Attachment 347510 Details for Bug 188745 – WIP

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Attachment 347510 Details for Bug 188745: Update libwebrtc up to 984f1a80c0

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[patch] WIP

patch (text/plain), 29.43 MB, created by youenn fablet on 2018-08-20 10:57:38 PDT

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Filename:

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Creator: youenn fablet

Created: 2018-08-20 10:57:38 PDT

Size: 29.43 MB

patch

obsolete

>diff --git a/LayoutTests/ChangeLog b/LayoutTests/ChangeLog
>index ee74960aa01..883cd0665b4 100644
>--- a/LayoutTests/ChangeLog
>+++ b/LayoutTests/ChangeLog
>@@ -1,3 +1,15 @@
>+2018-08-20  Youenn Fablet  <youenn@apple.com>
>+
>+        Update libwebrtc up to 984f1a80c0
>+        https://bugs.webkit.org/show_bug.cgi?id=188745
>+
>+        Reviewed by NOBODY (OOPS!).
>+
>+        Updated tests according new webrtc backend.
>+
>+        * webrtc/libwebrtc/setLocalDescriptionCrash.html:
>+        * webrtc/video-getParameters.html:
>+
> 2018-08-14  Antoine Quint  <graouts@apple.com>
> 
>         [Web Animations] Crash under AnimationTimeline::cancelOrRemoveDeclarativeAnimation()
>diff --git a/LayoutTests/webrtc/libwebrtc/setLocalDescriptionCrash.html b/LayoutTests/webrtc/libwebrtc/setLocalDescriptionCrash.html
>index 7a0142c2a47..886a97c52df 100644
>--- a/LayoutTests/webrtc/libwebrtc/setLocalDescriptionCrash.html
>+++ b/LayoutTests/webrtc/libwebrtc/setLocalDescriptionCrash.html
>@@ -10,7 +10,7 @@
>         <script>
> function badifyAnswer(answer)
> {
>-    return {type: answer.type, sdp: answer.sdp.replace("BUNDLE video", "BUNDLE")};
>+    return {type: answer.type, sdp: answer.sdp.replace("v=0", "v")};
> }
> 
> promise_test((test) => {
>@@ -29,9 +29,9 @@ promise_test((test) => {
>         badAnswer = badifyAnswer(answer);
>         return pc2.setLocalDescription(badAnswer);
>     }).then(assert_unreached, (e) => {
>-        assert_equals(e.message, "Failed to set local answer sdp: Failed to enable BUNDLE.");
>+        assert_equals(e.message, "Expect line: v=");
>         return pc2.setLocalDescription(badAnswer).then(assert_unreached, (e) => {
>-            assert_equals(e.message, "Failed to set local answer sdp: no pending remote description.");
>+            assert_equals(e.message, "Expect line: v=");
>         });
>     });
> }, "Testing calling twice setLocalDescription with a bad SDP");
>diff --git a/LayoutTests/webrtc/video-getParameters.html b/LayoutTests/webrtc/video-getParameters.html
>index c9c949f8a5e..3b269080b3d 100644
>--- a/LayoutTests/webrtc/video-getParameters.html
>+++ b/LayoutTests/webrtc/video-getParameters.html
>@@ -45,6 +45,9 @@ promise_test((test) => {
>         receiverParameters.encodings[0].fec.ssrc = 1;
>         receiverParameters.encodings[0].rtx.ssrc = 1;
> 
>+        senderParameters.transactionId = "";
>+        receiverParameters.transactionId = "";
>+
>         assert_equals(JSON.stringify(senderParameters), JSON.stringify(receiverParameters), "testing sender vs. receiver parameters");
> 
>         senderParameters.encodings[0].ssrc = 1;
>diff --git a/Source/ThirdParty/libwebrtc/ChangeLog b/Source/ThirdParty/libwebrtc/ChangeLog
>index 8f6aebb7b6a..4a016177e25 100644
>--- a/Source/ThirdParty/libwebrtc/ChangeLog
>+++ b/Source/ThirdParty/libwebrtc/ChangeLog
>@@ -1,3 +1,38 @@
>+2018-08-20  Youenn Fablet  <youenn@apple.com>
>+
>+        Update libwebrtc up to 984f1a80c0
>+        https://bugs.webkit.org/show_bug.cgi?id=188745
>+
>+        Reviewed by NOBODY (OOPS!).
>+
>+        Updated webrtc to 984f1a80c0.
>+        Updated libyuv accordingly.
>+        Added abseil-cpp and rnnoise third party source code.
>+        Removed WebKit specific patches that are no longer needed.
>+
>+        * Configurations/libwebrtc.iOS.exp: Updated to include new required symbols.
>+        * Configurations/libwebrtc.iOSsim.exp: Ditto.
>+        * Configurations/libwebrtc.mac.exp: Ditto.
>+        * Source/third_party/abseil-cpp: Added.
>+        * Source/third_party/rnnoise: Added.
>+        * Source/webrtc: Updated.
>+        * WebKit/0001-Adapting-libwebrtc-H264-codec.patch: Removed.
>+        * WebKit/0001-Disable-SIGPIPE-for-WebRTC-sockets.patch: Removed.
>+        * WebKit/0001-Update-RTCVideoEncoderH264.mm-for-WebKit.patch: Removed.
>+        * WebKit/0001-Using-VCP.patch: Removed.
>+        * WebKit/0003-Fixing-VP8-files.patch: Removed.
>+        * WebKit/0004-Removing-parameter-names-from-files-included-from-We.patch: Removed.
>+        * WebKit/0005-Fix-RTC_FATAL.patch: Removed.
>+        * WebKit/0006-Disabling-VP8.patch: Removed.
>+        * WebKit/0007-Fix-RTC_STRINGIZE.patch: Removed.
>+        * WebKit/0008-Fix-sanitizer.patch: Removed.
>+        * WebKit/0009-Remove-dispatch_set_target_queue.patch: Removed.
>+        * WebKit/0010-Fix-RTCVideoEncoderH264-CVPixelBuffer-leak.patch: Removed.
>+        * WebKit/0011-Fix-AudioDeviceID-array-leak.patch: Removed.
>+        * WebKit/0012-Add-WK-prefix-to-Objective-C-classes-and-protocols.patch: Removed.
>+        * WebKit/0013-Fix-SafeSetError-use-after-move.patch: Removed.
>+        * libwebrtc.xcodeproj/project.pbxproj:
>+
> 2018-08-06  David Kilzer  <ddkilzer@apple.com>
> 
>         [libwebrtc] SafeSetError() in peerconnection.cc contains use-after-move of webrtc::RTCError variable
>diff --git a/Source/ThirdParty/libwebrtc/Configurations/Base.xcconfig b/Source/ThirdParty/libwebrtc/Configurations/Base.xcconfig
>index 074d8b3be2d..b04182e31f8 100644
>--- a/Source/ThirdParty/libwebrtc/Configurations/Base.xcconfig
>+++ b/Source/ThirdParty/libwebrtc/Configurations/Base.xcconfig
>@@ -52,7 +52,7 @@ GCC_WARN_UNUSED_FUNCTION = YES;
> GCC_WARN_UNUSED_VARIABLE = YES;
> PREBINDING = NO;
> STRIP_INSTALLED_PRODUCT = NO;
>-WARNING_CFLAGS = -Wexit-time-destructors -Wglobal-constructors -Wthread-safety;
>+WARNING_CFLAGS = -Wthread-safety;
> 
> SUPPORTED_PLATFORMS = iphoneos iphonesimulator macosx;
> VALID_ARCHS = $(ARCHS_STANDARD_64_BIT);
>diff --git a/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.iOS.exp b/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.iOS.exp
>index 89f4e8b8a9c..8478b284152 100644
>--- a/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.iOS.exp
>+++ b/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.iOS.exp
>@@ -1,7 +1,4 @@
> __ZN3rtc10LogMessage10LogToDebugENS_15LoggingSeverityE
>-__ZN3rtc12FatalMessageC1EPKci
>-__ZN3rtc12FatalMessageC1EPKciPNSt3__112basic_stringIcNS3_11char_traitsIcEENS3_9allocatorIcEEEE
>-__ZN3rtc12FatalMessageD1Ev
> __ZN3rtc14MessageHandlerD2Ev
> __ZN3rtc14VideoSinkWantsC1Ev
> __ZN3rtc14VideoSinkWantsD1Ev
>@@ -9,7 +6,6 @@ __ZN3rtc17CopyOnWriteBuffer21CloneDataIfReferencedEm
> __ZN3rtc17CopyOnWriteBufferC1ERKS0_
> __ZN3rtc17CopyOnWriteBufferC1Emm
> __ZN3rtc17CopyOnWriteBufferD1Ev
>-__ZN3rtc6Thread14InvokeInternalERKNS_8LocationEPNS_14MessageHandlerE
> __ZN3rtc6Thread22CreateWithSocketServerEv
> __ZN3rtc6Thread5StartEPNS_8RunnableE
> __ZN3rtc6Thread6CreateEv
>@@ -34,15 +30,11 @@ __ZN6webrtc16ConvertVideoTypeENS_9VideoTypeE
> __ZN6webrtc18CreateIceCandidateERKNSt3__112basic_stringIcNS0_11char_traitsIcEENS0_9allocatorIcEEEEiS8_PNS_13SdpParseErrorE
> __ZN6webrtc19RTCCertificateStats5kTypeE
> __ZN6webrtc19RTCDataChannelStats5kTypeE
>-__ZN6webrtc20CoreVideoFrameBuffer6ToI420Ev
>-__ZN6webrtc20CoreVideoFrameBufferC2EP10__CVBuffer
>-__ZN6webrtc20CoreVideoFrameBufferD2Ev
> __ZN6webrtc24CreateSessionDescriptionERKNSt3__112basic_stringIcNS0_11char_traitsIcEENS0_9allocatorIcEEEES8_PNS_13SdpParseErrorE
> __ZN6webrtc24RTCIceCandidatePairStats5kTypeE
> __ZN6webrtc24RTCInboundRTPStreamStats5kTypeE
> __ZN6webrtc24RTCMediaStreamTrackStats5kTypeE
> __ZN6webrtc25RTCOutboundRTPStreamStats5kTypeE
>-__ZN6webrtc27CreatePeerConnectionFactoryEPN3rtc6ThreadES2_S2_NS0_13scoped_refptrINS_17AudioDeviceModuleEEENS3_INS_19AudioEncoderFactoryEEENS3_INS_19AudioDecoderFactoryEEENSt3__110unique_ptrINS_19VideoEncoderFactoryENSA_14default_deleteISC_EEEENSB_INS_19VideoDecoderFactoryENSD_ISG_EEEENS3_INS_10AudioMixerEEENS3_INS_15AudioProcessingEEE
> __ZN6webrtc27SessionDescriptionInterface6kOfferE
> __ZN6webrtc27SessionDescriptionInterface7kAnswerE
> __ZN6webrtc27SessionDescriptionInterface9kPrAnswerE
>@@ -50,7 +42,6 @@ __ZN6webrtc8internal21SynchronousMethodCall6InvokeERKN3rtc8LocationEPNS2_6Thread
> __ZN6webrtc8internal21SynchronousMethodCallC1EPN3rtc14MessageHandlerE
> __ZN6webrtc8internal21SynchronousMethodCallD1Ev
> __ZN7cricket17MediaTypeToStringENS_9MediaTypeE
>-__ZN7cricket18BasicPortAllocatorC1EPN3rtc14NetworkManagerEPNS1_19PacketSocketFactoryEPN6webrtc14TurnCustomizerEPNS_25RelayPortFactoryInterfaceE
> __ZNK6webrtc10VideoFrame18video_frame_bufferEv
> __ZNK6webrtc10VideoFrame5widthEv
> __ZNK6webrtc10VideoFrame6heightEv
>@@ -59,9 +50,6 @@ __ZNK6webrtc14RTCStatsReport13ConstIteratorneERKS1_
> __ZNK6webrtc14RTCStatsReport13ConstIteratorptEv
> __ZNK6webrtc14RTCStatsReport3endEv
> __ZNK6webrtc14RTCStatsReport5beginEv
>-__ZNK6webrtc20CoreVideoFrameBuffer4typeEv
>-__ZNK6webrtc20CoreVideoFrameBuffer5widthEv
>-__ZNK6webrtc20CoreVideoFrameBuffer6heightEv
> __ZNK6webrtc8RTCStats6ToJsonEv
> __ZTVN3rtc14MessageHandlerE
> _ConvertToI420
>@@ -95,13 +83,11 @@ __ZN3rtc13SocketAddressC1ERKS0_
> __ZN3rtc13SocketAddressaSERKS0_
> __ZNK3rtc13SocketAddress6ipaddrEv
> __ZNK3rtc18NetworkManagerBase11GetNetworksEPNSt3__16vectorIPNS_7NetworkENS1_9allocatorIS4_EEEE
>-__ZN3rtc19PacketSocketFactory21CreateClientTcpSocketERKNS_13SocketAddressES3_RKNS_9ProxyInfoERKNSt3__112basic_stringIcNS7_11char_traitsIcEENS7_9allocatorIcEEEERKNS_22PacketSocketTcpOptionsE
> __ZN3rtc22PacketTimeUpdateParamsC1Ev
> __ZN3rtc17AsyncPacketSocketD2Ev
> __ZN3rtc9ProxyInfoD1Ev
> __ZN3rtc18NetworkManagerBaseC2Ev
> __ZN3rtc7NetworkD1Ev
>-__ZTVN3rtc19PacketSocketFactoryE
> __ZN3rtc22PacketTimeUpdateParamsD1Ev
> __ZN3rtc22AsyncResolverInterfaceC2Ev
> __ZN3rtc18NetworkManagerBase16MergeNetworkListERKNSt3__16vectorIPNS_7NetworkENS1_9allocatorIS4_EEEEPb
>@@ -118,9 +104,6 @@ __ZN6webrtc32CreateBuiltinAudioDecoderFactoryEv
> __ZN6webrtc32CreateBuiltinAudioEncoderFactoryEv
> __ZN6webrtc27SessionDescriptionInterface16RemoveCandidatesERKNSt3__16vectorIN7cricket9CandidateENS1_9allocatorIS4_EEEE
> __ZNK6webrtc21IceCandidateInterface10server_urlEv
>-__ZNK6webrtc27SessionDescriptionInterface7GetTypeEv
>-__ZTVN6webrtc21IceCandidateInterfaceE
>-__ZTVN6webrtc27SessionDescriptionInterfaceE
> __ZN6webrtc20setApplicationStatusEb
> __ZN6webrtc32createVideoToolboxDecoderFactoryEv
> __ZN6webrtc32createVideoToolboxEncoderFactoryEv
>@@ -128,6 +111,98 @@ __ZN6webrtc29setH264HardwareEncoderAllowedEb
> __ZN6webrtc20pixelBufferFromFrameERKNS_10VideoFrameE
> __ZN6webrtc18pixelBufferToFrameEP10__CVBuffer
> __ZN3rtc24BasicPacketSocketFactory19CreateAsyncResolverEv
>-__ZN3rtc24BasicPacketSocketFactory21CreateClientTcpSocketERKNS_13SocketAddressES3_RKNS_9ProxyInfoERKNSt3__112basic_stringIcNS7_11char_traitsIcEENS7_9allocatorIcEEEERKNS_22PacketSocketTcpOptionsE
> __ZN3rtc24BasicPacketSocketFactoryC2Ev
> __ZN3rtc24BasicPacketSocketFactoryD2Ev
>+__ZNK3rtc6Thread9IsCurrentEv
>+__ZN3rtc6Thread14InvokeInternalERKNS_8LocationEPNS_14MessageHandlerE
>+__ZN3rtc5EventD1Ev
>+__ZN3rtc5EventC1Ebb
>+__ZN3rtc5Event4WaitEi
>+__ZN3rtc5Event3SetEv
>+__ZN3rtc18CreateRandomStringEm
>+__ZN6webrtc8RTCError2OKEv
>+__ZN3rtc18webrtc_checks_impl8FatalLogEPKciS2_PKNS0_12CheckArgTypeEz
>+__ZTVN6webrtc32CreateSessionDescriptionObserverE
>+__ZTVN6webrtc30PeerConnectionFactoryInterfaceE
>+__ZTVN6webrtc29SetSessionDescriptionObserverE
>+__ZNK3rtc17ThreadCheckerImpl19CalledOnValidThreadEv
>+__ZN7cricket18BasicPortAllocatorC1EPN3rtc14NetworkManagerEPNS1_19PacketSocketFactoryEPN6webrtc14TurnCustomizerEPNS_25RelayPortFactoryInterfaceE
>+__ZN7cricket12AudioOptionsD1Ev
>+__ZN6webrtc32CreateSessionDescriptionObserver9OnFailureENS_8RTCErrorE
>+__ZN6webrtc30PeerConnectionFactoryInterface17CreateVideoSourceEPN7cricket13VideoCapturerEPKNS_25MediaConstraintsInterfaceE
>+__ZN6webrtc30PeerConnectionFactoryInterface17CreateVideoSourceEPN7cricket13VideoCapturerE
>+__ZN6webrtc29SetSessionDescriptionObserver9OnFailureENS_8RTCErrorE
>+__ZN6webrtc28RtpHeaderExtensionCapabilityD1Ev
>+__ZN6webrtc26PeerConnectionDependenciesD1Ev
>+__ZN3rtc24BasicPacketSocketFactory21CreateClientTcpSocketERKNS_13SocketAddressES3_RKNS_9ProxyInfoERKNSt3__112basic_stringIcNS7_11char_traitsIcEENS7_9allocatorIcEEEERKNS_22PacketSocketTcpOptionsE
>+__ZN6webrtc12RtcpFeedbackC1ERKS0_
>+__ZN6webrtc12RtcpFeedbackD1Ev
>+__ZN6webrtc14I420BufferPoolC1Ev
>+__ZN6webrtc14I420BufferPoolD1Ev
>+__ZN6webrtc15RtpCapabilitiesC1Ev
>+__ZN6webrtc15RtpCapabilitiesD1Ev
>+__ZN6webrtc18RtpCodecCapabilityD1Ev
>+__ZN6webrtc23PeerConnectionInterface16RTCConfigurationC1ERKS1_
>+__ZN6webrtc23PeerConnectionInterface16RTCConfigurationC1Ev
>+__ZN6webrtc23PeerConnectionInterface16RTCConfigurationD1Ev
>+__ZN6webrtc23PeerConnectionInterface9IceServerC1ERKS1_
>+__ZN6webrtc23PeerConnectionInterface9IceServerC1Ev
>+__ZN6webrtc23PeerConnectionInterface9IceServerD1Ev
>+__ZN6webrtc26PeerConnectionDependenciesC1EOS0_
>+__ZN3rtc18webrtc_checks_impl8FatalLogEPKciS2_PKNS0_12CheckArgTypeEz
>+__ZN6webrtc12RtcpFeedbackC1ERKS0_
>+__ZN6webrtc12RtcpFeedbackD1Ev
>+__ZN6webrtc14I420BufferPoolC1Ev
>+__ZN6webrtc14I420BufferPoolD1Ev
>+__ZN6webrtc15RtpCapabilitiesC1Ev
>+__ZN6webrtc15RtpCapabilitiesD1Ev
>+__ZN6webrtc18RtpCodecCapabilityD1Ev
>+__ZN6webrtc23PeerConnectionInterface16RTCConfigurationC1ERKS1_
>+__ZTVN6webrtc27SessionDescriptionInterfaceE
>+__ZTVN6webrtc23PeerConnectionInterfaceE
>+__ZTVN6webrtc21IceCandidateInterfaceE
>+__ZTVN6webrtc20DataChannelInterfaceE
>+__ZTVN6webrtc19VideoTrackInterfaceE
>+__ZTVN6webrtc19AudioTrackInterfaceE
>+__ZNK6webrtc30PeerConnectionFactoryInterface26GetRtpReceiverCapabilitiesEN7cricket9MediaTypeE
>+__ZNK6webrtc30PeerConnectionFactoryInterface24GetRtpSenderCapabilitiesEN7cricket9MediaTypeE
>+__ZNK6webrtc27SessionDescriptionInterface7GetTypeEv
>+__ZNK6webrtc23PeerConnectionInterface26pending_remote_descriptionEv
>+__ZNK6webrtc23PeerConnectionInterface26current_remote_descriptionEv
>+__ZNK6webrtc23PeerConnectionInterface25pending_local_descriptionEv
>+__ZNK6webrtc23PeerConnectionInterface25current_local_descriptionEv
>+__ZNK6webrtc23PeerConnectionInterface15GetTransceiversEv
>+__ZNK6webrtc23PeerConnectionInterface12GetReceiversEv
>+__ZNK6webrtc23PeerConnectionInterface10GetSendersEv
>+__ZNK6webrtc20DataChannelInterface8protocolEv
>+__ZNK6webrtc20DataChannelInterface17maxRetransmitTimeEv
>+__ZNK6webrtc20DataChannelInterface14maxRetransmitsEv
>+__ZNK6webrtc20DataChannelInterface10negotiatedEv
>+__ZNK6webrtc19VideoTrackInterface12content_hintEv
>+__ZN6webrtc30PeerConnectionFactoryInterface20CreatePeerConnectionERKNS_23PeerConnectionInterface16RTCConfigurationENS_26PeerConnectionDependenciesE
>+__ZN6webrtc30PeerConnectionFactoryInterface17CreateVideoSourceENSt3__110unique_ptrIN7cricket13VideoCapturerENS1_14default_deleteIS4_EEEEPKNS_25MediaConstraintsInterfaceE
>+__ZN6webrtc30PeerConnectionFactoryInterface17CreateVideoSourceENSt3__110unique_ptrIN7cricket13VideoCapturerENS1_14default_deleteIS4_EEEE
>+__ZN6webrtc23PeerConnectionInterface19RemoveIceCandidatesERKNSt3__16vectorIN7cricket9CandidateENS1_9allocatorIS4_EEEE
>+__ZN6webrtc23PeerConnectionInterface16StartRtcEventLogEix
>+__ZN6webrtc23PeerConnectionInterface16StartRtcEventLogENSt3__110unique_ptrINS_17RtcEventLogOutputENS1_14default_deleteIS3_EEEEx
>+__ZN6webrtc23PeerConnectionInterface16SetConfigurationERKNS0_16RTCConfigurationEPNS_8RTCErrorE
>+__ZN6webrtc23PeerConnectionInterface16SetConfigurationERKNS0_16RTCConfigurationE
>+__ZN6webrtc19AudioTrackInterface14GetSignalLevelEPi
>+__ZN6webrtc19AudioTrackInterface17GetAudioProcessorEv
>+__ZN6webrtc23PeerConnectionInterface10SetBitrateERKNS_15BitrateSettingsE
>+__ZN6webrtc23PeerConnectionInterface12CreateSenderERKNSt3__112basic_stringIcNS1_11char_traitsIcEENS1_9allocatorIcEEEES9_
>+__ZN6webrtc23PeerConnectionInterface14AddTransceiverEN3rtc13scoped_refptrINS_25MediaStreamTrackInterfaceEEE
>+__ZN6webrtc23PeerConnectionInterface14AddTransceiverEN3rtc13scoped_refptrINS_25MediaStreamTrackInterfaceEEERKNS_18RtpTransceiverInitE
>+__ZN6webrtc23PeerConnectionInterface14AddTransceiverEN7cricket9MediaTypeE
>+__ZN6webrtc23PeerConnectionInterface14AddTransceiverEN7cricket9MediaTypeERKNS_18RtpTransceiverInitE
>+__ZN6webrtc23PeerConnectionInterface16GetConfigurationEv
>+__ZN6webrtc23PeerConnectionInterface14RemoveTrackNewEN3rtc13scoped_refptrINS_18RtpSenderInterfaceEEE
>+__ZN3rtc10SentPacketC1Exx
>+__ZN3rtc13PacketOptionsC1ERKS0_
>+__ZN3rtc10PacketInfoD1Ev
>+__ZN3rtc10PacketInfoC1ERKS0_
>+__ZTVN3rtc19PacketSocketFactoryE
>+__ZN3rtc13PacketOptionsD1Ev
>+__ZN3rtc19PacketSocketFactory21CreateClientTcpSocketERKNS_13SocketAddressES3_RKNS_9ProxyInfoERKNSt3__112basic_stringIcNS7_11char_traitsIcEENS7_9allocatorIcEEEERKNS_22PacketSocketTcpOptionsE
>+__ZN3rtc13PacketOptionsC1Ev
>+__ZN6webrtc27CreatePeerConnectionFactoryEPN3rtc6ThreadES2_S2_NS0_13scoped_refptrINS_17AudioDeviceModuleEEENS3_INS_19AudioEncoderFactoryEEENS3_INS_19AudioDecoderFactoryEEENSt3__110unique_ptrINS_19VideoEncoderFactoryENSA_14default_deleteISC_EEEENSB_INS_19VideoDecoderFactoryENSD_ISG_EEEENS3_INS_10AudioMixerEEENS3_INS_15AudioProcessingEEE
>diff --git a/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.iOSsim.exp b/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.iOSsim.exp
>index 89f4e8b8a9c..8478b284152 100644
>--- a/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.iOSsim.exp
>+++ b/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.iOSsim.exp
>@@ -1,7 +1,4 @@
> __ZN3rtc10LogMessage10LogToDebugENS_15LoggingSeverityE
>-__ZN3rtc12FatalMessageC1EPKci
>-__ZN3rtc12FatalMessageC1EPKciPNSt3__112basic_stringIcNS3_11char_traitsIcEENS3_9allocatorIcEEEE
>-__ZN3rtc12FatalMessageD1Ev
> __ZN3rtc14MessageHandlerD2Ev
> __ZN3rtc14VideoSinkWantsC1Ev
> __ZN3rtc14VideoSinkWantsD1Ev
>@@ -9,7 +6,6 @@ __ZN3rtc17CopyOnWriteBuffer21CloneDataIfReferencedEm
> __ZN3rtc17CopyOnWriteBufferC1ERKS0_
> __ZN3rtc17CopyOnWriteBufferC1Emm
> __ZN3rtc17CopyOnWriteBufferD1Ev
>-__ZN3rtc6Thread14InvokeInternalERKNS_8LocationEPNS_14MessageHandlerE
> __ZN3rtc6Thread22CreateWithSocketServerEv
> __ZN3rtc6Thread5StartEPNS_8RunnableE
> __ZN3rtc6Thread6CreateEv
>@@ -34,15 +30,11 @@ __ZN6webrtc16ConvertVideoTypeENS_9VideoTypeE
> __ZN6webrtc18CreateIceCandidateERKNSt3__112basic_stringIcNS0_11char_traitsIcEENS0_9allocatorIcEEEEiS8_PNS_13SdpParseErrorE
> __ZN6webrtc19RTCCertificateStats5kTypeE
> __ZN6webrtc19RTCDataChannelStats5kTypeE
>-__ZN6webrtc20CoreVideoFrameBuffer6ToI420Ev
>-__ZN6webrtc20CoreVideoFrameBufferC2EP10__CVBuffer
>-__ZN6webrtc20CoreVideoFrameBufferD2Ev
> __ZN6webrtc24CreateSessionDescriptionERKNSt3__112basic_stringIcNS0_11char_traitsIcEENS0_9allocatorIcEEEES8_PNS_13SdpParseErrorE
> __ZN6webrtc24RTCIceCandidatePairStats5kTypeE
> __ZN6webrtc24RTCInboundRTPStreamStats5kTypeE
> __ZN6webrtc24RTCMediaStreamTrackStats5kTypeE
> __ZN6webrtc25RTCOutboundRTPStreamStats5kTypeE
>-__ZN6webrtc27CreatePeerConnectionFactoryEPN3rtc6ThreadES2_S2_NS0_13scoped_refptrINS_17AudioDeviceModuleEEENS3_INS_19AudioEncoderFactoryEEENS3_INS_19AudioDecoderFactoryEEENSt3__110unique_ptrINS_19VideoEncoderFactoryENSA_14default_deleteISC_EEEENSB_INS_19VideoDecoderFactoryENSD_ISG_EEEENS3_INS_10AudioMixerEEENS3_INS_15AudioProcessingEEE
> __ZN6webrtc27SessionDescriptionInterface6kOfferE
> __ZN6webrtc27SessionDescriptionInterface7kAnswerE
> __ZN6webrtc27SessionDescriptionInterface9kPrAnswerE
>@@ -50,7 +42,6 @@ __ZN6webrtc8internal21SynchronousMethodCall6InvokeERKN3rtc8LocationEPNS2_6Thread
> __ZN6webrtc8internal21SynchronousMethodCallC1EPN3rtc14MessageHandlerE
> __ZN6webrtc8internal21SynchronousMethodCallD1Ev
> __ZN7cricket17MediaTypeToStringENS_9MediaTypeE
>-__ZN7cricket18BasicPortAllocatorC1EPN3rtc14NetworkManagerEPNS1_19PacketSocketFactoryEPN6webrtc14TurnCustomizerEPNS_25RelayPortFactoryInterfaceE
> __ZNK6webrtc10VideoFrame18video_frame_bufferEv
> __ZNK6webrtc10VideoFrame5widthEv
> __ZNK6webrtc10VideoFrame6heightEv
>@@ -59,9 +50,6 @@ __ZNK6webrtc14RTCStatsReport13ConstIteratorneERKS1_
> __ZNK6webrtc14RTCStatsReport13ConstIteratorptEv
> __ZNK6webrtc14RTCStatsReport3endEv
> __ZNK6webrtc14RTCStatsReport5beginEv
>-__ZNK6webrtc20CoreVideoFrameBuffer4typeEv
>-__ZNK6webrtc20CoreVideoFrameBuffer5widthEv
>-__ZNK6webrtc20CoreVideoFrameBuffer6heightEv
> __ZNK6webrtc8RTCStats6ToJsonEv
> __ZTVN3rtc14MessageHandlerE
> _ConvertToI420
>@@ -95,13 +83,11 @@ __ZN3rtc13SocketAddressC1ERKS0_
> __ZN3rtc13SocketAddressaSERKS0_
> __ZNK3rtc13SocketAddress6ipaddrEv
> __ZNK3rtc18NetworkManagerBase11GetNetworksEPNSt3__16vectorIPNS_7NetworkENS1_9allocatorIS4_EEEE
>-__ZN3rtc19PacketSocketFactory21CreateClientTcpSocketERKNS_13SocketAddressES3_RKNS_9ProxyInfoERKNSt3__112basic_stringIcNS7_11char_traitsIcEENS7_9allocatorIcEEEERKNS_22PacketSocketTcpOptionsE
> __ZN3rtc22PacketTimeUpdateParamsC1Ev
> __ZN3rtc17AsyncPacketSocketD2Ev
> __ZN3rtc9ProxyInfoD1Ev
> __ZN3rtc18NetworkManagerBaseC2Ev
> __ZN3rtc7NetworkD1Ev
>-__ZTVN3rtc19PacketSocketFactoryE
> __ZN3rtc22PacketTimeUpdateParamsD1Ev
> __ZN3rtc22AsyncResolverInterfaceC2Ev
> __ZN3rtc18NetworkManagerBase16MergeNetworkListERKNSt3__16vectorIPNS_7NetworkENS1_9allocatorIS4_EEEEPb
>@@ -118,9 +104,6 @@ __ZN6webrtc32CreateBuiltinAudioDecoderFactoryEv
> __ZN6webrtc32CreateBuiltinAudioEncoderFactoryEv
> __ZN6webrtc27SessionDescriptionInterface16RemoveCandidatesERKNSt3__16vectorIN7cricket9CandidateENS1_9allocatorIS4_EEEE
> __ZNK6webrtc21IceCandidateInterface10server_urlEv
>-__ZNK6webrtc27SessionDescriptionInterface7GetTypeEv
>-__ZTVN6webrtc21IceCandidateInterfaceE
>-__ZTVN6webrtc27SessionDescriptionInterfaceE
> __ZN6webrtc20setApplicationStatusEb
> __ZN6webrtc32createVideoToolboxDecoderFactoryEv
> __ZN6webrtc32createVideoToolboxEncoderFactoryEv
>@@ -128,6 +111,98 @@ __ZN6webrtc29setH264HardwareEncoderAllowedEb
> __ZN6webrtc20pixelBufferFromFrameERKNS_10VideoFrameE
> __ZN6webrtc18pixelBufferToFrameEP10__CVBuffer
> __ZN3rtc24BasicPacketSocketFactory19CreateAsyncResolverEv
>-__ZN3rtc24BasicPacketSocketFactory21CreateClientTcpSocketERKNS_13SocketAddressES3_RKNS_9ProxyInfoERKNSt3__112basic_stringIcNS7_11char_traitsIcEENS7_9allocatorIcEEEERKNS_22PacketSocketTcpOptionsE
> __ZN3rtc24BasicPacketSocketFactoryC2Ev
> __ZN3rtc24BasicPacketSocketFactoryD2Ev
>+__ZNK3rtc6Thread9IsCurrentEv
>+__ZN3rtc6Thread14InvokeInternalERKNS_8LocationEPNS_14MessageHandlerE
>+__ZN3rtc5EventD1Ev
>+__ZN3rtc5EventC1Ebb
>+__ZN3rtc5Event4WaitEi
>+__ZN3rtc5Event3SetEv
>+__ZN3rtc18CreateRandomStringEm
>+__ZN6webrtc8RTCError2OKEv
>+__ZN3rtc18webrtc_checks_impl8FatalLogEPKciS2_PKNS0_12CheckArgTypeEz
>+__ZTVN6webrtc32CreateSessionDescriptionObserverE
>+__ZTVN6webrtc30PeerConnectionFactoryInterfaceE
>+__ZTVN6webrtc29SetSessionDescriptionObserverE
>+__ZNK3rtc17ThreadCheckerImpl19CalledOnValidThreadEv
>+__ZN7cricket18BasicPortAllocatorC1EPN3rtc14NetworkManagerEPNS1_19PacketSocketFactoryEPN6webrtc14TurnCustomizerEPNS_25RelayPortFactoryInterfaceE
>+__ZN7cricket12AudioOptionsD1Ev
>+__ZN6webrtc32CreateSessionDescriptionObserver9OnFailureENS_8RTCErrorE
>+__ZN6webrtc30PeerConnectionFactoryInterface17CreateVideoSourceEPN7cricket13VideoCapturerEPKNS_25MediaConstraintsInterfaceE
>+__ZN6webrtc30PeerConnectionFactoryInterface17CreateVideoSourceEPN7cricket13VideoCapturerE
>+__ZN6webrtc29SetSessionDescriptionObserver9OnFailureENS_8RTCErrorE
>+__ZN6webrtc28RtpHeaderExtensionCapabilityD1Ev
>+__ZN6webrtc26PeerConnectionDependenciesD1Ev
>+__ZN3rtc24BasicPacketSocketFactory21CreateClientTcpSocketERKNS_13SocketAddressES3_RKNS_9ProxyInfoERKNSt3__112basic_stringIcNS7_11char_traitsIcEENS7_9allocatorIcEEEERKNS_22PacketSocketTcpOptionsE
>+__ZN6webrtc12RtcpFeedbackC1ERKS0_
>+__ZN6webrtc12RtcpFeedbackD1Ev
>+__ZN6webrtc14I420BufferPoolC1Ev
>+__ZN6webrtc14I420BufferPoolD1Ev
>+__ZN6webrtc15RtpCapabilitiesC1Ev
>+__ZN6webrtc15RtpCapabilitiesD1Ev
>+__ZN6webrtc18RtpCodecCapabilityD1Ev
>+__ZN6webrtc23PeerConnectionInterface16RTCConfigurationC1ERKS1_
>+__ZN6webrtc23PeerConnectionInterface16RTCConfigurationC1Ev
>+__ZN6webrtc23PeerConnectionInterface16RTCConfigurationD1Ev
>+__ZN6webrtc23PeerConnectionInterface9IceServerC1ERKS1_
>+__ZN6webrtc23PeerConnectionInterface9IceServerC1Ev
>+__ZN6webrtc23PeerConnectionInterface9IceServerD1Ev
>+__ZN6webrtc26PeerConnectionDependenciesC1EOS0_
>+__ZN3rtc18webrtc_checks_impl8FatalLogEPKciS2_PKNS0_12CheckArgTypeEz
>+__ZN6webrtc12RtcpFeedbackC1ERKS0_
>+__ZN6webrtc12RtcpFeedbackD1Ev
>+__ZN6webrtc14I420BufferPoolC1Ev
>+__ZN6webrtc14I420BufferPoolD1Ev
>+__ZN6webrtc15RtpCapabilitiesC1Ev
>+__ZN6webrtc15RtpCapabilitiesD1Ev
>+__ZN6webrtc18RtpCodecCapabilityD1Ev
>+__ZN6webrtc23PeerConnectionInterface16RTCConfigurationC1ERKS1_
>+__ZTVN6webrtc27SessionDescriptionInterfaceE
>+__ZTVN6webrtc23PeerConnectionInterfaceE
>+__ZTVN6webrtc21IceCandidateInterfaceE
>+__ZTVN6webrtc20DataChannelInterfaceE
>+__ZTVN6webrtc19VideoTrackInterfaceE
>+__ZTVN6webrtc19AudioTrackInterfaceE
>+__ZNK6webrtc30PeerConnectionFactoryInterface26GetRtpReceiverCapabilitiesEN7cricket9MediaTypeE
>+__ZNK6webrtc30PeerConnectionFactoryInterface24GetRtpSenderCapabilitiesEN7cricket9MediaTypeE
>+__ZNK6webrtc27SessionDescriptionInterface7GetTypeEv
>+__ZNK6webrtc23PeerConnectionInterface26pending_remote_descriptionEv
>+__ZNK6webrtc23PeerConnectionInterface26current_remote_descriptionEv
>+__ZNK6webrtc23PeerConnectionInterface25pending_local_descriptionEv
>+__ZNK6webrtc23PeerConnectionInterface25current_local_descriptionEv
>+__ZNK6webrtc23PeerConnectionInterface15GetTransceiversEv
>+__ZNK6webrtc23PeerConnectionInterface12GetReceiversEv
>+__ZNK6webrtc23PeerConnectionInterface10GetSendersEv
>+__ZNK6webrtc20DataChannelInterface8protocolEv
>+__ZNK6webrtc20DataChannelInterface17maxRetransmitTimeEv
>+__ZNK6webrtc20DataChannelInterface14maxRetransmitsEv
>+__ZNK6webrtc20DataChannelInterface10negotiatedEv
>+__ZNK6webrtc19VideoTrackInterface12content_hintEv
>+__ZN6webrtc30PeerConnectionFactoryInterface20CreatePeerConnectionERKNS_23PeerConnectionInterface16RTCConfigurationENS_26PeerConnectionDependenciesE
>+__ZN6webrtc30PeerConnectionFactoryInterface17CreateVideoSourceENSt3__110unique_ptrIN7cricket13VideoCapturerENS1_14default_deleteIS4_EEEEPKNS_25MediaConstraintsInterfaceE
>+__ZN6webrtc30PeerConnectionFactoryInterface17CreateVideoSourceENSt3__110unique_ptrIN7cricket13VideoCapturerENS1_14default_deleteIS4_EEEE
>+__ZN6webrtc23PeerConnectionInterface19RemoveIceCandidatesERKNSt3__16vectorIN7cricket9CandidateENS1_9allocatorIS4_EEEE
>+__ZN6webrtc23PeerConnectionInterface16StartRtcEventLogEix
>+__ZN6webrtc23PeerConnectionInterface16StartRtcEventLogENSt3__110unique_ptrINS_17RtcEventLogOutputENS1_14default_deleteIS3_EEEEx
>+__ZN6webrtc23PeerConnectionInterface16SetConfigurationERKNS0_16RTCConfigurationEPNS_8RTCErrorE
>+__ZN6webrtc23PeerConnectionInterface16SetConfigurationERKNS0_16RTCConfigurationE
>+__ZN6webrtc19AudioTrackInterface14GetSignalLevelEPi
>+__ZN6webrtc19AudioTrackInterface17GetAudioProcessorEv
>+__ZN6webrtc23PeerConnectionInterface10SetBitrateERKNS_15BitrateSettingsE
>+__ZN6webrtc23PeerConnectionInterface12CreateSenderERKNSt3__112basic_stringIcNS1_11char_traitsIcEENS1_9allocatorIcEEEES9_
>+__ZN6webrtc23PeerConnectionInterface14AddTransceiverEN3rtc13scoped_refptrINS_25MediaStreamTrackInterfaceEEE
>+__ZN6webrtc23PeerConnectionInterface14AddTransceiverEN3rtc13scoped_refptrINS_25MediaStreamTrackInterfaceEEERKNS_18RtpTransceiverInitE
>+__ZN6webrtc23PeerConnectionInterface14AddTransceiverEN7cricket9MediaTypeE
>+__ZN6webrtc23PeerConnectionInterface14AddTransceiverEN7cricket9MediaTypeERKNS_18RtpTransceiverInitE
>+__ZN6webrtc23PeerConnectionInterface16GetConfigurationEv
>+__ZN6webrtc23PeerConnectionInterface14RemoveTrackNewEN3rtc13scoped_refptrINS_18RtpSenderInterfaceEEE
>+__ZN3rtc10SentPacketC1Exx
>+__ZN3rtc13PacketOptionsC1ERKS0_
>+__ZN3rtc10PacketInfoD1Ev
>+__ZN3rtc10PacketInfoC1ERKS0_
>+__ZTVN3rtc19PacketSocketFactoryE
>+__ZN3rtc13PacketOptionsD1Ev
>+__ZN3rtc19PacketSocketFactory21CreateClientTcpSocketERKNS_13SocketAddressES3_RKNS_9ProxyInfoERKNSt3__112basic_stringIcNS7_11char_traitsIcEENS7_9allocatorIcEEEERKNS_22PacketSocketTcpOptionsE
>+__ZN3rtc13PacketOptionsC1Ev
>+__ZN6webrtc27CreatePeerConnectionFactoryEPN3rtc6ThreadES2_S2_NS0_13scoped_refptrINS_17AudioDeviceModuleEEENS3_INS_19AudioEncoderFactoryEEENS3_INS_19AudioDecoderFactoryEEENSt3__110unique_ptrINS_19VideoEncoderFactoryENSA_14default_deleteISC_EEEENSB_INS_19VideoDecoderFactoryENSD_ISG_EEEENS3_INS_10AudioMixerEEENS3_INS_15AudioProcessingEEE
>diff --git a/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.mac.exp b/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.mac.exp
>index 89f4e8b8a9c..8478b284152 100644
>--- a/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.mac.exp
>+++ b/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.mac.exp
>@@ -1,7 +1,4 @@
> __ZN3rtc10LogMessage10LogToDebugENS_15LoggingSeverityE
>-__ZN3rtc12FatalMessageC1EPKci
>-__ZN3rtc12FatalMessageC1EPKciPNSt3__112basic_stringIcNS3_11char_traitsIcEENS3_9allocatorIcEEEE
>-__ZN3rtc12FatalMessageD1Ev
> __ZN3rtc14MessageHandlerD2Ev
> __ZN3rtc14VideoSinkWantsC1Ev
> __ZN3rtc14VideoSinkWantsD1Ev
>@@ -9,7 +6,6 @@ __ZN3rtc17CopyOnWriteBuffer21CloneDataIfReferencedEm
> __ZN3rtc17CopyOnWriteBufferC1ERKS0_
> __ZN3rtc17CopyOnWriteBufferC1Emm
> __ZN3rtc17CopyOnWriteBufferD1Ev
>-__ZN3rtc6Thread14InvokeInternalERKNS_8LocationEPNS_14MessageHandlerE
> __ZN3rtc6Thread22CreateWithSocketServerEv
> __ZN3rtc6Thread5StartEPNS_8RunnableE
> __ZN3rtc6Thread6CreateEv
>@@ -34,15 +30,11 @@ __ZN6webrtc16ConvertVideoTypeENS_9VideoTypeE
> __ZN6webrtc18CreateIceCandidateERKNSt3__112basic_stringIcNS0_11char_traitsIcEENS0_9allocatorIcEEEEiS8_PNS_13SdpParseErrorE
> __ZN6webrtc19RTCCertificateStats5kTypeE
> __ZN6webrtc19RTCDataChannelStats5kTypeE
>-__ZN6webrtc20CoreVideoFrameBuffer6ToI420Ev
>-__ZN6webrtc20CoreVideoFrameBufferC2EP10__CVBuffer
>-__ZN6webrtc20CoreVideoFrameBufferD2Ev
> __ZN6webrtc24CreateSessionDescriptionERKNSt3__112basic_stringIcNS0_11char_traitsIcEENS0_9allocatorIcEEEES8_PNS_13SdpParseErrorE
> __ZN6webrtc24RTCIceCandidatePairStats5kTypeE
> __ZN6webrtc24RTCInboundRTPStreamStats5kTypeE
> __ZN6webrtc24RTCMediaStreamTrackStats5kTypeE
> __ZN6webrtc25RTCOutboundRTPStreamStats5kTypeE
>-__ZN6webrtc27CreatePeerConnectionFactoryEPN3rtc6ThreadES2_S2_NS0_13scoped_refptrINS_17AudioDeviceModuleEEENS3_INS_19AudioEncoderFactoryEEENS3_INS_19AudioDecoderFactoryEEENSt3__110unique_ptrINS_19VideoEncoderFactoryENSA_14default_deleteISC_EEEENSB_INS_19VideoDecoderFactoryENSD_ISG_EEEENS3_INS_10AudioMixerEEENS3_INS_15AudioProcessingEEE
> __ZN6webrtc27SessionDescriptionInterface6kOfferE
> __ZN6webrtc27SessionDescriptionInterface7kAnswerE
> __ZN6webrtc27SessionDescriptionInterface9kPrAnswerE
>@@ -50,7 +42,6 @@ __ZN6webrtc8internal21SynchronousMethodCall6InvokeERKN3rtc8LocationEPNS2_6Thread
> __ZN6webrtc8internal21SynchronousMethodCallC1EPN3rtc14MessageHandlerE
> __ZN6webrtc8internal21SynchronousMethodCallD1Ev
> __ZN7cricket17MediaTypeToStringENS_9MediaTypeE
>-__ZN7cricket18BasicPortAllocatorC1EPN3rtc14NetworkManagerEPNS1_19PacketSocketFactoryEPN6webrtc14TurnCustomizerEPNS_25RelayPortFactoryInterfaceE
> __ZNK6webrtc10VideoFrame18video_frame_bufferEv
> __ZNK6webrtc10VideoFrame5widthEv
> __ZNK6webrtc10VideoFrame6heightEv
>@@ -59,9 +50,6 @@ __ZNK6webrtc14RTCStatsReport13ConstIteratorneERKS1_
> __ZNK6webrtc14RTCStatsReport13ConstIteratorptEv
> __ZNK6webrtc14RTCStatsReport3endEv
> __ZNK6webrtc14RTCStatsReport5beginEv
>-__ZNK6webrtc20CoreVideoFrameBuffer4typeEv
>-__ZNK6webrtc20CoreVideoFrameBuffer5widthEv
>-__ZNK6webrtc20CoreVideoFrameBuffer6heightEv
> __ZNK6webrtc8RTCStats6ToJsonEv
> __ZTVN3rtc14MessageHandlerE
> _ConvertToI420
>@@ -95,13 +83,11 @@ __ZN3rtc13SocketAddressC1ERKS0_
> __ZN3rtc13SocketAddressaSERKS0_
> __ZNK3rtc13SocketAddress6ipaddrEv
> __ZNK3rtc18NetworkManagerBase11GetNetworksEPNSt3__16vectorIPNS_7NetworkENS1_9allocatorIS4_EEEE
>-__ZN3rtc19PacketSocketFactory21CreateClientTcpSocketERKNS_13SocketAddressES3_RKNS_9ProxyInfoERKNSt3__112basic_stringIcNS7_11char_traitsIcEENS7_9allocatorIcEEEERKNS_22PacketSocketTcpOptionsE
> __ZN3rtc22PacketTimeUpdateParamsC1Ev
> __ZN3rtc17AsyncPacketSocketD2Ev
> __ZN3rtc9ProxyInfoD1Ev
> __ZN3rtc18NetworkManagerBaseC2Ev
> __ZN3rtc7NetworkD1Ev
>-__ZTVN3rtc19PacketSocketFactoryE
> __ZN3rtc22PacketTimeUpdateParamsD1Ev
> __ZN3rtc22AsyncResolverInterfaceC2Ev
> __ZN3rtc18NetworkManagerBase16MergeNetworkListERKNSt3__16vectorIPNS_7NetworkENS1_9allocatorIS4_EEEEPb
>@@ -118,9 +104,6 @@ __ZN6webrtc32CreateBuiltinAudioDecoderFactoryEv
> __ZN6webrtc32CreateBuiltinAudioEncoderFactoryEv
> __ZN6webrtc27SessionDescriptionInterface16RemoveCandidatesERKNSt3__16vectorIN7cricket9CandidateENS1_9allocatorIS4_EEEE
> __ZNK6webrtc21IceCandidateInterface10server_urlEv
>-__ZNK6webrtc27SessionDescriptionInterface7GetTypeEv
>-__ZTVN6webrtc21IceCandidateInterfaceE
>-__ZTVN6webrtc27SessionDescriptionInterfaceE
> __ZN6webrtc20setApplicationStatusEb
> __ZN6webrtc32createVideoToolboxDecoderFactoryEv
> __ZN6webrtc32createVideoToolboxEncoderFactoryEv
>@@ -128,6 +111,98 @@ __ZN6webrtc29setH264HardwareEncoderAllowedEb
> __ZN6webrtc20pixelBufferFromFrameERKNS_10VideoFrameE
> __ZN6webrtc18pixelBufferToFrameEP10__CVBuffer
> __ZN3rtc24BasicPacketSocketFactory19CreateAsyncResolverEv
>-__ZN3rtc24BasicPacketSocketFactory21CreateClientTcpSocketERKNS_13SocketAddressES3_RKNS_9ProxyInfoERKNSt3__112basic_stringIcNS7_11char_traitsIcEENS7_9allocatorIcEEEERKNS_22PacketSocketTcpOptionsE
> __ZN3rtc24BasicPacketSocketFactoryC2Ev
> __ZN3rtc24BasicPacketSocketFactoryD2Ev
>+__ZNK3rtc6Thread9IsCurrentEv
>+__ZN3rtc6Thread14InvokeInternalERKNS_8LocationEPNS_14MessageHandlerE
>+__ZN3rtc5EventD1Ev
>+__ZN3rtc5EventC1Ebb
>+__ZN3rtc5Event4WaitEi
>+__ZN3rtc5Event3SetEv
>+__ZN3rtc18CreateRandomStringEm
>+__ZN6webrtc8RTCError2OKEv
>+__ZN3rtc18webrtc_checks_impl8FatalLogEPKciS2_PKNS0_12CheckArgTypeEz
>+__ZTVN6webrtc32CreateSessionDescriptionObserverE
>+__ZTVN6webrtc30PeerConnectionFactoryInterfaceE
>+__ZTVN6webrtc29SetSessionDescriptionObserverE
>+__ZNK3rtc17ThreadCheckerImpl19CalledOnValidThreadEv
>+__ZN7cricket18BasicPortAllocatorC1EPN3rtc14NetworkManagerEPNS1_19PacketSocketFactoryEPN6webrtc14TurnCustomizerEPNS_25RelayPortFactoryInterfaceE
>+__ZN7cricket12AudioOptionsD1Ev
>+__ZN6webrtc32CreateSessionDescriptionObserver9OnFailureENS_8RTCErrorE
>+__ZN6webrtc30PeerConnectionFactoryInterface17CreateVideoSourceEPN7cricket13VideoCapturerEPKNS_25MediaConstraintsInterfaceE
>+__ZN6webrtc30PeerConnectionFactoryInterface17CreateVideoSourceEPN7cricket13VideoCapturerE
>+__ZN6webrtc29SetSessionDescriptionObserver9OnFailureENS_8RTCErrorE
>+__ZN6webrtc28RtpHeaderExtensionCapabilityD1Ev
>+__ZN6webrtc26PeerConnectionDependenciesD1Ev
>+__ZN3rtc24BasicPacketSocketFactory21CreateClientTcpSocketERKNS_13SocketAddressES3_RKNS_9ProxyInfoERKNSt3__112basic_stringIcNS7_11char_traitsIcEENS7_9allocatorIcEEEERKNS_22PacketSocketTcpOptionsE
>+__ZN6webrtc12RtcpFeedbackC1ERKS0_
>+__ZN6webrtc12RtcpFeedbackD1Ev
>+__ZN6webrtc14I420BufferPoolC1Ev
>+__ZN6webrtc14I420BufferPoolD1Ev
>+__ZN6webrtc15RtpCapabilitiesC1Ev
>+__ZN6webrtc15RtpCapabilitiesD1Ev
>+__ZN6webrtc18RtpCodecCapabilityD1Ev
>+__ZN6webrtc23PeerConnectionInterface16RTCConfigurationC1ERKS1_
>+__ZN6webrtc23PeerConnectionInterface16RTCConfigurationC1Ev
>+__ZN6webrtc23PeerConnectionInterface16RTCConfigurationD1Ev
>+__ZN6webrtc23PeerConnectionInterface9IceServerC1ERKS1_
>+__ZN6webrtc23PeerConnectionInterface9IceServerC1Ev
>+__ZN6webrtc23PeerConnectionInterface9IceServerD1Ev
>+__ZN6webrtc26PeerConnectionDependenciesC1EOS0_
>+__ZN3rtc18webrtc_checks_impl8FatalLogEPKciS2_PKNS0_12CheckArgTypeEz
>+__ZN6webrtc12RtcpFeedbackC1ERKS0_
>+__ZN6webrtc12RtcpFeedbackD1Ev
>+__ZN6webrtc14I420BufferPoolC1Ev
>+__ZN6webrtc14I420BufferPoolD1Ev
>+__ZN6webrtc15RtpCapabilitiesC1Ev
>+__ZN6webrtc15RtpCapabilitiesD1Ev
>+__ZN6webrtc18RtpCodecCapabilityD1Ev
>+__ZN6webrtc23PeerConnectionInterface16RTCConfigurationC1ERKS1_
>+__ZTVN6webrtc27SessionDescriptionInterfaceE
>+__ZTVN6webrtc23PeerConnectionInterfaceE
>+__ZTVN6webrtc21IceCandidateInterfaceE
>+__ZTVN6webrtc20DataChannelInterfaceE
>+__ZTVN6webrtc19VideoTrackInterfaceE
>+__ZTVN6webrtc19AudioTrackInterfaceE
>+__ZNK6webrtc30PeerConnectionFactoryInterface26GetRtpReceiverCapabilitiesEN7cricket9MediaTypeE
>+__ZNK6webrtc30PeerConnectionFactoryInterface24GetRtpSenderCapabilitiesEN7cricket9MediaTypeE
>+__ZNK6webrtc27SessionDescriptionInterface7GetTypeEv
>+__ZNK6webrtc23PeerConnectionInterface26pending_remote_descriptionEv
>+__ZNK6webrtc23PeerConnectionInterface26current_remote_descriptionEv
>+__ZNK6webrtc23PeerConnectionInterface25pending_local_descriptionEv
>+__ZNK6webrtc23PeerConnectionInterface25current_local_descriptionEv
>+__ZNK6webrtc23PeerConnectionInterface15GetTransceiversEv
>+__ZNK6webrtc23PeerConnectionInterface12GetReceiversEv
>+__ZNK6webrtc23PeerConnectionInterface10GetSendersEv
>+__ZNK6webrtc20DataChannelInterface8protocolEv
>+__ZNK6webrtc20DataChannelInterface17maxRetransmitTimeEv
>+__ZNK6webrtc20DataChannelInterface14maxRetransmitsEv
>+__ZNK6webrtc20DataChannelInterface10negotiatedEv
>+__ZNK6webrtc19VideoTrackInterface12content_hintEv
>+__ZN6webrtc30PeerConnectionFactoryInterface20CreatePeerConnectionERKNS_23PeerConnectionInterface16RTCConfigurationENS_26PeerConnectionDependenciesE
>+__ZN6webrtc30PeerConnectionFactoryInterface17CreateVideoSourceENSt3__110unique_ptrIN7cricket13VideoCapturerENS1_14default_deleteIS4_EEEEPKNS_25MediaConstraintsInterfaceE
>+__ZN6webrtc30PeerConnectionFactoryInterface17CreateVideoSourceENSt3__110unique_ptrIN7cricket13VideoCapturerENS1_14default_deleteIS4_EEEE
>+__ZN6webrtc23PeerConnectionInterface19RemoveIceCandidatesERKNSt3__16vectorIN7cricket9CandidateENS1_9allocatorIS4_EEEE
>+__ZN6webrtc23PeerConnectionInterface16StartRtcEventLogEix
>+__ZN6webrtc23PeerConnectionInterface16StartRtcEventLogENSt3__110unique_ptrINS_17RtcEventLogOutputENS1_14default_deleteIS3_EEEEx
>+__ZN6webrtc23PeerConnectionInterface16SetConfigurationERKNS0_16RTCConfigurationEPNS_8RTCErrorE
>+__ZN6webrtc23PeerConnectionInterface16SetConfigurationERKNS0_16RTCConfigurationE
>+__ZN6webrtc19AudioTrackInterface14GetSignalLevelEPi
>+__ZN6webrtc19AudioTrackInterface17GetAudioProcessorEv
>+__ZN6webrtc23PeerConnectionInterface10SetBitrateERKNS_15BitrateSettingsE
>+__ZN6webrtc23PeerConnectionInterface12CreateSenderERKNSt3__112basic_stringIcNS1_11char_traitsIcEENS1_9allocatorIcEEEES9_
>+__ZN6webrtc23PeerConnectionInterface14AddTransceiverEN3rtc13scoped_refptrINS_25MediaStreamTrackInterfaceEEE
>+__ZN6webrtc23PeerConnectionInterface14AddTransceiverEN3rtc13scoped_refptrINS_25MediaStreamTrackInterfaceEEERKNS_18RtpTransceiverInitE
>+__ZN6webrtc23PeerConnectionInterface14AddTransceiverEN7cricket9MediaTypeE
>+__ZN6webrtc23PeerConnectionInterface14AddTransceiverEN7cricket9MediaTypeERKNS_18RtpTransceiverInitE
>+__ZN6webrtc23PeerConnectionInterface16GetConfigurationEv
>+__ZN6webrtc23PeerConnectionInterface14RemoveTrackNewEN3rtc13scoped_refptrINS_18RtpSenderInterfaceEEE
>+__ZN3rtc10SentPacketC1Exx
>+__ZN3rtc13PacketOptionsC1ERKS0_
>+__ZN3rtc10PacketInfoD1Ev
>+__ZN3rtc10PacketInfoC1ERKS0_
>+__ZTVN3rtc19PacketSocketFactoryE
>+__ZN3rtc13PacketOptionsD1Ev
>+__ZN3rtc19PacketSocketFactory21CreateClientTcpSocketERKNS_13SocketAddressES3_RKNS_9ProxyInfoERKNSt3__112basic_stringIcNS7_11char_traitsIcEENS7_9allocatorIcEEEERKNS_22PacketSocketTcpOptionsE
>+__ZN3rtc13PacketOptionsC1Ev
>+__ZN6webrtc27CreatePeerConnectionFactoryEPN3rtc6ThreadES2_S2_NS0_13scoped_refptrINS_17AudioDeviceModuleEEENS3_INS_19AudioEncoderFactoryEEENS3_INS_19AudioDecoderFactoryEEENSt3__110unique_ptrINS_19VideoEncoderFactoryENSA_14default_deleteISC_EEEENSB_INS_19VideoDecoderFactoryENSD_ISG_EEEENS3_INS_10AudioMixerEEENS3_INS_15AudioProcessingEEE
>diff --git a/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.xcconfig b/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.xcconfig
>index 0306586af6b..62a1196e337 100644
>--- a/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.xcconfig
>+++ b/Source/ThirdParty/libwebrtc/Configurations/libwebrtc.xcconfig
>@@ -26,7 +26,7 @@ INSTALLHDRS_SCRIPT_PHASE = YES;
> WARNING_CFLAGS = -Wno-deprecated-declarations $(inherited);
> 
> // FIXME: Set WEBRTC_USE_BUILTIN_ISAC_FIX and WEBRTC_USE_BUILTIN_ISAC_FLOAT for iOS and Mac
>-GCC_PREPROCESSOR_DEFINITIONS = GTEST_RELATIVE_PATH WEBRTC_OPUS_SUPPORT_120MS_PTIME=0 WEBRTC_POSIX WEBRTC_MAC SSL_USE_OPENSSL FEATURE_ENABLE_SSL HAVE_SRTP HAVE_OPENSSL_SSL_H SCTP_PROCESS_LEVEL_LOCKS SCTP_SIMPLE_ALLOCATOR SCTP_USE_OPENSSL_SHA1 __Userspace__ HAVE_SA_LEN HAVE_SCONN_LEN __APPLE_USE_RFC_2292 __Userspace_os_Darwin NON_WINDOWS_DEFINE HAVE_WEBRTC_VIDEO HAVE_WEBRTC_VOICE WEBRTC_INTELLIGIBILITY_ENHANCER=0 WEBRTC_APM_DEBUG_DUMP=0 WEBRTC_NS_FLOAT WEBRTC_USE_BUILTIN_ILBC WEBRTC_CODEC_ILBC WEBRTC_USE_BUILTIN_OPUS WEBRTC_CODEC_OPUS WEBRTC_CODEC_ISAC WEBRTC_CODEC_RED RTC_DISABLE_VP9 RTC_DISABLE_VP8 HAVE_STDINT_H HAVE_STDLIB_H HAVE_UINT64_T OPENSSL HAVE_CONFIG_H WEBRTC_WEBKIT_BUILD HAVE_PTHREAD_COND_TIMEDWAIT_RELATIVE  HAVE_SCTP WEBRTC_CODEC_G711 WEBRTC_CODEC_G722 WEBRTC_OPUS_VARIABLE_COMPLEXITY=0 WEBRTC_USE_BUILTIN_ISAC_FIX=1 WEBRTC_USE_BUILTIN_ISAC_FLOAT=0 $(inherited);
>+GCC_PREPROCESSOR_DEFINITIONS = GTEST_RELATIVE_PATH WEBRTC_OPUS_SUPPORT_120MS_PTIME=0 WEBRTC_POSIX WEBRTC_MAC SSL_USE_OPENSSL FEATURE_ENABLE_SSL HAVE_SRTP HAVE_OPENSSL_SSL_H SCTP_PROCESS_LEVEL_LOCKS SCTP_SIMPLE_ALLOCATOR SCTP_USE_OPENSSL_SHA1 __Userspace__ HAVE_SA_LEN HAVE_SCONN_LEN __APPLE_USE_RFC_2292 __Userspace_os_Darwin NON_WINDOWS_DEFINE HAVE_WEBRTC_VIDEO HAVE_WEBRTC_VOICE WEBRTC_INTELLIGIBILITY_ENHANCER=0 WEBRTC_APM_DEBUG_DUMP=0 WEBRTC_NS_FLOAT WEBRTC_USE_BUILTIN_ILBC WEBRTC_CODEC_ILBC WEBRTC_USE_BUILTIN_OPUS WEBRTC_CODEC_OPUS WEBRTC_CODEC_ISAC WEBRTC_CODEC_RED RTC_DISABLE_VP9 RTC_DISABLE_VP8 HAVE_STDINT_H HAVE_STDLIB_H HAVE_UINT64_T OPENSSL HAVE_CONFIG_H WEBRTC_WEBKIT_BUILD HAVE_PTHREAD_COND_TIMEDWAIT_RELATIVE  HAVE_SCTP WEBRTC_CODEC_G711 WEBRTC_CODEC_G722 WEBRTC_OPUS_VARIABLE_COMPLEXITY=0 WEBRTC_USE_BUILTIN_ISAC_FIX=1 WEBRTC_USE_BUILTIN_ISAC_FLOAT=0 USE_BUILTIN_SW_CODECS $(inherited);
> 
> GCC_PREPROCESSOR_DEFINITIONS[sdk=macosx*] = $(inherited) WEBRTC_USE_VTB_HARDWARE_ENCODER;
> GCC_PREPROCESSOR_DEFINITIONS[sdk=iphoneos*] = $(inherited) WEBRTC_IOS;
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/ABSEIL_ISSUE_TEMPLATE.md b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/ABSEIL_ISSUE_TEMPLATE.md
>new file mode 100644
>index 00000000000..ed5461f166c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/ABSEIL_ISSUE_TEMPLATE.md
>@@ -0,0 +1,22 @@
>+Please submit a new Abseil Issue using the template below:
>+
>+## [Short title of proposed API change(s)]
>+
>+--------------------------------------------------------------------------------
>+--------------------------------------------------------------------------------
>+
>+## Background
>+
>+[Provide the background information that is required in order to evaluate the
>+proposed API changes. No controversial claims should be made here. If there are
>+design constraints that need to be considered, they should be presented here
>+**along with justification for those constraints**. Linking to other docs is
>+good, but please keep the **pertinent information as self contained** as
>+possible in this section.]
>+
>+## Proposed API Change (s)
>+
>+[Please clearly describe the API change(s) being proposed. If multiple changes,
>+please keep them clearly distinguished. When possible, **use example code
>+snippets to illustrate before-after API usages**. List pros-n-cons. Highlight
>+the main questions that you want to be answered. Given the Abseil project compatibility requirements, describe why the API change is safe.]
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/AUTHORS b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/AUTHORS
>new file mode 100644
>index 00000000000..976d31defc2
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/AUTHORS
>@@ -0,0 +1,6 @@
>+# This is the list of Abseil authors for copyright purposes.
>+#
>+# This does not necessarily list everyone who has contributed code, since in
>+# some cases, their employer may be the copyright holder.  To see the full list
>+# of contributors, see the revision history in source control.
>+Google Inc.
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/BUILD.gn
>new file mode 100644
>index 00000000000..253a602e2f1
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/BUILD.gn
>@@ -0,0 +1,77 @@
>+# Copyright (c) 2018 The Chromium Authors. All rights reserved.
>+# Use of this source code is governed by a BSD-style license that can be
>+# found in the LICENSE file.
>+
>+# Flags specified here must not impact ABI. Code compiled with and without these
>+# opts will be linked together, and in some cases headers compiled with and
>+# without these options will be part of the same program.
>+
>+import("//build/toolchain/toolchain.gni")
>+
>+group("default") {
>+  deps = [
>+    "absl/types:any",
>+    "absl/types:bad_any_cast",
>+    "absl/types:span",
>+    "absl/types:optional",
>+    "absl/types:bad_optional_access",
>+  ]
>+}
>+
>+config("absl_include_config") {
>+  include_dirs = [ "." ]
>+}
>+
>+config("absl_define_config") {
>+  defines = [ "ABSL_ALLOCATOR_NOTHROW=1" ]
>+}
>+
>+config("absl_default_cflags_cc") {
>+  cflags_cc = []
>+  if (is_clang) {
>+    cflags_cc += [
>+      # TODO(crbug.com/588506): Explicitly enable conversion warnings.
>+      "-Wbool-conversion",
>+      "-Wconstant-conversion",
>+      "-Wenum-conversion",
>+      "-Wint-conversion",
>+      "-Wliteral-conversion",
>+      "-Wnon-literal-null-conversion",
>+      "-Wnull-conversion",
>+      "-Wobjc-literal-conversion",
>+      "-Wno-sign-conversion",
>+      "-Wstring-conversion",
>+    ]
>+    if (!is_nacl && !use_xcode_clang) {
>+      cflags_cc += [ "-Wbitfield-enum-conversion" ]
>+    }
>+  }
>+  if (is_win) {
>+    cflags_cc += [
>+      "/wd4005",  # macro-redefinition
>+      "/wd4018",  # sign-compare
>+      "/wd4068",  # unknown pragma
>+      "/wd4702",  # unreachable code
>+    ]
>+  }
>+}
>+
>+config("absl_test_cflags_cc") {
>+  cflags_cc = []
>+  if (is_clang || !is_win) {
>+    cflags_cc += [
>+      "-Wno-conversion-null",
>+      "-Wno-missing-declarations",
>+      "-Wno-sign-compare",
>+      "-Wno-unused-function",
>+      "-Wno-unused-parameter",
>+      "-Wno-unused-private-field",
>+    ]
>+  }
>+  if (is_win) {
>+    cflags_cc += [
>+      "/wd4018",  # signed/unsigned mismatch
>+      "/wd4101",  # unreferenced local variable
>+    ]
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/CMake/AbseilHelpers.cmake b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/CMake/AbseilHelpers.cmake
>new file mode 100644
>index 00000000000..e4eafe49e47
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/CMake/AbseilHelpers.cmake
>@@ -0,0 +1,170 @@
>+#
>+# Copyright 2017 The Abseil Authors.
>+#
>+# Licensed under the Apache License, Version 2.0 (the "License");
>+# you may not use this file except in compliance with the License.
>+# You may obtain a copy of the License at
>+#
>+#    http://www.apache.org/licenses/LICENSE-2.0
>+#
>+# Unless required by applicable law or agreed to in writing, software
>+# distributed under the License is distributed on an "AS IS" BASIS,
>+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+# See the License for the specific language governing permissions and
>+# limitations under the License.
>+#
>+
>+include(CMakeParseArguments)
>+
>+# The IDE folder for Abseil that will be used if Abseil is included in a CMake
>+# project that sets
>+#    set_property(GLOBAL PROPERTY USE_FOLDERS ON)
>+# For example, Visual Studio supports folders.
>+set(ABSL_IDE_FOLDER Abseil)
>+
>+#
>+# create a library in the absl namespace
>+#
>+# parameters
>+# SOURCES : sources files for the library
>+# PUBLIC_LIBRARIES: targets and flags for linking phase
>+# PRIVATE_COMPILE_FLAGS: compile flags for the library. Will not be exported.
>+# EXPORT_NAME: export name for the absl:: target export
>+# TARGET: target name
>+#
>+# create a target associated to <NAME>
>+# libraries are installed under CMAKE_INSTALL_FULL_LIBDIR by default
>+#
>+function(absl_library)
>+  cmake_parse_arguments(ABSL_LIB
>+    "DISABLE_INSTALL" # keep that in case we want to support installation one day
>+    "TARGET;EXPORT_NAME"
>+    "SOURCES;PUBLIC_LIBRARIES;PRIVATE_COMPILE_FLAGS"
>+    ${ARGN}
>+  )
>+
>+  set(_NAME ${ABSL_LIB_TARGET})
>+  string(TOUPPER ${_NAME} _UPPER_NAME)
>+
>+  add_library(${_NAME} STATIC ${ABSL_LIB_SOURCES})
>+
>+  target_compile_options(${_NAME} PRIVATE ${ABSL_COMPILE_CXXFLAGS} ${ABSL_LIB_PRIVATE_COMPILE_FLAGS})
>+  target_link_libraries(${_NAME} PUBLIC ${ABSL_LIB_PUBLIC_LIBRARIES})
>+  target_include_directories(${_NAME}
>+    PUBLIC ${ABSL_COMMON_INCLUDE_DIRS} ${ABSL_LIB_PUBLIC_INCLUDE_DIRS}
>+    PRIVATE ${ABSL_LIB_PRIVATE_INCLUDE_DIRS}
>+  )
>+  # Add all Abseil targets to a a folder in the IDE for organization.
>+  set_property(TARGET ${_NAME} PROPERTY FOLDER ${ABSL_IDE_FOLDER})
>+
>+  if(ABSL_LIB_EXPORT_NAME)
>+    add_library(absl::${ABSL_LIB_EXPORT_NAME} ALIAS ${_NAME})
>+  endif()
>+endfunction()
>+
>+
>+
>+#
>+# header only virtual target creation
>+#
>+function(absl_header_library)
>+  cmake_parse_arguments(ABSL_HO_LIB
>+    "DISABLE_INSTALL"
>+    "EXPORT_NAME;TARGET"
>+    "PUBLIC_LIBRARIES;PRIVATE_COMPILE_FLAGS;PUBLIC_INCLUDE_DIRS;PRIVATE_INCLUDE_DIRS"
>+    ${ARGN}
>+  )
>+
>+  set(_NAME ${ABSL_HO_LIB_TARGET})
>+
>+  set(__dummy_header_only_lib_file "${CMAKE_CURRENT_BINARY_DIR}/${_NAME}_header_only_dummy.cc")
>+
>+  if(NOT EXISTS ${__dummy_header_only_lib_file})
>+    file(WRITE ${__dummy_header_only_lib_file}
>+      "/* generated file for header-only cmake target */
>+
>+      namespace absl {
>+
>+       // single meaningless symbol
>+       void ${_NAME}__header_fakesym() {}
>+      }  // namespace absl
>+      "
>+    )
>+  endif()
>+
>+
>+  add_library(${_NAME} ${__dummy_header_only_lib_file})
>+  target_link_libraries(${_NAME} PUBLIC ${ABSL_HO_LIB_PUBLIC_LIBRARIES})
>+  target_include_directories(${_NAME}
>+    PUBLIC ${ABSL_COMMON_INCLUDE_DIRS} ${ABSL_HO_LIB_PUBLIC_INCLUDE_DIRS}
>+    PRIVATE ${ABSL_HO_LIB_PRIVATE_INCLUDE_DIRS}
>+  )
>+
>+  # Add all Abseil targets to a a folder in the IDE for organization.
>+  set_property(TARGET ${_NAME} PROPERTY FOLDER ${ABSL_IDE_FOLDER})
>+
>+  if(ABSL_HO_LIB_EXPORT_NAME)
>+    add_library(absl::${ABSL_HO_LIB_EXPORT_NAME} ALIAS ${_NAME})
>+  endif()
>+
>+endfunction()
>+
>+
>+#
>+# create an abseil unit_test and add it to the executed test list
>+#
>+# parameters
>+# TARGET: target name prefix
>+# SOURCES: sources files for the tests
>+# PUBLIC_LIBRARIES: targets and flags for linking phase.
>+# PRIVATE_COMPILE_FLAGS: compile flags for the test. Will not be exported.
>+#
>+# create a target associated to <NAME>_bin
>+#
>+# all tests will be register for execution with add_test()
>+#
>+# test compilation and execution is disable when BUILD_TESTING=OFF
>+#
>+function(absl_test)
>+
>+  cmake_parse_arguments(ABSL_TEST
>+    ""
>+    "TARGET"
>+    "SOURCES;PUBLIC_LIBRARIES;PRIVATE_COMPILE_FLAGS;PUBLIC_INCLUDE_DIRS"
>+    ${ARGN}
>+  )
>+
>+
>+  if(BUILD_TESTING)
>+
>+    set(_NAME ${ABSL_TEST_TARGET})
>+    string(TOUPPER ${_NAME} _UPPER_NAME)
>+
>+    add_executable(${_NAME}_bin ${ABSL_TEST_SOURCES})
>+
>+    target_compile_options(${_NAME}_bin PRIVATE ${ABSL_COMPILE_CXXFLAGS} ${ABSL_TEST_PRIVATE_COMPILE_FLAGS})
>+    target_link_libraries(${_NAME}_bin PUBLIC ${ABSL_TEST_PUBLIC_LIBRARIES} ${ABSL_TEST_COMMON_LIBRARIES})
>+    target_include_directories(${_NAME}_bin
>+      PUBLIC ${ABSL_COMMON_INCLUDE_DIRS} ${ABSL_TEST_PUBLIC_INCLUDE_DIRS}
>+      PRIVATE ${GMOCK_INCLUDE_DIRS} ${GTEST_INCLUDE_DIRS}
>+    )
>+
>+    # Add all Abseil targets to a a folder in the IDE for organization.
>+    set_property(TARGET ${_NAME}_bin PROPERTY FOLDER ${ABSL_IDE_FOLDER})
>+
>+    add_test(${_NAME} ${_NAME}_bin)
>+  endif(BUILD_TESTING)
>+
>+endfunction()
>+
>+
>+
>+
>+function(check_target my_target)
>+
>+  if(NOT TARGET ${my_target})
>+    message(FATAL_ERROR " ABSL: compiling absl requires a ${my_target} CMake target in your project,
>+                   see CMake/README.md for more details")
>+  endif(NOT TARGET ${my_target})
>+
>+endfunction()
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/CMake/CMakeLists.txt.in b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/CMake/CMakeLists.txt.in
>new file mode 100644
>index 00000000000..d60a33e9ac6
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/CMake/CMakeLists.txt.in
>@@ -0,0 +1,15 @@
>+cmake_minimum_required(VERSION 2.8.2)
>+
>+project(googletest-download NONE)
>+
>+include(ExternalProject)
>+ExternalProject_Add(googletest
>+  GIT_REPOSITORY    https://github.com/google/googletest.git
>+  GIT_TAG           master
>+  SOURCE_DIR        "${CMAKE_BINARY_DIR}/googletest-src"
>+  BINARY_DIR        "${CMAKE_BINARY_DIR}/googletest-build"
>+  CONFIGURE_COMMAND ""
>+  BUILD_COMMAND     ""
>+  INSTALL_COMMAND   ""
>+  TEST_COMMAND      ""
>+)
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/CMake/DownloadGTest.cmake b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/CMake/DownloadGTest.cmake
>new file mode 100644
>index 00000000000..9d4132158b8
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/CMake/DownloadGTest.cmake
>@@ -0,0 +1,32 @@
>+# Downloads and unpacks googletest at configure time.  Based on the instructions
>+# at https://github.com/google/googletest/tree/master/googletest#incorporating-into-an-existing-cmake-project
>+
>+# Download the latest googletest from Github master
>+configure_file(
>+  ${CMAKE_CURRENT_LIST_DIR}/CMakeLists.txt.in
>+  googletest-download/CMakeLists.txt
>+)
>+
>+# Configure and build the downloaded googletest source
>+execute_process(COMMAND ${CMAKE_COMMAND} -G "${CMAKE_GENERATOR}" .
>+  RESULT_VARIABLE result
>+  WORKING_DIRECTORY ${CMAKE_BINARY_DIR}/googletest-download )
>+if(result)
>+  message(FATAL_ERROR "CMake step for googletest failed: ${result}")
>+endif()
>+
>+execute_process(COMMAND ${CMAKE_COMMAND} --build .
>+  RESULT_VARIABLE result
>+  WORKING_DIRECTORY ${CMAKE_BINARY_DIR}/googletest-download)
>+if(result)
>+  message(FATAL_ERROR "Build step for googletest failed: ${result}")
>+endif()
>+
>+# Prevent overriding the parent project's compiler/linker settings on Windows
>+set(gtest_force_shared_crt ON CACHE BOOL "" FORCE)
>+
>+# Add googletest directly to our build. This defines the gtest and gtest_main
>+# targets.
>+add_subdirectory(${CMAKE_BINARY_DIR}/googletest-src
>+                 ${CMAKE_BINARY_DIR}/googletest-build
>+                 EXCLUDE_FROM_ALL)
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/CMake/README.md b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/CMake/README.md
>new file mode 100644
>index 00000000000..79bbe24d5ad
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/CMake/README.md
>@@ -0,0 +1,107 @@
>+# Abseil CMake Build Instructions
>+
>+Abseil comes with a CMake build script ([CMakeLists.txt](../CMakeLists.txt))
>+that can be used on a wide range of platforms ("C" stands for cross-platform.).
>+If you don't have CMake installed already, you can download it for free from
>+<http://www.cmake.org/>.
>+
>+CMake works by generating native makefiles or build projects that can
>+be used in the compiler environment of your choice.
>+
>+For API/ABI compatibility reasons, we strongly recommend building Abseil in a
>+subdirectory of your project or as an embedded dependency.
>+
>+## Incorporating Abseil Into a CMake Project
>+
>+The recommendations below are similar to those for using CMake within the
>+googletest framework
>+(<https://github.com/google/googletest/blob/master/googletest/README.md#incorporating-into-an-existing-cmake-project>)
>+
>+### Step-by-Step Instructions
>+
>+1. If you want to build the Abseil tests, integrate the Abseil dependency
>+[Google Test](https://github.com/google/googletest) into your CMake project. To disable Abseil tests, you have to pass
>+`-DBUILD_TESTING=OFF` when configuring your project with CMake.
>+
>+2. Download Abseil and copy it into a subdirectory in your CMake project or add
>+Abseil as a [git submodule](https://git-scm.com/docs/git-submodule) in your
>+CMake project.
>+
>+3. You can then use the CMake command
>+[`add_subdirectory()`](https://cmake.org/cmake/help/latest/command/add_subdirectory.html)
>+to include Abseil directly in your CMake project.
>+
>+4. Add the **absl::** target you wish to use to the
>+[`target_link_libraries()`](https://cmake.org/cmake/help/latest/command/target_link_libraries.html)
>+section of your executable or of your library.<br>
>+Here is a short CMakeLists.txt example of a project file using Abseil.
>+
>+```cmake
>+cmake_minimum_required(VERSION 2.8.12)
>+project(my_project)
>+
>+set(CMAKE_CXX_FLAGS "-std=c++11 -stdlib=libc++ ${CMAKE_CXX_FLAGS}")
>+
>+if(MSVC)
>+  # /wd4005  macro-redefinition
>+  # /wd4068  unknown pragma
>+  # /wd4244  conversion from 'type1' to 'type2'
>+  # /wd4267  conversion from 'size_t' to 'type2'
>+  # /wd4800  force value to bool 'true' or 'false' (performance warning)
>+  add_compile_options(/wd4005 /wd4068 /wd4244 /wd4267 /wd4800)
>+  add_definitions(/DNOMINMAX /DWIN32_LEAN_AND_MEAN=1 /D_CRT_SECURE_NO_WARNINGS)
>+endif()
>+
>+add_subdirectory(abseil-cpp)
>+
>+add_executable(my_exe source.cpp)
>+target_link_libraries(my_exe absl::base absl::synchronization absl::strings)
>+```
>+
>+### Running Abseil Tests with CMake
>+
>+Use the `-DABSL_RUN_TESTS=ON` flag to run Abseil tests.  Note that if the `-DBUILD_TESTING=OFF` flag is passed then Abseil tests will not be run.
>+
>+You will need to provide Abseil with a Googletest dependency.  There are two
>+options for how to do this:
>+
>+* Use `-DABSL_USE_GOOGLETEST_HEAD`.  This will automatically download the latest
>+Googletest source into the build directory at configure time.  Googletest will
>+then be compiled directly alongside Abseil's tests.
>+* Manually integrate Googletest with your build.  See
>+https://github.com/google/googletest/blob/master/googletest/README.md#using-cmake
>+for more information on using Googletest in a CMake project.
>+
>+For example, to run just the Abseil tests, you could use this script:
>+
>+```
>+cd path/to/abseil-cpp
>+mkdir build
>+cd build
>+cmake -DABSL_USE_GOOGLETEST_HEAD=ON -DABSL_RUN_TESTS=ON ..
>+make -j
>+ctest
>+```
>+
>+Currently, we only run our tests with CMake in a Linux environment, but we are
>+working on the rest of our supported platforms. See
>+https://github.com/abseil/abseil-cpp/projects/1 and
>+https://github.com/abseil/abseil-cpp/issues/109 for more information.
>+
>+### Available Abseil CMake Public Targets
>+
>+Here's a non-exhaustive list of Abseil CMake public targets:
>+
>+```cmake
>+absl::base
>+absl::algorithm
>+absl::container
>+absl::debugging
>+absl::memory
>+absl::meta
>+absl::numeric
>+absl::strings
>+absl::synchronization
>+absl::time
>+absl::utility
>+```
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/CMakeLists.txt b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/CMakeLists.txt
>new file mode 100644
>index 00000000000..89a3386f7e2
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/CMakeLists.txt
>@@ -0,0 +1,98 @@
>+#
>+# Copyright 2017 The Abseil Authors.
>+#
>+# Licensed under the Apache License, Version 2.0 (the "License");
>+# you may not use this file except in compliance with the License.
>+# You may obtain a copy of the License at
>+#
>+#      http://www.apache.org/licenses/LICENSE-2.0
>+#
>+# Unless required by applicable law or agreed to in writing, software
>+# distributed under the License is distributed on an "AS IS" BASIS,
>+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+# See the License for the specific language governing permissions and
>+# limitations under the License.
>+#
>+
>+# We require 3.0 for modern, target-based CMake.  We require 3.1 for the use of
>+# CXX_STANDARD in our targets.
>+cmake_minimum_required(VERSION 3.1)
>+project(absl)
>+
>+list(APPEND CMAKE_MODULE_PATH ${PROJECT_SOURCE_DIR}/CMake)
>+
>+include(GNUInstallDirs)
>+include(AbseilHelpers)
>+
>+
>+# config options
>+if (MSVC)
>+  # /wd4005  macro-redefinition
>+  # /wd4068  unknown pragma
>+  # /wd4244  conversion from 'type1' to 'type2'
>+  # /wd4267  conversion from 'size_t' to 'type2'
>+  # /wd4800  force value to bool 'true' or 'false' (performance warning)
>+  add_compile_options(/W3 /WX /wd4005 /wd4068 /wd4244 /wd4267 /wd4800)
>+  add_definitions(/DNOMINMAX /DWIN32_LEAN_AND_MEAN=1 /D_CRT_SECURE_NO_WARNINGS /D_SCL_SECURE_NO_WARNINGS)
>+else()
>+  set(ABSL_STD_CXX_FLAG "-std=c++11" CACHE STRING "c++ std flag (default: c++11)")
>+endif()
>+
>+
>+
>+##
>+## Using absl targets
>+##
>+## all public absl targets are
>+## exported with the absl:: prefix
>+##
>+## e.g absl::base absl::synchronization absl::strings ....
>+##
>+## DO NOT rely on the internal targets outside of the prefix
>+
>+
>+# include current path
>+list(APPEND ABSL_COMMON_INCLUDE_DIRS ${CMAKE_CURRENT_SOURCE_DIR})
>+
>+# -std=X
>+set(CMAKE_CXX_FLAGS "${ABSL_STD_CXX_FLAG} ${CMAKE_CXX_FLAGS}")
>+
>+# -fexceptions
>+set(ABSL_EXCEPTIONS_FLAG "${CMAKE_CXX_EXCEPTIONS}")
>+
>+# find dependencies
>+## pthread
>+find_package(Threads REQUIRED)
>+
>+option(ABSL_USE_GOOGLETEST_HEAD
>+  "If ON, abseil will download HEAD from googletest at config time." OFF)
>+
>+option(ABSL_RUN_TESTS "If ON, Abseil tests will be run." OFF)
>+
>+if(${ABSL_RUN_TESTS})
>+  # enable CTest.  This will set BUILD_TESTING to ON unless otherwise specified
>+  # on the command line
>+  include(CTest)
>+  enable_testing()
>+endif()
>+
>+## check targets
>+if(BUILD_TESTING)
>+
>+  if(${ABSL_USE_GOOGLETEST_HEAD})
>+    include(CMake/DownloadGTest.cmake)
>+  endif()
>+
>+  check_target(gtest)
>+  check_target(gtest_main)
>+  check_target(gmock)
>+
>+  list(APPEND ABSL_TEST_COMMON_LIBRARIES
>+    gtest_main
>+    gtest
>+    gmock
>+    ${CMAKE_THREAD_LIBS_INIT}
>+  )
>+endif()
>+
>+add_subdirectory(absl)
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/CONTRIBUTING.md b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/CONTRIBUTING.md
>new file mode 100644
>index 00000000000..40351ddcfaa
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/CONTRIBUTING.md
>@@ -0,0 +1,91 @@
>+# How to Contribute to Abseil
>+
>+We'd love to accept your patches and contributions to this project. There are
>+just a few small guidelines you need to follow.
>+
>+NOTE: If you are new to GitHub, please start by reading [Pull Request
>+howto](https://help.github.com/articles/about-pull-requests/)
>+
>+## Contributor License Agreement
>+
>+Contributions to this project must be accompanied by a Contributor License
>+Agreement. You (or your employer) retain the copyright to your contribution,
>+this simply gives us permission to use and redistribute your contributions as
>+part of the project. Head over to <https://cla.developers.google.com/> to see
>+your current agreements on file or to sign a new one.
>+
>+You generally only need to submit a CLA once, so if you've already submitted one
>+(even if it was for a different project), you probably don't need to do it
>+again.
>+
>+## Coding Style
>+
>+To keep the source consistent, readable, diffable and easy to merge, we use a
>+fairly rigid coding style, as defined by the
>+[google-styleguide](https://github.com/google/styleguide) project. All patches
>+will be expected to conform to the style outlined
>+[here](https://google.github.io/styleguide/cppguide.html).
>+
>+## Guidelines for Pull Requests
>+
>+*   If you are a Googler, it is preferable to first create an internal CL and
>+    have it reviewed and submitted. The code propagation process will deliver
>+    the change to GitHub.
>+
>+*   Create **small PRs** that are narrowly focused on **addressing a single
>+    concern**. We often receive PRs that are trying to fix several things at a
>+    time, but if only one fix is considered acceptable, nothing gets merged and
>+    both author's & review's time is wasted. Create more PRs to address
>+    different concerns and everyone will be happy.
>+
>+*   For speculative changes, consider opening an [Abseil
>+    issue](https://github.com/abseil/abseil-cpp/issues) and discussing it first.
>+    If you are suggesting a behavioral or API change, consider starting with an
>+    [Abseil proposal template](ABSEIL_ISSUE_TEMPLATE.md).
>+
>+*   Provide a good **PR description** as a record of **what** change is being
>+    made and **why** it was made. Link to a GitHub issue if it exists.
>+
>+*   Don't fix code style and formatting unless you are already changing that
>+    line to address an issue. Formatting of modified lines may be done using
>+   `git clang-format`. PRs with irrelevant changes won't be merged. If
>+    you do want to fix formatting or style, do that in a separate PR.
>+
>+*   Unless your PR is trivial, you should expect there will be reviewer comments
>+    that you'll need to address before merging. We expect you to be reasonably
>+    responsive to those comments, otherwise the PR will be closed after 2-3
>+    weeks of inactivity.
>+
>+*   Maintain **clean commit history** and use **meaningful commit messages**.
>+    PRs with messy commit history are difficult to review and won't be merged.
>+    Use `rebase -i upstream/master` to curate your commit history and/or to
>+    bring in latest changes from master (but avoid rebasing in the middle of a
>+    code review).
>+
>+*   Keep your PR up to date with upstream/master (if there are merge conflicts,
>+    we can't really merge your change).
>+
>+*   **All tests need to be passing** before your change can be merged. We
>+    recommend you **run tests locally** (see below)
>+
>+*   Exceptions to the rules can be made if there's a compelling reason for doing
>+    so. That is - the rules are here to serve us, not the other way around, and
>+    the rules need to be serving their intended purpose to be valuable.
>+
>+*   All submissions, including submissions by project members, require review.
>+
>+## Running Tests
>+
>+Use "bazel test <>" functionality to run the unit tests.
>+
>+Prerequisites for building and running tests are listed in
>+[README.md](README.md)
>+
>+## Abseil Committers
>+
>+The current members of the Abseil engineering team are the only committers at
>+present.
>+
>+## Release Process
>+
>+Abseil lives at head, where latest-and-greatest code can be found.
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/LICENSE b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/LICENSE
>new file mode 100644
>index 00000000000..fef7d967815
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/LICENSE
>@@ -0,0 +1,204 @@
>+
>+                                 Apache License
>+                           Version 2.0, January 2004
>+                        http://www.apache.org/licenses/
>+
>+   TERMS AND CONDITIONS FOR USE, REPRODUCTION, AND DISTRIBUTION
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>+      "License" shall mean the terms and conditions for use, reproduction,
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>+   5. Submission of Contributions. Unless You explicitly state otherwise,
>+      any Contribution intentionally submitted for inclusion in the Work
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>+   8. Limitation of Liability. In no event and under no legal theory,
>+      whether in tort (including negligence), contract, or otherwise,
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>+      negligent acts) or agreed to in writing, shall any Contributor be
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>+      on Your own behalf and on Your sole responsibility, not on behalf
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>+      of your accepting any such warranty or additional liability.
>+
>+   END OF TERMS AND CONDITIONS
>+
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>+
>+      To apply the Apache License to your work, attach the following
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>+      replaced with your own identifying information. (Don't include
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>+      file or class name and description of purpose be included on the
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>+   You may obtain a copy of the License at
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>+       http://www.apache.org/licenses/LICENSE-2.0
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>+   Unless required by applicable law or agreed to in writing, software
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>+   
>+   
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/LTS.md b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/LTS.md
>new file mode 100644
>index 00000000000..385b4f062bf
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/LTS.md
>@@ -0,0 +1,13 @@
>+# Long Term Support (LTS) Branches
>+
>+This repository contains periodic snapshots of the Abseil codebase that are
>+Long Term Support (LTS) branches. An LTS branch allows you to use a known
>+version of Abseil without interfering with other projects which may also, in
>+turn, use Abseil. (For more information about our releases, see the
>+[Abseil Release Management](https://abseil.io/about/releases) guide.)
>+
>+## LTS Branches
>+
>+The following lists LTS branches and the dates on which they have been released:
>+
>+* [LTS Branch June 20, 2018](https://github.com/abseil/abseil-cpp/tree/lts_2018_06_20/)
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/OWNERS b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/OWNERS
>new file mode 100644
>index 00000000000..ceca4a8ba2c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/OWNERS
>@@ -0,0 +1,2 @@
>+mbonadei@chromium.org
>+phoglund@chromium.org
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/README.chromium b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/README.chromium
>new file mode 100644
>index 00000000000..91d6d381a47
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/README.chromium
>@@ -0,0 +1,34 @@
>+Name: Abseil
>+Short Name: absl
>+URL: https://github.com/abseil/abseil-cpp
>+License: Apache 2.0
>+License File: LICENSE
>+Version: 0
>+Revision: bea85b52733022294eef108a2e42d77b616ddca2
>+Security Critical: yes
>+
>+Description:
>+This directory contains the source code of Abseil for C++. This can be used by
>+Chromium's dependencies, but shouldn't be used by Chromium itself.
>+See: https://goo.gl/TgnJb8.
>+
>+How to update Abseil:
>+
>+1. Download the code from the Abseil git repository (see URL).
>+
>+2. Copy the content of the Abseil git repo to //third_party/abseil-cpp.
>+
>+3. From //third_party/abseil-cpp/ launch ./rename_dynamic_annotations.sh.
>+   This script will rewrite dynamic_annotations macros and function inside
>+   Abseil in order to avoid ODR violations and macro clashing with Chromium
>+   (see: https://github.com/abseil/abseil-cpp/issues/122).
>+
>+Local Modifications:
>+
>+* absl/copts.bzl has been translated to //third_party/absl-cpp/BUILD.gn. Both
>+  files contain lists of compiler flags in order to reduce duplication.
>+
>+* All the BUILD.bazel files has been translated to BUILD.gn files.
>+
>+* Functions and macros in absl/base/dynamic_annotations.{h,cc} have been renamed
>+  to avoid ODR violations (see step 3).
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/README.md b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/README.md
>new file mode 100644
>index 00000000000..8eed5751bf5
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/README.md
>@@ -0,0 +1,108 @@
>+# Abseil - C++ Common Libraries
>+
>+The repository contains the Abseil C++ library code. Abseil is an open-source
>+collection of C++ code (compliant to C++11) designed to augment the C++
>+standard library.
>+
>+## Table of Contents
>+
>+- [About Abseil](#about)
>+- [Quickstart](#quickstart)
>+- [Building Abseil](#build)
>+- [Codemap](#codemap)
>+- [License](#license)
>+- [Links](#links)
>+
>+<a name="about"></a>
>+## About Abseil
>+
>+Abseil is an open-source collection of C++ library code designed to augment
>+the C++ standard library. The Abseil library code is collected from Google's
>+own C++ code base, has been extensively tested and used in production, and
>+is the same code we depend on in our daily coding lives.
>+
>+In some cases, Abseil provides pieces missing from the C++ standard; in
>+others, Abseil provides alternatives to the standard for special needs
>+we've found through usage in the Google code base. We denote those cases
>+clearly within the library code we provide you.
>+
>+Abseil is not meant to be a competitor to the standard library; we've
>+just found that many of these utilities serve a purpose within our code
>+base, and we now want to provide those resources to the C++ community as
>+a whole.
>+
>+<a name="quickstart"></a>
>+## Quickstart
>+
>+If you want to just get started, make sure you at least run through the
>+[Abseil Quickstart](https://abseil.io/docs/cpp/quickstart). The Quickstart
>+contains information about setting up your development environment, downloading
>+the Abseil code, running tests, and getting a simple binary working.
>+
>+<a name="build"></a>
>+## Building Abseil
>+
>+[Bazel](http://bazel.build) is the official build system for Abseil,
>+which is supported on most major platforms (Linux, Windows, MacOS, for example)
>+and compilers. See the [quickstart](https://abseil.io/docs/cpp/quickstart) for
>+more information on building Abseil using the Bazel build system.
>+
>+<a name="cmake"></a>
>+If you require CMake support, please check the
>+[CMake build instructions](CMake/README.md).
>+
>+## Codemap
>+
>+Abseil contains the following C++ library components:
>+
>+* [`base`](absl/base/) Abseil Fundamentals
>+  <br /> The `base` library contains initialization code and other code which
>+  all other Abseil code depends on. Code within `base` may not depend on any
>+  other code (other than the C++ standard library).
>+* [`algorithm`](absl/algorithm/)
>+  <br /> The `algorithm` library contains additions to the C++ `<algorithm>`
>+  library and container-based versions of such algorithms.
>+* [`container`](absl/container/)
>+  <br /> The `container` library contains additional STL-style containers.
>+* [`debugging`](absl/debugging/)
>+  <br /> The `debugging` library contains code useful for enabling leak
>+  checks. Future updates will add stacktrace and symbolization utilities.
>+* [`memory`](absl/memory/)
>+  <br /> The `memory` library contains C++11-compatible versions of
>+  `std::make_unique()` and related memory management facilities.
>+* [`meta`](absl/meta/)
>+  <br /> The `meta` library contains C++11-compatible versions of type checks
>+  available within C++14 and C++17 versions of the C++ `<type_traits>` library.
>+* [`numeric`](absl/numeric/)
>+  <br /> The `numeric` library contains C++11-compatible 128-bit integers.
>+* [`strings`](absl/strings/)
>+  <br /> The `strings` library contains a variety of strings routines and
>+  utilities, including a C++11-compatible version of the C++17
>+  `std::string_view` type.
>+* [`synchronization`](absl/synchronization/)
>+  <br /> The `synchronization` library contains concurrency primitives (Abseil's
>+  `absl::Mutex` class, an alternative to `std::mutex`) and a variety of
>+  synchronization abstractions.
>+* [`time`](absl/time/)
>+  <br /> The `time` library contains abstractions for computing with absolute
>+  points in time, durations of time, and formatting and parsing time within
>+  time zones.
>+* [`types`](absl/types/)
>+  <br /> The `types` library contains non-container utility types, like a
>+  C++11-compatible version of the C++17 `std::optional` type.
>+
>+## License
>+
>+The Abseil C++ library is licensed under the terms of the Apache
>+license. See [LICENSE](LICENSE) for more information.
>+
>+## Links
>+
>+For more information about Abseil:
>+
>+* Consult our [Abseil Introduction](http://abseil.io/about/intro)
>+* Read [Why Adopt Abseil](http://abseil.io/about/philosophy) to understand our
>+  design philosophy.
>+* Peruse our
>+  [Abseil Compatibility Guarantees](http://abseil.io/about/compatibility) to
>+  understand both what we promise to you, and what we expect of you in return.
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/WORKSPACE b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/WORKSPACE
>new file mode 100644
>index 00000000000..e4a911978dc
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/WORKSPACE
>@@ -0,0 +1,29 @@
>+workspace(name = "com_google_absl")
>+load("@bazel_tools//tools/build_defs/repo:http.bzl", "http_archive")
>+
>+# Bazel toolchains
>+http_archive(
>+  name = "bazel_toolchains",
>+  urls = [
>+    "https://mirror.bazel.build/github.com/bazelbuild/bazel-toolchains/archive/287b64e0a211fb7c23b74695f8d5f5205b61f4eb.tar.gz",
>+    "https://github.com/bazelbuild/bazel-toolchains/archive/287b64e0a211fb7c23b74695f8d5f5205b61f4eb.tar.gz",
>+  ],
>+  strip_prefix = "bazel-toolchains-287b64e0a211fb7c23b74695f8d5f5205b61f4eb",
>+  sha256 = "aca8ac6afd7745027ee4a43032b51a725a61a75a30f02cc58681ee87e4dcdf4b",
>+)
>+
>+# GoogleTest/GoogleMock framework. Used by most unit-tests.
>+http_archive(
>+     name = "com_google_googletest",
>+     urls = ["https://github.com/google/googletest/archive/b4d4438df9479675a632b2f11125e57133822ece.zip"],  # 2018-07-16
>+     strip_prefix = "googletest-b4d4438df9479675a632b2f11125e57133822ece",
>+     sha256 = "5aaa5d566517cae711e2a3505ea9a6438be1b37fcaae0ebcb96ccba9aa56f23a",
>+)
>+
>+# Google benchmark.
>+http_archive(
>+    name = "com_github_google_benchmark",
>+    urls = ["https://github.com/google/benchmark/archive/16703ff83c1ae6d53e5155df3bb3ab0bc96083be.zip"],
>+    strip_prefix = "benchmark-16703ff83c1ae6d53e5155df3bb3ab0bc96083be",
>+    sha256 = "59f918c8ccd4d74b6ac43484467b500f1d64b40cc1010daa055375b322a43ba3",
>+)
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/BUILD.bazel b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/BUILD.bazel
>new file mode 100644
>index 00000000000..edd0274c5d8
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/BUILD.bazel
>@@ -0,0 +1,51 @@
>+#
>+# Copyright 2017 The Abseil Authors.
>+#
>+# Licensed under the Apache License, Version 2.0 (the "License");
>+# you may not use this file except in compliance with the License.
>+# You may obtain a copy of the License at
>+#
>+#      http://www.apache.org/licenses/LICENSE-2.0
>+#
>+# Unless required by applicable law or agreed to in writing, software
>+# distributed under the License is distributed on an "AS IS" BASIS,
>+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+# See the License for the specific language governing permissions and
>+# limitations under the License.
>+#
>+
>+package(default_visibility = ["//visibility:public"])
>+
>+licenses(["notice"])  # Apache 2.0
>+
>+load(":compiler_config_setting.bzl", "create_llvm_config")
>+
>+create_llvm_config(
>+    name = "llvm_compiler",
>+    visibility = [":__subpackages__"],
>+)
>+
>+# following configs are based on mapping defined in: https://git.io/v5Ijz
>+config_setting(
>+    name = "ios",
>+    values = {
>+        "cpu": "darwin",
>+    },
>+    visibility = [":__subpackages__"],
>+)
>+
>+config_setting(
>+    name = "windows",
>+    values = {
>+        "cpu": "x64_windows",
>+    },
>+    visibility = [":__subpackages__"],
>+)
>+
>+config_setting(
>+    name = "ppc",
>+    values = {
>+        "cpu": "ppc",
>+    },
>+    visibility = [":__subpackages__"],
>+)
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/CMakeLists.txt b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/CMakeLists.txt
>new file mode 100644
>index 00000000000..689f64e258b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/CMakeLists.txt
>@@ -0,0 +1,30 @@
>+#
>+# Copyright 2017 The Abseil Authors.
>+#
>+# Licensed under the Apache License, Version 2.0 (the "License");
>+# you may not use this file except in compliance with the License.
>+# You may obtain a copy of the License at
>+#
>+#      http://www.apache.org/licenses/LICENSE-2.0
>+#
>+# Unless required by applicable law or agreed to in writing, software
>+# distributed under the License is distributed on an "AS IS" BASIS,
>+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+# See the License for the specific language governing permissions and
>+# limitations under the License.
>+#
>+
>+
>+
>+add_subdirectory(base)
>+add_subdirectory(algorithm)
>+add_subdirectory(container)
>+add_subdirectory(debugging)
>+add_subdirectory(memory)
>+add_subdirectory(meta)
>+add_subdirectory(numeric)
>+add_subdirectory(strings)
>+add_subdirectory(synchronization)
>+add_subdirectory(time)
>+add_subdirectory(types)
>+add_subdirectory(utility)
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/algorithm/BUILD.bazel b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/algorithm/BUILD.bazel
>new file mode 100644
>index 00000000000..d04dc71206e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/algorithm/BUILD.bazel
>@@ -0,0 +1,81 @@
>+#
>+# Copyright 2017 The Abseil Authors.
>+#
>+# Licensed under the Apache License, Version 2.0 (the "License");
>+# you may not use this file except in compliance with the License.
>+# You may obtain a copy of the License at
>+#
>+#      http://www.apache.org/licenses/LICENSE-2.0
>+#
>+# Unless required by applicable law or agreed to in writing, software
>+# distributed under the License is distributed on an "AS IS" BASIS,
>+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+# See the License for the specific language governing permissions and
>+# limitations under the License.
>+#
>+
>+load(
>+    "//absl:copts.bzl",
>+    "ABSL_DEFAULT_COPTS",
>+    "ABSL_TEST_COPTS",
>+)
>+
>+package(default_visibility = ["//visibility:public"])
>+
>+licenses(["notice"])  # Apache 2.0
>+
>+cc_library(
>+    name = "algorithm",
>+    hdrs = ["algorithm.h"],
>+    copts = ABSL_DEFAULT_COPTS,
>+)
>+
>+cc_test(
>+    name = "algorithm_test",
>+    size = "small",
>+    srcs = ["algorithm_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    deps = [
>+        ":algorithm",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "algorithm_benchmark",
>+    srcs = ["equal_benchmark.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    tags = ["benchmark"],
>+    deps = [
>+        ":algorithm",
>+        "//absl/base:core_headers",
>+        "@com_github_google_benchmark//:benchmark_main",
>+    ],
>+)
>+
>+cc_library(
>+    name = "container",
>+    hdrs = [
>+        "container.h",
>+    ],
>+    copts = ABSL_DEFAULT_COPTS,
>+    deps = [
>+        ":algorithm",
>+        "//absl/base:core_headers",
>+        "//absl/meta:type_traits",
>+    ],
>+)
>+
>+cc_test(
>+    name = "container_test",
>+    srcs = ["container_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    deps = [
>+        ":container",
>+        "//absl/base",
>+        "//absl/base:core_headers",
>+        "//absl/memory",
>+        "//absl/types:span",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/algorithm/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/algorithm/BUILD.gn
>new file mode 100644
>index 00000000000..37ec665764d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/algorithm/BUILD.gn
>@@ -0,0 +1,44 @@
>+# Copyright 2018 The Chromium Authors. All rights reserved.
>+# Use of this source code is governed by a BSD-style license that can be
>+# found in the LICENSE file.
>+
>+import("//build_overrides/build.gni")
>+
>+if (build_with_chromium) {
>+  visibility = [
>+    "//third_party/webrtc/*",
>+    "//third_party/abseil-cpp/*",
>+    "//third_party/googletest:gtest",
>+  ]
>+} else {
>+  visibility = [ "*" ]
>+}
>+
>+source_set("algorithm") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  public = [
>+    "algorithm.h",
>+  ]
>+}
>+
>+source_set("container") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  public = [
>+    "container.h",
>+  ]
>+  deps = [
>+    ":algorithm",
>+    "../base:core_headers",
>+    "../meta:type_traits",
>+  ]
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/algorithm/CMakeLists.txt b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/algorithm/CMakeLists.txt
>new file mode 100644
>index 00000000000..fdf45c55ed6
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/algorithm/CMakeLists.txt
>@@ -0,0 +1,63 @@
>+#
>+# Copyright 2017 The Abseil Authors.
>+#
>+# Licensed under the Apache License, Version 2.0 (the "License");
>+# you may not use this file except in compliance with the License.
>+# You may obtain a copy of the License at
>+#
>+#      http://www.apache.org/licenses/LICENSE-2.0
>+#
>+# Unless required by applicable law or agreed to in writing, software
>+# distributed under the License is distributed on an "AS IS" BASIS,
>+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+# See the License for the specific language governing permissions and
>+# limitations under the License.
>+#
>+
>+list(APPEND ALGORITHM_PUBLIC_HEADERS
>+  "algorithm.h"
>+  "container.h"
>+)
>+
>+
>+#
>+## TESTS
>+#
>+
>+# test algorithm_test
>+list(APPEND ALGORITHM_TEST_SRC
>+  "algorithm_test.cc"
>+  ${ALGORITHM_PUBLIC_HEADERS}
>+  ${ALGORITHM_INTERNAL_HEADERS}
>+)
>+
>+absl_header_library(
>+  TARGET
>+    absl_algorithm
>+  EXPORT_NAME
>+    algorithm
>+)
>+
>+absl_test(
>+  TARGET
>+    algorithm_test
>+  SOURCES
>+    ${ALGORITHM_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    absl::algorithm
>+)
>+
>+
>+
>+
>+# test container_test
>+set(CONTAINER_TEST_SRC "container_test.cc")
>+
>+absl_test(
>+  TARGET
>+    container_test
>+  SOURCES
>+    ${CONTAINER_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    absl::algorithm
>+)
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/algorithm/algorithm.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/algorithm/algorithm.h
>new file mode 100644
>index 00000000000..3d6586439fe
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/algorithm/algorithm.h
>@@ -0,0 +1,150 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// -----------------------------------------------------------------------------
>+// File: algorithm.h
>+// -----------------------------------------------------------------------------
>+//
>+// This header file contains Google extensions to the standard <algorithm> C++
>+// header.
>+
>+#ifndef ABSL_ALGORITHM_ALGORITHM_H_
>+#define ABSL_ALGORITHM_ALGORITHM_H_
>+
>+#include <algorithm>
>+#include <iterator>
>+#include <type_traits>
>+
>+namespace absl {
>+
>+namespace algorithm_internal {
>+
>+// Performs comparisons with operator==, similar to C++14's `std::equal_to<>`.
>+struct EqualTo {
>+  template <typename T, typename U>
>+  bool operator()(const T& a, const U& b) const {
>+    return a == b;
>+  }
>+};
>+
>+template <typename InputIter1, typename InputIter2, typename Pred>
>+bool EqualImpl(InputIter1 first1, InputIter1 last1, InputIter2 first2,
>+               InputIter2 last2, Pred pred, std::input_iterator_tag,
>+               std::input_iterator_tag) {
>+  while (true) {
>+    if (first1 == last1) return first2 == last2;
>+    if (first2 == last2) return false;
>+    if (!pred(*first1, *first2)) return false;
>+    ++first1;
>+    ++first2;
>+  }
>+}
>+
>+template <typename InputIter1, typename InputIter2, typename Pred>
>+bool EqualImpl(InputIter1 first1, InputIter1 last1, InputIter2 first2,
>+               InputIter2 last2, Pred&& pred, std::random_access_iterator_tag,
>+               std::random_access_iterator_tag) {
>+  return (last1 - first1 == last2 - first2) &&
>+         std::equal(first1, last1, first2, std::forward<Pred>(pred));
>+}
>+
>+// When we are using our own internal predicate that just applies operator==, we
>+// forward to the non-predicate form of std::equal. This enables an optimization
>+// in libstdc++ that can result in std::memcmp being used for integer types.
>+template <typename InputIter1, typename InputIter2>
>+bool EqualImpl(InputIter1 first1, InputIter1 last1, InputIter2 first2,
>+               InputIter2 last2, algorithm_internal::EqualTo /* unused */,
>+               std::random_access_iterator_tag,
>+               std::random_access_iterator_tag) {
>+  return (last1 - first1 == last2 - first2) &&
>+         std::equal(first1, last1, first2);
>+}
>+
>+template <typename It>
>+It RotateImpl(It first, It middle, It last, std::true_type) {
>+  return std::rotate(first, middle, last);
>+}
>+
>+template <typename It>
>+It RotateImpl(It first, It middle, It last, std::false_type) {
>+  std::rotate(first, middle, last);
>+  return std::next(first, std::distance(middle, last));
>+}
>+
>+}  // namespace algorithm_internal
>+
>+// Compares the equality of two ranges specified by pairs of iterators, using
>+// the given predicate, returning true iff for each corresponding iterator i1
>+// and i2 in the first and second range respectively, pred(*i1, *i2) == true
>+//
>+// This comparison takes at most min(`last1` - `first1`, `last2` - `first2`)
>+// invocations of the predicate. Additionally, if InputIter1 and InputIter2 are
>+// both random-access iterators, and `last1` - `first1` != `last2` - `first2`,
>+// then the predicate is never invoked and the function returns false.
>+//
>+// This is a C++11-compatible implementation of C++14 `std::equal`.  See
>+// http://en.cppreference.com/w/cpp/algorithm/equal for more information.
>+template <typename InputIter1, typename InputIter2, typename Pred>
>+bool equal(InputIter1 first1, InputIter1 last1, InputIter2 first2,
>+           InputIter2 last2, Pred&& pred) {
>+  return algorithm_internal::EqualImpl(
>+      first1, last1, first2, last2, std::forward<Pred>(pred),
>+      typename std::iterator_traits<InputIter1>::iterator_category{},
>+      typename std::iterator_traits<InputIter2>::iterator_category{});
>+}
>+
>+// Performs comparison of two ranges specified by pairs of iterators using
>+// operator==.
>+template <typename InputIter1, typename InputIter2>
>+bool equal(InputIter1 first1, InputIter1 last1, InputIter2 first2,
>+           InputIter2 last2) {
>+  return absl::equal(first1, last1, first2, last2,
>+                     algorithm_internal::EqualTo{});
>+}
>+
>+// Performs a linear search for `value` using the iterator `first` up to
>+// but not including `last`, returning true if [`first`, `last`) contains an
>+// element equal to `value`.
>+//
>+// A linear search is of O(n) complexity which is guaranteed to make at most
>+// n = (`last` - `first`) comparisons. A linear search over short containers
>+// may be faster than a binary search, even when the container is sorted.
>+template <typename InputIterator, typename EqualityComparable>
>+bool linear_search(InputIterator first, InputIterator last,
>+                   const EqualityComparable& value) {
>+  return std::find(first, last, value) != last;
>+}
>+
>+// Performs a left rotation on a range of elements (`first`, `last`) such that
>+// `middle` is now the first element. `rotate()` returns an iterator pointing to
>+// the first element before rotation. This function is exactly the same as
>+// `std::rotate`, but fixes a bug in gcc
>+// <= 4.9 where `std::rotate` returns `void` instead of an iterator.
>+//
>+// The complexity of this algorithm is the same as that of `std::rotate`, but if
>+// `ForwardIterator` is not a random-access iterator, then `absl::rotate`
>+// performs an additional pass over the range to construct the return value.
>+
>+template <typename ForwardIterator>
>+ForwardIterator rotate(ForwardIterator first, ForwardIterator middle,
>+                       ForwardIterator last) {
>+  return algorithm_internal::RotateImpl(
>+      first, middle, last,
>+      std::is_same<decltype(std::rotate(first, middle, last)),
>+                   ForwardIterator>());
>+}
>+
>+}  // namespace absl
>+
>+#endif  // ABSL_ALGORITHM_ALGORITHM_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/algorithm/algorithm_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/algorithm/algorithm_test.cc
>new file mode 100644
>index 00000000000..e4322bc4f20
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/algorithm/algorithm_test.cc
>@@ -0,0 +1,182 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/algorithm/algorithm.h"
>+
>+#include <algorithm>
>+#include <list>
>+#include <vector>
>+
>+#include "gmock/gmock.h"
>+#include "gtest/gtest.h"
>+
>+namespace {
>+
>+TEST(EqualTest, DefaultComparisonRandomAccess) {
>+  std::vector<int> v1{1, 2, 3};
>+  std::vector<int> v2 = v1;
>+  std::vector<int> v3 = {1, 2};
>+  std::vector<int> v4 = {1, 2, 4};
>+
>+  EXPECT_TRUE(absl::equal(v1.begin(), v1.end(), v2.begin(), v2.end()));
>+  EXPECT_FALSE(absl::equal(v1.begin(), v1.end(), v3.begin(), v3.end()));
>+  EXPECT_FALSE(absl::equal(v1.begin(), v1.end(), v4.begin(), v4.end()));
>+}
>+
>+TEST(EqualTest, DefaultComparison) {
>+  std::list<int> lst1{1, 2, 3};
>+  std::list<int> lst2 = lst1;
>+  std::list<int> lst3{1, 2};
>+  std::list<int> lst4{1, 2, 4};
>+
>+  EXPECT_TRUE(absl::equal(lst1.begin(), lst1.end(), lst2.begin(), lst2.end()));
>+  EXPECT_FALSE(absl::equal(lst1.begin(), lst1.end(), lst3.begin(), lst3.end()));
>+  EXPECT_FALSE(absl::equal(lst1.begin(), lst1.end(), lst4.begin(), lst4.end()));
>+}
>+
>+TEST(EqualTest, EmptyRange) {
>+  std::vector<int> v1{1, 2, 3};
>+  std::vector<int> empty1;
>+  std::vector<int> empty2;
>+
>+  EXPECT_FALSE(absl::equal(v1.begin(), v1.end(), empty1.begin(), empty1.end()));
>+  EXPECT_FALSE(absl::equal(empty1.begin(), empty1.end(), v1.begin(), v1.end()));
>+  EXPECT_TRUE(
>+      absl::equal(empty1.begin(), empty1.end(), empty2.begin(), empty2.end()));
>+}
>+
>+TEST(EqualTest, MixedIterTypes) {
>+  std::vector<int> v1{1, 2, 3};
>+  std::list<int> lst1{v1.begin(), v1.end()};
>+  std::list<int> lst2{1, 2, 4};
>+  std::list<int> lst3{1, 2};
>+
>+  EXPECT_TRUE(absl::equal(v1.begin(), v1.end(), lst1.begin(), lst1.end()));
>+  EXPECT_FALSE(absl::equal(v1.begin(), v1.end(), lst2.begin(), lst2.end()));
>+  EXPECT_FALSE(absl::equal(v1.begin(), v1.end(), lst3.begin(), lst3.end()));
>+}
>+
>+TEST(EqualTest, MixedValueTypes) {
>+  std::vector<int> v1{1, 2, 3};
>+  std::vector<char> v2{1, 2, 3};
>+  std::vector<char> v3{1, 2};
>+  std::vector<char> v4{1, 2, 4};
>+
>+  EXPECT_TRUE(absl::equal(v1.begin(), v1.end(), v2.begin(), v2.end()));
>+  EXPECT_FALSE(absl::equal(v1.begin(), v1.end(), v3.begin(), v3.end()));
>+  EXPECT_FALSE(absl::equal(v1.begin(), v1.end(), v4.begin(), v4.end()));
>+}
>+
>+TEST(EqualTest, WeirdIterators) {
>+  std::vector<bool> v1{true, false};
>+  std::vector<bool> v2 = v1;
>+  std::vector<bool> v3{true};
>+  std::vector<bool> v4{true, true, true};
>+
>+  EXPECT_TRUE(absl::equal(v1.begin(), v1.end(), v2.begin(), v2.end()));
>+  EXPECT_FALSE(absl::equal(v1.begin(), v1.end(), v3.begin(), v3.end()));
>+  EXPECT_FALSE(absl::equal(v1.begin(), v1.end(), v4.begin(), v4.end()));
>+}
>+
>+TEST(EqualTest, CustomComparison) {
>+  int n[] = {1, 2, 3, 4};
>+  std::vector<int*> v1{&n[0], &n[1], &n[2]};
>+  std::vector<int*> v2 = v1;
>+  std::vector<int*> v3{&n[0], &n[1], &n[3]};
>+  std::vector<int*> v4{&n[0], &n[1]};
>+
>+  auto eq = [](int* a, int* b) { return *a == *b; };
>+
>+  EXPECT_TRUE(absl::equal(v1.begin(), v1.end(), v2.begin(), v2.end(), eq));
>+  EXPECT_FALSE(absl::equal(v1.begin(), v1.end(), v3.begin(), v3.end(), eq));
>+  EXPECT_FALSE(absl::equal(v1.begin(), v1.end(), v4.begin(), v4.end(), eq));
>+}
>+
>+TEST(EqualTest, MoveOnlyPredicate) {
>+  std::vector<int> v1{1, 2, 3};
>+  std::vector<int> v2{4, 5, 6};
>+
>+  // move-only equality predicate
>+  struct Eq {
>+    Eq() = default;
>+    Eq(Eq &&) = default;
>+    Eq(const Eq &) = delete;
>+    Eq &operator=(const Eq &) = delete;
>+    bool operator()(const int a, const int b) const { return a == b; }
>+  };
>+
>+  EXPECT_TRUE(absl::equal(v1.begin(), v1.end(), v1.begin(), v1.end(), Eq()));
>+  EXPECT_FALSE(absl::equal(v1.begin(), v1.end(), v2.begin(), v2.end(), Eq()));
>+}
>+
>+struct CountingTrivialPred {
>+  int* count;
>+  bool operator()(int, int) const {
>+    ++*count;
>+    return true;
>+  }
>+};
>+
>+TEST(EqualTest, RandomAccessComplexity) {
>+  std::vector<int> v1{1, 1, 3};
>+  std::vector<int> v2 = v1;
>+  std::vector<int> v3{1, 2};
>+
>+  do {
>+    int count = 0;
>+    absl::equal(v1.begin(), v1.end(), v2.begin(), v2.end(),
>+                CountingTrivialPred{&count});
>+    EXPECT_LE(count, 3);
>+  } while (std::next_permutation(v2.begin(), v2.end()));
>+
>+  int count = 0;
>+  absl::equal(v1.begin(), v1.end(), v3.begin(), v3.end(),
>+              CountingTrivialPred{&count});
>+  EXPECT_EQ(count, 0);
>+}
>+
>+class LinearSearchTest : public testing::Test {
>+ protected:
>+  LinearSearchTest() : container_{1, 2, 3} {}
>+
>+  static bool Is3(int n) { return n == 3; }
>+  static bool Is4(int n) { return n == 4; }
>+
>+  std::vector<int> container_;
>+};
>+
>+TEST_F(LinearSearchTest, linear_search) {
>+  EXPECT_TRUE(absl::linear_search(container_.begin(), container_.end(), 3));
>+  EXPECT_FALSE(absl::linear_search(container_.begin(), container_.end(), 4));
>+}
>+
>+TEST_F(LinearSearchTest, linear_searchConst) {
>+  const std::vector<int> *const const_container = &container_;
>+  EXPECT_TRUE(
>+      absl::linear_search(const_container->begin(), const_container->end(), 3));
>+  EXPECT_FALSE(
>+      absl::linear_search(const_container->begin(), const_container->end(), 4));
>+}
>+
>+TEST(RotateTest, Rotate) {
>+  std::vector<int> v{0, 1, 2, 3, 4};
>+  EXPECT_EQ(*absl::rotate(v.begin(), v.begin() + 2, v.end()), 0);
>+  EXPECT_THAT(v, testing::ElementsAreArray({2, 3, 4, 0, 1}));
>+
>+  std::list<int> l{0, 1, 2, 3, 4};
>+  EXPECT_EQ(*absl::rotate(l.begin(), std::next(l.begin(), 3), l.end()), 0);
>+  EXPECT_THAT(l, testing::ElementsAreArray({3, 4, 0, 1, 2}));
>+}
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/algorithm/container.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/algorithm/container.h
>new file mode 100644
>index 00000000000..6af8c09799e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/algorithm/container.h
>@@ -0,0 +1,1642 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// -----------------------------------------------------------------------------
>+// File: container.h
>+// -----------------------------------------------------------------------------
>+//
>+// This header file provides Container-based versions of algorithmic functions
>+// within the C++ standard library. The following standard library sets of
>+// functions are covered within this file:
>+//
>+//   * Algorithmic <iterator> functions
>+//   * Algorithmic <numeric> functions
>+//   * <algorithm> functions
>+//
>+// The standard library functions operate on iterator ranges; the functions
>+// within this API operate on containers, though many return iterator ranges.
>+//
>+// All functions within this API are named with a `c_` prefix. Calls such as
>+// `absl::c_xx(container, ...) are equivalent to std:: functions such as
>+// `std::xx(std::begin(cont), std::end(cont), ...)`. Functions that act on
>+// iterators but not conceptually on iterator ranges (e.g. `std::iter_swap`)
>+// have no equivalent here.
>+//
>+// For template parameter and variable naming, `C` indicates the container type
>+// to which the function is applied, `Pred` indicates the predicate object type
>+// to be used by the function and `T` indicates the applicable element type.
>+//
>+
>+#ifndef ABSL_ALGORITHM_CONTAINER_H_
>+#define ABSL_ALGORITHM_CONTAINER_H_
>+
>+#include <algorithm>
>+#include <cassert>
>+#include <iterator>
>+#include <numeric>
>+#include <type_traits>
>+#include <utility>
>+#include <vector>
>+
>+#include "absl/algorithm/algorithm.h"
>+#include "absl/base/macros.h"
>+#include "absl/meta/type_traits.h"
>+
>+namespace absl {
>+
>+namespace container_algorithm_internal {
>+
>+// NOTE: it is important to defer to ADL lookup for building with C++ modules,
>+// especially for headers like <valarray> which are not visible from this file
>+// but specialize std::begin and std::end.
>+using std::begin;
>+using std::end;
>+
>+// The type of the iterator given by begin(c) (possibly std::begin(c)).
>+// ContainerIter<const vector<T>> gives vector<T>::const_iterator,
>+// while ContainerIter<vector<T>> gives vector<T>::iterator.
>+template <typename C>
>+using ContainerIter = decltype(begin(std::declval<C&>()));
>+
>+// An MSVC bug involving template parameter substitution requires us to use
>+// decltype() here instead of just std::pair.
>+template <typename C1, typename C2>
>+using ContainerIterPairType =
>+    decltype(std::make_pair(ContainerIter<C1>(), ContainerIter<C2>()));
>+
>+template <typename C>
>+using ContainerDifferenceType =
>+    decltype(std::distance(std::declval<ContainerIter<C>>(),
>+                           std::declval<ContainerIter<C>>()));
>+
>+template <typename C>
>+using ContainerPointerType =
>+    typename std::iterator_traits<ContainerIter<C>>::pointer;
>+
>+// container_algorithm_internal::c_begin and
>+// container_algorithm_internal::c_end are abbreviations for proper ADL
>+// lookup of std::begin and std::end, i.e.
>+//   using std::begin;
>+//   using std::end;
>+//   std::foo(begin(c), end(c);
>+// becomes
>+//   std::foo(container_algorithm_internal::begin(c),
>+//   container_algorithm_internal::end(c));
>+// These are meant for internal use only.
>+
>+template <typename C>
>+ContainerIter<C> c_begin(C& c) { return begin(c); }
>+
>+template <typename C>
>+ContainerIter<C> c_end(C& c) { return end(c); }
>+
>+}  // namespace container_algorithm_internal
>+
>+// PUBLIC API
>+
>+//------------------------------------------------------------------------------
>+// Abseil algorithm.h functions
>+//------------------------------------------------------------------------------
>+
>+// c_linear_search()
>+//
>+// Container-based version of absl::linear_search() for performing a linear
>+// search within a container.
>+template <typename C, typename EqualityComparable>
>+bool c_linear_search(const C& c, EqualityComparable&& value) {
>+  return linear_search(container_algorithm_internal::c_begin(c),
>+                       container_algorithm_internal::c_end(c),
>+                       std::forward<EqualityComparable>(value));
>+}
>+
>+//------------------------------------------------------------------------------
>+// <iterator> algorithms
>+//------------------------------------------------------------------------------
>+
>+// c_distance()
>+//
>+// Container-based version of the <iterator> `std::distance()` function to
>+// return the number of elements within a container.
>+template <typename C>
>+container_algorithm_internal::ContainerDifferenceType<const C> c_distance(
>+    const C& c) {
>+  return std::distance(container_algorithm_internal::c_begin(c),
>+                       container_algorithm_internal::c_end(c));
>+}
>+
>+//------------------------------------------------------------------------------
>+// <algorithm> Non-modifying sequence operations
>+//------------------------------------------------------------------------------
>+
>+// c_all_of()
>+//
>+// Container-based version of the <algorithm> `std::all_of()` function to
>+// test a condition on all elements within a container.
>+template <typename C, typename Pred>
>+bool c_all_of(const C& c, Pred&& pred) {
>+  return std::all_of(container_algorithm_internal::c_begin(c),
>+                     container_algorithm_internal::c_end(c),
>+                     std::forward<Pred>(pred));
>+}
>+
>+// c_any_of()
>+//
>+// Container-based version of the <algorithm> `std::any_of()` function to
>+// test if any element in a container fulfills a condition.
>+template <typename C, typename Pred>
>+bool c_any_of(const C& c, Pred&& pred) {
>+  return std::any_of(container_algorithm_internal::c_begin(c),
>+                     container_algorithm_internal::c_end(c),
>+                     std::forward<Pred>(pred));
>+}
>+
>+// c_none_of()
>+//
>+// Container-based version of the <algorithm> `std::none_of()` function to
>+// test if no elements in a container fulfil a condition.
>+template <typename C, typename Pred>
>+bool c_none_of(const C& c, Pred&& pred) {
>+  return std::none_of(container_algorithm_internal::c_begin(c),
>+                      container_algorithm_internal::c_end(c),
>+                      std::forward<Pred>(pred));
>+}
>+
>+// c_for_each()
>+//
>+// Container-based version of the <algorithm> `std::for_each()` function to
>+// apply a function to a container's elements.
>+template <typename C, typename Function>
>+decay_t<Function> c_for_each(C&& c, Function&& f) {
>+  return std::for_each(container_algorithm_internal::c_begin(c),
>+                       container_algorithm_internal::c_end(c),
>+                       std::forward<Function>(f));
>+}
>+
>+// c_find()
>+//
>+// Container-based version of the <algorithm> `std::find()` function to find
>+// the first element containing the passed value within a container value.
>+template <typename C, typename T>
>+container_algorithm_internal::ContainerIter<C> c_find(C& c, T&& value) {
>+  return std::find(container_algorithm_internal::c_begin(c),
>+                   container_algorithm_internal::c_end(c),
>+                   std::forward<T>(value));
>+}
>+
>+// c_find_if()
>+//
>+// Container-based version of the <algorithm> `std::find_if()` function to find
>+// the first element in a container matching the given condition.
>+template <typename C, typename Pred>
>+container_algorithm_internal::ContainerIter<C> c_find_if(C& c, Pred&& pred) {
>+  return std::find_if(container_algorithm_internal::c_begin(c),
>+                      container_algorithm_internal::c_end(c),
>+                      std::forward<Pred>(pred));
>+}
>+
>+// c_find_if_not()
>+//
>+// Container-based version of the <algorithm> `std::find_if_not()` function to
>+// find the first element in a container not matching the given condition.
>+template <typename C, typename Pred>
>+container_algorithm_internal::ContainerIter<C> c_find_if_not(C& c,
>+                                                             Pred&& pred) {
>+  return std::find_if_not(container_algorithm_internal::c_begin(c),
>+                          container_algorithm_internal::c_end(c),
>+                          std::forward<Pred>(pred));
>+}
>+
>+// c_find_end()
>+//
>+// Container-based version of the <algorithm> `std::find_end()` function to
>+// find the last subsequence within a container.
>+template <typename Sequence1, typename Sequence2>
>+container_algorithm_internal::ContainerIter<Sequence1> c_find_end(
>+    Sequence1& sequence, Sequence2& subsequence) {
>+  return std::find_end(container_algorithm_internal::c_begin(sequence),
>+                       container_algorithm_internal::c_end(sequence),
>+                       container_algorithm_internal::c_begin(subsequence),
>+                       container_algorithm_internal::c_end(subsequence));
>+}
>+
>+// Overload of c_find_end() for using a predicate evaluation other than `==` as
>+// the function's test condition.
>+template <typename Sequence1, typename Sequence2, typename BinaryPredicate>
>+container_algorithm_internal::ContainerIter<Sequence1> c_find_end(
>+    Sequence1& sequence, Sequence2& subsequence, BinaryPredicate&& pred) {
>+  return std::find_end(container_algorithm_internal::c_begin(sequence),
>+                       container_algorithm_internal::c_end(sequence),
>+                       container_algorithm_internal::c_begin(subsequence),
>+                       container_algorithm_internal::c_end(subsequence),
>+                       std::forward<BinaryPredicate>(pred));
>+}
>+
>+// c_find_first_of()
>+//
>+// Container-based version of the <algorithm> `std::find_first_of()` function to
>+// find the first elements in an ordered set within a container.
>+template <typename C1, typename C2>
>+container_algorithm_internal::ContainerIter<C1> c_find_first_of(C1& container,
>+                                                                C2& options) {
>+  return std::find_first_of(container_algorithm_internal::c_begin(container),
>+                            container_algorithm_internal::c_end(container),
>+                            container_algorithm_internal::c_begin(options),
>+                            container_algorithm_internal::c_end(options));
>+}
>+
>+// Overload of c_find_first_of() for using a predicate evaluation other than
>+// `==` as the function's test condition.
>+template <typename C1, typename C2, typename BinaryPredicate>
>+container_algorithm_internal::ContainerIter<C1> c_find_first_of(
>+    C1& container, C2& options, BinaryPredicate&& pred) {
>+  return std::find_first_of(container_algorithm_internal::c_begin(container),
>+                            container_algorithm_internal::c_end(container),
>+                            container_algorithm_internal::c_begin(options),
>+                            container_algorithm_internal::c_end(options),
>+                            std::forward<BinaryPredicate>(pred));
>+}
>+
>+// c_adjacent_find()
>+//
>+// Container-based version of the <algorithm> `std::adjacent_find()` function to
>+// find equal adjacent elements within a container.
>+template <typename Sequence>
>+container_algorithm_internal::ContainerIter<Sequence> c_adjacent_find(
>+    Sequence& sequence) {
>+  return std::adjacent_find(container_algorithm_internal::c_begin(sequence),
>+                            container_algorithm_internal::c_end(sequence));
>+}
>+
>+// Overload of c_adjacent_find() for using a predicate evaluation other than
>+// `==` as the function's test condition.
>+template <typename Sequence, typename BinaryPredicate>
>+container_algorithm_internal::ContainerIter<Sequence> c_adjacent_find(
>+    Sequence& sequence, BinaryPredicate&& pred) {
>+  return std::adjacent_find(container_algorithm_internal::c_begin(sequence),
>+                            container_algorithm_internal::c_end(sequence),
>+                            std::forward<BinaryPredicate>(pred));
>+}
>+
>+// c_count()
>+//
>+// Container-based version of the <algorithm> `std::count()` function to count
>+// values that match within a container.
>+template <typename C, typename T>
>+container_algorithm_internal::ContainerDifferenceType<const C> c_count(
>+    const C& c, T&& value) {
>+  return std::count(container_algorithm_internal::c_begin(c),
>+                    container_algorithm_internal::c_end(c),
>+                    std::forward<T>(value));
>+}
>+
>+// c_count_if()
>+//
>+// Container-based version of the <algorithm> `std::count_if()` function to
>+// count values matching a condition within a container.
>+template <typename C, typename Pred>
>+container_algorithm_internal::ContainerDifferenceType<const C> c_count_if(
>+    const C& c, Pred&& pred) {
>+  return std::count_if(container_algorithm_internal::c_begin(c),
>+                       container_algorithm_internal::c_end(c),
>+                       std::forward<Pred>(pred));
>+}
>+
>+// c_mismatch()
>+//
>+// Container-based version of the <algorithm> `std::mismatch()` function to
>+// return the first element where two ordered containers differ.
>+template <typename C1, typename C2>
>+container_algorithm_internal::ContainerIterPairType<C1, C2>
>+c_mismatch(C1& c1, C2& c2) {
>+  return std::mismatch(container_algorithm_internal::c_begin(c1),
>+                       container_algorithm_internal::c_end(c1),
>+                       container_algorithm_internal::c_begin(c2));
>+}
>+
>+// Overload of c_mismatch() for using a predicate evaluation other than `==` as
>+// the function's test condition.
>+template <typename C1, typename C2, typename BinaryPredicate>
>+container_algorithm_internal::ContainerIterPairType<C1, C2>
>+c_mismatch(C1& c1, C2& c2, BinaryPredicate&& pred) {
>+  return std::mismatch(container_algorithm_internal::c_begin(c1),
>+                       container_algorithm_internal::c_end(c1),
>+                       container_algorithm_internal::c_begin(c2),
>+                       std::forward<BinaryPredicate>(pred));
>+}
>+
>+// c_equal()
>+//
>+// Container-based version of the <algorithm> `std::equal()` function to
>+// test whether two containers are equal.
>+//
>+// NOTE: the semantics of c_equal() are slightly different than those of
>+// equal(): while the latter iterates over the second container only up to the
>+// size of the first container, c_equal() also checks whether the container
>+// sizes are equal.  This better matches expectations about c_equal() based on
>+// its signature.
>+//
>+// Example:
>+//   vector v1 = <1, 2, 3>;
>+//   vector v2 = <1, 2, 3, 4>;
>+//   equal(std::begin(v1), std::end(v1), std::begin(v2)) returns true
>+//   c_equal(v1, v2) returns false
>+
>+template <typename C1, typename C2>
>+bool c_equal(const C1& c1, const C2& c2) {
>+  return ((c1.size() == c2.size()) &&
>+          std::equal(container_algorithm_internal::c_begin(c1),
>+                     container_algorithm_internal::c_end(c1),
>+                     container_algorithm_internal::c_begin(c2)));
>+}
>+
>+// Overload of c_equal() for using a predicate evaluation other than `==` as
>+// the function's test condition.
>+template <typename C1, typename C2, typename BinaryPredicate>
>+bool c_equal(const C1& c1, const C2& c2, BinaryPredicate&& pred) {
>+  return ((c1.size() == c2.size()) &&
>+          std::equal(container_algorithm_internal::c_begin(c1),
>+                     container_algorithm_internal::c_end(c1),
>+                     container_algorithm_internal::c_begin(c2),
>+                     std::forward<BinaryPredicate>(pred)));
>+}
>+
>+// c_is_permutation()
>+//
>+// Container-based version of the <algorithm> `std::is_permutation()` function
>+// to test whether a container is a permutation of another.
>+template <typename C1, typename C2>
>+bool c_is_permutation(const C1& c1, const C2& c2) {
>+  using std::begin;
>+  using std::end;
>+  return c1.size() == c2.size() &&
>+         std::is_permutation(begin(c1), end(c1), begin(c2));
>+}
>+
>+// Overload of c_is_permutation() for using a predicate evaluation other than
>+// `==` as the function's test condition.
>+template <typename C1, typename C2, typename BinaryPredicate>
>+bool c_is_permutation(const C1& c1, const C2& c2, BinaryPredicate&& pred) {
>+  using std::begin;
>+  using std::end;
>+  return c1.size() == c2.size() &&
>+         std::is_permutation(begin(c1), end(c1), begin(c2),
>+                             std::forward<BinaryPredicate>(pred));
>+}
>+
>+// c_search()
>+//
>+// Container-based version of the <algorithm> `std::search()` function to search
>+// a container for a subsequence.
>+template <typename Sequence1, typename Sequence2>
>+container_algorithm_internal::ContainerIter<Sequence1> c_search(
>+    Sequence1& sequence, Sequence2& subsequence) {
>+  return std::search(container_algorithm_internal::c_begin(sequence),
>+                     container_algorithm_internal::c_end(sequence),
>+                     container_algorithm_internal::c_begin(subsequence),
>+                     container_algorithm_internal::c_end(subsequence));
>+}
>+
>+// Overload of c_search() for using a predicate evaluation other than
>+// `==` as the function's test condition.
>+template <typename Sequence1, typename Sequence2, typename BinaryPredicate>
>+container_algorithm_internal::ContainerIter<Sequence1> c_search(
>+    Sequence1& sequence, Sequence2& subsequence, BinaryPredicate&& pred) {
>+  return std::search(container_algorithm_internal::c_begin(sequence),
>+                     container_algorithm_internal::c_end(sequence),
>+                     container_algorithm_internal::c_begin(subsequence),
>+                     container_algorithm_internal::c_end(subsequence),
>+                     std::forward<BinaryPredicate>(pred));
>+}
>+
>+// c_search_n()
>+//
>+// Container-based version of the <algorithm> `std::search_n()` function to
>+// search a container for the first sequence of N elements.
>+template <typename Sequence, typename Size, typename T>
>+container_algorithm_internal::ContainerIter<Sequence> c_search_n(
>+    Sequence& sequence, Size count, T&& value) {
>+  return std::search_n(container_algorithm_internal::c_begin(sequence),
>+                       container_algorithm_internal::c_end(sequence), count,
>+                       std::forward<T>(value));
>+}
>+
>+// Overload of c_search_n() for using a predicate evaluation other than
>+// `==` as the function's test condition.
>+template <typename Sequence, typename Size, typename T,
>+          typename BinaryPredicate>
>+container_algorithm_internal::ContainerIter<Sequence> c_search_n(
>+    Sequence& sequence, Size count, T&& value, BinaryPredicate&& pred) {
>+  return std::search_n(container_algorithm_internal::c_begin(sequence),
>+                       container_algorithm_internal::c_end(sequence), count,
>+                       std::forward<T>(value),
>+                       std::forward<BinaryPredicate>(pred));
>+}
>+
>+//------------------------------------------------------------------------------
>+// <algorithm> Modifying sequence operations
>+//------------------------------------------------------------------------------
>+
>+// c_copy()
>+//
>+// Container-based version of the <algorithm> `std::copy()` function to copy a
>+// container's elements into an iterator.
>+template <typename InputSequence, typename OutputIterator>
>+OutputIterator c_copy(const InputSequence& input, OutputIterator output) {
>+  return std::copy(container_algorithm_internal::c_begin(input),
>+                   container_algorithm_internal::c_end(input), output);
>+}
>+
>+// c_copy_n()
>+//
>+// Container-based version of the <algorithm> `std::copy_n()` function to copy a
>+// container's first N elements into an iterator.
>+template <typename C, typename Size, typename OutputIterator>
>+OutputIterator c_copy_n(const C& input, Size n, OutputIterator output) {
>+  return std::copy_n(container_algorithm_internal::c_begin(input), n, output);
>+}
>+
>+// c_copy_if()
>+//
>+// Container-based version of the <algorithm> `std::copy_if()` function to copy
>+// a container's elements satisfying some condition into an iterator.
>+template <typename InputSequence, typename OutputIterator, typename Pred>
>+OutputIterator c_copy_if(const InputSequence& input, OutputIterator output,
>+                         Pred&& pred) {
>+  return std::copy_if(container_algorithm_internal::c_begin(input),
>+                      container_algorithm_internal::c_end(input), output,
>+                      std::forward<Pred>(pred));
>+}
>+
>+// c_copy_backward()
>+//
>+// Container-based version of the <algorithm> `std::copy_backward()` function to
>+// copy a container's elements in reverse order into an iterator.
>+template <typename C, typename BidirectionalIterator>
>+BidirectionalIterator c_copy_backward(const C& src,
>+                                      BidirectionalIterator dest) {
>+  return std::copy_backward(container_algorithm_internal::c_begin(src),
>+                            container_algorithm_internal::c_end(src), dest);
>+}
>+
>+// c_move()
>+//
>+// Container-based version of the <algorithm> `std::move()` function to move
>+// a container's elements into an iterator.
>+template <typename C, typename OutputIterator>
>+OutputIterator c_move(C& src, OutputIterator dest) {
>+  return std::move(container_algorithm_internal::c_begin(src),
>+                   container_algorithm_internal::c_end(src), dest);
>+}
>+
>+// c_swap_ranges()
>+//
>+// Container-based version of the <algorithm> `std::swap_ranges()` function to
>+// swap a container's elements with another container's elements.
>+template <typename C1, typename C2>
>+container_algorithm_internal::ContainerIter<C2> c_swap_ranges(C1& c1, C2& c2) {
>+  return std::swap_ranges(container_algorithm_internal::c_begin(c1),
>+                          container_algorithm_internal::c_end(c1),
>+                          container_algorithm_internal::c_begin(c2));
>+}
>+
>+// c_transform()
>+//
>+// Container-based version of the <algorithm> `std::transform()` function to
>+// transform a container's elements using the unary operation, storing the
>+// result in an iterator pointing to the last transformed element in the output
>+// range.
>+template <typename InputSequence, typename OutputIterator, typename UnaryOp>
>+OutputIterator c_transform(const InputSequence& input, OutputIterator output,
>+                           UnaryOp&& unary_op) {
>+  return std::transform(container_algorithm_internal::c_begin(input),
>+                        container_algorithm_internal::c_end(input), output,
>+                        std::forward<UnaryOp>(unary_op));
>+}
>+
>+// Overload of c_transform() for performing a transformation using a binary
>+// predicate.
>+template <typename InputSequence1, typename InputSequence2,
>+          typename OutputIterator, typename BinaryOp>
>+OutputIterator c_transform(const InputSequence1& input1,
>+                           const InputSequence2& input2, OutputIterator output,
>+                           BinaryOp&& binary_op) {
>+  return std::transform(container_algorithm_internal::c_begin(input1),
>+                        container_algorithm_internal::c_end(input1),
>+                        container_algorithm_internal::c_begin(input2), output,
>+                        std::forward<BinaryOp>(binary_op));
>+}
>+
>+// c_replace()
>+//
>+// Container-based version of the <algorithm> `std::replace()` function to
>+// replace a container's elements of some value with a new value. The container
>+// is modified in place.
>+template <typename Sequence, typename T>
>+void c_replace(Sequence& sequence, const T& old_value, const T& new_value) {
>+  std::replace(container_algorithm_internal::c_begin(sequence),
>+               container_algorithm_internal::c_end(sequence), old_value,
>+               new_value);
>+}
>+
>+// c_replace_if()
>+//
>+// Container-based version of the <algorithm> `std::replace_if()` function to
>+// replace a container's elements of some value with a new value based on some
>+// condition. The container is modified in place.
>+template <typename C, typename Pred, typename T>
>+void c_replace_if(C& c, Pred&& pred, T&& new_value) {
>+  std::replace_if(container_algorithm_internal::c_begin(c),
>+                  container_algorithm_internal::c_end(c),
>+                  std::forward<Pred>(pred), std::forward<T>(new_value));
>+}
>+
>+// c_replace_copy()
>+//
>+// Container-based version of the <algorithm> `std::replace_copy()` function to
>+// replace a container's elements of some value with a new value  and return the
>+// results within an iterator.
>+template <typename C, typename OutputIterator, typename T>
>+OutputIterator c_replace_copy(const C& c, OutputIterator result, T&& old_value,
>+                              T&& new_value) {
>+  return std::replace_copy(container_algorithm_internal::c_begin(c),
>+                           container_algorithm_internal::c_end(c), result,
>+                           std::forward<T>(old_value),
>+                           std::forward<T>(new_value));
>+}
>+
>+// c_replace_copy_if()
>+//
>+// Container-based version of the <algorithm> `std::replace_copy_if()` function
>+// to replace a container's elements of some value with a new value based on
>+// some condition, and return the results within an iterator.
>+template <typename C, typename OutputIterator, typename Pred, typename T>
>+OutputIterator c_replace_copy_if(const C& c, OutputIterator result, Pred&& pred,
>+                                 T&& new_value) {
>+  return std::replace_copy_if(container_algorithm_internal::c_begin(c),
>+                              container_algorithm_internal::c_end(c), result,
>+                              std::forward<Pred>(pred),
>+                              std::forward<T>(new_value));
>+}
>+
>+// c_fill()
>+//
>+// Container-based version of the <algorithm> `std::fill()` function to fill a
>+// container with some value.
>+template <typename C, typename T>
>+void c_fill(C& c, T&& value) {
>+  std::fill(container_algorithm_internal::c_begin(c),
>+            container_algorithm_internal::c_end(c), std::forward<T>(value));
>+}
>+
>+// c_fill_n()
>+//
>+// Container-based version of the <algorithm> `std::fill_n()` function to fill
>+// the first N elements in a container with some value.
>+template <typename C, typename Size, typename T>
>+void c_fill_n(C& c, Size n, T&& value) {
>+  std::fill_n(container_algorithm_internal::c_begin(c), n,
>+              std::forward<T>(value));
>+}
>+
>+// c_generate()
>+//
>+// Container-based version of the <algorithm> `std::generate()` function to
>+// assign a container's elements to the values provided by the given generator.
>+template <typename C, typename Generator>
>+void c_generate(C& c, Generator&& gen) {
>+  std::generate(container_algorithm_internal::c_begin(c),
>+                container_algorithm_internal::c_end(c),
>+                std::forward<Generator>(gen));
>+}
>+
>+// c_generate_n()
>+//
>+// Container-based version of the <algorithm> `std::generate_n()` function to
>+// assign a container's first N elements to the values provided by the given
>+// generator.
>+template <typename C, typename Size, typename Generator>
>+container_algorithm_internal::ContainerIter<C> c_generate_n(C& c, Size n,
>+                                                            Generator&& gen) {
>+  return std::generate_n(container_algorithm_internal::c_begin(c), n,
>+                         std::forward<Generator>(gen));
>+}
>+
>+// Note: `c_xx()` <algorithm> container versions for `remove()`, `remove_if()`,
>+// and `unique()` are omitted, because it's not clear whether or not such
>+// functions should call erase on their supplied sequences afterwards. Either
>+// behavior would be surprising for a different set of users.
>+//
>+
>+// c_remove_copy()
>+//
>+// Container-based version of the <algorithm> `std::remove_copy()` function to
>+// copy a container's elements while removing any elements matching the given
>+// `value`.
>+template <typename C, typename OutputIterator, typename T>
>+OutputIterator c_remove_copy(const C& c, OutputIterator result, T&& value) {
>+  return std::remove_copy(container_algorithm_internal::c_begin(c),
>+                          container_algorithm_internal::c_end(c), result,
>+                          std::forward<T>(value));
>+}
>+
>+// c_remove_copy_if()
>+//
>+// Container-based version of the <algorithm> `std::remove_copy_if()` function
>+// to copy a container's elements while removing any elements matching the given
>+// condition.
>+template <typename C, typename OutputIterator, typename Pred>
>+OutputIterator c_remove_copy_if(const C& c, OutputIterator result,
>+                                Pred&& pred) {
>+  return std::remove_copy_if(container_algorithm_internal::c_begin(c),
>+                             container_algorithm_internal::c_end(c), result,
>+                             std::forward<Pred>(pred));
>+}
>+
>+// c_unique_copy()
>+//
>+// Container-based version of the <algorithm> `std::unique_copy()` function to
>+// copy a container's elements while removing any elements containing duplicate
>+// values.
>+template <typename C, typename OutputIterator>
>+OutputIterator c_unique_copy(const C& c, OutputIterator result) {
>+  return std::unique_copy(container_algorithm_internal::c_begin(c),
>+                          container_algorithm_internal::c_end(c), result);
>+}
>+
>+// Overload of c_unique_copy() for using a predicate evaluation other than
>+// `==` for comparing uniqueness of the element values.
>+template <typename C, typename OutputIterator, typename BinaryPredicate>
>+OutputIterator c_unique_copy(const C& c, OutputIterator result,
>+                             BinaryPredicate&& pred) {
>+  return std::unique_copy(container_algorithm_internal::c_begin(c),
>+                          container_algorithm_internal::c_end(c), result,
>+                          std::forward<BinaryPredicate>(pred));
>+}
>+
>+// c_reverse()
>+//
>+// Container-based version of the <algorithm> `std::reverse()` function to
>+// reverse a container's elements.
>+template <typename Sequence>
>+void c_reverse(Sequence& sequence) {
>+  std::reverse(container_algorithm_internal::c_begin(sequence),
>+               container_algorithm_internal::c_end(sequence));
>+}
>+
>+// c_reverse_copy()
>+//
>+// Container-based version of the <algorithm> `std::reverse()` function to
>+// reverse a container's elements and write them to an iterator range.
>+template <typename C, typename OutputIterator>
>+OutputIterator c_reverse_copy(const C& sequence, OutputIterator result) {
>+  return std::reverse_copy(container_algorithm_internal::c_begin(sequence),
>+                           container_algorithm_internal::c_end(sequence),
>+                           result);
>+}
>+
>+// c_rotate()
>+//
>+// Container-based version of the <algorithm> `std::rotate()` function to
>+// shift a container's elements leftward such that the `middle` element becomes
>+// the first element in the container.
>+template <typename C,
>+          typename Iterator = container_algorithm_internal::ContainerIter<C>>
>+Iterator c_rotate(C& sequence, Iterator middle) {
>+  return absl::rotate(container_algorithm_internal::c_begin(sequence), middle,
>+                      container_algorithm_internal::c_end(sequence));
>+}
>+
>+// c_rotate_copy()
>+//
>+// Container-based version of the <algorithm> `std::rotate_copy()` function to
>+// shift a container's elements leftward such that the `middle` element becomes
>+// the first element in a new iterator range.
>+template <typename C, typename OutputIterator>
>+OutputIterator c_rotate_copy(
>+    const C& sequence,
>+    container_algorithm_internal::ContainerIter<const C> middle,
>+    OutputIterator result) {
>+  return std::rotate_copy(container_algorithm_internal::c_begin(sequence),
>+                          middle, container_algorithm_internal::c_end(sequence),
>+                          result);
>+}
>+
>+// c_shuffle()
>+//
>+// Container-based version of the <algorithm> `std::shuffle()` function to
>+// randomly shuffle elements within the container using a `gen()` uniform random
>+// number generator.
>+template <typename RandomAccessContainer, typename UniformRandomBitGenerator>
>+void c_shuffle(RandomAccessContainer& c, UniformRandomBitGenerator&& gen) {
>+  std::shuffle(container_algorithm_internal::c_begin(c),
>+               container_algorithm_internal::c_end(c),
>+               std::forward<UniformRandomBitGenerator>(gen));
>+}
>+
>+//------------------------------------------------------------------------------
>+// <algorithm> Partition functions
>+//------------------------------------------------------------------------------
>+
>+// c_is_partitioned()
>+//
>+// Container-based version of the <algorithm> `std::is_partitioned()` function
>+// to test whether all elements in the container for which `pred` returns `true`
>+// precede those for which `pred` is `false`.
>+template <typename C, typename Pred>
>+bool c_is_partitioned(const C& c, Pred&& pred) {
>+  return std::is_partitioned(container_algorithm_internal::c_begin(c),
>+                             container_algorithm_internal::c_end(c),
>+                             std::forward<Pred>(pred));
>+}
>+
>+// c_partition()
>+//
>+// Container-based version of the <algorithm> `std::partition()` function
>+// to rearrange all elements in a container in such a way that all elements for
>+// which `pred` returns `true` precede all those for which it returns `false`,
>+// returning an iterator to the first element of the second group.
>+template <typename C, typename Pred>
>+container_algorithm_internal::ContainerIter<C> c_partition(C& c, Pred&& pred) {
>+  return std::partition(container_algorithm_internal::c_begin(c),
>+                        container_algorithm_internal::c_end(c),
>+                        std::forward<Pred>(pred));
>+}
>+
>+// c_stable_partition()
>+//
>+// Container-based version of the <algorithm> `std::stable_partition()` function
>+// to rearrange all elements in a container in such a way that all elements for
>+// which `pred` returns `true` precede all those for which it returns `false`,
>+// preserving the relative ordering between the two groups. The function returns
>+// an iterator to the first element of the second group.
>+template <typename C, typename Pred>
>+container_algorithm_internal::ContainerIter<C> c_stable_partition(C& c,
>+                                                                  Pred&& pred) {
>+  return std::stable_partition(container_algorithm_internal::c_begin(c),
>+                               container_algorithm_internal::c_end(c),
>+                               std::forward<Pred>(pred));
>+}
>+
>+// c_partition_copy()
>+//
>+// Container-based version of the <algorithm> `std::partition_copy()` function
>+// to partition a container's elements and return them into two iterators: one
>+// for which `pred` returns `true`, and one for which `pred` returns `false.`
>+
>+template <typename C, typename OutputIterator1, typename OutputIterator2,
>+          typename Pred>
>+std::pair<OutputIterator1, OutputIterator2> c_partition_copy(
>+    const C& c, OutputIterator1 out_true, OutputIterator2 out_false,
>+    Pred&& pred) {
>+  return std::partition_copy(container_algorithm_internal::c_begin(c),
>+                             container_algorithm_internal::c_end(c), out_true,
>+                             out_false, std::forward<Pred>(pred));
>+}
>+
>+// c_partition_point()
>+//
>+// Container-based version of the <algorithm> `std::partition_point()` function
>+// to return the first element of an already partitioned container for which
>+// the given `pred` is not `true`.
>+template <typename C, typename Pred>
>+container_algorithm_internal::ContainerIter<C> c_partition_point(C& c,
>+                                                                 Pred&& pred) {
>+  return std::partition_point(container_algorithm_internal::c_begin(c),
>+                              container_algorithm_internal::c_end(c),
>+                              std::forward<Pred>(pred));
>+}
>+
>+//------------------------------------------------------------------------------
>+// <algorithm> Sorting functions
>+//------------------------------------------------------------------------------
>+
>+// c_sort()
>+//
>+// Container-based version of the <algorithm> `std::sort()` function
>+// to sort elements in ascending order of their values.
>+template <typename C>
>+void c_sort(C& c) {
>+  std::sort(container_algorithm_internal::c_begin(c),
>+            container_algorithm_internal::c_end(c));
>+}
>+
>+// Overload of c_sort() for performing a `comp` comparison other than the
>+// default `operator<`.
>+template <typename C, typename Compare>
>+void c_sort(C& c, Compare&& comp) {
>+  std::sort(container_algorithm_internal::c_begin(c),
>+            container_algorithm_internal::c_end(c),
>+            std::forward<Compare>(comp));
>+}
>+
>+// c_stable_sort()
>+//
>+// Container-based version of the <algorithm> `std::stable_sort()` function
>+// to sort elements in ascending order of their values, preserving the order
>+// of equivalents.
>+template <typename C>
>+void c_stable_sort(C& c) {
>+  std::stable_sort(container_algorithm_internal::c_begin(c),
>+                   container_algorithm_internal::c_end(c));
>+}
>+
>+// Overload of c_stable_sort() for performing a `comp` comparison other than the
>+// default `operator<`.
>+template <typename C, typename Compare>
>+void c_stable_sort(C& c, Compare&& comp) {
>+  std::stable_sort(container_algorithm_internal::c_begin(c),
>+                   container_algorithm_internal::c_end(c),
>+                   std::forward<Compare>(comp));
>+}
>+
>+// c_is_sorted()
>+//
>+// Container-based version of the <algorithm> `std::is_sorted()` function
>+// to evaluate whether the given container is sorted in ascending order.
>+template <typename C>
>+bool c_is_sorted(const C& c) {
>+  return std::is_sorted(container_algorithm_internal::c_begin(c),
>+                        container_algorithm_internal::c_end(c));
>+}
>+
>+// c_is_sorted() overload for performing a `comp` comparison other than the
>+// default `operator<`.
>+template <typename C, typename Compare>
>+bool c_is_sorted(const C& c, Compare&& comp) {
>+  return std::is_sorted(container_algorithm_internal::c_begin(c),
>+                        container_algorithm_internal::c_end(c),
>+                        std::forward<Compare>(comp));
>+}
>+
>+// c_partial_sort()
>+//
>+// Container-based version of the <algorithm> `std::partial_sort()` function
>+// to rearrange elements within a container such that elements before `middle`
>+// are sorted in ascending order.
>+template <typename RandomAccessContainer>
>+void c_partial_sort(
>+    RandomAccessContainer& sequence,
>+    container_algorithm_internal::ContainerIter<RandomAccessContainer> middle) {
>+  std::partial_sort(container_algorithm_internal::c_begin(sequence), middle,
>+                    container_algorithm_internal::c_end(sequence));
>+}
>+
>+// Overload of c_partial_sort() for performing a `comp` comparison other than
>+// the default `operator<`.
>+template <typename RandomAccessContainer, typename Compare>
>+void c_partial_sort(
>+    RandomAccessContainer& sequence,
>+    container_algorithm_internal::ContainerIter<RandomAccessContainer> middle,
>+    Compare&& comp) {
>+  std::partial_sort(container_algorithm_internal::c_begin(sequence), middle,
>+                    container_algorithm_internal::c_end(sequence),
>+                    std::forward<Compare>(comp));
>+}
>+
>+// c_partial_sort_copy()
>+//
>+// Container-based version of the <algorithm> `std::partial_sort_copy()`
>+// function to sort elements within a container such that elements before
>+// `middle` are sorted in ascending order, and return the result within an
>+// iterator.
>+template <typename C, typename RandomAccessContainer>
>+container_algorithm_internal::ContainerIter<RandomAccessContainer>
>+c_partial_sort_copy(const C& sequence, RandomAccessContainer& result) {
>+  return std::partial_sort_copy(container_algorithm_internal::c_begin(sequence),
>+                                container_algorithm_internal::c_end(sequence),
>+                                container_algorithm_internal::c_begin(result),
>+                                container_algorithm_internal::c_end(result));
>+}
>+
>+// Overload of c_partial_sort_copy() for performing a `comp` comparison other
>+// than the default `operator<`.
>+template <typename C, typename RandomAccessContainer, typename Compare>
>+container_algorithm_internal::ContainerIter<RandomAccessContainer>
>+c_partial_sort_copy(const C& sequence, RandomAccessContainer& result,
>+                    Compare&& comp) {
>+  return std::partial_sort_copy(container_algorithm_internal::c_begin(sequence),
>+                                container_algorithm_internal::c_end(sequence),
>+                                container_algorithm_internal::c_begin(result),
>+                                container_algorithm_internal::c_end(result),
>+                                std::forward<Compare>(comp));
>+}
>+
>+// c_is_sorted_until()
>+//
>+// Container-based version of the <algorithm> `std::is_sorted_until()` function
>+// to return the first element within a container that is not sorted in
>+// ascending order as an iterator.
>+template <typename C>
>+container_algorithm_internal::ContainerIter<C> c_is_sorted_until(C& c) {
>+  return std::is_sorted_until(container_algorithm_internal::c_begin(c),
>+                              container_algorithm_internal::c_end(c));
>+}
>+
>+// Overload of c_is_sorted_until() for performing a `comp` comparison other than
>+// the default `operator<`.
>+template <typename C, typename Compare>
>+container_algorithm_internal::ContainerIter<C> c_is_sorted_until(
>+    C& c, Compare&& comp) {
>+  return std::is_sorted_until(container_algorithm_internal::c_begin(c),
>+                              container_algorithm_internal::c_end(c),
>+                              std::forward<Compare>(comp));
>+}
>+
>+// c_nth_element()
>+//
>+// Container-based version of the <algorithm> `std::nth_element()` function
>+// to rearrange the elements within a container such that the `nth` element
>+// would be in that position in an ordered sequence; other elements may be in
>+// any order, except that all preceding `nth` will be less than that element,
>+// and all following `nth` will be greater than that element.
>+template <typename RandomAccessContainer>
>+void c_nth_element(
>+    RandomAccessContainer& sequence,
>+    container_algorithm_internal::ContainerIter<RandomAccessContainer> nth) {
>+  std::nth_element(container_algorithm_internal::c_begin(sequence), nth,
>+                   container_algorithm_internal::c_end(sequence));
>+}
>+
>+// Overload of c_nth_element() for performing a `comp` comparison other than
>+// the default `operator<`.
>+template <typename RandomAccessContainer, typename Compare>
>+void c_nth_element(
>+    RandomAccessContainer& sequence,
>+    container_algorithm_internal::ContainerIter<RandomAccessContainer> nth,
>+    Compare&& comp) {
>+  std::nth_element(container_algorithm_internal::c_begin(sequence), nth,
>+                   container_algorithm_internal::c_end(sequence),
>+                   std::forward<Compare>(comp));
>+}
>+
>+//------------------------------------------------------------------------------
>+// <algorithm> Binary Search
>+//------------------------------------------------------------------------------
>+
>+// c_lower_bound()
>+//
>+// Container-based version of the <algorithm> `std::lower_bound()` function
>+// to return an iterator pointing to the first element in a sorted container
>+// which does not compare less than `value`.
>+template <typename Sequence, typename T>
>+container_algorithm_internal::ContainerIter<Sequence> c_lower_bound(
>+    Sequence& sequence, T&& value) {
>+  return std::lower_bound(container_algorithm_internal::c_begin(sequence),
>+                          container_algorithm_internal::c_end(sequence),
>+                          std::forward<T>(value));
>+}
>+
>+// Overload of c_lower_bound() for performing a `comp` comparison other than
>+// the default `operator<`.
>+template <typename Sequence, typename T, typename Compare>
>+container_algorithm_internal::ContainerIter<Sequence> c_lower_bound(
>+    Sequence& sequence, T&& value, Compare&& comp) {
>+  return std::lower_bound(container_algorithm_internal::c_begin(sequence),
>+                          container_algorithm_internal::c_end(sequence),
>+                          std::forward<T>(value), std::forward<Compare>(comp));
>+}
>+
>+// c_upper_bound()
>+//
>+// Container-based version of the <algorithm> `std::upper_bound()` function
>+// to return an iterator pointing to the first element in a sorted container
>+// which is greater than `value`.
>+template <typename Sequence, typename T>
>+container_algorithm_internal::ContainerIter<Sequence> c_upper_bound(
>+    Sequence& sequence, T&& value) {
>+  return std::upper_bound(container_algorithm_internal::c_begin(sequence),
>+                          container_algorithm_internal::c_end(sequence),
>+                          std::forward<T>(value));
>+}
>+
>+// Overload of c_upper_bound() for performing a `comp` comparison other than
>+// the default `operator<`.
>+template <typename Sequence, typename T, typename Compare>
>+container_algorithm_internal::ContainerIter<Sequence> c_upper_bound(
>+    Sequence& sequence, T&& value, Compare&& comp) {
>+  return std::upper_bound(container_algorithm_internal::c_begin(sequence),
>+                          container_algorithm_internal::c_end(sequence),
>+                          std::forward<T>(value), std::forward<Compare>(comp));
>+}
>+
>+// c_equal_range()
>+//
>+// Container-based version of the <algorithm> `std::equal_range()` function
>+// to return an iterator pair pointing to the first and last elements in a
>+// sorted container which compare equal to `value`.
>+template <typename Sequence, typename T>
>+container_algorithm_internal::ContainerIterPairType<Sequence, Sequence>
>+c_equal_range(Sequence& sequence, T&& value) {
>+  return std::equal_range(container_algorithm_internal::c_begin(sequence),
>+                          container_algorithm_internal::c_end(sequence),
>+                          std::forward<T>(value));
>+}
>+
>+// Overload of c_equal_range() for performing a `comp` comparison other than
>+// the default `operator<`.
>+template <typename Sequence, typename T, typename Compare>
>+container_algorithm_internal::ContainerIterPairType<Sequence, Sequence>
>+c_equal_range(Sequence& sequence, T&& value, Compare&& comp) {
>+  return std::equal_range(container_algorithm_internal::c_begin(sequence),
>+                          container_algorithm_internal::c_end(sequence),
>+                          std::forward<T>(value), std::forward<Compare>(comp));
>+}
>+
>+// c_binary_search()
>+//
>+// Container-based version of the <algorithm> `std::binary_search()` function
>+// to test if any element in the sorted container contains a value equivalent to
>+// 'value'.
>+template <typename Sequence, typename T>
>+bool c_binary_search(Sequence&& sequence, T&& value) {
>+  return std::binary_search(container_algorithm_internal::c_begin(sequence),
>+                            container_algorithm_internal::c_end(sequence),
>+                            std::forward<T>(value));
>+}
>+
>+// Overload of c_binary_search() for performing a `comp` comparison other than
>+// the default `operator<`.
>+template <typename Sequence, typename T, typename Compare>
>+bool c_binary_search(Sequence&& sequence, T&& value, Compare&& comp) {
>+  return std::binary_search(container_algorithm_internal::c_begin(sequence),
>+                            container_algorithm_internal::c_end(sequence),
>+                            std::forward<T>(value),
>+                            std::forward<Compare>(comp));
>+}
>+
>+//------------------------------------------------------------------------------
>+// <algorithm> Merge functions
>+//------------------------------------------------------------------------------
>+
>+// c_merge()
>+//
>+// Container-based version of the <algorithm> `std::merge()` function
>+// to merge two sorted containers into a single sorted iterator.
>+template <typename C1, typename C2, typename OutputIterator>
>+OutputIterator c_merge(const C1& c1, const C2& c2, OutputIterator result) {
>+  return std::merge(container_algorithm_internal::c_begin(c1),
>+                    container_algorithm_internal::c_end(c1),
>+                    container_algorithm_internal::c_begin(c2),
>+                    container_algorithm_internal::c_end(c2), result);
>+}
>+
>+// Overload of c_merge() for performing a `comp` comparison other than
>+// the default `operator<`.
>+template <typename C1, typename C2, typename OutputIterator, typename Compare>
>+OutputIterator c_merge(const C1& c1, const C2& c2, OutputIterator result,
>+                       Compare&& comp) {
>+  return std::merge(container_algorithm_internal::c_begin(c1),
>+                    container_algorithm_internal::c_end(c1),
>+                    container_algorithm_internal::c_begin(c2),
>+                    container_algorithm_internal::c_end(c2), result,
>+                    std::forward<Compare>(comp));
>+}
>+
>+// c_inplace_merge()
>+//
>+// Container-based version of the <algorithm> `std::inplace_merge()` function
>+// to merge a supplied iterator `middle` into a container.
>+template <typename C>
>+void c_inplace_merge(C& c,
>+                     container_algorithm_internal::ContainerIter<C> middle) {
>+  std::inplace_merge(container_algorithm_internal::c_begin(c), middle,
>+                     container_algorithm_internal::c_end(c));
>+}
>+
>+// Overload of c_inplace_merge() for performing a merge using a `comp` other
>+// than `operator<`.
>+template <typename C, typename Compare>
>+void c_inplace_merge(C& c,
>+                     container_algorithm_internal::ContainerIter<C> middle,
>+                     Compare&& comp) {
>+  std::inplace_merge(container_algorithm_internal::c_begin(c), middle,
>+                     container_algorithm_internal::c_end(c),
>+                     std::forward<Compare>(comp));
>+}
>+
>+// c_includes()
>+//
>+// Container-based version of the <algorithm> `std::includes()` function
>+// to test whether a sorted container `c1` entirely contains another sorted
>+// container `c2`.
>+template <typename C1, typename C2>
>+bool c_includes(const C1& c1, const C2& c2) {
>+  return std::includes(container_algorithm_internal::c_begin(c1),
>+                       container_algorithm_internal::c_end(c1),
>+                       container_algorithm_internal::c_begin(c2),
>+                       container_algorithm_internal::c_end(c2));
>+}
>+
>+// Overload of c_includes() for performing a merge using a `comp` other than
>+// `operator<`.
>+template <typename C1, typename C2, typename Compare>
>+bool c_includes(const C1& c1, const C2& c2, Compare&& comp) {
>+  return std::includes(container_algorithm_internal::c_begin(c1),
>+                       container_algorithm_internal::c_end(c1),
>+                       container_algorithm_internal::c_begin(c2),
>+                       container_algorithm_internal::c_end(c2),
>+                       std::forward<Compare>(comp));
>+}
>+
>+// c_set_union()
>+//
>+// Container-based version of the <algorithm> `std::set_union()` function
>+// to return an iterator containing the union of two containers; duplicate
>+// values are not copied into the output.
>+template <typename C1, typename C2, typename OutputIterator>
>+OutputIterator c_set_union(const C1& c1, const C2& c2, OutputIterator output) {
>+  return std::set_union(container_algorithm_internal::c_begin(c1),
>+                        container_algorithm_internal::c_end(c1),
>+                        container_algorithm_internal::c_begin(c2),
>+                        container_algorithm_internal::c_end(c2), output);
>+}
>+
>+// Overload of c_set_union() for performing a merge using a `comp` other than
>+// `operator<`.
>+template <typename C1, typename C2, typename OutputIterator, typename Compare>
>+OutputIterator c_set_union(const C1& c1, const C2& c2, OutputIterator output,
>+                           Compare&& comp) {
>+  return std::set_union(container_algorithm_internal::c_begin(c1),
>+                        container_algorithm_internal::c_end(c1),
>+                        container_algorithm_internal::c_begin(c2),
>+                        container_algorithm_internal::c_end(c2), output,
>+                        std::forward<Compare>(comp));
>+}
>+
>+// c_set_intersection()
>+//
>+// Container-based version of the <algorithm> `std::set_intersection()` function
>+// to return an iterator containing the intersection of two containers.
>+template <typename C1, typename C2, typename OutputIterator>
>+OutputIterator c_set_intersection(const C1& c1, const C2& c2,
>+                                  OutputIterator output) {
>+  return std::set_intersection(container_algorithm_internal::c_begin(c1),
>+                               container_algorithm_internal::c_end(c1),
>+                               container_algorithm_internal::c_begin(c2),
>+                               container_algorithm_internal::c_end(c2), output);
>+}
>+
>+// Overload of c_set_intersection() for performing a merge using a `comp` other
>+// than `operator<`.
>+template <typename C1, typename C2, typename OutputIterator, typename Compare>
>+OutputIterator c_set_intersection(const C1& c1, const C2& c2,
>+                                  OutputIterator output, Compare&& comp) {
>+  return std::set_intersection(container_algorithm_internal::c_begin(c1),
>+                               container_algorithm_internal::c_end(c1),
>+                               container_algorithm_internal::c_begin(c2),
>+                               container_algorithm_internal::c_end(c2), output,
>+                               std::forward<Compare>(comp));
>+}
>+
>+// c_set_difference()
>+//
>+// Container-based version of the <algorithm> `std::set_difference()` function
>+// to return an iterator containing elements present in the first container but
>+// not in the second.
>+template <typename C1, typename C2, typename OutputIterator>
>+OutputIterator c_set_difference(const C1& c1, const C2& c2,
>+                                OutputIterator output) {
>+  return std::set_difference(container_algorithm_internal::c_begin(c1),
>+                             container_algorithm_internal::c_end(c1),
>+                             container_algorithm_internal::c_begin(c2),
>+                             container_algorithm_internal::c_end(c2), output);
>+}
>+
>+// Overload of c_set_difference() for performing a merge using a `comp` other
>+// than `operator<`.
>+template <typename C1, typename C2, typename OutputIterator, typename Compare>
>+OutputIterator c_set_difference(const C1& c1, const C2& c2,
>+                                OutputIterator output, Compare&& comp) {
>+  return std::set_difference(container_algorithm_internal::c_begin(c1),
>+                             container_algorithm_internal::c_end(c1),
>+                             container_algorithm_internal::c_begin(c2),
>+                             container_algorithm_internal::c_end(c2), output,
>+                             std::forward<Compare>(comp));
>+}
>+
>+// c_set_symmetric_difference()
>+//
>+// Container-based version of the <algorithm> `std::set_symmetric_difference()`
>+// function to return an iterator containing elements present in either one
>+// container or the other, but not both.
>+template <typename C1, typename C2, typename OutputIterator>
>+OutputIterator c_set_symmetric_difference(const C1& c1, const C2& c2,
>+                                          OutputIterator output) {
>+  return std::set_symmetric_difference(
>+      container_algorithm_internal::c_begin(c1),
>+      container_algorithm_internal::c_end(c1),
>+      container_algorithm_internal::c_begin(c2),
>+      container_algorithm_internal::c_end(c2), output);
>+}
>+
>+// Overload of c_set_symmetric_difference() for performing a merge using a
>+// `comp` other than `operator<`.
>+template <typename C1, typename C2, typename OutputIterator, typename Compare>
>+OutputIterator c_set_symmetric_difference(const C1& c1, const C2& c2,
>+                                          OutputIterator output,
>+                                          Compare&& comp) {
>+  return std::set_symmetric_difference(
>+      container_algorithm_internal::c_begin(c1),
>+      container_algorithm_internal::c_end(c1),
>+      container_algorithm_internal::c_begin(c2),
>+      container_algorithm_internal::c_end(c2), output,
>+      std::forward<Compare>(comp));
>+}
>+
>+//------------------------------------------------------------------------------
>+// <algorithm> Heap functions
>+//------------------------------------------------------------------------------
>+
>+// c_push_heap()
>+//
>+// Container-based version of the <algorithm> `std::push_heap()` function
>+// to push a value onto a container heap.
>+template <typename RandomAccessContainer>
>+void c_push_heap(RandomAccessContainer& sequence) {
>+  std::push_heap(container_algorithm_internal::c_begin(sequence),
>+                 container_algorithm_internal::c_end(sequence));
>+}
>+
>+// Overload of c_push_heap() for performing a push operation on a heap using a
>+// `comp` other than `operator<`.
>+template <typename RandomAccessContainer, typename Compare>
>+void c_push_heap(RandomAccessContainer& sequence, Compare&& comp) {
>+  std::push_heap(container_algorithm_internal::c_begin(sequence),
>+                 container_algorithm_internal::c_end(sequence),
>+                 std::forward<Compare>(comp));
>+}
>+
>+// c_pop_heap()
>+//
>+// Container-based version of the <algorithm> `std::pop_heap()` function
>+// to pop a value from a heap container.
>+template <typename RandomAccessContainer>
>+void c_pop_heap(RandomAccessContainer& sequence) {
>+  std::pop_heap(container_algorithm_internal::c_begin(sequence),
>+                container_algorithm_internal::c_end(sequence));
>+}
>+
>+// Overload of c_pop_heap() for performing a pop operation on a heap using a
>+// `comp` other than `operator<`.
>+template <typename RandomAccessContainer, typename Compare>
>+void c_pop_heap(RandomAccessContainer& sequence, Compare&& comp) {
>+  std::pop_heap(container_algorithm_internal::c_begin(sequence),
>+                container_algorithm_internal::c_end(sequence),
>+                std::forward<Compare>(comp));
>+}
>+
>+// c_make_heap()
>+//
>+// Container-based version of the <algorithm> `std::make_heap()` function
>+// to make a container a heap.
>+template <typename RandomAccessContainer>
>+void c_make_heap(RandomAccessContainer& sequence) {
>+  std::make_heap(container_algorithm_internal::c_begin(sequence),
>+                 container_algorithm_internal::c_end(sequence));
>+}
>+
>+// Overload of c_make_heap() for performing heap comparisons using a
>+// `comp` other than `operator<`
>+template <typename RandomAccessContainer, typename Compare>
>+void c_make_heap(RandomAccessContainer& sequence, Compare&& comp) {
>+  std::make_heap(container_algorithm_internal::c_begin(sequence),
>+                 container_algorithm_internal::c_end(sequence),
>+                 std::forward<Compare>(comp));
>+}
>+
>+// c_sort_heap()
>+//
>+// Container-based version of the <algorithm> `std::sort_heap()` function
>+// to sort a heap into ascending order (after which it is no longer a heap).
>+template <typename RandomAccessContainer>
>+void c_sort_heap(RandomAccessContainer& sequence) {
>+  std::sort_heap(container_algorithm_internal::c_begin(sequence),
>+                 container_algorithm_internal::c_end(sequence));
>+}
>+
>+// Overload of c_sort_heap() for performing heap comparisons using a
>+// `comp` other than `operator<`
>+template <typename RandomAccessContainer, typename Compare>
>+void c_sort_heap(RandomAccessContainer& sequence, Compare&& comp) {
>+  std::sort_heap(container_algorithm_internal::c_begin(sequence),
>+                 container_algorithm_internal::c_end(sequence),
>+                 std::forward<Compare>(comp));
>+}
>+
>+// c_is_heap()
>+//
>+// Container-based version of the <algorithm> `std::is_heap()` function
>+// to check whether the given container is a heap.
>+template <typename RandomAccessContainer>
>+bool c_is_heap(const RandomAccessContainer& sequence) {
>+  return std::is_heap(container_algorithm_internal::c_begin(sequence),
>+                      container_algorithm_internal::c_end(sequence));
>+}
>+
>+// Overload of c_is_heap() for performing heap comparisons using a
>+// `comp` other than `operator<`
>+template <typename RandomAccessContainer, typename Compare>
>+bool c_is_heap(const RandomAccessContainer& sequence, Compare&& comp) {
>+  return std::is_heap(container_algorithm_internal::c_begin(sequence),
>+                      container_algorithm_internal::c_end(sequence),
>+                      std::forward<Compare>(comp));
>+}
>+
>+// c_is_heap_until()
>+//
>+// Container-based version of the <algorithm> `std::is_heap_until()` function
>+// to find the first element in a given container which is not in heap order.
>+template <typename RandomAccessContainer>
>+container_algorithm_internal::ContainerIter<RandomAccessContainer>
>+c_is_heap_until(RandomAccessContainer& sequence) {
>+  return std::is_heap_until(container_algorithm_internal::c_begin(sequence),
>+                            container_algorithm_internal::c_end(sequence));
>+}
>+
>+// Overload of c_is_heap_until() for performing heap comparisons using a
>+// `comp` other than `operator<`
>+template <typename RandomAccessContainer, typename Compare>
>+container_algorithm_internal::ContainerIter<RandomAccessContainer>
>+c_is_heap_until(RandomAccessContainer& sequence, Compare&& comp) {
>+  return std::is_heap_until(container_algorithm_internal::c_begin(sequence),
>+                            container_algorithm_internal::c_end(sequence),
>+                            std::forward<Compare>(comp));
>+}
>+
>+//------------------------------------------------------------------------------
>+//  <algorithm> Min/max
>+//------------------------------------------------------------------------------
>+
>+// c_min_element()
>+//
>+// Container-based version of the <algorithm> `std::min_element()` function
>+// to return an iterator pointing to the element with the smallest value, using
>+// `operator<` to make the comparisons.
>+template <typename Sequence>
>+container_algorithm_internal::ContainerIter<Sequence> c_min_element(
>+    Sequence& sequence) {
>+  return std::min_element(container_algorithm_internal::c_begin(sequence),
>+                          container_algorithm_internal::c_end(sequence));
>+}
>+
>+// Overload of c_min_element() for performing a `comp` comparison other than
>+// `operator<`.
>+template <typename Sequence, typename Compare>
>+container_algorithm_internal::ContainerIter<Sequence> c_min_element(
>+    Sequence& sequence, Compare&& comp) {
>+  return std::min_element(container_algorithm_internal::c_begin(sequence),
>+                          container_algorithm_internal::c_end(sequence),
>+                          std::forward<Compare>(comp));
>+}
>+
>+// c_max_element()
>+//
>+// Container-based version of the <algorithm> `std::max_element()` function
>+// to return an iterator pointing to the element with the largest value, using
>+// `operator<` to make the comparisons.
>+template <typename Sequence>
>+container_algorithm_internal::ContainerIter<Sequence> c_max_element(
>+    Sequence& sequence) {
>+  return std::max_element(container_algorithm_internal::c_begin(sequence),
>+                          container_algorithm_internal::c_end(sequence));
>+}
>+
>+// Overload of c_max_element() for performing a `comp` comparison other than
>+// `operator<`.
>+template <typename Sequence, typename Compare>
>+container_algorithm_internal::ContainerIter<Sequence> c_max_element(
>+    Sequence& sequence, Compare&& comp) {
>+  return std::max_element(container_algorithm_internal::c_begin(sequence),
>+                          container_algorithm_internal::c_end(sequence),
>+                          std::forward<Compare>(comp));
>+}
>+
>+// c_minmax_element()
>+//
>+// Container-based version of the <algorithm> `std::minmax_element()` function
>+// to return a pair of iterators pointing to the elements containing the
>+// smallest and largest values, respectively, using `operator<` to make the
>+// comparisons.
>+template <typename C>
>+container_algorithm_internal::ContainerIterPairType<C, C>
>+c_minmax_element(C& c) {
>+  return std::minmax_element(container_algorithm_internal::c_begin(c),
>+                             container_algorithm_internal::c_end(c));
>+}
>+
>+// Overload of c_minmax_element() for performing `comp` comparisons other than
>+// `operator<`.
>+template <typename C, typename Compare>
>+container_algorithm_internal::ContainerIterPairType<C, C>
>+c_minmax_element(C& c, Compare&& comp) {
>+  return std::minmax_element(container_algorithm_internal::c_begin(c),
>+                             container_algorithm_internal::c_end(c),
>+                             std::forward<Compare>(comp));
>+}
>+
>+//------------------------------------------------------------------------------
>+//  <algorithm> Lexicographical Comparisons
>+//------------------------------------------------------------------------------
>+
>+// c_lexicographical_compare()
>+//
>+// Container-based version of the <algorithm> `std::lexicographical_compare()`
>+// function to lexicographically compare (e.g. sort words alphabetically) two
>+// container sequences. The comparison is performed using `operator<`. Note
>+// that capital letters ("A-Z") have ASCII values less than lowercase letters
>+// ("a-z").
>+template <typename Sequence1, typename Sequence2>
>+bool c_lexicographical_compare(Sequence1&& sequence1, Sequence2&& sequence2) {
>+  return std::lexicographical_compare(
>+      container_algorithm_internal::c_begin(sequence1),
>+      container_algorithm_internal::c_end(sequence1),
>+      container_algorithm_internal::c_begin(sequence2),
>+      container_algorithm_internal::c_end(sequence2));
>+}
>+
>+// Overload of c_lexicographical_compare() for performing a lexicographical
>+// comparison using a `comp` operator instead of `operator<`.
>+template <typename Sequence1, typename Sequence2, typename Compare>
>+bool c_lexicographical_compare(Sequence1&& sequence1, Sequence2&& sequence2,
>+                               Compare&& comp) {
>+  return std::lexicographical_compare(
>+      container_algorithm_internal::c_begin(sequence1),
>+      container_algorithm_internal::c_end(sequence1),
>+      container_algorithm_internal::c_begin(sequence2),
>+      container_algorithm_internal::c_end(sequence2),
>+      std::forward<Compare>(comp));
>+}
>+
>+// c_next_permutation()
>+//
>+// Container-based version of the <algorithm> `std::next_permutation()` function
>+// to rearrange a container's elements into the next lexicographically greater
>+// permutation.
>+template <typename C>
>+bool c_next_permutation(C& c) {
>+  return std::next_permutation(container_algorithm_internal::c_begin(c),
>+                               container_algorithm_internal::c_end(c));
>+}
>+
>+// Overload of c_next_permutation() for performing a lexicographical
>+// comparison using a `comp` operator instead of `operator<`.
>+template <typename C, typename Compare>
>+bool c_next_permutation(C& c, Compare&& comp) {
>+  return std::next_permutation(container_algorithm_internal::c_begin(c),
>+                               container_algorithm_internal::c_end(c),
>+                               std::forward<Compare>(comp));
>+}
>+
>+// c_prev_permutation()
>+//
>+// Container-based version of the <algorithm> `std::prev_permutation()` function
>+// to rearrange a container's elements into the next lexicographically lesser
>+// permutation.
>+template <typename C>
>+bool c_prev_permutation(C& c) {
>+  return std::prev_permutation(container_algorithm_internal::c_begin(c),
>+                               container_algorithm_internal::c_end(c));
>+}
>+
>+// Overload of c_prev_permutation() for performing a lexicographical
>+// comparison using a `comp` operator instead of `operator<`.
>+template <typename C, typename Compare>
>+bool c_prev_permutation(C& c, Compare&& comp) {
>+  return std::prev_permutation(container_algorithm_internal::c_begin(c),
>+                               container_algorithm_internal::c_end(c),
>+                               std::forward<Compare>(comp));
>+}
>+
>+//------------------------------------------------------------------------------
>+// <numeric> algorithms
>+//------------------------------------------------------------------------------
>+
>+// c_iota()
>+//
>+// Container-based version of the <algorithm> `std::iota()` function
>+// to compute successive values of `value`, as if incremented with `++value`
>+// after each element is written. and write them to the container.
>+template <typename Sequence, typename T>
>+void c_iota(Sequence& sequence, T&& value) {
>+  std::iota(container_algorithm_internal::c_begin(sequence),
>+            container_algorithm_internal::c_end(sequence),
>+            std::forward<T>(value));
>+}
>+// c_accumulate()
>+//
>+// Container-based version of the <algorithm> `std::accumulate()` function
>+// to accumulate the element values of a container to `init` and return that
>+// accumulation by value.
>+//
>+// Note: Due to a language technicality this function has return type
>+// absl::decay_t<T>. As a user of this function you can casually read
>+// this as "returns T by value" and assume it does the right thing.
>+template <typename Sequence, typename T>
>+decay_t<T> c_accumulate(const Sequence& sequence, T&& init) {
>+  return std::accumulate(container_algorithm_internal::c_begin(sequence),
>+                         container_algorithm_internal::c_end(sequence),
>+                         std::forward<T>(init));
>+}
>+
>+// Overload of c_accumulate() for using a binary operations other than
>+// addition for computing the accumulation.
>+template <typename Sequence, typename T, typename BinaryOp>
>+decay_t<T> c_accumulate(const Sequence& sequence, T&& init,
>+                        BinaryOp&& binary_op) {
>+  return std::accumulate(container_algorithm_internal::c_begin(sequence),
>+                         container_algorithm_internal::c_end(sequence),
>+                         std::forward<T>(init),
>+                         std::forward<BinaryOp>(binary_op));
>+}
>+
>+// c_inner_product()
>+//
>+// Container-based version of the <algorithm> `std::inner_product()` function
>+// to compute the cumulative inner product of container element pairs.
>+//
>+// Note: Due to a language technicality this function has return type
>+// absl::decay_t<T>. As a user of this function you can casually read
>+// this as "returns T by value" and assume it does the right thing.
>+template <typename Sequence1, typename Sequence2, typename T>
>+decay_t<T> c_inner_product(const Sequence1& factors1, const Sequence2& factors2,
>+                           T&& sum) {
>+  return std::inner_product(container_algorithm_internal::c_begin(factors1),
>+                            container_algorithm_internal::c_end(factors1),
>+                            container_algorithm_internal::c_begin(factors2),
>+                            std::forward<T>(sum));
>+}
>+
>+// Overload of c_inner_product() for using binary operations other than
>+// `operator+` (for computing the accumulation) and `operator*` (for computing
>+// the product between the two container's element pair).
>+template <typename Sequence1, typename Sequence2, typename T,
>+          typename BinaryOp1, typename BinaryOp2>
>+decay_t<T> c_inner_product(const Sequence1& factors1, const Sequence2& factors2,
>+                           T&& sum, BinaryOp1&& op1, BinaryOp2&& op2) {
>+  return std::inner_product(container_algorithm_internal::c_begin(factors1),
>+                            container_algorithm_internal::c_end(factors1),
>+                            container_algorithm_internal::c_begin(factors2),
>+                            std::forward<T>(sum), std::forward<BinaryOp1>(op1),
>+                            std::forward<BinaryOp2>(op2));
>+}
>+
>+// c_adjacent_difference()
>+//
>+// Container-based version of the <algorithm> `std::adjacent_difference()`
>+// function to compute the difference between each element and the one preceding
>+// it and write it to an iterator.
>+template <typename InputSequence, typename OutputIt>
>+OutputIt c_adjacent_difference(const InputSequence& input,
>+                               OutputIt output_first) {
>+  return std::adjacent_difference(container_algorithm_internal::c_begin(input),
>+                                  container_algorithm_internal::c_end(input),
>+                                  output_first);
>+}
>+
>+// Overload of c_adjacent_difference() for using a binary operation other than
>+// subtraction to compute the adjacent difference.
>+template <typename InputSequence, typename OutputIt, typename BinaryOp>
>+OutputIt c_adjacent_difference(const InputSequence& input,
>+                               OutputIt output_first, BinaryOp&& op) {
>+  return std::adjacent_difference(container_algorithm_internal::c_begin(input),
>+                                  container_algorithm_internal::c_end(input),
>+                                  output_first, std::forward<BinaryOp>(op));
>+}
>+
>+// c_partial_sum()
>+//
>+// Container-based version of the <algorithm> `std::partial_sum()` function
>+// to compute the partial sum of the elements in a sequence and write them
>+// to an iterator. The partial sum is the sum of all element values so far in
>+// the sequence.
>+template <typename InputSequence, typename OutputIt>
>+OutputIt c_partial_sum(const InputSequence& input, OutputIt output_first) {
>+  return std::partial_sum(container_algorithm_internal::c_begin(input),
>+                          container_algorithm_internal::c_end(input),
>+                          output_first);
>+}
>+
>+// Overload of c_partial_sum() for using a binary operation other than addition
>+// to compute the "partial sum".
>+template <typename InputSequence, typename OutputIt, typename BinaryOp>
>+OutputIt c_partial_sum(const InputSequence& input, OutputIt output_first,
>+                       BinaryOp&& op) {
>+  return std::partial_sum(container_algorithm_internal::c_begin(input),
>+                          container_algorithm_internal::c_end(input),
>+                          output_first, std::forward<BinaryOp>(op));
>+}
>+
>+}  // namespace absl
>+
>+#endif  // ABSL_ALGORITHM_CONTAINER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/algorithm/container_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/algorithm/container_test.cc
>new file mode 100644
>index 00000000000..de66f14680f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/algorithm/container_test.cc
>@@ -0,0 +1,997 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/algorithm/container.h"
>+
>+#include <functional>
>+#include <initializer_list>
>+#include <iterator>
>+#include <list>
>+#include <memory>
>+#include <ostream>
>+#include <random>
>+#include <set>
>+#include <unordered_set>
>+#include <utility>
>+#include <valarray>
>+#include <vector>
>+
>+#include "gmock/gmock.h"
>+#include "gtest/gtest.h"
>+#include "absl/base/casts.h"
>+#include "absl/base/macros.h"
>+#include "absl/memory/memory.h"
>+#include "absl/types/span.h"
>+
>+namespace {
>+
>+using ::testing::Each;
>+using ::testing::ElementsAre;
>+using ::testing::Gt;
>+using ::testing::IsNull;
>+using ::testing::Lt;
>+using ::testing::Pointee;
>+using ::testing::Truly;
>+using ::testing::UnorderedElementsAre;
>+
>+// Most of these tests just check that the code compiles, not that it
>+// does the right thing. That's fine since the functions just forward
>+// to the STL implementation.
>+class NonMutatingTest : public testing::Test {
>+ protected:
>+  std::unordered_set<int> container_ = {1, 2, 3};
>+  std::list<int> sequence_ = {1, 2, 3};
>+  std::vector<int> vector_ = {1, 2, 3};
>+  int array_[3] = {1, 2, 3};
>+};
>+
>+struct AccumulateCalls {
>+  void operator()(int value) {
>+    calls.push_back(value);
>+  }
>+  std::vector<int> calls;
>+};
>+
>+bool Predicate(int value) { return value < 3; }
>+bool BinPredicate(int v1, int v2) { return v1 < v2; }
>+bool Equals(int v1, int v2) { return v1 == v2; }
>+bool IsOdd(int x) { return x % 2 != 0; }
>+
>+
>+TEST_F(NonMutatingTest, Distance) {
>+  EXPECT_EQ(container_.size(), absl::c_distance(container_));
>+  EXPECT_EQ(sequence_.size(), absl::c_distance(sequence_));
>+  EXPECT_EQ(vector_.size(), absl::c_distance(vector_));
>+  EXPECT_EQ(ABSL_ARRAYSIZE(array_), absl::c_distance(array_));
>+
>+  // Works with a temporary argument.
>+  EXPECT_EQ(vector_.size(), absl::c_distance(std::vector<int>(vector_)));
>+}
>+
>+TEST_F(NonMutatingTest, Distance_OverloadedBeginEnd) {
>+  // Works with classes which have custom ADL-selected overloads of std::begin
>+  // and std::end.
>+  std::initializer_list<int> a = {1, 2, 3};
>+  std::valarray<int> b = {1, 2, 3};
>+  EXPECT_EQ(3, absl::c_distance(a));
>+  EXPECT_EQ(3, absl::c_distance(b));
>+
>+  // It is assumed that other c_* functions use the same mechanism for
>+  // ADL-selecting begin/end overloads.
>+}
>+
>+TEST_F(NonMutatingTest, ForEach) {
>+  AccumulateCalls c = absl::c_for_each(container_, AccumulateCalls());
>+  // Don't rely on the unordered_set's order.
>+  std::sort(c.calls.begin(), c.calls.end());
>+  EXPECT_EQ(vector_, c.calls);
>+
>+  // Works with temporary container, too.
>+  AccumulateCalls c2 =
>+      absl::c_for_each(std::unordered_set<int>(container_), AccumulateCalls());
>+  std::sort(c2.calls.begin(), c2.calls.end());
>+  EXPECT_EQ(vector_, c2.calls);
>+}
>+
>+TEST_F(NonMutatingTest, FindReturnsCorrectType) {
>+  auto it = absl::c_find(container_, 3);
>+  EXPECT_EQ(3, *it);
>+  absl::c_find(absl::implicit_cast<const std::list<int>&>(sequence_), 3);
>+}
>+
>+TEST_F(NonMutatingTest, FindIf) { absl::c_find_if(container_, Predicate); }
>+
>+TEST_F(NonMutatingTest, FindIfNot) {
>+  absl::c_find_if_not(container_, Predicate);
>+}
>+
>+TEST_F(NonMutatingTest, FindEnd) {
>+  absl::c_find_end(sequence_, vector_);
>+  absl::c_find_end(vector_, sequence_);
>+}
>+
>+TEST_F(NonMutatingTest, FindEndWithPredicate) {
>+  absl::c_find_end(sequence_, vector_, BinPredicate);
>+  absl::c_find_end(vector_, sequence_, BinPredicate);
>+}
>+
>+TEST_F(NonMutatingTest, FindFirstOf) {
>+  absl::c_find_first_of(container_, sequence_);
>+  absl::c_find_first_of(sequence_, container_);
>+}
>+
>+TEST_F(NonMutatingTest, FindFirstOfWithPredicate) {
>+  absl::c_find_first_of(container_, sequence_, BinPredicate);
>+  absl::c_find_first_of(sequence_, container_, BinPredicate);
>+}
>+
>+TEST_F(NonMutatingTest, AdjacentFind) { absl::c_adjacent_find(sequence_); }
>+
>+TEST_F(NonMutatingTest, AdjacentFindWithPredicate) {
>+  absl::c_adjacent_find(sequence_, BinPredicate);
>+}
>+
>+TEST_F(NonMutatingTest, Count) { EXPECT_EQ(1, absl::c_count(container_, 3)); }
>+
>+TEST_F(NonMutatingTest, CountIf) {
>+  EXPECT_EQ(2, absl::c_count_if(container_, Predicate));
>+  const std::unordered_set<int>& const_container = container_;
>+  EXPECT_EQ(2, absl::c_count_if(const_container, Predicate));
>+}
>+
>+TEST_F(NonMutatingTest, Mismatch) {
>+  absl::c_mismatch(container_, sequence_);
>+  absl::c_mismatch(sequence_, container_);
>+}
>+
>+TEST_F(NonMutatingTest, MismatchWithPredicate) {
>+  absl::c_mismatch(container_, sequence_, BinPredicate);
>+  absl::c_mismatch(sequence_, container_, BinPredicate);
>+}
>+
>+TEST_F(NonMutatingTest, Equal) {
>+  EXPECT_TRUE(absl::c_equal(vector_, sequence_));
>+  EXPECT_TRUE(absl::c_equal(sequence_, vector_));
>+
>+  // Test that behavior appropriately differs from that of equal().
>+  std::vector<int> vector_plus = {1, 2, 3};
>+  vector_plus.push_back(4);
>+  EXPECT_FALSE(absl::c_equal(vector_plus, sequence_));
>+  EXPECT_FALSE(absl::c_equal(sequence_, vector_plus));
>+}
>+
>+TEST_F(NonMutatingTest, EqualWithPredicate) {
>+  EXPECT_TRUE(absl::c_equal(vector_, sequence_, Equals));
>+  EXPECT_TRUE(absl::c_equal(sequence_, vector_, Equals));
>+
>+  // Test that behavior appropriately differs from that of equal().
>+  std::vector<int> vector_plus = {1, 2, 3};
>+  vector_plus.push_back(4);
>+  EXPECT_FALSE(absl::c_equal(vector_plus, sequence_, Equals));
>+  EXPECT_FALSE(absl::c_equal(sequence_, vector_plus, Equals));
>+}
>+
>+TEST_F(NonMutatingTest, IsPermutation) {
>+  auto vector_permut_ = vector_;
>+  std::next_permutation(vector_permut_.begin(), vector_permut_.end());
>+  EXPECT_TRUE(absl::c_is_permutation(vector_permut_, sequence_));
>+  EXPECT_TRUE(absl::c_is_permutation(sequence_, vector_permut_));
>+
>+  // Test that behavior appropriately differs from that of is_permutation().
>+  std::vector<int> vector_plus = {1, 2, 3};
>+  vector_plus.push_back(4);
>+  EXPECT_FALSE(absl::c_is_permutation(vector_plus, sequence_));
>+  EXPECT_FALSE(absl::c_is_permutation(sequence_, vector_plus));
>+}
>+
>+TEST_F(NonMutatingTest, IsPermutationWithPredicate) {
>+  auto vector_permut_ = vector_;
>+  std::next_permutation(vector_permut_.begin(), vector_permut_.end());
>+  EXPECT_TRUE(absl::c_is_permutation(vector_permut_, sequence_, Equals));
>+  EXPECT_TRUE(absl::c_is_permutation(sequence_, vector_permut_, Equals));
>+
>+  // Test that behavior appropriately differs from that of is_permutation().
>+  std::vector<int> vector_plus = {1, 2, 3};
>+  vector_plus.push_back(4);
>+  EXPECT_FALSE(absl::c_is_permutation(vector_plus, sequence_, Equals));
>+  EXPECT_FALSE(absl::c_is_permutation(sequence_, vector_plus, Equals));
>+}
>+
>+TEST_F(NonMutatingTest, Search) {
>+  absl::c_search(sequence_, vector_);
>+  absl::c_search(vector_, sequence_);
>+  absl::c_search(array_, sequence_);
>+}
>+
>+TEST_F(NonMutatingTest, SearchWithPredicate) {
>+  absl::c_search(sequence_, vector_, BinPredicate);
>+  absl::c_search(vector_, sequence_, BinPredicate);
>+}
>+
>+TEST_F(NonMutatingTest, SearchN) { absl::c_search_n(sequence_, 3, 1); }
>+
>+TEST_F(NonMutatingTest, SearchNWithPredicate) {
>+  absl::c_search_n(sequence_, 3, 1, BinPredicate);
>+}
>+
>+TEST_F(NonMutatingTest, LowerBound) {
>+  std::list<int>::iterator i = absl::c_lower_bound(sequence_, 3);
>+  ASSERT_TRUE(i != sequence_.end());
>+  EXPECT_EQ(2, std::distance(sequence_.begin(), i));
>+  EXPECT_EQ(3, *i);
>+}
>+
>+TEST_F(NonMutatingTest, LowerBoundWithPredicate) {
>+  std::vector<int> v(vector_);
>+  std::sort(v.begin(), v.end(), std::greater<int>());
>+  std::vector<int>::iterator i = absl::c_lower_bound(v, 3, std::greater<int>());
>+  EXPECT_TRUE(i == v.begin());
>+  EXPECT_EQ(3, *i);
>+}
>+
>+TEST_F(NonMutatingTest, UpperBound) {
>+  std::list<int>::iterator i = absl::c_upper_bound(sequence_, 1);
>+  ASSERT_TRUE(i != sequence_.end());
>+  EXPECT_EQ(1, std::distance(sequence_.begin(), i));
>+  EXPECT_EQ(2, *i);
>+}
>+
>+TEST_F(NonMutatingTest, UpperBoundWithPredicate) {
>+  std::vector<int> v(vector_);
>+  std::sort(v.begin(), v.end(), std::greater<int>());
>+  std::vector<int>::iterator i = absl::c_upper_bound(v, 1, std::greater<int>());
>+  EXPECT_EQ(3, i - v.begin());
>+  EXPECT_TRUE(i == v.end());
>+}
>+
>+TEST_F(NonMutatingTest, EqualRange) {
>+  std::pair<std::list<int>::iterator, std::list<int>::iterator> p =
>+      absl::c_equal_range(sequence_, 2);
>+  EXPECT_EQ(1, std::distance(sequence_.begin(), p.first));
>+  EXPECT_EQ(2, std::distance(sequence_.begin(), p.second));
>+}
>+
>+TEST_F(NonMutatingTest, EqualRangeArray) {
>+  auto p = absl::c_equal_range(array_, 2);
>+  EXPECT_EQ(1, std::distance(std::begin(array_), p.first));
>+  EXPECT_EQ(2, std::distance(std::begin(array_), p.second));
>+}
>+
>+TEST_F(NonMutatingTest, EqualRangeWithPredicate) {
>+  std::vector<int> v(vector_);
>+  std::sort(v.begin(), v.end(), std::greater<int>());
>+  std::pair<std::vector<int>::iterator, std::vector<int>::iterator> p =
>+      absl::c_equal_range(v, 2, std::greater<int>());
>+  EXPECT_EQ(1, std::distance(v.begin(), p.first));
>+  EXPECT_EQ(2, std::distance(v.begin(), p.second));
>+}
>+
>+TEST_F(NonMutatingTest, BinarySearch) {
>+  EXPECT_TRUE(absl::c_binary_search(vector_, 2));
>+  EXPECT_TRUE(absl::c_binary_search(std::vector<int>(vector_), 2));
>+}
>+
>+TEST_F(NonMutatingTest, BinarySearchWithPredicate) {
>+  std::vector<int> v(vector_);
>+  std::sort(v.begin(), v.end(), std::greater<int>());
>+  EXPECT_TRUE(absl::c_binary_search(v, 2, std::greater<int>()));
>+  EXPECT_TRUE(
>+      absl::c_binary_search(std::vector<int>(v), 2, std::greater<int>()));
>+}
>+
>+TEST_F(NonMutatingTest, MinElement) {
>+  std::list<int>::iterator i = absl::c_min_element(sequence_);
>+  ASSERT_TRUE(i != sequence_.end());
>+  EXPECT_EQ(*i, 1);
>+}
>+
>+TEST_F(NonMutatingTest, MinElementWithPredicate) {
>+  std::list<int>::iterator i =
>+      absl::c_min_element(sequence_, std::greater<int>());
>+  ASSERT_TRUE(i != sequence_.end());
>+  EXPECT_EQ(*i, 3);
>+}
>+
>+TEST_F(NonMutatingTest, MaxElement) {
>+  std::list<int>::iterator i = absl::c_max_element(sequence_);
>+  ASSERT_TRUE(i != sequence_.end());
>+  EXPECT_EQ(*i, 3);
>+}
>+
>+TEST_F(NonMutatingTest, MaxElementWithPredicate) {
>+  std::list<int>::iterator i =
>+      absl::c_max_element(sequence_, std::greater<int>());
>+  ASSERT_TRUE(i != sequence_.end());
>+  EXPECT_EQ(*i, 1);
>+}
>+
>+TEST_F(NonMutatingTest, LexicographicalCompare) {
>+  EXPECT_FALSE(absl::c_lexicographical_compare(sequence_, sequence_));
>+
>+  std::vector<int> v;
>+  v.push_back(1);
>+  v.push_back(2);
>+  v.push_back(4);
>+
>+  EXPECT_TRUE(absl::c_lexicographical_compare(sequence_, v));
>+  EXPECT_TRUE(absl::c_lexicographical_compare(std::list<int>(sequence_), v));
>+}
>+
>+TEST_F(NonMutatingTest, LexicographicalCopmareWithPredicate) {
>+  EXPECT_FALSE(absl::c_lexicographical_compare(sequence_, sequence_,
>+                                               std::greater<int>()));
>+
>+  std::vector<int> v;
>+  v.push_back(1);
>+  v.push_back(2);
>+  v.push_back(4);
>+
>+  EXPECT_TRUE(
>+      absl::c_lexicographical_compare(v, sequence_, std::greater<int>()));
>+  EXPECT_TRUE(absl::c_lexicographical_compare(
>+      std::vector<int>(v), std::list<int>(sequence_), std::greater<int>()));
>+}
>+
>+TEST_F(NonMutatingTest, Includes) {
>+  std::set<int> s(vector_.begin(), vector_.end());
>+  s.insert(4);
>+  EXPECT_TRUE(absl::c_includes(s, vector_));
>+}
>+
>+TEST_F(NonMutatingTest, IncludesWithPredicate) {
>+  std::vector<int> v = {3, 2, 1};
>+  std::set<int, std::greater<int>> s(v.begin(), v.end());
>+  s.insert(4);
>+  EXPECT_TRUE(absl::c_includes(s, v, std::greater<int>()));
>+}
>+
>+class NumericMutatingTest : public testing::Test {
>+ protected:
>+  std::list<int> list_ = {1, 2, 3};
>+  std::vector<int> output_;
>+};
>+
>+TEST_F(NumericMutatingTest, Iota) {
>+  absl::c_iota(list_, 5);
>+  std::list<int> expected{5, 6, 7};
>+  EXPECT_EQ(list_, expected);
>+}
>+
>+TEST_F(NonMutatingTest, Accumulate) {
>+  EXPECT_EQ(absl::c_accumulate(sequence_, 4), 1 + 2 + 3 + 4);
>+}
>+
>+TEST_F(NonMutatingTest, AccumulateWithBinaryOp) {
>+  EXPECT_EQ(absl::c_accumulate(sequence_, 4, std::multiplies<int>()),
>+            1 * 2 * 3 * 4);
>+}
>+
>+TEST_F(NonMutatingTest, AccumulateLvalueInit) {
>+  int lvalue = 4;
>+  EXPECT_EQ(absl::c_accumulate(sequence_, lvalue), 1 + 2 + 3 + 4);
>+}
>+
>+TEST_F(NonMutatingTest, AccumulateWithBinaryOpLvalueInit) {
>+  int lvalue = 4;
>+  EXPECT_EQ(absl::c_accumulate(sequence_, lvalue, std::multiplies<int>()),
>+            1 * 2 * 3 * 4);
>+}
>+
>+TEST_F(NonMutatingTest, InnerProduct) {
>+  EXPECT_EQ(absl::c_inner_product(sequence_, vector_, 1000),
>+            1000 + 1 * 1 + 2 * 2 + 3 * 3);
>+}
>+
>+TEST_F(NonMutatingTest, InnerProductWithBinaryOps) {
>+  EXPECT_EQ(absl::c_inner_product(sequence_, vector_, 10,
>+                                  std::multiplies<int>(), std::plus<int>()),
>+            10 * (1 + 1) * (2 + 2) * (3 + 3));
>+}
>+
>+TEST_F(NonMutatingTest, InnerProductLvalueInit) {
>+  int lvalue = 1000;
>+  EXPECT_EQ(absl::c_inner_product(sequence_, vector_, lvalue),
>+            1000 + 1 * 1 + 2 * 2 + 3 * 3);
>+}
>+
>+TEST_F(NonMutatingTest, InnerProductWithBinaryOpsLvalueInit) {
>+  int lvalue = 10;
>+  EXPECT_EQ(absl::c_inner_product(sequence_, vector_, lvalue,
>+                                  std::multiplies<int>(), std::plus<int>()),
>+            10 * (1 + 1) * (2 + 2) * (3 + 3));
>+}
>+
>+TEST_F(NumericMutatingTest, AdjacentDifference) {
>+  auto last = absl::c_adjacent_difference(list_, std::back_inserter(output_));
>+  *last = 1000;
>+  std::vector<int> expected{1, 2 - 1, 3 - 2, 1000};
>+  EXPECT_EQ(output_, expected);
>+}
>+
>+TEST_F(NumericMutatingTest, AdjacentDifferenceWithBinaryOp) {
>+  auto last = absl::c_adjacent_difference(list_, std::back_inserter(output_),
>+                                          std::multiplies<int>());
>+  *last = 1000;
>+  std::vector<int> expected{1, 2 * 1, 3 * 2, 1000};
>+  EXPECT_EQ(output_, expected);
>+}
>+
>+TEST_F(NumericMutatingTest, PartialSum) {
>+  auto last = absl::c_partial_sum(list_, std::back_inserter(output_));
>+  *last = 1000;
>+  std::vector<int> expected{1, 1 + 2, 1 + 2 + 3, 1000};
>+  EXPECT_EQ(output_, expected);
>+}
>+
>+TEST_F(NumericMutatingTest, PartialSumWithBinaryOp) {
>+  auto last = absl::c_partial_sum(list_, std::back_inserter(output_),
>+                                  std::multiplies<int>());
>+  *last = 1000;
>+  std::vector<int> expected{1, 1 * 2, 1 * 2 * 3, 1000};
>+  EXPECT_EQ(output_, expected);
>+}
>+
>+TEST_F(NonMutatingTest, LinearSearch) {
>+  EXPECT_TRUE(absl::c_linear_search(container_, 3));
>+  EXPECT_FALSE(absl::c_linear_search(container_, 4));
>+}
>+
>+TEST_F(NonMutatingTest, AllOf) {
>+  const std::vector<int>& v = vector_;
>+  EXPECT_FALSE(absl::c_all_of(v, [](int x) { return x > 1; }));
>+  EXPECT_TRUE(absl::c_all_of(v, [](int x) { return x > 0; }));
>+}
>+
>+TEST_F(NonMutatingTest, AnyOf) {
>+  const std::vector<int>& v = vector_;
>+  EXPECT_TRUE(absl::c_any_of(v, [](int x) { return x > 2; }));
>+  EXPECT_FALSE(absl::c_any_of(v, [](int x) { return x > 5; }));
>+}
>+
>+TEST_F(NonMutatingTest, NoneOf) {
>+  const std::vector<int>& v = vector_;
>+  EXPECT_FALSE(absl::c_none_of(v, [](int x) { return x > 2; }));
>+  EXPECT_TRUE(absl::c_none_of(v, [](int x) { return x > 5; }));
>+}
>+
>+TEST_F(NonMutatingTest, MinMaxElementLess) {
>+  std::pair<std::vector<int>::const_iterator, std::vector<int>::const_iterator>
>+      p = absl::c_minmax_element(vector_, std::less<int>());
>+  EXPECT_TRUE(p.first == vector_.begin());
>+  EXPECT_TRUE(p.second == vector_.begin() + 2);
>+}
>+
>+TEST_F(NonMutatingTest, MinMaxElementGreater) {
>+  std::pair<std::vector<int>::const_iterator, std::vector<int>::const_iterator>
>+      p = absl::c_minmax_element(vector_, std::greater<int>());
>+  EXPECT_TRUE(p.first == vector_.begin() + 2);
>+  EXPECT_TRUE(p.second == vector_.begin());
>+}
>+
>+TEST_F(NonMutatingTest, MinMaxElementNoPredicate) {
>+  std::pair<std::vector<int>::const_iterator, std::vector<int>::const_iterator>
>+      p = absl::c_minmax_element(vector_);
>+  EXPECT_TRUE(p.first == vector_.begin());
>+  EXPECT_TRUE(p.second == vector_.begin() + 2);
>+}
>+
>+class SortingTest : public testing::Test {
>+ protected:
>+  std::list<int> sorted_ = {1, 2, 3, 4};
>+  std::list<int> unsorted_ = {2, 4, 1, 3};
>+  std::list<int> reversed_ = {4, 3, 2, 1};
>+};
>+
>+TEST_F(SortingTest, IsSorted) {
>+  EXPECT_TRUE(absl::c_is_sorted(sorted_));
>+  EXPECT_FALSE(absl::c_is_sorted(unsorted_));
>+  EXPECT_FALSE(absl::c_is_sorted(reversed_));
>+}
>+
>+TEST_F(SortingTest, IsSortedWithPredicate) {
>+  EXPECT_FALSE(absl::c_is_sorted(sorted_, std::greater<int>()));
>+  EXPECT_FALSE(absl::c_is_sorted(unsorted_, std::greater<int>()));
>+  EXPECT_TRUE(absl::c_is_sorted(reversed_, std::greater<int>()));
>+}
>+
>+TEST_F(SortingTest, IsSortedUntil) {
>+  EXPECT_EQ(1, *absl::c_is_sorted_until(unsorted_));
>+  EXPECT_EQ(4, *absl::c_is_sorted_until(unsorted_, std::greater<int>()));
>+}
>+
>+TEST_F(SortingTest, NthElement) {
>+  std::vector<int> unsorted = {2, 4, 1, 3};
>+  absl::c_nth_element(unsorted, unsorted.begin() + 2);
>+  EXPECT_THAT(unsorted,
>+              ElementsAre(Lt(3), Lt(3), 3, Gt(3)));
>+  absl::c_nth_element(unsorted, unsorted.begin() + 2, std::greater<int>());
>+  EXPECT_THAT(unsorted,
>+              ElementsAre(Gt(2), Gt(2), 2, Lt(2)));
>+}
>+
>+TEST(MutatingTest, IsPartitioned) {
>+  EXPECT_TRUE(
>+      absl::c_is_partitioned(std::vector<int>{1, 3, 5, 2, 4, 6}, IsOdd));
>+  EXPECT_FALSE(
>+      absl::c_is_partitioned(std::vector<int>{1, 2, 3, 4, 5, 6}, IsOdd));
>+  EXPECT_FALSE(
>+      absl::c_is_partitioned(std::vector<int>{2, 4, 6, 1, 3, 5}, IsOdd));
>+}
>+
>+TEST(MutatingTest, Partition) {
>+  std::vector<int> actual = {1, 2, 3, 4, 5};
>+  absl::c_partition(actual, IsOdd);
>+  EXPECT_THAT(actual, Truly([](const std::vector<int>& c) {
>+                return absl::c_is_partitioned(c, IsOdd);
>+              }));
>+}
>+
>+TEST(MutatingTest, StablePartition) {
>+  std::vector<int> actual = {1, 2, 3, 4, 5};
>+  absl::c_stable_partition(actual, IsOdd);
>+  EXPECT_THAT(actual, ElementsAre(1, 3, 5, 2, 4));
>+}
>+
>+TEST(MutatingTest, PartitionCopy) {
>+  const std::vector<int> initial = {1, 2, 3, 4, 5};
>+  std::vector<int> odds, evens;
>+  auto ends = absl::c_partition_copy(initial, back_inserter(odds),
>+                                     back_inserter(evens), IsOdd);
>+  *ends.first = 7;
>+  *ends.second = 6;
>+  EXPECT_THAT(odds, ElementsAre(1, 3, 5, 7));
>+  EXPECT_THAT(evens, ElementsAre(2, 4, 6));
>+}
>+
>+TEST(MutatingTest, PartitionPoint) {
>+  const std::vector<int> initial = {1, 3, 5, 2, 4};
>+  auto middle = absl::c_partition_point(initial, IsOdd);
>+  EXPECT_EQ(2, *middle);
>+}
>+
>+TEST(MutatingTest, CopyMiddle) {
>+  const std::vector<int> initial = {4, -1, -2, -3, 5};
>+  const std::list<int> input = {1, 2, 3};
>+  const std::vector<int> expected = {4, 1, 2, 3, 5};
>+
>+  std::list<int> test_list(initial.begin(), initial.end());
>+  absl::c_copy(input, ++test_list.begin());
>+  EXPECT_EQ(std::list<int>(expected.begin(), expected.end()), test_list);
>+
>+  std::vector<int> test_vector = initial;
>+  absl::c_copy(input, test_vector.begin() + 1);
>+  EXPECT_EQ(expected, test_vector);
>+}
>+
>+TEST(MutatingTest, CopyFrontInserter) {
>+  const std::list<int> initial = {4, 5};
>+  const std::list<int> input = {1, 2, 3};
>+  const std::list<int> expected = {3, 2, 1, 4, 5};
>+
>+  std::list<int> test_list = initial;
>+  absl::c_copy(input, std::front_inserter(test_list));
>+  EXPECT_EQ(expected, test_list);
>+}
>+
>+TEST(MutatingTest, CopyBackInserter) {
>+  const std::vector<int> initial = {4, 5};
>+  const std::list<int> input = {1, 2, 3};
>+  const std::vector<int> expected = {4, 5, 1, 2, 3};
>+
>+  std::list<int> test_list(initial.begin(), initial.end());
>+  absl::c_copy(input, std::back_inserter(test_list));
>+  EXPECT_EQ(std::list<int>(expected.begin(), expected.end()), test_list);
>+
>+  std::vector<int> test_vector = initial;
>+  absl::c_copy(input, std::back_inserter(test_vector));
>+  EXPECT_EQ(expected, test_vector);
>+}
>+
>+TEST(MutatingTest, CopyN) {
>+  const std::vector<int> initial = {1, 2, 3, 4, 5};
>+  const std::vector<int> expected = {1, 2};
>+  std::vector<int> actual;
>+  absl::c_copy_n(initial, 2, back_inserter(actual));
>+  EXPECT_EQ(expected, actual);
>+}
>+
>+TEST(MutatingTest, CopyIf) {
>+  const std::list<int> input = {1, 2, 3};
>+  std::vector<int> output;
>+  absl::c_copy_if(input, std::back_inserter(output),
>+                  [](int i) { return i != 2; });
>+  EXPECT_THAT(output, ElementsAre(1, 3));
>+}
>+
>+TEST(MutatingTest, CopyBackward) {
>+  std::vector<int> actual = {1, 2, 3, 4, 5};
>+  std::vector<int> expected = {1, 2, 1, 2, 3};
>+  absl::c_copy_backward(absl::MakeSpan(actual.data(), 3), actual.end());
>+  EXPECT_EQ(expected, actual);
>+}
>+
>+TEST(MutatingTest, Move) {
>+  std::vector<std::unique_ptr<int>> src;
>+  src.emplace_back(absl::make_unique<int>(1));
>+  src.emplace_back(absl::make_unique<int>(2));
>+  src.emplace_back(absl::make_unique<int>(3));
>+  src.emplace_back(absl::make_unique<int>(4));
>+  src.emplace_back(absl::make_unique<int>(5));
>+
>+  std::vector<std::unique_ptr<int>> dest = {};
>+  absl::c_move(src, std::back_inserter(dest));
>+  EXPECT_THAT(src, Each(IsNull()));
>+  EXPECT_THAT(dest, ElementsAre(Pointee(1), Pointee(2), Pointee(3), Pointee(4),
>+                                Pointee(5)));
>+}
>+
>+TEST(MutatingTest, SwapRanges) {
>+  std::vector<int> odds = {2, 4, 6};
>+  std::vector<int> evens = {1, 3, 5};
>+  absl::c_swap_ranges(odds, evens);
>+  EXPECT_THAT(odds, ElementsAre(1, 3, 5));
>+  EXPECT_THAT(evens, ElementsAre(2, 4, 6));
>+}
>+
>+TEST_F(NonMutatingTest, Transform) {
>+  std::vector<int> x{0, 2, 4}, y, z;
>+  auto end = absl::c_transform(x, back_inserter(y), std::negate<int>());
>+  EXPECT_EQ(std::vector<int>({0, -2, -4}), y);
>+  *end = 7;
>+  EXPECT_EQ(std::vector<int>({0, -2, -4, 7}), y);
>+
>+  y = {1, 3, 0};
>+  end = absl::c_transform(x, y, back_inserter(z), std::plus<int>());
>+  EXPECT_EQ(std::vector<int>({1, 5, 4}), z);
>+  *end = 7;
>+  EXPECT_EQ(std::vector<int>({1, 5, 4, 7}), z);
>+}
>+
>+TEST(MutatingTest, Replace) {
>+  const std::vector<int> initial = {1, 2, 3, 1, 4, 5};
>+  const std::vector<int> expected = {4, 2, 3, 4, 4, 5};
>+
>+  std::vector<int> test_vector = initial;
>+  absl::c_replace(test_vector, 1, 4);
>+  EXPECT_EQ(expected, test_vector);
>+
>+  std::list<int> test_list(initial.begin(), initial.end());
>+  absl::c_replace(test_list, 1, 4);
>+  EXPECT_EQ(std::list<int>(expected.begin(), expected.end()), test_list);
>+}
>+
>+TEST(MutatingTest, ReplaceIf) {
>+  std::vector<int> actual = {1, 2, 3, 4, 5};
>+  const std::vector<int> expected = {0, 2, 0, 4, 0};
>+
>+  absl::c_replace_if(actual, IsOdd, 0);
>+  EXPECT_EQ(expected, actual);
>+}
>+
>+TEST(MutatingTest, ReplaceCopy) {
>+  const std::vector<int> initial = {1, 2, 3, 1, 4, 5};
>+  const std::vector<int> expected = {4, 2, 3, 4, 4, 5};
>+
>+  std::vector<int> actual;
>+  absl::c_replace_copy(initial, back_inserter(actual), 1, 4);
>+  EXPECT_EQ(expected, actual);
>+}
>+
>+TEST(MutatingTest, Sort) {
>+  std::vector<int> test_vector = {2, 3, 1, 4};
>+  absl::c_sort(test_vector);
>+  EXPECT_THAT(test_vector, ElementsAre(1, 2, 3, 4));
>+}
>+
>+TEST(MutatingTest, SortWithPredicate) {
>+  std::vector<int> test_vector = {2, 3, 1, 4};
>+  absl::c_sort(test_vector, std::greater<int>());
>+  EXPECT_THAT(test_vector, ElementsAre(4, 3, 2, 1));
>+}
>+
>+// For absl::c_stable_sort tests. Needs an operator< that does not cover all
>+// fields so that the test can check the sort preserves order of equal elements.
>+struct Element {
>+  int key;
>+  int value;
>+  friend bool operator<(const Element& e1, const Element& e2) {
>+    return e1.key < e2.key;
>+  }
>+  // Make gmock print useful diagnostics.
>+  friend std::ostream& operator<<(std::ostream& o, const Element& e) {
>+    return o << "{" << e.key << ", " << e.value << "}";
>+  }
>+};
>+
>+MATCHER_P2(IsElement, key, value, "") {
>+  return arg.key == key && arg.value == value;
>+}
>+
>+TEST(MutatingTest, StableSort) {
>+  std::vector<Element> test_vector = {{1, 1}, {2, 1}, {2, 0}, {1, 0}, {2, 2}};
>+  absl::c_stable_sort(test_vector);
>+  EXPECT_THAT(
>+      test_vector,
>+      ElementsAre(IsElement(1, 1), IsElement(1, 0), IsElement(2, 1),
>+                  IsElement(2, 0), IsElement(2, 2)));
>+}
>+
>+TEST(MutatingTest, StableSortWithPredicate) {
>+  std::vector<Element> test_vector = {{1, 1}, {2, 1}, {2, 0}, {1, 0}, {2, 2}};
>+  absl::c_stable_sort(test_vector, [](const Element& e1, const Element& e2) {
>+    return e2 < e1;
>+  });
>+  EXPECT_THAT(
>+      test_vector,
>+      ElementsAre(IsElement(2, 1), IsElement(2, 0), IsElement(2, 2),
>+                  IsElement(1, 1), IsElement(1, 0)));
>+}
>+
>+TEST(MutatingTest, ReplaceCopyIf) {
>+  const std::vector<int> initial = {1, 2, 3, 4, 5};
>+  const std::vector<int> expected = {0, 2, 0, 4, 0};
>+
>+  std::vector<int> actual;
>+  absl::c_replace_copy_if(initial, back_inserter(actual), IsOdd, 0);
>+  EXPECT_EQ(expected, actual);
>+}
>+
>+TEST(MutatingTest, Fill) {
>+  std::vector<int> actual(5);
>+  absl::c_fill(actual, 1);
>+  EXPECT_THAT(actual, ElementsAre(1, 1, 1, 1, 1));
>+}
>+
>+TEST(MutatingTest, FillN) {
>+  std::vector<int> actual(5, 0);
>+  absl::c_fill_n(actual, 2, 1);
>+  EXPECT_THAT(actual, ElementsAre(1, 1, 0, 0, 0));
>+}
>+
>+TEST(MutatingTest, Generate) {
>+  std::vector<int> actual(5);
>+  int x = 0;
>+  absl::c_generate(actual, [&x]() { return ++x; });
>+  EXPECT_THAT(actual, ElementsAre(1, 2, 3, 4, 5));
>+}
>+
>+TEST(MutatingTest, GenerateN) {
>+  std::vector<int> actual(5, 0);
>+  int x = 0;
>+  absl::c_generate_n(actual, 3, [&x]() { return ++x; });
>+  EXPECT_THAT(actual, ElementsAre(1, 2, 3, 0, 0));
>+}
>+
>+TEST(MutatingTest, RemoveCopy) {
>+  std::vector<int> actual;
>+  absl::c_remove_copy(std::vector<int>{1, 2, 3}, back_inserter(actual), 2);
>+  EXPECT_THAT(actual, ElementsAre(1, 3));
>+}
>+
>+TEST(MutatingTest, RemoveCopyIf) {
>+  std::vector<int> actual;
>+  absl::c_remove_copy_if(std::vector<int>{1, 2, 3}, back_inserter(actual),
>+                         IsOdd);
>+  EXPECT_THAT(actual, ElementsAre(2));
>+}
>+
>+TEST(MutatingTest, UniqueCopy) {
>+  std::vector<int> actual;
>+  absl::c_unique_copy(std::vector<int>{1, 2, 2, 2, 3, 3, 2},
>+                      back_inserter(actual));
>+  EXPECT_THAT(actual, ElementsAre(1, 2, 3, 2));
>+}
>+
>+TEST(MutatingTest, UniqueCopyWithPredicate) {
>+  std::vector<int> actual;
>+  absl::c_unique_copy(std::vector<int>{1, 2, 3, -1, -2, -3, 1},
>+                      back_inserter(actual),
>+                      [](int x, int y) { return (x < 0) == (y < 0); });
>+  EXPECT_THAT(actual, ElementsAre(1, -1, 1));
>+}
>+
>+TEST(MutatingTest, Reverse) {
>+  std::vector<int> test_vector = {1, 2, 3, 4};
>+  absl::c_reverse(test_vector);
>+  EXPECT_THAT(test_vector, ElementsAre(4, 3, 2, 1));
>+
>+  std::list<int> test_list = {1, 2, 3, 4};
>+  absl::c_reverse(test_list);
>+  EXPECT_THAT(test_list, ElementsAre(4, 3, 2, 1));
>+}
>+
>+TEST(MutatingTest, ReverseCopy) {
>+  std::vector<int> actual;
>+  absl::c_reverse_copy(std::vector<int>{1, 2, 3, 4}, back_inserter(actual));
>+  EXPECT_THAT(actual, ElementsAre(4, 3, 2, 1));
>+}
>+
>+TEST(MutatingTest, Rotate) {
>+  std::vector<int> actual = {1, 2, 3, 4};
>+  auto it = absl::c_rotate(actual, actual.begin() + 2);
>+  EXPECT_THAT(actual, testing::ElementsAreArray({3, 4, 1, 2}));
>+  EXPECT_EQ(*it, 1);
>+}
>+
>+TEST(MutatingTest, RotateCopy) {
>+  std::vector<int> initial = {1, 2, 3, 4};
>+  std::vector<int> actual;
>+  auto end =
>+      absl::c_rotate_copy(initial, initial.begin() + 2, back_inserter(actual));
>+  *end = 5;
>+  EXPECT_THAT(actual, ElementsAre(3, 4, 1, 2, 5));
>+}
>+
>+TEST(MutatingTest, Shuffle) {
>+  std::vector<int> actual = {1, 2, 3, 4, 5};
>+  absl::c_shuffle(actual, std::random_device());
>+  EXPECT_THAT(actual, UnorderedElementsAre(1, 2, 3, 4, 5));
>+}
>+
>+TEST(MutatingTest, PartialSort) {
>+  std::vector<int> sequence{5, 3, 42, 0};
>+  absl::c_partial_sort(sequence, sequence.begin() + 2);
>+  EXPECT_THAT(absl::MakeSpan(sequence.data(), 2), ElementsAre(0, 3));
>+  absl::c_partial_sort(sequence, sequence.begin() + 2, std::greater<int>());
>+  EXPECT_THAT(absl::MakeSpan(sequence.data(), 2), ElementsAre(42, 5));
>+}
>+
>+TEST(MutatingTest, PartialSortCopy) {
>+  const std::vector<int> initial = {5, 3, 42, 0};
>+  std::vector<int> actual(2);
>+  absl::c_partial_sort_copy(initial, actual);
>+  EXPECT_THAT(actual, ElementsAre(0, 3));
>+  absl::c_partial_sort_copy(initial, actual, std::greater<int>());
>+  EXPECT_THAT(actual, ElementsAre(42, 5));
>+}
>+
>+TEST(MutatingTest, Merge) {
>+  std::vector<int> actual;
>+  absl::c_merge(std::vector<int>{1, 3, 5}, std::vector<int>{2, 4},
>+                back_inserter(actual));
>+  EXPECT_THAT(actual, ElementsAre(1, 2, 3, 4, 5));
>+}
>+
>+TEST(MutatingTest, MergeWithComparator) {
>+  std::vector<int> actual;
>+  absl::c_merge(std::vector<int>{5, 3, 1}, std::vector<int>{4, 2},
>+                back_inserter(actual), std::greater<int>());
>+  EXPECT_THAT(actual, ElementsAre(5, 4, 3, 2, 1));
>+}
>+
>+TEST(MutatingTest, InplaceMerge) {
>+  std::vector<int> actual = {1, 3, 5, 2, 4};
>+  absl::c_inplace_merge(actual, actual.begin() + 3);
>+  EXPECT_THAT(actual, ElementsAre(1, 2, 3, 4, 5));
>+}
>+
>+TEST(MutatingTest, InplaceMergeWithComparator) {
>+  std::vector<int> actual = {5, 3, 1, 4, 2};
>+  absl::c_inplace_merge(actual, actual.begin() + 3, std::greater<int>());
>+  EXPECT_THAT(actual, ElementsAre(5, 4, 3, 2, 1));
>+}
>+
>+class SetOperationsTest : public testing::Test {
>+ protected:
>+  std::vector<int> a_ = {1, 2, 3};
>+  std::vector<int> b_ = {1, 3, 5};
>+
>+  std::vector<int> a_reversed_ = {3, 2, 1};
>+  std::vector<int> b_reversed_ = {5, 3, 1};
>+};
>+
>+TEST_F(SetOperationsTest, SetUnion) {
>+  std::vector<int> actual;
>+  absl::c_set_union(a_, b_, back_inserter(actual));
>+  EXPECT_THAT(actual, ElementsAre(1, 2, 3, 5));
>+}
>+
>+TEST_F(SetOperationsTest, SetUnionWithComparator) {
>+  std::vector<int> actual;
>+  absl::c_set_union(a_reversed_, b_reversed_, back_inserter(actual),
>+                    std::greater<int>());
>+  EXPECT_THAT(actual, ElementsAre(5, 3, 2, 1));
>+}
>+
>+TEST_F(SetOperationsTest, SetIntersection) {
>+  std::vector<int> actual;
>+  absl::c_set_intersection(a_, b_, back_inserter(actual));
>+  EXPECT_THAT(actual, ElementsAre(1, 3));
>+}
>+
>+TEST_F(SetOperationsTest, SetIntersectionWithComparator) {
>+  std::vector<int> actual;
>+  absl::c_set_intersection(a_reversed_, b_reversed_, back_inserter(actual),
>+                           std::greater<int>());
>+  EXPECT_THAT(actual, ElementsAre(3, 1));
>+}
>+
>+TEST_F(SetOperationsTest, SetDifference) {
>+  std::vector<int> actual;
>+  absl::c_set_difference(a_, b_, back_inserter(actual));
>+  EXPECT_THAT(actual, ElementsAre(2));
>+}
>+
>+TEST_F(SetOperationsTest, SetDifferenceWithComparator) {
>+  std::vector<int> actual;
>+  absl::c_set_difference(a_reversed_, b_reversed_, back_inserter(actual),
>+                         std::greater<int>());
>+  EXPECT_THAT(actual, ElementsAre(2));
>+}
>+
>+TEST_F(SetOperationsTest, SetSymmetricDifference) {
>+  std::vector<int> actual;
>+  absl::c_set_symmetric_difference(a_, b_, back_inserter(actual));
>+  EXPECT_THAT(actual, ElementsAre(2, 5));
>+}
>+
>+TEST_F(SetOperationsTest, SetSymmetricDifferenceWithComparator) {
>+  std::vector<int> actual;
>+  absl::c_set_symmetric_difference(a_reversed_, b_reversed_,
>+                                   back_inserter(actual), std::greater<int>());
>+  EXPECT_THAT(actual, ElementsAre(5, 2));
>+}
>+
>+TEST(HeapOperationsTest, WithoutComparator) {
>+  std::vector<int> heap = {1, 2, 3};
>+  EXPECT_FALSE(absl::c_is_heap(heap));
>+  absl::c_make_heap(heap);
>+  EXPECT_TRUE(absl::c_is_heap(heap));
>+  heap.push_back(4);
>+  EXPECT_EQ(3, absl::c_is_heap_until(heap) - heap.begin());
>+  absl::c_push_heap(heap);
>+  EXPECT_EQ(4, heap[0]);
>+  absl::c_pop_heap(heap);
>+  EXPECT_EQ(4, heap[3]);
>+  absl::c_make_heap(heap);
>+  absl::c_sort_heap(heap);
>+  EXPECT_THAT(heap, ElementsAre(1, 2, 3, 4));
>+  EXPECT_FALSE(absl::c_is_heap(heap));
>+}
>+
>+TEST(HeapOperationsTest, WithComparator) {
>+  using greater = std::greater<int>;
>+  std::vector<int> heap = {3, 2, 1};
>+  EXPECT_FALSE(absl::c_is_heap(heap, greater()));
>+  absl::c_make_heap(heap, greater());
>+  EXPECT_TRUE(absl::c_is_heap(heap, greater()));
>+  heap.push_back(0);
>+  EXPECT_EQ(3, absl::c_is_heap_until(heap, greater()) - heap.begin());
>+  absl::c_push_heap(heap, greater());
>+  EXPECT_EQ(0, heap[0]);
>+  absl::c_pop_heap(heap, greater());
>+  EXPECT_EQ(0, heap[3]);
>+  absl::c_make_heap(heap, greater());
>+  absl::c_sort_heap(heap, greater());
>+  EXPECT_THAT(heap, ElementsAre(3, 2, 1, 0));
>+  EXPECT_FALSE(absl::c_is_heap(heap, greater()));
>+}
>+
>+TEST(MutatingTest, PermutationOperations) {
>+  std::vector<int> initial = {1, 2, 3, 4};
>+  std::vector<int> permuted = initial;
>+
>+  absl::c_next_permutation(permuted);
>+  EXPECT_TRUE(absl::c_is_permutation(initial, permuted));
>+  EXPECT_TRUE(absl::c_is_permutation(initial, permuted, std::equal_to<int>()));
>+
>+  std::vector<int> permuted2 = initial;
>+  absl::c_prev_permutation(permuted2, std::greater<int>());
>+  EXPECT_EQ(permuted, permuted2);
>+
>+  absl::c_prev_permutation(permuted);
>+  EXPECT_EQ(initial, permuted);
>+}
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/algorithm/equal_benchmark.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/algorithm/equal_benchmark.cc
>new file mode 100644
>index 00000000000..19c0780ccd1
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/algorithm/equal_benchmark.cc
>@@ -0,0 +1,126 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include <cstdint>
>+#include <cstring>
>+
>+#include "benchmark/benchmark.h"
>+#include "absl/algorithm/algorithm.h"
>+
>+namespace {
>+
>+// The range of sequence sizes to benchmark.
>+constexpr int kMinBenchmarkSize = 1024;
>+constexpr int kMaxBenchmarkSize = 8 * 1024 * 1024;
>+
>+// A user-defined type for use in equality benchmarks. Note that we expect
>+// std::memcmp to win for this type: libstdc++'s std::equal only defers to
>+// memcmp for integral types. This is because it is not straightforward to
>+// guarantee that std::memcmp would produce a result "as-if" compared by
>+// operator== for other types (example gotchas: NaN floats, structs with
>+// padding).
>+struct EightBits {
>+  explicit EightBits(int /* unused */) : data(0) {}
>+  bool operator==(const EightBits& rhs) const { return data == rhs.data; }
>+  uint8_t data;
>+};
>+
>+template <typename T>
>+void BM_absl_equal_benchmark(benchmark::State& state) {
>+  std::vector<T> xs(state.range(0), T(0));
>+  std::vector<T> ys = xs;
>+  while (state.KeepRunning()) {
>+    const bool same = absl::equal(xs.begin(), xs.end(), ys.begin(), ys.end());
>+    benchmark::DoNotOptimize(same);
>+  }
>+}
>+
>+template <typename T>
>+void BM_std_equal_benchmark(benchmark::State& state) {
>+  std::vector<T> xs(state.range(0), T(0));
>+  std::vector<T> ys = xs;
>+  while (state.KeepRunning()) {
>+    const bool same = std::equal(xs.begin(), xs.end(), ys.begin());
>+    benchmark::DoNotOptimize(same);
>+  }
>+}
>+
>+template <typename T>
>+void BM_memcmp_benchmark(benchmark::State& state) {
>+  std::vector<T> xs(state.range(0), T(0));
>+  std::vector<T> ys = xs;
>+  while (state.KeepRunning()) {
>+    const bool same =
>+        std::memcmp(xs.data(), ys.data(), xs.size() * sizeof(T)) == 0;
>+    benchmark::DoNotOptimize(same);
>+  }
>+}
>+
>+// The expectation is that the compiler should be able to elide the equality
>+// comparison altogether for sufficiently simple types.
>+template <typename T>
>+void BM_absl_equal_self_benchmark(benchmark::State& state) {
>+  std::vector<T> xs(state.range(0), T(0));
>+  while (state.KeepRunning()) {
>+    const bool same = absl::equal(xs.begin(), xs.end(), xs.begin(), xs.end());
>+    benchmark::DoNotOptimize(same);
>+  }
>+}
>+
>+BENCHMARK_TEMPLATE(BM_absl_equal_benchmark, uint8_t)
>+    ->Range(kMinBenchmarkSize, kMaxBenchmarkSize);
>+BENCHMARK_TEMPLATE(BM_std_equal_benchmark, uint8_t)
>+    ->Range(kMinBenchmarkSize, kMaxBenchmarkSize);
>+BENCHMARK_TEMPLATE(BM_memcmp_benchmark, uint8_t)
>+    ->Range(kMinBenchmarkSize, kMaxBenchmarkSize);
>+BENCHMARK_TEMPLATE(BM_absl_equal_self_benchmark, uint8_t)
>+    ->Range(kMinBenchmarkSize, kMaxBenchmarkSize);
>+
>+BENCHMARK_TEMPLATE(BM_absl_equal_benchmark, uint16_t)
>+    ->Range(kMinBenchmarkSize, kMaxBenchmarkSize);
>+BENCHMARK_TEMPLATE(BM_std_equal_benchmark, uint16_t)
>+    ->Range(kMinBenchmarkSize, kMaxBenchmarkSize);
>+BENCHMARK_TEMPLATE(BM_memcmp_benchmark, uint16_t)
>+    ->Range(kMinBenchmarkSize, kMaxBenchmarkSize);
>+BENCHMARK_TEMPLATE(BM_absl_equal_self_benchmark, uint16_t)
>+    ->Range(kMinBenchmarkSize, kMaxBenchmarkSize);
>+
>+BENCHMARK_TEMPLATE(BM_absl_equal_benchmark, uint32_t)
>+    ->Range(kMinBenchmarkSize, kMaxBenchmarkSize);
>+BENCHMARK_TEMPLATE(BM_std_equal_benchmark, uint32_t)
>+    ->Range(kMinBenchmarkSize, kMaxBenchmarkSize);
>+BENCHMARK_TEMPLATE(BM_memcmp_benchmark, uint32_t)
>+    ->Range(kMinBenchmarkSize, kMaxBenchmarkSize);
>+BENCHMARK_TEMPLATE(BM_absl_equal_self_benchmark, uint32_t)
>+    ->Range(kMinBenchmarkSize, kMaxBenchmarkSize);
>+
>+BENCHMARK_TEMPLATE(BM_absl_equal_benchmark, uint64_t)
>+    ->Range(kMinBenchmarkSize, kMaxBenchmarkSize);
>+BENCHMARK_TEMPLATE(BM_std_equal_benchmark, uint64_t)
>+    ->Range(kMinBenchmarkSize, kMaxBenchmarkSize);
>+BENCHMARK_TEMPLATE(BM_memcmp_benchmark, uint64_t)
>+    ->Range(kMinBenchmarkSize, kMaxBenchmarkSize);
>+BENCHMARK_TEMPLATE(BM_absl_equal_self_benchmark, uint64_t)
>+    ->Range(kMinBenchmarkSize, kMaxBenchmarkSize);
>+
>+BENCHMARK_TEMPLATE(BM_absl_equal_benchmark, EightBits)
>+    ->Range(kMinBenchmarkSize, kMaxBenchmarkSize);
>+BENCHMARK_TEMPLATE(BM_std_equal_benchmark, EightBits)
>+    ->Range(kMinBenchmarkSize, kMaxBenchmarkSize);
>+BENCHMARK_TEMPLATE(BM_memcmp_benchmark, EightBits)
>+    ->Range(kMinBenchmarkSize, kMaxBenchmarkSize);
>+BENCHMARK_TEMPLATE(BM_absl_equal_self_benchmark, EightBits)
>+    ->Range(kMinBenchmarkSize, kMaxBenchmarkSize);
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/BUILD.bazel b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/BUILD.bazel
>new file mode 100644
>index 00000000000..06d092ebdfa
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/BUILD.bazel
>@@ -0,0 +1,423 @@
>+#
>+# Copyright 2017 The Abseil Authors.
>+#
>+# Licensed under the Apache License, Version 2.0 (the "License");
>+# you may not use this file except in compliance with the License.
>+# You may obtain a copy of the License at
>+#
>+#      http://www.apache.org/licenses/LICENSE-2.0
>+#
>+# Unless required by applicable law or agreed to in writing, software
>+# distributed under the License is distributed on an "AS IS" BASIS,
>+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+# See the License for the specific language governing permissions and
>+# limitations under the License.
>+#
>+
>+load(
>+    "//absl:copts.bzl",
>+    "ABSL_DEFAULT_COPTS",
>+    "ABSL_TEST_COPTS",
>+    "ABSL_EXCEPTIONS_FLAG",
>+)
>+
>+package(default_visibility = ["//visibility:public"])
>+
>+licenses(["notice"])  # Apache 2.0
>+
>+cc_library(
>+    name = "spinlock_wait",
>+    srcs = [
>+        "internal/spinlock_akaros.inc",
>+        "internal/spinlock_posix.inc",
>+        "internal/spinlock_wait.cc",
>+        "internal/spinlock_win32.inc",
>+    ],
>+    hdrs = [
>+        "internal/scheduling_mode.h",
>+        "internal/spinlock_wait.h",
>+    ],
>+    copts = ABSL_DEFAULT_COPTS,
>+    visibility = [
>+        "//absl/base:__pkg__",
>+    ],
>+    deps = [":core_headers"],
>+)
>+
>+cc_library(
>+    name = "config",
>+    hdrs = [
>+        "config.h",
>+        "policy_checks.h",
>+    ],
>+    copts = ABSL_DEFAULT_COPTS,
>+)
>+
>+cc_library(
>+    name = "dynamic_annotations",
>+    srcs = ["dynamic_annotations.cc"],
>+    hdrs = ["dynamic_annotations.h"],
>+    copts = ABSL_DEFAULT_COPTS,
>+    defines = ["__CLANG_SUPPORT_DYN_ANNOTATION__"],
>+)
>+
>+cc_library(
>+    name = "core_headers",
>+    hdrs = [
>+        "attributes.h",
>+        "macros.h",
>+        "optimization.h",
>+        "port.h",
>+        "thread_annotations.h",
>+    ],
>+    copts = ABSL_DEFAULT_COPTS,
>+    deps = [
>+        ":config",
>+        ":dynamic_annotations",
>+    ],
>+)
>+
>+cc_library(
>+    name = "malloc_internal",
>+    srcs = [
>+        "internal/low_level_alloc.cc",
>+    ],
>+    hdrs = [
>+        "internal/direct_mmap.h",
>+        "internal/low_level_alloc.h",
>+    ],
>+    copts = ABSL_DEFAULT_COPTS,
>+    visibility = [
>+        "//absl:__subpackages__",
>+    ],
>+    deps = [
>+        ":base",
>+        ":config",
>+        ":core_headers",
>+        ":dynamic_annotations",
>+        ":spinlock_wait",
>+    ],
>+)
>+
>+cc_library(
>+    name = "base_internal",
>+    hdrs = [
>+        "internal/hide_ptr.h",
>+        "internal/identity.h",
>+        "internal/inline_variable.h",
>+        "internal/invoke.h",
>+    ],
>+    copts = ABSL_DEFAULT_COPTS,
>+    visibility = [
>+        "//absl:__subpackages__",
>+    ],
>+)
>+
>+cc_library(
>+    name = "base",
>+    srcs = [
>+        "internal/cycleclock.cc",
>+        "internal/raw_logging.cc",
>+        "internal/spinlock.cc",
>+        "internal/sysinfo.cc",
>+        "internal/thread_identity.cc",
>+        "internal/unscaledcycleclock.cc",
>+    ],
>+    hdrs = [
>+        "call_once.h",
>+        "casts.h",
>+        "internal/atomic_hook.h",
>+        "internal/cycleclock.h",
>+        "internal/low_level_scheduling.h",
>+        "internal/per_thread_tls.h",
>+        "internal/raw_logging.h",
>+        "internal/spinlock.h",
>+        "internal/sysinfo.h",
>+        "internal/thread_identity.h",
>+        "internal/tsan_mutex_interface.h",
>+        "internal/unscaledcycleclock.h",
>+        "log_severity.h",
>+    ],
>+    copts = ABSL_DEFAULT_COPTS,
>+    deps = [
>+        ":base_internal",
>+        ":config",
>+        ":core_headers",
>+        ":dynamic_annotations",
>+        ":spinlock_wait",
>+    ],
>+)
>+
>+cc_test(
>+    name = "atomic_hook_test",
>+    size = "small",
>+    srcs = ["internal/atomic_hook_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    deps = [
>+        ":base",
>+        ":core_headers",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "bit_cast_test",
>+    size = "small",
>+    srcs = [
>+        "bit_cast_test.cc",
>+    ],
>+    copts = ABSL_TEST_COPTS,
>+    deps = [
>+        ":base",
>+        ":core_headers",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_library(
>+    name = "throw_delegate",
>+    srcs = ["internal/throw_delegate.cc"],
>+    hdrs = ["internal/throw_delegate.h"],
>+    copts = ABSL_DEFAULT_COPTS + ABSL_EXCEPTIONS_FLAG,
>+    visibility = [
>+        "//absl:__subpackages__",
>+    ],
>+    deps = [
>+        ":base",
>+        ":config",
>+        ":core_headers",
>+    ],
>+)
>+
>+cc_test(
>+    name = "throw_delegate_test",
>+    srcs = ["throw_delegate_test.cc"],
>+    copts = ABSL_TEST_COPTS + ABSL_EXCEPTIONS_FLAG,
>+    deps = [
>+        ":throw_delegate",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_library(
>+    name = "exception_testing",
>+    testonly = 1,
>+    hdrs = ["internal/exception_testing.h"],
>+    copts = ABSL_TEST_COPTS,
>+    visibility = [
>+        "//absl:__subpackages__",
>+    ],
>+    deps = [
>+        ":config",
>+        "@com_google_googletest//:gtest",
>+    ],
>+)
>+
>+cc_library(
>+    name = "pretty_function",
>+    hdrs = ["internal/pretty_function.h"],
>+    visibility = ["//absl:__subpackages__"],
>+)
>+
>+cc_library(
>+    name = "exception_safety_testing",
>+    testonly = 1,
>+    srcs = ["internal/exception_safety_testing.cc"],
>+    hdrs = ["internal/exception_safety_testing.h"],
>+    copts = ABSL_TEST_COPTS + ABSL_EXCEPTIONS_FLAG,
>+    deps = [
>+        ":base",
>+        ":config",
>+        ":pretty_function",
>+        "//absl/memory",
>+        "//absl/meta:type_traits",
>+        "//absl/strings",
>+        "//absl/types:optional",
>+        "@com_google_googletest//:gtest",
>+    ],
>+)
>+
>+cc_test(
>+    name = "exception_safety_testing_test",
>+    srcs = ["exception_safety_testing_test.cc"],
>+    copts = ABSL_TEST_COPTS + ABSL_EXCEPTIONS_FLAG,
>+    deps = [
>+        ":exception_safety_testing",
>+        "//absl/memory",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "inline_variable_test",
>+    size = "small",
>+    srcs = [
>+        "inline_variable_test.cc",
>+        "inline_variable_test_a.cc",
>+        "inline_variable_test_b.cc",
>+        "internal/inline_variable_testing.h",
>+    ],
>+    copts = ABSL_TEST_COPTS,
>+    deps = [
>+        ":base_internal",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "invoke_test",
>+    size = "small",
>+    srcs = ["invoke_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    deps = [
>+        ":base_internal",
>+        "//absl/memory",
>+        "//absl/strings",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+# Common test library made available for use in non-absl code that overrides
>+# AbslInternalSpinLockDelay and AbslInternalSpinLockWake.
>+cc_library(
>+    name = "spinlock_test_common",
>+    testonly = 1,
>+    srcs = ["spinlock_test_common.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    deps = [
>+        ":base",
>+        ":core_headers",
>+        ":spinlock_wait",
>+        "//absl/synchronization",
>+        "@com_google_googletest//:gtest",
>+    ],
>+    alwayslink = 1,
>+)
>+
>+cc_test(
>+    name = "spinlock_test",
>+    size = "medium",
>+    srcs = ["spinlock_test_common.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    deps = [
>+        ":base",
>+        ":core_headers",
>+        ":spinlock_wait",
>+        "//absl/synchronization",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_library(
>+    name = "endian",
>+    hdrs = [
>+        "internal/endian.h",
>+        "internal/unaligned_access.h",
>+    ],
>+    copts = ABSL_DEFAULT_COPTS,
>+    deps = [
>+        ":config",
>+        ":core_headers",
>+    ],
>+)
>+
>+cc_test(
>+    name = "endian_test",
>+    srcs = ["internal/endian_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    deps = [
>+        ":base",
>+        ":config",
>+        ":endian",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "config_test",
>+    srcs = ["config_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    deps = [
>+        ":config",
>+        "//absl/synchronization:thread_pool",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "call_once_test",
>+    srcs = ["call_once_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    deps = [
>+        ":base",
>+        ":core_headers",
>+        "//absl/synchronization",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "raw_logging_test",
>+    srcs = ["raw_logging_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    deps = [
>+        ":base",
>+        "//absl/strings",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "sysinfo_test",
>+    size = "small",
>+    srcs = ["internal/sysinfo_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    deps = [
>+        ":base",
>+        "//absl/synchronization",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "low_level_alloc_test",
>+    size = "small",
>+    srcs = ["internal/low_level_alloc_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    linkopts = select({
>+        "//absl:windows": [],
>+        "//conditions:default": ["-pthread"],
>+    }),
>+    tags = ["no_test_ios_x86_64"],
>+    deps = [":malloc_internal"],
>+)
>+
>+cc_test(
>+    name = "thread_identity_test",
>+    size = "small",
>+    srcs = ["internal/thread_identity_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    linkopts = select({
>+        "//absl:windows": [],
>+        "//conditions:default": ["-pthread"],
>+    }),
>+    deps = [
>+        ":base",
>+        ":core_headers",
>+        "//absl/synchronization",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "thread_identity_benchmark",
>+    srcs = ["internal/thread_identity_benchmark.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    tags = ["benchmark"],
>+    visibility = ["//visibility:private"],
>+    deps = [
>+        ":base",
>+        "//absl/synchronization",
>+        "@com_github_google_benchmark//:benchmark_main",
>+    ],
>+)
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/BUILD.gn
>new file mode 100644
>index 00000000000..a7146564622
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/BUILD.gn
>@@ -0,0 +1,298 @@
>+# Copyright 2018 The Chromium Authors. All rights reserved.
>+# Use of this source code is governed by a BSD-style license that can be
>+# found in the LICENSE file.
>+
>+import("//build_overrides/build.gni")
>+
>+if (build_with_chromium) {
>+  visibility = [
>+    "//third_party/webrtc/*",
>+    "//third_party/abseil-cpp/*",
>+    "//third_party/googletest:gtest",
>+  ]
>+} else {
>+  visibility = [ "*" ]
>+}
>+
>+source_set("spinlock_wait") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  sources = [
>+    "internal/spinlock_akaros.inc",
>+    "internal/spinlock_posix.inc",
>+    "internal/spinlock_wait.cc",
>+    "internal/spinlock_win32.inc",
>+  ]
>+  public = [
>+    "internal/scheduling_mode.h",
>+    "internal/spinlock_wait.h",
>+  ]
>+  deps = [
>+    ":core_headers",
>+  ]
>+  visibility = []
>+  visibility += [ "../base:*" ]
>+}
>+
>+source_set("config") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  public = [
>+    "config.h",
>+    "policy_checks.h",
>+  ]
>+}
>+
>+config("clang_support_dynamic_annotations") {
>+  cflags_cc = [ "-D__CLANG_SUPPORT_DYN_ANNOTATION__" ]
>+}
>+
>+source_set("dynamic_annotations") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [
>+    ":clang_support_dynamic_annotations",
>+    "//third_party/abseil-cpp:absl_include_config",
>+  ]
>+  sources = [
>+    "dynamic_annotations.cc",
>+  ]
>+  public = [
>+    "dynamic_annotations.h",
>+  ]
>+  # Abseil's dynamic annotations are only visible inside Abseil because
>+  # their usage is deprecated in Chromium (see README.chromium for more info).
>+  visibility = []
>+  visibility = [ "../*" ]
>+}
>+
>+source_set("core_headers") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  public = [
>+    "attributes.h",
>+    "macros.h",
>+    "optimization.h",
>+    "port.h",
>+    "thread_annotations.h",
>+  ]
>+  deps = [
>+    ":config",
>+    ":dynamic_annotations",
>+  ]
>+}
>+
>+source_set("malloc_internal") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  sources = [
>+    "internal/low_level_alloc.cc",
>+  ]
>+  public = [
>+    "internal/direct_mmap.h",
>+    "internal/low_level_alloc.h",
>+  ]
>+  deps = [
>+    ":base",
>+    ":config",
>+    ":core_headers",
>+    ":dynamic_annotations",
>+    ":spinlock_wait",
>+  ]
>+  visibility = []
>+  visibility += [ "../*" ]
>+}
>+
>+source_set("base_internal") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  public = [
>+    "internal/hide_ptr.h",
>+    "internal/identity.h",
>+    "internal/inline_variable.h",
>+    "internal/invoke.h",
>+  ]
>+  visibility = []
>+  visibility += [ "../*" ]
>+}
>+
>+source_set("base") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  sources = [
>+    "internal/cycleclock.cc",
>+    "internal/raw_logging.cc",
>+    "internal/spinlock.cc",
>+    "internal/sysinfo.cc",
>+    "internal/thread_identity.cc",
>+    "internal/unscaledcycleclock.cc",
>+  ]
>+  public = [
>+    "call_once.h",
>+    "casts.h",
>+    "internal/atomic_hook.h",
>+    "internal/cycleclock.h",
>+    "internal/low_level_scheduling.h",
>+    "internal/per_thread_tls.h",
>+    "internal/raw_logging.h",
>+    "internal/spinlock.h",
>+    "internal/sysinfo.h",
>+    "internal/thread_identity.h",
>+    "internal/tsan_mutex_interface.h",
>+    "internal/unscaledcycleclock.h",
>+    "log_severity.h",
>+  ]
>+  deps = [
>+    ":base_internal",
>+    ":config",
>+    ":core_headers",
>+    ":dynamic_annotations",
>+    ":spinlock_wait",
>+  ]
>+}
>+
>+source_set("throw_delegate") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  sources = [
>+    "internal/throw_delegate.cc",
>+  ]
>+  public = [
>+    "internal/throw_delegate.h",
>+  ]
>+  deps = [
>+    ":base",
>+    ":config",
>+    ":core_headers",
>+  ]
>+  visibility = []
>+  visibility += [ "../*" ]
>+}
>+
>+source_set("exception_testing") {
>+  testonly = true
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_test_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  public = [
>+    "internal/exception_testing.h",
>+  ]
>+  deps = [
>+    ":config",
>+  ]
>+  visibility = []
>+  visibility += [ "../*" ]
>+}
>+
>+source_set("pretty_function") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  public = [
>+    "internal/pretty_function.h",
>+  ]
>+  visibility = []
>+  visibility += [ "../*" ]
>+}
>+
>+# TODO(mbonadei): This target throws by design. We should probably
>+# just remove it.
>+# source_set("exception_safety_testing") {
>+#   testonly = true
>+#   configs -= [ "//build/config/compiler:chromium_code" ]
>+#   configs += [
>+#     "//build/config/compiler:no_chromium_code",
>+#     "//third_party/abseil-cpp:absl_test_cflags_cc",
>+#   ]
>+#   public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+#   sources = [
>+#     "internal/exception_safety_testing.cc",
>+#   ]
>+#   public = [
>+#     "internal/exception_safety_testing.h",
>+#   ]
>+#   deps = [
>+#     ":base",
>+#     ":config",
>+#     ":pretty_function",
>+#     "../memory",
>+#     "../meta:type_traits",
>+#     "../strings",
>+#     "../types:optional",
>+#     "//testing/gtest",
>+#   ]
>+# }
>+
>+source_set("spinlock_test_common") {
>+  testonly = true
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_test_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  sources = [
>+    "spinlock_test_common.cc",
>+  ]
>+  deps = [
>+    ":base",
>+    ":core_headers",
>+    ":spinlock_wait",
>+    "../synchronization",
>+    "//testing/gtest",
>+  ]
>+}
>+
>+source_set("endian") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  public = [
>+    "internal/endian.h",
>+    "internal/unaligned_access.h",
>+  ]
>+  deps = [
>+    ":config",
>+    ":core_headers",
>+  ]
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/CMakeLists.txt b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/CMakeLists.txt
>new file mode 100644
>index 00000000000..01d2af085f5
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/CMakeLists.txt
>@@ -0,0 +1,386 @@
>+#
>+# Copyright 2017 The Abseil Authors.
>+#
>+# Licensed under the Apache License, Version 2.0 (the "License");
>+# you may not use this file except in compliance with the License.
>+# You may obtain a copy of the License at
>+#
>+#    http://www.apache.org/licenses/LICENSE-2.0
>+#
>+# Unless required by applicable law or agreed to in writing, software
>+# distributed under the License is distributed on an "AS IS" BASIS,
>+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+# See the License for the specific language governing permissions and
>+# limitations under the License.
>+#
>+
>+list(APPEND BASE_PUBLIC_HEADERS
>+  "attributes.h"
>+  "call_once.h"
>+  "casts.h"
>+  "config.h"
>+  "dynamic_annotations.h"
>+  "log_severity.h"
>+  "macros.h"
>+  "optimization.h"
>+  "policy_checks.h"
>+  "port.h"
>+  "thread_annotations.h"
>+)
>+
>+
>+list(APPEND BASE_INTERNAL_HEADERS
>+  "internal/atomic_hook.h"
>+  "internal/cycleclock.h"
>+  "internal/direct_mmap.h"
>+  "internal/endian.h"
>+  "internal/exception_testing.h"
>+  "internal/exception_safety_testing.h"
>+  "internal/hide_ptr.h"
>+  "internal/identity.h"
>+  "internal/invoke.h"
>+  "internal/inline_variable.h"
>+  "internal/low_level_alloc.h"
>+  "internal/low_level_scheduling.h"
>+  "internal/per_thread_tls.h"
>+  "internal/pretty_function.h"
>+  "internal/raw_logging.h"
>+  "internal/scheduling_mode.h"
>+  "internal/spinlock.h"
>+  "internal/spinlock_wait.h"
>+  "internal/sysinfo.h"
>+  "internal/thread_identity.h"
>+  "internal/throw_delegate.h"
>+  "internal/tsan_mutex_interface.h"
>+  "internal/unaligned_access.h"
>+  "internal/unscaledcycleclock.h"
>+)
>+
>+
>+# absl_base main library
>+list(APPEND BASE_SRC
>+  "internal/cycleclock.cc"
>+  "internal/raw_logging.cc"
>+  "internal/spinlock.cc"
>+  "internal/sysinfo.cc"
>+  "internal/thread_identity.cc"
>+  "internal/unscaledcycleclock.cc"
>+  "internal/low_level_alloc.cc"
>+  ${BASE_PUBLIC_HEADERS}
>+  ${BASE_INTERNAL_HEADERS}
>+)
>+
>+absl_library(
>+  TARGET
>+    absl_base
>+  SOURCES
>+    ${BASE_SRC}
>+  PUBLIC_LIBRARIES
>+    absl_dynamic_annotations
>+    absl_spinlock_wait
>+  EXPORT_NAME
>+    base
>+)
>+
>+# throw delegate library
>+set(THROW_DELEGATE_SRC "internal/throw_delegate.cc")
>+
>+absl_library(
>+  TARGET
>+    absl_throw_delegate
>+  SOURCES
>+    ${THROW_DELEGATE_SRC}
>+  PUBLIC_LIBRARIES
>+    ${THROW_DELEGATE_PUBLIC_LIBRARIES}
>+  PRIVATE_COMPILE_FLAGS
>+    ${ABSL_EXCEPTIONS_FLAG}
>+  EXPORT_NAME
>+    throw_delegate
>+)
>+
>+if(BUILD_TESTING)
>+  # exception-safety testing library
>+  set(EXCEPTION_SAFETY_TESTING_SRC
>+    "internal/exception_safety_testing.h"
>+    "internal/exception_safety_testing.cc"
>+  )
>+  set(EXCEPTION_SAFETY_TESTING_PUBLIC_LIBRARIES
>+    ${ABSL_TEST_COMMON_LIBRARIES}
>+    absl::base
>+    absl::memory
>+    absl::meta
>+    absl::strings
>+    absl::optional
>+    gtest
>+  )
>+
>+absl_library(
>+  TARGET
>+    absl_base_internal_exception_safety_testing
>+  SOURCES
>+    ${EXCEPTION_SAFETY_TESTING_SRC}
>+  PUBLIC_LIBRARIES
>+    ${EXCEPTION_SAFETY_TESTING_PUBLIC_LIBRARIES}
>+  PRIVATE_COMPILE_FLAGS
>+    ${ABSL_EXCEPTIONS_FLAG}
>+)
>+endif()
>+
>+
>+# dynamic_annotations library
>+set(DYNAMIC_ANNOTATIONS_SRC "dynamic_annotations.cc")
>+
>+absl_library(
>+  TARGET
>+    absl_dynamic_annotations
>+  SOURCES
>+    ${DYNAMIC_ANNOTATIONS_SRC}
>+)
>+
>+
>+# spinlock_wait library
>+set(SPINLOCK_WAIT_SRC "internal/spinlock_wait.cc")
>+
>+absl_library(
>+  TARGET
>+    absl_spinlock_wait
>+  SOURCES
>+    ${SPINLOCK_WAIT_SRC}
>+)
>+
>+
>+# malloc_internal library
>+list(APPEND MALLOC_INTERNAL_SRC
>+  "internal/low_level_alloc.cc"
>+)
>+
>+absl_library(
>+  TARGET
>+    absl_malloc_internal
>+  SOURCES
>+    ${MALLOC_INTERNAL_SRC}
>+  PUBLIC_LIBRARIES
>+    absl_dynamic_annotations
>+)
>+
>+
>+
>+#
>+## TESTS
>+#
>+
>+# call once test
>+set(ATOMIC_HOOK_TEST_SRC "internal/atomic_hook_test.cc")
>+set(ATOMIC_HOOK_TEST_PUBLIC_LIBRARIES absl::base)
>+
>+absl_test(
>+  TARGET
>+    atomic_hook_test
>+  SOURCES
>+    ${ATOMIC_HOOK_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${ATOMIC_HOOK_TEST_PUBLIC_LIBRARIES}
>+)
>+
>+
>+# call once test
>+set(CALL_ONCE_TEST_SRC "call_once_test.cc")
>+set(CALL_ONCE_TEST_PUBLIC_LIBRARIES absl::base absl::synchronization)
>+
>+absl_test(
>+  TARGET
>+    call_once_test
>+  SOURCES
>+    ${CALL_ONCE_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${CALL_ONCE_TEST_PUBLIC_LIBRARIES}
>+)
>+
>+
>+# test bit_cast_test
>+set(BIT_CAST_TEST_SRC "bit_cast_test.cc")
>+
>+absl_test(
>+  TARGET
>+    bit_cast_test
>+  SOURCES
>+    ${BIT_CAST_TEST_SRC}
>+)
>+
>+
>+# test absl_throw_delegate_test
>+set(THROW_DELEGATE_TEST_SRC "throw_delegate_test.cc")
>+set(THROW_DELEGATE_TEST_PUBLIC_LIBRARIES absl::base absl_throw_delegate)
>+
>+absl_test(
>+  TARGET
>+    throw_delegate_test
>+  SOURCES
>+    ${THROW_DELEGATE_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${THROW_DELEGATE_TEST_PUBLIC_LIBRARIES}
>+)
>+
>+
>+# test invoke_test
>+set(INVOKE_TEST_SRC "invoke_test.cc")
>+set(INVOKE_TEST_PUBLIC_LIBRARIES absl::strings)
>+
>+absl_test(
>+  TARGET
>+    invoke_test
>+  SOURCES
>+    ${INVOKE_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${INVOKE_TEST_PUBLIC_LIBRARIES}
>+)
>+
>+
>+# test inline_variable_test
>+list(APPEND INLINE_VARIABLE_TEST_SRC
>+  "internal/inline_variable_testing.h"
>+  "inline_variable_test.cc"
>+  "inline_variable_test_a.cc"
>+  "inline_variable_test_b.cc"
>+)
>+
>+set(INLINE_VARIABLE_TEST_PUBLIC_LIBRARIES absl::base)
>+
>+absl_test(
>+  TARGET
>+    inline_variable_test
>+  SOURCES
>+    ${INLINE_VARIABLE_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${INLINE_VARIABLE_TEST_PUBLIC_LIBRARIES}
>+)
>+
>+
>+# test spinlock_test_common
>+set(SPINLOCK_TEST_COMMON_SRC "spinlock_test_common.cc")
>+set(SPINLOCK_TEST_COMMON_PUBLIC_LIBRARIES absl::base absl::synchronization)
>+
>+absl_test(
>+  TARGET
>+    spinlock_test_common
>+  SOURCES
>+    ${SPINLOCK_TEST_COMMON_SRC}
>+  PUBLIC_LIBRARIES
>+    ${SPINLOCK_TEST_COMMON_PUBLIC_LIBRARIES}
>+)
>+
>+
>+# test spinlock_test
>+set(SPINLOCK_TEST_SRC "spinlock_test_common.cc")
>+set(SPINLOCK_TEST_PUBLIC_LIBRARIES absl::base absl::synchronization)
>+
>+absl_test(
>+  TARGET
>+    spinlock_test
>+  SOURCES
>+    ${SPINLOCK_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${SPINLOCK_TEST_PUBLIC_LIBRARIES}
>+)
>+
>+
>+# test endian_test
>+set(ENDIAN_TEST_SRC "internal/endian_test.cc")
>+
>+absl_test(
>+  TARGET
>+    endian_test
>+  SOURCES
>+    ${ENDIAN_TEST_SRC}
>+)
>+
>+
>+# test config_test
>+set(CONFIG_TEST_SRC "config_test.cc")
>+set(CONFIG_TEST_PUBLIC_LIBRARIES absl::base absl::synchronization)
>+absl_test(
>+  TARGET
>+    config_test
>+  SOURCES
>+    ${CONFIG_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${CONFIG_TEST_PUBLIC_LIBRARIES}
>+)
>+
>+
>+# test raw_logging_test
>+set(RAW_LOGGING_TEST_SRC "raw_logging_test.cc")
>+set(RAW_LOGGING_TEST_PUBLIC_LIBRARIES absl::base absl::strings)
>+
>+absl_test(
>+  TARGET
>+    raw_logging_test
>+  SOURCES
>+    ${RAW_LOGGING_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${RAW_LOGGING_TEST_PUBLIC_LIBRARIES}
>+)
>+
>+
>+# test sysinfo_test
>+set(SYSINFO_TEST_SRC "internal/sysinfo_test.cc")
>+set(SYSINFO_TEST_PUBLIC_LIBRARIES absl::base absl::synchronization)
>+
>+absl_test(
>+  TARGET
>+    sysinfo_test
>+  SOURCES
>+    ${SYSINFO_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${SYSINFO_TEST_PUBLIC_LIBRARIES}
>+)
>+
>+
>+# test low_level_alloc_test
>+set(LOW_LEVEL_ALLOC_TEST_SRC "internal/low_level_alloc_test.cc")
>+set(LOW_LEVEL_ALLOC_TEST_PUBLIC_LIBRARIES absl::base)
>+
>+absl_test(
>+  TARGET
>+    low_level_alloc_test
>+  SOURCES
>+    ${LOW_LEVEL_ALLOC_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${LOW_LEVEL_ALLOC_TEST_PUBLIC_LIBRARIES}
>+)
>+
>+
>+# test thread_identity_test
>+set(THREAD_IDENTITY_TEST_SRC "internal/thread_identity_test.cc")
>+set(THREAD_IDENTITY_TEST_PUBLIC_LIBRARIES absl::base absl::synchronization)
>+
>+absl_test(
>+  TARGET
>+    thread_identity_test
>+  SOURCES
>+    ${THREAD_IDENTITY_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${THREAD_IDENTITY_TEST_PUBLIC_LIBRARIES}
>+)
>+
>+#test exceptions_safety_testing_test
>+set(EXCEPTION_SAFETY_TESTING_TEST_SRC "exception_safety_testing_test.cc")
>+set(EXCEPTION_SAFETY_TESTING_TEST_PUBLIC_LIBRARIES
>+  absl::base
>+  absl_base_internal_exception_safety_testing
>+  absl::memory
>+  absl::meta
>+  absl::strings
>+  absl::optional
>+)
>+
>+absl_test(
>+  TARGET
>+    absl_exception_safety_testing_test
>+  SOURCES
>+    ${EXCEPTION_SAFETY_TESTING_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${EXCEPTION_SAFETY_TESTING_TEST_PUBLIC_LIBRARIES}
>+  PRIVATE_COMPILE_FLAGS
>+    ${ABSL_EXCEPTIONS_FLAG}
>+)
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/attributes.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/attributes.h
>new file mode 100644
>index 00000000000..b1883b6d752
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/attributes.h
>@@ -0,0 +1,568 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// This header file defines macros for declaring attributes for functions,
>+// types, and variables.
>+//
>+// These macros are used within Abseil and allow the compiler to optimize, where
>+// applicable, certain function calls.
>+//
>+// This file is used for both C and C++!
>+//
>+// Most macros here are exposing GCC or Clang features, and are stubbed out for
>+// other compilers.
>+//
>+// GCC attributes documentation:
>+//   https://gcc.gnu.org/onlinedocs/gcc-4.7.0/gcc/Function-Attributes.html
>+//   https://gcc.gnu.org/onlinedocs/gcc-4.7.0/gcc/Variable-Attributes.html
>+//   https://gcc.gnu.org/onlinedocs/gcc-4.7.0/gcc/Type-Attributes.html
>+//
>+// Most attributes in this file are already supported by GCC 4.7. However, some
>+// of them are not supported in older version of Clang. Thus, we check
>+// `__has_attribute()` first. If the check fails, we check if we are on GCC and
>+// assume the attribute exists on GCC (which is verified on GCC 4.7).
>+//
>+// -----------------------------------------------------------------------------
>+// Sanitizer Attributes
>+// -----------------------------------------------------------------------------
>+//
>+// Sanitizer-related attributes are not "defined" in this file (and indeed
>+// are not defined as such in any file). To utilize the following
>+// sanitizer-related attributes within your builds, define the following macros
>+// within your build using a `-D` flag, along with the given value for
>+// `-fsanitize`:
>+//
>+//   * `ADDRESS_SANITIZER` + `-fsanitize=address` (Clang, GCC 4.8)
>+//   * `MEMORY_SANITIZER` + `-fsanitize=memory` (Clang-only)
>+//   * `THREAD_SANITIZER + `-fsanitize=thread` (Clang, GCC 4.8+)
>+//   * `UNDEFINED_BEHAVIOR_SANITIZER` + `-fsanitize=undefined` (Clang, GCC 4.9+)
>+//   * `CONTROL_FLOW_INTEGRITY` + -fsanitize=cfi (Clang-only)
>+//
>+// Example:
>+//
>+//   // Enable branches in the Abseil code that are tagged for ASan:
>+//   $ bazel build --copt=-DADDRESS_SANITIZER --copt=-fsanitize=address
>+//     --linkopt=-fsanitize=address *target*
>+//
>+// Since these macro names are only supported by GCC and Clang, we only check
>+// for `__GNUC__` (GCC or Clang) and the above macros.
>+#ifndef ABSL_BASE_ATTRIBUTES_H_
>+#define ABSL_BASE_ATTRIBUTES_H_
>+
>+// ABSL_HAVE_ATTRIBUTE
>+//
>+// A function-like feature checking macro that is a wrapper around
>+// `__has_attribute`, which is defined by GCC 5+ and Clang and evaluates to a
>+// nonzero constant integer if the attribute is supported or 0 if not.
>+//
>+// It evaluates to zero if `__has_attribute` is not defined by the compiler.
>+//
>+// GCC: https://gcc.gnu.org/gcc-5/changes.html
>+// Clang: https://clang.llvm.org/docs/LanguageExtensions.html
>+#ifdef __has_attribute
>+#define ABSL_HAVE_ATTRIBUTE(x) __has_attribute(x)
>+#else
>+#define ABSL_HAVE_ATTRIBUTE(x) 0
>+#endif
>+
>+// ABSL_HAVE_CPP_ATTRIBUTE
>+//
>+// A function-like feature checking macro that accepts C++11 style attributes.
>+// It's a wrapper around `__has_cpp_attribute`, defined by ISO C++ SD-6
>+// (http://en.cppreference.com/w/cpp/experimental/feature_test). If we don't
>+// find `__has_cpp_attribute`, will evaluate to 0.
>+#if defined(__cplusplus) && defined(__has_cpp_attribute)
>+// NOTE: requiring __cplusplus above should not be necessary, but
>+// works around https://bugs.llvm.org/show_bug.cgi?id=23435.
>+#define ABSL_HAVE_CPP_ATTRIBUTE(x) __has_cpp_attribute(x)
>+#else
>+#define ABSL_HAVE_CPP_ATTRIBUTE(x) 0
>+#endif
>+
>+// -----------------------------------------------------------------------------
>+// Function Attributes
>+// -----------------------------------------------------------------------------
>+//
>+// GCC: https://gcc.gnu.org/onlinedocs/gcc/Function-Attributes.html
>+// Clang: https://clang.llvm.org/docs/AttributeReference.html
>+
>+// ABSL_PRINTF_ATTRIBUTE
>+// ABSL_SCANF_ATTRIBUTE
>+//
>+// Tells the compiler to perform `printf` format std::string checking if the
>+// compiler supports it; see the 'format' attribute in
>+// <http://gcc.gnu.org/onlinedocs/gcc-4.7.0/gcc/Function-Attributes.html>.
>+//
>+// Note: As the GCC manual states, "[s]ince non-static C++ methods
>+// have an implicit 'this' argument, the arguments of such methods
>+// should be counted from two, not one."
>+#if ABSL_HAVE_ATTRIBUTE(format) || (defined(__GNUC__) && !defined(__clang__))
>+#define ABSL_PRINTF_ATTRIBUTE(string_index, first_to_check) \
>+  __attribute__((__format__(__printf__, string_index, first_to_check)))
>+#define ABSL_SCANF_ATTRIBUTE(string_index, first_to_check) \
>+  __attribute__((__format__(__scanf__, string_index, first_to_check)))
>+#else
>+#define ABSL_PRINTF_ATTRIBUTE(string_index, first_to_check)
>+#define ABSL_SCANF_ATTRIBUTE(string_index, first_to_check)
>+#endif
>+
>+// ABSL_ATTRIBUTE_ALWAYS_INLINE
>+// ABSL_ATTRIBUTE_NOINLINE
>+//
>+// Forces functions to either inline or not inline. Introduced in gcc 3.1.
>+#if ABSL_HAVE_ATTRIBUTE(always_inline) || \
>+    (defined(__GNUC__) && !defined(__clang__))
>+#define ABSL_ATTRIBUTE_ALWAYS_INLINE __attribute__((always_inline))
>+#define ABSL_HAVE_ATTRIBUTE_ALWAYS_INLINE 1
>+#else
>+#define ABSL_ATTRIBUTE_ALWAYS_INLINE
>+#endif
>+
>+#if ABSL_HAVE_ATTRIBUTE(noinline) || (defined(__GNUC__) && !defined(__clang__))
>+#define ABSL_ATTRIBUTE_NOINLINE __attribute__((noinline))
>+#define ABSL_HAVE_ATTRIBUTE_NOINLINE 1
>+#else
>+#define ABSL_ATTRIBUTE_NOINLINE
>+#endif
>+
>+// ABSL_ATTRIBUTE_NO_TAIL_CALL
>+//
>+// Prevents the compiler from optimizing away stack frames for functions which
>+// end in a call to another function.
>+#if ABSL_HAVE_ATTRIBUTE(disable_tail_calls)
>+#define ABSL_HAVE_ATTRIBUTE_NO_TAIL_CALL 1
>+#define ABSL_ATTRIBUTE_NO_TAIL_CALL __attribute__((disable_tail_calls))
>+#elif defined(__GNUC__) && !defined(__clang__)
>+#define ABSL_HAVE_ATTRIBUTE_NO_TAIL_CALL 1
>+#define ABSL_ATTRIBUTE_NO_TAIL_CALL \
>+  __attribute__((optimize("no-optimize-sibling-calls")))
>+#else
>+#define ABSL_ATTRIBUTE_NO_TAIL_CALL
>+#define ABSL_HAVE_ATTRIBUTE_NO_TAIL_CALL 0
>+#endif
>+
>+// ABSL_ATTRIBUTE_WEAK
>+//
>+// Tags a function as weak for the purposes of compilation and linking.
>+#if ABSL_HAVE_ATTRIBUTE(weak) || (defined(__GNUC__) && !defined(__clang__))
>+#undef ABSL_ATTRIBUTE_WEAK
>+#define ABSL_ATTRIBUTE_WEAK __attribute__((weak))
>+#define ABSL_HAVE_ATTRIBUTE_WEAK 1
>+#else
>+#define ABSL_ATTRIBUTE_WEAK
>+#define ABSL_HAVE_ATTRIBUTE_WEAK 0
>+#endif
>+
>+// ABSL_ATTRIBUTE_NONNULL
>+//
>+// Tells the compiler either (a) that a particular function parameter
>+// should be a non-null pointer, or (b) that all pointer arguments should
>+// be non-null.
>+//
>+// Note: As the GCC manual states, "[s]ince non-static C++ methods
>+// have an implicit 'this' argument, the arguments of such methods
>+// should be counted from two, not one."
>+//
>+// Args are indexed starting at 1.
>+//
>+// For non-static class member functions, the implicit `this` argument
>+// is arg 1, and the first explicit argument is arg 2. For static class member
>+// functions, there is no implicit `this`, and the first explicit argument is
>+// arg 1.
>+//
>+// Example:
>+//
>+//   /* arg_a cannot be null, but arg_b can */
>+//   void Function(void* arg_a, void* arg_b) ABSL_ATTRIBUTE_NONNULL(1);
>+//
>+//   class C {
>+//     /* arg_a cannot be null, but arg_b can */
>+//     void Method(void* arg_a, void* arg_b) ABSL_ATTRIBUTE_NONNULL(2);
>+//
>+//     /* arg_a cannot be null, but arg_b can */
>+//     static void StaticMethod(void* arg_a, void* arg_b)
>+//     ABSL_ATTRIBUTE_NONNULL(1);
>+//   };
>+//
>+// If no arguments are provided, then all pointer arguments should be non-null.
>+//
>+//  /* No pointer arguments may be null. */
>+//  void Function(void* arg_a, void* arg_b, int arg_c) ABSL_ATTRIBUTE_NONNULL();
>+//
>+// NOTE: The GCC nonnull attribute actually accepts a list of arguments, but
>+// ABSL_ATTRIBUTE_NONNULL does not.
>+#if ABSL_HAVE_ATTRIBUTE(nonnull) || (defined(__GNUC__) && !defined(__clang__))
>+#define ABSL_ATTRIBUTE_NONNULL(arg_index) __attribute__((nonnull(arg_index)))
>+#else
>+#define ABSL_ATTRIBUTE_NONNULL(...)
>+#endif
>+
>+// ABSL_ATTRIBUTE_NORETURN
>+//
>+// Tells the compiler that a given function never returns.
>+#if ABSL_HAVE_ATTRIBUTE(noreturn) || (defined(__GNUC__) && !defined(__clang__))
>+#define ABSL_ATTRIBUTE_NORETURN __attribute__((noreturn))
>+#elif defined(_MSC_VER)
>+#define ABSL_ATTRIBUTE_NORETURN __declspec(noreturn)
>+#else
>+#define ABSL_ATTRIBUTE_NORETURN
>+#endif
>+
>+// ABSL_ATTRIBUTE_NO_SANITIZE_ADDRESS
>+//
>+// Tells the AddressSanitizer (or other memory testing tools) to ignore a given
>+// function. Useful for cases when a function reads random locations on stack,
>+// calls _exit from a cloned subprocess, deliberately accesses buffer
>+// out of bounds or does other scary things with memory.
>+// NOTE: GCC supports AddressSanitizer(asan) since 4.8.
>+// https://gcc.gnu.org/gcc-4.8/changes.html
>+#if defined(__GNUC__) && defined(ADDRESS_SANITIZER)
>+#define ABSL_ATTRIBUTE_NO_SANITIZE_ADDRESS __attribute__((no_sanitize_address))
>+#else
>+#define ABSL_ATTRIBUTE_NO_SANITIZE_ADDRESS
>+#endif
>+
>+// ABSL_ATTRIBUTE_NO_SANITIZE_MEMORY
>+//
>+// Tells the  MemorySanitizer to relax the handling of a given function. All
>+// "Use of uninitialized value" warnings from such functions will be suppressed,
>+// and all values loaded from memory will be considered fully initialized.
>+// This attribute is similar to the ADDRESS_SANITIZER attribute above, but deals
>+// with initialized-ness rather than addressability issues.
>+// NOTE: MemorySanitizer(msan) is supported by Clang but not GCC.
>+#if defined(__GNUC__) && defined(MEMORY_SANITIZER)
>+#define ABSL_ATTRIBUTE_NO_SANITIZE_MEMORY __attribute__((no_sanitize_memory))
>+#else
>+#define ABSL_ATTRIBUTE_NO_SANITIZE_MEMORY
>+#endif
>+
>+// ABSL_ATTRIBUTE_NO_SANITIZE_THREAD
>+//
>+// Tells the ThreadSanitizer to not instrument a given function.
>+// NOTE: GCC supports ThreadSanitizer(tsan) since 4.8.
>+// https://gcc.gnu.org/gcc-4.8/changes.html
>+#if defined(__GNUC__) && defined(THREAD_SANITIZER)
>+#define ABSL_ATTRIBUTE_NO_SANITIZE_THREAD __attribute__((no_sanitize_thread))
>+#else
>+#define ABSL_ATTRIBUTE_NO_SANITIZE_THREAD
>+#endif
>+
>+// ABSL_ATTRIBUTE_NO_SANITIZE_UNDEFINED
>+//
>+// Tells the UndefinedSanitizer to ignore a given function. Useful for cases
>+// where certain behavior (eg. division by zero) is being used intentionally.
>+// NOTE: GCC supports UndefinedBehaviorSanitizer(ubsan) since 4.9.
>+// https://gcc.gnu.org/gcc-4.9/changes.html
>+#if defined(__GNUC__) && \
>+    (defined(UNDEFINED_BEHAVIOR_SANITIZER) || defined(ADDRESS_SANITIZER))
>+#define ABSL_ATTRIBUTE_NO_SANITIZE_UNDEFINED \
>+  __attribute__((no_sanitize("undefined")))
>+#else
>+#define ABSL_ATTRIBUTE_NO_SANITIZE_UNDEFINED
>+#endif
>+
>+// ABSL_ATTRIBUTE_NO_SANITIZE_CFI
>+//
>+// Tells the ControlFlowIntegrity sanitizer to not instrument a given function.
>+// See https://clang.llvm.org/docs/ControlFlowIntegrity.html for details.
>+#if defined(__GNUC__) && defined(CONTROL_FLOW_INTEGRITY)
>+#define ABSL_ATTRIBUTE_NO_SANITIZE_CFI __attribute__((no_sanitize("cfi")))
>+#else
>+#define ABSL_ATTRIBUTE_NO_SANITIZE_CFI
>+#endif
>+
>+// ABSL_ATTRIBUTE_RETURNS_NONNULL
>+//
>+// Tells the compiler that a particular function never returns a null pointer.
>+#if ABSL_HAVE_ATTRIBUTE(returns_nonnull) || \
>+    (defined(__GNUC__) && \
>+     (__GNUC__ > 5 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 9)) && \
>+     !defined(__clang__))
>+#define ABSL_ATTRIBUTE_RETURNS_NONNULL __attribute__((returns_nonnull))
>+#else
>+#define ABSL_ATTRIBUTE_RETURNS_NONNULL
>+#endif
>+
>+// ABSL_HAVE_ATTRIBUTE_SECTION
>+//
>+// Indicates whether labeled sections are supported. Labeled sections are not
>+// supported on Darwin/iOS.
>+#ifdef ABSL_HAVE_ATTRIBUTE_SECTION
>+#error ABSL_HAVE_ATTRIBUTE_SECTION cannot be directly set
>+#elif (ABSL_HAVE_ATTRIBUTE(section) ||                \
>+       (defined(__GNUC__) && !defined(__clang__))) && \
>+    !defined(__APPLE__)
>+#define ABSL_HAVE_ATTRIBUTE_SECTION 1
>+
>+// ABSL_ATTRIBUTE_SECTION
>+//
>+// Tells the compiler/linker to put a given function into a section and define
>+// `__start_ ## name` and `__stop_ ## name` symbols to bracket the section.
>+// This functionality is supported by GNU linker.  Any function annotated with
>+// `ABSL_ATTRIBUTE_SECTION` must not be inlined, or it will be placed into
>+// whatever section its caller is placed into.
>+//
>+#ifndef ABSL_ATTRIBUTE_SECTION
>+#define ABSL_ATTRIBUTE_SECTION(name) \
>+  __attribute__((section(#name))) __attribute__((noinline))
>+#endif
>+
>+
>+// ABSL_ATTRIBUTE_SECTION_VARIABLE
>+//
>+// Tells the compiler/linker to put a given variable into a section and define
>+// `__start_ ## name` and `__stop_ ## name` symbols to bracket the section.
>+// This functionality is supported by GNU linker.
>+#ifndef ABSL_ATTRIBUTE_SECTION_VARIABLE
>+#define ABSL_ATTRIBUTE_SECTION_VARIABLE(name) __attribute__((section(#name)))
>+#endif
>+
>+// ABSL_DECLARE_ATTRIBUTE_SECTION_VARS
>+//
>+// A weak section declaration to be used as a global declaration
>+// for ABSL_ATTRIBUTE_SECTION_START|STOP(name) to compile and link
>+// even without functions with ABSL_ATTRIBUTE_SECTION(name).
>+// ABSL_DEFINE_ATTRIBUTE_SECTION should be in the exactly one file; it's
>+// a no-op on ELF but not on Mach-O.
>+//
>+#ifndef ABSL_DECLARE_ATTRIBUTE_SECTION_VARS
>+#define ABSL_DECLARE_ATTRIBUTE_SECTION_VARS(name) \
>+  extern char __start_##name[] ABSL_ATTRIBUTE_WEAK;    \
>+  extern char __stop_##name[] ABSL_ATTRIBUTE_WEAK
>+#endif
>+#ifndef ABSL_DEFINE_ATTRIBUTE_SECTION_VARS
>+#define ABSL_INIT_ATTRIBUTE_SECTION_VARS(name)
>+#define ABSL_DEFINE_ATTRIBUTE_SECTION_VARS(name)
>+#endif
>+
>+// ABSL_ATTRIBUTE_SECTION_START
>+//
>+// Returns `void*` pointers to start/end of a section of code with
>+// functions having ABSL_ATTRIBUTE_SECTION(name).
>+// Returns 0 if no such functions exist.
>+// One must ABSL_DECLARE_ATTRIBUTE_SECTION_VARS(name) for this to compile and
>+// link.
>+//
>+#define ABSL_ATTRIBUTE_SECTION_START(name) \
>+  (reinterpret_cast<void *>(__start_##name))
>+#define ABSL_ATTRIBUTE_SECTION_STOP(name) \
>+  (reinterpret_cast<void *>(__stop_##name))
>+
>+#else  // !ABSL_HAVE_ATTRIBUTE_SECTION
>+
>+#define ABSL_HAVE_ATTRIBUTE_SECTION 0
>+
>+// provide dummy definitions
>+#define ABSL_ATTRIBUTE_SECTION(name)
>+#define ABSL_ATTRIBUTE_SECTION_VARIABLE(name)
>+#define ABSL_INIT_ATTRIBUTE_SECTION_VARS(name)
>+#define ABSL_DEFINE_ATTRIBUTE_SECTION_VARS(name)
>+#define ABSL_DECLARE_ATTRIBUTE_SECTION_VARS(name)
>+#define ABSL_ATTRIBUTE_SECTION_START(name) (reinterpret_cast<void *>(0))
>+#define ABSL_ATTRIBUTE_SECTION_STOP(name) (reinterpret_cast<void *>(0))
>+
>+#endif  // ABSL_ATTRIBUTE_SECTION
>+
>+// ABSL_ATTRIBUTE_STACK_ALIGN_FOR_OLD_LIBC
>+//
>+// Support for aligning the stack on 32-bit x86.
>+#if ABSL_HAVE_ATTRIBUTE(force_align_arg_pointer) || \
>+    (defined(__GNUC__) && !defined(__clang__))
>+#if defined(__i386__)
>+#define ABSL_ATTRIBUTE_STACK_ALIGN_FOR_OLD_LIBC \
>+  __attribute__((force_align_arg_pointer))
>+#define ABSL_REQUIRE_STACK_ALIGN_TRAMPOLINE (0)
>+#elif defined(__x86_64__)
>+#define ABSL_REQUIRE_STACK_ALIGN_TRAMPOLINE (1)
>+#define ABSL_ATTRIBUTE_STACK_ALIGN_FOR_OLD_LIBC
>+#else  // !__i386__ && !__x86_64
>+#define ABSL_REQUIRE_STACK_ALIGN_TRAMPOLINE (0)
>+#define ABSL_ATTRIBUTE_STACK_ALIGN_FOR_OLD_LIBC
>+#endif  // __i386__
>+#else
>+#define ABSL_ATTRIBUTE_STACK_ALIGN_FOR_OLD_LIBC
>+#define ABSL_REQUIRE_STACK_ALIGN_TRAMPOLINE (0)
>+#endif
>+
>+// ABSL_MUST_USE_RESULT
>+//
>+// Tells the compiler to warn about unused return values for functions declared
>+// with this macro. The macro must appear as the very first part of a function
>+// declaration or definition:
>+//
>+// Example:
>+//
>+//   ABSL_MUST_USE_RESULT Sprocket* AllocateSprocket();
>+//
>+// This placement has the broadest compatibility with GCC, Clang, and MSVC, with
>+// both defs and decls, and with GCC-style attributes, MSVC declspec, C++11
>+// and C++17 attributes.
>+//
>+// ABSL_MUST_USE_RESULT allows using cast-to-void to suppress the unused result
>+// warning. For that, warn_unused_result is used only for clang but not for gcc.
>+// https://gcc.gnu.org/bugzilla/show_bug.cgi?id=66425
>+//
>+// Note: past advice was to place the macro after the argument list.
>+#if ABSL_HAVE_ATTRIBUTE(nodiscard)
>+#define ABSL_MUST_USE_RESULT [[nodiscard]]
>+#elif defined(__clang__) && ABSL_HAVE_ATTRIBUTE(warn_unused_result)
>+#define ABSL_MUST_USE_RESULT __attribute__((warn_unused_result))
>+#else
>+#define ABSL_MUST_USE_RESULT
>+#endif
>+
>+// ABSL_ATTRIBUTE_HOT, ABSL_ATTRIBUTE_COLD
>+//
>+// Tells GCC that a function is hot or cold. GCC can use this information to
>+// improve static analysis, i.e. a conditional branch to a cold function
>+// is likely to be not-taken.
>+// This annotation is used for function declarations.
>+//
>+// Example:
>+//
>+//   int foo() ABSL_ATTRIBUTE_HOT;
>+#if ABSL_HAVE_ATTRIBUTE(hot) || (defined(__GNUC__) && !defined(__clang__))
>+#define ABSL_ATTRIBUTE_HOT __attribute__((hot))
>+#else
>+#define ABSL_ATTRIBUTE_HOT
>+#endif
>+
>+#if ABSL_HAVE_ATTRIBUTE(cold) || (defined(__GNUC__) && !defined(__clang__))
>+#define ABSL_ATTRIBUTE_COLD __attribute__((cold))
>+#else
>+#define ABSL_ATTRIBUTE_COLD
>+#endif
>+
>+// ABSL_XRAY_ALWAYS_INSTRUMENT, ABSL_XRAY_NEVER_INSTRUMENT, ABSL_XRAY_LOG_ARGS
>+//
>+// We define the ABSL_XRAY_ALWAYS_INSTRUMENT and ABSL_XRAY_NEVER_INSTRUMENT
>+// macro used as an attribute to mark functions that must always or never be
>+// instrumented by XRay. Currently, this is only supported in Clang/LLVM.
>+//
>+// For reference on the LLVM XRay instrumentation, see
>+// http://llvm.org/docs/XRay.html.
>+//
>+// A function with the XRAY_ALWAYS_INSTRUMENT macro attribute in its declaration
>+// will always get the XRay instrumentation sleds. These sleds may introduce
>+// some binary size and runtime overhead and must be used sparingly.
>+//
>+// These attributes only take effect when the following conditions are met:
>+//
>+//   * The file/target is built in at least C++11 mode, with a Clang compiler
>+//     that supports XRay attributes.
>+//   * The file/target is built with the -fxray-instrument flag set for the
>+//     Clang/LLVM compiler.
>+//   * The function is defined in the translation unit (the compiler honors the
>+//     attribute in either the definition or the declaration, and must match).
>+//
>+// There are cases when, even when building with XRay instrumentation, users
>+// might want to control specifically which functions are instrumented for a
>+// particular build using special-case lists provided to the compiler. These
>+// special case lists are provided to Clang via the
>+// -fxray-always-instrument=... and -fxray-never-instrument=... flags. The
>+// attributes in source take precedence over these special-case lists.
>+//
>+// To disable the XRay attributes at build-time, users may define
>+// ABSL_NO_XRAY_ATTRIBUTES. Do NOT define ABSL_NO_XRAY_ATTRIBUTES on specific
>+// packages/targets, as this may lead to conflicting definitions of functions at
>+// link-time.
>+//
>+#if ABSL_HAVE_CPP_ATTRIBUTE(clang::xray_always_instrument) && \
>+    !defined(ABSL_NO_XRAY_ATTRIBUTES)
>+#define ABSL_XRAY_ALWAYS_INSTRUMENT [[clang::xray_always_instrument]]
>+#define ABSL_XRAY_NEVER_INSTRUMENT [[clang::xray_never_instrument]]
>+#if ABSL_HAVE_CPP_ATTRIBUTE(clang::xray_log_args)
>+#define ABSL_XRAY_LOG_ARGS(N) \
>+    [[clang::xray_always_instrument, clang::xray_log_args(N)]]
>+#else
>+#define ABSL_XRAY_LOG_ARGS(N) [[clang::xray_always_instrument]]
>+#endif
>+#else
>+#define ABSL_XRAY_ALWAYS_INSTRUMENT
>+#define ABSL_XRAY_NEVER_INSTRUMENT
>+#define ABSL_XRAY_LOG_ARGS(N)
>+#endif
>+
>+// -----------------------------------------------------------------------------
>+// Variable Attributes
>+// -----------------------------------------------------------------------------
>+
>+// ABSL_ATTRIBUTE_UNUSED
>+//
>+// Prevents the compiler from complaining about or optimizing away variables
>+// that appear unused.
>+#if ABSL_HAVE_ATTRIBUTE(unused) || (defined(__GNUC__) && !defined(__clang__))
>+#undef ABSL_ATTRIBUTE_UNUSED
>+#define ABSL_ATTRIBUTE_UNUSED __attribute__((__unused__))
>+#else
>+#define ABSL_ATTRIBUTE_UNUSED
>+#endif
>+
>+// ABSL_ATTRIBUTE_INITIAL_EXEC
>+//
>+// Tells the compiler to use "initial-exec" mode for a thread-local variable.
>+// See http://people.redhat.com/drepper/tls.pdf for the gory details.
>+#if ABSL_HAVE_ATTRIBUTE(tls_model) || (defined(__GNUC__) && !defined(__clang__))
>+#define ABSL_ATTRIBUTE_INITIAL_EXEC __attribute__((tls_model("initial-exec")))
>+#else
>+#define ABSL_ATTRIBUTE_INITIAL_EXEC
>+#endif
>+
>+// ABSL_ATTRIBUTE_PACKED
>+//
>+// Prevents the compiler from padding a structure to natural alignment
>+#if ABSL_HAVE_ATTRIBUTE(packed) || (defined(__GNUC__) && !defined(__clang__))
>+#define ABSL_ATTRIBUTE_PACKED __attribute__((__packed__))
>+#else
>+#define ABSL_ATTRIBUTE_PACKED
>+#endif
>+
>+// ABSL_ATTRIBUTE_FUNC_ALIGN
>+//
>+// Tells the compiler to align the function start at least to certain
>+// alignment boundary
>+#if ABSL_HAVE_ATTRIBUTE(aligned) || (defined(__GNUC__) && !defined(__clang__))
>+#define ABSL_ATTRIBUTE_FUNC_ALIGN(bytes) __attribute__((aligned(bytes)))
>+#else
>+#define ABSL_ATTRIBUTE_FUNC_ALIGN(bytes)
>+#endif
>+
>+// ABSL_CONST_INIT
>+//
>+// A variable declaration annotated with the `ABSL_CONST_INIT` attribute will
>+// not compile (on supported platforms) unless the variable has a constant
>+// initializer. This is useful for variables with static and thread storage
>+// duration, because it guarantees that they will not suffer from the so-called
>+// "static init order fiasco".  Prefer to put this attribute on the most visible
>+// declaration of the variable, if there's more than one, because code that
>+// accesses the variable can then use the attribute for optimization.
>+//
>+// Example:
>+//
>+//   class MyClass {
>+//    public:
>+//     ABSL_CONST_INIT static MyType my_var;
>+//   };
>+//
>+//   MyType MyClass::my_var = MakeMyType(...);
>+//
>+// Note that this attribute is redundant if the variable is declared constexpr.
>+#if ABSL_HAVE_CPP_ATTRIBUTE(clang::require_constant_initialization)
>+// NOLINTNEXTLINE(whitespace/braces)
>+#define ABSL_CONST_INIT [[clang::require_constant_initialization]]
>+#else
>+#define ABSL_CONST_INIT
>+#endif  // ABSL_HAVE_CPP_ATTRIBUTE(clang::require_constant_initialization)
>+
>+#endif  // ABSL_BASE_ATTRIBUTES_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/bit_cast_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/bit_cast_test.cc
>new file mode 100644
>index 00000000000..8cd878d756e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/bit_cast_test.cc
>@@ -0,0 +1,107 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+// Unit test for bit_cast template.
>+
>+#include <cstdint>
>+#include <cstring>
>+
>+#include "gtest/gtest.h"
>+#include "absl/base/casts.h"
>+#include "absl/base/macros.h"
>+
>+namespace absl {
>+namespace {
>+
>+template <int N>
>+struct marshall { char buf[N]; };
>+
>+template <typename T>
>+void TestMarshall(const T values[], int num_values) {
>+  for (int i = 0; i < num_values; ++i) {
>+    T t0 = values[i];
>+    marshall<sizeof(T)> m0 = absl::bit_cast<marshall<sizeof(T)> >(t0);
>+    T t1 = absl::bit_cast<T>(m0);
>+    marshall<sizeof(T)> m1 = absl::bit_cast<marshall<sizeof(T)> >(t1);
>+    ASSERT_EQ(0, memcmp(&t0, &t1, sizeof(T)));
>+    ASSERT_EQ(0, memcmp(&m0, &m1, sizeof(T)));
>+  }
>+}
>+
>+// Convert back and forth to an integral type.  The C++ standard does
>+// not guarantee this will work, but we test that this works on all the
>+// platforms we support.
>+//
>+// Likewise, we below make assumptions about sizeof(float) and
>+// sizeof(double) which the standard does not guarantee, but which hold on the
>+// platforms we support.
>+
>+template <typename T, typename I>
>+void TestIntegral(const T values[], int num_values) {
>+  for (int i = 0; i < num_values; ++i) {
>+    T t0 = values[i];
>+    I i0 = absl::bit_cast<I>(t0);
>+    T t1 = absl::bit_cast<T>(i0);
>+    I i1 = absl::bit_cast<I>(t1);
>+    ASSERT_EQ(0, memcmp(&t0, &t1, sizeof(T)));
>+    ASSERT_EQ(i0, i1);
>+  }
>+}
>+
>+TEST(BitCast, Bool) {
>+  static const bool bool_list[] = { false, true };
>+  TestMarshall<bool>(bool_list, ABSL_ARRAYSIZE(bool_list));
>+}
>+
>+TEST(BitCast, Int32) {
>+  static const int32_t int_list[] =
>+    { 0, 1, 100, 2147483647, -1, -100, -2147483647, -2147483647-1 };
>+  TestMarshall<int32_t>(int_list, ABSL_ARRAYSIZE(int_list));
>+}
>+
>+TEST(BitCast, Int64) {
>+  static const int64_t int64_list[] =
>+    { 0, 1, 1LL << 40, -1, -(1LL<<40) };
>+  TestMarshall<int64_t>(int64_list, ABSL_ARRAYSIZE(int64_list));
>+}
>+
>+TEST(BitCast, Uint64) {
>+  static const uint64_t uint64_list[] =
>+    { 0, 1, 1LLU << 40, 1LLU << 63 };
>+  TestMarshall<uint64_t>(uint64_list, ABSL_ARRAYSIZE(uint64_list));
>+}
>+
>+TEST(BitCast, Float) {
>+  static const float float_list[] =
>+    { 0.0f, 1.0f, -1.0f, 10.0f, -10.0f,
>+      1e10f, 1e20f, 1e-10f, 1e-20f,
>+      2.71828f, 3.14159f };
>+  TestMarshall<float>(float_list, ABSL_ARRAYSIZE(float_list));
>+  TestIntegral<float, int>(float_list, ABSL_ARRAYSIZE(float_list));
>+  TestIntegral<float, unsigned>(float_list, ABSL_ARRAYSIZE(float_list));
>+}
>+
>+TEST(BitCast, Double) {
>+  static const double double_list[] =
>+    { 0.0, 1.0, -1.0, 10.0, -10.0,
>+      1e10, 1e100, 1e-10, 1e-100,
>+      2.718281828459045,
>+      3.141592653589793238462643383279502884197169399375105820974944 };
>+  TestMarshall<double>(double_list, ABSL_ARRAYSIZE(double_list));
>+  TestIntegral<double, int64_t>(double_list, ABSL_ARRAYSIZE(double_list));
>+  TestIntegral<double, uint64_t>(double_list, ABSL_ARRAYSIZE(double_list));
>+}
>+
>+}  // namespace
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/call_once.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/call_once.h
>new file mode 100644
>index 00000000000..532ee2e38bb
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/call_once.h
>@@ -0,0 +1,216 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// -----------------------------------------------------------------------------
>+// File: call_once.h
>+// -----------------------------------------------------------------------------
>+//
>+// This header file provides an Abseil version of `std::call_once` for invoking
>+// a given function at most once, across all threads. This Abseil version is
>+// faster than the C++11 version and incorporates the C++17 argument-passing
>+// fix, so that (for example) non-const references may be passed to the invoked
>+// function.
>+
>+#ifndef ABSL_BASE_CALL_ONCE_H_
>+#define ABSL_BASE_CALL_ONCE_H_
>+
>+#include <algorithm>
>+#include <atomic>
>+#include <cstdint>
>+#include <type_traits>
>+
>+#include "absl/base/internal/invoke.h"
>+#include "absl/base/internal/low_level_scheduling.h"
>+#include "absl/base/internal/raw_logging.h"
>+#include "absl/base/internal/scheduling_mode.h"
>+#include "absl/base/internal/spinlock_wait.h"
>+#include "absl/base/macros.h"
>+#include "absl/base/port.h"
>+
>+namespace absl {
>+
>+class once_flag;
>+
>+namespace base_internal {
>+std::atomic<uint32_t>* ControlWord(absl::once_flag* flag);
>+}  // namespace base_internal
>+
>+// call_once()
>+//
>+// For all invocations using a given `once_flag`, invokes a given `fn` exactly
>+// once across all threads. The first call to `call_once()` with a particular
>+// `once_flag` argument (that does not throw an exception) will run the
>+// specified function with the provided `args`; other calls with the same
>+// `once_flag` argument will not run the function, but will wait
>+// for the provided function to finish running (if it is still running).
>+//
>+// This mechanism provides a safe, simple, and fast mechanism for one-time
>+// initialization in a multi-threaded process.
>+//
>+// Example:
>+//
>+// class MyInitClass {
>+//  public:
>+//  ...
>+//  mutable absl::once_flag once_;
>+//
>+//  MyInitClass* init() const {
>+//    absl::call_once(once_, &MyInitClass::Init, this);
>+//    return ptr_;
>+//  }
>+//
>+template <typename Callable, typename... Args>
>+void call_once(absl::once_flag& flag, Callable&& fn, Args&&... args);
>+
>+// once_flag
>+//
>+// Objects of this type are used to distinguish calls to `call_once()` and
>+// ensure the provided function is only invoked once across all threads. This
>+// type is not copyable or movable. However, it has a `constexpr`
>+// constructor, and is safe to use as a namespace-scoped global variable.
>+class once_flag {
>+ public:
>+  constexpr once_flag() : control_(0) {}
>+  once_flag(const once_flag&) = delete;
>+  once_flag& operator=(const once_flag&) = delete;
>+
>+ private:
>+  friend std::atomic<uint32_t>* base_internal::ControlWord(once_flag* flag);
>+  std::atomic<uint32_t> control_;
>+};
>+
>+//------------------------------------------------------------------------------
>+// End of public interfaces.
>+// Implementation details follow.
>+//------------------------------------------------------------------------------
>+
>+namespace base_internal {
>+
>+// Like call_once, but uses KERNEL_ONLY scheduling. Intended to be used to
>+// initialize entities used by the scheduler implementation.
>+template <typename Callable, typename... Args>
>+void LowLevelCallOnce(absl::once_flag* flag, Callable&& fn, Args&&... args);
>+
>+// Disables scheduling while on stack when scheduling mode is non-cooperative.
>+// No effect for cooperative scheduling modes.
>+class SchedulingHelper {
>+ public:
>+  explicit SchedulingHelper(base_internal::SchedulingMode mode) : mode_(mode) {
>+    if (mode_ == base_internal::SCHEDULE_KERNEL_ONLY) {
>+      guard_result_ = base_internal::SchedulingGuard::DisableRescheduling();
>+    }
>+  }
>+
>+  ~SchedulingHelper() {
>+    if (mode_ == base_internal::SCHEDULE_KERNEL_ONLY) {
>+      base_internal::SchedulingGuard::EnableRescheduling(guard_result_);
>+    }
>+  }
>+
>+ private:
>+  base_internal::SchedulingMode mode_;
>+  bool guard_result_;
>+};
>+
>+// Bit patterns for call_once state machine values.  Internal implementation
>+// detail, not for use by clients.
>+//
>+// The bit patterns are arbitrarily chosen from unlikely values, to aid in
>+// debugging.  However, kOnceInit must be 0, so that a zero-initialized
>+// once_flag will be valid for immediate use.
>+enum {
>+  kOnceInit = 0,
>+  kOnceRunning = 0x65C2937B,
>+  kOnceWaiter = 0x05A308D2,
>+  // A very small constant is chosen for kOnceDone so that it fit in a single
>+  // compare with immediate instruction for most common ISAs.  This is verified
>+  // for x86, POWER and ARM.
>+  kOnceDone = 221,    // Random Number
>+};
>+
>+template <typename Callable, typename... Args>
>+void CallOnceImpl(std::atomic<uint32_t>* control,
>+                  base_internal::SchedulingMode scheduling_mode, Callable&& fn,
>+                  Args&&... args) {
>+#ifndef NDEBUG
>+  {
>+    uint32_t old_control = control->load(std::memory_order_acquire);
>+    if (old_control != kOnceInit &&
>+        old_control != kOnceRunning &&
>+        old_control != kOnceWaiter &&
>+        old_control != kOnceDone) {
>+      ABSL_RAW_LOG(
>+          FATAL,
>+          "Unexpected value for control word: %lx. Either the control word "
>+          "has non-static storage duration (where GoogleOnceDynamic might "
>+          "be appropriate), or there's been a memory corruption.",
>+          static_cast<unsigned long>(old_control)); // NOLINT
>+    }
>+  }
>+#endif  // NDEBUG
>+  static const base_internal::SpinLockWaitTransition trans[] = {
>+      {kOnceInit, kOnceRunning, true},
>+      {kOnceRunning, kOnceWaiter, false},
>+      {kOnceDone, kOnceDone, true}};
>+
>+  // Must do this before potentially modifying control word's state.
>+  base_internal::SchedulingHelper maybe_disable_scheduling(scheduling_mode);
>+  // Short circuit the simplest case to avoid procedure call overhead.
>+  uint32_t old_control = kOnceInit;
>+  if (control->compare_exchange_strong(old_control, kOnceRunning,
>+                                       std::memory_order_acquire,
>+                                       std::memory_order_relaxed) ||
>+      base_internal::SpinLockWait(control, ABSL_ARRAYSIZE(trans), trans,
>+                                  scheduling_mode) == kOnceInit) {
>+    base_internal::Invoke(std::forward<Callable>(fn),
>+                          std::forward<Args>(args)...);
>+    old_control = control->load(std::memory_order_relaxed);
>+    control->store(base_internal::kOnceDone, std::memory_order_release);
>+    if (old_control == base_internal::kOnceWaiter) {
>+      base_internal::SpinLockWake(control, true);
>+    }
>+  }  // else *control is already kOnceDone
>+}
>+
>+inline std::atomic<uint32_t>* ControlWord(once_flag* flag) {
>+  return &flag->control_;
>+}
>+
>+template <typename Callable, typename... Args>
>+void LowLevelCallOnce(absl::once_flag* flag, Callable&& fn, Args&&... args) {
>+  std::atomic<uint32_t>* once = base_internal::ControlWord(flag);
>+  uint32_t s = once->load(std::memory_order_acquire);
>+  if (ABSL_PREDICT_FALSE(s != base_internal::kOnceDone)) {
>+    base_internal::CallOnceImpl(once, base_internal::SCHEDULE_KERNEL_ONLY,
>+                                std::forward<Callable>(fn),
>+                                std::forward<Args>(args)...);
>+  }
>+}
>+
>+}  // namespace base_internal
>+
>+template <typename Callable, typename... Args>
>+void call_once(absl::once_flag& flag, Callable&& fn, Args&&... args) {
>+  std::atomic<uint32_t>* once = base_internal::ControlWord(&flag);
>+  uint32_t s = once->load(std::memory_order_acquire);
>+  if (ABSL_PREDICT_FALSE(s != base_internal::kOnceDone)) {
>+    base_internal::CallOnceImpl(
>+        once, base_internal::SCHEDULE_COOPERATIVE_AND_KERNEL,
>+        std::forward<Callable>(fn), std::forward<Args>(args)...);
>+  }
>+}
>+
>+}  // namespace absl
>+
>+#endif  // ABSL_BASE_CALL_ONCE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/call_once_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/call_once_test.cc
>new file mode 100644
>index 00000000000..cd58ee19f08
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/call_once_test.cc
>@@ -0,0 +1,102 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/base/call_once.h"
>+
>+#include <thread>
>+#include <vector>
>+
>+#include "gtest/gtest.h"
>+#include "absl/base/thread_annotations.h"
>+#include "absl/synchronization/mutex.h"
>+
>+namespace absl {
>+namespace {
>+
>+absl::once_flag once;
>+Mutex counters_mu;
>+
>+int running_thread_count GUARDED_BY(counters_mu) = 0;
>+int call_once_invoke_count GUARDED_BY(counters_mu) = 0;
>+int call_once_finished_count GUARDED_BY(counters_mu) = 0;
>+int call_once_return_count GUARDED_BY(counters_mu) = 0;
>+bool done_blocking GUARDED_BY(counters_mu) = false;
>+
>+// Function to be called from absl::call_once.  Waits for a notification.
>+void WaitAndIncrement() {
>+  counters_mu.Lock();
>+  ++call_once_invoke_count;
>+  counters_mu.Unlock();
>+
>+  counters_mu.LockWhen(Condition(&done_blocking));
>+  ++call_once_finished_count;
>+  counters_mu.Unlock();
>+}
>+
>+void ThreadBody() {
>+  counters_mu.Lock();
>+  ++running_thread_count;
>+  counters_mu.Unlock();
>+
>+  absl::call_once(once, WaitAndIncrement);
>+
>+  counters_mu.Lock();
>+  ++call_once_return_count;
>+  counters_mu.Unlock();
>+}
>+
>+// Returns true if all threads are set up for the test.
>+bool ThreadsAreSetup(void*) EXCLUSIVE_LOCKS_REQUIRED(counters_mu) {
>+  // All ten threads must be running, and WaitAndIncrement should be blocked.
>+  return running_thread_count == 10 && call_once_invoke_count == 1;
>+}
>+
>+TEST(CallOnceTest, ExecutionCount) {
>+  std::vector<std::thread> threads;
>+
>+  // Start 10 threads all calling call_once on the same once_flag.
>+  for (int i = 0; i < 10; ++i) {
>+    threads.emplace_back(ThreadBody);
>+  }
>+
>+
>+  // Wait until all ten threads have started, and WaitAndIncrement has been
>+  // invoked.
>+  counters_mu.LockWhen(Condition(ThreadsAreSetup, nullptr));
>+
>+  // WaitAndIncrement should have been invoked by exactly one call_once()
>+  // instance.  That thread should be blocking on a notification, and all other
>+  // call_once instances should be blocking as well.
>+  EXPECT_EQ(call_once_invoke_count, 1);
>+  EXPECT_EQ(call_once_finished_count, 0);
>+  EXPECT_EQ(call_once_return_count, 0);
>+
>+  // Allow WaitAndIncrement to finish executing.  Once it does, the other
>+  // call_once waiters will be unblocked.
>+  done_blocking = true;
>+  counters_mu.Unlock();
>+
>+  for (std::thread& thread : threads) {
>+    thread.join();
>+  }
>+
>+  counters_mu.Lock();
>+  EXPECT_EQ(call_once_invoke_count, 1);
>+  EXPECT_EQ(call_once_finished_count, 1);
>+  EXPECT_EQ(call_once_return_count, 10);
>+  counters_mu.Unlock();
>+}
>+
>+}  // namespace
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/casts.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/casts.h
>new file mode 100644
>index 00000000000..20fd34da701
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/casts.h
>@@ -0,0 +1,189 @@
>+//
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// -----------------------------------------------------------------------------
>+// File: casts.h
>+// -----------------------------------------------------------------------------
>+//
>+// This header file defines casting templates to fit use cases not covered by
>+// the standard casts provided in the C++ standard. As with all cast operations,
>+// use these with caution and only if alternatives do not exist.
>+
>+#ifndef ABSL_BASE_CASTS_H_
>+#define ABSL_BASE_CASTS_H_
>+
>+#include <cstring>
>+#include <memory>
>+#include <type_traits>
>+
>+#include "absl/base/internal/identity.h"
>+#include "absl/base/macros.h"
>+
>+namespace absl {
>+
>+namespace internal_casts {
>+
>+// NOTE: Not a fully compliant implementation of `std::is_trivially_copyable`.
>+// TODO(calabrese) Branch on implementations that directly provide
>+// `std::is_trivially_copyable`, create a more rigorous workaround, and publicly
>+// expose in meta/type_traits.
>+template <class T>
>+struct is_trivially_copyable
>+    : std::integral_constant<
>+          bool, std::is_destructible<T>::value&& __has_trivial_destructor(T) &&
>+                    __has_trivial_copy(T) && __has_trivial_assign(T)> {};
>+
>+template <class Dest, class Source>
>+struct is_bitcastable
>+    : std::integral_constant<bool,
>+                             sizeof(Dest) == sizeof(Source) &&
>+                                 is_trivially_copyable<Source>::value &&
>+                                 is_trivially_copyable<Dest>::value &&
>+                                 std::is_default_constructible<Dest>::value> {};
>+
>+}  // namespace internal_casts
>+
>+// implicit_cast()
>+//
>+// Performs an implicit conversion between types following the language
>+// rules for implicit conversion; if an implicit conversion is otherwise
>+// allowed by the language in the given context, this function performs such an
>+// implicit conversion.
>+//
>+// Example:
>+//
>+//   // If the context allows implicit conversion:
>+//   From from;
>+//   To to = from;
>+//
>+//   // Such code can be replaced by:
>+//   implicit_cast<To>(from);
>+//
>+// An `implicit_cast()` may also be used to annotate numeric type conversions
>+// that, although safe, may produce compiler warnings (such as `long` to `int`).
>+// Additionally, an `implicit_cast()` is also useful within return statements to
>+// indicate a specific implicit conversion is being undertaken.
>+//
>+// Example:
>+//
>+//   return implicit_cast<double>(size_in_bytes) / capacity_;
>+//
>+// Annotating code with `implicit_cast()` allows you to explicitly select
>+// particular overloads and template instantiations, while providing a safer
>+// cast than `reinterpret_cast()` or `static_cast()`.
>+//
>+// Additionally, an `implicit_cast()` can be used to allow upcasting within a
>+// type hierarchy where incorrect use of `static_cast()` could accidentally
>+// allow downcasting.
>+//
>+// Finally, an `implicit_cast()` can be used to perform implicit conversions
>+// from unrelated types that otherwise couldn't be implicitly cast directly;
>+// C++ will normally only implicitly cast "one step" in such conversions.
>+//
>+// That is, if C is a type which can be implicitly converted to B, with B being
>+// a type that can be implicitly converted to A, an `implicit_cast()` can be
>+// used to convert C to B (which the compiler can then implicitly convert to A
>+// using language rules).
>+//
>+// Example:
>+//
>+//   // Assume an object C is convertible to B, which is implicitly convertible
>+//   // to A
>+//   A a = implicit_cast<B>(C);
>+//
>+// Such implicit cast chaining may be useful within template logic.
>+template <typename To>
>+inline To implicit_cast(typename absl::internal::identity_t<To> to) {
>+  return to;
>+}
>+
>+// bit_cast()
>+//
>+// Performs a bitwise cast on a type without changing the underlying bit
>+// representation of that type's value. The two types must be of the same size
>+// and both types must be trivially copyable. As with most casts, use with
>+// caution. A `bit_cast()` might be needed when you need to temporarily treat a
>+// type as some other type, such as in the following cases:
>+//
>+//    * Serialization (casting temporarily to `char *` for those purposes is
>+//      always allowed by the C++ standard)
>+//    * Managing the individual bits of a type within mathematical operations
>+//      that are not normally accessible through that type
>+//    * Casting non-pointer types to pointer types (casting the other way is
>+//      allowed by `reinterpret_cast()` but round-trips cannot occur the other
>+//      way).
>+//
>+// Example:
>+//
>+//   float f = 3.14159265358979;
>+//   int i = bit_cast<int32_t>(f);
>+//   // i = 0x40490fdb
>+//
>+// Casting non-pointer types to pointer types and then dereferencing them
>+// traditionally produces undefined behavior.
>+//
>+// Example:
>+//
>+//   // WRONG
>+//   float f = 3.14159265358979;            // WRONG
>+//   int i = * reinterpret_cast<int*>(&f);  // WRONG
>+//
>+// The address-casting method produces undefined behavior according to the ISO
>+// C++ specification section [basic.lval]. Roughly, this section says: if an
>+// object in memory has one type, and a program accesses it with a different
>+// type, the result is undefined behavior for most values of "different type".
>+//
>+// Such casting results in type punning: holding an object in memory of one type
>+// and reading its bits back using a different type. A `bit_cast()` avoids this
>+// issue by implementing its casts using `memcpy()`, which avoids introducing
>+// this undefined behavior.
>+//
>+// NOTE: The requirements here are more strict than the bit_cast of standard
>+// proposal p0476 due to the need for workarounds and lack of intrinsics.
>+// Specifically, this implementation also requires `Dest` to be
>+// default-constructible.
>+template <
>+    typename Dest, typename Source,
>+    typename std::enable_if<internal_casts::is_bitcastable<Dest, Source>::value,
>+                            int>::type = 0>
>+inline Dest bit_cast(const Source& source) {
>+  Dest dest;
>+  memcpy(static_cast<void*>(std::addressof(dest)),
>+         static_cast<const void*>(std::addressof(source)), sizeof(dest));
>+  return dest;
>+}
>+
>+// NOTE: This overload is only picked if the requirements of bit_cast are not
>+// met. It is therefore UB, but is provided temporarily as previous versions of
>+// this function template were unchecked. Do not use this in new code.
>+template <
>+    typename Dest, typename Source,
>+    typename std::enable_if<
>+        !internal_casts::is_bitcastable<Dest, Source>::value, int>::type = 0>
>+ABSL_DEPRECATED(
>+    "absl::bit_cast type requirements were violated. Update the types being "
>+    "used such that they are the same size and are both TriviallyCopyable.")
>+inline Dest bit_cast(const Source& source) {
>+  static_assert(sizeof(Dest) == sizeof(Source),
>+                "Source and destination types should have equal sizes.");
>+
>+  Dest dest;
>+  memcpy(&dest, &source, sizeof(dest));
>+  return dest;
>+}
>+
>+}  // namespace absl
>+
>+#endif  // ABSL_BASE_CASTS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/config.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/config.h
>new file mode 100644
>index 00000000000..6890e313bf2
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/config.h
>@@ -0,0 +1,427 @@
>+//
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// -----------------------------------------------------------------------------
>+// File: config.h
>+// -----------------------------------------------------------------------------
>+//
>+// This header file defines a set of macros for checking the presence of
>+// important compiler and platform features. Such macros can be used to
>+// produce portable code by parameterizing compilation based on the presence or
>+// lack of a given feature.
>+//
>+// We define a "feature" as some interface we wish to program to: for example,
>+// a library function or system call. A value of `1` indicates support for
>+// that feature; any other value indicates the feature support is undefined.
>+//
>+// Example:
>+//
>+// Suppose a programmer wants to write a program that uses the 'mmap()' system
>+// call. The Abseil macro for that feature (`ABSL_HAVE_MMAP`) allows you to
>+// selectively include the `mmap.h` header and bracket code using that feature
>+// in the macro:
>+//
>+//   #include "absl/base/config.h"
>+//
>+//   #ifdef ABSL_HAVE_MMAP
>+//   #include "sys/mman.h"
>+//   #endif  //ABSL_HAVE_MMAP
>+//
>+//   ...
>+//   #ifdef ABSL_HAVE_MMAP
>+//   void *ptr = mmap(...);
>+//   ...
>+//   #endif  // ABSL_HAVE_MMAP
>+
>+#ifndef ABSL_BASE_CONFIG_H_
>+#define ABSL_BASE_CONFIG_H_
>+
>+// Included for the __GLIBC__ macro (or similar macros on other systems).
>+#include <limits.h>
>+
>+#ifdef __cplusplus
>+// Included for __GLIBCXX__, _LIBCPP_VERSION
>+#include <cstddef>
>+#endif  // __cplusplus
>+
>+#if defined(__APPLE__)
>+// Included for TARGET_OS_IPHONE, __IPHONE_OS_VERSION_MIN_REQUIRED,
>+// __IPHONE_8_0.
>+#include <Availability.h>
>+#include <TargetConditionals.h>
>+#endif
>+
>+#include "absl/base/policy_checks.h"
>+
>+// -----------------------------------------------------------------------------
>+// Compiler Feature Checks
>+// -----------------------------------------------------------------------------
>+
>+// ABSL_HAVE_BUILTIN()
>+//
>+// Checks whether the compiler supports a Clang Feature Checking Macro, and if
>+// so, checks whether it supports the provided builtin function "x" where x
>+// is one of the functions noted in
>+// https://clang.llvm.org/docs/LanguageExtensions.html
>+//
>+// Note: Use this macro to avoid an extra level of #ifdef __has_builtin check.
>+// http://releases.llvm.org/3.3/tools/clang/docs/LanguageExtensions.html
>+#ifdef __has_builtin
>+#define ABSL_HAVE_BUILTIN(x) __has_builtin(x)
>+#else
>+#define ABSL_HAVE_BUILTIN(x) 0
>+#endif
>+
>+// ABSL_HAVE_TLS is defined to 1 when __thread should be supported.
>+// We assume __thread is supported on Linux when compiled with Clang or compiled
>+// against libstdc++ with _GLIBCXX_HAVE_TLS defined.
>+#ifdef ABSL_HAVE_TLS
>+#error ABSL_HAVE_TLS cannot be directly set
>+#elif defined(__linux__) && (defined(__clang__) || defined(_GLIBCXX_HAVE_TLS))
>+#define ABSL_HAVE_TLS 1
>+#endif
>+
>+// ABSL_HAVE_STD_IS_TRIVIALLY_DESTRUCTIBLE
>+//
>+// Checks whether `std::is_trivially_destructible<T>` is supported.
>+//
>+// Notes: All supported compilers using libc++ support this feature, as does
>+// gcc >= 4.8.1 using libstdc++, and Visual Studio.
>+#ifdef ABSL_HAVE_STD_IS_TRIVIALLY_DESTRUCTIBLE
>+#error ABSL_HAVE_STD_IS_TRIVIALLY_DESTRUCTIBLE cannot be directly set
>+#elif defined(_LIBCPP_VERSION) ||                                        \
>+    (!defined(__clang__) && defined(__GNUC__) && defined(__GLIBCXX__) && \
>+     (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 8))) ||        \
>+    defined(_MSC_VER)
>+#define ABSL_HAVE_STD_IS_TRIVIALLY_DESTRUCTIBLE 1
>+#endif
>+
>+// ABSL_HAVE_STD_IS_TRIVIALLY_CONSTRUCTIBLE
>+//
>+// Checks whether `std::is_trivially_default_constructible<T>` and
>+// `std::is_trivially_copy_constructible<T>` are supported.
>+
>+// ABSL_HAVE_STD_IS_TRIVIALLY_ASSIGNABLE
>+//
>+// Checks whether `std::is_trivially_copy_assignable<T>` is supported.
>+
>+// Notes: Clang with libc++ supports these features, as does gcc >= 5.1 with
>+// either libc++ or libstdc++, and Visual Studio.
>+#if defined(ABSL_HAVE_STD_IS_TRIVIALLY_CONSTRUCTIBLE)
>+#error ABSL_HAVE_STD_IS_TRIVIALLY_CONSTRUCTIBLE cannot be directly set
>+#elif defined(ABSL_HAVE_STD_IS_TRIVIALLY_ASSIGNABLE)
>+#error ABSL_HAVE_STD_IS_TRIVIALLY_ASSIGNABLE cannot directly set
>+#elif (defined(__clang__) && defined(_LIBCPP_VERSION)) ||        \
>+    (!defined(__clang__) && defined(__GNUC__) &&                 \
>+     (__GNUC__ > 5 || (__GNUC__ == 5 && __GNUC_MINOR__ >= 1)) && \
>+     (defined(_LIBCPP_VERSION) || defined(__GLIBCXX__))) ||      \
>+    defined(_MSC_VER)
>+#define ABSL_HAVE_STD_IS_TRIVIALLY_CONSTRUCTIBLE 1
>+#define ABSL_HAVE_STD_IS_TRIVIALLY_ASSIGNABLE 1
>+#endif
>+
>+// ABSL_HAVE_THREAD_LOCAL
>+//
>+// Checks whether C++11's `thread_local` storage duration specifier is
>+// supported.
>+#ifdef ABSL_HAVE_THREAD_LOCAL
>+#error ABSL_HAVE_THREAD_LOCAL cannot be directly set
>+#elif defined(__APPLE__)
>+// Notes: Xcode's clang did not support `thread_local` until version
>+// 8, and even then not for all iOS < 9.0. Also, Xcode 9.3 started disallowing
>+// `thread_local` for 32-bit iOS simulator targeting iOS 9.x.
>+// `__has_feature` is only supported by Clang so it has be inside
>+// `defined(__APPLE__)` check.
>+#if __has_feature(cxx_thread_local)
>+#define ABSL_HAVE_THREAD_LOCAL 1
>+#endif
>+#else  // !defined(__APPLE__)
>+#define ABSL_HAVE_THREAD_LOCAL 1
>+#endif
>+
>+// There are platforms for which TLS should not be used even though the compiler
>+// makes it seem like it's supported (Android NDK < r12b for example).
>+// This is primarily because of linker problems and toolchain misconfiguration:
>+// Abseil does not intend to support this indefinitely. Currently, the newest
>+// toolchain that we intend to support that requires this behavior is the
>+// r11 NDK - allowing for a 5 year support window on that means this option
>+// is likely to be removed around June of 2021.
>+// TLS isn't supported until NDK r12b per
>+// https://developer.android.com/ndk/downloads/revision_history.html
>+// Since NDK r16, `__NDK_MAJOR__` and `__NDK_MINOR__` are defined in
>+// <android/ndk-version.h>. For NDK < r16, users should define these macros,
>+// e.g. `-D__NDK_MAJOR__=11 -D__NKD_MINOR__=0` for NDK r11.
>+#if defined(__ANDROID__) && defined(__clang__)
>+#if __has_include(<android/ndk-version.h>)
>+#include <android/ndk-version.h>
>+#endif  // __has_include(<android/ndk-version.h>)
>+#if defined(__ANDROID__) && defined(__clang__) && defined(__NDK_MAJOR__) && \
>+    defined(__NDK_MINOR__) &&                                               \
>+    ((__NDK_MAJOR__ < 12) || ((__NDK_MAJOR__ == 12) && (__NDK_MINOR__ < 1)))
>+#undef ABSL_HAVE_TLS
>+#undef ABSL_HAVE_THREAD_LOCAL
>+#endif
>+#endif  // defined(__ANDROID__) && defined(__clang__)
>+
>+// ABSL_HAVE_INTRINSIC_INT128
>+//
>+// Checks whether the __int128 compiler extension for a 128-bit integral type is
>+// supported.
>+//
>+// Note: __SIZEOF_INT128__ is defined by Clang and GCC when __int128 is
>+// supported, but we avoid using it in certain cases:
>+// * On Clang:
>+//   * Building using Clang for Windows, where the Clang runtime library has
>+//     128-bit support only on LP64 architectures, but Windows is LLP64.
>+//   * Building for aarch64, where __int128 exists but has exhibits a sporadic
>+//     compiler crashing bug.
>+// * On Nvidia's nvcc:
>+//   * nvcc also defines __GNUC__ and __SIZEOF_INT128__, but not all versions
>+//     actually support __int128.
>+#ifdef ABSL_HAVE_INTRINSIC_INT128
>+#error ABSL_HAVE_INTRINSIC_INT128 cannot be directly set
>+#elif defined(__SIZEOF_INT128__)
>+#if (defined(__clang__) && !defined(_WIN32) && !defined(__aarch64__)) || \
>+    (defined(__CUDACC__) && __CUDACC_VER_MAJOR__ >= 9) ||                \
>+    (defined(__GNUC__) && !defined(__clang__) && !defined(__CUDACC__))
>+#define ABSL_HAVE_INTRINSIC_INT128 1
>+#elif defined(__CUDACC__)
>+// __CUDACC_VER__ is a full version number before CUDA 9, and is defined to a
>+// std::string explaining that it has been removed starting with CUDA 9. We use
>+// nested #ifs because there is no short-circuiting in the preprocessor.
>+// NOTE: `__CUDACC__` could be undefined while `__CUDACC_VER__` is defined.
>+#if __CUDACC_VER__ >= 70000
>+#define ABSL_HAVE_INTRINSIC_INT128 1
>+#endif  // __CUDACC_VER__ >= 70000
>+#endif  // defined(__CUDACC__)
>+#endif  // ABSL_HAVE_INTRINSIC_INT128
>+
>+// ABSL_HAVE_EXCEPTIONS
>+//
>+// Checks whether the compiler both supports and enables exceptions. Many
>+// compilers support a "no exceptions" mode that disables exceptions.
>+//
>+// Generally, when ABSL_HAVE_EXCEPTIONS is not defined:
>+//
>+// * Code using `throw` and `try` may not compile.
>+// * The `noexcept` specifier will still compile and behave as normal.
>+// * The `noexcept` operator may still return `false`.
>+//
>+// For further details, consult the compiler's documentation.
>+#ifdef ABSL_HAVE_EXCEPTIONS
>+#error ABSL_HAVE_EXCEPTIONS cannot be directly set.
>+
>+#elif defined(__clang__)
>+// TODO(calabrese)
>+// Switch to using __cpp_exceptions when we no longer support versions < 3.6.
>+// For details on this check, see:
>+//   http://releases.llvm.org/3.6.0/tools/clang/docs/ReleaseNotes.html#the-exceptions-macro
>+#if defined(__EXCEPTIONS) && __has_feature(cxx_exceptions)
>+#define ABSL_HAVE_EXCEPTIONS 1
>+#endif  // defined(__EXCEPTIONS) && __has_feature(cxx_exceptions)
>+
>+// Handle remaining special cases and default to exceptions being supported.
>+#elif !(defined(__GNUC__) && (__GNUC__ < 5) && !defined(__EXCEPTIONS)) &&    \
>+    !(defined(__GNUC__) && (__GNUC__ >= 5) && !defined(__cpp_exceptions)) && \
>+    !(defined(_MSC_VER) && !defined(_CPPUNWIND))
>+#define ABSL_HAVE_EXCEPTIONS 1
>+#endif
>+
>+// -----------------------------------------------------------------------------
>+// Platform Feature Checks
>+// -----------------------------------------------------------------------------
>+
>+// Currently supported operating systems and associated preprocessor
>+// symbols:
>+//
>+//   Linux and Linux-derived           __linux__
>+//   Android                           __ANDROID__ (implies __linux__)
>+//   Linux (non-Android)               __linux__ && !__ANDROID__
>+//   Darwin (Mac OS X and iOS)         __APPLE__
>+//   Akaros (http://akaros.org)        __ros__
>+//   Windows                           _WIN32
>+//   NaCL                              __native_client__
>+//   AsmJS                             __asmjs__
>+//   WebAssembly                       __wasm__
>+//   Fuchsia                           __Fuchsia__
>+//
>+// Note that since Android defines both __ANDROID__ and __linux__, one
>+// may probe for either Linux or Android by simply testing for __linux__.
>+
>+// ABSL_HAVE_MMAP
>+//
>+// Checks whether the platform has an mmap(2) implementation as defined in
>+// POSIX.1-2001.
>+#ifdef ABSL_HAVE_MMAP
>+#error ABSL_HAVE_MMAP cannot be directly set
>+#elif defined(__linux__) || defined(__APPLE__) || defined(__FreeBSD__) ||   \
>+    defined(__ros__) || defined(__native_client__) || defined(__asmjs__) || \
>+    defined(__wasm__) || defined(__Fuchsia__) || defined(__sun)
>+#define ABSL_HAVE_MMAP 1
>+#endif
>+
>+// ABSL_HAVE_PTHREAD_GETSCHEDPARAM
>+//
>+// Checks whether the platform implements the pthread_(get|set)schedparam(3)
>+// functions as defined in POSIX.1-2001.
>+#ifdef ABSL_HAVE_PTHREAD_GETSCHEDPARAM
>+#error ABSL_HAVE_PTHREAD_GETSCHEDPARAM cannot be directly set
>+#elif defined(__linux__) || defined(__APPLE__) || defined(__FreeBSD__) || \
>+    defined(__ros__)
>+#define ABSL_HAVE_PTHREAD_GETSCHEDPARAM 1
>+#endif
>+
>+// ABSL_HAVE_SCHED_YIELD
>+//
>+// Checks whether the platform implements sched_yield(2) as defined in
>+// POSIX.1-2001.
>+#ifdef ABSL_HAVE_SCHED_YIELD
>+#error ABSL_HAVE_SCHED_YIELD cannot be directly set
>+#elif defined(__linux__) || defined(__ros__) || defined(__native_client__)
>+#define ABSL_HAVE_SCHED_YIELD 1
>+#endif
>+
>+// ABSL_HAVE_SEMAPHORE_H
>+//
>+// Checks whether the platform supports the <semaphore.h> header and sem_open(3)
>+// family of functions as standardized in POSIX.1-2001.
>+//
>+// Note: While Apple provides <semaphore.h> for both iOS and macOS, it is
>+// explicitly deprecated and will cause build failures if enabled for those
>+// platforms.  We side-step the issue by not defining it here for Apple
>+// platforms.
>+#ifdef ABSL_HAVE_SEMAPHORE_H
>+#error ABSL_HAVE_SEMAPHORE_H cannot be directly set
>+#elif defined(__linux__) || defined(__ros__)
>+#define ABSL_HAVE_SEMAPHORE_H 1
>+#endif
>+
>+// ABSL_HAVE_ALARM
>+//
>+// Checks whether the platform supports the <signal.h> header and alarm(2)
>+// function as standardized in POSIX.1-2001.
>+#ifdef ABSL_HAVE_ALARM
>+#error ABSL_HAVE_ALARM cannot be directly set
>+#elif defined(__GOOGLE_GRTE_VERSION__)
>+// feature tests for Google's GRTE
>+#define ABSL_HAVE_ALARM 1
>+#elif defined(__GLIBC__)
>+// feature test for glibc
>+#define ABSL_HAVE_ALARM 1
>+#elif defined(_MSC_VER)
>+// feature tests for Microsoft's library
>+#elif defined(__native_client__)
>+#else
>+// other standard libraries
>+#define ABSL_HAVE_ALARM 1
>+#endif
>+
>+// ABSL_IS_LITTLE_ENDIAN
>+// ABSL_IS_BIG_ENDIAN
>+//
>+// Checks the endianness of the platform.
>+//
>+// Notes: uses the built in endian macros provided by GCC (since 4.6) and
>+// Clang (since 3.2); see
>+// https://gcc.gnu.org/onlinedocs/cpp/Common-Predefined-Macros.html.
>+// Otherwise, if _WIN32, assume little endian. Otherwise, bail with an error.
>+#if defined(ABSL_IS_BIG_ENDIAN)
>+#error "ABSL_IS_BIG_ENDIAN cannot be directly set."
>+#endif
>+#if defined(ABSL_IS_LITTLE_ENDIAN)
>+#error "ABSL_IS_LITTLE_ENDIAN cannot be directly set."
>+#endif
>+
>+#if (defined(__BYTE_ORDER__) && defined(__ORDER_LITTLE_ENDIAN__) && \
>+     __BYTE_ORDER__ == __ORDER_LITTLE_ENDIAN__)
>+#define ABSL_IS_LITTLE_ENDIAN 1
>+#elif defined(__BYTE_ORDER__) && defined(__ORDER_BIG_ENDIAN__) && \
>+    __BYTE_ORDER__ == __ORDER_BIG_ENDIAN__
>+#define ABSL_IS_BIG_ENDIAN 1
>+#elif defined(_WIN32)
>+#define ABSL_IS_LITTLE_ENDIAN 1
>+#else
>+#error "absl endian detection needs to be set up for your compiler"
>+#endif
>+
>+// ABSL_HAVE_STD_ANY
>+//
>+// Checks whether C++17 std::any is available by checking whether <any> exists.
>+#ifdef ABSL_HAVE_STD_ANY
>+#error "ABSL_HAVE_STD_ANY cannot be directly set."
>+#endif
>+
>+#ifdef __has_include
>+#if __has_include(<any>) && __cplusplus >= 201703L
>+#define ABSL_HAVE_STD_ANY 1
>+#endif
>+#endif
>+
>+// ABSL_HAVE_STD_OPTIONAL
>+//
>+// Checks whether C++17 std::optional is available.
>+#ifdef ABSL_HAVE_STD_OPTIONAL
>+#error "ABSL_HAVE_STD_OPTIONAL cannot be directly set."
>+#endif
>+
>+#ifdef __has_include
>+#if __has_include(<optional>) && __cplusplus >= 201703L
>+#define ABSL_HAVE_STD_OPTIONAL 1
>+#endif
>+#endif
>+
>+// ABSL_HAVE_STD_VARIANT
>+//
>+// Checks whether C++17 std::variant is available.
>+#ifdef ABSL_HAVE_STD_VARIANT
>+#error "ABSL_HAVE_STD_VARIANT cannot be directly set."
>+#endif
>+
>+#ifdef __has_include
>+#if __has_include(<variant>) && __cplusplus >= 201703L
>+#define ABSL_HAVE_STD_VARIANT 1
>+#endif
>+#endif
>+
>+// ABSL_HAVE_STD_STRING_VIEW
>+//
>+// Checks whether C++17 std::string_view is available.
>+#ifdef ABSL_HAVE_STD_STRING_VIEW
>+#error "ABSL_HAVE_STD_STRING_VIEW cannot be directly set."
>+#endif
>+
>+#ifdef __has_include
>+#if __has_include(<string_view>) && __cplusplus >= 201703L
>+#define ABSL_HAVE_STD_STRING_VIEW 1
>+#endif
>+#endif
>+
>+// For MSVC, `__has_include` is supported in VS 2017 15.3, which is later than
>+// the support for <optional>, <any>, <string_view>, <variant>. So we use
>+// _MSC_VER to check whether we have VS 2017 RTM (when <optional>, <any>,
>+// <string_view>, <variant> is implemented) or higher. Also, `__cplusplus` is
>+// not correctly set by MSVC, so we use `_MSVC_LANG` to check the language
>+// version.
>+// TODO(zhangxy): fix tests before enabling aliasing for `std::any`,
>+// `std::string_view`.
>+#if defined(_MSC_VER) && _MSC_VER >= 1910 && \
>+    ((defined(_MSVC_LANG) && _MSVC_LANG > 201402) || __cplusplus > 201402)
>+// #define ABSL_HAVE_STD_ANY 1
>+#define ABSL_HAVE_STD_OPTIONAL 1
>+#define ABSL_HAVE_STD_VARIANT 1
>+// #define ABSL_HAVE_STD_STRING_VIEW 1
>+#endif
>+
>+#endif  // ABSL_BASE_CONFIG_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/config_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/config_test.cc
>new file mode 100644
>index 00000000000..c839712a537
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/config_test.cc
>@@ -0,0 +1,60 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/base/config.h"
>+
>+#include <cstdint>
>+
>+#include "gtest/gtest.h"
>+#include "absl/synchronization/internal/thread_pool.h"
>+
>+namespace {
>+
>+TEST(ConfigTest, Endianness) {
>+  union {
>+    uint32_t value;
>+    uint8_t data[sizeof(uint32_t)];
>+  } number;
>+  number.data[0] = 0x00;
>+  number.data[1] = 0x01;
>+  number.data[2] = 0x02;
>+  number.data[3] = 0x03;
>+#if defined(ABSL_IS_LITTLE_ENDIAN) && defined(ABSL_IS_BIG_ENDIAN)
>+#error Both ABSL_IS_LITTLE_ENDIAN and ABSL_IS_BIG_ENDIAN are defined
>+#elif defined(ABSL_IS_LITTLE_ENDIAN)
>+  EXPECT_EQ(UINT32_C(0x03020100), number.value);
>+#elif defined(ABSL_IS_BIG_ENDIAN)
>+  EXPECT_EQ(UINT32_C(0x00010203), number.value);
>+#else
>+#error Unknown endianness
>+#endif
>+}
>+
>+#if defined(ABSL_HAVE_THREAD_LOCAL)
>+TEST(ConfigTest, ThreadLocal) {
>+  static thread_local int mine_mine_mine = 16;
>+  EXPECT_EQ(16, mine_mine_mine);
>+  {
>+    absl::synchronization_internal::ThreadPool pool(1);
>+    pool.Schedule([&] {
>+      EXPECT_EQ(16, mine_mine_mine);
>+      mine_mine_mine = 32;
>+      EXPECT_EQ(32, mine_mine_mine);
>+    });
>+  }
>+  EXPECT_EQ(16, mine_mine_mine);
>+}
>+#endif
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/dynamic_annotations.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/dynamic_annotations.cc
>new file mode 100644
>index 00000000000..b97fa3a8b4a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/dynamic_annotations.cc
>@@ -0,0 +1,129 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include <stdlib.h>
>+#include <string.h>
>+
>+#include "absl/base/dynamic_annotations.h"
>+
>+#ifndef __has_feature
>+#define __has_feature(x) 0
>+#endif
>+
>+/* Compiler-based ThreadSanitizer defines
>+   ABSL_DYNAMIC_ANNOTATIONS_EXTERNAL_IMPL = 1
>+   and provides its own definitions of the functions. */
>+
>+#ifndef ABSL_DYNAMIC_ANNOTATIONS_EXTERNAL_IMPL
>+# define ABSL_DYNAMIC_ANNOTATIONS_EXTERNAL_IMPL 0
>+#endif
>+
>+/* Each function is empty and called (via a macro) only in debug mode.
>+   The arguments are captured by dynamic tools at runtime. */
>+
>+#if ABSL_DYNAMIC_ANNOTATIONS_EXTERNAL_IMPL == 0 && !defined(__native_client__)
>+
>+#if __has_feature(memory_sanitizer)
>+#include <sanitizer/msan_interface.h>
>+#endif
>+
>+#ifdef __cplusplus
>+extern "C" {
>+#endif
>+
>+void AbslAnnotateRWLockCreate(const char *, int,
>+                          const volatile void *){}
>+void AbslAnnotateRWLockDestroy(const char *, int,
>+                           const volatile void *){}
>+void AbslAnnotateRWLockAcquired(const char *, int,
>+                            const volatile void *, long){}
>+void AbslAnnotateRWLockReleased(const char *, int,
>+                            const volatile void *, long){}
>+void AbslAnnotateBenignRace(const char *, int,
>+                        const volatile void *,
>+                        const char *){}
>+void AbslAnnotateBenignRaceSized(const char *, int,
>+                             const volatile void *,
>+                             size_t,
>+                             const char *) {}
>+void AbslAnnotateThreadName(const char *, int,
>+                        const char *){}
>+void AbslAnnotateIgnoreReadsBegin(const char *, int){}
>+void AbslAnnotateIgnoreReadsEnd(const char *, int){}
>+void AbslAnnotateIgnoreWritesBegin(const char *, int){}
>+void AbslAnnotateIgnoreWritesEnd(const char *, int){}
>+void AbslAnnotateEnableRaceDetection(const char *, int, int){}
>+void AbslAnnotateMemoryIsInitialized(const char *, int,
>+                                 const volatile void *mem, size_t size) {
>+#if __has_feature(memory_sanitizer)
>+  __msan_unpoison(mem, size);
>+#else
>+  (void)mem;
>+  (void)size;
>+#endif
>+}
>+
>+void AbslAnnotateMemoryIsUninitialized(const char *, int,
>+                                   const volatile void *mem, size_t size) {
>+#if __has_feature(memory_sanitizer)
>+  __msan_allocated_memory(mem, size);
>+#else
>+  (void)mem;
>+  (void)size;
>+#endif
>+}
>+
>+static int AbslGetRunningOnValgrind(void) {
>+#ifdef RUNNING_ON_VALGRIND
>+  if (RUNNING_ON_VALGRIND) return 1;
>+#endif
>+  char *running_on_valgrind_str = getenv("RUNNING_ON_VALGRIND");
>+  if (running_on_valgrind_str) {
>+    return strcmp(running_on_valgrind_str, "0") != 0;
>+  }
>+  return 0;
>+}
>+
>+/* See the comments in dynamic_annotations.h */
>+int AbslRunningOnValgrind(void) {
>+  static volatile int running_on_valgrind = -1;
>+  int local_running_on_valgrind = running_on_valgrind;
>+  /* C doesn't have thread-safe initialization of statics, and we
>+     don't want to depend on pthread_once here, so hack it. */
>+  ABSL_ANNOTATE_BENIGN_RACE(&running_on_valgrind, "safe hack");
>+  if (local_running_on_valgrind == -1)
>+    running_on_valgrind = local_running_on_valgrind = AbslGetRunningOnValgrind();
>+  return local_running_on_valgrind;
>+}
>+
>+/* See the comments in dynamic_annotations.h */
>+double AbslValgrindSlowdown(void) {
>+  /* Same initialization hack as in AbslRunningOnValgrind(). */
>+  static volatile double slowdown = 0.0;
>+  double local_slowdown = slowdown;
>+  ABSL_ANNOTATE_BENIGN_RACE(&slowdown, "safe hack");
>+  if (AbslRunningOnValgrind() == 0) {
>+    return 1.0;
>+  }
>+  if (local_slowdown == 0.0) {
>+    char *env = getenv("VALGRIND_SLOWDOWN");
>+    slowdown = local_slowdown = env ? atof(env) : 50.0;
>+  }
>+  return local_slowdown;
>+}
>+
>+#ifdef __cplusplus
>+}  // extern "C"
>+#endif
>+#endif  /* ABSL_DYNAMIC_ANNOTATIONS_EXTERNAL_IMPL == 0 */
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/dynamic_annotations.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/dynamic_annotations.h
>new file mode 100644
>index 00000000000..88048b0fbb1
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/dynamic_annotations.h
>@@ -0,0 +1,388 @@
>+/*
>+ *  Copyright 2017 The Abseil Authors.
>+ *
>+ * Licensed under the Apache License, Version 2.0 (the "License");
>+ * you may not use this file except in compliance with the License.
>+ * You may obtain a copy of the License at
>+ *
>+ *      http://www.apache.org/licenses/LICENSE-2.0
>+ *
>+ * Unless required by applicable law or agreed to in writing, software
>+ * distributed under the License is distributed on an "AS IS" BASIS,
>+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+ * See the License for the specific language governing permissions and
>+ * limitations under the License.
>+ */
>+/* This file defines dynamic annotations for use with dynamic analysis
>+   tool such as valgrind, PIN, etc.
>+
>+   Dynamic annotation is a source code annotation that affects
>+   the generated code (that is, the annotation is not a comment).
>+   Each such annotation is attached to a particular
>+   instruction and/or to a particular object (address) in the program.
>+
>+   The annotations that should be used by users are macros in all upper-case
>+   (e.g., ABSL_ANNOTATE_THREAD_NAME).
>+
>+   Actual implementation of these macros may differ depending on the
>+   dynamic analysis tool being used.
>+
>+   This file supports the following configurations:
>+   - Dynamic Annotations enabled (with static thread-safety warnings disabled).
>+     In this case, macros expand to functions implemented by Thread Sanitizer,
>+     when building with TSan. When not provided an external implementation,
>+     dynamic_annotations.cc provides no-op implementations.
>+
>+   - Static Clang thread-safety warnings enabled.
>+     When building with a Clang compiler that supports thread-safety warnings,
>+     a subset of annotations can be statically-checked at compile-time. We
>+     expand these macros to static-inline functions that can be analyzed for
>+     thread-safety, but afterwards elided when building the final binary.
>+
>+   - All annotations are disabled.
>+     If neither Dynamic Annotations nor Clang thread-safety warnings are
>+     enabled, then all annotation-macros expand to empty. */
>+
>+#ifndef ABSL_BASE_DYNAMIC_ANNOTATIONS_H_
>+#define ABSL_BASE_DYNAMIC_ANNOTATIONS_H_
>+
>+#ifndef ABSL_DYNAMIC_ANNOTATIONS_ENABLED
>+# define ABSL_DYNAMIC_ANNOTATIONS_ENABLED 0
>+#endif
>+
>+#if ABSL_DYNAMIC_ANNOTATIONS_ENABLED != 0
>+
>+  /* -------------------------------------------------------------
>+     Annotations that suppress errors.  It is usually better to express the
>+     program's synchronization using the other annotations, but these can
>+     be used when all else fails. */
>+
>+  /* Report that we may have a benign race at "pointer", with size
>+     "sizeof(*(pointer))". "pointer" must be a non-void* pointer.  Insert at the
>+     point where "pointer" has been allocated, preferably close to the point
>+     where the race happens.  See also ABSL_ANNOTATE_BENIGN_RACE_STATIC. */
>+  #define ABSL_ANNOTATE_BENIGN_RACE(pointer, description) \
>+    AbslAnnotateBenignRaceSized(__FILE__, __LINE__, pointer, \
>+                            sizeof(*(pointer)), description)
>+
>+  /* Same as ABSL_ANNOTATE_BENIGN_RACE(address, description), but applies to
>+     the memory range [address, address+size). */
>+  #define ABSL_ANNOTATE_BENIGN_RACE_SIZED(address, size, description) \
>+    AbslAnnotateBenignRaceSized(__FILE__, __LINE__, address, size, description)
>+
>+  /* Enable (enable!=0) or disable (enable==0) race detection for all threads.
>+     This annotation could be useful if you want to skip expensive race analysis
>+     during some period of program execution, e.g. during initialization. */
>+  #define ABSL_ANNOTATE_ENABLE_RACE_DETECTION(enable) \
>+    AbslAnnotateEnableRaceDetection(__FILE__, __LINE__, enable)
>+
>+  /* -------------------------------------------------------------
>+     Annotations useful for debugging. */
>+
>+  /* Report the current thread name to a race detector. */
>+  #define ABSL_ANNOTATE_THREAD_NAME(name) \
>+    AbslAnnotateThreadName(__FILE__, __LINE__, name)
>+
>+  /* -------------------------------------------------------------
>+     Annotations useful when implementing locks.  They are not
>+     normally needed by modules that merely use locks.
>+     The "lock" argument is a pointer to the lock object. */
>+
>+  /* Report that a lock has been created at address "lock". */
>+  #define ABSL_ANNOTATE_RWLOCK_CREATE(lock) \
>+    AbslAnnotateRWLockCreate(__FILE__, __LINE__, lock)
>+
>+  /* Report that a linker initialized lock has been created at address "lock".
>+   */
>+#ifdef THREAD_SANITIZER
>+  #define ABSL_ANNOTATE_RWLOCK_CREATE_STATIC(lock) \
>+    AbslAnnotateRWLockCreateStatic(__FILE__, __LINE__, lock)
>+#else
>+  #define ABSL_ANNOTATE_RWLOCK_CREATE_STATIC(lock) ABSL_ANNOTATE_RWLOCK_CREATE(lock)
>+#endif
>+
>+  /* Report that the lock at address "lock" is about to be destroyed. */
>+  #define ABSL_ANNOTATE_RWLOCK_DESTROY(lock) \
>+    AbslAnnotateRWLockDestroy(__FILE__, __LINE__, lock)
>+
>+  /* Report that the lock at address "lock" has been acquired.
>+     is_w=1 for writer lock, is_w=0 for reader lock. */
>+  #define ABSL_ANNOTATE_RWLOCK_ACQUIRED(lock, is_w) \
>+    AbslAnnotateRWLockAcquired(__FILE__, __LINE__, lock, is_w)
>+
>+  /* Report that the lock at address "lock" is about to be released. */
>+  #define ABSL_ANNOTATE_RWLOCK_RELEASED(lock, is_w) \
>+    AbslAnnotateRWLockReleased(__FILE__, __LINE__, lock, is_w)
>+
>+#else  /* ABSL_DYNAMIC_ANNOTATIONS_ENABLED == 0 */
>+
>+  #define ABSL_ANNOTATE_RWLOCK_CREATE(lock) /* empty */
>+  #define ABSL_ANNOTATE_RWLOCK_CREATE_STATIC(lock) /* empty */
>+  #define ABSL_ANNOTATE_RWLOCK_DESTROY(lock) /* empty */
>+  #define ABSL_ANNOTATE_RWLOCK_ACQUIRED(lock, is_w) /* empty */
>+  #define ABSL_ANNOTATE_RWLOCK_RELEASED(lock, is_w) /* empty */
>+  #define ABSL_ANNOTATE_BENIGN_RACE(address, description) /* empty */
>+  #define ABSL_ANNOTATE_BENIGN_RACE_SIZED(address, size, description) /* empty */
>+  #define ABSL_ANNOTATE_THREAD_NAME(name) /* empty */
>+  #define ABSL_ANNOTATE_ENABLE_RACE_DETECTION(enable) /* empty */
>+
>+#endif  /* ABSL_DYNAMIC_ANNOTATIONS_ENABLED */
>+
>+/* These annotations are also made available to LLVM's Memory Sanitizer */
>+#if ABSL_DYNAMIC_ANNOTATIONS_ENABLED == 1 || defined(MEMORY_SANITIZER)
>+  #define ABSL_ANNOTATE_MEMORY_IS_INITIALIZED(address, size) \
>+    AbslAnnotateMemoryIsInitialized(__FILE__, __LINE__, address, size)
>+
>+  #define ABSL_ANNOTATE_MEMORY_IS_UNINITIALIZED(address, size) \
>+    AbslAnnotateMemoryIsUninitialized(__FILE__, __LINE__, address, size)
>+#else
>+  #define ABSL_ANNOTATE_MEMORY_IS_INITIALIZED(address, size) /* empty */
>+  #define ABSL_ANNOTATE_MEMORY_IS_UNINITIALIZED(address, size) /* empty */
>+#endif  /* ABSL_DYNAMIC_ANNOTATIONS_ENABLED || MEMORY_SANITIZER */
>+/* TODO(delesley) -- Replace __CLANG_SUPPORT_DYN_ANNOTATION__ with the
>+   appropriate feature ID. */
>+#if defined(__clang__) && (!defined(SWIG)) \
>+    && defined(__CLANG_SUPPORT_DYN_ANNOTATION__)
>+
>+  #if ABSL_DYNAMIC_ANNOTATIONS_ENABLED == 0
>+    #define ABSL_ANNOTALYSIS_ENABLED
>+  #endif
>+
>+  /* When running in opt-mode, GCC will issue a warning, if these attributes are
>+     compiled. Only include them when compiling using Clang. */
>+  #define ABSL_ATTRIBUTE_IGNORE_READS_BEGIN \
>+      __attribute((exclusive_lock_function("*")))
>+  #define ABSL_ATTRIBUTE_IGNORE_READS_END \
>+      __attribute((unlock_function("*")))
>+#else
>+  #define ABSL_ATTRIBUTE_IGNORE_READS_BEGIN  /* empty */
>+  #define ABSL_ATTRIBUTE_IGNORE_READS_END  /* empty */
>+#endif  /* defined(__clang__) && ... */
>+
>+#if (ABSL_DYNAMIC_ANNOTATIONS_ENABLED != 0) || defined(ABSL_ANNOTALYSIS_ENABLED)
>+  #define ABSL_ANNOTATIONS_ENABLED
>+#endif
>+
>+#if (ABSL_DYNAMIC_ANNOTATIONS_ENABLED != 0)
>+
>+  /* Request the analysis tool to ignore all reads in the current thread
>+     until ABSL_ANNOTATE_IGNORE_READS_END is called.
>+     Useful to ignore intentional racey reads, while still checking
>+     other reads and all writes.
>+     See also ABSL_ANNOTATE_UNPROTECTED_READ. */
>+  #define ABSL_ANNOTATE_IGNORE_READS_BEGIN() \
>+    AbslAnnotateIgnoreReadsBegin(__FILE__, __LINE__)
>+
>+  /* Stop ignoring reads. */
>+  #define ABSL_ANNOTATE_IGNORE_READS_END() \
>+    AbslAnnotateIgnoreReadsEnd(__FILE__, __LINE__)
>+
>+  /* Similar to ABSL_ANNOTATE_IGNORE_READS_BEGIN, but ignore writes instead. */
>+  #define ABSL_ANNOTATE_IGNORE_WRITES_BEGIN() \
>+    AbslAnnotateIgnoreWritesBegin(__FILE__, __LINE__)
>+
>+  /* Stop ignoring writes. */
>+  #define ABSL_ANNOTATE_IGNORE_WRITES_END() \
>+    AbslAnnotateIgnoreWritesEnd(__FILE__, __LINE__)
>+
>+/* Clang provides limited support for static thread-safety analysis
>+   through a feature called Annotalysis. We configure macro-definitions
>+   according to whether Annotalysis support is available. */
>+#elif defined(ABSL_ANNOTALYSIS_ENABLED)
>+
>+  #define ABSL_ANNOTATE_IGNORE_READS_BEGIN() \
>+    AbslStaticAnnotateIgnoreReadsBegin(__FILE__, __LINE__)
>+
>+  #define ABSL_ANNOTATE_IGNORE_READS_END() \
>+    AbslStaticAnnotateIgnoreReadsEnd(__FILE__, __LINE__)
>+
>+  #define ABSL_ANNOTATE_IGNORE_WRITES_BEGIN() \
>+    AbslStaticAnnotateIgnoreWritesBegin(__FILE__, __LINE__)
>+
>+  #define ABSL_ANNOTATE_IGNORE_WRITES_END() \
>+    AbslStaticAnnotateIgnoreWritesEnd(__FILE__, __LINE__)
>+
>+#else
>+  #define ABSL_ANNOTATE_IGNORE_READS_BEGIN()  /* empty */
>+  #define ABSL_ANNOTATE_IGNORE_READS_END()  /* empty */
>+  #define ABSL_ANNOTATE_IGNORE_WRITES_BEGIN()  /* empty */
>+  #define ABSL_ANNOTATE_IGNORE_WRITES_END()  /* empty */
>+#endif
>+
>+/* Implement the ANNOTATE_IGNORE_READS_AND_WRITES_* annotations using the more
>+   primitive annotations defined above. */
>+#if defined(ABSL_ANNOTATIONS_ENABLED)
>+
>+  /* Start ignoring all memory accesses (both reads and writes). */
>+  #define ABSL_ANNOTATE_IGNORE_READS_AND_WRITES_BEGIN() \
>+    do {                                           \
>+      ABSL_ANNOTATE_IGNORE_READS_BEGIN();               \
>+      ABSL_ANNOTATE_IGNORE_WRITES_BEGIN();              \
>+    }while (0)
>+
>+  /* Stop ignoring both reads and writes. */
>+  #define ABSL_ANNOTATE_IGNORE_READS_AND_WRITES_END()   \
>+    do {                                           \
>+      ABSL_ANNOTATE_IGNORE_WRITES_END();                \
>+      ABSL_ANNOTATE_IGNORE_READS_END();                 \
>+    }while (0)
>+
>+#else
>+  #define ABSL_ANNOTATE_IGNORE_READS_AND_WRITES_BEGIN()  /* empty */
>+  #define ABSL_ANNOTATE_IGNORE_READS_AND_WRITES_END()  /* empty */
>+#endif
>+
>+/* Use the macros above rather than using these functions directly. */
>+#include <stddef.h>
>+#ifdef __cplusplus
>+extern "C" {
>+#endif
>+void AbslAnnotateRWLockCreate(const char *file, int line,
>+                          const volatile void *lock);
>+void AbslAnnotateRWLockCreateStatic(const char *file, int line,
>+                          const volatile void *lock);
>+void AbslAnnotateRWLockDestroy(const char *file, int line,
>+                           const volatile void *lock);
>+void AbslAnnotateRWLockAcquired(const char *file, int line,
>+                            const volatile void *lock, long is_w);  /* NOLINT */
>+void AbslAnnotateRWLockReleased(const char *file, int line,
>+                            const volatile void *lock, long is_w);  /* NOLINT */
>+void AbslAnnotateBenignRace(const char *file, int line,
>+                        const volatile void *address,
>+                        const char *description);
>+void AbslAnnotateBenignRaceSized(const char *file, int line,
>+                        const volatile void *address,
>+                        size_t size,
>+                        const char *description);
>+void AbslAnnotateThreadName(const char *file, int line,
>+                        const char *name);
>+void AbslAnnotateEnableRaceDetection(const char *file, int line, int enable);
>+void AbslAnnotateMemoryIsInitialized(const char *file, int line,
>+                                 const volatile void *mem, size_t size);
>+void AbslAnnotateMemoryIsUninitialized(const char *file, int line,
>+                                   const volatile void *mem, size_t size);
>+
>+/* Annotations expand to these functions, when Dynamic Annotations are enabled.
>+   These functions are either implemented as no-op calls, if no Sanitizer is
>+   attached, or provided with externally-linked implementations by a library
>+   like ThreadSanitizer. */
>+void AbslAnnotateIgnoreReadsBegin(const char *file, int line)
>+    ABSL_ATTRIBUTE_IGNORE_READS_BEGIN;
>+void AbslAnnotateIgnoreReadsEnd(const char *file, int line)
>+    ABSL_ATTRIBUTE_IGNORE_READS_END;
>+void AbslAnnotateIgnoreWritesBegin(const char *file, int line);
>+void AbslAnnotateIgnoreWritesEnd(const char *file, int line);
>+
>+#if defined(ABSL_ANNOTALYSIS_ENABLED)
>+/* When Annotalysis is enabled without Dynamic Annotations, the use of
>+   static-inline functions allows the annotations to be read at compile-time,
>+   while still letting the compiler elide the functions from the final build.
>+
>+   TODO(delesley) -- The exclusive lock here ignores writes as well, but
>+   allows IGNORE_READS_AND_WRITES to work properly. */
>+#pragma GCC diagnostic push
>+#pragma GCC diagnostic ignored "-Wunused-function"
>+static inline void AbslStaticAnnotateIgnoreReadsBegin(const char *file, int line)
>+    ABSL_ATTRIBUTE_IGNORE_READS_BEGIN { (void)file; (void)line; }
>+static inline void AbslStaticAnnotateIgnoreReadsEnd(const char *file, int line)
>+    ABSL_ATTRIBUTE_IGNORE_READS_END { (void)file; (void)line; }
>+static inline void AbslStaticAnnotateIgnoreWritesBegin(
>+    const char *file, int line) { (void)file; (void)line; }
>+static inline void AbslStaticAnnotateIgnoreWritesEnd(
>+    const char *file, int line) { (void)file; (void)line; }
>+#pragma GCC diagnostic pop
>+#endif
>+
>+/* Return non-zero value if running under valgrind.
>+
>+  If "valgrind.h" is included into dynamic_annotations.cc,
>+  the regular valgrind mechanism will be used.
>+  See http://valgrind.org/docs/manual/manual-core-adv.html about
>+  RUNNING_ON_VALGRIND and other valgrind "client requests".
>+  The file "valgrind.h" may be obtained by doing
>+     svn co svn://svn.valgrind.org/valgrind/trunk/include
>+
>+  If for some reason you can't use "valgrind.h" or want to fake valgrind,
>+  there are two ways to make this function return non-zero:
>+    - Use environment variable: export RUNNING_ON_VALGRIND=1
>+    - Make your tool intercept the function AbslRunningOnValgrind() and
>+      change its return value.
>+ */
>+int AbslRunningOnValgrind(void);
>+
>+/* AbslValgrindSlowdown returns:
>+    * 1.0, if (AbslRunningOnValgrind() == 0)
>+    * 50.0, if (AbslRunningOnValgrind() != 0 && getenv("VALGRIND_SLOWDOWN") == NULL)
>+    * atof(getenv("VALGRIND_SLOWDOWN")) otherwise
>+   This function can be used to scale timeout values:
>+   EXAMPLE:
>+   for (;;) {
>+     DoExpensiveBackgroundTask();
>+     SleepForSeconds(5 * AbslValgrindSlowdown());
>+   }
>+ */
>+double AbslValgrindSlowdown(void);
>+
>+#ifdef __cplusplus
>+}
>+#endif
>+
>+/* ABSL_ANNOTATE_UNPROTECTED_READ is the preferred way to annotate racey reads.
>+
>+     Instead of doing
>+        ABSL_ANNOTATE_IGNORE_READS_BEGIN();
>+        ... = x;
>+        ABSL_ANNOTATE_IGNORE_READS_END();
>+     one can use
>+        ... = ABSL_ANNOTATE_UNPROTECTED_READ(x); */
>+#if defined(__cplusplus) && defined(ABSL_ANNOTATIONS_ENABLED)
>+template <typename T>
>+inline T ABSL_ANNOTATE_UNPROTECTED_READ(const volatile T &x) { /* NOLINT */
>+  ABSL_ANNOTATE_IGNORE_READS_BEGIN();
>+  T res = x;
>+  ABSL_ANNOTATE_IGNORE_READS_END();
>+  return res;
>+  }
>+#else
>+  #define ABSL_ANNOTATE_UNPROTECTED_READ(x) (x)
>+#endif
>+
>+#if ABSL_DYNAMIC_ANNOTATIONS_ENABLED != 0 && defined(__cplusplus)
>+  /* Apply ABSL_ANNOTATE_BENIGN_RACE_SIZED to a static variable. */
>+  #define ABSL_ANNOTATE_BENIGN_RACE_STATIC(static_var, description)        \
>+    namespace {                                                       \
>+      class static_var ## _annotator {                                \
>+       public:                                                        \
>+        static_var ## _annotator() {                                  \
>+          ABSL_ANNOTATE_BENIGN_RACE_SIZED(&static_var,                     \
>+                                      sizeof(static_var),             \
>+            # static_var ": " description);                           \
>+        }                                                             \
>+      };                                                              \
>+      static static_var ## _annotator the ## static_var ## _annotator;\
>+    }  // namespace
>+#else /* ABSL_DYNAMIC_ANNOTATIONS_ENABLED == 0 */
>+  #define ABSL_ANNOTATE_BENIGN_RACE_STATIC(static_var, description)  /* empty */
>+#endif /* ABSL_DYNAMIC_ANNOTATIONS_ENABLED */
>+
>+#ifdef ADDRESS_SANITIZER
>+/* Describe the current state of a contiguous container such as e.g.
>+ * std::vector or std::string. For more details see
>+ * sanitizer/common_interface_defs.h, which is provided by the compiler. */
>+#include <sanitizer/common_interface_defs.h>
>+#define ABSL_ANNOTATE_CONTIGUOUS_CONTAINER(beg, end, old_mid, new_mid) \
>+  __sanitizer_annotate_contiguous_container(beg, end, old_mid, new_mid)
>+#define ABSL_ADDRESS_SANITIZER_REDZONE(name)         \
>+  struct { char x[8] __attribute__ ((aligned (8))); } name
>+#else
>+#define ABSL_ANNOTATE_CONTIGUOUS_CONTAINER(beg, end, old_mid, new_mid)
>+#define ABSL_ADDRESS_SANITIZER_REDZONE(name)
>+#endif  // ADDRESS_SANITIZER
>+
>+/* Undefine the macros intended only in this file. */
>+#undef ABSL_ANNOTALYSIS_ENABLED
>+#undef ABSL_ANNOTATIONS_ENABLED
>+#undef ABSL_ATTRIBUTE_IGNORE_READS_BEGIN
>+#undef ABSL_ATTRIBUTE_IGNORE_READS_END
>+
>+#endif  /* ABSL_BASE_DYNAMIC_ANNOTATIONS_H_ */
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/exception_safety_testing_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/exception_safety_testing_test.cc
>new file mode 100644
>index 00000000000..97c8d6f8318
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/exception_safety_testing_test.cc
>@@ -0,0 +1,942 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/base/internal/exception_safety_testing.h"
>+
>+#include <cstddef>
>+#include <exception>
>+#include <iostream>
>+#include <list>
>+#include <type_traits>
>+#include <vector>
>+
>+#include "gtest/gtest-spi.h"
>+#include "gtest/gtest.h"
>+#include "absl/memory/memory.h"
>+
>+namespace testing {
>+
>+namespace {
>+
>+using ::testing::exceptions_internal::SetCountdown;
>+using ::testing::exceptions_internal::TestException;
>+using ::testing::exceptions_internal::UnsetCountdown;
>+
>+// EXPECT_NO_THROW can't inspect the thrown inspection in general.
>+template <typename F>
>+void ExpectNoThrow(const F& f) {
>+  try {
>+    f();
>+  } catch (TestException e) {
>+    ADD_FAILURE() << "Unexpected exception thrown from " << e.what();
>+  }
>+}
>+
>+TEST(ThrowingValueTest, Throws) {
>+  SetCountdown();
>+  EXPECT_THROW(ThrowingValue<> bomb, TestException);
>+
>+  // It's not guaranteed that every operator only throws *once*.  The default
>+  // ctor only throws once, though, so use it to make sure we only throw when
>+  // the countdown hits 0
>+  SetCountdown(2);
>+  ExpectNoThrow([]() { ThrowingValue<> bomb; });
>+  ExpectNoThrow([]() { ThrowingValue<> bomb; });
>+  EXPECT_THROW(ThrowingValue<> bomb, TestException);
>+
>+  UnsetCountdown();
>+}
>+
>+// Tests that an operation throws when the countdown is at 0, doesn't throw when
>+// the countdown doesn't hit 0, and doesn't modify the state of the
>+// ThrowingValue if it throws
>+template <typename F>
>+void TestOp(const F& f) {
>+  ExpectNoThrow(f);
>+
>+  SetCountdown();
>+  EXPECT_THROW(f(), TestException);
>+  UnsetCountdown();
>+}
>+
>+TEST(ThrowingValueTest, ThrowingCtors) {
>+  ThrowingValue<> bomb;
>+
>+  TestOp([]() { ThrowingValue<> bomb(1); });
>+  TestOp([&]() { ThrowingValue<> bomb1 = bomb; });
>+  TestOp([&]() { ThrowingValue<> bomb1 = std::move(bomb); });
>+}
>+
>+TEST(ThrowingValueTest, ThrowingAssignment) {
>+  ThrowingValue<> bomb, bomb1;
>+
>+  TestOp([&]() { bomb = bomb1; });
>+  TestOp([&]() { bomb = std::move(bomb1); });
>+
>+  // Test that when assignment throws, the assignment should fail (lhs != rhs)
>+  // and strong guarantee fails (lhs != lhs_copy).
>+  {
>+    ThrowingValue<> lhs(39), rhs(42);
>+    ThrowingValue<> lhs_copy(lhs);
>+    SetCountdown();
>+    EXPECT_THROW(lhs = rhs, TestException);
>+    UnsetCountdown();
>+    EXPECT_NE(lhs, rhs);
>+    EXPECT_NE(lhs_copy, lhs);
>+  }
>+  {
>+    ThrowingValue<> lhs(39), rhs(42);
>+    ThrowingValue<> lhs_copy(lhs), rhs_copy(rhs);
>+    SetCountdown();
>+    EXPECT_THROW(lhs = std::move(rhs), TestException);
>+    UnsetCountdown();
>+    EXPECT_NE(lhs, rhs_copy);
>+    EXPECT_NE(lhs_copy, lhs);
>+  }
>+}
>+
>+TEST(ThrowingValueTest, ThrowingComparisons) {
>+  ThrowingValue<> bomb1, bomb2;
>+  TestOp([&]() { return bomb1 == bomb2; });
>+  TestOp([&]() { return bomb1 != bomb2; });
>+  TestOp([&]() { return bomb1 < bomb2; });
>+  TestOp([&]() { return bomb1 <= bomb2; });
>+  TestOp([&]() { return bomb1 > bomb2; });
>+  TestOp([&]() { return bomb1 >= bomb2; });
>+}
>+
>+TEST(ThrowingValueTest, ThrowingArithmeticOps) {
>+  ThrowingValue<> bomb1(1), bomb2(2);
>+
>+  TestOp([&bomb1]() { +bomb1; });
>+  TestOp([&bomb1]() { -bomb1; });
>+  TestOp([&bomb1]() { ++bomb1; });
>+  TestOp([&bomb1]() { bomb1++; });
>+  TestOp([&bomb1]() { --bomb1; });
>+  TestOp([&bomb1]() { bomb1--; });
>+
>+  TestOp([&]() { bomb1 + bomb2; });
>+  TestOp([&]() { bomb1 - bomb2; });
>+  TestOp([&]() { bomb1* bomb2; });
>+  TestOp([&]() { bomb1 / bomb2; });
>+  TestOp([&]() { bomb1 << 1; });
>+  TestOp([&]() { bomb1 >> 1; });
>+}
>+
>+TEST(ThrowingValueTest, ThrowingLogicalOps) {
>+  ThrowingValue<> bomb1, bomb2;
>+
>+  TestOp([&bomb1]() { !bomb1; });
>+  TestOp([&]() { bomb1&& bomb2; });
>+  TestOp([&]() { bomb1 || bomb2; });
>+}
>+
>+TEST(ThrowingValueTest, ThrowingBitwiseOps) {
>+  ThrowingValue<> bomb1, bomb2;
>+
>+  TestOp([&bomb1]() { ~bomb1; });
>+  TestOp([&]() { bomb1& bomb2; });
>+  TestOp([&]() { bomb1 | bomb2; });
>+  TestOp([&]() { bomb1 ^ bomb2; });
>+}
>+
>+TEST(ThrowingValueTest, ThrowingCompoundAssignmentOps) {
>+  ThrowingValue<> bomb1(1), bomb2(2);
>+
>+  TestOp([&]() { bomb1 += bomb2; });
>+  TestOp([&]() { bomb1 -= bomb2; });
>+  TestOp([&]() { bomb1 *= bomb2; });
>+  TestOp([&]() { bomb1 /= bomb2; });
>+  TestOp([&]() { bomb1 %= bomb2; });
>+  TestOp([&]() { bomb1 &= bomb2; });
>+  TestOp([&]() { bomb1 |= bomb2; });
>+  TestOp([&]() { bomb1 ^= bomb2; });
>+  TestOp([&]() { bomb1 *= bomb2; });
>+}
>+
>+TEST(ThrowingValueTest, ThrowingStreamOps) {
>+  ThrowingValue<> bomb;
>+
>+  TestOp([&]() {
>+    std::istringstream stream;
>+    stream >> bomb;
>+  });
>+  TestOp([&]() {
>+    std::stringstream stream;
>+    stream << bomb;
>+  });
>+}
>+
>+// Tests the operator<< of ThrowingValue by forcing ConstructorTracker to emit
>+// a nonfatal failure that contains the std::string representation of the Thrower
>+TEST(ThrowingValueTest, StreamOpsOutput) {
>+  using ::testing::TypeSpec;
>+  exceptions_internal::ConstructorTracker ct(exceptions_internal::countdown);
>+
>+  // Test default spec list (kEverythingThrows)
>+  EXPECT_NONFATAL_FAILURE(
>+      {
>+        using Thrower = ThrowingValue<TypeSpec{}>;
>+        auto thrower = Thrower(123);
>+        thrower.~Thrower();
>+      },
>+      "ThrowingValue<>(123)");
>+
>+  // Test with one item in spec list (kNoThrowCopy)
>+  EXPECT_NONFATAL_FAILURE(
>+      {
>+        using Thrower = ThrowingValue<TypeSpec::kNoThrowCopy>;
>+        auto thrower = Thrower(234);
>+        thrower.~Thrower();
>+      },
>+      "ThrowingValue<kNoThrowCopy>(234)");
>+
>+  // Test with multiple items in spec list (kNoThrowMove, kNoThrowNew)
>+  EXPECT_NONFATAL_FAILURE(
>+      {
>+        using Thrower =
>+            ThrowingValue<TypeSpec::kNoThrowMove | TypeSpec::kNoThrowNew>;
>+        auto thrower = Thrower(345);
>+        thrower.~Thrower();
>+      },
>+      "ThrowingValue<kNoThrowMove | kNoThrowNew>(345)");
>+
>+  // Test with all items in spec list (kNoThrowCopy, kNoThrowMove, kNoThrowNew)
>+  EXPECT_NONFATAL_FAILURE(
>+      {
>+        using Thrower = ThrowingValue<static_cast<TypeSpec>(-1)>;
>+        auto thrower = Thrower(456);
>+        thrower.~Thrower();
>+      },
>+      "ThrowingValue<kNoThrowCopy | kNoThrowMove | kNoThrowNew>(456)");
>+}
>+
>+template <typename F>
>+void TestAllocatingOp(const F& f) {
>+  ExpectNoThrow(f);
>+
>+  SetCountdown();
>+  EXPECT_THROW(f(), exceptions_internal::TestBadAllocException);
>+  UnsetCountdown();
>+}
>+
>+TEST(ThrowingValueTest, ThrowingAllocatingOps) {
>+  // make_unique calls unqualified operator new, so these exercise the
>+  // ThrowingValue overloads.
>+  TestAllocatingOp([]() { return absl::make_unique<ThrowingValue<>>(1); });
>+  TestAllocatingOp([]() { return absl::make_unique<ThrowingValue<>[]>(2); });
>+}
>+
>+TEST(ThrowingValueTest, NonThrowingMoveCtor) {
>+  ThrowingValue<TypeSpec::kNoThrowMove> nothrow_ctor;
>+
>+  SetCountdown();
>+  ExpectNoThrow([&nothrow_ctor]() {
>+    ThrowingValue<TypeSpec::kNoThrowMove> nothrow1 = std::move(nothrow_ctor);
>+  });
>+  UnsetCountdown();
>+}
>+
>+TEST(ThrowingValueTest, NonThrowingMoveAssign) {
>+  ThrowingValue<TypeSpec::kNoThrowMove> nothrow_assign1, nothrow_assign2;
>+
>+  SetCountdown();
>+  ExpectNoThrow([&nothrow_assign1, &nothrow_assign2]() {
>+    nothrow_assign1 = std::move(nothrow_assign2);
>+  });
>+  UnsetCountdown();
>+}
>+
>+TEST(ThrowingValueTest, ThrowingCopyCtor) {
>+  ThrowingValue<> tv;
>+
>+  TestOp([&]() { ThrowingValue<> tv_copy(tv); });
>+}
>+
>+TEST(ThrowingValueTest, ThrowingCopyAssign) {
>+  ThrowingValue<> tv1, tv2;
>+
>+  TestOp([&]() { tv1 = tv2; });
>+}
>+
>+TEST(ThrowingValueTest, NonThrowingCopyCtor) {
>+  ThrowingValue<TypeSpec::kNoThrowCopy> nothrow_ctor;
>+
>+  SetCountdown();
>+  ExpectNoThrow([&nothrow_ctor]() {
>+    ThrowingValue<TypeSpec::kNoThrowCopy> nothrow1(nothrow_ctor);
>+  });
>+  UnsetCountdown();
>+}
>+
>+TEST(ThrowingValueTest, NonThrowingCopyAssign) {
>+  ThrowingValue<TypeSpec::kNoThrowCopy> nothrow_assign1, nothrow_assign2;
>+
>+  SetCountdown();
>+  ExpectNoThrow([&nothrow_assign1, &nothrow_assign2]() {
>+    nothrow_assign1 = nothrow_assign2;
>+  });
>+  UnsetCountdown();
>+}
>+
>+TEST(ThrowingValueTest, ThrowingSwap) {
>+  ThrowingValue<> bomb1, bomb2;
>+  TestOp([&]() { std::swap(bomb1, bomb2); });
>+}
>+
>+TEST(ThrowingValueTest, NonThrowingSwap) {
>+  ThrowingValue<TypeSpec::kNoThrowMove> bomb1, bomb2;
>+  ExpectNoThrow([&]() { std::swap(bomb1, bomb2); });
>+}
>+
>+TEST(ThrowingValueTest, NonThrowingAllocation) {
>+  ThrowingValue<TypeSpec::kNoThrowNew>* allocated;
>+  ThrowingValue<TypeSpec::kNoThrowNew>* array;
>+
>+  ExpectNoThrow([&allocated]() {
>+    allocated = new ThrowingValue<TypeSpec::kNoThrowNew>(1);
>+    delete allocated;
>+  });
>+  ExpectNoThrow([&array]() {
>+    array = new ThrowingValue<TypeSpec::kNoThrowNew>[2];
>+    delete[] array;
>+  });
>+}
>+
>+TEST(ThrowingValueTest, NonThrowingDelete) {
>+  auto* allocated = new ThrowingValue<>(1);
>+  auto* array = new ThrowingValue<>[2];
>+
>+  SetCountdown();
>+  ExpectNoThrow([allocated]() { delete allocated; });
>+  SetCountdown();
>+  ExpectNoThrow([array]() { delete[] array; });
>+
>+  UnsetCountdown();
>+}
>+
>+using Storage =
>+    absl::aligned_storage_t<sizeof(ThrowingValue<>), alignof(ThrowingValue<>)>;
>+
>+TEST(ThrowingValueTest, NonThrowingPlacementDelete) {
>+  constexpr int kArrayLen = 2;
>+  // We intentionally create extra space to store the tag allocated by placement
>+  // new[].
>+  constexpr int kStorageLen = 4;
>+
>+  Storage buf;
>+  Storage array_buf[kStorageLen];
>+  auto* placed = new (&buf) ThrowingValue<>(1);
>+  auto placed_array = new (&array_buf) ThrowingValue<>[kArrayLen];
>+
>+  SetCountdown();
>+  ExpectNoThrow([placed, &buf]() {
>+    placed->~ThrowingValue<>();
>+    ThrowingValue<>::operator delete(placed, &buf);
>+  });
>+
>+  SetCountdown();
>+  ExpectNoThrow([&, placed_array]() {
>+    for (int i = 0; i < kArrayLen; ++i) placed_array[i].~ThrowingValue<>();
>+    ThrowingValue<>::operator delete[](placed_array, &array_buf);
>+  });
>+
>+  UnsetCountdown();
>+}
>+
>+TEST(ThrowingValueTest, NonThrowingDestructor) {
>+  auto* allocated = new ThrowingValue<>();
>+
>+  SetCountdown();
>+  ExpectNoThrow([allocated]() { delete allocated; });
>+  UnsetCountdown();
>+}
>+
>+TEST(ThrowingBoolTest, ThrowingBool) {
>+  ThrowingBool t = true;
>+
>+  // Test that it's contextually convertible to bool
>+  if (t) {  // NOLINT(whitespace/empty_if_body)
>+  }
>+  EXPECT_TRUE(t);
>+
>+  TestOp([&]() { (void)!t; });
>+}
>+
>+TEST(ThrowingAllocatorTest, MemoryManagement) {
>+  // Just exercise the memory management capabilities under LSan to make sure we
>+  // don't leak.
>+  ThrowingAllocator<int> int_alloc;
>+  int* ip = int_alloc.allocate(1);
>+  int_alloc.deallocate(ip, 1);
>+  int* i_array = int_alloc.allocate(2);
>+  int_alloc.deallocate(i_array, 2);
>+
>+  ThrowingAllocator<ThrowingValue<>> tv_alloc;
>+  ThrowingValue<>* ptr = tv_alloc.allocate(1);
>+  tv_alloc.deallocate(ptr, 1);
>+  ThrowingValue<>* tv_array = tv_alloc.allocate(2);
>+  tv_alloc.deallocate(tv_array, 2);
>+}
>+
>+TEST(ThrowingAllocatorTest, CallsGlobalNew) {
>+  ThrowingAllocator<ThrowingValue<>, AllocSpec::kNoThrowAllocate> nothrow_alloc;
>+  ThrowingValue<>* ptr;
>+
>+  SetCountdown();
>+  // This will only throw if ThrowingValue::new is called.
>+  ExpectNoThrow([&]() { ptr = nothrow_alloc.allocate(1); });
>+  nothrow_alloc.deallocate(ptr, 1);
>+
>+  UnsetCountdown();
>+}
>+
>+TEST(ThrowingAllocatorTest, ThrowingConstructors) {
>+  ThrowingAllocator<int> int_alloc;
>+  int* ip = nullptr;
>+
>+  SetCountdown();
>+  EXPECT_THROW(ip = int_alloc.allocate(1), TestException);
>+  ExpectNoThrow([&]() { ip = int_alloc.allocate(1); });
>+
>+  *ip = 1;
>+  SetCountdown();
>+  EXPECT_THROW(int_alloc.construct(ip, 2), TestException);
>+  EXPECT_EQ(*ip, 1);
>+  int_alloc.deallocate(ip, 1);
>+
>+  UnsetCountdown();
>+}
>+
>+TEST(ThrowingAllocatorTest, NonThrowingConstruction) {
>+  {
>+    ThrowingAllocator<int, AllocSpec::kNoThrowAllocate> int_alloc;
>+    int* ip = nullptr;
>+
>+    SetCountdown();
>+    ExpectNoThrow([&]() { ip = int_alloc.allocate(1); });
>+
>+    SetCountdown();
>+    ExpectNoThrow([&]() { int_alloc.construct(ip, 2); });
>+
>+    EXPECT_EQ(*ip, 2);
>+    int_alloc.deallocate(ip, 1);
>+
>+    UnsetCountdown();
>+  }
>+
>+  {
>+    ThrowingAllocator<int> int_alloc;
>+    int* ip = nullptr;
>+    ExpectNoThrow([&]() { ip = int_alloc.allocate(1); });
>+    ExpectNoThrow([&]() { int_alloc.construct(ip, 2); });
>+    EXPECT_EQ(*ip, 2);
>+    int_alloc.deallocate(ip, 1);
>+  }
>+
>+  {
>+    ThrowingAllocator<ThrowingValue<>, AllocSpec::kNoThrowAllocate>
>+        nothrow_alloc;
>+    ThrowingValue<>* ptr;
>+
>+    SetCountdown();
>+    ExpectNoThrow([&]() { ptr = nothrow_alloc.allocate(1); });
>+
>+    SetCountdown();
>+    ExpectNoThrow(
>+        [&]() { nothrow_alloc.construct(ptr, 2, testing::nothrow_ctor); });
>+
>+    EXPECT_EQ(ptr->Get(), 2);
>+    nothrow_alloc.destroy(ptr);
>+    nothrow_alloc.deallocate(ptr, 1);
>+
>+    UnsetCountdown();
>+  }
>+
>+  {
>+    ThrowingAllocator<int> a;
>+
>+    SetCountdown();
>+    ExpectNoThrow([&]() { ThrowingAllocator<double> a1 = a; });
>+
>+    SetCountdown();
>+    ExpectNoThrow([&]() { ThrowingAllocator<double> a1 = std::move(a); });
>+
>+    UnsetCountdown();
>+  }
>+}
>+
>+TEST(ThrowingAllocatorTest, ThrowingAllocatorConstruction) {
>+  ThrowingAllocator<int> a;
>+  TestOp([]() { ThrowingAllocator<int> a; });
>+  TestOp([&]() { a.select_on_container_copy_construction(); });
>+}
>+
>+TEST(ThrowingAllocatorTest, State) {
>+  ThrowingAllocator<int> a1, a2;
>+  EXPECT_NE(a1, a2);
>+
>+  auto a3 = a1;
>+  EXPECT_EQ(a3, a1);
>+  int* ip = a1.allocate(1);
>+  EXPECT_EQ(a3, a1);
>+  a3.deallocate(ip, 1);
>+  EXPECT_EQ(a3, a1);
>+}
>+
>+TEST(ThrowingAllocatorTest, InVector) {
>+  std::vector<ThrowingValue<>, ThrowingAllocator<ThrowingValue<>>> v;
>+  for (int i = 0; i < 20; ++i) v.push_back({});
>+  for (int i = 0; i < 20; ++i) v.pop_back();
>+}
>+
>+TEST(ThrowingAllocatorTest, InList) {
>+  std::list<ThrowingValue<>, ThrowingAllocator<ThrowingValue<>>> l;
>+  for (int i = 0; i < 20; ++i) l.push_back({});
>+  for (int i = 0; i < 20; ++i) l.pop_back();
>+  for (int i = 0; i < 20; ++i) l.push_front({});
>+  for (int i = 0; i < 20; ++i) l.pop_front();
>+}
>+
>+template <typename TesterInstance, typename = void>
>+struct NullaryTestValidator : public std::false_type {};
>+
>+template <typename TesterInstance>
>+struct NullaryTestValidator<
>+    TesterInstance,
>+    absl::void_t<decltype(std::declval<TesterInstance>().Test())>>
>+    : public std::true_type {};
>+
>+template <typename TesterInstance>
>+bool HasNullaryTest(const TesterInstance&) {
>+  return NullaryTestValidator<TesterInstance>::value;
>+}
>+
>+void DummyOp(void*) {}
>+
>+template <typename TesterInstance, typename = void>
>+struct UnaryTestValidator : public std::false_type {};
>+
>+template <typename TesterInstance>
>+struct UnaryTestValidator<
>+    TesterInstance,
>+    absl::void_t<decltype(std::declval<TesterInstance>().Test(DummyOp))>>
>+    : public std::true_type {};
>+
>+template <typename TesterInstance>
>+bool HasUnaryTest(const TesterInstance&) {
>+  return UnaryTestValidator<TesterInstance>::value;
>+}
>+
>+TEST(ExceptionSafetyTesterTest, IncompleteTypesAreNotTestable) {
>+  using T = exceptions_internal::UninitializedT;
>+  auto op = [](T* t) {};
>+  auto inv = [](T*) { return testing::AssertionSuccess(); };
>+  auto fac = []() { return absl::make_unique<T>(); };
>+
>+  // Test that providing operation and inveriants still does not allow for the
>+  // the invocation of .Test() and .Test(op) because it lacks a factory
>+  auto without_fac =
>+      testing::MakeExceptionSafetyTester().WithOperation(op).WithInvariants(
>+          inv, testing::strong_guarantee);
>+  EXPECT_FALSE(HasNullaryTest(without_fac));
>+  EXPECT_FALSE(HasUnaryTest(without_fac));
>+
>+  // Test that providing invariants and factory allows the invocation of
>+  // .Test(op) but does not allow for .Test() because it lacks an operation
>+  auto without_op = testing::MakeExceptionSafetyTester()
>+                        .WithInvariants(inv, testing::strong_guarantee)
>+                        .WithFactory(fac);
>+  EXPECT_FALSE(HasNullaryTest(without_op));
>+  EXPECT_TRUE(HasUnaryTest(without_op));
>+
>+  // Test that providing operation and factory still does not allow for the
>+  // the invocation of .Test() and .Test(op) because it lacks invariants
>+  auto without_inv =
>+      testing::MakeExceptionSafetyTester().WithOperation(op).WithFactory(fac);
>+  EXPECT_FALSE(HasNullaryTest(without_inv));
>+  EXPECT_FALSE(HasUnaryTest(without_inv));
>+}
>+
>+struct ExampleStruct {};
>+
>+std::unique_ptr<ExampleStruct> ExampleFunctionFactory() {
>+  return absl::make_unique<ExampleStruct>();
>+}
>+
>+void ExampleFunctionOperation(ExampleStruct*) {}
>+
>+testing::AssertionResult ExampleFunctionInvariant(ExampleStruct*) {
>+  return testing::AssertionSuccess();
>+}
>+
>+struct {
>+  std::unique_ptr<ExampleStruct> operator()() const {
>+    return ExampleFunctionFactory();
>+  }
>+} example_struct_factory;
>+
>+struct {
>+  void operator()(ExampleStruct*) const {}
>+} example_struct_operation;
>+
>+struct {
>+  testing::AssertionResult operator()(ExampleStruct* example_struct) const {
>+    return ExampleFunctionInvariant(example_struct);
>+  }
>+} example_struct_invariant;
>+
>+auto example_lambda_factory = []() { return ExampleFunctionFactory(); };
>+
>+auto example_lambda_operation = [](ExampleStruct*) {};
>+
>+auto example_lambda_invariant = [](ExampleStruct* example_struct) {
>+  return ExampleFunctionInvariant(example_struct);
>+};
>+
>+// Testing that function references, pointers, structs with operator() and
>+// lambdas can all be used with ExceptionSafetyTester
>+TEST(ExceptionSafetyTesterTest, MixedFunctionTypes) {
>+  // function reference
>+  EXPECT_TRUE(testing::MakeExceptionSafetyTester()
>+                  .WithFactory(ExampleFunctionFactory)
>+                  .WithOperation(ExampleFunctionOperation)
>+                  .WithInvariants(ExampleFunctionInvariant)
>+                  .Test());
>+
>+  // function pointer
>+  EXPECT_TRUE(testing::MakeExceptionSafetyTester()
>+                  .WithFactory(&ExampleFunctionFactory)
>+                  .WithOperation(&ExampleFunctionOperation)
>+                  .WithInvariants(&ExampleFunctionInvariant)
>+                  .Test());
>+
>+  // struct
>+  EXPECT_TRUE(testing::MakeExceptionSafetyTester()
>+                  .WithFactory(example_struct_factory)
>+                  .WithOperation(example_struct_operation)
>+                  .WithInvariants(example_struct_invariant)
>+                  .Test());
>+
>+  // lambda
>+  EXPECT_TRUE(testing::MakeExceptionSafetyTester()
>+                  .WithFactory(example_lambda_factory)
>+                  .WithOperation(example_lambda_operation)
>+                  .WithInvariants(example_lambda_invariant)
>+                  .Test());
>+}
>+
>+struct NonNegative {
>+  bool operator==(const NonNegative& other) const { return i == other.i; }
>+  int i;
>+};
>+
>+testing::AssertionResult CheckNonNegativeInvariants(NonNegative* g) {
>+  if (g->i >= 0) {
>+    return testing::AssertionSuccess();
>+  }
>+  return testing::AssertionFailure()
>+         << "i should be non-negative but is " << g->i;
>+}
>+
>+struct {
>+  template <typename T>
>+  void operator()(T* t) const {
>+    (*t)();
>+  }
>+} invoker;
>+
>+auto tester =
>+    testing::MakeExceptionSafetyTester().WithOperation(invoker).WithInvariants(
>+        CheckNonNegativeInvariants);
>+auto strong_tester = tester.WithInvariants(testing::strong_guarantee);
>+
>+struct FailsBasicGuarantee : public NonNegative {
>+  void operator()() {
>+    --i;
>+    ThrowingValue<> bomb;
>+    ++i;
>+  }
>+};
>+
>+TEST(ExceptionCheckTest, BasicGuaranteeFailure) {
>+  EXPECT_FALSE(tester.WithInitialValue(FailsBasicGuarantee{}).Test());
>+}
>+
>+struct FollowsBasicGuarantee : public NonNegative {
>+  void operator()() {
>+    ++i;
>+    ThrowingValue<> bomb;
>+  }
>+};
>+
>+TEST(ExceptionCheckTest, BasicGuarantee) {
>+  EXPECT_TRUE(tester.WithInitialValue(FollowsBasicGuarantee{}).Test());
>+}
>+
>+TEST(ExceptionCheckTest, StrongGuaranteeFailure) {
>+  EXPECT_FALSE(strong_tester.WithInitialValue(FailsBasicGuarantee{}).Test());
>+  EXPECT_FALSE(strong_tester.WithInitialValue(FollowsBasicGuarantee{}).Test());
>+}
>+
>+struct BasicGuaranteeWithExtraInvariants : public NonNegative {
>+  // After operator(), i is incremented.  If operator() throws, i is set to 9999
>+  void operator()() {
>+    int old_i = i;
>+    i = kExceptionSentinel;
>+    ThrowingValue<> bomb;
>+    i = ++old_i;
>+  }
>+
>+  static constexpr int kExceptionSentinel = 9999;
>+};
>+constexpr int BasicGuaranteeWithExtraInvariants::kExceptionSentinel;
>+
>+TEST(ExceptionCheckTest, BasicGuaranteeWithExtraInvariants) {
>+  auto tester_with_val =
>+      tester.WithInitialValue(BasicGuaranteeWithExtraInvariants{});
>+  EXPECT_TRUE(tester_with_val.Test());
>+  EXPECT_TRUE(
>+      tester_with_val
>+          .WithInvariants([](BasicGuaranteeWithExtraInvariants* o) {
>+            if (o->i == BasicGuaranteeWithExtraInvariants::kExceptionSentinel) {
>+              return testing::AssertionSuccess();
>+            }
>+            return testing::AssertionFailure()
>+                   << "i should be "
>+                   << BasicGuaranteeWithExtraInvariants::kExceptionSentinel
>+                   << ", but is " << o->i;
>+          })
>+          .Test());
>+}
>+
>+struct FollowsStrongGuarantee : public NonNegative {
>+  void operator()() { ThrowingValue<> bomb; }
>+};
>+
>+TEST(ExceptionCheckTest, StrongGuarantee) {
>+  EXPECT_TRUE(tester.WithInitialValue(FollowsStrongGuarantee{}).Test());
>+  EXPECT_TRUE(strong_tester.WithInitialValue(FollowsStrongGuarantee{}).Test());
>+}
>+
>+struct HasReset : public NonNegative {
>+  void operator()() {
>+    i = -1;
>+    ThrowingValue<> bomb;
>+    i = 1;
>+  }
>+
>+  void reset() { i = 0; }
>+};
>+
>+testing::AssertionResult CheckHasResetInvariants(HasReset* h) {
>+  h->reset();
>+  return testing::AssertionResult(h->i == 0);
>+}
>+
>+TEST(ExceptionCheckTest, ModifyingChecker) {
>+  auto set_to_1000 = [](FollowsBasicGuarantee* g) {
>+    g->i = 1000;
>+    return testing::AssertionSuccess();
>+  };
>+  auto is_1000 = [](FollowsBasicGuarantee* g) {
>+    return testing::AssertionResult(g->i == 1000);
>+  };
>+  auto increment = [](FollowsStrongGuarantee* g) {
>+    ++g->i;
>+    return testing::AssertionSuccess();
>+  };
>+
>+  EXPECT_FALSE(tester.WithInitialValue(FollowsBasicGuarantee{})
>+                   .WithInvariants(set_to_1000, is_1000)
>+                   .Test());
>+  EXPECT_TRUE(strong_tester.WithInitialValue(FollowsStrongGuarantee{})
>+                  .WithInvariants(increment)
>+                  .Test());
>+  EXPECT_TRUE(testing::MakeExceptionSafetyTester()
>+                  .WithInitialValue(HasReset{})
>+                  .WithInvariants(CheckHasResetInvariants)
>+                  .Test(invoker));
>+}
>+
>+struct NonCopyable : public NonNegative {
>+  NonCopyable(const NonCopyable&) = delete;
>+  NonCopyable() : NonNegative{0} {}
>+
>+  void operator()() { ThrowingValue<> bomb; }
>+};
>+
>+TEST(ExceptionCheckTest, NonCopyable) {
>+  auto factory = []() { return absl::make_unique<NonCopyable>(); };
>+  EXPECT_TRUE(tester.WithFactory(factory).Test());
>+  EXPECT_TRUE(strong_tester.WithFactory(factory).Test());
>+}
>+
>+struct NonEqualityComparable : public NonNegative {
>+  void operator()() { ThrowingValue<> bomb; }
>+
>+  void ModifyOnThrow() {
>+    ++i;
>+    ThrowingValue<> bomb;
>+    static_cast<void>(bomb);
>+    --i;
>+  }
>+};
>+
>+TEST(ExceptionCheckTest, NonEqualityComparable) {
>+  auto nec_is_strong = [](NonEqualityComparable* nec) {
>+    return testing::AssertionResult(nec->i == NonEqualityComparable().i);
>+  };
>+  auto strong_nec_tester = tester.WithInitialValue(NonEqualityComparable{})
>+                               .WithInvariants(nec_is_strong);
>+
>+  EXPECT_TRUE(strong_nec_tester.Test());
>+  EXPECT_FALSE(strong_nec_tester.Test(
>+      [](NonEqualityComparable* n) { n->ModifyOnThrow(); }));
>+}
>+
>+template <typename T>
>+struct ExhaustivenessTester {
>+  void operator()() {
>+    successes |= 1;
>+    T b1;
>+    static_cast<void>(b1);
>+    successes |= (1 << 1);
>+    T b2;
>+    static_cast<void>(b2);
>+    successes |= (1 << 2);
>+    T b3;
>+    static_cast<void>(b3);
>+    successes |= (1 << 3);
>+  }
>+
>+  bool operator==(const ExhaustivenessTester<ThrowingValue<>>&) const {
>+    return true;
>+  }
>+
>+  static unsigned char successes;
>+};
>+
>+struct {
>+  template <typename T>
>+  testing::AssertionResult operator()(ExhaustivenessTester<T>*) const {
>+    return testing::AssertionSuccess();
>+  }
>+} CheckExhaustivenessTesterInvariants;
>+
>+template <typename T>
>+unsigned char ExhaustivenessTester<T>::successes = 0;
>+
>+TEST(ExceptionCheckTest, Exhaustiveness) {
>+  auto exhaust_tester = testing::MakeExceptionSafetyTester()
>+                            .WithInvariants(CheckExhaustivenessTesterInvariants)
>+                            .WithOperation(invoker);
>+
>+  EXPECT_TRUE(
>+      exhaust_tester.WithInitialValue(ExhaustivenessTester<int>{}).Test());
>+  EXPECT_EQ(ExhaustivenessTester<int>::successes, 0xF);
>+
>+  EXPECT_TRUE(
>+      exhaust_tester.WithInitialValue(ExhaustivenessTester<ThrowingValue<>>{})
>+          .WithInvariants(testing::strong_guarantee)
>+          .Test());
>+  EXPECT_EQ(ExhaustivenessTester<ThrowingValue<>>::successes, 0xF);
>+}
>+
>+struct LeaksIfCtorThrows : private exceptions_internal::TrackedObject {
>+  LeaksIfCtorThrows() : TrackedObject(ABSL_PRETTY_FUNCTION) {
>+    ++counter;
>+    ThrowingValue<> v;
>+    static_cast<void>(v);
>+    --counter;
>+  }
>+  LeaksIfCtorThrows(const LeaksIfCtorThrows&) noexcept
>+      : TrackedObject(ABSL_PRETTY_FUNCTION) {}
>+  static int counter;
>+};
>+int LeaksIfCtorThrows::counter = 0;
>+
>+TEST(ExceptionCheckTest, TestLeakyCtor) {
>+  testing::TestThrowingCtor<LeaksIfCtorThrows>();
>+  EXPECT_EQ(LeaksIfCtorThrows::counter, 1);
>+  LeaksIfCtorThrows::counter = 0;
>+}
>+
>+struct Tracked : private exceptions_internal::TrackedObject {
>+  Tracked() : TrackedObject(ABSL_PRETTY_FUNCTION) {}
>+};
>+
>+TEST(ConstructorTrackerTest, CreatedBefore) {
>+  Tracked a, b, c;
>+  exceptions_internal::ConstructorTracker ct(exceptions_internal::countdown);
>+}
>+
>+TEST(ConstructorTrackerTest, CreatedAfter) {
>+  exceptions_internal::ConstructorTracker ct(exceptions_internal::countdown);
>+  Tracked a, b, c;
>+}
>+
>+TEST(ConstructorTrackerTest, NotDestroyedAfter) {
>+  absl::aligned_storage_t<sizeof(Tracked), alignof(Tracked)> storage;
>+  EXPECT_NONFATAL_FAILURE(
>+      {
>+        exceptions_internal::ConstructorTracker ct(
>+            exceptions_internal::countdown);
>+        new (&storage) Tracked;
>+      },
>+      "not destroyed");
>+}
>+
>+TEST(ConstructorTrackerTest, DestroyedTwice) {
>+  exceptions_internal::ConstructorTracker ct(exceptions_internal::countdown);
>+  EXPECT_NONFATAL_FAILURE(
>+      {
>+        Tracked t;
>+        t.~Tracked();
>+      },
>+      "re-destroyed");
>+}
>+
>+TEST(ConstructorTrackerTest, ConstructedTwice) {
>+  exceptions_internal::ConstructorTracker ct(exceptions_internal::countdown);
>+  absl::aligned_storage_t<sizeof(Tracked), alignof(Tracked)> storage;
>+  EXPECT_NONFATAL_FAILURE(
>+      {
>+        new (&storage) Tracked;
>+        new (&storage) Tracked;
>+        reinterpret_cast<Tracked*>(&storage)->~Tracked();
>+      },
>+      "re-constructed");
>+}
>+
>+TEST(ThrowingValueTraitsTest, RelationalOperators) {
>+  ThrowingValue<> a, b;
>+  EXPECT_TRUE((std::is_convertible<decltype(a == b), bool>::value));
>+  EXPECT_TRUE((std::is_convertible<decltype(a != b), bool>::value));
>+  EXPECT_TRUE((std::is_convertible<decltype(a < b), bool>::value));
>+  EXPECT_TRUE((std::is_convertible<decltype(a <= b), bool>::value));
>+  EXPECT_TRUE((std::is_convertible<decltype(a > b), bool>::value));
>+  EXPECT_TRUE((std::is_convertible<decltype(a >= b), bool>::value));
>+}
>+
>+TEST(ThrowingAllocatorTraitsTest, Assignablility) {
>+  EXPECT_TRUE(std::is_move_assignable<ThrowingAllocator<int>>::value);
>+  EXPECT_TRUE(std::is_copy_assignable<ThrowingAllocator<int>>::value);
>+  EXPECT_TRUE(std::is_nothrow_move_assignable<ThrowingAllocator<int>>::value);
>+  EXPECT_TRUE(std::is_nothrow_copy_assignable<ThrowingAllocator<int>>::value);
>+}
>+
>+}  // namespace
>+
>+}  // namespace testing
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/inline_variable_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/inline_variable_test.cc
>new file mode 100644
>index 00000000000..5499189a3da
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/inline_variable_test.cc
>@@ -0,0 +1,62 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include <type_traits>
>+
>+#include "absl/base/internal/inline_variable.h"
>+#include "absl/base/internal/inline_variable_testing.h"
>+
>+#include "gtest/gtest.h"
>+
>+namespace absl {
>+namespace inline_variable_testing_internal {
>+namespace {
>+
>+TEST(InlineVariableTest, Constexpr) {
>+  static_assert(inline_variable_foo.value == 5, "");
>+  static_assert(other_inline_variable_foo.value == 5, "");
>+  static_assert(inline_variable_int == 5, "");
>+  static_assert(other_inline_variable_int == 5, "");
>+}
>+
>+TEST(InlineVariableTest, DefaultConstructedIdentityEquality) {
>+  EXPECT_EQ(get_foo_a().value, 5);
>+  EXPECT_EQ(get_foo_b().value, 5);
>+  EXPECT_EQ(&get_foo_a(), &get_foo_b());
>+}
>+
>+TEST(InlineVariableTest, DefaultConstructedIdentityInequality) {
>+  EXPECT_NE(&inline_variable_foo, &other_inline_variable_foo);
>+}
>+
>+TEST(InlineVariableTest, InitializedIdentityEquality) {
>+  EXPECT_EQ(get_int_a(), 5);
>+  EXPECT_EQ(get_int_b(), 5);
>+  EXPECT_EQ(&get_int_a(), &get_int_b());
>+}
>+
>+TEST(InlineVariableTest, InitializedIdentityInequality) {
>+  EXPECT_NE(&inline_variable_int, &other_inline_variable_int);
>+}
>+
>+TEST(InlineVariableTest, FunPtrType) {
>+  static_assert(
>+      std::is_same<void(*)(),
>+                   std::decay<decltype(inline_variable_fun_ptr)>::type>::value,
>+      "");
>+}
>+
>+}  // namespace
>+}  // namespace inline_variable_testing_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/inline_variable_test_a.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/inline_variable_test_a.cc
>new file mode 100644
>index 00000000000..a3bf3b68b3c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/inline_variable_test_a.cc
>@@ -0,0 +1,25 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/base/internal/inline_variable_testing.h"
>+
>+namespace absl {
>+namespace inline_variable_testing_internal {
>+
>+const Foo& get_foo_a() { return inline_variable_foo; }
>+
>+const int& get_int_a() { return inline_variable_int; }
>+
>+}  // namespace inline_variable_testing_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/inline_variable_test_b.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/inline_variable_test_b.cc
>new file mode 100644
>index 00000000000..b4b9393a558
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/inline_variable_test_b.cc
>@@ -0,0 +1,25 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/base/internal/inline_variable_testing.h"
>+
>+namespace absl {
>+namespace inline_variable_testing_internal {
>+
>+const Foo& get_foo_b() { return inline_variable_foo; }
>+
>+const int& get_int_b() { return inline_variable_int; }
>+
>+}  // namespace inline_variable_testing_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/atomic_hook.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/atomic_hook.h
>new file mode 100644
>index 00000000000..b458511b0c7
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/atomic_hook.h
>@@ -0,0 +1,165 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+
>+#ifndef ABSL_BASE_INTERNAL_ATOMIC_HOOK_H_
>+#define ABSL_BASE_INTERNAL_ATOMIC_HOOK_H_
>+
>+#include <atomic>
>+#include <cassert>
>+#include <cstdint>
>+#include <utility>
>+
>+#ifdef _MSC_FULL_VER
>+#define ABSL_HAVE_WORKING_ATOMIC_POINTER 0
>+#else
>+#define ABSL_HAVE_WORKING_ATOMIC_POINTER 1
>+#endif
>+
>+namespace absl {
>+namespace base_internal {
>+
>+template <typename T>
>+class AtomicHook;
>+
>+// AtomicHook is a helper class, templatized on a raw function pointer type, for
>+// implementing Abseil customization hooks.  It is a callable object that
>+// dispatches to the registered hook.
>+//
>+// A default constructed object performs a no-op (and returns a default
>+// constructed object) if no hook has been registered.
>+//
>+// Hooks can be pre-registered via constant initialization, for example,
>+// ABSL_CONST_INIT static AtomicHook<void(*)()> my_hook(DefaultAction);
>+// and then changed at runtime via a call to Store().
>+//
>+// Reads and writes guarantee memory_order_acquire/memory_order_release
>+// semantics.
>+template <typename ReturnType, typename... Args>
>+class AtomicHook<ReturnType (*)(Args...)> {
>+ public:
>+  using FnPtr = ReturnType (*)(Args...);
>+
>+  // Constructs an object that by default performs a no-op (and
>+  // returns a default constructed object) when no hook as been registered.
>+  constexpr AtomicHook() : AtomicHook(DummyFunction) {}
>+
>+  // Constructs an object that by default dispatches to/returns the
>+  // pre-registered default_fn when no hook has been registered at runtime.
>+#if ABSL_HAVE_WORKING_ATOMIC_POINTER
>+  explicit constexpr AtomicHook(FnPtr default_fn)
>+      : hook_(default_fn), default_fn_(default_fn) {}
>+#else
>+  explicit constexpr AtomicHook(FnPtr default_fn)
>+      : hook_(kUninitialized), default_fn_(default_fn) {}
>+#endif
>+
>+  // Stores the provided function pointer as the value for this hook.
>+  //
>+  // This is intended to be called once.  Multiple calls are legal only if the
>+  // same function pointer is provided for each call.  The store is implemented
>+  // as a memory_order_release operation, and read accesses are implemented as
>+  // memory_order_acquire.
>+  void Store(FnPtr fn) {
>+    bool success = DoStore(fn);
>+    static_cast<void>(success);
>+    assert(success);
>+  }
>+
>+  // Invokes the registered callback.  If no callback has yet been registered, a
>+  // default-constructed object of the appropriate type is returned instead.
>+  template <typename... CallArgs>
>+  ReturnType operator()(CallArgs&&... args) const {
>+    return DoLoad()(std::forward<CallArgs>(args)...);
>+  }
>+
>+  // Returns the registered callback, or nullptr if none has been registered.
>+  // Useful if client code needs to conditionalize behavior based on whether a
>+  // callback was registered.
>+  //
>+  // Note that atomic_hook.Load()() and atomic_hook() have different semantics:
>+  // operator()() will perform a no-op if no callback was registered, while
>+  // Load()() will dereference a null function pointer.  Prefer operator()() to
>+  // Load()() unless you must conditionalize behavior on whether a hook was
>+  // registered.
>+  FnPtr Load() const {
>+    FnPtr ptr = DoLoad();
>+    return (ptr == DummyFunction) ? nullptr : ptr;
>+  }
>+
>+ private:
>+  static ReturnType DummyFunction(Args...) {
>+    return ReturnType();
>+  }
>+
>+  // Current versions of MSVC (as of September 2017) have a broken
>+  // implementation of std::atomic<T*>:  Its constructor attempts to do the
>+  // equivalent of a reinterpret_cast in a constexpr context, which is not
>+  // allowed.
>+  //
>+  // This causes an issue when building with LLVM under Windows.  To avoid this,
>+  // we use a less-efficient, intptr_t-based implementation on Windows.
>+#if ABSL_HAVE_WORKING_ATOMIC_POINTER
>+  // Return the stored value, or DummyFunction if no value has been stored.
>+  FnPtr DoLoad() const { return hook_.load(std::memory_order_acquire); }
>+
>+  // Store the given value.  Returns false if a different value was already
>+  // stored to this object.
>+  bool DoStore(FnPtr fn) {
>+    assert(fn);
>+    FnPtr expected = default_fn_;
>+    const bool store_succeeded = hook_.compare_exchange_strong(
>+        expected, fn, std::memory_order_acq_rel, std::memory_order_acquire);
>+    const bool same_value_already_stored = (expected == fn);
>+    return store_succeeded || same_value_already_stored;
>+  }
>+
>+  std::atomic<FnPtr> hook_;
>+#else  // !ABSL_HAVE_WORKING_ATOMIC_POINTER
>+  // Use a sentinel value unlikely to be the address of an actual function.
>+  static constexpr intptr_t kUninitialized = 0;
>+
>+  static_assert(sizeof(intptr_t) >= sizeof(FnPtr),
>+                "intptr_t can't contain a function pointer");
>+
>+  FnPtr DoLoad() const {
>+    const intptr_t value = hook_.load(std::memory_order_acquire);
>+    if (value == kUninitialized) {
>+      return default_fn_;
>+    }
>+    return reinterpret_cast<FnPtr>(value);
>+  }
>+
>+  bool DoStore(FnPtr fn) {
>+    assert(fn);
>+    const auto value = reinterpret_cast<intptr_t>(fn);
>+    intptr_t expected = kUninitialized;
>+    const bool store_succeeded = hook_.compare_exchange_strong(
>+        expected, value, std::memory_order_acq_rel, std::memory_order_acquire);
>+    const bool same_value_already_stored = (expected == value);
>+    return store_succeeded || same_value_already_stored;
>+  }
>+
>+  std::atomic<intptr_t> hook_;
>+#endif
>+
>+  const FnPtr default_fn_;
>+};
>+
>+#undef ABSL_HAVE_WORKING_ATOMIC_POINTER
>+
>+}  // namespace base_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_BASE_INTERNAL_ATOMIC_HOOK_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/atomic_hook_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/atomic_hook_test.cc
>new file mode 100644
>index 00000000000..cf7407573a5
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/atomic_hook_test.cc
>@@ -0,0 +1,70 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/base/internal/atomic_hook.h"
>+
>+#include "gtest/gtest.h"
>+#include "absl/base/attributes.h"
>+
>+namespace {
>+
>+int value = 0;
>+void TestHook(int x) { value = x; }
>+
>+TEST(AtomicHookTest, NoDefaultFunction) {
>+  ABSL_CONST_INIT static absl::base_internal::AtomicHook<void(*)(int)> hook;
>+  value = 0;
>+
>+  // Test the default DummyFunction.
>+  EXPECT_TRUE(hook.Load() == nullptr);
>+  EXPECT_EQ(value, 0);
>+  hook(1);
>+  EXPECT_EQ(value, 0);
>+
>+  // Test a stored hook.
>+  hook.Store(TestHook);
>+  EXPECT_TRUE(hook.Load() == TestHook);
>+  EXPECT_EQ(value, 0);
>+  hook(1);
>+  EXPECT_EQ(value, 1);
>+
>+  // Calling Store() with the same hook should not crash.
>+  hook.Store(TestHook);
>+  EXPECT_TRUE(hook.Load() == TestHook);
>+  EXPECT_EQ(value, 1);
>+  hook(2);
>+  EXPECT_EQ(value, 2);
>+}
>+
>+TEST(AtomicHookTest, WithDefaultFunction) {
>+  // Set the default value to TestHook at compile-time.
>+  ABSL_CONST_INIT static absl::base_internal::AtomicHook<void (*)(int)> hook(
>+      TestHook);
>+  value = 0;
>+
>+  // Test the default value is TestHook.
>+  EXPECT_TRUE(hook.Load() == TestHook);
>+  EXPECT_EQ(value, 0);
>+  hook(1);
>+  EXPECT_EQ(value, 1);
>+
>+  // Calling Store() with the same hook should not crash.
>+  hook.Store(TestHook);
>+  EXPECT_TRUE(hook.Load() == TestHook);
>+  EXPECT_EQ(value, 1);
>+  hook(2);
>+  EXPECT_EQ(value, 2);
>+}
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/cycleclock.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/cycleclock.cc
>new file mode 100644
>index 00000000000..a742df01f94
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/cycleclock.cc
>@@ -0,0 +1,81 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+// The implementation of CycleClock::Frequency.
>+//
>+// NOTE: only i386 and x86_64 have been well tested.
>+// PPC, sparc, alpha, and ia64 are based on
>+//    http://peter.kuscsik.com/wordpress/?p=14
>+// with modifications by m3b.  See also
>+//    https://setisvn.ssl.berkeley.edu/svn/lib/fftw-3.0.1/kernel/cycle.h
>+
>+#include "absl/base/internal/cycleclock.h"
>+
>+#include <chrono>  // NOLINT(build/c++11)
>+
>+#include "absl/base/internal/unscaledcycleclock.h"
>+
>+namespace absl {
>+namespace base_internal {
>+
>+#if ABSL_USE_UNSCALED_CYCLECLOCK
>+
>+namespace {
>+
>+#ifdef NDEBUG
>+#ifdef ABSL_INTERNAL_UNSCALED_CYCLECLOCK_FREQUENCY_IS_CPU_FREQUENCY
>+// Not debug mode and the UnscaledCycleClock frequency is the CPU
>+// frequency.  Scale the CycleClock to prevent overflow if someone
>+// tries to represent the time as cycles since the Unix epoch.
>+static constexpr int32_t kShift = 1;
>+#else
>+// Not debug mode and the UnscaledCycleClock isn't operating at the
>+// raw CPU frequency. There is no need to do any scaling, so don't
>+// needlessly sacrifice precision.
>+static constexpr int32_t kShift = 0;
>+#endif
>+#else
>+// In debug mode use a different shift to discourage depending on a
>+// particular shift value.
>+static constexpr int32_t kShift = 2;
>+#endif
>+
>+static constexpr double kFrequencyScale = 1.0 / (1 << kShift);
>+
>+}  // namespace
>+
>+int64_t CycleClock::Now() {
>+  return base_internal::UnscaledCycleClock::Now() >> kShift;
>+}
>+
>+double CycleClock::Frequency() {
>+  return kFrequencyScale * base_internal::UnscaledCycleClock::Frequency();
>+}
>+
>+#else
>+
>+int64_t CycleClock::Now() {
>+  return std::chrono::duration_cast<std::chrono::nanoseconds>(
>+             std::chrono::steady_clock::now().time_since_epoch())
>+      .count();
>+}
>+
>+double CycleClock::Frequency() {
>+  return 1e9;
>+}
>+
>+#endif
>+
>+}  // namespace base_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/cycleclock.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/cycleclock.h
>new file mode 100644
>index 00000000000..60e971583c5
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/cycleclock.h
>@@ -0,0 +1,77 @@
>+//
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+
>+// -----------------------------------------------------------------------------
>+// File: cycleclock.h
>+// -----------------------------------------------------------------------------
>+//
>+// This header file defines a `CycleClock`, which yields the value and frequency
>+// of a cycle counter that increments at a rate that is approximately constant.
>+//
>+// NOTE:
>+//
>+// The cycle counter frequency is not necessarily related to the core clock
>+// frequency and should not be treated as such. That is, `CycleClock` cycles are
>+// not necessarily "CPU cycles" and code should not rely on that behavior, even
>+// if experimentally observed.
>+//
>+//
>+// An arbitrary offset may have been added to the counter at power on.
>+//
>+// On some platforms, the rate and offset of the counter may differ
>+// slightly when read from different CPUs of a multiprocessor. Usually,
>+// we try to ensure that the operating system adjusts values periodically
>+// so that values agree approximately.   If you need stronger guarantees,
>+// consider using alternate interfaces.
>+//
>+// The CPU is not required to maintain the ordering of a cycle counter read
>+// with respect to surrounding instructions.
>+
>+#ifndef ABSL_BASE_INTERNAL_CYCLECLOCK_H_
>+#define ABSL_BASE_INTERNAL_CYCLECLOCK_H_
>+
>+#include <cstdint>
>+
>+namespace absl {
>+namespace base_internal {
>+
>+// -----------------------------------------------------------------------------
>+// CycleClock
>+// -----------------------------------------------------------------------------
>+class CycleClock {
>+ public:
>+  // CycleClock::Now()
>+  //
>+  // Returns the value of a cycle counter that counts at a rate that is
>+  // approximately constant.
>+  static int64_t Now();
>+
>+  // CycleClock::Frequency()
>+  //
>+  // Returns the amount by which `CycleClock::Now()` increases per second. Note
>+  // that this value may not necessarily match the core CPU clock frequency.
>+  static double Frequency();
>+
>+ private:
>+  CycleClock() = delete;  // no instances
>+  CycleClock(const CycleClock&) = delete;
>+  CycleClock& operator=(const CycleClock&) = delete;
>+};
>+
>+}  // namespace base_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_BASE_INTERNAL_CYCLECLOCK_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/direct_mmap.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/direct_mmap.h
>new file mode 100644
>index 00000000000..0426e11890b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/direct_mmap.h
>@@ -0,0 +1,153 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// Functions for directly invoking mmap() via syscall, avoiding the case where
>+// mmap() has been locally overridden.
>+
>+#ifndef ABSL_BASE_INTERNAL_DIRECT_MMAP_H_
>+#define ABSL_BASE_INTERNAL_DIRECT_MMAP_H_
>+
>+#include "absl/base/config.h"
>+
>+#if ABSL_HAVE_MMAP
>+
>+#include <sys/mman.h>
>+
>+#ifdef __linux__
>+
>+#include <sys/types.h>
>+#ifdef __BIONIC__
>+#include <sys/syscall.h>
>+#else
>+#include <syscall.h>
>+#endif
>+
>+#include <linux/unistd.h>
>+#include <unistd.h>
>+#include <cerrno>
>+#include <cstdarg>
>+#include <cstdint>
>+
>+#ifdef __mips__
>+// Include definitions of the ABI currently in use.
>+#ifdef __BIONIC__
>+// Android doesn't have sgidefs.h, but does have asm/sgidefs.h, which has the
>+// definitions we need.
>+#include <asm/sgidefs.h>
>+#else
>+#include <sgidefs.h>
>+#endif  // __BIONIC__
>+#endif  // __mips__
>+
>+// SYS_mmap and SYS_munmap are not defined in Android.
>+#ifdef __BIONIC__
>+extern "C" void* __mmap2(void*, size_t, int, int, int, size_t);
>+#if defined(__NR_mmap) && !defined(SYS_mmap)
>+#define SYS_mmap __NR_mmap
>+#endif
>+#ifndef SYS_munmap
>+#define SYS_munmap __NR_munmap
>+#endif
>+#endif  // __BIONIC__
>+
>+namespace absl {
>+namespace base_internal {
>+
>+// Platform specific logic extracted from
>+// https://chromium.googlesource.com/linux-syscall-support/+/master/linux_syscall_support.h
>+inline void* DirectMmap(void* start, size_t length, int prot, int flags, int fd,
>+                        off64_t offset) noexcept {
>+#if defined(__i386__) || defined(__ARM_ARCH_3__) || defined(__ARM_EABI__) || \
>+    (defined(__mips__) && _MIPS_SIM == _MIPS_SIM_ABI32) ||                   \
>+    (defined(__PPC__) && !defined(__PPC64__)) ||                             \
>+    (defined(__s390__) && !defined(__s390x__))
>+  // On these architectures, implement mmap with mmap2.
>+  static int pagesize = 0;
>+  if (pagesize == 0) {
>+    pagesize = getpagesize();
>+  }
>+  if (offset < 0 || offset % pagesize != 0) {
>+    errno = EINVAL;
>+    return MAP_FAILED;
>+  }
>+#ifdef __BIONIC__
>+  // SYS_mmap2 has problems on Android API level <= 16.
>+  // Workaround by invoking __mmap2() instead.
>+  return __mmap2(start, length, prot, flags, fd, offset / pagesize);
>+#else
>+  return reinterpret_cast<void*>(
>+      syscall(SYS_mmap2, start, length, prot, flags, fd,
>+              static_cast<off_t>(offset / pagesize)));
>+#endif
>+#elif defined(__s390x__)
>+  // On s390x, mmap() arguments are passed in memory.
>+  unsigned long buf[6] = {reinterpret_cast<unsigned long>(start),  // NOLINT
>+                          static_cast<unsigned long>(length),      // NOLINT
>+                          static_cast<unsigned long>(prot),        // NOLINT
>+                          static_cast<unsigned long>(flags),       // NOLINT
>+                          static_cast<unsigned long>(fd),          // NOLINT
>+                          static_cast<unsigned long>(offset)};     // NOLINT
>+  return reinterpret_cast<void*>(syscall(SYS_mmap, buf));
>+#elif defined(__x86_64__)
>+// The x32 ABI has 32 bit longs, but the syscall interface is 64 bit.
>+// We need to explicitly cast to an unsigned 64 bit type to avoid implicit
>+// sign extension.  We can't cast pointers directly because those are
>+// 32 bits, and gcc will dump ugly warnings about casting from a pointer
>+// to an integer of a different size. We also need to make sure __off64_t
>+// isn't truncated to 32-bits under x32.
>+#define MMAP_SYSCALL_ARG(x) ((uint64_t)(uintptr_t)(x))
>+  return reinterpret_cast<void*>(
>+      syscall(SYS_mmap, MMAP_SYSCALL_ARG(start), MMAP_SYSCALL_ARG(length),
>+              MMAP_SYSCALL_ARG(prot), MMAP_SYSCALL_ARG(flags),
>+              MMAP_SYSCALL_ARG(fd), static_cast<uint64_t>(offset)));
>+#undef MMAP_SYSCALL_ARG
>+#else  // Remaining 64-bit aritectures.
>+  static_assert(sizeof(unsigned long) == 8, "Platform is not 64-bit");
>+  return reinterpret_cast<void*>(
>+      syscall(SYS_mmap, start, length, prot, flags, fd, offset));
>+#endif
>+}
>+
>+inline int DirectMunmap(void* start, size_t length) {
>+  return static_cast<int>(syscall(SYS_munmap, start, length));
>+}
>+
>+}  // namespace base_internal
>+}  // namespace absl
>+
>+#else  // !__linux__
>+
>+// For non-linux platforms where we have mmap, just dispatch directly to the
>+// actual mmap()/munmap() methods.
>+
>+namespace absl {
>+namespace base_internal {
>+
>+inline void* DirectMmap(void* start, size_t length, int prot, int flags, int fd,
>+                        off_t offset) {
>+  return mmap(start, length, prot, flags, fd, offset);
>+}
>+
>+inline int DirectMunmap(void* start, size_t length) {
>+  return munmap(start, length);
>+}
>+
>+}  // namespace base_internal
>+}  // namespace absl
>+
>+#endif  // __linux__
>+
>+#endif  // ABSL_HAVE_MMAP
>+
>+#endif  // ABSL_BASE_INTERNAL_DIRECT_MMAP_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/endian.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/endian.h
>new file mode 100644
>index 00000000000..edc10f10a5a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/endian.h
>@@ -0,0 +1,269 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+
>+#ifndef ABSL_BASE_INTERNAL_ENDIAN_H_
>+#define ABSL_BASE_INTERNAL_ENDIAN_H_
>+
>+// The following guarantees declaration of the byte swap functions
>+#ifdef _MSC_VER
>+#include <stdlib.h>  // NOLINT(build/include)
>+#elif defined(__APPLE__)
>+// Mac OS X / Darwin features
>+#include <libkern/OSByteOrder.h>
>+#elif defined(__FreeBSD__)
>+#include <sys/endian.h>
>+#elif defined(__GLIBC__)
>+#include <byteswap.h>  // IWYU pragma: export
>+#endif
>+
>+#include <cstdint>
>+#include "absl/base/config.h"
>+#include "absl/base/internal/unaligned_access.h"
>+#include "absl/base/port.h"
>+
>+namespace absl {
>+
>+// Use compiler byte-swapping intrinsics if they are available.  32-bit
>+// and 64-bit versions are available in Clang and GCC as of GCC 4.3.0.
>+// The 16-bit version is available in Clang and GCC only as of GCC 4.8.0.
>+// For simplicity, we enable them all only for GCC 4.8.0 or later.
>+#if defined(__clang__) || \
>+    (defined(__GNUC__) && \
>+     ((__GNUC__ == 4 && __GNUC_MINOR__ >= 8) || __GNUC__ >= 5))
>+inline uint64_t gbswap_64(uint64_t host_int) {
>+  return __builtin_bswap64(host_int);
>+}
>+inline uint32_t gbswap_32(uint32_t host_int) {
>+  return __builtin_bswap32(host_int);
>+}
>+inline uint16_t gbswap_16(uint16_t host_int) {
>+  return __builtin_bswap16(host_int);
>+}
>+
>+#elif defined(_MSC_VER)
>+inline uint64_t gbswap_64(uint64_t host_int) {
>+  return _byteswap_uint64(host_int);
>+}
>+inline uint32_t gbswap_32(uint32_t host_int) {
>+  return _byteswap_ulong(host_int);
>+}
>+inline uint16_t gbswap_16(uint16_t host_int) {
>+  return _byteswap_ushort(host_int);
>+}
>+
>+#elif defined(__APPLE__)
>+inline uint64_t gbswap_64(uint64_t host_int) { return OSSwapInt16(host_int); }
>+inline uint32_t gbswap_32(uint32_t host_int) { return OSSwapInt32(host_int); }
>+inline uint16_t gbswap_16(uint16_t host_int) { return OSSwapInt64(host_int); }
>+
>+#else
>+inline uint64_t gbswap_64(uint64_t host_int) {
>+#if defined(__GNUC__) && defined(__x86_64__) && !defined(__APPLE__)
>+  // Adapted from /usr/include/byteswap.h.  Not available on Mac.
>+  if (__builtin_constant_p(host_int)) {
>+    return __bswap_constant_64(host_int);
>+  } else {
>+    register uint64_t result;
>+    __asm__("bswap %0" : "=r"(result) : "0"(host_int));
>+    return result;
>+  }
>+#elif defined(__GLIBC__)
>+  return bswap_64(host_int);
>+#else
>+  return (((x & uint64_t{(0xFF}) << 56) |
>+          ((x & uint64_t{(0xFF00}) << 40) |
>+          ((x & uint64_t{(0xFF0000}) << 24) |
>+          ((x & uint64_t{(0xFF000000}) << 8) |
>+          ((x & uint64_t{(0xFF00000000}) >> 8) |
>+          ((x & uint64_t{(0xFF0000000000}) >> 24) |
>+          ((x & uint64_t{(0xFF000000000000}) >> 40) |
>+          ((x & uint64_t{(0xFF00000000000000}) >> 56));
>+#endif  // bswap_64
>+}
>+
>+inline uint32_t gbswap_32(uint32_t host_int) {
>+#if defined(__GLIBC__)
>+  return bswap_32(host_int);
>+#else
>+  return (((x & 0xFF) << 24) | ((x & 0xFF00) << 8) | ((x & 0xFF0000) >> 8) |
>+          ((x & 0xFF000000) >> 24));
>+#endif
>+}
>+
>+inline uint16_t gbswap_16(uint16_t host_int) {
>+#if defined(__GLIBC__)
>+  return bswap_16(host_int);
>+#else
>+  return uint16_t{((x & 0xFF) << 8) | ((x & 0xFF00) >> 8)};
>+#endif
>+}
>+
>+#endif  // intrinics available
>+
>+#ifdef ABSL_IS_LITTLE_ENDIAN
>+
>+// Definitions for ntohl etc. that don't require us to include
>+// netinet/in.h. We wrap gbswap_32 and gbswap_16 in functions rather
>+// than just #defining them because in debug mode, gcc doesn't
>+// correctly handle the (rather involved) definitions of bswap_32.
>+// gcc guarantees that inline functions are as fast as macros, so
>+// this isn't a performance hit.
>+inline uint16_t ghtons(uint16_t x) { return gbswap_16(x); }
>+inline uint32_t ghtonl(uint32_t x) { return gbswap_32(x); }
>+inline uint64_t ghtonll(uint64_t x) { return gbswap_64(x); }
>+
>+#elif defined ABSL_IS_BIG_ENDIAN
>+
>+// These definitions are simpler on big-endian machines
>+// These are functions instead of macros to avoid self-assignment warnings
>+// on calls such as "i = ghtnol(i);".  This also provides type checking.
>+inline uint16_t ghtons(uint16_t x) { return x; }
>+inline uint32_t ghtonl(uint32_t x) { return x; }
>+inline uint64_t ghtonll(uint64_t x) { return x; }
>+
>+#else
>+#error \
>+    "Unsupported byte order: Either ABSL_IS_BIG_ENDIAN or " \
>+       "ABSL_IS_LITTLE_ENDIAN must be defined"
>+#endif  // byte order
>+
>+inline uint16_t gntohs(uint16_t x) { return ghtons(x); }
>+inline uint32_t gntohl(uint32_t x) { return ghtonl(x); }
>+inline uint64_t gntohll(uint64_t x) { return ghtonll(x); }
>+
>+// Utilities to convert numbers between the current hosts's native byte
>+// order and little-endian byte order
>+//
>+// Load/Store methods are alignment safe
>+namespace little_endian {
>+// Conversion functions.
>+#ifdef ABSL_IS_LITTLE_ENDIAN
>+
>+inline uint16_t FromHost16(uint16_t x) { return x; }
>+inline uint16_t ToHost16(uint16_t x) { return x; }
>+
>+inline uint32_t FromHost32(uint32_t x) { return x; }
>+inline uint32_t ToHost32(uint32_t x) { return x; }
>+
>+inline uint64_t FromHost64(uint64_t x) { return x; }
>+inline uint64_t ToHost64(uint64_t x) { return x; }
>+
>+inline constexpr bool IsLittleEndian() { return true; }
>+
>+#elif defined ABSL_IS_BIG_ENDIAN
>+
>+inline uint16_t FromHost16(uint16_t x) { return gbswap_16(x); }
>+inline uint16_t ToHost16(uint16_t x) { return gbswap_16(x); }
>+
>+inline uint32_t FromHost32(uint32_t x) { return gbswap_32(x); }
>+inline uint32_t ToHost32(uint32_t x) { return gbswap_32(x); }
>+
>+inline uint64_t FromHost64(uint64_t x) { return gbswap_64(x); }
>+inline uint64_t ToHost64(uint64_t x) { return gbswap_64(x); }
>+
>+inline constexpr bool IsLittleEndian() { return false; }
>+
>+#endif /* ENDIAN */
>+
>+// Functions to do unaligned loads and stores in little-endian order.
>+inline uint16_t Load16(const void *p) {
>+  return ToHost16(ABSL_INTERNAL_UNALIGNED_LOAD16(p));
>+}
>+
>+inline void Store16(void *p, uint16_t v) {
>+  ABSL_INTERNAL_UNALIGNED_STORE16(p, FromHost16(v));
>+}
>+
>+inline uint32_t Load32(const void *p) {
>+  return ToHost32(ABSL_INTERNAL_UNALIGNED_LOAD32(p));
>+}
>+
>+inline void Store32(void *p, uint32_t v) {
>+  ABSL_INTERNAL_UNALIGNED_STORE32(p, FromHost32(v));
>+}
>+
>+inline uint64_t Load64(const void *p) {
>+  return ToHost64(ABSL_INTERNAL_UNALIGNED_LOAD64(p));
>+}
>+
>+inline void Store64(void *p, uint64_t v) {
>+  ABSL_INTERNAL_UNALIGNED_STORE64(p, FromHost64(v));
>+}
>+
>+}  // namespace little_endian
>+
>+// Utilities to convert numbers between the current hosts's native byte
>+// order and big-endian byte order (same as network byte order)
>+//
>+// Load/Store methods are alignment safe
>+namespace big_endian {
>+#ifdef ABSL_IS_LITTLE_ENDIAN
>+
>+inline uint16_t FromHost16(uint16_t x) { return gbswap_16(x); }
>+inline uint16_t ToHost16(uint16_t x) { return gbswap_16(x); }
>+
>+inline uint32_t FromHost32(uint32_t x) { return gbswap_32(x); }
>+inline uint32_t ToHost32(uint32_t x) { return gbswap_32(x); }
>+
>+inline uint64_t FromHost64(uint64_t x) { return gbswap_64(x); }
>+inline uint64_t ToHost64(uint64_t x) { return gbswap_64(x); }
>+
>+inline constexpr bool IsLittleEndian() { return true; }
>+
>+#elif defined ABSL_IS_BIG_ENDIAN
>+
>+inline uint16_t FromHost16(uint16_t x) { return x; }
>+inline uint16_t ToHost16(uint16_t x) { return x; }
>+
>+inline uint32_t FromHost32(uint32_t x) { return x; }
>+inline uint32_t ToHost32(uint32_t x) { return x; }
>+
>+inline uint64_t FromHost64(uint64_t x) { return x; }
>+inline uint64_t ToHost64(uint64_t x) { return x; }
>+
>+inline constexpr bool IsLittleEndian() { return false; }
>+
>+#endif /* ENDIAN */
>+
>+// Functions to do unaligned loads and stores in big-endian order.
>+inline uint16_t Load16(const void *p) {
>+  return ToHost16(ABSL_INTERNAL_UNALIGNED_LOAD16(p));
>+}
>+
>+inline void Store16(void *p, uint16_t v) {
>+  ABSL_INTERNAL_UNALIGNED_STORE16(p, FromHost16(v));
>+}
>+
>+inline uint32_t Load32(const void *p) {
>+  return ToHost32(ABSL_INTERNAL_UNALIGNED_LOAD32(p));
>+}
>+
>+inline void Store32(void *p, uint32_t v) {
>+  ABSL_INTERNAL_UNALIGNED_STORE32(p, FromHost32(v));
>+}
>+
>+inline uint64_t Load64(const void *p) {
>+  return ToHost64(ABSL_INTERNAL_UNALIGNED_LOAD64(p));
>+}
>+
>+inline void Store64(void *p, uint64_t v) {
>+  ABSL_INTERNAL_UNALIGNED_STORE64(p, FromHost64(v));
>+}
>+
>+}  // namespace big_endian
>+
>+}  // namespace absl
>+
>+#endif  // ABSL_BASE_INTERNAL_ENDIAN_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/endian_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/endian_test.cc
>new file mode 100644
>index 00000000000..e27691553bc
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/endian_test.cc
>@@ -0,0 +1,263 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/base/internal/endian.h"
>+
>+#include <algorithm>
>+#include <cstdint>
>+#include <limits>
>+#include <random>
>+#include <vector>
>+
>+#include "gtest/gtest.h"
>+#include "absl/base/config.h"
>+
>+namespace absl {
>+namespace {
>+
>+const uint64_t kInitialNumber{0x0123456789abcdef};
>+const uint64_t k64Value{kInitialNumber};
>+const uint32_t k32Value{0x01234567};
>+const uint16_t k16Value{0x0123};
>+const int kNumValuesToTest = 1000000;
>+const int kRandomSeed = 12345;
>+
>+#if defined(ABSL_IS_BIG_ENDIAN)
>+const uint64_t kInitialInNetworkOrder{kInitialNumber};
>+const uint64_t k64ValueLE{0xefcdab8967452301};
>+const uint32_t k32ValueLE{0x67452301};
>+const uint16_t k16ValueLE{0x2301};
>+
>+const uint64_t k64ValueBE{kInitialNumber};
>+const uint32_t k32ValueBE{k32Value};
>+const uint16_t k16ValueBE{k16Value};
>+#elif defined(ABSL_IS_LITTLE_ENDIAN)
>+const uint64_t kInitialInNetworkOrder{0xefcdab8967452301};
>+const uint64_t k64ValueLE{kInitialNumber};
>+const uint32_t k32ValueLE{k32Value};
>+const uint16_t k16ValueLE{k16Value};
>+
>+const uint64_t k64ValueBE{0xefcdab8967452301};
>+const uint32_t k32ValueBE{0x67452301};
>+const uint16_t k16ValueBE{0x2301};
>+#endif
>+
>+template<typename T>
>+std::vector<T> GenerateAllValuesForType() {
>+  std::vector<T> result;
>+  T next = std::numeric_limits<T>::min();
>+  while (true) {
>+    result.push_back(next);
>+    if (next == std::numeric_limits<T>::max()) {
>+      return result;
>+    }
>+    ++next;
>+  }
>+}
>+
>+template<typename T>
>+std::vector<T> GenerateRandomIntegers(size_t numValuesToTest) {
>+  std::vector<T> result;
>+  std::mt19937_64 rng(kRandomSeed);
>+  for (size_t i = 0; i < numValuesToTest; ++i) {
>+    result.push_back(rng());
>+  }
>+  return result;
>+}
>+
>+void ManualByteSwap(char* bytes, int length) {
>+  if (length == 1)
>+    return;
>+
>+  EXPECT_EQ(0, length % 2);
>+  for (int i = 0; i < length / 2; ++i) {
>+    int j = (length - 1) - i;
>+    using std::swap;
>+    swap(bytes[i], bytes[j]);
>+  }
>+}
>+
>+template<typename T>
>+inline T UnalignedLoad(const char* p) {
>+  static_assert(
>+      sizeof(T) == 1 || sizeof(T) == 2 || sizeof(T) == 4 || sizeof(T) == 8,
>+      "Unexpected type size");
>+
>+  switch (sizeof(T)) {
>+    case 1: return *reinterpret_cast<const T*>(p);
>+    case 2:
>+      return ABSL_INTERNAL_UNALIGNED_LOAD16(p);
>+    case 4:
>+      return ABSL_INTERNAL_UNALIGNED_LOAD32(p);
>+    case 8:
>+      return ABSL_INTERNAL_UNALIGNED_LOAD64(p);
>+    default:
>+      // Suppresses invalid "not all control paths return a value" on MSVC
>+      return {};
>+  }
>+}
>+
>+template <typename T, typename ByteSwapper>
>+static void GBSwapHelper(const std::vector<T>& host_values_to_test,
>+                         const ByteSwapper& byte_swapper) {
>+  // Test byte_swapper against a manual byte swap.
>+  for (typename std::vector<T>::const_iterator it = host_values_to_test.begin();
>+       it != host_values_to_test.end(); ++it) {
>+    T host_value = *it;
>+
>+    char actual_value[sizeof(host_value)];
>+    memcpy(actual_value, &host_value, sizeof(host_value));
>+    byte_swapper(actual_value);
>+
>+    char expected_value[sizeof(host_value)];
>+    memcpy(expected_value, &host_value, sizeof(host_value));
>+    ManualByteSwap(expected_value, sizeof(host_value));
>+
>+    ASSERT_EQ(0, memcmp(actual_value, expected_value, sizeof(host_value)))
>+        << "Swap output for 0x" << std::hex << host_value << " does not match. "
>+        << "Expected: 0x" << UnalignedLoad<T>(expected_value) << "; "
>+        << "actual: 0x" <<  UnalignedLoad<T>(actual_value);
>+  }
>+}
>+
>+void Swap16(char* bytes) {
>+  ABSL_INTERNAL_UNALIGNED_STORE16(
>+      bytes, gbswap_16(ABSL_INTERNAL_UNALIGNED_LOAD16(bytes)));
>+}
>+
>+void Swap32(char* bytes) {
>+  ABSL_INTERNAL_UNALIGNED_STORE32(
>+      bytes, gbswap_32(ABSL_INTERNAL_UNALIGNED_LOAD32(bytes)));
>+}
>+
>+void Swap64(char* bytes) {
>+  ABSL_INTERNAL_UNALIGNED_STORE64(
>+      bytes, gbswap_64(ABSL_INTERNAL_UNALIGNED_LOAD64(bytes)));
>+}
>+
>+TEST(EndianessTest, Uint16) {
>+  GBSwapHelper(GenerateAllValuesForType<uint16_t>(), &Swap16);
>+}
>+
>+TEST(EndianessTest, Uint32) {
>+  GBSwapHelper(GenerateRandomIntegers<uint32_t>(kNumValuesToTest), &Swap32);
>+}
>+
>+TEST(EndianessTest, Uint64) {
>+  GBSwapHelper(GenerateRandomIntegers<uint64_t>(kNumValuesToTest), &Swap64);
>+}
>+
>+TEST(EndianessTest, ghtonll_gntohll) {
>+  // Test that absl::ghtonl compiles correctly
>+  uint32_t test = 0x01234567;
>+  EXPECT_EQ(absl::gntohl(absl::ghtonl(test)), test);
>+
>+  uint64_t comp = absl::ghtonll(kInitialNumber);
>+  EXPECT_EQ(comp, kInitialInNetworkOrder);
>+  comp = absl::gntohll(kInitialInNetworkOrder);
>+  EXPECT_EQ(comp, kInitialNumber);
>+
>+  // Test that htonll and ntohll are each others' inverse functions on a
>+  // somewhat assorted batch of numbers. 37 is chosen to not be anything
>+  // particularly nice base 2.
>+  uint64_t value = 1;
>+  for (int i = 0; i < 100; ++i) {
>+    comp = absl::ghtonll(absl::gntohll(value));
>+    EXPECT_EQ(value, comp);
>+    comp = absl::gntohll(absl::ghtonll(value));
>+    EXPECT_EQ(value, comp);
>+    value *= 37;
>+  }
>+}
>+
>+TEST(EndianessTest, little_endian) {
>+  // Check little_endian uint16_t.
>+  uint64_t comp = little_endian::FromHost16(k16Value);
>+  EXPECT_EQ(comp, k16ValueLE);
>+  comp = little_endian::ToHost16(k16ValueLE);
>+  EXPECT_EQ(comp, k16Value);
>+
>+  // Check little_endian uint32_t.
>+  comp = little_endian::FromHost32(k32Value);
>+  EXPECT_EQ(comp, k32ValueLE);
>+  comp = little_endian::ToHost32(k32ValueLE);
>+  EXPECT_EQ(comp, k32Value);
>+
>+  // Check little_endian uint64_t.
>+  comp = little_endian::FromHost64(k64Value);
>+  EXPECT_EQ(comp, k64ValueLE);
>+  comp = little_endian::ToHost64(k64ValueLE);
>+  EXPECT_EQ(comp, k64Value);
>+
>+  // Check little-endian Load and store functions.
>+  uint16_t u16Buf;
>+  uint32_t u32Buf;
>+  uint64_t u64Buf;
>+
>+  little_endian::Store16(&u16Buf, k16Value);
>+  EXPECT_EQ(u16Buf, k16ValueLE);
>+  comp = little_endian::Load16(&u16Buf);
>+  EXPECT_EQ(comp, k16Value);
>+
>+  little_endian::Store32(&u32Buf, k32Value);
>+  EXPECT_EQ(u32Buf, k32ValueLE);
>+  comp = little_endian::Load32(&u32Buf);
>+  EXPECT_EQ(comp, k32Value);
>+
>+  little_endian::Store64(&u64Buf, k64Value);
>+  EXPECT_EQ(u64Buf, k64ValueLE);
>+  comp = little_endian::Load64(&u64Buf);
>+  EXPECT_EQ(comp, k64Value);
>+}
>+
>+TEST(EndianessTest, big_endian) {
>+  // Check big-endian Load and store functions.
>+  uint16_t u16Buf;
>+  uint32_t u32Buf;
>+  uint64_t u64Buf;
>+
>+  unsigned char buffer[10];
>+  big_endian::Store16(&u16Buf, k16Value);
>+  EXPECT_EQ(u16Buf, k16ValueBE);
>+  uint64_t comp = big_endian::Load16(&u16Buf);
>+  EXPECT_EQ(comp, k16Value);
>+
>+  big_endian::Store32(&u32Buf, k32Value);
>+  EXPECT_EQ(u32Buf, k32ValueBE);
>+  comp = big_endian::Load32(&u32Buf);
>+  EXPECT_EQ(comp, k32Value);
>+
>+  big_endian::Store64(&u64Buf, k64Value);
>+  EXPECT_EQ(u64Buf, k64ValueBE);
>+  comp = big_endian::Load64(&u64Buf);
>+  EXPECT_EQ(comp, k64Value);
>+
>+  big_endian::Store16(buffer + 1, k16Value);
>+  EXPECT_EQ(u16Buf, k16ValueBE);
>+  comp = big_endian::Load16(buffer + 1);
>+  EXPECT_EQ(comp, k16Value);
>+
>+  big_endian::Store32(buffer + 1, k32Value);
>+  EXPECT_EQ(u32Buf, k32ValueBE);
>+  comp = big_endian::Load32(buffer + 1);
>+  EXPECT_EQ(comp, k32Value);
>+
>+  big_endian::Store64(buffer + 1, k64Value);
>+  EXPECT_EQ(u64Buf, k64ValueBE);
>+  comp = big_endian::Load64(buffer + 1);
>+  EXPECT_EQ(comp, k64Value);
>+}
>+
>+}  // namespace
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/exception_safety_testing.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/exception_safety_testing.cc
>new file mode 100644
>index 00000000000..f1d081f7e50
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/exception_safety_testing.cc
>@@ -0,0 +1,71 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/base/internal/exception_safety_testing.h"
>+
>+#include "gtest/gtest.h"
>+#include "absl/meta/type_traits.h"
>+
>+namespace testing {
>+
>+exceptions_internal::NoThrowTag nothrow_ctor;
>+
>+exceptions_internal::StrongGuaranteeTagType strong_guarantee;
>+
>+namespace exceptions_internal {
>+
>+int countdown = -1;
>+
>+ConstructorTracker* ConstructorTracker::current_tracker_instance_ = nullptr;
>+
>+void MaybeThrow(absl::string_view msg, bool throw_bad_alloc) {
>+  if (countdown-- == 0) {
>+    if (throw_bad_alloc) throw TestBadAllocException(msg);
>+    throw TestException(msg);
>+  }
>+}
>+
>+testing::AssertionResult FailureMessage(const TestException& e,
>+                                        int countdown) noexcept {
>+  return testing::AssertionFailure() << "Exception thrown from " << e.what();
>+}
>+
>+std::string GetSpecString(TypeSpec spec) {
>+  std::string out;
>+  absl::string_view sep;
>+  const auto append = [&](absl::string_view s) {
>+    absl::StrAppend(&out, sep, s);
>+    sep = " | ";
>+  };
>+  if (static_cast<bool>(TypeSpec::kNoThrowCopy & spec)) {
>+    append("kNoThrowCopy");
>+  }
>+  if (static_cast<bool>(TypeSpec::kNoThrowMove & spec)) {
>+    append("kNoThrowMove");
>+  }
>+  if (static_cast<bool>(TypeSpec::kNoThrowNew & spec)) {
>+    append("kNoThrowNew");
>+  }
>+  return out;
>+}
>+
>+std::string GetSpecString(AllocSpec spec) {
>+  return static_cast<bool>(AllocSpec::kNoThrowAllocate & spec)
>+             ? "kNoThrowAllocate"
>+             : "";
>+}
>+
>+}  // namespace exceptions_internal
>+
>+}  // namespace testing
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/exception_safety_testing.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/exception_safety_testing.h
>new file mode 100644
>index 00000000000..8c2f5093fc4
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/exception_safety_testing.h
>@@ -0,0 +1,1109 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+// Utilities for testing exception-safety
>+
>+#ifndef ABSL_BASE_INTERNAL_EXCEPTION_SAFETY_TESTING_H_
>+#define ABSL_BASE_INTERNAL_EXCEPTION_SAFETY_TESTING_H_
>+
>+#include <cstddef>
>+#include <cstdint>
>+#include <functional>
>+#include <initializer_list>
>+#include <iosfwd>
>+#include <string>
>+#include <tuple>
>+#include <unordered_map>
>+
>+#include "gtest/gtest.h"
>+#include "absl/base/config.h"
>+#include "absl/base/internal/pretty_function.h"
>+#include "absl/memory/memory.h"
>+#include "absl/meta/type_traits.h"
>+#include "absl/strings/string_view.h"
>+#include "absl/strings/substitute.h"
>+#include "absl/types/optional.h"
>+
>+namespace testing {
>+
>+enum class TypeSpec;
>+enum class AllocSpec;
>+
>+constexpr TypeSpec operator|(TypeSpec a, TypeSpec b) {
>+  using T = absl::underlying_type_t<TypeSpec>;
>+  return static_cast<TypeSpec>(static_cast<T>(a) | static_cast<T>(b));
>+}
>+
>+constexpr TypeSpec operator&(TypeSpec a, TypeSpec b) {
>+  using T = absl::underlying_type_t<TypeSpec>;
>+  return static_cast<TypeSpec>(static_cast<T>(a) & static_cast<T>(b));
>+}
>+
>+constexpr AllocSpec operator|(AllocSpec a, AllocSpec b) {
>+  using T = absl::underlying_type_t<AllocSpec>;
>+  return static_cast<AllocSpec>(static_cast<T>(a) | static_cast<T>(b));
>+}
>+
>+constexpr AllocSpec operator&(AllocSpec a, AllocSpec b) {
>+  using T = absl::underlying_type_t<AllocSpec>;
>+  return static_cast<AllocSpec>(static_cast<T>(a) & static_cast<T>(b));
>+}
>+
>+namespace exceptions_internal {
>+
>+std::string GetSpecString(TypeSpec);
>+std::string GetSpecString(AllocSpec);
>+
>+struct NoThrowTag {};
>+struct StrongGuaranteeTagType {};
>+
>+// A simple exception class.  We throw this so that test code can catch
>+// exceptions specifically thrown by ThrowingValue.
>+class TestException {
>+ public:
>+  explicit TestException(absl::string_view msg) : msg_(msg) {}
>+  virtual ~TestException() {}
>+  virtual const char* what() const noexcept { return msg_.c_str(); }
>+
>+ private:
>+  std::string msg_;
>+};
>+
>+// TestBadAllocException exists because allocation functions must throw an
>+// exception which can be caught by a handler of std::bad_alloc.  We use a child
>+// class of std::bad_alloc so we can customise the error message, and also
>+// derive from TestException so we don't accidentally end up catching an actual
>+// bad_alloc exception in TestExceptionSafety.
>+class TestBadAllocException : public std::bad_alloc, public TestException {
>+ public:
>+  explicit TestBadAllocException(absl::string_view msg) : TestException(msg) {}
>+  using TestException::what;
>+};
>+
>+extern int countdown;
>+
>+// Allows the countdown variable to be set manually (defaulting to the initial
>+// value of 0)
>+inline void SetCountdown(int i = 0) { countdown = i; }
>+// Sets the countdown to the terminal value -1
>+inline void UnsetCountdown() { SetCountdown(-1); }
>+
>+void MaybeThrow(absl::string_view msg, bool throw_bad_alloc = false);
>+
>+testing::AssertionResult FailureMessage(const TestException& e,
>+                                        int countdown) noexcept;
>+
>+struct TrackedAddress {
>+  bool is_alive;
>+  std::string description;
>+};
>+
>+// Inspects the constructions and destructions of anything inheriting from
>+// TrackedObject. This allows us to safely "leak" TrackedObjects, as
>+// ConstructorTracker will destroy everything left over in its destructor.
>+class ConstructorTracker {
>+ public:
>+  explicit ConstructorTracker(int count) : countdown_(count) {
>+    assert(current_tracker_instance_ == nullptr);
>+    current_tracker_instance_ = this;
>+  }
>+
>+  ~ConstructorTracker() {
>+    assert(current_tracker_instance_ == this);
>+    current_tracker_instance_ = nullptr;
>+
>+    for (auto& it : address_map_) {
>+      void* address = it.first;
>+      TrackedAddress& tracked_address = it.second;
>+      if (tracked_address.is_alive) {
>+        ADD_FAILURE() << "Object at address " << address
>+                      << " with countdown of " << countdown_
>+                      << " was not destroyed [" << tracked_address.description
>+                      << "]";
>+      }
>+    }
>+  }
>+
>+  static void ObjectConstructed(void* address, std::string description) {
>+    if (!CurrentlyTracking()) return;
>+
>+    TrackedAddress& tracked_address =
>+        current_tracker_instance_->address_map_[address];
>+    if (tracked_address.is_alive) {
>+      ADD_FAILURE() << "Object at address " << address << " with countdown of "
>+                    << current_tracker_instance_->countdown_
>+                    << " was re-constructed. Previously: ["
>+                    << tracked_address.description << "] Now: [" << description
>+                    << "]";
>+    }
>+    tracked_address = {true, std::move(description)};
>+  }
>+
>+  static void ObjectDestructed(void* address) {
>+    if (!CurrentlyTracking()) return;
>+
>+    auto it = current_tracker_instance_->address_map_.find(address);
>+    // Not tracked. Ignore.
>+    if (it == current_tracker_instance_->address_map_.end()) return;
>+
>+    TrackedAddress& tracked_address = it->second;
>+    if (!tracked_address.is_alive) {
>+      ADD_FAILURE() << "Object at address " << address << " with countdown of "
>+                    << current_tracker_instance_->countdown_
>+                    << " was re-destroyed or created prior to construction "
>+                    << "tracking [" << tracked_address.description << "]";
>+    }
>+    tracked_address.is_alive = false;
>+  }
>+
>+ private:
>+  static bool CurrentlyTracking() {
>+    return current_tracker_instance_ != nullptr;
>+  }
>+
>+  std::unordered_map<void*, TrackedAddress> address_map_;
>+  int countdown_;
>+
>+  static ConstructorTracker* current_tracker_instance_;
>+};
>+
>+class TrackedObject {
>+ public:
>+  TrackedObject(const TrackedObject&) = delete;
>+  TrackedObject(TrackedObject&&) = delete;
>+
>+ protected:
>+  explicit TrackedObject(std::string description) {
>+    ConstructorTracker::ObjectConstructed(this, std::move(description));
>+  }
>+
>+  ~TrackedObject() noexcept { ConstructorTracker::ObjectDestructed(this); }
>+};
>+
>+template <typename Factory, typename Operation, typename Invariant>
>+absl::optional<testing::AssertionResult> TestSingleInvariantAtCountdownImpl(
>+    const Factory& factory, const Operation& operation, int count,
>+    const Invariant& invariant) {
>+  auto t_ptr = factory();
>+  absl::optional<testing::AssertionResult> current_res;
>+  SetCountdown(count);
>+  try {
>+    operation(t_ptr.get());
>+  } catch (const exceptions_internal::TestException& e) {
>+    current_res.emplace(invariant(t_ptr.get()));
>+    if (!current_res.value()) {
>+      *current_res << e.what() << " failed invariant check";
>+    }
>+  }
>+  UnsetCountdown();
>+  return current_res;
>+}
>+
>+template <typename Factory, typename Operation>
>+absl::optional<testing::AssertionResult> TestSingleInvariantAtCountdownImpl(
>+    const Factory& factory, const Operation& operation, int count,
>+    StrongGuaranteeTagType) {
>+  using TPtr = typename decltype(factory())::pointer;
>+  auto t_is_strong = [&](TPtr t) { return *t == *factory(); };
>+  return TestSingleInvariantAtCountdownImpl(factory, operation, count,
>+                                            t_is_strong);
>+}
>+
>+template <typename Factory, typename Operation, typename Invariant>
>+int TestSingleInvariantAtCountdown(
>+    const Factory& factory, const Operation& operation, int count,
>+    const Invariant& invariant,
>+    absl::optional<testing::AssertionResult>* reduced_res) {
>+  // If reduced_res is empty, it means the current call to
>+  // TestSingleInvariantAtCountdown(...) is the first test being run so we do
>+  // want to run it. Alternatively, if it's not empty (meaning a previous test
>+  // has run) we want to check if it passed. If the previous test did pass, we
>+  // want to contine running tests so we do want to run the current one. If it
>+  // failed, we want to short circuit so as not to overwrite the AssertionResult
>+  // output. If that's the case, we do not run the current test and instead we
>+  // simply return.
>+  if (!reduced_res->has_value() || reduced_res->value()) {
>+    *reduced_res = TestSingleInvariantAtCountdownImpl(factory, operation, count,
>+                                                      invariant);
>+  }
>+  return 0;
>+}
>+
>+template <typename Factory, typename Operation, typename... Invariants>
>+inline absl::optional<testing::AssertionResult> TestAllInvariantsAtCountdown(
>+    const Factory& factory, const Operation& operation, int count,
>+    const Invariants&... invariants) {
>+  absl::optional<testing::AssertionResult> reduced_res;
>+
>+  // Run each checker, short circuiting after the first failure
>+  int dummy[] = {
>+      0, (TestSingleInvariantAtCountdown(factory, operation, count, invariants,
>+                                         &reduced_res))...};
>+  static_cast<void>(dummy);
>+  return reduced_res;
>+}
>+
>+}  // namespace exceptions_internal
>+
>+extern exceptions_internal::NoThrowTag nothrow_ctor;
>+
>+extern exceptions_internal::StrongGuaranteeTagType strong_guarantee;
>+
>+// A test class which is convertible to bool.  The conversion can be
>+// instrumented to throw at a controlled time.
>+class ThrowingBool {
>+ public:
>+  ThrowingBool(bool b) noexcept : b_(b) {}  // NOLINT(runtime/explicit)
>+  operator bool() const {                   // NOLINT
>+    exceptions_internal::MaybeThrow(ABSL_PRETTY_FUNCTION);
>+    return b_;
>+  }
>+
>+ private:
>+  bool b_;
>+};
>+
>+/*
>+ * Configuration enum for the ThrowingValue type that defines behavior for the
>+ * lifetime of the instance. Use testing::nothrow_ctor to prevent the integer
>+ * constructor from throwing.
>+ *
>+ * kEverythingThrows: Every operation can throw an exception
>+ * kNoThrowCopy: Copy construction and copy assignment will not throw
>+ * kNoThrowMove: Move construction and move assignment will not throw
>+ * kNoThrowNew: Overloaded operators new and new[] will not throw
>+ */
>+enum class TypeSpec {
>+  kEverythingThrows = 0,
>+  kNoThrowCopy = 1,
>+  kNoThrowMove = 1 << 1,
>+  kNoThrowNew = 1 << 2,
>+};
>+
>+/*
>+ * A testing class instrumented to throw an exception at a controlled time.
>+ *
>+ * ThrowingValue implements a slightly relaxed version of the Regular concept --
>+ * that is it's a value type with the expected semantics.  It also implements
>+ * arithmetic operations.  It doesn't implement member and pointer operators
>+ * like operator-> or operator[].
>+ *
>+ * ThrowingValue can be instrumented to have certain operations be noexcept by
>+ * using compile-time bitfield template arguments.  That is, to make an
>+ * ThrowingValue which has noexcept move construction/assignment and noexcept
>+ * copy construction/assignment, use the following:
>+ *   ThrowingValue<testing::kNoThrowMove | testing::kNoThrowCopy> my_thrwr{val};
>+ */
>+template <TypeSpec Spec = TypeSpec::kEverythingThrows>
>+class ThrowingValue : private exceptions_internal::TrackedObject {
>+  static constexpr bool IsSpecified(TypeSpec spec) {
>+    return static_cast<bool>(Spec & spec);
>+  }
>+
>+  static constexpr int kDefaultValue = 0;
>+  static constexpr int kBadValue = 938550620;
>+
>+ public:
>+  ThrowingValue() : TrackedObject(GetInstanceString(kDefaultValue)) {
>+    exceptions_internal::MaybeThrow(ABSL_PRETTY_FUNCTION);
>+    dummy_ = kDefaultValue;
>+  }
>+
>+  ThrowingValue(const ThrowingValue& other) noexcept(
>+      IsSpecified(TypeSpec::kNoThrowCopy))
>+      : TrackedObject(GetInstanceString(other.dummy_)) {
>+    if (!IsSpecified(TypeSpec::kNoThrowCopy)) {
>+      exceptions_internal::MaybeThrow(ABSL_PRETTY_FUNCTION);
>+    }
>+    dummy_ = other.dummy_;
>+  }
>+
>+  ThrowingValue(ThrowingValue&& other) noexcept(
>+      IsSpecified(TypeSpec::kNoThrowMove))
>+      : TrackedObject(GetInstanceString(other.dummy_)) {
>+    if (!IsSpecified(TypeSpec::kNoThrowMove)) {
>+      exceptions_internal::MaybeThrow(ABSL_PRETTY_FUNCTION);
>+    }
>+    dummy_ = other.dummy_;
>+  }
>+
>+  explicit ThrowingValue(int i) : TrackedObject(GetInstanceString(i)) {
>+    exceptions_internal::MaybeThrow(ABSL_PRETTY_FUNCTION);
>+    dummy_ = i;
>+  }
>+
>+  ThrowingValue(int i, exceptions_internal::NoThrowTag) noexcept
>+      : TrackedObject(GetInstanceString(i)), dummy_(i) {}
>+
>+  // absl expects nothrow destructors
>+  ~ThrowingValue() noexcept = default;
>+
>+  ThrowingValue& operator=(const ThrowingValue& other) noexcept(
>+      IsSpecified(TypeSpec::kNoThrowCopy)) {
>+    dummy_ = kBadValue;
>+    if (!IsSpecified(TypeSpec::kNoThrowCopy)) {
>+      exceptions_internal::MaybeThrow(ABSL_PRETTY_FUNCTION);
>+    }
>+    dummy_ = other.dummy_;
>+    return *this;
>+  }
>+
>+  ThrowingValue& operator=(ThrowingValue&& other) noexcept(
>+      IsSpecified(TypeSpec::kNoThrowMove)) {
>+    dummy_ = kBadValue;
>+    if (!IsSpecified(TypeSpec::kNoThrowMove)) {
>+      exceptions_internal::MaybeThrow(ABSL_PRETTY_FUNCTION);
>+    }
>+    dummy_ = other.dummy_;
>+    return *this;
>+  }
>+
>+  // Arithmetic Operators
>+  ThrowingValue operator+(const ThrowingValue& other) const {
>+    exceptions_internal::MaybeThrow(ABSL_PRETTY_FUNCTION);
>+    return ThrowingValue(dummy_ + other.dummy_, nothrow_ctor);
>+  }
>+
>+  ThrowingValue operator+() const {
>+    exceptions_internal::MaybeThrow(ABSL_PRETTY_FUNCTION);
>+    return ThrowingValue(dummy_, nothrow_ctor);
>+  }
>+
>+  ThrowingValue operator-(const ThrowingValue& other) const {
>+    exceptions_internal::MaybeThrow(ABSL_PRETTY_FUNCTION);
>+    return ThrowingValue(dummy_ - other.dummy_, nothrow_ctor);
>+  }
>+
>+  ThrowingValue operator-() const {
>+    exceptions_internal::MaybeThrow(ABSL_PRETTY_FUNCTION);
>+    return ThrowingValue(-dummy_, nothrow_ctor);
>+  }
>+
>+  ThrowingValue& operator++() {
>+    exceptions_internal::MaybeThrow(ABSL_PRETTY_FUNCTION);
>+    ++dummy_;
>+    return *this;
>+  }
>+
>+  ThrowingValue operator++(int) {
>+    exceptions_internal::MaybeThrow(ABSL_PRETTY_FUNCTION);
>+    auto out = ThrowingValue(dummy_, nothrow_ctor);
>+    ++dummy_;
>+    return out;
>+  }
>+
>+  ThrowingValue& operator--() {
>+    exceptions_internal::MaybeThrow(ABSL_PRETTY_FUNCTION);
>+    --dummy_;
>+    return *this;
>+  }
>+
>+  ThrowingValue operator--(int) {
>+    exceptions_internal::MaybeThrow(ABSL_PRETTY_FUNCTION);
>+    auto out = ThrowingValue(dummy_, nothrow_ctor);
>+    --dummy_;
>+    return out;
>+  }
>+
>+  ThrowingValue operator*(const ThrowingValue& other) const {
>+    exceptions_internal::MaybeThrow(ABSL_PRETTY_FUNCTION);
>+    return ThrowingValue(dummy_ * other.dummy_, nothrow_ctor);
>+  }
>+
>+  ThrowingValue operator/(const ThrowingValue& other) const {
>+    exceptions_internal::MaybeThrow(ABSL_PRETTY_FUNCTION);
>+    return ThrowingValue(dummy_ / other.dummy_, nothrow_ctor);
>+  }
>+
>+  ThrowingValue operator%(const ThrowingValue& other) const {
>+    exceptions_internal::MaybeThrow(ABSL_PRETTY_FUNCTION);
>+    return ThrowingValue(dummy_ % other.dummy_, nothrow_ctor);
>+  }
>+
>+  ThrowingValue operator<<(int shift) const {
>+    exceptions_internal::MaybeThrow(ABSL_PRETTY_FUNCTION);
>+    return ThrowingValue(dummy_ << shift, nothrow_ctor);
>+  }
>+
>+  ThrowingValue operator>>(int shift) const {
>+    exceptions_internal::MaybeThrow(ABSL_PRETTY_FUNCTION);
>+    return ThrowingValue(dummy_ >> shift, nothrow_ctor);
>+  }
>+
>+  // Comparison Operators
>+  // NOTE: We use `ThrowingBool` instead of `bool` because most STL
>+  // types/containers requires T to be convertible to bool.
>+  friend ThrowingBool operator==(const ThrowingValue& a,
>+                                 const ThrowingValue& b) {
>+    exceptions_internal::MaybeThrow(ABSL_PRETTY_FUNCTION);
>+    return a.dummy_ == b.dummy_;
>+  }
>+  friend ThrowingBool operator!=(const ThrowingValue& a,
>+                                 const ThrowingValue& b) {
>+    exceptions_internal::MaybeThrow(ABSL_PRETTY_FUNCTION);
>+    return a.dummy_ != b.dummy_;
>+  }
>+  friend ThrowingBool operator<(const ThrowingValue& a,
>+                                const ThrowingValue& b) {
>+    exceptions_internal::MaybeThrow(ABSL_PRETTY_FUNCTION);
>+    return a.dummy_ < b.dummy_;
>+  }
>+  friend ThrowingBool operator<=(const ThrowingValue& a,
>+                                 const ThrowingValue& b) {
>+    exceptions_internal::MaybeThrow(ABSL_PRETTY_FUNCTION);
>+    return a.dummy_ <= b.dummy_;
>+  }
>+  friend ThrowingBool operator>(const ThrowingValue& a,
>+                                const ThrowingValue& b) {
>+    exceptions_internal::MaybeThrow(ABSL_PRETTY_FUNCTION);
>+    return a.dummy_ > b.dummy_;
>+  }
>+  friend ThrowingBool operator>=(const ThrowingValue& a,
>+                                 const ThrowingValue& b) {
>+    exceptions_internal::MaybeThrow(ABSL_PRETTY_FUNCTION);
>+    return a.dummy_ >= b.dummy_;
>+  }
>+
>+  // Logical Operators
>+  ThrowingBool operator!() const {
>+    exceptions_internal::MaybeThrow(ABSL_PRETTY_FUNCTION);
>+    return !dummy_;
>+  }
>+
>+  ThrowingBool operator&&(const ThrowingValue& other) const {
>+    exceptions_internal::MaybeThrow(ABSL_PRETTY_FUNCTION);
>+    return dummy_ && other.dummy_;
>+  }
>+
>+  ThrowingBool operator||(const ThrowingValue& other) const {
>+    exceptions_internal::MaybeThrow(ABSL_PRETTY_FUNCTION);
>+    return dummy_ || other.dummy_;
>+  }
>+
>+  // Bitwise Logical Operators
>+  ThrowingValue operator~() const {
>+    exceptions_internal::MaybeThrow(ABSL_PRETTY_FUNCTION);
>+    return ThrowingValue(~dummy_, nothrow_ctor);
>+  }
>+
>+  ThrowingValue operator&(const ThrowingValue& other) const {
>+    exceptions_internal::MaybeThrow(ABSL_PRETTY_FUNCTION);
>+    return ThrowingValue(dummy_ & other.dummy_, nothrow_ctor);
>+  }
>+
>+  ThrowingValue operator|(const ThrowingValue& other) const {
>+    exceptions_internal::MaybeThrow(ABSL_PRETTY_FUNCTION);
>+    return ThrowingValue(dummy_ | other.dummy_, nothrow_ctor);
>+  }
>+
>+  ThrowingValue operator^(const ThrowingValue& other) const {
>+    exceptions_internal::MaybeThrow(ABSL_PRETTY_FUNCTION);
>+    return ThrowingValue(dummy_ ^ other.dummy_, nothrow_ctor);
>+  }
>+
>+  // Compound Assignment operators
>+  ThrowingValue& operator+=(const ThrowingValue& other) {
>+    exceptions_internal::MaybeThrow(ABSL_PRETTY_FUNCTION);
>+    dummy_ += other.dummy_;
>+    return *this;
>+  }
>+
>+  ThrowingValue& operator-=(const ThrowingValue& other) {
>+    exceptions_internal::MaybeThrow(ABSL_PRETTY_FUNCTION);
>+    dummy_ -= other.dummy_;
>+    return *this;
>+  }
>+
>+  ThrowingValue& operator*=(const ThrowingValue& other) {
>+    exceptions_internal::MaybeThrow(ABSL_PRETTY_FUNCTION);
>+    dummy_ *= other.dummy_;
>+    return *this;
>+  }
>+
>+  ThrowingValue& operator/=(const ThrowingValue& other) {
>+    exceptions_internal::MaybeThrow(ABSL_PRETTY_FUNCTION);
>+    dummy_ /= other.dummy_;
>+    return *this;
>+  }
>+
>+  ThrowingValue& operator%=(const ThrowingValue& other) {
>+    exceptions_internal::MaybeThrow(ABSL_PRETTY_FUNCTION);
>+    dummy_ %= other.dummy_;
>+    return *this;
>+  }
>+
>+  ThrowingValue& operator&=(const ThrowingValue& other) {
>+    exceptions_internal::MaybeThrow(ABSL_PRETTY_FUNCTION);
>+    dummy_ &= other.dummy_;
>+    return *this;
>+  }
>+
>+  ThrowingValue& operator|=(const ThrowingValue& other) {
>+    exceptions_internal::MaybeThrow(ABSL_PRETTY_FUNCTION);
>+    dummy_ |= other.dummy_;
>+    return *this;
>+  }
>+
>+  ThrowingValue& operator^=(const ThrowingValue& other) {
>+    exceptions_internal::MaybeThrow(ABSL_PRETTY_FUNCTION);
>+    dummy_ ^= other.dummy_;
>+    return *this;
>+  }
>+
>+  ThrowingValue& operator<<=(int shift) {
>+    exceptions_internal::MaybeThrow(ABSL_PRETTY_FUNCTION);
>+    dummy_ <<= shift;
>+    return *this;
>+  }
>+
>+  ThrowingValue& operator>>=(int shift) {
>+    exceptions_internal::MaybeThrow(ABSL_PRETTY_FUNCTION);
>+    dummy_ >>= shift;
>+    return *this;
>+  }
>+
>+  // Pointer operators
>+  void operator&() const = delete;  // NOLINT(runtime/operator)
>+
>+  // Stream operators
>+  friend std::ostream& operator<<(std::ostream& os, const ThrowingValue& tv) {
>+    exceptions_internal::MaybeThrow(ABSL_PRETTY_FUNCTION);
>+    return os << GetInstanceString(tv.dummy_);
>+  }
>+
>+  friend std::istream& operator>>(std::istream& is, const ThrowingValue&) {
>+    exceptions_internal::MaybeThrow(ABSL_PRETTY_FUNCTION);
>+    return is;
>+  }
>+
>+  // Memory management operators
>+  // Args.. allows us to overload regular and placement new in one shot
>+  template <typename... Args>
>+  static void* operator new(size_t s, Args&&... args) noexcept(
>+      IsSpecified(TypeSpec::kNoThrowNew)) {
>+    if (!IsSpecified(TypeSpec::kNoThrowNew)) {
>+      exceptions_internal::MaybeThrow(ABSL_PRETTY_FUNCTION, true);
>+    }
>+    return ::operator new(s, std::forward<Args>(args)...);
>+  }
>+
>+  template <typename... Args>
>+  static void* operator new[](size_t s, Args&&... args) noexcept(
>+      IsSpecified(TypeSpec::kNoThrowNew)) {
>+    if (!IsSpecified(TypeSpec::kNoThrowNew)) {
>+      exceptions_internal::MaybeThrow(ABSL_PRETTY_FUNCTION, true);
>+    }
>+    return ::operator new[](s, std::forward<Args>(args)...);
>+  }
>+
>+  // Abseil doesn't support throwing overloaded operator delete.  These are
>+  // provided so a throwing operator-new can clean up after itself.
>+  //
>+  // We provide both regular and templated operator delete because if only the
>+  // templated version is provided as we did with operator new, the compiler has
>+  // no way of knowing which overload of operator delete to call. See
>+  // http://en.cppreference.com/w/cpp/memory/new/operator_delete and
>+  // http://en.cppreference.com/w/cpp/language/delete for the gory details.
>+  void operator delete(void* p) noexcept { ::operator delete(p); }
>+
>+  template <typename... Args>
>+  void operator delete(void* p, Args&&... args) noexcept {
>+    ::operator delete(p, std::forward<Args>(args)...);
>+  }
>+
>+  void operator delete[](void* p) noexcept { return ::operator delete[](p); }
>+
>+  template <typename... Args>
>+  void operator delete[](void* p, Args&&... args) noexcept {
>+    return ::operator delete[](p, std::forward<Args>(args)...);
>+  }
>+
>+  // Non-standard access to the actual contained value.  No need for this to
>+  // throw.
>+  int& Get() noexcept { return dummy_; }
>+  const int& Get() const noexcept { return dummy_; }
>+
>+ private:
>+  static std::string GetInstanceString(int dummy) {
>+    return absl::StrCat("ThrowingValue<",
>+                        exceptions_internal::GetSpecString(Spec), ">(", dummy,
>+                        ")");
>+  }
>+
>+  int dummy_;
>+};
>+// While not having to do with exceptions, explicitly delete comma operator, to
>+// make sure we don't use it on user-supplied types.
>+template <TypeSpec Spec, typename T>
>+void operator,(const ThrowingValue<Spec>&, T&&) = delete;
>+template <TypeSpec Spec, typename T>
>+void operator,(T&&, const ThrowingValue<Spec>&) = delete;
>+
>+/*
>+ * Configuration enum for the ThrowingAllocator type that defines behavior for
>+ * the lifetime of the instance.
>+ *
>+ * kEverythingThrows: Calls to the member functions may throw
>+ * kNoThrowAllocate: Calls to the member functions will not throw
>+ */
>+enum class AllocSpec {
>+  kEverythingThrows = 0,
>+  kNoThrowAllocate = 1,
>+};
>+
>+/*
>+ * An allocator type which is instrumented to throw at a controlled time, or not
>+ * to throw, using AllocSpec. The supported settings are the default of every
>+ * function which is allowed to throw in a conforming allocator possibly
>+ * throwing, or nothing throws, in line with the ABSL_ALLOCATOR_THROWS
>+ * configuration macro.
>+ */
>+template <typename T, AllocSpec Spec = AllocSpec::kEverythingThrows>
>+class ThrowingAllocator : private exceptions_internal::TrackedObject {
>+  static constexpr bool IsSpecified(AllocSpec spec) {
>+    return static_cast<bool>(Spec & spec);
>+  }
>+
>+ public:
>+  using pointer = T*;
>+  using const_pointer = const T*;
>+  using reference = T&;
>+  using const_reference = const T&;
>+  using void_pointer = void*;
>+  using const_void_pointer = const void*;
>+  using value_type = T;
>+  using size_type = size_t;
>+  using difference_type = ptrdiff_t;
>+
>+  using is_nothrow =
>+      std::integral_constant<bool, Spec == AllocSpec::kNoThrowAllocate>;
>+  using propagate_on_container_copy_assignment = std::true_type;
>+  using propagate_on_container_move_assignment = std::true_type;
>+  using propagate_on_container_swap = std::true_type;
>+  using is_always_equal = std::false_type;
>+
>+  ThrowingAllocator() : TrackedObject(GetInstanceString(next_id_)) {
>+    exceptions_internal::MaybeThrow(ABSL_PRETTY_FUNCTION);
>+    dummy_ = std::make_shared<const int>(next_id_++);
>+  }
>+
>+  template <typename U>
>+  ThrowingAllocator(const ThrowingAllocator<U, Spec>& other) noexcept  // NOLINT
>+      : TrackedObject(GetInstanceString(*other.State())),
>+        dummy_(other.State()) {}
>+
>+  // According to C++11 standard [17.6.3.5], Table 28, the move/copy ctors of
>+  // allocator shall not exit via an exception, thus they are marked noexcept.
>+  ThrowingAllocator(const ThrowingAllocator& other) noexcept
>+      : TrackedObject(GetInstanceString(*other.State())),
>+        dummy_(other.State()) {}
>+
>+  template <typename U>
>+  ThrowingAllocator(ThrowingAllocator<U, Spec>&& other) noexcept  // NOLINT
>+      : TrackedObject(GetInstanceString(*other.State())),
>+        dummy_(std::move(other.State())) {}
>+
>+  ThrowingAllocator(ThrowingAllocator&& other) noexcept
>+      : TrackedObject(GetInstanceString(*other.State())),
>+        dummy_(std::move(other.State())) {}
>+
>+  ~ThrowingAllocator() noexcept = default;
>+
>+  ThrowingAllocator& operator=(const ThrowingAllocator& other) noexcept {
>+    dummy_ = other.State();
>+    return *this;
>+  }
>+
>+  template <typename U>
>+  ThrowingAllocator& operator=(
>+      const ThrowingAllocator<U, Spec>& other) noexcept {
>+    dummy_ = other.State();
>+    return *this;
>+  }
>+
>+  template <typename U>
>+  ThrowingAllocator& operator=(ThrowingAllocator<U, Spec>&& other) noexcept {
>+    dummy_ = std::move(other.State());
>+    return *this;
>+  }
>+
>+  template <typename U>
>+  struct rebind {
>+    using other = ThrowingAllocator<U, Spec>;
>+  };
>+
>+  pointer allocate(size_type n) noexcept(
>+      IsSpecified(AllocSpec::kNoThrowAllocate)) {
>+    ReadStateAndMaybeThrow(ABSL_PRETTY_FUNCTION);
>+    return static_cast<pointer>(::operator new(n * sizeof(T)));
>+  }
>+
>+  pointer allocate(size_type n, const_void_pointer) noexcept(
>+      IsSpecified(AllocSpec::kNoThrowAllocate)) {
>+    return allocate(n);
>+  }
>+
>+  void deallocate(pointer ptr, size_type) noexcept {
>+    ReadState();
>+    ::operator delete(static_cast<void*>(ptr));
>+  }
>+
>+  template <typename U, typename... Args>
>+  void construct(U* ptr, Args&&... args) noexcept(
>+      IsSpecified(AllocSpec::kNoThrowAllocate)) {
>+    ReadStateAndMaybeThrow(ABSL_PRETTY_FUNCTION);
>+    ::new (static_cast<void*>(ptr)) U(std::forward<Args>(args)...);
>+  }
>+
>+  template <typename U>
>+  void destroy(U* p) noexcept {
>+    ReadState();
>+    p->~U();
>+  }
>+
>+  size_type max_size() const noexcept {
>+    return std::numeric_limits<difference_type>::max() / sizeof(value_type);
>+  }
>+
>+  ThrowingAllocator select_on_container_copy_construction() noexcept(
>+      IsSpecified(AllocSpec::kNoThrowAllocate)) {
>+    auto& out = *this;
>+    ReadStateAndMaybeThrow(ABSL_PRETTY_FUNCTION);
>+    return out;
>+  }
>+
>+  template <typename U>
>+  bool operator==(const ThrowingAllocator<U, Spec>& other) const noexcept {
>+    return dummy_ == other.dummy_;
>+  }
>+
>+  template <typename U>
>+  bool operator!=(const ThrowingAllocator<U, Spec>& other) const noexcept {
>+    return dummy_ != other.dummy_;
>+  }
>+
>+  template <typename, AllocSpec>
>+  friend class ThrowingAllocator;
>+
>+ private:
>+  static std::string GetInstanceString(int dummy) {
>+    return absl::StrCat("ThrowingAllocator<",
>+                        exceptions_internal::GetSpecString(Spec), ">(", dummy,
>+                        ")");
>+  }
>+
>+  const std::shared_ptr<const int>& State() const { return dummy_; }
>+  std::shared_ptr<const int>& State() { return dummy_; }
>+
>+  void ReadState() {
>+    // we know that this will never be true, but the compiler doesn't, so this
>+    // should safely force a read of the value.
>+    if (*dummy_ < 0) std::abort();
>+  }
>+
>+  void ReadStateAndMaybeThrow(absl::string_view msg) const {
>+    if (!IsSpecified(AllocSpec::kNoThrowAllocate)) {
>+      exceptions_internal::MaybeThrow(
>+          absl::Substitute("Allocator id $0 threw from $1", *dummy_, msg));
>+    }
>+  }
>+
>+  static int next_id_;
>+  std::shared_ptr<const int> dummy_;
>+};
>+
>+template <typename T, AllocSpec Spec>
>+int ThrowingAllocator<T, Spec>::next_id_ = 0;
>+
>+// Tests for resource leaks by attempting to construct a T using args repeatedly
>+// until successful, using the countdown method.  Side effects can then be
>+// tested for resource leaks.
>+template <typename T, typename... Args>
>+void TestThrowingCtor(Args&&... args) {
>+  struct Cleanup {
>+    ~Cleanup() { exceptions_internal::UnsetCountdown(); }
>+  } c;
>+  for (int count = 0;; ++count) {
>+    exceptions_internal::ConstructorTracker ct(count);
>+    exceptions_internal::SetCountdown(count);
>+    try {
>+      T temp(std::forward<Args>(args)...);
>+      static_cast<void>(temp);
>+      break;
>+    } catch (const exceptions_internal::TestException&) {
>+    }
>+  }
>+}
>+
>+// Tests the nothrow guarantee of the provided nullary operation. If the an
>+// exception is thrown, the result will be AssertionFailure(). Otherwise, it
>+// will be AssertionSuccess().
>+template <typename Operation>
>+testing::AssertionResult TestNothrowOp(const Operation& operation) {
>+  struct Cleanup {
>+    Cleanup() { exceptions_internal::SetCountdown(); }
>+    ~Cleanup() { exceptions_internal::UnsetCountdown(); }
>+  } c;
>+  try {
>+    operation();
>+    return testing::AssertionSuccess();
>+  } catch (exceptions_internal::TestException) {
>+    return testing::AssertionFailure()
>+           << "TestException thrown during call to operation() when nothrow "
>+              "guarantee was expected.";
>+  } catch (...) {
>+    return testing::AssertionFailure()
>+           << "Unknown exception thrown during call to operation() when "
>+              "nothrow guarantee was expected.";
>+  }
>+}
>+
>+namespace exceptions_internal {
>+
>+// Dummy struct for ExceptionSafetyTester<> partial state.
>+struct UninitializedT {};
>+
>+template <typename T>
>+class DefaultFactory {
>+ public:
>+  explicit DefaultFactory(const T& t) : t_(t) {}
>+  std::unique_ptr<T> operator()() const { return absl::make_unique<T>(t_); }
>+
>+ private:
>+  T t_;
>+};
>+
>+template <size_t LazyInvariantsCount, typename LazyFactory,
>+          typename LazyOperation>
>+using EnableIfTestable = typename absl::enable_if_t<
>+    LazyInvariantsCount != 0 &&
>+    !std::is_same<LazyFactory, UninitializedT>::value &&
>+    !std::is_same<LazyOperation, UninitializedT>::value>;
>+
>+template <typename Factory = UninitializedT,
>+          typename Operation = UninitializedT, typename... Invariants>
>+class ExceptionSafetyTester;
>+
>+}  // namespace exceptions_internal
>+
>+exceptions_internal::ExceptionSafetyTester<> MakeExceptionSafetyTester();
>+
>+namespace exceptions_internal {
>+
>+/*
>+ * Builds a tester object that tests if performing a operation on a T follows
>+ * exception safety guarantees. Verification is done via invariant assertion
>+ * callbacks applied to T instances post-throw.
>+ *
>+ * Template parameters for ExceptionSafetyTester:
>+ *
>+ * - Factory: The factory object (passed in via tester.WithFactory(...) or
>+ *   tester.WithInitialValue(...)) must be invocable with the signature
>+ *   `std::unique_ptr<T> operator()() const` where T is the type being tested.
>+ *   It is used for reliably creating identical T instances to test on.
>+ *
>+ * - Operation: The operation object (passsed in via tester.WithOperation(...)
>+ *   or tester.Test(...)) must be invocable with the signature
>+ *   `void operator()(T*) const` where T is the type being tested. It is used
>+ *   for performing steps on a T instance that may throw and that need to be
>+ *   checked for exception safety. Each call to the operation will receive a
>+ *   fresh T instance so it's free to modify and destroy the T instances as it
>+ *   pleases.
>+ *
>+ * - Invariants...: The invariant assertion callback objects (passed in via
>+ *   tester.WithInvariants(...)) must be invocable with the signature
>+ *   `testing::AssertionResult operator()(T*) const` where T is the type being
>+ *   tested. Invariant assertion callbacks are provided T instances post-throw.
>+ *   They must return testing::AssertionSuccess when the type invariants of the
>+ *   provided T instance hold. If the type invariants of the T instance do not
>+ *   hold, they must return testing::AssertionFailure. Execution order of
>+ *   Invariants... is unspecified. They will each individually get a fresh T
>+ *   instance so they are free to modify and destroy the T instances as they
>+ *   please.
>+ */
>+template <typename Factory, typename Operation, typename... Invariants>
>+class ExceptionSafetyTester {
>+ public:
>+  /*
>+   * Returns a new ExceptionSafetyTester with an included T factory based on the
>+   * provided T instance. The existing factory will not be included in the newly
>+   * created tester instance. The created factory returns a new T instance by
>+   * copy-constructing the provided const T& t.
>+   *
>+   * Preconditions for tester.WithInitialValue(const T& t):
>+   *
>+   * - The const T& t object must be copy-constructible where T is the type
>+   *   being tested. For non-copy-constructible objects, use the method
>+   *   tester.WithFactory(...).
>+   */
>+  template <typename T>
>+  ExceptionSafetyTester<DefaultFactory<T>, Operation, Invariants...>
>+  WithInitialValue(const T& t) const {
>+    return WithFactory(DefaultFactory<T>(t));
>+  }
>+
>+  /*
>+   * Returns a new ExceptionSafetyTester with the provided T factory included.
>+   * The existing factory will not be included in the newly-created tester
>+   * instance. This method is intended for use with types lacking a copy
>+   * constructor. Types that can be copy-constructed should instead use the
>+   * method tester.WithInitialValue(...).
>+   */
>+  template <typename NewFactory>
>+  ExceptionSafetyTester<absl::decay_t<NewFactory>, Operation, Invariants...>
>+  WithFactory(const NewFactory& new_factory) const {
>+    return {new_factory, operation_, invariants_};
>+  }
>+
>+  /*
>+   * Returns a new ExceptionSafetyTester with the provided testable operation
>+   * included. The existing operation will not be included in the newly created
>+   * tester.
>+   */
>+  template <typename NewOperation>
>+  ExceptionSafetyTester<Factory, absl::decay_t<NewOperation>, Invariants...>
>+  WithOperation(const NewOperation& new_operation) const {
>+    return {factory_, new_operation, invariants_};
>+  }
>+
>+  /*
>+   * Returns a new ExceptionSafetyTester with the provided MoreInvariants...
>+   * combined with the Invariants... that were already included in the instance
>+   * on which the method was called. Invariants... cannot be removed or replaced
>+   * once added to an ExceptionSafetyTester instance. A fresh object must be
>+   * created in order to get an empty Invariants... list.
>+   *
>+   * In addition to passing in custom invariant assertion callbacks, this method
>+   * accepts `testing::strong_guarantee` as an argument which checks T instances
>+   * post-throw against freshly created T instances via operator== to verify
>+   * that any state changes made during the execution of the operation were
>+   * properly rolled back.
>+   */
>+  template <typename... MoreInvariants>
>+  ExceptionSafetyTester<Factory, Operation, Invariants...,
>+                        absl::decay_t<MoreInvariants>...>
>+  WithInvariants(const MoreInvariants&... more_invariants) const {
>+    return {factory_, operation_,
>+            std::tuple_cat(invariants_,
>+                           std::tuple<absl::decay_t<MoreInvariants>...>(
>+                               more_invariants...))};
>+  }
>+
>+  /*
>+   * Returns a testing::AssertionResult that is the reduced result of the
>+   * exception safety algorithm. The algorithm short circuits and returns
>+   * AssertionFailure after the first invariant callback returns an
>+   * AssertionFailure. Otherwise, if all invariant callbacks return an
>+   * AssertionSuccess, the reduced result is AssertionSuccess.
>+   *
>+   * The passed-in testable operation will not be saved in a new tester instance
>+   * nor will it modify/replace the existing tester instance. This is useful
>+   * when each operation being tested is unique and does not need to be reused.
>+   *
>+   * Preconditions for tester.Test(const NewOperation& new_operation):
>+   *
>+   * - May only be called after at least one invariant assertion callback and a
>+   *   factory or initial value have been provided.
>+   */
>+  template <
>+      typename NewOperation,
>+      typename = EnableIfTestable<sizeof...(Invariants), Factory, NewOperation>>
>+  testing::AssertionResult Test(const NewOperation& new_operation) const {
>+    return TestImpl(new_operation, absl::index_sequence_for<Invariants...>());
>+  }
>+
>+  /*
>+   * Returns a testing::AssertionResult that is the reduced result of the
>+   * exception safety algorithm. The algorithm short circuits and returns
>+   * AssertionFailure after the first invariant callback returns an
>+   * AssertionFailure. Otherwise, if all invariant callbacks return an
>+   * AssertionSuccess, the reduced result is AssertionSuccess.
>+   *
>+   * Preconditions for tester.Test():
>+   *
>+   * - May only be called after at least one invariant assertion callback, a
>+   *   factory or initial value and a testable operation have been provided.
>+   */
>+  template <typename LazyOperation = Operation,
>+            typename =
>+                EnableIfTestable<sizeof...(Invariants), Factory, LazyOperation>>
>+  testing::AssertionResult Test() const {
>+    return TestImpl(operation_, absl::index_sequence_for<Invariants...>());
>+  }
>+
>+ private:
>+  template <typename, typename, typename...>
>+  friend class ExceptionSafetyTester;
>+
>+  friend ExceptionSafetyTester<> testing::MakeExceptionSafetyTester();
>+
>+  ExceptionSafetyTester() {}
>+
>+  ExceptionSafetyTester(const Factory& f, const Operation& o,
>+                        const std::tuple<Invariants...>& i)
>+      : factory_(f), operation_(o), invariants_(i) {}
>+
>+  template <typename SelectedOperation, size_t... Indices>
>+  testing::AssertionResult TestImpl(const SelectedOperation& selected_operation,
>+                                    absl::index_sequence<Indices...>) const {
>+    // Starting from 0 and counting upwards until one of the exit conditions is
>+    // hit...
>+    for (int count = 0;; ++count) {
>+      exceptions_internal::ConstructorTracker ct(count);
>+
>+      // Run the full exception safety test algorithm for the current countdown
>+      auto reduced_res =
>+          TestAllInvariantsAtCountdown(factory_, selected_operation, count,
>+                                       std::get<Indices>(invariants_)...);
>+      // If there is no value in the optional, no invariants were run because no
>+      // exception was thrown. This means that the test is complete and the loop
>+      // can exit successfully.
>+      if (!reduced_res.has_value()) {
>+        return testing::AssertionSuccess();
>+      }
>+      // If the optional is not empty and the value is falsy, an invariant check
>+      // failed so the test must exit to propegate the failure.
>+      if (!reduced_res.value()) {
>+        return reduced_res.value();
>+      }
>+      // If the optional is not empty and the value is not falsy, it means
>+      // exceptions were thrown but the invariants passed so the test must
>+      // continue to run.
>+    }
>+  }
>+
>+  Factory factory_;
>+  Operation operation_;
>+  std::tuple<Invariants...> invariants_;
>+};
>+
>+}  // namespace exceptions_internal
>+
>+/*
>+ * Constructs an empty ExceptionSafetyTester. All ExceptionSafetyTester
>+ * objects are immutable and all With[thing] mutation methods return new
>+ * instances of ExceptionSafetyTester.
>+ *
>+ * In order to test a T for exception safety, a factory for that T, a testable
>+ * operation, and at least one invariant callback returning an assertion
>+ * result must be applied using the respective methods.
>+ */
>+inline exceptions_internal::ExceptionSafetyTester<>
>+MakeExceptionSafetyTester() {
>+  return {};
>+}
>+
>+}  // namespace testing
>+
>+#endif  // ABSL_BASE_INTERNAL_EXCEPTION_SAFETY_TESTING_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/exception_testing.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/exception_testing.h
>new file mode 100644
>index 00000000000..0cf7918e4c6
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/exception_testing.h
>@@ -0,0 +1,42 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+// Testing utilities for ABSL types which throw exceptions.
>+
>+#ifndef ABSL_BASE_INTERNAL_EXCEPTION_TESTING_H_
>+#define ABSL_BASE_INTERNAL_EXCEPTION_TESTING_H_
>+
>+#include "gtest/gtest.h"
>+#include "absl/base/config.h"
>+
>+// ABSL_BASE_INTERNAL_EXPECT_FAIL tests either for a specified thrown exception
>+// if exceptions are enabled, or for death with a specified text in the error
>+// message
>+#ifdef ABSL_HAVE_EXCEPTIONS
>+
>+#define ABSL_BASE_INTERNAL_EXPECT_FAIL(expr, exception_t, text) \
>+  EXPECT_THROW(expr, exception_t)
>+
>+#elif defined(__ANDROID__)
>+// Android asserts do not log anywhere that gtest can currently inspect.
>+// So we expect exit, but cannot match the message.
>+#define ABSL_BASE_INTERNAL_EXPECT_FAIL(expr, exception_t, text) \
>+  EXPECT_DEATH(expr, ".*")
>+#else
>+#define ABSL_BASE_INTERNAL_EXPECT_FAIL(expr, exception_t, text) \
>+  EXPECT_DEATH_IF_SUPPORTED(expr, text)
>+
>+#endif
>+
>+#endif  // ABSL_BASE_INTERNAL_EXCEPTION_TESTING_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/hide_ptr.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/hide_ptr.h
>new file mode 100644
>index 00000000000..45cf438912c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/hide_ptr.h
>@@ -0,0 +1,47 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#ifndef ABSL_BASE_INTERNAL_HIDE_PTR_H_
>+#define ABSL_BASE_INTERNAL_HIDE_PTR_H_
>+
>+#include <cstdint>
>+
>+namespace absl {
>+namespace base_internal {
>+
>+// Arbitrary value with high bits set. Xor'ing with it is unlikely
>+// to map one valid pointer to another valid pointer.
>+constexpr uintptr_t HideMask() {
>+  return (uintptr_t{0xF03A5F7BU} << (sizeof(uintptr_t) - 4) * 8) | 0xF03A5F7BU;
>+}
>+
>+// Hide a pointer from the leak checker. For internal use only.
>+// Differs from absl::IgnoreLeak(ptr) in that absl::IgnoreLeak(ptr) causes ptr
>+// and all objects reachable from ptr to be ignored by the leak checker.
>+template <class T>
>+inline uintptr_t HidePtr(T* ptr) {
>+  return reinterpret_cast<uintptr_t>(ptr) ^ HideMask();
>+}
>+
>+// Return a pointer that has been hidden from the leak checker.
>+// For internal use only.
>+template <class T>
>+inline T* UnhidePtr(uintptr_t hidden) {
>+  return reinterpret_cast<T*>(hidden ^ HideMask());
>+}
>+
>+}  // namespace base_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_BASE_INTERNAL_HIDE_PTR_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/identity.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/identity.h
>new file mode 100644
>index 00000000000..a1a5d70a84d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/identity.h
>@@ -0,0 +1,33 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+
>+#ifndef ABSL_BASE_INTERNAL_IDENTITY_H_
>+#define ABSL_BASE_INTERNAL_IDENTITY_H_
>+
>+namespace absl {
>+namespace internal {
>+
>+template <typename T>
>+struct identity {
>+  typedef T type;
>+};
>+
>+template <typename T>
>+using identity_t = typename identity<T>::type;
>+
>+}  // namespace internal
>+}  // namespace absl
>+
>+#endif  // ABSL_BASE_INTERNAL_IDENTITY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/inline_variable.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/inline_variable.h
>new file mode 100644
>index 00000000000..f7bb8c56525
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/inline_variable.h
>@@ -0,0 +1,107 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#ifndef ABSL_BASE_INTERNAL_INLINE_VARIABLE_EMULATION_H_
>+#define ABSL_BASE_INTERNAL_INLINE_VARIABLE_EMULATION_H_
>+
>+#include <type_traits>
>+
>+#include "absl/base/internal/identity.h"
>+
>+// File:
>+//   This file define a macro that allows the creation of or emulation of C++17
>+//   inline variables based on whether or not the feature is supported.
>+
>+////////////////////////////////////////////////////////////////////////////////
>+// Macro: ABSL_INTERNAL_INLINE_CONSTEXPR(type, name, init)
>+//
>+// Description:
>+//   Expands to the equivalent of an inline constexpr instance of the specified
>+//   `type` and `name`, initialized to the value `init`. If the compiler being
>+//   used is detected as supporting actual inline variables as a language
>+//   feature, then the macro expands to an actual inline variable definition.
>+//
>+// Requires:
>+//   `type` is a type that is usable in an extern variable declaration.
>+//
>+// Requires: `name` is a valid identifier
>+//
>+// Requires:
>+//   `init` is an expression that can be used in the following definition:
>+//     constexpr type name = init;
>+//
>+// Usage:
>+//
>+//   // Equivalent to: `inline constexpr size_t variant_npos = -1;`
>+//   ABSL_INTERNAL_INLINE_CONSTEXPR(size_t, variant_npos, -1);
>+//
>+// Differences in implementation:
>+//   For a direct, language-level inline variable, decltype(name) will be the
>+//   type that was specified along with const qualification, whereas for
>+//   emulated inline variables, decltype(name) may be different (in practice
>+//   it will likely be a reference type).
>+////////////////////////////////////////////////////////////////////////////////
>+
>+#ifdef __cpp_inline_variables
>+
>+// Clang's -Wmissing-variable-declarations option erroneously warned that
>+// inline constexpr objects need to be pre-declared. This has now been fixed,
>+// but we will need to support this workaround for people building with older
>+// versions of clang.
>+//
>+// Bug: https://bugs.llvm.org/show_bug.cgi?id=35862
>+//
>+// Note:
>+//   identity_t is used here so that the const and name are in the
>+//   appropriate place for pointer types, reference types, function pointer
>+//   types, etc..
>+#if defined(__clang__)
>+#define ABSL_INTERNAL_EXTERN_DECL(type, name) \
>+  extern const ::absl::internal::identity_t<type> name;
>+#else  // Otherwise, just define the macro to do nothing.
>+#define ABSL_INTERNAL_EXTERN_DECL(type, name)
>+#endif  // defined(__clang__)
>+
>+// See above comment at top of file for details.
>+#define ABSL_INTERNAL_INLINE_CONSTEXPR(type, name, init) \
>+  ABSL_INTERNAL_EXTERN_DECL(type, name)                  \
>+  inline constexpr ::absl::internal::identity_t<type> name = init
>+
>+#else
>+
>+// See above comment at top of file for details.
>+//
>+// Note:
>+//   identity_t is used here so that the const and name are in the
>+//   appropriate place for pointer types, reference types, function pointer
>+//   types, etc..
>+#define ABSL_INTERNAL_INLINE_CONSTEXPR(var_type, name, init)                  \
>+  template <class /*AbslInternalDummy*/ = void>                               \
>+  struct AbslInternalInlineVariableHolder##name {                             \
>+    static constexpr ::absl::internal::identity_t<var_type> kInstance = init; \
>+  };                                                                          \
>+                                                                              \
>+  template <class AbslInternalDummy>                                          \
>+  constexpr ::absl::internal::identity_t<var_type>                            \
>+      AbslInternalInlineVariableHolder##name<AbslInternalDummy>::kInstance;   \
>+                                                                              \
>+  static constexpr const ::absl::internal::identity_t<var_type>&              \
>+      name = /* NOLINT */                                                     \
>+      AbslInternalInlineVariableHolder##name<>::kInstance;                    \
>+  static_assert(sizeof(void (*)(decltype(name))) != 0,                        \
>+                "Silence unused variable warnings.")
>+
>+#endif  // __cpp_inline_variables
>+
>+#endif  // ABSL_BASE_INTERNAL_INLINE_VARIABLE_EMULATION_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/inline_variable_testing.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/inline_variable_testing.h
>new file mode 100644
>index 00000000000..a0dd2bb28a4
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/inline_variable_testing.h
>@@ -0,0 +1,44 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#ifndef ABSL_BASE_INLINE_VARIABLE_TESTING_H_
>+#define ABSL_BASE_INLINE_VARIABLE_TESTING_H_
>+
>+#include "absl/base/internal/inline_variable.h"
>+
>+namespace absl {
>+namespace inline_variable_testing_internal {
>+
>+struct Foo {
>+  int value = 5;
>+};
>+
>+ABSL_INTERNAL_INLINE_CONSTEXPR(Foo, inline_variable_foo, {});
>+ABSL_INTERNAL_INLINE_CONSTEXPR(Foo, other_inline_variable_foo, {});
>+
>+ABSL_INTERNAL_INLINE_CONSTEXPR(int, inline_variable_int, 5);
>+ABSL_INTERNAL_INLINE_CONSTEXPR(int, other_inline_variable_int, 5);
>+
>+ABSL_INTERNAL_INLINE_CONSTEXPR(void(*)(), inline_variable_fun_ptr, nullptr);
>+
>+const Foo& get_foo_a();
>+const Foo& get_foo_b();
>+
>+const int& get_int_a();
>+const int& get_int_b();
>+
>+}  // namespace inline_variable_testing_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_BASE_INLINE_VARIABLE_TESTING_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/invoke.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/invoke.h
>new file mode 100644
>index 00000000000..8c3f4f60637
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/invoke.h
>@@ -0,0 +1,188 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// absl::base_internal::Invoke(f, args...) is an implementation of
>+// INVOKE(f, args...) from section [func.require] of the C++ standard.
>+//
>+// [func.require]
>+// Define INVOKE (f, t1, t2, ..., tN) as follows:
>+// 1. (t1.*f)(t2, ..., tN) when f is a pointer to a member function of a class T
>+//    and t1 is an object of type T or a reference to an object of type T or a
>+//    reference to an object of a type derived from T;
>+// 2. ((*t1).*f)(t2, ..., tN) when f is a pointer to a member function of a
>+//    class T and t1 is not one of the types described in the previous item;
>+// 3. t1.*f when N == 1 and f is a pointer to member data of a class T and t1 is
>+//    an object of type T or a reference to an object of type T or a reference
>+//    to an object of a type derived from T;
>+// 4. (*t1).*f when N == 1 and f is a pointer to member data of a class T and t1
>+//    is not one of the types described in the previous item;
>+// 5. f(t1, t2, ..., tN) in all other cases.
>+//
>+// The implementation is SFINAE-friendly: substitution failure within Invoke()
>+// isn't an error.
>+
>+#ifndef ABSL_BASE_INTERNAL_INVOKE_H_
>+#define ABSL_BASE_INTERNAL_INVOKE_H_
>+
>+#include <algorithm>
>+#include <type_traits>
>+#include <utility>
>+
>+// The following code is internal implementation detail.  See the comment at the
>+// top of this file for the API documentation.
>+
>+namespace absl {
>+namespace base_internal {
>+
>+// The five classes below each implement one of the clauses from the definition
>+// of INVOKE. The inner class template Accept<F, Args...> checks whether the
>+// clause is applicable; static function template Invoke(f, args...) does the
>+// invocation.
>+//
>+// By separating the clause selection logic from invocation we make sure that
>+// Invoke() does exactly what the standard says.
>+
>+template <typename Derived>
>+struct StrippedAccept {
>+  template <typename... Args>
>+  struct Accept : Derived::template AcceptImpl<typename std::remove_cv<
>+                      typename std::remove_reference<Args>::type>::type...> {};
>+};
>+
>+// (t1.*f)(t2, ..., tN) when f is a pointer to a member function of a class T
>+// and t1 is an object of type T or a reference to an object of type T or a
>+// reference to an object of a type derived from T.
>+struct MemFunAndRef : StrippedAccept<MemFunAndRef> {
>+  template <typename... Args>
>+  struct AcceptImpl : std::false_type {};
>+
>+  template <typename R, typename C, typename... Params, typename Obj,
>+            typename... Args>
>+  struct AcceptImpl<R (C::*)(Params...), Obj, Args...>
>+      : std::is_base_of<C, Obj> {};
>+
>+  template <typename R, typename C, typename... Params, typename Obj,
>+            typename... Args>
>+  struct AcceptImpl<R (C::*)(Params...) const, Obj, Args...>
>+      : std::is_base_of<C, Obj> {};
>+
>+  template <typename MemFun, typename Obj, typename... Args>
>+  static decltype((std::declval<Obj>().*
>+                   std::declval<MemFun>())(std::declval<Args>()...))
>+  Invoke(MemFun&& mem_fun, Obj&& obj, Args&&... args) {
>+    return (std::forward<Obj>(obj).*
>+            std::forward<MemFun>(mem_fun))(std::forward<Args>(args)...);
>+  }
>+};
>+
>+// ((*t1).*f)(t2, ..., tN) when f is a pointer to a member function of a
>+// class T and t1 is not one of the types described in the previous item.
>+struct MemFunAndPtr : StrippedAccept<MemFunAndPtr> {
>+  template <typename... Args>
>+  struct AcceptImpl : std::false_type {};
>+
>+  template <typename R, typename C, typename... Params, typename Ptr,
>+            typename... Args>
>+  struct AcceptImpl<R (C::*)(Params...), Ptr, Args...>
>+      : std::integral_constant<bool, !std::is_base_of<C, Ptr>::value> {};
>+
>+  template <typename R, typename C, typename... Params, typename Ptr,
>+            typename... Args>
>+  struct AcceptImpl<R (C::*)(Params...) const, Ptr, Args...>
>+      : std::integral_constant<bool, !std::is_base_of<C, Ptr>::value> {};
>+
>+  template <typename MemFun, typename Ptr, typename... Args>
>+  static decltype(((*std::declval<Ptr>()).*
>+                   std::declval<MemFun>())(std::declval<Args>()...))
>+  Invoke(MemFun&& mem_fun, Ptr&& ptr, Args&&... args) {
>+    return ((*std::forward<Ptr>(ptr)).*
>+            std::forward<MemFun>(mem_fun))(std::forward<Args>(args)...);
>+  }
>+};
>+
>+// t1.*f when N == 1 and f is a pointer to member data of a class T and t1 is
>+// an object of type T or a reference to an object of type T or a reference
>+// to an object of a type derived from T.
>+struct DataMemAndRef : StrippedAccept<DataMemAndRef> {
>+  template <typename... Args>
>+  struct AcceptImpl : std::false_type {};
>+
>+  template <typename R, typename C, typename Obj>
>+  struct AcceptImpl<R C::*, Obj> : std::is_base_of<C, Obj> {};
>+
>+  template <typename DataMem, typename Ref>
>+  static decltype(std::declval<Ref>().*std::declval<DataMem>()) Invoke(
>+      DataMem&& data_mem, Ref&& ref) {
>+    return std::forward<Ref>(ref).*std::forward<DataMem>(data_mem);
>+  }
>+};
>+
>+// (*t1).*f when N == 1 and f is a pointer to member data of a class T and t1
>+// is not one of the types described in the previous item.
>+struct DataMemAndPtr : StrippedAccept<DataMemAndPtr> {
>+  template <typename... Args>
>+  struct AcceptImpl : std::false_type {};
>+
>+  template <typename R, typename C, typename Ptr>
>+  struct AcceptImpl<R C::*, Ptr>
>+      : std::integral_constant<bool, !std::is_base_of<C, Ptr>::value> {};
>+
>+  template <typename DataMem, typename Ptr>
>+  static decltype((*std::declval<Ptr>()).*std::declval<DataMem>()) Invoke(
>+      DataMem&& data_mem, Ptr&& ptr) {
>+    return (*std::forward<Ptr>(ptr)).*std::forward<DataMem>(data_mem);
>+  }
>+};
>+
>+// f(t1, t2, ..., tN) in all other cases.
>+struct Callable {
>+  // Callable doesn't have Accept because it's the last clause that gets picked
>+  // when none of the previous clauses are applicable.
>+  template <typename F, typename... Args>
>+  static decltype(std::declval<F>()(std::declval<Args>()...)) Invoke(
>+      F&& f, Args&&... args) {
>+    return std::forward<F>(f)(std::forward<Args>(args)...);
>+  }
>+};
>+
>+// Resolves to the first matching clause.
>+template <typename... Args>
>+struct Invoker {
>+  typedef typename std::conditional<
>+      MemFunAndRef::Accept<Args...>::value, MemFunAndRef,
>+      typename std::conditional<
>+          MemFunAndPtr::Accept<Args...>::value, MemFunAndPtr,
>+          typename std::conditional<
>+              DataMemAndRef::Accept<Args...>::value, DataMemAndRef,
>+              typename std::conditional<DataMemAndPtr::Accept<Args...>::value,
>+                                        DataMemAndPtr, Callable>::type>::type>::
>+          type>::type type;
>+};
>+
>+// The result type of Invoke<F, Args...>.
>+template <typename F, typename... Args>
>+using InvokeT = decltype(Invoker<F, Args...>::type::Invoke(
>+    std::declval<F>(), std::declval<Args>()...));
>+
>+// Invoke(f, args...) is an implementation of INVOKE(f, args...) from section
>+// [func.require] of the C++ standard.
>+template <typename F, typename... Args>
>+InvokeT<F, Args...> Invoke(F&& f, Args&&... args) {
>+  return Invoker<F, Args...>::type::Invoke(std::forward<F>(f),
>+                                           std::forward<Args>(args)...);
>+}
>+}  // namespace base_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_BASE_INTERNAL_INVOKE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/low_level_alloc.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/low_level_alloc.cc
>new file mode 100644
>index 00000000000..0626cd5478d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/low_level_alloc.cc
>@@ -0,0 +1,604 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+// A low-level allocator that can be used by other low-level
>+// modules without introducing dependency cycles.
>+// This allocator is slow and wasteful of memory;
>+// it should not be used when performance is key.
>+
>+#include "absl/base/internal/low_level_alloc.h"
>+
>+#include <type_traits>
>+
>+#include "absl/base/call_once.h"
>+#include "absl/base/config.h"
>+#include "absl/base/internal/direct_mmap.h"
>+#include "absl/base/internal/scheduling_mode.h"
>+#include "absl/base/macros.h"
>+#include "absl/base/thread_annotations.h"
>+
>+// LowLevelAlloc requires that the platform support low-level
>+// allocation of virtual memory. Platforms lacking this cannot use
>+// LowLevelAlloc.
>+#ifndef ABSL_LOW_LEVEL_ALLOC_MISSING
>+
>+#ifndef _WIN32
>+#include <pthread.h>
>+#include <signal.h>
>+#include <sys/mman.h>
>+#include <unistd.h>
>+#else
>+#include <windows.h>
>+#endif
>+
>+#include <string.h>
>+#include <algorithm>
>+#include <atomic>
>+#include <cerrno>
>+#include <cstddef>
>+#include <new>                   // for placement-new
>+
>+#include "absl/base/dynamic_annotations.h"
>+#include "absl/base/internal/raw_logging.h"
>+#include "absl/base/internal/spinlock.h"
>+
>+// MAP_ANONYMOUS
>+#if defined(__APPLE__)
>+// For mmap, Linux defines both MAP_ANONYMOUS and MAP_ANON and says MAP_ANON is
>+// deprecated. In Darwin, MAP_ANON is all there is.
>+#if !defined MAP_ANONYMOUS
>+#define MAP_ANONYMOUS MAP_ANON
>+#endif  // !MAP_ANONYMOUS
>+#endif  // __APPLE__
>+
>+namespace absl {
>+namespace base_internal {
>+
>+// A first-fit allocator with amortized logarithmic free() time.
>+
>+// ---------------------------------------------------------------------------
>+static const int kMaxLevel = 30;
>+
>+namespace {
>+// This struct describes one allocated block, or one free block.
>+struct AllocList {
>+  struct Header {
>+    // Size of entire region, including this field. Must be
>+    // first. Valid in both allocated and unallocated blocks.
>+    uintptr_t size;
>+
>+    // kMagicAllocated or kMagicUnallocated xor this.
>+    uintptr_t magic;
>+
>+    // Pointer to parent arena.
>+    LowLevelAlloc::Arena *arena;
>+
>+    // Aligns regions to 0 mod 2*sizeof(void*).
>+    void *dummy_for_alignment;
>+  } header;
>+
>+  // Next two fields: in unallocated blocks: freelist skiplist data
>+  //                  in allocated blocks: overlaps with client data
>+
>+  // Levels in skiplist used.
>+  int levels;
>+
>+  // Actually has levels elements. The AllocList node may not have room
>+  // for all kMaxLevel entries. See max_fit in LLA_SkiplistLevels().
>+  AllocList *next[kMaxLevel];
>+};
>+}  // namespace
>+
>+// ---------------------------------------------------------------------------
>+// A trivial skiplist implementation.  This is used to keep the freelist
>+// in address order while taking only logarithmic time per insert and delete.
>+
>+// An integer approximation of log2(size/base)
>+// Requires size >= base.
>+static int IntLog2(size_t size, size_t base) {
>+  int result = 0;
>+  for (size_t i = size; i > base; i >>= 1) {  // i == floor(size/2**result)
>+    result++;
>+  }
>+  //    floor(size / 2**result) <= base < floor(size / 2**(result-1))
>+  // =>     log2(size/(base+1)) <= result < 1+log2(size/base)
>+  // => result ~= log2(size/base)
>+  return result;
>+}
>+
>+// Return a random integer n:  p(n)=1/(2**n) if 1 <= n; p(n)=0 if n < 1.
>+static int Random(uint32_t *state) {
>+  uint32_t r = *state;
>+  int result = 1;
>+  while ((((r = r*1103515245 + 12345) >> 30) & 1) == 0) {
>+    result++;
>+  }
>+  *state = r;
>+  return result;
>+}
>+
>+// Return a number of skiplist levels for a node of size bytes, where
>+// base is the minimum node size.  Compute level=log2(size / base)+n
>+// where n is 1 if random is false and otherwise a random number generated with
>+// the standard distribution for a skiplist:  See Random() above.
>+// Bigger nodes tend to have more skiplist levels due to the log2(size / base)
>+// term, so first-fit searches touch fewer nodes.  "level" is clipped so
>+// level<kMaxLevel and next[level-1] will fit in the node.
>+// 0 < LLA_SkiplistLevels(x,y,false) <= LLA_SkiplistLevels(x,y,true) < kMaxLevel
>+static int LLA_SkiplistLevels(size_t size, size_t base, uint32_t *random) {
>+  // max_fit is the maximum number of levels that will fit in a node for the
>+  // given size.   We can't return more than max_fit, no matter what the
>+  // random number generator says.
>+  size_t max_fit = (size - offsetof(AllocList, next)) / sizeof(AllocList *);
>+  int level = IntLog2(size, base) + (random != nullptr ? Random(random) : 1);
>+  if (static_cast<size_t>(level) > max_fit) level = static_cast<int>(max_fit);
>+  if (level > kMaxLevel-1) level = kMaxLevel - 1;
>+  ABSL_RAW_CHECK(level >= 1, "block not big enough for even one level");
>+  return level;
>+}
>+
>+// Return "atleast", the first element of AllocList *head s.t. *atleast >= *e.
>+// For 0 <= i < head->levels, set prev[i] to "no_greater", where no_greater
>+// points to the last element at level i in the AllocList less than *e, or is
>+// head if no such element exists.
>+static AllocList *LLA_SkiplistSearch(AllocList *head,
>+                                     AllocList *e, AllocList **prev) {
>+  AllocList *p = head;
>+  for (int level = head->levels - 1; level >= 0; level--) {
>+    for (AllocList *n; (n = p->next[level]) != nullptr && n < e; p = n) {
>+    }
>+    prev[level] = p;
>+  }
>+  return (head->levels == 0) ? nullptr : prev[0]->next[0];
>+}
>+
>+// Insert element *e into AllocList *head.  Set prev[] as LLA_SkiplistSearch.
>+// Requires that e->levels be previously set by the caller (using
>+// LLA_SkiplistLevels())
>+static void LLA_SkiplistInsert(AllocList *head, AllocList *e,
>+                               AllocList **prev) {
>+  LLA_SkiplistSearch(head, e, prev);
>+  for (; head->levels < e->levels; head->levels++) {  // extend prev pointers
>+    prev[head->levels] = head;                        // to all *e's levels
>+  }
>+  for (int i = 0; i != e->levels; i++) {  // add element to list
>+    e->next[i] = prev[i]->next[i];
>+    prev[i]->next[i] = e;
>+  }
>+}
>+
>+// Remove element *e from AllocList *head.  Set prev[] as LLA_SkiplistSearch().
>+// Requires that e->levels be previous set by the caller (using
>+// LLA_SkiplistLevels())
>+static void LLA_SkiplistDelete(AllocList *head, AllocList *e,
>+                               AllocList **prev) {
>+  AllocList *found = LLA_SkiplistSearch(head, e, prev);
>+  ABSL_RAW_CHECK(e == found, "element not in freelist");
>+  for (int i = 0; i != e->levels && prev[i]->next[i] == e; i++) {
>+    prev[i]->next[i] = e->next[i];
>+  }
>+  while (head->levels > 0 && head->next[head->levels - 1] == nullptr) {
>+    head->levels--;   // reduce head->levels if level unused
>+  }
>+}
>+
>+// ---------------------------------------------------------------------------
>+// Arena implementation
>+
>+// Metadata for an LowLevelAlloc arena instance.
>+struct LowLevelAlloc::Arena {
>+  // Constructs an arena with the given LowLevelAlloc flags.
>+  explicit Arena(uint32_t flags_value);
>+
>+  base_internal::SpinLock mu;
>+  // Head of free list, sorted by address
>+  AllocList freelist GUARDED_BY(mu);
>+  // Count of allocated blocks
>+  int32_t allocation_count GUARDED_BY(mu);
>+  // flags passed to NewArena
>+  const uint32_t flags;
>+  // Result of getpagesize()
>+  const size_t pagesize;
>+  // Lowest power of two >= max(16, sizeof(AllocList))
>+  const size_t roundup;
>+  // Smallest allocation block size
>+  const size_t min_size;
>+  // PRNG state
>+  uint32_t random GUARDED_BY(mu);
>+};
>+
>+namespace {
>+using ArenaStorage = std::aligned_storage<sizeof(LowLevelAlloc::Arena),
>+                                          alignof(LowLevelAlloc::Arena)>::type;
>+
>+// Static storage space for the lazily-constructed, default global arena
>+// instances.  We require this space because the whole point of LowLevelAlloc
>+// is to avoid relying on malloc/new.
>+ArenaStorage default_arena_storage;
>+ArenaStorage unhooked_arena_storage;
>+#ifndef ABSL_LOW_LEVEL_ALLOC_ASYNC_SIGNAL_SAFE_MISSING
>+ArenaStorage unhooked_async_sig_safe_arena_storage;
>+#endif
>+
>+// We must use LowLevelCallOnce here to construct the global arenas, rather than
>+// using function-level statics, to avoid recursively invoking the scheduler.
>+absl::once_flag create_globals_once;
>+
>+void CreateGlobalArenas() {
>+  new (&default_arena_storage)
>+      LowLevelAlloc::Arena(LowLevelAlloc::kCallMallocHook);
>+  new (&unhooked_arena_storage) LowLevelAlloc::Arena(0);
>+#ifndef ABSL_LOW_LEVEL_ALLOC_ASYNC_SIGNAL_SAFE_MISSING
>+  new (&unhooked_async_sig_safe_arena_storage)
>+      LowLevelAlloc::Arena(LowLevelAlloc::kAsyncSignalSafe);
>+#endif
>+}
>+
>+// Returns a global arena that does not call into hooks.  Used by NewArena()
>+// when kCallMallocHook is not set.
>+LowLevelAlloc::Arena* UnhookedArena() {
>+  base_internal::LowLevelCallOnce(&create_globals_once, CreateGlobalArenas);
>+  return reinterpret_cast<LowLevelAlloc::Arena*>(&unhooked_arena_storage);
>+}
>+
>+#ifndef ABSL_LOW_LEVEL_ALLOC_ASYNC_SIGNAL_SAFE_MISSING
>+// Returns a global arena that is async-signal safe.  Used by NewArena() when
>+// kAsyncSignalSafe is set.
>+LowLevelAlloc::Arena *UnhookedAsyncSigSafeArena() {
>+  base_internal::LowLevelCallOnce(&create_globals_once, CreateGlobalArenas);
>+  return reinterpret_cast<LowLevelAlloc::Arena *>(
>+      &unhooked_async_sig_safe_arena_storage);
>+}
>+#endif
>+
>+}  // namespace
>+
>+// Returns the default arena, as used by LowLevelAlloc::Alloc() and friends.
>+LowLevelAlloc::Arena *LowLevelAlloc::DefaultArena() {
>+  base_internal::LowLevelCallOnce(&create_globals_once, CreateGlobalArenas);
>+  return reinterpret_cast<LowLevelAlloc::Arena*>(&default_arena_storage);
>+}
>+
>+// magic numbers to identify allocated and unallocated blocks
>+static const uintptr_t kMagicAllocated = 0x4c833e95U;
>+static const uintptr_t kMagicUnallocated = ~kMagicAllocated;
>+
>+namespace {
>+class SCOPED_LOCKABLE ArenaLock {
>+ public:
>+  explicit ArenaLock(LowLevelAlloc::Arena *arena)
>+      EXCLUSIVE_LOCK_FUNCTION(arena->mu)
>+      : arena_(arena) {
>+#ifndef ABSL_LOW_LEVEL_ALLOC_ASYNC_SIGNAL_SAFE_MISSING
>+    if ((arena->flags & LowLevelAlloc::kAsyncSignalSafe) != 0) {
>+      sigset_t all;
>+      sigfillset(&all);
>+      mask_valid_ = pthread_sigmask(SIG_BLOCK, &all, &mask_) == 0;
>+    }
>+#endif
>+    arena_->mu.Lock();
>+  }
>+  ~ArenaLock() { ABSL_RAW_CHECK(left_, "haven't left Arena region"); }
>+  void Leave() UNLOCK_FUNCTION() {
>+    arena_->mu.Unlock();
>+#ifndef ABSL_LOW_LEVEL_ALLOC_ASYNC_SIGNAL_SAFE_MISSING
>+    if (mask_valid_) {
>+      pthread_sigmask(SIG_SETMASK, &mask_, nullptr);
>+    }
>+#endif
>+    left_ = true;
>+  }
>+
>+ private:
>+  bool left_ = false;  // whether left region
>+#ifndef ABSL_LOW_LEVEL_ALLOC_ASYNC_SIGNAL_SAFE_MISSING
>+  bool mask_valid_ = false;
>+  sigset_t mask_;  // old mask of blocked signals
>+#endif
>+  LowLevelAlloc::Arena *arena_;
>+  ArenaLock(const ArenaLock &) = delete;
>+  ArenaLock &operator=(const ArenaLock &) = delete;
>+};
>+}  // namespace
>+
>+// create an appropriate magic number for an object at "ptr"
>+// "magic" should be kMagicAllocated or kMagicUnallocated
>+inline static uintptr_t Magic(uintptr_t magic, AllocList::Header *ptr) {
>+  return magic ^ reinterpret_cast<uintptr_t>(ptr);
>+}
>+
>+namespace {
>+size_t GetPageSize() {
>+#ifdef _WIN32
>+  SYSTEM_INFO system_info;
>+  GetSystemInfo(&system_info);
>+  return std::max(system_info.dwPageSize, system_info.dwAllocationGranularity);
>+#else
>+  return getpagesize();
>+#endif
>+}
>+
>+size_t RoundedUpBlockSize() {
>+  // Round up block sizes to a power of two close to the header size.
>+  size_t roundup = 16;
>+  while (roundup < sizeof(AllocList::Header)) {
>+    roundup += roundup;
>+  }
>+  return roundup;
>+}
>+
>+}  // namespace
>+
>+LowLevelAlloc::Arena::Arena(uint32_t flags_value)
>+    : mu(base_internal::SCHEDULE_KERNEL_ONLY),
>+      allocation_count(0),
>+      flags(flags_value),
>+      pagesize(GetPageSize()),
>+      roundup(RoundedUpBlockSize()),
>+      min_size(2 * roundup),
>+      random(0) {
>+  freelist.header.size = 0;
>+  freelist.header.magic =
>+      Magic(kMagicUnallocated, &freelist.header);
>+  freelist.header.arena = this;
>+  freelist.levels = 0;
>+  memset(freelist.next, 0, sizeof(freelist.next));
>+}
>+
>+// L < meta_data_arena->mu
>+LowLevelAlloc::Arena *LowLevelAlloc::NewArena(int32_t flags) {
>+  Arena *meta_data_arena = DefaultArena();
>+#ifndef ABSL_LOW_LEVEL_ALLOC_ASYNC_SIGNAL_SAFE_MISSING
>+  if ((flags & LowLevelAlloc::kAsyncSignalSafe) != 0) {
>+    meta_data_arena = UnhookedAsyncSigSafeArena();
>+  } else  // NOLINT(readability/braces)
>+#endif
>+      if ((flags & LowLevelAlloc::kCallMallocHook) == 0) {
>+    meta_data_arena = UnhookedArena();
>+  }
>+  Arena *result =
>+    new (AllocWithArena(sizeof (*result), meta_data_arena)) Arena(flags);
>+  return result;
>+}
>+
>+// L < arena->mu, L < arena->arena->mu
>+bool LowLevelAlloc::DeleteArena(Arena *arena) {
>+  ABSL_RAW_CHECK(
>+      arena != nullptr && arena != DefaultArena() && arena != UnhookedArena(),
>+      "may not delete default arena");
>+  ArenaLock section(arena);
>+  if (arena->allocation_count != 0) {
>+    section.Leave();
>+    return false;
>+  }
>+  while (arena->freelist.next[0] != nullptr) {
>+    AllocList *region = arena->freelist.next[0];
>+    size_t size = region->header.size;
>+    arena->freelist.next[0] = region->next[0];
>+    ABSL_RAW_CHECK(
>+        region->header.magic == Magic(kMagicUnallocated, &region->header),
>+        "bad magic number in DeleteArena()");
>+    ABSL_RAW_CHECK(region->header.arena == arena,
>+                   "bad arena pointer in DeleteArena()");
>+    ABSL_RAW_CHECK(size % arena->pagesize == 0,
>+                   "empty arena has non-page-aligned block size");
>+    ABSL_RAW_CHECK(reinterpret_cast<uintptr_t>(region) % arena->pagesize == 0,
>+                   "empty arena has non-page-aligned block");
>+    int munmap_result;
>+#ifdef _WIN32
>+    munmap_result = VirtualFree(region, 0, MEM_RELEASE);
>+    ABSL_RAW_CHECK(munmap_result != 0,
>+                   "LowLevelAlloc::DeleteArena: VitualFree failed");
>+#else
>+    if ((arena->flags & LowLevelAlloc::kAsyncSignalSafe) == 0) {
>+      munmap_result = munmap(region, size);
>+    } else {
>+      munmap_result = base_internal::DirectMunmap(region, size);
>+    }
>+    if (munmap_result != 0) {
>+      ABSL_RAW_LOG(FATAL, "LowLevelAlloc::DeleteArena: munmap failed: %d",
>+                   errno);
>+    }
>+#endif
>+  }
>+  section.Leave();
>+  arena->~Arena();
>+  Free(arena);
>+  return true;
>+}
>+
>+// ---------------------------------------------------------------------------
>+
>+// Addition, checking for overflow.  The intent is to die if an external client
>+// manages to push through a request that would cause arithmetic to fail.
>+static inline uintptr_t CheckedAdd(uintptr_t a, uintptr_t b) {
>+  uintptr_t sum = a + b;
>+  ABSL_RAW_CHECK(sum >= a, "LowLevelAlloc arithmetic overflow");
>+  return sum;
>+}
>+
>+// Return value rounded up to next multiple of align.
>+// align must be a power of two.
>+static inline uintptr_t RoundUp(uintptr_t addr, uintptr_t align) {
>+  return CheckedAdd(addr, align - 1) & ~(align - 1);
>+}
>+
>+// Equivalent to "return prev->next[i]" but with sanity checking
>+// that the freelist is in the correct order, that it
>+// consists of regions marked "unallocated", and that no two regions
>+// are adjacent in memory (they should have been coalesced).
>+// L < arena->mu
>+static AllocList *Next(int i, AllocList *prev, LowLevelAlloc::Arena *arena) {
>+  ABSL_RAW_CHECK(i < prev->levels, "too few levels in Next()");
>+  AllocList *next = prev->next[i];
>+  if (next != nullptr) {
>+    ABSL_RAW_CHECK(
>+        next->header.magic == Magic(kMagicUnallocated, &next->header),
>+        "bad magic number in Next()");
>+    ABSL_RAW_CHECK(next->header.arena == arena, "bad arena pointer in Next()");
>+    if (prev != &arena->freelist) {
>+      ABSL_RAW_CHECK(prev < next, "unordered freelist");
>+      ABSL_RAW_CHECK(reinterpret_cast<char *>(prev) + prev->header.size <
>+                         reinterpret_cast<char *>(next),
>+                     "malformed freelist");
>+    }
>+  }
>+  return next;
>+}
>+
>+// Coalesce list item "a" with its successor if they are adjacent.
>+static void Coalesce(AllocList *a) {
>+  AllocList *n = a->next[0];
>+  if (n != nullptr && reinterpret_cast<char *>(a) + a->header.size ==
>+                          reinterpret_cast<char *>(n)) {
>+    LowLevelAlloc::Arena *arena = a->header.arena;
>+    a->header.size += n->header.size;
>+    n->header.magic = 0;
>+    n->header.arena = nullptr;
>+    AllocList *prev[kMaxLevel];
>+    LLA_SkiplistDelete(&arena->freelist, n, prev);
>+    LLA_SkiplistDelete(&arena->freelist, a, prev);
>+    a->levels = LLA_SkiplistLevels(a->header.size, arena->min_size,
>+                                   &arena->random);
>+    LLA_SkiplistInsert(&arena->freelist, a, prev);
>+  }
>+}
>+
>+// Adds block at location "v" to the free list
>+// L >= arena->mu
>+static void AddToFreelist(void *v, LowLevelAlloc::Arena *arena) {
>+  AllocList *f = reinterpret_cast<AllocList *>(
>+                        reinterpret_cast<char *>(v) - sizeof (f->header));
>+  ABSL_RAW_CHECK(f->header.magic == Magic(kMagicAllocated, &f->header),
>+                 "bad magic number in AddToFreelist()");
>+  ABSL_RAW_CHECK(f->header.arena == arena,
>+                 "bad arena pointer in AddToFreelist()");
>+  f->levels = LLA_SkiplistLevels(f->header.size, arena->min_size,
>+                                 &arena->random);
>+  AllocList *prev[kMaxLevel];
>+  LLA_SkiplistInsert(&arena->freelist, f, prev);
>+  f->header.magic = Magic(kMagicUnallocated, &f->header);
>+  Coalesce(f);                  // maybe coalesce with successor
>+  Coalesce(prev[0]);            // maybe coalesce with predecessor
>+}
>+
>+// Frees storage allocated by LowLevelAlloc::Alloc().
>+// L < arena->mu
>+void LowLevelAlloc::Free(void *v) {
>+  if (v != nullptr) {
>+    AllocList *f = reinterpret_cast<AllocList *>(
>+                        reinterpret_cast<char *>(v) - sizeof (f->header));
>+    ABSL_RAW_CHECK(f->header.magic == Magic(kMagicAllocated, &f->header),
>+                   "bad magic number in Free()");
>+    LowLevelAlloc::Arena *arena = f->header.arena;
>+    ArenaLock section(arena);
>+    AddToFreelist(v, arena);
>+    ABSL_RAW_CHECK(arena->allocation_count > 0, "nothing in arena to free");
>+    arena->allocation_count--;
>+    section.Leave();
>+  }
>+}
>+
>+// allocates and returns a block of size bytes, to be freed with Free()
>+// L < arena->mu
>+static void *DoAllocWithArena(size_t request, LowLevelAlloc::Arena *arena) {
>+  void *result = nullptr;
>+  if (request != 0) {
>+    AllocList *s;       // will point to region that satisfies request
>+    ArenaLock section(arena);
>+    // round up with header
>+    size_t req_rnd = RoundUp(CheckedAdd(request, sizeof (s->header)),
>+                             arena->roundup);
>+    for (;;) {      // loop until we find a suitable region
>+      // find the minimum levels that a block of this size must have
>+      int i = LLA_SkiplistLevels(req_rnd, arena->min_size, nullptr) - 1;
>+      if (i < arena->freelist.levels) {   // potential blocks exist
>+        AllocList *before = &arena->freelist;  // predecessor of s
>+        while ((s = Next(i, before, arena)) != nullptr &&
>+               s->header.size < req_rnd) {
>+          before = s;
>+        }
>+        if (s != nullptr) {       // we found a region
>+          break;
>+        }
>+      }
>+      // we unlock before mmap() both because mmap() may call a callback hook,
>+      // and because it may be slow.
>+      arena->mu.Unlock();
>+      // mmap generous 64K chunks to decrease
>+      // the chances/impact of fragmentation:
>+      size_t new_pages_size = RoundUp(req_rnd, arena->pagesize * 16);
>+      void *new_pages;
>+#ifdef _WIN32
>+      new_pages = VirtualAlloc(0, new_pages_size,
>+                               MEM_RESERVE | MEM_COMMIT, PAGE_READWRITE);
>+      ABSL_RAW_CHECK(new_pages != nullptr, "VirtualAlloc failed");
>+#else
>+      if ((arena->flags & LowLevelAlloc::kAsyncSignalSafe) != 0) {
>+        new_pages = base_internal::DirectMmap(nullptr, new_pages_size,
>+            PROT_WRITE|PROT_READ, MAP_ANONYMOUS|MAP_PRIVATE, -1, 0);
>+      } else {
>+        new_pages = mmap(nullptr, new_pages_size, PROT_WRITE | PROT_READ,
>+                         MAP_ANONYMOUS | MAP_PRIVATE, -1, 0);
>+      }
>+      if (new_pages == MAP_FAILED) {
>+        ABSL_RAW_LOG(FATAL, "mmap error: %d", errno);
>+      }
>+#endif
>+      arena->mu.Lock();
>+      s = reinterpret_cast<AllocList *>(new_pages);
>+      s->header.size = new_pages_size;
>+      // Pretend the block is allocated; call AddToFreelist() to free it.
>+      s->header.magic = Magic(kMagicAllocated, &s->header);
>+      s->header.arena = arena;
>+      AddToFreelist(&s->levels, arena);  // insert new region into free list
>+    }
>+    AllocList *prev[kMaxLevel];
>+    LLA_SkiplistDelete(&arena->freelist, s, prev);    // remove from free list
>+    // s points to the first free region that's big enough
>+    if (CheckedAdd(req_rnd, arena->min_size) <= s->header.size) {
>+      // big enough to split
>+      AllocList *n = reinterpret_cast<AllocList *>
>+                        (req_rnd + reinterpret_cast<char *>(s));
>+      n->header.size = s->header.size - req_rnd;
>+      n->header.magic = Magic(kMagicAllocated, &n->header);
>+      n->header.arena = arena;
>+      s->header.size = req_rnd;
>+      AddToFreelist(&n->levels, arena);
>+    }
>+    s->header.magic = Magic(kMagicAllocated, &s->header);
>+    ABSL_RAW_CHECK(s->header.arena == arena, "");
>+    arena->allocation_count++;
>+    section.Leave();
>+    result = &s->levels;
>+  }
>+  ABSL_ANNOTATE_MEMORY_IS_UNINITIALIZED(result, request);
>+  return result;
>+}
>+
>+void *LowLevelAlloc::Alloc(size_t request) {
>+  void *result = DoAllocWithArena(request, DefaultArena());
>+  return result;
>+}
>+
>+void *LowLevelAlloc::AllocWithArena(size_t request, Arena *arena) {
>+  ABSL_RAW_CHECK(arena != nullptr, "must pass a valid arena");
>+  void *result = DoAllocWithArena(request, arena);
>+  return result;
>+}
>+
>+}  // namespace base_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_LOW_LEVEL_ALLOC_MISSING
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/low_level_alloc.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/low_level_alloc.h
>new file mode 100644
>index 00000000000..3c15605bed3
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/low_level_alloc.h
>@@ -0,0 +1,119 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+
>+#ifndef ABSL_BASE_INTERNAL_LOW_LEVEL_ALLOC_H_
>+#define ABSL_BASE_INTERNAL_LOW_LEVEL_ALLOC_H_
>+
>+// A simple thread-safe memory allocator that does not depend on
>+// mutexes or thread-specific data.  It is intended to be used
>+// sparingly, and only when malloc() would introduce an unwanted
>+// dependency, such as inside the heap-checker, or the Mutex
>+// implementation.
>+
>+// IWYU pragma: private, include "base/low_level_alloc.h"
>+
>+#include <sys/types.h>
>+#include <cstdint>
>+
>+#include "absl/base/attributes.h"
>+#include "absl/base/config.h"
>+
>+// LowLevelAlloc requires that the platform support low-level
>+// allocation of virtual memory. Platforms lacking this cannot use
>+// LowLevelAlloc.
>+#ifdef ABSL_LOW_LEVEL_ALLOC_MISSING
>+#error ABSL_LOW_LEVEL_ALLOC_MISSING cannot be directly set
>+#elif !defined(ABSL_HAVE_MMAP) && !defined(_WIN32)
>+#define ABSL_LOW_LEVEL_ALLOC_MISSING 1
>+#endif
>+
>+// Using LowLevelAlloc with kAsyncSignalSafe isn't supported on Windows.
>+#ifdef ABSL_LOW_LEVEL_ALLOC_ASYNC_SIGNAL_SAFE_MISSING
>+#error ABSL_LOW_LEVEL_ALLOC_ASYNC_SIGNAL_SAFE_MISSING cannot be directly set
>+#elif defined(_WIN32)
>+#define ABSL_LOW_LEVEL_ALLOC_ASYNC_SIGNAL_SAFE_MISSING 1
>+#endif
>+
>+#include <cstddef>
>+
>+#include "absl/base/port.h"
>+
>+namespace absl {
>+namespace base_internal {
>+
>+class LowLevelAlloc {
>+ public:
>+  struct Arena;       // an arena from which memory may be allocated
>+
>+  // Returns a pointer to a block of at least "request" bytes
>+  // that have been newly allocated from the specific arena.
>+  // for Alloc() call the DefaultArena() is used.
>+  // Returns 0 if passed request==0.
>+  // Does not return 0 under other circumstances; it crashes if memory
>+  // is not available.
>+  static void *Alloc(size_t request) ABSL_ATTRIBUTE_SECTION(malloc_hook);
>+  static void *AllocWithArena(size_t request, Arena *arena)
>+      ABSL_ATTRIBUTE_SECTION(malloc_hook);
>+
>+  // Deallocates a region of memory that was previously allocated with
>+  // Alloc().   Does nothing if passed 0.   "s" must be either 0,
>+  // or must have been returned from a call to Alloc() and not yet passed to
>+  // Free() since that call to Alloc().  The space is returned to the arena
>+  // from which it was allocated.
>+  static void Free(void *s) ABSL_ATTRIBUTE_SECTION(malloc_hook);
>+
>+  // ABSL_ATTRIBUTE_SECTION(malloc_hook) for Alloc* and Free
>+  // are to put all callers of MallocHook::Invoke* in this module
>+  // into special section,
>+  // so that MallocHook::GetCallerStackTrace can function accurately.
>+
>+  // Create a new arena.
>+  // The root metadata for the new arena is allocated in the
>+  // meta_data_arena; the DefaultArena() can be passed for meta_data_arena.
>+  // These values may be ored into flags:
>+  enum {
>+    // Report calls to Alloc() and Free() via the MallocHook interface.
>+    // Set in the DefaultArena.
>+    kCallMallocHook = 0x0001,
>+
>+#ifndef ABSL_LOW_LEVEL_ALLOC_ASYNC_SIGNAL_SAFE_MISSING
>+    // Make calls to Alloc(), Free() be async-signal-safe. Not set in
>+    // DefaultArena(). Not supported on all platforms.
>+    kAsyncSignalSafe = 0x0002,
>+#endif
>+  };
>+  // Construct a new arena.  The allocation of the underlying metadata honors
>+  // the provided flags.  For example, the call NewArena(kAsyncSignalSafe)
>+  // is itself async-signal-safe, as well as generatating an arena that provides
>+  // async-signal-safe Alloc/Free.
>+  static Arena *NewArena(int32_t flags);
>+
>+  // Destroys an arena allocated by NewArena and returns true,
>+  // provided no allocated blocks remain in the arena.
>+  // If allocated blocks remain in the arena, does nothing and
>+  // returns false.
>+  // It is illegal to attempt to destroy the DefaultArena().
>+  static bool DeleteArena(Arena *arena);
>+
>+  // The default arena that always exists.
>+  static Arena *DefaultArena();
>+
>+ private:
>+  LowLevelAlloc();      // no instances
>+};
>+
>+}  // namespace base_internal
>+}  // namespace absl
>+#endif  // ABSL_BASE_INTERNAL_LOW_LEVEL_ALLOC_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/low_level_alloc_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/low_level_alloc_test.cc
>new file mode 100644
>index 00000000000..cf2b363299f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/low_level_alloc_test.cc
>@@ -0,0 +1,157 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/base/internal/low_level_alloc.h"
>+
>+#include <stdint.h>
>+#include <stdio.h>
>+#include <stdlib.h>
>+#include <thread>  // NOLINT(build/c++11)
>+#include <unordered_map>
>+#include <utility>
>+
>+namespace absl {
>+namespace base_internal {
>+namespace {
>+
>+// This test doesn't use gtest since it needs to test that everything
>+// works before main().
>+#define TEST_ASSERT(x)                                           \
>+  if (!(x)) {                                                    \
>+    printf("TEST_ASSERT(%s) FAILED ON LINE %d\n", #x, __LINE__); \
>+    abort();                                                     \
>+  }
>+
>+// a block of memory obtained from the allocator
>+struct BlockDesc {
>+  char *ptr;      // pointer to memory
>+  int len;        // number of bytes
>+  int fill;       // filled with data starting with this
>+};
>+
>+// Check that the pattern placed in the block d
>+// by RandomizeBlockDesc is still there.
>+static void CheckBlockDesc(const BlockDesc &d) {
>+  for (int i = 0; i != d.len; i++) {
>+    TEST_ASSERT((d.ptr[i] & 0xff) == ((d.fill + i) & 0xff));
>+  }
>+}
>+
>+// Fill the block "*d" with a pattern
>+// starting with a random byte.
>+static void RandomizeBlockDesc(BlockDesc *d) {
>+  d->fill = rand() & 0xff;
>+  for (int i = 0; i != d->len; i++) {
>+    d->ptr[i] = (d->fill + i) & 0xff;
>+  }
>+}
>+
>+// Use to indicate to the malloc hooks that
>+// this calls is from LowLevelAlloc.
>+static bool using_low_level_alloc = false;
>+
>+// n times, toss a coin, and based on the outcome
>+// either allocate a new block or deallocate an old block.
>+// New blocks are placed in a std::unordered_map with a random key
>+// and initialized with RandomizeBlockDesc().
>+// If keys conflict, the older block is freed.
>+// Old blocks are always checked with CheckBlockDesc()
>+// before being freed.  At the end of the run,
>+// all remaining allocated blocks are freed.
>+// If use_new_arena is true, use a fresh arena, and then delete it.
>+// If call_malloc_hook is true and user_arena is true,
>+// allocations and deallocations are reported via the MallocHook
>+// interface.
>+static void Test(bool use_new_arena, bool call_malloc_hook, int n) {
>+  typedef std::unordered_map<int, BlockDesc> AllocMap;
>+  AllocMap allocated;
>+  AllocMap::iterator it;
>+  BlockDesc block_desc;
>+  int rnd;
>+  LowLevelAlloc::Arena *arena = 0;
>+  if (use_new_arena) {
>+    int32_t flags = call_malloc_hook ? LowLevelAlloc::kCallMallocHook : 0;
>+    arena = LowLevelAlloc::NewArena(flags);
>+  }
>+  for (int i = 0; i != n; i++) {
>+    if (i != 0 && i % 10000 == 0) {
>+      printf(".");
>+      fflush(stdout);
>+    }
>+
>+    switch (rand() & 1) {      // toss a coin
>+    case 0:     // coin came up heads: add a block
>+      using_low_level_alloc = true;
>+      block_desc.len = rand() & 0x3fff;
>+      block_desc.ptr =
>+        reinterpret_cast<char *>(
>+                        arena == 0
>+                        ? LowLevelAlloc::Alloc(block_desc.len)
>+                        : LowLevelAlloc::AllocWithArena(block_desc.len, arena));
>+      using_low_level_alloc = false;
>+      RandomizeBlockDesc(&block_desc);
>+      rnd = rand();
>+      it = allocated.find(rnd);
>+      if (it != allocated.end()) {
>+        CheckBlockDesc(it->second);
>+        using_low_level_alloc = true;
>+        LowLevelAlloc::Free(it->second.ptr);
>+        using_low_level_alloc = false;
>+        it->second = block_desc;
>+      } else {
>+        allocated[rnd] = block_desc;
>+      }
>+      break;
>+    case 1:     // coin came up tails: remove a block
>+      it = allocated.begin();
>+      if (it != allocated.end()) {
>+        CheckBlockDesc(it->second);
>+        using_low_level_alloc = true;
>+        LowLevelAlloc::Free(it->second.ptr);
>+        using_low_level_alloc = false;
>+        allocated.erase(it);
>+      }
>+      break;
>+    }
>+  }
>+  // remove all remaining blocks
>+  while ((it = allocated.begin()) != allocated.end()) {
>+    CheckBlockDesc(it->second);
>+    using_low_level_alloc = true;
>+    LowLevelAlloc::Free(it->second.ptr);
>+    using_low_level_alloc = false;
>+    allocated.erase(it);
>+  }
>+  if (use_new_arena) {
>+    TEST_ASSERT(LowLevelAlloc::DeleteArena(arena));
>+  }
>+}
>+// LowLevelAlloc is designed to be safe to call before main().
>+static struct BeforeMain {
>+  BeforeMain() {
>+    Test(false, false, 50000);
>+    Test(true, false, 50000);
>+    Test(true, true, 50000);
>+  }
>+} before_main;
>+
>+}  // namespace
>+}  // namespace base_internal
>+}  // namespace absl
>+
>+int main(int argc, char *argv[]) {
>+  // The actual test runs in the global constructor of `before_main`.
>+  printf("PASS\n");
>+  return 0;
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/low_level_scheduling.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/low_level_scheduling.h
>new file mode 100644
>index 00000000000..e716f2b49fa
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/low_level_scheduling.h
>@@ -0,0 +1,104 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// Core interfaces and definitions used by by low-level interfaces such as
>+// SpinLock.
>+
>+#ifndef ABSL_BASE_INTERNAL_LOW_LEVEL_SCHEDULING_H_
>+#define ABSL_BASE_INTERNAL_LOW_LEVEL_SCHEDULING_H_
>+
>+#include "absl/base/internal/scheduling_mode.h"
>+#include "absl/base/macros.h"
>+
>+// The following two declarations exist so SchedulingGuard may friend them with
>+// the appropriate language linkage.  These callbacks allow libc internals, such
>+// as function level statics, to schedule cooperatively when locking.
>+extern "C" bool __google_disable_rescheduling(void);
>+extern "C" void __google_enable_rescheduling(bool disable_result);
>+
>+namespace absl {
>+namespace base_internal {
>+
>+class SchedulingHelper;  // To allow use of SchedulingGuard.
>+class SpinLock;          // To allow use of SchedulingGuard.
>+
>+// SchedulingGuard
>+// Provides guard semantics that may be used to disable cooperative rescheduling
>+// of the calling thread within specific program blocks.  This is used to
>+// protect resources (e.g. low-level SpinLocks or Domain code) that cooperative
>+// scheduling depends on.
>+//
>+// Domain implementations capable of rescheduling in reaction to involuntary
>+// kernel thread actions (e.g blocking due to a pagefault or syscall) must
>+// guarantee that an annotated thread is not allowed to (cooperatively)
>+// reschedule until the annotated region is complete.
>+//
>+// It is an error to attempt to use a cooperatively scheduled resource (e.g.
>+// Mutex) within a rescheduling-disabled region.
>+//
>+// All methods are async-signal safe.
>+class SchedulingGuard {
>+ public:
>+  // Returns true iff the calling thread may be cooperatively rescheduled.
>+  static bool ReschedulingIsAllowed();
>+
>+ private:
>+  // Disable cooperative rescheduling of the calling thread.  It may still
>+  // initiate scheduling operations (e.g. wake-ups), however, it may not itself
>+  // reschedule.  Nestable.  The returned result is opaque, clients should not
>+  // attempt to interpret it.
>+  // REQUIRES: Result must be passed to a pairing EnableScheduling().
>+  static bool DisableRescheduling();
>+
>+  // Marks the end of a rescheduling disabled region, previously started by
>+  // DisableRescheduling().
>+  // REQUIRES: Pairs with innermost call (and result) of DisableRescheduling().
>+  static void EnableRescheduling(bool disable_result);
>+
>+  // A scoped helper for {Disable, Enable}Rescheduling().
>+  // REQUIRES: destructor must run in same thread as constructor.
>+  struct ScopedDisable {
>+    ScopedDisable() { disabled = SchedulingGuard::DisableRescheduling(); }
>+    ~ScopedDisable() { SchedulingGuard::EnableRescheduling(disabled); }
>+
>+    bool disabled;
>+  };
>+
>+  // Access to SchedulingGuard is explicitly white-listed.
>+  friend class SchedulingHelper;
>+  friend class SpinLock;
>+
>+  SchedulingGuard(const SchedulingGuard&) = delete;
>+  SchedulingGuard& operator=(const SchedulingGuard&) = delete;
>+};
>+
>+//------------------------------------------------------------------------------
>+// End of public interfaces.
>+//------------------------------------------------------------------------------
>+inline bool SchedulingGuard::ReschedulingIsAllowed() {
>+  return false;
>+}
>+
>+inline bool SchedulingGuard::DisableRescheduling() {
>+  return false;
>+}
>+
>+inline void SchedulingGuard::EnableRescheduling(bool /* disable_result */) {
>+  return;
>+}
>+
>+
>+}  // namespace base_internal
>+}  // namespace absl
>+#endif  // ABSL_BASE_INTERNAL_LOW_LEVEL_SCHEDULING_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/per_thread_tls.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/per_thread_tls.h
>new file mode 100644
>index 00000000000..2428bdc1238
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/per_thread_tls.h
>@@ -0,0 +1,48 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#ifndef ABSL_BASE_INTERNAL_PER_THREAD_TLS_H_
>+#define ABSL_BASE_INTERNAL_PER_THREAD_TLS_H_
>+
>+// This header defines two macros:
>+// If the platform supports thread-local storage:
>+//   ABSL_PER_THREAD_TLS_KEYWORD is the C keyword needed to declare a
>+//   thread-local variable ABSL_PER_THREAD_TLS is 1
>+//
>+// Otherwise:
>+//   ABSL_PER_THREAD_TLS_KEYWORD is empty
>+//   ABSL_PER_THREAD_TLS is 0
>+//
>+// Microsoft C supports thread-local storage.
>+// GCC supports it if the appropriate version of glibc is available,
>+// which the programmer can indicate by defining ABSL_HAVE_TLS
>+
>+#include "absl/base/port.h"  // For ABSL_HAVE_TLS
>+
>+#if defined(ABSL_PER_THREAD_TLS)
>+#error ABSL_PER_THREAD_TLS cannot be directly set
>+#elif defined(ABSL_PER_THREAD_TLS_KEYWORD)
>+#error ABSL_PER_THREAD_TLS_KEYWORD cannot be directly set
>+#elif defined(ABSL_HAVE_TLS)
>+#define ABSL_PER_THREAD_TLS_KEYWORD __thread
>+#define ABSL_PER_THREAD_TLS 1
>+#elif defined(_MSC_VER)
>+#define ABSL_PER_THREAD_TLS_KEYWORD __declspec(thread)
>+#define ABSL_PER_THREAD_TLS 1
>+#else
>+#define ABSL_PER_THREAD_TLS_KEYWORD
>+#define ABSL_PER_THREAD_TLS 0
>+#endif
>+
>+#endif  // ABSL_BASE_INTERNAL_PER_THREAD_TLS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/pretty_function.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/pretty_function.h
>new file mode 100644
>index 00000000000..01b0547bd08
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/pretty_function.h
>@@ -0,0 +1,33 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#ifndef ABSL_BASE_INTERNAL_PRETTY_FUNCTION_H_
>+#define ABSL_BASE_INTERNAL_PRETTY_FUNCTION_H_
>+
>+// ABSL_PRETTY_FUNCTION
>+//
>+// In C++11, __func__ gives the undecorated name of the current function.  That
>+// is, "main", not "int main()".  Various compilers give extra macros to get the
>+// decorated function name, including return type and arguments, to
>+// differentiate between overload sets.  ABSL_PRETTY_FUNCTION is a portable
>+// version of these macros which forwards to the correct macro on each compiler.
>+#if defined(_MSC_VER)
>+#define ABSL_PRETTY_FUNCTION __FUNCSIG__
>+#elif defined(__GNUC__)
>+#define ABSL_PRETTY_FUNCTION __PRETTY_FUNCTION__
>+#else
>+#error "Unsupported compiler"
>+#endif
>+
>+#endif  // ABSL_BASE_INTERNAL_PRETTY_FUNCTION_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/raw_logging.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/raw_logging.cc
>new file mode 100644
>index 00000000000..d9485a66cc6
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/raw_logging.cc
>@@ -0,0 +1,234 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/base/internal/raw_logging.h"
>+
>+#include <stddef.h>
>+#include <cstdarg>
>+#include <cstdio>
>+#include <cstdlib>
>+#include <cstring>
>+
>+#include "absl/base/attributes.h"
>+#include "absl/base/config.h"
>+#include "absl/base/internal/atomic_hook.h"
>+#include "absl/base/log_severity.h"
>+
>+// We know how to perform low-level writes to stderr in POSIX and Windows.  For
>+// these platforms, we define the token ABSL_LOW_LEVEL_WRITE_SUPPORTED.
>+// Much of raw_logging.cc becomes a no-op when we can't output messages,
>+// although a FATAL ABSL_RAW_LOG message will still abort the process.
>+
>+// ABSL_HAVE_POSIX_WRITE is defined when the platform provides posix write()
>+// (as from unistd.h)
>+//
>+// This preprocessor token is also defined in raw_io.cc.  If you need to copy
>+// this, consider moving both to config.h instead.
>+#if defined(__linux__) || defined(__APPLE__) || defined(__FreeBSD__) || \
>+    defined(__Fuchsia__) || defined(__native_client__)
>+#include <unistd.h>
>+
>+
>+#define ABSL_HAVE_POSIX_WRITE 1
>+#define ABSL_LOW_LEVEL_WRITE_SUPPORTED 1
>+#else
>+#undef ABSL_HAVE_POSIX_WRITE
>+#endif
>+
>+// ABSL_HAVE_SYSCALL_WRITE is defined when the platform provides the syscall
>+//   syscall(SYS_write, /*int*/ fd, /*char* */ buf, /*size_t*/ len);
>+// for low level operations that want to avoid libc.
>+#if (defined(__linux__) || defined(__FreeBSD__)) && !defined(__ANDROID__)
>+#include <sys/syscall.h>
>+#define ABSL_HAVE_SYSCALL_WRITE 1
>+#define ABSL_LOW_LEVEL_WRITE_SUPPORTED 1
>+#else
>+#undef ABSL_HAVE_SYSCALL_WRITE
>+#endif
>+
>+#ifdef _WIN32
>+#include <io.h>
>+
>+#define ABSL_HAVE_RAW_IO 1
>+#define ABSL_LOW_LEVEL_WRITE_SUPPORTED 1
>+#else
>+#undef ABSL_HAVE_RAW_IO
>+#endif
>+
>+// TODO(gfalcon): We want raw-logging to work on as many platforms as possible.
>+// Explicitly #error out when not ABSL_LOW_LEVEL_WRITE_SUPPORTED, except for a
>+// whitelisted set of platforms for which we expect not to be able to raw log.
>+
>+ABSL_CONST_INIT static absl::base_internal::AtomicHook<
>+    absl::raw_logging_internal::LogPrefixHook> log_prefix_hook;
>+ABSL_CONST_INIT static absl::base_internal::AtomicHook<
>+    absl::raw_logging_internal::AbortHook> abort_hook;
>+
>+#ifdef ABSL_LOW_LEVEL_WRITE_SUPPORTED
>+static const char kTruncated[] = " ... (message truncated)\n";
>+
>+// sprintf the format to the buffer, adjusting *buf and *size to reflect the
>+// consumed bytes, and return whether the message fit without truncation.  If
>+// truncation occurred, if possible leave room in the buffer for the message
>+// kTruncated[].
>+inline static bool VADoRawLog(char** buf, int* size, const char* format,
>+                              va_list ap) ABSL_PRINTF_ATTRIBUTE(3, 0);
>+inline static bool VADoRawLog(char** buf, int* size,
>+                              const char* format, va_list ap) {
>+  int n = vsnprintf(*buf, *size, format, ap);
>+  bool result = true;
>+  if (n < 0 || n > *size) {
>+    result = false;
>+    if (static_cast<size_t>(*size) > sizeof(kTruncated)) {
>+      n = *size - sizeof(kTruncated);  // room for truncation message
>+    } else {
>+      n = 0;                           // no room for truncation message
>+    }
>+  }
>+  *size -= n;
>+  *buf += n;
>+  return result;
>+}
>+#endif  // ABSL_LOW_LEVEL_WRITE_SUPPORTED
>+
>+static constexpr int kLogBufSize = 3000;
>+
>+namespace {
>+
>+// CAVEAT: vsnprintf called from *DoRawLog below has some (exotic) code paths
>+// that invoke malloc() and getenv() that might acquire some locks.
>+
>+// Helper for RawLog below.
>+// *DoRawLog writes to *buf of *size and move them past the written portion.
>+// It returns true iff there was no overflow or error.
>+bool DoRawLog(char** buf, int* size, const char* format, ...)
>+    ABSL_PRINTF_ATTRIBUTE(3, 4);
>+bool DoRawLog(char** buf, int* size, const char* format, ...) {
>+  va_list ap;
>+  va_start(ap, format);
>+  int n = vsnprintf(*buf, *size, format, ap);
>+  va_end(ap);
>+  if (n < 0 || n > *size) return false;
>+  *size -= n;
>+  *buf += n;
>+  return true;
>+}
>+
>+void RawLogVA(absl::LogSeverity severity, const char* file, int line,
>+              const char* format, va_list ap) ABSL_PRINTF_ATTRIBUTE(4, 0);
>+void RawLogVA(absl::LogSeverity severity, const char* file, int line,
>+              const char* format, va_list ap) {
>+  char buffer[kLogBufSize];
>+  char* buf = buffer;
>+  int size = sizeof(buffer);
>+#ifdef ABSL_LOW_LEVEL_WRITE_SUPPORTED
>+  bool enabled = true;
>+#else
>+  bool enabled = false;
>+#endif
>+
>+#ifdef ABSL_MIN_LOG_LEVEL
>+  if (severity < static_cast<absl::LogSeverity>(ABSL_MIN_LOG_LEVEL) &&
>+      severity < absl::LogSeverity::kFatal) {
>+    enabled = false;
>+  }
>+#endif
>+
>+  auto log_prefix_hook_ptr = log_prefix_hook.Load();
>+  if (log_prefix_hook_ptr) {
>+    enabled = log_prefix_hook_ptr(severity, file, line, &buf, &size);
>+  } else {
>+    if (enabled) {
>+      DoRawLog(&buf, &size, "[%s : %d] RAW: ", file, line);
>+    }
>+  }
>+  const char* const prefix_end = buf;
>+
>+#ifdef ABSL_LOW_LEVEL_WRITE_SUPPORTED
>+  if (enabled) {
>+    bool no_chop = VADoRawLog(&buf, &size, format, ap);
>+    if (no_chop) {
>+      DoRawLog(&buf, &size, "\n");
>+    } else {
>+      DoRawLog(&buf, &size, "%s", kTruncated);
>+    }
>+    absl::raw_logging_internal::SafeWriteToStderr(buffer, strlen(buffer));
>+  }
>+#else
>+  static_cast<void>(format);
>+  static_cast<void>(ap);
>+#endif
>+
>+  // Abort the process after logging a FATAL message, even if the output itself
>+  // was suppressed.
>+  if (severity == absl::LogSeverity::kFatal) {
>+    abort_hook(file, line, buffer, prefix_end, buffer + kLogBufSize);
>+    abort();
>+  }
>+}
>+
>+}  // namespace
>+
>+namespace absl {
>+namespace raw_logging_internal {
>+void SafeWriteToStderr(const char *s, size_t len) {
>+#if defined(ABSL_HAVE_SYSCALL_WRITE)
>+  syscall(SYS_write, STDERR_FILENO, s, len);
>+#elif defined(ABSL_HAVE_POSIX_WRITE)
>+  write(STDERR_FILENO, s, len);
>+#elif defined(ABSL_HAVE_RAW_IO)
>+  _write(/* stderr */ 2, s, len);
>+#else
>+  // stderr logging unsupported on this platform
>+  (void) s;
>+  (void) len;
>+#endif
>+}
>+
>+void RawLog(absl::LogSeverity severity, const char* file, int line,
>+            const char* format, ...) ABSL_PRINTF_ATTRIBUTE(4, 5);
>+void RawLog(absl::LogSeverity severity, const char* file, int line,
>+            const char* format, ...) {
>+  va_list ap;
>+  va_start(ap, format);
>+  RawLogVA(severity, file, line, format, ap);
>+  va_end(ap);
>+}
>+
>+// Non-formatting version of RawLog().
>+//
>+// TODO(gfalcon): When string_view no longer depends on base, change this
>+// interface to take its message as a string_view instead.
>+static void DefaultInternalLog(absl::LogSeverity severity, const char* file,
>+                               int line, const std::string& message) {
>+  RawLog(severity, file, line, "%s", message.c_str());
>+}
>+
>+bool RawLoggingFullySupported() {
>+#ifdef ABSL_LOW_LEVEL_WRITE_SUPPORTED
>+  return true;
>+#else  // !ABSL_LOW_LEVEL_WRITE_SUPPORTED
>+  return false;
>+#endif  // !ABSL_LOW_LEVEL_WRITE_SUPPORTED
>+}
>+
>+ABSL_CONST_INIT absl::base_internal::AtomicHook<InternalLogFunction>
>+    internal_log_function(DefaultInternalLog);
>+
>+void RegisterInternalLogFunction(InternalLogFunction func) {
>+  internal_log_function.Store(func);
>+}
>+
>+}  // namespace raw_logging_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/raw_logging.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/raw_logging.h
>new file mode 100644
>index 00000000000..67abfd30798
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/raw_logging.h
>@@ -0,0 +1,180 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// Thread-safe logging routines that do not allocate any memory or
>+// acquire any locks, and can therefore be used by low-level memory
>+// allocation, synchronization, and signal-handling code.
>+
>+#ifndef ABSL_BASE_INTERNAL_RAW_LOGGING_H_
>+#define ABSL_BASE_INTERNAL_RAW_LOGGING_H_
>+
>+#include <string>
>+
>+#include "absl/base/attributes.h"
>+#include "absl/base/internal/atomic_hook.h"
>+#include "absl/base/log_severity.h"
>+#include "absl/base/macros.h"
>+#include "absl/base/port.h"
>+
>+// This is similar to LOG(severity) << format..., but
>+// * it is to be used ONLY by low-level modules that can't use normal LOG()
>+// * it is designed to be a low-level logger that does not allocate any
>+//   memory and does not need any locks, hence:
>+// * it logs straight and ONLY to STDERR w/o buffering
>+// * it uses an explicit printf-format and arguments list
>+// * it will silently chop off really long message strings
>+// Usage example:
>+//   ABSL_RAW_LOG(ERROR, "Failed foo with %i: %s", status, error);
>+// This will print an almost standard log line like this to stderr only:
>+//   E0821 211317 file.cc:123] RAW: Failed foo with 22: bad_file
>+#define ABSL_RAW_LOG(severity, ...)                                            \
>+  do {                                                                         \
>+    constexpr const char* absl_raw_logging_internal_basename =                 \
>+        ::absl::raw_logging_internal::Basename(__FILE__,                       \
>+                                               sizeof(__FILE__) - 1);          \
>+    ::absl::raw_logging_internal::RawLog(ABSL_RAW_LOGGING_INTERNAL_##severity, \
>+                                         absl_raw_logging_internal_basename,   \
>+                                         __LINE__, __VA_ARGS__);               \
>+  } while (0)
>+
>+// Similar to CHECK(condition) << message, but for low-level modules:
>+// we use only ABSL_RAW_LOG that does not allocate memory.
>+// We do not want to provide args list here to encourage this usage:
>+//   if (!cond)  ABSL_RAW_LOG(FATAL, "foo ...", hard_to_compute_args);
>+// so that the args are not computed when not needed.
>+#define ABSL_RAW_CHECK(condition, message)                             \
>+  do {                                                                 \
>+    if (ABSL_PREDICT_FALSE(!(condition))) {                            \
>+      ABSL_RAW_LOG(FATAL, "Check %s failed: %s", #condition, message); \
>+    }                                                                  \
>+  } while (0)
>+
>+// ABSL_INTERNAL_LOG and ABSL_INTERNAL_CHECK work like the RAW variants above,
>+// except that if the richer log library is linked into the binary, we dispatch
>+// to that instead.  This is potentially useful for internal logging and
>+// assertions, where we are using RAW_LOG neither for its async-signal-safety
>+// nor for its non-allocating nature, but rather because raw logging has very
>+// few other dependencies.
>+//
>+// The API is a subset of the above: each macro only takes two arguments.  Use
>+// StrCat if you need to build a richer message.
>+#define ABSL_INTERNAL_LOG(severity, message)                          \
>+  do {                                                                \
>+    constexpr const char* absl_raw_logging_internal_basename =        \
>+        ::absl::raw_logging_internal::Basename(__FILE__,              \
>+                                               sizeof(__FILE__) - 1); \
>+    ::absl::raw_logging_internal::internal_log_function(              \
>+        ABSL_RAW_LOGGING_INTERNAL_##severity,                         \
>+        absl_raw_logging_internal_basename, __LINE__, message);       \
>+  } while (0)
>+
>+#define ABSL_INTERNAL_CHECK(condition, message)               \
>+  do {                                                        \
>+    if (ABSL_PREDICT_FALSE(!(condition))) {                   \
>+      std::string death_message = "Check " #condition " failed: "; \
>+      death_message += std::string(message);                       \
>+      ABSL_INTERNAL_LOG(FATAL, death_message);                \
>+    }                                                         \
>+  } while (0)
>+
>+#define ABSL_RAW_LOGGING_INTERNAL_INFO ::absl::LogSeverity::kInfo
>+#define ABSL_RAW_LOGGING_INTERNAL_WARNING ::absl::LogSeverity::kWarning
>+#define ABSL_RAW_LOGGING_INTERNAL_ERROR ::absl::LogSeverity::kError
>+#define ABSL_RAW_LOGGING_INTERNAL_FATAL ::absl::LogSeverity::kFatal
>+#define ABSL_RAW_LOGGING_INTERNAL_LEVEL(severity) \
>+  ::absl::NormalizeLogSeverity(severity)
>+
>+namespace absl {
>+namespace raw_logging_internal {
>+
>+// Helper function to implement ABSL_RAW_LOG
>+// Logs format... at "severity" level, reporting it
>+// as called from file:line.
>+// This does not allocate memory or acquire locks.
>+void RawLog(absl::LogSeverity severity, const char* file, int line,
>+            const char* format, ...) ABSL_PRINTF_ATTRIBUTE(4, 5);
>+
>+// Writes the provided buffer directly to stderr, in a safe, low-level manner.
>+//
>+// In POSIX this means calling write(), which is async-signal safe and does
>+// not malloc.  If the platform supports the SYS_write syscall, we invoke that
>+// directly to side-step any libc interception.
>+void SafeWriteToStderr(const char *s, size_t len);
>+
>+// compile-time function to get the "base" filename, that is, the part of
>+// a filename after the last "/" or "\" path separator.  The search starts at
>+// the end of the std::string; the second parameter is the length of the std::string.
>+constexpr const char* Basename(const char* fname, int offset) {
>+  return offset == 0 || fname[offset - 1] == '/' || fname[offset - 1] == '\\'
>+             ? fname + offset
>+             : Basename(fname, offset - 1);
>+}
>+
>+// For testing only.
>+// Returns true if raw logging is fully supported. When it is not
>+// fully supported, no messages will be emitted, but a log at FATAL
>+// severity will cause an abort.
>+//
>+// TODO(gfalcon): Come up with a better name for this method.
>+bool RawLoggingFullySupported();
>+
>+// Function type for a raw_logging customization hook for suppressing messages
>+// by severity, and for writing custom prefixes on non-suppressed messages.
>+//
>+// The installed hook is called for every raw log invocation.  The message will
>+// be logged to stderr only if the hook returns true.  FATAL errors will cause
>+// the process to abort, even if writing to stderr is suppressed.  The hook is
>+// also provided with an output buffer, where it can write a custom log message
>+// prefix.
>+//
>+// The raw_logging system does not allocate memory or grab locks.  User-provided
>+// hooks must avoid these operations, and must not throw exceptions.
>+//
>+// 'severity' is the severity level of the message being written.
>+// 'file' and 'line' are the file and line number where the ABSL_RAW_LOG macro
>+// was located.
>+// 'buffer' and 'buf_size' are pointers to the buffer and buffer size.  If the
>+// hook writes a prefix, it must increment *buffer and decrement *buf_size
>+// accordingly.
>+using LogPrefixHook = bool (*)(absl::LogSeverity severity, const char* file,
>+                               int line, char** buffer, int* buf_size);
>+
>+// Function type for a raw_logging customization hook called to abort a process
>+// when a FATAL message is logged.  If the provided AbortHook() returns, the
>+// logging system will call abort().
>+//
>+// 'file' and 'line' are the file and line number where the ABSL_RAW_LOG macro
>+// was located.
>+// The null-terminated logged message lives in the buffer between 'buf_start'
>+// and 'buf_end'.  'prefix_end' points to the first non-prefix character of the
>+// buffer (as written by the LogPrefixHook.)
>+using AbortHook = void (*)(const char* file, int line, const char* buf_start,
>+                           const char* prefix_end, const char* buf_end);
>+
>+// Internal logging function for ABSL_INTERNAL_LOG to dispatch to.
>+//
>+// TODO(gfalcon): When string_view no longer depends on base, change this
>+// interface to take its message as a string_view instead.
>+using InternalLogFunction = void (*)(absl::LogSeverity severity,
>+                                     const char* file, int line,
>+                                     const std::string& message);
>+
>+extern base_internal::AtomicHook<InternalLogFunction> internal_log_function;
>+
>+void RegisterInternalLogFunction(InternalLogFunction func);
>+
>+}  // namespace raw_logging_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_BASE_INTERNAL_RAW_LOGGING_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/scheduling_mode.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/scheduling_mode.h
>new file mode 100644
>index 00000000000..1b6497ad875
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/scheduling_mode.h
>@@ -0,0 +1,54 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// Core interfaces and definitions used by by low-level interfaces such as
>+// SpinLock.
>+
>+#ifndef ABSL_BASE_INTERNAL_SCHEDULING_MODE_H_
>+#define ABSL_BASE_INTERNAL_SCHEDULING_MODE_H_
>+
>+namespace absl {
>+namespace base_internal {
>+
>+// Used to describe how a thread may be scheduled.  Typically associated with
>+// the declaration of a resource supporting synchronized access.
>+//
>+// SCHEDULE_COOPERATIVE_AND_KERNEL:
>+// Specifies that when waiting, a cooperative thread (e.g. a Fiber) may
>+// reschedule (using base::scheduling semantics); allowing other cooperative
>+// threads to proceed.
>+//
>+// SCHEDULE_KERNEL_ONLY: (Also described as "non-cooperative")
>+// Specifies that no cooperative scheduling semantics may be used, even if the
>+// current thread is itself cooperatively scheduled.  This means that
>+// cooperative threads will NOT allow other cooperative threads to execute in
>+// their place while waiting for a resource of this type.  Host operating system
>+// semantics (e.g. a futex) may still be used.
>+//
>+// When optional, clients should strongly prefer SCHEDULE_COOPERATIVE_AND_KERNEL
>+// by default.  SCHEDULE_KERNEL_ONLY should only be used for resources on which
>+// base::scheduling (e.g. the implementation of a Scheduler) may depend.
>+//
>+// NOTE: Cooperative resources may not be nested below non-cooperative ones.
>+// This means that it is invalid to to acquire a SCHEDULE_COOPERATIVE_AND_KERNEL
>+// resource if a SCHEDULE_KERNEL_ONLY resource is already held.
>+enum SchedulingMode {
>+  SCHEDULE_KERNEL_ONLY = 0,         // Allow scheduling only the host OS.
>+  SCHEDULE_COOPERATIVE_AND_KERNEL,  // Also allow cooperative scheduling.
>+};
>+
>+}  // namespace base_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_BASE_INTERNAL_SCHEDULING_MODE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/spinlock.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/spinlock.cc
>new file mode 100644
>index 00000000000..1b97efbccc5
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/spinlock.cc
>@@ -0,0 +1,228 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/base/internal/spinlock.h"
>+
>+#include <algorithm>
>+#include <atomic>
>+#include <limits>
>+
>+#include "absl/base/attributes.h"
>+#include "absl/base/internal/atomic_hook.h"
>+#include "absl/base/internal/cycleclock.h"
>+#include "absl/base/internal/spinlock_wait.h"
>+#include "absl/base/internal/sysinfo.h" /* For NumCPUs() */
>+#include "absl/base/call_once.h"
>+
>+// Description of lock-word:
>+//  31..00: [............................3][2][1][0]
>+//
>+//     [0]: kSpinLockHeld
>+//     [1]: kSpinLockCooperative
>+//     [2]: kSpinLockDisabledScheduling
>+// [31..3]: ONLY kSpinLockSleeper OR
>+//          Wait time in cycles >> PROFILE_TIMESTAMP_SHIFT
>+//
>+// Detailed descriptions:
>+//
>+// Bit [0]: The lock is considered held iff kSpinLockHeld is set.
>+//
>+// Bit [1]: Eligible waiters (e.g. Fibers) may co-operatively reschedule when
>+//          contended iff kSpinLockCooperative is set.
>+//
>+// Bit [2]: This bit is exclusive from bit [1].  It is used only by a
>+//          non-cooperative lock.  When set, indicates that scheduling was
>+//          successfully disabled when the lock was acquired.  May be unset,
>+//          even if non-cooperative, if a ThreadIdentity did not yet exist at
>+//          time of acquisition.
>+//
>+// Bit [3]: If this is the only upper bit ([31..3]) set then this lock was
>+//          acquired without contention, however, at least one waiter exists.
>+//
>+//          Otherwise, bits [31..3] represent the time spent by the current lock
>+//          holder to acquire the lock.  There may be outstanding waiter(s).
>+
>+namespace absl {
>+namespace base_internal {
>+
>+ABSL_CONST_INIT static base_internal::AtomicHook<void (*)(const void *lock,
>+                                                          int64_t wait_cycles)>
>+    submit_profile_data;
>+
>+void RegisterSpinLockProfiler(void (*fn)(const void *contendedlock,
>+                                         int64_t wait_cycles)) {
>+  submit_profile_data.Store(fn);
>+}
>+
>+// Uncommon constructors.
>+SpinLock::SpinLock(base_internal::SchedulingMode mode)
>+    : lockword_(IsCooperative(mode) ? kSpinLockCooperative : 0) {
>+  ABSL_TSAN_MUTEX_CREATE(this, __tsan_mutex_not_static);
>+}
>+
>+SpinLock::SpinLock(base_internal::LinkerInitialized,
>+                   base_internal::SchedulingMode mode) {
>+  ABSL_TSAN_MUTEX_CREATE(this, 0);
>+  if (IsCooperative(mode)) {
>+    InitLinkerInitializedAndCooperative();
>+  }
>+  // Otherwise, lockword_ is already initialized.
>+}
>+
>+// Static (linker initialized) spinlocks always start life as functional
>+// non-cooperative locks.  When their static constructor does run, it will call
>+// this initializer to augment the lockword with the cooperative bit.  By
>+// actually taking the lock when we do this we avoid the need for an atomic
>+// operation in the regular unlock path.
>+//
>+// SlowLock() must be careful to re-test for this bit so that any outstanding
>+// waiters may be upgraded to cooperative status.
>+void SpinLock::InitLinkerInitializedAndCooperative() {
>+  Lock();
>+  lockword_.fetch_or(kSpinLockCooperative, std::memory_order_relaxed);
>+  Unlock();
>+}
>+
>+// Monitor the lock to see if its value changes within some time period
>+// (adaptive_spin_count loop iterations).  A timestamp indicating
>+// when the thread initially started waiting for the lock is passed in via
>+// the initial_wait_timestamp value.  The total wait time in cycles for the
>+// lock is returned in the wait_cycles parameter.  The last value read
>+// from the lock is returned from the method.
>+uint32_t SpinLock::SpinLoop(int64_t initial_wait_timestamp,
>+                            uint32_t *wait_cycles) {
>+  // We are already in the slow path of SpinLock, initialize the
>+  // adaptive_spin_count here.
>+  ABSL_CONST_INIT static absl::once_flag init_adaptive_spin_count;
>+  ABSL_CONST_INIT static int adaptive_spin_count = 0;
>+  base_internal::LowLevelCallOnce(&init_adaptive_spin_count, []() {
>+    adaptive_spin_count = base_internal::NumCPUs() > 1 ? 1000 : 1;
>+  });
>+
>+  int c = adaptive_spin_count;
>+  uint32_t lock_value;
>+  do {
>+    lock_value = lockword_.load(std::memory_order_relaxed);
>+  } while ((lock_value & kSpinLockHeld) != 0 && --c > 0);
>+  uint32_t spin_loop_wait_cycles =
>+      EncodeWaitCycles(initial_wait_timestamp, CycleClock::Now());
>+  *wait_cycles = spin_loop_wait_cycles;
>+
>+  return TryLockInternal(lock_value, spin_loop_wait_cycles);
>+}
>+
>+void SpinLock::SlowLock() {
>+  // The lock was not obtained initially, so this thread needs to wait for
>+  // it.  Record the current timestamp in the local variable wait_start_time
>+  // so the total wait time can be stored in the lockword once this thread
>+  // obtains the lock.
>+  int64_t wait_start_time = CycleClock::Now();
>+  uint32_t wait_cycles;
>+  uint32_t lock_value = SpinLoop(wait_start_time, &wait_cycles);
>+
>+  int lock_wait_call_count = 0;
>+  while ((lock_value & kSpinLockHeld) != 0) {
>+    // If the lock is currently held, but not marked as having a sleeper, mark
>+    // it as having a sleeper.
>+    if ((lock_value & kWaitTimeMask) == 0) {
>+      // Here, just "mark" that the thread is going to sleep.  Don't store the
>+      // lock wait time in the lock as that will cause the current lock
>+      // owner to think it experienced contention.
>+      if (lockword_.compare_exchange_strong(
>+              lock_value, lock_value | kSpinLockSleeper,
>+              std::memory_order_acquire, std::memory_order_relaxed)) {
>+        // Successfully transitioned to kSpinLockSleeper.  Pass
>+        // kSpinLockSleeper to the SpinLockWait routine to properly indicate
>+        // the last lock_value observed.
>+        lock_value |= kSpinLockSleeper;
>+      } else if ((lock_value & kSpinLockHeld) == 0) {
>+        // Lock is free again, so try and acquire it before sleeping.  The
>+        // new lock state will be the number of cycles this thread waited if
>+        // this thread obtains the lock.
>+        lock_value = TryLockInternal(lock_value, wait_cycles);
>+        continue;   // Skip the delay at the end of the loop.
>+      }
>+    }
>+
>+    base_internal::SchedulingMode scheduling_mode;
>+    if ((lock_value & kSpinLockCooperative) != 0) {
>+      scheduling_mode = base_internal::SCHEDULE_COOPERATIVE_AND_KERNEL;
>+    } else {
>+      scheduling_mode = base_internal::SCHEDULE_KERNEL_ONLY;
>+    }
>+    // SpinLockDelay() calls into fiber scheduler, we need to see
>+    // synchronization there to avoid false positives.
>+    ABSL_TSAN_MUTEX_PRE_DIVERT(this, 0);
>+    // Wait for an OS specific delay.
>+    base_internal::SpinLockDelay(&lockword_, lock_value, ++lock_wait_call_count,
>+                                 scheduling_mode);
>+    ABSL_TSAN_MUTEX_POST_DIVERT(this, 0);
>+    // Spin again after returning from the wait routine to give this thread
>+    // some chance of obtaining the lock.
>+    lock_value = SpinLoop(wait_start_time, &wait_cycles);
>+  }
>+}
>+
>+void SpinLock::SlowUnlock(uint32_t lock_value) {
>+  base_internal::SpinLockWake(&lockword_,
>+                              false);  // wake waiter if necessary
>+
>+  // If our acquisition was contended, collect contentionz profile info.  We
>+  // reserve a unitary wait time to represent that a waiter exists without our
>+  // own acquisition having been contended.
>+  if ((lock_value & kWaitTimeMask) != kSpinLockSleeper) {
>+    const uint64_t wait_cycles = DecodeWaitCycles(lock_value);
>+    ABSL_TSAN_MUTEX_PRE_DIVERT(this, 0);
>+    submit_profile_data(this, wait_cycles);
>+    ABSL_TSAN_MUTEX_POST_DIVERT(this, 0);
>+  }
>+}
>+
>+// We use the upper 29 bits of the lock word to store the time spent waiting to
>+// acquire this lock.  This is reported by contentionz profiling.  Since the
>+// lower bits of the cycle counter wrap very quickly on high-frequency
>+// processors we divide to reduce the granularity to 2^PROFILE_TIMESTAMP_SHIFT
>+// sized units.  On a 4Ghz machine this will lose track of wait times greater
>+// than (2^29/4 Ghz)*128 =~ 17.2 seconds.  Such waits should be extremely rare.
>+enum { PROFILE_TIMESTAMP_SHIFT = 7 };
>+enum { LOCKWORD_RESERVED_SHIFT = 3 };  // We currently reserve the lower 3 bits.
>+
>+uint32_t SpinLock::EncodeWaitCycles(int64_t wait_start_time,
>+                                    int64_t wait_end_time) {
>+  static const int64_t kMaxWaitTime =
>+      std::numeric_limits<uint32_t>::max() >> LOCKWORD_RESERVED_SHIFT;
>+  int64_t scaled_wait_time =
>+      (wait_end_time - wait_start_time) >> PROFILE_TIMESTAMP_SHIFT;
>+
>+  // Return a representation of the time spent waiting that can be stored in
>+  // the lock word's upper bits.  bit_cast is required as Atomic32 is signed.
>+  const uint32_t clamped = static_cast<uint32_t>(
>+      std::min(scaled_wait_time, kMaxWaitTime) << LOCKWORD_RESERVED_SHIFT);
>+
>+  // bump up value if necessary to avoid returning kSpinLockSleeper.
>+  const uint32_t after_spinlock_sleeper =
>+     kSpinLockSleeper + (1 << LOCKWORD_RESERVED_SHIFT);
>+  return clamped == kSpinLockSleeper ? after_spinlock_sleeper : clamped;
>+}
>+
>+uint64_t SpinLock::DecodeWaitCycles(uint32_t lock_value) {
>+  // Cast to uint32_t first to ensure bits [63:32] are cleared.
>+  const uint64_t scaled_wait_time =
>+      static_cast<uint32_t>(lock_value & kWaitTimeMask);
>+  return scaled_wait_time
>+      << (PROFILE_TIMESTAMP_SHIFT - LOCKWORD_RESERVED_SHIFT);
>+}
>+
>+}  // namespace base_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/spinlock.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/spinlock.h
>new file mode 100644
>index 00000000000..212abc669e6
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/spinlock.h
>@@ -0,0 +1,239 @@
>+//
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+
>+//  Most users requiring mutual exclusion should use Mutex.
>+//  SpinLock is provided for use in three situations:
>+//   - for use in code that Mutex itself depends on
>+//   - to get a faster fast-path release under low contention (without an
>+//     atomic read-modify-write) In return, SpinLock has worse behaviour under
>+//     contention, which is why Mutex is preferred in most situations.
>+//   - for async signal safety (see below)
>+
>+// SpinLock is async signal safe.  If a spinlock is used within a signal
>+// handler, all code that acquires the lock must ensure that the signal cannot
>+// arrive while they are holding the lock.  Typically, this is done by blocking
>+// the signal.
>+
>+#ifndef ABSL_BASE_INTERNAL_SPINLOCK_H_
>+#define ABSL_BASE_INTERNAL_SPINLOCK_H_
>+
>+#include <stdint.h>
>+#include <sys/types.h>
>+#include <atomic>
>+
>+#include "absl/base/attributes.h"
>+#include "absl/base/dynamic_annotations.h"
>+#include "absl/base/internal/low_level_scheduling.h"
>+#include "absl/base/internal/raw_logging.h"
>+#include "absl/base/internal/scheduling_mode.h"
>+#include "absl/base/internal/tsan_mutex_interface.h"
>+#include "absl/base/macros.h"
>+#include "absl/base/port.h"
>+#include "absl/base/thread_annotations.h"
>+
>+namespace absl {
>+namespace base_internal {
>+
>+class LOCKABLE SpinLock {
>+ public:
>+  SpinLock() : lockword_(kSpinLockCooperative) {
>+    ABSL_TSAN_MUTEX_CREATE(this, __tsan_mutex_not_static);
>+  }
>+
>+  // Special constructor for use with static SpinLock objects.  E.g.,
>+  //
>+  //    static SpinLock lock(base_internal::kLinkerInitialized);
>+  //
>+  // When intialized using this constructor, we depend on the fact
>+  // that the linker has already initialized the memory appropriately.
>+  // A SpinLock constructed like this can be freely used from global
>+  // initializers without worrying about the order in which global
>+  // initializers run.
>+  explicit SpinLock(base_internal::LinkerInitialized) {
>+    // Does nothing; lockword_ is already initialized
>+    ABSL_TSAN_MUTEX_CREATE(this, 0);
>+  }
>+
>+  // Constructors that allow non-cooperative spinlocks to be created for use
>+  // inside thread schedulers.  Normal clients should not use these.
>+  explicit SpinLock(base_internal::SchedulingMode mode);
>+  SpinLock(base_internal::LinkerInitialized,
>+           base_internal::SchedulingMode mode);
>+
>+  ~SpinLock() { ABSL_TSAN_MUTEX_DESTROY(this, __tsan_mutex_not_static); }
>+
>+  // Acquire this SpinLock.
>+  inline void Lock() EXCLUSIVE_LOCK_FUNCTION() {
>+    ABSL_TSAN_MUTEX_PRE_LOCK(this, 0);
>+    if (!TryLockImpl()) {
>+      SlowLock();
>+    }
>+    ABSL_TSAN_MUTEX_POST_LOCK(this, 0, 0);
>+  }
>+
>+  // Try to acquire this SpinLock without blocking and return true if the
>+  // acquisition was successful.  If the lock was not acquired, false is
>+  // returned.  If this SpinLock is free at the time of the call, TryLock
>+  // will return true with high probability.
>+  inline bool TryLock() EXCLUSIVE_TRYLOCK_FUNCTION(true) {
>+    ABSL_TSAN_MUTEX_PRE_LOCK(this, __tsan_mutex_try_lock);
>+    bool res = TryLockImpl();
>+    ABSL_TSAN_MUTEX_POST_LOCK(
>+        this, __tsan_mutex_try_lock | (res ? 0 : __tsan_mutex_try_lock_failed),
>+        0);
>+    return res;
>+  }
>+
>+  // Release this SpinLock, which must be held by the calling thread.
>+  inline void Unlock() UNLOCK_FUNCTION() {
>+    ABSL_TSAN_MUTEX_PRE_UNLOCK(this, 0);
>+    uint32_t lock_value = lockword_.load(std::memory_order_relaxed);
>+    lockword_.store(lock_value & kSpinLockCooperative,
>+                    std::memory_order_release);
>+
>+    if ((lock_value & kSpinLockDisabledScheduling) != 0) {
>+      base_internal::SchedulingGuard::EnableRescheduling(true);
>+    }
>+    if ((lock_value & kWaitTimeMask) != 0) {
>+      // Collect contentionz profile info, and speed the wakeup of any waiter.
>+      // The wait_cycles value indicates how long this thread spent waiting
>+      // for the lock.
>+      SlowUnlock(lock_value);
>+    }
>+    ABSL_TSAN_MUTEX_POST_UNLOCK(this, 0);
>+  }
>+
>+  // Determine if the lock is held.  When the lock is held by the invoking
>+  // thread, true will always be returned. Intended to be used as
>+  // CHECK(lock.IsHeld()).
>+  inline bool IsHeld() const {
>+    return (lockword_.load(std::memory_order_relaxed) & kSpinLockHeld) != 0;
>+  }
>+
>+ protected:
>+  // These should not be exported except for testing.
>+
>+  // Store number of cycles between wait_start_time and wait_end_time in a
>+  // lock value.
>+  static uint32_t EncodeWaitCycles(int64_t wait_start_time,
>+                                   int64_t wait_end_time);
>+
>+  // Extract number of wait cycles in a lock value.
>+  static uint64_t DecodeWaitCycles(uint32_t lock_value);
>+
>+  // Provide access to protected method above.  Use for testing only.
>+  friend struct SpinLockTest;
>+
>+ private:
>+  // lockword_ is used to store the following:
>+  //
>+  // bit[0] encodes whether a lock is being held.
>+  // bit[1] encodes whether a lock uses cooperative scheduling.
>+  // bit[2] encodes whether a lock disables scheduling.
>+  // bit[3:31] encodes time a lock spent on waiting as a 29-bit unsigned int.
>+  enum { kSpinLockHeld = 1 };
>+  enum { kSpinLockCooperative = 2 };
>+  enum { kSpinLockDisabledScheduling = 4 };
>+  enum { kSpinLockSleeper = 8 };
>+  enum { kWaitTimeMask =                      // Includes kSpinLockSleeper.
>+    ~(kSpinLockHeld | kSpinLockCooperative | kSpinLockDisabledScheduling) };
>+
>+  // Returns true if the provided scheduling mode is cooperative.
>+  static constexpr bool IsCooperative(
>+      base_internal::SchedulingMode scheduling_mode) {
>+    return scheduling_mode == base_internal::SCHEDULE_COOPERATIVE_AND_KERNEL;
>+  }
>+
>+  uint32_t TryLockInternal(uint32_t lock_value, uint32_t wait_cycles);
>+  void InitLinkerInitializedAndCooperative();
>+  void SlowLock() ABSL_ATTRIBUTE_COLD;
>+  void SlowUnlock(uint32_t lock_value) ABSL_ATTRIBUTE_COLD;
>+  uint32_t SpinLoop(int64_t initial_wait_timestamp, uint32_t* wait_cycles);
>+
>+  inline bool TryLockImpl() {
>+    uint32_t lock_value = lockword_.load(std::memory_order_relaxed);
>+    return (TryLockInternal(lock_value, 0) & kSpinLockHeld) == 0;
>+  }
>+
>+  std::atomic<uint32_t> lockword_;
>+
>+  SpinLock(const SpinLock&) = delete;
>+  SpinLock& operator=(const SpinLock&) = delete;
>+};
>+
>+// Corresponding locker object that arranges to acquire a spinlock for
>+// the duration of a C++ scope.
>+class SCOPED_LOCKABLE SpinLockHolder {
>+ public:
>+  inline explicit SpinLockHolder(SpinLock* l) EXCLUSIVE_LOCK_FUNCTION(l)
>+      : lock_(l) {
>+    l->Lock();
>+  }
>+  inline ~SpinLockHolder() UNLOCK_FUNCTION() { lock_->Unlock(); }
>+
>+  SpinLockHolder(const SpinLockHolder&) = delete;
>+  SpinLockHolder& operator=(const SpinLockHolder&) = delete;
>+
>+ private:
>+  SpinLock* lock_;
>+};
>+
>+// Register a hook for profiling support.
>+//
>+// The function pointer registered here will be called whenever a spinlock is
>+// contended.  The callback is given an opaque handle to the contended spinlock
>+// and the number of wait cycles.  This is thread-safe, but only a single
>+// profiler can be registered.  It is an error to call this function multiple
>+// times with different arguments.
>+void RegisterSpinLockProfiler(void (*fn)(const void* lock,
>+                                         int64_t wait_cycles));
>+
>+//------------------------------------------------------------------------------
>+// Public interface ends here.
>+//------------------------------------------------------------------------------
>+
>+// If (result & kSpinLockHeld) == 0, then *this was successfully locked.
>+// Otherwise, returns last observed value for lockword_.
>+inline uint32_t SpinLock::TryLockInternal(uint32_t lock_value,
>+                                          uint32_t wait_cycles) {
>+  if ((lock_value & kSpinLockHeld) != 0) {
>+    return lock_value;
>+  }
>+
>+  uint32_t sched_disabled_bit = 0;
>+  if ((lock_value & kSpinLockCooperative) == 0) {
>+    // For non-cooperative locks we must make sure we mark ourselves as
>+    // non-reschedulable before we attempt to CompareAndSwap.
>+    if (base_internal::SchedulingGuard::DisableRescheduling()) {
>+      sched_disabled_bit = kSpinLockDisabledScheduling;
>+    }
>+  }
>+
>+  if (lockword_.compare_exchange_strong(
>+          lock_value,
>+          kSpinLockHeld | lock_value | wait_cycles | sched_disabled_bit,
>+          std::memory_order_acquire, std::memory_order_relaxed)) {
>+  } else {
>+    base_internal::SchedulingGuard::EnableRescheduling(sched_disabled_bit != 0);
>+  }
>+
>+  return lock_value;
>+}
>+
>+}  // namespace base_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_BASE_INTERNAL_SPINLOCK_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/spinlock_akaros.inc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/spinlock_akaros.inc
>new file mode 100644
>index 00000000000..051c8cf87fb
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/spinlock_akaros.inc
>@@ -0,0 +1,35 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// This file is an Akaros-specific part of spinlock_wait.cc
>+
>+#include <atomic>
>+
>+#include "absl/base/internal/scheduling_mode.h"
>+
>+extern "C" {
>+
>+ABSL_ATTRIBUTE_WEAK void AbslInternalSpinLockDelay(
>+    std::atomic<uint32_t>* /* lock_word */, uint32_t /* value */,
>+    int /* loop */, absl::base_internal::SchedulingMode /* mode */) {
>+  // In Akaros, one must take care not to call anything that could cause a
>+  // malloc(), a blocking system call, or a uthread_yield() while holding a
>+  // spinlock. Our callers assume will not call into libraries or other
>+  // arbitrary code.
>+}
>+
>+ABSL_ATTRIBUTE_WEAK void AbslInternalSpinLockWake(
>+    std::atomic<uint32_t>* /* lock_word */, bool /* all */) {}
>+
>+}  // extern "C"
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/spinlock_posix.inc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/spinlock_posix.inc
>new file mode 100644
>index 00000000000..0098c1c7601
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/spinlock_posix.inc
>@@ -0,0 +1,46 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// This file is a Posix-specific part of spinlock_wait.cc
>+
>+#include <sched.h>
>+#include <atomic>
>+#include <ctime>
>+#include <cerrno>
>+
>+#include "absl/base/internal/scheduling_mode.h"
>+#include "absl/base/port.h"
>+
>+extern "C" {
>+
>+ABSL_ATTRIBUTE_WEAK void AbslInternalSpinLockDelay(
>+    std::atomic<uint32_t>* /* lock_word */, uint32_t /* value */, int loop,
>+    absl::base_internal::SchedulingMode /* mode */) {
>+  int save_errno = errno;
>+  if (loop == 0) {
>+  } else if (loop == 1) {
>+    sched_yield();
>+  } else {
>+    struct timespec tm;
>+    tm.tv_sec = 0;
>+    tm.tv_nsec = absl::base_internal::SpinLockSuggestedDelayNS(loop);
>+    nanosleep(&tm, nullptr);
>+  }
>+  errno = save_errno;
>+}
>+
>+ABSL_ATTRIBUTE_WEAK void AbslInternalSpinLockWake(
>+    std::atomic<uint32_t>* /* lock_word */, bool /* all */) {}
>+
>+}  // extern "C"
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/spinlock_wait.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/spinlock_wait.cc
>new file mode 100644
>index 00000000000..9f6e9911e10
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/spinlock_wait.cc
>@@ -0,0 +1,79 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+// The OS-specific header included below must provide two calls:
>+// base::subtle::SpinLockDelay() and base::subtle::SpinLockWake().
>+// See spinlock_wait.h for the specs.
>+
>+#include <atomic>
>+#include <cstdint>
>+
>+#include "absl/base/internal/spinlock_wait.h"
>+
>+#if defined(_WIN32)
>+#include "absl/base/internal/spinlock_win32.inc"
>+#elif defined(__akaros__)
>+#include "absl/base/internal/spinlock_akaros.inc"
>+#else
>+#include "absl/base/internal/spinlock_posix.inc"
>+#endif
>+
>+namespace absl {
>+namespace base_internal {
>+
>+// See spinlock_wait.h for spec.
>+uint32_t SpinLockWait(std::atomic<uint32_t> *w, int n,
>+                      const SpinLockWaitTransition trans[],
>+                      base_internal::SchedulingMode scheduling_mode) {
>+  for (int loop = 0; ; loop++) {
>+    uint32_t v = w->load(std::memory_order_acquire);
>+    int i;
>+    for (i = 0; i != n && v != trans[i].from; i++) {
>+    }
>+    if (i == n) {
>+      SpinLockDelay(w, v, loop, scheduling_mode);  // no matching transition
>+    } else if (trans[i].to == v ||                 // null transition
>+               w->compare_exchange_strong(v, trans[i].to,
>+                                          std::memory_order_acquire,
>+                                          std::memory_order_relaxed)) {
>+      if (trans[i].done) return v;
>+    }
>+  }
>+}
>+
>+static std::atomic<uint64_t> delay_rand;
>+
>+// Return a suggested delay in nanoseconds for iteration number "loop"
>+int SpinLockSuggestedDelayNS(int loop) {
>+  // Weak pseudo-random number generator to get some spread between threads
>+  // when many are spinning.
>+  uint64_t r = delay_rand.load(std::memory_order_relaxed);
>+  r = 0x5deece66dLL * r + 0xb;   // numbers from nrand48()
>+  delay_rand.store(r, std::memory_order_relaxed);
>+
>+  r <<= 16;   // 48-bit random number now in top 48-bits.
>+  if (loop < 0 || loop > 32) {   // limit loop to 0..32
>+    loop = 32;
>+  }
>+  // loop>>3 cannot exceed 4 because loop cannot exceed 32.
>+  // Select top 20..24 bits of lower 48 bits,
>+  // giving approximately 0ms to 16ms.
>+  // Mean is exponential in loop for first 32 iterations, then 8ms.
>+  // The futex path multiplies this by 16, since we expect explicit wakeups
>+  // almost always on that path.
>+  return static_cast<int>(r >> (44 - (loop >> 3)));
>+}
>+
>+}  // namespace base_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/spinlock_wait.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/spinlock_wait.h
>new file mode 100644
>index 00000000000..5c6cc7fdba4
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/spinlock_wait.h
>@@ -0,0 +1,91 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#ifndef ABSL_BASE_INTERNAL_SPINLOCK_WAIT_H_
>+#define ABSL_BASE_INTERNAL_SPINLOCK_WAIT_H_
>+
>+// Operations to make atomic transitions on a word, and to allow
>+// waiting for those transitions to become possible.
>+
>+#include <stdint.h>
>+#include <atomic>
>+
>+#include "absl/base/internal/scheduling_mode.h"
>+
>+namespace absl {
>+namespace base_internal {
>+
>+// SpinLockWait() waits until it can perform one of several transitions from
>+// "from" to "to".  It returns when it performs a transition where done==true.
>+struct SpinLockWaitTransition {
>+  uint32_t from;
>+  uint32_t to;
>+  bool done;
>+};
>+
>+// Wait until *w can transition from trans[i].from to trans[i].to for some i
>+// satisfying 0<=i<n && trans[i].done, atomically make the transition,
>+// then return the old value of *w.   Make any other atomic transitions
>+// where !trans[i].done, but continue waiting.
>+uint32_t SpinLockWait(std::atomic<uint32_t> *w, int n,
>+                      const SpinLockWaitTransition trans[],
>+                      SchedulingMode scheduling_mode);
>+
>+// If possible, wake some thread that has called SpinLockDelay(w, ...). If
>+// "all" is true, wake all such threads.  This call is a hint, and on some
>+// systems it may be a no-op; threads calling SpinLockDelay() will always wake
>+// eventually even if SpinLockWake() is never called.
>+void SpinLockWake(std::atomic<uint32_t> *w, bool all);
>+
>+// Wait for an appropriate spin delay on iteration "loop" of a
>+// spin loop on location *w, whose previously observed value was "value".
>+// SpinLockDelay() may do nothing, may yield the CPU, may sleep a clock tick,
>+// or may wait for a delay that can be truncated by a call to SpinLockWake(w).
>+// In all cases, it must return in bounded time even if SpinLockWake() is not
>+// called.
>+void SpinLockDelay(std::atomic<uint32_t> *w, uint32_t value, int loop,
>+                   base_internal::SchedulingMode scheduling_mode);
>+
>+// Helper used by AbslInternalSpinLockDelay.
>+// Returns a suggested delay in nanoseconds for iteration number "loop".
>+int SpinLockSuggestedDelayNS(int loop);
>+
>+}  // namespace base_internal
>+}  // namespace absl
>+
>+// In some build configurations we pass --detect-odr-violations to the
>+// gold linker.  This causes it to flag weak symbol overrides as ODR
>+// violations.  Because ODR only applies to C++ and not C,
>+// --detect-odr-violations ignores symbols not mangled with C++ names.
>+// By changing our extension points to be extern "C", we dodge this
>+// check.
>+extern "C" {
>+void AbslInternalSpinLockWake(std::atomic<uint32_t> *w, bool all);
>+void AbslInternalSpinLockDelay(
>+    std::atomic<uint32_t> *w, uint32_t value, int loop,
>+    absl::base_internal::SchedulingMode scheduling_mode);
>+}
>+
>+inline void absl::base_internal::SpinLockWake(std::atomic<uint32_t> *w,
>+                                              bool all) {
>+  AbslInternalSpinLockWake(w, all);
>+}
>+
>+inline void absl::base_internal::SpinLockDelay(
>+    std::atomic<uint32_t> *w, uint32_t value, int loop,
>+    absl::base_internal::SchedulingMode scheduling_mode) {
>+  AbslInternalSpinLockDelay(w, value, loop, scheduling_mode);
>+}
>+
>+#endif  // ABSL_BASE_INTERNAL_SPINLOCK_WAIT_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/spinlock_win32.inc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/spinlock_win32.inc
>new file mode 100644
>index 00000000000..32c8fc0bb51
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/spinlock_win32.inc
>@@ -0,0 +1,37 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// This file is a Win32-specific part of spinlock_wait.cc
>+
>+#include <windows.h>
>+#include <atomic>
>+#include "absl/base/internal/scheduling_mode.h"
>+
>+extern "C" {
>+
>+void AbslInternalSpinLockDelay(std::atomic<uint32_t>* /* lock_word */,
>+                               uint32_t /* value */, int loop,
>+                               absl::base_internal::SchedulingMode /* mode */) {
>+  if (loop == 0) {
>+  } else if (loop == 1) {
>+    Sleep(0);
>+  } else {
>+    Sleep(absl::base_internal::SpinLockSuggestedDelayNS(loop) / 1000000);
>+  }
>+}
>+
>+void AbslInternalSpinLockWake(std::atomic<uint32_t>* /* lock_word */,
>+                              bool /* all */) {}
>+
>+}  // extern "C"
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/sysinfo.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/sysinfo.cc
>new file mode 100644
>index 00000000000..db41bacc844
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/sysinfo.cc
>@@ -0,0 +1,404 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/base/internal/sysinfo.h"
>+
>+#include "absl/base/attributes.h"
>+
>+#ifdef _WIN32
>+#include <shlwapi.h>
>+#include <windows.h>
>+#else
>+#include <fcntl.h>
>+#include <pthread.h>
>+#include <sys/stat.h>
>+#include <sys/types.h>
>+#include <unistd.h>
>+#endif
>+
>+#ifdef __linux__
>+#include <sys/syscall.h>
>+#endif
>+
>+#if defined(__APPLE__) || defined(__FreeBSD__)
>+#include <sys/sysctl.h>
>+#endif
>+
>+#if defined(__myriad2__)
>+#include <rtems.h>
>+#endif
>+
>+#include <string.h>
>+#include <cassert>
>+#include <cstdint>
>+#include <cstdio>
>+#include <cstdlib>
>+#include <ctime>
>+#include <limits>
>+#include <thread>  // NOLINT(build/c++11)
>+#include <utility>
>+#include <vector>
>+
>+#include "absl/base/call_once.h"
>+#include "absl/base/internal/raw_logging.h"
>+#include "absl/base/internal/spinlock.h"
>+#include "absl/base/internal/unscaledcycleclock.h"
>+
>+namespace absl {
>+namespace base_internal {
>+
>+static once_flag init_system_info_once;
>+static int num_cpus = 0;
>+static double nominal_cpu_frequency = 1.0;  // 0.0 might be dangerous.
>+
>+static int GetNumCPUs() {
>+#if defined(__myriad2__)
>+  return 1;
>+#else
>+  // Other possibilities:
>+  //  - Read /sys/devices/system/cpu/online and use cpumask_parse()
>+  //  - sysconf(_SC_NPROCESSORS_ONLN)
>+  return std::thread::hardware_concurrency();
>+#endif
>+}
>+
>+#if defined(_WIN32)
>+
>+static double GetNominalCPUFrequency() {
>+  DWORD data;
>+  DWORD data_size = sizeof(data);
>+  #pragma comment(lib, "shlwapi.lib")  // For SHGetValue().
>+  if (SUCCEEDED(
>+          SHGetValueA(HKEY_LOCAL_MACHINE,
>+                      "HARDWARE\\DESCRIPTION\\System\\CentralProcessor\\0",
>+                      "~MHz", nullptr, &data, &data_size))) {
>+    return data * 1e6;  // Value is MHz.
>+  }
>+  return 1.0;
>+}
>+
>+#elif defined(CTL_HW) && defined(HW_CPU_FREQ)
>+
>+static double GetNominalCPUFrequency() {
>+  unsigned freq;
>+  size_t size = sizeof(freq);
>+  int mib[2] = {CTL_HW, HW_CPU_FREQ};
>+  if (sysctl(mib, 2, &freq, &size, nullptr, 0) == 0) {
>+    return static_cast<double>(freq);
>+  }
>+  return 1.0;
>+}
>+
>+#else
>+
>+// Helper function for reading a long from a file. Returns true if successful
>+// and the memory location pointed to by value is set to the value read.
>+static bool ReadLongFromFile(const char *file, long *value) {
>+  bool ret = false;
>+  int fd = open(file, O_RDONLY);
>+  if (fd != -1) {
>+    char line[1024];
>+    char *err;
>+    memset(line, '\0', sizeof(line));
>+    int len = read(fd, line, sizeof(line) - 1);
>+    if (len <= 0) {
>+      ret = false;
>+    } else {
>+      const long temp_value = strtol(line, &err, 10);
>+      if (line[0] != '\0' && (*err == '\n' || *err == '\0')) {
>+        *value = temp_value;
>+        ret = true;
>+      }
>+    }
>+    close(fd);
>+  }
>+  return ret;
>+}
>+
>+#if defined(ABSL_INTERNAL_UNSCALED_CYCLECLOCK_FREQUENCY_IS_CPU_FREQUENCY)
>+
>+// Reads a monotonic time source and returns a value in
>+// nanoseconds. The returned value uses an arbitrary epoch, not the
>+// Unix epoch.
>+static int64_t ReadMonotonicClockNanos() {
>+  struct timespec t;
>+#ifdef CLOCK_MONOTONIC_RAW
>+  int rc = clock_gettime(CLOCK_MONOTONIC_RAW, &t);
>+#else
>+  int rc = clock_gettime(CLOCK_MONOTONIC, &t);
>+#endif
>+  if (rc != 0) {
>+    perror("clock_gettime() failed");
>+    abort();
>+  }
>+  return int64_t{t.tv_sec} * 1000000000 + t.tv_nsec;
>+}
>+
>+class UnscaledCycleClockWrapperForInitializeFrequency {
>+ public:
>+  static int64_t Now() { return base_internal::UnscaledCycleClock::Now(); }
>+};
>+
>+struct TimeTscPair {
>+  int64_t time;  // From ReadMonotonicClockNanos().
>+  int64_t tsc;   // From UnscaledCycleClock::Now().
>+};
>+
>+// Returns a pair of values (monotonic kernel time, TSC ticks) that
>+// approximately correspond to each other.  This is accomplished by
>+// doing several reads and picking the reading with the lowest
>+// latency.  This approach is used to minimize the probability that
>+// our thread was preempted between clock reads.
>+static TimeTscPair GetTimeTscPair() {
>+  int64_t best_latency = std::numeric_limits<int64_t>::max();
>+  TimeTscPair best;
>+  for (int i = 0; i < 10; ++i) {
>+    int64_t t0 = ReadMonotonicClockNanos();
>+    int64_t tsc = UnscaledCycleClockWrapperForInitializeFrequency::Now();
>+    int64_t t1 = ReadMonotonicClockNanos();
>+    int64_t latency = t1 - t0;
>+    if (latency < best_latency) {
>+      best_latency = latency;
>+      best.time = t0;
>+      best.tsc = tsc;
>+    }
>+  }
>+  return best;
>+}
>+
>+// Measures and returns the TSC frequency by taking a pair of
>+// measurements approximately `sleep_nanoseconds` apart.
>+static double MeasureTscFrequencyWithSleep(int sleep_nanoseconds) {
>+  auto t0 = GetTimeTscPair();
>+  struct timespec ts;
>+  ts.tv_sec = 0;
>+  ts.tv_nsec = sleep_nanoseconds;
>+  while (nanosleep(&ts, &ts) != 0 && errno == EINTR) {}
>+  auto t1 = GetTimeTscPair();
>+  double elapsed_ticks = t1.tsc - t0.tsc;
>+  double elapsed_time = (t1.time - t0.time) * 1e-9;
>+  return elapsed_ticks / elapsed_time;
>+}
>+
>+// Measures and returns the TSC frequency by calling
>+// MeasureTscFrequencyWithSleep(), doubling the sleep interval until the
>+// frequency measurement stabilizes.
>+static double MeasureTscFrequency() {
>+  double last_measurement = -1.0;
>+  int sleep_nanoseconds = 1000000;  // 1 millisecond.
>+  for (int i = 0; i < 8; ++i) {
>+    double measurement = MeasureTscFrequencyWithSleep(sleep_nanoseconds);
>+    if (measurement * 0.99 < last_measurement &&
>+        last_measurement < measurement * 1.01) {
>+      // Use the current measurement if it is within 1% of the
>+      // previous measurement.
>+      return measurement;
>+    }
>+    last_measurement = measurement;
>+    sleep_nanoseconds *= 2;
>+  }
>+  return last_measurement;
>+}
>+
>+#endif  // ABSL_INTERNAL_UNSCALED_CYCLECLOCK_FREQUENCY_IS_CPU_FREQUENCY
>+
>+static double GetNominalCPUFrequency() {
>+  long freq = 0;
>+
>+  // Google's production kernel has a patch to export the TSC
>+  // frequency through sysfs. If the kernel is exporting the TSC
>+  // frequency use that. There are issues where cpuinfo_max_freq
>+  // cannot be relied on because the BIOS may be exporting an invalid
>+  // p-state (on x86) or p-states may be used to put the processor in
>+  // a new mode (turbo mode). Essentially, those frequencies cannot
>+  // always be relied upon. The same reasons apply to /proc/cpuinfo as
>+  // well.
>+  if (ReadLongFromFile("/sys/devices/system/cpu/cpu0/tsc_freq_khz", &freq)) {
>+    return freq * 1e3;  // Value is kHz.
>+  }
>+
>+#if defined(ABSL_INTERNAL_UNSCALED_CYCLECLOCK_FREQUENCY_IS_CPU_FREQUENCY)
>+  // On these platforms, the TSC frequency is the nominal CPU
>+  // frequency.  But without having the kernel export it directly
>+  // though /sys/devices/system/cpu/cpu0/tsc_freq_khz, there is no
>+  // other way to reliably get the TSC frequency, so we have to
>+  // measure it ourselves.  Some CPUs abuse cpuinfo_max_freq by
>+  // exporting "fake" frequencies for implementing new features. For
>+  // example, Intel's turbo mode is enabled by exposing a p-state
>+  // value with a higher frequency than that of the real TSC
>+  // rate. Because of this, we prefer to measure the TSC rate
>+  // ourselves on i386 and x86-64.
>+  return MeasureTscFrequency();
>+#else
>+
>+  // If CPU scaling is in effect, we want to use the *maximum*
>+  // frequency, not whatever CPU speed some random processor happens
>+  // to be using now.
>+  if (ReadLongFromFile("/sys/devices/system/cpu/cpu0/cpufreq/cpuinfo_max_freq",
>+                       &freq)) {
>+    return freq * 1e3;  // Value is kHz.
>+  }
>+
>+  return 1.0;
>+#endif  // !ABSL_INTERNAL_UNSCALED_CYCLECLOCK_FREQUENCY_IS_CPU_FREQUENCY
>+}
>+
>+#endif
>+
>+// InitializeSystemInfo() may be called before main() and before
>+// malloc is properly initialized, therefore this must not allocate
>+// memory.
>+static void InitializeSystemInfo() {
>+  num_cpus = GetNumCPUs();
>+  nominal_cpu_frequency = GetNominalCPUFrequency();
>+}
>+
>+int NumCPUs() {
>+  base_internal::LowLevelCallOnce(&init_system_info_once, InitializeSystemInfo);
>+  return num_cpus;
>+}
>+
>+double NominalCPUFrequency() {
>+  base_internal::LowLevelCallOnce(&init_system_info_once, InitializeSystemInfo);
>+  return nominal_cpu_frequency;
>+}
>+
>+#if defined(_WIN32)
>+
>+pid_t GetTID() {
>+  return GetCurrentThreadId();
>+}
>+
>+#elif defined(__linux__)
>+
>+#ifndef SYS_gettid
>+#define SYS_gettid __NR_gettid
>+#endif
>+
>+pid_t GetTID() {
>+  return syscall(SYS_gettid);
>+}
>+
>+#elif defined(__akaros__)
>+
>+pid_t GetTID() {
>+  // Akaros has a concept of "vcore context", which is the state the program
>+  // is forced into when we need to make a user-level scheduling decision, or
>+  // run a signal handler.  This is analogous to the interrupt context that a
>+  // CPU might enter if it encounters some kind of exception.
>+  //
>+  // There is no current thread context in vcore context, but we need to give
>+  // a reasonable answer if asked for a thread ID (e.g., in a signal handler).
>+  // Thread 0 always exists, so if we are in vcore context, we return that.
>+  //
>+  // Otherwise, we know (since we are using pthreads) that the uthread struct
>+  // current_uthread is pointing to is the first element of a
>+  // struct pthread_tcb, so we extract and return the thread ID from that.
>+  //
>+  // TODO(dcross): Akaros anticipates moving the thread ID to the uthread
>+  // structure at some point. We should modify this code to remove the cast
>+  // when that happens.
>+  if (in_vcore_context())
>+    return 0;
>+  return reinterpret_cast<struct pthread_tcb *>(current_uthread)->id;
>+}
>+
>+#elif defined(__myriad2__)
>+
>+pid_t GetTID() {
>+  uint32_t tid;
>+  rtems_task_ident(RTEMS_SELF, 0, &tid);
>+  return tid;
>+}
>+
>+#else
>+
>+// Fallback implementation of GetTID using pthread_getspecific.
>+static once_flag tid_once;
>+static pthread_key_t tid_key;
>+static absl::base_internal::SpinLock tid_lock(
>+    absl::base_internal::kLinkerInitialized);
>+
>+// We set a bit per thread in this array to indicate that an ID is in
>+// use. ID 0 is unused because it is the default value returned by
>+// pthread_getspecific().
>+static std::vector<uint32_t>* tid_array GUARDED_BY(tid_lock) = nullptr;
>+static constexpr int kBitsPerWord = 32;  // tid_array is uint32_t.
>+
>+// Returns the TID to tid_array.
>+static void FreeTID(void *v) {
>+  intptr_t tid = reinterpret_cast<intptr_t>(v);
>+  int word = tid / kBitsPerWord;
>+  uint32_t mask = ~(1u << (tid % kBitsPerWord));
>+  absl::base_internal::SpinLockHolder lock(&tid_lock);
>+  assert(0 <= word && static_cast<size_t>(word) < tid_array->size());
>+  (*tid_array)[word] &= mask;
>+}
>+
>+static void InitGetTID() {
>+  if (pthread_key_create(&tid_key, FreeTID) != 0) {
>+    // The logging system calls GetTID() so it can't be used here.
>+    perror("pthread_key_create failed");
>+    abort();
>+  }
>+
>+  // Initialize tid_array.
>+  absl::base_internal::SpinLockHolder lock(&tid_lock);
>+  tid_array = new std::vector<uint32_t>(1);
>+  (*tid_array)[0] = 1;  // ID 0 is never-allocated.
>+}
>+
>+// Return a per-thread small integer ID from pthread's thread-specific data.
>+pid_t GetTID() {
>+  absl::call_once(tid_once, InitGetTID);
>+
>+  intptr_t tid = reinterpret_cast<intptr_t>(pthread_getspecific(tid_key));
>+  if (tid != 0) {
>+    return tid;
>+  }
>+
>+  int bit;  // tid_array[word] = 1u << bit;
>+  size_t word;
>+  {
>+    // Search for the first unused ID.
>+    absl::base_internal::SpinLockHolder lock(&tid_lock);
>+    // First search for a word in the array that is not all ones.
>+    word = 0;
>+    while (word < tid_array->size() && ~(*tid_array)[word] == 0) {
>+      ++word;
>+    }
>+    if (word == tid_array->size()) {
>+      tid_array->push_back(0);  // No space left, add kBitsPerWord more IDs.
>+    }
>+    // Search for a zero bit in the word.
>+    bit = 0;
>+    while (bit < kBitsPerWord && (((*tid_array)[word] >> bit) & 1) != 0) {
>+      ++bit;
>+    }
>+    tid = (word * kBitsPerWord) + bit;
>+    (*tid_array)[word] |= 1u << bit;  // Mark the TID as allocated.
>+  }
>+
>+  if (pthread_setspecific(tid_key, reinterpret_cast<void *>(tid)) != 0) {
>+    perror("pthread_setspecific failed");
>+    abort();
>+  }
>+
>+  return static_cast<pid_t>(tid);
>+}
>+
>+#endif
>+
>+}  // namespace base_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/sysinfo.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/sysinfo.h
>new file mode 100644
>index 00000000000..5bd1c500bd5
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/sysinfo.h
>@@ -0,0 +1,63 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// This file includes routines to find out characteristics
>+// of the machine a program is running on.  It is undoubtedly
>+// system-dependent.
>+
>+// Functions listed here that accept a pid_t as an argument act on the
>+// current process if the pid_t argument is 0
>+// All functions here are thread-hostile due to file caching unless
>+// commented otherwise.
>+
>+#ifndef ABSL_BASE_INTERNAL_SYSINFO_H_
>+#define ABSL_BASE_INTERNAL_SYSINFO_H_
>+
>+#ifndef _WIN32
>+#include <sys/types.h>
>+#else
>+#include <intsafe.h>
>+#endif
>+
>+#include "absl/base/port.h"
>+
>+namespace absl {
>+namespace base_internal {
>+
>+// Nominal core processor cycles per second of each processor.   This is _not_
>+// necessarily the frequency of the CycleClock counter (see cycleclock.h)
>+// Thread-safe.
>+double NominalCPUFrequency();
>+
>+// Number of logical processors (hyperthreads) in system. Thread-safe.
>+int NumCPUs();
>+
>+// Return the thread id of the current thread, as told by the system.
>+// No two currently-live threads implemented by the OS shall have the same ID.
>+// Thread ids of exited threads may be reused.   Multiple user-level threads
>+// may have the same thread ID if multiplexed on the same OS thread.
>+//
>+// On Linux, you may send a signal to the resulting ID with kill().  However,
>+// it is recommended for portability that you use pthread_kill() instead.
>+#ifdef _WIN32
>+// On Windows, process id and thread id are of the same type according to
>+// the return types of GetProcessId() and GetThreadId() are both DWORD.
>+using pid_t = DWORD;
>+#endif
>+pid_t GetTID();
>+
>+}  // namespace base_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_BASE_INTERNAL_SYSINFO_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/sysinfo_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/sysinfo_test.cc
>new file mode 100644
>index 00000000000..e0d9aab9bc2
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/sysinfo_test.cc
>@@ -0,0 +1,98 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/base/internal/sysinfo.h"
>+
>+#ifndef _WIN32
>+#include <sys/types.h>
>+#include <unistd.h>
>+#endif
>+
>+#include <thread>  // NOLINT(build/c++11)
>+#include <unordered_set>
>+#include <vector>
>+
>+#include "gtest/gtest.h"
>+#include "absl/synchronization/barrier.h"
>+#include "absl/synchronization/mutex.h"
>+
>+namespace absl {
>+namespace base_internal {
>+namespace {
>+
>+TEST(SysinfoTest, NumCPUs) {
>+  EXPECT_NE(NumCPUs(), 0)
>+      << "NumCPUs() should not have the default value of 0";
>+}
>+
>+TEST(SysinfoTest, NominalCPUFrequency) {
>+#if !(defined(__aarch64__) && defined(__linux__))
>+  EXPECT_GE(NominalCPUFrequency(), 1000.0)
>+      << "NominalCPUFrequency() did not return a reasonable value";
>+#else
>+  // TODO(absl-team): Aarch64 cannot read the CPU frequency from sysfs, so we
>+  // get back 1.0. Fix once the value is available.
>+  EXPECT_EQ(NominalCPUFrequency(), 1.0)
>+      << "CPU frequency detection was fixed! Please update unittest.";
>+#endif
>+}
>+
>+TEST(SysinfoTest, GetTID) {
>+  EXPECT_EQ(GetTID(), GetTID());  // Basic compile and equality test.
>+#ifdef __native_client__
>+  // Native Client has a race condition bug that leads to memory
>+  // exaustion when repeatedly creating and joining threads.
>+  // https://bugs.chromium.org/p/nativeclient/issues/detail?id=1027
>+  return;
>+#endif
>+  // Test that TIDs are unique to each thread.
>+  // Uses a few loops to exercise implementations that reallocate IDs.
>+  for (int i = 0; i < 32; ++i) {
>+    constexpr int kNumThreads = 64;
>+    Barrier all_threads_done(kNumThreads);
>+    std::vector<std::thread> threads;
>+
>+    Mutex mutex;
>+    std::unordered_set<pid_t> tids;
>+
>+    for (int j = 0; j < kNumThreads; ++j) {
>+      threads.push_back(std::thread([&]() {
>+        pid_t id = GetTID();
>+        {
>+          MutexLock lock(&mutex);
>+          ASSERT_TRUE(tids.find(id) == tids.end());
>+          tids.insert(id);
>+        }
>+        // We can't simply join the threads here. The threads need to
>+        // be alive otherwise the TID might have been reallocated to
>+        // another live thread.
>+        all_threads_done.Block();
>+      }));
>+    }
>+    for (auto& thread : threads) {
>+      thread.join();
>+    }
>+  }
>+}
>+
>+#ifdef __linux__
>+TEST(SysinfoTest, LinuxGetTID) {
>+  // On Linux, for the main thread, GetTID()==getpid() is guaranteed by the API.
>+  EXPECT_EQ(GetTID(), getpid());
>+}
>+#endif
>+
>+}  // namespace
>+}  // namespace base_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/thread_identity.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/thread_identity.cc
>new file mode 100644
>index 00000000000..678e8568d74
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/thread_identity.cc
>@@ -0,0 +1,123 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/base/internal/thread_identity.h"
>+
>+#ifndef _WIN32
>+#include <pthread.h>
>+#include <signal.h>
>+#endif
>+
>+#include <atomic>
>+#include <cassert>
>+#include <memory>
>+
>+#include "absl/base/call_once.h"
>+#include "absl/base/internal/raw_logging.h"
>+#include "absl/base/internal/spinlock.h"
>+
>+namespace absl {
>+namespace base_internal {
>+
>+#if ABSL_THREAD_IDENTITY_MODE != ABSL_THREAD_IDENTITY_MODE_USE_CPP11
>+namespace {
>+// Used to co-ordinate one-time creation of our pthread_key
>+absl::once_flag init_thread_identity_key_once;
>+pthread_key_t thread_identity_pthread_key;
>+std::atomic<bool> pthread_key_initialized(false);
>+
>+void AllocateThreadIdentityKey(ThreadIdentityReclaimerFunction reclaimer) {
>+  pthread_key_create(&thread_identity_pthread_key, reclaimer);
>+  pthread_key_initialized.store(true, std::memory_order_release);
>+}
>+}  // namespace
>+#endif
>+
>+#if ABSL_THREAD_IDENTITY_MODE == ABSL_THREAD_IDENTITY_MODE_USE_TLS || \
>+    ABSL_THREAD_IDENTITY_MODE == ABSL_THREAD_IDENTITY_MODE_USE_CPP11
>+// The actual TLS storage for a thread's currently associated ThreadIdentity.
>+// This is referenced by inline accessors in the header.
>+// "protected" visibility ensures that if multiple instances of Abseil code
>+// exist within a process (via dlopen() or similar), references to
>+// thread_identity_ptr from each instance of the code will refer to
>+// *different* instances of this ptr.
>+#ifdef __GNUC__
>+__attribute__((visibility("protected")))
>+#endif  // __GNUC__
>+  ABSL_PER_THREAD_TLS_KEYWORD ThreadIdentity* thread_identity_ptr;
>+#endif  // TLS or CPP11
>+
>+void SetCurrentThreadIdentity(
>+    ThreadIdentity* identity, ThreadIdentityReclaimerFunction reclaimer) {
>+  assert(CurrentThreadIdentityIfPresent() == nullptr);
>+  // Associate our destructor.
>+  // NOTE: This call to pthread_setspecific is currently the only immovable
>+  // barrier to CurrentThreadIdentity() always being async signal safe.
>+#if ABSL_THREAD_IDENTITY_MODE == ABSL_THREAD_IDENTITY_MODE_USE_POSIX_SETSPECIFIC
>+  // NOTE: Not async-safe.  But can be open-coded.
>+  absl::call_once(init_thread_identity_key_once, AllocateThreadIdentityKey,
>+                  reclaimer);
>+  // We must mask signals around the call to setspecific as with current glibc,
>+  // a concurrent getspecific (needed for GetCurrentThreadIdentityIfPresent())
>+  // may zero our value.
>+  //
>+  // While not officially async-signal safe, getspecific within a signal handler
>+  // is otherwise OK.
>+  sigset_t all_signals;
>+  sigset_t curr_signals;
>+  sigfillset(&all_signals);
>+  pthread_sigmask(SIG_SETMASK, &all_signals, &curr_signals);
>+  pthread_setspecific(thread_identity_pthread_key,
>+                      reinterpret_cast<void*>(identity));
>+  pthread_sigmask(SIG_SETMASK, &curr_signals, nullptr);
>+#elif ABSL_THREAD_IDENTITY_MODE == ABSL_THREAD_IDENTITY_MODE_USE_TLS
>+  // NOTE: Not async-safe.  But can be open-coded.
>+  absl::call_once(init_thread_identity_key_once, AllocateThreadIdentityKey,
>+                  reclaimer);
>+  pthread_setspecific(thread_identity_pthread_key,
>+                      reinterpret_cast<void*>(identity));
>+  thread_identity_ptr = identity;
>+#elif ABSL_THREAD_IDENTITY_MODE == ABSL_THREAD_IDENTITY_MODE_USE_CPP11
>+  thread_local std::unique_ptr<ThreadIdentity, ThreadIdentityReclaimerFunction>
>+      holder(identity, reclaimer);
>+  thread_identity_ptr = identity;
>+#else
>+#error Unimplemented ABSL_THREAD_IDENTITY_MODE
>+#endif
>+}
>+
>+void ClearCurrentThreadIdentity() {
>+#if ABSL_THREAD_IDENTITY_MODE == ABSL_THREAD_IDENTITY_MODE_USE_TLS || \
>+    ABSL_THREAD_IDENTITY_MODE == ABSL_THREAD_IDENTITY_MODE_USE_CPP11
>+  thread_identity_ptr = nullptr;
>+#elif ABSL_THREAD_IDENTITY_MODE == \
>+      ABSL_THREAD_IDENTITY_MODE_USE_POSIX_SETSPECIFIC
>+  // pthread_setspecific expected to clear value on destruction
>+  assert(CurrentThreadIdentityIfPresent() == nullptr);
>+#endif
>+}
>+
>+#if ABSL_THREAD_IDENTITY_MODE == ABSL_THREAD_IDENTITY_MODE_USE_POSIX_SETSPECIFIC
>+ThreadIdentity* CurrentThreadIdentityIfPresent() {
>+  bool initialized = pthread_key_initialized.load(std::memory_order_acquire);
>+  if (!initialized) {
>+    return nullptr;
>+  }
>+  return reinterpret_cast<ThreadIdentity*>(
>+      pthread_getspecific(thread_identity_pthread_key));
>+}
>+#endif
>+
>+}  // namespace base_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/thread_identity.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/thread_identity.h
>new file mode 100644
>index 00000000000..a51722f9d82
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/thread_identity.h
>@@ -0,0 +1,240 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// Each active thread has an ThreadIdentity that may represent the thread in
>+// various level interfaces.  ThreadIdentity objects are never deallocated.
>+// When a thread terminates, its ThreadIdentity object may be reused for a
>+// thread created later.
>+
>+#ifndef ABSL_BASE_INTERNAL_THREAD_IDENTITY_H_
>+#define ABSL_BASE_INTERNAL_THREAD_IDENTITY_H_
>+
>+#ifndef _WIN32
>+#include <pthread.h>
>+// Defines __GOOGLE_GRTE_VERSION__ (via glibc-specific features.h) when
>+// supported.
>+#include <unistd.h>
>+#endif
>+
>+#include <atomic>
>+#include <cstdint>
>+
>+#include "absl/base/internal/per_thread_tls.h"
>+
>+namespace absl {
>+
>+struct SynchLocksHeld;
>+struct SynchWaitParams;
>+
>+namespace base_internal {
>+
>+class SpinLock;
>+struct ThreadIdentity;
>+
>+// Used by the implementation of base::Mutex and base::CondVar.
>+struct PerThreadSynch {
>+  // The internal representation of base::Mutex and base::CondVar rely
>+  // on the alignment of PerThreadSynch. Both store the address of the
>+  // PerThreadSynch in the high-order bits of their internal state,
>+  // which means the low kLowZeroBits of the address of PerThreadSynch
>+  // must be zero.
>+  static constexpr int kLowZeroBits = 8;
>+  static constexpr int kAlignment = 1 << kLowZeroBits;
>+
>+  // Returns the associated ThreadIdentity.
>+  // This can be implemented as a cast because we guarantee
>+  // PerThreadSynch is the first element of ThreadIdentity.
>+  ThreadIdentity* thread_identity() {
>+    return reinterpret_cast<ThreadIdentity*>(this);
>+  }
>+
>+  PerThreadSynch *next;  // Circular waiter queue; initialized to 0.
>+  PerThreadSynch *skip;  // If non-zero, all entries in Mutex queue
>+                         // up to and including "skip" have same
>+                         // condition as this, and will be woken later
>+  bool may_skip;         // if false while on mutex queue, a mutex unlocker
>+                         // is using this PerThreadSynch as a terminator.  Its
>+                         // skip field must not be filled in because the loop
>+                         // might then skip over the terminator.
>+
>+  // The wait parameters of the current wait.  waitp is null if the
>+  // thread is not waiting. Transitions from null to non-null must
>+  // occur before the enqueue commit point (state = kQueued in
>+  // Enqueue() and CondVarEnqueue()). Transitions from non-null to
>+  // null must occur after the wait is finished (state = kAvailable in
>+  // Mutex::Block() and CondVar::WaitCommon()). This field may be
>+  // changed only by the thread that describes this PerThreadSynch.  A
>+  // special case is Fer(), which calls Enqueue() on another thread,
>+  // but with an identical SynchWaitParams pointer, thus leaving the
>+  // pointer unchanged.
>+  SynchWaitParams *waitp;
>+
>+  bool suppress_fatal_errors;  // If true, try to proceed even in the face of
>+                               // broken invariants.  This is used within fatal
>+                               // signal handlers to improve the chances of
>+                               // debug logging information being output
>+                               // successfully.
>+
>+  intptr_t readers;     // Number of readers in mutex.
>+  int priority;         // Priority of thread (updated every so often).
>+
>+  // When priority will next be read (cycles).
>+  int64_t next_priority_read_cycles;
>+
>+  // State values:
>+  //   kAvailable: This PerThreadSynch is available.
>+  //   kQueued: This PerThreadSynch is unavailable, it's currently queued on a
>+  //            Mutex or CondVar waistlist.
>+  //
>+  // Transitions from kQueued to kAvailable require a release
>+  // barrier. This is needed as a waiter may use "state" to
>+  // independently observe that it's no longer queued.
>+  //
>+  // Transitions from kAvailable to kQueued require no barrier, they
>+  // are externally ordered by the Mutex.
>+  enum State {
>+    kAvailable,
>+    kQueued
>+  };
>+  std::atomic<State> state;
>+
>+  bool maybe_unlocking;  // Valid at head of Mutex waiter queue;
>+                         // true if UnlockSlow could be searching
>+                         // for a waiter to wake.  Used for an optimization
>+                         // in Enqueue().  true is always a valid value.
>+                         // Can be reset to false when the unlocker or any
>+                         // writer releases the lock, or a reader fully releases
>+                         // the lock.  It may not be set to false by a reader
>+                         // that decrements the count to non-zero.
>+                         // protected by mutex spinlock
>+
>+  bool wake;  // This thread is to be woken from a Mutex.
>+
>+  // If "x" is on a waiter list for a mutex, "x->cond_waiter" is true iff the
>+  // waiter is waiting on the mutex as part of a CV Wait or Mutex Await.
>+  //
>+  // The value of "x->cond_waiter" is meaningless if "x" is not on a
>+  // Mutex waiter list.
>+  bool cond_waiter;
>+
>+  // Locks held; used during deadlock detection.
>+  // Allocated in Synch_GetAllLocks() and freed in ReclaimThreadIdentity().
>+  SynchLocksHeld *all_locks;
>+};
>+
>+struct ThreadIdentity {
>+  // Must be the first member.  The Mutex implementation requires that
>+  // the PerThreadSynch object associated with each thread is
>+  // PerThreadSynch::kAlignment aligned.  We provide this alignment on
>+  // ThreadIdentity itself.
>+  PerThreadSynch per_thread_synch;
>+
>+  // Private: Reserved for absl::synchronization_internal::Waiter.
>+  struct WaiterState {
>+    char data[128];
>+  } waiter_state;
>+
>+  // Used by PerThreadSem::{Get,Set}ThreadBlockedCounter().
>+  std::atomic<int>* blocked_count_ptr;
>+
>+  // The following variables are mostly read/written just by the
>+  // thread itself.  The only exception is that these are read by
>+  // a ticker thread as a hint.
>+  std::atomic<int> ticker;      // Tick counter, incremented once per second.
>+  std::atomic<int> wait_start;  // Ticker value when thread started waiting.
>+  std::atomic<bool> is_idle;    // Has thread become idle yet?
>+
>+  ThreadIdentity* next;
>+};
>+
>+// Returns the ThreadIdentity object representing the calling thread; guaranteed
>+// to be unique for its lifetime.  The returned object will remain valid for the
>+// program's lifetime; although it may be re-assigned to a subsequent thread.
>+// If one does not exist, return nullptr instead.
>+//
>+// Does not malloc(*), and is async-signal safe.
>+// [*] Technically pthread_setspecific() does malloc on first use; however this
>+// is handled internally within tcmalloc's initialization already.
>+//
>+// New ThreadIdentity objects can be constructed and associated with a thread
>+// by calling GetOrCreateCurrentThreadIdentity() in per-thread-sem.h.
>+ThreadIdentity* CurrentThreadIdentityIfPresent();
>+
>+using ThreadIdentityReclaimerFunction = void (*)(void*);
>+
>+// Sets the current thread identity to the given value.  'reclaimer' is a
>+// pointer to the global function for cleaning up instances on thread
>+// destruction.
>+void SetCurrentThreadIdentity(ThreadIdentity* identity,
>+                              ThreadIdentityReclaimerFunction reclaimer);
>+
>+// Removes the currently associated ThreadIdentity from the running thread.
>+// This must be called from inside the ThreadIdentityReclaimerFunction, and only
>+// from that function.
>+void ClearCurrentThreadIdentity();
>+
>+// May be chosen at compile time via: -DABSL_FORCE_THREAD_IDENTITY_MODE=<mode
>+// index>
>+#ifdef ABSL_THREAD_IDENTITY_MODE_USE_POSIX_SETSPECIFIC
>+#error ABSL_THREAD_IDENTITY_MODE_USE_POSIX_SETSPECIFIC cannot be direcly set
>+#else
>+#define ABSL_THREAD_IDENTITY_MODE_USE_POSIX_SETSPECIFIC 0
>+#endif
>+
>+#ifdef ABSL_THREAD_IDENTITY_MODE_USE_TLS
>+#error ABSL_THREAD_IDENTITY_MODE_USE_TLS cannot be direcly set
>+#else
>+#define ABSL_THREAD_IDENTITY_MODE_USE_TLS 1
>+#endif
>+
>+#ifdef ABSL_THREAD_IDENTITY_MODE_USE_CPP11
>+#error ABSL_THREAD_IDENTITY_MODE_USE_CPP11 cannot be direcly set
>+#else
>+#define ABSL_THREAD_IDENTITY_MODE_USE_CPP11 2
>+#endif
>+
>+#ifdef ABSL_THREAD_IDENTITY_MODE
>+#error ABSL_THREAD_IDENTITY_MODE cannot be direcly set
>+#elif defined(ABSL_FORCE_THREAD_IDENTITY_MODE)
>+#define ABSL_THREAD_IDENTITY_MODE ABSL_FORCE_THREAD_IDENTITY_MODE
>+#elif defined(_WIN32)
>+#define ABSL_THREAD_IDENTITY_MODE ABSL_THREAD_IDENTITY_MODE_USE_CPP11
>+#elif ABSL_PER_THREAD_TLS && defined(__GOOGLE_GRTE_VERSION__) && \
>+    (__GOOGLE_GRTE_VERSION__ >= 20140228L)
>+// Support for async-safe TLS was specifically added in GRTEv4.  It's not
>+// present in the upstream eglibc.
>+// Note:  Current default for production systems.
>+#define ABSL_THREAD_IDENTITY_MODE ABSL_THREAD_IDENTITY_MODE_USE_TLS
>+#else
>+#define ABSL_THREAD_IDENTITY_MODE \
>+  ABSL_THREAD_IDENTITY_MODE_USE_POSIX_SETSPECIFIC
>+#endif
>+
>+#if ABSL_THREAD_IDENTITY_MODE == ABSL_THREAD_IDENTITY_MODE_USE_TLS || \
>+    ABSL_THREAD_IDENTITY_MODE == ABSL_THREAD_IDENTITY_MODE_USE_CPP11
>+
>+extern ABSL_PER_THREAD_TLS_KEYWORD ThreadIdentity* thread_identity_ptr;
>+
>+inline ThreadIdentity* CurrentThreadIdentityIfPresent() {
>+  return thread_identity_ptr;
>+}
>+
>+#elif ABSL_THREAD_IDENTITY_MODE != \
>+    ABSL_THREAD_IDENTITY_MODE_USE_POSIX_SETSPECIFIC
>+#error Unknown ABSL_THREAD_IDENTITY_MODE
>+#endif
>+
>+}  // namespace base_internal
>+}  // namespace absl
>+#endif  // ABSL_BASE_INTERNAL_THREAD_IDENTITY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/thread_identity_benchmark.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/thread_identity_benchmark.cc
>new file mode 100644
>index 00000000000..242522b4418
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/thread_identity_benchmark.cc
>@@ -0,0 +1,38 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "benchmark/benchmark.h"
>+#include "absl/base/internal/thread_identity.h"
>+#include "absl/synchronization/internal/create_thread_identity.h"
>+#include "absl/synchronization/internal/per_thread_sem.h"
>+
>+namespace {
>+
>+void BM_SafeCurrentThreadIdentity(benchmark::State& state) {
>+  for (auto _ : state) {
>+    benchmark::DoNotOptimize(
>+        absl::synchronization_internal::GetOrCreateCurrentThreadIdentity());
>+  }
>+}
>+BENCHMARK(BM_SafeCurrentThreadIdentity);
>+
>+void BM_UnsafeCurrentThreadIdentity(benchmark::State& state) {
>+  for (auto _ : state) {
>+    benchmark::DoNotOptimize(
>+        absl::base_internal::CurrentThreadIdentityIfPresent());
>+  }
>+}
>+BENCHMARK(BM_UnsafeCurrentThreadIdentity);
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/thread_identity_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/thread_identity_test.cc
>new file mode 100644
>index 00000000000..ecb8af68982
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/thread_identity_test.cc
>@@ -0,0 +1,126 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/base/internal/thread_identity.h"
>+
>+#include <thread>  // NOLINT(build/c++11)
>+#include <vector>
>+
>+#include "gtest/gtest.h"
>+#include "absl/base/attributes.h"
>+#include "absl/base/internal/spinlock.h"
>+#include "absl/base/macros.h"
>+#include "absl/synchronization/internal/per_thread_sem.h"
>+#include "absl/synchronization/mutex.h"
>+
>+namespace absl {
>+namespace base_internal {
>+namespace {
>+
>+// protects num_identities_reused
>+static absl::base_internal::SpinLock map_lock(
>+    absl::base_internal::kLinkerInitialized);
>+static int num_identities_reused;
>+
>+static const void* const kCheckNoIdentity = reinterpret_cast<void*>(1);
>+
>+static void TestThreadIdentityCurrent(const void* assert_no_identity) {
>+  ThreadIdentity* identity;
>+
>+  // We have to test this conditionally, because if the test framework relies
>+  // on Abseil, then some previous action may have already allocated an
>+  // identity.
>+  if (assert_no_identity == kCheckNoIdentity) {
>+    identity = CurrentThreadIdentityIfPresent();
>+    EXPECT_TRUE(identity == nullptr);
>+  }
>+
>+  identity = synchronization_internal::GetOrCreateCurrentThreadIdentity();
>+  EXPECT_TRUE(identity != nullptr);
>+  ThreadIdentity* identity_no_init;
>+  identity_no_init = CurrentThreadIdentityIfPresent();
>+  EXPECT_TRUE(identity == identity_no_init);
>+
>+  // Check that per_thread_synch is correctly aligned.
>+  EXPECT_EQ(0, reinterpret_cast<intptr_t>(&identity->per_thread_synch) %
>+                   PerThreadSynch::kAlignment);
>+  EXPECT_EQ(identity, identity->per_thread_synch.thread_identity());
>+
>+  absl::base_internal::SpinLockHolder l(&map_lock);
>+  num_identities_reused++;
>+}
>+
>+TEST(ThreadIdentityTest, BasicIdentityWorks) {
>+  // This tests for the main() thread.
>+  TestThreadIdentityCurrent(nullptr);
>+}
>+
>+TEST(ThreadIdentityTest, BasicIdentityWorksThreaded) {
>+  // Now try the same basic test with multiple threads being created and
>+  // destroyed.  This makes sure that:
>+  // - New threads are created without a ThreadIdentity.
>+  // - We re-allocate ThreadIdentity objects from the free-list.
>+  // - If a thread implementation chooses to recycle threads, that
>+  //   correct re-initialization occurs.
>+  static const int kNumLoops = 3;
>+  static const int kNumThreads = 400;
>+  for (int iter = 0; iter < kNumLoops; iter++) {
>+    std::vector<std::thread> threads;
>+    for (int i = 0; i < kNumThreads; ++i) {
>+      threads.push_back(
>+          std::thread(TestThreadIdentityCurrent, kCheckNoIdentity));
>+    }
>+    for (auto& thread : threads) {
>+      thread.join();
>+    }
>+  }
>+
>+  // We should have recycled ThreadIdentity objects above; while (external)
>+  // library threads allocating their own identities may preclude some
>+  // reuse, we should have sufficient repetitions to exclude this.
>+  EXPECT_LT(kNumThreads, num_identities_reused);
>+}
>+
>+TEST(ThreadIdentityTest, ReusedThreadIdentityMutexTest) {
>+  // This test repeatly creates and joins a series of threads, each of
>+  // which acquires and releases shared Mutex locks. This verifies
>+  // Mutex operations work correctly under a reused
>+  // ThreadIdentity. Note that the most likely failure mode of this
>+  // test is a crash or deadlock.
>+  static const int kNumLoops = 10;
>+  static const int kNumThreads = 12;
>+  static const int kNumMutexes = 3;
>+  static const int kNumLockLoops = 5;
>+
>+  Mutex mutexes[kNumMutexes];
>+  for (int iter = 0; iter < kNumLoops; ++iter) {
>+    std::vector<std::thread> threads;
>+    for (int thread = 0; thread < kNumThreads; ++thread) {
>+      threads.push_back(std::thread([&]() {
>+        for (int l = 0; l < kNumLockLoops; ++l) {
>+          for (int m = 0; m < kNumMutexes; ++m) {
>+            MutexLock lock(&mutexes[m]);
>+          }
>+        }
>+      }));
>+    }
>+    for (auto& thread : threads) {
>+      thread.join();
>+    }
>+  }
>+}
>+
>+}  // namespace
>+}  // namespace base_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/throw_delegate.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/throw_delegate.cc
>new file mode 100644
>index 00000000000..46dc573cfa8
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/throw_delegate.cc
>@@ -0,0 +1,106 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/base/internal/throw_delegate.h"
>+
>+#include <cstdlib>
>+#include <functional>
>+#include <new>
>+#include <stdexcept>
>+#include "absl/base/config.h"
>+#include "absl/base/internal/raw_logging.h"
>+
>+namespace absl {
>+namespace base_internal {
>+
>+namespace {
>+template <typename T>
>+[[noreturn]] void Throw(const T& error) {
>+#ifdef ABSL_HAVE_EXCEPTIONS
>+  throw error;
>+#else
>+  ABSL_RAW_LOG(ERROR, "%s", error.what());
>+  abort();
>+#endif
>+}
>+}  // namespace
>+
>+void ThrowStdLogicError(const std::string& what_arg) {
>+  Throw(std::logic_error(what_arg));
>+}
>+void ThrowStdLogicError(const char* what_arg) {
>+  Throw(std::logic_error(what_arg));
>+}
>+void ThrowStdInvalidArgument(const std::string& what_arg) {
>+  Throw(std::invalid_argument(what_arg));
>+}
>+void ThrowStdInvalidArgument(const char* what_arg) {
>+  Throw(std::invalid_argument(what_arg));
>+}
>+
>+void ThrowStdDomainError(const std::string& what_arg) {
>+  Throw(std::domain_error(what_arg));
>+}
>+void ThrowStdDomainError(const char* what_arg) {
>+  Throw(std::domain_error(what_arg));
>+}
>+
>+void ThrowStdLengthError(const std::string& what_arg) {
>+  Throw(std::length_error(what_arg));
>+}
>+void ThrowStdLengthError(const char* what_arg) {
>+  Throw(std::length_error(what_arg));
>+}
>+
>+void ThrowStdOutOfRange(const std::string& what_arg) {
>+  Throw(std::out_of_range(what_arg));
>+}
>+void ThrowStdOutOfRange(const char* what_arg) {
>+  Throw(std::out_of_range(what_arg));
>+}
>+
>+void ThrowStdRuntimeError(const std::string& what_arg) {
>+  Throw(std::runtime_error(what_arg));
>+}
>+void ThrowStdRuntimeError(const char* what_arg) {
>+  Throw(std::runtime_error(what_arg));
>+}
>+
>+void ThrowStdRangeError(const std::string& what_arg) {
>+  Throw(std::range_error(what_arg));
>+}
>+void ThrowStdRangeError(const char* what_arg) {
>+  Throw(std::range_error(what_arg));
>+}
>+
>+void ThrowStdOverflowError(const std::string& what_arg) {
>+  Throw(std::overflow_error(what_arg));
>+}
>+void ThrowStdOverflowError(const char* what_arg) {
>+  Throw(std::overflow_error(what_arg));
>+}
>+
>+void ThrowStdUnderflowError(const std::string& what_arg) {
>+  Throw(std::underflow_error(what_arg));
>+}
>+void ThrowStdUnderflowError(const char* what_arg) {
>+  Throw(std::underflow_error(what_arg));
>+}
>+
>+void ThrowStdBadFunctionCall() { Throw(std::bad_function_call()); }
>+
>+void ThrowStdBadAlloc() { Throw(std::bad_alloc()); }
>+
>+}  // namespace base_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/throw_delegate.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/throw_delegate.h
>new file mode 100644
>index 00000000000..70e2d7709e5
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/throw_delegate.h
>@@ -0,0 +1,71 @@
>+//
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+
>+#ifndef ABSL_BASE_INTERNAL_THROW_DELEGATE_H_
>+#define ABSL_BASE_INTERNAL_THROW_DELEGATE_H_
>+
>+#include <string>
>+
>+namespace absl {
>+namespace base_internal {
>+
>+// Helper functions that allow throwing exceptions consistently from anywhere.
>+// The main use case is for header-based libraries (eg templates), as they will
>+// be built by many different targets with their own compiler options.
>+// In particular, this will allow a safe way to throw exceptions even if the
>+// caller is compiled with -fno-exceptions.  This is intended for implementing
>+// things like map<>::at(), which the standard documents as throwing an
>+// exception on error.
>+//
>+// Using other techniques like #if tricks could lead to ODR violations.
>+//
>+// You shouldn't use it unless you're writing code that you know will be built
>+// both with and without exceptions and you need to conform to an interface
>+// that uses exceptions.
>+
>+[[noreturn]] void ThrowStdLogicError(const std::string& what_arg);
>+[[noreturn]] void ThrowStdLogicError(const char* what_arg);
>+[[noreturn]] void ThrowStdInvalidArgument(const std::string& what_arg);
>+[[noreturn]] void ThrowStdInvalidArgument(const char* what_arg);
>+[[noreturn]] void ThrowStdDomainError(const std::string& what_arg);
>+[[noreturn]] void ThrowStdDomainError(const char* what_arg);
>+[[noreturn]] void ThrowStdLengthError(const std::string& what_arg);
>+[[noreturn]] void ThrowStdLengthError(const char* what_arg);
>+[[noreturn]] void ThrowStdOutOfRange(const std::string& what_arg);
>+[[noreturn]] void ThrowStdOutOfRange(const char* what_arg);
>+[[noreturn]] void ThrowStdRuntimeError(const std::string& what_arg);
>+[[noreturn]] void ThrowStdRuntimeError(const char* what_arg);
>+[[noreturn]] void ThrowStdRangeError(const std::string& what_arg);
>+[[noreturn]] void ThrowStdRangeError(const char* what_arg);
>+[[noreturn]] void ThrowStdOverflowError(const std::string& what_arg);
>+[[noreturn]] void ThrowStdOverflowError(const char* what_arg);
>+[[noreturn]] void ThrowStdUnderflowError(const std::string& what_arg);
>+[[noreturn]] void ThrowStdUnderflowError(const char* what_arg);
>+
>+[[noreturn]] void ThrowStdBadFunctionCall();
>+[[noreturn]] void ThrowStdBadAlloc();
>+
>+// ThrowStdBadArrayNewLength() cannot be consistently supported because
>+// std::bad_array_new_length is missing in libstdc++ until 4.9.0.
>+// https://gcc.gnu.org/onlinedocs/gcc-4.8.3/libstdc++/api/a01379_source.html
>+// https://gcc.gnu.org/onlinedocs/gcc-4.9.0/libstdc++/api/a01327_source.html
>+// libcxx (as of 3.2) and msvc (as of 2015) both have it.
>+// [[noreturn]] void ThrowStdBadArrayNewLength();
>+
>+}  // namespace base_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_BASE_INTERNAL_THROW_DELEGATE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/tsan_mutex_interface.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/tsan_mutex_interface.h
>new file mode 100644
>index 00000000000..6bb4faedb87
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/tsan_mutex_interface.h
>@@ -0,0 +1,66 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// This file is intended solely for spinlock.h.
>+// It provides ThreadSanitizer annotations for custom mutexes.
>+// See <sanitizer/tsan_interface.h> for meaning of these annotations.
>+
>+#ifndef ABSL_BASE_INTERNAL_TSAN_MUTEX_INTERFACE_H_
>+#define ABSL_BASE_INTERNAL_TSAN_MUTEX_INTERFACE_H_
>+
>+// ABSL_INTERNAL_HAVE_TSAN_INTERFACE
>+// Macro intended only for internal use.
>+//
>+// Checks whether LLVM Thread Sanitizer interfaces are available.
>+// First made available in LLVM 5.0 (Sep 2017).
>+#ifdef ABSL_INTERNAL_HAVE_TSAN_INTERFACE
>+#error "ABSL_INTERNAL_HAVE_TSAN_INTERFACE cannot be directly set."
>+#endif
>+
>+#if defined(THREAD_SANITIZER) && defined(__has_include)
>+#if __has_include(<sanitizer/tsan_interface.h>)
>+#define ABSL_INTERNAL_HAVE_TSAN_INTERFACE 1
>+#endif
>+#endif
>+
>+#ifdef ABSL_INTERNAL_HAVE_TSAN_INTERFACE
>+#include <sanitizer/tsan_interface.h>
>+
>+#define ABSL_TSAN_MUTEX_CREATE __tsan_mutex_create
>+#define ABSL_TSAN_MUTEX_DESTROY __tsan_mutex_destroy
>+#define ABSL_TSAN_MUTEX_PRE_LOCK __tsan_mutex_pre_lock
>+#define ABSL_TSAN_MUTEX_POST_LOCK __tsan_mutex_post_lock
>+#define ABSL_TSAN_MUTEX_PRE_UNLOCK __tsan_mutex_pre_unlock
>+#define ABSL_TSAN_MUTEX_POST_UNLOCK __tsan_mutex_post_unlock
>+#define ABSL_TSAN_MUTEX_PRE_SIGNAL __tsan_mutex_pre_signal
>+#define ABSL_TSAN_MUTEX_POST_SIGNAL __tsan_mutex_post_signal
>+#define ABSL_TSAN_MUTEX_PRE_DIVERT __tsan_mutex_pre_divert
>+#define ABSL_TSAN_MUTEX_POST_DIVERT __tsan_mutex_post_divert
>+
>+#else
>+
>+#define ABSL_TSAN_MUTEX_CREATE(...)
>+#define ABSL_TSAN_MUTEX_DESTROY(...)
>+#define ABSL_TSAN_MUTEX_PRE_LOCK(...)
>+#define ABSL_TSAN_MUTEX_POST_LOCK(...)
>+#define ABSL_TSAN_MUTEX_PRE_UNLOCK(...)
>+#define ABSL_TSAN_MUTEX_POST_UNLOCK(...)
>+#define ABSL_TSAN_MUTEX_PRE_SIGNAL(...)
>+#define ABSL_TSAN_MUTEX_POST_SIGNAL(...)
>+#define ABSL_TSAN_MUTEX_PRE_DIVERT(...)
>+#define ABSL_TSAN_MUTEX_POST_DIVERT(...)
>+
>+#endif
>+
>+#endif  // ABSL_BASE_INTERNAL_TSAN_MUTEX_INTERFACE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/unaligned_access.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/unaligned_access.h
>new file mode 100644
>index 00000000000..5c7517abd71
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/unaligned_access.h
>@@ -0,0 +1,297 @@
>+//
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+
>+#ifndef ABSL_BASE_INTERNAL_UNALIGNED_ACCESS_H_
>+#define ABSL_BASE_INTERNAL_UNALIGNED_ACCESS_H_
>+
>+#include <string.h>
>+#include <cstdint>
>+
>+#include "absl/base/attributes.h"
>+
>+// unaligned APIs
>+
>+// Portable handling of unaligned loads, stores, and copies.
>+// On some platforms, like ARM, the copy functions can be more efficient
>+// then a load and a store.
>+//
>+// It is possible to implement all of these these using constant-length memcpy
>+// calls, which is portable and will usually be inlined into simple loads and
>+// stores if the architecture supports it. However, such inlining usually
>+// happens in a pass that's quite late in compilation, which means the resulting
>+// loads and stores cannot participate in many other optimizations, leading to
>+// overall worse code.
>+
>+// The unaligned API is C++ only.  The declarations use C++ features
>+// (namespaces, inline) which are absent or incompatible in C.
>+#if defined(__cplusplus)
>+
>+#if defined(ADDRESS_SANITIZER) || defined(THREAD_SANITIZER) ||\
>+    defined(MEMORY_SANITIZER)
>+// Consider we have an unaligned load/store of 4 bytes from address 0x...05.
>+// AddressSanitizer will treat it as a 3-byte access to the range 05:07 and
>+// will miss a bug if 08 is the first unaddressable byte.
>+// ThreadSanitizer will also treat this as a 3-byte access to 05:07 and will
>+// miss a race between this access and some other accesses to 08.
>+// MemorySanitizer will correctly propagate the shadow on unaligned stores
>+// and correctly report bugs on unaligned loads, but it may not properly
>+// update and report the origin of the uninitialized memory.
>+// For all three tools, replacing an unaligned access with a tool-specific
>+// callback solves the problem.
>+
>+// Make sure uint16_t/uint32_t/uint64_t are defined.
>+#include <stdint.h>
>+
>+extern "C" {
>+uint16_t __sanitizer_unaligned_load16(const void *p);
>+uint32_t __sanitizer_unaligned_load32(const void *p);
>+uint64_t __sanitizer_unaligned_load64(const void *p);
>+void __sanitizer_unaligned_store16(void *p, uint16_t v);
>+void __sanitizer_unaligned_store32(void *p, uint32_t v);
>+void __sanitizer_unaligned_store64(void *p, uint64_t v);
>+}  // extern "C"
>+
>+namespace absl {
>+
>+inline uint16_t UnalignedLoad16(const void *p) {
>+  return __sanitizer_unaligned_load16(p);
>+}
>+
>+inline uint32_t UnalignedLoad32(const void *p) {
>+  return __sanitizer_unaligned_load32(p);
>+}
>+
>+inline uint64_t UnalignedLoad64(const void *p) {
>+  return __sanitizer_unaligned_load64(p);
>+}
>+
>+inline void UnalignedStore16(void *p, uint16_t v) {
>+  __sanitizer_unaligned_store16(p, v);
>+}
>+
>+inline void UnalignedStore32(void *p, uint32_t v) {
>+  __sanitizer_unaligned_store32(p, v);
>+}
>+
>+inline void UnalignedStore64(void *p, uint64_t v) {
>+  __sanitizer_unaligned_store64(p, v);
>+}
>+
>+}  // namespace absl
>+
>+#define ABSL_INTERNAL_UNALIGNED_LOAD16(_p) (absl::UnalignedLoad16(_p))
>+#define ABSL_INTERNAL_UNALIGNED_LOAD32(_p) (absl::UnalignedLoad32(_p))
>+#define ABSL_INTERNAL_UNALIGNED_LOAD64(_p) (absl::UnalignedLoad64(_p))
>+
>+#define ABSL_INTERNAL_UNALIGNED_STORE16(_p, _val) \
>+  (absl::UnalignedStore16(_p, _val))
>+#define ABSL_INTERNAL_UNALIGNED_STORE32(_p, _val) \
>+  (absl::UnalignedStore32(_p, _val))
>+#define ABSL_INTERNAL_UNALIGNED_STORE64(_p, _val) \
>+  (absl::UnalignedStore64(_p, _val))
>+
>+#elif defined(UNDEFINED_BEHAVIOR_SANITIZER)
>+
>+namespace absl {
>+
>+inline uint16_t UnalignedLoad16(const void *p) {
>+  uint16_t t;
>+  memcpy(&t, p, sizeof t);
>+  return t;
>+}
>+
>+inline uint32_t UnalignedLoad32(const void *p) {
>+  uint32_t t;
>+  memcpy(&t, p, sizeof t);
>+  return t;
>+}
>+
>+inline uint64_t UnalignedLoad64(const void *p) {
>+  uint64_t t;
>+  memcpy(&t, p, sizeof t);
>+  return t;
>+}
>+
>+inline void UnalignedStore16(void *p, uint16_t v) { memcpy(p, &v, sizeof v); }
>+
>+inline void UnalignedStore32(void *p, uint32_t v) { memcpy(p, &v, sizeof v); }
>+
>+inline void UnalignedStore64(void *p, uint64_t v) { memcpy(p, &v, sizeof v); }
>+
>+}  // namespace absl
>+
>+#define ABSL_INTERNAL_UNALIGNED_LOAD16(_p) (absl::UnalignedLoad16(_p))
>+#define ABSL_INTERNAL_UNALIGNED_LOAD32(_p) (absl::UnalignedLoad32(_p))
>+#define ABSL_INTERNAL_UNALIGNED_LOAD64(_p) (absl::UnalignedLoad64(_p))
>+
>+#define ABSL_INTERNAL_UNALIGNED_STORE16(_p, _val) \
>+  (absl::UnalignedStore16(_p, _val))
>+#define ABSL_INTERNAL_UNALIGNED_STORE32(_p, _val) \
>+  (absl::UnalignedStore32(_p, _val))
>+#define ABSL_INTERNAL_UNALIGNED_STORE64(_p, _val) \
>+  (absl::UnalignedStore64(_p, _val))
>+
>+#elif defined(__x86_64__) || defined(_M_X64) || defined(__i386) || \
>+    defined(_M_IX86) || defined(__ppc__) || defined(__PPC__) ||    \
>+    defined(__ppc64__) || defined(__PPC64__)
>+
>+// x86 and x86-64 can perform unaligned loads/stores directly;
>+// modern PowerPC hardware can also do unaligned integer loads and stores;
>+// but note: the FPU still sends unaligned loads and stores to a trap handler!
>+
>+#define ABSL_INTERNAL_UNALIGNED_LOAD16(_p) \
>+  (*reinterpret_cast<const uint16_t *>(_p))
>+#define ABSL_INTERNAL_UNALIGNED_LOAD32(_p) \
>+  (*reinterpret_cast<const uint32_t *>(_p))
>+#define ABSL_INTERNAL_UNALIGNED_LOAD64(_p) \
>+  (*reinterpret_cast<const uint64_t *>(_p))
>+
>+#define ABSL_INTERNAL_UNALIGNED_STORE16(_p, _val) \
>+  (*reinterpret_cast<uint16_t *>(_p) = (_val))
>+#define ABSL_INTERNAL_UNALIGNED_STORE32(_p, _val) \
>+  (*reinterpret_cast<uint32_t *>(_p) = (_val))
>+#define ABSL_INTERNAL_UNALIGNED_STORE64(_p, _val) \
>+  (*reinterpret_cast<uint64_t *>(_p) = (_val))
>+
>+#elif defined(__arm__) && \
>+      !defined(__ARM_ARCH_5__) && \
>+      !defined(__ARM_ARCH_5T__) && \
>+      !defined(__ARM_ARCH_5TE__) && \
>+      !defined(__ARM_ARCH_5TEJ__) && \
>+      !defined(__ARM_ARCH_6__) && \
>+      !defined(__ARM_ARCH_6J__) && \
>+      !defined(__ARM_ARCH_6K__) && \
>+      !defined(__ARM_ARCH_6Z__) && \
>+      !defined(__ARM_ARCH_6ZK__) && \
>+      !defined(__ARM_ARCH_6T2__)
>+
>+
>+// ARMv7 and newer support native unaligned accesses, but only of 16-bit
>+// and 32-bit values (not 64-bit); older versions either raise a fatal signal,
>+// do an unaligned read and rotate the words around a bit, or do the reads very
>+// slowly (trip through kernel mode). There's no simple #define that says just
>+// "ARMv7 or higher", so we have to filter away all ARMv5 and ARMv6
>+// sub-architectures. Newer gcc (>= 4.6) set an __ARM_FEATURE_ALIGNED #define,
>+// so in time, maybe we can move on to that.
>+//
>+// This is a mess, but there's not much we can do about it.
>+//
>+// To further complicate matters, only LDR instructions (single reads) are
>+// allowed to be unaligned, not LDRD (two reads) or LDM (many reads). Unless we
>+// explicitly tell the compiler that these accesses can be unaligned, it can and
>+// will combine accesses. On armcc, the way to signal this is done by accessing
>+// through the type (uint32_t __packed *), but GCC has no such attribute
>+// (it ignores __attribute__((packed)) on individual variables). However,
>+// we can tell it that a _struct_ is unaligned, which has the same effect,
>+// so we do that.
>+
>+namespace absl {
>+namespace internal {
>+
>+struct Unaligned16Struct {
>+  uint16_t value;
>+  uint8_t dummy;  // To make the size non-power-of-two.
>+} ABSL_ATTRIBUTE_PACKED;
>+
>+struct Unaligned32Struct {
>+  uint32_t value;
>+  uint8_t dummy;  // To make the size non-power-of-two.
>+} ABSL_ATTRIBUTE_PACKED;
>+
>+}  // namespace internal
>+}  // namespace absl
>+
>+#define ABSL_INTERNAL_UNALIGNED_LOAD16(_p) \
>+  ((reinterpret_cast<const ::absl::internal::Unaligned16Struct *>(_p))->value)
>+#define ABSL_INTERNAL_UNALIGNED_LOAD32(_p) \
>+  ((reinterpret_cast<const ::absl::internal::Unaligned32Struct *>(_p))->value)
>+
>+#define ABSL_INTERNAL_UNALIGNED_STORE16(_p, _val)                          \
>+  ((reinterpret_cast< ::absl::internal::Unaligned16Struct *>(_p))->value = \
>+       (_val))
>+#define ABSL_INTERNAL_UNALIGNED_STORE32(_p, _val)                          \
>+  ((reinterpret_cast< ::absl::internal::Unaligned32Struct *>(_p))->value = \
>+       (_val))
>+
>+namespace absl {
>+
>+inline uint64_t UnalignedLoad64(const void *p) {
>+  uint64_t t;
>+  memcpy(&t, p, sizeof t);
>+  return t;
>+}
>+
>+inline void UnalignedStore64(void *p, uint64_t v) { memcpy(p, &v, sizeof v); }
>+
>+}  // namespace absl
>+
>+#define ABSL_INTERNAL_UNALIGNED_LOAD64(_p) (absl::UnalignedLoad64(_p))
>+#define ABSL_INTERNAL_UNALIGNED_STORE64(_p, _val) \
>+  (absl::UnalignedStore64(_p, _val))
>+
>+#else
>+
>+// ABSL_INTERNAL_NEED_ALIGNED_LOADS is defined when the underlying platform
>+// doesn't support unaligned access.
>+#define ABSL_INTERNAL_NEED_ALIGNED_LOADS
>+
>+// These functions are provided for architectures that don't support
>+// unaligned loads and stores.
>+
>+namespace absl {
>+
>+inline uint16_t UnalignedLoad16(const void *p) {
>+  uint16_t t;
>+  memcpy(&t, p, sizeof t);
>+  return t;
>+}
>+
>+inline uint32_t UnalignedLoad32(const void *p) {
>+  uint32_t t;
>+  memcpy(&t, p, sizeof t);
>+  return t;
>+}
>+
>+inline uint64_t UnalignedLoad64(const void *p) {
>+  uint64_t t;
>+  memcpy(&t, p, sizeof t);
>+  return t;
>+}
>+
>+inline void UnalignedStore16(void *p, uint16_t v) { memcpy(p, &v, sizeof v); }
>+
>+inline void UnalignedStore32(void *p, uint32_t v) { memcpy(p, &v, sizeof v); }
>+
>+inline void UnalignedStore64(void *p, uint64_t v) { memcpy(p, &v, sizeof v); }
>+
>+}  // namespace absl
>+
>+#define ABSL_INTERNAL_UNALIGNED_LOAD16(_p) (absl::UnalignedLoad16(_p))
>+#define ABSL_INTERNAL_UNALIGNED_LOAD32(_p) (absl::UnalignedLoad32(_p))
>+#define ABSL_INTERNAL_UNALIGNED_LOAD64(_p) (absl::UnalignedLoad64(_p))
>+
>+#define ABSL_INTERNAL_UNALIGNED_STORE16(_p, _val) \
>+  (absl::UnalignedStore16(_p, _val))
>+#define ABSL_INTERNAL_UNALIGNED_STORE32(_p, _val) \
>+  (absl::UnalignedStore32(_p, _val))
>+#define ABSL_INTERNAL_UNALIGNED_STORE64(_p, _val) \
>+  (absl::UnalignedStore64(_p, _val))
>+
>+#endif
>+
>+#endif  // defined(__cplusplus), end of unaligned API
>+
>+#endif  // ABSL_BASE_INTERNAL_UNALIGNED_ACCESS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/unscaledcycleclock.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/unscaledcycleclock.cc
>new file mode 100644
>index 00000000000..a12d68bd10a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/unscaledcycleclock.cc
>@@ -0,0 +1,101 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/base/internal/unscaledcycleclock.h"
>+
>+#if ABSL_USE_UNSCALED_CYCLECLOCK
>+
>+#if defined(_WIN32)
>+#include <intrin.h>
>+#endif
>+
>+#if defined(__powerpc__) || defined(__ppc__)
>+#include <sys/platform/ppc.h>
>+#endif
>+
>+#include "absl/base/internal/sysinfo.h"
>+
>+namespace absl {
>+namespace base_internal {
>+
>+#if defined(__i386__)
>+
>+int64_t UnscaledCycleClock::Now() {
>+  int64_t ret;
>+  __asm__ volatile("rdtsc" : "=A"(ret));
>+  return ret;
>+}
>+
>+double UnscaledCycleClock::Frequency() {
>+  return base_internal::NominalCPUFrequency();
>+}
>+
>+#elif defined(__x86_64__)
>+
>+int64_t UnscaledCycleClock::Now() {
>+  uint64_t low, high;
>+  __asm__ volatile("rdtsc" : "=a"(low), "=d"(high));
>+  return (high << 32) | low;
>+}
>+
>+double UnscaledCycleClock::Frequency() {
>+  return base_internal::NominalCPUFrequency();
>+}
>+
>+#elif defined(__powerpc__) || defined(__ppc__)
>+
>+int64_t UnscaledCycleClock::Now() {
>+  return __ppc_get_timebase();
>+}
>+
>+double UnscaledCycleClock::Frequency() {
>+  return __ppc_get_timebase_freq();
>+}
>+
>+#elif defined(__aarch64__)
>+
>+// System timer of ARMv8 runs at a different frequency than the CPU's.
>+// The frequency is fixed, typically in the range 1-50MHz.  It can be
>+// read at CNTFRQ special register.  We assume the OS has set up
>+// the virtual timer properly.
>+int64_t UnscaledCycleClock::Now() {
>+  int64_t virtual_timer_value;
>+  asm volatile("mrs %0, cntvct_el0" : "=r"(virtual_timer_value));
>+  return virtual_timer_value;
>+}
>+
>+double UnscaledCycleClock::Frequency() {
>+  uint64_t aarch64_timer_frequency;
>+  asm volatile("mrs %0, cntfrq_el0" : "=r"(aarch64_timer_frequency));
>+  return aarch64_timer_frequency;
>+}
>+
>+#elif defined(_M_IX86) || defined(_M_X64)
>+
>+#pragma intrinsic(__rdtsc)
>+
>+int64_t UnscaledCycleClock::Now() {
>+  return __rdtsc();
>+}
>+
>+double UnscaledCycleClock::Frequency() {
>+  return base_internal::NominalCPUFrequency();
>+}
>+
>+#endif
>+
>+}  // namespace base_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_USE_UNSCALED_CYCLECLOCK
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/unscaledcycleclock.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/unscaledcycleclock.h
>new file mode 100644
>index 00000000000..049f1cace98
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/internal/unscaledcycleclock.h
>@@ -0,0 +1,119 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// UnscaledCycleClock
>+//    An UnscaledCycleClock yields the value and frequency of a cycle counter
>+//    that increments at a rate that is approximately constant.
>+//    This class is for internal / whitelisted use only, you should consider
>+//    using CycleClock instead.
>+//
>+// Notes:
>+// The cycle counter frequency is not necessarily the core clock frequency.
>+// That is, CycleCounter cycles are not necessarily "CPU cycles".
>+//
>+// An arbitrary offset may have been added to the counter at power on.
>+//
>+// On some platforms, the rate and offset of the counter may differ
>+// slightly when read from different CPUs of a multiprocessor.  Usually,
>+// we try to ensure that the operating system adjusts values periodically
>+// so that values agree approximately.   If you need stronger guarantees,
>+// consider using alternate interfaces.
>+//
>+// The CPU is not required to maintain the ordering of a cycle counter read
>+// with respect to surrounding instructions.
>+
>+#ifndef ABSL_BASE_INTERNAL_UNSCALEDCYCLECLOCK_H_
>+#define ABSL_BASE_INTERNAL_UNSCALEDCYCLECLOCK_H_
>+
>+#include <cstdint>
>+
>+#if defined(__APPLE__)
>+#include <TargetConditionals.h>
>+#endif
>+
>+#include "absl/base/port.h"
>+
>+// The following platforms have an implementation of a hardware counter.
>+#if defined(__i386__) || defined(__x86_64__) || defined(__aarch64__) || \
>+  defined(__powerpc__) || defined(__ppc__) || \
>+  defined(_M_IX86) || defined(_M_X64)
>+#define ABSL_HAVE_UNSCALED_CYCLECLOCK_IMPLEMENTATION 1
>+#else
>+#define ABSL_HAVE_UNSCALED_CYCLECLOCK_IMPLEMENTATION 0
>+#endif
>+
>+// The following platforms often disable access to the hardware
>+// counter (through a sandbox) even if the underlying hardware has a
>+// usable counter. The CycleTimer interface also requires a *scaled*
>+// CycleClock that runs at atleast 1 MHz. We've found some Android
>+// ARM64 devices where this is not the case, so we disable it by
>+// default on Android ARM64.
>+#if defined(__native_client__) || TARGET_OS_IPHONE || \
>+    (defined(__ANDROID__) && defined(__aarch64__))
>+#define ABSL_USE_UNSCALED_CYCLECLOCK_DEFAULT 0
>+#else
>+#define ABSL_USE_UNSCALED_CYCLECLOCK_DEFAULT 1
>+#endif
>+
>+// UnscaledCycleClock is an optional internal feature.
>+// Use "#if ABSL_USE_UNSCALED_CYCLECLOCK" to test for its presence.
>+// Can be overridden at compile-time via -DABSL_USE_UNSCALED_CYCLECLOCK=0|1
>+#if !defined(ABSL_USE_UNSCALED_CYCLECLOCK)
>+#define ABSL_USE_UNSCALED_CYCLECLOCK               \
>+  (ABSL_HAVE_UNSCALED_CYCLECLOCK_IMPLEMENTATION && \
>+   ABSL_USE_UNSCALED_CYCLECLOCK_DEFAULT)
>+#endif
>+
>+#if ABSL_USE_UNSCALED_CYCLECLOCK
>+
>+// This macro can be used to test if UnscaledCycleClock::Frequency()
>+// is NominalCPUFrequency() on a particular platform.
>+#if  (defined(__i386__) || defined(__x86_64__) || \
>+      defined(_M_IX86) || defined(_M_X64))
>+#define ABSL_INTERNAL_UNSCALED_CYCLECLOCK_FREQUENCY_IS_CPU_FREQUENCY
>+#endif
>+namespace absl {
>+namespace time_internal {
>+class UnscaledCycleClockWrapperForGetCurrentTime;
>+}  // namespace time_internal
>+
>+namespace base_internal {
>+class CycleClock;
>+class UnscaledCycleClockWrapperForInitializeFrequency;
>+
>+class UnscaledCycleClock {
>+ private:
>+  UnscaledCycleClock() = delete;
>+
>+  // Return the value of a cycle counter that counts at a rate that is
>+  // approximately constant.
>+  static int64_t Now();
>+
>+  // Return the how much UnscaledCycleClock::Now() increases per second.
>+  // This is not necessarily the core CPU clock frequency.
>+  // It may be the nominal value report by the kernel, rather than a measured
>+  // value.
>+  static double Frequency();
>+
>+  // Whitelisted friends.
>+  friend class base_internal::CycleClock;
>+  friend class time_internal::UnscaledCycleClockWrapperForGetCurrentTime;
>+  friend class base_internal::UnscaledCycleClockWrapperForInitializeFrequency;
>+};
>+
>+}  // namespace base_internal
>+}  // namespace absl
>+#endif  // ABSL_USE_UNSCALED_CYCLECLOCK
>+
>+#endif  // ABSL_BASE_INTERNAL_UNSCALEDCYCLECLOCK_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/invoke_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/invoke_test.cc
>new file mode 100644
>index 00000000000..466bf114a58
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/invoke_test.cc
>@@ -0,0 +1,200 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/base/internal/invoke.h"
>+
>+#include <functional>
>+#include <memory>
>+#include <string>
>+#include <utility>
>+
>+#include "gmock/gmock.h"
>+#include "gtest/gtest.h"
>+#include "absl/memory/memory.h"
>+#include "absl/strings/str_cat.h"
>+
>+namespace absl {
>+namespace base_internal {
>+namespace {
>+
>+int Function(int a, int b) { return a - b; }
>+
>+int Sink(std::unique_ptr<int> p) {
>+  return *p;
>+}
>+
>+std::unique_ptr<int> Factory(int n) {
>+  return make_unique<int>(n);
>+}
>+
>+void NoOp() {}
>+
>+struct ConstFunctor {
>+  int operator()(int a, int b) const { return a - b; }
>+};
>+
>+struct MutableFunctor {
>+  int operator()(int a, int b) { return a - b; }
>+};
>+
>+struct EphemeralFunctor {
>+  int operator()(int a, int b) && { return a - b; }
>+};
>+
>+struct OverloadedFunctor {
>+  template <typename... Args>
>+  std::string operator()(const Args&... args) & {
>+    return StrCat("&", args...);
>+  }
>+  template <typename... Args>
>+  std::string operator()(const Args&... args) const& {
>+    return StrCat("const&", args...);
>+  }
>+  template <typename... Args>
>+  std::string operator()(const Args&... args) && {
>+    return StrCat("&&", args...);
>+  }
>+};
>+
>+struct Class {
>+  int Method(int a, int b) { return a - b; }
>+  int ConstMethod(int a, int b) const { return a - b; }
>+
>+  int member;
>+};
>+
>+struct FlipFlop {
>+  int ConstMethod() const { return member; }
>+  FlipFlop operator*() const { return {-member}; }
>+
>+  int member;
>+};
>+
>+// CallMaybeWithArg(f) resolves either to Invoke(f) or Invoke(f, 42), depending
>+// on which one is valid.
>+template <typename F>
>+decltype(Invoke(std::declval<const F&>())) CallMaybeWithArg(const F& f) {
>+  return Invoke(f);
>+}
>+
>+template <typename F>
>+decltype(Invoke(std::declval<const F&>(), 42)) CallMaybeWithArg(const F& f) {
>+  return Invoke(f, 42);
>+}
>+
>+TEST(InvokeTest, Function) {
>+  EXPECT_EQ(1, Invoke(Function, 3, 2));
>+  EXPECT_EQ(1, Invoke(&Function, 3, 2));
>+}
>+
>+TEST(InvokeTest, NonCopyableArgument) {
>+  EXPECT_EQ(42, Invoke(Sink, make_unique<int>(42)));
>+}
>+
>+TEST(InvokeTest, NonCopyableResult) {
>+  EXPECT_THAT(Invoke(Factory, 42), ::testing::Pointee(42));
>+}
>+
>+TEST(InvokeTest, VoidResult) {
>+  Invoke(NoOp);
>+}
>+
>+TEST(InvokeTest, ConstFunctor) {
>+  EXPECT_EQ(1, Invoke(ConstFunctor(), 3, 2));
>+}
>+
>+TEST(InvokeTest, MutableFunctor) {
>+  MutableFunctor f;
>+  EXPECT_EQ(1, Invoke(f, 3, 2));
>+  EXPECT_EQ(1, Invoke(MutableFunctor(), 3, 2));
>+}
>+
>+TEST(InvokeTest, EphemeralFunctor) {
>+  EphemeralFunctor f;
>+  EXPECT_EQ(1, Invoke(std::move(f), 3, 2));
>+  EXPECT_EQ(1, Invoke(EphemeralFunctor(), 3, 2));
>+}
>+
>+TEST(InvokeTest, OverloadedFunctor) {
>+  OverloadedFunctor f;
>+  const OverloadedFunctor& cf = f;
>+
>+  EXPECT_EQ("&", Invoke(f));
>+  EXPECT_EQ("& 42", Invoke(f, " 42"));
>+
>+  EXPECT_EQ("const&", Invoke(cf));
>+  EXPECT_EQ("const& 42", Invoke(cf, " 42"));
>+
>+  EXPECT_EQ("&&", Invoke(std::move(f)));
>+  EXPECT_EQ("&& 42", Invoke(std::move(f), " 42"));
>+}
>+
>+TEST(InvokeTest, ReferenceWrapper) {
>+  ConstFunctor cf;
>+  MutableFunctor mf;
>+  EXPECT_EQ(1, Invoke(std::cref(cf), 3, 2));
>+  EXPECT_EQ(1, Invoke(std::ref(cf), 3, 2));
>+  EXPECT_EQ(1, Invoke(std::ref(mf), 3, 2));
>+}
>+
>+TEST(InvokeTest, MemberFunction) {
>+  std::unique_ptr<Class> p(new Class);
>+  std::unique_ptr<const Class> cp(new Class);
>+  EXPECT_EQ(1, Invoke(&Class::Method, p, 3, 2));
>+  EXPECT_EQ(1, Invoke(&Class::Method, p.get(), 3, 2));
>+
>+  EXPECT_EQ(1, Invoke(&Class::ConstMethod, p, 3, 2));
>+  EXPECT_EQ(1, Invoke(&Class::ConstMethod, p.get(), 3, 2));
>+  EXPECT_EQ(1, Invoke(&Class::ConstMethod, *p, 3, 2));
>+
>+  EXPECT_EQ(1, Invoke(&Class::ConstMethod, cp, 3, 2));
>+  EXPECT_EQ(1, Invoke(&Class::ConstMethod, cp.get(), 3, 2));
>+  EXPECT_EQ(1, Invoke(&Class::ConstMethod, *cp, 3, 2));
>+
>+  EXPECT_EQ(1, Invoke(&Class::Method, make_unique<Class>(), 3, 2));
>+  EXPECT_EQ(1, Invoke(&Class::ConstMethod, make_unique<Class>(), 3, 2));
>+  EXPECT_EQ(1, Invoke(&Class::ConstMethod, make_unique<const Class>(), 3, 2));
>+}
>+
>+TEST(InvokeTest, DataMember) {
>+  std::unique_ptr<Class> p(new Class{42});
>+  std::unique_ptr<const Class> cp(new Class{42});
>+  EXPECT_EQ(42, Invoke(&Class::member, p));
>+  EXPECT_EQ(42, Invoke(&Class::member, *p));
>+  EXPECT_EQ(42, Invoke(&Class::member, p.get()));
>+
>+  Invoke(&Class::member, p) = 42;
>+  Invoke(&Class::member, p.get()) = 42;
>+
>+  EXPECT_EQ(42, Invoke(&Class::member, cp));
>+  EXPECT_EQ(42, Invoke(&Class::member, *cp));
>+  EXPECT_EQ(42, Invoke(&Class::member, cp.get()));
>+}
>+
>+TEST(InvokeTest, FlipFlop) {
>+  FlipFlop obj = {42};
>+  // This call could resolve to (obj.*&FlipFlop::ConstMethod)() or
>+  // ((*obj).*&FlipFlop::ConstMethod)(). We verify that it's the former.
>+  EXPECT_EQ(42, Invoke(&FlipFlop::ConstMethod, obj));
>+  EXPECT_EQ(42, Invoke(&FlipFlop::member, obj));
>+}
>+
>+TEST(InvokeTest, SfinaeFriendly) {
>+  CallMaybeWithArg(NoOp);
>+  EXPECT_THAT(CallMaybeWithArg(Factory), ::testing::Pointee(42));
>+}
>+
>+}  // namespace
>+}  // namespace base_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/log_severity.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/log_severity.h
>new file mode 100644
>index 00000000000..e2931c34d1d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/log_severity.h
>@@ -0,0 +1,67 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+
>+#ifndef ABSL_BASE_INTERNAL_LOG_SEVERITY_H_
>+#define ABSL_BASE_INTERNAL_LOG_SEVERITY_H_
>+
>+#include <array>
>+
>+#include "absl/base/attributes.h"
>+
>+namespace absl {
>+
>+// Four severity levels are defined.  Logging APIs should terminate the program
>+// when a message is logged at severity `kFatal`; the other levels have no
>+// special semantics.
>+enum class LogSeverity : int {
>+  kInfo = 0,
>+  kWarning = 1,
>+  kError = 2,
>+  kFatal = 3,
>+};
>+
>+// Returns an iterable of all standard `absl::LogSeverity` values, ordered from
>+// least to most severe.
>+constexpr std::array<absl::LogSeverity, 4> LogSeverities() {
>+  return {{absl::LogSeverity::kInfo, absl::LogSeverity::kWarning,
>+           absl::LogSeverity::kError, absl::LogSeverity::kFatal}};
>+}
>+
>+// Returns the all-caps std::string representation (e.g. "INFO") of the specified
>+// severity level if it is one of the normal levels and "UNKNOWN" otherwise.
>+constexpr const char* LogSeverityName(absl::LogSeverity s) {
>+  return s == absl::LogSeverity::kInfo
>+             ? "INFO"
>+             : s == absl::LogSeverity::kWarning
>+                   ? "WARNING"
>+                   : s == absl::LogSeverity::kError
>+                         ? "ERROR"
>+                         : s == absl::LogSeverity::kFatal ? "FATAL" : "UNKNOWN";
>+}
>+
>+// Values less than `kInfo` normalize to `kInfo`; values greater than `kFatal`
>+// normalize to `kError` (**NOT** `kFatal`).
>+constexpr absl::LogSeverity NormalizeLogSeverity(absl::LogSeverity s) {
>+  return s < absl::LogSeverity::kInfo
>+             ? absl::LogSeverity::kInfo
>+             : s > absl::LogSeverity::kFatal ? absl::LogSeverity::kError : s;
>+}
>+constexpr absl::LogSeverity NormalizeLogSeverity(int s) {
>+  return NormalizeLogSeverity(static_cast<absl::LogSeverity>(s));
>+}
>+
>+}  // namespace absl
>+
>+#endif  // ABSL_BASE_INTERNAL_LOG_SEVERITY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/macros.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/macros.h
>new file mode 100644
>index 00000000000..ca3d5edb653
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/macros.h
>@@ -0,0 +1,202 @@
>+//
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// -----------------------------------------------------------------------------
>+// File: macros.h
>+// -----------------------------------------------------------------------------
>+//
>+// This header file defines the set of language macros used within Abseil code.
>+// For the set of macros used to determine supported compilers and platforms,
>+// see absl/base/config.h instead.
>+//
>+// This code is compiled directly on many platforms, including client
>+// platforms like Windows, Mac, and embedded systems.  Before making
>+// any changes here, make sure that you're not breaking any platforms.
>+//
>+
>+#ifndef ABSL_BASE_MACROS_H_
>+#define ABSL_BASE_MACROS_H_
>+
>+#include <cassert>
>+#include <cstddef>
>+
>+#include "absl/base/port.h"
>+
>+// ABSL_ARRAYSIZE()
>+//
>+// Returns the number of elements in an array as a compile-time constant, which
>+// can be used in defining new arrays. If you use this macro on a pointer by
>+// mistake, you will get a compile-time error.
>+#define ABSL_ARRAYSIZE(array) \
>+  (sizeof(::absl::macros_internal::ArraySizeHelper(array)))
>+
>+namespace absl {
>+namespace macros_internal {
>+// Note: this internal template function declaration is used by ABSL_ARRAYSIZE.
>+// The function doesn't need a definition, as we only use its type.
>+template <typename T, size_t N>
>+auto ArraySizeHelper(const T (&array)[N]) -> char (&)[N];
>+}  // namespace macros_internal
>+}  // namespace absl
>+
>+// kLinkerInitialized
>+//
>+// An enum used only as a constructor argument to indicate that a variable has
>+// static storage duration, and that the constructor should do nothing to its
>+// state. Use of this macro indicates to the reader that it is legal to
>+// declare a static instance of the class, provided the constructor is given
>+// the absl::base_internal::kLinkerInitialized argument.
>+//
>+// Normally, it is unsafe to declare a static variable that has a constructor or
>+// a destructor because invocation order is undefined. However, if the type can
>+// be zero-initialized (which the loader does for static variables) into a valid
>+// state and the type's destructor does not affect storage, then a constructor
>+// for static initialization can be declared.
>+//
>+// Example:
>+//       // Declaration
>+//       explicit MyClass(absl::base_internal:LinkerInitialized x) {}
>+//
>+//       // Invocation
>+//       static MyClass my_global(absl::base_internal::kLinkerInitialized);
>+namespace absl {
>+namespace base_internal {
>+enum LinkerInitialized {
>+  kLinkerInitialized = 0,
>+};
>+}  // namespace base_internal
>+}  // namespace absl
>+
>+// ABSL_FALLTHROUGH_INTENDED
>+//
>+// Annotates implicit fall-through between switch labels, allowing a case to
>+// indicate intentional fallthrough and turn off warnings about any lack of a
>+// `break` statement. The ABSL_FALLTHROUGH_INTENDED macro should be followed by
>+// a semicolon and can be used in most places where `break` can, provided that
>+// no statements exist between it and the next switch label.
>+//
>+// Example:
>+//
>+//  switch (x) {
>+//    case 40:
>+//    case 41:
>+//      if (truth_is_out_there) {
>+//        ++x;
>+//        ABSL_FALLTHROUGH_INTENDED;  // Use instead of/along with annotations
>+//                                    // in comments
>+//      } else {
>+//        return x;
>+//      }
>+//    case 42:
>+//      ...
>+//
>+// Notes: when compiled with clang in C++11 mode, the ABSL_FALLTHROUGH_INTENDED
>+// macro is expanded to the [[clang::fallthrough]] attribute, which is analysed
>+// when  performing switch labels fall-through diagnostic
>+// (`-Wimplicit-fallthrough`). See clang documentation on language extensions
>+// for details:
>+// http://clang.llvm.org/docs/AttributeReference.html#fallthrough-clang-fallthrough
>+//
>+// When used with unsupported compilers, the ABSL_FALLTHROUGH_INTENDED macro
>+// has no effect on diagnostics. In any case this macro has no effect on runtime
>+// behavior and performance of code.
>+#ifdef ABSL_FALLTHROUGH_INTENDED
>+#error "ABSL_FALLTHROUGH_INTENDED should not be defined."
>+#endif
>+
>+// TODO(zhangxy): Use c++17 standard [[fallthrough]] macro, when supported.
>+#if defined(__clang__) && defined(__has_warning)
>+#if __has_feature(cxx_attributes) && __has_warning("-Wimplicit-fallthrough")
>+#define ABSL_FALLTHROUGH_INTENDED [[clang::fallthrough]]
>+#endif
>+#elif defined(__GNUC__) && __GNUC__ >= 7
>+#define ABSL_FALLTHROUGH_INTENDED [[gnu::fallthrough]]
>+#endif
>+
>+#ifndef ABSL_FALLTHROUGH_INTENDED
>+#define ABSL_FALLTHROUGH_INTENDED \
>+  do {                            \
>+  } while (0)
>+#endif
>+
>+// ABSL_DEPRECATED()
>+//
>+// Marks a deprecated class, struct, enum, function, method and variable
>+// declarations. The macro argument is used as a custom diagnostic message (e.g.
>+// suggestion of a better alternative).
>+//
>+// Example:
>+//
>+//   class ABSL_DEPRECATED("Use Bar instead") Foo {...};
>+//   ABSL_DEPRECATED("Use Baz instead") void Bar() {...}
>+//
>+// Every usage of a deprecated entity will trigger a warning when compiled with
>+// clang's `-Wdeprecated-declarations` option. This option is turned off by
>+// default, but the warnings will be reported by clang-tidy.
>+#if defined(__clang__) && __cplusplus >= 201103L
>+#define ABSL_DEPRECATED(message) __attribute__((deprecated(message)))
>+#endif
>+
>+#ifndef ABSL_DEPRECATED
>+#define ABSL_DEPRECATED(message)
>+#endif
>+
>+// ABSL_BAD_CALL_IF()
>+//
>+// Used on a function overload to trap bad calls: any call that matches the
>+// overload will cause a compile-time error. This macro uses a clang-specific
>+// "enable_if" attribute, as described at
>+// http://clang.llvm.org/docs/AttributeReference.html#enable-if
>+//
>+// Overloads which use this macro should be bracketed by
>+// `#ifdef ABSL_BAD_CALL_IF`.
>+//
>+// Example:
>+//
>+//   int isdigit(int c);
>+//   #ifdef ABSL_BAD_CALL_IF
>+//   int isdigit(int c)
>+//     ABSL_BAD_CALL_IF(c <= -1 || c > 255,
>+//                       "'c' must have the value of an unsigned char or EOF");
>+//   #endif // ABSL_BAD_CALL_IF
>+
>+#if defined(__clang__)
>+# if __has_attribute(enable_if)
>+#  define ABSL_BAD_CALL_IF(expr, msg) \
>+    __attribute__((enable_if(expr, "Bad call trap"), unavailable(msg)))
>+# endif
>+#endif
>+
>+// ABSL_ASSERT()
>+//
>+// In C++11, `assert` can't be used portably within constexpr functions.
>+// ABSL_ASSERT functions as a runtime assert but works in C++11 constexpr
>+// functions.  Example:
>+//
>+// constexpr double Divide(double a, double b) {
>+//   return ABSL_ASSERT(b != 0), a / b;
>+// }
>+//
>+// This macro is inspired by
>+// https://akrzemi1.wordpress.com/2017/05/18/asserts-in-constexpr-functions/
>+#if defined(NDEBUG)
>+#define ABSL_ASSERT(expr) (false ? (void)(expr) : (void)0)
>+#else
>+#define ABSL_ASSERT(expr)              \
>+  (ABSL_PREDICT_TRUE((expr)) ? (void)0 \
>+                             : [] { assert(false && #expr); }())  // NOLINT
>+#endif
>+
>+#endif  // ABSL_BASE_MACROS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/optimization.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/optimization.h
>new file mode 100644
>index 00000000000..2fddfc800c1
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/optimization.h
>@@ -0,0 +1,165 @@
>+//
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// -----------------------------------------------------------------------------
>+// File: optimization.h
>+// -----------------------------------------------------------------------------
>+//
>+// This header file defines portable macros for performance optimization.
>+
>+#ifndef ABSL_BASE_OPTIMIZATION_H_
>+#define ABSL_BASE_OPTIMIZATION_H_
>+
>+#include "absl/base/config.h"
>+
>+// ABSL_BLOCK_TAIL_CALL_OPTIMIZATION
>+//
>+// Instructs the compiler to avoid optimizing tail-call recursion. Use of this
>+// macro is useful when you wish to preserve the existing function order within
>+// a stack trace for logging, debugging, or profiling purposes.
>+//
>+// Example:
>+//
>+//   int f() {
>+//     int result = g();
>+//     ABSL_BLOCK_TAIL_CALL_OPTIMIZATION();
>+//     return result;
>+//   }
>+#if defined(__pnacl__)
>+#define ABSL_BLOCK_TAIL_CALL_OPTIMIZATION() if (volatile int x = 0) { (void)x; }
>+#elif defined(__clang__)
>+// Clang will not tail call given inline volatile assembly.
>+#define ABSL_BLOCK_TAIL_CALL_OPTIMIZATION() __asm__ __volatile__("")
>+#elif defined(__GNUC__)
>+// GCC will not tail call given inline volatile assembly.
>+#define ABSL_BLOCK_TAIL_CALL_OPTIMIZATION() __asm__ __volatile__("")
>+#elif defined(_MSC_VER)
>+#include <intrin.h>
>+// The __nop() intrinsic blocks the optimisation.
>+#define ABSL_BLOCK_TAIL_CALL_OPTIMIZATION() __nop()
>+#else
>+#define ABSL_BLOCK_TAIL_CALL_OPTIMIZATION() if (volatile int x = 0) { (void)x; }
>+#endif
>+
>+// ABSL_CACHELINE_SIZE
>+//
>+// Explicitly defines the size of the L1 cache for purposes of alignment.
>+// Setting the cacheline size allows you to specify that certain objects be
>+// aligned on a cacheline boundary with `ABSL_CACHELINE_ALIGNED` declarations.
>+// (See below.)
>+//
>+// NOTE: this macro should be replaced with the following C++17 features, when
>+// those are generally available:
>+//
>+//   * `std::hardware_constructive_interference_size`
>+//   * `std::hardware_destructive_interference_size`
>+//
>+// See http://www.open-std.org/jtc1/sc22/wg21/docs/papers/2016/p0154r1.html
>+// for more information.
>+#if defined(__GNUC__)
>+// Cache line alignment
>+#if defined(__i386__) || defined(__x86_64__)
>+#define ABSL_CACHELINE_SIZE 64
>+#elif defined(__powerpc64__)
>+#define ABSL_CACHELINE_SIZE 128
>+#elif defined(__aarch64__)
>+// We would need to read special register ctr_el0 to find out L1 dcache size.
>+// This value is a good estimate based on a real aarch64 machine.
>+#define ABSL_CACHELINE_SIZE 64
>+#elif defined(__arm__)
>+// Cache line sizes for ARM: These values are not strictly correct since
>+// cache line sizes depend on implementations, not architectures.  There
>+// are even implementations with cache line sizes configurable at boot
>+// time.
>+#if defined(__ARM_ARCH_5T__)
>+#define ABSL_CACHELINE_SIZE 32
>+#elif defined(__ARM_ARCH_7A__)
>+#define ABSL_CACHELINE_SIZE 64
>+#endif
>+#endif
>+
>+#ifndef ABSL_CACHELINE_SIZE
>+// A reasonable default guess.  Note that overestimates tend to waste more
>+// space, while underestimates tend to waste more time.
>+#define ABSL_CACHELINE_SIZE 64
>+#endif
>+
>+// ABSL_CACHELINE_ALIGNED
>+//
>+// Indicates that the declared object be cache aligned using
>+// `ABSL_CACHELINE_SIZE` (see above). Cacheline aligning objects allows you to
>+// load a set of related objects in the L1 cache for performance improvements.
>+// Cacheline aligning objects properly allows constructive memory sharing and
>+// prevents destructive (or "false") memory sharing.
>+//
>+// NOTE: this macro should be replaced with usage of `alignas()` using
>+// `std::hardware_constructive_interference_size` and/or
>+// `std::hardware_destructive_interference_size` when available within C++17.
>+//
>+// See http://www.open-std.org/jtc1/sc22/wg21/docs/papers/2016/p0154r1.html
>+// for more information.
>+//
>+// On some compilers, `ABSL_CACHELINE_ALIGNED` expands to
>+// `__attribute__((aligned(ABSL_CACHELINE_SIZE)))`. For compilers where this is
>+// not known to work, the macro expands to nothing.
>+//
>+// No further guarantees are made here. The result of applying the macro
>+// to variables and types is always implementation-defined.
>+//
>+// WARNING: It is easy to use this attribute incorrectly, even to the point
>+// of causing bugs that are difficult to diagnose, crash, etc. It does not
>+// of itself guarantee that objects are aligned to a cache line.
>+//
>+// Recommendations:
>+//
>+// 1) Consult compiler documentation; this comment is not kept in sync as
>+//    toolchains evolve.
>+// 2) Verify your use has the intended effect. This often requires inspecting
>+//    the generated machine code.
>+// 3) Prefer applying this attribute to individual variables. Avoid
>+//    applying it to types. This tends to localize the effect.
>+#define ABSL_CACHELINE_ALIGNED __attribute__((aligned(ABSL_CACHELINE_SIZE)))
>+
>+#else  // not GCC
>+#define ABSL_CACHELINE_SIZE 64
>+#define ABSL_CACHELINE_ALIGNED
>+#endif
>+
>+// ABSL_PREDICT_TRUE, ABSL_PREDICT_FALSE
>+//
>+// Enables the compiler to prioritize compilation using static analysis for
>+// likely paths within a boolean branch.
>+//
>+// Example:
>+//
>+//   if (ABSL_PREDICT_TRUE(expression)) {
>+//     return result;                        // Faster if more likely
>+//   } else {
>+//     return 0;
>+//   }
>+//
>+// Compilers can use the information that a certain branch is not likely to be
>+// taken (for instance, a CHECK failure) to optimize for the common case in
>+// the absence of better information (ie. compiling gcc with `-fprofile-arcs`).
>+#if ABSL_HAVE_BUILTIN(__builtin_expect) || \
>+    (defined(__GNUC__) && !defined(__clang__))
>+#define ABSL_PREDICT_FALSE(x) (__builtin_expect(x, 0))
>+#define ABSL_PREDICT_TRUE(x) (__builtin_expect(!!(x), 1))
>+#else
>+#define ABSL_PREDICT_FALSE(x) (x)
>+#define ABSL_PREDICT_TRUE(x) (x)
>+#endif
>+
>+#endif  // ABSL_BASE_OPTIMIZATION_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/policy_checks.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/policy_checks.h
>new file mode 100644
>index 00000000000..0a07fc035e1
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/policy_checks.h
>@@ -0,0 +1,121 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// -----------------------------------------------------------------------------
>+// File: policy_checks.h
>+// -----------------------------------------------------------------------------
>+//
>+// This header enforces a minimum set of policies at build time, such as the
>+// supported compiler and library versions. Unsupported configurations are
>+// reported with `#error`. This enforcement is best effort, so successfully
>+// compiling this header does not guarantee a supported configuration.
>+
>+#ifndef ABSL_BASE_POLICY_CHECKS_H_
>+#define ABSL_BASE_POLICY_CHECKS_H_
>+
>+// Included for the __GLIBC_PREREQ macro used below.
>+#include <limits.h>
>+
>+// Included for the _STLPORT_VERSION macro used below.
>+#if defined(__cplusplus)
>+#include <cstddef>
>+#endif
>+
>+// -----------------------------------------------------------------------------
>+// Operating System Check
>+// -----------------------------------------------------------------------------
>+
>+#if defined(__CYGWIN__)
>+#error "Cygwin is not supported."
>+#endif
>+
>+// -----------------------------------------------------------------------------
>+// Compiler Check
>+// -----------------------------------------------------------------------------
>+
>+// We support MSVC++ 14.0 update 2 and later.
>+// This minimum will go up.
>+#if defined(_MSC_FULL_VER) && _MSC_FULL_VER < 190023918 && !defined(__clang__)
>+#error "This package requires Visual Studio 2015 Update 2 or higher."
>+#endif
>+
>+// We support gcc 4.7 and later.
>+// This minimum will go up.
>+#if defined(__GNUC__) && !defined(__clang__)
>+#if __GNUC__ < 4 || (__GNUC__ == 4 && __GNUC_MINOR__ < 7)
>+#error "This package requires gcc 4.7 or higher."
>+#endif
>+#endif
>+
>+// We support Apple Xcode clang 4.2.1 (version 421.11.65) and later.
>+// This corresponds to Apple Xcode version 4.5.
>+// This minimum will go up.
>+#if defined(__apple_build_version__) && __apple_build_version__ < 4211165
>+#error "This package requires __apple_build_version__ of 4211165 or higher."
>+#endif
>+
>+// -----------------------------------------------------------------------------
>+// C++ Version Check
>+// -----------------------------------------------------------------------------
>+
>+// Enforce C++11 as the minimum.  Note that Visual Studio has not
>+// advanced __cplusplus despite being good enough for our purposes, so
>+// so we exempt it from the check.
>+#if defined(__cplusplus) && !defined(_MSC_VER)
>+#if __cplusplus < 201103L
>+#error "C++ versions less than C++11 are not supported."
>+#endif
>+#endif
>+
>+// -----------------------------------------------------------------------------
>+// Standard Library Check
>+// -----------------------------------------------------------------------------
>+
>+// We have chosen glibc 2.12 as the minimum as it was tagged for release
>+// in May, 2010 and includes some functionality used in Google software
>+// (for instance pthread_setname_np):
>+// https://sourceware.org/ml/libc-alpha/2010-05/msg00000.html
>+#if defined(__GLIBC__) && defined(__GLIBC_PREREQ)
>+#if !__GLIBC_PREREQ(2, 12)
>+#error "Minimum required version of glibc is 2.12."
>+#endif
>+#endif
>+
>+#if defined(_STLPORT_VERSION)
>+#error "STLPort is not supported."
>+#endif
>+
>+// -----------------------------------------------------------------------------
>+// `char` Size Check
>+// -----------------------------------------------------------------------------
>+
>+// Abseil currently assumes CHAR_BIT == 8. If you would like to use Abseil on a
>+// platform where this is not the case, please provide us with the details about
>+// your platform so we can consider relaxing this requirement.
>+#if CHAR_BIT != 8
>+#error "Abseil assumes CHAR_BIT == 8."
>+#endif
>+
>+// -----------------------------------------------------------------------------
>+// `int` Size Check
>+// -----------------------------------------------------------------------------
>+
>+// Abseil currently assumes that an int is 4 bytes. If you would like to use
>+// Abseil on a platform where this is not the case, please provide us with the
>+// details about your platform so we can consider relaxing this requirement.
>+#if INT_MAX < 2147483647
>+#error "Abseil assumes that int is at least 4 bytes. "
>+#endif
>+
>+#endif  // ABSL_BASE_POLICY_CHECKS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/port.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/port.h
>new file mode 100644
>index 00000000000..1c67257fd8f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/port.h
>@@ -0,0 +1,26 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// This files is a forwarding header for other headers containing various
>+// portability macros and functions.
>+// This file is used for both C and C++!
>+
>+#ifndef ABSL_BASE_PORT_H_
>+#define ABSL_BASE_PORT_H_
>+
>+#include "absl/base/attributes.h"
>+#include "absl/base/config.h"
>+#include "absl/base/optimization.h"
>+
>+#endif  // ABSL_BASE_PORT_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/raw_logging_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/raw_logging_test.cc
>new file mode 100644
>index 00000000000..ebbc5db9067
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/raw_logging_test.cc
>@@ -0,0 +1,79 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+// This test serves primarily as a compilation test for base/raw_logging.h.
>+// Raw logging testing is covered by logging_unittest.cc, which is not as
>+// portable as this test.
>+
>+#include "absl/base/internal/raw_logging.h"
>+
>+#include <tuple>
>+
>+#include "gtest/gtest.h"
>+#include "absl/strings/str_cat.h"
>+
>+namespace {
>+
>+TEST(RawLoggingCompilationTest, Log) {
>+  ABSL_RAW_LOG(INFO, "RAW INFO: %d", 1);
>+  ABSL_RAW_LOG(INFO, "RAW INFO: %d %d", 1, 2);
>+  ABSL_RAW_LOG(INFO, "RAW INFO: %d %d %d", 1, 2, 3);
>+  ABSL_RAW_LOG(INFO, "RAW INFO: %d %d %d %d", 1, 2, 3, 4);
>+  ABSL_RAW_LOG(INFO, "RAW INFO: %d %d %d %d %d", 1, 2, 3, 4, 5);
>+  ABSL_RAW_LOG(WARNING, "RAW WARNING: %d", 1);
>+  ABSL_RAW_LOG(ERROR, "RAW ERROR: %d", 1);
>+}
>+
>+TEST(RawLoggingCompilationTest, PassingCheck) {
>+  ABSL_RAW_CHECK(true, "RAW CHECK");
>+}
>+
>+// Not all platforms support output from raw log, so we don't verify any
>+// particular output for RAW check failures (expecting the empty std::string
>+// accomplishes this).  This test is primarily a compilation test, but we
>+// are verifying process death when EXPECT_DEATH works for a platform.
>+const char kExpectedDeathOutput[] = "";
>+
>+TEST(RawLoggingDeathTest, FailingCheck) {
>+  EXPECT_DEATH_IF_SUPPORTED(ABSL_RAW_CHECK(1 == 0, "explanation"),
>+                            kExpectedDeathOutput);
>+}
>+
>+TEST(RawLoggingDeathTest, LogFatal) {
>+  EXPECT_DEATH_IF_SUPPORTED(ABSL_RAW_LOG(FATAL, "my dog has fleas"),
>+                            kExpectedDeathOutput);
>+}
>+
>+TEST(InternalLog, CompilationTest) {
>+  ABSL_INTERNAL_LOG(INFO, "Internal Log");
>+  std::string log_msg = "Internal Log";
>+  ABSL_INTERNAL_LOG(INFO, log_msg);
>+
>+  ABSL_INTERNAL_LOG(INFO, log_msg + " 2");
>+
>+  float d = 1.1f;
>+  ABSL_INTERNAL_LOG(INFO, absl::StrCat("Internal log ", 3, " + ", d));
>+}
>+
>+TEST(InternalLogDeathTest, FailingCheck) {
>+  EXPECT_DEATH_IF_SUPPORTED(ABSL_INTERNAL_CHECK(1 == 0, "explanation"),
>+                            kExpectedDeathOutput);
>+}
>+
>+TEST(InternalLogDeathTest, LogFatal) {
>+  EXPECT_DEATH_IF_SUPPORTED(ABSL_INTERNAL_LOG(FATAL, "my dog has fleas"),
>+                            kExpectedDeathOutput);
>+}
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/spinlock_test_common.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/spinlock_test_common.cc
>new file mode 100644
>index 00000000000..1b508848611
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/spinlock_test_common.cc
>@@ -0,0 +1,266 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+// A bunch of threads repeatedly hash an array of ints protected by a
>+// spinlock.  If the spinlock is working properly, all elements of the
>+// array should be equal at the end of the test.
>+
>+#include <cstdint>
>+#include <limits>
>+#include <random>
>+#include <thread>  // NOLINT(build/c++11)
>+#include <vector>
>+
>+#include "gtest/gtest.h"
>+#include "absl/base/attributes.h"
>+#include "absl/base/internal/low_level_scheduling.h"
>+#include "absl/base/internal/scheduling_mode.h"
>+#include "absl/base/internal/spinlock.h"
>+#include "absl/base/internal/sysinfo.h"
>+#include "absl/base/macros.h"
>+#include "absl/synchronization/blocking_counter.h"
>+#include "absl/synchronization/notification.h"
>+
>+constexpr int32_t kNumThreads = 10;
>+constexpr int32_t kIters = 1000;
>+
>+namespace absl {
>+namespace base_internal {
>+
>+// This is defined outside of anonymous namespace so that it can be
>+// a friend of SpinLock to access protected methods for testing.
>+struct SpinLockTest {
>+  static uint32_t EncodeWaitCycles(int64_t wait_start_time,
>+                                   int64_t wait_end_time) {
>+    return SpinLock::EncodeWaitCycles(wait_start_time, wait_end_time);
>+  }
>+  static uint64_t DecodeWaitCycles(uint32_t lock_value) {
>+    return SpinLock::DecodeWaitCycles(lock_value);
>+  }
>+};
>+
>+namespace {
>+
>+static constexpr int kArrayLength = 10;
>+static uint32_t values[kArrayLength];
>+static SpinLock static_spinlock(base_internal::kLinkerInitialized);
>+static SpinLock static_cooperative_spinlock(
>+    base_internal::kLinkerInitialized,
>+    base_internal::SCHEDULE_COOPERATIVE_AND_KERNEL);
>+static SpinLock static_noncooperative_spinlock(
>+    base_internal::kLinkerInitialized, base_internal::SCHEDULE_KERNEL_ONLY);
>+
>+
>+// Simple integer hash function based on the public domain lookup2 hash.
>+// http://burtleburtle.net/bob/c/lookup2.c
>+static uint32_t Hash32(uint32_t a, uint32_t c) {
>+  uint32_t b = 0x9e3779b9UL;  // The golden ratio; an arbitrary value.
>+  a -= b; a -= c; a ^= (c >> 13);
>+  b -= c; b -= a; b ^= (a << 8);
>+  c -= a; c -= b; c ^= (b >> 13);
>+  a -= b; a -= c; a ^= (c >> 12);
>+  b -= c; b -= a; b ^= (a << 16);
>+  c -= a; c -= b; c ^= (b >> 5);
>+  a -= b; a -= c; a ^= (c >> 3);
>+  b -= c; b -= a; b ^= (a << 10);
>+  c -= a; c -= b; c ^= (b >> 15);
>+  return c;
>+}
>+
>+static void TestFunction(int thread_salt, SpinLock* spinlock) {
>+  for (int i = 0; i < kIters; i++) {
>+    SpinLockHolder h(spinlock);
>+    for (int j = 0; j < kArrayLength; j++) {
>+      const int index = (j + thread_salt) % kArrayLength;
>+      values[index] = Hash32(values[index], thread_salt);
>+      std::this_thread::yield();
>+    }
>+  }
>+}
>+
>+static void ThreadedTest(SpinLock* spinlock) {
>+  std::vector<std::thread> threads;
>+  for (int i = 0; i < kNumThreads; ++i) {
>+    threads.push_back(std::thread(TestFunction, i, spinlock));
>+  }
>+  for (auto& thread : threads) {
>+    thread.join();
>+  }
>+
>+  SpinLockHolder h(spinlock);
>+  for (int i = 1; i < kArrayLength; i++) {
>+    EXPECT_EQ(values[0], values[i]);
>+  }
>+}
>+
>+TEST(SpinLock, StackNonCooperativeDisablesScheduling) {
>+  SpinLock spinlock(base_internal::SCHEDULE_KERNEL_ONLY);
>+  spinlock.Lock();
>+  EXPECT_FALSE(base_internal::SchedulingGuard::ReschedulingIsAllowed());
>+  spinlock.Unlock();
>+}
>+
>+TEST(SpinLock, StaticNonCooperativeDisablesScheduling) {
>+  static_noncooperative_spinlock.Lock();
>+  EXPECT_FALSE(base_internal::SchedulingGuard::ReschedulingIsAllowed());
>+  static_noncooperative_spinlock.Unlock();
>+}
>+
>+TEST(SpinLock, WaitCyclesEncoding) {
>+  // These are implementation details not exported by SpinLock.
>+  const int kProfileTimestampShift = 7;
>+  const int kLockwordReservedShift = 3;
>+  const uint32_t kSpinLockSleeper = 8;
>+
>+  // We should be able to encode up to (1^kMaxCycleBits - 1) without clamping
>+  // but the lower kProfileTimestampShift will be dropped.
>+  const int kMaxCyclesShift =
>+    32 - kLockwordReservedShift + kProfileTimestampShift;
>+  const uint64_t kMaxCycles = (int64_t{1} << kMaxCyclesShift) - 1;
>+
>+  // These bits should be zero after encoding.
>+  const uint32_t kLockwordReservedMask = (1 << kLockwordReservedShift) - 1;
>+
>+  // These bits are dropped when wait cycles are encoded.
>+  const uint64_t kProfileTimestampMask = (1 << kProfileTimestampShift) - 1;
>+
>+  // Test a bunch of random values
>+  std::default_random_engine generator;
>+  // Shift to avoid overflow below.
>+  std::uniform_int_distribution<uint64_t> time_distribution(
>+      0, std::numeric_limits<uint64_t>::max() >> 4);
>+  std::uniform_int_distribution<uint64_t> cycle_distribution(0, kMaxCycles);
>+
>+  for (int i = 0; i < 100; i++) {
>+    int64_t start_time = time_distribution(generator);
>+    int64_t cycles = cycle_distribution(generator);
>+    int64_t end_time = start_time + cycles;
>+    uint32_t lock_value = SpinLockTest::EncodeWaitCycles(start_time, end_time);
>+    EXPECT_EQ(0, lock_value & kLockwordReservedMask);
>+    uint64_t decoded = SpinLockTest::DecodeWaitCycles(lock_value);
>+    EXPECT_EQ(0, decoded & kProfileTimestampMask);
>+    EXPECT_EQ(cycles & ~kProfileTimestampMask, decoded);
>+  }
>+
>+  // Test corner cases
>+  int64_t start_time = time_distribution(generator);
>+  EXPECT_EQ(0, SpinLockTest::EncodeWaitCycles(start_time, start_time));
>+  EXPECT_EQ(0, SpinLockTest::DecodeWaitCycles(0));
>+  EXPECT_EQ(0, SpinLockTest::DecodeWaitCycles(kLockwordReservedMask));
>+  EXPECT_EQ(kMaxCycles & ~kProfileTimestampMask,
>+            SpinLockTest::DecodeWaitCycles(~kLockwordReservedMask));
>+
>+  // Check that we cannot produce kSpinLockSleeper during encoding.
>+  int64_t sleeper_cycles =
>+      kSpinLockSleeper << (kProfileTimestampShift - kLockwordReservedShift);
>+  uint32_t sleeper_value =
>+      SpinLockTest::EncodeWaitCycles(start_time, start_time + sleeper_cycles);
>+  EXPECT_NE(sleeper_value, kSpinLockSleeper);
>+
>+  // Test clamping
>+  uint32_t max_value =
>+    SpinLockTest::EncodeWaitCycles(start_time, start_time + kMaxCycles);
>+  uint64_t max_value_decoded = SpinLockTest::DecodeWaitCycles(max_value);
>+  uint64_t expected_max_value_decoded = kMaxCycles & ~kProfileTimestampMask;
>+  EXPECT_EQ(expected_max_value_decoded, max_value_decoded);
>+
>+  const int64_t step = (1 << kProfileTimestampShift);
>+  uint32_t after_max_value =
>+    SpinLockTest::EncodeWaitCycles(start_time, start_time + kMaxCycles + step);
>+  uint64_t after_max_value_decoded =
>+      SpinLockTest::DecodeWaitCycles(after_max_value);
>+  EXPECT_EQ(expected_max_value_decoded, after_max_value_decoded);
>+
>+  uint32_t before_max_value = SpinLockTest::EncodeWaitCycles(
>+      start_time, start_time + kMaxCycles - step);
>+  uint64_t before_max_value_decoded =
>+    SpinLockTest::DecodeWaitCycles(before_max_value);
>+  EXPECT_GT(expected_max_value_decoded, before_max_value_decoded);
>+}
>+TEST(SpinLockWithThreads, StaticSpinLock) {
>+  ThreadedTest(&static_spinlock);
>+}
>+TEST(SpinLockWithThreads, StackSpinLock) {
>+  SpinLock spinlock;
>+  ThreadedTest(&spinlock);
>+}
>+
>+TEST(SpinLockWithThreads, StackCooperativeSpinLock) {
>+  SpinLock spinlock(base_internal::SCHEDULE_COOPERATIVE_AND_KERNEL);
>+  ThreadedTest(&spinlock);
>+}
>+
>+TEST(SpinLockWithThreads, StackNonCooperativeSpinLock) {
>+  SpinLock spinlock(base_internal::SCHEDULE_KERNEL_ONLY);
>+  ThreadedTest(&spinlock);
>+}
>+
>+TEST(SpinLockWithThreads, StaticCooperativeSpinLock) {
>+  ThreadedTest(&static_cooperative_spinlock);
>+}
>+
>+TEST(SpinLockWithThreads, StaticNonCooperativeSpinLock) {
>+  ThreadedTest(&static_noncooperative_spinlock);
>+}
>+
>+TEST(SpinLockWithThreads, DoesNotDeadlock) {
>+  struct Helper {
>+    static void NotifyThenLock(Notification* locked, SpinLock* spinlock,
>+                               BlockingCounter* b) {
>+      locked->WaitForNotification();  // Wait for LockThenWait() to hold "s".
>+      b->DecrementCount();
>+      SpinLockHolder l(spinlock);
>+    }
>+
>+    static void LockThenWait(Notification* locked, SpinLock* spinlock,
>+                             BlockingCounter* b) {
>+      SpinLockHolder l(spinlock);
>+      locked->Notify();
>+      b->Wait();
>+    }
>+
>+    static void DeadlockTest(SpinLock* spinlock, int num_spinners) {
>+      Notification locked;
>+      BlockingCounter counter(num_spinners);
>+      std::vector<std::thread> threads;
>+
>+      threads.push_back(
>+          std::thread(Helper::LockThenWait, &locked, spinlock, &counter));
>+      for (int i = 0; i < num_spinners; ++i) {
>+        threads.push_back(
>+            std::thread(Helper::NotifyThenLock, &locked, spinlock, &counter));
>+      }
>+
>+      for (auto& thread : threads) {
>+        thread.join();
>+      }
>+    }
>+  };
>+
>+  SpinLock stack_cooperative_spinlock(
>+      base_internal::SCHEDULE_COOPERATIVE_AND_KERNEL);
>+  SpinLock stack_noncooperative_spinlock(base_internal::SCHEDULE_KERNEL_ONLY);
>+  Helper::DeadlockTest(&stack_cooperative_spinlock,
>+                       base_internal::NumCPUs() * 2);
>+  Helper::DeadlockTest(&stack_noncooperative_spinlock,
>+                       base_internal::NumCPUs() * 2);
>+  Helper::DeadlockTest(&static_cooperative_spinlock,
>+                       base_internal::NumCPUs() * 2);
>+  Helper::DeadlockTest(&static_noncooperative_spinlock,
>+                       base_internal::NumCPUs() * 2);
>+}
>+
>+}  // namespace
>+}  // namespace base_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/thread_annotations.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/thread_annotations.h
>new file mode 100644
>index 00000000000..fbb2797b825
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/thread_annotations.h
>@@ -0,0 +1,257 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// -----------------------------------------------------------------------------
>+// File: thread_annotations.h
>+// -----------------------------------------------------------------------------
>+//
>+// This header file contains macro definitions for thread safety annotations
>+// that allow developers to document the locking policies of multi-threaded
>+// code. The annotations can also help program analysis tools to identify
>+// potential thread safety issues.
>+//
>+//
>+// These annotations are implemented using compiler attributes. Using the macros
>+// defined here instead of raw attributes allow for portability and future
>+// compatibility.
>+//
>+// When referring to mutexes in the arguments of the attributes, you should
>+// use variable names or more complex expressions (e.g. my_object->mutex_)
>+// that evaluate to a concrete mutex object whenever possible. If the mutex
>+// you want to refer to is not in scope, you may use a member pointer
>+// (e.g. &MyClass::mutex_) to refer to a mutex in some (unknown) object.
>+
>+#ifndef ABSL_BASE_THREAD_ANNOTATIONS_H_
>+#define ABSL_BASE_THREAD_ANNOTATIONS_H_
>+#if defined(__clang__)
>+#define THREAD_ANNOTATION_ATTRIBUTE__(x)   __attribute__((x))
>+#else
>+#define THREAD_ANNOTATION_ATTRIBUTE__(x)   // no-op
>+#endif
>+
>+// GUARDED_BY()
>+//
>+// Documents if a shared field or global variable needs to be protected by a
>+// mutex. GUARDED_BY() allows the user to specify a particular mutex that
>+// should be held when accessing the annotated variable.
>+//
>+// Although this annotation (and PT_GUARDED_BY, below) cannot be applied to
>+// local variables, a local variable and its associated mutex can often be
>+// combined into a small class or struct, thereby allowing the annotation.
>+//
>+// Example:
>+//
>+//   class Foo {
>+//     Mutex mu_;
>+//     int p1_ GUARDED_BY(mu_);
>+//     ...
>+//   };
>+#define GUARDED_BY(x) THREAD_ANNOTATION_ATTRIBUTE__(guarded_by(x))
>+
>+// PT_GUARDED_BY()
>+//
>+// Documents if the memory location pointed to by a pointer should be guarded
>+// by a mutex when dereferencing the pointer.
>+//
>+// Example:
>+//   class Foo {
>+//     Mutex mu_;
>+//     int *p1_ PT_GUARDED_BY(mu_);
>+//     ...
>+//   };
>+//
>+// Note that a pointer variable to a shared memory location could itself be a
>+// shared variable.
>+//
>+// Example:
>+//
>+//   // `q_`, guarded by `mu1_`, points to a shared memory location that is
>+//   // guarded by `mu2_`:
>+//   int *q_ GUARDED_BY(mu1_) PT_GUARDED_BY(mu2_);
>+#define PT_GUARDED_BY(x) THREAD_ANNOTATION_ATTRIBUTE__(pt_guarded_by(x))
>+
>+// ACQUIRED_AFTER() / ACQUIRED_BEFORE()
>+//
>+// Documents the acquisition order between locks that can be held
>+// simultaneously by a thread. For any two locks that need to be annotated
>+// to establish an acquisition order, only one of them needs the annotation.
>+// (i.e. You don't have to annotate both locks with both ACQUIRED_AFTER
>+// and ACQUIRED_BEFORE.)
>+//
>+// As with GUARDED_BY, this is only applicable to mutexes that are shared
>+// fields or global variables.
>+//
>+// Example:
>+//
>+//   Mutex m1_;
>+//   Mutex m2_ ACQUIRED_AFTER(m1_);
>+#define ACQUIRED_AFTER(...) \
>+  THREAD_ANNOTATION_ATTRIBUTE__(acquired_after(__VA_ARGS__))
>+
>+#define ACQUIRED_BEFORE(...) \
>+  THREAD_ANNOTATION_ATTRIBUTE__(acquired_before(__VA_ARGS__))
>+
>+// EXCLUSIVE_LOCKS_REQUIRED() / SHARED_LOCKS_REQUIRED()
>+//
>+// Documents a function that expects a mutex to be held prior to entry.
>+// The mutex is expected to be held both on entry to, and exit from, the
>+// function.
>+//
>+// Example:
>+//
>+//   Mutex mu1, mu2;
>+//   int a GUARDED_BY(mu1);
>+//   int b GUARDED_BY(mu2);
>+//
>+//   void foo() EXCLUSIVE_LOCKS_REQUIRED(mu1, mu2) { ... };
>+#define EXCLUSIVE_LOCKS_REQUIRED(...) \
>+  THREAD_ANNOTATION_ATTRIBUTE__(exclusive_locks_required(__VA_ARGS__))
>+
>+#define SHARED_LOCKS_REQUIRED(...) \
>+  THREAD_ANNOTATION_ATTRIBUTE__(shared_locks_required(__VA_ARGS__))
>+
>+// LOCKS_EXCLUDED()
>+//
>+// Documents the locks acquired in the body of the function. These locks
>+// cannot be held when calling this function (as Abseil's `Mutex` locks are
>+// non-reentrant).
>+#define LOCKS_EXCLUDED(...) \
>+  THREAD_ANNOTATION_ATTRIBUTE__(locks_excluded(__VA_ARGS__))
>+
>+// LOCK_RETURNED()
>+//
>+// Documents a function that returns a mutex without acquiring it.  For example,
>+// a public getter method that returns a pointer to a private mutex should
>+// be annotated with LOCK_RETURNED.
>+#define LOCK_RETURNED(x) \
>+  THREAD_ANNOTATION_ATTRIBUTE__(lock_returned(x))
>+
>+// LOCKABLE
>+//
>+// Documents if a class/type is a lockable type (such as the `Mutex` class).
>+#define LOCKABLE \
>+  THREAD_ANNOTATION_ATTRIBUTE__(lockable)
>+
>+// SCOPED_LOCKABLE
>+//
>+// Documents if a class does RAII locking (such as the `MutexLock` class).
>+// The constructor should use `LOCK_FUNCTION()` to specify the mutex that is
>+// acquired, and the destructor should use `UNLOCK_FUNCTION()` with no
>+// arguments; the analysis will assume that the destructor unlocks whatever the
>+// constructor locked.
>+#define SCOPED_LOCKABLE \
>+  THREAD_ANNOTATION_ATTRIBUTE__(scoped_lockable)
>+
>+// EXCLUSIVE_LOCK_FUNCTION()
>+//
>+// Documents functions that acquire a lock in the body of a function, and do
>+// not release it.
>+#define EXCLUSIVE_LOCK_FUNCTION(...) \
>+  THREAD_ANNOTATION_ATTRIBUTE__(exclusive_lock_function(__VA_ARGS__))
>+
>+// SHARED_LOCK_FUNCTION()
>+//
>+// Documents functions that acquire a shared (reader) lock in the body of a
>+// function, and do not release it.
>+#define SHARED_LOCK_FUNCTION(...) \
>+  THREAD_ANNOTATION_ATTRIBUTE__(shared_lock_function(__VA_ARGS__))
>+
>+// UNLOCK_FUNCTION()
>+//
>+// Documents functions that expect a lock to be held on entry to the function,
>+// and release it in the body of the function.
>+#define UNLOCK_FUNCTION(...) \
>+  THREAD_ANNOTATION_ATTRIBUTE__(unlock_function(__VA_ARGS__))
>+
>+// EXCLUSIVE_TRYLOCK_FUNCTION() / SHARED_TRYLOCK_FUNCTION()
>+//
>+// Documents functions that try to acquire a lock, and return success or failure
>+// (or a non-boolean value that can be interpreted as a boolean).
>+// The first argument should be `true` for functions that return `true` on
>+// success, or `false` for functions that return `false` on success. The second
>+// argument specifies the mutex that is locked on success. If unspecified, this
>+// mutex is assumed to be `this`.
>+#define EXCLUSIVE_TRYLOCK_FUNCTION(...) \
>+  THREAD_ANNOTATION_ATTRIBUTE__(exclusive_trylock_function(__VA_ARGS__))
>+
>+#define SHARED_TRYLOCK_FUNCTION(...) \
>+  THREAD_ANNOTATION_ATTRIBUTE__(shared_trylock_function(__VA_ARGS__))
>+
>+// ASSERT_EXCLUSIVE_LOCK() / ASSERT_SHARED_LOCK()
>+//
>+// Documents functions that dynamically check to see if a lock is held, and fail
>+// if it is not held.
>+#define ASSERT_EXCLUSIVE_LOCK(...) \
>+  THREAD_ANNOTATION_ATTRIBUTE__(assert_exclusive_lock(__VA_ARGS__))
>+
>+#define ASSERT_SHARED_LOCK(...) \
>+  THREAD_ANNOTATION_ATTRIBUTE__(assert_shared_lock(__VA_ARGS__))
>+
>+// NO_THREAD_SAFETY_ANALYSIS
>+//
>+// Turns off thread safety checking within the body of a particular function.
>+// This annotation is used to mark functions that are known to be correct, but
>+// the locking behavior is more complicated than the analyzer can handle.
>+#define NO_THREAD_SAFETY_ANALYSIS \
>+  THREAD_ANNOTATION_ATTRIBUTE__(no_thread_safety_analysis)
>+
>+//------------------------------------------------------------------------------
>+// Tool-Supplied Annotations
>+//------------------------------------------------------------------------------
>+
>+// TS_UNCHECKED should be placed around lock expressions that are not valid
>+// C++ syntax, but which are present for documentation purposes.  These
>+// annotations will be ignored by the analysis.
>+#define TS_UNCHECKED(x) ""
>+
>+// TS_FIXME is used to mark lock expressions that are not valid C++ syntax.
>+// It is used by automated tools to mark and disable invalid expressions.
>+// The annotation should either be fixed, or changed to TS_UNCHECKED.
>+#define TS_FIXME(x) ""
>+
>+// Like NO_THREAD_SAFETY_ANALYSIS, this turns off checking within the body of
>+// a particular function.  However, this attribute is used to mark functions
>+// that are incorrect and need to be fixed.  It is used by automated tools to
>+// avoid breaking the build when the analysis is updated.
>+// Code owners are expected to eventually fix the routine.
>+#define NO_THREAD_SAFETY_ANALYSIS_FIXME  NO_THREAD_SAFETY_ANALYSIS
>+
>+// Similar to NO_THREAD_SAFETY_ANALYSIS_FIXME, this macro marks a GUARDED_BY
>+// annotation that needs to be fixed, because it is producing thread safety
>+// warning.  It disables the GUARDED_BY.
>+#define GUARDED_BY_FIXME(x)
>+
>+// Disables warnings for a single read operation.  This can be used to avoid
>+// warnings when it is known that the read is not actually involved in a race,
>+// but the compiler cannot confirm that.
>+#define TS_UNCHECKED_READ(x) thread_safety_analysis::ts_unchecked_read(x)
>+
>+
>+namespace thread_safety_analysis {
>+
>+// Takes a reference to a guarded data member, and returns an unguarded
>+// reference.
>+template <typename T>
>+inline const T& ts_unchecked_read(const T& v) NO_THREAD_SAFETY_ANALYSIS {
>+  return v;
>+}
>+
>+template <typename T>
>+inline T& ts_unchecked_read(T& v) NO_THREAD_SAFETY_ANALYSIS {
>+  return v;
>+}
>+
>+}  // namespace thread_safety_analysis
>+
>+#endif  // ABSL_BASE_THREAD_ANNOTATIONS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/throw_delegate_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/throw_delegate_test.cc
>new file mode 100644
>index 00000000000..0f15df0492b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/base/throw_delegate_test.cc
>@@ -0,0 +1,94 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/base/internal/throw_delegate.h"
>+
>+#include <functional>
>+#include <new>
>+#include <stdexcept>
>+
>+#include "gtest/gtest.h"
>+
>+namespace {
>+
>+using absl::base_internal::ThrowStdLogicError;
>+using absl::base_internal::ThrowStdInvalidArgument;
>+using absl::base_internal::ThrowStdDomainError;
>+using absl::base_internal::ThrowStdLengthError;
>+using absl::base_internal::ThrowStdOutOfRange;
>+using absl::base_internal::ThrowStdRuntimeError;
>+using absl::base_internal::ThrowStdRangeError;
>+using absl::base_internal::ThrowStdOverflowError;
>+using absl::base_internal::ThrowStdUnderflowError;
>+using absl::base_internal::ThrowStdBadFunctionCall;
>+using absl::base_internal::ThrowStdBadAlloc;
>+
>+constexpr const char* what_arg = "The quick brown fox jumps over the lazy dog";
>+
>+template <typename E>
>+void ExpectThrowChar(void (*f)(const char*)) {
>+  try {
>+    f(what_arg);
>+    FAIL() << "Didn't throw";
>+  } catch (const E& e) {
>+    EXPECT_STREQ(e.what(), what_arg);
>+  }
>+}
>+
>+template <typename E>
>+void ExpectThrowString(void (*f)(const std::string&)) {
>+  try {
>+    f(what_arg);
>+    FAIL() << "Didn't throw";
>+  } catch (const E& e) {
>+    EXPECT_STREQ(e.what(), what_arg);
>+  }
>+}
>+
>+template <typename E>
>+void ExpectThrowNoWhat(void (*f)()) {
>+  try {
>+    f();
>+    FAIL() << "Didn't throw";
>+  } catch (const E& e) {
>+  }
>+}
>+
>+TEST(ThrowHelper, Test) {
>+  // Not using EXPECT_THROW because we want to check the .what() message too.
>+  ExpectThrowChar<std::logic_error>(ThrowStdLogicError);
>+  ExpectThrowChar<std::invalid_argument>(ThrowStdInvalidArgument);
>+  ExpectThrowChar<std::domain_error>(ThrowStdDomainError);
>+  ExpectThrowChar<std::length_error>(ThrowStdLengthError);
>+  ExpectThrowChar<std::out_of_range>(ThrowStdOutOfRange);
>+  ExpectThrowChar<std::runtime_error>(ThrowStdRuntimeError);
>+  ExpectThrowChar<std::range_error>(ThrowStdRangeError);
>+  ExpectThrowChar<std::overflow_error>(ThrowStdOverflowError);
>+  ExpectThrowChar<std::underflow_error>(ThrowStdUnderflowError);
>+
>+  ExpectThrowString<std::logic_error>(ThrowStdLogicError);
>+  ExpectThrowString<std::invalid_argument>(ThrowStdInvalidArgument);
>+  ExpectThrowString<std::domain_error>(ThrowStdDomainError);
>+  ExpectThrowString<std::length_error>(ThrowStdLengthError);
>+  ExpectThrowString<std::out_of_range>(ThrowStdOutOfRange);
>+  ExpectThrowString<std::runtime_error>(ThrowStdRuntimeError);
>+  ExpectThrowString<std::range_error>(ThrowStdRangeError);
>+  ExpectThrowString<std::overflow_error>(ThrowStdOverflowError);
>+  ExpectThrowString<std::underflow_error>(ThrowStdUnderflowError);
>+
>+  ExpectThrowNoWhat<std::bad_function_call>(ThrowStdBadFunctionCall);
>+  ExpectThrowNoWhat<std::bad_alloc>(ThrowStdBadAlloc);
>+}
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/compiler_config_setting.bzl b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/compiler_config_setting.bzl
>new file mode 100644
>index 00000000000..b77c4f563b9
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/compiler_config_setting.bzl
>@@ -0,0 +1,39 @@
>+#
>+# Copyright 2018 The Abseil Authors.
>+#
>+# Licensed under the Apache License, Version 2.0 (the "License");
>+# you may not use this file except in compliance with the License.
>+# You may obtain a copy of the License at
>+#
>+#      http://www.apache.org/licenses/LICENSE-2.0
>+#
>+# Unless required by applicable law or agreed to in writing, software
>+# distributed under the License is distributed on an "AS IS" BASIS,
>+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+# See the License for the specific language governing permissions and
>+# limitations under the License.
>+#
>+
>+"""Creates config_setting that allows selecting based on 'compiler' value."""
>+
>+def create_llvm_config(name, visibility):
>+  # The "do_not_use_tools_cpp_compiler_present" attribute exists to
>+  # distinguish between older versions of Bazel that do not support
>+  # "@bazel_tools//tools/cpp:compiler" flag_value, and newer ones that do.
>+  # In the future, the only way to select on the compiler will be through
>+  # flag_values{"@bazel_tools//tools/cpp:compiler"} and the else branch can
>+  # be removed.
>+  if hasattr(cc_common, "do_not_use_tools_cpp_compiler_present"):
>+    native.config_setting(
>+      name = name,
>+      flag_values = {
>+          "@bazel_tools//tools/cpp:compiler": "llvm",
>+      },
>+      visibility = visibility,
>+    )
>+  else:
>+    native.config_setting(
>+        name = name,
>+        values = {"compiler": "llvm"},
>+        visibility = visibility,
>+    )
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/BUILD.bazel b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/BUILD.bazel
>new file mode 100644
>index 00000000000..6d5c958f382
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/BUILD.bazel
>@@ -0,0 +1,183 @@
>+#
>+# Copyright 2017 The Abseil Authors.
>+#
>+# Licensed under the Apache License, Version 2.0 (the "License");
>+# you may not use this file except in compliance with the License.
>+# You may obtain a copy of the License at
>+#
>+#      http://www.apache.org/licenses/LICENSE-2.0
>+#
>+# Unless required by applicable law or agreed to in writing, software
>+# distributed under the License is distributed on an "AS IS" BASIS,
>+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+# See the License for the specific language governing permissions and
>+# limitations under the License.
>+#
>+
>+load(
>+    "//absl:copts.bzl",
>+    "ABSL_DEFAULT_COPTS",
>+    "ABSL_TEST_COPTS",
>+    "ABSL_EXCEPTIONS_FLAG",
>+)
>+
>+package(default_visibility = ["//visibility:public"])
>+
>+licenses(["notice"])  # Apache 2.0
>+
>+cc_library(
>+    name = "compressed_tuple",
>+    hdrs = ["internal/compressed_tuple.h"],
>+    copts = ABSL_DEFAULT_COPTS,
>+    deps = [
>+        "//absl/utility",
>+    ],
>+)
>+
>+cc_test(
>+    name = "compressed_tuple_test",
>+    srcs = ["internal/compressed_tuple_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    deps = [
>+        ":compressed_tuple",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_library(
>+    name = "fixed_array",
>+    hdrs = ["fixed_array.h"],
>+    copts = ABSL_DEFAULT_COPTS,
>+    deps = [
>+        ":compressed_tuple",
>+        "//absl/algorithm",
>+        "//absl/base:core_headers",
>+        "//absl/base:dynamic_annotations",
>+        "//absl/base:throw_delegate",
>+        "//absl/memory",
>+    ],
>+)
>+
>+cc_test(
>+    name = "fixed_array_test",
>+    srcs = ["fixed_array_test.cc"],
>+    copts = ABSL_TEST_COPTS + ABSL_EXCEPTIONS_FLAG,
>+    deps = [
>+        ":fixed_array",
>+        "//absl/base:exception_testing",
>+        "//absl/memory",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "fixed_array_test_noexceptions",
>+    srcs = ["fixed_array_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    deps = [
>+        ":fixed_array",
>+        "//absl/base:exception_testing",
>+        "//absl/memory",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "fixed_array_exception_safety_test",
>+    srcs = ["fixed_array_exception_safety_test.cc"],
>+    copts = ABSL_TEST_COPTS + ABSL_EXCEPTIONS_FLAG,
>+    deps = [
>+        ":fixed_array",
>+        "//absl/base:exception_safety_testing",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "fixed_array_benchmark",
>+    srcs = ["fixed_array_benchmark.cc"],
>+    copts = ABSL_TEST_COPTS + ["$(STACK_FRAME_UNLIMITED)"],
>+    tags = ["benchmark"],
>+    deps = [
>+        ":fixed_array",
>+        "@com_github_google_benchmark//:benchmark_main",
>+    ],
>+)
>+
>+cc_library(
>+    name = "inlined_vector",
>+    hdrs = ["inlined_vector.h"],
>+    copts = ABSL_DEFAULT_COPTS,
>+    deps = [
>+        "//absl/algorithm",
>+        "//absl/base:core_headers",
>+        "//absl/base:throw_delegate",
>+        "//absl/memory",
>+    ],
>+)
>+
>+cc_test(
>+    name = "inlined_vector_test",
>+    srcs = ["inlined_vector_test.cc"],
>+    copts = ABSL_TEST_COPTS + ABSL_EXCEPTIONS_FLAG,
>+    deps = [
>+        ":inlined_vector",
>+        ":test_instance_tracker",
>+        "//absl/base",
>+        "//absl/base:core_headers",
>+        "//absl/base:exception_testing",
>+        "//absl/memory",
>+        "//absl/strings",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "inlined_vector_test_noexceptions",
>+    srcs = ["inlined_vector_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    deps = [
>+        ":inlined_vector",
>+        ":test_instance_tracker",
>+        "//absl/base",
>+        "//absl/base:core_headers",
>+        "//absl/base:exception_testing",
>+        "//absl/memory",
>+        "//absl/strings",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "inlined_vector_benchmark",
>+    srcs = ["inlined_vector_benchmark.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    tags = ["benchmark"],
>+    deps = [
>+        ":inlined_vector",
>+        "//absl/base",
>+        "//absl/strings",
>+        "@com_github_google_benchmark//:benchmark_main",
>+    ],
>+)
>+
>+cc_library(
>+    name = "test_instance_tracker",
>+    testonly = 1,
>+    srcs = ["internal/test_instance_tracker.cc"],
>+    hdrs = ["internal/test_instance_tracker.h"],
>+    copts = ABSL_DEFAULT_COPTS,
>+    visibility = [
>+        "//absl:__subpackages__",
>+    ],
>+)
>+
>+cc_test(
>+    name = "test_instance_tracker_test",
>+    srcs = ["internal/test_instance_tracker_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    deps = [
>+        ":test_instance_tracker",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/BUILD.gn
>new file mode 100644
>index 00000000000..001a2a36d48
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/BUILD.gn
>@@ -0,0 +1,86 @@
>+# Copyright 2018 The Chromium Authors. All rights reserved.
>+# Use of this source code is governed by a BSD-style license that can be
>+# found in the LICENSE file.
>+
>+import("//build_overrides/build.gni")
>+
>+if (build_with_chromium) {
>+  visibility = [
>+    "//third_party/webrtc/*",
>+    "//third_party/abseil-cpp/*",
>+    "//third_party/googletest:gtest",
>+  ]
>+} else {
>+  visibility = [ "*" ]
>+}
>+
>+source_set("compressed_tuple") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  public = [
>+    "internal/compressed_tuple.h",
>+  ]
>+  deps = [
>+    "../utility",
>+  ]
>+}
>+
>+source_set("fixed_array") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  public = [
>+    "fixed_array.h",
>+  ]
>+  deps = [
>+    ":compressed_tuple",
>+    "../algorithm",
>+    "../base:core_headers",
>+    "../base:dynamic_annotations",
>+    "../base:throw_delegate",
>+    "../memory",
>+  ]
>+}
>+
>+source_set("inlined_vector") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  public = [
>+    "inlined_vector.h",
>+  ]
>+  deps = [
>+    "../algorithm",
>+    "../base:core_headers",
>+    "../base:throw_delegate",
>+    "../memory",
>+  ]
>+}
>+
>+source_set("test_instance_tracker") {
>+  testonly = true
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  sources = [
>+    "internal/test_instance_tracker.cc",
>+  ]
>+  public = [
>+    "internal/test_instance_tracker.h",
>+  ]
>+  visibility = []
>+  visibility += [ "../*" ]
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/CMakeLists.txt b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/CMakeLists.txt
>new file mode 100644
>index 00000000000..123e4c4849a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/CMakeLists.txt
>@@ -0,0 +1,144 @@
>+#
>+# Copyright 2017 The Abseil Authors.
>+#
>+# Licensed under the Apache License, Version 2.0 (the "License");
>+# you may not use this file except in compliance with the License.
>+# You may obtain a copy of the License at
>+#
>+#      http://www.apache.org/licenses/LICENSE-2.0
>+#
>+# Unless required by applicable law or agreed to in writing, software
>+# distributed under the License is distributed on an "AS IS" BASIS,
>+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+# See the License for the specific language governing permissions and
>+# limitations under the License.
>+#
>+
>+
>+list(APPEND CONTAINER_PUBLIC_HEADERS
>+  "fixed_array.h"
>+  "inlined_vector.h"
>+)
>+
>+
>+list(APPEND CONTAINER_INTERNAL_HEADERS
>+  "internal/test_instance_tracker.h"
>+)
>+
>+
>+absl_header_library(
>+  TARGET
>+    absl_container
>+  EXPORT_NAME
>+    container
>+)
>+
>+
>+#
>+## TESTS
>+#
>+
>+list(APPEND TEST_INSTANCE_TRACKER_LIB_SRC
>+  "internal/test_instance_tracker.cc"
>+  ${CONTAINER_PUBLIC_HEADERS}
>+  ${CONTAINER_INTERNAL_HEADERS}
>+)
>+
>+
>+absl_library(
>+  TARGET
>+    test_instance_tracker_lib
>+  SOURCES
>+    ${TEST_INSTANCE_TRACKER_LIB_SRC}
>+  PUBLIC_LIBRARIES
>+    absl::container
>+)
>+
>+
>+
>+# test fixed_array_test
>+set(FIXED_ARRAY_TEST_SRC "fixed_array_test.cc")
>+set(FIXED_ARRAY_TEST_PUBLIC_LIBRARIES absl::base absl_throw_delegate test_instance_tracker_lib)
>+
>+absl_test(
>+  TARGET
>+    fixed_array_test
>+  SOURCES
>+    ${FIXED_ARRAY_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${FIXED_ARRAY_TEST_PUBLIC_LIBRARIES}
>+  PRIVATE_COMPILE_FLAGS
>+    ${ABSL_EXCEPTIONS_FLAG}
>+)
>+
>+
>+
>+absl_test(
>+  TARGET
>+    fixed_array_test_noexceptions
>+  SOURCES
>+    ${FIXED_ARRAY_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${FIXED_ARRAY_TEST_PUBLIC_LIBRARIES}
>+)
>+
>+
>+# test fixed_array_exception_safety_test
>+set(FIXED_ARRAY_EXCEPTION_SAFETY_TEST_SRC "fixed_array_exception_safety_test.cc")
>+set(FIXED_ARRAY_EXCEPTION_SAFETY_TEST_PUBLIC_LIBRARIES
>+  absl::container
>+  absl_base_internal_exception_safety_testing
>+)
>+
>+absl_test(
>+  TARGET
>+    fixed_array_exception_safety_test
>+  SOURCES
>+    ${FIXED_ARRAY_EXCEPTION_SAFETY_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${FIXED_ARRAY_EXCEPTION_SAFETY_TEST_PUBLIC_LIBRARIES}
>+  PRIVATE_COMPILE_FLAGS
>+    ${ABSL_EXCEPTIONS_FLAG}
>+)
>+
>+
>+# test inlined_vector_test
>+set(INLINED_VECTOR_TEST_SRC "inlined_vector_test.cc")
>+set(INLINED_VECTOR_TEST_PUBLIC_LIBRARIES absl::base absl_throw_delegate test_instance_tracker_lib)
>+
>+absl_test(
>+  TARGET
>+    inlined_vector_test
>+  SOURCES
>+    ${INLINED_VECTOR_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${INLINED_VECTOR_TEST_PUBLIC_LIBRARIES}
>+)
>+
>+absl_test(
>+  TARGET
>+    inlined_vector_test_noexceptions
>+  SOURCES
>+    ${INLINED_VECTOR_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${INLINED_VECTOR_TEST_PUBLIC_LIBRARIES}
>+  PRIVATE_COMPILE_FLAGS
>+    ${ABSL_NOEXCEPTION_CXXFLAGS}
>+)
>+
>+
>+# test test_instance_tracker_test
>+set(TEST_INSTANCE_TRACKER_TEST_SRC "internal/test_instance_tracker_test.cc")
>+set(TEST_INSTANCE_TRACKER_TEST_PUBLIC_LIBRARIES absl::base absl_throw_delegate test_instance_tracker_lib)
>+
>+
>+absl_test(
>+  TARGET
>+    test_instance_tracker_test
>+  SOURCES
>+    ${TEST_INSTANCE_TRACKER_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${TEST_INSTANCE_TRACKER_TEST_PUBLIC_LIBRARIES}
>+)
>+
>+
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/fixed_array.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/fixed_array.h
>new file mode 100644
>index 00000000000..182258f623e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/fixed_array.h
>@@ -0,0 +1,512 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// -----------------------------------------------------------------------------
>+// File: fixed_array.h
>+// -----------------------------------------------------------------------------
>+//
>+// A `FixedArray<T>` represents a non-resizable array of `T` where the length of
>+// the array can be determined at run-time. It is a good replacement for
>+// non-standard and deprecated uses of `alloca()` and variable length arrays
>+// within the GCC extension. (See
>+// https://gcc.gnu.org/onlinedocs/gcc/Variable-Length.html).
>+//
>+// `FixedArray` allocates small arrays inline, keeping performance fast by
>+// avoiding heap operations. It also helps reduce the chances of
>+// accidentally overflowing your stack if large input is passed to
>+// your function.
>+
>+#ifndef ABSL_CONTAINER_FIXED_ARRAY_H_
>+#define ABSL_CONTAINER_FIXED_ARRAY_H_
>+
>+#include <algorithm>
>+#include <array>
>+#include <cassert>
>+#include <cstddef>
>+#include <initializer_list>
>+#include <iterator>
>+#include <limits>
>+#include <memory>
>+#include <new>
>+#include <type_traits>
>+
>+#include "absl/algorithm/algorithm.h"
>+#include "absl/base/dynamic_annotations.h"
>+#include "absl/base/internal/throw_delegate.h"
>+#include "absl/base/macros.h"
>+#include "absl/base/optimization.h"
>+#include "absl/base/port.h"
>+#include "absl/container/internal/compressed_tuple.h"
>+#include "absl/memory/memory.h"
>+
>+namespace absl {
>+
>+constexpr static auto kFixedArrayUseDefault = static_cast<size_t>(-1);
>+
>+// -----------------------------------------------------------------------------
>+// FixedArray
>+// -----------------------------------------------------------------------------
>+//
>+// A `FixedArray` provides a run-time fixed-size array, allocating a small array
>+// inline for efficiency.
>+//
>+// Most users should not specify an `inline_elements` argument and let
>+// `FixedArray` automatically determine the number of elements
>+// to store inline based on `sizeof(T)`. If `inline_elements` is specified, the
>+// `FixedArray` implementation will use inline storage for arrays with a
>+// length <= `inline_elements`.
>+//
>+// Note that a `FixedArray` constructed with a `size_type` argument will
>+// default-initialize its values by leaving trivially constructible types
>+// uninitialized (e.g. int, int[4], double), and others default-constructed.
>+// This matches the behavior of c-style arrays and `std::array`, but not
>+// `std::vector`.
>+//
>+// Note that `FixedArray` does not provide a public allocator; if it requires a
>+// heap allocation, it will do so with global `::operator new[]()` and
>+// `::operator delete[]()`, even if T provides class-scope overrides for these
>+// operators.
>+template <typename T, size_t N = kFixedArrayUseDefault,
>+          typename A = std::allocator<T>>
>+class FixedArray {
>+  static_assert(!std::is_array<T>::value || std::extent<T>::value > 0,
>+                "Arrays with unknown bounds cannot be used with FixedArray.");
>+
>+  static constexpr size_t kInlineBytesDefault = 256;
>+
>+  using AllocatorTraits = std::allocator_traits<A>;
>+  // std::iterator_traits isn't guaranteed to be SFINAE-friendly until C++17,
>+  // but this seems to be mostly pedantic.
>+  template <typename Iterator>
>+  using EnableIfForwardIterator = absl::enable_if_t<std::is_convertible<
>+      typename std::iterator_traits<Iterator>::iterator_category,
>+      std::forward_iterator_tag>::value>;
>+  static constexpr bool NoexceptCopyable() {
>+    return std::is_nothrow_copy_constructible<StorageElement>::value &&
>+           absl::allocator_is_nothrow<allocator_type>::value;
>+  }
>+  static constexpr bool NoexceptMovable() {
>+    return std::is_nothrow_move_constructible<StorageElement>::value &&
>+           absl::allocator_is_nothrow<allocator_type>::value;
>+  }
>+  static constexpr bool DefaultConstructorIsNonTrivial() {
>+    return !absl::is_trivially_default_constructible<StorageElement>::value;
>+  }
>+
>+ public:
>+  using allocator_type = typename AllocatorTraits::allocator_type;
>+  using value_type = typename allocator_type::value_type;
>+  using pointer = typename allocator_type::pointer;
>+  using const_pointer = typename allocator_type::const_pointer;
>+  using reference = typename allocator_type::reference;
>+  using const_reference = typename allocator_type::const_reference;
>+  using size_type = typename allocator_type::size_type;
>+  using difference_type = typename allocator_type::difference_type;
>+  using iterator = pointer;
>+  using const_iterator = const_pointer;
>+  using reverse_iterator = std::reverse_iterator<iterator>;
>+  using const_reverse_iterator = std::reverse_iterator<const_iterator>;
>+
>+  static constexpr size_type inline_elements =
>+      (N == kFixedArrayUseDefault ? kInlineBytesDefault / sizeof(value_type)
>+                                  : static_cast<size_type>(N));
>+
>+  FixedArray(
>+      const FixedArray& other,
>+      const allocator_type& a = allocator_type()) noexcept(NoexceptCopyable())
>+      : FixedArray(other.begin(), other.end(), a) {}
>+
>+  FixedArray(
>+      FixedArray&& other,
>+      const allocator_type& a = allocator_type()) noexcept(NoexceptMovable())
>+      : FixedArray(std::make_move_iterator(other.begin()),
>+                   std::make_move_iterator(other.end()), a) {}
>+
>+  // Creates an array object that can store `n` elements.
>+  // Note that trivially constructible elements will be uninitialized.
>+  explicit FixedArray(size_type n, const allocator_type& a = allocator_type())
>+      : storage_(n, a) {
>+    if (DefaultConstructorIsNonTrivial()) {
>+      memory_internal::ConstructStorage(storage_.alloc(), storage_.begin(),
>+                                        storage_.end());
>+    }
>+  }
>+
>+  // Creates an array initialized with `n` copies of `val`.
>+  FixedArray(size_type n, const value_type& val,
>+             const allocator_type& a = allocator_type())
>+      : storage_(n, a) {
>+    memory_internal::ConstructStorage(storage_.alloc(), storage_.begin(),
>+                                      storage_.end(), val);
>+  }
>+
>+  // Creates an array initialized with the size and contents of `init_list`.
>+  FixedArray(std::initializer_list<value_type> init_list,
>+             const allocator_type& a = allocator_type())
>+      : FixedArray(init_list.begin(), init_list.end(), a) {}
>+
>+  // Creates an array initialized with the elements from the input
>+  // range. The array's size will always be `std::distance(first, last)`.
>+  // REQUIRES: Iterator must be a forward_iterator or better.
>+  template <typename Iterator, EnableIfForwardIterator<Iterator>* = nullptr>
>+  FixedArray(Iterator first, Iterator last,
>+             const allocator_type& a = allocator_type())
>+      : storage_(std::distance(first, last), a) {
>+    memory_internal::CopyToStorageFromRange(storage_.alloc(), storage_.begin(),
>+                                            first, last);
>+  }
>+
>+  ~FixedArray() noexcept {
>+    for (auto* cur = storage_.begin(); cur != storage_.end(); ++cur) {
>+      AllocatorTraits::destroy(*storage_.alloc(), cur);
>+    }
>+  }
>+
>+  // Assignments are deleted because they break the invariant that the size of a
>+  // `FixedArray` never changes.
>+  void operator=(FixedArray&&) = delete;
>+  void operator=(const FixedArray&) = delete;
>+
>+  // FixedArray::size()
>+  //
>+  // Returns the length of the fixed array.
>+  size_type size() const { return storage_.size(); }
>+
>+  // FixedArray::max_size()
>+  //
>+  // Returns the largest possible value of `std::distance(begin(), end())` for a
>+  // `FixedArray<T>`. This is equivalent to the most possible addressable bytes
>+  // over the number of bytes taken by T.
>+  constexpr size_type max_size() const {
>+    return std::numeric_limits<difference_type>::max() / sizeof(value_type);
>+  }
>+
>+  // FixedArray::empty()
>+  //
>+  // Returns whether or not the fixed array is empty.
>+  bool empty() const { return size() == 0; }
>+
>+  // FixedArray::memsize()
>+  //
>+  // Returns the memory size of the fixed array in bytes.
>+  size_t memsize() const { return size() * sizeof(value_type); }
>+
>+  // FixedArray::data()
>+  //
>+  // Returns a const T* pointer to elements of the `FixedArray`. This pointer
>+  // can be used to access (but not modify) the contained elements.
>+  const_pointer data() const { return AsValueType(storage_.begin()); }
>+
>+  // Overload of FixedArray::data() to return a T* pointer to elements of the
>+  // fixed array. This pointer can be used to access and modify the contained
>+  // elements.
>+  pointer data() { return AsValueType(storage_.begin()); }
>+
>+  // FixedArray::operator[]
>+  //
>+  // Returns a reference the ith element of the fixed array.
>+  // REQUIRES: 0 <= i < size()
>+  reference operator[](size_type i) {
>+    assert(i < size());
>+    return data()[i];
>+  }
>+
>+  // Overload of FixedArray::operator()[] to return a const reference to the
>+  // ith element of the fixed array.
>+  // REQUIRES: 0 <= i < size()
>+  const_reference operator[](size_type i) const {
>+    assert(i < size());
>+    return data()[i];
>+  }
>+
>+  // FixedArray::at
>+  //
>+  // Bounds-checked access.  Returns a reference to the ith element of the
>+  // fiexed array, or throws std::out_of_range
>+  reference at(size_type i) {
>+    if (ABSL_PREDICT_FALSE(i >= size())) {
>+      base_internal::ThrowStdOutOfRange("FixedArray::at failed bounds check");
>+    }
>+    return data()[i];
>+  }
>+
>+  // Overload of FixedArray::at() to return a const reference to the ith element
>+  // of the fixed array.
>+  const_reference at(size_type i) const {
>+    if (ABSL_PREDICT_FALSE(i >= size())) {
>+      base_internal::ThrowStdOutOfRange("FixedArray::at failed bounds check");
>+    }
>+    return data()[i];
>+  }
>+
>+  // FixedArray::front()
>+  //
>+  // Returns a reference to the first element of the fixed array.
>+  reference front() { return *begin(); }
>+
>+  // Overload of FixedArray::front() to return a reference to the first element
>+  // of a fixed array of const values.
>+  const_reference front() const { return *begin(); }
>+
>+  // FixedArray::back()
>+  //
>+  // Returns a reference to the last element of the fixed array.
>+  reference back() { return *(end() - 1); }
>+
>+  // Overload of FixedArray::back() to return a reference to the last element
>+  // of a fixed array of const values.
>+  const_reference back() const { return *(end() - 1); }
>+
>+  // FixedArray::begin()
>+  //
>+  // Returns an iterator to the beginning of the fixed array.
>+  iterator begin() { return data(); }
>+
>+  // Overload of FixedArray::begin() to return a const iterator to the
>+  // beginning of the fixed array.
>+  const_iterator begin() const { return data(); }
>+
>+  // FixedArray::cbegin()
>+  //
>+  // Returns a const iterator to the beginning of the fixed array.
>+  const_iterator cbegin() const { return begin(); }
>+
>+  // FixedArray::end()
>+  //
>+  // Returns an iterator to the end of the fixed array.
>+  iterator end() { return data() + size(); }
>+
>+  // Overload of FixedArray::end() to return a const iterator to the end of the
>+  // fixed array.
>+  const_iterator end() const { return data() + size(); }
>+
>+  // FixedArray::cend()
>+  //
>+  // Returns a const iterator to the end of the fixed array.
>+  const_iterator cend() const { return end(); }
>+
>+  // FixedArray::rbegin()
>+  //
>+  // Returns a reverse iterator from the end of the fixed array.
>+  reverse_iterator rbegin() { return reverse_iterator(end()); }
>+
>+  // Overload of FixedArray::rbegin() to return a const reverse iterator from
>+  // the end of the fixed array.
>+  const_reverse_iterator rbegin() const {
>+    return const_reverse_iterator(end());
>+  }
>+
>+  // FixedArray::crbegin()
>+  //
>+  // Returns a const reverse iterator from the end of the fixed array.
>+  const_reverse_iterator crbegin() const { return rbegin(); }
>+
>+  // FixedArray::rend()
>+  //
>+  // Returns a reverse iterator from the beginning of the fixed array.
>+  reverse_iterator rend() { return reverse_iterator(begin()); }
>+
>+  // Overload of FixedArray::rend() for returning a const reverse iterator
>+  // from the beginning of the fixed array.
>+  const_reverse_iterator rend() const {
>+    return const_reverse_iterator(begin());
>+  }
>+
>+  // FixedArray::crend()
>+  //
>+  // Returns a reverse iterator from the beginning of the fixed array.
>+  const_reverse_iterator crend() const { return rend(); }
>+
>+  // FixedArray::fill()
>+  //
>+  // Assigns the given `value` to all elements in the fixed array.
>+  void fill(const value_type& val) { std::fill(begin(), end(), val); }
>+
>+  // Relational operators. Equality operators are elementwise using
>+  // `operator==`, while order operators order FixedArrays lexicographically.
>+  friend bool operator==(const FixedArray& lhs, const FixedArray& rhs) {
>+    return absl::equal(lhs.begin(), lhs.end(), rhs.begin(), rhs.end());
>+  }
>+
>+  friend bool operator!=(const FixedArray& lhs, const FixedArray& rhs) {
>+    return !(lhs == rhs);
>+  }
>+
>+  friend bool operator<(const FixedArray& lhs, const FixedArray& rhs) {
>+    return std::lexicographical_compare(lhs.begin(), lhs.end(), rhs.begin(),
>+                                        rhs.end());
>+  }
>+
>+  friend bool operator>(const FixedArray& lhs, const FixedArray& rhs) {
>+    return rhs < lhs;
>+  }
>+
>+  friend bool operator<=(const FixedArray& lhs, const FixedArray& rhs) {
>+    return !(rhs < lhs);
>+  }
>+
>+  friend bool operator>=(const FixedArray& lhs, const FixedArray& rhs) {
>+    return !(lhs < rhs);
>+  }
>+ private:
>+  // StorageElement
>+  //
>+  // For FixedArrays with a C-style-array value_type, StorageElement is a POD
>+  // wrapper struct called StorageElementWrapper that holds the value_type
>+  // instance inside. This is needed for construction and destruction of the
>+  // entire array regardless of how many dimensions it has. For all other cases,
>+  // StorageElement is just an alias of value_type.
>+  //
>+  // Maintainer's Note: The simpler solution would be to simply wrap value_type
>+  // in a struct whether it's an array or not. That causes some paranoid
>+  // diagnostics to misfire, believing that 'data()' returns a pointer to a
>+  // single element, rather than the packed array that it really is.
>+  // e.g.:
>+  //
>+  //     FixedArray<char> buf(1);
>+  //     sprintf(buf.data(), "foo");
>+  //
>+  //     error: call to int __builtin___sprintf_chk(etc...)
>+  //     will always overflow destination buffer [-Werror]
>+  //
>+  template <typename OuterT = value_type,
>+            typename InnerT = absl::remove_extent_t<OuterT>,
>+            size_t InnerN = std::extent<OuterT>::value>
>+  struct StorageElementWrapper {
>+    InnerT array[InnerN];
>+  };
>+
>+  using StorageElement =
>+      absl::conditional_t<std::is_array<value_type>::value,
>+                          StorageElementWrapper<value_type>, value_type>;
>+  using StorageElementBuffer =
>+      absl::aligned_storage_t<sizeof(StorageElement), alignof(StorageElement)>;
>+
>+  static pointer AsValueType(pointer ptr) { return ptr; }
>+  static pointer AsValueType(StorageElementWrapper<value_type>* ptr) {
>+    return std::addressof(ptr->array);
>+  }
>+
>+  static_assert(sizeof(StorageElement) == sizeof(value_type), "");
>+  static_assert(alignof(StorageElement) == alignof(value_type), "");
>+
>+  struct NonEmptyInlinedStorage {
>+    StorageElement* data() {
>+      return reinterpret_cast<StorageElement*>(inlined_storage_.data());
>+    }
>+
>+#ifdef ADDRESS_SANITIZER
>+    void* RedzoneBegin() { return &redzone_begin_; }
>+    void* RedzoneEnd() { return &redzone_end_ + 1; }
>+#endif  // ADDRESS_SANITIZER
>+
>+    void AnnotateConstruct(size_type);
>+    void AnnotateDestruct(size_type);
>+
>+    ABSL_ADDRESS_SANITIZER_REDZONE(redzone_begin_);
>+    std::array<StorageElementBuffer, inline_elements> inlined_storage_;
>+    ABSL_ADDRESS_SANITIZER_REDZONE(redzone_end_);
>+  };
>+
>+  struct EmptyInlinedStorage {
>+    StorageElement* data() { return nullptr; }
>+    void AnnotateConstruct(size_type) {}
>+    void AnnotateDestruct(size_type) {}
>+  };
>+
>+  using InlinedStorage =
>+      absl::conditional_t<inline_elements == 0, EmptyInlinedStorage,
>+                          NonEmptyInlinedStorage>;
>+
>+  // Storage
>+  //
>+  // An instance of Storage manages the inline and out-of-line memory for
>+  // instances of FixedArray. This guarantees that even when construction of
>+  // individual elements fails in the FixedArray constructor body, the
>+  // destructor for Storage will still be called and out-of-line memory will be
>+  // properly deallocated.
>+  //
>+  class Storage : public InlinedStorage {
>+   public:
>+    Storage(size_type n, const allocator_type& a)
>+        : size_alloc_(n, a), data_(InitializeData()) {}
>+
>+    ~Storage() noexcept {
>+      if (UsingInlinedStorage(size())) {
>+        InlinedStorage::AnnotateDestruct(size());
>+      } else {
>+        AllocatorTraits::deallocate(*alloc(), AsValueType(begin()), size());
>+      }
>+    }
>+
>+    size_type size() const { return size_alloc_.template get<0>(); }
>+    StorageElement* begin() const { return data_; }
>+    StorageElement* end() const { return begin() + size(); }
>+    allocator_type* alloc() {
>+      return std::addressof(size_alloc_.template get<1>());
>+    }
>+
>+   private:
>+    static bool UsingInlinedStorage(size_type n) {
>+      return n <= inline_elements;
>+    }
>+
>+    StorageElement* InitializeData() {
>+      if (UsingInlinedStorage(size())) {
>+        InlinedStorage::AnnotateConstruct(size());
>+        return InlinedStorage::data();
>+      } else {
>+        return reinterpret_cast<StorageElement*>(
>+            AllocatorTraits::allocate(*alloc(), size()));
>+      }
>+    }
>+
>+    // `CompressedTuple` takes advantage of EBCO for stateless `allocator_type`s
>+    container_internal::CompressedTuple<size_type, allocator_type> size_alloc_;
>+    StorageElement* data_;
>+  };
>+
>+  Storage storage_;
>+};
>+
>+template <typename T, size_t N, typename A>
>+constexpr size_t FixedArray<T, N, A>::kInlineBytesDefault;
>+
>+template <typename T, size_t N, typename A>
>+constexpr typename FixedArray<T, N, A>::size_type
>+    FixedArray<T, N, A>::inline_elements;
>+
>+template <typename T, size_t N, typename A>
>+void FixedArray<T, N, A>::NonEmptyInlinedStorage::AnnotateConstruct(
>+    typename FixedArray<T, N, A>::size_type n) {
>+#ifdef ADDRESS_SANITIZER
>+  if (!n) return;
>+  ABSL_ANNOTATE_CONTIGUOUS_CONTAINER(data(), RedzoneEnd(), RedzoneEnd(), data() + n);
>+  ABSL_ANNOTATE_CONTIGUOUS_CONTAINER(RedzoneBegin(), data(), data(), RedzoneBegin());
>+#endif                   // ADDRESS_SANITIZER
>+  static_cast<void>(n);  // Mark used when not in asan mode
>+}
>+
>+template <typename T, size_t N, typename A>
>+void FixedArray<T, N, A>::NonEmptyInlinedStorage::AnnotateDestruct(
>+    typename FixedArray<T, N, A>::size_type n) {
>+#ifdef ADDRESS_SANITIZER
>+  if (!n) return;
>+  ABSL_ANNOTATE_CONTIGUOUS_CONTAINER(data(), RedzoneEnd(), data() + n, RedzoneEnd());
>+  ABSL_ANNOTATE_CONTIGUOUS_CONTAINER(RedzoneBegin(), data(), RedzoneBegin(), data());
>+#endif                   // ADDRESS_SANITIZER
>+  static_cast<void>(n);  // Mark used when not in asan mode
>+}
>+}  // namespace absl
>+#endif  // ABSL_CONTAINER_FIXED_ARRAY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/fixed_array_benchmark.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/fixed_array_benchmark.cc
>new file mode 100644
>index 00000000000..b4f0cf2aeb0
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/fixed_array_benchmark.cc
>@@ -0,0 +1,66 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/container/fixed_array.h"
>+
>+#include <stddef.h>
>+#include <string>
>+
>+#include "benchmark/benchmark.h"
>+
>+namespace {
>+
>+// For benchmarking -- simple class with constructor and destructor that
>+// set an int to a constant..
>+class SimpleClass {
>+ public:
>+  SimpleClass() : i(3) { }
>+  ~SimpleClass() { i = 0; }
>+ private:
>+  int i;
>+};
>+
>+template <typename C, size_t stack_size>
>+void BM_FixedArray(benchmark::State& state) {
>+  const int size = state.range(0);
>+  for (auto _ : state) {
>+    absl::FixedArray<C, stack_size> fa(size);
>+    benchmark::DoNotOptimize(fa.data());
>+  }
>+}
>+BENCHMARK_TEMPLATE(BM_FixedArray, char, absl::kFixedArrayUseDefault)
>+    ->Range(0, 1 << 16);
>+BENCHMARK_TEMPLATE(BM_FixedArray, char, 0)->Range(0, 1 << 16);
>+BENCHMARK_TEMPLATE(BM_FixedArray, char, 1)->Range(0, 1 << 16);
>+BENCHMARK_TEMPLATE(BM_FixedArray, char, 16)->Range(0, 1 << 16);
>+BENCHMARK_TEMPLATE(BM_FixedArray, char, 256)->Range(0, 1 << 16);
>+BENCHMARK_TEMPLATE(BM_FixedArray, char, 65536)->Range(0, 1 << 16);
>+
>+BENCHMARK_TEMPLATE(BM_FixedArray, SimpleClass, absl::kFixedArrayUseDefault)
>+    ->Range(0, 1 << 16);
>+BENCHMARK_TEMPLATE(BM_FixedArray, SimpleClass, 0)->Range(0, 1 << 16);
>+BENCHMARK_TEMPLATE(BM_FixedArray, SimpleClass, 1)->Range(0, 1 << 16);
>+BENCHMARK_TEMPLATE(BM_FixedArray, SimpleClass, 16)->Range(0, 1 << 16);
>+BENCHMARK_TEMPLATE(BM_FixedArray, SimpleClass, 256)->Range(0, 1 << 16);
>+BENCHMARK_TEMPLATE(BM_FixedArray, SimpleClass, 65536)->Range(0, 1 << 16);
>+
>+BENCHMARK_TEMPLATE(BM_FixedArray, std::string, absl::kFixedArrayUseDefault)
>+    ->Range(0, 1 << 16);
>+BENCHMARK_TEMPLATE(BM_FixedArray, std::string, 0)->Range(0, 1 << 16);
>+BENCHMARK_TEMPLATE(BM_FixedArray, std::string, 1)->Range(0, 1 << 16);
>+BENCHMARK_TEMPLATE(BM_FixedArray, std::string, 16)->Range(0, 1 << 16);
>+BENCHMARK_TEMPLATE(BM_FixedArray, std::string, 256)->Range(0, 1 << 16);
>+BENCHMARK_TEMPLATE(BM_FixedArray, std::string, 65536)->Range(0, 1 << 16);
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/fixed_array_exception_safety_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/fixed_array_exception_safety_test.cc
>new file mode 100644
>index 00000000000..c123c2a1c0d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/fixed_array_exception_safety_test.cc
>@@ -0,0 +1,117 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include <initializer_list>
>+
>+#include "absl/container/fixed_array.h"
>+
>+#include "gtest/gtest.h"
>+#include "absl/base/internal/exception_safety_testing.h"
>+
>+namespace absl {
>+
>+namespace {
>+
>+constexpr size_t kInlined = 25;
>+constexpr size_t kSmallSize = kInlined / 2;
>+constexpr size_t kLargeSize = kInlined * 2;
>+
>+constexpr int kInitialValue = 5;
>+constexpr int kUpdatedValue = 10;
>+
>+using ::testing::TestThrowingCtor;
>+
>+using Thrower = testing::ThrowingValue<testing::TypeSpec::kEverythingThrows>;
>+using FixedArr = absl::FixedArray<Thrower, kInlined>;
>+
>+using MoveThrower = testing::ThrowingValue<testing::TypeSpec::kNoThrowMove>;
>+using MoveFixedArr = absl::FixedArray<MoveThrower, kInlined>;
>+
>+TEST(FixedArrayExceptionSafety, CopyConstructor) {
>+  auto small = FixedArr(kSmallSize);
>+  TestThrowingCtor<FixedArr>(small);
>+
>+  auto large = FixedArr(kLargeSize);
>+  TestThrowingCtor<FixedArr>(large);
>+}
>+
>+TEST(FixedArrayExceptionSafety, MoveConstructor) {
>+  TestThrowingCtor<FixedArr>(FixedArr(kSmallSize));
>+  TestThrowingCtor<FixedArr>(FixedArr(kLargeSize));
>+
>+  // TypeSpec::kNoThrowMove
>+  TestThrowingCtor<MoveFixedArr>(MoveFixedArr(kSmallSize));
>+  TestThrowingCtor<MoveFixedArr>(MoveFixedArr(kLargeSize));
>+}
>+
>+TEST(FixedArrayExceptionSafety, SizeConstructor) {
>+  TestThrowingCtor<FixedArr>(kSmallSize);
>+  TestThrowingCtor<FixedArr>(kLargeSize);
>+}
>+
>+TEST(FixedArrayExceptionSafety, SizeValueConstructor) {
>+  TestThrowingCtor<FixedArr>(kSmallSize, Thrower());
>+  TestThrowingCtor<FixedArr>(kLargeSize, Thrower());
>+}
>+
>+TEST(FixedArrayExceptionSafety, IteratorConstructor) {
>+  auto small = FixedArr(kSmallSize);
>+  TestThrowingCtor<FixedArr>(small.begin(), small.end());
>+
>+  auto large = FixedArr(kLargeSize);
>+  TestThrowingCtor<FixedArr>(large.begin(), large.end());
>+}
>+
>+TEST(FixedArrayExceptionSafety, InitListConstructor) {
>+  constexpr int small_inlined = 3;
>+  using SmallFixedArr = absl::FixedArray<Thrower, small_inlined>;
>+
>+  TestThrowingCtor<SmallFixedArr>(std::initializer_list<Thrower>{});
>+  // Test inlined allocation
>+  TestThrowingCtor<SmallFixedArr>(
>+      std::initializer_list<Thrower>{Thrower{}, Thrower{}});
>+  // Test out of line allocation
>+  TestThrowingCtor<SmallFixedArr>(std::initializer_list<Thrower>{
>+      Thrower{}, Thrower{}, Thrower{}, Thrower{}, Thrower{}});
>+}
>+
>+testing::AssertionResult ReadMemory(FixedArr* fixed_arr) {
>+  // Marked volatile to prevent optimization. Used for running asan tests.
>+  volatile int sum = 0;
>+  for (const auto& thrower : *fixed_arr) {
>+    sum += thrower.Get();
>+  }
>+  return testing::AssertionSuccess() << "Values sum to [" << sum << "]";
>+}
>+
>+TEST(FixedArrayExceptionSafety, Fill) {
>+  auto test_fill = testing::MakeExceptionSafetyTester()
>+                       .WithInvariants(ReadMemory)
>+                       .WithOperation([&](FixedArr* fixed_arr_ptr) {
>+                         auto thrower =
>+                             Thrower(kUpdatedValue, testing::nothrow_ctor);
>+                         fixed_arr_ptr->fill(thrower);
>+                       });
>+
>+  EXPECT_TRUE(
>+      test_fill.WithInitialValue(FixedArr(kSmallSize, Thrower(kInitialValue)))
>+          .Test());
>+  EXPECT_TRUE(
>+      test_fill.WithInitialValue(FixedArr(kLargeSize, Thrower(kInitialValue)))
>+          .Test());
>+}
>+
>+}  // namespace
>+
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/fixed_array_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/fixed_array_test.cc
>new file mode 100644
>index 00000000000..b07ebcb6d9c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/fixed_array_test.cc
>@@ -0,0 +1,870 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/container/fixed_array.h"
>+
>+#include <stdio.h>
>+#include <cstring>
>+#include <list>
>+#include <memory>
>+#include <numeric>
>+#include <scoped_allocator>
>+#include <stdexcept>
>+#include <string>
>+#include <vector>
>+
>+#include "gmock/gmock.h"
>+#include "gtest/gtest.h"
>+#include "absl/base/internal/exception_testing.h"
>+#include "absl/memory/memory.h"
>+
>+using ::testing::ElementsAreArray;
>+
>+namespace {
>+
>+// Helper routine to determine if a absl::FixedArray used stack allocation.
>+template <typename ArrayType>
>+static bool IsOnStack(const ArrayType& a) {
>+  return a.size() <= ArrayType::inline_elements;
>+}
>+
>+class ConstructionTester {
>+ public:
>+  ConstructionTester()
>+      : self_ptr_(this),
>+        value_(0) {
>+    constructions++;
>+  }
>+  ~ConstructionTester() {
>+    assert(self_ptr_ == this);
>+    self_ptr_ = nullptr;
>+    destructions++;
>+  }
>+
>+  // These are incremented as elements are constructed and destructed so we can
>+  // be sure all elements are properly cleaned up.
>+  static int constructions;
>+  static int destructions;
>+
>+  void CheckConstructed() {
>+    assert(self_ptr_ == this);
>+  }
>+
>+  void set(int value) { value_ = value; }
>+  int get() { return value_; }
>+
>+ private:
>+  // self_ptr_ should always point to 'this' -- that's how we can be sure the
>+  // constructor has been called.
>+  ConstructionTester* self_ptr_;
>+  int value_;
>+};
>+
>+int ConstructionTester::constructions = 0;
>+int ConstructionTester::destructions = 0;
>+
>+// ThreeInts will initialize its three ints to the value stored in
>+// ThreeInts::counter. The constructor increments counter so that each object
>+// in an array of ThreeInts will have different values.
>+class ThreeInts {
>+ public:
>+  ThreeInts() {
>+    x_ = counter;
>+    y_ = counter;
>+    z_ = counter;
>+    ++counter;
>+  }
>+
>+  static int counter;
>+
>+  int x_, y_, z_;
>+};
>+
>+int ThreeInts::counter = 0;
>+
>+TEST(FixedArrayTest, CopyCtor) {
>+  absl::FixedArray<int, 10> on_stack(5);
>+  std::iota(on_stack.begin(), on_stack.end(), 0);
>+  absl::FixedArray<int, 10> stack_copy = on_stack;
>+  EXPECT_THAT(stack_copy, ElementsAreArray(on_stack));
>+  EXPECT_TRUE(IsOnStack(stack_copy));
>+
>+  absl::FixedArray<int, 10> allocated(15);
>+  std::iota(allocated.begin(), allocated.end(), 0);
>+  absl::FixedArray<int, 10> alloced_copy = allocated;
>+  EXPECT_THAT(alloced_copy, ElementsAreArray(allocated));
>+  EXPECT_FALSE(IsOnStack(alloced_copy));
>+}
>+
>+TEST(FixedArrayTest, MoveCtor) {
>+  absl::FixedArray<std::unique_ptr<int>, 10> on_stack(5);
>+  for (int i = 0; i < 5; ++i) {
>+    on_stack[i] = absl::make_unique<int>(i);
>+  }
>+
>+  absl::FixedArray<std::unique_ptr<int>, 10> stack_copy = std::move(on_stack);
>+  for (int i = 0; i < 5; ++i) EXPECT_EQ(*(stack_copy[i]), i);
>+  EXPECT_EQ(stack_copy.size(), on_stack.size());
>+
>+  absl::FixedArray<std::unique_ptr<int>, 10> allocated(15);
>+  for (int i = 0; i < 15; ++i) {
>+    allocated[i] = absl::make_unique<int>(i);
>+  }
>+
>+  absl::FixedArray<std::unique_ptr<int>, 10> alloced_copy =
>+      std::move(allocated);
>+  for (int i = 0; i < 15; ++i) EXPECT_EQ(*(alloced_copy[i]), i);
>+  EXPECT_EQ(allocated.size(), alloced_copy.size());
>+}
>+
>+TEST(FixedArrayTest, SmallObjects) {
>+  // Small object arrays
>+  {
>+    // Short arrays should be on the stack
>+    absl::FixedArray<int> array(4);
>+    EXPECT_TRUE(IsOnStack(array));
>+  }
>+
>+  {
>+    // Large arrays should be on the heap
>+    absl::FixedArray<int> array(1048576);
>+    EXPECT_FALSE(IsOnStack(array));
>+  }
>+
>+  {
>+    // Arrays of <= default size should be on the stack
>+    absl::FixedArray<int, 100> array(100);
>+    EXPECT_TRUE(IsOnStack(array));
>+  }
>+
>+  {
>+    // Arrays of > default size should be on the stack
>+    absl::FixedArray<int, 100> array(101);
>+    EXPECT_FALSE(IsOnStack(array));
>+  }
>+
>+  {
>+    // Arrays with different size elements should use approximately
>+    // same amount of stack space
>+    absl::FixedArray<int> array1(0);
>+    absl::FixedArray<char> array2(0);
>+    EXPECT_LE(sizeof(array1), sizeof(array2)+100);
>+    EXPECT_LE(sizeof(array2), sizeof(array1)+100);
>+  }
>+
>+  {
>+    // Ensure that vectors are properly constructed inside a fixed array.
>+    absl::FixedArray<std::vector<int> > array(2);
>+    EXPECT_EQ(0, array[0].size());
>+    EXPECT_EQ(0, array[1].size());
>+  }
>+
>+  {
>+    // Regardless of absl::FixedArray implementation, check that a type with a
>+    // low alignment requirement and a non power-of-two size is initialized
>+    // correctly.
>+    ThreeInts::counter = 1;
>+    absl::FixedArray<ThreeInts> array(2);
>+    EXPECT_EQ(1, array[0].x_);
>+    EXPECT_EQ(1, array[0].y_);
>+    EXPECT_EQ(1, array[0].z_);
>+    EXPECT_EQ(2, array[1].x_);
>+    EXPECT_EQ(2, array[1].y_);
>+    EXPECT_EQ(2, array[1].z_);
>+  }
>+}
>+
>+TEST(FixedArrayTest, AtThrows) {
>+  absl::FixedArray<int> a = {1, 2, 3};
>+  EXPECT_EQ(a.at(2), 3);
>+  ABSL_BASE_INTERNAL_EXPECT_FAIL(a.at(3), std::out_of_range,
>+                                 "failed bounds check");
>+}
>+
>+TEST(FixedArrayRelationalsTest, EqualArrays) {
>+  for (int i = 0; i < 10; ++i) {
>+    absl::FixedArray<int, 5> a1(i);
>+    std::iota(a1.begin(), a1.end(), 0);
>+    absl::FixedArray<int, 5> a2(a1.begin(), a1.end());
>+
>+    EXPECT_TRUE(a1 == a2);
>+    EXPECT_FALSE(a1 != a2);
>+    EXPECT_TRUE(a2 == a1);
>+    EXPECT_FALSE(a2 != a1);
>+    EXPECT_FALSE(a1 < a2);
>+    EXPECT_FALSE(a1 > a2);
>+    EXPECT_FALSE(a2 < a1);
>+    EXPECT_FALSE(a2 > a1);
>+    EXPECT_TRUE(a1 <= a2);
>+    EXPECT_TRUE(a1 >= a2);
>+    EXPECT_TRUE(a2 <= a1);
>+    EXPECT_TRUE(a2 >= a1);
>+  }
>+}
>+
>+TEST(FixedArrayRelationalsTest, UnequalArrays) {
>+  for (int i = 1; i < 10; ++i) {
>+    absl::FixedArray<int, 5> a1(i);
>+    std::iota(a1.begin(), a1.end(), 0);
>+    absl::FixedArray<int, 5> a2(a1.begin(), a1.end());
>+    --a2[i / 2];
>+
>+    EXPECT_FALSE(a1 == a2);
>+    EXPECT_TRUE(a1 != a2);
>+    EXPECT_FALSE(a2 == a1);
>+    EXPECT_TRUE(a2 != a1);
>+    EXPECT_FALSE(a1 < a2);
>+    EXPECT_TRUE(a1 > a2);
>+    EXPECT_TRUE(a2 < a1);
>+    EXPECT_FALSE(a2 > a1);
>+    EXPECT_FALSE(a1 <= a2);
>+    EXPECT_TRUE(a1 >= a2);
>+    EXPECT_TRUE(a2 <= a1);
>+    EXPECT_FALSE(a2 >= a1);
>+  }
>+}
>+
>+template <int stack_elements>
>+static void TestArray(int n) {
>+  SCOPED_TRACE(n);
>+  SCOPED_TRACE(stack_elements);
>+  ConstructionTester::constructions = 0;
>+  ConstructionTester::destructions = 0;
>+  {
>+    absl::FixedArray<ConstructionTester, stack_elements> array(n);
>+
>+    EXPECT_THAT(array.size(), n);
>+    EXPECT_THAT(array.memsize(), sizeof(ConstructionTester) * n);
>+    EXPECT_THAT(array.begin() + n, array.end());
>+
>+    // Check that all elements were constructed
>+    for (int i = 0; i < n; i++) {
>+      array[i].CheckConstructed();
>+    }
>+    // Check that no other elements were constructed
>+    EXPECT_THAT(ConstructionTester::constructions, n);
>+
>+    // Test operator[]
>+    for (int i = 0; i < n; i++) {
>+      array[i].set(i);
>+    }
>+    for (int i = 0; i < n; i++) {
>+      EXPECT_THAT(array[i].get(), i);
>+      EXPECT_THAT(array.data()[i].get(), i);
>+    }
>+
>+    // Test data()
>+    for (int i = 0; i < n; i++) {
>+      array.data()[i].set(i + 1);
>+    }
>+    for (int i = 0; i < n; i++) {
>+      EXPECT_THAT(array[i].get(), i+1);
>+      EXPECT_THAT(array.data()[i].get(), i+1);
>+    }
>+  }  // Close scope containing 'array'.
>+
>+  // Check that all constructed elements were destructed.
>+  EXPECT_EQ(ConstructionTester::constructions,
>+            ConstructionTester::destructions);
>+}
>+
>+template <int elements_per_inner_array, int inline_elements>
>+static void TestArrayOfArrays(int n) {
>+  SCOPED_TRACE(n);
>+  SCOPED_TRACE(inline_elements);
>+  SCOPED_TRACE(elements_per_inner_array);
>+  ConstructionTester::constructions = 0;
>+  ConstructionTester::destructions = 0;
>+  {
>+    using InnerArray = ConstructionTester[elements_per_inner_array];
>+    // Heap-allocate the FixedArray to avoid blowing the stack frame.
>+    auto array_ptr =
>+        absl::make_unique<absl::FixedArray<InnerArray, inline_elements>>(n);
>+    auto& array = *array_ptr;
>+
>+    ASSERT_EQ(array.size(), n);
>+    ASSERT_EQ(array.memsize(),
>+             sizeof(ConstructionTester) * elements_per_inner_array * n);
>+    ASSERT_EQ(array.begin() + n, array.end());
>+
>+    // Check that all elements were constructed
>+    for (int i = 0; i < n; i++) {
>+      for (int j = 0; j < elements_per_inner_array; j++) {
>+        (array[i])[j].CheckConstructed();
>+      }
>+    }
>+    // Check that no other elements were constructed
>+    ASSERT_EQ(ConstructionTester::constructions, n * elements_per_inner_array);
>+
>+    // Test operator[]
>+    for (int i = 0; i < n; i++) {
>+      for (int j = 0; j < elements_per_inner_array; j++) {
>+        (array[i])[j].set(i * elements_per_inner_array + j);
>+      }
>+    }
>+    for (int i = 0; i < n; i++) {
>+      for (int j = 0; j < elements_per_inner_array; j++) {
>+        ASSERT_EQ((array[i])[j].get(),  i * elements_per_inner_array + j);
>+        ASSERT_EQ((array.data()[i])[j].get(), i * elements_per_inner_array + j);
>+      }
>+    }
>+
>+    // Test data()
>+    for (int i = 0; i < n; i++) {
>+      for (int j = 0; j < elements_per_inner_array; j++) {
>+        (array.data()[i])[j].set((i + 1) * elements_per_inner_array + j);
>+      }
>+    }
>+    for (int i = 0; i < n; i++) {
>+      for (int j = 0; j < elements_per_inner_array; j++) {
>+        ASSERT_EQ((array[i])[j].get(),
>+                  (i + 1) * elements_per_inner_array + j);
>+        ASSERT_EQ((array.data()[i])[j].get(),
>+                  (i + 1) * elements_per_inner_array + j);
>+      }
>+    }
>+  }  // Close scope containing 'array'.
>+
>+  // Check that all constructed elements were destructed.
>+  EXPECT_EQ(ConstructionTester::constructions,
>+            ConstructionTester::destructions);
>+}
>+
>+TEST(IteratorConstructorTest, NonInline) {
>+  int const kInput[] = { 2, 3, 5, 7, 11, 13, 17 };
>+  absl::FixedArray<int, ABSL_ARRAYSIZE(kInput) - 1> const fixed(
>+      kInput, kInput + ABSL_ARRAYSIZE(kInput));
>+  ASSERT_EQ(ABSL_ARRAYSIZE(kInput), fixed.size());
>+  for (size_t i = 0; i < ABSL_ARRAYSIZE(kInput); ++i) {
>+    ASSERT_EQ(kInput[i], fixed[i]);
>+  }
>+}
>+
>+TEST(IteratorConstructorTest, Inline) {
>+  int const kInput[] = { 2, 3, 5, 7, 11, 13, 17 };
>+  absl::FixedArray<int, ABSL_ARRAYSIZE(kInput)> const fixed(
>+      kInput, kInput + ABSL_ARRAYSIZE(kInput));
>+  ASSERT_EQ(ABSL_ARRAYSIZE(kInput), fixed.size());
>+  for (size_t i = 0; i < ABSL_ARRAYSIZE(kInput); ++i) {
>+    ASSERT_EQ(kInput[i], fixed[i]);
>+  }
>+}
>+
>+TEST(IteratorConstructorTest, NonPod) {
>+  char const* kInput[] =
>+      { "red", "orange", "yellow", "green", "blue", "indigo", "violet" };
>+  absl::FixedArray<std::string> const fixed(kInput, kInput + ABSL_ARRAYSIZE(kInput));
>+  ASSERT_EQ(ABSL_ARRAYSIZE(kInput), fixed.size());
>+  for (size_t i = 0; i < ABSL_ARRAYSIZE(kInput); ++i) {
>+    ASSERT_EQ(kInput[i], fixed[i]);
>+  }
>+}
>+
>+TEST(IteratorConstructorTest, FromEmptyVector) {
>+  std::vector<int> const empty;
>+  absl::FixedArray<int> const fixed(empty.begin(), empty.end());
>+  EXPECT_EQ(0, fixed.size());
>+  EXPECT_EQ(empty.size(), fixed.size());
>+}
>+
>+TEST(IteratorConstructorTest, FromNonEmptyVector) {
>+  int const kInput[] = { 2, 3, 5, 7, 11, 13, 17 };
>+  std::vector<int> const items(kInput, kInput + ABSL_ARRAYSIZE(kInput));
>+  absl::FixedArray<int> const fixed(items.begin(), items.end());
>+  ASSERT_EQ(items.size(), fixed.size());
>+  for (size_t i = 0; i < items.size(); ++i) {
>+    ASSERT_EQ(items[i], fixed[i]);
>+  }
>+}
>+
>+TEST(IteratorConstructorTest, FromBidirectionalIteratorRange) {
>+  int const kInput[] = { 2, 3, 5, 7, 11, 13, 17 };
>+  std::list<int> const items(kInput, kInput + ABSL_ARRAYSIZE(kInput));
>+  absl::FixedArray<int> const fixed(items.begin(), items.end());
>+  EXPECT_THAT(fixed, testing::ElementsAreArray(kInput));
>+}
>+
>+TEST(InitListConstructorTest, InitListConstruction) {
>+  absl::FixedArray<int> fixed = {1, 2, 3};
>+  EXPECT_THAT(fixed, testing::ElementsAreArray({1, 2, 3}));
>+}
>+
>+TEST(FillConstructorTest, NonEmptyArrays) {
>+  absl::FixedArray<int> stack_array(4, 1);
>+  EXPECT_THAT(stack_array, testing::ElementsAreArray({1, 1, 1, 1}));
>+
>+  absl::FixedArray<int, 0> heap_array(4, 1);
>+  EXPECT_THAT(stack_array, testing::ElementsAreArray({1, 1, 1, 1}));
>+}
>+
>+TEST(FillConstructorTest, EmptyArray) {
>+  absl::FixedArray<int> empty_fill(0, 1);
>+  absl::FixedArray<int> empty_size(0);
>+  EXPECT_EQ(empty_fill, empty_size);
>+}
>+
>+TEST(FillConstructorTest, NotTriviallyCopyable) {
>+  std::string str = "abcd";
>+  absl::FixedArray<std::string> strings = {str, str, str, str};
>+
>+  absl::FixedArray<std::string> array(4, str);
>+  EXPECT_EQ(array, strings);
>+}
>+
>+TEST(FillConstructorTest, Disambiguation) {
>+  absl::FixedArray<size_t> a(1, 2);
>+  EXPECT_THAT(a, testing::ElementsAre(2));
>+}
>+
>+TEST(FixedArrayTest, ManySizedArrays) {
>+  std::vector<int> sizes;
>+  for (int i = 1; i < 100; i++) sizes.push_back(i);
>+  for (int i = 100; i <= 1000; i += 100) sizes.push_back(i);
>+  for (int n : sizes) {
>+    TestArray<0>(n);
>+    TestArray<1>(n);
>+    TestArray<64>(n);
>+    TestArray<1000>(n);
>+  }
>+}
>+
>+TEST(FixedArrayTest, ManySizedArraysOfArraysOf1) {
>+  for (int n = 1; n < 1000; n++) {
>+    ASSERT_NO_FATAL_FAILURE((TestArrayOfArrays<1, 0>(n)));
>+    ASSERT_NO_FATAL_FAILURE((TestArrayOfArrays<1, 1>(n)));
>+    ASSERT_NO_FATAL_FAILURE((TestArrayOfArrays<1, 64>(n)));
>+    ASSERT_NO_FATAL_FAILURE((TestArrayOfArrays<1, 1000>(n)));
>+  }
>+}
>+
>+TEST(FixedArrayTest, ManySizedArraysOfArraysOf2) {
>+  for (int n = 1; n < 1000; n++) {
>+    TestArrayOfArrays<2, 0>(n);
>+    TestArrayOfArrays<2, 1>(n);
>+    TestArrayOfArrays<2, 64>(n);
>+    TestArrayOfArrays<2, 1000>(n);
>+  }
>+}
>+
>+// If value_type is put inside of a struct container,
>+// we might evoke this error in a hardened build unless data() is carefully
>+// written, so check on that.
>+//     error: call to int __builtin___sprintf_chk(etc...)
>+//     will always overflow destination buffer [-Werror]
>+TEST(FixedArrayTest, AvoidParanoidDiagnostics) {
>+  absl::FixedArray<char, 32> buf(32);
>+  sprintf(buf.data(), "foo");  // NOLINT(runtime/printf)
>+}
>+
>+TEST(FixedArrayTest, TooBigInlinedSpace) {
>+  struct TooBig {
>+    char c[1 << 20];
>+  };  // too big for even one on the stack
>+
>+  // Simulate the data members of absl::FixedArray, a pointer and a size_t.
>+  struct Data {
>+    TooBig* p;
>+    size_t size;
>+  };
>+
>+  // Make sure TooBig objects are not inlined for 0 or default size.
>+  static_assert(sizeof(absl::FixedArray<TooBig, 0>) == sizeof(Data),
>+                "0-sized absl::FixedArray should have same size as Data.");
>+  static_assert(alignof(absl::FixedArray<TooBig, 0>) == alignof(Data),
>+                "0-sized absl::FixedArray should have same alignment as Data.");
>+  static_assert(sizeof(absl::FixedArray<TooBig>) == sizeof(Data),
>+                "default-sized absl::FixedArray should have same size as Data");
>+  static_assert(
>+      alignof(absl::FixedArray<TooBig>) == alignof(Data),
>+      "default-sized absl::FixedArray should have same alignment as Data.");
>+}
>+
>+// PickyDelete EXPECTs its class-scope deallocation funcs are unused.
>+struct PickyDelete {
>+  PickyDelete() {}
>+  ~PickyDelete() {}
>+  void operator delete(void* p) {
>+    EXPECT_TRUE(false) << __FUNCTION__;
>+    ::operator delete(p);
>+  }
>+  void operator delete[](void* p) {
>+    EXPECT_TRUE(false) << __FUNCTION__;
>+    ::operator delete[](p);
>+  }
>+};
>+
>+TEST(FixedArrayTest, UsesGlobalAlloc) { absl::FixedArray<PickyDelete, 0> a(5); }
>+
>+
>+TEST(FixedArrayTest, Data) {
>+  static const int kInput[] = { 2, 3, 5, 7, 11, 13, 17 };
>+  absl::FixedArray<int> fa(std::begin(kInput), std::end(kInput));
>+  EXPECT_EQ(fa.data(), &*fa.begin());
>+  EXPECT_EQ(fa.data(), &fa[0]);
>+
>+  const absl::FixedArray<int>& cfa = fa;
>+  EXPECT_EQ(cfa.data(), &*cfa.begin());
>+  EXPECT_EQ(cfa.data(), &cfa[0]);
>+}
>+
>+TEST(FixedArrayTest, Empty) {
>+  absl::FixedArray<int> empty(0);
>+  absl::FixedArray<int> inline_filled(1);
>+  absl::FixedArray<int, 0> heap_filled(1);
>+  EXPECT_TRUE(empty.empty());
>+  EXPECT_FALSE(inline_filled.empty());
>+  EXPECT_FALSE(heap_filled.empty());
>+}
>+
>+TEST(FixedArrayTest, FrontAndBack) {
>+  absl::FixedArray<int, 3 * sizeof(int)> inlined = {1, 2, 3};
>+  EXPECT_EQ(inlined.front(), 1);
>+  EXPECT_EQ(inlined.back(), 3);
>+
>+  absl::FixedArray<int, 0> allocated = {1, 2, 3};
>+  EXPECT_EQ(allocated.front(), 1);
>+  EXPECT_EQ(allocated.back(), 3);
>+
>+  absl::FixedArray<int> one_element = {1};
>+  EXPECT_EQ(one_element.front(), one_element.back());
>+}
>+
>+TEST(FixedArrayTest, ReverseIteratorInlined) {
>+  absl::FixedArray<int, 5 * sizeof(int)> a = {0, 1, 2, 3, 4};
>+
>+  int counter = 5;
>+  for (absl::FixedArray<int>::reverse_iterator iter = a.rbegin();
>+       iter != a.rend(); ++iter) {
>+    counter--;
>+    EXPECT_EQ(counter, *iter);
>+  }
>+  EXPECT_EQ(counter, 0);
>+
>+  counter = 5;
>+  for (absl::FixedArray<int>::const_reverse_iterator iter = a.rbegin();
>+       iter != a.rend(); ++iter) {
>+    counter--;
>+    EXPECT_EQ(counter, *iter);
>+  }
>+  EXPECT_EQ(counter, 0);
>+
>+  counter = 5;
>+  for (auto iter = a.crbegin(); iter != a.crend(); ++iter) {
>+    counter--;
>+    EXPECT_EQ(counter, *iter);
>+  }
>+  EXPECT_EQ(counter, 0);
>+}
>+
>+TEST(FixedArrayTest, ReverseIteratorAllocated) {
>+  absl::FixedArray<int, 0> a = {0, 1, 2, 3, 4};
>+
>+  int counter = 5;
>+  for (absl::FixedArray<int>::reverse_iterator iter = a.rbegin();
>+       iter != a.rend(); ++iter) {
>+    counter--;
>+    EXPECT_EQ(counter, *iter);
>+  }
>+  EXPECT_EQ(counter, 0);
>+
>+  counter = 5;
>+  for (absl::FixedArray<int>::const_reverse_iterator iter = a.rbegin();
>+       iter != a.rend(); ++iter) {
>+    counter--;
>+    EXPECT_EQ(counter, *iter);
>+  }
>+  EXPECT_EQ(counter, 0);
>+
>+  counter = 5;
>+  for (auto iter = a.crbegin(); iter != a.crend(); ++iter) {
>+    counter--;
>+    EXPECT_EQ(counter, *iter);
>+  }
>+  EXPECT_EQ(counter, 0);
>+}
>+
>+TEST(FixedArrayTest, Fill) {
>+  absl::FixedArray<int, 5 * sizeof(int)> inlined(5);
>+  int fill_val = 42;
>+  inlined.fill(fill_val);
>+  for (int i : inlined) EXPECT_EQ(i, fill_val);
>+
>+  absl::FixedArray<int, 0> allocated(5);
>+  allocated.fill(fill_val);
>+  for (int i : allocated) EXPECT_EQ(i, fill_val);
>+
>+  // It doesn't do anything, just make sure this compiles.
>+  absl::FixedArray<int> empty(0);
>+  empty.fill(fill_val);
>+}
>+
>+// TODO(johnsoncj): Investigate InlinedStorage default initialization in GCC 4.x
>+#ifndef __GNUC__
>+TEST(FixedArrayTest, DefaultCtorDoesNotValueInit) {
>+  using T = char;
>+  constexpr auto capacity = 10;
>+  using FixedArrType = absl::FixedArray<T, capacity>;
>+  using FixedArrBuffType =
>+      absl::aligned_storage_t<sizeof(FixedArrType), alignof(FixedArrType)>;
>+  constexpr auto scrubbed_bits = 0x95;
>+  constexpr auto length = capacity / 2;
>+
>+  FixedArrBuffType buff;
>+  std::memset(std::addressof(buff), scrubbed_bits, sizeof(FixedArrBuffType));
>+
>+  FixedArrType* arr =
>+      ::new (static_cast<void*>(std::addressof(buff))) FixedArrType(length);
>+  EXPECT_THAT(*arr, testing::Each(scrubbed_bits));
>+  arr->~FixedArrType();
>+}
>+#endif  // __GNUC__
>+
>+// This is a stateful allocator, but the state lives outside of the
>+// allocator (in whatever test is using the allocator). This is odd
>+// but helps in tests where the allocator is propagated into nested
>+// containers - that chain of allocators uses the same state and is
>+// thus easier to query for aggregate allocation information.
>+template <typename T>
>+class CountingAllocator : public std::allocator<T> {
>+ public:
>+  using Alloc = std::allocator<T>;
>+  using pointer = typename Alloc::pointer;
>+  using size_type = typename Alloc::size_type;
>+
>+  CountingAllocator() : bytes_used_(nullptr), instance_count_(nullptr) {}
>+  explicit CountingAllocator(int64_t* b)
>+      : bytes_used_(b), instance_count_(nullptr) {}
>+  CountingAllocator(int64_t* b, int64_t* a)
>+      : bytes_used_(b), instance_count_(a) {}
>+
>+  template <typename U>
>+  explicit CountingAllocator(const CountingAllocator<U>& x)
>+      : Alloc(x),
>+        bytes_used_(x.bytes_used_),
>+        instance_count_(x.instance_count_) {}
>+
>+  pointer allocate(size_type n, const void* const hint = nullptr) {
>+    assert(bytes_used_ != nullptr);
>+    *bytes_used_ += n * sizeof(T);
>+    return Alloc::allocate(n, hint);
>+  }
>+
>+  void deallocate(pointer p, size_type n) {
>+    Alloc::deallocate(p, n);
>+    assert(bytes_used_ != nullptr);
>+    *bytes_used_ -= n * sizeof(T);
>+  }
>+
>+  template <typename... Args>
>+  void construct(pointer p, Args&&... args) {
>+    Alloc::construct(p, absl::forward<Args>(args)...);
>+    if (instance_count_) {
>+      *instance_count_ += 1;
>+    }
>+  }
>+
>+  void destroy(pointer p) {
>+    Alloc::destroy(p);
>+    if (instance_count_) {
>+      *instance_count_ -= 1;
>+    }
>+  }
>+
>+  template <typename U>
>+  class rebind {
>+   public:
>+    using other = CountingAllocator<U>;
>+  };
>+
>+  int64_t* bytes_used_;
>+  int64_t* instance_count_;
>+};
>+
>+TEST(AllocatorSupportTest, CountInlineAllocations) {
>+  constexpr size_t inlined_size = 4;
>+  using Alloc = CountingAllocator<int>;
>+  using AllocFxdArr = absl::FixedArray<int, inlined_size, Alloc>;
>+
>+  int64_t allocated = 0;
>+  int64_t active_instances = 0;
>+
>+  {
>+    const int ia[] = {0, 1, 2, 3, 4, 5, 6, 7};
>+
>+    Alloc alloc(&allocated, &active_instances);
>+
>+    AllocFxdArr arr(ia, ia + inlined_size, alloc);
>+    static_cast<void>(arr);
>+  }
>+
>+  EXPECT_EQ(allocated, 0);
>+  EXPECT_EQ(active_instances, 0);
>+}
>+
>+TEST(AllocatorSupportTest, CountOutoflineAllocations) {
>+  constexpr size_t inlined_size = 4;
>+  using Alloc = CountingAllocator<int>;
>+  using AllocFxdArr = absl::FixedArray<int, inlined_size, Alloc>;
>+
>+  int64_t allocated = 0;
>+  int64_t active_instances = 0;
>+
>+  {
>+    const int ia[] = {0, 1, 2, 3, 4, 5, 6, 7};
>+    Alloc alloc(&allocated, &active_instances);
>+
>+    AllocFxdArr arr(ia, ia + ABSL_ARRAYSIZE(ia), alloc);
>+
>+    EXPECT_EQ(allocated, arr.size() * sizeof(int));
>+    static_cast<void>(arr);
>+  }
>+
>+  EXPECT_EQ(active_instances, 0);
>+}
>+
>+TEST(AllocatorSupportTest, CountCopyInlineAllocations) {
>+  constexpr size_t inlined_size = 4;
>+  using Alloc = CountingAllocator<int>;
>+  using AllocFxdArr = absl::FixedArray<int, inlined_size, Alloc>;
>+
>+  int64_t allocated1 = 0;
>+  int64_t allocated2 = 0;
>+  int64_t active_instances = 0;
>+  Alloc alloc(&allocated1, &active_instances);
>+  Alloc alloc2(&allocated2, &active_instances);
>+
>+  {
>+    int initial_value = 1;
>+
>+    AllocFxdArr arr1(inlined_size / 2, initial_value, alloc);
>+
>+    EXPECT_EQ(allocated1, 0);
>+
>+    AllocFxdArr arr2(arr1, alloc2);
>+
>+    EXPECT_EQ(allocated2, 0);
>+    static_cast<void>(arr1);
>+    static_cast<void>(arr2);
>+  }
>+
>+  EXPECT_EQ(active_instances, 0);
>+}
>+
>+TEST(AllocatorSupportTest, CountCopyOutoflineAllocations) {
>+  constexpr size_t inlined_size = 4;
>+  using Alloc = CountingAllocator<int>;
>+  using AllocFxdArr = absl::FixedArray<int, inlined_size, Alloc>;
>+
>+  int64_t allocated1 = 0;
>+  int64_t allocated2 = 0;
>+  int64_t active_instances = 0;
>+  Alloc alloc(&allocated1, &active_instances);
>+  Alloc alloc2(&allocated2, &active_instances);
>+
>+  {
>+    int initial_value = 1;
>+
>+    AllocFxdArr arr1(inlined_size * 2, initial_value, alloc);
>+
>+    EXPECT_EQ(allocated1, arr1.size() * sizeof(int));
>+
>+    AllocFxdArr arr2(arr1, alloc2);
>+
>+    EXPECT_EQ(allocated2, inlined_size * 2 * sizeof(int));
>+    static_cast<void>(arr1);
>+    static_cast<void>(arr2);
>+  }
>+
>+  EXPECT_EQ(active_instances, 0);
>+}
>+
>+TEST(AllocatorSupportTest, SizeValAllocConstructor) {
>+  using testing::AllOf;
>+  using testing::Each;
>+  using testing::SizeIs;
>+
>+  constexpr size_t inlined_size = 4;
>+  using Alloc = CountingAllocator<int>;
>+  using AllocFxdArr = absl::FixedArray<int, inlined_size, Alloc>;
>+
>+  {
>+    auto len = inlined_size / 2;
>+    auto val = 0;
>+    int64_t allocated = 0;
>+    AllocFxdArr arr(len, val, Alloc(&allocated));
>+
>+    EXPECT_EQ(allocated, 0);
>+    EXPECT_THAT(arr, AllOf(SizeIs(len), Each(0)));
>+  }
>+
>+  {
>+    auto len = inlined_size * 2;
>+    auto val = 0;
>+    int64_t allocated = 0;
>+    AllocFxdArr arr(len, val, Alloc(&allocated));
>+
>+    EXPECT_EQ(allocated, len * sizeof(int));
>+    EXPECT_THAT(arr, AllOf(SizeIs(len), Each(0)));
>+  }
>+}
>+
>+#ifdef ADDRESS_SANITIZER
>+TEST(FixedArrayTest, AddressSanitizerAnnotations1) {
>+  absl::FixedArray<int, 32> a(10);
>+  int *raw = a.data();
>+  raw[0] = 0;
>+  raw[9] = 0;
>+  EXPECT_DEATH(raw[-2] = 0, "container-overflow");
>+  EXPECT_DEATH(raw[-1] = 0, "container-overflow");
>+  EXPECT_DEATH(raw[10] = 0, "container-overflow");
>+  EXPECT_DEATH(raw[31] = 0, "container-overflow");
>+}
>+
>+TEST(FixedArrayTest, AddressSanitizerAnnotations2) {
>+  absl::FixedArray<char, 17> a(12);
>+  char *raw = a.data();
>+  raw[0] = 0;
>+  raw[11] = 0;
>+  EXPECT_DEATH(raw[-7] = 0, "container-overflow");
>+  EXPECT_DEATH(raw[-1] = 0, "container-overflow");
>+  EXPECT_DEATH(raw[12] = 0, "container-overflow");
>+  EXPECT_DEATH(raw[17] = 0, "container-overflow");
>+}
>+
>+TEST(FixedArrayTest, AddressSanitizerAnnotations3) {
>+  absl::FixedArray<uint64_t, 20> a(20);
>+  uint64_t *raw = a.data();
>+  raw[0] = 0;
>+  raw[19] = 0;
>+  EXPECT_DEATH(raw[-1] = 0, "container-overflow");
>+  EXPECT_DEATH(raw[20] = 0, "container-overflow");
>+}
>+
>+TEST(FixedArrayTest, AddressSanitizerAnnotations4) {
>+  absl::FixedArray<ThreeInts> a(10);
>+  ThreeInts *raw = a.data();
>+  raw[0] = ThreeInts();
>+  raw[9] = ThreeInts();
>+  // Note: raw[-1] is pointing to 12 bytes before the container range. However,
>+  // there is only a 8-byte red zone before the container range, so we only
>+  // access the last 4 bytes of the struct to make sure it stays within the red
>+  // zone.
>+  EXPECT_DEATH(raw[-1].z_ = 0, "container-overflow");
>+  EXPECT_DEATH(raw[10] = ThreeInts(), "container-overflow");
>+  // The actual size of storage is kDefaultBytes=256, 21*12 = 252,
>+  // so reading raw[21] should still trigger the correct warning.
>+  EXPECT_DEATH(raw[21] = ThreeInts(), "container-overflow");
>+}
>+#endif  // ADDRESS_SANITIZER
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/inlined_vector.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/inlined_vector.h
>new file mode 100644
>index 00000000000..ca36fd3699b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/inlined_vector.h
>@@ -0,0 +1,1386 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// -----------------------------------------------------------------------------
>+// File: inlined_vector.h
>+// -----------------------------------------------------------------------------
>+//
>+// This header file contains the declaration and definition of an "inlined
>+// vector" which behaves in an equivalent fashion to a `std::vector`, except
>+// that storage for small sequences of the vector are provided inline without
>+// requiring any heap allocation.
>+
>+// An `absl::InlinedVector<T,N>` specifies the size N at which to inline as one
>+// of its template parameters. Vectors of length <= N are provided inline.
>+// Typically N is very small (e.g., 4) so that sequences that are expected to be
>+// short do not require allocations.
>+
>+// An `absl::InlinedVector` does not usually require a specific allocator; if
>+// the inlined vector grows beyond its initial constraints, it will need to
>+// allocate (as any normal `std::vector` would) and it will generally use the
>+// default allocator in that case; optionally, a custom allocator may be
>+// specified using an `absl::InlinedVector<T,N,A>` construction.
>+
>+#ifndef ABSL_CONTAINER_INLINED_VECTOR_H_
>+#define ABSL_CONTAINER_INLINED_VECTOR_H_
>+
>+#include <algorithm>
>+#include <cassert>
>+#include <cstddef>
>+#include <cstdlib>
>+#include <cstring>
>+#include <initializer_list>
>+#include <iterator>
>+#include <memory>
>+#include <type_traits>
>+#include <utility>
>+
>+#include "absl/algorithm/algorithm.h"
>+#include "absl/base/internal/throw_delegate.h"
>+#include "absl/base/optimization.h"
>+#include "absl/base/port.h"
>+#include "absl/memory/memory.h"
>+
>+namespace absl {
>+
>+// -----------------------------------------------------------------------------
>+// InlinedVector
>+// -----------------------------------------------------------------------------
>+//
>+// An `absl::InlinedVector` is designed to be a drop-in replacement for
>+// `std::vector` for use cases where the vector's size is sufficiently small
>+// that it can be inlined. If the inlined vector does grow beyond its estimated
>+// size, it will trigger an initial allocation on the heap, and will behave as a
>+// `std:vector`. The API of the `absl::InlinedVector` within this file is
>+// designed to cover the same API footprint as covered by `std::vector`.
>+template <typename T, size_t N, typename A = std::allocator<T> >
>+class InlinedVector {
>+  using AllocatorTraits = std::allocator_traits<A>;
>+
>+ public:
>+  using allocator_type = A;
>+  using value_type = typename allocator_type::value_type;
>+  using pointer = typename allocator_type::pointer;
>+  using const_pointer = typename allocator_type::const_pointer;
>+  using reference = typename allocator_type::reference;
>+  using const_reference = typename allocator_type::const_reference;
>+  using size_type = typename allocator_type::size_type;
>+  using difference_type = typename allocator_type::difference_type;
>+  using iterator = pointer;
>+  using const_iterator = const_pointer;
>+  using reverse_iterator = std::reverse_iterator<iterator>;
>+  using const_reverse_iterator = std::reverse_iterator<const_iterator>;
>+
>+  InlinedVector() noexcept(noexcept(allocator_type()))
>+      : allocator_and_tag_(allocator_type()) {}
>+
>+  explicit InlinedVector(const allocator_type& alloc) noexcept
>+      : allocator_and_tag_(alloc) {}
>+
>+  // Create a vector with n copies of value_type().
>+  explicit InlinedVector(size_type n,
>+                         const allocator_type& alloc = allocator_type())
>+      : allocator_and_tag_(alloc) {
>+    InitAssign(n);
>+  }
>+
>+  // Create a vector with n copies of elem
>+  InlinedVector(size_type n, const value_type& elem,
>+                const allocator_type& alloc = allocator_type())
>+      : allocator_and_tag_(alloc) {
>+    InitAssign(n, elem);
>+  }
>+
>+  // Create and initialize with the elements [first .. last).
>+  // The unused enable_if argument restricts this constructor so that it is
>+  // elided when value_type is an integral type.  This prevents ambiguous
>+  // interpretation between a call to this constructor with two integral
>+  // arguments and a call to the preceding (n, elem) constructor.
>+  template <typename InputIterator>
>+  InlinedVector(
>+      InputIterator first, InputIterator last,
>+      const allocator_type& alloc = allocator_type(),
>+      typename std::enable_if<!std::is_integral<InputIterator>::value>::type* =
>+          nullptr)
>+      : allocator_and_tag_(alloc) {
>+    AppendRange(first, last);
>+  }
>+
>+  InlinedVector(std::initializer_list<value_type> init,
>+                const allocator_type& alloc = allocator_type())
>+      : allocator_and_tag_(alloc) {
>+    AppendRange(init.begin(), init.end());
>+  }
>+
>+  InlinedVector(const InlinedVector& v);
>+  InlinedVector(const InlinedVector& v, const allocator_type& alloc);
>+
>+  // This move constructor does not allocate and only moves the underlying
>+  // objects, so its `noexcept` specification depends on whether moving the
>+  // underlying objects can throw or not. We assume
>+  //  a) move constructors should only throw due to allocation failure and
>+  //  b) if `value_type`'s move constructor allocates, it uses the same
>+  //     allocation function as the `InlinedVector`'s allocator, so the move
>+  //     constructor is non-throwing if the allocator is non-throwing or
>+  //     `value_type`'s move constructor is specified as `noexcept`.
>+  InlinedVector(InlinedVector&& v) noexcept(
>+      absl::allocator_is_nothrow<allocator_type>::value ||
>+      std::is_nothrow_move_constructible<value_type>::value);
>+
>+  // This move constructor allocates and also moves the underlying objects, so
>+  // its `noexcept` specification depends on whether the allocation can throw
>+  // and whether moving the underlying objects can throw. Based on the same
>+  // assumptions above, the `noexcept` specification is dominated by whether the
>+  // allocation can throw regardless of whether `value_type`'s move constructor
>+  // is specified as `noexcept`.
>+  InlinedVector(InlinedVector&& v, const allocator_type& alloc) noexcept(
>+      absl::allocator_is_nothrow<allocator_type>::value);
>+
>+  ~InlinedVector() { clear(); }
>+
>+  InlinedVector& operator=(const InlinedVector& v) {
>+    if (this == &v) {
>+      return *this;
>+    }
>+    // Optimized to avoid reallocation.
>+    // Prefer reassignment to copy construction for elements.
>+    if (size() < v.size()) {  // grow
>+      reserve(v.size());
>+      std::copy(v.begin(), v.begin() + size(), begin());
>+      std::copy(v.begin() + size(), v.end(), std::back_inserter(*this));
>+    } else {  // maybe shrink
>+      erase(begin() + v.size(), end());
>+      std::copy(v.begin(), v.end(), begin());
>+    }
>+    return *this;
>+  }
>+
>+  InlinedVector& operator=(InlinedVector&& v) {
>+    if (this == &v) {
>+      return *this;
>+    }
>+    if (v.allocated()) {
>+      clear();
>+      tag().set_allocated_size(v.size());
>+      init_allocation(v.allocation());
>+      v.tag() = Tag();
>+    } else {
>+      if (allocated()) clear();
>+      // Both are inlined now.
>+      if (size() < v.size()) {
>+        auto mid = std::make_move_iterator(v.begin() + size());
>+        std::copy(std::make_move_iterator(v.begin()), mid, begin());
>+        UninitializedCopy(mid, std::make_move_iterator(v.end()), end());
>+      } else {
>+        auto new_end = std::copy(std::make_move_iterator(v.begin()),
>+                                 std::make_move_iterator(v.end()), begin());
>+        Destroy(new_end, end());
>+      }
>+      tag().set_inline_size(v.size());
>+    }
>+    return *this;
>+  }
>+
>+  InlinedVector& operator=(std::initializer_list<value_type> init) {
>+    AssignRange(init.begin(), init.end());
>+    return *this;
>+  }
>+
>+  // InlinedVector::assign()
>+  //
>+  // Replaces the contents of the inlined vector with copies of those in the
>+  // iterator range [first, last).
>+  template <typename InputIterator>
>+  void assign(
>+      InputIterator first, InputIterator last,
>+      typename std::enable_if<!std::is_integral<InputIterator>::value>::type* =
>+          nullptr) {
>+    AssignRange(first, last);
>+  }
>+
>+  // Overload of `InlinedVector::assign()` to take values from elements of an
>+  // initializer list
>+  void assign(std::initializer_list<value_type> init) {
>+    AssignRange(init.begin(), init.end());
>+  }
>+
>+  // Overload of `InlinedVector::assign()` to replace the first `n` elements of
>+  // the inlined vector with `elem` values.
>+  void assign(size_type n, const value_type& elem) {
>+    if (n <= size()) {  // Possibly shrink
>+      std::fill_n(begin(), n, elem);
>+      erase(begin() + n, end());
>+      return;
>+    }
>+    // Grow
>+    reserve(n);
>+    std::fill_n(begin(), size(), elem);
>+    if (allocated()) {
>+      UninitializedFill(allocated_space() + size(), allocated_space() + n,
>+                        elem);
>+      tag().set_allocated_size(n);
>+    } else {
>+      UninitializedFill(inlined_space() + size(), inlined_space() + n, elem);
>+      tag().set_inline_size(n);
>+    }
>+  }
>+
>+  // InlinedVector::size()
>+  //
>+  // Returns the number of elements in the inlined vector.
>+  size_type size() const noexcept { return tag().size(); }
>+
>+  // InlinedVector::empty()
>+  //
>+  // Checks if the inlined vector has no elements.
>+  bool empty() const noexcept { return (size() == 0); }
>+
>+  // InlinedVector::capacity()
>+  //
>+  // Returns the number of elements that can be stored in an inlined vector
>+  // without requiring a reallocation of underlying memory. Note that for
>+  // most inlined vectors, `capacity()` should equal its initial size `N`; for
>+  // inlined vectors which exceed this capacity, they will no longer be inlined,
>+  // and `capacity()` will equal its capacity on the allocated heap.
>+  size_type capacity() const noexcept {
>+    return allocated() ? allocation().capacity() : N;
>+  }
>+
>+  // InlinedVector::max_size()
>+  //
>+  // Returns the maximum number of elements the vector can hold.
>+  size_type max_size() const noexcept {
>+    // One bit of the size storage is used to indicate whether the inlined
>+    // vector is allocated; as a result, the maximum size of the container that
>+    // we can express is half of the max for our size type.
>+    return std::numeric_limits<size_type>::max() / 2;
>+  }
>+
>+  // InlinedVector::data()
>+  //
>+  // Returns a const T* pointer to elements of the inlined vector. This pointer
>+  // can be used to access (but not modify) the contained elements.
>+  // Only results within the range `[0,size())` are defined.
>+  const_pointer data() const noexcept {
>+    return allocated() ? allocated_space() : inlined_space();
>+  }
>+
>+  // Overload of InlinedVector::data() to return a T* pointer to elements of the
>+  // inlined vector. This pointer can be used to access and modify the contained
>+  // elements.
>+  pointer data() noexcept {
>+    return allocated() ? allocated_space() : inlined_space();
>+  }
>+
>+  // InlinedVector::clear()
>+  //
>+  // Removes all elements from the inlined vector.
>+  void clear() noexcept {
>+    size_type s = size();
>+    if (allocated()) {
>+      Destroy(allocated_space(), allocated_space() + s);
>+      allocation().Dealloc(allocator());
>+    } else if (s != 0) {  // do nothing for empty vectors
>+      Destroy(inlined_space(), inlined_space() + s);
>+    }
>+    tag() = Tag();
>+  }
>+
>+  // InlinedVector::at()
>+  //
>+  // Returns the ith element of an inlined vector.
>+  const value_type& at(size_type i) const {
>+    if (ABSL_PREDICT_FALSE(i >= size())) {
>+      base_internal::ThrowStdOutOfRange(
>+          "InlinedVector::at failed bounds check");
>+    }
>+    return data()[i];
>+  }
>+
>+  // InlinedVector::operator[]
>+  //
>+  // Returns the ith element of an inlined vector using the array operator.
>+  const value_type& operator[](size_type i) const {
>+    assert(i < size());
>+    return data()[i];
>+  }
>+
>+  // Overload of InlinedVector::at() to return the ith element of an inlined
>+  // vector.
>+  value_type& at(size_type i) {
>+    if (i >= size()) {
>+      base_internal::ThrowStdOutOfRange(
>+          "InlinedVector::at failed bounds check");
>+    }
>+    return data()[i];
>+  }
>+
>+  // Overload of InlinedVector::operator[] to return the ith element of an
>+  // inlined vector.
>+  value_type& operator[](size_type i) {
>+    assert(i < size());
>+    return data()[i];
>+  }
>+
>+  // InlinedVector::back()
>+  //
>+  // Returns a reference to the last element of an inlined vector.
>+  value_type& back() {
>+    assert(!empty());
>+    return at(size() - 1);
>+  }
>+
>+  // Overload of InlinedVector::back() returns a reference to the last element
>+  // of an inlined vector of const values.
>+  const value_type& back() const {
>+    assert(!empty());
>+    return at(size() - 1);
>+  }
>+
>+  // InlinedVector::front()
>+  //
>+  // Returns a reference to the first element of an inlined vector.
>+  value_type& front() {
>+    assert(!empty());
>+    return at(0);
>+  }
>+
>+  // Overload of InlinedVector::front() returns a reference to the first element
>+  // of an inlined vector of const values.
>+  const value_type& front() const {
>+    assert(!empty());
>+    return at(0);
>+  }
>+
>+  // InlinedVector::emplace_back()
>+  //
>+  // Constructs and appends an object to the inlined vector.
>+  //
>+  // Returns a reference to the inserted element.
>+  template <typename... Args>
>+  value_type& emplace_back(Args&&... args) {
>+    size_type s = size();
>+    assert(s <= capacity());
>+    if (ABSL_PREDICT_FALSE(s == capacity())) {
>+      return GrowAndEmplaceBack(std::forward<Args>(args)...);
>+    }
>+    assert(s < capacity());
>+
>+    value_type* space;
>+    if (allocated()) {
>+      tag().set_allocated_size(s + 1);
>+      space = allocated_space();
>+    } else {
>+      tag().set_inline_size(s + 1);
>+      space = inlined_space();
>+    }
>+    return Construct(space + s, std::forward<Args>(args)...);
>+  }
>+
>+  // InlinedVector::push_back()
>+  //
>+  // Appends a const element to the inlined vector.
>+  void push_back(const value_type& t) { emplace_back(t); }
>+
>+  // Overload of InlinedVector::push_back() to append a move-only element to the
>+  // inlined vector.
>+  void push_back(value_type&& t) { emplace_back(std::move(t)); }
>+
>+  // InlinedVector::pop_back()
>+  //
>+  // Removes the last element (which is destroyed) in the inlined vector.
>+  void pop_back() {
>+    assert(!empty());
>+    size_type s = size();
>+    if (allocated()) {
>+      Destroy(allocated_space() + s - 1, allocated_space() + s);
>+      tag().set_allocated_size(s - 1);
>+    } else {
>+      Destroy(inlined_space() + s - 1, inlined_space() + s);
>+      tag().set_inline_size(s - 1);
>+    }
>+  }
>+
>+  // InlinedVector::resize()
>+  //
>+  // Resizes the inlined vector to contain `n` elements. If `n` is smaller than
>+  // the inlined vector's current size, extra elements are destroyed. If `n` is
>+  // larger than the initial size, new elements are value-initialized.
>+  void resize(size_type n);
>+
>+  // Overload of InlinedVector::resize() to resize the inlined vector to contain
>+  // `n` elements. If `n` is larger than the current size, enough copies of
>+  // `elem` are appended to increase its size to `n`.
>+  void resize(size_type n, const value_type& elem);
>+
>+  // InlinedVector::begin()
>+  //
>+  // Returns an iterator to the beginning of the inlined vector.
>+  iterator begin() noexcept { return data(); }
>+
>+  // Overload of InlinedVector::begin() for returning a const iterator to the
>+  // beginning of the inlined vector.
>+  const_iterator begin() const noexcept { return data(); }
>+
>+  // InlinedVector::cbegin()
>+  //
>+  // Returns a const iterator to the beginning of the inlined vector.
>+  const_iterator cbegin() const noexcept { return begin(); }
>+
>+  // InlinedVector::end()
>+  //
>+  // Returns an iterator to the end of the inlined vector.
>+  iterator end() noexcept { return data() + size(); }
>+
>+  // Overload of InlinedVector::end() for returning a const iterator to the end
>+  // of the inlined vector.
>+  const_iterator end() const noexcept { return data() + size(); }
>+
>+  // InlinedVector::cend()
>+  //
>+  // Returns a const iterator to the end of the inlined vector.
>+  const_iterator cend() const noexcept { return end(); }
>+
>+  // InlinedVector::rbegin()
>+  //
>+  // Returns a reverse iterator from the end of the inlined vector.
>+  reverse_iterator rbegin() noexcept { return reverse_iterator(end()); }
>+
>+  // Overload of InlinedVector::rbegin() for returning a const reverse iterator
>+  // from the end of the inlined vector.
>+  const_reverse_iterator rbegin() const noexcept {
>+    return const_reverse_iterator(end());
>+  }
>+
>+  // InlinedVector::crbegin()
>+  //
>+  // Returns a const reverse iterator from the end of the inlined vector.
>+  const_reverse_iterator crbegin() const noexcept { return rbegin(); }
>+
>+  // InlinedVector::rend()
>+  //
>+  // Returns a reverse iterator from the beginning of the inlined vector.
>+  reverse_iterator rend() noexcept { return reverse_iterator(begin()); }
>+
>+  // Overload of InlinedVector::rend() for returning a const reverse iterator
>+  // from the beginning of the inlined vector.
>+  const_reverse_iterator rend() const noexcept {
>+    return const_reverse_iterator(begin());
>+  }
>+
>+  // InlinedVector::crend()
>+  //
>+  // Returns a reverse iterator from the beginning of the inlined vector.
>+  const_reverse_iterator crend() const noexcept { return rend(); }
>+
>+  // InlinedVector::emplace()
>+  //
>+  // Constructs and inserts an object to the inlined vector at the given
>+  // `position`, returning an iterator pointing to the newly emplaced element.
>+  template <typename... Args>
>+  iterator emplace(const_iterator position, Args&&... args);
>+
>+  // InlinedVector::insert()
>+  //
>+  // Inserts an element of the specified value at `position`, returning an
>+  // iterator pointing to the newly inserted element.
>+  iterator insert(const_iterator position, const value_type& v) {
>+    return emplace(position, v);
>+  }
>+
>+  // Overload of InlinedVector::insert() for inserting an element of the
>+  // specified rvalue, returning an iterator pointing to the newly inserted
>+  // element.
>+  iterator insert(const_iterator position, value_type&& v) {
>+    return emplace(position, std::move(v));
>+  }
>+
>+  // Overload of InlinedVector::insert() for inserting `n` elements of the
>+  // specified value at `position`, returning an iterator pointing to the first
>+  // of the newly inserted elements.
>+  iterator insert(const_iterator position, size_type n, const value_type& v) {
>+    return InsertWithCount(position, n, v);
>+  }
>+
>+  // Overload of `InlinedVector::insert()` to disambiguate the two
>+  // three-argument overloads of `insert()`, returning an iterator pointing to
>+  // the first of the newly inserted elements.
>+  template <typename InputIterator,
>+            typename = typename std::enable_if<std::is_convertible<
>+                typename std::iterator_traits<InputIterator>::iterator_category,
>+                std::input_iterator_tag>::value>::type>
>+  iterator insert(const_iterator position, InputIterator first,
>+                  InputIterator last) {
>+    using IterType =
>+        typename std::iterator_traits<InputIterator>::iterator_category;
>+    return InsertWithRange(position, first, last, IterType());
>+  }
>+
>+  // Overload of InlinedVector::insert() for inserting a list of elements at
>+  // `position`, returning an iterator pointing to the first of the newly
>+  // inserted elements.
>+  iterator insert(const_iterator position,
>+                  std::initializer_list<value_type> init) {
>+    return insert(position, init.begin(), init.end());
>+  }
>+
>+  // InlinedVector::erase()
>+  //
>+  // Erases the element at `position` of the inlined vector, returning an
>+  // iterator pointing to the following element or the container's end if the
>+  // last element was erased.
>+  iterator erase(const_iterator position) {
>+    assert(position >= begin());
>+    assert(position < end());
>+
>+    iterator pos = const_cast<iterator>(position);
>+    std::move(pos + 1, end(), pos);
>+    pop_back();
>+    return pos;
>+  }
>+
>+  // Overload of InlinedVector::erase() for erasing all elements in the
>+  // iterator range [first, last) in the inlined vector, returning an iterator
>+  // pointing to the first element following the range erased, or the
>+  // container's end if range included the container's last element.
>+  iterator erase(const_iterator first, const_iterator last);
>+
>+  // InlinedVector::reserve()
>+  //
>+  // Enlarges the underlying representation of the inlined vector so it can hold
>+  // at least `n` elements. This method does not change `size()` or the actual
>+  // contents of the vector.
>+  //
>+  // Note that if `n` does not exceed the inlined vector's initial size `N`,
>+  // `reserve()` will have no effect; if it does exceed its initial size,
>+  // `reserve()` will trigger an initial allocation and move the inlined vector
>+  // onto the heap. If the vector already exists on the heap and the requested
>+  // size exceeds it, a reallocation will be performed.
>+  void reserve(size_type n) {
>+    if (n > capacity()) {
>+      // Make room for new elements
>+      EnlargeBy(n - size());
>+    }
>+  }
>+
>+  // InlinedVector::shrink_to_fit()
>+  //
>+  // Reduces memory usage by freeing unused memory.
>+  // After this call `capacity()` will be equal to `max(N, size())`.
>+  //
>+  // If `size() <= N` and the elements are currently stored on the heap, they
>+  // will be moved to the inlined storage and the heap memory deallocated.
>+  // If `size() > N` and `size() < capacity()` the elements will be moved to
>+  // a reallocated storage on heap.
>+  void shrink_to_fit() {
>+    const auto s = size();
>+    if (!allocated() || s == capacity()) {
>+      // There's nothing to deallocate.
>+      return;
>+    }
>+
>+    if (s <= N) {
>+      // Move the elements to the inlined storage.
>+      // We have to do this using a temporary, because inlined_storage and
>+      // allocation_storage are in a union field.
>+      auto temp = std::move(*this);
>+      assign(std::make_move_iterator(temp.begin()),
>+             std::make_move_iterator(temp.end()));
>+      return;
>+    }
>+
>+    // Reallocate storage and move elements.
>+    // We can't simply use the same approach as above, because assign() would
>+    // call into reserve() internally and reserve larger capacity than we need.
>+    Allocation new_allocation(allocator(), s);
>+    UninitializedCopy(std::make_move_iterator(allocated_space()),
>+                      std::make_move_iterator(allocated_space() + s),
>+                      new_allocation.buffer());
>+    ResetAllocation(new_allocation, s);
>+  }
>+
>+  // InlinedVector::swap()
>+  //
>+  // Swaps the contents of this inlined vector with the contents of `other`.
>+  void swap(InlinedVector& other);
>+
>+  // InlinedVector::get_allocator()
>+  //
>+  // Returns the allocator of this inlined vector.
>+  allocator_type get_allocator() const { return allocator(); }
>+
>+ private:
>+  static_assert(N > 0, "inlined vector with nonpositive size");
>+
>+  // It holds whether the vector is allocated or not in the lowest bit.
>+  // The size is held in the high bits:
>+  //   size_ = (size << 1) | is_allocated;
>+  //
>+  // Maintainer's Note: size_type is user defined. The contract is limited to
>+  // arithmetic operators to avoid depending on compliant overloaded bitwise
>+  // operators.
>+  class Tag {
>+   public:
>+    Tag() : size_(0) {}
>+    size_type size() const { return size_ / 2; }
>+    void add_size(size_type n) { size_ += n * 2; }
>+    void set_inline_size(size_type n) { size_ = n * 2; }
>+    void set_allocated_size(size_type n) { size_ = (n * 2) + 1; }
>+    bool allocated() const { return size_ % 2; }
>+
>+   private:
>+    size_type size_;
>+  };
>+
>+  // Derives from allocator_type to use the empty base class optimization.
>+  // If the allocator_type is stateless, we can 'store'
>+  // our instance of it for free.
>+  class AllocatorAndTag : private allocator_type {
>+   public:
>+    explicit AllocatorAndTag(const allocator_type& a, Tag t = Tag())
>+        : allocator_type(a), tag_(t) {}
>+    Tag& tag() { return tag_; }
>+    const Tag& tag() const { return tag_; }
>+    allocator_type& allocator() { return *this; }
>+    const allocator_type& allocator() const { return *this; }
>+
>+   private:
>+    Tag tag_;
>+  };
>+
>+  class Allocation {
>+   public:
>+    Allocation(allocator_type& a,  // NOLINT(runtime/references)
>+               size_type capacity)
>+        : capacity_(capacity),
>+          buffer_(AllocatorTraits::allocate(a, capacity_)) {}
>+
>+    void Dealloc(allocator_type& a) {  // NOLINT(runtime/references)
>+      AllocatorTraits::deallocate(a, buffer(), capacity());
>+    }
>+
>+    size_type capacity() const { return capacity_; }
>+    const value_type* buffer() const { return buffer_; }
>+    value_type* buffer() { return buffer_; }
>+
>+   private:
>+    size_type capacity_;
>+    value_type* buffer_;
>+  };
>+
>+  const Tag& tag() const { return allocator_and_tag_.tag(); }
>+  Tag& tag() { return allocator_and_tag_.tag(); }
>+
>+  Allocation& allocation() {
>+    return reinterpret_cast<Allocation&>(rep_.allocation_storage.allocation);
>+  }
>+  const Allocation& allocation() const {
>+    return reinterpret_cast<const Allocation&>(
>+        rep_.allocation_storage.allocation);
>+  }
>+  void init_allocation(const Allocation& allocation) {
>+    new (&rep_.allocation_storage.allocation) Allocation(allocation);
>+  }
>+
>+  // TODO(absl-team): investigate whether the reinterpret_cast is appropriate.
>+  value_type* inlined_space() {
>+    return reinterpret_cast<value_type*>(
>+        std::addressof(rep_.inlined_storage.inlined[0]));
>+  }
>+  const value_type* inlined_space() const {
>+    return reinterpret_cast<const value_type*>(
>+        std::addressof(rep_.inlined_storage.inlined[0]));
>+  }
>+
>+  value_type* allocated_space() { return allocation().buffer(); }
>+  const value_type* allocated_space() const { return allocation().buffer(); }
>+
>+  const allocator_type& allocator() const {
>+    return allocator_and_tag_.allocator();
>+  }
>+  allocator_type& allocator() { return allocator_and_tag_.allocator(); }
>+
>+  bool allocated() const { return tag().allocated(); }
>+
>+  // Enlarge the underlying representation so we can store size_ + delta elems.
>+  // The size is not changed, and any newly added memory is not initialized.
>+  void EnlargeBy(size_type delta);
>+
>+  // Shift all elements from position to end() n places to the right.
>+  // If the vector needs to be enlarged, memory will be allocated.
>+  // Returns iterators pointing to the start of the previously-initialized
>+  // portion and the start of the uninitialized portion of the created gap.
>+  // The number of initialized spots is pair.second - pair.first;
>+  // the number of raw spots is n - (pair.second - pair.first).
>+  //
>+  // Updates the size of the InlinedVector internally.
>+  std::pair<iterator, iterator> ShiftRight(const_iterator position,
>+                                           size_type n);
>+
>+  void ResetAllocation(Allocation new_allocation, size_type new_size) {
>+    if (allocated()) {
>+      Destroy(allocated_space(), allocated_space() + size());
>+      assert(begin() == allocated_space());
>+      allocation().Dealloc(allocator());
>+      allocation() = new_allocation;
>+    } else {
>+      Destroy(inlined_space(), inlined_space() + size());
>+      init_allocation(new_allocation);  // bug: only init once
>+    }
>+    tag().set_allocated_size(new_size);
>+  }
>+
>+  template <typename... Args>
>+  value_type& GrowAndEmplaceBack(Args&&... args) {
>+    assert(size() == capacity());
>+    const size_type s = size();
>+
>+    Allocation new_allocation(allocator(), 2 * capacity());
>+
>+    value_type& new_element =
>+        Construct(new_allocation.buffer() + s, std::forward<Args>(args)...);
>+    UninitializedCopy(std::make_move_iterator(data()),
>+                      std::make_move_iterator(data() + s),
>+                      new_allocation.buffer());
>+
>+    ResetAllocation(new_allocation, s + 1);
>+
>+    return new_element;
>+  }
>+
>+  void InitAssign(size_type n);
>+  void InitAssign(size_type n, const value_type& t);
>+
>+  template <typename... Args>
>+  value_type& Construct(pointer p, Args&&... args) {
>+    AllocatorTraits::construct(allocator(), p, std::forward<Args>(args)...);
>+    return *p;
>+  }
>+
>+  template <typename Iter>
>+  void UninitializedCopy(Iter src, Iter src_last, value_type* dst) {
>+    for (; src != src_last; ++dst, ++src) Construct(dst, *src);
>+  }
>+
>+  template <typename... Args>
>+  void UninitializedFill(value_type* dst, value_type* dst_last,
>+                         const Args&... args) {
>+    for (; dst != dst_last; ++dst) Construct(dst, args...);
>+  }
>+
>+  // Destroy [ptr, ptr_last) in place.
>+  void Destroy(value_type* ptr, value_type* ptr_last);
>+
>+  template <typename Iter>
>+  void AppendRange(Iter first, Iter last, std::input_iterator_tag) {
>+    std::copy(first, last, std::back_inserter(*this));
>+  }
>+
>+  // Faster path for forward iterators.
>+  template <typename Iter>
>+  void AppendRange(Iter first, Iter last, std::forward_iterator_tag);
>+
>+  template <typename Iter>
>+  void AppendRange(Iter first, Iter last) {
>+    using IterTag = typename std::iterator_traits<Iter>::iterator_category;
>+    AppendRange(first, last, IterTag());
>+  }
>+
>+  template <typename Iter>
>+  void AssignRange(Iter first, Iter last, std::input_iterator_tag);
>+
>+  // Faster path for forward iterators.
>+  template <typename Iter>
>+  void AssignRange(Iter first, Iter last, std::forward_iterator_tag);
>+
>+  template <typename Iter>
>+  void AssignRange(Iter first, Iter last) {
>+    using IterTag = typename std::iterator_traits<Iter>::iterator_category;
>+    AssignRange(first, last, IterTag());
>+  }
>+
>+  iterator InsertWithCount(const_iterator position, size_type n,
>+                           const value_type& v);
>+
>+  template <typename InputIter>
>+  iterator InsertWithRange(const_iterator position, InputIter first,
>+                           InputIter last, std::input_iterator_tag);
>+  template <typename ForwardIter>
>+  iterator InsertWithRange(const_iterator position, ForwardIter first,
>+                           ForwardIter last, std::forward_iterator_tag);
>+
>+  AllocatorAndTag allocator_and_tag_;
>+
>+  // Either the inlined or allocated representation
>+  union Rep {
>+    // Use struct to perform indirection that solves a bizarre compilation
>+    // error on Visual Studio (all known versions).
>+    struct {
>+      typename std::aligned_storage<sizeof(value_type),
>+                                    alignof(value_type)>::type inlined[N];
>+    } inlined_storage;
>+    struct {
>+      typename std::aligned_storage<sizeof(Allocation),
>+                                    alignof(Allocation)>::type allocation;
>+    } allocation_storage;
>+  } rep_;
>+};
>+
>+// -----------------------------------------------------------------------------
>+// InlinedVector Non-Member Functions
>+// -----------------------------------------------------------------------------
>+
>+// swap()
>+//
>+// Swaps the contents of two inlined vectors. This convenience function
>+// simply calls InlinedVector::swap(other_inlined_vector).
>+template <typename T, size_t N, typename A>
>+void swap(InlinedVector<T, N, A>& a,
>+          InlinedVector<T, N, A>& b) noexcept(noexcept(a.swap(b))) {
>+  a.swap(b);
>+}
>+
>+// operator==()
>+//
>+// Tests the equivalency of the contents of two inlined vectors.
>+template <typename T, size_t N, typename A>
>+bool operator==(const InlinedVector<T, N, A>& a,
>+                const InlinedVector<T, N, A>& b) {
>+  return absl::equal(a.begin(), a.end(), b.begin(), b.end());
>+}
>+
>+// operator!=()
>+//
>+// Tests the inequality of the contents of two inlined vectors.
>+template <typename T, size_t N, typename A>
>+bool operator!=(const InlinedVector<T, N, A>& a,
>+                const InlinedVector<T, N, A>& b) {
>+  return !(a == b);
>+}
>+
>+// operator<()
>+//
>+// Tests whether the contents of one inlined vector are less than the contents
>+// of another through a lexicographical comparison operation.
>+template <typename T, size_t N, typename A>
>+bool operator<(const InlinedVector<T, N, A>& a,
>+               const InlinedVector<T, N, A>& b) {
>+  return std::lexicographical_compare(a.begin(), a.end(), b.begin(), b.end());
>+}
>+
>+// operator>()
>+//
>+// Tests whether the contents of one inlined vector are greater than the
>+// contents of another through a lexicographical comparison operation.
>+template <typename T, size_t N, typename A>
>+bool operator>(const InlinedVector<T, N, A>& a,
>+               const InlinedVector<T, N, A>& b) {
>+  return b < a;
>+}
>+
>+// operator<=()
>+//
>+// Tests whether the contents of one inlined vector are less than or equal to
>+// the contents of another through a lexicographical comparison operation.
>+template <typename T, size_t N, typename A>
>+bool operator<=(const InlinedVector<T, N, A>& a,
>+                const InlinedVector<T, N, A>& b) {
>+  return !(b < a);
>+}
>+
>+// operator>=()
>+//
>+// Tests whether the contents of one inlined vector are greater than or equal to
>+// the contents of another through a lexicographical comparison operation.
>+template <typename T, size_t N, typename A>
>+bool operator>=(const InlinedVector<T, N, A>& a,
>+                const InlinedVector<T, N, A>& b) {
>+  return !(a < b);
>+}
>+
>+// -----------------------------------------------------------------------------
>+// Implementation of InlinedVector
>+// -----------------------------------------------------------------------------
>+//
>+// Do not depend on any implementation details below this line.
>+
>+template <typename T, size_t N, typename A>
>+InlinedVector<T, N, A>::InlinedVector(const InlinedVector& v)
>+    : allocator_and_tag_(v.allocator()) {
>+  reserve(v.size());
>+  if (allocated()) {
>+    UninitializedCopy(v.begin(), v.end(), allocated_space());
>+    tag().set_allocated_size(v.size());
>+  } else {
>+    UninitializedCopy(v.begin(), v.end(), inlined_space());
>+    tag().set_inline_size(v.size());
>+  }
>+}
>+
>+template <typename T, size_t N, typename A>
>+InlinedVector<T, N, A>::InlinedVector(const InlinedVector& v,
>+                                      const allocator_type& alloc)
>+    : allocator_and_tag_(alloc) {
>+  reserve(v.size());
>+  if (allocated()) {
>+    UninitializedCopy(v.begin(), v.end(), allocated_space());
>+    tag().set_allocated_size(v.size());
>+  } else {
>+    UninitializedCopy(v.begin(), v.end(), inlined_space());
>+    tag().set_inline_size(v.size());
>+  }
>+}
>+
>+template <typename T, size_t N, typename A>
>+InlinedVector<T, N, A>::InlinedVector(InlinedVector&& v) noexcept(
>+    absl::allocator_is_nothrow<allocator_type>::value ||
>+    std::is_nothrow_move_constructible<value_type>::value)
>+    : allocator_and_tag_(v.allocator_and_tag_) {
>+  if (v.allocated()) {
>+    // We can just steal the underlying buffer from the source.
>+    // That leaves the source empty, so we clear its size.
>+    init_allocation(v.allocation());
>+    v.tag() = Tag();
>+  } else {
>+    UninitializedCopy(std::make_move_iterator(v.inlined_space()),
>+                      std::make_move_iterator(v.inlined_space() + v.size()),
>+                      inlined_space());
>+  }
>+}
>+
>+template <typename T, size_t N, typename A>
>+InlinedVector<T, N, A>::InlinedVector(
>+    InlinedVector&& v,
>+    const allocator_type&
>+        alloc) noexcept(absl::allocator_is_nothrow<allocator_type>::value)
>+    : allocator_and_tag_(alloc) {
>+  if (v.allocated()) {
>+    if (alloc == v.allocator()) {
>+      // We can just steal the allocation from the source.
>+      tag() = v.tag();
>+      init_allocation(v.allocation());
>+      v.tag() = Tag();
>+    } else {
>+      // We need to use our own allocator
>+      reserve(v.size());
>+      UninitializedCopy(std::make_move_iterator(v.begin()),
>+                        std::make_move_iterator(v.end()), allocated_space());
>+      tag().set_allocated_size(v.size());
>+    }
>+  } else {
>+    UninitializedCopy(std::make_move_iterator(v.inlined_space()),
>+                      std::make_move_iterator(v.inlined_space() + v.size()),
>+                      inlined_space());
>+    tag().set_inline_size(v.size());
>+  }
>+}
>+
>+template <typename T, size_t N, typename A>
>+void InlinedVector<T, N, A>::InitAssign(size_type n, const value_type& t) {
>+  if (n > static_cast<size_type>(N)) {
>+    Allocation new_allocation(allocator(), n);
>+    init_allocation(new_allocation);
>+    UninitializedFill(allocated_space(), allocated_space() + n, t);
>+    tag().set_allocated_size(n);
>+  } else {
>+    UninitializedFill(inlined_space(), inlined_space() + n, t);
>+    tag().set_inline_size(n);
>+  }
>+}
>+
>+template <typename T, size_t N, typename A>
>+void InlinedVector<T, N, A>::InitAssign(size_type n) {
>+  if (n > static_cast<size_type>(N)) {
>+    Allocation new_allocation(allocator(), n);
>+    init_allocation(new_allocation);
>+    UninitializedFill(allocated_space(), allocated_space() + n);
>+    tag().set_allocated_size(n);
>+  } else {
>+    UninitializedFill(inlined_space(), inlined_space() + n);
>+    tag().set_inline_size(n);
>+  }
>+}
>+
>+template <typename T, size_t N, typename A>
>+void InlinedVector<T, N, A>::resize(size_type n) {
>+  size_type s = size();
>+  if (n < s) {
>+    erase(begin() + n, end());
>+    return;
>+  }
>+  reserve(n);
>+  assert(capacity() >= n);
>+
>+  // Fill new space with elements constructed in-place.
>+  if (allocated()) {
>+    UninitializedFill(allocated_space() + s, allocated_space() + n);
>+    tag().set_allocated_size(n);
>+  } else {
>+    UninitializedFill(inlined_space() + s, inlined_space() + n);
>+    tag().set_inline_size(n);
>+  }
>+}
>+
>+template <typename T, size_t N, typename A>
>+void InlinedVector<T, N, A>::resize(size_type n, const value_type& elem) {
>+  size_type s = size();
>+  if (n < s) {
>+    erase(begin() + n, end());
>+    return;
>+  }
>+  reserve(n);
>+  assert(capacity() >= n);
>+
>+  // Fill new space with copies of 'elem'.
>+  if (allocated()) {
>+    UninitializedFill(allocated_space() + s, allocated_space() + n, elem);
>+    tag().set_allocated_size(n);
>+  } else {
>+    UninitializedFill(inlined_space() + s, inlined_space() + n, elem);
>+    tag().set_inline_size(n);
>+  }
>+}
>+
>+template <typename T, size_t N, typename A>
>+template <typename... Args>
>+typename InlinedVector<T, N, A>::iterator InlinedVector<T, N, A>::emplace(
>+    const_iterator position, Args&&... args) {
>+  assert(position >= begin());
>+  assert(position <= end());
>+  if (position == end()) {
>+    emplace_back(std::forward<Args>(args)...);
>+    return end() - 1;
>+  }
>+
>+  T new_t = T(std::forward<Args>(args)...);
>+
>+  auto range = ShiftRight(position, 1);
>+  if (range.first == range.second) {
>+    // constructing into uninitialized memory
>+    Construct(range.first, std::move(new_t));
>+  } else {
>+    // assigning into moved-from object
>+    *range.first = T(std::move(new_t));
>+  }
>+
>+  return range.first;
>+}
>+
>+template <typename T, size_t N, typename A>
>+typename InlinedVector<T, N, A>::iterator InlinedVector<T, N, A>::erase(
>+    const_iterator first, const_iterator last) {
>+  assert(begin() <= first);
>+  assert(first <= last);
>+  assert(last <= end());
>+
>+  iterator range_start = const_cast<iterator>(first);
>+  iterator range_end = const_cast<iterator>(last);
>+
>+  size_type s = size();
>+  ptrdiff_t erase_gap = std::distance(range_start, range_end);
>+  if (erase_gap > 0) {
>+    pointer space;
>+    if (allocated()) {
>+      space = allocated_space();
>+      tag().set_allocated_size(s - erase_gap);
>+    } else {
>+      space = inlined_space();
>+      tag().set_inline_size(s - erase_gap);
>+    }
>+    std::move(range_end, space + s, range_start);
>+    Destroy(space + s - erase_gap, space + s);
>+  }
>+  return range_start;
>+}
>+
>+template <typename T, size_t N, typename A>
>+void InlinedVector<T, N, A>::swap(InlinedVector& other) {
>+  using std::swap;  // Augment ADL with std::swap.
>+  if (&other == this) {
>+    return;
>+  }
>+  if (allocated() && other.allocated()) {
>+    // Both out of line, so just swap the tag, allocation, and allocator.
>+    swap(tag(), other.tag());
>+    swap(allocation(), other.allocation());
>+    swap(allocator(), other.allocator());
>+    return;
>+  }
>+  if (!allocated() && !other.allocated()) {
>+    // Both inlined: swap up to smaller size, then move remaining elements.
>+    InlinedVector* a = this;
>+    InlinedVector* b = &other;
>+    if (size() < other.size()) {
>+      swap(a, b);
>+    }
>+
>+    const size_type a_size = a->size();
>+    const size_type b_size = b->size();
>+    assert(a_size >= b_size);
>+    // 'a' is larger. Swap the elements up to the smaller array size.
>+    std::swap_ranges(a->inlined_space(), a->inlined_space() + b_size,
>+                     b->inlined_space());
>+
>+    // Move the remaining elements: A[b_size,a_size) -> B[b_size,a_size)
>+    b->UninitializedCopy(a->inlined_space() + b_size,
>+                         a->inlined_space() + a_size,
>+                         b->inlined_space() + b_size);
>+    a->Destroy(a->inlined_space() + b_size, a->inlined_space() + a_size);
>+
>+    swap(a->tag(), b->tag());
>+    swap(a->allocator(), b->allocator());
>+    assert(b->size() == a_size);
>+    assert(a->size() == b_size);
>+    return;
>+  }
>+  // One is out of line, one is inline.
>+  // We first move the elements from the inlined vector into the
>+  // inlined space in the other vector.  We then put the other vector's
>+  // pointer/capacity into the originally inlined vector and swap
>+  // the tags.
>+  InlinedVector* a = this;
>+  InlinedVector* b = &other;
>+  if (a->allocated()) {
>+    swap(a, b);
>+  }
>+  assert(!a->allocated());
>+  assert(b->allocated());
>+  const size_type a_size = a->size();
>+  const size_type b_size = b->size();
>+  // In an optimized build, b_size would be unused.
>+  (void)b_size;
>+
>+  // Made Local copies of size(), don't need tag() accurate anymore
>+  swap(a->tag(), b->tag());
>+
>+  // Copy b_allocation out before b's union gets clobbered by inline_space.
>+  Allocation b_allocation = b->allocation();
>+
>+  b->UninitializedCopy(a->inlined_space(), a->inlined_space() + a_size,
>+                       b->inlined_space());
>+  a->Destroy(a->inlined_space(), a->inlined_space() + a_size);
>+
>+  a->allocation() = b_allocation;
>+
>+  if (a->allocator() != b->allocator()) {
>+    swap(a->allocator(), b->allocator());
>+  }
>+
>+  assert(b->size() == a_size);
>+  assert(a->size() == b_size);
>+}
>+
>+template <typename T, size_t N, typename A>
>+void InlinedVector<T, N, A>::EnlargeBy(size_type delta) {
>+  const size_type s = size();
>+  assert(s <= capacity());
>+
>+  size_type target = std::max(static_cast<size_type>(N), s + delta);
>+
>+  // Compute new capacity by repeatedly doubling current capacity
>+  // TODO(psrc): Check and avoid overflow?
>+  size_type new_capacity = capacity();
>+  while (new_capacity < target) {
>+    new_capacity <<= 1;
>+  }
>+
>+  Allocation new_allocation(allocator(), new_capacity);
>+
>+  UninitializedCopy(std::make_move_iterator(data()),
>+                    std::make_move_iterator(data() + s),
>+                    new_allocation.buffer());
>+
>+  ResetAllocation(new_allocation, s);
>+}
>+
>+template <typename T, size_t N, typename A>
>+auto InlinedVector<T, N, A>::ShiftRight(const_iterator position, size_type n)
>+    -> std::pair<iterator, iterator> {
>+  iterator start_used = const_cast<iterator>(position);
>+  iterator start_raw = const_cast<iterator>(position);
>+  size_type s = size();
>+  size_type required_size = s + n;
>+
>+  if (required_size > capacity()) {
>+    // Compute new capacity by repeatedly doubling current capacity
>+    size_type new_capacity = capacity();
>+    while (new_capacity < required_size) {
>+      new_capacity <<= 1;
>+    }
>+    // Move everyone into the new allocation, leaving a gap of n for the
>+    // requested shift.
>+    Allocation new_allocation(allocator(), new_capacity);
>+    size_type index = position - begin();
>+    UninitializedCopy(std::make_move_iterator(data()),
>+                      std::make_move_iterator(data() + index),
>+                      new_allocation.buffer());
>+    UninitializedCopy(std::make_move_iterator(data() + index),
>+                      std::make_move_iterator(data() + s),
>+                      new_allocation.buffer() + index + n);
>+    ResetAllocation(new_allocation, s);
>+
>+    // New allocation means our iterator is invalid, so we'll recalculate.
>+    // Since the entire gap is in new space, there's no used space to reuse.
>+    start_raw = begin() + index;
>+    start_used = start_raw;
>+  } else {
>+    // If we had enough space, it's a two-part move. Elements going into
>+    // previously-unoccupied space need an UninitializedCopy. Elements
>+    // going into a previously-occupied space are just a move.
>+    iterator pos = const_cast<iterator>(position);
>+    iterator raw_space = end();
>+    size_type slots_in_used_space = raw_space - pos;
>+    size_type new_elements_in_used_space = std::min(n, slots_in_used_space);
>+    size_type new_elements_in_raw_space = n - new_elements_in_used_space;
>+    size_type old_elements_in_used_space =
>+        slots_in_used_space - new_elements_in_used_space;
>+
>+    UninitializedCopy(std::make_move_iterator(pos + old_elements_in_used_space),
>+                      std::make_move_iterator(raw_space),
>+                      raw_space + new_elements_in_raw_space);
>+    std::move_backward(pos, pos + old_elements_in_used_space, raw_space);
>+
>+    // If the gap is entirely in raw space, the used space starts where the raw
>+    // space starts, leaving no elements in used space. If the gap is entirely
>+    // in used space, the raw space starts at the end of the gap, leaving all
>+    // elements accounted for within the used space.
>+    start_used = pos;
>+    start_raw = pos + new_elements_in_used_space;
>+  }
>+  tag().add_size(n);
>+  return std::make_pair(start_used, start_raw);
>+}
>+
>+template <typename T, size_t N, typename A>
>+void InlinedVector<T, N, A>::Destroy(value_type* ptr, value_type* ptr_last) {
>+  for (value_type* p = ptr; p != ptr_last; ++p) {
>+    AllocatorTraits::destroy(allocator(), p);
>+  }
>+
>+  // Overwrite unused memory with 0xab so we can catch uninitialized usage.
>+  // Cast to void* to tell the compiler that we don't care that we might be
>+  // scribbling on a vtable pointer.
>+#ifndef NDEBUG
>+  if (ptr != ptr_last) {
>+    memset(reinterpret_cast<void*>(ptr), 0xab, sizeof(*ptr) * (ptr_last - ptr));
>+  }
>+#endif
>+}
>+
>+template <typename T, size_t N, typename A>
>+template <typename Iter>
>+void InlinedVector<T, N, A>::AppendRange(Iter first, Iter last,
>+                                         std::forward_iterator_tag) {
>+  using Length = typename std::iterator_traits<Iter>::difference_type;
>+  Length length = std::distance(first, last);
>+  reserve(size() + length);
>+  if (allocated()) {
>+    UninitializedCopy(first, last, allocated_space() + size());
>+    tag().set_allocated_size(size() + length);
>+  } else {
>+    UninitializedCopy(first, last, inlined_space() + size());
>+    tag().set_inline_size(size() + length);
>+  }
>+}
>+
>+template <typename T, size_t N, typename A>
>+template <typename Iter>
>+void InlinedVector<T, N, A>::AssignRange(Iter first, Iter last,
>+                                         std::input_iterator_tag) {
>+  // Optimized to avoid reallocation.
>+  // Prefer reassignment to copy construction for elements.
>+  iterator out = begin();
>+  for (; first != last && out != end(); ++first, ++out) {
>+    *out = *first;
>+  }
>+  erase(out, end());
>+  std::copy(first, last, std::back_inserter(*this));
>+}
>+
>+template <typename T, size_t N, typename A>
>+template <typename Iter>
>+void InlinedVector<T, N, A>::AssignRange(Iter first, Iter last,
>+                                         std::forward_iterator_tag) {
>+  using Length = typename std::iterator_traits<Iter>::difference_type;
>+  Length length = std::distance(first, last);
>+  // Prefer reassignment to copy construction for elements.
>+  if (static_cast<size_type>(length) <= size()) {
>+    erase(std::copy(first, last, begin()), end());
>+    return;
>+  }
>+  reserve(length);
>+  iterator out = begin();
>+  for (; out != end(); ++first, ++out) *out = *first;
>+  if (allocated()) {
>+    UninitializedCopy(first, last, out);
>+    tag().set_allocated_size(length);
>+  } else {
>+    UninitializedCopy(first, last, out);
>+    tag().set_inline_size(length);
>+  }
>+}
>+
>+template <typename T, size_t N, typename A>
>+auto InlinedVector<T, N, A>::InsertWithCount(const_iterator position,
>+                                             size_type n, const value_type& v)
>+    -> iterator {
>+  assert(position >= begin() && position <= end());
>+  if (n == 0) return const_cast<iterator>(position);
>+
>+  value_type copy = v;
>+  std::pair<iterator, iterator> it_pair = ShiftRight(position, n);
>+  std::fill(it_pair.first, it_pair.second, copy);
>+  UninitializedFill(it_pair.second, it_pair.first + n, copy);
>+
>+  return it_pair.first;
>+}
>+
>+template <typename T, size_t N, typename A>
>+template <typename InputIter>
>+auto InlinedVector<T, N, A>::InsertWithRange(const_iterator position,
>+                                             InputIter first, InputIter last,
>+                                             std::input_iterator_tag)
>+    -> iterator {
>+  assert(position >= begin() && position <= end());
>+  size_type index = position - cbegin();
>+  size_type i = index;
>+  while (first != last) insert(begin() + i++, *first++);
>+  return begin() + index;
>+}
>+
>+// Overload of InlinedVector::InsertWithRange()
>+template <typename T, size_t N, typename A>
>+template <typename ForwardIter>
>+auto InlinedVector<T, N, A>::InsertWithRange(const_iterator position,
>+                                             ForwardIter first,
>+                                             ForwardIter last,
>+                                             std::forward_iterator_tag)
>+    -> iterator {
>+  assert(position >= begin() && position <= end());
>+  if (first == last) {
>+    return const_cast<iterator>(position);
>+  }
>+  using Length = typename std::iterator_traits<ForwardIter>::difference_type;
>+  Length n = std::distance(first, last);
>+  std::pair<iterator, iterator> it_pair = ShiftRight(position, n);
>+  size_type used_spots = it_pair.second - it_pair.first;
>+  ForwardIter open_spot = std::next(first, used_spots);
>+  std::copy(first, open_spot, it_pair.first);
>+  UninitializedCopy(open_spot, last, it_pair.second);
>+  return it_pair.first;
>+}
>+
>+}  // namespace absl
>+
>+#endif  // ABSL_CONTAINER_INLINED_VECTOR_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/inlined_vector_benchmark.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/inlined_vector_benchmark.cc
>new file mode 100644
>index 00000000000..24f2174928a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/inlined_vector_benchmark.cc
>@@ -0,0 +1,385 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/container/inlined_vector.h"
>+
>+#include <string>
>+#include <vector>
>+
>+#include "benchmark/benchmark.h"
>+#include "absl/base/internal/raw_logging.h"
>+#include "absl/strings/str_cat.h"
>+
>+namespace {
>+
>+using IntVec = absl::InlinedVector<int, 8>;
>+
>+void BM_InlinedVectorFill(benchmark::State& state) {
>+  const int len = state.range(0);
>+  for (auto _ : state) {
>+    IntVec v;
>+    for (int i = 0; i < len; i++) {
>+      v.push_back(i);
>+    }
>+  }
>+  state.SetItemsProcessed(static_cast<int64_t>(state.iterations()) * len);
>+}
>+BENCHMARK(BM_InlinedVectorFill)->Range(0, 1024);
>+
>+void BM_InlinedVectorFillRange(benchmark::State& state) {
>+  const int len = state.range(0);
>+  std::unique_ptr<int[]> ia(new int[len]);
>+  for (int i = 0; i < len; i++) {
>+    ia[i] = i;
>+  }
>+  for (auto _ : state) {
>+    IntVec v(ia.get(), ia.get() + len);
>+    benchmark::DoNotOptimize(v);
>+  }
>+  state.SetItemsProcessed(static_cast<int64_t>(state.iterations()) * len);
>+}
>+BENCHMARK(BM_InlinedVectorFillRange)->Range(0, 1024);
>+
>+void BM_StdVectorFill(benchmark::State& state) {
>+  const int len = state.range(0);
>+  for (auto _ : state) {
>+    std::vector<int> v;
>+    for (int i = 0; i < len; i++) {
>+      v.push_back(i);
>+    }
>+  }
>+  state.SetItemsProcessed(static_cast<int64_t>(state.iterations()) * len);
>+}
>+BENCHMARK(BM_StdVectorFill)->Range(0, 1024);
>+
>+// The purpose of the next two benchmarks is to verify that
>+// absl::InlinedVector is efficient when moving is more efficent than
>+// copying. To do so, we use strings that are larger than the short
>+// std::string optimization.
>+bool StringRepresentedInline(std::string s) {
>+  const char* chars = s.data();
>+  std::string s1 = std::move(s);
>+  return s1.data() != chars;
>+}
>+
>+int GetNonShortStringOptimizationSize() {
>+  for (int i = 24; i <= 192; i *= 2) {
>+    if (!StringRepresentedInline(std::string(i, 'A'))) {
>+      return i;
>+    }
>+  }
>+  ABSL_RAW_LOG(
>+      FATAL,
>+      "Failed to find a std::string larger than the short std::string optimization");
>+  return -1;
>+}
>+
>+void BM_InlinedVectorFillString(benchmark::State& state) {
>+  const int len = state.range(0);
>+  const int no_sso = GetNonShortStringOptimizationSize();
>+  std::string strings[4] = {std::string(no_sso, 'A'), std::string(no_sso, 'B'),
>+                       std::string(no_sso, 'C'), std::string(no_sso, 'D')};
>+
>+  for (auto _ : state) {
>+    absl::InlinedVector<std::string, 8> v;
>+    for (int i = 0; i < len; i++) {
>+      v.push_back(strings[i & 3]);
>+    }
>+  }
>+  state.SetItemsProcessed(static_cast<int64_t>(state.iterations()) * len);
>+}
>+BENCHMARK(BM_InlinedVectorFillString)->Range(0, 1024);
>+
>+void BM_StdVectorFillString(benchmark::State& state) {
>+  const int len = state.range(0);
>+  const int no_sso = GetNonShortStringOptimizationSize();
>+  std::string strings[4] = {std::string(no_sso, 'A'), std::string(no_sso, 'B'),
>+                       std::string(no_sso, 'C'), std::string(no_sso, 'D')};
>+
>+  for (auto _ : state) {
>+    std::vector<std::string> v;
>+    for (int i = 0; i < len; i++) {
>+      v.push_back(strings[i & 3]);
>+    }
>+  }
>+  state.SetItemsProcessed(static_cast<int64_t>(state.iterations()) * len);
>+}
>+BENCHMARK(BM_StdVectorFillString)->Range(0, 1024);
>+
>+struct Buffer {  // some arbitrary structure for benchmarking.
>+  char* base;
>+  int length;
>+  int capacity;
>+  void* user_data;
>+};
>+
>+void BM_InlinedVectorTenAssignments(benchmark::State& state) {
>+  const int len = state.range(0);
>+  using BufferVec = absl::InlinedVector<Buffer, 2>;
>+
>+  BufferVec src;
>+  src.resize(len);
>+
>+  BufferVec dst;
>+  for (auto _ : state) {
>+    for (int i = 0; i < 10; ++i) {
>+      dst = src;
>+    }
>+  }
>+}
>+BENCHMARK(BM_InlinedVectorTenAssignments)
>+    ->Arg(0)->Arg(1)->Arg(2)->Arg(3)->Arg(4)->Arg(20);
>+
>+void BM_CreateFromContainer(benchmark::State& state) {
>+  for (auto _ : state) {
>+    absl::InlinedVector<int, 4> x(absl::InlinedVector<int, 4>{1, 2, 3});
>+    benchmark::DoNotOptimize(x);
>+  }
>+}
>+BENCHMARK(BM_CreateFromContainer);
>+
>+struct LargeCopyableOnly {
>+  LargeCopyableOnly() : d(1024, 17) {}
>+  LargeCopyableOnly(const LargeCopyableOnly& o) = default;
>+  LargeCopyableOnly& operator=(const LargeCopyableOnly& o) = default;
>+
>+  std::vector<int> d;
>+};
>+
>+struct LargeCopyableSwappable {
>+  LargeCopyableSwappable() : d(1024, 17) {}
>+  LargeCopyableSwappable(const LargeCopyableSwappable& o) = default;
>+  LargeCopyableSwappable(LargeCopyableSwappable&& o) = delete;
>+
>+  LargeCopyableSwappable& operator=(LargeCopyableSwappable o) {
>+    using std::swap;
>+    swap(*this, o);
>+    return *this;
>+  }
>+  LargeCopyableSwappable& operator=(LargeCopyableSwappable&& o) = delete;
>+
>+  friend void swap(LargeCopyableSwappable& a, LargeCopyableSwappable& b) {
>+    using std::swap;
>+    swap(a.d, b.d);
>+  }
>+
>+  std::vector<int> d;
>+};
>+
>+struct LargeCopyableMovable {
>+  LargeCopyableMovable() : d(1024, 17) {}
>+  // Use implicitly defined copy and move.
>+
>+  std::vector<int> d;
>+};
>+
>+struct LargeCopyableMovableSwappable {
>+  LargeCopyableMovableSwappable() : d(1024, 17) {}
>+  LargeCopyableMovableSwappable(const LargeCopyableMovableSwappable& o) =
>+      default;
>+  LargeCopyableMovableSwappable(LargeCopyableMovableSwappable&& o) = default;
>+
>+  LargeCopyableMovableSwappable& operator=(LargeCopyableMovableSwappable o) {
>+    using std::swap;
>+    swap(*this, o);
>+    return *this;
>+  }
>+  LargeCopyableMovableSwappable& operator=(LargeCopyableMovableSwappable&& o) =
>+      default;
>+
>+  friend void swap(LargeCopyableMovableSwappable& a,
>+                   LargeCopyableMovableSwappable& b) {
>+    using std::swap;
>+    swap(a.d, b.d);
>+  }
>+
>+  std::vector<int> d;
>+};
>+
>+template <typename ElementType>
>+void BM_SwapElements(benchmark::State& state) {
>+  const int len = state.range(0);
>+  using Vec = absl::InlinedVector<ElementType, 32>;
>+  Vec a(len);
>+  Vec b;
>+  for (auto _ : state) {
>+    using std::swap;
>+    swap(a, b);
>+  }
>+}
>+BENCHMARK_TEMPLATE(BM_SwapElements, LargeCopyableOnly)->Range(0, 1024);
>+BENCHMARK_TEMPLATE(BM_SwapElements, LargeCopyableSwappable)->Range(0, 1024);
>+BENCHMARK_TEMPLATE(BM_SwapElements, LargeCopyableMovable)->Range(0, 1024);
>+BENCHMARK_TEMPLATE(BM_SwapElements, LargeCopyableMovableSwappable)
>+    ->Range(0, 1024);
>+
>+// The following benchmark is meant to track the efficiency of the vector size
>+// as a function of stored type via the benchmark label. It is not meant to
>+// output useful sizeof operator performance. The loop is a dummy operation
>+// to fulfill the requirement of running the benchmark.
>+template <typename VecType>
>+void BM_Sizeof(benchmark::State& state) {
>+  int size = 0;
>+  for (auto _ : state) {
>+    VecType vec;
>+    size = sizeof(vec);
>+  }
>+  state.SetLabel(absl::StrCat("sz=", size));
>+}
>+BENCHMARK_TEMPLATE(BM_Sizeof, absl::InlinedVector<char, 1>);
>+BENCHMARK_TEMPLATE(BM_Sizeof, absl::InlinedVector<char, 4>);
>+BENCHMARK_TEMPLATE(BM_Sizeof, absl::InlinedVector<char, 7>);
>+BENCHMARK_TEMPLATE(BM_Sizeof, absl::InlinedVector<char, 8>);
>+
>+BENCHMARK_TEMPLATE(BM_Sizeof, absl::InlinedVector<int, 1>);
>+BENCHMARK_TEMPLATE(BM_Sizeof, absl::InlinedVector<int, 4>);
>+BENCHMARK_TEMPLATE(BM_Sizeof, absl::InlinedVector<int, 7>);
>+BENCHMARK_TEMPLATE(BM_Sizeof, absl::InlinedVector<int, 8>);
>+
>+BENCHMARK_TEMPLATE(BM_Sizeof, absl::InlinedVector<void*, 1>);
>+BENCHMARK_TEMPLATE(BM_Sizeof, absl::InlinedVector<void*, 4>);
>+BENCHMARK_TEMPLATE(BM_Sizeof, absl::InlinedVector<void*, 7>);
>+BENCHMARK_TEMPLATE(BM_Sizeof, absl::InlinedVector<void*, 8>);
>+
>+BENCHMARK_TEMPLATE(BM_Sizeof, absl::InlinedVector<std::string, 1>);
>+BENCHMARK_TEMPLATE(BM_Sizeof, absl::InlinedVector<std::string, 4>);
>+BENCHMARK_TEMPLATE(BM_Sizeof, absl::InlinedVector<std::string, 7>);
>+BENCHMARK_TEMPLATE(BM_Sizeof, absl::InlinedVector<std::string, 8>);
>+
>+void BM_InlinedVectorIndexInlined(benchmark::State& state) {
>+  absl::InlinedVector<int, 8> v = {1, 2, 3, 4, 5, 6, 7};
>+  for (auto _ : state) {
>+    for (int i = 0; i < 1000; ++i) {
>+      benchmark::DoNotOptimize(v);
>+      benchmark::DoNotOptimize(v[4]);
>+    }
>+  }
>+  state.SetItemsProcessed(1000 * static_cast<int64_t>(state.iterations()));
>+}
>+BENCHMARK(BM_InlinedVectorIndexInlined);
>+
>+void BM_InlinedVectorIndexExternal(benchmark::State& state) {
>+  absl::InlinedVector<int, 8> v = {1, 2, 3, 4, 5, 6, 7, 8, 9, 10};
>+  for (auto _ : state) {
>+    for (int i = 0; i < 1000; ++i) {
>+      benchmark::DoNotOptimize(v);
>+      benchmark::DoNotOptimize(v[4]);
>+    }
>+  }
>+  state.SetItemsProcessed(1000 * static_cast<int64_t>(state.iterations()));
>+}
>+BENCHMARK(BM_InlinedVectorIndexExternal);
>+
>+void BM_StdVectorIndex(benchmark::State& state) {
>+  std::vector<int> v = {1, 2, 3, 4, 5, 6, 7, 8, 9, 10};
>+  for (auto _ : state) {
>+    for (int i = 0; i < 1000; ++i) {
>+      benchmark::DoNotOptimize(v);
>+      benchmark::DoNotOptimize(v[4]);
>+    }
>+  }
>+  state.SetItemsProcessed(1000 * static_cast<int64_t>(state.iterations()));
>+}
>+BENCHMARK(BM_StdVectorIndex);
>+
>+#define UNROLL_2(x)            \
>+  benchmark::DoNotOptimize(x); \
>+  benchmark::DoNotOptimize(x);
>+
>+#define UNROLL_4(x) UNROLL_2(x) UNROLL_2(x)
>+#define UNROLL_8(x) UNROLL_4(x) UNROLL_4(x)
>+#define UNROLL_16(x) UNROLL_8(x) UNROLL_8(x);
>+
>+void BM_InlinedVectorDataInlined(benchmark::State& state) {
>+  absl::InlinedVector<int, 8> v = {1, 2, 3, 4, 5, 6, 7};
>+  for (auto _ : state) {
>+    UNROLL_16(v.data());
>+  }
>+  state.SetItemsProcessed(16 * static_cast<int64_t>(state.iterations()));
>+}
>+BENCHMARK(BM_InlinedVectorDataInlined);
>+
>+void BM_InlinedVectorDataExternal(benchmark::State& state) {
>+  absl::InlinedVector<int, 8> v = {1, 2, 3, 4, 5, 6, 7, 8, 9, 10};
>+  for (auto _ : state) {
>+    UNROLL_16(v.data());
>+  }
>+  state.SetItemsProcessed(16 * static_cast<int64_t>(state.iterations()));
>+}
>+BENCHMARK(BM_InlinedVectorDataExternal);
>+
>+void BM_StdVectorData(benchmark::State& state) {
>+  std::vector<int> v = {1, 2, 3, 4, 5, 6, 7, 8, 9, 10};
>+  for (auto _ : state) {
>+    UNROLL_16(v.data());
>+  }
>+  state.SetItemsProcessed(16 * static_cast<int64_t>(state.iterations()));
>+}
>+BENCHMARK(BM_StdVectorData);
>+
>+void BM_InlinedVectorSizeInlined(benchmark::State& state) {
>+  absl::InlinedVector<int, 8> v = {1, 2, 3, 4, 5, 6, 7};
>+  for (auto _ : state) {
>+    UNROLL_16(v.size());
>+  }
>+  state.SetItemsProcessed(16 * static_cast<int64_t>(state.iterations()));
>+}
>+BENCHMARK(BM_InlinedVectorSizeInlined);
>+
>+void BM_InlinedVectorSizeExternal(benchmark::State& state) {
>+  absl::InlinedVector<int, 8> v = {1, 2, 3, 4, 5, 6, 7, 8, 9, 10};
>+  for (auto _ : state) {
>+    UNROLL_16(v.size());
>+  }
>+  state.SetItemsProcessed(16 * static_cast<int64_t>(state.iterations()));
>+}
>+BENCHMARK(BM_InlinedVectorSizeExternal);
>+
>+void BM_StdVectorSize(benchmark::State& state) {
>+  std::vector<int> v = {1, 2, 3, 4, 5, 6, 7, 8, 9, 10};
>+  for (auto _ : state) {
>+    UNROLL_16(v.size());
>+  }
>+  state.SetItemsProcessed(16 * static_cast<int64_t>(state.iterations()));
>+}
>+BENCHMARK(BM_StdVectorSize);
>+
>+void BM_InlinedVectorEmptyInlined(benchmark::State& state) {
>+  absl::InlinedVector<int, 8> v = {1, 2, 3, 4, 5, 6, 7};
>+  for (auto _ : state) {
>+    UNROLL_16(v.empty());
>+  }
>+  state.SetItemsProcessed(16 * static_cast<int64_t>(state.iterations()));
>+}
>+BENCHMARK(BM_InlinedVectorEmptyInlined);
>+
>+void BM_InlinedVectorEmptyExternal(benchmark::State& state) {
>+  absl::InlinedVector<int, 8> v = {1, 2, 3, 4, 5, 6, 7, 8, 9, 10};
>+  for (auto _ : state) {
>+    UNROLL_16(v.empty());
>+  }
>+  state.SetItemsProcessed(16 * static_cast<int64_t>(state.iterations()));
>+}
>+BENCHMARK(BM_InlinedVectorEmptyExternal);
>+
>+void BM_StdVectorEmpty(benchmark::State& state) {
>+  std::vector<int> v = {1, 2, 3, 4, 5, 6, 7, 8, 9, 10};
>+  for (auto _ : state) {
>+    UNROLL_16(v.empty());
>+  }
>+  state.SetItemsProcessed(16 * static_cast<int64_t>(state.iterations()));
>+}
>+BENCHMARK(BM_StdVectorEmpty);
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/inlined_vector_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/inlined_vector_test.cc
>new file mode 100644
>index 00000000000..196a1bed976
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/inlined_vector_test.cc
>@@ -0,0 +1,1791 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/container/inlined_vector.h"
>+
>+#include <algorithm>
>+#include <forward_list>
>+#include <list>
>+#include <memory>
>+#include <scoped_allocator>
>+#include <sstream>
>+#include <stdexcept>
>+#include <string>
>+#include <vector>
>+
>+#include "gmock/gmock.h"
>+#include "gtest/gtest.h"
>+#include "absl/base/attributes.h"
>+#include "absl/base/internal/exception_testing.h"
>+#include "absl/base/internal/raw_logging.h"
>+#include "absl/base/macros.h"
>+#include "absl/container/internal/test_instance_tracker.h"
>+#include "absl/memory/memory.h"
>+#include "absl/strings/str_cat.h"
>+
>+namespace {
>+
>+using absl::test_internal::CopyableMovableInstance;
>+using absl::test_internal::CopyableOnlyInstance;
>+using absl::test_internal::InstanceTracker;
>+using testing::AllOf;
>+using testing::Each;
>+using testing::ElementsAre;
>+using testing::ElementsAreArray;
>+using testing::Eq;
>+using testing::Gt;
>+using testing::PrintToString;
>+
>+using IntVec = absl::InlinedVector<int, 8>;
>+
>+MATCHER_P(SizeIs, n, "") {
>+  return testing::ExplainMatchResult(n, arg.size(), result_listener);
>+}
>+
>+MATCHER_P(CapacityIs, n, "") {
>+  return testing::ExplainMatchResult(n, arg.capacity(), result_listener);
>+}
>+
>+MATCHER_P(ValueIs, e, "") {
>+  return testing::ExplainMatchResult(e, arg.value(), result_listener);
>+}
>+
>+// TODO(bsamwel): Add support for movable-only types.
>+
>+// Test fixture for typed tests on BaseCountedInstance derived classes, see
>+// test_instance_tracker.h.
>+template <typename T>
>+class InstanceTest : public ::testing::Test {};
>+TYPED_TEST_CASE_P(InstanceTest);
>+
>+// A simple reference counted class to make sure that the proper elements are
>+// destroyed in the erase(begin, end) test.
>+class RefCounted {
>+ public:
>+  RefCounted(int value, int* count) : value_(value), count_(count) {
>+    Ref();
>+  }
>+
>+  RefCounted(const RefCounted& v)
>+      : value_(v.value_), count_(v.count_) {
>+    Ref();
>+  }
>+
>+  ~RefCounted() {
>+    Unref();
>+    count_ = nullptr;
>+  }
>+
>+  friend void swap(RefCounted& a, RefCounted& b) {
>+    using std::swap;
>+    swap(a.value_, b.value_);
>+    swap(a.count_, b.count_);
>+  }
>+
>+  RefCounted& operator=(RefCounted v) {
>+    using std::swap;
>+    swap(*this, v);
>+    return *this;
>+  }
>+
>+  void Ref() const {
>+    ABSL_RAW_CHECK(count_ != nullptr, "");
>+    ++(*count_);
>+  }
>+
>+  void Unref() const {
>+    --(*count_);
>+    ABSL_RAW_CHECK(*count_ >= 0, "");
>+  }
>+
>+  int value_;
>+  int* count_;
>+};
>+
>+using RefCountedVec = absl::InlinedVector<RefCounted, 8>;
>+
>+// A class with a vtable pointer
>+class Dynamic {
>+ public:
>+  virtual ~Dynamic() {}
>+};
>+
>+using DynamicVec = absl::InlinedVector<Dynamic, 8>;
>+
>+// Append 0..len-1 to *v
>+template <typename Container>
>+static void Fill(Container* v, int len, int offset = 0) {
>+  for (int i = 0; i < len; i++) {
>+    v->push_back(i + offset);
>+  }
>+}
>+
>+static IntVec Fill(int len, int offset = 0) {
>+  IntVec v;
>+  Fill(&v, len, offset);
>+  return v;
>+}
>+
>+// This is a stateful allocator, but the state lives outside of the
>+// allocator (in whatever test is using the allocator). This is odd
>+// but helps in tests where the allocator is propagated into nested
>+// containers - that chain of allocators uses the same state and is
>+// thus easier to query for aggregate allocation information.
>+template <typename T>
>+class CountingAllocator : public std::allocator<T> {
>+ public:
>+  using Alloc = std::allocator<T>;
>+  using pointer = typename Alloc::pointer;
>+  using size_type = typename Alloc::size_type;
>+
>+  CountingAllocator() : bytes_used_(nullptr) {}
>+  explicit CountingAllocator(int64_t* b) : bytes_used_(b) {}
>+
>+  template <typename U>
>+  CountingAllocator(const CountingAllocator<U>& x)
>+      : Alloc(x), bytes_used_(x.bytes_used_) {}
>+
>+  pointer allocate(size_type n,
>+                   std::allocator<void>::const_pointer hint = nullptr) {
>+    assert(bytes_used_ != nullptr);
>+    *bytes_used_ += n * sizeof(T);
>+    return Alloc::allocate(n, hint);
>+  }
>+
>+  void deallocate(pointer p, size_type n) {
>+    Alloc::deallocate(p, n);
>+    assert(bytes_used_ != nullptr);
>+    *bytes_used_ -= n * sizeof(T);
>+  }
>+
>+  template<typename U>
>+  class rebind {
>+   public:
>+    using other = CountingAllocator<U>;
>+  };
>+
>+  friend bool operator==(const CountingAllocator& a,
>+                         const CountingAllocator& b) {
>+    return a.bytes_used_ == b.bytes_used_;
>+  }
>+
>+  friend bool operator!=(const CountingAllocator& a,
>+                         const CountingAllocator& b) {
>+    return !(a == b);
>+  }
>+
>+  int64_t* bytes_used_;
>+};
>+
>+TEST(IntVec, SimpleOps) {
>+  for (int len = 0; len < 20; len++) {
>+    IntVec v;
>+    const IntVec& cv = v;  // const alias
>+
>+    Fill(&v, len);
>+    EXPECT_EQ(len, v.size());
>+    EXPECT_LE(len, v.capacity());
>+
>+    for (int i = 0; i < len; i++) {
>+      EXPECT_EQ(i, v[i]);
>+      EXPECT_EQ(i, v.at(i));
>+    }
>+    EXPECT_EQ(v.begin(), v.data());
>+    EXPECT_EQ(cv.begin(), cv.data());
>+
>+    int counter = 0;
>+    for (IntVec::iterator iter = v.begin(); iter != v.end(); ++iter) {
>+      EXPECT_EQ(counter, *iter);
>+      counter++;
>+    }
>+    EXPECT_EQ(counter, len);
>+
>+    counter = 0;
>+    for (IntVec::const_iterator iter = v.begin(); iter != v.end(); ++iter) {
>+      EXPECT_EQ(counter, *iter);
>+      counter++;
>+    }
>+    EXPECT_EQ(counter, len);
>+
>+    counter = 0;
>+    for (IntVec::const_iterator iter = v.cbegin(); iter != v.cend(); ++iter) {
>+      EXPECT_EQ(counter, *iter);
>+      counter++;
>+    }
>+    EXPECT_EQ(counter, len);
>+
>+    if (len > 0) {
>+      EXPECT_EQ(0, v.front());
>+      EXPECT_EQ(len - 1, v.back());
>+      v.pop_back();
>+      EXPECT_EQ(len - 1, v.size());
>+      for (int i = 0; i < v.size(); ++i) {
>+        EXPECT_EQ(i, v[i]);
>+        EXPECT_EQ(i, v.at(i));
>+      }
>+    }
>+  }
>+}
>+
>+TEST(IntVec, AtThrows) {
>+  IntVec v = {1, 2, 3};
>+  EXPECT_EQ(v.at(2), 3);
>+  ABSL_BASE_INTERNAL_EXPECT_FAIL(v.at(3), std::out_of_range,
>+                                 "failed bounds check");
>+}
>+
>+TEST(IntVec, ReverseIterator) {
>+  for (int len = 0; len < 20; len++) {
>+    IntVec v;
>+    Fill(&v, len);
>+
>+    int counter = len;
>+    for (IntVec::reverse_iterator iter = v.rbegin(); iter != v.rend(); ++iter) {
>+      counter--;
>+      EXPECT_EQ(counter, *iter);
>+    }
>+    EXPECT_EQ(counter, 0);
>+
>+    counter = len;
>+    for (IntVec::const_reverse_iterator iter = v.rbegin(); iter != v.rend();
>+         ++iter) {
>+      counter--;
>+      EXPECT_EQ(counter, *iter);
>+    }
>+    EXPECT_EQ(counter, 0);
>+
>+    counter = len;
>+    for (IntVec::const_reverse_iterator iter = v.crbegin(); iter != v.crend();
>+         ++iter) {
>+      counter--;
>+      EXPECT_EQ(counter, *iter);
>+    }
>+    EXPECT_EQ(counter, 0);
>+  }
>+}
>+
>+TEST(IntVec, Erase) {
>+  for (int len = 1; len < 20; len++) {
>+    for (int i = 0; i < len; ++i) {
>+      IntVec v;
>+      Fill(&v, len);
>+      v.erase(v.begin() + i);
>+      EXPECT_EQ(len - 1, v.size());
>+      for (int j = 0; j < i; ++j) {
>+        EXPECT_EQ(j, v[j]);
>+      }
>+      for (int j = i; j < len - 1; ++j) {
>+        EXPECT_EQ(j + 1, v[j]);
>+      }
>+    }
>+  }
>+}
>+
>+// At the end of this test loop, the elements between [erase_begin, erase_end)
>+// should have reference counts == 0, and all others elements should have
>+// reference counts == 1.
>+TEST(RefCountedVec, EraseBeginEnd) {
>+  for (int len = 1; len < 20; ++len) {
>+    for (int erase_begin = 0; erase_begin < len; ++erase_begin) {
>+      for (int erase_end = erase_begin; erase_end <= len; ++erase_end) {
>+        std::vector<int> counts(len, 0);
>+        RefCountedVec v;
>+        for (int i = 0; i < len; ++i) {
>+          v.push_back(RefCounted(i, &counts[i]));
>+        }
>+
>+        int erase_len = erase_end - erase_begin;
>+
>+        v.erase(v.begin() + erase_begin, v.begin() + erase_end);
>+
>+        EXPECT_EQ(len - erase_len, v.size());
>+
>+        // Check the elements before the first element erased.
>+        for (int i = 0; i < erase_begin; ++i) {
>+          EXPECT_EQ(i, v[i].value_);
>+        }
>+
>+        // Check the elements after the first element erased.
>+        for (int i = erase_begin; i < v.size(); ++i) {
>+          EXPECT_EQ(i + erase_len, v[i].value_);
>+        }
>+
>+        // Check that the elements at the beginning are preserved.
>+        for (int i = 0; i < erase_begin; ++i) {
>+          EXPECT_EQ(1, counts[i]);
>+        }
>+
>+        // Check that the erased elements are destroyed
>+        for (int i = erase_begin; i < erase_end; ++i) {
>+          EXPECT_EQ(0, counts[i]);
>+        }
>+
>+        // Check that the elements at the end are preserved.
>+        for (int i = erase_end; i< len; ++i) {
>+          EXPECT_EQ(1, counts[i]);
>+        }
>+      }
>+    }
>+  }
>+}
>+
>+struct NoDefaultCtor {
>+  explicit NoDefaultCtor(int) {}
>+};
>+struct NoCopy {
>+  NoCopy() {}
>+  NoCopy(const NoCopy&) = delete;
>+};
>+struct NoAssign {
>+  NoAssign() {}
>+  NoAssign& operator=(const NoAssign&) = delete;
>+};
>+struct MoveOnly {
>+  MoveOnly() {}
>+  MoveOnly(MoveOnly&&) = default;
>+  MoveOnly& operator=(MoveOnly&&) = default;
>+};
>+TEST(InlinedVectorTest, NoDefaultCtor) {
>+  absl::InlinedVector<NoDefaultCtor, 1> v(10, NoDefaultCtor(2));
>+  (void)v;
>+}
>+TEST(InlinedVectorTest, NoCopy) {
>+  absl::InlinedVector<NoCopy, 1> v(10);
>+  (void)v;
>+}
>+TEST(InlinedVectorTest, NoAssign) {
>+  absl::InlinedVector<NoAssign, 1> v(10);
>+  (void)v;
>+}
>+TEST(InlinedVectorTest, MoveOnly) {
>+  absl::InlinedVector<MoveOnly, 2> v;
>+  v.push_back(MoveOnly{});
>+  v.push_back(MoveOnly{});
>+  v.push_back(MoveOnly{});
>+  v.erase(v.begin());
>+  v.push_back(MoveOnly{});
>+  v.erase(v.begin(), v.begin() + 1);
>+  v.insert(v.begin(), MoveOnly{});
>+  v.emplace(v.begin());
>+  v.emplace(v.begin(), MoveOnly{});
>+}
>+TEST(InlinedVectorTest, Noexcept) {
>+  EXPECT_TRUE(std::is_nothrow_move_constructible<IntVec>::value);
>+  EXPECT_TRUE((std::is_nothrow_move_constructible<
>+               absl::InlinedVector<MoveOnly, 2>>::value));
>+
>+  struct MoveCanThrow {
>+    MoveCanThrow(MoveCanThrow&&) {}
>+  };
>+  EXPECT_EQ(absl::default_allocator_is_nothrow::value,
>+            (std::is_nothrow_move_constructible<
>+                absl::InlinedVector<MoveCanThrow, 2>>::value));
>+}
>+
>+TEST(InlinedVectorTest, EmplaceBack) {
>+  absl::InlinedVector<std::pair<std::string, int>, 1> v;
>+
>+  auto& inlined_element = v.emplace_back("answer", 42);
>+  EXPECT_EQ(&inlined_element, &v[0]);
>+  EXPECT_EQ(inlined_element.first, "answer");
>+  EXPECT_EQ(inlined_element.second, 42);
>+
>+  auto& allocated_element = v.emplace_back("taxicab", 1729);
>+  EXPECT_EQ(&allocated_element, &v[1]);
>+  EXPECT_EQ(allocated_element.first, "taxicab");
>+  EXPECT_EQ(allocated_element.second, 1729);
>+}
>+
>+TEST(InlinedVectorTest, ShrinkToFitGrowingVector) {
>+  absl::InlinedVector<std::pair<std::string, int>, 1> v;
>+
>+  v.shrink_to_fit();
>+  EXPECT_EQ(v.capacity(), 1);
>+
>+  v.emplace_back("answer", 42);
>+  v.shrink_to_fit();
>+  EXPECT_EQ(v.capacity(), 1);
>+
>+  v.emplace_back("taxicab", 1729);
>+  EXPECT_GE(v.capacity(), 2);
>+  v.shrink_to_fit();
>+  EXPECT_EQ(v.capacity(), 2);
>+
>+  v.reserve(100);
>+  EXPECT_GE(v.capacity(), 100);
>+  v.shrink_to_fit();
>+  EXPECT_EQ(v.capacity(), 2);
>+}
>+
>+TEST(InlinedVectorTest, ShrinkToFitEdgeCases) {
>+  {
>+    absl::InlinedVector<std::pair<std::string, int>, 1> v;
>+    v.emplace_back("answer", 42);
>+    v.emplace_back("taxicab", 1729);
>+    EXPECT_GE(v.capacity(), 2);
>+    v.pop_back();
>+    v.shrink_to_fit();
>+    EXPECT_EQ(v.capacity(), 1);
>+    EXPECT_EQ(v[0].first, "answer");
>+    EXPECT_EQ(v[0].second, 42);
>+  }
>+
>+  {
>+    absl::InlinedVector<std::string, 2> v(100);
>+    v.resize(0);
>+    v.shrink_to_fit();
>+    EXPECT_EQ(v.capacity(), 2);  // inlined capacity
>+  }
>+
>+  {
>+    absl::InlinedVector<std::string, 2> v(100);
>+    v.resize(1);
>+    v.shrink_to_fit();
>+    EXPECT_EQ(v.capacity(), 2);  // inlined capacity
>+  }
>+
>+  {
>+    absl::InlinedVector<std::string, 2> v(100);
>+    v.resize(2);
>+    v.shrink_to_fit();
>+    EXPECT_EQ(v.capacity(), 2);
>+  }
>+
>+  {
>+    absl::InlinedVector<std::string, 2> v(100);
>+    v.resize(3);
>+    v.shrink_to_fit();
>+    EXPECT_EQ(v.capacity(), 3);
>+  }
>+}
>+
>+TEST(IntVec, Insert) {
>+  for (int len = 0; len < 20; len++) {
>+    for (int pos = 0; pos <= len; pos++) {
>+      {
>+        // Single element
>+        std::vector<int> std_v;
>+        Fill(&std_v, len);
>+        IntVec v;
>+        Fill(&v, len);
>+
>+        std_v.insert(std_v.begin() + pos, 9999);
>+        IntVec::iterator it = v.insert(v.cbegin() + pos, 9999);
>+        EXPECT_THAT(v, ElementsAreArray(std_v));
>+        EXPECT_EQ(it, v.cbegin() + pos);
>+      }
>+      {
>+        // n elements
>+        std::vector<int> std_v;
>+        Fill(&std_v, len);
>+        IntVec v;
>+        Fill(&v, len);
>+
>+        IntVec::size_type n = 5;
>+        std_v.insert(std_v.begin() + pos, n, 9999);
>+        IntVec::iterator it = v.insert(v.cbegin() + pos, n, 9999);
>+        EXPECT_THAT(v, ElementsAreArray(std_v));
>+        EXPECT_EQ(it, v.cbegin() + pos);
>+      }
>+      {
>+        // Iterator range (random access iterator)
>+        std::vector<int> std_v;
>+        Fill(&std_v, len);
>+        IntVec v;
>+        Fill(&v, len);
>+
>+        const std::vector<int> input = {9999, 8888, 7777};
>+        std_v.insert(std_v.begin() + pos, input.cbegin(), input.cend());
>+        IntVec::iterator it =
>+            v.insert(v.cbegin() + pos, input.cbegin(), input.cend());
>+        EXPECT_THAT(v, ElementsAreArray(std_v));
>+        EXPECT_EQ(it, v.cbegin() + pos);
>+      }
>+      {
>+        // Iterator range (forward iterator)
>+        std::vector<int> std_v;
>+        Fill(&std_v, len);
>+        IntVec v;
>+        Fill(&v, len);
>+
>+        const std::forward_list<int> input = {9999, 8888, 7777};
>+        std_v.insert(std_v.begin() + pos, input.cbegin(), input.cend());
>+        IntVec::iterator it =
>+            v.insert(v.cbegin() + pos, input.cbegin(), input.cend());
>+        EXPECT_THAT(v, ElementsAreArray(std_v));
>+        EXPECT_EQ(it, v.cbegin() + pos);
>+      }
>+      {
>+        // Iterator range (input iterator)
>+        std::vector<int> std_v;
>+        Fill(&std_v, len);
>+        IntVec v;
>+        Fill(&v, len);
>+
>+        std_v.insert(std_v.begin() + pos, {9999, 8888, 7777});
>+        std::istringstream input("9999 8888 7777");
>+        IntVec::iterator it =
>+            v.insert(v.cbegin() + pos, std::istream_iterator<int>(input),
>+                     std::istream_iterator<int>());
>+        EXPECT_THAT(v, ElementsAreArray(std_v));
>+        EXPECT_EQ(it, v.cbegin() + pos);
>+      }
>+      {
>+        // Initializer list
>+        std::vector<int> std_v;
>+        Fill(&std_v, len);
>+        IntVec v;
>+        Fill(&v, len);
>+
>+        std_v.insert(std_v.begin() + pos, {9999, 8888});
>+        IntVec::iterator it = v.insert(v.cbegin() + pos, {9999, 8888});
>+        EXPECT_THAT(v, ElementsAreArray(std_v));
>+        EXPECT_EQ(it, v.cbegin() + pos);
>+      }
>+    }
>+  }
>+}
>+
>+TEST(RefCountedVec, InsertConstructorDestructor) {
>+  // Make sure the proper construction/destruction happen during insert
>+  // operations.
>+  for (int len = 0; len < 20; len++) {
>+    SCOPED_TRACE(len);
>+    for (int pos = 0; pos <= len; pos++) {
>+      SCOPED_TRACE(pos);
>+      std::vector<int> counts(len, 0);
>+      int inserted_count = 0;
>+      RefCountedVec v;
>+      for (int i = 0; i < len; ++i) {
>+        SCOPED_TRACE(i);
>+        v.push_back(RefCounted(i, &counts[i]));
>+      }
>+
>+      EXPECT_THAT(counts, Each(Eq(1)));
>+
>+      RefCounted insert_element(9999, &inserted_count);
>+      EXPECT_EQ(1, inserted_count);
>+      v.insert(v.begin() + pos, insert_element);
>+      EXPECT_EQ(2, inserted_count);
>+      // Check that the elements at the end are preserved.
>+      EXPECT_THAT(counts, Each(Eq(1)));
>+      EXPECT_EQ(2, inserted_count);
>+    }
>+  }
>+}
>+
>+TEST(IntVec, Resize) {
>+  for (int len = 0; len < 20; len++) {
>+    IntVec v;
>+    Fill(&v, len);
>+
>+    // Try resizing up and down by k elements
>+    static const int kResizeElem = 1000000;
>+    for (int k = 0; k < 10; k++) {
>+      // Enlarging resize
>+      v.resize(len+k, kResizeElem);
>+      EXPECT_EQ(len+k, v.size());
>+      EXPECT_LE(len+k, v.capacity());
>+      for (int i = 0; i < len+k; i++) {
>+        if (i < len) {
>+          EXPECT_EQ(i, v[i]);
>+        } else {
>+          EXPECT_EQ(kResizeElem, v[i]);
>+        }
>+      }
>+
>+      // Shrinking resize
>+      v.resize(len, kResizeElem);
>+      EXPECT_EQ(len, v.size());
>+      EXPECT_LE(len, v.capacity());
>+      for (int i = 0; i < len; i++) {
>+        EXPECT_EQ(i, v[i]);
>+      }
>+    }
>+  }
>+}
>+
>+TEST(IntVec, InitWithLength) {
>+  for (int len = 0; len < 20; len++) {
>+    IntVec v(len, 7);
>+    EXPECT_EQ(len, v.size());
>+    EXPECT_LE(len, v.capacity());
>+    for (int i = 0; i < len; i++) {
>+      EXPECT_EQ(7, v[i]);
>+    }
>+  }
>+}
>+
>+TEST(IntVec, CopyConstructorAndAssignment) {
>+  for (int len = 0; len < 20; len++) {
>+    IntVec v;
>+    Fill(&v, len);
>+    EXPECT_EQ(len, v.size());
>+    EXPECT_LE(len, v.capacity());
>+
>+    IntVec v2(v);
>+    EXPECT_TRUE(v == v2) << PrintToString(v) << PrintToString(v2);
>+
>+    for (int start_len = 0; start_len < 20; start_len++) {
>+      IntVec v3;
>+      Fill(&v3, start_len, 99);  // Add dummy elements that should go away
>+      v3 = v;
>+      EXPECT_TRUE(v == v3) << PrintToString(v) << PrintToString(v3);
>+    }
>+  }
>+}
>+
>+TEST(IntVec, AliasingCopyAssignment) {
>+  for (int len = 0; len < 20; ++len) {
>+    IntVec original;
>+    Fill(&original, len);
>+    IntVec dup = original;
>+    dup = *&dup;
>+    EXPECT_EQ(dup, original);
>+  }
>+}
>+
>+TEST(IntVec, MoveConstructorAndAssignment) {
>+  for (int len = 0; len < 20; len++) {
>+    IntVec v_in;
>+    const int inlined_capacity = v_in.capacity();
>+    Fill(&v_in, len);
>+    EXPECT_EQ(len, v_in.size());
>+    EXPECT_LE(len, v_in.capacity());
>+
>+    {
>+      IntVec v_temp(v_in);
>+      auto* old_data = v_temp.data();
>+      IntVec v_out(std::move(v_temp));
>+      EXPECT_TRUE(v_in == v_out) << PrintToString(v_in) << PrintToString(v_out);
>+      if (v_in.size() > inlined_capacity) {
>+        // Allocation is moved as a whole, data stays in place.
>+        EXPECT_TRUE(v_out.data() == old_data);
>+      } else {
>+        EXPECT_FALSE(v_out.data() == old_data);
>+      }
>+    }
>+    for (int start_len = 0; start_len < 20; start_len++) {
>+      IntVec v_out;
>+      Fill(&v_out, start_len, 99);  // Add dummy elements that should go away
>+      IntVec v_temp(v_in);
>+      auto* old_data = v_temp.data();
>+      v_out = std::move(v_temp);
>+      EXPECT_TRUE(v_in == v_out) << PrintToString(v_in) << PrintToString(v_out);
>+      if (v_in.size() > inlined_capacity) {
>+        // Allocation is moved as a whole, data stays in place.
>+        EXPECT_TRUE(v_out.data() == old_data);
>+      } else {
>+        EXPECT_FALSE(v_out.data() == old_data);
>+      }
>+    }
>+  }
>+}
>+
>+class NotTriviallyDestructible {
>+ public:
>+  NotTriviallyDestructible() : p_(new int(1)) {}
>+  explicit NotTriviallyDestructible(int i) : p_(new int(i)) {}
>+
>+  NotTriviallyDestructible(const NotTriviallyDestructible& other)
>+      : p_(new int(*other.p_)) {}
>+
>+  NotTriviallyDestructible& operator=(const NotTriviallyDestructible& other) {
>+    p_ = absl::make_unique<int>(*other.p_);
>+    return *this;
>+  }
>+
>+  bool operator==(const NotTriviallyDestructible& other) const {
>+    return *p_ == *other.p_;
>+  }
>+
>+ private:
>+  std::unique_ptr<int> p_;
>+};
>+
>+TEST(AliasingTest, Emplace) {
>+  for (int i = 2; i < 20; ++i) {
>+    absl::InlinedVector<NotTriviallyDestructible, 10> vec;
>+    for (int j = 0; j < i; ++j) {
>+      vec.push_back(NotTriviallyDestructible(j));
>+    }
>+    vec.emplace(vec.begin(), vec[0]);
>+    EXPECT_EQ(vec[0], vec[1]);
>+    vec.emplace(vec.begin() + i / 2, vec[i / 2]);
>+    EXPECT_EQ(vec[i / 2], vec[i / 2 + 1]);
>+    vec.emplace(vec.end() - 1, vec.back());
>+    EXPECT_EQ(vec[vec.size() - 2], vec.back());
>+  }
>+}
>+
>+TEST(AliasingTest, InsertWithCount) {
>+  for (int i = 1; i < 20; ++i) {
>+    absl::InlinedVector<NotTriviallyDestructible, 10> vec;
>+    for (int j = 0; j < i; ++j) {
>+      vec.push_back(NotTriviallyDestructible(j));
>+    }
>+    for (int n = 0; n < 5; ++n) {
>+      // We use back where we can because it's guaranteed to become invalidated
>+      vec.insert(vec.begin(), n, vec.back());
>+      auto b = vec.begin();
>+      EXPECT_TRUE(
>+          std::all_of(b, b + n, [&vec](const NotTriviallyDestructible& x) {
>+            return x == vec.back();
>+          }));
>+
>+      auto m_idx = vec.size() / 2;
>+      vec.insert(vec.begin() + m_idx, n, vec.back());
>+      auto m = vec.begin() + m_idx;
>+      EXPECT_TRUE(
>+          std::all_of(m, m + n, [&vec](const NotTriviallyDestructible& x) {
>+            return x == vec.back();
>+          }));
>+
>+      // We want distinct values so the equality test is meaningful,
>+      // vec[vec.size() - 1] is also almost always invalidated.
>+      auto old_e = vec.size() - 1;
>+      auto val = vec[old_e];
>+      vec.insert(vec.end(), n, vec[old_e]);
>+      auto e = vec.begin() + old_e;
>+      EXPECT_TRUE(std::all_of(
>+          e, e + n,
>+          [&val](const NotTriviallyDestructible& x) { return x == val; }));
>+    }
>+  }
>+}
>+
>+TEST(OverheadTest, Storage) {
>+  // Check for size overhead.
>+  // In particular, ensure that std::allocator doesn't cost anything to store.
>+  // The union should be absorbing some of the allocation bookkeeping overhead
>+  // in the larger vectors, leaving only the size_ field as overhead.
>+  EXPECT_EQ(2 * sizeof(int*),
>+            sizeof(absl::InlinedVector<int*, 1>) - 1 * sizeof(int*));
>+  EXPECT_EQ(1 * sizeof(int*),
>+            sizeof(absl::InlinedVector<int*, 2>) - 2 * sizeof(int*));
>+  EXPECT_EQ(1 * sizeof(int*),
>+            sizeof(absl::InlinedVector<int*, 3>) - 3 * sizeof(int*));
>+  EXPECT_EQ(1 * sizeof(int*),
>+            sizeof(absl::InlinedVector<int*, 4>) - 4 * sizeof(int*));
>+  EXPECT_EQ(1 * sizeof(int*),
>+            sizeof(absl::InlinedVector<int*, 5>) - 5 * sizeof(int*));
>+  EXPECT_EQ(1 * sizeof(int*),
>+            sizeof(absl::InlinedVector<int*, 6>) - 6 * sizeof(int*));
>+  EXPECT_EQ(1 * sizeof(int*),
>+            sizeof(absl::InlinedVector<int*, 7>) - 7 * sizeof(int*));
>+  EXPECT_EQ(1 * sizeof(int*),
>+            sizeof(absl::InlinedVector<int*, 8>) - 8 * sizeof(int*));
>+}
>+
>+TEST(IntVec, Clear) {
>+  for (int len = 0; len < 20; len++) {
>+    SCOPED_TRACE(len);
>+    IntVec v;
>+    Fill(&v, len);
>+    v.clear();
>+    EXPECT_EQ(0, v.size());
>+    EXPECT_EQ(v.begin(), v.end());
>+  }
>+}
>+
>+TEST(IntVec, Reserve) {
>+  for (int len = 0; len < 20; len++) {
>+    IntVec v;
>+    Fill(&v, len);
>+
>+    for (int newlen = 0; newlen < 100; newlen++) {
>+      const int* start_rep = v.data();
>+      v.reserve(newlen);
>+      const int* final_rep = v.data();
>+      if (newlen <= len) {
>+        EXPECT_EQ(start_rep, final_rep);
>+      }
>+      EXPECT_LE(newlen, v.capacity());
>+
>+      // Filling up to newlen should not change rep
>+      while (v.size() < newlen) {
>+        v.push_back(0);
>+      }
>+      EXPECT_EQ(final_rep, v.data());
>+    }
>+  }
>+}
>+
>+TEST(StringVec, SelfRefPushBack) {
>+  std::vector<std::string> std_v;
>+  absl::InlinedVector<std::string, 4> v;
>+  const std::string s = "A quite long std::string to ensure heap.";
>+  std_v.push_back(s);
>+  v.push_back(s);
>+  for (int i = 0; i < 20; ++i) {
>+    EXPECT_THAT(v, ElementsAreArray(std_v));
>+
>+    v.push_back(v.back());
>+    std_v.push_back(std_v.back());
>+  }
>+  EXPECT_THAT(v, ElementsAreArray(std_v));
>+}
>+
>+TEST(StringVec, SelfRefPushBackWithMove) {
>+  std::vector<std::string> std_v;
>+  absl::InlinedVector<std::string, 4> v;
>+  const std::string s = "A quite long std::string to ensure heap.";
>+  std_v.push_back(s);
>+  v.push_back(s);
>+  for (int i = 0; i < 20; ++i) {
>+    EXPECT_EQ(v.back(), std_v.back());
>+
>+    v.push_back(std::move(v.back()));
>+    std_v.push_back(std::move(std_v.back()));
>+  }
>+  EXPECT_EQ(v.back(), std_v.back());
>+}
>+
>+TEST(StringVec, SelfMove) {
>+  const std::string s = "A quite long std::string to ensure heap.";
>+  for (int len = 0; len < 20; len++) {
>+    SCOPED_TRACE(len);
>+    absl::InlinedVector<std::string, 8> v;
>+    for (int i = 0; i < len; ++i) {
>+      SCOPED_TRACE(i);
>+      v.push_back(s);
>+    }
>+    // Indirection necessary to avoid compiler warning.
>+    v = std::move(*(&v));
>+    // Ensure that the inlined vector is still in a valid state by copying it.
>+    // We don't expect specific contents since a self-move results in an
>+    // unspecified valid state.
>+    std::vector<std::string> copy(v.begin(), v.end());
>+  }
>+}
>+
>+TEST(IntVec, Swap) {
>+  for (int l1 = 0; l1 < 20; l1++) {
>+    SCOPED_TRACE(l1);
>+    for (int l2 = 0; l2 < 20; l2++) {
>+      SCOPED_TRACE(l2);
>+      IntVec a = Fill(l1, 0);
>+      IntVec b = Fill(l2, 100);
>+      {
>+        using std::swap;
>+        swap(a, b);
>+      }
>+      EXPECT_EQ(l1, b.size());
>+      EXPECT_EQ(l2, a.size());
>+      for (int i = 0; i < l1; i++) {
>+        SCOPED_TRACE(i);
>+        EXPECT_EQ(i, b[i]);
>+      }
>+      for (int i = 0; i < l2; i++) {
>+        SCOPED_TRACE(i);
>+        EXPECT_EQ(100 + i, a[i]);
>+      }
>+    }
>+  }
>+}
>+
>+TYPED_TEST_P(InstanceTest, Swap) {
>+  using Instance = TypeParam;
>+  using InstanceVec = absl::InlinedVector<Instance, 8>;
>+  for (int l1 = 0; l1 < 20; l1++) {
>+    SCOPED_TRACE(l1);
>+    for (int l2 = 0; l2 < 20; l2++) {
>+      SCOPED_TRACE(l2);
>+      InstanceTracker tracker;
>+      InstanceVec a, b;
>+      const size_t inlined_capacity = a.capacity();
>+      for (int i = 0; i < l1; i++) a.push_back(Instance(i));
>+      for (int i = 0; i < l2; i++) b.push_back(Instance(100+i));
>+      EXPECT_EQ(tracker.instances(), l1 + l2);
>+      tracker.ResetCopiesMovesSwaps();
>+      {
>+        using std::swap;
>+        swap(a, b);
>+      }
>+      EXPECT_EQ(tracker.instances(), l1 + l2);
>+      if (a.size() > inlined_capacity && b.size() > inlined_capacity) {
>+        EXPECT_EQ(tracker.swaps(), 0);  // Allocations are swapped.
>+        EXPECT_EQ(tracker.moves(), 0);
>+      } else if (a.size() <= inlined_capacity && b.size() <= inlined_capacity) {
>+        EXPECT_EQ(tracker.swaps(), std::min(l1, l2));
>+        // TODO(bsamwel): This should use moves when the type is movable.
>+        EXPECT_EQ(tracker.copies(), std::max(l1, l2) - std::min(l1, l2));
>+      } else {
>+        // One is allocated and the other isn't. The allocation is transferred
>+        // without copying elements, and the inlined instances are copied/moved.
>+        EXPECT_EQ(tracker.swaps(), 0);
>+        // TODO(bsamwel): This should use moves when the type is movable.
>+        EXPECT_EQ(tracker.copies(), std::min(l1, l2));
>+      }
>+
>+      EXPECT_EQ(l1, b.size());
>+      EXPECT_EQ(l2, a.size());
>+      for (int i = 0; i < l1; i++) {
>+        EXPECT_EQ(i, b[i].value());
>+      }
>+      for (int i = 0; i < l2; i++) {
>+        EXPECT_EQ(100 + i, a[i].value());
>+      }
>+    }
>+  }
>+}
>+
>+TEST(IntVec, EqualAndNotEqual) {
>+  IntVec a, b;
>+  EXPECT_TRUE(a == b);
>+  EXPECT_FALSE(a != b);
>+
>+  a.push_back(3);
>+  EXPECT_FALSE(a == b);
>+  EXPECT_TRUE(a != b);
>+
>+  b.push_back(3);
>+  EXPECT_TRUE(a == b);
>+  EXPECT_FALSE(a != b);
>+
>+  b.push_back(7);
>+  EXPECT_FALSE(a == b);
>+  EXPECT_TRUE(a != b);
>+
>+  a.push_back(6);
>+  EXPECT_FALSE(a == b);
>+  EXPECT_TRUE(a != b);
>+
>+  a.clear();
>+  b.clear();
>+  for (int i = 0; i < 100; i++) {
>+    a.push_back(i);
>+    b.push_back(i);
>+    EXPECT_TRUE(a == b);
>+    EXPECT_FALSE(a != b);
>+
>+    b[i] = b[i] + 1;
>+    EXPECT_FALSE(a == b);
>+    EXPECT_TRUE(a != b);
>+
>+    b[i] = b[i] - 1;    // Back to before
>+    EXPECT_TRUE(a == b);
>+    EXPECT_FALSE(a != b);
>+  }
>+}
>+
>+TEST(IntVec, RelationalOps) {
>+  IntVec a, b;
>+  EXPECT_FALSE(a < b);
>+  EXPECT_FALSE(b < a);
>+  EXPECT_FALSE(a > b);
>+  EXPECT_FALSE(b > a);
>+  EXPECT_TRUE(a <= b);
>+  EXPECT_TRUE(b <= a);
>+  EXPECT_TRUE(a >= b);
>+  EXPECT_TRUE(b >= a);
>+  b.push_back(3);
>+  EXPECT_TRUE(a < b);
>+  EXPECT_FALSE(b < a);
>+  EXPECT_FALSE(a > b);
>+  EXPECT_TRUE(b > a);
>+  EXPECT_TRUE(a <= b);
>+  EXPECT_FALSE(b <= a);
>+  EXPECT_FALSE(a >= b);
>+  EXPECT_TRUE(b >= a);
>+}
>+
>+TYPED_TEST_P(InstanceTest, CountConstructorsDestructors) {
>+  using Instance = TypeParam;
>+  using InstanceVec = absl::InlinedVector<Instance, 8>;
>+  InstanceTracker tracker;
>+  for (int len = 0; len < 20; len++) {
>+    SCOPED_TRACE(len);
>+    tracker.ResetCopiesMovesSwaps();
>+
>+    InstanceVec v;
>+    const size_t inlined_capacity = v.capacity();
>+    for (int i = 0; i < len; i++) {
>+      v.push_back(Instance(i));
>+    }
>+    EXPECT_EQ(tracker.instances(), len);
>+    EXPECT_GE(tracker.copies() + tracker.moves(),
>+              len);  // More due to reallocation.
>+    tracker.ResetCopiesMovesSwaps();
>+
>+    // Enlarging resize() must construct some objects
>+    tracker.ResetCopiesMovesSwaps();
>+    v.resize(len + 10, Instance(100));
>+    EXPECT_EQ(tracker.instances(), len + 10);
>+    if (len <= inlined_capacity && len + 10 > inlined_capacity) {
>+      EXPECT_EQ(tracker.copies() + tracker.moves(), 10 + len);
>+    } else {
>+      // Only specify a minimum number of copies + moves. We don't want to
>+      // depend on the reallocation policy here.
>+      EXPECT_GE(tracker.copies() + tracker.moves(),
>+                10);  // More due to reallocation.
>+    }
>+
>+    // Shrinking resize() must destroy some objects
>+    tracker.ResetCopiesMovesSwaps();
>+    v.resize(len, Instance(100));
>+    EXPECT_EQ(tracker.instances(), len);
>+    EXPECT_EQ(tracker.copies(), 0);
>+    EXPECT_EQ(tracker.moves(), 0);
>+
>+    // reserve() must not increase the number of initialized objects
>+    SCOPED_TRACE("reserve");
>+    v.reserve(len+1000);
>+    EXPECT_EQ(tracker.instances(), len);
>+    EXPECT_EQ(tracker.copies() + tracker.moves(), len);
>+
>+    // pop_back() and erase() must destroy one object
>+    if (len > 0) {
>+      tracker.ResetCopiesMovesSwaps();
>+      v.pop_back();
>+      EXPECT_EQ(tracker.instances(), len - 1);
>+      EXPECT_EQ(tracker.copies(), 0);
>+      EXPECT_EQ(tracker.moves(), 0);
>+
>+      if (!v.empty()) {
>+        tracker.ResetCopiesMovesSwaps();
>+        v.erase(v.begin());
>+        EXPECT_EQ(tracker.instances(), len - 2);
>+        EXPECT_EQ(tracker.copies() + tracker.moves(), len - 2);
>+      }
>+    }
>+
>+    tracker.ResetCopiesMovesSwaps();
>+    int instances_before_empty_erase = tracker.instances();
>+    v.erase(v.begin(), v.begin());
>+    EXPECT_EQ(tracker.instances(), instances_before_empty_erase);
>+    EXPECT_EQ(tracker.copies() + tracker.moves(), 0);
>+  }
>+}
>+
>+TYPED_TEST_P(InstanceTest, CountConstructorsDestructorsOnCopyConstruction) {
>+  using Instance = TypeParam;
>+  using InstanceVec = absl::InlinedVector<Instance, 8>;
>+  InstanceTracker tracker;
>+  for (int len = 0; len < 20; len++) {
>+    SCOPED_TRACE(len);
>+    tracker.ResetCopiesMovesSwaps();
>+
>+    InstanceVec v;
>+    for (int i = 0; i < len; i++) {
>+      v.push_back(Instance(i));
>+    }
>+    EXPECT_EQ(tracker.instances(), len);
>+    EXPECT_GE(tracker.copies() + tracker.moves(),
>+              len);  // More due to reallocation.
>+    tracker.ResetCopiesMovesSwaps();
>+    {  // Copy constructor should create 'len' more instances.
>+      InstanceVec v_copy(v);
>+      EXPECT_EQ(tracker.instances(), len + len);
>+      EXPECT_EQ(tracker.copies(), len);
>+      EXPECT_EQ(tracker.moves(), 0);
>+    }
>+    EXPECT_EQ(tracker.instances(), len);
>+  }
>+}
>+
>+TYPED_TEST_P(InstanceTest, CountConstructorsDestructorsOnMoveConstruction) {
>+  using Instance = TypeParam;
>+  using InstanceVec = absl::InlinedVector<Instance, 8>;
>+  InstanceTracker tracker;
>+  for (int len = 0; len < 20; len++) {
>+    SCOPED_TRACE(len);
>+    tracker.ResetCopiesMovesSwaps();
>+
>+    InstanceVec v;
>+    const size_t inlined_capacity = v.capacity();
>+    for (int i = 0; i < len; i++) {
>+      v.push_back(Instance(i));
>+    }
>+    EXPECT_EQ(tracker.instances(), len);
>+    EXPECT_GE(tracker.copies() + tracker.moves(),
>+              len);  // More due to reallocation.
>+    tracker.ResetCopiesMovesSwaps();
>+    {
>+      InstanceVec v_copy(std::move(v));
>+      if (len > inlined_capacity) {
>+        // Allocation is moved as a whole.
>+        EXPECT_EQ(tracker.instances(), len);
>+        EXPECT_EQ(tracker.live_instances(), len);
>+        // Tests an implementation detail, don't rely on this in your code.
>+        EXPECT_EQ(v.size(), 0);  // NOLINT misc-use-after-move
>+        EXPECT_EQ(tracker.copies(), 0);
>+        EXPECT_EQ(tracker.moves(), 0);
>+      } else {
>+        EXPECT_EQ(tracker.instances(), len + len);
>+        if (Instance::supports_move()) {
>+          EXPECT_EQ(tracker.live_instances(), len);
>+          EXPECT_EQ(tracker.copies(), 0);
>+          EXPECT_EQ(tracker.moves(), len);
>+        } else {
>+          EXPECT_EQ(tracker.live_instances(), len + len);
>+          EXPECT_EQ(tracker.copies(), len);
>+          EXPECT_EQ(tracker.moves(), 0);
>+        }
>+      }
>+      EXPECT_EQ(tracker.swaps(), 0);
>+    }
>+  }
>+}
>+
>+TYPED_TEST_P(InstanceTest, CountConstructorsDestructorsOnAssignment) {
>+  using Instance = TypeParam;
>+  using InstanceVec = absl::InlinedVector<Instance, 8>;
>+  InstanceTracker tracker;
>+  for (int len = 0; len < 20; len++) {
>+    SCOPED_TRACE(len);
>+    for (int longorshort = 0; longorshort <= 1; ++longorshort) {
>+      SCOPED_TRACE(longorshort);
>+      tracker.ResetCopiesMovesSwaps();
>+
>+      InstanceVec longer, shorter;
>+      for (int i = 0; i < len; i++) {
>+        longer.push_back(Instance(i));
>+        shorter.push_back(Instance(i));
>+      }
>+      longer.push_back(Instance(len));
>+      EXPECT_EQ(tracker.instances(), len + len + 1);
>+      EXPECT_GE(tracker.copies() + tracker.moves(),
>+                len + len + 1);  // More due to reallocation.
>+
>+      tracker.ResetCopiesMovesSwaps();
>+      if (longorshort) {
>+        shorter = longer;
>+        EXPECT_EQ(tracker.instances(), (len + 1) + (len + 1));
>+        EXPECT_GE(tracker.copies() + tracker.moves(),
>+                  len + 1);  // More due to reallocation.
>+      } else {
>+        longer = shorter;
>+        EXPECT_EQ(tracker.instances(), len + len);
>+        EXPECT_EQ(tracker.copies() + tracker.moves(), len);
>+      }
>+    }
>+  }
>+}
>+
>+TYPED_TEST_P(InstanceTest, CountConstructorsDestructorsOnMoveAssignment) {
>+  using Instance = TypeParam;
>+  using InstanceVec = absl::InlinedVector<Instance, 8>;
>+  InstanceTracker tracker;
>+  for (int len = 0; len < 20; len++) {
>+    SCOPED_TRACE(len);
>+    for (int longorshort = 0; longorshort <= 1; ++longorshort) {
>+      SCOPED_TRACE(longorshort);
>+      tracker.ResetCopiesMovesSwaps();
>+
>+      InstanceVec longer, shorter;
>+      const int inlined_capacity = longer.capacity();
>+      for (int i = 0; i < len; i++) {
>+        longer.push_back(Instance(i));
>+        shorter.push_back(Instance(i));
>+      }
>+      longer.push_back(Instance(len));
>+      EXPECT_EQ(tracker.instances(), len + len + 1);
>+      EXPECT_GE(tracker.copies() + tracker.moves(),
>+                len + len + 1);  // More due to reallocation.
>+
>+      tracker.ResetCopiesMovesSwaps();
>+      int src_len;
>+      if (longorshort) {
>+        src_len = len + 1;
>+        shorter = std::move(longer);
>+      } else {
>+        src_len = len;
>+        longer = std::move(shorter);
>+      }
>+      if (src_len > inlined_capacity) {
>+        // Allocation moved as a whole.
>+        EXPECT_EQ(tracker.instances(), src_len);
>+        EXPECT_EQ(tracker.live_instances(), src_len);
>+        EXPECT_EQ(tracker.copies(), 0);
>+        EXPECT_EQ(tracker.moves(), 0);
>+      } else {
>+        // Elements are all copied.
>+        EXPECT_EQ(tracker.instances(), src_len + src_len);
>+        if (Instance::supports_move()) {
>+          EXPECT_EQ(tracker.copies(), 0);
>+          EXPECT_EQ(tracker.moves(), src_len);
>+          EXPECT_EQ(tracker.live_instances(), src_len);
>+        } else {
>+          EXPECT_EQ(tracker.copies(), src_len);
>+          EXPECT_EQ(tracker.moves(), 0);
>+          EXPECT_EQ(tracker.live_instances(), src_len + src_len);
>+        }
>+      }
>+      EXPECT_EQ(tracker.swaps(), 0);
>+    }
>+  }
>+}
>+
>+TEST(CountElemAssign, SimpleTypeWithInlineBacking) {
>+  for (size_t original_size = 0; original_size <= 5; ++original_size) {
>+    SCOPED_TRACE(original_size);
>+    // Original contents are [12345, 12345, ...]
>+    std::vector<int> original_contents(original_size, 12345);
>+
>+    absl::InlinedVector<int, 2> v(original_contents.begin(),
>+                                  original_contents.end());
>+    v.assign(2, 123);
>+    EXPECT_THAT(v, AllOf(SizeIs(2), ElementsAre(123, 123)));
>+    if (original_size <= 2) {
>+      // If the original had inline backing, it should stay inline.
>+      EXPECT_EQ(2, v.capacity());
>+    }
>+  }
>+}
>+
>+TEST(CountElemAssign, SimpleTypeWithAllocation) {
>+  for (size_t original_size = 0; original_size <= 5; ++original_size) {
>+    SCOPED_TRACE(original_size);
>+    // Original contents are [12345, 12345, ...]
>+    std::vector<int> original_contents(original_size, 12345);
>+
>+    absl::InlinedVector<int, 2> v(original_contents.begin(),
>+                                  original_contents.end());
>+    v.assign(3, 123);
>+    EXPECT_THAT(v, AllOf(SizeIs(3), ElementsAre(123, 123, 123)));
>+    EXPECT_LE(v.size(), v.capacity());
>+  }
>+}
>+
>+TYPED_TEST_P(InstanceTest, CountElemAssignInlineBacking) {
>+  using Instance = TypeParam;
>+  for (size_t original_size = 0; original_size <= 5; ++original_size) {
>+    SCOPED_TRACE(original_size);
>+    // Original contents are [12345, 12345, ...]
>+    std::vector<Instance> original_contents(original_size, Instance(12345));
>+
>+    absl::InlinedVector<Instance, 2> v(original_contents.begin(),
>+                                       original_contents.end());
>+    v.assign(2, Instance(123));
>+    EXPECT_THAT(v, AllOf(SizeIs(2), ElementsAre(ValueIs(123), ValueIs(123))));
>+    if (original_size <= 2) {
>+      // If the original had inline backing, it should stay inline.
>+      EXPECT_EQ(2, v.capacity());
>+    }
>+  }
>+}
>+
>+template <typename Instance>
>+void InstanceCountElemAssignWithAllocationTest() {
>+  for (size_t original_size = 0; original_size <= 5; ++original_size) {
>+    SCOPED_TRACE(original_size);
>+    // Original contents are [12345, 12345, ...]
>+    std::vector<Instance> original_contents(original_size, Instance(12345));
>+
>+    absl::InlinedVector<Instance, 2> v(original_contents.begin(),
>+                                       original_contents.end());
>+    v.assign(3, Instance(123));
>+    EXPECT_THAT(v,
>+                AllOf(SizeIs(3),
>+                      ElementsAre(ValueIs(123), ValueIs(123), ValueIs(123))));
>+    EXPECT_LE(v.size(), v.capacity());
>+  }
>+}
>+TEST(CountElemAssign, WithAllocationCopyableInstance) {
>+  InstanceCountElemAssignWithAllocationTest<CopyableOnlyInstance>();
>+}
>+TEST(CountElemAssign, WithAllocationCopyableMovableInstance) {
>+  InstanceCountElemAssignWithAllocationTest<CopyableMovableInstance>();
>+}
>+
>+TEST(RangedConstructor, SimpleType) {
>+  std::vector<int> source_v = {4, 5, 6};
>+  // First try to fit in inline backing
>+  absl::InlinedVector<int, 4> v(source_v.begin(), source_v.end());
>+  EXPECT_EQ(3, v.size());
>+  EXPECT_EQ(4, v.capacity());  // Indication that we're still on inlined storage
>+  EXPECT_EQ(4, v[0]);
>+  EXPECT_EQ(5, v[1]);
>+  EXPECT_EQ(6, v[2]);
>+
>+  // Now, force a re-allocate
>+  absl::InlinedVector<int, 2> realloc_v(source_v.begin(), source_v.end());
>+  EXPECT_EQ(3, realloc_v.size());
>+  EXPECT_LT(2, realloc_v.capacity());
>+  EXPECT_EQ(4, realloc_v[0]);
>+  EXPECT_EQ(5, realloc_v[1]);
>+  EXPECT_EQ(6, realloc_v[2]);
>+}
>+
>+// Test for ranged constructors using Instance as the element type and
>+// SourceContainer as the source container type.
>+template <typename Instance, typename SourceContainer, int inlined_capacity>
>+void InstanceRangedConstructorTestForContainer() {
>+  InstanceTracker tracker;
>+  SourceContainer source_v = {Instance(0), Instance(1)};
>+  tracker.ResetCopiesMovesSwaps();
>+  absl::InlinedVector<Instance, inlined_capacity> v(source_v.begin(),
>+                                                    source_v.end());
>+  EXPECT_EQ(2, v.size());
>+  EXPECT_LT(1, v.capacity());
>+  EXPECT_EQ(0, v[0].value());
>+  EXPECT_EQ(1, v[1].value());
>+  EXPECT_EQ(tracker.copies(), 2);
>+  EXPECT_EQ(tracker.moves(), 0);
>+}
>+
>+template <typename Instance, int inlined_capacity>
>+void InstanceRangedConstructorTestWithCapacity() {
>+  // Test with const and non-const, random access and non-random-access sources.
>+  // TODO(bsamwel): Test with an input iterator source.
>+  {
>+    SCOPED_TRACE("std::list");
>+    InstanceRangedConstructorTestForContainer<Instance, std::list<Instance>,
>+                                              inlined_capacity>();
>+    {
>+      SCOPED_TRACE("const std::list");
>+      InstanceRangedConstructorTestForContainer<
>+          Instance, const std::list<Instance>, inlined_capacity>();
>+    }
>+    {
>+      SCOPED_TRACE("std::vector");
>+      InstanceRangedConstructorTestForContainer<Instance, std::vector<Instance>,
>+                                                inlined_capacity>();
>+    }
>+    {
>+      SCOPED_TRACE("const std::vector");
>+      InstanceRangedConstructorTestForContainer<
>+          Instance, const std::vector<Instance>, inlined_capacity>();
>+    }
>+  }
>+}
>+
>+TYPED_TEST_P(InstanceTest, RangedConstructor) {
>+  using Instance = TypeParam;
>+  SCOPED_TRACE("capacity=1");
>+  InstanceRangedConstructorTestWithCapacity<Instance, 1>();
>+  SCOPED_TRACE("capacity=2");
>+  InstanceRangedConstructorTestWithCapacity<Instance, 2>();
>+}
>+
>+TEST(RangedConstructor, ElementsAreConstructed) {
>+  std::vector<std::string> source_v = {"cat", "dog"};
>+
>+  // Force expansion and re-allocation of v.  Ensures that when the vector is
>+  // expanded that new elements are constructed.
>+  absl::InlinedVector<std::string, 1> v(source_v.begin(), source_v.end());
>+  EXPECT_EQ("cat", v[0]);
>+  EXPECT_EQ("dog", v[1]);
>+}
>+
>+TEST(RangedAssign, SimpleType) {
>+  // Test for all combinations of original sizes (empty and non-empty inline,
>+  // and out of line) and target sizes.
>+  for (size_t original_size = 0; original_size <= 5; ++original_size) {
>+    SCOPED_TRACE(original_size);
>+    // Original contents are [12345, 12345, ...]
>+    std::vector<int> original_contents(original_size, 12345);
>+
>+    for (size_t target_size = 0; target_size <= 5; ++target_size) {
>+      SCOPED_TRACE(target_size);
>+
>+      // New contents are [3, 4, ...]
>+      std::vector<int> new_contents;
>+      for (size_t i = 0; i < target_size; ++i) {
>+        new_contents.push_back(i + 3);
>+      }
>+
>+      absl::InlinedVector<int, 3> v(original_contents.begin(),
>+                                    original_contents.end());
>+      v.assign(new_contents.begin(), new_contents.end());
>+
>+      EXPECT_EQ(new_contents.size(), v.size());
>+      EXPECT_LE(new_contents.size(), v.capacity());
>+      if (target_size <= 3 && original_size <= 3) {
>+        // Storage should stay inline when target size is small.
>+        EXPECT_EQ(3, v.capacity());
>+      }
>+      EXPECT_THAT(v, ElementsAreArray(new_contents));
>+    }
>+  }
>+}
>+
>+// Returns true if lhs and rhs have the same value.
>+template <typename Instance>
>+static bool InstanceValuesEqual(const Instance& lhs, const Instance& rhs) {
>+  return lhs.value() == rhs.value();
>+}
>+
>+// Test for ranged assign() using Instance as the element type and
>+// SourceContainer as the source container type.
>+template <typename Instance, typename SourceContainer>
>+void InstanceRangedAssignTestForContainer() {
>+  // Test for all combinations of original sizes (empty and non-empty inline,
>+  // and out of line) and target sizes.
>+  for (size_t original_size = 0; original_size <= 5; ++original_size) {
>+    SCOPED_TRACE(original_size);
>+    // Original contents are [12345, 12345, ...]
>+    std::vector<Instance> original_contents(original_size, Instance(12345));
>+
>+    for (size_t target_size = 0; target_size <= 5; ++target_size) {
>+      SCOPED_TRACE(target_size);
>+
>+      // New contents are [3, 4, ...]
>+      // Generate data using a non-const container, because SourceContainer
>+      // itself may be const.
>+      // TODO(bsamwel): Test with an input iterator.
>+      std::vector<Instance> new_contents_in;
>+      for (size_t i = 0; i < target_size; ++i) {
>+        new_contents_in.push_back(Instance(i + 3));
>+      }
>+      SourceContainer new_contents(new_contents_in.begin(),
>+                                   new_contents_in.end());
>+
>+      absl::InlinedVector<Instance, 3> v(original_contents.begin(),
>+                                         original_contents.end());
>+      v.assign(new_contents.begin(), new_contents.end());
>+
>+      EXPECT_EQ(new_contents.size(), v.size());
>+      EXPECT_LE(new_contents.size(), v.capacity());
>+      if (target_size <= 3 && original_size <= 3) {
>+        // Storage should stay inline when target size is small.
>+        EXPECT_EQ(3, v.capacity());
>+      }
>+      EXPECT_TRUE(std::equal(v.begin(), v.end(), new_contents.begin(),
>+                             InstanceValuesEqual<Instance>));
>+    }
>+  }
>+}
>+
>+TYPED_TEST_P(InstanceTest, RangedAssign) {
>+  using Instance = TypeParam;
>+  // Test with const and non-const, random access and non-random-access sources.
>+  // TODO(bsamwel): Test with an input iterator source.
>+  SCOPED_TRACE("std::list");
>+  InstanceRangedAssignTestForContainer<Instance, std::list<Instance>>();
>+  SCOPED_TRACE("const std::list");
>+  InstanceRangedAssignTestForContainer<Instance, const std::list<Instance>>();
>+  SCOPED_TRACE("std::vector");
>+  InstanceRangedAssignTestForContainer<Instance, std::vector<Instance>>();
>+  SCOPED_TRACE("const std::vector");
>+  InstanceRangedAssignTestForContainer<Instance, const std::vector<Instance>>();
>+}
>+
>+TEST(InitializerListConstructor, SimpleTypeWithInlineBacking) {
>+  EXPECT_THAT((absl::InlinedVector<int, 4>{4, 5, 6}),
>+              AllOf(SizeIs(3), CapacityIs(4), ElementsAre(4, 5, 6)));
>+}
>+
>+TEST(InitializerListConstructor, SimpleTypeWithReallocationRequired) {
>+  EXPECT_THAT((absl::InlinedVector<int, 2>{4, 5, 6}),
>+              AllOf(SizeIs(3), CapacityIs(Gt(2)), ElementsAre(4, 5, 6)));
>+}
>+
>+TEST(InitializerListConstructor, DisparateTypesInList) {
>+  EXPECT_THAT((absl::InlinedVector<int, 2>{-7, 8ULL}), ElementsAre(-7, 8));
>+
>+  EXPECT_THAT((absl::InlinedVector<std::string, 2>{"foo", std::string("bar")}),
>+              ElementsAre("foo", "bar"));
>+}
>+
>+TEST(InitializerListConstructor, ComplexTypeWithInlineBacking) {
>+  EXPECT_THAT((absl::InlinedVector<CopyableMovableInstance, 1>{
>+                  CopyableMovableInstance(0)}),
>+              AllOf(SizeIs(1), CapacityIs(1), ElementsAre(ValueIs(0))));
>+}
>+
>+TEST(InitializerListConstructor, ComplexTypeWithReallocationRequired) {
>+  EXPECT_THAT(
>+      (absl::InlinedVector<CopyableMovableInstance, 1>{
>+          CopyableMovableInstance(0), CopyableMovableInstance(1)}),
>+      AllOf(SizeIs(2), CapacityIs(Gt(1)), ElementsAre(ValueIs(0), ValueIs(1))));
>+}
>+
>+TEST(InitializerListAssign, SimpleTypeFitsInlineBacking) {
>+  for (size_t original_size = 0; original_size <= 4; ++original_size) {
>+    SCOPED_TRACE(original_size);
>+
>+    absl::InlinedVector<int, 2> v1(original_size, 12345);
>+    const size_t original_capacity_v1 = v1.capacity();
>+    v1.assign({3});
>+    EXPECT_THAT(
>+        v1, AllOf(SizeIs(1), CapacityIs(original_capacity_v1), ElementsAre(3)));
>+
>+    absl::InlinedVector<int, 2> v2(original_size, 12345);
>+    const size_t original_capacity_v2 = v2.capacity();
>+    v2 = {3};
>+    EXPECT_THAT(
>+        v2, AllOf(SizeIs(1), CapacityIs(original_capacity_v2), ElementsAre(3)));
>+  }
>+}
>+
>+TEST(InitializerListAssign, SimpleTypeDoesNotFitInlineBacking) {
>+  for (size_t original_size = 0; original_size <= 4; ++original_size) {
>+    SCOPED_TRACE(original_size);
>+    absl::InlinedVector<int, 2> v1(original_size, 12345);
>+    v1.assign({3, 4, 5});
>+    EXPECT_THAT(v1, AllOf(SizeIs(3), ElementsAre(3, 4, 5)));
>+    EXPECT_LE(3, v1.capacity());
>+
>+    absl::InlinedVector<int, 2> v2(original_size, 12345);
>+    v2 = {3, 4, 5};
>+    EXPECT_THAT(v2, AllOf(SizeIs(3), ElementsAre(3, 4, 5)));
>+    EXPECT_LE(3, v2.capacity());
>+  }
>+}
>+
>+TEST(InitializerListAssign, DisparateTypesInList) {
>+  absl::InlinedVector<int, 2> v_int1;
>+  v_int1.assign({-7, 8ULL});
>+  EXPECT_THAT(v_int1, ElementsAre(-7, 8));
>+
>+  absl::InlinedVector<int, 2> v_int2;
>+  v_int2 = {-7, 8ULL};
>+  EXPECT_THAT(v_int2, ElementsAre(-7, 8));
>+
>+  absl::InlinedVector<std::string, 2> v_string1;
>+  v_string1.assign({"foo", std::string("bar")});
>+  EXPECT_THAT(v_string1, ElementsAre("foo", "bar"));
>+
>+  absl::InlinedVector<std::string, 2> v_string2;
>+  v_string2 = {"foo", std::string("bar")};
>+  EXPECT_THAT(v_string2, ElementsAre("foo", "bar"));
>+}
>+
>+TYPED_TEST_P(InstanceTest, InitializerListAssign) {
>+  using Instance = TypeParam;
>+  for (size_t original_size = 0; original_size <= 4; ++original_size) {
>+    SCOPED_TRACE(original_size);
>+    absl::InlinedVector<Instance, 2> v(original_size, Instance(12345));
>+    const size_t original_capacity = v.capacity();
>+    v.assign({Instance(3)});
>+    EXPECT_THAT(v, AllOf(SizeIs(1), CapacityIs(original_capacity),
>+                         ElementsAre(ValueIs(3))));
>+  }
>+  for (size_t original_size = 0; original_size <= 4; ++original_size) {
>+    SCOPED_TRACE(original_size);
>+    absl::InlinedVector<Instance, 2> v(original_size, Instance(12345));
>+    v.assign({Instance(3), Instance(4), Instance(5)});
>+    EXPECT_THAT(v, AllOf(SizeIs(3),
>+                         ElementsAre(ValueIs(3), ValueIs(4), ValueIs(5))));
>+    EXPECT_LE(3, v.capacity());
>+  }
>+}
>+
>+REGISTER_TYPED_TEST_CASE_P(InstanceTest, Swap, CountConstructorsDestructors,
>+                           CountConstructorsDestructorsOnCopyConstruction,
>+                           CountConstructorsDestructorsOnMoveConstruction,
>+                           CountConstructorsDestructorsOnAssignment,
>+                           CountConstructorsDestructorsOnMoveAssignment,
>+                           CountElemAssignInlineBacking, RangedConstructor,
>+                           RangedAssign, InitializerListAssign);
>+
>+using InstanceTypes =
>+    ::testing::Types<CopyableOnlyInstance, CopyableMovableInstance>;
>+INSTANTIATE_TYPED_TEST_CASE_P(InstanceTestOnTypes, InstanceTest, InstanceTypes);
>+
>+TEST(DynamicVec, DynamicVecCompiles) {
>+  DynamicVec v;
>+  (void)v;
>+}
>+
>+TEST(AllocatorSupportTest, Constructors) {
>+  using MyAlloc = CountingAllocator<int>;
>+  using AllocVec = absl::InlinedVector<int, 4, MyAlloc>;
>+  const int ia[] = { 0, 1, 2, 3, 4, 5, 6, 7 };
>+  int64_t allocated = 0;
>+  MyAlloc alloc(&allocated);
>+  { AllocVec ABSL_ATTRIBUTE_UNUSED v; }
>+  { AllocVec ABSL_ATTRIBUTE_UNUSED v(alloc); }
>+  { AllocVec ABSL_ATTRIBUTE_UNUSED v(ia, ia + ABSL_ARRAYSIZE(ia), alloc); }
>+  { AllocVec ABSL_ATTRIBUTE_UNUSED v({1, 2, 3}, alloc); }
>+
>+  AllocVec v2;
>+  { AllocVec ABSL_ATTRIBUTE_UNUSED v(v2, alloc); }
>+  { AllocVec ABSL_ATTRIBUTE_UNUSED v(std::move(v2), alloc); }
>+}
>+
>+TEST(AllocatorSupportTest, CountAllocations) {
>+  using MyAlloc = CountingAllocator<int>;
>+  using AllocVec = absl::InlinedVector<int, 4, MyAlloc>;
>+  const int ia[] = { 0, 1, 2, 3, 4, 5, 6, 7 };
>+  int64_t allocated = 0;
>+  MyAlloc alloc(&allocated);
>+  {
>+    AllocVec ABSL_ATTRIBUTE_UNUSED v(ia, ia + 4, alloc);
>+    EXPECT_THAT(allocated, 0);
>+  }
>+  EXPECT_THAT(allocated, 0);
>+  {
>+    AllocVec ABSL_ATTRIBUTE_UNUSED v(ia, ia + ABSL_ARRAYSIZE(ia), alloc);
>+    EXPECT_THAT(allocated, v.size() * sizeof(int));
>+  }
>+  EXPECT_THAT(allocated, 0);
>+  {
>+    AllocVec v(4, 1, alloc);
>+    EXPECT_THAT(allocated, 0);
>+
>+    int64_t allocated2 = 0;
>+    MyAlloc alloc2(&allocated2);
>+    AllocVec v2(v, alloc2);
>+    EXPECT_THAT(allocated2, 0);
>+
>+    int64_t allocated3 = 0;
>+    MyAlloc alloc3(&allocated3);
>+    AllocVec v3(std::move(v), alloc3);
>+    EXPECT_THAT(allocated3, 0);
>+  }
>+  EXPECT_THAT(allocated, 0);
>+  {
>+    AllocVec v(8, 2, alloc);
>+    EXPECT_THAT(allocated, v.size() * sizeof(int));
>+
>+    int64_t allocated2 = 0;
>+    MyAlloc alloc2(&allocated2);
>+    AllocVec v2(v, alloc2);
>+    EXPECT_THAT(allocated2, v2.size() * sizeof(int));
>+
>+    int64_t allocated3 = 0;
>+    MyAlloc alloc3(&allocated3);
>+    AllocVec v3(std::move(v), alloc3);
>+    EXPECT_THAT(allocated3, v3.size() * sizeof(int));
>+  }
>+  EXPECT_EQ(allocated, 0);
>+  {
>+    // Test shrink_to_fit deallocations.
>+    AllocVec v(8, 2, alloc);
>+    EXPECT_EQ(allocated, 8 * sizeof(int));
>+    v.resize(5);
>+    EXPECT_EQ(allocated, 8 * sizeof(int));
>+    v.shrink_to_fit();
>+    EXPECT_EQ(allocated, 5 * sizeof(int));
>+    v.resize(4);
>+    EXPECT_EQ(allocated, 5 * sizeof(int));
>+    v.shrink_to_fit();
>+    EXPECT_EQ(allocated, 0);
>+  }
>+}
>+
>+TEST(AllocatorSupportTest, SwapBothAllocated) {
>+  using MyAlloc = CountingAllocator<int>;
>+  using AllocVec = absl::InlinedVector<int, 4, MyAlloc>;
>+  int64_t allocated1 = 0;
>+  int64_t allocated2 = 0;
>+  {
>+    const int ia1[] = { 0, 1, 2, 3, 4, 5, 6, 7 };
>+    const int ia2[] = { 0, 1, 2, 3, 4, 5, 6, 7, 8 };
>+    MyAlloc a1(&allocated1);
>+    MyAlloc a2(&allocated2);
>+    AllocVec v1(ia1, ia1 + ABSL_ARRAYSIZE(ia1), a1);
>+    AllocVec v2(ia2, ia2 + ABSL_ARRAYSIZE(ia2), a2);
>+    EXPECT_LT(v1.capacity(), v2.capacity());
>+    EXPECT_THAT(allocated1, v1.capacity() * sizeof(int));
>+    EXPECT_THAT(allocated2, v2.capacity() * sizeof(int));
>+    v1.swap(v2);
>+    EXPECT_THAT(v1, ElementsAreArray(ia2));
>+    EXPECT_THAT(v2, ElementsAreArray(ia1));
>+    EXPECT_THAT(allocated1, v2.capacity() * sizeof(int));
>+    EXPECT_THAT(allocated2, v1.capacity() * sizeof(int));
>+  }
>+  EXPECT_THAT(allocated1, 0);
>+  EXPECT_THAT(allocated2, 0);
>+}
>+
>+TEST(AllocatorSupportTest, SwapOneAllocated) {
>+  using MyAlloc = CountingAllocator<int>;
>+  using AllocVec = absl::InlinedVector<int, 4, MyAlloc>;
>+  int64_t allocated1 = 0;
>+  int64_t allocated2 = 0;
>+  {
>+    const int ia1[] = { 0, 1, 2, 3, 4, 5, 6, 7 };
>+    const int ia2[] = { 0, 1, 2, 3 };
>+    MyAlloc a1(&allocated1);
>+    MyAlloc a2(&allocated2);
>+    AllocVec v1(ia1, ia1 + ABSL_ARRAYSIZE(ia1), a1);
>+    AllocVec v2(ia2, ia2 + ABSL_ARRAYSIZE(ia2), a2);
>+    EXPECT_THAT(allocated1, v1.capacity() * sizeof(int));
>+    EXPECT_THAT(allocated2, 0);
>+    v1.swap(v2);
>+    EXPECT_THAT(v1, ElementsAreArray(ia2));
>+    EXPECT_THAT(v2, ElementsAreArray(ia1));
>+    EXPECT_THAT(allocated1, v2.capacity() * sizeof(int));
>+    EXPECT_THAT(allocated2, 0);
>+    EXPECT_TRUE(v2.get_allocator() == a1);
>+    EXPECT_TRUE(v1.get_allocator() == a2);
>+  }
>+  EXPECT_THAT(allocated1, 0);
>+  EXPECT_THAT(allocated2, 0);
>+}
>+
>+TEST(AllocatorSupportTest, ScopedAllocatorWorks) {
>+  using StdVector = std::vector<int, CountingAllocator<int>>;
>+  using MyAlloc =
>+      std::scoped_allocator_adaptor<CountingAllocator<StdVector>>;
>+  using AllocVec = absl::InlinedVector<StdVector, 4, MyAlloc>;
>+
>+  int64_t allocated = 0;
>+  AllocVec vec(MyAlloc{CountingAllocator<StdVector>{&allocated}});
>+  EXPECT_EQ(allocated, 0);
>+
>+  // This default constructs a vector<int>, but the allocator should pass itself
>+  // into the vector<int>.
>+  // The absl::InlinedVector does not allocate any memory.
>+  // The vector<int> does not allocate any memory.
>+  vec.resize(1);
>+  EXPECT_EQ(allocated, 0);
>+
>+  // We make vector<int> allocate memory.
>+  // It must go through the allocator even though we didn't construct the
>+  // vector directly.
>+  vec[0].push_back(1);
>+  EXPECT_EQ(allocated, sizeof(int) * 1);
>+
>+  // Another allocating vector.
>+  vec.push_back(vec[0]);
>+  EXPECT_EQ(allocated, sizeof(int) * 2);
>+
>+  // Overflow the inlined memory.
>+  // The absl::InlinedVector will now allocate.
>+  vec.resize(5);
>+  EXPECT_EQ(allocated, sizeof(int) * 2 + sizeof(StdVector) * 8);
>+
>+  // Adding one more in external mode should also work.
>+  vec.push_back(vec[0]);
>+  EXPECT_EQ(allocated, sizeof(int) * 3 + sizeof(StdVector) * 8);
>+
>+  // And extending these should still work.
>+  vec[0].push_back(1);
>+  EXPECT_EQ(allocated, sizeof(int) * 4 + sizeof(StdVector) * 8);
>+
>+  vec.clear();
>+  EXPECT_EQ(allocated, 0);
>+}
>+
>+TEST(AllocatorSupportTest, SizeAllocConstructor) {
>+  constexpr int inlined_size = 4;
>+  using Alloc = CountingAllocator<int>;
>+  using AllocVec = absl::InlinedVector<int, inlined_size, Alloc>;
>+
>+  {
>+    auto len = inlined_size / 2;
>+    int64_t allocated = 0;
>+    auto v = AllocVec(len, Alloc(&allocated));
>+
>+    // Inline storage used; allocator should not be invoked
>+    EXPECT_THAT(allocated, 0);
>+    EXPECT_THAT(v, AllOf(SizeIs(len), Each(0)));
>+  }
>+
>+  {
>+    auto len = inlined_size * 2;
>+    int64_t allocated = 0;
>+    auto v = AllocVec(len, Alloc(&allocated));
>+
>+    // Out of line storage used; allocation of 8 elements expected
>+    EXPECT_THAT(allocated, len * sizeof(int));
>+    EXPECT_THAT(v, AllOf(SizeIs(len), Each(0)));
>+  }
>+}
>+}  // anonymous namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/compressed_tuple.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/compressed_tuple.h
>new file mode 100644
>index 00000000000..cc52614f5b3
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/compressed_tuple.h
>@@ -0,0 +1,175 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// Helper class to perform the Empty Base Optimization.
>+// Ts can contain classes and non-classes, empty or not. For the ones that
>+// are empty classes, we perform the optimization. If all types in Ts are empty
>+// classes, then CompressedTuple<Ts...> is itself an empty class.
>+//
>+// To access the members, use member get<N>() function.
>+//
>+// Eg:
>+//   absl::container_internal::CompressedTuple<int, T1, T2, T3> value(7, t1, t2,
>+//                                                                    t3);
>+//   assert(value.get<0>() == 7);
>+//   T1& t1 = value.get<1>();
>+//   const T2& t2 = value.get<2>();
>+//   ...
>+//
>+// http://en.cppreference.com/w/cpp/language/ebo
>+
>+#ifndef ABSL_CONTAINER_INTERNAL_COMPRESSED_TUPLE_H_
>+#define ABSL_CONTAINER_INTERNAL_COMPRESSED_TUPLE_H_
>+
>+#include <tuple>
>+#include <type_traits>
>+#include <utility>
>+
>+#include "absl/utility/utility.h"
>+
>+#ifdef _MSC_VER
>+// We need to mark these classes with this declspec to ensure that
>+// CompressedTuple happens.
>+#define ABSL_INTERNAL_COMPRESSED_TUPLE_DECLSPEC __declspec(empty_bases)
>+#else  // _MSC_VER
>+#define ABSL_INTERNAL_COMPRESSED_TUPLE_DECLSPEC
>+#endif  // _MSC_VER
>+
>+namespace absl {
>+namespace container_internal {
>+
>+template <typename... Ts>
>+class CompressedTuple;
>+
>+namespace internal_compressed_tuple {
>+
>+template <typename D, size_t I>
>+struct Elem;
>+template <typename... B, size_t I>
>+struct Elem<CompressedTuple<B...>, I>
>+    : std::tuple_element<I, std::tuple<B...>> {};
>+template <typename D, size_t I>
>+using ElemT = typename Elem<D, I>::type;
>+
>+// Use the __is_final intrinsic if available. Where it's not available, classes
>+// declared with the 'final' specifier cannot be used as CompressedTuple
>+// elements.
>+// TODO(sbenza): Replace this with std::is_final in C++14.
>+template <typename T>
>+constexpr bool IsFinal() {
>+#if defined(__clang__) || defined(__GNUC__)
>+  return __is_final(T);
>+#else
>+  return false;
>+#endif
>+}
>+
>+template <typename T>
>+constexpr bool ShouldUseBase() {
>+  return std::is_class<T>::value && std::is_empty<T>::value && !IsFinal<T>();
>+}
>+
>+// The storage class provides two specializations:
>+//  - For empty classes, it stores T as a base class.
>+//  - For everything else, it stores T as a member.
>+template <typename D, size_t I, bool = ShouldUseBase<ElemT<D, I>>()>
>+struct Storage {
>+  using T = ElemT<D, I>;
>+  T value;
>+  constexpr Storage() = default;
>+  explicit constexpr Storage(T&& v) : value(absl::forward<T>(v)) {}
>+  constexpr const T& get() const { return value; }
>+  T& get() { return value; }
>+};
>+
>+template <typename D, size_t I>
>+struct ABSL_INTERNAL_COMPRESSED_TUPLE_DECLSPEC Storage<D, I, true>
>+    : ElemT<D, I> {
>+  using T = internal_compressed_tuple::ElemT<D, I>;
>+  constexpr Storage() = default;
>+  explicit constexpr Storage(T&& v) : T(absl::forward<T>(v)) {}
>+  constexpr const T& get() const { return *this; }
>+  T& get() { return *this; }
>+};
>+
>+template <typename D, typename I>
>+struct ABSL_INTERNAL_COMPRESSED_TUPLE_DECLSPEC CompressedTupleImpl;
>+
>+template <typename... Ts, size_t... I>
>+struct ABSL_INTERNAL_COMPRESSED_TUPLE_DECLSPEC
>+    CompressedTupleImpl<CompressedTuple<Ts...>, absl::index_sequence<I...>>
>+    // We use the dummy identity function through std::integral_constant to
>+    // convince MSVC of accepting and expanding I in that context. Without it
>+    // you would get:
>+    //   error C3548: 'I': parameter pack cannot be used in this context
>+    : Storage<CompressedTuple<Ts...>,
>+              std::integral_constant<size_t, I>::value>... {
>+  constexpr CompressedTupleImpl() = default;
>+  explicit constexpr CompressedTupleImpl(Ts&&... args)
>+      : Storage<CompressedTuple<Ts...>, I>(absl::forward<Ts>(args))... {}
>+};
>+
>+}  // namespace internal_compressed_tuple
>+
>+// Helper class to perform the Empty Base Class Optimization.
>+// Ts can contain classes and non-classes, empty or not. For the ones that
>+// are empty classes, we perform the CompressedTuple. If all types in Ts are
>+// empty classes, then CompressedTuple<Ts...> is itself an empty class.
>+//
>+// To access the members, use member .get<N>() function.
>+//
>+// Eg:
>+//   absl::container_internal::CompressedTuple<int, T1, T2, T3> value(7, t1, t2,
>+//                                                                    t3);
>+//   assert(value.get<0>() == 7);
>+//   T1& t1 = value.get<1>();
>+//   const T2& t2 = value.get<2>();
>+//   ...
>+//
>+// http://en.cppreference.com/w/cpp/language/ebo
>+template <typename... Ts>
>+class ABSL_INTERNAL_COMPRESSED_TUPLE_DECLSPEC CompressedTuple
>+    : private internal_compressed_tuple::CompressedTupleImpl<
>+          CompressedTuple<Ts...>, absl::index_sequence_for<Ts...>> {
>+ private:
>+  template <int I>
>+  using ElemT = internal_compressed_tuple::ElemT<CompressedTuple, I>;
>+
>+ public:
>+  constexpr CompressedTuple() = default;
>+  explicit constexpr CompressedTuple(Ts... base)
>+      : CompressedTuple::CompressedTupleImpl(absl::forward<Ts>(base)...) {}
>+
>+  template <int I>
>+  ElemT<I>& get() {
>+    return internal_compressed_tuple::Storage<CompressedTuple, I>::get();
>+  }
>+
>+  template <int I>
>+  constexpr const ElemT<I>& get() const {
>+    return internal_compressed_tuple::Storage<CompressedTuple, I>::get();
>+  }
>+};
>+
>+// Explicit specialization for a zero-element tuple
>+// (needed to avoid ambiguous overloads for the default constructor).
>+template <>
>+class ABSL_INTERNAL_COMPRESSED_TUPLE_DECLSPEC CompressedTuple<> {};
>+
>+}  // namespace container_internal
>+}  // namespace absl
>+
>+#undef ABSL_INTERNAL_COMPRESSED_TUPLE_DECLSPEC
>+
>+#endif  // ABSL_CONTAINER_INTERNAL_COMPRESSED_TUPLE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/compressed_tuple_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/compressed_tuple_test.cc
>new file mode 100644
>index 00000000000..45030c675ee
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/compressed_tuple_test.cc
>@@ -0,0 +1,166 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/container/internal/compressed_tuple.h"
>+
>+#include <string>
>+
>+#include "gmock/gmock.h"
>+#include "gtest/gtest.h"
>+
>+namespace absl {
>+namespace container_internal {
>+namespace {
>+
>+template <int>
>+struct Empty {};
>+
>+template <typename T>
>+struct NotEmpty {
>+  T value;
>+};
>+
>+template <typename T, typename U>
>+struct TwoValues {
>+  T value1;
>+  U value2;
>+};
>+
>+TEST(CompressedTupleTest, Sizeof) {
>+  EXPECT_EQ(sizeof(int), sizeof(CompressedTuple<int>));
>+  EXPECT_EQ(sizeof(int), sizeof(CompressedTuple<int, Empty<0>>));
>+  EXPECT_EQ(sizeof(int), sizeof(CompressedTuple<int, Empty<0>, Empty<1>>));
>+  EXPECT_EQ(sizeof(int),
>+            sizeof(CompressedTuple<int, Empty<0>, Empty<1>, Empty<2>>));
>+
>+  EXPECT_EQ(sizeof(TwoValues<int, double>),
>+            sizeof(CompressedTuple<int, NotEmpty<double>>));
>+  EXPECT_EQ(sizeof(TwoValues<int, double>),
>+            sizeof(CompressedTuple<int, Empty<0>, NotEmpty<double>>));
>+  EXPECT_EQ(sizeof(TwoValues<int, double>),
>+            sizeof(CompressedTuple<int, Empty<0>, NotEmpty<double>, Empty<1>>));
>+}
>+
>+TEST(CompressedTupleTest, Access) {
>+  struct S {
>+    std::string x;
>+  };
>+  CompressedTuple<int, Empty<0>, S> x(7, {}, S{"ABC"});
>+  EXPECT_EQ(sizeof(x), sizeof(TwoValues<int, S>));
>+  EXPECT_EQ(7, x.get<0>());
>+  EXPECT_EQ("ABC", x.get<2>().x);
>+}
>+
>+TEST(CompressedTupleTest, NonClasses) {
>+  CompressedTuple<int, const char*> x(7, "ABC");
>+  EXPECT_EQ(7, x.get<0>());
>+  EXPECT_STREQ("ABC", x.get<1>());
>+}
>+
>+TEST(CompressedTupleTest, MixClassAndNonClass) {
>+  CompressedTuple<int, const char*, Empty<0>, NotEmpty<double>> x(7, "ABC", {},
>+                                                                  {1.25});
>+  struct Mock {
>+    int v;
>+    const char* p;
>+    double d;
>+  };
>+  EXPECT_EQ(sizeof(x), sizeof(Mock));
>+  EXPECT_EQ(7, x.get<0>());
>+  EXPECT_STREQ("ABC", x.get<1>());
>+  EXPECT_EQ(1.25, x.get<3>().value);
>+}
>+
>+TEST(CompressedTupleTest, Nested) {
>+  CompressedTuple<int, CompressedTuple<int>,
>+                  CompressedTuple<int, CompressedTuple<int>>>
>+      x(1, CompressedTuple<int>(2),
>+        CompressedTuple<int, CompressedTuple<int>>(3, CompressedTuple<int>(4)));
>+  EXPECT_EQ(1, x.get<0>());
>+  EXPECT_EQ(2, x.get<1>().get<0>());
>+  EXPECT_EQ(3, x.get<2>().get<0>());
>+  EXPECT_EQ(4, x.get<2>().get<1>().get<0>());
>+
>+  CompressedTuple<Empty<0>, Empty<0>,
>+                  CompressedTuple<Empty<0>, CompressedTuple<Empty<0>>>>
>+      y;
>+  std::set<Empty<0>*> empties{&y.get<0>(), &y.get<1>(), &y.get<2>().get<0>(),
>+                              &y.get<2>().get<1>().get<0>()};
>+#ifdef _MSC_VER
>+  // MSVC has a bug where many instances of the same base class are layed out in
>+  // the same address when using __declspec(empty_bases).
>+  // This will be fixed in a future version of MSVC.
>+  int expected = 1;
>+#else
>+  int expected = 4;
>+#endif
>+  EXPECT_EQ(expected, sizeof(y));
>+  EXPECT_EQ(expected, empties.size());
>+  EXPECT_EQ(sizeof(y), sizeof(Empty<0>) * empties.size());
>+
>+  EXPECT_EQ(4 * sizeof(char),
>+            sizeof(CompressedTuple<CompressedTuple<char, char>,
>+                                   CompressedTuple<char, char>>));
>+  EXPECT_TRUE(
>+      (std::is_empty<CompressedTuple<CompressedTuple<Empty<0>>,
>+                                     CompressedTuple<Empty<1>>>>::value));
>+}
>+
>+TEST(CompressedTupleTest, Reference) {
>+  int i = 7;
>+  std::string s = "Very long std::string that goes in the heap";
>+  CompressedTuple<int, int&, std::string, std::string&> x(i, i, s, s);
>+
>+  // Sanity check. We should have not moved from `s`
>+  EXPECT_EQ(s, "Very long std::string that goes in the heap");
>+
>+  EXPECT_EQ(x.get<0>(), x.get<1>());
>+  EXPECT_NE(&x.get<0>(), &x.get<1>());
>+  EXPECT_EQ(&x.get<1>(), &i);
>+
>+  EXPECT_EQ(x.get<2>(), x.get<3>());
>+  EXPECT_NE(&x.get<2>(), &x.get<3>());
>+  EXPECT_EQ(&x.get<3>(), &s);
>+}
>+
>+TEST(CompressedTupleTest, NoElements) {
>+  CompressedTuple<> x;
>+  static_cast<void>(x);  // Silence -Wunused-variable.
>+  EXPECT_TRUE(std::is_empty<CompressedTuple<>>::value);
>+}
>+
>+TEST(CompressedTupleTest, Constexpr) {
>+  constexpr CompressedTuple<int, double, CompressedTuple<int>> x(
>+      7, 1.25, CompressedTuple<int>(5));
>+  constexpr int x0 = x.get<0>();
>+  constexpr double x1 = x.get<1>();
>+  constexpr int x2 = x.get<2>().get<0>();
>+  EXPECT_EQ(x0, 7);
>+  EXPECT_EQ(x1, 1.25);
>+  EXPECT_EQ(x2, 5);
>+}
>+
>+#if defined(__clang__) || defined(__GNUC__)
>+TEST(CompressedTupleTest, EmptyFinalClass) {
>+  struct S final {
>+    int f() const { return 5; }
>+  };
>+  CompressedTuple<S> x;
>+  EXPECT_EQ(x.get<0>().f(), 5);
>+}
>+#endif
>+
>+}  // namespace
>+}  // namespace container_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/test_instance_tracker.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/test_instance_tracker.cc
>new file mode 100644
>index 00000000000..fe00aca8fb9
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/test_instance_tracker.cc
>@@ -0,0 +1,26 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/container/internal/test_instance_tracker.h"
>+
>+namespace absl {
>+namespace test_internal {
>+int BaseCountedInstance::num_instances_ = 0;
>+int BaseCountedInstance::num_live_instances_ = 0;
>+int BaseCountedInstance::num_moves_ = 0;
>+int BaseCountedInstance::num_copies_ = 0;
>+int BaseCountedInstance::num_swaps_ = 0;
>+
>+}  // namespace test_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/test_instance_tracker.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/test_instance_tracker.h
>new file mode 100644
>index 00000000000..cf8f3a531e6
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/test_instance_tracker.h
>@@ -0,0 +1,220 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#ifndef ABSL_CONTAINER_INTERNAL_TEST_INSTANCE_TRACKER_H_
>+#define ABSL_CONTAINER_INTERNAL_TEST_INSTANCE_TRACKER_H_
>+
>+#include <cstdlib>
>+#include <ostream>
>+
>+namespace absl {
>+namespace test_internal {
>+
>+// A type that counts number of occurences of the type, the live occurrences of
>+// the type, as well as the number of copies, moves, and swaps that have
>+// occurred on the type. This is used as a base class for the copyable,
>+// copyable+movable, and movable types below that are used in actual tests. Use
>+// InstanceTracker in tests to track the number of instances.
>+class BaseCountedInstance {
>+ public:
>+  explicit BaseCountedInstance(int x) : value_(x) {
>+    ++num_instances_;
>+    ++num_live_instances_;
>+  }
>+  BaseCountedInstance(const BaseCountedInstance& x)
>+      : value_(x.value_), is_live_(x.is_live_) {
>+    ++num_instances_;
>+    if (is_live_) ++num_live_instances_;
>+    ++num_copies_;
>+  }
>+  BaseCountedInstance(BaseCountedInstance&& x)
>+      : value_(x.value_), is_live_(x.is_live_) {
>+    x.is_live_ = false;
>+    ++num_instances_;
>+    ++num_moves_;
>+  }
>+  ~BaseCountedInstance() {
>+    --num_instances_;
>+    if (is_live_) --num_live_instances_;
>+  }
>+
>+  BaseCountedInstance& operator=(const BaseCountedInstance& x) {
>+    value_ = x.value_;
>+    if (is_live_) --num_live_instances_;
>+    is_live_ = x.is_live_;
>+    if (is_live_) ++num_live_instances_;
>+    ++num_copies_;
>+    return *this;
>+  }
>+  BaseCountedInstance& operator=(BaseCountedInstance&& x) {
>+    value_ = x.value_;
>+    if (is_live_) --num_live_instances_;
>+    is_live_ = x.is_live_;
>+    x.is_live_ = false;
>+    ++num_moves_;
>+    return *this;
>+  }
>+
>+  int value() const {
>+    if (!is_live_) std::abort();
>+    return value_;
>+  }
>+
>+  friend std::ostream& operator<<(std::ostream& o,
>+                                  const BaseCountedInstance& v) {
>+    return o << "[value:" << v.value() << "]";
>+  }
>+
>+  // Implementation of efficient swap() that counts swaps.
>+  static void SwapImpl(
>+      BaseCountedInstance& lhs,    // NOLINT(runtime/references)
>+      BaseCountedInstance& rhs) {  // NOLINT(runtime/references)
>+    using std::swap;
>+    swap(lhs.value_, rhs.value_);
>+    swap(lhs.is_live_, rhs.is_live_);
>+    ++BaseCountedInstance::num_swaps_;
>+  }
>+
>+ private:
>+  friend class InstanceTracker;
>+
>+  int value_;
>+
>+  // Indicates if the value is live, ie it hasn't been moved away from.
>+  bool is_live_ = true;
>+
>+  // Number of instances.
>+  static int num_instances_;
>+
>+  // Number of live instances (those that have not been moved away from.)
>+  static int num_live_instances_;
>+
>+  // Number of times that BaseCountedInstance objects were moved.
>+  static int num_moves_;
>+
>+  // Number of times that BaseCountedInstance objects were copied.
>+  static int num_copies_;
>+
>+  // Number of times that BaseCountedInstance objects were swapped.
>+  static int num_swaps_;
>+};
>+
>+// Helper to track the BaseCountedInstance instance counters. Expects that the
>+// number of instances and live_instances are the same when it is constructed
>+// and when it is destructed.
>+class InstanceTracker {
>+ public:
>+  InstanceTracker()
>+      : start_instances_(BaseCountedInstance::num_instances_),
>+        start_live_instances_(BaseCountedInstance::num_live_instances_) {
>+    ResetCopiesMovesSwaps();
>+  }
>+  ~InstanceTracker() {
>+    if (instances() != 0) std::abort();
>+    if (live_instances() != 0) std::abort();
>+  }
>+
>+  // Returns the number of BaseCountedInstance instances both containing valid
>+  // values and those moved away from compared to when the InstanceTracker was
>+  // constructed
>+  int instances() const {
>+    return BaseCountedInstance::num_instances_ - start_instances_;
>+  }
>+
>+  // Returns the number of live BaseCountedInstance instances compared to when
>+  // the InstanceTracker was constructed
>+  int live_instances() const {
>+    return BaseCountedInstance::num_live_instances_ - start_live_instances_;
>+  }
>+
>+  // Returns the number of moves on BaseCountedInstance objects since
>+  // construction or since the last call to ResetCopiesMovesSwaps().
>+  int moves() const { return BaseCountedInstance::num_moves_ - start_moves_; }
>+
>+  // Returns the number of copies on BaseCountedInstance objects since
>+  // construction or the last call to ResetCopiesMovesSwaps().
>+  int copies() const {
>+    return BaseCountedInstance::num_copies_ - start_copies_;
>+  }
>+
>+  // Returns the number of swaps on BaseCountedInstance objects since
>+  // construction or the last call to ResetCopiesMovesSwaps().
>+  int swaps() const { return BaseCountedInstance::num_swaps_ - start_swaps_; }
>+
>+  // Resets the base values for moves, copies and swaps to the current values,
>+  // so that subsequent Get*() calls for moves, copies and swaps will compare to
>+  // the situation at the point of this call.
>+  void ResetCopiesMovesSwaps() {
>+    start_moves_ = BaseCountedInstance::num_moves_;
>+    start_copies_ = BaseCountedInstance::num_copies_;
>+    start_swaps_ = BaseCountedInstance::num_swaps_;
>+  }
>+
>+ private:
>+  int start_instances_;
>+  int start_live_instances_;
>+  int start_moves_;
>+  int start_copies_;
>+  int start_swaps_;
>+};
>+
>+// Copyable, not movable.
>+class CopyableOnlyInstance : public BaseCountedInstance {
>+ public:
>+  explicit CopyableOnlyInstance(int x) : BaseCountedInstance(x) {}
>+  CopyableOnlyInstance(const CopyableOnlyInstance& rhs) = default;
>+  CopyableOnlyInstance& operator=(const CopyableOnlyInstance& rhs) = default;
>+
>+  friend void swap(CopyableOnlyInstance& lhs, CopyableOnlyInstance& rhs) {
>+    BaseCountedInstance::SwapImpl(lhs, rhs);
>+  }
>+
>+  static bool supports_move() { return false; }
>+};
>+
>+// Copyable and movable.
>+class CopyableMovableInstance : public BaseCountedInstance {
>+ public:
>+  explicit CopyableMovableInstance(int x) : BaseCountedInstance(x) {}
>+  CopyableMovableInstance(const CopyableMovableInstance& rhs) = default;
>+  CopyableMovableInstance(CopyableMovableInstance&& rhs) = default;
>+  CopyableMovableInstance& operator=(const CopyableMovableInstance& rhs) =
>+      default;
>+  CopyableMovableInstance& operator=(CopyableMovableInstance&& rhs) = default;
>+
>+  friend void swap(CopyableMovableInstance& lhs, CopyableMovableInstance& rhs) {
>+    BaseCountedInstance::SwapImpl(lhs, rhs);
>+  }
>+
>+  static bool supports_move() { return true; }
>+};
>+
>+// Only movable, not default-constructible.
>+class MovableOnlyInstance : public BaseCountedInstance {
>+ public:
>+  explicit MovableOnlyInstance(int x) : BaseCountedInstance(x) {}
>+  MovableOnlyInstance(MovableOnlyInstance&& other) = default;
>+  MovableOnlyInstance& operator=(MovableOnlyInstance&& other) = default;
>+
>+  friend void swap(MovableOnlyInstance& lhs, MovableOnlyInstance& rhs) {
>+    BaseCountedInstance::SwapImpl(lhs, rhs);
>+  }
>+
>+  static bool supports_move() { return true; }
>+};
>+
>+}  // namespace test_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_CONTAINER_INTERNAL_TEST_INSTANCE_TRACKER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/test_instance_tracker_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/test_instance_tracker_test.cc
>new file mode 100644
>index 00000000000..9efb6771cf0
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/container/internal/test_instance_tracker_test.cc
>@@ -0,0 +1,160 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/container/internal/test_instance_tracker.h"
>+
>+#include "gtest/gtest.h"
>+
>+namespace {
>+
>+using absl::test_internal::CopyableMovableInstance;
>+using absl::test_internal::CopyableOnlyInstance;
>+using absl::test_internal::InstanceTracker;
>+using absl::test_internal::MovableOnlyInstance;
>+
>+TEST(TestInstanceTracker, CopyableMovable) {
>+  InstanceTracker tracker;
>+  CopyableMovableInstance src(1);
>+  EXPECT_EQ(1, src.value()) << src;
>+  CopyableMovableInstance copy(src);
>+  CopyableMovableInstance move(std::move(src));
>+  EXPECT_EQ(1, tracker.copies());
>+  EXPECT_EQ(1, tracker.moves());
>+  EXPECT_EQ(0, tracker.swaps());
>+  EXPECT_EQ(3, tracker.instances());
>+  EXPECT_EQ(2, tracker.live_instances());
>+  tracker.ResetCopiesMovesSwaps();
>+
>+  CopyableMovableInstance copy_assign(1);
>+  copy_assign = copy;
>+  CopyableMovableInstance move_assign(1);
>+  move_assign = std::move(move);
>+  EXPECT_EQ(1, tracker.copies());
>+  EXPECT_EQ(1, tracker.moves());
>+  EXPECT_EQ(0, tracker.swaps());
>+  EXPECT_EQ(5, tracker.instances());
>+  EXPECT_EQ(3, tracker.live_instances());
>+  tracker.ResetCopiesMovesSwaps();
>+
>+  {
>+    using std::swap;
>+    swap(move_assign, copy);
>+    swap(copy, move_assign);
>+    EXPECT_EQ(2, tracker.swaps());
>+    EXPECT_EQ(0, tracker.copies());
>+    EXPECT_EQ(0, tracker.moves());
>+    EXPECT_EQ(5, tracker.instances());
>+    EXPECT_EQ(3, tracker.live_instances());
>+  }
>+}
>+
>+TEST(TestInstanceTracker, CopyableOnly) {
>+  InstanceTracker tracker;
>+  CopyableOnlyInstance src(1);
>+  EXPECT_EQ(1, src.value()) << src;
>+  CopyableOnlyInstance copy(src);
>+  CopyableOnlyInstance copy2(std::move(src));  // NOLINT
>+  EXPECT_EQ(2, tracker.copies());
>+  EXPECT_EQ(0, tracker.moves());
>+  EXPECT_EQ(3, tracker.instances());
>+  EXPECT_EQ(3, tracker.live_instances());
>+  tracker.ResetCopiesMovesSwaps();
>+
>+  CopyableOnlyInstance copy_assign(1);
>+  copy_assign = copy;
>+  CopyableOnlyInstance copy_assign2(1);
>+  copy_assign2 = std::move(copy2);  // NOLINT
>+  EXPECT_EQ(2, tracker.copies());
>+  EXPECT_EQ(0, tracker.moves());
>+  EXPECT_EQ(5, tracker.instances());
>+  EXPECT_EQ(5, tracker.live_instances());
>+  tracker.ResetCopiesMovesSwaps();
>+
>+  {
>+    using std::swap;
>+    swap(src, copy);
>+    swap(copy, src);
>+    EXPECT_EQ(2, tracker.swaps());
>+    EXPECT_EQ(0, tracker.copies());
>+    EXPECT_EQ(0, tracker.moves());
>+    EXPECT_EQ(5, tracker.instances());
>+    EXPECT_EQ(5, tracker.live_instances());
>+  }
>+}
>+
>+TEST(TestInstanceTracker, MovableOnly) {
>+  InstanceTracker tracker;
>+  MovableOnlyInstance src(1);
>+  EXPECT_EQ(1, src.value()) << src;
>+  MovableOnlyInstance move(std::move(src));
>+  MovableOnlyInstance move_assign(2);
>+  move_assign = std::move(move);
>+  EXPECT_EQ(3, tracker.instances());
>+  EXPECT_EQ(1, tracker.live_instances());
>+  EXPECT_EQ(2, tracker.moves());
>+  EXPECT_EQ(0, tracker.copies());
>+  tracker.ResetCopiesMovesSwaps();
>+
>+  {
>+    using std::swap;
>+    MovableOnlyInstance other(2);
>+    swap(move_assign, other);
>+    swap(other, move_assign);
>+    EXPECT_EQ(2, tracker.swaps());
>+    EXPECT_EQ(0, tracker.copies());
>+    EXPECT_EQ(0, tracker.moves());
>+    EXPECT_EQ(4, tracker.instances());
>+    EXPECT_EQ(2, tracker.live_instances());
>+  }
>+}
>+
>+TEST(TestInstanceTracker, ExistingInstances) {
>+  CopyableMovableInstance uncounted_instance(1);
>+  CopyableMovableInstance uncounted_live_instance(
>+      std::move(uncounted_instance));
>+  InstanceTracker tracker;
>+  EXPECT_EQ(0, tracker.instances());
>+  EXPECT_EQ(0, tracker.live_instances());
>+  EXPECT_EQ(0, tracker.copies());
>+  {
>+    CopyableMovableInstance instance1(1);
>+    EXPECT_EQ(1, tracker.instances());
>+    EXPECT_EQ(1, tracker.live_instances());
>+    EXPECT_EQ(0, tracker.copies());
>+    EXPECT_EQ(0, tracker.moves());
>+    {
>+      InstanceTracker tracker2;
>+      CopyableMovableInstance instance2(instance1);
>+      CopyableMovableInstance instance3(std::move(instance2));
>+      EXPECT_EQ(3, tracker.instances());
>+      EXPECT_EQ(2, tracker.live_instances());
>+      EXPECT_EQ(1, tracker.copies());
>+      EXPECT_EQ(1, tracker.moves());
>+      EXPECT_EQ(2, tracker2.instances());
>+      EXPECT_EQ(1, tracker2.live_instances());
>+      EXPECT_EQ(1, tracker2.copies());
>+      EXPECT_EQ(1, tracker2.moves());
>+    }
>+    EXPECT_EQ(1, tracker.instances());
>+    EXPECT_EQ(1, tracker.live_instances());
>+    EXPECT_EQ(1, tracker.copies());
>+    EXPECT_EQ(1, tracker.moves());
>+  }
>+  EXPECT_EQ(0, tracker.instances());
>+  EXPECT_EQ(0, tracker.live_instances());
>+  EXPECT_EQ(1, tracker.copies());
>+  EXPECT_EQ(1, tracker.moves());
>+}
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/copts.bzl b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/copts.bzl
>new file mode 100644
>index 00000000000..0168ac5abdd
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/copts.bzl
>@@ -0,0 +1,154 @@
>+"""absl specific copts.
>+
>+Flags specified here must not impact ABI. Code compiled with and without these
>+opts will be linked together, and in some cases headers compiled with and
>+without these options will be part of the same program.
>+"""
>+GCC_FLAGS = [
>+    "-Wall",
>+    "-Wextra",
>+    "-Wcast-qual",
>+    "-Wconversion-null",
>+    "-Wmissing-declarations",
>+    "-Woverlength-strings",
>+    "-Wpointer-arith",
>+    "-Wunused-local-typedefs",
>+    "-Wunused-result",
>+    "-Wvarargs",
>+    "-Wvla",  # variable-length array
>+    "-Wwrite-strings",
>+    # Google style does not use unsigned integers, though STL containers
>+    # have unsigned types.
>+    "-Wno-sign-compare",
>+]
>+
>+GCC_TEST_FLAGS = [
>+    "-Wno-conversion-null",
>+    "-Wno-missing-declarations",
>+    "-Wno-sign-compare",
>+    "-Wno-unused-function",
>+    "-Wno-unused-parameter",
>+    "-Wno-unused-private-field",
>+]
>+
>+# Docs on single flags is preceded by a comment.
>+# Docs on groups of flags is preceded by ###.
>+
>+LLVM_FLAGS = [
>+    # All warnings are treated as errors by implicit -Werror flag
>+    "-Wall",
>+    "-Wextra",
>+    "-Weverything",
>+    # Abseil does not support C++98
>+    "-Wno-c++98-compat-pedantic",
>+    # Turns off all implicit conversion warnings. Most are re-enabled below.
>+    "-Wno-conversion",
>+    "-Wno-covered-switch-default",
>+    "-Wno-deprecated",
>+    "-Wno-disabled-macro-expansion",
>+    "-Wno-double-promotion",
>+    ###
>+    # Turned off as they include valid C++ code.
>+    "-Wno-comma",
>+    "-Wno-extra-semi",
>+    "-Wno-packed",
>+    "-Wno-padded",
>+    ###
>+    "-Wno-float-conversion",
>+    "-Wno-float-equal",
>+    "-Wno-format-nonliteral",
>+    # Too aggressive: warns on Clang extensions enclosed in Clang-only
>+    # compilation paths.
>+    "-Wno-gcc-compat",
>+    ###
>+    # Some internal globals are necessary. Don't do this at home.
>+    "-Wno-global-constructors",
>+    "-Wno-exit-time-destructors",
>+    ###
>+    "-Wno-nested-anon-types",
>+    "-Wno-non-modular-include-in-module",
>+    "-Wno-old-style-cast",
>+    # Warns on preferred usage of non-POD types such as string_view
>+    "-Wno-range-loop-analysis",
>+    "-Wno-reserved-id-macro",
>+    "-Wno-shorten-64-to-32",
>+    "-Wno-switch-enum",
>+    "-Wno-thread-safety-negative",
>+    "-Wno-undef",
>+    "-Wno-unknown-warning-option",
>+    "-Wno-unreachable-code",
>+    # Causes warnings on include guards
>+    "-Wno-unused-macros",
>+    "-Wno-weak-vtables",
>+    ###
>+    # Implicit conversion warnings turned off by -Wno-conversion
>+    # which are re-enabled below.
>+    "-Wbitfield-enum-conversion",
>+    "-Wbool-conversion",
>+    "-Wconstant-conversion",
>+    "-Wenum-conversion",
>+    "-Wint-conversion",
>+    "-Wliteral-conversion",
>+    "-Wnon-literal-null-conversion",
>+    "-Wnull-conversion",
>+    "-Wobjc-literal-conversion",
>+    "-Wno-sign-conversion",
>+    "-Wstring-conversion",
>+    ###
>+]
>+
>+LLVM_TEST_FLAGS = [
>+    "-Wno-c99-extensions",
>+    "-Wno-missing-noreturn",
>+    "-Wno-missing-prototypes",
>+    "-Wno-null-conversion",
>+    "-Wno-shadow",
>+    "-Wno-shift-sign-overflow",
>+    "-Wno-sign-compare",
>+    "-Wno-unused-function",
>+    "-Wno-unused-member-function",
>+    "-Wno-unused-parameter",
>+    "-Wno-unused-private-field",
>+    "-Wno-unused-template",
>+    "-Wno-used-but-marked-unused",
>+    "-Wno-zero-as-null-pointer-constant",
>+]
>+
>+MSVC_FLAGS = [
>+    "/W3",
>+    "/WX",
>+    "/wd4005",  # macro-redefinition
>+    "/wd4068",  # unknown pragma
>+    "/wd4244",  # conversion from 'type1' to 'type2', possible loss of data
>+    "/wd4267",  # conversion from 'size_t' to 'type', possible loss of data
>+    "/wd4800",  # forcing value to bool 'true' or 'false' (performance warning)
>+    "/DNOMINMAX",  # Don't define min and max macros (windows.h)
>+    "/DWIN32_LEAN_AND_MEAN",  # Don't bloat namespace with incompatible winsock versions.
>+    "/D_CRT_SECURE_NO_WARNINGS",  # Don't warn about usage of insecure C functions
>+]
>+
>+MSVC_TEST_FLAGS = [
>+    "/wd4018",  # signed/unsigned mismatch
>+    "/wd4101",  # unreferenced local variable
>+    "/wd4503",  # decorated name length exceeded, name was truncated
>+]
>+
>+# /Wall with msvc includes unhelpful warnings such as C4711, C4710, ...
>+ABSL_DEFAULT_COPTS = select({
>+    "//absl:windows": MSVC_FLAGS,
>+    "//absl:llvm_compiler": LLVM_FLAGS,
>+    "//conditions:default": GCC_FLAGS,
>+})
>+
>+# in absence of modules (--compiler=gcc or -c opt), cc_tests leak their copts
>+# to their (included header) dependencies and fail to build outside absl
>+ABSL_TEST_COPTS = ABSL_DEFAULT_COPTS + select({
>+    "//absl:windows": MSVC_TEST_FLAGS,
>+    "//absl:llvm_compiler": LLVM_TEST_FLAGS,
>+    "//conditions:default": GCC_TEST_FLAGS,
>+})
>+
>+ABSL_EXCEPTIONS_FLAG = select({
>+    "//absl:windows": ["/U_HAS_EXCEPTIONS", "/D_HAS_EXCEPTIONS=1", "/EHsc"],
>+    "//conditions:default": ["-fexceptions"],
>+})
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/BUILD.bazel b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/BUILD.bazel
>new file mode 100644
>index 00000000000..e1e7fced69f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/BUILD.bazel
>@@ -0,0 +1,305 @@
>+#
>+# Copyright 2017 The Abseil Authors.
>+#
>+# Licensed under the Apache License, Version 2.0 (the "License");
>+# you may not use this file except in compliance with the License.
>+# You may obtain a copy of the License at
>+#
>+#      http://www.apache.org/licenses/LICENSE-2.0
>+#
>+# Unless required by applicable law or agreed to in writing, software
>+# distributed under the License is distributed on an "AS IS" BASIS,
>+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+# See the License for the specific language governing permissions and
>+# limitations under the License.
>+#
>+
>+load(
>+    "//absl:copts.bzl",
>+    "ABSL_DEFAULT_COPTS",
>+    "ABSL_TEST_COPTS",
>+)
>+
>+package(
>+    default_visibility = ["//visibility:public"],
>+)
>+
>+licenses(["notice"])  # Apache 2.0
>+
>+cc_library(
>+    name = "stacktrace",
>+    srcs = [
>+        "stacktrace.cc",
>+    ],
>+    hdrs = ["stacktrace.h"],
>+    copts = ABSL_DEFAULT_COPTS,
>+    deps = [
>+        ":debugging_internal",
>+        "//absl/base",
>+        "//absl/base:core_headers",
>+    ],
>+)
>+
>+cc_library(
>+    name = "symbolize",
>+    srcs = [
>+        "symbolize.cc",
>+        "symbolize_elf.inc",
>+        "symbolize_unimplemented.inc",
>+        "symbolize_win32.inc",
>+    ],
>+    hdrs = [
>+        "internal/symbolize.h",
>+        "symbolize.h",
>+    ],
>+    copts = ABSL_DEFAULT_COPTS,
>+    deps = [
>+        ":debugging_internal",
>+        ":demangle_internal",
>+        "//absl/base",
>+        "//absl/base:core_headers",
>+        "//absl/base:malloc_internal",
>+    ],
>+)
>+
>+cc_test(
>+    name = "symbolize_test",
>+    srcs = ["symbolize_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    deps = [
>+        ":stack_consumption",
>+        ":symbolize",
>+        "//absl/base",
>+        "//absl/base:core_headers",
>+        "//absl/memory",
>+        "@com_google_googletest//:gtest",
>+    ],
>+)
>+
>+cc_library(
>+    name = "examine_stack",
>+    srcs = [
>+        "internal/examine_stack.cc",
>+    ],
>+    hdrs = [
>+        "internal/examine_stack.h",
>+    ],
>+    copts = ABSL_DEFAULT_COPTS,
>+    visibility = ["//visibility:private"],
>+    deps = [
>+        ":stacktrace",
>+        ":symbolize",
>+        "//absl/base",
>+        "//absl/base:core_headers",
>+    ],
>+)
>+
>+cc_library(
>+    name = "failure_signal_handler",
>+    srcs = ["failure_signal_handler.cc"],
>+    hdrs = ["failure_signal_handler.h"],
>+    copts = ABSL_DEFAULT_COPTS,
>+    deps = [
>+        ":examine_stack",
>+        ":stacktrace",
>+        "//absl/base",
>+        "//absl/base:config",
>+        "//absl/base:core_headers",
>+    ],
>+)
>+
>+cc_test(
>+    name = "failure_signal_handler_test",
>+    srcs = ["failure_signal_handler_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    linkopts = select({
>+        "//absl:windows": [],
>+        "//conditions:default": ["-pthread"],
>+    }),
>+    deps = [
>+        ":failure_signal_handler",
>+        ":stacktrace",
>+        ":symbolize",
>+        "//absl/base",
>+        "//absl/strings",
>+        "@com_google_googletest//:gtest",
>+    ],
>+)
>+
>+cc_library(
>+    name = "debugging_internal",
>+    srcs = [
>+        "internal/address_is_readable.cc",
>+        "internal/elf_mem_image.cc",
>+        "internal/vdso_support.cc",
>+    ],
>+    hdrs = [
>+        "internal/address_is_readable.h",
>+        "internal/elf_mem_image.h",
>+        "internal/stacktrace_aarch64-inl.inc",
>+        "internal/stacktrace_arm-inl.inc",
>+        "internal/stacktrace_config.h",
>+        "internal/stacktrace_generic-inl.inc",
>+        "internal/stacktrace_powerpc-inl.inc",
>+        "internal/stacktrace_unimplemented-inl.inc",
>+        "internal/stacktrace_win32-inl.inc",
>+        "internal/stacktrace_x86-inl.inc",
>+        "internal/vdso_support.h",
>+    ],
>+    copts = ABSL_DEFAULT_COPTS,
>+    deps = [
>+        "//absl/base",
>+        "//absl/base:dynamic_annotations",
>+    ],
>+)
>+
>+cc_library(
>+    name = "demangle_internal",
>+    srcs = ["internal/demangle.cc"],
>+    hdrs = ["internal/demangle.h"],
>+    copts = ABSL_DEFAULT_COPTS,
>+    deps = [
>+        "//absl/base",
>+        "//absl/base:core_headers",
>+    ],
>+)
>+
>+cc_test(
>+    name = "demangle_test",
>+    srcs = ["internal/demangle_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    deps = [
>+        ":demangle_internal",
>+        ":stack_consumption",
>+        "//absl/base",
>+        "//absl/base:core_headers",
>+        "//absl/memory",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_library(
>+    name = "leak_check",
>+    srcs = select({
>+        # The leak checking interface depends on weak function
>+        # declarations that may not necessarily have definitions.
>+        # Windows doesn't support this, and ios requires
>+        # guaranteed definitions for weak symbols.
>+        "//absl:ios": [],
>+        "//absl:windows": [],
>+        "//conditions:default": [
>+            "leak_check.cc",
>+        ],
>+    }),
>+    hdrs = select({
>+        "//absl:ios": [],
>+        "//absl:windows": [],
>+        "//conditions:default": ["leak_check.h"],
>+    }),
>+    deps = ["//absl/base:core_headers"],
>+)
>+
>+# Adding a dependency to leak_check_disable will disable
>+# sanitizer leak checking (asan/lsan) in a test without
>+# the need to mess around with build features.
>+cc_library(
>+    name = "leak_check_disable",
>+    srcs = ["leak_check_disable.cc"],
>+    linkstatic = 1,
>+    alwayslink = 1,
>+)
>+
>+# These targets exists for use in tests only, explicitly configuring the
>+# LEAK_SANITIZER macro. It must be linked with -fsanitize=leak for lsan.
>+ABSL_LSAN_LINKOPTS = select({
>+    "//absl:llvm_compiler": ["-fsanitize=leak"],
>+    "//conditions:default": [],
>+})
>+
>+cc_library(
>+    name = "leak_check_api_enabled_for_testing",
>+    testonly = 1,
>+    srcs = ["leak_check.cc"],
>+    hdrs = ["leak_check.h"],
>+    copts = select({
>+        "//absl:llvm_compiler": ["-DLEAK_SANITIZER"],
>+        "//conditions:default": [],
>+    }),
>+    visibility = ["//visibility:private"],
>+)
>+
>+cc_library(
>+    name = "leak_check_api_disabled_for_testing",
>+    testonly = 1,
>+    srcs = ["leak_check.cc"],
>+    hdrs = ["leak_check.h"],
>+    copts = ["-ULEAK_SANITIZER"],
>+    visibility = ["//visibility:private"],
>+)
>+
>+cc_test(
>+    name = "leak_check_test",
>+    srcs = ["leak_check_test.cc"],
>+    copts = select({
>+        "//absl:llvm_compiler": ["-DABSL_EXPECT_LEAK_SANITIZER"],
>+        "//conditions:default": [],
>+    }),
>+    linkopts = ABSL_LSAN_LINKOPTS,
>+    deps = [
>+        ":leak_check_api_enabled_for_testing",
>+        "//absl/base",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "leak_check_no_lsan_test",
>+    srcs = ["leak_check_test.cc"],
>+    copts = ["-UABSL_EXPECT_LEAK_SANITIZER"],
>+    deps = [
>+        ":leak_check_api_disabled_for_testing",
>+        "//absl/base",  # for raw_logging
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+# Test that leak checking is skipped when lsan is enabled but
>+# ":leak_check_disable" is linked in.
>+#
>+# This test should fail in the absence of a dependency on ":leak_check_disable"
>+cc_test(
>+    name = "disabled_leak_check_test",
>+    srcs = ["leak_check_fail_test.cc"],
>+    linkopts = ABSL_LSAN_LINKOPTS,
>+    deps = [
>+        ":leak_check_api_enabled_for_testing",
>+        ":leak_check_disable",
>+        "//absl/base",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_library(
>+    name = "stack_consumption",
>+    testonly = 1,
>+    srcs = ["internal/stack_consumption.cc"],
>+    hdrs = ["internal/stack_consumption.h"],
>+    copts = ABSL_DEFAULT_COPTS,
>+    visibility = ["//visibility:private"],
>+    deps = [
>+        "//absl/base",
>+        "//absl/base:core_headers",
>+    ],
>+)
>+
>+cc_test(
>+    name = "stack_consumption_test",
>+    srcs = ["internal/stack_consumption_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    deps = [
>+        ":stack_consumption",
>+        "//absl/base",
>+        "//absl/base:core_headers",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/BUILD.gn
>new file mode 100644
>index 00000000000..b50b0a6ddc8
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/BUILD.gn
>@@ -0,0 +1,245 @@
>+# Copyright 2018 The Chromium Authors. All rights reserved.
>+# Use of this source code is governed by a BSD-style license that can be
>+# found in the LICENSE file.
>+
>+import("//build_overrides/build.gni")
>+import("//build/config/sanitizers/sanitizers.gni")
>+
>+if (build_with_chromium) {
>+  visibility = [
>+    "//third_party/webrtc/*",
>+    "//third_party/abseil-cpp/*",
>+    "//third_party/googletest:gtest",
>+  ]
>+} else {
>+  visibility = [ "*" ]
>+}
>+
>+source_set("stacktrace") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  sources = [
>+    "stacktrace.cc",
>+  ]
>+  public = [
>+    "stacktrace.h",
>+  ]
>+  deps = [
>+    ":debugging_internal",
>+    "../base",
>+    "../base:core_headers",
>+  ]
>+}
>+
>+source_set("symbolize") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  sources = [
>+    "symbolize.cc",
>+    "symbolize_elf.inc",
>+    "symbolize_unimplemented.inc",
>+    "symbolize_win32.inc",
>+  ]
>+  public = [
>+    "internal/symbolize.h",
>+    "symbolize.h",
>+  ]
>+  deps = [
>+    ":debugging_internal",
>+    ":demangle_internal",
>+    "../base",
>+    "../base:core_headers",
>+    "../base:malloc_internal",
>+  ]
>+}
>+
>+source_set("examine_stack") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  sources = [
>+    "internal/examine_stack.cc",
>+  ]
>+  public = [
>+    "internal/examine_stack.h",
>+  ]
>+  visibility = []
>+  visibility += [ ":*" ]
>+  deps = [
>+    ":stacktrace",
>+    ":symbolize",
>+    "../base",
>+    "../base:core_headers",
>+  ]
>+}
>+
>+source_set("failure_signal_handler") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  sources = [
>+    "failure_signal_handler.cc"
>+  ]
>+  public = [
>+    "failure_signal_handler.h"
>+  ]
>+  deps = [
>+    ":examine_stack",
>+    ":stacktrace",
>+    "../base",
>+    "../base:config",
>+    "../base:core_headers",
>+  ]
>+}
>+
>+source_set("debugging_internal") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  sources = [
>+    "internal/address_is_readable.cc",
>+    "internal/elf_mem_image.cc",
>+    "internal/vdso_support.cc",
>+  ]
>+  public = [
>+    "internal/address_is_readable.h",
>+    "internal/elf_mem_image.h",
>+    "internal/stacktrace_config.h",
>+    "internal/vdso_support.h",
>+  ]
>+  deps = [
>+    "../base",
>+    "../base:dynamic_annotations",
>+  ]
>+}
>+
>+source_set("demangle_internal") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  sources = [
>+    "internal/demangle.cc",
>+  ]
>+  public = [
>+    "internal/demangle.h",
>+  ]
>+  deps = [
>+    "../base",
>+    "../base:core_headers",
>+  ]
>+}
>+
>+source_set("leak_check") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  if (is_ios || is_win) {
>+    sources = []
>+    public = []
>+  } else {
>+    sources = [
>+      "leak_check.cc",
>+    ]
>+    public = [
>+      "leak_check.h",
>+    ]
>+  }
>+  deps = [
>+    "../base:core_headers",
>+  ]
>+}
>+
>+source_set("leak_check_disable") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  sources = [
>+    "leak_check_disable.cc",
>+  ]
>+}
>+
>+if (is_lsan) {
>+  source_set("leak_check_api_enabled_for_testing") {
>+    testonly = true
>+    configs -= [ "//build/config/compiler:chromium_code" ]
>+    configs += [
>+      "//build/config/compiler:no_chromium_code",
>+      "//third_party/abseil-cpp:absl_default_cflags_cc",
>+    ]
>+    public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+    sources = [
>+      "leak_check.cc",
>+    ]
>+    public = [
>+      "leak_check.h",
>+    ]
>+    visibility = []
>+    visibility += [ ":*" ]
>+  }
>+} else {
>+  source_set("leak_check_api_disabled_for_testing") {
>+    testonly = true
>+    configs -= [ "//build/config/compiler:chromium_code" ]
>+    configs += [
>+      "//build/config/compiler:no_chromium_code",
>+      "//third_party/abseil-cpp:absl_default_cflags_cc",
>+    ]
>+    public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+    sources = [
>+      "leak_check.cc",
>+    ]
>+    public = [
>+      "leak_check.h",
>+    ]
>+    visibility = []
>+    visibility += [ ":*" ]
>+  }
>+}
>+
>+source_set("stack_consumption") {
>+  testonly = true
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  sources = [
>+    "internal/stack_consumption.cc",
>+  ]
>+  public = [
>+    "internal/stack_consumption.h",
>+  ]
>+  deps = [
>+    "../base",
>+    "../base:core_headers",
>+  ]
>+  visibility = []
>+  visibility += [ ":*" ]
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/CMakeLists.txt b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/CMakeLists.txt
>new file mode 100644
>index 00000000000..4af2ec8a411
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/CMakeLists.txt
>@@ -0,0 +1,217 @@
>+#
>+# Copyright 2017 The Abseil Authors.
>+#
>+# Licensed under the Apache License, Version 2.0 (the "License");
>+# you may not use this file except in compliance with the License.
>+# You may obtain a copy of the License at
>+#
>+#      http://www.apache.org/licenses/LICENSE-2.0
>+#
>+# Unless required by applicable law or agreed to in writing, software
>+# distributed under the License is distributed on an "AS IS" BASIS,
>+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+# See the License for the specific language governing permissions and
>+# limitations under the License.
>+#
>+
>+list(APPEND DEBUGGING_PUBLIC_HEADERS
>+  "failure_signal_handler.h"
>+  "leak_check.h"
>+  "stacktrace.h"
>+  "symbolize.h"
>+)
>+
>+# TODO(cohenjon) The below is all kinds of wrong.  Make this match what we do in
>+# Bazel
>+list(APPEND DEBUGGING_INTERNAL_HEADERS
>+  "internal/address_is_readable.h"
>+  "internal/demangle.h"
>+  "internal/elf_mem_image.h"
>+  "internal/examine_stack.h"
>+  "internal/stacktrace_config.h"
>+  "internal/symbolize.h"
>+  "internal/vdso_support.h"
>+)
>+
>+list(APPEND DEBUGGING_INTERNAL_SRC
>+  "internal/address_is_readable.cc"
>+  "internal/elf_mem_image.cc"
>+  "internal/vdso_support.cc"
>+)
>+
>+
>+list(APPEND STACKTRACE_SRC
>+  "stacktrace.cc"
>+  ${DEBUGGING_INTERNAL_SRC}
>+  ${DEBUGGING_PUBLIC_HEADERS}
>+  ${DEBUGGING_INTERNAL_HEADERS}
>+)
>+
>+list(APPEND SYMBOLIZE_SRC
>+  "symbolize.cc"
>+  "symbolize_elf.inc"
>+  "symbolize_unimplemented.inc"
>+  "symbolize_win32.inc"
>+  "internal/demangle.cc"
>+  ${DEBUGGING_PUBLIC_HEADERS}
>+  ${DEBUGGING_INTERNAL_HEADERS}
>+  ${DEBUGGING_INTERNAL_SRC}
>+)
>+
>+list(APPEND FAILURE_SIGNAL_HANDLER_SRC
>+  "failure_signal_handler.cc"
>+  ${DEBUGGING_PUBLIC_HEADERS}
>+)
>+
>+list(APPEND EXAMINE_STACK_SRC
>+  "internal/examine_stack.cc"
>+  ${DEBUGGING_PUBLIC_HEADERS}
>+  ${DEBUGGING_INTERNAL_HEADERS}
>+)
>+
>+absl_library(
>+  TARGET
>+    absl_stacktrace
>+  SOURCES
>+    ${STACKTRACE_SRC}
>+  EXPORT_NAME
>+    stacktrace
>+)
>+
>+absl_library(
>+  TARGET
>+    absl_symbolize
>+  SOURCES
>+    ${SYMBOLIZE_SRC}
>+  PUBLIC_LIBRARIES
>+    absl::base
>+    absl_malloc_internal
>+  EXPORT_NAME
>+    symbolize
>+)
>+
>+absl_library(
>+  TARGET
>+    absl_failure_signal_handler
>+  SOURCES
>+    ${FAILURE_SIGNAL_HANDLER_SRC}
>+  PUBLIC_LIBRARIES
>+    absl_base absl::examine_stack absl::stacktrace absl_synchronization
>+  EXPORT_NAME
>+    failure_signal_handler
>+)
>+
>+# Internal-only. Projects external to Abseil should not depend
>+# directly on this library.
>+absl_library(
>+  TARGET
>+    absl_examine_stack
>+  SOURCES
>+    ${EXAMINE_STACK_SRC}
>+  EXPORT_NAME
>+    examine_stack
>+)
>+
>+list(APPEND LEAK_CHECK_SRC
>+  "leak_check.cc"
>+)
>+
>+
>+# leak_check library
>+absl_library(
>+  TARGET
>+    absl_leak_check
>+  SOURCES
>+    ${LEAK_CHECK_SRC}
>+  PUBLIC_LIBRARIES
>+    absl_base
>+  EXPORT_NAME
>+    leak_check
>+)
>+
>+
>+# component target
>+absl_header_library(
>+  TARGET
>+    absl_debugging
>+  PUBLIC_LIBRARIES
>+    absl_stacktrace absl_leak_check
>+  EXPORT_NAME
>+    debugging
>+)
>+
>+#
>+## TESTS
>+#
>+
>+list(APPEND STACK_CONSUMPTION_SRC
>+  "internal/stack_consumption.cc"
>+  "internal/stack_consumption.h"
>+)
>+
>+absl_library(
>+  TARGET
>+    absl_stack_consumption
>+  SOURCES
>+    ${STACK_CONSUMPTION_SRC}
>+)
>+
>+absl_test(
>+  TARGET
>+    absl_stack_consumption_test
>+  SOURCES
>+    "internal/stack_consumption_test.cc"
>+  PUBLIC_LIBRARIES
>+    absl_stack_consumption
>+    absl::base
>+)
>+
>+list(APPEND DEMANGLE_TEST_SRC "internal/demangle_test.cc")
>+
>+absl_test(
>+  TARGET
>+    demangle_test
>+  SOURCES
>+    ${DEMANGLE_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    absl_symbolize absl_stack_consumption
>+)
>+
>+list(APPEND SYMBOLIZE_TEST_SRC "symbolize_test.cc")
>+
>+absl_test(
>+  TARGET
>+    symbolize_test
>+  SOURCES
>+    ${SYMBOLIZE_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    absl::base absl::memory absl_symbolize absl_stack_consumption
>+)
>+
>+list(APPEND FAILURE_SIGNAL_HANDLER_TEST_SRC "failure_signal_handler_test.cc")
>+
>+absl_test(
>+  TARGET
>+    failure_signal_handler_test
>+  SOURCES
>+    ${FAILURE_SIGNAL_HANDLER_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    absl_examine_stack
>+    absl_failure_signal_handler
>+    absl_stacktrace
>+    absl_symbolize
>+    absl::base
>+    absl::strings
>+)
>+
>+# test leak_check_test
>+list(APPEND LEAK_CHECK_TEST_SRC "leak_check_test.cc")
>+
>+absl_test(
>+  TARGET
>+    leak_check_test
>+  SOURCES
>+    ${LEAK_CHECK_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    absl_leak_check
>+)
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/failure_signal_handler.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/failure_signal_handler.cc
>new file mode 100644
>index 00000000000..d4b957bc2e4
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/failure_signal_handler.cc
>@@ -0,0 +1,355 @@
>+//
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+
>+#include "absl/debugging/failure_signal_handler.h"
>+
>+#include "absl/base/config.h"
>+
>+#ifdef _WIN32
>+#include <windows.h>
>+#else
>+#include <unistd.h>
>+#endif
>+
>+#ifdef ABSL_HAVE_MMAP
>+#include <sys/mman.h>
>+#endif
>+
>+#include <algorithm>
>+#include <atomic>
>+#include <cerrno>
>+#include <csignal>
>+#include <cstdio>
>+#include <cstring>
>+#include <ctime>
>+
>+#include "absl/base/attributes.h"
>+#include "absl/base/internal/raw_logging.h"
>+#include "absl/base/internal/sysinfo.h"
>+#include "absl/debugging/internal/examine_stack.h"
>+#include "absl/debugging/stacktrace.h"
>+
>+#ifndef _WIN32
>+#define ABSL_HAVE_SIGACTION
>+#endif
>+
>+namespace absl {
>+
>+ABSL_CONST_INIT static FailureSignalHandlerOptions fsh_options;
>+
>+// Resets the signal handler for signo to the default action for that
>+// signal, then raises the signal.
>+static void RaiseToDefaultHandler(int signo) {
>+  signal(signo, SIG_DFL);
>+  raise(signo);
>+}
>+
>+struct FailureSignalData {
>+  const int signo;
>+  const char* const as_string;
>+#ifdef ABSL_HAVE_SIGACTION
>+  struct sigaction previous_action;
>+  // StructSigaction is used to silence -Wmissing-field-initializers.
>+  using StructSigaction = struct sigaction;
>+  #define FSD_PREVIOUS_INIT FailureSignalData::StructSigaction()
>+#else
>+  void (*previous_handler)(int);
>+  #define FSD_PREVIOUS_INIT SIG_DFL
>+#endif
>+};
>+
>+ABSL_CONST_INIT static FailureSignalData failure_signal_data[] = {
>+    {SIGSEGV, "SIGSEGV", FSD_PREVIOUS_INIT},
>+    {SIGILL, "SIGILL", FSD_PREVIOUS_INIT},
>+    {SIGFPE, "SIGFPE", FSD_PREVIOUS_INIT},
>+    {SIGABRT, "SIGABRT", FSD_PREVIOUS_INIT},
>+    {SIGTERM, "SIGTERM", FSD_PREVIOUS_INIT},
>+#ifndef _WIN32
>+    {SIGBUS, "SIGBUS", FSD_PREVIOUS_INIT},
>+    {SIGTRAP, "SIGTRAP", FSD_PREVIOUS_INIT},
>+#endif
>+};
>+
>+#undef FSD_PREVIOUS_INIT
>+
>+static void RaiseToPreviousHandler(int signo) {
>+  // Search for the previous handler.
>+  for (const auto& it : failure_signal_data) {
>+    if (it.signo == signo) {
>+#ifdef ABSL_HAVE_SIGACTION
>+      sigaction(signo, &it.previous_action, nullptr);
>+#else
>+      signal(signo, it.previous_handler);
>+#endif
>+      raise(signo);
>+      return;
>+    }
>+  }
>+
>+  // Not found, use the default handler.
>+  RaiseToDefaultHandler(signo);
>+}
>+
>+namespace debugging_internal {
>+
>+const char* FailureSignalToString(int signo) {
>+  for (const auto& it : failure_signal_data) {
>+    if (it.signo == signo) {
>+      return it.as_string;
>+    }
>+  }
>+  return "";
>+}
>+
>+}  // namespace debugging_internal
>+
>+#ifndef _WIN32
>+
>+static bool SetupAlternateStackOnce() {
>+  const size_t page_mask = getpagesize() - 1;
>+  size_t stack_size = (std::max(SIGSTKSZ, 65536) + page_mask) & ~page_mask;
>+#if defined(ADDRESS_SANITIZER) || defined(MEMORY_SANITIZER) || \
>+    defined(THREAD_SANITIZER)
>+  // Account for sanitizer instrumentation requiring additional stack space.
>+  stack_size *= 5;
>+#endif
>+
>+  stack_t sigstk;
>+  memset(&sigstk, 0, sizeof(sigstk));
>+  sigstk.ss_size = stack_size;
>+
>+#ifdef ABSL_HAVE_MMAP
>+#ifndef MAP_STACK
>+#define MAP_STACK 0
>+#endif
>+#if defined(MAP_ANON) && !defined(MAP_ANONYMOUS)
>+#define MAP_ANONYMOUS MAP_ANON
>+#endif
>+  sigstk.ss_sp = mmap(nullptr, sigstk.ss_size, PROT_READ | PROT_WRITE,
>+                      MAP_PRIVATE | MAP_ANONYMOUS | MAP_STACK, -1, 0);
>+  if (sigstk.ss_sp == MAP_FAILED) {
>+    ABSL_RAW_LOG(FATAL, "mmap() for alternate signal stack failed");
>+  }
>+#else
>+  sigstk.ss_sp = malloc(sigstk.ss_size);
>+  if (sigstk.ss_sp == nullptr) {
>+    ABSL_RAW_LOG(FATAL, "malloc() for alternate signal stack failed");
>+  }
>+#endif
>+
>+  if (sigaltstack(&sigstk, nullptr) != 0) {
>+    ABSL_RAW_LOG(FATAL, "sigaltstack() failed with errno=%d", errno);
>+  }
>+  return true;
>+}
>+
>+#endif
>+
>+#ifdef ABSL_HAVE_SIGACTION
>+
>+// Sets up an alternate stack for signal handlers once.
>+// Returns the appropriate flag for sig_action.sa_flags
>+// if the system supports using an alternate stack.
>+static int MaybeSetupAlternateStack() {
>+#ifndef _WIN32
>+  ABSL_ATTRIBUTE_UNUSED static const bool kOnce = SetupAlternateStackOnce();
>+  return SA_ONSTACK;
>+#else
>+  return 0;
>+#endif
>+}
>+
>+static void InstallOneFailureHandler(FailureSignalData* data,
>+                                     void (*handler)(int, siginfo_t*, void*)) {
>+  struct sigaction act;
>+  memset(&act, 0, sizeof(act));
>+  sigemptyset(&act.sa_mask);
>+  act.sa_flags |= SA_SIGINFO;
>+  // SA_NODEFER is required to handle SIGABRT from
>+  // ImmediateAbortSignalHandler().
>+  act.sa_flags |= SA_NODEFER;
>+  if (fsh_options.use_alternate_stack) {
>+    act.sa_flags |= MaybeSetupAlternateStack();
>+  }
>+  act.sa_sigaction = handler;
>+  ABSL_RAW_CHECK(sigaction(data->signo, &act, &data->previous_action) == 0,
>+                 "sigaction() failed");
>+}
>+
>+#else
>+
>+static void InstallOneFailureHandler(FailureSignalData* data,
>+                                     void (*handler)(int)) {
>+  data->previous_handler = signal(data->signo, handler);
>+  ABSL_RAW_CHECK(data->previous_handler != SIG_ERR, "signal() failed");
>+}
>+
>+#endif
>+
>+static void WriteToStderr(const char* data) {
>+  int old_errno = errno;
>+  absl::raw_logging_internal::SafeWriteToStderr(data, strlen(data));
>+  errno = old_errno;
>+}
>+
>+static void WriteSignalMessage(int signo, void (*writerfn)(const char*)) {
>+  char buf[64];
>+  const char* const signal_string =
>+      debugging_internal::FailureSignalToString(signo);
>+  if (signal_string != nullptr && signal_string[0] != '\0') {
>+    snprintf(buf, sizeof(buf), "*** %s received at time=%ld ***\n",
>+             signal_string,
>+             static_cast<long>(time(nullptr)));  // NOLINT(runtime/int)
>+  } else {
>+    snprintf(buf, sizeof(buf), "*** Signal %d received at time=%ld ***\n",
>+             signo, static_cast<long>(time(nullptr)));  // NOLINT(runtime/int)
>+  }
>+  writerfn(buf);
>+}
>+
>+// `void*` might not be big enough to store `void(*)(const char*)`.
>+struct WriterFnStruct {
>+  void (*writerfn)(const char*);
>+};
>+
>+// Many of the absl::debugging_internal::Dump* functions in
>+// examine_stack.h take a writer function pointer that has a void* arg
>+// for historical reasons. failure_signal_handler_writer only takes a
>+// data pointer. This function converts between these types.
>+static void WriterFnWrapper(const char* data, void* arg) {
>+  static_cast<WriterFnStruct*>(arg)->writerfn(data);
>+}
>+
>+// Convenient wrapper around DumpPCAndFrameSizesAndStackTrace() for signal
>+// handlers. "noinline" so that GetStackFrames() skips the top-most stack
>+// frame for this function.
>+ABSL_ATTRIBUTE_NOINLINE static void WriteStackTrace(
>+    void* ucontext, bool symbolize_stacktrace,
>+    void (*writerfn)(const char*, void*), void* writerfn_arg) {
>+  constexpr int kNumStackFrames = 32;
>+  void* stack[kNumStackFrames];
>+  int frame_sizes[kNumStackFrames];
>+  int min_dropped_frames;
>+  int depth = absl::GetStackFramesWithContext(
>+      stack, frame_sizes, kNumStackFrames,
>+      1,  // Do not include this function in stack trace.
>+      ucontext, &min_dropped_frames);
>+  absl::debugging_internal::DumpPCAndFrameSizesAndStackTrace(
>+      absl::debugging_internal::GetProgramCounter(ucontext), stack, frame_sizes,
>+      depth, min_dropped_frames, symbolize_stacktrace, writerfn, writerfn_arg);
>+}
>+
>+// Called by AbslFailureSignalHandler() to write the failure info. It is
>+// called once with writerfn set to WriteToStderr() and then possibly
>+// with writerfn set to the user provided function.
>+static void WriteFailureInfo(int signo, void* ucontext,
>+                             void (*writerfn)(const char*)) {
>+  WriterFnStruct writerfn_struct{writerfn};
>+  WriteSignalMessage(signo, writerfn);
>+  WriteStackTrace(ucontext, fsh_options.symbolize_stacktrace, WriterFnWrapper,
>+                  &writerfn_struct);
>+}
>+
>+// absl::SleepFor() can't be used here since AbslInternalSleepFor()
>+// may be overridden to do something that isn't async-signal-safe on
>+// some platforms.
>+static void PortableSleepForSeconds(int seconds) {
>+#ifdef _WIN32
>+  Sleep(seconds * 1000);
>+#else
>+  struct timespec sleep_time;
>+  sleep_time.tv_sec = seconds;
>+  sleep_time.tv_nsec = 0;
>+  while (nanosleep(&sleep_time, &sleep_time) != 0 && errno == EINTR) {}
>+#endif
>+}
>+
>+#ifdef ABSL_HAVE_ALARM
>+// AbslFailureSignalHandler() installs this as a signal handler for
>+// SIGALRM, then sets an alarm to be delivered to the program after a
>+// set amount of time. If AbslFailureSignalHandler() hangs for more than
>+// the alarm timeout, ImmediateAbortSignalHandler() will abort the
>+// program.
>+static void ImmediateAbortSignalHandler(int) {
>+  RaiseToDefaultHandler(SIGABRT);
>+}
>+#endif
>+
>+// absl::base_internal::GetTID() returns pid_t on most platforms, but
>+// returns absl::base_internal::pid_t on Windows.
>+using GetTidType = decltype(absl::base_internal::GetTID());
>+ABSL_CONST_INIT static std::atomic<GetTidType> failed_tid(0);
>+
>+#ifndef ABSL_HAVE_SIGACTION
>+static void AbslFailureSignalHandler(int signo) {
>+  void* ucontext = nullptr;
>+#else
>+static void AbslFailureSignalHandler(int signo, siginfo_t*, void* ucontext) {
>+#endif
>+
>+  const GetTidType this_tid = absl::base_internal::GetTID();
>+  GetTidType previous_failed_tid = 0;
>+  if (!failed_tid.compare_exchange_strong(
>+          previous_failed_tid, static_cast<intptr_t>(this_tid),
>+          std::memory_order_acq_rel, std::memory_order_relaxed)) {
>+    ABSL_RAW_LOG(
>+        ERROR,
>+        "Signal %d raised at PC=%p while already in AbslFailureSignalHandler()",
>+        signo, absl::debugging_internal::GetProgramCounter(ucontext));
>+    if (this_tid != previous_failed_tid) {
>+      // Another thread is already in AbslFailureSignalHandler(), so wait
>+      // a bit for it to finish. If the other thread doesn't kill us,
>+      // we do so after sleeping.
>+      PortableSleepForSeconds(3);
>+      RaiseToDefaultHandler(signo);
>+      // The recursively raised signal may be blocked until we return.
>+      return;
>+    }
>+  }
>+
>+#ifdef ABSL_HAVE_ALARM
>+  // Set an alarm to abort the program in case this code hangs or deadlocks.
>+  if (fsh_options.alarm_on_failure_secs > 0) {
>+    alarm(0);  // Cancel any existing alarms.
>+    signal(SIGALRM, ImmediateAbortSignalHandler);
>+    alarm(fsh_options.alarm_on_failure_secs);
>+  }
>+#endif
>+
>+  // First write to stderr.
>+  WriteFailureInfo(signo, ucontext, WriteToStderr);
>+
>+  // Riskier code (because it is less likely to be async-signal-safe)
>+  // goes after this point.
>+  if (fsh_options.writerfn != nullptr) {
>+    WriteFailureInfo(signo, ucontext, fsh_options.writerfn);
>+  }
>+
>+  if (fsh_options.call_previous_handler) {
>+    RaiseToPreviousHandler(signo);
>+  } else {
>+    RaiseToDefaultHandler(signo);
>+  }
>+}
>+
>+void InstallFailureSignalHandler(const FailureSignalHandlerOptions& options) {
>+  fsh_options = options;
>+  for (auto& it : failure_signal_data) {
>+    InstallOneFailureHandler(&it, AbslFailureSignalHandler);
>+  }
>+}
>+
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/failure_signal_handler.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/failure_signal_handler.h
>new file mode 100644
>index 00000000000..c57954e5875
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/failure_signal_handler.h
>@@ -0,0 +1,117 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// -----------------------------------------------------------------------------
>+// File: failure_signal_handler.h
>+// -----------------------------------------------------------------------------
>+//
>+// This file configures the Abseil *failure signal handler* to capture and dump
>+// useful debugging information (such as a stacktrace) upon program failure.
>+//
>+// To use the failure signal handler, call `absl::InstallFailureSignalHandler()`
>+// very early in your program, usually in the first few lines of main():
>+//
>+// int main(int argc, char** argv) {
>+//   // Initialize the symbolizer to get a human-readable stack trace
>+//   absl::InitializeSymbolizer(argv[0]);
>+//
>+//   absl::FailureSignalHandlerOptions options;
>+//   absl::InstallFailureSignalHandler(options);
>+//   DoSomethingInteresting();
>+//   return 0;
>+// }
>+//
>+// Any program that raises a fatal signal (such as `SIGSEGV`, `SIGILL`,
>+// `SIGFPE`, `SIGABRT`, `SIGTERM`, `SIGBUG`, and `SIGTRAP`) will call the
>+// installed failure signal handler and provide debugging information to stderr.
>+//
>+// Note that you should *not* install the Abseil failure signal handler more
>+// than once. You may, of course, have another (non-Abseil) failure signal
>+// handler installed (which would be triggered if Abseil's failure signal
>+// handler sets `call_previous_handler` to `true`).
>+
>+#ifndef ABSL_DEBUGGING_FAILURE_SIGNAL_HANDLER_H_
>+#define ABSL_DEBUGGING_FAILURE_SIGNAL_HANDLER_H_
>+
>+namespace absl {
>+
>+// FailureSignalHandlerOptions
>+//
>+// Struct for holding `absl::InstallFailureSignalHandler()` configuration
>+// options.
>+struct FailureSignalHandlerOptions {
>+  // If true, try to symbolize the stacktrace emitted on failure, provided that
>+  // you have initialized a symbolizer for that purpose. (See symbolize.h for
>+  // more information.)
>+  bool symbolize_stacktrace = true;
>+
>+  // If true, try to run signal handlers on an alternate stack (if supported on
>+  // the given platform). An alternate stack is useful for program crashes due
>+  // to a stack overflow; by running on a alternate stack, the signal handler
>+  // may run even when normal stack space has been exausted. The downside of
>+  // using an alternate stack is that extra memory for the alternate stack needs
>+  // to be pre-allocated.
>+  bool use_alternate_stack = true;
>+
>+  // If positive, indicates the number of seconds after which the failure signal
>+  // handler is invoked to abort the program. Setting such an alarm is useful in
>+  // cases where the failure signal handler itself may become hung or
>+  // deadlocked.
>+  int alarm_on_failure_secs = 3;
>+
>+  // If true, call the previously registered signal handler for the signal that
>+  // was received (if one was registered) after the existing signal handler
>+  // runs. This mechanism can be used to chain signal handlers together.
>+  //
>+  // If false, the signal is raised to the default handler for that signal
>+  // (which normally terminates the program).
>+  //
>+  // IMPORTANT: If true, the chained fatal signal handlers must not try to
>+  // recover from the fatal signal. Instead, they should terminate the program
>+  // via some mechanism, like raising the default handler for the signal, or by
>+  // calling `_exit()`. Note that the failure signal handler may put parts of
>+  // the Abseil library into a state from which they cannot recover.
>+  bool call_previous_handler = false;
>+
>+  // If non-null, indicates a pointer to a callback function that will be called
>+  // upon failure, with a std::string argument containing failure data. This function
>+  // may be used as a hook to write failure data to a secondary location, such
>+  // as a log file. This function may also be called with null data, as a hint
>+  // to flush any buffered data before the program may be terminated. Consider
>+  // flushing any buffered data in all calls to this function.
>+  //
>+  // Since this function runs within a signal handler, it should be
>+  // async-signal-safe if possible.
>+  // See http://man7.org/linux/man-pages/man7/signal-safety.7.html
>+  void (*writerfn)(const char*) = nullptr;
>+};
>+
>+// InstallFailureSignalHandler()
>+//
>+// Installs a signal handler for the common failure signals `SIGSEGV`, `SIGILL`,
>+// `SIGFPE`, `SIGABRT`, `SIGTERM`, `SIGBUG`, and `SIGTRAP` (provided they exist
>+// on the given platform). The failure signal handler dumps program failure data
>+// useful for debugging in an unspecified format to stderr. This data may
>+// include the program counter, a stacktrace, and register information on some
>+// systems; do not rely on an exact format for the output, as it is subject to
>+// change.
>+void InstallFailureSignalHandler(const FailureSignalHandlerOptions& options);
>+
>+namespace debugging_internal {
>+const char* FailureSignalToString(int signo);
>+}  // namespace debugging_internal
>+
>+}  // namespace absl
>+
>+#endif  // ABSL_DEBUGGING_FAILURE_SIGNAL_HANDLER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/failure_signal_handler_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/failure_signal_handler_test.cc
>new file mode 100644
>index 00000000000..ba520910f47
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/failure_signal_handler_test.cc
>@@ -0,0 +1,155 @@
>+//
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+
>+#include "absl/debugging/failure_signal_handler.h"
>+
>+#include <csignal>
>+#include <cstdio>
>+#include <cstdlib>
>+#include <cstring>
>+#include <fstream>
>+
>+#include "gtest/gtest.h"
>+#include "absl/base/internal/raw_logging.h"
>+#include "absl/debugging/stacktrace.h"
>+#include "absl/debugging/symbolize.h"
>+#include "absl/strings/match.h"
>+#include "absl/strings/str_cat.h"
>+
>+namespace {
>+
>+#if GTEST_HAS_DEATH_TEST
>+
>+// For the parameterized death tests. GetParam() returns the signal number.
>+using FailureSignalHandlerDeathTest = ::testing::TestWithParam<int>;
>+
>+// This function runs in a fork()ed process on most systems.
>+void InstallHandlerAndRaise(int signo) {
>+  absl::InstallFailureSignalHandler(absl::FailureSignalHandlerOptions());
>+  raise(signo);
>+}
>+
>+TEST_P(FailureSignalHandlerDeathTest, AbslFailureSignal) {
>+  const int signo = GetParam();
>+  std::string exit_regex = absl::StrCat(
>+      "\\*\\*\\* ", absl::debugging_internal::FailureSignalToString(signo),
>+      " received at time=");
>+#ifndef _WIN32
>+  EXPECT_EXIT(InstallHandlerAndRaise(signo), testing::KilledBySignal(signo),
>+              exit_regex);
>+#else
>+  // Windows doesn't have testing::KilledBySignal().
>+  EXPECT_DEATH(InstallHandlerAndRaise(signo), exit_regex);
>+#endif
>+}
>+
>+ABSL_CONST_INIT FILE* error_file = nullptr;
>+
>+void WriteToErrorFile(const char* msg) {
>+  if (msg != nullptr) {
>+    ABSL_RAW_CHECK(fwrite(msg, strlen(msg), 1, error_file) == 1,
>+                   "fwrite() failed");
>+  }
>+  ABSL_RAW_CHECK(fflush(error_file) == 0, "fflush() failed");
>+}
>+
>+std::string GetTmpDir() {
>+  // TEST_TMPDIR is set by Bazel. Try the others when not running under Bazel.
>+  static const char* const kTmpEnvVars[] = {"TEST_TMPDIR", "TMPDIR", "TEMP",
>+                                            "TEMPDIR", "TMP"};
>+  for (const char* const var : kTmpEnvVars) {
>+    const char* tmp_dir = std::getenv(var);
>+    if (tmp_dir != nullptr) {
>+      return tmp_dir;
>+    }
>+  }
>+
>+  // Try something reasonable.
>+  return "/tmp";
>+}
>+
>+// This function runs in a fork()ed process on most systems.
>+void InstallHandlerWithWriteToFileAndRaise(const char* file, int signo) {
>+  error_file = fopen(file, "w");
>+  ABSL_RAW_CHECK(error_file != nullptr, "Failed create error_file");
>+  absl::FailureSignalHandlerOptions options;
>+  options.writerfn = WriteToErrorFile;
>+  absl::InstallFailureSignalHandler(options);
>+  raise(signo);
>+}
>+
>+TEST_P(FailureSignalHandlerDeathTest, AbslFatalSignalsWithWriterFn) {
>+  const int signo = GetParam();
>+  std::string tmp_dir = GetTmpDir();
>+  std::string file = absl::StrCat(tmp_dir, "/signo_", signo);
>+
>+  std::string exit_regex = absl::StrCat(
>+      "\\*\\*\\* ", absl::debugging_internal::FailureSignalToString(signo),
>+      " received at time=");
>+#ifndef _WIN32
>+  EXPECT_EXIT(InstallHandlerWithWriteToFileAndRaise(file.c_str(), signo),
>+              testing::KilledBySignal(signo), exit_regex);
>+#else
>+  // Windows doesn't have testing::KilledBySignal().
>+  EXPECT_DEATH(InstallHandlerWithWriteToFileAndRaise(file.c_str(), signo),
>+               exit_regex);
>+#endif
>+
>+  // Open the file in this process and check its contents.
>+  std::fstream error_output(file);
>+  ASSERT_TRUE(error_output.is_open()) << file;
>+  std::string error_line;
>+  std::getline(error_output, error_line);
>+  EXPECT_TRUE(absl::StartsWith(
>+      error_line,
>+      absl::StrCat("*** ",
>+                   absl::debugging_internal::FailureSignalToString(signo),
>+                   " received at ")));
>+
>+  if (absl::debugging_internal::StackTraceWorksForTest()) {
>+    std::getline(error_output, error_line);
>+    EXPECT_TRUE(absl::StartsWith(error_line, "PC: "));
>+  }
>+}
>+
>+constexpr int kFailureSignals[] = {
>+    SIGSEGV, SIGILL,  SIGFPE, SIGABRT, SIGTERM,
>+#ifndef _WIN32
>+    SIGBUS,  SIGTRAP,
>+#endif
>+};
>+
>+std::string SignalParamToString(const ::testing::TestParamInfo<int>& info) {
>+  std::string result = absl::debugging_internal::FailureSignalToString(info.param);
>+  if (result.empty()) {
>+    result = absl::StrCat(info.param);
>+  }
>+  return result;
>+}
>+
>+INSTANTIATE_TEST_CASE_P(AbslDeathTest, FailureSignalHandlerDeathTest,
>+                        ::testing::ValuesIn(kFailureSignals),
>+                        SignalParamToString);
>+
>+#endif  // GTEST_HAS_DEATH_TEST
>+
>+}  // namespace
>+
>+int main(int argc, char** argv) {
>+  absl::InitializeSymbolizer(argv[0]);
>+  testing::InitGoogleTest(&argc, argv);
>+  return RUN_ALL_TESTS();
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/address_is_readable.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/address_is_readable.cc
>new file mode 100644
>index 00000000000..7455aa0b416
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/address_is_readable.cc
>@@ -0,0 +1,133 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+// base::AddressIsReadable() probes an address to see whether it is readable,
>+// without faulting.
>+
>+#include "absl/debugging/internal/address_is_readable.h"
>+
>+#if !defined(__linux__) || defined(__ANDROID__)
>+
>+namespace absl {
>+namespace debugging_internal {
>+
>+// On platforms other than Linux, just return true.
>+bool AddressIsReadable(const void* /* addr */) { return true; }
>+
>+}  // namespace debugging_internal
>+}  // namespace absl
>+
>+#else
>+
>+#include <fcntl.h>
>+#include <sys/syscall.h>
>+#include <unistd.h>
>+#include <atomic>
>+#include <cerrno>
>+#include <cstdint>
>+
>+#include "absl/base/internal/raw_logging.h"
>+
>+namespace absl {
>+namespace debugging_internal {
>+
>+// Pack a pid and two file descriptors into a 64-bit word,
>+// using 16, 24, and 24 bits for each respectively.
>+static uint64_t Pack(uint64_t pid, uint64_t read_fd, uint64_t write_fd) {
>+  ABSL_RAW_CHECK((read_fd >> 24) == 0 && (write_fd >> 24) == 0,
>+                 "fd out of range");
>+  return (pid << 48) | ((read_fd & 0xffffff) << 24) | (write_fd & 0xffffff);
>+}
>+
>+// Unpack x into a pid and two file descriptors, where x was created with
>+// Pack().
>+static void Unpack(uint64_t x, int *pid, int *read_fd, int *write_fd) {
>+  *pid = x >> 48;
>+  *read_fd = (x >> 24) & 0xffffff;
>+  *write_fd = x & 0xffffff;
>+}
>+
>+// Return whether the byte at *addr is readable, without faulting.
>+// Save and restores errno.   Returns true on systems where
>+// unimplemented.
>+// This is a namespace-scoped variable for correct zero-initialization.
>+static std::atomic<uint64_t> pid_and_fds;  // initially 0, an invalid pid.
>+bool AddressIsReadable(const void *addr) {
>+  int save_errno = errno;
>+  // We test whether a byte is readable by using write().  Normally, this would
>+  // be done via a cached file descriptor to /dev/null, but linux fails to
>+  // check whether the byte is readable when the destination is /dev/null, so
>+  // we use a cached pipe.  We store the pid of the process that created the
>+  // pipe to handle the case where a process forks, and the child closes all
>+  // the file descriptors and then calls this routine.  This is not perfect:
>+  // the child could use the routine, then close all file descriptors and then
>+  // use this routine again.  But the likely use of this routine is when
>+  // crashing, to test the validity of pages when dumping the stack.  Beware
>+  // that we may leak file descriptors, but we're unlikely to leak many.
>+  int bytes_written;
>+  int current_pid = getpid() & 0xffff;   // we use only the low order 16 bits
>+  do {  // until we do not get EBADF trying to use file descriptors
>+    int pid;
>+    int read_fd;
>+    int write_fd;
>+    uint64_t local_pid_and_fds = pid_and_fds.load(std::memory_order_relaxed);
>+    Unpack(local_pid_and_fds, &pid, &read_fd, &write_fd);
>+    while (current_pid != pid) {
>+      int p[2];
>+      // new pipe
>+      if (pipe(p) != 0) {
>+        ABSL_RAW_LOG(FATAL, "Failed to create pipe, errno=%d", errno);
>+      }
>+      fcntl(p[0], F_SETFD, FD_CLOEXEC);
>+      fcntl(p[1], F_SETFD, FD_CLOEXEC);
>+      uint64_t new_pid_and_fds = Pack(current_pid, p[0], p[1]);
>+      if (pid_and_fds.compare_exchange_strong(
>+              local_pid_and_fds, new_pid_and_fds, std::memory_order_relaxed,
>+              std::memory_order_relaxed)) {
>+        local_pid_and_fds = new_pid_and_fds;  // fds exposed to other threads
>+      } else {  // fds not exposed to other threads; we can close them.
>+        close(p[0]);
>+        close(p[1]);
>+        local_pid_and_fds = pid_and_fds.load(std::memory_order_relaxed);
>+      }
>+      Unpack(local_pid_and_fds, &pid, &read_fd, &write_fd);
>+    }
>+    errno = 0;
>+    // Use syscall(SYS_write, ...) instead of write() to prevent ASAN
>+    // and other checkers from complaining about accesses to arbitrary
>+    // memory.
>+    do {
>+      bytes_written = syscall(SYS_write, write_fd, addr, 1);
>+    } while (bytes_written == -1 && errno == EINTR);
>+    if (bytes_written == 1) {   // remove the byte from the pipe
>+      char c;
>+      while (read(read_fd, &c, 1) == -1 && errno == EINTR) {
>+      }
>+    }
>+    if (errno == EBADF) {  // Descriptors invalid.
>+      // If pid_and_fds contains the problematic file descriptors we just used,
>+      // this call will forget them, and the loop will try again.
>+      pid_and_fds.compare_exchange_strong(local_pid_and_fds, 0,
>+                                          std::memory_order_relaxed,
>+                                          std::memory_order_relaxed);
>+    }
>+  } while (errno == EBADF);
>+  errno = save_errno;
>+  return bytes_written == 1;
>+}
>+
>+}  // namespace debugging_internal
>+}  // namespace absl
>+
>+#endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/address_is_readable.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/address_is_readable.h
>new file mode 100644
>index 00000000000..9d48030065c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/address_is_readable.h
>@@ -0,0 +1,29 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+
>+#ifndef ABSL_DEBUGGING_INTERNAL_ADDRESS_IS_READABLE_H_
>+#define ABSL_DEBUGGING_INTERNAL_ADDRESS_IS_READABLE_H_
>+
>+namespace absl {
>+namespace debugging_internal {
>+
>+// Return whether the byte at *addr is readable, without faulting.
>+// Save and restores errno.
>+bool AddressIsReadable(const void *addr);
>+
>+}  // namespace debugging_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_DEBUGGING_INTERNAL_ADDRESS_IS_READABLE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/demangle.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/demangle.cc
>new file mode 100644
>index 00000000000..c9ca2f3bdb1
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/demangle.cc
>@@ -0,0 +1,1862 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+// For reference check out:
>+// https://itanium-cxx-abi.github.io/cxx-abi/abi.html#mangling
>+//
>+// Note that we only have partial C++11 support yet.
>+
>+#include "absl/debugging/internal/demangle.h"
>+
>+#include <cstdint>
>+#include <cstdio>
>+#include <limits>
>+
>+namespace absl {
>+namespace debugging_internal {
>+
>+typedef struct {
>+  const char *abbrev;
>+  const char *real_name;
>+  // Number of arguments in <expression> context, or 0 if disallowed.
>+  int arity;
>+} AbbrevPair;
>+
>+// List of operators from Itanium C++ ABI.
>+static const AbbrevPair kOperatorList[] = {
>+    // New has special syntax (not currently supported).
>+    {"nw", "new", 0},
>+    {"na", "new[]", 0},
>+
>+    // Works except that the 'gs' prefix is not supported.
>+    {"dl", "delete", 1},
>+    {"da", "delete[]", 1},
>+
>+    {"ps", "+", 1},  // "positive"
>+    {"ng", "-", 1},  // "negative"
>+    {"ad", "&", 1},  // "address-of"
>+    {"de", "*", 1},  // "dereference"
>+    {"co", "~", 1},
>+
>+    {"pl", "+", 2},
>+    {"mi", "-", 2},
>+    {"ml", "*", 2},
>+    {"dv", "/", 2},
>+    {"rm", "%", 2},
>+    {"an", "&", 2},
>+    {"or", "|", 2},
>+    {"eo", "^", 2},
>+    {"aS", "=", 2},
>+    {"pL", "+=", 2},
>+    {"mI", "-=", 2},
>+    {"mL", "*=", 2},
>+    {"dV", "/=", 2},
>+    {"rM", "%=", 2},
>+    {"aN", "&=", 2},
>+    {"oR", "|=", 2},
>+    {"eO", "^=", 2},
>+    {"ls", "<<", 2},
>+    {"rs", ">>", 2},
>+    {"lS", "<<=", 2},
>+    {"rS", ">>=", 2},
>+    {"eq", "==", 2},
>+    {"ne", "!=", 2},
>+    {"lt", "<", 2},
>+    {"gt", ">", 2},
>+    {"le", "<=", 2},
>+    {"ge", ">=", 2},
>+    {"nt", "!", 1},
>+    {"aa", "&&", 2},
>+    {"oo", "||", 2},
>+    {"pp", "++", 1},
>+    {"mm", "--", 1},
>+    {"cm", ",", 2},
>+    {"pm", "->*", 2},
>+    {"pt", "->", 0},  // Special syntax
>+    {"cl", "()", 0},  // Special syntax
>+    {"ix", "[]", 2},
>+    {"qu", "?", 3},
>+    {"st", "sizeof", 0},  // Special syntax
>+    {"sz", "sizeof", 1},  // Not a real operator name, but used in expressions.
>+    {nullptr, nullptr, 0},
>+};
>+
>+// List of builtin types from Itanium C++ ABI.
>+static const AbbrevPair kBuiltinTypeList[] = {
>+    {"v", "void", 0},
>+    {"w", "wchar_t", 0},
>+    {"b", "bool", 0},
>+    {"c", "char", 0},
>+    {"a", "signed char", 0},
>+    {"h", "unsigned char", 0},
>+    {"s", "short", 0},
>+    {"t", "unsigned short", 0},
>+    {"i", "int", 0},
>+    {"j", "unsigned int", 0},
>+    {"l", "long", 0},
>+    {"m", "unsigned long", 0},
>+    {"x", "long long", 0},
>+    {"y", "unsigned long long", 0},
>+    {"n", "__int128", 0},
>+    {"o", "unsigned __int128", 0},
>+    {"f", "float", 0},
>+    {"d", "double", 0},
>+    {"e", "long double", 0},
>+    {"g", "__float128", 0},
>+    {"z", "ellipsis", 0},
>+    {nullptr, nullptr, 0},
>+};
>+
>+// List of substitutions Itanium C++ ABI.
>+static const AbbrevPair kSubstitutionList[] = {
>+    {"St", "", 0},
>+    {"Sa", "allocator", 0},
>+    {"Sb", "basic_string", 0},
>+    // std::basic_string<char, std::char_traits<char>,std::allocator<char> >
>+    {"Ss", "string", 0},
>+    // std::basic_istream<char, std::char_traits<char> >
>+    {"Si", "istream", 0},
>+    // std::basic_ostream<char, std::char_traits<char> >
>+    {"So", "ostream", 0},
>+    // std::basic_iostream<char, std::char_traits<char> >
>+    {"Sd", "iostream", 0},
>+    {nullptr, nullptr, 0},
>+};
>+
>+// State needed for demangling.  This struct is copied in almost every stack
>+// frame, so every byte counts.
>+typedef struct {
>+  int mangled_idx;                   // Cursor of mangled name.
>+  int out_cur_idx;                   // Cursor of output std::string.
>+  int prev_name_idx;                 // For constructors/destructors.
>+  signed int prev_name_length : 16;  // For constructors/destructors.
>+  signed int nest_level : 15;        // For nested names.
>+  unsigned int append : 1;           // Append flag.
>+  // Note: for some reason MSVC can't pack "bool append : 1" into the same int
>+  // with the above two fields, so we use an int instead.  Amusingly it can pack
>+  // "signed bool" as expected, but relying on that to continue to be a legal
>+  // type seems ill-advised (as it's illegal in at least clang).
>+} ParseState;
>+
>+static_assert(sizeof(ParseState) == 4 * sizeof(int),
>+              "unexpected size of ParseState");
>+
>+// One-off state for demangling that's not subject to backtracking -- either
>+// constant data, data that's intentionally immune to backtracking (steps), or
>+// data that would never be changed by backtracking anyway (recursion_depth).
>+//
>+// Only one copy of this exists for each call to Demangle, so the size of this
>+// struct is nearly inconsequential.
>+typedef struct {
>+  const char *mangled_begin;  // Beginning of input std::string.
>+  char *out;                  // Beginning of output std::string.
>+  int out_end_idx;            // One past last allowed output character.
>+  int recursion_depth;        // For stack exhaustion prevention.
>+  int steps;               // Cap how much work we'll do, regardless of depth.
>+  ParseState parse_state;  // Backtrackable state copied for most frames.
>+} State;
>+
>+namespace {
>+// Prevent deep recursion / stack exhaustion.
>+// Also prevent unbounded handling of complex inputs.
>+class ComplexityGuard {
>+ public:
>+  explicit ComplexityGuard(State *state) : state_(state) {
>+    ++state->recursion_depth;
>+    ++state->steps;
>+  }
>+  ~ComplexityGuard() { --state_->recursion_depth; }
>+
>+  // 256 levels of recursion seems like a reasonable upper limit on depth.
>+  // 128 is not enough to demagle synthetic tests from demangle_unittest.txt:
>+  // "_ZaaZZZZ..." and "_ZaaZcvZcvZ..."
>+  static constexpr int kRecursionDepthLimit = 256;
>+
>+  // We're trying to pick a charitable upper-limit on how many parse steps are
>+  // necessary to handle something that a human could actually make use of.
>+  // This is mostly in place as a bound on how much work we'll do if we are
>+  // asked to demangle an mangled name from an untrusted source, so it should be
>+  // much larger than the largest expected symbol, but much smaller than the
>+  // amount of work we can do in, e.g., a second.
>+  //
>+  // Some real-world symbols from an arbitrary binary started failing between
>+  // 2^12 and 2^13, so we multiply the latter by an extra factor of 16 to set
>+  // the limit.
>+  //
>+  // Spending one second on 2^17 parse steps would require each step to take
>+  // 7.6us, or ~30000 clock cycles, so it's safe to say this can be done in
>+  // under a second.
>+  static constexpr int kParseStepsLimit = 1 << 17;
>+
>+  bool IsTooComplex() const {
>+    return state_->recursion_depth > kRecursionDepthLimit ||
>+           state_->steps > kParseStepsLimit;
>+  }
>+
>+ private:
>+  State *state_;
>+};
>+}  // namespace
>+
>+// We don't use strlen() in libc since it's not guaranteed to be async
>+// signal safe.
>+static size_t StrLen(const char *str) {
>+  size_t len = 0;
>+  while (*str != '\0') {
>+    ++str;
>+    ++len;
>+  }
>+  return len;
>+}
>+
>+// Returns true if "str" has at least "n" characters remaining.
>+static bool AtLeastNumCharsRemaining(const char *str, int n) {
>+  for (int i = 0; i < n; ++i) {
>+    if (str[i] == '\0') {
>+      return false;
>+    }
>+  }
>+  return true;
>+}
>+
>+// Returns true if "str" has "prefix" as a prefix.
>+static bool StrPrefix(const char *str, const char *prefix) {
>+  size_t i = 0;
>+  while (str[i] != '\0' && prefix[i] != '\0' && str[i] == prefix[i]) {
>+    ++i;
>+  }
>+  return prefix[i] == '\0';  // Consumed everything in "prefix".
>+}
>+
>+static void InitState(State *state, const char *mangled, char *out,
>+                      int out_size) {
>+  state->mangled_begin = mangled;
>+  state->out = out;
>+  state->out_end_idx = out_size;
>+  state->recursion_depth = 0;
>+  state->steps = 0;
>+
>+  state->parse_state.mangled_idx = 0;
>+  state->parse_state.out_cur_idx = 0;
>+  state->parse_state.prev_name_idx = 0;
>+  state->parse_state.prev_name_length = -1;
>+  state->parse_state.nest_level = -1;
>+  state->parse_state.append = true;
>+}
>+
>+static inline const char *RemainingInput(State *state) {
>+  return &state->mangled_begin[state->parse_state.mangled_idx];
>+}
>+
>+// Returns true and advances "mangled_idx" if we find "one_char_token"
>+// at "mangled_idx" position.  It is assumed that "one_char_token" does
>+// not contain '\0'.
>+static bool ParseOneCharToken(State *state, const char one_char_token) {
>+  ComplexityGuard guard(state);
>+  if (guard.IsTooComplex()) return false;
>+  if (RemainingInput(state)[0] == one_char_token) {
>+    ++state->parse_state.mangled_idx;
>+    return true;
>+  }
>+  return false;
>+}
>+
>+// Returns true and advances "mangled_cur" if we find "two_char_token"
>+// at "mangled_cur" position.  It is assumed that "two_char_token" does
>+// not contain '\0'.
>+static bool ParseTwoCharToken(State *state, const char *two_char_token) {
>+  ComplexityGuard guard(state);
>+  if (guard.IsTooComplex()) return false;
>+  if (RemainingInput(state)[0] == two_char_token[0] &&
>+      RemainingInput(state)[1] == two_char_token[1]) {
>+    state->parse_state.mangled_idx += 2;
>+    return true;
>+  }
>+  return false;
>+}
>+
>+// Returns true and advances "mangled_cur" if we find any character in
>+// "char_class" at "mangled_cur" position.
>+static bool ParseCharClass(State *state, const char *char_class) {
>+  ComplexityGuard guard(state);
>+  if (guard.IsTooComplex()) return false;
>+  if (RemainingInput(state)[0] == '\0') {
>+    return false;
>+  }
>+  const char *p = char_class;
>+  for (; *p != '\0'; ++p) {
>+    if (RemainingInput(state)[0] == *p) {
>+      ++state->parse_state.mangled_idx;
>+      return true;
>+    }
>+  }
>+  return false;
>+}
>+
>+static bool ParseDigit(State *state, int *digit) {
>+  char c = RemainingInput(state)[0];
>+  if (ParseCharClass(state, "0123456789")) {
>+    if (digit != nullptr) {
>+      *digit = c - '0';
>+    }
>+    return true;
>+  }
>+  return false;
>+}
>+
>+// This function is used for handling an optional non-terminal.
>+static bool Optional(bool /*status*/) { return true; }
>+
>+// This function is used for handling <non-terminal>+ syntax.
>+typedef bool (*ParseFunc)(State *);
>+static bool OneOrMore(ParseFunc parse_func, State *state) {
>+  if (parse_func(state)) {
>+    while (parse_func(state)) {
>+    }
>+    return true;
>+  }
>+  return false;
>+}
>+
>+// This function is used for handling <non-terminal>* syntax. The function
>+// always returns true and must be followed by a termination token or a
>+// terminating sequence not handled by parse_func (e.g.
>+// ParseOneCharToken(state, 'E')).
>+static bool ZeroOrMore(ParseFunc parse_func, State *state) {
>+  while (parse_func(state)) {
>+  }
>+  return true;
>+}
>+
>+// Append "str" at "out_cur_idx".  If there is an overflow, out_cur_idx is
>+// set to out_end_idx+1.  The output std::string is ensured to
>+// always terminate with '\0' as long as there is no overflow.
>+static void Append(State *state, const char *const str, const int length) {
>+  for (int i = 0; i < length; ++i) {
>+    if (state->parse_state.out_cur_idx + 1 <
>+        state->out_end_idx) {  // +1 for '\0'
>+      state->out[state->parse_state.out_cur_idx++] = str[i];
>+    } else {
>+      // signal overflow
>+      state->parse_state.out_cur_idx = state->out_end_idx + 1;
>+      break;
>+    }
>+  }
>+  if (state->parse_state.out_cur_idx < state->out_end_idx) {
>+    state->out[state->parse_state.out_cur_idx] =
>+        '\0';  // Terminate it with '\0'
>+  }
>+}
>+
>+// We don't use equivalents in libc to avoid locale issues.
>+static bool IsLower(char c) { return c >= 'a' && c <= 'z'; }
>+
>+static bool IsAlpha(char c) {
>+  return (c >= 'a' && c <= 'z') || (c >= 'A' && c <= 'Z');
>+}
>+
>+static bool IsDigit(char c) { return c >= '0' && c <= '9'; }
>+
>+// Returns true if "str" is a function clone suffix.  These suffixes are used
>+// by GCC 4.5.x and later versions (and our locally-modified version of GCC
>+// 4.4.x) to indicate functions which have been cloned during optimization.
>+// We treat any sequence (.<alpha>+.<digit>+)+ as a function clone suffix.
>+static bool IsFunctionCloneSuffix(const char *str) {
>+  size_t i = 0;
>+  while (str[i] != '\0') {
>+    // Consume a single .<alpha>+.<digit>+ sequence.
>+    if (str[i] != '.' || !IsAlpha(str[i + 1])) {
>+      return false;
>+    }
>+    i += 2;
>+    while (IsAlpha(str[i])) {
>+      ++i;
>+    }
>+    if (str[i] != '.' || !IsDigit(str[i + 1])) {
>+      return false;
>+    }
>+    i += 2;
>+    while (IsDigit(str[i])) {
>+      ++i;
>+    }
>+  }
>+  return true;  // Consumed everything in "str".
>+}
>+
>+static bool EndsWith(State *state, const char chr) {
>+  return state->parse_state.out_cur_idx > 0 &&
>+         chr == state->out[state->parse_state.out_cur_idx - 1];
>+}
>+
>+// Append "str" with some tweaks, iff "append" state is true.
>+static void MaybeAppendWithLength(State *state, const char *const str,
>+                                  const int length) {
>+  if (state->parse_state.append && length > 0) {
>+    // Append a space if the output buffer ends with '<' and "str"
>+    // starts with '<' to avoid <<<.
>+    if (str[0] == '<' && EndsWith(state, '<')) {
>+      Append(state, " ", 1);
>+    }
>+    // Remember the last identifier name for ctors/dtors.
>+    if (IsAlpha(str[0]) || str[0] == '_') {
>+      state->parse_state.prev_name_idx = state->parse_state.out_cur_idx;
>+      state->parse_state.prev_name_length = length;
>+    }
>+    Append(state, str, length);
>+  }
>+}
>+
>+// Appends a positive decimal number to the output if appending is enabled.
>+static bool MaybeAppendDecimal(State *state, unsigned int val) {
>+  // Max {32-64}-bit unsigned int is 20 digits.
>+  constexpr size_t kMaxLength = 20;
>+  char buf[kMaxLength];
>+
>+  // We can't use itoa or sprintf as neither is specified to be
>+  // async-signal-safe.
>+  if (state->parse_state.append) {
>+    // We can't have a one-before-the-beginning pointer, so instead start with
>+    // one-past-the-end and manipulate one character before the pointer.
>+    char *p = &buf[kMaxLength];
>+    do {  // val=0 is the only input that should write a leading zero digit.
>+      *--p = (val % 10) + '0';
>+      val /= 10;
>+    } while (p > buf && val != 0);
>+
>+    // 'p' landed on the last character we set.  How convenient.
>+    Append(state, p, kMaxLength - (p - buf));
>+  }
>+
>+  return true;
>+}
>+
>+// A convenient wrapper around MaybeAppendWithLength().
>+// Returns true so that it can be placed in "if" conditions.
>+static bool MaybeAppend(State *state, const char *const str) {
>+  if (state->parse_state.append) {
>+    int length = StrLen(str);
>+    MaybeAppendWithLength(state, str, length);
>+  }
>+  return true;
>+}
>+
>+// This function is used for handling nested names.
>+static bool EnterNestedName(State *state) {
>+  state->parse_state.nest_level = 0;
>+  return true;
>+}
>+
>+// This function is used for handling nested names.
>+static bool LeaveNestedName(State *state, int16_t prev_value) {
>+  state->parse_state.nest_level = prev_value;
>+  return true;
>+}
>+
>+// Disable the append mode not to print function parameters, etc.
>+static bool DisableAppend(State *state) {
>+  state->parse_state.append = false;
>+  return true;
>+}
>+
>+// Restore the append mode to the previous state.
>+static bool RestoreAppend(State *state, bool prev_value) {
>+  state->parse_state.append = prev_value;
>+  return true;
>+}
>+
>+// Increase the nest level for nested names.
>+static void MaybeIncreaseNestLevel(State *state) {
>+  if (state->parse_state.nest_level > -1) {
>+    ++state->parse_state.nest_level;
>+  }
>+}
>+
>+// Appends :: for nested names if necessary.
>+static void MaybeAppendSeparator(State *state) {
>+  if (state->parse_state.nest_level >= 1) {
>+    MaybeAppend(state, "::");
>+  }
>+}
>+
>+// Cancel the last separator if necessary.
>+static void MaybeCancelLastSeparator(State *state) {
>+  if (state->parse_state.nest_level >= 1 && state->parse_state.append &&
>+      state->parse_state.out_cur_idx >= 2) {
>+    state->parse_state.out_cur_idx -= 2;
>+    state->out[state->parse_state.out_cur_idx] = '\0';
>+  }
>+}
>+
>+// Returns true if the identifier of the given length pointed to by
>+// "mangled_cur" is anonymous namespace.
>+static bool IdentifierIsAnonymousNamespace(State *state, int length) {
>+  // Returns true if "anon_prefix" is a proper prefix of "mangled_cur".
>+  static const char anon_prefix[] = "_GLOBAL__N_";
>+  return (length > static_cast<int>(sizeof(anon_prefix) - 1) &&
>+          StrPrefix(RemainingInput(state), anon_prefix));
>+}
>+
>+// Forward declarations of our parsing functions.
>+static bool ParseMangledName(State *state);
>+static bool ParseEncoding(State *state);
>+static bool ParseName(State *state);
>+static bool ParseUnscopedName(State *state);
>+static bool ParseNestedName(State *state);
>+static bool ParsePrefix(State *state);
>+static bool ParseUnqualifiedName(State *state);
>+static bool ParseSourceName(State *state);
>+static bool ParseLocalSourceName(State *state);
>+static bool ParseUnnamedTypeName(State *state);
>+static bool ParseNumber(State *state, int *number_out);
>+static bool ParseFloatNumber(State *state);
>+static bool ParseSeqId(State *state);
>+static bool ParseIdentifier(State *state, int length);
>+static bool ParseOperatorName(State *state, int *arity);
>+static bool ParseSpecialName(State *state);
>+static bool ParseCallOffset(State *state);
>+static bool ParseNVOffset(State *state);
>+static bool ParseVOffset(State *state);
>+static bool ParseCtorDtorName(State *state);
>+static bool ParseDecltype(State *state);
>+static bool ParseType(State *state);
>+static bool ParseCVQualifiers(State *state);
>+static bool ParseBuiltinType(State *state);
>+static bool ParseFunctionType(State *state);
>+static bool ParseBareFunctionType(State *state);
>+static bool ParseClassEnumType(State *state);
>+static bool ParseArrayType(State *state);
>+static bool ParsePointerToMemberType(State *state);
>+static bool ParseTemplateParam(State *state);
>+static bool ParseTemplateTemplateParam(State *state);
>+static bool ParseTemplateArgs(State *state);
>+static bool ParseTemplateArg(State *state);
>+static bool ParseBaseUnresolvedName(State *state);
>+static bool ParseUnresolvedName(State *state);
>+static bool ParseExpression(State *state);
>+static bool ParseExprPrimary(State *state);
>+static bool ParseExprCastValue(State *state);
>+static bool ParseLocalName(State *state);
>+static bool ParseLocalNameSuffix(State *state);
>+static bool ParseDiscriminator(State *state);
>+static bool ParseSubstitution(State *state, bool accept_std);
>+
>+// Implementation note: the following code is a straightforward
>+// translation of the Itanium C++ ABI defined in BNF with a couple of
>+// exceptions.
>+//
>+// - Support GNU extensions not defined in the Itanium C++ ABI
>+// - <prefix> and <template-prefix> are combined to avoid infinite loop
>+// - Reorder patterns to shorten the code
>+// - Reorder patterns to give greedier functions precedence
>+//   We'll mark "Less greedy than" for these cases in the code
>+//
>+// Each parsing function changes the parse state and returns true on
>+// success, or returns false and doesn't change the parse state (note:
>+// the parse-steps counter increases regardless of success or failure).
>+// To ensure that the parse state isn't changed in the latter case, we
>+// save the original state before we call multiple parsing functions
>+// consecutively with &&, and restore it if unsuccessful.  See
>+// ParseEncoding() as an example of this convention.  We follow the
>+// convention throughout the code.
>+//
>+// Originally we tried to do demangling without following the full ABI
>+// syntax but it turned out we needed to follow the full syntax to
>+// parse complicated cases like nested template arguments.  Note that
>+// implementing a full-fledged demangler isn't trivial (libiberty's
>+// cp-demangle.c has +4300 lines).
>+//
>+// Note that (foo) in <(foo) ...> is a modifier to be ignored.
>+//
>+// Reference:
>+// - Itanium C++ ABI
>+//   <https://mentorembedded.github.io/cxx-abi/abi.html#mangling>
>+
>+// <mangled-name> ::= _Z <encoding>
>+static bool ParseMangledName(State *state) {
>+  ComplexityGuard guard(state);
>+  if (guard.IsTooComplex()) return false;
>+  return ParseTwoCharToken(state, "_Z") && ParseEncoding(state);
>+}
>+
>+// <encoding> ::= <(function) name> <bare-function-type>
>+//            ::= <(data) name>
>+//            ::= <special-name>
>+static bool ParseEncoding(State *state) {
>+  ComplexityGuard guard(state);
>+  if (guard.IsTooComplex()) return false;
>+  // Implementing the first two productions together as <name>
>+  // [<bare-function-type>] avoids exponential blowup of backtracking.
>+  //
>+  // Since Optional(...) can't fail, there's no need to copy the state for
>+  // backtracking.
>+  if (ParseName(state) && Optional(ParseBareFunctionType(state))) {
>+    return true;
>+  }
>+
>+  if (ParseSpecialName(state)) {
>+    return true;
>+  }
>+  return false;
>+}
>+
>+// <name> ::= <nested-name>
>+//        ::= <unscoped-template-name> <template-args>
>+//        ::= <unscoped-name>
>+//        ::= <local-name>
>+static bool ParseName(State *state) {
>+  ComplexityGuard guard(state);
>+  if (guard.IsTooComplex()) return false;
>+  if (ParseNestedName(state) || ParseLocalName(state)) {
>+    return true;
>+  }
>+
>+  // We reorganize the productions to avoid re-parsing unscoped names.
>+  // - Inline <unscoped-template-name> productions:
>+  //   <name> ::= <substitution> <template-args>
>+  //          ::= <unscoped-name> <template-args>
>+  //          ::= <unscoped-name>
>+  // - Merge the two productions that start with unscoped-name:
>+  //   <name> ::= <unscoped-name> [<template-args>]
>+
>+  ParseState copy = state->parse_state;
>+  // "std<...>" isn't a valid name.
>+  if (ParseSubstitution(state, /*accept_std=*/false) &&
>+      ParseTemplateArgs(state)) {
>+    return true;
>+  }
>+  state->parse_state = copy;
>+
>+  // Note there's no need to restore state after this since only the first
>+  // subparser can fail.
>+  return ParseUnscopedName(state) && Optional(ParseTemplateArgs(state));
>+}
>+
>+// <unscoped-name> ::= <unqualified-name>
>+//                 ::= St <unqualified-name>
>+static bool ParseUnscopedName(State *state) {
>+  ComplexityGuard guard(state);
>+  if (guard.IsTooComplex()) return false;
>+  if (ParseUnqualifiedName(state)) {
>+    return true;
>+  }
>+
>+  ParseState copy = state->parse_state;
>+  if (ParseTwoCharToken(state, "St") && MaybeAppend(state, "std::") &&
>+      ParseUnqualifiedName(state)) {
>+    return true;
>+  }
>+  state->parse_state = copy;
>+  return false;
>+}
>+
>+// <ref-qualifer> ::= R // lvalue method reference qualifier
>+//                ::= O // rvalue method reference qualifier
>+static inline bool ParseRefQualifier(State *state) {
>+  return ParseCharClass(state, "OR");
>+}
>+
>+// <nested-name> ::= N [<CV-qualifiers>] [<ref-qualifier>] <prefix>
>+//                   <unqualified-name> E
>+//               ::= N [<CV-qualifiers>] [<ref-qualifier>] <template-prefix>
>+//                   <template-args> E
>+static bool ParseNestedName(State *state) {
>+  ComplexityGuard guard(state);
>+  if (guard.IsTooComplex()) return false;
>+  ParseState copy = state->parse_state;
>+  if (ParseOneCharToken(state, 'N') && EnterNestedName(state) &&
>+      Optional(ParseCVQualifiers(state)) &&
>+      Optional(ParseRefQualifier(state)) && ParsePrefix(state) &&
>+      LeaveNestedName(state, copy.nest_level) &&
>+      ParseOneCharToken(state, 'E')) {
>+    return true;
>+  }
>+  state->parse_state = copy;
>+  return false;
>+}
>+
>+// This part is tricky.  If we literally translate them to code, we'll
>+// end up infinite loop.  Hence we merge them to avoid the case.
>+//
>+// <prefix> ::= <prefix> <unqualified-name>
>+//          ::= <template-prefix> <template-args>
>+//          ::= <template-param>
>+//          ::= <substitution>
>+//          ::= # empty
>+// <template-prefix> ::= <prefix> <(template) unqualified-name>
>+//                   ::= <template-param>
>+//                   ::= <substitution>
>+static bool ParsePrefix(State *state) {
>+  ComplexityGuard guard(state);
>+  if (guard.IsTooComplex()) return false;
>+  bool has_something = false;
>+  while (true) {
>+    MaybeAppendSeparator(state);
>+    if (ParseTemplateParam(state) ||
>+        ParseSubstitution(state, /*accept_std=*/true) ||
>+        ParseUnscopedName(state) ||
>+        (ParseOneCharToken(state, 'M') && ParseUnnamedTypeName(state))) {
>+      has_something = true;
>+      MaybeIncreaseNestLevel(state);
>+      continue;
>+    }
>+    MaybeCancelLastSeparator(state);
>+    if (has_something && ParseTemplateArgs(state)) {
>+      return ParsePrefix(state);
>+    } else {
>+      break;
>+    }
>+  }
>+  return true;
>+}
>+
>+// <unqualified-name> ::= <operator-name>
>+//                    ::= <ctor-dtor-name>
>+//                    ::= <source-name>
>+//                    ::= <local-source-name> // GCC extension; see below.
>+//                    ::= <unnamed-type-name>
>+static bool ParseUnqualifiedName(State *state) {
>+  ComplexityGuard guard(state);
>+  if (guard.IsTooComplex()) return false;
>+  return (ParseOperatorName(state, nullptr) || ParseCtorDtorName(state) ||
>+          ParseSourceName(state) || ParseLocalSourceName(state) ||
>+          ParseUnnamedTypeName(state));
>+}
>+
>+// <source-name> ::= <positive length number> <identifier>
>+static bool ParseSourceName(State *state) {
>+  ComplexityGuard guard(state);
>+  if (guard.IsTooComplex()) return false;
>+  ParseState copy = state->parse_state;
>+  int length = -1;
>+  if (ParseNumber(state, &length) && ParseIdentifier(state, length)) {
>+    return true;
>+  }
>+  state->parse_state = copy;
>+  return false;
>+}
>+
>+// <local-source-name> ::= L <source-name> [<discriminator>]
>+//
>+// References:
>+//   http://gcc.gnu.org/bugzilla/show_bug.cgi?id=31775
>+//   http://gcc.gnu.org/viewcvs?view=rev&revision=124467
>+static bool ParseLocalSourceName(State *state) {
>+  ComplexityGuard guard(state);
>+  if (guard.IsTooComplex()) return false;
>+  ParseState copy = state->parse_state;
>+  if (ParseOneCharToken(state, 'L') && ParseSourceName(state) &&
>+      Optional(ParseDiscriminator(state))) {
>+    return true;
>+  }
>+  state->parse_state = copy;
>+  return false;
>+}
>+
>+// <unnamed-type-name> ::= Ut [<(nonnegative) number>] _
>+//                     ::= <closure-type-name>
>+// <closure-type-name> ::= Ul <lambda-sig> E [<(nonnegative) number>] _
>+// <lambda-sig>        ::= <(parameter) type>+
>+static bool ParseUnnamedTypeName(State *state) {
>+  ComplexityGuard guard(state);
>+  if (guard.IsTooComplex()) return false;
>+  ParseState copy = state->parse_state;
>+  // Type's 1-based index n is encoded as { "", n == 1; itoa(n-2), otherwise }.
>+  // Optionally parse the encoded value into 'which' and add 2 to get the index.
>+  int which = -1;
>+
>+  // Unnamed type local to function or class.
>+  if (ParseTwoCharToken(state, "Ut") && Optional(ParseNumber(state, &which)) &&
>+      which <= std::numeric_limits<int>::max() - 2 &&  // Don't overflow.
>+      ParseOneCharToken(state, '_')) {
>+    MaybeAppend(state, "{unnamed type#");
>+    MaybeAppendDecimal(state, 2 + which);
>+    MaybeAppend(state, "}");
>+    return true;
>+  }
>+  state->parse_state = copy;
>+
>+  // Closure type.
>+  which = -1;
>+  if (ParseTwoCharToken(state, "Ul") && DisableAppend(state) &&
>+      OneOrMore(ParseType, state) && RestoreAppend(state, copy.append) &&
>+      ParseOneCharToken(state, 'E') && Optional(ParseNumber(state, &which)) &&
>+      which <= std::numeric_limits<int>::max() - 2 &&  // Don't overflow.
>+      ParseOneCharToken(state, '_')) {
>+    MaybeAppend(state, "{lambda()#");
>+    MaybeAppendDecimal(state, 2 + which);
>+    MaybeAppend(state, "}");
>+    return true;
>+  }
>+  state->parse_state = copy;
>+
>+  return false;
>+}
>+
>+// <number> ::= [n] <non-negative decimal integer>
>+// If "number_out" is non-null, then *number_out is set to the value of the
>+// parsed number on success.
>+static bool ParseNumber(State *state, int *number_out) {
>+  ComplexityGuard guard(state);
>+  if (guard.IsTooComplex()) return false;
>+  bool negative = false;
>+  if (ParseOneCharToken(state, 'n')) {
>+    negative = true;
>+  }
>+  const char *p = RemainingInput(state);
>+  uint64_t number = 0;
>+  for (; *p != '\0'; ++p) {
>+    if (IsDigit(*p)) {
>+      number = number * 10 + (*p - '0');
>+    } else {
>+      break;
>+    }
>+  }
>+  // Apply the sign with uint64_t arithmetic so overflows aren't UB.  Gives
>+  // "incorrect" results for out-of-range inputs, but negative values only
>+  // appear for literals, which aren't printed.
>+  if (negative) {
>+    number = ~number + 1;
>+  }
>+  if (p != RemainingInput(state)) {  // Conversion succeeded.
>+    state->parse_state.mangled_idx += p - RemainingInput(state);
>+    if (number_out != nullptr) {
>+      // Note: possibly truncate "number".
>+      *number_out = number;
>+    }
>+    return true;
>+  }
>+  return false;
>+}
>+
>+// Floating-point literals are encoded using a fixed-length lowercase
>+// hexadecimal std::string.
>+static bool ParseFloatNumber(State *state) {
>+  ComplexityGuard guard(state);
>+  if (guard.IsTooComplex()) return false;
>+  const char *p = RemainingInput(state);
>+  for (; *p != '\0'; ++p) {
>+    if (!IsDigit(*p) && !(*p >= 'a' && *p <= 'f')) {
>+      break;
>+    }
>+  }
>+  if (p != RemainingInput(state)) {  // Conversion succeeded.
>+    state->parse_state.mangled_idx += p - RemainingInput(state);
>+    return true;
>+  }
>+  return false;
>+}
>+
>+// The <seq-id> is a sequence number in base 36,
>+// using digits and upper case letters
>+static bool ParseSeqId(State *state) {
>+  ComplexityGuard guard(state);
>+  if (guard.IsTooComplex()) return false;
>+  const char *p = RemainingInput(state);
>+  for (; *p != '\0'; ++p) {
>+    if (!IsDigit(*p) && !(*p >= 'A' && *p <= 'Z')) {
>+      break;
>+    }
>+  }
>+  if (p != RemainingInput(state)) {  // Conversion succeeded.
>+    state->parse_state.mangled_idx += p - RemainingInput(state);
>+    return true;
>+  }
>+  return false;
>+}
>+
>+// <identifier> ::= <unqualified source code identifier> (of given length)
>+static bool ParseIdentifier(State *state, int length) {
>+  ComplexityGuard guard(state);
>+  if (guard.IsTooComplex()) return false;
>+  if (length < 0 || !AtLeastNumCharsRemaining(RemainingInput(state), length)) {
>+    return false;
>+  }
>+  if (IdentifierIsAnonymousNamespace(state, length)) {
>+    MaybeAppend(state, "(anonymous namespace)");
>+  } else {
>+    MaybeAppendWithLength(state, RemainingInput(state), length);
>+  }
>+  state->parse_state.mangled_idx += length;
>+  return true;
>+}
>+
>+// <operator-name> ::= nw, and other two letters cases
>+//                 ::= cv <type>  # (cast)
>+//                 ::= v  <digit> <source-name> # vendor extended operator
>+static bool ParseOperatorName(State *state, int *arity) {
>+  ComplexityGuard guard(state);
>+  if (guard.IsTooComplex()) return false;
>+  if (!AtLeastNumCharsRemaining(RemainingInput(state), 2)) {
>+    return false;
>+  }
>+  // First check with "cv" (cast) case.
>+  ParseState copy = state->parse_state;
>+  if (ParseTwoCharToken(state, "cv") && MaybeAppend(state, "operator ") &&
>+      EnterNestedName(state) && ParseType(state) &&
>+      LeaveNestedName(state, copy.nest_level)) {
>+    if (arity != nullptr) {
>+      *arity = 1;
>+    }
>+    return true;
>+  }
>+  state->parse_state = copy;
>+
>+  // Then vendor extended operators.
>+  if (ParseOneCharToken(state, 'v') && ParseDigit(state, arity) &&
>+      ParseSourceName(state)) {
>+    return true;
>+  }
>+  state->parse_state = copy;
>+
>+  // Other operator names should start with a lower alphabet followed
>+  // by a lower/upper alphabet.
>+  if (!(IsLower(RemainingInput(state)[0]) &&
>+        IsAlpha(RemainingInput(state)[1]))) {
>+    return false;
>+  }
>+  // We may want to perform a binary search if we really need speed.
>+  const AbbrevPair *p;
>+  for (p = kOperatorList; p->abbrev != nullptr; ++p) {
>+    if (RemainingInput(state)[0] == p->abbrev[0] &&
>+        RemainingInput(state)[1] == p->abbrev[1]) {
>+      if (arity != nullptr) {
>+        *arity = p->arity;
>+      }
>+      MaybeAppend(state, "operator");
>+      if (IsLower(*p->real_name)) {  // new, delete, etc.
>+        MaybeAppend(state, " ");
>+      }
>+      MaybeAppend(state, p->real_name);
>+      state->parse_state.mangled_idx += 2;
>+      return true;
>+    }
>+  }
>+  return false;
>+}
>+
>+// <special-name> ::= TV <type>
>+//                ::= TT <type>
>+//                ::= TI <type>
>+//                ::= TS <type>
>+//                ::= Tc <call-offset> <call-offset> <(base) encoding>
>+//                ::= GV <(object) name>
>+//                ::= T <call-offset> <(base) encoding>
>+// G++ extensions:
>+//                ::= TC <type> <(offset) number> _ <(base) type>
>+//                ::= TF <type>
>+//                ::= TJ <type>
>+//                ::= GR <name>
>+//                ::= GA <encoding>
>+//                ::= Th <call-offset> <(base) encoding>
>+//                ::= Tv <call-offset> <(base) encoding>
>+//
>+// Note: we don't care much about them since they don't appear in
>+// stack traces.  The are special data.
>+static bool ParseSpecialName(State *state) {
>+  ComplexityGuard guard(state);
>+  if (guard.IsTooComplex()) return false;
>+  ParseState copy = state->parse_state;
>+  if (ParseOneCharToken(state, 'T') && ParseCharClass(state, "VTIS") &&
>+      ParseType(state)) {
>+    return true;
>+  }
>+  state->parse_state = copy;
>+
>+  if (ParseTwoCharToken(state, "Tc") && ParseCallOffset(state) &&
>+      ParseCallOffset(state) && ParseEncoding(state)) {
>+    return true;
>+  }
>+  state->parse_state = copy;
>+
>+  if (ParseTwoCharToken(state, "GV") && ParseName(state)) {
>+    return true;
>+  }
>+  state->parse_state = copy;
>+
>+  if (ParseOneCharToken(state, 'T') && ParseCallOffset(state) &&
>+      ParseEncoding(state)) {
>+    return true;
>+  }
>+  state->parse_state = copy;
>+
>+  // G++ extensions
>+  if (ParseTwoCharToken(state, "TC") && ParseType(state) &&
>+      ParseNumber(state, nullptr) && ParseOneCharToken(state, '_') &&
>+      DisableAppend(state) && ParseType(state)) {
>+    RestoreAppend(state, copy.append);
>+    return true;
>+  }
>+  state->parse_state = copy;
>+
>+  if (ParseOneCharToken(state, 'T') && ParseCharClass(state, "FJ") &&
>+      ParseType(state)) {
>+    return true;
>+  }
>+  state->parse_state = copy;
>+
>+  if (ParseTwoCharToken(state, "GR") && ParseName(state)) {
>+    return true;
>+  }
>+  state->parse_state = copy;
>+
>+  if (ParseTwoCharToken(state, "GA") && ParseEncoding(state)) {
>+    return true;
>+  }
>+  state->parse_state = copy;
>+
>+  if (ParseOneCharToken(state, 'T') && ParseCharClass(state, "hv") &&
>+      ParseCallOffset(state) && ParseEncoding(state)) {
>+    return true;
>+  }
>+  state->parse_state = copy;
>+  return false;
>+}
>+
>+// <call-offset> ::= h <nv-offset> _
>+//               ::= v <v-offset> _
>+static bool ParseCallOffset(State *state) {
>+  ComplexityGuard guard(state);
>+  if (guard.IsTooComplex()) return false;
>+  ParseState copy = state->parse_state;
>+  if (ParseOneCharToken(state, 'h') && ParseNVOffset(state) &&
>+      ParseOneCharToken(state, '_')) {
>+    return true;
>+  }
>+  state->parse_state = copy;
>+
>+  if (ParseOneCharToken(state, 'v') && ParseVOffset(state) &&
>+      ParseOneCharToken(state, '_')) {
>+    return true;
>+  }
>+  state->parse_state = copy;
>+
>+  return false;
>+}
>+
>+// <nv-offset> ::= <(offset) number>
>+static bool ParseNVOffset(State *state) {
>+  ComplexityGuard guard(state);
>+  if (guard.IsTooComplex()) return false;
>+  return ParseNumber(state, nullptr);
>+}
>+
>+// <v-offset>  ::= <(offset) number> _ <(virtual offset) number>
>+static bool ParseVOffset(State *state) {
>+  ComplexityGuard guard(state);
>+  if (guard.IsTooComplex()) return false;
>+  ParseState copy = state->parse_state;
>+  if (ParseNumber(state, nullptr) && ParseOneCharToken(state, '_') &&
>+      ParseNumber(state, nullptr)) {
>+    return true;
>+  }
>+  state->parse_state = copy;
>+  return false;
>+}
>+
>+// <ctor-dtor-name> ::= C1 | C2 | C3
>+//                  ::= D0 | D1 | D2
>+// # GCC extensions: "unified" constructor/destructor.  See
>+// # https://github.com/gcc-mirror/gcc/blob/7ad17b583c3643bd4557f29b8391ca7ef08391f5/gcc/cp/mangle.c#L1847
>+//                  ::= C4 | D4
>+static bool ParseCtorDtorName(State *state) {
>+  ComplexityGuard guard(state);
>+  if (guard.IsTooComplex()) return false;
>+  ParseState copy = state->parse_state;
>+  if (ParseOneCharToken(state, 'C') && ParseCharClass(state, "1234")) {
>+    const char *const prev_name = state->out + state->parse_state.prev_name_idx;
>+    MaybeAppendWithLength(state, prev_name,
>+                          state->parse_state.prev_name_length);
>+    return true;
>+  }
>+  state->parse_state = copy;
>+
>+  if (ParseOneCharToken(state, 'D') && ParseCharClass(state, "0124")) {
>+    const char *const prev_name = state->out + state->parse_state.prev_name_idx;
>+    MaybeAppend(state, "~");
>+    MaybeAppendWithLength(state, prev_name,
>+                          state->parse_state.prev_name_length);
>+    return true;
>+  }
>+  state->parse_state = copy;
>+  return false;
>+}
>+
>+// <decltype> ::= Dt <expression> E  # decltype of an id-expression or class
>+//                                   # member access (C++0x)
>+//            ::= DT <expression> E  # decltype of an expression (C++0x)
>+static bool ParseDecltype(State *state) {
>+  ComplexityGuard guard(state);
>+  if (guard.IsTooComplex()) return false;
>+
>+  ParseState copy = state->parse_state;
>+  if (ParseOneCharToken(state, 'D') && ParseCharClass(state, "tT") &&
>+      ParseExpression(state) && ParseOneCharToken(state, 'E')) {
>+    return true;
>+  }
>+  state->parse_state = copy;
>+
>+  return false;
>+}
>+
>+// <type> ::= <CV-qualifiers> <type>
>+//        ::= P <type>   # pointer-to
>+//        ::= R <type>   # reference-to
>+//        ::= O <type>   # rvalue reference-to (C++0x)
>+//        ::= C <type>   # complex pair (C 2000)
>+//        ::= G <type>   # imaginary (C 2000)
>+//        ::= U <source-name> <type>  # vendor extended type qualifier
>+//        ::= <builtin-type>
>+//        ::= <function-type>
>+//        ::= <class-enum-type>  # note: just an alias for <name>
>+//        ::= <array-type>
>+//        ::= <pointer-to-member-type>
>+//        ::= <template-template-param> <template-args>
>+//        ::= <template-param>
>+//        ::= <decltype>
>+//        ::= <substitution>
>+//        ::= Dp <type>          # pack expansion of (C++0x)
>+//
>+static bool ParseType(State *state) {
>+  ComplexityGuard guard(state);
>+  if (guard.IsTooComplex()) return false;
>+  ParseState copy = state->parse_state;
>+
>+  // We should check CV-qualifers, and PRGC things first.
>+  //
>+  // CV-qualifiers overlap with some operator names, but an operator name is not
>+  // valid as a type.  To avoid an ambiguity that can lead to exponential time
>+  // complexity, refuse to backtrack the CV-qualifiers.
>+  //
>+  // _Z4aoeuIrMvvE
>+  //  => _Z 4aoeuI        rM  v     v   E
>+  //         aoeu<operator%=, void, void>
>+  //  => _Z 4aoeuI r Mv v              E
>+  //         aoeu<void void::* restrict>
>+  //
>+  // By consuming the CV-qualifiers first, the former parse is disabled.
>+  if (ParseCVQualifiers(state)) {
>+    const bool result = ParseType(state);
>+    if (!result) state->parse_state = copy;
>+    return result;
>+  }
>+  state->parse_state = copy;
>+
>+  // Similarly, these tag characters can overlap with other <name>s resulting in
>+  // two different parse prefixes that land on <template-args> in the same
>+  // place, such as "C3r1xI...".  So, disable the "ctor-name = C3" parse by
>+  // refusing to backtrack the tag characters.
>+  if (ParseCharClass(state, "OPRCG")) {
>+    const bool result = ParseType(state);
>+    if (!result) state->parse_state = copy;
>+    return result;
>+  }
>+  state->parse_state = copy;
>+
>+  if (ParseTwoCharToken(state, "Dp") && ParseType(state)) {
>+    return true;
>+  }
>+  state->parse_state = copy;
>+
>+  if (ParseOneCharToken(state, 'U') && ParseSourceName(state) &&
>+      ParseType(state)) {
>+    return true;
>+  }
>+  state->parse_state = copy;
>+
>+  if (ParseBuiltinType(state) || ParseFunctionType(state) ||
>+      ParseClassEnumType(state) || ParseArrayType(state) ||
>+      ParsePointerToMemberType(state) || ParseDecltype(state) ||
>+      // "std" on its own isn't a type.
>+      ParseSubstitution(state, /*accept_std=*/false)) {
>+    return true;
>+  }
>+
>+  if (ParseTemplateTemplateParam(state) && ParseTemplateArgs(state)) {
>+    return true;
>+  }
>+  state->parse_state = copy;
>+
>+  // Less greedy than <template-template-param> <template-args>.
>+  if (ParseTemplateParam(state)) {
>+    return true;
>+  }
>+
>+  return false;
>+}
>+
>+// <CV-qualifiers> ::= [r] [V] [K]
>+// We don't allow empty <CV-qualifiers> to avoid infinite loop in
>+// ParseType().
>+static bool ParseCVQualifiers(State *state) {
>+  ComplexityGuard guard(state);
>+  if (guard.IsTooComplex()) return false;
>+  int num_cv_qualifiers = 0;
>+  num_cv_qualifiers += ParseOneCharToken(state, 'r');
>+  num_cv_qualifiers += ParseOneCharToken(state, 'V');
>+  num_cv_qualifiers += ParseOneCharToken(state, 'K');
>+  return num_cv_qualifiers > 0;
>+}
>+
>+// <builtin-type> ::= v, etc.
>+//                ::= u <source-name>
>+static bool ParseBuiltinType(State *state) {
>+  ComplexityGuard guard(state);
>+  if (guard.IsTooComplex()) return false;
>+  const AbbrevPair *p;
>+  for (p = kBuiltinTypeList; p->abbrev != nullptr; ++p) {
>+    if (RemainingInput(state)[0] == p->abbrev[0]) {
>+      MaybeAppend(state, p->real_name);
>+      ++state->parse_state.mangled_idx;
>+      return true;
>+    }
>+  }
>+
>+  ParseState copy = state->parse_state;
>+  if (ParseOneCharToken(state, 'u') && ParseSourceName(state)) {
>+    return true;
>+  }
>+  state->parse_state = copy;
>+  return false;
>+}
>+
>+// <function-type> ::= F [Y] <bare-function-type> E
>+static bool ParseFunctionType(State *state) {
>+  ComplexityGuard guard(state);
>+  if (guard.IsTooComplex()) return false;
>+  ParseState copy = state->parse_state;
>+  if (ParseOneCharToken(state, 'F') &&
>+      Optional(ParseOneCharToken(state, 'Y')) && ParseBareFunctionType(state) &&
>+      ParseOneCharToken(state, 'E')) {
>+    return true;
>+  }
>+  state->parse_state = copy;
>+  return false;
>+}
>+
>+// <bare-function-type> ::= <(signature) type>+
>+static bool ParseBareFunctionType(State *state) {
>+  ComplexityGuard guard(state);
>+  if (guard.IsTooComplex()) return false;
>+  ParseState copy = state->parse_state;
>+  DisableAppend(state);
>+  if (OneOrMore(ParseType, state)) {
>+    RestoreAppend(state, copy.append);
>+    MaybeAppend(state, "()");
>+    return true;
>+  }
>+  state->parse_state = copy;
>+  return false;
>+}
>+
>+// <class-enum-type> ::= <name>
>+static bool ParseClassEnumType(State *state) {
>+  ComplexityGuard guard(state);
>+  if (guard.IsTooComplex()) return false;
>+  return ParseName(state);
>+}
>+
>+// <array-type> ::= A <(positive dimension) number> _ <(element) type>
>+//              ::= A [<(dimension) expression>] _ <(element) type>
>+static bool ParseArrayType(State *state) {
>+  ComplexityGuard guard(state);
>+  if (guard.IsTooComplex()) return false;
>+  ParseState copy = state->parse_state;
>+  if (ParseOneCharToken(state, 'A') && ParseNumber(state, nullptr) &&
>+      ParseOneCharToken(state, '_') && ParseType(state)) {
>+    return true;
>+  }
>+  state->parse_state = copy;
>+
>+  if (ParseOneCharToken(state, 'A') && Optional(ParseExpression(state)) &&
>+      ParseOneCharToken(state, '_') && ParseType(state)) {
>+    return true;
>+  }
>+  state->parse_state = copy;
>+  return false;
>+}
>+
>+// <pointer-to-member-type> ::= M <(class) type> <(member) type>
>+static bool ParsePointerToMemberType(State *state) {
>+  ComplexityGuard guard(state);
>+  if (guard.IsTooComplex()) return false;
>+  ParseState copy = state->parse_state;
>+  if (ParseOneCharToken(state, 'M') && ParseType(state) && ParseType(state)) {
>+    return true;
>+  }
>+  state->parse_state = copy;
>+  return false;
>+}
>+
>+// <template-param> ::= T_
>+//                  ::= T <parameter-2 non-negative number> _
>+static bool ParseTemplateParam(State *state) {
>+  ComplexityGuard guard(state);
>+  if (guard.IsTooComplex()) return false;
>+  if (ParseTwoCharToken(state, "T_")) {
>+    MaybeAppend(state, "?");  // We don't support template substitutions.
>+    return true;
>+  }
>+
>+  ParseState copy = state->parse_state;
>+  if (ParseOneCharToken(state, 'T') && ParseNumber(state, nullptr) &&
>+      ParseOneCharToken(state, '_')) {
>+    MaybeAppend(state, "?");  // We don't support template substitutions.
>+    return true;
>+  }
>+  state->parse_state = copy;
>+  return false;
>+}
>+
>+// <template-template-param> ::= <template-param>
>+//                           ::= <substitution>
>+static bool ParseTemplateTemplateParam(State *state) {
>+  ComplexityGuard guard(state);
>+  if (guard.IsTooComplex()) return false;
>+  return (ParseTemplateParam(state) ||
>+          // "std" on its own isn't a template.
>+          ParseSubstitution(state, /*accept_std=*/false));
>+}
>+
>+// <template-args> ::= I <template-arg>+ E
>+static bool ParseTemplateArgs(State *state) {
>+  ComplexityGuard guard(state);
>+  if (guard.IsTooComplex()) return false;
>+  ParseState copy = state->parse_state;
>+  DisableAppend(state);
>+  if (ParseOneCharToken(state, 'I') && OneOrMore(ParseTemplateArg, state) &&
>+      ParseOneCharToken(state, 'E')) {
>+    RestoreAppend(state, copy.append);
>+    MaybeAppend(state, "<>");
>+    return true;
>+  }
>+  state->parse_state = copy;
>+  return false;
>+}
>+
>+// <template-arg>  ::= <type>
>+//                 ::= <expr-primary>
>+//                 ::= J <template-arg>* E        # argument pack
>+//                 ::= X <expression> E
>+static bool ParseTemplateArg(State *state) {
>+  ComplexityGuard guard(state);
>+  if (guard.IsTooComplex()) return false;
>+  ParseState copy = state->parse_state;
>+  if (ParseOneCharToken(state, 'J') && ZeroOrMore(ParseTemplateArg, state) &&
>+      ParseOneCharToken(state, 'E')) {
>+    return true;
>+  }
>+  state->parse_state = copy;
>+
>+  // There can be significant overlap between the following leading to
>+  // exponential backtracking:
>+  //
>+  //   <expr-primary> ::= L <type> <expr-cast-value> E
>+  //                 e.g. L 2xxIvE 1                 E
>+  //   <type>         ==> <local-source-name> <template-args>
>+  //                 e.g. L 2xx               IvE
>+  //
>+  // This means parsing an entire <type> twice, and <type> can contain
>+  // <template-arg>, so this can generate exponential backtracking.  There is
>+  // only overlap when the remaining input starts with "L <source-name>", so
>+  // parse all cases that can start this way jointly to share the common prefix.
>+  //
>+  // We have:
>+  //
>+  //   <template-arg> ::= <type>
>+  //                  ::= <expr-primary>
>+  //
>+  // First, drop all the productions of <type> that must start with something
>+  // other than 'L'.  All that's left is <class-enum-type>; inline it.
>+  //
>+  //   <type> ::= <nested-name> # starts with 'N'
>+  //          ::= <unscoped-name>
>+  //          ::= <unscoped-template-name> <template-args>
>+  //          ::= <local-name> # starts with 'Z'
>+  //
>+  // Drop and inline again:
>+  //
>+  //   <type> ::= <unscoped-name>
>+  //          ::= <unscoped-name> <template-args>
>+  //          ::= <substitution> <template-args> # starts with 'S'
>+  //
>+  // Merge the first two, inline <unscoped-name>, drop last:
>+  //
>+  //   <type> ::= <unqualified-name> [<template-args>]
>+  //          ::= St <unqualified-name> [<template-args>] # starts with 'S'
>+  //
>+  // Drop and inline:
>+  //
>+  //   <type> ::= <operator-name> [<template-args>] # starts with lowercase
>+  //          ::= <ctor-dtor-name> [<template-args>] # starts with 'C' or 'D'
>+  //          ::= <source-name> [<template-args>] # starts with digit
>+  //          ::= <local-source-name> [<template-args>]
>+  //          ::= <unnamed-type-name> [<template-args>] # starts with 'U'
>+  //
>+  // One more time:
>+  //
>+  //   <type> ::= L <source-name> [<template-args>]
>+  //
>+  // Likewise with <expr-primary>:
>+  //
>+  //   <expr-primary> ::= L <type> <expr-cast-value> E
>+  //                  ::= LZ <encoding> E # cannot overlap; drop
>+  //                  ::= L <mangled_name> E # cannot overlap; drop
>+  //
>+  // By similar reasoning as shown above, the only <type>s starting with
>+  // <source-name> are "<source-name> [<template-args>]".  Inline this.
>+  //
>+  //   <expr-primary> ::= L <source-name> [<template-args>] <expr-cast-value> E
>+  //
>+  // Now inline both of these into <template-arg>:
>+  //
>+  //   <template-arg> ::= L <source-name> [<template-args>]
>+  //                  ::= L <source-name> [<template-args>] <expr-cast-value> E
>+  //
>+  // Merge them and we're done:
>+  //   <template-arg>
>+  //     ::= L <source-name> [<template-args>] [<expr-cast-value> E]
>+  if (ParseLocalSourceName(state) && Optional(ParseTemplateArgs(state))) {
>+    copy = state->parse_state;
>+    if (ParseExprCastValue(state) && ParseOneCharToken(state, 'E')) {
>+      return true;
>+    }
>+    state->parse_state = copy;
>+    return true;
>+  }
>+
>+  // Now that the overlapping cases can't reach this code, we can safely call
>+  // both of these.
>+  if (ParseType(state) || ParseExprPrimary(state)) {
>+    return true;
>+  }
>+  state->parse_state = copy;
>+
>+  if (ParseOneCharToken(state, 'X') && ParseExpression(state) &&
>+      ParseOneCharToken(state, 'E')) {
>+    return true;
>+  }
>+  state->parse_state = copy;
>+  return false;
>+}
>+
>+// <unresolved-type> ::= <template-param> [<template-args>]
>+//                   ::= <decltype>
>+//                   ::= <substitution>
>+static inline bool ParseUnresolvedType(State *state) {
>+  // No ComplexityGuard because we don't copy the state in this stack frame.
>+  return (ParseTemplateParam(state) && Optional(ParseTemplateArgs(state))) ||
>+         ParseDecltype(state) || ParseSubstitution(state, /*accept_std=*/false);
>+}
>+
>+// <simple-id> ::= <source-name> [<template-args>]
>+static inline bool ParseSimpleId(State *state) {
>+  // No ComplexityGuard because we don't copy the state in this stack frame.
>+
>+  // Note: <simple-id> cannot be followed by a parameter pack; see comment in
>+  // ParseUnresolvedType.
>+  return ParseSourceName(state) && Optional(ParseTemplateArgs(state));
>+}
>+
>+// <base-unresolved-name> ::= <source-name> [<template-args>]
>+//                        ::= on <operator-name> [<template-args>]
>+//                        ::= dn <destructor-name>
>+static bool ParseBaseUnresolvedName(State *state) {
>+  ComplexityGuard guard(state);
>+  if (guard.IsTooComplex()) return false;
>+
>+  if (ParseSimpleId(state)) {
>+    return true;
>+  }
>+
>+  ParseState copy = state->parse_state;
>+  if (ParseTwoCharToken(state, "on") && ParseOperatorName(state, nullptr) &&
>+      Optional(ParseTemplateArgs(state))) {
>+    return true;
>+  }
>+  state->parse_state = copy;
>+
>+  if (ParseTwoCharToken(state, "dn") &&
>+      (ParseUnresolvedType(state) || ParseSimpleId(state))) {
>+    return true;
>+  }
>+  state->parse_state = copy;
>+
>+  return false;
>+}
>+
>+// <unresolved-name> ::= [gs] <base-unresolved-name>
>+//                   ::= sr <unresolved-type> <base-unresolved-name>
>+//                   ::= srN <unresolved-type> <unresolved-qualifier-level>+ E
>+//                         <base-unresolved-name>
>+//                   ::= [gs] sr <unresolved-qualifier-level>+ E
>+//                         <base-unresolved-name>
>+static bool ParseUnresolvedName(State *state) {
>+  ComplexityGuard guard(state);
>+  if (guard.IsTooComplex()) return false;
>+
>+  ParseState copy = state->parse_state;
>+  if (Optional(ParseTwoCharToken(state, "gs")) &&
>+      ParseBaseUnresolvedName(state)) {
>+    return true;
>+  }
>+  state->parse_state = copy;
>+
>+  if (ParseTwoCharToken(state, "sr") && ParseUnresolvedType(state) &&
>+      ParseBaseUnresolvedName(state)) {
>+    return true;
>+  }
>+  state->parse_state = copy;
>+
>+  if (ParseTwoCharToken(state, "sr") && ParseOneCharToken(state, 'N') &&
>+      ParseUnresolvedType(state) &&
>+      OneOrMore(/* <unresolved-qualifier-level> ::= */ ParseSimpleId, state) &&
>+      ParseOneCharToken(state, 'E') && ParseBaseUnresolvedName(state)) {
>+    return true;
>+  }
>+  state->parse_state = copy;
>+
>+  if (Optional(ParseTwoCharToken(state, "gs")) &&
>+      ParseTwoCharToken(state, "sr") &&
>+      OneOrMore(/* <unresolved-qualifier-level> ::= */ ParseSimpleId, state) &&
>+      ParseOneCharToken(state, 'E') && ParseBaseUnresolvedName(state)) {
>+    return true;
>+  }
>+  state->parse_state = copy;
>+
>+  return false;
>+}
>+
>+// <expression> ::= <1-ary operator-name> <expression>
>+//              ::= <2-ary operator-name> <expression> <expression>
>+//              ::= <3-ary operator-name> <expression> <expression> <expression>
>+//              ::= cl <expression>+ E
>+//              ::= cv <type> <expression>      # type (expression)
>+//              ::= cv <type> _ <expression>* E # type (expr-list)
>+//              ::= st <type>
>+//              ::= <template-param>
>+//              ::= <function-param>
>+//              ::= <expr-primary>
>+//              ::= dt <expression> <unresolved-name> # expr.name
>+//              ::= pt <expression> <unresolved-name> # expr->name
>+//              ::= sp <expression>         # argument pack expansion
>+//              ::= sr <type> <unqualified-name> <template-args>
>+//              ::= sr <type> <unqualified-name>
>+// <function-param> ::= fp <(top-level) CV-qualifiers> _
>+//                  ::= fp <(top-level) CV-qualifiers> <number> _
>+//                  ::= fL <number> p <(top-level) CV-qualifiers> _
>+//                  ::= fL <number> p <(top-level) CV-qualifiers> <number> _
>+static bool ParseExpression(State *state) {
>+  ComplexityGuard guard(state);
>+  if (guard.IsTooComplex()) return false;
>+  if (ParseTemplateParam(state) || ParseExprPrimary(state)) {
>+    return true;
>+  }
>+
>+  // Object/function call expression.
>+  ParseState copy = state->parse_state;
>+  if (ParseTwoCharToken(state, "cl") && OneOrMore(ParseExpression, state) &&
>+      ParseOneCharToken(state, 'E')) {
>+    return true;
>+  }
>+  state->parse_state = copy;
>+
>+  // Function-param expression (level 0).
>+  if (ParseTwoCharToken(state, "fp") && Optional(ParseCVQualifiers(state)) &&
>+      Optional(ParseNumber(state, nullptr)) && ParseOneCharToken(state, '_')) {
>+    return true;
>+  }
>+  state->parse_state = copy;
>+
>+  // Function-param expression (level 1+).
>+  if (ParseTwoCharToken(state, "fL") && Optional(ParseNumber(state, nullptr)) &&
>+      ParseOneCharToken(state, 'p') && Optional(ParseCVQualifiers(state)) &&
>+      Optional(ParseNumber(state, nullptr)) && ParseOneCharToken(state, '_')) {
>+    return true;
>+  }
>+  state->parse_state = copy;
>+
>+  // Parse the conversion expressions jointly to avoid re-parsing the <type> in
>+  // their common prefix.  Parsed as:
>+  // <expression> ::= cv <type> <conversion-args>
>+  // <conversion-args> ::= _ <expression>* E
>+  //                   ::= <expression>
>+  //
>+  // Also don't try ParseOperatorName after seeing "cv", since ParseOperatorName
>+  // also needs to accept "cv <type>" in other contexts.
>+  if (ParseTwoCharToken(state, "cv")) {
>+    if (ParseType(state)) {
>+      ParseState copy2 = state->parse_state;
>+      if (ParseOneCharToken(state, '_') && ZeroOrMore(ParseExpression, state) &&
>+          ParseOneCharToken(state, 'E')) {
>+        return true;
>+      }
>+      state->parse_state = copy2;
>+      if (ParseExpression(state)) {
>+        return true;
>+      }
>+    }
>+  } else {
>+    // Parse unary, binary, and ternary operator expressions jointly, taking
>+    // care not to re-parse subexpressions repeatedly. Parse like:
>+    //   <expression> ::= <operator-name> <expression>
>+    //                    [<one-to-two-expressions>]
>+    //   <one-to-two-expressions> ::= <expression> [<expression>]
>+    int arity = -1;
>+    if (ParseOperatorName(state, &arity) &&
>+        arity > 0 &&  // 0 arity => disabled.
>+        (arity < 3 || ParseExpression(state)) &&
>+        (arity < 2 || ParseExpression(state)) &&
>+        (arity < 1 || ParseExpression(state))) {
>+      return true;
>+    }
>+  }
>+  state->parse_state = copy;
>+
>+  // sizeof type
>+  if (ParseTwoCharToken(state, "st") && ParseType(state)) {
>+    return true;
>+  }
>+  state->parse_state = copy;
>+
>+  // Object and pointer member access expressions.
>+  if ((ParseTwoCharToken(state, "dt") || ParseTwoCharToken(state, "pt")) &&
>+      ParseExpression(state) && ParseType(state)) {
>+    return true;
>+  }
>+  state->parse_state = copy;
>+
>+  // Parameter pack expansion
>+  if (ParseTwoCharToken(state, "sp") && ParseExpression(state)) {
>+    return true;
>+  }
>+  state->parse_state = copy;
>+
>+  return ParseUnresolvedName(state);
>+}
>+
>+// <expr-primary> ::= L <type> <(value) number> E
>+//                ::= L <type> <(value) float> E
>+//                ::= L <mangled-name> E
>+//                // A bug in g++'s C++ ABI version 2 (-fabi-version=2).
>+//                ::= LZ <encoding> E
>+//
>+// Warning, subtle: the "bug" LZ production above is ambiguous with the first
>+// production where <type> starts with <local-name>, which can lead to
>+// exponential backtracking in two scenarios:
>+//
>+// - When whatever follows the E in the <local-name> in the first production is
>+//   not a name, we backtrack the whole <encoding> and re-parse the whole thing.
>+//
>+// - When whatever follows the <local-name> in the first production is not a
>+//   number and this <expr-primary> may be followed by a name, we backtrack the
>+//   <name> and re-parse it.
>+//
>+// Moreover this ambiguity isn't always resolved -- for example, the following
>+// has two different parses:
>+//
>+//   _ZaaILZ4aoeuE1x1EvE
>+//   => operator&&<aoeu, x, E, void>
>+//   => operator&&<(aoeu::x)(1), void>
>+//
>+// To resolve this, we just do what GCC's demangler does, and refuse to parse
>+// casts to <local-name> types.
>+static bool ParseExprPrimary(State *state) {
>+  ComplexityGuard guard(state);
>+  if (guard.IsTooComplex()) return false;
>+  ParseState copy = state->parse_state;
>+
>+  // The "LZ" special case: if we see LZ, we commit to accept "LZ <encoding> E"
>+  // or fail, no backtracking.
>+  if (ParseTwoCharToken(state, "LZ")) {
>+    if (ParseEncoding(state) && ParseOneCharToken(state, 'E')) {
>+      return true;
>+    }
>+
>+    state->parse_state = copy;
>+    return false;
>+  }
>+
>+  // The merged cast production.
>+  if (ParseOneCharToken(state, 'L') && ParseType(state) &&
>+      ParseExprCastValue(state)) {
>+    return true;
>+  }
>+  state->parse_state = copy;
>+
>+  if (ParseOneCharToken(state, 'L') && ParseMangledName(state) &&
>+      ParseOneCharToken(state, 'E')) {
>+    return true;
>+  }
>+  state->parse_state = copy;
>+
>+  return false;
>+}
>+
>+// <number> or <float>, followed by 'E', as described above ParseExprPrimary.
>+static bool ParseExprCastValue(State *state) {
>+  ComplexityGuard guard(state);
>+  if (guard.IsTooComplex()) return false;
>+  // We have to be able to backtrack after accepting a number because we could
>+  // have e.g. "7fffE", which will accept "7" as a number but then fail to find
>+  // the 'E'.
>+  ParseState copy = state->parse_state;
>+  if (ParseNumber(state, nullptr) && ParseOneCharToken(state, 'E')) {
>+    return true;
>+  }
>+  state->parse_state = copy;
>+
>+  if (ParseFloatNumber(state) && ParseOneCharToken(state, 'E')) {
>+    return true;
>+  }
>+  state->parse_state = copy;
>+
>+  return false;
>+}
>+
>+// <local-name> ::= Z <(function) encoding> E <(entity) name> [<discriminator>]
>+//              ::= Z <(function) encoding> E s [<discriminator>]
>+//
>+// Parsing a common prefix of these two productions together avoids an
>+// exponential blowup of backtracking.  Parse like:
>+//   <local-name> := Z <encoding> E <local-name-suffix>
>+//   <local-name-suffix> ::= s [<discriminator>]
>+//                       ::= <name> [<discriminator>]
>+
>+static bool ParseLocalNameSuffix(State *state) {
>+  ComplexityGuard guard(state);
>+  if (guard.IsTooComplex()) return false;
>+
>+  if (MaybeAppend(state, "::") && ParseName(state) &&
>+      Optional(ParseDiscriminator(state))) {
>+    return true;
>+  }
>+
>+  // Since we're not going to overwrite the above "::" by re-parsing the
>+  // <encoding> (whose trailing '\0' byte was in the byte now holding the
>+  // first ':'), we have to rollback the "::" if the <name> parse failed.
>+  if (state->parse_state.append) {
>+    state->out[state->parse_state.out_cur_idx - 2] = '\0';
>+  }
>+
>+  return ParseOneCharToken(state, 's') && Optional(ParseDiscriminator(state));
>+}
>+
>+static bool ParseLocalName(State *state) {
>+  ComplexityGuard guard(state);
>+  if (guard.IsTooComplex()) return false;
>+  ParseState copy = state->parse_state;
>+  if (ParseOneCharToken(state, 'Z') && ParseEncoding(state) &&
>+      ParseOneCharToken(state, 'E') && ParseLocalNameSuffix(state)) {
>+    return true;
>+  }
>+  state->parse_state = copy;
>+  return false;
>+}
>+
>+// <discriminator> := _ <(non-negative) number>
>+static bool ParseDiscriminator(State *state) {
>+  ComplexityGuard guard(state);
>+  if (guard.IsTooComplex()) return false;
>+  ParseState copy = state->parse_state;
>+  if (ParseOneCharToken(state, '_') && ParseNumber(state, nullptr)) {
>+    return true;
>+  }
>+  state->parse_state = copy;
>+  return false;
>+}
>+
>+// <substitution> ::= S_
>+//                ::= S <seq-id> _
>+//                ::= St, etc.
>+//
>+// "St" is special in that it's not valid as a standalone name, and it *is*
>+// allowed to precede a name without being wrapped in "N...E".  This means that
>+// if we accept it on its own, we can accept "St1a" and try to parse
>+// template-args, then fail and backtrack, accept "St" on its own, then "1a" as
>+// an unqualified name and re-parse the same template-args.  To block this
>+// exponential backtracking, we disable it with 'accept_std=false' in
>+// problematic contexts.
>+static bool ParseSubstitution(State *state, bool accept_std) {
>+  ComplexityGuard guard(state);
>+  if (guard.IsTooComplex()) return false;
>+  if (ParseTwoCharToken(state, "S_")) {
>+    MaybeAppend(state, "?");  // We don't support substitutions.
>+    return true;
>+  }
>+
>+  ParseState copy = state->parse_state;
>+  if (ParseOneCharToken(state, 'S') && ParseSeqId(state) &&
>+      ParseOneCharToken(state, '_')) {
>+    MaybeAppend(state, "?");  // We don't support substitutions.
>+    return true;
>+  }
>+  state->parse_state = copy;
>+
>+  // Expand abbreviations like "St" => "std".
>+  if (ParseOneCharToken(state, 'S')) {
>+    const AbbrevPair *p;
>+    for (p = kSubstitutionList; p->abbrev != nullptr; ++p) {
>+      if (RemainingInput(state)[0] == p->abbrev[1] &&
>+          (accept_std || p->abbrev[1] != 't')) {
>+        MaybeAppend(state, "std");
>+        if (p->real_name[0] != '\0') {
>+          MaybeAppend(state, "::");
>+          MaybeAppend(state, p->real_name);
>+        }
>+        ++state->parse_state.mangled_idx;
>+        return true;
>+      }
>+    }
>+  }
>+  state->parse_state = copy;
>+  return false;
>+}
>+
>+// Parse <mangled-name>, optionally followed by either a function-clone suffix
>+// or version suffix.  Returns true only if all of "mangled_cur" was consumed.
>+static bool ParseTopLevelMangledName(State *state) {
>+  ComplexityGuard guard(state);
>+  if (guard.IsTooComplex()) return false;
>+  if (ParseMangledName(state)) {
>+    if (RemainingInput(state)[0] != '\0') {
>+      // Drop trailing function clone suffix, if any.
>+      if (IsFunctionCloneSuffix(RemainingInput(state))) {
>+        return true;
>+      }
>+      // Append trailing version suffix if any.
>+      // ex. _Z3foo@@GLIBCXX_3.4
>+      if (RemainingInput(state)[0] == '@') {
>+        MaybeAppend(state, RemainingInput(state));
>+        return true;
>+      }
>+      return false;  // Unconsumed suffix.
>+    }
>+    return true;
>+  }
>+  return false;
>+}
>+
>+static bool Overflowed(const State *state) {
>+  return state->parse_state.out_cur_idx >= state->out_end_idx;
>+}
>+
>+// The demangler entry point.
>+bool Demangle(const char *mangled, char *out, int out_size) {
>+  State state;
>+  InitState(&state, mangled, out, out_size);
>+  return ParseTopLevelMangledName(&state) && !Overflowed(&state);
>+}
>+
>+}  // namespace debugging_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/demangle.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/demangle.h
>new file mode 100644
>index 00000000000..2e75564ed35
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/demangle.h
>@@ -0,0 +1,67 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+// An async-signal-safe and thread-safe demangler for Itanium C++ ABI
>+// (aka G++ V3 ABI).
>+//
>+// The demangler is implemented to be used in async signal handlers to
>+// symbolize stack traces.  We cannot use libstdc++'s
>+// abi::__cxa_demangle() in such signal handlers since it's not async
>+// signal safe (it uses malloc() internally).
>+//
>+// Note that this demangler doesn't support full demangling.  More
>+// specifically, it doesn't print types of function parameters and
>+// types of template arguments.  It just skips them.  However, it's
>+// still very useful to extract basic information such as class,
>+// function, constructor, destructor, and operator names.
>+//
>+// See the implementation note in demangle.cc if you are interested.
>+//
>+// Example:
>+//
>+// | Mangled Name  | The Demangler | abi::__cxa_demangle()
>+// |---------------|---------------|-----------------------
>+// | _Z1fv         | f()           | f()
>+// | _Z1fi         | f()           | f(int)
>+// | _Z3foo3bar    | foo()         | foo(bar)
>+// | _Z1fIiEvi     | f<>()         | void f<int>(int)
>+// | _ZN1N1fE      | N::f          | N::f
>+// | _ZN3Foo3BarEv | Foo::Bar()    | Foo::Bar()
>+// | _Zrm1XS_"     | operator%()   | operator%(X, X)
>+// | _ZN3FooC1Ev   | Foo::Foo()    | Foo::Foo()
>+// | _Z1fSs        | f()           | f(std::basic_string<char,
>+// |               |               |   std::char_traits<char>,
>+// |               |               |   std::allocator<char> >)
>+//
>+// See the unit test for more examples.
>+//
>+// Note: we might want to write demanglers for ABIs other than Itanium
>+// C++ ABI in the future.
>+//
>+
>+#ifndef ABSL_DEBUGGING_INTERNAL_DEMANGLE_H_
>+#define ABSL_DEBUGGING_INTERNAL_DEMANGLE_H_
>+
>+namespace absl {
>+namespace debugging_internal {
>+
>+// Demangle `mangled`.  On success, return true and write the
>+// demangled symbol name to `out`.  Otherwise, return false.
>+// `out` is modified even if demangling is unsuccessful.
>+bool Demangle(const char *mangled, char *out, int out_size);
>+
>+}  // namespace debugging_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_DEBUGGING_INTERNAL_DEMANGLE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/demangle_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/demangle_test.cc
>new file mode 100644
>index 00000000000..b9d9008f007
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/demangle_test.cc
>@@ -0,0 +1,191 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/debugging/internal/demangle.h"
>+
>+#include <cstdlib>
>+#include <string>
>+
>+#include "gtest/gtest.h"
>+#include "absl/base/internal/raw_logging.h"
>+#include "absl/debugging/internal/stack_consumption.h"
>+#include "absl/memory/memory.h"
>+
>+namespace absl {
>+namespace debugging_internal {
>+namespace {
>+
>+// A wrapper function for Demangle() to make the unit test simple.
>+static const char *DemangleIt(const char * const mangled) {
>+  static char demangled[4096];
>+  if (Demangle(mangled, demangled, sizeof(demangled))) {
>+    return demangled;
>+  } else {
>+    return mangled;
>+  }
>+}
>+
>+// Test corner cases of bounary conditions.
>+TEST(Demangle, CornerCases) {
>+  char tmp[10];
>+  EXPECT_TRUE(Demangle("_Z6foobarv", tmp, sizeof(tmp)));
>+  // sizeof("foobar()") == 9
>+  EXPECT_STREQ("foobar()", tmp);
>+  EXPECT_TRUE(Demangle("_Z6foobarv", tmp, 9));
>+  EXPECT_STREQ("foobar()", tmp);
>+  EXPECT_FALSE(Demangle("_Z6foobarv", tmp, 8));  // Not enough.
>+  EXPECT_FALSE(Demangle("_Z6foobarv", tmp, 1));
>+  EXPECT_FALSE(Demangle("_Z6foobarv", tmp, 0));
>+  EXPECT_FALSE(Demangle("_Z6foobarv", nullptr, 0));  // Should not cause SEGV.
>+  EXPECT_FALSE(Demangle("_Z1000000", tmp, 9));
>+}
>+
>+// Test handling of functions suffixed with .clone.N, which is used
>+// by GCC 4.5.x (and our locally-modified version of GCC 4.4.x), and
>+// .constprop.N and .isra.N, which are used by GCC 4.6.x.  These
>+// suffixes are used to indicate functions which have been cloned
>+// during optimization.  We ignore these suffixes.
>+TEST(Demangle, Clones) {
>+  char tmp[20];
>+  EXPECT_TRUE(Demangle("_ZL3Foov", tmp, sizeof(tmp)));
>+  EXPECT_STREQ("Foo()", tmp);
>+  EXPECT_TRUE(Demangle("_ZL3Foov.clone.3", tmp, sizeof(tmp)));
>+  EXPECT_STREQ("Foo()", tmp);
>+  EXPECT_TRUE(Demangle("_ZL3Foov.constprop.80", tmp, sizeof(tmp)));
>+  EXPECT_STREQ("Foo()", tmp);
>+  EXPECT_TRUE(Demangle("_ZL3Foov.isra.18", tmp, sizeof(tmp)));
>+  EXPECT_STREQ("Foo()", tmp);
>+  EXPECT_TRUE(Demangle("_ZL3Foov.isra.2.constprop.18", tmp, sizeof(tmp)));
>+  EXPECT_STREQ("Foo()", tmp);
>+  // Invalid (truncated), should not demangle.
>+  EXPECT_FALSE(Demangle("_ZL3Foov.clo", tmp, sizeof(tmp)));
>+  // Invalid (.clone. not followed by number), should not demangle.
>+  EXPECT_FALSE(Demangle("_ZL3Foov.clone.", tmp, sizeof(tmp)));
>+  // Invalid (.clone. followed by non-number), should not demangle.
>+  EXPECT_FALSE(Demangle("_ZL3Foov.clone.foo", tmp, sizeof(tmp)));
>+  // Invalid (.constprop. not followed by number), should not demangle.
>+  EXPECT_FALSE(Demangle("_ZL3Foov.isra.2.constprop.", tmp, sizeof(tmp)));
>+}
>+
>+// Tests that verify that Demangle footprint is within some limit.
>+// They are not to be run under sanitizers as the sanitizers increase
>+// stack consumption by about 4x.
>+#if defined(ABSL_INTERNAL_HAVE_DEBUGGING_STACK_CONSUMPTION) && \
>+    !ADDRESS_SANITIZER && !MEMORY_SANITIZER && !THREAD_SANITIZER
>+
>+static const char *g_mangled;
>+static char g_demangle_buffer[4096];
>+static char *g_demangle_result;
>+
>+static void DemangleSignalHandler(int signo) {
>+  if (Demangle(g_mangled, g_demangle_buffer, sizeof(g_demangle_buffer))) {
>+    g_demangle_result = g_demangle_buffer;
>+  } else {
>+    g_demangle_result = nullptr;
>+  }
>+}
>+
>+// Call Demangle and figure out the stack footprint of this call.
>+static const char *DemangleStackConsumption(const char *mangled,
>+                                            int *stack_consumed) {
>+  g_mangled = mangled;
>+  *stack_consumed = GetSignalHandlerStackConsumption(DemangleSignalHandler);
>+  ABSL_RAW_LOG(INFO, "Stack consumption of Demangle: %d", *stack_consumed);
>+  return g_demangle_result;
>+}
>+
>+// Demangle stack consumption should be within 8kB for simple mangled names
>+// with some level of nesting. With alternate signal stack we have 64K,
>+// but some signal handlers run on thread stack, and could have arbitrarily
>+// little space left (so we don't want to make this number too large).
>+const int kStackConsumptionUpperLimit = 8192;
>+
>+// Returns a mangled name nested to the given depth.
>+static std::string NestedMangledName(int depth) {
>+  std::string mangled_name = "_Z1a";
>+  if (depth > 0) {
>+    mangled_name += "IXL";
>+    mangled_name += NestedMangledName(depth - 1);
>+    mangled_name += "EEE";
>+  }
>+  return mangled_name;
>+}
>+
>+TEST(Demangle, DemangleStackConsumption) {
>+  // Measure stack consumption of Demangle for nested mangled names of varying
>+  // depth.  Since Demangle is implemented as a recursive descent parser,
>+  // stack consumption will grow as the nesting depth increases.  By measuring
>+  // the stack consumption for increasing depths, we can see the growing
>+  // impact of any stack-saving changes made to the code for Demangle.
>+  int stack_consumed = 0;
>+
>+  const char *demangled =
>+      DemangleStackConsumption("_Z6foobarv", &stack_consumed);
>+  EXPECT_STREQ("foobar()", demangled);
>+  EXPECT_GT(stack_consumed, 0);
>+  EXPECT_LT(stack_consumed, kStackConsumptionUpperLimit);
>+
>+  const std::string nested_mangled_name0 = NestedMangledName(0);
>+  demangled = DemangleStackConsumption(nested_mangled_name0.c_str(),
>+                                       &stack_consumed);
>+  EXPECT_STREQ("a", demangled);
>+  EXPECT_GT(stack_consumed, 0);
>+  EXPECT_LT(stack_consumed, kStackConsumptionUpperLimit);
>+
>+  const std::string nested_mangled_name1 = NestedMangledName(1);
>+  demangled = DemangleStackConsumption(nested_mangled_name1.c_str(),
>+                                       &stack_consumed);
>+  EXPECT_STREQ("a<>", demangled);
>+  EXPECT_GT(stack_consumed, 0);
>+  EXPECT_LT(stack_consumed, kStackConsumptionUpperLimit);
>+
>+  const std::string nested_mangled_name2 = NestedMangledName(2);
>+  demangled = DemangleStackConsumption(nested_mangled_name2.c_str(),
>+                                       &stack_consumed);
>+  EXPECT_STREQ("a<>", demangled);
>+  EXPECT_GT(stack_consumed, 0);
>+  EXPECT_LT(stack_consumed, kStackConsumptionUpperLimit);
>+
>+  const std::string nested_mangled_name3 = NestedMangledName(3);
>+  demangled = DemangleStackConsumption(nested_mangled_name3.c_str(),
>+                                       &stack_consumed);
>+  EXPECT_STREQ("a<>", demangled);
>+  EXPECT_GT(stack_consumed, 0);
>+  EXPECT_LT(stack_consumed, kStackConsumptionUpperLimit);
>+}
>+
>+#endif  // Stack consumption tests
>+
>+static void TestOnInput(const char* input) {
>+  static const int kOutSize = 1048576;
>+  auto out = absl::make_unique<char[]>(kOutSize);
>+  Demangle(input, out.get(), kOutSize);
>+}
>+
>+TEST(DemangleRegression, NegativeLength) {
>+  TestOnInput("_ZZn4");
>+}
>+TEST(DemangleRegression, DeeplyNestedArrayType) {
>+  const int depth = 100000;
>+  std::string data = "_ZStI";
>+  data.reserve(data.size() + 3 * depth + 1);
>+  for (int i = 0; i < depth; i++) {
>+    data += "A1_";
>+  }
>+  TestOnInput(data.c_str());
>+}
>+
>+}  // namespace
>+}  // namespace debugging_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/elf_mem_image.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/elf_mem_image.cc
>new file mode 100644
>index 00000000000..3f747e7f958
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/elf_mem_image.cc
>@@ -0,0 +1,380 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+// Allow dynamic symbol lookup in an in-memory Elf image.
>+//
>+
>+#include "absl/debugging/internal/elf_mem_image.h"
>+
>+#ifdef ABSL_HAVE_ELF_MEM_IMAGE  // defined in elf_mem_image.h
>+
>+#include <string.h>
>+#include <cassert>
>+#include <cstddef>
>+#include "absl/base/internal/raw_logging.h"
>+
>+// From binutils/include/elf/common.h (this doesn't appear to be documented
>+// anywhere else).
>+//
>+//   /* This flag appears in a Versym structure.  It means that the symbol
>+//      is hidden, and is only visible with an explicit version number.
>+//      This is a GNU extension.  */
>+//   #define VERSYM_HIDDEN           0x8000
>+//
>+//   /* This is the mask for the rest of the Versym information.  */
>+//   #define VERSYM_VERSION          0x7fff
>+
>+#define VERSYM_VERSION 0x7fff
>+
>+namespace absl {
>+namespace debugging_internal {
>+
>+namespace {
>+
>+#if __WORDSIZE == 32
>+const int kElfClass = ELFCLASS32;
>+int ElfBind(const ElfW(Sym) *symbol) { return ELF32_ST_BIND(symbol->st_info); }
>+int ElfType(const ElfW(Sym) *symbol) { return ELF32_ST_TYPE(symbol->st_info); }
>+#elif __WORDSIZE == 64
>+const int kElfClass = ELFCLASS64;
>+int ElfBind(const ElfW(Sym) *symbol) { return ELF64_ST_BIND(symbol->st_info); }
>+int ElfType(const ElfW(Sym) *symbol) { return ELF64_ST_TYPE(symbol->st_info); }
>+#else
>+const int kElfClass = -1;
>+int ElfBind(const ElfW(Sym) *) {
>+  ABSL_RAW_LOG(FATAL, "Unexpected word size");
>+  return 0;
>+}
>+int ElfType(const ElfW(Sym) *) {
>+  ABSL_RAW_LOG(FATAL, "Unexpected word size");
>+  return 0;
>+}
>+#endif
>+
>+// Extract an element from one of the ELF tables, cast it to desired type.
>+// This is just a simple arithmetic and a glorified cast.
>+// Callers are responsible for bounds checking.
>+template <typename T>
>+const T *GetTableElement(const ElfW(Ehdr) * ehdr, ElfW(Off) table_offset,
>+                         ElfW(Word) element_size, size_t index) {
>+  return reinterpret_cast<const T*>(reinterpret_cast<const char *>(ehdr)
>+                                    + table_offset
>+                                    + index * element_size);
>+}
>+
>+}  // namespace
>+
>+// The value of this variable doesn't matter; it's used only for its
>+// unique address.
>+const int ElfMemImage::kInvalidBaseSentinel = 0;
>+
>+ElfMemImage::ElfMemImage(const void *base) {
>+  ABSL_RAW_CHECK(base != kInvalidBase, "bad pointer");
>+  Init(base);
>+}
>+
>+int ElfMemImage::GetNumSymbols() const {
>+  if (!hash_) {
>+    return 0;
>+  }
>+  // See http://www.caldera.com/developers/gabi/latest/ch5.dynamic.html#hash
>+  return hash_[1];
>+}
>+
>+const ElfW(Sym) *ElfMemImage::GetDynsym(int index) const {
>+  ABSL_RAW_CHECK(index < GetNumSymbols(), "index out of range");
>+  return dynsym_ + index;
>+}
>+
>+const ElfW(Versym) *ElfMemImage::GetVersym(int index) const {
>+  ABSL_RAW_CHECK(index < GetNumSymbols(), "index out of range");
>+  return versym_ + index;
>+}
>+
>+const ElfW(Phdr) *ElfMemImage::GetPhdr(int index) const {
>+  ABSL_RAW_CHECK(index < ehdr_->e_phnum, "index out of range");
>+  return GetTableElement<ElfW(Phdr)>(ehdr_,
>+                                     ehdr_->e_phoff,
>+                                     ehdr_->e_phentsize,
>+                                     index);
>+}
>+
>+const char *ElfMemImage::GetDynstr(ElfW(Word) offset) const {
>+  ABSL_RAW_CHECK(offset < strsize_, "offset out of range");
>+  return dynstr_ + offset;
>+}
>+
>+const void *ElfMemImage::GetSymAddr(const ElfW(Sym) *sym) const {
>+  if (sym->st_shndx == SHN_UNDEF || sym->st_shndx >= SHN_LORESERVE) {
>+    // Symbol corresponds to "special" (e.g. SHN_ABS) section.
>+    return reinterpret_cast<const void *>(sym->st_value);
>+  }
>+  ABSL_RAW_CHECK(link_base_ < sym->st_value, "symbol out of range");
>+  return GetTableElement<char>(ehdr_, 0, 1, sym->st_value - link_base_);
>+}
>+
>+const ElfW(Verdef) *ElfMemImage::GetVerdef(int index) const {
>+  ABSL_RAW_CHECK(0 <= index && static_cast<size_t>(index) <= verdefnum_,
>+                 "index out of range");
>+  const ElfW(Verdef) *version_definition = verdef_;
>+  while (version_definition->vd_ndx < index && version_definition->vd_next) {
>+    const char *const version_definition_as_char =
>+        reinterpret_cast<const char *>(version_definition);
>+    version_definition =
>+        reinterpret_cast<const ElfW(Verdef) *>(version_definition_as_char +
>+                                               version_definition->vd_next);
>+  }
>+  return version_definition->vd_ndx == index ? version_definition : nullptr;
>+}
>+
>+const ElfW(Verdaux) *ElfMemImage::GetVerdefAux(
>+    const ElfW(Verdef) *verdef) const {
>+  return reinterpret_cast<const ElfW(Verdaux) *>(verdef+1);
>+}
>+
>+const char *ElfMemImage::GetVerstr(ElfW(Word) offset) const {
>+  ABSL_RAW_CHECK(offset < strsize_, "offset out of range");
>+  return dynstr_ + offset;
>+}
>+
>+void ElfMemImage::Init(const void *base) {
>+  ehdr_      = nullptr;
>+  dynsym_    = nullptr;
>+  dynstr_    = nullptr;
>+  versym_    = nullptr;
>+  verdef_    = nullptr;
>+  hash_      = nullptr;
>+  strsize_   = 0;
>+  verdefnum_ = 0;
>+  link_base_ = ~0L;  // Sentinel: PT_LOAD .p_vaddr can't possibly be this.
>+  if (!base) {
>+    return;
>+  }
>+  const char *const base_as_char = reinterpret_cast<const char *>(base);
>+  if (base_as_char[EI_MAG0] != ELFMAG0 || base_as_char[EI_MAG1] != ELFMAG1 ||
>+      base_as_char[EI_MAG2] != ELFMAG2 || base_as_char[EI_MAG3] != ELFMAG3) {
>+    assert(false);
>+    return;
>+  }
>+  int elf_class = base_as_char[EI_CLASS];
>+  if (elf_class != kElfClass) {
>+    assert(false);
>+    return;
>+  }
>+  switch (base_as_char[EI_DATA]) {
>+    case ELFDATA2LSB: {
>+      if (__LITTLE_ENDIAN != __BYTE_ORDER) {
>+        assert(false);
>+        return;
>+      }
>+      break;
>+    }
>+    case ELFDATA2MSB: {
>+      if (__BIG_ENDIAN != __BYTE_ORDER) {
>+        assert(false);
>+        return;
>+      }
>+      break;
>+    }
>+    default: {
>+      assert(false);
>+      return;
>+    }
>+  }
>+
>+  ehdr_ = reinterpret_cast<const ElfW(Ehdr) *>(base);
>+  const ElfW(Phdr) *dynamic_program_header = nullptr;
>+  for (int i = 0; i < ehdr_->e_phnum; ++i) {
>+    const ElfW(Phdr) *const program_header = GetPhdr(i);
>+    switch (program_header->p_type) {
>+      case PT_LOAD:
>+        if (!~link_base_) {
>+          link_base_ = program_header->p_vaddr;
>+        }
>+        break;
>+      case PT_DYNAMIC:
>+        dynamic_program_header = program_header;
>+        break;
>+    }
>+  }
>+  if (!~link_base_ || !dynamic_program_header) {
>+    assert(false);
>+    // Mark this image as not present. Can not recur infinitely.
>+    Init(nullptr);
>+    return;
>+  }
>+  ptrdiff_t relocation =
>+      base_as_char - reinterpret_cast<const char *>(link_base_);
>+  ElfW(Dyn) *dynamic_entry =
>+      reinterpret_cast<ElfW(Dyn) *>(dynamic_program_header->p_vaddr +
>+                                    relocation);
>+  for (; dynamic_entry->d_tag != DT_NULL; ++dynamic_entry) {
>+    const ElfW(Xword) value = dynamic_entry->d_un.d_val + relocation;
>+    switch (dynamic_entry->d_tag) {
>+      case DT_HASH:
>+        hash_ = reinterpret_cast<ElfW(Word) *>(value);
>+        break;
>+      case DT_SYMTAB:
>+        dynsym_ = reinterpret_cast<ElfW(Sym) *>(value);
>+        break;
>+      case DT_STRTAB:
>+        dynstr_ = reinterpret_cast<const char *>(value);
>+        break;
>+      case DT_VERSYM:
>+        versym_ = reinterpret_cast<ElfW(Versym) *>(value);
>+        break;
>+      case DT_VERDEF:
>+        verdef_ = reinterpret_cast<ElfW(Verdef) *>(value);
>+        break;
>+      case DT_VERDEFNUM:
>+        verdefnum_ = dynamic_entry->d_un.d_val;
>+        break;
>+      case DT_STRSZ:
>+        strsize_ = dynamic_entry->d_un.d_val;
>+        break;
>+      default:
>+        // Unrecognized entries explicitly ignored.
>+        break;
>+    }
>+  }
>+  if (!hash_ || !dynsym_ || !dynstr_ || !versym_ ||
>+      !verdef_ || !verdefnum_ || !strsize_) {
>+    assert(false);  // invalid VDSO
>+    // Mark this image as not present. Can not recur infinitely.
>+    Init(nullptr);
>+    return;
>+  }
>+}
>+
>+bool ElfMemImage::LookupSymbol(const char *name,
>+                               const char *version,
>+                               int type,
>+                               SymbolInfo *info_out) const {
>+  for (const SymbolInfo& info : *this) {
>+    if (strcmp(info.name, name) == 0 && strcmp(info.version, version) == 0 &&
>+        ElfType(info.symbol) == type) {
>+      if (info_out) {
>+        *info_out = info;
>+      }
>+      return true;
>+    }
>+  }
>+  return false;
>+}
>+
>+bool ElfMemImage::LookupSymbolByAddress(const void *address,
>+                                        SymbolInfo *info_out) const {
>+  for (const SymbolInfo& info : *this) {
>+    const char *const symbol_start =
>+        reinterpret_cast<const char *>(info.address);
>+    const char *const symbol_end = symbol_start + info.symbol->st_size;
>+    if (symbol_start <= address && address < symbol_end) {
>+      if (info_out) {
>+        // Client wants to know details for that symbol (the usual case).
>+        if (ElfBind(info.symbol) == STB_GLOBAL) {
>+          // Strong symbol; just return it.
>+          *info_out = info;
>+          return true;
>+        } else {
>+          // Weak or local. Record it, but keep looking for a strong one.
>+          *info_out = info;
>+        }
>+      } else {
>+        // Client only cares if there is an overlapping symbol.
>+        return true;
>+      }
>+    }
>+  }
>+  return false;
>+}
>+
>+ElfMemImage::SymbolIterator::SymbolIterator(const void *const image, int index)
>+    : index_(index), image_(image) {
>+}
>+
>+const ElfMemImage::SymbolInfo *ElfMemImage::SymbolIterator::operator->() const {
>+  return &info_;
>+}
>+
>+const ElfMemImage::SymbolInfo& ElfMemImage::SymbolIterator::operator*() const {
>+  return info_;
>+}
>+
>+bool ElfMemImage::SymbolIterator::operator==(const SymbolIterator &rhs) const {
>+  return this->image_ == rhs.image_ && this->index_ == rhs.index_;
>+}
>+
>+bool ElfMemImage::SymbolIterator::operator!=(const SymbolIterator &rhs) const {
>+  return !(*this == rhs);
>+}
>+
>+ElfMemImage::SymbolIterator &ElfMemImage::SymbolIterator::operator++() {
>+  this->Update(1);
>+  return *this;
>+}
>+
>+ElfMemImage::SymbolIterator ElfMemImage::begin() const {
>+  SymbolIterator it(this, 0);
>+  it.Update(0);
>+  return it;
>+}
>+
>+ElfMemImage::SymbolIterator ElfMemImage::end() const {
>+  return SymbolIterator(this, GetNumSymbols());
>+}
>+
>+void ElfMemImage::SymbolIterator::Update(int increment) {
>+  const ElfMemImage *image = reinterpret_cast<const ElfMemImage *>(image_);
>+  ABSL_RAW_CHECK(image->IsPresent() || increment == 0, "");
>+  if (!image->IsPresent()) {
>+    return;
>+  }
>+  index_ += increment;
>+  if (index_ >= image->GetNumSymbols()) {
>+    index_ = image->GetNumSymbols();
>+    return;
>+  }
>+  const ElfW(Sym)    *symbol = image->GetDynsym(index_);
>+  const ElfW(Versym) *version_symbol = image->GetVersym(index_);
>+  ABSL_RAW_CHECK(symbol && version_symbol, "");
>+  const char *const symbol_name = image->GetDynstr(symbol->st_name);
>+  const ElfW(Versym) version_index = version_symbol[0] & VERSYM_VERSION;
>+  const ElfW(Verdef) *version_definition = nullptr;
>+  const char *version_name = "";
>+  if (symbol->st_shndx == SHN_UNDEF) {
>+    // Undefined symbols reference DT_VERNEED, not DT_VERDEF, and
>+    // version_index could well be greater than verdefnum_, so calling
>+    // GetVerdef(version_index) may trigger assertion.
>+  } else {
>+    version_definition = image->GetVerdef(version_index);
>+  }
>+  if (version_definition) {
>+    // I am expecting 1 or 2 auxiliary entries: 1 for the version itself,
>+    // optional 2nd if the version has a parent.
>+    ABSL_RAW_CHECK(
>+        version_definition->vd_cnt == 1 || version_definition->vd_cnt == 2,
>+        "wrong number of entries");
>+    const ElfW(Verdaux) *version_aux = image->GetVerdefAux(version_definition);
>+    version_name = image->GetVerstr(version_aux->vda_name);
>+  }
>+  info_.name    = symbol_name;
>+  info_.version = version_name;
>+  info_.address = image->GetSymAddr(symbol);
>+  info_.symbol  = symbol;
>+}
>+
>+}  // namespace debugging_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_HAVE_ELF_MEM_IMAGE
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/elf_mem_image.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/elf_mem_image.h
>new file mode 100644
>index 00000000000..3b577268954
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/elf_mem_image.h
>@@ -0,0 +1,130 @@
>+/*
>+ * Copyright 2017 The Abseil Authors.
>+ *
>+ * Licensed under the Apache License, Version 2.0 (the "License");
>+ * you may not use this file except in compliance with the License.
>+ * You may obtain a copy of the License at
>+ *
>+ *      http://www.apache.org/licenses/LICENSE-2.0
>+ *
>+ * Unless required by applicable law or agreed to in writing, software
>+ * distributed under the License is distributed on an "AS IS" BASIS,
>+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+ * See the License for the specific language governing permissions and
>+ * limitations under the License.
>+ */
>+
>+// Allow dynamic symbol lookup for in-memory Elf images.
>+
>+#ifndef ABSL_DEBUGGING_INTERNAL_ELF_MEM_IMAGE_H_
>+#define ABSL_DEBUGGING_INTERNAL_ELF_MEM_IMAGE_H_
>+
>+// Including this will define the __GLIBC__ macro if glibc is being
>+// used.
>+#include <climits>
>+
>+// Maybe one day we can rewrite this file not to require the elf
>+// symbol extensions in glibc, but for right now we need them.
>+#ifdef ABSL_HAVE_ELF_MEM_IMAGE
>+#error ABSL_HAVE_ELF_MEM_IMAGE cannot be directly set
>+#endif
>+
>+#if defined(__ELF__) && defined(__GLIBC__) && !defined(__native_client__) && \
>+    !defined(__asmjs__) && !defined(__wasm__)
>+#define ABSL_HAVE_ELF_MEM_IMAGE 1
>+#endif
>+
>+#if ABSL_HAVE_ELF_MEM_IMAGE
>+
>+#include <link.h>  // for ElfW
>+
>+namespace absl {
>+namespace debugging_internal {
>+
>+// An in-memory ELF image (may not exist on disk).
>+class ElfMemImage {
>+ private:
>+  // Sentinel: there could never be an elf image at &kInvalidBaseSentinel.
>+  static const int kInvalidBaseSentinel;
>+
>+ public:
>+  // Sentinel: there could never be an elf image at this address.
>+  static constexpr const void *const kInvalidBase =
>+    static_cast<const void*>(&kInvalidBaseSentinel);
>+
>+  // Information about a single vdso symbol.
>+  // All pointers are into .dynsym, .dynstr, or .text of the VDSO.
>+  // Do not free() them or modify through them.
>+  struct SymbolInfo {
>+    const char      *name;      // E.g. "__vdso_getcpu"
>+    const char      *version;   // E.g. "LINUX_2.6", could be ""
>+                                // for unversioned symbol.
>+    const void      *address;   // Relocated symbol address.
>+    const ElfW(Sym) *symbol;    // Symbol in the dynamic symbol table.
>+  };
>+
>+  // Supports iteration over all dynamic symbols.
>+  class SymbolIterator {
>+   public:
>+    friend class ElfMemImage;
>+    const SymbolInfo *operator->() const;
>+    const SymbolInfo &operator*() const;
>+    SymbolIterator& operator++();
>+    bool operator!=(const SymbolIterator &rhs) const;
>+    bool operator==(const SymbolIterator &rhs) const;
>+   private:
>+    SymbolIterator(const void *const image, int index);
>+    void Update(int incr);
>+    SymbolInfo info_;
>+    int index_;
>+    const void *const image_;
>+  };
>+
>+
>+  explicit ElfMemImage(const void *base);
>+  void                 Init(const void *base);
>+  bool                 IsPresent() const { return ehdr_ != nullptr; }
>+  const ElfW(Phdr)*    GetPhdr(int index) const;
>+  const ElfW(Sym)*     GetDynsym(int index) const;
>+  const ElfW(Versym)*  GetVersym(int index) const;
>+  const ElfW(Verdef)*  GetVerdef(int index) const;
>+  const ElfW(Verdaux)* GetVerdefAux(const ElfW(Verdef) *verdef) const;
>+  const char*          GetDynstr(ElfW(Word) offset) const;
>+  const void*          GetSymAddr(const ElfW(Sym) *sym) const;
>+  const char*          GetVerstr(ElfW(Word) offset) const;
>+  int                  GetNumSymbols() const;
>+
>+  SymbolIterator begin() const;
>+  SymbolIterator end() const;
>+
>+  // Look up versioned dynamic symbol in the image.
>+  // Returns false if image is not present, or doesn't contain given
>+  // symbol/version/type combination.
>+  // If info_out is non-null, additional details are filled in.
>+  bool LookupSymbol(const char *name, const char *version,
>+                    int symbol_type, SymbolInfo *info_out) const;
>+
>+  // Find info about symbol (if any) which overlaps given address.
>+  // Returns true if symbol was found; false if image isn't present
>+  // or doesn't have a symbol overlapping given address.
>+  // If info_out is non-null, additional details are filled in.
>+  bool LookupSymbolByAddress(const void *address, SymbolInfo *info_out) const;
>+
>+ private:
>+  const ElfW(Ehdr) *ehdr_;
>+  const ElfW(Sym) *dynsym_;
>+  const ElfW(Versym) *versym_;
>+  const ElfW(Verdef) *verdef_;
>+  const ElfW(Word) *hash_;
>+  const char *dynstr_;
>+  size_t strsize_;
>+  size_t verdefnum_;
>+  ElfW(Addr) link_base_;     // Link-time base (p_vaddr of first PT_LOAD).
>+};
>+
>+}  // namespace debugging_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_HAVE_ELF_MEM_IMAGE
>+
>+#endif  // ABSL_DEBUGGING_INTERNAL_ELF_MEM_IMAGE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/examine_stack.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/examine_stack.cc
>new file mode 100644
>index 00000000000..faf88836735
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/examine_stack.cc
>@@ -0,0 +1,153 @@
>+//
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+
>+#include "absl/debugging/internal/examine_stack.h"
>+
>+#ifndef _WIN32
>+#include <unistd.h>
>+#endif
>+
>+#include <csignal>
>+#include <cstdio>
>+
>+#include "absl/base/attributes.h"
>+#include "absl/base/internal/raw_logging.h"
>+#include "absl/base/macros.h"
>+#include "absl/debugging/stacktrace.h"
>+#include "absl/debugging/symbolize.h"
>+
>+namespace absl {
>+namespace debugging_internal {
>+
>+// Returns the program counter from signal context, nullptr if
>+// unknown. vuc is a ucontext_t*. We use void* to avoid the use of
>+// ucontext_t on non-POSIX systems.
>+void* GetProgramCounter(void* vuc) {
>+#ifdef __linux__
>+  if (vuc != nullptr) {
>+    ucontext_t* context = reinterpret_cast<ucontext_t*>(vuc);
>+#if defined(__aarch64__)
>+    return reinterpret_cast<void*>(context->uc_mcontext.pc);
>+#elif defined(__arm__)
>+    return reinterpret_cast<void*>(context->uc_mcontext.arm_pc);
>+#elif defined(__i386__)
>+    if (14 < ABSL_ARRAYSIZE(context->uc_mcontext.gregs))
>+      return reinterpret_cast<void*>(context->uc_mcontext.gregs[14]);
>+#elif defined(__mips__)
>+    return reinterpret_cast<void*>(context->uc_mcontext.pc);
>+#elif defined(__powerpc64__)
>+    return reinterpret_cast<void*>(context->uc_mcontext.gp_regs[32]);
>+#elif defined(__powerpc__)
>+    return reinterpret_cast<void*>(context->uc_mcontext.regs->nip);
>+#elif defined(__s390__) && !defined(__s390x__)
>+    return reinterpret_cast<void*>(context->uc_mcontext.psw.addr & 0x7fffffff);
>+#elif defined(__s390__) && defined(__s390x__)
>+    return reinterpret_cast<void*>(context->uc_mcontext.psw.addr);
>+#elif defined(__x86_64__)
>+    if (16 < ABSL_ARRAYSIZE(context->uc_mcontext.gregs))
>+      return reinterpret_cast<void*>(context->uc_mcontext.gregs[16]);
>+#else
>+#error "Undefined Architecture."
>+#endif
>+  }
>+#elif defined(__akaros__)
>+  auto* ctx = reinterpret_cast<struct user_context*>(vuc);
>+  return reinterpret_cast<void*>(get_user_ctx_pc(ctx));
>+#endif
>+  static_cast<void>(vuc);
>+  return nullptr;
>+}
>+
>+// The %p field width for printf() functions is two characters per byte,
>+// and two extra for the leading "0x".
>+static constexpr int kPrintfPointerFieldWidth = 2 + 2 * sizeof(void*);
>+
>+// Print a program counter, its stack frame size, and its symbol name.
>+// Note that there is a separate symbolize_pc argument. Return addresses may be
>+// at the end of the function, and this allows the caller to back up from pc if
>+// appropriate.
>+static void DumpPCAndFrameSizeAndSymbol(void (*writerfn)(const char*, void*),
>+                                        void* writerfn_arg, void* pc,
>+                                        void* symbolize_pc, int framesize,
>+                                        const char* const prefix) {
>+  char tmp[1024];
>+  const char* symbol = "(unknown)";
>+  if (absl::Symbolize(symbolize_pc, tmp, sizeof(tmp))) {
>+    symbol = tmp;
>+  }
>+  char buf[1024];
>+  if (framesize <= 0) {
>+    snprintf(buf, sizeof(buf), "%s@ %*p  (unknown)  %s\n", prefix,
>+             kPrintfPointerFieldWidth, pc, symbol);
>+  } else {
>+    snprintf(buf, sizeof(buf), "%s@ %*p  %9d  %s\n", prefix,
>+             kPrintfPointerFieldWidth, pc, framesize, symbol);
>+  }
>+  writerfn(buf, writerfn_arg);
>+}
>+
>+// Print a program counter and the corresponding stack frame size.
>+static void DumpPCAndFrameSize(void (*writerfn)(const char*, void*),
>+                               void* writerfn_arg, void* pc, int framesize,
>+                               const char* const prefix) {
>+  char buf[100];
>+  if (framesize <= 0) {
>+    snprintf(buf, sizeof(buf), "%s@ %*p  (unknown)\n", prefix,
>+             kPrintfPointerFieldWidth, pc);
>+  } else {
>+    snprintf(buf, sizeof(buf), "%s@ %*p  %9d\n", prefix,
>+             kPrintfPointerFieldWidth, pc, framesize);
>+  }
>+  writerfn(buf, writerfn_arg);
>+}
>+
>+void DumpPCAndFrameSizesAndStackTrace(
>+    void* pc, void* const stack[], int frame_sizes[], int depth,
>+    int min_dropped_frames, bool symbolize_stacktrace,
>+    void (*writerfn)(const char*, void*), void* writerfn_arg) {
>+  if (pc != nullptr) {
>+    // We don't know the stack frame size for PC, use 0.
>+    if (symbolize_stacktrace) {
>+      DumpPCAndFrameSizeAndSymbol(writerfn, writerfn_arg, pc, pc, 0, "PC: ");
>+    } else {
>+      DumpPCAndFrameSize(writerfn, writerfn_arg, pc, 0, "PC: ");
>+    }
>+  }
>+  for (int i = 0; i < depth; i++) {
>+    if (symbolize_stacktrace) {
>+      // Pass the previous address of pc as the symbol address because pc is a
>+      // return address, and an overrun may occur when the function ends with a
>+      // call to a function annotated noreturn (e.g. CHECK). Note that we don't
>+      // do this for pc above, as the adjustment is only correct for return
>+      // addresses.
>+      DumpPCAndFrameSizeAndSymbol(writerfn, writerfn_arg, stack[i],
>+                                  reinterpret_cast<char*>(stack[i]) - 1,
>+                                  frame_sizes[i], "    ");
>+    } else {
>+      DumpPCAndFrameSize(writerfn, writerfn_arg, stack[i], frame_sizes[i],
>+                         "    ");
>+    }
>+  }
>+  if (min_dropped_frames > 0) {
>+    char buf[100];
>+    snprintf(buf, sizeof(buf), "    @ ... and at least %d more frames\n",
>+             min_dropped_frames);
>+    writerfn(buf, writerfn_arg);
>+  }
>+}
>+
>+}  // namespace debugging_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/examine_stack.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/examine_stack.h
>new file mode 100644
>index 00000000000..a16c03b2748
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/examine_stack.h
>@@ -0,0 +1,38 @@
>+//
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+
>+#ifndef ABSL_DEBUGGING_INTERNAL_EXAMINE_STACK_H_
>+#define ABSL_DEBUGGING_INTERNAL_EXAMINE_STACK_H_
>+
>+namespace absl {
>+namespace debugging_internal {
>+
>+// Returns the program counter from signal context, or nullptr if
>+// unknown. `vuc` is a ucontext_t*. We use void* to avoid the use of
>+// ucontext_t on non-POSIX systems.
>+void* GetProgramCounter(void* vuc);
>+
>+// Uses `writerfn` to dump the program counter, stack trace, and stack
>+// frame sizes.
>+void DumpPCAndFrameSizesAndStackTrace(
>+    void* pc, void* const stack[], int frame_sizes[], int depth,
>+    int min_dropped_frames, bool symbolize_stacktrace,
>+    void (*writerfn)(const char*, void*), void* writerfn_arg);
>+
>+}  // namespace debugging_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_DEBUGGING_INTERNAL_EXAMINE_STACK_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/stack_consumption.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/stack_consumption.cc
>new file mode 100644
>index 00000000000..2b3b972ea53
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/stack_consumption.cc
>@@ -0,0 +1,172 @@
>+//
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/debugging/internal/stack_consumption.h"
>+
>+#ifdef ABSL_INTERNAL_HAVE_DEBUGGING_STACK_CONSUMPTION
>+
>+#include <signal.h>
>+#include <sys/mman.h>
>+#include <unistd.h>
>+
>+#include <string.h>
>+
>+#include "absl/base/attributes.h"
>+#include "absl/base/internal/raw_logging.h"
>+
>+namespace absl {
>+namespace debugging_internal {
>+namespace {
>+
>+// This code requires that we know the direction in which the stack
>+// grows. It is commonly believed that this can be detected by putting
>+// a variable on the stack and then passing its address to a function
>+// that compares the address of this variable to the address of a
>+// variable on the function's own stack. However, this is unspecified
>+// behavior in C++: If two pointers p and q of the same type point to
>+// different objects that are not members of the same object or
>+// elements of the same array or to different functions, or if only
>+// one of them is null, the results of p<q, p>q, p<=q, and p>=q are
>+// unspecified. Therefore, instead we hardcode the direction of the
>+// stack on platforms we know about.
>+#if defined(__i386__) || defined(__x86_64__) || defined(__ppc__)
>+constexpr bool kStackGrowsDown = true;
>+#else
>+#error Need to define kStackGrowsDown
>+#endif
>+
>+// To measure the stack footprint of some code, we create a signal handler
>+// (for SIGUSR2 say) that exercises this code on an alternate stack. This
>+// alternate stack is initialized to some known pattern (0x55, 0x55, 0x55,
>+// ...). We then self-send this signal, and after the signal handler returns,
>+// look at the alternate stack buffer to see what portion has been touched.
>+//
>+// This trick gives us the the stack footprint of the signal handler.  But the
>+// signal handler, even before the code for it is exercised, consumes some
>+// stack already. We however only want the stack usage of the code inside the
>+// signal handler. To measure this accurately, we install two signal handlers:
>+// one that does nothing and just returns, and the user-provided signal
>+// handler. The difference between the stack consumption of these two signals
>+// handlers should give us the stack foorprint of interest.
>+
>+void EmptySignalHandler(int) {}
>+
>+// This is arbitrary value, and could be increase further, at the cost of
>+// memset()ting it all to known sentinel value.
>+constexpr int kAlternateStackSize = 64 << 10;  // 64KiB
>+
>+constexpr int kSafetyMargin = 32;
>+constexpr char kAlternateStackFillValue = 0x55;
>+
>+// These helper functions look at the alternate stack buffer, and figure
>+// out what portion of this buffer has been touched - this is the stack
>+// consumption of the signal handler running on this alternate stack.
>+// This function will return -1 if the alternate stack buffer has not been
>+// touched. It will abort the program if the buffer has overflowed or is about
>+// to overflow.
>+int GetStackConsumption(const void* const altstack) {
>+  const char* begin;
>+  int increment;
>+  if (kStackGrowsDown) {
>+    begin = reinterpret_cast<const char*>(altstack);
>+    increment = 1;
>+  } else {
>+    begin = reinterpret_cast<const char*>(altstack) + kAlternateStackSize - 1;
>+    increment = -1;
>+  }
>+
>+  for (int usage_count = kAlternateStackSize; usage_count > 0; --usage_count) {
>+    if (*begin != kAlternateStackFillValue) {
>+      ABSL_RAW_CHECK(usage_count <= kAlternateStackSize - kSafetyMargin,
>+                     "Buffer has overflowed or is about to overflow");
>+      return usage_count;
>+    }
>+    begin += increment;
>+  }
>+
>+  ABSL_RAW_LOG(FATAL, "Unreachable code");
>+  return -1;
>+}
>+
>+}  // namespace
>+
>+int GetSignalHandlerStackConsumption(void (*signal_handler)(int)) {
>+  // The alt-signal-stack cannot be heap allocated because there is a
>+  // bug in glibc-2.2 where some signal handler setup code looks at the
>+  // current stack pointer to figure out what thread is currently running.
>+  // Therefore, the alternate stack must be allocated from the main stack
>+  // itself.
>+  void* altstack = mmap(nullptr, kAlternateStackSize, PROT_READ | PROT_WRITE,
>+                        MAP_PRIVATE | MAP_ANONYMOUS, -1, 0);
>+  ABSL_RAW_CHECK(altstack != MAP_FAILED, "mmap() failed");
>+
>+  // Set up the alt-signal-stack (and save the older one).
>+  stack_t sigstk;
>+  memset(&sigstk, 0, sizeof(sigstk));
>+  stack_t old_sigstk;
>+  sigstk.ss_sp = altstack;
>+  sigstk.ss_size = kAlternateStackSize;
>+  sigstk.ss_flags = 0;
>+  ABSL_RAW_CHECK(sigaltstack(&sigstk, &old_sigstk) == 0,
>+                 "sigaltstack() failed");
>+
>+  // Set up SIGUSR1 and SIGUSR2 signal handlers (and save the older ones).
>+  struct sigaction sa;
>+  memset(&sa, 0, sizeof(sa));
>+  struct sigaction old_sa1, old_sa2;
>+  sigemptyset(&sa.sa_mask);
>+  sa.sa_flags = SA_ONSTACK;
>+
>+  // SIGUSR1 maps to EmptySignalHandler.
>+  sa.sa_handler = EmptySignalHandler;
>+  ABSL_RAW_CHECK(sigaction(SIGUSR1, &sa, &old_sa1) == 0, "sigaction() failed");
>+
>+  // SIGUSR2 maps to signal_handler.
>+  sa.sa_handler = signal_handler;
>+  ABSL_RAW_CHECK(sigaction(SIGUSR2, &sa, &old_sa2) == 0, "sigaction() failed");
>+
>+  // Send SIGUSR1 signal and measure the stack consumption of the empty
>+  // signal handler.
>+  // The first signal might use more stack space. Run once and ignore the
>+  // results to get that out of the way.
>+  ABSL_RAW_CHECK(kill(getpid(), SIGUSR1) == 0, "kill() failed");
>+
>+  memset(altstack, kAlternateStackFillValue, kAlternateStackSize);
>+  ABSL_RAW_CHECK(kill(getpid(), SIGUSR1) == 0, "kill() failed");
>+  int base_stack_consumption = GetStackConsumption(altstack);
>+
>+  // Send SIGUSR2 signal and measure the stack consumption of signal_handler.
>+  ABSL_RAW_CHECK(kill(getpid(), SIGUSR2) == 0, "kill() failed");
>+  int signal_handler_stack_consumption = GetStackConsumption(altstack);
>+
>+  // Now restore the old alt-signal-stack and signal handlers.
>+  ABSL_RAW_CHECK(sigaltstack(&old_sigstk, nullptr) == 0,
>+                 "sigaltstack() failed");
>+  ABSL_RAW_CHECK(sigaction(SIGUSR1, &old_sa1, nullptr) == 0,
>+                 "sigaction() failed");
>+  ABSL_RAW_CHECK(sigaction(SIGUSR2, &old_sa2, nullptr) == 0,
>+                 "sigaction() failed");
>+
>+  ABSL_RAW_CHECK(munmap(altstack, kAlternateStackSize) == 0, "munmap() failed");
>+  if (signal_handler_stack_consumption != -1 && base_stack_consumption != -1) {
>+    return signal_handler_stack_consumption - base_stack_consumption;
>+  }
>+  return -1;
>+}
>+
>+}  // namespace debugging_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_INTERNAL_HAVE_DEBUGGING_STACK_CONSUMPTION
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/stack_consumption.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/stack_consumption.h
>new file mode 100644
>index 00000000000..4c5fa0f0d82
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/stack_consumption.h
>@@ -0,0 +1,45 @@
>+//
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+// Helper function for measuring stack consumption of signal handlers.
>+
>+#ifndef ABSL_DEBUGGING_INTERNAL_STACK_CONSUMPTION_H_
>+#define ABSL_DEBUGGING_INTERNAL_STACK_CONSUMPTION_H_
>+
>+// The code in this module is not portable.
>+// Use this feature test macro to detect its availability.
>+#ifdef ABSL_INTERNAL_HAVE_DEBUGGING_STACK_CONSUMPTION
>+#error ABSL_INTERNAL_HAVE_DEBUGGING_STACK_CONSUMPTION cannot be set directly
>+#elif !defined(__APPLE__) && !defined(_WIN32) && \
>+    (defined(__i386__) || defined(__x86_64__) || defined(__ppc__))
>+#define ABSL_INTERNAL_HAVE_DEBUGGING_STACK_CONSUMPTION 1
>+
>+namespace absl {
>+namespace debugging_internal {
>+
>+// Returns the stack consumption in bytes for the code exercised by
>+// signal_handler.  To measure stack consumption, signal_handler is registered
>+// as a signal handler, so the code that it exercises must be async-signal
>+// safe.  The argument of signal_handler is an implementation detail of signal
>+// handlers and should ignored by the code for signal_handler.  Use global
>+// variables to pass information between your test code and signal_handler.
>+int GetSignalHandlerStackConsumption(void (*signal_handler)(int));
>+
>+}  // namespace debugging_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_INTERNAL_HAVE_DEBUGGING_STACK_CONSUMPTION
>+
>+#endif  // ABSL_DEBUGGING_INTERNAL_STACK_CONSUMPTION_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/stack_consumption_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/stack_consumption_test.cc
>new file mode 100644
>index 00000000000..5ce3846ec44
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/stack_consumption_test.cc
>@@ -0,0 +1,48 @@
>+//
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/debugging/internal/stack_consumption.h"
>+
>+#ifdef ABSL_INTERNAL_HAVE_DEBUGGING_STACK_CONSUMPTION
>+
>+#include <string.h>
>+
>+#include "gtest/gtest.h"
>+#include "absl/base/internal/raw_logging.h"
>+
>+namespace absl {
>+namespace debugging_internal {
>+namespace {
>+
>+static void SimpleSignalHandler(int signo) {
>+  char buf[100];
>+  memset(buf, 'a', sizeof(buf));
>+
>+  // Never true, but prevents compiler from optimizing buf out.
>+  if (signo == 0) {
>+    ABSL_RAW_LOG(INFO, "%p", static_cast<void*>(buf));
>+  }
>+}
>+
>+TEST(SignalHandlerStackConsumptionTest, MeasuresStackConsumption) {
>+  // Our handler should consume reasonable number of bytes.
>+  EXPECT_GE(GetSignalHandlerStackConsumption(SimpleSignalHandler), 100);
>+}
>+
>+}  // namespace
>+}  // namespace debugging_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_INTERNAL_HAVE_DEBUGGING_STACK_CONSUMPTION
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/stacktrace_aarch64-inl.inc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/stacktrace_aarch64-inl.inc
>new file mode 100644
>index 00000000000..7ed6b3eb82d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/stacktrace_aarch64-inl.inc
>@@ -0,0 +1,190 @@
>+#ifndef ABSL_DEBUGGING_INTERNAL_STACKTRACE_AARCH64_INL_H_
>+#define ABSL_DEBUGGING_INTERNAL_STACKTRACE_AARCH64_INL_H_
>+
>+// Generate stack tracer for aarch64
>+
>+#if defined(__linux__)
>+#include <sys/mman.h>
>+#include <ucontext.h>
>+#include <unistd.h>
>+#endif
>+
>+#include <atomic>
>+#include <cassert>
>+#include <cstdint>
>+#include <iostream>
>+
>+#include "absl/base/attributes.h"
>+#include "absl/debugging/internal/address_is_readable.h"
>+#include "absl/debugging/internal/vdso_support.h"  // a no-op on non-elf or non-glibc systems
>+#include "absl/debugging/stacktrace.h"
>+
>+static const uintptr_t kUnknownFrameSize = 0;
>+
>+#if defined(__linux__)
>+// Returns the address of the VDSO __kernel_rt_sigreturn function, if present.
>+static const unsigned char* GetKernelRtSigreturnAddress() {
>+  constexpr uintptr_t kImpossibleAddress = 1;
>+  ABSL_CONST_INIT static std::atomic<uintptr_t> memoized{kImpossibleAddress};
>+  uintptr_t address = memoized.load(std::memory_order_relaxed);
>+  if (address != kImpossibleAddress) {
>+    return reinterpret_cast<const unsigned char*>(address);
>+  }
>+
>+  address = reinterpret_cast<uintptr_t>(nullptr);
>+
>+#ifdef ABSL_HAVE_VDSO_SUPPORT
>+  absl::debugging_internal::VDSOSupport vdso;
>+  if (vdso.IsPresent()) {
>+    absl::debugging_internal::VDSOSupport::SymbolInfo symbol_info;
>+    if (!vdso.LookupSymbol("__kernel_rt_sigreturn", "LINUX_2.6.39", STT_FUNC,
>+                           &symbol_info) ||
>+        symbol_info.address == nullptr) {
>+      // Unexpected: VDSO is present, yet the expected symbol is missing
>+      // or null.
>+      assert(false && "VDSO is present, but doesn't have expected symbol");
>+    } else {
>+      if (reinterpret_cast<uintptr_t>(symbol_info.address) !=
>+          kImpossibleAddress) {
>+        address = reinterpret_cast<uintptr_t>(symbol_info.address);
>+      } else {
>+        assert(false && "VDSO returned invalid address");
>+      }
>+    }
>+  }
>+#endif
>+
>+  memoized.store(address, std::memory_order_relaxed);
>+  return reinterpret_cast<const unsigned char*>(address);
>+}
>+#endif  // __linux__
>+
>+// Compute the size of a stack frame in [low..high).  We assume that
>+// low < high.  Return size of kUnknownFrameSize.
>+template<typename T>
>+static inline uintptr_t ComputeStackFrameSize(const T* low,
>+                                              const T* high) {
>+  const char* low_char_ptr = reinterpret_cast<const char *>(low);
>+  const char* high_char_ptr = reinterpret_cast<const char *>(high);
>+  return low < high ? high_char_ptr - low_char_ptr : kUnknownFrameSize;
>+}
>+
>+// Given a pointer to a stack frame, locate and return the calling
>+// stackframe, or return null if no stackframe can be found. Perform sanity
>+// checks (the strictness of which is controlled by the boolean parameter
>+// "STRICT_UNWINDING") to reduce the chance that a bad pointer is returned.
>+template<bool STRICT_UNWINDING, bool WITH_CONTEXT>
>+static void **NextStackFrame(void **old_frame_pointer, const void *uc) {
>+  void **new_frame_pointer = reinterpret_cast<void**>(*old_frame_pointer);
>+  bool check_frame_size = true;
>+
>+#if defined(__linux__)
>+  if (WITH_CONTEXT && uc != nullptr) {
>+    // Check to see if next frame's return address is __kernel_rt_sigreturn.
>+    if (old_frame_pointer[1] == GetKernelRtSigreturnAddress()) {
>+      const ucontext_t *ucv = static_cast<const ucontext_t *>(uc);
>+      // old_frame_pointer[0] is not suitable for unwinding, look at
>+      // ucontext to discover frame pointer before signal.
>+      void **const pre_signal_frame_pointer =
>+          reinterpret_cast<void **>(ucv->uc_mcontext.regs[29]);
>+
>+      // Check that alleged frame pointer is actually readable. This is to
>+      // prevent "double fault" in case we hit the first fault due to e.g.
>+      // stack corruption.
>+      if (!absl::debugging_internal::AddressIsReadable(
>+              pre_signal_frame_pointer))
>+        return nullptr;
>+
>+      // Alleged frame pointer is readable, use it for further unwinding.
>+      new_frame_pointer = pre_signal_frame_pointer;
>+
>+      // Skip frame size check if we return from a signal. We may be using a
>+      // an alternate stack for signals.
>+      check_frame_size = false;
>+    }
>+  }
>+#endif
>+
>+  // aarch64 ABI requires stack pointer to be 16-byte-aligned.
>+  if ((reinterpret_cast<uintptr_t>(new_frame_pointer) & 15) != 0)
>+    return nullptr;
>+
>+  // Check frame size.  In strict mode, we assume frames to be under
>+  // 100,000 bytes.  In non-strict mode, we relax the limit to 1MB.
>+  if (check_frame_size) {
>+    const uintptr_t max_size = STRICT_UNWINDING ? 100000 : 1000000;
>+    const uintptr_t frame_size =
>+        ComputeStackFrameSize(old_frame_pointer, new_frame_pointer);
>+    if (frame_size == kUnknownFrameSize || frame_size > max_size)
>+      return nullptr;
>+  }
>+
>+  return new_frame_pointer;
>+}
>+
>+template <bool IS_STACK_FRAMES, bool IS_WITH_CONTEXT>
>+static int UnwindImpl(void** result, int* sizes, int max_depth, int skip_count,
>+                      const void *ucp, int *min_dropped_frames) {
>+#ifdef __GNUC__
>+  void **frame_pointer = reinterpret_cast<void**>(__builtin_frame_address(0));
>+#else
>+# error reading stack point not yet supported on this platform.
>+#endif
>+
>+  skip_count++;    // Skip the frame for this function.
>+  int n = 0;
>+
>+  // The frame pointer points to low address of a frame.  The first 64-bit
>+  // word of a frame points to the next frame up the call chain, which normally
>+  // is just after the high address of the current frame.  The second word of
>+  // a frame contains return adress of to the caller.   To find a pc value
>+  // associated with the current frame, we need to go down a level in the call
>+  // chain.  So we remember return the address of the last frame seen.  This
>+  // does not work for the first stack frame, which belongs to UnwindImp() but
>+  // we skip the frame for UnwindImp() anyway.
>+  void* prev_return_address = nullptr;
>+
>+  while (frame_pointer && n < max_depth) {
>+    // The absl::GetStackFrames routine is called when we are in some
>+    // informational context (the failure signal handler for example).
>+    // Use the non-strict unwinding rules to produce a stack trace
>+    // that is as complete as possible (even if it contains a few bogus
>+    // entries in some rare cases).
>+    void **next_frame_pointer =
>+        NextStackFrame<!IS_STACK_FRAMES, IS_WITH_CONTEXT>(frame_pointer, ucp);
>+
>+    if (skip_count > 0) {
>+      skip_count--;
>+    } else {
>+      result[n] = prev_return_address;
>+      if (IS_STACK_FRAMES) {
>+        sizes[n] = ComputeStackFrameSize(frame_pointer, next_frame_pointer);
>+      }
>+      n++;
>+    }
>+    prev_return_address = frame_pointer[1];
>+    frame_pointer = next_frame_pointer;
>+  }
>+  if (min_dropped_frames != nullptr) {
>+    // Implementation detail: we clamp the max of frames we are willing to
>+    // count, so as not to spend too much time in the loop below.
>+    const int kMaxUnwind = 200;
>+    int j = 0;
>+    for (; frame_pointer != nullptr && j < kMaxUnwind; j++) {
>+      frame_pointer =
>+          NextStackFrame<!IS_STACK_FRAMES, IS_WITH_CONTEXT>(frame_pointer, ucp);
>+    }
>+    *min_dropped_frames = j;
>+  }
>+  return n;
>+}
>+
>+namespace absl {
>+namespace debugging_internal {
>+bool StackTraceWorksForTest() {
>+  return true;
>+}
>+}  // namespace debugging_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_DEBUGGING_INTERNAL_STACKTRACE_AARCH64_INL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/stacktrace_arm-inl.inc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/stacktrace_arm-inl.inc
>new file mode 100644
>index 00000000000..c84083379bb
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/stacktrace_arm-inl.inc
>@@ -0,0 +1,123 @@
>+// Copyright 2011 and onwards Google Inc.
>+// All rights reserved.
>+//
>+// Author: Doug Kwan
>+// This is inspired by Craig Silverstein's PowerPC stacktrace code.
>+//
>+
>+#ifndef ABSL_DEBUGGING_INTERNAL_STACKTRACE_ARM_INL_H_
>+#define ABSL_DEBUGGING_INTERNAL_STACKTRACE_ARM_INL_H_
>+
>+#include <cstdint>
>+
>+#include "absl/debugging/stacktrace.h"
>+
>+// WARNING:
>+// This only works if all your code is in either ARM or THUMB mode.  With
>+// interworking, the frame pointer of the caller can either be in r11 (ARM
>+// mode) or r7 (THUMB mode).  A callee only saves the frame pointer of its
>+// mode in a fixed location on its stack frame.  If the caller is a different
>+// mode, there is no easy way to find the frame pointer.  It can either be
>+// still in the designated register or saved on stack along with other callee
>+// saved registers.
>+
>+// Given a pointer to a stack frame, locate and return the calling
>+// stackframe, or return nullptr if no stackframe can be found. Perform sanity
>+// checks (the strictness of which is controlled by the boolean parameter
>+// "STRICT_UNWINDING") to reduce the chance that a bad pointer is returned.
>+template<bool STRICT_UNWINDING>
>+static void **NextStackFrame(void **old_sp) {
>+  void **new_sp = (void**) old_sp[-1];
>+
>+  // Check that the transition from frame pointer old_sp to frame
>+  // pointer new_sp isn't clearly bogus
>+  if (STRICT_UNWINDING) {
>+    // With the stack growing downwards, older stack frame must be
>+    // at a greater address that the current one.
>+    if (new_sp <= old_sp) return nullptr;
>+    // Assume stack frames larger than 100,000 bytes are bogus.
>+    if ((uintptr_t)new_sp - (uintptr_t)old_sp > 100000) return nullptr;
>+  } else {
>+    // In the non-strict mode, allow discontiguous stack frames.
>+    // (alternate-signal-stacks for example).
>+    if (new_sp == old_sp) return nullptr;
>+    // And allow frames upto about 1MB.
>+    if ((new_sp > old_sp)
>+        && ((uintptr_t)new_sp - (uintptr_t)old_sp > 1000000)) return nullptr;
>+  }
>+  if ((uintptr_t)new_sp & (sizeof(void *) - 1)) return nullptr;
>+  return new_sp;
>+}
>+
>+// This ensures that absl::GetStackTrace sets up the Link Register properly.
>+#ifdef __GNUC__
>+void StacktraceArmDummyFunction() __attribute__((noinline));
>+void StacktraceArmDummyFunction() { __asm__ volatile(""); }
>+#else
>+# error StacktraceArmDummyFunction() needs to be ported to this platform.
>+#endif
>+
>+template <bool IS_STACK_FRAMES, bool IS_WITH_CONTEXT>
>+static int UnwindImpl(void** result, int* sizes, int max_depth, int skip_count,
>+                      const void * /* ucp */, int *min_dropped_frames) {
>+#ifdef __GNUC__
>+  void **sp = reinterpret_cast<void**>(__builtin_frame_address(0));
>+#else
>+# error reading stack point not yet supported on this platform.
>+#endif
>+
>+  // On ARM, the return address is stored in the link register (r14).
>+  // This is not saved on the stack frame of a leaf function.  To
>+  // simplify code that reads return addresses, we call a dummy
>+  // function so that the return address of this function is also
>+  // stored in the stack frame.  This works at least for gcc.
>+  StacktraceArmDummyFunction();
>+
>+  int n = 0;
>+  while (sp && n < max_depth) {
>+    // The absl::GetStackFrames routine is called when we are in some
>+    // informational context (the failure signal handler for example).
>+    // Use the non-strict unwinding rules to produce a stack trace
>+    // that is as complete as possible (even if it contains a few bogus
>+    // entries in some rare cases).
>+    void **next_sp = NextStackFrame<!IS_STACK_FRAMES>(sp);
>+
>+    if (skip_count > 0) {
>+      skip_count--;
>+    } else {
>+      result[n] = *sp;
>+
>+      if (IS_STACK_FRAMES) {
>+        if (next_sp > sp) {
>+          sizes[n] = (uintptr_t)next_sp - (uintptr_t)sp;
>+        } else {
>+          // A frame-size of 0 is used to indicate unknown frame size.
>+          sizes[n] = 0;
>+        }
>+      }
>+      n++;
>+    }
>+    sp = next_sp;
>+  }
>+  if (min_dropped_frames != nullptr) {
>+    // Implementation detail: we clamp the max of frames we are willing to
>+    // count, so as not to spend too much time in the loop below.
>+    const int kMaxUnwind = 200;
>+    int j = 0;
>+    for (; sp != nullptr && j < kMaxUnwind; j++) {
>+      sp = NextStackFrame<!IS_STACK_FRAMES>(sp);
>+    }
>+    *min_dropped_frames = j;
>+  }
>+  return n;
>+}
>+
>+namespace absl {
>+namespace debugging_internal {
>+bool StackTraceWorksForTest() {
>+  return false;
>+}
>+}  // namespace debugging_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_DEBUGGING_INTERNAL_STACKTRACE_ARM_INL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/stacktrace_config.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/stacktrace_config.h
>new file mode 100644
>index 00000000000..dd713da8c0c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/stacktrace_config.h
>@@ -0,0 +1,69 @@
>+/*
>+ * Copyright 2017 The Abseil Authors.
>+ *
>+ * Licensed under the Apache License, Version 2.0 (the "License");
>+ * you may not use this file except in compliance with the License.
>+ * You may obtain a copy of the License at
>+ *
>+ *      http://www.apache.org/licenses/LICENSE-2.0
>+ *
>+ * Unless required by applicable law or agreed to in writing, software
>+ * distributed under the License is distributed on an "AS IS" BASIS,
>+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+ * See the License for the specific language governing permissions and
>+ * limitations under the License.
>+
>+ * Defines ABSL_STACKTRACE_INL_HEADER to the *-inl.h containing
>+ * actual unwinder implementation.
>+ * This header is "private" to stacktrace.cc.
>+ * DO NOT include it into any other files.
>+*/
>+#ifndef ABSL_DEBUGGING_INTERNAL_STACKTRACE_CONFIG_H_
>+#define ABSL_DEBUGGING_INTERNAL_STACKTRACE_CONFIG_H_
>+
>+#if defined(ABSL_STACKTRACE_INL_HEADER)
>+#error ABSL_STACKTRACE_INL_HEADER cannot be directly set
>+
>+#elif defined(_WIN32)
>+#define ABSL_STACKTRACE_INL_HEADER \
>+    "absl/debugging/internal/stacktrace_win32-inl.inc"
>+
>+#elif defined(__linux__) && !defined(__ANDROID__)
>+
>+#if !defined(NO_FRAME_POINTER)
>+# if defined(__i386__) || defined(__x86_64__)
>+#define ABSL_STACKTRACE_INL_HEADER \
>+    "absl/debugging/internal/stacktrace_x86-inl.inc"
>+# elif defined(__ppc__) || defined(__PPC__)
>+#define ABSL_STACKTRACE_INL_HEADER \
>+    "absl/debugging/internal/stacktrace_powerpc-inl.inc"
>+# elif defined(__aarch64__)
>+#define ABSL_STACKTRACE_INL_HEADER \
>+    "absl/debugging/internal/stacktrace_aarch64-inl.inc"
>+# elif defined(__arm__)
>+#define ABSL_STACKTRACE_INL_HEADER \
>+    "absl/debugging/internal/stacktrace_arm-inl.inc"
>+# else
>+#define ABSL_STACKTRACE_INL_HEADER \
>+   "absl/debugging/internal/stacktrace_unimplemented-inl.inc"
>+# endif
>+#else  // defined(NO_FRAME_POINTER)
>+# if defined(__i386__) || defined(__x86_64__) || defined(__aarch64__)
>+#define ABSL_STACKTRACE_INL_HEADER \
>+    "absl/debugging/internal/stacktrace_generic-inl.inc"
>+# elif defined(__ppc__) || defined(__PPC__)
>+#define ABSL_STACKTRACE_INL_HEADER \
>+    "absl/debugging/internal/stacktrace_generic-inl.inc"
>+# else
>+#define ABSL_STACKTRACE_INL_HEADER \
>+   "absl/debugging/internal/stacktrace_unimplemented-inl.inc"
>+# endif
>+#endif  // NO_FRAME_POINTER
>+
>+#else
>+#define ABSL_STACKTRACE_INL_HEADER \
>+  "absl/debugging/internal/stacktrace_unimplemented-inl.inc"
>+
>+#endif
>+
>+#endif  // ABSL_DEBUGGING_INTERNAL_STACKTRACE_CONFIG_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/stacktrace_generic-inl.inc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/stacktrace_generic-inl.inc
>new file mode 100644
>index 00000000000..2c9ca410c60
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/stacktrace_generic-inl.inc
>@@ -0,0 +1,59 @@
>+// Copyright 2000 - 2007 Google Inc.
>+// All rights reserved.
>+//
>+// Author: Sanjay Ghemawat
>+//
>+// Portable implementation - just use glibc
>+//
>+// Note:  The glibc implementation may cause a call to malloc.
>+// This can cause a deadlock in HeapProfiler.
>+
>+#ifndef ABSL_DEBUGGING_INTERNAL_STACKTRACE_GENERIC_INL_H_
>+#define ABSL_DEBUGGING_INTERNAL_STACKTRACE_GENERIC_INL_H_
>+
>+#include <execinfo.h>
>+#include <cstring>
>+
>+#include "absl/debugging/stacktrace.h"
>+
>+template <bool IS_STACK_FRAMES, bool IS_WITH_CONTEXT>
>+static int UnwindImpl(void** result, int* sizes, int max_depth, int skip_count,
>+                      const void *ucp, int *min_dropped_frames) {
>+  static const int kStackLength = 64;
>+  void * stack[kStackLength];
>+  int size;
>+
>+  size = backtrace(stack, kStackLength);
>+  skip_count++;  // we want to skip the current frame as well
>+  int result_count = size - skip_count;
>+  if (result_count < 0)
>+    result_count = 0;
>+  if (result_count > max_depth)
>+    result_count = max_depth;
>+  for (int i = 0; i < result_count; i++)
>+    result[i] = stack[i + skip_count];
>+
>+  if (IS_STACK_FRAMES) {
>+    // No implementation for finding out the stack frame sizes yet.
>+    memset(sizes, 0, sizeof(*sizes) * result_count);
>+  }
>+  if (min_dropped_frames != nullptr) {
>+    if (size - skip_count - max_depth > 0) {
>+      *min_dropped_frames = size - skip_count - max_depth;
>+    } else {
>+      *min_dropped_frames = 0;
>+    }
>+  }
>+
>+  return result_count;
>+}
>+
>+namespace absl {
>+namespace debugging_internal {
>+bool StackTraceWorksForTest() {
>+  return true;
>+}
>+}  // namespace debugging_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_DEBUGGING_INTERNAL_STACKTRACE_GENERIC_INL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/stacktrace_powerpc-inl.inc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/stacktrace_powerpc-inl.inc
>new file mode 100644
>index 00000000000..860ac2b3e58
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/stacktrace_powerpc-inl.inc
>@@ -0,0 +1,246 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// Produce stack trace.  I'm guessing (hoping!) the code is much like
>+// for x86.  For apple machines, at least, it seems to be; see
>+//    http://developer.apple.com/documentation/mac/runtimehtml/RTArch-59.html
>+//    http://www.linux-foundation.org/spec/ELF/ppc64/PPC-elf64abi-1.9.html#STACK
>+// Linux has similar code: http://patchwork.ozlabs.org/linuxppc/patch?id=8882
>+
>+#ifndef ABSL_DEBUGGING_INTERNAL_STACKTRACE_POWERPC_INL_H_
>+#define ABSL_DEBUGGING_INTERNAL_STACKTRACE_POWERPC_INL_H_
>+
>+#if defined(__linux__)
>+#include <asm/ptrace.h>   // for PT_NIP.
>+#include <ucontext.h>     // for ucontext_t
>+#endif
>+
>+#include <unistd.h>
>+#include <cassert>
>+#include <cstdint>
>+#include <cstdio>
>+
>+#include "absl/base/attributes.h"
>+#include "absl/base/optimization.h"
>+#include "absl/base/port.h"
>+#include "absl/debugging/stacktrace.h"
>+#include "absl/debugging/internal/address_is_readable.h"
>+#include "absl/debugging/internal/vdso_support.h"  // a no-op on non-elf or non-glibc systems
>+
>+// Given a stack pointer, return the saved link register value.
>+// Note that this is the link register for a callee.
>+static inline void *StacktracePowerPCGetLR(void **sp) {
>+  // PowerPC has 3 main ABIs, which say where in the stack the
>+  // Link Register is.  For DARWIN and AIX (used by apple and
>+  // linux ppc64), it's in sp[2].  For SYSV (used by linux ppc),
>+  // it's in sp[1].
>+#if defined(_CALL_AIX) || defined(_CALL_DARWIN)
>+  return *(sp+2);
>+#elif defined(_CALL_SYSV)
>+  return *(sp+1);
>+#elif defined(__APPLE__) || defined(__FreeBSD__) || \
>+    (defined(__linux__) && defined(__PPC64__))
>+  // This check is in case the compiler doesn't define _CALL_AIX/etc.
>+  return *(sp+2);
>+#elif defined(__linux)
>+  // This check is in case the compiler doesn't define _CALL_SYSV.
>+  return *(sp+1);
>+#else
>+#error Need to specify the PPC ABI for your archiecture.
>+#endif
>+}
>+
>+// Given a pointer to a stack frame, locate and return the calling
>+// stackframe, or return null if no stackframe can be found. Perform sanity
>+// checks (the strictness of which is controlled by the boolean parameter
>+// "STRICT_UNWINDING") to reduce the chance that a bad pointer is returned.
>+template<bool STRICT_UNWINDING, bool IS_WITH_CONTEXT>
>+ABSL_ATTRIBUTE_NO_SANITIZE_ADDRESS  // May read random elements from stack.
>+ABSL_ATTRIBUTE_NO_SANITIZE_MEMORY   // May read random elements from stack.
>+static void **NextStackFrame(void **old_sp, const void *uc) {
>+  void **new_sp = (void **) *old_sp;
>+  enum { kStackAlignment = 16 };
>+
>+  // Check that the transition from frame pointer old_sp to frame
>+  // pointer new_sp isn't clearly bogus
>+  if (STRICT_UNWINDING) {
>+    // With the stack growing downwards, older stack frame must be
>+    // at a greater address that the current one.
>+    if (new_sp <= old_sp) return nullptr;
>+    // Assume stack frames larger than 100,000 bytes are bogus.
>+    if ((uintptr_t)new_sp - (uintptr_t)old_sp > 100000) return nullptr;
>+  } else {
>+    // In the non-strict mode, allow discontiguous stack frames.
>+    // (alternate-signal-stacks for example).
>+    if (new_sp == old_sp) return nullptr;
>+    // And allow frames upto about 1MB.
>+    if ((new_sp > old_sp)
>+        && ((uintptr_t)new_sp - (uintptr_t)old_sp > 1000000)) return nullptr;
>+  }
>+  if ((uintptr_t)new_sp % kStackAlignment != 0) return nullptr;
>+
>+#if defined(__linux__)
>+  enum StackTraceKernelSymbolStatus {
>+      kNotInitialized = 0, kAddressValid, kAddressInvalid };
>+
>+  if (IS_WITH_CONTEXT && uc != nullptr) {
>+    static StackTraceKernelSymbolStatus kernel_symbol_status =
>+        kNotInitialized;  // Sentinel: not computed yet.
>+    // Initialize with sentinel value: __kernel_rt_sigtramp_rt64 can not
>+    // possibly be there.
>+    static const unsigned char *kernel_sigtramp_rt64_address = nullptr;
>+    if (kernel_symbol_status == kNotInitialized) {
>+      absl::debugging_internal::VDSOSupport vdso;
>+      if (vdso.IsPresent()) {
>+        absl::debugging_internal::VDSOSupport::SymbolInfo
>+            sigtramp_rt64_symbol_info;
>+        if (!vdso.LookupSymbol(
>+                "__kernel_sigtramp_rt64", "LINUX_2.6.15",
>+                absl::debugging_internal::VDSOSupport::kVDSOSymbolType,
>+                &sigtramp_rt64_symbol_info) ||
>+            sigtramp_rt64_symbol_info.address == nullptr) {
>+          // Unexpected: VDSO is present, yet the expected symbol is missing
>+          // or null.
>+          assert(false && "VDSO is present, but doesn't have expected symbol");
>+          kernel_symbol_status = kAddressInvalid;
>+        } else {
>+          kernel_sigtramp_rt64_address =
>+              reinterpret_cast<const unsigned char *>(
>+                  sigtramp_rt64_symbol_info.address);
>+          kernel_symbol_status = kAddressValid;
>+        }
>+      } else {
>+        kernel_symbol_status = kAddressInvalid;
>+      }
>+    }
>+
>+    if (new_sp != nullptr &&
>+        kernel_symbol_status == kAddressValid &&
>+        StacktracePowerPCGetLR(new_sp) == kernel_sigtramp_rt64_address) {
>+      const ucontext_t* signal_context =
>+          reinterpret_cast<const ucontext_t*>(uc);
>+      void **const sp_before_signal =
>+          reinterpret_cast<void**>(signal_context->uc_mcontext.gp_regs[PT_R1]);
>+      // Check that alleged sp before signal is nonnull and is reasonably
>+      // aligned.
>+      if (sp_before_signal != nullptr &&
>+          ((uintptr_t)sp_before_signal % kStackAlignment) == 0) {
>+        // Check that alleged stack pointer is actually readable. This is to
>+        // prevent a "double fault" in case we hit the first fault due to e.g.
>+        // a stack corruption.
>+        if (absl::debugging_internal::AddressIsReadable(sp_before_signal)) {
>+          // Alleged stack pointer is readable, use it for further unwinding.
>+          new_sp = sp_before_signal;
>+        }
>+      }
>+    }
>+  }
>+#endif
>+
>+  return new_sp;
>+}
>+
>+// This ensures that absl::GetStackTrace sets up the Link Register properly.
>+ABSL_ATTRIBUTE_NOINLINE static void AbslStacktracePowerPCDummyFunction() {
>+  ABSL_BLOCK_TAIL_CALL_OPTIMIZATION();
>+}
>+
>+template <bool IS_STACK_FRAMES, bool IS_WITH_CONTEXT>
>+ABSL_ATTRIBUTE_NO_SANITIZE_ADDRESS  // May read random elements from stack.
>+ABSL_ATTRIBUTE_NO_SANITIZE_MEMORY   // May read random elements from stack.
>+static int UnwindImpl(void** result, int* sizes, int max_depth, int skip_count,
>+                      const void *ucp, int *min_dropped_frames) {
>+  void **sp;
>+  // Apple OS X uses an old version of gnu as -- both Darwin 7.9.0 (Panther)
>+  // and Darwin 8.8.1 (Tiger) use as 1.38.  This means we have to use a
>+  // different asm syntax.  I don't know quite the best way to discriminate
>+  // systems using the old as from the new one; I've gone with __APPLE__.
>+#ifdef __APPLE__
>+  __asm__ volatile ("mr %0,r1" : "=r" (sp));
>+#else
>+  __asm__ volatile ("mr %0,1" : "=r" (sp));
>+#endif
>+
>+  // On PowerPC, the "Link Register" or "Link Record" (LR), is a stack
>+  // entry that holds the return address of the subroutine call (what
>+  // instruction we run after our function finishes).  This is the
>+  // same as the stack-pointer of our parent routine, which is what we
>+  // want here.  While the compiler will always(?) set up LR for
>+  // subroutine calls, it may not for leaf functions (such as this one).
>+  // This routine forces the compiler (at least gcc) to push it anyway.
>+  AbslStacktracePowerPCDummyFunction();
>+
>+  // The LR save area is used by the callee, so the top entry is bogus.
>+  skip_count++;
>+
>+  int n = 0;
>+
>+  // Unlike ABIs of X86 and ARM, PowerPC ABIs say that return address (in
>+  // the link register) of a function call is stored in the caller's stack
>+  // frame instead of the callee's.  When we look for the return address
>+  // associated with a stack frame, we need to make sure that there is a
>+  // caller frame before it.  So we call NextStackFrame before entering the
>+  // loop below and check next_sp instead of sp for loop termination.
>+  // The outermost frame is set up by runtimes and it does not have a
>+  // caller frame, so it is skipped.
>+
>+  // The absl::GetStackFrames routine is called when we are in some
>+  // informational context (the failure signal handler for example).
>+  // Use the non-strict unwinding rules to produce a stack trace
>+  // that is as complete as possible (even if it contains a few
>+  // bogus entries in some rare cases).
>+  void **next_sp = NextStackFrame<!IS_STACK_FRAMES, IS_WITH_CONTEXT>(sp, ucp);
>+
>+  while (next_sp && n < max_depth) {
>+    if (skip_count > 0) {
>+      skip_count--;
>+    } else {
>+      result[n] = StacktracePowerPCGetLR(sp);
>+      if (IS_STACK_FRAMES) {
>+        if (next_sp > sp) {
>+          sizes[n] = (uintptr_t)next_sp - (uintptr_t)sp;
>+        } else {
>+          // A frame-size of 0 is used to indicate unknown frame size.
>+          sizes[n] = 0;
>+        }
>+      }
>+      n++;
>+    }
>+
>+    sp = next_sp;
>+    next_sp = NextStackFrame<!IS_STACK_FRAMES, IS_WITH_CONTEXT>(sp, ucp);
>+  }
>+
>+  if (min_dropped_frames != nullptr) {
>+    // Implementation detail: we clamp the max of frames we are willing to
>+    // count, so as not to spend too much time in the loop below.
>+    const int kMaxUnwind = 1000;
>+    int j = 0;
>+    for (; next_sp != nullptr && j < kMaxUnwind; j++) {
>+      next_sp = NextStackFrame<!IS_STACK_FRAMES, IS_WITH_CONTEXT>(next_sp, ucp);
>+    }
>+    *min_dropped_frames = j;
>+  }
>+  return n;
>+}
>+
>+namespace absl {
>+namespace debugging_internal {
>+bool StackTraceWorksForTest() {
>+  return true;
>+}
>+}  // namespace debugging_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_DEBUGGING_INTERNAL_STACKTRACE_POWERPC_INL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/stacktrace_unimplemented-inl.inc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/stacktrace_unimplemented-inl.inc
>new file mode 100644
>index 00000000000..e256fdd4ae7
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/stacktrace_unimplemented-inl.inc
>@@ -0,0 +1,22 @@
>+#ifndef ABSL_DEBUGGING_INTERNAL_STACKTRACE_UNIMPLEMENTED_INL_H_
>+#define ABSL_DEBUGGING_INTERNAL_STACKTRACE_UNIMPLEMENTED_INL_H_
>+
>+template <bool IS_STACK_FRAMES, bool IS_WITH_CONTEXT>
>+static int UnwindImpl(void** /* result */, int* /* sizes */,
>+                      int /* max_depth */, int /* skip_count */,
>+                      const void* /* ucp */, int *min_dropped_frames) {
>+  if (min_dropped_frames != nullptr) {
>+    *min_dropped_frames = 0;
>+  }
>+  return 0;
>+}
>+
>+namespace absl {
>+namespace debugging_internal {
>+bool StackTraceWorksForTest() {
>+  return false;
>+}
>+}  // namespace debugging_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_DEBUGGING_INTERNAL_STACKTRACE_UNIMPLEMENTED_INL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/stacktrace_win32-inl.inc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/stacktrace_win32-inl.inc
>new file mode 100644
>index 00000000000..a8f8a56afb3
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/stacktrace_win32-inl.inc
>@@ -0,0 +1,83 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// Produces a stack trace for Windows.  Normally, one could use
>+// stacktrace_x86-inl.h or stacktrace_x86_64-inl.h -- and indeed, that
>+// should work for binaries compiled using MSVC in "debug" mode.
>+// However, in "release" mode, Windows uses frame-pointer
>+// optimization, which makes getting a stack trace very difficult.
>+//
>+// There are several approaches one can take.  One is to use Windows
>+// intrinsics like StackWalk64.  These can work, but have restrictions
>+// on how successful they can be.  Another attempt is to write a
>+// version of stacktrace_x86-inl.h that has heuristic support for
>+// dealing with FPO, similar to what WinDbg does (see
>+// http://www.nynaeve.net/?p=97).  There are (non-working) examples of
>+// these approaches, complete with TODOs, in stacktrace_win32-inl.h#1
>+//
>+// The solution we've ended up doing is to call the undocumented
>+// windows function RtlCaptureStackBackTrace, which probably doesn't
>+// work with FPO but at least is fast, and doesn't require a symbol
>+// server.
>+//
>+// This code is inspired by a patch from David Vitek:
>+//   http://code.google.com/p/google-perftools/issues/detail?id=83
>+
>+#ifndef ABSL_DEBUGGING_INTERNAL_STACKTRACE_WIN32_INL_H_
>+#define ABSL_DEBUGGING_INTERNAL_STACKTRACE_WIN32_INL_H_
>+
>+#include <windows.h>    // for GetProcAddress and GetModuleHandle
>+#include <cassert>
>+
>+typedef USHORT NTAPI RtlCaptureStackBackTrace_Function(
>+    IN ULONG frames_to_skip,
>+    IN ULONG frames_to_capture,
>+    OUT PVOID *backtrace,
>+    OUT PULONG backtrace_hash);
>+
>+// Load the function we need at static init time, where we don't have
>+// to worry about someone else holding the loader's lock.
>+static RtlCaptureStackBackTrace_Function* const RtlCaptureStackBackTrace_fn =
>+   (RtlCaptureStackBackTrace_Function*)
>+   GetProcAddress(GetModuleHandleA("ntdll.dll"), "RtlCaptureStackBackTrace");
>+
>+template <bool IS_STACK_FRAMES, bool IS_WITH_CONTEXT>
>+static int UnwindImpl(void** result, int* sizes, int max_depth, int skip_count,
>+                      const void *ucp, int *min_dropped_frames) {
>+  int n = 0;
>+  if (!RtlCaptureStackBackTrace_fn) {
>+    // can't find a stacktrace with no function to call
>+  } else {
>+    n = (int)RtlCaptureStackBackTrace_fn(skip_count + 2, max_depth, result, 0);
>+  }
>+  if (IS_STACK_FRAMES) {
>+    // No implementation for finding out the stack frame sizes yet.
>+    memset(sizes, 0, sizeof(*sizes) * n);
>+  }
>+  if (min_dropped_frames != nullptr) {
>+    // Not implemented.
>+    *min_dropped_frames = 0;
>+  }
>+  return n;
>+}
>+
>+namespace absl {
>+namespace debugging_internal {
>+bool StackTraceWorksForTest() {
>+  return false;
>+}
>+}  // namespace debugging_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_DEBUGGING_INTERNAL_STACKTRACE_WIN32_INL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/stacktrace_x86-inl.inc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/stacktrace_x86-inl.inc
>new file mode 100644
>index 00000000000..ac85b920332
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/stacktrace_x86-inl.inc
>@@ -0,0 +1,337 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// Produce stack trace
>+
>+#ifndef ABSL_DEBUGGING_INTERNAL_STACKTRACE_X86_INL_INC_
>+#define ABSL_DEBUGGING_INTERNAL_STACKTRACE_X86_INL_INC_
>+
>+#if defined(__linux__) && (defined(__i386__) || defined(__x86_64__))
>+#include <ucontext.h>  // for ucontext_t
>+#endif
>+
>+#if !defined(_WIN32)
>+#include <unistd.h>
>+#endif
>+
>+#include <cassert>
>+#include <cstdint>
>+
>+#include "absl/base/macros.h"
>+#include "absl/base/port.h"
>+#include "absl/debugging/internal/address_is_readable.h"
>+#include "absl/debugging/internal/vdso_support.h"  // a no-op on non-elf or non-glibc systems
>+#include "absl/debugging/stacktrace.h"
>+#include "absl/base/internal/raw_logging.h"
>+
>+#if defined(__linux__) && defined(__i386__)
>+// Count "push %reg" instructions in VDSO __kernel_vsyscall(),
>+// preceeding "syscall" or "sysenter".
>+// If __kernel_vsyscall uses frame pointer, answer 0.
>+//
>+// kMaxBytes tells how many instruction bytes of __kernel_vsyscall
>+// to analyze before giving up. Up to kMaxBytes+1 bytes of
>+// instructions could be accessed.
>+//
>+// Here are known __kernel_vsyscall instruction sequences:
>+//
>+// SYSENTER (linux-2.6.26/arch/x86/vdso/vdso32/sysenter.S).
>+// Used on Intel.
>+//  0xffffe400 <__kernel_vsyscall+0>:       push   %ecx
>+//  0xffffe401 <__kernel_vsyscall+1>:       push   %edx
>+//  0xffffe402 <__kernel_vsyscall+2>:       push   %ebp
>+//  0xffffe403 <__kernel_vsyscall+3>:       mov    %esp,%ebp
>+//  0xffffe405 <__kernel_vsyscall+5>:       sysenter
>+//
>+// SYSCALL (see linux-2.6.26/arch/x86/vdso/vdso32/syscall.S).
>+// Used on AMD.
>+//  0xffffe400 <__kernel_vsyscall+0>:       push   %ebp
>+//  0xffffe401 <__kernel_vsyscall+1>:       mov    %ecx,%ebp
>+//  0xffffe403 <__kernel_vsyscall+3>:       syscall
>+//
>+
>+// The sequence below isn't actually expected in Google fleet,
>+// here only for completeness. Remove this comment from OSS release.
>+
>+// i386 (see linux-2.6.26/arch/x86/vdso/vdso32/int80.S)
>+//  0xffffe400 <__kernel_vsyscall+0>:       int $0x80
>+//  0xffffe401 <__kernel_vsyscall+1>:       ret
>+//
>+static const int kMaxBytes = 10;
>+
>+// We use assert()s instead of DCHECK()s -- this is too low level
>+// for DCHECK().
>+
>+static int CountPushInstructions(const unsigned char *const addr) {
>+  int result = 0;
>+  for (int i = 0; i < kMaxBytes; ++i) {
>+    if (addr[i] == 0x89) {
>+      // "mov reg,reg"
>+      if (addr[i + 1] == 0xE5) {
>+        // Found "mov %esp,%ebp".
>+        return 0;  
>+      }
>+      ++i;  // Skip register encoding byte.
>+    } else if (addr[i] == 0x0F &&
>+               (addr[i + 1] == 0x34 || addr[i + 1] == 0x05)) {
>+      // Found "sysenter" or "syscall".
>+      return result;
>+    } else if ((addr[i] & 0xF0) == 0x50) {
>+      // Found "push %reg".
>+      ++result;
>+    } else if (addr[i] == 0xCD && addr[i + 1] == 0x80) {
>+      // Found "int $0x80"
>+      assert(result == 0);
>+      return 0;
>+    } else {
>+      // Unexpected instruction.
>+      assert(false && "unexpected instruction in __kernel_vsyscall");
>+      return 0;
>+    }
>+  }
>+  // Unexpected: didn't find SYSENTER or SYSCALL in
>+  // [__kernel_vsyscall, __kernel_vsyscall + kMaxBytes) interval.
>+  assert(false && "did not find SYSENTER or SYSCALL in __kernel_vsyscall");
>+  return 0;
>+}
>+#endif
>+
>+// Assume stack frames larger than 100,000 bytes are bogus.
>+static const int kMaxFrameBytes = 100000;
>+
>+// Returns the stack frame pointer from signal context, 0 if unknown.
>+// vuc is a ucontext_t *.  We use void* to avoid the use
>+// of ucontext_t on non-POSIX systems.
>+static uintptr_t GetFP(const void *vuc) {
>+#if !defined(__linux__)
>+  static_cast<void>(vuc);  // Avoid an unused argument compiler warning.
>+#else
>+  if (vuc != nullptr) {
>+    auto *uc = reinterpret_cast<const ucontext_t *>(vuc);
>+#if defined(__i386__)
>+    const auto bp = uc->uc_mcontext.gregs[REG_EBP];
>+    const auto sp = uc->uc_mcontext.gregs[REG_ESP];
>+#elif defined(__x86_64__)
>+    const auto bp = uc->uc_mcontext.gregs[REG_RBP];
>+    const auto sp = uc->uc_mcontext.gregs[REG_RSP];
>+#else
>+    const uintptr_t bp = 0;
>+    const uintptr_t sp = 0;
>+#endif
>+    // Sanity-check that the base pointer is valid.  It should be as long as
>+    // SHRINK_WRAP_FRAME_POINTER is not set, but it's possible that some code in
>+    // the process is compiled with --copt=-fomit-frame-pointer or
>+    // --copt=-momit-leaf-frame-pointer.
>+    //
>+    // TODO(bcmills): -momit-leaf-frame-pointer is currently the default
>+    // behavior when building with clang.  Talk to the C++ toolchain team about
>+    // fixing that.
>+    if (bp >= sp && bp - sp <= kMaxFrameBytes) return bp;
>+
>+    // If bp isn't a plausible frame pointer, return the stack pointer instead.
>+    // If we're lucky, it points to the start of a stack frame; otherwise, we'll
>+    // get one frame of garbage in the stack trace and fail the sanity check on
>+    // the next iteration.
>+    return sp;
>+  }
>+#endif
>+  return 0;
>+}
>+
>+// Given a pointer to a stack frame, locate and return the calling
>+// stackframe, or return null if no stackframe can be found. Perform sanity
>+// checks (the strictness of which is controlled by the boolean parameter
>+// "STRICT_UNWINDING") to reduce the chance that a bad pointer is returned.
>+template <bool STRICT_UNWINDING, bool WITH_CONTEXT>
>+ABSL_ATTRIBUTE_NO_SANITIZE_ADDRESS  // May read random elements from stack.
>+ABSL_ATTRIBUTE_NO_SANITIZE_MEMORY   // May read random elements from stack.
>+static void **NextStackFrame(void **old_fp, const void *uc) {
>+  void **new_fp = (void **)*old_fp;
>+
>+#if defined(__linux__) && defined(__i386__)
>+  if (WITH_CONTEXT && uc != nullptr) {
>+    // How many "push %reg" instructions are there at __kernel_vsyscall?
>+    // This is constant for a given kernel and processor, so compute
>+    // it only once.
>+    static int num_push_instructions = -1;  // Sentinel: not computed yet.
>+    // Initialize with sentinel value: __kernel_rt_sigreturn can not possibly
>+    // be there.
>+    static const unsigned char *kernel_rt_sigreturn_address = nullptr;
>+    static const unsigned char *kernel_vsyscall_address = nullptr;
>+    if (num_push_instructions == -1) {
>+      absl::debugging_internal::VDSOSupport vdso;
>+      if (vdso.IsPresent()) {
>+        absl::debugging_internal::VDSOSupport::SymbolInfo
>+            rt_sigreturn_symbol_info;
>+        absl::debugging_internal::VDSOSupport::SymbolInfo vsyscall_symbol_info;
>+        if (!vdso.LookupSymbol("__kernel_rt_sigreturn", "LINUX_2.5", STT_FUNC,
>+                               &rt_sigreturn_symbol_info) ||
>+            !vdso.LookupSymbol("__kernel_vsyscall", "LINUX_2.5", STT_FUNC,
>+                               &vsyscall_symbol_info) ||
>+            rt_sigreturn_symbol_info.address == nullptr ||
>+            vsyscall_symbol_info.address == nullptr) {
>+          // Unexpected: 32-bit VDSO is present, yet one of the expected
>+          // symbols is missing or null.
>+          assert(false && "VDSO is present, but doesn't have expected symbols");
>+          num_push_instructions = 0;
>+        } else {
>+          kernel_rt_sigreturn_address =
>+              reinterpret_cast<const unsigned char *>(
>+                  rt_sigreturn_symbol_info.address);
>+          kernel_vsyscall_address =
>+              reinterpret_cast<const unsigned char *>(
>+                  vsyscall_symbol_info.address);
>+          num_push_instructions =
>+              CountPushInstructions(kernel_vsyscall_address);
>+        }
>+      } else {
>+        num_push_instructions = 0;
>+      }
>+    }
>+    if (num_push_instructions != 0 && kernel_rt_sigreturn_address != nullptr &&
>+        old_fp[1] == kernel_rt_sigreturn_address) {
>+      const ucontext_t *ucv = static_cast<const ucontext_t *>(uc);
>+      // This kernel does not use frame pointer in its VDSO code,
>+      // and so %ebp is not suitable for unwinding.
>+      void **const reg_ebp =
>+          reinterpret_cast<void **>(ucv->uc_mcontext.gregs[REG_EBP]);
>+      const unsigned char *const reg_eip =
>+          reinterpret_cast<unsigned char *>(ucv->uc_mcontext.gregs[REG_EIP]);
>+      if (new_fp == reg_ebp && kernel_vsyscall_address <= reg_eip &&
>+          reg_eip - kernel_vsyscall_address < kMaxBytes) {
>+        // We "stepped up" to __kernel_vsyscall, but %ebp is not usable.
>+        // Restore from 'ucv' instead.
>+        void **const reg_esp =
>+            reinterpret_cast<void **>(ucv->uc_mcontext.gregs[REG_ESP]);
>+        // Check that alleged %esp is not null and is reasonably aligned.
>+        if (reg_esp &&
>+            ((uintptr_t)reg_esp & (sizeof(reg_esp) - 1)) == 0) {
>+          // Check that alleged %esp is actually readable. This is to prevent
>+          // "double fault" in case we hit the first fault due to e.g. stack
>+          // corruption.
>+          void *const reg_esp2 = reg_esp[num_push_instructions - 1];
>+          if (absl::debugging_internal::AddressIsReadable(reg_esp2)) {
>+            // Alleged %esp is readable, use it for further unwinding.
>+            new_fp = reinterpret_cast<void **>(reg_esp2);
>+          }
>+        }
>+      }
>+    }
>+  }
>+#endif
>+
>+  const uintptr_t old_fp_u = reinterpret_cast<uintptr_t>(old_fp);
>+  const uintptr_t new_fp_u = reinterpret_cast<uintptr_t>(new_fp);
>+
>+  // Check that the transition from frame pointer old_fp to frame
>+  // pointer new_fp isn't clearly bogus.  Skip the checks if new_fp
>+  // matches the signal context, so that we don't skip out early when
>+  // using an alternate signal stack.
>+  //
>+  // TODO(bcmills): The GetFP call should be completely unnecessary when
>+  // SHRINK_WRAP_FRAME_POINTER is set (because we should be back in the thread's
>+  // stack by this point), but it is empirically still needed (e.g. when the
>+  // stack includes a call to abort).  unw_get_reg returns UNW_EBADREG for some
>+  // frames.  Figure out why GetValidFrameAddr and/or libunwind isn't doing what
>+  // it's supposed to.
>+  if (STRICT_UNWINDING &&
>+      (!WITH_CONTEXT || uc == nullptr || new_fp_u != GetFP(uc))) {
>+    // With the stack growing downwards, older stack frame must be
>+    // at a greater address that the current one.
>+    if (new_fp_u <= old_fp_u) return nullptr;
>+    if (new_fp_u - old_fp_u > kMaxFrameBytes) return nullptr;
>+  } else {
>+    if (new_fp == nullptr) return nullptr;  // skip AddressIsReadable() below
>+    // In the non-strict mode, allow discontiguous stack frames.
>+    // (alternate-signal-stacks for example).
>+    if (new_fp == old_fp) return nullptr;
>+  }
>+
>+  if (new_fp_u & (sizeof(void *) - 1)) return nullptr;
>+#ifdef __i386__
>+  // On 32-bit machines, the stack pointer can be very close to
>+  // 0xffffffff, so we explicitly check for a pointer into the
>+  // last two pages in the address space
>+  if (new_fp_u >= 0xffffe000) return nullptr;
>+#endif
>+#if !defined(_WIN32)
>+  if (!STRICT_UNWINDING) {
>+    // Lax sanity checks cause a crash in 32-bit tcmalloc/crash_reason_test
>+    // on AMD-based machines with VDSO-enabled kernels.
>+    // Make an extra sanity check to insure new_fp is readable.
>+    // Note: NextStackFrame<false>() is only called while the program
>+    //       is already on its last leg, so it's ok to be slow here.
>+
>+    if (!absl::debugging_internal::AddressIsReadable(new_fp)) {
>+      return nullptr;
>+    }
>+  }
>+#endif
>+  return new_fp;
>+}
>+
>+template <bool IS_STACK_FRAMES, bool IS_WITH_CONTEXT>
>+ABSL_ATTRIBUTE_NO_SANITIZE_ADDRESS  // May read random elements from stack.
>+ABSL_ATTRIBUTE_NO_SANITIZE_MEMORY   // May read random elements from stack.
>+ABSL_ATTRIBUTE_NOINLINE
>+static int UnwindImpl(void **result, int *sizes, int max_depth, int skip_count,
>+                      const void *ucp, int *min_dropped_frames) {
>+  int n = 0;
>+  void **fp = reinterpret_cast<void **>(__builtin_frame_address(0));
>+
>+  while (fp && n < max_depth) {
>+    if (*(fp + 1) == reinterpret_cast<void *>(0)) {
>+      // In 64-bit code, we often see a frame that
>+      // points to itself and has a return address of 0.
>+      break;
>+    }
>+    void **next_fp = NextStackFrame<!IS_STACK_FRAMES, IS_WITH_CONTEXT>(fp, ucp);
>+    if (skip_count > 0) {
>+      skip_count--;
>+    } else {
>+      result[n] = *(fp + 1);
>+      if (IS_STACK_FRAMES) {
>+        if (next_fp > fp) {
>+          sizes[n] = (uintptr_t)next_fp - (uintptr_t)fp;
>+        } else {
>+          // A frame-size of 0 is used to indicate unknown frame size.
>+          sizes[n] = 0;
>+        }
>+      }
>+      n++;
>+    }
>+    fp = next_fp;
>+  }
>+  if (min_dropped_frames != nullptr) {
>+    // Implementation detail: we clamp the max of frames we are willing to
>+    // count, so as not to spend too much time in the loop below.
>+    const int kMaxUnwind = 1000;
>+    int j = 0;
>+    for (; fp != nullptr && j < kMaxUnwind; j++) {
>+      fp = NextStackFrame<!IS_STACK_FRAMES, IS_WITH_CONTEXT>(fp, ucp);
>+    }
>+    *min_dropped_frames = j;
>+  }
>+  return n;
>+}
>+
>+namespace absl {
>+namespace debugging_internal {
>+bool StackTraceWorksForTest() {
>+  return true;
>+}
>+}  // namespace debugging_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_DEBUGGING_INTERNAL_STACKTRACE_X86_INL_INC_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/symbolize.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/symbolize.h
>new file mode 100644
>index 00000000000..8d926fec48a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/symbolize.h
>@@ -0,0 +1,123 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+// This file contains internal parts of the Abseil symbolizer.
>+// Do not depend on the anything in this file, it may change at anytime.
>+
>+#ifndef ABSL_DEBUGGING_INTERNAL_SYMBOLIZE_H_
>+#define ABSL_DEBUGGING_INTERNAL_SYMBOLIZE_H_
>+
>+#include <cstddef>
>+#include <cstdint>
>+#include "absl/base/port.h"  // Needed for string vs std::string
>+
>+#ifdef ABSL_INTERNAL_HAVE_ELF_SYMBOLIZE
>+#error ABSL_INTERNAL_HAVE_ELF_SYMBOLIZE cannot be directly set
>+#elif defined(__ELF__) && defined(__GLIBC__) && !defined(__native_client__) && \
>+    !defined(__asmjs__) && !defined(__wasm__)
>+#define ABSL_INTERNAL_HAVE_ELF_SYMBOLIZE 1
>+
>+#include <elf.h>
>+#include <link.h>  // For ElfW() macro.
>+#include <functional>
>+#include <string>
>+
>+namespace absl {
>+namespace debugging_internal {
>+
>+// Iterates over all sections, invoking callback on each with the section name
>+// and the section header.
>+//
>+// Returns true on success; otherwise returns false in case of errors.
>+//
>+// This is not async-signal-safe.
>+bool ForEachSection(
>+    int fd, const std::function<bool(const std::string& name, const ElfW(Shdr) &)>&
>+                callback);
>+
>+// Gets the section header for the given name, if it exists. Returns true on
>+// success. Otherwise, returns false.
>+bool GetSectionHeaderByName(int fd, const char *name, size_t name_len,
>+                            ElfW(Shdr) *out);
>+
>+}  // namespace debugging_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_INTERNAL_HAVE_ELF_SYMBOLIZE
>+
>+namespace absl {
>+namespace debugging_internal {
>+
>+struct SymbolDecoratorArgs {
>+  // The program counter we are getting symbolic name for.
>+  const void *pc;
>+  // 0 for main executable, load address for shared libraries.
>+  ptrdiff_t relocation;
>+  // Read-only file descriptor for ELF image covering "pc",
>+  // or -1 if no such ELF image exists in /proc/self/maps.
>+  int fd;
>+  // Output buffer, size.
>+  // Note: the buffer may not be empty -- default symbolizer may have already
>+  // produced some output, and earlier decorators may have adorned it in
>+  // some way. You are free to replace or augment the contents (within the
>+  // symbol_buf_size limit).
>+  char *const symbol_buf;
>+  size_t symbol_buf_size;
>+  // Temporary scratch space, size.
>+  // Use that space in preference to allocating your own stack buffer to
>+  // conserve stack.
>+  char *const tmp_buf;
>+  size_t tmp_buf_size;
>+  // User-provided argument
>+  void* arg;
>+};
>+using SymbolDecorator = void (*)(const SymbolDecoratorArgs *);
>+
>+// Installs a function-pointer as a decorator. Returns a value less than zero
>+// if the system cannot install the decorator. Otherwise, returns a unique
>+// identifier corresponding to the decorator. This identifier can be used to
>+// uninstall the decorator - See RemoveSymbolDecorator() below.
>+int InstallSymbolDecorator(SymbolDecorator decorator, void* arg);
>+
>+// Removes a previously installed function-pointer decorator. Parameter "ticket"
>+// is the return-value from calling InstallSymbolDecorator().
>+bool RemoveSymbolDecorator(int ticket);
>+
>+// Remove all installed decorators.  Returns true if successful, false if
>+// symbolization is currently in progress.
>+bool RemoveAllSymbolDecorators(void);
>+
>+// Registers an address range to a file mapping.
>+//
>+// Preconditions:
>+//   start <= end
>+//   filename != nullptr
>+//
>+// Returns true if the file was successfully registered.
>+bool RegisterFileMappingHint(
>+    const void* start, const void* end, uint64_t offset, const char* filename);
>+
>+// Looks up the file mapping registered by RegisterFileMappingHint for an
>+// address range. If there is one, the file name is stored in *filename and
>+// *start and *end are modified to reflect the registered mapping. Returns
>+// whether any hint was found.
>+bool GetFileMappingHint(const void** start,
>+                        const void** end,
>+                        uint64_t    *  offset,
>+                        const char** filename);
>+
>+}  // namespace debugging_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_DEBUGGING_INTERNAL_SYMBOLIZE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/vdso_support.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/vdso_support.cc
>new file mode 100644
>index 00000000000..e8129e87dca
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/vdso_support.cc
>@@ -0,0 +1,192 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+// Allow dynamic symbol lookup in the kernel VDSO page.
>+//
>+// VDSOSupport -- a class representing kernel VDSO (if present).
>+
>+#include "absl/debugging/internal/vdso_support.h"
>+
>+#ifdef ABSL_HAVE_VDSO_SUPPORT     // defined in vdso_support.h
>+
>+#include <errno.h>
>+#include <fcntl.h>
>+#include <sys/syscall.h>
>+#include <unistd.h>
>+
>+#if __GLIBC_PREREQ(2, 16)  // GLIBC-2.16 implements getauxval.
>+#include <sys/auxv.h>
>+#endif
>+
>+#include "absl/base/dynamic_annotations.h"
>+#include "absl/base/internal/raw_logging.h"
>+#include "absl/base/port.h"
>+
>+#ifndef AT_SYSINFO_EHDR
>+#define AT_SYSINFO_EHDR 33  // for crosstoolv10
>+#endif
>+
>+namespace absl {
>+namespace debugging_internal {
>+
>+ABSL_CONST_INIT
>+std::atomic<const void *> VDSOSupport::vdso_base_(
>+    debugging_internal::ElfMemImage::kInvalidBase);
>+
>+std::atomic<VDSOSupport::GetCpuFn> VDSOSupport::getcpu_fn_(&InitAndGetCPU);
>+VDSOSupport::VDSOSupport()
>+    // If vdso_base_ is still set to kInvalidBase, we got here
>+    // before VDSOSupport::Init has been called. Call it now.
>+    : image_(vdso_base_.load(std::memory_order_relaxed) ==
>+                     debugging_internal::ElfMemImage::kInvalidBase
>+                 ? Init()
>+                 : vdso_base_.load(std::memory_order_relaxed)) {}
>+
>+// NOTE: we can't use GoogleOnceInit() below, because we can be
>+// called by tcmalloc, and none of the *once* stuff may be functional yet.
>+//
>+// In addition, we hope that the VDSOSupportHelper constructor
>+// causes this code to run before there are any threads, and before
>+// InitGoogle() has executed any chroot or setuid calls.
>+//
>+// Finally, even if there is a race here, it is harmless, because
>+// the operation should be idempotent.
>+const void *VDSOSupport::Init() {
>+  const auto kInvalidBase = debugging_internal::ElfMemImage::kInvalidBase;
>+#if __GLIBC_PREREQ(2, 16)
>+  if (vdso_base_.load(std::memory_order_relaxed) == kInvalidBase) {
>+    errno = 0;
>+    const void *const sysinfo_ehdr =
>+        reinterpret_cast<const void *>(getauxval(AT_SYSINFO_EHDR));
>+    if (errno == 0) {
>+      vdso_base_.store(sysinfo_ehdr, std::memory_order_relaxed);
>+    }
>+  }
>+#endif  // __GLIBC_PREREQ(2, 16)
>+  if (vdso_base_.load(std::memory_order_relaxed) == kInvalidBase) {
>+    // Valgrind zaps AT_SYSINFO_EHDR and friends from the auxv[]
>+    // on stack, and so glibc works as if VDSO was not present.
>+    // But going directly to kernel via /proc/self/auxv below bypasses
>+    // Valgrind zapping. So we check for Valgrind separately.
>+    if (AbslRunningOnValgrind()) {
>+      vdso_base_.store(nullptr, std::memory_order_relaxed);
>+      getcpu_fn_.store(&GetCPUViaSyscall, std::memory_order_relaxed);
>+      return nullptr;
>+    }
>+    int fd = open("/proc/self/auxv", O_RDONLY);
>+    if (fd == -1) {
>+      // Kernel too old to have a VDSO.
>+      vdso_base_.store(nullptr, std::memory_order_relaxed);
>+      getcpu_fn_.store(&GetCPUViaSyscall, std::memory_order_relaxed);
>+      return nullptr;
>+    }
>+    ElfW(auxv_t) aux;
>+    while (read(fd, &aux, sizeof(aux)) == sizeof(aux)) {
>+      if (aux.a_type == AT_SYSINFO_EHDR) {
>+        vdso_base_.store(reinterpret_cast<void *>(aux.a_un.a_val),
>+                         std::memory_order_relaxed);
>+        break;
>+      }
>+    }
>+    close(fd);
>+    if (vdso_base_.load(std::memory_order_relaxed) == kInvalidBase) {
>+      // Didn't find AT_SYSINFO_EHDR in auxv[].
>+      vdso_base_.store(nullptr, std::memory_order_relaxed);
>+    }
>+  }
>+  GetCpuFn fn = &GetCPUViaSyscall;  // default if VDSO not present.
>+  if (vdso_base_.load(std::memory_order_relaxed)) {
>+    VDSOSupport vdso;
>+    SymbolInfo info;
>+    if (vdso.LookupSymbol("__vdso_getcpu", "LINUX_2.6", STT_FUNC, &info)) {
>+      fn = reinterpret_cast<GetCpuFn>(const_cast<void *>(info.address));
>+    }
>+  }
>+  // Subtle: this code runs outside of any locks; prevent compiler
>+  // from assigning to getcpu_fn_ more than once.
>+  getcpu_fn_.store(fn, std::memory_order_relaxed);
>+  return vdso_base_.load(std::memory_order_relaxed);
>+}
>+
>+const void *VDSOSupport::SetBase(const void *base) {
>+  ABSL_RAW_CHECK(base != debugging_internal::ElfMemImage::kInvalidBase,
>+                 "internal error");
>+  const void *old_base = vdso_base_.load(std::memory_order_relaxed);
>+  vdso_base_.store(base, std::memory_order_relaxed);
>+  image_.Init(base);
>+  // Also reset getcpu_fn_, so GetCPU could be tested with simulated VDSO.
>+  getcpu_fn_.store(&InitAndGetCPU, std::memory_order_relaxed);
>+  return old_base;
>+}
>+
>+bool VDSOSupport::LookupSymbol(const char *name,
>+                               const char *version,
>+                               int type,
>+                               SymbolInfo *info) const {
>+  return image_.LookupSymbol(name, version, type, info);
>+}
>+
>+bool VDSOSupport::LookupSymbolByAddress(const void *address,
>+                                        SymbolInfo *info_out) const {
>+  return image_.LookupSymbolByAddress(address, info_out);
>+}
>+
>+// NOLINT on 'long' because this routine mimics kernel api.
>+long VDSOSupport::GetCPUViaSyscall(unsigned *cpu,  // NOLINT(runtime/int)
>+                                   void *, void *) {
>+#ifdef SYS_getcpu
>+  return syscall(SYS_getcpu, cpu, nullptr, nullptr);
>+#else
>+  // x86_64 never implemented sys_getcpu(), except as a VDSO call.
>+  static_cast<void>(cpu);  // Avoid an unused argument compiler warning.
>+  errno = ENOSYS;
>+  return -1;
>+#endif
>+}
>+
>+// Use fast __vdso_getcpu if available.
>+long VDSOSupport::InitAndGetCPU(unsigned *cpu,  // NOLINT(runtime/int)
>+                                void *x, void *y) {
>+  Init();
>+  GetCpuFn fn = getcpu_fn_.load(std::memory_order_relaxed);
>+  ABSL_RAW_CHECK(fn != &InitAndGetCPU, "Init() did not set getcpu_fn_");
>+  return (*fn)(cpu, x, y);
>+}
>+
>+// This function must be very fast, and may be called from very
>+// low level (e.g. tcmalloc). Hence I avoid things like
>+// GoogleOnceInit() and ::operator new.
>+ABSL_ATTRIBUTE_NO_SANITIZE_MEMORY
>+int GetCPU() {
>+  unsigned cpu;
>+  int ret_code = (*VDSOSupport::getcpu_fn_)(&cpu, nullptr, nullptr);
>+  return ret_code == 0 ? cpu : ret_code;
>+}
>+
>+// We need to make sure VDSOSupport::Init() is called before
>+// InitGoogle() does any setuid or chroot calls.  If VDSOSupport
>+// is used in any global constructor, this will happen, since
>+// VDSOSupport's constructor calls Init.  But if not, we need to
>+// ensure it here, with a global constructor of our own.  This
>+// is an allowed exception to the normal rule against non-trivial
>+// global constructors.
>+static class VDSOInitHelper {
>+ public:
>+  VDSOInitHelper() { VDSOSupport::Init(); }
>+} vdso_init_helper;
>+
>+}  // namespace debugging_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_HAVE_VDSO_SUPPORT
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/vdso_support.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/vdso_support.h
>new file mode 100644
>index 00000000000..8002c7405fa
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/internal/vdso_support.h
>@@ -0,0 +1,156 @@
>+//
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+
>+// Allow dynamic symbol lookup in the kernel VDSO page.
>+//
>+// VDSO stands for "Virtual Dynamic Shared Object" -- a page of
>+// executable code, which looks like a shared library, but doesn't
>+// necessarily exist anywhere on disk, and which gets mmap()ed into
>+// every process by kernels which support VDSO, such as 2.6.x for 32-bit
>+// executables, and 2.6.24 and above for 64-bit executables.
>+//
>+// More details could be found here:
>+// http://www.trilithium.com/johan/2005/08/linux-gate/
>+//
>+// VDSOSupport -- a class representing kernel VDSO (if present).
>+//
>+// Example usage:
>+//  VDSOSupport vdso;
>+//  VDSOSupport::SymbolInfo info;
>+//  typedef (*FN)(unsigned *, void *, void *);
>+//  FN fn = nullptr;
>+//  if (vdso.LookupSymbol("__vdso_getcpu", "LINUX_2.6", STT_FUNC, &info)) {
>+//     fn = reinterpret_cast<FN>(info.address);
>+//  }
>+
>+#ifndef ABSL_DEBUGGING_INTERNAL_VDSO_SUPPORT_H_
>+#define ABSL_DEBUGGING_INTERNAL_VDSO_SUPPORT_H_
>+
>+#include <atomic>
>+
>+#include "absl/base/attributes.h"
>+#include "absl/debugging/internal/elf_mem_image.h"
>+
>+#ifdef ABSL_HAVE_ELF_MEM_IMAGE
>+
>+#ifdef ABSL_HAVE_VDSO_SUPPORT
>+#error ABSL_HAVE_VDSO_SUPPORT cannot be directly set
>+#else
>+#define ABSL_HAVE_VDSO_SUPPORT 1
>+#endif
>+
>+namespace absl {
>+namespace debugging_internal {
>+
>+// NOTE: this class may be used from within tcmalloc, and can not
>+// use any memory allocation routines.
>+class VDSOSupport {
>+ public:
>+  VDSOSupport();
>+
>+  typedef ElfMemImage::SymbolInfo SymbolInfo;
>+  typedef ElfMemImage::SymbolIterator SymbolIterator;
>+
>+  // On PowerPC64 VDSO symbols can either be of type STT_FUNC or STT_NOTYPE
>+  // depending on how the kernel is built.  The kernel is normally built with
>+  // STT_NOTYPE type VDSO symbols.  Let's make things simpler first by using a
>+  // compile-time constant.
>+#ifdef __powerpc64__
>+  enum { kVDSOSymbolType = STT_NOTYPE };
>+#else
>+  enum { kVDSOSymbolType = STT_FUNC };
>+#endif
>+
>+  // Answers whether we have a vdso at all.
>+  bool IsPresent() const { return image_.IsPresent(); }
>+
>+  // Allow to iterate over all VDSO symbols.
>+  SymbolIterator begin() const { return image_.begin(); }
>+  SymbolIterator end() const { return image_.end(); }
>+
>+  // Look up versioned dynamic symbol in the kernel VDSO.
>+  // Returns false if VDSO is not present, or doesn't contain given
>+  // symbol/version/type combination.
>+  // If info_out != nullptr, additional details are filled in.
>+  bool LookupSymbol(const char *name, const char *version,
>+                    int symbol_type, SymbolInfo *info_out) const;
>+
>+  // Find info about symbol (if any) which overlaps given address.
>+  // Returns true if symbol was found; false if VDSO isn't present
>+  // or doesn't have a symbol overlapping given address.
>+  // If info_out != nullptr, additional details are filled in.
>+  bool LookupSymbolByAddress(const void *address, SymbolInfo *info_out) const;
>+
>+  // Used only for testing. Replace real VDSO base with a mock.
>+  // Returns previous value of vdso_base_. After you are done testing,
>+  // you are expected to call SetBase() with previous value, in order to
>+  // reset state to the way it was.
>+  const void *SetBase(const void *s);
>+
>+  // Computes vdso_base_ and returns it. Should be called as early as
>+  // possible; before any thread creation, chroot or setuid.
>+  static const void *Init();
>+
>+ private:
>+  // image_ represents VDSO ELF image in memory.
>+  // image_.ehdr_ == nullptr implies there is no VDSO.
>+  ElfMemImage image_;
>+
>+  // Cached value of auxv AT_SYSINFO_EHDR, computed once.
>+  // This is a tri-state:
>+  //   kInvalidBase   => value hasn't been determined yet.
>+  //              0   => there is no VDSO.
>+  //           else   => vma of VDSO Elf{32,64}_Ehdr.
>+  //
>+  // When testing with mock VDSO, low bit is set.
>+  // The low bit is always available because vdso_base_ is
>+  // page-aligned.
>+  static std::atomic<const void *> vdso_base_;
>+
>+  // NOLINT on 'long' because these routines mimic kernel api.
>+  // The 'cache' parameter may be used by some versions of the kernel,
>+  // and should be nullptr or point to a static buffer containing at
>+  // least two 'long's.
>+  static long InitAndGetCPU(unsigned *cpu, void *cache,     // NOLINT 'long'.
>+                            void *unused);
>+  static long GetCPUViaSyscall(unsigned *cpu, void *cache,  // NOLINT 'long'.
>+                               void *unused);
>+  typedef long (*GetCpuFn)(unsigned *cpu, void *cache,      // NOLINT 'long'.
>+                           void *unused);
>+
>+  // This function pointer may point to InitAndGetCPU,
>+  // GetCPUViaSyscall, or __vdso_getcpu at different stages of initialization.
>+  ABSL_CONST_INIT static std::atomic<GetCpuFn> getcpu_fn_;
>+
>+  friend int GetCPU(void);  // Needs access to getcpu_fn_.
>+
>+  VDSOSupport(const VDSOSupport&) = delete;
>+  VDSOSupport& operator=(const VDSOSupport&) = delete;
>+};
>+
>+// Same as sched_getcpu() on later glibc versions.
>+// Return current CPU, using (fast) __vdso_getcpu@LINUX_2.6 if present,
>+// otherwise use syscall(SYS_getcpu,...).
>+// May return -1 with errno == ENOSYS if the kernel doesn't
>+// support SYS_getcpu.
>+int GetCPU();
>+
>+}  // namespace debugging_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_HAVE_ELF_MEM_IMAGE
>+
>+#endif  // ABSL_DEBUGGING_INTERNAL_VDSO_SUPPORT_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/leak_check.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/leak_check.cc
>new file mode 100644
>index 00000000000..e01e5f8c937
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/leak_check.cc
>@@ -0,0 +1,48 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+// Wrappers around lsan_interface functions.
>+// When lsan is not linked in, these functions are not available,
>+// therefore Abseil code which depends on these functions is conditioned on the
>+// definition of LEAK_SANITIZER.
>+#include "absl/debugging/leak_check.h"
>+
>+#ifndef LEAK_SANITIZER
>+
>+namespace absl {
>+bool HaveLeakSanitizer() { return false; }
>+void DoIgnoreLeak(const void*) { }
>+void RegisterLivePointers(const void*, size_t) { }
>+void UnRegisterLivePointers(const void*, size_t) { }
>+LeakCheckDisabler::LeakCheckDisabler() { }
>+LeakCheckDisabler::~LeakCheckDisabler() { }
>+}  // namespace absl
>+
>+#else
>+
>+#include <sanitizer/lsan_interface.h>
>+
>+namespace absl {
>+bool HaveLeakSanitizer() { return true; }
>+void DoIgnoreLeak(const void* ptr) { __lsan_ignore_object(ptr); }
>+void RegisterLivePointers(const void* ptr, size_t size) {
>+  __lsan_register_root_region(ptr, size);
>+}
>+void UnRegisterLivePointers(const void* ptr, size_t size) {
>+  __lsan_unregister_root_region(ptr, size);
>+}
>+LeakCheckDisabler::LeakCheckDisabler() { __lsan_disable(); }
>+LeakCheckDisabler::~LeakCheckDisabler() { __lsan_enable(); }
>+}  // namespace absl
>+
>+#endif  // LEAK_SANITIZER
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/leak_check.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/leak_check.h
>new file mode 100644
>index 00000000000..c930684e959
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/leak_check.h
>@@ -0,0 +1,109 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// -----------------------------------------------------------------------------
>+// File: leak_check.h
>+// -----------------------------------------------------------------------------
>+//
>+// This file contains functions that affect leak checking behavior within
>+// targets built with the LeakSanitizer (LSan), a memory leak detector that is
>+// integrated within the AddressSanitizer (ASan) as an additional component, or
>+// which can be used standalone. LSan and ASan are included (or can be provided)
>+// as additional components for most compilers such as Clang, gcc and MSVC.
>+// Note: this leak checking API is not yet supported in MSVC.
>+// Leak checking is enabled by default in all ASan builds.
>+//
>+// See https://github.com/google/sanitizers/wiki/AddressSanitizerLeakSanitizer
>+//
>+// -----------------------------------------------------------------------------
>+#ifndef ABSL_DEBUGGING_LEAK_CHECK_H_
>+#define ABSL_DEBUGGING_LEAK_CHECK_H_
>+
>+#include <cstddef>
>+
>+namespace absl {
>+
>+// HaveLeakSanitizer()
>+//
>+// Returns true if a leak-checking sanitizer (either ASan or standalone LSan) is
>+// currently built into this target.
>+bool HaveLeakSanitizer();
>+
>+// DoIgnoreLeak()
>+//
>+// Implements `IgnoreLeak()` below. This function should usually
>+// not be called directly; calling `IgnoreLeak()` is preferred.
>+void DoIgnoreLeak(const void* ptr);
>+
>+// IgnoreLeak()
>+//
>+// Instruct the leak sanitizer to ignore leak warnings on the object referenced
>+// by the passed pointer, as well as all heap objects transitively referenced
>+// by it. The passed object pointer can point to either the beginning of the
>+// object or anywhere within it.
>+//
>+// Example:
>+//
>+//   static T* obj = IgnoreLeak(new T(...));
>+//
>+// If the passed `ptr` does not point to an actively allocated object at the
>+// time `IgnoreLeak()` is called, the call is a no-op; if it is actively
>+// allocated, the object must not get deallocated later.
>+//
>+template <typename T>
>+T* IgnoreLeak(T* ptr) {
>+  DoIgnoreLeak(ptr);
>+  return ptr;
>+}
>+
>+// LeakCheckDisabler
>+//
>+// This helper class indicates that any heap allocations done in the code block
>+// covered by the scoped object, which should be allocated on the stack, will
>+// not be reported as leaks. Leak check disabling will occur within the code
>+// block and any nested function calls within the code block.
>+//
>+// Example:
>+//
>+//   void Foo() {
>+//     LeakCheckDisabler disabler;
>+//     ... code that allocates objects whose leaks should be ignored ...
>+//   }
>+//
>+// REQUIRES: Destructor runs in same thread as constructor
>+class LeakCheckDisabler {
>+ public:
>+  LeakCheckDisabler();
>+  LeakCheckDisabler(const LeakCheckDisabler&) = delete;
>+  LeakCheckDisabler& operator=(const LeakCheckDisabler&) = delete;
>+  ~LeakCheckDisabler();
>+};
>+
>+// RegisterLivePointers()
>+//
>+// Registers `ptr[0,size-1]` as pointers to memory that is still actively being
>+// referenced and for which leak checking should be ignored. This function is
>+// useful if you store pointers in mapped memory, for memory ranges that we know
>+// are correct but for which normal analysis would flag as leaked code.
>+void RegisterLivePointers(const void* ptr, size_t size);
>+
>+// UnRegisterLivePointers()
>+//
>+// Deregisters the pointers previously marked as active in
>+// `RegisterLivePointers()`, enabling leak checking of those pointers.
>+void UnRegisterLivePointers(const void* ptr, size_t size);
>+
>+}  // namespace absl
>+
>+#endif  // ABSL_DEBUGGING_LEAK_CHECK_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/leak_check_disable.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/leak_check_disable.cc
>new file mode 100644
>index 00000000000..df22c1cae91
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/leak_check_disable.cc
>@@ -0,0 +1,20 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+// Disable LeakSanitizer when this file is linked in.
>+// This function overrides __lsan_is_turned_off from sanitizer/lsan_interface.h
>+extern "C" int __lsan_is_turned_off();
>+extern "C" int __lsan_is_turned_off() {
>+  return 1;
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/leak_check_fail_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/leak_check_fail_test.cc
>new file mode 100644
>index 00000000000..bf541fe8415
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/leak_check_fail_test.cc
>@@ -0,0 +1,41 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include <memory>
>+#include "gtest/gtest.h"
>+#include "absl/base/internal/raw_logging.h"
>+#include "absl/debugging/leak_check.h"
>+
>+namespace {
>+
>+TEST(LeakCheckTest, LeakMemory) {
>+  // This test is expected to cause lsan failures on program exit. Therefore the
>+  // test will be run only by leak_check_test.sh, which will verify a
>+  // failed exit code.
>+
>+  char* foo = strdup("lsan should complain about this leaked string");
>+  ABSL_RAW_LOG(INFO, "Should detect leaked std::string %s", foo);
>+}
>+
>+TEST(LeakCheckTest, LeakMemoryAfterDisablerScope) {
>+  // This test is expected to cause lsan failures on program exit. Therefore the
>+  // test will be run only by external_leak_check_test.sh, which will verify a
>+  // failed exit code.
>+  { absl::LeakCheckDisabler disabler; }
>+  char* foo = strdup("lsan should also complain about this leaked string");
>+  ABSL_RAW_LOG(INFO, "Re-enabled leak detection.Should detect leaked std::string %s",
>+               foo);
>+}
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/leak_check_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/leak_check_test.cc
>new file mode 100644
>index 00000000000..febd1ee4f92
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/leak_check_test.cc
>@@ -0,0 +1,42 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include <string>
>+
>+#include "gtest/gtest.h"
>+#include "absl/base/internal/raw_logging.h"
>+#include "absl/debugging/leak_check.h"
>+
>+namespace {
>+
>+TEST(LeakCheckTest, DetectLeakSanitizer) {
>+#ifdef ABSL_EXPECT_LEAK_SANITIZER
>+  EXPECT_TRUE(absl::HaveLeakSanitizer());
>+#else
>+  EXPECT_FALSE(absl::HaveLeakSanitizer());
>+#endif
>+}
>+
>+TEST(LeakCheckTest, IgnoreLeakSuppressesLeakedMemoryErrors) {
>+  auto foo = absl::IgnoreLeak(new std::string("some ignored leaked string"));
>+  ABSL_RAW_LOG(INFO, "Ignoring leaked std::string %s", foo->c_str());
>+}
>+
>+TEST(LeakCheckTest, LeakCheckDisablerIgnoresLeak) {
>+  absl::LeakCheckDisabler disabler;
>+  auto foo = new std::string("some std::string leaked while checks are disabled");
>+  ABSL_RAW_LOG(INFO, "Ignoring leaked std::string %s", foo->c_str());
>+}
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/stacktrace.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/stacktrace.cc
>new file mode 100644
>index 00000000000..61fee6190f5
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/stacktrace.cc
>@@ -0,0 +1,133 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+// Produce stack trace.
>+//
>+// There are three different ways we can try to get the stack trace:
>+//
>+// 1) Our hand-coded stack-unwinder.  This depends on a certain stack
>+//    layout, which is used by gcc (and those systems using a
>+//    gcc-compatible ABI) on x86 systems, at least since gcc 2.95.
>+//    It uses the frame pointer to do its work.
>+//
>+// 2) The libunwind library.  This is still in development, and as a
>+//    separate library adds a new dependency, but doesn't need a frame
>+//    pointer.  It also doesn't call malloc.
>+//
>+// 3) The gdb unwinder -- also the one used by the c++ exception code.
>+//    It's obviously well-tested, but has a fatal flaw: it can call
>+//    malloc() from the unwinder.  This is a problem because we're
>+//    trying to use the unwinder to instrument malloc().
>+//
>+// Note: if you add a new implementation here, make sure it works
>+// correctly when absl::GetStackTrace() is called with max_depth == 0.
>+// Some code may do that.
>+
>+#include "absl/debugging/stacktrace.h"
>+
>+#include <atomic>
>+
>+#include "absl/base/attributes.h"
>+#include "absl/base/port.h"
>+#include "absl/debugging/internal/stacktrace_config.h"
>+
>+#if defined(ABSL_STACKTRACE_INL_HEADER)
>+#include ABSL_STACKTRACE_INL_HEADER
>+#else
>+# error Cannot calculate stack trace: will need to write for your environment
>+# include "absl/debugging/internal/stacktrace_aarch64-inl.inc"
>+# include "absl/debugging/internal/stacktrace_arm-inl.inc"
>+# include "absl/debugging/internal/stacktrace_generic-inl.inc"
>+# include "absl/debugging/internal/stacktrace_powerpc-inl.inc"
>+# include "absl/debugging/internal/stacktrace_unimplemented-inl.inc"
>+# include "absl/debugging/internal/stacktrace_win32-inl.inc"
>+# include "absl/debugging/internal/stacktrace_x86-inl.inc"
>+#endif
>+
>+namespace absl {
>+namespace {
>+
>+typedef int (*Unwinder)(void**, int*, int, int, const void*, int*);
>+std::atomic<Unwinder> custom;
>+
>+template <bool IS_STACK_FRAMES, bool IS_WITH_CONTEXT>
>+ABSL_ATTRIBUTE_ALWAYS_INLINE inline int Unwind(void** result, int* sizes,
>+                                               int max_depth, int skip_count,
>+                                               const void* uc,
>+                                               int* min_dropped_frames) {
>+  Unwinder f = &UnwindImpl<IS_STACK_FRAMES, IS_WITH_CONTEXT>;
>+  Unwinder g = custom.load(std::memory_order_acquire);
>+  if (g != nullptr) f = g;
>+
>+  // Add 1 to skip count for the unwinder function itself
>+  int size = (*f)(result, sizes, max_depth, skip_count + 1, uc,
>+                  min_dropped_frames);
>+  // To disable tail call to (*f)(...)
>+  ABSL_BLOCK_TAIL_CALL_OPTIMIZATION();
>+  return size;
>+}
>+
>+}  // anonymous namespace
>+
>+int GetStackFrames(void** result, int* sizes, int max_depth, int skip_count) {
>+  return Unwind<true, false>(result, sizes, max_depth, skip_count, nullptr,
>+                             nullptr);
>+}
>+
>+int GetStackFramesWithContext(void** result, int* sizes, int max_depth,
>+                              int skip_count, const void* uc,
>+                              int* min_dropped_frames) {
>+  return Unwind<true, true>(result, sizes, max_depth, skip_count, uc,
>+                            min_dropped_frames);
>+}
>+
>+int GetStackTrace(void** result, int max_depth, int skip_count) {
>+  return Unwind<false, false>(result, nullptr, max_depth, skip_count, nullptr,
>+                              nullptr);
>+}
>+
>+int GetStackTraceWithContext(void** result, int max_depth, int skip_count,
>+                             const void* uc, int* min_dropped_frames) {
>+  return Unwind<false, true>(result, nullptr, max_depth, skip_count, uc,
>+                             min_dropped_frames);
>+}
>+
>+void SetStackUnwinder(Unwinder w) {
>+  custom.store(w, std::memory_order_release);
>+}
>+
>+int DefaultStackUnwinder(void** pcs, int* sizes, int depth, int skip,
>+                         const void* uc, int* min_dropped_frames) {
>+  skip++;  // For this function
>+  Unwinder f = nullptr;
>+  if (sizes == nullptr) {
>+    if (uc == nullptr) {
>+      f = &UnwindImpl<false, false>;
>+    } else {
>+      f = &UnwindImpl<false, true>;
>+    }
>+  } else {
>+    if (uc == nullptr) {
>+      f = &UnwindImpl<true, false>;
>+    } else {
>+      f = &UnwindImpl<true, true>;
>+    }
>+  }
>+  volatile int x = 0;
>+  int n = (*f)(pcs, sizes, depth, skip, uc, min_dropped_frames);
>+  x = 1; (void) x;  // To disable tail call to (*f)(...)
>+  return n;
>+}
>+
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/stacktrace.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/stacktrace.h
>new file mode 100644
>index 00000000000..8b831e26815
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/stacktrace.h
>@@ -0,0 +1,225 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// -----------------------------------------------------------------------------
>+// File: stacktrace.h
>+// -----------------------------------------------------------------------------
>+//
>+// This file contains routines to extract the current stack trace and associated
>+// stack frames. These functions are thread-safe and async-signal-safe.
>+//
>+// Note that stack trace functionality is platform dependent and requires
>+// additional support from the compiler/build system in most cases. (That is,
>+// this functionality generally only works on platforms/builds that have been
>+// specifically configured to support it.)
>+//
>+// Note: stack traces in Abseil that do not utilize a symbolizer will result in
>+// frames consisting of function addresses rather than human-readable function
>+// names. (See symbolize.h for information on symbolizing these values.)
>+
>+#ifndef ABSL_DEBUGGING_STACKTRACE_H_
>+#define ABSL_DEBUGGING_STACKTRACE_H_
>+
>+namespace absl {
>+
>+// GetStackFrames()
>+//
>+// Records program counter values for up to `max_depth` frames, skipping the
>+// most recent `skip_count` stack frames, and stores their corresponding values
>+// and sizes in `results` and `sizes` buffers. (Note that the frame generated
>+// for the `absl::GetStackFrames()` routine itself is also skipped.)
>+// routine itself.
>+//
>+// Example:
>+//
>+//      main() { foo(); }
>+//      foo() { bar(); }
>+//      bar() {
>+//        void* result[10];
>+//        int sizes[10];
>+//        int depth = absl::GetStackFrames(result, sizes, 10, 1);
>+//      }
>+//
>+// The current stack frame would consist of three function calls: `bar()`,
>+// `foo()`, and then `main()`; however, since the `GetStackFrames()` call sets
>+// `skip_count` to `1`, it will skip the frame for `bar()`, the most recently
>+// invoked function call. It will therefore return two program counters and will
>+// produce values that map to the following function calls:
>+//
>+//      result[0]       foo()
>+//      result[1]       main()
>+//
>+// (Note: in practice, a few more entries after `main()` may be added to account
>+// for startup processes.)
>+//
>+// Corresponding stack frame sizes will also be recorded:
>+//
>+//    sizes[0]       16
>+//    sizes[1]       16
>+//
>+// (Stack frame sizes of `16` above are just for illustration purposes.)
>+//
>+// Stack frame sizes of 0 or less indicate that those frame sizes couldn't
>+// be identified.
>+//
>+// This routine may return fewer stack frame entries than are
>+// available. Also note that `result` and `sizes` must both be non-null.
>+extern int GetStackFrames(void** result, int* sizes, int max_depth,
>+                          int skip_count);
>+
>+// GetStackFramesWithContext()
>+//
>+// Records program counter values obtained from a signal handler. Records
>+// program counter values for up to `max_depth` frames, skipping the most recent
>+// `skip_count` stack frames, and stores their corresponding values and sizes in
>+// `results` and `sizes` buffers. (Note that the frame generated for the
>+// `absl::GetStackFramesWithContext()` routine itself is also skipped.)
>+//
>+// The `uc` parameter, if non-null, should be a pointer to a `ucontext_t` value
>+// passed to a signal handler registered via the `sa_sigaction` field of a
>+// `sigaction` struct. (See
>+// http://man7.org/linux/man-pages/man2/sigaction.2.html.) The `uc` value may
>+// help a stack unwinder to provide a better stack trace under certain
>+// conditions. `uc` may safely be null.
>+//
>+// The `min_dropped_frames` output parameter, if non-null, points to the
>+// location to note any dropped stack frames, if any, due to buffer limitations
>+// or other reasons. (This value will be set to `0` if no frames were dropped.)
>+// The number of total stack frames is guaranteed to be >= skip_count +
>+// max_depth + *min_dropped_frames.
>+extern int GetStackFramesWithContext(void** result, int* sizes, int max_depth,
>+                                     int skip_count, const void* uc,
>+                                     int* min_dropped_frames);
>+
>+// GetStackTrace()
>+//
>+// Records program counter values for up to `max_depth` frames, skipping the
>+// most recent `skip_count` stack frames, and stores their corresponding values
>+// in `results`. Note that this function is similar to `absl::GetStackFrames()`
>+// except that it returns the stack trace only, and not stack frame sizes.
>+//
>+// Example:
>+//
>+//      main() { foo(); }
>+//      foo() { bar(); }
>+//      bar() {
>+//        void* result[10];
>+//        int depth = absl::GetStackTrace(result, 10, 1);
>+//      }
>+//
>+// This produces:
>+//
>+//      result[0]       foo
>+//      result[1]       main
>+//           ....       ...
>+//
>+// `result` must not be null.
>+extern int GetStackTrace(void** result, int max_depth, int skip_count);
>+
>+// GetStackTraceWithContext()
>+//
>+// Records program counter values obtained from a signal handler. Records
>+// program counter values for up to `max_depth` frames, skipping the most recent
>+// `skip_count` stack frames, and stores their corresponding values in
>+// `results`. (Note that the frame generated for the
>+// `absl::GetStackFramesWithContext()` routine itself is also skipped.)
>+//
>+// The `uc` parameter, if non-null, should be a pointer to a `ucontext_t` value
>+// passed to a signal handler registered via the `sa_sigaction` field of a
>+// `sigaction` struct. (See
>+// http://man7.org/linux/man-pages/man2/sigaction.2.html.) The `uc` value may
>+// help a stack unwinder to provide a better stack trace under certain
>+// conditions. `uc` may safely be null.
>+//
>+// The `min_dropped_frames` output parameter, if non-null, points to the
>+// location to note any dropped stack frames, if any, due to buffer limitations
>+// or other reasons. (This value will be set to `0` if no frames were dropped.)
>+// The number of total stack frames is guaranteed to be >= skip_count +
>+// max_depth + *min_dropped_frames.
>+extern int GetStackTraceWithContext(void** result, int max_depth,
>+                                    int skip_count, const void* uc,
>+                                    int* min_dropped_frames);
>+
>+// SetStackUnwinder()
>+//
>+// Provides a custom function for unwinding stack frames that will be used in
>+// place of the default stack unwinder when invoking the static
>+// GetStack{Frames,Trace}{,WithContext}() functions above.
>+//
>+// The arguments passed to the unwinder function will match the
>+// arguments passed to `absl::GetStackFramesWithContext()` except that sizes
>+// will be non-null iff the caller is interested in frame sizes.
>+//
>+// If unwinder is set to null, we revert to the default stack-tracing behavior.
>+//
>+// *****************************************************************************
>+// WARNING
>+// *****************************************************************************
>+//
>+// absl::SetStackUnwinder is not suitable for general purpose use.  It is
>+// provided for custom runtimes.
>+// Some things to watch out for when calling `absl::SetStackUnwinder()`:
>+//
>+// (a) The unwinder may be called from within signal handlers and
>+// therefore must be async-signal-safe.
>+//
>+// (b) Even after a custom stack unwinder has been unregistered, other
>+// threads may still be in the process of using that unwinder.
>+// Therefore do not clean up any state that may be needed by an old
>+// unwinder.
>+// *****************************************************************************
>+extern void SetStackUnwinder(int (*unwinder)(void** pcs, int* sizes,
>+                                             int max_depth, int skip_count,
>+                                             const void* uc,
>+                                             int* min_dropped_frames));
>+
>+// DefaultStackUnwinder()
>+//
>+// Records program counter values of up to `max_depth` frames, skipping the most
>+// recent `skip_count` stack frames, and stores their corresponding values in
>+// `pcs`. (Note that the frame generated for this call itself is also skipped.)
>+// This function acts as a generic stack-unwinder; prefer usage of the more
>+// specific `GetStack{Trace,Frames}{,WithContext}()` functions above.
>+//
>+// If you have set your own stack unwinder (with the `SetStackUnwinder()`
>+// function above, you can still get the default stack unwinder by calling
>+// `DefaultStackUnwinder()`, which will ignore any previously set stack unwinder
>+// and use the default one instead.
>+//
>+// Because this function is generic, only `pcs` is guaranteed to be non-null
>+// upon return. It is legal for `sizes`, `uc`, and `min_dropped_frames` to all
>+// be null when called.
>+//
>+// The semantics are the same as the corresponding `GetStack*()` function in the
>+// case where `absl::SetStackUnwinder()` was never called. Equivalents are:
>+//
>+//                       null sizes         |        non-nullptr sizes
>+//             |==========================================================|
>+//     null uc | GetStackTrace()            | GetStackFrames()            |
>+// non-null uc | GetStackTraceWithContext() | GetStackFramesWithContext() |
>+//             |==========================================================|
>+extern int DefaultStackUnwinder(void** pcs, int* sizes, int max_depth,
>+                                int skip_count, const void* uc,
>+                                int* min_dropped_frames);
>+
>+namespace debugging_internal {
>+// Returns true for platforms which are expected to have functioning stack trace
>+// implementations. Intended to be used for tests which want to exclude
>+// verification of logic known to be broken because stack traces are not
>+// working.
>+extern bool StackTraceWorksForTest();
>+}  // namespace debugging_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_DEBUGGING_STACKTRACE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/symbolize.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/symbolize.cc
>new file mode 100644
>index 00000000000..a35e24cc29a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/symbolize.cc
>@@ -0,0 +1,28 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/debugging/symbolize.h"
>+
>+#if defined(ABSL_INTERNAL_HAVE_ELF_SYMBOLIZE)
>+#include "absl/debugging/symbolize_elf.inc"
>+#elif defined(_WIN32) && defined(_DEBUG)
>+// The Windows Symbolizer only works in debug mode. Note that _DEBUG
>+// is the macro that defines whether or not MS C-Runtime debug info is
>+// available. Note that the PDB files containing the debug info must
>+// also be available to the program at runtime for the symbolizer to
>+// work.
>+#include "absl/debugging/symbolize_win32.inc"
>+#else
>+#include "absl/debugging/symbolize_unimplemented.inc"
>+#endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/symbolize.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/symbolize.h
>new file mode 100644
>index 00000000000..24e6e6471c6
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/symbolize.h
>@@ -0,0 +1,97 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// -----------------------------------------------------------------------------
>+// File: symbolize.h
>+// -----------------------------------------------------------------------------
>+//
>+// This file configures the Abseil symbolizer for use in converting instruction
>+// pointer addresses (program counters) into human-readable names (function
>+// calls, etc.) within Abseil code.
>+//
>+// The symbolizer may be invoked from several sources:
>+//
>+//   * Implicitly, through the installation of an Abseil failure signal handler.
>+//     (See failure_signal_handler.h for more information.)
>+//   * By calling `Symbolize()` directly on a program counter you obtain through
>+//     `absl::GetStackTrace()` or `absl::GetStackFrames()`. (See stacktrace.h
>+//     for more information.
>+//   * By calling `Symbolize()` directly on a program counter you obtain through
>+//     other means (which would be platform-dependent).
>+//
>+// In all of the above cases, the symbolizer must first be initialized before
>+// any program counter values can be symbolized. If you are installing a failure
>+// signal handler, initialize the symbolizer before you do so.
>+//
>+// Example:
>+//
>+//   int main(int argc, char** argv) {
>+//     // Initialize the Symbolizer before installing the failure signal handler
>+//     absl::InitializeSymbolizer(argv[0]);
>+//
>+//     // Now you may install the failure signal handler
>+//     absl::FailureSignalHandlerOptions options;
>+//     absl::InstallFailureSignalHandler(options);
>+//
>+//     // Start running your main program
>+//     ...
>+//     return 0;
>+//  }
>+//
>+#ifndef ABSL_DEBUGGING_SYMBOLIZE_H_
>+#define ABSL_DEBUGGING_SYMBOLIZE_H_
>+
>+#include "absl/debugging/internal/symbolize.h"
>+
>+namespace absl {
>+
>+// InitializeSymbolizer()
>+//
>+// Initializes the program counter symbolizer, given the path of the program
>+// (typically obtained through `main()`s `argv[0]`). The Abseil symbolizer
>+// allows you to read program counters (instruction pointer values) using their
>+// human-readable names within output such as stack traces.
>+//
>+// Example:
>+//
>+// int main(int argc, char *argv[]) {
>+//   absl::InitializeSymbolizer(argv[0]);
>+//   // Now you can use the symbolizer
>+// }
>+void InitializeSymbolizer(const char* argv0);
>+
>+// Symbolize()
>+//
>+// Symbolizes a program counter (instruction pointer value) `pc` and, on
>+// success, writes the name to `out`. The symbol name is demangled, if possible.
>+// Note that the symbolized name may be truncated and will be NUL-terminated.
>+// Demangling is supported for symbols generated by GCC 3.x or newer). Returns
>+// `false` on failure.
>+//
>+// Example:
>+//
>+//   // Print a program counter and its symbol name.
>+//   static void DumpPCAndSymbol(void *pc) {
>+//     char tmp[1024];
>+//     const char *symbol = "(unknown)";
>+//     if (absl::Symbolize(pc, tmp, sizeof(tmp))) {
>+//       symbol = tmp;
>+//     }
>+//     absl::PrintF("%*p  %s\n", pc, symbol);
>+//  }
>+bool Symbolize(const void *pc, char *out, int out_size);
>+
>+}  // namespace absl
>+
>+#endif  // ABSL_DEBUGGING_SYMBOLIZE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/symbolize_elf.inc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/symbolize_elf.inc
>new file mode 100644
>index 00000000000..b16a42a62c9
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/symbolize_elf.inc
>@@ -0,0 +1,1473 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+// This library provides Symbolize() function that symbolizes program
>+// counters to their corresponding symbol names on linux platforms.
>+// This library has a minimal implementation of an ELF symbol table
>+// reader (i.e. it doesn't depend on libelf, etc.).
>+//
>+// The algorithm used in Symbolize() is as follows.
>+//
>+//   1. Go through a list of maps in /proc/self/maps and find the map
>+//   containing the program counter.
>+//
>+//   2. Open the mapped file and find a regular symbol table inside.
>+//   Iterate over symbols in the symbol table and look for the symbol
>+//   containing the program counter.  If such a symbol is found,
>+//   obtain the symbol name, and demangle the symbol if possible.
>+//   If the symbol isn't found in the regular symbol table (binary is
>+//   stripped), try the same thing with a dynamic symbol table.
>+//
>+// Note that Symbolize() is originally implemented to be used in
>+// signal handlers, hence it doesn't use malloc() and other unsafe
>+// operations.  It should be both thread-safe and async-signal-safe.
>+//
>+// Implementation note:
>+//
>+// We don't use heaps but only use stacks.  We want to reduce the
>+// stack consumption so that the symbolizer can run on small stacks.
>+//
>+// Here are some numbers collected with GCC 4.1.0 on x86:
>+// - sizeof(Elf32_Sym)  = 16
>+// - sizeof(Elf32_Shdr) = 40
>+// - sizeof(Elf64_Sym)  = 24
>+// - sizeof(Elf64_Shdr) = 64
>+//
>+// This implementation is intended to be async-signal-safe but uses some
>+// functions which are not guaranteed to be so, such as memchr() and
>+// memmove().  We assume they are async-signal-safe.
>+
>+#include <dlfcn.h>
>+#include <elf.h>
>+#include <fcntl.h>
>+#include <link.h>  // For ElfW() macro.
>+#include <sys/stat.h>
>+#include <sys/types.h>
>+#include <unistd.h>
>+
>+#include <algorithm>
>+#include <atomic>
>+#include <cerrno>
>+#include <cinttypes>
>+#include <climits>
>+#include <cstdint>
>+#include <cstdio>
>+#include <cstdlib>
>+#include <cstring>
>+
>+#include "absl/base/casts.h"
>+#include "absl/base/dynamic_annotations.h"
>+#include "absl/base/internal/low_level_alloc.h"
>+#include "absl/base/internal/raw_logging.h"
>+#include "absl/base/internal/spinlock.h"
>+#include "absl/base/port.h"
>+#include "absl/debugging/internal/demangle.h"
>+#include "absl/debugging/internal/vdso_support.h"
>+
>+namespace absl {
>+
>+// Value of argv[0]. Used by MaybeInitializeObjFile().
>+static char *argv0_value = nullptr;
>+
>+void InitializeSymbolizer(const char *argv0) {
>+  if (argv0_value != nullptr) {
>+    free(argv0_value);
>+    argv0_value = nullptr;
>+  }
>+  if (argv0 != nullptr && argv0[0] != '\0') {
>+    argv0_value = strdup(argv0);
>+  }
>+}
>+
>+namespace debugging_internal {
>+namespace {
>+
>+// Re-runs fn until it doesn't cause EINTR.
>+#define NO_INTR(fn) \
>+  do {              \
>+  } while ((fn) < 0 && errno == EINTR)
>+
>+// On Linux, ELF_ST_* are defined in <linux/elf.h>.  To make this portable
>+// we define our own ELF_ST_BIND and ELF_ST_TYPE if not available.
>+#ifndef ELF_ST_BIND
>+#define ELF_ST_BIND(info) (((unsigned char)(info)) >> 4)
>+#endif
>+
>+#ifndef ELF_ST_TYPE
>+#define ELF_ST_TYPE(info) (((unsigned char)(info)) & 0xF)
>+#endif
>+
>+// Some platforms use a special .opd section to store function pointers.
>+const char kOpdSectionName[] = ".opd";
>+
>+#if (defined(__powerpc__) && !(_CALL_ELF > 1)) || defined(__ia64)
>+// Use opd section for function descriptors on these platforms, the function
>+// address is the first word of the descriptor.
>+enum { kPlatformUsesOPDSections = 1 };
>+#else  // not PPC or IA64
>+enum { kPlatformUsesOPDSections = 0 };
>+#endif
>+
>+// This works for PowerPC & IA64 only.  A function descriptor consist of two
>+// pointers and the first one is the function's entry.
>+const size_t kFunctionDescriptorSize = sizeof(void *) * 2;
>+
>+const int kMaxDecorators = 10;  // Seems like a reasonable upper limit.
>+
>+struct InstalledSymbolDecorator {
>+  SymbolDecorator fn;
>+  void *arg;
>+  int ticket;
>+};
>+
>+int g_num_decorators;
>+InstalledSymbolDecorator g_decorators[kMaxDecorators];
>+
>+struct FileMappingHint {
>+  const void *start;
>+  const void *end;
>+  uint64_t offset;
>+  const char *filename;
>+};
>+
>+// Protects g_decorators.
>+// We are using SpinLock and not a Mutex here, because we may be called
>+// from inside Mutex::Lock itself, and it prohibits recursive calls.
>+// This happens in e.g. base/stacktrace_syscall_unittest.
>+// Moreover, we are using only TryLock(), if the decorator list
>+// is being modified (is busy), we skip all decorators, and possibly
>+// loose some info. Sorry, that's the best we could do.
>+base_internal::SpinLock g_decorators_mu(base_internal::kLinkerInitialized);
>+
>+const int kMaxFileMappingHints = 8;
>+int g_num_file_mapping_hints;
>+FileMappingHint g_file_mapping_hints[kMaxFileMappingHints];
>+// Protects g_file_mapping_hints.
>+base_internal::SpinLock g_file_mapping_mu(base_internal::kLinkerInitialized);
>+
>+// Async-signal-safe function to zero a buffer.
>+// memset() is not guaranteed to be async-signal-safe.
>+static void SafeMemZero(void* p, size_t size) {
>+  unsigned char *c = static_cast<unsigned char *>(p);
>+  while (size--) {
>+    *c++ = 0;
>+  }
>+}
>+
>+struct ObjFile {
>+  ObjFile()
>+      : filename(nullptr),
>+        start_addr(nullptr),
>+        end_addr(nullptr),
>+        offset(0),
>+        fd(-1),
>+        elf_type(-1) {
>+    SafeMemZero(&elf_header, sizeof(elf_header));
>+  }
>+
>+  char *filename;
>+  const void *start_addr;
>+  const void *end_addr;
>+  uint64_t offset;
>+
>+  // The following fields are initialized on the first access to the
>+  // object file.
>+  int fd;
>+  int elf_type;
>+  ElfW(Ehdr) elf_header;
>+};
>+
>+// Build 4-way associative cache for symbols. Within each cache line, symbols
>+// are replaced in LRU order.
>+enum {
>+  ASSOCIATIVITY = 4,
>+};
>+struct SymbolCacheLine {
>+  const void *pc[ASSOCIATIVITY];
>+  char *name[ASSOCIATIVITY];
>+
>+  // age[i] is incremented when a line is accessed. it's reset to zero if the
>+  // i'th entry is read.
>+  uint32_t age[ASSOCIATIVITY];
>+};
>+
>+// ---------------------------------------------------------------
>+// An async-signal-safe arena for LowLevelAlloc
>+static std::atomic<base_internal::LowLevelAlloc::Arena *> g_sig_safe_arena;
>+
>+static base_internal::LowLevelAlloc::Arena *SigSafeArena() {
>+  return g_sig_safe_arena.load(std::memory_order_acquire);
>+}
>+
>+static void InitSigSafeArena() {
>+  if (SigSafeArena() == nullptr) {
>+    base_internal::LowLevelAlloc::Arena *new_arena =
>+        base_internal::LowLevelAlloc::NewArena(
>+            base_internal::LowLevelAlloc::kAsyncSignalSafe);
>+    base_internal::LowLevelAlloc::Arena *old_value = nullptr;
>+    if (!g_sig_safe_arena.compare_exchange_strong(old_value, new_arena,
>+                                                  std::memory_order_release,
>+                                                  std::memory_order_relaxed)) {
>+      // We lost a race to allocate an arena; deallocate.
>+      base_internal::LowLevelAlloc::DeleteArena(new_arena);
>+    }
>+  }
>+}
>+
>+// ---------------------------------------------------------------
>+// An AddrMap is a vector of ObjFile, using SigSafeArena() for allocation.
>+
>+class AddrMap {
>+ public:
>+  AddrMap() : size_(0), allocated_(0), obj_(nullptr) {}
>+  ~AddrMap() { base_internal::LowLevelAlloc::Free(obj_); }
>+  int Size() const { return size_; }
>+  ObjFile *At(int i) { return &obj_[i]; }
>+  ObjFile *Add();
>+  void Clear();
>+
>+ private:
>+  int size_;       // count of valid elements (<= allocated_)
>+  int allocated_;  // count of allocated elements
>+  ObjFile *obj_;   // array of allocated_ elements
>+  AddrMap(const AddrMap &) = delete;
>+  AddrMap &operator=(const AddrMap &) = delete;
>+};
>+
>+void AddrMap::Clear() {
>+  for (int i = 0; i != size_; i++) {
>+    At(i)->~ObjFile();
>+  }
>+  size_ = 0;
>+}
>+
>+ObjFile *AddrMap::Add() {
>+  if (size_ == allocated_) {
>+    int new_allocated = allocated_ * 2 + 50;
>+    ObjFile *new_obj_ =
>+        static_cast<ObjFile *>(base_internal::LowLevelAlloc::AllocWithArena(
>+            new_allocated * sizeof(*new_obj_), SigSafeArena()));
>+    if (obj_) {
>+      memcpy(new_obj_, obj_, allocated_ * sizeof(*new_obj_));
>+      base_internal::LowLevelAlloc::Free(obj_);
>+    }
>+    obj_ = new_obj_;
>+    allocated_ = new_allocated;
>+  }
>+  return new (&obj_[size_++]) ObjFile;
>+}
>+
>+// ---------------------------------------------------------------
>+
>+enum FindSymbolResult { SYMBOL_NOT_FOUND = 1, SYMBOL_TRUNCATED, SYMBOL_FOUND };
>+
>+class Symbolizer {
>+ public:
>+  Symbolizer();
>+  ~Symbolizer();
>+  const char *GetSymbol(const void *const pc);
>+
>+ private:
>+  char *CopyString(const char *s) {
>+    int len = strlen(s);
>+    char *dst = static_cast<char *>(
>+        base_internal::LowLevelAlloc::AllocWithArena(len + 1, SigSafeArena()));
>+    ABSL_RAW_CHECK(dst != nullptr, "out of memory");
>+    memcpy(dst, s, len + 1);
>+    return dst;
>+  }
>+  ObjFile *FindObjFile(const void *const start,
>+                       size_t size) ABSL_ATTRIBUTE_NOINLINE;
>+  static bool RegisterObjFile(const char *filename,
>+                              const void *const start_addr,
>+                              const void *const end_addr, uint64_t offset,
>+                              void *arg);
>+  SymbolCacheLine *GetCacheLine(const void *const pc);
>+  const char *FindSymbolInCache(const void *const pc);
>+  const char *InsertSymbolInCache(const void *const pc, const char *name);
>+  void AgeSymbols(SymbolCacheLine *line);
>+  void ClearAddrMap();
>+  FindSymbolResult GetSymbolFromObjectFile(const ObjFile &obj,
>+                                           const void *const pc,
>+                                           const ptrdiff_t relocation,
>+                                           char *out, int out_size,
>+                                           char *tmp_buf, int tmp_buf_size);
>+
>+  enum {
>+    SYMBOL_BUF_SIZE = 2048,
>+    TMP_BUF_SIZE = 1024,
>+    SYMBOL_CACHE_LINES = 128,
>+  };
>+
>+  AddrMap addr_map_;
>+
>+  bool ok_;
>+  bool addr_map_read_;
>+
>+  char symbol_buf_[SYMBOL_BUF_SIZE];
>+
>+  // tmp_buf_ will be used to store arrays of ElfW(Shdr) and ElfW(Sym)
>+  // so we ensure that tmp_buf_ is properly aligned to store either.
>+  alignas(16) char tmp_buf_[TMP_BUF_SIZE];
>+  static_assert(alignof(ElfW(Shdr)) <= 16,
>+                "alignment of tmp buf too small for Shdr");
>+  static_assert(alignof(ElfW(Sym)) <= 16,
>+                "alignment of tmp buf too small for Sym");
>+
>+  SymbolCacheLine symbol_cache_[SYMBOL_CACHE_LINES];
>+};
>+
>+static std::atomic<Symbolizer *> g_cached_symbolizer;
>+
>+}  // namespace
>+
>+static int SymbolizerSize() {
>+  int pagesize = getpagesize();
>+  return ((sizeof(Symbolizer) - 1) / pagesize + 1) * pagesize;
>+}
>+
>+// Return (and set null) g_cached_symbolized_state if it is not null.
>+// Otherwise return a new symbolizer.
>+static Symbolizer *AllocateSymbolizer() {
>+  InitSigSafeArena();
>+  Symbolizer *symbolizer =
>+      g_cached_symbolizer.exchange(nullptr, std::memory_order_acquire);
>+  if (symbolizer != nullptr) {
>+    return symbolizer;
>+  }
>+  return new (base_internal::LowLevelAlloc::AllocWithArena(
>+      SymbolizerSize(), SigSafeArena())) Symbolizer();
>+}
>+
>+// Set g_cached_symbolize_state to s if it is null, otherwise
>+// delete s.
>+static void FreeSymbolizer(Symbolizer *s) {
>+  Symbolizer *old_cached_symbolizer = nullptr;
>+  if (!g_cached_symbolizer.compare_exchange_strong(old_cached_symbolizer, s,
>+                                                   std::memory_order_release,
>+                                                   std::memory_order_relaxed)) {
>+    s->~Symbolizer();
>+    base_internal::LowLevelAlloc::Free(s);
>+  }
>+}
>+
>+Symbolizer::Symbolizer() : ok_(true), addr_map_read_(false) {
>+  for (SymbolCacheLine &symbol_cache_line : symbol_cache_) {
>+    for (size_t j = 0; j < ABSL_ARRAYSIZE(symbol_cache_line.name); ++j) {
>+      symbol_cache_line.pc[j] = nullptr;
>+      symbol_cache_line.name[j] = nullptr;
>+      symbol_cache_line.age[j] = 0;
>+    }
>+  }
>+}
>+
>+Symbolizer::~Symbolizer() {
>+  for (SymbolCacheLine &symbol_cache_line : symbol_cache_) {
>+    for (char *s : symbol_cache_line.name) {
>+      base_internal::LowLevelAlloc::Free(s);
>+    }
>+  }
>+  ClearAddrMap();
>+}
>+
>+// We don't use assert() since it's not guaranteed to be
>+// async-signal-safe.  Instead we define a minimal assertion
>+// macro. So far, we don't need pretty printing for __FILE__, etc.
>+#define SAFE_ASSERT(expr) ((expr) ? static_cast<void>(0) : abort())
>+
>+// Read up to "count" bytes from file descriptor "fd" into the buffer
>+// starting at "buf" while handling short reads and EINTR.  On
>+// success, return the number of bytes read.  Otherwise, return -1.
>+static ssize_t ReadPersistent(int fd, void *buf, size_t count) {
>+  SAFE_ASSERT(fd >= 0);
>+  SAFE_ASSERT(count <= SSIZE_MAX);
>+  char *buf0 = reinterpret_cast<char *>(buf);
>+  size_t num_bytes = 0;
>+  while (num_bytes < count) {
>+    ssize_t len;
>+    NO_INTR(len = read(fd, buf0 + num_bytes, count - num_bytes));
>+    if (len < 0) {  // There was an error other than EINTR.
>+      ABSL_RAW_LOG(WARNING, "read failed: errno=%d", errno);
>+      return -1;
>+    }
>+    if (len == 0) {  // Reached EOF.
>+      break;
>+    }
>+    num_bytes += len;
>+  }
>+  SAFE_ASSERT(num_bytes <= count);
>+  return static_cast<ssize_t>(num_bytes);
>+}
>+
>+// Read up to "count" bytes from "offset" in the file pointed by file
>+// descriptor "fd" into the buffer starting at "buf".  On success,
>+// return the number of bytes read.  Otherwise, return -1.
>+static ssize_t ReadFromOffset(const int fd, void *buf, const size_t count,
>+                              const off_t offset) {
>+  off_t off = lseek(fd, offset, SEEK_SET);
>+  if (off == (off_t)-1) {
>+    ABSL_RAW_LOG(WARNING, "lseek(%d, %ju, SEEK_SET) failed: errno=%d", fd,
>+                 static_cast<uintmax_t>(offset), errno);
>+    return -1;
>+  }
>+  return ReadPersistent(fd, buf, count);
>+}
>+
>+// Try reading exactly "count" bytes from "offset" bytes in a file
>+// pointed by "fd" into the buffer starting at "buf" while handling
>+// short reads and EINTR.  On success, return true. Otherwise, return
>+// false.
>+static bool ReadFromOffsetExact(const int fd, void *buf, const size_t count,
>+                                const off_t offset) {
>+  ssize_t len = ReadFromOffset(fd, buf, count, offset);
>+  return len >= 0 && static_cast<size_t>(len) == count;
>+}
>+
>+// Returns elf_header.e_type if the file pointed by fd is an ELF binary.
>+static int FileGetElfType(const int fd) {
>+  ElfW(Ehdr) elf_header;
>+  if (!ReadFromOffsetExact(fd, &elf_header, sizeof(elf_header), 0)) {
>+    return -1;
>+  }
>+  if (memcmp(elf_header.e_ident, ELFMAG, SELFMAG) != 0) {
>+    return -1;
>+  }
>+  return elf_header.e_type;
>+}
>+
>+// Read the section headers in the given ELF binary, and if a section
>+// of the specified type is found, set the output to this section header
>+// and return true.  Otherwise, return false.
>+// To keep stack consumption low, we would like this function to not get
>+// inlined.
>+static ABSL_ATTRIBUTE_NOINLINE bool GetSectionHeaderByType(
>+    const int fd, ElfW(Half) sh_num, const off_t sh_offset, ElfW(Word) type,
>+    ElfW(Shdr) * out, char *tmp_buf, int tmp_buf_size) {
>+  ElfW(Shdr) *buf = reinterpret_cast<ElfW(Shdr) *>(tmp_buf);
>+  const int buf_entries = tmp_buf_size / sizeof(buf[0]);
>+  const int buf_bytes = buf_entries * sizeof(buf[0]);
>+
>+  for (int i = 0; i < sh_num;) {
>+    const ssize_t num_bytes_left = (sh_num - i) * sizeof(buf[0]);
>+    const ssize_t num_bytes_to_read =
>+        (buf_bytes > num_bytes_left) ? num_bytes_left : buf_bytes;
>+    const off_t offset = sh_offset + i * sizeof(buf[0]);
>+    const ssize_t len = ReadFromOffset(fd, buf, num_bytes_to_read, offset);
>+    if (len % sizeof(buf[0]) != 0) {
>+      ABSL_RAW_LOG(
>+          WARNING,
>+          "Reading %zd bytes from offset %ju returned %zd which is not a "
>+          "multiple of %zu.",
>+          num_bytes_to_read, static_cast<uintmax_t>(offset), len,
>+          sizeof(buf[0]));
>+      return false;
>+    }
>+    const ssize_t num_headers_in_buf = len / sizeof(buf[0]);
>+    SAFE_ASSERT(num_headers_in_buf <= buf_entries);
>+    for (int j = 0; j < num_headers_in_buf; ++j) {
>+      if (buf[j].sh_type == type) {
>+        *out = buf[j];
>+        return true;
>+      }
>+    }
>+    i += num_headers_in_buf;
>+  }
>+  return false;
>+}
>+
>+// There is no particular reason to limit section name to 63 characters,
>+// but there has (as yet) been no need for anything longer either.
>+const int kMaxSectionNameLen = 64;
>+
>+bool ForEachSection(int fd,
>+                    const std::function<bool(const std::string &name,
>+                                             const ElfW(Shdr) &)> &callback) {
>+  ElfW(Ehdr) elf_header;
>+  if (!ReadFromOffsetExact(fd, &elf_header, sizeof(elf_header), 0)) {
>+    return false;
>+  }
>+
>+  ElfW(Shdr) shstrtab;
>+  off_t shstrtab_offset =
>+      (elf_header.e_shoff + elf_header.e_shentsize * elf_header.e_shstrndx);
>+  if (!ReadFromOffsetExact(fd, &shstrtab, sizeof(shstrtab), shstrtab_offset)) {
>+    return false;
>+  }
>+
>+  for (int i = 0; i < elf_header.e_shnum; ++i) {
>+    ElfW(Shdr) out;
>+    off_t section_header_offset =
>+        (elf_header.e_shoff + elf_header.e_shentsize * i);
>+    if (!ReadFromOffsetExact(fd, &out, sizeof(out), section_header_offset)) {
>+      return false;
>+    }
>+    off_t name_offset = shstrtab.sh_offset + out.sh_name;
>+    char header_name[kMaxSectionNameLen + 1];
>+    ssize_t n_read =
>+        ReadFromOffset(fd, &header_name, kMaxSectionNameLen, name_offset);
>+    if (n_read == -1) {
>+      return false;
>+    } else if (n_read > kMaxSectionNameLen) {
>+      // Long read?
>+      return false;
>+    }
>+    header_name[n_read] = '\0';
>+
>+    std::string name(header_name);
>+    if (!callback(name, out)) {
>+      break;
>+    }
>+  }
>+  return true;
>+}
>+
>+// name_len should include terminating '\0'.
>+bool GetSectionHeaderByName(int fd, const char *name, size_t name_len,
>+                            ElfW(Shdr) * out) {
>+  char header_name[kMaxSectionNameLen];
>+  if (sizeof(header_name) < name_len) {
>+    ABSL_RAW_LOG(WARNING,
>+                 "Section name '%s' is too long (%zu); "
>+                 "section will not be found (even if present).",
>+                 name, name_len);
>+    // No point in even trying.
>+    return false;
>+  }
>+
>+  ElfW(Ehdr) elf_header;
>+  if (!ReadFromOffsetExact(fd, &elf_header, sizeof(elf_header), 0)) {
>+    return false;
>+  }
>+
>+  ElfW(Shdr) shstrtab;
>+  off_t shstrtab_offset =
>+      (elf_header.e_shoff + elf_header.e_shentsize * elf_header.e_shstrndx);
>+  if (!ReadFromOffsetExact(fd, &shstrtab, sizeof(shstrtab), shstrtab_offset)) {
>+    return false;
>+  }
>+
>+  for (int i = 0; i < elf_header.e_shnum; ++i) {
>+    off_t section_header_offset =
>+        (elf_header.e_shoff + elf_header.e_shentsize * i);
>+    if (!ReadFromOffsetExact(fd, out, sizeof(*out), section_header_offset)) {
>+      return false;
>+    }
>+    off_t name_offset = shstrtab.sh_offset + out->sh_name;
>+    ssize_t n_read = ReadFromOffset(fd, &header_name, name_len, name_offset);
>+    if (n_read < 0) {
>+      return false;
>+    } else if (static_cast<size_t>(n_read) != name_len) {
>+      // Short read -- name could be at end of file.
>+      continue;
>+    }
>+    if (memcmp(header_name, name, name_len) == 0) {
>+      return true;
>+    }
>+  }
>+  return false;
>+}
>+
>+// Compare symbols at in the same address.
>+// Return true if we should pick symbol1.
>+static bool ShouldPickFirstSymbol(const ElfW(Sym) & symbol1,
>+                                  const ElfW(Sym) & symbol2) {
>+  // If one of the symbols is weak and the other is not, pick the one
>+  // this is not a weak symbol.
>+  char bind1 = ELF_ST_BIND(symbol1.st_info);
>+  char bind2 = ELF_ST_BIND(symbol1.st_info);
>+  if (bind1 == STB_WEAK && bind2 != STB_WEAK) return false;
>+  if (bind2 == STB_WEAK && bind1 != STB_WEAK) return true;
>+
>+  // If one of the symbols has zero size and the other is not, pick the
>+  // one that has non-zero size.
>+  if (symbol1.st_size != 0 && symbol2.st_size == 0) {
>+    return true;
>+  }
>+  if (symbol1.st_size == 0 && symbol2.st_size != 0) {
>+    return false;
>+  }
>+
>+  // If one of the symbols has no type and the other is not, pick the
>+  // one that has a type.
>+  char type1 = ELF_ST_TYPE(symbol1.st_info);
>+  char type2 = ELF_ST_TYPE(symbol1.st_info);
>+  if (type1 != STT_NOTYPE && type2 == STT_NOTYPE) {
>+    return true;
>+  }
>+  if (type1 == STT_NOTYPE && type2 != STT_NOTYPE) {
>+    return false;
>+  }
>+
>+  // Pick the first one, if we still cannot decide.
>+  return true;
>+}
>+
>+// Return true if an address is inside a section.
>+static bool InSection(const void *address, const ElfW(Shdr) * section) {
>+  const char *start = reinterpret_cast<const char *>(section->sh_addr);
>+  size_t size = static_cast<size_t>(section->sh_size);
>+  return start <= address && address < (start + size);
>+}
>+
>+// Read a symbol table and look for the symbol containing the
>+// pc. Iterate over symbols in a symbol table and look for the symbol
>+// containing "pc".  If the symbol is found, and its name fits in
>+// out_size, the name is written into out and SYMBOL_FOUND is returned.
>+// If the name does not fit, truncated name is written into out,
>+// and SYMBOL_TRUNCATED is returned. Out is NUL-terminated.
>+// If the symbol is not found, SYMBOL_NOT_FOUND is returned;
>+// To keep stack consumption low, we would like this function to not get
>+// inlined.
>+static ABSL_ATTRIBUTE_NOINLINE FindSymbolResult FindSymbol(
>+    const void *const pc, const int fd, char *out, int out_size,
>+    ptrdiff_t relocation, const ElfW(Shdr) * strtab, const ElfW(Shdr) * symtab,
>+    const ElfW(Shdr) * opd, char *tmp_buf, int tmp_buf_size) {
>+  if (symtab == nullptr) {
>+    return SYMBOL_NOT_FOUND;
>+  }
>+
>+  // Read multiple symbols at once to save read() calls.
>+  ElfW(Sym) *buf = reinterpret_cast<ElfW(Sym) *>(tmp_buf);
>+  const int buf_entries = tmp_buf_size / sizeof(buf[0]);
>+
>+  const int num_symbols = symtab->sh_size / symtab->sh_entsize;
>+
>+  // On platforms using an .opd section (PowerPC & IA64), a function symbol
>+  // has the address of a function descriptor, which contains the real
>+  // starting address.  However, we do not always want to use the real
>+  // starting address because we sometimes want to symbolize a function
>+  // pointer into the .opd section, e.g. FindSymbol(&foo,...).
>+  const bool pc_in_opd =
>+      kPlatformUsesOPDSections && opd != nullptr && InSection(pc, opd);
>+  const bool deref_function_descriptor_pointer =
>+      kPlatformUsesOPDSections && opd != nullptr && !pc_in_opd;
>+
>+  ElfW(Sym) best_match;
>+  SafeMemZero(&best_match, sizeof(best_match));
>+  bool found_match = false;
>+  for (int i = 0; i < num_symbols;) {
>+    off_t offset = symtab->sh_offset + i * symtab->sh_entsize;
>+    const int num_remaining_symbols = num_symbols - i;
>+    const int entries_in_chunk = std::min(num_remaining_symbols, buf_entries);
>+    const int bytes_in_chunk = entries_in_chunk * sizeof(buf[0]);
>+    const ssize_t len = ReadFromOffset(fd, buf, bytes_in_chunk, offset);
>+    SAFE_ASSERT(len % sizeof(buf[0]) == 0);
>+    const ssize_t num_symbols_in_buf = len / sizeof(buf[0]);
>+    SAFE_ASSERT(num_symbols_in_buf <= entries_in_chunk);
>+    for (int j = 0; j < num_symbols_in_buf; ++j) {
>+      const ElfW(Sym) &symbol = buf[j];
>+
>+      // For a DSO, a symbol address is relocated by the loading address.
>+      // We keep the original address for opd redirection below.
>+      const char *const original_start_address =
>+          reinterpret_cast<const char *>(symbol.st_value);
>+      const char *start_address = original_start_address + relocation;
>+
>+      if (deref_function_descriptor_pointer &&
>+          InSection(original_start_address, opd)) {
>+        // The opd section is mapped into memory.  Just dereference
>+        // start_address to get the first double word, which points to the
>+        // function entry.
>+        start_address = *reinterpret_cast<const char *const *>(start_address);
>+      }
>+
>+      // If pc is inside the .opd section, it points to a function descriptor.
>+      const size_t size = pc_in_opd ? kFunctionDescriptorSize : symbol.st_size;
>+      const void *const end_address =
>+          reinterpret_cast<const char *>(start_address) + size;
>+      if (symbol.st_value != 0 &&  // Skip null value symbols.
>+          symbol.st_shndx != 0 &&  // Skip undefined symbols.
>+#ifdef STT_TLS
>+          ELF_ST_TYPE(symbol.st_info) != STT_TLS &&  // Skip thread-local data.
>+#endif                                               // STT_TLS
>+          ((start_address <= pc && pc < end_address) ||
>+           (start_address == pc && pc == end_address))) {
>+        if (!found_match || ShouldPickFirstSymbol(symbol, best_match)) {
>+          found_match = true;
>+          best_match = symbol;
>+        }
>+      }
>+    }
>+    i += num_symbols_in_buf;
>+  }
>+
>+  if (found_match) {
>+    const size_t off = strtab->sh_offset + best_match.st_name;
>+    const ssize_t n_read = ReadFromOffset(fd, out, out_size, off);
>+    if (n_read <= 0) {
>+      // This should never happen.
>+      ABSL_RAW_LOG(WARNING,
>+                   "Unable to read from fd %d at offset %zu: n_read = %zd", fd,
>+                   off, n_read);
>+      return SYMBOL_NOT_FOUND;
>+    }
>+    ABSL_RAW_CHECK(n_read <= out_size, "ReadFromOffset read too much data.");
>+
>+    // strtab->sh_offset points into .strtab-like section that contains
>+    // NUL-terminated strings: '\0foo\0barbaz\0...".
>+    //
>+    // sh_offset+st_name points to the start of symbol name, but we don't know
>+    // how long the symbol is, so we try to read as much as we have space for,
>+    // and usually over-read (i.e. there is a NUL somewhere before n_read).
>+    if (memchr(out, '\0', n_read) == nullptr) {
>+      // Either out_size was too small (n_read == out_size and no NUL), or
>+      // we tried to read past the EOF (n_read < out_size) and .strtab is
>+      // corrupt (missing terminating NUL; should never happen for valid ELF).
>+      out[n_read - 1] = '\0';
>+      return SYMBOL_TRUNCATED;
>+    }
>+    return SYMBOL_FOUND;
>+  }
>+
>+  return SYMBOL_NOT_FOUND;
>+}
>+
>+// Get the symbol name of "pc" from the file pointed by "fd".  Process
>+// both regular and dynamic symbol tables if necessary.
>+// See FindSymbol() comment for description of return value.
>+FindSymbolResult Symbolizer::GetSymbolFromObjectFile(
>+    const ObjFile &obj, const void *const pc, const ptrdiff_t relocation,
>+    char *out, int out_size, char *tmp_buf, int tmp_buf_size) {
>+  ElfW(Shdr) symtab;
>+  ElfW(Shdr) strtab;
>+  ElfW(Shdr) opd;
>+  ElfW(Shdr) *opd_ptr = nullptr;
>+
>+  // On platforms using an .opd sections for function descriptor, read
>+  // the section header.  The .opd section is in data segment and should be
>+  // loaded but we check that it is mapped just to be extra careful.
>+  if (kPlatformUsesOPDSections) {
>+    if (GetSectionHeaderByName(obj.fd, kOpdSectionName,
>+                               sizeof(kOpdSectionName) - 1, &opd) &&
>+        FindObjFile(reinterpret_cast<const char *>(opd.sh_addr) + relocation,
>+                    opd.sh_size) != nullptr) {
>+      opd_ptr = &opd;
>+    } else {
>+      return SYMBOL_NOT_FOUND;
>+    }
>+  }
>+
>+  // Consult a regular symbol table first.
>+  if (!GetSectionHeaderByType(obj.fd, obj.elf_header.e_shnum,
>+                              obj.elf_header.e_shoff, SHT_SYMTAB, &symtab,
>+                              tmp_buf, tmp_buf_size)) {
>+    return SYMBOL_NOT_FOUND;
>+  }
>+  if (!ReadFromOffsetExact(
>+          obj.fd, &strtab, sizeof(strtab),
>+          obj.elf_header.e_shoff + symtab.sh_link * sizeof(symtab))) {
>+    return SYMBOL_NOT_FOUND;
>+  }
>+  const FindSymbolResult rc =
>+      FindSymbol(pc, obj.fd, out, out_size, relocation, &strtab, &symtab,
>+                 opd_ptr, tmp_buf, tmp_buf_size);
>+  if (rc != SYMBOL_NOT_FOUND) {
>+    return rc;  // Found the symbol in a regular symbol table.
>+  }
>+
>+  // If the symbol is not found, then consult a dynamic symbol table.
>+  if (!GetSectionHeaderByType(obj.fd, obj.elf_header.e_shnum,
>+                              obj.elf_header.e_shoff, SHT_DYNSYM, &symtab,
>+                              tmp_buf, tmp_buf_size)) {
>+    return SYMBOL_NOT_FOUND;
>+  }
>+  if (!ReadFromOffsetExact(
>+          obj.fd, &strtab, sizeof(strtab),
>+          obj.elf_header.e_shoff + symtab.sh_link * sizeof(symtab))) {
>+    return SYMBOL_NOT_FOUND;
>+  }
>+  return FindSymbol(pc, obj.fd, out, out_size, relocation, &strtab, &symtab,
>+                    opd_ptr, tmp_buf, tmp_buf_size);
>+}
>+
>+namespace {
>+// Thin wrapper around a file descriptor so that the file descriptor
>+// gets closed for sure.
>+class FileDescriptor {
>+ public:
>+  explicit FileDescriptor(int fd) : fd_(fd) {}
>+  FileDescriptor(const FileDescriptor &) = delete;
>+  FileDescriptor &operator=(const FileDescriptor &) = delete;
>+
>+  ~FileDescriptor() {
>+    if (fd_ >= 0) {
>+      NO_INTR(close(fd_));
>+    }
>+  }
>+
>+  int get() const { return fd_; }
>+
>+ private:
>+  const int fd_;
>+};
>+
>+// Helper class for reading lines from file.
>+//
>+// Note: we don't use ProcMapsIterator since the object is big (it has
>+// a 5k array member) and uses async-unsafe functions such as sscanf()
>+// and snprintf().
>+class LineReader {
>+ public:
>+  explicit LineReader(int fd, char *buf, int buf_len)
>+      : fd_(fd),
>+        buf_len_(buf_len),
>+        buf_(buf),
>+        bol_(buf),
>+        eol_(buf),
>+        eod_(buf) {}
>+
>+  LineReader(const LineReader &) = delete;
>+  LineReader &operator=(const LineReader &) = delete;
>+
>+  // Read '\n'-terminated line from file.  On success, modify "bol"
>+  // and "eol", then return true.  Otherwise, return false.
>+  //
>+  // Note: if the last line doesn't end with '\n', the line will be
>+  // dropped.  It's an intentional behavior to make the code simple.
>+  bool ReadLine(const char **bol, const char **eol) {
>+    if (BufferIsEmpty()) {  // First time.
>+      const ssize_t num_bytes = ReadPersistent(fd_, buf_, buf_len_);
>+      if (num_bytes <= 0) {  // EOF or error.
>+        return false;
>+      }
>+      eod_ = buf_ + num_bytes;
>+      bol_ = buf_;
>+    } else {
>+      bol_ = eol_ + 1;            // Advance to the next line in the buffer.
>+      SAFE_ASSERT(bol_ <= eod_);  // "bol_" can point to "eod_".
>+      if (!HasCompleteLine()) {
>+        const int incomplete_line_length = eod_ - bol_;
>+        // Move the trailing incomplete line to the beginning.
>+        memmove(buf_, bol_, incomplete_line_length);
>+        // Read text from file and append it.
>+        char *const append_pos = buf_ + incomplete_line_length;
>+        const int capacity_left = buf_len_ - incomplete_line_length;
>+        const ssize_t num_bytes =
>+            ReadPersistent(fd_, append_pos, capacity_left);
>+        if (num_bytes <= 0) {  // EOF or error.
>+          return false;
>+        }
>+        eod_ = append_pos + num_bytes;
>+        bol_ = buf_;
>+      }
>+    }
>+    eol_ = FindLineFeed();
>+    if (eol_ == nullptr) {  // '\n' not found.  Malformed line.
>+      return false;
>+    }
>+    *eol_ = '\0';  // Replace '\n' with '\0'.
>+
>+    *bol = bol_;
>+    *eol = eol_;
>+    return true;
>+  }
>+
>+ private:
>+  char *FindLineFeed() const {
>+    return reinterpret_cast<char *>(memchr(bol_, '\n', eod_ - bol_));
>+  }
>+
>+  bool BufferIsEmpty() const { return buf_ == eod_; }
>+
>+  bool HasCompleteLine() const {
>+    return !BufferIsEmpty() && FindLineFeed() != nullptr;
>+  }
>+
>+  const int fd_;
>+  const int buf_len_;
>+  char *const buf_;
>+  char *bol_;
>+  char *eol_;
>+  const char *eod_;  // End of data in "buf_".
>+};
>+}  // namespace
>+
>+// Place the hex number read from "start" into "*hex".  The pointer to
>+// the first non-hex character or "end" is returned.
>+static const char *GetHex(const char *start, const char *end,
>+                          uint64_t *const value) {
>+  uint64_t hex = 0;
>+  const char *p;
>+  for (p = start; p < end; ++p) {
>+    int ch = *p;
>+    if ((ch >= '0' && ch <= '9') || (ch >= 'A' && ch <= 'F') ||
>+        (ch >= 'a' && ch <= 'f')) {
>+      hex = (hex << 4) | (ch < 'A' ? ch - '0' : (ch & 0xF) + 9);
>+    } else {  // Encountered the first non-hex character.
>+      break;
>+    }
>+  }
>+  SAFE_ASSERT(p <= end);
>+  *value = hex;
>+  return p;
>+}
>+
>+static const char *GetHex(const char *start, const char *end,
>+                          const void **const addr) {
>+  uint64_t hex = 0;
>+  const char *p = GetHex(start, end, &hex);
>+  *addr = reinterpret_cast<void *>(hex);
>+  return p;
>+}
>+
>+// Read /proc/self/maps and run "callback" for each mmapped file found.  If
>+// "callback" returns false, stop scanning and return true. Else continue
>+// scanning /proc/self/maps. Return true if no parse error is found.
>+static ABSL_ATTRIBUTE_NOINLINE bool ReadAddrMap(
>+    bool (*callback)(const char *filename, const void *const start_addr,
>+                     const void *const end_addr, uint64_t offset, void *arg),
>+    void *arg, void *tmp_buf, int tmp_buf_size) {
>+  // Use /proc/self/task/<pid>/maps instead of /proc/self/maps. The latter
>+  // requires kernel to stop all threads, and is significantly slower when there
>+  // are 1000s of threads.
>+  char maps_path[80];
>+  snprintf(maps_path, sizeof(maps_path), "/proc/self/task/%d/maps", getpid());
>+
>+  int maps_fd;
>+  NO_INTR(maps_fd = open(maps_path, O_RDONLY));
>+  FileDescriptor wrapped_maps_fd(maps_fd);
>+  if (wrapped_maps_fd.get() < 0) {
>+    ABSL_RAW_LOG(WARNING, "%s: errno=%d", maps_path, errno);
>+    return false;
>+  }
>+
>+  // Iterate over maps and look for the map containing the pc.  Then
>+  // look into the symbol tables inside.
>+  LineReader reader(wrapped_maps_fd.get(), static_cast<char *>(tmp_buf),
>+                    tmp_buf_size);
>+  while (true) {
>+    const char *cursor;
>+    const char *eol;
>+    if (!reader.ReadLine(&cursor, &eol)) {  // EOF or malformed line.
>+      break;
>+    }
>+
>+    const char *line = cursor;
>+    const void *start_address;
>+    // Start parsing line in /proc/self/maps.  Here is an example:
>+    //
>+    // 08048000-0804c000 r-xp 00000000 08:01 2142121    /bin/cat
>+    //
>+    // We want start address (08048000), end address (0804c000), flags
>+    // (r-xp) and file name (/bin/cat).
>+
>+    // Read start address.
>+    cursor = GetHex(cursor, eol, &start_address);
>+    if (cursor == eol || *cursor != '-') {
>+      ABSL_RAW_LOG(WARNING, "Corrupt /proc/self/maps line: %s", line);
>+      return false;
>+    }
>+    ++cursor;  // Skip '-'.
>+
>+    // Read end address.
>+    const void *end_address;
>+    cursor = GetHex(cursor, eol, &end_address);
>+    if (cursor == eol || *cursor != ' ') {
>+      ABSL_RAW_LOG(WARNING, "Corrupt /proc/self/maps line: %s", line);
>+      return false;
>+    }
>+    ++cursor;  // Skip ' '.
>+
>+    // Read flags.  Skip flags until we encounter a space or eol.
>+    const char *const flags_start = cursor;
>+    while (cursor < eol && *cursor != ' ') {
>+      ++cursor;
>+    }
>+    // We expect at least four letters for flags (ex. "r-xp").
>+    if (cursor == eol || cursor < flags_start + 4) {
>+      ABSL_RAW_LOG(WARNING, "Corrupt /proc/self/maps: %s", line);
>+      return false;
>+    }
>+
>+    // Check flags.  Normally we are only interested in "r-x" maps.  On
>+    // the PowerPC, function pointers point to descriptors in the .opd
>+    // section.  The descriptors themselves are not executable code.  So
>+    // we need to relax the check below to "r**".
>+    if (memcmp(flags_start, "r-x", 3) != 0 &&  // Not a "r-x" map.
>+        !(kPlatformUsesOPDSections && flags_start[0] == 'r')) {
>+      continue;  // We skip this map.
>+    }
>+    ++cursor;  // Skip ' '.
>+
>+    // Read file offset.
>+    uint64_t offset;
>+    cursor = GetHex(cursor, eol, &offset);
>+    ++cursor;  // Skip ' '.
>+
>+    // Skip to file name.  "cursor" now points to dev.  We need to skip at least
>+    // two spaces for dev and inode.
>+    int num_spaces = 0;
>+    while (cursor < eol) {
>+      if (*cursor == ' ') {
>+        ++num_spaces;
>+      } else if (num_spaces >= 2) {
>+        // The first non-space character after  skipping two spaces
>+        // is the beginning of the file name.
>+        break;
>+      }
>+      ++cursor;
>+    }
>+
>+    // Check whether this entry corresponds to our hint table for the true
>+    // filename.
>+    bool hinted =
>+        GetFileMappingHint(&start_address, &end_address, &offset, &cursor);
>+    if (!hinted && (cursor == eol || cursor[0] == '[')) {
>+      // not an object file, typically [vdso] or [vsyscall]
>+      continue;
>+    }
>+    if (!callback(cursor, start_address, end_address, offset, arg)) break;
>+  }
>+  return true;
>+}
>+
>+// Find the objfile mapped in address region containing [addr, addr + len).
>+ObjFile *Symbolizer::FindObjFile(const void *const addr, size_t len) {
>+  for (int i = 0; i < 2; ++i) {
>+    if (!ok_) return nullptr;
>+
>+    // Read /proc/self/maps if necessary
>+    if (!addr_map_read_) {
>+      addr_map_read_ = true;
>+      if (!ReadAddrMap(RegisterObjFile, this, tmp_buf_, TMP_BUF_SIZE)) {
>+        ok_ = false;
>+        return nullptr;
>+      }
>+    }
>+
>+    int lo = 0;
>+    int hi = addr_map_.Size();
>+    while (lo < hi) {
>+      int mid = (lo + hi) / 2;
>+      if (addr < addr_map_.At(mid)->end_addr) {
>+        hi = mid;
>+      } else {
>+        lo = mid + 1;
>+      }
>+    }
>+    if (lo != addr_map_.Size()) {
>+      ObjFile *obj = addr_map_.At(lo);
>+      SAFE_ASSERT(obj->end_addr > addr);
>+      if (addr >= obj->start_addr &&
>+          reinterpret_cast<const char *>(addr) + len <= obj->end_addr)
>+        return obj;
>+    }
>+
>+    // The address mapping may have changed since it was last read.  Retry.
>+    ClearAddrMap();
>+  }
>+  return nullptr;
>+}
>+
>+void Symbolizer::ClearAddrMap() {
>+  for (int i = 0; i != addr_map_.Size(); i++) {
>+    ObjFile *o = addr_map_.At(i);
>+    base_internal::LowLevelAlloc::Free(o->filename);
>+    if (o->fd >= 0) {
>+      NO_INTR(close(o->fd));
>+    }
>+  }
>+  addr_map_.Clear();
>+  addr_map_read_ = false;
>+}
>+
>+// Callback for ReadAddrMap to register objfiles in an in-memory table.
>+bool Symbolizer::RegisterObjFile(const char *filename,
>+                                 const void *const start_addr,
>+                                 const void *const end_addr, uint64_t offset,
>+                                 void *arg) {
>+  Symbolizer *impl = static_cast<Symbolizer *>(arg);
>+
>+  // Files are supposed to be added in the increasing address order.  Make
>+  // sure that's the case.
>+  int addr_map_size = impl->addr_map_.Size();
>+  if (addr_map_size != 0) {
>+    ObjFile *old = impl->addr_map_.At(addr_map_size - 1);
>+    if (old->end_addr > end_addr) {
>+      ABSL_RAW_LOG(ERROR,
>+                   "Unsorted addr map entry: 0x%" PRIxPTR ": %s <-> 0x%" PRIxPTR
>+                   ": %s",
>+                   reinterpret_cast<uintptr_t>(end_addr), filename,
>+                   reinterpret_cast<uintptr_t>(old->end_addr), old->filename);
>+      return true;
>+    } else if (old->end_addr == end_addr) {
>+      // The same entry appears twice. This sometimes happens for [vdso].
>+      if (old->start_addr != start_addr ||
>+          strcmp(old->filename, filename) != 0) {
>+        ABSL_RAW_LOG(ERROR,
>+                     "Duplicate addr 0x%" PRIxPTR ": %s <-> 0x%" PRIxPTR ": %s",
>+                     reinterpret_cast<uintptr_t>(end_addr), filename,
>+                     reinterpret_cast<uintptr_t>(old->end_addr), old->filename);
>+      }
>+      return true;
>+    }
>+  }
>+  ObjFile *obj = impl->addr_map_.Add();
>+  obj->filename = impl->CopyString(filename);
>+  obj->start_addr = start_addr;
>+  obj->end_addr = end_addr;
>+  obj->offset = offset;
>+  obj->elf_type = -1;  // filled on demand
>+  obj->fd = -1;        // opened on demand
>+  return true;
>+}
>+
>+// This function wraps the Demangle function to provide an interface
>+// where the input symbol is demangled in-place.
>+// To keep stack consumption low, we would like this function to not
>+// get inlined.
>+static ABSL_ATTRIBUTE_NOINLINE void DemangleInplace(char *out, int out_size,
>+                                                    char *tmp_buf,
>+                                                    int tmp_buf_size) {
>+  if (Demangle(out, tmp_buf, tmp_buf_size)) {
>+    // Demangling succeeded. Copy to out if the space allows.
>+    int len = strlen(tmp_buf);
>+    if (len + 1 <= out_size) {  // +1 for '\0'.
>+      SAFE_ASSERT(len < tmp_buf_size);
>+      memmove(out, tmp_buf, len + 1);
>+    }
>+  }
>+}
>+
>+SymbolCacheLine *Symbolizer::GetCacheLine(const void *const pc) {
>+  uintptr_t pc0 = reinterpret_cast<uintptr_t>(pc);
>+  pc0 >>= 3;  // drop the low 3 bits
>+
>+  // Shuffle bits.
>+  pc0 ^= (pc0 >> 6) ^ (pc0 >> 12) ^ (pc0 >> 18);
>+  return &symbol_cache_[pc0 % SYMBOL_CACHE_LINES];
>+}
>+
>+void Symbolizer::AgeSymbols(SymbolCacheLine *line) {
>+  for (uint32_t &age : line->age) {
>+    ++age;
>+  }
>+}
>+
>+const char *Symbolizer::FindSymbolInCache(const void *const pc) {
>+  if (pc == nullptr) return nullptr;
>+
>+  SymbolCacheLine *line = GetCacheLine(pc);
>+  for (size_t i = 0; i < ABSL_ARRAYSIZE(line->pc); ++i) {
>+    if (line->pc[i] == pc) {
>+      AgeSymbols(line);
>+      line->age[i] = 0;
>+      return line->name[i];
>+    }
>+  }
>+  return nullptr;
>+}
>+
>+const char *Symbolizer::InsertSymbolInCache(const void *const pc,
>+                                            const char *name) {
>+  SAFE_ASSERT(pc != nullptr);
>+
>+  SymbolCacheLine *line = GetCacheLine(pc);
>+  uint32_t max_age = 0;
>+  int oldest_index = -1;
>+  for (size_t i = 0; i < ABSL_ARRAYSIZE(line->pc); ++i) {
>+    if (line->pc[i] == nullptr) {
>+      AgeSymbols(line);
>+      line->pc[i] = pc;
>+      line->name[i] = CopyString(name);
>+      line->age[i] = 0;
>+      return line->name[i];
>+    }
>+    if (line->age[i] >= max_age) {
>+      max_age = line->age[i];
>+      oldest_index = i;
>+    }
>+  }
>+
>+  AgeSymbols(line);
>+  ABSL_RAW_CHECK(oldest_index >= 0, "Corrupt cache");
>+  base_internal::LowLevelAlloc::Free(line->name[oldest_index]);
>+  line->pc[oldest_index] = pc;
>+  line->name[oldest_index] = CopyString(name);
>+  line->age[oldest_index] = 0;
>+  return line->name[oldest_index];
>+}
>+
>+static void MaybeOpenFdFromSelfExe(ObjFile *obj) {
>+  if (memcmp(obj->start_addr, ELFMAG, SELFMAG) != 0) {
>+    return;
>+  }
>+  int fd = open("/proc/self/exe", O_RDONLY);
>+  if (fd == -1) {
>+    return;
>+  }
>+  // Verify that contents of /proc/self/exe matches in-memory image of
>+  // the binary. This can fail if the "deleted" binary is in fact not
>+  // the main executable, or for binaries that have the first PT_LOAD
>+  // segment smaller than 4K. We do it in four steps so that the
>+  // buffer is smaller and we don't consume too much stack space.
>+  const char *mem = reinterpret_cast<const char *>(obj->start_addr);
>+  for (int i = 0; i < 4; ++i) {
>+    char buf[1024];
>+    ssize_t n = read(fd, buf, sizeof(buf));
>+    if (n != sizeof(buf) || memcmp(buf, mem, sizeof(buf)) != 0) {
>+      close(fd);
>+      return;
>+    }
>+    mem += sizeof(buf);
>+  }
>+  obj->fd = fd;
>+}
>+
>+static bool MaybeInitializeObjFile(ObjFile *obj) {
>+  if (obj->fd < 0) {
>+    obj->fd = open(obj->filename, O_RDONLY);
>+
>+    if (obj->fd < 0) {
>+      // Getting /proc/self/exe here means that we were hinted.
>+      if (strcmp(obj->filename, "/proc/self/exe") == 0) {
>+        // /proc/self/exe may be inaccessible (due to setuid, etc.), so try
>+        // accessing the binary via argv0.
>+        if (argv0_value != nullptr) {
>+          obj->fd = open(argv0_value, O_RDONLY);
>+        }
>+      } else {
>+        MaybeOpenFdFromSelfExe(obj);
>+      }
>+    }
>+
>+    if (obj->fd < 0) {
>+      ABSL_RAW_LOG(WARNING, "%s: open failed: errno=%d", obj->filename, errno);
>+      return false;
>+    }
>+    obj->elf_type = FileGetElfType(obj->fd);
>+    if (obj->elf_type < 0) {
>+      ABSL_RAW_LOG(WARNING, "%s: wrong elf type: %d", obj->filename,
>+                   obj->elf_type);
>+      return false;
>+    }
>+
>+    if (!ReadFromOffsetExact(obj->fd, &obj->elf_header, sizeof(obj->elf_header),
>+                             0)) {
>+      ABSL_RAW_LOG(WARNING, "%s: failed to read elf header", obj->filename);
>+      return false;
>+    }
>+  }
>+  return true;
>+}
>+
>+// The implementation of our symbolization routine.  If it
>+// successfully finds the symbol containing "pc" and obtains the
>+// symbol name, returns pointer to that symbol. Otherwise, returns nullptr.
>+// If any symbol decorators have been installed via InstallSymbolDecorator(),
>+// they are called here as well.
>+// To keep stack consumption low, we would like this function to not
>+// get inlined.
>+const char *Symbolizer::GetSymbol(const void *const pc) {
>+  const char *entry = FindSymbolInCache(pc);
>+  if (entry != nullptr) {
>+    return entry;
>+  }
>+  symbol_buf_[0] = '\0';
>+
>+  ObjFile *const obj = FindObjFile(pc, 1);
>+  ptrdiff_t relocation = 0;
>+  int fd = -1;
>+  if (obj != nullptr) {
>+    if (MaybeInitializeObjFile(obj)) {
>+      if (obj->elf_type == ET_DYN &&
>+          reinterpret_cast<uint64_t>(obj->start_addr) >= obj->offset) {
>+        // This object was relocated.
>+        //
>+        // For obj->offset > 0, adjust the relocation since a mapping at offset
>+        // X in the file will have a start address of [true relocation]+X.
>+        relocation = reinterpret_cast<ptrdiff_t>(obj->start_addr) - obj->offset;
>+      }
>+
>+      fd = obj->fd;
>+    }
>+    if (GetSymbolFromObjectFile(*obj, pc, relocation, symbol_buf_,
>+                                sizeof(symbol_buf_), tmp_buf_,
>+                                sizeof(tmp_buf_)) == SYMBOL_FOUND) {
>+      // Only try to demangle the symbol name if it fit into symbol_buf_.
>+      DemangleInplace(symbol_buf_, sizeof(symbol_buf_), tmp_buf_,
>+                      sizeof(tmp_buf_));
>+    }
>+  } else {
>+#if ABSL_HAVE_VDSO_SUPPORT
>+    VDSOSupport vdso;
>+    if (vdso.IsPresent()) {
>+      VDSOSupport::SymbolInfo symbol_info;
>+      if (vdso.LookupSymbolByAddress(pc, &symbol_info)) {
>+        // All VDSO symbols are known to be short.
>+        size_t len = strlen(symbol_info.name);
>+        ABSL_RAW_CHECK(len + 1 < sizeof(symbol_buf_),
>+                       "VDSO symbol unexpectedly long");
>+        memcpy(symbol_buf_, symbol_info.name, len + 1);
>+      }
>+    }
>+#endif
>+  }
>+
>+  if (g_decorators_mu.TryLock()) {
>+    if (g_num_decorators > 0) {
>+      SymbolDecoratorArgs decorator_args = {
>+          pc,       relocation,       fd,     symbol_buf_, sizeof(symbol_buf_),
>+          tmp_buf_, sizeof(tmp_buf_), nullptr};
>+      for (int i = 0; i < g_num_decorators; ++i) {
>+        decorator_args.arg = g_decorators[i].arg;
>+        g_decorators[i].fn(&decorator_args);
>+      }
>+    }
>+    g_decorators_mu.Unlock();
>+  }
>+  if (symbol_buf_[0] == '\0') {
>+    return nullptr;
>+  }
>+  symbol_buf_[sizeof(symbol_buf_) - 1] = '\0';  // Paranoia.
>+  return InsertSymbolInCache(pc, symbol_buf_);
>+}
>+
>+bool RemoveAllSymbolDecorators(void) {
>+  if (!g_decorators_mu.TryLock()) {
>+    // Someone else is using decorators. Get out.
>+    return false;
>+  }
>+  g_num_decorators = 0;
>+  g_decorators_mu.Unlock();
>+  return true;
>+}
>+
>+bool RemoveSymbolDecorator(int ticket) {
>+  if (!g_decorators_mu.TryLock()) {
>+    // Someone else is using decorators. Get out.
>+    return false;
>+  }
>+  for (int i = 0; i < g_num_decorators; ++i) {
>+    if (g_decorators[i].ticket == ticket) {
>+      while (i < g_num_decorators - 1) {
>+        g_decorators[i] = g_decorators[i + 1];
>+        ++i;
>+      }
>+      g_num_decorators = i;
>+      break;
>+    }
>+  }
>+  g_decorators_mu.Unlock();
>+  return true;  // Decorator is known to be removed.
>+}
>+
>+int InstallSymbolDecorator(SymbolDecorator decorator, void *arg) {
>+  static int ticket = 0;
>+
>+  if (!g_decorators_mu.TryLock()) {
>+    // Someone else is using decorators. Get out.
>+    return false;
>+  }
>+  int ret = ticket;
>+  if (g_num_decorators >= kMaxDecorators) {
>+    ret = -1;
>+  } else {
>+    g_decorators[g_num_decorators] = {decorator, arg, ticket++};
>+    ++g_num_decorators;
>+  }
>+  g_decorators_mu.Unlock();
>+  return ret;
>+}
>+
>+bool RegisterFileMappingHint(const void *start, const void *end, uint64_t offset,
>+                             const char *filename) {
>+  SAFE_ASSERT(start <= end);
>+  SAFE_ASSERT(filename != nullptr);
>+
>+  InitSigSafeArena();
>+
>+  if (!g_file_mapping_mu.TryLock()) {
>+    return false;
>+  }
>+
>+  bool ret = true;
>+  if (g_num_file_mapping_hints >= kMaxFileMappingHints) {
>+    ret = false;
>+  } else {
>+    // TODO(ckennelly): Move this into a std::string copy routine.
>+    int len = strlen(filename);
>+    char *dst = static_cast<char *>(
>+        base_internal::LowLevelAlloc::AllocWithArena(len + 1, SigSafeArena()));
>+    ABSL_RAW_CHECK(dst != nullptr, "out of memory");
>+    memcpy(dst, filename, len + 1);
>+
>+    auto &hint = g_file_mapping_hints[g_num_file_mapping_hints++];
>+    hint.start = start;
>+    hint.end = end;
>+    hint.offset = offset;
>+    hint.filename = dst;
>+  }
>+
>+  g_file_mapping_mu.Unlock();
>+  return ret;
>+}
>+
>+bool GetFileMappingHint(const void **start, const void **end, uint64_t *offset,
>+                        const char **filename) {
>+  if (!g_file_mapping_mu.TryLock()) {
>+    return false;
>+  }
>+  bool found = false;
>+  for (int i = 0; i < g_num_file_mapping_hints; i++) {
>+    if (g_file_mapping_hints[i].start <= *start &&
>+        *end <= g_file_mapping_hints[i].end) {
>+      // We assume that the start_address for the mapping is the base
>+      // address of the ELF section, but when [start_address,end_address) is
>+      // not strictly equal to [hint.start, hint.end), that assumption is
>+      // invalid.
>+      //
>+      // This uses the hint's start address (even though hint.start is not
>+      // necessarily equal to start_address) to ensure the correct
>+      // relocation is computed later.
>+      *start = g_file_mapping_hints[i].start;
>+      *end = g_file_mapping_hints[i].end;
>+      *offset = g_file_mapping_hints[i].offset;
>+      *filename = g_file_mapping_hints[i].filename;
>+      found = true;
>+      break;
>+    }
>+  }
>+  g_file_mapping_mu.Unlock();
>+  return found;
>+}
>+
>+}  // namespace debugging_internal
>+
>+bool Symbolize(const void *pc, char *out, int out_size) {
>+  // Symbolization is very slow under tsan.
>+  ABSL_ANNOTATE_IGNORE_READS_AND_WRITES_BEGIN();
>+  SAFE_ASSERT(out_size >= 0);
>+  debugging_internal::Symbolizer *s = debugging_internal::AllocateSymbolizer();
>+  const char *name = s->GetSymbol(pc);
>+  bool ok = false;
>+  if (name != nullptr && out_size > 0) {
>+    strncpy(out, name, out_size);
>+    ok = true;
>+    if (out[out_size - 1] != '\0') {
>+      // strncpy() does not '\0' terminate when it truncates.  Do so, with
>+      // trailing ellipsis.
>+      static constexpr char kEllipsis[] = "...";
>+      int ellipsis_size =
>+          std::min(implicit_cast<int>(strlen(kEllipsis)), out_size - 1);
>+      memcpy(out + out_size - ellipsis_size - 1, kEllipsis, ellipsis_size);
>+      out[out_size - 1] = '\0';
>+    }
>+  }
>+  debugging_internal::FreeSymbolizer(s);
>+  ABSL_ANNOTATE_IGNORE_READS_AND_WRITES_END();
>+  return ok;
>+}
>+
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/symbolize_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/symbolize_test.cc
>new file mode 100644
>index 00000000000..5f2af47ee45
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/symbolize_test.cc
>@@ -0,0 +1,517 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/debugging/symbolize.h"
>+
>+#ifndef _WIN32
>+#include <fcntl.h>
>+#include <sys/mman.h>
>+#endif
>+
>+#include <cstring>
>+#include <iostream>
>+#include <memory>
>+
>+#include "gmock/gmock.h"
>+#include "gtest/gtest.h"
>+#include "absl/base/attributes.h"
>+#include "absl/base/casts.h"
>+#include "absl/base/internal/per_thread_tls.h"
>+#include "absl/base/internal/raw_logging.h"
>+#include "absl/base/optimization.h"
>+#include "absl/debugging/internal/stack_consumption.h"
>+#include "absl/memory/memory.h"
>+
>+using testing::Contains;
>+
>+// Functions to symbolize. Use C linkage to avoid mangled names.
>+extern "C" {
>+void nonstatic_func() { ABSL_BLOCK_TAIL_CALL_OPTIMIZATION(); }
>+static void static_func() { ABSL_BLOCK_TAIL_CALL_OPTIMIZATION(); }
>+}  // extern "C"
>+
>+struct Foo {
>+  static void func(int x);
>+};
>+
>+// A C++ method that should have a mangled name.
>+void ABSL_ATTRIBUTE_NOINLINE Foo::func(int) {
>+  ABSL_BLOCK_TAIL_CALL_OPTIMIZATION();
>+}
>+
>+// Create functions that will remain in different text sections in the
>+// final binary when linker option "-z,keep-text-section-prefix" is used.
>+int ABSL_ATTRIBUTE_SECTION_VARIABLE(.text.unlikely) unlikely_func() {
>+  return 0;
>+}
>+
>+int ABSL_ATTRIBUTE_SECTION_VARIABLE(.text.hot) hot_func() {
>+  return 0;
>+}
>+
>+int ABSL_ATTRIBUTE_SECTION_VARIABLE(.text.startup) startup_func() {
>+  return 0;
>+}
>+
>+int ABSL_ATTRIBUTE_SECTION_VARIABLE(.text.exit) exit_func() {
>+  return 0;
>+}
>+
>+int /*ABSL_ATTRIBUTE_SECTION_VARIABLE(.text)*/ regular_func() {
>+  return 0;
>+}
>+
>+// Thread-local data may confuse the symbolizer, ensure that it does not.
>+// Variable sizes and order are important.
>+#if ABSL_PER_THREAD_TLS
>+static ABSL_PER_THREAD_TLS_KEYWORD char symbolize_test_thread_small[1];
>+static ABSL_PER_THREAD_TLS_KEYWORD char
>+    symbolize_test_thread_big[2 * 1024 * 1024];
>+#endif
>+
>+// Used below to hopefully inhibit some compiler/linker optimizations
>+// that may remove kHpageTextPadding, kPadding0, and kPadding1 from
>+// the binary.
>+static volatile bool volatile_bool = false;
>+
>+// Force the binary to be large enough that a THP .text remap will succeed.
>+static constexpr size_t kHpageSize = 1 << 21;
>+const char kHpageTextPadding[kHpageSize * 4] ABSL_ATTRIBUTE_SECTION_VARIABLE(
>+    .text) = "";
>+
>+static char try_symbolize_buffer[4096];
>+
>+// A wrapper function for absl::Symbolize() to make the unit test simple.  The
>+// limit must be < sizeof(try_symbolize_buffer).  Returns null if
>+// absl::Symbolize() returns false, otherwise returns try_symbolize_buffer with
>+// the result of absl::Symbolize().
>+static const char *TrySymbolizeWithLimit(void *pc, int limit) {
>+  ABSL_RAW_CHECK(limit <= sizeof(try_symbolize_buffer),
>+                 "try_symbolize_buffer is too small");
>+
>+  // Use the heap to facilitate heap and buffer sanitizer tools.
>+  auto heap_buffer = absl::make_unique<char[]>(sizeof(try_symbolize_buffer));
>+  bool found = absl::Symbolize(pc, heap_buffer.get(), limit);
>+  if (found) {
>+    ABSL_RAW_CHECK(strnlen(heap_buffer.get(), limit) < limit,
>+                   "absl::Symbolize() did not properly terminate the string");
>+    strncpy(try_symbolize_buffer, heap_buffer.get(),
>+            sizeof(try_symbolize_buffer));
>+  }
>+
>+  return found ? try_symbolize_buffer : nullptr;
>+}
>+
>+// A wrapper for TrySymbolizeWithLimit(), with a large limit.
>+static const char *TrySymbolize(void *pc) {
>+  return TrySymbolizeWithLimit(pc, sizeof(try_symbolize_buffer));
>+}
>+
>+#ifdef ABSL_INTERNAL_HAVE_ELF_SYMBOLIZE
>+
>+TEST(Symbolize, Cached) {
>+  // Compilers should give us pointers to them.
>+  EXPECT_STREQ("nonstatic_func", TrySymbolize((void *)(&nonstatic_func)));
>+
>+  // The name of an internal linkage symbol is not specified; allow either a
>+  // mangled or an unmangled name here.
>+  const char *static_func_symbol = TrySymbolize((void *)(&static_func));
>+  EXPECT_TRUE(strcmp("static_func", static_func_symbol) == 0 ||
>+              strcmp("static_func()", static_func_symbol) == 0);
>+
>+  EXPECT_TRUE(nullptr == TrySymbolize(nullptr));
>+}
>+
>+TEST(Symbolize, Truncation) {
>+  constexpr char kNonStaticFunc[] = "nonstatic_func";
>+  EXPECT_STREQ("nonstatic_func",
>+               TrySymbolizeWithLimit((void *)(&nonstatic_func),
>+                                     strlen(kNonStaticFunc) + 1));
>+  EXPECT_STREQ("nonstatic_...",
>+               TrySymbolizeWithLimit((void *)(&nonstatic_func),
>+                                     strlen(kNonStaticFunc) + 0));
>+  EXPECT_STREQ("nonstatic...",
>+               TrySymbolizeWithLimit((void *)(&nonstatic_func),
>+                                     strlen(kNonStaticFunc) - 1));
>+  EXPECT_STREQ("n...", TrySymbolizeWithLimit((void *)(&nonstatic_func), 5));
>+  EXPECT_STREQ("...", TrySymbolizeWithLimit((void *)(&nonstatic_func), 4));
>+  EXPECT_STREQ("..", TrySymbolizeWithLimit((void *)(&nonstatic_func), 3));
>+  EXPECT_STREQ(".", TrySymbolizeWithLimit((void *)(&nonstatic_func), 2));
>+  EXPECT_STREQ("", TrySymbolizeWithLimit((void *)(&nonstatic_func), 1));
>+  EXPECT_EQ(nullptr, TrySymbolizeWithLimit((void *)(&nonstatic_func), 0));
>+}
>+
>+TEST(Symbolize, SymbolizeWithDemangling) {
>+  Foo::func(100);
>+  EXPECT_STREQ("Foo::func()", TrySymbolize((void *)(&Foo::func)));
>+}
>+
>+TEST(Symbolize, SymbolizeSplitTextSections) {
>+  EXPECT_STREQ("unlikely_func()", TrySymbolize((void *)(&unlikely_func)));
>+  EXPECT_STREQ("hot_func()", TrySymbolize((void *)(&hot_func)));
>+  EXPECT_STREQ("startup_func()", TrySymbolize((void *)(&startup_func)));
>+  EXPECT_STREQ("exit_func()", TrySymbolize((void *)(&exit_func)));
>+  EXPECT_STREQ("regular_func()", TrySymbolize((void *)(&regular_func)));
>+}
>+
>+// Tests that verify that Symbolize stack footprint is within some limit.
>+#ifdef ABSL_INTERNAL_HAVE_DEBUGGING_STACK_CONSUMPTION
>+
>+static void *g_pc_to_symbolize;
>+static char g_symbolize_buffer[4096];
>+static char *g_symbolize_result;
>+
>+static void SymbolizeSignalHandler(int signo) {
>+  if (absl::Symbolize(g_pc_to_symbolize, g_symbolize_buffer,
>+                      sizeof(g_symbolize_buffer))) {
>+    g_symbolize_result = g_symbolize_buffer;
>+  } else {
>+    g_symbolize_result = nullptr;
>+  }
>+}
>+
>+// Call Symbolize and figure out the stack footprint of this call.
>+static const char *SymbolizeStackConsumption(void *pc, int *stack_consumed) {
>+  g_pc_to_symbolize = pc;
>+  *stack_consumed = absl::debugging_internal::GetSignalHandlerStackConsumption(
>+      SymbolizeSignalHandler);
>+  return g_symbolize_result;
>+}
>+
>+static int GetStackConsumptionUpperLimit() {
>+  // Symbolize stack consumption should be within 2kB.
>+  int stack_consumption_upper_limit = 2048;
>+#if defined(ADDRESS_SANITIZER) || defined(MEMORY_SANITIZER) || \
>+    defined(THREAD_SANITIZER)
>+  // Account for sanitizer instrumentation requiring additional stack space.
>+  stack_consumption_upper_limit *= 5;
>+#endif
>+  return stack_consumption_upper_limit;
>+}
>+
>+TEST(Symbolize, SymbolizeStackConsumption) {
>+  int stack_consumed = 0;
>+
>+  const char *symbol =
>+      SymbolizeStackConsumption((void *)(&nonstatic_func), &stack_consumed);
>+  EXPECT_STREQ("nonstatic_func", symbol);
>+  EXPECT_GT(stack_consumed, 0);
>+  EXPECT_LT(stack_consumed, GetStackConsumptionUpperLimit());
>+
>+  // The name of an internal linkage symbol is not specified; allow either a
>+  // mangled or an unmangled name here.
>+  symbol = SymbolizeStackConsumption((void *)(&static_func), &stack_consumed);
>+  EXPECT_TRUE(strcmp("static_func", symbol) == 0 ||
>+              strcmp("static_func()", symbol) == 0);
>+  EXPECT_GT(stack_consumed, 0);
>+  EXPECT_LT(stack_consumed, GetStackConsumptionUpperLimit());
>+}
>+
>+TEST(Symbolize, SymbolizeWithDemanglingStackConsumption) {
>+  Foo::func(100);
>+  int stack_consumed = 0;
>+
>+  const char *symbol =
>+      SymbolizeStackConsumption((void *)(&Foo::func), &stack_consumed);
>+
>+  EXPECT_STREQ("Foo::func()", symbol);
>+  EXPECT_GT(stack_consumed, 0);
>+  EXPECT_LT(stack_consumed, GetStackConsumptionUpperLimit());
>+}
>+
>+#endif  // ABSL_INTERNAL_HAVE_DEBUGGING_STACK_CONSUMPTION
>+
>+// Use a 64K page size for PPC.
>+const size_t kPageSize = 64 << 10;
>+// We place a read-only symbols into the .text section and verify that we can
>+// symbolize them and other symbols after remapping them.
>+const char kPadding0[kPageSize * 4] ABSL_ATTRIBUTE_SECTION_VARIABLE(.text) =
>+    "";
>+const char kPadding1[kPageSize * 4] ABSL_ATTRIBUTE_SECTION_VARIABLE(.text) =
>+    "";
>+
>+static int FilterElfHeader(struct dl_phdr_info *info, size_t size, void *data) {
>+  for (int i = 0; i < info->dlpi_phnum; i++) {
>+    if (info->dlpi_phdr[i].p_type == PT_LOAD &&
>+        info->dlpi_phdr[i].p_flags == (PF_R | PF_X)) {
>+      const void *const vaddr =
>+          absl::bit_cast<void *>(info->dlpi_addr + info->dlpi_phdr[i].p_vaddr);
>+      const auto segsize = info->dlpi_phdr[i].p_memsz;
>+
>+      const char *self_exe;
>+      if (info->dlpi_name != nullptr && info->dlpi_name[0] != '\0') {
>+        self_exe = info->dlpi_name;
>+      } else {
>+        self_exe = "/proc/self/exe";
>+      }
>+
>+      absl::debugging_internal::RegisterFileMappingHint(
>+          vaddr, reinterpret_cast<const char *>(vaddr) + segsize,
>+          info->dlpi_phdr[i].p_offset, self_exe);
>+
>+      return 1;
>+    }
>+  }
>+
>+  return 1;
>+}
>+
>+TEST(Symbolize, SymbolizeWithMultipleMaps) {
>+  // Force kPadding0 and kPadding1 to be linked in.
>+  if (volatile_bool) {
>+    ABSL_RAW_LOG(INFO, "%s", kPadding0);
>+    ABSL_RAW_LOG(INFO, "%s", kPadding1);
>+  }
>+
>+  // Verify we can symbolize everything.
>+  char buf[512];
>+  memset(buf, 0, sizeof(buf));
>+  absl::Symbolize(kPadding0, buf, sizeof(buf));
>+  EXPECT_STREQ("kPadding0", buf);
>+
>+  memset(buf, 0, sizeof(buf));
>+  absl::Symbolize(kPadding1, buf, sizeof(buf));
>+  EXPECT_STREQ("kPadding1", buf);
>+
>+  // Specify a hint for the executable segment.
>+  dl_iterate_phdr(FilterElfHeader, nullptr);
>+
>+  // Reload at least one page out of kPadding0, kPadding1
>+  const char *ptrs[] = {kPadding0, kPadding1};
>+
>+  for (const char *ptr : ptrs) {
>+    const int kMapFlags = MAP_ANONYMOUS | MAP_PRIVATE;
>+    void *addr = mmap(nullptr, kPageSize, PROT_READ, kMapFlags, 0, 0);
>+    ASSERT_NE(addr, MAP_FAILED);
>+
>+    // kPadding[0-1] is full of zeroes, so we can remap anywhere within it, but
>+    // we ensure there is at least a full page of padding.
>+    void *remapped = reinterpret_cast<void *>(
>+        reinterpret_cast<uintptr_t>(ptr + kPageSize) & ~(kPageSize - 1ULL));
>+
>+    const int kMremapFlags = (MREMAP_MAYMOVE | MREMAP_FIXED);
>+    void *ret = mremap(addr, kPageSize, kPageSize, kMremapFlags, remapped);
>+    ASSERT_NE(ret, MAP_FAILED);
>+  }
>+
>+  // Invalidate the symbolization cache so we are forced to rely on the hint.
>+  absl::Symbolize(nullptr, buf, sizeof(buf));
>+
>+  // Verify we can still symbolize.
>+  const char *expected[] = {"kPadding0", "kPadding1"};
>+  const size_t offsets[] = {0, kPageSize, 2 * kPageSize, 3 * kPageSize};
>+
>+  for (int i = 0; i < 2; i++) {
>+    for (size_t offset : offsets) {
>+      memset(buf, 0, sizeof(buf));
>+      absl::Symbolize(ptrs[i] + offset, buf, sizeof(buf));
>+      EXPECT_STREQ(expected[i], buf);
>+    }
>+  }
>+}
>+
>+// Appends std::string(*args->arg) to args->symbol_buf.
>+static void DummySymbolDecorator(
>+    const absl::debugging_internal::SymbolDecoratorArgs *args) {
>+  std::string *message = static_cast<std::string *>(args->arg);
>+  strncat(args->symbol_buf, message->c_str(),
>+          args->symbol_buf_size - strlen(args->symbol_buf) - 1);
>+}
>+
>+TEST(Symbolize, InstallAndRemoveSymbolDecorators) {
>+  int ticket_a;
>+  std::string a_message("a");
>+  EXPECT_GE(ticket_a = absl::debugging_internal::InstallSymbolDecorator(
>+                DummySymbolDecorator, &a_message),
>+            0);
>+
>+  int ticket_b;
>+  std::string b_message("b");
>+  EXPECT_GE(ticket_b = absl::debugging_internal::InstallSymbolDecorator(
>+                DummySymbolDecorator, &b_message),
>+            0);
>+
>+  int ticket_c;
>+  std::string c_message("c");
>+  EXPECT_GE(ticket_c = absl::debugging_internal::InstallSymbolDecorator(
>+                DummySymbolDecorator, &c_message),
>+            0);
>+
>+  char *address = reinterpret_cast<char *>(1);
>+  EXPECT_STREQ("abc", TrySymbolize(address++));
>+
>+  EXPECT_TRUE(absl::debugging_internal::RemoveSymbolDecorator(ticket_b));
>+
>+  EXPECT_STREQ("ac", TrySymbolize(address++));
>+
>+  // Cleanup: remove all remaining decorators so other stack traces don't
>+  // get mystery "ac" decoration.
>+  EXPECT_TRUE(absl::debugging_internal::RemoveSymbolDecorator(ticket_a));
>+  EXPECT_TRUE(absl::debugging_internal::RemoveSymbolDecorator(ticket_c));
>+}
>+
>+// Some versions of Clang with optimizations enabled seem to be able
>+// to optimize away the .data section if no variables live in the
>+// section. This variable should get placed in the .data section, and
>+// the test below checks for the existence of a .data section.
>+static int in_data_section = 1;
>+
>+TEST(Symbolize, ForEachSection) {
>+  int fd = TEMP_FAILURE_RETRY(open("/proc/self/exe", O_RDONLY));
>+  ASSERT_NE(fd, -1);
>+
>+  std::vector<std::string> sections;
>+  ASSERT_TRUE(absl::debugging_internal::ForEachSection(
>+      fd, [&sections](const std::string &name, const ElfW(Shdr) &) {
>+        sections.push_back(name);
>+        return true;
>+      }));
>+
>+  // Check for the presence of common section names.
>+  EXPECT_THAT(sections, Contains(".text"));
>+  EXPECT_THAT(sections, Contains(".rodata"));
>+  EXPECT_THAT(sections, Contains(".bss"));
>+  ++in_data_section;
>+  EXPECT_THAT(sections, Contains(".data"));
>+
>+  close(fd);
>+}
>+
>+// x86 specific tests.  Uses some inline assembler.
>+extern "C" {
>+inline void *ABSL_ATTRIBUTE_ALWAYS_INLINE inline_func() {
>+  void *pc = nullptr;
>+#if defined(__i386__) || defined(__x86_64__)
>+  __asm__ __volatile__("call 1f; 1: pop %0" : "=r"(pc));
>+#endif
>+  return pc;
>+}
>+
>+void *ABSL_ATTRIBUTE_NOINLINE non_inline_func() {
>+  void *pc = nullptr;
>+#if defined(__i386__) || defined(__x86_64__)
>+  __asm__ __volatile__("call 1f; 1: pop %0" : "=r"(pc));
>+#endif
>+  return pc;
>+}
>+
>+void ABSL_ATTRIBUTE_NOINLINE TestWithPCInsideNonInlineFunction() {
>+#if defined(ABSL_HAVE_ATTRIBUTE_NOINLINE) && \
>+    (defined(__i386__) || defined(__x86_64__))
>+  void *pc = non_inline_func();
>+  const char *symbol = TrySymbolize(pc);
>+  ABSL_RAW_CHECK(symbol != nullptr, "TestWithPCInsideNonInlineFunction failed");
>+  ABSL_RAW_CHECK(strcmp(symbol, "non_inline_func") == 0,
>+                 "TestWithPCInsideNonInlineFunction failed");
>+  std::cout << "TestWithPCInsideNonInlineFunction passed" << std::endl;
>+#endif
>+}
>+
>+void ABSL_ATTRIBUTE_NOINLINE TestWithPCInsideInlineFunction() {
>+#if defined(ABSL_HAVE_ATTRIBUTE_ALWAYS_INLINE) && \
>+    (defined(__i386__) || defined(__x86_64__))
>+  void *pc = inline_func();  // Must be inlined.
>+  const char *symbol = TrySymbolize(pc);
>+  ABSL_RAW_CHECK(symbol != nullptr, "TestWithPCInsideInlineFunction failed");
>+  ABSL_RAW_CHECK(strcmp(symbol, __FUNCTION__) == 0,
>+                 "TestWithPCInsideInlineFunction failed");
>+  std::cout << "TestWithPCInsideInlineFunction passed" << std::endl;
>+#endif
>+}
>+}
>+
>+// Test with a return address.
>+void ABSL_ATTRIBUTE_NOINLINE TestWithReturnAddress() {
>+#if defined(ABSL_HAVE_ATTRIBUTE_NOINLINE)
>+  void *return_address = __builtin_return_address(0);
>+  const char *symbol = TrySymbolize(return_address);
>+  ABSL_RAW_CHECK(symbol != nullptr, "TestWithReturnAddress failed");
>+  ABSL_RAW_CHECK(strcmp(symbol, "main") == 0, "TestWithReturnAddress failed");
>+  std::cout << "TestWithReturnAddress passed" << std::endl;
>+#endif
>+}
>+
>+#elif defined(_WIN32) && defined(_DEBUG)
>+
>+TEST(Symbolize, Basics) {
>+  EXPECT_STREQ("nonstatic_func", TrySymbolize((void *)(&nonstatic_func)));
>+
>+  // The name of an internal linkage symbol is not specified; allow either a
>+  // mangled or an unmangled name here.
>+  const char* static_func_symbol = TrySymbolize((void *)(&static_func));
>+  ASSERT_TRUE(static_func_symbol != nullptr);
>+  EXPECT_TRUE(strstr(static_func_symbol, "static_func") != nullptr);
>+
>+  EXPECT_TRUE(nullptr == TrySymbolize(nullptr));
>+}
>+
>+TEST(Symbolize, Truncation) {
>+  constexpr char kNonStaticFunc[] = "nonstatic_func";
>+  EXPECT_STREQ("nonstatic_func",
>+               TrySymbolizeWithLimit((void *)(&nonstatic_func),
>+                                     strlen(kNonStaticFunc) + 1));
>+  EXPECT_STREQ("nonstatic_...",
>+               TrySymbolizeWithLimit((void *)(&nonstatic_func),
>+                                     strlen(kNonStaticFunc) + 0));
>+  EXPECT_STREQ("nonstatic...",
>+               TrySymbolizeWithLimit((void *)(&nonstatic_func),
>+                                     strlen(kNonStaticFunc) - 1));
>+  EXPECT_STREQ("n...", TrySymbolizeWithLimit((void *)(&nonstatic_func), 5));
>+  EXPECT_STREQ("...", TrySymbolizeWithLimit((void *)(&nonstatic_func), 4));
>+  EXPECT_STREQ("..", TrySymbolizeWithLimit((void *)(&nonstatic_func), 3));
>+  EXPECT_STREQ(".", TrySymbolizeWithLimit((void *)(&nonstatic_func), 2));
>+  EXPECT_STREQ("", TrySymbolizeWithLimit((void *)(&nonstatic_func), 1));
>+  EXPECT_EQ(nullptr, TrySymbolizeWithLimit((void *)(&nonstatic_func), 0));
>+}
>+
>+TEST(Symbolize, SymbolizeWithDemangling) {
>+  const char* result = TrySymbolize((void *)(&Foo::func));
>+  ASSERT_TRUE(result != nullptr);
>+  EXPECT_TRUE(strstr(result, "Foo::func") != nullptr) << result;
>+}
>+
>+#else  // Symbolizer unimplemented
>+
>+TEST(Symbolize, Unimplemented) {
>+  char buf[64];
>+  EXPECT_FALSE(absl::Symbolize((void *)(&nonstatic_func), buf, sizeof(buf)));
>+  EXPECT_FALSE(absl::Symbolize((void *)(&static_func), buf, sizeof(buf)));
>+  EXPECT_FALSE(absl::Symbolize((void *)(&Foo::func), buf, sizeof(buf)));
>+}
>+
>+#endif
>+
>+int main(int argc, char **argv) {
>+  // Make sure kHpageTextPadding is linked into the binary.
>+  if (volatile_bool) {
>+    ABSL_RAW_LOG(INFO, "%s", kHpageTextPadding);
>+  }
>+
>+#if ABSL_PER_THREAD_TLS
>+  // Touch the per-thread variables.
>+  symbolize_test_thread_small[0] = 0;
>+  symbolize_test_thread_big[0] = 0;
>+#endif
>+
>+  absl::InitializeSymbolizer(argv[0]);
>+  testing::InitGoogleTest(&argc, argv);
>+
>+#ifdef ABSL_INTERNAL_HAVE_ELF_SYMBOLIZE
>+  TestWithPCInsideInlineFunction();
>+  TestWithPCInsideNonInlineFunction();
>+  TestWithReturnAddress();
>+#endif
>+
>+  return RUN_ALL_TESTS();
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/symbolize_unimplemented.inc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/symbolize_unimplemented.inc
>new file mode 100644
>index 00000000000..2a3f4acb0e1
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/symbolize_unimplemented.inc
>@@ -0,0 +1,35 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include <cstdint>
>+
>+#include "absl/base/internal/raw_logging.h"
>+
>+namespace absl {
>+
>+namespace debugging_internal {
>+
>+int InstallSymbolDecorator(SymbolDecorator, void*) { return -1; }
>+bool RemoveSymbolDecorator(int) { return false; }
>+bool RemoveAllSymbolDecorators(void) { return false; }
>+bool RegisterFileMappingHint(const void *, const void *, uint64_t, const char *) {
>+  return false;
>+}
>+
>+}  // namespace debugging_internal
>+
>+void InitializeSymbolizer(const char*) {}
>+bool Symbolize(const void *, char *, int) { return false; }
>+
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/symbolize_win32.inc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/symbolize_win32.inc
>new file mode 100644
>index 00000000000..e3fff74d493
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/debugging/symbolize_win32.inc
>@@ -0,0 +1,74 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+// See "Retrieving Symbol Information by Address":
>+// https://msdn.microsoft.com/en-us/library/windows/desktop/ms680578(v=vs.85).aspx
>+
>+#include <windows.h>
>+#include <DbgHelp.h>
>+#pragma comment(lib, "DbgHelp")
>+
>+#include <algorithm>
>+#include <cstring>
>+
>+#include "absl/base/internal/raw_logging.h"
>+
>+namespace absl {
>+
>+static HANDLE process = NULL;
>+
>+void InitializeSymbolizer(const char *argv0) {
>+  if (process != nullptr) {
>+    return;
>+  }
>+  process = GetCurrentProcess();
>+
>+  // Symbols are not loaded until a reference is made requiring the
>+  // symbols be loaded. This is the fastest, most efficient way to use
>+  // the symbol handler.
>+  SymSetOptions(SYMOPT_DEFERRED_LOADS | SYMOPT_UNDNAME);
>+  if (!SymInitialize(process, nullptr, true)) {
>+    // GetLastError() returns a Win32 DWORD, but we assign to
>+    // unsigned long long to simplify the ABSL_RAW_LOG case below.  The uniform
>+    // initialization guarantees this is not a narrowing conversion.
>+    const unsigned long long error{GetLastError()};  // NOLINT(runtime/int)
>+    ABSL_RAW_LOG(FATAL, "SymInitialize() failed: %llu", error);
>+  }
>+}
>+
>+bool Symbolize(const void *pc, char *out, int out_size) {
>+  if (out_size <= 0) {
>+    return false;
>+  }
>+  std::aligned_storage<sizeof(SYMBOL_INFO) + MAX_SYM_NAME,
>+                       alignof(SYMBOL_INFO)>::type buf;
>+  SYMBOL_INFO *symbol = reinterpret_cast<SYMBOL_INFO *>(&buf);
>+  symbol->SizeOfStruct = sizeof(SYMBOL_INFO);
>+  symbol->MaxNameLen = MAX_SYM_NAME;
>+  if (!SymFromAddr(process, reinterpret_cast<DWORD64>(pc), nullptr, symbol)) {
>+    return false;
>+  }
>+  strncpy(out, symbol->Name, out_size);
>+  if (out[out_size - 1] != '\0') {
>+    // strncpy() does not '\0' terminate when it truncates.
>+    static constexpr char kEllipsis[] = "...";
>+    int ellipsis_size =
>+        std::min<int>(sizeof(kEllipsis) - 1, out_size - 1);
>+    memcpy(out + out_size - ellipsis_size - 1, kEllipsis, ellipsis_size);
>+    out[out_size - 1] = '\0';
>+  }
>+  return true;
>+}
>+
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/memory/BUILD.bazel b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/memory/BUILD.bazel
>new file mode 100644
>index 00000000000..46f47b12bb2
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/memory/BUILD.bazel
>@@ -0,0 +1,61 @@
>+#
>+# Copyright 2017 The Abseil Authors.
>+#
>+# Licensed under the Apache License, Version 2.0 (the "License");
>+# you may not use this file except in compliance with the License.
>+# You may obtain a copy of the License at
>+#
>+#      http://www.apache.org/licenses/LICENSE-2.0
>+#
>+# Unless required by applicable law or agreed to in writing, software
>+# distributed under the License is distributed on an "AS IS" BASIS,
>+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+# See the License for the specific language governing permissions and
>+# limitations under the License.
>+#
>+
>+load(
>+    "//absl:copts.bzl",
>+    "ABSL_DEFAULT_COPTS",
>+    "ABSL_TEST_COPTS",
>+    "ABSL_EXCEPTIONS_FLAG",
>+)
>+
>+package(default_visibility = ["//visibility:public"])
>+
>+licenses(["notice"])  # Apache 2.0
>+
>+cc_library(
>+    name = "memory",
>+    hdrs = ["memory.h"],
>+    copts = ABSL_DEFAULT_COPTS,
>+    deps = [
>+        "//absl/base:core_headers",
>+        "//absl/meta:type_traits",
>+    ],
>+)
>+
>+cc_test(
>+    name = "memory_test",
>+    srcs = ["memory_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    deps = [
>+        ":memory",
>+        "//absl/base",
>+        "//absl/base:core_headers",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "memory_exception_safety_test",
>+    srcs = [
>+        "memory_exception_safety_test.cc",
>+    ],
>+    copts = ABSL_TEST_COPTS + ABSL_EXCEPTIONS_FLAG,
>+    deps = [
>+        ":memory",
>+        "//absl/base:exception_safety_testing",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/memory/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/memory/BUILD.gn
>new file mode 100644
>index 00000000000..30224c6a112
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/memory/BUILD.gn
>@@ -0,0 +1,31 @@
>+# Copyright 2018 The Chromium Authors. All rights reserved.
>+# Use of this source code is governed by a BSD-style license that can be
>+# found in the LICENSE file.
>+
>+import("//build_overrides/build.gni")
>+
>+if (build_with_chromium) {
>+  visibility = [
>+    "//third_party/webrtc/*",
>+    "//third_party/abseil-cpp/*",
>+    "//third_party/googletest:gtest",
>+  ]
>+} else {
>+  visibility = [ "*" ]
>+}
>+
>+source_set("memory") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  public = [
>+    "memory.h",
>+  ]
>+  deps = [
>+    "../base:core_headers",
>+    "../meta:type_traits",
>+  ]
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/memory/CMakeLists.txt b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/memory/CMakeLists.txt
>new file mode 100644
>index 00000000000..5958f5c52bf
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/memory/CMakeLists.txt
>@@ -0,0 +1,71 @@
>+#
>+# Copyright 2017 The Abseil Authors.
>+#
>+# Licensed under the Apache License, Version 2.0 (the "License");
>+# you may not use this file except in compliance with the License.
>+# You may obtain a copy of the License at
>+#
>+#      http://www.apache.org/licenses/LICENSE-2.0
>+#
>+# Unless required by applicable law or agreed to in writing, software
>+# distributed under the License is distributed on an "AS IS" BASIS,
>+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+# See the License for the specific language governing permissions and
>+# limitations under the License.
>+#
>+
>+list(APPEND MEMORY_PUBLIC_HEADERS
>+  "memory.h"
>+)
>+
>+
>+absl_header_library(
>+  TARGET
>+    absl_memory
>+  EXPORT_NAME
>+    memory
>+)
>+
>+#
>+## TESTS
>+#
>+
>+# test memory_test
>+list(APPEND MEMORY_TEST_SRC
>+  "memory_test.cc"
>+  ${MEMORY_PUBLIC_HEADERS}
>+)
>+set(MEMORY_TEST_PUBLIC_LIBRARIES absl::base absl::memory)
>+
>+
>+
>+absl_test(
>+  TARGET
>+    memory_test
>+  SOURCES
>+    ${MEMORY_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${MEMORY_TEST_PUBLIC_LIBRARIES}
>+)
>+
>+
>+# test memory_exception_safety_test
>+set(MEMORY_EXCEPTION_SAFETY_TEST_SRC "memory_exception_safety_test.cc")
>+set(MEMORY_EXCEPTION_SAFETY_TEST_PUBLIC_LIBRARIES
>+  absl::memory
>+  absl_base_internal_exception_safety_testing
>+)
>+
>+absl_test(
>+  TARGET
>+    memory_exception_safety_test
>+  SOURCES
>+    ${MEMORY_EXCEPTION_SAFETY_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${MEMORY_EXCEPTION_SAFETY_TEST_PUBLIC_LIBRARIES}
>+  PRIVATE_COMPILE_FLAGS
>+    ${ABSL_EXCEPTIONS_FLAG}
>+)
>+
>+
>+
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/memory/memory.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/memory/memory.h
>new file mode 100644
>index 00000000000..4d9cc0075df
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/memory/memory.h
>@@ -0,0 +1,697 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// -----------------------------------------------------------------------------
>+// File: memory.h
>+// -----------------------------------------------------------------------------
>+//
>+// This header file contains utility functions for managing the creation and
>+// conversion of smart pointers. This file is an extension to the C++
>+// standard <memory> library header file.
>+
>+#ifndef ABSL_MEMORY_MEMORY_H_
>+#define ABSL_MEMORY_MEMORY_H_
>+
>+#include <cstddef>
>+#include <limits>
>+#include <memory>
>+#include <new>
>+#include <type_traits>
>+#include <utility>
>+
>+#include "absl/base/macros.h"
>+#include "absl/meta/type_traits.h"
>+
>+namespace absl {
>+
>+// -----------------------------------------------------------------------------
>+// Function Template: WrapUnique()
>+// -----------------------------------------------------------------------------
>+//
>+//  Adopts ownership from a raw pointer and transfers it to the returned
>+//  `std::unique_ptr`, whose type is deduced. DO NOT specify the template type T
>+//  when calling WrapUnique.
>+//
>+// Example:
>+//   X* NewX(int, int);
>+//   auto x = WrapUnique(NewX(1, 2));  // 'x' is std::unique_ptr<X>.
>+//
>+// `absl::WrapUnique` is useful for capturing the output of a raw pointer
>+// factory. However, prefer 'absl::make_unique<T>(args...) over
>+// 'absl::WrapUnique(new T(args...))'.
>+//
>+//   auto x = WrapUnique(new X(1, 2));  // works, but nonideal.
>+//   auto x = make_unique<X>(1, 2);     // safer, standard, avoids raw 'new'.
>+//
>+// Note that `absl::WrapUnique(p)` is valid only if `delete p` is a valid
>+// expression. In particular, `absl::WrapUnique()` cannot wrap pointers to
>+// arrays, functions or void, and it must not be used to capture pointers
>+// obtained from array-new expressions (even though that would compile!).
>+template <typename T>
>+std::unique_ptr<T> WrapUnique(T* ptr) {
>+  static_assert(!std::is_array<T>::value, "array types are unsupported");
>+  static_assert(std::is_object<T>::value, "non-object types are unsupported");
>+  return std::unique_ptr<T>(ptr);
>+}
>+
>+namespace memory_internal {
>+
>+// Traits to select proper overload and return type for `absl::make_unique<>`.
>+template <typename T>
>+struct MakeUniqueResult {
>+  using scalar = std::unique_ptr<T>;
>+};
>+template <typename T>
>+struct MakeUniqueResult<T[]> {
>+  using array = std::unique_ptr<T[]>;
>+};
>+template <typename T, size_t N>
>+struct MakeUniqueResult<T[N]> {
>+  using invalid = void;
>+};
>+
>+}  // namespace memory_internal
>+
>+// gcc 4.8 has __cplusplus at 201301 but doesn't define make_unique.  Other
>+// supported compilers either just define __cplusplus as 201103 but have
>+// make_unique (msvc), or have make_unique whenever __cplusplus > 201103 (clang)
>+#if (__cplusplus > 201103L || defined(_MSC_VER)) && \
>+    !(defined(__GNUC__) && __GNUC__ == 4 && __GNUC_MINOR__ == 8)
>+using std::make_unique;
>+#else
>+// -----------------------------------------------------------------------------
>+// Function Template: make_unique<T>()
>+// -----------------------------------------------------------------------------
>+//
>+// Creates a `std::unique_ptr<>`, while avoiding issues creating temporaries
>+// during the construction process. `absl::make_unique<>` also avoids redundant
>+// type declarations, by avoiding the need to explicitly use the `new` operator.
>+//
>+// This implementation of `absl::make_unique<>` is designed for C++11 code and
>+// will be replaced in C++14 by the equivalent `std::make_unique<>` abstraction.
>+// `absl::make_unique<>` is designed to be 100% compatible with
>+// `std::make_unique<>` so that the eventual migration will involve a simple
>+// rename operation.
>+//
>+// For more background on why `std::unique_ptr<T>(new T(a,b))` is problematic,
>+// see Herb Sutter's explanation on
>+// (Exception-Safe Function Calls)[http://herbsutter.com/gotw/_102/].
>+// (In general, reviewers should treat `new T(a,b)` with scrutiny.)
>+//
>+// Example usage:
>+//
>+//    auto p = make_unique<X>(args...);  // 'p'  is a std::unique_ptr<X>
>+//    auto pa = make_unique<X[]>(5);     // 'pa' is a std::unique_ptr<X[]>
>+//
>+// Three overloads of `absl::make_unique` are required:
>+//
>+//   - For non-array T:
>+//
>+//       Allocates a T with `new T(std::forward<Args> args...)`,
>+//       forwarding all `args` to T's constructor.
>+//       Returns a `std::unique_ptr<T>` owning that object.
>+//
>+//   - For an array of unknown bounds T[]:
>+//
>+//       `absl::make_unique<>` will allocate an array T of type U[] with
>+//       `new U[n]()` and return a `std::unique_ptr<U[]>` owning that array.
>+//
>+//       Note that 'U[n]()' is different from 'U[n]', and elements will be
>+//       value-initialized. Note as well that `std::unique_ptr` will perform its
>+//       own destruction of the array elements upon leaving scope, even though
>+//       the array [] does not have a default destructor.
>+//
>+//       NOTE: an array of unknown bounds T[] may still be (and often will be)
>+//       initialized to have a size, and will still use this overload. E.g:
>+//
>+//         auto my_array = absl::make_unique<int[]>(10);
>+//
>+//   - For an array of known bounds T[N]:
>+//
>+//       `absl::make_unique<>` is deleted (like with `std::make_unique<>`) as
>+//       this overload is not useful.
>+//
>+//       NOTE: an array of known bounds T[N] is not considered a useful
>+//       construction, and may cause undefined behavior in templates. E.g:
>+//
>+//         auto my_array = absl::make_unique<int[10]>();
>+//
>+//       In those cases, of course, you can still use the overload above and
>+//       simply initialize it to its desired size:
>+//
>+//         auto my_array = absl::make_unique<int[]>(10);
>+
>+// `absl::make_unique` overload for non-array types.
>+template <typename T, typename... Args>
>+typename memory_internal::MakeUniqueResult<T>::scalar make_unique(
>+    Args&&... args) {
>+  return std::unique_ptr<T>(new T(std::forward<Args>(args)...));
>+}
>+
>+// `absl::make_unique` overload for an array T[] of unknown bounds.
>+// The array allocation needs to use the `new T[size]` form and cannot take
>+// element constructor arguments. The `std::unique_ptr` will manage destructing
>+// these array elements.
>+template <typename T>
>+typename memory_internal::MakeUniqueResult<T>::array make_unique(size_t n) {
>+  return std::unique_ptr<T>(new typename absl::remove_extent_t<T>[n]());
>+}
>+
>+// `absl::make_unique` overload for an array T[N] of known bounds.
>+// This construction will be rejected.
>+template <typename T, typename... Args>
>+typename memory_internal::MakeUniqueResult<T>::invalid make_unique(
>+    Args&&... /* args */) = delete;
>+#endif
>+
>+// -----------------------------------------------------------------------------
>+// Function Template: RawPtr()
>+// -----------------------------------------------------------------------------
>+//
>+// Extracts the raw pointer from a pointer-like value `ptr`. `absl::RawPtr` is
>+// useful within templates that need to handle a complement of raw pointers,
>+// `std::nullptr_t`, and smart pointers.
>+template <typename T>
>+auto RawPtr(T&& ptr) -> decltype(std::addressof(*ptr)) {
>+  // ptr is a forwarding reference to support Ts with non-const operators.
>+  return (ptr != nullptr) ? std::addressof(*ptr) : nullptr;
>+}
>+inline std::nullptr_t RawPtr(std::nullptr_t) { return nullptr; }
>+
>+// -----------------------------------------------------------------------------
>+// Function Template: ShareUniquePtr()
>+// -----------------------------------------------------------------------------
>+//
>+// Adopts a `std::unique_ptr` rvalue and returns a `std::shared_ptr` of deduced
>+// type. Ownership (if any) of the held value is transferred to the returned
>+// shared pointer.
>+//
>+// Example:
>+//
>+//     auto up = absl::make_unique<int>(10);
>+//     auto sp = absl::ShareUniquePtr(std::move(up));  // shared_ptr<int>
>+//     CHECK_EQ(*sp, 10);
>+//     CHECK(up == nullptr);
>+//
>+// Note that this conversion is correct even when T is an array type, and more
>+// generally it works for *any* deleter of the `unique_ptr` (single-object
>+// deleter, array deleter, or any custom deleter), since the deleter is adopted
>+// by the shared pointer as well. The deleter is copied (unless it is a
>+// reference).
>+//
>+// Implements the resolution of [LWG 2415](http://wg21.link/lwg2415), by which a
>+// null shared pointer does not attempt to call the deleter.
>+template <typename T, typename D>
>+std::shared_ptr<T> ShareUniquePtr(std::unique_ptr<T, D>&& ptr) {
>+  return ptr ? std::shared_ptr<T>(std::move(ptr)) : std::shared_ptr<T>();
>+}
>+
>+// -----------------------------------------------------------------------------
>+// Function Template: WeakenPtr()
>+// -----------------------------------------------------------------------------
>+//
>+// Creates a weak pointer associated with a given shared pointer. The returned
>+// value is a `std::weak_ptr` of deduced type.
>+//
>+// Example:
>+//
>+//    auto sp = std::make_shared<int>(10);
>+//    auto wp = absl::WeakenPtr(sp);
>+//    CHECK_EQ(sp.get(), wp.lock().get());
>+//    sp.reset();
>+//    CHECK(wp.lock() == nullptr);
>+//
>+template <typename T>
>+std::weak_ptr<T> WeakenPtr(const std::shared_ptr<T>& ptr) {
>+  return std::weak_ptr<T>(ptr);
>+}
>+
>+namespace memory_internal {
>+
>+// ExtractOr<E, O, D>::type evaluates to E<O> if possible. Otherwise, D.
>+template <template <typename> class Extract, typename Obj, typename Default,
>+          typename>
>+struct ExtractOr {
>+  using type = Default;
>+};
>+
>+template <template <typename> class Extract, typename Obj, typename Default>
>+struct ExtractOr<Extract, Obj, Default, void_t<Extract<Obj>>> {
>+  using type = Extract<Obj>;
>+};
>+
>+template <template <typename> class Extract, typename Obj, typename Default>
>+using ExtractOrT = typename ExtractOr<Extract, Obj, Default, void>::type;
>+
>+// Extractors for the features of allocators.
>+template <typename T>
>+using GetPointer = typename T::pointer;
>+
>+template <typename T>
>+using GetConstPointer = typename T::const_pointer;
>+
>+template <typename T>
>+using GetVoidPointer = typename T::void_pointer;
>+
>+template <typename T>
>+using GetConstVoidPointer = typename T::const_void_pointer;
>+
>+template <typename T>
>+using GetDifferenceType = typename T::difference_type;
>+
>+template <typename T>
>+using GetSizeType = typename T::size_type;
>+
>+template <typename T>
>+using GetPropagateOnContainerCopyAssignment =
>+    typename T::propagate_on_container_copy_assignment;
>+
>+template <typename T>
>+using GetPropagateOnContainerMoveAssignment =
>+    typename T::propagate_on_container_move_assignment;
>+
>+template <typename T>
>+using GetPropagateOnContainerSwap = typename T::propagate_on_container_swap;
>+
>+template <typename T>
>+using GetIsAlwaysEqual = typename T::is_always_equal;
>+
>+template <typename T>
>+struct GetFirstArg;
>+
>+template <template <typename...> class Class, typename T, typename... Args>
>+struct GetFirstArg<Class<T, Args...>> {
>+  using type = T;
>+};
>+
>+template <typename Ptr, typename = void>
>+struct ElementType {
>+  using type = typename GetFirstArg<Ptr>::type;
>+};
>+
>+template <typename T>
>+struct ElementType<T, void_t<typename T::element_type>> {
>+  using type = typename T::element_type;
>+};
>+
>+template <typename T, typename U>
>+struct RebindFirstArg;
>+
>+template <template <typename...> class Class, typename T, typename... Args,
>+          typename U>
>+struct RebindFirstArg<Class<T, Args...>, U> {
>+  using type = Class<U, Args...>;
>+};
>+
>+template <typename T, typename U, typename = void>
>+struct RebindPtr {
>+  using type = typename RebindFirstArg<T, U>::type;
>+};
>+
>+template <typename T, typename U>
>+struct RebindPtr<T, U, void_t<typename T::template rebind<U>>> {
>+  using type = typename T::template rebind<U>;
>+};
>+
>+template <typename T, typename U>
>+constexpr bool HasRebindAlloc(...) {
>+  return false;
>+}
>+
>+template <typename T, typename U>
>+constexpr bool HasRebindAlloc(typename T::template rebind<U>::other*) {
>+  return true;
>+}
>+
>+template <typename T, typename U, bool = HasRebindAlloc<T, U>(nullptr)>
>+struct RebindAlloc {
>+  using type = typename RebindFirstArg<T, U>::type;
>+};
>+
>+template <typename T, typename U>
>+struct RebindAlloc<T, U, true> {
>+  using type = typename T::template rebind<U>::other;
>+};
>+
>+}  // namespace memory_internal
>+
>+// -----------------------------------------------------------------------------
>+// Class Template: pointer_traits
>+// -----------------------------------------------------------------------------
>+//
>+// An implementation of C++11's std::pointer_traits.
>+//
>+// Provided for portability on toolchains that have a working C++11 compiler,
>+// but the standard library is lacking in C++11 support. For example, some
>+// version of the Android NDK.
>+//
>+
>+template <typename Ptr>
>+struct pointer_traits {
>+  using pointer = Ptr;
>+
>+  // element_type:
>+  // Ptr::element_type if present. Otherwise T if Ptr is a template
>+  // instantiation Template<T, Args...>
>+  using element_type = typename memory_internal::ElementType<Ptr>::type;
>+
>+  // difference_type:
>+  // Ptr::difference_type if present, otherwise std::ptrdiff_t
>+  using difference_type =
>+      memory_internal::ExtractOrT<memory_internal::GetDifferenceType, Ptr,
>+                                  std::ptrdiff_t>;
>+
>+  // rebind:
>+  // Ptr::rebind<U> if exists, otherwise Template<U, Args...> if Ptr is a
>+  // template instantiation Template<T, Args...>
>+  template <typename U>
>+  using rebind = typename memory_internal::RebindPtr<Ptr, U>::type;
>+
>+  // pointer_to:
>+  // Calls Ptr::pointer_to(r)
>+  static pointer pointer_to(element_type& r) {  // NOLINT(runtime/references)
>+    return Ptr::pointer_to(r);
>+  }
>+};
>+
>+// Specialization for T*.
>+template <typename T>
>+struct pointer_traits<T*> {
>+  using pointer = T*;
>+  using element_type = T;
>+  using difference_type = std::ptrdiff_t;
>+
>+  template <typename U>
>+  using rebind = U*;
>+
>+  // pointer_to:
>+  // Calls std::addressof(r)
>+  static pointer pointer_to(
>+      element_type& r) noexcept {  // NOLINT(runtime/references)
>+    return std::addressof(r);
>+  }
>+};
>+
>+// -----------------------------------------------------------------------------
>+// Class Template: allocator_traits
>+// -----------------------------------------------------------------------------
>+//
>+// A C++11 compatible implementation of C++17's std::allocator_traits.
>+//
>+template <typename Alloc>
>+struct allocator_traits {
>+  using allocator_type = Alloc;
>+
>+  // value_type:
>+  // Alloc::value_type
>+  using value_type = typename Alloc::value_type;
>+
>+  // pointer:
>+  // Alloc::pointer if present, otherwise value_type*
>+  using pointer = memory_internal::ExtractOrT<memory_internal::GetPointer,
>+                                              Alloc, value_type*>;
>+
>+  // const_pointer:
>+  // Alloc::const_pointer if present, otherwise
>+  // absl::pointer_traits<pointer>::rebind<const value_type>
>+  using const_pointer =
>+      memory_internal::ExtractOrT<memory_internal::GetConstPointer, Alloc,
>+                                  typename absl::pointer_traits<pointer>::
>+                                      template rebind<const value_type>>;
>+
>+  // void_pointer:
>+  // Alloc::void_pointer if present, otherwise
>+  // absl::pointer_traits<pointer>::rebind<void>
>+  using void_pointer = memory_internal::ExtractOrT<
>+      memory_internal::GetVoidPointer, Alloc,
>+      typename absl::pointer_traits<pointer>::template rebind<void>>;
>+
>+  // const_void_pointer:
>+  // Alloc::const_void_pointer if present, otherwise
>+  // absl::pointer_traits<pointer>::rebind<const void>
>+  using const_void_pointer = memory_internal::ExtractOrT<
>+      memory_internal::GetConstVoidPointer, Alloc,
>+      typename absl::pointer_traits<pointer>::template rebind<const void>>;
>+
>+  // difference_type:
>+  // Alloc::difference_type if present, otherwise
>+  // absl::pointer_traits<pointer>::difference_type
>+  using difference_type = memory_internal::ExtractOrT<
>+      memory_internal::GetDifferenceType, Alloc,
>+      typename absl::pointer_traits<pointer>::difference_type>;
>+
>+  // size_type:
>+  // Alloc::size_type if present, otherwise
>+  // std::make_unsigned<difference_type>::type
>+  using size_type = memory_internal::ExtractOrT<
>+      memory_internal::GetSizeType, Alloc,
>+      typename std::make_unsigned<difference_type>::type>;
>+
>+  // propagate_on_container_copy_assignment:
>+  // Alloc::propagate_on_container_copy_assignment if present, otherwise
>+  // std::false_type
>+  using propagate_on_container_copy_assignment = memory_internal::ExtractOrT<
>+      memory_internal::GetPropagateOnContainerCopyAssignment, Alloc,
>+      std::false_type>;
>+
>+  // propagate_on_container_move_assignment:
>+  // Alloc::propagate_on_container_move_assignment if present, otherwise
>+  // std::false_type
>+  using propagate_on_container_move_assignment = memory_internal::ExtractOrT<
>+      memory_internal::GetPropagateOnContainerMoveAssignment, Alloc,
>+      std::false_type>;
>+
>+  // propagate_on_container_swap:
>+  // Alloc::propagate_on_container_swap if present, otherwise std::false_type
>+  using propagate_on_container_swap =
>+      memory_internal::ExtractOrT<memory_internal::GetPropagateOnContainerSwap,
>+                                  Alloc, std::false_type>;
>+
>+  // is_always_equal:
>+  // Alloc::is_always_equal if present, otherwise std::is_empty<Alloc>::type
>+  using is_always_equal =
>+      memory_internal::ExtractOrT<memory_internal::GetIsAlwaysEqual, Alloc,
>+                                  typename std::is_empty<Alloc>::type>;
>+
>+  // rebind_alloc:
>+  // Alloc::rebind<T>::other if present, otherwise Alloc<T, Args> if this Alloc
>+  // is Alloc<U, Args>
>+  template <typename T>
>+  using rebind_alloc = typename memory_internal::RebindAlloc<Alloc, T>::type;
>+
>+  // rebind_traits:
>+  // absl::allocator_traits<rebind_alloc<T>>
>+  template <typename T>
>+  using rebind_traits = absl::allocator_traits<rebind_alloc<T>>;
>+
>+  // allocate(Alloc& a, size_type n):
>+  // Calls a.allocate(n)
>+  static pointer allocate(Alloc& a,  // NOLINT(runtime/references)
>+                          size_type n) {
>+    return a.allocate(n);
>+  }
>+
>+  // allocate(Alloc& a, size_type n, const_void_pointer hint):
>+  // Calls a.allocate(n, hint) if possible.
>+  // If not possible, calls a.allocate(n)
>+  static pointer allocate(Alloc& a, size_type n,  // NOLINT(runtime/references)
>+                          const_void_pointer hint) {
>+    return allocate_impl(0, a, n, hint);
>+  }
>+
>+  // deallocate(Alloc& a, pointer p, size_type n):
>+  // Calls a.deallocate(p, n)
>+  static void deallocate(Alloc& a, pointer p,  // NOLINT(runtime/references)
>+                         size_type n) {
>+    a.deallocate(p, n);
>+  }
>+
>+  // construct(Alloc& a, T* p, Args&&... args):
>+  // Calls a.construct(p, std::forward<Args>(args)...) if possible.
>+  // If not possible, calls
>+  //   ::new (static_cast<void*>(p)) T(std::forward<Args>(args)...)
>+  template <typename T, typename... Args>
>+  static void construct(Alloc& a, T* p,  // NOLINT(runtime/references)
>+                        Args&&... args) {
>+    construct_impl(0, a, p, std::forward<Args>(args)...);
>+  }
>+
>+  // destroy(Alloc& a, T* p):
>+  // Calls a.destroy(p) if possible. If not possible, calls p->~T().
>+  template <typename T>
>+  static void destroy(Alloc& a, T* p) {  // NOLINT(runtime/references)
>+    destroy_impl(0, a, p);
>+  }
>+
>+  // max_size(const Alloc& a):
>+  // Returns a.max_size() if possible. If not possible, returns
>+  //   std::numeric_limits<size_type>::max() / sizeof(value_type)
>+  static size_type max_size(const Alloc& a) { return max_size_impl(0, a); }
>+
>+  // select_on_container_copy_construction(const Alloc& a):
>+  // Returns a.select_on_container_copy_construction() if possible.
>+  // If not possible, returns a.
>+  static Alloc select_on_container_copy_construction(const Alloc& a) {
>+    return select_on_container_copy_construction_impl(0, a);
>+  }
>+
>+ private:
>+  template <typename A>
>+  static auto allocate_impl(int, A& a,  // NOLINT(runtime/references)
>+                            size_type n, const_void_pointer hint)
>+      -> decltype(a.allocate(n, hint)) {
>+    return a.allocate(n, hint);
>+  }
>+  static pointer allocate_impl(char, Alloc& a,  // NOLINT(runtime/references)
>+                               size_type n, const_void_pointer) {
>+    return a.allocate(n);
>+  }
>+
>+  template <typename A, typename... Args>
>+  static auto construct_impl(int, A& a,  // NOLINT(runtime/references)
>+                             Args&&... args)
>+      -> decltype(a.construct(std::forward<Args>(args)...)) {
>+    a.construct(std::forward<Args>(args)...);
>+  }
>+
>+  template <typename T, typename... Args>
>+  static void construct_impl(char, Alloc&, T* p, Args&&... args) {
>+    ::new (static_cast<void*>(p)) T(std::forward<Args>(args)...);
>+  }
>+
>+  template <typename A, typename T>
>+  static auto destroy_impl(int, A& a,  // NOLINT(runtime/references)
>+                           T* p) -> decltype(a.destroy(p)) {
>+    a.destroy(p);
>+  }
>+  template <typename T>
>+  static void destroy_impl(char, Alloc&, T* p) {
>+    p->~T();
>+  }
>+
>+  template <typename A>
>+  static auto max_size_impl(int, const A& a) -> decltype(a.max_size()) {
>+    return a.max_size();
>+  }
>+  static size_type max_size_impl(char, const Alloc&) {
>+    return std::numeric_limits<size_type>::max() / sizeof(value_type);
>+  }
>+
>+  template <typename A>
>+  static auto select_on_container_copy_construction_impl(int, const A& a)
>+      -> decltype(a.select_on_container_copy_construction()) {
>+    return a.select_on_container_copy_construction();
>+  }
>+  static Alloc select_on_container_copy_construction_impl(char,
>+                                                          const Alloc& a) {
>+    return a;
>+  }
>+};
>+
>+namespace memory_internal {
>+
>+// This template alias transforms Alloc::is_nothrow into a metafunction with
>+// Alloc as a parameter so it can be used with ExtractOrT<>.
>+template <typename Alloc>
>+using GetIsNothrow = typename Alloc::is_nothrow;
>+
>+}  // namespace memory_internal
>+
>+// ABSL_ALLOCATOR_NOTHROW is a build time configuration macro for user to
>+// specify whether the default allocation function can throw or never throws.
>+// If the allocation function never throws, user should define it to a non-zero
>+// value (e.g. via `-DABSL_ALLOCATOR_NOTHROW`).
>+// If the allocation function can throw, user should leave it undefined or
>+// define it to zero.
>+//
>+// allocator_is_nothrow<Alloc> is a traits class that derives from
>+// Alloc::is_nothrow if present, otherwise std::false_type. It's specialized
>+// for Alloc = std::allocator<T> for any type T according to the state of
>+// ABSL_ALLOCATOR_NOTHROW.
>+//
>+// default_allocator_is_nothrow is a class that derives from std::true_type
>+// when the default allocator (global operator new) never throws, and
>+// std::false_type when it can throw. It is a convenience shorthand for writing
>+// allocator_is_nothrow<std::allocator<T>> (T can be any type).
>+// NOTE: allocator_is_nothrow<std::allocator<T>> is guaranteed to derive from
>+// the same type for all T, because users should specialize neither
>+// allocator_is_nothrow nor std::allocator.
>+template <typename Alloc>
>+struct allocator_is_nothrow
>+    : memory_internal::ExtractOrT<memory_internal::GetIsNothrow, Alloc,
>+                                  std::false_type> {};
>+
>+// FIXME: Define ABSL_ALLOCATOR_NOTHROW
>+#if 1
>+template <typename T>
>+struct allocator_is_nothrow<std::allocator<T>> : std::true_type {};
>+struct default_allocator_is_nothrow : std::true_type {};
>+#else
>+struct default_allocator_is_nothrow : std::false_type {};
>+#endif
>+
>+namespace memory_internal {
>+#ifdef ABSL_HAVE_EXCEPTIONS
>+template <typename Allocator, typename StorageElement, typename... Args>
>+void ConstructStorage(Allocator* alloc, StorageElement* first,
>+                      StorageElement* last, const Args&... args) {
>+  for (StorageElement* cur = first; cur != last; ++cur) {
>+    try {
>+      std::allocator_traits<Allocator>::construct(*alloc, cur, args...);
>+    } catch (...) {
>+      while (cur != first) {
>+        --cur;
>+        std::allocator_traits<Allocator>::destroy(*alloc, cur);
>+      }
>+      throw;
>+    }
>+  }
>+}
>+template <typename Allocator, typename StorageElement, typename Iterator>
>+void CopyToStorageFromRange(Allocator* alloc, StorageElement* destination,
>+                            Iterator first, Iterator last) {
>+  for (StorageElement* cur = destination; first != last;
>+       static_cast<void>(++cur), static_cast<void>(++first)) {
>+    try {
>+      std::allocator_traits<Allocator>::construct(*alloc, cur, *first);
>+    } catch (...) {
>+      while (cur != destination) {
>+        --cur;
>+        std::allocator_traits<Allocator>::destroy(*alloc, cur);
>+      }
>+      throw;
>+    }
>+  }
>+}
>+#else   // ABSL_HAVE_EXCEPTIONS
>+template <typename Allocator, typename StorageElement, typename... Args>
>+void ConstructStorage(Allocator* alloc, StorageElement* first,
>+                      StorageElement* last, const Args&... args) {
>+  for (; first != last; ++first) {
>+    std::allocator_traits<Allocator>::construct(*alloc, first, args...);
>+  }
>+}
>+template <typename Allocator, typename StorageElement, typename Iterator>
>+void CopyToStorageFromRange(Allocator* alloc, StorageElement* destination,
>+                            Iterator first, Iterator last) {
>+  for (; first != last;
>+       static_cast<void>(++destination), static_cast<void>(++first)) {
>+    std::allocator_traits<Allocator>::construct(*alloc, destination, *first);
>+  }
>+}
>+#endif  // ABSL_HAVE_EXCEPTIONS
>+}  // namespace memory_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_MEMORY_MEMORY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/memory/memory_exception_safety_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/memory/memory_exception_safety_test.cc
>new file mode 100644
>index 00000000000..d1f6e84f10f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/memory/memory_exception_safety_test.cc
>@@ -0,0 +1,52 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/memory/memory.h"
>+
>+#include "gtest/gtest.h"
>+#include "absl/base/internal/exception_safety_testing.h"
>+
>+namespace absl {
>+namespace {
>+
>+constexpr int kLength = 50;
>+using Thrower = testing::ThrowingValue<testing::TypeSpec::kEverythingThrows>;
>+using ThrowerStorage =
>+    absl::aligned_storage_t<sizeof(Thrower), alignof(Thrower)>;
>+using ThrowerList = std::array<ThrowerStorage, kLength>;
>+
>+TEST(MakeUnique, CheckForLeaks) {
>+  constexpr int kValue = 321;
>+  auto tester = testing::MakeExceptionSafetyTester()
>+                    .WithInitialValue(Thrower(kValue))
>+                    // Ensures make_unique does not modify the input. The real
>+                    // test, though, is ConstructorTracker checking for leaks.
>+                    .WithInvariants(testing::strong_guarantee);
>+
>+  EXPECT_TRUE(tester.Test([](Thrower* thrower) {
>+    static_cast<void>(absl::make_unique<Thrower>(*thrower));
>+  }));
>+
>+  EXPECT_TRUE(tester.Test([](Thrower* thrower) {
>+    static_cast<void>(absl::make_unique<Thrower>(std::move(*thrower)));
>+  }));
>+
>+  // Test T[n] overload
>+  EXPECT_TRUE(tester.Test([&](Thrower*) {
>+    static_cast<void>(absl::make_unique<Thrower[]>(kLength));
>+  }));
>+}
>+
>+}  // namespace
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/memory/memory_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/memory/memory_test.cc
>new file mode 100644
>index 00000000000..dee9b486a30
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/memory/memory_test.cc
>@@ -0,0 +1,614 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+// Tests for pointer utilities.
>+
>+#include "absl/memory/memory.h"
>+
>+#include <sys/types.h>
>+#include <cstddef>
>+#include <memory>
>+#include <string>
>+#include <type_traits>
>+#include <utility>
>+#include <vector>
>+
>+#include "gmock/gmock.h"
>+#include "gtest/gtest.h"
>+
>+namespace {
>+
>+using ::testing::ElementsAre;
>+using ::testing::Return;
>+
>+// This class creates observable behavior to verify that a destructor has
>+// been called, via the instance_count variable.
>+class DestructorVerifier {
>+ public:
>+  DestructorVerifier() { ++instance_count_;  }
>+  DestructorVerifier(const DestructorVerifier&) = delete;
>+  DestructorVerifier& operator=(const DestructorVerifier&) = delete;
>+  ~DestructorVerifier() {  --instance_count_; }
>+
>+  // The number of instances of this class currently active.
>+  static int instance_count() { return instance_count_; }
>+
>+ private:
>+  // The number of instances of this class currently active.
>+  static int instance_count_;
>+};
>+
>+int DestructorVerifier::instance_count_ = 0;
>+
>+TEST(WrapUniqueTest, WrapUnique) {
>+  // Test that the unique_ptr is constructed properly by verifying that the
>+  // destructor for its payload gets called at the proper time.
>+  {
>+    auto dv = new DestructorVerifier;
>+    EXPECT_EQ(1, DestructorVerifier::instance_count());
>+    std::unique_ptr<DestructorVerifier> ptr = absl::WrapUnique(dv);
>+    EXPECT_EQ(1, DestructorVerifier::instance_count());
>+  }
>+  EXPECT_EQ(0, DestructorVerifier::instance_count());
>+}
>+TEST(MakeUniqueTest, Basic) {
>+  std::unique_ptr<std::string> p = absl::make_unique<std::string>();
>+  EXPECT_EQ("", *p);
>+  p = absl::make_unique<std::string>("hi");
>+  EXPECT_EQ("hi", *p);
>+}
>+
>+struct MoveOnly {
>+  MoveOnly() = default;
>+  explicit MoveOnly(int i1) : ip1{new int{i1}} {}
>+  MoveOnly(int i1, int i2) : ip1{new int{i1}}, ip2{new int{i2}} {}
>+  std::unique_ptr<int> ip1;
>+  std::unique_ptr<int> ip2;
>+};
>+
>+struct AcceptMoveOnly {
>+  explicit AcceptMoveOnly(MoveOnly m) : m_(std::move(m)) {}
>+  MoveOnly m_;
>+};
>+
>+TEST(MakeUniqueTest, MoveOnlyTypeAndValue) {
>+  using ExpectedType = std::unique_ptr<MoveOnly>;
>+  {
>+    auto p = absl::make_unique<MoveOnly>();
>+    static_assert(std::is_same<decltype(p), ExpectedType>::value,
>+                  "unexpected return type");
>+    EXPECT_TRUE(!p->ip1);
>+    EXPECT_TRUE(!p->ip2);
>+  }
>+  {
>+    auto p = absl::make_unique<MoveOnly>(1);
>+    static_assert(std::is_same<decltype(p), ExpectedType>::value,
>+                  "unexpected return type");
>+    EXPECT_TRUE(p->ip1 && *p->ip1 == 1);
>+    EXPECT_TRUE(!p->ip2);
>+  }
>+  {
>+    auto p = absl::make_unique<MoveOnly>(1, 2);
>+    static_assert(std::is_same<decltype(p), ExpectedType>::value,
>+                  "unexpected return type");
>+    EXPECT_TRUE(p->ip1 && *p->ip1 == 1);
>+    EXPECT_TRUE(p->ip2 && *p->ip2 == 2);
>+  }
>+}
>+
>+TEST(MakeUniqueTest, AcceptMoveOnly) {
>+  auto p = absl::make_unique<AcceptMoveOnly>(MoveOnly());
>+  p = std::unique_ptr<AcceptMoveOnly>(new AcceptMoveOnly(MoveOnly()));
>+}
>+
>+struct ArrayWatch {
>+  void* operator new[](size_t n) {
>+    allocs().push_back(n);
>+    return ::operator new[](n);
>+  }
>+  void operator delete[](void* p) {
>+    return ::operator delete[](p);
>+  }
>+  static std::vector<size_t>& allocs() {
>+    static auto& v = *new std::vector<size_t>;
>+    return v;
>+  }
>+};
>+
>+TEST(Make_UniqueTest, Array) {
>+  // Ensure state is clean before we start so that these tests
>+  // are order-agnostic.
>+  ArrayWatch::allocs().clear();
>+
>+  auto p = absl::make_unique<ArrayWatch[]>(5);
>+  static_assert(std::is_same<decltype(p),
>+                             std::unique_ptr<ArrayWatch[]>>::value,
>+                "unexpected return type");
>+  EXPECT_THAT(ArrayWatch::allocs(), ElementsAre(5 * sizeof(ArrayWatch)));
>+}
>+
>+TEST(Make_UniqueTest, NotAmbiguousWithStdMakeUnique) {
>+  // Ensure that absl::make_unique is not ambiguous with std::make_unique.
>+  // In C++14 mode, the below call to make_unique has both types as candidates.
>+  struct TakesStdType {
>+    explicit TakesStdType(const std::vector<int> &vec) {}
>+  };
>+  using absl::make_unique;
>+  make_unique<TakesStdType>(std::vector<int>());
>+}
>+
>+#if 0
>+// TODO(billydonahue): Make a proper NC test.
>+// These tests shouldn't compile.
>+TEST(MakeUniqueTestNC, AcceptMoveOnlyLvalue) {
>+  auto m = MoveOnly();
>+  auto p = absl::make_unique<AcceptMoveOnly>(m);
>+}
>+TEST(MakeUniqueTestNC, KnownBoundArray) {
>+  auto p = absl::make_unique<ArrayWatch[5]>();
>+}
>+#endif
>+
>+TEST(RawPtrTest, RawPointer) {
>+  int i = 5;
>+  EXPECT_EQ(&i, absl::RawPtr(&i));
>+}
>+
>+TEST(RawPtrTest, SmartPointer) {
>+  int* o = new int(5);
>+  std::unique_ptr<int> p(o);
>+  EXPECT_EQ(o, absl::RawPtr(p));
>+}
>+
>+class IntPointerNonConstDeref {
>+ public:
>+  explicit IntPointerNonConstDeref(int* p) : p_(p) {}
>+  friend bool operator!=(const IntPointerNonConstDeref& a, std::nullptr_t) {
>+    return a.p_ != nullptr;
>+  }
>+  int& operator*() { return *p_; }
>+
>+ private:
>+  std::unique_ptr<int> p_;
>+};
>+
>+TEST(RawPtrTest, SmartPointerNonConstDereference) {
>+  int* o = new int(5);
>+  IntPointerNonConstDeref p(o);
>+  EXPECT_EQ(o, absl::RawPtr(p));
>+}
>+
>+TEST(RawPtrTest, NullValuedRawPointer) {
>+  int* p = nullptr;
>+  EXPECT_EQ(nullptr, absl::RawPtr(p));
>+}
>+
>+TEST(RawPtrTest, NullValuedSmartPointer) {
>+  std::unique_ptr<int> p;
>+  EXPECT_EQ(nullptr, absl::RawPtr(p));
>+}
>+
>+TEST(RawPtrTest, Nullptr) {
>+  auto p = absl::RawPtr(nullptr);
>+  EXPECT_TRUE((std::is_same<std::nullptr_t, decltype(p)>::value));
>+  EXPECT_EQ(nullptr, p);
>+}
>+
>+TEST(RawPtrTest, Null) {
>+  auto p = absl::RawPtr(nullptr);
>+  EXPECT_TRUE((std::is_same<std::nullptr_t, decltype(p)>::value));
>+  EXPECT_EQ(nullptr, p);
>+}
>+
>+TEST(RawPtrTest, Zero) {
>+  auto p = absl::RawPtr(nullptr);
>+  EXPECT_TRUE((std::is_same<std::nullptr_t, decltype(p)>::value));
>+  EXPECT_EQ(nullptr, p);
>+}
>+
>+TEST(ShareUniquePtrTest, Share) {
>+  auto up = absl::make_unique<int>();
>+  int* rp = up.get();
>+  auto sp = absl::ShareUniquePtr(std::move(up));
>+  EXPECT_EQ(sp.get(), rp);
>+}
>+
>+TEST(ShareUniquePtrTest, ShareNull) {
>+  struct NeverDie {
>+    using pointer = void*;
>+    void operator()(pointer) {
>+      ASSERT_TRUE(false) << "Deleter should not have been called.";
>+    }
>+  };
>+
>+  std::unique_ptr<void, NeverDie> up;
>+  auto sp = absl::ShareUniquePtr(std::move(up));
>+}
>+
>+TEST(WeakenPtrTest, Weak) {
>+  auto sp = std::make_shared<int>();
>+  auto wp = absl::WeakenPtr(sp);
>+  EXPECT_EQ(sp.get(), wp.lock().get());
>+  sp.reset();
>+  EXPECT_TRUE(wp.expired());
>+}
>+
>+// Should not compile.
>+/*
>+TEST(RawPtrTest, NotAPointer) {
>+  absl::RawPtr(1.5);
>+}
>+*/
>+
>+template <typename T>
>+struct SmartPointer {
>+  using difference_type = char;
>+};
>+
>+struct PointerWith {
>+  using element_type = int32_t;
>+  using difference_type = int16_t;
>+  template <typename U>
>+  using rebind = SmartPointer<U>;
>+
>+  static PointerWith pointer_to(
>+      element_type& r) {  // NOLINT(runtime/references)
>+    return PointerWith{&r};
>+  }
>+
>+  element_type* ptr;
>+};
>+
>+template <typename... Args>
>+struct PointerWithout {};
>+
>+TEST(PointerTraits, Types) {
>+  using TraitsWith = absl::pointer_traits<PointerWith>;
>+  EXPECT_TRUE((std::is_same<TraitsWith::pointer, PointerWith>::value));
>+  EXPECT_TRUE((std::is_same<TraitsWith::element_type, int32_t>::value));
>+  EXPECT_TRUE((std::is_same<TraitsWith::difference_type, int16_t>::value));
>+  EXPECT_TRUE((
>+      std::is_same<TraitsWith::rebind<int64_t>, SmartPointer<int64_t>>::value));
>+
>+  using TraitsWithout = absl::pointer_traits<PointerWithout<double, int>>;
>+  EXPECT_TRUE((std::is_same<TraitsWithout::pointer,
>+                            PointerWithout<double, int>>::value));
>+  EXPECT_TRUE((std::is_same<TraitsWithout::element_type, double>::value));
>+  EXPECT_TRUE(
>+      (std::is_same<TraitsWithout ::difference_type, std::ptrdiff_t>::value));
>+  EXPECT_TRUE((std::is_same<TraitsWithout::rebind<int64_t>,
>+                            PointerWithout<int64_t, int>>::value));
>+
>+  using TraitsRawPtr = absl::pointer_traits<char*>;
>+  EXPECT_TRUE((std::is_same<TraitsRawPtr::pointer, char*>::value));
>+  EXPECT_TRUE((std::is_same<TraitsRawPtr::element_type, char>::value));
>+  EXPECT_TRUE(
>+      (std::is_same<TraitsRawPtr::difference_type, std::ptrdiff_t>::value));
>+  EXPECT_TRUE((std::is_same<TraitsRawPtr::rebind<int64_t>, int64_t*>::value));
>+}
>+
>+TEST(PointerTraits, Functions) {
>+  int i;
>+  EXPECT_EQ(&i, absl::pointer_traits<PointerWith>::pointer_to(i).ptr);
>+  EXPECT_EQ(&i, absl::pointer_traits<int*>::pointer_to(i));
>+}
>+
>+TEST(AllocatorTraits, Typedefs) {
>+  struct A {
>+    struct value_type {};
>+  };
>+  EXPECT_TRUE((
>+      std::is_same<A,
>+                   typename absl::allocator_traits<A>::allocator_type>::value));
>+  EXPECT_TRUE(
>+      (std::is_same<A::value_type,
>+                    typename absl::allocator_traits<A>::value_type>::value));
>+
>+  struct X {};
>+  struct HasPointer {
>+    using value_type = X;
>+    using pointer = SmartPointer<X>;
>+  };
>+  EXPECT_TRUE((std::is_same<SmartPointer<X>, typename absl::allocator_traits<
>+                                                 HasPointer>::pointer>::value));
>+  EXPECT_TRUE(
>+      (std::is_same<A::value_type*,
>+                    typename absl::allocator_traits<A>::pointer>::value));
>+
>+  EXPECT_TRUE(
>+      (std::is_same<
>+          SmartPointer<const X>,
>+          typename absl::allocator_traits<HasPointer>::const_pointer>::value));
>+  EXPECT_TRUE(
>+      (std::is_same<const A::value_type*,
>+                    typename absl::allocator_traits<A>::const_pointer>::value));
>+
>+  struct HasVoidPointer {
>+    using value_type = X;
>+    struct void_pointer {};
>+  };
>+
>+  EXPECT_TRUE((std::is_same<HasVoidPointer::void_pointer,
>+                            typename absl::allocator_traits<
>+                                HasVoidPointer>::void_pointer>::value));
>+  EXPECT_TRUE(
>+      (std::is_same<SmartPointer<void>, typename absl::allocator_traits<
>+                                            HasPointer>::void_pointer>::value));
>+
>+  struct HasConstVoidPointer {
>+    using value_type = X;
>+    struct const_void_pointer {};
>+  };
>+
>+  EXPECT_TRUE(
>+      (std::is_same<HasConstVoidPointer::const_void_pointer,
>+                    typename absl::allocator_traits<
>+                        HasConstVoidPointer>::const_void_pointer>::value));
>+  EXPECT_TRUE((std::is_same<SmartPointer<const void>,
>+                            typename absl::allocator_traits<
>+                                HasPointer>::const_void_pointer>::value));
>+
>+  struct HasDifferenceType {
>+    using value_type = X;
>+    using difference_type = int;
>+  };
>+  EXPECT_TRUE(
>+      (std::is_same<int, typename absl::allocator_traits<
>+                             HasDifferenceType>::difference_type>::value));
>+  EXPECT_TRUE((std::is_same<char, typename absl::allocator_traits<
>+                                      HasPointer>::difference_type>::value));
>+
>+  struct HasSizeType {
>+    using value_type = X;
>+    using size_type = unsigned int;
>+  };
>+  EXPECT_TRUE((std::is_same<unsigned int, typename absl::allocator_traits<
>+                                              HasSizeType>::size_type>::value));
>+  EXPECT_TRUE((std::is_same<unsigned char, typename absl::allocator_traits<
>+                                               HasPointer>::size_type>::value));
>+
>+  struct HasPropagateOnCopy {
>+    using value_type = X;
>+    struct propagate_on_container_copy_assignment {};
>+  };
>+
>+  EXPECT_TRUE(
>+      (std::is_same<HasPropagateOnCopy::propagate_on_container_copy_assignment,
>+                    typename absl::allocator_traits<HasPropagateOnCopy>::
>+                        propagate_on_container_copy_assignment>::value));
>+  EXPECT_TRUE(
>+      (std::is_same<std::false_type,
>+                    typename absl::allocator_traits<
>+                        A>::propagate_on_container_copy_assignment>::value));
>+
>+  struct HasPropagateOnMove {
>+    using value_type = X;
>+    struct propagate_on_container_move_assignment {};
>+  };
>+
>+  EXPECT_TRUE(
>+      (std::is_same<HasPropagateOnMove::propagate_on_container_move_assignment,
>+                    typename absl::allocator_traits<HasPropagateOnMove>::
>+                        propagate_on_container_move_assignment>::value));
>+  EXPECT_TRUE(
>+      (std::is_same<std::false_type,
>+                    typename absl::allocator_traits<
>+                        A>::propagate_on_container_move_assignment>::value));
>+
>+  struct HasPropagateOnSwap {
>+    using value_type = X;
>+    struct propagate_on_container_swap {};
>+  };
>+
>+  EXPECT_TRUE(
>+      (std::is_same<HasPropagateOnSwap::propagate_on_container_swap,
>+                    typename absl::allocator_traits<HasPropagateOnSwap>::
>+                        propagate_on_container_swap>::value));
>+  EXPECT_TRUE(
>+      (std::is_same<std::false_type, typename absl::allocator_traits<A>::
>+                                         propagate_on_container_swap>::value));
>+
>+  struct HasIsAlwaysEqual {
>+    using value_type = X;
>+    struct is_always_equal {};
>+  };
>+
>+  EXPECT_TRUE((std::is_same<HasIsAlwaysEqual::is_always_equal,
>+                            typename absl::allocator_traits<
>+                                HasIsAlwaysEqual>::is_always_equal>::value));
>+  EXPECT_TRUE((std::is_same<std::true_type, typename absl::allocator_traits<
>+                                                A>::is_always_equal>::value));
>+  struct NonEmpty {
>+    using value_type = X;
>+    int i;
>+  };
>+  EXPECT_TRUE(
>+      (std::is_same<std::false_type,
>+                    absl::allocator_traits<NonEmpty>::is_always_equal>::value));
>+}
>+
>+template <typename T>
>+struct AllocWithPrivateInheritance : private std::allocator<T> {
>+  using value_type = T;
>+};
>+
>+TEST(AllocatorTraits, RebindWithPrivateInheritance) {
>+  // Regression test for some versions of gcc that do not like the sfinae we
>+  // used in combination with private inheritance.
>+  EXPECT_TRUE(
>+      (std::is_same<AllocWithPrivateInheritance<int>,
>+                    absl::allocator_traits<AllocWithPrivateInheritance<char>>::
>+                        rebind_alloc<int>>::value));
>+}
>+
>+template <typename T>
>+struct Rebound {};
>+
>+struct AllocWithRebind {
>+  using value_type = int;
>+  template <typename T>
>+  struct rebind {
>+    using other = Rebound<T>;
>+  };
>+};
>+
>+template <typename T, typename U>
>+struct AllocWithoutRebind {
>+  using value_type = int;
>+};
>+
>+TEST(AllocatorTraits, Rebind) {
>+  EXPECT_TRUE(
>+      (std::is_same<Rebound<int>,
>+                    typename absl::allocator_traits<
>+                        AllocWithRebind>::template rebind_alloc<int>>::value));
>+  EXPECT_TRUE(
>+      (std::is_same<absl::allocator_traits<Rebound<int>>,
>+                    typename absl::allocator_traits<
>+                        AllocWithRebind>::template rebind_traits<int>>::value));
>+
>+  EXPECT_TRUE(
>+      (std::is_same<AllocWithoutRebind<double, char>,
>+                    typename absl::allocator_traits<AllocWithoutRebind<
>+                        int, char>>::template rebind_alloc<double>>::value));
>+  EXPECT_TRUE(
>+      (std::is_same<absl::allocator_traits<AllocWithoutRebind<double, char>>,
>+                    typename absl::allocator_traits<AllocWithoutRebind<
>+                        int, char>>::template rebind_traits<double>>::value));
>+}
>+
>+struct TestValue {
>+  TestValue() {}
>+  explicit TestValue(int* trace) : trace(trace) { ++*trace; }
>+  ~TestValue() {
>+    if (trace) --*trace;
>+  }
>+  int* trace = nullptr;
>+};
>+
>+struct MinimalMockAllocator {
>+  MinimalMockAllocator() : value(0) {}
>+  explicit MinimalMockAllocator(int value) : value(value) {}
>+  MinimalMockAllocator(const MinimalMockAllocator& other)
>+      : value(other.value) {}
>+  using value_type = TestValue;
>+  MOCK_METHOD1(allocate, value_type*(size_t));
>+  MOCK_METHOD2(deallocate, void(value_type*, size_t));
>+
>+  int value;
>+};
>+
>+TEST(AllocatorTraits, FunctionsMinimal) {
>+  int trace = 0;
>+  int hint;
>+  TestValue x(&trace);
>+  MinimalMockAllocator mock;
>+  using Traits = absl::allocator_traits<MinimalMockAllocator>;
>+  EXPECT_CALL(mock, allocate(7)).WillRepeatedly(Return(&x));
>+  EXPECT_CALL(mock, deallocate(&x, 7));
>+
>+  EXPECT_EQ(&x, Traits::allocate(mock, 7));
>+  Traits::allocate(mock, 7, static_cast<const void*>(&hint));
>+  EXPECT_EQ(&x, Traits::allocate(mock, 7, static_cast<const void*>(&hint)));
>+  Traits::deallocate(mock, &x, 7);
>+
>+  EXPECT_EQ(1, trace);
>+  Traits::construct(mock, &x, &trace);
>+  EXPECT_EQ(2, trace);
>+  Traits::destroy(mock, &x);
>+  EXPECT_EQ(1, trace);
>+
>+  EXPECT_EQ(std::numeric_limits<size_t>::max() / sizeof(TestValue),
>+            Traits::max_size(mock));
>+
>+  EXPECT_EQ(0, mock.value);
>+  EXPECT_EQ(0, Traits::select_on_container_copy_construction(mock).value);
>+}
>+
>+struct FullMockAllocator {
>+  FullMockAllocator() : value(0) {}
>+  explicit FullMockAllocator(int value) : value(value) {}
>+  FullMockAllocator(const FullMockAllocator& other) : value(other.value) {}
>+  using value_type = TestValue;
>+  MOCK_METHOD1(allocate, value_type*(size_t));
>+  MOCK_METHOD2(allocate, value_type*(size_t, const void*));
>+  MOCK_METHOD2(construct, void(value_type*, int*));
>+  MOCK_METHOD1(destroy, void(value_type*));
>+  MOCK_CONST_METHOD0(max_size, size_t());
>+  MOCK_CONST_METHOD0(select_on_container_copy_construction,
>+                     FullMockAllocator());
>+
>+  int value;
>+};
>+
>+TEST(AllocatorTraits, FunctionsFull) {
>+  int trace = 0;
>+  int hint;
>+  TestValue x(&trace), y;
>+  FullMockAllocator mock;
>+  using Traits = absl::allocator_traits<FullMockAllocator>;
>+  EXPECT_CALL(mock, allocate(7)).WillRepeatedly(Return(&x));
>+  EXPECT_CALL(mock, allocate(13, &hint)).WillRepeatedly(Return(&y));
>+  EXPECT_CALL(mock, construct(&x, &trace));
>+  EXPECT_CALL(mock, destroy(&x));
>+  EXPECT_CALL(mock, max_size()).WillRepeatedly(Return(17));
>+  EXPECT_CALL(mock, select_on_container_copy_construction())
>+      .WillRepeatedly(Return(FullMockAllocator(23)));
>+
>+  EXPECT_EQ(&x, Traits::allocate(mock, 7));
>+  EXPECT_EQ(&y, Traits::allocate(mock, 13, static_cast<const void*>(&hint)));
>+
>+  EXPECT_EQ(1, trace);
>+  Traits::construct(mock, &x, &trace);
>+  EXPECT_EQ(1, trace);
>+  Traits::destroy(mock, &x);
>+  EXPECT_EQ(1, trace);
>+
>+  EXPECT_EQ(17, Traits::max_size(mock));
>+
>+  EXPECT_EQ(0, mock.value);
>+  EXPECT_EQ(23, Traits::select_on_container_copy_construction(mock).value);
>+}
>+
>+TEST(AllocatorNoThrowTest, DefaultAllocator) {
>+#if ABSL_ALLOCATOR_NOTHROW
>+  EXPECT_TRUE(absl::default_allocator_is_nothrow::value);
>+#else
>+  EXPECT_FALSE(absl::default_allocator_is_nothrow::value);
>+#endif
>+}
>+
>+TEST(AllocatorNoThrowTest, StdAllocator) {
>+#if ABSL_ALLOCATOR_NOTHROW
>+  EXPECT_TRUE(absl::allocator_is_nothrow<std::allocator<int>>::value);
>+#else
>+  EXPECT_FALSE(absl::allocator_is_nothrow<std::allocator<int>>::value);
>+#endif
>+}
>+
>+TEST(AllocatorNoThrowTest, CustomAllocator) {
>+  struct NoThrowAllocator {
>+    using is_nothrow = std::true_type;
>+  };
>+  struct CanThrowAllocator {
>+    using is_nothrow = std::false_type;
>+  };
>+  struct UnspecifiedAllocator {
>+  };
>+  EXPECT_TRUE(absl::allocator_is_nothrow<NoThrowAllocator>::value);
>+  EXPECT_FALSE(absl::allocator_is_nothrow<CanThrowAllocator>::value);
>+  EXPECT_FALSE(absl::allocator_is_nothrow<UnspecifiedAllocator>::value);
>+}
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/meta/BUILD.bazel b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/meta/BUILD.bazel
>new file mode 100644
>index 00000000000..dbc9717ddd3
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/meta/BUILD.bazel
>@@ -0,0 +1,29 @@
>+load(
>+    "//absl:copts.bzl",
>+    "ABSL_DEFAULT_COPTS",
>+    "ABSL_TEST_COPTS",
>+)
>+
>+package(default_visibility = ["//visibility:public"])
>+
>+licenses(["notice"])  # Apache 2.0
>+
>+cc_library(
>+    name = "type_traits",
>+    hdrs = ["type_traits.h"],
>+    copts = ABSL_DEFAULT_COPTS,
>+    deps = [
>+        "//absl/base:config",
>+    ],
>+)
>+
>+cc_test(
>+    name = "type_traits_test",
>+    srcs = ["type_traits_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    deps = [
>+        ":type_traits",
>+        "//absl/base:core_headers",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/meta/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/meta/BUILD.gn
>new file mode 100644
>index 00000000000..6757219c273
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/meta/BUILD.gn
>@@ -0,0 +1,30 @@
>+# Copyright 2018 The Chromium Authors. All rights reserved.
>+# Use of this source code is governed by a BSD-style license that can be
>+# found in the LICENSE file.
>+
>+import("//build_overrides/build.gni")
>+
>+if (build_with_chromium) {
>+  visibility = [
>+    "//third_party/webrtc/*",
>+    "//third_party/abseil-cpp/*",
>+    "//third_party/googletest:gtest",
>+  ]
>+} else {
>+  visibility = [ "*" ]
>+}
>+
>+source_set("type_traits") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  public = [
>+    "type_traits.h",
>+  ]
>+  deps = [
>+    "../base:config",
>+  ]
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/meta/CMakeLists.txt b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/meta/CMakeLists.txt
>new file mode 100644
>index 00000000000..adb0ceb754d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/meta/CMakeLists.txt
>@@ -0,0 +1,52 @@
>+#
>+# Copyright 2017 The Abseil Authors.
>+#
>+# Licensed under the Apache License, Version 2.0 (the "License");
>+# you may not use this file except in compliance with the License.
>+# You may obtain a copy of the License at
>+#
>+#      http://www.apache.org/licenses/LICENSE-2.0
>+#
>+# Unless required by applicable law or agreed to in writing, software
>+# distributed under the License is distributed on an "AS IS" BASIS,
>+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+# See the License for the specific language governing permissions and
>+# limitations under the License.
>+#
>+
>+list(APPEND META_PUBLIC_HEADERS
>+  "type_traits.h"
>+)
>+
>+
>+#
>+## TESTS
>+#
>+
>+# test type_traits_test
>+list(APPEND TYPE_TRAITS_TEST_SRC
>+  "type_traits_test.cc"
>+  ${META_PUBLIC_HEADERS}
>+)
>+
>+absl_header_library(
>+  TARGET
>+    absl_meta
>+  PUBLIC_LIBRARIES
>+    absl::base
>+  EXPORT_NAME
>+    meta
>+ )
>+
>+absl_test(
>+  TARGET
>+    type_traits_test
>+  SOURCES
>+    ${TYPE_TRAITS_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    absl::base
>+    absl::meta
>+)
>+
>+
>+
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/meta/type_traits.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/meta/type_traits.h
>new file mode 100644
>index 00000000000..457b890841a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/meta/type_traits.h
>@@ -0,0 +1,419 @@
>+//
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// -----------------------------------------------------------------------------
>+// type_traits.h
>+// -----------------------------------------------------------------------------
>+//
>+// This file contains C++11-compatible versions of standard <type_traits> API
>+// functions for determining the characteristics of types. Such traits can
>+// support type inference, classification, and transformation, as well as
>+// make it easier to write templates based on generic type behavior.
>+//
>+// See http://en.cppreference.com/w/cpp/header/type_traits
>+//
>+// WARNING: use of many of the constructs in this header will count as "complex
>+// template metaprogramming", so before proceeding, please carefully consider
>+// https://google.github.io/styleguide/cppguide.html#Template_metaprogramming
>+//
>+// WARNING: using template metaprogramming to detect or depend on API
>+// features is brittle and not guaranteed. Neither the standard library nor
>+// Abseil provides any guarantee that APIs are stable in the face of template
>+// metaprogramming. Use with caution.
>+#ifndef ABSL_META_TYPE_TRAITS_H_
>+#define ABSL_META_TYPE_TRAITS_H_
>+
>+#include <stddef.h>
>+#include <functional>
>+#include <type_traits>
>+
>+#include "absl/base/config.h"
>+
>+namespace absl {
>+
>+namespace type_traits_internal {
>+
>+template <typename... Ts>
>+struct VoidTImpl {
>+  using type = void;
>+};
>+
>+// This trick to retrieve a default alignment is necessary for our
>+// implementation of aligned_storage_t to be consistent with any implementation
>+// of std::aligned_storage.
>+template <size_t Len, typename T = std::aligned_storage<Len>>
>+struct default_alignment_of_aligned_storage;
>+
>+template <size_t Len, size_t Align>
>+struct default_alignment_of_aligned_storage<Len,
>+                                            std::aligned_storage<Len, Align>> {
>+  static constexpr size_t value = Align;
>+};
>+
>+////////////////////////////////
>+// Library Fundamentals V2 TS //
>+////////////////////////////////
>+
>+// NOTE: The `is_detected` family of templates here differ from the library
>+// fundamentals specification in that for library fundamentals, `Op<Args...>` is
>+// evaluated as soon as the type `is_detected<Op, Args...>` undergoes
>+// substitution, regardless of whether or not the `::value` is accessed. That
>+// is inconsistent with all other standard traits and prevents lazy evaluation
>+// in larger contexts (such as if the `is_detected` check is a trailing argument
>+// of a `conjunction`. This implementation opts to instead be lazy in the same
>+// way that the standard traits are (this "defect" of the detection idiom
>+// specifications has been reported).
>+
>+template <class Enabler, template <class...> class Op, class... Args>
>+struct is_detected_impl {
>+  using type = std::false_type;
>+};
>+
>+template <template <class...> class Op, class... Args>
>+struct is_detected_impl<typename VoidTImpl<Op<Args...>>::type, Op, Args...> {
>+  using type = std::true_type;
>+};
>+
>+template <template <class...> class Op, class... Args>
>+struct is_detected : is_detected_impl<void, Op, Args...>::type {};
>+
>+template <class Enabler, class To, template <class...> class Op, class... Args>
>+struct is_detected_convertible_impl {
>+  using type = std::false_type;
>+};
>+
>+template <class To, template <class...> class Op, class... Args>
>+struct is_detected_convertible_impl<
>+    typename std::enable_if<std::is_convertible<Op<Args...>, To>::value>::type,
>+    To, Op, Args...> {
>+  using type = std::true_type;
>+};
>+
>+template <class To, template <class...> class Op, class... Args>
>+struct is_detected_convertible
>+    : is_detected_convertible_impl<void, To, Op, Args...>::type {};
>+
>+}  // namespace type_traits_internal
>+
>+// void_t()
>+//
>+// Ignores the type of any its arguments and returns `void`. In general, this
>+// metafunction allows you to create a general case that maps to `void` while
>+// allowing specializations that map to specific types.
>+//
>+// This metafunction is designed to be a drop-in replacement for the C++17
>+// `std::void_t` metafunction.
>+//
>+// NOTE: `absl::void_t` does not use the standard-specified implementation so
>+// that it can remain compatible with gcc < 5.1. This can introduce slightly
>+// different behavior, such as when ordering partial specializations.
>+template <typename... Ts>
>+using void_t = typename type_traits_internal::VoidTImpl<Ts...>::type;
>+
>+// conjunction
>+//
>+// Performs a compile-time logical AND operation on the passed types (which
>+// must have  `::value` members convertible to `bool`. Short-circuits if it
>+// encounters any `false` members (and does not compare the `::value` members
>+// of any remaining arguments).
>+//
>+// This metafunction is designed to be a drop-in replacement for the C++17
>+// `std::conjunction` metafunction.
>+template <typename... Ts>
>+struct conjunction;
>+
>+template <typename T, typename... Ts>
>+struct conjunction<T, Ts...>
>+    : std::conditional<T::value, conjunction<Ts...>, T>::type {};
>+
>+template <typename T>
>+struct conjunction<T> : T {};
>+
>+template <>
>+struct conjunction<> : std::true_type {};
>+
>+// disjunction
>+//
>+// Performs a compile-time logical OR operation on the passed types (which
>+// must have  `::value` members convertible to `bool`. Short-circuits if it
>+// encounters any `true` members (and does not compare the `::value` members
>+// of any remaining arguments).
>+//
>+// This metafunction is designed to be a drop-in replacement for the C++17
>+// `std::disjunction` metafunction.
>+template <typename... Ts>
>+struct disjunction;
>+
>+template <typename T, typename... Ts>
>+struct disjunction<T, Ts...> :
>+      std::conditional<T::value, T, disjunction<Ts...>>::type {};
>+
>+template <typename T>
>+struct disjunction<T> : T {};
>+
>+template <>
>+struct disjunction<> : std::false_type {};
>+
>+// negation
>+//
>+// Performs a compile-time logical NOT operation on the passed type (which
>+// must have  `::value` members convertible to `bool`.
>+//
>+// This metafunction is designed to be a drop-in replacement for the C++17
>+// `std::negation` metafunction.
>+template <typename T>
>+struct negation : std::integral_constant<bool, !T::value> {};
>+
>+// is_trivially_destructible()
>+//
>+// Determines whether the passed type `T` is trivially destructable.
>+//
>+// This metafunction is designed to be a drop-in replacement for the C++11
>+// `std::is_trivially_destructible()` metafunction for platforms that have
>+// incomplete C++11 support (such as libstdc++ 4.x). On any platforms that do
>+// fully support C++11, we check whether this yields the same result as the std
>+// implementation.
>+//
>+// NOTE: the extensions (__has_trivial_xxx) are implemented in gcc (version >=
>+// 4.3) and clang. Since we are supporting libstdc++ > 4.7, they should always
>+// be present. These  extensions are documented at
>+// https://gcc.gnu.org/onlinedocs/gcc/Type-Traits.html#Type-Traits.
>+template <typename T>
>+struct is_trivially_destructible
>+    : std::integral_constant<bool, __has_trivial_destructor(T) &&
>+                                   std::is_destructible<T>::value> {
>+#ifdef ABSL_HAVE_STD_IS_TRIVIALLY_DESTRUCTIBLE
>+ private:
>+  static constexpr bool compliant = std::is_trivially_destructible<T>::value ==
>+                                    is_trivially_destructible::value;
>+  static_assert(compliant || std::is_trivially_destructible<T>::value,
>+                "Not compliant with std::is_trivially_destructible; "
>+                "Standard: false, Implementation: true");
>+  static_assert(compliant || !std::is_trivially_destructible<T>::value,
>+                "Not compliant with std::is_trivially_destructible; "
>+                "Standard: true, Implementation: false");
>+#endif  // ABSL_HAVE_STD_IS_TRIVIALLY_DESTRUCTIBLE
>+};
>+
>+// is_trivially_default_constructible()
>+//
>+// Determines whether the passed type `T` is trivially default constructible.
>+//
>+// This metafunction is designed to be a drop-in replacement for the C++11
>+// `std::is_trivially_default_constructible()` metafunction for platforms that
>+// have incomplete C++11 support (such as libstdc++ 4.x). On any platforms that
>+// do fully support C++11, we check whether this yields the same result as the
>+// std implementation.
>+//
>+// NOTE: according to the C++ standard, Section: 20.15.4.3 [meta.unary.prop]
>+// "The predicate condition for a template specialization is_constructible<T,
>+// Args...> shall be satisfied if and only if the following variable
>+// definition would be well-formed for some invented variable t:
>+//
>+// T t(declval<Args>()...);
>+//
>+// is_trivially_constructible<T, Args...> additionally requires that the
>+// variable definition does not call any operation that is not trivial.
>+// For the purposes of this check, the call to std::declval is considered
>+// trivial."
>+//
>+// Notes from http://en.cppreference.com/w/cpp/types/is_constructible:
>+// In many implementations, is_nothrow_constructible also checks if the
>+// destructor throws because it is effectively noexcept(T(arg)). Same
>+// applies to is_trivially_constructible, which, in these implementations, also
>+// requires that the destructor is trivial.
>+// GCC bug 51452: https://gcc.gnu.org/bugzilla/show_bug.cgi?id=51452
>+// LWG issue 2116: http://cplusplus.github.io/LWG/lwg-active.html#2116.
>+//
>+// "T obj();" need to be well-formed and not call any nontrivial operation.
>+// Nontrivially destructible types will cause the expression to be nontrivial.
>+template <typename T>
>+struct is_trivially_default_constructible
>+    : std::integral_constant<bool, __has_trivial_constructor(T) &&
>+                                   std::is_default_constructible<T>::value &&
>+                                   is_trivially_destructible<T>::value> {
>+#ifdef ABSL_HAVE_STD_IS_TRIVIALLY_CONSTRUCTIBLE
>+ private:
>+  static constexpr bool compliant =
>+      std::is_trivially_default_constructible<T>::value ==
>+      is_trivially_default_constructible::value;
>+  static_assert(compliant || std::is_trivially_default_constructible<T>::value,
>+                "Not compliant with std::is_trivially_default_constructible; "
>+                "Standard: false, Implementation: true");
>+  static_assert(compliant || !std::is_trivially_default_constructible<T>::value,
>+                "Not compliant with std::is_trivially_default_constructible; "
>+                "Standard: true, Implementation: false");
>+#endif  // ABSL_HAVE_STD_IS_TRIVIALLY_CONSTRUCTIBLE
>+};
>+
>+// is_trivially_copy_constructible()
>+//
>+// Determines whether the passed type `T` is trivially copy constructible.
>+//
>+// This metafunction is designed to be a drop-in replacement for the C++11
>+// `std::is_trivially_copy_constructible()` metafunction for platforms that have
>+// incomplete C++11 support (such as libstdc++ 4.x). On any platforms that do
>+// fully support C++11, we check whether this yields the same result as the std
>+// implementation.
>+//
>+// NOTE: `T obj(declval<const T&>());` needs to be well-formed and not call any
>+// nontrivial operation.  Nontrivially destructible types will cause the
>+// expression to be nontrivial.
>+template <typename T>
>+struct is_trivially_copy_constructible
>+    : std::integral_constant<bool, __has_trivial_copy(T) &&
>+                                   std::is_copy_constructible<T>::value &&
>+                                   is_trivially_destructible<T>::value> {
>+#ifdef ABSL_HAVE_STD_IS_TRIVIALLY_CONSTRUCTIBLE
>+ private:
>+  static constexpr bool compliant =
>+      std::is_trivially_copy_constructible<T>::value ==
>+      is_trivially_copy_constructible::value;
>+  static_assert(compliant || std::is_trivially_copy_constructible<T>::value,
>+                "Not compliant with std::is_trivially_copy_constructible; "
>+                "Standard: false, Implementation: true");
>+  static_assert(compliant || !std::is_trivially_copy_constructible<T>::value,
>+                "Not compliant with std::is_trivially_copy_constructible; "
>+                "Standard: true, Implementation: false");
>+#endif  // ABSL_HAVE_STD_IS_TRIVIALLY_CONSTRUCTIBLE
>+};
>+
>+// is_trivially_copy_assignable()
>+//
>+// Determines whether the passed type `T` is trivially copy assignable.
>+//
>+// This metafunction is designed to be a drop-in replacement for the C++11
>+// `std::is_trivially_copy_assignable()` metafunction for platforms that have
>+// incomplete C++11 support (such as libstdc++ 4.x). On any platforms that do
>+// fully support C++11, we check whether this yields the same result as the std
>+// implementation.
>+//
>+// NOTE: `is_assignable<T, U>::value` is `true` if the expression
>+// `declval<T>() = declval<U>()` is well-formed when treated as an unevaluated
>+// operand. `is_trivially_assignable<T, U>` requires the assignment to call no
>+// operation that is not trivial. `is_trivially_copy_assignable<T>` is simply
>+// `is_trivially_assignable<T&, const T&>`.
>+template <typename T>
>+struct is_trivially_copy_assignable
>+    : std::integral_constant<
>+          bool, __has_trivial_assign(typename std::remove_reference<T>::type) &&
>+                    std::is_copy_assignable<T>::value> {
>+#ifdef ABSL_HAVE_STD_IS_TRIVIALLY_ASSIGNABLE
>+ private:
>+  static constexpr bool compliant =
>+      std::is_trivially_copy_assignable<T>::value ==
>+      is_trivially_copy_assignable::value;
>+  static_assert(compliant || std::is_trivially_copy_assignable<T>::value,
>+                "Not compliant with std::is_trivially_copy_assignable; "
>+                "Standard: false, Implementation: true");
>+  static_assert(compliant || !std::is_trivially_copy_assignable<T>::value,
>+                "Not compliant with std::is_trivially_copy_assignable; "
>+                "Standard: true, Implementation: false");
>+#endif  // ABSL_HAVE_STD_IS_TRIVIALLY_ASSIGNABLE
>+};
>+
>+// -----------------------------------------------------------------------------
>+// C++14 "_t" trait aliases
>+// -----------------------------------------------------------------------------
>+
>+template <typename T>
>+using remove_cv_t = typename std::remove_cv<T>::type;
>+
>+template <typename T>
>+using remove_const_t = typename std::remove_const<T>::type;
>+
>+template <typename T>
>+using remove_volatile_t = typename std::remove_volatile<T>::type;
>+
>+template <typename T>
>+using add_cv_t = typename std::add_cv<T>::type;
>+
>+template <typename T>
>+using add_const_t = typename std::add_const<T>::type;
>+
>+template <typename T>
>+using add_volatile_t = typename std::add_volatile<T>::type;
>+
>+template <typename T>
>+using remove_reference_t = typename std::remove_reference<T>::type;
>+
>+template <typename T>
>+using add_lvalue_reference_t = typename std::add_lvalue_reference<T>::type;
>+
>+template <typename T>
>+using add_rvalue_reference_t = typename std::add_rvalue_reference<T>::type;
>+
>+template <typename T>
>+using remove_pointer_t = typename std::remove_pointer<T>::type;
>+
>+template <typename T>
>+using add_pointer_t = typename std::add_pointer<T>::type;
>+
>+template <typename T>
>+using make_signed_t = typename std::make_signed<T>::type;
>+
>+template <typename T>
>+using make_unsigned_t = typename std::make_unsigned<T>::type;
>+
>+template <typename T>
>+using remove_extent_t = typename std::remove_extent<T>::type;
>+
>+template <typename T>
>+using remove_all_extents_t = typename std::remove_all_extents<T>::type;
>+
>+template <size_t Len, size_t Align = type_traits_internal::
>+                          default_alignment_of_aligned_storage<Len>::value>
>+using aligned_storage_t = typename std::aligned_storage<Len, Align>::type;
>+
>+template <typename T>
>+using decay_t = typename std::decay<T>::type;
>+
>+template <bool B, typename T = void>
>+using enable_if_t = typename std::enable_if<B, T>::type;
>+
>+template <bool B, typename T, typename F>
>+using conditional_t = typename std::conditional<B, T, F>::type;
>+
>+template <typename... T>
>+using common_type_t = typename std::common_type<T...>::type;
>+
>+template <typename T>
>+using underlying_type_t = typename std::underlying_type<T>::type;
>+
>+template <typename T>
>+using result_of_t = typename std::result_of<T>::type;
>+
>+namespace type_traits_internal {
>+template <typename Key, typename = size_t>
>+struct IsHashable : std::false_type {};
>+
>+template <typename Key>
>+struct IsHashable<Key,
>+                  decltype(std::declval<std::hash<Key>>()(std::declval<Key>()))>
>+    : std::true_type {};
>+
>+template <typename Key>
>+struct IsHashEnabled
>+    : absl::conjunction<std::is_default_constructible<std::hash<Key>>,
>+                        std::is_copy_constructible<std::hash<Key>>,
>+                        std::is_destructible<std::hash<Key>>,
>+                        std::is_copy_assignable<std::hash<Key>>,
>+                        IsHashable<Key>> {};
>+}  // namespace type_traits_internal
>+
>+}  // namespace absl
>+
>+
>+#endif  // ABSL_META_TYPE_TRAITS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/meta/type_traits_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/meta/type_traits_test.cc
>new file mode 100644
>index 00000000000..81b4bd32397
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/meta/type_traits_test.cc
>@@ -0,0 +1,880 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/meta/type_traits.h"
>+
>+#include <cstdint>
>+#include <string>
>+#include <type_traits>
>+#include <utility>
>+#include <vector>
>+
>+#include "gtest/gtest.h"
>+
>+namespace {
>+
>+using ::testing::StaticAssertTypeEq;
>+
>+template <class T, class U>
>+struct simple_pair {
>+  T first;
>+  U second;
>+};
>+
>+struct Dummy {};
>+
>+struct ReturnType {};
>+struct ConvertibleToReturnType {
>+  operator ReturnType() const;  // NOLINT
>+};
>+
>+// Unique types used as parameter types for testing the detection idiom.
>+struct StructA {};
>+struct StructB {};
>+struct StructC {};
>+
>+struct TypeWithBarFunction {
>+  template <class T,
>+            absl::enable_if_t<std::is_same<T&&, StructA&>::value, int> = 0>
>+  ReturnType bar(T&&, const StructB&, StructC&&) &&;  // NOLINT
>+};
>+
>+struct TypeWithBarFunctionAndConvertibleReturnType {
>+  template <class T,
>+            absl::enable_if_t<std::is_same<T&&, StructA&>::value, int> = 0>
>+  ConvertibleToReturnType bar(T&&, const StructB&, StructC&&) &&;  // NOLINT
>+};
>+
>+template <class Class, class... Ts>
>+using BarIsCallableImpl =
>+    decltype(std::declval<Class>().bar(std::declval<Ts>()...));
>+
>+template <class Class, class... T>
>+using BarIsCallable =
>+    absl::type_traits_internal::is_detected<BarIsCallableImpl, Class, T...>;
>+
>+template <class Class, class... T>
>+using BarIsCallableConv = absl::type_traits_internal::is_detected_convertible<
>+    ReturnType, BarIsCallableImpl, Class, T...>;
>+
>+// NOTE: Test of detail type_traits_internal::is_detected.
>+TEST(IsDetectedTest, BasicUsage) {
>+  EXPECT_TRUE((BarIsCallable<TypeWithBarFunction, StructA&, const StructB&,
>+                             StructC>::value));
>+  EXPECT_TRUE(
>+      (BarIsCallable<TypeWithBarFunction, StructA&, StructB&, StructC>::value));
>+  EXPECT_TRUE(
>+      (BarIsCallable<TypeWithBarFunction, StructA&, StructB, StructC>::value));
>+
>+  EXPECT_FALSE((BarIsCallable<int, StructA&, const StructB&, StructC>::value));
>+  EXPECT_FALSE((BarIsCallable<TypeWithBarFunction&, StructA&, const StructB&,
>+                              StructC>::value));
>+  EXPECT_FALSE((BarIsCallable<TypeWithBarFunction, StructA, const StructB&,
>+                              StructC>::value));
>+}
>+
>+// NOTE: Test of detail type_traits_internal::is_detected_convertible.
>+TEST(IsDetectedConvertibleTest, BasicUsage) {
>+  EXPECT_TRUE((BarIsCallableConv<TypeWithBarFunction, StructA&, const StructB&,
>+                                 StructC>::value));
>+  EXPECT_TRUE((BarIsCallableConv<TypeWithBarFunction, StructA&, StructB&,
>+                                 StructC>::value));
>+  EXPECT_TRUE((BarIsCallableConv<TypeWithBarFunction, StructA&, StructB,
>+                                 StructC>::value));
>+  EXPECT_TRUE((BarIsCallableConv<TypeWithBarFunctionAndConvertibleReturnType,
>+                                 StructA&, const StructB&, StructC>::value));
>+  EXPECT_TRUE((BarIsCallableConv<TypeWithBarFunctionAndConvertibleReturnType,
>+                                 StructA&, StructB&, StructC>::value));
>+  EXPECT_TRUE((BarIsCallableConv<TypeWithBarFunctionAndConvertibleReturnType,
>+                                 StructA&, StructB, StructC>::value));
>+
>+  EXPECT_FALSE(
>+      (BarIsCallableConv<int, StructA&, const StructB&, StructC>::value));
>+  EXPECT_FALSE((BarIsCallableConv<TypeWithBarFunction&, StructA&,
>+                                  const StructB&, StructC>::value));
>+  EXPECT_FALSE((BarIsCallableConv<TypeWithBarFunction, StructA, const StructB&,
>+                                  StructC>::value));
>+  EXPECT_FALSE((BarIsCallableConv<TypeWithBarFunctionAndConvertibleReturnType&,
>+                                  StructA&, const StructB&, StructC>::value));
>+  EXPECT_FALSE((BarIsCallableConv<TypeWithBarFunctionAndConvertibleReturnType,
>+                                  StructA, const StructB&, StructC>::value));
>+}
>+
>+TEST(VoidTTest, BasicUsage) {
>+  StaticAssertTypeEq<void, absl::void_t<Dummy>>();
>+  StaticAssertTypeEq<void, absl::void_t<Dummy, Dummy, Dummy>>();
>+}
>+
>+TEST(ConjunctionTest, BasicBooleanLogic) {
>+  EXPECT_TRUE(absl::conjunction<>::value);
>+  EXPECT_TRUE(absl::conjunction<std::true_type>::value);
>+  EXPECT_TRUE((absl::conjunction<std::true_type, std::true_type>::value));
>+  EXPECT_FALSE((absl::conjunction<std::true_type, std::false_type>::value));
>+  EXPECT_FALSE((absl::conjunction<std::false_type, std::true_type>::value));
>+  EXPECT_FALSE((absl::conjunction<std::false_type, std::false_type>::value));
>+}
>+
>+struct MyTrueType {
>+  static constexpr bool value = true;
>+};
>+
>+struct MyFalseType {
>+  static constexpr bool value = false;
>+};
>+
>+TEST(ConjunctionTest, ShortCircuiting) {
>+  EXPECT_FALSE(
>+      (absl::conjunction<std::true_type, std::false_type, Dummy>::value));
>+  EXPECT_TRUE((std::is_base_of<MyFalseType,
>+                               absl::conjunction<std::true_type, MyFalseType,
>+                                                 std::false_type>>::value));
>+  EXPECT_TRUE(
>+      (std::is_base_of<MyTrueType,
>+                       absl::conjunction<std::true_type, MyTrueType>>::value));
>+}
>+
>+TEST(DisjunctionTest, BasicBooleanLogic) {
>+  EXPECT_FALSE(absl::disjunction<>::value);
>+  EXPECT_FALSE(absl::disjunction<std::false_type>::value);
>+  EXPECT_TRUE((absl::disjunction<std::true_type, std::true_type>::value));
>+  EXPECT_TRUE((absl::disjunction<std::true_type, std::false_type>::value));
>+  EXPECT_TRUE((absl::disjunction<std::false_type, std::true_type>::value));
>+  EXPECT_FALSE((absl::disjunction<std::false_type, std::false_type>::value));
>+}
>+
>+TEST(DisjunctionTest, ShortCircuiting) {
>+  EXPECT_TRUE(
>+      (absl::disjunction<std::false_type, std::true_type, Dummy>::value));
>+  EXPECT_TRUE((
>+      std::is_base_of<MyTrueType, absl::disjunction<std::false_type, MyTrueType,
>+                                                    std::true_type>>::value));
>+  EXPECT_TRUE((
>+      std::is_base_of<MyFalseType,
>+                      absl::disjunction<std::false_type, MyFalseType>>::value));
>+}
>+
>+TEST(NegationTest, BasicBooleanLogic) {
>+  EXPECT_FALSE(absl::negation<std::true_type>::value);
>+  EXPECT_FALSE(absl::negation<MyTrueType>::value);
>+  EXPECT_TRUE(absl::negation<std::false_type>::value);
>+  EXPECT_TRUE(absl::negation<MyFalseType>::value);
>+}
>+
>+// all member functions are trivial
>+class Trivial {
>+  int n_;
>+};
>+
>+struct TrivialDestructor {
>+  ~TrivialDestructor() = default;
>+};
>+
>+struct NontrivialDestructor {
>+  ~NontrivialDestructor() {}
>+};
>+
>+struct DeletedDestructor {
>+  ~DeletedDestructor() = delete;
>+};
>+
>+class TrivialDefaultCtor {
>+ public:
>+  TrivialDefaultCtor() = default;
>+  explicit TrivialDefaultCtor(int n) : n_(n) {}
>+
>+ private:
>+  int n_;
>+};
>+
>+class NontrivialDefaultCtor {
>+ public:
>+  NontrivialDefaultCtor() : n_(1) {}
>+
>+ private:
>+  int n_;
>+};
>+
>+class DeletedDefaultCtor {
>+ public:
>+  DeletedDefaultCtor() = delete;
>+  explicit DeletedDefaultCtor(int n) : n_(n) {}
>+
>+ private:
>+  int n_;
>+};
>+
>+class TrivialCopyCtor {
>+ public:
>+  explicit TrivialCopyCtor(int n) : n_(n) {}
>+  TrivialCopyCtor(const TrivialCopyCtor&) = default;
>+  TrivialCopyCtor& operator=(const TrivialCopyCtor& t) {
>+    n_ = t.n_;
>+    return *this;
>+  }
>+
>+ private:
>+  int n_;
>+};
>+
>+class NontrivialCopyCtor {
>+ public:
>+  explicit NontrivialCopyCtor(int n) : n_(n) {}
>+  NontrivialCopyCtor(const NontrivialCopyCtor& t) : n_(t.n_) {}
>+  NontrivialCopyCtor& operator=(const NontrivialCopyCtor&) = default;
>+
>+ private:
>+  int n_;
>+};
>+
>+class DeletedCopyCtor {
>+ public:
>+  explicit DeletedCopyCtor(int n) : n_(n) {}
>+  DeletedCopyCtor(const DeletedCopyCtor&) = delete;
>+  DeletedCopyCtor& operator=(const DeletedCopyCtor&) = default;
>+
>+ private:
>+  int n_;
>+};
>+
>+class TrivialCopyAssign {
>+ public:
>+  explicit TrivialCopyAssign(int n) : n_(n) {}
>+  TrivialCopyAssign(const TrivialCopyAssign& t) : n_(t.n_) {}
>+  TrivialCopyAssign& operator=(const TrivialCopyAssign& t) = default;
>+  ~TrivialCopyAssign() {}  // can have nontrivial destructor
>+ private:
>+  int n_;
>+};
>+
>+class NontrivialCopyAssign {
>+ public:
>+  explicit NontrivialCopyAssign(int n) : n_(n) {}
>+  NontrivialCopyAssign(const NontrivialCopyAssign&) = default;
>+  NontrivialCopyAssign& operator=(const NontrivialCopyAssign& t) {
>+    n_ = t.n_;
>+    return *this;
>+  }
>+
>+ private:
>+  int n_;
>+};
>+
>+class DeletedCopyAssign {
>+ public:
>+  explicit DeletedCopyAssign(int n) : n_(n) {}
>+  DeletedCopyAssign(const DeletedCopyAssign&) = default;
>+  DeletedCopyAssign& operator=(const DeletedCopyAssign&) = delete;
>+
>+ private:
>+  int n_;
>+};
>+
>+struct NonCopyable {
>+  NonCopyable() = default;
>+  NonCopyable(const NonCopyable&) = delete;
>+  NonCopyable& operator=(const NonCopyable&) = delete;
>+};
>+
>+class Base {
>+ public:
>+  virtual ~Base() {}
>+};
>+
>+// In GCC/Clang, std::is_trivially_constructible requires that the destructor is
>+// trivial. However, MSVC doesn't require that. This results in different
>+// behavior when checking is_trivially_constructible on any type with
>+// nontrivial destructor. Since absl::is_trivially_default_constructible and
>+// absl::is_trivially_copy_constructible both follows Clang/GCC's interpretation
>+// and check is_trivially_destructible, it results in inconsistency with
>+// std::is_trivially_xxx_constructible on MSVC. This macro is used to work
>+// around this issue in test. In practice, a trivially constructible type
>+// should also be trivially destructible.
>+// GCC bug 51452: https://gcc.gnu.org/bugzilla/show_bug.cgi?id=51452
>+// LWG issue 2116: http://cplusplus.github.io/LWG/lwg-active.html#2116
>+#ifndef _MSC_VER
>+#define ABSL_TRIVIALLY_CONSTRUCTIBLE_VERIFY_TRIVIALLY_DESTRUCTIBLE 1
>+#endif
>+
>+// Old versions of libc++, around Clang 3.5 to 3.6, consider deleted destructors
>+// as also being trivial. With the resolution of CWG 1928 and CWG 1734, this
>+// is no longer considered true and has thus been amended.
>+// Compiler Explorer: https://godbolt.org/g/zT59ZL
>+// CWG issue 1734: http://open-std.org/JTC1/SC22/WG21/docs/cwg_defects.html#1734
>+// CWG issue 1928: http://open-std.org/JTC1/SC22/WG21/docs/cwg_closed.html#1928
>+#if !defined(_LIBCPP_VERSION) || _LIBCPP_VERSION >= 3700
>+#define ABSL_TRIVIALLY_DESTRUCTIBLE_CONSIDER_DELETED_DESTRUCTOR_NOT_TRIVIAL 1
>+#endif
>+
>+// As of the moment, GCC versions >5.1 have a problem compiling for
>+// std::is_trivially_default_constructible<NontrivialDestructor[10]>, where
>+// NontrivialDestructor is a struct with a custom nontrivial destructor. Note
>+// that this problem only occurs for arrays of a known size, so something like
>+// std::is_trivially_default_constructible<NontrivialDestructor[]> does not
>+// have any problems.
>+// Compiler Explorer: https://godbolt.org/g/dXRbdK
>+// GCC bug 83689: https://gcc.gnu.org/bugzilla/show_bug.cgi?id=83689
>+#if defined(__clang__) || defined(_MSC_VER) || \
>+    (defined(__GNUC__) && __GNUC__ < 5)
>+#define ABSL_GCC_BUG_TRIVIALLY_CONSTRUCTIBLE_ON_ARRAY_OF_NONTRIVIAL 1
>+#endif
>+
>+TEST(TypeTraitsTest, TestTrivialDestructor) {
>+  // Verify that arithmetic types and pointers have trivial destructors.
>+  EXPECT_TRUE(absl::is_trivially_destructible<bool>::value);
>+  EXPECT_TRUE(absl::is_trivially_destructible<char>::value);
>+  EXPECT_TRUE(absl::is_trivially_destructible<unsigned char>::value);
>+  EXPECT_TRUE(absl::is_trivially_destructible<signed char>::value);
>+  EXPECT_TRUE(absl::is_trivially_destructible<wchar_t>::value);
>+  EXPECT_TRUE(absl::is_trivially_destructible<int>::value);
>+  EXPECT_TRUE(absl::is_trivially_destructible<unsigned int>::value);
>+  EXPECT_TRUE(absl::is_trivially_destructible<int16_t>::value);
>+  EXPECT_TRUE(absl::is_trivially_destructible<uint16_t>::value);
>+  EXPECT_TRUE(absl::is_trivially_destructible<int64_t>::value);
>+  EXPECT_TRUE(absl::is_trivially_destructible<uint64_t>::value);
>+  EXPECT_TRUE(absl::is_trivially_destructible<float>::value);
>+  EXPECT_TRUE(absl::is_trivially_destructible<double>::value);
>+  EXPECT_TRUE(absl::is_trivially_destructible<long double>::value);
>+  EXPECT_TRUE(absl::is_trivially_destructible<std::string*>::value);
>+  EXPECT_TRUE(absl::is_trivially_destructible<Trivial*>::value);
>+  EXPECT_TRUE(absl::is_trivially_destructible<const std::string*>::value);
>+  EXPECT_TRUE(absl::is_trivially_destructible<const Trivial*>::value);
>+  EXPECT_TRUE(absl::is_trivially_destructible<std::string**>::value);
>+  EXPECT_TRUE(absl::is_trivially_destructible<Trivial**>::value);
>+
>+  // classes with destructors
>+  EXPECT_TRUE(absl::is_trivially_destructible<Trivial>::value);
>+  EXPECT_TRUE(absl::is_trivially_destructible<TrivialDestructor>::value);
>+
>+  // Verify that types with a nontrivial or deleted destructor
>+  // are marked as such.
>+  EXPECT_FALSE(absl::is_trivially_destructible<NontrivialDestructor>::value);
>+#ifdef ABSL_TRIVIALLY_DESTRUCTIBLE_CONSIDER_DELETED_DESTRUCTOR_NOT_TRIVIAL
>+  EXPECT_FALSE(absl::is_trivially_destructible<DeletedDestructor>::value);
>+#endif
>+
>+  // simple_pair of such types is trivial
>+  EXPECT_TRUE((absl::is_trivially_destructible<simple_pair<int, int>>::value));
>+  EXPECT_TRUE((absl::is_trivially_destructible<
>+               simple_pair<Trivial, TrivialDestructor>>::value));
>+
>+  // Verify that types without trivial destructors are correctly marked as such.
>+  EXPECT_FALSE(absl::is_trivially_destructible<std::string>::value);
>+  EXPECT_FALSE(absl::is_trivially_destructible<std::vector<int>>::value);
>+
>+  // Verify that simple_pairs of types without trivial destructors
>+  // are not marked as trivial.
>+  EXPECT_FALSE((absl::is_trivially_destructible<
>+                simple_pair<int, std::string>>::value));
>+  EXPECT_FALSE((absl::is_trivially_destructible<
>+                simple_pair<std::string, int>>::value));
>+
>+  // array of such types is trivial
>+  using int10 = int[10];
>+  EXPECT_TRUE(absl::is_trivially_destructible<int10>::value);
>+  using Trivial10 = Trivial[10];
>+  EXPECT_TRUE(absl::is_trivially_destructible<Trivial10>::value);
>+  using TrivialDestructor10 = TrivialDestructor[10];
>+  EXPECT_TRUE(absl::is_trivially_destructible<TrivialDestructor10>::value);
>+
>+  // Conversely, the opposite also holds.
>+  using NontrivialDestructor10 = NontrivialDestructor[10];
>+  EXPECT_FALSE(absl::is_trivially_destructible<NontrivialDestructor10>::value);
>+}
>+
>+TEST(TypeTraitsTest, TestTrivialDefaultCtor) {
>+  // arithmetic types and pointers have trivial default constructors.
>+  EXPECT_TRUE(absl::is_trivially_default_constructible<bool>::value);
>+  EXPECT_TRUE(absl::is_trivially_default_constructible<char>::value);
>+  EXPECT_TRUE(absl::is_trivially_default_constructible<unsigned char>::value);
>+  EXPECT_TRUE(absl::is_trivially_default_constructible<signed char>::value);
>+  EXPECT_TRUE(absl::is_trivially_default_constructible<wchar_t>::value);
>+  EXPECT_TRUE(absl::is_trivially_default_constructible<int>::value);
>+  EXPECT_TRUE(absl::is_trivially_default_constructible<unsigned int>::value);
>+  EXPECT_TRUE(absl::is_trivially_default_constructible<int16_t>::value);
>+  EXPECT_TRUE(absl::is_trivially_default_constructible<uint16_t>::value);
>+  EXPECT_TRUE(absl::is_trivially_default_constructible<int64_t>::value);
>+  EXPECT_TRUE(absl::is_trivially_default_constructible<uint64_t>::value);
>+  EXPECT_TRUE(absl::is_trivially_default_constructible<float>::value);
>+  EXPECT_TRUE(absl::is_trivially_default_constructible<double>::value);
>+  EXPECT_TRUE(absl::is_trivially_default_constructible<long double>::value);
>+  EXPECT_TRUE(absl::is_trivially_default_constructible<std::string*>::value);
>+  EXPECT_TRUE(absl::is_trivially_default_constructible<Trivial*>::value);
>+  EXPECT_TRUE(
>+      absl::is_trivially_default_constructible<const std::string*>::value);
>+  EXPECT_TRUE(absl::is_trivially_default_constructible<const Trivial*>::value);
>+  EXPECT_TRUE(absl::is_trivially_default_constructible<std::string**>::value);
>+  EXPECT_TRUE(absl::is_trivially_default_constructible<Trivial**>::value);
>+
>+  // types with compiler generated default ctors
>+  EXPECT_TRUE(absl::is_trivially_default_constructible<Trivial>::value);
>+  EXPECT_TRUE(
>+      absl::is_trivially_default_constructible<TrivialDefaultCtor>::value);
>+
>+  // Verify that types without them are not.
>+  EXPECT_FALSE(
>+      absl::is_trivially_default_constructible<NontrivialDefaultCtor>::value);
>+  EXPECT_FALSE(
>+      absl::is_trivially_default_constructible<DeletedDefaultCtor>::value);
>+
>+#ifdef ABSL_TRIVIALLY_CONSTRUCTIBLE_VERIFY_TRIVIALLY_DESTRUCTIBLE
>+  // types with nontrivial destructor are nontrivial
>+  EXPECT_FALSE(
>+      absl::is_trivially_default_constructible<NontrivialDestructor>::value);
>+#endif
>+
>+  // types with vtables
>+  EXPECT_FALSE(absl::is_trivially_default_constructible<Base>::value);
>+
>+  // Verify that simple_pair has trivial constructors where applicable.
>+  EXPECT_TRUE((absl::is_trivially_default_constructible<
>+               simple_pair<int, char*>>::value));
>+  EXPECT_TRUE((absl::is_trivially_default_constructible<
>+               simple_pair<int, Trivial>>::value));
>+  EXPECT_TRUE((absl::is_trivially_default_constructible<
>+               simple_pair<int, TrivialDefaultCtor>>::value));
>+
>+  // Verify that types without trivial constructors are
>+  // correctly marked as such.
>+  EXPECT_FALSE(absl::is_trivially_default_constructible<std::string>::value);
>+  EXPECT_FALSE(
>+      absl::is_trivially_default_constructible<std::vector<int>>::value);
>+
>+  // Verify that simple_pairs of types without trivial constructors
>+  // are not marked as trivial.
>+  EXPECT_FALSE((absl::is_trivially_default_constructible<
>+                simple_pair<int, std::string>>::value));
>+  EXPECT_FALSE((absl::is_trivially_default_constructible<
>+                simple_pair<std::string, int>>::value));
>+
>+  // Verify that arrays of such types are trivially default constructible
>+  using int10 = int[10];
>+  EXPECT_TRUE(absl::is_trivially_default_constructible<int10>::value);
>+  using Trivial10 = Trivial[10];
>+  EXPECT_TRUE(absl::is_trivially_default_constructible<Trivial10>::value);
>+  using TrivialDefaultCtor10 = TrivialDefaultCtor[10];
>+  EXPECT_TRUE(
>+      absl::is_trivially_default_constructible<TrivialDefaultCtor10>::value);
>+
>+  // Conversely, the opposite also holds.
>+#ifdef ABSL_GCC_BUG_TRIVIALLY_CONSTRUCTIBLE_ON_ARRAY_OF_NONTRIVIAL
>+  using NontrivialDefaultCtor10 = NontrivialDefaultCtor[10];
>+  EXPECT_FALSE(
>+      absl::is_trivially_default_constructible<NontrivialDefaultCtor10>::value);
>+#endif
>+}
>+
>+TEST(TypeTraitsTest, TestTrivialCopyCtor) {
>+  // Verify that arithmetic types and pointers have trivial copy
>+  // constructors.
>+  EXPECT_TRUE(absl::is_trivially_copy_constructible<bool>::value);
>+  EXPECT_TRUE(absl::is_trivially_copy_constructible<char>::value);
>+  EXPECT_TRUE(absl::is_trivially_copy_constructible<unsigned char>::value);
>+  EXPECT_TRUE(absl::is_trivially_copy_constructible<signed char>::value);
>+  EXPECT_TRUE(absl::is_trivially_copy_constructible<wchar_t>::value);
>+  EXPECT_TRUE(absl::is_trivially_copy_constructible<int>::value);
>+  EXPECT_TRUE(absl::is_trivially_copy_constructible<unsigned int>::value);
>+  EXPECT_TRUE(absl::is_trivially_copy_constructible<int16_t>::value);
>+  EXPECT_TRUE(absl::is_trivially_copy_constructible<uint16_t>::value);
>+  EXPECT_TRUE(absl::is_trivially_copy_constructible<int64_t>::value);
>+  EXPECT_TRUE(absl::is_trivially_copy_constructible<uint64_t>::value);
>+  EXPECT_TRUE(absl::is_trivially_copy_constructible<float>::value);
>+  EXPECT_TRUE(absl::is_trivially_copy_constructible<double>::value);
>+  EXPECT_TRUE(absl::is_trivially_copy_constructible<long double>::value);
>+  EXPECT_TRUE(absl::is_trivially_copy_constructible<std::string*>::value);
>+  EXPECT_TRUE(absl::is_trivially_copy_constructible<Trivial*>::value);
>+  EXPECT_TRUE(absl::is_trivially_copy_constructible<const std::string*>::value);
>+  EXPECT_TRUE(absl::is_trivially_copy_constructible<const Trivial*>::value);
>+  EXPECT_TRUE(absl::is_trivially_copy_constructible<std::string**>::value);
>+  EXPECT_TRUE(absl::is_trivially_copy_constructible<Trivial**>::value);
>+
>+  // types with compiler generated copy ctors
>+  EXPECT_TRUE(absl::is_trivially_copy_constructible<Trivial>::value);
>+  EXPECT_TRUE(absl::is_trivially_copy_constructible<TrivialCopyCtor>::value);
>+
>+  // Verify that types without them (i.e. nontrivial or deleted) are not.
>+  EXPECT_FALSE(
>+      absl::is_trivially_copy_constructible<NontrivialCopyCtor>::value);
>+  EXPECT_FALSE(absl::is_trivially_copy_constructible<DeletedCopyCtor>::value);
>+  EXPECT_FALSE(
>+      absl::is_trivially_copy_constructible<NonCopyable>::value);
>+
>+#ifdef ABSL_TRIVIALLY_CONSTRUCTIBLE_VERIFY_TRIVIALLY_DESTRUCTIBLE
>+  // type with nontrivial destructor are nontrivial copy construbtible
>+  EXPECT_FALSE(
>+      absl::is_trivially_copy_constructible<NontrivialDestructor>::value);
>+#endif
>+
>+  // types with vtables
>+  EXPECT_FALSE(absl::is_trivially_copy_constructible<Base>::value);
>+
>+  // Verify that simple_pair of such types is trivially copy constructible
>+  EXPECT_TRUE(
>+      (absl::is_trivially_copy_constructible<simple_pair<int, char*>>::value));
>+  EXPECT_TRUE((
>+      absl::is_trivially_copy_constructible<simple_pair<int, Trivial>>::value));
>+  EXPECT_TRUE((absl::is_trivially_copy_constructible<
>+               simple_pair<int, TrivialCopyCtor>>::value));
>+
>+  // Verify that types without trivial copy constructors are
>+  // correctly marked as such.
>+  EXPECT_FALSE(absl::is_trivially_copy_constructible<std::string>::value);
>+  EXPECT_FALSE(absl::is_trivially_copy_constructible<std::vector<int>>::value);
>+
>+  // Verify that simple_pairs of types without trivial copy constructors
>+  // are not marked as trivial.
>+  EXPECT_FALSE((absl::is_trivially_copy_constructible<
>+                simple_pair<int, std::string>>::value));
>+  EXPECT_FALSE((absl::is_trivially_copy_constructible<
>+                simple_pair<std::string, int>>::value));
>+
>+  // Verify that arrays are not
>+  using int10 = int[10];
>+  EXPECT_FALSE(absl::is_trivially_copy_constructible<int10>::value);
>+}
>+
>+TEST(TypeTraitsTest, TestTrivialCopyAssign) {
>+  // Verify that arithmetic types and pointers have trivial copy
>+  // assignment operators.
>+  EXPECT_TRUE(absl::is_trivially_copy_assignable<bool>::value);
>+  EXPECT_TRUE(absl::is_trivially_copy_assignable<char>::value);
>+  EXPECT_TRUE(absl::is_trivially_copy_assignable<unsigned char>::value);
>+  EXPECT_TRUE(absl::is_trivially_copy_assignable<signed char>::value);
>+  EXPECT_TRUE(absl::is_trivially_copy_assignable<wchar_t>::value);
>+  EXPECT_TRUE(absl::is_trivially_copy_assignable<int>::value);
>+  EXPECT_TRUE(absl::is_trivially_copy_assignable<unsigned int>::value);
>+  EXPECT_TRUE(absl::is_trivially_copy_assignable<int16_t>::value);
>+  EXPECT_TRUE(absl::is_trivially_copy_assignable<uint16_t>::value);
>+  EXPECT_TRUE(absl::is_trivially_copy_assignable<int64_t>::value);
>+  EXPECT_TRUE(absl::is_trivially_copy_assignable<uint64_t>::value);
>+  EXPECT_TRUE(absl::is_trivially_copy_assignable<float>::value);
>+  EXPECT_TRUE(absl::is_trivially_copy_assignable<double>::value);
>+  EXPECT_TRUE(absl::is_trivially_copy_assignable<long double>::value);
>+  EXPECT_TRUE(absl::is_trivially_copy_assignable<std::string*>::value);
>+  EXPECT_TRUE(absl::is_trivially_copy_assignable<Trivial*>::value);
>+  EXPECT_TRUE(absl::is_trivially_copy_assignable<const std::string*>::value);
>+  EXPECT_TRUE(absl::is_trivially_copy_assignable<const Trivial*>::value);
>+  EXPECT_TRUE(absl::is_trivially_copy_assignable<std::string**>::value);
>+  EXPECT_TRUE(absl::is_trivially_copy_assignable<Trivial**>::value);
>+
>+  // const qualified types are not assignable
>+  EXPECT_FALSE(absl::is_trivially_copy_assignable<const int>::value);
>+
>+  // types with compiler generated copy assignment
>+  EXPECT_TRUE(absl::is_trivially_copy_assignable<Trivial>::value);
>+  EXPECT_TRUE(absl::is_trivially_copy_assignable<TrivialCopyAssign>::value);
>+
>+  // Verify that types without them (i.e. nontrivial or deleted) are not.
>+  EXPECT_FALSE(absl::is_trivially_copy_assignable<NontrivialCopyAssign>::value);
>+  EXPECT_FALSE(absl::is_trivially_copy_assignable<DeletedCopyAssign>::value);
>+  EXPECT_FALSE(absl::is_trivially_copy_assignable<NonCopyable>::value);
>+
>+  // types with vtables
>+  EXPECT_FALSE(absl::is_trivially_copy_assignable<Base>::value);
>+
>+  // Verify that simple_pair is trivially assignable
>+  EXPECT_TRUE(
>+      (absl::is_trivially_copy_assignable<simple_pair<int, char*>>::value));
>+  EXPECT_TRUE(
>+      (absl::is_trivially_copy_assignable<simple_pair<int, Trivial>>::value));
>+  EXPECT_TRUE((absl::is_trivially_copy_assignable<
>+               simple_pair<int, TrivialCopyAssign>>::value));
>+
>+  // Verify that types not trivially copy assignable are
>+  // correctly marked as such.
>+  EXPECT_FALSE(absl::is_trivially_copy_assignable<std::string>::value);
>+  EXPECT_FALSE(absl::is_trivially_copy_assignable<std::vector<int>>::value);
>+
>+  // Verify that simple_pairs of types not trivially copy assignable
>+  // are not marked as trivial.
>+  EXPECT_FALSE((absl::is_trivially_copy_assignable<
>+                simple_pair<int, std::string>>::value));
>+  EXPECT_FALSE((absl::is_trivially_copy_assignable<
>+                simple_pair<std::string, int>>::value));
>+
>+  // Verify that arrays are not trivially copy assignable
>+  using int10 = int[10];
>+  EXPECT_FALSE(absl::is_trivially_copy_assignable<int10>::value);
>+
>+  // Verify that references are handled correctly
>+  EXPECT_TRUE(absl::is_trivially_copy_assignable<Trivial&&>::value);
>+  EXPECT_TRUE(absl::is_trivially_copy_assignable<Trivial&>::value);
>+}
>+
>+#define ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(trait_name, ...)          \
>+  EXPECT_TRUE((std::is_same<typename std::trait_name<__VA_ARGS__>::type, \
>+                            absl::trait_name##_t<__VA_ARGS__>>::value))
>+
>+TEST(TypeTraitsTest, TestRemoveCVAliases) {
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(remove_cv, int);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(remove_cv, const int);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(remove_cv, volatile int);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(remove_cv, const volatile int);
>+
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(remove_const, int);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(remove_const, const int);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(remove_const, volatile int);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(remove_const, const volatile int);
>+
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(remove_volatile, int);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(remove_volatile, const int);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(remove_volatile, volatile int);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(remove_volatile, const volatile int);
>+}
>+
>+TEST(TypeTraitsTest, TestAddCVAliases) {
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(add_cv, int);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(add_cv, const int);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(add_cv, volatile int);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(add_cv, const volatile int);
>+
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(add_const, int);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(add_const, const int);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(add_const, volatile int);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(add_const, const volatile int);
>+
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(add_volatile, int);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(add_volatile, const int);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(add_volatile, volatile int);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(add_volatile, const volatile int);
>+}
>+
>+TEST(TypeTraitsTest, TestReferenceAliases) {
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(remove_reference, int);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(remove_reference, volatile int);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(remove_reference, int&);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(remove_reference, volatile int&);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(remove_reference, int&&);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(remove_reference, volatile int&&);
>+
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(add_lvalue_reference, int);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(add_lvalue_reference, volatile int);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(add_lvalue_reference, int&);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(add_lvalue_reference, volatile int&);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(add_lvalue_reference, int&&);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(add_lvalue_reference, volatile int&&);
>+
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(add_rvalue_reference, int);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(add_rvalue_reference, volatile int);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(add_rvalue_reference, int&);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(add_rvalue_reference, volatile int&);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(add_rvalue_reference, int&&);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(add_rvalue_reference, volatile int&&);
>+}
>+
>+TEST(TypeTraitsTest, TestPointerAliases) {
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(remove_pointer, int*);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(remove_pointer, volatile int*);
>+
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(add_pointer, int);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(add_pointer, volatile int);
>+}
>+
>+TEST(TypeTraitsTest, TestSignednessAliases) {
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(make_signed, int);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(make_signed, volatile int);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(make_signed, unsigned);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(make_signed, volatile unsigned);
>+
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(make_unsigned, int);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(make_unsigned, volatile int);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(make_unsigned, unsigned);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(make_unsigned, volatile unsigned);
>+}
>+
>+TEST(TypeTraitsTest, TestExtentAliases) {
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(remove_extent, int[]);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(remove_extent, int[1]);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(remove_extent, int[1][1]);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(remove_extent, int[][1]);
>+
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(remove_all_extents, int[]);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(remove_all_extents, int[1]);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(remove_all_extents, int[1][1]);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(remove_all_extents, int[][1]);
>+}
>+
>+TEST(TypeTraitsTest, TestAlignedStorageAlias) {
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 1);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 2);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 3);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 4);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 5);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 6);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 7);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 8);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 9);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 10);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 11);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 12);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 13);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 14);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 15);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 16);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 17);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 18);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 19);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 20);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 21);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 22);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 23);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 24);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 25);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 26);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 27);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 28);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 29);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 30);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 31);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 32);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 33);
>+
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 1, 128);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 2, 128);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 3, 128);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 4, 128);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 5, 128);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 6, 128);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 7, 128);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 8, 128);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 9, 128);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 10, 128);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 11, 128);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 12, 128);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 13, 128);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 14, 128);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 15, 128);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 16, 128);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 17, 128);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 18, 128);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 19, 128);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 20, 128);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 21, 128);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 22, 128);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 23, 128);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 24, 128);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 25, 128);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 26, 128);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 27, 128);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 28, 128);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 29, 128);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 30, 128);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 31, 128);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 32, 128);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(aligned_storage, 33, 128);
>+}
>+
>+TEST(TypeTraitsTest, TestDecay) {
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(decay, int);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(decay, const int);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(decay, volatile int);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(decay, const volatile int);
>+
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(decay, int&);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(decay, const int&);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(decay, volatile int&);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(decay, const volatile int&);
>+
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(decay, int&);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(decay, const int&);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(decay, volatile int&);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(decay, const volatile int&);
>+
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(decay, int[1]);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(decay, int[1][1]);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(decay, int[][1]);
>+
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(decay, int());
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(decay, int(float));  // NOLINT
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(decay, int(char, ...));  // NOLINT
>+}
>+
>+struct TypeA {};
>+struct TypeB {};
>+struct TypeC {};
>+struct TypeD {};
>+
>+template <typename T>
>+struct Wrap {};
>+
>+enum class TypeEnum { A, B, C, D };
>+
>+struct GetTypeT {
>+  template <typename T,
>+            absl::enable_if_t<std::is_same<T, TypeA>::value, int> = 0>
>+  TypeEnum operator()(Wrap<T>) const {
>+    return TypeEnum::A;
>+  }
>+
>+  template <typename T,
>+            absl::enable_if_t<std::is_same<T, TypeB>::value, int> = 0>
>+  TypeEnum operator()(Wrap<T>) const {
>+    return TypeEnum::B;
>+  }
>+
>+  template <typename T,
>+            absl::enable_if_t<std::is_same<T, TypeC>::value, int> = 0>
>+  TypeEnum operator()(Wrap<T>) const {
>+    return TypeEnum::C;
>+  }
>+
>+  // NOTE: TypeD is intentionally not handled
>+} constexpr GetType = {};
>+
>+TEST(TypeTraitsTest, TestEnableIf) {
>+  EXPECT_EQ(TypeEnum::A, GetType(Wrap<TypeA>()));
>+  EXPECT_EQ(TypeEnum::B, GetType(Wrap<TypeB>()));
>+  EXPECT_EQ(TypeEnum::C, GetType(Wrap<TypeC>()));
>+}
>+
>+TEST(TypeTraitsTest, TestConditional) {
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(conditional, true, int, char);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(conditional, false, int, char);
>+}
>+
>+// TODO(calabrese) Check with specialized std::common_type
>+TEST(TypeTraitsTest, TestCommonType) {
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(common_type, int);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(common_type, int, char);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(common_type, int, char, int);
>+
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(common_type, int&);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(common_type, int, char&);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(common_type, int, char, int&);
>+}
>+
>+TEST(TypeTraitsTest, TestUnderlyingType) {
>+  enum class enum_char : char {};
>+  enum class enum_long_long : long long {};  // NOLINT(runtime/int)
>+
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(underlying_type, enum_char);
>+  ABSL_INTERNAL_EXPECT_ALIAS_EQUIVALENCE(underlying_type, enum_long_long);
>+}
>+
>+struct GetTypeExtT {
>+  template <typename T>
>+  absl::result_of_t<const GetTypeT&(T)> operator()(T&& arg) const {
>+    return GetType(std::forward<T>(arg));
>+  }
>+
>+  TypeEnum operator()(Wrap<TypeD>) const { return TypeEnum::D; }
>+} constexpr GetTypeExt = {};
>+
>+TEST(TypeTraitsTest, TestResultOf) {
>+  EXPECT_EQ(TypeEnum::A, GetTypeExt(Wrap<TypeA>()));
>+  EXPECT_EQ(TypeEnum::B, GetTypeExt(Wrap<TypeB>()));
>+  EXPECT_EQ(TypeEnum::C, GetTypeExt(Wrap<TypeC>()));
>+  EXPECT_EQ(TypeEnum::D, GetTypeExt(Wrap<TypeD>()));
>+}
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/numeric/BUILD.bazel b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/numeric/BUILD.bazel
>new file mode 100644
>index 00000000000..f49571ebb3e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/numeric/BUILD.bazel
>@@ -0,0 +1,67 @@
>+# Copyright 2018 The Abseil Authors.
>+#
>+# Licensed under the Apache License, Version 2.0 (the "License");
>+# you may not use this file except in compliance with the License.
>+# You may obtain a copy of the License at
>+#
>+#      http://www.apache.org/licenses/LICENSE-2.0
>+#
>+# Unless required by applicable law or agreed to in writing, software
>+# distributed under the License is distributed on an "AS IS" BASIS,
>+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+# See the License for the specific language governing permissions and
>+# limitations under the License.
>+
>+load(
>+    "//absl:copts.bzl",
>+    "ABSL_DEFAULT_COPTS",
>+    "ABSL_TEST_COPTS",
>+)
>+
>+package(default_visibility = ["//visibility:public"])
>+
>+licenses(["notice"])  # Apache 2.0
>+
>+cc_library(
>+    name = "int128",
>+    srcs = [
>+        "int128.cc",
>+        "int128_have_intrinsic.inc",
>+        "int128_no_intrinsic.inc",
>+    ],
>+    hdrs = ["int128.h"],
>+    copts = ABSL_DEFAULT_COPTS,
>+    deps = [
>+        "//absl/base:config",
>+        "//absl/base:core_headers",
>+    ],
>+)
>+
>+cc_test(
>+    name = "int128_test",
>+    size = "small",
>+    srcs = [
>+        "int128_stream_test.cc",
>+        "int128_test.cc",
>+    ],
>+    copts = ABSL_TEST_COPTS,
>+    deps = [
>+        ":int128",
>+        "//absl/base",
>+        "//absl/base:core_headers",
>+        "//absl/meta:type_traits",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "int128_benchmark",
>+    srcs = ["int128_benchmark.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    tags = ["benchmark"],
>+    deps = [
>+        ":int128",
>+        "//absl/base:config",
>+        "@com_github_google_benchmark//:benchmark_main",
>+    ],
>+)
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/numeric/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/numeric/BUILD.gn
>new file mode 100644
>index 00000000000..fdfa2e271ff
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/numeric/BUILD.gn
>@@ -0,0 +1,36 @@
>+# Copyright 2018 The Chromium Authors. All rights reserved.
>+# Use of this source code is governed by a BSD-style license that can be
>+# found in the LICENSE file.
>+
>+import("//build_overrides/build.gni")
>+
>+if (build_with_chromium) {
>+  visibility = [
>+    "//third_party/webrtc/*",
>+    "//third_party/abseil-cpp/*",
>+    "//third_party/googletest:gtest",
>+  ]
>+} else {
>+  visibility = [ "*" ]
>+}
>+
>+source_set("int128") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  sources = [
>+    "int128.cc",
>+    "int128_have_intrinsic.inc",
>+    "int128_no_intrinsic.inc",
>+  ]
>+  public = [
>+    "int128.h",
>+  ]
>+  deps = [
>+    "../base:config",
>+    "../base:core_headers",
>+  ]
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/numeric/CMakeLists.txt b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/numeric/CMakeLists.txt
>new file mode 100644
>index 00000000000..3360b2ee726
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/numeric/CMakeLists.txt
>@@ -0,0 +1,62 @@
>+#
>+# Copyright 2017 The Abseil Authors.
>+#
>+# Licensed under the Apache License, Version 2.0 (the "License");
>+# you may not use this file except in compliance with the License.
>+# You may obtain a copy of the License at
>+#
>+#      http://www.apache.org/licenses/LICENSE-2.0
>+#
>+# Unless required by applicable law or agreed to in writing, software
>+# distributed under the License is distributed on an "AS IS" BASIS,
>+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+# See the License for the specific language governing permissions and
>+# limitations under the License.
>+#
>+
>+list(APPEND NUMERIC_PUBLIC_HEADERS
>+  "int128.h"
>+)
>+
>+
>+# library 128
>+list(APPEND INT128_SRC
>+  "int128.cc"
>+  ${NUMERIC_PUBLIC_HEADERS}
>+)
>+absl_library(
>+  TARGET
>+    absl_int128
>+  SOURCES
>+    ${INT128_SRC}
>+  PUBLIC_LIBRARIES
>+    ${INT128_PUBLIC_LIBRARIES}
>+  EXPORT_NAME
>+    int128
>+)
>+
>+
>+absl_header_library(
>+  TARGET
>+    absl_numeric
>+  PUBLIC_LIBRARIES
>+    absl::int128
>+  EXPORT_NAME
>+    numeric
>+)
>+
>+# test int128_test
>+set(INT128_TEST_SRC "int128_test.cc")
>+set(INT128_TEST_PUBLIC_LIBRARIES absl::numeric absl::base)
>+
>+absl_test(
>+  TARGET
>+    int128_test
>+  SOURCES
>+    ${INT128_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${INT128_TEST_PUBLIC_LIBRARIES}
>+)
>+
>+
>+
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/numeric/int128.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/numeric/int128.cc
>new file mode 100644
>index 00000000000..cd79534f3bf
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/numeric/int128.cc
>@@ -0,0 +1,251 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/numeric/int128.h"
>+
>+#include <stddef.h>
>+#include <cassert>
>+#include <iomanip>
>+#include <iostream>  // NOLINT(readability/streams)
>+#include <sstream>
>+#include <string>
>+#include <type_traits>
>+
>+namespace absl {
>+
>+const uint128 kuint128max = MakeUint128(std::numeric_limits<uint64_t>::max(),
>+                                        std::numeric_limits<uint64_t>::max());
>+
>+namespace {
>+
>+// Returns the 0-based position of the last set bit (i.e., most significant bit)
>+// in the given uint64_t. The argument may not be 0.
>+//
>+// For example:
>+//   Given: 5 (decimal) == 101 (binary)
>+//   Returns: 2
>+#define STEP(T, n, pos, sh)                   \
>+  do {                                        \
>+    if ((n) >= (static_cast<T>(1) << (sh))) { \
>+      (n) = (n) >> (sh);                      \
>+      (pos) |= (sh);                          \
>+    }                                         \
>+  } while (0)
>+static inline int Fls64(uint64_t n) {
>+  assert(n != 0);
>+  int pos = 0;
>+  STEP(uint64_t, n, pos, 0x20);
>+  uint32_t n32 = static_cast<uint32_t>(n);
>+  STEP(uint32_t, n32, pos, 0x10);
>+  STEP(uint32_t, n32, pos, 0x08);
>+  STEP(uint32_t, n32, pos, 0x04);
>+  return pos + ((uint64_t{0x3333333322221100} >> (n32 << 2)) & 0x3);
>+}
>+#undef STEP
>+
>+// Like Fls64() above, but returns the 0-based position of the last set bit
>+// (i.e., most significant bit) in the given uint128. The argument may not be 0.
>+static inline int Fls128(uint128 n) {
>+  if (uint64_t hi = Uint128High64(n)) {
>+    return Fls64(hi) + 64;
>+  }
>+  return Fls64(Uint128Low64(n));
>+}
>+
>+// Long division/modulo for uint128 implemented using the shift-subtract
>+// division algorithm adapted from:
>+// http://stackoverflow.com/questions/5386377/division-without-using
>+void DivModImpl(uint128 dividend, uint128 divisor, uint128* quotient_ret,
>+                uint128* remainder_ret) {
>+  assert(divisor != 0);
>+
>+  if (divisor > dividend) {
>+    *quotient_ret = 0;
>+    *remainder_ret = dividend;
>+    return;
>+  }
>+
>+  if (divisor == dividend) {
>+    *quotient_ret = 1;
>+    *remainder_ret = 0;
>+    return;
>+  }
>+
>+  uint128 denominator = divisor;
>+  uint128 quotient = 0;
>+
>+  // Left aligns the MSB of the denominator and the dividend.
>+  const int shift = Fls128(dividend) - Fls128(denominator);
>+  denominator <<= shift;
>+
>+  // Uses shift-subtract algorithm to divide dividend by denominator. The
>+  // remainder will be left in dividend.
>+  for (int i = 0; i <= shift; ++i) {
>+    quotient <<= 1;
>+    if (dividend >= denominator) {
>+      dividend -= denominator;
>+      quotient |= 1;
>+    }
>+    denominator >>= 1;
>+  }
>+
>+  *quotient_ret = quotient;
>+  *remainder_ret = dividend;
>+}
>+
>+template <typename T>
>+uint128 MakeUint128FromFloat(T v) {
>+  static_assert(std::is_floating_point<T>::value, "");
>+
>+  // Rounding behavior is towards zero, same as for built-in types.
>+
>+  // Undefined behavior if v is NaN or cannot fit into uint128.
>+  assert(std::isfinite(v) && v > -1 &&
>+         (std::numeric_limits<T>::max_exponent <= 128 ||
>+          v < std::ldexp(static_cast<T>(1), 128)));
>+
>+  if (v >= std::ldexp(static_cast<T>(1), 64)) {
>+    uint64_t hi = static_cast<uint64_t>(std::ldexp(v, -64));
>+    uint64_t lo = static_cast<uint64_t>(v - std::ldexp(static_cast<T>(hi), 64));
>+    return MakeUint128(hi, lo);
>+  }
>+
>+  return MakeUint128(0, static_cast<uint64_t>(v));
>+}
>+}  // namespace
>+
>+uint128::uint128(float v) : uint128(MakeUint128FromFloat(v)) {}
>+uint128::uint128(double v) : uint128(MakeUint128FromFloat(v)) {}
>+uint128::uint128(long double v) : uint128(MakeUint128FromFloat(v)) {}
>+
>+uint128 operator/(uint128 lhs, uint128 rhs) {
>+#if defined(ABSL_HAVE_INTRINSIC_INT128)
>+  return static_cast<unsigned __int128>(lhs) /
>+         static_cast<unsigned __int128>(rhs);
>+#else  // ABSL_HAVE_INTRINSIC_INT128
>+  uint128 quotient = 0;
>+  uint128 remainder = 0;
>+  DivModImpl(lhs, rhs, &quotient, &remainder);
>+  return quotient;
>+#endif  // ABSL_HAVE_INTRINSIC_INT128
>+}
>+uint128 operator%(uint128 lhs, uint128 rhs) {
>+#if defined(ABSL_HAVE_INTRINSIC_INT128)
>+  return static_cast<unsigned __int128>(lhs) %
>+         static_cast<unsigned __int128>(rhs);
>+#else  // ABSL_HAVE_INTRINSIC_INT128
>+  uint128 quotient = 0;
>+  uint128 remainder = 0;
>+  DivModImpl(lhs, rhs, &quotient, &remainder);
>+  return remainder;
>+#endif  // ABSL_HAVE_INTRINSIC_INT128
>+}
>+
>+namespace {
>+
>+std::string Uint128ToFormattedString(uint128 v, std::ios_base::fmtflags flags) {
>+  // Select a divisor which is the largest power of the base < 2^64.
>+  uint128 div;
>+  int div_base_log;
>+  switch (flags & std::ios::basefield) {
>+    case std::ios::hex:
>+      div = 0x1000000000000000;  // 16^15
>+      div_base_log = 15;
>+      break;
>+    case std::ios::oct:
>+      div = 01000000000000000000000;  // 8^21
>+      div_base_log = 21;
>+      break;
>+    default:  // std::ios::dec
>+      div = 10000000000000000000u;  // 10^19
>+      div_base_log = 19;
>+      break;
>+  }
>+
>+  // Now piece together the uint128 representation from three chunks of the
>+  // original value, each less than "div" and therefore representable as a
>+  // uint64_t.
>+  std::ostringstream os;
>+  std::ios_base::fmtflags copy_mask =
>+      std::ios::basefield | std::ios::showbase | std::ios::uppercase;
>+  os.setf(flags & copy_mask, copy_mask);
>+  uint128 high = v;
>+  uint128 low;
>+  DivModImpl(high, div, &high, &low);
>+  uint128 mid;
>+  DivModImpl(high, div, &high, &mid);
>+  if (Uint128Low64(high) != 0) {
>+    os << Uint128Low64(high);
>+    os << std::noshowbase << std::setfill('0') << std::setw(div_base_log);
>+    os << Uint128Low64(mid);
>+    os << std::setw(div_base_log);
>+  } else if (Uint128Low64(mid) != 0) {
>+    os << Uint128Low64(mid);
>+    os << std::noshowbase << std::setfill('0') << std::setw(div_base_log);
>+  }
>+  os << Uint128Low64(low);
>+  return os.str();
>+}
>+
>+}  // namespace
>+
>+std::ostream& operator<<(std::ostream& os, uint128 v) {
>+  std::ios_base::fmtflags flags = os.flags();
>+  std::string rep = Uint128ToFormattedString(v, flags);
>+
>+  // Add the requisite padding.
>+  std::streamsize width = os.width(0);
>+  if (static_cast<size_t>(width) > rep.size()) {
>+    std::ios::fmtflags adjustfield = flags & std::ios::adjustfield;
>+    if (adjustfield == std::ios::left) {
>+      rep.append(width - rep.size(), os.fill());
>+    } else if (adjustfield == std::ios::internal &&
>+               (flags & std::ios::showbase) &&
>+               (flags & std::ios::basefield) == std::ios::hex && v != 0) {
>+      rep.insert(2, width - rep.size(), os.fill());
>+    } else {
>+      rep.insert(0, width - rep.size(), os.fill());
>+    }
>+  }
>+
>+  return os << rep;
>+}
>+
>+}  // namespace absl
>+
>+namespace std {
>+constexpr bool numeric_limits<absl::uint128>::is_specialized;
>+constexpr bool numeric_limits<absl::uint128>::is_signed;
>+constexpr bool numeric_limits<absl::uint128>::is_integer;
>+constexpr bool numeric_limits<absl::uint128>::is_exact;
>+constexpr bool numeric_limits<absl::uint128>::has_infinity;
>+constexpr bool numeric_limits<absl::uint128>::has_quiet_NaN;
>+constexpr bool numeric_limits<absl::uint128>::has_signaling_NaN;
>+constexpr float_denorm_style numeric_limits<absl::uint128>::has_denorm;
>+constexpr bool numeric_limits<absl::uint128>::has_denorm_loss;
>+constexpr float_round_style numeric_limits<absl::uint128>::round_style;
>+constexpr bool numeric_limits<absl::uint128>::is_iec559;
>+constexpr bool numeric_limits<absl::uint128>::is_bounded;
>+constexpr bool numeric_limits<absl::uint128>::is_modulo;
>+constexpr int numeric_limits<absl::uint128>::digits;
>+constexpr int numeric_limits<absl::uint128>::digits10;
>+constexpr int numeric_limits<absl::uint128>::max_digits10;
>+constexpr int numeric_limits<absl::uint128>::radix;
>+constexpr int numeric_limits<absl::uint128>::min_exponent;
>+constexpr int numeric_limits<absl::uint128>::min_exponent10;
>+constexpr int numeric_limits<absl::uint128>::max_exponent;
>+constexpr int numeric_limits<absl::uint128>::max_exponent10;
>+constexpr bool numeric_limits<absl::uint128>::traps;
>+constexpr bool numeric_limits<absl::uint128>::tinyness_before;
>+}  // namespace std
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/numeric/int128.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/numeric/int128.h
>new file mode 100644
>index 00000000000..2d131b8bda1
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/numeric/int128.h
>@@ -0,0 +1,705 @@
>+//
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// -----------------------------------------------------------------------------
>+// File: int128.h
>+// -----------------------------------------------------------------------------
>+//
>+// This header file defines 128-bit integer types.
>+//
>+// Currently, this file defines `uint128`, an unsigned 128-bit integer; a signed
>+// 128-bit integer is forthcoming.
>+
>+#ifndef ABSL_NUMERIC_INT128_H_
>+#define ABSL_NUMERIC_INT128_H_
>+
>+#include <cassert>
>+#include <cmath>
>+#include <cstdint>
>+#include <cstring>
>+#include <iosfwd>
>+#include <limits>
>+#include <utility>
>+
>+#include "absl/base/config.h"
>+#include "absl/base/macros.h"
>+#include "absl/base/port.h"
>+
>+namespace absl {
>+
>+
>+// uint128
>+//
>+// An unsigned 128-bit integer type. The API is meant to mimic an intrinsic type
>+// as closely as is practical, including exhibiting undefined behavior in
>+// analogous cases (e.g. division by zero). This type is intended to be a
>+// drop-in replacement once C++ supports an intrinsic `uint128_t` type; when
>+// that occurs, existing well-behaved uses of `uint128` will continue to work
>+// using that new type.
>+//
>+// Note: code written with this type will continue to compile once `uint128_t`
>+// is introduced, provided the replacement helper functions
>+// `Uint128(Low|High)64()` and `MakeUint128()` are made.
>+//
>+// A `uint128` supports the following:
>+//
>+//   * Implicit construction from integral types
>+//   * Explicit conversion to integral types
>+//
>+// Additionally, if your compiler supports `__int128`, `uint128` is
>+// interoperable with that type. (Abseil checks for this compatibility through
>+// the `ABSL_HAVE_INTRINSIC_INT128` macro.)
>+//
>+// However, a `uint128` differs from intrinsic integral types in the following
>+// ways:
>+//
>+//   * Errors on implicit conversions that do not preserve value (such as
>+//     loss of precision when converting to float values).
>+//   * Requires explicit construction from and conversion to floating point
>+//     types.
>+//   * Conversion to integral types requires an explicit static_cast() to
>+//     mimic use of the `-Wnarrowing` compiler flag.
>+//   * The alignment requirement of `uint128` may differ from that of an
>+//     intrinsic 128-bit integer type depending on platform and build
>+//     configuration.
>+//
>+// Example:
>+//
>+//     float y = absl::Uint128Max();  // Error. uint128 cannot be implicitly
>+//                                    // converted to float.
>+//
>+//     absl::uint128 v;
>+//     uint64_t i = v;                         // Error
>+//     uint64_t i = static_cast<uint64_t>(v);  // OK
>+//
>+class
>+#if defined(ABSL_HAVE_INTRINSIC_INT128)
>+    alignas(unsigned __int128)
>+#endif  // ABSL_HAVE_INTRINSIC_INT128
>+        uint128 {
>+ public:
>+  uint128() = default;
>+
>+  // Constructors from arithmetic types
>+  constexpr uint128(int v);                 // NOLINT(runtime/explicit)
>+  constexpr uint128(unsigned int v);        // NOLINT(runtime/explicit)
>+  constexpr uint128(long v);                // NOLINT(runtime/int)
>+  constexpr uint128(unsigned long v);       // NOLINT(runtime/int)
>+  constexpr uint128(long long v);           // NOLINT(runtime/int)
>+  constexpr uint128(unsigned long long v);  // NOLINT(runtime/int)
>+#ifdef ABSL_HAVE_INTRINSIC_INT128
>+  constexpr uint128(__int128 v);           // NOLINT(runtime/explicit)
>+  constexpr uint128(unsigned __int128 v);  // NOLINT(runtime/explicit)
>+#endif  // ABSL_HAVE_INTRINSIC_INT128
>+  explicit uint128(float v);
>+  explicit uint128(double v);
>+  explicit uint128(long double v);
>+
>+  // Assignment operators from arithmetic types
>+  uint128& operator=(int v);
>+  uint128& operator=(unsigned int v);
>+  uint128& operator=(long v);                // NOLINT(runtime/int)
>+  uint128& operator=(unsigned long v);       // NOLINT(runtime/int)
>+  uint128& operator=(long long v);           // NOLINT(runtime/int)
>+  uint128& operator=(unsigned long long v);  // NOLINT(runtime/int)
>+#ifdef ABSL_HAVE_INTRINSIC_INT128
>+  uint128& operator=(__int128 v);
>+  uint128& operator=(unsigned __int128 v);
>+#endif  // ABSL_HAVE_INTRINSIC_INT128
>+
>+  // Conversion operators to other arithmetic types
>+  constexpr explicit operator bool() const;
>+  constexpr explicit operator char() const;
>+  constexpr explicit operator signed char() const;
>+  constexpr explicit operator unsigned char() const;
>+  constexpr explicit operator char16_t() const;
>+  constexpr explicit operator char32_t() const;
>+  constexpr explicit operator wchar_t() const;
>+  constexpr explicit operator short() const;  // NOLINT(runtime/int)
>+  // NOLINTNEXTLINE(runtime/int)
>+  constexpr explicit operator unsigned short() const;
>+  constexpr explicit operator int() const;
>+  constexpr explicit operator unsigned int() const;
>+  constexpr explicit operator long() const;  // NOLINT(runtime/int)
>+  // NOLINTNEXTLINE(runtime/int)
>+  constexpr explicit operator unsigned long() const;
>+  // NOLINTNEXTLINE(runtime/int)
>+  constexpr explicit operator long long() const;
>+  // NOLINTNEXTLINE(runtime/int)
>+  constexpr explicit operator unsigned long long() const;
>+#ifdef ABSL_HAVE_INTRINSIC_INT128
>+  constexpr explicit operator __int128() const;
>+  constexpr explicit operator unsigned __int128() const;
>+#endif  // ABSL_HAVE_INTRINSIC_INT128
>+  explicit operator float() const;
>+  explicit operator double() const;
>+  explicit operator long double() const;
>+
>+  // Trivial copy constructor, assignment operator and destructor.
>+
>+  // Arithmetic operators.
>+  uint128& operator+=(uint128 other);
>+  uint128& operator-=(uint128 other);
>+  uint128& operator*=(uint128 other);
>+  // Long division/modulo for uint128.
>+  uint128& operator/=(uint128 other);
>+  uint128& operator%=(uint128 other);
>+  uint128 operator++(int);
>+  uint128 operator--(int);
>+  uint128& operator<<=(int);
>+  uint128& operator>>=(int);
>+  uint128& operator&=(uint128 other);
>+  uint128& operator|=(uint128 other);
>+  uint128& operator^=(uint128 other);
>+  uint128& operator++();
>+  uint128& operator--();
>+
>+  // Uint128Low64()
>+  //
>+  // Returns the lower 64-bit value of a `uint128` value.
>+  friend constexpr uint64_t Uint128Low64(uint128 v);
>+
>+  // Uint128High64()
>+  //
>+  // Returns the higher 64-bit value of a `uint128` value.
>+  friend constexpr uint64_t Uint128High64(uint128 v);
>+
>+  // MakeUInt128()
>+  //
>+  // Constructs a `uint128` numeric value from two 64-bit unsigned integers.
>+  // Note that this factory function is the only way to construct a `uint128`
>+  // from integer values greater than 2^64.
>+  //
>+  // Example:
>+  //
>+  //   absl::uint128 big = absl::MakeUint128(1, 0);
>+  friend constexpr uint128 MakeUint128(uint64_t high, uint64_t low);
>+
>+  // Uint128Max()
>+  //
>+  // Returns the highest value for a 128-bit unsigned integer.
>+  friend constexpr uint128 Uint128Max();
>+
>+ private:
>+  constexpr uint128(uint64_t high, uint64_t low);
>+
>+  // TODO(strel) Update implementation to use __int128 once all users of
>+  // uint128 are fixed to not depend on alignof(uint128) == 8. Also add
>+  // alignas(16) to class definition to keep alignment consistent across
>+  // platforms.
>+#if defined(ABSL_IS_LITTLE_ENDIAN)
>+  uint64_t lo_;
>+  uint64_t hi_;
>+#elif defined(ABSL_IS_BIG_ENDIAN)
>+  uint64_t hi_;
>+  uint64_t lo_;
>+#else  // byte order
>+#error "Unsupported byte order: must be little-endian or big-endian."
>+#endif  // byte order
>+};
>+
>+// Prefer to use the constexpr `Uint128Max()`.
>+//
>+// TODO(absl-team) deprecate kuint128max once migration tool is released.
>+extern const uint128 kuint128max;
>+
>+// allow uint128 to be logged
>+std::ostream& operator<<(std::ostream& os, uint128 v);
>+
>+// TODO(strel) add operator>>(std::istream&, uint128)
>+
>+constexpr uint128 Uint128Max() {
>+  return uint128(std::numeric_limits<uint64_t>::max(),
>+                 std::numeric_limits<uint64_t>::max());
>+}
>+
>+}  // namespace absl
>+
>+// Specialized numeric_limits for uint128.
>+namespace std {
>+template <>
>+class numeric_limits<absl::uint128> {
>+ public:
>+  static constexpr bool is_specialized = true;
>+  static constexpr bool is_signed = false;
>+  static constexpr bool is_integer = true;
>+  static constexpr bool is_exact = true;
>+  static constexpr bool has_infinity = false;
>+  static constexpr bool has_quiet_NaN = false;
>+  static constexpr bool has_signaling_NaN = false;
>+  static constexpr float_denorm_style has_denorm = denorm_absent;
>+  static constexpr bool has_denorm_loss = false;
>+  static constexpr float_round_style round_style = round_toward_zero;
>+  static constexpr bool is_iec559 = false;
>+  static constexpr bool is_bounded = true;
>+  static constexpr bool is_modulo = true;
>+  static constexpr int digits = 128;
>+  static constexpr int digits10 = 38;
>+  static constexpr int max_digits10 = 0;
>+  static constexpr int radix = 2;
>+  static constexpr int min_exponent = 0;
>+  static constexpr int min_exponent10 = 0;
>+  static constexpr int max_exponent = 0;
>+  static constexpr int max_exponent10 = 0;
>+#ifdef ABSL_HAVE_INTRINSIC_INT128
>+  static constexpr bool traps = numeric_limits<unsigned __int128>::traps;
>+#else   // ABSL_HAVE_INTRINSIC_INT128
>+  static constexpr bool traps = numeric_limits<uint64_t>::traps;
>+#endif  // ABSL_HAVE_INTRINSIC_INT128
>+  static constexpr bool tinyness_before = false;
>+
>+  static constexpr absl::uint128 min() { return 0; }
>+  static constexpr absl::uint128 lowest() { return 0; }
>+  static constexpr absl::uint128 max() { return absl::Uint128Max(); }
>+  static constexpr absl::uint128 epsilon() { return 0; }
>+  static constexpr absl::uint128 round_error() { return 0; }
>+  static constexpr absl::uint128 infinity() { return 0; }
>+  static constexpr absl::uint128 quiet_NaN() { return 0; }
>+  static constexpr absl::uint128 signaling_NaN() { return 0; }
>+  static constexpr absl::uint128 denorm_min() { return 0; }
>+};
>+}  // namespace std
>+
>+// TODO(absl-team): Implement signed 128-bit type
>+
>+// --------------------------------------------------------------------------
>+//                      Implementation details follow
>+// --------------------------------------------------------------------------
>+namespace absl {
>+
>+constexpr uint128 MakeUint128(uint64_t high, uint64_t low) {
>+  return uint128(high, low);
>+}
>+
>+// Assignment from integer types.
>+
>+inline uint128& uint128::operator=(int v) { return *this = uint128(v); }
>+
>+inline uint128& uint128::operator=(unsigned int v) {
>+  return *this = uint128(v);
>+}
>+
>+inline uint128& uint128::operator=(long v) {  // NOLINT(runtime/int)
>+  return *this = uint128(v);
>+}
>+
>+// NOLINTNEXTLINE(runtime/int)
>+inline uint128& uint128::operator=(unsigned long v) {
>+  return *this = uint128(v);
>+}
>+
>+// NOLINTNEXTLINE(runtime/int)
>+inline uint128& uint128::operator=(long long v) {
>+  return *this = uint128(v);
>+}
>+
>+// NOLINTNEXTLINE(runtime/int)
>+inline uint128& uint128::operator=(unsigned long long v) {
>+  return *this = uint128(v);
>+}
>+
>+#ifdef ABSL_HAVE_INTRINSIC_INT128
>+inline uint128& uint128::operator=(__int128 v) {
>+  return *this = uint128(v);
>+}
>+
>+inline uint128& uint128::operator=(unsigned __int128 v) {
>+  return *this = uint128(v);
>+}
>+#endif  // ABSL_HAVE_INTRINSIC_INT128
>+
>+// Arithmetic operators.
>+
>+uint128 operator<<(uint128 lhs, int amount);
>+uint128 operator>>(uint128 lhs, int amount);
>+uint128 operator+(uint128 lhs, uint128 rhs);
>+uint128 operator-(uint128 lhs, uint128 rhs);
>+uint128 operator*(uint128 lhs, uint128 rhs);
>+uint128 operator/(uint128 lhs, uint128 rhs);
>+uint128 operator%(uint128 lhs, uint128 rhs);
>+
>+inline uint128& uint128::operator<<=(int amount) {
>+  *this = *this << amount;
>+  return *this;
>+}
>+
>+inline uint128& uint128::operator>>=(int amount) {
>+  *this = *this >> amount;
>+  return *this;
>+}
>+
>+inline uint128& uint128::operator+=(uint128 other) {
>+  *this = *this + other;
>+  return *this;
>+}
>+
>+inline uint128& uint128::operator-=(uint128 other) {
>+  *this = *this - other;
>+  return *this;
>+}
>+
>+inline uint128& uint128::operator*=(uint128 other) {
>+  *this = *this * other;
>+  return *this;
>+}
>+
>+inline uint128& uint128::operator/=(uint128 other) {
>+  *this = *this / other;
>+  return *this;
>+}
>+
>+inline uint128& uint128::operator%=(uint128 other) {
>+  *this = *this % other;
>+  return *this;
>+}
>+
>+constexpr uint64_t Uint128Low64(uint128 v) { return v.lo_; }
>+
>+constexpr uint64_t Uint128High64(uint128 v) { return v.hi_; }
>+
>+// Constructors from integer types.
>+
>+#if defined(ABSL_IS_LITTLE_ENDIAN)
>+
>+constexpr uint128::uint128(uint64_t high, uint64_t low)
>+    : lo_{low}, hi_{high} {}
>+
>+constexpr uint128::uint128(int v)
>+    : lo_{static_cast<uint64_t>(v)},
>+      hi_{v < 0 ? std::numeric_limits<uint64_t>::max() : 0} {}
>+constexpr uint128::uint128(long v)  // NOLINT(runtime/int)
>+    : lo_{static_cast<uint64_t>(v)},
>+      hi_{v < 0 ? std::numeric_limits<uint64_t>::max() : 0} {}
>+constexpr uint128::uint128(long long v)  // NOLINT(runtime/int)
>+    : lo_{static_cast<uint64_t>(v)},
>+      hi_{v < 0 ? std::numeric_limits<uint64_t>::max() : 0} {}
>+
>+constexpr uint128::uint128(unsigned int v) : lo_{v}, hi_{0} {}
>+// NOLINTNEXTLINE(runtime/int)
>+constexpr uint128::uint128(unsigned long v) : lo_{v}, hi_{0} {}
>+// NOLINTNEXTLINE(runtime/int)
>+constexpr uint128::uint128(unsigned long long v) : lo_{v}, hi_{0} {}
>+
>+#ifdef ABSL_HAVE_INTRINSIC_INT128
>+constexpr uint128::uint128(__int128 v)
>+    : lo_{static_cast<uint64_t>(v & ~uint64_t{0})},
>+      hi_{static_cast<uint64_t>(static_cast<unsigned __int128>(v) >> 64)} {}
>+constexpr uint128::uint128(unsigned __int128 v)
>+    : lo_{static_cast<uint64_t>(v & ~uint64_t{0})},
>+      hi_{static_cast<uint64_t>(v >> 64)} {}
>+#endif  // ABSL_HAVE_INTRINSIC_INT128
>+
>+#elif defined(ABSL_IS_BIG_ENDIAN)
>+
>+constexpr uint128::uint128(uint64_t high, uint64_t low)
>+    : hi_{high}, lo_{low} {}
>+
>+constexpr uint128::uint128(int v)
>+    : hi_{v < 0 ? std::numeric_limits<uint64_t>::max() : 0},
>+      lo_{static_cast<uint64_t>(v)} {}
>+constexpr uint128::uint128(long v)  // NOLINT(runtime/int)
>+    : hi_{v < 0 ? std::numeric_limits<uint64_t>::max() : 0},
>+      lo_{static_cast<uint64_t>(v)} {}
>+constexpr uint128::uint128(long long v)  // NOLINT(runtime/int)
>+    : hi_{v < 0 ? std::numeric_limits<uint64_t>::max() : 0},
>+      lo_{static_cast<uint64_t>(v)} {}
>+
>+constexpr uint128::uint128(unsigned int v) : hi_{0}, lo_{v} {}
>+// NOLINTNEXTLINE(runtime/int)
>+constexpr uint128::uint128(unsigned long v) : hi_{0}, lo_{v} {}
>+// NOLINTNEXTLINE(runtime/int)
>+constexpr uint128::uint128(unsigned long long v) : hi_{0}, lo_{v} {}
>+
>+#ifdef ABSL_HAVE_INTRINSIC_INT128
>+constexpr uint128::uint128(__int128 v)
>+    : hi_{static_cast<uint64_t>(static_cast<unsigned __int128>(v) >> 64)},
>+      lo_{static_cast<uint64_t>(v & ~uint64_t{0})} {}
>+constexpr uint128::uint128(unsigned __int128 v)
>+    : hi_{static_cast<uint64_t>(v >> 64)},
>+      lo_{static_cast<uint64_t>(v & ~uint64_t{0})} {}
>+#endif  // ABSL_HAVE_INTRINSIC_INT128
>+
>+#else  // byte order
>+#error "Unsupported byte order: must be little-endian or big-endian."
>+#endif  // byte order
>+
>+// Conversion operators to integer types.
>+
>+constexpr uint128::operator bool() const { return lo_ || hi_; }
>+
>+constexpr uint128::operator char() const { return static_cast<char>(lo_); }
>+
>+constexpr uint128::operator signed char() const {
>+  return static_cast<signed char>(lo_);
>+}
>+
>+constexpr uint128::operator unsigned char() const {
>+  return static_cast<unsigned char>(lo_);
>+}
>+
>+constexpr uint128::operator char16_t() const {
>+  return static_cast<char16_t>(lo_);
>+}
>+
>+constexpr uint128::operator char32_t() const {
>+  return static_cast<char32_t>(lo_);
>+}
>+
>+constexpr uint128::operator wchar_t() const {
>+  return static_cast<wchar_t>(lo_);
>+}
>+
>+// NOLINTNEXTLINE(runtime/int)
>+constexpr uint128::operator short() const { return static_cast<short>(lo_); }
>+
>+constexpr uint128::operator unsigned short() const {  // NOLINT(runtime/int)
>+  return static_cast<unsigned short>(lo_);            // NOLINT(runtime/int)
>+}
>+
>+constexpr uint128::operator int() const { return static_cast<int>(lo_); }
>+
>+constexpr uint128::operator unsigned int() const {
>+  return static_cast<unsigned int>(lo_);
>+}
>+
>+// NOLINTNEXTLINE(runtime/int)
>+constexpr uint128::operator long() const { return static_cast<long>(lo_); }
>+
>+constexpr uint128::operator unsigned long() const {  // NOLINT(runtime/int)
>+  return static_cast<unsigned long>(lo_);            // NOLINT(runtime/int)
>+}
>+
>+constexpr uint128::operator long long() const {  // NOLINT(runtime/int)
>+  return static_cast<long long>(lo_);            // NOLINT(runtime/int)
>+}
>+
>+constexpr uint128::operator unsigned long long() const {  // NOLINT(runtime/int)
>+  return static_cast<unsigned long long>(lo_);            // NOLINT(runtime/int)
>+}
>+
>+#ifdef ABSL_HAVE_INTRINSIC_INT128
>+constexpr uint128::operator __int128() const {
>+  return (static_cast<__int128>(hi_) << 64) + lo_;
>+}
>+
>+constexpr uint128::operator unsigned __int128() const {
>+  return (static_cast<unsigned __int128>(hi_) << 64) + lo_;
>+}
>+#endif  // ABSL_HAVE_INTRINSIC_INT128
>+
>+// Conversion operators to floating point types.
>+
>+inline uint128::operator float() const {
>+  return static_cast<float>(lo_) + std::ldexp(static_cast<float>(hi_), 64);
>+}
>+
>+inline uint128::operator double() const {
>+  return static_cast<double>(lo_) + std::ldexp(static_cast<double>(hi_), 64);
>+}
>+
>+inline uint128::operator long double() const {
>+  return static_cast<long double>(lo_) +
>+         std::ldexp(static_cast<long double>(hi_), 64);
>+}
>+
>+// Comparison operators.
>+
>+inline bool operator==(uint128 lhs, uint128 rhs) {
>+  return (Uint128Low64(lhs) == Uint128Low64(rhs) &&
>+          Uint128High64(lhs) == Uint128High64(rhs));
>+}
>+
>+inline bool operator!=(uint128 lhs, uint128 rhs) {
>+  return !(lhs == rhs);
>+}
>+
>+inline bool operator<(uint128 lhs, uint128 rhs) {
>+  return (Uint128High64(lhs) == Uint128High64(rhs))
>+             ? (Uint128Low64(lhs) < Uint128Low64(rhs))
>+             : (Uint128High64(lhs) < Uint128High64(rhs));
>+}
>+
>+inline bool operator>(uint128 lhs, uint128 rhs) {
>+  return (Uint128High64(lhs) == Uint128High64(rhs))
>+             ? (Uint128Low64(lhs) > Uint128Low64(rhs))
>+             : (Uint128High64(lhs) > Uint128High64(rhs));
>+}
>+
>+inline bool operator<=(uint128 lhs, uint128 rhs) {
>+  return (Uint128High64(lhs) == Uint128High64(rhs))
>+             ? (Uint128Low64(lhs) <= Uint128Low64(rhs))
>+             : (Uint128High64(lhs) <= Uint128High64(rhs));
>+}
>+
>+inline bool operator>=(uint128 lhs, uint128 rhs) {
>+  return (Uint128High64(lhs) == Uint128High64(rhs))
>+             ? (Uint128Low64(lhs) >= Uint128Low64(rhs))
>+             : (Uint128High64(lhs) >= Uint128High64(rhs));
>+}
>+
>+// Unary operators.
>+
>+inline uint128 operator-(uint128 val) {
>+  uint64_t hi = ~Uint128High64(val);
>+  uint64_t lo = ~Uint128Low64(val) + 1;
>+  if (lo == 0) ++hi;  // carry
>+  return MakeUint128(hi, lo);
>+}
>+
>+inline bool operator!(uint128 val) {
>+  return !Uint128High64(val) && !Uint128Low64(val);
>+}
>+
>+// Logical operators.
>+
>+inline uint128 operator~(uint128 val) {
>+  return MakeUint128(~Uint128High64(val), ~Uint128Low64(val));
>+}
>+
>+inline uint128 operator|(uint128 lhs, uint128 rhs) {
>+  return MakeUint128(Uint128High64(lhs) | Uint128High64(rhs),
>+                           Uint128Low64(lhs) | Uint128Low64(rhs));
>+}
>+
>+inline uint128 operator&(uint128 lhs, uint128 rhs) {
>+  return MakeUint128(Uint128High64(lhs) & Uint128High64(rhs),
>+                           Uint128Low64(lhs) & Uint128Low64(rhs));
>+}
>+
>+inline uint128 operator^(uint128 lhs, uint128 rhs) {
>+  return MakeUint128(Uint128High64(lhs) ^ Uint128High64(rhs),
>+                           Uint128Low64(lhs) ^ Uint128Low64(rhs));
>+}
>+
>+inline uint128& uint128::operator|=(uint128 other) {
>+  hi_ |= other.hi_;
>+  lo_ |= other.lo_;
>+  return *this;
>+}
>+
>+inline uint128& uint128::operator&=(uint128 other) {
>+  hi_ &= other.hi_;
>+  lo_ &= other.lo_;
>+  return *this;
>+}
>+
>+inline uint128& uint128::operator^=(uint128 other) {
>+  hi_ ^= other.hi_;
>+  lo_ ^= other.lo_;
>+  return *this;
>+}
>+
>+// Arithmetic operators.
>+
>+inline uint128 operator<<(uint128 lhs, int amount) {
>+  // uint64_t shifts of >= 64 are undefined, so we will need some
>+  // special-casing.
>+  if (amount < 64) {
>+    if (amount != 0) {
>+      return MakeUint128(
>+          (Uint128High64(lhs) << amount) | (Uint128Low64(lhs) >> (64 - amount)),
>+          Uint128Low64(lhs) << amount);
>+    }
>+    return lhs;
>+  }
>+  return MakeUint128(Uint128Low64(lhs) << (amount - 64), 0);
>+}
>+
>+inline uint128 operator>>(uint128 lhs, int amount) {
>+  // uint64_t shifts of >= 64 are undefined, so we will need some
>+  // special-casing.
>+  if (amount < 64) {
>+    if (amount != 0) {
>+      return MakeUint128(Uint128High64(lhs) >> amount,
>+                         (Uint128Low64(lhs) >> amount) |
>+                             (Uint128High64(lhs) << (64 - amount)));
>+    }
>+    return lhs;
>+  }
>+  return MakeUint128(0, Uint128High64(lhs) >> (amount - 64));
>+}
>+
>+inline uint128 operator+(uint128 lhs, uint128 rhs) {
>+  uint128 result = MakeUint128(Uint128High64(lhs) + Uint128High64(rhs),
>+                               Uint128Low64(lhs) + Uint128Low64(rhs));
>+  if (Uint128Low64(result) < Uint128Low64(lhs)) {  // check for carry
>+    return MakeUint128(Uint128High64(result) + 1, Uint128Low64(result));
>+  }
>+  return result;
>+}
>+
>+inline uint128 operator-(uint128 lhs, uint128 rhs) {
>+  uint128 result = MakeUint128(Uint128High64(lhs) - Uint128High64(rhs),
>+                               Uint128Low64(lhs) - Uint128Low64(rhs));
>+  if (Uint128Low64(lhs) < Uint128Low64(rhs)) {  // check for carry
>+    return MakeUint128(Uint128High64(result) - 1, Uint128Low64(result));
>+  }
>+  return result;
>+}
>+
>+inline uint128 operator*(uint128 lhs, uint128 rhs) {
>+#if defined(ABSL_HAVE_INTRINSIC_INT128)
>+  // TODO(strel) Remove once alignment issues are resolved and unsigned __int128
>+  // can be used for uint128 storage.
>+  return static_cast<unsigned __int128>(lhs) *
>+         static_cast<unsigned __int128>(rhs);
>+#else   // ABSL_HAVE_INTRINSIC128
>+  uint64_t a32 = Uint128Low64(lhs) >> 32;
>+  uint64_t a00 = Uint128Low64(lhs) & 0xffffffff;
>+  uint64_t b32 = Uint128Low64(rhs) >> 32;
>+  uint64_t b00 = Uint128Low64(rhs) & 0xffffffff;
>+  uint128 result =
>+      MakeUint128(Uint128High64(lhs) * Uint128Low64(rhs) +
>+                      Uint128Low64(lhs) * Uint128High64(rhs) + a32 * b32,
>+                  a00 * b00);
>+  result += uint128(a32 * b00) << 32;
>+  result += uint128(a00 * b32) << 32;
>+  return result;
>+#endif  // ABSL_HAVE_INTRINSIC128
>+}
>+
>+// Increment/decrement operators.
>+
>+inline uint128 uint128::operator++(int) {
>+  uint128 tmp(*this);
>+  *this += 1;
>+  return tmp;
>+}
>+
>+inline uint128 uint128::operator--(int) {
>+  uint128 tmp(*this);
>+  *this -= 1;
>+  return tmp;
>+}
>+
>+inline uint128& uint128::operator++() {
>+  *this += 1;
>+  return *this;
>+}
>+
>+inline uint128& uint128::operator--() {
>+  *this -= 1;
>+  return *this;
>+}
>+
>+#if defined(ABSL_HAVE_INTRINSIC_INT128)
>+#include "absl/numeric/int128_have_intrinsic.inc"
>+#else  // ABSL_HAVE_INTRINSIC_INT128
>+#include "absl/numeric/int128_no_intrinsic.inc"
>+#endif  // ABSL_HAVE_INTRINSIC_INT128
>+
>+}  // namespace absl
>+
>+#endif  // ABSL_NUMERIC_INT128_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/numeric/int128_benchmark.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/numeric/int128_benchmark.cc
>new file mode 100644
>index 00000000000..1cb7d0ed87e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/numeric/int128_benchmark.cc
>@@ -0,0 +1,221 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/numeric/int128.h"
>+
>+#include <algorithm>
>+#include <cstdint>
>+#include <random>
>+#include <vector>
>+
>+#include "benchmark/benchmark.h"
>+#include "absl/base/config.h"
>+
>+namespace {
>+
>+constexpr size_t kSampleSize = 1000000;
>+
>+std::mt19937 MakeRandomEngine() {
>+  std::random_device r;
>+  std::seed_seq seed({r(), r(), r(), r(), r(), r(), r(), r()});
>+  return std::mt19937(seed);
>+}
>+
>+std::vector<std::pair<absl::uint128, absl::uint128>>
>+GetRandomClass128SampleUniformDivisor() {
>+  std::vector<std::pair<absl::uint128, absl::uint128>> values;
>+  std::mt19937 random = MakeRandomEngine();
>+  std::uniform_int_distribution<uint64_t> uniform_uint64;
>+  values.reserve(kSampleSize);
>+  for (size_t i = 0; i < kSampleSize; ++i) {
>+    absl::uint128 a =
>+        absl::MakeUint128(uniform_uint64(random), uniform_uint64(random));
>+    absl::uint128 b =
>+        absl::MakeUint128(uniform_uint64(random), uniform_uint64(random));
>+    values.emplace_back(std::max(a, b),
>+                        std::max(absl::uint128(2), std::min(a, b)));
>+  }
>+  return values;
>+}
>+
>+void BM_DivideClass128UniformDivisor(benchmark::State& state) {
>+  auto values = GetRandomClass128SampleUniformDivisor();
>+  while (state.KeepRunningBatch(values.size())) {
>+    for (const auto& pair : values) {
>+      benchmark::DoNotOptimize(pair.first / pair.second);
>+    }
>+  }
>+}
>+BENCHMARK(BM_DivideClass128UniformDivisor);
>+
>+std::vector<std::pair<absl::uint128, uint64_t>>
>+GetRandomClass128SampleSmallDivisor() {
>+  std::vector<std::pair<absl::uint128, uint64_t>> values;
>+  std::mt19937 random = MakeRandomEngine();
>+  std::uniform_int_distribution<uint64_t> uniform_uint64;
>+  values.reserve(kSampleSize);
>+  for (size_t i = 0; i < kSampleSize; ++i) {
>+    absl::uint128 a =
>+        absl::MakeUint128(uniform_uint64(random), uniform_uint64(random));
>+    uint64_t b = std::max(uint64_t{2}, uniform_uint64(random));
>+    values.emplace_back(std::max(a, absl::uint128(b)), b);
>+  }
>+  return values;
>+}
>+
>+void BM_DivideClass128SmallDivisor(benchmark::State& state) {
>+  auto values = GetRandomClass128SampleSmallDivisor();
>+  while (state.KeepRunningBatch(values.size())) {
>+    for (const auto& pair : values) {
>+      benchmark::DoNotOptimize(pair.first / pair.second);
>+    }
>+  }
>+}
>+BENCHMARK(BM_DivideClass128SmallDivisor);
>+
>+std::vector<std::pair<absl::uint128, absl::uint128>> GetRandomClass128Sample() {
>+  std::vector<std::pair<absl::uint128, absl::uint128>> values;
>+  std::mt19937 random = MakeRandomEngine();
>+  std::uniform_int_distribution<uint64_t> uniform_uint64;
>+  values.reserve(kSampleSize);
>+  for (size_t i = 0; i < kSampleSize; ++i) {
>+    values.emplace_back(
>+        absl::MakeUint128(uniform_uint64(random), uniform_uint64(random)),
>+        absl::MakeUint128(uniform_uint64(random), uniform_uint64(random)));
>+  }
>+  return values;
>+}
>+
>+void BM_MultiplyClass128(benchmark::State& state) {
>+  auto values = GetRandomClass128Sample();
>+  while (state.KeepRunningBatch(values.size())) {
>+    for (const auto& pair : values) {
>+      benchmark::DoNotOptimize(pair.first * pair.second);
>+    }
>+  }
>+}
>+BENCHMARK(BM_MultiplyClass128);
>+
>+void BM_AddClass128(benchmark::State& state) {
>+  auto values = GetRandomClass128Sample();
>+  while (state.KeepRunningBatch(values.size())) {
>+    for (const auto& pair : values) {
>+      benchmark::DoNotOptimize(pair.first + pair.second);
>+    }
>+  }
>+}
>+BENCHMARK(BM_AddClass128);
>+
>+#ifdef ABSL_HAVE_INTRINSIC_INT128
>+
>+// Some implementations of <random> do not support __int128 when it is
>+// available, so we make our own uniform_int_distribution-like type.
>+class UniformIntDistribution128 {
>+ public:
>+  // NOLINTNEXTLINE: mimicking std::uniform_int_distribution API
>+  unsigned __int128 operator()(std::mt19937& generator) {
>+    return (static_cast<unsigned __int128>(dist64_(generator)) << 64) |
>+           dist64_(generator);
>+  }
>+
>+ private:
>+  std::uniform_int_distribution<uint64_t> dist64_;
>+};
>+
>+std::vector<std::pair<unsigned __int128, unsigned __int128>>
>+GetRandomIntrinsic128SampleUniformDivisor() {
>+  std::vector<std::pair<unsigned __int128, unsigned __int128>> values;
>+  std::mt19937 random = MakeRandomEngine();
>+  UniformIntDistribution128 uniform_uint128;
>+  values.reserve(kSampleSize);
>+  for (size_t i = 0; i < kSampleSize; ++i) {
>+    unsigned __int128 a = uniform_uint128(random);
>+    unsigned __int128 b = uniform_uint128(random);
>+    values.emplace_back(
>+        std::max(a, b),
>+        std::max(static_cast<unsigned __int128>(2), std::min(a, b)));
>+  }
>+  return values;
>+}
>+
>+void BM_DivideIntrinsic128UniformDivisor(benchmark::State& state) {
>+  auto values = GetRandomIntrinsic128SampleUniformDivisor();
>+  while (state.KeepRunningBatch(values.size())) {
>+    for (const auto& pair : values) {
>+      benchmark::DoNotOptimize(pair.first / pair.second);
>+    }
>+  }
>+}
>+BENCHMARK(BM_DivideIntrinsic128UniformDivisor);
>+
>+std::vector<std::pair<unsigned __int128, uint64_t>>
>+GetRandomIntrinsic128SampleSmallDivisor() {
>+  std::vector<std::pair<unsigned __int128, uint64_t>> values;
>+  std::mt19937 random = MakeRandomEngine();
>+  UniformIntDistribution128 uniform_uint128;
>+  std::uniform_int_distribution<uint64_t> uniform_uint64;
>+  values.reserve(kSampleSize);
>+  for (size_t i = 0; i < kSampleSize; ++i) {
>+    unsigned __int128 a = uniform_uint128(random);
>+    uint64_t b = std::max(uint64_t{2}, uniform_uint64(random));
>+    values.emplace_back(std::max(a, static_cast<unsigned __int128>(b)), b);
>+  }
>+  return values;
>+}
>+
>+void BM_DivideIntrinsic128SmallDivisor(benchmark::State& state) {
>+  auto values = GetRandomIntrinsic128SampleSmallDivisor();
>+  while (state.KeepRunningBatch(values.size())) {
>+    for (const auto& pair : values) {
>+      benchmark::DoNotOptimize(pair.first / pair.second);
>+    }
>+  }
>+}
>+BENCHMARK(BM_DivideIntrinsic128SmallDivisor);
>+
>+std::vector<std::pair<unsigned __int128, unsigned __int128>>
>+      GetRandomIntrinsic128Sample() {
>+  std::vector<std::pair<unsigned __int128, unsigned __int128>> values;
>+  std::mt19937 random = MakeRandomEngine();
>+  UniformIntDistribution128 uniform_uint128;
>+  values.reserve(kSampleSize);
>+  for (size_t i = 0; i < kSampleSize; ++i) {
>+    values.emplace_back(uniform_uint128(random), uniform_uint128(random));
>+  }
>+  return values;
>+}
>+
>+void BM_MultiplyIntrinsic128(benchmark::State& state) {
>+  auto values = GetRandomIntrinsic128Sample();
>+  while (state.KeepRunningBatch(values.size())) {
>+    for (const auto& pair : values) {
>+      benchmark::DoNotOptimize(pair.first * pair.second);
>+    }
>+  }
>+}
>+BENCHMARK(BM_MultiplyIntrinsic128);
>+
>+void BM_AddIntrinsic128(benchmark::State& state) {
>+  auto values = GetRandomIntrinsic128Sample();
>+  while (state.KeepRunningBatch(values.size())) {
>+    for (const auto& pair : values) {
>+      benchmark::DoNotOptimize(pair.first + pair.second);
>+    }
>+  }
>+}
>+BENCHMARK(BM_AddIntrinsic128);
>+
>+#endif  // ABSL_HAVE_INTRINSIC_INT128
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/numeric/int128_have_intrinsic.inc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/numeric/int128_have_intrinsic.inc
>new file mode 100644
>index 00000000000..ee2a093018d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/numeric/int128_have_intrinsic.inc
>@@ -0,0 +1,18 @@
>+//
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+// This file contains :int128 implementation details that depend on internal
>+// representation when ABSL_HAVE_INTRINSIC_INT128 is defined. This file is
>+// included by int128.h.
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/numeric/int128_no_intrinsic.inc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/numeric/int128_no_intrinsic.inc
>new file mode 100644
>index 00000000000..0d0b3cfdeb1
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/numeric/int128_no_intrinsic.inc
>@@ -0,0 +1,18 @@
>+//
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+// This file contains :int128 implementation details that depend on internal
>+// representation when ABSL_HAVE_INTRINSIC_INT128 is *not* defined. This file
>+// is included by int128.h.
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/numeric/int128_stream_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/numeric/int128_stream_test.cc
>new file mode 100644
>index 00000000000..09efaad4e7f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/numeric/int128_stream_test.cc
>@@ -0,0 +1,666 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/numeric/int128.h"
>+
>+#include <sstream>
>+#include <string>
>+
>+#include "gtest/gtest.h"
>+
>+namespace {
>+
>+struct Uint128TestCase {
>+  absl::uint128 value;
>+  std::ios_base::fmtflags flags;
>+  std::streamsize width;
>+  const char* expected;
>+};
>+
>+constexpr char kFill = '_';
>+
>+std::string StreamFormatToString(std::ios_base::fmtflags flags,
>+                                 std::streamsize width) {
>+  std::vector<const char*> flagstr;
>+  switch (flags & std::ios::basefield) {
>+    case std::ios::dec:
>+      flagstr.push_back("std::ios::dec");
>+      break;
>+    case std::ios::oct:
>+      flagstr.push_back("std::ios::oct");
>+      break;
>+    case std::ios::hex:
>+      flagstr.push_back("std::ios::hex");
>+      break;
>+    default:  // basefield not specified
>+      break;
>+  }
>+  switch (flags & std::ios::adjustfield) {
>+    case std::ios::left:
>+      flagstr.push_back("std::ios::left");
>+      break;
>+    case std::ios::internal:
>+      flagstr.push_back("std::ios::internal");
>+      break;
>+    case std::ios::right:
>+      flagstr.push_back("std::ios::right");
>+      break;
>+    default:  // adjustfield not specified
>+      break;
>+  }
>+  if (flags & std::ios::uppercase) flagstr.push_back("std::ios::uppercase");
>+  if (flags & std::ios::showbase) flagstr.push_back("std::ios::showbase");
>+  if (flags & std::ios::showpos) flagstr.push_back("std::ios::showpos");
>+
>+  std::ostringstream msg;
>+  msg << "\n  StreamFormatToString(test_case.flags, test_case.width)\n    "
>+         "flags: ";
>+  if (!flagstr.empty()) {
>+    for (size_t i = 0; i < flagstr.size() - 1; ++i) msg << flagstr[i] << " | ";
>+    msg << flagstr.back();
>+  } else {
>+    msg << "(default)";
>+  }
>+  msg << "\n    width: " << width << "\n    fill: '" << kFill << "'";
>+  return msg.str();
>+}
>+
>+void CheckUint128Case(const Uint128TestCase& test_case) {
>+  std::ostringstream os;
>+  os.flags(test_case.flags);
>+  os.width(test_case.width);
>+  os.fill(kFill);
>+  os << test_case.value;
>+  SCOPED_TRACE(StreamFormatToString(test_case.flags, test_case.width));
>+  EXPECT_EQ(test_case.expected, os.str());
>+}
>+
>+constexpr std::ios::fmtflags kDec = std::ios::dec;
>+constexpr std::ios::fmtflags kOct = std::ios::oct;
>+constexpr std::ios::fmtflags kHex = std::ios::hex;
>+constexpr std::ios::fmtflags kLeft = std::ios::left;
>+constexpr std::ios::fmtflags kInt = std::ios::internal;
>+constexpr std::ios::fmtflags kRight = std::ios::right;
>+constexpr std::ios::fmtflags kUpper = std::ios::uppercase;
>+constexpr std::ios::fmtflags kBase = std::ios::showbase;
>+constexpr std::ios::fmtflags kPos = std::ios::showpos;
>+
>+TEST(Uint128, OStreamValueTest) {
>+  CheckUint128Case({1, kDec, /*width = */ 0, "1"});
>+  CheckUint128Case({1, kOct, /*width = */ 0, "1"});
>+  CheckUint128Case({1, kHex, /*width = */ 0, "1"});
>+  CheckUint128Case({9, kDec, /*width = */ 0, "9"});
>+  CheckUint128Case({9, kOct, /*width = */ 0, "11"});
>+  CheckUint128Case({9, kHex, /*width = */ 0, "9"});
>+  CheckUint128Case({12345, kDec, /*width = */ 0, "12345"});
>+  CheckUint128Case({12345, kOct, /*width = */ 0, "30071"});
>+  CheckUint128Case({12345, kHex, /*width = */ 0, "3039"});
>+  CheckUint128Case(
>+      {0x8000000000000000, kDec, /*width = */ 0, "9223372036854775808"});
>+  CheckUint128Case(
>+      {0x8000000000000000, kOct, /*width = */ 0, "1000000000000000000000"});
>+  CheckUint128Case(
>+      {0x8000000000000000, kHex, /*width = */ 0, "8000000000000000"});
>+  CheckUint128Case({std::numeric_limits<uint64_t>::max(), kDec,
>+                    /*width = */ 0, "18446744073709551615"});
>+  CheckUint128Case({std::numeric_limits<uint64_t>::max(), kOct,
>+                    /*width = */ 0, "1777777777777777777777"});
>+  CheckUint128Case({std::numeric_limits<uint64_t>::max(), kHex,
>+                    /*width = */ 0, "ffffffffffffffff"});
>+  CheckUint128Case(
>+      {absl::MakeUint128(1, 0), kDec, /*width = */ 0, "18446744073709551616"});
>+  CheckUint128Case({absl::MakeUint128(1, 0), kOct, /*width = */ 0,
>+                    "2000000000000000000000"});
>+  CheckUint128Case(
>+      {absl::MakeUint128(1, 0), kHex, /*width = */ 0, "10000000000000000"});
>+  CheckUint128Case({absl::MakeUint128(0x8000000000000000, 0), kDec,
>+                    /*width = */ 0, "170141183460469231731687303715884105728"});
>+  CheckUint128Case({absl::MakeUint128(0x8000000000000000, 0), kOct,
>+                    /*width = */ 0,
>+                    "2000000000000000000000000000000000000000000"});
>+  CheckUint128Case({absl::MakeUint128(0x8000000000000000, 0), kHex,
>+                    /*width = */ 0, "80000000000000000000000000000000"});
>+  CheckUint128Case({absl::kuint128max, kDec, /*width = */ 0,
>+                    "340282366920938463463374607431768211455"});
>+  CheckUint128Case({absl::kuint128max, kOct, /*width = */ 0,
>+                    "3777777777777777777777777777777777777777777"});
>+  CheckUint128Case({absl::kuint128max, kHex, /*width = */ 0,
>+                    "ffffffffffffffffffffffffffffffff"});
>+}
>+
>+std::vector<Uint128TestCase> GetUint128FormatCases();
>+
>+TEST(Uint128, OStreamFormatTest) {
>+  for (const Uint128TestCase& test_case : GetUint128FormatCases()) {
>+    CheckUint128Case(test_case);
>+  }
>+}
>+
>+std::vector<Uint128TestCase> GetUint128FormatCases() {
>+  return {
>+      {0, std::ios_base::fmtflags(), /*width = */ 0, "0"},
>+      {0, std::ios_base::fmtflags(), /*width = */ 6, "_____0"},
>+      {0, kPos, /*width = */ 0, "0"},
>+      {0, kPos, /*width = */ 6, "_____0"},
>+      {0, kBase, /*width = */ 0, "0"},
>+      {0, kBase, /*width = */ 6, "_____0"},
>+      {0, kBase | kPos, /*width = */ 0, "0"},
>+      {0, kBase | kPos, /*width = */ 6, "_____0"},
>+      {0, kUpper, /*width = */ 0, "0"},
>+      {0, kUpper, /*width = */ 6, "_____0"},
>+      {0, kUpper | kPos, /*width = */ 0, "0"},
>+      {0, kUpper | kPos, /*width = */ 6, "_____0"},
>+      {0, kUpper | kBase, /*width = */ 0, "0"},
>+      {0, kUpper | kBase, /*width = */ 6, "_____0"},
>+      {0, kUpper | kBase | kPos, /*width = */ 0, "0"},
>+      {0, kUpper | kBase | kPos, /*width = */ 6, "_____0"},
>+      {0, kLeft, /*width = */ 0, "0"},
>+      {0, kLeft, /*width = */ 6, "0_____"},
>+      {0, kLeft | kPos, /*width = */ 0, "0"},
>+      {0, kLeft | kPos, /*width = */ 6, "0_____"},
>+      {0, kLeft | kBase, /*width = */ 0, "0"},
>+      {0, kLeft | kBase, /*width = */ 6, "0_____"},
>+      {0, kLeft | kBase | kPos, /*width = */ 0, "0"},
>+      {0, kLeft | kBase | kPos, /*width = */ 6, "0_____"},
>+      {0, kLeft | kUpper, /*width = */ 0, "0"},
>+      {0, kLeft | kUpper, /*width = */ 6, "0_____"},
>+      {0, kLeft | kUpper | kPos, /*width = */ 0, "0"},
>+      {0, kLeft | kUpper | kPos, /*width = */ 6, "0_____"},
>+      {0, kLeft | kUpper | kBase, /*width = */ 0, "0"},
>+      {0, kLeft | kUpper | kBase, /*width = */ 6, "0_____"},
>+      {0, kLeft | kUpper | kBase | kPos, /*width = */ 0, "0"},
>+      {0, kLeft | kUpper | kBase | kPos, /*width = */ 6, "0_____"},
>+      {0, kInt, /*width = */ 0, "0"},
>+      {0, kInt, /*width = */ 6, "_____0"},
>+      {0, kInt | kPos, /*width = */ 0, "0"},
>+      {0, kInt | kPos, /*width = */ 6, "_____0"},
>+      {0, kInt | kBase, /*width = */ 0, "0"},
>+      {0, kInt | kBase, /*width = */ 6, "_____0"},
>+      {0, kInt | kBase | kPos, /*width = */ 0, "0"},
>+      {0, kInt | kBase | kPos, /*width = */ 6, "_____0"},
>+      {0, kInt | kUpper, /*width = */ 0, "0"},
>+      {0, kInt | kUpper, /*width = */ 6, "_____0"},
>+      {0, kInt | kUpper | kPos, /*width = */ 0, "0"},
>+      {0, kInt | kUpper | kPos, /*width = */ 6, "_____0"},
>+      {0, kInt | kUpper | kBase, /*width = */ 0, "0"},
>+      {0, kInt | kUpper | kBase, /*width = */ 6, "_____0"},
>+      {0, kInt | kUpper | kBase | kPos, /*width = */ 0, "0"},
>+      {0, kInt | kUpper | kBase | kPos, /*width = */ 6, "_____0"},
>+      {0, kRight, /*width = */ 0, "0"},
>+      {0, kRight, /*width = */ 6, "_____0"},
>+      {0, kRight | kPos, /*width = */ 0, "0"},
>+      {0, kRight | kPos, /*width = */ 6, "_____0"},
>+      {0, kRight | kBase, /*width = */ 0, "0"},
>+      {0, kRight | kBase, /*width = */ 6, "_____0"},
>+      {0, kRight | kBase | kPos, /*width = */ 0, "0"},
>+      {0, kRight | kBase | kPos, /*width = */ 6, "_____0"},
>+      {0, kRight | kUpper, /*width = */ 0, "0"},
>+      {0, kRight | kUpper, /*width = */ 6, "_____0"},
>+      {0, kRight | kUpper | kPos, /*width = */ 0, "0"},
>+      {0, kRight | kUpper | kPos, /*width = */ 6, "_____0"},
>+      {0, kRight | kUpper | kBase, /*width = */ 0, "0"},
>+      {0, kRight | kUpper | kBase, /*width = */ 6, "_____0"},
>+      {0, kRight | kUpper | kBase | kPos, /*width = */ 0, "0"},
>+      {0, kRight | kUpper | kBase | kPos, /*width = */ 6, "_____0"},
>+      {0, kDec, /*width = */ 0, "0"},
>+      {0, kDec, /*width = */ 6, "_____0"},
>+      {0, kDec | kPos, /*width = */ 0, "0"},
>+      {0, kDec | kPos, /*width = */ 6, "_____0"},
>+      {0, kDec | kBase, /*width = */ 0, "0"},
>+      {0, kDec | kBase, /*width = */ 6, "_____0"},
>+      {0, kDec | kBase | kPos, /*width = */ 0, "0"},
>+      {0, kDec | kBase | kPos, /*width = */ 6, "_____0"},
>+      {0, kDec | kUpper, /*width = */ 0, "0"},
>+      {0, kDec | kUpper, /*width = */ 6, "_____0"},
>+      {0, kDec | kUpper | kPos, /*width = */ 0, "0"},
>+      {0, kDec | kUpper | kPos, /*width = */ 6, "_____0"},
>+      {0, kDec | kUpper | kBase, /*width = */ 0, "0"},
>+      {0, kDec | kUpper | kBase, /*width = */ 6, "_____0"},
>+      {0, kDec | kUpper | kBase | kPos, /*width = */ 0, "0"},
>+      {0, kDec | kUpper | kBase | kPos, /*width = */ 6, "_____0"},
>+      {0, kDec | kLeft, /*width = */ 0, "0"},
>+      {0, kDec | kLeft, /*width = */ 6, "0_____"},
>+      {0, kDec | kLeft | kPos, /*width = */ 0, "0"},
>+      {0, kDec | kLeft | kPos, /*width = */ 6, "0_____"},
>+      {0, kDec | kLeft | kBase, /*width = */ 0, "0"},
>+      {0, kDec | kLeft | kBase, /*width = */ 6, "0_____"},
>+      {0, kDec | kLeft | kBase | kPos, /*width = */ 0, "0"},
>+      {0, kDec | kLeft | kBase | kPos, /*width = */ 6, "0_____"},
>+      {0, kDec | kLeft | kUpper, /*width = */ 0, "0"},
>+      {0, kDec | kLeft | kUpper, /*width = */ 6, "0_____"},
>+      {0, kDec | kLeft | kUpper | kPos, /*width = */ 0, "0"},
>+      {0, kDec | kLeft | kUpper | kPos, /*width = */ 6, "0_____"},
>+      {0, kDec | kLeft | kUpper | kBase, /*width = */ 0, "0"},
>+      {0, kDec | kLeft | kUpper | kBase, /*width = */ 6, "0_____"},
>+      {0, kDec | kLeft | kUpper | kBase | kPos, /*width = */ 0, "0"},
>+      {0, kDec | kLeft | kUpper | kBase | kPos, /*width = */ 6, "0_____"},
>+      {0, kDec | kInt, /*width = */ 0, "0"},
>+      {0, kDec | kInt, /*width = */ 6, "_____0"},
>+      {0, kDec | kInt | kPos, /*width = */ 0, "0"},
>+      {0, kDec | kInt | kPos, /*width = */ 6, "_____0"},
>+      {0, kDec | kInt | kBase, /*width = */ 0, "0"},
>+      {0, kDec | kInt | kBase, /*width = */ 6, "_____0"},
>+      {0, kDec | kInt | kBase | kPos, /*width = */ 0, "0"},
>+      {0, kDec | kInt | kBase | kPos, /*width = */ 6, "_____0"},
>+      {0, kDec | kInt | kUpper, /*width = */ 0, "0"},
>+      {0, kDec | kInt | kUpper, /*width = */ 6, "_____0"},
>+      {0, kDec | kInt | kUpper | kPos, /*width = */ 0, "0"},
>+      {0, kDec | kInt | kUpper | kPos, /*width = */ 6, "_____0"},
>+      {0, kDec | kInt | kUpper | kBase, /*width = */ 0, "0"},
>+      {0, kDec | kInt | kUpper | kBase, /*width = */ 6, "_____0"},
>+      {0, kDec | kInt | kUpper | kBase | kPos, /*width = */ 0, "0"},
>+      {0, kDec | kInt | kUpper | kBase | kPos, /*width = */ 6, "_____0"},
>+      {0, kDec | kRight, /*width = */ 0, "0"},
>+      {0, kDec | kRight, /*width = */ 6, "_____0"},
>+      {0, kDec | kRight | kPos, /*width = */ 0, "0"},
>+      {0, kDec | kRight | kPos, /*width = */ 6, "_____0"},
>+      {0, kDec | kRight | kBase, /*width = */ 0, "0"},
>+      {0, kDec | kRight | kBase, /*width = */ 6, "_____0"},
>+      {0, kDec | kRight | kBase | kPos, /*width = */ 0, "0"},
>+      {0, kDec | kRight | kBase | kPos, /*width = */ 6, "_____0"},
>+      {0, kDec | kRight | kUpper, /*width = */ 0, "0"},
>+      {0, kDec | kRight | kUpper, /*width = */ 6, "_____0"},
>+      {0, kDec | kRight | kUpper | kPos, /*width = */ 0, "0"},
>+      {0, kDec | kRight | kUpper | kPos, /*width = */ 6, "_____0"},
>+      {0, kDec | kRight | kUpper | kBase, /*width = */ 0, "0"},
>+      {0, kDec | kRight | kUpper | kBase, /*width = */ 6, "_____0"},
>+      {0, kDec | kRight | kUpper | kBase | kPos, /*width = */ 0, "0"},
>+      {0, kDec | kRight | kUpper | kBase | kPos, /*width = */ 6, "_____0"},
>+      {0, kOct, /*width = */ 0, "0"},
>+      {0, kOct, /*width = */ 6, "_____0"},
>+      {0, kOct | kPos, /*width = */ 0, "0"},
>+      {0, kOct | kPos, /*width = */ 6, "_____0"},
>+      {0, kOct | kBase, /*width = */ 0, "0"},
>+      {0, kOct | kBase, /*width = */ 6, "_____0"},
>+      {0, kOct | kBase | kPos, /*width = */ 0, "0"},
>+      {0, kOct | kBase | kPos, /*width = */ 6, "_____0"},
>+      {0, kOct | kUpper, /*width = */ 0, "0"},
>+      {0, kOct | kUpper, /*width = */ 6, "_____0"},
>+      {0, kOct | kUpper | kPos, /*width = */ 0, "0"},
>+      {0, kOct | kUpper | kPos, /*width = */ 6, "_____0"},
>+      {0, kOct | kUpper | kBase, /*width = */ 0, "0"},
>+      {0, kOct | kUpper | kBase, /*width = */ 6, "_____0"},
>+      {0, kOct | kUpper | kBase | kPos, /*width = */ 0, "0"},
>+      {0, kOct | kUpper | kBase | kPos, /*width = */ 6, "_____0"},
>+      {0, kOct | kLeft, /*width = */ 0, "0"},
>+      {0, kOct | kLeft, /*width = */ 6, "0_____"},
>+      {0, kOct | kLeft | kPos, /*width = */ 0, "0"},
>+      {0, kOct | kLeft | kPos, /*width = */ 6, "0_____"},
>+      {0, kOct | kLeft | kBase, /*width = */ 0, "0"},
>+      {0, kOct | kLeft | kBase, /*width = */ 6, "0_____"},
>+      {0, kOct | kLeft | kBase | kPos, /*width = */ 0, "0"},
>+      {0, kOct | kLeft | kBase | kPos, /*width = */ 6, "0_____"},
>+      {0, kOct | kLeft | kUpper, /*width = */ 0, "0"},
>+      {0, kOct | kLeft | kUpper, /*width = */ 6, "0_____"},
>+      {0, kOct | kLeft | kUpper | kPos, /*width = */ 0, "0"},
>+      {0, kOct | kLeft | kUpper | kPos, /*width = */ 6, "0_____"},
>+      {0, kOct | kLeft | kUpper | kBase, /*width = */ 0, "0"},
>+      {0, kOct | kLeft | kUpper | kBase, /*width = */ 6, "0_____"},
>+      {0, kOct | kLeft | kUpper | kBase | kPos, /*width = */ 0, "0"},
>+      {0, kOct | kLeft | kUpper | kBase | kPos, /*width = */ 6, "0_____"},
>+      {0, kOct | kInt, /*width = */ 0, "0"},
>+      {0, kOct | kInt, /*width = */ 6, "_____0"},
>+      {0, kOct | kInt | kPos, /*width = */ 0, "0"},
>+      {0, kOct | kInt | kPos, /*width = */ 6, "_____0"},
>+      {0, kOct | kInt | kBase, /*width = */ 0, "0"},
>+      {0, kOct | kInt | kBase, /*width = */ 6, "_____0"},
>+      {0, kOct | kInt | kBase | kPos, /*width = */ 0, "0"},
>+      {0, kOct | kInt | kBase | kPos, /*width = */ 6, "_____0"},
>+      {0, kOct | kInt | kUpper, /*width = */ 0, "0"},
>+      {0, kOct | kInt | kUpper, /*width = */ 6, "_____0"},
>+      {0, kOct | kInt | kUpper | kPos, /*width = */ 0, "0"},
>+      {0, kOct | kInt | kUpper | kPos, /*width = */ 6, "_____0"},
>+      {0, kOct | kInt | kUpper | kBase, /*width = */ 0, "0"},
>+      {0, kOct | kInt | kUpper | kBase, /*width = */ 6, "_____0"},
>+      {0, kOct | kInt | kUpper | kBase | kPos, /*width = */ 0, "0"},
>+      {0, kOct | kInt | kUpper | kBase | kPos, /*width = */ 6, "_____0"},
>+      {0, kOct | kRight, /*width = */ 0, "0"},
>+      {0, kOct | kRight, /*width = */ 6, "_____0"},
>+      {0, kOct | kRight | kPos, /*width = */ 0, "0"},
>+      {0, kOct | kRight | kPos, /*width = */ 6, "_____0"},
>+      {0, kOct | kRight | kBase, /*width = */ 0, "0"},
>+      {0, kOct | kRight | kBase, /*width = */ 6, "_____0"},
>+      {0, kOct | kRight | kBase | kPos, /*width = */ 0, "0"},
>+      {0, kOct | kRight | kBase | kPos, /*width = */ 6, "_____0"},
>+      {0, kOct | kRight | kUpper, /*width = */ 0, "0"},
>+      {0, kOct | kRight | kUpper, /*width = */ 6, "_____0"},
>+      {0, kOct | kRight | kUpper | kPos, /*width = */ 0, "0"},
>+      {0, kOct | kRight | kUpper | kPos, /*width = */ 6, "_____0"},
>+      {0, kOct | kRight | kUpper | kBase, /*width = */ 0, "0"},
>+      {0, kOct | kRight | kUpper | kBase, /*width = */ 6, "_____0"},
>+      {0, kOct | kRight | kUpper | kBase | kPos, /*width = */ 0, "0"},
>+      {0, kOct | kRight | kUpper | kBase | kPos, /*width = */ 6, "_____0"},
>+      {0, kHex, /*width = */ 0, "0"},
>+      {0, kHex, /*width = */ 6, "_____0"},
>+      {0, kHex | kPos, /*width = */ 0, "0"},
>+      {0, kHex | kPos, /*width = */ 6, "_____0"},
>+      {0, kHex | kBase, /*width = */ 0, "0"},
>+      {0, kHex | kBase, /*width = */ 6, "_____0"},
>+      {0, kHex | kBase | kPos, /*width = */ 0, "0"},
>+      {0, kHex | kBase | kPos, /*width = */ 6, "_____0"},
>+      {0, kHex | kUpper, /*width = */ 0, "0"},
>+      {0, kHex | kUpper, /*width = */ 6, "_____0"},
>+      {0, kHex | kUpper | kPos, /*width = */ 0, "0"},
>+      {0, kHex | kUpper | kPos, /*width = */ 6, "_____0"},
>+      {0, kHex | kUpper | kBase, /*width = */ 0, "0"},
>+      {0, kHex | kUpper | kBase, /*width = */ 6, "_____0"},
>+      {0, kHex | kUpper | kBase | kPos, /*width = */ 0, "0"},
>+      {0, kHex | kUpper | kBase | kPos, /*width = */ 6, "_____0"},
>+      {0, kHex | kLeft, /*width = */ 0, "0"},
>+      {0, kHex | kLeft, /*width = */ 6, "0_____"},
>+      {0, kHex | kLeft | kPos, /*width = */ 0, "0"},
>+      {0, kHex | kLeft | kPos, /*width = */ 6, "0_____"},
>+      {0, kHex | kLeft | kBase, /*width = */ 0, "0"},
>+      {0, kHex | kLeft | kBase, /*width = */ 6, "0_____"},
>+      {0, kHex | kLeft | kBase | kPos, /*width = */ 0, "0"},
>+      {0, kHex | kLeft | kBase | kPos, /*width = */ 6, "0_____"},
>+      {0, kHex | kLeft | kUpper, /*width = */ 0, "0"},
>+      {0, kHex | kLeft | kUpper, /*width = */ 6, "0_____"},
>+      {0, kHex | kLeft | kUpper | kPos, /*width = */ 0, "0"},
>+      {0, kHex | kLeft | kUpper | kPos, /*width = */ 6, "0_____"},
>+      {0, kHex | kLeft | kUpper | kBase, /*width = */ 0, "0"},
>+      {0, kHex | kLeft | kUpper | kBase, /*width = */ 6, "0_____"},
>+      {0, kHex | kLeft | kUpper | kBase | kPos, /*width = */ 0, "0"},
>+      {0, kHex | kLeft | kUpper | kBase | kPos, /*width = */ 6, "0_____"},
>+      {0, kHex | kInt, /*width = */ 0, "0"},
>+      {0, kHex | kInt, /*width = */ 6, "_____0"},
>+      {0, kHex | kInt | kPos, /*width = */ 0, "0"},
>+      {0, kHex | kInt | kPos, /*width = */ 6, "_____0"},
>+      {0, kHex | kInt | kBase, /*width = */ 0, "0"},
>+      {0, kHex | kInt | kBase, /*width = */ 6, "_____0"},
>+      {0, kHex | kInt | kBase | kPos, /*width = */ 0, "0"},
>+      {0, kHex | kInt | kBase | kPos, /*width = */ 6, "_____0"},
>+      {0, kHex | kInt | kUpper, /*width = */ 0, "0"},
>+      {0, kHex | kInt | kUpper, /*width = */ 6, "_____0"},
>+      {0, kHex | kInt | kUpper | kPos, /*width = */ 0, "0"},
>+      {0, kHex | kInt | kUpper | kPos, /*width = */ 6, "_____0"},
>+      {0, kHex | kInt | kUpper | kBase, /*width = */ 0, "0"},
>+      {0, kHex | kInt | kUpper | kBase, /*width = */ 6, "_____0"},
>+      {0, kHex | kInt | kUpper | kBase | kPos, /*width = */ 0, "0"},
>+      {0, kHex | kInt | kUpper | kBase | kPos, /*width = */ 6, "_____0"},
>+      {0, kHex | kRight, /*width = */ 0, "0"},
>+      {0, kHex | kRight, /*width = */ 6, "_____0"},
>+      {0, kHex | kRight | kPos, /*width = */ 0, "0"},
>+      {0, kHex | kRight | kPos, /*width = */ 6, "_____0"},
>+      {0, kHex | kRight | kBase, /*width = */ 0, "0"},
>+      {0, kHex | kRight | kBase, /*width = */ 6, "_____0"},
>+      {0, kHex | kRight | kBase | kPos, /*width = */ 0, "0"},
>+      {0, kHex | kRight | kBase | kPos, /*width = */ 6, "_____0"},
>+      {0, kHex | kRight | kUpper, /*width = */ 0, "0"},
>+      {0, kHex | kRight | kUpper, /*width = */ 6, "_____0"},
>+      {0, kHex | kRight | kUpper | kPos, /*width = */ 0, "0"},
>+      {0, kHex | kRight | kUpper | kPos, /*width = */ 6, "_____0"},
>+      {0, kHex | kRight | kUpper | kBase, /*width = */ 0, "0"},
>+      {0, kHex | kRight | kUpper | kBase, /*width = */ 6, "_____0"},
>+      {0, kHex | kRight | kUpper | kBase | kPos, /*width = */ 0, "0"},
>+      {0, kHex | kRight | kUpper | kBase | kPos, /*width = */ 6, "_____0"},
>+      {37, std::ios_base::fmtflags(), /*width = */ 0, "37"},
>+      {37, std::ios_base::fmtflags(), /*width = */ 6, "____37"},
>+      {37, kPos, /*width = */ 0, "37"},
>+      {37, kPos, /*width = */ 6, "____37"},
>+      {37, kBase, /*width = */ 0, "37"},
>+      {37, kBase, /*width = */ 6, "____37"},
>+      {37, kBase | kPos, /*width = */ 0, "37"},
>+      {37, kBase | kPos, /*width = */ 6, "____37"},
>+      {37, kUpper, /*width = */ 0, "37"},
>+      {37, kUpper, /*width = */ 6, "____37"},
>+      {37, kUpper | kPos, /*width = */ 0, "37"},
>+      {37, kUpper | kPos, /*width = */ 6, "____37"},
>+      {37, kUpper | kBase, /*width = */ 0, "37"},
>+      {37, kUpper | kBase, /*width = */ 6, "____37"},
>+      {37, kUpper | kBase | kPos, /*width = */ 0, "37"},
>+      {37, kUpper | kBase | kPos, /*width = */ 6, "____37"},
>+      {37, kLeft, /*width = */ 0, "37"},
>+      {37, kLeft, /*width = */ 6, "37____"},
>+      {37, kLeft | kPos, /*width = */ 0, "37"},
>+      {37, kLeft | kPos, /*width = */ 6, "37____"},
>+      {37, kLeft | kBase, /*width = */ 0, "37"},
>+      {37, kLeft | kBase, /*width = */ 6, "37____"},
>+      {37, kLeft | kBase | kPos, /*width = */ 0, "37"},
>+      {37, kLeft | kBase | kPos, /*width = */ 6, "37____"},
>+      {37, kLeft | kUpper, /*width = */ 0, "37"},
>+      {37, kLeft | kUpper, /*width = */ 6, "37____"},
>+      {37, kLeft | kUpper | kPos, /*width = */ 0, "37"},
>+      {37, kLeft | kUpper | kPos, /*width = */ 6, "37____"},
>+      {37, kLeft | kUpper | kBase, /*width = */ 0, "37"},
>+      {37, kLeft | kUpper | kBase, /*width = */ 6, "37____"},
>+      {37, kLeft | kUpper | kBase | kPos, /*width = */ 0, "37"},
>+      {37, kLeft | kUpper | kBase | kPos, /*width = */ 6, "37____"},
>+      {37, kInt, /*width = */ 0, "37"},
>+      {37, kInt, /*width = */ 6, "____37"},
>+      {37, kInt | kPos, /*width = */ 0, "37"},
>+      {37, kInt | kPos, /*width = */ 6, "____37"},
>+      {37, kInt | kBase, /*width = */ 0, "37"},
>+      {37, kInt | kBase, /*width = */ 6, "____37"},
>+      {37, kInt | kBase | kPos, /*width = */ 0, "37"},
>+      {37, kInt | kBase | kPos, /*width = */ 6, "____37"},
>+      {37, kInt | kUpper, /*width = */ 0, "37"},
>+      {37, kInt | kUpper, /*width = */ 6, "____37"},
>+      {37, kInt | kUpper | kPos, /*width = */ 0, "37"},
>+      {37, kInt | kUpper | kPos, /*width = */ 6, "____37"},
>+      {37, kInt | kUpper | kBase, /*width = */ 0, "37"},
>+      {37, kInt | kUpper | kBase, /*width = */ 6, "____37"},
>+      {37, kInt | kUpper | kBase | kPos, /*width = */ 0, "37"},
>+      {37, kInt | kUpper | kBase | kPos, /*width = */ 6, "____37"},
>+      {37, kRight, /*width = */ 0, "37"},
>+      {37, kRight, /*width = */ 6, "____37"},
>+      {37, kRight | kPos, /*width = */ 0, "37"},
>+      {37, kRight | kPos, /*width = */ 6, "____37"},
>+      {37, kRight | kBase, /*width = */ 0, "37"},
>+      {37, kRight | kBase, /*width = */ 6, "____37"},
>+      {37, kRight | kBase | kPos, /*width = */ 0, "37"},
>+      {37, kRight | kBase | kPos, /*width = */ 6, "____37"},
>+      {37, kRight | kUpper, /*width = */ 0, "37"},
>+      {37, kRight | kUpper, /*width = */ 6, "____37"},
>+      {37, kRight | kUpper | kPos, /*width = */ 0, "37"},
>+      {37, kRight | kUpper | kPos, /*width = */ 6, "____37"},
>+      {37, kRight | kUpper | kBase, /*width = */ 0, "37"},
>+      {37, kRight | kUpper | kBase, /*width = */ 6, "____37"},
>+      {37, kRight | kUpper | kBase | kPos, /*width = */ 0, "37"},
>+      {37, kRight | kUpper | kBase | kPos, /*width = */ 6, "____37"},
>+      {37, kDec, /*width = */ 0, "37"},
>+      {37, kDec, /*width = */ 6, "____37"},
>+      {37, kDec | kPos, /*width = */ 0, "37"},
>+      {37, kDec | kPos, /*width = */ 6, "____37"},
>+      {37, kDec | kBase, /*width = */ 0, "37"},
>+      {37, kDec | kBase, /*width = */ 6, "____37"},
>+      {37, kDec | kBase | kPos, /*width = */ 0, "37"},
>+      {37, kDec | kBase | kPos, /*width = */ 6, "____37"},
>+      {37, kDec | kUpper, /*width = */ 0, "37"},
>+      {37, kDec | kUpper, /*width = */ 6, "____37"},
>+      {37, kDec | kUpper | kPos, /*width = */ 0, "37"},
>+      {37, kDec | kUpper | kPos, /*width = */ 6, "____37"},
>+      {37, kDec | kUpper | kBase, /*width = */ 0, "37"},
>+      {37, kDec | kUpper | kBase, /*width = */ 6, "____37"},
>+      {37, kDec | kUpper | kBase | kPos, /*width = */ 0, "37"},
>+      {37, kDec | kUpper | kBase | kPos, /*width = */ 6, "____37"},
>+      {37, kDec | kLeft, /*width = */ 0, "37"},
>+      {37, kDec | kLeft, /*width = */ 6, "37____"},
>+      {37, kDec | kLeft | kPos, /*width = */ 0, "37"},
>+      {37, kDec | kLeft | kPos, /*width = */ 6, "37____"},
>+      {37, kDec | kLeft | kBase, /*width = */ 0, "37"},
>+      {37, kDec | kLeft | kBase, /*width = */ 6, "37____"},
>+      {37, kDec | kLeft | kBase | kPos, /*width = */ 0, "37"},
>+      {37, kDec | kLeft | kBase | kPos, /*width = */ 6, "37____"},
>+      {37, kDec | kLeft | kUpper, /*width = */ 0, "37"},
>+      {37, kDec | kLeft | kUpper, /*width = */ 6, "37____"},
>+      {37, kDec | kLeft | kUpper | kPos, /*width = */ 0, "37"},
>+      {37, kDec | kLeft | kUpper | kPos, /*width = */ 6, "37____"},
>+      {37, kDec | kLeft | kUpper | kBase, /*width = */ 0, "37"},
>+      {37, kDec | kLeft | kUpper | kBase, /*width = */ 6, "37____"},
>+      {37, kDec | kLeft | kUpper | kBase | kPos, /*width = */ 0, "37"},
>+      {37, kDec | kLeft | kUpper | kBase | kPos, /*width = */ 6, "37____"},
>+      {37, kDec | kInt, /*width = */ 0, "37"},
>+      {37, kDec | kInt, /*width = */ 6, "____37"},
>+      {37, kDec | kInt | kPos, /*width = */ 0, "37"},
>+      {37, kDec | kInt | kPos, /*width = */ 6, "____37"},
>+      {37, kDec | kInt | kBase, /*width = */ 0, "37"},
>+      {37, kDec | kInt | kBase, /*width = */ 6, "____37"},
>+      {37, kDec | kInt | kBase | kPos, /*width = */ 0, "37"},
>+      {37, kDec | kInt | kBase | kPos, /*width = */ 6, "____37"},
>+      {37, kDec | kInt | kUpper, /*width = */ 0, "37"},
>+      {37, kDec | kInt | kUpper, /*width = */ 6, "____37"},
>+      {37, kDec | kInt | kUpper | kPos, /*width = */ 0, "37"},
>+      {37, kDec | kInt | kUpper | kPos, /*width = */ 6, "____37"},
>+      {37, kDec | kInt | kUpper | kBase, /*width = */ 0, "37"},
>+      {37, kDec | kInt | kUpper | kBase, /*width = */ 6, "____37"},
>+      {37, kDec | kInt | kUpper | kBase | kPos, /*width = */ 0, "37"},
>+      {37, kDec | kInt | kUpper | kBase | kPos, /*width = */ 6, "____37"},
>+      {37, kDec | kRight, /*width = */ 0, "37"},
>+      {37, kDec | kRight, /*width = */ 6, "____37"},
>+      {37, kDec | kRight | kPos, /*width = */ 0, "37"},
>+      {37, kDec | kRight | kPos, /*width = */ 6, "____37"},
>+      {37, kDec | kRight | kBase, /*width = */ 0, "37"},
>+      {37, kDec | kRight | kBase, /*width = */ 6, "____37"},
>+      {37, kDec | kRight | kBase | kPos, /*width = */ 0, "37"},
>+      {37, kDec | kRight | kBase | kPos, /*width = */ 6, "____37"},
>+      {37, kDec | kRight | kUpper, /*width = */ 0, "37"},
>+      {37, kDec | kRight | kUpper, /*width = */ 6, "____37"},
>+      {37, kDec | kRight | kUpper | kPos, /*width = */ 0, "37"},
>+      {37, kDec | kRight | kUpper | kPos, /*width = */ 6, "____37"},
>+      {37, kDec | kRight | kUpper | kBase, /*width = */ 0, "37"},
>+      {37, kDec | kRight | kUpper | kBase, /*width = */ 6, "____37"},
>+      {37, kDec | kRight | kUpper | kBase | kPos, /*width = */ 0, "37"},
>+      {37, kDec | kRight | kUpper | kBase | kPos, /*width = */ 6, "____37"},
>+      {37, kOct, /*width = */ 0, "45"},
>+      {37, kOct, /*width = */ 6, "____45"},
>+      {37, kOct | kPos, /*width = */ 0, "45"},
>+      {37, kOct | kPos, /*width = */ 6, "____45"},
>+      {37, kOct | kBase, /*width = */ 0, "045"},
>+      {37, kOct | kBase, /*width = */ 6, "___045"},
>+      {37, kOct | kBase | kPos, /*width = */ 0, "045"},
>+      {37, kOct | kBase | kPos, /*width = */ 6, "___045"},
>+      {37, kOct | kUpper, /*width = */ 0, "45"},
>+      {37, kOct | kUpper, /*width = */ 6, "____45"},
>+      {37, kOct | kUpper | kPos, /*width = */ 0, "45"},
>+      {37, kOct | kUpper | kPos, /*width = */ 6, "____45"},
>+      {37, kOct | kUpper | kBase, /*width = */ 0, "045"},
>+      {37, kOct | kUpper | kBase, /*width = */ 6, "___045"},
>+      {37, kOct | kUpper | kBase | kPos, /*width = */ 0, "045"},
>+      {37, kOct | kUpper | kBase | kPos, /*width = */ 6, "___045"},
>+      {37, kOct | kLeft, /*width = */ 0, "45"},
>+      {37, kOct | kLeft, /*width = */ 6, "45____"},
>+      {37, kOct | kLeft | kPos, /*width = */ 0, "45"},
>+      {37, kOct | kLeft | kPos, /*width = */ 6, "45____"},
>+      {37, kOct | kLeft | kBase, /*width = */ 0, "045"},
>+      {37, kOct | kLeft | kBase, /*width = */ 6, "045___"},
>+      {37, kOct | kLeft | kBase | kPos, /*width = */ 0, "045"},
>+      {37, kOct | kLeft | kBase | kPos, /*width = */ 6, "045___"},
>+      {37, kOct | kLeft | kUpper, /*width = */ 0, "45"},
>+      {37, kOct | kLeft | kUpper, /*width = */ 6, "45____"},
>+      {37, kOct | kLeft | kUpper | kPos, /*width = */ 0, "45"},
>+      {37, kOct | kLeft | kUpper | kPos, /*width = */ 6, "45____"},
>+      {37, kOct | kLeft | kUpper | kBase, /*width = */ 0, "045"},
>+      {37, kOct | kLeft | kUpper | kBase, /*width = */ 6, "045___"},
>+      {37, kOct | kLeft | kUpper | kBase | kPos, /*width = */ 0, "045"},
>+      {37, kOct | kLeft | kUpper | kBase | kPos, /*width = */ 6, "045___"},
>+      {37, kOct | kInt, /*width = */ 0, "45"},
>+      {37, kOct | kInt, /*width = */ 6, "____45"},
>+      {37, kOct | kInt | kPos, /*width = */ 0, "45"},
>+      {37, kOct | kInt | kPos, /*width = */ 6, "____45"},
>+      {37, kOct | kInt | kBase, /*width = */ 0, "045"},
>+      {37, kOct | kInt | kBase, /*width = */ 6, "___045"},
>+      {37, kOct | kInt | kBase | kPos, /*width = */ 0, "045"},
>+      {37, kOct | kInt | kBase | kPos, /*width = */ 6, "___045"},
>+      {37, kOct | kInt | kUpper, /*width = */ 0, "45"},
>+      {37, kOct | kInt | kUpper, /*width = */ 6, "____45"},
>+      {37, kOct | kInt | kUpper | kPos, /*width = */ 0, "45"},
>+      {37, kOct | kInt | kUpper | kPos, /*width = */ 6, "____45"},
>+      {37, kOct | kInt | kUpper | kBase, /*width = */ 0, "045"},
>+      {37, kOct | kInt | kUpper | kBase, /*width = */ 6, "___045"},
>+      {37, kOct | kInt | kUpper | kBase | kPos, /*width = */ 0, "045"},
>+      {37, kOct | kInt | kUpper | kBase | kPos, /*width = */ 6, "___045"},
>+      {37, kOct | kRight, /*width = */ 0, "45"},
>+      {37, kOct | kRight, /*width = */ 6, "____45"},
>+      {37, kOct | kRight | kPos, /*width = */ 0, "45"},
>+      {37, kOct | kRight | kPos, /*width = */ 6, "____45"},
>+      {37, kOct | kRight | kBase, /*width = */ 0, "045"},
>+      {37, kOct | kRight | kBase, /*width = */ 6, "___045"},
>+      {37, kOct | kRight | kBase | kPos, /*width = */ 0, "045"},
>+      {37, kOct | kRight | kBase | kPos, /*width = */ 6, "___045"},
>+      {37, kOct | kRight | kUpper, /*width = */ 0, "45"},
>+      {37, kOct | kRight | kUpper, /*width = */ 6, "____45"},
>+      {37, kOct | kRight | kUpper | kPos, /*width = */ 0, "45"},
>+      {37, kOct | kRight | kUpper | kPos, /*width = */ 6, "____45"},
>+      {37, kOct | kRight | kUpper | kBase, /*width = */ 0, "045"},
>+      {37, kOct | kRight | kUpper | kBase, /*width = */ 6, "___045"},
>+      {37, kOct | kRight | kUpper | kBase | kPos, /*width = */ 0, "045"},
>+      {37, kOct | kRight | kUpper | kBase | kPos, /*width = */ 6, "___045"},
>+      {37, kHex, /*width = */ 0, "25"},
>+      {37, kHex, /*width = */ 6, "____25"},
>+      {37, kHex | kPos, /*width = */ 0, "25"},
>+      {37, kHex | kPos, /*width = */ 6, "____25"},
>+      {37, kHex | kBase, /*width = */ 0, "0x25"},
>+      {37, kHex | kBase, /*width = */ 6, "__0x25"},
>+      {37, kHex | kBase | kPos, /*width = */ 0, "0x25"},
>+      {37, kHex | kBase | kPos, /*width = */ 6, "__0x25"},
>+      {37, kHex | kUpper, /*width = */ 0, "25"},
>+      {37, kHex | kUpper, /*width = */ 6, "____25"},
>+      {37, kHex | kUpper | kPos, /*width = */ 0, "25"},
>+      {37, kHex | kUpper | kPos, /*width = */ 6, "____25"},
>+      {37, kHex | kUpper | kBase, /*width = */ 0, "0X25"},
>+      {37, kHex | kUpper | kBase, /*width = */ 6, "__0X25"},
>+      {37, kHex | kUpper | kBase | kPos, /*width = */ 0, "0X25"},
>+      {37, kHex | kUpper | kBase | kPos, /*width = */ 6, "__0X25"},
>+      {37, kHex | kLeft, /*width = */ 0, "25"},
>+      {37, kHex | kLeft, /*width = */ 6, "25____"},
>+      {37, kHex | kLeft | kPos, /*width = */ 0, "25"},
>+      {37, kHex | kLeft | kPos, /*width = */ 6, "25____"},
>+      {37, kHex | kLeft | kBase, /*width = */ 0, "0x25"},
>+      {37, kHex | kLeft | kBase, /*width = */ 6, "0x25__"},
>+      {37, kHex | kLeft | kBase | kPos, /*width = */ 0, "0x25"},
>+      {37, kHex | kLeft | kBase | kPos, /*width = */ 6, "0x25__"},
>+      {37, kHex | kLeft | kUpper, /*width = */ 0, "25"},
>+      {37, kHex | kLeft | kUpper, /*width = */ 6, "25____"},
>+      {37, kHex | kLeft | kUpper | kPos, /*width = */ 0, "25"},
>+      {37, kHex | kLeft | kUpper | kPos, /*width = */ 6, "25____"},
>+      {37, kHex | kLeft | kUpper | kBase, /*width = */ 0, "0X25"},
>+      {37, kHex | kLeft | kUpper | kBase, /*width = */ 6, "0X25__"},
>+      {37, kHex | kLeft | kUpper | kBase | kPos, /*width = */ 0, "0X25"},
>+      {37, kHex | kLeft | kUpper | kBase | kPos, /*width = */ 6, "0X25__"},
>+      {37, kHex | kInt, /*width = */ 0, "25"},
>+      {37, kHex | kInt, /*width = */ 6, "____25"},
>+      {37, kHex | kInt | kPos, /*width = */ 0, "25"},
>+      {37, kHex | kInt | kPos, /*width = */ 6, "____25"},
>+      {37, kHex | kInt | kBase, /*width = */ 0, "0x25"},
>+      {37, kHex | kInt | kBase, /*width = */ 6, "0x__25"},
>+      {37, kHex | kInt | kBase | kPos, /*width = */ 0, "0x25"},
>+      {37, kHex | kInt | kBase | kPos, /*width = */ 6, "0x__25"},
>+      {37, kHex | kInt | kUpper, /*width = */ 0, "25"},
>+      {37, kHex | kInt | kUpper, /*width = */ 6, "____25"},
>+      {37, kHex | kInt | kUpper | kPos, /*width = */ 0, "25"},
>+      {37, kHex | kInt | kUpper | kPos, /*width = */ 6, "____25"},
>+      {37, kHex | kInt | kUpper | kBase, /*width = */ 0, "0X25"},
>+      {37, kHex | kInt | kUpper | kBase, /*width = */ 6, "0X__25"},
>+      {37, kHex | kInt | kUpper | kBase | kPos, /*width = */ 0, "0X25"},
>+      {37, kHex | kInt | kUpper | kBase | kPos, /*width = */ 6, "0X__25"},
>+      {37, kHex | kRight, /*width = */ 0, "25"},
>+      {37, kHex | kRight, /*width = */ 6, "____25"},
>+      {37, kHex | kRight | kPos, /*width = */ 0, "25"},
>+      {37, kHex | kRight | kPos, /*width = */ 6, "____25"},
>+      {37, kHex | kRight | kBase, /*width = */ 0, "0x25"},
>+      {37, kHex | kRight | kBase, /*width = */ 6, "__0x25"},
>+      {37, kHex | kRight | kBase | kPos, /*width = */ 0, "0x25"},
>+      {37, kHex | kRight | kBase | kPos, /*width = */ 6, "__0x25"},
>+      {37, kHex | kRight | kUpper, /*width = */ 0, "25"},
>+      {37, kHex | kRight | kUpper, /*width = */ 6, "____25"},
>+      {37, kHex | kRight | kUpper | kPos, /*width = */ 0, "25"},
>+      {37, kHex | kRight | kUpper | kPos, /*width = */ 6, "____25"},
>+      {37, kHex | kRight | kUpper | kBase, /*width = */ 0, "0X25"},
>+      {37, kHex | kRight | kUpper | kBase, /*width = */ 6, "__0X25"},
>+      {37, kHex | kRight | kUpper | kBase | kPos, /*width = */ 0, "0X25"},
>+      {37, kHex | kRight | kUpper | kBase | kPos, /*width = */ 6, "__0X25"}};
>+}
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/numeric/int128_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/numeric/int128_test.cc
>new file mode 100644
>index 00000000000..1eb3e0ec896
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/numeric/int128_test.cc
>@@ -0,0 +1,442 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/numeric/int128.h"
>+
>+#include <algorithm>
>+#include <limits>
>+#include <random>
>+#include <type_traits>
>+#include <utility>
>+#include <vector>
>+
>+#include "gtest/gtest.h"
>+#include "absl/base/internal/cycleclock.h"
>+#include "absl/meta/type_traits.h"
>+
>+#if defined(_MSC_VER) && _MSC_VER == 1900
>+// Disable "unary minus operator applied to unsigned type" warnings in Microsoft
>+// Visual C++ 14 (2015).
>+#pragma warning(disable:4146)
>+#endif
>+
>+namespace {
>+
>+template <typename T>
>+class Uint128IntegerTraitsTest : public ::testing::Test {};
>+typedef ::testing::Types<bool, char, signed char, unsigned char, char16_t,
>+                         char32_t, wchar_t,
>+                         short,           // NOLINT(runtime/int)
>+                         unsigned short,  // NOLINT(runtime/int)
>+                         int, unsigned int,
>+                         long,                // NOLINT(runtime/int)
>+                         unsigned long,       // NOLINT(runtime/int)
>+                         long long,           // NOLINT(runtime/int)
>+                         unsigned long long>  // NOLINT(runtime/int)
>+    IntegerTypes;
>+
>+template <typename T>
>+class Uint128FloatTraitsTest : public ::testing::Test {};
>+typedef ::testing::Types<float, double, long double> FloatingPointTypes;
>+
>+TYPED_TEST_CASE(Uint128IntegerTraitsTest, IntegerTypes);
>+
>+TYPED_TEST(Uint128IntegerTraitsTest, ConstructAssignTest) {
>+  static_assert(std::is_constructible<absl::uint128, TypeParam>::value,
>+                "absl::uint128 must be constructible from TypeParam");
>+  static_assert(std::is_assignable<absl::uint128&, TypeParam>::value,
>+                "absl::uint128 must be assignable from TypeParam");
>+  static_assert(!std::is_assignable<TypeParam&, absl::uint128>::value,
>+                "TypeParam must not be assignable from absl::uint128");
>+}
>+
>+TYPED_TEST_CASE(Uint128FloatTraitsTest, FloatingPointTypes);
>+
>+TYPED_TEST(Uint128FloatTraitsTest, ConstructAssignTest) {
>+  static_assert(std::is_constructible<absl::uint128, TypeParam>::value,
>+                "absl::uint128 must be constructible from TypeParam");
>+  static_assert(!std::is_assignable<absl::uint128&, TypeParam>::value,
>+                "absl::uint128 must not be assignable from TypeParam");
>+  static_assert(!std::is_assignable<TypeParam&, absl::uint128>::value,
>+                "TypeParam must not be assignable from absl::uint128");
>+}
>+
>+#ifdef ABSL_HAVE_INTRINSIC_INT128
>+// These type traits done separately as TYPED_TEST requires typeinfo, and not
>+// all platforms have this for __int128 even though they define the type.
>+TEST(Uint128, IntrinsicTypeTraitsTest) {
>+  static_assert(std::is_constructible<absl::uint128, __int128>::value,
>+                "absl::uint128 must be constructible from __int128");
>+  static_assert(std::is_assignable<absl::uint128&, __int128>::value,
>+                "absl::uint128 must be assignable from __int128");
>+  static_assert(!std::is_assignable<__int128&, absl::uint128>::value,
>+                "__int128 must not be assignable from absl::uint128");
>+
>+  static_assert(std::is_constructible<absl::uint128, unsigned __int128>::value,
>+                "absl::uint128 must be constructible from unsigned __int128");
>+  static_assert(std::is_assignable<absl::uint128&, unsigned __int128>::value,
>+                "absl::uint128 must be assignable from unsigned __int128");
>+  static_assert(!std::is_assignable<unsigned __int128&, absl::uint128>::value,
>+                "unsigned __int128 must not be assignable from absl::uint128");
>+}
>+#endif  // ABSL_HAVE_INTRINSIC_INT128
>+
>+TEST(Uint128, TrivialTraitsTest) {
>+  static_assert(absl::is_trivially_default_constructible<absl::uint128>::value,
>+                "");
>+  static_assert(absl::is_trivially_copy_constructible<absl::uint128>::value,
>+                "");
>+  static_assert(absl::is_trivially_copy_assignable<absl::uint128>::value, "");
>+  static_assert(std::is_trivially_destructible<absl::uint128>::value, "");
>+}
>+
>+TEST(Uint128, AllTests) {
>+  absl::uint128 zero = 0;
>+  absl::uint128 one = 1;
>+  absl::uint128 one_2arg = absl::MakeUint128(0, 1);
>+  absl::uint128 two = 2;
>+  absl::uint128 three = 3;
>+  absl::uint128 big = absl::MakeUint128(2000, 2);
>+  absl::uint128 big_minus_one = absl::MakeUint128(2000, 1);
>+  absl::uint128 bigger = absl::MakeUint128(2001, 1);
>+  absl::uint128 biggest = absl::Uint128Max();
>+  absl::uint128 high_low = absl::MakeUint128(1, 0);
>+  absl::uint128 low_high =
>+      absl::MakeUint128(0, std::numeric_limits<uint64_t>::max());
>+  EXPECT_LT(one, two);
>+  EXPECT_GT(two, one);
>+  EXPECT_LT(one, big);
>+  EXPECT_LT(one, big);
>+  EXPECT_EQ(one, one_2arg);
>+  EXPECT_NE(one, two);
>+  EXPECT_GT(big, one);
>+  EXPECT_GE(big, two);
>+  EXPECT_GE(big, big_minus_one);
>+  EXPECT_GT(big, big_minus_one);
>+  EXPECT_LT(big_minus_one, big);
>+  EXPECT_LE(big_minus_one, big);
>+  EXPECT_NE(big_minus_one, big);
>+  EXPECT_LT(big, biggest);
>+  EXPECT_LE(big, biggest);
>+  EXPECT_GT(biggest, big);
>+  EXPECT_GE(biggest, big);
>+  EXPECT_EQ(big, ~~big);
>+  EXPECT_EQ(one, one | one);
>+  EXPECT_EQ(big, big | big);
>+  EXPECT_EQ(one, one | zero);
>+  EXPECT_EQ(one, one & one);
>+  EXPECT_EQ(big, big & big);
>+  EXPECT_EQ(zero, one & zero);
>+  EXPECT_EQ(zero, big & ~big);
>+  EXPECT_EQ(zero, one ^ one);
>+  EXPECT_EQ(zero, big ^ big);
>+  EXPECT_EQ(one, one ^ zero);
>+
>+  // Shift operators.
>+  EXPECT_EQ(big, big << 0);
>+  EXPECT_EQ(big, big >> 0);
>+  EXPECT_GT(big << 1, big);
>+  EXPECT_LT(big >> 1, big);
>+  EXPECT_EQ(big, (big << 10) >> 10);
>+  EXPECT_EQ(big, (big >> 1) << 1);
>+  EXPECT_EQ(one, (one << 80) >> 80);
>+  EXPECT_EQ(zero, (one >> 80) << 80);
>+
>+  // Shift assignments.
>+  absl::uint128 big_copy = big;
>+  EXPECT_EQ(big << 0, big_copy <<= 0);
>+  big_copy = big;
>+  EXPECT_EQ(big >> 0, big_copy >>= 0);
>+  big_copy = big;
>+  EXPECT_EQ(big << 1, big_copy <<= 1);
>+  big_copy = big;
>+  EXPECT_EQ(big >> 1, big_copy >>= 1);
>+  big_copy = big;
>+  EXPECT_EQ(big << 10, big_copy <<= 10);
>+  big_copy = big;
>+  EXPECT_EQ(big >> 10, big_copy >>= 10);
>+  big_copy = big;
>+  EXPECT_EQ(big << 64, big_copy <<= 64);
>+  big_copy = big;
>+  EXPECT_EQ(big >> 64, big_copy >>= 64);
>+  big_copy = big;
>+  EXPECT_EQ(big << 73, big_copy <<= 73);
>+  big_copy = big;
>+  EXPECT_EQ(big >> 73, big_copy >>= 73);
>+
>+  EXPECT_EQ(absl::Uint128High64(biggest), std::numeric_limits<uint64_t>::max());
>+  EXPECT_EQ(absl::Uint128Low64(biggest), std::numeric_limits<uint64_t>::max());
>+  EXPECT_EQ(zero + one, one);
>+  EXPECT_EQ(one + one, two);
>+  EXPECT_EQ(big_minus_one + one, big);
>+  EXPECT_EQ(one - one, zero);
>+  EXPECT_EQ(one - zero, one);
>+  EXPECT_EQ(zero - one, biggest);
>+  EXPECT_EQ(big - big, zero);
>+  EXPECT_EQ(big - one, big_minus_one);
>+  EXPECT_EQ(big + std::numeric_limits<uint64_t>::max(), bigger);
>+  EXPECT_EQ(biggest + 1, zero);
>+  EXPECT_EQ(zero - 1, biggest);
>+  EXPECT_EQ(high_low - one, low_high);
>+  EXPECT_EQ(low_high + one, high_low);
>+  EXPECT_EQ(absl::Uint128High64((absl::uint128(1) << 64) - 1), 0);
>+  EXPECT_EQ(absl::Uint128Low64((absl::uint128(1) << 64) - 1),
>+            std::numeric_limits<uint64_t>::max());
>+  EXPECT_TRUE(!!one);
>+  EXPECT_TRUE(!!high_low);
>+  EXPECT_FALSE(!!zero);
>+  EXPECT_FALSE(!one);
>+  EXPECT_FALSE(!high_low);
>+  EXPECT_TRUE(!zero);
>+  EXPECT_TRUE(zero == 0);       // NOLINT(readability/check)
>+  EXPECT_FALSE(zero != 0);      // NOLINT(readability/check)
>+  EXPECT_FALSE(one == 0);       // NOLINT(readability/check)
>+  EXPECT_TRUE(one != 0);        // NOLINT(readability/check)
>+  EXPECT_FALSE(high_low == 0);  // NOLINT(readability/check)
>+  EXPECT_TRUE(high_low != 0);   // NOLINT(readability/check)
>+
>+  absl::uint128 test = zero;
>+  EXPECT_EQ(++test, one);
>+  EXPECT_EQ(test, one);
>+  EXPECT_EQ(test++, one);
>+  EXPECT_EQ(test, two);
>+  EXPECT_EQ(test -= 2, zero);
>+  EXPECT_EQ(test, zero);
>+  EXPECT_EQ(test += 2, two);
>+  EXPECT_EQ(test, two);
>+  EXPECT_EQ(--test, one);
>+  EXPECT_EQ(test, one);
>+  EXPECT_EQ(test--, one);
>+  EXPECT_EQ(test, zero);
>+  EXPECT_EQ(test |= three, three);
>+  EXPECT_EQ(test &= one, one);
>+  EXPECT_EQ(test ^= three, two);
>+  EXPECT_EQ(test >>= 1, one);
>+  EXPECT_EQ(test <<= 1, two);
>+
>+  EXPECT_EQ(big, -(-big));
>+  EXPECT_EQ(two, -((-one) - 1));
>+  EXPECT_EQ(absl::Uint128Max(), -one);
>+  EXPECT_EQ(zero, -zero);
>+
>+  EXPECT_EQ(absl::Uint128Max(), absl::kuint128max);
>+}
>+
>+TEST(Uint128, ConversionTests) {
>+  EXPECT_TRUE(absl::MakeUint128(1, 0));
>+
>+#ifdef ABSL_HAVE_INTRINSIC_INT128
>+  unsigned __int128 intrinsic =
>+      (static_cast<unsigned __int128>(0x3a5b76c209de76f6) << 64) +
>+      0x1f25e1d63a2b46c5;
>+  absl::uint128 custom =
>+      absl::MakeUint128(0x3a5b76c209de76f6, 0x1f25e1d63a2b46c5);
>+
>+  EXPECT_EQ(custom, absl::uint128(intrinsic));
>+  EXPECT_EQ(custom, absl::uint128(static_cast<__int128>(intrinsic)));
>+  EXPECT_EQ(intrinsic, static_cast<unsigned __int128>(custom));
>+  EXPECT_EQ(intrinsic, static_cast<__int128>(custom));
>+#endif  // ABSL_HAVE_INTRINSIC_INT128
>+
>+  // verify that an integer greater than 2**64 that can be stored precisely
>+  // inside a double is converted to a absl::uint128 without loss of
>+  // information.
>+  double precise_double = 0x530e * std::pow(2.0, 64.0) + 0xda74000000000000;
>+  absl::uint128 from_precise_double(precise_double);
>+  absl::uint128 from_precise_ints =
>+      absl::MakeUint128(0x530e, 0xda74000000000000);
>+  EXPECT_EQ(from_precise_double, from_precise_ints);
>+  EXPECT_DOUBLE_EQ(static_cast<double>(from_precise_ints), precise_double);
>+
>+  double approx_double = 0xffffeeeeddddcccc * std::pow(2.0, 64.0) +
>+                         0xbbbbaaaa99998888;
>+  absl::uint128 from_approx_double(approx_double);
>+  EXPECT_DOUBLE_EQ(static_cast<double>(from_approx_double), approx_double);
>+
>+  double round_to_zero = 0.7;
>+  double round_to_five = 5.8;
>+  double round_to_nine = 9.3;
>+  EXPECT_EQ(static_cast<absl::uint128>(round_to_zero), 0);
>+  EXPECT_EQ(static_cast<absl::uint128>(round_to_five), 5);
>+  EXPECT_EQ(static_cast<absl::uint128>(round_to_nine), 9);
>+}
>+
>+TEST(Uint128, OperatorAssignReturnRef) {
>+  absl::uint128 v(1);
>+  (v += 4) -= 3;
>+  EXPECT_EQ(2, v);
>+}
>+
>+TEST(Uint128, Multiply) {
>+  absl::uint128 a, b, c;
>+
>+  // Zero test.
>+  a = 0;
>+  b = 0;
>+  c = a * b;
>+  EXPECT_EQ(0, c);
>+
>+  // Max carries.
>+  a = absl::uint128(0) - 1;
>+  b = absl::uint128(0) - 1;
>+  c = a * b;
>+  EXPECT_EQ(1, c);
>+
>+  // Self-operation with max carries.
>+  c = absl::uint128(0) - 1;
>+  c *= c;
>+  EXPECT_EQ(1, c);
>+
>+  // 1-bit x 1-bit.
>+  for (int i = 0; i < 64; ++i) {
>+    for (int j = 0; j < 64; ++j) {
>+      a = absl::uint128(1) << i;
>+      b = absl::uint128(1) << j;
>+      c = a * b;
>+      EXPECT_EQ(absl::uint128(1) << (i + j), c);
>+    }
>+  }
>+
>+  // Verified with dc.
>+  a = absl::MakeUint128(0xffffeeeeddddcccc, 0xbbbbaaaa99998888);
>+  b = absl::MakeUint128(0x7777666655554444, 0x3333222211110000);
>+  c = a * b;
>+  EXPECT_EQ(absl::MakeUint128(0x530EDA741C71D4C3, 0xBF25975319080000), c);
>+  EXPECT_EQ(0, c - b * a);
>+  EXPECT_EQ(a*a - b*b, (a+b) * (a-b));
>+
>+  // Verified with dc.
>+  a = absl::MakeUint128(0x0123456789abcdef, 0xfedcba9876543210);
>+  b = absl::MakeUint128(0x02468ace13579bdf, 0xfdb97531eca86420);
>+  c = a * b;
>+  EXPECT_EQ(absl::MakeUint128(0x97a87f4f261ba3f2, 0x342d0bbf48948200), c);
>+  EXPECT_EQ(0, c - b * a);
>+  EXPECT_EQ(a*a - b*b, (a+b) * (a-b));
>+}
>+
>+TEST(Uint128, AliasTests) {
>+  absl::uint128 x1 = absl::MakeUint128(1, 2);
>+  absl::uint128 x2 = absl::MakeUint128(2, 4);
>+  x1 += x1;
>+  EXPECT_EQ(x2, x1);
>+
>+  absl::uint128 x3 = absl::MakeUint128(1, static_cast<uint64_t>(1) << 63);
>+  absl::uint128 x4 = absl::MakeUint128(3, 0);
>+  x3 += x3;
>+  EXPECT_EQ(x4, x3);
>+}
>+
>+TEST(Uint128, DivideAndMod) {
>+  using std::swap;
>+
>+  // a := q * b + r
>+  absl::uint128 a, b, q, r;
>+
>+  // Zero test.
>+  a = 0;
>+  b = 123;
>+  q = a / b;
>+  r = a % b;
>+  EXPECT_EQ(0, q);
>+  EXPECT_EQ(0, r);
>+
>+  a = absl::MakeUint128(0x530eda741c71d4c3, 0xbf25975319080000);
>+  q = absl::MakeUint128(0x4de2cab081, 0x14c34ab4676e4bab);
>+  b = absl::uint128(0x1110001);
>+  r = absl::uint128(0x3eb455);
>+  ASSERT_EQ(a, q * b + r);  // Sanity-check.
>+
>+  absl::uint128 result_q, result_r;
>+  result_q = a / b;
>+  result_r = a % b;
>+  EXPECT_EQ(q, result_q);
>+  EXPECT_EQ(r, result_r);
>+
>+  // Try the other way around.
>+  swap(q, b);
>+  result_q = a / b;
>+  result_r = a % b;
>+  EXPECT_EQ(q, result_q);
>+  EXPECT_EQ(r, result_r);
>+  // Restore.
>+  swap(b, q);
>+
>+  // Dividend < divisor; result should be q:0 r:<dividend>.
>+  swap(a, b);
>+  result_q = a / b;
>+  result_r = a % b;
>+  EXPECT_EQ(0, result_q);
>+  EXPECT_EQ(a, result_r);
>+  // Try the other way around.
>+  swap(a, q);
>+  result_q = a / b;
>+  result_r = a % b;
>+  EXPECT_EQ(0, result_q);
>+  EXPECT_EQ(a, result_r);
>+  // Restore.
>+  swap(q, a);
>+  swap(b, a);
>+
>+  // Try a large remainder.
>+  b = a / 2 + 1;
>+  absl::uint128 expected_r =
>+      absl::MakeUint128(0x29876d3a0e38ea61, 0xdf92cba98c83ffff);
>+  // Sanity checks.
>+  ASSERT_EQ(a / 2 - 1, expected_r);
>+  ASSERT_EQ(a, b + expected_r);
>+  result_q = a / b;
>+  result_r = a % b;
>+  EXPECT_EQ(1, result_q);
>+  EXPECT_EQ(expected_r, result_r);
>+}
>+
>+TEST(Uint128, DivideAndModRandomInputs) {
>+  const int kNumIters = 1 << 18;
>+  std::minstd_rand random(testing::UnitTest::GetInstance()->random_seed());
>+  std::uniform_int_distribution<uint64_t> uniform_uint64;
>+  for (int i = 0; i < kNumIters; ++i) {
>+    const absl::uint128 a =
>+        absl::MakeUint128(uniform_uint64(random), uniform_uint64(random));
>+    const absl::uint128 b =
>+        absl::MakeUint128(uniform_uint64(random), uniform_uint64(random));
>+    if (b == 0) {
>+      continue;  // Avoid a div-by-zero.
>+    }
>+    const absl::uint128 q = a / b;
>+    const absl::uint128 r = a % b;
>+    ASSERT_EQ(a, b * q + r);
>+  }
>+}
>+
>+TEST(Uint128, ConstexprTest) {
>+  constexpr absl::uint128 zero = absl::uint128();
>+  constexpr absl::uint128 one = 1;
>+  constexpr absl::uint128 minus_two = -2;
>+  EXPECT_EQ(zero, absl::uint128(0));
>+  EXPECT_EQ(one, absl::uint128(1));
>+  EXPECT_EQ(minus_two, absl::MakeUint128(-1, -2));
>+}
>+
>+TEST(Uint128, NumericLimitsTest) {
>+  static_assert(std::numeric_limits<absl::uint128>::is_specialized, "");
>+  static_assert(!std::numeric_limits<absl::uint128>::is_signed, "");
>+  static_assert(std::numeric_limits<absl::uint128>::is_integer, "");
>+  EXPECT_EQ(static_cast<int>(128 * std::log10(2)),
>+            std::numeric_limits<absl::uint128>::digits10);
>+  EXPECT_EQ(0, std::numeric_limits<absl::uint128>::min());
>+  EXPECT_EQ(0, std::numeric_limits<absl::uint128>::lowest());
>+  EXPECT_EQ(absl::Uint128Max(), std::numeric_limits<absl::uint128>::max());
>+}
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/BUILD.bazel b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/BUILD.bazel
>new file mode 100644
>index 00000000000..3a5f1332cda
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/BUILD.bazel
>@@ -0,0 +1,661 @@
>+#
>+# Copyright 2017 The Abseil Authors.
>+#
>+# Licensed under the Apache License, Version 2.0 (the "License");
>+# you may not use this file except in compliance with the License.
>+# You may obtain a copy of the License at
>+#
>+#      http://www.apache.org/licenses/LICENSE-2.0
>+#
>+# Unless required by applicable law or agreed to in writing, software
>+# distributed under the License is distributed on an "AS IS" BASIS,
>+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+# See the License for the specific language governing permissions and
>+# limitations under the License.
>+#
>+
>+load(
>+    "//absl:copts.bzl",
>+    "ABSL_DEFAULT_COPTS",
>+    "ABSL_TEST_COPTS",
>+    "ABSL_EXCEPTIONS_FLAG",
>+)
>+
>+package(
>+    default_visibility = ["//visibility:public"],
>+    features = ["parse_headers"],
>+)
>+
>+licenses(["notice"])  # Apache 2.0
>+
>+cc_library(
>+    name = "strings",
>+    srcs = [
>+        "ascii.cc",
>+        "charconv.cc",
>+        "escaping.cc",
>+        "internal/charconv_bigint.cc",
>+        "internal/charconv_bigint.h",
>+        "internal/charconv_parse.cc",
>+        "internal/charconv_parse.h",
>+        "internal/memutil.cc",
>+        "internal/memutil.h",
>+        "internal/stl_type_traits.h",
>+        "internal/str_join_internal.h",
>+        "internal/str_split_internal.h",
>+        "match.cc",
>+        "numbers.cc",
>+        "str_cat.cc",
>+        "str_replace.cc",
>+        "str_split.cc",
>+        "string_view.cc",
>+        "substitute.cc",
>+    ],
>+    hdrs = [
>+        "ascii.h",
>+        "charconv.h",
>+        "escaping.h",
>+        "match.h",
>+        "numbers.h",
>+        "str_cat.h",
>+        "str_join.h",
>+        "str_replace.h",
>+        "str_split.h",
>+        "string_view.h",
>+        "strip.h",
>+        "substitute.h",
>+    ],
>+    copts = ABSL_DEFAULT_COPTS,
>+    deps = [
>+        ":internal",
>+        "//absl/base",
>+        "//absl/base:config",
>+        "//absl/base:core_headers",
>+        "//absl/base:endian",
>+        "//absl/base:throw_delegate",
>+        "//absl/memory",
>+        "//absl/meta:type_traits",
>+        "//absl/numeric:int128",
>+    ],
>+)
>+
>+cc_library(
>+    name = "internal",
>+    srcs = [
>+        "internal/ostringstream.cc",
>+        "internal/utf8.cc",
>+    ],
>+    hdrs = [
>+        "internal/bits.h",
>+        "internal/char_map.h",
>+        "internal/ostringstream.h",
>+        "internal/resize_uninitialized.h",
>+        "internal/utf8.h",
>+    ],
>+    copts = ABSL_DEFAULT_COPTS,
>+    deps = [
>+        "//absl/base:core_headers",
>+        "//absl/base:endian",
>+        "//absl/meta:type_traits",
>+    ],
>+)
>+
>+cc_test(
>+    name = "match_test",
>+    size = "small",
>+    srcs = ["match_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    visibility = ["//visibility:private"],
>+    deps = [
>+        ":strings",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "escaping_test",
>+    size = "small",
>+    srcs = [
>+        "escaping_test.cc",
>+        "internal/escaping_test_common.h",
>+    ],
>+    copts = ABSL_TEST_COPTS,
>+    visibility = ["//visibility:private"],
>+    deps = [
>+        ":strings",
>+        "//absl/base:core_headers",
>+        "//absl/container:fixed_array",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "escaping_benchmark",
>+    srcs = [
>+        "escaping_benchmark.cc",
>+        "internal/escaping_test_common.h",
>+    ],
>+    copts = ABSL_TEST_COPTS,
>+    tags = ["benchmark"],
>+    visibility = ["//visibility:private"],
>+    deps = [
>+        ":strings",
>+        "//absl/base",
>+        "@com_github_google_benchmark//:benchmark_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "ascii_test",
>+    size = "small",
>+    srcs = ["ascii_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    visibility = ["//visibility:private"],
>+    deps = [
>+        ":strings",
>+        "//absl/base:core_headers",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "ascii_benchmark",
>+    srcs = ["ascii_benchmark.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    tags = ["benchmark"],
>+    visibility = ["//visibility:private"],
>+    deps = [
>+        ":strings",
>+        "@com_github_google_benchmark//:benchmark_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "memutil_benchmark",
>+    srcs = [
>+        "internal/memutil.h",
>+        "internal/memutil_benchmark.cc",
>+    ],
>+    copts = ABSL_TEST_COPTS,
>+    tags = ["benchmark"],
>+    visibility = ["//visibility:private"],
>+    deps = [
>+        ":strings",
>+        "//absl/base:core_headers",
>+        "@com_github_google_benchmark//:benchmark_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "memutil_test",
>+    size = "small",
>+    srcs = [
>+        "internal/memutil.h",
>+        "internal/memutil_test.cc",
>+    ],
>+    copts = ABSL_TEST_COPTS,
>+    visibility = ["//visibility:private"],
>+    deps = [
>+        ":strings",
>+        "//absl/base:core_headers",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "utf8_test",
>+    size = "small",
>+    srcs = [
>+        "internal/utf8_test.cc",
>+    ],
>+    copts = ABSL_TEST_COPTS,
>+    visibility = ["//visibility:private"],
>+    deps = [
>+        ":internal",
>+        ":strings",
>+        "//absl/base:core_headers",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "string_view_benchmark",
>+    srcs = ["string_view_benchmark.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    tags = ["benchmark"],
>+    visibility = ["//visibility:private"],
>+    deps = [
>+        ":strings",
>+        "//absl/base",
>+        "//absl/base:core_headers",
>+        "@com_github_google_benchmark//:benchmark_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "string_view_test",
>+    size = "small",
>+    srcs = ["string_view_test.cc"],
>+    copts = ABSL_TEST_COPTS + ABSL_EXCEPTIONS_FLAG,
>+    visibility = ["//visibility:private"],
>+    deps = [
>+        ":strings",
>+        "//absl/base:config",
>+        "//absl/base:core_headers",
>+        "//absl/base:dynamic_annotations",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "substitute_test",
>+    size = "small",
>+    srcs = ["substitute_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    visibility = ["//visibility:private"],
>+    deps = [
>+        ":strings",
>+        "//absl/base:core_headers",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "str_replace_benchmark",
>+    srcs = ["str_replace_benchmark.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    tags = ["benchmark"],
>+    visibility = ["//visibility:private"],
>+    deps = [
>+        ":strings",
>+        "//absl/base",
>+        "@com_github_google_benchmark//:benchmark_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "str_replace_test",
>+    size = "small",
>+    srcs = ["str_replace_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    visibility = ["//visibility:private"],
>+    deps = [
>+        ":strings",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "str_split_test",
>+    srcs = ["str_split_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    visibility = ["//visibility:private"],
>+    deps = [
>+        ":strings",
>+        "//absl/base:core_headers",
>+        "//absl/base:dynamic_annotations",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "str_split_benchmark",
>+    srcs = ["str_split_benchmark.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    tags = ["benchmark"],
>+    visibility = ["//visibility:private"],
>+    deps = [
>+        ":strings",
>+        "//absl/base",
>+        "@com_github_google_benchmark//:benchmark_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "ostringstream_test",
>+    size = "small",
>+    srcs = ["internal/ostringstream_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    visibility = ["//visibility:private"],
>+    deps = [
>+        ":internal",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "ostringstream_benchmark",
>+    srcs = ["internal/ostringstream_benchmark.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    tags = ["benchmark"],
>+    visibility = ["//visibility:private"],
>+    deps = [
>+        ":internal",
>+        "@com_github_google_benchmark//:benchmark_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "resize_uninitialized_test",
>+    size = "small",
>+    srcs = [
>+        "internal/resize_uninitialized.h",
>+        "internal/resize_uninitialized_test.cc",
>+    ],
>+    copts = ABSL_TEST_COPTS,
>+    visibility = ["//visibility:private"],
>+    deps = [
>+        "//absl/base:core_headers",
>+        "//absl/meta:type_traits",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "str_join_test",
>+    size = "small",
>+    srcs = ["str_join_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    visibility = ["//visibility:private"],
>+    deps = [
>+        ":strings",
>+        "//absl/base:core_headers",
>+        "//absl/memory",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "str_join_benchmark",
>+    srcs = ["str_join_benchmark.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    tags = ["benchmark"],
>+    visibility = ["//visibility:private"],
>+    deps = [
>+        ":strings",
>+        "//absl/memory",
>+        "@com_github_google_benchmark//:benchmark_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "str_cat_test",
>+    size = "small",
>+    srcs = ["str_cat_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    visibility = ["//visibility:private"],
>+    deps = [
>+        ":strings",
>+        "//absl/base:core_headers",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "str_cat_benchmark",
>+    srcs = ["str_cat_benchmark.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    tags = ["benchmark"],
>+    visibility = ["//visibility:private"],
>+    deps = [
>+        ":strings",
>+        "@com_github_google_benchmark//:benchmark_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "numbers_test",
>+    size = "small",
>+    srcs = [
>+        "internal/numbers_test_common.h",
>+        "numbers_test.cc",
>+    ],
>+    copts = ABSL_TEST_COPTS,
>+    visibility = ["//visibility:private"],
>+    deps = [
>+        ":strings",
>+        "//absl/base",
>+        "//absl/base:core_headers",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "numbers_benchmark",
>+    srcs = ["numbers_benchmark.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    tags = ["benchmark"],
>+    visibility = ["//visibility:private"],
>+    deps = [
>+        ":strings",
>+        "//absl/base",
>+        "@com_github_google_benchmark//:benchmark_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "strip_test",
>+    size = "small",
>+    srcs = ["strip_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    visibility = ["//visibility:private"],
>+    deps = [
>+        ":strings",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "char_map_test",
>+    srcs = ["internal/char_map_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    deps = [
>+        ":internal",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "char_map_benchmark",
>+    srcs = ["internal/char_map_benchmark.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    tags = ["benchmark"],
>+    deps = [
>+        ":internal",
>+        "@com_github_google_benchmark//:benchmark_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "charconv_test",
>+    srcs = ["charconv_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    deps = [
>+        ":strings",
>+        "//absl/base",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "charconv_parse_test",
>+    srcs = [
>+        "internal/charconv_parse.h",
>+        "internal/charconv_parse_test.cc",
>+    ],
>+    copts = ABSL_TEST_COPTS,
>+    deps = [
>+        ":strings",
>+        "//absl/base",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "charconv_bigint_test",
>+    srcs = [
>+        "internal/charconv_bigint.h",
>+        "internal/charconv_bigint_test.cc",
>+        "internal/charconv_parse.h",
>+    ],
>+    copts = ABSL_TEST_COPTS,
>+    deps = [
>+        ":strings",
>+        "//absl/base",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "charconv_benchmark",
>+    srcs = [
>+        "charconv_benchmark.cc",
>+    ],
>+    tags = [
>+        "benchmark",
>+    ],
>+    deps = [
>+        ":strings",
>+        "//absl/base",
>+        "@com_github_google_benchmark//:benchmark_main",
>+    ],
>+)
>+
>+cc_library(
>+    name = "str_format",
>+    hdrs = [
>+        "str_format.h",
>+    ],
>+    copts = ABSL_DEFAULT_COPTS,
>+    deps = [
>+        ":str_format_internal",
>+    ],
>+)
>+
>+cc_library(
>+    name = "str_format_internal",
>+    srcs = [
>+        "internal/str_format/arg.cc",
>+        "internal/str_format/bind.cc",
>+        "internal/str_format/extension.cc",
>+        "internal/str_format/float_conversion.cc",
>+        "internal/str_format/output.cc",
>+        "internal/str_format/parser.cc",
>+    ],
>+    hdrs = [
>+        "internal/str_format/arg.h",
>+        "internal/str_format/bind.h",
>+        "internal/str_format/checker.h",
>+        "internal/str_format/extension.h",
>+        "internal/str_format/float_conversion.h",
>+        "internal/str_format/output.h",
>+        "internal/str_format/parser.h",
>+    ],
>+    copts = ABSL_DEFAULT_COPTS,
>+    visibility = ["//visibility:private"],
>+    deps = [
>+        ":strings",
>+        "//absl/base:core_headers",
>+        "//absl/container:inlined_vector",
>+        "//absl/meta:type_traits",
>+        "//absl/numeric:int128",
>+        "//absl/types:span",
>+    ],
>+)
>+
>+cc_test(
>+    name = "str_format_test",
>+    srcs = ["str_format_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    visibility = ["//visibility:private"],
>+    deps = [
>+        ":str_format",
>+        ":strings",
>+        "//absl/base:core_headers",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "str_format_extension_test",
>+    srcs = [
>+        "internal/str_format/extension_test.cc",
>+    ],
>+    copts = ABSL_TEST_COPTS,
>+    visibility = ["//visibility:private"],
>+    deps = [
>+        ":str_format",
>+        ":str_format_internal",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "str_format_arg_test",
>+    srcs = ["internal/str_format/arg_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    visibility = ["//visibility:private"],
>+    deps = [
>+        ":str_format",
>+        ":str_format_internal",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "str_format_bind_test",
>+    srcs = ["internal/str_format/bind_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    visibility = ["//visibility:private"],
>+    deps = [
>+        ":str_format_internal",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "str_format_checker_test",
>+    srcs = ["internal/str_format/checker_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    visibility = ["//visibility:private"],
>+    deps = [
>+        ":str_format",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "str_format_convert_test",
>+    size = "small",
>+    srcs = ["internal/str_format/convert_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    visibility = ["//visibility:private"],
>+    deps = [
>+        ":str_format_internal",
>+        "//absl/numeric:int128",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "str_format_output_test",
>+    srcs = ["internal/str_format/output_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    visibility = ["//visibility:private"],
>+    deps = [
>+        ":str_format_internal",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "str_format_parser_test",
>+    srcs = ["internal/str_format/parser_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    visibility = ["//visibility:private"],
>+    deps = [
>+        ":str_format_internal",
>+        "//absl/base:core_headers",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/BUILD.gn
>new file mode 100644
>index 00000000000..9a8a10d0e71
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/BUILD.gn
>@@ -0,0 +1,146 @@
>+# Copyright 2018 The Chromium Authors. All rights reserved.
>+# Use of this source code is governed by a BSD-style license that can be
>+# found in the LICENSE file.
>+
>+import("//build_overrides/build.gni")
>+
>+if (build_with_chromium) {
>+  visibility = [
>+    "//third_party/webrtc/*",
>+    "//third_party/abseil-cpp/*",
>+    "//third_party/googletest:gtest",
>+  ]
>+} else {
>+  visibility = [ "*" ]
>+}
>+
>+source_set("strings") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  sources = [
>+    "ascii.cc",
>+    "charconv.cc",
>+    "escaping.cc",
>+    "internal/charconv_bigint.cc",
>+    "internal/charconv_bigint.h",
>+    "internal/charconv_parse.cc",
>+    "internal/charconv_parse.h",
>+    "internal/memutil.cc",
>+    "internal/memutil.h",
>+    "internal/stl_type_traits.h",
>+    "internal/str_join_internal.h",
>+    "internal/str_split_internal.h",
>+    "match.cc",
>+    "numbers.cc",
>+    "str_cat.cc",
>+    "str_replace.cc",
>+    "str_split.cc",
>+    "string_view.cc",
>+    "substitute.cc",
>+  ]
>+  public = [
>+    "ascii.h",
>+    "charconv.h",
>+    "escaping.h",
>+    "match.h",
>+    "numbers.h",
>+    "str_cat.h",
>+    "str_join.h",
>+    "str_replace.h",
>+    "str_split.h",
>+    "string_view.h",
>+    "strip.h",
>+    "substitute.h",
>+  ]
>+  deps = [
>+    ":internal",
>+    "../base",
>+    "../base:config",
>+    "../base:core_headers",
>+    "../base:endian",
>+    "../base:throw_delegate",
>+    "../memory",
>+    "../meta:type_traits",
>+    "../numeric:int128",
>+  ]
>+}
>+
>+source_set("internal") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  sources = [
>+    "internal/ostringstream.cc",
>+    "internal/utf8.cc",
>+  ]
>+  public = [
>+    "internal/bits.h",
>+    "internal/char_map.h",
>+    "internal/ostringstream.h",
>+    "internal/resize_uninitialized.h",
>+    "internal/utf8.h",
>+  ]
>+  deps = [
>+    "../base:core_headers",
>+    "../base:endian",
>+    "../meta:type_traits",
>+  ]
>+}
>+
>+source_set("str_format") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  public = [
>+    "str_format.h",
>+  ]
>+  deps = [
>+    ":str_format_internal",
>+  ]
>+}
>+
>+source_set("str_format_internal") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  sources = [
>+    "internal/str_format/arg.cc",
>+    "internal/str_format/bind.cc",
>+    "internal/str_format/extension.cc",
>+    "internal/str_format/float_conversion.cc",
>+    "internal/str_format/output.cc",
>+    "internal/str_format/parser.cc",
>+  ]
>+  public = [
>+    "internal/str_format/arg.h",
>+    "internal/str_format/bind.h",
>+    "internal/str_format/checker.h",
>+    "internal/str_format/extension.h",
>+    "internal/str_format/float_conversion.h",
>+    "internal/str_format/output.h",
>+    "internal/str_format/parser.h",
>+  ]
>+  visibility = []
>+  visibility += [ ":*" ]
>+  deps = [
>+    ":strings",
>+    "../base:core_headers",
>+    "../container:inlined_vector",
>+    "../meta:type_traits",
>+    "../numeric:int128",
>+    "../types:span",
>+  ]
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/CMakeLists.txt b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/CMakeLists.txt
>new file mode 100644
>index 00000000000..cd122134729
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/CMakeLists.txt
>@@ -0,0 +1,464 @@
>+#
>+# Copyright 2017 The Abseil Authors.
>+#
>+# Licensed under the Apache License, Version 2.0 (the "License");
>+# you may not use this file except in compliance with the License.
>+# You may obtain a copy of the License at
>+#
>+#      http://www.apache.org/licenses/LICENSE-2.0
>+#
>+# Unless required by applicable law or agreed to in writing, software
>+# distributed under the License is distributed on an "AS IS" BASIS,
>+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+# See the License for the specific language governing permissions and
>+# limitations under the License.
>+#
>+
>+
>+list(APPEND STRINGS_PUBLIC_HEADERS
>+  "ascii.h"
>+  "charconv.h"
>+  "escaping.h"
>+  "match.h"
>+  "numbers.h"
>+  "str_cat.h"
>+  "string_view.h"
>+  "strip.h"
>+  "str_join.h"
>+  "str_replace.h"
>+  "str_split.h"
>+  "substitute.h"
>+)
>+
>+
>+list(APPEND STRINGS_INTERNAL_HEADERS
>+  "internal/bits.h"
>+  "internal/char_map.h"
>+  "internal/charconv_bigint.h"
>+  "internal/charconv_parse.h"
>+  "internal/memutil.h"
>+  "internal/ostringstream.h"
>+  "internal/resize_uninitialized.h"
>+  "internal/stl_type_traits.h"
>+  "internal/str_join_internal.h"
>+  "internal/str_split_internal.h"
>+  "internal/utf8.h"
>+)
>+
>+
>+
>+# add string library
>+list(APPEND STRINGS_SRC
>+  "ascii.cc"
>+  "charconv.cc"
>+  "escaping.cc"
>+  "internal/charconv_bigint.cc"
>+  "internal/charconv_parse.cc"
>+  "internal/memutil.cc"
>+  "internal/memutil.h"
>+  "internal/utf8.cc"
>+  "internal/ostringstream.cc"
>+  "match.cc"
>+  "numbers.cc"
>+  "str_cat.cc"
>+  "str_replace.cc"
>+  "str_split.cc"
>+  "string_view.cc"
>+  "substitute.cc"
>+  ${STRINGS_PUBLIC_HEADERS}
>+  ${STRINGS_INTERNAL_HEADERS}
>+)
>+set(STRINGS_PUBLIC_LIBRARIES absl::base absl_throw_delegate)
>+
>+absl_library(
>+  TARGET
>+    absl_strings
>+  SOURCES
>+    ${STRINGS_SRC}
>+  PUBLIC_LIBRARIES
>+    ${STRINGS_PUBLIC_LIBRARIES}
>+  EXPORT_NAME
>+    strings
>+)
>+
>+# add str_format library
>+absl_header_library(
>+  TARGET
>+    absl_str_format
>+  PUBLIC_LIBRARIES
>+    str_format_internal
>+  EXPORT_NAME
>+    str_format
>+)
>+
>+# str_format_internal
>+absl_library(
>+  TARGET
>+    str_format_internal
>+  SOURCES
>+    "internal/str_format/arg.cc"
>+    "internal/str_format/bind.cc"
>+    "internal/str_format/extension.cc"
>+    "internal/str_format/float_conversion.cc"
>+    "internal/str_format/output.cc"
>+    "internal/str_format/parser.cc"
>+    "internal/str_format/arg.h"
>+    "internal/str_format/bind.h"
>+    "internal/str_format/checker.h"
>+    "internal/str_format/extension.h"
>+    "internal/str_format/float_conversion.h"
>+    "internal/str_format/output.h"
>+    "internal/str_format/parser.h"
>+  PUBLIC_LIBRARIES
>+    str_format_extension_internal
>+    absl::strings
>+    absl::base
>+    absl::numeric
>+    absl::container
>+    absl::span
>+)
>+
>+# str_format_extension_internal
>+absl_library(
>+  TARGET
>+    str_format_extension_internal
>+  SOURCES
>+    "internal/str_format/extension.cc"
>+    "internal/str_format/extension.h"
>+    "internal/str_format/output.cc"
>+    "internal/str_format/output.h"
>+  PUBLIC_LIBRARIES
>+    absl::base
>+    absl::strings
>+)
>+
>+#
>+## TESTS
>+#
>+
>+# test match_test
>+set(MATCH_TEST_SRC "match_test.cc")
>+set(MATCH_TEST_PUBLIC_LIBRARIES absl::strings)
>+
>+absl_test(
>+  TARGET
>+    match_test
>+  SOURCES
>+    ${MATCH_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${MATCH_TEST_PUBLIC_LIBRARIES}
>+)
>+
>+
>+# test escaping_test
>+set(ESCAPING_TEST_SRC "escaping_test.cc")
>+set(ESCAPING_TEST_PUBLIC_LIBRARIES absl::strings absl::base)
>+
>+absl_test(
>+  TARGET
>+    escaping_test
>+  SOURCES
>+    ${ESCAPING_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${ESCAPING_TEST_PUBLIC_LIBRARIES}
>+)
>+
>+
>+# test ascii_test
>+set(ASCII_TEST_SRC "ascii_test.cc")
>+set(ASCII_TEST_PUBLIC_LIBRARIES absl::strings)
>+
>+absl_test(
>+  TARGET
>+    ascii_test
>+  SOURCES
>+    ${ASCII_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${ASCII_TEST_PUBLIC_LIBRARIES}
>+)
>+
>+
>+# test memutil_test
>+set(MEMUTIL_TEST_SRC "internal/memutil_test.cc")
>+set(MEMUTIL_TEST_PUBLIC_LIBRARIES absl::strings)
>+
>+absl_test(
>+  TARGET
>+    memutil_test
>+  SOURCES
>+    ${MEMUTIL_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${MEMUTIL_TEST_PUBLIC_LIBRARIES}
>+)
>+
>+
>+# test utf8_test
>+set(UTF8_TEST_SRC "internal/utf8_test.cc")
>+set(UTF8_TEST_PUBLIC_LIBRARIES absl::strings absl::base)
>+
>+absl_test(
>+  TARGET
>+    utf8_test
>+  SOURCES
>+    ${UTF8_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${UTF8_TEST_PUBLIC_LIBRARIES}
>+)
>+
>+
>+# test string_view_test
>+set(STRING_VIEW_TEST_SRC "string_view_test.cc")
>+set(STRING_VIEW_TEST_PUBLIC_LIBRARIES absl::strings absl_throw_delegate absl::base)
>+
>+absl_test(
>+  TARGET
>+    string_view_test
>+  SOURCES
>+    ${STRING_VIEW_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${STRING_VIEW_TEST_PUBLIC_LIBRARIES}
>+)
>+
>+
>+# test substitute_test
>+set(SUBSTITUTE_TEST_SRC "substitute_test.cc")
>+set(SUBSTITUTE_TEST_PUBLIC_LIBRARIES absl::strings absl::base)
>+
>+absl_test(
>+  TARGET
>+    substitute_test
>+  SOURCES
>+    ${SUBSTITUTE_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${SUBSTITUTE_TEST_PUBLIC_LIBRARIES}
>+)
>+
>+
>+# test str_replace_test
>+set(STR_REPLACE_TEST_SRC "str_replace_test.cc")
>+set(STR_REPLACE_TEST_PUBLIC_LIBRARIES absl::strings absl::base absl_throw_delegate)
>+
>+absl_test(
>+  TARGET
>+    str_replace_test
>+  SOURCES
>+    ${STR_REPLACE_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${STR_REPLACE_TEST_PUBLIC_LIBRARIES}
>+)
>+
>+
>+# test str_split_test
>+set(STR_SPLIT_TEST_SRC "str_split_test.cc")
>+set(STR_SPLIT_TEST_PUBLIC_LIBRARIES absl::strings absl::base absl_throw_delegate)
>+
>+absl_test(
>+  TARGET
>+    str_split_test
>+  SOURCES
>+    ${STR_SPLIT_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${STR_SPLIT_TEST_PUBLIC_LIBRARIES}
>+)
>+
>+
>+# test ostringstream_test
>+set(OSTRINGSTREAM_TEST_SRC "internal/ostringstream_test.cc")
>+set(OSTRINGSTREAM_TEST_PUBLIC_LIBRARIES absl::strings)
>+
>+absl_test(
>+  TARGET
>+    ostringstream_test
>+  SOURCES
>+    ${OSTRINGSTREAM_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${OSTRINGSTREAM_TEST_PUBLIC_LIBRARIES}
>+)
>+
>+
>+# test resize_uninitialized_test
>+set(RESIZE_UNINITIALIZED_TEST_SRC "internal/resize_uninitialized_test.cc")
>+
>+absl_test(
>+  TARGET
>+    resize_uninitialized_test
>+  SOURCES
>+    ${RESIZE_UNINITIALIZED_TEST_SRC}
>+)
>+
>+
>+# test str_join_test
>+set(STR_JOIN_TEST_SRC "str_join_test.cc")
>+set(STR_JOIN_TEST_PUBLIC_LIBRARIES absl::strings)
>+
>+absl_test(
>+  TARGET
>+    str_join_test
>+  SOURCES
>+    ${STR_JOIN_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${STR_JOIN_TEST_PUBLIC_LIBRARIES}
>+)
>+
>+
>+# test str_cat_test
>+set(STR_CAT_TEST_SRC "str_cat_test.cc")
>+set(STR_CAT_TEST_PUBLIC_LIBRARIES absl::strings)
>+
>+absl_test(
>+  TARGET
>+    str_cat_test
>+  SOURCES
>+    ${STR_CAT_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${STR_CAT_TEST_PUBLIC_LIBRARIES}
>+)
>+
>+
>+# test numbers_test
>+set(NUMBERS_TEST_SRC "numbers_test.cc")
>+set(NUMBERS_TEST_PUBLIC_LIBRARIES absl::strings)
>+
>+absl_test(
>+  TARGET
>+    numbers_test
>+  SOURCES
>+    ${NUMBERS_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${NUMBERS_TEST_PUBLIC_LIBRARIES}
>+)
>+
>+
>+# test strip_test
>+set(STRIP_TEST_SRC "strip_test.cc")
>+set(STRIP_TEST_PUBLIC_LIBRARIES absl::strings)
>+
>+absl_test(
>+  TARGET
>+    strip_test
>+  SOURCES
>+    ${STRIP_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${STRIP_TEST_PUBLIC_LIBRARIES}
>+)
>+
>+
>+# test char_map_test
>+set(CHAR_MAP_TEST_SRC "internal/char_map_test.cc")
>+set(CHAR_MAP_TEST_PUBLIC_LIBRARIES absl::strings)
>+
>+absl_test(
>+  TARGET
>+    char_map_test
>+  SOURCES
>+    ${CHAR_MAP_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${CHAR_MAP_TEST_PUBLIC_LIBRARIES}
>+)
>+
>+
>+# test charconv_test
>+set(CHARCONV_TEST_SRC "charconv_test.cc")
>+set(CHARCONV_TEST_PUBLIC_LIBRARIES absl::strings)
>+
>+absl_test(
>+  TARGET
>+    charconv_test
>+  SOURCES
>+    ${CHARCONV_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${CHARCONV_TEST_PUBLIC_LIBRARIES}
>+)
>+
>+
>+# test charconv_parse_test
>+set(CHARCONV_PARSE_TEST_SRC "internal/charconv_parse_test.cc")
>+set(CHARCONV_PARSE_TEST_PUBLIC_LIBRARIES absl::strings)
>+
>+absl_test(
>+  TARGET
>+    charconv_parse_test
>+  SOURCES
>+    ${CHARCONV_PARSE_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${CHARCONV_PARSE_TEST_PUBLIC_LIBRARIES}
>+)
>+
>+
>+# test charconv_bigint_test
>+set(CHARCONV_BIGINT_TEST_SRC "internal/charconv_bigint_test.cc")
>+set(CHARCONV_BIGINT_TEST_PUBLIC_LIBRARIES absl::strings)
>+
>+absl_test(
>+  TARGET
>+    charconv_bigint_test
>+  SOURCES
>+    ${CHARCONV_BIGINT_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${CHARCONV_BIGINT_TEST_PUBLIC_LIBRARIES}
>+)
>+# test str_format_test
>+absl_test(
>+  TARGET
>+    str_format_test
>+  SOURCES
>+    "str_format_test.cc"
>+  PUBLIC_LIBRARIES
>+    absl::base
>+    absl::str_format
>+    absl::strings
>+)
>+
>+# test str_format_bind_test
>+absl_test(
>+  TARGET
>+    str_format_bind_test
>+  SOURCES
>+    "internal/str_format/bind_test.cc"
>+  PUBLIC_LIBRARIES
>+    str_format_internal
>+)
>+
>+# test str_format_checker_test
>+absl_test(
>+  TARGET
>+    str_format_checker_test
>+  SOURCES
>+    "internal/str_format/checker_test.cc"
>+  PUBLIC_LIBRARIES
>+    absl::str_format
>+)
>+
>+# test str_format_convert_test
>+absl_test(
>+  TARGET
>+    str_format_convert_test
>+  SOURCES
>+    "internal/str_format/convert_test.cc"
>+  PUBLIC_LIBRARIES
>+    str_format_internal
>+    absl::numeric
>+)
>+
>+# test str_format_output_test
>+absl_test(
>+  TARGET
>+    str_format_output_test
>+  SOURCES
>+    "internal/str_format/output_test.cc"
>+  PUBLIC_LIBRARIES
>+    str_format_extension_internal
>+)
>+
>+# test str_format_parser_test
>+absl_test(
>+  TARGET
>+    str_format_parser_test
>+  SOURCES
>+    "internal/str_format/parser_test.cc"
>+  PUBLIC_LIBRARIES
>+    str_format_internal
>+    absl::base
>+)
>+
>+
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/ascii.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/ascii.cc
>new file mode 100644
>index 00000000000..c9481e88654
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/ascii.cc
>@@ -0,0 +1,198 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/strings/ascii.h"
>+
>+namespace absl {
>+namespace ascii_internal {
>+
>+// # Table generated by this Python code (bit 0x02 is currently unused):
>+// TODO(mbar) Move Python code for generation of table to BUILD and link here.
>+
>+// NOTE: The kAsciiPropertyBits table used within this code was generated by
>+// Python code of the following form. (Bit 0x02 is currently unused and
>+// available.)
>+//
>+// def Hex2(n):
>+//   return '0x' + hex(n/16)[2:] + hex(n%16)[2:]
>+// def IsPunct(ch):
>+//   return (ord(ch) >= 32 and ord(ch) < 127 and
>+//           not ch.isspace() and not ch.isalnum())
>+// def IsBlank(ch):
>+//   return ch in ' \t'
>+// def IsCntrl(ch):
>+//   return ord(ch) < 32 or ord(ch) == 127
>+// def IsXDigit(ch):
>+//   return ch.isdigit() or ch.lower() in 'abcdef'
>+// for i in range(128):
>+//   ch = chr(i)
>+//   mask = ((ch.isalpha() and 0x01 or 0) |
>+//           (ch.isalnum() and 0x04 or 0) |
>+//           (ch.isspace() and 0x08 or 0) |
>+//           (IsPunct(ch) and 0x10 or 0) |
>+//           (IsBlank(ch) and 0x20 or 0) |
>+//           (IsCntrl(ch) and 0x40 or 0) |
>+//           (IsXDigit(ch) and 0x80 or 0))
>+//   print Hex2(mask) + ',',
>+//   if i % 16 == 7:
>+//     print ' //', Hex2(i & 0x78)
>+//   elif i % 16 == 15:
>+//     print
>+
>+// clang-format off
>+// Array of bitfields holding character information. Each bit value corresponds
>+// to a particular character feature. For readability, and because the value
>+// of these bits is tightly coupled to this implementation, the individual bits
>+// are not named. Note that bitfields for all characters above ASCII 127 are
>+// zero-initialized.
>+const unsigned char kPropertyBits[256] = {
>+    0x40, 0x40, 0x40, 0x40, 0x40, 0x40, 0x40, 0x40,  // 0x00
>+    0x40, 0x68, 0x48, 0x48, 0x48, 0x48, 0x40, 0x40,
>+    0x40, 0x40, 0x40, 0x40, 0x40, 0x40, 0x40, 0x40,  // 0x10
>+    0x40, 0x40, 0x40, 0x40, 0x40, 0x40, 0x40, 0x40,
>+    0x28, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10,  // 0x20
>+    0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10,
>+    0x84, 0x84, 0x84, 0x84, 0x84, 0x84, 0x84, 0x84,  // 0x30
>+    0x84, 0x84, 0x10, 0x10, 0x10, 0x10, 0x10, 0x10,
>+    0x10, 0x85, 0x85, 0x85, 0x85, 0x85, 0x85, 0x05,  // 0x40
>+    0x05, 0x05, 0x05, 0x05, 0x05, 0x05, 0x05, 0x05,
>+    0x05, 0x05, 0x05, 0x05, 0x05, 0x05, 0x05, 0x05,  // 0x50
>+    0x05, 0x05, 0x05, 0x10, 0x10, 0x10, 0x10, 0x10,
>+    0x10, 0x85, 0x85, 0x85, 0x85, 0x85, 0x85, 0x05,  // 0x60
>+    0x05, 0x05, 0x05, 0x05, 0x05, 0x05, 0x05, 0x05,
>+    0x05, 0x05, 0x05, 0x05, 0x05, 0x05, 0x05, 0x05,  // 0x70
>+    0x05, 0x05, 0x05, 0x10, 0x10, 0x10, 0x10, 0x40,
>+};
>+
>+// Array of characters for the ascii_tolower() function. For values 'A'
>+// through 'Z', return the lower-case character; otherwise, return the
>+// identity of the passed character.
>+const char kToLower[256] = {
>+  '\x00', '\x01', '\x02', '\x03', '\x04', '\x05', '\x06', '\x07',
>+  '\x08', '\x09', '\x0a', '\x0b', '\x0c', '\x0d', '\x0e', '\x0f',
>+  '\x10', '\x11', '\x12', '\x13', '\x14', '\x15', '\x16', '\x17',
>+  '\x18', '\x19', '\x1a', '\x1b', '\x1c', '\x1d', '\x1e', '\x1f',
>+  '\x20', '\x21', '\x22', '\x23', '\x24', '\x25', '\x26', '\x27',
>+  '\x28', '\x29', '\x2a', '\x2b', '\x2c', '\x2d', '\x2e', '\x2f',
>+  '\x30', '\x31', '\x32', '\x33', '\x34', '\x35', '\x36', '\x37',
>+  '\x38', '\x39', '\x3a', '\x3b', '\x3c', '\x3d', '\x3e', '\x3f',
>+  '\x40',    'a',    'b',    'c',    'd',    'e',    'f',    'g',
>+     'h',    'i',    'j',    'k',    'l',    'm',    'n',    'o',
>+     'p',    'q',    'r',    's',    't',    'u',    'v',    'w',
>+     'x',    'y',    'z', '\x5b', '\x5c', '\x5d', '\x5e', '\x5f',
>+  '\x60', '\x61', '\x62', '\x63', '\x64', '\x65', '\x66', '\x67',
>+  '\x68', '\x69', '\x6a', '\x6b', '\x6c', '\x6d', '\x6e', '\x6f',
>+  '\x70', '\x71', '\x72', '\x73', '\x74', '\x75', '\x76', '\x77',
>+  '\x78', '\x79', '\x7a', '\x7b', '\x7c', '\x7d', '\x7e', '\x7f',
>+  '\x80', '\x81', '\x82', '\x83', '\x84', '\x85', '\x86', '\x87',
>+  '\x88', '\x89', '\x8a', '\x8b', '\x8c', '\x8d', '\x8e', '\x8f',
>+  '\x90', '\x91', '\x92', '\x93', '\x94', '\x95', '\x96', '\x97',
>+  '\x98', '\x99', '\x9a', '\x9b', '\x9c', '\x9d', '\x9e', '\x9f',
>+  '\xa0', '\xa1', '\xa2', '\xa3', '\xa4', '\xa5', '\xa6', '\xa7',
>+  '\xa8', '\xa9', '\xaa', '\xab', '\xac', '\xad', '\xae', '\xaf',
>+  '\xb0', '\xb1', '\xb2', '\xb3', '\xb4', '\xb5', '\xb6', '\xb7',
>+  '\xb8', '\xb9', '\xba', '\xbb', '\xbc', '\xbd', '\xbe', '\xbf',
>+  '\xc0', '\xc1', '\xc2', '\xc3', '\xc4', '\xc5', '\xc6', '\xc7',
>+  '\xc8', '\xc9', '\xca', '\xcb', '\xcc', '\xcd', '\xce', '\xcf',
>+  '\xd0', '\xd1', '\xd2', '\xd3', '\xd4', '\xd5', '\xd6', '\xd7',
>+  '\xd8', '\xd9', '\xda', '\xdb', '\xdc', '\xdd', '\xde', '\xdf',
>+  '\xe0', '\xe1', '\xe2', '\xe3', '\xe4', '\xe5', '\xe6', '\xe7',
>+  '\xe8', '\xe9', '\xea', '\xeb', '\xec', '\xed', '\xee', '\xef',
>+  '\xf0', '\xf1', '\xf2', '\xf3', '\xf4', '\xf5', '\xf6', '\xf7',
>+  '\xf8', '\xf9', '\xfa', '\xfb', '\xfc', '\xfd', '\xfe', '\xff',
>+};
>+
>+// Array of characters for the ascii_toupper() function. For values 'a'
>+// through 'z', return the upper-case character; otherwise, return the
>+// identity of the passed character.
>+const char kToUpper[256] = {
>+  '\x00', '\x01', '\x02', '\x03', '\x04', '\x05', '\x06', '\x07',
>+  '\x08', '\x09', '\x0a', '\x0b', '\x0c', '\x0d', '\x0e', '\x0f',
>+  '\x10', '\x11', '\x12', '\x13', '\x14', '\x15', '\x16', '\x17',
>+  '\x18', '\x19', '\x1a', '\x1b', '\x1c', '\x1d', '\x1e', '\x1f',
>+  '\x20', '\x21', '\x22', '\x23', '\x24', '\x25', '\x26', '\x27',
>+  '\x28', '\x29', '\x2a', '\x2b', '\x2c', '\x2d', '\x2e', '\x2f',
>+  '\x30', '\x31', '\x32', '\x33', '\x34', '\x35', '\x36', '\x37',
>+  '\x38', '\x39', '\x3a', '\x3b', '\x3c', '\x3d', '\x3e', '\x3f',
>+  '\x40', '\x41', '\x42', '\x43', '\x44', '\x45', '\x46', '\x47',
>+  '\x48', '\x49', '\x4a', '\x4b', '\x4c', '\x4d', '\x4e', '\x4f',
>+  '\x50', '\x51', '\x52', '\x53', '\x54', '\x55', '\x56', '\x57',
>+  '\x58', '\x59', '\x5a', '\x5b', '\x5c', '\x5d', '\x5e', '\x5f',
>+  '\x60',    'A',    'B',    'C',    'D',    'E',    'F',    'G',
>+     'H',    'I',    'J',    'K',    'L',    'M',    'N',    'O',
>+     'P',    'Q',    'R',    'S',    'T',    'U',    'V',    'W',
>+     'X',    'Y',    'Z', '\x7b', '\x7c', '\x7d', '\x7e', '\x7f',
>+  '\x80', '\x81', '\x82', '\x83', '\x84', '\x85', '\x86', '\x87',
>+  '\x88', '\x89', '\x8a', '\x8b', '\x8c', '\x8d', '\x8e', '\x8f',
>+  '\x90', '\x91', '\x92', '\x93', '\x94', '\x95', '\x96', '\x97',
>+  '\x98', '\x99', '\x9a', '\x9b', '\x9c', '\x9d', '\x9e', '\x9f',
>+  '\xa0', '\xa1', '\xa2', '\xa3', '\xa4', '\xa5', '\xa6', '\xa7',
>+  '\xa8', '\xa9', '\xaa', '\xab', '\xac', '\xad', '\xae', '\xaf',
>+  '\xb0', '\xb1', '\xb2', '\xb3', '\xb4', '\xb5', '\xb6', '\xb7',
>+  '\xb8', '\xb9', '\xba', '\xbb', '\xbc', '\xbd', '\xbe', '\xbf',
>+  '\xc0', '\xc1', '\xc2', '\xc3', '\xc4', '\xc5', '\xc6', '\xc7',
>+  '\xc8', '\xc9', '\xca', '\xcb', '\xcc', '\xcd', '\xce', '\xcf',
>+  '\xd0', '\xd1', '\xd2', '\xd3', '\xd4', '\xd5', '\xd6', '\xd7',
>+  '\xd8', '\xd9', '\xda', '\xdb', '\xdc', '\xdd', '\xde', '\xdf',
>+  '\xe0', '\xe1', '\xe2', '\xe3', '\xe4', '\xe5', '\xe6', '\xe7',
>+  '\xe8', '\xe9', '\xea', '\xeb', '\xec', '\xed', '\xee', '\xef',
>+  '\xf0', '\xf1', '\xf2', '\xf3', '\xf4', '\xf5', '\xf6', '\xf7',
>+  '\xf8', '\xf9', '\xfa', '\xfb', '\xfc', '\xfd', '\xfe', '\xff',
>+};
>+// clang-format on
>+
>+}  // namespace ascii_internal
>+
>+void AsciiStrToLower(std::string* s) {
>+  for (auto& ch : *s) {
>+    ch = absl::ascii_tolower(ch);
>+  }
>+}
>+
>+void AsciiStrToUpper(std::string* s) {
>+  for (auto& ch : *s) {
>+    ch = absl::ascii_toupper(ch);
>+  }
>+}
>+
>+void RemoveExtraAsciiWhitespace(std::string* str) {
>+  auto stripped = StripAsciiWhitespace(*str);
>+
>+  if (stripped.empty()) {
>+    str->clear();
>+    return;
>+  }
>+
>+  auto input_it = stripped.begin();
>+  auto input_end = stripped.end();
>+  auto output_it = &(*str)[0];
>+  bool is_ws = false;
>+
>+  for (; input_it < input_end; ++input_it) {
>+    if (is_ws) {
>+      // Consecutive whitespace?  Keep only the last.
>+      is_ws = absl::ascii_isspace(*input_it);
>+      if (is_ws) --output_it;
>+    } else {
>+      is_ws = absl::ascii_isspace(*input_it);
>+    }
>+
>+    *output_it = *input_it;
>+    ++output_it;
>+  }
>+
>+  str->erase(output_it - &(*str)[0]);
>+}
>+
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/ascii.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/ascii.h
>new file mode 100644
>index 00000000000..96a64541c32
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/ascii.h
>@@ -0,0 +1,239 @@
>+//
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// -----------------------------------------------------------------------------
>+// File: ascii.h
>+// -----------------------------------------------------------------------------
>+//
>+// This package contains functions operating on characters and strings
>+// restricted to standard ASCII. These include character classification
>+// functions analogous to those found in the ANSI C Standard Library <ctype.h>
>+// header file.
>+//
>+// C++ implementations provide <ctype.h> functionality based on their
>+// C environment locale. In general, reliance on such a locale is not ideal, as
>+// the locale standard is problematic (and may not return invariant information
>+// for the same character set, for example). These `ascii_*()` functions are
>+// hard-wired for standard ASCII, much faster, and guaranteed to behave
>+// consistently.  They will never be overloaded, nor will their function
>+// signature change.
>+//
>+// `ascii_isalnum()`, `ascii_isalpha()`, `ascii_isascii()`, `ascii_isblank()`,
>+// `ascii_iscntrl()`, `ascii_isdigit()`, `ascii_isgraph()`, `ascii_islower()`,
>+// `ascii_isprint()`, `ascii_ispunct()`, `ascii_isspace()`, `ascii_isupper()`,
>+// `ascii_isxdigit()`
>+//   Analogous to the <ctype.h> functions with similar names, these
>+//   functions take an unsigned char and return a bool, based on whether the
>+//   character matches the condition specified.
>+//
>+//   If the input character has a numerical value greater than 127, these
>+//   functions return `false`.
>+//
>+// `ascii_tolower()`, `ascii_toupper()`
>+//   Analogous to the <ctype.h> functions with similar names, these functions
>+//   take an unsigned char and return a char.
>+//
>+//   If the input character is not an ASCII {lower,upper}-case letter (including
>+//   numerical values greater than 127) then the functions return the same value
>+//   as the input character.
>+
>+#ifndef ABSL_STRINGS_ASCII_H_
>+#define ABSL_STRINGS_ASCII_H_
>+
>+#include <algorithm>
>+#include <string>
>+
>+#include "absl/base/attributes.h"
>+#include "absl/strings/string_view.h"
>+
>+namespace absl {
>+namespace ascii_internal {
>+
>+// Declaration for an array of bitfields holding character information.
>+extern const unsigned char kPropertyBits[256];
>+
>+// Declaration for the array of characters to upper-case characters.
>+extern const char kToUpper[256];
>+
>+// Declaration for the array of characters to lower-case characters.
>+extern const char kToLower[256];
>+
>+}  // namespace ascii_internal
>+
>+// ascii_isalpha()
>+//
>+// Determines whether the given character is an alphabetic character.
>+inline bool ascii_isalpha(unsigned char c) {
>+  return (ascii_internal::kPropertyBits[c] & 0x01) != 0;
>+}
>+
>+// ascii_isalnum()
>+//
>+// Determines whether the given character is an alphanumeric character.
>+inline bool ascii_isalnum(unsigned char c) {
>+  return (ascii_internal::kPropertyBits[c] & 0x04) != 0;
>+}
>+
>+// ascii_isspace()
>+//
>+// Determines whether the given character is a whitespace character (space,
>+// tab, vertical tab, formfeed, linefeed, or carriage return).
>+inline bool ascii_isspace(unsigned char c) {
>+  return (ascii_internal::kPropertyBits[c] & 0x08) != 0;
>+}
>+
>+// ascii_ispunct()
>+//
>+// Determines whether the given character is a punctuation character.
>+inline bool ascii_ispunct(unsigned char c) {
>+  return (ascii_internal::kPropertyBits[c] & 0x10) != 0;
>+}
>+
>+// ascii_isblank()
>+//
>+// Determines whether the given character is a blank character (tab or space).
>+inline bool ascii_isblank(unsigned char c) {
>+  return (ascii_internal::kPropertyBits[c] & 0x20) != 0;
>+}
>+
>+// ascii_iscntrl()
>+//
>+// Determines whether the given character is a control character.
>+inline bool ascii_iscntrl(unsigned char c) {
>+  return (ascii_internal::kPropertyBits[c] & 0x40) != 0;
>+}
>+
>+// ascii_isxdigit()
>+//
>+// Determines whether the given character can be represented as a hexadecimal
>+// digit character (i.e. {0-9} or {A-F}).
>+inline bool ascii_isxdigit(unsigned char c) {
>+  return (ascii_internal::kPropertyBits[c] & 0x80) != 0;
>+}
>+
>+// ascii_isdigit()
>+//
>+// Determines whether the given character can be represented as a decimal
>+// digit character (i.e. {0-9}).
>+inline bool ascii_isdigit(unsigned char c) { return c >= '0' && c <= '9'; }
>+
>+// ascii_isprint()
>+//
>+// Determines whether the given character is printable, including whitespace.
>+inline bool ascii_isprint(unsigned char c) { return c >= 32 && c < 127; }
>+
>+// ascii_isgraph()
>+//
>+// Determines whether the given character has a graphical representation.
>+inline bool ascii_isgraph(unsigned char c) { return c > 32 && c < 127; }
>+
>+// ascii_isupper()
>+//
>+// Determines whether the given character is uppercase.
>+inline bool ascii_isupper(unsigned char c) { return c >= 'A' && c <= 'Z'; }
>+
>+// ascii_islower()
>+//
>+// Determines whether the given character is lowercase.
>+inline bool ascii_islower(unsigned char c) { return c >= 'a' && c <= 'z'; }
>+
>+// ascii_isascii()
>+//
>+// Determines whether the given character is ASCII.
>+inline bool ascii_isascii(unsigned char c) { return c < 128; }
>+
>+// ascii_tolower()
>+//
>+// Returns an ASCII character, converting to lowercase if uppercase is
>+// passed. Note that character values > 127 are simply returned.
>+inline char ascii_tolower(unsigned char c) {
>+  return ascii_internal::kToLower[c];
>+}
>+
>+// Converts the characters in `s` to lowercase, changing the contents of `s`.
>+void AsciiStrToLower(std::string* s);
>+
>+// Creates a lowercase std::string from a given absl::string_view.
>+ABSL_MUST_USE_RESULT inline std::string AsciiStrToLower(absl::string_view s) {
>+  std::string result(s);
>+  absl::AsciiStrToLower(&result);
>+  return result;
>+}
>+
>+// ascii_toupper()
>+//
>+// Returns the ASCII character, converting to upper-case if lower-case is
>+// passed. Note that characters values > 127 are simply returned.
>+inline char ascii_toupper(unsigned char c) {
>+  return ascii_internal::kToUpper[c];
>+}
>+
>+// Converts the characters in `s` to uppercase, changing the contents of `s`.
>+void AsciiStrToUpper(std::string* s);
>+
>+// Creates an uppercase std::string from a given absl::string_view.
>+ABSL_MUST_USE_RESULT inline std::string AsciiStrToUpper(absl::string_view s) {
>+  std::string result(s);
>+  absl::AsciiStrToUpper(&result);
>+  return result;
>+}
>+
>+// Returns absl::string_view with whitespace stripped from the beginning of the
>+// given string_view.
>+ABSL_MUST_USE_RESULT inline absl::string_view StripLeadingAsciiWhitespace(
>+    absl::string_view str) {
>+  auto it = std::find_if_not(str.begin(), str.end(), absl::ascii_isspace);
>+  return absl::string_view(it, str.end() - it);
>+}
>+
>+// Strips in place whitespace from the beginning of the given std::string.
>+inline void StripLeadingAsciiWhitespace(std::string* str) {
>+  auto it = std::find_if_not(str->begin(), str->end(), absl::ascii_isspace);
>+  str->erase(str->begin(), it);
>+}
>+
>+// Returns absl::string_view with whitespace stripped from the end of the given
>+// string_view.
>+ABSL_MUST_USE_RESULT inline absl::string_view StripTrailingAsciiWhitespace(
>+    absl::string_view str) {
>+  auto it = std::find_if_not(str.rbegin(), str.rend(), absl::ascii_isspace);
>+  return absl::string_view(str.begin(), str.rend() - it);
>+}
>+
>+// Strips in place whitespace from the end of the given std::string
>+inline void StripTrailingAsciiWhitespace(std::string* str) {
>+  auto it = std::find_if_not(str->rbegin(), str->rend(), absl::ascii_isspace);
>+  str->erase(str->rend() - it);
>+}
>+
>+// Returns absl::string_view with whitespace stripped from both ends of the
>+// given string_view.
>+ABSL_MUST_USE_RESULT inline absl::string_view StripAsciiWhitespace(
>+    absl::string_view str) {
>+  return StripTrailingAsciiWhitespace(StripLeadingAsciiWhitespace(str));
>+}
>+
>+// Strips in place whitespace from both ends of the given std::string
>+inline void StripAsciiWhitespace(std::string* str) {
>+  StripTrailingAsciiWhitespace(str);
>+  StripLeadingAsciiWhitespace(str);
>+}
>+
>+// Removes leading, trailing, and consecutive internal whitespace.
>+void RemoveExtraAsciiWhitespace(std::string*);
>+
>+}  // namespace absl
>+
>+#endif  // ABSL_STRINGS_ASCII_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/ascii_benchmark.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/ascii_benchmark.cc
>new file mode 100644
>index 00000000000..8dea4b8cfbf
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/ascii_benchmark.cc
>@@ -0,0 +1,120 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/strings/ascii.h"
>+
>+#include <cctype>
>+#include <string>
>+#include <array>
>+#include <random>
>+
>+#include "benchmark/benchmark.h"
>+
>+namespace {
>+
>+std::array<unsigned char, 256> MakeShuffledBytes() {
>+  std::array<unsigned char, 256> bytes;
>+  for (size_t i = 0; i < 256; ++i) bytes[i] = static_cast<unsigned char>(i);
>+  std::random_device rd;
>+  std::seed_seq seed({rd(), rd(), rd(), rd(), rd(), rd(), rd(), rd()});
>+  std::mt19937 g(seed);
>+  std::shuffle(bytes.begin(), bytes.end(), g);
>+  return bytes;
>+}
>+
>+template <typename Function>
>+void AsciiBenchmark(benchmark::State& state, Function f) {
>+  std::array<unsigned char, 256> bytes = MakeShuffledBytes();
>+  size_t sum = 0;
>+  for (auto _ : state) {
>+    for (unsigned char b : bytes) sum += f(b) ? 1 : 0;
>+  }
>+  // Make a copy of `sum` before calling `DoNotOptimize` to make sure that `sum`
>+  // can be put in a CPU register and not degrade performance in the loop above.
>+  size_t sum2 = sum;
>+  benchmark::DoNotOptimize(sum2);
>+  state.SetBytesProcessed(state.iterations() * bytes.size());
>+}
>+
>+using StdAsciiFunction = int (*)(int);
>+template <StdAsciiFunction f>
>+void BM_Ascii(benchmark::State& state) {
>+  AsciiBenchmark(state, f);
>+}
>+
>+using AbslAsciiIsFunction = bool (*)(unsigned char);
>+template <AbslAsciiIsFunction f>
>+void BM_Ascii(benchmark::State& state) {
>+  AsciiBenchmark(state, f);
>+}
>+
>+using AbslAsciiToFunction = char (*)(unsigned char);
>+template <AbslAsciiToFunction f>
>+void BM_Ascii(benchmark::State& state) {
>+  AsciiBenchmark(state, f);
>+}
>+
>+inline char Noop(unsigned char b) { return static_cast<char>(b); }
>+
>+BENCHMARK_TEMPLATE(BM_Ascii, Noop);
>+BENCHMARK_TEMPLATE(BM_Ascii, std::isalpha);
>+BENCHMARK_TEMPLATE(BM_Ascii, absl::ascii_isalpha);
>+BENCHMARK_TEMPLATE(BM_Ascii, std::isdigit);
>+BENCHMARK_TEMPLATE(BM_Ascii, absl::ascii_isdigit);
>+BENCHMARK_TEMPLATE(BM_Ascii, std::isalnum);
>+BENCHMARK_TEMPLATE(BM_Ascii, absl::ascii_isalnum);
>+BENCHMARK_TEMPLATE(BM_Ascii, std::isspace);
>+BENCHMARK_TEMPLATE(BM_Ascii, absl::ascii_isspace);
>+BENCHMARK_TEMPLATE(BM_Ascii, std::ispunct);
>+BENCHMARK_TEMPLATE(BM_Ascii, absl::ascii_ispunct);
>+BENCHMARK_TEMPLATE(BM_Ascii, std::isblank);
>+BENCHMARK_TEMPLATE(BM_Ascii, absl::ascii_isblank);
>+BENCHMARK_TEMPLATE(BM_Ascii, std::iscntrl);
>+BENCHMARK_TEMPLATE(BM_Ascii, absl::ascii_iscntrl);
>+BENCHMARK_TEMPLATE(BM_Ascii, std::isxdigit);
>+BENCHMARK_TEMPLATE(BM_Ascii, absl::ascii_isxdigit);
>+BENCHMARK_TEMPLATE(BM_Ascii, std::isprint);
>+BENCHMARK_TEMPLATE(BM_Ascii, absl::ascii_isprint);
>+BENCHMARK_TEMPLATE(BM_Ascii, std::isgraph);
>+BENCHMARK_TEMPLATE(BM_Ascii, absl::ascii_isgraph);
>+BENCHMARK_TEMPLATE(BM_Ascii, std::isupper);
>+BENCHMARK_TEMPLATE(BM_Ascii, absl::ascii_isupper);
>+BENCHMARK_TEMPLATE(BM_Ascii, std::islower);
>+BENCHMARK_TEMPLATE(BM_Ascii, absl::ascii_islower);
>+BENCHMARK_TEMPLATE(BM_Ascii, isascii);
>+BENCHMARK_TEMPLATE(BM_Ascii, absl::ascii_isascii);
>+BENCHMARK_TEMPLATE(BM_Ascii, std::tolower);
>+BENCHMARK_TEMPLATE(BM_Ascii, absl::ascii_tolower);
>+BENCHMARK_TEMPLATE(BM_Ascii, std::toupper);
>+BENCHMARK_TEMPLATE(BM_Ascii, absl::ascii_toupper);
>+
>+static void BM_StrToLower(benchmark::State& state) {
>+  const int size = state.range(0);
>+  std::string s(size, 'X');
>+  for (auto _ : state) {
>+    benchmark::DoNotOptimize(absl::AsciiStrToLower(s));
>+  }
>+}
>+BENCHMARK(BM_StrToLower)->Range(1, 1 << 20);
>+
>+static void BM_StrToUpper(benchmark::State& state) {
>+  const int size = state.range(0);
>+  std::string s(size, 'x');
>+  for (auto _ : state) {
>+    benchmark::DoNotOptimize(absl::AsciiStrToUpper(s));
>+  }
>+}
>+BENCHMARK(BM_StrToUpper)->Range(1, 1 << 20);
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/ascii_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/ascii_test.cc
>new file mode 100644
>index 00000000000..9903b0496b7
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/ascii_test.cc
>@@ -0,0 +1,361 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/strings/ascii.h"
>+
>+#include <cctype>
>+#include <clocale>
>+#include <cstring>
>+#include <string>
>+
>+#include "gtest/gtest.h"
>+#include "absl/base/macros.h"
>+#include "absl/base/port.h"
>+
>+namespace {
>+
>+TEST(AsciiIsFoo, All) {
>+  for (int i = 0; i < 256; i++) {
>+    if ((i >= 'a' && i <= 'z') || (i >= 'A' && i <= 'Z'))
>+      EXPECT_TRUE(absl::ascii_isalpha(i)) << ": failed on " << i;
>+    else
>+      EXPECT_TRUE(!absl::ascii_isalpha(i)) << ": failed on " << i;
>+  }
>+  for (int i = 0; i < 256; i++) {
>+    if ((i >= '0' && i <= '9'))
>+      EXPECT_TRUE(absl::ascii_isdigit(i)) << ": failed on " << i;
>+    else
>+      EXPECT_TRUE(!absl::ascii_isdigit(i)) << ": failed on " << i;
>+  }
>+  for (int i = 0; i < 256; i++) {
>+    if (absl::ascii_isalpha(i) || absl::ascii_isdigit(i))
>+      EXPECT_TRUE(absl::ascii_isalnum(i)) << ": failed on " << i;
>+    else
>+      EXPECT_TRUE(!absl::ascii_isalnum(i)) << ": failed on " << i;
>+  }
>+  for (int i = 0; i < 256; i++) {
>+    if (i != '\0' && strchr(" \r\n\t\v\f", i))
>+      EXPECT_TRUE(absl::ascii_isspace(i)) << ": failed on " << i;
>+    else
>+      EXPECT_TRUE(!absl::ascii_isspace(i)) << ": failed on " << i;
>+  }
>+  for (int i = 0; i < 256; i++) {
>+    if (i >= 32 && i < 127)
>+      EXPECT_TRUE(absl::ascii_isprint(i)) << ": failed on " << i;
>+    else
>+      EXPECT_TRUE(!absl::ascii_isprint(i)) << ": failed on " << i;
>+  }
>+  for (int i = 0; i < 256; i++) {
>+    if (absl::ascii_isprint(i) && !absl::ascii_isspace(i) &&
>+        !absl::ascii_isalnum(i))
>+      EXPECT_TRUE(absl::ascii_ispunct(i)) << ": failed on " << i;
>+    else
>+      EXPECT_TRUE(!absl::ascii_ispunct(i)) << ": failed on " << i;
>+  }
>+  for (int i = 0; i < 256; i++) {
>+    if (i == ' ' || i == '\t')
>+      EXPECT_TRUE(absl::ascii_isblank(i)) << ": failed on " << i;
>+    else
>+      EXPECT_TRUE(!absl::ascii_isblank(i)) << ": failed on " << i;
>+  }
>+  for (int i = 0; i < 256; i++) {
>+    if (i < 32 || i == 127)
>+      EXPECT_TRUE(absl::ascii_iscntrl(i)) << ": failed on " << i;
>+    else
>+      EXPECT_TRUE(!absl::ascii_iscntrl(i)) << ": failed on " << i;
>+  }
>+  for (int i = 0; i < 256; i++) {
>+    if (absl::ascii_isdigit(i) || (i >= 'A' && i <= 'F') ||
>+        (i >= 'a' && i <= 'f'))
>+      EXPECT_TRUE(absl::ascii_isxdigit(i)) << ": failed on " << i;
>+    else
>+      EXPECT_TRUE(!absl::ascii_isxdigit(i)) << ": failed on " << i;
>+  }
>+  for (int i = 0; i < 256; i++) {
>+    if (i > 32 && i < 127)
>+      EXPECT_TRUE(absl::ascii_isgraph(i)) << ": failed on " << i;
>+    else
>+      EXPECT_TRUE(!absl::ascii_isgraph(i)) << ": failed on " << i;
>+  }
>+  for (int i = 0; i < 256; i++) {
>+    if (i >= 'A' && i <= 'Z')
>+      EXPECT_TRUE(absl::ascii_isupper(i)) << ": failed on " << i;
>+    else
>+      EXPECT_TRUE(!absl::ascii_isupper(i)) << ": failed on " << i;
>+  }
>+  for (int i = 0; i < 256; i++) {
>+    if (i >= 'a' && i <= 'z')
>+      EXPECT_TRUE(absl::ascii_islower(i)) << ": failed on " << i;
>+    else
>+      EXPECT_TRUE(!absl::ascii_islower(i)) << ": failed on " << i;
>+  }
>+  for (int i = 0; i < 128; i++) {
>+    EXPECT_TRUE(absl::ascii_isascii(i)) << ": failed on " << i;
>+  }
>+  for (int i = 128; i < 256; i++) {
>+    EXPECT_TRUE(!absl::ascii_isascii(i)) << ": failed on " << i;
>+  }
>+
>+  // The official is* functions don't accept negative signed chars, but
>+  // our absl::ascii_is* functions do.
>+  for (int i = 0; i < 256; i++) {
>+    signed char sc = static_cast<signed char>(static_cast<unsigned char>(i));
>+    EXPECT_EQ(absl::ascii_isalpha(i), absl::ascii_isalpha(sc)) << i;
>+    EXPECT_EQ(absl::ascii_isdigit(i), absl::ascii_isdigit(sc)) << i;
>+    EXPECT_EQ(absl::ascii_isalnum(i), absl::ascii_isalnum(sc)) << i;
>+    EXPECT_EQ(absl::ascii_isspace(i), absl::ascii_isspace(sc)) << i;
>+    EXPECT_EQ(absl::ascii_ispunct(i), absl::ascii_ispunct(sc)) << i;
>+    EXPECT_EQ(absl::ascii_isblank(i), absl::ascii_isblank(sc)) << i;
>+    EXPECT_EQ(absl::ascii_iscntrl(i), absl::ascii_iscntrl(sc)) << i;
>+    EXPECT_EQ(absl::ascii_isxdigit(i), absl::ascii_isxdigit(sc)) << i;
>+    EXPECT_EQ(absl::ascii_isprint(i), absl::ascii_isprint(sc)) << i;
>+    EXPECT_EQ(absl::ascii_isgraph(i), absl::ascii_isgraph(sc)) << i;
>+    EXPECT_EQ(absl::ascii_isupper(i), absl::ascii_isupper(sc)) << i;
>+    EXPECT_EQ(absl::ascii_islower(i), absl::ascii_islower(sc)) << i;
>+    EXPECT_EQ(absl::ascii_isascii(i), absl::ascii_isascii(sc)) << i;
>+  }
>+}
>+
>+// Checks that absl::ascii_isfoo returns the same value as isfoo in the C
>+// locale.
>+TEST(AsciiIsFoo, SameAsIsFoo) {
>+#ifndef __ANDROID__
>+  // temporarily change locale to C. It should already be C, but just for safety
>+  const char* old_locale = setlocale(LC_CTYPE, "C");
>+  ASSERT_TRUE(old_locale != nullptr);
>+#endif
>+
>+  for (int i = 0; i < 256; i++) {
>+    EXPECT_EQ(isalpha(i) != 0, absl::ascii_isalpha(i)) << i;
>+    EXPECT_EQ(isdigit(i) != 0, absl::ascii_isdigit(i)) << i;
>+    EXPECT_EQ(isalnum(i) != 0, absl::ascii_isalnum(i)) << i;
>+    EXPECT_EQ(isspace(i) != 0, absl::ascii_isspace(i)) << i;
>+    EXPECT_EQ(ispunct(i) != 0, absl::ascii_ispunct(i)) << i;
>+    EXPECT_EQ(isblank(i) != 0, absl::ascii_isblank(i)) << i;
>+    EXPECT_EQ(iscntrl(i) != 0, absl::ascii_iscntrl(i)) << i;
>+    EXPECT_EQ(isxdigit(i) != 0, absl::ascii_isxdigit(i)) << i;
>+    EXPECT_EQ(isprint(i) != 0, absl::ascii_isprint(i)) << i;
>+    EXPECT_EQ(isgraph(i) != 0, absl::ascii_isgraph(i)) << i;
>+    EXPECT_EQ(isupper(i) != 0, absl::ascii_isupper(i)) << i;
>+    EXPECT_EQ(islower(i) != 0, absl::ascii_islower(i)) << i;
>+    EXPECT_EQ(isascii(i) != 0, absl::ascii_isascii(i)) << i;
>+  }
>+
>+#ifndef __ANDROID__
>+  // restore the old locale.
>+  ASSERT_TRUE(setlocale(LC_CTYPE, old_locale));
>+#endif
>+}
>+
>+TEST(AsciiToFoo, All) {
>+#ifndef __ANDROID__
>+  // temporarily change locale to C. It should already be C, but just for safety
>+  const char* old_locale = setlocale(LC_CTYPE, "C");
>+  ASSERT_TRUE(old_locale != nullptr);
>+#endif
>+
>+  for (int i = 0; i < 256; i++) {
>+    if (absl::ascii_islower(i))
>+      EXPECT_EQ(absl::ascii_toupper(i), 'A' + (i - 'a')) << i;
>+    else
>+      EXPECT_EQ(absl::ascii_toupper(i), static_cast<char>(i)) << i;
>+
>+    if (absl::ascii_isupper(i))
>+      EXPECT_EQ(absl::ascii_tolower(i), 'a' + (i - 'A')) << i;
>+    else
>+      EXPECT_EQ(absl::ascii_tolower(i), static_cast<char>(i)) << i;
>+
>+    // These CHECKs only hold in a C locale.
>+    EXPECT_EQ(static_cast<char>(tolower(i)), absl::ascii_tolower(i)) << i;
>+    EXPECT_EQ(static_cast<char>(toupper(i)), absl::ascii_toupper(i)) << i;
>+
>+    // The official to* functions don't accept negative signed chars, but
>+    // our absl::ascii_to* functions do.
>+    signed char sc = static_cast<signed char>(static_cast<unsigned char>(i));
>+    EXPECT_EQ(absl::ascii_tolower(i), absl::ascii_tolower(sc)) << i;
>+    EXPECT_EQ(absl::ascii_toupper(i), absl::ascii_toupper(sc)) << i;
>+  }
>+#ifndef __ANDROID__
>+  // restore the old locale.
>+  ASSERT_TRUE(setlocale(LC_CTYPE, old_locale));
>+#endif
>+}
>+
>+TEST(AsciiStrTo, Lower) {
>+  const char buf[] = "ABCDEF";
>+  const std::string str("GHIJKL");
>+  const std::string str2("MNOPQR");
>+  const absl::string_view sp(str2);
>+
>+  EXPECT_EQ("abcdef", absl::AsciiStrToLower(buf));
>+  EXPECT_EQ("ghijkl", absl::AsciiStrToLower(str));
>+  EXPECT_EQ("mnopqr", absl::AsciiStrToLower(sp));
>+
>+  char mutable_buf[] = "Mutable";
>+  std::transform(mutable_buf, mutable_buf + strlen(mutable_buf),
>+                 mutable_buf, absl::ascii_tolower);
>+  EXPECT_STREQ("mutable", mutable_buf);
>+}
>+
>+TEST(AsciiStrTo, Upper) {
>+  const char buf[] = "abcdef";
>+  const std::string str("ghijkl");
>+  const std::string str2("mnopqr");
>+  const absl::string_view sp(str2);
>+
>+  EXPECT_EQ("ABCDEF", absl::AsciiStrToUpper(buf));
>+  EXPECT_EQ("GHIJKL", absl::AsciiStrToUpper(str));
>+  EXPECT_EQ("MNOPQR", absl::AsciiStrToUpper(sp));
>+
>+  char mutable_buf[] = "Mutable";
>+  std::transform(mutable_buf, mutable_buf + strlen(mutable_buf),
>+                 mutable_buf, absl::ascii_toupper);
>+  EXPECT_STREQ("MUTABLE", mutable_buf);
>+}
>+
>+TEST(StripLeadingAsciiWhitespace, FromStringView) {
>+  EXPECT_EQ(absl::string_view{},
>+            absl::StripLeadingAsciiWhitespace(absl::string_view{}));
>+  EXPECT_EQ("foo", absl::StripLeadingAsciiWhitespace({"foo"}));
>+  EXPECT_EQ("foo", absl::StripLeadingAsciiWhitespace({"\t  \n\f\r\n\vfoo"}));
>+  EXPECT_EQ("foo foo\n ",
>+            absl::StripLeadingAsciiWhitespace({"\t  \n\f\r\n\vfoo foo\n "}));
>+  EXPECT_EQ(absl::string_view{}, absl::StripLeadingAsciiWhitespace(
>+                                     {"\t  \n\f\r\v\n\t  \n\f\r\v\n"}));
>+}
>+
>+TEST(StripLeadingAsciiWhitespace, InPlace) {
>+  std::string str;
>+
>+  absl::StripLeadingAsciiWhitespace(&str);
>+  EXPECT_EQ("", str);
>+
>+  str = "foo";
>+  absl::StripLeadingAsciiWhitespace(&str);
>+  EXPECT_EQ("foo", str);
>+
>+  str = "\t  \n\f\r\n\vfoo";
>+  absl::StripLeadingAsciiWhitespace(&str);
>+  EXPECT_EQ("foo", str);
>+
>+  str = "\t  \n\f\r\n\vfoo foo\n ";
>+  absl::StripLeadingAsciiWhitespace(&str);
>+  EXPECT_EQ("foo foo\n ", str);
>+
>+  str = "\t  \n\f\r\v\n\t  \n\f\r\v\n";
>+  absl::StripLeadingAsciiWhitespace(&str);
>+  EXPECT_EQ(absl::string_view{}, str);
>+}
>+
>+TEST(StripTrailingAsciiWhitespace, FromStringView) {
>+  EXPECT_EQ(absl::string_view{},
>+            absl::StripTrailingAsciiWhitespace(absl::string_view{}));
>+  EXPECT_EQ("foo", absl::StripTrailingAsciiWhitespace({"foo"}));
>+  EXPECT_EQ("foo", absl::StripTrailingAsciiWhitespace({"foo\t  \n\f\r\n\v"}));
>+  EXPECT_EQ(" \nfoo foo",
>+            absl::StripTrailingAsciiWhitespace({" \nfoo foo\t  \n\f\r\n\v"}));
>+  EXPECT_EQ(absl::string_view{}, absl::StripTrailingAsciiWhitespace(
>+                                     {"\t  \n\f\r\v\n\t  \n\f\r\v\n"}));
>+}
>+
>+TEST(StripTrailingAsciiWhitespace, InPlace) {
>+  std::string str;
>+
>+  absl::StripTrailingAsciiWhitespace(&str);
>+  EXPECT_EQ("", str);
>+
>+  str = "foo";
>+  absl::StripTrailingAsciiWhitespace(&str);
>+  EXPECT_EQ("foo", str);
>+
>+  str = "foo\t  \n\f\r\n\v";
>+  absl::StripTrailingAsciiWhitespace(&str);
>+  EXPECT_EQ("foo", str);
>+
>+  str = " \nfoo foo\t  \n\f\r\n\v";
>+  absl::StripTrailingAsciiWhitespace(&str);
>+  EXPECT_EQ(" \nfoo foo", str);
>+
>+  str = "\t  \n\f\r\v\n\t  \n\f\r\v\n";
>+  absl::StripTrailingAsciiWhitespace(&str);
>+  EXPECT_EQ(absl::string_view{}, str);
>+}
>+
>+TEST(StripAsciiWhitespace, FromStringView) {
>+  EXPECT_EQ(absl::string_view{},
>+            absl::StripAsciiWhitespace(absl::string_view{}));
>+  EXPECT_EQ("foo", absl::StripAsciiWhitespace({"foo"}));
>+  EXPECT_EQ("foo",
>+            absl::StripAsciiWhitespace({"\t  \n\f\r\n\vfoo\t  \n\f\r\n\v"}));
>+  EXPECT_EQ("foo foo", absl::StripAsciiWhitespace(
>+                           {"\t  \n\f\r\n\vfoo foo\t  \n\f\r\n\v"}));
>+  EXPECT_EQ(absl::string_view{},
>+            absl::StripAsciiWhitespace({"\t  \n\f\r\v\n\t  \n\f\r\v\n"}));
>+}
>+
>+TEST(StripAsciiWhitespace, InPlace) {
>+  std::string str;
>+
>+  absl::StripAsciiWhitespace(&str);
>+  EXPECT_EQ("", str);
>+
>+  str = "foo";
>+  absl::StripAsciiWhitespace(&str);
>+  EXPECT_EQ("foo", str);
>+
>+  str = "\t  \n\f\r\n\vfoo\t  \n\f\r\n\v";
>+  absl::StripAsciiWhitespace(&str);
>+  EXPECT_EQ("foo", str);
>+
>+  str = "\t  \n\f\r\n\vfoo foo\t  \n\f\r\n\v";
>+  absl::StripAsciiWhitespace(&str);
>+  EXPECT_EQ("foo foo", str);
>+
>+  str = "\t  \n\f\r\v\n\t  \n\f\r\v\n";
>+  absl::StripAsciiWhitespace(&str);
>+  EXPECT_EQ(absl::string_view{}, str);
>+}
>+
>+TEST(RemoveExtraAsciiWhitespace, InPlace) {
>+  const char* inputs[] = {"No extra space",
>+                          "  Leading whitespace",
>+                          "Trailing whitespace  ",
>+                          "  Leading and trailing  ",
>+                          " Whitespace \t  in\v   middle  ",
>+                          "'Eeeeep!  \n Newlines!\n",
>+                          "nospaces",
>+                          "",
>+                          "\n\t a\t\n\nb \t\n"};
>+
>+  const char* outputs[] = {
>+      "No extra space",
>+      "Leading whitespace",
>+      "Trailing whitespace",
>+      "Leading and trailing",
>+      "Whitespace in middle",
>+      "'Eeeeep! Newlines!",
>+      "nospaces",
>+      "",
>+      "a\nb",
>+  };
>+  const int NUM_TESTS = ABSL_ARRAYSIZE(inputs);
>+
>+  for (int i = 0; i < NUM_TESTS; i++) {
>+    std::string s(inputs[i]);
>+    absl::RemoveExtraAsciiWhitespace(&s);
>+    EXPECT_EQ(outputs[i], s);
>+  }
>+}
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/charconv.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/charconv.cc
>new file mode 100644
>index 00000000000..08c3947eccd
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/charconv.cc
>@@ -0,0 +1,982 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/strings/charconv.h"
>+
>+#include <algorithm>
>+#include <cassert>
>+#include <cmath>
>+#include <cstring>
>+
>+#include "absl/base/casts.h"
>+#include "absl/numeric/int128.h"
>+#include "absl/strings/internal/bits.h"
>+#include "absl/strings/internal/charconv_bigint.h"
>+#include "absl/strings/internal/charconv_parse.h"
>+
>+// The macro ABSL_BIT_PACK_FLOATS is defined on x86-64, where IEEE floating
>+// point numbers have the same endianness in memory as a bitfield struct
>+// containing the corresponding parts.
>+//
>+// When set, we replace calls to ldexp() with manual bit packing, which is
>+// faster and is unaffected by floating point environment.
>+#ifdef ABSL_BIT_PACK_FLOATS
>+#error ABSL_BIT_PACK_FLOATS cannot be directly set
>+#elif defined(__x86_64__) || defined(_M_X64)
>+#define ABSL_BIT_PACK_FLOATS 1
>+#endif
>+
>+// A note about subnormals:
>+//
>+// The code below talks about "normals" and "subnormals".  A normal IEEE float
>+// has a fixed-width mantissa and power of two exponent.  For example, a normal
>+// `double` has a 53-bit mantissa.  Because the high bit is always 1, it is not
>+// stored in the representation.  The implicit bit buys an extra bit of
>+// resolution in the datatype.
>+//
>+// The downside of this scheme is that there is a large gap between DBL_MIN and
>+// zero.  (Large, at least, relative to the different between DBL_MIN and the
>+// next representable number).  This gap is softened by the "subnormal" numbers,
>+// which have the same power-of-two exponent as DBL_MIN, but no implicit 53rd
>+// bit.  An all-bits-zero exponent in the encoding represents subnormals.  (Zero
>+// is represented as a subnormal with an all-bits-zero mantissa.)
>+//
>+// The code below, in calculations, represents the mantissa as a uint64_t.  The
>+// end result normally has the 53rd bit set.  It represents subnormals by using
>+// narrower mantissas.
>+
>+namespace absl {
>+namespace {
>+
>+template <typename FloatType>
>+struct FloatTraits;
>+
>+template <>
>+struct FloatTraits<double> {
>+  // The number of mantissa bits in the given float type.  This includes the
>+  // implied high bit.
>+  static constexpr int kTargetMantissaBits = 53;
>+
>+  // The largest supported IEEE exponent, in our integral mantissa
>+  // representation.
>+  //
>+  // If `m` is the largest possible int kTargetMantissaBits bits wide, then
>+  // m * 2**kMaxExponent is exactly equal to DBL_MAX.
>+  static constexpr int kMaxExponent = 971;
>+
>+  // The smallest supported IEEE normal exponent, in our integral mantissa
>+  // representation.
>+  //
>+  // If `m` is the smallest possible int kTargetMantissaBits bits wide, then
>+  // m * 2**kMinNormalExponent is exactly equal to DBL_MIN.
>+  static constexpr int kMinNormalExponent = -1074;
>+
>+  static double MakeNan(const char* tagp) {
>+    // Support nan no matter which namespace it's in.  Some platforms
>+    // incorrectly don't put it in namespace std.
>+    using namespace std;  // NOLINT
>+    return nan(tagp);
>+  }
>+
>+  // Builds a nonzero floating point number out of the provided parts.
>+  //
>+  // This is intended to do the same operation as ldexp(mantissa, exponent),
>+  // but using purely integer math, to avoid -ffastmath and floating
>+  // point environment issues.  Using type punning is also faster. We fall back
>+  // to ldexp on a per-platform basis for portability.
>+  //
>+  // `exponent` must be between kMinNormalExponent and kMaxExponent.
>+  //
>+  // `mantissa` must either be exactly kTargetMantissaBits wide, in which case
>+  // a normal value is made, or it must be less narrow than that, in which case
>+  // `exponent` must be exactly kMinNormalExponent, and a subnormal value is
>+  // made.
>+  static double Make(uint64_t mantissa, int exponent, bool sign) {
>+#ifndef ABSL_BIT_PACK_FLOATS
>+    // Support ldexp no matter which namespace it's in.  Some platforms
>+    // incorrectly don't put it in namespace std.
>+    using namespace std;  // NOLINT
>+    return sign ? -ldexp(mantissa, exponent) : ldexp(mantissa, exponent);
>+#else
>+    constexpr uint64_t kMantissaMask =
>+        (uint64_t(1) << (kTargetMantissaBits - 1)) - 1;
>+    uint64_t dbl = static_cast<uint64_t>(sign) << 63;
>+    if (mantissa > kMantissaMask) {
>+      // Normal value.
>+      // Adjust by 1023 for the exponent representation bias, and an additional
>+      // 52 due to the implied decimal point in the IEEE mantissa represenation.
>+      dbl += uint64_t{exponent + 1023u + kTargetMantissaBits - 1} << 52;
>+      mantissa &= kMantissaMask;
>+    } else {
>+      // subnormal value
>+      assert(exponent == kMinNormalExponent);
>+    }
>+    dbl += mantissa;
>+    return absl::bit_cast<double>(dbl);
>+#endif  // ABSL_BIT_PACK_FLOATS
>+  }
>+};
>+
>+// Specialization of floating point traits for the `float` type.  See the
>+// FloatTraits<double> specialization above for meaning of each of the following
>+// members and methods.
>+template <>
>+struct FloatTraits<float> {
>+  static constexpr int kTargetMantissaBits = 24;
>+  static constexpr int kMaxExponent = 104;
>+  static constexpr int kMinNormalExponent = -149;
>+  static float MakeNan(const char* tagp) {
>+    // Support nanf no matter which namespace it's in.  Some platforms
>+    // incorrectly don't put it in namespace std.
>+    using namespace std;  // NOLINT
>+    return nanf(tagp);
>+  }
>+  static float Make(uint32_t mantissa, int exponent, bool sign) {
>+#ifndef ABSL_BIT_PACK_FLOATS
>+    // Support ldexpf no matter which namespace it's in.  Some platforms
>+    // incorrectly don't put it in namespace std.
>+    using namespace std;  // NOLINT
>+    return sign ? -ldexpf(mantissa, exponent) : ldexpf(mantissa, exponent);
>+#else
>+    constexpr uint32_t kMantissaMask =
>+        (uint32_t(1) << (kTargetMantissaBits - 1)) - 1;
>+    uint32_t flt = static_cast<uint32_t>(sign) << 31;
>+    if (mantissa > kMantissaMask) {
>+      // Normal value.
>+      // Adjust by 127 for the exponent representation bias, and an additional
>+      // 23 due to the implied decimal point in the IEEE mantissa represenation.
>+      flt += uint32_t{exponent + 127u + kTargetMantissaBits - 1} << 23;
>+      mantissa &= kMantissaMask;
>+    } else {
>+      // subnormal value
>+      assert(exponent == kMinNormalExponent);
>+    }
>+    flt += mantissa;
>+    return absl::bit_cast<float>(flt);
>+#endif  // ABSL_BIT_PACK_FLOATS
>+  }
>+};
>+
>+// Decimal-to-binary conversions require coercing powers of 10 into a mantissa
>+// and a power of 2.  The two helper functions Power10Mantissa(n) and
>+// Power10Exponent(n) perform this task.  Together, these represent a hand-
>+// rolled floating point value which is equal to or just less than 10**n.
>+//
>+// The return values satisfy two range guarantees:
>+//
>+//   Power10Mantissa(n) * 2**Power10Exponent(n) <= 10**n
>+//     < (Power10Mantissa(n) + 1) * 2**Power10Exponent(n)
>+//
>+//   2**63 <= Power10Mantissa(n) < 2**64.
>+//
>+// Lookups into the power-of-10 table must first check the Power10Overflow() and
>+// Power10Underflow() functions, to avoid out-of-bounds table access.
>+//
>+// Indexes into these tables are biased by -kPower10TableMin, and the table has
>+// values in the range [kPower10TableMin, kPower10TableMax].
>+extern const uint64_t kPower10MantissaTable[];
>+extern const int16_t kPower10ExponentTable[];
>+
>+// The smallest allowed value for use with the Power10Mantissa() and
>+// Power10Exponent() functions below.  (If a smaller exponent is needed in
>+// calculations, the end result is guaranteed to underflow.)
>+constexpr int kPower10TableMin = -342;
>+
>+// The largest allowed value for use with the Power10Mantissa() and
>+// Power10Exponent() functions below.  (If a smaller exponent is needed in
>+// calculations, the end result is guaranteed to overflow.)
>+constexpr int kPower10TableMax = 308;
>+
>+uint64_t Power10Mantissa(int n) {
>+  return kPower10MantissaTable[n - kPower10TableMin];
>+}
>+
>+int Power10Exponent(int n) {
>+  return kPower10ExponentTable[n - kPower10TableMin];
>+}
>+
>+// Returns true if n is large enough that 10**n always results in an IEEE
>+// overflow.
>+bool Power10Overflow(int n) { return n > kPower10TableMax; }
>+
>+// Returns true if n is small enough that 10**n times a ParsedFloat mantissa
>+// always results in an IEEE underflow.
>+bool Power10Underflow(int n) { return n < kPower10TableMin; }
>+
>+// Returns true if Power10Mantissa(n) * 2**Power10Exponent(n) is exactly equal
>+// to 10**n numerically.  Put another way, this returns true if there is no
>+// truncation error in Power10Mantissa(n).
>+bool Power10Exact(int n) { return n >= 0 && n <= 27; }
>+
>+// Sentinel exponent values for representing numbers too large or too close to
>+// zero to represent in a double.
>+constexpr int kOverflow = 99999;
>+constexpr int kUnderflow = -99999;
>+
>+// Struct representing the calculated conversion result of a positive (nonzero)
>+// floating point number.
>+//
>+// The calculated number is mantissa * 2**exponent (mantissa is treated as an
>+// integer.)  `mantissa` is chosen to be the correct width for the IEEE float
>+// representation being calculated.  (`mantissa` will always have the same bit
>+// width for normal values, and narrower bit widths for subnormals.)
>+//
>+// If the result of conversion was an underflow or overflow, exponent is set
>+// to kUnderflow or kOverflow.
>+struct CalculatedFloat {
>+  uint64_t mantissa = 0;
>+  int exponent = 0;
>+};
>+
>+// Returns the bit width of the given uint128.  (Equivalently, returns 128
>+// minus the number of leading zero bits.)
>+int BitWidth(uint128 value) {
>+  if (Uint128High64(value) == 0) {
>+    return 64 - strings_internal::CountLeadingZeros64(Uint128Low64(value));
>+  }
>+  return 128 - strings_internal::CountLeadingZeros64(Uint128High64(value));
>+}
>+
>+// Calculates how far to the right a mantissa needs to be shifted to create a
>+// properly adjusted mantissa for an IEEE floating point number.
>+//
>+// `mantissa_width` is the bit width of the mantissa to be shifted, and
>+// `binary_exponent` is the exponent of the number before the shift.
>+//
>+// This accounts for subnormal values, and will return a larger-than-normal
>+// shift if binary_exponent would otherwise be too low.
>+template <typename FloatType>
>+int NormalizedShiftSize(int mantissa_width, int binary_exponent) {
>+  const int normal_shift =
>+      mantissa_width - FloatTraits<FloatType>::kTargetMantissaBits;
>+  const int minimum_shift =
>+      FloatTraits<FloatType>::kMinNormalExponent - binary_exponent;
>+  return std::max(normal_shift, minimum_shift);
>+}
>+
>+// Right shifts a uint128 so that it has the requested bit width.  (The
>+// resulting value will have 128 - bit_width leading zeroes.)  The initial
>+// `value` must be wider than the requested bit width.
>+//
>+// Returns the number of bits shifted.
>+int TruncateToBitWidth(int bit_width, uint128* value) {
>+  const int current_bit_width = BitWidth(*value);
>+  const int shift = current_bit_width - bit_width;
>+  *value >>= shift;
>+  return shift;
>+}
>+
>+// Checks if the given ParsedFloat represents one of the edge cases that are
>+// not dependent on number base: zero, infinity, or NaN.  If so, sets *value
>+// the appropriate double, and returns true.
>+template <typename FloatType>
>+bool HandleEdgeCase(const strings_internal::ParsedFloat& input, bool negative,
>+                    FloatType* value) {
>+  if (input.type == strings_internal::FloatType::kNan) {
>+    // A bug in both clang and gcc would cause the compiler to optimize away the
>+    // buffer we are building below.  Declaring the buffer volatile avoids the
>+    // issue, and has no measurable performance impact in microbenchmarks.
>+    //
>+    // https://bugs.llvm.org/show_bug.cgi?id=37778
>+    // https://gcc.gnu.org/bugzilla/show_bug.cgi?id=86113
>+    constexpr ptrdiff_t kNanBufferSize = 128;
>+    volatile char n_char_sequence[kNanBufferSize];
>+    if (input.subrange_begin == nullptr) {
>+      n_char_sequence[0] = '\0';
>+    } else {
>+      ptrdiff_t nan_size = input.subrange_end - input.subrange_begin;
>+      nan_size = std::min(nan_size, kNanBufferSize - 1);
>+      std::copy_n(input.subrange_begin, nan_size, n_char_sequence);
>+      n_char_sequence[nan_size] = '\0';
>+    }
>+    char* nan_argument = const_cast<char*>(n_char_sequence);
>+    *value = negative ? -FloatTraits<FloatType>::MakeNan(nan_argument)
>+                      : FloatTraits<FloatType>::MakeNan(nan_argument);
>+    return true;
>+  }
>+  if (input.type == strings_internal::FloatType::kInfinity) {
>+    *value = negative ? -std::numeric_limits<FloatType>::infinity()
>+                      : std::numeric_limits<FloatType>::infinity();
>+    return true;
>+  }
>+  if (input.mantissa == 0) {
>+    *value = negative ? -0.0 : 0.0;
>+    return true;
>+  }
>+  return false;
>+}
>+
>+// Given a CalculatedFloat result of a from_chars conversion, generate the
>+// correct output values.
>+//
>+// CalculatedFloat can represent an underflow or overflow, in which case the
>+// error code in *result is set.  Otherwise, the calculated floating point
>+// number is stored in *value.
>+template <typename FloatType>
>+void EncodeResult(const CalculatedFloat& calculated, bool negative,
>+                  absl::from_chars_result* result, FloatType* value) {
>+  if (calculated.exponent == kOverflow) {
>+    result->ec = std::errc::result_out_of_range;
>+    *value = negative ? -std::numeric_limits<FloatType>::max()
>+                      : std::numeric_limits<FloatType>::max();
>+    return;
>+  } else if (calculated.mantissa == 0 || calculated.exponent == kUnderflow) {
>+    result->ec = std::errc::result_out_of_range;
>+    *value = negative ? -0.0 : 0.0;
>+    return;
>+  }
>+  *value = FloatTraits<FloatType>::Make(calculated.mantissa,
>+                                        calculated.exponent, negative);
>+}
>+
>+// Returns the given uint128 shifted to the right by `shift` bits, and rounds
>+// the remaining bits using round_to_nearest logic.  The value is returned as a
>+// uint64_t, since this is the type used by this library for storing calculated
>+// floating point mantissas.
>+//
>+// It is expected that the width of the input value shifted by `shift` will
>+// be the correct bit-width for the target mantissa, which is strictly narrower
>+// than a uint64_t.
>+//
>+// If `input_exact` is false, then a nonzero error epsilon is assumed.  For
>+// rounding purposes, the true value being rounded is strictly greater than the
>+// input value.  The error may represent a single lost carry bit.
>+//
>+// When input_exact, shifted bits of the form 1000000... represent a tie, which
>+// is broken by rounding to even -- the rounding direction is chosen so the low
>+// bit of the returned value is 0.
>+//
>+// When !input_exact, shifted bits of the form 10000000... represent a value
>+// strictly greater than one half (due to the error epsilon), and so ties are
>+// always broken by rounding up.
>+//
>+// When !input_exact, shifted bits of the form 01111111... are uncertain;
>+// the true value may or may not be greater than 10000000..., due to the
>+// possible lost carry bit.  The correct rounding direction is unknown.  In this
>+// case, the result is rounded down, and `output_exact` is set to false.
>+//
>+// Zero and negative values of `shift` are accepted, in which case the word is
>+// shifted left, as necessary.
>+uint64_t ShiftRightAndRound(uint128 value, int shift, bool input_exact,
>+                            bool* output_exact) {
>+  if (shift <= 0) {
>+    *output_exact = input_exact;
>+    return static_cast<uint64_t>(value << -shift);
>+  }
>+  if (shift >= 128) {
>+    // Exponent is so small that we are shifting away all significant bits.
>+    // Answer will not be representable, even as a subnormal, so return a zero
>+    // mantissa (which represents underflow).
>+    *output_exact = true;
>+    return 0;
>+  }
>+
>+  *output_exact = true;
>+  const uint128 shift_mask = (uint128(1) << shift) - 1;
>+  const uint128 halfway_point = uint128(1) << (shift - 1);
>+
>+  const uint128 shifted_bits = value & shift_mask;
>+  value >>= shift;
>+  if (shifted_bits > halfway_point) {
>+    // Shifted bits greater than 10000... require rounding up.
>+    return static_cast<uint64_t>(value + 1);
>+  }
>+  if (shifted_bits == halfway_point) {
>+    // In exact mode, shifted bits of 10000... mean we're exactly halfway
>+    // between two numbers, and we must round to even.  So only round up if
>+    // the low bit of `value` is set.
>+    //
>+    // In inexact mode, the nonzero error means the actual value is greater
>+    // than the halfway point and we must alway round up.
>+    if ((value & 1) == 1 || !input_exact) {
>+      ++value;
>+    }
>+    return static_cast<uint64_t>(value);
>+  }
>+  if (!input_exact && shifted_bits == halfway_point - 1) {
>+    // Rounding direction is unclear, due to error.
>+    *output_exact = false;
>+  }
>+  // Otherwise, round down.
>+  return static_cast<uint64_t>(value);
>+}
>+
>+// Checks if a floating point guess needs to be rounded up, using high precision
>+// math.
>+//
>+// `guess_mantissa` and `guess_exponent` represent a candidate guess for the
>+// number represented by `parsed_decimal`.
>+//
>+// The exact number represented by `parsed_decimal` must lie between the two
>+// numbers:
>+//   A = `guess_mantissa * 2**guess_exponent`
>+//   B = `(guess_mantissa + 1) * 2**guess_exponent`
>+//
>+// This function returns false if `A` is the better guess, and true if `B` is
>+// the better guess, with rounding ties broken by rounding to even.
>+bool MustRoundUp(uint64_t guess_mantissa, int guess_exponent,
>+                 const strings_internal::ParsedFloat& parsed_decimal) {
>+  // 768 is the number of digits needed in the worst case.  We could determine a
>+  // better limit dynamically based on the value of parsed_decimal.exponent.
>+  // This would optimize pathological input cases only.  (Sane inputs won't have
>+  // hundreds of digits of mantissa.)
>+  absl::strings_internal::BigUnsigned<84> exact_mantissa;
>+  int exact_exponent = exact_mantissa.ReadFloatMantissa(parsed_decimal, 768);
>+
>+  // Adjust the `guess` arguments to be halfway between A and B.
>+  guess_mantissa = guess_mantissa * 2 + 1;
>+  guess_exponent -= 1;
>+
>+  // In our comparison:
>+  // lhs = exact = exact_mantissa * 10**exact_exponent
>+  //             = exact_mantissa * 5**exact_exponent * 2**exact_exponent
>+  // rhs = guess = guess_mantissa * 2**guess_exponent
>+  //
>+  // Because we are doing integer math, we can't directly deal with negative
>+  // exponents.  We instead move these to the other side of the inequality.
>+  absl::strings_internal::BigUnsigned<84>& lhs = exact_mantissa;
>+  int comparison;
>+  if (exact_exponent >= 0) {
>+    lhs.MultiplyByFiveToTheNth(exact_exponent);
>+    absl::strings_internal::BigUnsigned<84> rhs(guess_mantissa);
>+    // There are powers of 2 on both sides of the inequality; reduce this to
>+    // a single bit-shift.
>+    if (exact_exponent > guess_exponent) {
>+      lhs.ShiftLeft(exact_exponent - guess_exponent);
>+    } else {
>+      rhs.ShiftLeft(guess_exponent - exact_exponent);
>+    }
>+    comparison = Compare(lhs, rhs);
>+  } else {
>+    // Move the power of 5 to the other side of the equation, giving us:
>+    // lhs = exact_mantissa * 2**exact_exponent
>+    // rhs = guess_mantissa * 5**(-exact_exponent) * 2**guess_exponent
>+    absl::strings_internal::BigUnsigned<84> rhs =
>+        absl::strings_internal::BigUnsigned<84>::FiveToTheNth(-exact_exponent);
>+    rhs.MultiplyBy(guess_mantissa);
>+    if (exact_exponent > guess_exponent) {
>+      lhs.ShiftLeft(exact_exponent - guess_exponent);
>+    } else {
>+      rhs.ShiftLeft(guess_exponent - exact_exponent);
>+    }
>+    comparison = Compare(lhs, rhs);
>+  }
>+  if (comparison < 0) {
>+    return false;
>+  } else if (comparison > 0) {
>+    return true;
>+  } else {
>+    // When lhs == rhs, the decimal input is exactly between A and B.
>+    // Round towards even -- round up only if the low bit of the initial
>+    // `guess_mantissa` was a 1.  We shifted guess_mantissa left 1 bit at
>+    // the beginning of this function, so test the 2nd bit here.
>+    return (guess_mantissa & 2) == 2;
>+  }
>+}
>+
>+// Constructs a CalculatedFloat from a given mantissa and exponent, but
>+// with the following normalizations applied:
>+//
>+// If rounding has caused mantissa to increase just past the allowed bit
>+// width, shift and adjust exponent.
>+//
>+// If exponent is too high, sets kOverflow.
>+//
>+// If mantissa is zero (representing a non-zero value not representable, even
>+// as a subnormal), sets kUnderflow.
>+template <typename FloatType>
>+CalculatedFloat CalculatedFloatFromRawValues(uint64_t mantissa, int exponent) {
>+  CalculatedFloat result;
>+  if (mantissa == uint64_t(1) << FloatTraits<FloatType>::kTargetMantissaBits) {
>+    mantissa >>= 1;
>+    exponent += 1;
>+  }
>+  if (exponent > FloatTraits<FloatType>::kMaxExponent) {
>+    result.exponent = kOverflow;
>+  } else if (mantissa == 0) {
>+    result.exponent = kUnderflow;
>+  } else {
>+    result.exponent = exponent;
>+    result.mantissa = mantissa;
>+  }
>+  return result;
>+}
>+
>+template <typename FloatType>
>+CalculatedFloat CalculateFromParsedHexadecimal(
>+    const strings_internal::ParsedFloat& parsed_hex) {
>+  uint64_t mantissa = parsed_hex.mantissa;
>+  int exponent = parsed_hex.exponent;
>+  int mantissa_width = 64 - strings_internal::CountLeadingZeros64(mantissa);
>+  const int shift = NormalizedShiftSize<FloatType>(mantissa_width, exponent);
>+  bool result_exact;
>+  exponent += shift;
>+  mantissa = ShiftRightAndRound(mantissa, shift,
>+                                /* input exact= */ true, &result_exact);
>+  // ParseFloat handles rounding in the hexadecimal case, so we don't have to
>+  // check `result_exact` here.
>+  return CalculatedFloatFromRawValues<FloatType>(mantissa, exponent);
>+}
>+
>+template <typename FloatType>
>+CalculatedFloat CalculateFromParsedDecimal(
>+    const strings_internal::ParsedFloat& parsed_decimal) {
>+  CalculatedFloat result;
>+
>+  // Large or small enough decimal exponents will always result in overflow
>+  // or underflow.
>+  if (Power10Underflow(parsed_decimal.exponent)) {
>+    result.exponent = kUnderflow;
>+    return result;
>+  } else if (Power10Overflow(parsed_decimal.exponent)) {
>+    result.exponent = kOverflow;
>+    return result;
>+  }
>+
>+  // Otherwise convert our power of 10 into a power of 2 times an integer
>+  // mantissa, and multiply this by our parsed decimal mantissa.
>+  uint128 wide_binary_mantissa = parsed_decimal.mantissa;
>+  wide_binary_mantissa *= Power10Mantissa(parsed_decimal.exponent);
>+  int binary_exponent = Power10Exponent(parsed_decimal.exponent);
>+
>+  // Discard bits that are inaccurate due to truncation error.  The magic
>+  // `mantissa_width` constants below are justified in charconv_algorithm.md.
>+  // They represent the number of bits in `wide_binary_mantissa` that are
>+  // guaranteed to be unaffected by error propagation.
>+  bool mantissa_exact;
>+  int mantissa_width;
>+  if (parsed_decimal.subrange_begin) {
>+    // Truncated mantissa
>+    mantissa_width = 58;
>+    mantissa_exact = false;
>+    binary_exponent +=
>+        TruncateToBitWidth(mantissa_width, &wide_binary_mantissa);
>+  } else if (!Power10Exact(parsed_decimal.exponent)) {
>+    // Exact mantissa, truncated power of ten
>+    mantissa_width = 63;
>+    mantissa_exact = false;
>+    binary_exponent +=
>+        TruncateToBitWidth(mantissa_width, &wide_binary_mantissa);
>+  } else {
>+    // Product is exact
>+    mantissa_width = BitWidth(wide_binary_mantissa);
>+    mantissa_exact = true;
>+  }
>+
>+  // Shift into an FloatType-sized mantissa, and round to nearest.
>+  const int shift =
>+      NormalizedShiftSize<FloatType>(mantissa_width, binary_exponent);
>+  bool result_exact;
>+  binary_exponent += shift;
>+  uint64_t binary_mantissa = ShiftRightAndRound(wide_binary_mantissa, shift,
>+                                                mantissa_exact, &result_exact);
>+  if (!result_exact) {
>+    // We could not determine the rounding direction using int128 math.  Use
>+    // full resolution math instead.
>+    if (MustRoundUp(binary_mantissa, binary_exponent, parsed_decimal)) {
>+      binary_mantissa += 1;
>+    }
>+  }
>+
>+  return CalculatedFloatFromRawValues<FloatType>(binary_mantissa,
>+                                                 binary_exponent);
>+}
>+
>+template <typename FloatType>
>+from_chars_result FromCharsImpl(const char* first, const char* last,
>+                                FloatType& value, chars_format fmt_flags) {
>+  from_chars_result result;
>+  result.ptr = first;  // overwritten on successful parse
>+  result.ec = std::errc();
>+
>+  bool negative = false;
>+  if (first != last && *first == '-') {
>+    ++first;
>+    negative = true;
>+  }
>+  // If the `hex` flag is *not* set, then we will accept a 0x prefix and try
>+  // to parse a hexadecimal float.
>+  if ((fmt_flags & chars_format::hex) == chars_format{} && last - first >= 2 &&
>+      *first == '0' && (first[1] == 'x' || first[1] == 'X')) {
>+    const char* hex_first = first + 2;
>+    strings_internal::ParsedFloat hex_parse =
>+        strings_internal::ParseFloat<16>(hex_first, last, fmt_flags);
>+    if (hex_parse.end == nullptr ||
>+        hex_parse.type != strings_internal::FloatType::kNumber) {
>+      // Either we failed to parse a hex float after the "0x", or we read
>+      // "0xinf" or "0xnan" which we don't want to match.
>+      //
>+      // However, a std::string that begins with "0x" also begins with "0", which
>+      // is normally a valid match for the number zero.  So we want these
>+      // strings to match zero unless fmt_flags is `scientific`.  (This flag
>+      // means an exponent is required, which the std::string "0" does not have.)
>+      if (fmt_flags == chars_format::scientific) {
>+        result.ec = std::errc::invalid_argument;
>+      } else {
>+        result.ptr = first + 1;
>+        value = negative ? -0.0 : 0.0;
>+      }
>+      return result;
>+    }
>+    // We matched a value.
>+    result.ptr = hex_parse.end;
>+    if (HandleEdgeCase(hex_parse, negative, &value)) {
>+      return result;
>+    }
>+    CalculatedFloat calculated =
>+        CalculateFromParsedHexadecimal<FloatType>(hex_parse);
>+    EncodeResult(calculated, negative, &result, &value);
>+    return result;
>+  }
>+  // Otherwise, we choose the number base based on the flags.
>+  if ((fmt_flags & chars_format::hex) == chars_format::hex) {
>+    strings_internal::ParsedFloat hex_parse =
>+        strings_internal::ParseFloat<16>(first, last, fmt_flags);
>+    if (hex_parse.end == nullptr) {
>+      result.ec = std::errc::invalid_argument;
>+      return result;
>+    }
>+    result.ptr = hex_parse.end;
>+    if (HandleEdgeCase(hex_parse, negative, &value)) {
>+      return result;
>+    }
>+    CalculatedFloat calculated =
>+        CalculateFromParsedHexadecimal<FloatType>(hex_parse);
>+    EncodeResult(calculated, negative, &result, &value);
>+    return result;
>+  } else {
>+    strings_internal::ParsedFloat decimal_parse =
>+        strings_internal::ParseFloat<10>(first, last, fmt_flags);
>+    if (decimal_parse.end == nullptr) {
>+      result.ec = std::errc::invalid_argument;
>+      return result;
>+    }
>+    result.ptr = decimal_parse.end;
>+    if (HandleEdgeCase(decimal_parse, negative, &value)) {
>+      return result;
>+    }
>+    CalculatedFloat calculated =
>+        CalculateFromParsedDecimal<FloatType>(decimal_parse);
>+    EncodeResult(calculated, negative, &result, &value);
>+    return result;
>+  }
>+  return result;
>+}
>+}  // namespace
>+
>+from_chars_result from_chars(const char* first, const char* last, double& value,
>+                             chars_format fmt) {
>+  return FromCharsImpl(first, last, value, fmt);
>+}
>+
>+from_chars_result from_chars(const char* first, const char* last, float& value,
>+                             chars_format fmt) {
>+  return FromCharsImpl(first, last, value, fmt);
>+}
>+
>+namespace {
>+
>+// Table of powers of 10, from kPower10TableMin to kPower10TableMax.
>+//
>+// kPower10MantissaTable[i - kPower10TableMin] stores the 64-bit mantissa (high
>+// bit always on), and kPower10ExponentTable[i - kPower10TableMin] stores the
>+// power-of-two exponent.  For a given number i, this gives the unique mantissa
>+// and exponent such that mantissa * 2**exponent <= 10**i < (mantissa + 1) *
>+// 2**exponent.
>+
>+const uint64_t kPower10MantissaTable[] = {
>+    0xeef453d6923bd65aU, 0x9558b4661b6565f8U, 0xbaaee17fa23ebf76U,
>+    0xe95a99df8ace6f53U, 0x91d8a02bb6c10594U, 0xb64ec836a47146f9U,
>+    0xe3e27a444d8d98b7U, 0x8e6d8c6ab0787f72U, 0xb208ef855c969f4fU,
>+    0xde8b2b66b3bc4723U, 0x8b16fb203055ac76U, 0xaddcb9e83c6b1793U,
>+    0xd953e8624b85dd78U, 0x87d4713d6f33aa6bU, 0xa9c98d8ccb009506U,
>+    0xd43bf0effdc0ba48U, 0x84a57695fe98746dU, 0xa5ced43b7e3e9188U,
>+    0xcf42894a5dce35eaU, 0x818995ce7aa0e1b2U, 0xa1ebfb4219491a1fU,
>+    0xca66fa129f9b60a6U, 0xfd00b897478238d0U, 0x9e20735e8cb16382U,
>+    0xc5a890362fddbc62U, 0xf712b443bbd52b7bU, 0x9a6bb0aa55653b2dU,
>+    0xc1069cd4eabe89f8U, 0xf148440a256e2c76U, 0x96cd2a865764dbcaU,
>+    0xbc807527ed3e12bcU, 0xeba09271e88d976bU, 0x93445b8731587ea3U,
>+    0xb8157268fdae9e4cU, 0xe61acf033d1a45dfU, 0x8fd0c16206306babU,
>+    0xb3c4f1ba87bc8696U, 0xe0b62e2929aba83cU, 0x8c71dcd9ba0b4925U,
>+    0xaf8e5410288e1b6fU, 0xdb71e91432b1a24aU, 0x892731ac9faf056eU,
>+    0xab70fe17c79ac6caU, 0xd64d3d9db981787dU, 0x85f0468293f0eb4eU,
>+    0xa76c582338ed2621U, 0xd1476e2c07286faaU, 0x82cca4db847945caU,
>+    0xa37fce126597973cU, 0xcc5fc196fefd7d0cU, 0xff77b1fcbebcdc4fU,
>+    0x9faacf3df73609b1U, 0xc795830d75038c1dU, 0xf97ae3d0d2446f25U,
>+    0x9becce62836ac577U, 0xc2e801fb244576d5U, 0xf3a20279ed56d48aU,
>+    0x9845418c345644d6U, 0xbe5691ef416bd60cU, 0xedec366b11c6cb8fU,
>+    0x94b3a202eb1c3f39U, 0xb9e08a83a5e34f07U, 0xe858ad248f5c22c9U,
>+    0x91376c36d99995beU, 0xb58547448ffffb2dU, 0xe2e69915b3fff9f9U,
>+    0x8dd01fad907ffc3bU, 0xb1442798f49ffb4aU, 0xdd95317f31c7fa1dU,
>+    0x8a7d3eef7f1cfc52U, 0xad1c8eab5ee43b66U, 0xd863b256369d4a40U,
>+    0x873e4f75e2224e68U, 0xa90de3535aaae202U, 0xd3515c2831559a83U,
>+    0x8412d9991ed58091U, 0xa5178fff668ae0b6U, 0xce5d73ff402d98e3U,
>+    0x80fa687f881c7f8eU, 0xa139029f6a239f72U, 0xc987434744ac874eU,
>+    0xfbe9141915d7a922U, 0x9d71ac8fada6c9b5U, 0xc4ce17b399107c22U,
>+    0xf6019da07f549b2bU, 0x99c102844f94e0fbU, 0xc0314325637a1939U,
>+    0xf03d93eebc589f88U, 0x96267c7535b763b5U, 0xbbb01b9283253ca2U,
>+    0xea9c227723ee8bcbU, 0x92a1958a7675175fU, 0xb749faed14125d36U,
>+    0xe51c79a85916f484U, 0x8f31cc0937ae58d2U, 0xb2fe3f0b8599ef07U,
>+    0xdfbdcece67006ac9U, 0x8bd6a141006042bdU, 0xaecc49914078536dU,
>+    0xda7f5bf590966848U, 0x888f99797a5e012dU, 0xaab37fd7d8f58178U,
>+    0xd5605fcdcf32e1d6U, 0x855c3be0a17fcd26U, 0xa6b34ad8c9dfc06fU,
>+    0xd0601d8efc57b08bU, 0x823c12795db6ce57U, 0xa2cb1717b52481edU,
>+    0xcb7ddcdda26da268U, 0xfe5d54150b090b02U, 0x9efa548d26e5a6e1U,
>+    0xc6b8e9b0709f109aU, 0xf867241c8cc6d4c0U, 0x9b407691d7fc44f8U,
>+    0xc21094364dfb5636U, 0xf294b943e17a2bc4U, 0x979cf3ca6cec5b5aU,
>+    0xbd8430bd08277231U, 0xece53cec4a314ebdU, 0x940f4613ae5ed136U,
>+    0xb913179899f68584U, 0xe757dd7ec07426e5U, 0x9096ea6f3848984fU,
>+    0xb4bca50b065abe63U, 0xe1ebce4dc7f16dfbU, 0x8d3360f09cf6e4bdU,
>+    0xb080392cc4349decU, 0xdca04777f541c567U, 0x89e42caaf9491b60U,
>+    0xac5d37d5b79b6239U, 0xd77485cb25823ac7U, 0x86a8d39ef77164bcU,
>+    0xa8530886b54dbdebU, 0xd267caa862a12d66U, 0x8380dea93da4bc60U,
>+    0xa46116538d0deb78U, 0xcd795be870516656U, 0x806bd9714632dff6U,
>+    0xa086cfcd97bf97f3U, 0xc8a883c0fdaf7df0U, 0xfad2a4b13d1b5d6cU,
>+    0x9cc3a6eec6311a63U, 0xc3f490aa77bd60fcU, 0xf4f1b4d515acb93bU,
>+    0x991711052d8bf3c5U, 0xbf5cd54678eef0b6U, 0xef340a98172aace4U,
>+    0x9580869f0e7aac0eU, 0xbae0a846d2195712U, 0xe998d258869facd7U,
>+    0x91ff83775423cc06U, 0xb67f6455292cbf08U, 0xe41f3d6a7377eecaU,
>+    0x8e938662882af53eU, 0xb23867fb2a35b28dU, 0xdec681f9f4c31f31U,
>+    0x8b3c113c38f9f37eU, 0xae0b158b4738705eU, 0xd98ddaee19068c76U,
>+    0x87f8a8d4cfa417c9U, 0xa9f6d30a038d1dbcU, 0xd47487cc8470652bU,
>+    0x84c8d4dfd2c63f3bU, 0xa5fb0a17c777cf09U, 0xcf79cc9db955c2ccU,
>+    0x81ac1fe293d599bfU, 0xa21727db38cb002fU, 0xca9cf1d206fdc03bU,
>+    0xfd442e4688bd304aU, 0x9e4a9cec15763e2eU, 0xc5dd44271ad3cdbaU,
>+    0xf7549530e188c128U, 0x9a94dd3e8cf578b9U, 0xc13a148e3032d6e7U,
>+    0xf18899b1bc3f8ca1U, 0x96f5600f15a7b7e5U, 0xbcb2b812db11a5deU,
>+    0xebdf661791d60f56U, 0x936b9fcebb25c995U, 0xb84687c269ef3bfbU,
>+    0xe65829b3046b0afaU, 0x8ff71a0fe2c2e6dcU, 0xb3f4e093db73a093U,
>+    0xe0f218b8d25088b8U, 0x8c974f7383725573U, 0xafbd2350644eeacfU,
>+    0xdbac6c247d62a583U, 0x894bc396ce5da772U, 0xab9eb47c81f5114fU,
>+    0xd686619ba27255a2U, 0x8613fd0145877585U, 0xa798fc4196e952e7U,
>+    0xd17f3b51fca3a7a0U, 0x82ef85133de648c4U, 0xa3ab66580d5fdaf5U,
>+    0xcc963fee10b7d1b3U, 0xffbbcfe994e5c61fU, 0x9fd561f1fd0f9bd3U,
>+    0xc7caba6e7c5382c8U, 0xf9bd690a1b68637bU, 0x9c1661a651213e2dU,
>+    0xc31bfa0fe5698db8U, 0xf3e2f893dec3f126U, 0x986ddb5c6b3a76b7U,
>+    0xbe89523386091465U, 0xee2ba6c0678b597fU, 0x94db483840b717efU,
>+    0xba121a4650e4ddebU, 0xe896a0d7e51e1566U, 0x915e2486ef32cd60U,
>+    0xb5b5ada8aaff80b8U, 0xe3231912d5bf60e6U, 0x8df5efabc5979c8fU,
>+    0xb1736b96b6fd83b3U, 0xddd0467c64bce4a0U, 0x8aa22c0dbef60ee4U,
>+    0xad4ab7112eb3929dU, 0xd89d64d57a607744U, 0x87625f056c7c4a8bU,
>+    0xa93af6c6c79b5d2dU, 0xd389b47879823479U, 0x843610cb4bf160cbU,
>+    0xa54394fe1eedb8feU, 0xce947a3da6a9273eU, 0x811ccc668829b887U,
>+    0xa163ff802a3426a8U, 0xc9bcff6034c13052U, 0xfc2c3f3841f17c67U,
>+    0x9d9ba7832936edc0U, 0xc5029163f384a931U, 0xf64335bcf065d37dU,
>+    0x99ea0196163fa42eU, 0xc06481fb9bcf8d39U, 0xf07da27a82c37088U,
>+    0x964e858c91ba2655U, 0xbbe226efb628afeaU, 0xeadab0aba3b2dbe5U,
>+    0x92c8ae6b464fc96fU, 0xb77ada0617e3bbcbU, 0xe55990879ddcaabdU,
>+    0x8f57fa54c2a9eab6U, 0xb32df8e9f3546564U, 0xdff9772470297ebdU,
>+    0x8bfbea76c619ef36U, 0xaefae51477a06b03U, 0xdab99e59958885c4U,
>+    0x88b402f7fd75539bU, 0xaae103b5fcd2a881U, 0xd59944a37c0752a2U,
>+    0x857fcae62d8493a5U, 0xa6dfbd9fb8e5b88eU, 0xd097ad07a71f26b2U,
>+    0x825ecc24c873782fU, 0xa2f67f2dfa90563bU, 0xcbb41ef979346bcaU,
>+    0xfea126b7d78186bcU, 0x9f24b832e6b0f436U, 0xc6ede63fa05d3143U,
>+    0xf8a95fcf88747d94U, 0x9b69dbe1b548ce7cU, 0xc24452da229b021bU,
>+    0xf2d56790ab41c2a2U, 0x97c560ba6b0919a5U, 0xbdb6b8e905cb600fU,
>+    0xed246723473e3813U, 0x9436c0760c86e30bU, 0xb94470938fa89bceU,
>+    0xe7958cb87392c2c2U, 0x90bd77f3483bb9b9U, 0xb4ecd5f01a4aa828U,
>+    0xe2280b6c20dd5232U, 0x8d590723948a535fU, 0xb0af48ec79ace837U,
>+    0xdcdb1b2798182244U, 0x8a08f0f8bf0f156bU, 0xac8b2d36eed2dac5U,
>+    0xd7adf884aa879177U, 0x86ccbb52ea94baeaU, 0xa87fea27a539e9a5U,
>+    0xd29fe4b18e88640eU, 0x83a3eeeef9153e89U, 0xa48ceaaab75a8e2bU,
>+    0xcdb02555653131b6U, 0x808e17555f3ebf11U, 0xa0b19d2ab70e6ed6U,
>+    0xc8de047564d20a8bU, 0xfb158592be068d2eU, 0x9ced737bb6c4183dU,
>+    0xc428d05aa4751e4cU, 0xf53304714d9265dfU, 0x993fe2c6d07b7fabU,
>+    0xbf8fdb78849a5f96U, 0xef73d256a5c0f77cU, 0x95a8637627989aadU,
>+    0xbb127c53b17ec159U, 0xe9d71b689dde71afU, 0x9226712162ab070dU,
>+    0xb6b00d69bb55c8d1U, 0xe45c10c42a2b3b05U, 0x8eb98a7a9a5b04e3U,
>+    0xb267ed1940f1c61cU, 0xdf01e85f912e37a3U, 0x8b61313bbabce2c6U,
>+    0xae397d8aa96c1b77U, 0xd9c7dced53c72255U, 0x881cea14545c7575U,
>+    0xaa242499697392d2U, 0xd4ad2dbfc3d07787U, 0x84ec3c97da624ab4U,
>+    0xa6274bbdd0fadd61U, 0xcfb11ead453994baU, 0x81ceb32c4b43fcf4U,
>+    0xa2425ff75e14fc31U, 0xcad2f7f5359a3b3eU, 0xfd87b5f28300ca0dU,
>+    0x9e74d1b791e07e48U, 0xc612062576589ddaU, 0xf79687aed3eec551U,
>+    0x9abe14cd44753b52U, 0xc16d9a0095928a27U, 0xf1c90080baf72cb1U,
>+    0x971da05074da7beeU, 0xbce5086492111aeaU, 0xec1e4a7db69561a5U,
>+    0x9392ee8e921d5d07U, 0xb877aa3236a4b449U, 0xe69594bec44de15bU,
>+    0x901d7cf73ab0acd9U, 0xb424dc35095cd80fU, 0xe12e13424bb40e13U,
>+    0x8cbccc096f5088cbU, 0xafebff0bcb24aafeU, 0xdbe6fecebdedd5beU,
>+    0x89705f4136b4a597U, 0xabcc77118461cefcU, 0xd6bf94d5e57a42bcU,
>+    0x8637bd05af6c69b5U, 0xa7c5ac471b478423U, 0xd1b71758e219652bU,
>+    0x83126e978d4fdf3bU, 0xa3d70a3d70a3d70aU, 0xccccccccccccccccU,
>+    0x8000000000000000U, 0xa000000000000000U, 0xc800000000000000U,
>+    0xfa00000000000000U, 0x9c40000000000000U, 0xc350000000000000U,
>+    0xf424000000000000U, 0x9896800000000000U, 0xbebc200000000000U,
>+    0xee6b280000000000U, 0x9502f90000000000U, 0xba43b74000000000U,
>+    0xe8d4a51000000000U, 0x9184e72a00000000U, 0xb5e620f480000000U,
>+    0xe35fa931a0000000U, 0x8e1bc9bf04000000U, 0xb1a2bc2ec5000000U,
>+    0xde0b6b3a76400000U, 0x8ac7230489e80000U, 0xad78ebc5ac620000U,
>+    0xd8d726b7177a8000U, 0x878678326eac9000U, 0xa968163f0a57b400U,
>+    0xd3c21bcecceda100U, 0x84595161401484a0U, 0xa56fa5b99019a5c8U,
>+    0xcecb8f27f4200f3aU, 0x813f3978f8940984U, 0xa18f07d736b90be5U,
>+    0xc9f2c9cd04674edeU, 0xfc6f7c4045812296U, 0x9dc5ada82b70b59dU,
>+    0xc5371912364ce305U, 0xf684df56c3e01bc6U, 0x9a130b963a6c115cU,
>+    0xc097ce7bc90715b3U, 0xf0bdc21abb48db20U, 0x96769950b50d88f4U,
>+    0xbc143fa4e250eb31U, 0xeb194f8e1ae525fdU, 0x92efd1b8d0cf37beU,
>+    0xb7abc627050305adU, 0xe596b7b0c643c719U, 0x8f7e32ce7bea5c6fU,
>+    0xb35dbf821ae4f38bU, 0xe0352f62a19e306eU, 0x8c213d9da502de45U,
>+    0xaf298d050e4395d6U, 0xdaf3f04651d47b4cU, 0x88d8762bf324cd0fU,
>+    0xab0e93b6efee0053U, 0xd5d238a4abe98068U, 0x85a36366eb71f041U,
>+    0xa70c3c40a64e6c51U, 0xd0cf4b50cfe20765U, 0x82818f1281ed449fU,
>+    0xa321f2d7226895c7U, 0xcbea6f8ceb02bb39U, 0xfee50b7025c36a08U,
>+    0x9f4f2726179a2245U, 0xc722f0ef9d80aad6U, 0xf8ebad2b84e0d58bU,
>+    0x9b934c3b330c8577U, 0xc2781f49ffcfa6d5U, 0xf316271c7fc3908aU,
>+    0x97edd871cfda3a56U, 0xbde94e8e43d0c8ecU, 0xed63a231d4c4fb27U,
>+    0x945e455f24fb1cf8U, 0xb975d6b6ee39e436U, 0xe7d34c64a9c85d44U,
>+    0x90e40fbeea1d3a4aU, 0xb51d13aea4a488ddU, 0xe264589a4dcdab14U,
>+    0x8d7eb76070a08aecU, 0xb0de65388cc8ada8U, 0xdd15fe86affad912U,
>+    0x8a2dbf142dfcc7abU, 0xacb92ed9397bf996U, 0xd7e77a8f87daf7fbU,
>+    0x86f0ac99b4e8dafdU, 0xa8acd7c0222311bcU, 0xd2d80db02aabd62bU,
>+    0x83c7088e1aab65dbU, 0xa4b8cab1a1563f52U, 0xcde6fd5e09abcf26U,
>+    0x80b05e5ac60b6178U, 0xa0dc75f1778e39d6U, 0xc913936dd571c84cU,
>+    0xfb5878494ace3a5fU, 0x9d174b2dcec0e47bU, 0xc45d1df942711d9aU,
>+    0xf5746577930d6500U, 0x9968bf6abbe85f20U, 0xbfc2ef456ae276e8U,
>+    0xefb3ab16c59b14a2U, 0x95d04aee3b80ece5U, 0xbb445da9ca61281fU,
>+    0xea1575143cf97226U, 0x924d692ca61be758U, 0xb6e0c377cfa2e12eU,
>+    0xe498f455c38b997aU, 0x8edf98b59a373fecU, 0xb2977ee300c50fe7U,
>+    0xdf3d5e9bc0f653e1U, 0x8b865b215899f46cU, 0xae67f1e9aec07187U,
>+    0xda01ee641a708de9U, 0x884134fe908658b2U, 0xaa51823e34a7eedeU,
>+    0xd4e5e2cdc1d1ea96U, 0x850fadc09923329eU, 0xa6539930bf6bff45U,
>+    0xcfe87f7cef46ff16U, 0x81f14fae158c5f6eU, 0xa26da3999aef7749U,
>+    0xcb090c8001ab551cU, 0xfdcb4fa002162a63U, 0x9e9f11c4014dda7eU,
>+    0xc646d63501a1511dU, 0xf7d88bc24209a565U, 0x9ae757596946075fU,
>+    0xc1a12d2fc3978937U, 0xf209787bb47d6b84U, 0x9745eb4d50ce6332U,
>+    0xbd176620a501fbffU, 0xec5d3fa8ce427affU, 0x93ba47c980e98cdfU,
>+    0xb8a8d9bbe123f017U, 0xe6d3102ad96cec1dU, 0x9043ea1ac7e41392U,
>+    0xb454e4a179dd1877U, 0xe16a1dc9d8545e94U, 0x8ce2529e2734bb1dU,
>+    0xb01ae745b101e9e4U, 0xdc21a1171d42645dU, 0x899504ae72497ebaU,
>+    0xabfa45da0edbde69U, 0xd6f8d7509292d603U, 0x865b86925b9bc5c2U,
>+    0xa7f26836f282b732U, 0xd1ef0244af2364ffU, 0x8335616aed761f1fU,
>+    0xa402b9c5a8d3a6e7U, 0xcd036837130890a1U, 0x802221226be55a64U,
>+    0xa02aa96b06deb0fdU, 0xc83553c5c8965d3dU, 0xfa42a8b73abbf48cU,
>+    0x9c69a97284b578d7U, 0xc38413cf25e2d70dU, 0xf46518c2ef5b8cd1U,
>+    0x98bf2f79d5993802U, 0xbeeefb584aff8603U, 0xeeaaba2e5dbf6784U,
>+    0x952ab45cfa97a0b2U, 0xba756174393d88dfU, 0xe912b9d1478ceb17U,
>+    0x91abb422ccb812eeU, 0xb616a12b7fe617aaU, 0xe39c49765fdf9d94U,
>+    0x8e41ade9fbebc27dU, 0xb1d219647ae6b31cU, 0xde469fbd99a05fe3U,
>+    0x8aec23d680043beeU, 0xada72ccc20054ae9U, 0xd910f7ff28069da4U,
>+    0x87aa9aff79042286U, 0xa99541bf57452b28U, 0xd3fa922f2d1675f2U,
>+    0x847c9b5d7c2e09b7U, 0xa59bc234db398c25U, 0xcf02b2c21207ef2eU,
>+    0x8161afb94b44f57dU, 0xa1ba1ba79e1632dcU, 0xca28a291859bbf93U,
>+    0xfcb2cb35e702af78U, 0x9defbf01b061adabU, 0xc56baec21c7a1916U,
>+    0xf6c69a72a3989f5bU, 0x9a3c2087a63f6399U, 0xc0cb28a98fcf3c7fU,
>+    0xf0fdf2d3f3c30b9fU, 0x969eb7c47859e743U, 0xbc4665b596706114U,
>+    0xeb57ff22fc0c7959U, 0x9316ff75dd87cbd8U, 0xb7dcbf5354e9beceU,
>+    0xe5d3ef282a242e81U, 0x8fa475791a569d10U, 0xb38d92d760ec4455U,
>+    0xe070f78d3927556aU, 0x8c469ab843b89562U, 0xaf58416654a6babbU,
>+    0xdb2e51bfe9d0696aU, 0x88fcf317f22241e2U, 0xab3c2fddeeaad25aU,
>+    0xd60b3bd56a5586f1U, 0x85c7056562757456U, 0xa738c6bebb12d16cU,
>+    0xd106f86e69d785c7U, 0x82a45b450226b39cU, 0xa34d721642b06084U,
>+    0xcc20ce9bd35c78a5U, 0xff290242c83396ceU, 0x9f79a169bd203e41U,
>+    0xc75809c42c684dd1U, 0xf92e0c3537826145U, 0x9bbcc7a142b17ccbU,
>+    0xc2abf989935ddbfeU, 0xf356f7ebf83552feU, 0x98165af37b2153deU,
>+    0xbe1bf1b059e9a8d6U, 0xeda2ee1c7064130cU, 0x9485d4d1c63e8be7U,
>+    0xb9a74a0637ce2ee1U, 0xe8111c87c5c1ba99U, 0x910ab1d4db9914a0U,
>+    0xb54d5e4a127f59c8U, 0xe2a0b5dc971f303aU, 0x8da471a9de737e24U,
>+    0xb10d8e1456105dadU, 0xdd50f1996b947518U, 0x8a5296ffe33cc92fU,
>+    0xace73cbfdc0bfb7bU, 0xd8210befd30efa5aU, 0x8714a775e3e95c78U,
>+    0xa8d9d1535ce3b396U, 0xd31045a8341ca07cU, 0x83ea2b892091e44dU,
>+    0xa4e4b66b68b65d60U, 0xce1de40642e3f4b9U, 0x80d2ae83e9ce78f3U,
>+    0xa1075a24e4421730U, 0xc94930ae1d529cfcU, 0xfb9b7cd9a4a7443cU,
>+    0x9d412e0806e88aa5U, 0xc491798a08a2ad4eU, 0xf5b5d7ec8acb58a2U,
>+    0x9991a6f3d6bf1765U, 0xbff610b0cc6edd3fU, 0xeff394dcff8a948eU,
>+    0x95f83d0a1fb69cd9U, 0xbb764c4ca7a4440fU, 0xea53df5fd18d5513U,
>+    0x92746b9be2f8552cU, 0xb7118682dbb66a77U, 0xe4d5e82392a40515U,
>+    0x8f05b1163ba6832dU, 0xb2c71d5bca9023f8U, 0xdf78e4b2bd342cf6U,
>+    0x8bab8eefb6409c1aU, 0xae9672aba3d0c320U, 0xda3c0f568cc4f3e8U,
>+    0x8865899617fb1871U, 0xaa7eebfb9df9de8dU, 0xd51ea6fa85785631U,
>+    0x8533285c936b35deU, 0xa67ff273b8460356U, 0xd01fef10a657842cU,
>+    0x8213f56a67f6b29bU, 0xa298f2c501f45f42U, 0xcb3f2f7642717713U,
>+    0xfe0efb53d30dd4d7U, 0x9ec95d1463e8a506U, 0xc67bb4597ce2ce48U,
>+    0xf81aa16fdc1b81daU, 0x9b10a4e5e9913128U, 0xc1d4ce1f63f57d72U,
>+    0xf24a01a73cf2dccfU, 0x976e41088617ca01U, 0xbd49d14aa79dbc82U,
>+    0xec9c459d51852ba2U, 0x93e1ab8252f33b45U, 0xb8da1662e7b00a17U,
>+    0xe7109bfba19c0c9dU, 0x906a617d450187e2U, 0xb484f9dc9641e9daU,
>+    0xe1a63853bbd26451U, 0x8d07e33455637eb2U, 0xb049dc016abc5e5fU,
>+    0xdc5c5301c56b75f7U, 0x89b9b3e11b6329baU, 0xac2820d9623bf429U,
>+    0xd732290fbacaf133U, 0x867f59a9d4bed6c0U, 0xa81f301449ee8c70U,
>+    0xd226fc195c6a2f8cU, 0x83585d8fd9c25db7U, 0xa42e74f3d032f525U,
>+    0xcd3a1230c43fb26fU, 0x80444b5e7aa7cf85U, 0xa0555e361951c366U,
>+    0xc86ab5c39fa63440U, 0xfa856334878fc150U, 0x9c935e00d4b9d8d2U,
>+    0xc3b8358109e84f07U, 0xf4a642e14c6262c8U, 0x98e7e9cccfbd7dbdU,
>+    0xbf21e44003acdd2cU, 0xeeea5d5004981478U, 0x95527a5202df0ccbU,
>+    0xbaa718e68396cffdU, 0xe950df20247c83fdU, 0x91d28b7416cdd27eU,
>+    0xb6472e511c81471dU, 0xe3d8f9e563a198e5U, 0x8e679c2f5e44ff8fU,
>+};
>+
>+const int16_t kPower10ExponentTable[] = {
>+    -1200, -1196, -1193, -1190, -1186, -1183, -1180, -1176, -1173, -1170, -1166,
>+    -1163, -1160, -1156, -1153, -1150, -1146, -1143, -1140, -1136, -1133, -1130,
>+    -1127, -1123, -1120, -1117, -1113, -1110, -1107, -1103, -1100, -1097, -1093,
>+    -1090, -1087, -1083, -1080, -1077, -1073, -1070, -1067, -1063, -1060, -1057,
>+    -1053, -1050, -1047, -1043, -1040, -1037, -1034, -1030, -1027, -1024, -1020,
>+    -1017, -1014, -1010, -1007, -1004, -1000, -997,  -994,  -990,  -987,  -984,
>+    -980,  -977,  -974,  -970,  -967,  -964,  -960,  -957,  -954,  -950,  -947,
>+    -944,  -940,  -937,  -934,  -931,  -927,  -924,  -921,  -917,  -914,  -911,
>+    -907,  -904,  -901,  -897,  -894,  -891,  -887,  -884,  -881,  -877,  -874,
>+    -871,  -867,  -864,  -861,  -857,  -854,  -851,  -847,  -844,  -841,  -838,
>+    -834,  -831,  -828,  -824,  -821,  -818,  -814,  -811,  -808,  -804,  -801,
>+    -798,  -794,  -791,  -788,  -784,  -781,  -778,  -774,  -771,  -768,  -764,
>+    -761,  -758,  -754,  -751,  -748,  -744,  -741,  -738,  -735,  -731,  -728,
>+    -725,  -721,  -718,  -715,  -711,  -708,  -705,  -701,  -698,  -695,  -691,
>+    -688,  -685,  -681,  -678,  -675,  -671,  -668,  -665,  -661,  -658,  -655,
>+    -651,  -648,  -645,  -642,  -638,  -635,  -632,  -628,  -625,  -622,  -618,
>+    -615,  -612,  -608,  -605,  -602,  -598,  -595,  -592,  -588,  -585,  -582,
>+    -578,  -575,  -572,  -568,  -565,  -562,  -558,  -555,  -552,  -549,  -545,
>+    -542,  -539,  -535,  -532,  -529,  -525,  -522,  -519,  -515,  -512,  -509,
>+    -505,  -502,  -499,  -495,  -492,  -489,  -485,  -482,  -479,  -475,  -472,
>+    -469,  -465,  -462,  -459,  -455,  -452,  -449,  -446,  -442,  -439,  -436,
>+    -432,  -429,  -426,  -422,  -419,  -416,  -412,  -409,  -406,  -402,  -399,
>+    -396,  -392,  -389,  -386,  -382,  -379,  -376,  -372,  -369,  -366,  -362,
>+    -359,  -356,  -353,  -349,  -346,  -343,  -339,  -336,  -333,  -329,  -326,
>+    -323,  -319,  -316,  -313,  -309,  -306,  -303,  -299,  -296,  -293,  -289,
>+    -286,  -283,  -279,  -276,  -273,  -269,  -266,  -263,  -259,  -256,  -253,
>+    -250,  -246,  -243,  -240,  -236,  -233,  -230,  -226,  -223,  -220,  -216,
>+    -213,  -210,  -206,  -203,  -200,  -196,  -193,  -190,  -186,  -183,  -180,
>+    -176,  -173,  -170,  -166,  -163,  -160,  -157,  -153,  -150,  -147,  -143,
>+    -140,  -137,  -133,  -130,  -127,  -123,  -120,  -117,  -113,  -110,  -107,
>+    -103,  -100,  -97,   -93,   -90,   -87,   -83,   -80,   -77,   -73,   -70,
>+    -67,   -63,   -60,   -57,   -54,   -50,   -47,   -44,   -40,   -37,   -34,
>+    -30,   -27,   -24,   -20,   -17,   -14,   -10,   -7,    -4,    0,     3,
>+    6,     10,    13,    16,    20,    23,    26,    30,    33,    36,    39,
>+    43,    46,    49,    53,    56,    59,    63,    66,    69,    73,    76,
>+    79,    83,    86,    89,    93,    96,    99,    103,   106,   109,   113,
>+    116,   119,   123,   126,   129,   132,   136,   139,   142,   146,   149,
>+    152,   156,   159,   162,   166,   169,   172,   176,   179,   182,   186,
>+    189,   192,   196,   199,   202,   206,   209,   212,   216,   219,   222,
>+    226,   229,   232,   235,   239,   242,   245,   249,   252,   255,   259,
>+    262,   265,   269,   272,   275,   279,   282,   285,   289,   292,   295,
>+    299,   302,   305,   309,   312,   315,   319,   322,   325,   328,   332,
>+    335,   338,   342,   345,   348,   352,   355,   358,   362,   365,   368,
>+    372,   375,   378,   382,   385,   388,   392,   395,   398,   402,   405,
>+    408,   412,   415,   418,   422,   425,   428,   431,   435,   438,   441,
>+    445,   448,   451,   455,   458,   461,   465,   468,   471,   475,   478,
>+    481,   485,   488,   491,   495,   498,   501,   505,   508,   511,   515,
>+    518,   521,   524,   528,   531,   534,   538,   541,   544,   548,   551,
>+    554,   558,   561,   564,   568,   571,   574,   578,   581,   584,   588,
>+    591,   594,   598,   601,   604,   608,   611,   614,   617,   621,   624,
>+    627,   631,   634,   637,   641,   644,   647,   651,   654,   657,   661,
>+    664,   667,   671,   674,   677,   681,   684,   687,   691,   694,   697,
>+    701,   704,   707,   711,   714,   717,   720,   724,   727,   730,   734,
>+    737,   740,   744,   747,   750,   754,   757,   760,   764,   767,   770,
>+    774,   777,   780,   784,   787,   790,   794,   797,   800,   804,   807,
>+    810,   813,   817,   820,   823,   827,   830,   833,   837,   840,   843,
>+    847,   850,   853,   857,   860,   863,   867,   870,   873,   877,   880,
>+    883,   887,   890,   893,   897,   900,   903,   907,   910,   913,   916,
>+    920,   923,   926,   930,   933,   936,   940,   943,   946,   950,   953,
>+    956,   960,
>+};
>+
>+}  // namespace
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/charconv.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/charconv.h
>new file mode 100644
>index 00000000000..3e313679c96
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/charconv.h
>@@ -0,0 +1,115 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#ifndef ABSL_STRINGS_CHARCONV_H_
>+#define ABSL_STRINGS_CHARCONV_H_
>+
>+#include <system_error>  // NOLINT(build/c++11)
>+
>+namespace absl {
>+
>+// Workalike compatibilty version of std::chars_format from C++17.
>+//
>+// This is an bitfield enumerator which can be passed to absl::from_chars to
>+// configure the std::string-to-float conversion.
>+enum class chars_format {
>+  scientific = 1,
>+  fixed = 2,
>+  hex = 4,
>+  general = fixed | scientific,
>+};
>+
>+// The return result of a std::string-to-number conversion.
>+//
>+// `ec` will be set to `invalid_argument` if a well-formed number was not found
>+// at the start of the input range, `result_out_of_range` if a well-formed
>+// number was found, but it was out of the representable range of the requested
>+// type, or to std::errc() otherwise.
>+//
>+// If a well-formed number was found, `ptr` is set to one past the sequence of
>+// characters that were successfully parsed.  If none was found, `ptr` is set
>+// to the `first` argument to from_chars.
>+struct from_chars_result {
>+  const char* ptr;
>+  std::errc ec;
>+};
>+
>+// Workalike compatibilty version of std::from_chars from C++17.  Currently
>+// this only supports the `double` and `float` types.
>+//
>+// This interface incorporates the proposed resolutions for library issues
>+// DR 3800 and DR 3801.  If these are adopted with different wording,
>+// Abseil's behavior will change to match the standard.  (The behavior most
>+// likely to change is for DR 3801, which says what `value` will be set to in
>+// the case of overflow and underflow.  Code that wants to avoid possible
>+// breaking changes in this area should not depend on `value` when the returned
>+// from_chars_result indicates a range error.)
>+//
>+// Searches the range [first, last) for the longest matching pattern beginning
>+// at `first` that represents a floating point number.  If one is found, store
>+// the result in `value`.
>+//
>+// The matching pattern format is almost the same as that of strtod(), except
>+// that C locale is not respected, and an initial '+' character in the input
>+// range will never be matched.
>+//
>+// If `fmt` is set, it must be one of the enumerator values of the chars_format.
>+// (This is despite the fact that chars_format is a bitmask type.)  If set to
>+// `scientific`, a matching number must contain an exponent.  If set to `fixed`,
>+// then an exponent will never match.  (For example, the std::string "1e5" will be
>+// parsed as "1".)  If set to `hex`, then a hexadecimal float is parsed in the
>+// format that strtod() accepts, except that a "0x" prefix is NOT matched.
>+// (In particular, in `hex` mode, the input "0xff" results in the largest
>+// matching pattern "0".)
>+absl::from_chars_result from_chars(const char* first, const char* last,
>+                                   double& value,  // NOLINT
>+                                   chars_format fmt = chars_format::general);
>+
>+absl::from_chars_result from_chars(const char* first, const char* last,
>+                                   float& value,  // NOLINT
>+                                   chars_format fmt = chars_format::general);
>+
>+// std::chars_format is specified as a bitmask type, which means the following
>+// operations must be provided:
>+inline constexpr chars_format operator&(chars_format lhs, chars_format rhs) {
>+  return static_cast<chars_format>(static_cast<int>(lhs) &
>+                                   static_cast<int>(rhs));
>+}
>+inline constexpr chars_format operator|(chars_format lhs, chars_format rhs) {
>+  return static_cast<chars_format>(static_cast<int>(lhs) |
>+                                   static_cast<int>(rhs));
>+}
>+inline constexpr chars_format operator^(chars_format lhs, chars_format rhs) {
>+  return static_cast<chars_format>(static_cast<int>(lhs) ^
>+                                   static_cast<int>(rhs));
>+}
>+inline constexpr chars_format operator~(chars_format arg) {
>+  return static_cast<chars_format>(~static_cast<int>(arg));
>+}
>+inline chars_format& operator&=(chars_format& lhs, chars_format rhs) {
>+  lhs = lhs & rhs;
>+  return lhs;
>+}
>+inline chars_format& operator|=(chars_format& lhs, chars_format rhs) {
>+  lhs = lhs | rhs;
>+  return lhs;
>+}
>+inline chars_format& operator^=(chars_format& lhs, chars_format rhs) {
>+  lhs = lhs ^ rhs;
>+  return lhs;
>+}
>+
>+}  // namespace absl
>+
>+#endif  // ABSL_STRINGS_CHARCONV_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/charconv_benchmark.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/charconv_benchmark.cc
>new file mode 100644
>index 00000000000..fd83f44e3d0
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/charconv_benchmark.cc
>@@ -0,0 +1,204 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/strings/charconv.h"
>+
>+#include <cstdlib>
>+#include <cstring>
>+#include <string>
>+
>+#include "benchmark/benchmark.h"
>+
>+namespace {
>+
>+void BM_Strtod_Pi(benchmark::State& state) {
>+  const char* pi = "3.14159";
>+  for (auto s : state) {
>+    benchmark::DoNotOptimize(pi);
>+    benchmark::DoNotOptimize(strtod(pi, nullptr));
>+  }
>+}
>+BENCHMARK(BM_Strtod_Pi);
>+
>+void BM_Absl_Pi(benchmark::State& state) {
>+  const char* pi = "3.14159";
>+  const char* pi_end = pi + strlen(pi);
>+  for (auto s : state) {
>+    benchmark::DoNotOptimize(pi);
>+    double v;
>+    absl::from_chars(pi, pi_end, v);
>+    benchmark::DoNotOptimize(v);
>+  }
>+}
>+BENCHMARK(BM_Absl_Pi);
>+
>+void BM_Strtod_Pi_float(benchmark::State& state) {
>+  const char* pi = "3.14159";
>+  for (auto s : state) {
>+    benchmark::DoNotOptimize(pi);
>+    benchmark::DoNotOptimize(strtof(pi, nullptr));
>+  }
>+}
>+BENCHMARK(BM_Strtod_Pi_float);
>+
>+void BM_Absl_Pi_float(benchmark::State& state) {
>+  const char* pi = "3.14159";
>+  const char* pi_end = pi + strlen(pi);
>+  for (auto s : state) {
>+    benchmark::DoNotOptimize(pi);
>+    float v;
>+    absl::from_chars(pi, pi_end, v);
>+    benchmark::DoNotOptimize(v);
>+  }
>+}
>+BENCHMARK(BM_Absl_Pi_float);
>+
>+void BM_Strtod_HardLarge(benchmark::State& state) {
>+  const char* num = "272104041512242479.e200";
>+  for (auto s : state) {
>+    benchmark::DoNotOptimize(num);
>+    benchmark::DoNotOptimize(strtod(num, nullptr));
>+  }
>+}
>+BENCHMARK(BM_Strtod_HardLarge);
>+
>+void BM_Absl_HardLarge(benchmark::State& state) {
>+  const char* numstr = "272104041512242479.e200";
>+  const char* numstr_end = numstr + strlen(numstr);
>+  for (auto s : state) {
>+    benchmark::DoNotOptimize(numstr);
>+    double v;
>+    absl::from_chars(numstr, numstr_end, v);
>+    benchmark::DoNotOptimize(v);
>+  }
>+}
>+BENCHMARK(BM_Absl_HardLarge);
>+
>+void BM_Strtod_HardSmall(benchmark::State& state) {
>+  const char* num = "94080055902682397.e-242";
>+  for (auto s : state) {
>+    benchmark::DoNotOptimize(num);
>+    benchmark::DoNotOptimize(strtod(num, nullptr));
>+  }
>+}
>+BENCHMARK(BM_Strtod_HardSmall);
>+
>+void BM_Absl_HardSmall(benchmark::State& state) {
>+  const char* numstr = "94080055902682397.e-242";
>+  const char* numstr_end = numstr + strlen(numstr);
>+  for (auto s : state) {
>+    benchmark::DoNotOptimize(numstr);
>+    double v;
>+    absl::from_chars(numstr, numstr_end, v);
>+    benchmark::DoNotOptimize(v);
>+  }
>+}
>+BENCHMARK(BM_Absl_HardSmall);
>+
>+void BM_Strtod_HugeMantissa(benchmark::State& state) {
>+  std::string huge(200, '3');
>+  const char* num = huge.c_str();
>+  for (auto s : state) {
>+    benchmark::DoNotOptimize(num);
>+    benchmark::DoNotOptimize(strtod(num, nullptr));
>+  }
>+}
>+BENCHMARK(BM_Strtod_HugeMantissa);
>+
>+void BM_Absl_HugeMantissa(benchmark::State& state) {
>+  std::string huge(200, '3');
>+  const char* num = huge.c_str();
>+  const char* num_end = num + 200;
>+  for (auto s : state) {
>+    benchmark::DoNotOptimize(num);
>+    double v;
>+    absl::from_chars(num, num_end, v);
>+    benchmark::DoNotOptimize(v);
>+  }
>+}
>+BENCHMARK(BM_Absl_HugeMantissa);
>+
>+std::string MakeHardCase(int length) {
>+  // The number 1.1521...e-297 is exactly halfway between 12345 * 2**-1000 and
>+  // the next larger representable number.  The digits of this number are in
>+  // the std::string below.
>+  const std::string digits =
>+      "1."
>+      "152113937042223790993097181572444900347587985074226836242307364987727724"
>+      "831384300183638649152607195040591791364113930628852279348613864894524591"
>+      "272746490313676832900762939595690019745859128071117417798540258114233761"
>+      "012939937017879509401007964861774960297319002612457273148497158989073482"
>+      "171377406078223015359818300988676687994537274548940612510414856761641652"
>+      "513434981938564294004070500716200446656421722229202383105446378511678258"
>+      "370570631774499359748259931676320916632111681001853983492795053244971606"
>+      "922718923011680846577744433974087653954904214152517799883551075537146316"
>+      "168973685866425605046988661997658648354773076621610279716804960009043764"
>+      "038392994055171112475093876476783502487512538082706095923790634572014823"
>+      "78877699375152587890625" +
>+      std::string(5000, '0');
>+  // generate the hard cases on either side for the given length.
>+  // Lengths between 3 and 1000 are reasonable.
>+  return digits.substr(0, length) + "1e-297";
>+}
>+
>+void BM_Strtod_Big_And_Difficult(benchmark::State& state) {
>+  std::string testcase = MakeHardCase(state.range(0));
>+  const char* begin = testcase.c_str();
>+  for (auto s : state) {
>+    benchmark::DoNotOptimize(begin);
>+    benchmark::DoNotOptimize(strtod(begin, nullptr));
>+  }
>+}
>+BENCHMARK(BM_Strtod_Big_And_Difficult)->Range(3, 5000);
>+
>+void BM_Absl_Big_And_Difficult(benchmark::State& state) {
>+  std::string testcase = MakeHardCase(state.range(0));
>+  const char* begin = testcase.c_str();
>+  const char* end = begin + testcase.size();
>+  for (auto s : state) {
>+    benchmark::DoNotOptimize(begin);
>+    double v;
>+    absl::from_chars(begin, end, v);
>+    benchmark::DoNotOptimize(v);
>+  }
>+}
>+BENCHMARK(BM_Absl_Big_And_Difficult)->Range(3, 5000);
>+
>+}  // namespace
>+
>+// ------------------------------------------------------------------------
>+// Benchmark                                 Time           CPU Iterations
>+// ------------------------------------------------------------------------
>+// BM_Strtod_Pi                             96 ns         96 ns    6337454
>+// BM_Absl_Pi                               35 ns         35 ns   20031996
>+// BM_Strtod_Pi_float                       91 ns         91 ns    7745851
>+// BM_Absl_Pi_float                         35 ns         35 ns   20430298
>+// BM_Strtod_HardLarge                     133 ns        133 ns    5288341
>+// BM_Absl_HardLarge                       181 ns        181 ns    3855615
>+// BM_Strtod_HardSmall                     279 ns        279 ns    2517243
>+// BM_Absl_HardSmall                       287 ns        287 ns    2458744
>+// BM_Strtod_HugeMantissa                  433 ns        433 ns    1604293
>+// BM_Absl_HugeMantissa                    160 ns        160 ns    4403671
>+// BM_Strtod_Big_And_Difficult/3           236 ns        236 ns    2942496
>+// BM_Strtod_Big_And_Difficult/8           232 ns        232 ns    2983796
>+// BM_Strtod_Big_And_Difficult/64          437 ns        437 ns    1591951
>+// BM_Strtod_Big_And_Difficult/512        1738 ns       1738 ns     402519
>+// BM_Strtod_Big_And_Difficult/4096       3943 ns       3943 ns     176128
>+// BM_Strtod_Big_And_Difficult/5000       4397 ns       4397 ns     157878
>+// BM_Absl_Big_And_Difficult/3              39 ns         39 ns   17799583
>+// BM_Absl_Big_And_Difficult/8              43 ns         43 ns   16096859
>+// BM_Absl_Big_And_Difficult/64            550 ns        550 ns    1259717
>+// BM_Absl_Big_And_Difficult/512          4167 ns       4167 ns     171414
>+// BM_Absl_Big_And_Difficult/4096         9160 ns       9159 ns      76297
>+// BM_Absl_Big_And_Difficult/5000         9738 ns       9738 ns      70140
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/charconv_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/charconv_test.cc
>new file mode 100644
>index 00000000000..f8d71cc6c44
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/charconv_test.cc
>@@ -0,0 +1,766 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/strings/charconv.h"
>+
>+#include <cstdlib>
>+#include <string>
>+
>+#include "gmock/gmock.h"
>+#include "gtest/gtest.h"
>+#include "absl/strings/str_cat.h"
>+
>+#ifdef _MSC_FULL_VER
>+#define ABSL_COMPILER_DOES_EXACT_ROUNDING 0
>+#define ABSL_STRTOD_HANDLES_NAN_CORRECTLY 0
>+#else
>+#define ABSL_COMPILER_DOES_EXACT_ROUNDING 1
>+#define ABSL_STRTOD_HANDLES_NAN_CORRECTLY 1
>+#endif
>+
>+namespace {
>+
>+#if ABSL_COMPILER_DOES_EXACT_ROUNDING
>+
>+// Tests that the given std::string is accepted by absl::from_chars, and that it
>+// converts exactly equal to the given number.
>+void TestDoubleParse(absl::string_view str, double expected_number) {
>+  SCOPED_TRACE(str);
>+  double actual_number = 0.0;
>+  absl::from_chars_result result =
>+      absl::from_chars(str.data(), str.data() + str.length(), actual_number);
>+  EXPECT_EQ(result.ec, std::errc());
>+  EXPECT_EQ(result.ptr, str.data() + str.length());
>+  EXPECT_EQ(actual_number, expected_number);
>+}
>+
>+void TestFloatParse(absl::string_view str, float expected_number) {
>+  SCOPED_TRACE(str);
>+  float actual_number = 0.0;
>+  absl::from_chars_result result =
>+      absl::from_chars(str.data(), str.data() + str.length(), actual_number);
>+  EXPECT_EQ(result.ec, std::errc());
>+  EXPECT_EQ(result.ptr, str.data() + str.length());
>+  EXPECT_EQ(actual_number, expected_number);
>+}
>+
>+// Tests that the given double or single precision floating point literal is
>+// parsed correctly by absl::from_chars.
>+//
>+// These convenience macros assume that the C++ compiler being used also does
>+// fully correct decimal-to-binary conversions.
>+#define FROM_CHARS_TEST_DOUBLE(number)     \
>+  {                                        \
>+    TestDoubleParse(#number, number);      \
>+    TestDoubleParse("-" #number, -number); \
>+  }
>+
>+#define FROM_CHARS_TEST_FLOAT(number)        \
>+  {                                          \
>+    TestFloatParse(#number, number##f);      \
>+    TestFloatParse("-" #number, -number##f); \
>+  }
>+
>+TEST(FromChars, NearRoundingCases) {
>+  // Cases from "A Program for Testing IEEE Decimal-Binary Conversion"
>+  // by Vern Paxson.
>+
>+  // Forms that should round towards zero.  (These are the hardest cases for
>+  // each decimal mantissa size.)
>+  FROM_CHARS_TEST_DOUBLE(5.e125);
>+  FROM_CHARS_TEST_DOUBLE(69.e267);
>+  FROM_CHARS_TEST_DOUBLE(999.e-026);
>+  FROM_CHARS_TEST_DOUBLE(7861.e-034);
>+  FROM_CHARS_TEST_DOUBLE(75569.e-254);
>+  FROM_CHARS_TEST_DOUBLE(928609.e-261);
>+  FROM_CHARS_TEST_DOUBLE(9210917.e080);
>+  FROM_CHARS_TEST_DOUBLE(84863171.e114);
>+  FROM_CHARS_TEST_DOUBLE(653777767.e273);
>+  FROM_CHARS_TEST_DOUBLE(5232604057.e-298);
>+  FROM_CHARS_TEST_DOUBLE(27235667517.e-109);
>+  FROM_CHARS_TEST_DOUBLE(653532977297.e-123);
>+  FROM_CHARS_TEST_DOUBLE(3142213164987.e-294);
>+  FROM_CHARS_TEST_DOUBLE(46202199371337.e-072);
>+  FROM_CHARS_TEST_DOUBLE(231010996856685.e-073);
>+  FROM_CHARS_TEST_DOUBLE(9324754620109615.e212);
>+  FROM_CHARS_TEST_DOUBLE(78459735791271921.e049);
>+  FROM_CHARS_TEST_DOUBLE(272104041512242479.e200);
>+  FROM_CHARS_TEST_DOUBLE(6802601037806061975.e198);
>+  FROM_CHARS_TEST_DOUBLE(20505426358836677347.e-221);
>+  FROM_CHARS_TEST_DOUBLE(836168422905420598437.e-234);
>+  FROM_CHARS_TEST_DOUBLE(4891559871276714924261.e222);
>+  FROM_CHARS_TEST_FLOAT(5.e-20);
>+  FROM_CHARS_TEST_FLOAT(67.e14);
>+  FROM_CHARS_TEST_FLOAT(985.e15);
>+  FROM_CHARS_TEST_FLOAT(7693.e-42);
>+  FROM_CHARS_TEST_FLOAT(55895.e-16);
>+  FROM_CHARS_TEST_FLOAT(996622.e-44);
>+  FROM_CHARS_TEST_FLOAT(7038531.e-32);
>+  FROM_CHARS_TEST_FLOAT(60419369.e-46);
>+  FROM_CHARS_TEST_FLOAT(702990899.e-20);
>+  FROM_CHARS_TEST_FLOAT(6930161142.e-48);
>+  FROM_CHARS_TEST_FLOAT(25933168707.e-13);
>+  FROM_CHARS_TEST_FLOAT(596428896559.e20);
>+
>+  // Similarly, forms that should round away from zero.
>+  FROM_CHARS_TEST_DOUBLE(9.e-265);
>+  FROM_CHARS_TEST_DOUBLE(85.e-037);
>+  FROM_CHARS_TEST_DOUBLE(623.e100);
>+  FROM_CHARS_TEST_DOUBLE(3571.e263);
>+  FROM_CHARS_TEST_DOUBLE(81661.e153);
>+  FROM_CHARS_TEST_DOUBLE(920657.e-023);
>+  FROM_CHARS_TEST_DOUBLE(4603285.e-024);
>+  FROM_CHARS_TEST_DOUBLE(87575437.e-309);
>+  FROM_CHARS_TEST_DOUBLE(245540327.e122);
>+  FROM_CHARS_TEST_DOUBLE(6138508175.e120);
>+  FROM_CHARS_TEST_DOUBLE(83356057653.e193);
>+  FROM_CHARS_TEST_DOUBLE(619534293513.e124);
>+  FROM_CHARS_TEST_DOUBLE(2335141086879.e218);
>+  FROM_CHARS_TEST_DOUBLE(36167929443327.e-159);
>+  FROM_CHARS_TEST_DOUBLE(609610927149051.e-255);
>+  FROM_CHARS_TEST_DOUBLE(3743626360493413.e-165);
>+  FROM_CHARS_TEST_DOUBLE(94080055902682397.e-242);
>+  FROM_CHARS_TEST_DOUBLE(899810892172646163.e283);
>+  FROM_CHARS_TEST_DOUBLE(7120190517612959703.e120);
>+  FROM_CHARS_TEST_DOUBLE(25188282901709339043.e-252);
>+  FROM_CHARS_TEST_DOUBLE(308984926168550152811.e-052);
>+  FROM_CHARS_TEST_DOUBLE(6372891218502368041059.e064);
>+  FROM_CHARS_TEST_FLOAT(3.e-23);
>+  FROM_CHARS_TEST_FLOAT(57.e18);
>+  FROM_CHARS_TEST_FLOAT(789.e-35);
>+  FROM_CHARS_TEST_FLOAT(2539.e-18);
>+  FROM_CHARS_TEST_FLOAT(76173.e28);
>+  FROM_CHARS_TEST_FLOAT(887745.e-11);
>+  FROM_CHARS_TEST_FLOAT(5382571.e-37);
>+  FROM_CHARS_TEST_FLOAT(82381273.e-35);
>+  FROM_CHARS_TEST_FLOAT(750486563.e-38);
>+  FROM_CHARS_TEST_FLOAT(3752432815.e-39);
>+  FROM_CHARS_TEST_FLOAT(75224575729.e-45);
>+  FROM_CHARS_TEST_FLOAT(459926601011.e15);
>+}
>+
>+#undef FROM_CHARS_TEST_DOUBLE
>+#undef FROM_CHARS_TEST_FLOAT
>+#endif
>+
>+float ToFloat(absl::string_view s) {
>+  float f;
>+  absl::from_chars(s.data(), s.data() + s.size(), f);
>+  return f;
>+}
>+
>+double ToDouble(absl::string_view s) {
>+  double d;
>+  absl::from_chars(s.data(), s.data() + s.size(), d);
>+  return d;
>+}
>+
>+// A duplication of the test cases in "NearRoundingCases" above, but with
>+// expected values expressed with integers, using ldexp/ldexpf.  These test
>+// cases will work even on compilers that do not accurately round floating point
>+// literals.
>+TEST(FromChars, NearRoundingCasesExplicit) {
>+  EXPECT_EQ(ToDouble("5.e125"), ldexp(6653062250012735, 365));
>+  EXPECT_EQ(ToDouble("69.e267"), ldexp(4705683757438170, 841));
>+  EXPECT_EQ(ToDouble("999.e-026"), ldexp(6798841691080350, -129));
>+  EXPECT_EQ(ToDouble("7861.e-034"), ldexp(8975675289889240, -153));
>+  EXPECT_EQ(ToDouble("75569.e-254"), ldexp(6091718967192243, -880));
>+  EXPECT_EQ(ToDouble("928609.e-261"), ldexp(7849264900213743, -900));
>+  EXPECT_EQ(ToDouble("9210917.e080"), ldexp(8341110837370930, 236));
>+  EXPECT_EQ(ToDouble("84863171.e114"), ldexp(4625202867375927, 353));
>+  EXPECT_EQ(ToDouble("653777767.e273"), ldexp(5068902999763073, 884));
>+  EXPECT_EQ(ToDouble("5232604057.e-298"), ldexp(5741343011915040, -1010));
>+  EXPECT_EQ(ToDouble("27235667517.e-109"), ldexp(6707124626673586, -380));
>+  EXPECT_EQ(ToDouble("653532977297.e-123"), ldexp(7078246407265384, -422));
>+  EXPECT_EQ(ToDouble("3142213164987.e-294"), ldexp(8219991337640559, -988));
>+  EXPECT_EQ(ToDouble("46202199371337.e-072"), ldexp(5224462102115359, -246));
>+  EXPECT_EQ(ToDouble("231010996856685.e-073"), ldexp(5224462102115359, -247));
>+  EXPECT_EQ(ToDouble("9324754620109615.e212"), ldexp(5539753864394442, 705));
>+  EXPECT_EQ(ToDouble("78459735791271921.e049"), ldexp(8388176519442766, 166));
>+  EXPECT_EQ(ToDouble("272104041512242479.e200"), ldexp(5554409530847367, 670));
>+  EXPECT_EQ(ToDouble("6802601037806061975.e198"), ldexp(5554409530847367, 668));
>+  EXPECT_EQ(ToDouble("20505426358836677347.e-221"),
>+            ldexp(4524032052079546, -722));
>+  EXPECT_EQ(ToDouble("836168422905420598437.e-234"),
>+            ldexp(5070963299887562, -760));
>+  EXPECT_EQ(ToDouble("4891559871276714924261.e222"),
>+            ldexp(6452687840519111, 757));
>+  EXPECT_EQ(ToFloat("5.e-20"), ldexpf(15474250, -88));
>+  EXPECT_EQ(ToFloat("67.e14"), ldexpf(12479722, 29));
>+  EXPECT_EQ(ToFloat("985.e15"), ldexpf(14333636, 36));
>+  EXPECT_EQ(ToFloat("7693.e-42"), ldexpf(10979816, -150));
>+  EXPECT_EQ(ToFloat("55895.e-16"), ldexpf(12888509, -61));
>+  EXPECT_EQ(ToFloat("996622.e-44"), ldexpf(14224264, -150));
>+  EXPECT_EQ(ToFloat("7038531.e-32"), ldexpf(11420669, -107));
>+  EXPECT_EQ(ToFloat("60419369.e-46"), ldexpf(8623340, -150));
>+  EXPECT_EQ(ToFloat("702990899.e-20"), ldexpf(16209866, -61));
>+  EXPECT_EQ(ToFloat("6930161142.e-48"), ldexpf(9891056, -150));
>+  EXPECT_EQ(ToFloat("25933168707.e-13"), ldexpf(11138211, -32));
>+  EXPECT_EQ(ToFloat("596428896559.e20"), ldexpf(12333860, 82));
>+
>+
>+  EXPECT_EQ(ToDouble("9.e-265"), ldexp(8168427841980010, -930));
>+  EXPECT_EQ(ToDouble("85.e-037"), ldexp(6360455125664090, -169));
>+  EXPECT_EQ(ToDouble("623.e100"), ldexp(6263531988747231, 289));
>+  EXPECT_EQ(ToDouble("3571.e263"), ldexp(6234526311072170, 833));
>+  EXPECT_EQ(ToDouble("81661.e153"), ldexp(6696636728760206, 472));
>+  EXPECT_EQ(ToDouble("920657.e-023"), ldexp(5975405561110124, -109));
>+  EXPECT_EQ(ToDouble("4603285.e-024"), ldexp(5975405561110124, -110));
>+  EXPECT_EQ(ToDouble("87575437.e-309"), ldexp(8452160731874668, -1053));
>+  EXPECT_EQ(ToDouble("245540327.e122"), ldexp(4985336549131723, 381));
>+  EXPECT_EQ(ToDouble("6138508175.e120"), ldexp(4985336549131723, 379));
>+  EXPECT_EQ(ToDouble("83356057653.e193"), ldexp(5986732817132056, 625));
>+  EXPECT_EQ(ToDouble("619534293513.e124"), ldexp(4798406992060657, 399));
>+  EXPECT_EQ(ToDouble("2335141086879.e218"), ldexp(5419088166961646, 713));
>+  EXPECT_EQ(ToDouble("36167929443327.e-159"), ldexp(8135819834632444, -536));
>+  EXPECT_EQ(ToDouble("609610927149051.e-255"), ldexp(4576664294594737, -850));
>+  EXPECT_EQ(ToDouble("3743626360493413.e-165"), ldexp(6898586531774201, -549));
>+  EXPECT_EQ(ToDouble("94080055902682397.e-242"), ldexp(6273271706052298, -800));
>+  EXPECT_EQ(ToDouble("899810892172646163.e283"), ldexp(7563892574477827, 947));
>+  EXPECT_EQ(ToDouble("7120190517612959703.e120"), ldexp(5385467232557565, 409));
>+  EXPECT_EQ(ToDouble("25188282901709339043.e-252"),
>+            ldexp(5635662608542340, -825));
>+  EXPECT_EQ(ToDouble("308984926168550152811.e-052"),
>+            ldexp(5644774693823803, -157));
>+  EXPECT_EQ(ToDouble("6372891218502368041059.e064"),
>+            ldexp(4616868614322430, 233));
>+
>+  EXPECT_EQ(ToFloat("3.e-23"), ldexpf(9507380, -98));
>+  EXPECT_EQ(ToFloat("57.e18"), ldexpf(12960300, 42));
>+  EXPECT_EQ(ToFloat("789.e-35"), ldexpf(10739312, -130));
>+  EXPECT_EQ(ToFloat("2539.e-18"), ldexpf(11990089, -72));
>+  EXPECT_EQ(ToFloat("76173.e28"), ldexpf(9845130, 86));
>+  EXPECT_EQ(ToFloat("887745.e-11"), ldexpf(9760860, -40));
>+  EXPECT_EQ(ToFloat("5382571.e-37"), ldexpf(11447463, -124));
>+  EXPECT_EQ(ToFloat("82381273.e-35"), ldexpf(8554961, -113));
>+  EXPECT_EQ(ToFloat("750486563.e-38"), ldexpf(9975678, -120));
>+  EXPECT_EQ(ToFloat("3752432815.e-39"), ldexpf(9975678, -121));
>+  EXPECT_EQ(ToFloat("75224575729.e-45"), ldexpf(13105970, -137));
>+  EXPECT_EQ(ToFloat("459926601011.e15"), ldexpf(12466336, 65));
>+}
>+
>+// Common test logic for converting a std::string which lies exactly halfway between
>+// two target floats.
>+//
>+// mantissa and exponent represent the precise value between two floating point
>+// numbers, `expected_low` and `expected_high`.  The floating point
>+// representation to parse in `StrCat(mantissa, "e", exponent)`.
>+//
>+// This function checks that an input just slightly less than the exact value
>+// is rounded down to `expected_low`, and an input just slightly greater than
>+// the exact value is rounded up to `expected_high`.
>+//
>+// The exact value should round to `expected_half`, which must be either
>+// `expected_low` or `expected_high`.
>+template <typename FloatType>
>+void TestHalfwayValue(const std::string& mantissa, int exponent,
>+                      FloatType expected_low, FloatType expected_high,
>+                      FloatType expected_half) {
>+  std::string low_rep = mantissa;
>+  low_rep[low_rep.size() - 1] -= 1;
>+  absl::StrAppend(&low_rep, std::string(1000, '9'), "e", exponent);
>+
>+  FloatType actual_low = 0;
>+  absl::from_chars(low_rep.data(), low_rep.data() + low_rep.size(), actual_low);
>+  EXPECT_EQ(expected_low, actual_low);
>+
>+  std::string high_rep = absl::StrCat(mantissa, std::string(1000, '0'), "1e", exponent);
>+  FloatType actual_high = 0;
>+  absl::from_chars(high_rep.data(), high_rep.data() + high_rep.size(),
>+                   actual_high);
>+  EXPECT_EQ(expected_high, actual_high);
>+
>+  std::string halfway_rep = absl::StrCat(mantissa, "e", exponent);
>+  FloatType actual_half = 0;
>+  absl::from_chars(halfway_rep.data(), halfway_rep.data() + halfway_rep.size(),
>+                   actual_half);
>+  EXPECT_EQ(expected_half, actual_half);
>+}
>+
>+TEST(FromChars, DoubleRounding) {
>+  const double zero = 0.0;
>+  const double first_subnormal = nextafter(zero, 1.0);
>+  const double second_subnormal = nextafter(first_subnormal, 1.0);
>+
>+  const double first_normal = DBL_MIN;
>+  const double last_subnormal = nextafter(first_normal, 0.0);
>+  const double second_normal = nextafter(first_normal, 1.0);
>+
>+  const double last_normal = DBL_MAX;
>+  const double penultimate_normal = nextafter(last_normal, 0.0);
>+
>+  // Various test cases for numbers between two representable floats.  Each
>+  // call to TestHalfwayValue tests a number just below and just above the
>+  // halfway point, as well as the number exactly between them.
>+
>+  // Test between zero and first_subnormal.  Round-to-even tie rounds down.
>+  TestHalfwayValue(
>+      "2."
>+      "470328229206232720882843964341106861825299013071623822127928412503377536"
>+      "351043759326499181808179961898982823477228588654633283551779698981993873"
>+      "980053909390631503565951557022639229085839244910518443593180284993653615"
>+      "250031937045767824921936562366986365848075700158576926990370631192827955"
>+      "855133292783433840935197801553124659726357957462276646527282722005637400"
>+      "648549997709659947045402082816622623785739345073633900796776193057750674"
>+      "017632467360096895134053553745851666113422376667860416215968046191446729"
>+      "184030053005753084904876539171138659164623952491262365388187963623937328"
>+      "042389101867234849766823508986338858792562830275599565752445550725518931"
>+      "369083625477918694866799496832404970582102851318545139621383772282614543"
>+      "7693412532098591327667236328125",
>+      -324, zero, first_subnormal, zero);
>+
>+  // first_subnormal and second_subnormal.  Round-to-even tie rounds up.
>+  TestHalfwayValue(
>+      "7."
>+      "410984687618698162648531893023320585475897039214871466383785237510132609"
>+      "053131277979497545424539885696948470431685765963899850655339096945981621"
>+      "940161728171894510697854671067917687257517734731555330779540854980960845"
>+      "750095811137303474765809687100959097544227100475730780971111893578483867"
>+      "565399878350301522805593404659373979179073872386829939581848166016912201"
>+      "945649993128979841136206248449867871357218035220901702390328579173252022"
>+      "052897402080290685402160661237554998340267130003581248647904138574340187"
>+      "552090159017259254714629617513415977493871857473787096164563890871811984"
>+      "127167305601704549300470526959016576377688490826798697257336652176556794"
>+      "107250876433756084600398490497214911746308553955635418864151316847843631"
>+      "3080237596295773983001708984375",
>+      -324, first_subnormal, second_subnormal, second_subnormal);
>+
>+  // last_subnormal and first_normal.  Round-to-even tie rounds up.
>+  TestHalfwayValue(
>+      "2."
>+      "225073858507201136057409796709131975934819546351645648023426109724822222"
>+      "021076945516529523908135087914149158913039621106870086438694594645527657"
>+      "207407820621743379988141063267329253552286881372149012981122451451889849"
>+      "057222307285255133155755015914397476397983411801999323962548289017107081"
>+      "850690630666655994938275772572015763062690663332647565300009245888316433"
>+      "037779791869612049497390377829704905051080609940730262937128958950003583"
>+      "799967207254304360284078895771796150945516748243471030702609144621572289"
>+      "880258182545180325707018860872113128079512233426288368622321503775666622"
>+      "503982534335974568884423900265498198385487948292206894721689831099698365"
>+      "846814022854243330660339850886445804001034933970427567186443383770486037"
>+      "86162277173854562306587467901408672332763671875",
>+      -308, last_subnormal, first_normal, first_normal);
>+
>+  // first_normal and second_normal.  Round-to-even tie rounds down.
>+  TestHalfwayValue(
>+      "2."
>+      "225073858507201630123055637955676152503612414573018013083228724049586647"
>+      "606759446192036794116886953213985520549032000903434781884412325572184367"
>+      "563347617020518175998922941393629966742598285899994830148971433555578567"
>+      "693279306015978183162142425067962460785295885199272493577688320732492479"
>+      "924816869232247165964934329258783950102250973957579510571600738343645738"
>+      "494324192997092179207389919761694314131497173265255020084997973676783743"
>+      "155205818804439163810572367791175177756227497413804253387084478193655533"
>+      "073867420834526162513029462022730109054820067654020201547112002028139700"
>+      "141575259123440177362244273712468151750189745559978653234255886219611516"
>+      "335924167958029604477064946470184777360934300451421683607013647479513962"
>+      "13837722826145437693412532098591327667236328125",
>+      -308, first_normal, second_normal, first_normal);
>+
>+  // penultimate_normal and last_normal.  Round-to-even rounds down.
>+  TestHalfwayValue(
>+      "1."
>+      "797693134862315608353258760581052985162070023416521662616611746258695532"
>+      "672923265745300992879465492467506314903358770175220871059269879629062776"
>+      "047355692132901909191523941804762171253349609463563872612866401980290377"
>+      "995141836029815117562837277714038305214839639239356331336428021390916694"
>+      "57927874464075218944",
>+      308, penultimate_normal, last_normal, penultimate_normal);
>+}
>+
>+// Same test cases as DoubleRounding, now with new and improved Much Smaller
>+// Precision!
>+TEST(FromChars, FloatRounding) {
>+  const float zero = 0.0;
>+  const float first_subnormal = nextafterf(zero, 1.0);
>+  const float second_subnormal = nextafterf(first_subnormal, 1.0);
>+
>+  const float first_normal = FLT_MIN;
>+  const float last_subnormal = nextafterf(first_normal, 0.0);
>+  const float second_normal = nextafterf(first_normal, 1.0);
>+
>+  const float last_normal = FLT_MAX;
>+  const float penultimate_normal = nextafterf(last_normal, 0.0);
>+
>+  // Test between zero and first_subnormal.  Round-to-even tie rounds down.
>+  TestHalfwayValue(
>+      "7."
>+      "006492321624085354618647916449580656401309709382578858785341419448955413"
>+      "42930300743319094181060791015625",
>+      -46, zero, first_subnormal, zero);
>+
>+  // first_subnormal and second_subnormal.  Round-to-even tie rounds up.
>+  TestHalfwayValue(
>+      "2."
>+      "101947696487225606385594374934874196920392912814773657635602425834686624"
>+      "028790902229957282543182373046875",
>+      -45, first_subnormal, second_subnormal, second_subnormal);
>+
>+  // last_subnormal and first_normal.  Round-to-even tie rounds up.
>+  TestHalfwayValue(
>+      "1."
>+      "175494280757364291727882991035766513322858992758990427682963118425003064"
>+      "9651730385585324256680905818939208984375",
>+      -38, last_subnormal, first_normal, first_normal);
>+
>+  // first_normal and second_normal.  Round-to-even tie rounds down.
>+  TestHalfwayValue(
>+      "1."
>+      "175494420887210724209590083408724842314472120785184615334540294131831453"
>+      "9442813071445925743319094181060791015625",
>+      -38, first_normal, second_normal, first_normal);
>+
>+  // penultimate_normal and last_normal.  Round-to-even rounds down.
>+  TestHalfwayValue("3.40282336497324057985868971510891282432", 38,
>+                   penultimate_normal, last_normal, penultimate_normal);
>+}
>+
>+TEST(FromChars, Underflow) {
>+  // Check that underflow is handled correctly, according to the specification
>+  // in DR 3081.
>+  double d;
>+  float f;
>+  absl::from_chars_result result;
>+
>+  std::string negative_underflow = "-1e-1000";
>+  const char* begin = negative_underflow.data();
>+  const char* end = begin + negative_underflow.size();
>+  d = 100.0;
>+  result = absl::from_chars(begin, end, d);
>+  EXPECT_EQ(result.ptr, end);
>+  EXPECT_EQ(result.ec, std::errc::result_out_of_range);
>+  EXPECT_TRUE(std::signbit(d));  // negative
>+  EXPECT_GE(d, -std::numeric_limits<double>::min());
>+  f = 100.0;
>+  result = absl::from_chars(begin, end, f);
>+  EXPECT_EQ(result.ptr, end);
>+  EXPECT_EQ(result.ec, std::errc::result_out_of_range);
>+  EXPECT_TRUE(std::signbit(f));  // negative
>+  EXPECT_GE(f, -std::numeric_limits<float>::min());
>+
>+  std::string positive_underflow = "1e-1000";
>+  begin = positive_underflow.data();
>+  end = begin + positive_underflow.size();
>+  d = -100.0;
>+  result = absl::from_chars(begin, end, d);
>+  EXPECT_EQ(result.ptr, end);
>+  EXPECT_EQ(result.ec, std::errc::result_out_of_range);
>+  EXPECT_FALSE(std::signbit(d));  // positive
>+  EXPECT_LE(d, std::numeric_limits<double>::min());
>+  f = -100.0;
>+  result = absl::from_chars(begin, end, f);
>+  EXPECT_EQ(result.ptr, end);
>+  EXPECT_EQ(result.ec, std::errc::result_out_of_range);
>+  EXPECT_FALSE(std::signbit(f));  // positive
>+  EXPECT_LE(f, std::numeric_limits<float>::min());
>+}
>+
>+TEST(FromChars, Overflow) {
>+  // Check that overflow is handled correctly, according to the specification
>+  // in DR 3081.
>+  double d;
>+  float f;
>+  absl::from_chars_result result;
>+
>+  std::string negative_overflow = "-1e1000";
>+  const char* begin = negative_overflow.data();
>+  const char* end = begin + negative_overflow.size();
>+  d = 100.0;
>+  result = absl::from_chars(begin, end, d);
>+  EXPECT_EQ(result.ptr, end);
>+  EXPECT_EQ(result.ec, std::errc::result_out_of_range);
>+  EXPECT_TRUE(std::signbit(d));  // negative
>+  EXPECT_EQ(d, -std::numeric_limits<double>::max());
>+  f = 100.0;
>+  result = absl::from_chars(begin, end, f);
>+  EXPECT_EQ(result.ptr, end);
>+  EXPECT_EQ(result.ec, std::errc::result_out_of_range);
>+  EXPECT_TRUE(std::signbit(f));  // negative
>+  EXPECT_EQ(f, -std::numeric_limits<float>::max());
>+
>+  std::string positive_overflow = "1e1000";
>+  begin = positive_overflow.data();
>+  end = begin + positive_overflow.size();
>+  d = -100.0;
>+  result = absl::from_chars(begin, end, d);
>+  EXPECT_EQ(result.ptr, end);
>+  EXPECT_EQ(result.ec, std::errc::result_out_of_range);
>+  EXPECT_FALSE(std::signbit(d));  // positive
>+  EXPECT_EQ(d, std::numeric_limits<double>::max());
>+  f = -100.0;
>+  result = absl::from_chars(begin, end, f);
>+  EXPECT_EQ(result.ptr, end);
>+  EXPECT_EQ(result.ec, std::errc::result_out_of_range);
>+  EXPECT_FALSE(std::signbit(f));  // positive
>+  EXPECT_EQ(f, std::numeric_limits<float>::max());
>+}
>+
>+TEST(FromChars, ReturnValuePtr) {
>+  // Check that `ptr` points one past the number scanned, even if that number
>+  // is not representable.
>+  double d;
>+  absl::from_chars_result result;
>+
>+  std::string normal = "3.14@#$%@#$%";
>+  result = absl::from_chars(normal.data(), normal.data() + normal.size(), d);
>+  EXPECT_EQ(result.ec, std::errc());
>+  EXPECT_EQ(result.ptr - normal.data(), 4);
>+
>+  std::string overflow = "1e1000@#$%@#$%";
>+  result = absl::from_chars(overflow.data(),
>+                            overflow.data() + overflow.size(), d);
>+  EXPECT_EQ(result.ec, std::errc::result_out_of_range);
>+  EXPECT_EQ(result.ptr - overflow.data(), 6);
>+
>+  std::string garbage = "#$%@#$%";
>+  result = absl::from_chars(garbage.data(),
>+                            garbage.data() + garbage.size(), d);
>+  EXPECT_EQ(result.ec, std::errc::invalid_argument);
>+  EXPECT_EQ(result.ptr - garbage.data(), 0);
>+}
>+
>+// Check for a wide range of inputs that strtod() and absl::from_chars() exactly
>+// agree on the conversion amount.
>+//
>+// This test assumes the platform's strtod() uses perfect round_to_nearest
>+// rounding.
>+TEST(FromChars, TestVersusStrtod) {
>+  for (int mantissa = 1000000; mantissa <= 9999999; mantissa += 501) {
>+    for (int exponent = -300; exponent < 300; ++exponent) {
>+      std::string candidate = absl::StrCat(mantissa, "e", exponent);
>+      double strtod_value = strtod(candidate.c_str(), nullptr);
>+      double absl_value = 0;
>+      absl::from_chars(candidate.data(), candidate.data() + candidate.size(),
>+                       absl_value);
>+      ASSERT_EQ(strtod_value, absl_value) << candidate;
>+    }
>+  }
>+}
>+
>+// Check for a wide range of inputs that strtof() and absl::from_chars() exactly
>+// agree on the conversion amount.
>+//
>+// This test assumes the platform's strtof() uses perfect round_to_nearest
>+// rounding.
>+TEST(FromChars, TestVersusStrtof) {
>+  for (int mantissa = 1000000; mantissa <= 9999999; mantissa += 501) {
>+    for (int exponent = -43; exponent < 32; ++exponent) {
>+      std::string candidate = absl::StrCat(mantissa, "e", exponent);
>+      float strtod_value = strtof(candidate.c_str(), nullptr);
>+      float absl_value = 0;
>+      absl::from_chars(candidate.data(), candidate.data() + candidate.size(),
>+                       absl_value);
>+      ASSERT_EQ(strtod_value, absl_value) << candidate;
>+    }
>+  }
>+}
>+
>+// Tests if two floating point values have identical bit layouts.  (EXPECT_EQ
>+// is not suitable for NaN testing, since NaNs are never equal.)
>+template <typename Float>
>+bool Identical(Float a, Float b) {
>+  return 0 == memcmp(&a, &b, sizeof(Float));
>+}
>+
>+// Check that NaNs are parsed correctly.  The spec requires that
>+// std::from_chars on "NaN(123abc)" return the same value as std::nan("123abc").
>+// How such an n-char-sequence affects the generated NaN is unspecified, so we
>+// just test for symmetry with std::nan and strtod here.
>+//
>+// (In Linux, this parses the value as a number and stuffs that number into the
>+// free bits of a quiet NaN.)
>+TEST(FromChars, NaNDoubles) {
>+  for (std::string n_char_sequence :
>+       {"", "1", "2", "3", "fff", "FFF", "200000", "400000", "4000000000000",
>+        "8000000000000", "abc123", "legal_but_unexpected",
>+        "99999999999999999999999", "_"}) {
>+    std::string input = absl::StrCat("nan(", n_char_sequence, ")");
>+    SCOPED_TRACE(input);
>+    double from_chars_double;
>+    absl::from_chars(input.data(), input.data() + input.size(),
>+                     from_chars_double);
>+    double std_nan_double = std::nan(n_char_sequence.c_str());
>+    EXPECT_TRUE(Identical(from_chars_double, std_nan_double));
>+
>+    // Also check that we match strtod()'s behavior.  This test assumes that the
>+    // platform has a compliant strtod().
>+#if ABSL_STRTOD_HANDLES_NAN_CORRECTLY
>+    double strtod_double = strtod(input.c_str(), nullptr);
>+    EXPECT_TRUE(Identical(from_chars_double, strtod_double));
>+#endif  // ABSL_STRTOD_HANDLES_NAN_CORRECTLY
>+
>+    // Check that we can parse a negative NaN
>+    std::string negative_input = "-" + input;
>+    double negative_from_chars_double;
>+    absl::from_chars(negative_input.data(),
>+                     negative_input.data() + negative_input.size(),
>+                     negative_from_chars_double);
>+    EXPECT_TRUE(std::signbit(negative_from_chars_double));
>+    EXPECT_FALSE(Identical(negative_from_chars_double, from_chars_double));
>+    from_chars_double = std::copysign(from_chars_double, -1.0);
>+    EXPECT_TRUE(Identical(negative_from_chars_double, from_chars_double));
>+  }
>+}
>+
>+TEST(FromChars, NaNFloats) {
>+  for (std::string n_char_sequence :
>+       {"", "1", "2", "3", "fff", "FFF", "200000", "400000", "4000000000000",
>+        "8000000000000", "abc123", "legal_but_unexpected",
>+        "99999999999999999999999", "_"}) {
>+    std::string input = absl::StrCat("nan(", n_char_sequence, ")");
>+    SCOPED_TRACE(input);
>+    float from_chars_float;
>+    absl::from_chars(input.data(), input.data() + input.size(),
>+                     from_chars_float);
>+    float std_nan_float = std::nanf(n_char_sequence.c_str());
>+    EXPECT_TRUE(Identical(from_chars_float, std_nan_float));
>+
>+    // Also check that we match strtof()'s behavior.  This test assumes that the
>+    // platform has a compliant strtof().
>+#if ABSL_STRTOD_HANDLES_NAN_CORRECTLY
>+    float strtof_float = strtof(input.c_str(), nullptr);
>+    EXPECT_TRUE(Identical(from_chars_float, strtof_float));
>+#endif  // ABSL_STRTOD_HANDLES_NAN_CORRECTLY
>+
>+    // Check that we can parse a negative NaN
>+    std::string negative_input = "-" + input;
>+    float negative_from_chars_float;
>+    absl::from_chars(negative_input.data(),
>+                     negative_input.data() + negative_input.size(),
>+                     negative_from_chars_float);
>+    EXPECT_TRUE(std::signbit(negative_from_chars_float));
>+    EXPECT_FALSE(Identical(negative_from_chars_float, from_chars_float));
>+    from_chars_float = std::copysign(from_chars_float, -1.0);
>+    EXPECT_TRUE(Identical(negative_from_chars_float, from_chars_float));
>+  }
>+}
>+
>+// Returns an integer larger than step.  The values grow exponentially.
>+int NextStep(int step) {
>+  return step + (step >> 2) + 1;
>+}
>+
>+// Test a conversion on a family of input strings, checking that the calculation
>+// is correct for in-bounds values, and that overflow and underflow are done
>+// correctly for out-of-bounds values.
>+//
>+// input_generator maps from an integer index to a std::string to test.
>+// expected_generator maps from an integer index to an expected Float value.
>+// from_chars conversion of input_generator(i) should result in
>+// expected_generator(i).
>+//
>+// lower_bound and upper_bound denote the smallest and largest values for which
>+// the conversion is expected to succeed.
>+template <typename Float>
>+void TestOverflowAndUnderflow(
>+    const std::function<std::string(int)>& input_generator,
>+    const std::function<Float(int)>& expected_generator, int lower_bound,
>+    int upper_bound) {
>+  // test legal values near lower_bound
>+  int index, step;
>+  for (index = lower_bound, step = 1; index < upper_bound;
>+       index += step, step = NextStep(step)) {
>+    std::string input = input_generator(index);
>+    SCOPED_TRACE(input);
>+    Float expected = expected_generator(index);
>+    Float actual;
>+    auto result =
>+        absl::from_chars(input.data(), input.data() + input.size(), actual);
>+    EXPECT_EQ(result.ec, std::errc());
>+    EXPECT_EQ(expected, actual);
>+  }
>+  // test legal values near upper_bound
>+  for (index = upper_bound, step = 1; index > lower_bound;
>+       index -= step, step = NextStep(step)) {
>+    std::string input = input_generator(index);
>+    SCOPED_TRACE(input);
>+    Float expected = expected_generator(index);
>+    Float actual;
>+    auto result =
>+        absl::from_chars(input.data(), input.data() + input.size(), actual);
>+    EXPECT_EQ(result.ec, std::errc());
>+    EXPECT_EQ(expected, actual);
>+  }
>+  // Test underflow values below lower_bound
>+  for (index = lower_bound - 1, step = 1; index > -1000000;
>+       index -= step, step = NextStep(step)) {
>+    std::string input = input_generator(index);
>+    SCOPED_TRACE(input);
>+    Float actual;
>+    auto result =
>+        absl::from_chars(input.data(), input.data() + input.size(), actual);
>+    EXPECT_EQ(result.ec, std::errc::result_out_of_range);
>+    EXPECT_LT(actual, 1.0);  // check for underflow
>+  }
>+  // Test overflow values above upper_bound
>+  for (index = upper_bound + 1, step = 1; index < 1000000;
>+       index += step, step = NextStep(step)) {
>+    std::string input = input_generator(index);
>+    SCOPED_TRACE(input);
>+    Float actual;
>+    auto result =
>+        absl::from_chars(input.data(), input.data() + input.size(), actual);
>+    EXPECT_EQ(result.ec, std::errc::result_out_of_range);
>+    EXPECT_GT(actual, 1.0);  // check for overflow
>+  }
>+}
>+
>+// Check that overflow and underflow are caught correctly for hex doubles.
>+//
>+// The largest representable double is 0x1.fffffffffffffp+1023, and the
>+// smallest representable subnormal is 0x0.0000000000001p-1022, which equals
>+// 0x1p-1074.  Therefore 1023 and -1074 are the limits of acceptable exponents
>+// in this test.
>+TEST(FromChars, HexdecimalDoubleLimits) {
>+  auto input_gen = [](int index) { return absl::StrCat("0x1.0p", index); };
>+  auto expected_gen = [](int index) { return std::ldexp(1.0, index); };
>+  TestOverflowAndUnderflow<double>(input_gen, expected_gen, -1074, 1023);
>+}
>+
>+// Check that overflow and underflow are caught correctly for hex floats.
>+//
>+// The largest representable float is 0x1.fffffep+127, and the smallest
>+// representable subnormal is 0x0.000002p-126, which equals 0x1p-149.
>+// Therefore 127 and -149 are the limits of acceptable exponents in this test.
>+TEST(FromChars, HexdecimalFloatLimits) {
>+  auto input_gen = [](int index) { return absl::StrCat("0x1.0p", index); };
>+  auto expected_gen = [](int index) { return std::ldexp(1.0f, index); };
>+  TestOverflowAndUnderflow<float>(input_gen, expected_gen, -149, 127);
>+}
>+
>+// Check that overflow and underflow are caught correctly for decimal doubles.
>+//
>+// The largest representable double is about 1.8e308, and the smallest
>+// representable subnormal is about 5e-324.  '1e-324' therefore rounds away from
>+// the smallest representable positive value.  -323 and 308 are the limits of
>+// acceptable exponents in this test.
>+TEST(FromChars, DecimalDoubleLimits) {
>+  auto input_gen = [](int index) { return absl::StrCat("1.0e", index); };
>+  auto expected_gen = [](int index) { return std::pow(10.0, index); };
>+  TestOverflowAndUnderflow<double>(input_gen, expected_gen, -323, 308);
>+}
>+
>+// Check that overflow and underflow are caught correctly for decimal floats.
>+//
>+// The largest representable float is about 3.4e38, and the smallest
>+// representable subnormal is about 1.45e-45.  '1e-45' therefore rounds towards
>+// the smallest representable positive value.  -45 and 38 are the limits of
>+// acceptable exponents in this test.
>+TEST(FromChars, DecimalFloatLimits) {
>+  auto input_gen = [](int index) { return absl::StrCat("1.0e", index); };
>+  auto expected_gen = [](int index) { return std::pow(10.0, index); };
>+  TestOverflowAndUnderflow<float>(input_gen, expected_gen, -45, 38);
>+}
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/escaping.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/escaping.cc
>new file mode 100644
>index 00000000000..fbc9f756315
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/escaping.cc
>@@ -0,0 +1,1109 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/strings/escaping.h"
>+
>+#include <algorithm>
>+#include <cassert>
>+#include <cstdint>
>+#include <cstring>
>+#include <iterator>
>+#include <limits>
>+#include <string>
>+
>+#include "absl/base/internal/endian.h"
>+#include "absl/base/internal/raw_logging.h"
>+#include "absl/base/internal/unaligned_access.h"
>+#include "absl/strings/internal/char_map.h"
>+#include "absl/strings/internal/resize_uninitialized.h"
>+#include "absl/strings/internal/utf8.h"
>+#include "absl/strings/str_cat.h"
>+#include "absl/strings/str_join.h"
>+#include "absl/strings/string_view.h"
>+
>+namespace absl {
>+namespace {
>+
>+// Digit conversion.
>+constexpr char kHexChar[] = "0123456789abcdef";
>+
>+constexpr char kHexTable[513] =
>+    "000102030405060708090a0b0c0d0e0f"
>+    "101112131415161718191a1b1c1d1e1f"
>+    "202122232425262728292a2b2c2d2e2f"
>+    "303132333435363738393a3b3c3d3e3f"
>+    "404142434445464748494a4b4c4d4e4f"
>+    "505152535455565758595a5b5c5d5e5f"
>+    "606162636465666768696a6b6c6d6e6f"
>+    "707172737475767778797a7b7c7d7e7f"
>+    "808182838485868788898a8b8c8d8e8f"
>+    "909192939495969798999a9b9c9d9e9f"
>+    "a0a1a2a3a4a5a6a7a8a9aaabacadaeaf"
>+    "b0b1b2b3b4b5b6b7b8b9babbbcbdbebf"
>+    "c0c1c2c3c4c5c6c7c8c9cacbcccdcecf"
>+    "d0d1d2d3d4d5d6d7d8d9dadbdcdddedf"
>+    "e0e1e2e3e4e5e6e7e8e9eaebecedeeef"
>+    "f0f1f2f3f4f5f6f7f8f9fafbfcfdfeff";
>+
>+// These are used for the leave_nulls_escaped argument to CUnescapeInternal().
>+constexpr bool kUnescapeNulls = false;
>+
>+inline bool is_octal_digit(char c) { return ('0' <= c) && (c <= '7'); }
>+
>+inline int hex_digit_to_int(char c) {
>+  static_assert('0' == 0x30 && 'A' == 0x41 && 'a' == 0x61,
>+                "Character set must be ASCII.");
>+  assert(absl::ascii_isxdigit(c));
>+  int x = static_cast<unsigned char>(c);
>+  if (x > '9') {
>+    x += 9;
>+  }
>+  return x & 0xf;
>+}
>+
>+inline bool IsSurrogate(char32_t c, absl::string_view src, std::string* error) {
>+  if (c >= 0xD800 && c <= 0xDFFF) {
>+    if (error) {
>+      *error = absl::StrCat("invalid surrogate character (0xD800-DFFF): \\",
>+                            src);
>+    }
>+    return true;
>+  }
>+  return false;
>+}
>+
>+// ----------------------------------------------------------------------
>+// CUnescapeInternal()
>+//    Implements both CUnescape() and CUnescapeForNullTerminatedString().
>+//
>+//    Unescapes C escape sequences and is the reverse of CEscape().
>+//
>+//    If 'source' is valid, stores the unescaped std::string and its size in
>+//    'dest' and 'dest_len' respectively, and returns true. Otherwise
>+//    returns false and optionally stores the error description in
>+//    'error'. Set 'error' to nullptr to disable error reporting.
>+//
>+//    'dest' should point to a buffer that is at least as big as 'source'.
>+//    'source' and 'dest' may be the same.
>+//
>+//     NOTE: any changes to this function must also be reflected in the older
>+//     UnescapeCEscapeSequences().
>+// ----------------------------------------------------------------------
>+bool CUnescapeInternal(absl::string_view source, bool leave_nulls_escaped,
>+                       char* dest, ptrdiff_t* dest_len, std::string* error) {
>+  char* d = dest;
>+  const char* p = source.data();
>+  const char* end = source.end();
>+  const char* last_byte = end - 1;
>+
>+  // Small optimization for case where source = dest and there's no escaping
>+  while (p == d && p < end && *p != '\\') p++, d++;
>+
>+  while (p < end) {
>+    if (*p != '\\') {
>+      *d++ = *p++;
>+    } else {
>+      if (++p > last_byte) {  // skip past the '\\'
>+        if (error) *error = "String cannot end with \\";
>+        return false;
>+      }
>+      switch (*p) {
>+        case 'a':  *d++ = '\a';  break;
>+        case 'b':  *d++ = '\b';  break;
>+        case 'f':  *d++ = '\f';  break;
>+        case 'n':  *d++ = '\n';  break;
>+        case 'r':  *d++ = '\r';  break;
>+        case 't':  *d++ = '\t';  break;
>+        case 'v':  *d++ = '\v';  break;
>+        case '\\': *d++ = '\\';  break;
>+        case '?':  *d++ = '\?';  break;    // \?  Who knew?
>+        case '\'': *d++ = '\'';  break;
>+        case '"':  *d++ = '\"';  break;
>+        case '0':
>+        case '1':
>+        case '2':
>+        case '3':
>+        case '4':
>+        case '5':
>+        case '6':
>+        case '7': {
>+          // octal digit: 1 to 3 digits
>+          const char* octal_start = p;
>+          unsigned int ch = *p - '0';
>+          if (p < last_byte && is_octal_digit(p[1])) ch = ch * 8 + *++p - '0';
>+          if (p < last_byte && is_octal_digit(p[1]))
>+            ch = ch * 8 + *++p - '0';      // now points at last digit
>+          if (ch > 0xff) {
>+            if (error) {
>+              *error = "Value of \\" +
>+                       std::string(octal_start, p + 1 - octal_start) +
>+                       " exceeds 0xff";
>+            }
>+            return false;
>+          }
>+          if ((ch == 0) && leave_nulls_escaped) {
>+            // Copy the escape sequence for the null character
>+            const ptrdiff_t octal_size = p + 1 - octal_start;
>+            *d++ = '\\';
>+            memcpy(d, octal_start, octal_size);
>+            d += octal_size;
>+            break;
>+          }
>+          *d++ = ch;
>+          break;
>+        }
>+        case 'x':
>+        case 'X': {
>+          if (p >= last_byte) {
>+            if (error) *error = "String cannot end with \\x";
>+            return false;
>+          } else if (!absl::ascii_isxdigit(p[1])) {
>+            if (error) *error = "\\x cannot be followed by a non-hex digit";
>+            return false;
>+          }
>+          unsigned int ch = 0;
>+          const char* hex_start = p;
>+          while (p < last_byte && absl::ascii_isxdigit(p[1]))
>+            // Arbitrarily many hex digits
>+            ch = (ch << 4) + hex_digit_to_int(*++p);
>+          if (ch > 0xFF) {
>+            if (error) {
>+              *error = "Value of \\" + std::string(hex_start, p + 1 - hex_start) +
>+                       " exceeds 0xff";
>+            }
>+            return false;
>+          }
>+          if ((ch == 0) && leave_nulls_escaped) {
>+            // Copy the escape sequence for the null character
>+            const ptrdiff_t hex_size = p + 1 - hex_start;
>+            *d++ = '\\';
>+            memcpy(d, hex_start, hex_size);
>+            d += hex_size;
>+            break;
>+          }
>+          *d++ = ch;
>+          break;
>+        }
>+        case 'u': {
>+          // \uhhhh => convert 4 hex digits to UTF-8
>+          char32_t rune = 0;
>+          const char* hex_start = p;
>+          if (p + 4 >= end) {
>+            if (error) {
>+              *error = "\\u must be followed by 4 hex digits: \\" +
>+                       std::string(hex_start, p + 1 - hex_start);
>+            }
>+            return false;
>+          }
>+          for (int i = 0; i < 4; ++i) {
>+            // Look one char ahead.
>+            if (absl::ascii_isxdigit(p[1])) {
>+              rune = (rune << 4) + hex_digit_to_int(*++p);  // Advance p.
>+            } else {
>+              if (error) {
>+                *error = "\\u must be followed by 4 hex digits: \\" +
>+                         std::string(hex_start, p + 1 - hex_start);
>+              }
>+              return false;
>+            }
>+          }
>+          if ((rune == 0) && leave_nulls_escaped) {
>+            // Copy the escape sequence for the null character
>+            *d++ = '\\';
>+            memcpy(d, hex_start, 5);  // u0000
>+            d += 5;
>+            break;
>+          }
>+          if (IsSurrogate(rune, absl::string_view(hex_start, 5), error)) {
>+            return false;
>+          }
>+          d += strings_internal::EncodeUTF8Char(d, rune);
>+          break;
>+        }
>+        case 'U': {
>+          // \Uhhhhhhhh => convert 8 hex digits to UTF-8
>+          char32_t rune = 0;
>+          const char* hex_start = p;
>+          if (p + 8 >= end) {
>+            if (error) {
>+              *error = "\\U must be followed by 8 hex digits: \\" +
>+                       std::string(hex_start, p + 1 - hex_start);
>+            }
>+            return false;
>+          }
>+          for (int i = 0; i < 8; ++i) {
>+            // Look one char ahead.
>+            if (absl::ascii_isxdigit(p[1])) {
>+              // Don't change rune until we're sure this
>+              // is within the Unicode limit, but do advance p.
>+              uint32_t newrune = (rune << 4) + hex_digit_to_int(*++p);
>+              if (newrune > 0x10FFFF) {
>+                if (error) {
>+                  *error = "Value of \\" +
>+                           std::string(hex_start, p + 1 - hex_start) +
>+                           " exceeds Unicode limit (0x10FFFF)";
>+                }
>+                return false;
>+              } else {
>+                rune = newrune;
>+              }
>+            } else {
>+              if (error) {
>+                *error = "\\U must be followed by 8 hex digits: \\" +
>+                         std::string(hex_start, p + 1 - hex_start);
>+              }
>+              return false;
>+            }
>+          }
>+          if ((rune == 0) && leave_nulls_escaped) {
>+            // Copy the escape sequence for the null character
>+            *d++ = '\\';
>+            memcpy(d, hex_start, 9);  // U00000000
>+            d += 9;
>+            break;
>+          }
>+          if (IsSurrogate(rune, absl::string_view(hex_start, 9), error)) {
>+            return false;
>+          }
>+          d += strings_internal::EncodeUTF8Char(d, rune);
>+          break;
>+        }
>+        default: {
>+          if (error) *error = std::string("Unknown escape sequence: \\") + *p;
>+          return false;
>+        }
>+      }
>+      p++;                                 // read past letter we escaped
>+    }
>+  }
>+  *dest_len = d - dest;
>+  return true;
>+}
>+
>+// ----------------------------------------------------------------------
>+// CUnescapeInternal()
>+//
>+//    Same as above but uses a C++ std::string for output. 'source' and 'dest'
>+//    may be the same.
>+// ----------------------------------------------------------------------
>+bool CUnescapeInternal(absl::string_view source, bool leave_nulls_escaped,
>+                       std::string* dest, std::string* error) {
>+  strings_internal::STLStringResizeUninitialized(dest, source.size());
>+
>+  ptrdiff_t dest_size;
>+  if (!CUnescapeInternal(source,
>+                         leave_nulls_escaped,
>+                         const_cast<char*>(dest->data()),
>+                         &dest_size,
>+                         error)) {
>+    return false;
>+  }
>+  dest->erase(dest_size);
>+  return true;
>+}
>+
>+// ----------------------------------------------------------------------
>+// CEscape()
>+// CHexEscape()
>+// Utf8SafeCEscape()
>+// Utf8SafeCHexEscape()
>+//    Escapes 'src' using C-style escape sequences.  This is useful for
>+//    preparing query flags.  The 'Hex' version uses hexadecimal rather than
>+//    octal sequences.  The 'Utf8Safe' version does not touch UTF-8 bytes.
>+//
>+//    Escaped chars: \n, \r, \t, ", ', \, and !absl::ascii_isprint().
>+// ----------------------------------------------------------------------
>+std::string CEscapeInternal(absl::string_view src, bool use_hex, bool utf8_safe) {
>+  std::string dest;
>+  bool last_hex_escape = false;  // true if last output char was \xNN.
>+
>+  for (unsigned char c : src) {
>+    bool is_hex_escape = false;
>+    switch (c) {
>+      case '\n': dest.append("\\" "n"); break;
>+      case '\r': dest.append("\\" "r"); break;
>+      case '\t': dest.append("\\" "t"); break;
>+      case '\"': dest.append("\\" "\""); break;
>+      case '\'': dest.append("\\" "'"); break;
>+      case '\\': dest.append("\\" "\\"); break;
>+      default:
>+        // Note that if we emit \xNN and the src character after that is a hex
>+        // digit then that digit must be escaped too to prevent it being
>+        // interpreted as part of the character code by C.
>+        if ((!utf8_safe || c < 0x80) &&
>+            (!absl::ascii_isprint(c) ||
>+             (last_hex_escape && absl::ascii_isxdigit(c)))) {
>+          if (use_hex) {
>+            dest.append("\\" "x");
>+            dest.push_back(kHexChar[c / 16]);
>+            dest.push_back(kHexChar[c % 16]);
>+            is_hex_escape = true;
>+          } else {
>+            dest.append("\\");
>+            dest.push_back(kHexChar[c / 64]);
>+            dest.push_back(kHexChar[(c % 64) / 8]);
>+            dest.push_back(kHexChar[c % 8]);
>+          }
>+        } else {
>+          dest.push_back(c);
>+          break;
>+        }
>+    }
>+    last_hex_escape = is_hex_escape;
>+  }
>+
>+  return dest;
>+}
>+
>+/* clang-format off */
>+constexpr char c_escaped_len[256] = {
>+    4, 4, 4, 4, 4, 4, 4, 4, 4, 2, 2, 4, 4, 2, 4, 4,  // \t, \n, \r
>+    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4,
>+    1, 1, 2, 1, 1, 1, 1, 2, 1, 1, 1, 1, 1, 1, 1, 1,  // ", '
>+    1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1,  // '0'..'9'
>+    1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1,  // 'A'..'O'
>+    1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 2, 1, 1, 1,  // 'P'..'Z', '\'
>+    1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1,  // 'a'..'o'
>+    1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 4,  // 'p'..'z', DEL
>+    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4,
>+    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4,
>+    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4,
>+    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4,
>+    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4,
>+    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4,
>+    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4,
>+    4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4,
>+};
>+/* clang-format on */
>+
>+// Calculates the length of the C-style escaped version of 'src'.
>+// Assumes that non-printable characters are escaped using octal sequences, and
>+// that UTF-8 bytes are not handled specially.
>+inline size_t CEscapedLength(absl::string_view src) {
>+  size_t escaped_len = 0;
>+  for (unsigned char c : src) escaped_len += c_escaped_len[c];
>+  return escaped_len;
>+}
>+
>+void CEscapeAndAppendInternal(absl::string_view src, std::string* dest) {
>+  size_t escaped_len = CEscapedLength(src);
>+  if (escaped_len == src.size()) {
>+    dest->append(src.data(), src.size());
>+    return;
>+  }
>+
>+  size_t cur_dest_len = dest->size();
>+  strings_internal::STLStringResizeUninitialized(dest,
>+                                                 cur_dest_len + escaped_len);
>+  char* append_ptr = &(*dest)[cur_dest_len];
>+
>+  for (unsigned char c : src) {
>+    int char_len = c_escaped_len[c];
>+    if (char_len == 1) {
>+      *append_ptr++ = c;
>+    } else if (char_len == 2) {
>+      switch (c) {
>+        case '\n':
>+          *append_ptr++ = '\\';
>+          *append_ptr++ = 'n';
>+          break;
>+        case '\r':
>+          *append_ptr++ = '\\';
>+          *append_ptr++ = 'r';
>+          break;
>+        case '\t':
>+          *append_ptr++ = '\\';
>+          *append_ptr++ = 't';
>+          break;
>+        case '\"':
>+          *append_ptr++ = '\\';
>+          *append_ptr++ = '\"';
>+          break;
>+        case '\'':
>+          *append_ptr++ = '\\';
>+          *append_ptr++ = '\'';
>+          break;
>+        case '\\':
>+          *append_ptr++ = '\\';
>+          *append_ptr++ = '\\';
>+          break;
>+      }
>+    } else {
>+      *append_ptr++ = '\\';
>+      *append_ptr++ = '0' + c / 64;
>+      *append_ptr++ = '0' + (c % 64) / 8;
>+      *append_ptr++ = '0' + c % 8;
>+    }
>+  }
>+}
>+
>+bool Base64UnescapeInternal(const char* src_param, size_t szsrc, char* dest,
>+                            size_t szdest, const signed char* unbase64,
>+                            size_t* len) {
>+  static const char kPad64Equals = '=';
>+  static const char kPad64Dot = '.';
>+
>+  size_t destidx = 0;
>+  int decode = 0;
>+  int state = 0;
>+  unsigned int ch = 0;
>+  unsigned int temp = 0;
>+
>+  // If "char" is signed by default, using *src as an array index results in
>+  // accessing negative array elements. Treat the input as a pointer to
>+  // unsigned char to avoid this.
>+  const unsigned char* src = reinterpret_cast<const unsigned char*>(src_param);
>+
>+  // The GET_INPUT macro gets the next input character, skipping
>+  // over any whitespace, and stopping when we reach the end of the
>+  // std::string or when we read any non-data character.  The arguments are
>+  // an arbitrary identifier (used as a label for goto) and the number
>+  // of data bytes that must remain in the input to avoid aborting the
>+  // loop.
>+#define GET_INPUT(label, remain)                                \
>+  label:                                                        \
>+  --szsrc;                                                      \
>+  ch = *src++;                                                  \
>+  decode = unbase64[ch];                                        \
>+  if (decode < 0) {                                             \
>+    if (absl::ascii_isspace(ch) && szsrc >= remain) goto label; \
>+    state = 4 - remain;                                         \
>+    break;                                                      \
>+  }
>+
>+  // if dest is null, we're just checking to see if it's legal input
>+  // rather than producing output.  (I suspect this could just be done
>+  // with a regexp...).  We duplicate the loop so this test can be
>+  // outside it instead of in every iteration.
>+
>+  if (dest) {
>+    // This loop consumes 4 input bytes and produces 3 output bytes
>+    // per iteration.  We can't know at the start that there is enough
>+    // data left in the std::string for a full iteration, so the loop may
>+    // break out in the middle; if so 'state' will be set to the
>+    // number of input bytes read.
>+
>+    while (szsrc >= 4) {
>+      // We'll start by optimistically assuming that the next four
>+      // bytes of the std::string (src[0..3]) are four good data bytes
>+      // (that is, no nulls, whitespace, padding chars, or illegal
>+      // chars).  We need to test src[0..2] for nulls individually
>+      // before constructing temp to preserve the property that we
>+      // never read past a null in the std::string (no matter how long
>+      // szsrc claims the std::string is).
>+
>+      if (!src[0] || !src[1] || !src[2] ||
>+          ((temp = ((unsigned(unbase64[src[0]]) << 18) |
>+                    (unsigned(unbase64[src[1]]) << 12) |
>+                    (unsigned(unbase64[src[2]]) << 6) |
>+                    (unsigned(unbase64[src[3]])))) &
>+           0x80000000)) {
>+        // Iff any of those four characters was bad (null, illegal,
>+        // whitespace, padding), then temp's high bit will be set
>+        // (because unbase64[] is -1 for all bad characters).
>+        //
>+        // We'll back up and resort to the slower decoder, which knows
>+        // how to handle those cases.
>+
>+        GET_INPUT(first, 4);
>+        temp = decode;
>+        GET_INPUT(second, 3);
>+        temp = (temp << 6) | decode;
>+        GET_INPUT(third, 2);
>+        temp = (temp << 6) | decode;
>+        GET_INPUT(fourth, 1);
>+        temp = (temp << 6) | decode;
>+      } else {
>+        // We really did have four good data bytes, so advance four
>+        // characters in the std::string.
>+
>+        szsrc -= 4;
>+        src += 4;
>+      }
>+
>+      // temp has 24 bits of input, so write that out as three bytes.
>+
>+      if (destidx + 3 > szdest) return false;
>+      dest[destidx + 2] = temp;
>+      temp >>= 8;
>+      dest[destidx + 1] = temp;
>+      temp >>= 8;
>+      dest[destidx] = temp;
>+      destidx += 3;
>+    }
>+  } else {
>+    while (szsrc >= 4) {
>+      if (!src[0] || !src[1] || !src[2] ||
>+          ((temp = ((unsigned(unbase64[src[0]]) << 18) |
>+                    (unsigned(unbase64[src[1]]) << 12) |
>+                    (unsigned(unbase64[src[2]]) << 6) |
>+                    (unsigned(unbase64[src[3]])))) &
>+           0x80000000)) {
>+        GET_INPUT(first_no_dest, 4);
>+        GET_INPUT(second_no_dest, 3);
>+        GET_INPUT(third_no_dest, 2);
>+        GET_INPUT(fourth_no_dest, 1);
>+      } else {
>+        szsrc -= 4;
>+        src += 4;
>+      }
>+      destidx += 3;
>+    }
>+  }
>+
>+#undef GET_INPUT
>+
>+  // if the loop terminated because we read a bad character, return
>+  // now.
>+  if (decode < 0 && ch != kPad64Equals && ch != kPad64Dot &&
>+      !absl::ascii_isspace(ch))
>+    return false;
>+
>+  if (ch == kPad64Equals || ch == kPad64Dot) {
>+    // if we stopped by hitting an '=' or '.', un-read that character -- we'll
>+    // look at it again when we count to check for the proper number of
>+    // equals signs at the end.
>+    ++szsrc;
>+    --src;
>+  } else {
>+    // This loop consumes 1 input byte per iteration.  It's used to
>+    // clean up the 0-3 input bytes remaining when the first, faster
>+    // loop finishes.  'temp' contains the data from 'state' input
>+    // characters read by the first loop.
>+    while (szsrc > 0) {
>+      --szsrc;
>+      ch = *src++;
>+      decode = unbase64[ch];
>+      if (decode < 0) {
>+        if (absl::ascii_isspace(ch)) {
>+          continue;
>+        } else if (ch == kPad64Equals || ch == kPad64Dot) {
>+          // back up one character; we'll read it again when we check
>+          // for the correct number of pad characters at the end.
>+          ++szsrc;
>+          --src;
>+          break;
>+        } else {
>+          return false;
>+        }
>+      }
>+
>+      // Each input character gives us six bits of output.
>+      temp = (temp << 6) | decode;
>+      ++state;
>+      if (state == 4) {
>+        // If we've accumulated 24 bits of output, write that out as
>+        // three bytes.
>+        if (dest) {
>+          if (destidx + 3 > szdest) return false;
>+          dest[destidx + 2] = temp;
>+          temp >>= 8;
>+          dest[destidx + 1] = temp;
>+          temp >>= 8;
>+          dest[destidx] = temp;
>+        }
>+        destidx += 3;
>+        state = 0;
>+        temp = 0;
>+      }
>+    }
>+  }
>+
>+  // Process the leftover data contained in 'temp' at the end of the input.
>+  int expected_equals = 0;
>+  switch (state) {
>+    case 0:
>+      // Nothing left over; output is a multiple of 3 bytes.
>+      break;
>+
>+    case 1:
>+      // Bad input; we have 6 bits left over.
>+      return false;
>+
>+    case 2:
>+      // Produce one more output byte from the 12 input bits we have left.
>+      if (dest) {
>+        if (destidx + 1 > szdest) return false;
>+        temp >>= 4;
>+        dest[destidx] = temp;
>+      }
>+      ++destidx;
>+      expected_equals = 2;
>+      break;
>+
>+    case 3:
>+      // Produce two more output bytes from the 18 input bits we have left.
>+      if (dest) {
>+        if (destidx + 2 > szdest) return false;
>+        temp >>= 2;
>+        dest[destidx + 1] = temp;
>+        temp >>= 8;
>+        dest[destidx] = temp;
>+      }
>+      destidx += 2;
>+      expected_equals = 1;
>+      break;
>+
>+    default:
>+      // state should have no other values at this point.
>+      ABSL_RAW_LOG(FATAL, "This can't happen; base64 decoder state = %d",
>+                   state);
>+  }
>+
>+  // The remainder of the std::string should be all whitespace, mixed with
>+  // exactly 0 equals signs, or exactly 'expected_equals' equals
>+  // signs.  (Always accepting 0 equals signs is an Abseil extension
>+  // not covered in the RFC, as is accepting dot as the pad character.)
>+
>+  int equals = 0;
>+  while (szsrc > 0) {
>+    if (*src == kPad64Equals || *src == kPad64Dot)
>+      ++equals;
>+    else if (!absl::ascii_isspace(*src))
>+      return false;
>+    --szsrc;
>+    ++src;
>+  }
>+
>+  const bool ok = (equals == 0 || equals == expected_equals);
>+  if (ok) *len = destidx;
>+  return ok;
>+}
>+
>+// The arrays below were generated by the following code
>+// #include <sys/time.h>
>+// #include <stdlib.h>
>+// #include <std::string.h>
>+// main()
>+// {
>+//   static const char Base64[] =
>+//     "ABCDEFGHIJKLMNOPQRSTUVWXYZabcdefghijklmnopqrstuvwxyz0123456789+/";
>+//   char* pos;
>+//   int idx, i, j;
>+//   printf("    ");
>+//   for (i = 0; i < 255; i += 8) {
>+//     for (j = i; j < i + 8; j++) {
>+//       pos = strchr(Base64, j);
>+//       if ((pos == nullptr) || (j == 0))
>+//         idx = -1;
>+//       else
>+//         idx = pos - Base64;
>+//       if (idx == -1)
>+//         printf(" %2d,     ", idx);
>+//       else
>+//         printf(" %2d/*%c*/,", idx, j);
>+//     }
>+//     printf("\n    ");
>+//   }
>+// }
>+//
>+// where the value of "Base64[]" was replaced by one of the base-64 conversion
>+// tables from the functions below.
>+/* clang-format off */
>+constexpr signed char kUnBase64[] = {
>+    -1,      -1,      -1,      -1,      -1,      -1,      -1,      -1,
>+    -1,      -1,      -1,      -1,      -1,      -1,      -1,      -1,
>+    -1,      -1,      -1,      -1,      -1,      -1,      -1,      -1,
>+    -1,      -1,      -1,      -1,      -1,      -1,      -1,      -1,
>+    -1,      -1,      -1,      -1,      -1,      -1,      -1,      -1,
>+    -1,      -1,      -1,      62/*+*/, -1,      -1,      -1,      63/*/ */,
>+    52/*0*/, 53/*1*/, 54/*2*/, 55/*3*/, 56/*4*/, 57/*5*/, 58/*6*/, 59/*7*/,
>+    60/*8*/, 61/*9*/, -1,      -1,      -1,      -1,      -1,      -1,
>+    -1,       0/*A*/,  1/*B*/,  2/*C*/,  3/*D*/,  4/*E*/,  5/*F*/,  6/*G*/,
>+    07/*H*/,  8/*I*/,  9/*J*/, 10/*K*/, 11/*L*/, 12/*M*/, 13/*N*/, 14/*O*/,
>+    15/*P*/, 16/*Q*/, 17/*R*/, 18/*S*/, 19/*T*/, 20/*U*/, 21/*V*/, 22/*W*/,
>+    23/*X*/, 24/*Y*/, 25/*Z*/, -1,      -1,      -1,      -1,      -1,
>+    -1,      26/*a*/, 27/*b*/, 28/*c*/, 29/*d*/, 30/*e*/, 31/*f*/, 32/*g*/,
>+    33/*h*/, 34/*i*/, 35/*j*/, 36/*k*/, 37/*l*/, 38/*m*/, 39/*n*/, 40/*o*/,
>+    41/*p*/, 42/*q*/, 43/*r*/, 44/*s*/, 45/*t*/, 46/*u*/, 47/*v*/, 48/*w*/,
>+    49/*x*/, 50/*y*/, 51/*z*/, -1,      -1,      -1,      -1,      -1,
>+    -1,      -1,      -1,      -1,      -1,      -1,      -1,      -1,
>+    -1,      -1,      -1,      -1,      -1,      -1,      -1,      -1,
>+    -1,      -1,      -1,      -1,      -1,      -1,      -1,      -1,
>+    -1,      -1,      -1,      -1,      -1,      -1,      -1,      -1,
>+    -1,      -1,      -1,      -1,      -1,      -1,      -1,      -1,
>+    -1,      -1,      -1,      -1,      -1,      -1,      -1,      -1,
>+    -1,      -1,      -1,      -1,      -1,      -1,      -1,      -1,
>+    -1,      -1,      -1,      -1,      -1,      -1,      -1,      -1,
>+    -1,      -1,      -1,      -1,      -1,      -1,      -1,      -1,
>+    -1,      -1,      -1,      -1,      -1,      -1,      -1,      -1,
>+    -1,      -1,      -1,      -1,      -1,      -1,      -1,      -1,
>+    -1,      -1,      -1,      -1,      -1,      -1,      -1,      -1,
>+    -1,      -1,      -1,      -1,      -1,      -1,      -1,      -1,
>+    -1,      -1,      -1,      -1,      -1,      -1,      -1,      -1,
>+    -1,      -1,      -1,      -1,      -1,      -1,      -1,      -1,
>+    -1,      -1,      -1,      -1,      -1,      -1,      -1,      -1
>+};
>+
>+constexpr signed char kUnWebSafeBase64[] = {
>+    -1,      -1,      -1,      -1,      -1,      -1,      -1,      -1,
>+    -1,      -1,      -1,      -1,      -1,      -1,      -1,      -1,
>+    -1,      -1,      -1,      -1,      -1,      -1,      -1,      -1,
>+    -1,      -1,      -1,      -1,      -1,      -1,      -1,      -1,
>+    -1,      -1,      -1,      -1,      -1,      -1,      -1,      -1,
>+    -1,      -1,      -1,      -1,      -1,      62/*-*/, -1,      -1,
>+    52/*0*/, 53/*1*/, 54/*2*/, 55/*3*/, 56/*4*/, 57/*5*/, 58/*6*/, 59/*7*/,
>+    60/*8*/, 61/*9*/, -1,      -1,      -1,      -1,      -1,      -1,
>+    -1,       0/*A*/,  1/*B*/,  2/*C*/,  3/*D*/,  4/*E*/,  5/*F*/,  6/*G*/,
>+    07/*H*/,  8/*I*/,  9/*J*/, 10/*K*/, 11/*L*/, 12/*M*/, 13/*N*/, 14/*O*/,
>+    15/*P*/, 16/*Q*/, 17/*R*/, 18/*S*/, 19/*T*/, 20/*U*/, 21/*V*/, 22/*W*/,
>+    23/*X*/, 24/*Y*/, 25/*Z*/, -1,      -1,      -1,      -1,      63/*_*/,
>+    -1,      26/*a*/, 27/*b*/, 28/*c*/, 29/*d*/, 30/*e*/, 31/*f*/, 32/*g*/,
>+    33/*h*/, 34/*i*/, 35/*j*/, 36/*k*/, 37/*l*/, 38/*m*/, 39/*n*/, 40/*o*/,
>+    41/*p*/, 42/*q*/, 43/*r*/, 44/*s*/, 45/*t*/, 46/*u*/, 47/*v*/, 48/*w*/,
>+    49/*x*/, 50/*y*/, 51/*z*/, -1,      -1,      -1,      -1,      -1,
>+    -1,      -1,      -1,      -1,      -1,      -1,      -1,      -1,
>+    -1,      -1,      -1,      -1,      -1,      -1,      -1,      -1,
>+    -1,      -1,      -1,      -1,      -1,      -1,      -1,      -1,
>+    -1,      -1,      -1,      -1,      -1,      -1,      -1,      -1,
>+    -1,      -1,      -1,      -1,      -1,      -1,      -1,      -1,
>+    -1,      -1,      -1,      -1,      -1,      -1,      -1,      -1,
>+    -1,      -1,      -1,      -1,      -1,      -1,      -1,      -1,
>+    -1,      -1,      -1,      -1,      -1,      -1,      -1,      -1,
>+    -1,      -1,      -1,      -1,      -1,      -1,      -1,      -1,
>+    -1,      -1,      -1,      -1,      -1,      -1,      -1,      -1,
>+    -1,      -1,      -1,      -1,      -1,      -1,      -1,      -1,
>+    -1,      -1,      -1,      -1,      -1,      -1,      -1,      -1,
>+    -1,      -1,      -1,      -1,      -1,      -1,      -1,      -1,
>+    -1,      -1,      -1,      -1,      -1,      -1,      -1,      -1,
>+    -1,      -1,      -1,      -1,      -1,      -1,      -1,      -1,
>+    -1,      -1,      -1,      -1,      -1,      -1,      -1,      -1
>+};
>+/* clang-format on */
>+
>+size_t CalculateBase64EscapedLenInternal(size_t input_len, bool do_padding) {
>+  // Base64 encodes three bytes of input at a time. If the input is not
>+  // divisible by three, we pad as appropriate.
>+  //
>+  // (from http://tools.ietf.org/html/rfc3548)
>+  // Special processing is performed if fewer than 24 bits are available
>+  // at the end of the data being encoded.  A full encoding quantum is
>+  // always completed at the end of a quantity.  When fewer than 24 input
>+  // bits are available in an input group, zero bits are added (on the
>+  // right) to form an integral number of 6-bit groups.  Padding at the
>+  // end of the data is performed using the '=' character.  Since all base
>+  // 64 input is an integral number of octets, only the following cases
>+  // can arise:
>+
>+  // Base64 encodes each three bytes of input into four bytes of output.
>+  size_t len = (input_len / 3) * 4;
>+
>+  if (input_len % 3 == 0) {
>+    // (from http://tools.ietf.org/html/rfc3548)
>+    // (1) the final quantum of encoding input is an integral multiple of 24
>+    // bits; here, the final unit of encoded output will be an integral
>+    // multiple of 4 characters with no "=" padding,
>+  } else if (input_len % 3 == 1) {
>+    // (from http://tools.ietf.org/html/rfc3548)
>+    // (2) the final quantum of encoding input is exactly 8 bits; here, the
>+    // final unit of encoded output will be two characters followed by two
>+    // "=" padding characters, or
>+    len += 2;
>+    if (do_padding) {
>+      len += 2;
>+    }
>+  } else {  // (input_len % 3 == 2)
>+    // (from http://tools.ietf.org/html/rfc3548)
>+    // (3) the final quantum of encoding input is exactly 16 bits; here, the
>+    // final unit of encoded output will be three characters followed by one
>+    // "=" padding character.
>+    len += 3;
>+    if (do_padding) {
>+      len += 1;
>+    }
>+  }
>+
>+  assert(len >= input_len);  // make sure we didn't overflow
>+  return len;
>+}
>+
>+size_t Base64EscapeInternal(const unsigned char* src, size_t szsrc, char* dest,
>+                            size_t szdest, const char* base64,
>+                            bool do_padding) {
>+  static const char kPad64 = '=';
>+
>+  if (szsrc * 4 > szdest * 3) return 0;
>+
>+  char* cur_dest = dest;
>+  const unsigned char* cur_src = src;
>+
>+  char* const limit_dest = dest + szdest;
>+  const unsigned char* const limit_src = src + szsrc;
>+
>+  // Three bytes of data encodes to four characters of cyphertext.
>+  // So we can pump through three-byte chunks atomically.
>+  if (szsrc >= 3) {  // "limit_src - 3" is UB if szsrc < 3
>+    while (cur_src < limit_src - 3) {  // as long as we have >= 32 bits
>+      uint32_t in = absl::big_endian::Load32(cur_src) >> 8;
>+
>+      cur_dest[0] = base64[in >> 18];
>+      in &= 0x3FFFF;
>+      cur_dest[1] = base64[in >> 12];
>+      in &= 0xFFF;
>+      cur_dest[2] = base64[in >> 6];
>+      in &= 0x3F;
>+      cur_dest[3] = base64[in];
>+
>+      cur_dest += 4;
>+      cur_src += 3;
>+    }
>+  }
>+  // To save time, we didn't update szdest or szsrc in the loop.  So do it now.
>+  szdest = limit_dest - cur_dest;
>+  szsrc = limit_src - cur_src;
>+
>+  /* now deal with the tail (<=3 bytes) */
>+  switch (szsrc) {
>+    case 0:
>+      // Nothing left; nothing more to do.
>+      break;
>+    case 1: {
>+      // One byte left: this encodes to two characters, and (optionally)
>+      // two pad characters to round out the four-character cypherblock.
>+      if (szdest < 2) return 0;
>+      uint32_t in = cur_src[0];
>+      cur_dest[0] = base64[in >> 2];
>+      in &= 0x3;
>+      cur_dest[1] = base64[in << 4];
>+      cur_dest += 2;
>+      szdest -= 2;
>+      if (do_padding) {
>+        if (szdest < 2) return 0;
>+        cur_dest[0] = kPad64;
>+        cur_dest[1] = kPad64;
>+        cur_dest += 2;
>+        szdest -= 2;
>+      }
>+      break;
>+    }
>+    case 2: {
>+      // Two bytes left: this encodes to three characters, and (optionally)
>+      // one pad character to round out the four-character cypherblock.
>+      if (szdest < 3) return 0;
>+      uint32_t in = absl::big_endian::Load16(cur_src);
>+      cur_dest[0] = base64[in >> 10];
>+      in &= 0x3FF;
>+      cur_dest[1] = base64[in >> 4];
>+      in &= 0x00F;
>+      cur_dest[2] = base64[in << 2];
>+      cur_dest += 3;
>+      szdest -= 3;
>+      if (do_padding) {
>+        if (szdest < 1) return 0;
>+        cur_dest[0] = kPad64;
>+        cur_dest += 1;
>+        szdest -= 1;
>+      }
>+      break;
>+    }
>+    case 3: {
>+      // Three bytes left: same as in the big loop above.  We can't do this in
>+      // the loop because the loop above always reads 4 bytes, and the fourth
>+      // byte is past the end of the input.
>+      if (szdest < 4) return 0;
>+      uint32_t in = (cur_src[0] << 16) + absl::big_endian::Load16(cur_src + 1);
>+      cur_dest[0] = base64[in >> 18];
>+      in &= 0x3FFFF;
>+      cur_dest[1] = base64[in >> 12];
>+      in &= 0xFFF;
>+      cur_dest[2] = base64[in >> 6];
>+      in &= 0x3F;
>+      cur_dest[3] = base64[in];
>+      cur_dest += 4;
>+      szdest -= 4;
>+      break;
>+    }
>+    default:
>+      // Should not be reached: blocks of 4 bytes are handled
>+      // in the while loop before this switch statement.
>+      ABSL_RAW_LOG(FATAL, "Logic problem? szsrc = %zu", szsrc);
>+      break;
>+  }
>+  return (cur_dest - dest);
>+}
>+
>+constexpr char kBase64Chars[] =
>+    "ABCDEFGHIJKLMNOPQRSTUVWXYZabcdefghijklmnopqrstuvwxyz0123456789+/";
>+
>+constexpr char kWebSafeBase64Chars[] =
>+    "ABCDEFGHIJKLMNOPQRSTUVWXYZabcdefghijklmnopqrstuvwxyz0123456789-_";
>+
>+void Base64EscapeInternal(const unsigned char* src, size_t szsrc, std::string* dest,
>+                          bool do_padding, const char* base64_chars) {
>+  const size_t calc_escaped_size =
>+      CalculateBase64EscapedLenInternal(szsrc, do_padding);
>+  strings_internal::STLStringResizeUninitialized(dest, calc_escaped_size);
>+
>+  const size_t escaped_len = Base64EscapeInternal(
>+      src, szsrc, &(*dest)[0], dest->size(), base64_chars, do_padding);
>+  assert(calc_escaped_size == escaped_len);
>+  dest->erase(escaped_len);
>+}
>+
>+bool Base64UnescapeInternal(const char* src, size_t slen, std::string* dest,
>+                            const signed char* unbase64) {
>+  // Determine the size of the output std::string.  Base64 encodes every 3 bytes into
>+  // 4 characters.  any leftover chars are added directly for good measure.
>+  // This is documented in the base64 RFC: http://tools.ietf.org/html/rfc3548
>+  const size_t dest_len = 3 * (slen / 4) + (slen % 4);
>+
>+  strings_internal::STLStringResizeUninitialized(dest, dest_len);
>+
>+  // We are getting the destination buffer by getting the beginning of the
>+  // std::string and converting it into a char *.
>+  size_t len;
>+  const bool ok =
>+      Base64UnescapeInternal(src, slen, &(*dest)[0], dest_len, unbase64, &len);
>+  if (!ok) {
>+    dest->clear();
>+    return false;
>+  }
>+
>+  // could be shorter if there was padding
>+  assert(len <= dest_len);
>+  dest->erase(len);
>+
>+  return true;
>+}
>+
>+/* clang-format off */
>+constexpr char kHexValue[256] = {
>+    0,  0,  0,  0,  0,  0,  0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
>+    0,  0,  0,  0,  0,  0,  0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
>+    0,  0,  0,  0,  0,  0,  0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
>+    0,  1,  2,  3,  4,  5,  6, 7, 8, 9, 0, 0, 0, 0, 0, 0,  // '0'..'9'
>+    0, 10, 11, 12, 13, 14, 15, 0, 0, 0, 0, 0, 0, 0, 0, 0,  // 'A'..'F'
>+    0,  0,  0,  0,  0,  0,  0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
>+    0, 10, 11, 12, 13, 14, 15, 0, 0, 0, 0, 0, 0, 0, 0, 0,  // 'a'..'f'
>+    0,  0,  0,  0,  0,  0,  0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
>+    0,  0,  0,  0,  0,  0,  0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
>+    0,  0,  0,  0,  0,  0,  0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
>+    0,  0,  0,  0,  0,  0,  0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
>+    0,  0,  0,  0,  0,  0,  0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
>+    0,  0,  0,  0,  0,  0,  0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
>+    0,  0,  0,  0,  0,  0,  0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
>+    0,  0,  0,  0,  0,  0,  0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
>+    0,  0,  0,  0,  0,  0,  0, 0, 0, 0, 0, 0, 0, 0, 0, 0
>+};
>+/* clang-format on */
>+
>+// This is a templated function so that T can be either a char*
>+// or a std::string.  This works because we use the [] operator to access
>+// individual characters at a time.
>+template <typename T>
>+void HexStringToBytesInternal(const char* from, T to, ptrdiff_t num) {
>+  for (int i = 0; i < num; i++) {
>+    to[i] = (kHexValue[from[i * 2] & 0xFF] << 4) +
>+            (kHexValue[from[i * 2 + 1] & 0xFF]);
>+  }
>+}
>+
>+// This is a templated function so that T can be either a char* or a std::string.
>+template <typename T>
>+void BytesToHexStringInternal(const unsigned char* src, T dest, ptrdiff_t num) {
>+  auto dest_ptr = &dest[0];
>+  for (auto src_ptr = src; src_ptr != (src + num); ++src_ptr, dest_ptr += 2) {
>+    const char* hex_p = &kHexTable[*src_ptr * 2];
>+    std::copy(hex_p, hex_p + 2, dest_ptr);
>+  }
>+}
>+
>+}  // namespace
>+
>+// ----------------------------------------------------------------------
>+// CUnescape()
>+//
>+// See CUnescapeInternal() for implementation details.
>+// ----------------------------------------------------------------------
>+bool CUnescape(absl::string_view source, std::string* dest, std::string* error) {
>+  return CUnescapeInternal(source, kUnescapeNulls, dest, error);
>+}
>+
>+std::string CEscape(absl::string_view src) {
>+  std::string dest;
>+  CEscapeAndAppendInternal(src, &dest);
>+  return dest;
>+}
>+
>+std::string CHexEscape(absl::string_view src) {
>+  return CEscapeInternal(src, true, false);
>+}
>+
>+std::string Utf8SafeCEscape(absl::string_view src) {
>+  return CEscapeInternal(src, false, true);
>+}
>+
>+std::string Utf8SafeCHexEscape(absl::string_view src) {
>+  return CEscapeInternal(src, true, true);
>+}
>+
>+// ----------------------------------------------------------------------
>+// ptrdiff_t Base64Unescape() - base64 decoder
>+// ptrdiff_t Base64Escape() - base64 encoder
>+// ptrdiff_t WebSafeBase64Unescape() - Google's variation of base64 decoder
>+// ptrdiff_t WebSafeBase64Escape() - Google's variation of base64 encoder
>+//
>+// Check out
>+// http://tools.ietf.org/html/rfc2045 for formal description, but what we
>+// care about is that...
>+//   Take the encoded stuff in groups of 4 characters and turn each
>+//   character into a code 0 to 63 thus:
>+//           A-Z map to 0 to 25
>+//           a-z map to 26 to 51
>+//           0-9 map to 52 to 61
>+//           +(- for WebSafe) maps to 62
>+//           /(_ for WebSafe) maps to 63
>+//   There will be four numbers, all less than 64 which can be represented
>+//   by a 6 digit binary number (aaaaaa, bbbbbb, cccccc, dddddd respectively).
>+//   Arrange the 6 digit binary numbers into three bytes as such:
>+//   aaaaaabb bbbbcccc ccdddddd
>+//   Equals signs (one or two) are used at the end of the encoded block to
>+//   indicate that the text was not an integer multiple of three bytes long.
>+// ----------------------------------------------------------------------
>+
>+bool Base64Unescape(absl::string_view src, std::string* dest) {
>+  return Base64UnescapeInternal(src.data(), src.size(), dest, kUnBase64);
>+}
>+
>+bool WebSafeBase64Unescape(absl::string_view src, std::string* dest) {
>+  return Base64UnescapeInternal(src.data(), src.size(), dest, kUnWebSafeBase64);
>+}
>+
>+void Base64Escape(absl::string_view src, std::string* dest) {
>+  Base64EscapeInternal(reinterpret_cast<const unsigned char*>(src.data()),
>+                       src.size(), dest, true, kBase64Chars);
>+}
>+
>+void WebSafeBase64Escape(absl::string_view src, std::string* dest) {
>+  Base64EscapeInternal(reinterpret_cast<const unsigned char*>(src.data()),
>+                       src.size(), dest, false, kWebSafeBase64Chars);
>+}
>+
>+std::string HexStringToBytes(absl::string_view from) {
>+  std::string result;
>+  const auto num = from.size() / 2;
>+  strings_internal::STLStringResizeUninitialized(&result, num);
>+  absl::HexStringToBytesInternal<std::string&>(from.data(), result, num);
>+  return result;
>+}
>+
>+std::string BytesToHexString(absl::string_view from) {
>+  std::string result;
>+  strings_internal::STLStringResizeUninitialized(&result, 2 * from.size());
>+  absl::BytesToHexStringInternal<std::string&>(
>+      reinterpret_cast<const unsigned char*>(from.data()), result, from.size());
>+  return result;
>+}
>+
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/escaping.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/escaping.h
>new file mode 100644
>index 00000000000..7f1ab96d8ff
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/escaping.h
>@@ -0,0 +1,161 @@
>+//
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// -----------------------------------------------------------------------------
>+// File: escaping.h
>+// -----------------------------------------------------------------------------
>+//
>+// This header file contains std::string utilities involved in escaping and
>+// unescaping strings in various ways.
>+//
>+
>+#ifndef ABSL_STRINGS_ESCAPING_H_
>+#define ABSL_STRINGS_ESCAPING_H_
>+
>+#include <cstddef>
>+#include <string>
>+#include <vector>
>+
>+#include "absl/base/macros.h"
>+#include "absl/strings/ascii.h"
>+#include "absl/strings/str_join.h"
>+#include "absl/strings/string_view.h"
>+
>+namespace absl {
>+
>+// CUnescape()
>+//
>+// Unescapes a `source` std::string and copies it into `dest`, rewriting C-style
>+// escape sequences (http://en.cppreference.com/w/cpp/language/escape) into
>+// their proper code point equivalents, returning `true` if successful.
>+//
>+// The following unescape sequences can be handled:
>+//
>+//   * ASCII escape sequences ('\n','\r','\\', etc.) to their ASCII equivalents
>+//   * Octal escape sequences ('\nnn') to byte nnn. The unescaped value must
>+//     resolve to a single byte or an error will occur. E.g. values greater than
>+//     0xff will produce an error.
>+//   * Hexadecimal escape sequences ('\xnn') to byte nn. While an arbitrary
>+//     number of following digits are allowed, the unescaped value must resolve
>+//     to a single byte or an error will occur. E.g. '\x0045' is equivalent to
>+//     '\x45', but '\x1234' will produce an error.
>+//   * Unicode escape sequences ('\unnnn' for exactly four hex digits or
>+//     '\Unnnnnnnn' for exactly eight hex digits, which will be encoded in
>+//     UTF-8. (E.g., `\u2019` unescapes to the three bytes 0xE2, 0x80, and
>+//     0x99).
>+//
>+//
>+// If any errors are encountered, this function returns `false` and stores the
>+// first encountered error in `error`. To disable error reporting, set `error`
>+// to `nullptr` or use the overload with no error reporting below.
>+//
>+// Example:
>+//
>+//   std::string s = "foo\\rbar\\nbaz\\t";
>+//   std::string unescaped_s;
>+//   if (!absl::CUnescape(s, &unescaped_s) {
>+//     ...
>+//   }
>+//   EXPECT_EQ(unescaped_s, "foo\rbar\nbaz\t");
>+bool CUnescape(absl::string_view source, std::string* dest, std::string* error);
>+
>+// Overload of `CUnescape()` with no error reporting.
>+inline bool CUnescape(absl::string_view source, std::string* dest) {
>+  return CUnescape(source, dest, nullptr);
>+}
>+
>+// CEscape()
>+//
>+// Escapes a 'src' std::string using C-style escapes sequences
>+// (http://en.cppreference.com/w/cpp/language/escape), escaping other
>+// non-printable/non-whitespace bytes as octal sequences (e.g. "\377").
>+//
>+// Example:
>+//
>+//   std::string s = "foo\rbar\tbaz\010\011\012\013\014\x0d\n";
>+//   std::string escaped_s = absl::CEscape(s);
>+//   EXPECT_EQ(escaped_s, "foo\\rbar\\tbaz\\010\\t\\n\\013\\014\\r\\n");
>+std::string CEscape(absl::string_view src);
>+
>+// CHexEscape()
>+//
>+// Escapes a 'src' std::string using C-style escape sequences, escaping
>+// other non-printable/non-whitespace bytes as hexadecimal sequences (e.g.
>+// "\xFF").
>+//
>+// Example:
>+//
>+//   std::string s = "foo\rbar\tbaz\010\011\012\013\014\x0d\n";
>+//   std::string escaped_s = absl::CHexEscape(s);
>+//   EXPECT_EQ(escaped_s, "foo\\rbar\\tbaz\\x08\\t\\n\\x0b\\x0c\\r\\n");
>+std::string CHexEscape(absl::string_view src);
>+
>+// Utf8SafeCEscape()
>+//
>+// Escapes a 'src' std::string using C-style escape sequences, escaping bytes as
>+// octal sequences, and passing through UTF-8 characters without conversion.
>+// I.e., when encountering any bytes with their high bit set, this function
>+// will not escape those values, whether or not they are valid UTF-8.
>+std::string Utf8SafeCEscape(absl::string_view src);
>+
>+// Utf8SafeCHexEscape()
>+//
>+// Escapes a 'src' std::string using C-style escape sequences, escaping bytes as
>+// hexadecimal sequences, and passing through UTF-8 characters without
>+// conversion.
>+std::string Utf8SafeCHexEscape(absl::string_view src);
>+
>+// Base64Unescape()
>+//
>+// Converts a `src` std::string encoded in Base64 to its binary equivalent, writing
>+// it to a `dest` buffer, returning `true` on success. If `src` contains invalid
>+// characters, `dest` is cleared and returns `false`.
>+bool Base64Unescape(absl::string_view src, std::string* dest);
>+
>+// WebSafeBase64Unescape(absl::string_view, std::string*)
>+//
>+// Converts a `src` std::string encoded in Base64 to its binary equivalent, writing
>+// it to a `dest` buffer, but using '-' instead of '+', and '_' instead of '/'.
>+// If `src` contains invalid characters, `dest` is cleared and returns `false`.
>+bool WebSafeBase64Unescape(absl::string_view src, std::string* dest);
>+
>+// Base64Escape()
>+//
>+// Encodes a `src` std::string into a `dest` buffer using base64 encoding, with
>+// padding characters. This function conforms with RFC 4648 section 4 (base64).
>+void Base64Escape(absl::string_view src, std::string* dest);
>+
>+// WebSafeBase64Escape()
>+//
>+// Encodes a `src` std::string into a `dest` buffer using '-' instead of '+' and
>+// '_' instead of '/', and without padding. This function conforms with RFC 4648
>+// section 5 (base64url).
>+void WebSafeBase64Escape(absl::string_view src, std::string* dest);
>+
>+// HexStringToBytes()
>+//
>+// Converts an ASCII hex std::string into bytes, returning binary data of length
>+// `from.size()/2`.
>+std::string HexStringToBytes(absl::string_view from);
>+
>+// BytesToHexString()
>+//
>+// Converts binary data into an ASCII text std::string, returning a std::string of size
>+// `2*from.size()`.
>+std::string BytesToHexString(absl::string_view from);
>+
>+}  // namespace absl
>+
>+#endif  // ABSL_STRINGS_ESCAPING_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/escaping_benchmark.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/escaping_benchmark.cc
>new file mode 100644
>index 00000000000..0f791f4e875
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/escaping_benchmark.cc
>@@ -0,0 +1,94 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/strings/escaping.h"
>+
>+#include <cstdio>
>+#include <cstring>
>+#include <random>
>+
>+#include "benchmark/benchmark.h"
>+#include "absl/base/internal/raw_logging.h"
>+#include "absl/strings/internal/escaping_test_common.h"
>+
>+namespace {
>+
>+void BM_CUnescapeHexString(benchmark::State& state) {
>+  std::string src;
>+  for (int i = 0; i < 50; i++) {
>+    src += "\\x55";
>+  }
>+  std::string dest;
>+  for (auto _ : state) {
>+    absl::CUnescape(src, &dest);
>+  }
>+}
>+BENCHMARK(BM_CUnescapeHexString);
>+
>+void BM_WebSafeBase64Escape_string(benchmark::State& state) {
>+  std::string raw;
>+  for (int i = 0; i < 10; ++i) {
>+    for (const auto& test_set : absl::strings_internal::base64_strings()) {
>+      raw += std::string(test_set.plaintext);
>+    }
>+  }
>+
>+  // The actual benchmark loop is tiny...
>+  std::string escaped;
>+  for (auto _ : state) {
>+    absl::WebSafeBase64Escape(raw, &escaped);
>+  }
>+
>+  // We want to be sure the compiler doesn't throw away the loop above,
>+  // and the easiest way to ensure that is to round-trip the results and verify
>+  // them.
>+  std::string round_trip;
>+  absl::WebSafeBase64Unescape(escaped, &round_trip);
>+  ABSL_RAW_CHECK(round_trip == raw, "");
>+}
>+BENCHMARK(BM_WebSafeBase64Escape_string);
>+
>+// Used for the CEscape benchmarks
>+const char kStringValueNoEscape[] = "1234567890";
>+const char kStringValueSomeEscaped[] = "123\n56789\xA1";
>+const char kStringValueMostEscaped[] = "\xA1\xA2\ny\xA4\xA5\xA6z\b\r";
>+
>+void CEscapeBenchmarkHelper(benchmark::State& state, const char* string_value,
>+                            int max_len) {
>+  std::string src;
>+  while (src.size() < max_len) {
>+    absl::StrAppend(&src, string_value);
>+  }
>+
>+  for (auto _ : state) {
>+    absl::CEscape(src);
>+  }
>+}
>+
>+void BM_CEscape_NoEscape(benchmark::State& state) {
>+  CEscapeBenchmarkHelper(state, kStringValueNoEscape, state.range(0));
>+}
>+BENCHMARK(BM_CEscape_NoEscape)->Range(1, 1 << 14);
>+
>+void BM_CEscape_SomeEscaped(benchmark::State& state) {
>+  CEscapeBenchmarkHelper(state, kStringValueSomeEscaped, state.range(0));
>+}
>+BENCHMARK(BM_CEscape_SomeEscaped)->Range(1, 1 << 14);
>+
>+void BM_CEscape_MostEscaped(benchmark::State& state) {
>+  CEscapeBenchmarkHelper(state, kStringValueMostEscaped, state.range(0));
>+}
>+BENCHMARK(BM_CEscape_MostEscaped)->Range(1, 1 << 14);
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/escaping_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/escaping_test.cc
>new file mode 100644
>index 00000000000..3f65ec107f4
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/escaping_test.cc
>@@ -0,0 +1,641 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/strings/escaping.h"
>+
>+#include <array>
>+#include <cstdio>
>+#include <cstring>
>+#include <memory>
>+#include <vector>
>+
>+#include "gmock/gmock.h"
>+#include "gtest/gtest.h"
>+#include "absl/container/fixed_array.h"
>+#include "absl/strings/str_cat.h"
>+
>+#include "absl/strings/internal/escaping_test_common.h"
>+
>+namespace {
>+
>+struct epair {
>+  std::string escaped;
>+  std::string unescaped;
>+};
>+
>+TEST(CEscape, EscapeAndUnescape) {
>+  const std::string inputs[] = {
>+    std::string("foo\nxx\r\b\0023"),
>+    std::string(""),
>+    std::string("abc"),
>+    std::string("\1chad_rules"),
>+    std::string("\1arnar_drools"),
>+    std::string("xxxx\r\t'\"\\"),
>+    std::string("\0xx\0", 4),
>+    std::string("\x01\x31"),
>+    std::string("abc\xb\x42\141bc"),
>+    std::string("123\1\x31\x32\x33"),
>+    std::string("\xc1\xca\x1b\x62\x19o\xcc\x04"),
>+    std::string("\\\"\xe8\xb0\xb7\xe6\xad\x8c\\\" is Google\\\'s Chinese name"),
>+  };
>+  // Do this twice, once for octal escapes and once for hex escapes.
>+  for (int kind = 0; kind < 4; kind++) {
>+    for (const std::string& original : inputs) {
>+      std::string escaped;
>+      switch (kind) {
>+        case 0:
>+          escaped = absl::CEscape(original);
>+          break;
>+        case 1:
>+          escaped = absl::CHexEscape(original);
>+          break;
>+        case 2:
>+          escaped = absl::Utf8SafeCEscape(original);
>+          break;
>+        case 3:
>+          escaped = absl::Utf8SafeCHexEscape(original);
>+          break;
>+      }
>+      std::string unescaped_str;
>+      EXPECT_TRUE(absl::CUnescape(escaped, &unescaped_str));
>+      EXPECT_EQ(unescaped_str, original);
>+
>+      // Check in-place unescaping
>+      std::string s = escaped;
>+      EXPECT_TRUE(absl::CUnescape(s, &s));
>+      ASSERT_EQ(s, original);
>+    }
>+  }
>+  // Check that all possible two character strings can be escaped then
>+  // unescaped successfully.
>+  for (int char0 = 0; char0 < 256; char0++) {
>+    for (int char1 = 0; char1 < 256; char1++) {
>+      char chars[2];
>+      chars[0] = char0;
>+      chars[1] = char1;
>+      std::string s(chars, 2);
>+      std::string escaped = absl::CHexEscape(s);
>+      std::string unescaped;
>+      EXPECT_TRUE(absl::CUnescape(escaped, &unescaped));
>+      EXPECT_EQ(s, unescaped);
>+    }
>+  }
>+}
>+
>+TEST(CEscape, BasicEscaping) {
>+  epair oct_values[] = {
>+      {"foo\\rbar\\nbaz\\t", "foo\rbar\nbaz\t"},
>+      {"\\'full of \\\"sound\\\" and \\\"fury\\\"\\'",
>+       "'full of \"sound\" and \"fury\"'"},
>+      {"signi\\\\fying\\\\ nothing\\\\", "signi\\fying\\ nothing\\"},
>+      {"\\010\\t\\n\\013\\014\\r", "\010\011\012\013\014\015"}
>+  };
>+  epair hex_values[] = {
>+      {"ubik\\rubik\\nubik\\t", "ubik\rubik\nubik\t"},
>+      {"I\\\'ve just seen a \\\"face\\\"",
>+       "I've just seen a \"face\""},
>+      {"hel\\\\ter\\\\skel\\\\ter\\\\", "hel\\ter\\skel\\ter\\"},
>+      {"\\x08\\t\\n\\x0b\\x0c\\r", "\010\011\012\013\014\015"}
>+  };
>+  epair utf8_oct_values[] = {
>+      {"\xe8\xb0\xb7\xe6\xad\x8c\\r\xe8\xb0\xb7\xe6\xad\x8c\\nbaz\\t",
>+       "\xe8\xb0\xb7\xe6\xad\x8c\r\xe8\xb0\xb7\xe6\xad\x8c\nbaz\t"},
>+      {"\\\"\xe8\xb0\xb7\xe6\xad\x8c\\\" is Google\\\'s Chinese name",
>+       "\"\xe8\xb0\xb7\xe6\xad\x8c\" is Google\'s Chinese name"},
>+      {"\xe3\x83\xa1\xe3\x83\xbc\xe3\x83\xab\\\\are\\\\Japanese\\\\chars\\\\",
>+       "\xe3\x83\xa1\xe3\x83\xbc\xe3\x83\xab\\are\\Japanese\\chars\\"},
>+      {"\xed\x81\xac\xeb\xa1\xac\\010\\t\\n\\013\\014\\r",
>+       "\xed\x81\xac\xeb\xa1\xac\010\011\012\013\014\015"}
>+  };
>+  epair utf8_hex_values[] = {
>+      {"\x20\xe4\xbd\xa0\\t\xe5\xa5\xbd,\\r!\\n",
>+       "\x20\xe4\xbd\xa0\t\xe5\xa5\xbd,\r!\n"},
>+      {"\xe8\xa9\xa6\xe9\xa8\x93\\\' means \\\"test\\\"",
>+       "\xe8\xa9\xa6\xe9\xa8\x93\' means \"test\""},
>+      {"\\\\\xe6\x88\x91\\\\:\\\\\xe6\x9d\xa8\xe6\xac\xa2\\\\",
>+       "\\\xe6\x88\x91\\:\\\xe6\x9d\xa8\xe6\xac\xa2\\"},
>+      {"\xed\x81\xac\xeb\xa1\xac\\x08\\t\\n\\x0b\\x0c\\r",
>+       "\xed\x81\xac\xeb\xa1\xac\010\011\012\013\014\015"}
>+  };
>+
>+  for (const epair& val : oct_values) {
>+    std::string escaped = absl::CEscape(val.unescaped);
>+    EXPECT_EQ(escaped, val.escaped);
>+  }
>+  for (const epair& val : hex_values) {
>+    std::string escaped = absl::CHexEscape(val.unescaped);
>+    EXPECT_EQ(escaped, val.escaped);
>+  }
>+  for (const epair& val : utf8_oct_values) {
>+    std::string escaped = absl::Utf8SafeCEscape(val.unescaped);
>+    EXPECT_EQ(escaped, val.escaped);
>+  }
>+  for (const epair& val : utf8_hex_values) {
>+    std::string escaped = absl::Utf8SafeCHexEscape(val.unescaped);
>+    EXPECT_EQ(escaped, val.escaped);
>+  }
>+}
>+
>+TEST(Unescape, BasicFunction) {
>+  epair tests[] =
>+    {{"\\u0030", "0"},
>+     {"\\u00A3", "\xC2\xA3"},
>+     {"\\u22FD", "\xE2\x8B\xBD"},
>+     {"\\U00010000", "\xF0\x90\x80\x80"},
>+     {"\\U0010FFFD", "\xF4\x8F\xBF\xBD"}};
>+  for (const epair& val : tests) {
>+    std::string out;
>+    EXPECT_TRUE(absl::CUnescape(val.escaped, &out));
>+    EXPECT_EQ(out, val.unescaped);
>+  }
>+  std::string bad[] =
>+     {"\\u1",         // too short
>+      "\\U1",         // too short
>+      "\\Uffffff",    // exceeds 0x10ffff (largest Unicode)
>+      "\\U00110000",  // exceeds 0x10ffff (largest Unicode)
>+      "\\uD835",      // surrogate character (D800-DFFF)
>+      "\\U0000DD04",  // surrogate character (D800-DFFF)
>+      "\\777",        // exceeds 0xff
>+      "\\xABCD"};     // exceeds 0xff
>+  for (const std::string& e : bad) {
>+    std::string error;
>+    std::string out;
>+    EXPECT_FALSE(absl::CUnescape(e, &out, &error));
>+    EXPECT_FALSE(error.empty());
>+  }
>+}
>+
>+class CUnescapeTest : public testing::Test {
>+ protected:
>+  static const char kStringWithMultipleOctalNulls[];
>+  static const char kStringWithMultipleHexNulls[];
>+  static const char kStringWithMultipleUnicodeNulls[];
>+
>+  std::string result_string_;
>+};
>+
>+const char CUnescapeTest::kStringWithMultipleOctalNulls[] =
>+    "\\0\\n"    // null escape \0 plus newline
>+    "0\\n"      // just a number 0 (not a null escape) plus newline
>+    "\\00\\12"  // null escape \00 plus octal newline code
>+    "\\000";    // null escape \000
>+
>+// This has the same ingredients as kStringWithMultipleOctalNulls
>+// but with \x hex escapes instead of octal escapes.
>+const char CUnescapeTest::kStringWithMultipleHexNulls[] =
>+    "\\x0\\n"
>+    "0\\n"
>+    "\\x00\\xa"
>+    "\\x000";
>+
>+const char CUnescapeTest::kStringWithMultipleUnicodeNulls[] =
>+    "\\u0000\\n"    // short-form (4-digit) null escape plus newline
>+    "0\\n"          // just a number 0 (not a null escape) plus newline
>+    "\\U00000000";  // long-form (8-digit) null escape
>+
>+TEST_F(CUnescapeTest, Unescapes1CharOctalNull) {
>+  std::string original_string = "\\0";
>+  EXPECT_TRUE(absl::CUnescape(original_string, &result_string_));
>+  EXPECT_EQ(std::string("\0", 1), result_string_);
>+}
>+
>+TEST_F(CUnescapeTest, Unescapes2CharOctalNull) {
>+  std::string original_string = "\\00";
>+  EXPECT_TRUE(absl::CUnescape(original_string, &result_string_));
>+  EXPECT_EQ(std::string("\0", 1), result_string_);
>+}
>+
>+TEST_F(CUnescapeTest, Unescapes3CharOctalNull) {
>+  std::string original_string = "\\000";
>+  EXPECT_TRUE(absl::CUnescape(original_string, &result_string_));
>+  EXPECT_EQ(std::string("\0", 1), result_string_);
>+}
>+
>+TEST_F(CUnescapeTest, Unescapes1CharHexNull) {
>+  std::string original_string = "\\x0";
>+  EXPECT_TRUE(absl::CUnescape(original_string, &result_string_));
>+  EXPECT_EQ(std::string("\0", 1), result_string_);
>+}
>+
>+TEST_F(CUnescapeTest, Unescapes2CharHexNull) {
>+  std::string original_string = "\\x00";
>+  EXPECT_TRUE(absl::CUnescape(original_string, &result_string_));
>+  EXPECT_EQ(std::string("\0", 1), result_string_);
>+}
>+
>+TEST_F(CUnescapeTest, Unescapes3CharHexNull) {
>+  std::string original_string = "\\x000";
>+  EXPECT_TRUE(absl::CUnescape(original_string, &result_string_));
>+  EXPECT_EQ(std::string("\0", 1), result_string_);
>+}
>+
>+TEST_F(CUnescapeTest, Unescapes4CharUnicodeNull) {
>+  std::string original_string = "\\u0000";
>+  EXPECT_TRUE(absl::CUnescape(original_string, &result_string_));
>+  EXPECT_EQ(std::string("\0", 1), result_string_);
>+}
>+
>+TEST_F(CUnescapeTest, Unescapes8CharUnicodeNull) {
>+  std::string original_string = "\\U00000000";
>+  EXPECT_TRUE(absl::CUnescape(original_string, &result_string_));
>+  EXPECT_EQ(std::string("\0", 1), result_string_);
>+}
>+
>+TEST_F(CUnescapeTest, UnescapesMultipleOctalNulls) {
>+  std::string original_string(kStringWithMultipleOctalNulls);
>+  EXPECT_TRUE(absl::CUnescape(original_string, &result_string_));
>+  // All escapes, including newlines and null escapes, should have been
>+  // converted to the equivalent characters.
>+  EXPECT_EQ(std::string("\0\n"
>+                   "0\n"
>+                   "\0\n"
>+                   "\0", 7), result_string_);
>+}
>+
>+
>+TEST_F(CUnescapeTest, UnescapesMultipleHexNulls) {
>+  std::string original_string(kStringWithMultipleHexNulls);
>+  EXPECT_TRUE(absl::CUnescape(original_string, &result_string_));
>+  EXPECT_EQ(std::string("\0\n"
>+                   "0\n"
>+                   "\0\n"
>+                   "\0", 7), result_string_);
>+}
>+
>+TEST_F(CUnescapeTest, UnescapesMultipleUnicodeNulls) {
>+  std::string original_string(kStringWithMultipleUnicodeNulls);
>+  EXPECT_TRUE(absl::CUnescape(original_string, &result_string_));
>+  EXPECT_EQ(std::string("\0\n"
>+                   "0\n"
>+                   "\0", 5), result_string_);
>+}
>+
>+static struct {
>+  absl::string_view plaintext;
>+  absl::string_view cyphertext;
>+} const base64_tests[] = {
>+    // Empty std::string.
>+    {{"", 0}, {"", 0}},
>+    {{nullptr, 0},
>+     {"", 0}},  // if length is zero, plaintext ptr must be ignored!
>+
>+    // Basic bit patterns;
>+    // values obtained with "echo -n '...' | uuencode -m test"
>+
>+    {{"\000", 1}, "AA=="},
>+    {{"\001", 1}, "AQ=="},
>+    {{"\002", 1}, "Ag=="},
>+    {{"\004", 1}, "BA=="},
>+    {{"\010", 1}, "CA=="},
>+    {{"\020", 1}, "EA=="},
>+    {{"\040", 1}, "IA=="},
>+    {{"\100", 1}, "QA=="},
>+    {{"\200", 1}, "gA=="},
>+
>+    {{"\377", 1}, "/w=="},
>+    {{"\376", 1}, "/g=="},
>+    {{"\375", 1}, "/Q=="},
>+    {{"\373", 1}, "+w=="},
>+    {{"\367", 1}, "9w=="},
>+    {{"\357", 1}, "7w=="},
>+    {{"\337", 1}, "3w=="},
>+    {{"\277", 1}, "vw=="},
>+    {{"\177", 1}, "fw=="},
>+    {{"\000\000", 2}, "AAA="},
>+    {{"\000\001", 2}, "AAE="},
>+    {{"\000\002", 2}, "AAI="},
>+    {{"\000\004", 2}, "AAQ="},
>+    {{"\000\010", 2}, "AAg="},
>+    {{"\000\020", 2}, "ABA="},
>+    {{"\000\040", 2}, "ACA="},
>+    {{"\000\100", 2}, "AEA="},
>+    {{"\000\200", 2}, "AIA="},
>+    {{"\001\000", 2}, "AQA="},
>+    {{"\002\000", 2}, "AgA="},
>+    {{"\004\000", 2}, "BAA="},
>+    {{"\010\000", 2}, "CAA="},
>+    {{"\020\000", 2}, "EAA="},
>+    {{"\040\000", 2}, "IAA="},
>+    {{"\100\000", 2}, "QAA="},
>+    {{"\200\000", 2}, "gAA="},
>+
>+    {{"\377\377", 2}, "//8="},
>+    {{"\377\376", 2}, "//4="},
>+    {{"\377\375", 2}, "//0="},
>+    {{"\377\373", 2}, "//s="},
>+    {{"\377\367", 2}, "//c="},
>+    {{"\377\357", 2}, "/+8="},
>+    {{"\377\337", 2}, "/98="},
>+    {{"\377\277", 2}, "/78="},
>+    {{"\377\177", 2}, "/38="},
>+    {{"\376\377", 2}, "/v8="},
>+    {{"\375\377", 2}, "/f8="},
>+    {{"\373\377", 2}, "+/8="},
>+    {{"\367\377", 2}, "9/8="},
>+    {{"\357\377", 2}, "7/8="},
>+    {{"\337\377", 2}, "3/8="},
>+    {{"\277\377", 2}, "v/8="},
>+    {{"\177\377", 2}, "f/8="},
>+
>+    {{"\000\000\000", 3}, "AAAA"},
>+    {{"\000\000\001", 3}, "AAAB"},
>+    {{"\000\000\002", 3}, "AAAC"},
>+    {{"\000\000\004", 3}, "AAAE"},
>+    {{"\000\000\010", 3}, "AAAI"},
>+    {{"\000\000\020", 3}, "AAAQ"},
>+    {{"\000\000\040", 3}, "AAAg"},
>+    {{"\000\000\100", 3}, "AABA"},
>+    {{"\000\000\200", 3}, "AACA"},
>+    {{"\000\001\000", 3}, "AAEA"},
>+    {{"\000\002\000", 3}, "AAIA"},
>+    {{"\000\004\000", 3}, "AAQA"},
>+    {{"\000\010\000", 3}, "AAgA"},
>+    {{"\000\020\000", 3}, "ABAA"},
>+    {{"\000\040\000", 3}, "ACAA"},
>+    {{"\000\100\000", 3}, "AEAA"},
>+    {{"\000\200\000", 3}, "AIAA"},
>+    {{"\001\000\000", 3}, "AQAA"},
>+    {{"\002\000\000", 3}, "AgAA"},
>+    {{"\004\000\000", 3}, "BAAA"},
>+    {{"\010\000\000", 3}, "CAAA"},
>+    {{"\020\000\000", 3}, "EAAA"},
>+    {{"\040\000\000", 3}, "IAAA"},
>+    {{"\100\000\000", 3}, "QAAA"},
>+    {{"\200\000\000", 3}, "gAAA"},
>+
>+    {{"\377\377\377", 3}, "////"},
>+    {{"\377\377\376", 3}, "///+"},
>+    {{"\377\377\375", 3}, "///9"},
>+    {{"\377\377\373", 3}, "///7"},
>+    {{"\377\377\367", 3}, "///3"},
>+    {{"\377\377\357", 3}, "///v"},
>+    {{"\377\377\337", 3}, "///f"},
>+    {{"\377\377\277", 3}, "//+/"},
>+    {{"\377\377\177", 3}, "//9/"},
>+    {{"\377\376\377", 3}, "//7/"},
>+    {{"\377\375\377", 3}, "//3/"},
>+    {{"\377\373\377", 3}, "//v/"},
>+    {{"\377\367\377", 3}, "//f/"},
>+    {{"\377\357\377", 3}, "/+//"},
>+    {{"\377\337\377", 3}, "/9//"},
>+    {{"\377\277\377", 3}, "/7//"},
>+    {{"\377\177\377", 3}, "/3//"},
>+    {{"\376\377\377", 3}, "/v//"},
>+    {{"\375\377\377", 3}, "/f//"},
>+    {{"\373\377\377", 3}, "+///"},
>+    {{"\367\377\377", 3}, "9///"},
>+    {{"\357\377\377", 3}, "7///"},
>+    {{"\337\377\377", 3}, "3///"},
>+    {{"\277\377\377", 3}, "v///"},
>+    {{"\177\377\377", 3}, "f///"},
>+
>+    // Random numbers: values obtained with
>+    //
>+    //  #! /bin/bash
>+    //  dd bs=$1 count=1 if=/dev/random of=/tmp/bar.random
>+    //  od -N $1 -t o1 /tmp/bar.random
>+    //  uuencode -m test < /tmp/bar.random
>+    //
>+    // where $1 is the number of bytes (2, 3)
>+
>+    {{"\243\361", 2}, "o/E="},
>+    {{"\024\167", 2}, "FHc="},
>+    {{"\313\252", 2}, "y6o="},
>+    {{"\046\041", 2}, "JiE="},
>+    {{"\145\236", 2}, "ZZ4="},
>+    {{"\254\325", 2}, "rNU="},
>+    {{"\061\330", 2}, "Mdg="},
>+    {{"\245\032", 2}, "pRo="},
>+    {{"\006\000", 2}, "BgA="},
>+    {{"\375\131", 2}, "/Vk="},
>+    {{"\303\210", 2}, "w4g="},
>+    {{"\040\037", 2}, "IB8="},
>+    {{"\261\372", 2}, "sfo="},
>+    {{"\335\014", 2}, "3Qw="},
>+    {{"\233\217", 2}, "m48="},
>+    {{"\373\056", 2}, "+y4="},
>+    {{"\247\232", 2}, "p5o="},
>+    {{"\107\053", 2}, "Rys="},
>+    {{"\204\077", 2}, "hD8="},
>+    {{"\276\211", 2}, "vok="},
>+    {{"\313\110", 2}, "y0g="},
>+    {{"\363\376", 2}, "8/4="},
>+    {{"\251\234", 2}, "qZw="},
>+    {{"\103\262", 2}, "Q7I="},
>+    {{"\142\312", 2}, "Yso="},
>+    {{"\067\211", 2}, "N4k="},
>+    {{"\220\001", 2}, "kAE="},
>+    {{"\152\240", 2}, "aqA="},
>+    {{"\367\061", 2}, "9zE="},
>+    {{"\133\255", 2}, "W60="},
>+    {{"\176\035", 2}, "fh0="},
>+    {{"\032\231", 2}, "Gpk="},
>+
>+    {{"\013\007\144", 3}, "Cwdk"},
>+    {{"\030\112\106", 3}, "GEpG"},
>+    {{"\047\325\046", 3}, "J9Um"},
>+    {{"\310\160\022", 3}, "yHAS"},
>+    {{"\131\100\237", 3}, "WUCf"},
>+    {{"\064\342\134", 3}, "NOJc"},
>+    {{"\010\177\004", 3}, "CH8E"},
>+    {{"\345\147\205", 3}, "5WeF"},
>+    {{"\300\343\360", 3}, "wOPw"},
>+    {{"\061\240\201", 3}, "MaCB"},
>+    {{"\225\333\044", 3}, "ldsk"},
>+    {{"\215\137\352", 3}, "jV/q"},
>+    {{"\371\147\160", 3}, "+Wdw"},
>+    {{"\030\320\051", 3}, "GNAp"},
>+    {{"\044\174\241", 3}, "JHyh"},
>+    {{"\260\127\037", 3}, "sFcf"},
>+    {{"\111\045\033", 3}, "SSUb"},
>+    {{"\202\114\107", 3}, "gkxH"},
>+    {{"\057\371\042", 3}, "L/ki"},
>+    {{"\223\247\244", 3}, "k6ek"},
>+    {{"\047\216\144", 3}, "J45k"},
>+    {{"\203\070\327", 3}, "gzjX"},
>+    {{"\247\140\072", 3}, "p2A6"},
>+    {{"\124\115\116", 3}, "VE1O"},
>+    {{"\157\162\050", 3}, "b3Io"},
>+    {{"\357\223\004", 3}, "75ME"},
>+    {{"\052\117\156", 3}, "Kk9u"},
>+    {{"\347\154\000", 3}, "52wA"},
>+    {{"\303\012\142", 3}, "wwpi"},
>+    {{"\060\035\362", 3}, "MB3y"},
>+    {{"\130\226\361", 3}, "WJbx"},
>+    {{"\173\013\071", 3}, "ews5"},
>+    {{"\336\004\027", 3}, "3gQX"},
>+    {{"\357\366\234", 3}, "7/ac"},
>+    {{"\353\304\111", 3}, "68RJ"},
>+    {{"\024\264\131", 3}, "FLRZ"},
>+    {{"\075\114\251", 3}, "PUyp"},
>+    {{"\315\031\225", 3}, "zRmV"},
>+    {{"\154\201\276", 3}, "bIG+"},
>+    {{"\200\066\072", 3}, "gDY6"},
>+    {{"\142\350\267", 3}, "Yui3"},
>+    {{"\033\000\166", 3}, "GwB2"},
>+    {{"\210\055\077", 3}, "iC0/"},
>+    {{"\341\037\124", 3}, "4R9U"},
>+    {{"\161\103\152", 3}, "cUNq"},
>+    {{"\270\142\131", 3}, "uGJZ"},
>+    {{"\337\076\074", 3}, "3z48"},
>+    {{"\375\106\362", 3}, "/Uby"},
>+    {{"\227\301\127", 3}, "l8FX"},
>+    {{"\340\002\234", 3}, "4AKc"},
>+    {{"\121\064\033", 3}, "UTQb"},
>+    {{"\157\134\143", 3}, "b1xj"},
>+    {{"\247\055\327", 3}, "py3X"},
>+    {{"\340\142\005", 3}, "4GIF"},
>+    {{"\060\260\143", 3}, "MLBj"},
>+    {{"\075\203\170", 3}, "PYN4"},
>+    {{"\143\160\016", 3}, "Y3AO"},
>+    {{"\313\013\063", 3}, "ywsz"},
>+    {{"\174\236\135", 3}, "fJ5d"},
>+    {{"\103\047\026", 3}, "QycW"},
>+    {{"\365\005\343", 3}, "9QXj"},
>+    {{"\271\160\223", 3}, "uXCT"},
>+    {{"\362\255\172", 3}, "8q16"},
>+    {{"\113\012\015", 3}, "SwoN"},
>+
>+    // various lengths, generated by this python script:
>+    //
>+    // from std::string import lowercase as lc
>+    // for i in range(27):
>+    //   print '{ %2d, "%s",%s "%s" },' % (i, lc[:i], ' ' * (26-i),
>+    //                                     lc[:i].encode('base64').strip())
>+
>+    {{"", 0}, {"", 0}},
>+    {"a", "YQ=="},
>+    {"ab", "YWI="},
>+    {"abc", "YWJj"},
>+    {"abcd", "YWJjZA=="},
>+    {"abcde", "YWJjZGU="},
>+    {"abcdef", "YWJjZGVm"},
>+    {"abcdefg", "YWJjZGVmZw=="},
>+    {"abcdefgh", "YWJjZGVmZ2g="},
>+    {"abcdefghi", "YWJjZGVmZ2hp"},
>+    {"abcdefghij", "YWJjZGVmZ2hpag=="},
>+    {"abcdefghijk", "YWJjZGVmZ2hpams="},
>+    {"abcdefghijkl", "YWJjZGVmZ2hpamts"},
>+    {"abcdefghijklm", "YWJjZGVmZ2hpamtsbQ=="},
>+    {"abcdefghijklmn", "YWJjZGVmZ2hpamtsbW4="},
>+    {"abcdefghijklmno", "YWJjZGVmZ2hpamtsbW5v"},
>+    {"abcdefghijklmnop", "YWJjZGVmZ2hpamtsbW5vcA=="},
>+    {"abcdefghijklmnopq", "YWJjZGVmZ2hpamtsbW5vcHE="},
>+    {"abcdefghijklmnopqr", "YWJjZGVmZ2hpamtsbW5vcHFy"},
>+    {"abcdefghijklmnopqrs", "YWJjZGVmZ2hpamtsbW5vcHFycw=="},
>+    {"abcdefghijklmnopqrst", "YWJjZGVmZ2hpamtsbW5vcHFyc3Q="},
>+    {"abcdefghijklmnopqrstu", "YWJjZGVmZ2hpamtsbW5vcHFyc3R1"},
>+    {"abcdefghijklmnopqrstuv", "YWJjZGVmZ2hpamtsbW5vcHFyc3R1dg=="},
>+    {"abcdefghijklmnopqrstuvw", "YWJjZGVmZ2hpamtsbW5vcHFyc3R1dnc="},
>+    {"abcdefghijklmnopqrstuvwx", "YWJjZGVmZ2hpamtsbW5vcHFyc3R1dnd4"},
>+    {"abcdefghijklmnopqrstuvwxy", "YWJjZGVmZ2hpamtsbW5vcHFyc3R1dnd4eQ=="},
>+    {"abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2hpamtsbW5vcHFyc3R1dnd4eXo="},
>+};
>+
>+TEST(Base64, EscapeAndUnescape) {
>+  // Check the short strings; this tests the math (and boundaries)
>+  for (const auto& tc : base64_tests) {
>+    std::string encoded("this junk should be ignored");
>+    absl::Base64Escape(tc.plaintext, &encoded);
>+    EXPECT_EQ(encoded, tc.cyphertext);
>+
>+    std::string decoded("this junk should be ignored");
>+    EXPECT_TRUE(absl::Base64Unescape(encoded, &decoded));
>+    EXPECT_EQ(decoded, tc.plaintext);
>+
>+    std::string websafe(tc.cyphertext);
>+    for (int c = 0; c < websafe.size(); ++c) {
>+      if ('+' == websafe[c]) websafe[c] = '-';
>+      if ('/' == websafe[c]) websafe[c] = '_';
>+      if ('=' == websafe[c]) {
>+        websafe.resize(c);
>+        break;
>+      }
>+    }
>+
>+    encoded = "this junk should be ignored";
>+    absl::WebSafeBase64Escape(tc.plaintext, &encoded);
>+    EXPECT_EQ(encoded, websafe);
>+
>+    // Let's try the std::string version of the decoder
>+    decoded = "this junk should be ignored";
>+    EXPECT_TRUE(absl::WebSafeBase64Unescape(websafe, &decoded));
>+    EXPECT_EQ(decoded, tc.plaintext);
>+  }
>+
>+  // Now try the long strings, this tests the streaming
>+  for (const auto& tc : absl::strings_internal::base64_strings()) {
>+    std::string buffer;
>+    absl::WebSafeBase64Escape(tc.plaintext, &buffer);
>+    EXPECT_EQ(tc.cyphertext, buffer);
>+  }
>+
>+  // Verify the behavior when decoding bad data
>+  {
>+    absl::string_view data_set[] = {"ab-/", absl::string_view("\0bcd", 4),
>+                                    absl::string_view("abc.\0", 5)};
>+    for (absl::string_view bad_data : data_set) {
>+      std::string buf;
>+      EXPECT_FALSE(absl::Base64Unescape(bad_data, &buf));
>+      EXPECT_FALSE(absl::WebSafeBase64Unescape(bad_data, &buf));
>+      EXPECT_TRUE(buf.empty());
>+    }
>+  }
>+}
>+
>+TEST(Base64, DISABLED_HugeData) {
>+  const size_t kSize = size_t(3) * 1000 * 1000 * 1000;
>+  static_assert(kSize % 3 == 0, "kSize must be divisible by 3");
>+  const std::string huge(kSize, 'x');
>+
>+  std::string escaped;
>+  absl::Base64Escape(huge, &escaped);
>+
>+  // Generates the std::string that should match a base64 encoded "xxx..." std::string.
>+  // "xxx" in base64 is "eHh4".
>+  std::string expected_encoding;
>+  expected_encoding.reserve(kSize / 3 * 4);
>+  for (size_t i = 0; i < kSize / 3; ++i) {
>+    expected_encoding.append("eHh4");
>+  }
>+  EXPECT_EQ(expected_encoding, escaped);
>+
>+  std::string unescaped;
>+  EXPECT_TRUE(absl::Base64Unescape(escaped, &unescaped));
>+  EXPECT_EQ(huge, unescaped);
>+}
>+
>+TEST(HexAndBack, HexStringToBytes_and_BytesToHexString) {
>+  std::string hex_mixed = "0123456789abcdefABCDEF";
>+  std::string bytes_expected = "\x01\x23\x45\x67\x89\xab\xcd\xef\xAB\xCD\xEF";
>+  std::string hex_only_lower = "0123456789abcdefabcdef";
>+
>+  std::string bytes_result = absl::HexStringToBytes(hex_mixed);
>+  EXPECT_EQ(bytes_expected, bytes_result);
>+
>+  std::string prefix_valid = hex_mixed + "?";
>+  std::string prefix_valid_result = absl::HexStringToBytes(
>+      absl::string_view(prefix_valid.data(), prefix_valid.size() - 1));
>+  EXPECT_EQ(bytes_expected, prefix_valid_result);
>+
>+  std::string infix_valid = "?" + hex_mixed + "???";
>+  std::string infix_valid_result = absl::HexStringToBytes(
>+      absl::string_view(infix_valid.data() + 1, hex_mixed.size()));
>+  EXPECT_EQ(bytes_expected, infix_valid_result);
>+
>+  std::string hex_result = absl::BytesToHexString(bytes_expected);
>+  EXPECT_EQ(hex_only_lower, hex_result);
>+}
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/bits.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/bits.h
>new file mode 100644
>index 00000000000..901082ccddf
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/bits.h
>@@ -0,0 +1,53 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#ifndef ABSL_STRINGS_INTERNAL_BITS_H_
>+#define ABSL_STRINGS_INTERNAL_BITS_H_
>+
>+#include <cstdint>
>+
>+#if defined(_MSC_VER) && defined(_M_X64)
>+#include <intrin.h>
>+#pragma intrinsic(_BitScanReverse64)
>+#endif
>+
>+namespace absl {
>+namespace strings_internal {
>+
>+// Returns the number of leading 0 bits in a 64-bit value.
>+inline int CountLeadingZeros64(uint64_t n) {
>+#if defined(__GNUC__)
>+  static_assert(sizeof(unsigned long long) == sizeof(n),  // NOLINT(runtime/int)
>+                "__builtin_clzll does not take 64bit arg");
>+  return n == 0 ? 64 : __builtin_clzll(n);
>+#elif defined(_MSC_VER) && defined(_M_X64)
>+  unsigned long result;  // NOLINT(runtime/int)
>+  if (_BitScanReverse64(&result, n)) {
>+    return 63 - result;
>+  }
>+  return 64;
>+#else
>+  int zeroes = 60;
>+  if (n >> 32) zeroes -= 32, n >>= 32;
>+  if (n >> 16) zeroes -= 16, n >>= 16;
>+  if (n >> 8) zeroes -= 8, n >>= 8;
>+  if (n >> 4) zeroes -= 4, n >>= 4;
>+  return "\4\3\2\2\1\1\1\1\0\0\0\0\0\0\0\0"[n] + zeroes;
>+#endif
>+}
>+
>+}  // namespace strings_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_STRINGS_INTERNAL_BITS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/char_map.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/char_map.h
>new file mode 100644
>index 00000000000..8d92963a4de
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/char_map.h
>@@ -0,0 +1,154 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// Character Map Class
>+//
>+// A fast, bit-vector map for 8-bit unsigned characters.
>+// This class is useful for non-character purposes as well.
>+
>+#ifndef ABSL_STRINGS_INTERNAL_CHAR_MAP_H_
>+#define ABSL_STRINGS_INTERNAL_CHAR_MAP_H_
>+
>+#include <cstddef>
>+#include <cstdint>
>+#include <cstring>
>+
>+#include "absl/base/macros.h"
>+#include "absl/base/port.h"
>+
>+namespace absl {
>+namespace strings_internal {
>+
>+class Charmap {
>+ public:
>+  constexpr Charmap() : m_() {}
>+
>+  // Initializes with a given char*.  Note that NUL is not treated as
>+  // a terminator, but rather a char to be flicked.
>+  Charmap(const char* str, int len) : m_() {
>+    while (len--) SetChar(*str++);
>+  }
>+
>+  // Initializes with a given char*.  NUL is treated as a terminator
>+  // and will not be in the charmap.
>+  explicit Charmap(const char* str) : m_() {
>+    while (*str) SetChar(*str++);
>+  }
>+
>+  constexpr bool contains(unsigned char c) const {
>+    return (m_[c / 64] >> (c % 64)) & 0x1;
>+  }
>+
>+  // Returns true if and only if a character exists in both maps.
>+  bool IntersectsWith(const Charmap& c) const {
>+    for (size_t i = 0; i < ABSL_ARRAYSIZE(m_); ++i) {
>+      if ((m_[i] & c.m_[i]) != 0) return true;
>+    }
>+    return false;
>+  }
>+
>+  bool IsZero() const {
>+    for (uint64_t c : m_) {
>+      if (c != 0) return false;
>+    }
>+    return true;
>+  }
>+
>+  // Containing only a single specified char.
>+  static constexpr Charmap Char(char x) {
>+    return Charmap(CharMaskForWord(x, 0), CharMaskForWord(x, 1),
>+                   CharMaskForWord(x, 2), CharMaskForWord(x, 3));
>+  }
>+
>+  // Containing all the chars in the C-std::string 's'.
>+  // Note that this is expensively recursive because of the C++11 constexpr
>+  // formulation. Use only in constexpr initializers.
>+  static constexpr Charmap FromString(const char* s) {
>+    return *s == 0 ? Charmap() : (Char(*s) | FromString(s + 1));
>+  }
>+
>+  // Containing all the chars in the closed interval [lo,hi].
>+  static constexpr Charmap Range(char lo, char hi) {
>+    return Charmap(RangeForWord(lo, hi, 0), RangeForWord(lo, hi, 1),
>+                   RangeForWord(lo, hi, 2), RangeForWord(lo, hi, 3));
>+  }
>+
>+  friend constexpr Charmap operator&(const Charmap& a, const Charmap& b) {
>+    return Charmap(a.m_[0] & b.m_[0], a.m_[1] & b.m_[1], a.m_[2] & b.m_[2],
>+                   a.m_[3] & b.m_[3]);
>+  }
>+
>+  friend constexpr Charmap operator|(const Charmap& a, const Charmap& b) {
>+    return Charmap(a.m_[0] | b.m_[0], a.m_[1] | b.m_[1], a.m_[2] | b.m_[2],
>+                   a.m_[3] | b.m_[3]);
>+  }
>+
>+  friend constexpr Charmap operator~(const Charmap& a) {
>+    return Charmap(~a.m_[0], ~a.m_[1], ~a.m_[2], ~a.m_[3]);
>+  }
>+
>+ private:
>+  constexpr Charmap(uint64_t b0, uint64_t b1, uint64_t b2, uint64_t b3)
>+      : m_{b0, b1, b2, b3} {}
>+
>+  static constexpr uint64_t RangeForWord(unsigned char lo, unsigned char hi,
>+                                         uint64_t word) {
>+    return OpenRangeFromZeroForWord(hi + 1, word) &
>+           ~OpenRangeFromZeroForWord(lo, word);
>+  }
>+
>+  // All the chars in the specified word of the range [0, upper).
>+  static constexpr uint64_t OpenRangeFromZeroForWord(uint64_t upper,
>+                                                     uint64_t word) {
>+    return (upper <= 64 * word)
>+               ? 0
>+               : (upper >= 64 * (word + 1))
>+                     ? ~static_cast<uint64_t>(0)
>+                     : (~static_cast<uint64_t>(0) >> (64 - upper % 64));
>+  }
>+
>+  static constexpr uint64_t CharMaskForWord(unsigned char x, uint64_t word) {
>+    return (x / 64 == word) ? (static_cast<uint64_t>(1) << (x % 64)) : 0;
>+  }
>+
>+ private:
>+  void SetChar(unsigned char c) {
>+    m_[c / 64] |= static_cast<uint64_t>(1) << (c % 64);
>+  }
>+
>+  uint64_t m_[4];
>+};
>+
>+// Mirror the char-classifying predicates in <cctype>
>+constexpr Charmap UpperCharmap() { return Charmap::Range('A', 'Z'); }
>+constexpr Charmap LowerCharmap() { return Charmap::Range('a', 'z'); }
>+constexpr Charmap DigitCharmap() { return Charmap::Range('0', '9'); }
>+constexpr Charmap AlphaCharmap() { return LowerCharmap() | UpperCharmap(); }
>+constexpr Charmap AlnumCharmap() { return DigitCharmap() | AlphaCharmap(); }
>+constexpr Charmap XDigitCharmap() {
>+  return DigitCharmap() | Charmap::Range('A', 'F') | Charmap::Range('a', 'f');
>+}
>+constexpr Charmap PrintCharmap() { return Charmap::Range(0x20, 0x7e); }
>+constexpr Charmap SpaceCharmap() { return Charmap::FromString("\t\n\v\f\r "); }
>+constexpr Charmap CntrlCharmap() {
>+  return Charmap::Range(0, 0x7f) & ~PrintCharmap();
>+}
>+constexpr Charmap BlankCharmap() { return Charmap::FromString("\t "); }
>+constexpr Charmap GraphCharmap() { return PrintCharmap() & ~SpaceCharmap(); }
>+constexpr Charmap PunctCharmap() { return GraphCharmap() & ~AlnumCharmap(); }
>+
>+}  // namespace strings_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_STRINGS_INTERNAL_CHAR_MAP_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/char_map_benchmark.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/char_map_benchmark.cc
>new file mode 100644
>index 00000000000..c45f3157c29
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/char_map_benchmark.cc
>@@ -0,0 +1,61 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/strings/internal/char_map.h"
>+
>+#include <cstdint>
>+
>+#include "benchmark/benchmark.h"
>+
>+namespace {
>+
>+absl::strings_internal::Charmap MakeBenchmarkMap() {
>+  absl::strings_internal::Charmap m;
>+  uint32_t x[] = {0x0, 0x1, 0x2, 0x3, 0xf, 0xe, 0xd, 0xc};
>+  for (uint32_t& t : x) t *= static_cast<uint32_t>(0x11111111UL);
>+  for (uint32_t i = 0; i < 256; ++i) {
>+    if ((x[i / 32] >> (i % 32)) & 1)
>+      m = m | absl::strings_internal::Charmap::Char(i);
>+  }
>+  return m;
>+}
>+
>+// Micro-benchmark for Charmap::contains.
>+void BM_Contains(benchmark::State& state) {
>+  // Loop-body replicated 10 times to increase time per iteration.
>+  // Argument continuously changed to avoid generating common subexpressions.
>+  const absl::strings_internal::Charmap benchmark_map = MakeBenchmarkMap();
>+  unsigned char c = 0;
>+  int ops = 0;
>+  for (auto _ : state) {
>+    ops += benchmark_map.contains(c++);
>+    ops += benchmark_map.contains(c++);
>+    ops += benchmark_map.contains(c++);
>+    ops += benchmark_map.contains(c++);
>+    ops += benchmark_map.contains(c++);
>+    ops += benchmark_map.contains(c++);
>+    ops += benchmark_map.contains(c++);
>+    ops += benchmark_map.contains(c++);
>+    ops += benchmark_map.contains(c++);
>+    ops += benchmark_map.contains(c++);
>+  }
>+  benchmark::DoNotOptimize(ops);
>+}
>+BENCHMARK(BM_Contains);
>+
>+// We don't bother benchmarking Charmap::IsZero or Charmap::IntersectsWith;
>+// their running time is data-dependent and it is not worth characterizing
>+// "typical" data.
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/char_map_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/char_map_test.cc
>new file mode 100644
>index 00000000000..c3601e101e2
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/char_map_test.cc
>@@ -0,0 +1,172 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/strings/internal/char_map.h"
>+
>+#include <cctype>
>+#include <string>
>+#include <vector>
>+
>+#include "gmock/gmock.h"
>+#include "gtest/gtest.h"
>+
>+namespace {
>+
>+constexpr absl::strings_internal::Charmap everything_map =
>+    ~absl::strings_internal::Charmap();
>+constexpr absl::strings_internal::Charmap nothing_map{};
>+
>+TEST(Charmap, AllTests) {
>+  const absl::strings_internal::Charmap also_nothing_map("", 0);
>+  ASSERT_TRUE(everything_map.contains('\0'));
>+  ASSERT_TRUE(!nothing_map.contains('\0'));
>+  ASSERT_TRUE(!also_nothing_map.contains('\0'));
>+  for (unsigned char ch = 1; ch != 0; ++ch) {
>+    ASSERT_TRUE(everything_map.contains(ch));
>+    ASSERT_TRUE(!nothing_map.contains(ch));
>+    ASSERT_TRUE(!also_nothing_map.contains(ch));
>+  }
>+
>+  const absl::strings_internal::Charmap symbols("&@#@^!@?", 5);
>+  ASSERT_TRUE(symbols.contains('&'));
>+  ASSERT_TRUE(symbols.contains('@'));
>+  ASSERT_TRUE(symbols.contains('#'));
>+  ASSERT_TRUE(symbols.contains('^'));
>+  ASSERT_TRUE(!symbols.contains('!'));
>+  ASSERT_TRUE(!symbols.contains('?'));
>+  int cnt = 0;
>+  for (unsigned char ch = 1; ch != 0; ++ch)
>+    cnt += symbols.contains(ch);
>+  ASSERT_EQ(cnt, 4);
>+
>+  const absl::strings_internal::Charmap lets("^abcde", 3);
>+  const absl::strings_internal::Charmap lets2("fghij\0klmnop", 10);
>+  const absl::strings_internal::Charmap lets3("fghij\0klmnop");
>+  ASSERT_TRUE(lets2.contains('k'));
>+  ASSERT_TRUE(!lets3.contains('k'));
>+
>+  ASSERT_TRUE(symbols.IntersectsWith(lets));
>+  ASSERT_TRUE(!lets2.IntersectsWith(lets));
>+  ASSERT_TRUE(lets.IntersectsWith(symbols));
>+  ASSERT_TRUE(!lets.IntersectsWith(lets2));
>+
>+  ASSERT_TRUE(nothing_map.IsZero());
>+  ASSERT_TRUE(!lets.IsZero());
>+}
>+
>+namespace {
>+std::string Members(const absl::strings_internal::Charmap& m) {
>+  std::string r;
>+  for (size_t i = 0; i < 256; ++i)
>+    if (m.contains(i)) r.push_back(i);
>+  return r;
>+}
>+
>+std::string ClosedRangeString(unsigned char lo, unsigned char hi) {
>+  // Don't depend on lo<hi. Just increment until lo==hi.
>+  std::string s;
>+  while (true) {
>+    s.push_back(lo);
>+    if (lo == hi) break;
>+    ++lo;
>+  }
>+  return s;
>+}
>+
>+}  // namespace
>+
>+TEST(Charmap, Constexpr) {
>+  constexpr absl::strings_internal::Charmap kEmpty = nothing_map;
>+  EXPECT_THAT(Members(kEmpty), "");
>+  constexpr absl::strings_internal::Charmap kA =
>+      absl::strings_internal::Charmap::Char('A');
>+  EXPECT_THAT(Members(kA), "A");
>+  constexpr absl::strings_internal::Charmap kAZ =
>+      absl::strings_internal::Charmap::Range('A', 'Z');
>+  EXPECT_THAT(Members(kAZ), "ABCDEFGHIJKLMNOPQRSTUVWXYZ");
>+  constexpr absl::strings_internal::Charmap kIdentifier =
>+      absl::strings_internal::Charmap::Range('0', '9') |
>+      absl::strings_internal::Charmap::Range('A', 'Z') |
>+      absl::strings_internal::Charmap::Range('a', 'z') |
>+      absl::strings_internal::Charmap::Char('_');
>+  EXPECT_THAT(Members(kIdentifier),
>+              "0123456789"
>+              "ABCDEFGHIJKLMNOPQRSTUVWXYZ"
>+              "_"
>+              "abcdefghijklmnopqrstuvwxyz");
>+  constexpr absl::strings_internal::Charmap kAll = everything_map;
>+  for (size_t i = 0; i < 256; ++i) {
>+    EXPECT_TRUE(kAll.contains(i)) << i;
>+  }
>+  constexpr absl::strings_internal::Charmap kHello =
>+      absl::strings_internal::Charmap::FromString("Hello, world!");
>+  EXPECT_THAT(Members(kHello), " !,Hdelorw");
>+
>+  // test negation and intersection
>+  constexpr absl::strings_internal::Charmap kABC =
>+      absl::strings_internal::Charmap::Range('A', 'Z') &
>+      ~absl::strings_internal::Charmap::Range('D', 'Z');
>+  EXPECT_THAT(Members(kABC), "ABC");
>+}
>+
>+TEST(Charmap, Range) {
>+  // Exhaustive testing takes too long, so test some of the boundaries that
>+  // are perhaps going to cause trouble.
>+  std::vector<size_t> poi = {0,   1,   2,   3,   4,   7,   8,   9,  15,
>+                             16,  17,  30,  31,  32,  33,  63,  64, 65,
>+                             127, 128, 129, 223, 224, 225, 254, 255};
>+  for (auto lo = poi.begin(); lo != poi.end(); ++lo) {
>+    SCOPED_TRACE(*lo);
>+    for (auto hi = lo; hi != poi.end(); ++hi) {
>+      SCOPED_TRACE(*hi);
>+      EXPECT_THAT(Members(absl::strings_internal::Charmap::Range(*lo, *hi)),
>+                  ClosedRangeString(*lo, *hi));
>+    }
>+  }
>+}
>+
>+bool AsBool(int x) { return static_cast<bool>(x); }
>+
>+TEST(CharmapCtype, Match) {
>+  for (int c = 0; c < 256; ++c) {
>+    SCOPED_TRACE(c);
>+    SCOPED_TRACE(static_cast<char>(c));
>+    EXPECT_EQ(AsBool(std::isupper(c)),
>+              absl::strings_internal::UpperCharmap().contains(c));
>+    EXPECT_EQ(AsBool(std::islower(c)),
>+              absl::strings_internal::LowerCharmap().contains(c));
>+    EXPECT_EQ(AsBool(std::isdigit(c)),
>+              absl::strings_internal::DigitCharmap().contains(c));
>+    EXPECT_EQ(AsBool(std::isalpha(c)),
>+              absl::strings_internal::AlphaCharmap().contains(c));
>+    EXPECT_EQ(AsBool(std::isalnum(c)),
>+              absl::strings_internal::AlnumCharmap().contains(c));
>+    EXPECT_EQ(AsBool(std::isxdigit(c)),
>+              absl::strings_internal::XDigitCharmap().contains(c));
>+    EXPECT_EQ(AsBool(std::isprint(c)),
>+              absl::strings_internal::PrintCharmap().contains(c));
>+    EXPECT_EQ(AsBool(std::isspace(c)),
>+              absl::strings_internal::SpaceCharmap().contains(c));
>+    EXPECT_EQ(AsBool(std::iscntrl(c)),
>+              absl::strings_internal::CntrlCharmap().contains(c));
>+    EXPECT_EQ(AsBool(std::isblank(c)),
>+              absl::strings_internal::BlankCharmap().contains(c));
>+    EXPECT_EQ(AsBool(std::isgraph(c)),
>+              absl::strings_internal::GraphCharmap().contains(c));
>+    EXPECT_EQ(AsBool(std::ispunct(c)),
>+              absl::strings_internal::PunctCharmap().contains(c));
>+  }
>+}
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/charconv_bigint.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/charconv_bigint.cc
>new file mode 100644
>index 00000000000..3e7296e7068
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/charconv_bigint.cc
>@@ -0,0 +1,357 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/strings/internal/charconv_bigint.h"
>+
>+#include <algorithm>
>+#include <cassert>
>+#include <string>
>+
>+namespace absl {
>+namespace strings_internal {
>+
>+namespace {
>+
>+// Table containing some large powers of 5, for fast computation.
>+
>+// Constant step size for entries in the kLargePowersOfFive table.  Each entry
>+// is larger than the previous entry by a factor of 5**kLargePowerOfFiveStep
>+// (or 5**27).
>+//
>+// In other words, the Nth entry in the table is 5**(27*N).
>+//
>+// 5**27 is the largest power of 5 that fits in 64 bits.
>+constexpr int kLargePowerOfFiveStep = 27;
>+
>+// The largest legal index into the kLargePowersOfFive table.
>+//
>+// In other words, the largest precomputed power of 5 is 5**(27*20).
>+constexpr int kLargestPowerOfFiveIndex = 20;
>+
>+// Table of powers of (5**27), up to (5**27)**20 == 5**540.
>+//
>+// Used to generate large powers of 5 while limiting the number of repeated
>+// multiplications required.
>+//
>+// clang-format off
>+const uint32_t kLargePowersOfFive[] = {
>+// 5**27 (i=1), start=0, end=2
>+  0xfa10079dU, 0x6765c793U,
>+// 5**54 (i=2), start=2, end=6
>+  0x97d9f649U, 0x6664242dU, 0x29939b14U, 0x29c30f10U,
>+// 5**81 (i=3), start=6, end=12
>+  0xc4f809c5U, 0x7bf3f22aU, 0x67bdae34U, 0xad340517U, 0x369d1b5fU, 0x10de1593U,
>+// 5**108 (i=4), start=12, end=20
>+  0x92b260d1U, 0x9efff7c7U, 0x81de0ec6U, 0xaeba5d56U, 0x410664a4U, 0x4f40737aU,
>+  0x20d3846fU, 0x06d00f73U,
>+// 5**135 (i=5), start=20, end=30
>+  0xff1b172dU, 0x13a1d71cU, 0xefa07617U, 0x7f682d3dU, 0xff8c90c0U, 0x3f0131e7U,
>+  0x3fdcb9feU, 0x917b0177U, 0x16c407a7U, 0x02c06b9dU,
>+// 5**162 (i=6), start=30, end=42
>+  0x960f7199U, 0x056667ecU, 0xe07aefd8U, 0x80f2b9ccU, 0x8273f5e3U, 0xeb9a214aU,
>+  0x40b38005U, 0x0e477ad4U, 0x277d08e6U, 0xfa28b11eU, 0xd3f7d784U, 0x011c835bU,
>+// 5**189 (i=7), start=42, end=56
>+  0xf723d9d5U, 0x3282d3f3U, 0xe00857d1U, 0x69659d25U, 0x2cf117cfU, 0x24da6d07U,
>+  0x954d1417U, 0x3e5d8cedU, 0x7a8bb766U, 0xfd785ae6U, 0x645436d2U, 0x40c78b34U,
>+  0x94151217U, 0x0072e9f7U,
>+// 5**216 (i=8), start=56, end=72
>+  0x2b416aa1U, 0x7893c5a7U, 0xe37dc6d4U, 0x2bad2beaU, 0xf0fc846cU, 0x7575ae4bU,
>+  0x62587b14U, 0x83b67a34U, 0x02110cdbU, 0xf7992f55U, 0x00deb022U, 0xa4a23becU,
>+  0x8af5c5cdU, 0xb85b654fU, 0x818df38bU, 0x002e69d2U,
>+// 5**243 (i=9), start=72, end=90
>+  0x3518cbbdU, 0x20b0c15fU, 0x38756c2fU, 0xfb5dc3ddU, 0x22ad2d94U, 0xbf35a952U,
>+  0xa699192aU, 0x9a613326U, 0xad2a9cedU, 0xd7f48968U, 0xe87dfb54U, 0xc8f05db6U,
>+  0x5ef67531U, 0x31c1ab49U, 0xe202ac9fU, 0x9b2957b5U, 0xa143f6d3U, 0x0012bf07U,
>+// 5**270 (i=10), start=90, end=110
>+  0x8b971de9U, 0x21aba2e1U, 0x63944362U, 0x57172336U, 0xd9544225U, 0xfb534166U,
>+  0x08c563eeU, 0x14640ee2U, 0x24e40d31U, 0x02b06537U, 0x03887f14U, 0x0285e533U,
>+  0xb744ef26U, 0x8be3a6c4U, 0x266979b4U, 0x6761ece2U, 0xd9cb39e4U, 0xe67de319U,
>+  0x0d39e796U, 0x00079250U,
>+// 5**297 (i=11), start=110, end=132
>+  0x260eb6e5U, 0xf414a796U, 0xee1a7491U, 0xdb9368ebU, 0xf50c105bU, 0x59157750U,
>+  0x9ed2fb5cU, 0xf6e56d8bU, 0xeaee8d23U, 0x0f319f75U, 0x2aa134d6U, 0xac2908e9U,
>+  0xd4413298U, 0x02f02a55U, 0x989d5a7aU, 0x70dde184U, 0xba8040a7U, 0x03200981U,
>+  0xbe03b11cU, 0x3c1c2a18U, 0xd60427a1U, 0x00030ee0U,
>+// 5**324 (i=12), start=132, end=156
>+  0xce566d71U, 0xf1c4aa25U, 0x4e93ca53U, 0xa72283d0U, 0x551a73eaU, 0x3d0538e2U,
>+  0x8da4303fU, 0x6a58de60U, 0x0e660221U, 0x49cf61a6U, 0x8d058fc1U, 0xb9d1a14cU,
>+  0x4bab157dU, 0xc85c6932U, 0x518c8b9eU, 0x9b92b8d0U, 0x0d8a0e21U, 0xbd855df9U,
>+  0xb3ea59a1U, 0x8da29289U, 0x4584d506U, 0x3752d80fU, 0xb72569c6U, 0x00013c33U,
>+// 5**351 (i=13), start=156, end=182
>+  0x190f354dU, 0x83695cfeU, 0xe5a4d0c7U, 0xb60fb7e8U, 0xee5bbcc4U, 0xb922054cU,
>+  0xbb4f0d85U, 0x48394028U, 0x1d8957dbU, 0x0d7edb14U, 0x4ecc7587U, 0x505e9e02U,
>+  0x4c87f36bU, 0x99e66bd6U, 0x44b9ed35U, 0x753037d4U, 0xe5fe5f27U, 0x2742c203U,
>+  0x13b2ed2bU, 0xdc525d2cU, 0xe6fde59aU, 0x77ffb18fU, 0x13c5752cU, 0x08a84bccU,
>+  0x859a4940U, 0x00007fb6U,
>+// 5**378 (i=14), start=182, end=210
>+  0x4f98cb39U, 0xa60edbbcU, 0x83b5872eU, 0xa501acffU, 0x9cc76f78U, 0xbadd4c73U,
>+  0x43e989faU, 0xca7acf80U, 0x2e0c824fU, 0xb19f4ffcU, 0x092fd81cU, 0xe4eb645bU,
>+  0xa1ff84c2U, 0x8a5a83baU, 0xa8a1fae9U, 0x1db43609U, 0xb0fed50bU, 0x0dd7d2bdU,
>+  0x7d7accd8U, 0x91fa640fU, 0x37dcc6c5U, 0x1c417fd5U, 0xe4d462adU, 0xe8a43399U,
>+  0x131bf9a5U, 0x8df54d29U, 0x36547dc1U, 0x00003395U,
>+// 5**405 (i=15), start=210, end=240
>+  0x5bd330f5U, 0x77d21967U, 0x1ac481b7U, 0x6be2f7ceU, 0x7f4792a9U, 0xe84c2c52U,
>+  0x84592228U, 0x9dcaf829U, 0xdab44ce1U, 0x3d0c311bU, 0x532e297dU, 0x4704e8b4U,
>+  0x9cdc32beU, 0x41e64d9dU, 0x7717bea1U, 0xa824c00dU, 0x08f50b27U, 0x0f198d77U,
>+  0x49bbfdf0U, 0x025c6c69U, 0xd4e55cd3U, 0xf083602bU, 0xb9f0fecdU, 0xc0864aeaU,
>+  0x9cb98681U, 0xaaf620e9U, 0xacb6df30U, 0x4faafe66U, 0x8af13c3bU, 0x000014d5U,
>+// 5**432 (i=16), start=240, end=272
>+  0x682bb941U, 0x89a9f297U, 0xcba75d7bU, 0x404217b1U, 0xb4e519e9U, 0xa1bc162bU,
>+  0xf7f5910aU, 0x98715af5U, 0x2ff53e57U, 0xe3ef118cU, 0x490c4543U, 0xbc9b1734U,
>+  0x2affbe4dU, 0x4cedcb4cU, 0xfb14e99eU, 0x35e34212U, 0xece39c24U, 0x07673ab3U,
>+  0xe73115ddU, 0xd15d38e7U, 0x093eed3bU, 0xf8e7eac5U, 0x78a8cc80U, 0x25227aacU,
>+  0x3f590551U, 0x413da1cbU, 0xdf643a55U, 0xab65ad44U, 0xd70b23d7U, 0xc672cd76U,
>+  0x3364ea62U, 0x0000086aU,
>+// 5**459 (i=17), start=272, end=306
>+  0x22f163ddU, 0x23cf07acU, 0xbe2af6c2U, 0xf412f6f6U, 0xc3ff541eU, 0x6eeaf7deU,
>+  0xa47047e0U, 0x408cda92U, 0x0f0eeb08U, 0x56deba9dU, 0xcfc6b090U, 0x8bbbdf04U,
>+  0x3933cdb3U, 0x9e7bb67dU, 0x9f297035U, 0x38946244U, 0xee1d37bbU, 0xde898174U,
>+  0x63f3559dU, 0x705b72fbU, 0x138d27d9U, 0xf8603a78U, 0x735eec44U, 0xe30987d5U,
>+  0xc6d38070U, 0x9cfe548eU, 0x9ff01422U, 0x7c564aa8U, 0x91cc60baU, 0xcbc3565dU,
>+  0x7550a50bU, 0x6909aeadU, 0x13234c45U, 0x00000366U,
>+// 5**486 (i=18), start=306, end=342
>+  0x17954989U, 0x3a7d7709U, 0x98042de5U, 0xa9011443U, 0x45e723c2U, 0x269ffd6fU,
>+  0x58852a46U, 0xaaa1042aU, 0x2eee8153U, 0xb2b6c39eU, 0xaf845b65U, 0xf6c365d7U,
>+  0xe4cffb2bU, 0xc840e90cU, 0xabea8abbU, 0x5c58f8d2U, 0x5c19fa3aU, 0x4670910aU,
>+  0x4449f21cU, 0xefa645b3U, 0xcc427decU, 0x083c3d73U, 0x467cb413U, 0x6fe10ae4U,
>+  0x3caffc72U, 0x9f8da55eU, 0x5e5c8ea7U, 0x490594bbU, 0xf0871b0bU, 0xdd89816cU,
>+  0x8e931df8U, 0xe85ce1c9U, 0xcca090a5U, 0x575fa16bU, 0x6b9f106cU, 0x0000015fU,
>+// 5**513 (i=19), start=342, end=380
>+  0xee20d805U, 0x57bc3c07U, 0xcdea624eU, 0xd3f0f52dU, 0x9924b4f4U, 0xcf968640U,
>+  0x61d41962U, 0xe87fb464U, 0xeaaf51c7U, 0x564c8b60U, 0xccda4028U, 0x529428bbU,
>+  0x313a1fa8U, 0x96bd0f94U, 0x7a82ebaaU, 0xad99e7e9U, 0xf2668cd4U, 0xbe33a45eU,
>+  0xfd0db669U, 0x87ee369fU, 0xd3ec20edU, 0x9c4d7db7U, 0xdedcf0d8U, 0x7cd2ca64U,
>+  0xe25a6577U, 0x61003fd4U, 0xe56f54ccU, 0x10b7c748U, 0x40526e5eU, 0x7300ae87U,
>+  0x5c439261U, 0x2c0ff469U, 0xbf723f12U, 0xb2379b61U, 0xbf59b4f5U, 0xc91b1c3fU,
>+  0xf0046d27U, 0x0000008dU,
>+// 5**540 (i=20), start=380, end=420
>+  0x525c9e11U, 0xf4e0eb41U, 0xebb2895dU, 0x5da512f9U, 0x7d9b29d4U, 0x452f4edcU,
>+  0x0b90bc37U, 0x341777cbU, 0x63d269afU, 0x1da77929U, 0x0a5c1826U, 0x77991898U,
>+  0x5aeddf86U, 0xf853a877U, 0x538c31ccU, 0xe84896daU, 0xb7a0010bU, 0x17ef4de5U,
>+  0xa52a2adeU, 0x029fd81cU, 0x987ce701U, 0x27fefd77U, 0xdb46c66fU, 0x5d301900U,
>+  0x496998c0U, 0xbb6598b9U, 0x5eebb607U, 0xe547354aU, 0xdf4a2f7eU, 0xf06c4955U,
>+  0x96242ffaU, 0x1775fb27U, 0xbecc58ceU, 0xebf2a53bU, 0x3eaad82aU, 0xf41137baU,
>+  0x573e6fbaU, 0xfb4866b8U, 0x54002148U, 0x00000039U,
>+};
>+// clang-format on
>+
>+// Returns a pointer to the big integer data for (5**27)**i.  i must be
>+// between 1 and 20, inclusive.
>+const uint32_t* LargePowerOfFiveData(int i) {
>+  return kLargePowersOfFive + i * (i - 1);
>+}
>+
>+// Returns the size of the big integer data for (5**27)**i, in words.  i must be
>+// between 1 and 20, inclusive.
>+int LargePowerOfFiveSize(int i) { return 2 * i; }
>+}  // namespace
>+
>+const uint32_t kFiveToNth[14] = {
>+    1,     5,      25,      125,     625,      3125,      15625,
>+    78125, 390625, 1953125, 9765625, 48828125, 244140625, 1220703125,
>+};
>+
>+const uint32_t kTenToNth[10] = {
>+    1, 10, 100, 1000, 10000, 100000, 1000000, 10000000, 100000000, 1000000000,
>+};
>+
>+template <int max_words>
>+int BigUnsigned<max_words>::ReadFloatMantissa(const ParsedFloat& fp,
>+                                              int significant_digits) {
>+  SetToZero();
>+  assert(fp.type == FloatType::kNumber);
>+
>+  if (fp.subrange_begin == nullptr) {
>+    // We already exactly parsed the mantissa, so no more work is necessary.
>+    words_[0] = fp.mantissa & 0xffffffffu;
>+    words_[1] = fp.mantissa >> 32;
>+    if (words_[1]) {
>+      size_ = 2;
>+    } else if (words_[0]) {
>+      size_ = 1;
>+    }
>+    return fp.exponent;
>+  }
>+  int exponent_adjust =
>+      ReadDigits(fp.subrange_begin, fp.subrange_end, significant_digits);
>+  return fp.literal_exponent + exponent_adjust;
>+}
>+
>+template <int max_words>
>+int BigUnsigned<max_words>::ReadDigits(const char* begin, const char* end,
>+                                       int significant_digits) {
>+  assert(significant_digits <= Digits10() + 1);
>+  SetToZero();
>+
>+  bool after_decimal_point = false;
>+  // Discard any leading zeroes before the decimal point
>+  while (begin < end && *begin == '0') {
>+    ++begin;
>+  }
>+  int dropped_digits = 0;
>+  // Discard any trailing zeroes.  These may or may not be after the decimal
>+  // point.
>+  while (begin < end && *std::prev(end) == '0') {
>+    --end;
>+    ++dropped_digits;
>+  }
>+  if (begin < end && *std::prev(end) == '.') {
>+    // If the std::string ends in '.', either before or after dropping zeroes, then
>+    // drop the decimal point and look for more digits to drop.
>+    dropped_digits = 0;
>+    --end;
>+    while (begin < end && *std::prev(end) == '0') {
>+      --end;
>+      ++dropped_digits;
>+    }
>+  } else if (dropped_digits) {
>+    // We dropped digits, and aren't sure if they're before or after the decimal
>+    // point.  Figure that out now.
>+    const char* dp = std::find(begin, end, '.');
>+    if (dp != end) {
>+      // The dropped trailing digits were after the decimal point, so don't
>+      // count them.
>+      dropped_digits = 0;
>+    }
>+  }
>+  // Any non-fraction digits we dropped need to be accounted for in our exponent
>+  // adjustment.
>+  int exponent_adjust = dropped_digits;
>+
>+  uint32_t queued = 0;
>+  int digits_queued = 0;
>+  for (; begin != end && significant_digits > 0; ++begin) {
>+    if (*begin == '.') {
>+      after_decimal_point = true;
>+      continue;
>+    }
>+    if (after_decimal_point) {
>+      // For each fractional digit we emit in our parsed integer, adjust our
>+      // decimal exponent to compensate.
>+      --exponent_adjust;
>+    }
>+    int digit = (*begin - '0');
>+    --significant_digits;
>+    if (significant_digits == 0 && std::next(begin) != end &&
>+        (digit == 0 || digit == 5)) {
>+      // If this is the very last significant digit, but insignificant digits
>+      // remain, we know that the last of those remaining significant digits is
>+      // nonzero.  (If it wasn't, we would have stripped it before we got here.)
>+      // So if this final digit is a 0 or 5, adjust it upward by 1.
>+      //
>+      // This adjustment is what allows incredibly large mantissas ending in
>+      // 500000...000000000001 to correctly round up, rather than to nearest.
>+      ++digit;
>+    }
>+    queued = 10 * queued + digit;
>+    ++digits_queued;
>+    if (digits_queued == kMaxSmallPowerOfTen) {
>+      MultiplyBy(kTenToNth[kMaxSmallPowerOfTen]);
>+      AddWithCarry(0, queued);
>+      queued = digits_queued = 0;
>+    }
>+  }
>+  // Encode any remaining digits.
>+  if (digits_queued) {
>+    MultiplyBy(kTenToNth[digits_queued]);
>+    AddWithCarry(0, queued);
>+  }
>+
>+  // If any insignificant digits remain, we will drop them.  But if we have not
>+  // yet read the decimal point, then we have to adjust the exponent to account
>+  // for the dropped digits.
>+  if (begin < end && !after_decimal_point) {
>+    // This call to std::find will result in a pointer either to the decimal
>+    // point, or to the end of our buffer if there was none.
>+    //
>+    // Either way, [begin, decimal_point) will contain the set of dropped digits
>+    // that require an exponent adjustment.
>+    const char* decimal_point = std::find(begin, end, '.');
>+    exponent_adjust += (decimal_point - begin);
>+  }
>+  return exponent_adjust;
>+}
>+
>+template <int max_words>
>+/* static */ BigUnsigned<max_words> BigUnsigned<max_words>::FiveToTheNth(
>+    int n) {
>+  BigUnsigned answer(1u);
>+
>+  // Seed from the table of large powers, if possible.
>+  bool first_pass = true;
>+  while (n >= kLargePowerOfFiveStep) {
>+    int big_power =
>+        std::min(n / kLargePowerOfFiveStep, kLargestPowerOfFiveIndex);
>+    if (first_pass) {
>+      // just copy, rather than multiplying by 1
>+      std::copy(
>+          LargePowerOfFiveData(big_power),
>+          LargePowerOfFiveData(big_power) + LargePowerOfFiveSize(big_power),
>+          answer.words_);
>+      answer.size_ = LargePowerOfFiveSize(big_power);
>+      first_pass = false;
>+    } else {
>+      answer.MultiplyBy(LargePowerOfFiveSize(big_power),
>+                        LargePowerOfFiveData(big_power));
>+    }
>+    n -= kLargePowerOfFiveStep * big_power;
>+  }
>+  answer.MultiplyByFiveToTheNth(n);
>+  return answer;
>+}
>+
>+template <int max_words>
>+void BigUnsigned<max_words>::MultiplyStep(int original_size,
>+                                          const uint32_t* other_words,
>+                                          int other_size, int step) {
>+  int this_i = std::min(original_size - 1, step);
>+  int other_i = step - this_i;
>+
>+  uint64_t this_word = 0;
>+  uint64_t carry = 0;
>+  for (; this_i >= 0 && other_i < other_size; --this_i, ++other_i) {
>+    uint64_t product = words_[this_i];
>+    product *= other_words[other_i];
>+    this_word += product;
>+    carry += (this_word >> 32);
>+    this_word &= 0xffffffff;
>+  }
>+  AddWithCarry(step + 1, carry);
>+  words_[step] = this_word & 0xffffffff;
>+  if (this_word > 0 && size_ <= step) {
>+    size_ = step + 1;
>+  }
>+}
>+
>+template <int max_words>
>+std::string BigUnsigned<max_words>::ToString() const {
>+  BigUnsigned<max_words> copy = *this;
>+  std::string result;
>+  // Build result in reverse order
>+  while (copy.size() > 0) {
>+    int next_digit = copy.DivMod<10>();
>+    result.push_back('0' + next_digit);
>+  }
>+  if (result.empty()) {
>+    result.push_back('0');
>+  }
>+  std::reverse(result.begin(), result.end());
>+  return result;
>+}
>+
>+template class BigUnsigned<4>;
>+template class BigUnsigned<84>;
>+
>+}  // namespace strings_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/charconv_bigint.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/charconv_bigint.h
>new file mode 100644
>index 00000000000..aa70af2c289
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/charconv_bigint.h
>@@ -0,0 +1,426 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#ifndef ABSL_STRINGS_INTERNAL_CHARCONV_BIGINT_H_
>+#define ABSL_STRINGS_INTERNAL_CHARCONV_BIGINT_H_
>+
>+#include <algorithm>
>+#include <cstdint>
>+#include <iostream>
>+#include <string>
>+
>+#include "absl/strings/ascii.h"
>+#include "absl/strings/internal/charconv_parse.h"
>+#include "absl/strings/string_view.h"
>+
>+namespace absl {
>+namespace strings_internal {
>+
>+// The largest power that 5 that can be raised to, and still fit in a uint32_t.
>+constexpr int kMaxSmallPowerOfFive = 13;
>+// The largest power that 10 that can be raised to, and still fit in a uint32_t.
>+constexpr int kMaxSmallPowerOfTen = 9;
>+
>+extern const uint32_t kFiveToNth[kMaxSmallPowerOfFive + 1];
>+extern const uint32_t kTenToNth[kMaxSmallPowerOfTen + 1];
>+
>+// Large, fixed-width unsigned integer.
>+//
>+// Exact rounding for decimal-to-binary floating point conversion requires very
>+// large integer math, but a design goal of absl::from_chars is to avoid
>+// allocating memory.  The integer precision needed for decimal-to-binary
>+// conversions is large but bounded, so a huge fixed-width integer class
>+// suffices.
>+//
>+// This is an intentionally limited big integer class.  Only needed operations
>+// are implemented.  All storage lives in an array data member, and all
>+// arithmetic is done in-place, to avoid requiring separate storage for operand
>+// and result.
>+//
>+// This is an internal class.  Some methods live in the .cc file, and are
>+// instantiated only for the values of max_words we need.
>+template <int max_words>
>+class BigUnsigned {
>+ public:
>+  static_assert(max_words == 4 || max_words == 84,
>+                "unsupported max_words value");
>+
>+  BigUnsigned() : size_(0), words_{} {}
>+  explicit BigUnsigned(uint32_t v) : size_(v > 0 ? 1 : 0), words_{v} {}
>+  explicit BigUnsigned(uint64_t v)
>+      : size_(0),
>+        words_{static_cast<uint32_t>(v & 0xffffffff),
>+               static_cast<uint32_t>(v >> 32)} {
>+    if (words_[1]) {
>+      size_ = 2;
>+    } else if (words_[0]) {
>+      size_ = 1;
>+    }
>+  }
>+
>+  // Constructs a BigUnsigned from the given string_view containing a decimal
>+  // value.  If the input std::string is not a decimal integer, constructs a 0
>+  // instead.
>+  explicit BigUnsigned(absl::string_view sv) : size_(0), words_{} {
>+    // Check for valid input, returning a 0 otherwise.  This is reasonable
>+    // behavior only because this constructor is for unit tests.
>+    if (std::find_if_not(sv.begin(), sv.end(), ascii_isdigit) != sv.end() ||
>+        sv.empty()) {
>+      return;
>+    }
>+    int exponent_adjust =
>+        ReadDigits(sv.data(), sv.data() + sv.size(), Digits10() + 1);
>+    if (exponent_adjust > 0) {
>+      MultiplyByTenToTheNth(exponent_adjust);
>+    }
>+  }
>+
>+  // Loads the mantissa value of a previously-parsed float.
>+  //
>+  // Returns the associated decimal exponent.  The value of the parsed float is
>+  // exactly *this * 10**exponent.
>+  int ReadFloatMantissa(const ParsedFloat& fp, int significant_digits);
>+
>+  // Returns the number of decimal digits of precision this type provides.  All
>+  // numbers with this many decimal digits or fewer are representable by this
>+  // type.
>+  //
>+  // Analagous to std::numeric_limits<BigUnsigned>::digits10.
>+  static constexpr int Digits10() {
>+    // 9975007/1035508 is very slightly less than log10(2**32).
>+    return static_cast<uint64_t>(max_words) * 9975007 / 1035508;
>+  }
>+
>+  // Shifts left by the given number of bits.
>+  void ShiftLeft(int count) {
>+    if (count > 0) {
>+      const int word_shift = count / 32;
>+      if (word_shift >= max_words) {
>+        SetToZero();
>+        return;
>+      }
>+      size_ = std::min(size_ + word_shift, max_words);
>+      count %= 32;
>+      if (count == 0) {
>+        std::copy_backward(words_, words_ + size_ - word_shift, words_ + size_);
>+      } else {
>+        for (int i = std::min(size_, max_words - 1); i > word_shift; --i) {
>+          words_[i] = (words_[i - word_shift] << count) |
>+                      (words_[i - word_shift - 1] >> (32 - count));
>+        }
>+        words_[word_shift] = words_[0] << count;
>+        // Grow size_ if necessary.
>+        if (size_ < max_words && words_[size_]) {
>+          ++size_;
>+        }
>+      }
>+      std::fill(words_, words_ + word_shift, 0u);
>+    }
>+  }
>+
>+
>+  // Multiplies by v in-place.
>+  void MultiplyBy(uint32_t v) {
>+    if (size_ == 0 || v == 1) {
>+      return;
>+    }
>+    if (v == 0) {
>+      SetToZero();
>+      return;
>+    }
>+    const uint64_t factor = v;
>+    uint64_t window = 0;
>+    for (int i = 0; i < size_; ++i) {
>+      window += factor * words_[i];
>+      words_[i] = window & 0xffffffff;
>+      window >>= 32;
>+    }
>+    // If carry bits remain and there's space for them, grow size_.
>+    if (window && size_ < max_words) {
>+      words_[size_] = window & 0xffffffff;
>+      ++size_;
>+    }
>+  }
>+
>+  void MultiplyBy(uint64_t v) {
>+    uint32_t words[2];
>+    words[0] = static_cast<uint32_t>(v);
>+    words[1] = static_cast<uint32_t>(v >> 32);
>+    if (words[1] == 0) {
>+      MultiplyBy(words[0]);
>+    } else {
>+      MultiplyBy(2, words);
>+    }
>+  }
>+
>+  // Multiplies in place by 5 to the power of n.  n must be non-negative.
>+  void MultiplyByFiveToTheNth(int n) {
>+    while (n >= kMaxSmallPowerOfFive) {
>+      MultiplyBy(kFiveToNth[kMaxSmallPowerOfFive]);
>+      n -= kMaxSmallPowerOfFive;
>+    }
>+    if (n > 0) {
>+      MultiplyBy(kFiveToNth[n]);
>+    }
>+  }
>+
>+  // Multiplies in place by 10 to the power of n.  n must be non-negative.
>+  void MultiplyByTenToTheNth(int n) {
>+    if (n > kMaxSmallPowerOfTen) {
>+      // For large n, raise to a power of 5, then shift left by the same amount.
>+      // (10**n == 5**n * 2**n.)  This requires fewer multiplications overall.
>+      MultiplyByFiveToTheNth(n);
>+      ShiftLeft(n);
>+    } else if (n > 0) {
>+      // We can do this more quickly for very small N by using a single
>+      // multiplication.
>+      MultiplyBy(kTenToNth[n]);
>+    }
>+  }
>+
>+  // Returns the value of 5**n, for non-negative n.  This implementation uses
>+  // a lookup table, and is faster then seeding a BigUnsigned with 1 and calling
>+  // MultiplyByFiveToTheNth().
>+  static BigUnsigned FiveToTheNth(int n);
>+
>+  // Multiplies by another BigUnsigned, in-place.
>+  template <int M>
>+  void MultiplyBy(const BigUnsigned<M>& other) {
>+    MultiplyBy(other.size(), other.words());
>+  }
>+
>+  void SetToZero() {
>+    std::fill(words_, words_ + size_, 0u);
>+    size_ = 0;
>+  }
>+
>+  // Returns the value of the nth word of this BigUnsigned.  This is
>+  // range-checked, and returns 0 on out-of-bounds accesses.
>+  uint32_t GetWord(int index) const {
>+    if (index < 0 || index >= size_) {
>+      return 0;
>+    }
>+    return words_[index];
>+  }
>+
>+  // Returns this integer as a decimal std::string.  This is not used in the decimal-
>+  // to-binary conversion; it is intended to aid in testing.
>+  std::string ToString() const;
>+
>+  int size() const { return size_; }
>+  const uint32_t* words() const { return words_; }
>+
>+ private:
>+  // Reads the number between [begin, end), possibly containing a decimal point,
>+  // into this BigUnsigned.
>+  //
>+  // Callers are required to ensure [begin, end) contains a valid number, with
>+  // one or more decimal digits and at most one decimal point.  This routine
>+  // will behave unpredictably if these preconditions are not met.
>+  //
>+  // Only the first `significant_digits` digits are read.  Digits beyond this
>+  // limit are "sticky": If the final significant digit is 0 or 5, and if any
>+  // dropped digit is nonzero, then that final significant digit is adjusted up
>+  // to 1 or 6.  This adjustment allows for precise rounding.
>+  //
>+  // Returns `exponent_adjustment`, a power-of-ten exponent adjustment to
>+  // account for the decimal point and for dropped significant digits.  After
>+  // this function returns,
>+  //   actual_value_of_parsed_string ~= *this * 10**exponent_adjustment.
>+  int ReadDigits(const char* begin, const char* end, int significant_digits);
>+
>+  // Performs a step of big integer multiplication.  This computes the full
>+  // (64-bit-wide) values that should be added at the given index (step), and
>+  // adds to that location in-place.
>+  //
>+  // Because our math all occurs in place, we must multiply starting from the
>+  // highest word working downward.  (This is a bit more expensive due to the
>+  // extra carries involved.)
>+  //
>+  // This must be called in steps, for each word to be calculated, starting from
>+  // the high end and working down to 0.  The first value of `step` should be
>+  //   `std::min(original_size + other.size_ - 2, max_words - 1)`.
>+  // The reason for this expression is that multiplying the i'th word from one
>+  // multiplicand and the j'th word of another multiplicand creates a
>+  // two-word-wide value to be stored at the (i+j)'th element.  The highest
>+  // word indices we will access are `original_size - 1` from this object, and
>+  // `other.size_ - 1` from our operand.  Therefore,
>+  // `original_size + other.size_ - 2` is the first step we should calculate,
>+  // but limited on an upper bound by max_words.
>+
>+  // Working from high-to-low ensures that we do not overwrite the portions of
>+  // the initial value of *this which are still needed for later steps.
>+  //
>+  // Once called with step == 0, *this contains the result of the
>+  // multiplication.
>+  //
>+  // `original_size` is the size_ of *this before the first call to
>+  // MultiplyStep().  `other_words` and `other_size` are the contents of our
>+  // operand.  `step` is the step to perform, as described above.
>+  void MultiplyStep(int original_size, const uint32_t* other_words,
>+                    int other_size, int step);
>+
>+  void MultiplyBy(int other_size, const uint32_t* other_words) {
>+    const int original_size = size_;
>+    const int first_step =
>+        std::min(original_size + other_size - 2, max_words - 1);
>+    for (int step = first_step; step >= 0; --step) {
>+      MultiplyStep(original_size, other_words, other_size, step);
>+    }
>+  }
>+
>+  // Adds a 32-bit value to the index'th word, with carry.
>+  void AddWithCarry(int index, uint32_t value) {
>+    if (value) {
>+      while (index < max_words && value > 0) {
>+        words_[index] += value;
>+        // carry if we overflowed in this word:
>+        if (value > words_[index]) {
>+          value = 1;
>+          ++index;
>+        } else {
>+          value = 0;
>+        }
>+      }
>+      size_ = std::min(max_words, std::max(index + 1, size_));
>+    }
>+  }
>+
>+  void AddWithCarry(int index, uint64_t value) {
>+    if (value && index < max_words) {
>+      uint32_t high = value >> 32;
>+      uint32_t low = value & 0xffffffff;
>+      words_[index] += low;
>+      if (words_[index] < low) {
>+        ++high;
>+        if (high == 0) {
>+          // Carry from the low word caused our high word to overflow.
>+          // Short circuit here to do the right thing.
>+          AddWithCarry(index + 2, static_cast<uint32_t>(1));
>+          return;
>+        }
>+      }
>+      if (high > 0) {
>+        AddWithCarry(index + 1, high);
>+      } else {
>+        // Normally 32-bit AddWithCarry() sets size_, but since we don't call
>+        // it when `high` is 0, do it ourselves here.
>+        size_ = std::min(max_words, std::max(index + 1, size_));
>+      }
>+    }
>+  }
>+
>+  // Divide this in place by a constant divisor.  Returns the remainder of the
>+  // division.
>+  template <uint32_t divisor>
>+  uint32_t DivMod() {
>+    uint64_t accumulator = 0;
>+    for (int i = size_ - 1; i >= 0; --i) {
>+      accumulator <<= 32;
>+      accumulator += words_[i];
>+      // accumulator / divisor will never overflow an int32_t in this loop
>+      words_[i] = static_cast<uint32_t>(accumulator / divisor);
>+      accumulator = accumulator % divisor;
>+    }
>+    while (size_ > 0 && words_[size_ - 1] == 0) {
>+      --size_;
>+    }
>+    return static_cast<uint32_t>(accumulator);
>+  }
>+
>+  // The number of elements in words_ that may carry significant values.
>+  // All elements beyond this point are 0.
>+  //
>+  // When size_ is 0, this BigUnsigned stores the value 0.
>+  // When size_ is nonzero, is *not* guaranteed that words_[size_ - 1] is
>+  // nonzero.  This can occur due to overflow truncation.
>+  // In particular, x.size_ != y.size_ does *not* imply x != y.
>+  int size_;
>+  uint32_t words_[max_words];
>+};
>+
>+// Compares two big integer instances.
>+//
>+// Returns -1 if lhs < rhs, 0 if lhs == rhs, and 1 if lhs > rhs.
>+template <int N, int M>
>+int Compare(const BigUnsigned<N>& lhs, const BigUnsigned<M>& rhs) {
>+  int limit = std::max(lhs.size(), rhs.size());
>+  for (int i = limit - 1; i >= 0; --i) {
>+    const uint32_t lhs_word = lhs.GetWord(i);
>+    const uint32_t rhs_word = rhs.GetWord(i);
>+    if (lhs_word < rhs_word) {
>+      return -1;
>+    } else if (lhs_word > rhs_word) {
>+      return 1;
>+    }
>+  }
>+  return 0;
>+}
>+
>+template <int N, int M>
>+bool operator==(const BigUnsigned<N>& lhs, const BigUnsigned<M>& rhs) {
>+  int limit = std::max(lhs.size(), rhs.size());
>+  for (int i = 0; i < limit; ++i) {
>+    if (lhs.GetWord(i) != rhs.GetWord(i)) {
>+      return false;
>+    }
>+  }
>+  return true;
>+}
>+
>+template <int N, int M>
>+bool operator!=(const BigUnsigned<N>& lhs, const BigUnsigned<M>& rhs) {
>+  return !(lhs == rhs);
>+}
>+
>+template <int N, int M>
>+bool operator<(const BigUnsigned<N>& lhs, const BigUnsigned<M>& rhs) {
>+  return Compare(lhs, rhs) == -1;
>+}
>+
>+template <int N, int M>
>+bool operator>(const BigUnsigned<N>& lhs, const BigUnsigned<M>& rhs) {
>+  return rhs < lhs;
>+}
>+template <int N, int M>
>+bool operator<=(const BigUnsigned<N>& lhs, const BigUnsigned<M>& rhs) {
>+  return !(rhs < lhs);
>+}
>+template <int N, int M>
>+bool operator>=(const BigUnsigned<N>& lhs, const BigUnsigned<M>& rhs) {
>+  return !(lhs < rhs);
>+}
>+
>+// Output operator for BigUnsigned, for testing purposes only.
>+template <int N>
>+std::ostream& operator<<(std::ostream& os, const BigUnsigned<N>& num) {
>+  return os << num.ToString();
>+}
>+
>+// Explicit instantiation declarations for the sizes of BigUnsigned that we
>+// are using.
>+//
>+// For now, the choices of 4 and 84 are arbitrary; 4 is a small value that is
>+// still bigger than an int128, and 84 is a large value we will want to use
>+// in the from_chars implementation.
>+//
>+// Comments justifying the use of 84 belong in the from_chars implementation,
>+// and will be added in a follow-up CL.
>+extern template class BigUnsigned<4>;
>+extern template class BigUnsigned<84>;
>+
>+}  // namespace strings_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_STRINGS_INTERNAL_CHARCONV_BIGINT_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/charconv_bigint_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/charconv_bigint_test.cc
>new file mode 100644
>index 00000000000..9b6357888fb
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/charconv_bigint_test.cc
>@@ -0,0 +1,203 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/strings/internal/charconv_bigint.h"
>+
>+#include <string>
>+
>+#include "gtest/gtest.h"
>+
>+namespace absl {
>+namespace strings_internal {
>+
>+TEST(BigUnsigned, ShiftLeft) {
>+  {
>+    // Check that 3 * 2**100 is calculated correctly
>+    BigUnsigned<4> num(3u);
>+    num.ShiftLeft(100);
>+    EXPECT_EQ(num, BigUnsigned<4>("3802951800684688204490109616128"));
>+  }
>+  {
>+    // Test that overflow is truncated properly.
>+    // 15 is 4 bits long, and BigUnsigned<4> is a 128-bit bigint.
>+    // Shifting left by 125 bits should truncate off the high bit, so that
>+    //   15 << 125 == 7 << 125
>+    // after truncation.
>+    BigUnsigned<4> a(15u);
>+    BigUnsigned<4> b(7u);
>+    BigUnsigned<4> c(3u);
>+    a.ShiftLeft(125);
>+    b.ShiftLeft(125);
>+    c.ShiftLeft(125);
>+    EXPECT_EQ(a, b);
>+    EXPECT_NE(a, c);
>+  }
>+  {
>+    // Same test, larger bigint:
>+    BigUnsigned<84> a(15u);
>+    BigUnsigned<84> b(7u);
>+    BigUnsigned<84> c(3u);
>+    a.ShiftLeft(84 * 32 - 3);
>+    b.ShiftLeft(84 * 32 - 3);
>+    c.ShiftLeft(84 * 32 - 3);
>+    EXPECT_EQ(a, b);
>+    EXPECT_NE(a, c);
>+  }
>+  {
>+    // Check that incrementally shifting has the same result as doing it all at
>+    // once (attempting to capture corner cases.)
>+    const std::string seed = "1234567890123456789012345678901234567890";
>+    BigUnsigned<84> a(seed);
>+    for (int i = 1; i <= 84 * 32; ++i) {
>+      a.ShiftLeft(1);
>+      BigUnsigned<84> b(seed);
>+      b.ShiftLeft(i);
>+      EXPECT_EQ(a, b);
>+    }
>+    // And we should have fully rotated all bits off by now:
>+    EXPECT_EQ(a, BigUnsigned<84>(0u));
>+  }
>+}
>+
>+TEST(BigUnsigned, MultiplyByUint32) {
>+  const BigUnsigned<84> factorial_100(
>+      "933262154439441526816992388562667004907159682643816214685929638952175999"
>+      "932299156089414639761565182862536979208272237582511852109168640000000000"
>+      "00000000000000");
>+  BigUnsigned<84> a(1u);
>+  for (uint32_t i = 1; i <= 100; ++i) {
>+    a.MultiplyBy(i);
>+  }
>+  EXPECT_EQ(a, BigUnsigned<84>(factorial_100));
>+}
>+
>+TEST(BigUnsigned, MultiplyByBigUnsigned) {
>+  {
>+    // Put the terms of factorial_200 into two bigints, and multiply them
>+    // together.
>+    const BigUnsigned<84> factorial_200(
>+        "7886578673647905035523632139321850622951359776871732632947425332443594"
>+        "4996340334292030428401198462390417721213891963883025764279024263710506"
>+        "1926624952829931113462857270763317237396988943922445621451664240254033"
>+        "2918641312274282948532775242424075739032403212574055795686602260319041"
>+        "7032406235170085879617892222278962370389737472000000000000000000000000"
>+        "0000000000000000000000000");
>+    BigUnsigned<84> evens(1u);
>+    BigUnsigned<84> odds(1u);
>+    for (uint32_t i = 1; i < 200; i += 2) {
>+      odds.MultiplyBy(i);
>+      evens.MultiplyBy(i + 1);
>+    }
>+    evens.MultiplyBy(odds);
>+    EXPECT_EQ(evens, factorial_200);
>+  }
>+  {
>+    // Multiply various powers of 10 together.
>+    for (int a = 0 ; a < 700; a += 25) {
>+      SCOPED_TRACE(a);
>+      BigUnsigned<84> a_value("3" + std::string(a, '0'));
>+      for (int b = 0; b < (700 - a); b += 25) {
>+        SCOPED_TRACE(b);
>+        BigUnsigned<84> b_value("2" + std::string(b, '0'));
>+        BigUnsigned<84> expected_product("6" + std::string(a + b, '0'));
>+        b_value.MultiplyBy(a_value);
>+        EXPECT_EQ(b_value, expected_product);
>+      }
>+    }
>+  }
>+}
>+
>+TEST(BigUnsigned, MultiplyByOverflow) {
>+  {
>+    // Check that multiplcation overflow predictably truncates.
>+
>+    // A big int with all bits on.
>+    BigUnsigned<4> all_bits_on("340282366920938463463374607431768211455");
>+    // Modulo 2**128, this is equal to -1.  Therefore the square of this,
>+    // modulo 2**128, should be 1.
>+    all_bits_on.MultiplyBy(all_bits_on);
>+    EXPECT_EQ(all_bits_on, BigUnsigned<4>(1u));
>+  }
>+  {
>+    // Try multiplying a large bigint by 2**50, and compare the result to
>+    // shifting.
>+    BigUnsigned<4> value_1("12345678901234567890123456789012345678");
>+    BigUnsigned<4> value_2("12345678901234567890123456789012345678");
>+    BigUnsigned<4> two_to_fiftieth(1u);
>+    two_to_fiftieth.ShiftLeft(50);
>+
>+    value_1.ShiftLeft(50);
>+    value_2.MultiplyBy(two_to_fiftieth);
>+    EXPECT_EQ(value_1, value_2);
>+  }
>+}
>+
>+TEST(BigUnsigned, FiveToTheNth) {
>+  {
>+    // Sanity check that MultiplyByFiveToTheNth gives consistent answers, up to
>+    // and including overflow.
>+    for (int i = 0; i < 1160; ++i) {
>+      SCOPED_TRACE(i);
>+      BigUnsigned<84> value_1(123u);
>+      BigUnsigned<84> value_2(123u);
>+      value_1.MultiplyByFiveToTheNth(i);
>+      for (int j = 0; j < i; j++) {
>+        value_2.MultiplyBy(5u);
>+      }
>+      EXPECT_EQ(value_1, value_2);
>+    }
>+  }
>+  {
>+    // Check that the faster, table-lookup-based static method returns the same
>+    // result that multiplying in-place would return, up to and including
>+    // overflow.
>+    for (int i = 0; i < 1160; ++i) {
>+      SCOPED_TRACE(i);
>+      BigUnsigned<84> value_1(1u);
>+      value_1.MultiplyByFiveToTheNth(i);
>+      BigUnsigned<84> value_2 = BigUnsigned<84>::FiveToTheNth(i);
>+      EXPECT_EQ(value_1, value_2);
>+    }
>+  }
>+}
>+
>+TEST(BigUnsigned, TenToTheNth) {
>+  {
>+    // Sanity check MultiplyByTenToTheNth.
>+    for (int i = 0; i < 800; ++i) {
>+      SCOPED_TRACE(i);
>+      BigUnsigned<84> value_1(123u);
>+      BigUnsigned<84> value_2(123u);
>+      value_1.MultiplyByTenToTheNth(i);
>+      for (int j = 0; j < i; j++) {
>+        value_2.MultiplyBy(10u);
>+      }
>+      EXPECT_EQ(value_1, value_2);
>+    }
>+  }
>+  {
>+    // Alternate testing approach, taking advantage of the decimal parser.
>+    for (int i = 0; i < 200; ++i) {
>+      SCOPED_TRACE(i);
>+      BigUnsigned<84> value_1(135u);
>+      value_1.MultiplyByTenToTheNth(i);
>+      BigUnsigned<84> value_2("135" + std::string(i, '0'));
>+      EXPECT_EQ(value_1, value_2);
>+    }
>+  }
>+}
>+
>+
>+}  // namespace strings_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/charconv_parse.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/charconv_parse.cc
>new file mode 100644
>index 00000000000..a04cc67669a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/charconv_parse.cc
>@@ -0,0 +1,496 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/strings/internal/charconv_parse.h"
>+#include "absl/strings/charconv.h"
>+
>+#include <cassert>
>+#include <cstdint>
>+#include <limits>
>+
>+#include "absl/strings/internal/memutil.h"
>+
>+namespace absl {
>+namespace {
>+
>+// ParseFloat<10> will read the first 19 significant digits of the mantissa.
>+// This number was chosen for multiple reasons.
>+//
>+// (a) First, for whatever integer type we choose to represent the mantissa, we
>+// want to choose the largest possible number of decimal digits for that integer
>+// type.  We are using uint64_t, which can express any 19-digit unsigned
>+// integer.
>+//
>+// (b) Second, we need to parse enough digits that the binary value of any
>+// mantissa we capture has more bits of resolution than the mantissa
>+// representation in the target float.  Our algorithm requires at least 3 bits
>+// of headway, but 19 decimal digits give a little more than that.
>+//
>+// The following static assertions verify the above comments:
>+constexpr int kDecimalMantissaDigitsMax = 19;
>+
>+static_assert(std::numeric_limits<uint64_t>::digits10 ==
>+                  kDecimalMantissaDigitsMax,
>+              "(a) above");
>+
>+// IEEE doubles, which we assume in Abseil, have 53 binary bits of mantissa.
>+static_assert(std::numeric_limits<double>::is_iec559, "IEEE double assumed");
>+static_assert(std::numeric_limits<double>::radix == 2, "IEEE double fact");
>+static_assert(std::numeric_limits<double>::digits == 53, "IEEE double fact");
>+
>+// The lowest valued 19-digit decimal mantissa we can read still contains
>+// sufficient information to reconstruct a binary mantissa.
>+static_assert(1000000000000000000u > (uint64_t(1) << (53 + 3)), "(b) above");
>+
>+// ParseFloat<16> will read the first 15 significant digits of the mantissa.
>+//
>+// Because a base-16-to-base-2 conversion can be done exactly, we do not need
>+// to maximize the number of scanned hex digits to improve our conversion.  What
>+// is required is to scan two more bits than the mantissa can represent, so that
>+// we always round correctly.
>+//
>+// (One extra bit does not suffice to perform correct rounding, since a number
>+// exactly halfway between two representable floats has unique rounding rules,
>+// so we need to differentiate between a "halfway between" number and a "closer
>+// to the larger value" number.)
>+constexpr int kHexadecimalMantissaDigitsMax = 15;
>+
>+// The minimum number of significant bits that will be read from
>+// kHexadecimalMantissaDigitsMax hex digits.  We must subtract by three, since
>+// the most significant digit can be a "1", which only contributes a single
>+// significant bit.
>+constexpr int kGuaranteedHexadecimalMantissaBitPrecision =
>+    4 * kHexadecimalMantissaDigitsMax - 3;
>+
>+static_assert(kGuaranteedHexadecimalMantissaBitPrecision >
>+                  std::numeric_limits<double>::digits + 2,
>+              "kHexadecimalMantissaDigitsMax too small");
>+
>+// We also impose a limit on the number of significant digits we will read from
>+// an exponent, to avoid having to deal with integer overflow.  We use 9 for
>+// this purpose.
>+//
>+// If we read a 9 digit exponent, the end result of the conversion will
>+// necessarily be infinity or zero, depending on the sign of the exponent.
>+// Therefore we can just drop extra digits on the floor without any extra
>+// logic.
>+constexpr int kDecimalExponentDigitsMax = 9;
>+static_assert(std::numeric_limits<int>::digits10 >= kDecimalExponentDigitsMax,
>+              "int type too small");
>+
>+// To avoid incredibly large inputs causing integer overflow for our exponent,
>+// we impose an arbitrary but very large limit on the number of significant
>+// digits we will accept.  The implementation refuses to match a std::string with
>+// more consecutive significant mantissa digits than this.
>+constexpr int kDecimalDigitLimit = 50000000;
>+
>+// Corresponding limit for hexadecimal digit inputs.  This is one fourth the
>+// amount of kDecimalDigitLimit, since each dropped hexadecimal digit requires
>+// a binary exponent adjustment of 4.
>+constexpr int kHexadecimalDigitLimit = kDecimalDigitLimit / 4;
>+
>+// The largest exponent we can read is 999999999 (per
>+// kDecimalExponentDigitsMax), and the largest exponent adjustment we can get
>+// from dropped mantissa digits is 2 * kDecimalDigitLimit, and the sum of these
>+// comfortably fits in an integer.
>+//
>+// We count kDecimalDigitLimit twice because there are independent limits for
>+// numbers before and after the decimal point.  (In the case where there are no
>+// significant digits before the decimal point, there are independent limits for
>+// post-decimal-point leading zeroes and for significant digits.)
>+static_assert(999999999 + 2 * kDecimalDigitLimit <
>+                  std::numeric_limits<int>::max(),
>+              "int type too small");
>+static_assert(999999999 + 2 * (4 * kHexadecimalDigitLimit) <
>+                  std::numeric_limits<int>::max(),
>+              "int type too small");
>+
>+// Returns true if the provided bitfield allows parsing an exponent value
>+// (e.g., "1.5e100").
>+bool AllowExponent(chars_format flags) {
>+  bool fixed = (flags & chars_format::fixed) == chars_format::fixed;
>+  bool scientific =
>+      (flags & chars_format::scientific) == chars_format::scientific;
>+  return scientific || !fixed;
>+}
>+
>+// Returns true if the provided bitfield requires an exponent value be present.
>+bool RequireExponent(chars_format flags) {
>+  bool fixed = (flags & chars_format::fixed) == chars_format::fixed;
>+  bool scientific =
>+      (flags & chars_format::scientific) == chars_format::scientific;
>+  return scientific && !fixed;
>+}
>+
>+const int8_t kAsciiToInt[256] = {
>+    -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1,
>+    -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1,
>+    -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, 0,  1,  2,  3,  4,  5,  6,  7,  8,
>+    9,  -1, -1, -1, -1, -1, -1, -1, 10, 11, 12, 13, 14, 15, -1, -1, -1, -1, -1,
>+    -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1,
>+    -1, -1, 10, 11, 12, 13, 14, 15, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1,
>+    -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1,
>+    -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1,
>+    -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1,
>+    -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1,
>+    -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1,
>+    -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1,
>+    -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1, -1,
>+    -1, -1, -1, -1, -1, -1, -1, -1, -1};
>+
>+// Returns true if `ch` is a digit in the given base
>+template <int base>
>+bool IsDigit(char ch);
>+
>+// Converts a valid `ch` to its digit value in the given base.
>+template <int base>
>+unsigned ToDigit(char ch);
>+
>+// Returns true if `ch` is the exponent delimiter for the given base.
>+template <int base>
>+bool IsExponentCharacter(char ch);
>+
>+// Returns the maximum number of significant digits we will read for a float
>+// in the given base.
>+template <int base>
>+constexpr int MantissaDigitsMax();
>+
>+// Returns the largest consecutive run of digits we will accept when parsing a
>+// number in the given base.
>+template <int base>
>+constexpr int DigitLimit();
>+
>+// Returns the amount the exponent must be adjusted by for each dropped digit.
>+// (For decimal this is 1, since the digits are in base 10 and the exponent base
>+// is also 10, but for hexadecimal this is 4, since the digits are base 16 but
>+// the exponent base is 2.)
>+template <int base>
>+constexpr int DigitMagnitude();
>+
>+template <>
>+bool IsDigit<10>(char ch) {
>+  return ch >= '0' && ch <= '9';
>+}
>+template <>
>+bool IsDigit<16>(char ch) {
>+  return kAsciiToInt[static_cast<unsigned char>(ch)] >= 0;
>+}
>+
>+template <>
>+unsigned ToDigit<10>(char ch) {
>+  return ch - '0';
>+}
>+template <>
>+unsigned ToDigit<16>(char ch) {
>+  return kAsciiToInt[static_cast<unsigned char>(ch)];
>+}
>+
>+template <>
>+bool IsExponentCharacter<10>(char ch) {
>+  return ch == 'e' || ch == 'E';
>+}
>+
>+template <>
>+bool IsExponentCharacter<16>(char ch) {
>+  return ch == 'p' || ch == 'P';
>+}
>+
>+template <>
>+constexpr int MantissaDigitsMax<10>() {
>+  return kDecimalMantissaDigitsMax;
>+}
>+template <>
>+constexpr int MantissaDigitsMax<16>() {
>+  return kHexadecimalMantissaDigitsMax;
>+}
>+
>+template <>
>+constexpr int DigitLimit<10>() {
>+  return kDecimalDigitLimit;
>+}
>+template <>
>+constexpr int DigitLimit<16>() {
>+  return kHexadecimalDigitLimit;
>+}
>+
>+template <>
>+constexpr int DigitMagnitude<10>() {
>+  return 1;
>+}
>+template <>
>+constexpr int DigitMagnitude<16>() {
>+  return 4;
>+}
>+
>+// Reads decimal digits from [begin, end) into *out.  Returns the number of
>+// digits consumed.
>+//
>+// After max_digits has been read, keeps consuming characters, but no longer
>+// adjusts *out.  If a nonzero digit is dropped this way, *dropped_nonzero_digit
>+// is set; otherwise, it is left unmodified.
>+//
>+// If no digits are matched, returns 0 and leaves *out unchanged.
>+//
>+// ConsumeDigits does not protect against overflow on *out; max_digits must
>+// be chosen with respect to type T to avoid the possibility of overflow.
>+template <int base, typename T>
>+std::size_t ConsumeDigits(const char* begin, const char* end, int max_digits,
>+                          T* out, bool* dropped_nonzero_digit) {
>+  if (base == 10) {
>+    assert(max_digits <= std::numeric_limits<T>::digits10);
>+  } else if (base == 16) {
>+    assert(max_digits * 4 <= std::numeric_limits<T>::digits);
>+  }
>+  const char* const original_begin = begin;
>+  T accumulator = *out;
>+  const char* significant_digits_end =
>+      (end - begin > max_digits) ? begin + max_digits : end;
>+  while (begin < significant_digits_end && IsDigit<base>(*begin)) {
>+    // Do not guard against *out overflow; max_digits was chosen to avoid this.
>+    // Do assert against it, to detect problems in debug builds.
>+    auto digit = static_cast<T>(ToDigit<base>(*begin));
>+    assert(accumulator * base >= accumulator);
>+    accumulator *= base;
>+    assert(accumulator + digit >= accumulator);
>+    accumulator += digit;
>+    ++begin;
>+  }
>+  bool dropped_nonzero = false;
>+  while (begin < end && IsDigit<base>(*begin)) {
>+    dropped_nonzero = dropped_nonzero || (*begin != '0');
>+    ++begin;
>+  }
>+  if (dropped_nonzero && dropped_nonzero_digit != nullptr) {
>+    *dropped_nonzero_digit = true;
>+  }
>+  *out = accumulator;
>+  return begin - original_begin;
>+}
>+
>+// Returns true if `v` is one of the chars allowed inside parentheses following
>+// a NaN.
>+bool IsNanChar(char v) {
>+  return (v == '_') || (v >= '0' && v <= '9') || (v >= 'a' && v <= 'z') ||
>+         (v >= 'A' && v <= 'Z');
>+}
>+
>+// Checks the range [begin, end) for a strtod()-formatted infinity or NaN.  If
>+// one is found, sets `out` appropriately and returns true.
>+bool ParseInfinityOrNan(const char* begin, const char* end,
>+                        strings_internal::ParsedFloat* out) {
>+  if (end - begin < 3) {
>+    return false;
>+  }
>+  switch (*begin) {
>+    case 'i':
>+    case 'I': {
>+      // An infinity std::string consists of the characters "inf" or "infinity",
>+      // case insensitive.
>+      if (strings_internal::memcasecmp(begin + 1, "nf", 2) != 0) {
>+        return false;
>+      }
>+      out->type = strings_internal::FloatType::kInfinity;
>+      if (end - begin >= 8 &&
>+          strings_internal::memcasecmp(begin + 3, "inity", 5) == 0) {
>+        out->end = begin + 8;
>+      } else {
>+        out->end = begin + 3;
>+      }
>+      return true;
>+    }
>+    case 'n':
>+    case 'N': {
>+      // A NaN consists of the characters "nan", case insensitive, optionally
>+      // followed by a parenthesized sequence of zero or more alphanumeric
>+      // characters and/or underscores.
>+      if (strings_internal::memcasecmp(begin + 1, "an", 2) != 0) {
>+        return false;
>+      }
>+      out->type = strings_internal::FloatType::kNan;
>+      out->end = begin + 3;
>+      // NaN is allowed to be followed by a parenthesized std::string, consisting of
>+      // only the characters [a-zA-Z0-9_].  Match that if it's present.
>+      begin += 3;
>+      if (begin < end && *begin == '(') {
>+        const char* nan_begin = begin + 1;
>+        while (nan_begin < end && IsNanChar(*nan_begin)) {
>+          ++nan_begin;
>+        }
>+        if (nan_begin < end && *nan_begin == ')') {
>+          // We found an extra NaN specifier range
>+          out->subrange_begin = begin + 1;
>+          out->subrange_end = nan_begin;
>+          out->end = nan_begin + 1;
>+        }
>+      }
>+      return true;
>+    }
>+    default:
>+      return false;
>+  }
>+}
>+}  // namespace
>+
>+namespace strings_internal {
>+
>+template <int base>
>+strings_internal::ParsedFloat ParseFloat(const char* begin, const char* end,
>+                                         chars_format format_flags) {
>+  strings_internal::ParsedFloat result;
>+
>+  // Exit early if we're given an empty range.
>+  if (begin == end) return result;
>+
>+  // Handle the infinity and NaN cases.
>+  if (ParseInfinityOrNan(begin, end, &result)) {
>+    return result;
>+  }
>+
>+  const char* const mantissa_begin = begin;
>+  while (begin < end && *begin == '0') {
>+    ++begin;  // skip leading zeros
>+  }
>+  uint64_t mantissa = 0;
>+
>+  int exponent_adjustment = 0;
>+  bool mantissa_is_inexact = false;
>+  std::size_t pre_decimal_digits = ConsumeDigits<base>(
>+      begin, end, MantissaDigitsMax<base>(), &mantissa, &mantissa_is_inexact);
>+  begin += pre_decimal_digits;
>+  int digits_left;
>+  if (pre_decimal_digits >= DigitLimit<base>()) {
>+    // refuse to parse pathological inputs
>+    return result;
>+  } else if (pre_decimal_digits > MantissaDigitsMax<base>()) {
>+    // We dropped some non-fraction digits on the floor.  Adjust our exponent
>+    // to compensate.
>+    exponent_adjustment =
>+        static_cast<int>(pre_decimal_digits - MantissaDigitsMax<base>());
>+    digits_left = 0;
>+  } else {
>+    digits_left =
>+        static_cast<int>(MantissaDigitsMax<base>() - pre_decimal_digits);
>+  }
>+  if (begin < end && *begin == '.') {
>+    ++begin;
>+    if (mantissa == 0) {
>+      // If we haven't seen any nonzero digits yet, keep skipping zeros.  We
>+      // have to adjust the exponent to reflect the changed place value.
>+      const char* begin_zeros = begin;
>+      while (begin < end && *begin == '0') {
>+        ++begin;
>+      }
>+      std::size_t zeros_skipped = begin - begin_zeros;
>+      if (zeros_skipped >= DigitLimit<base>()) {
>+        // refuse to parse pathological inputs
>+        return result;
>+      }
>+      exponent_adjustment -= static_cast<int>(zeros_skipped);
>+    }
>+    std::size_t post_decimal_digits = ConsumeDigits<base>(
>+        begin, end, digits_left, &mantissa, &mantissa_is_inexact);
>+    begin += post_decimal_digits;
>+
>+    // Since `mantissa` is an integer, each significant digit we read after
>+    // the decimal point requires an adjustment to the exponent. "1.23e0" will
>+    // be stored as `mantissa` == 123 and `exponent` == -2 (that is,
>+    // "123e-2").
>+    if (post_decimal_digits >= DigitLimit<base>()) {
>+      // refuse to parse pathological inputs
>+      return result;
>+    } else if (post_decimal_digits > digits_left) {
>+      exponent_adjustment -= digits_left;
>+    } else {
>+      exponent_adjustment -= post_decimal_digits;
>+    }
>+  }
>+  // If we've found no mantissa whatsoever, this isn't a number.
>+  if (mantissa_begin == begin) {
>+    return result;
>+  }
>+  // A bare "." doesn't count as a mantissa either.
>+  if (begin - mantissa_begin == 1 && *mantissa_begin == '.') {
>+    return result;
>+  }
>+
>+  if (mantissa_is_inexact) {
>+    // We dropped significant digits on the floor.  Handle this appropriately.
>+    if (base == 10) {
>+      // If we truncated significant decimal digits, store the full range of the
>+      // mantissa for future big integer math for exact rounding.
>+      result.subrange_begin = mantissa_begin;
>+      result.subrange_end = begin;
>+    } else if (base == 16) {
>+      // If we truncated hex digits, reflect this fact by setting the low
>+      // ("sticky") bit.  This allows for correct rounding in all cases.
>+      mantissa |= 1;
>+    }
>+  }
>+  result.mantissa = mantissa;
>+
>+  const char* const exponent_begin = begin;
>+  result.literal_exponent = 0;
>+  bool found_exponent = false;
>+  if (AllowExponent(format_flags) && begin < end &&
>+      IsExponentCharacter<base>(*begin)) {
>+    bool negative_exponent = false;
>+    ++begin;
>+    if (begin < end && *begin == '-') {
>+      negative_exponent = true;
>+      ++begin;
>+    } else if (begin < end && *begin == '+') {
>+      ++begin;
>+    }
>+    const char* const exponent_digits_begin = begin;
>+    // Exponent is always expressed in decimal, even for hexadecimal floats.
>+    begin += ConsumeDigits<10>(begin, end, kDecimalExponentDigitsMax,
>+                               &result.literal_exponent, nullptr);
>+    if (begin == exponent_digits_begin) {
>+      // there were no digits where we expected an exponent.  We failed to read
>+      // an exponent and should not consume the 'e' after all.  Rewind 'begin'.
>+      found_exponent = false;
>+      begin = exponent_begin;
>+    } else {
>+      found_exponent = true;
>+      if (negative_exponent) {
>+        result.literal_exponent = -result.literal_exponent;
>+      }
>+    }
>+  }
>+
>+  if (!found_exponent && RequireExponent(format_flags)) {
>+    // Provided flags required an exponent, but none was found.  This results
>+    // in a failure to scan.
>+    return result;
>+  }
>+
>+  // Success!
>+  result.type = strings_internal::FloatType::kNumber;
>+  if (result.mantissa > 0) {
>+    result.exponent = result.literal_exponent +
>+                      (DigitMagnitude<base>() * exponent_adjustment);
>+  } else {
>+    result.exponent = 0;
>+  }
>+  result.end = begin;
>+  return result;
>+}
>+
>+template ParsedFloat ParseFloat<10>(const char* begin, const char* end,
>+                                    chars_format format_flags);
>+template ParsedFloat ParseFloat<16>(const char* begin, const char* end,
>+                                    chars_format format_flags);
>+
>+}  // namespace strings_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/charconv_parse.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/charconv_parse.h
>new file mode 100644
>index 00000000000..7a5c0874b80
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/charconv_parse.h
>@@ -0,0 +1,96 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#ifndef ABSL_STRINGS_INTERNAL_CHARCONV_PARSE_H_
>+#define ABSL_STRINGS_INTERNAL_CHARCONV_PARSE_H_
>+
>+#include <cstdint>
>+
>+#include "absl/strings/charconv.h"
>+
>+namespace absl {
>+namespace strings_internal {
>+
>+// Enum indicating whether a parsed float is a number or special value.
>+enum class FloatType { kNumber, kInfinity, kNan };
>+
>+// The decomposed parts of a parsed `float` or `double`.
>+struct ParsedFloat {
>+  // Representation of the parsed mantissa, with the decimal point adjusted to
>+  // make it an integer.
>+  //
>+  // During decimal scanning, this contains 19 significant digits worth of
>+  // mantissa value.  If digits beyond this point are found, they
>+  // are truncated, and if any of these dropped digits are nonzero, then
>+  // `mantissa` is inexact, and the full mantissa is stored in [subrange_begin,
>+  // subrange_end).
>+  //
>+  // During hexadecimal scanning, this contains 15 significant hex digits worth
>+  // of mantissa value.  Digits beyond this point are sticky -- they are
>+  // truncated, but if any dropped digits are nonzero, the low bit of mantissa
>+  // will be set.  (This allows for precise rounding, and avoids the need
>+  // to store the full mantissa in [subrange_begin, subrange_end).)
>+  uint64_t mantissa = 0;
>+
>+  // Floating point expontent.  This reflects any decimal point adjustments and
>+  // any truncated digits from the mantissa.  The absolute value of the parsed
>+  // number is represented by mantissa * (base ** exponent), where base==10 for
>+  // decimal floats, and base==2 for hexadecimal floats.
>+  int exponent = 0;
>+
>+  // The literal exponent value scanned from the input, or 0 if none was
>+  // present.  This does not reflect any adjustments applied to mantissa.
>+  int literal_exponent = 0;
>+
>+  // The type of number scanned.
>+  FloatType type = FloatType::kNumber;
>+
>+  // When non-null, [subrange_begin, subrange_end) marks a range of characters
>+  // that require further processing.  The meaning is dependent on float type.
>+  // If type == kNumber and this is set, this is a "wide input": the input
>+  // mantissa contained more than 19 digits.  The range contains the full
>+  // mantissa.  It plus `literal_exponent` need to be examined to find the best
>+  // floating point match.
>+  // If type == kNan and this is set, the range marks the contents of a
>+  // matched parenthesized character region after the NaN.
>+  const char* subrange_begin = nullptr;
>+  const char* subrange_end = nullptr;
>+
>+  // One-past-the-end of the successfully parsed region, or nullptr if no
>+  // matching pattern was found.
>+  const char* end = nullptr;
>+};
>+
>+// Read the floating point number in the provided range, and populate
>+// ParsedFloat accordingly.
>+//
>+// format_flags is a bitmask value specifying what patterns this API will match.
>+// `scientific` and `fixed`  are honored per std::from_chars rules
>+// ([utility.from.chars], C++17): if exactly one of these bits is set, then an
>+// exponent is required, or dislallowed, respectively.
>+//
>+// Template parameter `base` must be either 10 or 16.  For base 16, a "0x" is
>+// *not* consumed.  The `hex` bit from format_flags is ignored by ParseFloat.
>+template <int base>
>+ParsedFloat ParseFloat(const char* begin, const char* end,
>+                       absl::chars_format format_flags);
>+
>+extern template ParsedFloat ParseFloat<10>(const char* begin, const char* end,
>+                                           absl::chars_format format_flags);
>+extern template ParsedFloat ParseFloat<16>(const char* begin, const char* end,
>+                                           absl::chars_format format_flags);
>+
>+}  // namespace strings_internal
>+}  // namespace absl
>+#endif  // ABSL_STRINGS_INTERNAL_CHARCONV_PARSE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/charconv_parse_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/charconv_parse_test.cc
>new file mode 100644
>index 00000000000..1ff86004973
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/charconv_parse_test.cc
>@@ -0,0 +1,357 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/strings/internal/charconv_parse.h"
>+
>+#include <string>
>+#include <utility>
>+
>+#include "gmock/gmock.h"
>+#include "gtest/gtest.h"
>+#include "absl/base/internal/raw_logging.h"
>+#include "absl/strings/str_cat.h"
>+
>+using absl::chars_format;
>+using absl::strings_internal::FloatType;
>+using absl::strings_internal::ParsedFloat;
>+using absl::strings_internal::ParseFloat;
>+
>+namespace {
>+
>+// Check that a given std::string input is parsed to the expected mantissa and
>+// exponent.
>+//
>+// Input std::string `s` must contain a '$' character.  It marks the end of the
>+// characters that should be consumed by the match.  It is stripped from the
>+// input to ParseFloat.
>+//
>+// If input std::string `s` contains '[' and ']' characters, these mark the region
>+// of characters that should be marked as the "subrange".  For NaNs, this is
>+// the location of the extended NaN std::string.  For numbers, this is the location
>+// of the full, over-large mantissa.
>+template <int base>
>+void ExpectParsedFloat(std::string s, absl::chars_format format_flags,
>+                       FloatType expected_type, uint64_t expected_mantissa,
>+                       int expected_exponent,
>+                       int expected_literal_exponent = -999) {
>+  SCOPED_TRACE(s);
>+
>+  int begin_subrange = -1;
>+  int end_subrange = -1;
>+  // If s contains '[' and ']', then strip these characters and set the subrange
>+  // indices appropriately.
>+  std::string::size_type open_bracket_pos = s.find('[');
>+  if (open_bracket_pos != std::string::npos) {
>+    begin_subrange = static_cast<int>(open_bracket_pos);
>+    s.replace(open_bracket_pos, 1, "");
>+    std::string::size_type close_bracket_pos = s.find(']');
>+    ABSL_RAW_CHECK(close_bracket_pos != absl::string_view::npos,
>+                   "Test input contains [ without matching ]");
>+    end_subrange = static_cast<int>(close_bracket_pos);
>+    s.replace(close_bracket_pos, 1, "");
>+  }
>+  const std::string::size_type expected_characters_matched = s.find('$');
>+  ABSL_RAW_CHECK(expected_characters_matched != std::string::npos,
>+                 "Input std::string must contain $");
>+  s.replace(expected_characters_matched, 1, "");
>+
>+  ParsedFloat parsed =
>+      ParseFloat<base>(s.data(), s.data() + s.size(), format_flags);
>+
>+  EXPECT_NE(parsed.end, nullptr);
>+  if (parsed.end == nullptr) {
>+    return;  // The following tests are not useful if we fully failed to parse
>+  }
>+  EXPECT_EQ(parsed.type, expected_type);
>+  if (begin_subrange == -1) {
>+    EXPECT_EQ(parsed.subrange_begin, nullptr);
>+    EXPECT_EQ(parsed.subrange_end, nullptr);
>+  } else {
>+    EXPECT_EQ(parsed.subrange_begin, s.data() + begin_subrange);
>+    EXPECT_EQ(parsed.subrange_end, s.data() + end_subrange);
>+  }
>+  if (parsed.type == FloatType::kNumber) {
>+    EXPECT_EQ(parsed.mantissa, expected_mantissa);
>+    EXPECT_EQ(parsed.exponent, expected_exponent);
>+    if (expected_literal_exponent != -999) {
>+      EXPECT_EQ(parsed.literal_exponent, expected_literal_exponent);
>+    }
>+  }
>+  auto characters_matched = static_cast<int>(parsed.end - s.data());
>+  EXPECT_EQ(characters_matched, expected_characters_matched);
>+}
>+
>+// Check that a given std::string input is parsed to the expected mantissa and
>+// exponent.
>+//
>+// Input std::string `s` must contain a '$' character.  It marks the end of the
>+// characters that were consumed by the match.
>+template <int base>
>+void ExpectNumber(std::string s, absl::chars_format format_flags,
>+                  uint64_t expected_mantissa, int expected_exponent,
>+                  int expected_literal_exponent = -999) {
>+  ExpectParsedFloat<base>(std::move(s), format_flags, FloatType::kNumber,
>+                          expected_mantissa, expected_exponent,
>+                          expected_literal_exponent);
>+}
>+
>+// Check that a given std::string input is parsed to the given special value.
>+//
>+// This tests against both number bases, since infinities and NaNs have
>+// identical representations in both modes.
>+void ExpectSpecial(const std::string& s, absl::chars_format format_flags,
>+                   FloatType type) {
>+  ExpectParsedFloat<10>(s, format_flags, type, 0, 0);
>+  ExpectParsedFloat<16>(s, format_flags, type, 0, 0);
>+}
>+
>+// Check that a given input std::string is not matched by Float.
>+template <int base>
>+void ExpectFailedParse(absl::string_view s, absl::chars_format format_flags) {
>+  ParsedFloat parsed =
>+      ParseFloat<base>(s.data(), s.data() + s.size(), format_flags);
>+  EXPECT_EQ(parsed.end, nullptr);
>+}
>+
>+TEST(ParseFloat, SimpleValue) {
>+  // Test that various forms of floating point numbers all parse correctly.
>+  ExpectNumber<10>("1.23456789e5$", chars_format::general, 123456789, -3);
>+  ExpectNumber<10>("1.23456789e+5$", chars_format::general, 123456789, -3);
>+  ExpectNumber<10>("1.23456789E5$", chars_format::general, 123456789, -3);
>+  ExpectNumber<10>("1.23456789e05$", chars_format::general, 123456789, -3);
>+  ExpectNumber<10>("123.456789e3$", chars_format::general, 123456789, -3);
>+  ExpectNumber<10>("0.000123456789e9$", chars_format::general, 123456789, -3);
>+  ExpectNumber<10>("123456.789$", chars_format::general, 123456789, -3);
>+  ExpectNumber<10>("123456789e-3$", chars_format::general, 123456789, -3);
>+
>+  ExpectNumber<16>("1.234abcdefp28$", chars_format::general, 0x1234abcdef, -8);
>+  ExpectNumber<16>("1.234abcdefp+28$", chars_format::general, 0x1234abcdef, -8);
>+  ExpectNumber<16>("1.234ABCDEFp28$", chars_format::general, 0x1234abcdef, -8);
>+  ExpectNumber<16>("1.234AbCdEfP0028$", chars_format::general, 0x1234abcdef,
>+                   -8);
>+  ExpectNumber<16>("123.4abcdefp20$", chars_format::general, 0x1234abcdef, -8);
>+  ExpectNumber<16>("0.0001234abcdefp44$", chars_format::general, 0x1234abcdef,
>+                   -8);
>+  ExpectNumber<16>("1234abcd.ef$", chars_format::general, 0x1234abcdef, -8);
>+  ExpectNumber<16>("1234abcdefp-8$", chars_format::general, 0x1234abcdef, -8);
>+
>+  // ExpectNumber does not attempt to drop trailing zeroes.
>+  ExpectNumber<10>("0001.2345678900e005$", chars_format::general, 12345678900,
>+                   -5);
>+  ExpectNumber<16>("0001.234abcdef000p28$", chars_format::general,
>+                   0x1234abcdef000, -20);
>+
>+  // Ensure non-matching characters after a number are ignored, even when they
>+  // look like potentially matching characters.
>+  ExpectNumber<10>("1.23456789e5$   ", chars_format::general, 123456789, -3);
>+  ExpectNumber<10>("1.23456789e5$e5e5", chars_format::general, 123456789, -3);
>+  ExpectNumber<10>("1.23456789e5$.25", chars_format::general, 123456789, -3);
>+  ExpectNumber<10>("1.23456789e5$-", chars_format::general, 123456789, -3);
>+  ExpectNumber<10>("1.23456789e5$PUPPERS!!!", chars_format::general, 123456789,
>+                   -3);
>+  ExpectNumber<10>("123456.789$efghij", chars_format::general, 123456789, -3);
>+  ExpectNumber<10>("123456.789$e", chars_format::general, 123456789, -3);
>+  ExpectNumber<10>("123456.789$p5", chars_format::general, 123456789, -3);
>+  ExpectNumber<10>("123456.789$.10", chars_format::general, 123456789, -3);
>+
>+  ExpectNumber<16>("1.234abcdefp28$   ", chars_format::general, 0x1234abcdef,
>+                   -8);
>+  ExpectNumber<16>("1.234abcdefp28$p28", chars_format::general, 0x1234abcdef,
>+                   -8);
>+  ExpectNumber<16>("1.234abcdefp28$.125", chars_format::general, 0x1234abcdef,
>+                   -8);
>+  ExpectNumber<16>("1.234abcdefp28$-", chars_format::general, 0x1234abcdef, -8);
>+  ExpectNumber<16>("1.234abcdefp28$KITTEHS!!!", chars_format::general,
>+                   0x1234abcdef, -8);
>+  ExpectNumber<16>("1234abcd.ef$ghijk", chars_format::general, 0x1234abcdef,
>+                   -8);
>+  ExpectNumber<16>("1234abcd.ef$p", chars_format::general, 0x1234abcdef, -8);
>+  ExpectNumber<16>("1234abcd.ef$.10", chars_format::general, 0x1234abcdef, -8);
>+
>+  // Ensure we can read a full resolution mantissa without overflow.
>+  ExpectNumber<10>("9999999999999999999$", chars_format::general,
>+                   9999999999999999999u, 0);
>+  ExpectNumber<16>("fffffffffffffff$", chars_format::general,
>+                   0xfffffffffffffffu, 0);
>+
>+  // Check that zero is consistently read.
>+  ExpectNumber<10>("0$", chars_format::general, 0, 0);
>+  ExpectNumber<16>("0$", chars_format::general, 0, 0);
>+  ExpectNumber<10>("000000000000000000000000000000000000000$",
>+                   chars_format::general, 0, 0);
>+  ExpectNumber<16>("000000000000000000000000000000000000000$",
>+                   chars_format::general, 0, 0);
>+  ExpectNumber<10>("0000000000000000000000.000000000000000000$",
>+                   chars_format::general, 0, 0);
>+  ExpectNumber<16>("0000000000000000000000.000000000000000000$",
>+                   chars_format::general, 0, 0);
>+  ExpectNumber<10>("0.00000000000000000000000000000000e123456$",
>+                   chars_format::general, 0, 0);
>+  ExpectNumber<16>("0.00000000000000000000000000000000p123456$",
>+                   chars_format::general, 0, 0);
>+}
>+
>+TEST(ParseFloat, LargeDecimalMantissa) {
>+  // After 19 significant decimal digits in the mantissa, ParsedFloat will
>+  // truncate additional digits.  We need to test that:
>+  //   1) the truncation to 19 digits happens
>+  //   2) the returned exponent reflects the dropped significant digits
>+  //   3) a correct literal_exponent is set
>+  //
>+  // If and only if a significant digit is found after 19 digits, then the
>+  // entirety of the mantissa in case the exact value is needed to make a
>+  // rounding decision.  The [ and ] characters below denote where such a
>+  // subregion was marked by by ParseFloat.  They are not part of the input.
>+
>+  // Mark a capture group only if a dropped digit is significant (nonzero).
>+  ExpectNumber<10>("100000000000000000000000000$", chars_format::general,
>+                   1000000000000000000,
>+                   /* adjusted exponent */ 8);
>+
>+  ExpectNumber<10>("123456789123456789100000000$", chars_format::general,
>+                   1234567891234567891,
>+                   /* adjusted exponent */ 8);
>+
>+  ExpectNumber<10>("[123456789123456789123456789]$", chars_format::general,
>+                   1234567891234567891,
>+                   /* adjusted exponent */ 8,
>+                   /* literal exponent */ 0);
>+
>+  ExpectNumber<10>("[123456789123456789100000009]$", chars_format::general,
>+                   1234567891234567891,
>+                   /* adjusted exponent */ 8,
>+                   /* literal exponent */ 0);
>+
>+  ExpectNumber<10>("[123456789123456789120000000]$", chars_format::general,
>+                   1234567891234567891,
>+                   /* adjusted exponent */ 8,
>+                   /* literal exponent */ 0);
>+
>+  // Leading zeroes should not count towards the 19 significant digit limit
>+  ExpectNumber<10>("[00000000123456789123456789123456789]$",
>+                   chars_format::general, 1234567891234567891,
>+                   /* adjusted exponent */ 8,
>+                   /* literal exponent */ 0);
>+
>+  ExpectNumber<10>("00000000123456789123456789100000000$",
>+                   chars_format::general, 1234567891234567891,
>+                   /* adjusted exponent */ 8);
>+
>+  // Truncated digits after the decimal point should not cause a further
>+  // exponent adjustment.
>+  ExpectNumber<10>("1.234567891234567891e123$", chars_format::general,
>+                   1234567891234567891, 105);
>+  ExpectNumber<10>("[1.23456789123456789123456789]e123$", chars_format::general,
>+                   1234567891234567891,
>+                   /* adjusted exponent */ 105,
>+                   /* literal exponent */ 123);
>+
>+  // Ensure we truncate, and not round.  (The from_chars algorithm we use
>+  // depends on our guess missing low, if it misses, so we need the rounding
>+  // error to be downward.)
>+  ExpectNumber<10>("[1999999999999999999999]$", chars_format::general,
>+                   1999999999999999999,
>+                   /* adjusted exponent */ 3,
>+                   /* literal exponent */ 0);
>+}
>+
>+TEST(ParseFloat, LargeHexadecimalMantissa) {
>+  // After 15 significant hex digits in the mantissa, ParsedFloat will treat
>+  // additional digits as sticky,  We need to test that:
>+  //   1) The truncation to 15 digits happens
>+  //   2) The returned exponent reflects the dropped significant digits
>+  //   3) If a nonzero digit is dropped, the low bit of mantissa is set.
>+
>+  ExpectNumber<16>("123456789abcdef123456789abcdef$", chars_format::general,
>+                   0x123456789abcdef, 60);
>+
>+  // Leading zeroes should not count towards the 15 significant digit limit
>+  ExpectNumber<16>("000000123456789abcdef123456789abcdef$",
>+                   chars_format::general, 0x123456789abcdef, 60);
>+
>+  // Truncated digits after the radix point should not cause a further
>+  // exponent adjustment.
>+  ExpectNumber<16>("1.23456789abcdefp100$", chars_format::general,
>+                   0x123456789abcdef, 44);
>+  ExpectNumber<16>("1.23456789abcdef123456789abcdefp100$",
>+                   chars_format::general, 0x123456789abcdef, 44);
>+
>+  // test sticky digit behavior.  The low bit should be set iff any dropped
>+  // digit is nonzero.
>+  ExpectNumber<16>("123456789abcdee123456789abcdee$", chars_format::general,
>+                   0x123456789abcdef, 60);
>+  ExpectNumber<16>("123456789abcdee000000000000001$", chars_format::general,
>+                   0x123456789abcdef, 60);
>+  ExpectNumber<16>("123456789abcdee000000000000000$", chars_format::general,
>+                   0x123456789abcdee, 60);
>+}
>+
>+TEST(ParseFloat, ScientificVsFixed) {
>+  // In fixed mode, an exponent is never matched (but the remainder of the
>+  // number will be matched.)
>+  ExpectNumber<10>("1.23456789$e5", chars_format::fixed, 123456789, -8);
>+  ExpectNumber<10>("123456.789$", chars_format::fixed, 123456789, -3);
>+  ExpectNumber<16>("1.234abcdef$p28", chars_format::fixed, 0x1234abcdef, -36);
>+  ExpectNumber<16>("1234abcd.ef$", chars_format::fixed, 0x1234abcdef, -8);
>+
>+  // In scientific mode, numbers don't match *unless* they have an exponent.
>+  ExpectNumber<10>("1.23456789e5$", chars_format::scientific, 123456789, -3);
>+  ExpectFailedParse<10>("-123456.789$", chars_format::scientific);
>+  ExpectNumber<16>("1.234abcdefp28$", chars_format::scientific, 0x1234abcdef,
>+                   -8);
>+  ExpectFailedParse<16>("1234abcd.ef$", chars_format::scientific);
>+}
>+
>+TEST(ParseFloat, Infinity) {
>+  ExpectFailedParse<10>("in", chars_format::general);
>+  ExpectFailedParse<16>("in", chars_format::general);
>+  ExpectFailedParse<10>("inx", chars_format::general);
>+  ExpectFailedParse<16>("inx", chars_format::general);
>+  ExpectSpecial("inf$", chars_format::general, FloatType::kInfinity);
>+  ExpectSpecial("Inf$", chars_format::general, FloatType::kInfinity);
>+  ExpectSpecial("INF$", chars_format::general, FloatType::kInfinity);
>+  ExpectSpecial("inf$inite", chars_format::general, FloatType::kInfinity);
>+  ExpectSpecial("iNfInItY$", chars_format::general, FloatType::kInfinity);
>+  ExpectSpecial("infinity$!!!", chars_format::general, FloatType::kInfinity);
>+}
>+
>+TEST(ParseFloat, NaN) {
>+  ExpectFailedParse<10>("na", chars_format::general);
>+  ExpectFailedParse<16>("na", chars_format::general);
>+  ExpectFailedParse<10>("nah", chars_format::general);
>+  ExpectFailedParse<16>("nah", chars_format::general);
>+  ExpectSpecial("nan$", chars_format::general, FloatType::kNan);
>+  ExpectSpecial("NaN$", chars_format::general, FloatType::kNan);
>+  ExpectSpecial("nAn$", chars_format::general, FloatType::kNan);
>+  ExpectSpecial("NAN$", chars_format::general, FloatType::kNan);
>+  ExpectSpecial("NaN$aNaNaNaNaBatman!", chars_format::general, FloatType::kNan);
>+
>+  // A parenthesized sequence of the characters [a-zA-Z0-9_] is allowed to
>+  // appear after an NaN.  Check that this is allowed, and that the correct
>+  // characters are grouped.
>+  //
>+  // (The characters [ and ] in the pattern below delimit the expected matched
>+  // subgroup; they are not part of the input passed to ParseFloat.)
>+  ExpectSpecial("nan([0xabcdef])$", chars_format::general, FloatType::kNan);
>+  ExpectSpecial("nan([0xabcdef])$...", chars_format::general, FloatType::kNan);
>+  ExpectSpecial("nan([0xabcdef])$)...", chars_format::general, FloatType::kNan);
>+  ExpectSpecial("nan([])$", chars_format::general, FloatType::kNan);
>+  ExpectSpecial("nan([aAzZ09_])$", chars_format::general, FloatType::kNan);
>+  // If the subgroup contains illegal characters, don't match it at all.
>+  ExpectSpecial("nan$(bad-char)", chars_format::general, FloatType::kNan);
>+  // Also cope with a missing close paren.
>+  ExpectSpecial("nan$(0xabcdef", chars_format::general, FloatType::kNan);
>+}
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/escaping_test_common.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/escaping_test_common.h
>new file mode 100644
>index 00000000000..cc41f4312c7
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/escaping_test_common.h
>@@ -0,0 +1,131 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// This test contains common things needed by both escaping_test.cc and
>+// escaping_benchmark.cc.
>+
>+#ifndef ABSL_STRINGS_INTERNAL_ESCAPING_TEST_COMMON_H_
>+#define ABSL_STRINGS_INTERNAL_ESCAPING_TEST_COMMON_H_
>+
>+#include <array>
>+#include "absl/strings/string_view.h"
>+
>+namespace absl {
>+namespace strings_internal {
>+
>+struct base64_testcase {
>+  absl::string_view plaintext;
>+  absl::string_view cyphertext;
>+};
>+
>+inline const std::array<base64_testcase, 5>& base64_strings() {
>+  static const std::array<base64_testcase, 5> testcase{{
>+      // Some google quotes
>+      // Cyphertext created with "uuencode (GNU sharutils) 4.6.3"
>+      // (Note that we're testing the websafe encoding, though, so if
>+      // you add messages, be sure to run "tr -- '+/' '-_'" on the output)
>+      { "I was always good at math and science, and I never realized "
>+        "that was unusual or somehow undesirable. So one of the things "
>+        "I care a lot about is helping to remove that stigma, "
>+        "to show girls that you can be feminine, you can like the things "
>+        "that girls like, but you can also be really good at technology. "
>+        "You can be really good at building things."
>+        " - Marissa Meyer, Newsweek, 2010-12-22" "\n",
>+
>+        "SSB3YXMgYWx3YXlzIGdvb2QgYXQgbWF0aCBhbmQgc2NpZW5jZSwgYW5kIEkg"
>+        "bmV2ZXIgcmVhbGl6ZWQgdGhhdCB3YXMgdW51c3VhbCBvciBzb21laG93IHVu"
>+        "ZGVzaXJhYmxlLiBTbyBvbmUgb2YgdGhlIHRoaW5ncyBJIGNhcmUgYSBsb3Qg"
>+        "YWJvdXQgaXMgaGVscGluZyB0byByZW1vdmUgdGhhdCBzdGlnbWEsIHRvIHNo"
>+        "b3cgZ2lybHMgdGhhdCB5b3UgY2FuIGJlIGZlbWluaW5lLCB5b3UgY2FuIGxp"
>+        "a2UgdGhlIHRoaW5ncyB0aGF0IGdpcmxzIGxpa2UsIGJ1dCB5b3UgY2FuIGFs"
>+        "c28gYmUgcmVhbGx5IGdvb2QgYXQgdGVjaG5vbG9neS4gWW91IGNhbiBiZSBy"
>+        "ZWFsbHkgZ29vZCBhdCBidWlsZGluZyB0aGluZ3MuIC0gTWFyaXNzYSBNZXll"
>+        "ciwgTmV3c3dlZWssIDIwMTAtMTItMjIK" },
>+
>+      { "Typical first year for a new cluster: "
>+        "~0.5 overheating "
>+        "~1 PDU failure "
>+        "~1 rack-move "
>+        "~1 network rewiring "
>+        "~20 rack failures "
>+        "~5 racks go wonky "
>+        "~8 network maintenances "
>+        "~12 router reloads "
>+        "~3 router failures "
>+        "~dozens of minor 30-second blips for dns "
>+        "~1000 individual machine failures "
>+        "~thousands of hard drive failures "
>+        "slow disks, bad memory, misconfigured machines, flaky machines, etc."
>+        " - Jeff Dean, The Joys of Real Hardware" "\n",
>+
>+        "VHlwaWNhbCBmaXJzdCB5ZWFyIGZvciBhIG5ldyBjbHVzdGVyOiB-MC41IG92"
>+        "ZXJoZWF0aW5nIH4xIFBEVSBmYWlsdXJlIH4xIHJhY2stbW92ZSB-MSBuZXR3"
>+        "b3JrIHJld2lyaW5nIH4yMCByYWNrIGZhaWx1cmVzIH41IHJhY2tzIGdvIHdv"
>+        "bmt5IH44IG5ldHdvcmsgbWFpbnRlbmFuY2VzIH4xMiByb3V0ZXIgcmVsb2Fk"
>+        "cyB-MyByb3V0ZXIgZmFpbHVyZXMgfmRvemVucyBvZiBtaW5vciAzMC1zZWNv"
>+        "bmQgYmxpcHMgZm9yIGRucyB-MTAwMCBpbmRpdmlkdWFsIG1hY2hpbmUgZmFp"
>+        "bHVyZXMgfnRob3VzYW5kcyBvZiBoYXJkIGRyaXZlIGZhaWx1cmVzIHNsb3cg"
>+        "ZGlza3MsIGJhZCBtZW1vcnksIG1pc2NvbmZpZ3VyZWQgbWFjaGluZXMsIGZs"
>+        "YWt5IG1hY2hpbmVzLCBldGMuIC0gSmVmZiBEZWFuLCBUaGUgSm95cyBvZiBS"
>+        "ZWFsIEhhcmR3YXJlCg" },
>+
>+      { "I'm the head of the webspam team at Google.  "
>+        "That means that if you type your name into Google and get porn back, "
>+        "it's my fault. Unless you're a porn star, in which case porn is a "
>+        "completely reasonable response."
>+        " - Matt Cutts, Google Plus" "\n",
>+
>+        "SSdtIHRoZSBoZWFkIG9mIHRoZSB3ZWJzcGFtIHRlYW0gYXQgR29vZ2xlLiAg"
>+        "VGhhdCBtZWFucyB0aGF0IGlmIHlvdSB0eXBlIHlvdXIgbmFtZSBpbnRvIEdv"
>+        "b2dsZSBhbmQgZ2V0IHBvcm4gYmFjaywgaXQncyBteSBmYXVsdC4gVW5sZXNz"
>+        "IHlvdSdyZSBhIHBvcm4gc3RhciwgaW4gd2hpY2ggY2FzZSBwb3JuIGlzIGEg"
>+        "Y29tcGxldGVseSByZWFzb25hYmxlIHJlc3BvbnNlLiAtIE1hdHQgQ3V0dHMs"
>+        "IEdvb2dsZSBQbHVzCg" },
>+
>+      { "It will still be a long time before machines approach human "
>+        "intelligence. "
>+        "But luckily, machines don't actually have to be intelligent; "
>+        "they just have to fake it. Access to a wealth of information, "
>+        "combined with a rudimentary decision-making capacity, "
>+        "can often be almost as useful. Of course, the results are better yet "
>+        "when coupled with intelligence. A reference librarian with access to "
>+        "a good search engine is a formidable tool."
>+        " - Craig Silverstein, Siemens Pictures of the Future, Spring 2004"
>+        "\n",
>+
>+        "SXQgd2lsbCBzdGlsbCBiZSBhIGxvbmcgdGltZSBiZWZvcmUgbWFjaGluZXMg"
>+        "YXBwcm9hY2ggaHVtYW4gaW50ZWxsaWdlbmNlLiBCdXQgbHVja2lseSwgbWFj"
>+        "aGluZXMgZG9uJ3QgYWN0dWFsbHkgaGF2ZSB0byBiZSBpbnRlbGxpZ2VudDsg"
>+        "dGhleSBqdXN0IGhhdmUgdG8gZmFrZSBpdC4gQWNjZXNzIHRvIGEgd2VhbHRo"
>+        "IG9mIGluZm9ybWF0aW9uLCBjb21iaW5lZCB3aXRoIGEgcnVkaW1lbnRhcnkg"
>+        "ZGVjaXNpb24tbWFraW5nIGNhcGFjaXR5LCBjYW4gb2Z0ZW4gYmUgYWxtb3N0"
>+        "IGFzIHVzZWZ1bC4gT2YgY291cnNlLCB0aGUgcmVzdWx0cyBhcmUgYmV0dGVy"
>+        "IHlldCB3aGVuIGNvdXBsZWQgd2l0aCBpbnRlbGxpZ2VuY2UuIEEgcmVmZXJl"
>+        "bmNlIGxpYnJhcmlhbiB3aXRoIGFjY2VzcyB0byBhIGdvb2Qgc2VhcmNoIGVu"
>+        "Z2luZSBpcyBhIGZvcm1pZGFibGUgdG9vbC4gLSBDcmFpZyBTaWx2ZXJzdGVp"
>+        "biwgU2llbWVucyBQaWN0dXJlcyBvZiB0aGUgRnV0dXJlLCBTcHJpbmcgMjAw"
>+        "NAo" },
>+
>+      // Degenerate edge case
>+      { "",
>+        "" },
>+  }};
>+
>+  return testcase;
>+}
>+
>+}  // namespace strings_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_STRINGS_INTERNAL_ESCAPING_TEST_COMMON_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/memutil.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/memutil.cc
>new file mode 100644
>index 00000000000..a0de70dffdb
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/memutil.cc
>@@ -0,0 +1,110 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/strings/internal/memutil.h"
>+
>+#include <cstdlib>
>+
>+namespace absl {
>+namespace strings_internal {
>+
>+int memcasecmp(const char* s1, const char* s2, size_t len) {
>+  const unsigned char* us1 = reinterpret_cast<const unsigned char*>(s1);
>+  const unsigned char* us2 = reinterpret_cast<const unsigned char*>(s2);
>+
>+  for (size_t i = 0; i < len; i++) {
>+    const int diff =
>+        int{static_cast<unsigned char>(absl::ascii_tolower(us1[i]))} -
>+        int{static_cast<unsigned char>(absl::ascii_tolower(us2[i]))};
>+    if (diff != 0) return diff;
>+  }
>+  return 0;
>+}
>+
>+char* memdup(const char* s, size_t slen) {
>+  void* copy;
>+  if ((copy = malloc(slen)) == nullptr) return nullptr;
>+  memcpy(copy, s, slen);
>+  return reinterpret_cast<char*>(copy);
>+}
>+
>+char* memrchr(const char* s, int c, size_t slen) {
>+  for (const char* e = s + slen - 1; e >= s; e--) {
>+    if (*e == c) return const_cast<char*>(e);
>+  }
>+  return nullptr;
>+}
>+
>+size_t memspn(const char* s, size_t slen, const char* accept) {
>+  const char* p = s;
>+  const char* spanp;
>+  char c, sc;
>+
>+cont:
>+  c = *p++;
>+  if (slen-- == 0) return p - 1 - s;
>+  for (spanp = accept; (sc = *spanp++) != '\0';)
>+    if (sc == c) goto cont;
>+  return p - 1 - s;
>+}
>+
>+size_t memcspn(const char* s, size_t slen, const char* reject) {
>+  const char* p = s;
>+  const char* spanp;
>+  char c, sc;
>+
>+  while (slen-- != 0) {
>+    c = *p++;
>+    for (spanp = reject; (sc = *spanp++) != '\0';)
>+      if (sc == c) return p - 1 - s;
>+  }
>+  return p - s;
>+}
>+
>+char* mempbrk(const char* s, size_t slen, const char* accept) {
>+  const char* scanp;
>+  int sc;
>+
>+  for (; slen; ++s, --slen) {
>+    for (scanp = accept; (sc = *scanp++) != '\0';)
>+      if (sc == *s) return const_cast<char*>(s);
>+  }
>+  return nullptr;
>+}
>+
>+// This is significantly faster for case-sensitive matches with very
>+// few possible matches.  See unit test for benchmarks.
>+const char* memmatch(const char* phaystack, size_t haylen, const char* pneedle,
>+                     size_t neelen) {
>+  if (0 == neelen) {
>+    return phaystack;  // even if haylen is 0
>+  }
>+  if (haylen < neelen) return nullptr;
>+
>+  const char* match;
>+  const char* hayend = phaystack + haylen - neelen + 1;
>+  // A static cast is used here to work around the fact that memchr returns
>+  // a void* on Posix-compliant systems and const void* on Windows.
>+  while ((match = static_cast<const char*>(
>+              memchr(phaystack, pneedle[0], hayend - phaystack)))) {
>+    if (memcmp(match, pneedle, neelen) == 0)
>+      return match;
>+    else
>+      phaystack = match + 1;
>+  }
>+  return nullptr;
>+}
>+
>+}  // namespace strings_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/memutil.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/memutil.h
>new file mode 100644
>index 00000000000..a6f1c69138e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/memutil.h
>@@ -0,0 +1,146 @@
>+//
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+
>+// These routines provide mem versions of standard C std::string routines,
>+// such as strpbrk.  They function exactly the same as the str versions,
>+// so if you wonder what they are, replace the word "mem" by
>+// "str" and check out the man page.  I could return void*, as the
>+// strutil.h mem*() routines tend to do, but I return char* instead
>+// since this is by far the most common way these functions are called.
>+//
>+// The difference between the mem and str versions is the mem version
>+// takes a pointer and a length, rather than a '\0'-terminated std::string.
>+// The memcase* routines defined here assume the locale is "C"
>+// (they use absl::ascii_tolower instead of tolower).
>+//
>+// These routines are based on the BSD library.
>+//
>+// Here's a list of routines from std::string.h, and their mem analogues.
>+// Functions in lowercase are defined in std::string.h; those in UPPERCASE
>+// are defined here:
>+//
>+// strlen                  --
>+// strcat strncat          MEMCAT
>+// strcpy strncpy          memcpy
>+// --                      memccpy   (very cool function, btw)
>+// --                      memmove
>+// --                      memset
>+// strcmp strncmp          memcmp
>+// strcasecmp strncasecmp  MEMCASECMP
>+// strchr                  memchr
>+// strcoll                 --
>+// strxfrm                 --
>+// strdup strndup          MEMDUP
>+// strrchr                 MEMRCHR
>+// strspn                  MEMSPN
>+// strcspn                 MEMCSPN
>+// strpbrk                 MEMPBRK
>+// strstr                  MEMSTR MEMMEM
>+// (g)strcasestr           MEMCASESTR MEMCASEMEM
>+// strtok                  --
>+// strprefix               MEMPREFIX      (strprefix is from strutil.h)
>+// strcaseprefix           MEMCASEPREFIX  (strcaseprefix is from strutil.h)
>+// strsuffix               MEMSUFFIX      (strsuffix is from strutil.h)
>+// strcasesuffix           MEMCASESUFFIX  (strcasesuffix is from strutil.h)
>+// --                      MEMIS
>+// --                      MEMCASEIS
>+// strcount                MEMCOUNT       (strcount is from strutil.h)
>+
>+#ifndef ABSL_STRINGS_INTERNAL_MEMUTIL_H_
>+#define ABSL_STRINGS_INTERNAL_MEMUTIL_H_
>+
>+#include <cstddef>
>+#include <cstring>
>+
>+#include "absl/base/port.h"  // disable some warnings on Windows
>+#include "absl/strings/ascii.h"  // for absl::ascii_tolower
>+
>+namespace absl {
>+namespace strings_internal {
>+
>+inline char* memcat(char* dest, size_t destlen, const char* src,
>+                    size_t srclen) {
>+  return reinterpret_cast<char*>(memcpy(dest + destlen, src, srclen));
>+}
>+
>+int memcasecmp(const char* s1, const char* s2, size_t len);
>+char* memdup(const char* s, size_t slen);
>+char* memrchr(const char* s, int c, size_t slen);
>+size_t memspn(const char* s, size_t slen, const char* accept);
>+size_t memcspn(const char* s, size_t slen, const char* reject);
>+char* mempbrk(const char* s, size_t slen, const char* accept);
>+
>+// This is for internal use only.  Don't call this directly
>+template <bool case_sensitive>
>+const char* int_memmatch(const char* haystack, size_t haylen,
>+                         const char* needle, size_t neelen) {
>+  if (0 == neelen) {
>+    return haystack;  // even if haylen is 0
>+  }
>+  const char* hayend = haystack + haylen;
>+  const char* needlestart = needle;
>+  const char* needleend = needlestart + neelen;
>+
>+  for (; haystack < hayend; ++haystack) {
>+    char hay = case_sensitive
>+                   ? *haystack
>+                   : absl::ascii_tolower(static_cast<unsigned char>(*haystack));
>+    char nee = case_sensitive
>+                   ? *needle
>+                   : absl::ascii_tolower(static_cast<unsigned char>(*needle));
>+    if (hay == nee) {
>+      if (++needle == needleend) {
>+        return haystack + 1 - neelen;
>+      }
>+    } else if (needle != needlestart) {
>+      // must back up haystack in case a prefix matched (find "aab" in "aaab")
>+      haystack -= needle - needlestart;  // for loop will advance one more
>+      needle = needlestart;
>+    }
>+  }
>+  return nullptr;
>+}
>+
>+// These are the guys you can call directly
>+inline const char* memstr(const char* phaystack, size_t haylen,
>+                          const char* pneedle) {
>+  return int_memmatch<true>(phaystack, haylen, pneedle, strlen(pneedle));
>+}
>+
>+inline const char* memcasestr(const char* phaystack, size_t haylen,
>+                              const char* pneedle) {
>+  return int_memmatch<false>(phaystack, haylen, pneedle, strlen(pneedle));
>+}
>+
>+inline const char* memmem(const char* phaystack, size_t haylen,
>+                          const char* pneedle, size_t needlelen) {
>+  return int_memmatch<true>(phaystack, haylen, pneedle, needlelen);
>+}
>+
>+inline const char* memcasemem(const char* phaystack, size_t haylen,
>+                              const char* pneedle, size_t needlelen) {
>+  return int_memmatch<false>(phaystack, haylen, pneedle, needlelen);
>+}
>+
>+// This is significantly faster for case-sensitive matches with very
>+// few possible matches.  See unit test for benchmarks.
>+const char* memmatch(const char* phaystack, size_t haylen, const char* pneedle,
>+                     size_t neelen);
>+
>+}  // namespace strings_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_STRINGS_INTERNAL_MEMUTIL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/memutil_benchmark.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/memutil_benchmark.cc
>new file mode 100644
>index 00000000000..77915adb958
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/memutil_benchmark.cc
>@@ -0,0 +1,323 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/strings/internal/memutil.h"
>+
>+#include <algorithm>
>+#include <cstdlib>
>+
>+#include "benchmark/benchmark.h"
>+#include "absl/strings/ascii.h"
>+
>+// We fill the haystack with aaaaaaaaaaaaaaaaaa...aaaab.
>+// That gives us:
>+// - an easy search: 'b'
>+// - a medium search: 'ab'.  That means every letter is a possible match.
>+// - a pathological search: 'aaaaaa.......aaaaab' (half as many a's as haytack)
>+// We benchmark case-sensitive and case-insensitive versions of
>+// three memmem implementations:
>+// - memmem() from memutil.h
>+// - search() from STL
>+// - memmatch(), a custom implementation using memchr and memcmp.
>+// Here are sample results:
>+//
>+// Run on (12 X 3800 MHz CPU s)
>+// CPU Caches:
>+//   L1 Data 32K (x6)
>+//   L1 Instruction 32K (x6)
>+//   L2 Unified 256K (x6)
>+//   L3 Unified 15360K (x1)
>+// ----------------------------------------------------------------
>+// Benchmark                           Time          CPU Iterations
>+// ----------------------------------------------------------------
>+// BM_Memmem                        3583 ns      3582 ns     196469  2.59966GB/s
>+// BM_MemmemMedium                 13743 ns     13742 ns      50901  693.986MB/s
>+// BM_MemmemPathological        13695030 ns  13693977 ns         51  713.133kB/s
>+// BM_Memcasemem                    3299 ns      3299 ns     212942  2.82309GB/s
>+// BM_MemcasememMedium             16407 ns     16406 ns      42170  581.309MB/s
>+// BM_MemcasememPathological    17267745 ns  17266030 ns         41  565.598kB/s
>+// BM_Search                        1610 ns      1609 ns     431321  5.78672GB/s
>+// BM_SearchMedium                 11111 ns     11110 ns      63001  858.414MB/s
>+// BM_SearchPathological        12117390 ns  12116397 ns         58  805.984kB/s
>+// BM_Searchcase                    3081 ns      3081 ns     229949  3.02313GB/s
>+// BM_SearchcaseMedium             16003 ns     16001 ns      44170  595.998MB/s
>+// BM_SearchcasePathological    15823413 ns  15821909 ns         44  617.222kB/s
>+// BM_Memmatch                       197 ns       197 ns    3584225  47.2951GB/s
>+// BM_MemmatchMedium               52333 ns     52329 ns      13280  182.244MB/s
>+// BM_MemmatchPathological        659799 ns    659727 ns       1058  14.4556MB/s
>+// BM_Memcasematch                  5460 ns      5460 ns     127606  1.70586GB/s
>+// BM_MemcasematchMedium           32861 ns     32857 ns      21258  290.248MB/s
>+// BM_MemcasematchPathological  15154243 ns  15153089 ns         46  644.464kB/s
>+// BM_MemmemStartup                    5 ns         5 ns  150821500
>+// BM_SearchStartup                    5 ns         5 ns  150644203
>+// BM_MemmatchStartup                  7 ns         7 ns   97068802
>+//
>+// Conclusions:
>+//
>+// The following recommendations are based on the sample results above. However,
>+// we have found that the performance of STL search can vary significantly
>+// depending on compiler and standard library implementation. We recommend you
>+// run the benchmarks for yourself on relevant platforms.
>+//
>+// If you need case-insensitive, STL search is slightly better than memmem for
>+// all cases.
>+//
>+// Case-sensitive is more subtle:
>+// Custom memmatch is _very_ fast at scanning, so if you have very few possible
>+// matches in your haystack, that's the way to go. Performance drops
>+// significantly with more matches.
>+//
>+// STL search is slightly faster than memmem in the medium and pathological
>+// benchmarks. However, the performance of memmem is currently more dependable
>+// across platforms and build configurations.
>+
>+namespace {
>+
>+constexpr int kHaystackSize = 10000;
>+constexpr int64_t kHaystackSize64 = kHaystackSize;
>+const char* MakeHaystack() {
>+  char* haystack = new char[kHaystackSize];
>+  for (int i = 0; i < kHaystackSize - 1; ++i) haystack[i] = 'a';
>+  haystack[kHaystackSize - 1] = 'b';
>+  return haystack;
>+}
>+const char* const kHaystack = MakeHaystack();
>+
>+void BM_Memmem(benchmark::State& state) {
>+  for (auto _ : state) {
>+    benchmark::DoNotOptimize(
>+        absl::strings_internal::memmem(kHaystack, kHaystackSize, "b", 1));
>+  }
>+  state.SetBytesProcessed(kHaystackSize64 * state.iterations());
>+}
>+BENCHMARK(BM_Memmem);
>+
>+void BM_MemmemMedium(benchmark::State& state) {
>+  for (auto _ : state) {
>+    benchmark::DoNotOptimize(
>+        absl::strings_internal::memmem(kHaystack, kHaystackSize, "ab", 2));
>+  }
>+  state.SetBytesProcessed(kHaystackSize64 * state.iterations());
>+}
>+BENCHMARK(BM_MemmemMedium);
>+
>+void BM_MemmemPathological(benchmark::State& state) {
>+  for (auto _ : state) {
>+    benchmark::DoNotOptimize(absl::strings_internal::memmem(
>+        kHaystack, kHaystackSize, kHaystack + kHaystackSize / 2,
>+        kHaystackSize - kHaystackSize / 2));
>+  }
>+  state.SetBytesProcessed(kHaystackSize64 * state.iterations());
>+}
>+BENCHMARK(BM_MemmemPathological);
>+
>+void BM_Memcasemem(benchmark::State& state) {
>+  for (auto _ : state) {
>+    benchmark::DoNotOptimize(
>+        absl::strings_internal::memcasemem(kHaystack, kHaystackSize, "b", 1));
>+  }
>+  state.SetBytesProcessed(kHaystackSize64 * state.iterations());
>+}
>+BENCHMARK(BM_Memcasemem);
>+
>+void BM_MemcasememMedium(benchmark::State& state) {
>+  for (auto _ : state) {
>+    benchmark::DoNotOptimize(
>+        absl::strings_internal::memcasemem(kHaystack, kHaystackSize, "ab", 2));
>+  }
>+  state.SetBytesProcessed(kHaystackSize64 * state.iterations());
>+}
>+BENCHMARK(BM_MemcasememMedium);
>+
>+void BM_MemcasememPathological(benchmark::State& state) {
>+  for (auto _ : state) {
>+    benchmark::DoNotOptimize(absl::strings_internal::memcasemem(
>+        kHaystack, kHaystackSize, kHaystack + kHaystackSize / 2,
>+        kHaystackSize - kHaystackSize / 2));
>+  }
>+  state.SetBytesProcessed(kHaystackSize64 * state.iterations());
>+}
>+BENCHMARK(BM_MemcasememPathological);
>+
>+bool case_eq(const char a, const char b) {
>+  return absl::ascii_tolower(a) == absl::ascii_tolower(b);
>+}
>+
>+void BM_Search(benchmark::State& state) {
>+  for (auto _ : state) {
>+    benchmark::DoNotOptimize(std::search(kHaystack, kHaystack + kHaystackSize,
>+                                         kHaystack + kHaystackSize - 1,
>+                                         kHaystack + kHaystackSize));
>+  }
>+  state.SetBytesProcessed(kHaystackSize64 * state.iterations());
>+}
>+BENCHMARK(BM_Search);
>+
>+void BM_SearchMedium(benchmark::State& state) {
>+  for (auto _ : state) {
>+    benchmark::DoNotOptimize(std::search(kHaystack, kHaystack + kHaystackSize,
>+                                         kHaystack + kHaystackSize - 2,
>+                                         kHaystack + kHaystackSize));
>+  }
>+  state.SetBytesProcessed(kHaystackSize64 * state.iterations());
>+}
>+BENCHMARK(BM_SearchMedium);
>+
>+void BM_SearchPathological(benchmark::State& state) {
>+  for (auto _ : state) {
>+    benchmark::DoNotOptimize(std::search(kHaystack, kHaystack + kHaystackSize,
>+                                         kHaystack + kHaystackSize / 2,
>+                                         kHaystack + kHaystackSize));
>+  }
>+  state.SetBytesProcessed(kHaystackSize64 * state.iterations());
>+}
>+BENCHMARK(BM_SearchPathological);
>+
>+void BM_Searchcase(benchmark::State& state) {
>+  for (auto _ : state) {
>+    benchmark::DoNotOptimize(std::search(kHaystack, kHaystack + kHaystackSize,
>+                                         kHaystack + kHaystackSize - 1,
>+                                         kHaystack + kHaystackSize, case_eq));
>+  }
>+  state.SetBytesProcessed(kHaystackSize64 * state.iterations());
>+}
>+BENCHMARK(BM_Searchcase);
>+
>+void BM_SearchcaseMedium(benchmark::State& state) {
>+  for (auto _ : state) {
>+    benchmark::DoNotOptimize(std::search(kHaystack, kHaystack + kHaystackSize,
>+                                         kHaystack + kHaystackSize - 2,
>+                                         kHaystack + kHaystackSize, case_eq));
>+  }
>+  state.SetBytesProcessed(kHaystackSize64 * state.iterations());
>+}
>+BENCHMARK(BM_SearchcaseMedium);
>+
>+void BM_SearchcasePathological(benchmark::State& state) {
>+  for (auto _ : state) {
>+    benchmark::DoNotOptimize(std::search(kHaystack, kHaystack + kHaystackSize,
>+                                         kHaystack + kHaystackSize / 2,
>+                                         kHaystack + kHaystackSize, case_eq));
>+  }
>+  state.SetBytesProcessed(kHaystackSize64 * state.iterations());
>+}
>+BENCHMARK(BM_SearchcasePathological);
>+
>+char* memcasechr(const char* s, int c, size_t slen) {
>+  c = absl::ascii_tolower(c);
>+  for (; slen; ++s, --slen) {
>+    if (absl::ascii_tolower(*s) == c) return const_cast<char*>(s);
>+  }
>+  return nullptr;
>+}
>+
>+const char* memcasematch(const char* phaystack, size_t haylen,
>+                         const char* pneedle, size_t neelen) {
>+  if (0 == neelen) {
>+    return phaystack;  // even if haylen is 0
>+  }
>+  if (haylen < neelen) return nullptr;
>+
>+  const char* match;
>+  const char* hayend = phaystack + haylen - neelen + 1;
>+  while ((match = static_cast<char*>(
>+              memcasechr(phaystack, pneedle[0], hayend - phaystack)))) {
>+    if (absl::strings_internal::memcasecmp(match, pneedle, neelen) == 0)
>+      return match;
>+    else
>+      phaystack = match + 1;
>+  }
>+  return nullptr;
>+}
>+
>+void BM_Memmatch(benchmark::State& state) {
>+  for (auto _ : state) {
>+    benchmark::DoNotOptimize(
>+        absl::strings_internal::memmatch(kHaystack, kHaystackSize, "b", 1));
>+  }
>+  state.SetBytesProcessed(kHaystackSize64 * state.iterations());
>+}
>+BENCHMARK(BM_Memmatch);
>+
>+void BM_MemmatchMedium(benchmark::State& state) {
>+  for (auto _ : state) {
>+    benchmark::DoNotOptimize(
>+        absl::strings_internal::memmatch(kHaystack, kHaystackSize, "ab", 2));
>+  }
>+  state.SetBytesProcessed(kHaystackSize64 * state.iterations());
>+}
>+BENCHMARK(BM_MemmatchMedium);
>+
>+void BM_MemmatchPathological(benchmark::State& state) {
>+  for (auto _ : state) {
>+    benchmark::DoNotOptimize(absl::strings_internal::memmatch(
>+        kHaystack, kHaystackSize, kHaystack + kHaystackSize / 2,
>+        kHaystackSize - kHaystackSize / 2));
>+  }
>+  state.SetBytesProcessed(kHaystackSize64 * state.iterations());
>+}
>+BENCHMARK(BM_MemmatchPathological);
>+
>+void BM_Memcasematch(benchmark::State& state) {
>+  for (auto _ : state) {
>+    benchmark::DoNotOptimize(memcasematch(kHaystack, kHaystackSize, "b", 1));
>+  }
>+  state.SetBytesProcessed(kHaystackSize64 * state.iterations());
>+}
>+BENCHMARK(BM_Memcasematch);
>+
>+void BM_MemcasematchMedium(benchmark::State& state) {
>+  for (auto _ : state) {
>+    benchmark::DoNotOptimize(memcasematch(kHaystack, kHaystackSize, "ab", 2));
>+  }
>+  state.SetBytesProcessed(kHaystackSize64 * state.iterations());
>+}
>+BENCHMARK(BM_MemcasematchMedium);
>+
>+void BM_MemcasematchPathological(benchmark::State& state) {
>+  for (auto _ : state) {
>+    benchmark::DoNotOptimize(memcasematch(kHaystack, kHaystackSize,
>+                                          kHaystack + kHaystackSize / 2,
>+                                          kHaystackSize - kHaystackSize / 2));
>+  }
>+  state.SetBytesProcessed(kHaystackSize64 * state.iterations());
>+}
>+BENCHMARK(BM_MemcasematchPathological);
>+
>+void BM_MemmemStartup(benchmark::State& state) {
>+  for (auto _ : state) {
>+    benchmark::DoNotOptimize(absl::strings_internal::memmem(
>+        kHaystack + kHaystackSize - 10, 10, kHaystack + kHaystackSize - 1, 1));
>+  }
>+}
>+BENCHMARK(BM_MemmemStartup);
>+
>+void BM_SearchStartup(benchmark::State& state) {
>+  for (auto _ : state) {
>+    benchmark::DoNotOptimize(
>+        std::search(kHaystack + kHaystackSize - 10, kHaystack + kHaystackSize,
>+                    kHaystack + kHaystackSize - 1, kHaystack + kHaystackSize));
>+  }
>+}
>+BENCHMARK(BM_SearchStartup);
>+
>+void BM_MemmatchStartup(benchmark::State& state) {
>+  for (auto _ : state) {
>+    benchmark::DoNotOptimize(absl::strings_internal::memmatch(
>+        kHaystack + kHaystackSize - 10, 10, kHaystack + kHaystackSize - 1, 1));
>+  }
>+}
>+BENCHMARK(BM_MemmatchStartup);
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/memutil_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/memutil_test.cc
>new file mode 100644
>index 00000000000..09424de9a47
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/memutil_test.cc
>@@ -0,0 +1,179 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+// Unit test for memutil.cc
>+
>+#include "absl/strings/internal/memutil.h"
>+
>+#include <cstdlib>
>+
>+#include "gtest/gtest.h"
>+#include "absl/strings/ascii.h"
>+
>+namespace {
>+
>+static char* memcasechr(const char* s, int c, size_t slen) {
>+  c = absl::ascii_tolower(c);
>+  for (; slen; ++s, --slen) {
>+    if (absl::ascii_tolower(*s) == c) return const_cast<char*>(s);
>+  }
>+  return nullptr;
>+}
>+
>+static const char* memcasematch(const char* phaystack, size_t haylen,
>+                                const char* pneedle, size_t neelen) {
>+  if (0 == neelen) {
>+    return phaystack;  // even if haylen is 0
>+  }
>+  if (haylen < neelen) return nullptr;
>+
>+  const char* match;
>+  const char* hayend = phaystack + haylen - neelen + 1;
>+  while ((match = static_cast<char*>(
>+              memcasechr(phaystack, pneedle[0], hayend - phaystack)))) {
>+    if (absl::strings_internal::memcasecmp(match, pneedle, neelen) == 0)
>+      return match;
>+    else
>+      phaystack = match + 1;
>+  }
>+  return nullptr;
>+}
>+
>+TEST(MemUtilTest, AllTests) {
>+  // check memutil functions
>+  char a[1000];
>+  absl::strings_internal::memcat(a, 0, "hello", sizeof("hello") - 1);
>+  absl::strings_internal::memcat(a, 5, " there", sizeof(" there") - 1);
>+
>+  EXPECT_EQ(absl::strings_internal::memcasecmp(a, "heLLO there",
>+                                               sizeof("hello there") - 1),
>+            0);
>+  EXPECT_EQ(absl::strings_internal::memcasecmp(a, "heLLO therf",
>+                                               sizeof("hello there") - 1),
>+            -1);
>+  EXPECT_EQ(absl::strings_internal::memcasecmp(a, "heLLO therf",
>+                                               sizeof("hello there") - 2),
>+            0);
>+  EXPECT_EQ(absl::strings_internal::memcasecmp(a, "whatever", 0), 0);
>+
>+  char* p = absl::strings_internal::memdup("hello", 5);
>+  free(p);
>+
>+  p = absl::strings_internal::memrchr("hello there", 'e',
>+                                      sizeof("hello there") - 1);
>+  EXPECT_TRUE(p && p[-1] == 'r');
>+  p = absl::strings_internal::memrchr("hello there", 'e',
>+                                      sizeof("hello there") - 2);
>+  EXPECT_TRUE(p && p[-1] == 'h');
>+  p = absl::strings_internal::memrchr("hello there", 'u',
>+                                      sizeof("hello there") - 1);
>+  EXPECT_TRUE(p == nullptr);
>+
>+  int len = absl::strings_internal::memspn("hello there",
>+                                           sizeof("hello there") - 1, "hole");
>+  EXPECT_EQ(len, sizeof("hello") - 1);
>+  len = absl::strings_internal::memspn("hello there", sizeof("hello there") - 1,
>+                                       "u");
>+  EXPECT_EQ(len, 0);
>+  len = absl::strings_internal::memspn("hello there", sizeof("hello there") - 1,
>+                                       "");
>+  EXPECT_EQ(len, 0);
>+  len = absl::strings_internal::memspn("hello there", sizeof("hello there") - 1,
>+                                       "trole h");
>+  EXPECT_EQ(len, sizeof("hello there") - 1);
>+  len = absl::strings_internal::memspn("hello there!",
>+                                       sizeof("hello there!") - 1, "trole h");
>+  EXPECT_EQ(len, sizeof("hello there") - 1);
>+  len = absl::strings_internal::memspn("hello there!",
>+                                       sizeof("hello there!") - 2, "trole h!");
>+  EXPECT_EQ(len, sizeof("hello there!") - 2);
>+
>+  len = absl::strings_internal::memcspn("hello there",
>+                                        sizeof("hello there") - 1, "leho");
>+  EXPECT_EQ(len, 0);
>+  len = absl::strings_internal::memcspn("hello there",
>+                                        sizeof("hello there") - 1, "u");
>+  EXPECT_EQ(len, sizeof("hello there") - 1);
>+  len = absl::strings_internal::memcspn("hello there",
>+                                        sizeof("hello there") - 1, "");
>+  EXPECT_EQ(len, sizeof("hello there") - 1);
>+  len = absl::strings_internal::memcspn("hello there",
>+                                        sizeof("hello there") - 1, " ");
>+  EXPECT_EQ(len, 5);
>+
>+  p = absl::strings_internal::mempbrk("hello there", sizeof("hello there") - 1,
>+                                      "leho");
>+  EXPECT_TRUE(p && p[1] == 'e' && p[2] == 'l');
>+  p = absl::strings_internal::mempbrk("hello there", sizeof("hello there") - 1,
>+                                      "nu");
>+  EXPECT_TRUE(p == nullptr);
>+  p = absl::strings_internal::mempbrk("hello there!",
>+                                      sizeof("hello there!") - 2, "!");
>+  EXPECT_TRUE(p == nullptr);
>+  p = absl::strings_internal::mempbrk("hello there", sizeof("hello there") - 1,
>+                                      " t ");
>+  EXPECT_TRUE(p && p[-1] == 'o' && p[1] == 't');
>+
>+  {
>+    const char kHaystack[] = "0123456789";
>+    EXPECT_EQ(absl::strings_internal::memmem(kHaystack, 0, "", 0), kHaystack);
>+    EXPECT_EQ(absl::strings_internal::memmem(kHaystack, 10, "012", 3),
>+              kHaystack);
>+    EXPECT_EQ(absl::strings_internal::memmem(kHaystack, 10, "0xx", 1),
>+              kHaystack);
>+    EXPECT_EQ(absl::strings_internal::memmem(kHaystack, 10, "789", 3),
>+              kHaystack + 7);
>+    EXPECT_EQ(absl::strings_internal::memmem(kHaystack, 10, "9xx", 1),
>+              kHaystack + 9);
>+    EXPECT_TRUE(absl::strings_internal::memmem(kHaystack, 10, "9xx", 3) ==
>+                nullptr);
>+    EXPECT_TRUE(absl::strings_internal::memmem(kHaystack, 10, "xxx", 1) ==
>+                nullptr);
>+  }
>+  {
>+    const char kHaystack[] = "aBcDeFgHiJ";
>+    EXPECT_EQ(absl::strings_internal::memcasemem(kHaystack, 0, "", 0),
>+              kHaystack);
>+    EXPECT_EQ(absl::strings_internal::memcasemem(kHaystack, 10, "Abc", 3),
>+              kHaystack);
>+    EXPECT_EQ(absl::strings_internal::memcasemem(kHaystack, 10, "Axx", 1),
>+              kHaystack);
>+    EXPECT_EQ(absl::strings_internal::memcasemem(kHaystack, 10, "hIj", 3),
>+              kHaystack + 7);
>+    EXPECT_EQ(absl::strings_internal::memcasemem(kHaystack, 10, "jxx", 1),
>+              kHaystack + 9);
>+    EXPECT_TRUE(absl::strings_internal::memcasemem(kHaystack, 10, "jxx", 3) ==
>+                nullptr);
>+    EXPECT_TRUE(absl::strings_internal::memcasemem(kHaystack, 10, "xxx", 1) ==
>+                nullptr);
>+  }
>+  {
>+    const char kHaystack[] = "0123456789";
>+    EXPECT_EQ(absl::strings_internal::memmatch(kHaystack, 0, "", 0), kHaystack);
>+    EXPECT_EQ(absl::strings_internal::memmatch(kHaystack, 10, "012", 3),
>+              kHaystack);
>+    EXPECT_EQ(absl::strings_internal::memmatch(kHaystack, 10, "0xx", 1),
>+              kHaystack);
>+    EXPECT_EQ(absl::strings_internal::memmatch(kHaystack, 10, "789", 3),
>+              kHaystack + 7);
>+    EXPECT_EQ(absl::strings_internal::memmatch(kHaystack, 10, "9xx", 1),
>+              kHaystack + 9);
>+    EXPECT_TRUE(absl::strings_internal::memmatch(kHaystack, 10, "9xx", 3) ==
>+                nullptr);
>+    EXPECT_TRUE(absl::strings_internal::memmatch(kHaystack, 10, "xxx", 1) ==
>+                nullptr);
>+  }
>+}
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/numbers_test_common.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/numbers_test_common.h
>new file mode 100644
>index 00000000000..20e3af51141
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/numbers_test_common.h
>@@ -0,0 +1,178 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// This file contains common things needed by numbers_test.cc,
>+// numbers_legacy_test.cc and numbers_benchmark.cc.
>+
>+#ifndef ABSL_STRINGS_INTERNAL_NUMBERS_TEST_COMMON_H_
>+#define ABSL_STRINGS_INTERNAL_NUMBERS_TEST_COMMON_H_
>+
>+#include <array>
>+#include <cstdint>
>+#include <limits>
>+#include <string>
>+
>+namespace absl {
>+namespace strings_internal {
>+
>+template <typename IntType>
>+inline bool Itoa(IntType value, int base, std::string* destination) {
>+  destination->clear();
>+  if (base <= 1 || base > 36) {
>+    return false;
>+  }
>+
>+  if (value == 0) {
>+    destination->push_back('0');
>+    return true;
>+  }
>+
>+  bool negative = value < 0;
>+  while (value != 0) {
>+    const IntType next_value = value / base;
>+    // Can't use std::abs here because of problems when IntType is unsigned.
>+    int remainder = value > next_value * base ? value - next_value * base
>+                                              : next_value * base - value;
>+    char c = remainder < 10 ? '0' + remainder : 'A' + remainder - 10;
>+    destination->insert(0, 1, c);
>+    value = next_value;
>+  }
>+
>+  if (negative) {
>+    destination->insert(0, 1, '-');
>+  }
>+  return true;
>+}
>+
>+struct uint32_test_case {
>+  const char* str;
>+  bool expect_ok;
>+  int base;  // base to pass to the conversion function
>+  uint32_t expected;
>+};
>+
>+inline const std::array<uint32_test_case, 27>& strtouint32_test_cases() {
>+  static const std::array<uint32_test_case, 27> test_cases{{
>+      {"0xffffffff", true, 16, std::numeric_limits<uint32_t>::max()},
>+      {"0x34234324", true, 16, 0x34234324},
>+      {"34234324", true, 16, 0x34234324},
>+      {"0", true, 16, 0},
>+      {" \t\n 0xffffffff", true, 16, std::numeric_limits<uint32_t>::max()},
>+      {" \f\v 46", true, 10, 46},  // must accept weird whitespace
>+      {" \t\n 72717222", true, 8, 072717222},
>+      {" \t\n 072717222", true, 8, 072717222},
>+      {" \t\n 072717228", false, 8, 07271722},
>+      {"0", true, 0, 0},
>+
>+      // Base-10 version.
>+      {"34234324", true, 0, 34234324},
>+      {"4294967295", true, 0, std::numeric_limits<uint32_t>::max()},
>+      {"34234324 \n\t", true, 10, 34234324},
>+
>+      // Unusual base
>+      {"0", true, 3, 0},
>+      {"2", true, 3, 2},
>+      {"11", true, 3, 4},
>+
>+      // Invalid uints.
>+      {"", false, 0, 0},
>+      {"  ", false, 0, 0},
>+      {"abc", false, 0, 0},  // would be valid hex, but prefix is missing
>+      {"34234324a", false, 0, 34234324},
>+      {"34234.3", false, 0, 34234},
>+      {"-1", false, 0, 0},
>+      {"   -123", false, 0, 0},
>+      {" \t\n -123", false, 0, 0},
>+
>+      // Out of bounds.
>+      {"4294967296", false, 0, std::numeric_limits<uint32_t>::max()},
>+      {"0x100000000", false, 0, std::numeric_limits<uint32_t>::max()},
>+      {nullptr, false, 0, 0},
>+  }};
>+  return test_cases;
>+}
>+
>+struct uint64_test_case {
>+  const char* str;
>+  bool expect_ok;
>+  int base;
>+  uint64_t expected;
>+};
>+
>+inline const std::array<uint64_test_case, 34>& strtouint64_test_cases() {
>+  static const std::array<uint64_test_case, 34> test_cases{{
>+      {"0x3423432448783446", true, 16, int64_t{0x3423432448783446}},
>+      {"3423432448783446", true, 16, int64_t{0x3423432448783446}},
>+
>+      {"0", true, 16, 0},
>+      {"000", true, 0, 0},
>+      {"0", true, 0, 0},
>+      {" \t\n 0xffffffffffffffff", true, 16,
>+       std::numeric_limits<uint64_t>::max()},
>+
>+      {"012345670123456701234", true, 8, int64_t{012345670123456701234}},
>+      {"12345670123456701234", true, 8, int64_t{012345670123456701234}},
>+
>+      {"12845670123456701234", false, 8, 0},
>+
>+      // Base-10 version.
>+      {"34234324487834466", true, 0, int64_t{34234324487834466}},
>+
>+      {" \t\n 18446744073709551615", true, 0,
>+       std::numeric_limits<uint64_t>::max()},
>+
>+      {"34234324487834466 \n\t ", true, 0, int64_t{34234324487834466}},
>+
>+      {" \f\v 46", true, 10, 46},  // must accept weird whitespace
>+
>+      // Unusual base
>+      {"0", true, 3, 0},
>+      {"2", true, 3, 2},
>+      {"11", true, 3, 4},
>+
>+      {"0", true, 0, 0},
>+
>+      // Invalid uints.
>+      {"", false, 0, 0},
>+      {"  ", false, 0, 0},
>+      {"abc", false, 0, 0},
>+      {"34234324487834466a", false, 0, 0},
>+      {"34234487834466.3", false, 0, 0},
>+      {"-1", false, 0, 0},
>+      {"   -123", false, 0, 0},
>+      {" \t\n -123", false, 0, 0},
>+
>+      // Out of bounds.
>+      {"18446744073709551616", false, 10, 0},
>+      {"18446744073709551616", false, 0, 0},
>+      {"0x10000000000000000", false, 16, std::numeric_limits<uint64_t>::max()},
>+      {"0X10000000000000000", false, 16,
>+       std::numeric_limits<uint64_t>::max()},  // 0X versus 0x.
>+      {"0x10000000000000000", false, 0, std::numeric_limits<uint64_t>::max()},
>+      {"0X10000000000000000", false, 0,
>+       std::numeric_limits<uint64_t>::max()},  // 0X versus 0x.
>+
>+      {"0x1234", true, 16, 0x1234},
>+
>+      // Base-10 std::string version.
>+      {"1234", true, 0, 1234},
>+      {nullptr, false, 0, 0},
>+  }};
>+  return test_cases;
>+}
>+
>+}  // namespace strings_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_STRINGS_INTERNAL_NUMBERS_TEST_COMMON_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/ostringstream.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/ostringstream.cc
>new file mode 100644
>index 00000000000..6ee2b109088
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/ostringstream.cc
>@@ -0,0 +1,34 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/strings/internal/ostringstream.h"
>+
>+namespace absl {
>+namespace strings_internal {
>+
>+OStringStream::Buf::int_type OStringStream::overflow(int c) {
>+  assert(s_);
>+  if (!Buf::traits_type::eq_int_type(c, Buf::traits_type::eof()))
>+    s_->push_back(static_cast<char>(c));
>+  return 1;
>+}
>+
>+std::streamsize OStringStream::xsputn(const char* s, std::streamsize n) {
>+  assert(s_);
>+  s_->append(s, n);
>+  return n;
>+}
>+
>+}  // namespace strings_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/ostringstream.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/ostringstream.h
>new file mode 100644
>index 00000000000..6e1325b9140
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/ostringstream.h
>@@ -0,0 +1,87 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#ifndef ABSL_STRINGS_INTERNAL_OSTRINGSTREAM_H_
>+#define ABSL_STRINGS_INTERNAL_OSTRINGSTREAM_H_
>+
>+#include <cassert>
>+#include <ostream>
>+#include <streambuf>
>+#include <string>
>+
>+#include "absl/base/port.h"
>+
>+namespace absl {
>+namespace strings_internal {
>+
>+// The same as std::ostringstream but appends to a user-specified std::string,
>+// and is faster. It is ~70% faster to create, ~50% faster to write to, and
>+// completely free to extract the result std::string.
>+//
>+//   std::string s;
>+//   OStringStream strm(&s);
>+//   strm << 42 << ' ' << 3.14;  // appends to `s`
>+//
>+// The stream object doesn't have to be named. Starting from C++11 operator<<
>+// works with rvalues of std::ostream.
>+//
>+//   std::string s;
>+//   OStringStream(&s) << 42 << ' ' << 3.14;  // appends to `s`
>+//
>+// OStringStream is faster to create than std::ostringstream but it's still
>+// relatively slow. Avoid creating multiple streams where a single stream will
>+// do.
>+//
>+// Creates unnecessary instances of OStringStream: slow.
>+//
>+//   std::string s;
>+//   OStringStream(&s) << 42;
>+//   OStringStream(&s) << ' ';
>+//   OStringStream(&s) << 3.14;
>+//
>+// Creates a single instance of OStringStream and reuses it: fast.
>+//
>+//   std::string s;
>+//   OStringStream strm(&s);
>+//   strm << 42;
>+//   strm << ' ';
>+//   strm << 3.14;
>+//
>+// Note: flush() has no effect. No reason to call it.
>+class OStringStream : private std::basic_streambuf<char>, public std::ostream {
>+ public:
>+  // The argument can be null, in which case you'll need to call str(p) with a
>+  // non-null argument before you can write to the stream.
>+  //
>+  // The destructor of OStringStream doesn't use the std::string. It's OK to destroy
>+  // the std::string before the stream.
>+  explicit OStringStream(std::string* s) : std::ostream(this), s_(s) {}
>+
>+  std::string* str() { return s_; }
>+  const std::string* str() const { return s_; }
>+  void str(std::string* s) { s_ = s; }
>+
>+ private:
>+  using Buf = std::basic_streambuf<char>;
>+
>+  Buf::int_type overflow(int c) override;
>+  std::streamsize xsputn(const char* s, std::streamsize n) override;
>+
>+  std::string* s_;
>+};
>+
>+}  // namespace strings_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_STRINGS_INTERNAL_OSTRINGSTREAM_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/ostringstream_benchmark.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/ostringstream_benchmark.cc
>new file mode 100644
>index 00000000000..c93f96909d8
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/ostringstream_benchmark.cc
>@@ -0,0 +1,106 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/strings/internal/ostringstream.h"
>+
>+#include <sstream>
>+#include <string>
>+
>+#include "benchmark/benchmark.h"
>+
>+namespace {
>+
>+enum StringType {
>+  kNone,
>+  kStdString,
>+};
>+
>+// Benchmarks for std::ostringstream.
>+template <StringType kOutput>
>+void BM_StdStream(benchmark::State& state) {
>+  const int num_writes = state.range(0);
>+  const int bytes_per_write = state.range(1);
>+  const std::string payload(bytes_per_write, 'x');
>+  for (auto _ : state) {
>+    std::ostringstream strm;
>+    benchmark::DoNotOptimize(strm);
>+    for (int i = 0; i != num_writes; ++i) {
>+      strm << payload;
>+    }
>+    switch (kOutput) {
>+      case kNone: {
>+        break;
>+      }
>+      case kStdString: {
>+        std::string s = strm.str();
>+        benchmark::DoNotOptimize(s);
>+        break;
>+      }
>+    }
>+  }
>+}
>+
>+// Create the stream, optionally write to it, then destroy it.
>+BENCHMARK_TEMPLATE(BM_StdStream, kNone)
>+    ->ArgPair(0, 0)
>+    ->ArgPair(1, 16)   // 16 bytes is small enough for SSO
>+    ->ArgPair(1, 256)  // 256 bytes requires heap allocation
>+    ->ArgPair(1024, 256);
>+// Create the stream, write to it, get std::string out, then destroy.
>+BENCHMARK_TEMPLATE(BM_StdStream, kStdString)
>+    ->ArgPair(1, 16)   // 16 bytes is small enough for SSO
>+    ->ArgPair(1, 256)  // 256 bytes requires heap allocation
>+    ->ArgPair(1024, 256);
>+
>+// Benchmarks for OStringStream.
>+template <StringType kOutput>
>+void BM_CustomStream(benchmark::State& state) {
>+  const int num_writes = state.range(0);
>+  const int bytes_per_write = state.range(1);
>+  const std::string payload(bytes_per_write, 'x');
>+  for (auto _ : state) {
>+    std::string out;
>+    absl::strings_internal::OStringStream strm(&out);
>+    benchmark::DoNotOptimize(strm);
>+    for (int i = 0; i != num_writes; ++i) {
>+      strm << payload;
>+    }
>+    switch (kOutput) {
>+      case kNone: {
>+        break;
>+      }
>+      case kStdString: {
>+        std::string s = out;
>+        benchmark::DoNotOptimize(s);
>+        break;
>+      }
>+    }
>+  }
>+}
>+
>+// Create the stream, optionally write to it, then destroy it.
>+BENCHMARK_TEMPLATE(BM_CustomStream, kNone)
>+    ->ArgPair(0, 0)
>+    ->ArgPair(1, 16)   // 16 bytes is small enough for SSO
>+    ->ArgPair(1, 256)  // 256 bytes requires heap allocation
>+    ->ArgPair(1024, 256);
>+// Create the stream, write to it, get std::string out, then destroy.
>+// It's not useful in practice to extract std::string from OStringStream; we
>+// measure it for completeness.
>+BENCHMARK_TEMPLATE(BM_CustomStream, kStdString)
>+    ->ArgPair(1, 16)   // 16 bytes is small enough for SSO
>+    ->ArgPair(1, 256)  // 256 bytes requires heap allocation
>+    ->ArgPair(1024, 256);
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/ostringstream_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/ostringstream_test.cc
>new file mode 100644
>index 00000000000..069a0e1fbbb
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/ostringstream_test.cc
>@@ -0,0 +1,102 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/strings/internal/ostringstream.h"
>+
>+#include <memory>
>+#include <ostream>
>+#include <string>
>+#include <type_traits>
>+
>+#include "gtest/gtest.h"
>+
>+namespace {
>+
>+TEST(OStringStream, IsOStream) {
>+  static_assert(
>+      std::is_base_of<std::ostream, absl::strings_internal::OStringStream>(),
>+      "");
>+}
>+
>+TEST(OStringStream, ConstructDestroy) {
>+  {
>+    absl::strings_internal::OStringStream strm(nullptr);
>+    EXPECT_EQ(nullptr, strm.str());
>+  }
>+  {
>+    std::string s = "abc";
>+    {
>+      absl::strings_internal::OStringStream strm(&s);
>+      EXPECT_EQ(&s, strm.str());
>+    }
>+    EXPECT_EQ("abc", s);
>+  }
>+  {
>+    std::unique_ptr<std::string> s(new std::string);
>+    absl::strings_internal::OStringStream strm(s.get());
>+    s.reset();
>+  }
>+}
>+
>+TEST(OStringStream, Str) {
>+  std::string s1;
>+  absl::strings_internal::OStringStream strm(&s1);
>+  const absl::strings_internal::OStringStream& c_strm(strm);
>+
>+  static_assert(std::is_same<decltype(strm.str()), std::string*>(), "");
>+  static_assert(std::is_same<decltype(c_strm.str()), const std::string*>(), "");
>+
>+  EXPECT_EQ(&s1, strm.str());
>+  EXPECT_EQ(&s1, c_strm.str());
>+
>+  strm.str(&s1);
>+  EXPECT_EQ(&s1, strm.str());
>+  EXPECT_EQ(&s1, c_strm.str());
>+
>+  std::string s2;
>+  strm.str(&s2);
>+  EXPECT_EQ(&s2, strm.str());
>+  EXPECT_EQ(&s2, c_strm.str());
>+
>+  strm.str(nullptr);
>+  EXPECT_EQ(nullptr, strm.str());
>+  EXPECT_EQ(nullptr, c_strm.str());
>+}
>+
>+TEST(OStreamStream, WriteToLValue) {
>+  std::string s = "abc";
>+  {
>+    absl::strings_internal::OStringStream strm(&s);
>+    EXPECT_EQ("abc", s);
>+    strm << "";
>+    EXPECT_EQ("abc", s);
>+    strm << 42;
>+    EXPECT_EQ("abc42", s);
>+    strm << 'x' << 'y';
>+    EXPECT_EQ("abc42xy", s);
>+  }
>+  EXPECT_EQ("abc42xy", s);
>+}
>+
>+TEST(OStreamStream, WriteToRValue) {
>+  std::string s = "abc";
>+  absl::strings_internal::OStringStream(&s) << "";
>+  EXPECT_EQ("abc", s);
>+  absl::strings_internal::OStringStream(&s) << 42;
>+  EXPECT_EQ("abc42", s);
>+  absl::strings_internal::OStringStream(&s) << 'x' << 'y';
>+  EXPECT_EQ("abc42xy", s);
>+}
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/resize_uninitialized.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/resize_uninitialized.h
>new file mode 100644
>index 00000000000..0157ca0245f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/resize_uninitialized.h
>@@ -0,0 +1,69 @@
>+//
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+
>+#ifndef ABSL_STRINGS_INTERNAL_RESIZE_UNINITIALIZED_H_
>+#define ABSL_STRINGS_INTERNAL_RESIZE_UNINITIALIZED_H_
>+
>+#include <string>
>+#include <utility>
>+
>+#include "absl/base/port.h"
>+#include "absl/meta/type_traits.h"  //  for void_t
>+
>+namespace absl {
>+namespace strings_internal {
>+
>+// Is a subclass of true_type or false_type, depending on whether or not
>+// T has a resize_uninitialized member.
>+template <typename T, typename = void>
>+struct HasResizeUninitialized : std::false_type {};
>+template <typename T>
>+struct HasResizeUninitialized<
>+    T, absl::void_t<decltype(std::declval<T>().resize_uninitialized(237))>>
>+    : std::true_type {};
>+
>+template <typename string_type>
>+void ResizeUninit(string_type* s, size_t new_size, std::true_type) {
>+  s->resize_uninitialized(new_size);
>+}
>+template <typename string_type>
>+void ResizeUninit(string_type* s, size_t new_size, std::false_type) {
>+  s->resize(new_size);
>+}
>+
>+// Returns true if the std::string implementation supports a resize where
>+// the new characters added to the std::string are left untouched.
>+//
>+// (A better name might be "STLStringSupportsUninitializedResize", alluding to
>+// the previous function.)
>+template <typename string_type>
>+inline constexpr bool STLStringSupportsNontrashingResize(string_type*) {
>+  return HasResizeUninitialized<string_type>();
>+}
>+
>+// Like str->resize(new_size), except any new characters added to "*str" as a
>+// result of resizing may be left uninitialized, rather than being filled with
>+// '0' bytes. Typically used when code is then going to overwrite the backing
>+// store of the std::string with known data. Uses a Google extension to std::string.
>+template <typename string_type, typename = void>
>+inline void STLStringResizeUninitialized(string_type* s, size_t new_size) {
>+  ResizeUninit(s, new_size, HasResizeUninitialized<string_type>());
>+}
>+
>+}  // namespace strings_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_STRINGS_INTERNAL_RESIZE_UNINITIALIZED_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/resize_uninitialized_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/resize_uninitialized_test.cc
>new file mode 100644
>index 00000000000..ad282efcd9b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/resize_uninitialized_test.cc
>@@ -0,0 +1,68 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/strings/internal/resize_uninitialized.h"
>+
>+#include "gtest/gtest.h"
>+
>+namespace {
>+
>+int resize_call_count = 0;
>+
>+struct resizable_string {
>+  void resize(size_t) { resize_call_count += 1; }
>+};
>+
>+int resize_uninitialized_call_count = 0;
>+
>+struct resize_uninitializable_string {
>+  void resize(size_t) { resize_call_count += 1; }
>+  void resize_uninitialized(size_t) { resize_uninitialized_call_count += 1; }
>+};
>+
>+TEST(ResizeUninit, WithAndWithout) {
>+  resize_call_count = 0;
>+  resize_uninitialized_call_count = 0;
>+  {
>+    resizable_string rs;
>+
>+    EXPECT_EQ(resize_call_count, 0);
>+    EXPECT_EQ(resize_uninitialized_call_count, 0);
>+    EXPECT_FALSE(
>+        absl::strings_internal::STLStringSupportsNontrashingResize(&rs));
>+    EXPECT_EQ(resize_call_count, 0);
>+    EXPECT_EQ(resize_uninitialized_call_count, 0);
>+    absl::strings_internal::STLStringResizeUninitialized(&rs, 237);
>+    EXPECT_EQ(resize_call_count, 1);
>+    EXPECT_EQ(resize_uninitialized_call_count, 0);
>+  }
>+
>+  resize_call_count = 0;
>+  resize_uninitialized_call_count = 0;
>+  {
>+    resize_uninitializable_string rus;
>+
>+    EXPECT_EQ(resize_call_count, 0);
>+    EXPECT_EQ(resize_uninitialized_call_count, 0);
>+    EXPECT_TRUE(
>+        absl::strings_internal::STLStringSupportsNontrashingResize(&rus));
>+    EXPECT_EQ(resize_call_count, 0);
>+    EXPECT_EQ(resize_uninitialized_call_count, 0);
>+    absl::strings_internal::STLStringResizeUninitialized(&rus, 237);
>+    EXPECT_EQ(resize_call_count, 0);
>+    EXPECT_EQ(resize_uninitialized_call_count, 1);
>+  }
>+}
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/stl_type_traits.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/stl_type_traits.h
>new file mode 100644
>index 00000000000..04c4a5323b8
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/stl_type_traits.h
>@@ -0,0 +1,246 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+
>+// Thie file provides the IsStrictlyBaseOfAndConvertibleToSTLContainer type
>+// trait metafunction to assist in working with the _GLIBCXX_DEBUG debug
>+// wrappers of STL containers.
>+//
>+// DO NOT INCLUDE THIS FILE DIRECTLY. Use this file by including
>+// absl/strings/str_split.h.
>+//
>+// IWYU pragma: private, include "absl/strings/str_split.h"
>+
>+#ifndef ABSL_STRINGS_INTERNAL_STL_TYPE_TRAITS_H_
>+#define ABSL_STRINGS_INTERNAL_STL_TYPE_TRAITS_H_
>+
>+#include <array>
>+#include <bitset>
>+#include <deque>
>+#include <forward_list>
>+#include <list>
>+#include <map>
>+#include <set>
>+#include <type_traits>
>+#include <unordered_map>
>+#include <unordered_set>
>+#include <vector>
>+
>+#include "absl/meta/type_traits.h"
>+
>+namespace absl {
>+namespace strings_internal {
>+
>+template <typename C, template <typename...> class T>
>+struct IsSpecializationImpl : std::false_type {};
>+template <template <typename...> class T, typename... Args>
>+struct IsSpecializationImpl<T<Args...>, T> : std::true_type {};
>+template <typename C, template <typename...> class T>
>+using IsSpecialization = IsSpecializationImpl<absl::decay_t<C>, T>;
>+
>+template <typename C>
>+struct IsArrayImpl : std::false_type {};
>+template <template <typename, size_t> class A, typename T, size_t N>
>+struct IsArrayImpl<A<T, N>> : std::is_same<A<T, N>, std::array<T, N>> {};
>+template <typename C>
>+using IsArray = IsArrayImpl<absl::decay_t<C>>;
>+
>+template <typename C>
>+struct IsBitsetImpl : std::false_type {};
>+template <template <size_t> class B, size_t N>
>+struct IsBitsetImpl<B<N>> : std::is_same<B<N>, std::bitset<N>> {};
>+template <typename C>
>+using IsBitset = IsBitsetImpl<absl::decay_t<C>>;
>+
>+template <typename C>
>+struct IsSTLContainer
>+    : absl::disjunction<
>+          IsArray<C>, IsBitset<C>, IsSpecialization<C, std::deque>,
>+          IsSpecialization<C, std::forward_list>,
>+          IsSpecialization<C, std::list>, IsSpecialization<C, std::map>,
>+          IsSpecialization<C, std::multimap>, IsSpecialization<C, std::set>,
>+          IsSpecialization<C, std::multiset>,
>+          IsSpecialization<C, std::unordered_map>,
>+          IsSpecialization<C, std::unordered_multimap>,
>+          IsSpecialization<C, std::unordered_set>,
>+          IsSpecialization<C, std::unordered_multiset>,
>+          IsSpecialization<C, std::vector>> {};
>+
>+template <typename C, template <typename...> class T, typename = void>
>+struct IsBaseOfSpecializationImpl : std::false_type {};
>+// IsBaseOfSpecializationImpl needs multiple partial specializations to SFINAE
>+// on the existence of container dependent types and plug them into the STL
>+// template.
>+template <typename C, template <typename, typename> class T>
>+struct IsBaseOfSpecializationImpl<
>+    C, T, absl::void_t<typename C::value_type, typename C::allocator_type>>
>+    : std::is_base_of<C,
>+                      T<typename C::value_type, typename C::allocator_type>> {};
>+template <typename C, template <typename, typename, typename> class T>
>+struct IsBaseOfSpecializationImpl<
>+    C, T,
>+    absl::void_t<typename C::key_type, typename C::key_compare,
>+                 typename C::allocator_type>>
>+    : std::is_base_of<C, T<typename C::key_type, typename C::key_compare,
>+                           typename C::allocator_type>> {};
>+template <typename C, template <typename, typename, typename, typename> class T>
>+struct IsBaseOfSpecializationImpl<
>+    C, T,
>+    absl::void_t<typename C::key_type, typename C::mapped_type,
>+                 typename C::key_compare, typename C::allocator_type>>
>+    : std::is_base_of<C,
>+                      T<typename C::key_type, typename C::mapped_type,
>+                        typename C::key_compare, typename C::allocator_type>> {
>+};
>+template <typename C, template <typename, typename, typename, typename> class T>
>+struct IsBaseOfSpecializationImpl<
>+    C, T,
>+    absl::void_t<typename C::key_type, typename C::hasher,
>+                 typename C::key_equal, typename C::allocator_type>>
>+    : std::is_base_of<C, T<typename C::key_type, typename C::hasher,
>+                           typename C::key_equal, typename C::allocator_type>> {
>+};
>+template <typename C,
>+          template <typename, typename, typename, typename, typename> class T>
>+struct IsBaseOfSpecializationImpl<
>+    C, T,
>+    absl::void_t<typename C::key_type, typename C::mapped_type,
>+                 typename C::hasher, typename C::key_equal,
>+                 typename C::allocator_type>>
>+    : std::is_base_of<C, T<typename C::key_type, typename C::mapped_type,
>+                           typename C::hasher, typename C::key_equal,
>+                           typename C::allocator_type>> {};
>+template <typename C, template <typename...> class T>
>+using IsBaseOfSpecialization = IsBaseOfSpecializationImpl<absl::decay_t<C>, T>;
>+
>+template <typename C>
>+struct IsBaseOfArrayImpl : std::false_type {};
>+template <template <typename, size_t> class A, typename T, size_t N>
>+struct IsBaseOfArrayImpl<A<T, N>> : std::is_base_of<A<T, N>, std::array<T, N>> {
>+};
>+template <typename C>
>+using IsBaseOfArray = IsBaseOfArrayImpl<absl::decay_t<C>>;
>+
>+template <typename C>
>+struct IsBaseOfBitsetImpl : std::false_type {};
>+template <template <size_t> class B, size_t N>
>+struct IsBaseOfBitsetImpl<B<N>> : std::is_base_of<B<N>, std::bitset<N>> {};
>+template <typename C>
>+using IsBaseOfBitset = IsBaseOfBitsetImpl<absl::decay_t<C>>;
>+
>+template <typename C>
>+struct IsBaseOfSTLContainer
>+    : absl::disjunction<IsBaseOfArray<C>, IsBaseOfBitset<C>,
>+                        IsBaseOfSpecialization<C, std::deque>,
>+                        IsBaseOfSpecialization<C, std::forward_list>,
>+                        IsBaseOfSpecialization<C, std::list>,
>+                        IsBaseOfSpecialization<C, std::map>,
>+                        IsBaseOfSpecialization<C, std::multimap>,
>+                        IsBaseOfSpecialization<C, std::set>,
>+                        IsBaseOfSpecialization<C, std::multiset>,
>+                        IsBaseOfSpecialization<C, std::unordered_map>,
>+                        IsBaseOfSpecialization<C, std::unordered_multimap>,
>+                        IsBaseOfSpecialization<C, std::unordered_set>,
>+                        IsBaseOfSpecialization<C, std::unordered_multiset>,
>+                        IsBaseOfSpecialization<C, std::vector>> {};
>+
>+template <typename C, template <typename...> class T, typename = void>
>+struct IsConvertibleToSpecializationImpl : std::false_type {};
>+// IsConvertibleToSpecializationImpl needs multiple partial specializations to
>+// SFINAE on the existence of container dependent types and plug them into the
>+// STL template.
>+template <typename C, template <typename, typename> class T>
>+struct IsConvertibleToSpecializationImpl<
>+    C, T, absl::void_t<typename C::value_type, typename C::allocator_type>>
>+    : std::is_convertible<
>+          C, T<typename C::value_type, typename C::allocator_type>> {};
>+template <typename C, template <typename, typename, typename> class T>
>+struct IsConvertibleToSpecializationImpl<
>+    C, T,
>+    absl::void_t<typename C::key_type, typename C::key_compare,
>+                 typename C::allocator_type>>
>+    : std::is_convertible<C, T<typename C::key_type, typename C::key_compare,
>+                               typename C::allocator_type>> {};
>+template <typename C, template <typename, typename, typename, typename> class T>
>+struct IsConvertibleToSpecializationImpl<
>+    C, T,
>+    absl::void_t<typename C::key_type, typename C::mapped_type,
>+                 typename C::key_compare, typename C::allocator_type>>
>+    : std::is_convertible<
>+          C, T<typename C::key_type, typename C::mapped_type,
>+               typename C::key_compare, typename C::allocator_type>> {};
>+template <typename C, template <typename, typename, typename, typename> class T>
>+struct IsConvertibleToSpecializationImpl<
>+    C, T,
>+    absl::void_t<typename C::key_type, typename C::hasher,
>+                 typename C::key_equal, typename C::allocator_type>>
>+    : std::is_convertible<
>+          C, T<typename C::key_type, typename C::hasher, typename C::key_equal,
>+               typename C::allocator_type>> {};
>+template <typename C,
>+          template <typename, typename, typename, typename, typename> class T>
>+struct IsConvertibleToSpecializationImpl<
>+    C, T,
>+    absl::void_t<typename C::key_type, typename C::mapped_type,
>+                 typename C::hasher, typename C::key_equal,
>+                 typename C::allocator_type>>
>+    : std::is_convertible<C, T<typename C::key_type, typename C::mapped_type,
>+                               typename C::hasher, typename C::key_equal,
>+                               typename C::allocator_type>> {};
>+template <typename C, template <typename...> class T>
>+using IsConvertibleToSpecialization =
>+    IsConvertibleToSpecializationImpl<absl::decay_t<C>, T>;
>+
>+template <typename C>
>+struct IsConvertibleToArrayImpl : std::false_type {};
>+template <template <typename, size_t> class A, typename T, size_t N>
>+struct IsConvertibleToArrayImpl<A<T, N>>
>+    : std::is_convertible<A<T, N>, std::array<T, N>> {};
>+template <typename C>
>+using IsConvertibleToArray = IsConvertibleToArrayImpl<absl::decay_t<C>>;
>+
>+template <typename C>
>+struct IsConvertibleToBitsetImpl : std::false_type {};
>+template <template <size_t> class B, size_t N>
>+struct IsConvertibleToBitsetImpl<B<N>>
>+    : std::is_convertible<B<N>, std::bitset<N>> {};
>+template <typename C>
>+using IsConvertibleToBitset = IsConvertibleToBitsetImpl<absl::decay_t<C>>;
>+
>+template <typename C>
>+struct IsConvertibleToSTLContainer
>+    : absl::disjunction<
>+          IsConvertibleToArray<C>, IsConvertibleToBitset<C>,
>+          IsConvertibleToSpecialization<C, std::deque>,
>+          IsConvertibleToSpecialization<C, std::forward_list>,
>+          IsConvertibleToSpecialization<C, std::list>,
>+          IsConvertibleToSpecialization<C, std::map>,
>+          IsConvertibleToSpecialization<C, std::multimap>,
>+          IsConvertibleToSpecialization<C, std::set>,
>+          IsConvertibleToSpecialization<C, std::multiset>,
>+          IsConvertibleToSpecialization<C, std::unordered_map>,
>+          IsConvertibleToSpecialization<C, std::unordered_multimap>,
>+          IsConvertibleToSpecialization<C, std::unordered_set>,
>+          IsConvertibleToSpecialization<C, std::unordered_multiset>,
>+          IsConvertibleToSpecialization<C, std::vector>> {};
>+
>+template <typename C>
>+struct IsStrictlyBaseOfAndConvertibleToSTLContainer
>+    : absl::conjunction<absl::negation<IsSTLContainer<C>>,
>+                        IsBaseOfSTLContainer<C>,
>+                        IsConvertibleToSTLContainer<C>> {};
>+
>+}  // namespace strings_internal
>+}  // namespace absl
>+#endif  // ABSL_STRINGS_INTERNAL_STL_TYPE_TRAITS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/arg.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/arg.cc
>new file mode 100644
>index 00000000000..eafb068fe28
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/arg.cc
>@@ -0,0 +1,399 @@
>+//
>+// POSIX spec:
>+//   http://pubs.opengroup.org/onlinepubs/009695399/functions/fprintf.html
>+//
>+#include "absl/strings/internal/str_format/arg.h"
>+
>+#include <cassert>
>+#include <cerrno>
>+#include <cstdlib>
>+#include <string>
>+#include <type_traits>
>+
>+#include "absl/base/port.h"
>+#include "absl/strings/internal/str_format/float_conversion.h"
>+
>+namespace absl {
>+namespace str_format_internal {
>+namespace {
>+
>+const char kDigit[2][32] = { "0123456789abcdef", "0123456789ABCDEF" };
>+
>+// Reduce *capacity by s.size(), clipped to a 0 minimum.
>+void ReducePadding(string_view s, size_t *capacity) {
>+  *capacity = Excess(s.size(), *capacity);
>+}
>+
>+// Reduce *capacity by n, clipped to a 0 minimum.
>+void ReducePadding(size_t n, size_t *capacity) {
>+  *capacity = Excess(n, *capacity);
>+}
>+
>+template <typename T>
>+struct MakeUnsigned : std::make_unsigned<T> {};
>+template <>
>+struct MakeUnsigned<absl::uint128> {
>+  using type = absl::uint128;
>+};
>+
>+template <typename T>
>+struct IsSigned : std::is_signed<T> {};
>+template <>
>+struct IsSigned<absl::uint128> : std::false_type {};
>+
>+class ConvertedIntInfo {
>+ public:
>+  template <typename T>
>+  ConvertedIntInfo(T v, ConversionChar conv) {
>+    using Unsigned = typename MakeUnsigned<T>::type;
>+    auto u = static_cast<Unsigned>(v);
>+    if (IsNeg(v)) {
>+      is_neg_ = true;
>+      u = Unsigned{} - u;
>+    } else {
>+      is_neg_ = false;
>+    }
>+    UnsignedToStringRight(u, conv);
>+  }
>+
>+  string_view digits() const {
>+    return {end() - size_, static_cast<size_t>(size_)};
>+  }
>+  bool is_neg() const { return is_neg_; }
>+
>+ private:
>+  template <typename T, bool IsSigned>
>+  struct IsNegImpl {
>+    static bool Eval(T v) { return v < 0; }
>+  };
>+  template <typename T>
>+  struct IsNegImpl<T, false> {
>+    static bool Eval(T) {
>+      return false;
>+    }
>+  };
>+
>+  template <typename T>
>+  bool IsNeg(T v) {
>+    return IsNegImpl<T, IsSigned<T>::value>::Eval(v);
>+  }
>+
>+  template <typename T>
>+  void UnsignedToStringRight(T u, ConversionChar conv) {
>+    char *p = end();
>+    switch (conv.radix()) {
>+      default:
>+      case 10:
>+        for (; u; u /= 10)
>+          *--p = static_cast<char>('0' + static_cast<size_t>(u % 10));
>+        break;
>+      case 8:
>+        for (; u; u /= 8)
>+          *--p = static_cast<char>('0' + static_cast<size_t>(u % 8));
>+        break;
>+      case 16: {
>+        const char *digits = kDigit[conv.upper() ? 1 : 0];
>+        for (; u; u /= 16) *--p = digits[static_cast<size_t>(u % 16)];
>+        break;
>+      }
>+    }
>+    size_ = static_cast<int>(end() - p);
>+  }
>+
>+  const char *end() const { return storage_ + sizeof(storage_); }
>+  char *end() { return storage_ + sizeof(storage_); }
>+
>+  bool is_neg_;
>+  int size_;
>+  // Max size: 128 bit value as octal -> 43 digits
>+  char storage_[128 / 3 + 1];
>+};
>+
>+// Note: 'o' conversions do not have a base indicator, it's just that
>+// the '#' flag is specified to modify the precision for 'o' conversions.
>+string_view BaseIndicator(const ConvertedIntInfo &info,
>+                          const ConversionSpec &conv) {
>+  bool alt = conv.flags().alt;
>+  int radix = conv.conv().radix();
>+  if (conv.conv().id() == ConversionChar::p)
>+    alt = true;  // always show 0x for %p.
>+  // From the POSIX description of '#' flag:
>+  //   "For x or X conversion specifiers, a non-zero result shall have
>+  //   0x (or 0X) prefixed to it."
>+  if (alt && radix == 16 && !info.digits().empty()) {
>+    if (conv.conv().upper()) return "0X";
>+    return "0x";
>+  }
>+  return {};
>+}
>+
>+string_view SignColumn(bool neg, const ConversionSpec &conv) {
>+  if (conv.conv().is_signed()) {
>+    if (neg) return "-";
>+    if (conv.flags().show_pos) return "+";
>+    if (conv.flags().sign_col) return " ";
>+  }
>+  return {};
>+}
>+
>+bool ConvertCharImpl(unsigned char v, const ConversionSpec &conv,
>+                     FormatSinkImpl *sink) {
>+  size_t fill = 0;
>+  if (conv.width() >= 0) fill = conv.width();
>+  ReducePadding(1, &fill);
>+  if (!conv.flags().left) sink->Append(fill, ' ');
>+  sink->Append(1, v);
>+  if (conv.flags().left) sink->Append(fill, ' ');
>+  return true;
>+}
>+
>+bool ConvertIntImplInner(const ConvertedIntInfo &info,
>+                         const ConversionSpec &conv, FormatSinkImpl *sink) {
>+  // Print as a sequence of Substrings:
>+  //   [left_spaces][sign][base_indicator][zeroes][formatted][right_spaces]
>+  size_t fill = 0;
>+  if (conv.width() >= 0) fill = conv.width();
>+
>+  string_view formatted = info.digits();
>+  ReducePadding(formatted, &fill);
>+
>+  string_view sign = SignColumn(info.is_neg(), conv);
>+  ReducePadding(sign, &fill);
>+
>+  string_view base_indicator = BaseIndicator(info, conv);
>+  ReducePadding(base_indicator, &fill);
>+
>+  int precision = conv.precision();
>+  bool precision_specified = precision >= 0;
>+  if (!precision_specified)
>+    precision = 1;
>+
>+  if (conv.flags().alt && conv.conv().id() == ConversionChar::o) {
>+    // From POSIX description of the '#' (alt) flag:
>+    //   "For o conversion, it increases the precision (if necessary) to
>+    //   force the first digit of the result to be zero."
>+    if (formatted.empty() || *formatted.begin() != '0') {
>+      int needed = static_cast<int>(formatted.size()) + 1;
>+      precision = std::max(precision, needed);
>+    }
>+  }
>+
>+  size_t num_zeroes = Excess(formatted.size(), precision);
>+  ReducePadding(num_zeroes, &fill);
>+
>+  size_t num_left_spaces = !conv.flags().left ? fill : 0;
>+  size_t num_right_spaces = conv.flags().left ? fill : 0;
>+
>+  // From POSIX description of the '0' (zero) flag:
>+  //   "For d, i, o, u, x, and X conversion specifiers, if a precision
>+  //   is specified, the '0' flag is ignored."
>+  if (!precision_specified && conv.flags().zero) {
>+    num_zeroes += num_left_spaces;
>+    num_left_spaces = 0;
>+  }
>+
>+  sink->Append(num_left_spaces, ' ');
>+  sink->Append(sign);
>+  sink->Append(base_indicator);
>+  sink->Append(num_zeroes, '0');
>+  sink->Append(formatted);
>+  sink->Append(num_right_spaces, ' ');
>+  return true;
>+}
>+
>+template <typename T>
>+bool ConvertIntImplInner(T v, const ConversionSpec &conv,
>+                         FormatSinkImpl *sink) {
>+  ConvertedIntInfo info(v, conv.conv());
>+  if (conv.flags().basic && conv.conv().id() != ConversionChar::p) {
>+    if (info.is_neg()) sink->Append(1, '-');
>+    if (info.digits().empty()) {
>+      sink->Append(1, '0');
>+    } else {
>+      sink->Append(info.digits());
>+    }
>+    return true;
>+  }
>+  return ConvertIntImplInner(info, conv, sink);
>+}
>+
>+template <typename T>
>+bool ConvertIntArg(T v, const ConversionSpec &conv, FormatSinkImpl *sink) {
>+  if (conv.conv().is_float()) {
>+    return FormatConvertImpl(static_cast<double>(v), conv, sink).value;
>+  }
>+  if (conv.conv().id() == ConversionChar::c)
>+    return ConvertCharImpl(static_cast<unsigned char>(v), conv, sink);
>+  if (!conv.conv().is_integral())
>+    return false;
>+  if (!conv.conv().is_signed() && IsSigned<T>::value) {
>+    using U = typename MakeUnsigned<T>::type;
>+    return FormatConvertImpl(static_cast<U>(v), conv, sink).value;
>+  }
>+  return ConvertIntImplInner(v, conv, sink);
>+}
>+
>+template <typename T>
>+bool ConvertFloatArg(T v, const ConversionSpec &conv, FormatSinkImpl *sink) {
>+  return conv.conv().is_float() && ConvertFloatImpl(v, conv, sink);
>+}
>+
>+inline bool ConvertStringArg(string_view v, const ConversionSpec &conv,
>+                             FormatSinkImpl *sink) {
>+  if (conv.conv().id() != ConversionChar::s)
>+    return false;
>+  if (conv.flags().basic) {
>+    sink->Append(v);
>+    return true;
>+  }
>+  return sink->PutPaddedString(v, conv.width(), conv.precision(),
>+                               conv.flags().left);
>+}
>+
>+}  // namespace
>+
>+// ==================== Strings ====================
>+ConvertResult<Conv::s> FormatConvertImpl(const std::string &v,
>+                                         const ConversionSpec &conv,
>+                                         FormatSinkImpl *sink) {
>+  return {ConvertStringArg(v, conv, sink)};
>+}
>+
>+ConvertResult<Conv::s> FormatConvertImpl(string_view v,
>+                                         const ConversionSpec &conv,
>+                                         FormatSinkImpl *sink) {
>+  return {ConvertStringArg(v, conv, sink)};
>+}
>+
>+ConvertResult<Conv::s | Conv::p> FormatConvertImpl(const char *v,
>+                                                   const ConversionSpec &conv,
>+                                                   FormatSinkImpl *sink) {
>+  if (conv.conv().id() == ConversionChar::p)
>+    return {FormatConvertImpl(VoidPtr(v), conv, sink).value};
>+  size_t len;
>+  if (v == nullptr) {
>+    len = 0;
>+  } else if (conv.precision() < 0) {
>+    len = std::strlen(v);
>+  } else {
>+    // If precision is set, we look for the null terminator on the valid range.
>+    len = std::find(v, v + conv.precision(), '\0') - v;
>+  }
>+  return {ConvertStringArg(string_view(v, len), conv, sink)};
>+}
>+
>+// ==================== Raw pointers ====================
>+ConvertResult<Conv::p> FormatConvertImpl(VoidPtr v, const ConversionSpec &conv,
>+                                         FormatSinkImpl *sink) {
>+  if (conv.conv().id() != ConversionChar::p)
>+    return {false};
>+  if (!v.value) {
>+    sink->Append("(nil)");
>+    return {true};
>+  }
>+  return {ConvertIntImplInner(v.value, conv, sink)};
>+}
>+
>+// ==================== Floats ====================
>+FloatingConvertResult FormatConvertImpl(float v, const ConversionSpec &conv,
>+                                        FormatSinkImpl *sink) {
>+  return {ConvertFloatArg(v, conv, sink)};
>+}
>+FloatingConvertResult FormatConvertImpl(double v, const ConversionSpec &conv,
>+                                        FormatSinkImpl *sink) {
>+  return {ConvertFloatArg(v, conv, sink)};
>+}
>+FloatingConvertResult FormatConvertImpl(long double v,
>+                                        const ConversionSpec &conv,
>+                                        FormatSinkImpl *sink) {
>+  return {ConvertFloatArg(v, conv, sink)};
>+}
>+
>+// ==================== Chars ====================
>+IntegralConvertResult FormatConvertImpl(char v, const ConversionSpec &conv,
>+                                        FormatSinkImpl *sink) {
>+  return {ConvertIntArg(v, conv, sink)};
>+}
>+IntegralConvertResult FormatConvertImpl(signed char v,
>+                                        const ConversionSpec &conv,
>+                                        FormatSinkImpl *sink) {
>+  return {ConvertIntArg(v, conv, sink)};
>+}
>+IntegralConvertResult FormatConvertImpl(unsigned char v,
>+                                        const ConversionSpec &conv,
>+                                        FormatSinkImpl *sink) {
>+  return {ConvertIntArg(v, conv, sink)};
>+}
>+
>+// ==================== Ints ====================
>+IntegralConvertResult FormatConvertImpl(short v,  // NOLINT
>+                                        const ConversionSpec &conv,
>+                                        FormatSinkImpl *sink) {
>+  return {ConvertIntArg(v, conv, sink)};
>+}
>+IntegralConvertResult FormatConvertImpl(unsigned short v,  // NOLINT
>+                                        const ConversionSpec &conv,
>+                                        FormatSinkImpl *sink) {
>+  return {ConvertIntArg(v, conv, sink)};
>+}
>+IntegralConvertResult FormatConvertImpl(int v, const ConversionSpec &conv,
>+                                        FormatSinkImpl *sink) {
>+  return {ConvertIntArg(v, conv, sink)};
>+}
>+IntegralConvertResult FormatConvertImpl(unsigned v, const ConversionSpec &conv,
>+                                        FormatSinkImpl *sink) {
>+  return {ConvertIntArg(v, conv, sink)};
>+}
>+IntegralConvertResult FormatConvertImpl(long v,  // NOLINT
>+                                        const ConversionSpec &conv,
>+                                        FormatSinkImpl *sink) {
>+  return {ConvertIntArg(v, conv, sink)};
>+}
>+IntegralConvertResult FormatConvertImpl(unsigned long v,  // NOLINT
>+                                        const ConversionSpec &conv,
>+                                        FormatSinkImpl *sink) {
>+  return {ConvertIntArg(v, conv, sink)};
>+}
>+IntegralConvertResult FormatConvertImpl(long long v,  // NOLINT
>+                                        const ConversionSpec &conv,
>+                                        FormatSinkImpl *sink) {
>+  return {ConvertIntArg(v, conv, sink)};
>+}
>+IntegralConvertResult FormatConvertImpl(unsigned long long v,  // NOLINT
>+                                        const ConversionSpec &conv,
>+                                        FormatSinkImpl *sink) {
>+  return {ConvertIntArg(v, conv, sink)};
>+}
>+IntegralConvertResult FormatConvertImpl(absl::uint128 v,
>+                                        const ConversionSpec &conv,
>+                                        FormatSinkImpl *sink) {
>+  return {ConvertIntArg(v, conv, sink)};
>+}
>+
>+template struct FormatArgImpl::TypedVTable<str_format_internal::VoidPtr>;
>+
>+template struct FormatArgImpl::TypedVTable<bool>;
>+template struct FormatArgImpl::TypedVTable<char>;
>+template struct FormatArgImpl::TypedVTable<signed char>;
>+template struct FormatArgImpl::TypedVTable<unsigned char>;
>+template struct FormatArgImpl::TypedVTable<short>;           // NOLINT
>+template struct FormatArgImpl::TypedVTable<unsigned short>;  // NOLINT
>+template struct FormatArgImpl::TypedVTable<int>;
>+template struct FormatArgImpl::TypedVTable<unsigned>;
>+template struct FormatArgImpl::TypedVTable<long>;                // NOLINT
>+template struct FormatArgImpl::TypedVTable<unsigned long>;       // NOLINT
>+template struct FormatArgImpl::TypedVTable<long long>;           // NOLINT
>+template struct FormatArgImpl::TypedVTable<unsigned long long>;  // NOLINT
>+template struct FormatArgImpl::TypedVTable<absl::uint128>;
>+
>+template struct FormatArgImpl::TypedVTable<float>;
>+template struct FormatArgImpl::TypedVTable<double>;
>+template struct FormatArgImpl::TypedVTable<long double>;
>+
>+template struct FormatArgImpl::TypedVTable<const char *>;
>+template struct FormatArgImpl::TypedVTable<std::string>;
>+template struct FormatArgImpl::TypedVTable<string_view>;
>+
>+}  // namespace str_format_internal
>+
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/arg.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/arg.h
>new file mode 100644
>index 00000000000..a9562188ea9
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/arg.h
>@@ -0,0 +1,434 @@
>+#ifndef ABSL_STRINGS_INTERNAL_STR_FORMAT_ARG_H_
>+#define ABSL_STRINGS_INTERNAL_STR_FORMAT_ARG_H_
>+
>+#include <string.h>
>+#include <wchar.h>
>+
>+#include <cstdio>
>+#include <iomanip>
>+#include <limits>
>+#include <sstream>
>+#include <string>
>+#include <type_traits>
>+
>+#include "absl/base/port.h"
>+#include "absl/meta/type_traits.h"
>+#include "absl/numeric/int128.h"
>+#include "absl/strings/internal/str_format/extension.h"
>+#include "absl/strings/string_view.h"
>+
>+class Cord;
>+class CordReader;
>+
>+namespace absl {
>+
>+class FormatCountCapture;
>+class FormatSink;
>+
>+namespace str_format_internal {
>+
>+template <typename T, typename = void>
>+struct HasUserDefinedConvert : std::false_type {};
>+
>+template <typename T>
>+struct HasUserDefinedConvert<
>+    T, void_t<decltype(AbslFormatConvert(
>+           std::declval<const T&>(), std::declval<const ConversionSpec&>(),
>+           std::declval<FormatSink*>()))>> : std::true_type {};
>+template <typename T>
>+class StreamedWrapper;
>+
>+// If 'v' can be converted (in the printf sense) according to 'conv',
>+// then convert it, appending to `sink` and return `true`.
>+// Otherwise fail and return `false`.
>+// Raw pointers.
>+struct VoidPtr {
>+  VoidPtr() = default;
>+  template <typename T,
>+            decltype(reinterpret_cast<uintptr_t>(std::declval<T*>())) = 0>
>+  VoidPtr(T* ptr)  // NOLINT
>+      : value(ptr ? reinterpret_cast<uintptr_t>(ptr) : 0) {}
>+  uintptr_t value;
>+};
>+ConvertResult<Conv::p> FormatConvertImpl(VoidPtr v, const ConversionSpec& conv,
>+                                         FormatSinkImpl* sink);
>+
>+// Strings.
>+ConvertResult<Conv::s> FormatConvertImpl(const std::string& v,
>+                                         const ConversionSpec& conv,
>+                                         FormatSinkImpl* sink);
>+ConvertResult<Conv::s> FormatConvertImpl(string_view v,
>+                                         const ConversionSpec& conv,
>+                                         FormatSinkImpl* sink);
>+ConvertResult<Conv::s | Conv::p> FormatConvertImpl(const char* v,
>+                                                   const ConversionSpec& conv,
>+                                                   FormatSinkImpl* sink);
>+template <class AbslCord,
>+          typename std::enable_if<
>+              std::is_same<AbslCord, ::Cord>::value>::type* = nullptr,
>+          class AbslCordReader = ::CordReader>
>+ConvertResult<Conv::s> FormatConvertImpl(const AbslCord& value,
>+                                         const ConversionSpec& conv,
>+                                         FormatSinkImpl* sink) {
>+  if (conv.conv().id() != ConversionChar::s) return {false};
>+
>+  bool is_left = conv.flags().left;
>+  size_t space_remaining = 0;
>+
>+  int width = conv.width();
>+  if (width >= 0) space_remaining = width;
>+
>+  size_t to_write = value.size();
>+
>+  int precision = conv.precision();
>+  if (precision >= 0)
>+    to_write = std::min(to_write, static_cast<size_t>(precision));
>+
>+  space_remaining = Excess(to_write, space_remaining);
>+
>+  if (space_remaining > 0 && !is_left) sink->Append(space_remaining, ' ');
>+
>+  string_view piece;
>+  for (AbslCordReader reader(value);
>+       to_write > 0 && reader.ReadFragment(&piece); to_write -= piece.size()) {
>+    if (piece.size() > to_write) piece.remove_suffix(piece.size() - to_write);
>+    sink->Append(piece);
>+  }
>+
>+  if (space_remaining > 0 && is_left) sink->Append(space_remaining, ' ');
>+  return {true};
>+}
>+
>+using IntegralConvertResult =
>+    ConvertResult<Conv::c | Conv::numeric | Conv::star>;
>+using FloatingConvertResult = ConvertResult<Conv::floating>;
>+
>+// Floats.
>+FloatingConvertResult FormatConvertImpl(float v, const ConversionSpec& conv,
>+                                        FormatSinkImpl* sink);
>+FloatingConvertResult FormatConvertImpl(double v, const ConversionSpec& conv,
>+                                        FormatSinkImpl* sink);
>+FloatingConvertResult FormatConvertImpl(long double v,
>+                                        const ConversionSpec& conv,
>+                                        FormatSinkImpl* sink);
>+
>+// Chars.
>+IntegralConvertResult FormatConvertImpl(char v, const ConversionSpec& conv,
>+                                        FormatSinkImpl* sink);
>+IntegralConvertResult FormatConvertImpl(signed char v,
>+                                        const ConversionSpec& conv,
>+                                        FormatSinkImpl* sink);
>+IntegralConvertResult FormatConvertImpl(unsigned char v,
>+                                        const ConversionSpec& conv,
>+                                        FormatSinkImpl* sink);
>+
>+// Ints.
>+IntegralConvertResult FormatConvertImpl(short v,  // NOLINT
>+                                        const ConversionSpec& conv,
>+                                        FormatSinkImpl* sink);
>+IntegralConvertResult FormatConvertImpl(unsigned short v,  // NOLINT
>+                                        const ConversionSpec& conv,
>+                                        FormatSinkImpl* sink);
>+IntegralConvertResult FormatConvertImpl(int v, const ConversionSpec& conv,
>+                                        FormatSinkImpl* sink);
>+IntegralConvertResult FormatConvertImpl(unsigned v, const ConversionSpec& conv,
>+                                        FormatSinkImpl* sink);
>+IntegralConvertResult FormatConvertImpl(long v,  // NOLINT
>+                                        const ConversionSpec& conv,
>+                                        FormatSinkImpl* sink);
>+IntegralConvertResult FormatConvertImpl(unsigned long v,  // NOLINT
>+                                        const ConversionSpec& conv,
>+                                        FormatSinkImpl* sink);
>+IntegralConvertResult FormatConvertImpl(long long v,  // NOLINT
>+                                        const ConversionSpec& conv,
>+                                        FormatSinkImpl* sink);
>+IntegralConvertResult FormatConvertImpl(unsigned long long v,  // NOLINT
>+                                        const ConversionSpec& conv,
>+                                        FormatSinkImpl* sink);
>+IntegralConvertResult FormatConvertImpl(uint128 v, const ConversionSpec& conv,
>+                                        FormatSinkImpl* sink);
>+template <typename T, enable_if_t<std::is_same<T, bool>::value, int> = 0>
>+IntegralConvertResult FormatConvertImpl(T v, const ConversionSpec& conv,
>+                                        FormatSinkImpl* sink) {
>+  return FormatConvertImpl(static_cast<int>(v), conv, sink);
>+}
>+
>+// We provide this function to help the checker, but it is never defined.
>+// FormatArgImpl will use the underlying Convert functions instead.
>+template <typename T>
>+typename std::enable_if<std::is_enum<T>::value &&
>+                            !HasUserDefinedConvert<T>::value,
>+                        IntegralConvertResult>::type
>+FormatConvertImpl(T v, const ConversionSpec& conv, FormatSinkImpl* sink);
>+
>+template <typename T>
>+ConvertResult<Conv::s> FormatConvertImpl(const StreamedWrapper<T>& v,
>+                                         const ConversionSpec& conv,
>+                                         FormatSinkImpl* out) {
>+  std::ostringstream oss;
>+  oss << v.v_;
>+  if (!oss) return {false};
>+  return str_format_internal::FormatConvertImpl(oss.str(), conv, out);
>+}
>+
>+// Use templates and dependent types to delay evaluation of the function
>+// until after FormatCountCapture is fully defined.
>+struct FormatCountCaptureHelper {
>+  template <class T = int>
>+  static ConvertResult<Conv::n> ConvertHelper(const FormatCountCapture& v,
>+                                              const ConversionSpec& conv,
>+                                              FormatSinkImpl* sink) {
>+    const absl::enable_if_t<sizeof(T) != 0, FormatCountCapture>& v2 = v;
>+
>+    if (conv.conv().id() != str_format_internal::ConversionChar::n)
>+      return {false};
>+    *v2.p_ = static_cast<int>(sink->size());
>+    return {true};
>+  }
>+};
>+
>+template <class T = int>
>+ConvertResult<Conv::n> FormatConvertImpl(const FormatCountCapture& v,
>+                                         const ConversionSpec& conv,
>+                                         FormatSinkImpl* sink) {
>+  return FormatCountCaptureHelper::ConvertHelper(v, conv, sink);
>+}
>+
>+// Helper friend struct to hide implementation details from the public API of
>+// FormatArgImpl.
>+struct FormatArgImplFriend {
>+  template <typename Arg>
>+  static bool ToInt(Arg arg, int* out) {
>+    if (!arg.vtbl_->to_int) return false;
>+    *out = arg.vtbl_->to_int(arg.data_);
>+    return true;
>+  }
>+
>+  template <typename Arg>
>+  static bool Convert(Arg arg, const str_format_internal::ConversionSpec& conv,
>+                      FormatSinkImpl* out) {
>+    return arg.vtbl_->convert(arg.data_, conv, out);
>+  }
>+
>+  template <typename Arg>
>+  static const void* GetVTablePtrForTest(Arg arg) {
>+    return arg.vtbl_;
>+  }
>+};
>+
>+// A type-erased handle to a format argument.
>+class FormatArgImpl {
>+ private:
>+  enum { kInlinedSpace = 8 };
>+
>+  using VoidPtr = str_format_internal::VoidPtr;
>+
>+  union Data {
>+    const void* ptr;
>+    const volatile void* volatile_ptr;
>+    char buf[kInlinedSpace];
>+  };
>+
>+  struct VTable {
>+    bool (*convert)(Data, const str_format_internal::ConversionSpec& conv,
>+                    FormatSinkImpl* out);
>+    int (*to_int)(Data);
>+  };
>+
>+  template <typename T>
>+  struct store_by_value
>+      : std::integral_constant<bool, (sizeof(T) <= kInlinedSpace) &&
>+                                         (std::is_integral<T>::value ||
>+                                          std::is_floating_point<T>::value ||
>+                                          std::is_pointer<T>::value ||
>+                                          std::is_same<VoidPtr, T>::value)> {};
>+
>+  enum StoragePolicy { ByPointer, ByVolatilePointer, ByValue };
>+  template <typename T>
>+  struct storage_policy
>+      : std::integral_constant<StoragePolicy,
>+                               (std::is_volatile<T>::value
>+                                    ? ByVolatilePointer
>+                                    : (store_by_value<T>::value ? ByValue
>+                                                                : ByPointer))> {
>+  };
>+
>+  // An instance of an FormatArgImpl::VTable suitable for 'T'.
>+  template <typename T>
>+  struct TypedVTable;
>+
>+  // To reduce the number of vtables we will decay values before hand.
>+  // Anything with a user-defined Convert will get its own vtable.
>+  // For everything else:
>+  //   - Decay char* and char arrays into `const char*`
>+  //   - Decay any other pointer to `const void*`
>+  //   - Decay all enums to their underlying type.
>+  //   - Decay function pointers to void*.
>+  template <typename T, typename = void>
>+  struct DecayType {
>+    static constexpr bool kHasUserDefined =
>+        str_format_internal::HasUserDefinedConvert<T>::value;
>+    using type = typename std::conditional<
>+        !kHasUserDefined && std::is_convertible<T, const char*>::value,
>+        const char*,
>+        typename std::conditional<!kHasUserDefined &&
>+                                      std::is_convertible<T, VoidPtr>::value,
>+                                  VoidPtr, const T&>::type>::type;
>+  };
>+  template <typename T>
>+  struct DecayType<T,
>+                   typename std::enable_if<
>+                       !str_format_internal::HasUserDefinedConvert<T>::value &&
>+                       std::is_enum<T>::value>::type> {
>+    using type = typename std::underlying_type<T>::type;
>+  };
>+
>+ public:
>+  template <typename T>
>+  explicit FormatArgImpl(const T& value) {
>+    using D = typename DecayType<T>::type;
>+    static_assert(
>+        std::is_same<D, const T&>::value || storage_policy<D>::value == ByValue,
>+        "Decayed types must be stored by value");
>+    Init(static_cast<D>(value));
>+  }
>+
>+ private:
>+  friend struct str_format_internal::FormatArgImplFriend;
>+  template <typename T, StoragePolicy = storage_policy<T>::value>
>+  struct Manager;
>+
>+  template <typename T>
>+  struct Manager<T, ByPointer> {
>+    static Data SetValue(const T& value) {
>+      Data data;
>+      data.ptr = &value;
>+      return data;
>+    }
>+
>+    static const T& Value(Data arg) { return *static_cast<const T*>(arg.ptr); }
>+  };
>+
>+  template <typename T>
>+  struct Manager<T, ByVolatilePointer> {
>+    static Data SetValue(const T& value) {
>+      Data data;
>+      data.volatile_ptr = &value;
>+      return data;
>+    }
>+
>+    static const T& Value(Data arg) {
>+      return *static_cast<const T*>(arg.volatile_ptr);
>+    }
>+  };
>+
>+  template <typename T>
>+  struct Manager<T, ByValue> {
>+    static Data SetValue(const T& value) {
>+      Data data;
>+      memcpy(data.buf, &value, sizeof(value));
>+      return data;
>+    }
>+
>+    static T Value(Data arg) {
>+      T value;
>+      memcpy(&value, arg.buf, sizeof(T));
>+      return value;
>+    }
>+  };
>+
>+  template <typename T>
>+  void Init(const T& value);
>+
>+  template <typename T>
>+  static int ToIntVal(const T& val) {
>+    using CommonType = typename std::conditional<std::is_signed<T>::value,
>+                                                 int64_t, uint64_t>::type;
>+    if (static_cast<CommonType>(val) >
>+        static_cast<CommonType>(std::numeric_limits<int>::max())) {
>+      return std::numeric_limits<int>::max();
>+    } else if (std::is_signed<T>::value &&
>+               static_cast<CommonType>(val) <
>+                   static_cast<CommonType>(std::numeric_limits<int>::min())) {
>+      return std::numeric_limits<int>::min();
>+    }
>+    return static_cast<int>(val);
>+  }
>+
>+  Data data_;
>+  const VTable* vtbl_;
>+};
>+
>+template <typename T>
>+struct FormatArgImpl::TypedVTable {
>+ private:
>+  static bool ConvertImpl(Data arg,
>+                          const str_format_internal::ConversionSpec& conv,
>+                          FormatSinkImpl* out) {
>+    return str_format_internal::FormatConvertImpl(Manager<T>::Value(arg), conv,
>+                                                  out)
>+        .value;
>+  }
>+
>+  template <typename U = T, typename = void>
>+  struct ToIntImpl {
>+    static constexpr int (*value)(Data) = nullptr;
>+  };
>+
>+  template <typename U>
>+  struct ToIntImpl<U,
>+                   typename std::enable_if<std::is_integral<U>::value>::type> {
>+    static int Invoke(Data arg) { return ToIntVal(Manager<T>::Value(arg)); }
>+    static constexpr int (*value)(Data) = &Invoke;
>+  };
>+
>+  template <typename U>
>+  struct ToIntImpl<U, typename std::enable_if<std::is_enum<U>::value>::type> {
>+    static int Invoke(Data arg) {
>+      return ToIntVal(static_cast<typename std::underlying_type<T>::type>(
>+          Manager<T>::Value(arg)));
>+    }
>+    static constexpr int (*value)(Data) = &Invoke;
>+  };
>+
>+ public:
>+  static constexpr VTable value{&ConvertImpl, ToIntImpl<>::value};
>+};
>+
>+template <typename T>
>+constexpr FormatArgImpl::VTable FormatArgImpl::TypedVTable<T>::value;
>+
>+template <typename T>
>+void FormatArgImpl::Init(const T& value) {
>+  data_ = Manager<T>::SetValue(value);
>+  vtbl_ = &TypedVTable<T>::value;
>+}
>+
>+extern template struct FormatArgImpl::TypedVTable<str_format_internal::VoidPtr>;
>+
>+extern template struct FormatArgImpl::TypedVTable<bool>;
>+extern template struct FormatArgImpl::TypedVTable<char>;
>+extern template struct FormatArgImpl::TypedVTable<signed char>;
>+extern template struct FormatArgImpl::TypedVTable<unsigned char>;
>+extern template struct FormatArgImpl::TypedVTable<short>;           // NOLINT
>+extern template struct FormatArgImpl::TypedVTable<unsigned short>;  // NOLINT
>+extern template struct FormatArgImpl::TypedVTable<int>;
>+extern template struct FormatArgImpl::TypedVTable<unsigned>;
>+extern template struct FormatArgImpl::TypedVTable<long>;           // NOLINT
>+extern template struct FormatArgImpl::TypedVTable<unsigned long>;  // NOLINT
>+extern template struct FormatArgImpl::TypedVTable<long long>;      // NOLINT
>+extern template struct FormatArgImpl::TypedVTable<
>+    unsigned long long>;  // NOLINT
>+extern template struct FormatArgImpl::TypedVTable<uint128>;
>+
>+extern template struct FormatArgImpl::TypedVTable<float>;
>+extern template struct FormatArgImpl::TypedVTable<double>;
>+extern template struct FormatArgImpl::TypedVTable<long double>;
>+
>+extern template struct FormatArgImpl::TypedVTable<const char*>;
>+extern template struct FormatArgImpl::TypedVTable<std::string>;
>+extern template struct FormatArgImpl::TypedVTable<string_view>;
>+}  // namespace str_format_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_STRINGS_INTERNAL_STR_FORMAT_ARG_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/arg_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/arg_test.cc
>new file mode 100644
>index 00000000000..83d59048ea2
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/arg_test.cc
>@@ -0,0 +1,111 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+#include "absl/strings/internal/str_format/arg.h"
>+
>+#include <ostream>
>+#include <string>
>+#include "gtest/gtest.h"
>+#include "absl/strings/str_format.h"
>+
>+namespace absl {
>+namespace str_format_internal {
>+namespace {
>+
>+class FormatArgImplTest : public ::testing::Test {
>+ public:
>+  enum Color { kRed, kGreen, kBlue };
>+
>+  static const char *hi() { return "hi"; }
>+};
>+
>+TEST_F(FormatArgImplTest, ToInt) {
>+  int out = 0;
>+  EXPECT_TRUE(FormatArgImplFriend::ToInt(FormatArgImpl(1), &out));
>+  EXPECT_EQ(1, out);
>+  EXPECT_TRUE(FormatArgImplFriend::ToInt(FormatArgImpl(-1), &out));
>+  EXPECT_EQ(-1, out);
>+  EXPECT_TRUE(
>+      FormatArgImplFriend::ToInt(FormatArgImpl(static_cast<char>(64)), &out));
>+  EXPECT_EQ(64, out);
>+  EXPECT_TRUE(FormatArgImplFriend::ToInt(
>+      FormatArgImpl(static_cast<unsigned long long>(123456)), &out));  // NOLINT
>+  EXPECT_EQ(123456, out);
>+  EXPECT_TRUE(FormatArgImplFriend::ToInt(
>+      FormatArgImpl(static_cast<unsigned long long>(  // NOLINT
>+                        std::numeric_limits<int>::max()) +
>+                    1),
>+      &out));
>+  EXPECT_EQ(std::numeric_limits<int>::max(), out);
>+  EXPECT_TRUE(FormatArgImplFriend::ToInt(
>+      FormatArgImpl(static_cast<long long>(  // NOLINT
>+                        std::numeric_limits<int>::min()) -
>+                    10),
>+      &out));
>+  EXPECT_EQ(std::numeric_limits<int>::min(), out);
>+  EXPECT_TRUE(FormatArgImplFriend::ToInt(FormatArgImpl(false), &out));
>+  EXPECT_EQ(0, out);
>+  EXPECT_TRUE(FormatArgImplFriend::ToInt(FormatArgImpl(true), &out));
>+  EXPECT_EQ(1, out);
>+  EXPECT_FALSE(FormatArgImplFriend::ToInt(FormatArgImpl(2.2), &out));
>+  EXPECT_FALSE(FormatArgImplFriend::ToInt(FormatArgImpl(3.2f), &out));
>+  EXPECT_FALSE(FormatArgImplFriend::ToInt(
>+      FormatArgImpl(static_cast<int *>(nullptr)), &out));
>+  EXPECT_FALSE(FormatArgImplFriend::ToInt(FormatArgImpl(hi()), &out));
>+  EXPECT_FALSE(FormatArgImplFriend::ToInt(FormatArgImpl("hi"), &out));
>+  EXPECT_TRUE(FormatArgImplFriend::ToInt(FormatArgImpl(kBlue), &out));
>+  EXPECT_EQ(2, out);
>+}
>+
>+extern const char kMyArray[];
>+
>+TEST_F(FormatArgImplTest, CharArraysDecayToCharPtr) {
>+  const char* a = "";
>+  EXPECT_EQ(FormatArgImplFriend::GetVTablePtrForTest(FormatArgImpl(a)),
>+            FormatArgImplFriend::GetVTablePtrForTest(FormatArgImpl("")));
>+  EXPECT_EQ(FormatArgImplFriend::GetVTablePtrForTest(FormatArgImpl(a)),
>+            FormatArgImplFriend::GetVTablePtrForTest(FormatArgImpl("A")));
>+  EXPECT_EQ(FormatArgImplFriend::GetVTablePtrForTest(FormatArgImpl(a)),
>+            FormatArgImplFriend::GetVTablePtrForTest(FormatArgImpl("ABC")));
>+  EXPECT_EQ(FormatArgImplFriend::GetVTablePtrForTest(FormatArgImpl(a)),
>+            FormatArgImplFriend::GetVTablePtrForTest(FormatArgImpl(kMyArray)));
>+}
>+
>+TEST_F(FormatArgImplTest, OtherPtrDecayToVoidPtr) {
>+  auto expected = FormatArgImplFriend::GetVTablePtrForTest(
>+      FormatArgImpl(static_cast<void *>(nullptr)));
>+  EXPECT_EQ(FormatArgImplFriend::GetVTablePtrForTest(
>+                FormatArgImpl(static_cast<int *>(nullptr))),
>+            expected);
>+  EXPECT_EQ(FormatArgImplFriend::GetVTablePtrForTest(
>+                FormatArgImpl(static_cast<volatile int *>(nullptr))),
>+            expected);
>+
>+  auto p = static_cast<void (*)()>([] {});
>+  EXPECT_EQ(FormatArgImplFriend::GetVTablePtrForTest(FormatArgImpl(p)),
>+            expected);
>+}
>+
>+TEST_F(FormatArgImplTest, WorksWithCharArraysOfUnknownSize) {
>+  std::string s;
>+  FormatSinkImpl sink(&s);
>+  ConversionSpec conv;
>+  conv.set_conv(ConversionChar::FromChar('s'));
>+  conv.set_flags(Flags());
>+  conv.set_width(-1);
>+  conv.set_precision(-1);
>+  EXPECT_TRUE(
>+      FormatArgImplFriend::Convert(FormatArgImpl(kMyArray), conv, &sink));
>+  sink.Flush();
>+  EXPECT_EQ("ABCDE", s);
>+}
>+const char kMyArray[] = "ABCDE";
>+
>+}  // namespace
>+}  // namespace str_format_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/bind.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/bind.cc
>new file mode 100644
>index 00000000000..33e8641558a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/bind.cc
>@@ -0,0 +1,232 @@
>+#include "absl/strings/internal/str_format/bind.h"
>+
>+#include <cerrno>
>+#include <limits>
>+#include <sstream>
>+#include <string>
>+
>+namespace absl {
>+namespace str_format_internal {
>+
>+namespace {
>+
>+inline bool BindFromPosition(int position, int* value,
>+                             absl::Span<const FormatArgImpl> pack) {
>+  assert(position > 0);
>+  if (static_cast<size_t>(position) > pack.size()) {
>+    return false;
>+  }
>+  // -1 because positions are 1-based
>+  return FormatArgImplFriend::ToInt(pack[position - 1], value);
>+}
>+
>+class ArgContext {
>+ public:
>+  explicit ArgContext(absl::Span<const FormatArgImpl> pack) : pack_(pack) {}
>+
>+  // Fill 'bound' with the results of applying the context's argument pack
>+  // to the specified 'props'. We synthesize a BoundConversion by
>+  // lining up a UnboundConversion with a user argument. We also
>+  // resolve any '*' specifiers for width and precision, so after
>+  // this call, 'bound' has all the information it needs to be formatted.
>+  // Returns false on failure.
>+  bool Bind(const UnboundConversion *props, BoundConversion *bound);
>+
>+ private:
>+  absl::Span<const FormatArgImpl> pack_;
>+};
>+
>+inline bool ArgContext::Bind(const UnboundConversion* unbound,
>+                             BoundConversion* bound) {
>+  const FormatArgImpl* arg = nullptr;
>+  int arg_position = unbound->arg_position;
>+  if (static_cast<size_t>(arg_position - 1) >= pack_.size()) return false;
>+  arg = &pack_[arg_position - 1];  // 1-based
>+
>+  if (!unbound->flags.basic) {
>+    int width = unbound->width.value();
>+    bool force_left = false;
>+    if (unbound->width.is_from_arg()) {
>+      if (!BindFromPosition(unbound->width.get_from_arg(), &width, pack_))
>+        return false;
>+      if (width < 0) {
>+        // "A negative field width is taken as a '-' flag followed by a
>+        // positive field width."
>+        force_left = true;
>+        width = -width;
>+      }
>+    }
>+
>+    int precision = unbound->precision.value();
>+    if (unbound->precision.is_from_arg()) {
>+      if (!BindFromPosition(unbound->precision.get_from_arg(), &precision,
>+                            pack_))
>+        return false;
>+    }
>+
>+    bound->set_width(width);
>+    bound->set_precision(precision);
>+    bound->set_flags(unbound->flags);
>+    if (force_left)
>+      bound->set_left(true);
>+  } else {
>+    bound->set_flags(unbound->flags);
>+    bound->set_width(-1);
>+    bound->set_precision(-1);
>+  }
>+
>+  bound->set_length_mod(unbound->length_mod);
>+  bound->set_conv(unbound->conv);
>+  bound->set_arg(arg);
>+  return true;
>+}
>+
>+template <typename Converter>
>+class ConverterConsumer {
>+ public:
>+  ConverterConsumer(Converter converter, absl::Span<const FormatArgImpl> pack)
>+      : converter_(converter), arg_context_(pack) {}
>+
>+  bool Append(string_view s) {
>+    converter_.Append(s);
>+    return true;
>+  }
>+  bool ConvertOne(const UnboundConversion& conv, string_view conv_string) {
>+    BoundConversion bound;
>+    if (!arg_context_.Bind(&conv, &bound)) return false;
>+    return converter_.ConvertOne(bound, conv_string);
>+  }
>+
>+ private:
>+  Converter converter_;
>+  ArgContext arg_context_;
>+};
>+
>+template <typename Converter>
>+bool ConvertAll(const UntypedFormatSpecImpl& format,
>+                absl::Span<const FormatArgImpl> args,
>+                const Converter& converter) {
>+  const ParsedFormatBase* pc = format.parsed_conversion();
>+  if (pc)
>+    return pc->ProcessFormat(ConverterConsumer<Converter>(converter, args));
>+
>+  return ParseFormatString(format.str(),
>+                           ConverterConsumer<Converter>(converter, args));
>+}
>+
>+class DefaultConverter {
>+ public:
>+  explicit DefaultConverter(FormatSinkImpl* sink) : sink_(sink) {}
>+
>+  void Append(string_view s) const { sink_->Append(s); }
>+
>+  bool ConvertOne(const BoundConversion& bound, string_view /*conv*/) const {
>+    return FormatArgImplFriend::Convert(*bound.arg(), bound, sink_);
>+  }
>+
>+ private:
>+  FormatSinkImpl* sink_;
>+};
>+
>+class SummarizingConverter {
>+ public:
>+  explicit SummarizingConverter(FormatSinkImpl* sink) : sink_(sink) {}
>+
>+  void Append(string_view s) const { sink_->Append(s); }
>+
>+  bool ConvertOne(const BoundConversion& bound, string_view /*conv*/) const {
>+    UntypedFormatSpecImpl spec("%d");
>+
>+    std::ostringstream ss;
>+    ss << "{" << Streamable(spec, {*bound.arg()}) << ":" << bound.flags();
>+    if (bound.width() >= 0) ss << bound.width();
>+    if (bound.precision() >= 0) ss << "." << bound.precision();
>+    ss << bound.length_mod() << bound.conv() << "}";
>+    Append(ss.str());
>+    return true;
>+  }
>+
>+ private:
>+  FormatSinkImpl* sink_;
>+};
>+
>+}  // namespace
>+
>+bool BindWithPack(const UnboundConversion* props,
>+                  absl::Span<const FormatArgImpl> pack,
>+                  BoundConversion* bound) {
>+  return ArgContext(pack).Bind(props, bound);
>+}
>+
>+std::string Summarize(const UntypedFormatSpecImpl& format,
>+                 absl::Span<const FormatArgImpl> args) {
>+  typedef SummarizingConverter Converter;
>+  std::string out;
>+  {
>+    // inner block to destroy sink before returning out. It ensures a last
>+    // flush.
>+    FormatSinkImpl sink(&out);
>+    if (!ConvertAll(format, args, Converter(&sink))) {
>+      sink.Flush();
>+      out.clear();
>+    }
>+  }
>+  return out;
>+}
>+
>+bool FormatUntyped(FormatRawSinkImpl raw_sink,
>+                   const UntypedFormatSpecImpl& format,
>+                   absl::Span<const FormatArgImpl> args) {
>+  FormatSinkImpl sink(raw_sink);
>+  using Converter = DefaultConverter;
>+  if (!ConvertAll(format, args, Converter(&sink))) {
>+    sink.Flush();
>+    return false;
>+  }
>+  return true;
>+}
>+
>+std::ostream& Streamable::Print(std::ostream& os) const {
>+  if (!FormatUntyped(&os, format_, args_)) os.setstate(std::ios::failbit);
>+  return os;
>+}
>+
>+std::string& AppendPack(std::string* out, const UntypedFormatSpecImpl& format,
>+                   absl::Span<const FormatArgImpl> args) {
>+  size_t orig = out->size();
>+  if (!FormatUntyped(out, format, args)) out->resize(orig);
>+  return *out;
>+}
>+
>+int FprintF(std::FILE* output, const UntypedFormatSpecImpl& format,
>+            absl::Span<const FormatArgImpl> args) {
>+  FILERawSink sink(output);
>+  if (!FormatUntyped(&sink, format, args)) {
>+    errno = EINVAL;
>+    return -1;
>+  }
>+  if (sink.error()) {
>+    errno = sink.error();
>+    return -1;
>+  }
>+  if (sink.count() > std::numeric_limits<int>::max()) {
>+    errno = EFBIG;
>+    return -1;
>+  }
>+  return static_cast<int>(sink.count());
>+}
>+
>+int SnprintF(char* output, size_t size, const UntypedFormatSpecImpl& format,
>+             absl::Span<const FormatArgImpl> args) {
>+  BufferRawSink sink(output, size ? size - 1 : 0);
>+  if (!FormatUntyped(&sink, format, args)) {
>+    errno = EINVAL;
>+    return -1;
>+  }
>+  size_t total = sink.total_written();
>+  if (size) output[std::min(total, size - 1)] = 0;
>+  return static_cast<int>(total);
>+}
>+
>+}  // namespace str_format_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/bind.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/bind.h
>new file mode 100644
>index 00000000000..4008611211c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/bind.h
>@@ -0,0 +1,189 @@
>+#ifndef ABSL_STRINGS_INTERNAL_STR_FORMAT_BIND_H_
>+#define ABSL_STRINGS_INTERNAL_STR_FORMAT_BIND_H_
>+
>+#include <array>
>+#include <cstdio>
>+#include <sstream>
>+#include <string>
>+
>+#include "absl/base/port.h"
>+#include "absl/container/inlined_vector.h"
>+#include "absl/strings/internal/str_format/arg.h"
>+#include "absl/strings/internal/str_format/checker.h"
>+#include "absl/strings/internal/str_format/parser.h"
>+#include "absl/types/span.h"
>+
>+namespace absl {
>+
>+class UntypedFormatSpec;
>+
>+namespace str_format_internal {
>+
>+class BoundConversion : public ConversionSpec {
>+ public:
>+  const FormatArgImpl* arg() const { return arg_; }
>+  void set_arg(const FormatArgImpl* a) { arg_ = a; }
>+
>+ private:
>+  const FormatArgImpl* arg_;
>+};
>+
>+// This is the type-erased class that the implementation uses.
>+class UntypedFormatSpecImpl {
>+ public:
>+  UntypedFormatSpecImpl() = delete;
>+
>+  explicit UntypedFormatSpecImpl(string_view s) : str_(s), pc_() {}
>+  explicit UntypedFormatSpecImpl(
>+      const str_format_internal::ParsedFormatBase* pc)
>+      : pc_(pc) {}
>+  string_view str() const { return str_; }
>+  const str_format_internal::ParsedFormatBase* parsed_conversion() const {
>+    return pc_;
>+  }
>+
>+  template <typename T>
>+  static const UntypedFormatSpecImpl& Extract(const T& s) {
>+    return s.spec_;
>+  }
>+
>+ private:
>+  string_view str_;
>+  const str_format_internal::ParsedFormatBase* pc_;
>+};
>+
>+template <typename T, typename...>
>+struct MakeDependent {
>+  using type = T;
>+};
>+
>+// Implicitly convertible from `const char*`, `string_view`, and the
>+// `ExtendedParsedFormat` type. This abstraction allows all format functions to
>+// operate on any without providing too many overloads.
>+template <typename... Args>
>+class FormatSpecTemplate
>+    : public MakeDependent<UntypedFormatSpec, Args...>::type {
>+  using Base = typename MakeDependent<UntypedFormatSpec, Args...>::type;
>+
>+ public:
>+#if ABSL_INTERNAL_ENABLE_FORMAT_CHECKER
>+
>+  // Honeypot overload for when the std::string is not constexpr.
>+  // We use the 'unavailable' attribute to give a better compiler error than
>+  // just 'method is deleted'.
>+  FormatSpecTemplate(...)  // NOLINT
>+      __attribute__((unavailable("Format std::string is not constexpr.")));
>+
>+  // Honeypot overload for when the format is constexpr and invalid.
>+  // We use the 'unavailable' attribute to give a better compiler error than
>+  // just 'method is deleted'.
>+  // To avoid checking the format twice, we just check that the format is
>+  // constexpr. If is it valid, then the overload below will kick in.
>+  // We add the template here to make this overload have lower priority.
>+  template <typename = void>
>+  FormatSpecTemplate(const char* s)  // NOLINT
>+      __attribute__((
>+          enable_if(str_format_internal::EnsureConstexpr(s), "constexpr trap"),
>+          unavailable(
>+              "Format specified does not match the arguments passed.")));
>+
>+  template <typename T = void>
>+  FormatSpecTemplate(string_view s)  // NOLINT
>+      __attribute__((enable_if(str_format_internal::EnsureConstexpr(s),
>+                               "constexpr trap"))) {
>+    static_assert(sizeof(T*) == 0,
>+                  "Format specified does not match the arguments passed.");
>+  }
>+
>+  // Good format overload.
>+  FormatSpecTemplate(const char* s)  // NOLINT
>+      __attribute__((enable_if(ValidFormatImpl<ArgumentToConv<Args>()...>(s),
>+                               "bad format trap")))
>+      : Base(s) {}
>+
>+  FormatSpecTemplate(string_view s)  // NOLINT
>+      __attribute__((enable_if(ValidFormatImpl<ArgumentToConv<Args>()...>(s),
>+                               "bad format trap")))
>+      : Base(s) {}
>+
>+#else  // ABSL_INTERNAL_ENABLE_FORMAT_CHECKER
>+
>+  FormatSpecTemplate(const char* s) : Base(s) {}  // NOLINT
>+  FormatSpecTemplate(string_view s) : Base(s) {}  // NOLINT
>+
>+#endif  // ABSL_INTERNAL_ENABLE_FORMAT_CHECKER
>+
>+  template <Conv... C, typename = typename std::enable_if<
>+                           sizeof...(C) == sizeof...(Args) &&
>+                           AllOf(Contains(ArgumentToConv<Args>(),
>+                                          C)...)>::type>
>+  FormatSpecTemplate(const ExtendedParsedFormat<C...>& pc)  // NOLINT
>+      : Base(&pc) {}
>+};
>+
>+template <typename... Args>
>+struct FormatSpecDeductionBarrier {
>+  using type = FormatSpecTemplate<Args...>;
>+};
>+
>+class Streamable {
>+ public:
>+  Streamable(const UntypedFormatSpecImpl& format,
>+             absl::Span<const FormatArgImpl> args)
>+      : format_(format), args_(args.begin(), args.end()) {}
>+
>+  std::ostream& Print(std::ostream& os) const;
>+
>+  friend std::ostream& operator<<(std::ostream& os, const Streamable& l) {
>+    return l.Print(os);
>+  }
>+
>+ private:
>+  const UntypedFormatSpecImpl& format_;
>+  absl::InlinedVector<FormatArgImpl, 4> args_;
>+};
>+
>+// for testing
>+std::string Summarize(const UntypedFormatSpecImpl& format,
>+                 absl::Span<const FormatArgImpl> args);
>+bool BindWithPack(const UnboundConversion* props,
>+                  absl::Span<const FormatArgImpl> pack, BoundConversion* bound);
>+
>+bool FormatUntyped(FormatRawSinkImpl raw_sink,
>+                   const UntypedFormatSpecImpl& format,
>+                   absl::Span<const FormatArgImpl> args);
>+
>+std::string& AppendPack(std::string* out, const UntypedFormatSpecImpl& format,
>+                   absl::Span<const FormatArgImpl> args);
>+
>+inline std::string FormatPack(const UntypedFormatSpecImpl& format,
>+                         absl::Span<const FormatArgImpl> args) {
>+  std::string out;
>+  AppendPack(&out, format, args);
>+  return out;
>+}
>+
>+int FprintF(std::FILE* output, const UntypedFormatSpecImpl& format,
>+            absl::Span<const FormatArgImpl> args);
>+int SnprintF(char* output, size_t size, const UntypedFormatSpecImpl& format,
>+             absl::Span<const FormatArgImpl> args);
>+
>+// Returned by Streamed(v). Converts via '%s' to the std::string created
>+// by std::ostream << v.
>+template <typename T>
>+class StreamedWrapper {
>+ public:
>+  explicit StreamedWrapper(const T& v) : v_(v) { }
>+
>+ private:
>+  template <typename S>
>+  friend ConvertResult<Conv::s> FormatConvertImpl(const StreamedWrapper<S>& v,
>+                                                  const ConversionSpec& conv,
>+                                                  FormatSinkImpl* out);
>+  const T& v_;
>+};
>+
>+}  // namespace str_format_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_STRINGS_INTERNAL_STR_FORMAT_BIND_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/bind_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/bind_test.cc
>new file mode 100644
>index 00000000000..47575739ba1
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/bind_test.cc
>@@ -0,0 +1,131 @@
>+#include "absl/strings/internal/str_format/bind.h"
>+
>+#include <string.h>
>+
>+#include "gtest/gtest.h"
>+
>+namespace absl {
>+namespace str_format_internal {
>+namespace {
>+
>+template <typename T, size_t N>
>+size_t ArraySize(T (&)[N]) {
>+  return N;
>+}
>+
>+class FormatBindTest : public ::testing::Test {
>+ public:
>+  bool Extract(const char *s, UnboundConversion *props, int *next) const {
>+    absl::string_view src = s;
>+    return ConsumeUnboundConversion(&src, props, next) && src.empty();
>+  }
>+};
>+
>+TEST_F(FormatBindTest, BindSingle) {
>+  struct Expectation {
>+    int line;
>+    const char *fmt;
>+    int ok_phases;
>+    const FormatArgImpl *arg;
>+    int width;
>+    int precision;
>+    int next_arg;
>+  };
>+  const int no = -1;
>+  const int ia[] = { 10, 20, 30, 40};
>+  const FormatArgImpl args[] = {FormatArgImpl(ia[0]), FormatArgImpl(ia[1]),
>+                                FormatArgImpl(ia[2]), FormatArgImpl(ia[3])};
>+#pragma GCC diagnostic push
>+#pragma GCC diagnostic ignored "-Wmissing-field-initializers"
>+  const Expectation kExpect[] = {
>+    {__LINE__, "d",          2, &args[0], no, no, 2},
>+    {__LINE__, "4d",         2, &args[0],  4, no, 2},
>+    {__LINE__, ".5d",        2, &args[0], no,  5, 2},
>+    {__LINE__, "4.5d",       2, &args[0],  4,  5, 2},
>+    {__LINE__, "*d",         2, &args[1], 10, no, 3},
>+    {__LINE__, ".*d",        2, &args[1], no, 10, 3},
>+    {__LINE__, "*.*d",       2, &args[2], 10, 20, 4},
>+    {__LINE__, "1$d",        2, &args[0], no, no, 0},
>+    {__LINE__, "2$d",        2, &args[1], no, no, 0},
>+    {__LINE__, "3$d",        2, &args[2], no, no, 0},
>+    {__LINE__, "4$d",        2, &args[3], no, no, 0},
>+    {__LINE__, "2$*1$d",     2, &args[1], 10, no, 0},
>+    {__LINE__, "2$*2$d",     2, &args[1], 20, no, 0},
>+    {__LINE__, "2$*3$d",     2, &args[1], 30, no, 0},
>+    {__LINE__, "2$.*1$d",    2, &args[1], no, 10, 0},
>+    {__LINE__, "2$.*2$d",    2, &args[1], no, 20, 0},
>+    {__LINE__, "2$.*3$d",    2, &args[1], no, 30, 0},
>+    {__LINE__, "2$*3$.*1$d", 2, &args[1], 30, 10, 0},
>+    {__LINE__, "2$*2$.*2$d", 2, &args[1], 20, 20, 0},
>+    {__LINE__, "2$*1$.*3$d", 2, &args[1], 10, 30, 0},
>+    {__LINE__, "2$*3$.*1$d", 2, &args[1], 30, 10, 0},
>+    {__LINE__, "1$*d",       0},  // indexed, then positional
>+    {__LINE__, "*2$d",       0},  // positional, then indexed
>+    {__LINE__, "6$d",        1},  // arg position out of bounds
>+    {__LINE__, "1$6$d",      0},  // width position incorrectly specified
>+    {__LINE__, "1$.6$d",     0},  // precision position incorrectly specified
>+    {__LINE__, "1$*6$d",     1},  // width position out of bounds
>+    {__LINE__, "1$.*6$d",    1},  // precision position out of bounds
>+  };
>+#pragma GCC diagnostic pop
>+  for (const Expectation &e : kExpect) {
>+    SCOPED_TRACE(e.line);
>+    SCOPED_TRACE(e.fmt);
>+    UnboundConversion props;
>+    BoundConversion bound;
>+    int ok_phases = 0;
>+    int next = 0;
>+    if (Extract(e.fmt, &props, &next)) {
>+      ++ok_phases;
>+      if (BindWithPack(&props, args, &bound)) {
>+        ++ok_phases;
>+      }
>+    }
>+    EXPECT_EQ(e.ok_phases, ok_phases);
>+    if (e.ok_phases < 2) continue;
>+    if (e.arg != nullptr) {
>+      EXPECT_EQ(e.arg, bound.arg());
>+    }
>+    EXPECT_EQ(e.width, bound.width());
>+    EXPECT_EQ(e.precision, bound.precision());
>+  }
>+}
>+
>+TEST_F(FormatBindTest, FormatPack) {
>+  struct Expectation {
>+    int line;
>+    const char *fmt;
>+    const char *summary;
>+  };
>+  const int ia[] = { 10, 20, 30, 40, -10 };
>+  const FormatArgImpl args[] = {FormatArgImpl(ia[0]), FormatArgImpl(ia[1]),
>+                                FormatArgImpl(ia[2]), FormatArgImpl(ia[3]),
>+                                FormatArgImpl(ia[4])};
>+  const Expectation kExpect[] = {
>+    {__LINE__, "a%4db%dc", "a{10:4d}b{20:d}c"},
>+    {__LINE__, "a%.4db%dc", "a{10:.4d}b{20:d}c"},
>+    {__LINE__, "a%4.5db%dc", "a{10:4.5d}b{20:d}c"},
>+    {__LINE__, "a%db%4.5dc", "a{10:d}b{20:4.5d}c"},
>+    {__LINE__, "a%db%*.*dc", "a{10:d}b{40:20.30d}c"},
>+    {__LINE__, "a%.*fb", "a{20:.10f}b"},
>+    {__LINE__, "a%1$db%2$*3$.*4$dc", "a{10:d}b{20:30.40d}c"},
>+    {__LINE__, "a%4$db%3$*2$.*1$dc", "a{40:d}b{30:20.10d}c"},
>+    {__LINE__, "a%04ldb", "a{10:04ld}b"},
>+    {__LINE__, "a%-#04lldb", "a{10:-#04lld}b"},
>+    {__LINE__, "a%1$*5$db", "a{10:-10d}b"},
>+    {__LINE__, "a%1$.*5$db", "a{10:d}b"},
>+  };
>+  for (const Expectation &e : kExpect) {
>+    absl::string_view fmt = e.fmt;
>+    SCOPED_TRACE(e.line);
>+    SCOPED_TRACE(e.fmt);
>+    UntypedFormatSpecImpl format(fmt);
>+    EXPECT_EQ(e.summary,
>+              str_format_internal::Summarize(format, absl::MakeSpan(args)))
>+        << "line:" << e.line;
>+  }
>+}
>+
>+}  // namespace
>+}  // namespace str_format_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/checker.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/checker.h
>new file mode 100644
>index 00000000000..8b594f2d5cc
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/checker.h
>@@ -0,0 +1,325 @@
>+#ifndef ABSL_STRINGS_INTERNAL_STR_FORMAT_CHECKER_H_
>+#define ABSL_STRINGS_INTERNAL_STR_FORMAT_CHECKER_H_
>+
>+#include "absl/strings/internal/str_format/arg.h"
>+#include "absl/strings/internal/str_format/extension.h"
>+
>+// Compile time check support for entry points.
>+
>+#ifndef ABSL_INTERNAL_ENABLE_FORMAT_CHECKER
>+#if defined(__clang__) && !defined(__native_client__)
>+#if __has_attribute(enable_if)
>+#define ABSL_INTERNAL_ENABLE_FORMAT_CHECKER 1
>+#endif  // __has_attribute(enable_if)
>+#endif  // defined(__clang__) && !defined(__native_client__)
>+#endif  // ABSL_INTERNAL_ENABLE_FORMAT_CHECKER
>+
>+namespace absl {
>+namespace str_format_internal {
>+
>+constexpr bool AllOf() { return true; }
>+
>+template <typename... T>
>+constexpr bool AllOf(bool b, T... t) {
>+  return b && AllOf(t...);
>+}
>+
>+template <typename Arg>
>+constexpr Conv ArgumentToConv() {
>+  return decltype(str_format_internal::FormatConvertImpl(
>+      std::declval<const Arg&>(), std::declval<const ConversionSpec&>(),
>+      std::declval<FormatSinkImpl*>()))::kConv;
>+}
>+
>+#if ABSL_INTERNAL_ENABLE_FORMAT_CHECKER
>+
>+constexpr bool ContainsChar(const char* chars, char c) {
>+  return *chars == c || (*chars && ContainsChar(chars + 1, c));
>+}
>+
>+// A constexpr compatible list of Convs.
>+struct ConvList {
>+  const Conv* array;
>+  int count;
>+
>+  // We do the bound check here to avoid having to do it on the callers.
>+  // Returning an empty Conv has the same effect as short circuiting because it
>+  // will never match any conversion.
>+  constexpr Conv operator[](int i) const {
>+    return i < count ? array[i] : Conv{};
>+  }
>+
>+  constexpr ConvList without_front() const {
>+    return count != 0 ? ConvList{array + 1, count - 1} : *this;
>+  }
>+};
>+
>+template <size_t count>
>+struct ConvListT {
>+  // Make sure the array has size > 0.
>+  Conv list[count ? count : 1];
>+};
>+
>+constexpr char GetChar(string_view str, size_t index) {
>+  return index < str.size() ? str[index] : char{};
>+}
>+
>+constexpr string_view ConsumeFront(string_view str, size_t len = 1) {
>+  return len <= str.size() ? string_view(str.data() + len, str.size() - len)
>+                           : string_view();
>+}
>+
>+constexpr string_view ConsumeAnyOf(string_view format, const char* chars) {
>+  return ContainsChar(chars, GetChar(format, 0))
>+             ? ConsumeAnyOf(ConsumeFront(format), chars)
>+             : format;
>+}
>+
>+constexpr bool IsDigit(char c) { return c >= '0' && c <= '9'; }
>+
>+// Helper class for the ParseDigits function.
>+// It encapsulates the two return values we need there.
>+struct Integer {
>+  string_view format;
>+  int value;
>+
>+  // If the next character is a '$', consume it.
>+  // Otherwise, make `this` an invalid positional argument.
>+  constexpr Integer ConsumePositionalDollar() const {
>+    return GetChar(format, 0) == '$' ? Integer{ConsumeFront(format), value}
>+                                     : Integer{format, 0};
>+  }
>+};
>+
>+constexpr Integer ParseDigits(string_view format, int value = 0) {
>+  return IsDigit(GetChar(format, 0))
>+             ? ParseDigits(ConsumeFront(format),
>+                           10 * value + GetChar(format, 0) - '0')
>+             : Integer{format, value};
>+}
>+
>+// Parse digits for a positional argument.
>+// The parsing also consumes the '$'.
>+constexpr Integer ParsePositional(string_view format) {
>+  return ParseDigits(format).ConsumePositionalDollar();
>+}
>+
>+// Parses a single conversion specifier.
>+// See ConvParser::Run() for post conditions.
>+class ConvParser {
>+  constexpr ConvParser SetFormat(string_view format) const {
>+    return ConvParser(format, args_, error_, arg_position_, is_positional_);
>+  }
>+
>+  constexpr ConvParser SetArgs(ConvList args) const {
>+    return ConvParser(format_, args, error_, arg_position_, is_positional_);
>+  }
>+
>+  constexpr ConvParser SetError(bool error) const {
>+    return ConvParser(format_, args_, error_ || error, arg_position_,
>+                      is_positional_);
>+  }
>+
>+  constexpr ConvParser SetArgPosition(int arg_position) const {
>+    return ConvParser(format_, args_, error_, arg_position, is_positional_);
>+  }
>+
>+  // Consumes the next arg and verifies that it matches `conv`.
>+  // `error_` is set if there is no next arg or if it doesn't match `conv`.
>+  constexpr ConvParser ConsumeNextArg(char conv) const {
>+    return SetArgs(args_.without_front()).SetError(!Contains(args_[0], conv));
>+  }
>+
>+  // Verify that positional argument `i.value` matches `conv`.
>+  // `error_` is set if `i.value` is not a valid argument or if it doesn't
>+  // match.
>+  constexpr ConvParser VerifyPositional(Integer i, char conv) const {
>+    return SetFormat(i.format).SetError(!Contains(args_[i.value - 1], conv));
>+  }
>+
>+  // Parse the position of the arg and store it in `arg_position_`.
>+  constexpr ConvParser ParseArgPosition(Integer arg) const {
>+    return SetFormat(arg.format).SetArgPosition(arg.value);
>+  }
>+
>+  // Consume the flags.
>+  constexpr ConvParser ParseFlags() const {
>+    return SetFormat(ConsumeAnyOf(format_, "-+ #0"));
>+  }
>+
>+  // Consume the width.
>+  // If it is '*', we verify that it matches `args_`. `error_` is set if it
>+  // doesn't match.
>+  constexpr ConvParser ParseWidth() const {
>+    return IsDigit(GetChar(format_, 0))
>+               ? SetFormat(ParseDigits(format_).format)
>+               : GetChar(format_, 0) == '*'
>+                     ? is_positional_
>+                           ? VerifyPositional(
>+                                 ParsePositional(ConsumeFront(format_)), '*')
>+                           : SetFormat(ConsumeFront(format_))
>+                                 .ConsumeNextArg('*')
>+                     : *this;
>+  }
>+
>+  // Consume the precision.
>+  // If it is '*', we verify that it matches `args_`. `error_` is set if it
>+  // doesn't match.
>+  constexpr ConvParser ParsePrecision() const {
>+    return GetChar(format_, 0) != '.'
>+               ? *this
>+               : GetChar(format_, 1) == '*'
>+                     ? is_positional_
>+                           ? VerifyPositional(
>+                                 ParsePositional(ConsumeFront(format_, 2)), '*')
>+                           : SetFormat(ConsumeFront(format_, 2))
>+                                 .ConsumeNextArg('*')
>+                     : SetFormat(ParseDigits(ConsumeFront(format_)).format);
>+  }
>+
>+  // Consume the length characters.
>+  constexpr ConvParser ParseLength() const {
>+    return SetFormat(ConsumeAnyOf(format_, "lLhjztq"));
>+  }
>+
>+  // Consume the conversion character and verify that it matches `args_`.
>+  // `error_` is set if it doesn't match.
>+  constexpr ConvParser ParseConversion() const {
>+    return is_positional_
>+               ? VerifyPositional({ConsumeFront(format_), arg_position_},
>+                                  GetChar(format_, 0))
>+               : ConsumeNextArg(GetChar(format_, 0))
>+                     .SetFormat(ConsumeFront(format_));
>+  }
>+
>+  constexpr ConvParser(string_view format, ConvList args, bool error,
>+                       int arg_position, bool is_positional)
>+      : format_(format),
>+        args_(args),
>+        error_(error),
>+        arg_position_(arg_position),
>+        is_positional_(is_positional) {}
>+
>+ public:
>+  constexpr ConvParser(string_view format, ConvList args, bool is_positional)
>+      : format_(format),
>+        args_(args),
>+        error_(false),
>+        arg_position_(0),
>+        is_positional_(is_positional) {}
>+
>+  // Consume the whole conversion specifier.
>+  // `format()` will be set to the character after the conversion character.
>+  // `error()` will be set if any of the arguments do not match.
>+  constexpr ConvParser Run() const {
>+    return (is_positional_ ? ParseArgPosition(ParsePositional(format_)) : *this)
>+        .ParseFlags()
>+        .ParseWidth()
>+        .ParsePrecision()
>+        .ParseLength()
>+        .ParseConversion();
>+  }
>+
>+  constexpr string_view format() const { return format_; }
>+  constexpr ConvList args() const { return args_; }
>+  constexpr bool error() const { return error_; }
>+  constexpr bool is_positional() const { return is_positional_; }
>+
>+ private:
>+  string_view format_;
>+  // Current list of arguments. If we are not in positional mode we will consume
>+  // from the front.
>+  ConvList args_;
>+  bool error_;
>+  // Holds the argument position of the conversion character, if we are in
>+  // positional mode. Otherwise, it is unspecified.
>+  int arg_position_;
>+  // Whether we are in positional mode.
>+  // It changes the behavior of '*' and where to find the converted argument.
>+  bool is_positional_;
>+};
>+
>+// Parses a whole format expression.
>+// See FormatParser::Run().
>+class FormatParser {
>+  static constexpr bool FoundPercent(string_view format) {
>+    return format.empty() ||
>+           (GetChar(format, 0) == '%' && GetChar(format, 1) != '%');
>+  }
>+
>+  // We use an inner function to increase the recursion limit.
>+  // The inner function consumes up to `limit` characters on every run.
>+  // This increases the limit from 512 to ~512*limit.
>+  static constexpr string_view ConsumeNonPercentInner(string_view format,
>+                                                      int limit = 20) {
>+    return FoundPercent(format) || !limit
>+               ? format
>+               : ConsumeNonPercentInner(
>+                     ConsumeFront(format, GetChar(format, 0) == '%' &&
>+                                                  GetChar(format, 1) == '%'
>+                                              ? 2
>+                                              : 1),
>+                     limit - 1);
>+  }
>+
>+  // Consume characters until the next conversion spec %.
>+  // It skips %%.
>+  static constexpr string_view ConsumeNonPercent(string_view format) {
>+    return FoundPercent(format)
>+               ? format
>+               : ConsumeNonPercent(ConsumeNonPercentInner(format));
>+  }
>+
>+  static constexpr bool IsPositional(string_view format) {
>+    return IsDigit(GetChar(format, 0)) ? IsPositional(ConsumeFront(format))
>+                                       : GetChar(format, 0) == '$';
>+  }
>+
>+  constexpr bool RunImpl(bool is_positional) const {
>+    // In non-positional mode we require all arguments to be consumed.
>+    // In positional mode just reaching the end of the format without errors is
>+    // enough.
>+    return (format_.empty() && (is_positional || args_.count == 0)) ||
>+           (!format_.empty() &&
>+            ValidateArg(
>+                ConvParser(ConsumeFront(format_), args_, is_positional).Run()));
>+  }
>+
>+  constexpr bool ValidateArg(ConvParser conv) const {
>+    return !conv.error() && FormatParser(conv.format(), conv.args())
>+                                .RunImpl(conv.is_positional());
>+  }
>+
>+ public:
>+  constexpr FormatParser(string_view format, ConvList args)
>+      : format_(ConsumeNonPercent(format)), args_(args) {}
>+
>+  // Runs the parser for `format` and `args`.
>+  // It verifies that the format is valid and that all conversion specifiers
>+  // match the arguments passed.
>+  // In non-positional mode it also verfies that all arguments are consumed.
>+  constexpr bool Run() const {
>+    return RunImpl(!format_.empty() && IsPositional(ConsumeFront(format_)));
>+  }
>+
>+ private:
>+  string_view format_;
>+  // Current list of arguments.
>+  // If we are not in positional mode we will consume from the front and will
>+  // have to be empty in the end.
>+  ConvList args_;
>+};
>+
>+template <Conv... C>
>+constexpr bool ValidFormatImpl(string_view format) {
>+  return FormatParser(format,
>+                      {ConvListT<sizeof...(C)>{{C...}}.list, sizeof...(C)})
>+      .Run();
>+}
>+
>+#endif  // ABSL_INTERNAL_ENABLE_FORMAT_CHECKER
>+
>+}  // namespace str_format_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_STRINGS_INTERNAL_STR_FORMAT_CHECKER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/checker_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/checker_test.cc
>new file mode 100644
>index 00000000000..14d11ea8bd3
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/checker_test.cc
>@@ -0,0 +1,150 @@
>+#include <string>
>+
>+#include "gmock/gmock.h"
>+#include "gtest/gtest.h"
>+#include "absl/strings/str_format.h"
>+
>+namespace absl {
>+namespace str_format_internal {
>+namespace {
>+
>+std::string ConvToString(Conv conv) {
>+  std::string out;
>+#define CONV_SET_CASE(c) \
>+  if (Contains(conv, Conv::c)) out += #c;
>+  ABSL_CONVERSION_CHARS_EXPAND_(CONV_SET_CASE, )
>+#undef CONV_SET_CASE
>+  if (Contains(conv, Conv::star)) out += "*";
>+  return out;
>+}
>+
>+TEST(StrFormatChecker, ArgumentToConv) {
>+  Conv conv = ArgumentToConv<std::string>();
>+  EXPECT_EQ(ConvToString(conv), "s");
>+
>+  conv = ArgumentToConv<const char*>();
>+  EXPECT_EQ(ConvToString(conv), "sp");
>+
>+  conv = ArgumentToConv<double>();
>+  EXPECT_EQ(ConvToString(conv), "fFeEgGaA");
>+
>+  conv = ArgumentToConv<int>();
>+  EXPECT_EQ(ConvToString(conv), "cdiouxXfFeEgGaA*");
>+
>+  conv = ArgumentToConv<std::string*>();
>+  EXPECT_EQ(ConvToString(conv), "p");
>+}
>+
>+#if ABSL_INTERNAL_ENABLE_FORMAT_CHECKER
>+
>+struct Case {
>+  bool result;
>+  const char* format;
>+};
>+
>+template <typename... Args>
>+constexpr Case ValidFormat(const char* format) {
>+  return {ValidFormatImpl<ArgumentToConv<Args>()...>(format), format};
>+}
>+
>+TEST(StrFormatChecker, ValidFormat) {
>+  // We want to make sure these expressions are constexpr and they have the
>+  // expected value.
>+  // If they are not constexpr the attribute will just ignore them and not give
>+  // a compile time error.
>+  enum e {};
>+  enum class e2 {};
>+  constexpr Case trues[] = {
>+      ValidFormat<>("abc"),  //
>+
>+      ValidFormat<e>("%d"),                             //
>+      ValidFormat<e2>("%d"),                            //
>+      ValidFormat<int>("%% %d"),                        //
>+      ValidFormat<int>("%ld"),                          //
>+      ValidFormat<int>("%lld"),                         //
>+      ValidFormat<std::string>("%s"),                        //
>+      ValidFormat<std::string>("%10s"),                      //
>+      ValidFormat<int>("%.10x"),                        //
>+      ValidFormat<int, int>("%*.3x"),                   //
>+      ValidFormat<int>("%1.d"),                         //
>+      ValidFormat<int>("%.d"),                          //
>+      ValidFormat<int, double>("%d %g"),                //
>+      ValidFormat<int, std::string>("%*s"),                  //
>+      ValidFormat<int, double>("%.*f"),                 //
>+      ValidFormat<void (*)(), volatile int*>("%p %p"),  //
>+      ValidFormat<string_view, const char*, double, void*>(
>+          "string_view=%s const char*=%s double=%f void*=%p)"),
>+
>+      ValidFormat<int>("%% %1$d"),            //
>+      ValidFormat<int>("%1$ld"),              //
>+      ValidFormat<int>("%1$lld"),             //
>+      ValidFormat<std::string>("%1$s"),            //
>+      ValidFormat<std::string>("%1$10s"),          //
>+      ValidFormat<int>("%1$.10x"),            //
>+      ValidFormat<int>("%1$*1$.*1$d"),        //
>+      ValidFormat<int, int>("%1$*2$.3x"),     //
>+      ValidFormat<int>("%1$1.d"),             //
>+      ValidFormat<int>("%1$.d"),              //
>+      ValidFormat<double, int>("%2$d %1$g"),  //
>+      ValidFormat<int, std::string>("%2$*1$s"),    //
>+      ValidFormat<int, double>("%2$.*1$f"),   //
>+      ValidFormat<void*, string_view, const char*, double>(
>+          "string_view=%2$s const char*=%3$s double=%4$f void*=%1$p "
>+          "repeat=%3$s)")};
>+
>+  for (Case c : trues) {
>+    EXPECT_TRUE(c.result) << c.format;
>+  }
>+
>+  constexpr Case falses[] = {
>+      ValidFormat<int>(""),  //
>+
>+      ValidFormat<e>("%s"),             //
>+      ValidFormat<e2>("%s"),            //
>+      ValidFormat<>("%s"),              //
>+      ValidFormat<>("%r"),              //
>+      ValidFormat<int>("%s"),           //
>+      ValidFormat<int>("%.1.d"),        //
>+      ValidFormat<int>("%*1d"),         //
>+      ValidFormat<int>("%1-d"),         //
>+      ValidFormat<std::string, int>("%*s"),  //
>+      ValidFormat<int>("%*d"),          //
>+      ValidFormat<std::string>("%p"),        //
>+      ValidFormat<int (*)(int)>("%d"),  //
>+
>+      ValidFormat<>("%3$d"),                //
>+      ValidFormat<>("%1$r"),                //
>+      ValidFormat<int>("%1$s"),             //
>+      ValidFormat<int>("%1$.1.d"),          //
>+      ValidFormat<int>("%1$*2$1d"),         //
>+      ValidFormat<int>("%1$1-d"),           //
>+      ValidFormat<std::string, int>("%2$*1$s"),  //
>+      ValidFormat<std::string>("%1$p"),
>+
>+      ValidFormat<int, int>("%d %2$d"),  //
>+  };
>+
>+  for (Case c : falses) {
>+    EXPECT_FALSE(c.result) << c.format;
>+  }
>+}
>+
>+TEST(StrFormatChecker, LongFormat) {
>+#define CHARS_X_40 "1234567890123456789012345678901234567890"
>+#define CHARS_X_400                                                            \
>+  CHARS_X_40 CHARS_X_40 CHARS_X_40 CHARS_X_40 CHARS_X_40 CHARS_X_40 CHARS_X_40 \
>+      CHARS_X_40 CHARS_X_40 CHARS_X_40
>+#define CHARS_X_4000                                                      \
>+  CHARS_X_400 CHARS_X_400 CHARS_X_400 CHARS_X_400 CHARS_X_400 CHARS_X_400 \
>+      CHARS_X_400 CHARS_X_400 CHARS_X_400 CHARS_X_400
>+  constexpr char long_format[] =
>+      CHARS_X_4000 "%d" CHARS_X_4000 "%s" CHARS_X_4000;
>+  constexpr bool is_valid = ValidFormat<int, std::string>(long_format).result;
>+  EXPECT_TRUE(is_valid);
>+}
>+
>+#endif  // ABSL_INTERNAL_ENABLE_FORMAT_CHECKER
>+
>+}  // namespace
>+}  // namespace str_format_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/convert_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/convert_test.cc
>new file mode 100644
>index 00000000000..32f8a0f9ad1
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/convert_test.cc
>@@ -0,0 +1,575 @@
>+#include <errno.h>
>+#include <stdarg.h>
>+#include <stdio.h>
>+#include <cmath>
>+#include <string>
>+
>+#include "gtest/gtest.h"
>+#include "absl/strings/internal/str_format/bind.h"
>+
>+namespace absl {
>+namespace str_format_internal {
>+namespace {
>+
>+template <typename T, size_t N>
>+size_t ArraySize(T (&)[N]) {
>+  return N;
>+}
>+
>+std::string LengthModFor(float) { return ""; }
>+std::string LengthModFor(double) { return ""; }
>+std::string LengthModFor(long double) { return "L"; }
>+std::string LengthModFor(char) { return "hh"; }
>+std::string LengthModFor(signed char) { return "hh"; }
>+std::string LengthModFor(unsigned char) { return "hh"; }
>+std::string LengthModFor(short) { return "h"; }           // NOLINT
>+std::string LengthModFor(unsigned short) { return "h"; }  // NOLINT
>+std::string LengthModFor(int) { return ""; }
>+std::string LengthModFor(unsigned) { return ""; }
>+std::string LengthModFor(long) { return "l"; }                 // NOLINT
>+std::string LengthModFor(unsigned long) { return "l"; }        // NOLINT
>+std::string LengthModFor(long long) { return "ll"; }           // NOLINT
>+std::string LengthModFor(unsigned long long) { return "ll"; }  // NOLINT
>+
>+std::string EscCharImpl(int v) {
>+  if (isprint(v)) return std::string(1, static_cast<char>(v));
>+  char buf[64];
>+  int n = snprintf(buf, sizeof(buf), "\\%#.2x",
>+                   static_cast<unsigned>(v & 0xff));
>+  assert(n > 0 && n < sizeof(buf));
>+  return std::string(buf, n);
>+}
>+
>+std::string Esc(char v) { return EscCharImpl(v); }
>+std::string Esc(signed char v) { return EscCharImpl(v); }
>+std::string Esc(unsigned char v) { return EscCharImpl(v); }
>+
>+template <typename T>
>+std::string Esc(const T &v) {
>+  std::ostringstream oss;
>+  oss << v;
>+  return oss.str();
>+}
>+
>+void StrAppend(std::string *dst, const char *format, va_list ap) {
>+  // First try with a small fixed size buffer
>+  static const int kSpaceLength = 1024;
>+  char space[kSpaceLength];
>+
>+  // It's possible for methods that use a va_list to invalidate
>+  // the data in it upon use.  The fix is to make a copy
>+  // of the structure before using it and use that copy instead.
>+  va_list backup_ap;
>+  va_copy(backup_ap, ap);
>+  int result = vsnprintf(space, kSpaceLength, format, backup_ap);
>+  va_end(backup_ap);
>+  if (result < kSpaceLength) {
>+    if (result >= 0) {
>+      // Normal case -- everything fit.
>+      dst->append(space, result);
>+      return;
>+    }
>+    if (result < 0) {
>+      // Just an error.
>+      return;
>+    }
>+  }
>+
>+  // Increase the buffer size to the size requested by vsnprintf,
>+  // plus one for the closing \0.
>+  int length = result + 1;
>+  char *buf = new char[length];
>+
>+  // Restore the va_list before we use it again
>+  va_copy(backup_ap, ap);
>+  result = vsnprintf(buf, length, format, backup_ap);
>+  va_end(backup_ap);
>+
>+  if (result >= 0 && result < length) {
>+    // It fit
>+    dst->append(buf, result);
>+  }
>+  delete[] buf;
>+}
>+
>+std::string StrPrint(const char *format, ...) {
>+  va_list ap;
>+  va_start(ap, format);
>+  std::string result;
>+  StrAppend(&result, format, ap);
>+  va_end(ap);
>+  return result;
>+}
>+
>+class FormatConvertTest : public ::testing::Test { };
>+
>+template <typename T>
>+void TestStringConvert(const T& str) {
>+  const FormatArgImpl args[] = {FormatArgImpl(str)};
>+  struct Expectation {
>+    const char *out;
>+    const char *fmt;
>+  };
>+  const Expectation kExpect[] = {
>+    {"hello",  "%1$s"      },
>+    {"",       "%1$.s"     },
>+    {"",       "%1$.0s"    },
>+    {"h",      "%1$.1s"    },
>+    {"he",     "%1$.2s"    },
>+    {"hello",  "%1$.10s"   },
>+    {" hello", "%1$6s"     },
>+    {"   he",  "%1$5.2s"   },
>+    {"he   ",  "%1$-5.2s"  },
>+    {"hello ", "%1$-6.10s" },
>+  };
>+  for (const Expectation &e : kExpect) {
>+    UntypedFormatSpecImpl format(e.fmt);
>+    EXPECT_EQ(e.out, FormatPack(format, absl::MakeSpan(args)));
>+  }
>+}
>+
>+TEST_F(FormatConvertTest, BasicString) {
>+  TestStringConvert("hello");  // As char array.
>+  TestStringConvert(static_cast<const char*>("hello"));
>+  TestStringConvert(std::string("hello"));
>+  TestStringConvert(string_view("hello"));
>+}
>+
>+TEST_F(FormatConvertTest, NullString) {
>+  const char* p = nullptr;
>+  UntypedFormatSpecImpl format("%s");
>+  EXPECT_EQ("", FormatPack(format, {FormatArgImpl(p)}));
>+}
>+
>+TEST_F(FormatConvertTest, StringPrecision) {
>+  // We cap at the precision.
>+  char c = 'a';
>+  const char* p = &c;
>+  UntypedFormatSpecImpl format("%.1s");
>+  EXPECT_EQ("a", FormatPack(format, {FormatArgImpl(p)}));
>+
>+  // We cap at the nul terminator.
>+  p = "ABC";
>+  UntypedFormatSpecImpl format2("%.10s");
>+  EXPECT_EQ("ABC", FormatPack(format2, {FormatArgImpl(p)}));
>+}
>+
>+TEST_F(FormatConvertTest, Pointer) {
>+#if _MSC_VER
>+  // MSVC's printf implementation prints pointers differently. We can't easily
>+  // compare our implementation to theirs.
>+  return;
>+#endif
>+  static int x = 0;
>+  const int *xp = &x;
>+  char c = 'h';
>+  char *mcp = &c;
>+  const char *cp = "hi";
>+  const char *cnil = nullptr;
>+  const int *inil = nullptr;
>+  using VoidF = void (*)();
>+  VoidF fp = [] {}, fnil = nullptr;
>+  volatile char vc;
>+  volatile char* vcp = &vc;
>+  volatile char* vcnil = nullptr;
>+  const FormatArgImpl args[] = {
>+      FormatArgImpl(xp),   FormatArgImpl(cp),  FormatArgImpl(inil),
>+      FormatArgImpl(cnil), FormatArgImpl(mcp), FormatArgImpl(fp),
>+      FormatArgImpl(fnil), FormatArgImpl(vcp), FormatArgImpl(vcnil),
>+  };
>+  struct Expectation {
>+    std::string out;
>+    const char *fmt;
>+  };
>+  const Expectation kExpect[] = {
>+      {StrPrint("%p", &x), "%p"},
>+      {StrPrint("%20p", &x), "%20p"},
>+      {StrPrint("%.1p", &x), "%.1p"},
>+      {StrPrint("%.20p", &x), "%.20p"},
>+      {StrPrint("%30.20p", &x), "%30.20p"},
>+
>+      {StrPrint("%-p", &x), "%-p"},
>+      {StrPrint("%-20p", &x), "%-20p"},
>+      {StrPrint("%-.1p", &x), "%-.1p"},
>+      {StrPrint("%.20p", &x), "%.20p"},
>+      {StrPrint("%-30.20p", &x), "%-30.20p"},
>+
>+      {StrPrint("%p", cp), "%2$p"},   // const char*
>+      {"(nil)", "%3$p"},              // null const char *
>+      {"(nil)", "%4$p"},              // null const int *
>+      {StrPrint("%p", mcp), "%5$p"},  // nonconst char*
>+
>+      {StrPrint("%p", fp), "%6$p"},   // function pointer
>+      {StrPrint("%p", vcp), "%8$p"},  // function pointer
>+
>+#ifndef __APPLE__
>+      // Apple's printf differs here (0x0 vs. nil)
>+      {StrPrint("%p", fnil), "%7$p"},   // null function pointer
>+      {StrPrint("%p", vcnil), "%9$p"},  // null function pointer
>+#endif
>+  };
>+  for (const Expectation &e : kExpect) {
>+    UntypedFormatSpecImpl format(e.fmt);
>+    EXPECT_EQ(e.out, FormatPack(format, absl::MakeSpan(args))) << e.fmt;
>+  }
>+}
>+
>+struct Cardinal {
>+  enum Pos { k1 = 1, k2 = 2, k3 = 3 };
>+  enum Neg { kM1 = -1, kM2 = -2, kM3 = -3 };
>+};
>+
>+TEST_F(FormatConvertTest, Enum) {
>+  const Cardinal::Pos k3 = Cardinal::k3;
>+  const Cardinal::Neg km3 = Cardinal::kM3;
>+  const FormatArgImpl args[] = {FormatArgImpl(k3), FormatArgImpl(km3)};
>+  UntypedFormatSpecImpl format("%1$d");
>+  UntypedFormatSpecImpl format2("%2$d");
>+  EXPECT_EQ("3", FormatPack(format, absl::MakeSpan(args)));
>+  EXPECT_EQ("-3", FormatPack(format2, absl::MakeSpan(args)));
>+}
>+
>+template <typename T>
>+class TypedFormatConvertTest : public FormatConvertTest { };
>+
>+TYPED_TEST_CASE_P(TypedFormatConvertTest);
>+
>+std::vector<std::string> AllFlagCombinations() {
>+  const char kFlags[] = {'-', '#', '0', '+', ' '};
>+  std::vector<std::string> result;
>+  for (size_t fsi = 0; fsi < (1ull << ArraySize(kFlags)); ++fsi) {
>+    std::string flag_set;
>+    for (size_t fi = 0; fi < ArraySize(kFlags); ++fi)
>+      if (fsi & (1ull << fi))
>+        flag_set += kFlags[fi];
>+    result.push_back(flag_set);
>+  }
>+  return result;
>+}
>+
>+TYPED_TEST_P(TypedFormatConvertTest, AllIntsWithFlags) {
>+  typedef TypeParam T;
>+  typedef typename std::make_unsigned<T>::type UnsignedT;
>+  using remove_volatile_t = typename std::remove_volatile<T>::type;
>+  const T kMin = std::numeric_limits<remove_volatile_t>::min();
>+  const T kMax = std::numeric_limits<remove_volatile_t>::max();
>+  const T kVals[] = {
>+      remove_volatile_t(1),
>+      remove_volatile_t(2),
>+      remove_volatile_t(3),
>+      remove_volatile_t(123),
>+      remove_volatile_t(-1),
>+      remove_volatile_t(-2),
>+      remove_volatile_t(-3),
>+      remove_volatile_t(-123),
>+      remove_volatile_t(0),
>+      kMax - remove_volatile_t(1),
>+      kMax,
>+      kMin + remove_volatile_t(1),
>+      kMin,
>+  };
>+  const char kConvChars[] = {'d', 'i', 'u', 'o', 'x', 'X'};
>+  const std::string kWid[] = {"", "4", "10"};
>+  const std::string kPrec[] = {"", ".", ".0", ".4", ".10"};
>+
>+  const std::vector<std::string> flag_sets = AllFlagCombinations();
>+
>+  for (size_t vi = 0; vi < ArraySize(kVals); ++vi) {
>+    const T val = kVals[vi];
>+    SCOPED_TRACE(Esc(val));
>+    const FormatArgImpl args[] = {FormatArgImpl(val)};
>+    for (size_t ci = 0; ci < ArraySize(kConvChars); ++ci) {
>+      const char conv_char = kConvChars[ci];
>+      for (size_t fsi = 0; fsi < flag_sets.size(); ++fsi) {
>+        const std::string &flag_set = flag_sets[fsi];
>+        for (size_t wi = 0; wi < ArraySize(kWid); ++wi) {
>+          const std::string &wid = kWid[wi];
>+          for (size_t pi = 0; pi < ArraySize(kPrec); ++pi) {
>+            const std::string &prec = kPrec[pi];
>+
>+            const bool is_signed_conv = (conv_char == 'd' || conv_char == 'i');
>+            const bool is_unsigned_to_signed =
>+                !std::is_signed<T>::value && is_signed_conv;
>+            // Don't consider sign-related flags '+' and ' ' when doing
>+            // unsigned to signed conversions.
>+            if (is_unsigned_to_signed &&
>+                flag_set.find_first_of("+ ") != std::string::npos) {
>+              continue;
>+            }
>+
>+            std::string new_fmt("%");
>+            new_fmt += flag_set;
>+            new_fmt += wid;
>+            new_fmt += prec;
>+            // old and new always agree up to here.
>+            std::string old_fmt = new_fmt;
>+            new_fmt += conv_char;
>+            std::string old_result;
>+            if (is_unsigned_to_signed) {
>+              // don't expect agreement on unsigned formatted as signed,
>+              // as printf can't do that conversion properly. For those
>+              // cases, we do expect agreement with printf with a "%u"
>+              // and the unsigned equivalent of 'val'.
>+              UnsignedT uval = val;
>+              old_fmt += LengthModFor(uval);
>+              old_fmt += "u";
>+              old_result = StrPrint(old_fmt.c_str(), uval);
>+            } else {
>+              old_fmt += LengthModFor(val);
>+              old_fmt += conv_char;
>+              old_result = StrPrint(old_fmt.c_str(), val);
>+            }
>+
>+            SCOPED_TRACE(std::string() + " old_fmt: \"" + old_fmt +
>+                         "\"'"
>+                         " new_fmt: \"" +
>+                         new_fmt + "\"");
>+            UntypedFormatSpecImpl format(new_fmt);
>+            EXPECT_EQ(old_result, FormatPack(format, absl::MakeSpan(args)));
>+          }
>+        }
>+      }
>+    }
>+  }
>+}
>+
>+TYPED_TEST_P(TypedFormatConvertTest, Char) {
>+  typedef TypeParam T;
>+  using remove_volatile_t = typename std::remove_volatile<T>::type;
>+  static const T kMin = std::numeric_limits<remove_volatile_t>::min();
>+  static const T kMax = std::numeric_limits<remove_volatile_t>::max();
>+  T kVals[] = {
>+    remove_volatile_t(1), remove_volatile_t(2), remove_volatile_t(10),
>+    remove_volatile_t(-1), remove_volatile_t(-2), remove_volatile_t(-10),
>+    remove_volatile_t(0),
>+    kMin + remove_volatile_t(1), kMin,
>+    kMax - remove_volatile_t(1), kMax
>+  };
>+  for (const T &c : kVals) {
>+    const FormatArgImpl args[] = {FormatArgImpl(c)};
>+    UntypedFormatSpecImpl format("%c");
>+    EXPECT_EQ(StrPrint("%c", c), FormatPack(format, absl::MakeSpan(args)));
>+  }
>+}
>+
>+REGISTER_TYPED_TEST_CASE_P(TypedFormatConvertTest, AllIntsWithFlags, Char);
>+
>+typedef ::testing::Types<
>+    int, unsigned, volatile int,
>+    short, unsigned short,
>+    long, unsigned long,
>+    long long, unsigned long long,
>+    signed char, unsigned char, char>
>+    AllIntTypes;
>+INSTANTIATE_TYPED_TEST_CASE_P(TypedFormatConvertTestWithAllIntTypes,
>+                              TypedFormatConvertTest, AllIntTypes);
>+TEST_F(FormatConvertTest, Uint128) {
>+  absl::uint128 v = static_cast<absl::uint128>(0x1234567890abcdef) * 1979;
>+  absl::uint128 max = absl::Uint128Max();
>+  const FormatArgImpl args[] = {FormatArgImpl(v), FormatArgImpl(max)};
>+
>+  struct Case {
>+    const char* format;
>+    const char* expected;
>+  } cases[] = {
>+      {"%1$d", "2595989796776606496405"},
>+      {"%1$30d", "        2595989796776606496405"},
>+      {"%1$-30d", "2595989796776606496405        "},
>+      {"%1$u", "2595989796776606496405"},
>+      {"%1$x", "8cba9876066020f695"},
>+      {"%2$d", "340282366920938463463374607431768211455"},
>+      {"%2$u", "340282366920938463463374607431768211455"},
>+      {"%2$x", "ffffffffffffffffffffffffffffffff"},
>+  };
>+
>+  for (auto c : cases) {
>+    UntypedFormatSpecImpl format(c.format);
>+    EXPECT_EQ(c.expected, FormatPack(format, absl::MakeSpan(args)));
>+  }
>+}
>+
>+TEST_F(FormatConvertTest, Float) {
>+#if _MSC_VER
>+  // MSVC has a different rounding policy than us so we can't test our
>+  // implementation against the native one there.
>+  return;
>+#endif  // _MSC_VER
>+
>+  const char *const kFormats[] = {
>+      "%",  "%.3",  "%8.5",   "%9",   "%.60", "%.30",   "%03",    "%+",
>+      "% ", "%-10", "%#15.3", "%#.0", "%.0",  "%1$*2$", "%1$.*2$"};
>+
>+  std::vector<double> doubles = {0.0,
>+                                 -0.0,
>+                                 .99999999999999,
>+                                 99999999999999.,
>+                                 std::numeric_limits<double>::max(),
>+                                 -std::numeric_limits<double>::max(),
>+                                 std::numeric_limits<double>::min(),
>+                                 -std::numeric_limits<double>::min(),
>+                                 std::numeric_limits<double>::lowest(),
>+                                 -std::numeric_limits<double>::lowest(),
>+                                 std::numeric_limits<double>::epsilon(),
>+                                 std::numeric_limits<double>::epsilon() + 1,
>+                                 std::numeric_limits<double>::infinity(),
>+                                 -std::numeric_limits<double>::infinity()};
>+
>+#ifndef __APPLE__
>+  // Apple formats NaN differently (+nan) vs. (nan)
>+  doubles.push_back(std::nan(""));
>+#endif
>+
>+  // Some regression tests.
>+  doubles.push_back(0.99999999999999989);
>+
>+  if (std::numeric_limits<double>::has_denorm != std::denorm_absent) {
>+    doubles.push_back(std::numeric_limits<double>::denorm_min());
>+    doubles.push_back(-std::numeric_limits<double>::denorm_min());
>+  }
>+
>+  for (double base :
>+       {1., 12., 123., 1234., 12345., 123456., 1234567., 12345678., 123456789.,
>+        1234567890., 12345678901., 123456789012., 1234567890123.}) {
>+    for (int exp = -123; exp <= 123; ++exp) {
>+      for (int sign : {1, -1}) {
>+        doubles.push_back(sign * std::ldexp(base, exp));
>+      }
>+    }
>+  }
>+
>+  for (const char *fmt : kFormats) {
>+    for (char f : {'f', 'F',  //
>+                   'g', 'G',  //
>+                   'a', 'A',  //
>+                   'e', 'E'}) {
>+      std::string fmt_str = std::string(fmt) + f;
>+      for (double d : doubles) {
>+        int i = -10;
>+        FormatArgImpl args[2] = {FormatArgImpl(d), FormatArgImpl(i)};
>+        UntypedFormatSpecImpl format(fmt_str);
>+        // We use ASSERT_EQ here because failures are usually correlated and a
>+        // bug would print way too many failed expectations causing the test to
>+        // time out.
>+        ASSERT_EQ(StrPrint(fmt_str.c_str(), d, i),
>+                  FormatPack(format, absl::MakeSpan(args)))
>+            << fmt_str << " " << StrPrint("%.18g", d) << " "
>+            << StrPrint("%.999f", d);
>+      }
>+    }
>+  }
>+}
>+
>+TEST_F(FormatConvertTest, LongDouble) {
>+  const char *const kFormats[] = {"%",    "%.3", "%8.5", "%9",
>+                                  "%.60", "%+",  "% ",   "%-10"};
>+
>+  // This value is not representable in double, but it is in long double that
>+  // uses the extended format.
>+  // This is to verify that we are not truncating the value mistakenly through a
>+  // double.
>+  long double very_precise = 10000000000000000.25L;
>+
>+  std::vector<long double> doubles = {
>+      0.0,
>+      -0.0,
>+      very_precise,
>+      1 / very_precise,
>+      std::numeric_limits<long double>::max(),
>+      -std::numeric_limits<long double>::max(),
>+      std::numeric_limits<long double>::min(),
>+      -std::numeric_limits<long double>::min(),
>+      std::numeric_limits<long double>::infinity(),
>+      -std::numeric_limits<long double>::infinity()};
>+
>+  for (const char *fmt : kFormats) {
>+    for (char f : {'f', 'F',  //
>+                   'g', 'G',  //
>+                   'a', 'A',  //
>+                   'e', 'E'}) {
>+      std::string fmt_str = std::string(fmt) + 'L' + f;
>+      for (auto d : doubles) {
>+        FormatArgImpl arg(d);
>+        UntypedFormatSpecImpl format(fmt_str);
>+        // We use ASSERT_EQ here because failures are usually correlated and a
>+        // bug would print way too many failed expectations causing the test to
>+        // time out.
>+        ASSERT_EQ(StrPrint(fmt_str.c_str(), d),
>+                  FormatPack(format, {&arg, 1}))
>+            << fmt_str << " " << StrPrint("%.18Lg", d) << " "
>+            << StrPrint("%.999Lf", d);
>+      }
>+    }
>+  }
>+}
>+
>+TEST_F(FormatConvertTest, IntAsFloat) {
>+  const int kMin = std::numeric_limits<int>::min();
>+  const int kMax = std::numeric_limits<int>::max();
>+  const int ia[] = {
>+    1, 2, 3, 123,
>+    -1, -2, -3, -123,
>+    0, kMax - 1, kMax, kMin + 1, kMin };
>+  for (const int fx : ia) {
>+    SCOPED_TRACE(fx);
>+    const FormatArgImpl args[] = {FormatArgImpl(fx)};
>+    struct Expectation {
>+      int line;
>+      std::string out;
>+      const char *fmt;
>+    };
>+    const double dx = static_cast<double>(fx);
>+    const Expectation kExpect[] = {
>+      { __LINE__, StrPrint("%f", dx), "%f" },
>+      { __LINE__, StrPrint("%12f", dx), "%12f" },
>+      { __LINE__, StrPrint("%.12f", dx), "%.12f" },
>+      { __LINE__, StrPrint("%12a", dx), "%12a" },
>+      { __LINE__, StrPrint("%.12a", dx), "%.12a" },
>+    };
>+    for (const Expectation &e : kExpect) {
>+      SCOPED_TRACE(e.line);
>+      SCOPED_TRACE(e.fmt);
>+      UntypedFormatSpecImpl format(e.fmt);
>+      EXPECT_EQ(e.out, FormatPack(format, absl::MakeSpan(args)));
>+    }
>+  }
>+}
>+
>+template <typename T>
>+bool FormatFails(const char* test_format, T value) {
>+  std::string format_string = std::string("<<") + test_format + ">>";
>+  UntypedFormatSpecImpl format(format_string);
>+
>+  int one = 1;
>+  const FormatArgImpl args[] = {FormatArgImpl(value), FormatArgImpl(one)};
>+  EXPECT_EQ(FormatPack(format, absl::MakeSpan(args)), "")
>+      << "format=" << test_format << " value=" << value;
>+  return FormatPack(format, absl::MakeSpan(args)).empty();
>+}
>+
>+TEST_F(FormatConvertTest, ExpectedFailures) {
>+  // Int input
>+  EXPECT_TRUE(FormatFails("%p", 1));
>+  EXPECT_TRUE(FormatFails("%s", 1));
>+  EXPECT_TRUE(FormatFails("%n", 1));
>+
>+  // Double input
>+  EXPECT_TRUE(FormatFails("%p", 1.));
>+  EXPECT_TRUE(FormatFails("%s", 1.));
>+  EXPECT_TRUE(FormatFails("%n", 1.));
>+  EXPECT_TRUE(FormatFails("%c", 1.));
>+  EXPECT_TRUE(FormatFails("%d", 1.));
>+  EXPECT_TRUE(FormatFails("%x", 1.));
>+  EXPECT_TRUE(FormatFails("%*d", 1.));
>+
>+  // String input
>+  EXPECT_TRUE(FormatFails("%n", ""));
>+  EXPECT_TRUE(FormatFails("%c", ""));
>+  EXPECT_TRUE(FormatFails("%d", ""));
>+  EXPECT_TRUE(FormatFails("%x", ""));
>+  EXPECT_TRUE(FormatFails("%f", ""));
>+  EXPECT_TRUE(FormatFails("%*d", ""));
>+}
>+
>+}  // namespace
>+}  // namespace str_format_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/extension.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/extension.cc
>new file mode 100644
>index 00000000000..c2174703c3e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/extension.cc
>@@ -0,0 +1,84 @@
>+//
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/strings/internal/str_format/extension.h"
>+
>+#include <errno.h>
>+#include <algorithm>
>+#include <string>
>+
>+namespace absl {
>+namespace str_format_internal {
>+namespace {
>+// clang-format off
>+#define ABSL_LENGTH_MODS_EXPAND_ \
>+  X_VAL(h) X_SEP \
>+  X_VAL(hh) X_SEP \
>+  X_VAL(l) X_SEP \
>+  X_VAL(ll) X_SEP \
>+  X_VAL(L) X_SEP \
>+  X_VAL(j) X_SEP \
>+  X_VAL(z) X_SEP \
>+  X_VAL(t) X_SEP \
>+  X_VAL(q)
>+// clang-format on
>+}  // namespace
>+
>+const LengthMod::Spec LengthMod::kSpecs[] = {
>+#define X_VAL(id) { LengthMod::id, #id, strlen(#id) }
>+#define X_SEP ,
>+    ABSL_LENGTH_MODS_EXPAND_, {LengthMod::none, "", 0}
>+#undef X_VAL
>+#undef X_SEP
>+};
>+
>+const ConversionChar::Spec ConversionChar::kSpecs[] = {
>+#define X_VAL(id) { ConversionChar::id, #id[0] }
>+#define X_SEP ,
>+    ABSL_CONVERSION_CHARS_EXPAND_(X_VAL, X_SEP),
>+    {ConversionChar::none, '\0'},
>+#undef X_VAL
>+#undef X_SEP
>+};
>+
>+std::string Flags::ToString() const {
>+  std::string s;
>+  s.append(left     ? "-" : "");
>+  s.append(show_pos ? "+" : "");
>+  s.append(sign_col ? " " : "");
>+  s.append(alt      ? "#" : "");
>+  s.append(zero     ? "0" : "");
>+  return s;
>+}
>+
>+const size_t LengthMod::kNumValues;
>+
>+const size_t ConversionChar::kNumValues;
>+
>+bool FormatSinkImpl::PutPaddedString(string_view v, int w, int p, bool l) {
>+  size_t space_remaining = 0;
>+  if (w >= 0) space_remaining = w;
>+  size_t n = v.size();
>+  if (p >= 0) n = std::min(n, static_cast<size_t>(p));
>+  string_view shown(v.data(), n);
>+  space_remaining = Excess(shown.size(), space_remaining);
>+  if (!l) Append(space_remaining, ' ');
>+  Append(shown);
>+  if (l) Append(space_remaining, ' ');
>+  return true;
>+}
>+
>+}  // namespace str_format_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/extension.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/extension.h
>new file mode 100644
>index 00000000000..810330b9d71
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/extension.h
>@@ -0,0 +1,406 @@
>+//
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+//
>+#ifndef ABSL_STRINGS_INTERNAL_STR_FORMAT_EXTENSION_H_
>+#define ABSL_STRINGS_INTERNAL_STR_FORMAT_EXTENSION_H_
>+
>+#include <limits.h>
>+#include <cstring>
>+#include <ostream>
>+
>+#include "absl/base/port.h"
>+#include "absl/strings/internal/str_format/output.h"
>+#include "absl/strings/string_view.h"
>+
>+class Cord;
>+
>+namespace absl {
>+
>+namespace str_format_internal {
>+
>+class FormatRawSinkImpl {
>+ public:
>+  // Implicitly convert from any type that provides the hook function as
>+  // described above.
>+  template <typename T, decltype(str_format_internal::InvokeFlush(
>+                            std::declval<T*>(), string_view()))* = nullptr>
>+  FormatRawSinkImpl(T* raw)  // NOLINT
>+      : sink_(raw), write_(&FormatRawSinkImpl::Flush<T>) {}
>+
>+  void Write(string_view s) { write_(sink_, s); }
>+
>+  template <typename T>
>+  static FormatRawSinkImpl Extract(T s) {
>+    return s.sink_;
>+  }
>+
>+ private:
>+  template <typename T>
>+  static void Flush(void* r, string_view s) {
>+    str_format_internal::InvokeFlush(static_cast<T*>(r), s);
>+  }
>+
>+  void* sink_;
>+  void (*write_)(void*, string_view);
>+};
>+
>+// An abstraction to which conversions write their std::string data.
>+class FormatSinkImpl {
>+ public:
>+  explicit FormatSinkImpl(FormatRawSinkImpl raw) : raw_(raw) {}
>+
>+  ~FormatSinkImpl() { Flush(); }
>+
>+  void Flush() {
>+    raw_.Write(string_view(buf_, pos_ - buf_));
>+    pos_ = buf_;
>+  }
>+
>+  void Append(size_t n, char c) {
>+    if (n == 0) return;
>+    size_ += n;
>+    auto raw_append = [&](size_t count) {
>+      memset(pos_, c, count);
>+      pos_ += count;
>+    };
>+    while (n > Avail()) {
>+      n -= Avail();
>+      if (Avail() > 0) {
>+        raw_append(Avail());
>+      }
>+      Flush();
>+    }
>+    raw_append(n);
>+  }
>+
>+  void Append(string_view v) {
>+    size_t n = v.size();
>+    if (n == 0) return;
>+    size_ += n;
>+    if (n >= Avail()) {
>+      Flush();
>+      raw_.Write(v);
>+      return;
>+    }
>+    memcpy(pos_, v.data(), n);
>+    pos_ += n;
>+  }
>+
>+  size_t size() const { return size_; }
>+
>+  // Put 'v' to 'sink' with specified width, precision, and left flag.
>+  bool PutPaddedString(string_view v, int w, int p, bool l);
>+
>+  template <typename T>
>+  T Wrap() {
>+    return T(this);
>+  }
>+
>+  template <typename T>
>+  static FormatSinkImpl* Extract(T* s) {
>+    return s->sink_;
>+  }
>+
>+ private:
>+  size_t Avail() const { return buf_ + sizeof(buf_) - pos_; }
>+
>+  FormatRawSinkImpl raw_;
>+  size_t size_ = 0;
>+  char* pos_ = buf_;
>+  char buf_[1024];
>+};
>+
>+struct Flags {
>+  bool basic : 1;     // fastest conversion: no flags, width, or precision
>+  bool left : 1;      // "-"
>+  bool show_pos : 1;  // "+"
>+  bool sign_col : 1;  // " "
>+  bool alt : 1;       // "#"
>+  bool zero : 1;      // "0"
>+  std::string ToString() const;
>+  friend std::ostream& operator<<(std::ostream& os, const Flags& v) {
>+    return os << v.ToString();
>+  }
>+};
>+
>+struct LengthMod {
>+ public:
>+  enum Id : uint8_t {
>+    h, hh, l, ll, L, j, z, t, q, none
>+  };
>+  static const size_t kNumValues = none + 1;
>+
>+  LengthMod() : id_(none) {}
>+
>+  // Index into the opaque array of LengthMod enums.
>+  // Requires: i < kNumValues
>+  static LengthMod FromIndex(size_t i) {
>+    return LengthMod(kSpecs[i].value);
>+  }
>+
>+  static LengthMod FromId(Id id) { return LengthMod(id); }
>+
>+  // The length modifier std::string associated with a specified LengthMod.
>+  string_view name() const {
>+    const Spec& spec = kSpecs[id_];
>+    return {spec.name, spec.name_length};
>+  }
>+
>+  Id id() const { return id_; }
>+
>+  friend bool operator==(const LengthMod& a, const LengthMod& b) {
>+    return a.id() == b.id();
>+  }
>+  friend bool operator!=(const LengthMod& a, const LengthMod& b) {
>+    return !(a == b);
>+  }
>+  friend std::ostream& operator<<(std::ostream& os, const LengthMod& v) {
>+    return os << v.name();
>+  }
>+
>+ private:
>+  struct Spec {
>+    Id value;
>+    const char *name;
>+    size_t name_length;
>+  };
>+  static const Spec kSpecs[];
>+
>+  explicit LengthMod(Id id) : id_(id) {}
>+
>+  Id id_;
>+};
>+
>+// clang-format off
>+#define ABSL_CONVERSION_CHARS_EXPAND_(X_VAL, X_SEP) \
>+  /* text */ \
>+  X_VAL(c) X_SEP X_VAL(C) X_SEP X_VAL(s) X_SEP X_VAL(S) X_SEP \
>+  /* ints */ \
>+  X_VAL(d) X_SEP X_VAL(i) X_SEP X_VAL(o) X_SEP \
>+  X_VAL(u) X_SEP X_VAL(x) X_SEP X_VAL(X) X_SEP \
>+  /* floats */ \
>+  X_VAL(f) X_SEP X_VAL(F) X_SEP X_VAL(e) X_SEP X_VAL(E) X_SEP \
>+  X_VAL(g) X_SEP X_VAL(G) X_SEP X_VAL(a) X_SEP X_VAL(A) X_SEP \
>+  /* misc */ \
>+  X_VAL(n) X_SEP X_VAL(p)
>+// clang-format on
>+
>+struct ConversionChar {
>+ public:
>+  enum Id : uint8_t {
>+    c, C, s, S,              // text
>+    d, i, o, u, x, X,        // int
>+    f, F, e, E, g, G, a, A,  // float
>+    n, p,                    // misc
>+    none
>+  };
>+  static const size_t kNumValues = none + 1;
>+
>+  ConversionChar() : id_(none) {}
>+
>+ public:
>+  // Index into the opaque array of ConversionChar enums.
>+  // Requires: i < kNumValues
>+  static ConversionChar FromIndex(size_t i) {
>+    return ConversionChar(kSpecs[i].value);
>+  }
>+
>+  static ConversionChar FromChar(char c) {
>+    ConversionChar::Id out_id = ConversionChar::none;
>+    switch (c) {
>+#define X_VAL(id)                \
>+  case #id[0]:                   \
>+    out_id = ConversionChar::id; \
>+    break;
>+      ABSL_CONVERSION_CHARS_EXPAND_(X_VAL, )
>+#undef X_VAL
>+      default:
>+        break;
>+    }
>+    return ConversionChar(out_id);
>+  }
>+
>+  static ConversionChar FromId(Id id) { return ConversionChar(id); }
>+  Id id() const { return id_; }
>+
>+  int radix() const {
>+    switch (id()) {
>+      case x: case X: case a: case A: case p: return 16;
>+      case o: return 8;
>+      default: return 10;
>+    }
>+  }
>+
>+  bool upper() const {
>+    switch (id()) {
>+      case X: case F: case E: case G: case A: return true;
>+      default: return false;
>+    }
>+  }
>+
>+  bool is_signed() const {
>+    switch (id()) {
>+      case d: case i: return true;
>+      default: return false;
>+    }
>+  }
>+
>+  bool is_integral() const {
>+    switch (id()) {
>+      case d: case i: case u: case o: case x: case X:
>+        return true;
>+      default: return false;
>+    }
>+  }
>+
>+  bool is_float() const {
>+    switch (id()) {
>+      case a: case e: case f: case g: case A: case E: case F: case G:
>+        return true;
>+      default: return false;
>+    }
>+  }
>+
>+  bool IsValid() const { return id() != none; }
>+
>+  // The associated char.
>+  char Char() const { return kSpecs[id_].name; }
>+
>+  friend bool operator==(const ConversionChar& a, const ConversionChar& b) {
>+    return a.id() == b.id();
>+  }
>+  friend bool operator!=(const ConversionChar& a, const ConversionChar& b) {
>+    return !(a == b);
>+  }
>+  friend std::ostream& operator<<(std::ostream& os, const ConversionChar& v) {
>+    char c = v.Char();
>+    if (!c) c = '?';
>+    return os << c;
>+  }
>+
>+ private:
>+  struct Spec {
>+    Id value;
>+    char name;
>+  };
>+  static const Spec kSpecs[];
>+
>+  explicit ConversionChar(Id id) : id_(id) {}
>+
>+  Id id_;
>+};
>+
>+class ConversionSpec {
>+ public:
>+  Flags flags() const { return flags_; }
>+  LengthMod length_mod() const { return length_mod_; }
>+  ConversionChar conv() const { return conv_; }
>+
>+  // Returns the specified width. If width is unspecfied, it returns a negative
>+  // value.
>+  int width() const { return width_; }
>+  // Returns the specified precision. If precision is unspecfied, it returns a
>+  // negative value.
>+  int precision() const { return precision_; }
>+
>+  void set_flags(Flags f) { flags_ = f; }
>+  void set_length_mod(LengthMod lm) { length_mod_ = lm; }
>+  void set_conv(ConversionChar c) { conv_ = c; }
>+  void set_width(int w) { width_ = w; }
>+  void set_precision(int p) { precision_ = p; }
>+  void set_left(bool b) { flags_.left = b; }
>+
>+ private:
>+  Flags flags_;
>+  LengthMod length_mod_;
>+  ConversionChar conv_;
>+  int width_;
>+  int precision_;
>+};
>+
>+constexpr uint64_t ConversionCharToConvValue(char conv) {
>+  return
>+#define CONV_SET_CASE(c) \
>+  conv == #c[0] ? (uint64_t{1} << (1 + ConversionChar::Id::c)):
>+      ABSL_CONVERSION_CHARS_EXPAND_(CONV_SET_CASE, )
>+#undef CONV_SET_CASE
>+                  conv == '*'
>+          ? 1
>+          : 0;
>+}
>+
>+enum class Conv : uint64_t {
>+#define CONV_SET_CASE(c) c = ConversionCharToConvValue(#c[0]),
>+  ABSL_CONVERSION_CHARS_EXPAND_(CONV_SET_CASE, )
>+#undef CONV_SET_CASE
>+
>+  // Used for width/precision '*' specification.
>+  star = ConversionCharToConvValue('*'),
>+
>+  // Some predefined values:
>+  integral = d | i | u | o | x | X,
>+  floating = a | e | f | g | A | E | F | G,
>+  numeric = integral | floating,
>+  string = s,  // absl:ignore(std::string)
>+  pointer = p
>+};
>+
>+// Type safe OR operator.
>+// We need this for two reasons:
>+//  1. operator| on enums makes them decay to integers and the result is an
>+//     integer. We need the result to stay as an enum.
>+//  2. We use "enum class" which would not work even if we accepted the decay.
>+constexpr Conv operator|(Conv a, Conv b) {
>+  return Conv(static_cast<uint64_t>(a) | static_cast<uint64_t>(b));
>+}
>+
>+// Get a conversion with a single character in it.
>+constexpr Conv ConversionCharToConv(char c) {
>+  return Conv(ConversionCharToConvValue(c));
>+}
>+
>+// Checks whether `c` exists in `set`.
>+constexpr bool Contains(Conv set, char c) {
>+  return (static_cast<uint64_t>(set) & ConversionCharToConvValue(c)) != 0;
>+}
>+
>+// Checks whether all the characters in `c` are contained in `set`
>+constexpr bool Contains(Conv set, Conv c) {
>+  return (static_cast<uint64_t>(set) & static_cast<uint64_t>(c)) ==
>+         static_cast<uint64_t>(c);
>+}
>+
>+// Return type of the AbslFormatConvert() functions.
>+// The Conv template parameter is used to inform the framework of what
>+// conversion characters are supported by that AbslFormatConvert routine.
>+template <Conv C>
>+struct ConvertResult {
>+  static constexpr Conv kConv = C;
>+  bool value;
>+};
>+template <Conv C>
>+constexpr Conv ConvertResult<C>::kConv;
>+
>+// Return capacity - used, clipped to a minimum of 0.
>+inline size_t Excess(size_t used, size_t capacity) {
>+  return used < capacity ? capacity - used : 0;
>+}
>+
>+}  // namespace str_format_internal
>+
>+}  // namespace absl
>+
>+#endif  // ABSL_STRINGS_STR_FORMAT_EXTENSION_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/extension_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/extension_test.cc
>new file mode 100644
>index 00000000000..224fc923d3e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/extension_test.cc
>@@ -0,0 +1,65 @@
>+//
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+
>+#include "absl/strings/internal/str_format/extension.h"
>+
>+#include <random>
>+#include <string>
>+#include "absl/strings/str_format.h"
>+
>+#include "gtest/gtest.h"
>+
>+namespace {
>+
>+std::string MakeRandomString(size_t len) {
>+  std::random_device rd;
>+  std::mt19937 gen(rd());
>+  std::uniform_int_distribution<> dis('a', 'z');
>+  std::string s(len, '0');
>+  for (char& c : s) {
>+    c = dis(gen);
>+  }
>+  return s;
>+}
>+
>+TEST(FormatExtensionTest, SinkAppendSubstring) {
>+  for (size_t chunk_size : {1, 10, 100, 1000, 10000}) {
>+    std::string expected, actual;
>+    absl::str_format_internal::FormatSinkImpl sink(&actual);
>+    for (size_t chunks = 0; chunks < 10; ++chunks) {
>+      std::string rand = MakeRandomString(chunk_size);
>+      expected += rand;
>+      sink.Append(rand);
>+    }
>+    sink.Flush();
>+    EXPECT_EQ(actual, expected);
>+  }
>+}
>+
>+TEST(FormatExtensionTest, SinkAppendChars) {
>+  for (size_t chunk_size : {1, 10, 100, 1000, 10000}) {
>+    std::string expected, actual;
>+    absl::str_format_internal::FormatSinkImpl sink(&actual);
>+    for (size_t chunks = 0; chunks < 10; ++chunks) {
>+      std::string rand = MakeRandomString(1);
>+      expected.append(chunk_size, rand[0]);
>+      sink.Append(chunk_size, rand[0]);
>+    }
>+    sink.Flush();
>+    EXPECT_EQ(actual, expected);
>+  }
>+}
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/float_conversion.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/float_conversion.cc
>new file mode 100644
>index 00000000000..6176db9cb5a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/float_conversion.cc
>@@ -0,0 +1,483 @@
>+#include "absl/strings/internal/str_format/float_conversion.h"
>+
>+#include <string.h>
>+#include <algorithm>
>+#include <cassert>
>+#include <cmath>
>+#include <string>
>+
>+namespace absl {
>+namespace str_format_internal {
>+
>+namespace {
>+
>+char *CopyStringTo(string_view v, char *out) {
>+  std::memcpy(out, v.data(), v.size());
>+  return out + v.size();
>+}
>+
>+template <typename Float>
>+bool FallbackToSnprintf(const Float v, const ConversionSpec &conv,
>+                        FormatSinkImpl *sink) {
>+  int w = conv.width() >= 0 ? conv.width() : 0;
>+  int p = conv.precision() >= 0 ? conv.precision() : -1;
>+  char fmt[32];
>+  {
>+    char *fp = fmt;
>+    *fp++ = '%';
>+    fp = CopyStringTo(conv.flags().ToString(), fp);
>+    fp = CopyStringTo("*.*", fp);
>+    if (std::is_same<long double, Float>()) {
>+      *fp++ = 'L';
>+    }
>+    *fp++ = conv.conv().Char();
>+    *fp = 0;
>+    assert(fp < fmt + sizeof(fmt));
>+  }
>+  std::string space(512, '\0');
>+  string_view result;
>+  while (true) {
>+    int n = snprintf(&space[0], space.size(), fmt, w, p, v);
>+    if (n < 0) return false;
>+    if (static_cast<size_t>(n) < space.size()) {
>+      result = string_view(space.data(), n);
>+      break;
>+    }
>+    space.resize(n + 1);
>+  }
>+  sink->Append(result);
>+  return true;
>+}
>+
>+// 128-bits in decimal: ceil(128*log(2)/log(10))
>+//   or std::numeric_limits<__uint128_t>::digits10
>+constexpr int kMaxFixedPrecision = 39;
>+
>+constexpr int kBufferLength = /*sign*/ 1 +
>+                              /*integer*/ kMaxFixedPrecision +
>+                              /*point*/ 1 +
>+                              /*fraction*/ kMaxFixedPrecision +
>+                              /*exponent e+123*/ 5;
>+
>+struct Buffer {
>+  void push_front(char c) {
>+    assert(begin > data);
>+    *--begin = c;
>+  }
>+  void push_back(char c) {
>+    assert(end < data + sizeof(data));
>+    *end++ = c;
>+  }
>+  void pop_back() {
>+    assert(begin < end);
>+    --end;
>+  }
>+
>+  char &back() {
>+    assert(begin < end);
>+    return end[-1];
>+  }
>+
>+  char last_digit() const { return end[-1] == '.' ? end[-2] : end[-1]; }
>+
>+  int size() const { return static_cast<int>(end - begin); }
>+
>+  char data[kBufferLength];
>+  char *begin;
>+  char *end;
>+};
>+
>+enum class FormatStyle { Fixed, Precision };
>+
>+// If the value is Inf or Nan, print it and return true.
>+// Otherwise, return false.
>+template <typename Float>
>+bool ConvertNonNumericFloats(char sign_char, Float v,
>+                             const ConversionSpec &conv, FormatSinkImpl *sink) {
>+  char text[4], *ptr = text;
>+  if (sign_char) *ptr++ = sign_char;
>+  if (std::isnan(v)) {
>+    ptr = std::copy_n(conv.conv().upper() ? "NAN" : "nan", 3, ptr);
>+  } else if (std::isinf(v)) {
>+    ptr = std::copy_n(conv.conv().upper() ? "INF" : "inf", 3, ptr);
>+  } else {
>+    return false;
>+  }
>+
>+  return sink->PutPaddedString(string_view(text, ptr - text), conv.width(), -1,
>+                               conv.flags().left);
>+}
>+
>+// Round up the last digit of the value.
>+// It will carry over and potentially overflow. 'exp' will be adjusted in that
>+// case.
>+template <FormatStyle mode>
>+void RoundUp(Buffer *buffer, int *exp) {
>+  char *p = &buffer->back();
>+  while (p >= buffer->begin && (*p == '9' || *p == '.')) {
>+    if (*p == '9') *p = '0';
>+    --p;
>+  }
>+
>+  if (p < buffer->begin) {
>+    *p = '1';
>+    buffer->begin = p;
>+    if (mode == FormatStyle::Precision) {
>+      std::swap(p[1], p[2]);  // move the .
>+      ++*exp;
>+      buffer->pop_back();
>+    }
>+  } else {
>+    ++*p;
>+  }
>+}
>+
>+void PrintExponent(int exp, char e, Buffer *out) {
>+  out->push_back(e);
>+  if (exp < 0) {
>+    out->push_back('-');
>+    exp = -exp;
>+  } else {
>+    out->push_back('+');
>+  }
>+  // Exponent digits.
>+  if (exp > 99) {
>+    out->push_back(exp / 100 + '0');
>+    out->push_back(exp / 10 % 10 + '0');
>+    out->push_back(exp % 10 + '0');
>+  } else {
>+    out->push_back(exp / 10 + '0');
>+    out->push_back(exp % 10 + '0');
>+  }
>+}
>+
>+template <typename Float, typename Int>
>+constexpr bool CanFitMantissa() {
>+  return
>+#if defined(__clang__) && !defined(__SSE3__)
>+      // Workaround for clang bug: https://bugs.llvm.org/show_bug.cgi?id=38289
>+      // Casting from long double to uint64_t is miscompiled and drops bits.
>+      (!std::is_same<Float, long double>::value ||
>+       !std::is_same<Int, uint64_t>::value) &&
>+#endif
>+      std::numeric_limits<Float>::digits <= std::numeric_limits<Int>::digits;
>+}
>+
>+template <typename Float>
>+struct Decomposed {
>+  Float mantissa;
>+  int exponent;
>+};
>+
>+// Decompose the double into an integer mantissa and an exponent.
>+template <typename Float>
>+Decomposed<Float> Decompose(Float v) {
>+  int exp;
>+  Float m = std::frexp(v, &exp);
>+  m = std::ldexp(m, std::numeric_limits<Float>::digits);
>+  exp -= std::numeric_limits<Float>::digits;
>+  return {m, exp};
>+}
>+
>+// Print 'digits' as decimal.
>+// In Fixed mode, we add a '.' at the end.
>+// In Precision mode, we add a '.' after the first digit.
>+template <FormatStyle mode, typename Int>
>+int PrintIntegralDigits(Int digits, Buffer *out) {
>+  int printed = 0;
>+  if (digits) {
>+    for (; digits; digits /= 10) out->push_front(digits % 10 + '0');
>+    printed = out->size();
>+    if (mode == FormatStyle::Precision) {
>+      out->push_front(*out->begin);
>+      out->begin[1] = '.';
>+    } else {
>+      out->push_back('.');
>+    }
>+  } else if (mode == FormatStyle::Fixed) {
>+    out->push_front('0');
>+    out->push_back('.');
>+    printed = 1;
>+  }
>+  return printed;
>+}
>+
>+// Back out 'extra_digits' digits and round up if necessary.
>+bool RemoveExtraPrecision(int extra_digits, bool has_leftover_value,
>+                          Buffer *out, int *exp_out) {
>+  if (extra_digits <= 0) return false;
>+
>+  // Back out the extra digits
>+  out->end -= extra_digits;
>+
>+  bool needs_to_round_up = [&] {
>+    // We look at the digit just past the end.
>+    // There must be 'extra_digits' extra valid digits after end.
>+    if (*out->end > '5') return true;
>+    if (*out->end < '5') return false;
>+    if (has_leftover_value || std::any_of(out->end + 1, out->end + extra_digits,
>+                                          [](char c) { return c != '0'; }))
>+      return true;
>+
>+    // Ends in ...50*, round to even.
>+    return out->last_digit() % 2 == 1;
>+  }();
>+
>+  if (needs_to_round_up) {
>+    RoundUp<FormatStyle::Precision>(out, exp_out);
>+  }
>+  return true;
>+}
>+
>+// Print the value into the buffer.
>+// This will not include the exponent, which will be returned in 'exp_out' for
>+// Precision mode.
>+template <typename Int, typename Float, FormatStyle mode>
>+bool FloatToBufferImpl(Int int_mantissa, int exp, int precision, Buffer *out,
>+                       int *exp_out) {
>+  assert((CanFitMantissa<Float, Int>()));
>+
>+  const int int_bits = std::numeric_limits<Int>::digits;
>+
>+  // In precision mode, we start printing one char to the right because it will
>+  // also include the '.'
>+  // In fixed mode we put the dot afterwards on the right.
>+  out->begin = out->end =
>+      out->data + 1 + kMaxFixedPrecision + (mode == FormatStyle::Precision);
>+
>+  if (exp >= 0) {
>+    if (std::numeric_limits<Float>::digits + exp > int_bits) {
>+      // The value will overflow the Int
>+      return false;
>+    }
>+    int digits_printed = PrintIntegralDigits<mode>(int_mantissa << exp, out);
>+    int digits_to_zero_pad = precision;
>+    if (mode == FormatStyle::Precision) {
>+      *exp_out = digits_printed - 1;
>+      digits_to_zero_pad -= digits_printed - 1;
>+      if (RemoveExtraPrecision(-digits_to_zero_pad, false, out, exp_out)) {
>+        return true;
>+      }
>+    }
>+    for (; digits_to_zero_pad-- > 0;) out->push_back('0');
>+    return true;
>+  }
>+
>+  exp = -exp;
>+  // We need at least 4 empty bits for the next decimal digit.
>+  // We will multiply by 10.
>+  if (exp > int_bits - 4) return false;
>+
>+  const Int mask = (Int{1} << exp) - 1;
>+
>+  // Print the integral part first.
>+  int digits_printed = PrintIntegralDigits<mode>(int_mantissa >> exp, out);
>+  int_mantissa &= mask;
>+
>+  int fractional_count = precision;
>+  if (mode == FormatStyle::Precision) {
>+    if (digits_printed == 0) {
>+      // Find the first non-zero digit, when in Precision mode.
>+      *exp_out = 0;
>+      if (int_mantissa) {
>+        while (int_mantissa <= mask) {
>+          int_mantissa *= 10;
>+          --*exp_out;
>+        }
>+      }
>+      out->push_front(static_cast<char>(int_mantissa >> exp) + '0');
>+      out->push_back('.');
>+      int_mantissa &= mask;
>+    } else {
>+      // We already have a digit, and a '.'
>+      *exp_out = digits_printed - 1;
>+      fractional_count -= *exp_out;
>+      if (RemoveExtraPrecision(-fractional_count, int_mantissa != 0, out,
>+                               exp_out)) {
>+        // If we had enough digits, return right away.
>+        // The code below will try to round again otherwise.
>+        return true;
>+      }
>+    }
>+  }
>+
>+  auto get_next_digit = [&] {
>+    int_mantissa *= 10;
>+    int digit = static_cast<int>(int_mantissa >> exp);
>+    int_mantissa &= mask;
>+    return digit;
>+  };
>+
>+  // Print fractional_count more digits, if available.
>+  for (; fractional_count > 0; --fractional_count) {
>+    out->push_back(get_next_digit() + '0');
>+  }
>+
>+  int next_digit = get_next_digit();
>+  if (next_digit > 5 ||
>+      (next_digit == 5 && (int_mantissa || out->last_digit() % 2 == 1))) {
>+    RoundUp<mode>(out, exp_out);
>+  }
>+
>+  return true;
>+}
>+
>+template <FormatStyle mode, typename Float>
>+bool FloatToBuffer(Decomposed<Float> decomposed, int precision, Buffer *out,
>+                   int *exp) {
>+  if (precision > kMaxFixedPrecision) return false;
>+
>+  // Try with uint64_t.
>+  if (CanFitMantissa<Float, std::uint64_t>() &&
>+      FloatToBufferImpl<std::uint64_t, Float, mode>(
>+          static_cast<std::uint64_t>(decomposed.mantissa),
>+          static_cast<std::uint64_t>(decomposed.exponent), precision, out, exp))
>+    return true;
>+
>+#if defined(__SIZEOF_INT128__)
>+  // If that is not enough, try with __uint128_t.
>+  return CanFitMantissa<Float, __uint128_t>() &&
>+         FloatToBufferImpl<__uint128_t, Float, mode>(
>+             static_cast<__uint128_t>(decomposed.mantissa),
>+             static_cast<__uint128_t>(decomposed.exponent), precision, out,
>+             exp);
>+#endif
>+  return false;
>+}
>+
>+void WriteBufferToSink(char sign_char, string_view str,
>+                       const ConversionSpec &conv, FormatSinkImpl *sink) {
>+  int left_spaces = 0, zeros = 0, right_spaces = 0;
>+  int missing_chars =
>+      conv.width() >= 0 ? std::max(conv.width() - static_cast<int>(str.size()) -
>+                                       static_cast<int>(sign_char != 0),
>+                                   0)
>+                        : 0;
>+  if (conv.flags().left) {
>+    right_spaces = missing_chars;
>+  } else if (conv.flags().zero) {
>+    zeros = missing_chars;
>+  } else {
>+    left_spaces = missing_chars;
>+  }
>+
>+  sink->Append(left_spaces, ' ');
>+  if (sign_char) sink->Append(1, sign_char);
>+  sink->Append(zeros, '0');
>+  sink->Append(str);
>+  sink->Append(right_spaces, ' ');
>+}
>+
>+template <typename Float>
>+bool FloatToSink(const Float v, const ConversionSpec &conv,
>+                 FormatSinkImpl *sink) {
>+  // Print the sign or the sign column.
>+  Float abs_v = v;
>+  char sign_char = 0;
>+  if (std::signbit(abs_v)) {
>+    sign_char = '-';
>+    abs_v = -abs_v;
>+  } else if (conv.flags().show_pos) {
>+    sign_char = '+';
>+  } else if (conv.flags().sign_col) {
>+    sign_char = ' ';
>+  }
>+
>+  // Print nan/inf.
>+  if (ConvertNonNumericFloats(sign_char, abs_v, conv, sink)) {
>+    return true;
>+  }
>+
>+  int precision = conv.precision() < 0 ? 6 : conv.precision();
>+
>+  int exp = 0;
>+
>+  auto decomposed = Decompose(abs_v);
>+
>+  Buffer buffer;
>+
>+  switch (conv.conv().id()) {
>+    case ConversionChar::f:
>+    case ConversionChar::F:
>+      if (!FloatToBuffer<FormatStyle::Fixed>(decomposed, precision, &buffer,
>+                                             nullptr)) {
>+        return FallbackToSnprintf(v, conv, sink);
>+      }
>+      if (!conv.flags().alt && buffer.back() == '.') buffer.pop_back();
>+      break;
>+
>+    case ConversionChar::e:
>+    case ConversionChar::E:
>+      if (!FloatToBuffer<FormatStyle::Precision>(decomposed, precision, &buffer,
>+                                                 &exp)) {
>+        return FallbackToSnprintf(v, conv, sink);
>+      }
>+      if (!conv.flags().alt && buffer.back() == '.') buffer.pop_back();
>+      PrintExponent(exp, conv.conv().upper() ? 'E' : 'e', &buffer);
>+      break;
>+
>+    case ConversionChar::g:
>+    case ConversionChar::G:
>+      precision = std::max(0, precision - 1);
>+      if (!FloatToBuffer<FormatStyle::Precision>(decomposed, precision, &buffer,
>+                                                 &exp)) {
>+        return FallbackToSnprintf(v, conv, sink);
>+      }
>+      if (precision + 1 > exp && exp >= -4) {
>+        if (exp < 0) {
>+          // Have 1.23456, needs 0.00123456
>+          // Move the first digit
>+          buffer.begin[1] = *buffer.begin;
>+          // Add some zeros
>+          for (; exp < -1; ++exp) *buffer.begin-- = '0';
>+          *buffer.begin-- = '.';
>+          *buffer.begin = '0';
>+        } else if (exp > 0) {
>+          // Have 1.23456, needs 1234.56
>+          // Move the '.' exp positions to the right.
>+          std::rotate(buffer.begin + 1, buffer.begin + 2,
>+                      buffer.begin + exp + 2);
>+        }
>+        exp = 0;
>+      }
>+      if (!conv.flags().alt) {
>+        while (buffer.back() == '0') buffer.pop_back();
>+        if (buffer.back() == '.') buffer.pop_back();
>+      }
>+      if (exp) PrintExponent(exp, conv.conv().upper() ? 'E' : 'e', &buffer);
>+      break;
>+
>+    case ConversionChar::a:
>+    case ConversionChar::A:
>+      return FallbackToSnprintf(v, conv, sink);
>+
>+    default:
>+      return false;
>+  }
>+
>+  WriteBufferToSink(sign_char,
>+                    string_view(buffer.begin, buffer.end - buffer.begin), conv,
>+                    sink);
>+
>+  return true;
>+}
>+
>+}  // namespace
>+
>+bool ConvertFloatImpl(long double v, const ConversionSpec &conv,
>+                      FormatSinkImpl *sink) {
>+  return FloatToSink(v, conv, sink);
>+}
>+
>+bool ConvertFloatImpl(float v, const ConversionSpec &conv,
>+                      FormatSinkImpl *sink) {
>+  return FloatToSink(v, conv, sink);
>+}
>+
>+bool ConvertFloatImpl(double v, const ConversionSpec &conv,
>+                      FormatSinkImpl *sink) {
>+  return FloatToSink(v, conv, sink);
>+}
>+
>+}  // namespace str_format_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/float_conversion.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/float_conversion.h
>new file mode 100644
>index 00000000000..8ba5566d3ee
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/float_conversion.h
>@@ -0,0 +1,21 @@
>+#ifndef ABSL_STRINGS_INTERNAL_STR_FORMAT_FLOAT_CONVERSION_H_
>+#define ABSL_STRINGS_INTERNAL_STR_FORMAT_FLOAT_CONVERSION_H_
>+
>+#include "absl/strings/internal/str_format/extension.h"
>+
>+namespace absl {
>+namespace str_format_internal {
>+
>+bool ConvertFloatImpl(float v, const ConversionSpec &conv,
>+                      FormatSinkImpl *sink);
>+
>+bool ConvertFloatImpl(double v, const ConversionSpec &conv,
>+                      FormatSinkImpl *sink);
>+
>+bool ConvertFloatImpl(long double v, const ConversionSpec &conv,
>+                      FormatSinkImpl *sink);
>+
>+}  // namespace str_format_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_STRINGS_INTERNAL_STR_FORMAT_FLOAT_CONVERSION_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/output.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/output.cc
>new file mode 100644
>index 00000000000..5c3795b737c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/output.cc
>@@ -0,0 +1,47 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/strings/internal/str_format/output.h"
>+
>+#include <errno.h>
>+#include <cstring>
>+
>+namespace absl {
>+namespace str_format_internal {
>+
>+void BufferRawSink::Write(string_view v) {
>+  size_t to_write = std::min(v.size(), size_);
>+  std::memcpy(buffer_, v.data(), to_write);
>+  buffer_ += to_write;
>+  size_ -= to_write;
>+  total_written_ += v.size();
>+}
>+
>+void FILERawSink::Write(string_view v) {
>+  while (!v.empty() && !error_) {
>+    if (size_t result = std::fwrite(v.data(), 1, v.size(), output_)) {
>+      // Some progress was made.
>+      count_ += result;
>+      v.remove_prefix(result);
>+    } else {
>+      // Some error occurred.
>+      if (errno != EINTR) {
>+        error_ = errno;
>+      }
>+    }
>+  }
>+}
>+
>+}  // namespace str_format_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/output.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/output.h
>new file mode 100644
>index 00000000000..3b0aa5e7157
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/output.h
>@@ -0,0 +1,101 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// Output extension hooks for the Format library.
>+// `internal::InvokeFlush` calls the appropriate flush function for the
>+// specified output argument.
>+// `BufferRawSink` is a simple output sink for a char buffer. Used by SnprintF.
>+// `FILERawSink` is a std::FILE* based sink. Used by PrintF and FprintF.
>+
>+#ifndef ABSL_STRINGS_INTERNAL_STR_FORMAT_OUTPUT_H_
>+#define ABSL_STRINGS_INTERNAL_STR_FORMAT_OUTPUT_H_
>+
>+#include <cstdio>
>+#include <ostream>
>+#include <string>
>+
>+#include "absl/base/port.h"
>+#include "absl/strings/string_view.h"
>+
>+class Cord;
>+
>+namespace absl {
>+namespace str_format_internal {
>+
>+// RawSink implementation that writes into a char* buffer.
>+// It will not overflow the buffer, but will keep the total count of chars
>+// that would have been written.
>+class BufferRawSink {
>+ public:
>+  BufferRawSink(char* buffer, size_t size) : buffer_(buffer), size_(size) {}
>+
>+  size_t total_written() const { return total_written_; }
>+  void Write(string_view v);
>+
>+ private:
>+  char* buffer_;
>+  size_t size_;
>+  size_t total_written_ = 0;
>+};
>+
>+// RawSink implementation that writes into a FILE*.
>+// It keeps track of the total number of bytes written and any error encountered
>+// during the writes.
>+class FILERawSink {
>+ public:
>+  explicit FILERawSink(std::FILE* output) : output_(output) {}
>+
>+  void Write(string_view v);
>+
>+  size_t count() const { return count_; }
>+  int error() const { return error_; }
>+
>+ private:
>+  std::FILE* output_;
>+  int error_ = 0;
>+  size_t count_ = 0;
>+};
>+
>+// Provide RawSink integration with common types from the STL.
>+inline void AbslFormatFlush(std::string* out, string_view s) {
>+  out->append(s.begin(), s.size());
>+}
>+inline void AbslFormatFlush(std::ostream* out, string_view s) {
>+  out->write(s.begin(), s.size());
>+}
>+
>+template <class AbslCord, typename = typename std::enable_if<
>+                              std::is_same<AbslCord, ::Cord>::value>::type>
>+inline void AbslFormatFlush(AbslCord* out, string_view s) {
>+  out->Append(s);
>+}
>+
>+inline void AbslFormatFlush(FILERawSink* sink, string_view v) {
>+  sink->Write(v);
>+}
>+
>+inline void AbslFormatFlush(BufferRawSink* sink, string_view v) {
>+  sink->Write(v);
>+}
>+
>+template <typename T>
>+auto InvokeFlush(T* out, string_view s)
>+    -> decltype(str_format_internal::AbslFormatFlush(out, s)) {
>+  str_format_internal::AbslFormatFlush(out, s);
>+}
>+
>+}  // namespace str_format_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_STRINGS_INTERNAL_STR_FORMAT_OUTPUT_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/output_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/output_test.cc
>new file mode 100644
>index 00000000000..cc3c615557f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/output_test.cc
>@@ -0,0 +1,78 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/strings/internal/str_format/output.h"
>+
>+#include <sstream>
>+#include <string>
>+
>+
>+#include "gmock/gmock.h"
>+#include "gtest/gtest.h"
>+
>+namespace absl {
>+namespace {
>+
>+TEST(InvokeFlush, String) {
>+  std::string str = "ABC";
>+  str_format_internal::InvokeFlush(&str, "DEF");
>+  EXPECT_EQ(str, "ABCDEF");
>+
>+#if UTIL_FORMAT_HAS_GLOBAL_STRING
>+  std::string str2 = "ABC";
>+  str_format_internal::InvokeFlush(&str2, "DEF");
>+  EXPECT_EQ(str2, "ABCDEF");
>+#endif  // UTIL_FORMAT_HAS_GLOBAL_STRING
>+}
>+
>+TEST(InvokeFlush, Stream) {
>+  std::stringstream str;
>+  str << "ABC";
>+  str_format_internal::InvokeFlush(&str, "DEF");
>+  EXPECT_EQ(str.str(), "ABCDEF");
>+}
>+
>+TEST(BufferRawSink, Limits) {
>+  char buf[16];
>+  {
>+    std::fill(std::begin(buf), std::end(buf), 'x');
>+    str_format_internal::BufferRawSink bufsink(buf, sizeof(buf) - 1);
>+    str_format_internal::InvokeFlush(&bufsink, "Hello World237");
>+    EXPECT_EQ(std::string(buf, sizeof(buf)), "Hello World237xx");
>+  }
>+  {
>+    std::fill(std::begin(buf), std::end(buf), 'x');
>+    str_format_internal::BufferRawSink bufsink(buf, sizeof(buf) - 1);
>+    str_format_internal::InvokeFlush(&bufsink, "Hello World237237");
>+    EXPECT_EQ(std::string(buf, sizeof(buf)), "Hello World2372x");
>+  }
>+  {
>+    std::fill(std::begin(buf), std::end(buf), 'x');
>+    str_format_internal::BufferRawSink bufsink(buf, sizeof(buf) - 1);
>+    str_format_internal::InvokeFlush(&bufsink, "Hello World");
>+    str_format_internal::InvokeFlush(&bufsink, "237");
>+    EXPECT_EQ(std::string(buf, sizeof(buf)), "Hello World237xx");
>+  }
>+  {
>+    std::fill(std::begin(buf), std::end(buf), 'x');
>+    str_format_internal::BufferRawSink bufsink(buf, sizeof(buf) - 1);
>+    str_format_internal::InvokeFlush(&bufsink, "Hello World");
>+    str_format_internal::InvokeFlush(&bufsink, "237237");
>+    EXPECT_EQ(std::string(buf, sizeof(buf)), "Hello World2372x");
>+  }
>+}
>+
>+}  // namespace
>+}  // namespace absl
>+
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/parser.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/parser.cc
>new file mode 100644
>index 00000000000..10114f489c0
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/parser.cc
>@@ -0,0 +1,294 @@
>+#include "absl/strings/internal/str_format/parser.h"
>+
>+#include <assert.h>
>+#include <string.h>
>+#include <wchar.h>
>+#include <cctype>
>+#include <cstdint>
>+
>+#include <algorithm>
>+#include <initializer_list>
>+#include <limits>
>+#include <ostream>
>+#include <string>
>+#include <unordered_set>
>+
>+namespace absl {
>+namespace str_format_internal {
>+namespace {
>+
>+bool CheckFastPathSetting(const UnboundConversion& conv) {
>+  bool should_be_basic = !conv.flags.left &&      //
>+                         !conv.flags.show_pos &&  //
>+                         !conv.flags.sign_col &&  //
>+                         !conv.flags.alt &&       //
>+                         !conv.flags.zero &&      //
>+                         (conv.width.value() == -1) &&
>+                         (conv.precision.value() == -1);
>+  if (should_be_basic != conv.flags.basic) {
>+    fprintf(stderr,
>+            "basic=%d left=%d show_pos=%d sign_col=%d alt=%d zero=%d "
>+            "width=%d precision=%d\n",
>+            conv.flags.basic, conv.flags.left, conv.flags.show_pos,
>+            conv.flags.sign_col, conv.flags.alt, conv.flags.zero,
>+            conv.width.value(), conv.precision.value());
>+  }
>+  return should_be_basic == conv.flags.basic;
>+}
>+
>+// Keep a single table for all the conversion chars and length modifiers.
>+// We invert the length modifiers to make them negative so that we can easily
>+// test for them.
>+// Everything else is `none`, which is a negative constant.
>+using CC = ConversionChar::Id;
>+using LM = LengthMod::Id;
>+static constexpr std::int8_t none = -128;
>+static constexpr std::int8_t kIds[] = {
>+    none,   none,   none,   none,  none,   none,  none,  none,   // 00-07
>+    none,   none,   none,   none,  none,   none,  none,  none,   // 08-0f
>+    none,   none,   none,   none,  none,   none,  none,  none,   // 10-17
>+    none,   none,   none,   none,  none,   none,  none,  none,   // 18-1f
>+    none,   none,   none,   none,  none,   none,  none,  none,   // 20-27
>+    none,   none,   none,   none,  none,   none,  none,  none,   // 28-2f
>+    none,   none,   none,   none,  none,   none,  none,  none,   // 30-37
>+    none,   none,   none,   none,  none,   none,  none,  none,   // 38-3f
>+    none,   CC::A,  none,   CC::C, none,   CC::E, CC::F, CC::G,  // @ABCDEFG
>+    none,   none,   none,   none,  ~LM::L, none,  none,  none,   // HIJKLMNO
>+    none,   none,   none,   CC::S, none,   none,  none,  none,   // PQRSTUVW
>+    CC::X,  none,   none,   none,  none,   none,  none,  none,   // XYZ[\]^_
>+    none,   CC::a,  none,   CC::c, CC::d,  CC::e, CC::f, CC::g,  // `abcdefg
>+    ~LM::h, CC::i,  ~LM::j, none,  ~LM::l, none,  CC::n, CC::o,  // hijklmno
>+    CC::p,  ~LM::q, none,   CC::s, ~LM::t, CC::u, none,  none,   // pqrstuvw
>+    CC::x,  none,   ~LM::z, none,  none,   none,  none,  none,   // xyz{|}~!
>+    none,   none,   none,   none,  none,   none,  none,  none,   // 80-87
>+    none,   none,   none,   none,  none,   none,  none,  none,   // 88-8f
>+    none,   none,   none,   none,  none,   none,  none,  none,   // 90-97
>+    none,   none,   none,   none,  none,   none,  none,  none,   // 98-9f
>+    none,   none,   none,   none,  none,   none,  none,  none,   // a0-a7
>+    none,   none,   none,   none,  none,   none,  none,  none,   // a8-af
>+    none,   none,   none,   none,  none,   none,  none,  none,   // b0-b7
>+    none,   none,   none,   none,  none,   none,  none,  none,   // b8-bf
>+    none,   none,   none,   none,  none,   none,  none,  none,   // c0-c7
>+    none,   none,   none,   none,  none,   none,  none,  none,   // c8-cf
>+    none,   none,   none,   none,  none,   none,  none,  none,   // d0-d7
>+    none,   none,   none,   none,  none,   none,  none,  none,   // d8-df
>+    none,   none,   none,   none,  none,   none,  none,  none,   // e0-e7
>+    none,   none,   none,   none,  none,   none,  none,  none,   // e8-ef
>+    none,   none,   none,   none,  none,   none,  none,  none,   // f0-f7
>+    none,   none,   none,   none,  none,   none,  none,  none,   // f8-ff
>+};
>+
>+template <bool is_positional>
>+bool ConsumeConversion(string_view *src, UnboundConversion *conv,
>+                       int *next_arg) {
>+  const char *pos = src->begin();
>+  const char *const end = src->end();
>+  char c;
>+  // Read the next char into `c` and update `pos`. Reads '\0' if at end.
>+  const auto get_char = [&] { c = pos == end ? '\0' : *pos++; };
>+
>+  const auto parse_digits = [&] {
>+    int digits = c - '0';
>+    // We do not want to overflow `digits` so we consume at most digits10-1
>+    // digits. If there are more digits the parsing will fail later on when the
>+    // digit doesn't match the expected characters.
>+    int num_digits = std::numeric_limits<int>::digits10 - 2;
>+    for (get_char(); num_digits && std::isdigit(c); get_char()) {
>+      --num_digits;
>+      digits = 10 * digits + c - '0';
>+    }
>+    return digits;
>+  };
>+
>+  if (is_positional) {
>+    get_char();
>+    if (c < '1' || c > '9') return false;
>+    conv->arg_position = parse_digits();
>+    assert(conv->arg_position > 0);
>+    if (c != '$') return false;
>+  }
>+
>+  get_char();
>+
>+  // We should start with the basic flag on.
>+  assert(conv->flags.basic);
>+
>+  // Any non alpha character makes this conversion not basic.
>+  // This includes flags (-+ #0), width (1-9, *) or precision (.).
>+  // All conversion characters and length modifiers are alpha characters.
>+  if (c < 'A') {
>+    conv->flags.basic = false;
>+
>+    for (; c <= '0'; get_char()) {
>+      switch (c) {
>+        case '-':
>+          conv->flags.left = true;
>+          continue;
>+        case '+':
>+          conv->flags.show_pos = true;
>+          continue;
>+        case ' ':
>+          conv->flags.sign_col = true;
>+          continue;
>+        case '#':
>+          conv->flags.alt = true;
>+          continue;
>+        case '0':
>+          conv->flags.zero = true;
>+          continue;
>+      }
>+      break;
>+    }
>+
>+    if (c <= '9') {
>+      if (c >= '0') {
>+        int maybe_width = parse_digits();
>+        if (!is_positional && c == '$') {
>+          if (*next_arg != 0) return false;
>+          // Positional conversion.
>+          *next_arg = -1;
>+          conv->flags = Flags();
>+          conv->flags.basic = true;
>+          return ConsumeConversion<true>(src, conv, next_arg);
>+        }
>+        conv->width.set_value(maybe_width);
>+      } else if (c == '*') {
>+        get_char();
>+        if (is_positional) {
>+          if (c < '1' || c > '9') return false;
>+          conv->width.set_from_arg(parse_digits());
>+          if (c != '$') return false;
>+          get_char();
>+        } else {
>+          conv->width.set_from_arg(++*next_arg);
>+        }
>+      }
>+    }
>+
>+    if (c == '.') {
>+      get_char();
>+      if (std::isdigit(c)) {
>+        conv->precision.set_value(parse_digits());
>+      } else if (c == '*') {
>+        get_char();
>+        if (is_positional) {
>+          if (c < '1' || c > '9') return false;
>+          conv->precision.set_from_arg(parse_digits());
>+          if (c != '$') return false;
>+          get_char();
>+        } else {
>+          conv->precision.set_from_arg(++*next_arg);
>+        }
>+      } else {
>+        conv->precision.set_value(0);
>+      }
>+    }
>+  }
>+
>+  std::int8_t id = kIds[static_cast<unsigned char>(c)];
>+
>+  if (id < 0) {
>+    if (id == none) return false;
>+
>+    // It is a length modifier.
>+    using str_format_internal::LengthMod;
>+    LengthMod length_mod = LengthMod::FromId(static_cast<LM>(~id));
>+    get_char();
>+    if (c == 'h' && length_mod.id() == LengthMod::h) {
>+      conv->length_mod = LengthMod::FromId(LengthMod::hh);
>+      get_char();
>+    } else if (c == 'l' && length_mod.id() == LengthMod::l) {
>+      conv->length_mod = LengthMod::FromId(LengthMod::ll);
>+      get_char();
>+    } else {
>+      conv->length_mod = length_mod;
>+    }
>+    id = kIds[static_cast<unsigned char>(c)];
>+    if (id < 0) return false;
>+  }
>+
>+  assert(CheckFastPathSetting(*conv));
>+  (void)(&CheckFastPathSetting);
>+
>+  conv->conv = ConversionChar::FromId(static_cast<CC>(id));
>+  if (!is_positional) conv->arg_position = ++*next_arg;
>+  *src = string_view(pos, end - pos);
>+  return true;
>+}
>+
>+}  // namespace
>+
>+bool ConsumeUnboundConversion(string_view *src, UnboundConversion *conv,
>+                              int *next_arg) {
>+  if (*next_arg < 0) return ConsumeConversion<true>(src, conv, next_arg);
>+  return ConsumeConversion<false>(src, conv, next_arg);
>+}
>+
>+struct ParsedFormatBase::ParsedFormatConsumer {
>+  explicit ParsedFormatConsumer(ParsedFormatBase *parsedformat)
>+      : parsed(parsedformat), data_pos(parsedformat->data_.get()) {}
>+
>+  bool Append(string_view s) {
>+    if (s.empty()) return true;
>+
>+    size_t text_end = AppendText(s);
>+
>+    if (!parsed->items_.empty() && !parsed->items_.back().is_conversion) {
>+      // Let's extend the existing text run.
>+      parsed->items_.back().text_end = text_end;
>+    } else {
>+      // Let's make a new text run.
>+      parsed->items_.push_back({false, text_end, {}});
>+    }
>+    return true;
>+  }
>+
>+  bool ConvertOne(const UnboundConversion &conv, string_view s) {
>+    size_t text_end = AppendText(s);
>+    parsed->items_.push_back({true, text_end, conv});
>+    return true;
>+  }
>+
>+  size_t AppendText(string_view s) {
>+    memcpy(data_pos, s.data(), s.size());
>+    data_pos += s.size();
>+    return static_cast<size_t>(data_pos - parsed->data_.get());
>+  }
>+
>+  ParsedFormatBase *parsed;
>+  char* data_pos;
>+};
>+
>+ParsedFormatBase::ParsedFormatBase(string_view format, bool allow_ignored,
>+                                   std::initializer_list<Conv> convs)
>+    : data_(format.empty() ? nullptr : new char[format.size()]) {
>+  has_error_ = !ParseFormatString(format, ParsedFormatConsumer(this)) ||
>+               !MatchesConversions(allow_ignored, convs);
>+}
>+
>+bool ParsedFormatBase::MatchesConversions(
>+    bool allow_ignored, std::initializer_list<Conv> convs) const {
>+  std::unordered_set<int> used;
>+  auto add_if_valid_conv = [&](int pos, char c) {
>+      if (static_cast<size_t>(pos) > convs.size() ||
>+          !Contains(convs.begin()[pos - 1], c))
>+        return false;
>+      used.insert(pos);
>+      return true;
>+  };
>+  for (const ConversionItem &item : items_) {
>+    if (!item.is_conversion) continue;
>+    auto &conv = item.conv;
>+    if (conv.precision.is_from_arg() &&
>+        !add_if_valid_conv(conv.precision.get_from_arg(), '*'))
>+      return false;
>+    if (conv.width.is_from_arg() &&
>+        !add_if_valid_conv(conv.width.get_from_arg(), '*'))
>+      return false;
>+    if (!add_if_valid_conv(conv.arg_position, conv.conv.Char())) return false;
>+  }
>+  return used.size() == convs.size() || allow_ignored;
>+}
>+
>+}  // namespace str_format_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/parser.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/parser.h
>new file mode 100644
>index 00000000000..5bebc95540e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/parser.h
>@@ -0,0 +1,291 @@
>+#ifndef ABSL_STRINGS_INTERNAL_STR_FORMAT_PARSER_H_
>+#define ABSL_STRINGS_INTERNAL_STR_FORMAT_PARSER_H_
>+
>+#include <limits.h>
>+#include <stddef.h>
>+#include <stdlib.h>
>+
>+#include <cassert>
>+#include <initializer_list>
>+#include <iosfwd>
>+#include <iterator>
>+#include <memory>
>+#include <vector>
>+
>+#include "absl/strings/internal/str_format/checker.h"
>+#include "absl/strings/internal/str_format/extension.h"
>+
>+namespace absl {
>+namespace str_format_internal {
>+
>+// The analyzed properties of a single specified conversion.
>+struct UnboundConversion {
>+  UnboundConversion()
>+      : flags() /* This is required to zero all the fields of flags. */ {
>+    flags.basic = true;
>+  }
>+
>+  class InputValue {
>+   public:
>+    void set_value(int value) {
>+      assert(value >= 0);
>+      value_ = value;
>+    }
>+    int value() const { return value_; }
>+
>+    // Marks the value as "from arg". aka the '*' format.
>+    // Requires `value >= 1`.
>+    // When set, is_from_arg() return true and get_from_arg() returns the
>+    // original value.
>+    // `value()`'s return value is unspecfied in this state.
>+    void set_from_arg(int value) {
>+      assert(value > 0);
>+      value_ = -value - 1;
>+    }
>+    bool is_from_arg() const { return value_ < -1; }
>+    int get_from_arg() const {
>+      assert(is_from_arg());
>+      return -value_ - 1;
>+    }
>+
>+   private:
>+    int value_ = -1;
>+  };
>+
>+  // No need to initialize. It will always be set in the parser.
>+  int arg_position;
>+
>+  InputValue width;
>+  InputValue precision;
>+
>+  Flags flags;
>+  LengthMod length_mod;
>+  ConversionChar conv;
>+};
>+
>+// Consume conversion spec prefix (not including '%') of '*src' if valid.
>+// Examples of valid specs would be e.g.: "s", "d", "-12.6f".
>+// If valid, the front of src is advanced such that src becomes the
>+// part following the conversion spec, and the spec part is broken down and
>+// returned in 'conv'.
>+// If invalid, returns false and leaves 'src' unmodified.
>+// For example:
>+//   Given "d9", returns "d", and leaves src="9",
>+//   Given "!", returns "" and leaves src="!".
>+bool ConsumeUnboundConversion(string_view* src, UnboundConversion* conv,
>+                              int* next_arg);
>+
>+// Parse the format std::string provided in 'src' and pass the identified items into
>+// 'consumer'.
>+// Text runs will be passed by calling
>+//   Consumer::Append(string_view);
>+// ConversionItems will be passed by calling
>+//   Consumer::ConvertOne(UnboundConversion, string_view);
>+// In the case of ConvertOne, the string_view that is passed is the
>+// portion of the format std::string corresponding to the conversion, not including
>+// the leading %. On success, it returns true. On failure, it stops and returns
>+// false.
>+template <typename Consumer>
>+bool ParseFormatString(string_view src, Consumer consumer) {
>+  int next_arg = 0;
>+  while (!src.empty()) {
>+    const char* percent =
>+        static_cast<const char*>(memchr(src.begin(), '%', src.size()));
>+    if (!percent) {
>+      // We found the last substring.
>+      return consumer.Append(src);
>+    }
>+    // We found a percent, so push the text run then process the percent.
>+    size_t percent_loc = percent - src.data();
>+    if (!consumer.Append(string_view(src.data(), percent_loc))) return false;
>+    if (percent + 1 >= src.end()) return false;
>+
>+    UnboundConversion conv;
>+
>+    switch (percent[1]) {
>+      case '%':
>+        if (!consumer.Append("%")) return false;
>+        src.remove_prefix(percent_loc + 2);
>+        continue;
>+
>+#define PARSER_CASE(ch)                                     \
>+  case #ch[0]:                                              \
>+    src.remove_prefix(percent_loc + 2);                     \
>+    conv.conv = ConversionChar::FromId(ConversionChar::ch); \
>+    conv.arg_position = ++next_arg;                         \
>+    break;
>+        ABSL_CONVERSION_CHARS_EXPAND_(PARSER_CASE, );
>+#undef PARSER_CASE
>+
>+      default:
>+        src.remove_prefix(percent_loc + 1);
>+        if (!ConsumeUnboundConversion(&src, &conv, &next_arg)) return false;
>+        break;
>+    }
>+    if (next_arg == 0) {
>+      // This indicates an error in the format std::string.
>+      // The only way to get next_arg == 0 is to have a positional argument
>+      // first which sets next_arg to -1 and then a non-positional argument
>+      // which does ++next_arg.
>+      // Checking here seems to be the cheapeast place to do it.
>+      return false;
>+    }
>+    if (!consumer.ConvertOne(
>+            conv, string_view(percent + 1, src.data() - (percent + 1)))) {
>+      return false;
>+    }
>+  }
>+  return true;
>+}
>+
>+// Always returns true, or fails to compile in a constexpr context if s does not
>+// point to a constexpr char array.
>+constexpr bool EnsureConstexpr(string_view s) {
>+  return s.empty() || s[0] == s[0];
>+}
>+
>+class ParsedFormatBase {
>+ public:
>+  explicit ParsedFormatBase(string_view format, bool allow_ignored,
>+                            std::initializer_list<Conv> convs);
>+
>+  ParsedFormatBase(const ParsedFormatBase& other) { *this = other; }
>+
>+  ParsedFormatBase(ParsedFormatBase&& other) { *this = std::move(other); }
>+
>+  ParsedFormatBase& operator=(const ParsedFormatBase& other) {
>+    if (this == &other) return *this;
>+    has_error_ = other.has_error_;
>+    items_ = other.items_;
>+    size_t text_size = items_.empty() ? 0 : items_.back().text_end;
>+    data_.reset(new char[text_size]);
>+    memcpy(data_.get(), other.data_.get(), text_size);
>+    return *this;
>+  }
>+
>+  ParsedFormatBase& operator=(ParsedFormatBase&& other) {
>+    if (this == &other) return *this;
>+    has_error_ = other.has_error_;
>+    data_ = std::move(other.data_);
>+    items_ = std::move(other.items_);
>+    // Reset the vector to make sure the invariants hold.
>+    other.items_.clear();
>+    return *this;
>+  }
>+
>+  template <typename Consumer>
>+  bool ProcessFormat(Consumer consumer) const {
>+    const char* const base = data_.get();
>+    string_view text(base, 0);
>+    for (const auto& item : items_) {
>+      text = string_view(text.end(), (base + item.text_end) - text.end());
>+      if (item.is_conversion) {
>+        if (!consumer.ConvertOne(item.conv, text)) return false;
>+      } else {
>+        if (!consumer.Append(text)) return false;
>+      }
>+    }
>+    return !has_error_;
>+  }
>+
>+  bool has_error() const { return has_error_; }
>+
>+ private:
>+  // Returns whether the conversions match and if !allow_ignored it verifies
>+  // that all conversions are used by the format.
>+  bool MatchesConversions(bool allow_ignored,
>+                          std::initializer_list<Conv> convs) const;
>+
>+  struct ParsedFormatConsumer;
>+
>+  struct ConversionItem {
>+    bool is_conversion;
>+    // Points to the past-the-end location of this element in the data_ array.
>+    size_t text_end;
>+    UnboundConversion conv;
>+  };
>+
>+  bool has_error_;
>+  std::unique_ptr<char[]> data_;
>+  std::vector<ConversionItem> items_;
>+};
>+
>+
>+// A value type representing a preparsed format.  These can be created, copied
>+// around, and reused to speed up formatting loops.
>+// The user must specify through the template arguments the conversion
>+// characters used in the format. This will be checked at compile time.
>+//
>+// This class uses Conv enum values to specify each argument.
>+// This allows for more flexibility as you can specify multiple possible
>+// conversion characters for each argument.
>+// ParsedFormat<char...> is a simplified alias for when the user only
>+// needs to specify a single conversion character for each argument.
>+//
>+// Example:
>+//   // Extended format supports multiple characters per argument:
>+//   using MyFormat = ExtendedParsedFormat<Conv::d | Conv::x>;
>+//   MyFormat GetFormat(bool use_hex) {
>+//     if (use_hex) return MyFormat("foo %x bar");
>+//     return MyFormat("foo %d bar");
>+//   }
>+//   // 'format' can be used with any value that supports 'd' and 'x',
>+//   // like `int`.
>+//   auto format = GetFormat(use_hex);
>+//   value = StringF(format, i);
>+//
>+// This class also supports runtime format checking with the ::New() and
>+// ::NewAllowIgnored() factory functions.
>+// This is the only API that allows the user to pass a runtime specified format
>+// std::string. These factory functions will return NULL if the format does not match
>+// the conversions requested by the user.
>+template <str_format_internal::Conv... C>
>+class ExtendedParsedFormat : public str_format_internal::ParsedFormatBase {
>+ public:
>+  explicit ExtendedParsedFormat(string_view format)
>+#if ABSL_INTERNAL_ENABLE_FORMAT_CHECKER
>+      __attribute__((
>+          enable_if(str_format_internal::EnsureConstexpr(format),
>+                    "Format std::string is not constexpr."),
>+          enable_if(str_format_internal::ValidFormatImpl<C...>(format),
>+                    "Format specified does not match the template arguments.")))
>+#endif  // ABSL_INTERNAL_ENABLE_FORMAT_CHECKER
>+      : ExtendedParsedFormat(format, false) {
>+  }
>+
>+  // ExtendedParsedFormat factory function.
>+  // The user still has to specify the conversion characters, but they will not
>+  // be checked at compile time. Instead, it will be checked at runtime.
>+  // This delays the checking to runtime, but allows the user to pass
>+  // dynamically sourced formats.
>+  // It returns NULL if the format does not match the conversion characters.
>+  // The user is responsible for checking the return value before using it.
>+  //
>+  // The 'New' variant will check that all the specified arguments are being
>+  // consumed by the format and return NULL if any argument is being ignored.
>+  // The 'NewAllowIgnored' variant will not verify this and will allow formats
>+  // that ignore arguments.
>+  static std::unique_ptr<ExtendedParsedFormat> New(string_view format) {
>+    return New(format, false);
>+  }
>+  static std::unique_ptr<ExtendedParsedFormat> NewAllowIgnored(
>+      string_view format) {
>+    return New(format, true);
>+  }
>+
>+ private:
>+  static std::unique_ptr<ExtendedParsedFormat> New(string_view format,
>+                                                   bool allow_ignored) {
>+    std::unique_ptr<ExtendedParsedFormat> conv(
>+        new ExtendedParsedFormat(format, allow_ignored));
>+    if (conv->has_error()) return nullptr;
>+    return conv;
>+  }
>+
>+  ExtendedParsedFormat(string_view s, bool allow_ignored)
>+      : ParsedFormatBase(s, allow_ignored, {C...}) {}
>+};
>+}  // namespace str_format_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_STRINGS_INTERNAL_STR_FORMAT_PARSER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/parser_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/parser_test.cc
>new file mode 100644
>index 00000000000..e698020b1ab
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_format/parser_test.cc
>@@ -0,0 +1,379 @@
>+#include "absl/strings/internal/str_format/parser.h"
>+
>+#include <string.h>
>+#include "gtest/gtest.h"
>+#include "absl/base/macros.h"
>+
>+namespace absl {
>+namespace str_format_internal {
>+
>+namespace {
>+
>+TEST(LengthModTest, Names) {
>+  struct Expectation {
>+    int line;
>+    LengthMod::Id id;
>+    const char *name;
>+  };
>+  const Expectation kExpect[] = {
>+    {__LINE__, LengthMod::none, ""  },
>+    {__LINE__, LengthMod::h,    "h" },
>+    {__LINE__, LengthMod::hh,   "hh"},
>+    {__LINE__, LengthMod::l,    "l" },
>+    {__LINE__, LengthMod::ll,   "ll"},
>+    {__LINE__, LengthMod::L,    "L" },
>+    {__LINE__, LengthMod::j,    "j" },
>+    {__LINE__, LengthMod::z,    "z" },
>+    {__LINE__, LengthMod::t,    "t" },
>+    {__LINE__, LengthMod::q,    "q" },
>+  };
>+  EXPECT_EQ(ABSL_ARRAYSIZE(kExpect), LengthMod::kNumValues);
>+  for (auto e : kExpect) {
>+    SCOPED_TRACE(e.line);
>+    LengthMod mod = LengthMod::FromId(e.id);
>+    EXPECT_EQ(e.id, mod.id());
>+    EXPECT_EQ(e.name, mod.name());
>+  }
>+}
>+
>+TEST(ConversionCharTest, Names) {
>+  struct Expectation {
>+    ConversionChar::Id id;
>+    char name;
>+  };
>+  // clang-format off
>+  const Expectation kExpect[] = {
>+#define X(c) {ConversionChar::c, #c[0]}
>+    X(c), X(C), X(s), X(S),                          // text
>+    X(d), X(i), X(o), X(u), X(x), X(X),              // int
>+    X(f), X(F), X(e), X(E), X(g), X(G), X(a), X(A),  // float
>+    X(n), X(p),                                      // misc
>+#undef X
>+    {ConversionChar::none, '\0'},
>+  };
>+  // clang-format on
>+  EXPECT_EQ(ABSL_ARRAYSIZE(kExpect), ConversionChar::kNumValues);
>+  for (auto e : kExpect) {
>+    SCOPED_TRACE(e.name);
>+    ConversionChar v = ConversionChar::FromId(e.id);
>+    EXPECT_EQ(e.id, v.id());
>+    EXPECT_EQ(e.name, v.Char());
>+  }
>+}
>+
>+class ConsumeUnboundConversionTest : public ::testing::Test {
>+ public:
>+  typedef UnboundConversion Props;
>+  string_view Consume(string_view* src) {
>+    int next = 0;
>+    const char* prev_begin = src->begin();
>+    o = UnboundConversion();  // refresh
>+    ConsumeUnboundConversion(src, &o, &next);
>+    return {prev_begin, static_cast<size_t>(src->begin() - prev_begin)};
>+  }
>+
>+  bool Run(const char *fmt, bool force_positional = false) {
>+    string_view src = fmt;
>+    int next = force_positional ? -1 : 0;
>+    o = UnboundConversion();  // refresh
>+    return ConsumeUnboundConversion(&src, &o, &next) && src.empty();
>+  }
>+  UnboundConversion o;
>+};
>+
>+TEST_F(ConsumeUnboundConversionTest, ConsumeSpecification) {
>+  struct Expectation {
>+    int line;
>+    const char *src;
>+    const char *out;
>+    const char *src_post;
>+  };
>+  const Expectation kExpect[] = {
>+    {__LINE__, "",     "",     ""  },
>+    {__LINE__, "b",    "",     "b" },  // 'b' is invalid
>+    {__LINE__, "ba",   "",     "ba"},  // 'b' is invalid
>+    {__LINE__, "l",    "",     "l" },  // just length mod isn't okay
>+    {__LINE__, "d",    "d",    ""  },  // basic
>+    {__LINE__, "d ",   "d",    " " },  // leave suffix
>+    {__LINE__, "dd",   "d",    "d" },  // don't be greedy
>+    {__LINE__, "d9",   "d",    "9" },  // leave non-space suffix
>+    {__LINE__, "dzz",  "d",    "zz"},  // length mod as suffix
>+    {__LINE__, "1$*2$d", "1$*2$d", ""  },  // arg indexing and * allowed.
>+    {__LINE__, "0-14.3hhd", "0-14.3hhd", ""},  // precision, width
>+    {__LINE__, " 0-+#14.3hhd", " 0-+#14.3hhd", ""},  // flags
>+  };
>+  for (const auto& e : kExpect) {
>+    SCOPED_TRACE(e.line);
>+    string_view src = e.src;
>+    EXPECT_EQ(e.src, src);
>+    string_view out = Consume(&src);
>+    EXPECT_EQ(e.out, out);
>+    EXPECT_EQ(e.src_post, src);
>+  }
>+}
>+
>+TEST_F(ConsumeUnboundConversionTest, BasicConversion) {
>+  EXPECT_FALSE(Run(""));
>+  EXPECT_FALSE(Run("z"));
>+
>+  EXPECT_FALSE(Run("dd"));  // no excess allowed
>+
>+  EXPECT_TRUE(Run("d"));
>+  EXPECT_EQ('d', o.conv.Char());
>+  EXPECT_FALSE(o.width.is_from_arg());
>+  EXPECT_LT(o.width.value(), 0);
>+  EXPECT_FALSE(o.precision.is_from_arg());
>+  EXPECT_LT(o.precision.value(), 0);
>+  EXPECT_EQ(1, o.arg_position);
>+  EXPECT_EQ(LengthMod::none, o.length_mod.id());
>+}
>+
>+TEST_F(ConsumeUnboundConversionTest, ArgPosition) {
>+  EXPECT_TRUE(Run("d"));
>+  EXPECT_EQ(1, o.arg_position);
>+  EXPECT_TRUE(Run("3$d"));
>+  EXPECT_EQ(3, o.arg_position);
>+  EXPECT_TRUE(Run("1$d"));
>+  EXPECT_EQ(1, o.arg_position);
>+  EXPECT_TRUE(Run("1$d", true));
>+  EXPECT_EQ(1, o.arg_position);
>+  EXPECT_TRUE(Run("123$d"));
>+  EXPECT_EQ(123, o.arg_position);
>+  EXPECT_TRUE(Run("123$d", true));
>+  EXPECT_EQ(123, o.arg_position);
>+  EXPECT_TRUE(Run("10$d"));
>+  EXPECT_EQ(10, o.arg_position);
>+  EXPECT_TRUE(Run("10$d", true));
>+  EXPECT_EQ(10, o.arg_position);
>+
>+  // Position can't be zero.
>+  EXPECT_FALSE(Run("0$d"));
>+  EXPECT_FALSE(Run("0$d", true));
>+  EXPECT_FALSE(Run("1$*0$d"));
>+  EXPECT_FALSE(Run("1$.*0$d"));
>+
>+  // Position can't start with a zero digit at all. That is not a 'decimal'.
>+  EXPECT_FALSE(Run("01$p"));
>+  EXPECT_FALSE(Run("01$p", true));
>+  EXPECT_FALSE(Run("1$*01$p"));
>+  EXPECT_FALSE(Run("1$.*01$p"));
>+}
>+
>+TEST_F(ConsumeUnboundConversionTest, WidthAndPrecision) {
>+  EXPECT_TRUE(Run("14d"));
>+  EXPECT_EQ('d', o.conv.Char());
>+  EXPECT_FALSE(o.width.is_from_arg());
>+  EXPECT_EQ(14, o.width.value());
>+  EXPECT_FALSE(o.precision.is_from_arg());
>+  EXPECT_LT(o.precision.value(), 0);
>+
>+  EXPECT_TRUE(Run("14.d"));
>+  EXPECT_FALSE(o.width.is_from_arg());
>+  EXPECT_FALSE(o.precision.is_from_arg());
>+  EXPECT_EQ(14, o.width.value());
>+  EXPECT_EQ(0, o.precision.value());
>+
>+  EXPECT_TRUE(Run(".d"));
>+  EXPECT_FALSE(o.width.is_from_arg());
>+  EXPECT_LT(o.width.value(), 0);
>+  EXPECT_FALSE(o.precision.is_from_arg());
>+  EXPECT_EQ(0, o.precision.value());
>+
>+  EXPECT_TRUE(Run(".5d"));
>+  EXPECT_FALSE(o.width.is_from_arg());
>+  EXPECT_LT(o.width.value(), 0);
>+  EXPECT_FALSE(o.precision.is_from_arg());
>+  EXPECT_EQ(5, o.precision.value());
>+
>+  EXPECT_TRUE(Run(".0d"));
>+  EXPECT_FALSE(o.width.is_from_arg());
>+  EXPECT_LT(o.width.value(), 0);
>+  EXPECT_FALSE(o.precision.is_from_arg());
>+  EXPECT_EQ(0, o.precision.value());
>+
>+  EXPECT_TRUE(Run("14.5d"));
>+  EXPECT_FALSE(o.width.is_from_arg());
>+  EXPECT_FALSE(o.precision.is_from_arg());
>+  EXPECT_EQ(14, o.width.value());
>+  EXPECT_EQ(5, o.precision.value());
>+
>+  EXPECT_TRUE(Run("*.*d"));
>+  EXPECT_TRUE(o.width.is_from_arg());
>+  EXPECT_EQ(1, o.width.get_from_arg());
>+  EXPECT_TRUE(o.precision.is_from_arg());
>+  EXPECT_EQ(2, o.precision.get_from_arg());
>+  EXPECT_EQ(3, o.arg_position);
>+
>+  EXPECT_TRUE(Run("*d"));
>+  EXPECT_TRUE(o.width.is_from_arg());
>+  EXPECT_EQ(1, o.width.get_from_arg());
>+  EXPECT_FALSE(o.precision.is_from_arg());
>+  EXPECT_LT(o.precision.value(), 0);
>+  EXPECT_EQ(2, o.arg_position);
>+
>+  EXPECT_TRUE(Run(".*d"));
>+  EXPECT_FALSE(o.width.is_from_arg());
>+  EXPECT_LT(o.width.value(), 0);
>+  EXPECT_TRUE(o.precision.is_from_arg());
>+  EXPECT_EQ(1, o.precision.get_from_arg());
>+  EXPECT_EQ(2, o.arg_position);
>+
>+  // mixed implicit and explicit: didn't specify arg position.
>+  EXPECT_FALSE(Run("*23$.*34$d"));
>+
>+  EXPECT_TRUE(Run("12$*23$.*34$d"));
>+  EXPECT_EQ(12, o.arg_position);
>+  EXPECT_TRUE(o.width.is_from_arg());
>+  EXPECT_EQ(23, o.width.get_from_arg());
>+  EXPECT_TRUE(o.precision.is_from_arg());
>+  EXPECT_EQ(34, o.precision.get_from_arg());
>+
>+  EXPECT_TRUE(Run("2$*5$.*9$d"));
>+  EXPECT_EQ(2, o.arg_position);
>+  EXPECT_TRUE(o.width.is_from_arg());
>+  EXPECT_EQ(5, o.width.get_from_arg());
>+  EXPECT_TRUE(o.precision.is_from_arg());
>+  EXPECT_EQ(9, o.precision.get_from_arg());
>+
>+  EXPECT_FALSE(Run(".*0$d")) << "no arg 0";
>+}
>+
>+TEST_F(ConsumeUnboundConversionTest, Flags) {
>+  static const char kAllFlags[] = "-+ #0";
>+  static const int kNumFlags = ABSL_ARRAYSIZE(kAllFlags) - 1;
>+  for (int rev = 0; rev < 2; ++rev) {
>+    for (int i = 0; i < 1 << kNumFlags; ++i) {
>+      std::string fmt;
>+      for (int k = 0; k < kNumFlags; ++k)
>+        if ((i >> k) & 1) fmt += kAllFlags[k];
>+      // flag order shouldn't matter
>+      if (rev == 1) { std::reverse(fmt.begin(), fmt.end()); }
>+      fmt += 'd';
>+      SCOPED_TRACE(fmt);
>+      EXPECT_TRUE(Run(fmt.c_str()));
>+      EXPECT_EQ(fmt.find('-') == std::string::npos, !o.flags.left);
>+      EXPECT_EQ(fmt.find('+') == std::string::npos, !o.flags.show_pos);
>+      EXPECT_EQ(fmt.find(' ') == std::string::npos, !o.flags.sign_col);
>+      EXPECT_EQ(fmt.find('#') == std::string::npos, !o.flags.alt);
>+      EXPECT_EQ(fmt.find('0') == std::string::npos, !o.flags.zero);
>+    }
>+  }
>+}
>+
>+TEST_F(ConsumeUnboundConversionTest, BasicFlag) {
>+  // Flag is on
>+  for (const char* fmt : {"d", "llx", "G", "1$X"}) {
>+    SCOPED_TRACE(fmt);
>+    EXPECT_TRUE(Run(fmt));
>+    EXPECT_TRUE(o.flags.basic);
>+  }
>+
>+  // Flag is off
>+  for (const char* fmt : {"3d", ".llx", "-G", "1$#X"}) {
>+    SCOPED_TRACE(fmt);
>+    EXPECT_TRUE(Run(fmt));
>+    EXPECT_FALSE(o.flags.basic);
>+  }
>+}
>+
>+struct SummarizeConsumer {
>+  std::string* out;
>+  explicit SummarizeConsumer(std::string* out) : out(out) {}
>+
>+  bool Append(string_view s) {
>+    *out += "[" + std::string(s) + "]";
>+    return true;
>+  }
>+
>+  bool ConvertOne(const UnboundConversion& conv, string_view s) {
>+    *out += "{";
>+    *out += std::string(s);
>+    *out += ":";
>+    *out += std::to_string(conv.arg_position) + "$";
>+    if (conv.width.is_from_arg()) {
>+      *out += std::to_string(conv.width.get_from_arg()) + "$*";
>+    }
>+    if (conv.precision.is_from_arg()) {
>+      *out += "." + std::to_string(conv.precision.get_from_arg()) + "$*";
>+    }
>+    *out += conv.conv.Char();
>+    *out += "}";
>+    return true;
>+  }
>+};
>+
>+std::string SummarizeParsedFormat(const ParsedFormatBase& pc) {
>+  std::string out;
>+  if (!pc.ProcessFormat(SummarizeConsumer(&out))) out += "!";
>+  return out;
>+}
>+
>+class ParsedFormatTest : public testing::Test {};
>+
>+TEST_F(ParsedFormatTest, ValueSemantics) {
>+  ParsedFormatBase p1({}, true, {});  // empty format
>+  EXPECT_EQ("", SummarizeParsedFormat(p1));
>+
>+  ParsedFormatBase p2 = p1;  // copy construct (empty)
>+  EXPECT_EQ(SummarizeParsedFormat(p1), SummarizeParsedFormat(p2));
>+
>+  p1 = ParsedFormatBase("hello%s", true, {Conv::s});  // move assign
>+  EXPECT_EQ("[hello]{s:1$s}", SummarizeParsedFormat(p1));
>+
>+  ParsedFormatBase p3 = p1;  // copy construct (nonempty)
>+  EXPECT_EQ(SummarizeParsedFormat(p1), SummarizeParsedFormat(p3));
>+
>+  using std::swap;
>+  swap(p1, p2);
>+  EXPECT_EQ("", SummarizeParsedFormat(p1));
>+  EXPECT_EQ("[hello]{s:1$s}", SummarizeParsedFormat(p2));
>+  swap(p1, p2);  // undo
>+
>+  p2 = p1;  // copy assign
>+  EXPECT_EQ(SummarizeParsedFormat(p1), SummarizeParsedFormat(p2));
>+}
>+
>+struct ExpectParse {
>+  const char* in;
>+  std::initializer_list<Conv> conv_set;
>+  const char* out;
>+};
>+
>+TEST_F(ParsedFormatTest, Parsing) {
>+  // Parse should be equivalent to that obtained by ConversionParseIterator.
>+  // No need to retest the parsing edge cases here.
>+  const ExpectParse kExpect[] = {
>+      {"", {}, ""},
>+      {"ab", {}, "[ab]"},
>+      {"a%d", {Conv::d}, "[a]{d:1$d}"},
>+      {"a%+d", {Conv::d}, "[a]{+d:1$d}"},
>+      {"a% d", {Conv::d}, "[a]{ d:1$d}"},
>+      {"a%b %d", {}, "[a]!"},  // stop after error
>+  };
>+  for (const auto& e : kExpect) {
>+    SCOPED_TRACE(e.in);
>+    EXPECT_EQ(e.out,
>+              SummarizeParsedFormat(ParsedFormatBase(e.in, false, e.conv_set)));
>+  }
>+}
>+
>+TEST_F(ParsedFormatTest, ParsingFlagOrder) {
>+  const ExpectParse kExpect[] = {
>+      {"a%+ 0d", {Conv::d}, "[a]{+ 0d:1$d}"},
>+      {"a%+0 d", {Conv::d}, "[a]{+0 d:1$d}"},
>+      {"a%0+ d", {Conv::d}, "[a]{0+ d:1$d}"},
>+      {"a% +0d", {Conv::d}, "[a]{ +0d:1$d}"},
>+      {"a%0 +d", {Conv::d}, "[a]{0 +d:1$d}"},
>+      {"a% 0+d", {Conv::d}, "[a]{ 0+d:1$d}"},
>+      {"a%+   0+d", {Conv::d}, "[a]{+   0+d:1$d}"},
>+  };
>+  for (const auto& e : kExpect) {
>+    SCOPED_TRACE(e.in);
>+    EXPECT_EQ(e.out,
>+              SummarizeParsedFormat(ParsedFormatBase(e.in, false, e.conv_set)));
>+  }
>+}
>+
>+}  // namespace
>+}  // namespace str_format_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_join_internal.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_join_internal.h
>new file mode 100644
>index 00000000000..a734758c21e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_join_internal.h
>@@ -0,0 +1,311 @@
>+//
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+
>+// This file declares INTERNAL parts of the Join API that are inlined/templated
>+// or otherwise need to be available at compile time. The main abstractions
>+// defined in this file are:
>+//
>+//   - A handful of default Formatters
>+//   - JoinAlgorithm() overloads
>+//   - JoinRange() overloads
>+//   - JoinTuple()
>+//
>+// DO NOT INCLUDE THIS FILE DIRECTLY. Use this file by including
>+// absl/strings/str_join.h
>+//
>+// IWYU pragma: private, include "absl/strings/str_join.h"
>+
>+#ifndef ABSL_STRINGS_INTERNAL_STR_JOIN_INTERNAL_H_
>+#define ABSL_STRINGS_INTERNAL_STR_JOIN_INTERNAL_H_
>+
>+#include <cstring>
>+#include <iterator>
>+#include <memory>
>+#include <string>
>+#include <type_traits>
>+#include <utility>
>+
>+#include "absl/strings/internal/ostringstream.h"
>+#include "absl/strings/internal/resize_uninitialized.h"
>+#include "absl/strings/str_cat.h"
>+
>+namespace absl {
>+namespace strings_internal {
>+
>+//
>+// Formatter objects
>+//
>+// The following are implementation classes for standard Formatter objects. The
>+// factory functions that users will call to create and use these formatters are
>+// defined and documented in strings/join.h.
>+//
>+
>+// The default formatter. Converts alpha-numeric types to strings.
>+struct AlphaNumFormatterImpl {
>+  // This template is needed in order to support passing in a dereferenced
>+  // vector<bool>::iterator
>+  template <typename T>
>+  void operator()(std::string* out, const T& t) const {
>+    StrAppend(out, AlphaNum(t));
>+  }
>+
>+  void operator()(std::string* out, const AlphaNum& t) const {
>+    StrAppend(out, t);
>+  }
>+};
>+
>+// A type that's used to overload the JoinAlgorithm() function (defined below)
>+// for ranges that do not require additional formatting (e.g., a range of
>+// strings).
>+
>+struct NoFormatter : public AlphaNumFormatterImpl {};
>+
>+// Formats types to strings using the << operator.
>+class StreamFormatterImpl {
>+ public:
>+  // The method isn't const because it mutates state. Making it const will
>+  // render StreamFormatterImpl thread-hostile.
>+  template <typename T>
>+  void operator()(std::string* out, const T& t) {
>+    // The stream is created lazily to avoid paying the relatively high cost
>+    // of its construction when joining an empty range.
>+    if (strm_) {
>+      strm_->clear();  // clear the bad, fail and eof bits in case they were set
>+      strm_->str(out);
>+    } else {
>+      strm_.reset(new strings_internal::OStringStream(out));
>+    }
>+    *strm_ << t;
>+  }
>+
>+ private:
>+  std::unique_ptr<strings_internal::OStringStream> strm_;
>+};
>+
>+// Formats a std::pair<>. The 'first' member is formatted using f1_ and the
>+// 'second' member is formatted using f2_. sep_ is the separator.
>+template <typename F1, typename F2>
>+class PairFormatterImpl {
>+ public:
>+  PairFormatterImpl(F1 f1, absl::string_view sep, F2 f2)
>+      : f1_(std::move(f1)), sep_(sep), f2_(std::move(f2)) {}
>+
>+  template <typename T>
>+  void operator()(std::string* out, const T& p) {
>+    f1_(out, p.first);
>+    out->append(sep_);
>+    f2_(out, p.second);
>+  }
>+
>+  template <typename T>
>+  void operator()(std::string* out, const T& p) const {
>+    f1_(out, p.first);
>+    out->append(sep_);
>+    f2_(out, p.second);
>+  }
>+
>+ private:
>+  F1 f1_;
>+  std::string sep_;
>+  F2 f2_;
>+};
>+
>+// Wraps another formatter and dereferences the argument to operator() then
>+// passes the dereferenced argument to the wrapped formatter. This can be
>+// useful, for example, to join a std::vector<int*>.
>+template <typename Formatter>
>+class DereferenceFormatterImpl {
>+ public:
>+  DereferenceFormatterImpl() : f_() {}
>+  explicit DereferenceFormatterImpl(Formatter&& f)
>+      : f_(std::forward<Formatter>(f)) {}
>+
>+  template <typename T>
>+  void operator()(std::string* out, const T& t) {
>+    f_(out, *t);
>+  }
>+
>+  template <typename T>
>+  void operator()(std::string* out, const T& t) const {
>+    f_(out, *t);
>+  }
>+
>+ private:
>+  Formatter f_;
>+};
>+
>+// DefaultFormatter<T> is a traits class that selects a default Formatter to use
>+// for the given type T. The ::Type member names the Formatter to use. This is
>+// used by the strings::Join() functions that do NOT take a Formatter argument,
>+// in which case a default Formatter must be chosen.
>+//
>+// AlphaNumFormatterImpl is the default in the base template, followed by
>+// specializations for other types.
>+template <typename ValueType>
>+struct DefaultFormatter {
>+  typedef AlphaNumFormatterImpl Type;
>+};
>+template <>
>+struct DefaultFormatter<const char*> {
>+  typedef AlphaNumFormatterImpl Type;
>+};
>+template <>
>+struct DefaultFormatter<char*> {
>+  typedef AlphaNumFormatterImpl Type;
>+};
>+template <>
>+struct DefaultFormatter<std::string> {
>+  typedef NoFormatter Type;
>+};
>+template <>
>+struct DefaultFormatter<absl::string_view> {
>+  typedef NoFormatter Type;
>+};
>+template <typename ValueType>
>+struct DefaultFormatter<ValueType*> {
>+  typedef DereferenceFormatterImpl<typename DefaultFormatter<ValueType>::Type>
>+      Type;
>+};
>+
>+template <typename ValueType>
>+struct DefaultFormatter<std::unique_ptr<ValueType>>
>+    : public DefaultFormatter<ValueType*> {};
>+
>+//
>+// JoinAlgorithm() functions
>+//
>+
>+// The main joining algorithm. This simply joins the elements in the given
>+// iterator range, each separated by the given separator, into an output std::string,
>+// and formats each element using the provided Formatter object.
>+template <typename Iterator, typename Formatter>
>+std::string JoinAlgorithm(Iterator start, Iterator end, absl::string_view s,
>+                     Formatter&& f) {
>+  std::string result;
>+  absl::string_view sep("");
>+  for (Iterator it = start; it != end; ++it) {
>+    result.append(sep.data(), sep.size());
>+    f(&result, *it);
>+    sep = s;
>+  }
>+  return result;
>+}
>+
>+// A joining algorithm that's optimized for a forward iterator range of
>+// std::string-like objects that do not need any additional formatting. This is to
>+// optimize the common case of joining, say, a std::vector<std::string> or a
>+// std::vector<absl::string_view>.
>+//
>+// This is an overload of the previous JoinAlgorithm() function. Here the
>+// Formatter argument is of type NoFormatter. Since NoFormatter is an internal
>+// type, this overload is only invoked when strings::Join() is called with a
>+// range of std::string-like objects (e.g., std::string, absl::string_view), and an
>+// explicit Formatter argument was NOT specified.
>+//
>+// The optimization is that the needed space will be reserved in the output
>+// std::string to avoid the need to resize while appending. To do this, the iterator
>+// range will be traversed twice: once to calculate the total needed size, and
>+// then again to copy the elements and delimiters to the output std::string.
>+template <typename Iterator,
>+          typename = typename std::enable_if<std::is_convertible<
>+              typename std::iterator_traits<Iterator>::iterator_category,
>+              std::forward_iterator_tag>::value>::type>
>+std::string JoinAlgorithm(Iterator start, Iterator end, absl::string_view s,
>+                     NoFormatter) {
>+  std::string result;
>+  if (start != end) {
>+    // Sums size
>+    size_t result_size = start->size();
>+    for (Iterator it = start; ++it != end;) {
>+      result_size += s.size();
>+      result_size += it->size();
>+    }
>+
>+    if (result_size > 0) {
>+      STLStringResizeUninitialized(&result, result_size);
>+
>+      // Joins strings
>+      char* result_buf = &*result.begin();
>+      memcpy(result_buf, start->data(), start->size());
>+      result_buf += start->size();
>+      for (Iterator it = start; ++it != end;) {
>+        memcpy(result_buf, s.data(), s.size());
>+        result_buf += s.size();
>+        memcpy(result_buf, it->data(), it->size());
>+        result_buf += it->size();
>+      }
>+    }
>+  }
>+
>+  return result;
>+}
>+
>+// JoinTupleLoop implements a loop over the elements of a std::tuple, which
>+// are heterogeneous. The primary template matches the tuple interior case. It
>+// continues the iteration after appending a separator (for nonzero indices)
>+// and formatting an element of the tuple. The specialization for the I=N case
>+// matches the end-of-tuple, and terminates the iteration.
>+template <size_t I, size_t N>
>+struct JoinTupleLoop {
>+  template <typename Tup, typename Formatter>
>+  void operator()(std::string* out, const Tup& tup, absl::string_view sep,
>+                  Formatter&& fmt) {
>+    if (I > 0) out->append(sep.data(), sep.size());
>+    fmt(out, std::get<I>(tup));
>+    JoinTupleLoop<I + 1, N>()(out, tup, sep, fmt);
>+  }
>+};
>+template <size_t N>
>+struct JoinTupleLoop<N, N> {
>+  template <typename Tup, typename Formatter>
>+  void operator()(std::string*, const Tup&, absl::string_view, Formatter&&) {}
>+};
>+
>+template <typename... T, typename Formatter>
>+std::string JoinAlgorithm(const std::tuple<T...>& tup, absl::string_view sep,
>+                     Formatter&& fmt) {
>+  std::string result;
>+  JoinTupleLoop<0, sizeof...(T)>()(&result, tup, sep, fmt);
>+  return result;
>+}
>+
>+template <typename Iterator>
>+std::string JoinRange(Iterator first, Iterator last, absl::string_view separator) {
>+  // No formatter was explicitly given, so a default must be chosen.
>+  typedef typename std::iterator_traits<Iterator>::value_type ValueType;
>+  typedef typename DefaultFormatter<ValueType>::Type Formatter;
>+  return JoinAlgorithm(first, last, separator, Formatter());
>+}
>+
>+template <typename Range, typename Formatter>
>+std::string JoinRange(const Range& range, absl::string_view separator,
>+                 Formatter&& fmt) {
>+  using std::begin;
>+  using std::end;
>+  return JoinAlgorithm(begin(range), end(range), separator, fmt);
>+}
>+
>+template <typename Range>
>+std::string JoinRange(const Range& range, absl::string_view separator) {
>+  using std::begin;
>+  using std::end;
>+  return JoinRange(begin(range), end(range), separator);
>+}
>+
>+}  // namespace strings_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_STRINGS_INTERNAL_STR_JOIN_INTERNAL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_split_internal.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_split_internal.h
>new file mode 100644
>index 00000000000..9cf0833f490
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/str_split_internal.h
>@@ -0,0 +1,453 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+
>+// This file declares INTERNAL parts of the Split API that are inline/templated
>+// or otherwise need to be available at compile time. The main abstractions
>+// defined in here are
>+//
>+//   - ConvertibleToStringView
>+//   - SplitIterator<>
>+//   - Splitter<>
>+//
>+// DO NOT INCLUDE THIS FILE DIRECTLY. Use this file by including
>+// absl/strings/str_split.h.
>+//
>+// IWYU pragma: private, include "absl/strings/str_split.h"
>+
>+#ifndef ABSL_STRINGS_INTERNAL_STR_SPLIT_INTERNAL_H_
>+#define ABSL_STRINGS_INTERNAL_STR_SPLIT_INTERNAL_H_
>+
>+#include <array>
>+#include <initializer_list>
>+#include <iterator>
>+#include <map>
>+#include <type_traits>
>+#include <utility>
>+#include <vector>
>+
>+#include "absl/base/macros.h"
>+#include "absl/base/port.h"
>+#include "absl/meta/type_traits.h"
>+#include "absl/strings/string_view.h"
>+
>+#ifdef _GLIBCXX_DEBUG
>+#include "absl/strings/internal/stl_type_traits.h"
>+#endif  // _GLIBCXX_DEBUG
>+
>+namespace absl {
>+namespace strings_internal {
>+
>+// This class is implicitly constructible from everything that absl::string_view
>+// is implicitly constructible from. If it's constructed from a temporary
>+// std::string, the data is moved into a data member so its lifetime matches that of
>+// the ConvertibleToStringView instance.
>+class ConvertibleToStringView {
>+ public:
>+  ConvertibleToStringView(const char* s)  // NOLINT(runtime/explicit)
>+      : value_(s) {}
>+  ConvertibleToStringView(char* s) : value_(s) {}  // NOLINT(runtime/explicit)
>+  ConvertibleToStringView(absl::string_view s)     // NOLINT(runtime/explicit)
>+      : value_(s) {}
>+  ConvertibleToStringView(const std::string& s)  // NOLINT(runtime/explicit)
>+      : value_(s) {}
>+
>+  // Matches rvalue strings and moves their data to a member.
>+ConvertibleToStringView(std::string&& s)  // NOLINT(runtime/explicit)
>+    : copy_(std::move(s)), value_(copy_) {}
>+
>+  ConvertibleToStringView(const ConvertibleToStringView& other)
>+      : copy_(other.copy_),
>+        value_(other.IsSelfReferential() ? copy_ : other.value_) {}
>+
>+  ConvertibleToStringView(ConvertibleToStringView&& other) {
>+    StealMembers(std::move(other));
>+  }
>+
>+  ConvertibleToStringView& operator=(ConvertibleToStringView other) {
>+    StealMembers(std::move(other));
>+    return *this;
>+  }
>+
>+  absl::string_view value() const { return value_; }
>+
>+ private:
>+  // Returns true if ctsp's value refers to its internal copy_ member.
>+  bool IsSelfReferential() const { return value_.data() == copy_.data(); }
>+
>+  void StealMembers(ConvertibleToStringView&& other) {
>+    if (other.IsSelfReferential()) {
>+      copy_ = std::move(other.copy_);
>+      value_ = copy_;
>+      other.value_ = other.copy_;
>+    } else {
>+      value_ = other.value_;
>+    }
>+  }
>+
>+  // Holds the data moved from temporary std::string arguments. Declared first so
>+  // that 'value' can refer to 'copy_'.
>+  std::string copy_;
>+  absl::string_view value_;
>+};
>+
>+// An iterator that enumerates the parts of a std::string from a Splitter. The text
>+// to be split, the Delimiter, and the Predicate are all taken from the given
>+// Splitter object. Iterators may only be compared if they refer to the same
>+// Splitter instance.
>+//
>+// This class is NOT part of the public splitting API.
>+template <typename Splitter>
>+class SplitIterator {
>+ public:
>+  using iterator_category = std::input_iterator_tag;
>+  using value_type = absl::string_view;
>+  using difference_type = ptrdiff_t;
>+  using pointer = const value_type*;
>+  using reference = const value_type&;
>+
>+  enum State { kInitState, kLastState, kEndState };
>+  SplitIterator(State state, const Splitter* splitter)
>+      : pos_(0),
>+        state_(state),
>+        splitter_(splitter),
>+        delimiter_(splitter->delimiter()),
>+        predicate_(splitter->predicate()) {
>+    // Hack to maintain backward compatibility. This one block makes it so an
>+    // empty absl::string_view whose .data() happens to be nullptr behaves
>+    // *differently* from an otherwise empty absl::string_view whose .data() is
>+    // not nullptr. This is an undesirable difference in general, but this
>+    // behavior is maintained to avoid breaking existing code that happens to
>+    // depend on this old behavior/bug. Perhaps it will be fixed one day. The
>+    // difference in behavior is as follows:
>+    //   Split(absl::string_view(""), '-');  // {""}
>+    //   Split(absl::string_view(), '-');    // {}
>+    if (splitter_->text().data() == nullptr) {
>+      state_ = kEndState;
>+      pos_ = splitter_->text().size();
>+      return;
>+    }
>+
>+    if (state_ == kEndState) {
>+      pos_ = splitter_->text().size();
>+    } else {
>+      ++(*this);
>+    }
>+  }
>+
>+  bool at_end() const { return state_ == kEndState; }
>+
>+  reference operator*() const { return curr_; }
>+  pointer operator->() const { return &curr_; }
>+
>+  SplitIterator& operator++() {
>+    do {
>+      if (state_ == kLastState) {
>+        state_ = kEndState;
>+        return *this;
>+      }
>+      const absl::string_view text = splitter_->text();
>+      const absl::string_view d = delimiter_.Find(text, pos_);
>+      if (d.data() == text.end()) state_ = kLastState;
>+      curr_ = text.substr(pos_, d.data() - (text.data() + pos_));
>+      pos_ += curr_.size() + d.size();
>+    } while (!predicate_(curr_));
>+    return *this;
>+  }
>+
>+  SplitIterator operator++(int) {
>+    SplitIterator old(*this);
>+    ++(*this);
>+    return old;
>+  }
>+
>+  friend bool operator==(const SplitIterator& a, const SplitIterator& b) {
>+    return a.state_ == b.state_ && a.pos_ == b.pos_;
>+  }
>+
>+  friend bool operator!=(const SplitIterator& a, const SplitIterator& b) {
>+    return !(a == b);
>+  }
>+
>+ private:
>+  size_t pos_;
>+  State state_;
>+  absl::string_view curr_;
>+  const Splitter* splitter_;
>+  typename Splitter::DelimiterType delimiter_;
>+  typename Splitter::PredicateType predicate_;
>+};
>+
>+// HasMappedType<T>::value is true iff there exists a type T::mapped_type.
>+template <typename T, typename = void>
>+struct HasMappedType : std::false_type {};
>+template <typename T>
>+struct HasMappedType<T, absl::void_t<typename T::mapped_type>>
>+    : std::true_type {};
>+
>+// HasValueType<T>::value is true iff there exists a type T::value_type.
>+template <typename T, typename = void>
>+struct HasValueType : std::false_type {};
>+template <typename T>
>+struct HasValueType<T, absl::void_t<typename T::value_type>> : std::true_type {
>+};
>+
>+// HasConstIterator<T>::value is true iff there exists a type T::const_iterator.
>+template <typename T, typename = void>
>+struct HasConstIterator : std::false_type {};
>+template <typename T>
>+struct HasConstIterator<T, absl::void_t<typename T::const_iterator>>
>+    : std::true_type {};
>+
>+// IsInitializerList<T>::value is true iff T is an std::initializer_list. More
>+// details below in Splitter<> where this is used.
>+std::false_type IsInitializerListDispatch(...);  // default: No
>+template <typename T>
>+std::true_type IsInitializerListDispatch(std::initializer_list<T>*);
>+template <typename T>
>+struct IsInitializerList
>+    : decltype(IsInitializerListDispatch(static_cast<T*>(nullptr))) {};
>+
>+// A SplitterIsConvertibleTo<C>::type alias exists iff the specified condition
>+// is true for type 'C'.
>+//
>+// Restricts conversion to container-like types (by testing for the presence of
>+// a const_iterator member type) and also to disable conversion to an
>+// std::initializer_list (which also has a const_iterator). Otherwise, code
>+// compiled in C++11 will get an error due to ambiguous conversion paths (in
>+// C++11 std::vector<T>::operator= is overloaded to take either a std::vector<T>
>+// or an std::initializer_list<T>).
>+
>+template <typename C, bool has_value_type, bool has_mapped_type>
>+struct SplitterIsConvertibleToImpl : std::false_type {};
>+
>+template <typename C>
>+struct SplitterIsConvertibleToImpl<C, true, false>
>+    : std::is_constructible<typename C::value_type, absl::string_view> {};
>+
>+template <typename C>
>+struct SplitterIsConvertibleToImpl<C, true, true>
>+    : absl::conjunction<
>+          std::is_constructible<typename C::key_type, absl::string_view>,
>+          std::is_constructible<typename C::mapped_type, absl::string_view>> {};
>+
>+template <typename C>
>+struct SplitterIsConvertibleTo
>+    : SplitterIsConvertibleToImpl<
>+          C,
>+#ifdef _GLIBCXX_DEBUG
>+          !IsStrictlyBaseOfAndConvertibleToSTLContainer<C>::value &&
>+#endif  // _GLIBCXX_DEBUG
>+              !IsInitializerList<
>+                  typename std::remove_reference<C>::type>::value &&
>+              HasValueType<C>::value && HasConstIterator<C>::value,
>+          HasMappedType<C>::value> {
>+};
>+
>+// This class implements the range that is returned by absl::StrSplit(). This
>+// class has templated conversion operators that allow it to be implicitly
>+// converted to a variety of types that the caller may have specified on the
>+// left-hand side of an assignment.
>+//
>+// The main interface for interacting with this class is through its implicit
>+// conversion operators. However, this class may also be used like a container
>+// in that it has .begin() and .end() member functions. It may also be used
>+// within a range-for loop.
>+//
>+// Output containers can be collections of any type that is constructible from
>+// an absl::string_view.
>+//
>+// An Predicate functor may be supplied. This predicate will be used to filter
>+// the split strings: only strings for which the predicate returns true will be
>+// kept. A Predicate object is any unary functor that takes an absl::string_view
>+// and returns bool.
>+template <typename Delimiter, typename Predicate>
>+class Splitter {
>+ public:
>+  using DelimiterType = Delimiter;
>+  using PredicateType = Predicate;
>+  using const_iterator = strings_internal::SplitIterator<Splitter>;
>+  using value_type = typename std::iterator_traits<const_iterator>::value_type;
>+
>+  Splitter(ConvertibleToStringView input_text, Delimiter d, Predicate p)
>+      : text_(std::move(input_text)),
>+        delimiter_(std::move(d)),
>+        predicate_(std::move(p)) {}
>+
>+  absl::string_view text() const { return text_.value(); }
>+  const Delimiter& delimiter() const { return delimiter_; }
>+  const Predicate& predicate() const { return predicate_; }
>+
>+  // Range functions that iterate the split substrings as absl::string_view
>+  // objects. These methods enable a Splitter to be used in a range-based for
>+  // loop.
>+  const_iterator begin() const { return {const_iterator::kInitState, this}; }
>+  const_iterator end() const { return {const_iterator::kEndState, this}; }
>+
>+  // An implicit conversion operator that is restricted to only those containers
>+  // that the splitter is convertible to.
>+  template <typename Container,
>+            typename = typename std::enable_if<
>+                SplitterIsConvertibleTo<Container>::value>::type>
>+  operator Container() const {  // NOLINT(runtime/explicit)
>+    return ConvertToContainer<Container, typename Container::value_type,
>+                              HasMappedType<Container>::value>()(*this);
>+  }
>+
>+  // Returns a pair with its .first and .second members set to the first two
>+  // strings returned by the begin() iterator. Either/both of .first and .second
>+  // will be constructed with empty strings if the iterator doesn't have a
>+  // corresponding value.
>+  template <typename First, typename Second>
>+  operator std::pair<First, Second>() const {  // NOLINT(runtime/explicit)
>+    absl::string_view first, second;
>+    auto it = begin();
>+    if (it != end()) {
>+      first = *it;
>+      if (++it != end()) {
>+        second = *it;
>+      }
>+    }
>+    return {First(first), Second(second)};
>+  }
>+
>+ private:
>+  // ConvertToContainer is a functor converting a Splitter to the requested
>+  // Container of ValueType. It is specialized below to optimize splitting to
>+  // certain combinations of Container and ValueType.
>+  //
>+  // This base template handles the generic case of storing the split results in
>+  // the requested non-map-like container and converting the split substrings to
>+  // the requested type.
>+  template <typename Container, typename ValueType, bool is_map = false>
>+  struct ConvertToContainer {
>+    Container operator()(const Splitter& splitter) const {
>+      Container c;
>+      auto it = std::inserter(c, c.end());
>+      for (const auto sp : splitter) {
>+        *it++ = ValueType(sp);
>+      }
>+      return c;
>+    }
>+  };
>+
>+  // Partial specialization for a std::vector<absl::string_view>.
>+  //
>+  // Optimized for the common case of splitting to a
>+  // std::vector<absl::string_view>. In this case we first split the results to
>+  // a small array of absl::string_view on the stack, to reduce reallocations.
>+  template <typename A>
>+  struct ConvertToContainer<std::vector<absl::string_view, A>,
>+                            absl::string_view, false> {
>+    std::vector<absl::string_view, A> operator()(
>+        const Splitter& splitter) const {
>+      struct raw_view {
>+        const char* data;
>+        size_t size;
>+        operator absl::string_view() const {  // NOLINT(runtime/explicit)
>+          return {data, size};
>+        }
>+      };
>+      std::vector<absl::string_view, A> v;
>+      std::array<raw_view, 16> ar;
>+      for (auto it = splitter.begin(); !it.at_end();) {
>+        size_t index = 0;
>+        do {
>+          ar[index].data = it->data();
>+          ar[index].size = it->size();
>+          ++it;
>+        } while (++index != ar.size() && !it.at_end());
>+        v.insert(v.end(), ar.begin(), ar.begin() + index);
>+      }
>+      return v;
>+    }
>+  };
>+
>+  // Partial specialization for a std::vector<std::string>.
>+  //
>+  // Optimized for the common case of splitting to a std::vector<std::string>. In
>+  // this case we first split the results to a std::vector<absl::string_view> so
>+  // the returned std::vector<std::string> can have space reserved to avoid std::string
>+  // moves.
>+  template <typename A>
>+  struct ConvertToContainer<std::vector<std::string, A>, std::string, false> {
>+    std::vector<std::string, A> operator()(const Splitter& splitter) const {
>+      const std::vector<absl::string_view> v = splitter;
>+      return std::vector<std::string, A>(v.begin(), v.end());
>+    }
>+  };
>+
>+  // Partial specialization for containers of pairs (e.g., maps).
>+  //
>+  // The algorithm is to insert a new pair into the map for each even-numbered
>+  // item, with the even-numbered item as the key with a default-constructed
>+  // value. Each odd-numbered item will then be assigned to the last pair's
>+  // value.
>+  template <typename Container, typename First, typename Second>
>+  struct ConvertToContainer<Container, std::pair<const First, Second>, true> {
>+    Container operator()(const Splitter& splitter) const {
>+      Container m;
>+      typename Container::iterator it;
>+      bool insert = true;
>+      for (const auto sp : splitter) {
>+        if (insert) {
>+          it = Inserter<Container>::Insert(&m, First(sp), Second());
>+        } else {
>+          it->second = Second(sp);
>+        }
>+        insert = !insert;
>+      }
>+      return m;
>+    }
>+
>+    // Inserts the key and value into the given map, returning an iterator to
>+    // the inserted item. Specialized for std::map and std::multimap to use
>+    // emplace() and adapt emplace()'s return value.
>+    template <typename Map>
>+    struct Inserter {
>+      using M = Map;
>+      template <typename... Args>
>+      static typename M::iterator Insert(M* m, Args&&... args) {
>+        return m->insert(std::make_pair(std::forward<Args>(args)...)).first;
>+      }
>+    };
>+
>+    template <typename... Ts>
>+    struct Inserter<std::map<Ts...>> {
>+      using M = std::map<Ts...>;
>+      template <typename... Args>
>+      static typename M::iterator Insert(M* m, Args&&... args) {
>+        return m->emplace(std::make_pair(std::forward<Args>(args)...)).first;
>+      }
>+    };
>+
>+    template <typename... Ts>
>+    struct Inserter<std::multimap<Ts...>> {
>+      using M = std::multimap<Ts...>;
>+      template <typename... Args>
>+      static typename M::iterator Insert(M* m, Args&&... args) {
>+        return m->emplace(std::make_pair(std::forward<Args>(args)...));
>+      }
>+    };
>+  };
>+
>+  ConvertibleToStringView text_;
>+  Delimiter delimiter_;
>+  Predicate predicate_;
>+};
>+
>+}  // namespace strings_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_STRINGS_INTERNAL_STR_SPLIT_INTERNAL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/utf8.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/utf8.cc
>new file mode 100644
>index 00000000000..2415c2ccc45
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/utf8.cc
>@@ -0,0 +1,51 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+// UTF8 utilities, implemented to reduce dependencies.
>+
>+#include "absl/strings/internal/utf8.h"
>+
>+namespace absl {
>+namespace strings_internal {
>+
>+size_t EncodeUTF8Char(char *buffer, char32_t utf8_char) {
>+  if (utf8_char <= 0x7F) {
>+    *buffer = static_cast<char>(utf8_char);
>+    return 1;
>+  } else if (utf8_char <= 0x7FF) {
>+    buffer[1] = 0x80 | (utf8_char & 0x3F);
>+    utf8_char >>= 6;
>+    buffer[0] = 0xC0 | utf8_char;
>+    return 2;
>+  } else if (utf8_char <= 0xFFFF) {
>+    buffer[2] = 0x80 | (utf8_char & 0x3F);
>+    utf8_char >>= 6;
>+    buffer[1] = 0x80 | (utf8_char & 0x3F);
>+    utf8_char >>= 6;
>+    buffer[0] = 0xE0 | utf8_char;
>+    return 3;
>+  } else {
>+    buffer[3] = 0x80 | (utf8_char & 0x3F);
>+    utf8_char >>= 6;
>+    buffer[2] = 0x80 | (utf8_char & 0x3F);
>+    utf8_char >>= 6;
>+    buffer[1] = 0x80 | (utf8_char & 0x3F);
>+    utf8_char >>= 6;
>+    buffer[0] = 0xF0 | utf8_char;
>+    return 4;
>+  }
>+}
>+
>+}  // namespace strings_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/utf8.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/utf8.h
>new file mode 100644
>index 00000000000..d2c3c0b0120
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/utf8.h
>@@ -0,0 +1,47 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// UTF8 utilities, implemented to reduce dependencies.
>+//
>+
>+#ifndef ABSL_STRINGS_INTERNAL_UTF8_H_
>+#define ABSL_STRINGS_INTERNAL_UTF8_H_
>+
>+#include <cstddef>
>+#include <cstdint>
>+
>+namespace absl {
>+namespace strings_internal {
>+
>+// For Unicode code points 0 through 0x10FFFF, EncodeUTF8Char writes
>+// out the UTF-8 encoding into buffer, and returns the number of chars
>+// it wrote.
>+//
>+// As described in https://tools.ietf.org/html/rfc3629#section-3 , the encodings
>+// are:
>+//    00 -     7F : 0xxxxxxx
>+//    80 -    7FF : 110xxxxx 10xxxxxx
>+//   800 -   FFFF : 1110xxxx 10xxxxxx 10xxxxxx
>+// 10000 - 10FFFF : 11110xxx 10xxxxxx 10xxxxxx 10xxxxxx
>+//
>+// Values greater than 0x10FFFF are not supported and may or may not write
>+// characters into buffer, however never will more than kMaxEncodedUTF8Size
>+// bytes be written, regardless of the value of utf8_char.
>+enum { kMaxEncodedUTF8Size = 4 };
>+size_t EncodeUTF8Char(char *buffer, char32_t utf8_char);
>+
>+}  // namespace strings_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_STRINGS_INTERNAL_UTF8_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/utf8_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/utf8_test.cc
>new file mode 100644
>index 00000000000..64cec70df7e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/internal/utf8_test.cc
>@@ -0,0 +1,57 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/strings/internal/utf8.h"
>+
>+#include <cstdint>
>+#include <utility>
>+
>+#include "gtest/gtest.h"
>+#include "absl/base/port.h"
>+
>+namespace {
>+
>+TEST(EncodeUTF8Char, BasicFunction) {
>+  std::pair<char32_t, std::string> tests[] = {{0x0030, u8"\u0030"},
>+                                         {0x00A3, u8"\u00A3"},
>+                                         {0x00010000, u8"\U00010000"},
>+                                         {0x0000FFFF, u8"\U0000FFFF"},
>+                                         {0x0010FFFD, u8"\U0010FFFD"}};
>+  for (auto &test : tests) {
>+    char buf0[7] = {'\x00', '\x00', '\x00', '\x00', '\x00', '\x00', '\x00'};
>+    char buf1[7] = {'\xFF', '\xFF', '\xFF', '\xFF', '\xFF', '\xFF', '\xFF'};
>+    char *buf0_written =
>+        &buf0[absl::strings_internal::EncodeUTF8Char(buf0, test.first)];
>+    char *buf1_written =
>+        &buf1[absl::strings_internal::EncodeUTF8Char(buf1, test.first)];
>+    int apparent_length = 7;
>+    while (buf0[apparent_length - 1] == '\x00' &&
>+           buf1[apparent_length - 1] == '\xFF') {
>+      if (--apparent_length == 0) break;
>+    }
>+    EXPECT_EQ(apparent_length, buf0_written - buf0);
>+    EXPECT_EQ(apparent_length, buf1_written - buf1);
>+    EXPECT_EQ(apparent_length, test.second.length());
>+    EXPECT_EQ(std::string(buf0, apparent_length), test.second);
>+    EXPECT_EQ(std::string(buf1, apparent_length), test.second);
>+  }
>+  char buf[32] = "Don't Tread On Me";
>+  EXPECT_LE(absl::strings_internal::EncodeUTF8Char(buf, 0x00110000),
>+            absl::strings_internal::kMaxEncodedUTF8Size);
>+  char buf2[32] = "Negative is invalid but sane";
>+  EXPECT_LE(absl::strings_internal::EncodeUTF8Char(buf2, -1),
>+            absl::strings_internal::kMaxEncodedUTF8Size);
>+}
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/match.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/match.cc
>new file mode 100644
>index 00000000000..25bd7f0b828
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/match.cc
>@@ -0,0 +1,40 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/strings/match.h"
>+
>+#include "absl/strings/internal/memutil.h"
>+
>+namespace absl {
>+
>+namespace {
>+bool CaseEqual(absl::string_view piece1, absl::string_view piece2) {
>+  return (piece1.size() == piece2.size() &&
>+          0 == strings_internal::memcasecmp(piece1.data(), piece2.data(),
>+                                            piece1.size()));
>+  // memcasecmp uses ascii_tolower().
>+}
>+}  // namespace
>+
>+bool StartsWithIgnoreCase(absl::string_view text, absl::string_view prefix) {
>+  return (text.size() >= prefix.size()) &&
>+         CaseEqual(text.substr(0, prefix.size()), prefix);
>+}
>+
>+bool EndsWithIgnoreCase(absl::string_view text, absl::string_view suffix) {
>+  return (text.size() >= suffix.size()) &&
>+         CaseEqual(text.substr(text.size() - suffix.size()), suffix);
>+}
>+
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/match.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/match.h
>new file mode 100644
>index 00000000000..108b6048b04
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/match.h
>@@ -0,0 +1,83 @@
>+//
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// -----------------------------------------------------------------------------
>+// File: match.h
>+// -----------------------------------------------------------------------------
>+//
>+// This file contains simple utilities for performing std::string matching checks.
>+// All of these function parameters are specified as `absl::string_view`,
>+// meaning that these functions can accept `std::string`, `absl::string_view` or
>+// nul-terminated C-style strings.
>+//
>+// Examples:
>+//   std::string s = "foo";
>+//   absl::string_view sv = "f";
>+//   assert(absl::StrContains(s, sv));
>+//
>+// Note: The order of parameters in these functions is designed to mimic the
>+// order an equivalent member function would exhibit;
>+// e.g. `s.Contains(x)` ==> `absl::StrContains(s, x).
>+#ifndef ABSL_STRINGS_MATCH_H_
>+#define ABSL_STRINGS_MATCH_H_
>+
>+#include <cstring>
>+
>+#include "absl/strings/string_view.h"
>+
>+namespace absl {
>+
>+// StrContains()
>+//
>+// Returns whether a given std::string `haystack` contains the substring `needle`.
>+inline bool StrContains(absl::string_view haystack, absl::string_view needle) {
>+  return haystack.find(needle, 0) != haystack.npos;
>+}
>+
>+// StartsWith()
>+//
>+// Returns whether a given std::string `text` begins with `prefix`.
>+inline bool StartsWith(absl::string_view text, absl::string_view prefix) {
>+  return prefix.empty() ||
>+         (text.size() >= prefix.size() &&
>+          memcmp(text.data(), prefix.data(), prefix.size()) == 0);
>+}
>+
>+// EndsWith()
>+//
>+// Returns whether a given std::string `text` ends with `suffix`.
>+inline bool EndsWith(absl::string_view text, absl::string_view suffix) {
>+  return suffix.empty() ||
>+         (text.size() >= suffix.size() &&
>+          memcmp(text.data() + (text.size() - suffix.size()), suffix.data(),
>+                 suffix.size()) == 0
>+         );
>+}
>+
>+// StartsWithIgnoreCase()
>+//
>+// Returns whether a given std::string `text` starts with `starts_with`, ignoring
>+// case in the comparison.
>+bool StartsWithIgnoreCase(absl::string_view text, absl::string_view prefix);
>+
>+// EndsWithIgnoreCase()
>+//
>+// Returns whether a given std::string `text` ends with `ends_with`, ignoring case
>+// in the comparison.
>+bool EndsWithIgnoreCase(absl::string_view text, absl::string_view suffix);
>+
>+}  // namespace absl
>+
>+#endif  // ABSL_STRINGS_MATCH_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/match_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/match_test.cc
>new file mode 100644
>index 00000000000..d194f0e689b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/match_test.cc
>@@ -0,0 +1,99 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/strings/match.h"
>+
>+#include "gtest/gtest.h"
>+
>+namespace {
>+
>+TEST(MatchTest, StartsWith) {
>+  const std::string s1("123" "\0" "456", 7);
>+  const absl::string_view a("foobar");
>+  const absl::string_view b(s1);
>+  const absl::string_view e;
>+  EXPECT_TRUE(absl::StartsWith(a, a));
>+  EXPECT_TRUE(absl::StartsWith(a, "foo"));
>+  EXPECT_TRUE(absl::StartsWith(a, e));
>+  EXPECT_TRUE(absl::StartsWith(b, s1));
>+  EXPECT_TRUE(absl::StartsWith(b, b));
>+  EXPECT_TRUE(absl::StartsWith(b, e));
>+  EXPECT_TRUE(absl::StartsWith(e, ""));
>+  EXPECT_FALSE(absl::StartsWith(a, b));
>+  EXPECT_FALSE(absl::StartsWith(b, a));
>+  EXPECT_FALSE(absl::StartsWith(e, a));
>+}
>+
>+TEST(MatchTest, EndsWith) {
>+  const std::string s1("123" "\0" "456", 7);
>+  const absl::string_view a("foobar");
>+  const absl::string_view b(s1);
>+  const absl::string_view e;
>+  EXPECT_TRUE(absl::EndsWith(a, a));
>+  EXPECT_TRUE(absl::EndsWith(a, "bar"));
>+  EXPECT_TRUE(absl::EndsWith(a, e));
>+  EXPECT_TRUE(absl::EndsWith(b, s1));
>+  EXPECT_TRUE(absl::EndsWith(b, b));
>+  EXPECT_TRUE(absl::EndsWith(b, e));
>+  EXPECT_TRUE(absl::EndsWith(e, ""));
>+  EXPECT_FALSE(absl::EndsWith(a, b));
>+  EXPECT_FALSE(absl::EndsWith(b, a));
>+  EXPECT_FALSE(absl::EndsWith(e, a));
>+}
>+
>+TEST(MatchTest, Contains) {
>+  absl::string_view a("abcdefg");
>+  absl::string_view b("abcd");
>+  absl::string_view c("efg");
>+  absl::string_view d("gh");
>+  EXPECT_TRUE(absl::StrContains(a, a));
>+  EXPECT_TRUE(absl::StrContains(a, b));
>+  EXPECT_TRUE(absl::StrContains(a, c));
>+  EXPECT_FALSE(absl::StrContains(a, d));
>+  EXPECT_TRUE(absl::StrContains("", ""));
>+  EXPECT_TRUE(absl::StrContains("abc", ""));
>+  EXPECT_FALSE(absl::StrContains("", "a"));
>+}
>+
>+TEST(MatchTest, ContainsNull) {
>+  const std::string s = "foo";
>+  const char* cs = "foo";
>+  const absl::string_view sv("foo");
>+  const absl::string_view sv2("foo\0bar", 4);
>+  EXPECT_EQ(s, "foo");
>+  EXPECT_EQ(sv, "foo");
>+  EXPECT_NE(sv2, "foo");
>+  EXPECT_TRUE(absl::EndsWith(s, sv));
>+  EXPECT_TRUE(absl::StartsWith(cs, sv));
>+  EXPECT_TRUE(absl::StrContains(cs, sv));
>+  EXPECT_FALSE(absl::StrContains(cs, sv2));
>+}
>+
>+TEST(MatchTest, StartsWithIgnoreCase) {
>+  EXPECT_TRUE(absl::StartsWithIgnoreCase("foo", "foo"));
>+  EXPECT_TRUE(absl::StartsWithIgnoreCase("foo", "Fo"));
>+  EXPECT_TRUE(absl::StartsWithIgnoreCase("foo", ""));
>+  EXPECT_FALSE(absl::StartsWithIgnoreCase("foo", "fooo"));
>+  EXPECT_FALSE(absl::StartsWithIgnoreCase("", "fo"));
>+}
>+
>+TEST(MatchTest, EndsWithIgnoreCase) {
>+  EXPECT_TRUE(absl::EndsWithIgnoreCase("foo", "foo"));
>+  EXPECT_TRUE(absl::EndsWithIgnoreCase("foo", "Oo"));
>+  EXPECT_TRUE(absl::EndsWithIgnoreCase("foo", ""));
>+  EXPECT_FALSE(absl::EndsWithIgnoreCase("foo", "fooo"));
>+  EXPECT_FALSE(absl::EndsWithIgnoreCase("", "fo"));
>+}
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/numbers.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/numbers.cc
>new file mode 100644
>index 00000000000..f842ed85e9f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/numbers.cc
>@@ -0,0 +1,912 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+// This file contains std::string processing functions related to
>+// numeric values.
>+
>+#include "absl/strings/numbers.h"
>+
>+#include <algorithm>
>+#include <cassert>
>+#include <cfloat>          // for DBL_DIG and FLT_DIG
>+#include <cmath>           // for HUGE_VAL
>+#include <cstdint>
>+#include <cstdio>
>+#include <cstdlib>
>+#include <cstring>
>+#include <iterator>
>+#include <limits>
>+#include <memory>
>+#include <utility>
>+
>+#include "absl/base/internal/raw_logging.h"
>+#include "absl/strings/ascii.h"
>+#include "absl/strings/charconv.h"
>+#include "absl/strings/internal/bits.h"
>+#include "absl/strings/internal/memutil.h"
>+#include "absl/strings/str_cat.h"
>+
>+namespace absl {
>+
>+bool SimpleAtof(absl::string_view str, float* value) {
>+  *value = 0.0;
>+  str = StripAsciiWhitespace(str);
>+  if (!str.empty() && str[0] == '+') {
>+    str.remove_prefix(1);
>+  }
>+  auto result = absl::from_chars(str.data(), str.data() + str.size(), *value);
>+  if (result.ec == std::errc::invalid_argument) {
>+    return false;
>+  }
>+  if (result.ptr != str.data() + str.size()) {
>+    // not all non-whitespace characters consumed
>+    return false;
>+  }
>+  // from_chars() with DR 3801's current wording will return max() on
>+  // overflow.  SimpleAtof returns infinity instead.
>+  if (result.ec == std::errc::result_out_of_range) {
>+    if (*value > 1.0) {
>+      *value = std::numeric_limits<float>::infinity();
>+    } else if (*value < -1.0) {
>+      *value = -std::numeric_limits<float>::infinity();
>+    }
>+  }
>+  return true;
>+}
>+
>+bool SimpleAtod(absl::string_view str, double* value) {
>+  *value = 0.0;
>+  str = StripAsciiWhitespace(str);
>+  if (!str.empty() && str[0] == '+') {
>+    str.remove_prefix(1);
>+  }
>+  auto result = absl::from_chars(str.data(), str.data() + str.size(), *value);
>+  if (result.ec == std::errc::invalid_argument) {
>+    return false;
>+  }
>+  if (result.ptr != str.data() + str.size()) {
>+    // not all non-whitespace characters consumed
>+    return false;
>+  }
>+  // from_chars() with DR 3801's current wording will return max() on
>+  // overflow.  SimpleAtod returns infinity instead.
>+  if (result.ec == std::errc::result_out_of_range) {
>+    if (*value > 1.0) {
>+      *value = std::numeric_limits<double>::infinity();
>+    } else if (*value < -1.0) {
>+      *value = -std::numeric_limits<double>::infinity();
>+    }
>+  }
>+  return true;
>+}
>+
>+namespace {
>+
>+// TODO(rogeeff): replace with the real released thing once we figure out what
>+// it is.
>+inline bool CaseEqual(absl::string_view piece1, absl::string_view piece2) {
>+  return (piece1.size() == piece2.size() &&
>+          0 == strings_internal::memcasecmp(piece1.data(), piece2.data(),
>+                                            piece1.size()));
>+}
>+
>+// Writes a two-character representation of 'i' to 'buf'. 'i' must be in the
>+// range 0 <= i < 100, and buf must have space for two characters. Example:
>+//   char buf[2];
>+//   PutTwoDigits(42, buf);
>+//   // buf[0] == '4'
>+//   // buf[1] == '2'
>+inline void PutTwoDigits(size_t i, char* buf) {
>+  static const char two_ASCII_digits[100][2] = {
>+    {'0', '0'}, {'0', '1'}, {'0', '2'}, {'0', '3'}, {'0', '4'},
>+    {'0', '5'}, {'0', '6'}, {'0', '7'}, {'0', '8'}, {'0', '9'},
>+    {'1', '0'}, {'1', '1'}, {'1', '2'}, {'1', '3'}, {'1', '4'},
>+    {'1', '5'}, {'1', '6'}, {'1', '7'}, {'1', '8'}, {'1', '9'},
>+    {'2', '0'}, {'2', '1'}, {'2', '2'}, {'2', '3'}, {'2', '4'},
>+    {'2', '5'}, {'2', '6'}, {'2', '7'}, {'2', '8'}, {'2', '9'},
>+    {'3', '0'}, {'3', '1'}, {'3', '2'}, {'3', '3'}, {'3', '4'},
>+    {'3', '5'}, {'3', '6'}, {'3', '7'}, {'3', '8'}, {'3', '9'},
>+    {'4', '0'}, {'4', '1'}, {'4', '2'}, {'4', '3'}, {'4', '4'},
>+    {'4', '5'}, {'4', '6'}, {'4', '7'}, {'4', '8'}, {'4', '9'},
>+    {'5', '0'}, {'5', '1'}, {'5', '2'}, {'5', '3'}, {'5', '4'},
>+    {'5', '5'}, {'5', '6'}, {'5', '7'}, {'5', '8'}, {'5', '9'},
>+    {'6', '0'}, {'6', '1'}, {'6', '2'}, {'6', '3'}, {'6', '4'},
>+    {'6', '5'}, {'6', '6'}, {'6', '7'}, {'6', '8'}, {'6', '9'},
>+    {'7', '0'}, {'7', '1'}, {'7', '2'}, {'7', '3'}, {'7', '4'},
>+    {'7', '5'}, {'7', '6'}, {'7', '7'}, {'7', '8'}, {'7', '9'},
>+    {'8', '0'}, {'8', '1'}, {'8', '2'}, {'8', '3'}, {'8', '4'},
>+    {'8', '5'}, {'8', '6'}, {'8', '7'}, {'8', '8'}, {'8', '9'},
>+    {'9', '0'}, {'9', '1'}, {'9', '2'}, {'9', '3'}, {'9', '4'},
>+    {'9', '5'}, {'9', '6'}, {'9', '7'}, {'9', '8'}, {'9', '9'}
>+  };
>+  assert(i < 100);
>+  memcpy(buf, two_ASCII_digits[i], 2);
>+}
>+
>+}  // namespace
>+
>+bool SimpleAtob(absl::string_view str, bool* value) {
>+  ABSL_RAW_CHECK(value != nullptr, "Output pointer must not be nullptr.");
>+  if (CaseEqual(str, "true") || CaseEqual(str, "t") ||
>+      CaseEqual(str, "yes") || CaseEqual(str, "y") ||
>+      CaseEqual(str, "1")) {
>+    *value = true;
>+    return true;
>+  }
>+  if (CaseEqual(str, "false") || CaseEqual(str, "f") ||
>+      CaseEqual(str, "no") || CaseEqual(str, "n") ||
>+      CaseEqual(str, "0")) {
>+    *value = false;
>+    return true;
>+  }
>+  return false;
>+}
>+
>+// ----------------------------------------------------------------------
>+// FastIntToBuffer() overloads
>+//
>+// Like the Fast*ToBuffer() functions above, these are intended for speed.
>+// Unlike the Fast*ToBuffer() functions, however, these functions write
>+// their output to the beginning of the buffer.  The caller is responsible
>+// for ensuring that the buffer has enough space to hold the output.
>+//
>+// Returns a pointer to the end of the std::string (i.e. the null character
>+// terminating the std::string).
>+// ----------------------------------------------------------------------
>+
>+namespace {
>+
>+// Used to optimize printing a decimal number's final digit.
>+const char one_ASCII_final_digits[10][2] {
>+  {'0', 0}, {'1', 0}, {'2', 0}, {'3', 0}, {'4', 0},
>+  {'5', 0}, {'6', 0}, {'7', 0}, {'8', 0}, {'9', 0},
>+};
>+
>+}  // namespace
>+
>+char* numbers_internal::FastIntToBuffer(uint32_t i, char* buffer) {
>+  uint32_t digits;
>+  // The idea of this implementation is to trim the number of divides to as few
>+  // as possible, and also reducing memory stores and branches, by going in
>+  // steps of two digits at a time rather than one whenever possible.
>+  // The huge-number case is first, in the hopes that the compiler will output
>+  // that case in one branch-free block of code, and only output conditional
>+  // branches into it from below.
>+  if (i >= 1000000000) {     // >= 1,000,000,000
>+    digits = i / 100000000;  //      100,000,000
>+    i -= digits * 100000000;
>+    PutTwoDigits(digits, buffer);
>+    buffer += 2;
>+  lt100_000_000:
>+    digits = i / 1000000;  // 1,000,000
>+    i -= digits * 1000000;
>+    PutTwoDigits(digits, buffer);
>+    buffer += 2;
>+  lt1_000_000:
>+    digits = i / 10000;  // 10,000
>+    i -= digits * 10000;
>+    PutTwoDigits(digits, buffer);
>+    buffer += 2;
>+  lt10_000:
>+    digits = i / 100;
>+    i -= digits * 100;
>+    PutTwoDigits(digits, buffer);
>+    buffer += 2;
>+ lt100:
>+    digits = i;
>+    PutTwoDigits(digits, buffer);
>+    buffer += 2;
>+    *buffer = 0;
>+    return buffer;
>+  }
>+
>+  if (i < 100) {
>+    digits = i;
>+    if (i >= 10) goto lt100;
>+    memcpy(buffer, one_ASCII_final_digits[i], 2);
>+    return buffer + 1;
>+  }
>+  if (i < 10000) {  //    10,000
>+    if (i >= 1000) goto lt10_000;
>+    digits = i / 100;
>+    i -= digits * 100;
>+    *buffer++ = '0' + digits;
>+    goto lt100;
>+  }
>+  if (i < 1000000) {  //    1,000,000
>+    if (i >= 100000) goto lt1_000_000;
>+    digits = i / 10000;  //    10,000
>+    i -= digits * 10000;
>+    *buffer++ = '0' + digits;
>+    goto lt10_000;
>+  }
>+  if (i < 100000000) {  //    100,000,000
>+    if (i >= 10000000) goto lt100_000_000;
>+    digits = i / 1000000;  //   1,000,000
>+    i -= digits * 1000000;
>+    *buffer++ = '0' + digits;
>+    goto lt1_000_000;
>+  }
>+  // we already know that i < 1,000,000,000
>+  digits = i / 100000000;  //   100,000,000
>+  i -= digits * 100000000;
>+  *buffer++ = '0' + digits;
>+  goto lt100_000_000;
>+}
>+
>+char* numbers_internal::FastIntToBuffer(int32_t i, char* buffer) {
>+  uint32_t u = i;
>+  if (i < 0) {
>+    *buffer++ = '-';
>+    // We need to do the negation in modular (i.e., "unsigned")
>+    // arithmetic; MSVC++ apprently warns for plain "-u", so
>+    // we write the equivalent expression "0 - u" instead.
>+    u = 0 - u;
>+  }
>+  return numbers_internal::FastIntToBuffer(u, buffer);
>+}
>+
>+char* numbers_internal::FastIntToBuffer(uint64_t i, char* buffer) {
>+  uint32_t u32 = static_cast<uint32_t>(i);
>+  if (u32 == i) return numbers_internal::FastIntToBuffer(u32, buffer);
>+
>+  // Here we know i has at least 10 decimal digits.
>+  uint64_t top_1to11 = i / 1000000000;
>+  u32 = static_cast<uint32_t>(i - top_1to11 * 1000000000);
>+  uint32_t top_1to11_32 = static_cast<uint32_t>(top_1to11);
>+
>+  if (top_1to11_32 == top_1to11) {
>+    buffer = numbers_internal::FastIntToBuffer(top_1to11_32, buffer);
>+  } else {
>+    // top_1to11 has more than 32 bits too; print it in two steps.
>+    uint32_t top_8to9 = static_cast<uint32_t>(top_1to11 / 100);
>+    uint32_t mid_2 = static_cast<uint32_t>(top_1to11 - top_8to9 * 100);
>+    buffer = numbers_internal::FastIntToBuffer(top_8to9, buffer);
>+    PutTwoDigits(mid_2, buffer);
>+    buffer += 2;
>+  }
>+
>+  // We have only 9 digits now, again the maximum uint32_t can handle fully.
>+  uint32_t digits = u32 / 10000000;  // 10,000,000
>+  u32 -= digits * 10000000;
>+  PutTwoDigits(digits, buffer);
>+  buffer += 2;
>+  digits = u32 / 100000;  // 100,000
>+  u32 -= digits * 100000;
>+  PutTwoDigits(digits, buffer);
>+  buffer += 2;
>+  digits = u32 / 1000;  // 1,000
>+  u32 -= digits * 1000;
>+  PutTwoDigits(digits, buffer);
>+  buffer += 2;
>+  digits = u32 / 10;
>+  u32 -= digits * 10;
>+  PutTwoDigits(digits, buffer);
>+  buffer += 2;
>+  memcpy(buffer, one_ASCII_final_digits[u32], 2);
>+  return buffer + 1;
>+}
>+
>+char* numbers_internal::FastIntToBuffer(int64_t i, char* buffer) {
>+  uint64_t u = i;
>+  if (i < 0) {
>+    *buffer++ = '-';
>+    u = 0 - u;
>+  }
>+  return numbers_internal::FastIntToBuffer(u, buffer);
>+}
>+
>+// Given a 128-bit number expressed as a pair of uint64_t, high half first,
>+// return that number multiplied by the given 32-bit value.  If the result is
>+// too large to fit in a 128-bit number, divide it by 2 until it fits.
>+static std::pair<uint64_t, uint64_t> Mul32(std::pair<uint64_t, uint64_t> num,
>+                                           uint32_t mul) {
>+  uint64_t bits0_31 = num.second & 0xFFFFFFFF;
>+  uint64_t bits32_63 = num.second >> 32;
>+  uint64_t bits64_95 = num.first & 0xFFFFFFFF;
>+  uint64_t bits96_127 = num.first >> 32;
>+
>+  // The picture so far: each of these 64-bit values has only the lower 32 bits
>+  // filled in.
>+  // bits96_127:          [ 00000000 xxxxxxxx ]
>+  // bits64_95:                    [ 00000000 xxxxxxxx ]
>+  // bits32_63:                             [ 00000000 xxxxxxxx ]
>+  // bits0_31:                                       [ 00000000 xxxxxxxx ]
>+
>+  bits0_31 *= mul;
>+  bits32_63 *= mul;
>+  bits64_95 *= mul;
>+  bits96_127 *= mul;
>+
>+  // Now the top halves may also have value, though all 64 of their bits will
>+  // never be set at the same time, since they are a result of a 32x32 bit
>+  // multiply.  This makes the carry calculation slightly easier.
>+  // bits96_127:          [ mmmmmmmm | mmmmmmmm ]
>+  // bits64_95:                    [ | mmmmmmmm mmmmmmmm | ]
>+  // bits32_63:                      |        [ mmmmmmmm | mmmmmmmm ]
>+  // bits0_31:                       |                 [ | mmmmmmmm mmmmmmmm ]
>+  // eventually:        [ bits128_up | ...bits64_127.... | ..bits0_63... ]
>+
>+  uint64_t bits0_63 = bits0_31 + (bits32_63 << 32);
>+  uint64_t bits64_127 = bits64_95 + (bits96_127 << 32) + (bits32_63 >> 32) +
>+                        (bits0_63 < bits0_31);
>+  uint64_t bits128_up = (bits96_127 >> 32) + (bits64_127 < bits64_95);
>+  if (bits128_up == 0) return {bits64_127, bits0_63};
>+
>+  int shift = 64 - strings_internal::CountLeadingZeros64(bits128_up);
>+  uint64_t lo = (bits0_63 >> shift) + (bits64_127 << (64 - shift));
>+  uint64_t hi = (bits64_127 >> shift) + (bits128_up << (64 - shift));
>+  return {hi, lo};
>+}
>+
>+// Compute num * 5 ^ expfive, and return the first 128 bits of the result,
>+// where the first bit is always a one.  So PowFive(1, 0) starts 0b100000,
>+// PowFive(1, 1) starts 0b101000, PowFive(1, 2) starts 0b110010, etc.
>+static std::pair<uint64_t, uint64_t> PowFive(uint64_t num, int expfive) {
>+  std::pair<uint64_t, uint64_t> result = {num, 0};
>+  while (expfive >= 13) {
>+    // 5^13 is the highest power of five that will fit in a 32-bit integer.
>+    result = Mul32(result, 5 * 5 * 5 * 5 * 5 * 5 * 5 * 5 * 5 * 5 * 5 * 5 * 5);
>+    expfive -= 13;
>+  }
>+  constexpr int powers_of_five[13] = {
>+      1,
>+      5,
>+      5 * 5,
>+      5 * 5 * 5,
>+      5 * 5 * 5 * 5,
>+      5 * 5 * 5 * 5 * 5,
>+      5 * 5 * 5 * 5 * 5 * 5,
>+      5 * 5 * 5 * 5 * 5 * 5 * 5,
>+      5 * 5 * 5 * 5 * 5 * 5 * 5 * 5,
>+      5 * 5 * 5 * 5 * 5 * 5 * 5 * 5 * 5,
>+      5 * 5 * 5 * 5 * 5 * 5 * 5 * 5 * 5 * 5,
>+      5 * 5 * 5 * 5 * 5 * 5 * 5 * 5 * 5 * 5 * 5,
>+      5 * 5 * 5 * 5 * 5 * 5 * 5 * 5 * 5 * 5 * 5 * 5};
>+  result = Mul32(result, powers_of_five[expfive & 15]);
>+  int shift = strings_internal::CountLeadingZeros64(result.first);
>+  if (shift != 0) {
>+    result.first = (result.first << shift) + (result.second >> (64 - shift));
>+    result.second = (result.second << shift);
>+  }
>+  return result;
>+}
>+
>+struct ExpDigits {
>+  int32_t exponent;
>+  char digits[6];
>+};
>+
>+// SplitToSix converts value, a positive double-precision floating-point number,
>+// into a base-10 exponent and 6 ASCII digits, where the first digit is never
>+// zero.  For example, SplitToSix(1) returns an exponent of zero and a digits
>+// array of {'1', '0', '0', '0', '0', '0'}.  If value is exactly halfway between
>+// two possible representations, e.g. value = 100000.5, then "round to even" is
>+// performed.
>+static ExpDigits SplitToSix(const double value) {
>+  ExpDigits exp_dig;
>+  int exp = 5;
>+  double d = value;
>+  // First step: calculate a close approximation of the output, where the
>+  // value d will be between 100,000 and 999,999, representing the digits
>+  // in the output ASCII array, and exp is the base-10 exponent.  It would be
>+  // faster to use a table here, and to look up the base-2 exponent of value,
>+  // however value is an IEEE-754 64-bit number, so the table would have 2,000
>+  // entries, which is not cache-friendly.
>+  if (d >= 999999.5) {
>+    if (d >= 1e+261) exp += 256, d *= 1e-256;
>+    if (d >= 1e+133) exp += 128, d *= 1e-128;
>+    if (d >= 1e+69) exp += 64, d *= 1e-64;
>+    if (d >= 1e+37) exp += 32, d *= 1e-32;
>+    if (d >= 1e+21) exp += 16, d *= 1e-16;
>+    if (d >= 1e+13) exp += 8, d *= 1e-8;
>+    if (d >= 1e+9) exp += 4, d *= 1e-4;
>+    if (d >= 1e+7) exp += 2, d *= 1e-2;
>+    if (d >= 1e+6) exp += 1, d *= 1e-1;
>+  } else {
>+    if (d < 1e-250) exp -= 256, d *= 1e256;
>+    if (d < 1e-122) exp -= 128, d *= 1e128;
>+    if (d < 1e-58) exp -= 64, d *= 1e64;
>+    if (d < 1e-26) exp -= 32, d *= 1e32;
>+    if (d < 1e-10) exp -= 16, d *= 1e16;
>+    if (d < 1e-2) exp -= 8, d *= 1e8;
>+    if (d < 1e+2) exp -= 4, d *= 1e4;
>+    if (d < 1e+4) exp -= 2, d *= 1e2;
>+    if (d < 1e+5) exp -= 1, d *= 1e1;
>+  }
>+  // At this point, d is in the range [99999.5..999999.5) and exp is in the
>+  // range [-324..308]. Since we need to round d up, we want to add a half
>+  // and truncate.
>+  // However, the technique above may have lost some precision, due to its
>+  // repeated multiplication by constants that each may be off by half a bit
>+  // of precision.  This only matters if we're close to the edge though.
>+  // Since we'd like to know if the fractional part of d is close to a half,
>+  // we multiply it by 65536 and see if the fractional part is close to 32768.
>+  // (The number doesn't have to be a power of two,but powers of two are faster)
>+  uint64_t d64k = d * 65536;
>+  int dddddd;  // A 6-digit decimal integer.
>+  if ((d64k % 65536) == 32767 || (d64k % 65536) == 32768) {
>+    // OK, it's fairly likely that precision was lost above, which is
>+    // not a surprise given only 52 mantissa bits are available.  Therefore
>+    // redo the calculation using 128-bit numbers.  (64 bits are not enough).
>+
>+    // Start out with digits rounded down; maybe add one below.
>+    dddddd = static_cast<int>(d64k / 65536);
>+
>+    // mantissa is a 64-bit integer representing M.mmm... * 2^63.  The actual
>+    // value we're representing, of course, is M.mmm... * 2^exp2.
>+    int exp2;
>+    double m = std::frexp(value, &exp2);
>+    uint64_t mantissa = m * (32768.0 * 65536.0 * 65536.0 * 65536.0);
>+    // std::frexp returns an m value in the range [0.5, 1.0), however we
>+    // can't multiply it by 2^64 and convert to an integer because some FPUs
>+    // throw an exception when converting an number higher than 2^63 into an
>+    // integer - even an unsigned 64-bit integer!  Fortunately it doesn't matter
>+    // since m only has 52 significant bits anyway.
>+    mantissa <<= 1;
>+    exp2 -= 64;  // not needed, but nice for debugging
>+
>+    // OK, we are here to compare:
>+    //     (dddddd + 0.5) * 10^(exp-5)  vs.  mantissa * 2^exp2
>+    // so we can round up dddddd if appropriate.  Those values span the full
>+    // range of 600 orders of magnitude of IEE 64-bit floating-point.
>+    // Fortunately, we already know they are very close, so we don't need to
>+    // track the base-2 exponent of both sides.  This greatly simplifies the
>+    // the math since the 2^exp2 calculation is unnecessary and the power-of-10
>+    // calculation can become a power-of-5 instead.
>+
>+    std::pair<uint64_t, uint64_t> edge, val;
>+    if (exp >= 6) {
>+      // Compare (dddddd + 0.5) * 5 ^ (exp - 5) to mantissa
>+      // Since we're tossing powers of two, 2 * dddddd + 1 is the
>+      // same as dddddd + 0.5
>+      edge = PowFive(2 * dddddd + 1, exp - 5);
>+
>+      val.first = mantissa;
>+      val.second = 0;
>+    } else {
>+      // We can't compare (dddddd + 0.5) * 5 ^ (exp - 5) to mantissa as we did
>+      // above because (exp - 5) is negative.  So we compare (dddddd + 0.5) to
>+      // mantissa * 5 ^ (5 - exp)
>+      edge = PowFive(2 * dddddd + 1, 0);
>+
>+      val = PowFive(mantissa, 5 - exp);
>+    }
>+    // printf("exp=%d %016lx %016lx vs %016lx %016lx\n", exp, val.first,
>+    //        val.second, edge.first, edge.second);
>+    if (val > edge) {
>+      dddddd++;
>+    } else if (val == edge) {
>+      dddddd += (dddddd & 1);
>+    }
>+  } else {
>+    // Here, we are not close to the edge.
>+    dddddd = static_cast<int>((d64k + 32768) / 65536);
>+  }
>+  if (dddddd == 1000000) {
>+    dddddd = 100000;
>+    exp += 1;
>+  }
>+  exp_dig.exponent = exp;
>+
>+  int two_digits = dddddd / 10000;
>+  dddddd -= two_digits * 10000;
>+  PutTwoDigits(two_digits, &exp_dig.digits[0]);
>+
>+  two_digits = dddddd / 100;
>+  dddddd -= two_digits * 100;
>+  PutTwoDigits(two_digits, &exp_dig.digits[2]);
>+
>+  PutTwoDigits(dddddd, &exp_dig.digits[4]);
>+  return exp_dig;
>+}
>+
>+// Helper function for fast formatting of floating-point.
>+// The result is the same as "%g", a.k.a. "%.6g".
>+size_t numbers_internal::SixDigitsToBuffer(double d, char* const buffer) {
>+  static_assert(std::numeric_limits<float>::is_iec559,
>+                "IEEE-754/IEC-559 support only");
>+
>+  char* out = buffer;  // we write data to out, incrementing as we go, but
>+                       // FloatToBuffer always returns the address of the buffer
>+                       // passed in.
>+
>+  if (std::isnan(d)) {
>+    strcpy(out, "nan");  // NOLINT(runtime/printf)
>+    return 3;
>+  }
>+  if (d == 0) {  // +0 and -0 are handled here
>+    if (std::signbit(d)) *out++ = '-';
>+    *out++ = '0';
>+    *out = 0;
>+    return out - buffer;
>+  }
>+  if (d < 0) {
>+    *out++ = '-';
>+    d = -d;
>+  }
>+  if (std::isinf(d)) {
>+    strcpy(out, "inf");  // NOLINT(runtime/printf)
>+    return out + 3 - buffer;
>+  }
>+
>+  auto exp_dig = SplitToSix(d);
>+  int exp = exp_dig.exponent;
>+  const char* digits = exp_dig.digits;
>+  out[0] = '0';
>+  out[1] = '.';
>+  switch (exp) {
>+    case 5:
>+      memcpy(out, &digits[0], 6), out += 6;
>+      *out = 0;
>+      return out - buffer;
>+    case 4:
>+      memcpy(out, &digits[0], 5), out += 5;
>+      if (digits[5] != '0') {
>+        *out++ = '.';
>+        *out++ = digits[5];
>+      }
>+      *out = 0;
>+      return out - buffer;
>+    case 3:
>+      memcpy(out, &digits[0], 4), out += 4;
>+      if ((digits[5] | digits[4]) != '0') {
>+        *out++ = '.';
>+        *out++ = digits[4];
>+        if (digits[5] != '0') *out++ = digits[5];
>+      }
>+      *out = 0;
>+      return out - buffer;
>+    case 2:
>+      memcpy(out, &digits[0], 3), out += 3;
>+      *out++ = '.';
>+      memcpy(out, &digits[3], 3);
>+      out += 3;
>+      while (out[-1] == '0') --out;
>+      if (out[-1] == '.') --out;
>+      *out = 0;
>+      return out - buffer;
>+    case 1:
>+      memcpy(out, &digits[0], 2), out += 2;
>+      *out++ = '.';
>+      memcpy(out, &digits[2], 4);
>+      out += 4;
>+      while (out[-1] == '0') --out;
>+      if (out[-1] == '.') --out;
>+      *out = 0;
>+      return out - buffer;
>+    case 0:
>+      memcpy(out, &digits[0], 1), out += 1;
>+      *out++ = '.';
>+      memcpy(out, &digits[1], 5);
>+      out += 5;
>+      while (out[-1] == '0') --out;
>+      if (out[-1] == '.') --out;
>+      *out = 0;
>+      return out - buffer;
>+    case -4:
>+      out[2] = '0';
>+      ++out;
>+      ABSL_FALLTHROUGH_INTENDED;
>+    case -3:
>+      out[2] = '0';
>+      ++out;
>+      ABSL_FALLTHROUGH_INTENDED;
>+    case -2:
>+      out[2] = '0';
>+      ++out;
>+      ABSL_FALLTHROUGH_INTENDED;
>+    case -1:
>+      out += 2;
>+      memcpy(out, &digits[0], 6);
>+      out += 6;
>+      while (out[-1] == '0') --out;
>+      *out = 0;
>+      return out - buffer;
>+  }
>+  assert(exp < -4 || exp >= 6);
>+  out[0] = digits[0];
>+  assert(out[1] == '.');
>+  out += 2;
>+  memcpy(out, &digits[1], 5), out += 5;
>+  while (out[-1] == '0') --out;
>+  if (out[-1] == '.') --out;
>+  *out++ = 'e';
>+  if (exp > 0) {
>+    *out++ = '+';
>+  } else {
>+    *out++ = '-';
>+    exp = -exp;
>+  }
>+  if (exp > 99) {
>+    int dig1 = exp / 100;
>+    exp -= dig1 * 100;
>+    *out++ = '0' + dig1;
>+  }
>+  PutTwoDigits(exp, out);
>+  out += 2;
>+  *out = 0;
>+  return out - buffer;
>+}
>+
>+namespace {
>+// Represents integer values of digits.
>+// Uses 36 to indicate an invalid character since we support
>+// bases up to 36.
>+static const int8_t kAsciiToInt[256] = {
>+    36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36,  // 16 36s.
>+    36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36,
>+    36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 0,  1,  2,  3,  4,  5,
>+    6,  7,  8,  9,  36, 36, 36, 36, 36, 36, 36, 10, 11, 12, 13, 14, 15, 16, 17,
>+    18, 19, 20, 21, 22, 23, 24, 25, 26, 27, 28, 29, 30, 31, 32, 33, 34, 35, 36,
>+    36, 36, 36, 36, 36, 10, 11, 12, 13, 14, 15, 16, 17, 18, 19, 20, 21, 22, 23,
>+    24, 25, 26, 27, 28, 29, 30, 31, 32, 33, 34, 35, 36, 36, 36, 36, 36, 36, 36,
>+    36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36,
>+    36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36,
>+    36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36,
>+    36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36,
>+    36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36,
>+    36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36,
>+    36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36, 36};
>+
>+// Parse the sign and optional hex or oct prefix in text.
>+inline bool safe_parse_sign_and_base(absl::string_view* text /*inout*/,
>+                                     int* base_ptr /*inout*/,
>+                                     bool* negative_ptr /*output*/) {
>+  if (text->data() == nullptr) {
>+    return false;
>+  }
>+
>+  const char* start = text->data();
>+  const char* end = start + text->size();
>+  int base = *base_ptr;
>+
>+  // Consume whitespace.
>+  while (start < end && absl::ascii_isspace(start[0])) {
>+    ++start;
>+  }
>+  while (start < end && absl::ascii_isspace(end[-1])) {
>+    --end;
>+  }
>+  if (start >= end) {
>+    return false;
>+  }
>+
>+  // Consume sign.
>+  *negative_ptr = (start[0] == '-');
>+  if (*negative_ptr || start[0] == '+') {
>+    ++start;
>+    if (start >= end) {
>+      return false;
>+    }
>+  }
>+
>+  // Consume base-dependent prefix.
>+  //  base 0: "0x" -> base 16, "0" -> base 8, default -> base 10
>+  //  base 16: "0x" -> base 16
>+  // Also validate the base.
>+  if (base == 0) {
>+    if (end - start >= 2 && start[0] == '0' &&
>+        (start[1] == 'x' || start[1] == 'X')) {
>+      base = 16;
>+      start += 2;
>+      if (start >= end) {
>+        // "0x" with no digits after is invalid.
>+        return false;
>+      }
>+    } else if (end - start >= 1 && start[0] == '0') {
>+      base = 8;
>+      start += 1;
>+    } else {
>+      base = 10;
>+    }
>+  } else if (base == 16) {
>+    if (end - start >= 2 && start[0] == '0' &&
>+        (start[1] == 'x' || start[1] == 'X')) {
>+      start += 2;
>+      if (start >= end) {
>+        // "0x" with no digits after is invalid.
>+        return false;
>+      }
>+    }
>+  } else if (base >= 2 && base <= 36) {
>+    // okay
>+  } else {
>+    return false;
>+  }
>+  *text = absl::string_view(start, end - start);
>+  *base_ptr = base;
>+  return true;
>+}
>+
>+// Consume digits.
>+//
>+// The classic loop:
>+//
>+//   for each digit
>+//     value = value * base + digit
>+//   value *= sign
>+//
>+// The classic loop needs overflow checking.  It also fails on the most
>+// negative integer, -2147483648 in 32-bit two's complement representation.
>+//
>+// My improved loop:
>+//
>+//  if (!negative)
>+//    for each digit
>+//      value = value * base
>+//      value = value + digit
>+//  else
>+//    for each digit
>+//      value = value * base
>+//      value = value - digit
>+//
>+// Overflow checking becomes simple.
>+
>+// Lookup tables per IntType:
>+// vmax/base and vmin/base are precomputed because division costs at least 8ns.
>+// TODO(junyer): Doing this per base instead (i.e. an array of structs, not a
>+// struct of arrays) would probably be better in terms of d-cache for the most
>+// commonly used bases.
>+template <typename IntType>
>+struct LookupTables {
>+  static const IntType kVmaxOverBase[];
>+  static const IntType kVminOverBase[];
>+};
>+
>+// An array initializer macro for X/base where base in [0, 36].
>+// However, note that lookups for base in [0, 1] should never happen because
>+// base has been validated to be in [2, 36] by safe_parse_sign_and_base().
>+#define X_OVER_BASE_INITIALIZER(X)                                        \
>+  {                                                                       \
>+    0, 0, X / 2, X / 3, X / 4, X / 5, X / 6, X / 7, X / 8, X / 9, X / 10, \
>+        X / 11, X / 12, X / 13, X / 14, X / 15, X / 16, X / 17, X / 18,   \
>+        X / 19, X / 20, X / 21, X / 22, X / 23, X / 24, X / 25, X / 26,   \
>+        X / 27, X / 28, X / 29, X / 30, X / 31, X / 32, X / 33, X / 34,   \
>+        X / 35, X / 36,                                                   \
>+  }
>+
>+template <typename IntType>
>+const IntType LookupTables<IntType>::kVmaxOverBase[] =
>+    X_OVER_BASE_INITIALIZER(std::numeric_limits<IntType>::max());
>+
>+template <typename IntType>
>+const IntType LookupTables<IntType>::kVminOverBase[] =
>+    X_OVER_BASE_INITIALIZER(std::numeric_limits<IntType>::min());
>+
>+#undef X_OVER_BASE_INITIALIZER
>+
>+template <typename IntType>
>+inline bool safe_parse_positive_int(absl::string_view text, int base,
>+                                    IntType* value_p) {
>+  IntType value = 0;
>+  const IntType vmax = std::numeric_limits<IntType>::max();
>+  assert(vmax > 0);
>+  assert(base >= 0);
>+  assert(vmax >= static_cast<IntType>(base));
>+  const IntType vmax_over_base = LookupTables<IntType>::kVmaxOverBase[base];
>+  const char* start = text.data();
>+  const char* end = start + text.size();
>+  // loop over digits
>+  for (; start < end; ++start) {
>+    unsigned char c = static_cast<unsigned char>(start[0]);
>+    int digit = kAsciiToInt[c];
>+    if (digit >= base) {
>+      *value_p = value;
>+      return false;
>+    }
>+    if (value > vmax_over_base) {
>+      *value_p = vmax;
>+      return false;
>+    }
>+    value *= base;
>+    if (value > vmax - digit) {
>+      *value_p = vmax;
>+      return false;
>+    }
>+    value += digit;
>+  }
>+  *value_p = value;
>+  return true;
>+}
>+
>+template <typename IntType>
>+inline bool safe_parse_negative_int(absl::string_view text, int base,
>+                                    IntType* value_p) {
>+  IntType value = 0;
>+  const IntType vmin = std::numeric_limits<IntType>::min();
>+  assert(vmin < 0);
>+  assert(vmin <= 0 - base);
>+  IntType vmin_over_base = LookupTables<IntType>::kVminOverBase[base];
>+  // 2003 c++ standard [expr.mul]
>+  // "... the sign of the remainder is implementation-defined."
>+  // Although (vmin/base)*base + vmin%base is always vmin.
>+  // 2011 c++ standard tightens the spec but we cannot rely on it.
>+  // TODO(junyer): Handle this in the lookup table generation.
>+  if (vmin % base > 0) {
>+    vmin_over_base += 1;
>+  }
>+  const char* start = text.data();
>+  const char* end = start + text.size();
>+  // loop over digits
>+  for (; start < end; ++start) {
>+    unsigned char c = static_cast<unsigned char>(start[0]);
>+    int digit = kAsciiToInt[c];
>+    if (digit >= base) {
>+      *value_p = value;
>+      return false;
>+    }
>+    if (value < vmin_over_base) {
>+      *value_p = vmin;
>+      return false;
>+    }
>+    value *= base;
>+    if (value < vmin + digit) {
>+      *value_p = vmin;
>+      return false;
>+    }
>+    value -= digit;
>+  }
>+  *value_p = value;
>+  return true;
>+}
>+
>+// Input format based on POSIX.1-2008 strtol
>+// http://pubs.opengroup.org/onlinepubs/9699919799/functions/strtol.html
>+template <typename IntType>
>+inline bool safe_int_internal(absl::string_view text, IntType* value_p,
>+                              int base) {
>+  *value_p = 0;
>+  bool negative;
>+  if (!safe_parse_sign_and_base(&text, &base, &negative)) {
>+    return false;
>+  }
>+  if (!negative) {
>+    return safe_parse_positive_int(text, base, value_p);
>+  } else {
>+    return safe_parse_negative_int(text, base, value_p);
>+  }
>+}
>+
>+template <typename IntType>
>+inline bool safe_uint_internal(absl::string_view text, IntType* value_p,
>+                               int base) {
>+  *value_p = 0;
>+  bool negative;
>+  if (!safe_parse_sign_and_base(&text, &base, &negative) || negative) {
>+    return false;
>+  }
>+  return safe_parse_positive_int(text, base, value_p);
>+}
>+}  // anonymous namespace
>+
>+namespace numbers_internal {
>+bool safe_strto32_base(absl::string_view text, int32_t* value, int base) {
>+  return safe_int_internal<int32_t>(text, value, base);
>+}
>+
>+bool safe_strto64_base(absl::string_view text, int64_t* value, int base) {
>+  return safe_int_internal<int64_t>(text, value, base);
>+}
>+
>+bool safe_strtou32_base(absl::string_view text, uint32_t* value, int base) {
>+  return safe_uint_internal<uint32_t>(text, value, base);
>+}
>+
>+bool safe_strtou64_base(absl::string_view text, uint64_t* value, int base) {
>+  return safe_uint_internal<uint64_t>(text, value, base);
>+}
>+}  // namespace numbers_internal
>+
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/numbers.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/numbers.h
>new file mode 100644
>index 00000000000..cf3c597266c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/numbers.h
>@@ -0,0 +1,184 @@
>+//
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// -----------------------------------------------------------------------------
>+// File: numbers.h
>+// -----------------------------------------------------------------------------
>+//
>+// This package contains functions for converting strings to numbers. For
>+// converting numbers to strings, use `StrCat()` or `StrAppend()` in str_cat.h,
>+// which automatically detect and convert most number values appropriately.
>+
>+#ifndef ABSL_STRINGS_NUMBERS_H_
>+#define ABSL_STRINGS_NUMBERS_H_
>+
>+#include <cstddef>
>+#include <cstdlib>
>+#include <cstring>
>+#include <ctime>
>+#include <limits>
>+#include <string>
>+#include <type_traits>
>+
>+#include "absl/base/macros.h"
>+#include "absl/base/port.h"
>+#include "absl/numeric/int128.h"
>+#include "absl/strings/string_view.h"
>+
>+namespace absl {
>+
>+// SimpleAtoi()
>+//
>+// Converts the given std::string into an integer value, returning `true` if
>+// successful. The std::string must reflect a base-10 integer (optionally followed or
>+// preceded by ASCII whitespace) whose value falls within the range of the
>+// integer type,
>+template <typename int_type>
>+ABSL_MUST_USE_RESULT bool SimpleAtoi(absl::string_view s, int_type* out);
>+
>+// SimpleAtof()
>+//
>+// Converts the given std::string (optionally followed or preceded by ASCII
>+// whitespace) into a float, which may be rounded on overflow or underflow.
>+// See http://en.cppreference.com/w/c/std::string/byte/strtof for details about the
>+// allowed formats for `str`.
>+ABSL_MUST_USE_RESULT bool SimpleAtof(absl::string_view str, float* value);
>+
>+// SimpleAtod()
>+//
>+// Converts the given std::string (optionally followed or preceded by ASCII
>+// whitespace) into a double, which may be rounded on overflow or underflow.
>+// See http://en.cppreference.com/w/c/std::string/byte/strtof for details about the
>+// allowed formats for `str`.
>+ABSL_MUST_USE_RESULT bool SimpleAtod(absl::string_view str, double* value);
>+
>+// SimpleAtob()
>+//
>+// Converts the given std::string into a boolean, returning `true` if successful.
>+// The following case-insensitive strings are interpreted as boolean `true`:
>+// "true", "t", "yes", "y", "1". The following case-insensitive strings
>+// are interpreted as boolean `false`: "false", "f", "no", "n", "0".
>+ABSL_MUST_USE_RESULT bool SimpleAtob(absl::string_view str, bool* value);
>+
>+}  // namespace absl
>+
>+// End of public API.  Implementation details follow.
>+
>+namespace absl {
>+namespace numbers_internal {
>+
>+// safe_strto?() functions for implementing SimpleAtoi()
>+bool safe_strto32_base(absl::string_view text, int32_t* value, int base);
>+bool safe_strto64_base(absl::string_view text, int64_t* value, int base);
>+bool safe_strtou32_base(absl::string_view text, uint32_t* value, int base);
>+bool safe_strtou64_base(absl::string_view text, uint64_t* value, int base);
>+
>+static const int kFastToBufferSize = 32;
>+static const int kSixDigitsToBufferSize = 16;
>+
>+// Helper function for fast formatting of floating-point values.
>+// The result is the same as printf's "%g", a.k.a. "%.6g"; that is, six
>+// significant digits are returned, trailing zeros are removed, and numbers
>+// outside the range 0.0001-999999 are output using scientific notation
>+// (1.23456e+06). This routine is heavily optimized.
>+// Required buffer size is `kSixDigitsToBufferSize`.
>+size_t SixDigitsToBuffer(double d, char* buffer);
>+
>+// These functions are intended for speed. All functions take an output buffer
>+// as an argument and return a pointer to the last byte they wrote, which is the
>+// terminating '\0'. At most `kFastToBufferSize` bytes are written.
>+char* FastIntToBuffer(int32_t, char*);
>+char* FastIntToBuffer(uint32_t, char*);
>+char* FastIntToBuffer(int64_t, char*);
>+char* FastIntToBuffer(uint64_t, char*);
>+
>+// For enums and integer types that are not an exact match for the types above,
>+// use templates to call the appropriate one of the four overloads above.
>+template <typename int_type>
>+char* FastIntToBuffer(int_type i, char* buffer) {
>+  static_assert(sizeof(i) <= 64 / 8,
>+                "FastIntToBuffer works only with 64-bit-or-less integers.");
>+  // TODO(jorg): This signed-ness check is used because it works correctly
>+  // with enums, and it also serves to check that int_type is not a pointer.
>+  // If one day something like std::is_signed<enum E> works, switch to it.
>+  if (static_cast<int_type>(1) - 2 < 0) {  // Signed
>+    if (sizeof(i) > 32 / 8) {           // 33-bit to 64-bit
>+      return FastIntToBuffer(static_cast<int64_t>(i), buffer);
>+    } else {  // 32-bit or less
>+      return FastIntToBuffer(static_cast<int32_t>(i), buffer);
>+    }
>+  } else {                     // Unsigned
>+    if (sizeof(i) > 32 / 8) {  // 33-bit to 64-bit
>+      return FastIntToBuffer(static_cast<uint64_t>(i), buffer);
>+    } else {  // 32-bit or less
>+      return FastIntToBuffer(static_cast<uint32_t>(i), buffer);
>+    }
>+  }
>+}
>+
>+// Implementation of SimpleAtoi, generalized to support arbitrary base (used
>+// with base different from 10 elsewhere in Abseil implementation).
>+template <typename int_type>
>+ABSL_MUST_USE_RESULT bool safe_strtoi_base(absl::string_view s, int_type* out,
>+                                           int base) {
>+  static_assert(sizeof(*out) == 4 || sizeof(*out) == 8,
>+                "SimpleAtoi works only with 32-bit or 64-bit integers.");
>+  static_assert(!std::is_floating_point<int_type>::value,
>+                "Use SimpleAtof or SimpleAtod instead.");
>+  bool parsed;
>+  // TODO(jorg): This signed-ness check is used because it works correctly
>+  // with enums, and it also serves to check that int_type is not a pointer.
>+  // If one day something like std::is_signed<enum E> works, switch to it.
>+  if (static_cast<int_type>(1) - 2 < 0) {  // Signed
>+    if (sizeof(*out) == 64 / 8) {       // 64-bit
>+      int64_t val;
>+      parsed = numbers_internal::safe_strto64_base(s, &val, base);
>+      *out = static_cast<int_type>(val);
>+    } else {  // 32-bit
>+      int32_t val;
>+      parsed = numbers_internal::safe_strto32_base(s, &val, base);
>+      *out = static_cast<int_type>(val);
>+    }
>+  } else {                         // Unsigned
>+    if (sizeof(*out) == 64 / 8) {  // 64-bit
>+      uint64_t val;
>+      parsed = numbers_internal::safe_strtou64_base(s, &val, base);
>+      *out = static_cast<int_type>(val);
>+    } else {  // 32-bit
>+      uint32_t val;
>+      parsed = numbers_internal::safe_strtou32_base(s, &val, base);
>+      *out = static_cast<int_type>(val);
>+    }
>+  }
>+  return parsed;
>+}
>+
>+}  // namespace numbers_internal
>+
>+// SimpleAtoi()
>+//
>+// Converts a std::string to an integer, using `safe_strto?()` functions for actual
>+// parsing, returning `true` if successful. The `safe_strto?()` functions apply
>+// strict checking; the std::string must be a base-10 integer, optionally followed or
>+// preceded by ASCII whitespace, with a value in the range of the corresponding
>+// integer type.
>+template <typename int_type>
>+ABSL_MUST_USE_RESULT bool SimpleAtoi(absl::string_view s, int_type* out) {
>+  return numbers_internal::safe_strtoi_base(s, out, 10);
>+}
>+
>+}  // namespace absl
>+
>+#endif  // ABSL_STRINGS_NUMBERS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/numbers_benchmark.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/numbers_benchmark.cc
>new file mode 100644
>index 00000000000..8ef650b9a3e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/numbers_benchmark.cc
>@@ -0,0 +1,263 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include <cstdint>
>+#include <random>
>+#include <string>
>+#include <type_traits>
>+#include <vector>
>+
>+#include "benchmark/benchmark.h"
>+#include "absl/base/internal/raw_logging.h"
>+#include "absl/strings/numbers.h"
>+
>+namespace {
>+
>+template <typename T>
>+void BM_FastIntToBuffer(benchmark::State& state) {
>+  const int inc = state.range(0);
>+  char buf[absl::numbers_internal::kFastToBufferSize];
>+  // Use the unsigned type to increment to take advantage of well-defined
>+  // modular arithmetic.
>+  typename std::make_unsigned<T>::type x = 0;
>+  for (auto _ : state) {
>+    absl::numbers_internal::FastIntToBuffer(static_cast<T>(x), buf);
>+    x += inc;
>+  }
>+}
>+BENCHMARK_TEMPLATE(BM_FastIntToBuffer, int32_t)->Range(0, 1 << 15);
>+BENCHMARK_TEMPLATE(BM_FastIntToBuffer, int64_t)->Range(0, 1 << 30);
>+
>+// Creates an integer that would be printed as `num_digits` repeated 7s in the
>+// given `base`. `base` must be greater than or equal to 8.
>+int64_t RepeatedSevens(int num_digits, int base) {
>+  ABSL_RAW_CHECK(base >= 8, "");
>+  int64_t num = 7;
>+  while (--num_digits) num = base * num + 7;
>+  return num;
>+}
>+
>+void BM_safe_strto32_string(benchmark::State& state) {
>+  const int digits = state.range(0);
>+  const int base = state.range(1);
>+  std::string str(digits, '7');  // valid in octal, decimal and hex
>+  int32_t value = 0;
>+  for (auto _ : state) {
>+    benchmark::DoNotOptimize(
>+        absl::numbers_internal::safe_strto32_base(str, &value, base));
>+  }
>+  ABSL_RAW_CHECK(value == RepeatedSevens(digits, base), "");
>+}
>+BENCHMARK(BM_safe_strto32_string)
>+    ->ArgPair(1, 8)
>+    ->ArgPair(1, 10)
>+    ->ArgPair(1, 16)
>+    ->ArgPair(2, 8)
>+    ->ArgPair(2, 10)
>+    ->ArgPair(2, 16)
>+    ->ArgPair(4, 8)
>+    ->ArgPair(4, 10)
>+    ->ArgPair(4, 16)
>+    ->ArgPair(8, 8)
>+    ->ArgPair(8, 10)
>+    ->ArgPair(8, 16)
>+    ->ArgPair(10, 8)
>+    ->ArgPair(9, 10);
>+
>+void BM_safe_strto64_string(benchmark::State& state) {
>+  const int digits = state.range(0);
>+  const int base = state.range(1);
>+  std::string str(digits, '7');  // valid in octal, decimal and hex
>+  int64_t value = 0;
>+  for (auto _ : state) {
>+    benchmark::DoNotOptimize(
>+        absl::numbers_internal::safe_strto64_base(str, &value, base));
>+  }
>+  ABSL_RAW_CHECK(value == RepeatedSevens(digits, base), "");
>+}
>+BENCHMARK(BM_safe_strto64_string)
>+    ->ArgPair(1, 8)
>+    ->ArgPair(1, 10)
>+    ->ArgPair(1, 16)
>+    ->ArgPair(2, 8)
>+    ->ArgPair(2, 10)
>+    ->ArgPair(2, 16)
>+    ->ArgPair(4, 8)
>+    ->ArgPair(4, 10)
>+    ->ArgPair(4, 16)
>+    ->ArgPair(8, 8)
>+    ->ArgPair(8, 10)
>+    ->ArgPair(8, 16)
>+    ->ArgPair(16, 8)
>+    ->ArgPair(16, 10)
>+    ->ArgPair(16, 16);
>+
>+void BM_safe_strtou32_string(benchmark::State& state) {
>+  const int digits = state.range(0);
>+  const int base = state.range(1);
>+  std::string str(digits, '7');  // valid in octal, decimal and hex
>+  uint32_t value = 0;
>+  for (auto _ : state) {
>+    benchmark::DoNotOptimize(
>+        absl::numbers_internal::safe_strtou32_base(str, &value, base));
>+  }
>+  ABSL_RAW_CHECK(value == RepeatedSevens(digits, base), "");
>+}
>+BENCHMARK(BM_safe_strtou32_string)
>+    ->ArgPair(1, 8)
>+    ->ArgPair(1, 10)
>+    ->ArgPair(1, 16)
>+    ->ArgPair(2, 8)
>+    ->ArgPair(2, 10)
>+    ->ArgPair(2, 16)
>+    ->ArgPair(4, 8)
>+    ->ArgPair(4, 10)
>+    ->ArgPair(4, 16)
>+    ->ArgPair(8, 8)
>+    ->ArgPair(8, 10)
>+    ->ArgPair(8, 16)
>+    ->ArgPair(10, 8)
>+    ->ArgPair(9, 10);
>+
>+void BM_safe_strtou64_string(benchmark::State& state) {
>+  const int digits = state.range(0);
>+  const int base = state.range(1);
>+  std::string str(digits, '7');  // valid in octal, decimal and hex
>+  uint64_t value = 0;
>+  for (auto _ : state) {
>+    benchmark::DoNotOptimize(
>+        absl::numbers_internal::safe_strtou64_base(str, &value, base));
>+  }
>+  ABSL_RAW_CHECK(value == RepeatedSevens(digits, base), "");
>+}
>+BENCHMARK(BM_safe_strtou64_string)
>+    ->ArgPair(1, 8)
>+    ->ArgPair(1, 10)
>+    ->ArgPair(1, 16)
>+    ->ArgPair(2, 8)
>+    ->ArgPair(2, 10)
>+    ->ArgPair(2, 16)
>+    ->ArgPair(4, 8)
>+    ->ArgPair(4, 10)
>+    ->ArgPair(4, 16)
>+    ->ArgPair(8, 8)
>+    ->ArgPair(8, 10)
>+    ->ArgPair(8, 16)
>+    ->ArgPair(16, 8)
>+    ->ArgPair(16, 10)
>+    ->ArgPair(16, 16);
>+
>+// Returns a vector of `num_strings` strings. Each std::string represents a
>+// floating point number with `num_digits` digits before the decimal point and
>+// another `num_digits` digits after.
>+std::vector<std::string> MakeFloatStrings(int num_strings, int num_digits) {
>+  // For convenience, use a random number generator to generate the test data.
>+  // We don't actually need random properties, so use a fixed seed.
>+  std::minstd_rand0 rng(1);
>+  std::uniform_int_distribution<int> random_digit('0', '9');
>+
>+  std::vector<std::string> float_strings(num_strings);
>+  for (std::string& s : float_strings) {
>+    s.reserve(2 * num_digits + 1);
>+    for (int i = 0; i < num_digits; ++i) {
>+      s.push_back(static_cast<char>(random_digit(rng)));
>+    }
>+    s.push_back('.');
>+    for (int i = 0; i < num_digits; ++i) {
>+      s.push_back(static_cast<char>(random_digit(rng)));
>+    }
>+  }
>+  return float_strings;
>+}
>+
>+template <typename StringType>
>+StringType GetStringAs(const std::string& s) {
>+  return static_cast<StringType>(s);
>+}
>+template <>
>+const char* GetStringAs<const char*>(const std::string& s) {
>+  return s.c_str();
>+}
>+
>+template <typename StringType>
>+std::vector<StringType> GetStringsAs(const std::vector<std::string>& strings) {
>+  std::vector<StringType> result;
>+  result.reserve(strings.size());
>+  for (const std::string& s : strings) {
>+    result.push_back(GetStringAs<StringType>(s));
>+  }
>+  return result;
>+}
>+
>+template <typename T>
>+void BM_SimpleAtof(benchmark::State& state) {
>+  const int num_strings = state.range(0);
>+  const int num_digits = state.range(1);
>+  std::vector<std::string> backing_strings =
>+      MakeFloatStrings(num_strings, num_digits);
>+  std::vector<T> inputs = GetStringsAs<T>(backing_strings);
>+  float value;
>+  for (auto _ : state) {
>+    for (const T& input : inputs) {
>+      benchmark::DoNotOptimize(absl::SimpleAtof(input, &value));
>+    }
>+  }
>+}
>+BENCHMARK_TEMPLATE(BM_SimpleAtof, absl::string_view)
>+    ->ArgPair(10, 1)
>+    ->ArgPair(10, 2)
>+    ->ArgPair(10, 4)
>+    ->ArgPair(10, 8);
>+BENCHMARK_TEMPLATE(BM_SimpleAtof, const char*)
>+    ->ArgPair(10, 1)
>+    ->ArgPair(10, 2)
>+    ->ArgPair(10, 4)
>+    ->ArgPair(10, 8);
>+BENCHMARK_TEMPLATE(BM_SimpleAtof, std::string)
>+    ->ArgPair(10, 1)
>+    ->ArgPair(10, 2)
>+    ->ArgPair(10, 4)
>+    ->ArgPair(10, 8);
>+
>+template <typename T>
>+void BM_SimpleAtod(benchmark::State& state) {
>+  const int num_strings = state.range(0);
>+  const int num_digits = state.range(1);
>+  std::vector<std::string> backing_strings =
>+      MakeFloatStrings(num_strings, num_digits);
>+  std::vector<T> inputs = GetStringsAs<T>(backing_strings);
>+  double value;
>+  for (auto _ : state) {
>+    for (const T& input : inputs) {
>+      benchmark::DoNotOptimize(absl::SimpleAtod(input, &value));
>+    }
>+  }
>+}
>+BENCHMARK_TEMPLATE(BM_SimpleAtod, absl::string_view)
>+    ->ArgPair(10, 1)
>+    ->ArgPair(10, 2)
>+    ->ArgPair(10, 4)
>+    ->ArgPair(10, 8);
>+BENCHMARK_TEMPLATE(BM_SimpleAtod, const char*)
>+    ->ArgPair(10, 1)
>+    ->ArgPair(10, 2)
>+    ->ArgPair(10, 4)
>+    ->ArgPair(10, 8);
>+BENCHMARK_TEMPLATE(BM_SimpleAtod, std::string)
>+    ->ArgPair(10, 1)
>+    ->ArgPair(10, 2)
>+    ->ArgPair(10, 4)
>+    ->ArgPair(10, 8);
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/numbers_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/numbers_test.cc
>new file mode 100644
>index 00000000000..27cc0479e34
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/numbers_test.cc
>@@ -0,0 +1,1185 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+// This file tests std::string processing functions related to numeric values.
>+
>+#include "absl/strings/numbers.h"
>+
>+#include <sys/types.h>
>+#include <cfenv>  // NOLINT(build/c++11)
>+#include <cinttypes>
>+#include <climits>
>+#include <cmath>
>+#include <cstddef>
>+#include <cstdint>
>+#include <cstdio>
>+#include <cstdlib>
>+#include <cstring>
>+#include <limits>
>+#include <numeric>
>+#include <random>
>+#include <set>
>+#include <string>
>+#include <vector>
>+
>+#include "gmock/gmock.h"
>+#include "gtest/gtest.h"
>+#include "absl/base/internal/raw_logging.h"
>+#include "absl/strings/str_cat.h"
>+
>+#include "absl/strings/internal/numbers_test_common.h"
>+
>+namespace {
>+
>+using absl::numbers_internal::kSixDigitsToBufferSize;
>+using absl::numbers_internal::safe_strto32_base;
>+using absl::numbers_internal::safe_strto64_base;
>+using absl::numbers_internal::safe_strtou32_base;
>+using absl::numbers_internal::safe_strtou64_base;
>+using absl::numbers_internal::SixDigitsToBuffer;
>+using absl::strings_internal::Itoa;
>+using absl::strings_internal::strtouint32_test_cases;
>+using absl::strings_internal::strtouint64_test_cases;
>+using absl::SimpleAtoi;
>+using testing::Eq;
>+using testing::MatchesRegex;
>+
>+// Number of floats to test with.
>+// 5,000,000 is a reasonable default for a test that only takes a few seconds.
>+// 1,000,000,000+ triggers checking for all possible mantissa values for
>+// double-precision tests. 2,000,000,000+ triggers checking for every possible
>+// single-precision float.
>+const int kFloatNumCases = 5000000;
>+
>+// This is a slow, brute-force routine to compute the exact base-10
>+// representation of a double-precision floating-point number.  It
>+// is useful for debugging only.
>+std::string PerfectDtoa(double d) {
>+  if (d == 0) return "0";
>+  if (d < 0) return "-" + PerfectDtoa(-d);
>+
>+  // Basic theory: decompose d into mantissa and exp, where
>+  // d = mantissa * 2^exp, and exp is as close to zero as possible.
>+  int64_t mantissa, exp = 0;
>+  while (d >= 1ULL << 63) ++exp, d *= 0.5;
>+  while ((mantissa = d) != d) --exp, d *= 2.0;
>+
>+  // Then convert mantissa to ASCII, and either double it (if
>+  // exp > 0) or halve it (if exp < 0) repeatedly.  "halve it"
>+  // in this case means multiplying it by five and dividing by 10.
>+  constexpr int maxlen = 1100;  // worst case is actually 1030 or so.
>+  char buf[maxlen + 5];
>+  for (int64_t num = mantissa, pos = maxlen; --pos >= 0;) {
>+    buf[pos] = '0' + (num % 10);
>+    num /= 10;
>+  }
>+  char* begin = &buf[0];
>+  char* end = buf + maxlen;
>+  for (int i = 0; i != exp; i += (exp > 0) ? 1 : -1) {
>+    int carry = 0;
>+    for (char* p = end; --p != begin;) {
>+      int dig = *p - '0';
>+      dig = dig * (exp > 0 ? 2 : 5) + carry;
>+      carry = dig / 10;
>+      dig %= 10;
>+      *p = '0' + dig;
>+    }
>+  }
>+  if (exp < 0) {
>+    // "dividing by 10" above means we have to add the decimal point.
>+    memmove(end + 1 + exp, end + exp, 1 - exp);
>+    end[exp] = '.';
>+    ++end;
>+  }
>+  while (*begin == '0' && begin[1] != '.') ++begin;
>+  return {begin, end};
>+}
>+
>+TEST(ToString, PerfectDtoa) {
>+  EXPECT_THAT(PerfectDtoa(1), Eq("1"));
>+  EXPECT_THAT(PerfectDtoa(0.1),
>+              Eq("0.1000000000000000055511151231257827021181583404541015625"));
>+  EXPECT_THAT(PerfectDtoa(1e24), Eq("999999999999999983222784"));
>+  EXPECT_THAT(PerfectDtoa(5e-324), MatchesRegex("0.0000.*625"));
>+  for (int i = 0; i < 100; ++i) {
>+    for (double multiplier :
>+         {1e-300, 1e-200, 1e-100, 0.1, 1.0, 10.0, 1e100, 1e300}) {
>+      double d = multiplier * i;
>+      std::string s = PerfectDtoa(d);
>+      EXPECT_DOUBLE_EQ(d, strtod(s.c_str(), nullptr));
>+    }
>+  }
>+}
>+
>+template <typename integer>
>+struct MyInteger {
>+  integer i;
>+  explicit constexpr MyInteger(integer i) : i(i) {}
>+  constexpr operator integer() const { return i; }
>+
>+  constexpr MyInteger operator+(MyInteger other) const { return i + other.i; }
>+  constexpr MyInteger operator-(MyInteger other) const { return i - other.i; }
>+  constexpr MyInteger operator*(MyInteger other) const { return i * other.i; }
>+  constexpr MyInteger operator/(MyInteger other) const { return i / other.i; }
>+
>+  constexpr bool operator<(MyInteger other) const { return i < other.i; }
>+  constexpr bool operator<=(MyInteger other) const { return i <= other.i; }
>+  constexpr bool operator==(MyInteger other) const { return i == other.i; }
>+  constexpr bool operator>=(MyInteger other) const { return i >= other.i; }
>+  constexpr bool operator>(MyInteger other) const { return i > other.i; }
>+  constexpr bool operator!=(MyInteger other) const { return i != other.i; }
>+
>+  integer as_integer() const { return i; }
>+};
>+
>+typedef MyInteger<int64_t> MyInt64;
>+typedef MyInteger<uint64_t> MyUInt64;
>+
>+void CheckInt32(int32_t x) {
>+  char buffer[absl::numbers_internal::kFastToBufferSize];
>+  char* actual = absl::numbers_internal::FastIntToBuffer(x, buffer);
>+  std::string expected = std::to_string(x);
>+  EXPECT_EQ(expected, std::string(buffer, actual)) << " Input " << x;
>+
>+  char* generic_actual = absl::numbers_internal::FastIntToBuffer(x, buffer);
>+  EXPECT_EQ(expected, std::string(buffer, generic_actual)) << " Input " << x;
>+}
>+
>+void CheckInt64(int64_t x) {
>+  char buffer[absl::numbers_internal::kFastToBufferSize + 3];
>+  buffer[0] = '*';
>+  buffer[23] = '*';
>+  buffer[24] = '*';
>+  char* actual = absl::numbers_internal::FastIntToBuffer(x, &buffer[1]);
>+  std::string expected = std::to_string(x);
>+  EXPECT_EQ(expected, std::string(&buffer[1], actual)) << " Input " << x;
>+  EXPECT_EQ(buffer[0], '*');
>+  EXPECT_EQ(buffer[23], '*');
>+  EXPECT_EQ(buffer[24], '*');
>+
>+  char* my_actual =
>+      absl::numbers_internal::FastIntToBuffer(MyInt64(x), &buffer[1]);
>+  EXPECT_EQ(expected, std::string(&buffer[1], my_actual)) << " Input " << x;
>+}
>+
>+void CheckUInt32(uint32_t x) {
>+  char buffer[absl::numbers_internal::kFastToBufferSize];
>+  char* actual = absl::numbers_internal::FastIntToBuffer(x, buffer);
>+  std::string expected = std::to_string(x);
>+  EXPECT_EQ(expected, std::string(buffer, actual)) << " Input " << x;
>+
>+  char* generic_actual = absl::numbers_internal::FastIntToBuffer(x, buffer);
>+  EXPECT_EQ(expected, std::string(buffer, generic_actual)) << " Input " << x;
>+}
>+
>+void CheckUInt64(uint64_t x) {
>+  char buffer[absl::numbers_internal::kFastToBufferSize + 1];
>+  char* actual = absl::numbers_internal::FastIntToBuffer(x, &buffer[1]);
>+  std::string expected = std::to_string(x);
>+  EXPECT_EQ(expected, std::string(&buffer[1], actual)) << " Input " << x;
>+
>+  char* generic_actual = absl::numbers_internal::FastIntToBuffer(x, &buffer[1]);
>+  EXPECT_EQ(expected, std::string(&buffer[1], generic_actual)) << " Input " << x;
>+
>+  char* my_actual =
>+      absl::numbers_internal::FastIntToBuffer(MyUInt64(x), &buffer[1]);
>+  EXPECT_EQ(expected, std::string(&buffer[1], my_actual)) << " Input " << x;
>+}
>+
>+void CheckHex64(uint64_t v) {
>+  char expected[16 + 1];
>+  std::string actual = absl::StrCat(absl::Hex(v, absl::kZeroPad16));
>+  snprintf(expected, sizeof(expected), "%016" PRIx64, static_cast<uint64_t>(v));
>+  EXPECT_EQ(expected, actual) << " Input " << v;
>+}
>+
>+TEST(Numbers, TestFastPrints) {
>+  for (int i = -100; i <= 100; i++) {
>+    CheckInt32(i);
>+    CheckInt64(i);
>+  }
>+  for (int i = 0; i <= 100; i++) {
>+    CheckUInt32(i);
>+    CheckUInt64(i);
>+  }
>+  // Test min int to make sure that works
>+  CheckInt32(INT_MIN);
>+  CheckInt32(INT_MAX);
>+  CheckInt64(LONG_MIN);
>+  CheckInt64(uint64_t{1000000000});
>+  CheckInt64(uint64_t{9999999999});
>+  CheckInt64(uint64_t{100000000000000});
>+  CheckInt64(uint64_t{999999999999999});
>+  CheckInt64(uint64_t{1000000000000000000});
>+  CheckInt64(uint64_t{1199999999999999999});
>+  CheckInt64(int64_t{-700000000000000000});
>+  CheckInt64(LONG_MAX);
>+  CheckUInt32(std::numeric_limits<uint32_t>::max());
>+  CheckUInt64(uint64_t{1000000000});
>+  CheckUInt64(uint64_t{9999999999});
>+  CheckUInt64(uint64_t{100000000000000});
>+  CheckUInt64(uint64_t{999999999999999});
>+  CheckUInt64(uint64_t{1000000000000000000});
>+  CheckUInt64(uint64_t{1199999999999999999});
>+  CheckUInt64(std::numeric_limits<uint64_t>::max());
>+
>+  for (int i = 0; i < 10000; i++) {
>+    CheckHex64(i);
>+  }
>+  CheckHex64(uint64_t{0x123456789abcdef0});
>+}
>+
>+template <typename int_type, typename in_val_type>
>+void VerifySimpleAtoiGood(in_val_type in_value, int_type exp_value) {
>+  std::string s = absl::StrCat(in_value);
>+  int_type x = static_cast<int_type>(~exp_value);
>+  EXPECT_TRUE(SimpleAtoi(s, &x))
>+      << "in_value=" << in_value << " s=" << s << " x=" << x;
>+  EXPECT_EQ(exp_value, x);
>+  x = static_cast<int_type>(~exp_value);
>+  EXPECT_TRUE(SimpleAtoi(s.c_str(), &x));
>+  EXPECT_EQ(exp_value, x);
>+}
>+
>+template <typename int_type, typename in_val_type>
>+void VerifySimpleAtoiBad(in_val_type in_value) {
>+  std::string s = absl::StrCat(in_value);
>+  int_type x;
>+  EXPECT_FALSE(SimpleAtoi(s, &x));
>+  EXPECT_FALSE(SimpleAtoi(s.c_str(), &x));
>+}
>+
>+TEST(NumbersTest, Atoi) {
>+  // SimpleAtoi(absl::string_view, int32_t)
>+  VerifySimpleAtoiGood<int32_t>(0, 0);
>+  VerifySimpleAtoiGood<int32_t>(42, 42);
>+  VerifySimpleAtoiGood<int32_t>(-42, -42);
>+
>+  VerifySimpleAtoiGood<int32_t>(std::numeric_limits<int32_t>::min(),
>+                                std::numeric_limits<int32_t>::min());
>+  VerifySimpleAtoiGood<int32_t>(std::numeric_limits<int32_t>::max(),
>+                                std::numeric_limits<int32_t>::max());
>+
>+  // SimpleAtoi(absl::string_view, uint32_t)
>+  VerifySimpleAtoiGood<uint32_t>(0, 0);
>+  VerifySimpleAtoiGood<uint32_t>(42, 42);
>+  VerifySimpleAtoiBad<uint32_t>(-42);
>+
>+  VerifySimpleAtoiBad<uint32_t>(std::numeric_limits<int32_t>::min());
>+  VerifySimpleAtoiGood<uint32_t>(std::numeric_limits<int32_t>::max(),
>+                                 std::numeric_limits<int32_t>::max());
>+  VerifySimpleAtoiGood<uint32_t>(std::numeric_limits<uint32_t>::max(),
>+                                 std::numeric_limits<uint32_t>::max());
>+  VerifySimpleAtoiBad<uint32_t>(std::numeric_limits<int64_t>::min());
>+  VerifySimpleAtoiBad<uint32_t>(std::numeric_limits<int64_t>::max());
>+  VerifySimpleAtoiBad<uint32_t>(std::numeric_limits<uint64_t>::max());
>+
>+  // SimpleAtoi(absl::string_view, int64_t)
>+  VerifySimpleAtoiGood<int64_t>(0, 0);
>+  VerifySimpleAtoiGood<int64_t>(42, 42);
>+  VerifySimpleAtoiGood<int64_t>(-42, -42);
>+
>+  VerifySimpleAtoiGood<int64_t>(std::numeric_limits<int32_t>::min(),
>+                                std::numeric_limits<int32_t>::min());
>+  VerifySimpleAtoiGood<int64_t>(std::numeric_limits<int32_t>::max(),
>+                                std::numeric_limits<int32_t>::max());
>+  VerifySimpleAtoiGood<int64_t>(std::numeric_limits<uint32_t>::max(),
>+                                std::numeric_limits<uint32_t>::max());
>+  VerifySimpleAtoiGood<int64_t>(std::numeric_limits<int64_t>::min(),
>+                                std::numeric_limits<int64_t>::min());
>+  VerifySimpleAtoiGood<int64_t>(std::numeric_limits<int64_t>::max(),
>+                                std::numeric_limits<int64_t>::max());
>+  VerifySimpleAtoiBad<int64_t>(std::numeric_limits<uint64_t>::max());
>+
>+  // SimpleAtoi(absl::string_view, uint64_t)
>+  VerifySimpleAtoiGood<uint64_t>(0, 0);
>+  VerifySimpleAtoiGood<uint64_t>(42, 42);
>+  VerifySimpleAtoiBad<uint64_t>(-42);
>+
>+  VerifySimpleAtoiBad<uint64_t>(std::numeric_limits<int32_t>::min());
>+  VerifySimpleAtoiGood<uint64_t>(std::numeric_limits<int32_t>::max(),
>+                                 std::numeric_limits<int32_t>::max());
>+  VerifySimpleAtoiGood<uint64_t>(std::numeric_limits<uint32_t>::max(),
>+                                 std::numeric_limits<uint32_t>::max());
>+  VerifySimpleAtoiBad<uint64_t>(std::numeric_limits<int64_t>::min());
>+  VerifySimpleAtoiGood<uint64_t>(std::numeric_limits<int64_t>::max(),
>+                                 std::numeric_limits<int64_t>::max());
>+  VerifySimpleAtoiGood<uint64_t>(std::numeric_limits<uint64_t>::max(),
>+                                 std::numeric_limits<uint64_t>::max());
>+
>+  // Some other types
>+  VerifySimpleAtoiGood<int>(-42, -42);
>+  VerifySimpleAtoiGood<int32_t>(-42, -42);
>+  VerifySimpleAtoiGood<uint32_t>(42, 42);
>+  VerifySimpleAtoiGood<unsigned int>(42, 42);
>+  VerifySimpleAtoiGood<int64_t>(-42, -42);
>+  VerifySimpleAtoiGood<long>(-42, -42);  // NOLINT(runtime/int)
>+  VerifySimpleAtoiGood<uint64_t>(42, 42);
>+  VerifySimpleAtoiGood<size_t>(42, 42);
>+  VerifySimpleAtoiGood<std::string::size_type>(42, 42);
>+}
>+
>+TEST(NumbersTest, Atoenum) {
>+  enum E01 {
>+    E01_zero = 0,
>+    E01_one = 1,
>+  };
>+
>+  VerifySimpleAtoiGood<E01>(E01_zero, E01_zero);
>+  VerifySimpleAtoiGood<E01>(E01_one, E01_one);
>+
>+  enum E_101 {
>+    E_101_minusone = -1,
>+    E_101_zero = 0,
>+    E_101_one = 1,
>+  };
>+
>+  VerifySimpleAtoiGood<E_101>(E_101_minusone, E_101_minusone);
>+  VerifySimpleAtoiGood<E_101>(E_101_zero, E_101_zero);
>+  VerifySimpleAtoiGood<E_101>(E_101_one, E_101_one);
>+
>+  enum E_bigint {
>+    E_bigint_zero = 0,
>+    E_bigint_one = 1,
>+    E_bigint_max31 = static_cast<int32_t>(0x7FFFFFFF),
>+  };
>+
>+  VerifySimpleAtoiGood<E_bigint>(E_bigint_zero, E_bigint_zero);
>+  VerifySimpleAtoiGood<E_bigint>(E_bigint_one, E_bigint_one);
>+  VerifySimpleAtoiGood<E_bigint>(E_bigint_max31, E_bigint_max31);
>+
>+  enum E_fullint {
>+    E_fullint_zero = 0,
>+    E_fullint_one = 1,
>+    E_fullint_max31 = static_cast<int32_t>(0x7FFFFFFF),
>+    E_fullint_min32 = INT32_MIN,
>+  };
>+
>+  VerifySimpleAtoiGood<E_fullint>(E_fullint_zero, E_fullint_zero);
>+  VerifySimpleAtoiGood<E_fullint>(E_fullint_one, E_fullint_one);
>+  VerifySimpleAtoiGood<E_fullint>(E_fullint_max31, E_fullint_max31);
>+  VerifySimpleAtoiGood<E_fullint>(E_fullint_min32, E_fullint_min32);
>+
>+  enum E_biguint {
>+    E_biguint_zero = 0,
>+    E_biguint_one = 1,
>+    E_biguint_max31 = static_cast<uint32_t>(0x7FFFFFFF),
>+    E_biguint_max32 = static_cast<uint32_t>(0xFFFFFFFF),
>+  };
>+
>+  VerifySimpleAtoiGood<E_biguint>(E_biguint_zero, E_biguint_zero);
>+  VerifySimpleAtoiGood<E_biguint>(E_biguint_one, E_biguint_one);
>+  VerifySimpleAtoiGood<E_biguint>(E_biguint_max31, E_biguint_max31);
>+  VerifySimpleAtoiGood<E_biguint>(E_biguint_max32, E_biguint_max32);
>+}
>+
>+TEST(stringtest, safe_strto32_base) {
>+  int32_t value;
>+  EXPECT_TRUE(safe_strto32_base("0x34234324", &value, 16));
>+  EXPECT_EQ(0x34234324, value);
>+
>+  EXPECT_TRUE(safe_strto32_base("0X34234324", &value, 16));
>+  EXPECT_EQ(0x34234324, value);
>+
>+  EXPECT_TRUE(safe_strto32_base("34234324", &value, 16));
>+  EXPECT_EQ(0x34234324, value);
>+
>+  EXPECT_TRUE(safe_strto32_base("0", &value, 16));
>+  EXPECT_EQ(0, value);
>+
>+  EXPECT_TRUE(safe_strto32_base(" \t\n -0x34234324", &value, 16));
>+  EXPECT_EQ(-0x34234324, value);
>+
>+  EXPECT_TRUE(safe_strto32_base(" \t\n -34234324", &value, 16));
>+  EXPECT_EQ(-0x34234324, value);
>+
>+  EXPECT_TRUE(safe_strto32_base("7654321", &value, 8));
>+  EXPECT_EQ(07654321, value);
>+
>+  EXPECT_TRUE(safe_strto32_base("-01234", &value, 8));
>+  EXPECT_EQ(-01234, value);
>+
>+  EXPECT_FALSE(safe_strto32_base("1834", &value, 8));
>+
>+  // Autodetect base.
>+  EXPECT_TRUE(safe_strto32_base("0", &value, 0));
>+  EXPECT_EQ(0, value);
>+
>+  EXPECT_TRUE(safe_strto32_base("077", &value, 0));
>+  EXPECT_EQ(077, value);  // Octal interpretation
>+
>+  // Leading zero indicates octal, but then followed by invalid digit.
>+  EXPECT_FALSE(safe_strto32_base("088", &value, 0));
>+
>+  // Leading 0x indicated hex, but then followed by invalid digit.
>+  EXPECT_FALSE(safe_strto32_base("0xG", &value, 0));
>+
>+  // Base-10 version.
>+  EXPECT_TRUE(safe_strto32_base("34234324", &value, 10));
>+  EXPECT_EQ(34234324, value);
>+
>+  EXPECT_TRUE(safe_strto32_base("0", &value, 10));
>+  EXPECT_EQ(0, value);
>+
>+  EXPECT_TRUE(safe_strto32_base(" \t\n -34234324", &value, 10));
>+  EXPECT_EQ(-34234324, value);
>+
>+  EXPECT_TRUE(safe_strto32_base("34234324 \n\t ", &value, 10));
>+  EXPECT_EQ(34234324, value);
>+
>+  // Invalid ints.
>+  EXPECT_FALSE(safe_strto32_base("", &value, 10));
>+  EXPECT_FALSE(safe_strto32_base("  ", &value, 10));
>+  EXPECT_FALSE(safe_strto32_base("abc", &value, 10));
>+  EXPECT_FALSE(safe_strto32_base("34234324a", &value, 10));
>+  EXPECT_FALSE(safe_strto32_base("34234.3", &value, 10));
>+
>+  // Out of bounds.
>+  EXPECT_FALSE(safe_strto32_base("2147483648", &value, 10));
>+  EXPECT_FALSE(safe_strto32_base("-2147483649", &value, 10));
>+
>+  // String version.
>+  EXPECT_TRUE(safe_strto32_base(std::string("0x1234"), &value, 16));
>+  EXPECT_EQ(0x1234, value);
>+
>+  // Base-10 std::string version.
>+  EXPECT_TRUE(safe_strto32_base("1234", &value, 10));
>+  EXPECT_EQ(1234, value);
>+}
>+
>+TEST(stringtest, safe_strto32_range) {
>+  // These tests verify underflow/overflow behaviour.
>+  int32_t value;
>+  EXPECT_FALSE(safe_strto32_base("2147483648", &value, 10));
>+  EXPECT_EQ(std::numeric_limits<int32_t>::max(), value);
>+
>+  EXPECT_TRUE(safe_strto32_base("-2147483648", &value, 10));
>+  EXPECT_EQ(std::numeric_limits<int32_t>::min(), value);
>+
>+  EXPECT_FALSE(safe_strto32_base("-2147483649", &value, 10));
>+  EXPECT_EQ(std::numeric_limits<int32_t>::min(), value);
>+}
>+
>+TEST(stringtest, safe_strto64_range) {
>+  // These tests verify underflow/overflow behaviour.
>+  int64_t value;
>+  EXPECT_FALSE(safe_strto64_base("9223372036854775808", &value, 10));
>+  EXPECT_EQ(std::numeric_limits<int64_t>::max(), value);
>+
>+  EXPECT_TRUE(safe_strto64_base("-9223372036854775808", &value, 10));
>+  EXPECT_EQ(std::numeric_limits<int64_t>::min(), value);
>+
>+  EXPECT_FALSE(safe_strto64_base("-9223372036854775809", &value, 10));
>+  EXPECT_EQ(std::numeric_limits<int64_t>::min(), value);
>+}
>+
>+TEST(stringtest, safe_strto32_leading_substring) {
>+  // These tests verify this comment in numbers.h:
>+  // On error, returns false, and sets *value to: [...]
>+  //   conversion of leading substring if available ("123@@@" -> 123)
>+  //   0 if no leading substring available
>+  int32_t value;
>+  EXPECT_FALSE(safe_strto32_base("04069@@@", &value, 10));
>+  EXPECT_EQ(4069, value);
>+
>+  EXPECT_FALSE(safe_strto32_base("04069@@@", &value, 8));
>+  EXPECT_EQ(0406, value);
>+
>+  EXPECT_FALSE(safe_strto32_base("04069balloons", &value, 10));
>+  EXPECT_EQ(4069, value);
>+
>+  EXPECT_FALSE(safe_strto32_base("04069balloons", &value, 16));
>+  EXPECT_EQ(0x4069ba, value);
>+
>+  EXPECT_FALSE(safe_strto32_base("@@@", &value, 10));
>+  EXPECT_EQ(0, value);  // there was no leading substring
>+}
>+
>+TEST(stringtest, safe_strto64_leading_substring) {
>+  // These tests verify this comment in numbers.h:
>+  // On error, returns false, and sets *value to: [...]
>+  //   conversion of leading substring if available ("123@@@" -> 123)
>+  //   0 if no leading substring available
>+  int64_t value;
>+  EXPECT_FALSE(safe_strto64_base("04069@@@", &value, 10));
>+  EXPECT_EQ(4069, value);
>+
>+  EXPECT_FALSE(safe_strto64_base("04069@@@", &value, 8));
>+  EXPECT_EQ(0406, value);
>+
>+  EXPECT_FALSE(safe_strto64_base("04069balloons", &value, 10));
>+  EXPECT_EQ(4069, value);
>+
>+  EXPECT_FALSE(safe_strto64_base("04069balloons", &value, 16));
>+  EXPECT_EQ(0x4069ba, value);
>+
>+  EXPECT_FALSE(safe_strto64_base("@@@", &value, 10));
>+  EXPECT_EQ(0, value);  // there was no leading substring
>+}
>+
>+TEST(stringtest, safe_strto64_base) {
>+  int64_t value;
>+  EXPECT_TRUE(safe_strto64_base("0x3423432448783446", &value, 16));
>+  EXPECT_EQ(int64_t{0x3423432448783446}, value);
>+
>+  EXPECT_TRUE(safe_strto64_base("3423432448783446", &value, 16));
>+  EXPECT_EQ(int64_t{0x3423432448783446}, value);
>+
>+  EXPECT_TRUE(safe_strto64_base("0", &value, 16));
>+  EXPECT_EQ(0, value);
>+
>+  EXPECT_TRUE(safe_strto64_base(" \t\n -0x3423432448783446", &value, 16));
>+  EXPECT_EQ(int64_t{-0x3423432448783446}, value);
>+
>+  EXPECT_TRUE(safe_strto64_base(" \t\n -3423432448783446", &value, 16));
>+  EXPECT_EQ(int64_t{-0x3423432448783446}, value);
>+
>+  EXPECT_TRUE(safe_strto64_base("123456701234567012", &value, 8));
>+  EXPECT_EQ(int64_t{0123456701234567012}, value);
>+
>+  EXPECT_TRUE(safe_strto64_base("-017777777777777", &value, 8));
>+  EXPECT_EQ(int64_t{-017777777777777}, value);
>+
>+  EXPECT_FALSE(safe_strto64_base("19777777777777", &value, 8));
>+
>+  // Autodetect base.
>+  EXPECT_TRUE(safe_strto64_base("0", &value, 0));
>+  EXPECT_EQ(0, value);
>+
>+  EXPECT_TRUE(safe_strto64_base("077", &value, 0));
>+  EXPECT_EQ(077, value);  // Octal interpretation
>+
>+  // Leading zero indicates octal, but then followed by invalid digit.
>+  EXPECT_FALSE(safe_strto64_base("088", &value, 0));
>+
>+  // Leading 0x indicated hex, but then followed by invalid digit.
>+  EXPECT_FALSE(safe_strto64_base("0xG", &value, 0));
>+
>+  // Base-10 version.
>+  EXPECT_TRUE(safe_strto64_base("34234324487834466", &value, 10));
>+  EXPECT_EQ(int64_t{34234324487834466}, value);
>+
>+  EXPECT_TRUE(safe_strto64_base("0", &value, 10));
>+  EXPECT_EQ(0, value);
>+
>+  EXPECT_TRUE(safe_strto64_base(" \t\n -34234324487834466", &value, 10));
>+  EXPECT_EQ(int64_t{-34234324487834466}, value);
>+
>+  EXPECT_TRUE(safe_strto64_base("34234324487834466 \n\t ", &value, 10));
>+  EXPECT_EQ(int64_t{34234324487834466}, value);
>+
>+  // Invalid ints.
>+  EXPECT_FALSE(safe_strto64_base("", &value, 10));
>+  EXPECT_FALSE(safe_strto64_base("  ", &value, 10));
>+  EXPECT_FALSE(safe_strto64_base("abc", &value, 10));
>+  EXPECT_FALSE(safe_strto64_base("34234324487834466a", &value, 10));
>+  EXPECT_FALSE(safe_strto64_base("34234487834466.3", &value, 10));
>+
>+  // Out of bounds.
>+  EXPECT_FALSE(safe_strto64_base("9223372036854775808", &value, 10));
>+  EXPECT_FALSE(safe_strto64_base("-9223372036854775809", &value, 10));
>+
>+  // String version.
>+  EXPECT_TRUE(safe_strto64_base(std::string("0x1234"), &value, 16));
>+  EXPECT_EQ(0x1234, value);
>+
>+  // Base-10 std::string version.
>+  EXPECT_TRUE(safe_strto64_base("1234", &value, 10));
>+  EXPECT_EQ(1234, value);
>+}
>+
>+const size_t kNumRandomTests = 10000;
>+
>+template <typename IntType>
>+void test_random_integer_parse_base(bool (*parse_func)(absl::string_view,
>+                                                       IntType* value,
>+                                                       int base)) {
>+  using RandomEngine = std::minstd_rand0;
>+  std::random_device rd;
>+  RandomEngine rng(rd());
>+  std::uniform_int_distribution<IntType> random_int(
>+      std::numeric_limits<IntType>::min());
>+  std::uniform_int_distribution<int> random_base(2, 35);
>+  for (size_t i = 0; i < kNumRandomTests; i++) {
>+    IntType value = random_int(rng);
>+    int base = random_base(rng);
>+    std::string str_value;
>+    EXPECT_TRUE(Itoa<IntType>(value, base, &str_value));
>+    IntType parsed_value;
>+
>+    // Test successful parse
>+    EXPECT_TRUE(parse_func(str_value, &parsed_value, base));
>+    EXPECT_EQ(parsed_value, value);
>+
>+    // Test overflow
>+    EXPECT_FALSE(
>+        parse_func(absl::StrCat(std::numeric_limits<IntType>::max(), value),
>+                   &parsed_value, base));
>+
>+    // Test underflow
>+    if (std::numeric_limits<IntType>::min() < 0) {
>+      EXPECT_FALSE(
>+          parse_func(absl::StrCat(std::numeric_limits<IntType>::min(), value),
>+                     &parsed_value, base));
>+    } else {
>+      EXPECT_FALSE(parse_func(absl::StrCat("-", value), &parsed_value, base));
>+    }
>+  }
>+}
>+
>+TEST(stringtest, safe_strto32_random) {
>+  test_random_integer_parse_base<int32_t>(&safe_strto32_base);
>+}
>+TEST(stringtest, safe_strto64_random) {
>+  test_random_integer_parse_base<int64_t>(&safe_strto64_base);
>+}
>+TEST(stringtest, safe_strtou32_random) {
>+  test_random_integer_parse_base<uint32_t>(&safe_strtou32_base);
>+}
>+TEST(stringtest, safe_strtou64_random) {
>+  test_random_integer_parse_base<uint64_t>(&safe_strtou64_base);
>+}
>+
>+TEST(stringtest, safe_strtou32_base) {
>+  for (int i = 0; strtouint32_test_cases()[i].str != nullptr; ++i) {
>+    const auto& e = strtouint32_test_cases()[i];
>+    uint32_t value;
>+    EXPECT_EQ(e.expect_ok, safe_strtou32_base(e.str, &value, e.base))
>+        << "str=\"" << e.str << "\" base=" << e.base;
>+    if (e.expect_ok) {
>+      EXPECT_EQ(e.expected, value) << "i=" << i << " str=\"" << e.str
>+                                   << "\" base=" << e.base;
>+    }
>+  }
>+}
>+
>+TEST(stringtest, safe_strtou32_base_length_delimited) {
>+  for (int i = 0; strtouint32_test_cases()[i].str != nullptr; ++i) {
>+    const auto& e = strtouint32_test_cases()[i];
>+    std::string tmp(e.str);
>+    tmp.append("12");  // Adds garbage at the end.
>+
>+    uint32_t value;
>+    EXPECT_EQ(e.expect_ok,
>+              safe_strtou32_base(absl::string_view(tmp.data(), strlen(e.str)),
>+                                 &value, e.base))
>+        << "str=\"" << e.str << "\" base=" << e.base;
>+    if (e.expect_ok) {
>+      EXPECT_EQ(e.expected, value) << "i=" << i << " str=" << e.str
>+                                   << " base=" << e.base;
>+    }
>+  }
>+}
>+
>+TEST(stringtest, safe_strtou64_base) {
>+  for (int i = 0; strtouint64_test_cases()[i].str != nullptr; ++i) {
>+    const auto& e = strtouint64_test_cases()[i];
>+    uint64_t value;
>+    EXPECT_EQ(e.expect_ok, safe_strtou64_base(e.str, &value, e.base))
>+        << "str=\"" << e.str << "\" base=" << e.base;
>+    if (e.expect_ok) {
>+      EXPECT_EQ(e.expected, value) << "str=" << e.str << " base=" << e.base;
>+    }
>+  }
>+}
>+
>+TEST(stringtest, safe_strtou64_base_length_delimited) {
>+  for (int i = 0; strtouint64_test_cases()[i].str != nullptr; ++i) {
>+    const auto& e = strtouint64_test_cases()[i];
>+    std::string tmp(e.str);
>+    tmp.append("12");  // Adds garbage at the end.
>+
>+    uint64_t value;
>+    EXPECT_EQ(e.expect_ok,
>+              safe_strtou64_base(absl::string_view(tmp.data(), strlen(e.str)),
>+                                 &value, e.base))
>+        << "str=\"" << e.str << "\" base=" << e.base;
>+    if (e.expect_ok) {
>+      EXPECT_EQ(e.expected, value) << "str=\"" << e.str << "\" base=" << e.base;
>+    }
>+  }
>+}
>+
>+// feenableexcept() and fedisableexcept() are missing on Mac OS X, MSVC,
>+// and WebAssembly.
>+#if defined(_MSC_VER) || defined(__APPLE__) || defined(__EMSCRIPTEN__)
>+#define ABSL_MISSING_FEENABLEEXCEPT 1
>+#define ABSL_MISSING_FEDISABLEEXCEPT 1
>+#endif
>+
>+class SimpleDtoaTest : public testing::Test {
>+ protected:
>+  void SetUp() override {
>+    // Store the current floating point env & clear away any pending exceptions.
>+    feholdexcept(&fp_env_);
>+#ifndef ABSL_MISSING_FEENABLEEXCEPT
>+    // Turn on floating point exceptions.
>+    feenableexcept(FE_DIVBYZERO | FE_INVALID | FE_OVERFLOW);
>+#endif
>+  }
>+
>+  void TearDown() override {
>+    // Restore the floating point environment to the original state.
>+    // In theory fedisableexcept is unnecessary; fesetenv will also do it.
>+    // In practice, our toolchains have subtle bugs.
>+#ifndef ABSL_MISSING_FEDISABLEEXCEPT
>+    fedisableexcept(FE_DIVBYZERO | FE_INVALID | FE_OVERFLOW);
>+#endif
>+    fesetenv(&fp_env_);
>+  }
>+
>+  std::string ToNineDigits(double value) {
>+    char buffer[16];  // more than enough for %.9g
>+    snprintf(buffer, sizeof(buffer), "%.9g", value);
>+    return buffer;
>+  }
>+
>+  fenv_t fp_env_;
>+};
>+
>+// Run the given runnable functor for "cases" test cases, chosen over the
>+// available range of float.  pi and e and 1/e are seeded, and then all
>+// available integer powers of 2 and 10 are multiplied against them.  In
>+// addition to trying all those values, we try the next higher and next lower
>+// float, and then we add additional test cases evenly distributed between them.
>+// Each test case is passed to runnable as both a positive and negative value.
>+template <typename R>
>+void ExhaustiveFloat(uint32_t cases, R&& runnable) {
>+  runnable(0.0f);
>+  runnable(-0.0f);
>+  if (cases >= 2e9) {  // more than 2 billion?  Might as well run them all.
>+    for (float f = 0; f < std::numeric_limits<float>::max(); ) {
>+      f = nextafterf(f, std::numeric_limits<float>::max());
>+      runnable(-f);
>+      runnable(f);
>+    }
>+    return;
>+  }
>+  std::set<float> floats = {3.4028234e38f};
>+  for (float f : {1.0, 3.14159265, 2.718281828, 1 / 2.718281828}) {
>+    for (float testf = f; testf != 0; testf *= 0.1f) floats.insert(testf);
>+    for (float testf = f; testf != 0; testf *= 0.5f) floats.insert(testf);
>+    for (float testf = f; testf < 3e38f / 2; testf *= 2.0f)
>+      floats.insert(testf);
>+    for (float testf = f; testf < 3e38f / 10; testf *= 10) floats.insert(testf);
>+  }
>+
>+  float last = *floats.begin();
>+
>+  runnable(last);
>+  runnable(-last);
>+  int iters_per_float = cases / floats.size();
>+  if (iters_per_float == 0) iters_per_float = 1;
>+  for (float f : floats) {
>+    if (f == last) continue;
>+    float testf = nextafter(last, std::numeric_limits<float>::max());
>+    runnable(testf);
>+    runnable(-testf);
>+    last = testf;
>+    if (f == last) continue;
>+    double step = (double{f} - last) / iters_per_float;
>+    for (double d = last + step; d < f; d += step) {
>+      testf = d;
>+      if (testf != last) {
>+        runnable(testf);
>+        runnable(-testf);
>+        last = testf;
>+      }
>+    }
>+    testf = nextafter(f, 0.0f);
>+    if (testf > last) {
>+      runnable(testf);
>+      runnable(-testf);
>+      last = testf;
>+    }
>+    if (f != last) {
>+      runnable(f);
>+      runnable(-f);
>+      last = f;
>+    }
>+  }
>+}
>+
>+TEST_F(SimpleDtoaTest, ExhaustiveDoubleToSixDigits) {
>+  uint64_t test_count = 0;
>+  std::vector<double> mismatches;
>+  auto checker = [&](double d) {
>+    if (d != d) return;  // rule out NaNs
>+    ++test_count;
>+    char sixdigitsbuf[kSixDigitsToBufferSize] = {0};
>+    SixDigitsToBuffer(d, sixdigitsbuf);
>+    char snprintfbuf[kSixDigitsToBufferSize] = {0};
>+    snprintf(snprintfbuf, kSixDigitsToBufferSize, "%g", d);
>+    if (strcmp(sixdigitsbuf, snprintfbuf) != 0) {
>+      mismatches.push_back(d);
>+      if (mismatches.size() < 10) {
>+        ABSL_RAW_LOG(ERROR, "%s",
>+                     absl::StrCat("Six-digit failure with double.  ", "d=", d,
>+                                  "=", d, " sixdigits=", sixdigitsbuf,
>+                                  " printf(%g)=", snprintfbuf)
>+                         .c_str());
>+      }
>+    }
>+  };
>+  // Some quick sanity checks...
>+  checker(5e-324);
>+  checker(1e-308);
>+  checker(1.0);
>+  checker(1.000005);
>+  checker(1.7976931348623157e308);
>+  checker(0.00390625);
>+#ifndef _MSC_VER
>+  // on MSVC, snprintf() rounds it to 0.00195313. SixDigitsToBuffer() rounds it
>+  // to 0.00195312 (round half to even).
>+  checker(0.001953125);
>+#endif
>+  checker(0.005859375);
>+  // Some cases where the rounding is very very close
>+  checker(1.089095e-15);
>+  checker(3.274195e-55);
>+  checker(6.534355e-146);
>+  checker(2.920845e+234);
>+
>+  if (mismatches.empty()) {
>+    test_count = 0;
>+    ExhaustiveFloat(kFloatNumCases, checker);
>+
>+    test_count = 0;
>+    std::vector<int> digit_testcases{
>+        100000, 100001, 100002, 100005, 100010, 100020, 100050, 100100,  // misc
>+        195312, 195313,  // 1.953125 is a case where we round down, just barely.
>+        200000, 500000, 800000,  // misc mid-range cases
>+        585937, 585938,  // 5.859375 is a case where we round up, just barely.
>+        900000, 990000, 999000, 999900, 999990, 999996, 999997, 999998, 999999};
>+    if (kFloatNumCases >= 1e9) {
>+      // If at least 1 billion test cases were requested, user wants an
>+      // exhaustive test. So let's test all mantissas, too.
>+      constexpr int min_mantissa = 100000, max_mantissa = 999999;
>+      digit_testcases.resize(max_mantissa - min_mantissa + 1);
>+      std::iota(digit_testcases.begin(), digit_testcases.end(), min_mantissa);
>+    }
>+
>+    for (int exponent = -324; exponent <= 308; ++exponent) {
>+      double powten = pow(10.0, exponent);
>+      if (powten == 0) powten = 5e-324;
>+      if (kFloatNumCases >= 1e9) {
>+        // The exhaustive test takes a very long time, so log progress.
>+        char buf[kSixDigitsToBufferSize];
>+        ABSL_RAW_LOG(
>+            INFO, "%s",
>+            absl::StrCat("Exp ", exponent, " powten=", powten, "(",
>+                         powten, ") (",
>+                         std::string(buf, SixDigitsToBuffer(powten, buf)), ")")
>+                .c_str());
>+      }
>+      for (int digits : digit_testcases) {
>+        if (exponent == 308 && digits >= 179769) break;  // don't overflow!
>+        double digiform = (digits + 0.5) * 0.00001;
>+        double testval = digiform * powten;
>+        double pretestval = nextafter(testval, 0);
>+        double posttestval = nextafter(testval, 1.7976931348623157e308);
>+        checker(testval);
>+        checker(pretestval);
>+        checker(posttestval);
>+      }
>+    }
>+  } else {
>+    EXPECT_EQ(mismatches.size(), 0);
>+    for (size_t i = 0; i < mismatches.size(); ++i) {
>+      if (i > 100) i = mismatches.size() - 1;
>+      double d = mismatches[i];
>+      char sixdigitsbuf[kSixDigitsToBufferSize] = {0};
>+      SixDigitsToBuffer(d, sixdigitsbuf);
>+      char snprintfbuf[kSixDigitsToBufferSize] = {0};
>+      snprintf(snprintfbuf, kSixDigitsToBufferSize, "%g", d);
>+      double before = nextafter(d, 0.0);
>+      double after = nextafter(d, 1.7976931348623157e308);
>+      char b1[32], b2[kSixDigitsToBufferSize];
>+      ABSL_RAW_LOG(
>+          ERROR, "%s",
>+          absl::StrCat(
>+              "Mismatch #", i, "  d=", d, " (", ToNineDigits(d), ")",
>+              " sixdigits='", sixdigitsbuf, "'", " snprintf='", snprintfbuf,
>+              "'", " Before.=", PerfectDtoa(before), " ",
>+              (SixDigitsToBuffer(before, b2), b2),
>+              " vs snprintf=", (snprintf(b1, sizeof(b1), "%g", before), b1),
>+              " Perfect=", PerfectDtoa(d), " ", (SixDigitsToBuffer(d, b2), b2),
>+              " vs snprintf=", (snprintf(b1, sizeof(b1), "%g", d), b1),
>+              " After.=.", PerfectDtoa(after), " ",
>+              (SixDigitsToBuffer(after, b2), b2),
>+              " vs snprintf=", (snprintf(b1, sizeof(b1), "%g", after), b1))
>+              .c_str());
>+    }
>+  }
>+}
>+
>+TEST(StrToInt32, Partial) {
>+  struct Int32TestLine {
>+    std::string input;
>+    bool status;
>+    int32_t value;
>+  };
>+  const int32_t int32_min = std::numeric_limits<int32_t>::min();
>+  const int32_t int32_max = std::numeric_limits<int32_t>::max();
>+  Int32TestLine int32_test_line[] = {
>+      {"", false, 0},
>+      {" ", false, 0},
>+      {"-", false, 0},
>+      {"123@@@", false, 123},
>+      {absl::StrCat(int32_min, int32_max), false, int32_min},
>+      {absl::StrCat(int32_max, int32_max), false, int32_max},
>+  };
>+
>+  for (const Int32TestLine& test_line : int32_test_line) {
>+    int32_t value = -2;
>+    bool status = safe_strto32_base(test_line.input, &value, 10);
>+    EXPECT_EQ(test_line.status, status) << test_line.input;
>+    EXPECT_EQ(test_line.value, value) << test_line.input;
>+    value = -2;
>+    status = safe_strto32_base(test_line.input, &value, 10);
>+    EXPECT_EQ(test_line.status, status) << test_line.input;
>+    EXPECT_EQ(test_line.value, value) << test_line.input;
>+    value = -2;
>+    status = safe_strto32_base(absl::string_view(test_line.input), &value, 10);
>+    EXPECT_EQ(test_line.status, status) << test_line.input;
>+    EXPECT_EQ(test_line.value, value) << test_line.input;
>+  }
>+}
>+
>+TEST(StrToUint32, Partial) {
>+  struct Uint32TestLine {
>+    std::string input;
>+    bool status;
>+    uint32_t value;
>+  };
>+  const uint32_t uint32_max = std::numeric_limits<uint32_t>::max();
>+  Uint32TestLine uint32_test_line[] = {
>+      {"", false, 0},
>+      {" ", false, 0},
>+      {"-", false, 0},
>+      {"123@@@", false, 123},
>+      {absl::StrCat(uint32_max, uint32_max), false, uint32_max},
>+  };
>+
>+  for (const Uint32TestLine& test_line : uint32_test_line) {
>+    uint32_t value = 2;
>+    bool status = safe_strtou32_base(test_line.input, &value, 10);
>+    EXPECT_EQ(test_line.status, status) << test_line.input;
>+    EXPECT_EQ(test_line.value, value) << test_line.input;
>+    value = 2;
>+    status = safe_strtou32_base(test_line.input, &value, 10);
>+    EXPECT_EQ(test_line.status, status) << test_line.input;
>+    EXPECT_EQ(test_line.value, value) << test_line.input;
>+    value = 2;
>+    status = safe_strtou32_base(absl::string_view(test_line.input), &value, 10);
>+    EXPECT_EQ(test_line.status, status) << test_line.input;
>+    EXPECT_EQ(test_line.value, value) << test_line.input;
>+  }
>+}
>+
>+TEST(StrToInt64, Partial) {
>+  struct Int64TestLine {
>+    std::string input;
>+    bool status;
>+    int64_t value;
>+  };
>+  const int64_t int64_min = std::numeric_limits<int64_t>::min();
>+  const int64_t int64_max = std::numeric_limits<int64_t>::max();
>+  Int64TestLine int64_test_line[] = {
>+      {"", false, 0},
>+      {" ", false, 0},
>+      {"-", false, 0},
>+      {"123@@@", false, 123},
>+      {absl::StrCat(int64_min, int64_max), false, int64_min},
>+      {absl::StrCat(int64_max, int64_max), false, int64_max},
>+  };
>+
>+  for (const Int64TestLine& test_line : int64_test_line) {
>+    int64_t value = -2;
>+    bool status = safe_strto64_base(test_line.input, &value, 10);
>+    EXPECT_EQ(test_line.status, status) << test_line.input;
>+    EXPECT_EQ(test_line.value, value) << test_line.input;
>+    value = -2;
>+    status = safe_strto64_base(test_line.input, &value, 10);
>+    EXPECT_EQ(test_line.status, status) << test_line.input;
>+    EXPECT_EQ(test_line.value, value) << test_line.input;
>+    value = -2;
>+    status = safe_strto64_base(absl::string_view(test_line.input), &value, 10);
>+    EXPECT_EQ(test_line.status, status) << test_line.input;
>+    EXPECT_EQ(test_line.value, value) << test_line.input;
>+  }
>+}
>+
>+TEST(StrToUint64, Partial) {
>+  struct Uint64TestLine {
>+    std::string input;
>+    bool status;
>+    uint64_t value;
>+  };
>+  const uint64_t uint64_max = std::numeric_limits<uint64_t>::max();
>+  Uint64TestLine uint64_test_line[] = {
>+      {"", false, 0},
>+      {" ", false, 0},
>+      {"-", false, 0},
>+      {"123@@@", false, 123},
>+      {absl::StrCat(uint64_max, uint64_max), false, uint64_max},
>+  };
>+
>+  for (const Uint64TestLine& test_line : uint64_test_line) {
>+    uint64_t value = 2;
>+    bool status = safe_strtou64_base(test_line.input, &value, 10);
>+    EXPECT_EQ(test_line.status, status) << test_line.input;
>+    EXPECT_EQ(test_line.value, value) << test_line.input;
>+    value = 2;
>+    status = safe_strtou64_base(test_line.input, &value, 10);
>+    EXPECT_EQ(test_line.status, status) << test_line.input;
>+    EXPECT_EQ(test_line.value, value) << test_line.input;
>+    value = 2;
>+    status = safe_strtou64_base(absl::string_view(test_line.input), &value, 10);
>+    EXPECT_EQ(test_line.status, status) << test_line.input;
>+    EXPECT_EQ(test_line.value, value) << test_line.input;
>+  }
>+}
>+
>+TEST(StrToInt32Base, PrefixOnly) {
>+  struct Int32TestLine {
>+    std::string input;
>+    bool status;
>+    int32_t value;
>+  };
>+  Int32TestLine int32_test_line[] = {
>+    { "", false, 0 },
>+    { "-", false, 0 },
>+    { "-0", true, 0 },
>+    { "0", true, 0 },
>+    { "0x", false, 0 },
>+    { "-0x", false, 0 },
>+  };
>+  const int base_array[] = { 0, 2, 8, 10, 16 };
>+
>+  for (const Int32TestLine& line : int32_test_line) {
>+    for (const int base : base_array) {
>+      int32_t value = 2;
>+      bool status = safe_strto32_base(line.input.c_str(), &value, base);
>+      EXPECT_EQ(line.status, status) << line.input << " " << base;
>+      EXPECT_EQ(line.value, value) << line.input << " " << base;
>+      value = 2;
>+      status = safe_strto32_base(line.input, &value, base);
>+      EXPECT_EQ(line.status, status) << line.input << " " << base;
>+      EXPECT_EQ(line.value, value) << line.input << " " << base;
>+      value = 2;
>+      status = safe_strto32_base(absl::string_view(line.input), &value, base);
>+      EXPECT_EQ(line.status, status) << line.input << " " << base;
>+      EXPECT_EQ(line.value, value) << line.input << " " << base;
>+    }
>+  }
>+}
>+
>+TEST(StrToUint32Base, PrefixOnly) {
>+  struct Uint32TestLine {
>+    std::string input;
>+    bool status;
>+    uint32_t value;
>+  };
>+  Uint32TestLine uint32_test_line[] = {
>+    { "", false, 0 },
>+    { "0", true, 0 },
>+    { "0x", false, 0 },
>+  };
>+  const int base_array[] = { 0, 2, 8, 10, 16 };
>+
>+  for (const Uint32TestLine& line : uint32_test_line) {
>+    for (const int base : base_array) {
>+      uint32_t value = 2;
>+      bool status = safe_strtou32_base(line.input.c_str(), &value, base);
>+      EXPECT_EQ(line.status, status) << line.input << " " << base;
>+      EXPECT_EQ(line.value, value) << line.input << " " << base;
>+      value = 2;
>+      status = safe_strtou32_base(line.input, &value, base);
>+      EXPECT_EQ(line.status, status) << line.input << " " << base;
>+      EXPECT_EQ(line.value, value) << line.input << " " << base;
>+      value = 2;
>+      status = safe_strtou32_base(absl::string_view(line.input), &value, base);
>+      EXPECT_EQ(line.status, status) << line.input << " " << base;
>+      EXPECT_EQ(line.value, value) << line.input << " " << base;
>+    }
>+  }
>+}
>+
>+TEST(StrToInt64Base, PrefixOnly) {
>+  struct Int64TestLine {
>+    std::string input;
>+    bool status;
>+    int64_t value;
>+  };
>+  Int64TestLine int64_test_line[] = {
>+    { "", false, 0 },
>+    { "-", false, 0 },
>+    { "-0", true, 0 },
>+    { "0", true, 0 },
>+    { "0x", false, 0 },
>+    { "-0x", false, 0 },
>+  };
>+  const int base_array[] = { 0, 2, 8, 10, 16 };
>+
>+  for (const Int64TestLine& line : int64_test_line) {
>+    for (const int base : base_array) {
>+      int64_t value = 2;
>+      bool status = safe_strto64_base(line.input.c_str(), &value, base);
>+      EXPECT_EQ(line.status, status) << line.input << " " << base;
>+      EXPECT_EQ(line.value, value) << line.input << " " << base;
>+      value = 2;
>+      status = safe_strto64_base(line.input, &value, base);
>+      EXPECT_EQ(line.status, status) << line.input << " " << base;
>+      EXPECT_EQ(line.value, value) << line.input << " " << base;
>+      value = 2;
>+      status = safe_strto64_base(absl::string_view(line.input), &value, base);
>+      EXPECT_EQ(line.status, status) << line.input << " " << base;
>+      EXPECT_EQ(line.value, value) << line.input << " " << base;
>+    }
>+  }
>+}
>+
>+TEST(StrToUint64Base, PrefixOnly) {
>+  struct Uint64TestLine {
>+    std::string input;
>+    bool status;
>+    uint64_t value;
>+  };
>+  Uint64TestLine uint64_test_line[] = {
>+    { "", false, 0 },
>+    { "0", true, 0 },
>+    { "0x", false, 0 },
>+  };
>+  const int base_array[] = { 0, 2, 8, 10, 16 };
>+
>+  for (const Uint64TestLine& line : uint64_test_line) {
>+    for (const int base : base_array) {
>+      uint64_t value = 2;
>+      bool status = safe_strtou64_base(line.input.c_str(), &value, base);
>+      EXPECT_EQ(line.status, status) << line.input << " " << base;
>+      EXPECT_EQ(line.value, value) << line.input << " " << base;
>+      value = 2;
>+      status = safe_strtou64_base(line.input, &value, base);
>+      EXPECT_EQ(line.status, status) << line.input << " " << base;
>+      EXPECT_EQ(line.value, value) << line.input << " " << base;
>+      value = 2;
>+      status = safe_strtou64_base(absl::string_view(line.input), &value, base);
>+      EXPECT_EQ(line.status, status) << line.input << " " << base;
>+      EXPECT_EQ(line.value, value) << line.input << " " << base;
>+    }
>+  }
>+}
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_cat.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_cat.cc
>new file mode 100644
>index 00000000000..3fe8c95eca9
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_cat.cc
>@@ -0,0 +1,239 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/strings/str_cat.h"
>+
>+#include <assert.h>
>+#include <algorithm>
>+#include <cstdint>
>+#include <cstring>
>+
>+#include "absl/strings/ascii.h"
>+#include "absl/strings/internal/resize_uninitialized.h"
>+
>+namespace absl {
>+
>+AlphaNum::AlphaNum(Hex hex) {
>+  char* const end = &digits_[numbers_internal::kFastToBufferSize];
>+  char* writer = end;
>+  uint64_t value = hex.value;
>+  static const char hexdigits[] = "0123456789abcdef";
>+  do {
>+    *--writer = hexdigits[value & 0xF];
>+    value >>= 4;
>+  } while (value != 0);
>+
>+  char* beg;
>+  if (end - writer < hex.width) {
>+    beg = end - hex.width;
>+    std::fill_n(beg, writer - beg, hex.fill);
>+  } else {
>+    beg = writer;
>+  }
>+
>+  piece_ = absl::string_view(beg, end - beg);
>+}
>+
>+AlphaNum::AlphaNum(Dec dec) {
>+  assert(dec.width <= numbers_internal::kFastToBufferSize);
>+  char* const end = &digits_[numbers_internal::kFastToBufferSize];
>+  char* const minfill = end - dec.width;
>+  char* writer = end;
>+  uint64_t value = dec.value;
>+  bool neg = dec.neg;
>+  while (value > 9) {
>+    *--writer = '0' + (value % 10);
>+    value /= 10;
>+  }
>+  *--writer = '0' + value;
>+  if (neg) *--writer = '-';
>+
>+  ptrdiff_t fillers = writer - minfill;
>+  if (fillers > 0) {
>+    // Tricky: if the fill character is ' ', then it's <fill><+/-><digits>
>+    // But...: if the fill character is '0', then it's <+/-><fill><digits>
>+    bool add_sign_again = false;
>+    if (neg && dec.fill == '0') {  // If filling with '0',
>+      ++writer;                    // ignore the sign we just added
>+      add_sign_again = true;       // and re-add the sign later.
>+    }
>+    writer -= fillers;
>+    std::fill_n(writer, fillers, dec.fill);
>+    if (add_sign_again) *--writer = '-';
>+  }
>+
>+  piece_ = absl::string_view(writer, end - writer);
>+}
>+
>+// ----------------------------------------------------------------------
>+// StrCat()
>+//    This merges the given strings or integers, with no delimiter.  This
>+//    is designed to be the fastest possible way to construct a std::string out
>+//    of a mix of raw C strings, StringPieces, strings, and integer values.
>+// ----------------------------------------------------------------------
>+
>+// Append is merely a version of memcpy that returns the address of the byte
>+// after the area just overwritten.
>+static char* Append(char* out, const AlphaNum& x) {
>+  // memcpy is allowed to overwrite arbitrary memory, so doing this after the
>+  // call would force an extra fetch of x.size().
>+  char* after = out + x.size();
>+  memcpy(out, x.data(), x.size());
>+  return after;
>+}
>+
>+std::string StrCat(const AlphaNum& a, const AlphaNum& b) {
>+  std::string result;
>+  absl::strings_internal::STLStringResizeUninitialized(&result,
>+                                                       a.size() + b.size());
>+  char* const begin = &*result.begin();
>+  char* out = begin;
>+  out = Append(out, a);
>+  out = Append(out, b);
>+  assert(out == begin + result.size());
>+  return result;
>+}
>+
>+std::string StrCat(const AlphaNum& a, const AlphaNum& b, const AlphaNum& c) {
>+  std::string result;
>+  strings_internal::STLStringResizeUninitialized(
>+      &result, a.size() + b.size() + c.size());
>+  char* const begin = &*result.begin();
>+  char* out = begin;
>+  out = Append(out, a);
>+  out = Append(out, b);
>+  out = Append(out, c);
>+  assert(out == begin + result.size());
>+  return result;
>+}
>+
>+std::string StrCat(const AlphaNum& a, const AlphaNum& b, const AlphaNum& c,
>+              const AlphaNum& d) {
>+  std::string result;
>+  strings_internal::STLStringResizeUninitialized(
>+      &result, a.size() + b.size() + c.size() + d.size());
>+  char* const begin = &*result.begin();
>+  char* out = begin;
>+  out = Append(out, a);
>+  out = Append(out, b);
>+  out = Append(out, c);
>+  out = Append(out, d);
>+  assert(out == begin + result.size());
>+  return result;
>+}
>+
>+namespace strings_internal {
>+
>+// Do not call directly - these are not part of the public API.
>+std::string CatPieces(std::initializer_list<absl::string_view> pieces) {
>+  std::string result;
>+  size_t total_size = 0;
>+  for (const absl::string_view piece : pieces) total_size += piece.size();
>+  strings_internal::STLStringResizeUninitialized(&result, total_size);
>+
>+  char* const begin = &*result.begin();
>+  char* out = begin;
>+  for (const absl::string_view piece : pieces) {
>+    const size_t this_size = piece.size();
>+    memcpy(out, piece.data(), this_size);
>+    out += this_size;
>+  }
>+  assert(out == begin + result.size());
>+  return result;
>+}
>+
>+// It's possible to call StrAppend with an absl::string_view that is itself a
>+// fragment of the std::string we're appending to.  However the results of this are
>+// random. Therefore, check for this in debug mode.  Use unsigned math so we
>+// only have to do one comparison. Note, there's an exception case: appending an
>+// empty std::string is always allowed.
>+#define ASSERT_NO_OVERLAP(dest, src) \
>+  assert(((src).size() == 0) ||      \
>+         (uintptr_t((src).data() - (dest).data()) > uintptr_t((dest).size())))
>+
>+void AppendPieces(std::string* dest,
>+                  std::initializer_list<absl::string_view> pieces) {
>+  size_t old_size = dest->size();
>+  size_t total_size = old_size;
>+  for (const absl::string_view piece : pieces) {
>+    ASSERT_NO_OVERLAP(*dest, piece);
>+    total_size += piece.size();
>+  }
>+  strings_internal::STLStringResizeUninitialized(dest, total_size);
>+
>+  char* const begin = &*dest->begin();
>+  char* out = begin + old_size;
>+  for (const absl::string_view piece : pieces) {
>+    const size_t this_size = piece.size();
>+    memcpy(out, piece.data(), this_size);
>+    out += this_size;
>+  }
>+  assert(out == begin + dest->size());
>+}
>+
>+}  // namespace strings_internal
>+
>+void StrAppend(std::string* dest, const AlphaNum& a) {
>+  ASSERT_NO_OVERLAP(*dest, a);
>+  dest->append(a.data(), a.size());
>+}
>+
>+void StrAppend(std::string* dest, const AlphaNum& a, const AlphaNum& b) {
>+  ASSERT_NO_OVERLAP(*dest, a);
>+  ASSERT_NO_OVERLAP(*dest, b);
>+  std::string::size_type old_size = dest->size();
>+  strings_internal::STLStringResizeUninitialized(
>+      dest, old_size + a.size() + b.size());
>+  char* const begin = &*dest->begin();
>+  char* out = begin + old_size;
>+  out = Append(out, a);
>+  out = Append(out, b);
>+  assert(out == begin + dest->size());
>+}
>+
>+void StrAppend(std::string* dest, const AlphaNum& a, const AlphaNum& b,
>+               const AlphaNum& c) {
>+  ASSERT_NO_OVERLAP(*dest, a);
>+  ASSERT_NO_OVERLAP(*dest, b);
>+  ASSERT_NO_OVERLAP(*dest, c);
>+  std::string::size_type old_size = dest->size();
>+  strings_internal::STLStringResizeUninitialized(
>+      dest, old_size + a.size() + b.size() + c.size());
>+  char* const begin = &*dest->begin();
>+  char* out = begin + old_size;
>+  out = Append(out, a);
>+  out = Append(out, b);
>+  out = Append(out, c);
>+  assert(out == begin + dest->size());
>+}
>+
>+void StrAppend(std::string* dest, const AlphaNum& a, const AlphaNum& b,
>+               const AlphaNum& c, const AlphaNum& d) {
>+  ASSERT_NO_OVERLAP(*dest, a);
>+  ASSERT_NO_OVERLAP(*dest, b);
>+  ASSERT_NO_OVERLAP(*dest, c);
>+  ASSERT_NO_OVERLAP(*dest, d);
>+  std::string::size_type old_size = dest->size();
>+  strings_internal::STLStringResizeUninitialized(
>+      dest, old_size + a.size() + b.size() + c.size() + d.size());
>+  char* const begin = &*dest->begin();
>+  char* out = begin + old_size;
>+  out = Append(out, a);
>+  out = Append(out, b);
>+  out = Append(out, c);
>+  out = Append(out, d);
>+  assert(out == begin + dest->size());
>+}
>+
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_cat.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_cat.h
>new file mode 100644
>index 00000000000..e5501a5012e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_cat.h
>@@ -0,0 +1,385 @@
>+//
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// -----------------------------------------------------------------------------
>+// File: str_cat.h
>+// -----------------------------------------------------------------------------
>+//
>+// This package contains functions for efficiently concatenating and appending
>+// strings: `StrCat()` and `StrAppend()`. Most of the work within these routines
>+// is actually handled through use of a special AlphaNum type, which was
>+// designed to be used as a parameter type that efficiently manages conversion
>+// to strings and avoids copies in the above operations.
>+//
>+// Any routine accepting either a std::string or a number may accept `AlphaNum`.
>+// The basic idea is that by accepting a `const AlphaNum &` as an argument
>+// to your function, your callers will automagically convert bools, integers,
>+// and floating point values to strings for you.
>+//
>+// NOTE: Use of `AlphaNum` outside of the //absl/strings package is unsupported
>+// except for the specific case of function parameters of type `AlphaNum` or
>+// `const AlphaNum &`. In particular, instantiating `AlphaNum` directly as a
>+// stack variable is not supported.
>+//
>+// Conversion from 8-bit values is not accepted because, if it were, then an
>+// attempt to pass ':' instead of ":" might result in a 58 ending up in your
>+// result.
>+//
>+// Bools convert to "0" or "1".
>+//
>+// Floating point numbers are formatted with six-digit precision, which is
>+// the default for "std::cout <<" or printf "%g" (the same as "%.6g").
>+//
>+//
>+// You can convert to hexadecimal output rather than decimal output using the
>+// `Hex` type contained here. To do so, pass `Hex(my_int)` as a parameter to
>+// `StrCat()` or `StrAppend()`. You may specify a minimum hex field width using
>+// a `PadSpec` enum.
>+//
>+// -----------------------------------------------------------------------------
>+
>+#ifndef ABSL_STRINGS_STR_CAT_H_
>+#define ABSL_STRINGS_STR_CAT_H_
>+
>+#include <array>
>+#include <cstdint>
>+#include <string>
>+#include <type_traits>
>+
>+#include "absl/base/port.h"
>+#include "absl/strings/numbers.h"
>+#include "absl/strings/string_view.h"
>+
>+namespace absl {
>+
>+namespace strings_internal {
>+// AlphaNumBuffer allows a way to pass a std::string to StrCat without having to do
>+// memory allocation.  It is simply a pair of a fixed-size character array, and
>+// a size.  Please don't use outside of absl, yet.
>+template <size_t max_size>
>+struct AlphaNumBuffer {
>+  std::array<char, max_size> data;
>+  size_t size;
>+};
>+
>+}  // namespace strings_internal
>+
>+// Enum that specifies the number of significant digits to return in a `Hex` or
>+// `Dec` conversion and fill character to use. A `kZeroPad2` value, for example,
>+// would produce hexadecimal strings such as "0A","0F" and a 'kSpacePad5' value
>+// would produce hexadecimal strings such as "    A","    F".
>+enum PadSpec : uint8_t {
>+  kNoPad = 1,
>+  kZeroPad2,
>+  kZeroPad3,
>+  kZeroPad4,
>+  kZeroPad5,
>+  kZeroPad6,
>+  kZeroPad7,
>+  kZeroPad8,
>+  kZeroPad9,
>+  kZeroPad10,
>+  kZeroPad11,
>+  kZeroPad12,
>+  kZeroPad13,
>+  kZeroPad14,
>+  kZeroPad15,
>+  kZeroPad16,
>+
>+  kSpacePad2 = kZeroPad2 + 64,
>+  kSpacePad3,
>+  kSpacePad4,
>+  kSpacePad5,
>+  kSpacePad6,
>+  kSpacePad7,
>+  kSpacePad8,
>+  kSpacePad9,
>+  kSpacePad10,
>+  kSpacePad11,
>+  kSpacePad12,
>+  kSpacePad13,
>+  kSpacePad14,
>+  kSpacePad15,
>+  kSpacePad16,
>+};
>+
>+// -----------------------------------------------------------------------------
>+// Hex
>+// -----------------------------------------------------------------------------
>+//
>+// `Hex` stores a set of hexadecimal std::string conversion parameters for use
>+// within `AlphaNum` std::string conversions.
>+struct Hex {
>+  uint64_t value;
>+  uint8_t width;
>+  char fill;
>+
>+  template <typename Int>
>+  explicit Hex(
>+      Int v, PadSpec spec = absl::kNoPad,
>+      typename std::enable_if<sizeof(Int) == 1 &&
>+                              !std::is_pointer<Int>::value>::type* = nullptr)
>+      : Hex(spec, static_cast<uint8_t>(v)) {}
>+  template <typename Int>
>+  explicit Hex(
>+      Int v, PadSpec spec = absl::kNoPad,
>+      typename std::enable_if<sizeof(Int) == 2 &&
>+                              !std::is_pointer<Int>::value>::type* = nullptr)
>+      : Hex(spec, static_cast<uint16_t>(v)) {}
>+  template <typename Int>
>+  explicit Hex(
>+      Int v, PadSpec spec = absl::kNoPad,
>+      typename std::enable_if<sizeof(Int) == 4 &&
>+                              !std::is_pointer<Int>::value>::type* = nullptr)
>+      : Hex(spec, static_cast<uint32_t>(v)) {}
>+  template <typename Int>
>+  explicit Hex(
>+      Int v, PadSpec spec = absl::kNoPad,
>+      typename std::enable_if<sizeof(Int) == 8 &&
>+                              !std::is_pointer<Int>::value>::type* = nullptr)
>+      : Hex(spec, static_cast<uint64_t>(v)) {}
>+  template <typename Pointee>
>+  explicit Hex(Pointee* v, PadSpec spec = absl::kNoPad)
>+      : Hex(spec, reinterpret_cast<uintptr_t>(v)) {}
>+
>+ private:
>+  Hex(PadSpec spec, uint64_t v)
>+      : value(v),
>+        width(spec == absl::kNoPad
>+                  ? 1
>+                  : spec >= absl::kSpacePad2 ? spec - absl::kSpacePad2 + 2
>+                                             : spec - absl::kZeroPad2 + 2),
>+        fill(spec >= absl::kSpacePad2 ? ' ' : '0') {}
>+};
>+
>+// -----------------------------------------------------------------------------
>+// Dec
>+// -----------------------------------------------------------------------------
>+//
>+// `Dec` stores a set of decimal std::string conversion parameters for use
>+// within `AlphaNum` std::string conversions.  Dec is slower than the default
>+// integer conversion, so use it only if you need padding.
>+struct Dec {
>+  uint64_t value;
>+  uint8_t width;
>+  char fill;
>+  bool neg;
>+
>+  template <typename Int>
>+  explicit Dec(Int v, PadSpec spec = absl::kNoPad,
>+               typename std::enable_if<(sizeof(Int) <= 8)>::type* = nullptr)
>+      : value(v >= 0 ? static_cast<uint64_t>(v)
>+                     : uint64_t{0} - static_cast<uint64_t>(v)),
>+        width(spec == absl::kNoPad
>+                  ? 1
>+                  : spec >= absl::kSpacePad2 ? spec - absl::kSpacePad2 + 2
>+                                             : spec - absl::kZeroPad2 + 2),
>+        fill(spec >= absl::kSpacePad2 ? ' ' : '0'),
>+        neg(v < 0) {}
>+};
>+
>+// -----------------------------------------------------------------------------
>+// AlphaNum
>+// -----------------------------------------------------------------------------
>+//
>+// The `AlphaNum` class acts as the main parameter type for `StrCat()` and
>+// `StrAppend()`, providing efficient conversion of numeric, boolean, and
>+// hexadecimal values (through the `Hex` type) into strings.
>+
>+class AlphaNum {
>+ public:
>+  // No bool ctor -- bools convert to an integral type.
>+  // A bool ctor would also convert incoming pointers (bletch).
>+
>+  AlphaNum(int x)  // NOLINT(runtime/explicit)
>+      : piece_(digits_,
>+               numbers_internal::FastIntToBuffer(x, digits_) - &digits_[0]) {}
>+  AlphaNum(unsigned int x)  // NOLINT(runtime/explicit)
>+      : piece_(digits_,
>+               numbers_internal::FastIntToBuffer(x, digits_) - &digits_[0]) {}
>+  AlphaNum(long x)  // NOLINT(*)
>+      : piece_(digits_,
>+               numbers_internal::FastIntToBuffer(x, digits_) - &digits_[0]) {}
>+  AlphaNum(unsigned long x)  // NOLINT(*)
>+      : piece_(digits_,
>+               numbers_internal::FastIntToBuffer(x, digits_) - &digits_[0]) {}
>+  AlphaNum(long long x)  // NOLINT(*)
>+      : piece_(digits_,
>+               numbers_internal::FastIntToBuffer(x, digits_) - &digits_[0]) {}
>+  AlphaNum(unsigned long long x)  // NOLINT(*)
>+      : piece_(digits_,
>+               numbers_internal::FastIntToBuffer(x, digits_) - &digits_[0]) {}
>+
>+  AlphaNum(float f)  // NOLINT(runtime/explicit)
>+      : piece_(digits_, numbers_internal::SixDigitsToBuffer(f, digits_)) {}
>+  AlphaNum(double f)  // NOLINT(runtime/explicit)
>+      : piece_(digits_, numbers_internal::SixDigitsToBuffer(f, digits_)) {}
>+
>+  AlphaNum(Hex hex);  // NOLINT(runtime/explicit)
>+  AlphaNum(Dec dec);  // NOLINT(runtime/explicit)
>+
>+  template <size_t size>
>+  AlphaNum(  // NOLINT(runtime/explicit)
>+      const strings_internal::AlphaNumBuffer<size>& buf)
>+      : piece_(&buf.data[0], buf.size) {}
>+
>+  AlphaNum(const char* c_str) : piece_(c_str) {}  // NOLINT(runtime/explicit)
>+  AlphaNum(absl::string_view pc) : piece_(pc) {}  // NOLINT(runtime/explicit)
>+  template <typename Allocator>
>+  AlphaNum(  // NOLINT(runtime/explicit)
>+      const std::basic_string<char, std::char_traits<char>, Allocator>& str)
>+      : piece_(str) {}
>+
>+  // Use std::string literals ":" instead of character literals ':'.
>+  AlphaNum(char c) = delete;  // NOLINT(runtime/explicit)
>+
>+  AlphaNum(const AlphaNum&) = delete;
>+  AlphaNum& operator=(const AlphaNum&) = delete;
>+
>+  absl::string_view::size_type size() const { return piece_.size(); }
>+  const char* data() const { return piece_.data(); }
>+  absl::string_view Piece() const { return piece_; }
>+
>+  // Normal enums are already handled by the integer formatters.
>+  // This overload matches only scoped enums.
>+  template <typename T,
>+            typename = typename std::enable_if<
>+                std::is_enum<T>{} && !std::is_convertible<T, int>{}>::type>
>+  AlphaNum(T e)  // NOLINT(runtime/explicit)
>+      : AlphaNum(static_cast<typename std::underlying_type<T>::type>(e)) {}
>+
>+ private:
>+  absl::string_view piece_;
>+  char digits_[numbers_internal::kFastToBufferSize];
>+};
>+
>+// -----------------------------------------------------------------------------
>+// StrCat()
>+// -----------------------------------------------------------------------------
>+//
>+// Merges given strings or numbers, using no delimiter(s).
>+//
>+// `StrCat()` is designed to be the fastest possible way to construct a std::string
>+// out of a mix of raw C strings, string_views, strings, bool values,
>+// and numeric values.
>+//
>+// Don't use `StrCat()` for user-visible strings. The localization process
>+// works poorly on strings built up out of fragments.
>+//
>+// For clarity and performance, don't use `StrCat()` when appending to a
>+// std::string. Use `StrAppend()` instead. In particular, avoid using any of these
>+// (anti-)patterns:
>+//
>+//   str.append(StrCat(...))
>+//   str += StrCat(...)
>+//   str = StrCat(str, ...)
>+//
>+// The last case is the worst, with a potential to change a loop
>+// from a linear time operation with O(1) dynamic allocations into a
>+// quadratic time operation with O(n) dynamic allocations.
>+//
>+// See `StrAppend()` below for more information.
>+
>+namespace strings_internal {
>+
>+// Do not call directly - this is not part of the public API.
>+std::string CatPieces(std::initializer_list<absl::string_view> pieces);
>+void AppendPieces(std::string* dest,
>+                  std::initializer_list<absl::string_view> pieces);
>+
>+}  // namespace strings_internal
>+
>+ABSL_MUST_USE_RESULT inline std::string StrCat() { return std::string(); }
>+
>+ABSL_MUST_USE_RESULT inline std::string StrCat(const AlphaNum& a) {
>+  return std::string(a.data(), a.size());
>+}
>+
>+ABSL_MUST_USE_RESULT std::string StrCat(const AlphaNum& a, const AlphaNum& b);
>+ABSL_MUST_USE_RESULT std::string StrCat(const AlphaNum& a, const AlphaNum& b,
>+                                   const AlphaNum& c);
>+ABSL_MUST_USE_RESULT std::string StrCat(const AlphaNum& a, const AlphaNum& b,
>+                                   const AlphaNum& c, const AlphaNum& d);
>+
>+// Support 5 or more arguments
>+template <typename... AV>
>+ABSL_MUST_USE_RESULT inline std::string StrCat(const AlphaNum& a, const AlphaNum& b,
>+                                          const AlphaNum& c, const AlphaNum& d,
>+                                          const AlphaNum& e,
>+                                          const AV&... args) {
>+  return strings_internal::CatPieces(
>+      {a.Piece(), b.Piece(), c.Piece(), d.Piece(), e.Piece(),
>+       static_cast<const AlphaNum&>(args).Piece()...});
>+}
>+
>+// -----------------------------------------------------------------------------
>+// StrAppend()
>+// -----------------------------------------------------------------------------
>+//
>+// Appends a std::string or set of strings to an existing std::string, in a similar
>+// fashion to `StrCat()`.
>+//
>+// WARNING: `StrAppend(&str, a, b, c, ...)` requires that none of the
>+// a, b, c, parameters be a reference into str. For speed, `StrAppend()` does
>+// not try to check each of its input arguments to be sure that they are not
>+// a subset of the std::string being appended to. That is, while this will work:
>+//
>+//   std::string s = "foo";
>+//   s += s;
>+//
>+// This output is undefined:
>+//
>+//   std::string s = "foo";
>+//   StrAppend(&s, s);
>+//
>+// This output is undefined as well, since `absl::string_view` does not own its
>+// data:
>+//
>+//   std::string s = "foobar";
>+//   absl::string_view p = s;
>+//   StrAppend(&s, p);
>+
>+inline void StrAppend(std::string*) {}
>+void StrAppend(std::string* dest, const AlphaNum& a);
>+void StrAppend(std::string* dest, const AlphaNum& a, const AlphaNum& b);
>+void StrAppend(std::string* dest, const AlphaNum& a, const AlphaNum& b,
>+               const AlphaNum& c);
>+void StrAppend(std::string* dest, const AlphaNum& a, const AlphaNum& b,
>+               const AlphaNum& c, const AlphaNum& d);
>+
>+// Support 5 or more arguments
>+template <typename... AV>
>+inline void StrAppend(std::string* dest, const AlphaNum& a, const AlphaNum& b,
>+                      const AlphaNum& c, const AlphaNum& d, const AlphaNum& e,
>+                      const AV&... args) {
>+  strings_internal::AppendPieces(
>+      dest, {a.Piece(), b.Piece(), c.Piece(), d.Piece(), e.Piece(),
>+             static_cast<const AlphaNum&>(args).Piece()...});
>+}
>+
>+// Helper function for the future StrCat default floating-point format, %.6g
>+// This is fast.
>+inline strings_internal::AlphaNumBuffer<
>+    numbers_internal::kSixDigitsToBufferSize>
>+SixDigits(double d) {
>+  strings_internal::AlphaNumBuffer<numbers_internal::kSixDigitsToBufferSize>
>+      result;
>+  result.size = numbers_internal::SixDigitsToBuffer(d, &result.data[0]);
>+  return result;
>+}
>+
>+}  // namespace absl
>+
>+#endif  // ABSL_STRINGS_STR_CAT_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_cat_benchmark.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_cat_benchmark.cc
>new file mode 100644
>index 00000000000..b6df9e309c4
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_cat_benchmark.cc
>@@ -0,0 +1,140 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/strings/str_cat.h"
>+
>+#include <cstdint>
>+#include <string>
>+
>+#include "benchmark/benchmark.h"
>+#include "absl/strings/substitute.h"
>+
>+namespace {
>+
>+const char kStringOne[] = "Once Upon A Time, ";
>+const char kStringTwo[] = "There was a std::string benchmark";
>+
>+// We want to include negative numbers in the benchmark, so this function
>+// is used to count 0, 1, -1, 2, -2, 3, -3, ...
>+inline int IncrementAlternatingSign(int i) {
>+  return i > 0 ? -i : 1 - i;
>+}
>+
>+void BM_Sum_By_StrCat(benchmark::State& state) {
>+  int i = 0;
>+  char foo[100];
>+  for (auto _ : state) {
>+    // NOLINTNEXTLINE(runtime/printf)
>+    strcpy(foo, absl::StrCat(kStringOne, i, kStringTwo, i * 65536ULL).c_str());
>+    int sum = 0;
>+    for (char* f = &foo[0]; *f != 0; ++f) {
>+      sum += *f;
>+    }
>+    benchmark::DoNotOptimize(sum);
>+    i = IncrementAlternatingSign(i);
>+  }
>+}
>+BENCHMARK(BM_Sum_By_StrCat);
>+
>+void BM_StrCat_By_snprintf(benchmark::State& state) {
>+  int i = 0;
>+  char on_stack[1000];
>+  for (auto _ : state) {
>+    snprintf(on_stack, sizeof(on_stack), "%s %s:%d", kStringOne, kStringTwo, i);
>+    i = IncrementAlternatingSign(i);
>+  }
>+}
>+BENCHMARK(BM_StrCat_By_snprintf);
>+
>+void BM_StrCat_By_Strings(benchmark::State& state) {
>+  int i = 0;
>+  for (auto _ : state) {
>+    std::string result =
>+        std::string(kStringOne) + " " + kStringTwo + ":" + absl::StrCat(i);
>+    benchmark::DoNotOptimize(result);
>+    i = IncrementAlternatingSign(i);
>+  }
>+}
>+BENCHMARK(BM_StrCat_By_Strings);
>+
>+void BM_StrCat_By_StringOpPlus(benchmark::State& state) {
>+  int i = 0;
>+  for (auto _ : state) {
>+    std::string result = kStringOne;
>+    result += " ";
>+    result += kStringTwo;
>+    result += ":";
>+    result += absl::StrCat(i);
>+    benchmark::DoNotOptimize(result);
>+    i = IncrementAlternatingSign(i);
>+  }
>+}
>+BENCHMARK(BM_StrCat_By_StringOpPlus);
>+
>+void BM_StrCat_By_StrCat(benchmark::State& state) {
>+  int i = 0;
>+  for (auto _ : state) {
>+    std::string result = absl::StrCat(kStringOne, " ", kStringTwo, ":", i);
>+    benchmark::DoNotOptimize(result);
>+    i = IncrementAlternatingSign(i);
>+  }
>+}
>+BENCHMARK(BM_StrCat_By_StrCat);
>+
>+void BM_HexCat_By_StrCat(benchmark::State& state) {
>+  int i = 0;
>+  for (auto _ : state) {
>+    std::string result =
>+        absl::StrCat(kStringOne, " ", absl::Hex(int64_t{i} + 0x10000000));
>+    benchmark::DoNotOptimize(result);
>+    i = IncrementAlternatingSign(i);
>+  }
>+}
>+BENCHMARK(BM_HexCat_By_StrCat);
>+
>+void BM_HexCat_By_Substitute(benchmark::State& state) {
>+  int i = 0;
>+  for (auto _ : state) {
>+    std::string result = absl::Substitute(
>+        "$0 $1", kStringOne, reinterpret_cast<void*>(int64_t{i} + 0x10000000));
>+    benchmark::DoNotOptimize(result);
>+    i = IncrementAlternatingSign(i);
>+  }
>+}
>+BENCHMARK(BM_HexCat_By_Substitute);
>+
>+void BM_FloatToString_By_StrCat(benchmark::State& state) {
>+  int i = 0;
>+  float foo = 0.0f;
>+  for (auto _ : state) {
>+    std::string result = absl::StrCat(foo += 1.001f, " != ", int64_t{i});
>+    benchmark::DoNotOptimize(result);
>+    i = IncrementAlternatingSign(i);
>+  }
>+}
>+BENCHMARK(BM_FloatToString_By_StrCat);
>+
>+void BM_DoubleToString_By_SixDigits(benchmark::State& state) {
>+  int i = 0;
>+  double foo = 0.0;
>+  for (auto _ : state) {
>+    std::string result =
>+        absl::StrCat(absl::SixDigits(foo += 1.001), " != ", int64_t{i});
>+    benchmark::DoNotOptimize(result);
>+    i = IncrementAlternatingSign(i);
>+  }
>+}
>+BENCHMARK(BM_DoubleToString_By_SixDigits);
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_cat_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_cat_test.cc
>new file mode 100644
>index 00000000000..e9d67690638
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_cat_test.cc
>@@ -0,0 +1,549 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+// Unit tests for all str_cat.h functions
>+
>+#include "absl/strings/str_cat.h"
>+
>+#include <cstdint>
>+#include <string>
>+
>+#include "gtest/gtest.h"
>+#include "absl/strings/substitute.h"
>+
>+#ifdef __ANDROID__
>+// Android assert messages only go to system log, so death tests cannot inspect
>+// the message for matching.
>+#define ABSL_EXPECT_DEBUG_DEATH(statement, regex) \
>+  EXPECT_DEBUG_DEATH(statement, ".*")
>+#else
>+#define ABSL_EXPECT_DEBUG_DEATH EXPECT_DEBUG_DEATH
>+#endif
>+
>+namespace {
>+
>+// Test absl::StrCat of ints and longs of various sizes and signdedness.
>+TEST(StrCat, Ints) {
>+  const short s = -1;  // NOLINT(runtime/int)
>+  const uint16_t us = 2;
>+  const int i = -3;
>+  const unsigned int ui = 4;
>+  const long l = -5;                 // NOLINT(runtime/int)
>+  const unsigned long ul = 6;        // NOLINT(runtime/int)
>+  const long long ll = -7;           // NOLINT(runtime/int)
>+  const unsigned long long ull = 8;  // NOLINT(runtime/int)
>+  const ptrdiff_t ptrdiff = -9;
>+  const size_t size = 10;
>+  const intptr_t intptr = -12;
>+  const uintptr_t uintptr = 13;
>+  std::string answer;
>+  answer = absl::StrCat(s, us);
>+  EXPECT_EQ(answer, "-12");
>+  answer = absl::StrCat(i, ui);
>+  EXPECT_EQ(answer, "-34");
>+  answer = absl::StrCat(l, ul);
>+  EXPECT_EQ(answer, "-56");
>+  answer = absl::StrCat(ll, ull);
>+  EXPECT_EQ(answer, "-78");
>+  answer = absl::StrCat(ptrdiff, size);
>+  EXPECT_EQ(answer, "-910");
>+  answer = absl::StrCat(ptrdiff, intptr);
>+  EXPECT_EQ(answer, "-9-12");
>+  answer = absl::StrCat(uintptr, 0);
>+  EXPECT_EQ(answer, "130");
>+}
>+
>+TEST(StrCat, Enums) {
>+  enum SmallNumbers { One = 1, Ten = 10 } e = Ten;
>+  EXPECT_EQ("10", absl::StrCat(e));
>+  EXPECT_EQ("-5", absl::StrCat(SmallNumbers(-5)));
>+
>+  enum class Option { Boxers = 1, Briefs = -1 };
>+
>+  EXPECT_EQ("-1", absl::StrCat(Option::Briefs));
>+
>+  enum class Airplane : uint64_t {
>+    Airbus = 1,
>+    Boeing = 1000,
>+    Canary = 10000000000  // too big for "int"
>+  };
>+
>+  EXPECT_EQ("10000000000", absl::StrCat(Airplane::Canary));
>+
>+  enum class TwoGig : int32_t {
>+    TwoToTheZero = 1,
>+    TwoToTheSixteenth = 1 << 16,
>+    TwoToTheThirtyFirst = INT32_MIN
>+  };
>+  EXPECT_EQ("65536", absl::StrCat(TwoGig::TwoToTheSixteenth));
>+  EXPECT_EQ("-2147483648", absl::StrCat(TwoGig::TwoToTheThirtyFirst));
>+  EXPECT_EQ("-1", absl::StrCat(static_cast<TwoGig>(-1)));
>+
>+  enum class FourGig : uint32_t {
>+    TwoToTheZero = 1,
>+    TwoToTheSixteenth = 1 << 16,
>+    TwoToTheThirtyFirst = 1U << 31  // too big for "int"
>+  };
>+  EXPECT_EQ("65536", absl::StrCat(FourGig::TwoToTheSixteenth));
>+  EXPECT_EQ("2147483648", absl::StrCat(FourGig::TwoToTheThirtyFirst));
>+  EXPECT_EQ("4294967295", absl::StrCat(static_cast<FourGig>(-1)));
>+
>+  EXPECT_EQ("10000000000", absl::StrCat(Airplane::Canary));
>+}
>+
>+TEST(StrCat, Basics) {
>+  std::string result;
>+
>+  std::string strs[] = {
>+    "Hello",
>+    "Cruel",
>+    "World"
>+  };
>+
>+  std::string stdstrs[] = {
>+    "std::Hello",
>+    "std::Cruel",
>+    "std::World"
>+  };
>+
>+  absl::string_view pieces[] = {"Hello", "Cruel", "World"};
>+
>+  const char* c_strs[] = {
>+    "Hello",
>+    "Cruel",
>+    "World"
>+  };
>+
>+  int32_t i32s[] = {'H', 'C', 'W'};
>+  uint64_t ui64s[] = {12345678910LL, 10987654321LL};
>+
>+  EXPECT_EQ(absl::StrCat(), "");
>+
>+  result = absl::StrCat(false, true, 2, 3);
>+  EXPECT_EQ(result, "0123");
>+
>+  result = absl::StrCat(-1);
>+  EXPECT_EQ(result, "-1");
>+
>+  result = absl::StrCat(absl::SixDigits(0.5));
>+  EXPECT_EQ(result, "0.5");
>+
>+  result = absl::StrCat(strs[1], pieces[2]);
>+  EXPECT_EQ(result, "CruelWorld");
>+
>+  result = absl::StrCat(stdstrs[1], " ", stdstrs[2]);
>+  EXPECT_EQ(result, "std::Cruel std::World");
>+
>+  result = absl::StrCat(strs[0], ", ", pieces[2]);
>+  EXPECT_EQ(result, "Hello, World");
>+
>+  result = absl::StrCat(strs[0], ", ", strs[1], " ", strs[2], "!");
>+  EXPECT_EQ(result, "Hello, Cruel World!");
>+
>+  result = absl::StrCat(pieces[0], ", ", pieces[1], " ", pieces[2]);
>+  EXPECT_EQ(result, "Hello, Cruel World");
>+
>+  result = absl::StrCat(c_strs[0], ", ", c_strs[1], " ", c_strs[2]);
>+  EXPECT_EQ(result, "Hello, Cruel World");
>+
>+  result = absl::StrCat("ASCII ", i32s[0], ", ", i32s[1], " ", i32s[2], "!");
>+  EXPECT_EQ(result, "ASCII 72, 67 87!");
>+
>+  result = absl::StrCat(ui64s[0], ", ", ui64s[1], "!");
>+  EXPECT_EQ(result, "12345678910, 10987654321!");
>+
>+  std::string one = "1";  // Actually, it's the size of this std::string that we want; a
>+                     // 64-bit build distinguishes between size_t and uint64_t,
>+                     // even though they're both unsigned 64-bit values.
>+  result = absl::StrCat("And a ", one.size(), " and a ",
>+                        &result[2] - &result[0], " and a ", one, " 2 3 4", "!");
>+  EXPECT_EQ(result, "And a 1 and a 2 and a 1 2 3 4!");
>+
>+  // result = absl::StrCat("Single chars won't compile", '!');
>+  // result = absl::StrCat("Neither will nullptrs", nullptr);
>+  result =
>+      absl::StrCat("To output a char by ASCII/numeric value, use +: ", '!' + 0);
>+  EXPECT_EQ(result, "To output a char by ASCII/numeric value, use +: 33");
>+
>+  float f = 100000.5;
>+  result = absl::StrCat("A hundred K and a half is ", absl::SixDigits(f));
>+  EXPECT_EQ(result, "A hundred K and a half is 100000");
>+
>+  f = 100001.5;
>+  result =
>+      absl::StrCat("A hundred K and one and a half is ", absl::SixDigits(f));
>+  EXPECT_EQ(result, "A hundred K and one and a half is 100002");
>+
>+  double d = 100000.5;
>+  d *= d;
>+  result =
>+      absl::StrCat("A hundred K and a half squared is ", absl::SixDigits(d));
>+  EXPECT_EQ(result, "A hundred K and a half squared is 1.00001e+10");
>+
>+  result = absl::StrCat(1, 2, 333, 4444, 55555, 666666, 7777777, 88888888,
>+                        999999999);
>+  EXPECT_EQ(result, "12333444455555666666777777788888888999999999");
>+}
>+
>+// A minimal allocator that uses malloc().
>+template <typename T>
>+struct Mallocator {
>+  typedef T value_type;
>+  typedef size_t size_type;
>+  typedef ptrdiff_t difference_type;
>+  typedef T* pointer;
>+  typedef const T* const_pointer;
>+  typedef T& reference;
>+  typedef const T& const_reference;
>+
>+  size_type max_size() const {
>+    return size_t(std::numeric_limits<size_type>::max()) / sizeof(value_type);
>+  }
>+  template <typename U>
>+  struct rebind {
>+    typedef Mallocator<U> other;
>+  };
>+  Mallocator() = default;
>+  template <class U>
>+  Mallocator(const Mallocator<U>&) {}  // NOLINT(runtime/explicit)
>+
>+  T* allocate(size_t n) { return static_cast<T*>(std::malloc(n * sizeof(T))); }
>+  void deallocate(T* p, size_t) { std::free(p); }
>+};
>+template <typename T, typename U>
>+bool operator==(const Mallocator<T>&, const Mallocator<U>&) {
>+  return true;
>+}
>+template <typename T, typename U>
>+bool operator!=(const Mallocator<T>&, const Mallocator<U>&) {
>+  return false;
>+}
>+
>+TEST(StrCat, CustomAllocator) {
>+  using mstring =
>+      std::basic_string<char, std::char_traits<char>, Mallocator<char>>;
>+  const mstring str1("PARACHUTE OFF A BLIMP INTO MOSCONE!!");
>+
>+  const mstring str2("Read this book about coffee tables");
>+
>+  std::string result = absl::StrCat(str1, str2);
>+  EXPECT_EQ(result,
>+            "PARACHUTE OFF A BLIMP INTO MOSCONE!!"
>+            "Read this book about coffee tables");
>+}
>+
>+TEST(StrCat, MaxArgs) {
>+  std::string result;
>+  // Test 10 up to 26 arguments, the old maximum
>+  result = absl::StrCat(1, 2, 3, 4, 5, 6, 7, 8, 9, "a");
>+  EXPECT_EQ(result, "123456789a");
>+  result = absl::StrCat(1, 2, 3, 4, 5, 6, 7, 8, 9, "a", "b");
>+  EXPECT_EQ(result, "123456789ab");
>+  result = absl::StrCat(1, 2, 3, 4, 5, 6, 7, 8, 9, "a", "b", "c");
>+  EXPECT_EQ(result, "123456789abc");
>+  result = absl::StrCat(1, 2, 3, 4, 5, 6, 7, 8, 9, "a", "b", "c", "d");
>+  EXPECT_EQ(result, "123456789abcd");
>+  result = absl::StrCat(1, 2, 3, 4, 5, 6, 7, 8, 9, "a", "b", "c", "d", "e");
>+  EXPECT_EQ(result, "123456789abcde");
>+  result =
>+      absl::StrCat(1, 2, 3, 4, 5, 6, 7, 8, 9, "a", "b", "c", "d", "e", "f");
>+  EXPECT_EQ(result, "123456789abcdef");
>+  result = absl::StrCat(1, 2, 3, 4, 5, 6, 7, 8, 9, "a", "b", "c", "d", "e", "f",
>+                        "g");
>+  EXPECT_EQ(result, "123456789abcdefg");
>+  result = absl::StrCat(1, 2, 3, 4, 5, 6, 7, 8, 9, "a", "b", "c", "d", "e", "f",
>+                        "g", "h");
>+  EXPECT_EQ(result, "123456789abcdefgh");
>+  result = absl::StrCat(1, 2, 3, 4, 5, 6, 7, 8, 9, "a", "b", "c", "d", "e", "f",
>+                        "g", "h", "i");
>+  EXPECT_EQ(result, "123456789abcdefghi");
>+  result = absl::StrCat(1, 2, 3, 4, 5, 6, 7, 8, 9, "a", "b", "c", "d", "e", "f",
>+                        "g", "h", "i", "j");
>+  EXPECT_EQ(result, "123456789abcdefghij");
>+  result = absl::StrCat(1, 2, 3, 4, 5, 6, 7, 8, 9, "a", "b", "c", "d", "e", "f",
>+                        "g", "h", "i", "j", "k");
>+  EXPECT_EQ(result, "123456789abcdefghijk");
>+  result = absl::StrCat(1, 2, 3, 4, 5, 6, 7, 8, 9, "a", "b", "c", "d", "e", "f",
>+                        "g", "h", "i", "j", "k", "l");
>+  EXPECT_EQ(result, "123456789abcdefghijkl");
>+  result = absl::StrCat(1, 2, 3, 4, 5, 6, 7, 8, 9, "a", "b", "c", "d", "e", "f",
>+                        "g", "h", "i", "j", "k", "l", "m");
>+  EXPECT_EQ(result, "123456789abcdefghijklm");
>+  result = absl::StrCat(1, 2, 3, 4, 5, 6, 7, 8, 9, "a", "b", "c", "d", "e", "f",
>+                        "g", "h", "i", "j", "k", "l", "m", "n");
>+  EXPECT_EQ(result, "123456789abcdefghijklmn");
>+  result = absl::StrCat(1, 2, 3, 4, 5, 6, 7, 8, 9, "a", "b", "c", "d", "e", "f",
>+                        "g", "h", "i", "j", "k", "l", "m", "n", "o");
>+  EXPECT_EQ(result, "123456789abcdefghijklmno");
>+  result = absl::StrCat(1, 2, 3, 4, 5, 6, 7, 8, 9, "a", "b", "c", "d", "e", "f",
>+                        "g", "h", "i", "j", "k", "l", "m", "n", "o", "p");
>+  EXPECT_EQ(result, "123456789abcdefghijklmnop");
>+  result = absl::StrCat(1, 2, 3, 4, 5, 6, 7, 8, 9, "a", "b", "c", "d", "e", "f",
>+                        "g", "h", "i", "j", "k", "l", "m", "n", "o", "p", "q");
>+  EXPECT_EQ(result, "123456789abcdefghijklmnopq");
>+  // No limit thanks to C++11's variadic templates
>+  result = absl::StrCat(
>+      1, 2, 3, 4, 5, 6, 7, 8, 9, 10, "a", "b", "c", "d", "e", "f", "g", "h",
>+      "i", "j", "k", "l", "m", "n", "o", "p", "q", "r", "s", "t", "u", "v", "w",
>+      "x", "y", "z", "A", "B", "C", "D", "E", "F", "G", "H", "I", "J", "K", "L",
>+      "M", "N", "O", "P", "Q", "R", "S", "T", "U", "V", "W", "X", "Y", "Z");
>+  EXPECT_EQ(result,
>+            "12345678910abcdefghijklmnopqrstuvwxyzABCDEFGHIJKLMNOPQRSTUVWXYZ");
>+}
>+
>+TEST(StrAppend, Basics) {
>+  std::string result = "existing text";
>+
>+  std::string strs[] = {
>+    "Hello",
>+    "Cruel",
>+    "World"
>+  };
>+
>+  std::string stdstrs[] = {
>+    "std::Hello",
>+    "std::Cruel",
>+    "std::World"
>+  };
>+
>+  absl::string_view pieces[] = {"Hello", "Cruel", "World"};
>+
>+  const char* c_strs[] = {
>+    "Hello",
>+    "Cruel",
>+    "World"
>+  };
>+
>+  int32_t i32s[] = {'H', 'C', 'W'};
>+  uint64_t ui64s[] = {12345678910LL, 10987654321LL};
>+
>+  std::string::size_type old_size = result.size();
>+  absl::StrAppend(&result);
>+  EXPECT_EQ(result.size(), old_size);
>+
>+  old_size = result.size();
>+  absl::StrAppend(&result, strs[0]);
>+  EXPECT_EQ(result.substr(old_size), "Hello");
>+
>+  old_size = result.size();
>+  absl::StrAppend(&result, strs[1], pieces[2]);
>+  EXPECT_EQ(result.substr(old_size), "CruelWorld");
>+
>+  old_size = result.size();
>+  absl::StrAppend(&result, stdstrs[0], ", ", pieces[2]);
>+  EXPECT_EQ(result.substr(old_size), "std::Hello, World");
>+
>+  old_size = result.size();
>+  absl::StrAppend(&result, strs[0], ", ", stdstrs[1], " ", strs[2], "!");
>+  EXPECT_EQ(result.substr(old_size), "Hello, std::Cruel World!");
>+
>+  old_size = result.size();
>+  absl::StrAppend(&result, pieces[0], ", ", pieces[1], " ", pieces[2]);
>+  EXPECT_EQ(result.substr(old_size), "Hello, Cruel World");
>+
>+  old_size = result.size();
>+  absl::StrAppend(&result, c_strs[0], ", ", c_strs[1], " ", c_strs[2]);
>+  EXPECT_EQ(result.substr(old_size), "Hello, Cruel World");
>+
>+  old_size = result.size();
>+  absl::StrAppend(&result, "ASCII ", i32s[0], ", ", i32s[1], " ", i32s[2], "!");
>+  EXPECT_EQ(result.substr(old_size), "ASCII 72, 67 87!");
>+
>+  old_size = result.size();
>+  absl::StrAppend(&result, ui64s[0], ", ", ui64s[1], "!");
>+  EXPECT_EQ(result.substr(old_size), "12345678910, 10987654321!");
>+
>+  std::string one = "1";  // Actually, it's the size of this std::string that we want; a
>+                     // 64-bit build distinguishes between size_t and uint64_t,
>+                     // even though they're both unsigned 64-bit values.
>+  old_size = result.size();
>+  absl::StrAppend(&result, "And a ", one.size(), " and a ",
>+                  &result[2] - &result[0], " and a ", one, " 2 3 4", "!");
>+  EXPECT_EQ(result.substr(old_size), "And a 1 and a 2 and a 1 2 3 4!");
>+
>+  // result = absl::StrCat("Single chars won't compile", '!');
>+  // result = absl::StrCat("Neither will nullptrs", nullptr);
>+  old_size = result.size();
>+  absl::StrAppend(&result,
>+                  "To output a char by ASCII/numeric value, use +: ", '!' + 0);
>+  EXPECT_EQ(result.substr(old_size),
>+            "To output a char by ASCII/numeric value, use +: 33");
>+
>+  // Test 9 arguments, the old maximum
>+  old_size = result.size();
>+  absl::StrAppend(&result, 1, 22, 333, 4444, 55555, 666666, 7777777, 88888888,
>+                  9);
>+  EXPECT_EQ(result.substr(old_size), "1223334444555556666667777777888888889");
>+
>+  // No limit thanks to C++11's variadic templates
>+  old_size = result.size();
>+  absl::StrAppend(
>+      &result, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10,                           //
>+      "a", "b", "c", "d", "e", "f", "g", "h", "i", "j", "k", "l", "m",  //
>+      "n", "o", "p", "q", "r", "s", "t", "u", "v", "w", "x", "y", "z",  //
>+      "A", "B", "C", "D", "E", "F", "G", "H", "I", "J", "K", "L", "M",  //
>+      "N", "O", "P", "Q", "R", "S", "T", "U", "V", "W", "X", "Y", "Z",  //
>+      "No limit thanks to C++11's variadic templates");
>+  EXPECT_EQ(result.substr(old_size),
>+            "12345678910abcdefghijklmnopqrstuvwxyzABCDEFGHIJKLMNOPQRSTUVWXYZ"
>+            "No limit thanks to C++11's variadic templates");
>+}
>+
>+#ifdef GTEST_HAS_DEATH_TEST
>+TEST(StrAppend, Death) {
>+  std::string s = "self";
>+  // on linux it's "assertion", on mac it's "Assertion",
>+  // on chromiumos it's "Assertion ... failed".
>+  ABSL_EXPECT_DEBUG_DEATH(absl::StrAppend(&s, s.c_str() + 1),
>+                          "ssertion.*failed");
>+  ABSL_EXPECT_DEBUG_DEATH(absl::StrAppend(&s, s), "ssertion.*failed");
>+}
>+#endif  // GTEST_HAS_DEATH_TEST
>+
>+TEST(StrAppend, EmptyString) {
>+  std::string s = "";
>+  absl::StrAppend(&s, s);
>+  EXPECT_EQ(s, "");
>+}
>+
>+template <typename IntType>
>+void CheckHex(IntType v, const char* nopad_format, const char* zeropad_format,
>+              const char* spacepad_format) {
>+  char expected[256];
>+
>+  std::string actual = absl::StrCat(absl::Hex(v, absl::kNoPad));
>+  snprintf(expected, sizeof(expected), nopad_format, v);
>+  EXPECT_EQ(expected, actual) << " decimal value " << v;
>+
>+  for (int spec = absl::kZeroPad2; spec <= absl::kZeroPad16; ++spec) {
>+    std::string actual =
>+        absl::StrCat(absl::Hex(v, static_cast<absl::PadSpec>(spec)));
>+    snprintf(expected, sizeof(expected), zeropad_format,
>+             spec - absl::kZeroPad2 + 2, v);
>+    EXPECT_EQ(expected, actual) << " decimal value " << v;
>+  }
>+
>+  for (int spec = absl::kSpacePad2; spec <= absl::kSpacePad16; ++spec) {
>+    std::string actual =
>+        absl::StrCat(absl::Hex(v, static_cast<absl::PadSpec>(spec)));
>+    snprintf(expected, sizeof(expected), spacepad_format,
>+             spec - absl::kSpacePad2 + 2, v);
>+    EXPECT_EQ(expected, actual) << " decimal value " << v;
>+  }
>+}
>+
>+template <typename IntType>
>+void CheckDec(IntType v, const char* nopad_format, const char* zeropad_format,
>+              const char* spacepad_format) {
>+  char expected[256];
>+
>+  std::string actual = absl::StrCat(absl::Dec(v, absl::kNoPad));
>+  snprintf(expected, sizeof(expected), nopad_format, v);
>+  EXPECT_EQ(expected, actual) << " decimal value " << v;
>+
>+  for (int spec = absl::kZeroPad2; spec <= absl::kZeroPad16; ++spec) {
>+    std::string actual =
>+        absl::StrCat(absl::Dec(v, static_cast<absl::PadSpec>(spec)));
>+    snprintf(expected, sizeof(expected), zeropad_format,
>+             spec - absl::kZeroPad2 + 2, v);
>+    EXPECT_EQ(expected, actual)
>+        << " decimal value " << v << " format '" << zeropad_format
>+        << "' digits " << (spec - absl::kZeroPad2 + 2);
>+  }
>+
>+  for (int spec = absl::kSpacePad2; spec <= absl::kSpacePad16; ++spec) {
>+    std::string actual =
>+        absl::StrCat(absl::Dec(v, static_cast<absl::PadSpec>(spec)));
>+    snprintf(expected, sizeof(expected), spacepad_format,
>+             spec - absl::kSpacePad2 + 2, v);
>+    EXPECT_EQ(expected, actual)
>+        << " decimal value " << v << " format '" << spacepad_format
>+        << "' digits " << (spec - absl::kSpacePad2 + 2);
>+  }
>+}
>+
>+void CheckHexDec64(uint64_t v) {
>+  unsigned long long ullv = v;  // NOLINT(runtime/int)
>+
>+  CheckHex(ullv, "%llx", "%0*llx", "%*llx");
>+  CheckDec(ullv, "%llu", "%0*llu", "%*llu");
>+
>+  long long llv = static_cast<long long>(ullv);  // NOLINT(runtime/int)
>+  CheckDec(llv, "%lld", "%0*lld", "%*lld");
>+
>+  if (sizeof(v) == sizeof(&v)) {
>+    auto uintptr = static_cast<uintptr_t>(v);
>+    void* ptr = reinterpret_cast<void*>(uintptr);
>+    CheckHex(ptr, "%llx", "%0*llx", "%*llx");
>+  }
>+}
>+
>+void CheckHexDec32(uint32_t uv) {
>+  CheckHex(uv, "%x", "%0*x", "%*x");
>+  CheckDec(uv, "%u", "%0*u", "%*u");
>+  int32_t v = static_cast<int32_t>(uv);
>+  CheckDec(v, "%d", "%0*d", "%*d");
>+
>+  if (sizeof(v) == sizeof(&v)) {
>+    auto uintptr = static_cast<uintptr_t>(v);
>+    void* ptr = reinterpret_cast<void*>(uintptr);
>+    CheckHex(ptr, "%x", "%0*x", "%*x");
>+  }
>+}
>+
>+void CheckAll(uint64_t v) {
>+  CheckHexDec64(v);
>+  CheckHexDec32(static_cast<uint32_t>(v));
>+}
>+
>+void TestFastPrints() {
>+  // Test all small ints; there aren't many and they're common.
>+  for (int i = 0; i < 10000; i++) {
>+    CheckAll(i);
>+  }
>+
>+  CheckAll(std::numeric_limits<uint64_t>::max());
>+  CheckAll(std::numeric_limits<uint64_t>::max() - 1);
>+  CheckAll(std::numeric_limits<int64_t>::min());
>+  CheckAll(std::numeric_limits<int64_t>::min() + 1);
>+  CheckAll(std::numeric_limits<uint32_t>::max());
>+  CheckAll(std::numeric_limits<uint32_t>::max() - 1);
>+  CheckAll(std::numeric_limits<int32_t>::min());
>+  CheckAll(std::numeric_limits<int32_t>::min() + 1);
>+  CheckAll(999999999);              // fits in 32 bits
>+  CheckAll(1000000000);             // fits in 32 bits
>+  CheckAll(9999999999);             // doesn't fit in 32 bits
>+  CheckAll(10000000000);            // doesn't fit in 32 bits
>+  CheckAll(999999999999999999);     // fits in signed 64-bit
>+  CheckAll(9999999999999999999u);   // fits in unsigned 64-bit, but not signed.
>+  CheckAll(1000000000000000000);    // fits in signed 64-bit
>+  CheckAll(10000000000000000000u);  // fits in unsigned 64-bit, but not signed.
>+
>+  CheckAll(999999999876543210);    // check all decimal digits, signed
>+  CheckAll(9999999999876543210u);  // check all decimal digits, unsigned.
>+  CheckAll(0x123456789abcdef0);    // check all hex digits
>+  CheckAll(0x12345678);
>+
>+  int8_t minus_one_8bit = -1;
>+  EXPECT_EQ("ff", absl::StrCat(absl::Hex(minus_one_8bit)));
>+
>+  int16_t minus_one_16bit = -1;
>+  EXPECT_EQ("ffff", absl::StrCat(absl::Hex(minus_one_16bit)));
>+}
>+
>+TEST(Numbers, TestFunctionsMovedOverFromNumbersMain) {
>+  TestFastPrints();
>+}
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_format.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_format.h
>new file mode 100644
>index 00000000000..70a811b75e2
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_format.h
>@@ -0,0 +1,512 @@
>+//
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// -----------------------------------------------------------------------------
>+// File: str_format.h
>+// -----------------------------------------------------------------------------
>+//
>+// The `str_format` library is a typesafe replacement for the family of
>+// `printf()` std::string formatting routines within the `<cstdio>` standard library
>+// header. Like the `printf` family, the `str_format` uses a "format string" to
>+// perform argument substitutions based on types.
>+//
>+// Example:
>+//
>+//   std::string s = absl::StrFormat("%s %s You have $%d!", "Hello", name, dollars);
>+//
>+// The library consists of the following basic utilities:
>+//
>+//   * `absl::StrFormat()`, a type-safe replacement for `std::sprintf()`, to
>+//     write a format std::string to a `string` value.
>+//   * `absl::StrAppendFormat()` to append a format std::string to a `string`
>+//   * `absl::StreamFormat()` to more efficiently write a format std::string to a
>+//     stream, such as`std::cout`.
>+//   * `absl::PrintF()`, `absl::FPrintF()` and `absl::SNPrintF()` as
>+//     replacements for `std::printf()`, `std::fprintf()` and `std::snprintf()`.
>+//
>+//     Note: a version of `std::sprintf()` is not supported as it is
>+//     generally unsafe due to buffer overflows.
>+//
>+// Additionally, you can provide a format std::string (and its associated arguments)
>+// using one of the following abstractions:
>+//
>+//   * A `FormatSpec` class template fully encapsulates a format std::string and its
>+//     type arguments and is usually provided to `str_format` functions as a
>+//     variadic argument of type `FormatSpec<Arg...>`. The `FormatSpec<Args...>`
>+//     template is evaluated at compile-time, providing type safety.
>+//   * A `ParsedFormat` instance, which encapsulates a specific, pre-compiled
>+//     format std::string for a specific set of type(s), and which can be passed
>+//     between API boundaries. (The `FormatSpec` type should not be used
>+//     directly.)
>+//
>+// The `str_format` library provides the ability to output its format strings to
>+// arbitrary sink types:
>+//
>+//   * A generic `Format()` function to write outputs to arbitrary sink types,
>+//     which must implement a `RawSinkFormat` interface. (See
>+//     `str_format_sink.h` for more information.)
>+//
>+//   * A `FormatUntyped()` function that is similar to `Format()` except it is
>+//     loosely typed. `FormatUntyped()` is not a template and does not perform
>+//     any compile-time checking of the format std::string; instead, it returns a
>+//     boolean from a runtime check.
>+//
>+// In addition, the `str_format` library provides extension points for
>+// augmenting formatting to new types. These extensions are fully documented
>+// within the `str_format_extension.h` header file.
>+#ifndef ABSL_STRINGS_STR_FORMAT_H_
>+#define ABSL_STRINGS_STR_FORMAT_H_
>+
>+#include <cstdio>
>+#include <string>
>+
>+#include "absl/strings/internal/str_format/arg.h"  // IWYU pragma: export
>+#include "absl/strings/internal/str_format/bind.h"  // IWYU pragma: export
>+#include "absl/strings/internal/str_format/checker.h"  // IWYU pragma: export
>+#include "absl/strings/internal/str_format/extension.h"  // IWYU pragma: export
>+#include "absl/strings/internal/str_format/parser.h"  // IWYU pragma: export
>+
>+namespace absl {
>+
>+// UntypedFormatSpec
>+//
>+// A type-erased class that can be used directly within untyped API entry
>+// points. An `UntypedFormatSpec` is specifically used as an argument to
>+// `FormatUntyped()`.
>+//
>+// Example:
>+//
>+//   absl::UntypedFormatSpec format("%d");
>+//   std::string out;
>+//   CHECK(absl::FormatUntyped(&out, format, {absl::FormatArg(1)}));
>+class UntypedFormatSpec {
>+ public:
>+  UntypedFormatSpec() = delete;
>+  UntypedFormatSpec(const UntypedFormatSpec&) = delete;
>+  UntypedFormatSpec& operator=(const UntypedFormatSpec&) = delete;
>+
>+  explicit UntypedFormatSpec(string_view s) : spec_(s) {}
>+
>+ protected:
>+  explicit UntypedFormatSpec(const str_format_internal::ParsedFormatBase* pc)
>+      : spec_(pc) {}
>+
>+ private:
>+  friend str_format_internal::UntypedFormatSpecImpl;
>+  str_format_internal::UntypedFormatSpecImpl spec_;
>+};
>+
>+// FormatStreamed()
>+//
>+// Takes a streamable argument and returns an object that can print it
>+// with '%s'. Allows printing of types that have an `operator<<` but no
>+// intrinsic type support within `StrFormat()` itself.
>+//
>+// Example:
>+//
>+//   absl::StrFormat("%s", absl::FormatStreamed(obj));
>+template <typename T>
>+str_format_internal::StreamedWrapper<T> FormatStreamed(const T& v) {
>+  return str_format_internal::StreamedWrapper<T>(v);
>+}
>+
>+// FormatCountCapture
>+//
>+// This class provides a way to safely wrap `StrFormat()` captures of `%n`
>+// conversions, which denote the number of characters written by a formatting
>+// operation to this point, into an integer value.
>+//
>+// This wrapper is designed to allow safe usage of `%n` within `StrFormat(); in
>+// the `printf()` family of functions, `%n` is not safe to use, as the `int *`
>+// buffer can be used to capture arbitrary data.
>+//
>+// Example:
>+//
>+//   int n = 0;
>+//   std::string s = absl::StrFormat("%s%d%n", "hello", 123,
>+//                   absl::FormatCountCapture(&n));
>+//   EXPECT_EQ(8, n);
>+class FormatCountCapture {
>+ public:
>+  explicit FormatCountCapture(int* p) : p_(p) {}
>+
>+ private:
>+  // FormatCountCaptureHelper is used to define FormatConvertImpl() for this
>+  // class.
>+  friend struct str_format_internal::FormatCountCaptureHelper;
>+  // Unused() is here because of the false positive from -Wunused-private-field
>+  // p_ is used in the templated function of the friend FormatCountCaptureHelper
>+  // class.
>+  int* Unused() { return p_; }
>+  int* p_;
>+};
>+
>+// FormatSpec
>+//
>+// The `FormatSpec` type defines the makeup of a format std::string within the
>+// `str_format` library. You should not need to use or manipulate this type
>+// directly. A `FormatSpec` is a variadic class template that is evaluated at
>+// compile-time, according to the format std::string and arguments that are passed
>+// to it.
>+//
>+// For a `FormatSpec` to be valid at compile-time, it must be provided as
>+// either:
>+//
>+// * A `constexpr` literal or `absl::string_view`, which is how it most often
>+//   used.
>+// * A `ParsedFormat` instantiation, which ensures the format std::string is
>+//   valid before use. (See below.)
>+//
>+// Example:
>+//
>+//   // Provided as a std::string literal.
>+//   absl::StrFormat("Welcome to %s, Number %d!", "The Village", 6);
>+//
>+//   // Provided as a constexpr absl::string_view.
>+//   constexpr absl::string_view formatString = "Welcome to %s, Number %d!";
>+//   absl::StrFormat(formatString, "The Village", 6);
>+//
>+//   // Provided as a pre-compiled ParsedFormat object.
>+//   // Note that this example is useful only for illustration purposes.
>+//   absl::ParsedFormat<'s', 'd'> formatString("Welcome to %s, Number %d!");
>+//   absl::StrFormat(formatString, "TheVillage", 6);
>+//
>+// A format std::string generally follows the POSIX syntax as used within the POSIX
>+// `printf` specification.
>+//
>+// (See http://pubs.opengroup.org/onlinepubs/9699919799/utilities/printf.html.)
>+//
>+// In specific, the `FormatSpec` supports the following type specifiers:
>+//   * `c` for characters
>+//   * `s` for strings
>+//   * `d` or `i` for integers
>+//   * `o` for unsigned integer conversions into octal
>+//   * `x` or `X` for unsigned integer conversions into hex
>+//   * `u` for unsigned integers
>+//   * `f` or `F` for floating point values into decimal notation
>+//   * `e` or `E` for floating point values into exponential notation
>+//   * `a` or `A` for floating point values into hex exponential notation
>+//   * `g` or `G` for floating point values into decimal or exponential
>+//     notation based on their precision
>+//   * `p` for pointer address values
>+//   * `n` for the special case of writing out the number of characters
>+//     written to this point. The resulting value must be captured within an
>+//     `absl::FormatCountCapture` type.
>+//
>+// NOTE: `o`, `x\X` and `u` will convert signed values to their unsigned
>+// counterpart before formatting.
>+//
>+// Examples:
>+//     "%c", 'a'                -> "a"
>+//     "%c", 32                 -> " "
>+//     "%s", "C"                -> "C"
>+//     "%s", std::string("C++") -> "C++"
>+//     "%d", -10                -> "-10"
>+//     "%o", 10                 -> "12"
>+//     "%x", 16                 -> "10"
>+//     "%f", 123456789          -> "123456789.000000"
>+//     "%e", .01                -> "1.00000e-2"
>+//     "%a", -3.0               -> "-0x1.8p+1"
>+//     "%g", .01                -> "1e-2"
>+//     "%p", *int               -> "0x7ffdeb6ad2a4"
>+//
>+//     int n = 0;
>+//     std::string s = absl::StrFormat(
>+//         "%s%d%n", "hello", 123, absl::FormatCountCapture(&n));
>+//     EXPECT_EQ(8, n);
>+//
>+// The `FormatSpec` intrinsically supports all of these fundamental C++ types:
>+//
>+// *   Characters: `char`, `signed char`, `unsigned char`
>+// *   Integers: `int`, `short`, `unsigned short`, `unsigned`, `long`,
>+//         `unsigned long`, `long long`, `unsigned long long`
>+// *   Floating-point: `float`, `double`, `long double`
>+//
>+// However, in the `str_format` library, a format conversion specifies a broader
>+// C++ conceptual category instead of an exact type. For example, `%s` binds to
>+// any std::string-like argument, so `std::string`, `absl::string_view`, and
>+// `const char*` are all accepted. Likewise, `%d` accepts any integer-like
>+// argument, etc.
>+
>+template <typename... Args>
>+using FormatSpec =
>+    typename str_format_internal::FormatSpecDeductionBarrier<Args...>::type;
>+
>+// ParsedFormat
>+//
>+// A `ParsedFormat` is a class template representing a preparsed `FormatSpec`,
>+// with template arguments specifying the conversion characters used within the
>+// format std::string. Such characters must be valid format type specifiers, and
>+// these type specifiers are checked at compile-time.
>+//
>+// Instances of `ParsedFormat` can be created, copied, and reused to speed up
>+// formatting loops. A `ParsedFormat` may either be constructed statically, or
>+// dynamically through its `New()` factory function, which only constructs a
>+// runtime object if the format is valid at that time.
>+//
>+// Example:
>+//
>+//   // Verified at compile time.
>+//   absl::ParsedFormat<'s', 'd'> formatString("Welcome to %s, Number %d!");
>+//   absl::StrFormat(formatString, "TheVillage", 6);
>+//
>+//   // Verified at runtime.
>+//   auto format_runtime = absl::ParsedFormat<'d'>::New(format_string);
>+//   if (format_runtime) {
>+//     value = absl::StrFormat(*format_runtime, i);
>+//   } else {
>+//     ... error case ...
>+//   }
>+template <char... Conv>
>+using ParsedFormat = str_format_internal::ExtendedParsedFormat<
>+    str_format_internal::ConversionCharToConv(Conv)...>;
>+
>+// StrFormat()
>+//
>+// Returns a `string` given a `printf()`-style format std::string and zero or more
>+// additional arguments. Use it as you would `sprintf()`. `StrFormat()` is the
>+// primary formatting function within the `str_format` library, and should be
>+// used in most cases where you need type-safe conversion of types into
>+// formatted strings.
>+//
>+// The format std::string generally consists of ordinary character data along with
>+// one or more format conversion specifiers (denoted by the `%` character).
>+// Ordinary character data is returned unchanged into the result std::string, while
>+// each conversion specification performs a type substitution from
>+// `StrFormat()`'s other arguments. See the comments for `FormatSpec` for full
>+// information on the makeup of this format std::string.
>+//
>+// Example:
>+//
>+//   std::string s = absl::StrFormat(
>+//       "Welcome to %s, Number %d!", "The Village", 6);
>+//   EXPECT_EQ("Welcome to The Village, Number 6!", s);
>+//
>+// Returns an empty std::string in case of error.
>+template <typename... Args>
>+ABSL_MUST_USE_RESULT std::string StrFormat(const FormatSpec<Args...>& format,
>+                                      const Args&... args) {
>+  return str_format_internal::FormatPack(
>+      str_format_internal::UntypedFormatSpecImpl::Extract(format),
>+      {str_format_internal::FormatArgImpl(args)...});
>+}
>+
>+// StrAppendFormat()
>+//
>+// Appends to a `dst` std::string given a format std::string, and zero or more additional
>+// arguments, returning `*dst` as a convenience for chaining purposes. Appends
>+// nothing in case of error (but possibly alters its capacity).
>+//
>+// Example:
>+//
>+//   std::string orig("For example PI is approximately ");
>+//   std::cout << StrAppendFormat(&orig, "%12.6f", 3.14);
>+template <typename... Args>
>+std::string& StrAppendFormat(std::string* dst, const FormatSpec<Args...>& format,
>+                        const Args&... args) {
>+  return str_format_internal::AppendPack(
>+      dst, str_format_internal::UntypedFormatSpecImpl::Extract(format),
>+      {str_format_internal::FormatArgImpl(args)...});
>+}
>+
>+// StreamFormat()
>+//
>+// Writes to an output stream given a format std::string and zero or more arguments,
>+// generally in a manner that is more efficient than streaming the result of
>+// `absl:: StrFormat()`. The returned object must be streamed before the full
>+// expression ends.
>+//
>+// Example:
>+//
>+//   std::cout << StreamFormat("%12.6f", 3.14);
>+template <typename... Args>
>+ABSL_MUST_USE_RESULT str_format_internal::Streamable StreamFormat(
>+    const FormatSpec<Args...>& format, const Args&... args) {
>+  return str_format_internal::Streamable(
>+      str_format_internal::UntypedFormatSpecImpl::Extract(format),
>+      {str_format_internal::FormatArgImpl(args)...});
>+}
>+
>+// PrintF()
>+//
>+// Writes to stdout given a format std::string and zero or more arguments. This
>+// function is functionally equivalent to `std::printf()` (and type-safe);
>+// prefer `absl::PrintF()` over `std::printf()`.
>+//
>+// Example:
>+//
>+//   std::string_view s = "Ulaanbaatar";
>+//   absl::PrintF("The capital of Mongolia is %s", s);
>+//
>+//   Outputs: "The capital of Mongolia is Ulaanbaatar"
>+//
>+template <typename... Args>
>+int PrintF(const FormatSpec<Args...>& format, const Args&... args) {
>+  return str_format_internal::FprintF(
>+      stdout, str_format_internal::UntypedFormatSpecImpl::Extract(format),
>+      {str_format_internal::FormatArgImpl(args)...});
>+}
>+
>+// FPrintF()
>+//
>+// Writes to a file given a format std::string and zero or more arguments. This
>+// function is functionally equivalent to `std::fprintf()` (and type-safe);
>+// prefer `absl::FPrintF()` over `std::fprintf()`.
>+//
>+// Example:
>+//
>+//   std::string_view s = "Ulaanbaatar";
>+//   absl::FPrintF("The capital of Mongolia is %s", s);
>+//
>+//   Outputs: "The capital of Mongolia is Ulaanbaatar"
>+//
>+template <typename... Args>
>+int FPrintF(std::FILE* output, const FormatSpec<Args...>& format,
>+            const Args&... args) {
>+  return str_format_internal::FprintF(
>+      output, str_format_internal::UntypedFormatSpecImpl::Extract(format),
>+      {str_format_internal::FormatArgImpl(args)...});
>+}
>+
>+// SNPrintF()
>+//
>+// Writes to a sized buffer given a format std::string and zero or more arguments.
>+// This function is functionally equivalent to `std::snprintf()` (and
>+// type-safe); prefer `absl::SNPrintF()` over `std::snprintf()`.
>+//
>+// Example:
>+//
>+//   std::string_view s = "Ulaanbaatar";
>+//   char output[128];
>+//   absl::SNPrintF(output, sizeof(output),
>+//                  "The capital of Mongolia is %s", s);
>+//
>+//   Post-condition: output == "The capital of Mongolia is Ulaanbaatar"
>+//
>+template <typename... Args>
>+int SNPrintF(char* output, std::size_t size, const FormatSpec<Args...>& format,
>+             const Args&... args) {
>+  return str_format_internal::SnprintF(
>+      output, size, str_format_internal::UntypedFormatSpecImpl::Extract(format),
>+      {str_format_internal::FormatArgImpl(args)...});
>+}
>+
>+// -----------------------------------------------------------------------------
>+// Custom Output Formatting Functions
>+// -----------------------------------------------------------------------------
>+
>+// FormatRawSink
>+//
>+// FormatRawSink is a type erased wrapper around arbitrary sink objects
>+// specifically used as an argument to `Format()`.
>+// FormatRawSink does not own the passed sink object. The passed object must
>+// outlive the FormatRawSink.
>+class FormatRawSink {
>+ public:
>+  // Implicitly convert from any type that provides the hook function as
>+  // described above.
>+  template <typename T,
>+            typename = typename std::enable_if<std::is_constructible<
>+                str_format_internal::FormatRawSinkImpl, T*>::value>::type>
>+  FormatRawSink(T* raw)  // NOLINT
>+      : sink_(raw) {}
>+
>+ private:
>+  friend str_format_internal::FormatRawSinkImpl;
>+  str_format_internal::FormatRawSinkImpl sink_;
>+};
>+
>+// Format()
>+//
>+// Writes a formatted std::string to an arbitrary sink object (implementing the
>+// `absl::FormatRawSink` interface), using a format std::string and zero or more
>+// additional arguments.
>+//
>+// By default, `string` and `std::ostream` are supported as destination objects.
>+//
>+// `absl::Format()` is a generic version of `absl::StrFormat(), for custom
>+// sinks. The format std::string, like format strings for `StrFormat()`, is checked
>+// at compile-time.
>+//
>+// On failure, this function returns `false` and the state of the sink is
>+// unspecified.
>+template <typename... Args>
>+bool Format(FormatRawSink raw_sink, const FormatSpec<Args...>& format,
>+            const Args&... args) {
>+  return str_format_internal::FormatUntyped(
>+      str_format_internal::FormatRawSinkImpl::Extract(raw_sink),
>+      str_format_internal::UntypedFormatSpecImpl::Extract(format),
>+      {str_format_internal::FormatArgImpl(args)...});
>+}
>+
>+// FormatArg
>+//
>+// A type-erased handle to a format argument specifically used as an argument to
>+// `FormatUntyped()`. You may construct `FormatArg` by passing
>+// reference-to-const of any printable type. `FormatArg` is both copyable and
>+// assignable. The source data must outlive the `FormatArg` instance. See
>+// example below.
>+//
>+using FormatArg = str_format_internal::FormatArgImpl;
>+
>+// FormatUntyped()
>+//
>+// Writes a formatted std::string to an arbitrary sink object (implementing the
>+// `absl::FormatRawSink` interface), using an `UntypedFormatSpec` and zero or
>+// more additional arguments.
>+//
>+// This function acts as the most generic formatting function in the
>+// `str_format` library. The caller provides a raw sink, an unchecked format
>+// std::string, and (usually) a runtime specified list of arguments; no compile-time
>+// checking of formatting is performed within this function. As a result, a
>+// caller should check the return value to verify that no error occurred.
>+// On failure, this function returns `false` and the state of the sink is
>+// unspecified.
>+//
>+// The arguments are provided in an `absl::Span<const absl::FormatArg>`.
>+// Each `absl::FormatArg` object binds to a single argument and keeps a
>+// reference to it. The values used to create the `FormatArg` objects must
>+// outlive this function call. (See `str_format_arg.h` for information on
>+// the `FormatArg` class.)_
>+//
>+// Example:
>+//
>+//   std::optional<std::string> FormatDynamic(const std::string& in_format,
>+//                                       const vector<std::string>& in_args) {
>+//     std::string out;
>+//     std::vector<absl::FormatArg> args;
>+//     for (const auto& v : in_args) {
>+//       // It is important that 'v' is a reference to the objects in in_args.
>+//       // The values we pass to FormatArg must outlive the call to
>+//       // FormatUntyped.
>+//       args.emplace_back(v);
>+//     }
>+//     absl::UntypedFormatSpec format(in_format);
>+//     if (!absl::FormatUntyped(&out, format, args)) {
>+//       return std::nullopt;
>+//     }
>+//     return std::move(out);
>+//   }
>+//
>+ABSL_MUST_USE_RESULT inline bool FormatUntyped(
>+    FormatRawSink raw_sink, const UntypedFormatSpec& format,
>+    absl::Span<const FormatArg> args) {
>+  return str_format_internal::FormatUntyped(
>+      str_format_internal::FormatRawSinkImpl::Extract(raw_sink),
>+      str_format_internal::UntypedFormatSpecImpl::Extract(format), args);
>+}
>+
>+}  // namespace absl
>+#endif  // ABSL_STRINGS_STR_FORMAT_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_format_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_format_test.cc
>new file mode 100644
>index 00000000000..fed75fafb5c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_format_test.cc
>@@ -0,0 +1,603 @@
>+
>+#include <cstdarg>
>+#include <cstdint>
>+#include <cstdio>
>+#include <string>
>+
>+#include "gmock/gmock.h"
>+#include "gtest/gtest.h"
>+#include "absl/strings/str_format.h"
>+#include "absl/strings/string_view.h"
>+
>+namespace absl {
>+namespace {
>+using str_format_internal::FormatArgImpl;
>+
>+class FormatEntryPointTest : public ::testing::Test { };
>+
>+TEST_F(FormatEntryPointTest, Format) {
>+  std::string sink;
>+  EXPECT_TRUE(Format(&sink, "A format %d", 123));
>+  EXPECT_EQ("A format 123", sink);
>+  sink.clear();
>+
>+  ParsedFormat<'d'> pc("A format %d");
>+  EXPECT_TRUE(Format(&sink, pc, 123));
>+  EXPECT_EQ("A format 123", sink);
>+}
>+TEST_F(FormatEntryPointTest, UntypedFormat) {
>+  constexpr const char* formats[] = {
>+    "",
>+    "a",
>+    "%80d",
>+#if !defined(_MSC_VER) && !defined(__ANDROID__)
>+    // MSVC and Android don't support positional syntax.
>+    "complicated multipart %% %1$d format %1$0999d",
>+#endif  // _MSC_VER
>+  };
>+  for (const char* fmt : formats) {
>+    std::string actual;
>+    int i = 123;
>+    FormatArgImpl arg_123(i);
>+    absl::Span<const FormatArgImpl> args(&arg_123, 1);
>+    UntypedFormatSpec format(fmt);
>+
>+    EXPECT_TRUE(FormatUntyped(&actual, format, args));
>+    char buf[4096]{};
>+    snprintf(buf, sizeof(buf), fmt, 123);
>+    EXPECT_EQ(
>+        str_format_internal::FormatPack(
>+            str_format_internal::UntypedFormatSpecImpl::Extract(format), args),
>+        buf);
>+    EXPECT_EQ(actual, buf);
>+  }
>+  // The internal version works with a preparsed format.
>+  ParsedFormat<'d'> pc("A format %d");
>+  int i = 345;
>+  FormatArg arg(i);
>+  std::string out;
>+  EXPECT_TRUE(str_format_internal::FormatUntyped(
>+      &out, str_format_internal::UntypedFormatSpecImpl(&pc), {&arg, 1}));
>+  EXPECT_EQ("A format 345", out);
>+}
>+
>+TEST_F(FormatEntryPointTest, StringFormat) {
>+  EXPECT_EQ("123", StrFormat("%d", 123));
>+  constexpr absl::string_view view("=%d=", 4);
>+  EXPECT_EQ("=123=", StrFormat(view, 123));
>+}
>+
>+TEST_F(FormatEntryPointTest, AppendFormat) {
>+  std::string s;
>+  std::string& r = StrAppendFormat(&s, "%d", 123);
>+  EXPECT_EQ(&s, &r);  // should be same object
>+  EXPECT_EQ("123", r);
>+}
>+
>+TEST_F(FormatEntryPointTest, AppendFormatFail) {
>+  std::string s = "orig";
>+
>+  UntypedFormatSpec format(" more %d");
>+  FormatArgImpl arg("not an int");
>+
>+  EXPECT_EQ("orig",
>+            str_format_internal::AppendPack(
>+                &s, str_format_internal::UntypedFormatSpecImpl::Extract(format),
>+                {&arg, 1}));
>+}
>+
>+
>+TEST_F(FormatEntryPointTest, ManyArgs) {
>+  EXPECT_EQ("24", StrFormat("%24$d", 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13,
>+                            14, 15, 16, 17, 18, 19, 20, 21, 22, 23, 24));
>+  EXPECT_EQ("60", StrFormat("%60$d", 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13,
>+                            14, 15, 16, 17, 18, 19, 20, 21, 22, 23, 24, 25, 26,
>+                            27, 28, 29, 30, 31, 32, 33, 34, 35, 36, 37, 38, 39,
>+                            40, 41, 42, 43, 44, 45, 46, 47, 48, 49, 50, 51, 52,
>+                            53, 54, 55, 56, 57, 58, 59, 60));
>+}
>+
>+TEST_F(FormatEntryPointTest, Preparsed) {
>+  ParsedFormat<'d'> pc("%d");
>+  EXPECT_EQ("123", StrFormat(pc, 123));
>+  // rvalue ok?
>+  EXPECT_EQ("123", StrFormat(ParsedFormat<'d'>("%d"), 123));
>+  constexpr absl::string_view view("=%d=", 4);
>+  EXPECT_EQ("=123=", StrFormat(ParsedFormat<'d'>(view), 123));
>+}
>+
>+TEST_F(FormatEntryPointTest, FormatCountCapture) {
>+  int n = 0;
>+  EXPECT_EQ("", StrFormat("%n", FormatCountCapture(&n)));
>+  EXPECT_EQ(0, n);
>+  EXPECT_EQ("123", StrFormat("%d%n", 123, FormatCountCapture(&n)));
>+  EXPECT_EQ(3, n);
>+}
>+
>+TEST_F(FormatEntryPointTest, FormatCountCaptureWrongType) {
>+  // Should reject int*.
>+  int n = 0;
>+  UntypedFormatSpec format("%d%n");
>+  int i = 123, *ip = &n;
>+  FormatArgImpl args[2] = {FormatArgImpl(i), FormatArgImpl(ip)};
>+
>+  EXPECT_EQ("", str_format_internal::FormatPack(
>+                    str_format_internal::UntypedFormatSpecImpl::Extract(format),
>+                    absl::MakeSpan(args)));
>+}
>+
>+TEST_F(FormatEntryPointTest, FormatCountCaptureMultiple) {
>+  int n1 = 0;
>+  int n2 = 0;
>+  EXPECT_EQ("    1         2",
>+            StrFormat("%5d%n%10d%n", 1, FormatCountCapture(&n1), 2,
>+                      FormatCountCapture(&n2)));
>+  EXPECT_EQ(5, n1);
>+  EXPECT_EQ(15, n2);
>+}
>+
>+TEST_F(FormatEntryPointTest, FormatCountCaptureExample) {
>+  int n;
>+  std::string s;
>+  StrAppendFormat(&s, "%s: %n%s\n", "(1,1)", FormatCountCapture(&n), "(1,2)");
>+  StrAppendFormat(&s, "%*s%s\n", n, "", "(2,2)");
>+  EXPECT_EQ(7, n);
>+  EXPECT_EQ(
>+      "(1,1): (1,2)\n"
>+      "       (2,2)\n",
>+      s);
>+}
>+
>+TEST_F(FormatEntryPointTest, Stream) {
>+  const std::string formats[] = {
>+    "",
>+    "a",
>+    "%80d",
>+#if !defined(_MSC_VER) && !defined(__ANDROID__)
>+    // MSVC doesn't support positional syntax.
>+    "complicated multipart %% %1$d format %1$080d",
>+#endif  // _MSC_VER
>+  };
>+  std::string buf(4096, '\0');
>+  for (const auto& fmt : formats) {
>+    const auto parsed = ParsedFormat<'d'>::NewAllowIgnored(fmt);
>+    std::ostringstream oss;
>+    oss << StreamFormat(*parsed, 123);
>+    int fmt_result = snprintf(&*buf.begin(), buf.size(), fmt.c_str(), 123);
>+    ASSERT_TRUE(oss) << fmt;
>+    ASSERT_TRUE(fmt_result >= 0 && static_cast<size_t>(fmt_result) < buf.size())
>+        << fmt_result;
>+    EXPECT_EQ(buf.c_str(), oss.str());
>+  }
>+}
>+
>+TEST_F(FormatEntryPointTest, StreamOk) {
>+  std::ostringstream oss;
>+  oss << StreamFormat("hello %d", 123);
>+  EXPECT_EQ("hello 123", oss.str());
>+  EXPECT_TRUE(oss.good());
>+}
>+
>+TEST_F(FormatEntryPointTest, StreamFail) {
>+  std::ostringstream oss;
>+  UntypedFormatSpec format("hello %d");
>+  FormatArgImpl arg("non-numeric");
>+  oss << str_format_internal::Streamable(
>+      str_format_internal::UntypedFormatSpecImpl::Extract(format), {&arg, 1});
>+  EXPECT_EQ("hello ", oss.str());  // partial write
>+  EXPECT_TRUE(oss.fail());
>+}
>+
>+std::string WithSnprintf(const char* fmt, ...) {
>+  std::string buf;
>+  buf.resize(128);
>+  va_list va;
>+  va_start(va, fmt);
>+  int r = vsnprintf(&*buf.begin(), buf.size(), fmt, va);
>+  va_end(va);
>+  EXPECT_GE(r, 0);
>+  EXPECT_LT(r, buf.size());
>+  buf.resize(r);
>+  return buf;
>+}
>+
>+TEST_F(FormatEntryPointTest, FloatPrecisionArg) {
>+  // Test that positional parameters for width and precision
>+  // are indexed to precede the value.
>+  // Also sanity check the same formats against snprintf.
>+  EXPECT_EQ("0.1", StrFormat("%.1f", 0.1));
>+  EXPECT_EQ("0.1", WithSnprintf("%.1f", 0.1));
>+  EXPECT_EQ("  0.1", StrFormat("%*.1f", 5, 0.1));
>+  EXPECT_EQ("  0.1", WithSnprintf("%*.1f", 5, 0.1));
>+  EXPECT_EQ("0.1", StrFormat("%.*f", 1, 0.1));
>+  EXPECT_EQ("0.1", WithSnprintf("%.*f", 1, 0.1));
>+  EXPECT_EQ("  0.1", StrFormat("%*.*f", 5, 1, 0.1));
>+  EXPECT_EQ("  0.1", WithSnprintf("%*.*f", 5, 1, 0.1));
>+}
>+namespace streamed_test {
>+struct X {};
>+std::ostream& operator<<(std::ostream& os, const X&) {
>+  return os << "X";
>+}
>+}  // streamed_test
>+
>+TEST_F(FormatEntryPointTest, FormatStreamed) {
>+  EXPECT_EQ("123", StrFormat("%s", FormatStreamed(123)));
>+  EXPECT_EQ("  123", StrFormat("%5s", FormatStreamed(123)));
>+  EXPECT_EQ("123  ", StrFormat("%-5s", FormatStreamed(123)));
>+  EXPECT_EQ("X", StrFormat("%s", FormatStreamed(streamed_test::X())));
>+  EXPECT_EQ("123", StrFormat("%s", FormatStreamed(StreamFormat("%d", 123))));
>+}
>+
>+// Helper class that creates a temporary file and exposes a FILE* to it.
>+// It will close the file on destruction.
>+class TempFile {
>+ public:
>+  TempFile() : file_(std::tmpfile()) {}
>+  ~TempFile() { std::fclose(file_); }
>+
>+  std::FILE* file() const { return file_; }
>+
>+  // Read the file into a std::string.
>+  std::string ReadFile() {
>+    std::fseek(file_, 0, SEEK_END);
>+    int size = std::ftell(file_);
>+    std::rewind(file_);
>+    std::string str(2 * size, ' ');
>+    int read_bytes = std::fread(&str[0], 1, str.size(), file_);
>+    EXPECT_EQ(read_bytes, size);
>+    str.resize(read_bytes);
>+    EXPECT_TRUE(std::feof(file_));
>+    return str;
>+  }
>+
>+ private:
>+  std::FILE* file_;
>+};
>+
>+TEST_F(FormatEntryPointTest, FPrintF) {
>+  TempFile tmp;
>+  int result =
>+      FPrintF(tmp.file(), "STRING: %s NUMBER: %010d", std::string("ABC"), -19);
>+  EXPECT_EQ(result, 30);
>+  EXPECT_EQ(tmp.ReadFile(), "STRING: ABC NUMBER: -000000019");
>+}
>+
>+TEST_F(FormatEntryPointTest, FPrintFError) {
>+  errno = 0;
>+  int result = FPrintF(stdin, "ABC");
>+  EXPECT_LT(result, 0);
>+  EXPECT_EQ(errno, EBADF);
>+}
>+
>+#if __GNUC__
>+TEST_F(FormatEntryPointTest, FprintfTooLarge) {
>+  std::FILE* f = std::fopen("/dev/null", "w");
>+  int width = 2000000000;
>+  errno = 0;
>+  int result = FPrintF(f, "%*d %*d", width, 0, width, 0);
>+  EXPECT_LT(result, 0);
>+  EXPECT_EQ(errno, EFBIG);
>+  std::fclose(f);
>+}
>+
>+TEST_F(FormatEntryPointTest, PrintF) {
>+  int stdout_tmp = dup(STDOUT_FILENO);
>+
>+  TempFile tmp;
>+  std::fflush(stdout);
>+  dup2(fileno(tmp.file()), STDOUT_FILENO);
>+
>+  int result = PrintF("STRING: %s NUMBER: %010d", std::string("ABC"), -19);
>+
>+  std::fflush(stdout);
>+  dup2(stdout_tmp, STDOUT_FILENO);
>+  close(stdout_tmp);
>+
>+  EXPECT_EQ(result, 30);
>+  EXPECT_EQ(tmp.ReadFile(), "STRING: ABC NUMBER: -000000019");
>+}
>+#endif  // __GNUC__
>+
>+TEST_F(FormatEntryPointTest, SNPrintF) {
>+  char buffer[16];
>+  int result =
>+      SNPrintF(buffer, sizeof(buffer), "STRING: %s", std::string("ABC"));
>+  EXPECT_EQ(result, 11);
>+  EXPECT_EQ(std::string(buffer), "STRING: ABC");
>+
>+  result = SNPrintF(buffer, sizeof(buffer), "NUMBER: %d", 123456);
>+  EXPECT_EQ(result, 14);
>+  EXPECT_EQ(std::string(buffer), "NUMBER: 123456");
>+
>+  result = SNPrintF(buffer, sizeof(buffer), "NUMBER: %d", 1234567);
>+  EXPECT_EQ(result, 15);
>+  EXPECT_EQ(std::string(buffer), "NUMBER: 1234567");
>+
>+  result = SNPrintF(buffer, sizeof(buffer), "NUMBER: %d", 12345678);
>+  EXPECT_EQ(result, 16);
>+  EXPECT_EQ(std::string(buffer), "NUMBER: 1234567");
>+
>+  result = SNPrintF(buffer, sizeof(buffer), "NUMBER: %d", 123456789);
>+  EXPECT_EQ(result, 17);
>+  EXPECT_EQ(std::string(buffer), "NUMBER: 1234567");
>+
>+  result = SNPrintF(nullptr, 0, "Just checking the %s of the output.", "size");
>+  EXPECT_EQ(result, 37);
>+}
>+
>+TEST(StrFormat, BehavesAsDocumented) {
>+  std::string s = absl::StrFormat("%s, %d!", "Hello", 123);
>+  EXPECT_EQ("Hello, 123!", s);
>+  // The format of a replacement is
>+  // '%'[position][flags][width['.'precision]][length_modifier][format]
>+  EXPECT_EQ(absl::StrFormat("%1$+3.2Lf", 1.1), "+1.10");
>+  // Text conversion:
>+  //     "c" - Character.              Eg: 'a' -> "A", 20 -> " "
>+  EXPECT_EQ(StrFormat("%c", 'a'), "a");
>+  EXPECT_EQ(StrFormat("%c", 0x20), " ");
>+  //           Formats char and integral types: int, long, uint64_t, etc.
>+  EXPECT_EQ(StrFormat("%c", int{'a'}), "a");
>+  EXPECT_EQ(StrFormat("%c", long{'a'}), "a");  // NOLINT
>+  EXPECT_EQ(StrFormat("%c", uint64_t{'a'}), "a");
>+  //     "s" - std::string                  Eg: "C" -> "C", std::string("C++") -> "C++"
>+  //           Formats std::string, char*, string_view, and Cord.
>+  EXPECT_EQ(StrFormat("%s", "C"), "C");
>+  EXPECT_EQ(StrFormat("%s", std::string("C++")), "C++");
>+  EXPECT_EQ(StrFormat("%s", string_view("view")), "view");
>+  // Integral Conversion
>+  //     These format integral types: char, int, long, uint64_t, etc.
>+  EXPECT_EQ(StrFormat("%d", char{10}), "10");
>+  EXPECT_EQ(StrFormat("%d", int{10}), "10");
>+  EXPECT_EQ(StrFormat("%d", long{10}), "10");  // NOLINT
>+  EXPECT_EQ(StrFormat("%d", uint64_t{10}), "10");
>+  //     d,i - signed decimal          Eg: -10 -> "-10"
>+  EXPECT_EQ(StrFormat("%d", -10), "-10");
>+  EXPECT_EQ(StrFormat("%i", -10), "-10");
>+  //      o  - octal                   Eg:  10 -> "12"
>+  EXPECT_EQ(StrFormat("%o", 10), "12");
>+  //      u  - unsigned decimal        Eg:  10 -> "10"
>+  EXPECT_EQ(StrFormat("%u", 10), "10");
>+  //     x/X - lower,upper case hex    Eg:  10 -> "a"/"A"
>+  EXPECT_EQ(StrFormat("%x", 10), "a");
>+  EXPECT_EQ(StrFormat("%X", 10), "A");
>+  // Floating-point, with upper/lower-case output.
>+  //     These format floating points types: float, double, long double, etc.
>+  EXPECT_EQ(StrFormat("%.1f", float{1}), "1.0");
>+  EXPECT_EQ(StrFormat("%.1f", double{1}), "1.0");
>+  const long double long_double = 1.0;
>+  EXPECT_EQ(StrFormat("%.1f", long_double), "1.0");
>+  //     These also format integral types: char, int, long, uint64_t, etc.:
>+  EXPECT_EQ(StrFormat("%.1f", char{1}), "1.0");
>+  EXPECT_EQ(StrFormat("%.1f", int{1}), "1.0");
>+  EXPECT_EQ(StrFormat("%.1f", long{1}), "1.0");  // NOLINT
>+  EXPECT_EQ(StrFormat("%.1f", uint64_t{1}), "1.0");
>+  //     f/F - decimal.                Eg: 123456789 -> "123456789.000000"
>+  EXPECT_EQ(StrFormat("%f", 123456789), "123456789.000000");
>+  EXPECT_EQ(StrFormat("%F", 123456789), "123456789.000000");
>+  //     e/E - exponentiated           Eg: .01 -> "1.00000e-2"/"1.00000E-2"
>+  EXPECT_EQ(StrFormat("%e", .01), "1.000000e-02");
>+  EXPECT_EQ(StrFormat("%E", .01), "1.000000E-02");
>+  //     g/G - exponentiate to fit     Eg: .01 -> "0.01", 1e10 ->"1e+10"/"1E+10"
>+  EXPECT_EQ(StrFormat("%g", .01), "0.01");
>+  EXPECT_EQ(StrFormat("%g", 1e10), "1e+10");
>+  EXPECT_EQ(StrFormat("%G", 1e10), "1E+10");
>+  //     a/A - lower,upper case hex    Eg: -3.0 -> "-0x1.8p+1"/"-0X1.8P+1"
>+
>+// On Android platform <=21, there is a regression in hexfloat formatting.
>+#if !defined(__ANDROID_API__) || __ANDROID_API__ > 21
>+  EXPECT_EQ(StrFormat("%.1a", -3.0), "-0x1.8p+1");  // .1 to fix MSVC output
>+  EXPECT_EQ(StrFormat("%.1A", -3.0), "-0X1.8P+1");  // .1 to fix MSVC output
>+#endif
>+
>+  // Other conversion
>+  int64_t value = 0x7ffdeb6;
>+  auto ptr_value = static_cast<uintptr_t>(value);
>+  const int& something = *reinterpret_cast<const int*>(ptr_value);
>+  EXPECT_EQ(StrFormat("%p", &something), StrFormat("0x%x", ptr_value));
>+
>+  // Output widths are supported, with optional flags.
>+  EXPECT_EQ(StrFormat("%3d", 1), "  1");
>+  EXPECT_EQ(StrFormat("%3d", 123456), "123456");
>+  EXPECT_EQ(StrFormat("%06.2f", 1.234), "001.23");
>+  EXPECT_EQ(StrFormat("%+d", 1), "+1");
>+  EXPECT_EQ(StrFormat("% d", 1), " 1");
>+  EXPECT_EQ(StrFormat("%-4d", -1), "-1  ");
>+  EXPECT_EQ(StrFormat("%#o", 10), "012");
>+  EXPECT_EQ(StrFormat("%#x", 15), "0xf");
>+  EXPECT_EQ(StrFormat("%04d", 8), "0008");
>+  // Posix positional substitution.
>+  EXPECT_EQ(absl::StrFormat("%2$s, %3$s, %1$s!", "vici", "veni", "vidi"),
>+            "veni, vidi, vici!");
>+  // Length modifiers are ignored.
>+  EXPECT_EQ(StrFormat("%hhd", int{1}), "1");
>+  EXPECT_EQ(StrFormat("%hd", int{1}), "1");
>+  EXPECT_EQ(StrFormat("%ld", int{1}), "1");
>+  EXPECT_EQ(StrFormat("%lld", int{1}), "1");
>+  EXPECT_EQ(StrFormat("%Ld", int{1}), "1");
>+  EXPECT_EQ(StrFormat("%jd", int{1}), "1");
>+  EXPECT_EQ(StrFormat("%zd", int{1}), "1");
>+  EXPECT_EQ(StrFormat("%td", int{1}), "1");
>+  EXPECT_EQ(StrFormat("%qd", int{1}), "1");
>+}
>+
>+using str_format_internal::ExtendedParsedFormat;
>+using str_format_internal::ParsedFormatBase;
>+
>+struct SummarizeConsumer {
>+  std::string* out;
>+  explicit SummarizeConsumer(std::string* out) : out(out) {}
>+
>+  bool Append(string_view s) {
>+    *out += "[" + std::string(s) + "]";
>+    return true;
>+  }
>+
>+  bool ConvertOne(const str_format_internal::UnboundConversion& conv,
>+                  string_view s) {
>+    *out += "{";
>+    *out += std::string(s);
>+    *out += ":";
>+    *out += std::to_string(conv.arg_position) + "$";
>+    if (conv.width.is_from_arg()) {
>+      *out += std::to_string(conv.width.get_from_arg()) + "$*";
>+    }
>+    if (conv.precision.is_from_arg()) {
>+      *out += "." + std::to_string(conv.precision.get_from_arg()) + "$*";
>+    }
>+    *out += conv.conv.Char();
>+    *out += "}";
>+    return true;
>+  }
>+};
>+
>+std::string SummarizeParsedFormat(const ParsedFormatBase& pc) {
>+  std::string out;
>+  if (!pc.ProcessFormat(SummarizeConsumer(&out))) out += "!";
>+  return out;
>+}
>+
>+class ParsedFormatTest : public testing::Test {};
>+
>+TEST_F(ParsedFormatTest, SimpleChecked) {
>+  EXPECT_EQ("[ABC]{d:1$d}[DEF]",
>+            SummarizeParsedFormat(ParsedFormat<'d'>("ABC%dDEF")));
>+  EXPECT_EQ("{s:1$s}[FFF]{d:2$d}[ZZZ]{f:3$f}",
>+            SummarizeParsedFormat(ParsedFormat<'s', 'd', 'f'>("%sFFF%dZZZ%f")));
>+  EXPECT_EQ("{s:1$s}[ ]{.*d:3$.2$*d}",
>+            SummarizeParsedFormat(ParsedFormat<'s', '*', 'd'>("%s %.*d")));
>+}
>+
>+TEST_F(ParsedFormatTest, SimpleUncheckedCorrect) {
>+  auto f = ParsedFormat<'d'>::New("ABC%dDEF");
>+  ASSERT_TRUE(f);
>+  EXPECT_EQ("[ABC]{d:1$d}[DEF]", SummarizeParsedFormat(*f));
>+
>+  std::string format = "%sFFF%dZZZ%f";
>+  auto f2 = ParsedFormat<'s', 'd', 'f'>::New(format);
>+
>+  ASSERT_TRUE(f2);
>+  EXPECT_EQ("{s:1$s}[FFF]{d:2$d}[ZZZ]{f:3$f}", SummarizeParsedFormat(*f2));
>+
>+  f2 = ParsedFormat<'s', 'd', 'f'>::New("%s %d %f");
>+
>+  ASSERT_TRUE(f2);
>+  EXPECT_EQ("{s:1$s}[ ]{d:2$d}[ ]{f:3$f}", SummarizeParsedFormat(*f2));
>+
>+  auto star = ParsedFormat<'*', 'd'>::New("%*d");
>+  ASSERT_TRUE(star);
>+  EXPECT_EQ("{*d:2$1$*d}", SummarizeParsedFormat(*star));
>+
>+  auto dollar = ParsedFormat<'d', 's'>::New("%2$s %1$d");
>+  ASSERT_TRUE(dollar);
>+  EXPECT_EQ("{2$s:2$s}[ ]{1$d:1$d}", SummarizeParsedFormat(*dollar));
>+  // with reuse
>+  dollar = ParsedFormat<'d', 's'>::New("%2$s %1$d %1$d");
>+  ASSERT_TRUE(dollar);
>+  EXPECT_EQ("{2$s:2$s}[ ]{1$d:1$d}[ ]{1$d:1$d}",
>+            SummarizeParsedFormat(*dollar));
>+}
>+
>+TEST_F(ParsedFormatTest, SimpleUncheckedIgnoredArgs) {
>+  EXPECT_FALSE((ParsedFormat<'d', 's'>::New("ABC")));
>+  EXPECT_FALSE((ParsedFormat<'d', 's'>::New("%dABC")));
>+  EXPECT_FALSE((ParsedFormat<'d', 's'>::New("ABC%2$s")));
>+  auto f = ParsedFormat<'d', 's'>::NewAllowIgnored("ABC");
>+  ASSERT_TRUE(f);
>+  EXPECT_EQ("[ABC]", SummarizeParsedFormat(*f));
>+  f = ParsedFormat<'d', 's'>::NewAllowIgnored("%dABC");
>+  ASSERT_TRUE(f);
>+  EXPECT_EQ("{d:1$d}[ABC]", SummarizeParsedFormat(*f));
>+  f = ParsedFormat<'d', 's'>::NewAllowIgnored("ABC%2$s");
>+  ASSERT_TRUE(f);
>+  EXPECT_EQ("[ABC]{2$s:2$s}", SummarizeParsedFormat(*f));
>+}
>+
>+TEST_F(ParsedFormatTest, SimpleUncheckedUnsupported) {
>+  EXPECT_FALSE(ParsedFormat<'d'>::New("%1$d %1$x"));
>+  EXPECT_FALSE(ParsedFormat<'x'>::New("%1$d %1$x"));
>+}
>+
>+TEST_F(ParsedFormatTest, SimpleUncheckedIncorrect) {
>+  EXPECT_FALSE(ParsedFormat<'d'>::New(""));
>+
>+  EXPECT_FALSE(ParsedFormat<'d'>::New("ABC%dDEF%d"));
>+
>+  std::string format = "%sFFF%dZZZ%f";
>+  EXPECT_FALSE((ParsedFormat<'s', 'd', 'g'>::New(format)));
>+}
>+
>+using str_format_internal::Conv;
>+
>+TEST_F(ParsedFormatTest, UncheckedCorrect) {
>+  auto f = ExtendedParsedFormat<Conv::d>::New("ABC%dDEF");
>+  ASSERT_TRUE(f);
>+  EXPECT_EQ("[ABC]{d:1$d}[DEF]", SummarizeParsedFormat(*f));
>+
>+  std::string format = "%sFFF%dZZZ%f";
>+  auto f2 =
>+      ExtendedParsedFormat<Conv::string, Conv::d, Conv::floating>::New(format);
>+
>+  ASSERT_TRUE(f2);
>+  EXPECT_EQ("{s:1$s}[FFF]{d:2$d}[ZZZ]{f:3$f}", SummarizeParsedFormat(*f2));
>+
>+  f2 = ExtendedParsedFormat<Conv::string, Conv::d, Conv::floating>::New(
>+      "%s %d %f");
>+
>+  ASSERT_TRUE(f2);
>+  EXPECT_EQ("{s:1$s}[ ]{d:2$d}[ ]{f:3$f}", SummarizeParsedFormat(*f2));
>+
>+  auto star = ExtendedParsedFormat<Conv::star, Conv::d>::New("%*d");
>+  ASSERT_TRUE(star);
>+  EXPECT_EQ("{*d:2$1$*d}", SummarizeParsedFormat(*star));
>+
>+  auto dollar = ExtendedParsedFormat<Conv::d, Conv::s>::New("%2$s %1$d");
>+  ASSERT_TRUE(dollar);
>+  EXPECT_EQ("{2$s:2$s}[ ]{1$d:1$d}", SummarizeParsedFormat(*dollar));
>+  // with reuse
>+  dollar = ExtendedParsedFormat<Conv::d, Conv::s>::New("%2$s %1$d %1$d");
>+  ASSERT_TRUE(dollar);
>+  EXPECT_EQ("{2$s:2$s}[ ]{1$d:1$d}[ ]{1$d:1$d}",
>+            SummarizeParsedFormat(*dollar));
>+}
>+
>+TEST_F(ParsedFormatTest, UncheckedIgnoredArgs) {
>+  EXPECT_FALSE((ExtendedParsedFormat<Conv::d, Conv::s>::New("ABC")));
>+  EXPECT_FALSE((ExtendedParsedFormat<Conv::d, Conv::s>::New("%dABC")));
>+  EXPECT_FALSE((ExtendedParsedFormat<Conv::d, Conv::s>::New("ABC%2$s")));
>+  auto f = ExtendedParsedFormat<Conv::d, Conv::s>::NewAllowIgnored("ABC");
>+  ASSERT_TRUE(f);
>+  EXPECT_EQ("[ABC]", SummarizeParsedFormat(*f));
>+  f = ExtendedParsedFormat<Conv::d, Conv::s>::NewAllowIgnored("%dABC");
>+  ASSERT_TRUE(f);
>+  EXPECT_EQ("{d:1$d}[ABC]", SummarizeParsedFormat(*f));
>+  f = ExtendedParsedFormat<Conv::d, Conv::s>::NewAllowIgnored("ABC%2$s");
>+  ASSERT_TRUE(f);
>+  EXPECT_EQ("[ABC]{2$s:2$s}", SummarizeParsedFormat(*f));
>+}
>+
>+TEST_F(ParsedFormatTest, UncheckedMultipleTypes) {
>+  auto dx = ExtendedParsedFormat<Conv::d | Conv::x>::New("%1$d %1$x");
>+  EXPECT_TRUE(dx);
>+  EXPECT_EQ("{1$d:1$d}[ ]{1$x:1$x}", SummarizeParsedFormat(*dx));
>+
>+  dx = ExtendedParsedFormat<Conv::d | Conv::x>::New("%1$d");
>+  EXPECT_TRUE(dx);
>+  EXPECT_EQ("{1$d:1$d}", SummarizeParsedFormat(*dx));
>+}
>+
>+TEST_F(ParsedFormatTest, UncheckedIncorrect) {
>+  EXPECT_FALSE(ExtendedParsedFormat<Conv::d>::New(""));
>+
>+  EXPECT_FALSE(ExtendedParsedFormat<Conv::d>::New("ABC%dDEF%d"));
>+
>+  std::string format = "%sFFF%dZZZ%f";
>+  EXPECT_FALSE((ExtendedParsedFormat<Conv::s, Conv::d, Conv::g>::New(format)));
>+}
>+
>+TEST_F(ParsedFormatTest, RegressionMixPositional) {
>+  EXPECT_FALSE((ExtendedParsedFormat<Conv::d, Conv::o>::New("%1$d %o")));
>+}
>+
>+}  // namespace
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_join.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_join.h
>new file mode 100644
>index 00000000000..bd4d0e1d932
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_join.h
>@@ -0,0 +1,288 @@
>+//
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// -----------------------------------------------------------------------------
>+// File: str_join.h
>+// -----------------------------------------------------------------------------
>+//
>+// This header file contains functions for joining a range of elements and
>+// returning the result as a std::string. StrJoin operations are specified by passing
>+// a range, a separator std::string to use between the elements joined, and an
>+// optional Formatter responsible for converting each argument in the range to a
>+// std::string. If omitted, a default `AlphaNumFormatter()` is called on the elements
>+// to be joined, using the same formatting that `absl::StrCat()` uses. This
>+// package defines a number of default formatters, and you can define your own
>+// implementations.
>+//
>+// Ranges are specified by passing a container with `std::begin()` and
>+// `std::end()` iterators, container-specific `begin()` and `end()` iterators, a
>+// brace-initialized `std::initializer_list`, or a `std::tuple` of heterogeneous
>+// objects. The separator std::string is specified as an `absl::string_view`.
>+//
>+// Because the default formatter uses the `absl::AlphaNum` class,
>+// `absl::StrJoin()`, like `absl::StrCat()`, will work out-of-the-box on
>+// collections of strings, ints, floats, doubles, etc.
>+//
>+// Example:
>+//
>+//   std::vector<std::string> v = {"foo", "bar", "baz"};
>+//   std::string s = absl::StrJoin(v, "-");
>+//   EXPECT_EQ("foo-bar-baz", s);
>+//
>+// See comments on the `absl::StrJoin()` function for more examples.
>+
>+#ifndef ABSL_STRINGS_STR_JOIN_H_
>+#define ABSL_STRINGS_STR_JOIN_H_
>+
>+#include <cstdio>
>+#include <cstring>
>+#include <initializer_list>
>+#include <iterator>
>+#include <string>
>+#include <tuple>
>+#include <utility>
>+
>+#include "absl/base/macros.h"
>+#include "absl/strings/internal/str_join_internal.h"
>+#include "absl/strings/string_view.h"
>+
>+namespace absl {
>+
>+// -----------------------------------------------------------------------------
>+// Concept: Formatter
>+// -----------------------------------------------------------------------------
>+//
>+// A Formatter is a function object that is responsible for formatting its
>+// argument as a std::string and appending it to a given output std::string. Formatters
>+// may be implemented as function objects, lambdas, or normal functions. You may
>+// provide your own Formatter to enable `absl::StrJoin()` to work with arbitrary
>+// types.
>+//
>+// The following is an example of a custom Formatter that simply uses
>+// `std::to_string()` to format an integer as a std::string.
>+//
>+//   struct MyFormatter {
>+//     void operator()(std::string* out, int i) const {
>+//       out->append(std::to_string(i));
>+//     }
>+//   };
>+//
>+// You would use the above formatter by passing an instance of it as the final
>+// argument to `absl::StrJoin()`:
>+//
>+//   std::vector<int> v = {1, 2, 3, 4};
>+//   std::string s = absl::StrJoin(v, "-", MyFormatter());
>+//   EXPECT_EQ("1-2-3-4", s);
>+//
>+// The following standard formatters are provided within this file:
>+//
>+// - `AlphaNumFormatter()` (the default)
>+// - `StreamFormatter()`
>+// - `PairFormatter()`
>+// - `DereferenceFormatter()`
>+
>+// AlphaNumFormatter()
>+//
>+// Default formatter used if none is specified. Uses `absl::AlphaNum` to convert
>+// numeric arguments to strings.
>+inline strings_internal::AlphaNumFormatterImpl AlphaNumFormatter() {
>+  return strings_internal::AlphaNumFormatterImpl();
>+}
>+
>+// StreamFormatter()
>+//
>+// Formats its argument using the << operator.
>+inline strings_internal::StreamFormatterImpl StreamFormatter() {
>+  return strings_internal::StreamFormatterImpl();
>+}
>+
>+// Function Template: PairFormatter(Formatter, absl::string_view, Formatter)
>+//
>+// Formats a `std::pair` by putting a given separator between the pair's
>+// `.first` and `.second` members. This formatter allows you to specify
>+// custom Formatters for both the first and second member of each pair.
>+template <typename FirstFormatter, typename SecondFormatter>
>+inline strings_internal::PairFormatterImpl<FirstFormatter, SecondFormatter>
>+PairFormatter(FirstFormatter f1, absl::string_view sep, SecondFormatter f2) {
>+  return strings_internal::PairFormatterImpl<FirstFormatter, SecondFormatter>(
>+      std::move(f1), sep, std::move(f2));
>+}
>+
>+// Function overload of PairFormatter() for using a default
>+// `AlphaNumFormatter()` for each Formatter in the pair.
>+inline strings_internal::PairFormatterImpl<
>+    strings_internal::AlphaNumFormatterImpl,
>+    strings_internal::AlphaNumFormatterImpl>
>+PairFormatter(absl::string_view sep) {
>+  return PairFormatter(AlphaNumFormatter(), sep, AlphaNumFormatter());
>+}
>+
>+// Function Template: DereferenceFormatter(Formatter)
>+//
>+// Formats its argument by dereferencing it and then applying the given
>+// formatter. This formatter is useful for formatting a container of
>+// pointer-to-T. This pattern often shows up when joining repeated fields in
>+// protocol buffers.
>+template <typename Formatter>
>+strings_internal::DereferenceFormatterImpl<Formatter> DereferenceFormatter(
>+    Formatter&& f) {
>+  return strings_internal::DereferenceFormatterImpl<Formatter>(
>+      std::forward<Formatter>(f));
>+}
>+
>+// Function overload of `DererefenceFormatter()` for using a default
>+// `AlphaNumFormatter()`.
>+inline strings_internal::DereferenceFormatterImpl<
>+    strings_internal::AlphaNumFormatterImpl>
>+DereferenceFormatter() {
>+  return strings_internal::DereferenceFormatterImpl<
>+      strings_internal::AlphaNumFormatterImpl>(AlphaNumFormatter());
>+}
>+
>+// -----------------------------------------------------------------------------
>+// StrJoin()
>+// -----------------------------------------------------------------------------
>+//
>+// Joins a range of elements and returns the result as a std::string.
>+// `absl::StrJoin()` takes a range, a separator std::string to use between the
>+// elements joined, and an optional Formatter responsible for converting each
>+// argument in the range to a std::string.
>+//
>+// If omitted, the default `AlphaNumFormatter()` is called on the elements to be
>+// joined.
>+//
>+// Example 1:
>+//   // Joins a collection of strings. This pattern also works with a collection
>+//   // of `absl::string_view` or even `const char*`.
>+//   std::vector<std::string> v = {"foo", "bar", "baz"};
>+//   std::string s = absl::StrJoin(v, "-");
>+//   EXPECT_EQ("foo-bar-baz", s);
>+//
>+// Example 2:
>+//   // Joins the values in the given `std::initializer_list<>` specified using
>+//   // brace initialization. This pattern also works with an initializer_list
>+//   // of ints or `absl::string_view` -- any `AlphaNum`-compatible type.
>+//   std::string s = absl::StrJoin({"foo", "bar", "baz"}, "-");
>+//   EXPECT_EQ("foo-bar-baz", s);
>+//
>+// Example 3:
>+//   // Joins a collection of ints. This pattern also works with floats,
>+//   // doubles, int64s -- any `StrCat()`-compatible type.
>+//   std::vector<int> v = {1, 2, 3, -4};
>+//   std::string s = absl::StrJoin(v, "-");
>+//   EXPECT_EQ("1-2-3--4", s);
>+//
>+// Example 4:
>+//   // Joins a collection of pointer-to-int. By default, pointers are
>+//   // dereferenced and the pointee is formatted using the default format for
>+//   // that type; such dereferencing occurs for all levels of indirection, so
>+//   // this pattern works just as well for `std::vector<int**>` as for
>+//   // `std::vector<int*>`.
>+//   int x = 1, y = 2, z = 3;
>+//   std::vector<int*> v = {&x, &y, &z};
>+//   std::string s = absl::StrJoin(v, "-");
>+//   EXPECT_EQ("1-2-3", s);
>+//
>+// Example 5:
>+//   // Dereferencing of `std::unique_ptr<>` is also supported:
>+//   std::vector<std::unique_ptr<int>> v
>+//   v.emplace_back(new int(1));
>+//   v.emplace_back(new int(2));
>+//   v.emplace_back(new int(3));
>+//   std::string s = absl::StrJoin(v, "-");
>+//   EXPECT_EQ("1-2-3", s);
>+//
>+// Example 6:
>+//   // Joins a `std::map`, with each key-value pair separated by an equals
>+//   // sign. This pattern would also work with, say, a
>+//   // `std::vector<std::pair<>>`.
>+//   std::map<std::string, int> m = {
>+//       std::make_pair("a", 1),
>+//       std::make_pair("b", 2),
>+//       std::make_pair("c", 3)};
>+//   std::string s = absl::StrJoin(m, ",", absl::PairFormatter("="));
>+//   EXPECT_EQ("a=1,b=2,c=3", s);
>+//
>+// Example 7:
>+//   // These examples show how `absl::StrJoin()` handles a few common edge
>+//   // cases:
>+//   std::vector<std::string> v_empty;
>+//   EXPECT_EQ("", absl::StrJoin(v_empty, "-"));
>+//
>+//   std::vector<std::string> v_one_item = {"foo"};
>+//   EXPECT_EQ("foo", absl::StrJoin(v_one_item, "-"));
>+//
>+//   std::vector<std::string> v_empty_string = {""};
>+//   EXPECT_EQ("", absl::StrJoin(v_empty_string, "-"));
>+//
>+//   std::vector<std::string> v_one_item_empty_string = {"a", ""};
>+//   EXPECT_EQ("a-", absl::StrJoin(v_one_item_empty_string, "-"));
>+//
>+//   std::vector<std::string> v_two_empty_string = {"", ""};
>+//   EXPECT_EQ("-", absl::StrJoin(v_two_empty_string, "-"));
>+//
>+// Example 8:
>+//   // Joins a `std::tuple<T...>` of heterogeneous types, converting each to
>+//   // a std::string using the `absl::AlphaNum` class.
>+//   std::string s = absl::StrJoin(std::make_tuple(123, "abc", 0.456), "-");
>+//   EXPECT_EQ("123-abc-0.456", s);
>+
>+template <typename Iterator, typename Formatter>
>+std::string StrJoin(Iterator start, Iterator end, absl::string_view sep,
>+               Formatter&& fmt) {
>+  return strings_internal::JoinAlgorithm(start, end, sep, fmt);
>+}
>+
>+template <typename Range, typename Formatter>
>+std::string StrJoin(const Range& range, absl::string_view separator,
>+               Formatter&& fmt) {
>+  return strings_internal::JoinRange(range, separator, fmt);
>+}
>+
>+template <typename T, typename Formatter>
>+std::string StrJoin(std::initializer_list<T> il, absl::string_view separator,
>+               Formatter&& fmt) {
>+  return strings_internal::JoinRange(il, separator, fmt);
>+}
>+
>+template <typename... T, typename Formatter>
>+std::string StrJoin(const std::tuple<T...>& value, absl::string_view separator,
>+               Formatter&& fmt) {
>+  return strings_internal::JoinAlgorithm(value, separator, fmt);
>+}
>+
>+template <typename Iterator>
>+std::string StrJoin(Iterator start, Iterator end, absl::string_view separator) {
>+  return strings_internal::JoinRange(start, end, separator);
>+}
>+
>+template <typename Range>
>+std::string StrJoin(const Range& range, absl::string_view separator) {
>+  return strings_internal::JoinRange(range, separator);
>+}
>+
>+template <typename T>
>+std::string StrJoin(std::initializer_list<T> il, absl::string_view separator) {
>+  return strings_internal::JoinRange(il, separator);
>+}
>+
>+template <typename... T>
>+std::string StrJoin(const std::tuple<T...>& value, absl::string_view separator) {
>+  return strings_internal::JoinAlgorithm(value, separator, AlphaNumFormatter());
>+}
>+
>+}  // namespace absl
>+
>+#endif  // ABSL_STRINGS_STR_JOIN_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_join_benchmark.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_join_benchmark.cc
>new file mode 100644
>index 00000000000..7fb0e4973cb
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_join_benchmark.cc
>@@ -0,0 +1,96 @@
>+//
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/strings/str_join.h"
>+
>+#include <string>
>+#include <vector>
>+#include <utility>
>+
>+#include "benchmark/benchmark.h"
>+
>+namespace {
>+
>+void BM_Join2_Strings(benchmark::State& state) {
>+  const int string_len = state.range(0);
>+  const int num_strings = state.range(1);
>+  const std::string s(string_len, 'x');
>+  const std::vector<std::string> v(num_strings, s);
>+  for (auto _ : state) {
>+    std::string s = absl::StrJoin(v, "-");
>+    benchmark::DoNotOptimize(s);
>+  }
>+}
>+BENCHMARK(BM_Join2_Strings)
>+    ->ArgPair(1 << 0, 1 << 3)
>+    ->ArgPair(1 << 10, 1 << 3)
>+    ->ArgPair(1 << 13, 1 << 3)
>+    ->ArgPair(1 << 0, 1 << 10)
>+    ->ArgPair(1 << 10, 1 << 10)
>+    ->ArgPair(1 << 13, 1 << 10)
>+    ->ArgPair(1 << 0, 1 << 13)
>+    ->ArgPair(1 << 10, 1 << 13)
>+    ->ArgPair(1 << 13, 1 << 13);
>+
>+void BM_Join2_Ints(benchmark::State& state) {
>+  const int num_ints = state.range(0);
>+  const std::vector<int> v(num_ints, 42);
>+  for (auto _ : state) {
>+    std::string s = absl::StrJoin(v, "-");
>+    benchmark::DoNotOptimize(s);
>+  }
>+}
>+BENCHMARK(BM_Join2_Ints)->Range(0, 1 << 13);
>+
>+void BM_Join2_KeysAndValues(benchmark::State& state) {
>+  const int string_len = state.range(0);
>+  const int num_pairs = state.range(1);
>+  const std::string s(string_len, 'x');
>+  const std::vector<std::pair<std::string, int>> v(num_pairs, std::make_pair(s, 42));
>+  for (auto _ : state) {
>+    std::string s = absl::StrJoin(v, ",", absl::PairFormatter("="));
>+    benchmark::DoNotOptimize(s);
>+  }
>+}
>+BENCHMARK(BM_Join2_KeysAndValues)
>+    ->ArgPair(1 << 0, 1 << 3)
>+    ->ArgPair(1 << 10, 1 << 3)
>+    ->ArgPair(1 << 13, 1 << 3)
>+    ->ArgPair(1 << 0, 1 << 10)
>+    ->ArgPair(1 << 10, 1 << 10)
>+    ->ArgPair(1 << 13, 1 << 10)
>+    ->ArgPair(1 << 0, 1 << 13)
>+    ->ArgPair(1 << 10, 1 << 13)
>+    ->ArgPair(1 << 13, 1 << 13);
>+
>+void BM_JoinStreamable(benchmark::State& state) {
>+  const int string_len = state.range(0);
>+  const int num_strings = state.range(1);
>+  const std::vector<std::string> v(num_strings, std::string(string_len, 'x'));
>+  for (auto _ : state) {
>+    std::string s = absl::StrJoin(v, "", absl::StreamFormatter());
>+    benchmark::DoNotOptimize(s);
>+  }
>+}
>+BENCHMARK(BM_JoinStreamable)
>+    ->ArgPair(0, 0)
>+    ->ArgPair(16, 1)
>+    ->ArgPair(256, 1)
>+    ->ArgPair(16, 16)
>+    ->ArgPair(256, 16)
>+    ->ArgPair(16, 256)
>+    ->ArgPair(256, 256);
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_join_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_join_test.cc
>new file mode 100644
>index 00000000000..c941f9c80d4
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_join_test.cc
>@@ -0,0 +1,472 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+// Unit tests for all join.h functions
>+
>+#include "absl/strings/str_join.h"
>+
>+#include <cstddef>
>+#include <cstdint>
>+#include <cstdio>
>+#include <functional>
>+#include <initializer_list>
>+#include <map>
>+#include <memory>
>+#include <ostream>
>+#include <tuple>
>+#include <type_traits>
>+#include <vector>
>+
>+#include "gtest/gtest.h"
>+#include "absl/base/macros.h"
>+#include "absl/memory/memory.h"
>+#include "absl/strings/str_cat.h"
>+#include "absl/strings/str_split.h"
>+
>+namespace {
>+
>+TEST(StrJoin, APIExamples) {
>+  {
>+    // Collection of strings
>+    std::vector<std::string> v = {"foo", "bar", "baz"};
>+    EXPECT_EQ("foo-bar-baz", absl::StrJoin(v, "-"));
>+  }
>+
>+  {
>+    // Collection of absl::string_view
>+    std::vector<absl::string_view> v = {"foo", "bar", "baz"};
>+    EXPECT_EQ("foo-bar-baz", absl::StrJoin(v, "-"));
>+  }
>+
>+  {
>+    // Collection of const char*
>+    std::vector<const char*> v = {"foo", "bar", "baz"};
>+    EXPECT_EQ("foo-bar-baz", absl::StrJoin(v, "-"));
>+  }
>+
>+  {
>+    // Collection of non-const char*
>+    std::string a = "foo", b = "bar", c = "baz";
>+    std::vector<char*> v = {&a[0], &b[0], &c[0]};
>+    EXPECT_EQ("foo-bar-baz", absl::StrJoin(v, "-"));
>+  }
>+
>+  {
>+    // Collection of ints
>+    std::vector<int> v = {1, 2, 3, -4};
>+    EXPECT_EQ("1-2-3--4", absl::StrJoin(v, "-"));
>+  }
>+
>+  {
>+    // Literals passed as a std::initializer_list
>+    std::string s = absl::StrJoin({"a", "b", "c"}, "-");
>+    EXPECT_EQ("a-b-c", s);
>+  }
>+  {
>+    // Join a std::tuple<T...>.
>+    std::string s = absl::StrJoin(std::make_tuple(123, "abc", 0.456), "-");
>+    EXPECT_EQ("123-abc-0.456", s);
>+  }
>+
>+  {
>+    // Collection of unique_ptrs
>+    std::vector<std::unique_ptr<int>> v;
>+    v.emplace_back(new int(1));
>+    v.emplace_back(new int(2));
>+    v.emplace_back(new int(3));
>+    EXPECT_EQ("1-2-3", absl::StrJoin(v, "-"));
>+  }
>+
>+  {
>+    // Array of ints
>+    const int a[] = {1, 2, 3, -4};
>+    EXPECT_EQ("1-2-3--4", absl::StrJoin(a, a + ABSL_ARRAYSIZE(a), "-"));
>+  }
>+
>+  {
>+    // Collection of pointers
>+    int x = 1, y = 2, z = 3;
>+    std::vector<int*> v = {&x, &y, &z};
>+    EXPECT_EQ("1-2-3", absl::StrJoin(v, "-"));
>+  }
>+
>+  {
>+    // Collection of pointers to pointers
>+    int x = 1, y = 2, z = 3;
>+    int *px = &x, *py = &y, *pz = &z;
>+    std::vector<int**> v = {&px, &py, &pz};
>+    EXPECT_EQ("1-2-3", absl::StrJoin(v, "-"));
>+  }
>+
>+  {
>+    // Collection of pointers to std::string
>+    std::string a("a"), b("b");
>+    std::vector<std::string*> v = {&a, &b};
>+    EXPECT_EQ("a-b", absl::StrJoin(v, "-"));
>+  }
>+
>+  {
>+    // A std::map, which is a collection of std::pair<>s.
>+    std::map<std::string, int> m = { {"a", 1}, {"b", 2}, {"c", 3} };
>+    EXPECT_EQ("a=1,b=2,c=3", absl::StrJoin(m, ",", absl::PairFormatter("=")));
>+  }
>+
>+  {
>+    // Shows absl::StrSplit and absl::StrJoin working together. This example is
>+    // equivalent to s/=/-/g.
>+    const std::string s = "a=b=c=d";
>+    EXPECT_EQ("a-b-c-d", absl::StrJoin(absl::StrSplit(s, "="), "-"));
>+  }
>+
>+  //
>+  // A few examples of edge cases
>+  //
>+
>+  {
>+    // Empty range yields an empty std::string.
>+    std::vector<std::string> v;
>+    EXPECT_EQ("", absl::StrJoin(v, "-"));
>+  }
>+
>+  {
>+    // A range of 1 element gives a std::string with that element but no separator.
>+    std::vector<std::string> v = {"foo"};
>+    EXPECT_EQ("foo", absl::StrJoin(v, "-"));
>+  }
>+
>+  {
>+    // A range with a single empty std::string element
>+    std::vector<std::string> v = {""};
>+    EXPECT_EQ("", absl::StrJoin(v, "-"));
>+  }
>+
>+  {
>+    // A range with 2 elements, one of which is an empty std::string
>+    std::vector<std::string> v = {"a", ""};
>+    EXPECT_EQ("a-", absl::StrJoin(v, "-"));
>+  }
>+
>+  {
>+    // A range with 2 empty elements.
>+    std::vector<std::string> v = {"", ""};
>+    EXPECT_EQ("-", absl::StrJoin(v, "-"));
>+  }
>+
>+  {
>+    // A std::vector of bool.
>+    std::vector<bool> v = {true, false, true};
>+    EXPECT_EQ("1-0-1", absl::StrJoin(v, "-"));
>+  }
>+}
>+
>+TEST(StrJoin, CustomFormatter) {
>+  std::vector<std::string> v{"One", "Two", "Three"};
>+  {
>+    std::string joined = absl::StrJoin(v, "", [](std::string* out, const std::string& in) {
>+      absl::StrAppend(out, "(", in, ")");
>+    });
>+    EXPECT_EQ("(One)(Two)(Three)", joined);
>+  }
>+  {
>+    class ImmovableFormatter {
>+     public:
>+      void operator()(std::string* out, const std::string& in) {
>+        absl::StrAppend(out, "(", in, ")");
>+      }
>+      ImmovableFormatter() {}
>+      ImmovableFormatter(const ImmovableFormatter&) = delete;
>+    };
>+    EXPECT_EQ("(One)(Two)(Three)", absl::StrJoin(v, "", ImmovableFormatter()));
>+  }
>+  {
>+    class OverloadedFormatter {
>+     public:
>+      void operator()(std::string* out, const std::string& in) {
>+        absl::StrAppend(out, "(", in, ")");
>+      }
>+      void operator()(std::string* out, const std::string& in) const {
>+        absl::StrAppend(out, "[", in, "]");
>+      }
>+    };
>+    EXPECT_EQ("(One)(Two)(Three)", absl::StrJoin(v, "", OverloadedFormatter()));
>+    const OverloadedFormatter fmt = {};
>+    EXPECT_EQ("[One][Two][Three]", absl::StrJoin(v, "", fmt));
>+  }
>+}
>+
>+//
>+// Tests the Formatters
>+//
>+
>+TEST(AlphaNumFormatter, FormatterAPI) {
>+  // Not an exhaustive test. See strings/strcat_test.h for the exhaustive test
>+  // of what AlphaNum can convert.
>+  auto f = absl::AlphaNumFormatter();
>+  std::string s;
>+  f(&s, "Testing: ");
>+  f(&s, static_cast<int>(1));
>+  f(&s, static_cast<int16_t>(2));
>+  f(&s, static_cast<int64_t>(3));
>+  f(&s, static_cast<float>(4));
>+  f(&s, static_cast<double>(5));
>+  f(&s, static_cast<unsigned>(6));
>+  f(&s, static_cast<size_t>(7));
>+  f(&s, absl::string_view(" OK"));
>+  EXPECT_EQ("Testing: 1234567 OK", s);
>+}
>+
>+// Make sure people who are mistakenly using std::vector<bool> even though
>+// they're not memory-constrained can use absl::AlphaNumFormatter().
>+TEST(AlphaNumFormatter, VectorOfBool) {
>+  auto f = absl::AlphaNumFormatter();
>+  std::string s;
>+  std::vector<bool> v = {true, false, true};
>+  f(&s, *v.cbegin());
>+  f(&s, *v.begin());
>+  f(&s, v[1]);
>+  EXPECT_EQ("110", s);
>+}
>+
>+TEST(AlphaNumFormatter, AlphaNum) {
>+  auto f = absl::AlphaNumFormatter();
>+  std::string s;
>+  f(&s, absl::AlphaNum("hello"));
>+  EXPECT_EQ("hello", s);
>+}
>+
>+struct StreamableType {
>+  std::string contents;
>+};
>+inline std::ostream& operator<<(std::ostream& os, const StreamableType& t) {
>+  os << "Streamable:" << t.contents;
>+  return os;
>+}
>+
>+TEST(StreamFormatter, FormatterAPI) {
>+  auto f = absl::StreamFormatter();
>+  std::string s;
>+  f(&s, "Testing: ");
>+  f(&s, static_cast<int>(1));
>+  f(&s, static_cast<int16_t>(2));
>+  f(&s, static_cast<int64_t>(3));
>+  f(&s, static_cast<float>(4));
>+  f(&s, static_cast<double>(5));
>+  f(&s, static_cast<unsigned>(6));
>+  f(&s, static_cast<size_t>(7));
>+  f(&s, absl::string_view(" OK "));
>+  StreamableType streamable = {"object"};
>+  f(&s, streamable);
>+  EXPECT_EQ("Testing: 1234567 OK Streamable:object", s);
>+}
>+
>+// A dummy formatter that wraps each element in parens. Used in some tests
>+// below.
>+struct TestingParenFormatter {
>+  template <typename T>
>+  void operator()(std::string* s, const T& t) {
>+    absl::StrAppend(s, "(", t, ")");
>+  }
>+};
>+
>+TEST(PairFormatter, FormatterAPI) {
>+  {
>+    // Tests default PairFormatter(sep) that uses AlphaNumFormatter for the
>+    // 'first' and 'second' members.
>+    const auto f = absl::PairFormatter("=");
>+    std::string s;
>+    f(&s, std::make_pair("a", "b"));
>+    f(&s, std::make_pair(1, 2));
>+    EXPECT_EQ("a=b1=2", s);
>+  }
>+
>+  {
>+    // Tests using a custom formatter for the 'first' and 'second' members.
>+    auto f = absl::PairFormatter(TestingParenFormatter(), "=",
>+                                 TestingParenFormatter());
>+    std::string s;
>+    f(&s, std::make_pair("a", "b"));
>+    f(&s, std::make_pair(1, 2));
>+    EXPECT_EQ("(a)=(b)(1)=(2)", s);
>+  }
>+}
>+
>+TEST(DereferenceFormatter, FormatterAPI) {
>+  {
>+    // Tests wrapping the default AlphaNumFormatter.
>+    const absl::strings_internal::DereferenceFormatterImpl<
>+        absl::strings_internal::AlphaNumFormatterImpl>
>+        f;
>+    int x = 1, y = 2, z = 3;
>+    std::string s;
>+    f(&s, &x);
>+    f(&s, &y);
>+    f(&s, &z);
>+    EXPECT_EQ("123", s);
>+  }
>+
>+  {
>+    // Tests wrapping std::string's default formatter.
>+    absl::strings_internal::DereferenceFormatterImpl<
>+        absl::strings_internal::DefaultFormatter<std::string>::Type>
>+        f;
>+
>+    std::string x = "x";
>+    std::string y = "y";
>+    std::string z = "z";
>+    std::string s;
>+    f(&s, &x);
>+    f(&s, &y);
>+    f(&s, &z);
>+    EXPECT_EQ(s, "xyz");
>+  }
>+
>+  {
>+    // Tests wrapping a custom formatter.
>+    auto f = absl::DereferenceFormatter(TestingParenFormatter());
>+    int x = 1, y = 2, z = 3;
>+    std::string s;
>+    f(&s, &x);
>+    f(&s, &y);
>+    f(&s, &z);
>+    EXPECT_EQ("(1)(2)(3)", s);
>+  }
>+
>+  {
>+    absl::strings_internal::DereferenceFormatterImpl<
>+        absl::strings_internal::AlphaNumFormatterImpl>
>+        f;
>+    auto x = std::unique_ptr<int>(new int(1));
>+    auto y = std::unique_ptr<int>(new int(2));
>+    auto z = std::unique_ptr<int>(new int(3));
>+    std::string s;
>+    f(&s, x);
>+    f(&s, y);
>+    f(&s, z);
>+    EXPECT_EQ("123", s);
>+  }
>+}
>+
>+//
>+// Tests the interfaces for the 4 public Join function overloads. The semantics
>+// of the algorithm is covered in the above APIExamples test.
>+//
>+TEST(StrJoin, PublicAPIOverloads) {
>+  std::vector<std::string> v = {"a", "b", "c"};
>+
>+  // Iterators + formatter
>+  EXPECT_EQ("a-b-c",
>+            absl::StrJoin(v.begin(), v.end(), "-", absl::AlphaNumFormatter()));
>+  // Range + formatter
>+  EXPECT_EQ("a-b-c", absl::StrJoin(v, "-", absl::AlphaNumFormatter()));
>+  // Iterators, no formatter
>+  EXPECT_EQ("a-b-c", absl::StrJoin(v.begin(), v.end(), "-"));
>+  // Range, no formatter
>+  EXPECT_EQ("a-b-c", absl::StrJoin(v, "-"));
>+}
>+
>+TEST(StrJoin, Array) {
>+  const absl::string_view a[] = {"a", "b", "c"};
>+  EXPECT_EQ("a-b-c", absl::StrJoin(a, "-"));
>+}
>+
>+TEST(StrJoin, InitializerList) {
>+  { EXPECT_EQ("a-b-c", absl::StrJoin({"a", "b", "c"}, "-")); }
>+
>+  {
>+    auto a = {"a", "b", "c"};
>+    EXPECT_EQ("a-b-c", absl::StrJoin(a, "-"));
>+  }
>+
>+  {
>+    std::initializer_list<const char*> a = {"a", "b", "c"};
>+    EXPECT_EQ("a-b-c", absl::StrJoin(a, "-"));
>+  }
>+
>+  {
>+    std::initializer_list<std::string> a = {"a", "b", "c"};
>+    EXPECT_EQ("a-b-c", absl::StrJoin(a, "-"));
>+  }
>+
>+  {
>+    std::initializer_list<absl::string_view> a = {"a", "b", "c"};
>+    EXPECT_EQ("a-b-c", absl::StrJoin(a, "-"));
>+  }
>+
>+  {
>+    // Tests initializer_list with a non-default formatter
>+    auto a = {"a", "b", "c"};
>+    TestingParenFormatter f;
>+    EXPECT_EQ("(a)-(b)-(c)", absl::StrJoin(a, "-", f));
>+  }
>+
>+  {
>+    // initializer_list of ints
>+    EXPECT_EQ("1-2-3", absl::StrJoin({1, 2, 3}, "-"));
>+  }
>+
>+  {
>+    // Tests initializer_list of ints with a non-default formatter
>+    auto a = {1, 2, 3};
>+    TestingParenFormatter f;
>+    EXPECT_EQ("(1)-(2)-(3)", absl::StrJoin(a, "-", f));
>+  }
>+}
>+
>+TEST(StrJoin, Tuple) {
>+  EXPECT_EQ("", absl::StrJoin(std::make_tuple(), "-"));
>+  EXPECT_EQ("hello", absl::StrJoin(std::make_tuple("hello"), "-"));
>+
>+  int x(10);
>+  std::string y("hello");
>+  double z(3.14);
>+  EXPECT_EQ("10-hello-3.14", absl::StrJoin(std::make_tuple(x, y, z), "-"));
>+
>+  // Faster! Faster!!
>+  EXPECT_EQ("10-hello-3.14",
>+            absl::StrJoin(std::make_tuple(x, std::cref(y), z), "-"));
>+
>+  struct TestFormatter {
>+    char buffer[128];
>+    void operator()(std::string* out, int v) {
>+      snprintf(buffer, sizeof(buffer), "%#.8x", v);
>+      out->append(buffer);
>+    }
>+    void operator()(std::string* out, double v) {
>+      snprintf(buffer, sizeof(buffer), "%#.0f", v);
>+      out->append(buffer);
>+    }
>+    void operator()(std::string* out, const std::string& v) {
>+      snprintf(buffer, sizeof(buffer), "%.4s", v.c_str());
>+      out->append(buffer);
>+    }
>+  };
>+  EXPECT_EQ("0x0000000a-hell-3.",
>+            absl::StrJoin(std::make_tuple(x, y, z), "-", TestFormatter()));
>+  EXPECT_EQ(
>+      "0x0000000a-hell-3.",
>+      absl::StrJoin(std::make_tuple(x, std::cref(y), z), "-", TestFormatter()));
>+  EXPECT_EQ("0x0000000a-hell-3.",
>+            absl::StrJoin(std::make_tuple(&x, &y, &z), "-",
>+                          absl::DereferenceFormatter(TestFormatter())));
>+  EXPECT_EQ("0x0000000a-hell-3.",
>+            absl::StrJoin(std::make_tuple(absl::make_unique<int>(x),
>+                                          absl::make_unique<std::string>(y),
>+                                          absl::make_unique<double>(z)),
>+                          "-", absl::DereferenceFormatter(TestFormatter())));
>+  EXPECT_EQ("0x0000000a-hell-3.",
>+            absl::StrJoin(std::make_tuple(absl::make_unique<int>(x), &y, &z),
>+                          "-", absl::DereferenceFormatter(TestFormatter())));
>+}
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_replace.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_replace.cc
>new file mode 100644
>index 00000000000..69efa357138
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_replace.cc
>@@ -0,0 +1,79 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/strings/str_replace.h"
>+
>+#include "absl/strings/str_cat.h"
>+
>+namespace absl {
>+namespace strings_internal {
>+
>+using FixedMapping =
>+    std::initializer_list<std::pair<absl::string_view, absl::string_view>>;
>+
>+// Applies the ViableSubstitutions in subs_ptr to the absl::string_view s, and
>+// stores the result in *result_ptr. Returns the number of substitutions that
>+// occurred.
>+int ApplySubstitutions(
>+    absl::string_view s,
>+    std::vector<strings_internal::ViableSubstitution>* subs_ptr,
>+    std::string* result_ptr) {
>+  auto& subs = *subs_ptr;
>+  int substitutions = 0;
>+  size_t pos = 0;
>+  while (!subs.empty()) {
>+    auto& sub = subs.back();
>+    if (sub.offset >= pos) {
>+      if (pos <= s.size()) {
>+        StrAppend(result_ptr, s.substr(pos, sub.offset - pos), sub.replacement);
>+      }
>+      pos = sub.offset + sub.old.size();
>+      substitutions += 1;
>+    }
>+    sub.offset = s.find(sub.old, pos);
>+    if (sub.offset == s.npos) {
>+      subs.pop_back();
>+    } else {
>+      // Insertion sort to ensure the last ViableSubstitution continues to be
>+      // before all the others.
>+      size_t index = subs.size();
>+      while (--index && subs[index - 1].OccursBefore(subs[index])) {
>+        std::swap(subs[index], subs[index - 1]);
>+      }
>+    }
>+  }
>+  result_ptr->append(s.data() + pos, s.size() - pos);
>+  return substitutions;
>+}
>+
>+}  // namespace strings_internal
>+
>+// We can implement this in terms of the generic StrReplaceAll, but
>+// we must specify the template overload because C++ cannot deduce the type
>+// of an initializer_list parameter to a function, and also if we don't specify
>+// the type, we just call ourselves.
>+//
>+// Note that we implement them here, rather than in the header, so that they
>+// aren't inlined.
>+
>+std::string StrReplaceAll(absl::string_view s,
>+                     strings_internal::FixedMapping replacements) {
>+  return StrReplaceAll<strings_internal::FixedMapping>(s, replacements);
>+}
>+
>+int StrReplaceAll(strings_internal::FixedMapping replacements, std::string* target) {
>+  return StrReplaceAll<strings_internal::FixedMapping>(replacements, target);
>+}
>+
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_replace.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_replace.h
>new file mode 100644
>index 00000000000..f4d9bb9545d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_replace.h
>@@ -0,0 +1,213 @@
>+//
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// -----------------------------------------------------------------------------
>+// File: str_replace.h
>+// -----------------------------------------------------------------------------
>+//
>+// This file defines `absl::StrReplaceAll()`, a general-purpose std::string
>+// replacement function designed for large, arbitrary text substitutions,
>+// especially on strings which you are receiving from some other system for
>+// further processing (e.g. processing regular expressions, escaping HTML
>+// entities, etc. `StrReplaceAll` is designed to be efficient even when only
>+// one substitution is being performed, or when substitution is rare.
>+//
>+// If the std::string being modified is known at compile-time, and the substitutions
>+// vary, `absl::Substitute()` may be a better choice.
>+//
>+// Example:
>+//
>+// std::string html_escaped = absl::StrReplaceAll(user_input, {
>+//                                           {"&", "&amp;"},
>+//                                           {"<", "&lt;"},
>+//                                           {">", "&gt;"},
>+//                                           {"\"", "&quot;"},
>+//                                           {"'", "&#39;"}});
>+#ifndef ABSL_STRINGS_STR_REPLACE_H_
>+#define ABSL_STRINGS_STR_REPLACE_H_
>+
>+#include <string>
>+#include <utility>
>+#include <vector>
>+
>+#include "absl/base/attributes.h"
>+#include "absl/strings/string_view.h"
>+
>+namespace absl {
>+
>+// StrReplaceAll()
>+//
>+// Replaces character sequences within a given std::string with replacements provided
>+// within an initializer list of key/value pairs. Candidate replacements are
>+// considered in order as they occur within the std::string, with earlier matches
>+// taking precedence, and longer matches taking precedence for candidates
>+// starting at the same position in the std::string. Once a substitution is made, the
>+// replaced text is not considered for any further substitutions.
>+//
>+// Example:
>+//
>+//   std::string s = absl::StrReplaceAll("$who bought $count #Noun. Thanks $who!",
>+//                                  {{"$count", absl::StrCat(5)},
>+//                                   {"$who", "Bob"},
>+//                                   {"#Noun", "Apples"}});
>+//   EXPECT_EQ("Bob bought 5 Apples. Thanks Bob!", s);
>+ABSL_MUST_USE_RESULT std::string StrReplaceAll(
>+    absl::string_view s,
>+    std::initializer_list<std::pair<absl::string_view, absl::string_view>>
>+        replacements);
>+
>+// Overload of `StrReplaceAll()` to accept a container of key/value replacement
>+// pairs (typically either an associative map or a `std::vector` of `std::pair`
>+// elements). A vector of pairs is generally more efficient.
>+//
>+// Examples:
>+//
>+//   std::map<const absl::string_view, const absl::string_view> replacements;
>+//   replacements["$who"] = "Bob";
>+//   replacements["$count"] = "5";
>+//   replacements["#Noun"] = "Apples";
>+//   std::string s = absl::StrReplaceAll("$who bought $count #Noun. Thanks $who!",
>+//                                  replacements);
>+//   EXPECT_EQ("Bob bought 5 Apples. Thanks Bob!", s);
>+//
>+//   // A std::vector of std::pair elements can be more efficient.
>+//   std::vector<std::pair<const absl::string_view, std::string>> replacements;
>+//   replacements.push_back({"&", "&amp;"});
>+//   replacements.push_back({"<", "&lt;"});
>+//   replacements.push_back({">", "&gt;"});
>+//   std::string s = absl::StrReplaceAll("if (ptr < &foo)",
>+//                                  replacements);
>+//   EXPECT_EQ("if (ptr &lt; &amp;foo)", s);
>+template <typename StrToStrMapping>
>+std::string StrReplaceAll(absl::string_view s, const StrToStrMapping& replacements);
>+
>+// Overload of `StrReplaceAll()` to replace character sequences within a given
>+// output std::string *in place* with replacements provided within an initializer
>+// list of key/value pairs, returning the number of substitutions that occurred.
>+//
>+// Example:
>+//
>+//   std::string s = std::string("$who bought $count #Noun. Thanks $who!");
>+//   int count;
>+//   count = absl::StrReplaceAll({{"$count", absl::StrCat(5)},
>+//                               {"$who", "Bob"},
>+//                               {"#Noun", "Apples"}}, &s);
>+//  EXPECT_EQ(count, 4);
>+//  EXPECT_EQ("Bob bought 5 Apples. Thanks Bob!", s);
>+int StrReplaceAll(
>+    std::initializer_list<std::pair<absl::string_view, absl::string_view>>
>+        replacements,
>+    std::string* target);
>+
>+// Overload of `StrReplaceAll()` to replace patterns within a given output
>+// std::string *in place* with replacements provided within a container of key/value
>+// pairs.
>+//
>+// Example:
>+//
>+//   std::string s = std::string("if (ptr < &foo)");
>+//   int count = absl::StrReplaceAll({{"&", "&amp;"},
>+//                                    {"<", "&lt;"},
>+//                                    {">", "&gt;"}}, &s);
>+//  EXPECT_EQ(count, 2);
>+//  EXPECT_EQ("if (ptr &lt; &amp;foo)", s);
>+template <typename StrToStrMapping>
>+int StrReplaceAll(const StrToStrMapping& replacements, std::string* target);
>+
>+// Implementation details only, past this point.
>+namespace strings_internal {
>+
>+struct ViableSubstitution {
>+  absl::string_view old;
>+  absl::string_view replacement;
>+  size_t offset;
>+
>+  ViableSubstitution(absl::string_view old_str,
>+                     absl::string_view replacement_str, size_t offset_val)
>+      : old(old_str), replacement(replacement_str), offset(offset_val) {}
>+
>+  // One substitution occurs "before" another (takes priority) if either
>+  // it has the lowest offset, or it has the same offset but a larger size.
>+  bool OccursBefore(const ViableSubstitution& y) const {
>+    if (offset != y.offset) return offset < y.offset;
>+    return old.size() > y.old.size();
>+  }
>+};
>+
>+// Build a vector of ViableSubstitutions based on the given list of
>+// replacements. subs can be implemented as a priority_queue. However, it turns
>+// out that most callers have small enough a list of substitutions that the
>+// overhead of such a queue isn't worth it.
>+template <typename StrToStrMapping>
>+std::vector<ViableSubstitution> FindSubstitutions(
>+    absl::string_view s, const StrToStrMapping& replacements) {
>+  std::vector<ViableSubstitution> subs;
>+  subs.reserve(replacements.size());
>+
>+  for (const auto& rep : replacements) {
>+    using std::get;
>+    absl::string_view old(get<0>(rep));
>+
>+    size_t pos = s.find(old);
>+    if (pos == s.npos) continue;
>+
>+    // Ignore attempts to replace "". This condition is almost never true,
>+    // but above condition is frequently true. That's why we test for this
>+    // now and not before.
>+    if (old.empty()) continue;
>+
>+    subs.emplace_back(old, get<1>(rep), pos);
>+
>+    // Insertion sort to ensure the last ViableSubstitution comes before
>+    // all the others.
>+    size_t index = subs.size();
>+    while (--index && subs[index - 1].OccursBefore(subs[index])) {
>+      std::swap(subs[index], subs[index - 1]);
>+    }
>+  }
>+  return subs;
>+}
>+
>+int ApplySubstitutions(absl::string_view s,
>+                       std::vector<ViableSubstitution>* subs_ptr,
>+                       std::string* result_ptr);
>+
>+}  // namespace strings_internal
>+
>+template <typename StrToStrMapping>
>+std::string StrReplaceAll(absl::string_view s, const StrToStrMapping& replacements) {
>+  auto subs = strings_internal::FindSubstitutions(s, replacements);
>+  std::string result;
>+  result.reserve(s.size());
>+  strings_internal::ApplySubstitutions(s, &subs, &result);
>+  return result;
>+}
>+
>+template <typename StrToStrMapping>
>+int StrReplaceAll(const StrToStrMapping& replacements, std::string* target) {
>+  auto subs = strings_internal::FindSubstitutions(*target, replacements);
>+  if (subs.empty()) return 0;
>+
>+  std::string result;
>+  result.reserve(target->size());
>+  int substitutions =
>+      strings_internal::ApplySubstitutions(*target, &subs, &result);
>+  target->swap(result);
>+  return substitutions;
>+}
>+
>+}  // namespace absl
>+
>+#endif  // ABSL_STRINGS_STR_REPLACE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_replace_benchmark.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_replace_benchmark.cc
>new file mode 100644
>index 00000000000..e608de8d19e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_replace_benchmark.cc
>@@ -0,0 +1,122 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/strings/str_replace.h"
>+
>+#include <cstring>
>+#include <string>
>+
>+#include "benchmark/benchmark.h"
>+#include "absl/base/internal/raw_logging.h"
>+
>+namespace {
>+
>+std::string* big_string;
>+std::string* after_replacing_the;
>+std::string* after_replacing_many;
>+
>+struct Replacement {
>+  const char* needle;
>+  const char* replacement;
>+} replacements[] = {
>+    {"the", "box"},          //
>+    {"brown", "quick"},      //
>+    {"jumped", "liquored"},  //
>+    {"dozen", "brown"},      //
>+    {"lazy", "pack"},        //
>+    {"liquor", "shakes"},    //
>+};
>+
>+// Here, we set up a std::string for use in global-replace benchmarks.
>+// We started with a million blanks, and then deterministically insert
>+// 10,000 copies each of two pangrams.  The result is a std::string that is
>+// 40% blank space and 60% these words.  'the' occurs 18,247 times and
>+// all the substitutions together occur 49,004 times.
>+//
>+// We then create "after_replacing_the" to be a std::string that is a result of
>+// replacing "the" with "box" in big_string.
>+//
>+// And then we create "after_replacing_many" to be a std::string that is result
>+// of preferring several substitutions.
>+void SetUpStrings() {
>+  if (big_string == nullptr) {
>+    size_t r = 0;
>+    big_string = new std::string(1000 * 1000, ' ');
>+    for (std::string phrase : {"the quick brown fox jumped over the lazy dogs",
>+                          "pack my box with the five dozen liquor jugs"}) {
>+      for (int i = 0; i < 10 * 1000; ++i) {
>+        r = r * 237 + 41;  // not very random.
>+        memcpy(&(*big_string)[r % (big_string->size() - phrase.size())],
>+               phrase.data(), phrase.size());
>+      }
>+    }
>+    // big_string->resize(50);
>+    // OK, we've set up the std::string, now let's set up expectations - first by
>+    // just replacing "the" with "box"
>+    after_replacing_the = new std::string(*big_string);
>+    for (size_t pos = 0;
>+         (pos = after_replacing_the->find("the", pos)) != std::string::npos;) {
>+      memcpy(&(*after_replacing_the)[pos], "box", 3);
>+    }
>+    // And then with all the replacements.
>+    after_replacing_many = new std::string(*big_string);
>+    for (size_t pos = 0;;) {
>+      size_t next_pos = static_cast<size_t>(-1);
>+      const char* needle_string = nullptr;
>+      const char* replacement_string = nullptr;
>+      for (const auto& r : replacements) {
>+        auto needlepos = after_replacing_many->find(r.needle, pos);
>+        if (needlepos != std::string::npos && needlepos < next_pos) {
>+          next_pos = needlepos;
>+          needle_string = r.needle;
>+          replacement_string = r.replacement;
>+        }
>+      }
>+      if (next_pos > after_replacing_many->size()) break;
>+      after_replacing_many->replace(next_pos, strlen(needle_string),
>+                                    replacement_string);
>+      next_pos += strlen(replacement_string);
>+      pos = next_pos;
>+    }
>+  }
>+}
>+
>+void BM_StrReplaceAllOneReplacement(benchmark::State& state) {
>+  SetUpStrings();
>+  std::string src = *big_string;
>+  for (auto _ : state) {
>+    std::string dest = absl::StrReplaceAll(src, {{"the", "box"}});
>+    ABSL_RAW_CHECK(dest == *after_replacing_the,
>+                   "not benchmarking intended behavior");
>+  }
>+}
>+BENCHMARK(BM_StrReplaceAllOneReplacement);
>+
>+void BM_StrReplaceAll(benchmark::State& state) {
>+  SetUpStrings();
>+  std::string src = *big_string;
>+  for (auto _ : state) {
>+    std::string dest = absl::StrReplaceAll(src, {{"the", "box"},
>+                                            {"brown", "quick"},
>+                                            {"jumped", "liquored"},
>+                                            {"dozen", "brown"},
>+                                            {"lazy", "pack"},
>+                                            {"liquor", "shakes"}});
>+    ABSL_RAW_CHECK(dest == *after_replacing_many,
>+                   "not benchmarking intended behavior");
>+  }
>+}
>+BENCHMARK(BM_StrReplaceAll);
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_replace_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_replace_test.cc
>new file mode 100644
>index 00000000000..5d003a224a4
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_replace_test.cc
>@@ -0,0 +1,341 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/strings/str_replace.h"
>+
>+#include <list>
>+#include <map>
>+#include <tuple>
>+
>+#include "gtest/gtest.h"
>+#include "absl/strings/str_cat.h"
>+#include "absl/strings/str_split.h"
>+
>+TEST(StrReplaceAll, OneReplacement) {
>+  std::string s;
>+
>+  // Empty std::string.
>+  s = absl::StrReplaceAll(s, {{"", ""}});
>+  EXPECT_EQ(s, "");
>+  s = absl::StrReplaceAll(s, {{"x", ""}});
>+  EXPECT_EQ(s, "");
>+  s = absl::StrReplaceAll(s, {{"", "y"}});
>+  EXPECT_EQ(s, "");
>+  s = absl::StrReplaceAll(s, {{"x", "y"}});
>+  EXPECT_EQ(s, "");
>+
>+  // Empty substring.
>+  s = absl::StrReplaceAll("abc", {{"", ""}});
>+  EXPECT_EQ(s, "abc");
>+  s = absl::StrReplaceAll("abc", {{"", "y"}});
>+  EXPECT_EQ(s, "abc");
>+  s = absl::StrReplaceAll("abc", {{"x", ""}});
>+  EXPECT_EQ(s, "abc");
>+
>+  // Substring not found.
>+  s = absl::StrReplaceAll("abc", {{"xyz", "123"}});
>+  EXPECT_EQ(s, "abc");
>+
>+  // Replace entire std::string.
>+  s = absl::StrReplaceAll("abc", {{"abc", "xyz"}});
>+  EXPECT_EQ(s, "xyz");
>+
>+  // Replace once at the start.
>+  s = absl::StrReplaceAll("abc", {{"a", "x"}});
>+  EXPECT_EQ(s, "xbc");
>+
>+  // Replace once in the middle.
>+  s = absl::StrReplaceAll("abc", {{"b", "x"}});
>+  EXPECT_EQ(s, "axc");
>+
>+  // Replace once at the end.
>+  s = absl::StrReplaceAll("abc", {{"c", "x"}});
>+  EXPECT_EQ(s, "abx");
>+
>+  // Replace multiple times with varying lengths of original/replacement.
>+  s = absl::StrReplaceAll("ababa", {{"a", "xxx"}});
>+  EXPECT_EQ(s, "xxxbxxxbxxx");
>+
>+  s = absl::StrReplaceAll("ababa", {{"b", "xxx"}});
>+  EXPECT_EQ(s, "axxxaxxxa");
>+
>+  s = absl::StrReplaceAll("aaabaaabaaa", {{"aaa", "x"}});
>+  EXPECT_EQ(s, "xbxbx");
>+
>+  s = absl::StrReplaceAll("abbbabbba", {{"bbb", "x"}});
>+  EXPECT_EQ(s, "axaxa");
>+
>+  // Overlapping matches are replaced greedily.
>+  s = absl::StrReplaceAll("aaa", {{"aa", "x"}});
>+  EXPECT_EQ(s, "xa");
>+
>+  // The replacements are not recursive.
>+  s = absl::StrReplaceAll("aaa", {{"aa", "a"}});
>+  EXPECT_EQ(s, "aa");
>+}
>+
>+TEST(StrReplaceAll, ManyReplacements) {
>+  std::string s;
>+
>+  // Empty std::string.
>+  s = absl::StrReplaceAll("", {{"", ""}, {"x", ""}, {"", "y"}, {"x", "y"}});
>+  EXPECT_EQ(s, "");
>+
>+  // Empty substring.
>+  s = absl::StrReplaceAll("abc", {{"", ""}, {"", "y"}, {"x", ""}});
>+  EXPECT_EQ(s, "abc");
>+
>+  // Replace entire std::string, one char at a time
>+  s = absl::StrReplaceAll("abc", {{"a", "x"}, {"b", "y"}, {"c", "z"}});
>+  EXPECT_EQ(s, "xyz");
>+  s = absl::StrReplaceAll("zxy", {{"z", "x"}, {"x", "y"}, {"y", "z"}});
>+  EXPECT_EQ(s, "xyz");
>+
>+  // Replace once at the start (longer matches take precedence)
>+  s = absl::StrReplaceAll("abc", {{"a", "x"}, {"ab", "xy"}, {"abc", "xyz"}});
>+  EXPECT_EQ(s, "xyz");
>+
>+  // Replace once in the middle.
>+  s = absl::StrReplaceAll(
>+      "Abc!", {{"a", "x"}, {"ab", "xy"}, {"b", "y"}, {"bc", "yz"}, {"c", "z"}});
>+  EXPECT_EQ(s, "Ayz!");
>+
>+  // Replace once at the end.
>+  s = absl::StrReplaceAll(
>+      "Abc!",
>+      {{"a", "x"}, {"ab", "xy"}, {"b", "y"}, {"bc!", "yz?"}, {"c!", "z;"}});
>+  EXPECT_EQ(s, "Ayz?");
>+
>+  // Replace multiple times with varying lengths of original/replacement.
>+  s = absl::StrReplaceAll("ababa", {{"a", "xxx"}, {"b", "XXXX"}});
>+  EXPECT_EQ(s, "xxxXXXXxxxXXXXxxx");
>+
>+  // Overlapping matches are replaced greedily.
>+  s = absl::StrReplaceAll("aaa", {{"aa", "x"}, {"a", "X"}});
>+  EXPECT_EQ(s, "xX");
>+  s = absl::StrReplaceAll("aaa", {{"a", "X"}, {"aa", "x"}});
>+  EXPECT_EQ(s, "xX");
>+
>+  // Two well-known sentences
>+  s = absl::StrReplaceAll("the quick brown fox jumped over the lazy dogs",
>+                          {
>+                              {"brown", "box"},
>+                              {"dogs", "jugs"},
>+                              {"fox", "with"},
>+                              {"jumped", "five"},
>+                              {"over", "dozen"},
>+                              {"quick", "my"},
>+                              {"the", "pack"},
>+                              {"the lazy", "liquor"},
>+                          });
>+  EXPECT_EQ(s, "pack my box with five dozen liquor jugs");
>+}
>+
>+TEST(StrReplaceAll, ManyReplacementsInMap) {
>+  std::map<const char *, const char *> replacements;
>+  replacements["$who"] = "Bob";
>+  replacements["$count"] = "5";
>+  replacements["#Noun"] = "Apples";
>+  std::string s = absl::StrReplaceAll("$who bought $count #Noun. Thanks $who!",
>+                                 replacements);
>+  EXPECT_EQ("Bob bought 5 Apples. Thanks Bob!", s);
>+}
>+
>+TEST(StrReplaceAll, ReplacementsInPlace) {
>+  std::string s = std::string("$who bought $count #Noun. Thanks $who!");
>+  int count;
>+  count = absl::StrReplaceAll({{"$count", absl::StrCat(5)},
>+                              {"$who", "Bob"},
>+                              {"#Noun", "Apples"}}, &s);
>+  EXPECT_EQ(count, 4);
>+  EXPECT_EQ("Bob bought 5 Apples. Thanks Bob!", s);
>+}
>+
>+TEST(StrReplaceAll, ReplacementsInPlaceInMap) {
>+  std::string s = std::string("$who bought $count #Noun. Thanks $who!");
>+  std::map<absl::string_view, absl::string_view> replacements;
>+  replacements["$who"] = "Bob";
>+  replacements["$count"] = "5";
>+  replacements["#Noun"] = "Apples";
>+  int count;
>+  count = absl::StrReplaceAll(replacements, &s);
>+  EXPECT_EQ(count, 4);
>+  EXPECT_EQ("Bob bought 5 Apples. Thanks Bob!", s);
>+}
>+
>+struct Cont {
>+  Cont() {}
>+  explicit Cont(absl::string_view src) : data(src) {}
>+
>+  absl::string_view data;
>+};
>+
>+template <int index>
>+absl::string_view get(const Cont& c) {
>+  auto splitter = absl::StrSplit(c.data, ':');
>+  auto it = splitter.begin();
>+  for (int i = 0; i < index; ++i) ++it;
>+
>+  return *it;
>+}
>+
>+TEST(StrReplaceAll, VariableNumber) {
>+  std::string s;
>+  {
>+    std::vector<std::pair<std::string, std::string>> replacements;
>+
>+    s = "abc";
>+    EXPECT_EQ(0, absl::StrReplaceAll(replacements, &s));
>+    EXPECT_EQ("abc", s);
>+
>+    s = "abc";
>+    replacements.push_back({"a", "A"});
>+    EXPECT_EQ(1, absl::StrReplaceAll(replacements, &s));
>+    EXPECT_EQ("Abc", s);
>+
>+    s = "abc";
>+    replacements.push_back({"b", "B"});
>+    EXPECT_EQ(2, absl::StrReplaceAll(replacements, &s));
>+    EXPECT_EQ("ABc", s);
>+
>+    s = "abc";
>+    replacements.push_back({"d", "D"});
>+    EXPECT_EQ(2, absl::StrReplaceAll(replacements, &s));
>+    EXPECT_EQ("ABc", s);
>+
>+    EXPECT_EQ("ABcABc", absl::StrReplaceAll("abcabc", replacements));
>+  }
>+
>+  {
>+    std::map<const char*, const char*> replacements;
>+    replacements["aa"] = "x";
>+    replacements["a"] = "X";
>+    s = "aaa";
>+    EXPECT_EQ(2, absl::StrReplaceAll(replacements, &s));
>+    EXPECT_EQ("xX", s);
>+
>+    EXPECT_EQ("xxX", absl::StrReplaceAll("aaaaa", replacements));
>+  }
>+
>+  {
>+    std::list<std::pair<absl::string_view, absl::string_view>> replacements = {
>+        {"a", "x"}, {"b", "y"}, {"c", "z"}};
>+
>+    std::string s = absl::StrReplaceAll("abc", replacements);
>+    EXPECT_EQ(s, "xyz");
>+  }
>+
>+  {
>+    using X = std::tuple<absl::string_view, std::string, int>;
>+    std::vector<X> replacements(3);
>+    replacements[0] = X{"a", "x", 1};
>+    replacements[1] = X{"b", "y", 0};
>+    replacements[2] = X{"c", "z", -1};
>+
>+    std::string s = absl::StrReplaceAll("abc", replacements);
>+    EXPECT_EQ(s, "xyz");
>+  }
>+
>+  {
>+    std::vector<Cont> replacements(3);
>+    replacements[0] = Cont{"a:x"};
>+    replacements[1] = Cont{"b:y"};
>+    replacements[2] = Cont{"c:z"};
>+
>+    std::string s = absl::StrReplaceAll("abc", replacements);
>+    EXPECT_EQ(s, "xyz");
>+  }
>+}
>+
>+// Same as above, but using the in-place variant of absl::StrReplaceAll,
>+// that returns the # of replacements performed.
>+TEST(StrReplaceAll, Inplace) {
>+  std::string s;
>+  int reps;
>+
>+  // Empty std::string.
>+  s = "";
>+  reps = absl::StrReplaceAll({{"", ""}, {"x", ""}, {"", "y"}, {"x", "y"}}, &s);
>+  EXPECT_EQ(reps, 0);
>+  EXPECT_EQ(s, "");
>+
>+  // Empty substring.
>+  s = "abc";
>+  reps = absl::StrReplaceAll({{"", ""}, {"", "y"}, {"x", ""}}, &s);
>+  EXPECT_EQ(reps, 0);
>+  EXPECT_EQ(s, "abc");
>+
>+  // Replace entire std::string, one char at a time
>+  s = "abc";
>+  reps = absl::StrReplaceAll({{"a", "x"}, {"b", "y"}, {"c", "z"}}, &s);
>+  EXPECT_EQ(reps, 3);
>+  EXPECT_EQ(s, "xyz");
>+  s = "zxy";
>+  reps = absl::StrReplaceAll({{"z", "x"}, {"x", "y"}, {"y", "z"}}, &s);
>+  EXPECT_EQ(reps, 3);
>+  EXPECT_EQ(s, "xyz");
>+
>+  // Replace once at the start (longer matches take precedence)
>+  s = "abc";
>+  reps = absl::StrReplaceAll({{"a", "x"}, {"ab", "xy"}, {"abc", "xyz"}}, &s);
>+  EXPECT_EQ(reps, 1);
>+  EXPECT_EQ(s, "xyz");
>+
>+  // Replace once in the middle.
>+  s = "Abc!";
>+  reps = absl::StrReplaceAll(
>+      {{"a", "x"}, {"ab", "xy"}, {"b", "y"}, {"bc", "yz"}, {"c", "z"}}, &s);
>+  EXPECT_EQ(reps, 1);
>+  EXPECT_EQ(s, "Ayz!");
>+
>+  // Replace once at the end.
>+  s = "Abc!";
>+  reps = absl::StrReplaceAll(
>+      {{"a", "x"}, {"ab", "xy"}, {"b", "y"}, {"bc!", "yz?"}, {"c!", "z;"}}, &s);
>+  EXPECT_EQ(reps, 1);
>+  EXPECT_EQ(s, "Ayz?");
>+
>+  // Replace multiple times with varying lengths of original/replacement.
>+  s = "ababa";
>+  reps = absl::StrReplaceAll({{"a", "xxx"}, {"b", "XXXX"}}, &s);
>+  EXPECT_EQ(reps, 5);
>+  EXPECT_EQ(s, "xxxXXXXxxxXXXXxxx");
>+
>+  // Overlapping matches are replaced greedily.
>+  s = "aaa";
>+  reps = absl::StrReplaceAll({{"aa", "x"}, {"a", "X"}}, &s);
>+  EXPECT_EQ(reps, 2);
>+  EXPECT_EQ(s, "xX");
>+  s = "aaa";
>+  reps = absl::StrReplaceAll({{"a", "X"}, {"aa", "x"}}, &s);
>+  EXPECT_EQ(reps, 2);
>+  EXPECT_EQ(s, "xX");
>+
>+  // Two well-known sentences
>+  s = "the quick brown fox jumped over the lazy dogs";
>+  reps = absl::StrReplaceAll(
>+      {
>+          {"brown", "box"},
>+          {"dogs", "jugs"},
>+          {"fox", "with"},
>+          {"jumped", "five"},
>+          {"over", "dozen"},
>+          {"quick", "my"},
>+          {"the", "pack"},
>+          {"the lazy", "liquor"},
>+      },
>+      &s);
>+  EXPECT_EQ(reps, 8);
>+  EXPECT_EQ(s, "pack my box with five dozen liquor jugs");
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_split.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_split.cc
>new file mode 100644
>index 00000000000..0207213c26b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_split.cc
>@@ -0,0 +1,136 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/strings/str_split.h"
>+
>+#include <algorithm>
>+#include <cassert>
>+#include <cstdint>
>+#include <cstdlib>
>+#include <cstring>
>+#include <iterator>
>+#include <limits>
>+#include <memory>
>+
>+#include "absl/base/internal/raw_logging.h"
>+#include "absl/strings/ascii.h"
>+
>+namespace absl {
>+
>+namespace {
>+
>+// This GenericFind() template function encapsulates the finding algorithm
>+// shared between the ByString and ByAnyChar delimiters. The FindPolicy
>+// template parameter allows each delimiter to customize the actual find
>+// function to use and the length of the found delimiter. For example, the
>+// Literal delimiter will ultimately use absl::string_view::find(), and the
>+// AnyOf delimiter will use absl::string_view::find_first_of().
>+template <typename FindPolicy>
>+absl::string_view GenericFind(absl::string_view text,
>+                              absl::string_view delimiter, size_t pos,
>+                              FindPolicy find_policy) {
>+  if (delimiter.empty() && text.length() > 0) {
>+    // Special case for empty std::string delimiters: always return a zero-length
>+    // absl::string_view referring to the item at position 1 past pos.
>+    return absl::string_view(text.begin() + pos + 1, 0);
>+  }
>+  size_t found_pos = absl::string_view::npos;
>+  absl::string_view found(text.end(), 0);  // By default, not found
>+  found_pos = find_policy.Find(text, delimiter, pos);
>+  if (found_pos != absl::string_view::npos) {
>+    found = absl::string_view(text.data() + found_pos,
>+                              find_policy.Length(delimiter));
>+  }
>+  return found;
>+}
>+
>+// Finds using absl::string_view::find(), therefore the length of the found
>+// delimiter is delimiter.length().
>+struct LiteralPolicy {
>+  size_t Find(absl::string_view text, absl::string_view delimiter, size_t pos) {
>+    return text.find(delimiter, pos);
>+  }
>+  size_t Length(absl::string_view delimiter) { return delimiter.length(); }
>+};
>+
>+// Finds using absl::string_view::find_first_of(), therefore the length of the
>+// found delimiter is 1.
>+struct AnyOfPolicy {
>+  size_t Find(absl::string_view text, absl::string_view delimiter, size_t pos) {
>+    return text.find_first_of(delimiter, pos);
>+  }
>+  size_t Length(absl::string_view /* delimiter */) { return 1; }
>+};
>+
>+}  // namespace
>+
>+//
>+// ByString
>+//
>+
>+ByString::ByString(absl::string_view sp) : delimiter_(sp) {}
>+
>+absl::string_view ByString::Find(absl::string_view text, size_t pos) const {
>+  if (delimiter_.length() == 1) {
>+    // Much faster to call find on a single character than on an
>+    // absl::string_view.
>+    size_t found_pos = text.find(delimiter_[0], pos);
>+    if (found_pos == absl::string_view::npos)
>+      return absl::string_view(text.end(), 0);
>+    return text.substr(found_pos, 1);
>+  }
>+  return GenericFind(text, delimiter_, pos, LiteralPolicy());
>+}
>+
>+//
>+// ByChar
>+//
>+
>+absl::string_view ByChar::Find(absl::string_view text, size_t pos) const {
>+  size_t found_pos = text.find(c_, pos);
>+  if (found_pos == absl::string_view::npos)
>+    return absl::string_view(text.end(), 0);
>+  return text.substr(found_pos, 1);
>+}
>+
>+//
>+// ByAnyChar
>+//
>+
>+ByAnyChar::ByAnyChar(absl::string_view sp) : delimiters_(sp) {}
>+
>+absl::string_view ByAnyChar::Find(absl::string_view text, size_t pos) const {
>+  return GenericFind(text, delimiters_, pos, AnyOfPolicy());
>+}
>+
>+//
>+// ByLength
>+//
>+ByLength::ByLength(ptrdiff_t length) : length_(length) {
>+  ABSL_RAW_CHECK(length > 0, "");
>+}
>+
>+absl::string_view ByLength::Find(absl::string_view text,
>+                                      size_t pos) const {
>+  pos = std::min(pos, text.size());  // truncate `pos`
>+  absl::string_view substr = text.substr(pos);
>+  // If the std::string is shorter than the chunk size we say we
>+  // "can't find the delimiter" so this will be the last chunk.
>+  if (substr.length() <= static_cast<size_t>(length_))
>+    return absl::string_view(text.end(), 0);
>+
>+  return absl::string_view(substr.begin() + length_, 0);
>+}
>+
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_split.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_split.h
>new file mode 100644
>index 00000000000..9a7be2b0534
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_split.h
>@@ -0,0 +1,511 @@
>+//
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// -----------------------------------------------------------------------------
>+// File: str_split.h
>+// -----------------------------------------------------------------------------
>+//
>+// This file contains functions for splitting strings. It defines the main
>+// `StrSplit()` function, several delimiters for determining the boundaries on
>+// which to split the std::string, and predicates for filtering delimited results.
>+// `StrSplit()` adapts the returned collection to the type specified by the
>+// caller.
>+//
>+// Example:
>+//
>+//   // Splits the given std::string on commas. Returns the results in a
>+//   // vector of strings.
>+//   std::vector<std::string> v = absl::StrSplit("a,b,c", ',');
>+//   // Can also use ","
>+//   // v[0] == "a", v[1] == "b", v[2] == "c"
>+//
>+// See StrSplit() below for more information.
>+#ifndef ABSL_STRINGS_STR_SPLIT_H_
>+#define ABSL_STRINGS_STR_SPLIT_H_
>+
>+#include <algorithm>
>+#include <cstddef>
>+#include <map>
>+#include <set>
>+#include <string>
>+#include <utility>
>+#include <vector>
>+
>+#include "absl/base/internal/raw_logging.h"
>+#include "absl/strings/internal/str_split_internal.h"
>+#include "absl/strings/string_view.h"
>+#include "absl/strings/strip.h"
>+
>+namespace absl {
>+
>+//------------------------------------------------------------------------------
>+// Delimiters
>+//------------------------------------------------------------------------------
>+//
>+// `StrSplit()` uses delimiters to define the boundaries between elements in the
>+// provided input. Several `Delimiter` types are defined below. If a std::string
>+// (`const char*`, `std::string`, or `absl::string_view`) is passed in place of
>+// an explicit `Delimiter` object, `StrSplit()` treats it the same way as if it
>+// were passed a `ByString` delimiter.
>+//
>+// A `Delimiter` is an object with a `Find()` function that knows how to find
>+// the first occurrence of itself in a given `absl::string_view`.
>+//
>+// The following `Delimiter` types are available for use within `StrSplit()`:
>+//
>+//   - `ByString` (default for std::string arguments)
>+//   - `ByChar` (default for a char argument)
>+//   - `ByAnyChar`
>+//   - `ByLength`
>+//   - `MaxSplits`
>+//
>+//
>+// A Delimiter's Find() member function will be passed the input text that is to
>+// be split and the position to begin searching for the next delimiter in the
>+// input text. The returned absl::string_view should refer to the next
>+// occurrence (after pos) of the represented delimiter; this returned
>+// absl::string_view represents the next location where the input std::string should
>+// be broken. The returned absl::string_view may be zero-length if the Delimiter
>+// does not represent a part of the std::string (e.g., a fixed-length delimiter). If
>+// no delimiter is found in the given text, a zero-length absl::string_view
>+// referring to text.end() should be returned (e.g.,
>+// absl::string_view(text.end(), 0)). It is important that the returned
>+// absl::string_view always be within the bounds of input text given as an
>+// argument--it must not refer to a std::string that is physically located outside of
>+// the given std::string.
>+//
>+// The following example is a simple Delimiter object that is created with a
>+// single char and will look for that char in the text passed to the Find()
>+// function:
>+//
>+//   struct SimpleDelimiter {
>+//     const char c_;
>+//     explicit SimpleDelimiter(char c) : c_(c) {}
>+//     absl::string_view Find(absl::string_view text, size_t pos) {
>+//       auto found = text.find(c_, pos);
>+//       if (found == absl::string_view::npos)
>+//         return absl::string_view(text.end(), 0);
>+//
>+//       return absl::string_view(text, found, 1);
>+//     }
>+//   };
>+
>+// ByString
>+//
>+// A sub-std::string delimiter. If `StrSplit()` is passed a std::string in place of a
>+// `Delimiter` object, the std::string will be implicitly converted into a
>+// `ByString` delimiter.
>+//
>+// Example:
>+//
>+//   // Because a std::string literal is converted to an `absl::ByString`,
>+//   // the following two splits are equivalent.
>+//
>+//   std::vector<std::string> v1 = absl::StrSplit("a, b, c", ", ");
>+//
>+//   using absl::ByString;
>+//   std::vector<std::string> v2 = absl::StrSplit("a, b, c",
>+//                                                ByString(", "));
>+//   // v[0] == "a", v[1] == "b", v[2] == "c"
>+class ByString {
>+ public:
>+  explicit ByString(absl::string_view sp);
>+  absl::string_view Find(absl::string_view text, size_t pos) const;
>+
>+ private:
>+  const std::string delimiter_;
>+};
>+
>+// ByChar
>+//
>+// A single character delimiter. `ByChar` is functionally equivalent to a
>+// 1-char std::string within a `ByString` delimiter, but slightly more
>+// efficient.
>+//
>+// Example:
>+//
>+//   // Because a char literal is converted to a absl::ByChar,
>+//   // the following two splits are equivalent.
>+//   std::vector<std::string> v1 = absl::StrSplit("a,b,c", ',');
>+//   using absl::ByChar;
>+//   std::vector<std::string> v2 = absl::StrSplit("a,b,c", ByChar(','));
>+//   // v[0] == "a", v[1] == "b", v[2] == "c"
>+//
>+// `ByChar` is also the default delimiter if a single character is given
>+// as the delimiter to `StrSplit()`. For example, the following calls are
>+// equivalent:
>+//
>+//   std::vector<std::string> v = absl::StrSplit("a-b", '-');
>+//
>+//   using absl::ByChar;
>+//   std::vector<std::string> v = absl::StrSplit("a-b", ByChar('-'));
>+//
>+class ByChar {
>+ public:
>+  explicit ByChar(char c) : c_(c) {}
>+  absl::string_view Find(absl::string_view text, size_t pos) const;
>+
>+ private:
>+  char c_;
>+};
>+
>+// ByAnyChar
>+//
>+// A delimiter that will match any of the given byte-sized characters within
>+// its provided std::string.
>+//
>+// Note: this delimiter works with single-byte std::string data, but does not work
>+// with variable-width encodings, such as UTF-8.
>+//
>+// Example:
>+//
>+//   using absl::ByAnyChar;
>+//   std::vector<std::string> v = absl::StrSplit("a,b=c", ByAnyChar(",="));
>+//   // v[0] == "a", v[1] == "b", v[2] == "c"
>+//
>+// If `ByAnyChar` is given the empty std::string, it behaves exactly like
>+// `ByString` and matches each individual character in the input std::string.
>+//
>+class ByAnyChar {
>+ public:
>+  explicit ByAnyChar(absl::string_view sp);
>+  absl::string_view Find(absl::string_view text, size_t pos) const;
>+
>+ private:
>+  const std::string delimiters_;
>+};
>+
>+// ByLength
>+//
>+// A delimiter for splitting into equal-length strings. The length argument to
>+// the constructor must be greater than 0.
>+//
>+// Note: this delimiter works with single-byte std::string data, but does not work
>+// with variable-width encodings, such as UTF-8.
>+//
>+// Example:
>+//
>+//   using absl::ByLength;
>+//   std::vector<std::string> v = absl::StrSplit("123456789", ByLength(3));
>+
>+//   // v[0] == "123", v[1] == "456", v[2] == "789"
>+//
>+// Note that the std::string does not have to be a multiple of the fixed split
>+// length. In such a case, the last substring will be shorter.
>+//
>+//   using absl::ByLength;
>+//   std::vector<std::string> v = absl::StrSplit("12345", ByLength(2));
>+//
>+//   // v[0] == "12", v[1] == "34", v[2] == "5"
>+class ByLength {
>+ public:
>+  explicit ByLength(ptrdiff_t length);
>+  absl::string_view Find(absl::string_view text, size_t pos) const;
>+
>+ private:
>+  const ptrdiff_t length_;
>+};
>+
>+namespace strings_internal {
>+
>+// A traits-like metafunction for selecting the default Delimiter object type
>+// for a particular Delimiter type. The base case simply exposes type Delimiter
>+// itself as the delimiter's Type. However, there are specializations for
>+// std::string-like objects that map them to the ByString delimiter object.
>+// This allows functions like absl::StrSplit() and absl::MaxSplits() to accept
>+// std::string-like objects (e.g., ',') as delimiter arguments but they will be
>+// treated as if a ByString delimiter was given.
>+template <typename Delimiter>
>+struct SelectDelimiter {
>+  using type = Delimiter;
>+};
>+
>+template <>
>+struct SelectDelimiter<char> {
>+  using type = ByChar;
>+};
>+template <>
>+struct SelectDelimiter<char*> {
>+  using type = ByString;
>+};
>+template <>
>+struct SelectDelimiter<const char*> {
>+  using type = ByString;
>+};
>+template <>
>+struct SelectDelimiter<absl::string_view> {
>+  using type = ByString;
>+};
>+template <>
>+struct SelectDelimiter<std::string> {
>+  using type = ByString;
>+};
>+
>+// Wraps another delimiter and sets a max number of matches for that delimiter.
>+template <typename Delimiter>
>+class MaxSplitsImpl {
>+ public:
>+  MaxSplitsImpl(Delimiter delimiter, int limit)
>+      : delimiter_(delimiter), limit_(limit), count_(0) {}
>+  absl::string_view Find(absl::string_view text, size_t pos) {
>+    if (count_++ == limit_) {
>+      return absl::string_view(text.end(), 0);  // No more matches.
>+    }
>+    return delimiter_.Find(text, pos);
>+  }
>+
>+ private:
>+  Delimiter delimiter_;
>+  const int limit_;
>+  int count_;
>+};
>+
>+}  // namespace strings_internal
>+
>+// MaxSplits()
>+//
>+// A delimiter that limits the number of matches which can occur to the passed
>+// `limit`. The last element in the returned collection will contain all
>+// remaining unsplit pieces, which may contain instances of the delimiter.
>+// The collection will contain at most `limit` + 1 elements.
>+// Example:
>+//
>+//   using absl::MaxSplits;
>+//   std::vector<std::string> v = absl::StrSplit("a,b,c", MaxSplits(',', 1));
>+//
>+//   // v[0] == "a", v[1] == "b,c"
>+template <typename Delimiter>
>+inline strings_internal::MaxSplitsImpl<
>+    typename strings_internal::SelectDelimiter<Delimiter>::type>
>+MaxSplits(Delimiter delimiter, int limit) {
>+  typedef
>+      typename strings_internal::SelectDelimiter<Delimiter>::type DelimiterType;
>+  return strings_internal::MaxSplitsImpl<DelimiterType>(
>+      DelimiterType(delimiter), limit);
>+}
>+
>+//------------------------------------------------------------------------------
>+// Predicates
>+//------------------------------------------------------------------------------
>+//
>+// Predicates filter the results of a `StrSplit()` by determining whether or not
>+// a resultant element is included in the result set. A predicate may be passed
>+// as an optional third argument to the `StrSplit()` function.
>+//
>+// Predicates are unary functions (or functors) that take a single
>+// `absl::string_view` argument and return a bool indicating whether the
>+// argument should be included (`true`) or excluded (`false`).
>+//
>+// Predicates are useful when filtering out empty substrings. By default, empty
>+// substrings may be returned by `StrSplit()`, which is similar to the way split
>+// functions work in other programming languages.
>+
>+// AllowEmpty()
>+//
>+// Always returns `true`, indicating that all strings--including empty
>+// strings--should be included in the split output. This predicate is not
>+// strictly needed because this is the default behavior of `StrSplit()`;
>+// however, it might be useful at some call sites to make the intent explicit.
>+//
>+// Example:
>+//
>+//  std::vector<std::string> v = absl::StrSplit(" a , ,,b,", ',', AllowEmpty());
>+//
>+//  // v[0] == " a ", v[1] == " ", v[2] == "", v[3] = "b", v[4] == ""
>+struct AllowEmpty {
>+  bool operator()(absl::string_view) const { return true; }
>+};
>+
>+// SkipEmpty()
>+//
>+// Returns `false` if the given `absl::string_view` is empty, indicating that
>+// `StrSplit()` should omit the empty std::string.
>+//
>+// Example:
>+//
>+//   std::vector<std::string> v = absl::StrSplit(",a,,b,", ',', SkipEmpty());
>+//
>+//   // v[0] == "a", v[1] == "b"
>+//
>+// Note: `SkipEmpty()` does not consider a std::string containing only whitespace
>+// to be empty. To skip such whitespace as well, use the `SkipWhitespace()`
>+// predicate.
>+struct SkipEmpty {
>+  bool operator()(absl::string_view sp) const { return !sp.empty(); }
>+};
>+
>+// SkipWhitespace()
>+//
>+// Returns `false` if the given `absl::string_view` is empty *or* contains only
>+// whitespace, indicating that `StrSplit()` should omit the std::string.
>+//
>+// Example:
>+//
>+//   std::vector<std::string> v = absl::StrSplit(" a , ,,b,",
>+//                                               ',', SkipWhitespace());
>+//   // v[0] == " a ", v[1] == "b"
>+//
>+//   // SkipEmpty() would return whitespace elements
>+//   std::vector<std::string> v = absl::StrSplit(" a , ,,b,", ',', SkipEmpty());
>+//   // v[0] == " a ", v[1] == " ", v[2] == "b"
>+struct SkipWhitespace {
>+  bool operator()(absl::string_view sp) const {
>+    sp = absl::StripAsciiWhitespace(sp);
>+    return !sp.empty();
>+  }
>+};
>+
>+//------------------------------------------------------------------------------
>+//                                  StrSplit()
>+//------------------------------------------------------------------------------
>+
>+// StrSplit()
>+//
>+// Splits a given std::string based on the provided `Delimiter` object, returning the
>+// elements within the type specified by the caller. Optionally, you may pass a
>+// `Predicate` to `StrSplit()` indicating whether to include or exclude the
>+// resulting element within the final result set. (See the overviews for
>+// Delimiters and Predicates above.)
>+//
>+// Example:
>+//
>+//   std::vector<std::string> v = absl::StrSplit("a,b,c,d", ',');
>+//   // v[0] == "a", v[1] == "b", v[2] == "c", v[3] == "d"
>+//
>+// You can also provide an explicit `Delimiter` object:
>+//
>+// Example:
>+//
>+//   using absl::ByAnyChar;
>+//   std::vector<std::string> v = absl::StrSplit("a,b=c", ByAnyChar(",="));
>+//   // v[0] == "a", v[1] == "b", v[2] == "c"
>+//
>+// See above for more information on delimiters.
>+//
>+// By default, empty strings are included in the result set. You can optionally
>+// include a third `Predicate` argument to apply a test for whether the
>+// resultant element should be included in the result set:
>+//
>+// Example:
>+//
>+//   std::vector<std::string> v = absl::StrSplit(" a , ,,b,",
>+//                                               ',', SkipWhitespace());
>+//   // v[0] == " a ", v[1] == "b"
>+//
>+// See above for more information on predicates.
>+//
>+//------------------------------------------------------------------------------
>+// StrSplit() Return Types
>+//------------------------------------------------------------------------------
>+//
>+// The `StrSplit()` function adapts the returned collection to the collection
>+// specified by the caller (e.g. `std::vector` above). The returned collections
>+// may contain `string`, `absl::string_view` (in which case the original std::string
>+// being split must ensure that it outlives the collection), or any object that
>+// can be explicitly created from an `absl::string_view`. This behavior works
>+// for:
>+//
>+// 1) All standard STL containers including `std::vector`, `std::list`,
>+//    `std::deque`, `std::set`,`std::multiset`, 'std::map`, and `std::multimap`
>+// 2) `std::pair` (which is not actually a container). See below.
>+//
>+// Example:
>+//
>+//   // The results are returned as `absl::string_view` objects. Note that we
>+//   // have to ensure that the input std::string outlives any results.
>+//   std::vector<absl::string_view> v = absl::StrSplit("a,b,c", ',');
>+//
>+//   // Stores results in a std::set<std::string>, which also performs
>+//   // de-duplication and orders the elements in ascending order.
>+//   std::set<std::string> a = absl::StrSplit("b,a,c,a,b", ',');
>+//   // v[0] == "a", v[1] == "b", v[2] = "c"
>+//
>+//   // `StrSplit()` can be used within a range-based for loop, in which case
>+//   // each element will be of type `absl::string_view`.
>+//   std::vector<std::string> v;
>+//   for (const auto sv : absl::StrSplit("a,b,c", ',')) {
>+//     if (sv != "b") v.emplace_back(sv);
>+//   }
>+//   // v[0] == "a", v[1] == "c"
>+//
>+//   // Stores results in a map. The map implementation assumes that the input
>+//   // is provided as a series of key/value pairs. For example, the 0th element
>+//   // resulting from the split will be stored as a key to the 1st element. If
>+//   // an odd number of elements are resolved, the last element is paired with
>+//   // a default-constructed value (e.g., empty std::string).
>+//   std::map<std::string, std::string> m = absl::StrSplit("a,b,c", ',');
>+//   // m["a"] == "b", m["c"] == ""     // last component value equals ""
>+//
>+// Splitting to `std::pair` is an interesting case because it can hold only two
>+// elements and is not a collection type. When splitting to a `std::pair` the
>+// first two split strings become the `std::pair` `.first` and `.second`
>+// members, respectively. The remaining split substrings are discarded. If there
>+// are less than two split substrings, the empty std::string is used for the
>+// corresponding
>+// `std::pair` member.
>+//
>+// Example:
>+//
>+//   // Stores first two split strings as the members in a std::pair.
>+//   std::pair<std::string, std::string> p = absl::StrSplit("a,b,c", ',');
>+//   // p.first == "a", p.second == "b"       // "c" is omitted.
>+//
>+// The `StrSplit()` function can be used multiple times to perform more
>+// complicated splitting logic, such as intelligently parsing key-value pairs.
>+//
>+// Example:
>+//
>+//   // The input std::string "a=b=c,d=e,f=,g" becomes
>+//   // { "a" => "b=c", "d" => "e", "f" => "", "g" => "" }
>+//   std::map<std::string, std::string> m;
>+//   for (absl::string_view sp : absl::StrSplit("a=b=c,d=e,f=,g", ',')) {
>+//     m.insert(absl::StrSplit(sp, absl::MaxSplits('=', 1)));
>+//   }
>+//   EXPECT_EQ("b=c", m.find("a")->second);
>+//   EXPECT_EQ("e", m.find("d")->second);
>+//   EXPECT_EQ("", m.find("f")->second);
>+//   EXPECT_EQ("", m.find("g")->second);
>+//
>+// WARNING: Due to a legacy bug that is maintained for backward compatibility,
>+// splitting the following empty string_views produces different results:
>+//
>+//   absl::StrSplit(absl::string_view(""), '-');  // {""}
>+//   absl::StrSplit(absl::string_view(), '-');    // {}, but should be {""}
>+//
>+// Try not to depend on this distinction because the bug may one day be fixed.
>+template <typename Delimiter>
>+strings_internal::Splitter<
>+    typename strings_internal::SelectDelimiter<Delimiter>::type, AllowEmpty>
>+StrSplit(strings_internal::ConvertibleToStringView text, Delimiter d) {
>+  using DelimiterType =
>+      typename strings_internal::SelectDelimiter<Delimiter>::type;
>+  return strings_internal::Splitter<DelimiterType, AllowEmpty>(
>+      std::move(text), DelimiterType(d), AllowEmpty());
>+}
>+
>+template <typename Delimiter, typename Predicate>
>+strings_internal::Splitter<
>+    typename strings_internal::SelectDelimiter<Delimiter>::type, Predicate>
>+StrSplit(strings_internal::ConvertibleToStringView text, Delimiter d,
>+         Predicate p) {
>+  using DelimiterType =
>+      typename strings_internal::SelectDelimiter<Delimiter>::type;
>+  return strings_internal::Splitter<DelimiterType, Predicate>(
>+      std::move(text), DelimiterType(d), std::move(p));
>+}
>+
>+}  // namespace absl
>+
>+#endif  // ABSL_STRINGS_STR_SPLIT_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_split_benchmark.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_split_benchmark.cc
>new file mode 100644
>index 00000000000..326ff744ebd
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_split_benchmark.cc
>@@ -0,0 +1,156 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/strings/str_split.h"
>+
>+#include <iterator>
>+#include <string>
>+#include <unordered_map>
>+#include <unordered_set>
>+#include <vector>
>+
>+#include "benchmark/benchmark.h"
>+#include "absl/base/internal/raw_logging.h"
>+#include "absl/strings/string_view.h"
>+
>+namespace {
>+
>+std::string MakeTestString(int desired_length) {
>+  static const int kAverageValueLen = 25;
>+  std::string test(desired_length * kAverageValueLen, 'x');
>+  for (int i = 1; i < test.size(); i += kAverageValueLen) {
>+    test[i] = ';';
>+  }
>+  return test;
>+}
>+
>+void BM_Split2StringPiece(benchmark::State& state) {
>+  std::string test = MakeTestString(state.range(0));
>+  for (auto _ : state) {
>+    std::vector<absl::string_view> result = absl::StrSplit(test, ';');
>+    benchmark::DoNotOptimize(result);
>+  }
>+}
>+BENCHMARK_RANGE(BM_Split2StringPiece, 0, 1 << 20);
>+
>+void BM_Split2StringPieceLifted(benchmark::State& state) {
>+  std::string test = MakeTestString(state.range(0));
>+  std::vector<absl::string_view> result;
>+  for (auto _ : state) {
>+    result = absl::StrSplit(test, ';');
>+  }
>+  benchmark::DoNotOptimize(result);
>+}
>+BENCHMARK_RANGE(BM_Split2StringPieceLifted, 0, 1 << 20);
>+
>+void BM_Split2String(benchmark::State& state) {
>+  std::string test = MakeTestString(state.range(0));
>+  for (auto _ : state) {
>+    std::vector<std::string> result = absl::StrSplit(test, ';');
>+    benchmark::DoNotOptimize(result);
>+  }
>+}
>+BENCHMARK_RANGE(BM_Split2String, 0, 1 << 20);
>+
>+// This benchmark is for comparing Split2 to Split1 (SplitStringUsing). In
>+// particular, this benchmark uses SkipEmpty() to match SplitStringUsing's
>+// behavior.
>+void BM_Split2SplitStringUsing(benchmark::State& state) {
>+  std::string test = MakeTestString(state.range(0));
>+  for (auto _ : state) {
>+    std::vector<std::string> result = absl::StrSplit(test, ';', absl::SkipEmpty());
>+    benchmark::DoNotOptimize(result);
>+  }
>+}
>+BENCHMARK_RANGE(BM_Split2SplitStringUsing, 0, 1 << 20);
>+
>+void BM_SplitStringToUnorderedSet(benchmark::State& state) {
>+  const int len = state.range(0);
>+  std::string test(len, 'x');
>+  for (int i = 1; i < len; i += 2) {
>+    test[i] = ';';
>+  }
>+  for (auto _ : state) {
>+    std::unordered_set<std::string> result =
>+        absl::StrSplit(test, ':', absl::SkipEmpty());
>+    benchmark::DoNotOptimize(result);
>+  }
>+}
>+BENCHMARK_RANGE(BM_SplitStringToUnorderedSet, 0, 1 << 20);
>+
>+void BM_SplitStringToUnorderedMap(benchmark::State& state) {
>+  const int len = state.range(0);
>+  std::string test(len, 'x');
>+  for (int i = 1; i < len; i += 2) {
>+    test[i] = ';';
>+  }
>+  for (auto _ : state) {
>+    std::unordered_map<std::string, std::string> result =
>+        absl::StrSplit(test, ':', absl::SkipEmpty());
>+    benchmark::DoNotOptimize(result);
>+  }
>+}
>+BENCHMARK_RANGE(BM_SplitStringToUnorderedMap, 0, 1 << 20);
>+
>+void BM_SplitStringAllowEmpty(benchmark::State& state) {
>+  const int len = state.range(0);
>+  std::string test(len, 'x');
>+  for (int i = 1; i < len; i += 2) {
>+    test[i] = ';';
>+  }
>+  for (auto _ : state) {
>+    std::vector<std::string> result = absl::StrSplit(test, ';');
>+    benchmark::DoNotOptimize(result);
>+  }
>+}
>+BENCHMARK_RANGE(BM_SplitStringAllowEmpty, 0, 1 << 20);
>+
>+struct OneCharLiteral {
>+  char operator()() const { return 'X'; }
>+};
>+
>+struct OneCharStringLiteral {
>+  const char* operator()() const { return "X"; }
>+};
>+
>+template <typename DelimiterFactory>
>+void BM_SplitStringWithOneChar(benchmark::State& state) {
>+  const auto delimiter = DelimiterFactory()();
>+  std::vector<absl::string_view> pieces;
>+  size_t v = 0;
>+  for (auto _ : state) {
>+    pieces = absl::StrSplit("The quick brown fox jumps over the lazy dog",
>+                            delimiter);
>+    v += pieces.size();
>+  }
>+  ABSL_RAW_CHECK(v == state.iterations(), "");
>+}
>+BENCHMARK_TEMPLATE(BM_SplitStringWithOneChar, OneCharLiteral);
>+BENCHMARK_TEMPLATE(BM_SplitStringWithOneChar, OneCharStringLiteral);
>+
>+template <typename DelimiterFactory>
>+void BM_SplitStringWithOneCharNoVector(benchmark::State& state) {
>+  const auto delimiter = DelimiterFactory()();
>+  size_t v = 0;
>+  for (auto _ : state) {
>+    auto splitter = absl::StrSplit(
>+        "The quick brown fox jumps over the lazy dog", delimiter);
>+    v += std::distance(splitter.begin(), splitter.end());
>+  }
>+  ABSL_RAW_CHECK(v == state.iterations(), "");
>+}
>+BENCHMARK_TEMPLATE(BM_SplitStringWithOneCharNoVector, OneCharLiteral);
>+BENCHMARK_TEMPLATE(BM_SplitStringWithOneCharNoVector, OneCharStringLiteral);
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_split_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_split_test.cc
>new file mode 100644
>index 00000000000..29a68047c99
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/str_split_test.cc
>@@ -0,0 +1,935 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/strings/str_split.h"
>+
>+#include <deque>
>+#include <initializer_list>
>+#include <list>
>+#include <map>
>+#include <memory>
>+#include <string>
>+#include <type_traits>
>+#include <unordered_map>
>+#include <unordered_set>
>+#include <vector>
>+
>+#include "gmock/gmock.h"
>+#include "gtest/gtest.h"
>+#include "absl/base/dynamic_annotations.h"  // for AbslRunningOnValgrind
>+#include "absl/base/macros.h"
>+#include "absl/strings/numbers.h"
>+
>+namespace {
>+
>+using ::testing::ElementsAre;
>+using ::testing::Pair;
>+using ::testing::UnorderedElementsAre;
>+
>+TEST(Split, TraitsTest) {
>+  static_assert(!absl::strings_internal::SplitterIsConvertibleTo<int>::value,
>+                "");
>+  static_assert(!absl::strings_internal::SplitterIsConvertibleTo<std::string>::value,
>+                "");
>+  static_assert(absl::strings_internal::SplitterIsConvertibleTo<
>+                    std::vector<std::string>>::value,
>+                "");
>+  static_assert(
>+      !absl::strings_internal::SplitterIsConvertibleTo<std::vector<int>>::value,
>+      "");
>+  static_assert(absl::strings_internal::SplitterIsConvertibleTo<
>+                    std::vector<absl::string_view>>::value,
>+                "");
>+  static_assert(absl::strings_internal::SplitterIsConvertibleTo<
>+                    std::map<std::string, std::string>>::value,
>+                "");
>+  static_assert(absl::strings_internal::SplitterIsConvertibleTo<
>+                    std::map<absl::string_view, absl::string_view>>::value,
>+                "");
>+  static_assert(!absl::strings_internal::SplitterIsConvertibleTo<
>+                    std::map<int, std::string>>::value,
>+                "");
>+  static_assert(!absl::strings_internal::SplitterIsConvertibleTo<
>+                    std::map<std::string, int>>::value,
>+                "");
>+}
>+
>+// This tests the overall split API, which is made up of the absl::StrSplit()
>+// function and the Delimiter objects in the absl:: namespace.
>+// This TEST macro is outside of any namespace to require full specification of
>+// namespaces just like callers will need to use.
>+TEST(Split, APIExamples) {
>+  {
>+    // Passes std::string delimiter. Assumes the default of Literal.
>+    std::vector<std::string> v = absl::StrSplit("a,b,c", ',');
>+    EXPECT_THAT(v, ElementsAre("a", "b", "c"));
>+
>+    // Equivalent to...
>+    using absl::ByString;
>+    v = absl::StrSplit("a,b,c", ByString(","));
>+    EXPECT_THAT(v, ElementsAre("a", "b", "c"));
>+
>+    // Equivalent to...
>+    EXPECT_THAT(absl::StrSplit("a,b,c", ByString(",")),
>+                ElementsAre("a", "b", "c"));
>+  }
>+
>+  {
>+    // Same as above, but using a single character as the delimiter.
>+    std::vector<std::string> v = absl::StrSplit("a,b,c", ',');
>+    EXPECT_THAT(v, ElementsAre("a", "b", "c"));
>+
>+    // Equivalent to...
>+    using absl::ByChar;
>+    v = absl::StrSplit("a,b,c", ByChar(','));
>+    EXPECT_THAT(v, ElementsAre("a", "b", "c"));
>+  }
>+
>+  {
>+    // Same as above, but using std::string
>+    std::vector<std::string> v = absl::StrSplit("a,b,c", ',');
>+    EXPECT_THAT(v, ElementsAre("a", "b", "c"));
>+
>+    // Equivalent to...
>+    using absl::ByChar;
>+    v = absl::StrSplit("a,b,c", ByChar(','));
>+    EXPECT_THAT(v, ElementsAre("a", "b", "c"));
>+  }
>+
>+  {
>+    // Uses the Literal std::string "=>" as the delimiter.
>+    const std::vector<std::string> v = absl::StrSplit("a=>b=>c", "=>");
>+    EXPECT_THAT(v, ElementsAre("a", "b", "c"));
>+  }
>+
>+  {
>+    // The substrings are returned as string_views, eliminating copying.
>+    std::vector<absl::string_view> v = absl::StrSplit("a,b,c", ',');
>+    EXPECT_THAT(v, ElementsAre("a", "b", "c"));
>+  }
>+
>+  {
>+    // Leading and trailing empty substrings.
>+    std::vector<std::string> v = absl::StrSplit(",a,b,c,", ',');
>+    EXPECT_THAT(v, ElementsAre("", "a", "b", "c", ""));
>+  }
>+
>+  {
>+    // Splits on a delimiter that is not found.
>+    std::vector<std::string> v = absl::StrSplit("abc", ',');
>+    EXPECT_THAT(v, ElementsAre("abc"));
>+  }
>+
>+  {
>+    // Splits the input std::string into individual characters by using an empty
>+    // std::string as the delimiter.
>+    std::vector<std::string> v = absl::StrSplit("abc", "");
>+    EXPECT_THAT(v, ElementsAre("a", "b", "c"));
>+  }
>+
>+  {
>+    // Splits std::string data with embedded NUL characters, using NUL as the
>+    // delimiter. A simple delimiter of "\0" doesn't work because strlen() will
>+    // say that's the empty std::string when constructing the absl::string_view
>+    // delimiter. Instead, a non-empty std::string containing NUL can be used as the
>+    // delimiter.
>+    std::string embedded_nulls("a\0b\0c", 5);
>+    std::string null_delim("\0", 1);
>+    std::vector<std::string> v = absl::StrSplit(embedded_nulls, null_delim);
>+    EXPECT_THAT(v, ElementsAre("a", "b", "c"));
>+  }
>+
>+  {
>+    // Stores first two split strings as the members in a std::pair.
>+    std::pair<std::string, std::string> p = absl::StrSplit("a,b,c", ',');
>+    EXPECT_EQ("a", p.first);
>+    EXPECT_EQ("b", p.second);
>+    // "c" is omitted because std::pair can hold only two elements.
>+  }
>+
>+  {
>+    // Results stored in std::set<std::string>
>+    std::set<std::string> v = absl::StrSplit("a,b,c,a,b,c,a,b,c", ',');
>+    EXPECT_THAT(v, ElementsAre("a", "b", "c"));
>+  }
>+
>+  {
>+    // Uses a non-const char* delimiter.
>+    char a[] = ",";
>+    char* d = a + 0;
>+    std::vector<std::string> v = absl::StrSplit("a,b,c", d);
>+    EXPECT_THAT(v, ElementsAre("a", "b", "c"));
>+  }
>+
>+  {
>+    // Results split using either of , or ;
>+    using absl::ByAnyChar;
>+    std::vector<std::string> v = absl::StrSplit("a,b;c", ByAnyChar(",;"));
>+    EXPECT_THAT(v, ElementsAre("a", "b", "c"));
>+  }
>+
>+  {
>+    // Uses the SkipWhitespace predicate.
>+    using absl::SkipWhitespace;
>+    std::vector<std::string> v = absl::StrSplit(" a , ,,b,", ',', SkipWhitespace());
>+    EXPECT_THAT(v, ElementsAre(" a ", "b"));
>+  }
>+
>+  {
>+    // Uses the ByLength delimiter.
>+    using absl::ByLength;
>+    std::vector<std::string> v = absl::StrSplit("abcdefg", ByLength(3));
>+    EXPECT_THAT(v, ElementsAre("abc", "def", "g"));
>+  }
>+
>+  {
>+    // Different forms of initialization / conversion.
>+    std::vector<std::string> v1 = absl::StrSplit("a,b,c", ',');
>+    EXPECT_THAT(v1, ElementsAre("a", "b", "c"));
>+    std::vector<std::string> v2(absl::StrSplit("a,b,c", ','));
>+    EXPECT_THAT(v2, ElementsAre("a", "b", "c"));
>+    auto v3 = std::vector<std::string>(absl::StrSplit("a,b,c", ','));
>+    EXPECT_THAT(v3, ElementsAre("a", "b", "c"));
>+    v3 = absl::StrSplit("a,b,c", ',');
>+    EXPECT_THAT(v3, ElementsAre("a", "b", "c"));
>+  }
>+
>+  {
>+    // Results stored in a std::map.
>+    std::map<std::string, std::string> m = absl::StrSplit("a,1,b,2,a,3", ',');
>+    EXPECT_EQ(2, m.size());
>+    EXPECT_EQ("3", m["a"]);
>+    EXPECT_EQ("2", m["b"]);
>+  }
>+
>+  {
>+    // Results stored in a std::multimap.
>+    std::multimap<std::string, std::string> m = absl::StrSplit("a,1,b,2,a,3", ',');
>+    EXPECT_EQ(3, m.size());
>+    auto it = m.find("a");
>+    EXPECT_EQ("1", it->second);
>+    ++it;
>+    EXPECT_EQ("3", it->second);
>+    it = m.find("b");
>+    EXPECT_EQ("2", it->second);
>+  }
>+
>+  {
>+    // Demonstrates use in a range-based for loop in C++11.
>+    std::string s = "x,x,x,x,x,x,x";
>+    for (absl::string_view sp : absl::StrSplit(s, ',')) {
>+      EXPECT_EQ("x", sp);
>+    }
>+  }
>+
>+  {
>+    // Demonstrates use with a Predicate in a range-based for loop.
>+    using absl::SkipWhitespace;
>+    std::string s = " ,x,,x,,x,x,x,,";
>+    for (absl::string_view sp : absl::StrSplit(s, ',', SkipWhitespace())) {
>+      EXPECT_EQ("x", sp);
>+    }
>+  }
>+
>+  {
>+    // Demonstrates a "smart" split to std::map using two separate calls to
>+    // absl::StrSplit. One call to split the records, and another call to split
>+    // the keys and values. This also uses the Limit delimiter so that the
>+    // std::string "a=b=c" will split to "a" -> "b=c".
>+    std::map<std::string, std::string> m;
>+    for (absl::string_view sp : absl::StrSplit("a=b=c,d=e,f=,g", ',')) {
>+      m.insert(absl::StrSplit(sp, absl::MaxSplits('=', 1)));
>+    }
>+    EXPECT_EQ("b=c", m.find("a")->second);
>+    EXPECT_EQ("e", m.find("d")->second);
>+    EXPECT_EQ("", m.find("f")->second);
>+    EXPECT_EQ("", m.find("g")->second);
>+  }
>+}
>+
>+//
>+// Tests for SplitIterator
>+//
>+
>+TEST(SplitIterator, Basics) {
>+  auto splitter = absl::StrSplit("a,b", ',');
>+  auto it = splitter.begin();
>+  auto end = splitter.end();
>+
>+  EXPECT_NE(it, end);
>+  EXPECT_EQ("a", *it);  // tests dereference
>+  ++it;                 // tests preincrement
>+  EXPECT_NE(it, end);
>+  EXPECT_EQ("b", std::string(it->data(), it->size()));  // tests dereference as ptr
>+  it++;                                            // tests postincrement
>+  EXPECT_EQ(it, end);
>+}
>+
>+// Simple Predicate to skip a particular std::string.
>+class Skip {
>+ public:
>+  explicit Skip(const std::string& s) : s_(s) {}
>+  bool operator()(absl::string_view sp) { return sp != s_; }
>+
>+ private:
>+  std::string s_;
>+};
>+
>+TEST(SplitIterator, Predicate) {
>+  auto splitter = absl::StrSplit("a,b,c", ',', Skip("b"));
>+  auto it = splitter.begin();
>+  auto end = splitter.end();
>+
>+  EXPECT_NE(it, end);
>+  EXPECT_EQ("a", *it);  // tests dereference
>+  ++it;                 // tests preincrement -- "b" should be skipped here.
>+  EXPECT_NE(it, end);
>+  EXPECT_EQ("c", std::string(it->data(), it->size()));  // tests dereference as ptr
>+  it++;                                            // tests postincrement
>+  EXPECT_EQ(it, end);
>+}
>+
>+TEST(SplitIterator, EdgeCases) {
>+  // Expected input and output, assuming a delimiter of ','
>+  struct {
>+    std::string in;
>+    std::vector<std::string> expect;
>+  } specs[] = {
>+      {"", {""}},
>+      {"foo", {"foo"}},
>+      {",", {"", ""}},
>+      {",foo", {"", "foo"}},
>+      {"foo,", {"foo", ""}},
>+      {",foo,", {"", "foo", ""}},
>+      {"foo,bar", {"foo", "bar"}},
>+  };
>+
>+  for (const auto& spec : specs) {
>+    SCOPED_TRACE(spec.in);
>+    auto splitter = absl::StrSplit(spec.in, ',');
>+    auto it = splitter.begin();
>+    auto end = splitter.end();
>+    for (const auto& expected : spec.expect) {
>+      EXPECT_NE(it, end);
>+      EXPECT_EQ(expected, *it++);
>+    }
>+    EXPECT_EQ(it, end);
>+  }
>+}
>+
>+TEST(Splitter, Const) {
>+  const auto splitter = absl::StrSplit("a,b,c", ',');
>+  EXPECT_THAT(splitter, ElementsAre("a", "b", "c"));
>+}
>+
>+TEST(Split, EmptyAndNull) {
>+  // Attention: Splitting a null absl::string_view is different than splitting
>+  // an empty absl::string_view even though both string_views are considered
>+  // equal. This behavior is likely surprising and undesirable. However, to
>+  // maintain backward compatibility, there is a small "hack" in
>+  // str_split_internal.h that preserves this behavior. If that behavior is ever
>+  // changed/fixed, this test will need to be updated.
>+  EXPECT_THAT(absl::StrSplit(absl::string_view(""), '-'), ElementsAre(""));
>+  EXPECT_THAT(absl::StrSplit(absl::string_view(), '-'), ElementsAre());
>+}
>+
>+TEST(SplitIterator, EqualityAsEndCondition) {
>+  auto splitter = absl::StrSplit("a,b,c", ',');
>+  auto it = splitter.begin();
>+  auto it2 = it;
>+
>+  // Increments it2 twice to point to "c" in the input text.
>+  ++it2;
>+  ++it2;
>+  EXPECT_EQ("c", *it2);
>+
>+  // This test uses a non-end SplitIterator as the terminating condition in a
>+  // for loop. This relies on SplitIterator equality for non-end SplitIterators
>+  // working correctly. At this point it2 points to "c", and we use that as the
>+  // "end" condition in this test.
>+  std::vector<absl::string_view> v;
>+  for (; it != it2; ++it) {
>+    v.push_back(*it);
>+  }
>+  EXPECT_THAT(v, ElementsAre("a", "b"));
>+}
>+
>+//
>+// Tests for Splitter
>+//
>+
>+TEST(Splitter, RangeIterators) {
>+  auto splitter = absl::StrSplit("a,b,c", ',');
>+  std::vector<absl::string_view> output;
>+  for (const absl::string_view p : splitter) {
>+    output.push_back(p);
>+  }
>+  EXPECT_THAT(output, ElementsAre("a", "b", "c"));
>+}
>+
>+// Some template functions for use in testing conversion operators
>+template <typename ContainerType, typename Splitter>
>+void TestConversionOperator(const Splitter& splitter) {
>+  ContainerType output = splitter;
>+  EXPECT_THAT(output, UnorderedElementsAre("a", "b", "c", "d"));
>+}
>+
>+template <typename MapType, typename Splitter>
>+void TestMapConversionOperator(const Splitter& splitter) {
>+  MapType m = splitter;
>+  EXPECT_THAT(m, UnorderedElementsAre(Pair("a", "b"), Pair("c", "d")));
>+}
>+
>+template <typename FirstType, typename SecondType, typename Splitter>
>+void TestPairConversionOperator(const Splitter& splitter) {
>+  std::pair<FirstType, SecondType> p = splitter;
>+  EXPECT_EQ(p, (std::pair<FirstType, SecondType>("a", "b")));
>+}
>+
>+TEST(Splitter, ConversionOperator) {
>+  auto splitter = absl::StrSplit("a,b,c,d", ',');
>+
>+  TestConversionOperator<std::vector<absl::string_view>>(splitter);
>+  TestConversionOperator<std::vector<std::string>>(splitter);
>+  TestConversionOperator<std::list<absl::string_view>>(splitter);
>+  TestConversionOperator<std::list<std::string>>(splitter);
>+  TestConversionOperator<std::deque<absl::string_view>>(splitter);
>+  TestConversionOperator<std::deque<std::string>>(splitter);
>+  TestConversionOperator<std::set<absl::string_view>>(splitter);
>+  TestConversionOperator<std::set<std::string>>(splitter);
>+  TestConversionOperator<std::multiset<absl::string_view>>(splitter);
>+  TestConversionOperator<std::multiset<std::string>>(splitter);
>+  TestConversionOperator<std::unordered_set<std::string>>(splitter);
>+
>+  // Tests conversion to map-like objects.
>+
>+  TestMapConversionOperator<std::map<absl::string_view, absl::string_view>>(
>+      splitter);
>+  TestMapConversionOperator<std::map<absl::string_view, std::string>>(splitter);
>+  TestMapConversionOperator<std::map<std::string, absl::string_view>>(splitter);
>+  TestMapConversionOperator<std::map<std::string, std::string>>(splitter);
>+  TestMapConversionOperator<
>+      std::multimap<absl::string_view, absl::string_view>>(splitter);
>+  TestMapConversionOperator<std::multimap<absl::string_view, std::string>>(splitter);
>+  TestMapConversionOperator<std::multimap<std::string, absl::string_view>>(splitter);
>+  TestMapConversionOperator<std::multimap<std::string, std::string>>(splitter);
>+  TestMapConversionOperator<std::unordered_map<std::string, std::string>>(splitter);
>+
>+  // Tests conversion to std::pair
>+
>+  TestPairConversionOperator<absl::string_view, absl::string_view>(splitter);
>+  TestPairConversionOperator<absl::string_view, std::string>(splitter);
>+  TestPairConversionOperator<std::string, absl::string_view>(splitter);
>+  TestPairConversionOperator<std::string, std::string>(splitter);
>+}
>+
>+// A few additional tests for conversion to std::pair. This conversion is
>+// different from others because a std::pair always has exactly two elements:
>+// .first and .second. The split has to work even when the split has
>+// less-than, equal-to, and more-than 2 strings.
>+TEST(Splitter, ToPair) {
>+  {
>+    // Empty std::string
>+    std::pair<std::string, std::string> p = absl::StrSplit("", ',');
>+    EXPECT_EQ("", p.first);
>+    EXPECT_EQ("", p.second);
>+  }
>+
>+  {
>+    // Only first
>+    std::pair<std::string, std::string> p = absl::StrSplit("a", ',');
>+    EXPECT_EQ("a", p.first);
>+    EXPECT_EQ("", p.second);
>+  }
>+
>+  {
>+    // Only second
>+    std::pair<std::string, std::string> p = absl::StrSplit(",b", ',');
>+    EXPECT_EQ("", p.first);
>+    EXPECT_EQ("b", p.second);
>+  }
>+
>+  {
>+    // First and second.
>+    std::pair<std::string, std::string> p = absl::StrSplit("a,b", ',');
>+    EXPECT_EQ("a", p.first);
>+    EXPECT_EQ("b", p.second);
>+  }
>+
>+  {
>+    // First and second and then more stuff that will be ignored.
>+    std::pair<std::string, std::string> p = absl::StrSplit("a,b,c", ',');
>+    EXPECT_EQ("a", p.first);
>+    EXPECT_EQ("b", p.second);
>+    // "c" is omitted.
>+  }
>+}
>+
>+TEST(Splitter, Predicates) {
>+  static const char kTestChars[] = ",a, ,b,";
>+  using absl::AllowEmpty;
>+  using absl::SkipEmpty;
>+  using absl::SkipWhitespace;
>+
>+  {
>+    // No predicate. Does not skip empties.
>+    auto splitter = absl::StrSplit(kTestChars, ',');
>+    std::vector<std::string> v = splitter;
>+    EXPECT_THAT(v, ElementsAre("", "a", " ", "b", ""));
>+  }
>+
>+  {
>+    // Allows empty strings. Same behavior as no predicate at all.
>+    auto splitter = absl::StrSplit(kTestChars, ',', AllowEmpty());
>+    std::vector<std::string> v_allowempty = splitter;
>+    EXPECT_THAT(v_allowempty, ElementsAre("", "a", " ", "b", ""));
>+
>+    // Ensures AllowEmpty equals the behavior with no predicate.
>+    auto splitter_nopredicate = absl::StrSplit(kTestChars, ',');
>+    std::vector<std::string> v_nopredicate = splitter_nopredicate;
>+    EXPECT_EQ(v_allowempty, v_nopredicate);
>+  }
>+
>+  {
>+    // Skips empty strings.
>+    auto splitter = absl::StrSplit(kTestChars, ',', SkipEmpty());
>+    std::vector<std::string> v = splitter;
>+    EXPECT_THAT(v, ElementsAre("a", " ", "b"));
>+  }
>+
>+  {
>+    // Skips empty and all-whitespace strings.
>+    auto splitter = absl::StrSplit(kTestChars, ',', SkipWhitespace());
>+    std::vector<std::string> v = splitter;
>+    EXPECT_THAT(v, ElementsAre("a", "b"));
>+  }
>+}
>+
>+//
>+// Tests for StrSplit()
>+//
>+
>+TEST(Split, Basics) {
>+  {
>+    // Doesn't really do anything useful because the return value is ignored,
>+    // but it should work.
>+    absl::StrSplit("a,b,c", ',');
>+  }
>+
>+  {
>+    std::vector<absl::string_view> v = absl::StrSplit("a,b,c", ',');
>+    EXPECT_THAT(v, ElementsAre("a", "b", "c"));
>+  }
>+
>+  {
>+    std::vector<std::string> v = absl::StrSplit("a,b,c", ',');
>+    EXPECT_THAT(v, ElementsAre("a", "b", "c"));
>+  }
>+
>+  {
>+    // Ensures that assignment works. This requires a little extra work with
>+    // C++11 because of overloads with initializer_list.
>+    std::vector<std::string> v;
>+    v = absl::StrSplit("a,b,c", ',');
>+
>+    EXPECT_THAT(v, ElementsAre("a", "b", "c"));
>+    std::map<std::string, std::string> m;
>+    m = absl::StrSplit("a,b,c", ',');
>+    EXPECT_EQ(2, m.size());
>+    std::unordered_map<std::string, std::string> hm;
>+    hm = absl::StrSplit("a,b,c", ',');
>+    EXPECT_EQ(2, hm.size());
>+  }
>+}
>+
>+absl::string_view ReturnStringView() { return "Hello World"; }
>+const char* ReturnConstCharP() { return "Hello World"; }
>+char* ReturnCharP() { return const_cast<char*>("Hello World"); }
>+
>+TEST(Split, AcceptsCertainTemporaries) {
>+  std::vector<std::string> v;
>+  v = absl::StrSplit(ReturnStringView(), ' ');
>+  EXPECT_THAT(v, ElementsAre("Hello", "World"));
>+  v = absl::StrSplit(ReturnConstCharP(), ' ');
>+  EXPECT_THAT(v, ElementsAre("Hello", "World"));
>+  v = absl::StrSplit(ReturnCharP(), ' ');
>+  EXPECT_THAT(v, ElementsAre("Hello", "World"));
>+}
>+
>+TEST(Split, Temporary) {
>+  // Use a std::string longer than the small-std::string-optimization length, so that when
>+  // the temporary is destroyed, if the splitter keeps a reference to the
>+  // std::string's contents, it'll reference freed memory instead of just dead
>+  // on-stack memory.
>+  const char input[] = "a,b,c,d,e,f,g,h,i,j,k,l,m,n,o,p,q,r,s,t,u";
>+  EXPECT_LT(sizeof(std::string), ABSL_ARRAYSIZE(input))
>+      << "Input should be larger than fits on the stack.";
>+
>+  // This happens more often in C++11 as part of a range-based for loop.
>+  auto splitter = absl::StrSplit(std::string(input), ',');
>+  std::string expected = "a";
>+  for (absl::string_view letter : splitter) {
>+    EXPECT_EQ(expected, letter);
>+    ++expected[0];
>+  }
>+  EXPECT_EQ("v", expected);
>+
>+  // This happens more often in C++11 as part of a range-based for loop.
>+  auto std_splitter = absl::StrSplit(std::string(input), ',');
>+  expected = "a";
>+  for (absl::string_view letter : std_splitter) {
>+    EXPECT_EQ(expected, letter);
>+    ++expected[0];
>+  }
>+  EXPECT_EQ("v", expected);
>+}
>+
>+template <typename T>
>+static std::unique_ptr<T> CopyToHeap(const T& value) {
>+  return std::unique_ptr<T>(new T(value));
>+}
>+
>+TEST(Split, LvalueCaptureIsCopyable) {
>+  std::string input = "a,b";
>+  auto heap_splitter = CopyToHeap(absl::StrSplit(input, ','));
>+  auto stack_splitter = *heap_splitter;
>+  heap_splitter.reset();
>+  std::vector<std::string> result = stack_splitter;
>+  EXPECT_THAT(result, testing::ElementsAre("a", "b"));
>+}
>+
>+TEST(Split, TemporaryCaptureIsCopyable) {
>+  auto heap_splitter = CopyToHeap(absl::StrSplit(std::string("a,b"), ','));
>+  auto stack_splitter = *heap_splitter;
>+  heap_splitter.reset();
>+  std::vector<std::string> result = stack_splitter;
>+  EXPECT_THAT(result, testing::ElementsAre("a", "b"));
>+}
>+
>+TEST(Split, SplitterIsCopyableAndMoveable) {
>+  auto a = absl::StrSplit("foo", '-');
>+
>+  // Ensures that the following expressions compile.
>+  auto b = a;             // Copy construct
>+  auto c = std::move(a);  // Move construct
>+  b = c;                  // Copy assign
>+  c = std::move(b);       // Move assign
>+
>+  EXPECT_THAT(c, ElementsAre("foo"));
>+}
>+
>+TEST(Split, StringDelimiter) {
>+  {
>+    std::vector<absl::string_view> v = absl::StrSplit("a,b", ',');
>+    EXPECT_THAT(v, ElementsAre("a", "b"));
>+  }
>+
>+  {
>+    std::vector<absl::string_view> v = absl::StrSplit("a,b", std::string(","));
>+    EXPECT_THAT(v, ElementsAre("a", "b"));
>+  }
>+
>+  {
>+    std::vector<absl::string_view> v =
>+        absl::StrSplit("a,b", absl::string_view(","));
>+    EXPECT_THAT(v, ElementsAre("a", "b"));
>+  }
>+}
>+
>+TEST(Split, UTF8) {
>+  // Tests splitting utf8 strings and utf8 delimiters.
>+  std::string utf8_string = u8"\u03BA\u1F79\u03C3\u03BC\u03B5";
>+  {
>+    // A utf8 input std::string with an ascii delimiter.
>+    std::string to_split = "a," + utf8_string;
>+    std::vector<absl::string_view> v = absl::StrSplit(to_split, ',');
>+    EXPECT_THAT(v, ElementsAre("a", utf8_string));
>+  }
>+
>+  {
>+    // A utf8 input std::string and a utf8 delimiter.
>+    std::string to_split = "a," + utf8_string + ",b";
>+    std::string unicode_delimiter = "," + utf8_string + ",";
>+    std::vector<absl::string_view> v =
>+        absl::StrSplit(to_split, unicode_delimiter);
>+    EXPECT_THAT(v, ElementsAre("a", "b"));
>+  }
>+
>+  {
>+    // A utf8 input std::string and ByAnyChar with ascii chars.
>+    std::vector<absl::string_view> v =
>+        absl::StrSplit(u8"Foo h\u00E4llo th\u4E1Ere", absl::ByAnyChar(" \t"));
>+    EXPECT_THAT(v, ElementsAre("Foo", u8"h\u00E4llo", u8"th\u4E1Ere"));
>+  }
>+}
>+
>+TEST(Split, EmptyStringDelimiter) {
>+  {
>+    std::vector<std::string> v = absl::StrSplit("", "");
>+    EXPECT_THAT(v, ElementsAre(""));
>+  }
>+
>+  {
>+    std::vector<std::string> v = absl::StrSplit("a", "");
>+    EXPECT_THAT(v, ElementsAre("a"));
>+  }
>+
>+  {
>+    std::vector<std::string> v = absl::StrSplit("ab", "");
>+    EXPECT_THAT(v, ElementsAre("a", "b"));
>+  }
>+
>+  {
>+    std::vector<std::string> v = absl::StrSplit("a b", "");
>+    EXPECT_THAT(v, ElementsAre("a", " ", "b"));
>+  }
>+}
>+
>+TEST(Split, SubstrDelimiter) {
>+  std::vector<absl::string_view> results;
>+  absl::string_view delim("//");
>+
>+  results = absl::StrSplit("", delim);
>+  EXPECT_THAT(results, ElementsAre(""));
>+
>+  results = absl::StrSplit("//", delim);
>+  EXPECT_THAT(results, ElementsAre("", ""));
>+
>+  results = absl::StrSplit("ab", delim);
>+  EXPECT_THAT(results, ElementsAre("ab"));
>+
>+  results = absl::StrSplit("ab//", delim);
>+  EXPECT_THAT(results, ElementsAre("ab", ""));
>+
>+  results = absl::StrSplit("ab/", delim);
>+  EXPECT_THAT(results, ElementsAre("ab/"));
>+
>+  results = absl::StrSplit("a/b", delim);
>+  EXPECT_THAT(results, ElementsAre("a/b"));
>+
>+  results = absl::StrSplit("a//b", delim);
>+  EXPECT_THAT(results, ElementsAre("a", "b"));
>+
>+  results = absl::StrSplit("a///b", delim);
>+  EXPECT_THAT(results, ElementsAre("a", "/b"));
>+
>+  results = absl::StrSplit("a////b", delim);
>+  EXPECT_THAT(results, ElementsAre("a", "", "b"));
>+}
>+
>+TEST(Split, EmptyResults) {
>+  std::vector<absl::string_view> results;
>+
>+  results = absl::StrSplit("", '#');
>+  EXPECT_THAT(results, ElementsAre(""));
>+
>+  results = absl::StrSplit("#", '#');
>+  EXPECT_THAT(results, ElementsAre("", ""));
>+
>+  results = absl::StrSplit("#cd", '#');
>+  EXPECT_THAT(results, ElementsAre("", "cd"));
>+
>+  results = absl::StrSplit("ab#cd#", '#');
>+  EXPECT_THAT(results, ElementsAre("ab", "cd", ""));
>+
>+  results = absl::StrSplit("ab##cd", '#');
>+  EXPECT_THAT(results, ElementsAre("ab", "", "cd"));
>+
>+  results = absl::StrSplit("ab##", '#');
>+  EXPECT_THAT(results, ElementsAre("ab", "", ""));
>+
>+  results = absl::StrSplit("ab#ab#", '#');
>+  EXPECT_THAT(results, ElementsAre("ab", "ab", ""));
>+
>+  results = absl::StrSplit("aaaa", 'a');
>+  EXPECT_THAT(results, ElementsAre("", "", "", "", ""));
>+
>+  results = absl::StrSplit("", '#', absl::SkipEmpty());
>+  EXPECT_THAT(results, ElementsAre());
>+}
>+
>+template <typename Delimiter>
>+static bool IsFoundAtStartingPos(absl::string_view text, Delimiter d,
>+                                 size_t starting_pos, int expected_pos) {
>+  absl::string_view found = d.Find(text, starting_pos);
>+  return found.data() != text.end() &&
>+         expected_pos == found.data() - text.data();
>+}
>+
>+// Helper function for testing Delimiter objects. Returns true if the given
>+// Delimiter is found in the given std::string at the given position. This function
>+// tests two cases:
>+//   1. The actual text given, staring at position 0
>+//   2. The text given with leading padding that should be ignored
>+template <typename Delimiter>
>+static bool IsFoundAt(absl::string_view text, Delimiter d, int expected_pos) {
>+  const std::string leading_text = ",x,y,z,";
>+  return IsFoundAtStartingPos(text, d, 0, expected_pos) &&
>+         IsFoundAtStartingPos(leading_text + std::string(text), d,
>+                              leading_text.length(),
>+                              expected_pos + leading_text.length());
>+}
>+
>+//
>+// Tests for Literal
>+//
>+
>+// Tests using any delimiter that represents a single comma.
>+template <typename Delimiter>
>+void TestComma(Delimiter d) {
>+  EXPECT_TRUE(IsFoundAt(",", d, 0));
>+  EXPECT_TRUE(IsFoundAt("a,", d, 1));
>+  EXPECT_TRUE(IsFoundAt(",b", d, 0));
>+  EXPECT_TRUE(IsFoundAt("a,b", d, 1));
>+  EXPECT_TRUE(IsFoundAt("a,b,", d, 1));
>+  EXPECT_TRUE(IsFoundAt("a,b,c", d, 1));
>+  EXPECT_FALSE(IsFoundAt("", d, -1));
>+  EXPECT_FALSE(IsFoundAt(" ", d, -1));
>+  EXPECT_FALSE(IsFoundAt("a", d, -1));
>+  EXPECT_FALSE(IsFoundAt("a b c", d, -1));
>+  EXPECT_FALSE(IsFoundAt("a;b;c", d, -1));
>+  EXPECT_FALSE(IsFoundAt(";", d, -1));
>+}
>+
>+TEST(Delimiter, Literal) {
>+  using absl::ByString;
>+  TestComma(ByString(","));
>+
>+  // Works as named variable.
>+  ByString comma_string(",");
>+  TestComma(comma_string);
>+
>+  // The first occurrence of empty std::string ("") in a std::string is at position 0.
>+  // There is a test below that demonstrates this for absl::string_view::find().
>+  // If the ByString delimiter returned position 0 for this, there would
>+  // be an infinite loop in the SplitIterator code. To avoid this, empty std::string
>+  // is a special case in that it always returns the item at position 1.
>+  absl::string_view abc("abc");
>+  EXPECT_EQ(0, abc.find(""));  // "" is found at position 0
>+  ByString empty("");
>+  EXPECT_FALSE(IsFoundAt("", empty, 0));
>+  EXPECT_FALSE(IsFoundAt("a", empty, 0));
>+  EXPECT_TRUE(IsFoundAt("ab", empty, 1));
>+  EXPECT_TRUE(IsFoundAt("abc", empty, 1));
>+}
>+
>+TEST(Split, ByChar) {
>+  using absl::ByChar;
>+  TestComma(ByChar(','));
>+
>+  // Works as named variable.
>+  ByChar comma_char(',');
>+  TestComma(comma_char);
>+}
>+
>+//
>+// Tests for ByAnyChar
>+//
>+
>+TEST(Delimiter, ByAnyChar) {
>+  using absl::ByAnyChar;
>+  ByAnyChar one_delim(",");
>+  // Found
>+  EXPECT_TRUE(IsFoundAt(",", one_delim, 0));
>+  EXPECT_TRUE(IsFoundAt("a,", one_delim, 1));
>+  EXPECT_TRUE(IsFoundAt("a,b", one_delim, 1));
>+  EXPECT_TRUE(IsFoundAt(",b", one_delim, 0));
>+  // Not found
>+  EXPECT_FALSE(IsFoundAt("", one_delim, -1));
>+  EXPECT_FALSE(IsFoundAt(" ", one_delim, -1));
>+  EXPECT_FALSE(IsFoundAt("a", one_delim, -1));
>+  EXPECT_FALSE(IsFoundAt("a;b;c", one_delim, -1));
>+  EXPECT_FALSE(IsFoundAt(";", one_delim, -1));
>+
>+  ByAnyChar two_delims(",;");
>+  // Found
>+  EXPECT_TRUE(IsFoundAt(",", two_delims, 0));
>+  EXPECT_TRUE(IsFoundAt(";", two_delims, 0));
>+  EXPECT_TRUE(IsFoundAt(",;", two_delims, 0));
>+  EXPECT_TRUE(IsFoundAt(";,", two_delims, 0));
>+  EXPECT_TRUE(IsFoundAt(",;b", two_delims, 0));
>+  EXPECT_TRUE(IsFoundAt(";,b", two_delims, 0));
>+  EXPECT_TRUE(IsFoundAt("a;,", two_delims, 1));
>+  EXPECT_TRUE(IsFoundAt("a,;", two_delims, 1));
>+  EXPECT_TRUE(IsFoundAt("a;,b", two_delims, 1));
>+  EXPECT_TRUE(IsFoundAt("a,;b", two_delims, 1));
>+  // Not found
>+  EXPECT_FALSE(IsFoundAt("", two_delims, -1));
>+  EXPECT_FALSE(IsFoundAt(" ", two_delims, -1));
>+  EXPECT_FALSE(IsFoundAt("a", two_delims, -1));
>+  EXPECT_FALSE(IsFoundAt("a=b=c", two_delims, -1));
>+  EXPECT_FALSE(IsFoundAt("=", two_delims, -1));
>+
>+  // ByAnyChar behaves just like ByString when given a delimiter of empty
>+  // std::string. That is, it always returns a zero-length absl::string_view
>+  // referring to the item at position 1, not position 0.
>+  ByAnyChar empty("");
>+  EXPECT_FALSE(IsFoundAt("", empty, 0));
>+  EXPECT_FALSE(IsFoundAt("a", empty, 0));
>+  EXPECT_TRUE(IsFoundAt("ab", empty, 1));
>+  EXPECT_TRUE(IsFoundAt("abc", empty, 1));
>+}
>+
>+//
>+// Tests for ByLength
>+//
>+
>+TEST(Delimiter, ByLength) {
>+  using absl::ByLength;
>+
>+  ByLength four_char_delim(4);
>+
>+  // Found
>+  EXPECT_TRUE(IsFoundAt("abcde", four_char_delim, 4));
>+  EXPECT_TRUE(IsFoundAt("abcdefghijklmnopqrstuvwxyz", four_char_delim, 4));
>+  EXPECT_TRUE(IsFoundAt("a b,c\nd", four_char_delim, 4));
>+  // Not found
>+  EXPECT_FALSE(IsFoundAt("", four_char_delim, 0));
>+  EXPECT_FALSE(IsFoundAt("a", four_char_delim, 0));
>+  EXPECT_FALSE(IsFoundAt("ab", four_char_delim, 0));
>+  EXPECT_FALSE(IsFoundAt("abc", four_char_delim, 0));
>+  EXPECT_FALSE(IsFoundAt("abcd", four_char_delim, 0));
>+}
>+
>+TEST(Split, WorksWithLargeStrings) {
>+  if (sizeof(size_t) > 4) {
>+    std::string s((uint32_t{1} << 31) + 1, 'x');  // 2G + 1 byte
>+    s.back() = '-';
>+    std::vector<absl::string_view> v = absl::StrSplit(s, '-');
>+    EXPECT_EQ(2, v.size());
>+    // The first element will contain 2G of 'x's.
>+    // testing::StartsWith is too slow with a 2G std::string.
>+    EXPECT_EQ('x', v[0][0]);
>+    EXPECT_EQ('x', v[0][1]);
>+    EXPECT_EQ('x', v[0][3]);
>+    EXPECT_EQ("", v[1]);
>+  }
>+}
>+
>+TEST(SplitInternalTest, TypeTraits) {
>+  EXPECT_FALSE(absl::strings_internal::HasMappedType<int>::value);
>+  EXPECT_TRUE(
>+      (absl::strings_internal::HasMappedType<std::map<int, int>>::value));
>+  EXPECT_FALSE(absl::strings_internal::HasValueType<int>::value);
>+  EXPECT_TRUE(
>+      (absl::strings_internal::HasValueType<std::map<int, int>>::value));
>+  EXPECT_FALSE(absl::strings_internal::HasConstIterator<int>::value);
>+  EXPECT_TRUE(
>+      (absl::strings_internal::HasConstIterator<std::map<int, int>>::value));
>+  EXPECT_FALSE(absl::strings_internal::IsInitializerList<int>::value);
>+  EXPECT_TRUE((absl::strings_internal::IsInitializerList<
>+               std::initializer_list<int>>::value));
>+}
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/string_view.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/string_view.cc
>new file mode 100644
>index 00000000000..4ceeb6bff58
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/string_view.cc
>@@ -0,0 +1,245 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/strings/string_view.h"
>+
>+#ifndef ABSL_HAVE_STD_STRING_VIEW
>+
>+#include <algorithm>
>+#include <climits>
>+#include <cstring>
>+#include <ostream>
>+
>+#include "absl/strings/internal/memutil.h"
>+
>+namespace absl {
>+
>+namespace {
>+void WritePadding(std::ostream& o, size_t pad) {
>+  char fill_buf[32];
>+  memset(fill_buf, o.fill(), sizeof(fill_buf));
>+  while (pad) {
>+    size_t n = std::min(pad, sizeof(fill_buf));
>+    o.write(fill_buf, n);
>+    pad -= n;
>+  }
>+}
>+
>+class LookupTable {
>+ public:
>+  // For each character in wanted, sets the index corresponding
>+  // to the ASCII code of that character. This is used by
>+  // the find_.*_of methods below to tell whether or not a character is in
>+  // the lookup table in constant time.
>+  explicit LookupTable(string_view wanted) {
>+    for (char c : wanted) {
>+      table_[Index(c)] = true;
>+    }
>+  }
>+  bool operator[](char c) const { return table_[Index(c)]; }
>+
>+ private:
>+  static unsigned char Index(char c) { return static_cast<unsigned char>(c); }
>+  bool table_[UCHAR_MAX + 1] = {};
>+};
>+
>+}  // namespace
>+
>+std::ostream& operator<<(std::ostream& o, string_view piece) {
>+  std::ostream::sentry sentry(o);
>+  if (sentry) {
>+    size_t lpad = 0;
>+    size_t rpad = 0;
>+    if (static_cast<size_t>(o.width()) > piece.size()) {
>+      size_t pad = o.width() - piece.size();
>+      if ((o.flags() & o.adjustfield) == o.left) {
>+        rpad = pad;
>+      } else {
>+        lpad = pad;
>+      }
>+    }
>+    if (lpad) WritePadding(o, lpad);
>+    o.write(piece.data(), piece.size());
>+    if (rpad) WritePadding(o, rpad);
>+    o.width(0);
>+  }
>+  return o;
>+}
>+
>+string_view::size_type string_view::copy(char* buf, size_type n,
>+                                         size_type pos) const {
>+  size_type ulen = length_;
>+  assert(pos <= ulen);
>+  size_type rlen = std::min(ulen - pos, n);
>+  if (rlen > 0) {
>+    const char* start = ptr_ + pos;
>+    std::copy(start, start + rlen, buf);
>+  }
>+  return rlen;
>+}
>+
>+string_view::size_type string_view::find(string_view s, size_type pos) const
>+    noexcept {
>+  if (empty() || pos > length_) {
>+    if (empty() && pos == 0 && s.empty()) return 0;
>+    return npos;
>+  }
>+  const char* result =
>+      strings_internal::memmatch(ptr_ + pos, length_ - pos, s.ptr_, s.length_);
>+  return result ? result - ptr_ : npos;
>+}
>+
>+string_view::size_type string_view::find(char c, size_type pos) const noexcept {
>+  if (empty() || pos >= length_) {
>+    return npos;
>+  }
>+  const char* result =
>+      static_cast<const char*>(memchr(ptr_ + pos, c, length_ - pos));
>+  return result != nullptr ? result - ptr_ : npos;
>+}
>+
>+string_view::size_type string_view::rfind(string_view s, size_type pos) const
>+    noexcept {
>+  if (length_ < s.length_) return npos;
>+  if (s.empty()) return std::min(length_, pos);
>+  const char* last = ptr_ + std::min(length_ - s.length_, pos) + s.length_;
>+  const char* result = std::find_end(ptr_, last, s.ptr_, s.ptr_ + s.length_);
>+  return result != last ? result - ptr_ : npos;
>+}
>+
>+// Search range is [0..pos] inclusive.  If pos == npos, search everything.
>+string_view::size_type string_view::rfind(char c, size_type pos) const
>+    noexcept {
>+  // Note: memrchr() is not available on Windows.
>+  if (empty()) return npos;
>+  for (size_type i = std::min(pos, length_ - 1);; --i) {
>+    if (ptr_[i] == c) {
>+      return i;
>+    }
>+    if (i == 0) break;
>+  }
>+  return npos;
>+}
>+
>+string_view::size_type string_view::find_first_of(string_view s,
>+                                                  size_type pos) const
>+    noexcept {
>+  if (empty() || s.empty()) {
>+    return npos;
>+  }
>+  // Avoid the cost of LookupTable() for a single-character search.
>+  if (s.length_ == 1) return find_first_of(s.ptr_[0], pos);
>+  LookupTable tbl(s);
>+  for (size_type i = pos; i < length_; ++i) {
>+    if (tbl[ptr_[i]]) {
>+      return i;
>+    }
>+  }
>+  return npos;
>+}
>+
>+string_view::size_type string_view::find_first_not_of(string_view s,
>+                                                      size_type pos) const
>+    noexcept {
>+  if (empty()) return npos;
>+  // Avoid the cost of LookupTable() for a single-character search.
>+  if (s.length_ == 1) return find_first_not_of(s.ptr_[0], pos);
>+  LookupTable tbl(s);
>+  for (size_type i = pos; i < length_; ++i) {
>+    if (!tbl[ptr_[i]]) {
>+      return i;
>+    }
>+  }
>+  return npos;
>+}
>+
>+string_view::size_type string_view::find_first_not_of(char c,
>+                                                      size_type pos) const
>+    noexcept {
>+  if (empty()) return npos;
>+  for (; pos < length_; ++pos) {
>+    if (ptr_[pos] != c) {
>+      return pos;
>+    }
>+  }
>+  return npos;
>+}
>+
>+string_view::size_type string_view::find_last_of(string_view s,
>+                                                 size_type pos) const noexcept {
>+  if (empty() || s.empty()) return npos;
>+  // Avoid the cost of LookupTable() for a single-character search.
>+  if (s.length_ == 1) return find_last_of(s.ptr_[0], pos);
>+  LookupTable tbl(s);
>+  for (size_type i = std::min(pos, length_ - 1);; --i) {
>+    if (tbl[ptr_[i]]) {
>+      return i;
>+    }
>+    if (i == 0) break;
>+  }
>+  return npos;
>+}
>+
>+string_view::size_type string_view::find_last_not_of(string_view s,
>+                                                     size_type pos) const
>+    noexcept {
>+  if (empty()) return npos;
>+  size_type i = std::min(pos, length_ - 1);
>+  if (s.empty()) return i;
>+  // Avoid the cost of LookupTable() for a single-character search.
>+  if (s.length_ == 1) return find_last_not_of(s.ptr_[0], pos);
>+  LookupTable tbl(s);
>+  for (;; --i) {
>+    if (!tbl[ptr_[i]]) {
>+      return i;
>+    }
>+    if (i == 0) break;
>+  }
>+  return npos;
>+}
>+
>+string_view::size_type string_view::find_last_not_of(char c,
>+                                                     size_type pos) const
>+    noexcept {
>+  if (empty()) return npos;
>+  size_type i = std::min(pos, length_ - 1);
>+  for (;; --i) {
>+    if (ptr_[i] != c) {
>+      return i;
>+    }
>+    if (i == 0) break;
>+  }
>+  return npos;
>+}
>+
>+// MSVC has non-standard behavior that implicitly creates definitions for static
>+// const members. These implicit definitions conflict with explicit out-of-class
>+// member definitions that are required by the C++ standard, resulting in
>+// LNK1169 "multiply defined" errors at link time. __declspec(selectany) asks
>+// MSVC to choose only one definition for the symbol it decorates. See details
>+// at http://msdn.microsoft.com/en-us/library/34h23df8(v=vs.100).aspx
>+#ifdef _MSC_VER
>+#define ABSL_STRING_VIEW_SELECTANY __declspec(selectany)
>+#else
>+#define ABSL_STRING_VIEW_SELECTANY
>+#endif
>+
>+ABSL_STRING_VIEW_SELECTANY
>+constexpr string_view::size_type string_view::npos;
>+ABSL_STRING_VIEW_SELECTANY
>+constexpr string_view::size_type string_view::kMaxSize;
>+
>+}  // namespace absl
>+
>+#endif  // ABSL_HAVE_STD_STRING_VIEW
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/string_view.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/string_view.h
>new file mode 100644
>index 00000000000..a7f9199240a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/string_view.h
>@@ -0,0 +1,570 @@
>+//
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// -----------------------------------------------------------------------------
>+// File: string_view.h
>+// -----------------------------------------------------------------------------
>+//
>+// This file contains the definition of the `absl::string_view` class. A
>+// `string_view` points to a contiguous span of characters, often part or all of
>+// another `std::string`, double-quoted std::string literal, character array, or even
>+// another `string_view`.
>+//
>+// This `absl::string_view` abstraction is designed to be a drop-in
>+// replacement for the C++17 `std::string_view` abstraction.
>+#ifndef ABSL_STRINGS_STRING_VIEW_H_
>+#define ABSL_STRINGS_STRING_VIEW_H_
>+
>+#include <algorithm>
>+#include "absl/base/config.h"
>+
>+#ifdef ABSL_HAVE_STD_STRING_VIEW
>+
>+#include <string_view>
>+
>+namespace absl {
>+using std::string_view;
>+}  // namespace absl
>+
>+#else  // ABSL_HAVE_STD_STRING_VIEW
>+
>+#include <cassert>
>+#include <cstddef>
>+#include <cstring>
>+#include <iosfwd>
>+#include <iterator>
>+#include <limits>
>+#include <string>
>+
>+#include "absl/base/internal/throw_delegate.h"
>+#include "absl/base/macros.h"
>+#include "absl/base/port.h"
>+
>+namespace absl {
>+
>+// absl::string_view
>+//
>+// A `string_view` provides a lightweight view into the std::string data provided by
>+// a `std::string`, double-quoted std::string literal, character array, or even
>+// another `string_view`. A `string_view` does *not* own the std::string to which it
>+// points, and that data cannot be modified through the view.
>+//
>+// You can use `string_view` as a function or method parameter anywhere a
>+// parameter can receive a double-quoted std::string literal, `const char*`,
>+// `std::string`, or another `absl::string_view` argument with no need to copy
>+// the std::string data. Systematic use of `string_view` within function arguments
>+// reduces data copies and `strlen()` calls.
>+//
>+// Because of its small size, prefer passing `string_view` by value:
>+//
>+//   void MyFunction(absl::string_view arg);
>+//
>+// If circumstances require, you may also pass one by const reference:
>+//
>+//   void MyFunction(const absl::string_view& arg);  // not preferred
>+//
>+// Passing by value generates slightly smaller code for many architectures.
>+//
>+// In either case, the source data of the `string_view` must outlive the
>+// `string_view` itself.
>+//
>+// A `string_view` is also suitable for local variables if you know that the
>+// lifetime of the underlying object is longer than the lifetime of your
>+// `string_view` variable. However, beware of binding a `string_view` to a
>+// temporary value:
>+//
>+//   // BAD use of string_view: lifetime problem
>+//   absl::string_view sv = obj.ReturnAString();
>+//
>+//   // GOOD use of string_view: str outlives sv
>+//   std::string str = obj.ReturnAString();
>+//   absl::string_view sv = str;
>+//
>+// Due to lifetime issues, a `string_view` is sometimes a poor choice for a
>+// return value and usually a poor choice for a data member. If you do use a
>+// `string_view` this way, it is your responsibility to ensure that the object
>+// pointed to by the `string_view` outlives the `string_view`.
>+//
>+// A `string_view` may represent a whole std::string or just part of a std::string. For
>+// example, when splitting a std::string, `std::vector<absl::string_view>` is a
>+// natural data type for the output.
>+//
>+//
>+// When constructed from a source which is nul-terminated, the `string_view`
>+// itself will not include the nul-terminator unless a specific size (including
>+// the nul) is passed to the constructor. As a result, common idioms that work
>+// on nul-terminated strings do not work on `string_view` objects. If you write
>+// code that scans a `string_view`, you must check its length rather than test
>+// for nul, for example. Note, however, that nuls may still be embedded within
>+// a `string_view` explicitly.
>+//
>+// You may create a null `string_view` in two ways:
>+//
>+//   absl::string_view sv();
>+//   absl::string_view sv(nullptr, 0);
>+//
>+// For the above, `sv.data() == nullptr`, `sv.length() == 0`, and
>+// `sv.empty() == true`. Also, if you create a `string_view` with a non-null
>+// pointer then `sv.data() != nullptr`. Thus, you can use `string_view()` to
>+// signal an undefined value that is different from other `string_view` values
>+// in a similar fashion to how `const char* p1 = nullptr;` is different from
>+// `const char* p2 = "";`. However, in practice, it is not recommended to rely
>+// on this behavior.
>+//
>+// Be careful not to confuse a null `string_view` with an empty one. A null
>+// `string_view` is an empty `string_view`, but some empty `string_view`s are
>+// not null. Prefer checking for emptiness over checking for null.
>+//
>+// There are many ways to create an empty string_view:
>+//
>+//   const char* nullcp = nullptr;
>+//   // string_view.size() will return 0 in all cases.
>+//   absl::string_view();
>+//   absl::string_view(nullcp, 0);
>+//   absl::string_view("");
>+//   absl::string_view("", 0);
>+//   absl::string_view("abcdef", 0);
>+//   absl::string_view("abcdef" + 6, 0);
>+//
>+// All empty `string_view` objects whether null or not, are equal:
>+//
>+//   absl::string_view() == absl::string_view("", 0)
>+//   absl::string_view(nullptr, 0) == absl:: string_view("abcdef"+6, 0)
>+class string_view {
>+ public:
>+  using traits_type = std::char_traits<char>;
>+  using value_type = char;
>+  using pointer = char*;
>+  using const_pointer = const char*;
>+  using reference = char&;
>+  using const_reference = const char&;
>+  using const_iterator = const char*;
>+  using iterator = const_iterator;
>+  using const_reverse_iterator = std::reverse_iterator<const_iterator>;
>+  using reverse_iterator = const_reverse_iterator;
>+  using size_type = size_t;
>+  using difference_type = std::ptrdiff_t;
>+
>+  static constexpr size_type npos = static_cast<size_type>(-1);
>+
>+  // Null `string_view` constructor
>+  constexpr string_view() noexcept : ptr_(nullptr), length_(0) {}
>+
>+  // Implicit constructors
>+
>+  template <typename Allocator>
>+  string_view(  // NOLINT(runtime/explicit)
>+      const std::basic_string<char, std::char_traits<char>, Allocator>&
>+          str) noexcept
>+      : ptr_(str.data()), length_(CheckLengthInternal(str.size())) {}
>+
>+  // Implicit constructor of a `string_view` from nul-terminated `str`. When
>+  // accepting possibly null strings, use `absl::NullSafeStringView(str)`
>+  // instead (see below).
>+  constexpr string_view(const char* str)  // NOLINT(runtime/explicit)
>+      : ptr_(str), length_(CheckLengthInternal(StrLenInternal(str))) {}
>+
>+  // Implicit constructor of a `string_view` from a `const char*` and length.
>+  constexpr string_view(const char* data, size_type len)
>+      : ptr_(data), length_(CheckLengthInternal(len)) {}
>+
>+  // NOTE: Harmlessly omitted to work around gdb bug.
>+  //   constexpr string_view(const string_view&) noexcept = default;
>+  //   string_view& operator=(const string_view&) noexcept = default;
>+
>+  // Iterators
>+
>+  // string_view::begin()
>+  //
>+  // Returns an iterator pointing to the first character at the beginning of the
>+  // `string_view`, or `end()` if the `string_view` is empty.
>+  constexpr const_iterator begin() const noexcept { return ptr_; }
>+
>+  // string_view::end()
>+  //
>+  // Returns an iterator pointing just beyond the last character at the end of
>+  // the `string_view`. This iterator acts as a placeholder; attempting to
>+  // access it results in undefined behavior.
>+  constexpr const_iterator end() const noexcept { return ptr_ + length_; }
>+
>+  // string_view::cbegin()
>+  //
>+  // Returns a const iterator pointing to the first character at the beginning
>+  // of the `string_view`, or `end()` if the `string_view` is empty.
>+  constexpr const_iterator cbegin() const noexcept { return begin(); }
>+
>+  // string_view::cend()
>+  //
>+  // Returns a const iterator pointing just beyond the last character at the end
>+  // of the `string_view`. This pointer acts as a placeholder; attempting to
>+  // access its element results in undefined behavior.
>+  constexpr const_iterator cend() const noexcept { return end(); }
>+
>+  // string_view::rbegin()
>+  //
>+  // Returns a reverse iterator pointing to the last character at the end of the
>+  // `string_view`, or `rend()` if the `string_view` is empty.
>+  const_reverse_iterator rbegin() const noexcept {
>+    return const_reverse_iterator(end());
>+  }
>+
>+  // string_view::rend()
>+  //
>+  // Returns a reverse iterator pointing just before the first character at the
>+  // beginning of the `string_view`. This pointer acts as a placeholder;
>+  // attempting to access its element results in undefined behavior.
>+  const_reverse_iterator rend() const noexcept {
>+    return const_reverse_iterator(begin());
>+  }
>+
>+  // string_view::crbegin()
>+  //
>+  // Returns a const reverse iterator pointing to the last character at the end
>+  // of the `string_view`, or `crend()` if the `string_view` is empty.
>+  const_reverse_iterator crbegin() const noexcept { return rbegin(); }
>+
>+  // string_view::crend()
>+  //
>+  // Returns a const reverse iterator pointing just before the first character
>+  // at the beginning of the `string_view`. This pointer acts as a placeholder;
>+  // attempting to access its element results in undefined behavior.
>+  const_reverse_iterator crend() const noexcept { return rend(); }
>+
>+  // Capacity Utilities
>+
>+  // string_view::size()
>+  //
>+  // Returns the number of characters in the `string_view`.
>+  constexpr size_type size() const noexcept {
>+    return length_;
>+  }
>+
>+  // string_view::length()
>+  //
>+  // Returns the number of characters in the `string_view`. Alias for `size()`.
>+  constexpr size_type length() const noexcept { return size(); }
>+
>+  // string_view::max_size()
>+  //
>+  // Returns the maximum number of characters the `string_view` can hold.
>+  constexpr size_type max_size() const noexcept { return kMaxSize; }
>+
>+  // string_view::empty()
>+  //
>+  // Checks if the `string_view` is empty (refers to no characters).
>+  constexpr bool empty() const noexcept { return length_ == 0; }
>+
>+  // std::string:view::operator[]
>+  //
>+  // Returns the ith element of an `string_view` using the array operator.
>+  // Note that this operator does not perform any bounds checking.
>+  constexpr const_reference operator[](size_type i) const { return ptr_[i]; }
>+
>+  // string_view::front()
>+  //
>+  // Returns the first element of a `string_view`.
>+  constexpr const_reference front() const { return ptr_[0]; }
>+
>+  // string_view::back()
>+  //
>+  // Returns the last element of a `string_view`.
>+  constexpr const_reference back() const { return ptr_[size() - 1]; }
>+
>+  // string_view::data()
>+  //
>+  // Returns a pointer to the underlying character array (which is of course
>+  // stored elsewhere). Note that `string_view::data()` may contain embedded nul
>+  // characters, but the returned buffer may or may not be nul-terminated;
>+  // therefore, do not pass `data()` to a routine that expects a nul-terminated
>+  // std::string.
>+  constexpr const_pointer data() const noexcept { return ptr_; }
>+
>+  // Modifiers
>+
>+  // string_view::remove_prefix()
>+  //
>+  // Removes the first `n` characters from the `string_view`. Note that the
>+  // underlying std::string is not changed, only the view.
>+  void remove_prefix(size_type n) {
>+    assert(n <= length_);
>+    ptr_ += n;
>+    length_ -= n;
>+  }
>+
>+  // string_view::remove_suffix()
>+  //
>+  // Removes the last `n` characters from the `string_view`. Note that the
>+  // underlying std::string is not changed, only the view.
>+  void remove_suffix(size_type n) {
>+    assert(n <= length_);
>+    length_ -= n;
>+  }
>+
>+  // string_view::swap()
>+  //
>+  // Swaps this `string_view` with another `string_view`.
>+  void swap(string_view& s) noexcept {
>+    auto t = *this;
>+    *this = s;
>+    s = t;
>+  }
>+
>+  // Explicit conversion operators
>+
>+  // Converts to `std::basic_string`.
>+  template <typename A>
>+  explicit operator std::basic_string<char, traits_type, A>() const {
>+    if (!data()) return {};
>+    return std::basic_string<char, traits_type, A>(data(), size());
>+  }
>+
>+  // string_view::copy()
>+  //
>+  // Copies the contents of the `string_view` at offset `pos` and length `n`
>+  // into `buf`.
>+  size_type copy(char* buf, size_type n, size_type pos = 0) const;
>+
>+  // string_view::substr()
>+  //
>+  // Returns a "substring" of the `string_view` (at offset `pos` and length
>+  // `n`) as another string_view. This function throws `std::out_of_bounds` if
>+  // `pos > size'.
>+  string_view substr(size_type pos, size_type n = npos) const {
>+    if (ABSL_PREDICT_FALSE(pos > length_))
>+      base_internal::ThrowStdOutOfRange("absl::string_view::substr");
>+    n = std::min(n, length_ - pos);
>+    return string_view(ptr_ + pos, n);
>+  }
>+
>+  // string_view::compare()
>+  //
>+  // Performs a lexicographical comparison between the `string_view` and
>+  // another `absl::string_view), returning -1 if `this` is less than, 0 if
>+  // `this` is equal to, and 1 if `this` is greater than the passed std::string
>+  // view. Note that in the case of data equality, a further comparison is made
>+  // on the respective sizes of the two `string_view`s to determine which is
>+  // smaller, equal, or greater.
>+  int compare(string_view x) const noexcept {
>+    auto min_length = std::min(length_, x.length_);
>+    if (min_length > 0) {
>+      int r = memcmp(ptr_, x.ptr_, min_length);
>+      if (r < 0) return -1;
>+      if (r > 0) return 1;
>+    }
>+    if (length_ < x.length_) return -1;
>+    if (length_ > x.length_) return 1;
>+    return 0;
>+  }
>+
>+  // Overload of `string_view::compare()` for comparing a substring of the
>+  // 'string_view` and another `absl::string_view`.
>+  int compare(size_type pos1, size_type count1, string_view v) const {
>+    return substr(pos1, count1).compare(v);
>+  }
>+
>+  // Overload of `string_view::compare()` for comparing a substring of the
>+  // `string_view` and a substring of another `absl::string_view`.
>+  int compare(size_type pos1, size_type count1, string_view v, size_type pos2,
>+              size_type count2) const {
>+    return substr(pos1, count1).compare(v.substr(pos2, count2));
>+  }
>+
>+  // Overload of `string_view::compare()` for comparing a `string_view` and a
>+  // a different  C-style std::string `s`.
>+  int compare(const char* s) const { return compare(string_view(s)); }
>+
>+  // Overload of `string_view::compare()` for comparing a substring of the
>+  // `string_view` and a different std::string C-style std::string `s`.
>+  int compare(size_type pos1, size_type count1, const char* s) const {
>+    return substr(pos1, count1).compare(string_view(s));
>+  }
>+
>+  // Overload of `string_view::compare()` for comparing a substring of the
>+  // `string_view` and a substring of a different C-style std::string `s`.
>+  int compare(size_type pos1, size_type count1, const char* s,
>+              size_type count2) const {
>+    return substr(pos1, count1).compare(string_view(s, count2));
>+  }
>+
>+  // Find Utilities
>+
>+  // string_view::find()
>+  //
>+  // Finds the first occurrence of the substring `s` within the `string_view`,
>+  // returning the position of the first character's match, or `npos` if no
>+  // match was found.
>+  size_type find(string_view s, size_type pos = 0) const noexcept;
>+
>+  // Overload of `string_view::find()` for finding the given character `c`
>+  // within the `string_view`.
>+  size_type find(char c, size_type pos = 0) const noexcept;
>+
>+  // string_view::rfind()
>+  //
>+  // Finds the last occurrence of a substring `s` within the `string_view`,
>+  // returning the position of the first character's match, or `npos` if no
>+  // match was found.
>+  size_type rfind(string_view s, size_type pos = npos) const
>+      noexcept;
>+
>+  // Overload of `string_view::rfind()` for finding the last given character `c`
>+  // within the `string_view`.
>+  size_type rfind(char c, size_type pos = npos) const noexcept;
>+
>+  // string_view::find_first_of()
>+  //
>+  // Finds the first occurrence of any of the characters in `s` within the
>+  // `string_view`, returning the start position of the match, or `npos` if no
>+  // match was found.
>+  size_type find_first_of(string_view s, size_type pos = 0) const
>+      noexcept;
>+
>+  // Overload of `string_view::find_first_of()` for finding a character `c`
>+  // within the `string_view`.
>+  size_type find_first_of(char c, size_type pos = 0) const
>+      noexcept {
>+    return find(c, pos);
>+  }
>+
>+  // string_view::find_last_of()
>+  //
>+  // Finds the last occurrence of any of the characters in `s` within the
>+  // `string_view`, returning the start position of the match, or `npos` if no
>+  // match was found.
>+  size_type find_last_of(string_view s, size_type pos = npos) const
>+      noexcept;
>+
>+  // Overload of `string_view::find_last_of()` for finding a character `c`
>+  // within the `string_view`.
>+  size_type find_last_of(char c, size_type pos = npos) const
>+      noexcept {
>+    return rfind(c, pos);
>+  }
>+
>+  // string_view::find_first_not_of()
>+  //
>+  // Finds the first occurrence of any of the characters not in `s` within the
>+  // `string_view`, returning the start position of the first non-match, or
>+  // `npos` if no non-match was found.
>+  size_type find_first_not_of(string_view s, size_type pos = 0) const noexcept;
>+
>+  // Overload of `string_view::find_first_not_of()` for finding a character
>+  // that is not `c` within the `string_view`.
>+  size_type find_first_not_of(char c, size_type pos = 0) const noexcept;
>+
>+  // string_view::find_last_not_of()
>+  //
>+  // Finds the last occurrence of any of the characters not in `s` within the
>+  // `string_view`, returning the start position of the last non-match, or
>+  // `npos` if no non-match was found.
>+  size_type find_last_not_of(string_view s,
>+                                          size_type pos = npos) const noexcept;
>+
>+  // Overload of `string_view::find_last_not_of()` for finding a character
>+  // that is not `c` within the `string_view`.
>+  size_type find_last_not_of(char c, size_type pos = npos) const
>+      noexcept;
>+
>+ private:
>+  static constexpr size_type kMaxSize =
>+      std::numeric_limits<difference_type>::max();
>+
>+  // check whether __builtin_strlen is provided by the compiler.
>+  // GCC doesn't have __has_builtin()
>+  // (https://gcc.gnu.org/bugzilla/show_bug.cgi?id=66970),
>+  // but has __builtin_strlen according to
>+  // https://gcc.gnu.org/onlinedocs/gcc-4.7.0/gcc/Other-Builtins.html.
>+#if ABSL_HAVE_BUILTIN(__builtin_strlen) || \
>+    (defined(__GNUC__) && !defined(__clang__))
>+  static constexpr size_type StrLenInternal(const char* str) {
>+    return str ? __builtin_strlen(str) : 0;
>+  }
>+#else
>+  static constexpr size_type StrLenInternal(const char* str) {
>+    return str ? strlen(str) : 0;
>+  }
>+#endif
>+
>+  static constexpr size_type CheckLengthInternal(size_type len) {
>+    return ABSL_ASSERT(len <= kMaxSize), len;
>+  }
>+
>+  const char* ptr_;
>+  size_type length_;
>+};
>+
>+// This large function is defined inline so that in a fairly common case where
>+// one of the arguments is a literal, the compiler can elide a lot of the
>+// following comparisons.
>+inline bool operator==(string_view x, string_view y) noexcept {
>+  auto len = x.size();
>+  if (len != y.size()) {
>+    return false;
>+  }
>+  return x.data() == y.data() || len <= 0 ||
>+         memcmp(x.data(), y.data(), len) == 0;
>+}
>+
>+inline bool operator!=(string_view x, string_view y) noexcept {
>+  return !(x == y);
>+}
>+
>+inline bool operator<(string_view x, string_view y) noexcept {
>+  auto min_size = std::min(x.size(), y.size());
>+  const int r = min_size == 0 ? 0 : memcmp(x.data(), y.data(), min_size);
>+  return (r < 0) || (r == 0 && x.size() < y.size());
>+}
>+
>+inline bool operator>(string_view x, string_view y) noexcept { return y < x; }
>+
>+inline bool operator<=(string_view x, string_view y) noexcept {
>+  return !(y < x);
>+}
>+
>+inline bool operator>=(string_view x, string_view y) noexcept {
>+  return !(x < y);
>+}
>+
>+// IO Insertion Operator
>+std::ostream& operator<<(std::ostream& o, string_view piece);
>+
>+}  // namespace absl
>+
>+#endif  // ABSL_HAVE_STD_STRING_VIEW
>+
>+namespace absl {
>+
>+// ClippedSubstr()
>+//
>+// Like `s.substr(pos, n)`, but clips `pos` to an upper bound of `s.size()`.
>+// Provided because std::string_view::substr throws if `pos > size()`
>+inline string_view ClippedSubstr(string_view s, size_t pos,
>+                                 size_t n = string_view::npos) {
>+  pos = std::min(pos, static_cast<size_t>(s.size()));
>+  return s.substr(pos, n);
>+}
>+
>+// NullSafeStringView()
>+//
>+// Creates an `absl::string_view` from a pointer `p` even if it's null-valued.
>+// This function should be used where an `absl::string_view` can be created from
>+// a possibly-null pointer.
>+inline string_view NullSafeStringView(const char* p) {
>+  return p ? string_view(p) : string_view();
>+}
>+
>+}  // namespace absl
>+
>+#endif  // ABSL_STRINGS_STRING_VIEW_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/string_view_benchmark.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/string_view_benchmark.cc
>new file mode 100644
>index 00000000000..fb46db18b3c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/string_view_benchmark.cc
>@@ -0,0 +1,329 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/strings/string_view.h"
>+
>+#include <algorithm>
>+#include <cstdint>
>+#include <map>
>+#include <random>
>+#include <string>
>+#include <unordered_set>
>+#include <vector>
>+
>+#include "benchmark/benchmark.h"
>+#include "absl/base/attributes.h"
>+#include "absl/base/internal/raw_logging.h"
>+#include "absl/base/macros.h"
>+#include "absl/strings/str_cat.h"
>+
>+namespace {
>+
>+// Provide a forcibly out-of-line wrapper for operator== that can be used in
>+// benchmarks to measure the impact of inlining.
>+ABSL_ATTRIBUTE_NOINLINE
>+bool NonInlinedEq(absl::string_view a, absl::string_view b) { return a == b; }
>+
>+// We use functions that cannot be inlined to perform the comparison loops so
>+// that inlining of the operator== can't optimize away *everything*.
>+ABSL_ATTRIBUTE_NOINLINE
>+void DoEqualityComparisons(benchmark::State& state, absl::string_view a,
>+                           absl::string_view b) {
>+  for (auto _ : state) {
>+    benchmark::DoNotOptimize(a == b);
>+  }
>+}
>+
>+void BM_EqualIdentical(benchmark::State& state) {
>+  std::string x(state.range(0), 'a');
>+  DoEqualityComparisons(state, x, x);
>+}
>+BENCHMARK(BM_EqualIdentical)->DenseRange(0, 3)->Range(4, 1 << 10);
>+
>+void BM_EqualSame(benchmark::State& state) {
>+  std::string x(state.range(0), 'a');
>+  std::string y = x;
>+  DoEqualityComparisons(state, x, y);
>+}
>+BENCHMARK(BM_EqualSame)
>+    ->DenseRange(0, 10)
>+    ->Arg(20)
>+    ->Arg(40)
>+    ->Arg(70)
>+    ->Arg(110)
>+    ->Range(160, 4096);
>+
>+void BM_EqualDifferent(benchmark::State& state) {
>+  const int len = state.range(0);
>+  std::string x(len, 'a');
>+  std::string y = x;
>+  if (len > 0) {
>+    y[len - 1] = 'b';
>+  }
>+  DoEqualityComparisons(state, x, y);
>+}
>+BENCHMARK(BM_EqualDifferent)->DenseRange(0, 3)->Range(4, 1 << 10);
>+
>+// This benchmark is intended to check that important simplifications can be
>+// made with absl::string_view comparisons against constant strings. The idea is
>+// that if constant strings cause redundant components of the comparison, the
>+// compiler should detect and eliminate them. Here we use 8 different strings,
>+// each with the same size. Provided our comparison makes the implementation
>+// inline-able by the compiler, it should fold all of these away into a single
>+// size check once per loop iteration.
>+ABSL_ATTRIBUTE_NOINLINE
>+void DoConstantSizeInlinedEqualityComparisons(benchmark::State& state,
>+                                              absl::string_view a) {
>+  for (auto _ : state) {
>+    benchmark::DoNotOptimize(a == "aaa");
>+    benchmark::DoNotOptimize(a == "bbb");
>+    benchmark::DoNotOptimize(a == "ccc");
>+    benchmark::DoNotOptimize(a == "ddd");
>+    benchmark::DoNotOptimize(a == "eee");
>+    benchmark::DoNotOptimize(a == "fff");
>+    benchmark::DoNotOptimize(a == "ggg");
>+    benchmark::DoNotOptimize(a == "hhh");
>+  }
>+}
>+void BM_EqualConstantSizeInlined(benchmark::State& state) {
>+  std::string x(state.range(0), 'a');
>+  DoConstantSizeInlinedEqualityComparisons(state, x);
>+}
>+// We only need to check for size of 3, and <> 3 as this benchmark only has to
>+// do with size differences.
>+BENCHMARK(BM_EqualConstantSizeInlined)->DenseRange(2, 4);
>+
>+// This benchmark exists purely to give context to the above timings: this is
>+// what they would look like if the compiler is completely unable to simplify
>+// between two comparisons when they are comparing against constant strings.
>+ABSL_ATTRIBUTE_NOINLINE
>+void DoConstantSizeNonInlinedEqualityComparisons(benchmark::State& state,
>+                                                 absl::string_view a) {
>+  for (auto _ : state) {
>+    // Force these out-of-line to compare with the above function.
>+    benchmark::DoNotOptimize(NonInlinedEq(a, "aaa"));
>+    benchmark::DoNotOptimize(NonInlinedEq(a, "bbb"));
>+    benchmark::DoNotOptimize(NonInlinedEq(a, "ccc"));
>+    benchmark::DoNotOptimize(NonInlinedEq(a, "ddd"));
>+    benchmark::DoNotOptimize(NonInlinedEq(a, "eee"));
>+    benchmark::DoNotOptimize(NonInlinedEq(a, "fff"));
>+    benchmark::DoNotOptimize(NonInlinedEq(a, "ggg"));
>+    benchmark::DoNotOptimize(NonInlinedEq(a, "hhh"));
>+  }
>+}
>+
>+void BM_EqualConstantSizeNonInlined(benchmark::State& state) {
>+  std::string x(state.range(0), 'a');
>+  DoConstantSizeNonInlinedEqualityComparisons(state, x);
>+}
>+// We only need to check for size of 3, and <> 3 as this benchmark only has to
>+// do with size differences.
>+BENCHMARK(BM_EqualConstantSizeNonInlined)->DenseRange(2, 4);
>+
>+void BM_CompareSame(benchmark::State& state) {
>+  const int len = state.range(0);
>+  std::string x;
>+  for (int i = 0; i < len; i++) {
>+    x += 'a';
>+  }
>+  std::string y = x;
>+  absl::string_view a = x;
>+  absl::string_view b = y;
>+
>+  for (auto _ : state) {
>+    benchmark::DoNotOptimize(a.compare(b));
>+  }
>+}
>+BENCHMARK(BM_CompareSame)->DenseRange(0, 3)->Range(4, 1 << 10);
>+
>+void BM_find_string_view_len_one(benchmark::State& state) {
>+  std::string haystack(state.range(0), '0');
>+  absl::string_view s(haystack);
>+  for (auto _ : state) {
>+    s.find("x");  // not present; length 1
>+  }
>+}
>+BENCHMARK(BM_find_string_view_len_one)->Range(1, 1 << 20);
>+
>+void BM_find_string_view_len_two(benchmark::State& state) {
>+  std::string haystack(state.range(0), '0');
>+  absl::string_view s(haystack);
>+  for (auto _ : state) {
>+    s.find("xx");  // not present; length 2
>+  }
>+}
>+BENCHMARK(BM_find_string_view_len_two)->Range(1, 1 << 20);
>+
>+void BM_find_one_char(benchmark::State& state) {
>+  std::string haystack(state.range(0), '0');
>+  absl::string_view s(haystack);
>+  for (auto _ : state) {
>+    s.find('x');  // not present
>+  }
>+}
>+BENCHMARK(BM_find_one_char)->Range(1, 1 << 20);
>+
>+void BM_rfind_one_char(benchmark::State& state) {
>+  std::string haystack(state.range(0), '0');
>+  absl::string_view s(haystack);
>+  for (auto _ : state) {
>+    s.rfind('x');  // not present
>+  }
>+}
>+BENCHMARK(BM_rfind_one_char)->Range(1, 1 << 20);
>+
>+void BM_worst_case_find_first_of(benchmark::State& state, int haystack_len) {
>+  const int needle_len = state.range(0);
>+  std::string needle;
>+  for (int i = 0; i < needle_len; ++i) {
>+    needle += 'a' + i;
>+  }
>+  std::string haystack(haystack_len, '0');  // 1000 zeros.
>+
>+  absl::string_view s(haystack);
>+  for (auto _ : state) {
>+    s.find_first_of(needle);
>+  }
>+}
>+
>+void BM_find_first_of_short(benchmark::State& state) {
>+  BM_worst_case_find_first_of(state, 10);
>+}
>+
>+void BM_find_first_of_medium(benchmark::State& state) {
>+  BM_worst_case_find_first_of(state, 100);
>+}
>+
>+void BM_find_first_of_long(benchmark::State& state) {
>+  BM_worst_case_find_first_of(state, 1000);
>+}
>+
>+BENCHMARK(BM_find_first_of_short)->DenseRange(0, 4)->Arg(8)->Arg(16)->Arg(32);
>+BENCHMARK(BM_find_first_of_medium)->DenseRange(0, 4)->Arg(8)->Arg(16)->Arg(32);
>+BENCHMARK(BM_find_first_of_long)->DenseRange(0, 4)->Arg(8)->Arg(16)->Arg(32);
>+
>+struct EasyMap : public std::map<absl::string_view, uint64_t> {
>+  explicit EasyMap(size_t) {}
>+};
>+
>+// This templated benchmark helper function is intended to stress operator== or
>+// operator< in a realistic test.  It surely isn't entirely realistic, but it's
>+// a start.  The test creates a map of type Map, a template arg, and populates
>+// it with table_size key/value pairs. Each key has WordsPerKey words.  After
>+// creating the map, a number of lookups are done in random order.  Some keys
>+// are used much more frequently than others in this phase of the test.
>+template <typename Map, int WordsPerKey>
>+void StringViewMapBenchmark(benchmark::State& state) {
>+  const int table_size = state.range(0);
>+  const double kFractionOfKeysThatAreHot = 0.2;
>+  const int kNumLookupsOfHotKeys = 20;
>+  const int kNumLookupsOfColdKeys = 1;
>+  const char* words[] = {"the",   "quick",  "brown",    "fox",      "jumped",
>+                         "over",  "the",    "lazy",     "dog",      "and",
>+                         "found", "a",      "large",    "mushroom", "and",
>+                         "a",     "couple", "crickets", "eating",   "pie"};
>+  // Create some keys that consist of words in random order.
>+  std::random_device r;
>+  std::seed_seq seed({r(), r(), r(), r(), r(), r(), r(), r()});
>+  std::mt19937 rng(seed);
>+  std::vector<std::string> keys(table_size);
>+  std::vector<int> all_indices;
>+  const int kBlockSize = 1 << 12;
>+  std::unordered_set<std::string> t(kBlockSize);
>+  std::uniform_int_distribution<int> uniform(0, ABSL_ARRAYSIZE(words) - 1);
>+  for (int i = 0; i < table_size; i++) {
>+    all_indices.push_back(i);
>+    do {
>+      keys[i].clear();
>+      for (int j = 0; j < WordsPerKey; j++) {
>+        absl::StrAppend(&keys[i], j > 0 ? " " : "", words[uniform(rng)]);
>+      }
>+    } while (!t.insert(keys[i]).second);
>+  }
>+
>+  // Create a list of strings to lookup: a permutation of the array of
>+  // keys we just created, with repeats.  "Hot" keys get repeated more.
>+  std::shuffle(all_indices.begin(), all_indices.end(), rng);
>+  const int num_hot = table_size * kFractionOfKeysThatAreHot;
>+  const int num_cold = table_size - num_hot;
>+  std::vector<int> hot_indices(all_indices.begin(),
>+                               all_indices.begin() + num_hot);
>+  std::vector<int> indices;
>+  for (int i = 0; i < kNumLookupsOfColdKeys; i++) {
>+    indices.insert(indices.end(), all_indices.begin(), all_indices.end());
>+  }
>+  for (int i = 0; i < kNumLookupsOfHotKeys - kNumLookupsOfColdKeys; i++) {
>+    indices.insert(indices.end(), hot_indices.begin(), hot_indices.end());
>+  }
>+  std::shuffle(indices.begin(), indices.end(), rng);
>+  ABSL_RAW_CHECK(
>+      num_cold * kNumLookupsOfColdKeys + num_hot * kNumLookupsOfHotKeys ==
>+          indices.size(),
>+      "");
>+  // After constructing the array we probe it with absl::string_views built from
>+  // test_strings.  This means operator== won't see equal pointers, so
>+  // it'll have to check for equal lengths and equal characters.
>+  std::vector<std::string> test_strings(indices.size());
>+  for (int i = 0; i < indices.size(); i++) {
>+    test_strings[i] = keys[indices[i]];
>+  }
>+
>+  // Run the benchmark. It includes map construction but is mostly
>+  // map lookups.
>+  for (auto _ : state) {
>+    Map h(table_size);
>+    for (int i = 0; i < table_size; i++) {
>+      h[keys[i]] = i * 2;
>+    }
>+    ABSL_RAW_CHECK(h.size() == table_size, "");
>+    uint64_t sum = 0;
>+    for (int i = 0; i < indices.size(); i++) {
>+      sum += h[test_strings[i]];
>+    }
>+    benchmark::DoNotOptimize(sum);
>+  }
>+}
>+
>+void BM_StdMap_4(benchmark::State& state) {
>+  StringViewMapBenchmark<EasyMap, 4>(state);
>+}
>+BENCHMARK(BM_StdMap_4)->Range(1 << 10, 1 << 16);
>+
>+void BM_StdMap_8(benchmark::State& state) {
>+  StringViewMapBenchmark<EasyMap, 8>(state);
>+}
>+BENCHMARK(BM_StdMap_8)->Range(1 << 10, 1 << 16);
>+
>+void BM_CopyToStringNative(benchmark::State& state) {
>+  std::string src(state.range(0), 'x');
>+  absl::string_view sv(src);
>+  std::string dst;
>+  for (auto _ : state) {
>+    dst.assign(sv.begin(), sv.end());
>+  }
>+}
>+BENCHMARK(BM_CopyToStringNative)->Range(1 << 3, 1 << 12);
>+
>+void BM_AppendToStringNative(benchmark::State& state) {
>+  std::string src(state.range(0), 'x');
>+  absl::string_view sv(src);
>+  std::string dst;
>+  for (auto _ : state) {
>+    dst.clear();
>+    dst.insert(dst.end(), sv.begin(), sv.end());
>+  }
>+}
>+BENCHMARK(BM_AppendToStringNative)->Range(1 << 3, 1 << 12);
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/string_view_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/string_view_test.cc
>new file mode 100644
>index 00000000000..b19d07c7fd6
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/string_view_test.cc
>@@ -0,0 +1,1146 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/strings/string_view.h"
>+
>+#include <stdlib.h>
>+#include <iomanip>
>+#include <iterator>
>+#include <limits>
>+#include <map>
>+#include <sstream>
>+#include <stdexcept>
>+#include <string>
>+#include <type_traits>
>+#include <utility>
>+
>+#include "gtest/gtest.h"
>+#include "absl/base/config.h"
>+#include "absl/base/dynamic_annotations.h"
>+
>+#ifdef __ANDROID__
>+// Android assert messages only go to system log, so death tests cannot inspect
>+// the message for matching.
>+#define ABSL_EXPECT_DEATH_IF_SUPPORTED(statement, regex) \
>+  EXPECT_DEATH_IF_SUPPORTED(statement, ".*")
>+#else
>+#define ABSL_EXPECT_DEATH_IF_SUPPORTED EXPECT_DEATH_IF_SUPPORTED
>+#endif
>+
>+namespace {
>+
>+// A minimal allocator that uses malloc().
>+template <typename T>
>+struct Mallocator {
>+  typedef T value_type;
>+  typedef size_t size_type;
>+  typedef ptrdiff_t difference_type;
>+  typedef T* pointer;
>+  typedef const T* const_pointer;
>+  typedef T& reference;
>+  typedef const T& const_reference;
>+
>+  size_type max_size() const {
>+    return size_t(std::numeric_limits<size_type>::max()) / sizeof(value_type);
>+  }
>+  template <typename U>
>+  struct rebind {
>+    typedef Mallocator<U> other;
>+  };
>+  Mallocator() = default;
>+  template <class U>
>+  Mallocator(const Mallocator<U>&) {}  // NOLINT(runtime/explicit)
>+
>+  T* allocate(size_t n) { return static_cast<T*>(std::malloc(n * sizeof(T))); }
>+  void deallocate(T* p, size_t) { std::free(p); }
>+};
>+template <typename T, typename U>
>+bool operator==(const Mallocator<T>&, const Mallocator<U>&) {
>+  return true;
>+}
>+template <typename T, typename U>
>+bool operator!=(const Mallocator<T>&, const Mallocator<U>&) {
>+  return false;
>+}
>+
>+TEST(StringViewTest, Ctor) {
>+  {
>+    // Null.
>+    absl::string_view s10;
>+    EXPECT_TRUE(s10.data() == nullptr);
>+    EXPECT_EQ(0, s10.length());
>+  }
>+
>+  {
>+    // const char* without length.
>+    const char* hello = "hello";
>+    absl::string_view s20(hello);
>+    EXPECT_TRUE(s20.data() == hello);
>+    EXPECT_EQ(5, s20.length());
>+
>+    // const char* with length.
>+    absl::string_view s21(hello, 4);
>+    EXPECT_TRUE(s21.data() == hello);
>+    EXPECT_EQ(4, s21.length());
>+
>+    // Not recommended, but valid C++
>+    absl::string_view s22(hello, 6);
>+    EXPECT_TRUE(s22.data() == hello);
>+    EXPECT_EQ(6, s22.length());
>+  }
>+
>+  {
>+    // std::string.
>+    std::string hola = "hola";
>+    absl::string_view s30(hola);
>+    EXPECT_TRUE(s30.data() == hola.data());
>+    EXPECT_EQ(4, s30.length());
>+
>+    // std::string with embedded '\0'.
>+    hola.push_back('\0');
>+    hola.append("h2");
>+    hola.push_back('\0');
>+    absl::string_view s31(hola);
>+    EXPECT_TRUE(s31.data() == hola.data());
>+    EXPECT_EQ(8, s31.length());
>+  }
>+
>+  {
>+    using mstring =
>+        std::basic_string<char, std::char_traits<char>, Mallocator<char>>;
>+    mstring str1("BUNGIE-JUMPING!");
>+    const mstring str2("SLEEPING!");
>+
>+    absl::string_view s1(str1);
>+    s1.remove_prefix(strlen("BUNGIE-JUM"));
>+
>+    absl::string_view s2(str2);
>+    s2.remove_prefix(strlen("SLEE"));
>+
>+    EXPECT_EQ(s1, s2);
>+    EXPECT_EQ(s1, "PING!");
>+  }
>+
>+  // TODO(mec): absl::string_view(const absl::string_view&);
>+}
>+
>+TEST(StringViewTest, Swap) {
>+  absl::string_view a("a");
>+  absl::string_view b("bbb");
>+  EXPECT_TRUE(noexcept(a.swap(b)));
>+  a.swap(b);
>+  EXPECT_EQ(a, "bbb");
>+  EXPECT_EQ(b, "a");
>+  a.swap(b);
>+  EXPECT_EQ(a, "a");
>+  EXPECT_EQ(b, "bbb");
>+}
>+
>+TEST(StringViewTest, STLComparator) {
>+  std::string s1("foo");
>+  std::string s2("bar");
>+  std::string s3("baz");
>+
>+  absl::string_view p1(s1);
>+  absl::string_view p2(s2);
>+  absl::string_view p3(s3);
>+
>+  typedef std::map<absl::string_view, int> TestMap;
>+  TestMap map;
>+
>+  map.insert(std::make_pair(p1, 0));
>+  map.insert(std::make_pair(p2, 1));
>+  map.insert(std::make_pair(p3, 2));
>+  EXPECT_EQ(map.size(), 3);
>+
>+  TestMap::const_iterator iter = map.begin();
>+  EXPECT_EQ(iter->second, 1);
>+  ++iter;
>+  EXPECT_EQ(iter->second, 2);
>+  ++iter;
>+  EXPECT_EQ(iter->second, 0);
>+  ++iter;
>+  EXPECT_TRUE(iter == map.end());
>+
>+  TestMap::iterator new_iter = map.find("zot");
>+  EXPECT_TRUE(new_iter == map.end());
>+
>+  new_iter = map.find("bar");
>+  EXPECT_TRUE(new_iter != map.end());
>+
>+  map.erase(new_iter);
>+  EXPECT_EQ(map.size(), 2);
>+
>+  iter = map.begin();
>+  EXPECT_EQ(iter->second, 2);
>+  ++iter;
>+  EXPECT_EQ(iter->second, 0);
>+  ++iter;
>+  EXPECT_TRUE(iter == map.end());
>+}
>+
>+#define COMPARE(result, op, x, y)                                      \
>+  EXPECT_EQ(result, absl::string_view((x)) op absl::string_view((y))); \
>+  EXPECT_EQ(result, absl::string_view((x)).compare(absl::string_view((y))) op 0)
>+
>+TEST(StringViewTest, ComparisonOperators) {
>+  COMPARE(true, ==, "",   "");
>+  COMPARE(true, ==, "", absl::string_view());
>+  COMPARE(true, ==, absl::string_view(), "");
>+  COMPARE(true, ==, "a",  "a");
>+  COMPARE(true, ==, "aa", "aa");
>+  COMPARE(false, ==, "a",  "");
>+  COMPARE(false, ==, "",   "a");
>+  COMPARE(false, ==, "a",  "b");
>+  COMPARE(false, ==, "a",  "aa");
>+  COMPARE(false, ==, "aa", "a");
>+
>+  COMPARE(false, !=, "",   "");
>+  COMPARE(false, !=, "a",  "a");
>+  COMPARE(false, !=, "aa", "aa");
>+  COMPARE(true, !=, "a",  "");
>+  COMPARE(true, !=, "",   "a");
>+  COMPARE(true, !=, "a",  "b");
>+  COMPARE(true, !=, "a",  "aa");
>+  COMPARE(true, !=, "aa", "a");
>+
>+  COMPARE(true, <, "a",  "b");
>+  COMPARE(true, <, "a",  "aa");
>+  COMPARE(true, <, "aa", "b");
>+  COMPARE(true, <, "aa", "bb");
>+  COMPARE(false, <, "a",  "a");
>+  COMPARE(false, <, "b",  "a");
>+  COMPARE(false, <, "aa", "a");
>+  COMPARE(false, <, "b",  "aa");
>+  COMPARE(false, <, "bb", "aa");
>+
>+  COMPARE(true, <=, "a",  "a");
>+  COMPARE(true, <=, "a",  "b");
>+  COMPARE(true, <=, "a",  "aa");
>+  COMPARE(true, <=, "aa", "b");
>+  COMPARE(true, <=, "aa", "bb");
>+  COMPARE(false, <=, "b",  "a");
>+  COMPARE(false, <=, "aa", "a");
>+  COMPARE(false, <=, "b",  "aa");
>+  COMPARE(false, <=, "bb", "aa");
>+
>+  COMPARE(false, >=, "a",  "b");
>+  COMPARE(false, >=, "a",  "aa");
>+  COMPARE(false, >=, "aa", "b");
>+  COMPARE(false, >=, "aa", "bb");
>+  COMPARE(true, >=, "a",  "a");
>+  COMPARE(true, >=, "b",  "a");
>+  COMPARE(true, >=, "aa", "a");
>+  COMPARE(true, >=, "b",  "aa");
>+  COMPARE(true, >=, "bb", "aa");
>+
>+  COMPARE(false, >, "a",  "a");
>+  COMPARE(false, >, "a",  "b");
>+  COMPARE(false, >, "a",  "aa");
>+  COMPARE(false, >, "aa", "b");
>+  COMPARE(false, >, "aa", "bb");
>+  COMPARE(true, >, "b",  "a");
>+  COMPARE(true, >, "aa", "a");
>+  COMPARE(true, >, "b",  "aa");
>+  COMPARE(true, >, "bb", "aa");
>+}
>+
>+TEST(StringViewTest, ComparisonOperatorsByCharacterPosition) {
>+  std::string x;
>+  for (int i = 0; i < 256; i++) {
>+    x += 'a';
>+    std::string y = x;
>+    COMPARE(true, ==, x, y);
>+    for (int j = 0; j < i; j++) {
>+      std::string z = x;
>+      z[j] = 'b';       // Differs in position 'j'
>+      COMPARE(false, ==, x, z);
>+      COMPARE(true, <, x, z);
>+      COMPARE(true, >, z, x);
>+      if (j + 1 < i) {
>+        z[j + 1] = 'A';  // Differs in position 'j+1' as well
>+        COMPARE(false, ==, x, z);
>+        COMPARE(true, <, x, z);
>+        COMPARE(true, >, z, x);
>+        z[j + 1] = 'z';  // Differs in position 'j+1' as well
>+        COMPARE(false, ==, x, z);
>+        COMPARE(true, <, x, z);
>+        COMPARE(true, >, z, x);
>+      }
>+    }
>+  }
>+}
>+#undef COMPARE
>+
>+// Sadly, our users often confuse std::string::npos with absl::string_view::npos;
>+// So much so that we test here that they are the same.  They need to
>+// both be unsigned, and both be the maximum-valued integer of their type.
>+
>+template <typename T>
>+struct is_type {
>+  template <typename U>
>+  static bool same(U) {
>+    return false;
>+  }
>+  static bool same(T) { return true; }
>+};
>+
>+TEST(StringViewTest, NposMatchesStdStringView) {
>+  EXPECT_EQ(absl::string_view::npos, std::string::npos);
>+
>+  EXPECT_TRUE(is_type<size_t>::same(absl::string_view::npos));
>+  EXPECT_FALSE(is_type<size_t>::same(""));
>+
>+  // Make sure absl::string_view::npos continues to be a header constant.
>+  char test[absl::string_view::npos & 1] = {0};
>+  EXPECT_EQ(0, test[0]);
>+}
>+
>+TEST(StringViewTest, STL1) {
>+  const absl::string_view a("abcdefghijklmnopqrstuvwxyz");
>+  const absl::string_view b("abc");
>+  const absl::string_view c("xyz");
>+  const absl::string_view d("foobar");
>+  const absl::string_view e;
>+  std::string temp("123");
>+  temp += '\0';
>+  temp += "456";
>+  const absl::string_view f(temp);
>+
>+  EXPECT_EQ(a[6], 'g');
>+  EXPECT_EQ(b[0], 'a');
>+  EXPECT_EQ(c[2], 'z');
>+  EXPECT_EQ(f[3], '\0');
>+  EXPECT_EQ(f[5], '5');
>+
>+  EXPECT_EQ(*d.data(), 'f');
>+  EXPECT_EQ(d.data()[5], 'r');
>+  EXPECT_TRUE(e.data() == nullptr);
>+
>+  EXPECT_EQ(*a.begin(), 'a');
>+  EXPECT_EQ(*(b.begin() + 2), 'c');
>+  EXPECT_EQ(*(c.end() - 1), 'z');
>+
>+  EXPECT_EQ(*a.rbegin(), 'z');
>+  EXPECT_EQ(*(b.rbegin() + 2), 'a');
>+  EXPECT_EQ(*(c.rend() - 1), 'x');
>+  EXPECT_TRUE(a.rbegin() + 26 == a.rend());
>+
>+  EXPECT_EQ(a.size(), 26);
>+  EXPECT_EQ(b.size(), 3);
>+  EXPECT_EQ(c.size(), 3);
>+  EXPECT_EQ(d.size(), 6);
>+  EXPECT_EQ(e.size(), 0);
>+  EXPECT_EQ(f.size(), 7);
>+
>+  EXPECT_TRUE(!d.empty());
>+  EXPECT_TRUE(d.begin() != d.end());
>+  EXPECT_TRUE(d.begin() + 6 == d.end());
>+
>+  EXPECT_TRUE(e.empty());
>+  EXPECT_TRUE(e.begin() == e.end());
>+
>+  char buf[4] = { '%', '%', '%', '%' };
>+  EXPECT_EQ(a.copy(buf, 4), 4);
>+  EXPECT_EQ(buf[0], a[0]);
>+  EXPECT_EQ(buf[1], a[1]);
>+  EXPECT_EQ(buf[2], a[2]);
>+  EXPECT_EQ(buf[3], a[3]);
>+  EXPECT_EQ(a.copy(buf, 3, 7), 3);
>+  EXPECT_EQ(buf[0], a[7]);
>+  EXPECT_EQ(buf[1], a[8]);
>+  EXPECT_EQ(buf[2], a[9]);
>+  EXPECT_EQ(buf[3], a[3]);
>+  EXPECT_EQ(c.copy(buf, 99), 3);
>+  EXPECT_EQ(buf[0], c[0]);
>+  EXPECT_EQ(buf[1], c[1]);
>+  EXPECT_EQ(buf[2], c[2]);
>+  EXPECT_EQ(buf[3], a[3]);
>+}
>+
>+// Separated from STL1() because some compilers produce an overly
>+// large stack frame for the combined function.
>+TEST(StringViewTest, STL2) {
>+  const absl::string_view a("abcdefghijklmnopqrstuvwxyz");
>+  const absl::string_view b("abc");
>+  const absl::string_view c("xyz");
>+  absl::string_view d("foobar");
>+  const absl::string_view e;
>+  const absl::string_view f(
>+      "123"
>+      "\0"
>+      "456",
>+      7);
>+
>+  d = absl::string_view();
>+  EXPECT_EQ(d.size(), 0);
>+  EXPECT_TRUE(d.empty());
>+  EXPECT_TRUE(d.data() == nullptr);
>+  EXPECT_TRUE(d.begin() == d.end());
>+
>+  EXPECT_EQ(a.find(b), 0);
>+  EXPECT_EQ(a.find(b, 1), absl::string_view::npos);
>+  EXPECT_EQ(a.find(c), 23);
>+  EXPECT_EQ(a.find(c, 9), 23);
>+  EXPECT_EQ(a.find(c, absl::string_view::npos), absl::string_view::npos);
>+  EXPECT_EQ(b.find(c), absl::string_view::npos);
>+  EXPECT_EQ(b.find(c, absl::string_view::npos), absl::string_view::npos);
>+  EXPECT_EQ(a.find(d), 0);
>+  EXPECT_EQ(a.find(e), 0);
>+  EXPECT_EQ(a.find(d, 12), 12);
>+  EXPECT_EQ(a.find(e, 17), 17);
>+  absl::string_view g("xx not found bb");
>+  EXPECT_EQ(a.find(g), absl::string_view::npos);
>+  // empty std::string nonsense
>+  EXPECT_EQ(d.find(b), absl::string_view::npos);
>+  EXPECT_EQ(e.find(b), absl::string_view::npos);
>+  EXPECT_EQ(d.find(b, 4), absl::string_view::npos);
>+  EXPECT_EQ(e.find(b, 7), absl::string_view::npos);
>+
>+  size_t empty_search_pos = std::string().find(std::string());
>+  EXPECT_EQ(d.find(d), empty_search_pos);
>+  EXPECT_EQ(d.find(e), empty_search_pos);
>+  EXPECT_EQ(e.find(d), empty_search_pos);
>+  EXPECT_EQ(e.find(e), empty_search_pos);
>+  EXPECT_EQ(d.find(d, 4), std::string().find(std::string(), 4));
>+  EXPECT_EQ(d.find(e, 4), std::string().find(std::string(), 4));
>+  EXPECT_EQ(e.find(d, 4), std::string().find(std::string(), 4));
>+  EXPECT_EQ(e.find(e, 4), std::string().find(std::string(), 4));
>+
>+  EXPECT_EQ(a.find('a'), 0);
>+  EXPECT_EQ(a.find('c'), 2);
>+  EXPECT_EQ(a.find('z'), 25);
>+  EXPECT_EQ(a.find('$'), absl::string_view::npos);
>+  EXPECT_EQ(a.find('\0'), absl::string_view::npos);
>+  EXPECT_EQ(f.find('\0'), 3);
>+  EXPECT_EQ(f.find('3'), 2);
>+  EXPECT_EQ(f.find('5'), 5);
>+  EXPECT_EQ(g.find('o'), 4);
>+  EXPECT_EQ(g.find('o', 4), 4);
>+  EXPECT_EQ(g.find('o', 5), 8);
>+  EXPECT_EQ(a.find('b', 5), absl::string_view::npos);
>+  // empty std::string nonsense
>+  EXPECT_EQ(d.find('\0'), absl::string_view::npos);
>+  EXPECT_EQ(e.find('\0'), absl::string_view::npos);
>+  EXPECT_EQ(d.find('\0', 4), absl::string_view::npos);
>+  EXPECT_EQ(e.find('\0', 7), absl::string_view::npos);
>+  EXPECT_EQ(d.find('x'), absl::string_view::npos);
>+  EXPECT_EQ(e.find('x'), absl::string_view::npos);
>+  EXPECT_EQ(d.find('x', 4), absl::string_view::npos);
>+  EXPECT_EQ(e.find('x', 7), absl::string_view::npos);
>+
>+  EXPECT_EQ(a.rfind(b), 0);
>+  EXPECT_EQ(a.rfind(b, 1), 0);
>+  EXPECT_EQ(a.rfind(c), 23);
>+  EXPECT_EQ(a.rfind(c, 22), absl::string_view::npos);
>+  EXPECT_EQ(a.rfind(c, 1), absl::string_view::npos);
>+  EXPECT_EQ(a.rfind(c, 0), absl::string_view::npos);
>+  EXPECT_EQ(b.rfind(c), absl::string_view::npos);
>+  EXPECT_EQ(b.rfind(c, 0), absl::string_view::npos);
>+  EXPECT_EQ(a.rfind(d), std::string(a).rfind(std::string()));
>+  EXPECT_EQ(a.rfind(e), std::string(a).rfind(std::string()));
>+  EXPECT_EQ(a.rfind(d, 12), 12);
>+  EXPECT_EQ(a.rfind(e, 17), 17);
>+  EXPECT_EQ(a.rfind(g), absl::string_view::npos);
>+  EXPECT_EQ(d.rfind(b), absl::string_view::npos);
>+  EXPECT_EQ(e.rfind(b), absl::string_view::npos);
>+  EXPECT_EQ(d.rfind(b, 4), absl::string_view::npos);
>+  EXPECT_EQ(e.rfind(b, 7), absl::string_view::npos);
>+  // empty std::string nonsense
>+  EXPECT_EQ(d.rfind(d, 4), std::string().rfind(std::string()));
>+  EXPECT_EQ(e.rfind(d, 7), std::string().rfind(std::string()));
>+  EXPECT_EQ(d.rfind(e, 4), std::string().rfind(std::string()));
>+  EXPECT_EQ(e.rfind(e, 7), std::string().rfind(std::string()));
>+  EXPECT_EQ(d.rfind(d), std::string().rfind(std::string()));
>+  EXPECT_EQ(e.rfind(d), std::string().rfind(std::string()));
>+  EXPECT_EQ(d.rfind(e), std::string().rfind(std::string()));
>+  EXPECT_EQ(e.rfind(e), std::string().rfind(std::string()));
>+
>+  EXPECT_EQ(g.rfind('o'), 8);
>+  EXPECT_EQ(g.rfind('q'), absl::string_view::npos);
>+  EXPECT_EQ(g.rfind('o', 8), 8);
>+  EXPECT_EQ(g.rfind('o', 7), 4);
>+  EXPECT_EQ(g.rfind('o', 3), absl::string_view::npos);
>+  EXPECT_EQ(f.rfind('\0'), 3);
>+  EXPECT_EQ(f.rfind('\0', 12), 3);
>+  EXPECT_EQ(f.rfind('3'), 2);
>+  EXPECT_EQ(f.rfind('5'), 5);
>+  // empty std::string nonsense
>+  EXPECT_EQ(d.rfind('o'), absl::string_view::npos);
>+  EXPECT_EQ(e.rfind('o'), absl::string_view::npos);
>+  EXPECT_EQ(d.rfind('o', 4), absl::string_view::npos);
>+  EXPECT_EQ(e.rfind('o', 7), absl::string_view::npos);
>+}
>+
>+// Continued from STL2
>+TEST(StringViewTest, STL2FindFirst) {
>+  const absl::string_view a("abcdefghijklmnopqrstuvwxyz");
>+  const absl::string_view b("abc");
>+  const absl::string_view c("xyz");
>+  absl::string_view d("foobar");
>+  const absl::string_view e;
>+  const absl::string_view f(
>+      "123"
>+      "\0"
>+      "456",
>+      7);
>+  absl::string_view g("xx not found bb");
>+
>+  d = absl::string_view();
>+  EXPECT_EQ(a.find_first_of(b), 0);
>+  EXPECT_EQ(a.find_first_of(b, 0), 0);
>+  EXPECT_EQ(a.find_first_of(b, 1), 1);
>+  EXPECT_EQ(a.find_first_of(b, 2), 2);
>+  EXPECT_EQ(a.find_first_of(b, 3), absl::string_view::npos);
>+  EXPECT_EQ(a.find_first_of(c), 23);
>+  EXPECT_EQ(a.find_first_of(c, 23), 23);
>+  EXPECT_EQ(a.find_first_of(c, 24), 24);
>+  EXPECT_EQ(a.find_first_of(c, 25), 25);
>+  EXPECT_EQ(a.find_first_of(c, 26), absl::string_view::npos);
>+  EXPECT_EQ(g.find_first_of(b), 13);
>+  EXPECT_EQ(g.find_first_of(c), 0);
>+  EXPECT_EQ(a.find_first_of(f), absl::string_view::npos);
>+  EXPECT_EQ(f.find_first_of(a), absl::string_view::npos);
>+  // empty std::string nonsense
>+  EXPECT_EQ(a.find_first_of(d), absl::string_view::npos);
>+  EXPECT_EQ(a.find_first_of(e), absl::string_view::npos);
>+  EXPECT_EQ(d.find_first_of(b), absl::string_view::npos);
>+  EXPECT_EQ(e.find_first_of(b), absl::string_view::npos);
>+  EXPECT_EQ(d.find_first_of(d), absl::string_view::npos);
>+  EXPECT_EQ(e.find_first_of(d), absl::string_view::npos);
>+  EXPECT_EQ(d.find_first_of(e), absl::string_view::npos);
>+  EXPECT_EQ(e.find_first_of(e), absl::string_view::npos);
>+
>+  EXPECT_EQ(a.find_first_not_of(b), 3);
>+  EXPECT_EQ(a.find_first_not_of(c), 0);
>+  EXPECT_EQ(b.find_first_not_of(a), absl::string_view::npos);
>+  EXPECT_EQ(c.find_first_not_of(a), absl::string_view::npos);
>+  EXPECT_EQ(f.find_first_not_of(a), 0);
>+  EXPECT_EQ(a.find_first_not_of(f), 0);
>+  EXPECT_EQ(a.find_first_not_of(d), 0);
>+  EXPECT_EQ(a.find_first_not_of(e), 0);
>+  // empty std::string nonsense
>+  EXPECT_EQ(a.find_first_not_of(d), 0);
>+  EXPECT_EQ(a.find_first_not_of(e), 0);
>+  EXPECT_EQ(a.find_first_not_of(d, 1), 1);
>+  EXPECT_EQ(a.find_first_not_of(e, 1), 1);
>+  EXPECT_EQ(a.find_first_not_of(d, a.size() - 1), a.size() - 1);
>+  EXPECT_EQ(a.find_first_not_of(e, a.size() - 1), a.size() - 1);
>+  EXPECT_EQ(a.find_first_not_of(d, a.size()), absl::string_view::npos);
>+  EXPECT_EQ(a.find_first_not_of(e, a.size()), absl::string_view::npos);
>+  EXPECT_EQ(a.find_first_not_of(d, absl::string_view::npos),
>+            absl::string_view::npos);
>+  EXPECT_EQ(a.find_first_not_of(e, absl::string_view::npos),
>+            absl::string_view::npos);
>+  EXPECT_EQ(d.find_first_not_of(a), absl::string_view::npos);
>+  EXPECT_EQ(e.find_first_not_of(a), absl::string_view::npos);
>+  EXPECT_EQ(d.find_first_not_of(d), absl::string_view::npos);
>+  EXPECT_EQ(e.find_first_not_of(d), absl::string_view::npos);
>+  EXPECT_EQ(d.find_first_not_of(e), absl::string_view::npos);
>+  EXPECT_EQ(e.find_first_not_of(e), absl::string_view::npos);
>+
>+  absl::string_view h("====");
>+  EXPECT_EQ(h.find_first_not_of('='), absl::string_view::npos);
>+  EXPECT_EQ(h.find_first_not_of('=', 3), absl::string_view::npos);
>+  EXPECT_EQ(h.find_first_not_of('\0'), 0);
>+  EXPECT_EQ(g.find_first_not_of('x'), 2);
>+  EXPECT_EQ(f.find_first_not_of('\0'), 0);
>+  EXPECT_EQ(f.find_first_not_of('\0', 3), 4);
>+  EXPECT_EQ(f.find_first_not_of('\0', 2), 2);
>+  // empty std::string nonsense
>+  EXPECT_EQ(d.find_first_not_of('x'), absl::string_view::npos);
>+  EXPECT_EQ(e.find_first_not_of('x'), absl::string_view::npos);
>+  EXPECT_EQ(d.find_first_not_of('\0'), absl::string_view::npos);
>+  EXPECT_EQ(e.find_first_not_of('\0'), absl::string_view::npos);
>+}
>+
>+// Continued from STL2
>+TEST(StringViewTest, STL2FindLast) {
>+  const absl::string_view a("abcdefghijklmnopqrstuvwxyz");
>+  const absl::string_view b("abc");
>+  const absl::string_view c("xyz");
>+  absl::string_view d("foobar");
>+  const absl::string_view e;
>+  const absl::string_view f(
>+      "123"
>+      "\0"
>+      "456",
>+      7);
>+  absl::string_view g("xx not found bb");
>+  absl::string_view h("====");
>+  absl::string_view i("56");
>+
>+  d = absl::string_view();
>+  EXPECT_EQ(h.find_last_of(a), absl::string_view::npos);
>+  EXPECT_EQ(g.find_last_of(a), g.size()-1);
>+  EXPECT_EQ(a.find_last_of(b), 2);
>+  EXPECT_EQ(a.find_last_of(c), a.size()-1);
>+  EXPECT_EQ(f.find_last_of(i), 6);
>+  EXPECT_EQ(a.find_last_of('a'), 0);
>+  EXPECT_EQ(a.find_last_of('b'), 1);
>+  EXPECT_EQ(a.find_last_of('z'), 25);
>+  EXPECT_EQ(a.find_last_of('a', 5), 0);
>+  EXPECT_EQ(a.find_last_of('b', 5), 1);
>+  EXPECT_EQ(a.find_last_of('b', 0), absl::string_view::npos);
>+  EXPECT_EQ(a.find_last_of('z', 25), 25);
>+  EXPECT_EQ(a.find_last_of('z', 24), absl::string_view::npos);
>+  EXPECT_EQ(f.find_last_of(i, 5), 5);
>+  EXPECT_EQ(f.find_last_of(i, 6), 6);
>+  EXPECT_EQ(f.find_last_of(a, 4), absl::string_view::npos);
>+  // empty std::string nonsense
>+  EXPECT_EQ(f.find_last_of(d), absl::string_view::npos);
>+  EXPECT_EQ(f.find_last_of(e), absl::string_view::npos);
>+  EXPECT_EQ(f.find_last_of(d, 4), absl::string_view::npos);
>+  EXPECT_EQ(f.find_last_of(e, 4), absl::string_view::npos);
>+  EXPECT_EQ(d.find_last_of(d), absl::string_view::npos);
>+  EXPECT_EQ(d.find_last_of(e), absl::string_view::npos);
>+  EXPECT_EQ(e.find_last_of(d), absl::string_view::npos);
>+  EXPECT_EQ(e.find_last_of(e), absl::string_view::npos);
>+  EXPECT_EQ(d.find_last_of(f), absl::string_view::npos);
>+  EXPECT_EQ(e.find_last_of(f), absl::string_view::npos);
>+  EXPECT_EQ(d.find_last_of(d, 4), absl::string_view::npos);
>+  EXPECT_EQ(d.find_last_of(e, 4), absl::string_view::npos);
>+  EXPECT_EQ(e.find_last_of(d, 4), absl::string_view::npos);
>+  EXPECT_EQ(e.find_last_of(e, 4), absl::string_view::npos);
>+  EXPECT_EQ(d.find_last_of(f, 4), absl::string_view::npos);
>+  EXPECT_EQ(e.find_last_of(f, 4), absl::string_view::npos);
>+
>+  EXPECT_EQ(a.find_last_not_of(b), a.size()-1);
>+  EXPECT_EQ(a.find_last_not_of(c), 22);
>+  EXPECT_EQ(b.find_last_not_of(a), absl::string_view::npos);
>+  EXPECT_EQ(b.find_last_not_of(b), absl::string_view::npos);
>+  EXPECT_EQ(f.find_last_not_of(i), 4);
>+  EXPECT_EQ(a.find_last_not_of(c, 24), 22);
>+  EXPECT_EQ(a.find_last_not_of(b, 3), 3);
>+  EXPECT_EQ(a.find_last_not_of(b, 2), absl::string_view::npos);
>+  // empty std::string nonsense
>+  EXPECT_EQ(f.find_last_not_of(d), f.size()-1);
>+  EXPECT_EQ(f.find_last_not_of(e), f.size()-1);
>+  EXPECT_EQ(f.find_last_not_of(d, 4), 4);
>+  EXPECT_EQ(f.find_last_not_of(e, 4), 4);
>+  EXPECT_EQ(d.find_last_not_of(d), absl::string_view::npos);
>+  EXPECT_EQ(d.find_last_not_of(e), absl::string_view::npos);
>+  EXPECT_EQ(e.find_last_not_of(d), absl::string_view::npos);
>+  EXPECT_EQ(e.find_last_not_of(e), absl::string_view::npos);
>+  EXPECT_EQ(d.find_last_not_of(f), absl::string_view::npos);
>+  EXPECT_EQ(e.find_last_not_of(f), absl::string_view::npos);
>+  EXPECT_EQ(d.find_last_not_of(d, 4), absl::string_view::npos);
>+  EXPECT_EQ(d.find_last_not_of(e, 4), absl::string_view::npos);
>+  EXPECT_EQ(e.find_last_not_of(d, 4), absl::string_view::npos);
>+  EXPECT_EQ(e.find_last_not_of(e, 4), absl::string_view::npos);
>+  EXPECT_EQ(d.find_last_not_of(f, 4), absl::string_view::npos);
>+  EXPECT_EQ(e.find_last_not_of(f, 4), absl::string_view::npos);
>+
>+  EXPECT_EQ(h.find_last_not_of('x'), h.size() - 1);
>+  EXPECT_EQ(h.find_last_not_of('='), absl::string_view::npos);
>+  EXPECT_EQ(b.find_last_not_of('c'), 1);
>+  EXPECT_EQ(h.find_last_not_of('x', 2), 2);
>+  EXPECT_EQ(h.find_last_not_of('=', 2), absl::string_view::npos);
>+  EXPECT_EQ(b.find_last_not_of('b', 1), 0);
>+  // empty std::string nonsense
>+  EXPECT_EQ(d.find_last_not_of('x'), absl::string_view::npos);
>+  EXPECT_EQ(e.find_last_not_of('x'), absl::string_view::npos);
>+  EXPECT_EQ(d.find_last_not_of('\0'), absl::string_view::npos);
>+  EXPECT_EQ(e.find_last_not_of('\0'), absl::string_view::npos);
>+}
>+
>+// Continued from STL2
>+TEST(StringViewTest, STL2Substr) {
>+  const absl::string_view a("abcdefghijklmnopqrstuvwxyz");
>+  const absl::string_view b("abc");
>+  const absl::string_view c("xyz");
>+  absl::string_view d("foobar");
>+  const absl::string_view e;
>+
>+  d = absl::string_view();
>+  EXPECT_EQ(a.substr(0, 3), b);
>+  EXPECT_EQ(a.substr(23), c);
>+  EXPECT_EQ(a.substr(23, 3), c);
>+  EXPECT_EQ(a.substr(23, 99), c);
>+  EXPECT_EQ(a.substr(0), a);
>+  EXPECT_EQ(a.substr(3, 2), "de");
>+  // empty std::string nonsense
>+  EXPECT_EQ(d.substr(0, 99), e);
>+  // use of npos
>+  EXPECT_EQ(a.substr(0, absl::string_view::npos), a);
>+  EXPECT_EQ(a.substr(23, absl::string_view::npos), c);
>+  // throw exception
>+#ifdef ABSL_HAVE_EXCEPTIONS
>+  EXPECT_THROW(a.substr(99, 2), std::out_of_range);
>+#else
>+  EXPECT_DEATH(a.substr(99, 2), "absl::string_view::substr");
>+#endif
>+}
>+
>+TEST(StringViewTest, TruncSubstr) {
>+  const absl::string_view hi("hi");
>+  EXPECT_EQ("", absl::ClippedSubstr(hi, 0, 0));
>+  EXPECT_EQ("h", absl::ClippedSubstr(hi, 0, 1));
>+  EXPECT_EQ("hi", absl::ClippedSubstr(hi, 0));
>+  EXPECT_EQ("i", absl::ClippedSubstr(hi, 1));
>+  EXPECT_EQ("", absl::ClippedSubstr(hi, 2));
>+  EXPECT_EQ("", absl::ClippedSubstr(hi, 3));  // truncation
>+  EXPECT_EQ("", absl::ClippedSubstr(hi, 3, 2));  // truncation
>+}
>+
>+TEST(StringViewTest, UTF8) {
>+  std::string utf8 = "\u00E1";
>+  std::string utf8_twice = utf8 + " " + utf8;
>+  int utf8_len = strlen(utf8.data());
>+  EXPECT_EQ(utf8_len, absl::string_view(utf8_twice).find_first_of(" "));
>+  EXPECT_EQ(utf8_len, absl::string_view(utf8_twice).find_first_of(" \t"));
>+}
>+
>+TEST(StringViewTest, FindConformance) {
>+  struct {
>+    std::string haystack;
>+    std::string needle;
>+  } specs[] = {
>+    {"", ""},
>+    {"", "a"},
>+    {"a", ""},
>+    {"a", "a"},
>+    {"a", "b"},
>+    {"aa", ""},
>+    {"aa", "a"},
>+    {"aa", "b"},
>+    {"ab", "a"},
>+    {"ab", "b"},
>+    {"abcd", ""},
>+    {"abcd", "a"},
>+    {"abcd", "d"},
>+    {"abcd", "ab"},
>+    {"abcd", "bc"},
>+    {"abcd", "cd"},
>+    {"abcd", "abcd"},
>+  };
>+  for (const auto& s : specs) {
>+    SCOPED_TRACE(s.haystack);
>+    SCOPED_TRACE(s.needle);
>+    std::string st = s.haystack;
>+    absl::string_view sp = s.haystack;
>+    for (size_t i = 0; i <= sp.size(); ++i) {
>+      size_t pos = (i == sp.size()) ? absl::string_view::npos : i;
>+      SCOPED_TRACE(pos);
>+      EXPECT_EQ(sp.find(s.needle, pos),
>+                st.find(s.needle, pos));
>+      EXPECT_EQ(sp.rfind(s.needle, pos),
>+                st.rfind(s.needle, pos));
>+      EXPECT_EQ(sp.find_first_of(s.needle, pos),
>+                st.find_first_of(s.needle, pos));
>+      EXPECT_EQ(sp.find_first_not_of(s.needle, pos),
>+                st.find_first_not_of(s.needle, pos));
>+      EXPECT_EQ(sp.find_last_of(s.needle, pos),
>+                st.find_last_of(s.needle, pos));
>+      EXPECT_EQ(sp.find_last_not_of(s.needle, pos),
>+                st.find_last_not_of(s.needle, pos));
>+    }
>+  }
>+}
>+
>+TEST(StringViewTest, Remove) {
>+  absl::string_view a("foobar");
>+  std::string s1("123");
>+  s1 += '\0';
>+  s1 += "456";
>+  absl::string_view b(s1);
>+  absl::string_view e;
>+  std::string s2;
>+
>+  // remove_prefix
>+  absl::string_view c(a);
>+  c.remove_prefix(3);
>+  EXPECT_EQ(c, "bar");
>+  c = a;
>+  c.remove_prefix(0);
>+  EXPECT_EQ(c, a);
>+  c.remove_prefix(c.size());
>+  EXPECT_EQ(c, e);
>+
>+  // remove_suffix
>+  c = a;
>+  c.remove_suffix(3);
>+  EXPECT_EQ(c, "foo");
>+  c = a;
>+  c.remove_suffix(0);
>+  EXPECT_EQ(c, a);
>+  c.remove_suffix(c.size());
>+  EXPECT_EQ(c, e);
>+}
>+
>+TEST(StringViewTest, Set) {
>+  absl::string_view a("foobar");
>+  absl::string_view empty;
>+  absl::string_view b;
>+
>+  // set
>+  b = absl::string_view("foobar", 6);
>+  EXPECT_EQ(b, a);
>+  b = absl::string_view("foobar", 0);
>+  EXPECT_EQ(b, empty);
>+  b = absl::string_view("foobar", 7);
>+  EXPECT_NE(b, a);
>+
>+  b = absl::string_view("foobar");
>+  EXPECT_EQ(b, a);
>+}
>+
>+TEST(StringViewTest, FrontBack) {
>+  static const char arr[] = "abcd";
>+  const absl::string_view csp(arr, 4);
>+  EXPECT_EQ(&arr[0], &csp.front());
>+  EXPECT_EQ(&arr[3], &csp.back());
>+}
>+
>+TEST(StringViewTest, FrontBackSingleChar) {
>+  static const char c = 'a';
>+  const absl::string_view csp(&c, 1);
>+  EXPECT_EQ(&c, &csp.front());
>+  EXPECT_EQ(&c, &csp.back());
>+}
>+
>+// `std::string_view::string_view(const char*)` calls
>+// `std::char_traits<char>::length(const char*)` to get the std::string length. In
>+// libc++, it doesn't allow `nullptr` in the constexpr context, with the error
>+// "read of dereferenced null pointer is not allowed in a constant expression".
>+// At run time, the behavior of `std::char_traits::length()` on `nullptr` is
>+// undefined by the standard and usually results in crash with libc++. This
>+// conforms to the standard, but `absl::string_view` implements a different
>+// behavior for historical reasons. We work around tests that construct
>+// `string_view` from `nullptr` when using libc++.
>+#if !defined(ABSL_HAVE_STD_STRING_VIEW) || !defined(_LIBCPP_VERSION)
>+#define ABSL_HAVE_STRING_VIEW_FROM_NULLPTR 1
>+#endif  // !defined(ABSL_HAVE_STD_STRING_VIEW) || !defined(_LIBCPP_VERSION)
>+
>+TEST(StringViewTest, NULLInput) {
>+  absl::string_view s;
>+  EXPECT_EQ(s.data(), nullptr);
>+  EXPECT_EQ(s.size(), 0);
>+
>+#ifdef ABSL_HAVE_STRING_VIEW_FROM_NULLPTR
>+  s = absl::string_view(nullptr);
>+  EXPECT_EQ(s.data(), nullptr);
>+  EXPECT_EQ(s.size(), 0);
>+
>+  // .ToString() on a absl::string_view with nullptr should produce the empty
>+  // std::string.
>+  EXPECT_EQ("", std::string(s));
>+#endif  // ABSL_HAVE_STRING_VIEW_FROM_NULLPTR
>+}
>+
>+TEST(StringViewTest, Comparisons2) {
>+  // The `compare` member has 6 overloads (v: string_view, s: const char*):
>+  //  (1) compare(v)
>+  //  (2) compare(pos1, count1, v)
>+  //  (3) compare(pos1, count1, v, pos2, count2)
>+  //  (4) compare(s)
>+  //  (5) compare(pos1, count1, s)
>+  //  (6) compare(pos1, count1, s, count2)
>+
>+  absl::string_view abc("abcdefghijklmnopqrstuvwxyz");
>+
>+  // check comparison operations on strings longer than 4 bytes.
>+  EXPECT_EQ(abc, absl::string_view("abcdefghijklmnopqrstuvwxyz"));
>+  EXPECT_EQ(abc.compare(absl::string_view("abcdefghijklmnopqrstuvwxyz")), 0);
>+
>+  EXPECT_LT(abc, absl::string_view("abcdefghijklmnopqrstuvwxzz"));
>+  EXPECT_LT(abc.compare(absl::string_view("abcdefghijklmnopqrstuvwxzz")), 0);
>+
>+  EXPECT_GT(abc, absl::string_view("abcdefghijklmnopqrstuvwxyy"));
>+  EXPECT_GT(abc.compare(absl::string_view("abcdefghijklmnopqrstuvwxyy")), 0);
>+
>+  // The "substr" variants of `compare`.
>+  absl::string_view digits("0123456789");
>+  auto npos = absl::string_view::npos;
>+
>+  // Taking string_view
>+  EXPECT_EQ(digits.compare(3, npos, absl::string_view("3456789")), 0);  // 2
>+  EXPECT_EQ(digits.compare(3, 4, absl::string_view("3456")), 0);        // 2
>+  EXPECT_EQ(digits.compare(10, 0, absl::string_view()), 0);             // 2
>+  EXPECT_EQ(digits.compare(3, 4, absl::string_view("0123456789"), 3, 4),
>+            0);  // 3
>+  EXPECT_LT(digits.compare(3, 4, absl::string_view("0123456789"), 3, 5),
>+            0);  // 3
>+  EXPECT_LT(digits.compare(0, npos, absl::string_view("0123456789"), 3, 5),
>+            0);  // 3
>+  // Taking const char*
>+  EXPECT_EQ(digits.compare(3, 4, "3456"), 0);                 // 5
>+  EXPECT_EQ(digits.compare(3, npos, "3456789"), 0);           // 5
>+  EXPECT_EQ(digits.compare(10, 0, ""), 0);                    // 5
>+  EXPECT_EQ(digits.compare(3, 4, "0123456789", 3, 4), 0);     // 6
>+  EXPECT_LT(digits.compare(3, 4, "0123456789", 3, 5), 0);     // 6
>+  EXPECT_LT(digits.compare(0, npos, "0123456789", 3, 5), 0);  // 6
>+}
>+
>+struct MyCharAlloc : std::allocator<char> {};
>+
>+TEST(StringViewTest, ExplicitConversionOperator) {
>+  absl::string_view sp = "hi";
>+  EXPECT_EQ(sp, std::string(sp));
>+}
>+
>+TEST(StringViewTest, NullSafeStringView) {
>+  {
>+    absl::string_view s = absl::NullSafeStringView(nullptr);
>+    EXPECT_EQ(nullptr, s.data());
>+    EXPECT_EQ(0, s.size());
>+    EXPECT_EQ(absl::string_view(), s);
>+  }
>+  {
>+    static const char kHi[] = "hi";
>+    absl::string_view s = absl::NullSafeStringView(kHi);
>+    EXPECT_EQ(kHi, s.data());
>+    EXPECT_EQ(strlen(kHi), s.size());
>+    EXPECT_EQ(absl::string_view("hi"), s);
>+  }
>+}
>+
>+TEST(StringViewTest, ConstexprCompiles) {
>+  constexpr absl::string_view sp;
>+#ifdef ABSL_HAVE_STRING_VIEW_FROM_NULLPTR
>+  constexpr absl::string_view cstr(nullptr);
>+#endif
>+  constexpr absl::string_view cstr_len("cstr", 4);
>+
>+#if defined(ABSL_HAVE_STD_STRING_VIEW)
>+  // In libstdc++ (as of 7.2), `std::string_view::string_view(const char*)`
>+  // calls `std::char_traits<char>::length(const char*)` to get the std::string
>+  // length, but it is not marked constexpr yet. See GCC bug:
>+  // https://gcc.gnu.org/bugzilla/show_bug.cgi?id=78156
>+  // Also, there is a LWG issue that adds constexpr to length() which was just
>+  // resolved 2017-06-02. See
>+  // http://www.open-std.org/jtc1/sc22/wg21/docs/lwg-defects.html#2232
>+  // TODO(zhangxy): Update the condition when libstdc++ adopts the constexpr
>+  // length().
>+#if !defined(__GLIBCXX__)
>+#define ABSL_HAVE_CONSTEXPR_STRING_VIEW_FROM_CSTR 1
>+#endif  // !__GLIBCXX__
>+
>+#else  // ABSL_HAVE_STD_STRING_VIEW
>+
>+// This duplicates the check for __builtin_strlen in the header.
>+#if ABSL_HAVE_BUILTIN(__builtin_strlen) || \
>+    (defined(__GNUC__) && !defined(__clang__))
>+#define ABSL_HAVE_CONSTEXPR_STRING_VIEW_FROM_CSTR 1
>+#elif defined(__GNUC__)  // GCC or clang
>+#error GCC/clang should have constexpr string_view.
>+#endif
>+
>+#endif  // ABSL_HAVE_STD_STRING_VIEW
>+
>+#ifdef ABSL_HAVE_CONSTEXPR_STRING_VIEW_FROM_CSTR
>+  constexpr absl::string_view cstr_strlen("foo");
>+  EXPECT_EQ(cstr_strlen.length(), 3);
>+  constexpr absl::string_view cstr_strlen2 = "bar";
>+  EXPECT_EQ(cstr_strlen2, "bar");
>+#endif
>+
>+#if !defined(__clang__) || 3 < __clang_major__ || \
>+  (3 == __clang_major__ && 4 < __clang_minor__)
>+  // older clang versions (< 3.5) complain that:
>+  //   "cannot perform pointer arithmetic on null pointer"
>+  constexpr absl::string_view::iterator const_begin_empty = sp.begin();
>+  constexpr absl::string_view::iterator const_end_empty = sp.end();
>+  EXPECT_EQ(const_begin_empty, const_end_empty);
>+
>+#ifdef ABSL_HAVE_STRING_VIEW_FROM_NULLPTR
>+  constexpr absl::string_view::iterator const_begin_nullptr = cstr.begin();
>+  constexpr absl::string_view::iterator const_end_nullptr = cstr.end();
>+  EXPECT_EQ(const_begin_nullptr, const_end_nullptr);
>+#endif  // ABSL_HAVE_STRING_VIEW_FROM_NULLPTR
>+#endif  // !defined(__clang__) || ...
>+
>+  constexpr absl::string_view::iterator const_begin = cstr_len.begin();
>+  constexpr absl::string_view::iterator const_end = cstr_len.end();
>+  constexpr absl::string_view::size_type const_size = cstr_len.size();
>+  constexpr absl::string_view::size_type const_length = cstr_len.length();
>+  EXPECT_EQ(const_begin + const_size, const_end);
>+  EXPECT_EQ(const_begin + const_length, const_end);
>+
>+  constexpr bool isempty = sp.empty();
>+  EXPECT_TRUE(isempty);
>+
>+  constexpr const char c = cstr_len[2];
>+  EXPECT_EQ(c, 't');
>+
>+  constexpr const char cfront = cstr_len.front();
>+  constexpr const char cback = cstr_len.back();
>+  EXPECT_EQ(cfront, 'c');
>+  EXPECT_EQ(cback, 'r');
>+
>+  constexpr const char* np = sp.data();
>+  constexpr const char* cstr_ptr = cstr_len.data();
>+  EXPECT_EQ(np, nullptr);
>+  EXPECT_NE(cstr_ptr, nullptr);
>+
>+  constexpr size_t sp_npos = sp.npos;
>+  EXPECT_EQ(sp_npos, -1);
>+}
>+
>+TEST(StringViewTest, Noexcept) {
>+  EXPECT_TRUE((std::is_nothrow_constructible<absl::string_view,
>+                                             const std::string&>::value));
>+  EXPECT_TRUE(
>+      (std::is_nothrow_constructible<absl::string_view, const std::string&>::value));
>+  EXPECT_TRUE(std::is_nothrow_constructible<absl::string_view>::value);
>+  constexpr absl::string_view sp;
>+  EXPECT_TRUE(noexcept(sp.begin()));
>+  EXPECT_TRUE(noexcept(sp.end()));
>+  EXPECT_TRUE(noexcept(sp.cbegin()));
>+  EXPECT_TRUE(noexcept(sp.cend()));
>+  EXPECT_TRUE(noexcept(sp.rbegin()));
>+  EXPECT_TRUE(noexcept(sp.rend()));
>+  EXPECT_TRUE(noexcept(sp.crbegin()));
>+  EXPECT_TRUE(noexcept(sp.crend()));
>+  EXPECT_TRUE(noexcept(sp.size()));
>+  EXPECT_TRUE(noexcept(sp.length()));
>+  EXPECT_TRUE(noexcept(sp.empty()));
>+  EXPECT_TRUE(noexcept(sp.data()));
>+  EXPECT_TRUE(noexcept(sp.compare(sp)));
>+  EXPECT_TRUE(noexcept(sp.find(sp)));
>+  EXPECT_TRUE(noexcept(sp.find('f')));
>+  EXPECT_TRUE(noexcept(sp.rfind(sp)));
>+  EXPECT_TRUE(noexcept(sp.rfind('f')));
>+  EXPECT_TRUE(noexcept(sp.find_first_of(sp)));
>+  EXPECT_TRUE(noexcept(sp.find_first_of('f')));
>+  EXPECT_TRUE(noexcept(sp.find_last_of(sp)));
>+  EXPECT_TRUE(noexcept(sp.find_last_of('f')));
>+  EXPECT_TRUE(noexcept(sp.find_first_not_of(sp)));
>+  EXPECT_TRUE(noexcept(sp.find_first_not_of('f')));
>+  EXPECT_TRUE(noexcept(sp.find_last_not_of(sp)));
>+  EXPECT_TRUE(noexcept(sp.find_last_not_of('f')));
>+}
>+
>+TEST(ComparisonOpsTest, StringCompareNotAmbiguous) {
>+  EXPECT_EQ("hello", std::string("hello"));
>+  EXPECT_LT("hello", std::string("world"));
>+}
>+
>+TEST(ComparisonOpsTest, HeterogenousStringViewEquals) {
>+  EXPECT_EQ(absl::string_view("hello"), std::string("hello"));
>+  EXPECT_EQ("hello", absl::string_view("hello"));
>+}
>+
>+TEST(FindOneCharTest, EdgeCases) {
>+  absl::string_view a("xxyyyxx");
>+
>+  // Set a = "xyyyx".
>+  a.remove_prefix(1);
>+  a.remove_suffix(1);
>+
>+  EXPECT_EQ(0, a.find('x'));
>+  EXPECT_EQ(0, a.find('x', 0));
>+  EXPECT_EQ(4, a.find('x', 1));
>+  EXPECT_EQ(4, a.find('x', 4));
>+  EXPECT_EQ(absl::string_view::npos, a.find('x', 5));
>+
>+  EXPECT_EQ(4, a.rfind('x'));
>+  EXPECT_EQ(4, a.rfind('x', 5));
>+  EXPECT_EQ(4, a.rfind('x', 4));
>+  EXPECT_EQ(0, a.rfind('x', 3));
>+  EXPECT_EQ(0, a.rfind('x', 0));
>+
>+  // Set a = "yyy".
>+  a.remove_prefix(1);
>+  a.remove_suffix(1);
>+
>+  EXPECT_EQ(absl::string_view::npos, a.find('x'));
>+  EXPECT_EQ(absl::string_view::npos, a.rfind('x'));
>+}
>+
>+#ifndef THREAD_SANITIZER  // Allocates too much memory for tsan.
>+TEST(HugeStringView, TwoPointTwoGB) {
>+  if (sizeof(size_t) <= 4 || AbslRunningOnValgrind())
>+    return;
>+  // Try a huge std::string piece.
>+  const size_t size = size_t{2200} * 1000 * 1000;
>+  std::string s(size, 'a');
>+  absl::string_view sp(s);
>+  EXPECT_EQ(size, sp.length());
>+  sp.remove_prefix(1);
>+  EXPECT_EQ(size - 1, sp.length());
>+  sp.remove_suffix(2);
>+  EXPECT_EQ(size - 1 - 2, sp.length());
>+}
>+#endif  // THREAD_SANITIZER
>+
>+#if !defined(NDEBUG) && !defined(ABSL_HAVE_STD_STRING_VIEW)
>+TEST(NonNegativeLenTest, NonNegativeLen) {
>+  ABSL_EXPECT_DEATH_IF_SUPPORTED(absl::string_view("xyz", -1),
>+                                 "len <= kMaxSize");
>+}
>+
>+TEST(LenExceedsMaxSizeTest, LenExceedsMaxSize) {
>+  auto max_size = absl::string_view().max_size();
>+
>+  // This should construct ok (although the view itself is obviously invalid).
>+  absl::string_view ok_view("", max_size);
>+
>+  // Adding one to the max should trigger an assertion.
>+  ABSL_EXPECT_DEATH_IF_SUPPORTED(absl::string_view("", max_size + 1),
>+                                 "len <= kMaxSize");
>+}
>+#endif  // !defined(NDEBUG) && !defined(ABSL_HAVE_STD_STRING_VIEW)
>+
>+class StringViewStreamTest : public ::testing::Test {
>+ public:
>+  // Set negative 'width' for right justification.
>+  template <typename T>
>+  std::string Pad(const T& s, int width, char fill = 0) {
>+    std::ostringstream oss;
>+    if (fill != 0) {
>+      oss << std::setfill(fill);
>+    }
>+    if (width < 0) {
>+      width = -width;
>+      oss << std::right;
>+    }
>+    oss << std::setw(width) << s;
>+    return oss.str();
>+  }
>+};
>+
>+TEST_F(StringViewStreamTest, Padding) {
>+  std::string s("hello");
>+  absl::string_view sp(s);
>+  for (int w = -64; w < 64; ++w) {
>+    SCOPED_TRACE(w);
>+    EXPECT_EQ(Pad(s, w), Pad(sp, w));
>+  }
>+  for (int w = -64; w < 64; ++w) {
>+    SCOPED_TRACE(w);
>+    EXPECT_EQ(Pad(s, w, '#'), Pad(sp, w, '#'));
>+  }
>+}
>+
>+TEST_F(StringViewStreamTest, ResetsWidth) {
>+  // Width should reset after one formatted write.
>+  // If we weren't resetting width after formatting the string_view,
>+  // we'd have width=5 carrying over to the printing of the "]",
>+  // creating "[###hi####]".
>+  std::string s = "hi";
>+  absl::string_view sp = s;
>+  {
>+    std::ostringstream oss;
>+    oss << "[" << std::setfill('#') << std::setw(5) << s << "]";
>+    ASSERT_EQ("[###hi]", oss.str());
>+  }
>+  {
>+    std::ostringstream oss;
>+    oss << "[" << std::setfill('#') << std::setw(5) << sp << "]";
>+    EXPECT_EQ("[###hi]", oss.str());
>+  }
>+}
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/strip.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/strip.h
>new file mode 100644
>index 00000000000..2f8d21f7deb
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/strip.h
>@@ -0,0 +1,89 @@
>+//
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// -----------------------------------------------------------------------------
>+// File: strip.h
>+// -----------------------------------------------------------------------------
>+//
>+// This file contains various functions for stripping substrings from a std::string.
>+#ifndef ABSL_STRINGS_STRIP_H_
>+#define ABSL_STRINGS_STRIP_H_
>+
>+#include <cstddef>
>+#include <string>
>+
>+#include "absl/base/macros.h"
>+#include "absl/strings/ascii.h"
>+#include "absl/strings/match.h"
>+#include "absl/strings/string_view.h"
>+
>+namespace absl {
>+
>+// ConsumePrefix()
>+//
>+// Strips the `expected` prefix from the start of the given std::string, returning
>+// `true` if the strip operation succeeded or false otherwise.
>+//
>+// Example:
>+//
>+//   absl::string_view input("abc");
>+//   EXPECT_TRUE(absl::ConsumePrefix(&input, "a"));
>+//   EXPECT_EQ(input, "bc");
>+inline bool ConsumePrefix(absl::string_view* str, absl::string_view expected) {
>+  if (!absl::StartsWith(*str, expected)) return false;
>+  str->remove_prefix(expected.size());
>+  return true;
>+}
>+// ConsumeSuffix()
>+//
>+// Strips the `expected` suffix from the end of the given std::string, returning
>+// `true` if the strip operation succeeded or false otherwise.
>+//
>+// Example:
>+//
>+//   absl::string_view input("abcdef");
>+//   EXPECT_TRUE(absl::ConsumeSuffix(&input, "def"));
>+//   EXPECT_EQ(input, "abc");
>+inline bool ConsumeSuffix(absl::string_view* str, absl::string_view expected) {
>+  if (!absl::EndsWith(*str, expected)) return false;
>+  str->remove_suffix(expected.size());
>+  return true;
>+}
>+
>+// StripPrefix()
>+//
>+// Returns a view into the input std::string 'str' with the given 'prefix' removed,
>+// but leaving the original std::string intact. If the prefix does not match at the
>+// start of the std::string, returns the original std::string instead.
>+ABSL_MUST_USE_RESULT inline absl::string_view StripPrefix(
>+    absl::string_view str, absl::string_view prefix) {
>+  if (absl::StartsWith(str, prefix)) str.remove_prefix(prefix.size());
>+  return str;
>+}
>+
>+// StripSuffix()
>+//
>+// Returns a view into the input std::string 'str' with the given 'suffix' removed,
>+// but leaving the original std::string intact. If the suffix does not match at the
>+// end of the std::string, returns the original std::string instead.
>+ABSL_MUST_USE_RESULT inline absl::string_view StripSuffix(
>+    absl::string_view str, absl::string_view suffix) {
>+  if (absl::EndsWith(str, suffix)) str.remove_suffix(suffix.size());
>+  return str;
>+}
>+
>+}  // namespace absl
>+
>+#endif  // ABSL_STRINGS_STRIP_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/strip_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/strip_test.cc
>new file mode 100644
>index 00000000000..205c160c192
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/strip_test.cc
>@@ -0,0 +1,201 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+// This file contains functions that remove a defined part from the std::string,
>+// i.e., strip the std::string.
>+
>+#include "absl/strings/strip.h"
>+
>+#include <cassert>
>+#include <cstdio>
>+#include <cstring>
>+
>+#include "gmock/gmock.h"
>+#include "gtest/gtest.h"
>+#include "absl/strings/string_view.h"
>+
>+namespace {
>+
>+using testing::ElementsAre;
>+using testing::IsEmpty;
>+
>+TEST(Strip, ConsumePrefixOneChar) {
>+  absl::string_view input("abc");
>+  EXPECT_TRUE(absl::ConsumePrefix(&input, "a"));
>+  EXPECT_EQ(input, "bc");
>+
>+  EXPECT_FALSE(absl::ConsumePrefix(&input, "x"));
>+  EXPECT_EQ(input, "bc");
>+
>+  EXPECT_TRUE(absl::ConsumePrefix(&input, "b"));
>+  EXPECT_EQ(input, "c");
>+
>+  EXPECT_TRUE(absl::ConsumePrefix(&input, "c"));
>+  EXPECT_EQ(input, "");
>+
>+  EXPECT_FALSE(absl::ConsumePrefix(&input, "a"));
>+  EXPECT_EQ(input, "");
>+}
>+
>+TEST(Strip, ConsumePrefix) {
>+  absl::string_view input("abcdef");
>+  EXPECT_FALSE(absl::ConsumePrefix(&input, "abcdefg"));
>+  EXPECT_EQ(input, "abcdef");
>+
>+  EXPECT_FALSE(absl::ConsumePrefix(&input, "abce"));
>+  EXPECT_EQ(input, "abcdef");
>+
>+  EXPECT_TRUE(absl::ConsumePrefix(&input, ""));
>+  EXPECT_EQ(input, "abcdef");
>+
>+  EXPECT_FALSE(absl::ConsumePrefix(&input, "abcdeg"));
>+  EXPECT_EQ(input, "abcdef");
>+
>+  EXPECT_TRUE(absl::ConsumePrefix(&input, "abcdef"));
>+  EXPECT_EQ(input, "");
>+
>+  input = "abcdef";
>+  EXPECT_TRUE(absl::ConsumePrefix(&input, "abcde"));
>+  EXPECT_EQ(input, "f");
>+}
>+
>+TEST(Strip, ConsumeSuffix) {
>+  absl::string_view input("abcdef");
>+  EXPECT_FALSE(absl::ConsumeSuffix(&input, "abcdefg"));
>+  EXPECT_EQ(input, "abcdef");
>+
>+  EXPECT_TRUE(absl::ConsumeSuffix(&input, ""));
>+  EXPECT_EQ(input, "abcdef");
>+
>+  EXPECT_TRUE(absl::ConsumeSuffix(&input, "def"));
>+  EXPECT_EQ(input, "abc");
>+
>+  input = "abcdef";
>+  EXPECT_FALSE(absl::ConsumeSuffix(&input, "abcdeg"));
>+  EXPECT_EQ(input, "abcdef");
>+
>+  EXPECT_TRUE(absl::ConsumeSuffix(&input, "f"));
>+  EXPECT_EQ(input, "abcde");
>+
>+  EXPECT_TRUE(absl::ConsumeSuffix(&input, "abcde"));
>+  EXPECT_EQ(input, "");
>+}
>+
>+TEST(Strip, StripPrefix) {
>+  const absl::string_view null_str;
>+
>+  EXPECT_EQ(absl::StripPrefix("foobar", "foo"), "bar");
>+  EXPECT_EQ(absl::StripPrefix("foobar", ""), "foobar");
>+  EXPECT_EQ(absl::StripPrefix("foobar", null_str), "foobar");
>+  EXPECT_EQ(absl::StripPrefix("foobar", "foobar"), "");
>+  EXPECT_EQ(absl::StripPrefix("foobar", "bar"), "foobar");
>+  EXPECT_EQ(absl::StripPrefix("foobar", "foobarr"), "foobar");
>+  EXPECT_EQ(absl::StripPrefix("", ""), "");
>+}
>+
>+TEST(Strip, StripSuffix) {
>+  const absl::string_view null_str;
>+
>+  EXPECT_EQ(absl::StripSuffix("foobar", "bar"), "foo");
>+  EXPECT_EQ(absl::StripSuffix("foobar", ""), "foobar");
>+  EXPECT_EQ(absl::StripSuffix("foobar", null_str), "foobar");
>+  EXPECT_EQ(absl::StripSuffix("foobar", "foobar"), "");
>+  EXPECT_EQ(absl::StripSuffix("foobar", "foo"), "foobar");
>+  EXPECT_EQ(absl::StripSuffix("foobar", "ffoobar"), "foobar");
>+  EXPECT_EQ(absl::StripSuffix("", ""), "");
>+}
>+
>+TEST(Strip, RemoveExtraAsciiWhitespace) {
>+  const char* inputs[] = {
>+      "No extra space",
>+      "  Leading whitespace",
>+      "Trailing whitespace  ",
>+      "  Leading and trailing  ",
>+      " Whitespace \t  in\v   middle  ",
>+      "'Eeeeep!  \n Newlines!\n",
>+      "nospaces",
>+  };
>+  const char* outputs[] = {
>+      "No extra space",
>+      "Leading whitespace",
>+      "Trailing whitespace",
>+      "Leading and trailing",
>+      "Whitespace in middle",
>+      "'Eeeeep! Newlines!",
>+      "nospaces",
>+  };
>+  int NUM_TESTS = 7;
>+
>+  for (int i = 0; i < NUM_TESTS; i++) {
>+    std::string s(inputs[i]);
>+    absl::RemoveExtraAsciiWhitespace(&s);
>+    EXPECT_STREQ(outputs[i], s.c_str());
>+  }
>+
>+  // Test that absl::RemoveExtraAsciiWhitespace returns immediately for empty
>+  // strings (It was adding the \0 character to the C++ std::string, which broke
>+  // tests involving empty())
>+  std::string zero_string = "";
>+  assert(zero_string.empty());
>+  absl::RemoveExtraAsciiWhitespace(&zero_string);
>+  EXPECT_EQ(zero_string.size(), 0);
>+  EXPECT_TRUE(zero_string.empty());
>+}
>+
>+TEST(Strip, StripTrailingAsciiWhitespace) {
>+  std::string test = "foo  ";
>+  absl::StripTrailingAsciiWhitespace(&test);
>+  EXPECT_EQ(test, "foo");
>+
>+  test = "   ";
>+  absl::StripTrailingAsciiWhitespace(&test);
>+  EXPECT_EQ(test, "");
>+
>+  test = "";
>+  absl::StripTrailingAsciiWhitespace(&test);
>+  EXPECT_EQ(test, "");
>+
>+  test = " abc\t";
>+  absl::StripTrailingAsciiWhitespace(&test);
>+  EXPECT_EQ(test, " abc");
>+}
>+
>+TEST(String, StripLeadingAsciiWhitespace) {
>+  absl::string_view orig = "\t  \n\f\r\n\vfoo";
>+  EXPECT_EQ("foo", absl::StripLeadingAsciiWhitespace(orig));
>+  orig = "\t  \n\f\r\v\n\t  \n\f\r\v\n";
>+  EXPECT_EQ(absl::string_view(), absl::StripLeadingAsciiWhitespace(orig));
>+}
>+
>+TEST(Strip, StripAsciiWhitespace) {
>+  std::string test2 = "\t  \f\r\n\vfoo \t\f\r\v\n";
>+  absl::StripAsciiWhitespace(&test2);
>+  EXPECT_EQ(test2, "foo");
>+  std::string test3 = "bar";
>+  absl::StripAsciiWhitespace(&test3);
>+  EXPECT_EQ(test3, "bar");
>+  std::string test4 = "\t  \f\r\n\vfoo";
>+  absl::StripAsciiWhitespace(&test4);
>+  EXPECT_EQ(test4, "foo");
>+  std::string test5 = "foo \t\f\r\v\n";
>+  absl::StripAsciiWhitespace(&test5);
>+  EXPECT_EQ(test5, "foo");
>+  absl::string_view test6("\t  \f\r\n\vfoo \t\f\r\v\n");
>+  test6 = absl::StripAsciiWhitespace(test6);
>+  EXPECT_EQ(test6, "foo");
>+  test6 = absl::StripAsciiWhitespace(test6);
>+  EXPECT_EQ(test6, "foo");  // already stripped
>+}
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/substitute.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/substitute.cc
>new file mode 100644
>index 00000000000..3b200594509
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/substitute.cc
>@@ -0,0 +1,170 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/strings/substitute.h"
>+
>+#include <algorithm>
>+
>+#include "absl/base/internal/raw_logging.h"
>+#include "absl/strings/ascii.h"
>+#include "absl/strings/escaping.h"
>+#include "absl/strings/internal/resize_uninitialized.h"
>+#include "absl/strings/string_view.h"
>+
>+namespace absl {
>+namespace substitute_internal {
>+
>+void SubstituteAndAppendArray(std::string* output, absl::string_view format,
>+                              const absl::string_view* args_array,
>+                              size_t num_args) {
>+  // Determine total size needed.
>+  size_t size = 0;
>+  for (size_t i = 0; i < format.size(); i++) {
>+    if (format[i] == '$') {
>+      if (i + 1 >= format.size()) {
>+#ifndef NDEBUG
>+        ABSL_RAW_LOG(FATAL,
>+                     "Invalid strings::Substitute() format std::string: \"%s\".",
>+                     absl::CEscape(format).c_str());
>+#endif
>+        return;
>+      } else if (absl::ascii_isdigit(format[i + 1])) {
>+        int index = format[i + 1] - '0';
>+        if (static_cast<size_t>(index) >= num_args) {
>+#ifndef NDEBUG
>+          ABSL_RAW_LOG(
>+              FATAL,
>+              "Invalid strings::Substitute() format std::string: asked for \"$"
>+              "%d\", but only %d args were given.  Full format std::string was: "
>+              "\"%s\".",
>+              index, static_cast<int>(num_args), absl::CEscape(format).c_str());
>+#endif
>+          return;
>+        }
>+        size += args_array[index].size();
>+        ++i;  // Skip next char.
>+      } else if (format[i + 1] == '$') {
>+        ++size;
>+        ++i;  // Skip next char.
>+      } else {
>+#ifndef NDEBUG
>+        ABSL_RAW_LOG(FATAL,
>+                     "Invalid strings::Substitute() format std::string: \"%s\".",
>+                     absl::CEscape(format).c_str());
>+#endif
>+        return;
>+      }
>+    } else {
>+      ++size;
>+    }
>+  }
>+
>+  if (size == 0) return;
>+
>+  // Build the std::string.
>+  size_t original_size = output->size();
>+  strings_internal::STLStringResizeUninitialized(output, original_size + size);
>+  char* target = &(*output)[original_size];
>+  for (size_t i = 0; i < format.size(); i++) {
>+    if (format[i] == '$') {
>+      if (absl::ascii_isdigit(format[i + 1])) {
>+        const absl::string_view src = args_array[format[i + 1] - '0'];
>+        target = std::copy(src.begin(), src.end(), target);
>+        ++i;  // Skip next char.
>+      } else if (format[i + 1] == '$') {
>+        *target++ = '$';
>+        ++i;  // Skip next char.
>+      }
>+    } else {
>+      *target++ = format[i];
>+    }
>+  }
>+
>+  assert(target == output->data() + output->size());
>+}
>+
>+static const char kHexDigits[] = "0123456789abcdef";
>+Arg::Arg(const void* value) {
>+  static_assert(sizeof(scratch_) >= sizeof(value) * 2 + 2,
>+                "fix sizeof(scratch_)");
>+  if (value == nullptr) {
>+    piece_ = "NULL";
>+  } else {
>+    char* ptr = scratch_ + sizeof(scratch_);
>+    uintptr_t num = reinterpret_cast<uintptr_t>(value);
>+    do {
>+      *--ptr = kHexDigits[num & 0xf];
>+      num >>= 4;
>+    } while (num != 0);
>+    *--ptr = 'x';
>+    *--ptr = '0';
>+    piece_ = absl::string_view(ptr, scratch_ + sizeof(scratch_) - ptr);
>+  }
>+}
>+
>+// TODO(jorg): Don't duplicate so much code between here and str_cat.cc
>+Arg::Arg(Hex hex) {
>+  char* const end = &scratch_[numbers_internal::kFastToBufferSize];
>+  char* writer = end;
>+  uint64_t value = hex.value;
>+  do {
>+    *--writer = kHexDigits[value & 0xF];
>+    value >>= 4;
>+  } while (value != 0);
>+
>+  char* beg;
>+  if (end - writer < hex.width) {
>+    beg = end - hex.width;
>+    std::fill_n(beg, writer - beg, hex.fill);
>+  } else {
>+    beg = writer;
>+  }
>+
>+  piece_ = absl::string_view(beg, end - beg);
>+}
>+
>+// TODO(jorg): Don't duplicate so much code between here and str_cat.cc
>+Arg::Arg(Dec dec) {
>+  assert(dec.width <= numbers_internal::kFastToBufferSize);
>+  char* const end = &scratch_[numbers_internal::kFastToBufferSize];
>+  char* const minfill = end - dec.width;
>+  char* writer = end;
>+  uint64_t value = dec.value;
>+  bool neg = dec.neg;
>+  while (value > 9) {
>+    *--writer = '0' + (value % 10);
>+    value /= 10;
>+  }
>+  *--writer = '0' + value;
>+  if (neg) *--writer = '-';
>+
>+  ptrdiff_t fillers = writer - minfill;
>+  if (fillers > 0) {
>+    // Tricky: if the fill character is ' ', then it's <fill><+/-><digits>
>+    // But...: if the fill character is '0', then it's <+/-><fill><digits>
>+    bool add_sign_again = false;
>+    if (neg && dec.fill == '0') {  // If filling with '0',
>+      ++writer;                    // ignore the sign we just added
>+      add_sign_again = true;       // and re-add the sign later.
>+    }
>+    writer -= fillers;
>+    std::fill_n(writer, fillers, dec.fill);
>+    if (add_sign_again) *--writer = '-';
>+  }
>+
>+  piece_ = absl::string_view(writer, end - writer);
>+}
>+
>+}  // namespace substitute_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/substitute.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/substitute.h
>new file mode 100644
>index 00000000000..c4b25ba7095
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/substitute.h
>@@ -0,0 +1,671 @@
>+//
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// -----------------------------------------------------------------------------
>+// File: substitute.h
>+// -----------------------------------------------------------------------------
>+//
>+// This package contains functions for efficiently performing std::string
>+// substitutions using a format std::string with positional notation:
>+// `Substitute()` and `SubstituteAndAppend()`.
>+//
>+// Unlike printf-style format specifiers, `Substitute()` functions do not need
>+// to specify the type of the substitution arguments. Supported arguments
>+// following the format std::string, such as strings, string_views, ints,
>+// floats, and bools, are automatically converted to strings during the
>+// substitution process. (See below for a full list of supported types.)
>+//
>+// `Substitute()` does not allow you to specify *how* to format a value, beyond
>+// the default conversion to std::string. For example, you cannot format an integer
>+// in hex.
>+//
>+// The format std::string uses positional identifiers indicated by a dollar sign ($)
>+// and single digit positional ids to indicate which substitution arguments to
>+// use at that location within the format std::string.
>+//
>+// Example 1:
>+//   std::string s = Substitute("$1 purchased $0 $2. Thanks $1!",
>+//                         5, "Bob", "Apples");
>+//   EXPECT_EQ("Bob purchased 5 Apples. Thanks Bob!", s);
>+//
>+// Example 2:
>+//   std::string s = "Hi. ";
>+//   SubstituteAndAppend(&s, "My name is $0 and I am $1 years old.", "Bob", 5);
>+//   EXPECT_EQ("Hi. My name is Bob and I am 5 years old.", s);
>+//
>+//
>+// Supported types:
>+//   * absl::string_view, std::string, const char* (null is equivalent to "")
>+//   * int32_t, int64_t, uint32_t, uint64
>+//   * float, double
>+//   * bool (Printed as "true" or "false")
>+//   * pointer types other than char* (Printed as "0x<lower case hex std::string>",
>+//     except that null is printed as "NULL")
>+//
>+// If an invalid format std::string is provided, Substitute returns an empty std::string
>+// and SubstituteAndAppend does not change the provided output std::string.
>+// A format std::string is invalid if it:
>+//   * ends in an unescaped $ character,
>+//     e.g. "Hello $", or
>+//   * calls for a position argument which is not provided,
>+//     e.g. Substitute("Hello $2", "world"), or
>+//   * specifies a non-digit, non-$ character after an unescaped $ character,
>+//     e.g. "Hello $f".
>+// In debug mode, i.e. #ifndef NDEBUG, such errors terminate the program.
>+
>+#ifndef ABSL_STRINGS_SUBSTITUTE_H_
>+#define ABSL_STRINGS_SUBSTITUTE_H_
>+
>+#include <cstring>
>+#include <string>
>+
>+#include "absl/base/macros.h"
>+#include "absl/base/port.h"
>+#include "absl/strings/ascii.h"
>+#include "absl/strings/escaping.h"
>+#include "absl/strings/numbers.h"
>+#include "absl/strings/str_cat.h"
>+#include "absl/strings/str_split.h"
>+#include "absl/strings/string_view.h"
>+#include "absl/strings/strip.h"
>+
>+namespace absl {
>+namespace substitute_internal {
>+
>+// Arg
>+//
>+// This class provides an argument type for `absl::Substitute()` and
>+// `absl::SubstituteAndAppend()`. `Arg` handles implicit conversion of various
>+// types to a std::string. (`Arg` is very similar to the `AlphaNum` class in
>+// `StrCat()`.)
>+//
>+// This class has implicit constructors.
>+class Arg {
>+ public:
>+  // Overloads for std::string-y things
>+  //
>+  // Explicitly overload `const char*` so the compiler doesn't cast to `bool`.
>+  Arg(const char* value)  // NOLINT(runtime/explicit)
>+      : piece_(absl::NullSafeStringView(value)) {}
>+  template <typename Allocator>
>+  Arg(  // NOLINT
>+      const std::basic_string<char, std::char_traits<char>, Allocator>&
>+          value) noexcept
>+      : piece_(value) {}
>+  Arg(absl::string_view value)  // NOLINT(runtime/explicit)
>+      : piece_(value) {}
>+
>+  // Overloads for primitives
>+  //
>+  // No overloads are available for signed and unsigned char because if people
>+  // are explicitly declaring their chars as signed or unsigned then they are
>+  // probably using them as 8-bit integers and would probably prefer an integer
>+  // representation. However, we can't really know, so we make the caller decide
>+  // what to do.
>+  Arg(char value)  // NOLINT(runtime/explicit)
>+      : piece_(scratch_, 1) { scratch_[0] = value; }
>+  Arg(short value)  // NOLINT(*)
>+      : piece_(scratch_,
>+               numbers_internal::FastIntToBuffer(value, scratch_) - scratch_) {}
>+  Arg(unsigned short value)  // NOLINT(*)
>+      : piece_(scratch_,
>+               numbers_internal::FastIntToBuffer(value, scratch_) - scratch_) {}
>+  Arg(int value)  // NOLINT(runtime/explicit)
>+      : piece_(scratch_,
>+               numbers_internal::FastIntToBuffer(value, scratch_) - scratch_) {}
>+  Arg(unsigned int value)  // NOLINT(runtime/explicit)
>+      : piece_(scratch_,
>+               numbers_internal::FastIntToBuffer(value, scratch_) - scratch_) {}
>+  Arg(long value)  // NOLINT(*)
>+      : piece_(scratch_,
>+               numbers_internal::FastIntToBuffer(value, scratch_) - scratch_) {}
>+  Arg(unsigned long value)  // NOLINT(*)
>+      : piece_(scratch_,
>+               numbers_internal::FastIntToBuffer(value, scratch_) - scratch_) {}
>+  Arg(long long value)  // NOLINT(*)
>+      : piece_(scratch_,
>+               numbers_internal::FastIntToBuffer(value, scratch_) - scratch_) {}
>+  Arg(unsigned long long value)  // NOLINT(*)
>+      : piece_(scratch_,
>+               numbers_internal::FastIntToBuffer(value, scratch_) - scratch_) {}
>+  Arg(float value)  // NOLINT(runtime/explicit)
>+      : piece_(scratch_, numbers_internal::SixDigitsToBuffer(value, scratch_)) {
>+  }
>+  Arg(double value)  // NOLINT(runtime/explicit)
>+      : piece_(scratch_, numbers_internal::SixDigitsToBuffer(value, scratch_)) {
>+  }
>+  Arg(bool value)  // NOLINT(runtime/explicit)
>+      : piece_(value ? "true" : "false") {}
>+
>+  Arg(Hex hex);  // NOLINT(runtime/explicit)
>+  Arg(Dec dec);  // NOLINT(runtime/explicit)
>+
>+  // `void*` values, with the exception of `char*`, are printed as
>+  // "0x<hex value>". However, in the case of `nullptr`, "NULL" is printed.
>+  Arg(const void* value);  // NOLINT(runtime/explicit)
>+
>+  Arg(const Arg&) = delete;
>+  Arg& operator=(const Arg&) = delete;
>+
>+  absl::string_view piece() const { return piece_; }
>+
>+ private:
>+  absl::string_view piece_;
>+  char scratch_[numbers_internal::kFastToBufferSize];
>+};
>+
>+// Internal helper function. Don't call this from outside this implementation.
>+// This interface may change without notice.
>+void SubstituteAndAppendArray(std::string* output, absl::string_view format,
>+                              const absl::string_view* args_array,
>+                              size_t num_args);
>+
>+#if defined(ABSL_BAD_CALL_IF)
>+constexpr int CalculateOneBit(const char* format) {
>+  return (*format < '0' || *format > '9') ? 0 : (1 << (*format - '0'));
>+}
>+
>+constexpr const char* SkipNumber(const char* format) {
>+  return !*format ? format : (format + 1);
>+}
>+
>+constexpr int PlaceholderBitmask(const char* format) {
>+  return !*format ? 0 : *format != '$'
>+                             ? PlaceholderBitmask(format + 1)
>+                             : (CalculateOneBit(format + 1) |
>+                                   PlaceholderBitmask(SkipNumber(format + 1)));
>+}
>+#endif  // ABSL_BAD_CALL_IF
>+
>+}  // namespace substitute_internal
>+
>+//
>+// PUBLIC API
>+//
>+
>+// SubstituteAndAppend()
>+//
>+// Substitutes variables into a given format std::string and appends to a given
>+// output std::string. See file comments above for usage.
>+//
>+// The declarations of `SubstituteAndAppend()` below consist of overloads
>+// for passing 0 to 10 arguments, respectively.
>+//
>+// NOTE: A zero-argument `SubstituteAndAppend()` may be used within variadic
>+// templates to allow a variable number of arguments.
>+//
>+// Example:
>+//  template <typename... Args>
>+//  void VarMsg(std::string* boilerplate, absl::string_view format,
>+//      const Args&... args) {
>+//    absl::SubstituteAndAppend(boilerplate, format, args...);
>+//  }
>+//
>+inline void SubstituteAndAppend(std::string* output, absl::string_view format) {
>+  substitute_internal::SubstituteAndAppendArray(output, format, nullptr, 0);
>+}
>+
>+inline void SubstituteAndAppend(std::string* output, absl::string_view format,
>+                                const substitute_internal::Arg& a0) {
>+  const absl::string_view args[] = {a0.piece()};
>+  substitute_internal::SubstituteAndAppendArray(output, format, args,
>+                                                ABSL_ARRAYSIZE(args));
>+}
>+
>+inline void SubstituteAndAppend(std::string* output, absl::string_view format,
>+                                const substitute_internal::Arg& a0,
>+                                const substitute_internal::Arg& a1) {
>+  const absl::string_view args[] = {a0.piece(), a1.piece()};
>+  substitute_internal::SubstituteAndAppendArray(output, format, args,
>+                                                ABSL_ARRAYSIZE(args));
>+}
>+
>+inline void SubstituteAndAppend(std::string* output, absl::string_view format,
>+                                const substitute_internal::Arg& a0,
>+                                const substitute_internal::Arg& a1,
>+                                const substitute_internal::Arg& a2) {
>+  const absl::string_view args[] = {a0.piece(), a1.piece(), a2.piece()};
>+  substitute_internal::SubstituteAndAppendArray(output, format, args,
>+                                                ABSL_ARRAYSIZE(args));
>+}
>+
>+inline void SubstituteAndAppend(std::string* output, absl::string_view format,
>+                                const substitute_internal::Arg& a0,
>+                                const substitute_internal::Arg& a1,
>+                                const substitute_internal::Arg& a2,
>+                                const substitute_internal::Arg& a3) {
>+  const absl::string_view args[] = {a0.piece(), a1.piece(), a2.piece(),
>+                                    a3.piece()};
>+  substitute_internal::SubstituteAndAppendArray(output, format, args,
>+                                                ABSL_ARRAYSIZE(args));
>+}
>+
>+inline void SubstituteAndAppend(std::string* output, absl::string_view format,
>+                                const substitute_internal::Arg& a0,
>+                                const substitute_internal::Arg& a1,
>+                                const substitute_internal::Arg& a2,
>+                                const substitute_internal::Arg& a3,
>+                                const substitute_internal::Arg& a4) {
>+  const absl::string_view args[] = {a0.piece(), a1.piece(), a2.piece(),
>+                                    a3.piece(), a4.piece()};
>+  substitute_internal::SubstituteAndAppendArray(output, format, args,
>+                                                ABSL_ARRAYSIZE(args));
>+}
>+
>+inline void SubstituteAndAppend(std::string* output, absl::string_view format,
>+                                const substitute_internal::Arg& a0,
>+                                const substitute_internal::Arg& a1,
>+                                const substitute_internal::Arg& a2,
>+                                const substitute_internal::Arg& a3,
>+                                const substitute_internal::Arg& a4,
>+                                const substitute_internal::Arg& a5) {
>+  const absl::string_view args[] = {a0.piece(), a1.piece(), a2.piece(),
>+                                    a3.piece(), a4.piece(), a5.piece()};
>+  substitute_internal::SubstituteAndAppendArray(output, format, args,
>+                                                ABSL_ARRAYSIZE(args));
>+}
>+
>+inline void SubstituteAndAppend(std::string* output, absl::string_view format,
>+                                const substitute_internal::Arg& a0,
>+                                const substitute_internal::Arg& a1,
>+                                const substitute_internal::Arg& a2,
>+                                const substitute_internal::Arg& a3,
>+                                const substitute_internal::Arg& a4,
>+                                const substitute_internal::Arg& a5,
>+                                const substitute_internal::Arg& a6) {
>+  const absl::string_view args[] = {a0.piece(), a1.piece(), a2.piece(),
>+                                    a3.piece(), a4.piece(), a5.piece(),
>+                                    a6.piece()};
>+  substitute_internal::SubstituteAndAppendArray(output, format, args,
>+                                                ABSL_ARRAYSIZE(args));
>+}
>+
>+inline void SubstituteAndAppend(
>+    std::string* output, absl::string_view format,
>+    const substitute_internal::Arg& a0, const substitute_internal::Arg& a1,
>+    const substitute_internal::Arg& a2, const substitute_internal::Arg& a3,
>+    const substitute_internal::Arg& a4, const substitute_internal::Arg& a5,
>+    const substitute_internal::Arg& a6, const substitute_internal::Arg& a7) {
>+  const absl::string_view args[] = {a0.piece(), a1.piece(), a2.piece(),
>+                                    a3.piece(), a4.piece(), a5.piece(),
>+                                    a6.piece(), a7.piece()};
>+  substitute_internal::SubstituteAndAppendArray(output, format, args,
>+                                                ABSL_ARRAYSIZE(args));
>+}
>+
>+inline void SubstituteAndAppend(
>+    std::string* output, absl::string_view format,
>+    const substitute_internal::Arg& a0, const substitute_internal::Arg& a1,
>+    const substitute_internal::Arg& a2, const substitute_internal::Arg& a3,
>+    const substitute_internal::Arg& a4, const substitute_internal::Arg& a5,
>+    const substitute_internal::Arg& a6, const substitute_internal::Arg& a7,
>+    const substitute_internal::Arg& a8) {
>+  const absl::string_view args[] = {a0.piece(), a1.piece(), a2.piece(),
>+                                    a3.piece(), a4.piece(), a5.piece(),
>+                                    a6.piece(), a7.piece(), a8.piece()};
>+  substitute_internal::SubstituteAndAppendArray(output, format, args,
>+                                                ABSL_ARRAYSIZE(args));
>+}
>+
>+inline void SubstituteAndAppend(
>+    std::string* output, absl::string_view format,
>+    const substitute_internal::Arg& a0, const substitute_internal::Arg& a1,
>+    const substitute_internal::Arg& a2, const substitute_internal::Arg& a3,
>+    const substitute_internal::Arg& a4, const substitute_internal::Arg& a5,
>+    const substitute_internal::Arg& a6, const substitute_internal::Arg& a7,
>+    const substitute_internal::Arg& a8, const substitute_internal::Arg& a9) {
>+  const absl::string_view args[] = {
>+      a0.piece(), a1.piece(), a2.piece(), a3.piece(), a4.piece(),
>+      a5.piece(), a6.piece(), a7.piece(), a8.piece(), a9.piece()};
>+  substitute_internal::SubstituteAndAppendArray(output, format, args,
>+                                                ABSL_ARRAYSIZE(args));
>+}
>+
>+#if defined(ABSL_BAD_CALL_IF)
>+// This body of functions catches cases where the number of placeholders
>+// doesn't match the number of data arguments.
>+void SubstituteAndAppend(std::string* output, const char* format)
>+    ABSL_BAD_CALL_IF(substitute_internal::PlaceholderBitmask(format) != 0,
>+                     "There were no substitution arguments "
>+                     "but this format std::string has a $[0-9] in it");
>+
>+void SubstituteAndAppend(std::string* output, const char* format,
>+                         const substitute_internal::Arg& a0)
>+    ABSL_BAD_CALL_IF(substitute_internal::PlaceholderBitmask(format) != 1,
>+                     "There was 1 substitution argument given, but "
>+                     "this format std::string is either missing its $0, or "
>+                     "contains one of $1-$9");
>+
>+void SubstituteAndAppend(std::string* output, const char* format,
>+                         const substitute_internal::Arg& a0,
>+                         const substitute_internal::Arg& a1)
>+    ABSL_BAD_CALL_IF(substitute_internal::PlaceholderBitmask(format) != 3,
>+                     "There were 2 substitution arguments given, but "
>+                     "this format std::string is either missing its $0/$1, or "
>+                     "contains one of $2-$9");
>+
>+void SubstituteAndAppend(std::string* output, const char* format,
>+                         const substitute_internal::Arg& a0,
>+                         const substitute_internal::Arg& a1,
>+                         const substitute_internal::Arg& a2)
>+    ABSL_BAD_CALL_IF(substitute_internal::PlaceholderBitmask(format) != 7,
>+                     "There were 3 substitution arguments given, but "
>+                     "this format std::string is either missing its $0/$1/$2, or "
>+                     "contains one of $3-$9");
>+
>+void SubstituteAndAppend(std::string* output, const char* format,
>+                         const substitute_internal::Arg& a0,
>+                         const substitute_internal::Arg& a1,
>+                         const substitute_internal::Arg& a2,
>+                         const substitute_internal::Arg& a3)
>+    ABSL_BAD_CALL_IF(substitute_internal::PlaceholderBitmask(format) != 15,
>+                     "There were 4 substitution arguments given, but "
>+                     "this format std::string is either missing its $0-$3, or "
>+                     "contains one of $4-$9");
>+
>+void SubstituteAndAppend(std::string* output, const char* format,
>+                         const substitute_internal::Arg& a0,
>+                         const substitute_internal::Arg& a1,
>+                         const substitute_internal::Arg& a2,
>+                         const substitute_internal::Arg& a3,
>+                         const substitute_internal::Arg& a4)
>+    ABSL_BAD_CALL_IF(substitute_internal::PlaceholderBitmask(format) != 31,
>+                     "There were 5 substitution arguments given, but "
>+                     "this format std::string is either missing its $0-$4, or "
>+                     "contains one of $5-$9");
>+
>+void SubstituteAndAppend(std::string* output, const char* format,
>+                         const substitute_internal::Arg& a0,
>+                         const substitute_internal::Arg& a1,
>+                         const substitute_internal::Arg& a2,
>+                         const substitute_internal::Arg& a3,
>+                         const substitute_internal::Arg& a4,
>+                         const substitute_internal::Arg& a5)
>+    ABSL_BAD_CALL_IF(substitute_internal::PlaceholderBitmask(format) != 63,
>+                     "There were 6 substitution arguments given, but "
>+                     "this format std::string is either missing its $0-$5, or "
>+                     "contains one of $6-$9");
>+
>+void SubstituteAndAppend(
>+    std::string* output, const char* format, const substitute_internal::Arg& a0,
>+    const substitute_internal::Arg& a1, const substitute_internal::Arg& a2,
>+    const substitute_internal::Arg& a3, const substitute_internal::Arg& a4,
>+    const substitute_internal::Arg& a5, const substitute_internal::Arg& a6)
>+    ABSL_BAD_CALL_IF(substitute_internal::PlaceholderBitmask(format) != 127,
>+                     "There were 7 substitution arguments given, but "
>+                     "this format std::string is either missing its $0-$6, or "
>+                     "contains one of $7-$9");
>+
>+void SubstituteAndAppend(
>+    std::string* output, const char* format, const substitute_internal::Arg& a0,
>+    const substitute_internal::Arg& a1, const substitute_internal::Arg& a2,
>+    const substitute_internal::Arg& a3, const substitute_internal::Arg& a4,
>+    const substitute_internal::Arg& a5, const substitute_internal::Arg& a6,
>+    const substitute_internal::Arg& a7)
>+    ABSL_BAD_CALL_IF(substitute_internal::PlaceholderBitmask(format) != 255,
>+                     "There were 8 substitution arguments given, but "
>+                     "this format std::string is either missing its $0-$7, or "
>+                     "contains one of $8-$9");
>+
>+void SubstituteAndAppend(
>+    std::string* output, const char* format, const substitute_internal::Arg& a0,
>+    const substitute_internal::Arg& a1, const substitute_internal::Arg& a2,
>+    const substitute_internal::Arg& a3, const substitute_internal::Arg& a4,
>+    const substitute_internal::Arg& a5, const substitute_internal::Arg& a6,
>+    const substitute_internal::Arg& a7, const substitute_internal::Arg& a8)
>+    ABSL_BAD_CALL_IF(
>+        substitute_internal::PlaceholderBitmask(format) != 511,
>+        "There were 9 substitution arguments given, but "
>+        "this format std::string is either missing its $0-$8, or contains a $9");
>+
>+void SubstituteAndAppend(
>+    std::string* output, const char* format, const substitute_internal::Arg& a0,
>+    const substitute_internal::Arg& a1, const substitute_internal::Arg& a2,
>+    const substitute_internal::Arg& a3, const substitute_internal::Arg& a4,
>+    const substitute_internal::Arg& a5, const substitute_internal::Arg& a6,
>+    const substitute_internal::Arg& a7, const substitute_internal::Arg& a8,
>+    const substitute_internal::Arg& a9)
>+    ABSL_BAD_CALL_IF(substitute_internal::PlaceholderBitmask(format) != 1023,
>+                     "There were 10 substitution arguments given, but this "
>+                     "format std::string doesn't contain all of $0 through $9");
>+#endif  // ABSL_BAD_CALL_IF
>+
>+// Substitute()
>+//
>+// Substitutes variables into a given format std::string. See file comments above
>+// for usage.
>+//
>+// The declarations of `Substitute()` below consist of overloads for passing 0
>+// to 10 arguments, respectively.
>+//
>+// NOTE: A zero-argument `Substitute()` may be used within variadic templates to
>+// allow a variable number of arguments.
>+//
>+// Example:
>+//  template <typename... Args>
>+//  void VarMsg(absl::string_view format, const Args&... args) {
>+//    std::string s = absl::Substitute(format, args...);
>+
>+ABSL_MUST_USE_RESULT inline std::string Substitute(absl::string_view format) {
>+  std::string result;
>+  SubstituteAndAppend(&result, format);
>+  return result;
>+}
>+
>+ABSL_MUST_USE_RESULT inline std::string Substitute(
>+    absl::string_view format, const substitute_internal::Arg& a0) {
>+  std::string result;
>+  SubstituteAndAppend(&result, format, a0);
>+  return result;
>+}
>+
>+ABSL_MUST_USE_RESULT inline std::string Substitute(
>+    absl::string_view format, const substitute_internal::Arg& a0,
>+    const substitute_internal::Arg& a1) {
>+  std::string result;
>+  SubstituteAndAppend(&result, format, a0, a1);
>+  return result;
>+}
>+
>+ABSL_MUST_USE_RESULT inline std::string Substitute(
>+    absl::string_view format, const substitute_internal::Arg& a0,
>+    const substitute_internal::Arg& a1, const substitute_internal::Arg& a2) {
>+  std::string result;
>+  SubstituteAndAppend(&result, format, a0, a1, a2);
>+  return result;
>+}
>+
>+ABSL_MUST_USE_RESULT inline std::string Substitute(
>+    absl::string_view format, const substitute_internal::Arg& a0,
>+    const substitute_internal::Arg& a1, const substitute_internal::Arg& a2,
>+    const substitute_internal::Arg& a3) {
>+  std::string result;
>+  SubstituteAndAppend(&result, format, a0, a1, a2, a3);
>+  return result;
>+}
>+
>+ABSL_MUST_USE_RESULT inline std::string Substitute(
>+    absl::string_view format, const substitute_internal::Arg& a0,
>+    const substitute_internal::Arg& a1, const substitute_internal::Arg& a2,
>+    const substitute_internal::Arg& a3, const substitute_internal::Arg& a4) {
>+  std::string result;
>+  SubstituteAndAppend(&result, format, a0, a1, a2, a3, a4);
>+  return result;
>+}
>+
>+ABSL_MUST_USE_RESULT inline std::string Substitute(
>+    absl::string_view format, const substitute_internal::Arg& a0,
>+    const substitute_internal::Arg& a1, const substitute_internal::Arg& a2,
>+    const substitute_internal::Arg& a3, const substitute_internal::Arg& a4,
>+    const substitute_internal::Arg& a5) {
>+  std::string result;
>+  SubstituteAndAppend(&result, format, a0, a1, a2, a3, a4, a5);
>+  return result;
>+}
>+
>+ABSL_MUST_USE_RESULT inline std::string Substitute(
>+    absl::string_view format, const substitute_internal::Arg& a0,
>+    const substitute_internal::Arg& a1, const substitute_internal::Arg& a2,
>+    const substitute_internal::Arg& a3, const substitute_internal::Arg& a4,
>+    const substitute_internal::Arg& a5, const substitute_internal::Arg& a6) {
>+  std::string result;
>+  SubstituteAndAppend(&result, format, a0, a1, a2, a3, a4, a5, a6);
>+  return result;
>+}
>+
>+ABSL_MUST_USE_RESULT inline std::string Substitute(
>+    absl::string_view format, const substitute_internal::Arg& a0,
>+    const substitute_internal::Arg& a1, const substitute_internal::Arg& a2,
>+    const substitute_internal::Arg& a3, const substitute_internal::Arg& a4,
>+    const substitute_internal::Arg& a5, const substitute_internal::Arg& a6,
>+    const substitute_internal::Arg& a7) {
>+  std::string result;
>+  SubstituteAndAppend(&result, format, a0, a1, a2, a3, a4, a5, a6, a7);
>+  return result;
>+}
>+
>+ABSL_MUST_USE_RESULT inline std::string Substitute(
>+    absl::string_view format, const substitute_internal::Arg& a0,
>+    const substitute_internal::Arg& a1, const substitute_internal::Arg& a2,
>+    const substitute_internal::Arg& a3, const substitute_internal::Arg& a4,
>+    const substitute_internal::Arg& a5, const substitute_internal::Arg& a6,
>+    const substitute_internal::Arg& a7, const substitute_internal::Arg& a8) {
>+  std::string result;
>+  SubstituteAndAppend(&result, format, a0, a1, a2, a3, a4, a5, a6, a7, a8);
>+  return result;
>+}
>+
>+ABSL_MUST_USE_RESULT inline std::string Substitute(
>+    absl::string_view format, const substitute_internal::Arg& a0,
>+    const substitute_internal::Arg& a1, const substitute_internal::Arg& a2,
>+    const substitute_internal::Arg& a3, const substitute_internal::Arg& a4,
>+    const substitute_internal::Arg& a5, const substitute_internal::Arg& a6,
>+    const substitute_internal::Arg& a7, const substitute_internal::Arg& a8,
>+    const substitute_internal::Arg& a9) {
>+  std::string result;
>+  SubstituteAndAppend(&result, format, a0, a1, a2, a3, a4, a5, a6, a7, a8, a9);
>+  return result;
>+}
>+
>+#if defined(ABSL_BAD_CALL_IF)
>+// This body of functions catches cases where the number of placeholders
>+// doesn't match the number of data arguments.
>+std::string Substitute(const char* format)
>+    ABSL_BAD_CALL_IF(substitute_internal::PlaceholderBitmask(format) != 0,
>+                     "There were no substitution arguments "
>+                     "but this format std::string has a $[0-9] in it");
>+
>+std::string Substitute(const char* format, const substitute_internal::Arg& a0)
>+    ABSL_BAD_CALL_IF(substitute_internal::PlaceholderBitmask(format) != 1,
>+                     "There was 1 substitution argument given, but "
>+                     "this format std::string is either missing its $0, or "
>+                     "contains one of $1-$9");
>+
>+std::string Substitute(const char* format, const substitute_internal::Arg& a0,
>+                  const substitute_internal::Arg& a1)
>+    ABSL_BAD_CALL_IF(substitute_internal::PlaceholderBitmask(format) != 3,
>+                     "There were 2 substitution arguments given, but "
>+                     "this format std::string is either missing its $0/$1, or "
>+                     "contains one of $2-$9");
>+
>+std::string Substitute(const char* format, const substitute_internal::Arg& a0,
>+                  const substitute_internal::Arg& a1,
>+                  const substitute_internal::Arg& a2)
>+    ABSL_BAD_CALL_IF(substitute_internal::PlaceholderBitmask(format) != 7,
>+                     "There were 3 substitution arguments given, but "
>+                     "this format std::string is either missing its $0/$1/$2, or "
>+                     "contains one of $3-$9");
>+
>+std::string Substitute(const char* format, const substitute_internal::Arg& a0,
>+                  const substitute_internal::Arg& a1,
>+                  const substitute_internal::Arg& a2,
>+                  const substitute_internal::Arg& a3)
>+    ABSL_BAD_CALL_IF(substitute_internal::PlaceholderBitmask(format) != 15,
>+                     "There were 4 substitution arguments given, but "
>+                     "this format std::string is either missing its $0-$3, or "
>+                     "contains one of $4-$9");
>+
>+std::string Substitute(const char* format, const substitute_internal::Arg& a0,
>+                  const substitute_internal::Arg& a1,
>+                  const substitute_internal::Arg& a2,
>+                  const substitute_internal::Arg& a3,
>+                  const substitute_internal::Arg& a4)
>+    ABSL_BAD_CALL_IF(substitute_internal::PlaceholderBitmask(format) != 31,
>+                     "There were 5 substitution arguments given, but "
>+                     "this format std::string is either missing its $0-$4, or "
>+                     "contains one of $5-$9");
>+
>+std::string Substitute(const char* format, const substitute_internal::Arg& a0,
>+                  const substitute_internal::Arg& a1,
>+                  const substitute_internal::Arg& a2,
>+                  const substitute_internal::Arg& a3,
>+                  const substitute_internal::Arg& a4,
>+                  const substitute_internal::Arg& a5)
>+    ABSL_BAD_CALL_IF(substitute_internal::PlaceholderBitmask(format) != 63,
>+                     "There were 6 substitution arguments given, but "
>+                     "this format std::string is either missing its $0-$5, or "
>+                     "contains one of $6-$9");
>+
>+std::string Substitute(const char* format, const substitute_internal::Arg& a0,
>+                  const substitute_internal::Arg& a1,
>+                  const substitute_internal::Arg& a2,
>+                  const substitute_internal::Arg& a3,
>+                  const substitute_internal::Arg& a4,
>+                  const substitute_internal::Arg& a5,
>+                  const substitute_internal::Arg& a6)
>+    ABSL_BAD_CALL_IF(substitute_internal::PlaceholderBitmask(format) != 127,
>+                     "There were 7 substitution arguments given, but "
>+                     "this format std::string is either missing its $0-$6, or "
>+                     "contains one of $7-$9");
>+
>+std::string Substitute(const char* format, const substitute_internal::Arg& a0,
>+                  const substitute_internal::Arg& a1,
>+                  const substitute_internal::Arg& a2,
>+                  const substitute_internal::Arg& a3,
>+                  const substitute_internal::Arg& a4,
>+                  const substitute_internal::Arg& a5,
>+                  const substitute_internal::Arg& a6,
>+                  const substitute_internal::Arg& a7)
>+    ABSL_BAD_CALL_IF(substitute_internal::PlaceholderBitmask(format) != 255,
>+                     "There were 8 substitution arguments given, but "
>+                     "this format std::string is either missing its $0-$7, or "
>+                     "contains one of $8-$9");
>+
>+std::string Substitute(
>+    const char* format, const substitute_internal::Arg& a0,
>+    const substitute_internal::Arg& a1, const substitute_internal::Arg& a2,
>+    const substitute_internal::Arg& a3, const substitute_internal::Arg& a4,
>+    const substitute_internal::Arg& a5, const substitute_internal::Arg& a6,
>+    const substitute_internal::Arg& a7, const substitute_internal::Arg& a8)
>+    ABSL_BAD_CALL_IF(
>+        substitute_internal::PlaceholderBitmask(format) != 511,
>+        "There were 9 substitution arguments given, but "
>+        "this format std::string is either missing its $0-$8, or contains a $9");
>+
>+std::string Substitute(
>+    const char* format, const substitute_internal::Arg& a0,
>+    const substitute_internal::Arg& a1, const substitute_internal::Arg& a2,
>+    const substitute_internal::Arg& a3, const substitute_internal::Arg& a4,
>+    const substitute_internal::Arg& a5, const substitute_internal::Arg& a6,
>+    const substitute_internal::Arg& a7, const substitute_internal::Arg& a8,
>+    const substitute_internal::Arg& a9)
>+    ABSL_BAD_CALL_IF(substitute_internal::PlaceholderBitmask(format) != 1023,
>+                     "There were 10 substitution arguments given, but this "
>+                     "format std::string doesn't contain all of $0 through $9");
>+#endif  // ABSL_BAD_CALL_IF
>+
>+}  // namespace absl
>+
>+#endif  // ABSL_STRINGS_SUBSTITUTE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/substitute_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/substitute_test.cc
>new file mode 100644
>index 00000000000..144df01ec18
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/substitute_test.cc
>@@ -0,0 +1,192 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/strings/substitute.h"
>+
>+#include <cstdint>
>+
>+#include "gtest/gtest.h"
>+#include "absl/strings/str_cat.h"
>+
>+namespace {
>+
>+TEST(SubstituteTest, Substitute) {
>+  // Basic.
>+  EXPECT_EQ("Hello, world!", absl::Substitute("$0, $1!", "Hello", "world"));
>+
>+  // Non-char* types.
>+  EXPECT_EQ("123 0.2 0.1 foo true false x",
>+            absl::Substitute("$0 $1 $2 $3 $4 $5 $6", 123, 0.2, 0.1f,
>+                             std::string("foo"), true, false, 'x'));
>+
>+  // All int types.
>+  EXPECT_EQ(
>+      "-32767 65535 "
>+      "-1234567890 3234567890 "
>+      "-1234567890 3234567890 "
>+      "-1234567890123456789 9234567890123456789",
>+      absl::Substitute(
>+          "$0 $1 $2 $3 $4 $5 $6 $7",
>+          static_cast<short>(-32767),          // NOLINT(runtime/int)
>+          static_cast<unsigned short>(65535),  // NOLINT(runtime/int)
>+          -1234567890, 3234567890U, -1234567890L, 3234567890UL,
>+          -int64_t{1234567890123456789}, uint64_t{9234567890123456789u}));
>+
>+  // Hex format
>+  EXPECT_EQ("0 1 f ffff0ffff 0123456789abcdef",
>+            absl::Substitute("$0$1$2$3$4 $5",  //
>+                             absl::Hex(0), absl::Hex(1, absl::kSpacePad2),
>+                             absl::Hex(0xf, absl::kSpacePad2),
>+                             absl::Hex(int16_t{-1}, absl::kSpacePad5),
>+                             absl::Hex(int16_t{-1}, absl::kZeroPad5),
>+                             absl::Hex(0x123456789abcdef, absl::kZeroPad16)));
>+
>+  // Dec format
>+  EXPECT_EQ("0 115   -1-0001 81985529216486895",
>+            absl::Substitute("$0$1$2$3$4 $5",  //
>+                             absl::Dec(0), absl::Dec(1, absl::kSpacePad2),
>+                             absl::Dec(0xf, absl::kSpacePad2),
>+                             absl::Dec(int16_t{-1}, absl::kSpacePad5),
>+                             absl::Dec(int16_t{-1}, absl::kZeroPad5),
>+                             absl::Dec(0x123456789abcdef, absl::kZeroPad16)));
>+
>+  // Pointer.
>+  const int* int_p = reinterpret_cast<const int*>(0x12345);
>+  std::string str = absl::Substitute("$0", int_p);
>+  EXPECT_EQ(absl::StrCat("0x", absl::Hex(int_p)), str);
>+
>+  // Volatile Pointer.
>+  // Like C++ streamed I/O, such pointers implicitly become bool
>+  volatile int vol = 237;
>+  volatile int *volatile volptr = &vol;
>+  str = absl::Substitute("$0", volptr);
>+  EXPECT_EQ("true", str);
>+
>+  // null is special. StrCat prints 0x0. Substitute prints NULL.
>+  const uint64_t* null_p = nullptr;
>+  str = absl::Substitute("$0", null_p);
>+  EXPECT_EQ("NULL", str);
>+
>+  // char* is also special.
>+  const char* char_p = "print me";
>+  str = absl::Substitute("$0", char_p);
>+  EXPECT_EQ("print me", str);
>+
>+  char char_buf[16];
>+  strncpy(char_buf, "print me too", sizeof(char_buf));
>+  str = absl::Substitute("$0", char_buf);
>+  EXPECT_EQ("print me too", str);
>+
>+  // null char* is "doubly" special. Represented as the empty std::string.
>+  char_p = nullptr;
>+  str = absl::Substitute("$0", char_p);
>+  EXPECT_EQ("", str);
>+
>+  // Out-of-order.
>+  EXPECT_EQ("b, a, c, b", absl::Substitute("$1, $0, $2, $1", "a", "b", "c"));
>+
>+  // Literal $
>+  EXPECT_EQ("$", absl::Substitute("$$"));
>+
>+  EXPECT_EQ("$1", absl::Substitute("$$1"));
>+
>+  // Test all overloads.
>+  EXPECT_EQ("a", absl::Substitute("$0", "a"));
>+  EXPECT_EQ("a b", absl::Substitute("$0 $1", "a", "b"));
>+  EXPECT_EQ("a b c", absl::Substitute("$0 $1 $2", "a", "b", "c"));
>+  EXPECT_EQ("a b c d", absl::Substitute("$0 $1 $2 $3", "a", "b", "c", "d"));
>+  EXPECT_EQ("a b c d e",
>+            absl::Substitute("$0 $1 $2 $3 $4", "a", "b", "c", "d", "e"));
>+  EXPECT_EQ("a b c d e f", absl::Substitute("$0 $1 $2 $3 $4 $5", "a", "b", "c",
>+                                            "d", "e", "f"));
>+  EXPECT_EQ("a b c d e f g", absl::Substitute("$0 $1 $2 $3 $4 $5 $6", "a", "b",
>+                                              "c", "d", "e", "f", "g"));
>+  EXPECT_EQ("a b c d e f g h",
>+            absl::Substitute("$0 $1 $2 $3 $4 $5 $6 $7", "a", "b", "c", "d", "e",
>+                             "f", "g", "h"));
>+  EXPECT_EQ("a b c d e f g h i",
>+            absl::Substitute("$0 $1 $2 $3 $4 $5 $6 $7 $8", "a", "b", "c", "d",
>+                             "e", "f", "g", "h", "i"));
>+  EXPECT_EQ("a b c d e f g h i j",
>+            absl::Substitute("$0 $1 $2 $3 $4 $5 $6 $7 $8 $9", "a", "b", "c",
>+                             "d", "e", "f", "g", "h", "i", "j"));
>+  EXPECT_EQ("a b c d e f g h i j b0",
>+            absl::Substitute("$0 $1 $2 $3 $4 $5 $6 $7 $8 $9 $10", "a", "b", "c",
>+                             "d", "e", "f", "g", "h", "i", "j"));
>+
>+  const char* null_cstring = nullptr;
>+  EXPECT_EQ("Text: ''", absl::Substitute("Text: '$0'", null_cstring));
>+}
>+
>+TEST(SubstituteTest, SubstituteAndAppend) {
>+  std::string str = "Hello";
>+  absl::SubstituteAndAppend(&str, ", $0!", "world");
>+  EXPECT_EQ("Hello, world!", str);
>+
>+  // Test all overloads.
>+  str.clear();
>+  absl::SubstituteAndAppend(&str, "$0", "a");
>+  EXPECT_EQ("a", str);
>+  str.clear();
>+  absl::SubstituteAndAppend(&str, "$0 $1", "a", "b");
>+  EXPECT_EQ("a b", str);
>+  str.clear();
>+  absl::SubstituteAndAppend(&str, "$0 $1 $2", "a", "b", "c");
>+  EXPECT_EQ("a b c", str);
>+  str.clear();
>+  absl::SubstituteAndAppend(&str, "$0 $1 $2 $3", "a", "b", "c", "d");
>+  EXPECT_EQ("a b c d", str);
>+  str.clear();
>+  absl::SubstituteAndAppend(&str, "$0 $1 $2 $3 $4", "a", "b", "c", "d", "e");
>+  EXPECT_EQ("a b c d e", str);
>+  str.clear();
>+  absl::SubstituteAndAppend(&str, "$0 $1 $2 $3 $4 $5", "a", "b", "c", "d", "e",
>+                            "f");
>+  EXPECT_EQ("a b c d e f", str);
>+  str.clear();
>+  absl::SubstituteAndAppend(&str, "$0 $1 $2 $3 $4 $5 $6", "a", "b", "c", "d",
>+                            "e", "f", "g");
>+  EXPECT_EQ("a b c d e f g", str);
>+  str.clear();
>+  absl::SubstituteAndAppend(&str, "$0 $1 $2 $3 $4 $5 $6 $7", "a", "b", "c", "d",
>+                            "e", "f", "g", "h");
>+  EXPECT_EQ("a b c d e f g h", str);
>+  str.clear();
>+  absl::SubstituteAndAppend(&str, "$0 $1 $2 $3 $4 $5 $6 $7 $8", "a", "b", "c",
>+                            "d", "e", "f", "g", "h", "i");
>+  EXPECT_EQ("a b c d e f g h i", str);
>+  str.clear();
>+  absl::SubstituteAndAppend(&str, "$0 $1 $2 $3 $4 $5 $6 $7 $8 $9", "a", "b",
>+                            "c", "d", "e", "f", "g", "h", "i", "j");
>+  EXPECT_EQ("a b c d e f g h i j", str);
>+}
>+
>+#ifdef GTEST_HAS_DEATH_TEST
>+
>+TEST(SubstituteDeathTest, SubstituteDeath) {
>+  EXPECT_DEBUG_DEATH(
>+      static_cast<void>(absl::Substitute(absl::string_view("-$2"), "a", "b")),
>+      "Invalid strings::Substitute\\(\\) format std::string: asked for \"\\$2\", "
>+      "but only 2 args were given.");
>+  EXPECT_DEBUG_DEATH(
>+      static_cast<void>(absl::Substitute("-$z-")),
>+      "Invalid strings::Substitute\\(\\) format std::string: \"-\\$z-\"");
>+  EXPECT_DEBUG_DEATH(
>+      static_cast<void>(absl::Substitute("-$")),
>+      "Invalid strings::Substitute\\(\\) format std::string: \"-\\$\"");
>+}
>+
>+#endif  // GTEST_HAS_DEATH_TEST
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/testdata/getline-1.txt b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/testdata/getline-1.txt
>new file mode 100644
>index 00000000000..19b909733ae
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/testdata/getline-1.txt
>@@ -0,0 +1,3 @@
>+alpha
>+
>+beta gamma
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/testdata/getline-2.txt b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/testdata/getline-2.txt
>new file mode 100644
>index 00000000000..d6842d8ee36
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/strings/testdata/getline-2.txt
>@@ -0,0 +1 @@
>+one.two.three
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/BUILD.bazel b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/BUILD.bazel
>new file mode 100644
>index 00000000000..8d302e01223
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/BUILD.bazel
>@@ -0,0 +1,234 @@
>+#
>+# Copyright 2017 The Abseil Authors.
>+#
>+# Licensed under the Apache License, Version 2.0 (the "License");
>+# you may not use this file except in compliance with the License.
>+# You may obtain a copy of the License at
>+#
>+#      http://www.apache.org/licenses/LICENSE-2.0
>+#
>+# Unless required by applicable law or agreed to in writing, software
>+# distributed under the License is distributed on an "AS IS" BASIS,
>+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+# See the License for the specific language governing permissions and
>+# limitations under the License.
>+#
>+
>+load(
>+    "//absl:copts.bzl",
>+    "ABSL_DEFAULT_COPTS",
>+    "ABSL_TEST_COPTS",
>+)
>+
>+package(default_visibility = ["//visibility:public"])
>+
>+licenses(["notice"])  # Apache 2.0
>+
>+# Internal data structure for efficiently detecting mutex dependency cycles
>+cc_library(
>+    name = "graphcycles_internal",
>+    srcs = [
>+        "internal/graphcycles.cc",
>+    ],
>+    hdrs = [
>+        "internal/graphcycles.h",
>+    ],
>+    copts = ABSL_DEFAULT_COPTS,
>+    visibility = [
>+        "//absl:__subpackages__",
>+    ],
>+    deps = [
>+        "//absl/base",
>+        "//absl/base:base_internal",
>+        "//absl/base:core_headers",
>+        "//absl/base:malloc_internal",
>+    ],
>+)
>+
>+cc_library(
>+    name = "synchronization",
>+    srcs = [
>+        "barrier.cc",
>+        "blocking_counter.cc",
>+        "internal/create_thread_identity.cc",
>+        "internal/per_thread_sem.cc",
>+        "internal/waiter.cc",
>+        "notification.cc",
>+    ] + select({
>+        "//conditions:default": ["mutex.cc"],
>+    }),
>+    hdrs = [
>+        "barrier.h",
>+        "blocking_counter.h",
>+        "internal/create_thread_identity.h",
>+        "internal/kernel_timeout.h",
>+        "internal/mutex_nonprod.inc",
>+        "internal/per_thread_sem.h",
>+        "internal/waiter.h",
>+        "mutex.h",
>+        "notification.h",
>+    ],
>+    copts = ABSL_DEFAULT_COPTS,
>+    deps = [
>+        ":graphcycles_internal",
>+        "//absl/base",
>+        "//absl/base:base_internal",
>+        "//absl/base:config",
>+        "//absl/base:core_headers",
>+        "//absl/base:dynamic_annotations",
>+        "//absl/base:malloc_internal",
>+        "//absl/debugging:stacktrace",
>+        "//absl/debugging:symbolize",
>+        "//absl/time",
>+    ],
>+)
>+
>+cc_test(
>+    name = "barrier_test",
>+    size = "small",
>+    srcs = ["barrier_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    deps = [
>+        ":synchronization",
>+        "//absl/time",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "blocking_counter_test",
>+    size = "small",
>+    srcs = ["blocking_counter_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    deps = [
>+        ":synchronization",
>+        "//absl/time",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "graphcycles_test",
>+    size = "medium",
>+    srcs = ["internal/graphcycles_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    deps = [
>+        ":graphcycles_internal",
>+        "//absl/base",
>+        "//absl/base:core_headers",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "graphcycles_benchmark",
>+    srcs = ["internal/graphcycles_benchmark.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    tags = [
>+        "benchmark",
>+    ],
>+    deps = [
>+        ":graphcycles_internal",
>+        "//absl/base",
>+        "@com_github_google_benchmark//:benchmark_main",
>+    ],
>+)
>+
>+cc_library(
>+    name = "thread_pool",
>+    testonly = 1,
>+    hdrs = ["internal/thread_pool.h"],
>+    deps = [
>+        ":synchronization",
>+        "//absl/base:core_headers",
>+    ],
>+)
>+
>+cc_test(
>+    name = "mutex_test",
>+    size = "large",
>+    srcs = ["mutex_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    deps = [
>+        ":synchronization",
>+        ":thread_pool",
>+        "//absl/base",
>+        "//absl/base:core_headers",
>+        "//absl/memory",
>+        "//absl/time",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "mutex_benchmark",
>+    srcs = ["mutex_benchmark.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    tags = ["benchmark"],
>+    visibility = ["//visibility:private"],
>+    deps = [
>+        ":synchronization",
>+        ":thread_pool",
>+        "//absl/base",
>+        "@com_github_google_benchmark//:benchmark_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "notification_test",
>+    size = "small",
>+    srcs = ["notification_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    deps = [
>+        ":synchronization",
>+        "//absl/time",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_library(
>+    name = "per_thread_sem_test_common",
>+    testonly = 1,
>+    srcs = ["internal/per_thread_sem_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    deps = [
>+        ":synchronization",
>+        "//absl/base",
>+        "//absl/strings",
>+        "//absl/time",
>+        "@com_google_googletest//:gtest",
>+    ],
>+    alwayslink = 1,
>+)
>+
>+cc_test(
>+    name = "per_thread_sem_test",
>+    size = "medium",
>+    copts = ABSL_TEST_COPTS,
>+    deps = [
>+        ":per_thread_sem_test_common",
>+        ":synchronization",
>+        "//absl/base",
>+        "//absl/strings",
>+        "//absl/time",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "lifetime_test",
>+    srcs = [
>+        "lifetime_test.cc",
>+    ],
>+    copts = ABSL_TEST_COPTS,
>+    linkopts = select({
>+        "//absl:windows": [],
>+        "//conditions:default": ["-pthread"],
>+    }),
>+    tags = ["no_test_ios_x86_64"],
>+    deps = [
>+        ":synchronization",
>+        "//absl/base",
>+        "//absl/base:core_headers",
>+    ],
>+)
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/BUILD.gn
>new file mode 100644
>index 00000000000..3664aa13274
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/BUILD.gn
>@@ -0,0 +1,116 @@
>+# Copyright 2018 The Chromium Authors. All rights reserved.
>+# Use of this source code is governed by a BSD-style license that can be
>+# found in the LICENSE file.
>+
>+import("//build_overrides/build.gni")
>+
>+if (build_with_chromium) {
>+  visibility = [
>+    "//third_party/webrtc/*",
>+    "//third_party/abseil-cpp/*",
>+    "//third_party/googletest:gtest",
>+  ]
>+} else {
>+  visibility = [ "*" ]
>+}
>+
>+source_set("graphcycles_internal") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  sources = [
>+    "internal/graphcycles.cc",
>+  ]
>+  public = [
>+    "internal/graphcycles.h",
>+  ]
>+  deps = [
>+    "../base",
>+    "../base:base_internal",
>+    "../base:core_headers",
>+    "../base:malloc_internal",
>+  ]
>+  visibility = []
>+  visibility += [ "../*" ]
>+}
>+
>+source_set("synchronization") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  sources = [
>+    "barrier.cc",
>+    "blocking_counter.cc",
>+    "internal/create_thread_identity.cc",
>+    "internal/per_thread_sem.cc",
>+    "internal/waiter.cc",
>+    "notification.cc",
>+    "mutex.cc"
>+  ]
>+  public = [
>+    "barrier.h",
>+    "blocking_counter.h",
>+    "internal/create_thread_identity.h",
>+    "internal/kernel_timeout.h",
>+    "internal/mutex_nonprod.inc",
>+    "internal/per_thread_sem.h",
>+    "internal/waiter.h",
>+    "mutex.h",
>+    "notification.h",
>+  ]
>+  deps = [
>+    ":graphcycles_internal",
>+    "../base",
>+    "../base:base_internal",
>+    "../base:config",
>+    "../base:core_headers",
>+    "../base:dynamic_annotations",
>+    "../base:malloc_internal",
>+    "../debugging:stacktrace",
>+    "../debugging:symbolize",
>+    "../time",
>+  ]
>+}
>+
>+source_set("thread_pool") {
>+  testonly = true
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  public = [
>+    "internal/thread_pool.h",
>+  ]
>+  deps = [
>+    ":synchronization",
>+    "../base:core_headers",
>+  ]
>+}
>+
>+source_set("per_thread_sem_test_common") {
>+  testonly = true
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_test_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  sources = [
>+    "internal/per_thread_sem_test.cc",
>+  ]
>+  deps = [
>+    ":synchronization",
>+    "../base",
>+    "../strings",
>+    "../time",
>+    "//testing/gtest",
>+  ]
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/CMakeLists.txt b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/CMakeLists.txt
>new file mode 100644
>index 00000000000..de0d7b7d450
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/CMakeLists.txt
>@@ -0,0 +1,155 @@
>+#
>+# Copyright 2017 The Abseil Authors.
>+#
>+# Licensed under the Apache License, Version 2.0 (the "License");
>+# you may not use this file except in compliance with the License.
>+# You may obtain a copy of the License at
>+#
>+#      http://www.apache.org/licenses/LICENSE-2.0
>+#
>+# Unless required by applicable law or agreed to in writing, software
>+# distributed under the License is distributed on an "AS IS" BASIS,
>+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+# See the License for the specific language governing permissions and
>+# limitations under the License.
>+#
>+
>+list(APPEND SYNCHRONIZATION_PUBLIC_HEADERS
>+  "barrier.h"
>+  "blocking_counter.h"
>+  "mutex.h"
>+  "notification.h"
>+)
>+
>+
>+list(APPEND SYNCHRONIZATION_INTERNAL_HEADERS
>+  "internal/create_thread_identity.h"
>+  "internal/graphcycles.h"
>+  "internal/kernel_timeout.h"
>+  "internal/per_thread_sem.h"
>+  "internal/thread_pool.h"
>+  "internal/waiter.h"
>+)
>+
>+
>+
>+# synchronization library
>+list(APPEND SYNCHRONIZATION_SRC
>+  "barrier.cc"
>+  "blocking_counter.cc"
>+  "internal/create_thread_identity.cc"
>+  "internal/per_thread_sem.cc"
>+  "internal/waiter.cc"
>+  "internal/graphcycles.cc"
>+  "notification.cc"
>+  "mutex.cc"
>+)
>+
>+set(SYNCHRONIZATION_PUBLIC_LIBRARIES absl::base absl::stacktrace absl::symbolize absl::time)
>+
>+absl_library(
>+  TARGET
>+    absl_synchronization
>+  SOURCES
>+    ${SYNCHRONIZATION_SRC}
>+  PUBLIC_LIBRARIES
>+    ${SYNCHRONIZATION_PUBLIC_LIBRARIES}
>+  EXPORT_NAME
>+    synchronization
>+)
>+
>+
>+#
>+## TESTS
>+#
>+
>+
>+# test barrier_test
>+set(BARRIER_TEST_SRC "barrier_test.cc")
>+set(BARRIER_TEST_PUBLIC_LIBRARIES absl::synchronization)
>+
>+absl_test(
>+  TARGET
>+    barrier_test
>+  SOURCES
>+    ${BARRIER_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${BARRIER_TEST_PUBLIC_LIBRARIES}
>+)
>+
>+
>+# test blocking_counter_test
>+set(BLOCKING_COUNTER_TEST_SRC "blocking_counter_test.cc")
>+set(BLOCKING_COUNTER_TEST_PUBLIC_LIBRARIES absl::synchronization)
>+
>+absl_test(
>+  TARGET
>+    blocking_counter_test
>+  SOURCES
>+    ${BLOCKING_COUNTER_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${BLOCKING_COUNTER_TEST_PUBLIC_LIBRARIES}
>+)
>+
>+
>+# test graphcycles_test
>+set(GRAPHCYCLES_TEST_SRC "internal/graphcycles_test.cc")
>+set(GRAPHCYCLES_TEST_PUBLIC_LIBRARIES absl::synchronization)
>+
>+absl_test(
>+  TARGET
>+    graphcycles_test
>+  SOURCES
>+    ${GRAPHCYCLES_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${GRAPHCYCLES_TEST_PUBLIC_LIBRARIES}
>+)
>+
>+
>+# test mutex_test
>+set(MUTEX_TEST_SRC "mutex_test.cc")
>+set(MUTEX_TEST_PUBLIC_LIBRARIES absl::synchronization)
>+
>+absl_test(
>+  TARGET
>+    mutex_test
>+  SOURCES
>+    ${MUTEX_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${MUTEX_TEST_PUBLIC_LIBRARIES}
>+)
>+
>+
>+# test notification_test
>+set(NOTIFICATION_TEST_SRC "notification_test.cc")
>+set(NOTIFICATION_TEST_PUBLIC_LIBRARIES absl::synchronization)
>+
>+absl_test(
>+  TARGET
>+    notification_test
>+  SOURCES
>+    ${NOTIFICATION_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${NOTIFICATION_TEST_PUBLIC_LIBRARIES}
>+)
>+
>+
>+# test per_thread_sem_test_common
>+set(PER_THREAD_SEM_TEST_COMMON_SRC "internal/per_thread_sem_test.cc")
>+set(PER_THREAD_SEM_TEST_COMMON_PUBLIC_LIBRARIES absl::synchronization absl::strings)
>+
>+absl_test(
>+  TARGET
>+    per_thread_sem_test_common
>+  SOURCES
>+    ${PER_THREAD_SEM_TEST_COMMON_SRC}
>+  PUBLIC_LIBRARIES
>+    ${PER_THREAD_SEM_TEST_COMMON_PUBLIC_LIBRARIES}
>+)
>+
>+
>+
>+
>+
>+
>+
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/barrier.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/barrier.cc
>new file mode 100644
>index 00000000000..a1b3ad5c217
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/barrier.cc
>@@ -0,0 +1,50 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/synchronization/barrier.h"
>+
>+#include "absl/base/internal/raw_logging.h"
>+#include "absl/synchronization/mutex.h"
>+
>+namespace absl {
>+
>+// Return whether int *arg is zero.
>+static bool IsZero(void *arg) {
>+  return 0 == *reinterpret_cast<int *>(arg);
>+}
>+
>+bool Barrier::Block() {
>+  MutexLock l(&this->lock_);
>+
>+  this->num_to_block_--;
>+  if (this->num_to_block_ < 0) {
>+    ABSL_RAW_LOG(
>+        FATAL,
>+        "Block() called too many times.  num_to_block_=%d out of total=%d",
>+        this->num_to_block_, this->num_to_exit_);
>+  }
>+
>+  this->lock_.Await(Condition(IsZero, &this->num_to_block_));
>+
>+  // Determine which thread can safely delete this Barrier object
>+  this->num_to_exit_--;
>+  ABSL_RAW_CHECK(this->num_to_exit_ >= 0, "barrier underflow");
>+
>+  // If num_to_exit_ == 0 then all other threads in the barrier have
>+  // exited the Wait() and have released the Mutex so this thread is
>+  // free to delete the barrier.
>+  return this->num_to_exit_ == 0;
>+}
>+
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/barrier.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/barrier.h
>new file mode 100644
>index 00000000000..f834feec11d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/barrier.h
>@@ -0,0 +1,77 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// -----------------------------------------------------------------------------
>+// barrier.h
>+// -----------------------------------------------------------------------------
>+
>+#ifndef ABSL_SYNCHRONIZATION_BARRIER_H_
>+#define ABSL_SYNCHRONIZATION_BARRIER_H_
>+
>+#include "absl/base/thread_annotations.h"
>+#include "absl/synchronization/mutex.h"
>+
>+namespace absl {
>+
>+// Barrier
>+//
>+// This class creates a barrier which blocks threads until a prespecified
>+// threshold of threads (`num_threads`) utilizes the barrier. A thread utilizes
>+// the `Barrier` by calling `Block()` on the barrier, which will block that
>+// thread; no call to `Block()` will return until `num_threads` threads have
>+// called it.
>+//
>+// Exactly one call to `Block()` will return `true`, which is then responsible
>+// for destroying the barrier; because stack allocation will cause the barrier
>+// to be deleted when it is out of scope, barriers should not be stack
>+// allocated.
>+//
>+// Example:
>+//
>+//   // Main thread creates a `Barrier`:
>+//   barrier = new Barrier(num_threads);
>+//
>+//   // Each participating thread could then call:
>+//   if (barrier->Block()) delete barrier;  // Exactly one call to `Block()`
>+//                                          // returns `true`; that call
>+//                                          // deletes the barrier.
>+class Barrier {
>+ public:
>+  // `num_threads` is the number of threads that will participate in the barrier
>+  explicit Barrier(int num_threads)
>+      : num_to_block_(num_threads), num_to_exit_(num_threads) {}
>+
>+  Barrier(const Barrier&) = delete;
>+  Barrier& operator=(const Barrier&) = delete;
>+
>+  // Barrier::Block()
>+  //
>+  // Blocks the current thread, and returns only when the `num_threads`
>+  // threshold of threads utilizing this barrier has been reached. `Block()`
>+  // returns `true` for precisely one caller, which may then destroy the
>+  // barrier.
>+  //
>+  // Memory ordering: For any threads X and Y, any action taken by X
>+  // before X calls `Block()` will be visible to Y after Y returns from
>+  // `Block()`.
>+  bool Block();
>+
>+ private:
>+  Mutex lock_;
>+  int num_to_block_ GUARDED_BY(lock_);
>+  int num_to_exit_ GUARDED_BY(lock_);
>+};
>+
>+}  // namespace absl
>+#endif  // ABSL_SYNCHRONIZATION_BARRIER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/barrier_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/barrier_test.cc
>new file mode 100644
>index 00000000000..d6cababdf73
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/barrier_test.cc
>@@ -0,0 +1,75 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/synchronization/barrier.h"
>+
>+#include <thread>  // NOLINT(build/c++11)
>+#include <vector>
>+
>+#include "gtest/gtest.h"
>+#include "absl/synchronization/mutex.h"
>+#include "absl/time/clock.h"
>+
>+
>+TEST(Barrier, SanityTest) {
>+  constexpr int kNumThreads = 10;
>+  absl::Barrier* barrier = new absl::Barrier(kNumThreads);
>+
>+  absl::Mutex mutex;
>+  int counter = 0;  // Guarded by mutex.
>+
>+  auto thread_func = [&] {
>+    if (barrier->Block()) {
>+      // This thread is the last thread to reach the barrier so it is
>+      // responsible for deleting it.
>+      delete barrier;
>+    }
>+
>+    // Increment the counter.
>+    absl::MutexLock lock(&mutex);
>+    ++counter;
>+  };
>+
>+  // Start (kNumThreads - 1) threads running thread_func.
>+  std::vector<std::thread> threads;
>+  for (int i = 0; i < kNumThreads - 1; ++i) {
>+    threads.push_back(std::thread(thread_func));
>+  }
>+
>+  // Give (kNumThreads - 1) threads a chance to reach the barrier.
>+  // This test assumes at least one thread will have run after the
>+  // sleep has elapsed. Sleeping in a test is usually bad form, but we
>+  // need to make sure that we are testing the barrier instead of some
>+  // other synchronization method.
>+  absl::SleepFor(absl::Seconds(1));
>+
>+  // The counter should still be zero since no thread should have
>+  // been able to pass the barrier yet.
>+  {
>+    absl::MutexLock lock(&mutex);
>+    EXPECT_EQ(counter, 0);
>+  }
>+
>+  // Start 1 more thread. This should make all threads pass the barrier.
>+  threads.push_back(std::thread(thread_func));
>+
>+  // All threads should now be able to proceed and finish.
>+  for (auto& thread : threads) {
>+    thread.join();
>+  }
>+
>+  // All threads should now have incremented the counter.
>+  absl::MutexLock lock(&mutex);
>+  EXPECT_EQ(counter, kNumThreads);
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/blocking_counter.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/blocking_counter.cc
>new file mode 100644
>index 00000000000..7e68e960bc3
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/blocking_counter.cc
>@@ -0,0 +1,55 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/synchronization/blocking_counter.h"
>+
>+#include "absl/base/internal/raw_logging.h"
>+
>+namespace absl {
>+
>+// Return whether int *arg is zero.
>+static bool IsZero(void *arg) {
>+  return 0 == *reinterpret_cast<int *>(arg);
>+}
>+
>+bool BlockingCounter::DecrementCount() {
>+  MutexLock l(&lock_);
>+  count_--;
>+  if (count_ < 0) {
>+    ABSL_RAW_LOG(
>+        FATAL,
>+        "BlockingCounter::DecrementCount() called too many times.  count=%d",
>+        count_);
>+  }
>+  return count_ == 0;
>+}
>+
>+void BlockingCounter::Wait() {
>+  MutexLock l(&this->lock_);
>+  ABSL_RAW_CHECK(count_ >= 0, "BlockingCounter underflow");
>+
>+  // only one thread may call Wait(). To support more than one thread,
>+  // implement a counter num_to_exit, like in the Barrier class.
>+  ABSL_RAW_CHECK(num_waiting_ == 0, "multiple threads called Wait()");
>+  num_waiting_++;
>+
>+  this->lock_.Await(Condition(IsZero, &this->count_));
>+
>+  // At this point, We know that all threads executing DecrementCount have
>+  // released the lock, and so will not touch this object again.
>+  // Therefore, the thread calling this method is free to delete the object
>+  // after we return from this method.
>+}
>+
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/blocking_counter.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/blocking_counter.h
>new file mode 100644
>index 00000000000..557ed028fe5
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/blocking_counter.h
>@@ -0,0 +1,97 @@
>+//
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// -----------------------------------------------------------------------------
>+// blocking_counter.h
>+// -----------------------------------------------------------------------------
>+
>+#ifndef ABSL_SYNCHRONIZATION_BLOCKING_COUNTER_H_
>+#define ABSL_SYNCHRONIZATION_BLOCKING_COUNTER_H_
>+
>+#include "absl/base/thread_annotations.h"
>+#include "absl/synchronization/mutex.h"
>+
>+namespace absl {
>+
>+// BlockingCounter
>+//
>+// This class allows a thread to block for a pre-specified number of actions.
>+// `BlockingCounter` maintains a single non-negative abstract integer "count"
>+// with an initial value `initial_count`. A thread can then call `Wait()` on
>+// this blocking counter to block until the specified number of events occur;
>+// worker threads then call 'DecrementCount()` on the counter upon completion of
>+// their work. Once the counter's internal "count" reaches zero, the blocked
>+// thread unblocks.
>+//
>+// A `BlockingCounter` requires the following:
>+//     - its `initial_count` is non-negative.
>+//     - the number of calls to `DecrementCount()` on it is at most
>+//       `initial_count`.
>+//     - `Wait()` is called at most once on it.
>+//
>+// Given the above requirements, a `BlockingCounter` provides the following
>+// guarantees:
>+//     - Once its internal "count" reaches zero, no legal action on the object
>+//       can further change the value of "count".
>+//     - When `Wait()` returns, it is legal to destroy the `BlockingCounter`.
>+//     - When `Wait()` returns, the number of calls to `DecrementCount()` on
>+//       this blocking counter exactly equals `initial_count`.
>+//
>+// Example:
>+//     BlockingCounter bcount(N);         // there are N items of work
>+//     ... Allow worker threads to start.
>+//     ... On completing each work item, workers do:
>+//     ... bcount.DecrementCount();      // an item of work has been completed
>+//
>+//     bcount.Wait();                    // wait for all work to be complete
>+//
>+class BlockingCounter {
>+ public:
>+  explicit BlockingCounter(int initial_count)
>+      : count_(initial_count), num_waiting_(0) {}
>+
>+  BlockingCounter(const BlockingCounter&) = delete;
>+  BlockingCounter& operator=(const BlockingCounter&) = delete;
>+
>+  // BlockingCounter::DecrementCount()
>+  //
>+  // Decrements the counter's "count" by one, and return "count == 0". This
>+  // function requires that "count != 0" when it is called.
>+  //
>+  // Memory ordering: For any threads X and Y, any action taken by X
>+  // before it calls `DecrementCount()` is visible to thread Y after
>+  // Y's call to `DecrementCount()`, provided Y's call returns `true`.
>+  bool DecrementCount();
>+
>+  // BlockingCounter::Wait()
>+  //
>+  // Blocks until the counter reaches zero. This function may be called at most
>+  // once. On return, `DecrementCount()` will have been called "initial_count"
>+  // times and the blocking counter may be destroyed.
>+  //
>+  // Memory ordering: For any threads X and Y, any action taken by X
>+  // before X calls `DecrementCount()` is visible to Y after Y returns
>+  // from `Wait()`.
>+  void Wait();
>+
>+ private:
>+  Mutex lock_;
>+  int count_ GUARDED_BY(lock_);
>+  int num_waiting_ GUARDED_BY(lock_);
>+};
>+
>+}  // namespace absl
>+
>+#endif  // ABSL_SYNCHRONIZATION_BLOCKING_COUNTER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/blocking_counter_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/blocking_counter_test.cc
>new file mode 100644
>index 00000000000..e8223f841a5
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/blocking_counter_test.cc
>@@ -0,0 +1,66 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/synchronization/blocking_counter.h"
>+
>+#include <thread>  // NOLINT(build/c++11)
>+#include <vector>
>+
>+#include "gtest/gtest.h"
>+#include "absl/time/clock.h"
>+#include "absl/time/time.h"
>+
>+namespace absl {
>+namespace {
>+
>+void PauseAndDecreaseCounter(BlockingCounter* counter, int* done) {
>+  absl::SleepFor(absl::Seconds(1));
>+  *done = 1;
>+  counter->DecrementCount();
>+}
>+
>+TEST(BlockingCounterTest, BasicFunctionality) {
>+  // This test verifies that BlockingCounter functions correctly. Starts a
>+  // number of threads that just sleep for a second and decrement a counter.
>+
>+  // Initialize the counter.
>+  const int num_workers = 10;
>+  BlockingCounter counter(num_workers);
>+
>+  std::vector<std::thread> workers;
>+  std::vector<int> done(num_workers, 0);
>+
>+  // Start a number of parallel tasks that will just wait for a seconds and
>+  // then decrement the count.
>+  workers.reserve(num_workers);
>+  for (int k = 0; k < num_workers; k++) {
>+    workers.emplace_back(
>+        [&counter, &done, k] { PauseAndDecreaseCounter(&counter, &done[k]); });
>+  }
>+
>+  // Wait for the threads to have all finished.
>+  counter.Wait();
>+
>+  // Check that all the workers have completed.
>+  for (int k = 0; k < num_workers; k++) {
>+    EXPECT_EQ(1, done[k]);
>+  }
>+
>+  for (std::thread& w : workers) {
>+    w.join();
>+  }
>+}
>+
>+}  // namespace
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/internal/create_thread_identity.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/internal/create_thread_identity.cc
>new file mode 100644
>index 00000000000..e7a65cd884a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/internal/create_thread_identity.cc
>@@ -0,0 +1,112 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include <stdint.h>
>+#include <new>
>+
>+// This file is a no-op if the required LowLevelAlloc support is missing.
>+#include "absl/base/internal/low_level_alloc.h"
>+#ifndef ABSL_LOW_LEVEL_ALLOC_MISSING
>+
>+#include <string.h>
>+
>+#include "absl/base/attributes.h"
>+#include "absl/base/internal/spinlock.h"
>+#include "absl/base/internal/thread_identity.h"
>+#include "absl/synchronization/internal/per_thread_sem.h"
>+
>+namespace absl {
>+namespace synchronization_internal {
>+
>+// ThreadIdentity storage is persistent, we maintain a free-list of previously
>+// released ThreadIdentity objects.
>+static base_internal::SpinLock freelist_lock(base_internal::kLinkerInitialized);
>+static base_internal::ThreadIdentity* thread_identity_freelist;
>+
>+// A per-thread destructor for reclaiming associated ThreadIdentity objects.
>+// Since we must preserve their storage we cache them for re-use.
>+static void ReclaimThreadIdentity(void* v) {
>+  base_internal::ThreadIdentity* identity =
>+      static_cast<base_internal::ThreadIdentity*>(v);
>+
>+  // all_locks might have been allocated by the Mutex implementation.
>+  // We free it here when we are notified that our thread is dying.
>+  if (identity->per_thread_synch.all_locks != nullptr) {
>+    base_internal::LowLevelAlloc::Free(identity->per_thread_synch.all_locks);
>+  }
>+
>+  // We must explicitly clear the current thread's identity:
>+  // (a) Subsequent (unrelated) per-thread destructors may require an identity.
>+  //     We must guarantee a new identity is used in this case (this instructor
>+  //     will be reinvoked up to PTHREAD_DESTRUCTOR_ITERATIONS in this case).
>+  // (b) ThreadIdentity implementations may depend on memory that is not
>+  //     reinitialized before reuse.  We must allow explicit clearing of the
>+  //     association state in this case.
>+  base_internal::ClearCurrentThreadIdentity();
>+  {
>+    base_internal::SpinLockHolder l(&freelist_lock);
>+    identity->next = thread_identity_freelist;
>+    thread_identity_freelist = identity;
>+  }
>+}
>+
>+// Return value rounded up to next multiple of align.
>+// Align must be a power of two.
>+static intptr_t RoundUp(intptr_t addr, intptr_t align) {
>+  return (addr + align - 1) & ~(align - 1);
>+}
>+
>+static base_internal::ThreadIdentity* NewThreadIdentity() {
>+  base_internal::ThreadIdentity* identity = nullptr;
>+
>+  {
>+    // Re-use a previously released object if possible.
>+    base_internal::SpinLockHolder l(&freelist_lock);
>+    if (thread_identity_freelist) {
>+      identity = thread_identity_freelist;  // Take list-head.
>+      thread_identity_freelist = thread_identity_freelist->next;
>+    }
>+  }
>+
>+  if (identity == nullptr) {
>+    // Allocate enough space to align ThreadIdentity to a multiple of
>+    // PerThreadSynch::kAlignment. This space is never released (it is
>+    // added to a freelist by ReclaimThreadIdentity instead).
>+    void* allocation = base_internal::LowLevelAlloc::Alloc(
>+        sizeof(*identity) + base_internal::PerThreadSynch::kAlignment - 1);
>+    // Round up the address to the required alignment.
>+    identity = reinterpret_cast<base_internal::ThreadIdentity*>(
>+        RoundUp(reinterpret_cast<intptr_t>(allocation),
>+                base_internal::PerThreadSynch::kAlignment));
>+  }
>+  memset(identity, 0, sizeof(*identity));
>+
>+  return identity;
>+}
>+
>+// Allocates and attaches ThreadIdentity object for the calling thread.  Returns
>+// the new identity.
>+// REQUIRES: CurrentThreadIdentity(false) == nullptr
>+base_internal::ThreadIdentity* CreateThreadIdentity() {
>+  base_internal::ThreadIdentity* identity = NewThreadIdentity();
>+  PerThreadSem::Init(identity);
>+  // Associate the value with the current thread, and attach our destructor.
>+  base_internal::SetCurrentThreadIdentity(identity, ReclaimThreadIdentity);
>+  return identity;
>+}
>+
>+}  // namespace synchronization_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_LOW_LEVEL_ALLOC_MISSING
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/internal/create_thread_identity.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/internal/create_thread_identity.h
>new file mode 100644
>index 00000000000..1bb87dee630
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/internal/create_thread_identity.h
>@@ -0,0 +1,53 @@
>+/*
>+ * Copyright 2017 The Abseil Authors.
>+ *
>+ * Licensed under the Apache License, Version 2.0 (the "License");
>+ * you may not use this file except in compliance with the License.
>+ * You may obtain a copy of the License at
>+ *
>+ *      http://www.apache.org/licenses/LICENSE-2.0
>+ *
>+ * Unless required by applicable law or agreed to in writing, software
>+ * distributed under the License is distributed on an "AS IS" BASIS,
>+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+ * See the License for the specific language governing permissions and
>+ * limitations under the License.
>+ */
>+
>+// Interface for getting the current ThreadIdentity, creating one if necessary.
>+// See thread_identity.h.
>+//
>+// This file is separate from thread_identity.h because creating a new
>+// ThreadIdentity requires slightly higher level libraries (per_thread_sem
>+// and low_level_alloc) than accessing an existing one.  This separation allows
>+// us to have a smaller //absl/base:base.
>+
>+#ifndef ABSL_SYNCHRONIZATION_INTERNAL_CREATE_THREAD_IDENTITY_H_
>+#define ABSL_SYNCHRONIZATION_INTERNAL_CREATE_THREAD_IDENTITY_H_
>+
>+#include "absl/base/internal/thread_identity.h"
>+#include "absl/base/port.h"
>+
>+namespace absl {
>+namespace synchronization_internal {
>+
>+// Allocates and attaches a ThreadIdentity object for the calling thread.
>+// For private use only.
>+base_internal::ThreadIdentity* CreateThreadIdentity();
>+
>+// Returns the ThreadIdentity object representing the calling thread; guaranteed
>+// to be unique for its lifetime.  The returned object will remain valid for the
>+// program's lifetime; although it may be re-assigned to a subsequent thread.
>+// If one does not exist for the calling thread, allocate it now.
>+inline base_internal::ThreadIdentity* GetOrCreateCurrentThreadIdentity() {
>+  base_internal::ThreadIdentity* identity =
>+      base_internal::CurrentThreadIdentityIfPresent();
>+  if (ABSL_PREDICT_FALSE(identity == nullptr)) {
>+    return CreateThreadIdentity();
>+  }
>+  return identity;
>+}
>+
>+}  // namespace synchronization_internal
>+}  // namespace absl
>+#endif  // ABSL_SYNCHRONIZATION_INTERNAL_CREATE_THREAD_IDENTITY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/internal/graphcycles.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/internal/graphcycles.cc
>new file mode 100644
>index 00000000000..ab1f3f84e74
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/internal/graphcycles.cc
>@@ -0,0 +1,699 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+// GraphCycles provides incremental cycle detection on a dynamic
>+// graph using the following algorithm:
>+//
>+// A dynamic topological sort algorithm for directed acyclic graphs
>+// David J. Pearce, Paul H. J. Kelly
>+// Journal of Experimental Algorithmics (JEA) JEA Homepage archive
>+// Volume 11, 2006, Article No. 1.7
>+//
>+// Brief summary of the algorithm:
>+//
>+// (1) Maintain a rank for each node that is consistent
>+//     with the topological sort of the graph. I.e., path from x to y
>+//     implies rank[x] < rank[y].
>+// (2) When a new edge (x->y) is inserted, do nothing if rank[x] < rank[y].
>+// (3) Otherwise: adjust ranks in the neighborhood of x and y.
>+
>+#include "absl/base/attributes.h"
>+// This file is a no-op if the required LowLevelAlloc support is missing.
>+#include "absl/base/internal/low_level_alloc.h"
>+#ifndef ABSL_LOW_LEVEL_ALLOC_MISSING
>+
>+#include "absl/synchronization/internal/graphcycles.h"
>+
>+#include <algorithm>
>+#include <array>
>+#include "absl/base/internal/hide_ptr.h"
>+#include "absl/base/internal/raw_logging.h"
>+#include "absl/base/internal/spinlock.h"
>+
>+// Do not use STL.   This module does not use standard memory allocation.
>+
>+namespace absl {
>+namespace synchronization_internal {
>+
>+namespace {
>+
>+// Avoid LowLevelAlloc's default arena since it calls malloc hooks in
>+// which people are doing things like acquiring Mutexes.
>+static absl::base_internal::SpinLock arena_mu(
>+    absl::base_internal::kLinkerInitialized);
>+static base_internal::LowLevelAlloc::Arena* arena;
>+
>+static void InitArenaIfNecessary() {
>+  arena_mu.Lock();
>+  if (arena == nullptr) {
>+    arena = base_internal::LowLevelAlloc::NewArena(0);
>+  }
>+  arena_mu.Unlock();
>+}
>+
>+// Number of inlined elements in Vec.  Hash table implementation
>+// relies on this being a power of two.
>+static const uint32_t kInline = 8;
>+
>+// A simple LowLevelAlloc based resizable vector with inlined storage
>+// for a few elements.  T must be a plain type since constructor
>+// and destructor are not run on elements of type T managed by Vec.
>+template <typename T>
>+class Vec {
>+ public:
>+  Vec() { Init(); }
>+  ~Vec() { Discard(); }
>+
>+  void clear() {
>+    Discard();
>+    Init();
>+  }
>+
>+  bool empty() const { return size_ == 0; }
>+  uint32_t size() const { return size_; }
>+  T* begin() { return ptr_; }
>+  T* end() { return ptr_ + size_; }
>+  const T& operator[](uint32_t i) const { return ptr_[i]; }
>+  T& operator[](uint32_t i) { return ptr_[i]; }
>+  const T& back() const { return ptr_[size_-1]; }
>+  void pop_back() { size_--; }
>+
>+  void push_back(const T& v) {
>+    if (size_ == capacity_) Grow(size_ + 1);
>+    ptr_[size_] = v;
>+    size_++;
>+  }
>+
>+  void resize(uint32_t n) {
>+    if (n > capacity_) Grow(n);
>+    size_ = n;
>+  }
>+
>+  void fill(const T& val) {
>+    for (uint32_t i = 0; i < size(); i++) {
>+      ptr_[i] = val;
>+    }
>+  }
>+
>+  // Guarantees src is empty at end.
>+  // Provided for the hash table resizing code below.
>+  void MoveFrom(Vec<T>* src) {
>+    if (src->ptr_ == src->space_) {
>+      // Need to actually copy
>+      resize(src->size_);
>+      std::copy(src->ptr_, src->ptr_ + src->size_, ptr_);
>+      src->size_ = 0;
>+    } else {
>+      Discard();
>+      ptr_ = src->ptr_;
>+      size_ = src->size_;
>+      capacity_ = src->capacity_;
>+      src->Init();
>+    }
>+  }
>+
>+ private:
>+  T* ptr_;
>+  T space_[kInline];
>+  uint32_t size_;
>+  uint32_t capacity_;
>+
>+  void Init() {
>+    ptr_ = space_;
>+    size_ = 0;
>+    capacity_ = kInline;
>+  }
>+
>+  void Discard() {
>+    if (ptr_ != space_) base_internal::LowLevelAlloc::Free(ptr_);
>+  }
>+
>+  void Grow(uint32_t n) {
>+    while (capacity_ < n) {
>+      capacity_ *= 2;
>+    }
>+    size_t request = static_cast<size_t>(capacity_) * sizeof(T);
>+    T* copy = static_cast<T*>(
>+        base_internal::LowLevelAlloc::AllocWithArena(request, arena));
>+    std::copy(ptr_, ptr_ + size_, copy);
>+    Discard();
>+    ptr_ = copy;
>+  }
>+
>+  Vec(const Vec&) = delete;
>+  Vec& operator=(const Vec&) = delete;
>+};
>+
>+// A hash set of non-negative int32_t that uses Vec for its underlying storage.
>+class NodeSet {
>+ public:
>+  NodeSet() { Init(); }
>+
>+  void clear() { Init(); }
>+  bool contains(int32_t v) const { return table_[FindIndex(v)] == v; }
>+
>+  bool insert(int32_t v) {
>+    uint32_t i = FindIndex(v);
>+    if (table_[i] == v) {
>+      return false;
>+    }
>+    if (table_[i] == kEmpty) {
>+      // Only inserting over an empty cell increases the number of occupied
>+      // slots.
>+      occupied_++;
>+    }
>+    table_[i] = v;
>+    // Double when 75% full.
>+    if (occupied_ >= table_.size() - table_.size()/4) Grow();
>+    return true;
>+  }
>+
>+  void erase(uint32_t v) {
>+    uint32_t i = FindIndex(v);
>+    if (static_cast<uint32_t>(table_[i]) == v) {
>+      table_[i] = kDel;
>+    }
>+  }
>+
>+  // Iteration: is done via HASH_FOR_EACH
>+  // Example:
>+  //    HASH_FOR_EACH(elem, node->out) { ... }
>+#define HASH_FOR_EACH(elem, eset) \
>+  for (int32_t elem, _cursor = 0; (eset).Next(&_cursor, &elem); )
>+  bool Next(int32_t* cursor, int32_t* elem) {
>+    while (static_cast<uint32_t>(*cursor) < table_.size()) {
>+      int32_t v = table_[*cursor];
>+      (*cursor)++;
>+      if (v >= 0) {
>+        *elem = v;
>+        return true;
>+      }
>+    }
>+    return false;
>+  }
>+
>+ private:
>+  static const int32_t kEmpty;
>+  static const int32_t kDel;
>+  Vec<int32_t> table_;
>+  uint32_t occupied_;     // Count of non-empty slots (includes deleted slots)
>+
>+  static uint32_t Hash(uint32_t a) { return a * 41; }
>+
>+  // Return index for storing v.  May return an empty index or deleted index
>+  int FindIndex(int32_t v) const {
>+    // Search starting at hash index.
>+    const uint32_t mask = table_.size() - 1;
>+    uint32_t i = Hash(v) & mask;
>+    int deleted_index = -1;  // If >= 0, index of first deleted element we see
>+    while (true) {
>+      int32_t e = table_[i];
>+      if (v == e) {
>+        return i;
>+      } else if (e == kEmpty) {
>+        // Return any previously encountered deleted slot.
>+        return (deleted_index >= 0) ? deleted_index : i;
>+      } else if (e == kDel && deleted_index < 0) {
>+        // Keep searching since v might be present later.
>+        deleted_index = i;
>+      }
>+      i = (i + 1) & mask;  // Linear probing; quadratic is slightly slower.
>+    }
>+  }
>+
>+  void Init() {
>+    table_.clear();
>+    table_.resize(kInline);
>+    table_.fill(kEmpty);
>+    occupied_ = 0;
>+  }
>+
>+  void Grow() {
>+    Vec<int32_t> copy;
>+    copy.MoveFrom(&table_);
>+    occupied_ = 0;
>+    table_.resize(copy.size() * 2);
>+    table_.fill(kEmpty);
>+
>+    for (const auto& e : copy) {
>+      if (e >= 0) insert(e);
>+    }
>+  }
>+
>+  NodeSet(const NodeSet&) = delete;
>+  NodeSet& operator=(const NodeSet&) = delete;
>+};
>+
>+const int32_t NodeSet::kEmpty = -1;
>+const int32_t NodeSet::kDel = -2;
>+
>+// We encode a node index and a node version in GraphId.  The version
>+// number is incremented when the GraphId is freed which automatically
>+// invalidates all copies of the GraphId.
>+
>+inline GraphId MakeId(int32_t index, uint32_t version) {
>+  GraphId g;
>+  g.handle =
>+      (static_cast<uint64_t>(version) << 32) | static_cast<uint32_t>(index);
>+  return g;
>+}
>+
>+inline int32_t NodeIndex(GraphId id) {
>+  return static_cast<uint32_t>(id.handle & 0xfffffffful);
>+}
>+
>+inline uint32_t NodeVersion(GraphId id) {
>+  return static_cast<uint32_t>(id.handle >> 32);
>+}
>+
>+struct Node {
>+  int32_t rank;               // rank number assigned by Pearce-Kelly algorithm
>+  uint32_t version;           // Current version number
>+  int32_t next_hash;          // Next entry in hash table
>+  bool visited;               // Temporary marker used by depth-first-search
>+  uintptr_t masked_ptr;       // User-supplied pointer
>+  NodeSet in;                 // List of immediate predecessor nodes in graph
>+  NodeSet out;                // List of immediate successor nodes in graph
>+  int priority;               // Priority of recorded stack trace.
>+  int nstack;                 // Depth of recorded stack trace.
>+  void* stack[40];            // stack[0,nstack-1] holds stack trace for node.
>+};
>+
>+// Hash table for pointer to node index lookups.
>+class PointerMap {
>+ public:
>+  explicit PointerMap(const Vec<Node*>* nodes) : nodes_(nodes) {
>+    table_.fill(-1);
>+  }
>+
>+  int32_t Find(void* ptr) {
>+    auto masked = base_internal::HidePtr(ptr);
>+    for (int32_t i = table_[Hash(ptr)]; i != -1;) {
>+      Node* n = (*nodes_)[i];
>+      if (n->masked_ptr == masked) return i;
>+      i = n->next_hash;
>+    }
>+    return -1;
>+  }
>+
>+  void Add(void* ptr, int32_t i) {
>+    int32_t* head = &table_[Hash(ptr)];
>+    (*nodes_)[i]->next_hash = *head;
>+    *head = i;
>+  }
>+
>+  int32_t Remove(void* ptr) {
>+    // Advance through linked list while keeping track of the
>+    // predecessor slot that points to the current entry.
>+    auto masked = base_internal::HidePtr(ptr);
>+    for (int32_t* slot = &table_[Hash(ptr)]; *slot != -1; ) {
>+      int32_t index = *slot;
>+      Node* n = (*nodes_)[index];
>+      if (n->masked_ptr == masked) {
>+        *slot = n->next_hash;  // Remove n from linked list
>+        n->next_hash = -1;
>+        return index;
>+      }
>+      slot = &n->next_hash;
>+    }
>+    return -1;
>+  }
>+
>+ private:
>+  // Number of buckets in hash table for pointer lookups.
>+  static constexpr uint32_t kHashTableSize = 8171;  // should be prime
>+
>+  const Vec<Node*>* nodes_;
>+  std::array<int32_t, kHashTableSize> table_;
>+
>+  static uint32_t Hash(void* ptr) {
>+    return reinterpret_cast<uintptr_t>(ptr) % kHashTableSize;
>+  }
>+};
>+
>+}  // namespace
>+
>+struct GraphCycles::Rep {
>+  Vec<Node*> nodes_;
>+  Vec<int32_t> free_nodes_;  // Indices for unused entries in nodes_
>+  PointerMap ptrmap_;
>+
>+  // Temporary state.
>+  Vec<int32_t> deltaf_;  // Results of forward DFS
>+  Vec<int32_t> deltab_;  // Results of backward DFS
>+  Vec<int32_t> list_;    // All nodes to reprocess
>+  Vec<int32_t> merged_;  // Rank values to assign to list_ entries
>+  Vec<int32_t> stack_;   // Emulates recursion stack for depth-first searches
>+
>+  Rep() : ptrmap_(&nodes_) {}
>+};
>+
>+static Node* FindNode(GraphCycles::Rep* rep, GraphId id) {
>+  Node* n = rep->nodes_[NodeIndex(id)];
>+  return (n->version == NodeVersion(id)) ? n : nullptr;
>+}
>+
>+GraphCycles::GraphCycles() {
>+  InitArenaIfNecessary();
>+  rep_ = new (base_internal::LowLevelAlloc::AllocWithArena(sizeof(Rep), arena))
>+      Rep;
>+}
>+
>+GraphCycles::~GraphCycles() {
>+  for (auto* node : rep_->nodes_) {
>+    node->Node::~Node();
>+    base_internal::LowLevelAlloc::Free(node);
>+  }
>+  rep_->Rep::~Rep();
>+  base_internal::LowLevelAlloc::Free(rep_);
>+}
>+
>+bool GraphCycles::CheckInvariants() const {
>+  Rep* r = rep_;
>+  NodeSet ranks;  // Set of ranks seen so far.
>+  for (uint32_t x = 0; x < r->nodes_.size(); x++) {
>+    Node* nx = r->nodes_[x];
>+    void* ptr = base_internal::UnhidePtr<void>(nx->masked_ptr);
>+    if (ptr != nullptr && static_cast<uint32_t>(r->ptrmap_.Find(ptr)) != x) {
>+      ABSL_RAW_LOG(FATAL, "Did not find live node in hash table %u %p", x, ptr);
>+    }
>+    if (nx->visited) {
>+      ABSL_RAW_LOG(FATAL, "Did not clear visited marker on node %u", x);
>+    }
>+    if (!ranks.insert(nx->rank)) {
>+      ABSL_RAW_LOG(FATAL, "Duplicate occurrence of rank %d", nx->rank);
>+    }
>+    HASH_FOR_EACH(y, nx->out) {
>+      Node* ny = r->nodes_[y];
>+      if (nx->rank >= ny->rank) {
>+        ABSL_RAW_LOG(FATAL, "Edge %u->%d has bad rank assignment %d->%d", x, y,
>+                     nx->rank, ny->rank);
>+      }
>+    }
>+  }
>+  return true;
>+}
>+
>+GraphId GraphCycles::GetId(void* ptr) {
>+  int32_t i = rep_->ptrmap_.Find(ptr);
>+  if (i != -1) {
>+    return MakeId(i, rep_->nodes_[i]->version);
>+  } else if (rep_->free_nodes_.empty()) {
>+    Node* n =
>+        new (base_internal::LowLevelAlloc::AllocWithArena(sizeof(Node), arena))
>+            Node;
>+    n->version = 1;  // Avoid 0 since it is used by InvalidGraphId()
>+    n->visited = false;
>+    n->rank = rep_->nodes_.size();
>+    n->masked_ptr = base_internal::HidePtr(ptr);
>+    n->nstack = 0;
>+    n->priority = 0;
>+    rep_->nodes_.push_back(n);
>+    rep_->ptrmap_.Add(ptr, n->rank);
>+    return MakeId(n->rank, n->version);
>+  } else {
>+    // Preserve preceding rank since the set of ranks in use must be
>+    // a permutation of [0,rep_->nodes_.size()-1].
>+    int32_t r = rep_->free_nodes_.back();
>+    rep_->free_nodes_.pop_back();
>+    Node* n = rep_->nodes_[r];
>+    n->masked_ptr = base_internal::HidePtr(ptr);
>+    n->nstack = 0;
>+    n->priority = 0;
>+    rep_->ptrmap_.Add(ptr, r);
>+    return MakeId(r, n->version);
>+  }
>+}
>+
>+void GraphCycles::RemoveNode(void* ptr) {
>+  int32_t i = rep_->ptrmap_.Remove(ptr);
>+  if (i == -1) {
>+    return;
>+  }
>+  Node* x = rep_->nodes_[i];
>+  HASH_FOR_EACH(y, x->out) {
>+    rep_->nodes_[y]->in.erase(i);
>+  }
>+  HASH_FOR_EACH(y, x->in) {
>+    rep_->nodes_[y]->out.erase(i);
>+  }
>+  x->in.clear();
>+  x->out.clear();
>+  x->masked_ptr = base_internal::HidePtr<void>(nullptr);
>+  if (x->version == std::numeric_limits<uint32_t>::max()) {
>+    // Cannot use x any more
>+  } else {
>+    x->version++;  // Invalidates all copies of node.
>+    rep_->free_nodes_.push_back(i);
>+  }
>+}
>+
>+void* GraphCycles::Ptr(GraphId id) {
>+  Node* n = FindNode(rep_, id);
>+  return n == nullptr ? nullptr
>+                      : base_internal::UnhidePtr<void>(n->masked_ptr);
>+}
>+
>+bool GraphCycles::HasNode(GraphId node) {
>+  return FindNode(rep_, node) != nullptr;
>+}
>+
>+bool GraphCycles::HasEdge(GraphId x, GraphId y) const {
>+  Node* xn = FindNode(rep_, x);
>+  return xn && FindNode(rep_, y) && xn->out.contains(NodeIndex(y));
>+}
>+
>+void GraphCycles::RemoveEdge(GraphId x, GraphId y) {
>+  Node* xn = FindNode(rep_, x);
>+  Node* yn = FindNode(rep_, y);
>+  if (xn && yn) {
>+    xn->out.erase(NodeIndex(y));
>+    yn->in.erase(NodeIndex(x));
>+    // No need to update the rank assignment since a previous valid
>+    // rank assignment remains valid after an edge deletion.
>+  }
>+}
>+
>+static bool ForwardDFS(GraphCycles::Rep* r, int32_t n, int32_t upper_bound);
>+static void BackwardDFS(GraphCycles::Rep* r, int32_t n, int32_t lower_bound);
>+static void Reorder(GraphCycles::Rep* r);
>+static void Sort(const Vec<Node*>&, Vec<int32_t>* delta);
>+static void MoveToList(
>+    GraphCycles::Rep* r, Vec<int32_t>* src, Vec<int32_t>* dst);
>+
>+bool GraphCycles::InsertEdge(GraphId idx, GraphId idy) {
>+  Rep* r = rep_;
>+  const int32_t x = NodeIndex(idx);
>+  const int32_t y = NodeIndex(idy);
>+  Node* nx = FindNode(r, idx);
>+  Node* ny = FindNode(r, idy);
>+  if (nx == nullptr || ny == nullptr) return true;  // Expired ids
>+
>+  if (nx == ny) return false;  // Self edge
>+  if (!nx->out.insert(y)) {
>+    // Edge already exists.
>+    return true;
>+  }
>+
>+  ny->in.insert(x);
>+
>+  if (nx->rank <= ny->rank) {
>+    // New edge is consistent with existing rank assignment.
>+    return true;
>+  }
>+
>+  // Current rank assignments are incompatible with the new edge.  Recompute.
>+  // We only need to consider nodes that fall in the range [ny->rank,nx->rank].
>+  if (!ForwardDFS(r, y, nx->rank)) {
>+    // Found a cycle.  Undo the insertion and tell caller.
>+    nx->out.erase(y);
>+    ny->in.erase(x);
>+    // Since we do not call Reorder() on this path, clear any visited
>+    // markers left by ForwardDFS.
>+    for (const auto& d : r->deltaf_) {
>+      r->nodes_[d]->visited = false;
>+    }
>+    return false;
>+  }
>+  BackwardDFS(r, x, ny->rank);
>+  Reorder(r);
>+  return true;
>+}
>+
>+static bool ForwardDFS(GraphCycles::Rep* r, int32_t n, int32_t upper_bound) {
>+  // Avoid recursion since stack space might be limited.
>+  // We instead keep a stack of nodes to visit.
>+  r->deltaf_.clear();
>+  r->stack_.clear();
>+  r->stack_.push_back(n);
>+  while (!r->stack_.empty()) {
>+    n = r->stack_.back();
>+    r->stack_.pop_back();
>+    Node* nn = r->nodes_[n];
>+    if (nn->visited) continue;
>+
>+    nn->visited = true;
>+    r->deltaf_.push_back(n);
>+
>+    HASH_FOR_EACH(w, nn->out) {
>+      Node* nw = r->nodes_[w];
>+      if (nw->rank == upper_bound) {
>+        return false;  // Cycle
>+      }
>+      if (!nw->visited && nw->rank < upper_bound) {
>+        r->stack_.push_back(w);
>+      }
>+    }
>+  }
>+  return true;
>+}
>+
>+static void BackwardDFS(GraphCycles::Rep* r, int32_t n, int32_t lower_bound) {
>+  r->deltab_.clear();
>+  r->stack_.clear();
>+  r->stack_.push_back(n);
>+  while (!r->stack_.empty()) {
>+    n = r->stack_.back();
>+    r->stack_.pop_back();
>+    Node* nn = r->nodes_[n];
>+    if (nn->visited) continue;
>+
>+    nn->visited = true;
>+    r->deltab_.push_back(n);
>+
>+    HASH_FOR_EACH(w, nn->in) {
>+      Node* nw = r->nodes_[w];
>+      if (!nw->visited && lower_bound < nw->rank) {
>+        r->stack_.push_back(w);
>+      }
>+    }
>+  }
>+}
>+
>+static void Reorder(GraphCycles::Rep* r) {
>+  Sort(r->nodes_, &r->deltab_);
>+  Sort(r->nodes_, &r->deltaf_);
>+
>+  // Adds contents of delta lists to list_ (backwards deltas first).
>+  r->list_.clear();
>+  MoveToList(r, &r->deltab_, &r->list_);
>+  MoveToList(r, &r->deltaf_, &r->list_);
>+
>+  // Produce sorted list of all ranks that will be reassigned.
>+  r->merged_.resize(r->deltab_.size() + r->deltaf_.size());
>+  std::merge(r->deltab_.begin(), r->deltab_.end(),
>+             r->deltaf_.begin(), r->deltaf_.end(),
>+             r->merged_.begin());
>+
>+  // Assign the ranks in order to the collected list.
>+  for (uint32_t i = 0; i < r->list_.size(); i++) {
>+    r->nodes_[r->list_[i]]->rank = r->merged_[i];
>+  }
>+}
>+
>+static void Sort(const Vec<Node*>& nodes, Vec<int32_t>* delta) {
>+  struct ByRank {
>+    const Vec<Node*>* nodes;
>+    bool operator()(int32_t a, int32_t b) const {
>+      return (*nodes)[a]->rank < (*nodes)[b]->rank;
>+    }
>+  };
>+  ByRank cmp;
>+  cmp.nodes = &nodes;
>+  std::sort(delta->begin(), delta->end(), cmp);
>+}
>+
>+static void MoveToList(
>+    GraphCycles::Rep* r, Vec<int32_t>* src, Vec<int32_t>* dst) {
>+  for (auto& v : *src) {
>+    int32_t w = v;
>+    v = r->nodes_[w]->rank;         // Replace v entry with its rank
>+    r->nodes_[w]->visited = false;  // Prepare for future DFS calls
>+    dst->push_back(w);
>+  }
>+}
>+
>+int GraphCycles::FindPath(GraphId idx, GraphId idy, int max_path_len,
>+                          GraphId path[]) const {
>+  Rep* r = rep_;
>+  if (FindNode(r, idx) == nullptr || FindNode(r, idy) == nullptr) return 0;
>+  const int32_t x = NodeIndex(idx);
>+  const int32_t y = NodeIndex(idy);
>+
>+  // Forward depth first search starting at x until we hit y.
>+  // As we descend into a node, we push it onto the path.
>+  // As we leave a node, we remove it from the path.
>+  int path_len = 0;
>+
>+  NodeSet seen;
>+  r->stack_.clear();
>+  r->stack_.push_back(x);
>+  while (!r->stack_.empty()) {
>+    int32_t n = r->stack_.back();
>+    r->stack_.pop_back();
>+    if (n < 0) {
>+      // Marker to indicate that we are leaving a node
>+      path_len--;
>+      continue;
>+    }
>+
>+    if (path_len < max_path_len) {
>+      path[path_len] = MakeId(n, rep_->nodes_[n]->version);
>+    }
>+    path_len++;
>+    r->stack_.push_back(-1);  // Will remove tentative path entry
>+
>+    if (n == y) {
>+      return path_len;
>+    }
>+
>+    HASH_FOR_EACH(w, r->nodes_[n]->out) {
>+      if (seen.insert(w)) {
>+        r->stack_.push_back(w);
>+      }
>+    }
>+  }
>+
>+  return 0;
>+}
>+
>+bool GraphCycles::IsReachable(GraphId x, GraphId y) const {
>+  return FindPath(x, y, 0, nullptr) > 0;
>+}
>+
>+void GraphCycles::UpdateStackTrace(GraphId id, int priority,
>+                                   int (*get_stack_trace)(void** stack, int)) {
>+  Node* n = FindNode(rep_, id);
>+  if (n == nullptr || n->priority >= priority) {
>+    return;
>+  }
>+  n->nstack = (*get_stack_trace)(n->stack, ABSL_ARRAYSIZE(n->stack));
>+  n->priority = priority;
>+}
>+
>+int GraphCycles::GetStackTrace(GraphId id, void*** ptr) {
>+  Node* n = FindNode(rep_, id);
>+  if (n == nullptr) {
>+    *ptr = nullptr;
>+    return 0;
>+  } else {
>+    *ptr = n->stack;
>+    return n->nstack;
>+  }
>+}
>+
>+}  // namespace synchronization_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_LOW_LEVEL_ALLOC_MISSING
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/internal/graphcycles.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/internal/graphcycles.h
>new file mode 100644
>index 00000000000..2e6686a4c06
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/internal/graphcycles.h
>@@ -0,0 +1,137 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+
>+#ifndef ABSL_SYNCHRONIZATION_INTERNAL_GRAPHCYCLES_H_
>+#define ABSL_SYNCHRONIZATION_INTERNAL_GRAPHCYCLES_H_
>+
>+// GraphCycles detects the introduction of a cycle into a directed
>+// graph that is being built up incrementally.
>+//
>+// Nodes are identified by small integers.  It is not possible to
>+// record multiple edges with the same (source, destination) pair;
>+// requests to add an edge where one already exists are silently
>+// ignored.
>+//
>+// It is also not possible to introduce a cycle; an attempt to insert
>+// an edge that would introduce a cycle fails and returns false.
>+//
>+// GraphCycles uses no internal locking; calls into it should be
>+// serialized externally.
>+
>+// Performance considerations:
>+//   Works well on sparse graphs, poorly on dense graphs.
>+//   Extra information is maintained incrementally to detect cycles quickly.
>+//   InsertEdge() is very fast when the edge already exists, and reasonably fast
>+//   otherwise.
>+//   FindPath() is linear in the size of the graph.
>+// The current implemenation uses O(|V|+|E|) space.
>+
>+#include <cstdint>
>+
>+namespace absl {
>+namespace synchronization_internal {
>+
>+// Opaque identifier for a graph node.
>+struct GraphId {
>+  uint64_t handle;
>+
>+  bool operator==(const GraphId& x) const { return handle == x.handle; }
>+  bool operator!=(const GraphId& x) const { return handle != x.handle; }
>+};
>+
>+// Return an invalid graph id that will never be assigned by GraphCycles.
>+inline GraphId InvalidGraphId() {
>+  return GraphId{0};
>+}
>+
>+class GraphCycles {
>+ public:
>+  GraphCycles();
>+  ~GraphCycles();
>+
>+  // Return the id to use for ptr, assigning one if necessary.
>+  // Subsequent calls with the same ptr value will return the same id
>+  // until Remove().
>+  GraphId GetId(void* ptr);
>+
>+  // Remove "ptr" from the graph.  Its corresponding node and all
>+  // edges to and from it are removed.
>+  void RemoveNode(void* ptr);
>+
>+  // Return the pointer associated with id, or nullptr if id is not
>+  // currently in the graph.
>+  void* Ptr(GraphId id);
>+
>+  // Attempt to insert an edge from source_node to dest_node.  If the
>+  // edge would introduce a cycle, return false without making any
>+  // changes. Otherwise add the edge and return true.
>+  bool InsertEdge(GraphId source_node, GraphId dest_node);
>+
>+  // Remove any edge that exists from source_node to dest_node.
>+  void RemoveEdge(GraphId source_node, GraphId dest_node);
>+
>+  // Return whether node exists in the graph.
>+  bool HasNode(GraphId node);
>+
>+  // Return whether there is an edge directly from source_node to dest_node.
>+  bool HasEdge(GraphId source_node, GraphId dest_node) const;
>+
>+  // Return whether dest_node is reachable from source_node
>+  // by following edges.
>+  bool IsReachable(GraphId source_node, GraphId dest_node) const;
>+
>+  // Find a path from "source" to "dest".  If such a path exists,
>+  // place the nodes on the path in the array path[], and return
>+  // the number of nodes on the path.  If the path is longer than
>+  // max_path_len nodes, only the first max_path_len nodes are placed
>+  // in path[].  The client should compare the return value with
>+  // max_path_len" to see when this occurs.  If no path exists, return
>+  // 0.  Any valid path stored in path[] will start with "source" and
>+  // end with "dest".  There is no guarantee that the path is the
>+  // shortest, but no node will appear twice in the path, except the
>+  // source and destination node if they are identical; therefore, the
>+  // return value is at most one greater than the number of nodes in
>+  // the graph.
>+  int FindPath(GraphId source, GraphId dest, int max_path_len,
>+               GraphId path[]) const;
>+
>+  // Update the stack trace recorded for id with the current stack
>+  // trace if the last time it was updated had a smaller priority
>+  // than the priority passed on this call.
>+  //
>+  // *get_stack_trace is called to get the stack trace.
>+  void UpdateStackTrace(GraphId id, int priority,
>+                        int (*get_stack_trace)(void**, int));
>+
>+  // Set *ptr to the beginning of the array that holds the recorded
>+  // stack trace for id and return the depth of the stack trace.
>+  int GetStackTrace(GraphId id, void*** ptr);
>+
>+  // Check internal invariants. Crashes on failure, returns true on success.
>+  // Expensive: should only be called from graphcycles_test.cc.
>+  bool CheckInvariants() const;
>+
>+  // ----------------------------------------------------
>+  struct Rep;
>+ private:
>+  Rep *rep_;      // opaque representation
>+  GraphCycles(const GraphCycles&) = delete;
>+  GraphCycles& operator=(const GraphCycles&) = delete;
>+};
>+
>+}  // namespace synchronization_internal
>+}  // namespace absl
>+
>+#endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/internal/graphcycles_benchmark.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/internal/graphcycles_benchmark.cc
>new file mode 100644
>index 00000000000..a239c25c3a8
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/internal/graphcycles_benchmark.cc
>@@ -0,0 +1,44 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/synchronization/internal/graphcycles.h"
>+
>+#include <algorithm>
>+#include <cstdint>
>+#include <vector>
>+
>+#include "benchmark/benchmark.h"
>+#include "absl/base/internal/raw_logging.h"
>+
>+namespace {
>+
>+void BM_StressTest(benchmark::State& state) {
>+  const int num_nodes = state.range(0);
>+  while (state.KeepRunningBatch(num_nodes)) {
>+    absl::synchronization_internal::GraphCycles g;
>+    std::vector<absl::synchronization_internal::GraphId> nodes(num_nodes);
>+    for (int i = 0; i < num_nodes; i++) {
>+      nodes[i] = g.GetId(reinterpret_cast<void*>(static_cast<uintptr_t>(i)));
>+    }
>+    for (int i = 0; i < num_nodes; i++) {
>+      int end = std::min(num_nodes, i + 5);
>+      for (int j = i + 1; j < end; j++) {
>+        ABSL_RAW_CHECK(g.InsertEdge(nodes[i], nodes[j]), "");
>+      }
>+    }
>+  }
>+}
>+BENCHMARK(BM_StressTest)->Range(2048, 1048576);
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/internal/graphcycles_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/internal/graphcycles_test.cc
>new file mode 100644
>index 00000000000..9a85b39074d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/internal/graphcycles_test.cc
>@@ -0,0 +1,462 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/synchronization/internal/graphcycles.h"
>+
>+#include <map>
>+#include <random>
>+#include <unordered_set>
>+#include <utility>
>+#include <vector>
>+
>+#include "gtest/gtest.h"
>+#include "absl/base/internal/raw_logging.h"
>+#include "absl/base/macros.h"
>+
>+namespace absl {
>+namespace synchronization_internal {
>+
>+// We emulate a GraphCycles object with a node vector and an edge vector.
>+// We then compare the two implementations.
>+
>+using Nodes = std::vector<int>;
>+struct Edge {
>+  int from;
>+  int to;
>+};
>+using Edges = std::vector<Edge>;
>+using RandomEngine = std::mt19937_64;
>+
>+// Mapping from integer index to GraphId.
>+typedef std::map<int, GraphId> IdMap;
>+static GraphId Get(const IdMap& id, int num) {
>+  auto iter = id.find(num);
>+  return (iter == id.end()) ? InvalidGraphId() : iter->second;
>+}
>+
>+// Return whether "to" is reachable from "from".
>+static bool IsReachable(Edges *edges, int from, int to,
>+                        std::unordered_set<int> *seen) {
>+  seen->insert(from);     // we are investigating "from"; don't do it again
>+  if (from == to) return true;
>+  for (const auto &edge : *edges) {
>+    if (edge.from == from) {
>+      if (edge.to == to) {  // success via edge directly
>+        return true;
>+      } else if (seen->find(edge.to) == seen->end() &&  // success via edge
>+                 IsReachable(edges, edge.to, to, seen)) {
>+        return true;
>+      }
>+    }
>+  }
>+  return false;
>+}
>+
>+static void PrintEdges(Edges *edges) {
>+  ABSL_RAW_LOG(INFO, "EDGES (%zu)", edges->size());
>+  for (const auto &edge : *edges) {
>+    int a = edge.from;
>+    int b = edge.to;
>+    ABSL_RAW_LOG(INFO, "%d %d", a, b);
>+  }
>+  ABSL_RAW_LOG(INFO, "---");
>+}
>+
>+static void PrintGCEdges(Nodes *nodes, const IdMap &id, GraphCycles *gc) {
>+  ABSL_RAW_LOG(INFO, "GC EDGES");
>+  for (int a : *nodes) {
>+    for (int b : *nodes) {
>+      if (gc->HasEdge(Get(id, a), Get(id, b))) {
>+        ABSL_RAW_LOG(INFO, "%d %d", a, b);
>+      }
>+    }
>+  }
>+  ABSL_RAW_LOG(INFO, "---");
>+}
>+
>+static void PrintTransitiveClosure(Nodes *nodes, Edges *edges) {
>+  ABSL_RAW_LOG(INFO, "Transitive closure");
>+  for (int a : *nodes) {
>+    for (int b : *nodes) {
>+      std::unordered_set<int> seen;
>+      if (IsReachable(edges, a, b, &seen)) {
>+        ABSL_RAW_LOG(INFO, "%d %d", a, b);
>+      }
>+    }
>+  }
>+  ABSL_RAW_LOG(INFO, "---");
>+}
>+
>+static void PrintGCTransitiveClosure(Nodes *nodes, const IdMap &id,
>+                                     GraphCycles *gc) {
>+  ABSL_RAW_LOG(INFO, "GC Transitive closure");
>+  for (int a : *nodes) {
>+    for (int b : *nodes) {
>+      if (gc->IsReachable(Get(id, a), Get(id, b))) {
>+        ABSL_RAW_LOG(INFO, "%d %d", a, b);
>+      }
>+    }
>+  }
>+  ABSL_RAW_LOG(INFO, "---");
>+}
>+
>+static void CheckTransitiveClosure(Nodes *nodes, Edges *edges, const IdMap &id,
>+                                   GraphCycles *gc) {
>+  std::unordered_set<int> seen;
>+  for (const auto &a : *nodes) {
>+    for (const auto &b : *nodes) {
>+      seen.clear();
>+      bool gc_reachable = gc->IsReachable(Get(id, a), Get(id, b));
>+      bool reachable = IsReachable(edges, a, b, &seen);
>+      if (gc_reachable != reachable) {
>+        PrintEdges(edges);
>+        PrintGCEdges(nodes, id, gc);
>+        PrintTransitiveClosure(nodes, edges);
>+        PrintGCTransitiveClosure(nodes, id, gc);
>+        ABSL_RAW_LOG(FATAL, "gc_reachable %s reachable %s a %d b %d",
>+                     gc_reachable ? "true" : "false",
>+                     reachable ? "true" : "false", a, b);
>+      }
>+    }
>+  }
>+}
>+
>+static void CheckEdges(Nodes *nodes, Edges *edges, const IdMap &id,
>+                       GraphCycles *gc) {
>+  int count = 0;
>+  for (const auto &edge : *edges) {
>+    int a = edge.from;
>+    int b = edge.to;
>+    if (!gc->HasEdge(Get(id, a), Get(id, b))) {
>+      PrintEdges(edges);
>+      PrintGCEdges(nodes, id, gc);
>+      ABSL_RAW_LOG(FATAL, "!gc->HasEdge(%d, %d)", a, b);
>+    }
>+  }
>+  for (const auto &a : *nodes) {
>+    for (const auto &b : *nodes) {
>+      if (gc->HasEdge(Get(id, a), Get(id, b))) {
>+        count++;
>+      }
>+    }
>+  }
>+  if (count != edges->size()) {
>+    PrintEdges(edges);
>+    PrintGCEdges(nodes, id, gc);
>+    ABSL_RAW_LOG(FATAL, "edges->size() %zu  count %d", edges->size(), count);
>+  }
>+}
>+
>+static void CheckInvariants(const GraphCycles &gc) {
>+  if (ABSL_PREDICT_FALSE(!gc.CheckInvariants()))
>+    ABSL_RAW_LOG(FATAL, "CheckInvariants");
>+}
>+
>+// Returns the index of a randomly chosen node in *nodes.
>+// Requires *nodes be non-empty.
>+static int RandomNode(RandomEngine* rng, Nodes *nodes) {
>+  std::uniform_int_distribution<int> uniform(0, nodes->size()-1);
>+  return uniform(*rng);
>+}
>+
>+// Returns the index of a randomly chosen edge in *edges.
>+// Requires *edges be non-empty.
>+static int RandomEdge(RandomEngine* rng, Edges *edges) {
>+  std::uniform_int_distribution<int> uniform(0, edges->size()-1);
>+  return uniform(*rng);
>+}
>+
>+// Returns the index of edge (from, to) in *edges or -1 if it is not in *edges.
>+static int EdgeIndex(Edges *edges, int from, int to) {
>+  int i = 0;
>+  while (i != edges->size() &&
>+         ((*edges)[i].from != from || (*edges)[i].to != to)) {
>+    i++;
>+  }
>+  return i == edges->size()? -1 : i;
>+}
>+
>+TEST(GraphCycles, RandomizedTest) {
>+  int next_node = 0;
>+  Nodes nodes;
>+  Edges edges;   // from, to
>+  IdMap id;
>+  GraphCycles graph_cycles;
>+  static const int kMaxNodes = 7;  // use <= 7 nodes to keep test short
>+  static const int kDataOffset = 17;  // an offset to the node-specific data
>+  int n = 100000;
>+  int op = 0;
>+  RandomEngine rng(testing::UnitTest::GetInstance()->random_seed());
>+  std::uniform_int_distribution<int> uniform(0, 5);
>+
>+  auto ptr = [](intptr_t i) {
>+    return reinterpret_cast<void*>(i + kDataOffset);
>+  };
>+
>+  for (int iter = 0; iter != n; iter++) {
>+    for (const auto &node : nodes) {
>+      ASSERT_EQ(graph_cycles.Ptr(Get(id, node)), ptr(node)) << " node " << node;
>+    }
>+    CheckEdges(&nodes, &edges, id, &graph_cycles);
>+    CheckTransitiveClosure(&nodes, &edges, id, &graph_cycles);
>+    op = uniform(rng);
>+    switch (op) {
>+    case 0:     // Add a node
>+      if (nodes.size() < kMaxNodes) {
>+        int new_node = next_node++;
>+        GraphId new_gnode = graph_cycles.GetId(ptr(new_node));
>+        ASSERT_NE(new_gnode, InvalidGraphId());
>+        id[new_node] = new_gnode;
>+        ASSERT_EQ(ptr(new_node), graph_cycles.Ptr(new_gnode));
>+        nodes.push_back(new_node);
>+      }
>+      break;
>+
>+    case 1:    // Remove a node
>+      if (nodes.size() > 0) {
>+        int node_index = RandomNode(&rng, &nodes);
>+        int node = nodes[node_index];
>+        nodes[node_index] = nodes.back();
>+        nodes.pop_back();
>+        graph_cycles.RemoveNode(ptr(node));
>+        ASSERT_EQ(graph_cycles.Ptr(Get(id, node)), nullptr);
>+        id.erase(node);
>+        int i = 0;
>+        while (i != edges.size()) {
>+          if (edges[i].from == node || edges[i].to == node) {
>+            edges[i] = edges.back();
>+            edges.pop_back();
>+          } else {
>+            i++;
>+          }
>+        }
>+      }
>+      break;
>+
>+    case 2:   // Add an edge
>+      if (nodes.size() > 0) {
>+        int from = RandomNode(&rng, &nodes);
>+        int to = RandomNode(&rng, &nodes);
>+        if (EdgeIndex(&edges, nodes[from], nodes[to]) == -1) {
>+          if (graph_cycles.InsertEdge(id[nodes[from]], id[nodes[to]])) {
>+            Edge new_edge;
>+            new_edge.from = nodes[from];
>+            new_edge.to = nodes[to];
>+            edges.push_back(new_edge);
>+          } else {
>+            std::unordered_set<int> seen;
>+            ASSERT_TRUE(IsReachable(&edges, nodes[to], nodes[from], &seen))
>+                << "Edge " << nodes[to] << "->" << nodes[from];
>+          }
>+        }
>+      }
>+      break;
>+
>+    case 3:    // Remove an edge
>+      if (edges.size() > 0) {
>+        int i = RandomEdge(&rng, &edges);
>+        int from = edges[i].from;
>+        int to = edges[i].to;
>+        ASSERT_EQ(i, EdgeIndex(&edges, from, to));
>+        edges[i] = edges.back();
>+        edges.pop_back();
>+        ASSERT_EQ(-1, EdgeIndex(&edges, from, to));
>+        graph_cycles.RemoveEdge(id[from], id[to]);
>+      }
>+      break;
>+
>+    case 4:   // Check a path
>+      if (nodes.size() > 0) {
>+        int from = RandomNode(&rng, &nodes);
>+        int to = RandomNode(&rng, &nodes);
>+        GraphId path[2*kMaxNodes];
>+        int path_len = graph_cycles.FindPath(id[nodes[from]], id[nodes[to]],
>+                                             ABSL_ARRAYSIZE(path), path);
>+        std::unordered_set<int> seen;
>+        bool reachable = IsReachable(&edges, nodes[from], nodes[to], &seen);
>+        bool gc_reachable =
>+            graph_cycles.IsReachable(Get(id, nodes[from]), Get(id, nodes[to]));
>+        ASSERT_EQ(path_len != 0, reachable);
>+        ASSERT_EQ(path_len != 0, gc_reachable);
>+        // In the following line, we add one because a node can appear
>+        // twice, if the path is from that node to itself, perhaps via
>+        // every other node.
>+        ASSERT_LE(path_len, kMaxNodes + 1);
>+        if (path_len != 0) {
>+          ASSERT_EQ(id[nodes[from]], path[0]);
>+          ASSERT_EQ(id[nodes[to]], path[path_len-1]);
>+          for (int i = 1; i < path_len; i++) {
>+            ASSERT_TRUE(graph_cycles.HasEdge(path[i-1], path[i]));
>+          }
>+        }
>+      }
>+      break;
>+
>+    case 5:  // Check invariants
>+      CheckInvariants(graph_cycles);
>+      break;
>+
>+    default:
>+      ABSL_RAW_LOG(FATAL, "op %d", op);
>+    }
>+
>+    // Very rarely, test graph expansion by adding then removing many nodes.
>+    std::bernoulli_distribution one_in_1024(1.0 / 1024);
>+    if (one_in_1024(rng)) {
>+      CheckEdges(&nodes, &edges, id, &graph_cycles);
>+      CheckTransitiveClosure(&nodes, &edges, id, &graph_cycles);
>+      for (int i = 0; i != 256; i++) {
>+        int new_node = next_node++;
>+        GraphId new_gnode = graph_cycles.GetId(ptr(new_node));
>+        ASSERT_NE(InvalidGraphId(), new_gnode);
>+        id[new_node] = new_gnode;
>+        ASSERT_EQ(ptr(new_node), graph_cycles.Ptr(new_gnode));
>+        for (const auto &node : nodes) {
>+          ASSERT_NE(node, new_node);
>+        }
>+        nodes.push_back(new_node);
>+      }
>+      for (int i = 0; i != 256; i++) {
>+        ASSERT_GT(nodes.size(), 0);
>+        int node_index = RandomNode(&rng, &nodes);
>+        int node = nodes[node_index];
>+        nodes[node_index] = nodes.back();
>+        nodes.pop_back();
>+        graph_cycles.RemoveNode(ptr(node));
>+        id.erase(node);
>+        int j = 0;
>+        while (j != edges.size()) {
>+          if (edges[j].from == node || edges[j].to == node) {
>+            edges[j] = edges.back();
>+            edges.pop_back();
>+          } else {
>+            j++;
>+          }
>+        }
>+      }
>+      CheckInvariants(graph_cycles);
>+    }
>+  }
>+}
>+
>+class GraphCyclesTest : public ::testing::Test {
>+ public:
>+  IdMap id_;
>+  GraphCycles g_;
>+
>+  static void* Ptr(int i) {
>+    return reinterpret_cast<void*>(static_cast<uintptr_t>(i));
>+  }
>+
>+  static int Num(void* ptr) {
>+    return static_cast<int>(reinterpret_cast<uintptr_t>(ptr));
>+  }
>+
>+  // Test relies on ith NewNode() call returning Node numbered i
>+  GraphCyclesTest() {
>+    for (int i = 0; i < 100; i++) {
>+      id_[i] = g_.GetId(Ptr(i));
>+    }
>+    CheckInvariants(g_);
>+  }
>+
>+  bool AddEdge(int x, int y) {
>+    return g_.InsertEdge(Get(id_, x), Get(id_, y));
>+  }
>+
>+  void AddMultiples() {
>+    // For every node x > 0: add edge to 2*x, 3*x
>+    for (int x = 1; x < 25; x++) {
>+      EXPECT_TRUE(AddEdge(x, 2*x)) << x;
>+      EXPECT_TRUE(AddEdge(x, 3*x)) << x;
>+    }
>+    CheckInvariants(g_);
>+  }
>+
>+  std::string Path(int x, int y) {
>+    GraphId path[5];
>+    int np = g_.FindPath(Get(id_, x), Get(id_, y), ABSL_ARRAYSIZE(path), path);
>+    std::string result;
>+    for (int i = 0; i < np; i++) {
>+      if (i >= ABSL_ARRAYSIZE(path)) {
>+        result += " ...";
>+        break;
>+      }
>+      if (!result.empty()) result.push_back(' ');
>+      char buf[20];
>+      snprintf(buf, sizeof(buf), "%d", Num(g_.Ptr(path[i])));
>+      result += buf;
>+    }
>+    return result;
>+  }
>+};
>+
>+TEST_F(GraphCyclesTest, NoCycle) {
>+  AddMultiples();
>+  CheckInvariants(g_);
>+}
>+
>+TEST_F(GraphCyclesTest, SimpleCycle) {
>+  AddMultiples();
>+  EXPECT_FALSE(AddEdge(8, 4));
>+  EXPECT_EQ("4 8", Path(4, 8));
>+  CheckInvariants(g_);
>+}
>+
>+TEST_F(GraphCyclesTest, IndirectCycle) {
>+  AddMultiples();
>+  EXPECT_TRUE(AddEdge(16, 9));
>+  CheckInvariants(g_);
>+  EXPECT_FALSE(AddEdge(9, 2));
>+  EXPECT_EQ("2 4 8 16 9", Path(2, 9));
>+  CheckInvariants(g_);
>+}
>+
>+TEST_F(GraphCyclesTest, LongPath) {
>+  ASSERT_TRUE(AddEdge(2, 4));
>+  ASSERT_TRUE(AddEdge(4, 6));
>+  ASSERT_TRUE(AddEdge(6, 8));
>+  ASSERT_TRUE(AddEdge(8, 10));
>+  ASSERT_TRUE(AddEdge(10, 12));
>+  ASSERT_FALSE(AddEdge(12, 2));
>+  EXPECT_EQ("2 4 6 8 10 ...", Path(2, 12));
>+  CheckInvariants(g_);
>+}
>+
>+TEST_F(GraphCyclesTest, RemoveNode) {
>+  ASSERT_TRUE(AddEdge(1, 2));
>+  ASSERT_TRUE(AddEdge(2, 3));
>+  ASSERT_TRUE(AddEdge(3, 4));
>+  ASSERT_TRUE(AddEdge(4, 5));
>+  g_.RemoveNode(g_.Ptr(id_[3]));
>+  id_.erase(3);
>+  ASSERT_TRUE(AddEdge(5, 1));
>+}
>+
>+TEST_F(GraphCyclesTest, ManyEdges) {
>+  const int N = 50;
>+  for (int i = 0; i < N; i++) {
>+    for (int j = 1; j < N; j++) {
>+      ASSERT_TRUE(AddEdge(i, i+j));
>+    }
>+  }
>+  CheckInvariants(g_);
>+  ASSERT_TRUE(AddEdge(2*N-1, 0));
>+  CheckInvariants(g_);
>+  ASSERT_FALSE(AddEdge(10, 9));
>+  CheckInvariants(g_);
>+}
>+
>+}  // namespace synchronization_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/internal/kernel_timeout.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/internal/kernel_timeout.h
>new file mode 100644
>index 00000000000..bb7080004c7
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/internal/kernel_timeout.h
>@@ -0,0 +1,151 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+
>+// An optional absolute timeout, with nanosecond granularity,
>+// compatible with absl::Time. Suitable for in-register
>+// parameter-passing (e.g. syscalls.)
>+// Constructible from a absl::Time (for a timeout to be respected) or {}
>+// (for "no timeout".)
>+// This is a private low-level API for use by a handful of low-level
>+// components that are friends of this class. Higher-level components
>+// should build APIs based on absl::Time and absl::Duration.
>+
>+#ifndef ABSL_SYNCHRONIZATION_INTERNAL_KERNEL_TIMEOUT_H_
>+#define ABSL_SYNCHRONIZATION_INTERNAL_KERNEL_TIMEOUT_H_
>+
>+#include <time.h>
>+#include <algorithm>
>+#include <limits>
>+
>+#include "absl/base/internal/raw_logging.h"
>+#include "absl/time/clock.h"
>+#include "absl/time/time.h"
>+
>+namespace absl {
>+namespace synchronization_internal {
>+
>+class Futex;
>+class Waiter;
>+
>+class KernelTimeout {
>+ public:
>+  // A timeout that should expire at <t>.  Any value, in the full
>+  // InfinitePast() to InfiniteFuture() range, is valid here and will be
>+  // respected.
>+  explicit KernelTimeout(absl::Time t) : ns_(MakeNs(t)) {}
>+  // No timeout.
>+  KernelTimeout() : ns_(0) {}
>+
>+  // A more explicit factory for those who prefer it.  Equivalent to {}.
>+  static KernelTimeout Never() { return {}; }
>+
>+  // We explicitly do not support other custom formats: timespec, int64_t nanos.
>+  // Unify on this and absl::Time, please.
>+  bool has_timeout() const { return ns_ != 0; }
>+
>+ private:
>+  // internal rep, not user visible: ns after unix epoch.
>+  // zero = no timeout.
>+  // Negative we treat as an unlikely (and certainly expired!) but valid
>+  // timeout.
>+  int64_t ns_;
>+
>+  static int64_t MakeNs(absl::Time t) {
>+    // optimization--InfiniteFuture is common "no timeout" value
>+    // and cheaper to compare than convert.
>+    if (t == absl::InfiniteFuture()) return 0;
>+    int64_t x = ToUnixNanos(t);
>+
>+    // A timeout that lands exactly on the epoch (x=0) needs to be respected,
>+    // so we alter it unnoticably to 1.  Negative timeouts are in
>+    // theory supported, but handled poorly by the kernel (long
>+    // delays) so push them forward too; since all such times have
>+    // already passed, it's indistinguishable.
>+    if (x <= 0) x = 1;
>+    // A time larger than what can be represented to the kernel is treated
>+    // as no timeout.
>+    if (x == std::numeric_limits<int64_t>::max()) x = 0;
>+    return x;
>+  }
>+
>+  // Convert to parameter for sem_timedwait/futex/similar.  Only for approved
>+  // users.  Do not call if !has_timeout.
>+  struct timespec MakeAbsTimespec() {
>+    int64_t n = ns_;
>+    static const int64_t kNanosPerSecond = 1000 * 1000 * 1000;
>+    if (n == 0) {
>+      ABSL_RAW_LOG(
>+          ERROR,
>+          "Tried to create a timespec from a non-timeout; never do this.");
>+      // But we'll try to continue sanely.  no-timeout ~= saturated timeout.
>+      n = std::numeric_limits<int64_t>::max();
>+    }
>+
>+    // Kernel APIs validate timespecs as being at or after the epoch,
>+    // despite the kernel time type being signed.  However, no one can
>+    // tell the difference between a timeout at or before the epoch (since
>+    // all such timeouts have expired!)
>+    if (n < 0) n = 0;
>+
>+    struct timespec abstime;
>+    int64_t seconds = std::min(n / kNanosPerSecond,
>+                             int64_t{std::numeric_limits<time_t>::max()});
>+    abstime.tv_sec = static_cast<time_t>(seconds);
>+    abstime.tv_nsec =
>+        static_cast<decltype(abstime.tv_nsec)>(n % kNanosPerSecond);
>+    return abstime;
>+  }
>+
>+#ifdef _WIN32
>+  // Converts to milliseconds from now, or INFINITE when
>+  // !has_timeout(). For use by SleepConditionVariableSRW on
>+  // Windows. Callers should recognize that the return value is a
>+  // relative duration (it should be recomputed by calling this method
>+  // in the case of a spurious wakeup).
>+  // This header file may be included transitively by public header files,
>+  // so we define our own DWORD and INFINITE instead of getting them from
>+  // <intsafe.h> and <WinBase.h>.
>+  typedef unsigned long DWord;  // NOLINT
>+  DWord InMillisecondsFromNow() const {
>+    constexpr DWord kInfinite = std::numeric_limits<DWord>::max();
>+    if (!has_timeout()) {
>+      return kInfinite;
>+    }
>+    // The use of absl::Now() to convert from absolute time to
>+    // relative time means that absl::Now() cannot use anything that
>+    // depends on KernelTimeout (for example, Mutex) on Windows.
>+    int64_t now = ToUnixNanos(absl::Now());
>+    if (ns_ >= now) {
>+      // Round up so that Now() + ms_from_now >= ns_.
>+      constexpr uint64_t max_nanos =
>+          std::numeric_limits<int64_t>::max() - 999999u;
>+      uint64_t ms_from_now =
>+          (std::min<uint64_t>(max_nanos, ns_ - now) + 999999u) / 1000000u;
>+      if (ms_from_now > kInfinite) {
>+        return kInfinite;
>+      }
>+      return static_cast<DWord>(ms_from_now);
>+    }
>+    return 0;
>+  }
>+#endif
>+
>+  friend class Futex;
>+  friend class Waiter;
>+};
>+
>+}  // namespace synchronization_internal
>+}  // namespace absl
>+#endif  // ABSL_SYNCHRONIZATION_INTERNAL_KERNEL_TIMEOUT_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/internal/mutex_nonprod.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/internal/mutex_nonprod.cc
>new file mode 100644
>index 00000000000..45c60326dba
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/internal/mutex_nonprod.cc
>@@ -0,0 +1,318 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+// Implementation of a small subset of Mutex and CondVar functionality
>+// for platforms where the production implementation hasn't been fully
>+// ported yet.
>+
>+#include "absl/synchronization/mutex.h"
>+
>+#if defined(_WIN32)
>+#include <chrono>  // NOLINT(build/c++11)
>+#else
>+#include <sys/time.h>
>+#include <time.h>
>+#endif
>+
>+#include <algorithm>
>+
>+#include "absl/base/internal/raw_logging.h"
>+#include "absl/time/time.h"
>+
>+namespace absl {
>+namespace synchronization_internal {
>+
>+namespace {
>+
>+// Return the current time plus the timeout.
>+absl::Time DeadlineFromTimeout(absl::Duration timeout) {
>+  return absl::Now() + timeout;
>+}
>+
>+// Limit the deadline to a positive, 32-bit time_t value to accommodate
>+// implementation restrictions.  This also deals with InfinitePast and
>+// InfiniteFuture.
>+absl::Time LimitedDeadline(absl::Time deadline) {
>+  deadline = std::max(absl::FromTimeT(0), deadline);
>+  deadline = std::min(deadline, absl::FromTimeT(0x7fffffff));
>+  return deadline;
>+}
>+
>+}  // namespace
>+
>+#if defined(_WIN32)
>+
>+MutexImpl::MutexImpl() {}
>+
>+MutexImpl::~MutexImpl() {
>+  if (locked_) {
>+    std_mutex_.unlock();
>+  }
>+}
>+
>+void MutexImpl::Lock() {
>+  std_mutex_.lock();
>+  locked_ = true;
>+}
>+
>+bool MutexImpl::TryLock() {
>+  bool locked = std_mutex_.try_lock();
>+  if (locked) locked_ = true;
>+  return locked;
>+}
>+
>+void MutexImpl::Unlock() {
>+  locked_ = false;
>+  released_.SignalAll();
>+  std_mutex_.unlock();
>+}
>+
>+CondVarImpl::CondVarImpl() {}
>+
>+CondVarImpl::~CondVarImpl() {}
>+
>+void CondVarImpl::Signal() { std_cv_.notify_one(); }
>+
>+void CondVarImpl::SignalAll() { std_cv_.notify_all(); }
>+
>+void CondVarImpl::Wait(MutexImpl* mu) {
>+  mu->released_.SignalAll();
>+  std_cv_.wait(mu->std_mutex_);
>+}
>+
>+bool CondVarImpl::WaitWithDeadline(MutexImpl* mu, absl::Time deadline) {
>+  mu->released_.SignalAll();
>+  time_t when = ToTimeT(deadline);
>+  int64_t nanos = ToInt64Nanoseconds(deadline - absl::FromTimeT(when));
>+  std::chrono::system_clock::time_point deadline_tp =
>+      std::chrono::system_clock::from_time_t(when) +
>+      std::chrono::duration_cast<std::chrono::system_clock::duration>(
>+          std::chrono::nanoseconds(nanos));
>+  auto deadline_since_epoch =
>+      std::chrono::duration_cast<std::chrono::duration<double>>(
>+          deadline_tp - std::chrono::system_clock::from_time_t(0));
>+  return std_cv_.wait_until(mu->std_mutex_, deadline_tp) ==
>+         std::cv_status::timeout;
>+}
>+
>+#else  // ! _WIN32
>+
>+MutexImpl::MutexImpl() {
>+  ABSL_RAW_CHECK(pthread_mutex_init(&pthread_mutex_, nullptr) == 0,
>+                 "pthread error");
>+}
>+
>+MutexImpl::~MutexImpl() {
>+  if (locked_) {
>+    ABSL_RAW_CHECK(pthread_mutex_unlock(&pthread_mutex_) == 0, "pthread error");
>+  }
>+  ABSL_RAW_CHECK(pthread_mutex_destroy(&pthread_mutex_) == 0, "pthread error");
>+}
>+
>+void MutexImpl::Lock() {
>+  ABSL_RAW_CHECK(pthread_mutex_lock(&pthread_mutex_) == 0, "pthread error");
>+  locked_ = true;
>+}
>+
>+bool MutexImpl::TryLock() {
>+  bool locked = (0 == pthread_mutex_trylock(&pthread_mutex_));
>+  if (locked) locked_ = true;
>+  return locked;
>+}
>+
>+void MutexImpl::Unlock() {
>+  locked_ = false;
>+  released_.SignalAll();
>+  ABSL_RAW_CHECK(pthread_mutex_unlock(&pthread_mutex_) == 0, "pthread error");
>+}
>+
>+CondVarImpl::CondVarImpl() {
>+  ABSL_RAW_CHECK(pthread_cond_init(&pthread_cv_, nullptr) == 0,
>+                 "pthread error");
>+}
>+
>+CondVarImpl::~CondVarImpl() {
>+  ABSL_RAW_CHECK(pthread_cond_destroy(&pthread_cv_) == 0, "pthread error");
>+}
>+
>+void CondVarImpl::Signal() {
>+  ABSL_RAW_CHECK(pthread_cond_signal(&pthread_cv_) == 0, "pthread error");
>+}
>+
>+void CondVarImpl::SignalAll() {
>+  ABSL_RAW_CHECK(pthread_cond_broadcast(&pthread_cv_) == 0, "pthread error");
>+}
>+
>+void CondVarImpl::Wait(MutexImpl* mu) {
>+  mu->released_.SignalAll();
>+  ABSL_RAW_CHECK(pthread_cond_wait(&pthread_cv_, &mu->pthread_mutex_) == 0,
>+                 "pthread error");
>+}
>+
>+bool CondVarImpl::WaitWithDeadline(MutexImpl* mu, absl::Time deadline) {
>+  mu->released_.SignalAll();
>+  struct timespec ts = ToTimespec(deadline);
>+  int rc = pthread_cond_timedwait(&pthread_cv_, &mu->pthread_mutex_, &ts);
>+  if (rc == ETIMEDOUT) return true;
>+  ABSL_RAW_CHECK(rc == 0, "pthread error");
>+  return false;
>+}
>+
>+#endif  // ! _WIN32
>+
>+void MutexImpl::Await(const Condition& cond) {
>+  if (cond.Eval()) return;
>+  released_.SignalAll();
>+  do {
>+    released_.Wait(this);
>+  } while (!cond.Eval());
>+}
>+
>+bool MutexImpl::AwaitWithDeadline(const Condition& cond, absl::Time deadline) {
>+  if (cond.Eval()) return true;
>+  released_.SignalAll();
>+  while (true) {
>+    if (released_.WaitWithDeadline(this, deadline)) return false;
>+    if (cond.Eval()) return true;
>+  }
>+}
>+
>+}  // namespace synchronization_internal
>+
>+Mutex::Mutex() {}
>+
>+Mutex::~Mutex() {}
>+
>+void Mutex::Lock() { impl()->Lock(); }
>+
>+void Mutex::Unlock() { impl()->Unlock(); }
>+
>+bool Mutex::TryLock() { return impl()->TryLock(); }
>+
>+void Mutex::ReaderLock() { Lock(); }
>+
>+void Mutex::ReaderUnlock() { Unlock(); }
>+
>+void Mutex::Await(const Condition& cond) { impl()->Await(cond); }
>+
>+void Mutex::LockWhen(const Condition& cond) {
>+  Lock();
>+  Await(cond);
>+}
>+
>+bool Mutex::AwaitWithDeadline(const Condition& cond, absl::Time deadline) {
>+  return impl()->AwaitWithDeadline(
>+      cond, synchronization_internal::LimitedDeadline(deadline));
>+}
>+
>+bool Mutex::AwaitWithTimeout(const Condition& cond, absl::Duration timeout) {
>+  return AwaitWithDeadline(
>+      cond, synchronization_internal::DeadlineFromTimeout(timeout));
>+}
>+
>+bool Mutex::LockWhenWithDeadline(const Condition& cond, absl::Time deadline) {
>+  Lock();
>+  return AwaitWithDeadline(cond, deadline);
>+}
>+
>+bool Mutex::LockWhenWithTimeout(const Condition& cond, absl::Duration timeout) {
>+  return LockWhenWithDeadline(
>+      cond, synchronization_internal::DeadlineFromTimeout(timeout));
>+}
>+
>+void Mutex::ReaderLockWhen(const Condition& cond) {
>+  ReaderLock();
>+  Await(cond);
>+}
>+
>+bool Mutex::ReaderLockWhenWithTimeout(const Condition& cond,
>+                                      absl::Duration timeout) {
>+  return LockWhenWithTimeout(cond, timeout);
>+}
>+bool Mutex::ReaderLockWhenWithDeadline(const Condition& cond,
>+                                       absl::Time deadline) {
>+  return LockWhenWithDeadline(cond, deadline);
>+}
>+
>+void Mutex::EnableDebugLog(const char*) {}
>+void Mutex::EnableInvariantDebugging(void (*)(void*), void*) {}
>+void Mutex::ForgetDeadlockInfo() {}
>+void Mutex::AssertHeld() const {}
>+void Mutex::AssertReaderHeld() const {}
>+void Mutex::AssertNotHeld() const {}
>+
>+CondVar::CondVar() {}
>+
>+CondVar::~CondVar() {}
>+
>+void CondVar::Signal() { impl()->Signal(); }
>+
>+void CondVar::SignalAll() { impl()->SignalAll(); }
>+
>+void CondVar::Wait(Mutex* mu) { return impl()->Wait(mu->impl()); }
>+
>+bool CondVar::WaitWithDeadline(Mutex* mu, absl::Time deadline) {
>+  return impl()->WaitWithDeadline(
>+      mu->impl(), synchronization_internal::LimitedDeadline(deadline));
>+}
>+
>+bool CondVar::WaitWithTimeout(Mutex* mu, absl::Duration timeout) {
>+  return WaitWithDeadline(mu, absl::Now() + timeout);
>+}
>+
>+void CondVar::EnableDebugLog(const char*) {}
>+
>+#ifdef THREAD_SANITIZER
>+extern "C" void __tsan_read1(void *addr);
>+#else
>+#define __tsan_read1(addr)  // do nothing if TSan not enabled
>+#endif
>+
>+// A function that just returns its argument, dereferenced
>+static bool Dereference(void *arg) {
>+  // ThreadSanitizer does not instrument this file for memory accesses.
>+  // This function dereferences a user variable that can participate
>+  // in a data race, so we need to manually tell TSan about this memory access.
>+  __tsan_read1(arg);
>+  return *(static_cast<bool *>(arg));
>+}
>+
>+Condition::Condition() {}   // null constructor, used for kTrue only
>+const Condition Condition::kTrue;
>+
>+Condition::Condition(bool (*func)(void *), void *arg)
>+    : eval_(&CallVoidPtrFunction),
>+      function_(func),
>+      method_(nullptr),
>+      arg_(arg) {}
>+
>+bool Condition::CallVoidPtrFunction(const Condition *c) {
>+  return (*c->function_)(c->arg_);
>+}
>+
>+Condition::Condition(const bool *cond)
>+    : eval_(CallVoidPtrFunction),
>+      function_(Dereference),
>+      method_(nullptr),
>+      // const_cast is safe since Dereference does not modify arg
>+      arg_(const_cast<bool *>(cond)) {}
>+
>+bool Condition::Eval() const {
>+  // eval_ == null for kTrue
>+  return (this->eval_ == nullptr) || (*this->eval_)(this);
>+}
>+
>+void RegisterSymbolizer(bool (*)(const void*, char*, int)) {}
>+
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/internal/mutex_nonprod.inc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/internal/mutex_nonprod.inc
>new file mode 100644
>index 00000000000..0aab3d1314e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/internal/mutex_nonprod.inc
>@@ -0,0 +1,256 @@
>+// Do not include.  This is an implementation detail of base/mutex.h.
>+//
>+// Declares three classes:
>+//
>+// base::internal::MutexImpl - implementation helper for Mutex
>+// base::internal::CondVarImpl - implementation helper for CondVar
>+// base::internal::SynchronizationStorage<T> - implementation helper for
>+//                                             Mutex, CondVar
>+
>+#include <type_traits>
>+
>+#if defined(_WIN32)
>+#include <condition_variable>
>+#include <mutex>
>+#else
>+#include <pthread.h>
>+#endif
>+
>+#include "absl/base/call_once.h"
>+#include "absl/time/time.h"
>+
>+// Declare that Mutex::ReaderLock is actually Lock().  Intended primarily
>+// for tests, and even then as a last resort.
>+#ifdef ABSL_MUTEX_READER_LOCK_IS_EXCLUSIVE
>+#error ABSL_MUTEX_READER_LOCK_IS_EXCLUSIVE cannot be directly set
>+#else
>+#define ABSL_MUTEX_READER_LOCK_IS_EXCLUSIVE 1
>+#endif
>+
>+// Declare that Mutex::EnableInvariantDebugging is not implemented.
>+// Intended primarily for tests, and even then as a last resort.
>+#ifdef ABSL_MUTEX_ENABLE_INVARIANT_DEBUGGING_NOT_IMPLEMENTED
>+#error ABSL_MUTEX_ENABLE_INVARIANT_DEBUGGING_NOT_IMPLEMENTED cannot be directly set
>+#else
>+#define ABSL_MUTEX_ENABLE_INVARIANT_DEBUGGING_NOT_IMPLEMENTED 1
>+#endif
>+
>+namespace absl {
>+class Condition;
>+
>+namespace synchronization_internal {
>+
>+class MutexImpl;
>+
>+// Do not use this implementation detail of CondVar. Provides most of the
>+// implementation, but should not be placed directly in static storage
>+// because it will not linker initialize properly. See
>+// SynchronizationStorage<T> below for what we mean by linker
>+// initialization.
>+class CondVarImpl {
>+ public:
>+  CondVarImpl();
>+  CondVarImpl(const CondVarImpl&) = delete;
>+  CondVarImpl& operator=(const CondVarImpl&) = delete;
>+  ~CondVarImpl();
>+
>+  void Signal();
>+  void SignalAll();
>+  void Wait(MutexImpl* mutex);
>+  bool WaitWithDeadline(MutexImpl* mutex, absl::Time deadline);
>+
>+ private:
>+#if defined(_WIN32)
>+  std::condition_variable_any std_cv_;
>+#else
>+  pthread_cond_t pthread_cv_;
>+#endif
>+};
>+
>+// Do not use this implementation detail of Mutex. Provides most of the
>+// implementation, but should not be placed directly in static storage
>+// because it will not linker initialize properly. See
>+// SynchronizationStorage<T> below for what we mean by linker
>+// initialization.
>+class MutexImpl {
>+ public:
>+  MutexImpl();
>+  MutexImpl(const MutexImpl&) = delete;
>+  MutexImpl& operator=(const MutexImpl&) = delete;
>+  ~MutexImpl();
>+
>+  void Lock();
>+  bool TryLock();
>+  void Unlock();
>+  void Await(const Condition& cond);
>+  bool AwaitWithDeadline(const Condition& cond, absl::Time deadline);
>+
>+ private:
>+  friend class CondVarImpl;
>+
>+#if defined(_WIN32)
>+  std::mutex std_mutex_;
>+#else
>+  pthread_mutex_t pthread_mutex_;
>+#endif
>+
>+  // True if the underlying mutex is locked.  If the destructor is entered
>+  // while locked_, the underlying mutex is unlocked.  Mutex supports
>+  // destruction while locked, but the same is undefined behavior for both
>+  // pthread_mutex_t and std::mutex.
>+  bool locked_ = false;
>+
>+  // Signaled before releasing the lock, in support of Await.
>+  CondVarImpl released_;
>+};
>+
>+// Do not use this implementation detail of CondVar and Mutex.  A storage
>+// space for T that supports a LinkerInitialized constructor. T must
>+// have a default constructor, which is called by the first call to
>+// get(). T's destructor is never called if the LinkerInitialized
>+// constructor is called.
>+//
>+// Objects constructed with the default constructor are constructed and
>+// destructed like any other object, and should never be allocated in
>+// static storage.
>+//
>+// Objects constructed with the LinkerInitialized constructor should
>+// always be in static storage. For such objects, calls to get() are always
>+// valid, except from signal handlers.
>+//
>+// Note that this implementation relies on undefined language behavior that
>+// are known to hold for the set of supported compilers. An analysis
>+// follows.
>+//
>+// From the C++11 standard:
>+//
>+// [basic.life] says an object has non-trivial initialization if it is of
>+// class type and it is initialized by a constructor other than a trivial
>+// default constructor.  (the LinkerInitialized constructor is
>+// non-trivial)
>+//
>+// [basic.life] says the lifetime of an object with a non-trivial
>+// constructor begins when the call to the constructor is complete.
>+//
>+// [basic.life] says the lifetime of an object with non-trivial destructor
>+// ends when the call to the destructor begins.
>+//
>+// [basic.life] p5 specifies undefined behavior when accessing non-static
>+// members of an instance outside its
>+// lifetime. (SynchronizationStorage::get() access non-static members)
>+//
>+// So, LinkerInitialized object of SynchronizationStorage uses a
>+// non-trivial constructor, which is called at some point during dynamic
>+// initialization, and is therefore subject to order of dynamic
>+// initialization bugs, where get() is called before the object's
>+// constructor is, resulting in undefined behavior.
>+//
>+// Similarly, a LinkerInitialized SynchronizationStorage object has a
>+// non-trivial destructor, and so its lifetime ends at some point during
>+// destruction of objects with static storage duration [basic.start.term]
>+// p4. There is a window where other exit code could call get() after this
>+// occurs, resulting in undefined behavior.
>+//
>+// Combined, these statements imply that LinkerInitialized instances
>+// of SynchronizationStorage<T> rely on undefined behavior.
>+//
>+// However, in practice, the implementation works on all supported
>+// compilers. Specifically, we rely on:
>+//
>+// a) zero-initialization being sufficient to initialize
>+// LinkerInitialized instances for the purposes of calling
>+// get(), regardless of when the constructor is called. This is
>+// because the is_dynamic_ boolean is correctly zero-initialized to
>+// false.
>+//
>+// b) the LinkerInitialized constructor is a NOP, and immaterial to
>+// even to concurrent calls to get().
>+//
>+// c) the destructor being a NOP for LinkerInitialized objects
>+// (guaranteed by a check for !is_dynamic_), and so any concurrent and
>+// subsequent calls to get() functioning as if the destructor were not
>+// called, by virtue of the instances' storage remaining valid after the
>+// destructor runs.
>+//
>+// d) That a-c apply transitively when SynchronizationStorage<T> is the
>+// only member of a class allocated in static storage.
>+//
>+// Nothing in the language standard guarantees that a-d hold.  In practice,
>+// these hold in all supported compilers.
>+//
>+// Future direction:
>+//
>+// Ideally, we would simply use std::mutex or a similar class, which when
>+// allocated statically would support use immediately after static
>+// initialization up until static storage is reclaimed (i.e. the properties
>+// we require of all "linker initialized" instances).
>+//
>+// Regarding construction in static storage, std::mutex is required to
>+// provide a constexpr default constructor [thread.mutex.class], which
>+// ensures the instance's lifetime begins with static initialization
>+// [basic.start.init], and so is immune to any problems caused by the order
>+// of dynamic initialization. However, as of this writing Microsoft's
>+// Visual Studio does not provide a constexpr constructor for std::mutex.
>+// See
>+// https://blogs.msdn.microsoft.com/vcblog/2015/06/02/constexpr-complete-for-vs-2015-rtm-c11-compiler-c17-stl/
>+//
>+// Regarding destruction of instances in static storage, [basic.life] does
>+// say an object ends when storage in which the occupies is released, in
>+// the case of non-trivial destructor. However, std::mutex is not specified
>+// to have a trivial destructor.
>+//
>+// So, we would need a class with a constexpr default constructor and a
>+// trivial destructor. Today, we can achieve neither desired property using
>+// std::mutex directly.
>+template <typename T>
>+class SynchronizationStorage {
>+ public:
>+  // Instances allocated on the heap or on the stack should use the default
>+  // constructor.
>+  SynchronizationStorage()
>+      : is_dynamic_(true), once_() {}
>+
>+  // Instances allocated in static storage (not on the heap, not on the
>+  // stack) should use this constructor.
>+  explicit SynchronizationStorage(base_internal::LinkerInitialized) {}
>+
>+  SynchronizationStorage(SynchronizationStorage&) = delete;
>+  SynchronizationStorage& operator=(SynchronizationStorage&) = delete;
>+
>+  ~SynchronizationStorage() {
>+    if (is_dynamic_) {
>+      get()->~T();
>+    }
>+  }
>+
>+  // Retrieve the object in storage. This is fast and thread safe, but does
>+  // incur the cost of absl::call_once().
>+  //
>+  // For instances in static storage constructed with the
>+  // LinkerInitialized constructor, may be called at any time without
>+  // regard for order of dynamic initialization or destruction of objects
>+  // in static storage. See the class comment for caveats.
>+  T* get() {
>+    absl::call_once(once_, SynchronizationStorage::Construct, this);
>+    return reinterpret_cast<T*>(&space_);
>+  }
>+
>+ private:
>+  static void Construct(SynchronizationStorage<T>* self) {
>+    new (&self->space_) T();
>+  }
>+
>+  // When true, T's destructor is run when this is destructed.
>+  //
>+  // The LinkerInitialized constructor assumes this value will be set
>+  // false by static initialization.
>+  bool is_dynamic_;
>+
>+  absl::once_flag once_;
>+
>+  // An aligned space for T.
>+  typename std::aligned_storage<sizeof(T), alignof(T)>::type space_;
>+};
>+
>+}  // namespace synchronization_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/internal/per_thread_sem.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/internal/per_thread_sem.cc
>new file mode 100644
>index 00000000000..caa2baf64bd
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/internal/per_thread_sem.cc
>@@ -0,0 +1,99 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+// This file is a no-op if the required LowLevelAlloc support is missing.
>+#include "absl/base/internal/low_level_alloc.h"
>+#ifndef ABSL_LOW_LEVEL_ALLOC_MISSING
>+
>+#include "absl/synchronization/internal/per_thread_sem.h"
>+
>+#include <atomic>
>+
>+#include "absl/base/attributes.h"
>+#include "absl/base/internal/thread_identity.h"
>+#include "absl/synchronization/internal/waiter.h"
>+
>+namespace absl {
>+namespace synchronization_internal {
>+
>+void PerThreadSem::SetThreadBlockedCounter(std::atomic<int> *counter) {
>+  base_internal::ThreadIdentity *identity;
>+  identity = GetOrCreateCurrentThreadIdentity();
>+  identity->blocked_count_ptr = counter;
>+}
>+
>+std::atomic<int> *PerThreadSem::GetThreadBlockedCounter() {
>+  base_internal::ThreadIdentity *identity;
>+  identity = GetOrCreateCurrentThreadIdentity();
>+  return identity->blocked_count_ptr;
>+}
>+
>+void PerThreadSem::Init(base_internal::ThreadIdentity *identity) {
>+  Waiter::GetWaiter(identity)->Init();
>+  identity->ticker.store(0, std::memory_order_relaxed);
>+  identity->wait_start.store(0, std::memory_order_relaxed);
>+  identity->is_idle.store(false, std::memory_order_relaxed);
>+}
>+
>+void PerThreadSem::Tick(base_internal::ThreadIdentity *identity) {
>+  const int ticker =
>+      identity->ticker.fetch_add(1, std::memory_order_relaxed) + 1;
>+  const int wait_start = identity->wait_start.load(std::memory_order_relaxed);
>+  const bool is_idle = identity->is_idle.load(std::memory_order_relaxed);
>+  if (wait_start && (ticker - wait_start > Waiter::kIdlePeriods) && !is_idle) {
>+    // Wakeup the waiting thread since it is time for it to become idle.
>+    Waiter::GetWaiter(identity)->Poke();
>+  }
>+}
>+
>+}  // namespace synchronization_internal
>+}  // namespace absl
>+
>+extern "C" {
>+
>+ABSL_ATTRIBUTE_WEAK void AbslInternalPerThreadSemPost(
>+    absl::base_internal::ThreadIdentity *identity) {
>+  absl::synchronization_internal::Waiter::GetWaiter(identity)->Post();
>+}
>+
>+ABSL_ATTRIBUTE_WEAK bool AbslInternalPerThreadSemWait(
>+    absl::synchronization_internal::KernelTimeout t) {
>+  bool timeout = false;
>+  absl::base_internal::ThreadIdentity *identity;
>+  identity = absl::synchronization_internal::GetOrCreateCurrentThreadIdentity();
>+
>+  // Ensure wait_start != 0.
>+  int ticker = identity->ticker.load(std::memory_order_relaxed);
>+  identity->wait_start.store(ticker ? ticker : 1, std::memory_order_relaxed);
>+  identity->is_idle.store(false, std::memory_order_relaxed);
>+
>+  if (identity->blocked_count_ptr != nullptr) {
>+    // Increment count of threads blocked in a given thread pool.
>+    identity->blocked_count_ptr->fetch_add(1, std::memory_order_relaxed);
>+  }
>+
>+  timeout =
>+      !absl::synchronization_internal::Waiter::GetWaiter(identity)->Wait(t);
>+
>+  if (identity->blocked_count_ptr != nullptr) {
>+    identity->blocked_count_ptr->fetch_sub(1, std::memory_order_relaxed);
>+  }
>+  identity->is_idle.store(false, std::memory_order_relaxed);
>+  identity->wait_start.store(0, std::memory_order_relaxed);
>+  return !timeout;
>+}
>+
>+}  // extern "C"
>+
>+#endif  // ABSL_LOW_LEVEL_ALLOC_MISSING
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/internal/per_thread_sem.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/internal/per_thread_sem.h
>new file mode 100644
>index 00000000000..678b69e487a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/internal/per_thread_sem.h
>@@ -0,0 +1,107 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+
>+// PerThreadSem is a low-level synchronization primitive controlling the
>+// runnability of a single thread, used internally by Mutex and CondVar.
>+//
>+// This is NOT a general-purpose synchronization mechanism, and should not be
>+// used directly by applications.  Applications should use Mutex and CondVar.
>+//
>+// The semantics of PerThreadSem are the same as that of a counting semaphore.
>+// Each thread maintains an abstract "count" value associated with its identity.
>+
>+#ifndef ABSL_SYNCHRONIZATION_INTERNAL_PER_THREAD_SEM_H_
>+#define ABSL_SYNCHRONIZATION_INTERNAL_PER_THREAD_SEM_H_
>+
>+#include <atomic>
>+
>+#include "absl/base/internal/thread_identity.h"
>+#include "absl/synchronization/internal/create_thread_identity.h"
>+#include "absl/synchronization/internal/kernel_timeout.h"
>+
>+namespace absl {
>+
>+class Mutex;
>+
>+namespace synchronization_internal {
>+
>+class PerThreadSem {
>+ public:
>+  PerThreadSem() = delete;
>+  PerThreadSem(const PerThreadSem&) = delete;
>+  PerThreadSem& operator=(const PerThreadSem&) = delete;
>+
>+  // Routine invoked periodically (once a second) by a background thread.
>+  // Has no effect on user-visible state.
>+  static void Tick(base_internal::ThreadIdentity* identity);
>+
>+  // ---------------------------------------------------------------------------
>+  // Routines used by autosizing threadpools to detect when threads are
>+  // blocked.  Each thread has a counter pointer, initially zero.  If non-zero,
>+  // the implementation atomically increments the counter when it blocks on a
>+  // semaphore, a decrements it again when it wakes.  This allows a threadpool
>+  // to keep track of how many of its threads are blocked.
>+  // SetThreadBlockedCounter() should be used only by threadpool
>+  // implementations.  GetThreadBlockedCounter() should be used by modules that
>+  // block threads; if the pointer returned is non-zero, the location should be
>+  // incremented before the thread blocks, and decremented after it wakes.
>+  static void SetThreadBlockedCounter(std::atomic<int> *counter);
>+  static std::atomic<int> *GetThreadBlockedCounter();
>+
>+ private:
>+  // Create the PerThreadSem associated with "identity".  Initializes count=0.
>+  // REQUIRES: May only be called by ThreadIdentity.
>+  static void Init(base_internal::ThreadIdentity* identity);
>+
>+  // Increments "identity"'s count.
>+  static inline void Post(base_internal::ThreadIdentity* identity);
>+
>+  // Waits until either our count > 0 or t has expired.
>+  // If count > 0, decrements count and returns true.  Otherwise returns false.
>+  // !t.has_timeout() => Wait(t) will return true.
>+  static inline bool Wait(KernelTimeout t);
>+
>+  // White-listed callers.
>+  friend class PerThreadSemTest;
>+  friend class absl::Mutex;
>+  friend absl::base_internal::ThreadIdentity* CreateThreadIdentity();
>+};
>+
>+}  // namespace synchronization_internal
>+}  // namespace absl
>+
>+// In some build configurations we pass --detect-odr-violations to the
>+// gold linker.  This causes it to flag weak symbol overrides as ODR
>+// violations.  Because ODR only applies to C++ and not C,
>+// --detect-odr-violations ignores symbols not mangled with C++ names.
>+// By changing our extension points to be extern "C", we dodge this
>+// check.
>+extern "C" {
>+void AbslInternalPerThreadSemPost(
>+    absl::base_internal::ThreadIdentity* identity);
>+bool AbslInternalPerThreadSemWait(
>+    absl::synchronization_internal::KernelTimeout t);
>+}  // extern "C"
>+
>+void absl::synchronization_internal::PerThreadSem::Post(
>+    absl::base_internal::ThreadIdentity* identity) {
>+  AbslInternalPerThreadSemPost(identity);
>+}
>+
>+bool absl::synchronization_internal::PerThreadSem::Wait(
>+    absl::synchronization_internal::KernelTimeout t) {
>+  return AbslInternalPerThreadSemWait(t);
>+}
>+#endif  // ABSL_SYNCHRONIZATION_INTERNAL_PER_THREAD_SEM_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/internal/per_thread_sem_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/internal/per_thread_sem_test.cc
>new file mode 100644
>index 00000000000..2b52ea76ab0
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/internal/per_thread_sem_test.cc
>@@ -0,0 +1,172 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/synchronization/internal/per_thread_sem.h"
>+
>+#include <atomic>
>+#include <condition_variable>  // NOLINT(build/c++11)
>+#include <functional>
>+#include <limits>
>+#include <mutex>               // NOLINT(build/c++11)
>+#include <string>
>+#include <thread>              // NOLINT(build/c++11)
>+
>+#include "gtest/gtest.h"
>+#include "absl/base/internal/cycleclock.h"
>+#include "absl/base/internal/thread_identity.h"
>+#include "absl/strings/str_cat.h"
>+#include "absl/time/clock.h"
>+#include "absl/time/time.h"
>+
>+// In this test we explicitly avoid the use of synchronization
>+// primitives which might use PerThreadSem, most notably absl::Mutex.
>+
>+namespace absl {
>+namespace synchronization_internal {
>+
>+class SimpleSemaphore {
>+ public:
>+  SimpleSemaphore() : count_(0) {}
>+
>+  // Decrements (locks) the semaphore. If the semaphore's value is
>+  // greater than zero, then the decrement proceeds, and the function
>+  // returns, immediately. If the semaphore currently has the value
>+  // zero, then the call blocks until it becomes possible to perform
>+  // the decrement.
>+  void Wait() {
>+    std::unique_lock<std::mutex> lock(mu_);
>+    cv_.wait(lock, [this]() { return count_ > 0; });
>+    --count_;
>+    cv_.notify_one();
>+  }
>+
>+  // Increments (unlocks) the semaphore. If the semaphore's value
>+  // consequently becomes greater than zero, then another thread
>+  // blocked Wait() call will be woken up and proceed to lock the
>+  // semaphore.
>+  void Post() {
>+    std::lock_guard<std::mutex> lock(mu_);
>+    ++count_;
>+    cv_.notify_one();
>+  }
>+
>+ private:
>+  std::mutex mu_;
>+  std::condition_variable cv_;
>+  int count_;
>+};
>+
>+struct ThreadData {
>+  int num_iterations;                 // Number of replies to send.
>+  SimpleSemaphore identity2_written;  // Posted by thread writing identity2.
>+  base_internal::ThreadIdentity *identity1;  // First Post()-er.
>+  base_internal::ThreadIdentity *identity2;  // First Wait()-er.
>+  KernelTimeout timeout;
>+};
>+
>+// Need friendship with PerThreadSem.
>+class PerThreadSemTest : public testing::Test {
>+ public:
>+  static void TimingThread(ThreadData* t) {
>+    t->identity2 = GetOrCreateCurrentThreadIdentity();
>+    t->identity2_written.Post();
>+    while (t->num_iterations--) {
>+      Wait(t->timeout);
>+      Post(t->identity1);
>+    }
>+  }
>+
>+  void TestTiming(const char *msg, bool timeout) {
>+    static const int kNumIterations = 100;
>+    ThreadData t;
>+    t.num_iterations = kNumIterations;
>+    t.timeout = timeout ?
>+        KernelTimeout(absl::Now() + absl::Seconds(10000))  // far in the future
>+        : KernelTimeout::Never();
>+    t.identity1 = GetOrCreateCurrentThreadIdentity();
>+
>+    // We can't use the Thread class here because it uses the Mutex
>+    // class which will invoke PerThreadSem, so we use std::thread instead.
>+    std::thread partner_thread(std::bind(TimingThread, &t));
>+
>+    // Wait for our partner thread to register their identity.
>+    t.identity2_written.Wait();
>+
>+    int64_t min_cycles = std::numeric_limits<int64_t>::max();
>+    int64_t total_cycles = 0;
>+    for (int i = 0; i < kNumIterations; ++i) {
>+      absl::SleepFor(absl::Milliseconds(20));
>+      int64_t cycles = base_internal::CycleClock::Now();
>+      Post(t.identity2);
>+      Wait(t.timeout);
>+      cycles = base_internal::CycleClock::Now() - cycles;
>+      min_cycles = std::min(min_cycles, cycles);
>+      total_cycles += cycles;
>+    }
>+    std::string out =
>+        StrCat(msg, "min cycle count=", min_cycles, " avg cycle count=",
>+               absl::SixDigits(static_cast<double>(total_cycles) /
>+                               kNumIterations));
>+    printf("%s\n", out.c_str());
>+
>+    partner_thread.join();
>+  }
>+
>+ protected:
>+  static void Post(base_internal::ThreadIdentity *id) {
>+    PerThreadSem::Post(id);
>+  }
>+  static bool Wait(KernelTimeout t) {
>+    return PerThreadSem::Wait(t);
>+  }
>+
>+  // convenience overload
>+  static bool Wait(absl::Time t) {
>+    return Wait(KernelTimeout(t));
>+  }
>+
>+  static void Tick(base_internal::ThreadIdentity *identity) {
>+    PerThreadSem::Tick(identity);
>+  }
>+};
>+
>+namespace {
>+
>+TEST_F(PerThreadSemTest, WithoutTimeout) {
>+  PerThreadSemTest::TestTiming("Without timeout: ", false);
>+}
>+
>+TEST_F(PerThreadSemTest, WithTimeout) {
>+  PerThreadSemTest::TestTiming("With timeout:    ", true);
>+}
>+
>+TEST_F(PerThreadSemTest, Timeouts) {
>+  absl::Time timeout = absl::Now() + absl::Milliseconds(50);
>+  EXPECT_FALSE(Wait(timeout));
>+  EXPECT_LE(timeout, absl::Now());
>+
>+  absl::Time negative_timeout = absl::UnixEpoch() - absl::Milliseconds(100);
>+  EXPECT_FALSE(Wait(negative_timeout));
>+  EXPECT_LE(negative_timeout, absl::Now());  // trivially true :)
>+
>+  Post(GetOrCreateCurrentThreadIdentity());
>+  // The wait here has an expired timeout, but we have a wake to consume,
>+  // so this should succeed
>+  EXPECT_TRUE(Wait(negative_timeout));
>+}
>+
>+}  // namespace
>+
>+}  // namespace synchronization_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/internal/thread_pool.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/internal/thread_pool.h
>new file mode 100644
>index 00000000000..846404277a0
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/internal/thread_pool.h
>@@ -0,0 +1,90 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#ifndef ABSL_SYNCHRONIZATION_INTERNAL_THREAD_POOL_H_
>+#define ABSL_SYNCHRONIZATION_INTERNAL_THREAD_POOL_H_
>+
>+#include <cassert>
>+#include <functional>
>+#include <queue>
>+#include <thread>  // NOLINT(build/c++11)
>+#include <vector>
>+
>+#include "absl/base/thread_annotations.h"
>+#include "absl/synchronization/mutex.h"
>+
>+namespace absl {
>+namespace synchronization_internal {
>+
>+// A simple ThreadPool implementation for tests.
>+class ThreadPool {
>+ public:
>+  explicit ThreadPool(int num_threads) {
>+    for (int i = 0; i < num_threads; ++i) {
>+      threads_.push_back(std::thread(&ThreadPool::WorkLoop, this));
>+    }
>+  }
>+
>+  ThreadPool(const ThreadPool &) = delete;
>+  ThreadPool &operator=(const ThreadPool &) = delete;
>+
>+  ~ThreadPool() {
>+    {
>+      absl::MutexLock l(&mu_);
>+      for (int i = 0; i < threads_.size(); ++i) {
>+        queue_.push(nullptr);  // Shutdown signal.
>+      }
>+    }
>+    for (auto &t : threads_) {
>+      t.join();
>+    }
>+  }
>+
>+  // Schedule a function to be run on a ThreadPool thread immediately.
>+  void Schedule(std::function<void()> func) {
>+    assert(func != nullptr);
>+    absl::MutexLock l(&mu_);
>+    queue_.push(std::move(func));
>+  }
>+
>+ private:
>+  bool WorkAvailable() const EXCLUSIVE_LOCKS_REQUIRED(mu_) {
>+    return !queue_.empty();
>+  }
>+
>+  void WorkLoop() {
>+    while (true) {
>+      std::function<void()> func;
>+      {
>+        absl::MutexLock l(&mu_);
>+        mu_.Await(absl::Condition(this, &ThreadPool::WorkAvailable));
>+        func = std::move(queue_.front());
>+        queue_.pop();
>+      }
>+      if (func == nullptr) {  // Shutdown signal.
>+        break;
>+      }
>+      func();
>+    }
>+  }
>+
>+  absl::Mutex mu_;
>+  std::queue<std::function<void()>> queue_ GUARDED_BY(mu_);
>+  std::vector<std::thread> threads_;
>+};
>+
>+}  // namespace synchronization_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_SYNCHRONIZATION_INTERNAL_THREAD_POOL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/internal/waiter.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/internal/waiter.cc
>new file mode 100644
>index 00000000000..768c52085cd
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/internal/waiter.cc
>@@ -0,0 +1,412 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/synchronization/internal/waiter.h"
>+
>+#include "absl/base/config.h"
>+
>+#ifdef _WIN32
>+#include <windows.h>
>+#else
>+#include <pthread.h>
>+#include <sys/time.h>
>+#include <unistd.h>
>+#endif
>+
>+#ifdef __linux__
>+#include <linux/futex.h>
>+#include <sys/syscall.h>
>+#endif
>+
>+#ifdef ABSL_HAVE_SEMAPHORE_H
>+#include <semaphore.h>
>+#endif
>+
>+#include <errno.h>
>+#include <stdio.h>
>+#include <time.h>
>+
>+#include <atomic>
>+#include <cassert>
>+#include <cstdint>
>+#include "absl/base/internal/raw_logging.h"
>+#include "absl/base/internal/thread_identity.h"
>+#include "absl/base/optimization.h"
>+#include "absl/synchronization/internal/kernel_timeout.h"
>+
>+namespace absl {
>+namespace synchronization_internal {
>+
>+static void MaybeBecomeIdle() {
>+  base_internal::ThreadIdentity *identity =
>+      base_internal::CurrentThreadIdentityIfPresent();
>+  assert(identity != nullptr);
>+  const bool is_idle = identity->is_idle.load(std::memory_order_relaxed);
>+  const int ticker = identity->ticker.load(std::memory_order_relaxed);
>+  const int wait_start = identity->wait_start.load(std::memory_order_relaxed);
>+  if (!is_idle && ticker - wait_start > Waiter::kIdlePeriods) {
>+    identity->is_idle.store(true, std::memory_order_relaxed);
>+  }
>+}
>+
>+#if ABSL_WAITER_MODE == ABSL_WAITER_MODE_FUTEX
>+
>+// Some Android headers are missing these definitions even though they
>+// support these futex operations.
>+#ifdef __BIONIC__
>+#ifndef SYS_futex
>+#define SYS_futex __NR_futex
>+#endif
>+#ifndef FUTEX_WAIT_BITSET
>+#define FUTEX_WAIT_BITSET 9
>+#endif
>+#ifndef FUTEX_PRIVATE_FLAG
>+#define FUTEX_PRIVATE_FLAG 128
>+#endif
>+#ifndef FUTEX_CLOCK_REALTIME
>+#define FUTEX_CLOCK_REALTIME 256
>+#endif
>+#ifndef FUTEX_BITSET_MATCH_ANY
>+#define FUTEX_BITSET_MATCH_ANY 0xFFFFFFFF
>+#endif
>+#endif
>+class Futex {
>+ public:
>+  static int WaitUntil(std::atomic<int32_t> *v, int32_t val,
>+                       KernelTimeout t) {
>+    int err = 0;
>+    if (t.has_timeout()) {
>+      // https://locklessinc.com/articles/futex_cheat_sheet/
>+      // Unlike FUTEX_WAIT, FUTEX_WAIT_BITSET uses absolute time.
>+      struct timespec abs_timeout = t.MakeAbsTimespec();
>+      // Atomically check that the futex value is still 0, and if it
>+      // is, sleep until abs_timeout or until woken by FUTEX_WAKE.
>+      err = syscall(
>+          SYS_futex, reinterpret_cast<int32_t *>(v),
>+          FUTEX_WAIT_BITSET | FUTEX_PRIVATE_FLAG | FUTEX_CLOCK_REALTIME, val,
>+          &abs_timeout, nullptr, FUTEX_BITSET_MATCH_ANY);
>+    } else {
>+      // Atomically check that the futex value is still 0, and if it
>+      // is, sleep until woken by FUTEX_WAKE.
>+      err = syscall(SYS_futex, reinterpret_cast<int32_t *>(v),
>+                    FUTEX_WAIT | FUTEX_PRIVATE_FLAG, val, nullptr);
>+    }
>+    if (err != 0) {
>+      err = -errno;
>+    }
>+    return err;
>+  }
>+
>+  static int Wake(std::atomic<int32_t> *v, int32_t count) {
>+    int err = syscall(SYS_futex, reinterpret_cast<int32_t *>(v),
>+                      FUTEX_WAKE | FUTEX_PRIVATE_FLAG, count);
>+    if (ABSL_PREDICT_FALSE(err < 0)) {
>+      err = -errno;
>+    }
>+    return err;
>+  }
>+};
>+
>+void Waiter::Init() {
>+  futex_.store(0, std::memory_order_relaxed);
>+}
>+
>+bool Waiter::Wait(KernelTimeout t) {
>+  // Loop until we can atomically decrement futex from a positive
>+  // value, waiting on a futex while we believe it is zero.
>+  while (true) {
>+    int32_t x = futex_.load(std::memory_order_relaxed);
>+    if (x != 0) {
>+      if (!futex_.compare_exchange_weak(x, x - 1,
>+                                        std::memory_order_acquire,
>+                                        std::memory_order_relaxed)) {
>+        continue;  // Raced with someone, retry.
>+      }
>+      return true;  // Consumed a wakeup, we are done.
>+    }
>+
>+    const int err = Futex::WaitUntil(&futex_, 0, t);
>+    if (err != 0) {
>+      if (err == -EINTR || err == -EWOULDBLOCK) {
>+        // Do nothing, the loop will retry.
>+      } else if (err == -ETIMEDOUT) {
>+        return false;
>+      } else {
>+        ABSL_RAW_LOG(FATAL, "Futex operation failed with error %d\n", err);
>+      }
>+    }
>+
>+    MaybeBecomeIdle();
>+  }
>+}
>+
>+void Waiter::Post() {
>+  if (futex_.fetch_add(1, std::memory_order_release) == 0) {
>+    // We incremented from 0, need to wake a potential waker.
>+    Poke();
>+  }
>+}
>+
>+void Waiter::Poke() {
>+  // Wake one thread waiting on the futex.
>+  const int err = Futex::Wake(&futex_, 1);
>+  if (ABSL_PREDICT_FALSE(err < 0)) {
>+    ABSL_RAW_LOG(FATAL, "Futex operation failed with error %d\n", err);
>+  }
>+}
>+
>+#elif ABSL_WAITER_MODE == ABSL_WAITER_MODE_CONDVAR
>+
>+class PthreadMutexHolder {
>+ public:
>+  explicit PthreadMutexHolder(pthread_mutex_t *mu) : mu_(mu) {
>+    const int err = pthread_mutex_lock(mu_);
>+    if (err != 0) {
>+      ABSL_RAW_LOG(FATAL, "pthread_mutex_lock failed: %d", err);
>+    }
>+  }
>+
>+  PthreadMutexHolder(const PthreadMutexHolder &rhs) = delete;
>+  PthreadMutexHolder &operator=(const PthreadMutexHolder &rhs) = delete;
>+
>+  ~PthreadMutexHolder() {
>+    const int err = pthread_mutex_unlock(mu_);
>+    if (err != 0) {
>+      ABSL_RAW_LOG(FATAL, "pthread_mutex_unlock failed: %d", err);
>+    }
>+  }
>+
>+ private:
>+  pthread_mutex_t *mu_;
>+};
>+
>+void Waiter::Init() {
>+  const int err = pthread_mutex_init(&mu_, 0);
>+  if (err != 0) {
>+    ABSL_RAW_LOG(FATAL, "pthread_mutex_init failed: %d", err);
>+  }
>+
>+  const int err2 = pthread_cond_init(&cv_, 0);
>+  if (err2 != 0) {
>+    ABSL_RAW_LOG(FATAL, "pthread_cond_init failed: %d", err2);
>+  }
>+
>+  waiter_count_.store(0, std::memory_order_relaxed);
>+  wakeup_count_.store(0, std::memory_order_relaxed);
>+}
>+
>+bool Waiter::Wait(KernelTimeout t) {
>+  struct timespec abs_timeout;
>+  if (t.has_timeout()) {
>+    abs_timeout = t.MakeAbsTimespec();
>+  }
>+
>+  PthreadMutexHolder h(&mu_);
>+  waiter_count_.fetch_add(1, std::memory_order_relaxed);
>+  // Loop until we find a wakeup to consume or timeout.
>+  while (true) {
>+    int x = wakeup_count_.load(std::memory_order_relaxed);
>+    if (x != 0) {
>+      if (!wakeup_count_.compare_exchange_weak(x, x - 1,
>+                                               std::memory_order_acquire,
>+                                               std::memory_order_relaxed)) {
>+        continue;  // Raced with someone, retry.
>+      }
>+      // Successfully consumed a wakeup, we're done.
>+      waiter_count_.fetch_sub(1, std::memory_order_relaxed);
>+      return true;
>+    }
>+
>+    // No wakeups available, time to wait.
>+    if (!t.has_timeout()) {
>+      const int err = pthread_cond_wait(&cv_, &mu_);
>+      if (err != 0) {
>+        ABSL_RAW_LOG(FATAL, "pthread_cond_wait failed: %d", err);
>+      }
>+    } else {
>+      const int err = pthread_cond_timedwait(&cv_, &mu_, &abs_timeout);
>+      if (err == ETIMEDOUT) {
>+        waiter_count_.fetch_sub(1, std::memory_order_relaxed);
>+        return false;
>+      }
>+      if (err != 0) {
>+        ABSL_RAW_LOG(FATAL, "pthread_cond_wait failed: %d", err);
>+      }
>+    }
>+    MaybeBecomeIdle();
>+  }
>+}
>+
>+void Waiter::Post() {
>+  wakeup_count_.fetch_add(1, std::memory_order_release);
>+  Poke();
>+}
>+
>+void Waiter::Poke() {
>+  if (waiter_count_.load(std::memory_order_relaxed) == 0) {
>+    return;
>+  }
>+  // Potentially a waker. Take the lock and check again.
>+  PthreadMutexHolder h(&mu_);
>+  if (waiter_count_.load(std::memory_order_relaxed) == 0) {
>+    return;
>+  }
>+  const int err = pthread_cond_signal(&cv_);
>+  if (err != 0) {
>+    ABSL_RAW_LOG(FATAL, "pthread_cond_signal failed: %d", err);
>+  }
>+}
>+
>+#elif ABSL_WAITER_MODE == ABSL_WAITER_MODE_SEM
>+
>+void Waiter::Init() {
>+  if (sem_init(&sem_, 0, 0) != 0) {
>+    ABSL_RAW_LOG(FATAL, "sem_init failed with errno %d\n", errno);
>+  }
>+  wakeups_.store(0, std::memory_order_relaxed);
>+}
>+
>+bool Waiter::Wait(KernelTimeout t) {
>+  struct timespec abs_timeout;
>+  if (t.has_timeout()) {
>+    abs_timeout = t.MakeAbsTimespec();
>+  }
>+
>+  // Loop until we timeout or consume a wakeup.
>+  while (true) {
>+    int x = wakeups_.load(std::memory_order_relaxed);
>+    if (x != 0) {
>+      if (!wakeups_.compare_exchange_weak(x, x - 1,
>+                                          std::memory_order_acquire,
>+                                          std::memory_order_relaxed)) {
>+        continue;  // Raced with someone, retry.
>+      }
>+      // Successfully consumed a wakeup, we're done.
>+      return true;
>+    }
>+
>+    // Nothing to consume, wait (looping on EINTR).
>+    while (true) {
>+      if (!t.has_timeout()) {
>+        if (sem_wait(&sem_) == 0) break;
>+        if (errno == EINTR) continue;
>+        ABSL_RAW_LOG(FATAL, "sem_wait failed: %d", errno);
>+      } else {
>+        if (sem_timedwait(&sem_, &abs_timeout) == 0) break;
>+        if (errno == EINTR) continue;
>+        if (errno == ETIMEDOUT) return false;
>+        ABSL_RAW_LOG(FATAL, "sem_timedwait failed: %d", errno);
>+      }
>+    }
>+    MaybeBecomeIdle();
>+  }
>+}
>+
>+void Waiter::Post() {
>+  wakeups_.fetch_add(1, std::memory_order_release);  // Post a wakeup.
>+  Poke();
>+}
>+
>+void Waiter::Poke() {
>+  if (sem_post(&sem_) != 0) {  // Wake any semaphore waiter.
>+    ABSL_RAW_LOG(FATAL, "sem_post failed with errno %d\n", errno);
>+  }
>+}
>+
>+#elif ABSL_WAITER_MODE == ABSL_WAITER_MODE_WIN32
>+
>+class LockHolder {
>+ public:
>+  explicit LockHolder(SRWLOCK* mu) : mu_(mu) {
>+    AcquireSRWLockExclusive(mu_);
>+  }
>+
>+  LockHolder(const LockHolder&) = delete;
>+  LockHolder& operator=(const LockHolder&) = delete;
>+
>+  ~LockHolder() {
>+    ReleaseSRWLockExclusive(mu_);
>+  }
>+
>+ private:
>+  SRWLOCK* mu_;
>+};
>+
>+void Waiter::Init() {
>+  InitializeSRWLock(&mu_);
>+  InitializeConditionVariable(&cv_);
>+  waiter_count_.store(0, std::memory_order_relaxed);
>+  wakeup_count_.store(0, std::memory_order_relaxed);
>+}
>+
>+bool Waiter::Wait(KernelTimeout t) {
>+  LockHolder h(&mu_);
>+  waiter_count_.fetch_add(1, std::memory_order_relaxed);
>+
>+  // Loop until we find a wakeup to consume or timeout.
>+  while (true) {
>+    int x = wakeup_count_.load(std::memory_order_relaxed);
>+    if (x != 0) {
>+      if (!wakeup_count_.compare_exchange_weak(x, x - 1,
>+                                               std::memory_order_acquire,
>+                                               std::memory_order_relaxed)) {
>+        continue;  // Raced with someone, retry.
>+      }
>+      // Successfully consumed a wakeup, we're done.
>+      waiter_count_.fetch_sub(1, std::memory_order_relaxed);
>+      return true;
>+    }
>+
>+    // No wakeups available, time to wait.
>+    if (!SleepConditionVariableSRW(
>+            &cv_, &mu_, t.InMillisecondsFromNow(), 0)) {
>+      // GetLastError() returns a Win32 DWORD, but we assign to
>+      // unsigned long to simplify the ABSL_RAW_LOG case below.  The uniform
>+      // initialization guarantees this is not a narrowing conversion.
>+      const unsigned long err{GetLastError()};  // NOLINT(runtime/int)
>+      if (err == ERROR_TIMEOUT) {
>+        waiter_count_.fetch_sub(1, std::memory_order_relaxed);
>+        return false;
>+      } else {
>+        ABSL_RAW_LOG(FATAL, "SleepConditionVariableSRW failed: %lu", err);
>+      }
>+    }
>+
>+    MaybeBecomeIdle();
>+  }
>+}
>+
>+void Waiter::Post() {
>+  wakeup_count_.fetch_add(1, std::memory_order_release);
>+  Poke();
>+}
>+
>+void Waiter::Poke() {
>+  if (waiter_count_.load(std::memory_order_relaxed) == 0) {
>+    return;
>+  }
>+  // Potentially a waker. Take the lock and check again.
>+  LockHolder h(&mu_);
>+  if (waiter_count_.load(std::memory_order_relaxed) == 0) {
>+    return;
>+  }
>+  WakeConditionVariable(&cv_);
>+}
>+
>+#else
>+#error Unknown ABSL_WAITER_MODE
>+#endif
>+
>+}  // namespace synchronization_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/internal/waiter.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/internal/waiter.h
>new file mode 100644
>index 00000000000..23166f4bfa3
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/internal/waiter.h
>@@ -0,0 +1,139 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+
>+#ifndef ABSL_SYNCHRONIZATION_INTERNAL_WAITER_H_
>+#define ABSL_SYNCHRONIZATION_INTERNAL_WAITER_H_
>+
>+#include "absl/base/config.h"
>+
>+#ifdef _WIN32
>+#include <windows.h>
>+#else
>+#include <pthread.h>
>+#endif
>+
>+#ifdef ABSL_HAVE_SEMAPHORE_H
>+#include <semaphore.h>
>+#endif
>+
>+#include <atomic>
>+#include <cstdint>
>+
>+#include "absl/base/internal/thread_identity.h"
>+#include "absl/synchronization/internal/kernel_timeout.h"
>+
>+// May be chosen at compile time via -DABSL_FORCE_WAITER_MODE=<index>
>+#define ABSL_WAITER_MODE_FUTEX 0
>+#define ABSL_WAITER_MODE_SEM 1
>+#define ABSL_WAITER_MODE_CONDVAR 2
>+#define ABSL_WAITER_MODE_WIN32 3
>+
>+#if defined(ABSL_FORCE_WAITER_MODE)
>+#define ABSL_WAITER_MODE ABSL_FORCE_WAITER_MODE
>+#elif defined(_WIN32)
>+#define ABSL_WAITER_MODE ABSL_WAITER_MODE_WIN32
>+#elif defined(__linux__)
>+#define ABSL_WAITER_MODE ABSL_WAITER_MODE_FUTEX
>+#elif defined(ABSL_HAVE_SEMAPHORE_H)
>+#define ABSL_WAITER_MODE ABSL_WAITER_MODE_SEM
>+#else
>+#define ABSL_WAITER_MODE ABSL_WAITER_MODE_CONDVAR
>+#endif
>+
>+namespace absl {
>+namespace synchronization_internal {
>+
>+// Waiter is an OS-specific semaphore.
>+class Waiter {
>+ public:
>+  // No constructor, instances use the reserved space in ThreadIdentity.
>+  // All initialization logic belongs in `Init()`.
>+  Waiter() = delete;
>+  Waiter(const Waiter&) = delete;
>+  Waiter& operator=(const Waiter&) = delete;
>+
>+  // Prepare any data to track waits.
>+  void Init();
>+
>+  // Blocks the calling thread until a matching call to `Post()` or
>+  // `t` has passed. Returns `true` if woken (`Post()` called),
>+  // `false` on timeout.
>+  bool Wait(KernelTimeout t);
>+
>+  // Restart the caller of `Wait()` as with a normal semaphore.
>+  void Post();
>+
>+  // If anyone is waiting, wake them up temporarily and cause them to
>+  // call `MaybeBecomeIdle()`. They will then return to waiting for a
>+  // `Post()` or timeout.
>+  void Poke();
>+
>+  // Returns the Waiter associated with the identity.
>+  static Waiter* GetWaiter(base_internal::ThreadIdentity* identity) {
>+    static_assert(
>+        sizeof(Waiter) <= sizeof(base_internal::ThreadIdentity::WaiterState),
>+        "Insufficient space for Waiter");
>+    return reinterpret_cast<Waiter*>(identity->waiter_state.data);
>+  }
>+
>+  // How many periods to remain idle before releasing resources
>+#ifndef THREAD_SANITIZER
>+  static const int kIdlePeriods = 60;
>+#else
>+  // Memory consumption under ThreadSanitizer is a serious concern,
>+  // so we release resources sooner. The value of 1 leads to 1 to 2 second
>+  // delay before marking a thread as idle.
>+  static const int kIdlePeriods = 1;
>+#endif
>+
>+ private:
>+#if ABSL_WAITER_MODE == ABSL_WAITER_MODE_FUTEX
>+  // Futexes are defined by specification to be 32-bits.
>+  // Thus std::atomic<int32_t> must be just an int32_t with lockfree methods.
>+  std::atomic<int32_t> futex_;
>+  static_assert(sizeof(int32_t) == sizeof(futex_), "Wrong size for futex");
>+
>+#elif ABSL_WAITER_MODE == ABSL_WAITER_MODE_CONDVAR
>+  pthread_mutex_t mu_;
>+  pthread_cond_t cv_;
>+  std::atomic<int> waiter_count_;
>+  std::atomic<int> wakeup_count_;  // Unclaimed wakeups, written under lock.
>+
>+#elif ABSL_WAITER_MODE == ABSL_WAITER_MODE_SEM
>+  sem_t sem_;
>+  // This seems superfluous, but for Poke() we need to cause spurious
>+  // wakeups on the semaphore. Hence we can't actually use the
>+  // semaphore's count.
>+  std::atomic<int> wakeups_;
>+
>+#elif ABSL_WAITER_MODE == ABSL_WAITER_MODE_WIN32
>+  // The Windows API has lots of choices for synchronization
>+  // primivitives.  We are using SRWLOCK and CONDITION_VARIABLE
>+  // because they don't require a destructor to release system
>+  // resources.
>+  SRWLOCK mu_;
>+  CONDITION_VARIABLE cv_;
>+  std::atomic<int> waiter_count_;
>+  std::atomic<int> wakeup_count_;
>+
>+#else
>+  #error Unknown ABSL_WAITER_MODE
>+#endif
>+};
>+
>+}  // namespace synchronization_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_SYNCHRONIZATION_INTERNAL_WAITER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/lifetime_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/lifetime_test.cc
>new file mode 100644
>index 00000000000..90c9009b18f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/lifetime_test.cc
>@@ -0,0 +1,132 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include <cstdlib>
>+#include <thread>  // NOLINT(build/c++11), Abseil test
>+#include <type_traits>
>+
>+#include "absl/base/attributes.h"
>+#include "absl/base/internal/raw_logging.h"
>+#include "absl/base/thread_annotations.h"
>+#include "absl/synchronization/mutex.h"
>+#include "absl/synchronization/notification.h"
>+
>+namespace {
>+
>+// A two-threaded test which checks that Mutex, CondVar, and Notification have
>+// correct basic functionality.  The intent is to establish that they
>+// function correctly in various phases of construction and destruction.
>+//
>+// Thread one acquires a lock on 'mutex', wakes thread two via 'notification',
>+// then waits for 'state' to be set, as signalled by 'condvar'.
>+//
>+// Thread two waits on 'notification', then sets 'state' inside the 'mutex',
>+// signalling the change via 'condvar'.
>+//
>+// These tests use ABSL_RAW_CHECK to validate invariants, rather than EXPECT or
>+// ASSERT from gUnit, because we need to invoke them during global destructors,
>+// when gUnit teardown would have already begun.
>+void ThreadOne(absl::Mutex* mutex, absl::CondVar* condvar,
>+               absl::Notification* notification, bool* state) {
>+  // Test that the notification is in a valid initial state.
>+  ABSL_RAW_CHECK(!notification->HasBeenNotified(), "invalid Notification");
>+  ABSL_RAW_CHECK(*state == false, "*state not initialized");
>+
>+  {
>+    absl::MutexLock lock(mutex);
>+
>+    notification->Notify();
>+    ABSL_RAW_CHECK(notification->HasBeenNotified(), "invalid Notification");
>+
>+    while (*state == false) {
>+      condvar->Wait(mutex);
>+    }
>+  }
>+}
>+
>+void ThreadTwo(absl::Mutex* mutex, absl::CondVar* condvar,
>+               absl::Notification* notification, bool* state) {
>+  ABSL_RAW_CHECK(*state == false, "*state not initialized");
>+
>+  // Wake thread one
>+  notification->WaitForNotification();
>+  ABSL_RAW_CHECK(notification->HasBeenNotified(), "invalid Notification");
>+  {
>+    absl::MutexLock lock(mutex);
>+    *state = true;
>+    condvar->Signal();
>+  }
>+}
>+
>+// Launch thread 1 and thread 2, and block on their completion.
>+// If any of 'mutex', 'condvar', or 'notification' is nullptr, use a locally
>+// constructed instance instead.
>+void RunTests(absl::Mutex* mutex, absl::CondVar* condvar,
>+              absl::Notification* notification) {
>+  absl::Mutex default_mutex;
>+  absl::CondVar default_condvar;
>+  absl::Notification default_notification;
>+  if (!mutex) {
>+    mutex = &default_mutex;
>+  }
>+  if (!condvar) {
>+    condvar = &default_condvar;
>+  }
>+  if (!notification) {
>+    notification = &default_notification;
>+  }
>+  bool state = false;
>+  std::thread thread_one(ThreadOne, mutex, condvar, notification, &state);
>+  std::thread thread_two(ThreadTwo, mutex, condvar, notification, &state);
>+  thread_one.join();
>+  thread_two.join();
>+}
>+
>+void TestLocals() {
>+  absl::Mutex mutex;
>+  absl::CondVar condvar;
>+  absl::Notification notification;
>+  RunTests(&mutex, &condvar, &notification);
>+}
>+
>+// Global variables during start and termination
>+//
>+// In a translation unit, static storage duration variables are initialized in
>+// the order of their definitions, and destroyed in the reverse order of their
>+// definitions.  We can use this to arrange for tests to be run on these objects
>+// before they are created, and after they are destroyed.
>+
>+using Function = void (*)();
>+
>+class OnConstruction {
>+ public:
>+  explicit OnConstruction(Function fn) { fn(); }
>+};
>+
>+class OnDestruction {
>+ public:
>+  explicit OnDestruction(Function fn) : fn_(fn) {}
>+  ~OnDestruction() { fn_(); }
>+ private:
>+  Function fn_;
>+};
>+
>+}  // namespace
>+
>+int main() {
>+  TestLocals();
>+  // Explicitly call exit(0) here, to make it clear that we intend for the
>+  // above global object destructors to run.
>+  std::exit(0);
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/mutex.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/mutex.cc
>new file mode 100644
>index 00000000000..80f34f035fc
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/mutex.cc
>@@ -0,0 +1,2687 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/synchronization/mutex.h"
>+
>+#ifdef _WIN32
>+#include <windows.h>
>+#ifdef ERROR
>+#undef ERROR
>+#endif
>+#else
>+#include <fcntl.h>
>+#include <pthread.h>
>+#include <sched.h>
>+#include <sys/time.h>
>+#endif
>+
>+#include <assert.h>
>+#include <errno.h>
>+#include <stdio.h>
>+#include <stdlib.h>
>+#include <string.h>
>+#include <time.h>
>+
>+#include <algorithm>
>+#include <atomic>
>+#include <cinttypes>
>+#include <thread>  // NOLINT(build/c++11)
>+
>+#include "absl/base/attributes.h"
>+#include "absl/base/config.h"
>+#include "absl/base/dynamic_annotations.h"
>+#include "absl/base/internal/atomic_hook.h"
>+#include "absl/base/internal/cycleclock.h"
>+#include "absl/base/internal/hide_ptr.h"
>+#include "absl/base/internal/low_level_alloc.h"
>+#include "absl/base/internal/raw_logging.h"
>+#include "absl/base/internal/spinlock.h"
>+#include "absl/base/internal/sysinfo.h"
>+#include "absl/base/internal/thread_identity.h"
>+#include "absl/base/port.h"
>+#include "absl/debugging/stacktrace.h"
>+#include "absl/debugging/symbolize.h"
>+#include "absl/synchronization/internal/graphcycles.h"
>+#include "absl/synchronization/internal/per_thread_sem.h"
>+#include "absl/time/time.h"
>+
>+using absl::base_internal::CurrentThreadIdentityIfPresent;
>+using absl::base_internal::PerThreadSynch;
>+using absl::base_internal::ThreadIdentity;
>+using absl::synchronization_internal::GetOrCreateCurrentThreadIdentity;
>+using absl::synchronization_internal::GraphCycles;
>+using absl::synchronization_internal::GraphId;
>+using absl::synchronization_internal::InvalidGraphId;
>+using absl::synchronization_internal::KernelTimeout;
>+using absl::synchronization_internal::PerThreadSem;
>+
>+extern "C" {
>+ABSL_ATTRIBUTE_WEAK void AbslInternalMutexYield() { std::this_thread::yield(); }
>+}  // extern "C"
>+
>+namespace absl {
>+
>+namespace {
>+
>+#if defined(THREAD_SANITIZER)
>+constexpr OnDeadlockCycle kDeadlockDetectionDefault = OnDeadlockCycle::kIgnore;
>+#else
>+constexpr OnDeadlockCycle kDeadlockDetectionDefault = OnDeadlockCycle::kAbort;
>+#endif
>+
>+ABSL_CONST_INIT std::atomic<OnDeadlockCycle> synch_deadlock_detection(
>+    kDeadlockDetectionDefault);
>+ABSL_CONST_INIT std::atomic<bool> synch_check_invariants(false);
>+
>+// ------------------------------------------ spinlock support
>+
>+// Make sure read-only globals used in the Mutex code are contained on the
>+// same cacheline and cacheline aligned to eliminate any false sharing with
>+// other globals from this and other modules.
>+static struct MutexGlobals {
>+  MutexGlobals() {
>+    // Find machine-specific data needed for Delay() and
>+    // TryAcquireWithSpinning(). This runs in the global constructor
>+    // sequence, and before that zeros are safe values.
>+    num_cpus = absl::base_internal::NumCPUs();
>+    spinloop_iterations = num_cpus > 1 ? 1500 : 0;
>+  }
>+  int num_cpus;
>+  int spinloop_iterations;
>+  // Pad this struct to a full cacheline to prevent false sharing.
>+  char padding[ABSL_CACHELINE_SIZE - 2 * sizeof(int)];
>+} ABSL_CACHELINE_ALIGNED mutex_globals;
>+static_assert(
>+    sizeof(MutexGlobals) == ABSL_CACHELINE_SIZE,
>+    "MutexGlobals must occupy an entire cacheline to prevent false sharing");
>+
>+ABSL_CONST_INIT absl::base_internal::AtomicHook<void (*)(int64_t wait_cycles)>
>+    submit_profile_data;
>+ABSL_CONST_INIT absl::base_internal::AtomicHook<
>+    void (*)(const char *msg, const void *obj, int64_t wait_cycles)> mutex_tracer;
>+ABSL_CONST_INIT absl::base_internal::AtomicHook<
>+    void (*)(const char *msg, const void *cv)> cond_var_tracer;
>+ABSL_CONST_INIT absl::base_internal::AtomicHook<
>+    bool (*)(const void *pc, char *out, int out_size)>
>+    symbolizer(absl::Symbolize);
>+
>+}  // namespace
>+
>+void RegisterMutexProfiler(void (*fn)(int64_t wait_timestamp)) {
>+  submit_profile_data.Store(fn);
>+}
>+
>+void RegisterMutexTracer(void (*fn)(const char *msg, const void *obj,
>+                                    int64_t wait_cycles)) {
>+  mutex_tracer.Store(fn);
>+}
>+
>+void RegisterCondVarTracer(void (*fn)(const char *msg, const void *cv)) {
>+  cond_var_tracer.Store(fn);
>+}
>+
>+void RegisterSymbolizer(bool (*fn)(const void *pc, char *out, int out_size)) {
>+  symbolizer.Store(fn);
>+}
>+
>+// spinlock delay on iteration c.  Returns new c.
>+namespace {
>+  enum DelayMode { AGGRESSIVE, GENTLE };
>+};
>+static int Delay(int32_t c, DelayMode mode) {
>+  // If this a uniprocessor, only yield/sleep.  Otherwise, if the mode is
>+  // aggressive then spin many times before yielding.  If the mode is
>+  // gentle then spin only a few times before yielding.  Aggressive spinning is
>+  // used to ensure that an Unlock() call, which  must get the spin lock for
>+  // any thread to make progress gets it without undue delay.
>+  int32_t limit = (mutex_globals.num_cpus > 1) ?
>+      ((mode == AGGRESSIVE) ? 5000 : 250) : 0;
>+  if (c < limit) {
>+    c++;               // spin
>+  } else {
>+    ABSL_TSAN_MUTEX_PRE_DIVERT(0, 0);
>+    if (c == limit) {  // yield once
>+      AbslInternalMutexYield();
>+      c++;
>+    } else {           // then wait
>+      absl::SleepFor(absl::Microseconds(10));
>+      c = 0;
>+    }
>+    ABSL_TSAN_MUTEX_POST_DIVERT(0, 0);
>+  }
>+  return (c);
>+}
>+
>+// --------------------------Generic atomic ops
>+// Ensure that "(*pv & bits) == bits" by doing an atomic update of "*pv" to
>+// "*pv | bits" if necessary.  Wait until (*pv & wait_until_clear)==0
>+// before making any change.
>+// This is used to set flags in mutex and condition variable words.
>+static void AtomicSetBits(std::atomic<intptr_t>* pv, intptr_t bits,
>+                          intptr_t wait_until_clear) {
>+  intptr_t v;
>+  do {
>+    v = pv->load(std::memory_order_relaxed);
>+  } while ((v & bits) != bits &&
>+           ((v & wait_until_clear) != 0 ||
>+            !pv->compare_exchange_weak(v, v | bits,
>+                                       std::memory_order_release,
>+                                       std::memory_order_relaxed)));
>+}
>+
>+// Ensure that "(*pv & bits) == 0" by doing an atomic update of "*pv" to
>+// "*pv & ~bits" if necessary.  Wait until (*pv & wait_until_clear)==0
>+// before making any change.
>+// This is used to unset flags in mutex and condition variable words.
>+static void AtomicClearBits(std::atomic<intptr_t>* pv, intptr_t bits,
>+                            intptr_t wait_until_clear) {
>+  intptr_t v;
>+  do {
>+    v = pv->load(std::memory_order_relaxed);
>+  } while ((v & bits) != 0 &&
>+           ((v & wait_until_clear) != 0 ||
>+            !pv->compare_exchange_weak(v, v & ~bits,
>+                                       std::memory_order_release,
>+                                       std::memory_order_relaxed)));
>+}
>+
>+//------------------------------------------------------------------
>+
>+// Data for doing deadlock detection.
>+static absl::base_internal::SpinLock deadlock_graph_mu(
>+    absl::base_internal::kLinkerInitialized);
>+
>+// graph used to detect deadlocks.
>+static GraphCycles *deadlock_graph GUARDED_BY(deadlock_graph_mu)
>+    PT_GUARDED_BY(deadlock_graph_mu);
>+
>+//------------------------------------------------------------------
>+// An event mechanism for debugging mutex use.
>+// It also allows mutexes to be given names for those who can't handle
>+// addresses, and instead like to give their data structures names like
>+// "Henry", "Fido", or "Rupert IV, King of Yondavia".
>+
>+namespace {  // to prevent name pollution
>+enum {       // Mutex and CondVar events passed as "ev" to PostSynchEvent
>+             // Mutex events
>+  SYNCH_EV_TRYLOCK_SUCCESS,
>+  SYNCH_EV_TRYLOCK_FAILED,
>+  SYNCH_EV_READERTRYLOCK_SUCCESS,
>+  SYNCH_EV_READERTRYLOCK_FAILED,
>+  SYNCH_EV_LOCK,
>+  SYNCH_EV_LOCK_RETURNING,
>+  SYNCH_EV_READERLOCK,
>+  SYNCH_EV_READERLOCK_RETURNING,
>+  SYNCH_EV_UNLOCK,
>+  SYNCH_EV_READERUNLOCK,
>+
>+  // CondVar events
>+  SYNCH_EV_WAIT,
>+  SYNCH_EV_WAIT_RETURNING,
>+  SYNCH_EV_SIGNAL,
>+  SYNCH_EV_SIGNALALL,
>+};
>+
>+enum {                 // Event flags
>+  SYNCH_F_R = 0x01,    // reader event
>+  SYNCH_F_LCK = 0x02,  // PostSynchEvent called with mutex held
>+  SYNCH_F_ACQ = 0x04,  // event is an acquire
>+
>+  SYNCH_F_LCK_W = SYNCH_F_LCK,
>+  SYNCH_F_LCK_R = SYNCH_F_LCK | SYNCH_F_R,
>+  SYNCH_F_ACQ_W = SYNCH_F_ACQ,
>+  SYNCH_F_ACQ_R = SYNCH_F_ACQ | SYNCH_F_R,
>+};
>+}  // anonymous namespace
>+
>+// Properties of the events.
>+static const struct {
>+  int flags;
>+  const char *msg;
>+} event_properties[] = {
>+  { SYNCH_F_LCK_W|SYNCH_F_ACQ_W, "TryLock succeeded " },
>+  { 0,                           "TryLock failed " },
>+  { SYNCH_F_LCK_R|SYNCH_F_ACQ_R, "ReaderTryLock succeeded " },
>+  { 0,                           "ReaderTryLock failed " },
>+  {               SYNCH_F_ACQ_W, "Lock blocking " },
>+  { SYNCH_F_LCK_W,               "Lock returning " },
>+  {               SYNCH_F_ACQ_R, "ReaderLock blocking " },
>+  { SYNCH_F_LCK_R,               "ReaderLock returning " },
>+  { SYNCH_F_LCK_W,               "Unlock " },
>+  { SYNCH_F_LCK_R,               "ReaderUnlock " },
>+  { 0,                           "Wait on " },
>+  { 0,                           "Wait unblocked " },
>+  { 0,                           "Signal on " },
>+  { 0,                           "SignalAll on " },
>+};
>+static absl::base_internal::SpinLock synch_event_mu(
>+    absl::base_internal::kLinkerInitialized);
>+// protects synch_event
>+
>+// Hash table size; should be prime > 2.
>+// Can't be too small, as it's used for deadlock detection information.
>+static const uint32_t kNSynchEvent = 1031;
>+
>+static struct SynchEvent {     // this is a trivial hash table for the events
>+  // struct is freed when refcount reaches 0
>+  int refcount GUARDED_BY(synch_event_mu);
>+
>+  // buckets have linear, 0-terminated  chains
>+  SynchEvent *next GUARDED_BY(synch_event_mu);
>+
>+  // Constant after initialization
>+  uintptr_t masked_addr;  // object at this address is called "name"
>+
>+  // No explicit synchronization used.  Instead we assume that the
>+  // client who enables/disables invariants/logging on a Mutex does so
>+  // while the Mutex is not being concurrently accessed by others.
>+  void (*invariant)(void *arg);  // called on each event
>+  void *arg;            // first arg to (*invariant)()
>+  bool log;             // logging turned on
>+
>+  // Constant after initialization
>+  char name[1];         // actually longer---null-terminated std::string
>+} *synch_event[kNSynchEvent] GUARDED_BY(synch_event_mu);
>+
>+// Ensure that the object at "addr" has a SynchEvent struct associated with it,
>+// set "bits" in the word there (waiting until lockbit is clear before doing
>+// so), and return a refcounted reference that will remain valid until
>+// UnrefSynchEvent() is called.  If a new SynchEvent is allocated,
>+// the std::string name is copied into it.
>+// When used with a mutex, the caller should also ensure that kMuEvent
>+// is set in the mutex word, and similarly for condition variables and kCVEvent.
>+static SynchEvent *EnsureSynchEvent(std::atomic<intptr_t> *addr,
>+                                    const char *name, intptr_t bits,
>+                                    intptr_t lockbit) {
>+  uint32_t h = reinterpret_cast<intptr_t>(addr) % kNSynchEvent;
>+  SynchEvent *e;
>+  // first look for existing SynchEvent struct..
>+  synch_event_mu.Lock();
>+  for (e = synch_event[h];
>+       e != nullptr && e->masked_addr != base_internal::HidePtr(addr);
>+       e = e->next) {
>+  }
>+  if (e == nullptr) {  // no SynchEvent struct found; make one.
>+    if (name == nullptr) {
>+      name = "";
>+    }
>+    size_t l = strlen(name);
>+    e = reinterpret_cast<SynchEvent *>(
>+        base_internal::LowLevelAlloc::Alloc(sizeof(*e) + l));
>+    e->refcount = 2;    // one for return value, one for linked list
>+    e->masked_addr = base_internal::HidePtr(addr);
>+    e->invariant = nullptr;
>+    e->arg = nullptr;
>+    e->log = false;
>+    strcpy(e->name, name);  // NOLINT(runtime/printf)
>+    e->next = synch_event[h];
>+    AtomicSetBits(addr, bits, lockbit);
>+    synch_event[h] = e;
>+  } else {
>+    e->refcount++;      // for return value
>+  }
>+  synch_event_mu.Unlock();
>+  return e;
>+}
>+
>+// Deallocate the SynchEvent *e, whose refcount has fallen to zero.
>+static void DeleteSynchEvent(SynchEvent *e) {
>+  base_internal::LowLevelAlloc::Free(e);
>+}
>+
>+// Decrement the reference count of *e, or do nothing if e==null.
>+static void UnrefSynchEvent(SynchEvent *e) {
>+  if (e != nullptr) {
>+    synch_event_mu.Lock();
>+    bool del = (--(e->refcount) == 0);
>+    synch_event_mu.Unlock();
>+    if (del) {
>+      DeleteSynchEvent(e);
>+    }
>+  }
>+}
>+
>+// Forget the mapping from the object (Mutex or CondVar) at address addr
>+// to SynchEvent object, and clear "bits" in its word (waiting until lockbit
>+// is clear before doing so).
>+static void ForgetSynchEvent(std::atomic<intptr_t> *addr, intptr_t bits,
>+                             intptr_t lockbit) {
>+  uint32_t h = reinterpret_cast<intptr_t>(addr) % kNSynchEvent;
>+  SynchEvent **pe;
>+  SynchEvent *e;
>+  synch_event_mu.Lock();
>+  for (pe = &synch_event[h];
>+       (e = *pe) != nullptr && e->masked_addr != base_internal::HidePtr(addr);
>+       pe = &e->next) {
>+  }
>+  bool del = false;
>+  if (e != nullptr) {
>+    *pe = e->next;
>+    del = (--(e->refcount) == 0);
>+  }
>+  AtomicClearBits(addr, bits, lockbit);
>+  synch_event_mu.Unlock();
>+  if (del) {
>+    DeleteSynchEvent(e);
>+  }
>+}
>+
>+// Return a refcounted reference to the SynchEvent of the object at address
>+// "addr", if any.  The pointer returned is valid until the UnrefSynchEvent() is
>+// called.
>+static SynchEvent *GetSynchEvent(const void *addr) {
>+  uint32_t h = reinterpret_cast<intptr_t>(addr) % kNSynchEvent;
>+  SynchEvent *e;
>+  synch_event_mu.Lock();
>+  for (e = synch_event[h];
>+       e != nullptr && e->masked_addr != base_internal::HidePtr(addr);
>+       e = e->next) {
>+  }
>+  if (e != nullptr) {
>+    e->refcount++;
>+  }
>+  synch_event_mu.Unlock();
>+  return e;
>+}
>+
>+// Called when an event "ev" occurs on a Mutex of CondVar "obj"
>+// if event recording is on
>+static void PostSynchEvent(void *obj, int ev) {
>+  SynchEvent *e = GetSynchEvent(obj);
>+  // logging is on if event recording is on and either there's no event struct,
>+  // or it explicitly says to log
>+  if (e == nullptr || e->log) {
>+    void *pcs[40];
>+    int n = absl::GetStackTrace(pcs, ABSL_ARRAYSIZE(pcs), 1);
>+    // A buffer with enough space for the ASCII for all the PCs, even on a
>+    // 64-bit machine.
>+    char buffer[ABSL_ARRAYSIZE(pcs) * 24];
>+    int pos = snprintf(buffer, sizeof (buffer), " @");
>+    for (int i = 0; i != n; i++) {
>+      pos += snprintf(&buffer[pos], sizeof (buffer) - pos, " %p", pcs[i]);
>+    }
>+    ABSL_RAW_LOG(INFO, "%s%p %s %s", event_properties[ev].msg, obj,
>+                 (e == nullptr ? "" : e->name), buffer);
>+  }
>+  if ((event_properties[ev].flags & SYNCH_F_LCK) != 0 && e != nullptr &&
>+      e->invariant != nullptr) {
>+    (*e->invariant)(e->arg);
>+  }
>+  UnrefSynchEvent(e);
>+}
>+
>+//------------------------------------------------------------------
>+
>+// The SynchWaitParams struct encapsulates the way in which a thread is waiting:
>+// whether it has a timeout, the condition, exclusive/shared, and whether a
>+// condition variable wait has an associated Mutex (as opposed to another
>+// type of lock).  It also points to the PerThreadSynch struct of its thread.
>+// cv_word tells Enqueue() to enqueue on a CondVar using CondVarEnqueue().
>+//
>+// This structure is held on the stack rather than directly in
>+// PerThreadSynch because a thread can be waiting on multiple Mutexes if,
>+// while waiting on one Mutex, the implementation calls a client callback
>+// (such as a Condition function) that acquires another Mutex. We don't
>+// strictly need to allow this, but programmers become confused if we do not
>+// allow them to use functions such a LOG() within Condition functions.  The
>+// PerThreadSynch struct points at the most recent SynchWaitParams struct when
>+// the thread is on a Mutex's waiter queue.
>+struct SynchWaitParams {
>+  SynchWaitParams(Mutex::MuHow how_arg, const Condition *cond_arg,
>+                  KernelTimeout timeout_arg, Mutex *cvmu_arg,
>+                  PerThreadSynch *thread_arg,
>+                  std::atomic<intptr_t> *cv_word_arg)
>+      : how(how_arg),
>+        cond(cond_arg),
>+        timeout(timeout_arg),
>+        cvmu(cvmu_arg),
>+        thread(thread_arg),
>+        cv_word(cv_word_arg),
>+        contention_start_cycles(base_internal::CycleClock::Now()) {}
>+
>+  const Mutex::MuHow how;  // How this thread needs to wait.
>+  const Condition *cond;  // The condition that this thread is waiting for.
>+                          // In Mutex, this field is set to zero if a timeout
>+                          // expires.
>+  KernelTimeout timeout;  // timeout expiry---absolute time
>+                          // In Mutex, this field is set to zero if a timeout
>+                          // expires.
>+  Mutex *const cvmu;      // used for transfer from cond var to mutex
>+  PerThreadSynch *const thread;  // thread that is waiting
>+
>+  // If not null, thread should be enqueued on the CondVar whose state
>+  // word is cv_word instead of queueing normally on the Mutex.
>+  std::atomic<intptr_t> *cv_word;
>+
>+  int64_t contention_start_cycles;  // Time (in cycles) when this thread started
>+                                  // to contend for the mutex.
>+};
>+
>+struct SynchLocksHeld {
>+  int n;              // number of valid entries in locks[]
>+  bool overflow;      // true iff we overflowed the array at some point
>+  struct {
>+    Mutex *mu;        // lock acquired
>+    int32_t count;      // times acquired
>+    GraphId id;       // deadlock_graph id of acquired lock
>+  } locks[40];
>+  // If a thread overfills the array during deadlock detection, we
>+  // continue, discarding information as needed.  If no overflow has
>+  // taken place, we can provide more error checking, such as
>+  // detecting when a thread releases a lock it does not hold.
>+};
>+
>+// A sentinel value in lists that is not 0.
>+// A 0 value is used to mean "not on a list".
>+static PerThreadSynch *const kPerThreadSynchNull =
>+  reinterpret_cast<PerThreadSynch *>(1);
>+
>+static SynchLocksHeld *LocksHeldAlloc() {
>+  SynchLocksHeld *ret = reinterpret_cast<SynchLocksHeld *>(
>+      base_internal::LowLevelAlloc::Alloc(sizeof(SynchLocksHeld)));
>+  ret->n = 0;
>+  ret->overflow = false;
>+  return ret;
>+}
>+
>+// Return the PerThreadSynch-struct for this thread.
>+static PerThreadSynch *Synch_GetPerThread() {
>+  ThreadIdentity *identity = GetOrCreateCurrentThreadIdentity();
>+  return &identity->per_thread_synch;
>+}
>+
>+static PerThreadSynch *Synch_GetPerThreadAnnotated(Mutex *mu) {
>+  if (mu) {
>+    ABSL_TSAN_MUTEX_PRE_DIVERT(mu, 0);
>+  }
>+  PerThreadSynch *w = Synch_GetPerThread();
>+  if (mu) {
>+    ABSL_TSAN_MUTEX_POST_DIVERT(mu, 0);
>+  }
>+  return w;
>+}
>+
>+static SynchLocksHeld *Synch_GetAllLocks() {
>+  PerThreadSynch *s = Synch_GetPerThread();
>+  if (s->all_locks == nullptr) {
>+    s->all_locks = LocksHeldAlloc();  // Freed by ReclaimThreadIdentity.
>+  }
>+  return s->all_locks;
>+}
>+
>+// Post on "w"'s associated PerThreadSem.
>+inline void Mutex::IncrementSynchSem(Mutex *mu, PerThreadSynch *w) {
>+  if (mu) {
>+    ABSL_TSAN_MUTEX_PRE_DIVERT(mu, 0);
>+  }
>+  PerThreadSem::Post(w->thread_identity());
>+  if (mu) {
>+    ABSL_TSAN_MUTEX_POST_DIVERT(mu, 0);
>+  }
>+}
>+
>+// Wait on "w"'s associated PerThreadSem; returns false if timeout expired.
>+bool Mutex::DecrementSynchSem(Mutex *mu, PerThreadSynch *w, KernelTimeout t) {
>+  if (mu) {
>+    ABSL_TSAN_MUTEX_PRE_DIVERT(mu, 0);
>+  }
>+  assert(w == Synch_GetPerThread());
>+  static_cast<void>(w);
>+  bool res = PerThreadSem::Wait(t);
>+  if (mu) {
>+    ABSL_TSAN_MUTEX_POST_DIVERT(mu, 0);
>+  }
>+  return res;
>+}
>+
>+// We're in a fatal signal handler that hopes to use Mutex and to get
>+// lucky by not deadlocking.  We try to improve its chances of success
>+// by effectively disabling some of the consistency checks.  This will
>+// prevent certain ABSL_RAW_CHECK() statements from being triggered when
>+// re-rentry is detected.  The ABSL_RAW_CHECK() statements are those in the
>+// Mutex code checking that the "waitp" field has not been reused.
>+void Mutex::InternalAttemptToUseMutexInFatalSignalHandler() {
>+  // Fix the per-thread state only if it exists.
>+  ThreadIdentity *identity = CurrentThreadIdentityIfPresent();
>+  if (identity != nullptr) {
>+    identity->per_thread_synch.suppress_fatal_errors = true;
>+  }
>+  // Don't do deadlock detection when we are already failing.
>+  synch_deadlock_detection.store(OnDeadlockCycle::kIgnore,
>+                                 std::memory_order_release);
>+}
>+
>+// --------------------------time support
>+
>+// Return the current time plus the timeout.  Use the same clock as
>+// PerThreadSem::Wait() for consistency.  Unfortunately, we don't have
>+// such a choice when a deadline is given directly.
>+static absl::Time DeadlineFromTimeout(absl::Duration timeout) {
>+#ifndef _WIN32
>+  struct timeval tv;
>+  gettimeofday(&tv, nullptr);
>+  return absl::TimeFromTimeval(tv) + timeout;
>+#else
>+  return absl::Now() + timeout;
>+#endif
>+}
>+
>+// --------------------------Mutexes
>+
>+// In the layout below, the msb of the bottom byte is currently unused.  Also,
>+// the following constraints were considered in choosing the layout:
>+//  o Both the debug allocator's "uninitialized" and "freed" patterns (0xab and
>+//    0xcd) are illegal: reader and writer lock both held.
>+//  o kMuWriter and kMuEvent should exceed kMuDesig and kMuWait, to enable the
>+//    bit-twiddling trick in Mutex::Unlock().
>+//  o kMuWriter / kMuReader == kMuWrWait / kMuWait,
>+//    to enable the bit-twiddling trick in CheckForMutexCorruption().
>+static const intptr_t kMuReader      = 0x0001L;  // a reader holds the lock
>+static const intptr_t kMuDesig       = 0x0002L;  // there's a designated waker
>+static const intptr_t kMuWait        = 0x0004L;  // threads are waiting
>+static const intptr_t kMuWriter      = 0x0008L;  // a writer holds the lock
>+static const intptr_t kMuEvent       = 0x0010L;  // record this mutex's events
>+// INVARIANT1:  there's a thread that was blocked on the mutex, is
>+// no longer, yet has not yet acquired the mutex.  If there's a
>+// designated waker, all threads can avoid taking the slow path in
>+// unlock because the designated waker will subsequently acquire
>+// the lock and wake someone.  To maintain INVARIANT1 the bit is
>+// set when a thread is unblocked(INV1a), and threads that were
>+// unblocked reset the bit when they either acquire or re-block
>+// (INV1b).
>+static const intptr_t kMuWrWait      = 0x0020L;  // runnable writer is waiting
>+                                                 // for a reader
>+static const intptr_t kMuSpin        = 0x0040L;  // spinlock protects wait list
>+static const intptr_t kMuLow         = 0x00ffL;  // mask all mutex bits
>+static const intptr_t kMuHigh        = ~kMuLow;  // mask pointer/reader count
>+
>+// Hack to make constant values available to gdb pretty printer
>+enum {
>+  kGdbMuSpin = kMuSpin,
>+  kGdbMuEvent = kMuEvent,
>+  kGdbMuWait = kMuWait,
>+  kGdbMuWriter = kMuWriter,
>+  kGdbMuDesig = kMuDesig,
>+  kGdbMuWrWait = kMuWrWait,
>+  kGdbMuReader = kMuReader,
>+  kGdbMuLow = kMuLow,
>+};
>+
>+// kMuWrWait implies kMuWait.
>+// kMuReader and kMuWriter are mutually exclusive.
>+// If kMuReader is zero, there are no readers.
>+// Otherwise, if kMuWait is zero, the high order bits contain a count of the
>+// number of readers.  Otherwise, the reader count is held in
>+// PerThreadSynch::readers of the most recently queued waiter, again in the
>+// bits above kMuLow.
>+static const intptr_t kMuOne = 0x0100;  // a count of one reader
>+
>+// flags passed to Enqueue and LockSlow{,WithTimeout,Loop}
>+static const int kMuHasBlocked = 0x01;  // already blocked (MUST == 1)
>+static const int kMuIsCond = 0x02;      // conditional waiter (CV or Condition)
>+
>+static_assert(PerThreadSynch::kAlignment > kMuLow,
>+              "PerThreadSynch::kAlignment must be greater than kMuLow");
>+
>+// This struct contains various bitmasks to be used in
>+// acquiring and releasing a mutex in a particular mode.
>+struct MuHowS {
>+  // if all the bits in fast_need_zero are zero, the lock can be acquired by
>+  // adding fast_add and oring fast_or.  The bit kMuDesig should be reset iff
>+  // this is the designated waker.
>+  intptr_t fast_need_zero;
>+  intptr_t fast_or;
>+  intptr_t fast_add;
>+
>+  intptr_t slow_need_zero;  // fast_need_zero with events (e.g. logging)
>+
>+  intptr_t slow_inc_need_zero;  // if all the bits in slow_inc_need_zero are
>+                                // zero a reader can acquire a read share by
>+                                // setting the reader bit and incrementing
>+                                // the reader count (in last waiter since
>+                                // we're now slow-path).  kMuWrWait be may
>+                                // be ignored if we already waited once.
>+};
>+
>+static const MuHowS kSharedS = {
>+    // shared or read lock
>+    kMuWriter | kMuWait | kMuEvent,   // fast_need_zero
>+    kMuReader,                        // fast_or
>+    kMuOne,                           // fast_add
>+    kMuWriter | kMuWait,              // slow_need_zero
>+    kMuSpin | kMuWriter | kMuWrWait,  // slow_inc_need_zero
>+};
>+static const MuHowS kExclusiveS = {
>+    // exclusive or write lock
>+    kMuWriter | kMuReader | kMuEvent,  // fast_need_zero
>+    kMuWriter,                         // fast_or
>+    0,                                 // fast_add
>+    kMuWriter | kMuReader,             // slow_need_zero
>+    ~static_cast<intptr_t>(0),         // slow_inc_need_zero
>+};
>+static const Mutex::MuHow kShared = &kSharedS;        // shared lock
>+static const Mutex::MuHow kExclusive = &kExclusiveS;  // exclusive lock
>+
>+#ifdef NDEBUG
>+static constexpr bool kDebugMode = false;
>+#else
>+static constexpr bool kDebugMode = true;
>+#endif
>+
>+#ifdef THREAD_SANITIZER
>+static unsigned TsanFlags(Mutex::MuHow how) {
>+  return how == kShared ? __tsan_mutex_read_lock : 0;
>+}
>+#endif
>+
>+static bool DebugOnlyIsExiting() {
>+  return false;
>+}
>+
>+Mutex::~Mutex() {
>+  intptr_t v = mu_.load(std::memory_order_relaxed);
>+  if ((v & kMuEvent) != 0 && !DebugOnlyIsExiting()) {
>+    ForgetSynchEvent(&this->mu_, kMuEvent, kMuSpin);
>+  }
>+  if (kDebugMode) {
>+    this->ForgetDeadlockInfo();
>+  }
>+  ABSL_TSAN_MUTEX_DESTROY(this, __tsan_mutex_not_static);
>+}
>+
>+void Mutex::EnableDebugLog(const char *name) {
>+  SynchEvent *e = EnsureSynchEvent(&this->mu_, name, kMuEvent, kMuSpin);
>+  e->log = true;
>+  UnrefSynchEvent(e);
>+}
>+
>+void EnableMutexInvariantDebugging(bool enabled) {
>+  synch_check_invariants.store(enabled, std::memory_order_release);
>+}
>+
>+void Mutex::EnableInvariantDebugging(void (*invariant)(void *),
>+                                     void *arg) {
>+  if (synch_check_invariants.load(std::memory_order_acquire) &&
>+      invariant != nullptr) {
>+    SynchEvent *e = EnsureSynchEvent(&this->mu_, nullptr, kMuEvent, kMuSpin);
>+    e->invariant = invariant;
>+    e->arg = arg;
>+    UnrefSynchEvent(e);
>+  }
>+}
>+
>+void SetMutexDeadlockDetectionMode(OnDeadlockCycle mode) {
>+  synch_deadlock_detection.store(mode, std::memory_order_release);
>+}
>+
>+// Return true iff threads x and y are waiting on the same condition for the
>+// same type of lock.  Requires that x and y be waiting on the same Mutex
>+// queue.
>+static bool MuSameCondition(PerThreadSynch *x, PerThreadSynch *y) {
>+  return x->waitp->how == y->waitp->how &&
>+         Condition::GuaranteedEqual(x->waitp->cond, y->waitp->cond);
>+}
>+
>+// Given the contents of a mutex word containing a PerThreadSynch pointer,
>+// return the pointer.
>+static inline PerThreadSynch *GetPerThreadSynch(intptr_t v) {
>+  return reinterpret_cast<PerThreadSynch *>(v & kMuHigh);
>+}
>+
>+// The next several routines maintain the per-thread next and skip fields
>+// used in the Mutex waiter queue.
>+// The queue is a circular singly-linked list, of which the "head" is the
>+// last element, and head->next if the first element.
>+// The skip field has the invariant:
>+//   For thread x, x->skip is one of:
>+//     - invalid (iff x is not in a Mutex wait queue),
>+//     - null, or
>+//     - a pointer to a distinct thread waiting later in the same Mutex queue
>+//       such that all threads in [x, x->skip] have the same condition and
>+//       lock type (MuSameCondition() is true for all pairs in [x, x->skip]).
>+// In addition, if x->skip is  valid, (x->may_skip || x->skip == null)
>+//
>+// By the spec of MuSameCondition(), it is not necessary when removing the
>+// first runnable thread y from the front a Mutex queue to adjust the skip
>+// field of another thread x because if x->skip==y, x->skip must (have) become
>+// invalid before y is removed.  The function TryRemove can remove a specified
>+// thread from an arbitrary position in the queue whether runnable or not, so
>+// it fixes up skip fields that would otherwise be left dangling.
>+// The statement
>+//     if (x->may_skip && MuSameCondition(x, x->next)) { x->skip = x->next; }
>+// maintains the invariant provided x is not the last waiter in a Mutex queue
>+// The statement
>+//          if (x->skip != null) { x->skip = x->skip->skip; }
>+// maintains the invariant.
>+
>+// Returns the last thread y in a mutex waiter queue such that all threads in
>+// [x, y] inclusive share the same condition.  Sets skip fields of some threads
>+// in that range to optimize future evaluation of Skip() on x values in
>+// the range.  Requires thread x is in a mutex waiter queue.
>+// The locking is unusual.  Skip() is called under these conditions:
>+//   - spinlock is held in call from Enqueue(), with maybe_unlocking == false
>+//   - Mutex is held in call from UnlockSlow() by last unlocker, with
>+//     maybe_unlocking == true
>+//   - both Mutex and spinlock are held in call from DequeueAllWakeable() (from
>+//     UnlockSlow()) and TryRemove()
>+// These cases are mutually exclusive, so Skip() never runs concurrently
>+// with itself on the same Mutex.   The skip chain is used in these other places
>+// that cannot occur concurrently:
>+//   - FixSkip() (from TryRemove()) - spinlock and Mutex are held)
>+//   - Dequeue() (with spinlock and Mutex held)
>+//   - UnlockSlow() (with spinlock and Mutex held)
>+// A more complex case is Enqueue()
>+//   - Enqueue() (with spinlock held and maybe_unlocking == false)
>+//               This is the first case in which Skip is called, above.
>+//   - Enqueue() (without spinlock held; but queue is empty and being freshly
>+//                formed)
>+//   - Enqueue() (with spinlock held and maybe_unlocking == true)
>+// The first case has mutual exclusion, and the second isolation through
>+// working on an otherwise unreachable data structure.
>+// In the last case, Enqueue() is required to change no skip/next pointers
>+// except those in the added node and the former "head" node.  This implies
>+// that the new node is added after head, and so must be the new head or the
>+// new front of the queue.
>+static PerThreadSynch *Skip(PerThreadSynch *x) {
>+  PerThreadSynch *x0 = nullptr;
>+  PerThreadSynch *x1 = x;
>+  PerThreadSynch *x2 = x->skip;
>+  if (x2 != nullptr) {
>+    // Each iteration attempts to advance sequence (x0,x1,x2) to next sequence
>+    // such that   x1 == x0->skip && x2 == x1->skip
>+    while ((x0 = x1, x1 = x2, x2 = x2->skip) != nullptr) {
>+      x0->skip = x2;      // short-circuit skip from x0 to x2
>+    }
>+    x->skip = x1;         // short-circuit skip from x to result
>+  }
>+  return x1;
>+}
>+
>+// "ancestor" appears before "to_be_removed" in the same Mutex waiter queue.
>+// The latter is going to be removed out of order, because of a timeout.
>+// Check whether "ancestor" has a skip field pointing to "to_be_removed",
>+// and fix it if it does.
>+static void FixSkip(PerThreadSynch *ancestor, PerThreadSynch *to_be_removed) {
>+  if (ancestor->skip == to_be_removed) {  // ancestor->skip left dangling
>+    if (to_be_removed->skip != nullptr) {
>+      ancestor->skip = to_be_removed->skip;  // can skip past to_be_removed
>+    } else if (ancestor->next != to_be_removed) {  // they are not adjacent
>+      ancestor->skip = ancestor->next;             // can skip one past ancestor
>+    } else {
>+      ancestor->skip = nullptr;  // can't skip at all
>+    }
>+  }
>+}
>+
>+static void CondVarEnqueue(SynchWaitParams *waitp);
>+
>+// Enqueue thread "waitp->thread" on a waiter queue.
>+// Called with mutex spinlock held if head != nullptr
>+// If head==nullptr and waitp->cv_word==nullptr, then Enqueue() is
>+// idempotent; it alters no state associated with the existing (empty)
>+// queue.
>+//
>+// If waitp->cv_word == nullptr, queue the thread at either the front or
>+// the end (according to its priority) of the circular mutex waiter queue whose
>+// head is "head", and return the new head.  mu is the previous mutex state,
>+// which contains the reader count (perhaps adjusted for the operation in
>+// progress) if the list was empty and a read lock held, and the holder hint if
>+// the list was empty and a write lock held.  (flags & kMuIsCond) indicates
>+// whether this thread was transferred from a CondVar or is waiting for a
>+// non-trivial condition.  In this case, Enqueue() never returns nullptr
>+//
>+// If waitp->cv_word != nullptr, CondVarEnqueue() is called, and "head" is
>+// returned. This mechanism is used by CondVar to queue a thread on the
>+// condition variable queue instead of the mutex queue in implementing Wait().
>+// In this case, Enqueue() can return nullptr (if head==nullptr).
>+static PerThreadSynch *Enqueue(PerThreadSynch *head,
>+                               SynchWaitParams *waitp, intptr_t mu, int flags) {
>+  // If we have been given a cv_word, call CondVarEnqueue() and return
>+  // the previous head of the Mutex waiter queue.
>+  if (waitp->cv_word != nullptr) {
>+    CondVarEnqueue(waitp);
>+    return head;
>+  }
>+
>+  PerThreadSynch *s = waitp->thread;
>+  ABSL_RAW_CHECK(
>+      s->waitp == nullptr ||    // normal case
>+          s->waitp == waitp ||  // Fer()---transfer from condition variable
>+          s->suppress_fatal_errors,
>+      "detected illegal recursion into Mutex code");
>+  s->waitp = waitp;
>+  s->skip = nullptr;             // maintain skip invariant (see above)
>+  s->may_skip = true;            // always true on entering queue
>+  s->wake = false;               // not being woken
>+  s->cond_waiter = ((flags & kMuIsCond) != 0);
>+  if (head == nullptr) {         // s is the only waiter
>+    s->next = s;                 // it's the only entry in the cycle
>+    s->readers = mu;             // reader count is from mu word
>+    s->maybe_unlocking = false;  // no one is searching an empty list
>+    head = s;                    // s is new head
>+  } else {
>+    PerThreadSynch *enqueue_after = nullptr;  // we'll put s after this element
>+#ifdef ABSL_HAVE_PTHREAD_GETSCHEDPARAM
>+    int64_t now_cycles = base_internal::CycleClock::Now();
>+    if (s->next_priority_read_cycles < now_cycles) {
>+      // Every so often, update our idea of the thread's priority.
>+      // pthread_getschedparam() is 5% of the block/wakeup time;
>+      // base_internal::CycleClock::Now() is 0.5%.
>+      int policy;
>+      struct sched_param param;
>+      pthread_getschedparam(pthread_self(), &policy, &param);
>+      s->priority = param.sched_priority;
>+      s->next_priority_read_cycles =
>+          now_cycles +
>+          static_cast<int64_t>(base_internal::CycleClock::Frequency());
>+    }
>+    if (s->priority > head->priority) {  // s's priority is above head's
>+      // try to put s in priority-fifo order, or failing that at the front.
>+      if (!head->maybe_unlocking) {
>+        // No unlocker can be scanning the queue, so we can insert between
>+        // skip-chains, and within a skip-chain if it has the same condition as
>+        // s.  We insert in priority-fifo order, examining the end of every
>+        // skip-chain, plus every element with the same condition as s.
>+        PerThreadSynch *advance_to = head;    // next value of enqueue_after
>+        PerThreadSynch *cur;                  // successor of enqueue_after
>+        do {
>+          enqueue_after = advance_to;
>+          cur = enqueue_after->next;  // this advance ensures progress
>+          advance_to = Skip(cur);   // normally, advance to end of skip chain
>+                                    // (side-effect: optimizes skip chain)
>+          if (advance_to != cur && s->priority > advance_to->priority &&
>+              MuSameCondition(s, cur)) {
>+            // but this skip chain is not a singleton, s has higher priority
>+            // than its tail and has the same condition as the chain,
>+            // so we can insert within the skip-chain
>+            advance_to = cur;         // advance by just one
>+          }
>+        } while (s->priority <= advance_to->priority);
>+              // termination guaranteed because s->priority > head->priority
>+              // and head is the end of a skip chain
>+      } else if (waitp->how == kExclusive &&
>+                 Condition::GuaranteedEqual(waitp->cond, nullptr)) {
>+        // An unlocker could be scanning the queue, but we know it will recheck
>+        // the queue front for writers that have no condition, which is what s
>+        // is, so an insert at front is safe.
>+        enqueue_after = head;       // add after head, at front
>+      }
>+    }
>+#endif
>+    if (enqueue_after != nullptr) {
>+      s->next = enqueue_after->next;
>+      enqueue_after->next = s;
>+
>+      // enqueue_after can be: head, Skip(...), or cur.
>+      // The first two imply enqueue_after->skip == nullptr, and
>+      // the last is used only if MuSameCondition(s, cur).
>+      // We require this because clearing enqueue_after->skip
>+      // is impossible; enqueue_after's predecessors might also
>+      // incorrectly skip over s if we were to allow other
>+      // insertion points.
>+      ABSL_RAW_CHECK(
>+          enqueue_after->skip == nullptr || MuSameCondition(enqueue_after, s),
>+          "Mutex Enqueue failure");
>+
>+      if (enqueue_after != head && enqueue_after->may_skip &&
>+          MuSameCondition(enqueue_after, enqueue_after->next)) {
>+        // enqueue_after can skip to its new successor, s
>+        enqueue_after->skip = enqueue_after->next;
>+      }
>+      if (MuSameCondition(s, s->next)) {  // s->may_skip is known to be true
>+        s->skip = s->next;                // s may skip to its successor
>+      }
>+    } else {   // enqueue not done any other way, so
>+               // we're inserting s at the back
>+      // s will become new head; copy data from head into it
>+      s->next = head->next;        // add s after head
>+      head->next = s;
>+      s->readers = head->readers;  // reader count is from previous head
>+      s->maybe_unlocking = head->maybe_unlocking;  // same for unlock hint
>+      if (head->may_skip && MuSameCondition(head, s)) {
>+        // head now has successor; may skip
>+        head->skip = s;
>+      }
>+      head = s;  // s is new head
>+    }
>+  }
>+  s->state.store(PerThreadSynch::kQueued, std::memory_order_relaxed);
>+  return head;
>+}
>+
>+// Dequeue the successor pw->next of thread pw from the Mutex waiter queue
>+// whose last element is head.  The new head element is returned, or null
>+// if the list is made empty.
>+// Dequeue is called with both spinlock and Mutex held.
>+static PerThreadSynch *Dequeue(PerThreadSynch *head, PerThreadSynch *pw) {
>+  PerThreadSynch *w = pw->next;
>+  pw->next = w->next;         // snip w out of list
>+  if (head == w) {            // we removed the head
>+    head = (pw == w) ? nullptr : pw;  // either emptied list, or pw is new head
>+  } else if (pw != head && MuSameCondition(pw, pw->next)) {
>+    // pw can skip to its new successor
>+    if (pw->next->skip !=
>+        nullptr) {  // either skip to its successors skip target
>+      pw->skip = pw->next->skip;
>+    } else {                   // or to pw's successor
>+      pw->skip = pw->next;
>+    }
>+  }
>+  return head;
>+}
>+
>+// Traverse the elements [ pw->next, h] of the circular list whose last element
>+// is head.
>+// Remove all elements with wake==true and place them in the
>+// singly-linked list wake_list in the order found.   Assumes that
>+// there is only one such element if the element has how == kExclusive.
>+// Return the new head.
>+static PerThreadSynch *DequeueAllWakeable(PerThreadSynch *head,
>+                                          PerThreadSynch *pw,
>+                                          PerThreadSynch **wake_tail) {
>+  PerThreadSynch *orig_h = head;
>+  PerThreadSynch *w = pw->next;
>+  bool skipped = false;
>+  do {
>+    if (w->wake) {                    // remove this element
>+      ABSL_RAW_CHECK(pw->skip == nullptr, "bad skip in DequeueAllWakeable");
>+      // we're removing pw's successor so either pw->skip is zero or we should
>+      // already have removed pw since if pw->skip!=null, pw has the same
>+      // condition as w.
>+      head = Dequeue(head, pw);
>+      w->next = *wake_tail;           // keep list terminated
>+      *wake_tail = w;                 // add w to wake_list;
>+      wake_tail = &w->next;           // next addition to end
>+      if (w->waitp->how == kExclusive) {  // wake at most 1 writer
>+        break;
>+      }
>+    } else {                // not waking this one; skip
>+      pw = Skip(w);       // skip as much as possible
>+      skipped = true;
>+    }
>+    w = pw->next;
>+    // We want to stop processing after we've considered the original head,
>+    // orig_h.  We can't test for w==orig_h in the loop because w may skip over
>+    // it; we are guaranteed only that w's predecessor will not skip over
>+    // orig_h.  When we've considered orig_h, either we've processed it and
>+    // removed it (so orig_h != head), or we considered it and skipped it (so
>+    // skipped==true && pw == head because skipping from head always skips by
>+    // just one, leaving pw pointing at head).  So we want to
>+    // continue the loop with the negation of that expression.
>+  } while (orig_h == head && (pw != head || !skipped));
>+  return head;
>+}
>+
>+// Try to remove thread s from the list of waiters on this mutex.
>+// Does nothing if s is not on the waiter list.
>+void Mutex::TryRemove(PerThreadSynch *s) {
>+  intptr_t v = mu_.load(std::memory_order_relaxed);
>+  // acquire spinlock & lock
>+  if ((v & (kMuWait | kMuSpin | kMuWriter | kMuReader)) == kMuWait &&
>+      mu_.compare_exchange_strong(v, v | kMuSpin | kMuWriter,
>+                                  std::memory_order_acquire,
>+                                  std::memory_order_relaxed)) {
>+    PerThreadSynch *h = GetPerThreadSynch(v);
>+    if (h != nullptr) {
>+      PerThreadSynch *pw = h;   // pw is w's predecessor
>+      PerThreadSynch *w;
>+      if ((w = pw->next) != s) {  // search for thread,
>+        do {                      // processing at least one element
>+          if (!MuSameCondition(s, w)) {  // seeking different condition
>+            pw = Skip(w);                // so skip all that won't match
>+            // we don't have to worry about dangling skip fields
>+            // in the threads we skipped; none can point to s
>+            // because their condition differs from s
>+          } else {          // seeking same condition
>+            FixSkip(w, s);  // fix up any skip pointer from w to s
>+            pw = w;
>+          }
>+          // don't search further if we found the thread, or we're about to
>+          // process the first thread again.
>+        } while ((w = pw->next) != s && pw != h);
>+      }
>+      if (w == s) {                 // found thread; remove it
>+        // pw->skip may be non-zero here; the loop above ensured that
>+        // no ancestor of s can skip to s, so removal is safe anyway.
>+        h = Dequeue(h, pw);
>+        s->next = nullptr;
>+        s->state.store(PerThreadSynch::kAvailable, std::memory_order_release);
>+      }
>+    }
>+    intptr_t nv;
>+    do {                        // release spinlock and lock
>+      v = mu_.load(std::memory_order_relaxed);
>+      nv = v & (kMuDesig | kMuEvent);
>+      if (h != nullptr) {
>+        nv |= kMuWait | reinterpret_cast<intptr_t>(h);
>+        h->readers = 0;            // we hold writer lock
>+        h->maybe_unlocking = false;  // finished unlocking
>+      }
>+    } while (!mu_.compare_exchange_weak(v, nv,
>+                                        std::memory_order_release,
>+                                        std::memory_order_relaxed));
>+  }
>+}
>+
>+// Wait until thread "s", which must be the current thread, is removed from the
>+// this mutex's waiter queue.  If "s->waitp->timeout" has a timeout, wake up
>+// if the wait extends past the absolute time specified, even if "s" is still
>+// on the mutex queue.  In this case, remove "s" from the queue and return
>+// true, otherwise return false.
>+void Mutex::Block(PerThreadSynch *s) {
>+  while (s->state.load(std::memory_order_acquire) == PerThreadSynch::kQueued) {
>+    if (!DecrementSynchSem(this, s, s->waitp->timeout)) {
>+      // After a timeout, we go into a spin loop until we remove ourselves
>+      // from the queue, or someone else removes us.  We can't be sure to be
>+      // able to remove ourselves in a single lock acquisition because this
>+      // mutex may be held, and the holder has the right to read the centre
>+      // of the waiter queue without holding the spinlock.
>+      this->TryRemove(s);
>+      int c = 0;
>+      while (s->next != nullptr) {
>+        c = Delay(c, GENTLE);
>+        this->TryRemove(s);
>+      }
>+      if (kDebugMode) {
>+        // This ensures that we test the case that TryRemove() is called when s
>+        // is not on the queue.
>+        this->TryRemove(s);
>+      }
>+      s->waitp->timeout = KernelTimeout::Never();      // timeout is satisfied
>+      s->waitp->cond = nullptr;  // condition no longer relevant for wakeups
>+    }
>+  }
>+  ABSL_RAW_CHECK(s->waitp != nullptr || s->suppress_fatal_errors,
>+                 "detected illegal recursion in Mutex code");
>+  s->waitp = nullptr;
>+}
>+
>+// Wake thread w, and return the next thread in the list.
>+PerThreadSynch *Mutex::Wakeup(PerThreadSynch *w) {
>+  PerThreadSynch *next = w->next;
>+  w->next = nullptr;
>+  w->state.store(PerThreadSynch::kAvailable, std::memory_order_release);
>+  IncrementSynchSem(this, w);
>+
>+  return next;
>+}
>+
>+static GraphId GetGraphIdLocked(Mutex *mu)
>+    EXCLUSIVE_LOCKS_REQUIRED(deadlock_graph_mu) {
>+  if (!deadlock_graph) {  // (re)create the deadlock graph.
>+    deadlock_graph =
>+        new (base_internal::LowLevelAlloc::Alloc(sizeof(*deadlock_graph)))
>+            GraphCycles;
>+  }
>+  return deadlock_graph->GetId(mu);
>+}
>+
>+static GraphId GetGraphId(Mutex *mu) LOCKS_EXCLUDED(deadlock_graph_mu) {
>+  deadlock_graph_mu.Lock();
>+  GraphId id = GetGraphIdLocked(mu);
>+  deadlock_graph_mu.Unlock();
>+  return id;
>+}
>+
>+// Record a lock acquisition.  This is used in debug mode for deadlock
>+// detection.  The held_locks pointer points to the relevant data
>+// structure for each case.
>+static void LockEnter(Mutex* mu, GraphId id, SynchLocksHeld *held_locks) {
>+  int n = held_locks->n;
>+  int i = 0;
>+  while (i != n && held_locks->locks[i].id != id) {
>+    i++;
>+  }
>+  if (i == n) {
>+    if (n == ABSL_ARRAYSIZE(held_locks->locks)) {
>+      held_locks->overflow = true;  // lost some data
>+    } else {                        // we have room for lock
>+      held_locks->locks[i].mu = mu;
>+      held_locks->locks[i].count = 1;
>+      held_locks->locks[i].id = id;
>+      held_locks->n = n + 1;
>+    }
>+  } else {
>+    held_locks->locks[i].count++;
>+  }
>+}
>+
>+// Record a lock release.  Each call to LockEnter(mu, id, x) should be
>+// eventually followed by a call to LockLeave(mu, id, x) by the same thread.
>+// It does not process the event if is not needed when deadlock detection is
>+// disabled.
>+static void LockLeave(Mutex* mu, GraphId id, SynchLocksHeld *held_locks) {
>+  int n = held_locks->n;
>+  int i = 0;
>+  while (i != n && held_locks->locks[i].id != id) {
>+    i++;
>+  }
>+  if (i == n) {
>+    if (!held_locks->overflow) {
>+      // The deadlock id may have been reassigned after ForgetDeadlockInfo,
>+      // but in that case mu should still be present.
>+      i = 0;
>+      while (i != n && held_locks->locks[i].mu != mu) {
>+        i++;
>+      }
>+      if (i == n) {  // mu missing means releasing unheld lock
>+        SynchEvent *mu_events = GetSynchEvent(mu);
>+        ABSL_RAW_LOG(FATAL,
>+                     "thread releasing lock it does not hold: %p %s; "
>+                     ,
>+                     static_cast<void *>(mu),
>+                     mu_events == nullptr ? "" : mu_events->name);
>+      }
>+    }
>+  } else if (held_locks->locks[i].count == 1) {
>+    held_locks->n = n - 1;
>+    held_locks->locks[i] = held_locks->locks[n - 1];
>+    held_locks->locks[n - 1].id = InvalidGraphId();
>+    held_locks->locks[n - 1].mu =
>+        nullptr;  // clear mu to please the leak detector.
>+  } else {
>+    assert(held_locks->locks[i].count > 0);
>+    held_locks->locks[i].count--;
>+  }
>+}
>+
>+// Call LockEnter() if in debug mode and deadlock detection is enabled.
>+static inline void DebugOnlyLockEnter(Mutex *mu) {
>+  if (kDebugMode) {
>+    if (synch_deadlock_detection.load(std::memory_order_acquire) !=
>+        OnDeadlockCycle::kIgnore) {
>+      LockEnter(mu, GetGraphId(mu), Synch_GetAllLocks());
>+    }
>+  }
>+}
>+
>+// Call LockEnter() if in debug mode and deadlock detection is enabled.
>+static inline void DebugOnlyLockEnter(Mutex *mu, GraphId id) {
>+  if (kDebugMode) {
>+    if (synch_deadlock_detection.load(std::memory_order_acquire) !=
>+        OnDeadlockCycle::kIgnore) {
>+      LockEnter(mu, id, Synch_GetAllLocks());
>+    }
>+  }
>+}
>+
>+// Call LockLeave() if in debug mode and deadlock detection is enabled.
>+static inline void DebugOnlyLockLeave(Mutex *mu) {
>+  if (kDebugMode) {
>+    if (synch_deadlock_detection.load(std::memory_order_acquire) !=
>+        OnDeadlockCycle::kIgnore) {
>+      LockLeave(mu, GetGraphId(mu), Synch_GetAllLocks());
>+    }
>+  }
>+}
>+
>+static char *StackString(void **pcs, int n, char *buf, int maxlen,
>+                         bool symbolize) {
>+  static const int kSymLen = 200;
>+  char sym[kSymLen];
>+  int len = 0;
>+  for (int i = 0; i != n; i++) {
>+    if (symbolize) {
>+      if (!symbolizer(pcs[i], sym, kSymLen)) {
>+        sym[0] = '\0';
>+      }
>+      snprintf(buf + len, maxlen - len, "%s\t@ %p %s\n",
>+               (i == 0 ? "\n" : ""),
>+               pcs[i], sym);
>+    } else {
>+      snprintf(buf + len, maxlen - len, " %p", pcs[i]);
>+    }
>+    len += strlen(&buf[len]);
>+  }
>+  return buf;
>+}
>+
>+static char *CurrentStackString(char *buf, int maxlen, bool symbolize) {
>+  void *pcs[40];
>+  return StackString(pcs, absl::GetStackTrace(pcs, ABSL_ARRAYSIZE(pcs), 2), buf,
>+                     maxlen, symbolize);
>+}
>+
>+namespace {
>+enum { kMaxDeadlockPathLen = 10 };  // maximum length of a deadlock cycle;
>+                                    // a path this long would be remarkable
>+// Buffers required to report a deadlock.
>+// We do not allocate them on stack to avoid large stack frame.
>+struct DeadlockReportBuffers {
>+  char buf[6100];
>+  GraphId path[kMaxDeadlockPathLen];
>+};
>+
>+struct ScopedDeadlockReportBuffers {
>+  ScopedDeadlockReportBuffers() {
>+    b = reinterpret_cast<DeadlockReportBuffers *>(
>+        base_internal::LowLevelAlloc::Alloc(sizeof(*b)));
>+  }
>+  ~ScopedDeadlockReportBuffers() { base_internal::LowLevelAlloc::Free(b); }
>+  DeadlockReportBuffers *b;
>+};
>+
>+// Helper to pass to GraphCycles::UpdateStackTrace.
>+int GetStack(void** stack, int max_depth) {
>+  return absl::GetStackTrace(stack, max_depth, 3);
>+}
>+}  // anonymous namespace
>+
>+// Called in debug mode when a thread is about to acquire a lock in a way that
>+// may block.
>+static GraphId DeadlockCheck(Mutex *mu) {
>+  if (synch_deadlock_detection.load(std::memory_order_acquire) ==
>+      OnDeadlockCycle::kIgnore) {
>+    return InvalidGraphId();
>+  }
>+
>+  SynchLocksHeld *all_locks = Synch_GetAllLocks();
>+
>+  absl::base_internal::SpinLockHolder lock(&deadlock_graph_mu);
>+  const GraphId mu_id = GetGraphIdLocked(mu);
>+
>+  if (all_locks->n == 0) {
>+    // There are no other locks held. Return now so that we don't need to
>+    // call GetSynchEvent(). This way we do not record the stack trace
>+    // for this Mutex. It's ok, since if this Mutex is involved in a deadlock,
>+    // it can't always be the first lock acquired by a thread.
>+    return mu_id;
>+  }
>+
>+  // We prefer to keep stack traces that show a thread holding and acquiring
>+  // as many locks as possible.  This increases the chances that a given edge
>+  // in the acquires-before graph will be represented in the stack traces
>+  // recorded for the locks.
>+  deadlock_graph->UpdateStackTrace(mu_id, all_locks->n + 1, GetStack);
>+
>+  // For each other mutex already held by this thread:
>+  for (int i = 0; i != all_locks->n; i++) {
>+    const GraphId other_node_id = all_locks->locks[i].id;
>+    const Mutex *other =
>+        static_cast<const Mutex *>(deadlock_graph->Ptr(other_node_id));
>+    if (other == nullptr) {
>+      // Ignore stale lock
>+      continue;
>+    }
>+
>+    // Add the acquired-before edge to the graph.
>+    if (!deadlock_graph->InsertEdge(other_node_id, mu_id)) {
>+      ScopedDeadlockReportBuffers scoped_buffers;
>+      DeadlockReportBuffers *b = scoped_buffers.b;
>+      static int number_of_reported_deadlocks = 0;
>+      number_of_reported_deadlocks++;
>+      // Symbolize only 2 first deadlock report to avoid huge slowdowns.
>+      bool symbolize = number_of_reported_deadlocks <= 2;
>+      ABSL_RAW_LOG(ERROR, "Potential Mutex deadlock: %s",
>+                   CurrentStackString(b->buf, sizeof (b->buf), symbolize));
>+      int len = 0;
>+      for (int j = 0; j != all_locks->n; j++) {
>+        void* pr = deadlock_graph->Ptr(all_locks->locks[j].id);
>+        if (pr != nullptr) {
>+          snprintf(b->buf + len, sizeof (b->buf) - len, " %p", pr);
>+          len += static_cast<int>(strlen(&b->buf[len]));
>+        }
>+      }
>+      ABSL_RAW_LOG(ERROR, "Acquiring %p    Mutexes held: %s",
>+                   static_cast<void *>(mu), b->buf);
>+      ABSL_RAW_LOG(ERROR, "Cycle: ");
>+      int path_len = deadlock_graph->FindPath(
>+          mu_id, other_node_id, ABSL_ARRAYSIZE(b->path), b->path);
>+      for (int j = 0; j != path_len; j++) {
>+        GraphId id = b->path[j];
>+        Mutex *path_mu = static_cast<Mutex *>(deadlock_graph->Ptr(id));
>+        if (path_mu == nullptr) continue;
>+        void** stack;
>+        int depth = deadlock_graph->GetStackTrace(id, &stack);
>+        snprintf(b->buf, sizeof(b->buf),
>+                 "mutex@%p stack: ", static_cast<void *>(path_mu));
>+        StackString(stack, depth, b->buf + strlen(b->buf),
>+                    static_cast<int>(sizeof(b->buf) - strlen(b->buf)),
>+                    symbolize);
>+        ABSL_RAW_LOG(ERROR, "%s", b->buf);
>+      }
>+      if (synch_deadlock_detection.load(std::memory_order_acquire) ==
>+          OnDeadlockCycle::kAbort) {
>+        deadlock_graph_mu.Unlock();  // avoid deadlock in fatal sighandler
>+        ABSL_RAW_LOG(FATAL, "dying due to potential deadlock");
>+        return mu_id;
>+      }
>+      break;   // report at most one potential deadlock per acquisition
>+    }
>+  }
>+
>+  return mu_id;
>+}
>+
>+// Invoke DeadlockCheck() iff we're in debug mode and
>+// deadlock checking has been enabled.
>+static inline GraphId DebugOnlyDeadlockCheck(Mutex *mu) {
>+  if (kDebugMode && synch_deadlock_detection.load(std::memory_order_acquire) !=
>+                        OnDeadlockCycle::kIgnore) {
>+    return DeadlockCheck(mu);
>+  } else {
>+    return InvalidGraphId();
>+  }
>+}
>+
>+void Mutex::ForgetDeadlockInfo() {
>+  if (kDebugMode && synch_deadlock_detection.load(std::memory_order_acquire) !=
>+                        OnDeadlockCycle::kIgnore) {
>+    deadlock_graph_mu.Lock();
>+    if (deadlock_graph != nullptr) {
>+      deadlock_graph->RemoveNode(this);
>+    }
>+    deadlock_graph_mu.Unlock();
>+  }
>+}
>+
>+void Mutex::AssertNotHeld() const {
>+  // We have the data to allow this check only if in debug mode and deadlock
>+  // detection is enabled.
>+  if (kDebugMode &&
>+      (mu_.load(std::memory_order_relaxed) & (kMuWriter | kMuReader)) != 0 &&
>+      synch_deadlock_detection.load(std::memory_order_acquire) !=
>+          OnDeadlockCycle::kIgnore) {
>+    GraphId id = GetGraphId(const_cast<Mutex *>(this));
>+    SynchLocksHeld *locks = Synch_GetAllLocks();
>+    for (int i = 0; i != locks->n; i++) {
>+      if (locks->locks[i].id == id) {
>+        SynchEvent *mu_events = GetSynchEvent(this);
>+        ABSL_RAW_LOG(FATAL, "thread should not hold mutex %p %s",
>+                     static_cast<const void *>(this),
>+                     (mu_events == nullptr ? "" : mu_events->name));
>+      }
>+    }
>+  }
>+}
>+
>+// Attempt to acquire *mu, and return whether successful.  The implementation
>+// may spin for a short while if the lock cannot be acquired immediately.
>+static bool TryAcquireWithSpinning(std::atomic<intptr_t>* mu) {
>+  int c = mutex_globals.spinloop_iterations;
>+  int result = -1;  // result of operation:  0=false, 1=true, -1=unknown
>+
>+  do {  // do/while somewhat faster on AMD
>+    intptr_t v = mu->load(std::memory_order_relaxed);
>+    if ((v & (kMuReader|kMuEvent)) != 0) {  // a reader or tracing -> give up
>+      result = 0;
>+    } else if (((v & kMuWriter) == 0) &&  // no holder -> try to acquire
>+               mu->compare_exchange_strong(v, kMuWriter | v,
>+                                           std::memory_order_acquire,
>+                                           std::memory_order_relaxed)) {
>+      result = 1;
>+    }
>+  } while (result == -1 && --c > 0);
>+  return result == 1;
>+}
>+
>+ABSL_XRAY_LOG_ARGS(1) void Mutex::Lock() {
>+  ABSL_TSAN_MUTEX_PRE_LOCK(this, 0);
>+  GraphId id = DebugOnlyDeadlockCheck(this);
>+  intptr_t v = mu_.load(std::memory_order_relaxed);
>+  // try fast acquire, then spin loop
>+  if ((v & (kMuWriter | kMuReader | kMuEvent)) != 0 ||
>+      !mu_.compare_exchange_strong(v, kMuWriter | v,
>+                                   std::memory_order_acquire,
>+                                   std::memory_order_relaxed)) {
>+    // try spin acquire, then slow loop
>+    if (!TryAcquireWithSpinning(&this->mu_)) {
>+      this->LockSlow(kExclusive, nullptr, 0);
>+    }
>+  }
>+  DebugOnlyLockEnter(this, id);
>+  ABSL_TSAN_MUTEX_POST_LOCK(this, 0, 0);
>+}
>+
>+ABSL_XRAY_LOG_ARGS(1) void Mutex::ReaderLock() {
>+  ABSL_TSAN_MUTEX_PRE_LOCK(this, __tsan_mutex_read_lock);
>+  GraphId id = DebugOnlyDeadlockCheck(this);
>+  intptr_t v = mu_.load(std::memory_order_relaxed);
>+  // try fast acquire, then slow loop
>+  if ((v & (kMuWriter | kMuWait | kMuEvent)) != 0 ||
>+      !mu_.compare_exchange_strong(v, (kMuReader | v) + kMuOne,
>+                                   std::memory_order_acquire,
>+                                   std::memory_order_relaxed)) {
>+    this->LockSlow(kShared, nullptr, 0);
>+  }
>+  DebugOnlyLockEnter(this, id);
>+  ABSL_TSAN_MUTEX_POST_LOCK(this, __tsan_mutex_read_lock, 0);
>+}
>+
>+void Mutex::LockWhen(const Condition &cond) {
>+  ABSL_TSAN_MUTEX_PRE_LOCK(this, 0);
>+  GraphId id = DebugOnlyDeadlockCheck(this);
>+  this->LockSlow(kExclusive, &cond, 0);
>+  DebugOnlyLockEnter(this, id);
>+  ABSL_TSAN_MUTEX_POST_LOCK(this, 0, 0);
>+}
>+
>+bool Mutex::LockWhenWithTimeout(const Condition &cond, absl::Duration timeout) {
>+  return LockWhenWithDeadline(cond, DeadlineFromTimeout(timeout));
>+}
>+
>+bool Mutex::LockWhenWithDeadline(const Condition &cond, absl::Time deadline) {
>+  ABSL_TSAN_MUTEX_PRE_LOCK(this, 0);
>+  GraphId id = DebugOnlyDeadlockCheck(this);
>+  bool res = LockSlowWithDeadline(kExclusive, &cond,
>+                                  KernelTimeout(deadline), 0);
>+  DebugOnlyLockEnter(this, id);
>+  ABSL_TSAN_MUTEX_POST_LOCK(this, 0, 0);
>+  return res;
>+}
>+
>+void Mutex::ReaderLockWhen(const Condition &cond) {
>+  ABSL_TSAN_MUTEX_PRE_LOCK(this, __tsan_mutex_read_lock);
>+  GraphId id = DebugOnlyDeadlockCheck(this);
>+  this->LockSlow(kShared, &cond, 0);
>+  DebugOnlyLockEnter(this, id);
>+  ABSL_TSAN_MUTEX_POST_LOCK(this, __tsan_mutex_read_lock, 0);
>+}
>+
>+bool Mutex::ReaderLockWhenWithTimeout(const Condition &cond,
>+                                      absl::Duration timeout) {
>+  return ReaderLockWhenWithDeadline(cond, DeadlineFromTimeout(timeout));
>+}
>+
>+bool Mutex::ReaderLockWhenWithDeadline(const Condition &cond,
>+                                       absl::Time deadline) {
>+  ABSL_TSAN_MUTEX_PRE_LOCK(this, __tsan_mutex_read_lock);
>+  GraphId id = DebugOnlyDeadlockCheck(this);
>+  bool res = LockSlowWithDeadline(kShared, &cond, KernelTimeout(deadline), 0);
>+  DebugOnlyLockEnter(this, id);
>+  ABSL_TSAN_MUTEX_POST_LOCK(this, __tsan_mutex_read_lock, 0);
>+  return res;
>+}
>+
>+void Mutex::Await(const Condition &cond) {
>+  if (cond.Eval()) {    // condition already true; nothing to do
>+    if (kDebugMode) {
>+      this->AssertReaderHeld();
>+    }
>+  } else {              // normal case
>+    ABSL_RAW_CHECK(this->AwaitCommon(cond, KernelTimeout::Never()),
>+                   "condition untrue on return from Await");
>+  }
>+}
>+
>+bool Mutex::AwaitWithTimeout(const Condition &cond, absl::Duration timeout) {
>+  return AwaitWithDeadline(cond, DeadlineFromTimeout(timeout));
>+}
>+
>+bool Mutex::AwaitWithDeadline(const Condition &cond, absl::Time deadline) {
>+  if (cond.Eval()) {      // condition already true; nothing to do
>+    if (kDebugMode) {
>+      this->AssertReaderHeld();
>+    }
>+    return true;
>+  }
>+
>+  KernelTimeout t{deadline};
>+  bool res = this->AwaitCommon(cond, t);
>+  ABSL_RAW_CHECK(res || t.has_timeout(),
>+                 "condition untrue on return from Await");
>+  return res;
>+}
>+
>+bool Mutex::AwaitCommon(const Condition &cond, KernelTimeout t) {
>+  this->AssertReaderHeld();
>+  MuHow how =
>+      (mu_.load(std::memory_order_relaxed) & kMuWriter) ? kExclusive : kShared;
>+  ABSL_TSAN_MUTEX_PRE_UNLOCK(this, TsanFlags(how));
>+  SynchWaitParams waitp(
>+      how, &cond, t, nullptr /*no cvmu*/, Synch_GetPerThreadAnnotated(this),
>+      nullptr /*no cv_word*/);
>+  int flags = kMuHasBlocked;
>+  if (!Condition::GuaranteedEqual(&cond, nullptr)) {
>+    flags |= kMuIsCond;
>+  }
>+  this->UnlockSlow(&waitp);
>+  this->Block(waitp.thread);
>+  ABSL_TSAN_MUTEX_POST_UNLOCK(this, TsanFlags(how));
>+  ABSL_TSAN_MUTEX_PRE_LOCK(this, TsanFlags(how));
>+  this->LockSlowLoop(&waitp, flags);
>+  bool res = waitp.cond != nullptr ||  // => cond known true from LockSlowLoop
>+             cond.Eval();
>+  ABSL_TSAN_MUTEX_POST_LOCK(this, TsanFlags(how), 0);
>+  return res;
>+}
>+
>+ABSL_XRAY_LOG_ARGS(1) bool Mutex::TryLock() {
>+  ABSL_TSAN_MUTEX_PRE_LOCK(this, __tsan_mutex_try_lock);
>+  intptr_t v = mu_.load(std::memory_order_relaxed);
>+  if ((v & (kMuWriter | kMuReader | kMuEvent)) == 0 &&  // try fast acquire
>+      mu_.compare_exchange_strong(v, kMuWriter | v,
>+                                  std::memory_order_acquire,
>+                                  std::memory_order_relaxed)) {
>+    DebugOnlyLockEnter(this);
>+    ABSL_TSAN_MUTEX_POST_LOCK(this, __tsan_mutex_try_lock, 0);
>+    return true;
>+  }
>+  if ((v & kMuEvent) != 0) {              // we're recording events
>+    if ((v & kExclusive->slow_need_zero) == 0 &&  // try fast acquire
>+        mu_.compare_exchange_strong(
>+            v, (kExclusive->fast_or | v) + kExclusive->fast_add,
>+            std::memory_order_acquire, std::memory_order_relaxed)) {
>+      DebugOnlyLockEnter(this);
>+      PostSynchEvent(this, SYNCH_EV_TRYLOCK_SUCCESS);
>+      ABSL_TSAN_MUTEX_POST_LOCK(this, __tsan_mutex_try_lock, 0);
>+      return true;
>+    } else {
>+      PostSynchEvent(this, SYNCH_EV_TRYLOCK_FAILED);
>+    }
>+  }
>+  ABSL_TSAN_MUTEX_POST_LOCK(
>+      this, __tsan_mutex_try_lock | __tsan_mutex_try_lock_failed, 0);
>+  return false;
>+}
>+
>+ABSL_XRAY_LOG_ARGS(1) bool Mutex::ReaderTryLock() {
>+  ABSL_TSAN_MUTEX_PRE_LOCK(this,
>+                           __tsan_mutex_read_lock | __tsan_mutex_try_lock);
>+  intptr_t v = mu_.load(std::memory_order_relaxed);
>+  // The while-loops (here and below) iterate only if the mutex word keeps
>+  // changing (typically because the reader count changes) under the CAS.  We
>+  // limit the number of attempts to avoid having to think about livelock.
>+  int loop_limit = 5;
>+  while ((v & (kMuWriter|kMuWait|kMuEvent)) == 0 && loop_limit != 0) {
>+    if (mu_.compare_exchange_strong(v, (kMuReader | v) + kMuOne,
>+                                    std::memory_order_acquire,
>+                                    std::memory_order_relaxed)) {
>+      DebugOnlyLockEnter(this);
>+      ABSL_TSAN_MUTEX_POST_LOCK(
>+          this, __tsan_mutex_read_lock | __tsan_mutex_try_lock, 0);
>+      return true;
>+    }
>+    loop_limit--;
>+    v = mu_.load(std::memory_order_relaxed);
>+  }
>+  if ((v & kMuEvent) != 0) {   // we're recording events
>+    loop_limit = 5;
>+    while ((v & kShared->slow_need_zero) == 0 && loop_limit != 0) {
>+      if (mu_.compare_exchange_strong(v, (kMuReader | v) + kMuOne,
>+                                      std::memory_order_acquire,
>+                                      std::memory_order_relaxed)) {
>+        DebugOnlyLockEnter(this);
>+        PostSynchEvent(this, SYNCH_EV_READERTRYLOCK_SUCCESS);
>+        ABSL_TSAN_MUTEX_POST_LOCK(
>+            this, __tsan_mutex_read_lock | __tsan_mutex_try_lock, 0);
>+        return true;
>+      }
>+      loop_limit--;
>+      v = mu_.load(std::memory_order_relaxed);
>+    }
>+    if ((v & kMuEvent) != 0) {
>+      PostSynchEvent(this, SYNCH_EV_READERTRYLOCK_FAILED);
>+    }
>+  }
>+  ABSL_TSAN_MUTEX_POST_LOCK(this,
>+                            __tsan_mutex_read_lock | __tsan_mutex_try_lock |
>+                                __tsan_mutex_try_lock_failed,
>+                            0);
>+  return false;
>+}
>+
>+ABSL_XRAY_LOG_ARGS(1) void Mutex::Unlock() {
>+  ABSL_TSAN_MUTEX_PRE_UNLOCK(this, 0);
>+  DebugOnlyLockLeave(this);
>+  intptr_t v = mu_.load(std::memory_order_relaxed);
>+
>+  if (kDebugMode && ((v & (kMuWriter | kMuReader)) != kMuWriter)) {
>+    ABSL_RAW_LOG(FATAL, "Mutex unlocked when destroyed or not locked: v=0x%x",
>+                 static_cast<unsigned>(v));
>+  }
>+
>+  // should_try_cas is whether we'll try a compare-and-swap immediately.
>+  // NOTE: optimized out when kDebugMode is false.
>+  bool should_try_cas = ((v & (kMuEvent | kMuWriter)) == kMuWriter &&
>+                          (v & (kMuWait | kMuDesig)) != kMuWait);
>+  // But, we can use an alternate computation of it, that compilers
>+  // currently don't find on their own.  When that changes, this function
>+  // can be simplified.
>+  intptr_t x = (v ^ (kMuWriter | kMuWait)) & (kMuWriter | kMuEvent);
>+  intptr_t y = (v ^ (kMuWriter | kMuWait)) & (kMuWait | kMuDesig);
>+  // Claim: "x == 0 && y > 0" is equal to should_try_cas.
>+  // Also, because kMuWriter and kMuEvent exceed kMuDesig and kMuWait,
>+  // all possible non-zero values for x exceed all possible values for y.
>+  // Therefore, (x == 0 && y > 0) == (x < y).
>+  if (kDebugMode && should_try_cas != (x < y)) {
>+    // We would usually use PRIdPTR here, but is not correctly implemented
>+    // within the android toolchain.
>+    ABSL_RAW_LOG(FATAL, "internal logic error %llx %llx %llx\n",
>+                 static_cast<long long>(v), static_cast<long long>(x),
>+                 static_cast<long long>(y));
>+  }
>+  if (x < y &&
>+      mu_.compare_exchange_strong(v, v & ~(kMuWrWait | kMuWriter),
>+                                  std::memory_order_release,
>+                                  std::memory_order_relaxed)) {
>+    // fast writer release (writer with no waiters or with designated waker)
>+  } else {
>+    this->UnlockSlow(nullptr /*no waitp*/);  // take slow path
>+  }
>+  ABSL_TSAN_MUTEX_POST_UNLOCK(this, 0);
>+}
>+
>+// Requires v to represent a reader-locked state.
>+static bool ExactlyOneReader(intptr_t v) {
>+  assert((v & (kMuWriter|kMuReader)) == kMuReader);
>+  assert((v & kMuHigh) != 0);
>+  // The more straightforward "(v & kMuHigh) == kMuOne" also works, but
>+  // on some architectures the following generates slightly smaller code.
>+  // It may be faster too.
>+  constexpr intptr_t kMuMultipleWaitersMask = kMuHigh ^ kMuOne;
>+  return (v & kMuMultipleWaitersMask) == 0;
>+}
>+
>+ABSL_XRAY_LOG_ARGS(1) void Mutex::ReaderUnlock() {
>+  ABSL_TSAN_MUTEX_PRE_UNLOCK(this, __tsan_mutex_read_lock);
>+  DebugOnlyLockLeave(this);
>+  intptr_t v = mu_.load(std::memory_order_relaxed);
>+  assert((v & (kMuWriter|kMuReader)) == kMuReader);
>+  if ((v & (kMuReader|kMuWait|kMuEvent)) == kMuReader) {
>+    // fast reader release (reader with no waiters)
>+    intptr_t clear = ExactlyOneReader(v) ? kMuReader|kMuOne : kMuOne;
>+    if (mu_.compare_exchange_strong(v, v - clear,
>+                                    std::memory_order_release,
>+                                    std::memory_order_relaxed)) {
>+      ABSL_TSAN_MUTEX_POST_UNLOCK(this, __tsan_mutex_read_lock);
>+      return;
>+    }
>+  }
>+  this->UnlockSlow(nullptr /*no waitp*/);  // take slow path
>+  ABSL_TSAN_MUTEX_POST_UNLOCK(this, __tsan_mutex_read_lock);
>+}
>+
>+// The zap_desig_waker bitmask is used to clear the designated waker flag in
>+// the mutex if this thread has blocked, and therefore may be the designated
>+// waker.
>+static const intptr_t zap_desig_waker[] = {
>+    ~static_cast<intptr_t>(0),  // not blocked
>+    ~static_cast<intptr_t>(
>+        kMuDesig)  // blocked; turn off the designated waker bit
>+};
>+
>+// The ignore_waiting_writers bitmask is used to ignore the existence
>+// of waiting writers if a reader that has already blocked once
>+// wakes up.
>+static const intptr_t ignore_waiting_writers[] = {
>+    ~static_cast<intptr_t>(0),  // not blocked
>+    ~static_cast<intptr_t>(
>+        kMuWrWait)  // blocked; pretend there are no waiting writers
>+};
>+
>+// Internal version of LockWhen().  See LockSlowWithDeadline()
>+void Mutex::LockSlow(MuHow how, const Condition *cond, int flags) {
>+  ABSL_RAW_CHECK(
>+      this->LockSlowWithDeadline(how, cond, KernelTimeout::Never(), flags),
>+      "condition untrue on return from LockSlow");
>+}
>+
>+// Compute cond->Eval() and tell race detectors that we do it under mutex mu.
>+static inline bool EvalConditionAnnotated(const Condition *cond, Mutex *mu,
>+                                          bool locking, Mutex::MuHow how) {
>+  // Delicate annotation dance.
>+  // We are currently inside of read/write lock/unlock operation.
>+  // All memory accesses are ignored inside of mutex operations + for unlock
>+  // operation tsan considers that we've already released the mutex.
>+  bool res = false;
>+  if (locking) {
>+    // For lock we pretend that we have finished the operation,
>+    // evaluate the predicate, then unlock the mutex and start locking it again
>+    // to match the annotation at the end of outer lock operation.
>+    // Note: we can't simply do POST_LOCK, Eval, PRE_LOCK, because then tsan
>+    // will think the lock acquisition is recursive which will trigger
>+    // deadlock detector.
>+    ABSL_TSAN_MUTEX_POST_LOCK(mu, TsanFlags(how), 0);
>+    res = cond->Eval();
>+    ABSL_TSAN_MUTEX_PRE_UNLOCK(mu, TsanFlags(how));
>+    ABSL_TSAN_MUTEX_POST_UNLOCK(mu, TsanFlags(how));
>+    ABSL_TSAN_MUTEX_PRE_LOCK(mu, TsanFlags(how));
>+  } else {
>+    // Similarly, for unlock we pretend that we have unlocked the mutex,
>+    // lock the mutex, evaluate the predicate, and start unlocking it again
>+    // to match the annotation at the end of outer unlock operation.
>+    ABSL_TSAN_MUTEX_POST_UNLOCK(mu, TsanFlags(how));
>+    ABSL_TSAN_MUTEX_PRE_LOCK(mu, TsanFlags(how));
>+    ABSL_TSAN_MUTEX_POST_LOCK(mu, TsanFlags(how), 0);
>+    res = cond->Eval();
>+    ABSL_TSAN_MUTEX_PRE_UNLOCK(mu, TsanFlags(how));
>+  }
>+  // Prevent unused param warnings in non-TSAN builds.
>+  static_cast<void>(mu);
>+  static_cast<void>(how);
>+  return res;
>+}
>+
>+// Compute cond->Eval() hiding it from race detectors.
>+// We are hiding it because inside of UnlockSlow we can evaluate a predicate
>+// that was just added by a concurrent Lock operation; Lock adds the predicate
>+// to the internal Mutex list without actually acquiring the Mutex
>+// (it only acquires the internal spinlock, which is rightfully invisible for
>+// tsan). As the result there is no tsan-visible synchronization between the
>+// addition and this thread. So if we would enable race detection here,
>+// it would race with the predicate initialization.
>+static inline bool EvalConditionIgnored(Mutex *mu, const Condition *cond) {
>+  // Memory accesses are already ignored inside of lock/unlock operations,
>+  // but synchronization operations are also ignored. When we evaluate the
>+  // predicate we must ignore only memory accesses but not synchronization,
>+  // because missed synchronization can lead to false reports later.
>+  // So we "divert" (which un-ignores both memory accesses and synchronization)
>+  // and then separately turn on ignores of memory accesses.
>+  ABSL_TSAN_MUTEX_PRE_DIVERT(mu, 0);
>+  ABSL_ANNOTATE_IGNORE_READS_AND_WRITES_BEGIN();
>+  bool res = cond->Eval();
>+  ABSL_ANNOTATE_IGNORE_READS_AND_WRITES_END();
>+  ABSL_TSAN_MUTEX_POST_DIVERT(mu, 0);
>+  static_cast<void>(mu);  // Prevent unused param warning in non-TSAN builds.
>+  return res;
>+}
>+
>+// Internal equivalent of *LockWhenWithDeadline(), where
>+//   "t" represents the absolute timeout; !t.has_timeout() means "forever".
>+//   "how" is "kShared" (for ReaderLockWhen) or "kExclusive" (for LockWhen)
>+// In flags, bits are ored together:
>+// - kMuHasBlocked indicates that the client has already blocked on the call so
>+//   the designated waker bit must be cleared and waiting writers should not
>+//   obstruct this call
>+// - kMuIsCond indicates that this is a conditional acquire (condition variable,
>+//   Await,  LockWhen) so contention profiling should be suppressed.
>+bool Mutex::LockSlowWithDeadline(MuHow how, const Condition *cond,
>+                                 KernelTimeout t, int flags) {
>+  intptr_t v = mu_.load(std::memory_order_relaxed);
>+  bool unlock = false;
>+  if ((v & how->fast_need_zero) == 0 &&  // try fast acquire
>+      mu_.compare_exchange_strong(
>+          v, (how->fast_or | (v & zap_desig_waker[flags & kMuHasBlocked])) +
>+                 how->fast_add,
>+          std::memory_order_acquire, std::memory_order_relaxed)) {
>+    if (cond == nullptr || EvalConditionAnnotated(cond, this, true, how)) {
>+      return true;
>+    }
>+    unlock = true;
>+  }
>+  SynchWaitParams waitp(
>+      how, cond, t, nullptr /*no cvmu*/, Synch_GetPerThreadAnnotated(this),
>+      nullptr /*no cv_word*/);
>+  if (!Condition::GuaranteedEqual(cond, nullptr)) {
>+    flags |= kMuIsCond;
>+  }
>+  if (unlock) {
>+    this->UnlockSlow(&waitp);
>+    this->Block(waitp.thread);
>+    flags |= kMuHasBlocked;
>+  }
>+  this->LockSlowLoop(&waitp, flags);
>+  return waitp.cond != nullptr ||  // => cond known true from LockSlowLoop
>+         cond == nullptr || EvalConditionAnnotated(cond, this, true, how);
>+}
>+
>+// RAW_CHECK_FMT() takes a condition, a printf-style format std::string, and
>+// the printf-style argument list.   The format std::string must be a literal.
>+// Arguments after the first are not evaluated unless the condition is true.
>+#define RAW_CHECK_FMT(cond, ...)                                   \
>+  do {                                                             \
>+    if (ABSL_PREDICT_FALSE(!(cond))) {                             \
>+      ABSL_RAW_LOG(FATAL, "Check " #cond " failed: " __VA_ARGS__); \
>+    }                                                              \
>+  } while (0)
>+
>+static void CheckForMutexCorruption(intptr_t v, const char* label) {
>+  // Test for either of two situations that should not occur in v:
>+  //   kMuWriter and kMuReader
>+  //   kMuWrWait and !kMuWait
>+  const intptr_t w = v ^ kMuWait;
>+  // By flipping that bit, we can now test for:
>+  //   kMuWriter and kMuReader in w
>+  //   kMuWrWait and kMuWait in w
>+  // We've chosen these two pairs of values to be so that they will overlap,
>+  // respectively, when the word is left shifted by three.  This allows us to
>+  // save a branch in the common (correct) case of them not being coincident.
>+  static_assert(kMuReader << 3 == kMuWriter, "must match");
>+  static_assert(kMuWait << 3 == kMuWrWait, "must match");
>+  if (ABSL_PREDICT_TRUE((w & (w << 3) & (kMuWriter | kMuWrWait)) == 0)) return;
>+  RAW_CHECK_FMT((v & (kMuWriter | kMuReader)) != (kMuWriter | kMuReader),
>+                "%s: Mutex corrupt: both reader and writer lock held: %p",
>+                label, reinterpret_cast<void *>(v));
>+  RAW_CHECK_FMT((v & (kMuWait | kMuWrWait)) != kMuWrWait,
>+                "%s: Mutex corrupt: waiting writer with no waiters: %p",
>+                label, reinterpret_cast<void *>(v));
>+  assert(false);
>+}
>+
>+void Mutex::LockSlowLoop(SynchWaitParams *waitp, int flags) {
>+  int c = 0;
>+  intptr_t v = mu_.load(std::memory_order_relaxed);
>+  if ((v & kMuEvent) != 0) {
>+    PostSynchEvent(this,
>+         waitp->how == kExclusive?  SYNCH_EV_LOCK: SYNCH_EV_READERLOCK);
>+  }
>+  ABSL_RAW_CHECK(
>+      waitp->thread->waitp == nullptr || waitp->thread->suppress_fatal_errors,
>+      "detected illegal recursion into Mutex code");
>+  for (;;) {
>+    v = mu_.load(std::memory_order_relaxed);
>+    CheckForMutexCorruption(v, "Lock");
>+    if ((v & waitp->how->slow_need_zero) == 0) {
>+      if (mu_.compare_exchange_strong(
>+              v, (waitp->how->fast_or |
>+                  (v & zap_desig_waker[flags & kMuHasBlocked])) +
>+                     waitp->how->fast_add,
>+              std::memory_order_acquire, std::memory_order_relaxed)) {
>+        if (waitp->cond == nullptr ||
>+            EvalConditionAnnotated(waitp->cond, this, true, waitp->how)) {
>+          break;  // we timed out, or condition true, so return
>+        }
>+        this->UnlockSlow(waitp);  // got lock but condition false
>+        this->Block(waitp->thread);
>+        flags |= kMuHasBlocked;
>+        c = 0;
>+      }
>+    } else {                      // need to access waiter list
>+      bool dowait = false;
>+      if ((v & (kMuSpin|kMuWait)) == 0) {   // no waiters
>+        // This thread tries to become the one and only waiter.
>+        PerThreadSynch *new_h = Enqueue(nullptr, waitp, v, flags);
>+        intptr_t nv = (v & zap_desig_waker[flags & kMuHasBlocked] & kMuLow) |
>+                      kMuWait;
>+        ABSL_RAW_CHECK(new_h != nullptr, "Enqueue to empty list failed");
>+        if (waitp->how == kExclusive && (v & kMuReader) != 0) {
>+          nv |= kMuWrWait;
>+        }
>+        if (mu_.compare_exchange_strong(
>+                v, reinterpret_cast<intptr_t>(new_h) | nv,
>+                std::memory_order_release, std::memory_order_relaxed)) {
>+          dowait = true;
>+        } else {            // attempted Enqueue() failed
>+          // zero out the waitp field set by Enqueue()
>+          waitp->thread->waitp = nullptr;
>+        }
>+      } else if ((v & waitp->how->slow_inc_need_zero &
>+                  ignore_waiting_writers[flags & kMuHasBlocked]) == 0) {
>+        // This is a reader that needs to increment the reader count,
>+        // but the count is currently held in the last waiter.
>+        if (mu_.compare_exchange_strong(
>+                v, (v & zap_desig_waker[flags & kMuHasBlocked]) | kMuSpin |
>+                       kMuReader,
>+                std::memory_order_acquire, std::memory_order_relaxed)) {
>+          PerThreadSynch *h = GetPerThreadSynch(v);
>+          h->readers += kMuOne;       // inc reader count in waiter
>+          do {                        // release spinlock
>+            v = mu_.load(std::memory_order_relaxed);
>+          } while (!mu_.compare_exchange_weak(v, (v & ~kMuSpin) | kMuReader,
>+                                              std::memory_order_release,
>+                                              std::memory_order_relaxed));
>+          if (waitp->cond == nullptr ||
>+              EvalConditionAnnotated(waitp->cond, this, true, waitp->how)) {
>+            break;  // we timed out, or condition true, so return
>+          }
>+          this->UnlockSlow(waitp);           // got lock but condition false
>+          this->Block(waitp->thread);
>+          flags |= kMuHasBlocked;
>+          c = 0;
>+        }
>+      } else if ((v & kMuSpin) == 0 &&  // attempt to queue ourselves
>+                 mu_.compare_exchange_strong(
>+                     v, (v & zap_desig_waker[flags & kMuHasBlocked]) | kMuSpin |
>+                            kMuWait,
>+                     std::memory_order_acquire, std::memory_order_relaxed)) {
>+        PerThreadSynch *h = GetPerThreadSynch(v);
>+        PerThreadSynch *new_h = Enqueue(h, waitp, v, flags);
>+        intptr_t wr_wait = 0;
>+        ABSL_RAW_CHECK(new_h != nullptr, "Enqueue to list failed");
>+        if (waitp->how == kExclusive && (v & kMuReader) != 0) {
>+          wr_wait = kMuWrWait;      // give priority to a waiting writer
>+        }
>+        do {                        // release spinlock
>+          v = mu_.load(std::memory_order_relaxed);
>+        } while (!mu_.compare_exchange_weak(
>+            v, (v & (kMuLow & ~kMuSpin)) | kMuWait | wr_wait |
>+            reinterpret_cast<intptr_t>(new_h),
>+            std::memory_order_release, std::memory_order_relaxed));
>+        dowait = true;
>+      }
>+      if (dowait) {
>+        this->Block(waitp->thread);  // wait until removed from list or timeout
>+        flags |= kMuHasBlocked;
>+        c = 0;
>+      }
>+    }
>+    ABSL_RAW_CHECK(
>+        waitp->thread->waitp == nullptr || waitp->thread->suppress_fatal_errors,
>+        "detected illegal recursion into Mutex code");
>+    c = Delay(c, GENTLE);          // delay, then try again
>+  }
>+  ABSL_RAW_CHECK(
>+      waitp->thread->waitp == nullptr || waitp->thread->suppress_fatal_errors,
>+      "detected illegal recursion into Mutex code");
>+  if ((v & kMuEvent) != 0) {
>+    PostSynchEvent(this,
>+                   waitp->how == kExclusive? SYNCH_EV_LOCK_RETURNING :
>+                                      SYNCH_EV_READERLOCK_RETURNING);
>+  }
>+}
>+
>+// Unlock this mutex, which is held by the current thread.
>+// If waitp is non-zero, it must be the wait parameters for the current thread
>+// which holds the lock but is not runnable because its condition is false
>+// or it n the process of blocking on a condition variable; it must requeue
>+// itself on the mutex/condvar to wait for its condition to become true.
>+void Mutex::UnlockSlow(SynchWaitParams *waitp) {
>+  intptr_t v = mu_.load(std::memory_order_relaxed);
>+  this->AssertReaderHeld();
>+  CheckForMutexCorruption(v, "Unlock");
>+  if ((v & kMuEvent) != 0) {
>+    PostSynchEvent(this,
>+                (v & kMuWriter) != 0? SYNCH_EV_UNLOCK: SYNCH_EV_READERUNLOCK);
>+  }
>+  int c = 0;
>+  // the waiter under consideration to wake, or zero
>+  PerThreadSynch *w = nullptr;
>+  // the predecessor to w or zero
>+  PerThreadSynch *pw = nullptr;
>+  // head of the list searched previously, or zero
>+  PerThreadSynch *old_h = nullptr;
>+  // a condition that's known to be false.
>+  const Condition *known_false = nullptr;
>+  PerThreadSynch *wake_list = kPerThreadSynchNull;   // list of threads to wake
>+  intptr_t wr_wait = 0;        // set to kMuWrWait if we wake a reader and a
>+                               // later writer could have acquired the lock
>+                               // (starvation avoidance)
>+  ABSL_RAW_CHECK(waitp == nullptr || waitp->thread->waitp == nullptr ||
>+                     waitp->thread->suppress_fatal_errors,
>+                 "detected illegal recursion into Mutex code");
>+  // This loop finds threads wake_list to wakeup if any, and removes them from
>+  // the list of waiters.  In addition, it places waitp.thread on the queue of
>+  // waiters if waitp is non-zero.
>+  for (;;) {
>+    v = mu_.load(std::memory_order_relaxed);
>+    if ((v & kMuWriter) != 0 && (v & (kMuWait | kMuDesig)) != kMuWait &&
>+        waitp == nullptr) {
>+      // fast writer release (writer with no waiters or with designated waker)
>+      if (mu_.compare_exchange_strong(v, v & ~(kMuWrWait | kMuWriter),
>+                                      std::memory_order_release,
>+                                      std::memory_order_relaxed)) {
>+        return;
>+      }
>+    } else if ((v & (kMuReader | kMuWait)) == kMuReader && waitp == nullptr) {
>+      // fast reader release (reader with no waiters)
>+      intptr_t clear = ExactlyOneReader(v) ? kMuReader | kMuOne : kMuOne;
>+      if (mu_.compare_exchange_strong(v, v - clear,
>+                                      std::memory_order_release,
>+                                      std::memory_order_relaxed)) {
>+        return;
>+      }
>+    } else if ((v & kMuSpin) == 0 &&  // attempt to get spinlock
>+               mu_.compare_exchange_strong(v, v | kMuSpin,
>+                                           std::memory_order_acquire,
>+                                           std::memory_order_relaxed)) {
>+      if ((v & kMuWait) == 0) {       // no one to wake
>+        intptr_t nv;
>+        bool do_enqueue = true;  // always Enqueue() the first time
>+        ABSL_RAW_CHECK(waitp != nullptr,
>+                       "UnlockSlow is confused");  // about to sleep
>+        do {    // must loop to release spinlock as reader count may change
>+          v = mu_.load(std::memory_order_relaxed);
>+          // decrement reader count if there are readers
>+          intptr_t new_readers = (v >= kMuOne)?  v - kMuOne : v;
>+          PerThreadSynch *new_h = nullptr;
>+          if (do_enqueue) {
>+            // If we are enqueuing on a CondVar (waitp->cv_word != nullptr) then
>+            // we must not retry here.  The initial attempt will always have
>+            // succeeded, further attempts would enqueue us against *this due to
>+            // Fer() handling.
>+            do_enqueue = (waitp->cv_word == nullptr);
>+            new_h = Enqueue(nullptr, waitp, new_readers, kMuIsCond);
>+          }
>+          intptr_t clear = kMuWrWait | kMuWriter;  // by default clear write bit
>+          if ((v & kMuWriter) == 0 && ExactlyOneReader(v)) {  // last reader
>+            clear = kMuWrWait | kMuReader;                    // clear read bit
>+          }
>+          nv = (v & kMuLow & ~clear & ~kMuSpin);
>+          if (new_h != nullptr) {
>+            nv |= kMuWait | reinterpret_cast<intptr_t>(new_h);
>+          } else {  // new_h could be nullptr if we queued ourselves on a
>+                    // CondVar
>+            // In that case, we must place the reader count back in the mutex
>+            // word, as Enqueue() did not store it in the new waiter.
>+            nv |= new_readers & kMuHigh;
>+          }
>+          // release spinlock & our lock; retry if reader-count changed
>+          // (writer count cannot change since we hold lock)
>+        } while (!mu_.compare_exchange_weak(v, nv,
>+                                            std::memory_order_release,
>+                                            std::memory_order_relaxed));
>+        break;
>+      }
>+
>+      // There are waiters.
>+      // Set h to the head of the circular waiter list.
>+      PerThreadSynch *h = GetPerThreadSynch(v);
>+      if ((v & kMuReader) != 0 && (h->readers & kMuHigh) > kMuOne) {
>+        // a reader but not the last
>+        h->readers -= kMuOne;  // release our lock
>+        intptr_t nv = v;       // normally just release spinlock
>+        if (waitp != nullptr) {  // but waitp!=nullptr => must queue ourselves
>+          PerThreadSynch *new_h = Enqueue(h, waitp, v, kMuIsCond);
>+          ABSL_RAW_CHECK(new_h != nullptr,
>+                         "waiters disappeared during Enqueue()!");
>+          nv &= kMuLow;
>+          nv |= kMuWait | reinterpret_cast<intptr_t>(new_h);
>+        }
>+        mu_.store(nv, std::memory_order_release);  // release spinlock
>+        // can release with a store because there were waiters
>+        break;
>+      }
>+
>+      // Either we didn't search before, or we marked the queue
>+      // as "maybe_unlocking" and no one else should have changed it.
>+      ABSL_RAW_CHECK(old_h == nullptr || h->maybe_unlocking,
>+                     "Mutex queue changed beneath us");
>+
>+      // The lock is becoming free, and there's a waiter
>+      if (old_h != nullptr &&
>+          !old_h->may_skip) {                  // we used old_h as a terminator
>+        old_h->may_skip = true;                // allow old_h to skip once more
>+        ABSL_RAW_CHECK(old_h->skip == nullptr, "illegal skip from head");
>+        if (h != old_h && MuSameCondition(old_h, old_h->next)) {
>+          old_h->skip = old_h->next;  // old_h not head & can skip to successor
>+        }
>+      }
>+      if (h->next->waitp->how == kExclusive &&
>+          Condition::GuaranteedEqual(h->next->waitp->cond, nullptr)) {
>+        // easy case: writer with no condition; no need to search
>+        pw = h;                       // wake w, the successor of h (=pw)
>+        w = h->next;
>+        w->wake = true;
>+        // We are waking up a writer.  This writer may be racing against
>+        // an already awake reader for the lock.  We want the
>+        // writer to usually win this race,
>+        // because if it doesn't, we can potentially keep taking a reader
>+        // perpetually and writers will starve.  Worse than
>+        // that, this can also starve other readers if kMuWrWait gets set
>+        // later.
>+        wr_wait = kMuWrWait;
>+      } else if (w != nullptr && (w->waitp->how == kExclusive || h == old_h)) {
>+        // we found a waiter w to wake on a previous iteration and either it's
>+        // a writer, or we've searched the entire list so we have all the
>+        // readers.
>+        if (pw == nullptr) {  // if w's predecessor is unknown, it must be h
>+          pw = h;
>+        }
>+      } else {
>+        // At this point we don't know all the waiters to wake, and the first
>+        // waiter has a condition or is a reader.  We avoid searching over
>+        // waiters we've searched on previous iterations by starting at
>+        // old_h if it's set.  If old_h==h, there's no one to wakeup at all.
>+        if (old_h == h) {      // we've searched before, and nothing's new
>+                               // so there's no one to wake.
>+          intptr_t nv = (v & ~(kMuReader|kMuWriter|kMuWrWait));
>+          h->readers = 0;
>+          h->maybe_unlocking = false;   // finished unlocking
>+          if (waitp != nullptr) {       // we must queue ourselves and sleep
>+            PerThreadSynch *new_h = Enqueue(h, waitp, v, kMuIsCond);
>+            nv &= kMuLow;
>+            if (new_h != nullptr) {
>+              nv |= kMuWait | reinterpret_cast<intptr_t>(new_h);
>+            }  // else new_h could be nullptr if we queued ourselves on a
>+               // CondVar
>+          }
>+          // release spinlock & lock
>+          // can release with a store because there were waiters
>+          mu_.store(nv, std::memory_order_release);
>+          break;
>+        }
>+
>+        // set up to walk the list
>+        PerThreadSynch *w_walk;   // current waiter during list walk
>+        PerThreadSynch *pw_walk;  // previous waiter during list walk
>+        if (old_h != nullptr) {  // we've searched up to old_h before
>+          pw_walk = old_h;
>+          w_walk = old_h->next;
>+        } else {            // no prior search, start at beginning
>+          pw_walk =
>+              nullptr;  // h->next's predecessor may change; don't record it
>+          w_walk = h->next;
>+        }
>+
>+        h->may_skip = false;  // ensure we never skip past h in future searches
>+                              // even if other waiters are queued after it.
>+        ABSL_RAW_CHECK(h->skip == nullptr, "illegal skip from head");
>+
>+        h->maybe_unlocking = true;  // we're about to scan the waiter list
>+                                    // without the spinlock held.
>+                                    // Enqueue must be conservative about
>+                                    // priority queuing.
>+
>+        // We must release the spinlock to evaluate the conditions.
>+        mu_.store(v, std::memory_order_release);  // release just spinlock
>+        // can release with a store because there were waiters
>+
>+        // h is the last waiter queued, and w_walk the first unsearched waiter.
>+        // Without the spinlock, the locations mu_ and h->next may now change
>+        // underneath us, but since we hold the lock itself, the only legal
>+        // change is to add waiters between h and w_walk.  Therefore, it's safe
>+        // to walk the path from w_walk to h inclusive. (TryRemove() can remove
>+        // a waiter anywhere, but it acquires both the spinlock and the Mutex)
>+
>+        old_h = h;        // remember we searched to here
>+
>+        // Walk the path upto and including h looking for waiters we can wake.
>+        while (pw_walk != h) {
>+          w_walk->wake = false;
>+          if (w_walk->waitp->cond ==
>+                  nullptr ||  // no condition => vacuously true OR
>+              (w_walk->waitp->cond != known_false &&
>+               // this thread's condition is not known false, AND
>+               //  is in fact true
>+               EvalConditionIgnored(this, w_walk->waitp->cond))) {
>+            if (w == nullptr) {
>+              w_walk->wake = true;    // can wake this waiter
>+              w = w_walk;
>+              pw = pw_walk;
>+              if (w_walk->waitp->how == kExclusive) {
>+                wr_wait = kMuWrWait;
>+                break;                // bail if waking this writer
>+              }
>+            } else if (w_walk->waitp->how == kShared) {  // wake if a reader
>+              w_walk->wake = true;
>+            } else {   // writer with true condition
>+              wr_wait = kMuWrWait;
>+            }
>+          } else {                  // can't wake; condition false
>+            known_false = w_walk->waitp->cond;  // remember last false condition
>+          }
>+          if (w_walk->wake) {   // we're waking reader w_walk
>+            pw_walk = w_walk;   // don't skip similar waiters
>+          } else {              // not waking; skip as much as possible
>+            pw_walk = Skip(w_walk);
>+          }
>+          // If pw_walk == h, then load of pw_walk->next can race with
>+          // concurrent write in Enqueue(). However, at the same time
>+          // we do not need to do the load, because we will bail out
>+          // from the loop anyway.
>+          if (pw_walk != h) {
>+            w_walk = pw_walk->next;
>+          }
>+        }
>+
>+        continue;  // restart for(;;)-loop to wakeup w or to find more waiters
>+      }
>+      ABSL_RAW_CHECK(pw->next == w, "pw not w's predecessor");
>+      // The first (and perhaps only) waiter we've chosen to wake is w, whose
>+      // predecessor is pw.  If w is a reader, we must wake all the other
>+      // waiters with wake==true as well.  We may also need to queue
>+      // ourselves if waitp != null.  The spinlock and the lock are still
>+      // held.
>+
>+      // This traverses the list in [ pw->next, h ], where h is the head,
>+      // removing all elements with wake==true and placing them in the
>+      // singly-linked list wake_list.  Returns the new head.
>+      h = DequeueAllWakeable(h, pw, &wake_list);
>+
>+      intptr_t nv = (v & kMuEvent) | kMuDesig;
>+                                             // assume no waiters left,
>+                                             // set kMuDesig for INV1a
>+
>+      if (waitp != nullptr) {  // we must queue ourselves and sleep
>+        h = Enqueue(h, waitp, v, kMuIsCond);
>+        // h is new last waiter; could be null if we queued ourselves on a
>+        // CondVar
>+      }
>+
>+      ABSL_RAW_CHECK(wake_list != kPerThreadSynchNull,
>+                     "unexpected empty wake list");
>+
>+      if (h != nullptr) {  // there are waiters left
>+        h->readers = 0;
>+        h->maybe_unlocking = false;     // finished unlocking
>+        nv |= wr_wait | kMuWait | reinterpret_cast<intptr_t>(h);
>+      }
>+
>+      // release both spinlock & lock
>+      // can release with a store because there were waiters
>+      mu_.store(nv, std::memory_order_release);
>+      break;  // out of for(;;)-loop
>+    }
>+    c = Delay(c, AGGRESSIVE);  // aggressive here; no one can proceed till we do
>+  }                            // end of for(;;)-loop
>+
>+  if (wake_list != kPerThreadSynchNull) {
>+    int64_t enqueue_timestamp = wake_list->waitp->contention_start_cycles;
>+    bool cond_waiter = wake_list->cond_waiter;
>+    do {
>+      wake_list = Wakeup(wake_list);              // wake waiters
>+    } while (wake_list != kPerThreadSynchNull);
>+    if (!cond_waiter) {
>+      // Sample lock contention events only if the (first) waiter was trying to
>+      // acquire the lock, not waiting on a condition variable or Condition.
>+      int64_t wait_cycles = base_internal::CycleClock::Now() - enqueue_timestamp;
>+      mutex_tracer("slow release", this, wait_cycles);
>+      ABSL_TSAN_MUTEX_PRE_DIVERT(this, 0);
>+      submit_profile_data(enqueue_timestamp);
>+      ABSL_TSAN_MUTEX_POST_DIVERT(this, 0);
>+    }
>+  }
>+}
>+
>+// Used by CondVar implementation to reacquire mutex after waking from
>+// condition variable.  This routine is used instead of Lock() because the
>+// waiting thread may have been moved from the condition variable queue to the
>+// mutex queue without a wakeup, by Trans().  In that case, when the thread is
>+// finally woken, the woken thread will believe it has been woken from the
>+// condition variable (i.e. its PC will be in when in the CondVar code), when
>+// in fact it has just been woken from the mutex.  Thus, it must enter the slow
>+// path of the mutex in the same state as if it had just woken from the mutex.
>+// That is, it must ensure to clear kMuDesig (INV1b).
>+void Mutex::Trans(MuHow how) {
>+  this->LockSlow(how, nullptr, kMuHasBlocked | kMuIsCond);
>+}
>+
>+// Used by CondVar implementation to effectively wake thread w from the
>+// condition variable.  If this mutex is free, we simply wake the thread.
>+// It will later acquire the mutex with high probability.  Otherwise, we
>+// enqueue thread w on this mutex.
>+void Mutex::Fer(PerThreadSynch *w) {
>+  int c = 0;
>+  ABSL_RAW_CHECK(w->waitp->cond == nullptr,
>+                 "Mutex::Fer while waiting on Condition");
>+  ABSL_RAW_CHECK(!w->waitp->timeout.has_timeout(),
>+                 "Mutex::Fer while in timed wait");
>+  ABSL_RAW_CHECK(w->waitp->cv_word == nullptr,
>+                 "Mutex::Fer with pending CondVar queueing");
>+  for (;;) {
>+    intptr_t v = mu_.load(std::memory_order_relaxed);
>+    // Note: must not queue if the mutex is unlocked (nobody will wake it).
>+    // For example, we can have only kMuWait (conditional) or maybe
>+    // kMuWait|kMuWrWait.
>+    // conflicting != 0 implies that the waking thread cannot currently take
>+    // the mutex, which in turn implies that someone else has it and can wake
>+    // us if we queue.
>+    const intptr_t conflicting =
>+        kMuWriter | (w->waitp->how == kShared ? 0 : kMuReader);
>+    if ((v & conflicting) == 0) {
>+      w->next = nullptr;
>+      w->state.store(PerThreadSynch::kAvailable, std::memory_order_release);
>+      IncrementSynchSem(this, w);
>+      return;
>+    } else {
>+      if ((v & (kMuSpin|kMuWait)) == 0) {       // no waiters
>+        // This thread tries to become the one and only waiter.
>+        PerThreadSynch *new_h = Enqueue(nullptr, w->waitp, v, kMuIsCond);
>+        ABSL_RAW_CHECK(new_h != nullptr,
>+                       "Enqueue failed");  // we must queue ourselves
>+        if (mu_.compare_exchange_strong(
>+                v, reinterpret_cast<intptr_t>(new_h) | (v & kMuLow) | kMuWait,
>+                std::memory_order_release, std::memory_order_relaxed)) {
>+          return;
>+        }
>+      } else if ((v & kMuSpin) == 0 &&
>+                 mu_.compare_exchange_strong(v, v | kMuSpin | kMuWait)) {
>+        PerThreadSynch *h = GetPerThreadSynch(v);
>+        PerThreadSynch *new_h = Enqueue(h, w->waitp, v, kMuIsCond);
>+        ABSL_RAW_CHECK(new_h != nullptr,
>+                       "Enqueue failed");  // we must queue ourselves
>+        do {
>+          v = mu_.load(std::memory_order_relaxed);
>+        } while (!mu_.compare_exchange_weak(
>+            v,
>+            (v & kMuLow & ~kMuSpin) | kMuWait |
>+                reinterpret_cast<intptr_t>(new_h),
>+            std::memory_order_release, std::memory_order_relaxed));
>+        return;
>+      }
>+    }
>+    c = Delay(c, GENTLE);
>+  }
>+}
>+
>+void Mutex::AssertHeld() const {
>+  if ((mu_.load(std::memory_order_relaxed) & kMuWriter) == 0) {
>+    SynchEvent *e = GetSynchEvent(this);
>+    ABSL_RAW_LOG(FATAL, "thread should hold write lock on Mutex %p %s",
>+                 static_cast<const void *>(this),
>+                 (e == nullptr ? "" : e->name));
>+  }
>+}
>+
>+void Mutex::AssertReaderHeld() const {
>+  if ((mu_.load(std::memory_order_relaxed) & (kMuReader | kMuWriter)) == 0) {
>+    SynchEvent *e = GetSynchEvent(this);
>+    ABSL_RAW_LOG(
>+        FATAL, "thread should hold at least a read lock on Mutex %p %s",
>+        static_cast<const void *>(this), (e == nullptr ? "" : e->name));
>+  }
>+}
>+
>+// -------------------------------- condition variables
>+static const intptr_t kCvSpin = 0x0001L;   // spinlock protects waiter list
>+static const intptr_t kCvEvent = 0x0002L;  // record events
>+
>+static const intptr_t kCvLow = 0x0003L;  // low order bits of CV
>+
>+// Hack to make constant values available to gdb pretty printer
>+enum { kGdbCvSpin = kCvSpin, kGdbCvEvent = kCvEvent, kGdbCvLow = kCvLow, };
>+
>+static_assert(PerThreadSynch::kAlignment > kCvLow,
>+              "PerThreadSynch::kAlignment must be greater than kCvLow");
>+
>+void CondVar::EnableDebugLog(const char *name) {
>+  SynchEvent *e = EnsureSynchEvent(&this->cv_, name, kCvEvent, kCvSpin);
>+  e->log = true;
>+  UnrefSynchEvent(e);
>+}
>+
>+CondVar::~CondVar() {
>+  if ((cv_.load(std::memory_order_relaxed) & kCvEvent) != 0) {
>+    ForgetSynchEvent(&this->cv_, kCvEvent, kCvSpin);
>+  }
>+}
>+
>+
>+// Remove thread s from the list of waiters on this condition variable.
>+void CondVar::Remove(PerThreadSynch *s) {
>+  intptr_t v;
>+  int c = 0;
>+  for (v = cv_.load(std::memory_order_relaxed);;
>+       v = cv_.load(std::memory_order_relaxed)) {
>+    if ((v & kCvSpin) == 0 &&  // attempt to acquire spinlock
>+        cv_.compare_exchange_strong(v, v | kCvSpin,
>+                                    std::memory_order_acquire,
>+                                    std::memory_order_relaxed)) {
>+      PerThreadSynch *h = reinterpret_cast<PerThreadSynch *>(v & ~kCvLow);
>+      if (h != nullptr) {
>+        PerThreadSynch *w = h;
>+        while (w->next != s && w->next != h) {  // search for thread
>+          w = w->next;
>+        }
>+        if (w->next == s) {           // found thread; remove it
>+          w->next = s->next;
>+          if (h == s) {
>+            h = (w == s) ? nullptr : w;
>+          }
>+          s->next = nullptr;
>+          s->state.store(PerThreadSynch::kAvailable, std::memory_order_release);
>+        }
>+      }
>+                                      // release spinlock
>+      cv_.store((v & kCvEvent) | reinterpret_cast<intptr_t>(h),
>+                std::memory_order_release);
>+      return;
>+    } else {
>+      c = Delay(c, GENTLE);            // try again after a delay
>+    }
>+  }
>+}
>+
>+// Queue thread waitp->thread on condition variable word cv_word using
>+// wait parameters waitp.
>+// We split this into a separate routine, rather than simply doing it as part
>+// of WaitCommon().  If we were to queue ourselves on the condition variable
>+// before calling Mutex::UnlockSlow(), the Mutex code might be re-entered (via
>+// the logging code, or via a Condition function) and might potentially attempt
>+// to block this thread.  That would be a problem if the thread were already on
>+// a the condition variable waiter queue.  Thus, we use the waitp->cv_word
>+// to tell the unlock code to call CondVarEnqueue() to queue the thread on the
>+// condition variable queue just before the mutex is to be unlocked, and (most
>+// importantly) after any call to an external routine that might re-enter the
>+// mutex code.
>+static void CondVarEnqueue(SynchWaitParams *waitp) {
>+  // This thread might be transferred to the Mutex queue by Fer() when
>+  // we are woken.  To make sure that is what happens, Enqueue() doesn't
>+  // call CondVarEnqueue() again but instead uses its normal code.  We
>+  // must do this before we queue ourselves so that cv_word will be null
>+  // when seen by the dequeuer, who may wish immediately to requeue
>+  // this thread on another queue.
>+  std::atomic<intptr_t> *cv_word = waitp->cv_word;
>+  waitp->cv_word = nullptr;
>+
>+  intptr_t v = cv_word->load(std::memory_order_relaxed);
>+  int c = 0;
>+  while ((v & kCvSpin) != 0 ||  // acquire spinlock
>+         !cv_word->compare_exchange_weak(v, v | kCvSpin,
>+                                         std::memory_order_acquire,
>+                                         std::memory_order_relaxed)) {
>+    c = Delay(c, GENTLE);
>+    v = cv_word->load(std::memory_order_relaxed);
>+  }
>+  ABSL_RAW_CHECK(waitp->thread->waitp == nullptr, "waiting when shouldn't be");
>+  waitp->thread->waitp = waitp;      // prepare ourselves for waiting
>+  PerThreadSynch *h = reinterpret_cast<PerThreadSynch *>(v & ~kCvLow);
>+  if (h == nullptr) {  // add this thread to waiter list
>+    waitp->thread->next = waitp->thread;
>+  } else {
>+    waitp->thread->next = h->next;
>+    h->next = waitp->thread;
>+  }
>+  waitp->thread->state.store(PerThreadSynch::kQueued,
>+                             std::memory_order_relaxed);
>+  cv_word->store((v & kCvEvent) | reinterpret_cast<intptr_t>(waitp->thread),
>+                 std::memory_order_release);
>+}
>+
>+bool CondVar::WaitCommon(Mutex *mutex, KernelTimeout t) {
>+  bool rc = false;          // return value; true iff we timed-out
>+
>+  intptr_t mutex_v = mutex->mu_.load(std::memory_order_relaxed);
>+  Mutex::MuHow mutex_how = ((mutex_v & kMuWriter) != 0) ? kExclusive : kShared;
>+  ABSL_TSAN_MUTEX_PRE_UNLOCK(mutex, TsanFlags(mutex_how));
>+
>+  // maybe trace this call
>+  intptr_t v = cv_.load(std::memory_order_relaxed);
>+  cond_var_tracer("Wait", this);
>+  if ((v & kCvEvent) != 0) {
>+    PostSynchEvent(this, SYNCH_EV_WAIT);
>+  }
>+
>+  // Release mu and wait on condition variable.
>+  SynchWaitParams waitp(mutex_how, nullptr, t, mutex,
>+                        Synch_GetPerThreadAnnotated(mutex), &cv_);
>+  // UnlockSlow() will call CondVarEnqueue() just before releasing the
>+  // Mutex, thus queuing this thread on the condition variable.  See
>+  // CondVarEnqueue() for the reasons.
>+  mutex->UnlockSlow(&waitp);
>+
>+  // wait for signal
>+  while (waitp.thread->state.load(std::memory_order_acquire) ==
>+         PerThreadSynch::kQueued) {
>+    if (!Mutex::DecrementSynchSem(mutex, waitp.thread, t)) {
>+      this->Remove(waitp.thread);
>+      rc = true;
>+    }
>+  }
>+
>+  ABSL_RAW_CHECK(waitp.thread->waitp != nullptr, "not waiting when should be");
>+  waitp.thread->waitp = nullptr;  // cleanup
>+
>+  // maybe trace this call
>+  cond_var_tracer("Unwait", this);
>+  if ((v & kCvEvent) != 0) {
>+    PostSynchEvent(this, SYNCH_EV_WAIT_RETURNING);
>+  }
>+
>+  // From synchronization point of view Wait is unlock of the mutex followed
>+  // by lock of the mutex. We've annotated start of unlock in the beginning
>+  // of the function. Now, finish unlock and annotate lock of the mutex.
>+  // (Trans is effectively lock).
>+  ABSL_TSAN_MUTEX_POST_UNLOCK(mutex, TsanFlags(mutex_how));
>+  ABSL_TSAN_MUTEX_PRE_LOCK(mutex, TsanFlags(mutex_how));
>+  mutex->Trans(mutex_how);  // Reacquire mutex
>+  ABSL_TSAN_MUTEX_POST_LOCK(mutex, TsanFlags(mutex_how), 0);
>+  return rc;
>+}
>+
>+bool CondVar::WaitWithTimeout(Mutex *mu, absl::Duration timeout) {
>+  return WaitWithDeadline(mu, DeadlineFromTimeout(timeout));
>+}
>+
>+bool CondVar::WaitWithDeadline(Mutex *mu, absl::Time deadline) {
>+  return WaitCommon(mu, KernelTimeout(deadline));
>+}
>+
>+void CondVar::Wait(Mutex *mu) {
>+  WaitCommon(mu, KernelTimeout::Never());
>+}
>+
>+// Wake thread w
>+// If it was a timed wait, w will be waiting on w->cv
>+// Otherwise, if it was not a Mutex mutex, w will be waiting on w->sem
>+// Otherwise, w is transferred to the Mutex mutex via Mutex::Fer().
>+void CondVar::Wakeup(PerThreadSynch *w) {
>+  if (w->waitp->timeout.has_timeout() || w->waitp->cvmu == nullptr) {
>+    // The waiting thread only needs to observe "w->state == kAvailable" to be
>+    // released, we must cache "cvmu" before clearing "next".
>+    Mutex *mu = w->waitp->cvmu;
>+    w->next = nullptr;
>+    w->state.store(PerThreadSynch::kAvailable, std::memory_order_release);
>+    Mutex::IncrementSynchSem(mu, w);
>+  } else {
>+    w->waitp->cvmu->Fer(w);
>+  }
>+}
>+
>+void CondVar::Signal() {
>+  ABSL_TSAN_MUTEX_PRE_SIGNAL(0, 0);
>+  intptr_t v;
>+  int c = 0;
>+  for (v = cv_.load(std::memory_order_relaxed); v != 0;
>+       v = cv_.load(std::memory_order_relaxed)) {
>+    if ((v & kCvSpin) == 0 &&  // attempt to acquire spinlock
>+        cv_.compare_exchange_strong(v, v | kCvSpin,
>+                                    std::memory_order_acquire,
>+                                    std::memory_order_relaxed)) {
>+      PerThreadSynch *h = reinterpret_cast<PerThreadSynch *>(v & ~kCvLow);
>+      PerThreadSynch *w = nullptr;
>+      if (h != nullptr) {  // remove first waiter
>+        w = h->next;
>+        if (w == h) {
>+          h = nullptr;
>+        } else {
>+          h->next = w->next;
>+        }
>+      }
>+                                      // release spinlock
>+      cv_.store((v & kCvEvent) | reinterpret_cast<intptr_t>(h),
>+                std::memory_order_release);
>+      if (w != nullptr) {
>+        CondVar::Wakeup(w);                // wake waiter, if there was one
>+        cond_var_tracer("Signal wakeup", this);
>+      }
>+      if ((v & kCvEvent) != 0) {
>+        PostSynchEvent(this, SYNCH_EV_SIGNAL);
>+      }
>+      ABSL_TSAN_MUTEX_POST_SIGNAL(0, 0);
>+      return;
>+    } else {
>+      c = Delay(c, GENTLE);
>+    }
>+  }
>+  ABSL_TSAN_MUTEX_POST_SIGNAL(0, 0);
>+}
>+
>+void CondVar::SignalAll () {
>+  ABSL_TSAN_MUTEX_PRE_SIGNAL(0, 0);
>+  intptr_t v;
>+  int c = 0;
>+  for (v = cv_.load(std::memory_order_relaxed); v != 0;
>+       v = cv_.load(std::memory_order_relaxed)) {
>+    // empty the list if spinlock free
>+    // We do this by simply setting the list to empty using
>+    // compare and swap.   We then have the entire list in our hands,
>+    // which cannot be changing since we grabbed it while no one
>+    // held the lock.
>+    if ((v & kCvSpin) == 0 &&
>+        cv_.compare_exchange_strong(v, v & kCvEvent, std::memory_order_acquire,
>+                                    std::memory_order_relaxed)) {
>+      PerThreadSynch *h = reinterpret_cast<PerThreadSynch *>(v & ~kCvLow);
>+      if (h != nullptr) {
>+        PerThreadSynch *w;
>+        PerThreadSynch *n = h->next;
>+        do {                          // for every thread, wake it up
>+          w = n;
>+          n = n->next;
>+          CondVar::Wakeup(w);
>+        } while (w != h);
>+        cond_var_tracer("SignalAll wakeup", this);
>+      }
>+      if ((v & kCvEvent) != 0) {
>+        PostSynchEvent(this, SYNCH_EV_SIGNALALL);
>+      }
>+      ABSL_TSAN_MUTEX_POST_SIGNAL(0, 0);
>+      return;
>+    } else {
>+      c = Delay(c, GENTLE);           // try again after a delay
>+    }
>+  }
>+  ABSL_TSAN_MUTEX_POST_SIGNAL(0, 0);
>+}
>+
>+void ReleasableMutexLock::Release() {
>+  ABSL_RAW_CHECK(this->mu_ != nullptr,
>+                 "ReleasableMutexLock::Release may only be called once");
>+  this->mu_->Unlock();
>+  this->mu_ = nullptr;
>+}
>+
>+#ifdef THREAD_SANITIZER
>+extern "C" void __tsan_read1(void *addr);
>+#else
>+#define __tsan_read1(addr)  // do nothing if TSan not enabled
>+#endif
>+
>+// A function that just returns its argument, dereferenced
>+static bool Dereference(void *arg) {
>+  // ThreadSanitizer does not instrument this file for memory accesses.
>+  // This function dereferences a user variable that can participate
>+  // in a data race, so we need to manually tell TSan about this memory access.
>+  __tsan_read1(arg);
>+  return *(static_cast<bool *>(arg));
>+}
>+
>+Condition::Condition() {}   // null constructor, used for kTrue only
>+const Condition Condition::kTrue;
>+
>+Condition::Condition(bool (*func)(void *), void *arg)
>+    : eval_(&CallVoidPtrFunction),
>+      function_(func),
>+      method_(nullptr),
>+      arg_(arg) {}
>+
>+bool Condition::CallVoidPtrFunction(const Condition *c) {
>+  return (*c->function_)(c->arg_);
>+}
>+
>+Condition::Condition(const bool *cond)
>+    : eval_(CallVoidPtrFunction),
>+      function_(Dereference),
>+      method_(nullptr),
>+      // const_cast is safe since Dereference does not modify arg
>+      arg_(const_cast<bool *>(cond)) {}
>+
>+bool Condition::Eval() const {
>+  // eval_ == null for kTrue
>+  return (this->eval_ == nullptr) || (*this->eval_)(this);
>+}
>+
>+bool Condition::GuaranteedEqual(const Condition *a, const Condition *b) {
>+  if (a == nullptr) {
>+    return b == nullptr || b->eval_ == nullptr;
>+  }
>+  if (b == nullptr || b->eval_ == nullptr) {
>+    return a->eval_ == nullptr;
>+  }
>+  return a->eval_ == b->eval_ && a->function_ == b->function_ &&
>+         a->arg_ == b->arg_ && a->method_ == b->method_;
>+}
>+
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/mutex.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/mutex.h
>new file mode 100644
>index 00000000000..83c214867ac
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/mutex.h
>@@ -0,0 +1,1028 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// -----------------------------------------------------------------------------
>+// mutex.h
>+// -----------------------------------------------------------------------------
>+//
>+// This header file defines a `Mutex` -- a mutually exclusive lock -- and the
>+// most common type of synchronization primitive for facilitating locks on
>+// shared resources. A mutex is used to prevent multiple threads from accessing
>+// and/or writing to a shared resource concurrently.
>+//
>+// Unlike a `std::mutex`, the Abseil `Mutex` provides the following additional
>+// features:
>+//   * Conditional predicates intrinsic to the `Mutex` object
>+//   * Shared/reader locks, in addition to standard exclusive/writer locks
>+//   * Deadlock detection and debug support.
>+//
>+// The following helper classes are also defined within this file:
>+//
>+//  MutexLock - An RAII wrapper to acquire and release a `Mutex` for exclusive/
>+//              write access within the current scope.
>+//  ReaderMutexLock
>+//            - An RAII wrapper to acquire and release a `Mutex` for shared/read
>+//              access within the current scope.
>+//
>+//  WriterMutexLock
>+//            - Alias for `MutexLock` above, designed for use in distinguishing
>+//              reader and writer locks within code.
>+//
>+// In addition to simple mutex locks, this file also defines ways to perform
>+// locking under certain conditions.
>+//
>+//  Condition   - (Preferred) Used to wait for a particular predicate that
>+//                depends on state protected by the `Mutex` to become true.
>+//  CondVar     - A lower-level variant of `Condition` that relies on
>+//                application code to explicitly signal the `CondVar` when
>+//                a condition has been met.
>+//
>+// See below for more information on using `Condition` or `CondVar`.
>+//
>+// Mutexes and mutex behavior can be quite complicated. The information within
>+// this header file is limited, as a result. Please consult the Mutex guide for
>+// more complete information and examples.
>+
>+#ifndef ABSL_SYNCHRONIZATION_MUTEX_H_
>+#define ABSL_SYNCHRONIZATION_MUTEX_H_
>+
>+#include <atomic>
>+#include <cstdint>
>+#include <string>
>+
>+#include "absl/base/internal/identity.h"
>+#include "absl/base/internal/low_level_alloc.h"
>+#include "absl/base/internal/thread_identity.h"
>+#include "absl/base/internal/tsan_mutex_interface.h"
>+#include "absl/base/port.h"
>+#include "absl/base/thread_annotations.h"
>+#include "absl/synchronization/internal/kernel_timeout.h"
>+#include "absl/synchronization/internal/per_thread_sem.h"
>+#include "absl/time/time.h"
>+
>+// Decide if we should use the non-production implementation because
>+// the production implementation hasn't been fully ported yet.
>+#ifdef ABSL_INTERNAL_USE_NONPROD_MUTEX
>+#error ABSL_INTERNAL_USE_NONPROD_MUTEX cannot be directly set
>+#elif defined(ABSL_LOW_LEVEL_ALLOC_MISSING)
>+#define ABSL_INTERNAL_USE_NONPROD_MUTEX 1
>+#include "absl/synchronization/internal/mutex_nonprod.inc"
>+#endif
>+
>+namespace absl {
>+
>+class Condition;
>+struct SynchWaitParams;
>+
>+// -----------------------------------------------------------------------------
>+// Mutex
>+// -----------------------------------------------------------------------------
>+//
>+// A `Mutex` is a non-reentrant (aka non-recursive) Mutually Exclusive lock
>+// on some resource, typically a variable or data structure with associated
>+// invariants. Proper usage of mutexes prevents concurrent access by different
>+// threads to the same resource.
>+//
>+// A `Mutex` has two basic operations: `Mutex::Lock()` and `Mutex::Unlock()`.
>+// The `Lock()` operation *acquires* a `Mutex` (in a state known as an
>+// *exclusive* -- or write -- lock), while the `Unlock()` operation *releases* a
>+// Mutex. During the span of time between the Lock() and Unlock() operations,
>+// a mutex is said to be *held*. By design all mutexes support exclusive/write
>+// locks, as this is the most common way to use a mutex.
>+//
>+// The `Mutex` state machine for basic lock/unlock operations is quite simple:
>+//
>+// |                | Lock()     | Unlock() |
>+// |----------------+------------+----------|
>+// | Free           | Exclusive  | invalid  |
>+// | Exclusive      | blocks     | Free     |
>+//
>+// Attempts to `Unlock()` must originate from the thread that performed the
>+// corresponding `Lock()` operation.
>+//
>+// An "invalid" operation is disallowed by the API. The `Mutex` implementation
>+// is allowed to do anything on an invalid call, including but not limited to
>+// crashing with a useful error message, silently succeeding, or corrupting
>+// data structures. In debug mode, the implementation attempts to crash with a
>+// useful error message.
>+//
>+// `Mutex` is not guaranteed to be "fair" in prioritizing waiting threads; it
>+// is, however, approximately fair over long periods, and starvation-free for
>+// threads at the same priority.
>+//
>+// The lock/unlock primitives are now annotated with lock annotations
>+// defined in (base/thread_annotations.h). When writing multi-threaded code,
>+// you should use lock annotations whenever possible to document your lock
>+// synchronization policy. Besides acting as documentation, these annotations
>+// also help compilers or static analysis tools to identify and warn about
>+// issues that could potentially result in race conditions and deadlocks.
>+//
>+// For more information about the lock annotations, please see
>+// [Thread Safety Analysis](http://clang.llvm.org/docs/ThreadSafetyAnalysis.html)
>+// in the Clang documentation.
>+//
>+// See also `MutexLock`, below, for scoped `Mutex` acquisition.
>+
>+class LOCKABLE Mutex {
>+ public:
>+  Mutex();
>+  ~Mutex();
>+
>+  // Mutex::Lock()
>+  //
>+  // Blocks the calling thread, if necessary, until this `Mutex` is free, and
>+  // then acquires it exclusively. (This lock is also known as a "write lock.")
>+  void Lock() EXCLUSIVE_LOCK_FUNCTION();
>+
>+  // Mutex::Unlock()
>+  //
>+  // Releases this `Mutex` and returns it from the exclusive/write state to the
>+  // free state. Caller must hold the `Mutex` exclusively.
>+  void Unlock() UNLOCK_FUNCTION();
>+
>+  // Mutex::TryLock()
>+  //
>+  // If the mutex can be acquired without blocking, does so exclusively and
>+  // returns `true`. Otherwise, returns `false`. Returns `true` with high
>+  // probability if the `Mutex` was free.
>+  bool TryLock() EXCLUSIVE_TRYLOCK_FUNCTION(true);
>+
>+  // Mutex::AssertHeld()
>+  //
>+  // Return immediately if this thread holds the `Mutex` exclusively (in write
>+  // mode). Otherwise, may report an error (typically by crashing with a
>+  // diagnostic), or may return immediately.
>+  void AssertHeld() const ASSERT_EXCLUSIVE_LOCK();
>+
>+  // ---------------------------------------------------------------------------
>+  // Reader-Writer Locking
>+  // ---------------------------------------------------------------------------
>+
>+  // A Mutex can also be used as a starvation-free reader-writer lock.
>+  // Neither read-locks nor write-locks are reentrant/recursive to avoid
>+  // potential client programming errors.
>+  //
>+  // The Mutex API provides `Writer*()` aliases for the existing `Lock()`,
>+  // `Unlock()` and `TryLock()` methods for use within applications mixing
>+  // reader/writer locks. Using `Reader*()` and `Writer*()` operations in this
>+  // manner can make locking behavior clearer when mixing read and write modes.
>+  //
>+  // Introducing reader locks necessarily complicates the `Mutex` state
>+  // machine somewhat. The table below illustrates the allowed state transitions
>+  // of a mutex in such cases. Note that ReaderLock() may block even if the lock
>+  // is held in shared mode; this occurs when another thread is blocked on a
>+  // call to WriterLock().
>+  //
>+  // ---------------------------------------------------------------------------
>+  //     Operation: WriterLock() Unlock()  ReaderLock()           ReaderUnlock()
>+  // ---------------------------------------------------------------------------
>+  // State
>+  // ---------------------------------------------------------------------------
>+  // Free           Exclusive    invalid   Shared(1)              invalid
>+  // Shared(1)      blocks       invalid   Shared(2) or blocks    Free
>+  // Shared(n) n>1  blocks       invalid   Shared(n+1) or blocks  Shared(n-1)
>+  // Exclusive      blocks       Free      blocks                 invalid
>+  // ---------------------------------------------------------------------------
>+  //
>+  // In comments below, "shared" refers to a state of Shared(n) for any n > 0.
>+
>+  // Mutex::ReaderLock()
>+  //
>+  // Blocks the calling thread, if necessary, until this `Mutex` is either free,
>+  // or in shared mode, and then acquires a share of it. Note that
>+  // `ReaderLock()` will block if some other thread has an exclusive/writer lock
>+  // on the mutex.
>+
>+  void ReaderLock() SHARED_LOCK_FUNCTION();
>+
>+  // Mutex::ReaderUnlock()
>+  //
>+  // Releases a read share of this `Mutex`. `ReaderUnlock` may return a mutex to
>+  // the free state if this thread holds the last reader lock on the mutex. Note
>+  // that you cannot call `ReaderUnlock()` on a mutex held in write mode.
>+  void ReaderUnlock() UNLOCK_FUNCTION();
>+
>+  // Mutex::ReaderTryLock()
>+  //
>+  // If the mutex can be acquired without blocking, acquires this mutex for
>+  // shared access and returns `true`. Otherwise, returns `false`. Returns
>+  // `true` with high probability if the `Mutex` was free or shared.
>+  bool ReaderTryLock() SHARED_TRYLOCK_FUNCTION(true);
>+
>+  // Mutex::AssertReaderHeld()
>+  //
>+  // Returns immediately if this thread holds the `Mutex` in at least shared
>+  // mode (read mode). Otherwise, may report an error (typically by
>+  // crashing with a diagnostic), or may return immediately.
>+  void AssertReaderHeld() const ASSERT_SHARED_LOCK();
>+
>+  // Mutex::WriterLock()
>+  // Mutex::WriterUnlock()
>+  // Mutex::WriterTryLock()
>+  //
>+  // Aliases for `Mutex::Lock()`, `Mutex::Unlock()`, and `Mutex::TryLock()`.
>+  //
>+  // These methods may be used (along with the complementary `Reader*()`
>+  // methods) to distingish simple exclusive `Mutex` usage (`Lock()`,
>+  // etc.) from reader/writer lock usage.
>+  void WriterLock() EXCLUSIVE_LOCK_FUNCTION() { this->Lock(); }
>+
>+  void WriterUnlock() UNLOCK_FUNCTION() { this->Unlock(); }
>+
>+  bool WriterTryLock() EXCLUSIVE_TRYLOCK_FUNCTION(true) {
>+    return this->TryLock();
>+  }
>+
>+  // ---------------------------------------------------------------------------
>+  // Conditional Critical Regions
>+  // ---------------------------------------------------------------------------
>+
>+  // Conditional usage of a `Mutex` can occur using two distinct paradigms:
>+  //
>+  //   * Use of `Mutex` member functions with `Condition` objects.
>+  //   * Use of the separate `CondVar` abstraction.
>+  //
>+  // In general, prefer use of `Condition` and the `Mutex` member functions
>+  // listed below over `CondVar`. When there are multiple threads waiting on
>+  // distinctly different conditions, however, a battery of `CondVar`s may be
>+  // more efficient. This section discusses use of `Condition` objects.
>+  //
>+  // `Mutex` contains member functions for performing lock operations only under
>+  // certain conditions, of class `Condition`. For correctness, the `Condition`
>+  // must return a boolean that is a pure function, only of state protected by
>+  // the `Mutex`. The condition must be invariant w.r.t. environmental state
>+  // such as thread, cpu id, or time, and must be `noexcept`. The condition will
>+  // always be invoked with the mutex held in at least read mode, so you should
>+  // not block it for long periods or sleep it on a timer.
>+  //
>+  // Since a condition must not depend directly on the current time, use
>+  // `*WithTimeout()` member function variants to make your condition
>+  // effectively true after a given duration, or `*WithDeadline()` variants to
>+  // make your condition effectively true after a given time.
>+  //
>+  // The condition function should have no side-effects aside from debug
>+  // logging; as a special exception, the function may acquire other mutexes
>+  // provided it releases all those that it acquires.  (This exception was
>+  // required to allow logging.)
>+
>+  // Mutex::Await()
>+  //
>+  // Unlocks this `Mutex` and blocks until simultaneously both `cond` is `true`
>+  // and this `Mutex` can be reacquired, then reacquires this `Mutex` in the
>+  // same mode in which it was previously held. If the condition is initially
>+  // `true`, `Await()` *may* skip the release/re-acquire step.
>+  //
>+  // `Await()` requires that this thread holds this `Mutex` in some mode.
>+  void Await(const Condition &cond);
>+
>+  // Mutex::LockWhen()
>+  // Mutex::ReaderLockWhen()
>+  // Mutex::WriterLockWhen()
>+  //
>+  // Blocks until simultaneously both `cond` is `true` and this `Mutex` can
>+  // be acquired, then atomically acquires this `Mutex`. `LockWhen()` is
>+  // logically equivalent to `*Lock(); Await();` though they may have different
>+  // performance characteristics.
>+  void LockWhen(const Condition &cond) EXCLUSIVE_LOCK_FUNCTION();
>+
>+  void ReaderLockWhen(const Condition &cond) SHARED_LOCK_FUNCTION();
>+
>+  void WriterLockWhen(const Condition &cond) EXCLUSIVE_LOCK_FUNCTION() {
>+    this->LockWhen(cond);
>+  }
>+
>+  // ---------------------------------------------------------------------------
>+  // Mutex Variants with Timeouts/Deadlines
>+  // ---------------------------------------------------------------------------
>+
>+  // Mutex::AwaitWithTimeout()
>+  // Mutex::AwaitWithDeadline()
>+  //
>+  // If `cond` is initially true, do nothing, or act as though `cond` is
>+  // initially false.
>+  //
>+  // If `cond` is initially false, unlock this `Mutex` and block until
>+  // simultaneously:
>+  //   - either `cond` is true or the {timeout has expired, deadline has passed}
>+  //     and
>+  //   - this `Mutex` can be reacquired,
>+  // then reacquire this `Mutex` in the same mode in which it was previously
>+  // held, returning `true` iff `cond` is `true` on return.
>+  //
>+  // Deadlines in the past are equivalent to an immediate deadline.
>+  // Negative timeouts are equivalent to a zero timeout.
>+  //
>+  // This method requires that this thread holds this `Mutex` in some mode.
>+  bool AwaitWithTimeout(const Condition &cond, absl::Duration timeout);
>+
>+  bool AwaitWithDeadline(const Condition &cond, absl::Time deadline);
>+
>+  // Mutex::LockWhenWithTimeout()
>+  // Mutex::ReaderLockWhenWithTimeout()
>+  // Mutex::WriterLockWhenWithTimeout()
>+  //
>+  // Blocks until simultaneously both:
>+  //   - either `cond` is `true` or the timeout has expired, and
>+  //   - this `Mutex` can be acquired,
>+  // then atomically acquires this `Mutex`, returning `true` iff `cond` is
>+  // `true` on return.
>+  //
>+  // Negative timeouts are equivalent to a zero timeout.
>+  bool LockWhenWithTimeout(const Condition &cond, absl::Duration timeout)
>+      EXCLUSIVE_LOCK_FUNCTION();
>+  bool ReaderLockWhenWithTimeout(const Condition &cond, absl::Duration timeout)
>+      SHARED_LOCK_FUNCTION();
>+  bool WriterLockWhenWithTimeout(const Condition &cond, absl::Duration timeout)
>+      EXCLUSIVE_LOCK_FUNCTION() {
>+    return this->LockWhenWithTimeout(cond, timeout);
>+  }
>+
>+  // Mutex::LockWhenWithDeadline()
>+  // Mutex::ReaderLockWhenWithDeadline()
>+  // Mutex::WriterLockWhenWithDeadline()
>+  //
>+  // Blocks until simultaneously both:
>+  //   - either `cond` is `true` or the deadline has been passed, and
>+  //   - this `Mutex` can be acquired,
>+  // then atomically acquires this Mutex, returning `true` iff `cond` is `true`
>+  // on return.
>+  //
>+  // Deadlines in the past are equivalent to an immediate deadline.
>+  bool LockWhenWithDeadline(const Condition &cond, absl::Time deadline)
>+      EXCLUSIVE_LOCK_FUNCTION();
>+  bool ReaderLockWhenWithDeadline(const Condition &cond, absl::Time deadline)
>+      SHARED_LOCK_FUNCTION();
>+  bool WriterLockWhenWithDeadline(const Condition &cond, absl::Time deadline)
>+      EXCLUSIVE_LOCK_FUNCTION() {
>+    return this->LockWhenWithDeadline(cond, deadline);
>+  }
>+
>+  // ---------------------------------------------------------------------------
>+  // Debug Support: Invariant Checking, Deadlock Detection, Logging.
>+  // ---------------------------------------------------------------------------
>+
>+  // Mutex::EnableInvariantDebugging()
>+  //
>+  // If `invariant`!=null and if invariant debugging has been enabled globally,
>+  // cause `(*invariant)(arg)` to be called at moments when the invariant for
>+  // this `Mutex` should hold (for example: just after acquire, just before
>+  // release).
>+  //
>+  // The routine `invariant` should have no side-effects since it is not
>+  // guaranteed how many times it will be called; it should check the invariant
>+  // and crash if it does not hold. Enabling global invariant debugging may
>+  // substantially reduce `Mutex` performance; it should be set only for
>+  // non-production runs.  Optimization options may also disable invariant
>+  // checks.
>+  void EnableInvariantDebugging(void (*invariant)(void *), void *arg);
>+
>+  // Mutex::EnableDebugLog()
>+  //
>+  // Cause all subsequent uses of this `Mutex` to be logged via
>+  // `ABSL_RAW_LOG(INFO)`. Log entries are tagged with `name` if no previous
>+  // call to `EnableInvariantDebugging()` or `EnableDebugLog()` has been made.
>+  //
>+  // Note: This method substantially reduces `Mutex` performance.
>+  void EnableDebugLog(const char *name);
>+
>+  // Deadlock detection
>+
>+  // Mutex::ForgetDeadlockInfo()
>+  //
>+  // Forget any deadlock-detection information previously gathered
>+  // about this `Mutex`. Call this method in debug mode when the lock ordering
>+  // of a `Mutex` changes.
>+  void ForgetDeadlockInfo();
>+
>+  // Mutex::AssertNotHeld()
>+  //
>+  // Return immediately if this thread does not hold this `Mutex` in any
>+  // mode; otherwise, may report an error (typically by crashing with a
>+  // diagnostic), or may return immediately.
>+  //
>+  // Currently this check is performed only if all of:
>+  //    - in debug mode
>+  //    - SetMutexDeadlockDetectionMode() has been set to kReport or kAbort
>+  //    - number of locks concurrently held by this thread is not large.
>+  // are true.
>+  void AssertNotHeld() const;
>+
>+  // Special cases.
>+
>+  // A `MuHow` is a constant that indicates how a lock should be acquired.
>+  // Internal implementation detail.  Clients should ignore.
>+  typedef const struct MuHowS *MuHow;
>+
>+  // Mutex::InternalAttemptToUseMutexInFatalSignalHandler()
>+  //
>+  // Causes the `Mutex` implementation to prepare itself for re-entry caused by
>+  // future use of `Mutex` within a fatal signal handler. This method is
>+  // intended for use only for last-ditch attempts to log crash information.
>+  // It does not guarantee that attempts to use Mutexes within the handler will
>+  // not deadlock; it merely makes other faults less likely.
>+  //
>+  // WARNING:  This routine must be invoked from a signal handler, and the
>+  // signal handler must either loop forever or terminate the process.
>+  // Attempts to return from (or `longjmp` out of) the signal handler once this
>+  // call has been made may cause arbitrary program behaviour including
>+  // crashes and deadlocks.
>+  static void InternalAttemptToUseMutexInFatalSignalHandler();
>+
>+ private:
>+#ifdef ABSL_INTERNAL_USE_NONPROD_MUTEX
>+  friend class CondVar;
>+
>+  synchronization_internal::MutexImpl *impl() { return impl_.get(); }
>+
>+  synchronization_internal::SynchronizationStorage<
>+      synchronization_internal::MutexImpl>
>+      impl_;
>+#else
>+  std::atomic<intptr_t> mu_;  // The Mutex state.
>+
>+  // Post()/Wait() versus associated PerThreadSem; in class for required
>+  // friendship with PerThreadSem.
>+  static inline void IncrementSynchSem(Mutex *mu,
>+                                       base_internal::PerThreadSynch *w);
>+  static inline bool DecrementSynchSem(
>+      Mutex *mu, base_internal::PerThreadSynch *w,
>+      synchronization_internal::KernelTimeout t);
>+
>+  // slow path acquire
>+  void LockSlowLoop(SynchWaitParams *waitp, int flags);
>+  // wrappers around LockSlowLoop()
>+  bool LockSlowWithDeadline(MuHow how, const Condition *cond,
>+                            synchronization_internal::KernelTimeout t,
>+                            int flags);
>+  void LockSlow(MuHow how, const Condition *cond,
>+                int flags) ABSL_ATTRIBUTE_COLD;
>+  // slow path release
>+  void UnlockSlow(SynchWaitParams *waitp) ABSL_ATTRIBUTE_COLD;
>+  // Common code between Await() and AwaitWithTimeout/Deadline()
>+  bool AwaitCommon(const Condition &cond,
>+                   synchronization_internal::KernelTimeout t);
>+  // Attempt to remove thread s from queue.
>+  void TryRemove(base_internal::PerThreadSynch *s);
>+  // Block a thread on mutex.
>+  void Block(base_internal::PerThreadSynch *s);
>+  // Wake a thread; return successor.
>+  base_internal::PerThreadSynch *Wakeup(base_internal::PerThreadSynch *w);
>+
>+  friend class CondVar;   // for access to Trans()/Fer().
>+  void Trans(MuHow how);  // used for CondVar->Mutex transfer
>+  void Fer(
>+      base_internal::PerThreadSynch *w);  // used for CondVar->Mutex transfer
>+#endif
>+
>+  // Catch the error of writing Mutex when intending MutexLock.
>+  Mutex(const volatile Mutex * /*ignored*/) {}  // NOLINT(runtime/explicit)
>+
>+  Mutex(const Mutex&) = delete;
>+  Mutex& operator=(const Mutex&) = delete;
>+};
>+
>+// -----------------------------------------------------------------------------
>+// Mutex RAII Wrappers
>+// -----------------------------------------------------------------------------
>+
>+// MutexLock
>+//
>+// `MutexLock` is a helper class, which acquires and releases a `Mutex` via
>+// RAII.
>+//
>+// Example:
>+//
>+// Class Foo {
>+//
>+//   Foo::Bar* Baz() {
>+//     MutexLock l(&lock_);
>+//     ...
>+//     return bar;
>+//   }
>+//
>+// private:
>+//   Mutex lock_;
>+// };
>+class SCOPED_LOCKABLE MutexLock {
>+ public:
>+  explicit MutexLock(Mutex *mu) EXCLUSIVE_LOCK_FUNCTION(mu) : mu_(mu) {
>+    this->mu_->Lock();
>+  }
>+
>+  MutexLock(const MutexLock &) = delete;  // NOLINT(runtime/mutex)
>+  MutexLock(MutexLock&&) = delete;  // NOLINT(runtime/mutex)
>+  MutexLock& operator=(const MutexLock&) = delete;
>+  MutexLock& operator=(MutexLock&&) = delete;
>+
>+  ~MutexLock() UNLOCK_FUNCTION() { this->mu_->Unlock(); }
>+
>+ private:
>+  Mutex *const mu_;
>+};
>+
>+// ReaderMutexLock
>+//
>+// The `ReaderMutexLock` is a helper class, like `MutexLock`, which acquires and
>+// releases a shared lock on a `Mutex` via RAII.
>+class SCOPED_LOCKABLE ReaderMutexLock {
>+ public:
>+  explicit ReaderMutexLock(Mutex *mu) SHARED_LOCK_FUNCTION(mu)
>+      :  mu_(mu) {
>+    mu->ReaderLock();
>+  }
>+
>+  ReaderMutexLock(const ReaderMutexLock&) = delete;
>+  ReaderMutexLock(ReaderMutexLock&&) = delete;
>+  ReaderMutexLock& operator=(const ReaderMutexLock&) = delete;
>+  ReaderMutexLock& operator=(ReaderMutexLock&&) = delete;
>+
>+  ~ReaderMutexLock() UNLOCK_FUNCTION() {
>+    this->mu_->ReaderUnlock();
>+  }
>+
>+ private:
>+  Mutex *const mu_;
>+};
>+
>+// WriterMutexLock
>+//
>+// The `WriterMutexLock` is a helper class, like `MutexLock`, which acquires and
>+// releases a write (exclusive) lock on a `Mutex` via RAII.
>+class SCOPED_LOCKABLE WriterMutexLock {
>+ public:
>+  explicit WriterMutexLock(Mutex *mu) EXCLUSIVE_LOCK_FUNCTION(mu)
>+      : mu_(mu) {
>+    mu->WriterLock();
>+  }
>+
>+  WriterMutexLock(const WriterMutexLock&) = delete;
>+  WriterMutexLock(WriterMutexLock&&) = delete;
>+  WriterMutexLock& operator=(const WriterMutexLock&) = delete;
>+  WriterMutexLock& operator=(WriterMutexLock&&) = delete;
>+
>+  ~WriterMutexLock() UNLOCK_FUNCTION() {
>+    this->mu_->WriterUnlock();
>+  }
>+
>+ private:
>+  Mutex *const mu_;
>+};
>+
>+// -----------------------------------------------------------------------------
>+// Condition
>+// -----------------------------------------------------------------------------
>+//
>+// As noted above, `Mutex` contains a number of member functions which take a
>+// `Condition` as a argument; clients can wait for conditions to become `true`
>+// before attempting to acquire the mutex. These sections are known as
>+// "condition critical" sections. To use a `Condition`, you simply need to
>+// construct it, and use within an appropriate `Mutex` member function;
>+// everything else in the `Condition` class is an implementation detail.
>+//
>+// A `Condition` is specified as a function pointer which returns a boolean.
>+// `Condition` functions should be pure functions -- their results should depend
>+// only on passed arguments, should not consult any external state (such as
>+// clocks), and should have no side-effects, aside from debug logging. Any
>+// objects that the function may access should be limited to those which are
>+// constant while the mutex is blocked on the condition (e.g. a stack variable),
>+// or objects of state protected explicitly by the mutex.
>+//
>+// No matter which construction is used for `Condition`, the underlying
>+// function pointer / functor / callable must not throw any
>+// exceptions. Correctness of `Mutex` / `Condition` is not guaranteed in
>+// the face of a throwing `Condition`. (When Abseil is allowed to depend
>+// on C++17, these function pointers will be explicitly marked
>+// `noexcept`; until then this requirement cannot be enforced in the
>+// type system.)
>+//
>+// Note: to use a `Condition`, you need only construct it and pass it within the
>+// appropriate `Mutex' member function, such as `Mutex::Await()`.
>+//
>+// Example:
>+//
>+//   // assume count_ is not internal reference count
>+//   int count_ GUARDED_BY(mu_);
>+//
>+//   mu_.LockWhen(Condition(+[](int* count) { return *count == 0; },
>+//         &count_));
>+//
>+// When multiple threads are waiting on exactly the same condition, make sure
>+// that they are constructed with the same parameters (same pointer to function
>+// + arg, or same pointer to object + method), so that the mutex implementation
>+// can avoid redundantly evaluating the same condition for each thread.
>+class Condition {
>+ public:
>+  // A Condition that returns the result of "(*func)(arg)"
>+  Condition(bool (*func)(void *), void *arg);
>+
>+  // Templated version for people who are averse to casts.
>+  //
>+  // To use a lambda, prepend it with unary plus, which converts the lambda
>+  // into a function pointer:
>+  //     Condition(+[](T* t) { return ...; }, arg).
>+  //
>+  // Note: lambdas in this case must contain no bound variables.
>+  //
>+  // See class comment for performance advice.
>+  template<typename T>
>+  Condition(bool (*func)(T *), T *arg);
>+
>+  // Templated version for invoking a method that returns a `bool`.
>+  //
>+  // `Condition(object, &Class::Method)` constructs a `Condition` that evaluates
>+  // `object->Method()`.
>+  //
>+  // Implementation Note: `absl::internal::identity` is used to allow methods to
>+  // come from base classes. A simpler signature like
>+  // `Condition(T*, bool (T::*)())` does not suffice.
>+  template<typename T>
>+  Condition(T *object, bool (absl::internal::identity<T>::type::* method)());
>+
>+  // Same as above, for const members
>+  template<typename T>
>+  Condition(const T *object,
>+            bool (absl::internal::identity<T>::type::* method)() const);
>+
>+  // A Condition that returns the value of `*cond`
>+  explicit Condition(const bool *cond);
>+
>+  // Templated version for invoking a functor that returns a `bool`.
>+  // This approach accepts pointers to non-mutable lambdas, `std::function`,
>+  // the result of` std::bind` and user-defined functors that define
>+  // `bool F::operator()() const`.
>+  //
>+  // Example:
>+  //
>+  //   auto reached = [this, current]() {
>+  //     mu_.AssertReaderHeld();                // For annotalysis.
>+  //     return processed_ >= current;
>+  //   };
>+  //   mu_.Await(Condition(&reached));
>+
>+  // See class comment for performance advice. In particular, if there
>+  // might be more than one waiter for the same condition, make sure
>+  // that all waiters construct the condition with the same pointers.
>+
>+  // Implementation note: The second template parameter ensures that this
>+  // constructor doesn't participate in overload resolution if T doesn't have
>+  // `bool operator() const`.
>+  template <typename T, typename E = decltype(
>+      static_cast<bool (T::*)() const>(&T::operator()))>
>+  explicit Condition(const T *obj)
>+      : Condition(obj, static_cast<bool (T::*)() const>(&T::operator())) {}
>+
>+  // A Condition that always returns `true`.
>+  static const Condition kTrue;
>+
>+  // Evaluates the condition.
>+  bool Eval() const;
>+
>+  // Returns `true` if the two conditions are guaranteed to return the same
>+  // value if evaluated at the same time, `false` if the evaluation *may* return
>+  // different results.
>+  //
>+  // Two `Condition` values are guaranteed equal if both their `func` and `arg`
>+  // components are the same. A null pointer is equivalent to a `true`
>+  // condition.
>+  static bool GuaranteedEqual(const Condition *a, const Condition *b);
>+
>+ private:
>+  typedef bool (*InternalFunctionType)(void * arg);
>+  typedef bool (Condition::*InternalMethodType)();
>+  typedef bool (*InternalMethodCallerType)(void * arg,
>+                                           InternalMethodType internal_method);
>+
>+  bool (*eval_)(const Condition*);  // Actual evaluator
>+  InternalFunctionType function_;   // function taking pointer returning bool
>+  InternalMethodType method_;       // method returning bool
>+  void *arg_;                       // arg of function_ or object of method_
>+
>+  Condition();        // null constructor used only to create kTrue
>+
>+  // Various functions eval_ can point to:
>+  static bool CallVoidPtrFunction(const Condition*);
>+  template <typename T> static bool CastAndCallFunction(const Condition* c);
>+  template <typename T> static bool CastAndCallMethod(const Condition* c);
>+};
>+
>+// -----------------------------------------------------------------------------
>+// CondVar
>+// -----------------------------------------------------------------------------
>+//
>+// A condition variable, reflecting state evaluated separately outside of the
>+// `Mutex` object, which can be signaled to wake callers.
>+// This class is not normally needed; use `Mutex` member functions such as
>+// `Mutex::Await()` and intrinsic `Condition` abstractions. In rare cases
>+// with many threads and many conditions, `CondVar` may be faster.
>+//
>+// The implementation may deliver signals to any condition variable at
>+// any time, even when no call to `Signal()` or `SignalAll()` is made; as a
>+// result, upon being awoken, you must check the logical condition you have
>+// been waiting upon.
>+//
>+// Examples:
>+//
>+// Usage for a thread waiting for some condition C protected by mutex mu:
>+//       mu.Lock();
>+//       while (!C) { cv->Wait(&mu); }        // releases and reacquires mu
>+//       //  C holds; process data
>+//       mu.Unlock();
>+//
>+// Usage to wake T is:
>+//       mu.Lock();
>+//      // process data, possibly establishing C
>+//      if (C) { cv->Signal(); }
>+//      mu.Unlock();
>+//
>+// If C may be useful to more than one waiter, use `SignalAll()` instead of
>+// `Signal()`.
>+//
>+// With this implementation it is efficient to use `Signal()/SignalAll()` inside
>+// the locked region; this usage can make reasoning about your program easier.
>+//
>+class CondVar {
>+ public:
>+  CondVar();
>+  ~CondVar();
>+
>+  // CondVar::Wait()
>+  //
>+  // Atomically releases a `Mutex` and blocks on this condition variable.
>+  // Waits until awakened by a call to `Signal()` or `SignalAll()` (or a
>+  // spurious wakeup), then reacquires the `Mutex` and returns.
>+  //
>+  // Requires and ensures that the current thread holds the `Mutex`.
>+  void Wait(Mutex *mu);
>+
>+  // CondVar::WaitWithTimeout()
>+  //
>+  // Atomically releases a `Mutex` and blocks on this condition variable.
>+  // Waits until awakened by a call to `Signal()` or `SignalAll()` (or a
>+  // spurious wakeup), or until the timeout has expired, then reacquires
>+  // the `Mutex` and returns.
>+  //
>+  // Returns true if the timeout has expired without this `CondVar`
>+  // being signalled in any manner. If both the timeout has expired
>+  // and this `CondVar` has been signalled, the implementation is free
>+  // to return `true` or `false`.
>+  //
>+  // Requires and ensures that the current thread holds the `Mutex`.
>+  bool WaitWithTimeout(Mutex *mu, absl::Duration timeout);
>+
>+  // CondVar::WaitWithDeadline()
>+  //
>+  // Atomically releases a `Mutex` and blocks on this condition variable.
>+  // Waits until awakened by a call to `Signal()` or `SignalAll()` (or a
>+  // spurious wakeup), or until the deadline has passed, then reacquires
>+  // the `Mutex` and returns.
>+  //
>+  // Deadlines in the past are equivalent to an immediate deadline.
>+  //
>+  // Returns true if the deadline has passed without this `CondVar`
>+  // being signalled in any manner. If both the deadline has passed
>+  // and this `CondVar` has been signalled, the implementation is free
>+  // to return `true` or `false`.
>+  //
>+  // Requires and ensures that the current thread holds the `Mutex`.
>+  bool WaitWithDeadline(Mutex *mu, absl::Time deadline);
>+
>+  // CondVar::Signal()
>+  //
>+  // Signal this `CondVar`; wake at least one waiter if one exists.
>+  void Signal();
>+
>+  // CondVar::SignalAll()
>+  //
>+  // Signal this `CondVar`; wake all waiters.
>+  void SignalAll();
>+
>+  // CondVar::EnableDebugLog()
>+  //
>+  // Causes all subsequent uses of this `CondVar` to be logged via
>+  // `ABSL_RAW_LOG(INFO)`. Log entries are tagged with `name` if `name != 0`.
>+  // Note: this method substantially reduces `CondVar` performance.
>+  void EnableDebugLog(const char *name);
>+
>+ private:
>+#ifdef ABSL_INTERNAL_USE_NONPROD_MUTEX
>+  synchronization_internal::CondVarImpl *impl() { return impl_.get(); }
>+  synchronization_internal::SynchronizationStorage<
>+      synchronization_internal::CondVarImpl>
>+      impl_;
>+#else
>+  bool WaitCommon(Mutex *mutex, synchronization_internal::KernelTimeout t);
>+  void Remove(base_internal::PerThreadSynch *s);
>+  void Wakeup(base_internal::PerThreadSynch *w);
>+  std::atomic<intptr_t> cv_;  // Condition variable state.
>+#endif
>+  CondVar(const CondVar&) = delete;
>+  CondVar& operator=(const CondVar&) = delete;
>+};
>+
>+
>+// Variants of MutexLock.
>+//
>+// If you find yourself using one of these, consider instead using
>+// Mutex::Unlock() and/or if-statements for clarity.
>+
>+// MutexLockMaybe
>+//
>+// MutexLockMaybe is like MutexLock, but is a no-op when mu is null.
>+class SCOPED_LOCKABLE MutexLockMaybe {
>+ public:
>+  explicit MutexLockMaybe(Mutex *mu) EXCLUSIVE_LOCK_FUNCTION(mu)
>+      : mu_(mu) { if (this->mu_ != nullptr) { this->mu_->Lock(); } }
>+  ~MutexLockMaybe() UNLOCK_FUNCTION() {
>+    if (this->mu_ != nullptr) { this->mu_->Unlock(); }
>+  }
>+ private:
>+  Mutex *const mu_;
>+  MutexLockMaybe(const MutexLockMaybe&) = delete;
>+  MutexLockMaybe(MutexLockMaybe&&) = delete;
>+  MutexLockMaybe& operator=(const MutexLockMaybe&) = delete;
>+  MutexLockMaybe& operator=(MutexLockMaybe&&) = delete;
>+};
>+
>+// ReleasableMutexLock
>+//
>+// ReleasableMutexLock is like MutexLock, but permits `Release()` of its
>+// mutex before destruction. `Release()` may be called at most once.
>+class SCOPED_LOCKABLE ReleasableMutexLock {
>+ public:
>+  explicit ReleasableMutexLock(Mutex *mu) EXCLUSIVE_LOCK_FUNCTION(mu)
>+      : mu_(mu) {
>+    this->mu_->Lock();
>+  }
>+  ~ReleasableMutexLock() UNLOCK_FUNCTION() {
>+    if (this->mu_ != nullptr) { this->mu_->Unlock(); }
>+  }
>+
>+  void Release() UNLOCK_FUNCTION();
>+
>+ private:
>+  Mutex *mu_;
>+  ReleasableMutexLock(const ReleasableMutexLock&) = delete;
>+  ReleasableMutexLock(ReleasableMutexLock&&) = delete;
>+  ReleasableMutexLock& operator=(const ReleasableMutexLock&) = delete;
>+  ReleasableMutexLock& operator=(ReleasableMutexLock&&) = delete;
>+};
>+
>+#ifdef ABSL_INTERNAL_USE_NONPROD_MUTEX
>+#else
>+inline Mutex::Mutex() : mu_(0) {
>+  ABSL_TSAN_MUTEX_CREATE(this, __tsan_mutex_not_static);
>+}
>+
>+inline CondVar::CondVar() : cv_(0) {}
>+#endif
>+
>+// static
>+template <typename T>
>+bool Condition::CastAndCallMethod(const Condition *c) {
>+  typedef bool (T::*MemberType)();
>+  MemberType rm = reinterpret_cast<MemberType>(c->method_);
>+  T *x = static_cast<T *>(c->arg_);
>+  return (x->*rm)();
>+}
>+
>+// static
>+template <typename T>
>+bool Condition::CastAndCallFunction(const Condition *c) {
>+  typedef bool (*FuncType)(T *);
>+  FuncType fn = reinterpret_cast<FuncType>(c->function_);
>+  T *x = static_cast<T *>(c->arg_);
>+  return (*fn)(x);
>+}
>+
>+template <typename T>
>+inline Condition::Condition(bool (*func)(T *), T *arg)
>+    : eval_(&CastAndCallFunction<T>),
>+      function_(reinterpret_cast<InternalFunctionType>(func)),
>+      method_(nullptr),
>+      arg_(const_cast<void *>(static_cast<const void *>(arg))) {}
>+
>+template <typename T>
>+inline Condition::Condition(T *object,
>+                            bool (absl::internal::identity<T>::type::*method)())
>+    : eval_(&CastAndCallMethod<T>),
>+      function_(nullptr),
>+      method_(reinterpret_cast<InternalMethodType>(method)),
>+      arg_(object) {}
>+
>+template <typename T>
>+inline Condition::Condition(const T *object,
>+                            bool (absl::internal::identity<T>::type::*method)()
>+                                const)
>+    : eval_(&CastAndCallMethod<T>),
>+      function_(nullptr),
>+      method_(reinterpret_cast<InternalMethodType>(method)),
>+      arg_(reinterpret_cast<void *>(const_cast<T *>(object))) {}
>+
>+// Register a hook for profiling support.
>+//
>+// The function pointer registered here will be called whenever a mutex is
>+// contended.  The callback is given the absl/base/cycleclock.h timestamp when
>+// waiting began.
>+//
>+// Calls to this function do not race or block, but there is no ordering
>+// guaranteed between calls to this function and call to the provided hook.
>+// In particular, the previously registered hook may still be called for some
>+// time after this function returns.
>+void RegisterMutexProfiler(void (*fn)(int64_t wait_timestamp));
>+
>+// Register a hook for Mutex tracing.
>+//
>+// The function pointer registered here will be called whenever a mutex is
>+// contended.  The callback is given an opaque handle to the contended mutex,
>+// an event name, and the number of wait cycles (as measured by
>+// //absl/base/internal/cycleclock.h, and which may not be real
>+// "cycle" counts.)
>+//
>+// The only event name currently sent is "slow release".
>+//
>+// This has the same memory ordering concerns as RegisterMutexProfiler() above.
>+void RegisterMutexTracer(void (*fn)(const char *msg, const void *obj,
>+                              int64_t wait_cycles));
>+
>+// TODO(gfalcon): Combine RegisterMutexProfiler() and RegisterMutexTracer()
>+// into a single interface, since they are only ever called in pairs.
>+
>+// Register a hook for CondVar tracing.
>+//
>+// The function pointer registered here will be called here on various CondVar
>+// events.  The callback is given an opaque handle to the CondVar object and
>+// a std::string identifying the event.  This is thread-safe, but only a single
>+// tracer can be registered.
>+//
>+// Events that can be sent are "Wait", "Unwait", "Signal wakeup", and
>+// "SignalAll wakeup".
>+//
>+// This has the same memory ordering concerns as RegisterMutexProfiler() above.
>+void RegisterCondVarTracer(void (*fn)(const char *msg, const void *cv));
>+
>+// Register a hook for symbolizing stack traces in deadlock detector reports.
>+//
>+// 'pc' is the program counter being symbolized, 'out' is the buffer to write
>+// into, and 'out_size' is the size of the buffer.  This function can return
>+// false if symbolizing failed, or true if a null-terminated symbol was written
>+// to 'out.'
>+//
>+// This has the same memory ordering concerns as RegisterMutexProfiler() above.
>+//
>+// DEPRECATED: The default symbolizer function is absl::Symbolize() and the
>+// ability to register a different hook for symbolizing stack traces will be
>+// removed on or after 2023-05-01.
>+ABSL_DEPRECATED("absl::RegisterSymbolizer() is deprecated and will be removed "
>+                "on or after 2023-05-01")
>+void RegisterSymbolizer(bool (*fn)(const void *pc, char *out, int out_size));
>+
>+// EnableMutexInvariantDebugging()
>+//
>+// Enable or disable global support for Mutex invariant debugging.  If enabled,
>+// then invariant predicates can be registered per-Mutex for debug checking.
>+// See Mutex::EnableInvariantDebugging().
>+void EnableMutexInvariantDebugging(bool enabled);
>+
>+// When in debug mode, and when the feature has been enabled globally, the
>+// implementation will keep track of lock ordering and complain (or optionally
>+// crash) if a cycle is detected in the acquired-before graph.
>+
>+// Possible modes of operation for the deadlock detector in debug mode.
>+enum class OnDeadlockCycle {
>+  kIgnore,  // Neither report on nor attempt to track cycles in lock ordering
>+  kReport,  // Report lock cycles to stderr when detected
>+  kAbort,  // Report lock cycles to stderr when detected, then abort
>+};
>+
>+// SetMutexDeadlockDetectionMode()
>+//
>+// Enable or disable global support for detection of potential deadlocks
>+// due to Mutex lock ordering inversions.  When set to 'kIgnore', tracking of
>+// lock ordering is disabled.  Otherwise, in debug builds, a lock ordering graph
>+// will be maintained internally, and detected cycles will be reported in
>+// the manner chosen here.
>+void SetMutexDeadlockDetectionMode(OnDeadlockCycle mode);
>+
>+}  // namespace absl
>+
>+// In some build configurations we pass --detect-odr-violations to the
>+// gold linker.  This causes it to flag weak symbol overrides as ODR
>+// violations.  Because ODR only applies to C++ and not C,
>+// --detect-odr-violations ignores symbols not mangled with C++ names.
>+// By changing our extension points to be extern "C", we dodge this
>+// check.
>+extern "C" {
>+void AbslInternalMutexYield();
>+}  // extern "C"
>+#endif  // ABSL_SYNCHRONIZATION_MUTEX_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/mutex_benchmark.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/mutex_benchmark.cc
>new file mode 100644
>index 00000000000..1e019e001ae
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/mutex_benchmark.cc
>@@ -0,0 +1,94 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include <vector>
>+
>+#include "benchmark/benchmark.h"
>+#include "absl/base/internal/sysinfo.h"
>+#include "absl/synchronization/blocking_counter.h"
>+#include "absl/synchronization/internal/thread_pool.h"
>+#include "absl/synchronization/mutex.h"
>+
>+namespace {
>+
>+// Measure the overhead of conditions on mutex release (when they must be
>+// evaluated).  Mutex has (some) support for equivalence classes allowing
>+// Conditions with the same function/argument to potentially not be multiply
>+// evaluated.
>+//
>+// num_classes==0 is used for the special case of every waiter being distinct.
>+void BM_ConditionWaiters(benchmark::State& state) {
>+  int num_classes = state.range(0);
>+  int num_waiters = state.range(1);
>+
>+  struct Helper {
>+    static void Waiter(absl::BlockingCounter* init, absl::Mutex* m, int* p) {
>+      init->DecrementCount();
>+      m->LockWhen(absl::Condition(
>+          static_cast<bool (*)(int*)>([](int* v) { return *v == 0; }), p));
>+      m->Unlock();
>+    }
>+  };
>+
>+  if (num_classes == 0) {
>+    // No equivalence classes.
>+    num_classes = num_waiters;
>+  }
>+
>+  absl::BlockingCounter init(num_waiters);
>+  absl::Mutex mu;
>+  std::vector<int> equivalence_classes(num_classes, 1);
>+
>+  // Must be declared last to be destroyed first.
>+  absl::synchronization_internal::ThreadPool pool(num_waiters);
>+
>+  for (int i = 0; i < num_waiters; i++) {
>+    // Mutex considers Conditions with the same function and argument
>+    // to be equivalent.
>+    pool.Schedule([&, i] {
>+      Helper::Waiter(&init, &mu, &equivalence_classes[i % num_classes]);
>+    });
>+  }
>+  init.Wait();
>+
>+  for (auto _ : state) {
>+    mu.Lock();
>+    mu.Unlock();  // Each unlock requires Condition evaluation for our waiters.
>+  }
>+
>+  mu.Lock();
>+  for (int i = 0; i < num_classes; i++) {
>+    equivalence_classes[i] = 0;
>+  }
>+  mu.Unlock();
>+}
>+
>+// Some configurations have higher thread limits than others.
>+#if defined(__linux__) && !defined(THREAD_SANITIZER)
>+constexpr int kMaxConditionWaiters = 8192;
>+#else
>+constexpr int kMaxConditionWaiters = 1024;
>+#endif
>+BENCHMARK(BM_ConditionWaiters)->RangePair(0, 2, 1, kMaxConditionWaiters);
>+
>+void BM_ContendedMutex(benchmark::State& state) {
>+  static absl::Mutex* mu = new absl::Mutex;
>+  for (auto _ : state) {
>+    absl::MutexLock lock(mu);
>+  }
>+}
>+BENCHMARK(BM_ContendedMutex)->Threads(1);
>+BENCHMARK(BM_ContendedMutex)->ThreadPerCpu();
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/mutex_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/mutex_test.cc
>new file mode 100644
>index 00000000000..b2820e2068d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/mutex_test.cc
>@@ -0,0 +1,1660 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/synchronization/mutex.h"
>+
>+#ifdef WIN32
>+#include <windows.h>
>+#endif
>+
>+#include <algorithm>
>+#include <atomic>
>+#include <cstdlib>
>+#include <functional>
>+#include <memory>
>+#include <random>
>+#include <string>
>+#include <thread>  // NOLINT(build/c++11)
>+#include <vector>
>+
>+#include "gtest/gtest.h"
>+#include "absl/base/attributes.h"
>+#include "absl/base/internal/raw_logging.h"
>+#include "absl/base/internal/sysinfo.h"
>+#include "absl/memory/memory.h"
>+#include "absl/synchronization/internal/thread_pool.h"
>+#include "absl/time/clock.h"
>+#include "absl/time/time.h"
>+
>+namespace {
>+
>+// TODO(dmauro): Replace with a commandline flag.
>+static constexpr bool kExtendedTest = false;
>+
>+std::unique_ptr<absl::synchronization_internal::ThreadPool> CreatePool(
>+    int threads) {
>+  return absl::make_unique<absl::synchronization_internal::ThreadPool>(threads);
>+}
>+
>+std::unique_ptr<absl::synchronization_internal::ThreadPool>
>+CreateDefaultPool() {
>+  return CreatePool(kExtendedTest ? 32 : 10);
>+}
>+
>+// Hack to schedule a function to run on a thread pool thread after a
>+// duration has elapsed.
>+static void ScheduleAfter(absl::synchronization_internal::ThreadPool *tp,
>+                          absl::Duration after,
>+                          const std::function<void()> &func) {
>+  tp->Schedule([func, after] {
>+    absl::SleepFor(after);
>+    func();
>+  });
>+}
>+
>+struct TestContext {
>+  int iterations;
>+  int threads;
>+  int g0;  // global 0
>+  int g1;  // global 1
>+  absl::Mutex mu;
>+  absl::CondVar cv;
>+};
>+
>+// To test whether the invariant check call occurs
>+static std::atomic<bool> invariant_checked;
>+
>+static bool GetInvariantChecked() {
>+  return invariant_checked.load(std::memory_order_relaxed);
>+}
>+
>+static void SetInvariantChecked(bool new_value) {
>+  invariant_checked.store(new_value, std::memory_order_relaxed);
>+}
>+
>+static void CheckSumG0G1(void *v) {
>+  TestContext *cxt = static_cast<TestContext *>(v);
>+  ABSL_RAW_CHECK(cxt->g0 == -cxt->g1, "Error in CheckSumG0G1");
>+  SetInvariantChecked(true);
>+}
>+
>+static void TestMu(TestContext *cxt, int c) {
>+  for (int i = 0; i != cxt->iterations; i++) {
>+    absl::MutexLock l(&cxt->mu);
>+    int a = cxt->g0 + 1;
>+    cxt->g0 = a;
>+    cxt->g1--;
>+  }
>+}
>+
>+static void TestTry(TestContext *cxt, int c) {
>+  for (int i = 0; i != cxt->iterations; i++) {
>+    do {
>+      std::this_thread::yield();
>+    } while (!cxt->mu.TryLock());
>+    int a = cxt->g0 + 1;
>+    cxt->g0 = a;
>+    cxt->g1--;
>+    cxt->mu.Unlock();
>+  }
>+}
>+
>+static void TestR20ms(TestContext *cxt, int c) {
>+  for (int i = 0; i != cxt->iterations; i++) {
>+    absl::ReaderMutexLock l(&cxt->mu);
>+    absl::SleepFor(absl::Milliseconds(20));
>+    cxt->mu.AssertReaderHeld();
>+  }
>+}
>+
>+static void TestRW(TestContext *cxt, int c) {
>+  if ((c & 1) == 0) {
>+    for (int i = 0; i != cxt->iterations; i++) {
>+      absl::WriterMutexLock l(&cxt->mu);
>+      cxt->g0++;
>+      cxt->g1--;
>+      cxt->mu.AssertHeld();
>+      cxt->mu.AssertReaderHeld();
>+    }
>+  } else {
>+    for (int i = 0; i != cxt->iterations; i++) {
>+      absl::ReaderMutexLock l(&cxt->mu);
>+      ABSL_RAW_CHECK(cxt->g0 == -cxt->g1, "Error in TestRW");
>+      cxt->mu.AssertReaderHeld();
>+    }
>+  }
>+}
>+
>+struct MyContext {
>+  int target;
>+  TestContext *cxt;
>+  bool MyTurn();
>+};
>+
>+bool MyContext::MyTurn() {
>+  TestContext *cxt = this->cxt;
>+  return cxt->g0 == this->target || cxt->g0 == cxt->iterations;
>+}
>+
>+static void TestAwait(TestContext *cxt, int c) {
>+  MyContext mc;
>+  mc.target = c;
>+  mc.cxt = cxt;
>+  absl::MutexLock l(&cxt->mu);
>+  cxt->mu.AssertHeld();
>+  while (cxt->g0 < cxt->iterations) {
>+    cxt->mu.Await(absl::Condition(&mc, &MyContext::MyTurn));
>+    ABSL_RAW_CHECK(mc.MyTurn(), "Error in TestAwait");
>+    cxt->mu.AssertHeld();
>+    if (cxt->g0 < cxt->iterations) {
>+      int a = cxt->g0 + 1;
>+      cxt->g0 = a;
>+      mc.target += cxt->threads;
>+    }
>+  }
>+}
>+
>+static void TestSignalAll(TestContext *cxt, int c) {
>+  int target = c;
>+  absl::MutexLock l(&cxt->mu);
>+  cxt->mu.AssertHeld();
>+  while (cxt->g0 < cxt->iterations) {
>+    while (cxt->g0 != target && cxt->g0 != cxt->iterations) {
>+      cxt->cv.Wait(&cxt->mu);
>+    }
>+    if (cxt->g0 < cxt->iterations) {
>+      int a = cxt->g0 + 1;
>+      cxt->g0 = a;
>+      cxt->cv.SignalAll();
>+      target += cxt->threads;
>+    }
>+  }
>+}
>+
>+static void TestSignal(TestContext *cxt, int c) {
>+  ABSL_RAW_CHECK(cxt->threads == 2, "TestSignal should use 2 threads");
>+  int target = c;
>+  absl::MutexLock l(&cxt->mu);
>+  cxt->mu.AssertHeld();
>+  while (cxt->g0 < cxt->iterations) {
>+    while (cxt->g0 != target && cxt->g0 != cxt->iterations) {
>+      cxt->cv.Wait(&cxt->mu);
>+    }
>+    if (cxt->g0 < cxt->iterations) {
>+      int a = cxt->g0 + 1;
>+      cxt->g0 = a;
>+      cxt->cv.Signal();
>+      target += cxt->threads;
>+    }
>+  }
>+}
>+
>+static void TestCVTimeout(TestContext *cxt, int c) {
>+  int target = c;
>+  absl::MutexLock l(&cxt->mu);
>+  cxt->mu.AssertHeld();
>+  while (cxt->g0 < cxt->iterations) {
>+    while (cxt->g0 != target && cxt->g0 != cxt->iterations) {
>+      cxt->cv.WaitWithTimeout(&cxt->mu, absl::Seconds(100));
>+    }
>+    if (cxt->g0 < cxt->iterations) {
>+      int a = cxt->g0 + 1;
>+      cxt->g0 = a;
>+      cxt->cv.SignalAll();
>+      target += cxt->threads;
>+    }
>+  }
>+}
>+
>+static bool G0GE2(TestContext *cxt) { return cxt->g0 >= 2; }
>+
>+static void TestTime(TestContext *cxt, int c, bool use_cv) {
>+  ABSL_RAW_CHECK(cxt->iterations == 1, "TestTime should only use 1 iteration");
>+  ABSL_RAW_CHECK(cxt->threads > 2, "TestTime should use more than 2 threads");
>+  const bool kFalse = false;
>+  absl::Condition false_cond(&kFalse);
>+  absl::Condition g0ge2(G0GE2, cxt);
>+  if (c == 0) {
>+    absl::MutexLock l(&cxt->mu);
>+
>+    absl::Time start = absl::Now();
>+    if (use_cv) {
>+      cxt->cv.WaitWithTimeout(&cxt->mu, absl::Seconds(1));
>+    } else {
>+      ABSL_RAW_CHECK(!cxt->mu.AwaitWithTimeout(false_cond, absl::Seconds(1)),
>+                     "TestTime failed");
>+    }
>+    absl::Duration elapsed = absl::Now() - start;
>+    ABSL_RAW_CHECK(
>+        absl::Seconds(0.9) <= elapsed && elapsed <= absl::Seconds(2.0),
>+        "TestTime failed");
>+    ABSL_RAW_CHECK(cxt->g0 == 1, "TestTime failed");
>+
>+    start = absl::Now();
>+    if (use_cv) {
>+      cxt->cv.WaitWithTimeout(&cxt->mu, absl::Seconds(1));
>+    } else {
>+      ABSL_RAW_CHECK(!cxt->mu.AwaitWithTimeout(false_cond, absl::Seconds(1)),
>+                     "TestTime failed");
>+    }
>+    elapsed = absl::Now() - start;
>+    ABSL_RAW_CHECK(
>+        absl::Seconds(0.9) <= elapsed && elapsed <= absl::Seconds(2.0),
>+        "TestTime failed");
>+    cxt->g0++;
>+    if (use_cv) {
>+      cxt->cv.Signal();
>+    }
>+
>+    start = absl::Now();
>+    if (use_cv) {
>+      cxt->cv.WaitWithTimeout(&cxt->mu, absl::Seconds(4));
>+    } else {
>+      ABSL_RAW_CHECK(!cxt->mu.AwaitWithTimeout(false_cond, absl::Seconds(4)),
>+                     "TestTime failed");
>+    }
>+    elapsed = absl::Now() - start;
>+    ABSL_RAW_CHECK(
>+        absl::Seconds(3.9) <= elapsed && elapsed <= absl::Seconds(6.0),
>+        "TestTime failed");
>+    ABSL_RAW_CHECK(cxt->g0 >= 3, "TestTime failed");
>+
>+    start = absl::Now();
>+    if (use_cv) {
>+      cxt->cv.WaitWithTimeout(&cxt->mu, absl::Seconds(1));
>+    } else {
>+      ABSL_RAW_CHECK(!cxt->mu.AwaitWithTimeout(false_cond, absl::Seconds(1)),
>+                     "TestTime failed");
>+    }
>+    elapsed = absl::Now() - start;
>+    ABSL_RAW_CHECK(
>+        absl::Seconds(0.9) <= elapsed && elapsed <= absl::Seconds(2.0),
>+        "TestTime failed");
>+    if (use_cv) {
>+      cxt->cv.SignalAll();
>+    }
>+
>+    start = absl::Now();
>+    if (use_cv) {
>+      cxt->cv.WaitWithTimeout(&cxt->mu, absl::Seconds(1));
>+    } else {
>+      ABSL_RAW_CHECK(!cxt->mu.AwaitWithTimeout(false_cond, absl::Seconds(1)),
>+                     "TestTime failed");
>+    }
>+    elapsed = absl::Now() - start;
>+    ABSL_RAW_CHECK(absl::Seconds(0.9) <= elapsed &&
>+                   elapsed <= absl::Seconds(2.0), "TestTime failed");
>+    ABSL_RAW_CHECK(cxt->g0 == cxt->threads, "TestTime failed");
>+
>+  } else if (c == 1) {
>+    absl::MutexLock l(&cxt->mu);
>+    const absl::Time start = absl::Now();
>+    if (use_cv) {
>+      cxt->cv.WaitWithTimeout(&cxt->mu, absl::Milliseconds(500));
>+    } else {
>+      ABSL_RAW_CHECK(
>+          !cxt->mu.AwaitWithTimeout(false_cond, absl::Milliseconds(500)),
>+          "TestTime failed");
>+    }
>+    const absl::Duration elapsed = absl::Now() - start;
>+    ABSL_RAW_CHECK(
>+        absl::Seconds(0.4) <= elapsed && elapsed <= absl::Seconds(0.9),
>+        "TestTime failed");
>+    cxt->g0++;
>+  } else if (c == 2) {
>+    absl::MutexLock l(&cxt->mu);
>+    if (use_cv) {
>+      while (cxt->g0 < 2) {
>+        cxt->cv.WaitWithTimeout(&cxt->mu, absl::Seconds(100));
>+      }
>+    } else {
>+      ABSL_RAW_CHECK(cxt->mu.AwaitWithTimeout(g0ge2, absl::Seconds(100)),
>+                     "TestTime failed");
>+    }
>+    cxt->g0++;
>+  } else {
>+    absl::MutexLock l(&cxt->mu);
>+    if (use_cv) {
>+      while (cxt->g0 < 2) {
>+        cxt->cv.Wait(&cxt->mu);
>+      }
>+    } else {
>+      cxt->mu.Await(g0ge2);
>+    }
>+    cxt->g0++;
>+  }
>+}
>+
>+static void TestMuTime(TestContext *cxt, int c) { TestTime(cxt, c, false); }
>+
>+static void TestCVTime(TestContext *cxt, int c) { TestTime(cxt, c, true); }
>+
>+static void EndTest(int *c0, int *c1, absl::Mutex *mu, absl::CondVar *cv,
>+                    const std::function<void(int)>& cb) {
>+  mu->Lock();
>+  int c = (*c0)++;
>+  mu->Unlock();
>+  cb(c);
>+  absl::MutexLock l(mu);
>+  (*c1)++;
>+  cv->Signal();
>+}
>+
>+// Code common to RunTest() and RunTestWithInvariantDebugging().
>+static int RunTestCommon(TestContext *cxt, void (*test)(TestContext *cxt, int),
>+                         int threads, int iterations, int operations) {
>+  absl::Mutex mu2;
>+  absl::CondVar cv2;
>+  int c0 = 0;
>+  int c1 = 0;
>+  cxt->g0 = 0;
>+  cxt->g1 = 0;
>+  cxt->iterations = iterations;
>+  cxt->threads = threads;
>+  absl::synchronization_internal::ThreadPool tp(threads);
>+  for (int i = 0; i != threads; i++) {
>+    tp.Schedule(std::bind(&EndTest, &c0, &c1, &mu2, &cv2,
>+                          std::function<void(int)>(
>+                              std::bind(test, cxt, std::placeholders::_1))));
>+  }
>+  mu2.Lock();
>+  while (c1 != threads) {
>+    cv2.Wait(&mu2);
>+  }
>+  mu2.Unlock();
>+  return cxt->g0;
>+}
>+
>+// Basis for the parameterized tests configured below.
>+static int RunTest(void (*test)(TestContext *cxt, int), int threads,
>+                   int iterations, int operations) {
>+  TestContext cxt;
>+  return RunTestCommon(&cxt, test, threads, iterations, operations);
>+}
>+
>+// Like RunTest(), but sets an invariant on the tested Mutex and
>+// verifies that the invariant check happened. The invariant function
>+// will be passed the TestContext* as its arg and must call
>+// SetInvariantChecked(true);
>+#if !defined(ABSL_MUTEX_ENABLE_INVARIANT_DEBUGGING_NOT_IMPLEMENTED)
>+static int RunTestWithInvariantDebugging(void (*test)(TestContext *cxt, int),
>+                                         int threads, int iterations,
>+                                         int operations,
>+                                         void (*invariant)(void *)) {
>+  absl::EnableMutexInvariantDebugging(true);
>+  SetInvariantChecked(false);
>+  TestContext cxt;
>+  cxt.mu.EnableInvariantDebugging(invariant, &cxt);
>+  int ret = RunTestCommon(&cxt, test, threads, iterations, operations);
>+  ABSL_RAW_CHECK(GetInvariantChecked(), "Invariant not checked");
>+  absl::EnableMutexInvariantDebugging(false);  // Restore.
>+  return ret;
>+}
>+#endif
>+
>+// --------------------------------------------------------
>+// Test for fix of bug in TryRemove()
>+struct TimeoutBugStruct {
>+  absl::Mutex mu;
>+  bool a;
>+  int a_waiter_count;
>+};
>+
>+static void WaitForA(TimeoutBugStruct *x) {
>+  x->mu.LockWhen(absl::Condition(&x->a));
>+  x->a_waiter_count--;
>+  x->mu.Unlock();
>+}
>+
>+static bool NoAWaiters(TimeoutBugStruct *x) { return x->a_waiter_count == 0; }
>+
>+// Test that a CondVar.Wait(&mutex) can un-block a call to mutex.Await() in
>+// another thread.
>+TEST(Mutex, CondVarWaitSignalsAwait) {
>+  // Use a struct so the lock annotations apply.
>+  struct {
>+    absl::Mutex barrier_mu;
>+    bool barrier GUARDED_BY(barrier_mu) = false;
>+
>+    absl::Mutex release_mu;
>+    bool release GUARDED_BY(release_mu) = false;
>+    absl::CondVar released_cv;
>+  } state;
>+
>+  auto pool = CreateDefaultPool();
>+
>+  // Thread A.  Sets barrier, waits for release using Mutex::Await, then
>+  // signals released_cv.
>+  pool->Schedule([&state] {
>+    state.release_mu.Lock();
>+
>+    state.barrier_mu.Lock();
>+    state.barrier = true;
>+    state.barrier_mu.Unlock();
>+
>+    state.release_mu.Await(absl::Condition(&state.release));
>+    state.released_cv.Signal();
>+    state.release_mu.Unlock();
>+  });
>+
>+  state.barrier_mu.LockWhen(absl::Condition(&state.barrier));
>+  state.barrier_mu.Unlock();
>+  state.release_mu.Lock();
>+  // Thread A is now blocked on release by way of Mutex::Await().
>+
>+  // Set release.  Calling released_cv.Wait() should un-block thread A,
>+  // which will signal released_cv.  If not, the test will hang.
>+  state.release = true;
>+  state.released_cv.Wait(&state.release_mu);
>+  state.release_mu.Unlock();
>+}
>+
>+// Test that a CondVar.WaitWithTimeout(&mutex) can un-block a call to
>+// mutex.Await() in another thread.
>+TEST(Mutex, CondVarWaitWithTimeoutSignalsAwait) {
>+  // Use a struct so the lock annotations apply.
>+  struct {
>+    absl::Mutex barrier_mu;
>+    bool barrier GUARDED_BY(barrier_mu) = false;
>+
>+    absl::Mutex release_mu;
>+    bool release GUARDED_BY(release_mu) = false;
>+    absl::CondVar released_cv;
>+  } state;
>+
>+  auto pool = CreateDefaultPool();
>+
>+  // Thread A.  Sets barrier, waits for release using Mutex::Await, then
>+  // signals released_cv.
>+  pool->Schedule([&state] {
>+    state.release_mu.Lock();
>+
>+    state.barrier_mu.Lock();
>+    state.barrier = true;
>+    state.barrier_mu.Unlock();
>+
>+    state.release_mu.Await(absl::Condition(&state.release));
>+    state.released_cv.Signal();
>+    state.release_mu.Unlock();
>+  });
>+
>+  state.barrier_mu.LockWhen(absl::Condition(&state.barrier));
>+  state.barrier_mu.Unlock();
>+  state.release_mu.Lock();
>+  // Thread A is now blocked on release by way of Mutex::Await().
>+
>+  // Set release.  Calling released_cv.Wait() should un-block thread A,
>+  // which will signal released_cv.  If not, the test will hang.
>+  state.release = true;
>+  EXPECT_TRUE(
>+      !state.released_cv.WaitWithTimeout(&state.release_mu, absl::Seconds(10)))
>+      << "; Unrecoverable test failure: CondVar::WaitWithTimeout did not "
>+         "unblock the absl::Mutex::Await call in another thread.";
>+
>+  state.release_mu.Unlock();
>+}
>+
>+// Test for regression of a bug in loop of TryRemove()
>+TEST(Mutex, MutexTimeoutBug) {
>+  auto tp = CreateDefaultPool();
>+
>+  TimeoutBugStruct x;
>+  x.a = false;
>+  x.a_waiter_count = 2;
>+  tp->Schedule(std::bind(&WaitForA, &x));
>+  tp->Schedule(std::bind(&WaitForA, &x));
>+  absl::SleepFor(absl::Seconds(1));  // Allow first two threads to hang.
>+  // The skip field of the second will point to the first because there are
>+  // only two.
>+
>+  // Now cause a thread waiting on an always-false to time out
>+  // This would deadlock when the bug was present.
>+  bool always_false = false;
>+  x.mu.LockWhenWithTimeout(absl::Condition(&always_false),
>+                           absl::Milliseconds(500));
>+
>+  // if we get here, the bug is not present.   Cleanup the state.
>+
>+  x.a = true;                                    // wakeup the two waiters on A
>+  x.mu.Await(absl::Condition(&NoAWaiters, &x));  // wait for them to exit
>+  x.mu.Unlock();
>+}
>+
>+struct CondVarWaitDeadlock : testing::TestWithParam<int> {
>+  absl::Mutex mu;
>+  absl::CondVar cv;
>+  bool cond1 = false;
>+  bool cond2 = false;
>+  bool read_lock1;
>+  bool read_lock2;
>+  bool signal_unlocked;
>+
>+  CondVarWaitDeadlock() {
>+    read_lock1 = GetParam() & (1 << 0);
>+    read_lock2 = GetParam() & (1 << 1);
>+    signal_unlocked = GetParam() & (1 << 2);
>+  }
>+
>+  void Waiter1() {
>+    if (read_lock1) {
>+      mu.ReaderLock();
>+      while (!cond1) {
>+        cv.Wait(&mu);
>+      }
>+      mu.ReaderUnlock();
>+    } else {
>+      mu.Lock();
>+      while (!cond1) {
>+        cv.Wait(&mu);
>+      }
>+      mu.Unlock();
>+    }
>+  }
>+
>+  void Waiter2() {
>+    if (read_lock2) {
>+      mu.ReaderLockWhen(absl::Condition(&cond2));
>+      mu.ReaderUnlock();
>+    } else {
>+      mu.LockWhen(absl::Condition(&cond2));
>+      mu.Unlock();
>+    }
>+  }
>+};
>+
>+// Test for a deadlock bug in Mutex::Fer().
>+// The sequence of events that lead to the deadlock is:
>+// 1. waiter1 blocks on cv in read mode (mu bits = 0).
>+// 2. waiter2 blocks on mu in either mode (mu bits = kMuWait).
>+// 3. main thread locks mu, sets cond1, unlocks mu (mu bits = kMuWait).
>+// 4. main thread signals on cv and this eventually calls Mutex::Fer().
>+// Currently Fer wakes waiter1 since mu bits = kMuWait (mutex is unlocked).
>+// Before the bug fix Fer neither woke waiter1 nor queued it on mutex,
>+// which resulted in deadlock.
>+TEST_P(CondVarWaitDeadlock, Test) {
>+  auto waiter1 = CreatePool(1);
>+  auto waiter2 = CreatePool(1);
>+  waiter1->Schedule([this] { this->Waiter1(); });
>+  waiter2->Schedule([this] { this->Waiter2(); });
>+
>+  // Wait while threads block (best-effort is fine).
>+  absl::SleepFor(absl::Milliseconds(100));
>+
>+  // Wake condwaiter.
>+  mu.Lock();
>+  cond1 = true;
>+  if (signal_unlocked) {
>+    mu.Unlock();
>+    cv.Signal();
>+  } else {
>+    cv.Signal();
>+    mu.Unlock();
>+  }
>+  waiter1.reset();  // "join" waiter1
>+
>+  // Wake waiter.
>+  mu.Lock();
>+  cond2 = true;
>+  mu.Unlock();
>+  waiter2.reset();  // "join" waiter2
>+}
>+
>+INSTANTIATE_TEST_CASE_P(CondVarWaitDeadlockTest, CondVarWaitDeadlock,
>+                        ::testing::Range(0, 8),
>+                        ::testing::PrintToStringParamName());
>+
>+// --------------------------------------------------------
>+// Test for fix of bug in DequeueAllWakeable()
>+// Bug was that if there was more than one waiting reader
>+// and all should be woken, the most recently blocked one
>+// would not be.
>+
>+struct DequeueAllWakeableBugStruct {
>+  absl::Mutex mu;
>+  absl::Mutex mu2;       // protects all fields below
>+  int unfinished_count;  // count of unfinished readers; under mu2
>+  bool done1;            // unfinished_count == 0; under mu2
>+  int finished_count;    // count of finished readers, under mu2
>+  bool done2;            // finished_count == 0; under mu2
>+};
>+
>+// Test for regression of a bug in loop of DequeueAllWakeable()
>+static void AcquireAsReader(DequeueAllWakeableBugStruct *x) {
>+  x->mu.ReaderLock();
>+  x->mu2.Lock();
>+  x->unfinished_count--;
>+  x->done1 = (x->unfinished_count == 0);
>+  x->mu2.Unlock();
>+  // make sure that both readers acquired mu before we release it.
>+  absl::SleepFor(absl::Seconds(2));
>+  x->mu.ReaderUnlock();
>+
>+  x->mu2.Lock();
>+  x->finished_count--;
>+  x->done2 = (x->finished_count == 0);
>+  x->mu2.Unlock();
>+}
>+
>+// Test for regression of a bug in loop of DequeueAllWakeable()
>+TEST(Mutex, MutexReaderWakeupBug) {
>+  auto tp = CreateDefaultPool();
>+
>+  DequeueAllWakeableBugStruct x;
>+  x.unfinished_count = 2;
>+  x.done1 = false;
>+  x.finished_count = 2;
>+  x.done2 = false;
>+  x.mu.Lock();  // acquire mu exclusively
>+  // queue two thread that will block on reader locks on x.mu
>+  tp->Schedule(std::bind(&AcquireAsReader, &x));
>+  tp->Schedule(std::bind(&AcquireAsReader, &x));
>+  absl::SleepFor(absl::Seconds(1));  // give time for reader threads to block
>+  x.mu.Unlock();                     // wake them up
>+
>+  // both readers should finish promptly
>+  EXPECT_TRUE(
>+      x.mu2.LockWhenWithTimeout(absl::Condition(&x.done1), absl::Seconds(10)));
>+  x.mu2.Unlock();
>+
>+  EXPECT_TRUE(
>+      x.mu2.LockWhenWithTimeout(absl::Condition(&x.done2), absl::Seconds(10)));
>+  x.mu2.Unlock();
>+}
>+
>+struct LockWhenTestStruct {
>+  absl::Mutex mu1;
>+  bool cond = false;
>+
>+  absl::Mutex mu2;
>+  bool waiting = false;
>+};
>+
>+static bool LockWhenTestIsCond(LockWhenTestStruct* s) {
>+  s->mu2.Lock();
>+  s->waiting = true;
>+  s->mu2.Unlock();
>+  return s->cond;
>+}
>+
>+static void LockWhenTestWaitForIsCond(LockWhenTestStruct* s) {
>+  s->mu1.LockWhen(absl::Condition(&LockWhenTestIsCond, s));
>+  s->mu1.Unlock();
>+}
>+
>+TEST(Mutex, LockWhen) {
>+  LockWhenTestStruct s;
>+
>+  std::thread t(LockWhenTestWaitForIsCond, &s);
>+  s.mu2.LockWhen(absl::Condition(&s.waiting));
>+  s.mu2.Unlock();
>+
>+  s.mu1.Lock();
>+  s.cond = true;
>+  s.mu1.Unlock();
>+
>+  t.join();
>+}
>+
>+// --------------------------------------------------------
>+// The following test requires Mutex::ReaderLock to be a real shared
>+// lock, which is not the case in all builds.
>+#if !defined(ABSL_MUTEX_READER_LOCK_IS_EXCLUSIVE)
>+
>+// Test for fix of bug in UnlockSlow() that incorrectly decremented the reader
>+// count when putting a thread to sleep waiting for a false condition when the
>+// lock was not held.
>+
>+// For this bug to strike, we make a thread wait on a free mutex with no
>+// waiters by causing its wakeup condition to be false.   Then the
>+// next two acquirers must be readers.   The bug causes the lock
>+// to be released when one reader unlocks, rather than both.
>+
>+struct ReaderDecrementBugStruct {
>+  bool cond;  // to delay first thread (under mu)
>+  int done;   // reference count (under mu)
>+  absl::Mutex mu;
>+
>+  bool waiting_on_cond;   // under mu2
>+  bool have_reader_lock;  // under mu2
>+  bool complete;          // under mu2
>+  absl::Mutex mu2;        // > mu
>+};
>+
>+// L >= mu, L < mu_waiting_on_cond
>+static bool IsCond(void *v) {
>+  ReaderDecrementBugStruct *x = reinterpret_cast<ReaderDecrementBugStruct *>(v);
>+  x->mu2.Lock();
>+  x->waiting_on_cond = true;
>+  x->mu2.Unlock();
>+  return x->cond;
>+}
>+
>+// L >= mu
>+static bool AllDone(void *v) {
>+  ReaderDecrementBugStruct *x = reinterpret_cast<ReaderDecrementBugStruct *>(v);
>+  return x->done == 0;
>+}
>+
>+// L={}
>+static void WaitForCond(ReaderDecrementBugStruct *x) {
>+  absl::Mutex dummy;
>+  absl::MutexLock l(&dummy);
>+  x->mu.LockWhen(absl::Condition(&IsCond, x));
>+  x->done--;
>+  x->mu.Unlock();
>+}
>+
>+// L={}
>+static void GetReadLock(ReaderDecrementBugStruct *x) {
>+  x->mu.ReaderLock();
>+  x->mu2.Lock();
>+  x->have_reader_lock = true;
>+  x->mu2.Await(absl::Condition(&x->complete));
>+  x->mu2.Unlock();
>+  x->mu.ReaderUnlock();
>+  x->mu.Lock();
>+  x->done--;
>+  x->mu.Unlock();
>+}
>+
>+// Test for reader counter being decremented incorrectly by waiter
>+// with false condition.
>+TEST(Mutex, MutexReaderDecrementBug) NO_THREAD_SAFETY_ANALYSIS {
>+  ReaderDecrementBugStruct x;
>+  x.cond = false;
>+  x.waiting_on_cond = false;
>+  x.have_reader_lock = false;
>+  x.complete = false;
>+  x.done = 2;  // initial ref count
>+
>+  // Run WaitForCond() and wait for it to sleep
>+  std::thread thread1(WaitForCond, &x);
>+  x.mu2.LockWhen(absl::Condition(&x.waiting_on_cond));
>+  x.mu2.Unlock();
>+
>+  // Run GetReadLock(), and wait for it to get the read lock
>+  std::thread thread2(GetReadLock, &x);
>+  x.mu2.LockWhen(absl::Condition(&x.have_reader_lock));
>+  x.mu2.Unlock();
>+
>+  // Get the reader lock ourselves, and release it.
>+  x.mu.ReaderLock();
>+  x.mu.ReaderUnlock();
>+
>+  // The lock should be held in read mode by GetReadLock().
>+  // If we have the bug, the lock will be free.
>+  x.mu.AssertReaderHeld();
>+
>+  // Wake up all the threads.
>+  x.mu2.Lock();
>+  x.complete = true;
>+  x.mu2.Unlock();
>+
>+  // TODO(delesley): turn on analysis once lock upgrading is supported.
>+  // (This call upgrades the lock from shared to exclusive.)
>+  x.mu.Lock();
>+  x.cond = true;
>+  x.mu.Await(absl::Condition(&AllDone, &x));
>+  x.mu.Unlock();
>+
>+  thread1.join();
>+  thread2.join();
>+}
>+#endif  // !ABSL_MUTEX_READER_LOCK_IS_EXCLUSIVE
>+
>+// Test that we correctly handle the situation when a lock is
>+// held and then destroyed (w/o unlocking).
>+TEST(Mutex, LockedMutexDestructionBug) NO_THREAD_SAFETY_ANALYSIS {
>+  for (int i = 0; i != 10; i++) {
>+    // Create, lock and destroy 10 locks.
>+    const int kNumLocks = 10;
>+    auto mu = absl::make_unique<absl::Mutex[]>(kNumLocks);
>+    for (int j = 0; j != kNumLocks; j++) {
>+      if ((j % 2) == 0) {
>+        mu[j].WriterLock();
>+      } else {
>+        mu[j].ReaderLock();
>+      }
>+    }
>+  }
>+}
>+
>+// --------------------------------------------------------
>+// Test for bug with pattern of readers using a condvar.  The bug was that if a
>+// reader went to sleep on a condition variable while one or more other readers
>+// held the lock, but there were no waiters, the reader count (held in the
>+// mutex word) would be lost.  (This is because Enqueue() had at one time
>+// always placed the thread on the Mutex queue.  Later (CL 4075610), to
>+// tolerate re-entry into Mutex from a Condition predicate, Enqueue() was
>+// changed so that it could also place a thread on a condition-variable.  This
>+// introduced the case where Enqueue() returned with an empty queue, and this
>+// case was handled incorrectly in one place.)
>+
>+static void ReaderForReaderOnCondVar(absl::Mutex *mu, absl::CondVar *cv,
>+                                     int *running) {
>+  std::random_device dev;
>+  std::mt19937 gen(dev());
>+  std::uniform_int_distribution<int> random_millis(0, 15);
>+  mu->ReaderLock();
>+  while (*running == 3) {
>+    absl::SleepFor(absl::Milliseconds(random_millis(gen)));
>+    cv->WaitWithTimeout(mu, absl::Milliseconds(random_millis(gen)));
>+  }
>+  mu->ReaderUnlock();
>+  mu->Lock();
>+  (*running)--;
>+  mu->Unlock();
>+}
>+
>+struct True {
>+  template <class... Args>
>+  bool operator()(Args...) const {
>+    return true;
>+  }
>+};
>+
>+struct DerivedTrue : True {};
>+
>+TEST(Mutex, FunctorCondition) {
>+  {  // Variadic
>+    True f;
>+    EXPECT_TRUE(absl::Condition(&f).Eval());
>+  }
>+
>+  {  // Inherited
>+    DerivedTrue g;
>+    EXPECT_TRUE(absl::Condition(&g).Eval());
>+  }
>+
>+  {  // lambda
>+    int value = 3;
>+    auto is_zero = [&value] { return value == 0; };
>+    absl::Condition c(&is_zero);
>+    EXPECT_FALSE(c.Eval());
>+    value = 0;
>+    EXPECT_TRUE(c.Eval());
>+  }
>+
>+  {  // bind
>+    int value = 0;
>+    auto is_positive = std::bind(std::less<int>(), 0, std::cref(value));
>+    absl::Condition c(&is_positive);
>+    EXPECT_FALSE(c.Eval());
>+    value = 1;
>+    EXPECT_TRUE(c.Eval());
>+  }
>+
>+  {  // std::function
>+    int value = 3;
>+    std::function<bool()> is_zero = [&value] { return value == 0; };
>+    absl::Condition c(&is_zero);
>+    EXPECT_FALSE(c.Eval());
>+    value = 0;
>+    EXPECT_TRUE(c.Eval());
>+  }
>+}
>+
>+static bool IntIsZero(int *x) { return *x == 0; }
>+
>+// Test for reader waiting condition variable when there are other readers
>+// but no waiters.
>+TEST(Mutex, TestReaderOnCondVar) {
>+  auto tp = CreateDefaultPool();
>+  absl::Mutex mu;
>+  absl::CondVar cv;
>+  int running = 3;
>+  tp->Schedule(std::bind(&ReaderForReaderOnCondVar, &mu, &cv, &running));
>+  tp->Schedule(std::bind(&ReaderForReaderOnCondVar, &mu, &cv, &running));
>+  absl::SleepFor(absl::Seconds(2));
>+  mu.Lock();
>+  running--;
>+  mu.Await(absl::Condition(&IntIsZero, &running));
>+  mu.Unlock();
>+}
>+
>+// --------------------------------------------------------
>+struct AcquireFromConditionStruct {
>+  absl::Mutex mu0;   // protects value, done
>+  int value;         // times condition function is called; under mu0,
>+  bool done;         // done with test?  under mu0
>+  absl::Mutex mu1;   // used to attempt to mess up state of mu0
>+  absl::CondVar cv;  // so the condition function can be invoked from
>+                     // CondVar::Wait().
>+};
>+
>+static bool ConditionWithAcquire(AcquireFromConditionStruct *x) {
>+  x->value++;  // count times this function is called
>+
>+  if (x->value == 2 || x->value == 3) {
>+    // On the second and third invocation of this function, sleep for 100ms,
>+    // but with the side-effect of altering the state of a Mutex other than
>+    // than one for which this is a condition.  The spec now explicitly allows
>+    // this side effect; previously it did not.  it was illegal.
>+    bool always_false = false;
>+    x->mu1.LockWhenWithTimeout(absl::Condition(&always_false),
>+                               absl::Milliseconds(100));
>+    x->mu1.Unlock();
>+  }
>+  ABSL_RAW_CHECK(x->value < 4, "should not be invoked a fourth time");
>+
>+  // We arrange for the condition to return true on only the 2nd and 3rd calls.
>+  return x->value == 2 || x->value == 3;
>+}
>+
>+static void WaitForCond2(AcquireFromConditionStruct *x) {
>+  // wait for cond0 to become true
>+  x->mu0.LockWhen(absl::Condition(&ConditionWithAcquire, x));
>+  x->done = true;
>+  x->mu0.Unlock();
>+}
>+
>+// Test for Condition whose function acquires other Mutexes
>+TEST(Mutex, AcquireFromCondition) {
>+  auto tp = CreateDefaultPool();
>+
>+  AcquireFromConditionStruct x;
>+  x.value = 0;
>+  x.done = false;
>+  tp->Schedule(
>+      std::bind(&WaitForCond2, &x));  // run WaitForCond2() in a thread T
>+  // T will hang because the first invocation of ConditionWithAcquire() will
>+  // return false.
>+  absl::SleepFor(absl::Milliseconds(500));  // allow T time to hang
>+
>+  x.mu0.Lock();
>+  x.cv.WaitWithTimeout(&x.mu0, absl::Milliseconds(500));  // wake T
>+  // T will be woken because the Wait() will call ConditionWithAcquire()
>+  // for the second time, and it will return true.
>+
>+  x.mu0.Unlock();
>+
>+  // T will then acquire the lock and recheck its own condition.
>+  // It will find the condition true, as this is the third invocation,
>+  // but the use of another Mutex by the calling function will
>+  // cause the old mutex implementation to think that the outer
>+  // LockWhen() has timed out because the inner LockWhenWithTimeout() did.
>+  // T will then check the condition a fourth time because it finds a
>+  // timeout occurred.  This should not happen in the new
>+  // implementation that allows the Condition function to use Mutexes.
>+
>+  // It should also succeed, even though the Condition function
>+  // is being invoked from CondVar::Wait, and thus this thread
>+  // is conceptually waiting both on the condition variable, and on mu2.
>+
>+  x.mu0.LockWhen(absl::Condition(&x.done));
>+  x.mu0.Unlock();
>+}
>+
>+// The deadlock detector is not part of non-prod builds, so do not test it.
>+#if !defined(ABSL_INTERNAL_USE_NONPROD_MUTEX)
>+
>+TEST(Mutex, DeadlockDetector) {
>+  absl::SetMutexDeadlockDetectionMode(absl::OnDeadlockCycle::kAbort);
>+
>+  // check that we can call ForgetDeadlockInfo() on a lock with the lock held
>+  absl::Mutex m1;
>+  absl::Mutex m2;
>+  absl::Mutex m3;
>+  absl::Mutex m4;
>+
>+  m1.Lock();  // m1 gets ID1
>+  m2.Lock();  // m2 gets ID2
>+  m3.Lock();  // m3 gets ID3
>+  m3.Unlock();
>+  m2.Unlock();
>+  // m1 still held
>+  m1.ForgetDeadlockInfo();  // m1 loses ID
>+  m2.Lock();                // m2 gets ID2
>+  m3.Lock();                // m3 gets ID3
>+  m4.Lock();                // m4 gets ID4
>+  m3.Unlock();
>+  m2.Unlock();
>+  m4.Unlock();
>+  m1.Unlock();
>+}
>+
>+// Bazel has a test "warning" file that programs can write to if the
>+// test should pass with a warning.  This class disables the warning
>+// file until it goes out of scope.
>+class ScopedDisableBazelTestWarnings {
>+ public:
>+  ScopedDisableBazelTestWarnings() {
>+#ifdef WIN32
>+    char file[MAX_PATH];
>+    if (GetEnvironmentVariable(kVarName, file, sizeof(file)) < sizeof(file)) {
>+      warnings_output_file_ = file;
>+      SetEnvironmentVariable(kVarName, nullptr);
>+    }
>+#else
>+    const char *file = getenv(kVarName);
>+    if (file != nullptr) {
>+      warnings_output_file_ = file;
>+      unsetenv(kVarName);
>+    }
>+#endif
>+  }
>+
>+  ~ScopedDisableBazelTestWarnings() {
>+    if (!warnings_output_file_.empty()) {
>+#ifdef WIN32
>+      SetEnvironmentVariable(kVarName, warnings_output_file_.c_str());
>+#else
>+      setenv(kVarName, warnings_output_file_.c_str(), 0);
>+#endif
>+    }
>+  }
>+
>+ private:
>+  static const char kVarName[];
>+  std::string warnings_output_file_;
>+};
>+const char ScopedDisableBazelTestWarnings::kVarName[] =
>+    "TEST_WARNINGS_OUTPUT_FILE";
>+
>+TEST(Mutex, DeadlockDetectorBazelWarning) {
>+  absl::SetMutexDeadlockDetectionMode(absl::OnDeadlockCycle::kReport);
>+
>+  // Cause deadlock detection to detect something, if it's
>+  // compiled in and enabled.  But turn off the bazel warning.
>+  ScopedDisableBazelTestWarnings disable_bazel_test_warnings;
>+
>+  absl::Mutex mu0;
>+  absl::Mutex mu1;
>+  bool got_mu0 = mu0.TryLock();
>+  mu1.Lock();  // acquire mu1 while holding mu0
>+  if (got_mu0) {
>+    mu0.Unlock();
>+  }
>+  if (mu0.TryLock()) {  // try lock shouldn't cause deadlock detector to fire
>+    mu0.Unlock();
>+  }
>+  mu0.Lock();  // acquire mu0 while holding mu1; should get one deadlock
>+               // report here
>+  mu0.Unlock();
>+  mu1.Unlock();
>+
>+  absl::SetMutexDeadlockDetectionMode(absl::OnDeadlockCycle::kAbort);
>+}
>+
>+// This test is tagged with NO_THREAD_SAFETY_ANALYSIS because the
>+// annotation-based static thread-safety analysis is not currently
>+// predicate-aware and cannot tell if the two for-loops that acquire and
>+// release the locks have the same predicates.
>+TEST(Mutex, DeadlockDetectorStessTest) NO_THREAD_SAFETY_ANALYSIS {
>+  // Stress test: Here we create a large number of locks and use all of them.
>+  // If a deadlock detector keeps a full graph of lock acquisition order,
>+  // it will likely be too slow for this test to pass.
>+  const int n_locks = 1 << 17;
>+  auto array_of_locks = absl::make_unique<absl::Mutex[]>(n_locks);
>+  for (int i = 0; i < n_locks; i++) {
>+    int end = std::min(n_locks, i + 5);
>+    // acquire and then release locks i, i+1, ..., i+4
>+    for (int j = i; j < end; j++) {
>+      array_of_locks[j].Lock();
>+    }
>+    for (int j = i; j < end; j++) {
>+      array_of_locks[j].Unlock();
>+    }
>+  }
>+}
>+
>+TEST(Mutex, DeadlockIdBug) NO_THREAD_SAFETY_ANALYSIS {
>+  // Test a scenario where a cached deadlock graph node id in the
>+  // list of held locks is not invalidated when the corresponding
>+  // mutex is deleted.
>+  absl::SetMutexDeadlockDetectionMode(absl::OnDeadlockCycle::kAbort);
>+  // Mutex that will be destroyed while being held
>+  absl::Mutex *a = new absl::Mutex;
>+  // Other mutexes needed by test
>+  absl::Mutex b, c;
>+
>+  // Hold mutex.
>+  a->Lock();
>+
>+  // Force deadlock id assignment by acquiring another lock.
>+  b.Lock();
>+  b.Unlock();
>+
>+  // Delete the mutex. The Mutex destructor tries to remove held locks,
>+  // but the attempt isn't foolproof.  It can fail if:
>+  //   (a) Deadlock detection is currently disabled.
>+  //   (b) The destruction is from another thread.
>+  // We exploit (a) by temporarily disabling deadlock detection.
>+  absl::SetMutexDeadlockDetectionMode(absl::OnDeadlockCycle::kIgnore);
>+  delete a;
>+  absl::SetMutexDeadlockDetectionMode(absl::OnDeadlockCycle::kAbort);
>+
>+  // Now acquire another lock which will force a deadlock id assignment.
>+  // We should end up getting assigned the same deadlock id that was
>+  // freed up when "a" was deleted, which will cause a spurious deadlock
>+  // report if the held lock entry for "a" was not invalidated.
>+  c.Lock();
>+  c.Unlock();
>+}
>+#endif  // !defined(ABSL_INTERNAL_USE_NONPROD_MUTEX)
>+
>+// --------------------------------------------------------
>+// Test for timeouts/deadlines on condition waits that are specified using
>+// absl::Duration and absl::Time.  For each waiting function we test with
>+// a timeout/deadline that has already expired/passed, one that is infinite
>+// and so never expires/passes, and one that will expire/pass in the near
>+// future.
>+
>+static absl::Duration TimeoutTestAllowedSchedulingDelay() {
>+  // Note: we use a function here because Microsoft Visual Studio fails to
>+  // properly initialize constexpr static absl::Duration variables.
>+  return absl::Milliseconds(150);
>+}
>+
>+// Returns true if `actual_delay` is close enough to `expected_delay` to pass
>+// the timeouts/deadlines test.  Otherwise, logs warnings and returns false.
>+ABSL_MUST_USE_RESULT
>+static bool DelayIsWithinBounds(absl::Duration expected_delay,
>+                                absl::Duration actual_delay) {
>+  bool pass = true;
>+  // Do not allow the observed delay to be less than expected.  This may occur
>+  // in practice due to clock skew or when the synchronization primitives use a
>+  // different clock than absl::Now(), but these cases should be handled by the
>+  // the retry mechanism in each TimeoutTest.
>+  if (actual_delay < expected_delay) {
>+    ABSL_RAW_LOG(WARNING,
>+                 "Actual delay %s was too short, expected %s (difference %s)",
>+                 absl::FormatDuration(actual_delay).c_str(),
>+                 absl::FormatDuration(expected_delay).c_str(),
>+                 absl::FormatDuration(actual_delay - expected_delay).c_str());
>+    pass = false;
>+  }
>+  // If the expected delay is <= zero then allow a small error tolerance, since
>+  // we do not expect context switches to occur during test execution.
>+  // Otherwise, thread scheduling delays may be substantial in rare cases, so
>+  // tolerate up to kTimeoutTestAllowedSchedulingDelay of error.
>+  absl::Duration tolerance = expected_delay <= absl::ZeroDuration()
>+                                 ? absl::Milliseconds(10)
>+                                 : TimeoutTestAllowedSchedulingDelay();
>+  if (actual_delay > expected_delay + tolerance) {
>+    ABSL_RAW_LOG(WARNING,
>+                 "Actual delay %s was too long, expected %s (difference %s)",
>+                 absl::FormatDuration(actual_delay).c_str(),
>+                 absl::FormatDuration(expected_delay).c_str(),
>+                 absl::FormatDuration(actual_delay - expected_delay).c_str());
>+    pass = false;
>+  }
>+  return pass;
>+}
>+
>+// Parameters for TimeoutTest, below.
>+struct TimeoutTestParam {
>+  // The file and line number (used for logging purposes only).
>+  const char *from_file;
>+  int from_line;
>+
>+  // Should the absolute deadline API based on absl::Time be tested?  If false,
>+  // the relative deadline API based on absl::Duration is tested.
>+  bool use_absolute_deadline;
>+
>+  // The deadline/timeout used when calling the API being tested
>+  // (e.g. Mutex::LockWhenWithDeadline).
>+  absl::Duration wait_timeout;
>+
>+  // The delay before the condition will be set true by the test code.  If zero
>+  // or negative, the condition is set true immediately (before calling the API
>+  // being tested).  Otherwise, if infinite, the condition is never set true.
>+  // Otherwise a closure is scheduled for the future that sets the condition
>+  // true.
>+  absl::Duration satisfy_condition_delay;
>+
>+  // The expected result of the condition after the call to the API being
>+  // tested. Generally `true` means the condition was true when the API returns,
>+  // `false` indicates an expected timeout.
>+  bool expected_result;
>+
>+  // The expected delay before the API under test returns.  This is inherently
>+  // flaky, so some slop is allowed (see `DelayIsWithinBounds` above), and the
>+  // test keeps trying indefinitely until this constraint passes.
>+  absl::Duration expected_delay;
>+};
>+
>+// Print a `TimeoutTestParam` to a debug log.
>+std::ostream &operator<<(std::ostream &os, const TimeoutTestParam &param) {
>+  return os << "from: " << param.from_file << ":" << param.from_line
>+            << " use_absolute_deadline: "
>+            << (param.use_absolute_deadline ? "true" : "false")
>+            << " wait_timeout: " << param.wait_timeout
>+            << " satisfy_condition_delay: " << param.satisfy_condition_delay
>+            << " expected_result: "
>+            << (param.expected_result ? "true" : "false")
>+            << " expected_delay: " << param.expected_delay;
>+}
>+
>+std::string FormatString(const TimeoutTestParam &param) {
>+  std::ostringstream os;
>+  os << param;
>+  return os.str();
>+}
>+
>+// Like `thread::Executor::ScheduleAt` except:
>+// a) Delays zero or negative are executed immediately in the current thread.
>+// b) Infinite delays are never scheduled.
>+// c) Calls this test's `ScheduleAt` helper instead of using `pool` directly.
>+static void RunAfterDelay(absl::Duration delay,
>+                          absl::synchronization_internal::ThreadPool *pool,
>+                          const std::function<void()> &callback) {
>+  if (delay <= absl::ZeroDuration()) {
>+    callback();  // immediate
>+  } else if (delay != absl::InfiniteDuration()) {
>+    ScheduleAfter(pool, delay, callback);
>+  }
>+}
>+
>+class TimeoutTest : public ::testing::Test,
>+                    public ::testing::WithParamInterface<TimeoutTestParam> {};
>+
>+std::vector<TimeoutTestParam> MakeTimeoutTestParamValues() {
>+  // The `finite` delay is a finite, relatively short, delay.  We make it larger
>+  // than our allowed scheduling delay (slop factor) to avoid confusion when
>+  // diagnosing test failures.  The other constants here have clear meanings.
>+  const absl::Duration finite = 3 * TimeoutTestAllowedSchedulingDelay();
>+  const absl::Duration never = absl::InfiniteDuration();
>+  const absl::Duration negative = -absl::InfiniteDuration();
>+  const absl::Duration immediate = absl::ZeroDuration();
>+
>+  // Every test case is run twice; once using the absolute deadline API and once
>+  // using the relative timeout API.
>+  std::vector<TimeoutTestParam> values;
>+  for (bool use_absolute_deadline : {false, true}) {
>+    // Tests with a negative timeout (deadline in the past), which should
>+    // immediately return current state of the condition.
>+
>+    // The condition is already true:
>+    values.push_back(TimeoutTestParam{
>+        __FILE__, __LINE__, use_absolute_deadline,
>+        negative,   // wait_timeout
>+        immediate,  // satisfy_condition_delay
>+        true,       // expected_result
>+        immediate,  // expected_delay
>+    });
>+
>+    // The condition becomes true, but the timeout has already expired:
>+    values.push_back(TimeoutTestParam{
>+        __FILE__, __LINE__, use_absolute_deadline,
>+        negative,  // wait_timeout
>+        finite,    // satisfy_condition_delay
>+        false,     // expected_result
>+        immediate  // expected_delay
>+    });
>+
>+    // The condition never becomes true:
>+    values.push_back(TimeoutTestParam{
>+        __FILE__, __LINE__, use_absolute_deadline,
>+        negative,  // wait_timeout
>+        never,     // satisfy_condition_delay
>+        false,     // expected_result
>+        immediate  // expected_delay
>+    });
>+
>+    // Tests with an infinite timeout (deadline in the infinite future), which
>+    // should only return when the condition becomes true.
>+
>+    // The condition is already true:
>+    values.push_back(TimeoutTestParam{
>+        __FILE__, __LINE__, use_absolute_deadline,
>+        never,      // wait_timeout
>+        immediate,  // satisfy_condition_delay
>+        true,       // expected_result
>+        immediate   // expected_delay
>+    });
>+
>+    // The condition becomes true before the (infinite) expiry:
>+    values.push_back(TimeoutTestParam{
>+        __FILE__, __LINE__, use_absolute_deadline,
>+        never,   // wait_timeout
>+        finite,  // satisfy_condition_delay
>+        true,    // expected_result
>+        finite,  // expected_delay
>+    });
>+
>+    // Tests with a (small) finite timeout (deadline soon), with the condition
>+    // becoming true both before and after its expiry.
>+
>+    // The condition is already true:
>+    values.push_back(TimeoutTestParam{
>+        __FILE__, __LINE__, use_absolute_deadline,
>+        never,      // wait_timeout
>+        immediate,  // satisfy_condition_delay
>+        true,       // expected_result
>+        immediate   // expected_delay
>+    });
>+
>+    // The condition becomes true before the expiry:
>+    values.push_back(TimeoutTestParam{
>+        __FILE__, __LINE__, use_absolute_deadline,
>+        finite * 2,  // wait_timeout
>+        finite,      // satisfy_condition_delay
>+        true,        // expected_result
>+        finite       // expected_delay
>+    });
>+
>+    // The condition becomes true, but the timeout has already expired:
>+    values.push_back(TimeoutTestParam{
>+        __FILE__, __LINE__, use_absolute_deadline,
>+        finite,      // wait_timeout
>+        finite * 2,  // satisfy_condition_delay
>+        false,       // expected_result
>+        finite       // expected_delay
>+    });
>+
>+    // The condition never becomes true:
>+    values.push_back(TimeoutTestParam{
>+        __FILE__, __LINE__, use_absolute_deadline,
>+        finite,  // wait_timeout
>+        never,   // satisfy_condition_delay
>+        false,   // expected_result
>+        finite   // expected_delay
>+    });
>+  }
>+  return values;
>+}
>+
>+// Instantiate `TimeoutTest` with `MakeTimeoutTestParamValues()`.
>+INSTANTIATE_TEST_CASE_P(All, TimeoutTest,
>+                        testing::ValuesIn(MakeTimeoutTestParamValues()));
>+
>+TEST_P(TimeoutTest, Await) {
>+  const TimeoutTestParam params = GetParam();
>+  ABSL_RAW_LOG(INFO, "Params: %s", FormatString(params).c_str());
>+
>+  // Because this test asserts bounds on scheduling delays it is flaky.  To
>+  // compensate it loops forever until it passes.  Failures express as test
>+  // timeouts, in which case the test log can be used to diagnose the issue.
>+  for (int attempt = 1;; ++attempt) {
>+    ABSL_RAW_LOG(INFO, "Attempt %d", attempt);
>+
>+    absl::Mutex mu;
>+    bool value = false;  // condition value (under mu)
>+
>+    std::unique_ptr<absl::synchronization_internal::ThreadPool> pool =
>+        CreateDefaultPool();
>+    RunAfterDelay(params.satisfy_condition_delay, pool.get(), [&] {
>+      absl::MutexLock l(&mu);
>+      value = true;
>+    });
>+
>+    absl::MutexLock lock(&mu);
>+    absl::Time start_time = absl::Now();
>+    absl::Condition cond(&value);
>+    bool result =
>+        params.use_absolute_deadline
>+            ? mu.AwaitWithDeadline(cond, start_time + params.wait_timeout)
>+            : mu.AwaitWithTimeout(cond, params.wait_timeout);
>+    if (DelayIsWithinBounds(params.expected_delay, absl::Now() - start_time)) {
>+      EXPECT_EQ(params.expected_result, result);
>+      break;
>+    }
>+  }
>+}
>+
>+TEST_P(TimeoutTest, LockWhen) {
>+  const TimeoutTestParam params = GetParam();
>+  ABSL_RAW_LOG(INFO, "Params: %s", FormatString(params).c_str());
>+
>+  // Because this test asserts bounds on scheduling delays it is flaky.  To
>+  // compensate it loops forever until it passes.  Failures express as test
>+  // timeouts, in which case the test log can be used to diagnose the issue.
>+  for (int attempt = 1;; ++attempt) {
>+    ABSL_RAW_LOG(INFO, "Attempt %d", attempt);
>+
>+    absl::Mutex mu;
>+    bool value = false;  // condition value (under mu)
>+
>+    std::unique_ptr<absl::synchronization_internal::ThreadPool> pool =
>+        CreateDefaultPool();
>+    RunAfterDelay(params.satisfy_condition_delay, pool.get(), [&] {
>+      absl::MutexLock l(&mu);
>+      value = true;
>+    });
>+
>+    absl::Time start_time = absl::Now();
>+    absl::Condition cond(&value);
>+    bool result =
>+        params.use_absolute_deadline
>+            ? mu.LockWhenWithDeadline(cond, start_time + params.wait_timeout)
>+            : mu.LockWhenWithTimeout(cond, params.wait_timeout);
>+    mu.Unlock();
>+
>+    if (DelayIsWithinBounds(params.expected_delay, absl::Now() - start_time)) {
>+      EXPECT_EQ(params.expected_result, result);
>+      break;
>+    }
>+  }
>+}
>+
>+TEST_P(TimeoutTest, ReaderLockWhen) {
>+  const TimeoutTestParam params = GetParam();
>+  ABSL_RAW_LOG(INFO, "Params: %s", FormatString(params).c_str());
>+
>+  // Because this test asserts bounds on scheduling delays it is flaky.  To
>+  // compensate it loops forever until it passes.  Failures express as test
>+  // timeouts, in which case the test log can be used to diagnose the issue.
>+  for (int attempt = 0;; ++attempt) {
>+    ABSL_RAW_LOG(INFO, "Attempt %d", attempt);
>+
>+    absl::Mutex mu;
>+    bool value = false;  // condition value (under mu)
>+
>+    std::unique_ptr<absl::synchronization_internal::ThreadPool> pool =
>+        CreateDefaultPool();
>+    RunAfterDelay(params.satisfy_condition_delay, pool.get(), [&] {
>+      absl::MutexLock l(&mu);
>+      value = true;
>+    });
>+
>+    absl::Time start_time = absl::Now();
>+    bool result =
>+        params.use_absolute_deadline
>+            ? mu.ReaderLockWhenWithDeadline(absl::Condition(&value),
>+                                            start_time + params.wait_timeout)
>+            : mu.ReaderLockWhenWithTimeout(absl::Condition(&value),
>+                                           params.wait_timeout);
>+    mu.ReaderUnlock();
>+
>+    if (DelayIsWithinBounds(params.expected_delay, absl::Now() - start_time)) {
>+      EXPECT_EQ(params.expected_result, result);
>+      break;
>+    }
>+  }
>+}
>+
>+TEST_P(TimeoutTest, Wait) {
>+  const TimeoutTestParam params = GetParam();
>+  ABSL_RAW_LOG(INFO, "Params: %s", FormatString(params).c_str());
>+
>+  // Because this test asserts bounds on scheduling delays it is flaky.  To
>+  // compensate it loops forever until it passes.  Failures express as test
>+  // timeouts, in which case the test log can be used to diagnose the issue.
>+  for (int attempt = 0;; ++attempt) {
>+    ABSL_RAW_LOG(INFO, "Attempt %d", attempt);
>+
>+    absl::Mutex mu;
>+    bool value = false;  // condition value (under mu)
>+    absl::CondVar cv;    // signals a change of `value`
>+
>+    std::unique_ptr<absl::synchronization_internal::ThreadPool> pool =
>+        CreateDefaultPool();
>+    RunAfterDelay(params.satisfy_condition_delay, pool.get(), [&] {
>+      absl::MutexLock l(&mu);
>+      value = true;
>+      cv.Signal();
>+    });
>+
>+    absl::MutexLock lock(&mu);
>+    absl::Time start_time = absl::Now();
>+    absl::Duration timeout = params.wait_timeout;
>+    absl::Time deadline = start_time + timeout;
>+    while (!value) {
>+      if (params.use_absolute_deadline ? cv.WaitWithDeadline(&mu, deadline)
>+                                       : cv.WaitWithTimeout(&mu, timeout)) {
>+        break;  // deadline/timeout exceeded
>+      }
>+      timeout = deadline - absl::Now();  // recompute
>+    }
>+    bool result = value;  // note: `mu` is still held
>+
>+    if (DelayIsWithinBounds(params.expected_delay, absl::Now() - start_time)) {
>+      EXPECT_EQ(params.expected_result, result);
>+      break;
>+    }
>+  }
>+}
>+
>+TEST(Mutex, Logging) {
>+  // Allow user to look at logging output
>+  absl::Mutex logged_mutex;
>+  logged_mutex.EnableDebugLog("fido_mutex");
>+  absl::CondVar logged_cv;
>+  logged_cv.EnableDebugLog("rover_cv");
>+  logged_mutex.Lock();
>+  logged_cv.WaitWithTimeout(&logged_mutex, absl::Milliseconds(20));
>+  logged_mutex.Unlock();
>+  logged_mutex.ReaderLock();
>+  logged_mutex.ReaderUnlock();
>+  logged_mutex.Lock();
>+  logged_mutex.Unlock();
>+  logged_cv.Signal();
>+  logged_cv.SignalAll();
>+}
>+
>+// --------------------------------------------------------
>+
>+// Generate the vector of thread counts for tests parameterized on thread count.
>+static std::vector<int> AllThreadCountValues() {
>+  if (kExtendedTest) {
>+    return {2, 4, 8, 10, 16, 20, 24, 30, 32};
>+  }
>+  return {2, 4, 10};
>+}
>+
>+// A test fixture parameterized by thread count.
>+class MutexVariableThreadCountTest : public ::testing::TestWithParam<int> {};
>+
>+// Instantiate the above with AllThreadCountOptions().
>+INSTANTIATE_TEST_CASE_P(ThreadCounts, MutexVariableThreadCountTest,
>+                        ::testing::ValuesIn(AllThreadCountValues()),
>+                        ::testing::PrintToStringParamName());
>+
>+// Reduces iterations by some factor for slow platforms
>+// (determined empirically).
>+static int ScaleIterations(int x) {
>+  // ABSL_MUTEX_READER_LOCK_IS_EXCLUSIVE is set in the implementation
>+  // of Mutex that uses either std::mutex or pthread_mutex_t. Use
>+  // these as keys to determine the slow implementation.
>+#if defined(ABSL_MUTEX_READER_LOCK_IS_EXCLUSIVE)
>+  return x / 10;
>+#else
>+  return x;
>+#endif
>+}
>+
>+TEST_P(MutexVariableThreadCountTest, Mutex) {
>+  int threads = GetParam();
>+  int iterations = ScaleIterations(10000000) / threads;
>+  int operations = threads * iterations;
>+  EXPECT_EQ(RunTest(&TestMu, threads, iterations, operations), operations);
>+#if !defined(ABSL_MUTEX_ENABLE_INVARIANT_DEBUGGING_NOT_IMPLEMENTED)
>+  iterations = std::min(iterations, 10);
>+  operations = threads * iterations;
>+  EXPECT_EQ(RunTestWithInvariantDebugging(&TestMu, threads, iterations,
>+                                          operations, CheckSumG0G1),
>+            operations);
>+#endif
>+}
>+
>+TEST_P(MutexVariableThreadCountTest, Try) {
>+  int threads = GetParam();
>+  int iterations = 1000000 / threads;
>+  int operations = iterations * threads;
>+  EXPECT_EQ(RunTest(&TestTry, threads, iterations, operations), operations);
>+#if !defined(ABSL_MUTEX_ENABLE_INVARIANT_DEBUGGING_NOT_IMPLEMENTED)
>+  iterations = std::min(iterations, 10);
>+  operations = threads * iterations;
>+  EXPECT_EQ(RunTestWithInvariantDebugging(&TestTry, threads, iterations,
>+                                          operations, CheckSumG0G1),
>+            operations);
>+#endif
>+}
>+
>+TEST_P(MutexVariableThreadCountTest, R20ms) {
>+  int threads = GetParam();
>+  int iterations = 100;
>+  int operations = iterations * threads;
>+  EXPECT_EQ(RunTest(&TestR20ms, threads, iterations, operations), 0);
>+}
>+
>+TEST_P(MutexVariableThreadCountTest, RW) {
>+  int threads = GetParam();
>+  int iterations = ScaleIterations(20000000) / threads;
>+  int operations = iterations * threads;
>+  EXPECT_EQ(RunTest(&TestRW, threads, iterations, operations), operations / 2);
>+#if !defined(ABSL_MUTEX_ENABLE_INVARIANT_DEBUGGING_NOT_IMPLEMENTED)
>+  iterations = std::min(iterations, 10);
>+  operations = threads * iterations;
>+  EXPECT_EQ(RunTestWithInvariantDebugging(&TestRW, threads, iterations,
>+                                          operations, CheckSumG0G1),
>+            operations / 2);
>+#endif
>+}
>+
>+TEST_P(MutexVariableThreadCountTest, Await) {
>+  int threads = GetParam();
>+  int iterations = ScaleIterations(500000);
>+  int operations = iterations;
>+  EXPECT_EQ(RunTest(&TestAwait, threads, iterations, operations), operations);
>+}
>+
>+TEST_P(MutexVariableThreadCountTest, SignalAll) {
>+  int threads = GetParam();
>+  int iterations = 200000 / threads;
>+  int operations = iterations;
>+  EXPECT_EQ(RunTest(&TestSignalAll, threads, iterations, operations),
>+            operations);
>+}
>+
>+TEST(Mutex, Signal) {
>+  int threads = 2;  // TestSignal must use two threads
>+  int iterations = 200000;
>+  int operations = iterations;
>+  EXPECT_EQ(RunTest(&TestSignal, threads, iterations, operations), operations);
>+}
>+
>+TEST(Mutex, Timed) {
>+  int threads = 10;  // Use a fixed thread count of 10
>+  int iterations = 1000;
>+  int operations = iterations;
>+  EXPECT_EQ(RunTest(&TestCVTimeout, threads, iterations, operations),
>+            operations);
>+}
>+
>+TEST(Mutex, CVTime) {
>+  int threads = 10;  // Use a fixed thread count of 10
>+  int iterations = 1;
>+  EXPECT_EQ(RunTest(&TestCVTime, threads, iterations, 1),
>+            threads * iterations);
>+}
>+
>+TEST(Mutex, MuTime) {
>+  int threads = 10;  // Use a fixed thread count of 10
>+  int iterations = 1;
>+  EXPECT_EQ(RunTest(&TestMuTime, threads, iterations, 1), threads * iterations);
>+}
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/notification.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/notification.cc
>new file mode 100644
>index 00000000000..ed8cc906774
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/notification.cc
>@@ -0,0 +1,84 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/synchronization/notification.h"
>+
>+#include <atomic>
>+
>+#include "absl/base/attributes.h"
>+#include "absl/base/internal/raw_logging.h"
>+#include "absl/synchronization/mutex.h"
>+#include "absl/time/time.h"
>+
>+namespace absl {
>+void Notification::Notify() {
>+  MutexLock l(&this->mutex_);
>+
>+#ifndef NDEBUG
>+  if (ABSL_PREDICT_FALSE(notified_yet_.load(std::memory_order_relaxed))) {
>+    ABSL_RAW_LOG(
>+        FATAL,
>+        "Notify() method called more than once for Notification object %p",
>+        static_cast<void *>(this));
>+  }
>+#endif
>+
>+  notified_yet_.store(true, std::memory_order_release);
>+}
>+
>+Notification::~Notification() {
>+  // Make sure that the thread running Notify() exits before the object is
>+  // destructed.
>+  MutexLock l(&this->mutex_);
>+}
>+
>+static inline bool HasBeenNotifiedInternal(
>+    const std::atomic<bool> *notified_yet) {
>+  return notified_yet->load(std::memory_order_acquire);
>+}
>+
>+bool Notification::HasBeenNotified() const {
>+  return HasBeenNotifiedInternal(&this->notified_yet_);
>+}
>+
>+void Notification::WaitForNotification() const {
>+  if (!HasBeenNotifiedInternal(&this->notified_yet_)) {
>+    this->mutex_.LockWhen(Condition(&HasBeenNotifiedInternal,
>+                                    &this->notified_yet_));
>+    this->mutex_.Unlock();
>+  }
>+}
>+
>+bool Notification::WaitForNotificationWithTimeout(
>+    absl::Duration timeout) const {
>+  bool notified = HasBeenNotifiedInternal(&this->notified_yet_);
>+  if (!notified) {
>+    notified = this->mutex_.LockWhenWithTimeout(
>+        Condition(&HasBeenNotifiedInternal, &this->notified_yet_), timeout);
>+    this->mutex_.Unlock();
>+  }
>+  return notified;
>+}
>+
>+bool Notification::WaitForNotificationWithDeadline(absl::Time deadline) const {
>+  bool notified = HasBeenNotifiedInternal(&this->notified_yet_);
>+  if (!notified) {
>+    notified = this->mutex_.LockWhenWithDeadline(
>+        Condition(&HasBeenNotifiedInternal, &this->notified_yet_), deadline);
>+    this->mutex_.Unlock();
>+  }
>+  return notified;
>+}
>+
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/notification.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/notification.h
>new file mode 100644
>index 00000000000..107932f2909
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/notification.h
>@@ -0,0 +1,112 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// -----------------------------------------------------------------------------
>+// notification.h
>+// -----------------------------------------------------------------------------
>+//
>+// This header file defines a `Notification` abstraction, which allows threads
>+// to receive notification of a single occurrence of a single event.
>+//
>+// The `Notification` object maintains a private boolean "notified" state that
>+// transitions to `true` at most once. The `Notification` class provides the
>+// following primary member functions:
>+//   * `HasBeenNotified() `to query its state
>+//   * `WaitForNotification*()` to have threads wait until the "notified" state
>+//      is `true`.
>+//   * `Notify()` to set the notification's "notified" state to `true` and
>+//     notify all waiting threads that the event has occurred.
>+//     This method may only be called once.
>+//
>+// Note that while `Notify()` may only be called once, it is perfectly valid to
>+// call any of the `WaitForNotification*()` methods multiple times, from
>+// multiple threads -- even after the notification's "notified" state has been
>+// set -- in which case those methods will immediately return.
>+//
>+// Note that the lifetime of a `Notification` requires careful consideration;
>+// it might not be safe to destroy a notification after calling `Notify()` since
>+// it is still legal for other threads to call `WaitForNotification*()` methods
>+// on the notification. However, observers responding to a "notified" state of
>+// `true` can safely delete the notification without interfering with the call
>+// to `Notify()` in the other thread.
>+//
>+// Memory ordering: For any threads X and Y, if X calls `Notify()`, then any
>+// action taken by X before it calls `Notify()` is visible to thread Y after:
>+//  * Y returns from `WaitForNotification()`, or
>+//  * Y receives a `true` return value from either `HasBeenNotified()` or
>+//    `WaitForNotificationWithTimeout()`.
>+
>+#ifndef ABSL_SYNCHRONIZATION_NOTIFICATION_H_
>+#define ABSL_SYNCHRONIZATION_NOTIFICATION_H_
>+
>+#include <atomic>
>+
>+#include "absl/synchronization/mutex.h"
>+#include "absl/time/time.h"
>+
>+namespace absl {
>+
>+// -----------------------------------------------------------------------------
>+// Notification
>+// -----------------------------------------------------------------------------
>+class Notification {
>+ public:
>+  // Initializes the "notified" state to unnotified.
>+  Notification() : notified_yet_(false) {}
>+  explicit Notification(bool prenotify) : notified_yet_(prenotify) {}
>+  Notification(const Notification&) = delete;
>+  Notification& operator=(const Notification&) = delete;
>+  ~Notification();
>+
>+  // Notification::HasBeenNotified()
>+  //
>+  // Returns the value of the notification's internal "notified" state.
>+  bool HasBeenNotified() const;
>+
>+  // Notification::WaitForNotification()
>+  //
>+  // Blocks the calling thread until the notification's "notified" state is
>+  // `true`. Note that if `Notify()` has been previously called on this
>+  // notification, this function will immediately return.
>+  void WaitForNotification() const;
>+
>+  // Notification::WaitForNotificationWithTimeout()
>+  //
>+  // Blocks until either the notification's "notified" state is `true` (which
>+  // may occur immediately) or the timeout has elapsed, returning the value of
>+  // its "notified" state in either case.
>+  bool WaitForNotificationWithTimeout(absl::Duration timeout) const;
>+
>+  // Notification::WaitForNotificationWithDeadline()
>+  //
>+  // Blocks until either the notification's "notified" state is `true` (which
>+  // may occur immediately) or the deadline has expired, returning the value of
>+  // its "notified" state in either case.
>+  bool WaitForNotificationWithDeadline(absl::Time deadline) const;
>+
>+  // Notification::Notify()
>+  //
>+  // Sets the "notified" state of this notification to `true` and wakes waiting
>+  // threads. Note: do not call `Notify()` multiple times on the same
>+  // `Notification`; calling `Notify()` more than once on the same notification
>+  // results in undefined behavior.
>+  void Notify();
>+
>+ private:
>+  mutable Mutex mutex_;
>+  std::atomic<bool> notified_yet_;  // written under mutex_
>+};
>+
>+}  // namespace absl
>+#endif  // ABSL_SYNCHRONIZATION_NOTIFICATION_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/notification_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/notification_test.cc
>new file mode 100644
>index 00000000000..9b3b6a5a9e8
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/synchronization/notification_test.cc
>@@ -0,0 +1,124 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/synchronization/notification.h"
>+
>+#include <thread>  // NOLINT(build/c++11)
>+#include <vector>
>+
>+#include "gtest/gtest.h"
>+#include "absl/synchronization/mutex.h"
>+
>+namespace absl {
>+
>+// A thread-safe class that holds a counter.
>+class ThreadSafeCounter {
>+ public:
>+  ThreadSafeCounter() : count_(0) {}
>+
>+  void Increment() {
>+    MutexLock lock(&mutex_);
>+    ++count_;
>+  }
>+
>+  int Get() const {
>+    MutexLock lock(&mutex_);
>+    return count_;
>+  }
>+
>+  void WaitUntilGreaterOrEqual(int n) {
>+    MutexLock lock(&mutex_);
>+    auto cond = [this, n]() { return count_ >= n; };
>+    mutex_.Await(Condition(&cond));
>+  }
>+
>+ private:
>+  mutable Mutex mutex_;
>+  int count_;
>+};
>+
>+// Runs the |i|'th worker thread for the tests in BasicTests().  Increments the
>+// |ready_counter|, waits on the |notification|, and then increments the
>+// |done_counter|.
>+static void RunWorker(int i, ThreadSafeCounter* ready_counter,
>+                      Notification* notification,
>+                      ThreadSafeCounter* done_counter) {
>+  ready_counter->Increment();
>+  notification->WaitForNotification();
>+  done_counter->Increment();
>+}
>+
>+// Tests that the |notification| properly blocks and awakens threads.  Assumes
>+// that the |notification| is not yet triggered.  If |notify_before_waiting| is
>+// true, the |notification| is triggered before any threads are created, so the
>+// threads never block in WaitForNotification().  Otherwise, the |notification|
>+// is triggered at a later point when most threads are likely to be blocking in
>+// WaitForNotification().
>+static void BasicTests(bool notify_before_waiting, Notification* notification) {
>+  EXPECT_FALSE(notification->HasBeenNotified());
>+  EXPECT_FALSE(
>+      notification->WaitForNotificationWithTimeout(absl::Milliseconds(0)));
>+  EXPECT_FALSE(notification->WaitForNotificationWithDeadline(absl::Now()));
>+
>+  absl::Time start = absl::Now();
>+  EXPECT_FALSE(
>+      notification->WaitForNotificationWithTimeout(absl::Milliseconds(50)));
>+  EXPECT_LE(start + absl::Milliseconds(50), absl::Now());
>+
>+  ThreadSafeCounter ready_counter;
>+  ThreadSafeCounter done_counter;
>+
>+  if (notify_before_waiting) {
>+    notification->Notify();
>+  }
>+
>+  // Create a bunch of threads that increment the |done_counter| after being
>+  // notified.
>+  const int kNumThreads = 10;
>+  std::vector<std::thread> workers;
>+  for (int i = 0; i < kNumThreads; ++i) {
>+    workers.push_back(std::thread(&RunWorker, i, &ready_counter, notification,
>+                                  &done_counter));
>+  }
>+
>+  if (!notify_before_waiting) {
>+    ready_counter.WaitUntilGreaterOrEqual(kNumThreads);
>+
>+    // Workers have not been notified yet, so the |done_counter| should be
>+    // unmodified.
>+    EXPECT_EQ(0, done_counter.Get());
>+
>+    notification->Notify();
>+  }
>+
>+  // After notifying and then joining the workers, both counters should be
>+  // fully incremented.
>+  notification->WaitForNotification();  // should exit immediately
>+  EXPECT_TRUE(notification->HasBeenNotified());
>+  EXPECT_TRUE(notification->WaitForNotificationWithTimeout(absl::Seconds(0)));
>+  EXPECT_TRUE(notification->WaitForNotificationWithDeadline(absl::Now()));
>+  for (std::thread& worker : workers) {
>+    worker.join();
>+  }
>+  EXPECT_EQ(kNumThreads, ready_counter.Get());
>+  EXPECT_EQ(kNumThreads, done_counter.Get());
>+}
>+
>+TEST(NotificationTest, SanityTest) {
>+  Notification local_notification1, local_notification2;
>+  BasicTests(false, &local_notification1);
>+  BasicTests(true, &local_notification2);
>+}
>+
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/BUILD.bazel b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/BUILD.bazel
>new file mode 100644
>index 00000000000..c7c16d4376c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/BUILD.bazel
>@@ -0,0 +1,112 @@
>+#
>+# Copyright 2017 The Abseil Authors.
>+#
>+# Licensed under the Apache License, Version 2.0 (the "License");
>+# you may not use this file except in compliance with the License.
>+# You may obtain a copy of the License at
>+#
>+#      http://www.apache.org/licenses/LICENSE-2.0
>+#
>+# Unless required by applicable law or agreed to in writing, software
>+# distributed under the License is distributed on an "AS IS" BASIS,
>+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+# See the License for the specific language governing permissions and
>+# limitations under the License.
>+#
>+
>+load(
>+    "//absl:copts.bzl",
>+    "ABSL_DEFAULT_COPTS",
>+    "ABSL_TEST_COPTS",
>+)
>+
>+package(default_visibility = ["//visibility:public"])
>+
>+licenses(["notice"])  # Apache 2.0
>+
>+cc_library(
>+    name = "time",
>+    srcs = [
>+        "clock.cc",
>+        "duration.cc",
>+        "format.cc",
>+        "internal/get_current_time_chrono.inc",
>+        "internal/get_current_time_posix.inc",
>+        "time.cc",
>+    ],
>+    hdrs = [
>+        "clock.h",
>+        "time.h",
>+    ],
>+    copts = ABSL_DEFAULT_COPTS,
>+    deps = [
>+        "//absl/base",
>+        "//absl/base:core_headers",
>+        "//absl/numeric:int128",
>+        "//absl/strings",
>+        "//absl/time/internal/cctz:civil_time",
>+        "//absl/time/internal/cctz:time_zone",
>+    ],
>+)
>+
>+cc_library(
>+    name = "test_util",
>+    testonly = 1,
>+    srcs = [
>+        "internal/test_util.cc",
>+        "internal/zoneinfo.inc",
>+    ],
>+    hdrs = ["internal/test_util.h"],
>+    copts = ABSL_DEFAULT_COPTS,
>+    visibility = [
>+        "//absl/time:__pkg__",
>+    ],
>+    deps = [
>+        ":time",
>+        "//absl/base",
>+        "//absl/time/internal/cctz:time_zone",
>+        "@com_google_googletest//:gtest",
>+    ],
>+)
>+
>+cc_test(
>+    name = "time_test",
>+    srcs = [
>+        "clock_test.cc",
>+        "duration_test.cc",
>+        "format_test.cc",
>+        "time_norm_test.cc",
>+        "time_test.cc",
>+        "time_zone_test.cc",
>+    ],
>+    copts = ABSL_TEST_COPTS,
>+    deps = [
>+        ":test_util",
>+        ":time",
>+        "//absl/base",
>+        "//absl/base:config",
>+        "//absl/base:core_headers",
>+        "//absl/time/internal/cctz:time_zone",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "time_benchmark",
>+    srcs = [
>+        "clock_benchmark.cc",
>+        "duration_benchmark.cc",
>+        "format_benchmark.cc",
>+        "time_benchmark.cc",
>+    ],
>+    copts = ABSL_TEST_COPTS,
>+    tags = [
>+        "benchmark",
>+    ],
>+    deps = [
>+        ":test_util",
>+        ":time",
>+        "//absl/base",
>+        "@com_github_google_benchmark//:benchmark_main",
>+    ],
>+)
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/BUILD.gn
>new file mode 100644
>index 00000000000..9927af8343c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/BUILD.gn
>@@ -0,0 +1,70 @@
>+# Copyright 2018 The Chromium Authors. All rights reserved.
>+# Use of this source code is governed by a BSD-style license that can be
>+# found in the LICENSE file.
>+
>+import("//build_overrides/build.gni")
>+
>+if (build_with_chromium) {
>+  visibility = [
>+    "//third_party/webrtc/*",
>+    "//third_party/abseil-cpp/*",
>+    "//third_party/googletest:gtest",
>+  ]
>+} else {
>+  visibility = [ "*" ]
>+}
>+
>+source_set("time") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  sources = [
>+    "clock.cc",
>+    "duration.cc",
>+    "format.cc",
>+    "internal/get_current_time_chrono.inc",
>+    "internal/get_current_time_posix.inc",
>+    "time.cc",
>+  ]
>+  public = [
>+    "clock.h",
>+    "time.h",
>+  ]
>+  deps = [
>+    "../base",
>+    "../base:core_headers",
>+    "../numeric:int128",
>+    "../strings",
>+    "../time/internal/cctz:civil_time",
>+    "../time/internal/cctz:time_zone",
>+  ]
>+}
>+
>+source_set("test_util") {
>+  testonly = true
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  sources = [
>+    "internal/test_util.cc",
>+    "internal/zoneinfo.inc",
>+  ]
>+  public = [
>+    "internal/test_util.h",
>+  ]
>+  deps = [
>+    ":time",
>+    "../base",
>+    "../time/internal/cctz:time_zone",
>+    "//testing/gtest",
>+    "//testing/gmock",
>+  ]
>+  visibility = []
>+  visibility += [ "../time:*" ]
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/CMakeLists.txt b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/CMakeLists.txt
>new file mode 100644
>index 00000000000..06272364f14
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/CMakeLists.txt
>@@ -0,0 +1,96 @@
>+#
>+# Copyright 2017 The Abseil Authors.
>+#
>+# Licensed under the Apache License, Version 2.0 (the "License");
>+# you may not use this file except in compliance with the License.
>+# You may obtain a copy of the License at
>+#
>+#      http://www.apache.org/licenses/LICENSE-2.0
>+#
>+# Unless required by applicable law or agreed to in writing, software
>+# distributed under the License is distributed on an "AS IS" BASIS,
>+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+# See the License for the specific language governing permissions and
>+# limitations under the License.
>+#
>+
>+list(APPEND TIME_PUBLIC_HEADERS
>+  "clock.h"
>+  "time.h"
>+)
>+
>+
>+list(APPEND TIME_INTERNAL_HEADERS
>+  "internal/test_util.h"
>+  "internal/cctz/include/cctz/civil_time.h"
>+  "internal/cctz/include/cctz/civil_time_detail.h"
>+  "internal/cctz/include/cctz/time_zone.h"
>+  "internal/cctz/include/cctz/zone_info_source.h"
>+)
>+
>+list(APPEND TIME_SRC
>+  "time.cc"
>+  "clock.cc"
>+  "duration.cc"
>+  "format.cc"
>+  "internal/cctz/src/civil_time_detail.cc"
>+  "internal/cctz/src/time_zone_fixed.cc"
>+  "internal/cctz/src/time_zone_fixed.h"
>+  "internal/cctz/src/time_zone_format.cc"
>+  "internal/cctz/src/time_zone_if.cc"
>+  "internal/cctz/src/time_zone_if.h"
>+  "internal/cctz/src/time_zone_impl.cc"
>+  "internal/cctz/src/time_zone_impl.h"
>+  "internal/cctz/src/time_zone_info.cc"
>+  "internal/cctz/src/time_zone_info.h"
>+  "internal/cctz/src/time_zone_libc.cc"
>+  "internal/cctz/src/time_zone_libc.h"
>+  "internal/cctz/src/time_zone_lookup.cc"
>+  "internal/cctz/src/time_zone_posix.cc"
>+  "internal/cctz/src/time_zone_posix.h"
>+  "internal/cctz/src/tzfile.h"
>+  "internal/cctz/src/zone_info_source.cc"
>+  ${TIME_PUBLIC_HEADERS}
>+  ${TIME_INTERNAL_HEADERS}
>+)
>+set(TIME_PUBLIC_LIBRARIES absl::base absl::stacktrace absl::int128 absl::strings)
>+
>+absl_library(
>+  TARGET
>+    absl_time
>+  SOURCES
>+    ${TIME_SRC}
>+  PUBLIC_LIBRARIES
>+    ${TIME_PUBLIC_LIBRARIES}
>+  EXPORT_NAME
>+    time
>+)
>+
>+
>+
>+#
>+## TESTS
>+#
>+
>+# test time_test
>+list(APPEND TIME_TEST_SRC
>+  "time_test.cc"
>+  "clock_test.cc"
>+  "duration_test.cc"
>+  "format_test.cc"
>+  "time_norm_test.cc"
>+  "time_test.cc"
>+  "time_zone_test.cc"
>+  "internal/test_util.cc"
>+)
>+set(TIME_TEST_PUBLIC_LIBRARIES absl::time)
>+
>+absl_test(
>+  TARGET
>+    time_test
>+  SOURCES
>+    ${TIME_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${TIME_TEST_PUBLIC_LIBRARIES}
>+)
>+
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/clock.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/clock.cc
>new file mode 100644
>index 00000000000..74ee1401b98
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/clock.cc
>@@ -0,0 +1,561 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/time/clock.h"
>+
>+#include "absl/base/attributes.h"
>+
>+#ifdef _WIN32
>+#include <windows.h>
>+#endif
>+
>+#include <algorithm>
>+#include <atomic>
>+#include <cerrno>
>+#include <cstdint>
>+#include <ctime>
>+#include <limits>
>+
>+#include "absl/base/internal/spinlock.h"
>+#include "absl/base/internal/unscaledcycleclock.h"
>+#include "absl/base/macros.h"
>+#include "absl/base/port.h"
>+#include "absl/base/thread_annotations.h"
>+
>+namespace absl {
>+Time Now() {
>+  // TODO(bww): Get a timespec instead so we don't have to divide.
>+  int64_t n = absl::GetCurrentTimeNanos();
>+  if (n >= 0) {
>+    return time_internal::FromUnixDuration(
>+        time_internal::MakeDuration(n / 1000000000, n % 1000000000 * 4));
>+  }
>+  return time_internal::FromUnixDuration(absl::Nanoseconds(n));
>+}
>+}  // namespace absl
>+
>+// Decide if we should use the fast GetCurrentTimeNanos() algorithm
>+// based on the cyclecounter, otherwise just get the time directly
>+// from the OS on every call. This can be chosen at compile-time via
>+// -DABSL_USE_CYCLECLOCK_FOR_GET_CURRENT_TIME_NANOS=[0|1]
>+#ifndef ABSL_USE_CYCLECLOCK_FOR_GET_CURRENT_TIME_NANOS
>+#if ABSL_USE_UNSCALED_CYCLECLOCK
>+#define ABSL_USE_CYCLECLOCK_FOR_GET_CURRENT_TIME_NANOS 1
>+#else
>+#define ABSL_USE_CYCLECLOCK_FOR_GET_CURRENT_TIME_NANOS 0
>+#endif
>+#endif
>+
>+#if defined(__APPLE__) || defined(_WIN32)
>+#include "absl/time/internal/get_current_time_chrono.inc"
>+#else
>+#include "absl/time/internal/get_current_time_posix.inc"
>+#endif
>+
>+// Allows override by test.
>+#ifndef GET_CURRENT_TIME_NANOS_FROM_SYSTEM
>+#define GET_CURRENT_TIME_NANOS_FROM_SYSTEM() \
>+  ::absl::time_internal::GetCurrentTimeNanosFromSystem()
>+#endif
>+
>+#if !ABSL_USE_CYCLECLOCK_FOR_GET_CURRENT_TIME_NANOS
>+namespace absl {
>+int64_t GetCurrentTimeNanos() {
>+  return GET_CURRENT_TIME_NANOS_FROM_SYSTEM();
>+}
>+}  // namespace absl
>+#else  // Use the cyclecounter-based implementation below.
>+
>+// Allows override by test.
>+#ifndef GET_CURRENT_TIME_NANOS_CYCLECLOCK_NOW
>+#define GET_CURRENT_TIME_NANOS_CYCLECLOCK_NOW() \
>+  ::absl::time_internal::UnscaledCycleClockWrapperForGetCurrentTime::Now()
>+#endif
>+
>+// The following counters are used only by the test code.
>+static int64_t stats_initializations;
>+static int64_t stats_reinitializations;
>+static int64_t stats_calibrations;
>+static int64_t stats_slow_paths;
>+static int64_t stats_fast_slow_paths;
>+
>+namespace absl {
>+namespace time_internal {
>+// This is a friend wrapper around UnscaledCycleClock::Now()
>+// (needed to access UnscaledCycleClock).
>+class UnscaledCycleClockWrapperForGetCurrentTime {
>+ public:
>+  static int64_t Now() { return base_internal::UnscaledCycleClock::Now(); }
>+};
>+}  // namespace time_internal
>+
>+// uint64_t is used in this module to provide an extra bit in multiplications
>+
>+// Return the time in ns as told by the kernel interface.  Place in *cycleclock
>+// the value of the cycleclock at about the time of the syscall.
>+// This call represents the time base that this module synchronizes to.
>+// Ensures that *cycleclock does not step back by up to (1 << 16) from
>+// last_cycleclock, to discard small backward counter steps.  (Larger steps are
>+// assumed to be complete resyncs, which shouldn't happen.  If they do, a full
>+// reinitialization of the outer algorithm should occur.)
>+static int64_t GetCurrentTimeNanosFromKernel(uint64_t last_cycleclock,
>+                                             uint64_t *cycleclock) {
>+  // We try to read clock values at about the same time as the kernel clock.
>+  // This value gets adjusted up or down as estimate of how long that should
>+  // take, so we can reject attempts that take unusually long.
>+  static std::atomic<uint64_t> approx_syscall_time_in_cycles{10 * 1000};
>+
>+  uint64_t local_approx_syscall_time_in_cycles =  // local copy
>+      approx_syscall_time_in_cycles.load(std::memory_order_relaxed);
>+
>+  int64_t current_time_nanos_from_system;
>+  uint64_t before_cycles;
>+  uint64_t after_cycles;
>+  uint64_t elapsed_cycles;
>+  int loops = 0;
>+  do {
>+    before_cycles = GET_CURRENT_TIME_NANOS_CYCLECLOCK_NOW();
>+    current_time_nanos_from_system = GET_CURRENT_TIME_NANOS_FROM_SYSTEM();
>+    after_cycles = GET_CURRENT_TIME_NANOS_CYCLECLOCK_NOW();
>+    // elapsed_cycles is unsigned, so is large on overflow
>+    elapsed_cycles = after_cycles - before_cycles;
>+    if (elapsed_cycles >= local_approx_syscall_time_in_cycles &&
>+        ++loops == 20) {  // clock changed frequencies?  Back off.
>+      loops = 0;
>+      if (local_approx_syscall_time_in_cycles < 1000 * 1000) {
>+        local_approx_syscall_time_in_cycles =
>+            (local_approx_syscall_time_in_cycles + 1) << 1;
>+      }
>+      approx_syscall_time_in_cycles.store(
>+          local_approx_syscall_time_in_cycles,
>+          std::memory_order_relaxed);
>+    }
>+  } while (elapsed_cycles >= local_approx_syscall_time_in_cycles ||
>+           last_cycleclock - after_cycles < (static_cast<uint64_t>(1) << 16));
>+
>+  // Number of times in a row we've seen a kernel time call take substantially
>+  // less than approx_syscall_time_in_cycles.
>+  static std::atomic<uint32_t> seen_smaller{ 0 };
>+
>+  // Adjust approx_syscall_time_in_cycles to be within a factor of 2
>+  // of the typical time to execute one iteration of the loop above.
>+  if ((local_approx_syscall_time_in_cycles >> 1) < elapsed_cycles) {
>+    // measured time is no smaller than half current approximation
>+    seen_smaller.store(0, std::memory_order_relaxed);
>+  } else if (seen_smaller.fetch_add(1, std::memory_order_relaxed) >= 3) {
>+    // smaller delays several times in a row; reduce approximation by 12.5%
>+    const uint64_t new_approximation =
>+        local_approx_syscall_time_in_cycles -
>+        (local_approx_syscall_time_in_cycles >> 3);
>+    approx_syscall_time_in_cycles.store(new_approximation,
>+                                        std::memory_order_relaxed);
>+    seen_smaller.store(0, std::memory_order_relaxed);
>+  }
>+
>+  *cycleclock = after_cycles;
>+  return current_time_nanos_from_system;
>+}
>+
>+
>+// ---------------------------------------------------------------------
>+// An implementation of reader-write locks that use no atomic ops in the read
>+// case.  This is a generalization of Lamport's method for reading a multiword
>+// clock.  Increment a word on each write acquisition, using the low-order bit
>+// as a spinlock; the word is the high word of the "clock".  Readers read the
>+// high word, then all other data, then the high word again, and repeat the
>+// read if the reads of the high words yields different answers, or an odd
>+// value (either case suggests possible interference from a writer).
>+// Here we use a spinlock to ensure only one writer at a time, rather than
>+// spinning on the bottom bit of the word to benefit from SpinLock
>+// spin-delay tuning.
>+
>+// Acquire seqlock (*seq) and return the value to be written to unlock.
>+static inline uint64_t SeqAcquire(std::atomic<uint64_t> *seq) {
>+  uint64_t x = seq->fetch_add(1, std::memory_order_relaxed);
>+
>+  // We put a release fence between update to *seq and writes to shared data.
>+  // Thus all stores to shared data are effectively release operations and
>+  // update to *seq above cannot be re-ordered past any of them.  Note that
>+  // this barrier is not for the fetch_add above.  A release barrier for the
>+  // fetch_add would be before it, not after.
>+  std::atomic_thread_fence(std::memory_order_release);
>+
>+  return x + 2;   // original word plus 2
>+}
>+
>+// Release seqlock (*seq) by writing x to it---a value previously returned by
>+// SeqAcquire.
>+static inline void SeqRelease(std::atomic<uint64_t> *seq, uint64_t x) {
>+  // The unlock store to *seq must have release ordering so that all
>+  // updates to shared data must finish before this store.
>+  seq->store(x, std::memory_order_release);  // release lock for readers
>+}
>+
>+// ---------------------------------------------------------------------
>+
>+// "nsscaled" is unit of time equal to a (2**kScale)th of a nanosecond.
>+enum { kScale = 30 };
>+
>+// The minimum interval between samples of the time base.
>+// We pick enough time to amortize the cost of the sample,
>+// to get a reasonably accurate cycle counter rate reading,
>+// and not so much that calculations will overflow 64-bits.
>+static const uint64_t kMinNSBetweenSamples = 2000 << 20;
>+
>+// We require that kMinNSBetweenSamples shifted by kScale
>+// have at least a bit left over for 64-bit calculations.
>+static_assert(((kMinNSBetweenSamples << (kScale + 1)) >> (kScale + 1)) ==
>+               kMinNSBetweenSamples,
>+               "cannot represent kMaxBetweenSamplesNSScaled");
>+
>+// A reader-writer lock protecting the static locations below.
>+// See SeqAcquire() and SeqRelease() above.
>+static absl::base_internal::SpinLock lock(
>+    absl::base_internal::kLinkerInitialized);
>+static std::atomic<uint64_t> seq(0);
>+
>+// data from a sample of the kernel's time value
>+struct TimeSampleAtomic {
>+  std::atomic<uint64_t> raw_ns;              // raw kernel time
>+  std::atomic<uint64_t> base_ns;             // our estimate of time
>+  std::atomic<uint64_t> base_cycles;         // cycle counter reading
>+  std::atomic<uint64_t> nsscaled_per_cycle;  // cycle period
>+  // cycles before we'll sample again (a scaled reciprocal of the period,
>+  // to avoid a division on the fast path).
>+  std::atomic<uint64_t> min_cycles_per_sample;
>+};
>+// Same again, but with non-atomic types
>+struct TimeSample {
>+  uint64_t raw_ns;                 // raw kernel time
>+  uint64_t base_ns;                // our estimate of time
>+  uint64_t base_cycles;            // cycle counter reading
>+  uint64_t nsscaled_per_cycle;     // cycle period
>+  uint64_t min_cycles_per_sample;  // approx cycles before next sample
>+};
>+
>+static struct TimeSampleAtomic last_sample;   // the last sample; under seq
>+
>+static int64_t GetCurrentTimeNanosSlowPath() ABSL_ATTRIBUTE_COLD;
>+
>+// Read the contents of *atomic into *sample.
>+// Each field is read atomically, but to maintain atomicity between fields,
>+// the access must be done under a lock.
>+static void ReadTimeSampleAtomic(const struct TimeSampleAtomic *atomic,
>+                                 struct TimeSample *sample) {
>+  sample->base_ns = atomic->base_ns.load(std::memory_order_relaxed);
>+  sample->base_cycles = atomic->base_cycles.load(std::memory_order_relaxed);
>+  sample->nsscaled_per_cycle =
>+      atomic->nsscaled_per_cycle.load(std::memory_order_relaxed);
>+  sample->min_cycles_per_sample =
>+      atomic->min_cycles_per_sample.load(std::memory_order_relaxed);
>+  sample->raw_ns = atomic->raw_ns.load(std::memory_order_relaxed);
>+}
>+
>+// Public routine.
>+// Algorithm:  We wish to compute real time from a cycle counter.  In normal
>+// operation, we construct a piecewise linear approximation to the kernel time
>+// source, using the cycle counter value.  The start of each line segment is at
>+// the same point as the end of the last, but may have a different slope (that
>+// is, a different idea of the cycle counter frequency).  Every couple of
>+// seconds, the kernel time source is sampled and compared with the current
>+// approximation.  A new slope is chosen that, if followed for another couple
>+// of seconds, will correct the error at the current position.  The information
>+// for a sample is in the "last_sample" struct.  The linear approximation is
>+//   estimated_time = last_sample.base_ns +
>+//     last_sample.ns_per_cycle * (counter_reading - last_sample.base_cycles)
>+// (ns_per_cycle is actually stored in different units and scaled, to avoid
>+// overflow).  The base_ns of the next linear approximation is the
>+// estimated_time using the last approximation; the base_cycles is the cycle
>+// counter value at that time; the ns_per_cycle is the number of ns per cycle
>+// measured since the last sample, but adjusted so that most of the difference
>+// between the estimated_time and the kernel time will be corrected by the
>+// estimated time to the next sample.  In normal operation, this algorithm
>+// relies on:
>+// - the cycle counter and kernel time rates not changing a lot in a few
>+//   seconds.
>+// - the client calling into the code often compared to a couple of seconds, so
>+//   the time to the next correction can be estimated.
>+// Any time ns_per_cycle is not known, a major error is detected, or the
>+// assumption about frequent calls is violated, the implementation returns the
>+// kernel time.  It records sufficient data that a linear approximation can
>+// resume a little later.
>+
>+int64_t GetCurrentTimeNanos() {
>+  // read the data from the "last_sample" struct (but don't need raw_ns yet)
>+  // The reads of "seq" and test of the values emulate a reader lock.
>+  uint64_t base_ns;
>+  uint64_t base_cycles;
>+  uint64_t nsscaled_per_cycle;
>+  uint64_t min_cycles_per_sample;
>+  uint64_t seq_read0;
>+  uint64_t seq_read1;
>+
>+  // If we have enough information to interpolate, the value returned will be
>+  // derived from this cycleclock-derived time estimate.  On some platforms
>+  // (POWER) the function to retrieve this value has enough complexity to
>+  // contribute to register pressure - reading it early before initializing
>+  // the other pieces of the calculation minimizes spill/restore instructions,
>+  // minimizing icache cost.
>+  uint64_t now_cycles = GET_CURRENT_TIME_NANOS_CYCLECLOCK_NOW();
>+
>+  // Acquire pairs with the barrier in SeqRelease - if this load sees that
>+  // store, the shared-data reads necessarily see that SeqRelease's updates
>+  // to the same shared data.
>+  seq_read0 = seq.load(std::memory_order_acquire);
>+
>+  base_ns = last_sample.base_ns.load(std::memory_order_relaxed);
>+  base_cycles = last_sample.base_cycles.load(std::memory_order_relaxed);
>+  nsscaled_per_cycle =
>+      last_sample.nsscaled_per_cycle.load(std::memory_order_relaxed);
>+  min_cycles_per_sample =
>+      last_sample.min_cycles_per_sample.load(std::memory_order_relaxed);
>+
>+  // This acquire fence pairs with the release fence in SeqAcquire.  Since it
>+  // is sequenced between reads of shared data and seq_read1, the reads of
>+  // shared data are effectively acquiring.
>+  std::atomic_thread_fence(std::memory_order_acquire);
>+
>+  // The shared-data reads are effectively acquire ordered, and the
>+  // shared-data writes are effectively release ordered. Therefore if our
>+  // shared-data reads see any of a particular update's shared-data writes,
>+  // seq_read1 is guaranteed to see that update's SeqAcquire.
>+  seq_read1 = seq.load(std::memory_order_relaxed);
>+
>+  // Fast path.  Return if min_cycles_per_sample has not yet elapsed since the
>+  // last sample, and we read a consistent sample.  The fast path activates
>+  // only when min_cycles_per_sample is non-zero, which happens when we get an
>+  // estimate for the cycle time.  The predicate will fail if now_cycles <
>+  // base_cycles, or if some other thread is in the slow path.
>+  //
>+  // Since we now read now_cycles before base_ns, it is possible for now_cycles
>+  // to be less than base_cycles (if we were interrupted between those loads and
>+  // last_sample was updated). This is harmless, because delta_cycles will wrap
>+  // and report a time much much bigger than min_cycles_per_sample. In that case
>+  // we will take the slow path.
>+  uint64_t delta_cycles = now_cycles - base_cycles;
>+  if (seq_read0 == seq_read1 && (seq_read0 & 1) == 0 &&
>+      delta_cycles < min_cycles_per_sample) {
>+    return base_ns + ((delta_cycles * nsscaled_per_cycle) >> kScale);
>+  }
>+  return GetCurrentTimeNanosSlowPath();
>+}
>+
>+// Return (a << kScale)/b.
>+// Zero is returned if b==0.   Scaling is performed internally to
>+// preserve precision without overflow.
>+static uint64_t SafeDivideAndScale(uint64_t a, uint64_t b) {
>+  // Find maximum safe_shift so that
>+  //  0 <= safe_shift <= kScale  and  (a << safe_shift) does not overflow.
>+  int safe_shift = kScale;
>+  while (((a << safe_shift) >> safe_shift) != a) {
>+    safe_shift--;
>+  }
>+  uint64_t scaled_b = b >> (kScale - safe_shift);
>+  uint64_t quotient = 0;
>+  if (scaled_b != 0) {
>+    quotient = (a << safe_shift) / scaled_b;
>+  }
>+  return quotient;
>+}
>+
>+static uint64_t UpdateLastSample(
>+    uint64_t now_cycles, uint64_t now_ns, uint64_t delta_cycles,
>+    const struct TimeSample *sample) ABSL_ATTRIBUTE_COLD;
>+
>+// The slow path of GetCurrentTimeNanos().  This is taken while gathering
>+// initial samples, when enough time has elapsed since the last sample, and if
>+// any other thread is writing to last_sample.
>+//
>+// Manually mark this 'noinline' to minimize stack frame size of the fast
>+// path.  Without this, sometimes a compiler may inline this big block of code
>+// into the fast past.  That causes lots of register spills and reloads that
>+// are unnecessary unless the slow path is taken.
>+//
>+// TODO(absl-team): Remove this attribute when our compiler is smart enough
>+// to do the right thing.
>+ABSL_ATTRIBUTE_NOINLINE
>+static int64_t GetCurrentTimeNanosSlowPath() LOCKS_EXCLUDED(lock) {
>+  // Serialize access to slow-path.  Fast-path readers are not blocked yet, and
>+  // code below must not modify last_sample until the seqlock is acquired.
>+  lock.Lock();
>+
>+  // Sample the kernel time base.  This is the definition of
>+  // "now" if we take the slow path.
>+  static uint64_t last_now_cycles;  // protected by lock
>+  uint64_t now_cycles;
>+  uint64_t now_ns = GetCurrentTimeNanosFromKernel(last_now_cycles, &now_cycles);
>+  last_now_cycles = now_cycles;
>+
>+  uint64_t estimated_base_ns;
>+
>+  // ----------
>+  // Read the "last_sample" values again; this time holding the write lock.
>+  struct TimeSample sample;
>+  ReadTimeSampleAtomic(&last_sample, &sample);
>+
>+  // ----------
>+  // Try running the fast path again; another thread may have updated the
>+  // sample between our run of the fast path and the sample we just read.
>+  uint64_t delta_cycles = now_cycles - sample.base_cycles;
>+  if (delta_cycles < sample.min_cycles_per_sample) {
>+    // Another thread updated the sample.  This path does not take the seqlock
>+    // so that blocked readers can make progress without blocking new readers.
>+    estimated_base_ns = sample.base_ns +
>+        ((delta_cycles * sample.nsscaled_per_cycle) >> kScale);
>+    stats_fast_slow_paths++;
>+  } else {
>+    estimated_base_ns =
>+        UpdateLastSample(now_cycles, now_ns, delta_cycles, &sample);
>+  }
>+
>+  lock.Unlock();
>+
>+  return estimated_base_ns;
>+}
>+
>+// Main part of the algorithm.  Locks out readers, updates the approximation
>+// using the new sample from the kernel, and stores the result in last_sample
>+// for readers.  Returns the new estimated time.
>+static uint64_t UpdateLastSample(uint64_t now_cycles, uint64_t now_ns,
>+                                 uint64_t delta_cycles,
>+                                 const struct TimeSample *sample)
>+    EXCLUSIVE_LOCKS_REQUIRED(lock) {
>+  uint64_t estimated_base_ns = now_ns;
>+  uint64_t lock_value = SeqAcquire(&seq);  // acquire seqlock to block readers
>+
>+  // The 5s in the next if-statement limits the time for which we will trust
>+  // the cycle counter and our last sample to give a reasonable result.
>+  // Errors in the rate of the source clock can be multiplied by the ratio
>+  // between this limit and kMinNSBetweenSamples.
>+  if (sample->raw_ns == 0 ||  // no recent sample, or clock went backwards
>+      sample->raw_ns + static_cast<uint64_t>(5) * 1000 * 1000 * 1000 < now_ns ||
>+      now_ns < sample->raw_ns || now_cycles < sample->base_cycles) {
>+    // record this sample, and forget any previously known slope.
>+    last_sample.raw_ns.store(now_ns, std::memory_order_relaxed);
>+    last_sample.base_ns.store(estimated_base_ns, std::memory_order_relaxed);
>+    last_sample.base_cycles.store(now_cycles, std::memory_order_relaxed);
>+    last_sample.nsscaled_per_cycle.store(0, std::memory_order_relaxed);
>+    last_sample.min_cycles_per_sample.store(0, std::memory_order_relaxed);
>+    stats_initializations++;
>+  } else if (sample->raw_ns + 500 * 1000 * 1000 < now_ns &&
>+             sample->base_cycles + 100 < now_cycles) {
>+    // Enough time has passed to compute the cycle time.
>+    if (sample->nsscaled_per_cycle != 0) {  // Have a cycle time estimate.
>+      // Compute time from counter reading, but avoiding overflow
>+      // delta_cycles may be larger than on the fast path.
>+      uint64_t estimated_scaled_ns;
>+      int s = -1;
>+      do {
>+        s++;
>+        estimated_scaled_ns = (delta_cycles >> s) * sample->nsscaled_per_cycle;
>+      } while (estimated_scaled_ns / sample->nsscaled_per_cycle !=
>+               (delta_cycles >> s));
>+      estimated_base_ns = sample->base_ns +
>+                          (estimated_scaled_ns >> (kScale - s));
>+    }
>+
>+    // Compute the assumed cycle time kMinNSBetweenSamples ns into the future
>+    // assuming the cycle counter rate stays the same as the last interval.
>+    uint64_t ns = now_ns - sample->raw_ns;
>+    uint64_t measured_nsscaled_per_cycle = SafeDivideAndScale(ns, delta_cycles);
>+
>+    uint64_t assumed_next_sample_delta_cycles =
>+        SafeDivideAndScale(kMinNSBetweenSamples, measured_nsscaled_per_cycle);
>+
>+    int64_t diff_ns = now_ns - estimated_base_ns;  // estimate low by this much
>+
>+    // We want to set nsscaled_per_cycle so that our estimate of the ns time
>+    // at the assumed cycle time is the assumed ns time.
>+    // That is, we want to set nsscaled_per_cycle so:
>+    //  kMinNSBetweenSamples + diff_ns  ==
>+    //  (assumed_next_sample_delta_cycles * nsscaled_per_cycle) >> kScale
>+    // But we wish to damp oscillations, so instead correct only most
>+    // of our current error, by solving:
>+    //  kMinNSBetweenSamples + diff_ns - (diff_ns / 16) ==
>+    //  (assumed_next_sample_delta_cycles * nsscaled_per_cycle) >> kScale
>+    ns = kMinNSBetweenSamples + diff_ns - (diff_ns / 16);
>+    uint64_t new_nsscaled_per_cycle =
>+        SafeDivideAndScale(ns, assumed_next_sample_delta_cycles);
>+    if (new_nsscaled_per_cycle != 0 &&
>+        diff_ns < 100 * 1000 * 1000 && -diff_ns < 100 * 1000 * 1000) {
>+      // record the cycle time measurement
>+      last_sample.nsscaled_per_cycle.store(
>+          new_nsscaled_per_cycle, std::memory_order_relaxed);
>+      uint64_t new_min_cycles_per_sample =
>+          SafeDivideAndScale(kMinNSBetweenSamples, new_nsscaled_per_cycle);
>+      last_sample.min_cycles_per_sample.store(
>+          new_min_cycles_per_sample, std::memory_order_relaxed);
>+      stats_calibrations++;
>+    } else {  // something went wrong; forget the slope
>+      last_sample.nsscaled_per_cycle.store(0, std::memory_order_relaxed);
>+      last_sample.min_cycles_per_sample.store(0, std::memory_order_relaxed);
>+      estimated_base_ns = now_ns;
>+      stats_reinitializations++;
>+    }
>+    last_sample.raw_ns.store(now_ns, std::memory_order_relaxed);
>+    last_sample.base_ns.store(estimated_base_ns, std::memory_order_relaxed);
>+    last_sample.base_cycles.store(now_cycles, std::memory_order_relaxed);
>+  } else {
>+    // have a sample, but no slope; waiting for enough time for a calibration
>+    stats_slow_paths++;
>+  }
>+
>+  SeqRelease(&seq, lock_value);  // release the readers
>+
>+  return estimated_base_ns;
>+}
>+}  // namespace absl
>+#endif  // ABSL_USE_CYCLECLOCK_FOR_GET_CURRENT_TIME_NANOS
>+
>+namespace absl {
>+namespace {
>+
>+// Returns the maximum duration that SleepOnce() can sleep for.
>+constexpr absl::Duration MaxSleep() {
>+#ifdef _WIN32
>+  // Windows Sleep() takes unsigned long argument in milliseconds.
>+  return absl::Milliseconds(
>+      std::numeric_limits<unsigned long>::max());  // NOLINT(runtime/int)
>+#else
>+  return absl::Seconds(std::numeric_limits<time_t>::max());
>+#endif
>+}
>+
>+// Sleeps for the given duration.
>+// REQUIRES: to_sleep <= MaxSleep().
>+void SleepOnce(absl::Duration to_sleep) {
>+#ifdef _WIN32
>+  Sleep(to_sleep / absl::Milliseconds(1));
>+#else
>+  struct timespec sleep_time = absl::ToTimespec(to_sleep);
>+  while (nanosleep(&sleep_time, &sleep_time) != 0 && errno == EINTR) {
>+    // Ignore signals and wait for the full interval to elapse.
>+  }
>+#endif
>+}
>+
>+}  // namespace
>+}  // namespace absl
>+
>+extern "C" {
>+
>+ABSL_ATTRIBUTE_WEAK void AbslInternalSleepFor(absl::Duration duration) {
>+  while (duration > absl::ZeroDuration()) {
>+    absl::Duration to_sleep = std::min(duration, absl::MaxSleep());
>+    absl::SleepOnce(to_sleep);
>+    duration -= to_sleep;
>+  }
>+}
>+
>+}  // extern "C"
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/clock.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/clock.h
>new file mode 100644
>index 00000000000..3753d4ee43a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/clock.h
>@@ -0,0 +1,72 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// -----------------------------------------------------------------------------
>+// File: clock.h
>+// -----------------------------------------------------------------------------
>+//
>+// This header file contains utility functions for working with the system-wide
>+// realtime clock. For descriptions of the main time abstractions used within
>+// this header file, consult the time.h header file.
>+#ifndef ABSL_TIME_CLOCK_H_
>+#define ABSL_TIME_CLOCK_H_
>+
>+#include "absl/base/macros.h"
>+#include "absl/time/time.h"
>+
>+namespace absl {
>+
>+// Now()
>+//
>+// Returns the current time, expressed as an `absl::Time` absolute time value.
>+absl::Time Now();
>+
>+// GetCurrentTimeNanos()
>+//
>+// Returns the current time, expressed as a count of nanoseconds since the Unix
>+// Epoch (https://en.wikipedia.org/wiki/Unix_time). Prefer `absl::Now()` instead
>+// for all but the most performance-sensitive cases (i.e. when you are calling
>+// this function hundreds of thousands of times per second).
>+int64_t GetCurrentTimeNanos();
>+
>+// SleepFor()
>+//
>+// Sleeps for the specified duration, expressed as an `absl::Duration`.
>+//
>+// Notes:
>+// * Signal interruptions will not reduce the sleep duration.
>+// * Returns immediately when passed a nonpositive duration.
>+void SleepFor(absl::Duration duration);
>+
>+}  // namespace absl
>+
>+// -----------------------------------------------------------------------------
>+// Implementation Details
>+// -----------------------------------------------------------------------------
>+
>+// In some build configurations we pass --detect-odr-violations to the
>+// gold linker.  This causes it to flag weak symbol overrides as ODR
>+// violations.  Because ODR only applies to C++ and not C,
>+// --detect-odr-violations ignores symbols not mangled with C++ names.
>+// By changing our extension points to be extern "C", we dodge this
>+// check.
>+extern "C" {
>+void AbslInternalSleepFor(absl::Duration duration);
>+}  // extern "C"
>+
>+inline void absl::SleepFor(absl::Duration duration) {
>+  AbslInternalSleepFor(duration);
>+}
>+
>+#endif  // ABSL_TIME_CLOCK_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/clock_benchmark.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/clock_benchmark.cc
>new file mode 100644
>index 00000000000..3d3cd9d5c42
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/clock_benchmark.cc
>@@ -0,0 +1,72 @@
>+// Copyright 2018 The Abseil Authors.
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/time/clock.h"
>+
>+#if !defined(_WIN32)
>+#include <sys/time.h>
>+#endif  // _WIN32
>+#include <cstdio>
>+
>+#include "absl/base/internal/cycleclock.h"
>+#include "benchmark/benchmark.h"
>+
>+namespace {
>+
>+void BM_Clock_Now_AbslTime(benchmark::State& state) {
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(absl::Now());
>+  }
>+}
>+BENCHMARK(BM_Clock_Now_AbslTime);
>+
>+void BM_Clock_Now_GetCurrentTimeNanos(benchmark::State& state) {
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(absl::GetCurrentTimeNanos());
>+  }
>+}
>+BENCHMARK(BM_Clock_Now_GetCurrentTimeNanos);
>+
>+void BM_Clock_Now_AbslTime_ToUnixNanos(benchmark::State& state) {
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(absl::ToUnixNanos(absl::Now()));
>+  }
>+}
>+BENCHMARK(BM_Clock_Now_AbslTime_ToUnixNanos);
>+
>+void BM_Clock_Now_CycleClock(benchmark::State& state) {
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(absl::base_internal::CycleClock::Now());
>+  }
>+}
>+BENCHMARK(BM_Clock_Now_CycleClock);
>+
>+#if !defined(_WIN32)
>+static void BM_Clock_Now_gettimeofday(benchmark::State& state) {
>+  struct timeval tv;
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(gettimeofday(&tv, nullptr));
>+  }
>+}
>+BENCHMARK(BM_Clock_Now_gettimeofday);
>+
>+static void BM_Clock_Now_clock_gettime(benchmark::State& state) {
>+  struct timespec ts;
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(clock_gettime(CLOCK_REALTIME, &ts));
>+  }
>+}
>+BENCHMARK(BM_Clock_Now_clock_gettime);
>+#endif  // _WIN32
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/clock_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/clock_test.cc
>new file mode 100644
>index 00000000000..707166d0c28
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/clock_test.cc
>@@ -0,0 +1,118 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/time/clock.h"
>+
>+#include "absl/base/config.h"
>+#if defined(ABSL_HAVE_ALARM)
>+#include <signal.h>
>+#include <unistd.h>
>+#elif defined(__linux__) || defined(__APPLE__)
>+#error all known Linux and Apple targets have alarm
>+#endif
>+
>+#include "gtest/gtest.h"
>+#include "absl/time/time.h"
>+
>+namespace {
>+
>+TEST(Time, Now) {
>+  const absl::Time before = absl::FromUnixNanos(absl::GetCurrentTimeNanos());
>+  const absl::Time now = absl::Now();
>+  const absl::Time after = absl::FromUnixNanos(absl::GetCurrentTimeNanos());
>+  EXPECT_GE(now, before);
>+  EXPECT_GE(after, now);
>+}
>+
>+enum class AlarmPolicy { kWithoutAlarm, kWithAlarm };
>+
>+#if defined(ABSL_HAVE_ALARM)
>+bool alarm_handler_invoked = false;
>+
>+void AlarmHandler(int signo) {
>+  ASSERT_EQ(signo, SIGALRM);
>+  alarm_handler_invoked = true;
>+}
>+#endif
>+
>+// Does SleepFor(d) take between lower_bound and upper_bound at least
>+// once between now and (now + timeout)?  If requested (and supported),
>+// add an alarm for the middle of the sleep period and expect it to fire.
>+bool SleepForBounded(absl::Duration d, absl::Duration lower_bound,
>+                     absl::Duration upper_bound, absl::Duration timeout,
>+                     AlarmPolicy alarm_policy, int* attempts) {
>+  const absl::Time deadline = absl::Now() + timeout;
>+  while (absl::Now() < deadline) {
>+#if defined(ABSL_HAVE_ALARM)
>+    sig_t old_alarm = SIG_DFL;
>+    if (alarm_policy == AlarmPolicy::kWithAlarm) {
>+      alarm_handler_invoked = false;
>+      old_alarm = signal(SIGALRM, AlarmHandler);
>+      alarm(absl::ToInt64Seconds(d / 2));
>+    }
>+#else
>+    EXPECT_EQ(alarm_policy, AlarmPolicy::kWithoutAlarm);
>+#endif
>+    ++*attempts;
>+    absl::Time start = absl::Now();
>+    absl::SleepFor(d);
>+    absl::Duration actual = absl::Now() - start;
>+#if defined(ABSL_HAVE_ALARM)
>+    if (alarm_policy == AlarmPolicy::kWithAlarm) {
>+      signal(SIGALRM, old_alarm);
>+      if (!alarm_handler_invoked) continue;
>+    }
>+#endif
>+    if (lower_bound <= actual && actual <= upper_bound) {
>+      return true;  // yes, the SleepFor() was correctly bounded
>+    }
>+  }
>+  return false;
>+}
>+
>+testing::AssertionResult AssertSleepForBounded(absl::Duration d,
>+                                               absl::Duration early,
>+                                               absl::Duration late,
>+                                               absl::Duration timeout,
>+                                               AlarmPolicy alarm_policy) {
>+  const absl::Duration lower_bound = d - early;
>+  const absl::Duration upper_bound = d + late;
>+  int attempts = 0;
>+  if (SleepForBounded(d, lower_bound, upper_bound, timeout, alarm_policy,
>+                      &attempts)) {
>+    return testing::AssertionSuccess();
>+  }
>+  return testing::AssertionFailure()
>+         << "SleepFor(" << d << ") did not return within [" << lower_bound
>+         << ":" << upper_bound << "] in " << attempts << " attempt"
>+         << (attempts == 1 ? "" : "s") << " over " << timeout
>+         << (alarm_policy == AlarmPolicy::kWithAlarm ? " with" : " without")
>+         << " an alarm";
>+}
>+
>+// Tests that SleepFor() returns neither too early nor too late.
>+TEST(SleepFor, Bounded) {
>+  const absl::Duration d = absl::Milliseconds(2500);
>+  const absl::Duration early = absl::Milliseconds(100);
>+  const absl::Duration late = absl::Milliseconds(300);
>+  const absl::Duration timeout = 48 * d;
>+  EXPECT_TRUE(AssertSleepForBounded(d, early, late, timeout,
>+                                    AlarmPolicy::kWithoutAlarm));
>+#if defined(ABSL_HAVE_ALARM)
>+  EXPECT_TRUE(AssertSleepForBounded(d, early, late, timeout,
>+                                    AlarmPolicy::kWithAlarm));
>+#endif
>+}
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/duration.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/duration.cc
>new file mode 100644
>index 00000000000..f402137b0a6
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/duration.cc
>@@ -0,0 +1,904 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+// The implementation of the absl::Duration class, which is declared in
>+// //absl/time.h.  This class behaves like a numeric type; it has no public
>+// methods and is used only through the operators defined here.
>+//
>+// Implementation notes:
>+//
>+// An absl::Duration is represented as
>+//
>+//   rep_hi_ : (int64_t)  Whole seconds
>+//   rep_lo_ : (uint32_t) Fractions of a second
>+//
>+// The seconds value (rep_hi_) may be positive or negative as appropriate.
>+// The fractional seconds (rep_lo_) is always a positive offset from rep_hi_.
>+// The API for Duration guarantees at least nanosecond resolution, which
>+// means rep_lo_ could have a max value of 1B - 1 if it stored nanoseconds.
>+// However, to utilize more of the available 32 bits of space in rep_lo_,
>+// we instead store quarters of a nanosecond in rep_lo_ resulting in a max
>+// value of 4B - 1.  This allows us to correctly handle calculations like
>+// 0.5 nanos + 0.5 nanos = 1 nano.  The following example shows the actual
>+// Duration rep using quarters of a nanosecond.
>+//
>+//    2.5 sec = {rep_hi_=2,  rep_lo_=2000000000}  // lo = 4 * 500000000
>+//   -2.5 sec = {rep_hi_=-3, rep_lo_=2000000000}
>+//
>+// Infinite durations are represented as Durations with the rep_lo_ field set
>+// to all 1s.
>+//
>+//   +InfiniteDuration:
>+//     rep_hi_ : kint64max
>+//     rep_lo_ : ~0U
>+//
>+//   -InfiniteDuration:
>+//     rep_hi_ : kint64min
>+//     rep_lo_ : ~0U
>+//
>+// Arithmetic overflows/underflows to +/- infinity and saturates.
>+
>+#include <algorithm>
>+#include <cassert>
>+#include <cctype>
>+#include <cerrno>
>+#include <cmath>
>+#include <cstdint>
>+#include <cstdlib>
>+#include <cstring>
>+#include <ctime>
>+#include <functional>
>+#include <limits>
>+#include <string>
>+
>+#include "absl/base/casts.h"
>+#include "absl/numeric/int128.h"
>+#include "absl/time/time.h"
>+
>+namespace absl {
>+
>+namespace {
>+
>+using time_internal::kTicksPerNanosecond;
>+using time_internal::kTicksPerSecond;
>+
>+constexpr int64_t kint64max = std::numeric_limits<int64_t>::max();
>+constexpr int64_t kint64min = std::numeric_limits<int64_t>::min();
>+
>+// Can't use std::isinfinite() because it doesn't exist on windows.
>+inline bool IsFinite(double d) {
>+  return d != std::numeric_limits<double>::infinity() &&
>+         d != -std::numeric_limits<double>::infinity();
>+}
>+
>+// Can't use std::round() because it is only available in C++11.
>+// Note that we ignore the possibility of floating-point over/underflow.
>+template <typename Double>
>+inline double Round(Double d) {
>+  return d < 0 ? std::ceil(d - 0.5) : std::floor(d + 0.5);
>+}
>+
>+// *sec may be positive or negative.  *ticks must be in the range
>+// -kTicksPerSecond < *ticks < kTicksPerSecond.  If *ticks is negative it
>+// will be normalized to a positive value by adjusting *sec accordingly.
>+inline void NormalizeTicks(int64_t* sec, int64_t* ticks) {
>+  if (*ticks < 0) {
>+    --*sec;
>+    *ticks += kTicksPerSecond;
>+  }
>+}
>+
>+// Makes a uint128 from the absolute value of the given scalar.
>+inline uint128 MakeU128(int64_t a) {
>+  uint128 u128 = 0;
>+  if (a < 0) {
>+    ++u128;
>+    ++a;  // Makes it safe to negate 'a'
>+    a = -a;
>+  }
>+  u128 += static_cast<uint64_t>(a);
>+  return u128;
>+}
>+
>+// Makes a uint128 count of ticks out of the absolute value of the Duration.
>+inline uint128 MakeU128Ticks(Duration d) {
>+  int64_t rep_hi = time_internal::GetRepHi(d);
>+  uint32_t rep_lo = time_internal::GetRepLo(d);
>+  if (rep_hi < 0) {
>+    ++rep_hi;
>+    rep_hi = -rep_hi;
>+    rep_lo = kTicksPerSecond - rep_lo;
>+  }
>+  uint128 u128 = static_cast<uint64_t>(rep_hi);
>+  u128 *= static_cast<uint64_t>(kTicksPerSecond);
>+  u128 += rep_lo;
>+  return u128;
>+}
>+
>+// Breaks a uint128 of ticks into a Duration.
>+inline Duration MakeDurationFromU128(uint128 u128, bool is_neg) {
>+  int64_t rep_hi;
>+  uint32_t rep_lo;
>+  const uint64_t h64 = Uint128High64(u128);
>+  const uint64_t l64 = Uint128Low64(u128);
>+  if (h64 == 0) {  // fastpath
>+    const uint64_t hi = l64 / kTicksPerSecond;
>+    rep_hi = static_cast<int64_t>(hi);
>+    rep_lo = static_cast<uint32_t>(l64 - hi * kTicksPerSecond);
>+  } else {
>+    // kMaxRepHi64 is the high 64 bits of (2^63 * kTicksPerSecond).
>+    // Any positive tick count whose high 64 bits are >= kMaxRepHi64
>+    // is not representable as a Duration.  A negative tick count can
>+    // have its high 64 bits == kMaxRepHi64 but only when the low 64
>+    // bits are all zero, otherwise it is not representable either.
>+    const uint64_t kMaxRepHi64 = 0x77359400UL;
>+    if (h64 >= kMaxRepHi64) {
>+      if (is_neg && h64 == kMaxRepHi64 && l64 == 0) {
>+        // Avoid trying to represent -kint64min below.
>+        return time_internal::MakeDuration(kint64min);
>+      }
>+      return is_neg ? -InfiniteDuration() : InfiniteDuration();
>+    }
>+    const uint128 kTicksPerSecond128 = static_cast<uint64_t>(kTicksPerSecond);
>+    const uint128 hi = u128 / kTicksPerSecond128;
>+    rep_hi = static_cast<int64_t>(Uint128Low64(hi));
>+    rep_lo =
>+        static_cast<uint32_t>(Uint128Low64(u128 - hi * kTicksPerSecond128));
>+  }
>+  if (is_neg) {
>+    rep_hi = -rep_hi;
>+    if (rep_lo != 0) {
>+      --rep_hi;
>+      rep_lo = kTicksPerSecond - rep_lo;
>+    }
>+  }
>+  return time_internal::MakeDuration(rep_hi, rep_lo);
>+}
>+
>+// Convert between int64_t and uint64_t, preserving representation. This
>+// allows us to do arithmetic in the unsigned domain, where overflow has
>+// well-defined behavior. See operator+=() and operator-=().
>+//
>+// C99 7.20.1.1.1, as referenced by C++11 18.4.1.2, says, "The typedef
>+// name intN_t designates a signed integer type with width N, no padding
>+// bits, and a two's complement representation." So, we can convert to
>+// and from the corresponding uint64_t value using a bit cast.
>+inline uint64_t EncodeTwosComp(int64_t v) {
>+  return absl::bit_cast<uint64_t>(v);
>+}
>+inline int64_t DecodeTwosComp(uint64_t v) { return absl::bit_cast<int64_t>(v); }
>+
>+// Note: The overflow detection in this function is done using greater/less *or
>+// equal* because kint64max/min is too large to be represented exactly in a
>+// double (which only has 53 bits of precision). In order to avoid assigning to
>+// rep->hi a double value that is too large for an int64_t (and therefore is
>+// undefined), we must consider computations that equal kint64max/min as a
>+// double as overflow cases.
>+inline bool SafeAddRepHi(double a_hi, double b_hi, Duration* d) {
>+  double c = a_hi + b_hi;
>+  if (c >= kint64max) {
>+    *d = InfiniteDuration();
>+    return false;
>+  }
>+  if (c <= kint64min) {
>+    *d = -InfiniteDuration();
>+    return false;
>+  }
>+  *d = time_internal::MakeDuration(c, time_internal::GetRepLo(*d));
>+  return true;
>+}
>+
>+// A functor that's similar to std::multiplies<T>, except this returns the max
>+// T value instead of overflowing. This is only defined for uint128.
>+template <typename Ignored>
>+struct SafeMultiply {
>+  uint128 operator()(uint128 a, uint128 b) const {
>+    // b hi is always zero because it originated as an int64_t.
>+    assert(Uint128High64(b) == 0);
>+    // Fastpath to avoid the expensive overflow check with division.
>+    if (Uint128High64(a) == 0) {
>+      return (((Uint128Low64(a) | Uint128Low64(b)) >> 32) == 0)
>+                 ? static_cast<uint128>(Uint128Low64(a) * Uint128Low64(b))
>+                 : a * b;
>+    }
>+    return b == 0 ? b : (a > kuint128max / b) ? kuint128max : a * b;
>+  }
>+};
>+
>+// Scales (i.e., multiplies or divides, depending on the Operation template)
>+// the Duration d by the int64_t r.
>+template <template <typename> class Operation>
>+inline Duration ScaleFixed(Duration d, int64_t r) {
>+  const uint128 a = MakeU128Ticks(d);
>+  const uint128 b = MakeU128(r);
>+  const uint128 q = Operation<uint128>()(a, b);
>+  const bool is_neg = (time_internal::GetRepHi(d) < 0) != (r < 0);
>+  return MakeDurationFromU128(q, is_neg);
>+}
>+
>+// Scales (i.e., multiplies or divides, depending on the Operation template)
>+// the Duration d by the double r.
>+template <template <typename> class Operation>
>+inline Duration ScaleDouble(Duration d, double r) {
>+  Operation<double> op;
>+  double hi_doub = op(time_internal::GetRepHi(d), r);
>+  double lo_doub = op(time_internal::GetRepLo(d), r);
>+
>+  double hi_int = 0;
>+  double hi_frac = std::modf(hi_doub, &hi_int);
>+
>+  // Moves hi's fractional bits to lo.
>+  lo_doub /= kTicksPerSecond;
>+  lo_doub += hi_frac;
>+
>+  double lo_int = 0;
>+  double lo_frac = std::modf(lo_doub, &lo_int);
>+
>+  // Rolls lo into hi if necessary.
>+  int64_t lo64 = Round(lo_frac * kTicksPerSecond);
>+
>+  Duration ans;
>+  if (!SafeAddRepHi(hi_int, lo_int, &ans)) return ans;
>+  int64_t hi64 = time_internal::GetRepHi(ans);
>+  if (!SafeAddRepHi(hi64, lo64 / kTicksPerSecond, &ans)) return ans;
>+  hi64 = time_internal::GetRepHi(ans);
>+  lo64 %= kTicksPerSecond;
>+  NormalizeTicks(&hi64, &lo64);
>+  return time_internal::MakeDuration(hi64, lo64);
>+}
>+
>+// Tries to divide num by den as fast as possible by looking for common, easy
>+// cases. If the division was done, the quotient is in *q and the remainder is
>+// in *rem and true will be returned.
>+inline bool IDivFastPath(const Duration num, const Duration den, int64_t* q,
>+                         Duration* rem) {
>+  // Bail if num or den is an infinity.
>+  if (time_internal::IsInfiniteDuration(num) ||
>+      time_internal::IsInfiniteDuration(den))
>+    return false;
>+
>+  int64_t num_hi = time_internal::GetRepHi(num);
>+  uint32_t num_lo = time_internal::GetRepLo(num);
>+  int64_t den_hi = time_internal::GetRepHi(den);
>+  uint32_t den_lo = time_internal::GetRepLo(den);
>+
>+  if (den_hi == 0 && den_lo == kTicksPerNanosecond) {
>+    // Dividing by 1ns
>+    if (num_hi >= 0 && num_hi < (kint64max - kTicksPerSecond) / 1000000000) {
>+      *q = num_hi * 1000000000 + num_lo / kTicksPerNanosecond;
>+      *rem = time_internal::MakeDuration(0, num_lo % den_lo);
>+      return true;
>+    }
>+  } else if (den_hi == 0 && den_lo == 100 * kTicksPerNanosecond) {
>+    // Dividing by 100ns (common when converting to Universal time)
>+    if (num_hi >= 0 && num_hi < (kint64max - kTicksPerSecond) / 10000000) {
>+      *q = num_hi * 10000000 + num_lo / (100 * kTicksPerNanosecond);
>+      *rem = time_internal::MakeDuration(0, num_lo % den_lo);
>+      return true;
>+    }
>+  } else if (den_hi == 0 && den_lo == 1000 * kTicksPerNanosecond) {
>+    // Dividing by 1us
>+    if (num_hi >= 0 && num_hi < (kint64max - kTicksPerSecond) / 1000000) {
>+      *q = num_hi * 1000000 + num_lo / (1000 * kTicksPerNanosecond);
>+      *rem = time_internal::MakeDuration(0, num_lo % den_lo);
>+      return true;
>+    }
>+  } else if (den_hi == 0 && den_lo == 1000000 * kTicksPerNanosecond) {
>+    // Dividing by 1ms
>+    if (num_hi >= 0 && num_hi < (kint64max - kTicksPerSecond) / 1000) {
>+      *q = num_hi * 1000 + num_lo / (1000000 * kTicksPerNanosecond);
>+      *rem = time_internal::MakeDuration(0, num_lo % den_lo);
>+      return true;
>+    }
>+  } else if (den_hi > 0 && den_lo == 0) {
>+    // Dividing by positive multiple of 1s
>+    if (num_hi >= 0) {
>+      if (den_hi == 1) {
>+        *q = num_hi;
>+        *rem = time_internal::MakeDuration(0, num_lo);
>+        return true;
>+      }
>+      *q = num_hi / den_hi;
>+      *rem = time_internal::MakeDuration(num_hi % den_hi, num_lo);
>+      return true;
>+    }
>+    if (num_lo != 0) {
>+      num_hi += 1;
>+    }
>+    int64_t quotient = num_hi / den_hi;
>+    int64_t rem_sec = num_hi % den_hi;
>+    if (rem_sec > 0) {
>+      rem_sec -= den_hi;
>+      quotient += 1;
>+    }
>+    if (num_lo != 0) {
>+      rem_sec -= 1;
>+    }
>+    *q = quotient;
>+    *rem = time_internal::MakeDuration(rem_sec, num_lo);
>+    return true;
>+  }
>+
>+  return false;
>+}
>+
>+}  // namespace
>+
>+namespace time_internal {
>+
>+// The 'satq' argument indicates whether the quotient should saturate at the
>+// bounds of int64_t.  If it does saturate, the difference will spill over to
>+// the remainder.  If it does not saturate, the remainder remain accurate,
>+// but the returned quotient will over/underflow int64_t and should not be used.
>+int64_t IDivDuration(bool satq, const Duration num, const Duration den,
>+                   Duration* rem) {
>+  int64_t q = 0;
>+  if (IDivFastPath(num, den, &q, rem)) {
>+    return q;
>+  }
>+
>+  const bool num_neg = num < ZeroDuration();
>+  const bool den_neg = den < ZeroDuration();
>+  const bool quotient_neg = num_neg != den_neg;
>+
>+  if (time_internal::IsInfiniteDuration(num) || den == ZeroDuration()) {
>+    *rem = num_neg ? -InfiniteDuration() : InfiniteDuration();
>+    return quotient_neg ? kint64min : kint64max;
>+  }
>+  if (time_internal::IsInfiniteDuration(den)) {
>+    *rem = num;
>+    return 0;
>+  }
>+
>+  const uint128 a = MakeU128Ticks(num);
>+  const uint128 b = MakeU128Ticks(den);
>+  uint128 quotient128 = a / b;
>+
>+  if (satq) {
>+    // Limits the quotient to the range of int64_t.
>+    if (quotient128 > uint128(static_cast<uint64_t>(kint64max))) {
>+      quotient128 = quotient_neg ? uint128(static_cast<uint64_t>(kint64min))
>+                                 : uint128(static_cast<uint64_t>(kint64max));
>+    }
>+  }
>+
>+  const uint128 remainder128 = a - quotient128 * b;
>+  *rem = MakeDurationFromU128(remainder128, num_neg);
>+
>+  if (!quotient_neg || quotient128 == 0) {
>+    return Uint128Low64(quotient128) & kint64max;
>+  }
>+  // The quotient needs to be negated, but we need to carefully handle
>+  // quotient128s with the top bit on.
>+  return -static_cast<int64_t>(Uint128Low64(quotient128 - 1) & kint64max) - 1;
>+}
>+
>+}  // namespace time_internal
>+
>+//
>+// Additive operators.
>+//
>+
>+Duration& Duration::operator+=(Duration rhs) {
>+  if (time_internal::IsInfiniteDuration(*this)) return *this;
>+  if (time_internal::IsInfiniteDuration(rhs)) return *this = rhs;
>+  const int64_t orig_rep_hi = rep_hi_;
>+  rep_hi_ =
>+      DecodeTwosComp(EncodeTwosComp(rep_hi_) + EncodeTwosComp(rhs.rep_hi_));
>+  if (rep_lo_ >= kTicksPerSecond - rhs.rep_lo_) {
>+    rep_hi_ = DecodeTwosComp(EncodeTwosComp(rep_hi_) + 1);
>+    rep_lo_ -= kTicksPerSecond;
>+  }
>+  rep_lo_ += rhs.rep_lo_;
>+  if (rhs.rep_hi_ < 0 ? rep_hi_ > orig_rep_hi : rep_hi_ < orig_rep_hi) {
>+    return *this = rhs.rep_hi_ < 0 ? -InfiniteDuration() : InfiniteDuration();
>+  }
>+  return *this;
>+}
>+
>+Duration& Duration::operator-=(Duration rhs) {
>+  if (time_internal::IsInfiniteDuration(*this)) return *this;
>+  if (time_internal::IsInfiniteDuration(rhs)) {
>+    return *this = rhs.rep_hi_ >= 0 ? -InfiniteDuration() : InfiniteDuration();
>+  }
>+  const int64_t orig_rep_hi = rep_hi_;
>+  rep_hi_ =
>+      DecodeTwosComp(EncodeTwosComp(rep_hi_) - EncodeTwosComp(rhs.rep_hi_));
>+  if (rep_lo_ < rhs.rep_lo_) {
>+    rep_hi_ = DecodeTwosComp(EncodeTwosComp(rep_hi_) - 1);
>+    rep_lo_ += kTicksPerSecond;
>+  }
>+  rep_lo_ -= rhs.rep_lo_;
>+  if (rhs.rep_hi_ < 0 ? rep_hi_ < orig_rep_hi : rep_hi_ > orig_rep_hi) {
>+    return *this = rhs.rep_hi_ >= 0 ? -InfiniteDuration() : InfiniteDuration();
>+  }
>+  return *this;
>+}
>+
>+//
>+// Multiplicative operators.
>+//
>+
>+Duration& Duration::operator*=(int64_t r) {
>+  if (time_internal::IsInfiniteDuration(*this)) {
>+    const bool is_neg = (r < 0) != (rep_hi_ < 0);
>+    return *this = is_neg ? -InfiniteDuration() : InfiniteDuration();
>+  }
>+  return *this = ScaleFixed<SafeMultiply>(*this, r);
>+}
>+
>+Duration& Duration::operator*=(double r) {
>+  if (time_internal::IsInfiniteDuration(*this) || !IsFinite(r)) {
>+    const bool is_neg = (std::signbit(r) != 0) != (rep_hi_ < 0);
>+    return *this = is_neg ? -InfiniteDuration() : InfiniteDuration();
>+  }
>+  return *this = ScaleDouble<std::multiplies>(*this, r);
>+}
>+
>+Duration& Duration::operator/=(int64_t r) {
>+  if (time_internal::IsInfiniteDuration(*this) || r == 0) {
>+    const bool is_neg = (r < 0) != (rep_hi_ < 0);
>+    return *this = is_neg ? -InfiniteDuration() : InfiniteDuration();
>+  }
>+  return *this = ScaleFixed<std::divides>(*this, r);
>+}
>+
>+Duration& Duration::operator/=(double r) {
>+  if (time_internal::IsInfiniteDuration(*this) || r == 0.0) {
>+    const bool is_neg = (std::signbit(r) != 0) != (rep_hi_ < 0);
>+    return *this = is_neg ? -InfiniteDuration() : InfiniteDuration();
>+  }
>+  return *this = ScaleDouble<std::divides>(*this, r);
>+}
>+
>+Duration& Duration::operator%=(Duration rhs) {
>+  time_internal::IDivDuration(false, *this, rhs, this);
>+  return *this;
>+}
>+
>+double FDivDuration(Duration num, Duration den) {
>+  // Arithmetic with infinity is sticky.
>+  if (time_internal::IsInfiniteDuration(num) || den == ZeroDuration()) {
>+    return (num < ZeroDuration()) == (den < ZeroDuration())
>+               ? std::numeric_limits<double>::infinity()
>+               : -std::numeric_limits<double>::infinity();
>+  }
>+  if (time_internal::IsInfiniteDuration(den)) return 0.0;
>+
>+  double a =
>+      static_cast<double>(time_internal::GetRepHi(num)) * kTicksPerSecond +
>+      time_internal::GetRepLo(num);
>+  double b =
>+      static_cast<double>(time_internal::GetRepHi(den)) * kTicksPerSecond +
>+      time_internal::GetRepLo(den);
>+  return a / b;
>+}
>+
>+//
>+// Trunc/Floor/Ceil.
>+//
>+
>+Duration Trunc(Duration d, Duration unit) {
>+  return d - (d % unit);
>+}
>+
>+Duration Floor(const Duration d, const Duration unit) {
>+  const absl::Duration td = Trunc(d, unit);
>+  return td <= d ? td : td - AbsDuration(unit);
>+}
>+
>+Duration Ceil(const Duration d, const Duration unit) {
>+  const absl::Duration td = Trunc(d, unit);
>+  return td >= d ? td : td + AbsDuration(unit);
>+}
>+
>+//
>+// Factory functions.
>+//
>+
>+Duration DurationFromTimespec(timespec ts) {
>+  if (static_cast<uint64_t>(ts.tv_nsec) < 1000 * 1000 * 1000) {
>+    int64_t ticks = ts.tv_nsec * kTicksPerNanosecond;
>+    return time_internal::MakeDuration(ts.tv_sec, ticks);
>+  }
>+  return Seconds(ts.tv_sec) + Nanoseconds(ts.tv_nsec);
>+}
>+
>+Duration DurationFromTimeval(timeval tv) {
>+  if (static_cast<uint64_t>(tv.tv_usec) < 1000 * 1000) {
>+    int64_t ticks = tv.tv_usec * 1000 * kTicksPerNanosecond;
>+    return time_internal::MakeDuration(tv.tv_sec, ticks);
>+  }
>+  return Seconds(tv.tv_sec) + Microseconds(tv.tv_usec);
>+}
>+
>+//
>+// Conversion to other duration types.
>+//
>+
>+int64_t ToInt64Nanoseconds(Duration d) {
>+  if (time_internal::GetRepHi(d) >= 0 &&
>+      time_internal::GetRepHi(d) >> 33 == 0) {
>+    return (time_internal::GetRepHi(d) * 1000 * 1000 * 1000) +
>+           (time_internal::GetRepLo(d) / kTicksPerNanosecond);
>+  }
>+  return d / Nanoseconds(1);
>+}
>+int64_t ToInt64Microseconds(Duration d) {
>+  if (time_internal::GetRepHi(d) >= 0 &&
>+      time_internal::GetRepHi(d) >> 43 == 0) {
>+    return (time_internal::GetRepHi(d) * 1000 * 1000) +
>+           (time_internal::GetRepLo(d) / (kTicksPerNanosecond * 1000));
>+  }
>+  return d / Microseconds(1);
>+}
>+int64_t ToInt64Milliseconds(Duration d) {
>+  if (time_internal::GetRepHi(d) >= 0 &&
>+      time_internal::GetRepHi(d) >> 53 == 0) {
>+    return (time_internal::GetRepHi(d) * 1000) +
>+           (time_internal::GetRepLo(d) / (kTicksPerNanosecond * 1000 * 1000));
>+  }
>+  return d / Milliseconds(1);
>+}
>+int64_t ToInt64Seconds(Duration d) {
>+  int64_t hi = time_internal::GetRepHi(d);
>+  if (time_internal::IsInfiniteDuration(d)) return hi;
>+  if (hi < 0 && time_internal::GetRepLo(d) != 0) ++hi;
>+  return hi;
>+}
>+int64_t ToInt64Minutes(Duration d) {
>+  int64_t hi = time_internal::GetRepHi(d);
>+  if (time_internal::IsInfiniteDuration(d)) return hi;
>+  if (hi < 0 && time_internal::GetRepLo(d) != 0) ++hi;
>+  return hi / 60;
>+}
>+int64_t ToInt64Hours(Duration d) {
>+  int64_t hi = time_internal::GetRepHi(d);
>+  if (time_internal::IsInfiniteDuration(d)) return hi;
>+  if (hi < 0 && time_internal::GetRepLo(d) != 0) ++hi;
>+  return hi / (60 * 60);
>+}
>+
>+double ToDoubleNanoseconds(Duration d) {
>+  return FDivDuration(d, Nanoseconds(1));
>+}
>+double ToDoubleMicroseconds(Duration d) {
>+  return FDivDuration(d, Microseconds(1));
>+}
>+double ToDoubleMilliseconds(Duration d) {
>+  return FDivDuration(d, Milliseconds(1));
>+}
>+double ToDoubleSeconds(Duration d) {
>+  return FDivDuration(d, Seconds(1));
>+}
>+double ToDoubleMinutes(Duration d) {
>+  return FDivDuration(d, Minutes(1));
>+}
>+double ToDoubleHours(Duration d) {
>+  return FDivDuration(d, Hours(1));
>+}
>+
>+timespec ToTimespec(Duration d) {
>+  timespec ts;
>+  if (!time_internal::IsInfiniteDuration(d)) {
>+    int64_t rep_hi = time_internal::GetRepHi(d);
>+    uint32_t rep_lo = time_internal::GetRepLo(d);
>+    if (rep_hi < 0) {
>+      // Tweak the fields so that unsigned division of rep_lo
>+      // maps to truncation (towards zero) for the timespec.
>+      rep_lo += kTicksPerNanosecond - 1;
>+      if (rep_lo >= kTicksPerSecond) {
>+        rep_hi += 1;
>+        rep_lo -= kTicksPerSecond;
>+      }
>+    }
>+    ts.tv_sec = rep_hi;
>+    if (ts.tv_sec == rep_hi) {  // no time_t narrowing
>+      ts.tv_nsec = rep_lo / kTicksPerNanosecond;
>+      return ts;
>+    }
>+  }
>+  if (d >= ZeroDuration()) {
>+    ts.tv_sec = std::numeric_limits<time_t>::max();
>+    ts.tv_nsec = 1000 * 1000 * 1000 - 1;
>+  } else {
>+    ts.tv_sec = std::numeric_limits<time_t>::min();
>+    ts.tv_nsec = 0;
>+  }
>+  return ts;
>+}
>+
>+timeval ToTimeval(Duration d) {
>+  timeval tv;
>+  timespec ts = ToTimespec(d);
>+  if (ts.tv_sec < 0) {
>+    // Tweak the fields so that positive division of tv_nsec
>+    // maps to truncation (towards zero) for the timeval.
>+    ts.tv_nsec += 1000 - 1;
>+    if (ts.tv_nsec >= 1000 * 1000 * 1000) {
>+      ts.tv_sec += 1;
>+      ts.tv_nsec -= 1000 * 1000 * 1000;
>+    }
>+  }
>+  tv.tv_sec = ts.tv_sec;
>+  if (tv.tv_sec != ts.tv_sec) {  // narrowing
>+    if (ts.tv_sec < 0) {
>+      tv.tv_sec = std::numeric_limits<decltype(tv.tv_sec)>::min();
>+      tv.tv_usec = 0;
>+    } else {
>+      tv.tv_sec = std::numeric_limits<decltype(tv.tv_sec)>::max();
>+      tv.tv_usec = 1000 * 1000 - 1;
>+    }
>+    return tv;
>+  }
>+  tv.tv_usec = static_cast<int>(ts.tv_nsec / 1000);  // suseconds_t
>+  return tv;
>+}
>+
>+std::chrono::nanoseconds ToChronoNanoseconds(Duration d) {
>+  return time_internal::ToChronoDuration<std::chrono::nanoseconds>(d);
>+}
>+std::chrono::microseconds ToChronoMicroseconds(Duration d) {
>+  return time_internal::ToChronoDuration<std::chrono::microseconds>(d);
>+}
>+std::chrono::milliseconds ToChronoMilliseconds(Duration d) {
>+  return time_internal::ToChronoDuration<std::chrono::milliseconds>(d);
>+}
>+std::chrono::seconds ToChronoSeconds(Duration d) {
>+  return time_internal::ToChronoDuration<std::chrono::seconds>(d);
>+}
>+std::chrono::minutes ToChronoMinutes(Duration d) {
>+  return time_internal::ToChronoDuration<std::chrono::minutes>(d);
>+}
>+std::chrono::hours ToChronoHours(Duration d) {
>+  return time_internal::ToChronoDuration<std::chrono::hours>(d);
>+}
>+
>+//
>+// To/From std::string formatting.
>+//
>+
>+namespace {
>+
>+// Formats a positive 64-bit integer in the given field width.  Note that
>+// it is up to the caller of Format64() to ensure that there is sufficient
>+// space before ep to hold the conversion.
>+char* Format64(char* ep, int width, int64_t v) {
>+  do {
>+    --width;
>+    *--ep = '0' + (v % 10);  // contiguous digits
>+  } while (v /= 10);
>+  while (--width >= 0) *--ep = '0';  // zero pad
>+  return ep;
>+}
>+
>+// Helpers for FormatDuration() that format 'n' and append it to 'out'
>+// followed by the given 'unit'.  If 'n' formats to "0", nothing is
>+// appended (not even the unit).
>+
>+// A type that encapsulates how to display a value of a particular unit. For
>+// values that are displayed with fractional parts, the precision indicates
>+// where to round the value. The precision varies with the display unit because
>+// a Duration can hold only quarters of a nanosecond, so displaying information
>+// beyond that is just noise.
>+//
>+// For example, a microsecond value of 42.00025xxxxx should not display beyond 5
>+// fractional digits, because it is in the noise of what a Duration can
>+// represent.
>+struct DisplayUnit {
>+  const char* abbr;
>+  int prec;
>+  double pow10;
>+};
>+const DisplayUnit kDisplayNano = {"ns", 2, 1e2};
>+const DisplayUnit kDisplayMicro = {"us", 5, 1e5};
>+const DisplayUnit kDisplayMilli = {"ms", 8, 1e8};
>+const DisplayUnit kDisplaySec = {"s", 11, 1e11};
>+const DisplayUnit kDisplayMin = {"m", -1, 0.0};   // prec ignored
>+const DisplayUnit kDisplayHour = {"h", -1, 0.0};  // prec ignored
>+
>+void AppendNumberUnit(std::string* out, int64_t n, DisplayUnit unit) {
>+  char buf[sizeof("2562047788015216")];  // hours in max duration
>+  char* const ep = buf + sizeof(buf);
>+  char* bp = Format64(ep, 0, n);
>+  if (*bp != '0' || bp + 1 != ep) {
>+    out->append(bp, ep - bp);
>+    out->append(unit.abbr);
>+  }
>+}
>+
>+// Note: unit.prec is limited to double's digits10 value (typically 15) so it
>+// always fits in buf[].
>+void AppendNumberUnit(std::string* out, double n, DisplayUnit unit) {
>+  const int buf_size = std::numeric_limits<double>::digits10;
>+  const int prec = std::min(buf_size, unit.prec);
>+  char buf[buf_size];  // also large enough to hold integer part
>+  char* ep = buf + sizeof(buf);
>+  double d = 0;
>+  int64_t frac_part = Round(std::modf(n, &d) * unit.pow10);
>+  int64_t int_part = d;
>+  if (int_part != 0 || frac_part != 0) {
>+    char* bp = Format64(ep, 0, int_part);  // always < 1000
>+    out->append(bp, ep - bp);
>+    if (frac_part != 0) {
>+      out->push_back('.');
>+      bp = Format64(ep, prec, frac_part);
>+      while (ep[-1] == '0') --ep;
>+      out->append(bp, ep - bp);
>+    }
>+    out->append(unit.abbr);
>+  }
>+}
>+
>+}  // namespace
>+
>+// From Go's doc at http://golang.org/pkg/time/#Duration.String
>+//   [FormatDuration] returns a std::string representing the duration in the
>+//   form "72h3m0.5s".  Leading zero units are omitted.  As a special
>+//   case, durations less than one second format use a smaller unit
>+//   (milli-, micro-, or nanoseconds) to ensure that the leading digit
>+//   is non-zero.  The zero duration formats as 0, with no unit.
>+std::string FormatDuration(Duration d) {
>+  const Duration min_duration = Seconds(kint64min);
>+  if (d == min_duration) {
>+    // Avoid needing to negate kint64min by directly returning what the
>+    // following code should produce in that case.
>+    return "-2562047788015215h30m8s";
>+  }
>+  std::string s;
>+  if (d < ZeroDuration()) {
>+    s.append("-");
>+    d = -d;
>+  }
>+  if (d == InfiniteDuration()) {
>+    s.append("inf");
>+  } else if (d < Seconds(1)) {
>+    // Special case for durations with a magnitude < 1 second.  The duration
>+    // is printed as a fraction of a single unit, e.g., "1.2ms".
>+    if (d < Microseconds(1)) {
>+      AppendNumberUnit(&s, FDivDuration(d, Nanoseconds(1)), kDisplayNano);
>+    } else if (d < Milliseconds(1)) {
>+      AppendNumberUnit(&s, FDivDuration(d, Microseconds(1)), kDisplayMicro);
>+    } else {
>+      AppendNumberUnit(&s, FDivDuration(d, Milliseconds(1)), kDisplayMilli);
>+    }
>+  } else {
>+    AppendNumberUnit(&s, IDivDuration(d, Hours(1), &d), kDisplayHour);
>+    AppendNumberUnit(&s, IDivDuration(d, Minutes(1), &d), kDisplayMin);
>+    AppendNumberUnit(&s, FDivDuration(d, Seconds(1)), kDisplaySec);
>+  }
>+  if (s.empty() || s == "-") {
>+    s = "0";
>+  }
>+  return s;
>+}
>+
>+namespace {
>+
>+// A helper for ParseDuration() that parses a leading number from the given
>+// std::string and stores the result in *int_part/*frac_part/*frac_scale.  The
>+// given std::string pointer is modified to point to the first unconsumed char.
>+bool ConsumeDurationNumber(const char** dpp, int64_t* int_part,
>+                           int64_t* frac_part, int64_t* frac_scale) {
>+  *int_part = 0;
>+  *frac_part = 0;
>+  *frac_scale = 1;  // invariant: *frac_part < *frac_scale
>+  const char* start = *dpp;
>+  for (; std::isdigit(**dpp); *dpp += 1) {
>+    const int d = **dpp - '0';  // contiguous digits
>+    if (*int_part > kint64max / 10) return false;
>+    *int_part *= 10;
>+    if (*int_part > kint64max - d) return false;
>+    *int_part += d;
>+  }
>+  const bool int_part_empty = (*dpp == start);
>+  if (**dpp != '.') return !int_part_empty;
>+  for (*dpp += 1; std::isdigit(**dpp); *dpp += 1) {
>+    const int d = **dpp - '0';  // contiguous digits
>+    if (*frac_scale <= kint64max / 10) {
>+      *frac_part *= 10;
>+      *frac_part += d;
>+      *frac_scale *= 10;
>+    }
>+  }
>+  return !int_part_empty || *frac_scale != 1;
>+}
>+
>+// A helper for ParseDuration() that parses a leading unit designator (e.g.,
>+// ns, us, ms, s, m, h) from the given std::string and stores the resulting unit
>+// in "*unit".  The given std::string pointer is modified to point to the first
>+// unconsumed char.
>+bool ConsumeDurationUnit(const char** start, Duration* unit) {
>+  const char *s = *start;
>+  bool ok = true;
>+  if (strncmp(s, "ns", 2) == 0) {
>+    s += 2;
>+    *unit = Nanoseconds(1);
>+  } else if (strncmp(s, "us", 2) == 0) {
>+    s += 2;
>+    *unit = Microseconds(1);
>+  } else if (strncmp(s, "ms", 2) == 0) {
>+    s += 2;
>+    *unit = Milliseconds(1);
>+  } else if (strncmp(s, "s", 1) == 0) {
>+    s += 1;
>+    *unit = Seconds(1);
>+  } else if (strncmp(s, "m", 1) == 0) {
>+    s += 1;
>+    *unit = Minutes(1);
>+  } else if (strncmp(s, "h", 1) == 0) {
>+    s += 1;
>+    *unit = Hours(1);
>+  } else {
>+    ok = false;
>+  }
>+  *start = s;
>+  return ok;
>+}
>+
>+}  // namespace
>+
>+// From Go's doc at http://golang.org/pkg/time/#ParseDuration
>+//   [ParseDuration] parses a duration std::string.  A duration std::string is
>+//   a possibly signed sequence of decimal numbers, each with optional
>+//   fraction and a unit suffix, such as "300ms", "-1.5h" or "2h45m".
>+//   Valid time units are "ns", "us" "ms", "s", "m", "h".
>+bool ParseDuration(const std::string& dur_string, Duration* d) {
>+  const char* start = dur_string.c_str();
>+  int sign = 1;
>+
>+  if (*start == '-' || *start == '+') {
>+    sign = *start == '-' ? -1 : 1;
>+    ++start;
>+  }
>+
>+  // Can't parse a duration from an empty std::string.
>+  if (*start == '\0') {
>+    return false;
>+  }
>+
>+  // Special case for a std::string of "0".
>+  if (*start == '0' && *(start + 1) == '\0') {
>+    *d = ZeroDuration();
>+    return true;
>+  }
>+
>+  if (strcmp(start, "inf") == 0) {
>+    *d = sign * InfiniteDuration();
>+    return true;
>+  }
>+
>+  Duration dur;
>+  while (*start != '\0') {
>+    int64_t int_part;
>+    int64_t frac_part;
>+    int64_t frac_scale;
>+    Duration unit;
>+    if (!ConsumeDurationNumber(&start, &int_part, &frac_part, &frac_scale) ||
>+        !ConsumeDurationUnit(&start, &unit)) {
>+      return false;
>+    }
>+    if (int_part != 0) dur += sign * int_part * unit;
>+    if (frac_part != 0) dur += sign * frac_part * unit / frac_scale;
>+  }
>+  *d = dur;
>+  return true;
>+}
>+bool ParseFlag(const std::string& text, Duration* dst, std::string* ) {
>+  return ParseDuration(text, dst);
>+}
>+
>+std::string UnparseFlag(Duration d) { return FormatDuration(d); }
>+
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/duration_benchmark.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/duration_benchmark.cc
>new file mode 100644
>index 00000000000..d5657bd576a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/duration_benchmark.cc
>@@ -0,0 +1,427 @@
>+// Copyright 2018 The Abseil Authors.
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include <cmath>
>+#include <cstddef>
>+#include <cstdint>
>+#include <ctime>
>+#include <string>
>+
>+#include "absl/time/time.h"
>+#include "benchmark/benchmark.h"
>+
>+namespace {
>+
>+//
>+// Factory functions
>+//
>+
>+void BM_Duration_Factory_Nanoseconds(benchmark::State& state) {
>+  int64_t i = 0;
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(absl::Nanoseconds(i));
>+    i += 314159;
>+  }
>+}
>+BENCHMARK(BM_Duration_Factory_Nanoseconds);
>+
>+void BM_Duration_Factory_Microseconds(benchmark::State& state) {
>+  int64_t i = 0;
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(absl::Microseconds(i));
>+    i += 314;
>+  }
>+}
>+BENCHMARK(BM_Duration_Factory_Microseconds);
>+
>+void BM_Duration_Factory_Milliseconds(benchmark::State& state) {
>+  int64_t i = 0;
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(absl::Milliseconds(i));
>+    i += 1;
>+  }
>+}
>+BENCHMARK(BM_Duration_Factory_Milliseconds);
>+
>+void BM_Duration_Factory_Seconds(benchmark::State& state) {
>+  int64_t i = 0;
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(absl::Seconds(i));
>+    i += 1;
>+  }
>+}
>+BENCHMARK(BM_Duration_Factory_Seconds);
>+
>+void BM_Duration_Factory_Minutes(benchmark::State& state) {
>+  int64_t i = 0;
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(absl::Minutes(i));
>+    i += 1;
>+  }
>+}
>+BENCHMARK(BM_Duration_Factory_Minutes);
>+
>+void BM_Duration_Factory_Hours(benchmark::State& state) {
>+  int64_t i = 0;
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(absl::Hours(i));
>+    i += 1;
>+  }
>+}
>+BENCHMARK(BM_Duration_Factory_Hours);
>+
>+void BM_Duration_Factory_DoubleNanoseconds(benchmark::State& state) {
>+  double d = 1;
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(absl::Nanoseconds(d));
>+    d = d * 1.00000001 + 1;
>+  }
>+}
>+BENCHMARK(BM_Duration_Factory_DoubleNanoseconds);
>+
>+void BM_Duration_Factory_DoubleMicroseconds(benchmark::State& state) {
>+  double d = 1e-3;
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(absl::Microseconds(d));
>+    d = d * 1.00000001 + 1e-3;
>+  }
>+}
>+BENCHMARK(BM_Duration_Factory_DoubleMicroseconds);
>+
>+void BM_Duration_Factory_DoubleMilliseconds(benchmark::State& state) {
>+  double d = 1e-6;
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(absl::Milliseconds(d));
>+    d = d * 1.00000001 + 1e-6;
>+  }
>+}
>+BENCHMARK(BM_Duration_Factory_DoubleMilliseconds);
>+
>+void BM_Duration_Factory_DoubleSeconds(benchmark::State& state) {
>+  double d = 1e-9;
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(absl::Seconds(d));
>+    d = d * 1.00000001 + 1e-9;
>+  }
>+}
>+BENCHMARK(BM_Duration_Factory_DoubleSeconds);
>+
>+void BM_Duration_Factory_DoubleMinutes(benchmark::State& state) {
>+  double d = 1e-9;
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(absl::Minutes(d));
>+    d = d * 1.00000001 + 1e-9;
>+  }
>+}
>+BENCHMARK(BM_Duration_Factory_DoubleMinutes);
>+
>+void BM_Duration_Factory_DoubleHours(benchmark::State& state) {
>+  double d = 1e-9;
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(absl::Hours(d));
>+    d = d * 1.00000001 + 1e-9;
>+  }
>+}
>+BENCHMARK(BM_Duration_Factory_DoubleHours);
>+
>+//
>+// Arithmetic
>+//
>+
>+void BM_Duration_Addition(benchmark::State& state) {
>+  absl::Duration d = absl::Nanoseconds(1);
>+  absl::Duration step = absl::Milliseconds(1);
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(d += step);
>+  }
>+}
>+BENCHMARK(BM_Duration_Addition);
>+
>+void BM_Duration_Subtraction(benchmark::State& state) {
>+  absl::Duration d = absl::Seconds(std::numeric_limits<int64_t>::max());
>+  absl::Duration step = absl::Milliseconds(1);
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(d -= step);
>+  }
>+}
>+BENCHMARK(BM_Duration_Subtraction);
>+
>+void BM_Duration_Multiplication_Fixed(benchmark::State& state) {
>+  absl::Duration d = absl::Milliseconds(1);
>+  absl::Duration s;
>+  int i = 0;
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(s += d * (i + 1));
>+    ++i;
>+  }
>+}
>+BENCHMARK(BM_Duration_Multiplication_Fixed);
>+
>+void BM_Duration_Multiplication_Double(benchmark::State& state) {
>+  absl::Duration d = absl::Milliseconds(1);
>+  absl::Duration s;
>+  int i = 0;
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(s += d * (i + 1.0));
>+    ++i;
>+  }
>+}
>+BENCHMARK(BM_Duration_Multiplication_Double);
>+
>+void BM_Duration_Division_Fixed(benchmark::State& state) {
>+  absl::Duration d = absl::Seconds(1);
>+  int i = 0;
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(d /= i + 1);
>+    ++i;
>+  }
>+}
>+BENCHMARK(BM_Duration_Division_Fixed);
>+
>+void BM_Duration_Division_Double(benchmark::State& state) {
>+  absl::Duration d = absl::Seconds(1);
>+  int i = 0;
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(d /= i + 1.0);
>+    ++i;
>+  }
>+}
>+BENCHMARK(BM_Duration_Division_Double);
>+
>+void BM_Duration_FDivDuration_Nanoseconds(benchmark::State& state) {
>+  double d = 1;
>+  int i = 0;
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(
>+        d += absl::FDivDuration(absl::Milliseconds(i), absl::Nanoseconds(1)));
>+    ++i;
>+  }
>+}
>+BENCHMARK(BM_Duration_FDivDuration_Nanoseconds);
>+
>+void BM_Duration_IDivDuration_Nanoseconds(benchmark::State& state) {
>+  int64_t a = 1;
>+  absl::Duration ignore;
>+  int i = 0;
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(a +=
>+                             absl::IDivDuration(absl::Nanoseconds(i),
>+                                                absl::Nanoseconds(1), &ignore));
>+    ++i;
>+  }
>+}
>+BENCHMARK(BM_Duration_IDivDuration_Nanoseconds);
>+
>+void BM_Duration_IDivDuration_Microseconds(benchmark::State& state) {
>+  int64_t a = 1;
>+  absl::Duration ignore;
>+  int i = 0;
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(a += absl::IDivDuration(absl::Microseconds(i),
>+                                                     absl::Microseconds(1),
>+                                                     &ignore));
>+    ++i;
>+  }
>+}
>+BENCHMARK(BM_Duration_IDivDuration_Microseconds);
>+
>+void BM_Duration_IDivDuration_Milliseconds(benchmark::State& state) {
>+  int64_t a = 1;
>+  absl::Duration ignore;
>+  int i = 0;
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(a += absl::IDivDuration(absl::Milliseconds(i),
>+                                                     absl::Milliseconds(1),
>+                                                     &ignore));
>+    ++i;
>+  }
>+}
>+BENCHMARK(BM_Duration_IDivDuration_Milliseconds);
>+
>+void BM_Duration_IDivDuration_Seconds(benchmark::State& state) {
>+  int64_t a = 1;
>+  absl::Duration ignore;
>+  int i = 0;
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(
>+        a += absl::IDivDuration(absl::Seconds(i), absl::Seconds(1), &ignore));
>+    ++i;
>+  }
>+}
>+BENCHMARK(BM_Duration_IDivDuration_Seconds);
>+
>+void BM_Duration_IDivDuration_Minutes(benchmark::State& state) {
>+  int64_t a = 1;
>+  absl::Duration ignore;
>+  int i = 0;
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(
>+        a += absl::IDivDuration(absl::Minutes(i), absl::Minutes(1), &ignore));
>+    ++i;
>+  }
>+}
>+BENCHMARK(BM_Duration_IDivDuration_Minutes);
>+
>+void BM_Duration_IDivDuration_Hours(benchmark::State& state) {
>+  int64_t a = 1;
>+  absl::Duration ignore;
>+  int i = 0;
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(
>+        a += absl::IDivDuration(absl::Hours(i), absl::Hours(1), &ignore));
>+    ++i;
>+  }
>+}
>+BENCHMARK(BM_Duration_IDivDuration_Hours);
>+
>+void BM_Duration_ToInt64Nanoseconds(benchmark::State& state) {
>+  absl::Duration d = absl::Seconds(100000);
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(absl::ToInt64Nanoseconds(d));
>+  }
>+}
>+BENCHMARK(BM_Duration_ToInt64Nanoseconds);
>+
>+void BM_Duration_ToInt64Microseconds(benchmark::State& state) {
>+  absl::Duration d = absl::Seconds(100000);
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(absl::ToInt64Microseconds(d));
>+  }
>+}
>+BENCHMARK(BM_Duration_ToInt64Microseconds);
>+
>+void BM_Duration_ToInt64Milliseconds(benchmark::State& state) {
>+  absl::Duration d = absl::Seconds(100000);
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(absl::ToInt64Milliseconds(d));
>+  }
>+}
>+BENCHMARK(BM_Duration_ToInt64Milliseconds);
>+
>+void BM_Duration_ToInt64Seconds(benchmark::State& state) {
>+  absl::Duration d = absl::Seconds(100000);
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(absl::ToInt64Seconds(d));
>+  }
>+}
>+BENCHMARK(BM_Duration_ToInt64Seconds);
>+
>+void BM_Duration_ToInt64Minutes(benchmark::State& state) {
>+  absl::Duration d = absl::Seconds(100000);
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(absl::ToInt64Minutes(d));
>+  }
>+}
>+BENCHMARK(BM_Duration_ToInt64Minutes);
>+
>+void BM_Duration_ToInt64Hours(benchmark::State& state) {
>+  absl::Duration d = absl::Seconds(100000);
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(absl::ToInt64Hours(d));
>+  }
>+}
>+BENCHMARK(BM_Duration_ToInt64Hours);
>+
>+//
>+// To/FromTimespec
>+//
>+
>+void BM_Duration_ToTimespec_AbslTime(benchmark::State& state) {
>+  absl::Duration d = absl::Seconds(1);
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(absl::ToTimespec(d));
>+  }
>+}
>+BENCHMARK(BM_Duration_ToTimespec_AbslTime);
>+
>+ABSL_ATTRIBUTE_NOINLINE timespec DoubleToTimespec(double seconds) {
>+  timespec ts;
>+  ts.tv_sec = seconds;
>+  ts.tv_nsec = (seconds - ts.tv_sec) * (1000 * 1000 * 1000);
>+  return ts;
>+}
>+
>+void BM_Duration_ToTimespec_Double(benchmark::State& state) {
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(DoubleToTimespec(1.0));
>+  }
>+}
>+BENCHMARK(BM_Duration_ToTimespec_Double);
>+
>+void BM_Duration_FromTimespec_AbslTime(benchmark::State& state) {
>+  timespec ts;
>+  ts.tv_sec = 0;
>+  ts.tv_nsec = 0;
>+  while (state.KeepRunning()) {
>+    if (++ts.tv_nsec == 1000 * 1000 * 1000) {
>+      ++ts.tv_sec;
>+      ts.tv_nsec = 0;
>+    }
>+    benchmark::DoNotOptimize(absl::DurationFromTimespec(ts));
>+  }
>+}
>+BENCHMARK(BM_Duration_FromTimespec_AbslTime);
>+
>+ABSL_ATTRIBUTE_NOINLINE double TimespecToDouble(timespec ts) {
>+  return ts.tv_sec + (ts.tv_nsec / (1000 * 1000 * 1000));
>+}
>+
>+void BM_Duration_FromTimespec_Double(benchmark::State& state) {
>+  timespec ts;
>+  ts.tv_sec = 0;
>+  ts.tv_nsec = 0;
>+  while (state.KeepRunning()) {
>+    if (++ts.tv_nsec == 1000 * 1000 * 1000) {
>+      ++ts.tv_sec;
>+      ts.tv_nsec = 0;
>+    }
>+    benchmark::DoNotOptimize(TimespecToDouble(ts));
>+  }
>+}
>+BENCHMARK(BM_Duration_FromTimespec_Double);
>+
>+//
>+// String conversions
>+//
>+
>+const char* const kDurations[] = {
>+    "0",                                   // 0
>+    "123ns",                               // 1
>+    "1h2m3s",                              // 2
>+    "-2h3m4.005006007s",                   // 3
>+    "2562047788015215h30m7.99999999975s",  // 4
>+};
>+const int kNumDurations = sizeof(kDurations) / sizeof(kDurations[0]);
>+
>+void BM_Duration_FormatDuration(benchmark::State& state) {
>+  const std::string s = kDurations[state.range(0)];
>+  state.SetLabel(s);
>+  absl::Duration d;
>+  absl::ParseDuration(kDurations[state.range(0)], &d);
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(absl::FormatDuration(d));
>+  }
>+}
>+BENCHMARK(BM_Duration_FormatDuration)->DenseRange(0, kNumDurations - 1);
>+
>+void BM_Duration_ParseDuration(benchmark::State& state) {
>+  const std::string s = kDurations[state.range(0)];
>+  state.SetLabel(s);
>+  absl::Duration d;
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(absl::ParseDuration(s, &d));
>+  }
>+}
>+BENCHMARK(BM_Duration_ParseDuration)->DenseRange(0, kNumDurations - 1);
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/duration_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/duration_test.cc
>new file mode 100644
>index 00000000000..7ae25dc68f9
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/duration_test.cc
>@@ -0,0 +1,1764 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include <chrono>  // NOLINT(build/c++11)
>+#include <cmath>
>+#include <cstdint>
>+#include <ctime>
>+#include <iomanip>
>+#include <limits>
>+#include <random>
>+#include <string>
>+
>+#include "gmock/gmock.h"
>+#include "gtest/gtest.h"
>+#include "absl/time/time.h"
>+
>+namespace {
>+
>+constexpr int64_t kint64max = std::numeric_limits<int64_t>::max();
>+constexpr int64_t kint64min = std::numeric_limits<int64_t>::min();
>+
>+// Approximates the given number of years. This is only used to make some test
>+// code more readable.
>+absl::Duration ApproxYears(int64_t n) { return absl::Hours(n) * 365 * 24; }
>+
>+// A gMock matcher to match timespec values. Use this matcher like:
>+// timespec ts1, ts2;
>+// EXPECT_THAT(ts1, TimespecMatcher(ts2));
>+MATCHER_P(TimespecMatcher, ts, "") {
>+  if (ts.tv_sec == arg.tv_sec && ts.tv_nsec == arg.tv_nsec)
>+    return true;
>+  *result_listener << "expected: {" << ts.tv_sec << ", " << ts.tv_nsec << "} ";
>+  *result_listener << "actual: {" << arg.tv_sec << ", " << arg.tv_nsec << "}";
>+  return false;
>+}
>+
>+// A gMock matcher to match timeval values. Use this matcher like:
>+// timeval tv1, tv2;
>+// EXPECT_THAT(tv1, TimevalMatcher(tv2));
>+MATCHER_P(TimevalMatcher, tv, "") {
>+  if (tv.tv_sec == arg.tv_sec && tv.tv_usec == arg.tv_usec)
>+    return true;
>+  *result_listener << "expected: {" << tv.tv_sec << ", " << tv.tv_usec << "} ";
>+  *result_listener << "actual: {" << arg.tv_sec << ", " << arg.tv_usec << "}";
>+  return false;
>+}
>+
>+TEST(Duration, ValueSemantics) {
>+  // If this compiles, the test passes.
>+  constexpr absl::Duration a;      // Default construction
>+  constexpr absl::Duration b = a;  // Copy construction
>+  constexpr absl::Duration c(b);   // Copy construction (again)
>+
>+  absl::Duration d;
>+  d = c;  // Assignment
>+}
>+
>+TEST(Duration, Factories) {
>+  constexpr absl::Duration zero = absl::ZeroDuration();
>+  constexpr absl::Duration nano = absl::Nanoseconds(1);
>+  constexpr absl::Duration micro = absl::Microseconds(1);
>+  constexpr absl::Duration milli = absl::Milliseconds(1);
>+  constexpr absl::Duration sec = absl::Seconds(1);
>+  constexpr absl::Duration min = absl::Minutes(1);
>+  constexpr absl::Duration hour = absl::Hours(1);
>+
>+  EXPECT_EQ(zero, absl::Duration());
>+  EXPECT_EQ(zero, absl::Seconds(0));
>+  EXPECT_EQ(nano, absl::Nanoseconds(1));
>+  EXPECT_EQ(micro, absl::Nanoseconds(1000));
>+  EXPECT_EQ(milli, absl::Microseconds(1000));
>+  EXPECT_EQ(sec, absl::Milliseconds(1000));
>+  EXPECT_EQ(min, absl::Seconds(60));
>+  EXPECT_EQ(hour, absl::Minutes(60));
>+
>+  // Tests factory limits
>+  const absl::Duration inf = absl::InfiniteDuration();
>+
>+  EXPECT_GT(inf, absl::Seconds(kint64max));
>+  EXPECT_LT(-inf, absl::Seconds(kint64min));
>+  EXPECT_LT(-inf, absl::Seconds(-kint64max));
>+
>+  EXPECT_EQ(inf, absl::Minutes(kint64max));
>+  EXPECT_EQ(-inf, absl::Minutes(kint64min));
>+  EXPECT_EQ(-inf, absl::Minutes(-kint64max));
>+  EXPECT_GT(inf, absl::Minutes(kint64max / 60));
>+  EXPECT_LT(-inf, absl::Minutes(kint64min / 60));
>+  EXPECT_LT(-inf, absl::Minutes(-kint64max / 60));
>+
>+  EXPECT_EQ(inf, absl::Hours(kint64max));
>+  EXPECT_EQ(-inf, absl::Hours(kint64min));
>+  EXPECT_EQ(-inf, absl::Hours(-kint64max));
>+  EXPECT_GT(inf, absl::Hours(kint64max / 3600));
>+  EXPECT_LT(-inf, absl::Hours(kint64min / 3600));
>+  EXPECT_LT(-inf, absl::Hours(-kint64max / 3600));
>+}
>+
>+TEST(Duration, ToConversion) {
>+#define TEST_DURATION_CONVERSION(UNIT)                                  \
>+  do {                                                                  \
>+    const absl::Duration d = absl::UNIT(1.5);                           \
>+    constexpr absl::Duration z = absl::ZeroDuration();                  \
>+    constexpr absl::Duration inf = absl::InfiniteDuration();            \
>+    constexpr double dbl_inf = std::numeric_limits<double>::infinity(); \
>+    EXPECT_EQ(kint64min, absl::ToInt64##UNIT(-inf));                    \
>+    EXPECT_EQ(-1, absl::ToInt64##UNIT(-d));                             \
>+    EXPECT_EQ(0, absl::ToInt64##UNIT(z));                               \
>+    EXPECT_EQ(1, absl::ToInt64##UNIT(d));                               \
>+    EXPECT_EQ(kint64max, absl::ToInt64##UNIT(inf));                     \
>+    EXPECT_EQ(-dbl_inf, absl::ToDouble##UNIT(-inf));                    \
>+    EXPECT_EQ(-1.5, absl::ToDouble##UNIT(-d));                          \
>+    EXPECT_EQ(0, absl::ToDouble##UNIT(z));                              \
>+    EXPECT_EQ(1.5, absl::ToDouble##UNIT(d));                            \
>+    EXPECT_EQ(dbl_inf, absl::ToDouble##UNIT(inf));                      \
>+  } while (0)
>+
>+  TEST_DURATION_CONVERSION(Nanoseconds);
>+  TEST_DURATION_CONVERSION(Microseconds);
>+  TEST_DURATION_CONVERSION(Milliseconds);
>+  TEST_DURATION_CONVERSION(Seconds);
>+  TEST_DURATION_CONVERSION(Minutes);
>+  TEST_DURATION_CONVERSION(Hours);
>+
>+#undef TEST_DURATION_CONVERSION
>+}
>+
>+template <int64_t N>
>+void TestToConversion() {
>+  constexpr absl::Duration nano = absl::Nanoseconds(N);
>+  EXPECT_EQ(N, absl::ToInt64Nanoseconds(nano));
>+  EXPECT_EQ(0, absl::ToInt64Microseconds(nano));
>+  EXPECT_EQ(0, absl::ToInt64Milliseconds(nano));
>+  EXPECT_EQ(0, absl::ToInt64Seconds(nano));
>+  EXPECT_EQ(0, absl::ToInt64Minutes(nano));
>+  EXPECT_EQ(0, absl::ToInt64Hours(nano));
>+  const absl::Duration micro = absl::Microseconds(N);
>+  EXPECT_EQ(N * 1000, absl::ToInt64Nanoseconds(micro));
>+  EXPECT_EQ(N, absl::ToInt64Microseconds(micro));
>+  EXPECT_EQ(0, absl::ToInt64Milliseconds(micro));
>+  EXPECT_EQ(0, absl::ToInt64Seconds(micro));
>+  EXPECT_EQ(0, absl::ToInt64Minutes(micro));
>+  EXPECT_EQ(0, absl::ToInt64Hours(micro));
>+  const absl::Duration milli = absl::Milliseconds(N);
>+  EXPECT_EQ(N * 1000 * 1000, absl::ToInt64Nanoseconds(milli));
>+  EXPECT_EQ(N * 1000, absl::ToInt64Microseconds(milli));
>+  EXPECT_EQ(N, absl::ToInt64Milliseconds(milli));
>+  EXPECT_EQ(0, absl::ToInt64Seconds(milli));
>+  EXPECT_EQ(0, absl::ToInt64Minutes(milli));
>+  EXPECT_EQ(0, absl::ToInt64Hours(milli));
>+  const absl::Duration sec = absl::Seconds(N);
>+  EXPECT_EQ(N * 1000 * 1000 * 1000, absl::ToInt64Nanoseconds(sec));
>+  EXPECT_EQ(N * 1000 * 1000, absl::ToInt64Microseconds(sec));
>+  EXPECT_EQ(N * 1000, absl::ToInt64Milliseconds(sec));
>+  EXPECT_EQ(N, absl::ToInt64Seconds(sec));
>+  EXPECT_EQ(0, absl::ToInt64Minutes(sec));
>+  EXPECT_EQ(0, absl::ToInt64Hours(sec));
>+  const absl::Duration min = absl::Minutes(N);
>+  EXPECT_EQ(N * 60 * 1000 * 1000 * 1000, absl::ToInt64Nanoseconds(min));
>+  EXPECT_EQ(N * 60 * 1000 * 1000, absl::ToInt64Microseconds(min));
>+  EXPECT_EQ(N * 60 * 1000, absl::ToInt64Milliseconds(min));
>+  EXPECT_EQ(N * 60, absl::ToInt64Seconds(min));
>+  EXPECT_EQ(N, absl::ToInt64Minutes(min));
>+  EXPECT_EQ(0, absl::ToInt64Hours(min));
>+  const absl::Duration hour = absl::Hours(N);
>+  EXPECT_EQ(N * 60 * 60 * 1000 * 1000 * 1000, absl::ToInt64Nanoseconds(hour));
>+  EXPECT_EQ(N * 60 * 60 * 1000 * 1000, absl::ToInt64Microseconds(hour));
>+  EXPECT_EQ(N * 60 * 60 * 1000, absl::ToInt64Milliseconds(hour));
>+  EXPECT_EQ(N * 60 * 60, absl::ToInt64Seconds(hour));
>+  EXPECT_EQ(N * 60, absl::ToInt64Minutes(hour));
>+  EXPECT_EQ(N, absl::ToInt64Hours(hour));
>+}
>+
>+TEST(Duration, ToConversionDeprecated) {
>+  TestToConversion<43>();
>+  TestToConversion<1>();
>+  TestToConversion<0>();
>+  TestToConversion<-1>();
>+  TestToConversion<-43>();
>+}
>+
>+template <int64_t N>
>+void TestFromChronoBasicEquality() {
>+  using std::chrono::nanoseconds;
>+  using std::chrono::microseconds;
>+  using std::chrono::milliseconds;
>+  using std::chrono::seconds;
>+  using std::chrono::minutes;
>+  using std::chrono::hours;
>+
>+  static_assert(absl::Nanoseconds(N) == absl::FromChrono(nanoseconds(N)), "");
>+  static_assert(absl::Microseconds(N) == absl::FromChrono(microseconds(N)), "");
>+  static_assert(absl::Milliseconds(N) == absl::FromChrono(milliseconds(N)), "");
>+  static_assert(absl::Seconds(N) == absl::FromChrono(seconds(N)), "");
>+  static_assert(absl::Minutes(N) == absl::FromChrono(minutes(N)), "");
>+  static_assert(absl::Hours(N) == absl::FromChrono(hours(N)), "");
>+}
>+
>+TEST(Duration, FromChrono) {
>+  TestFromChronoBasicEquality<-123>();
>+  TestFromChronoBasicEquality<-1>();
>+  TestFromChronoBasicEquality<0>();
>+  TestFromChronoBasicEquality<1>();
>+  TestFromChronoBasicEquality<123>();
>+
>+  // Minutes (might, depending on the platform) saturate at +inf.
>+  const auto chrono_minutes_max = std::chrono::minutes::max();
>+  const auto minutes_max = absl::FromChrono(chrono_minutes_max);
>+  const int64_t minutes_max_count = chrono_minutes_max.count();
>+  if (minutes_max_count > kint64max / 60) {
>+    EXPECT_EQ(absl::InfiniteDuration(), minutes_max);
>+  } else {
>+    EXPECT_EQ(absl::Minutes(minutes_max_count), minutes_max);
>+  }
>+
>+  // Minutes (might, depending on the platform) saturate at -inf.
>+  const auto chrono_minutes_min = std::chrono::minutes::min();
>+  const auto minutes_min = absl::FromChrono(chrono_minutes_min);
>+  const int64_t minutes_min_count = chrono_minutes_min.count();
>+  if (minutes_min_count < kint64min / 60) {
>+    EXPECT_EQ(-absl::InfiniteDuration(), minutes_min);
>+  } else {
>+    EXPECT_EQ(absl::Minutes(minutes_min_count), minutes_min);
>+  }
>+
>+  // Hours (might, depending on the platform) saturate at +inf.
>+  const auto chrono_hours_max = std::chrono::hours::max();
>+  const auto hours_max = absl::FromChrono(chrono_hours_max);
>+  const int64_t hours_max_count = chrono_hours_max.count();
>+  if (hours_max_count > kint64max / 3600) {
>+    EXPECT_EQ(absl::InfiniteDuration(), hours_max);
>+  } else {
>+    EXPECT_EQ(absl::Hours(hours_max_count), hours_max);
>+  }
>+
>+  // Hours (might, depending on the platform) saturate at -inf.
>+  const auto chrono_hours_min = std::chrono::hours::min();
>+  const auto hours_min = absl::FromChrono(chrono_hours_min);
>+  const int64_t hours_min_count = chrono_hours_min.count();
>+  if (hours_min_count < kint64min / 3600) {
>+    EXPECT_EQ(-absl::InfiniteDuration(), hours_min);
>+  } else {
>+    EXPECT_EQ(absl::Hours(hours_min_count), hours_min);
>+  }
>+}
>+
>+template <int64_t N>
>+void TestToChrono() {
>+  using std::chrono::nanoseconds;
>+  using std::chrono::microseconds;
>+  using std::chrono::milliseconds;
>+  using std::chrono::seconds;
>+  using std::chrono::minutes;
>+  using std::chrono::hours;
>+
>+  EXPECT_EQ(nanoseconds(N), absl::ToChronoNanoseconds(absl::Nanoseconds(N)));
>+  EXPECT_EQ(microseconds(N), absl::ToChronoMicroseconds(absl::Microseconds(N)));
>+  EXPECT_EQ(milliseconds(N), absl::ToChronoMilliseconds(absl::Milliseconds(N)));
>+  EXPECT_EQ(seconds(N), absl::ToChronoSeconds(absl::Seconds(N)));
>+
>+  constexpr auto absl_minutes = absl::Minutes(N);
>+  auto chrono_minutes = minutes(N);
>+  if (absl_minutes == -absl::InfiniteDuration()) {
>+    chrono_minutes = minutes::min();
>+  } else if (absl_minutes == absl::InfiniteDuration()) {
>+    chrono_minutes = minutes::max();
>+  }
>+  EXPECT_EQ(chrono_minutes, absl::ToChronoMinutes(absl_minutes));
>+
>+  constexpr auto absl_hours = absl::Hours(N);
>+  auto chrono_hours = hours(N);
>+  if (absl_hours == -absl::InfiniteDuration()) {
>+    chrono_hours = hours::min();
>+  } else if (absl_hours == absl::InfiniteDuration()) {
>+    chrono_hours = hours::max();
>+  }
>+  EXPECT_EQ(chrono_hours, absl::ToChronoHours(absl_hours));
>+}
>+
>+TEST(Duration, ToChrono) {
>+  using std::chrono::nanoseconds;
>+  using std::chrono::microseconds;
>+  using std::chrono::milliseconds;
>+  using std::chrono::seconds;
>+  using std::chrono::minutes;
>+  using std::chrono::hours;
>+
>+  TestToChrono<kint64min>();
>+  TestToChrono<-1>();
>+  TestToChrono<0>();
>+  TestToChrono<1>();
>+  TestToChrono<kint64max>();
>+
>+  // Verify truncation toward zero.
>+  const auto tick = absl::Nanoseconds(1) / 4;
>+  EXPECT_EQ(nanoseconds(0), absl::ToChronoNanoseconds(tick));
>+  EXPECT_EQ(nanoseconds(0), absl::ToChronoNanoseconds(-tick));
>+  EXPECT_EQ(microseconds(0), absl::ToChronoMicroseconds(tick));
>+  EXPECT_EQ(microseconds(0), absl::ToChronoMicroseconds(-tick));
>+  EXPECT_EQ(milliseconds(0), absl::ToChronoMilliseconds(tick));
>+  EXPECT_EQ(milliseconds(0), absl::ToChronoMilliseconds(-tick));
>+  EXPECT_EQ(seconds(0), absl::ToChronoSeconds(tick));
>+  EXPECT_EQ(seconds(0), absl::ToChronoSeconds(-tick));
>+  EXPECT_EQ(minutes(0), absl::ToChronoMinutes(tick));
>+  EXPECT_EQ(minutes(0), absl::ToChronoMinutes(-tick));
>+  EXPECT_EQ(hours(0), absl::ToChronoHours(tick));
>+  EXPECT_EQ(hours(0), absl::ToChronoHours(-tick));
>+
>+  // Verifies +/- infinity saturation at max/min.
>+  constexpr auto inf = absl::InfiniteDuration();
>+  EXPECT_EQ(nanoseconds::min(), absl::ToChronoNanoseconds(-inf));
>+  EXPECT_EQ(nanoseconds::max(), absl::ToChronoNanoseconds(inf));
>+  EXPECT_EQ(microseconds::min(), absl::ToChronoMicroseconds(-inf));
>+  EXPECT_EQ(microseconds::max(), absl::ToChronoMicroseconds(inf));
>+  EXPECT_EQ(milliseconds::min(), absl::ToChronoMilliseconds(-inf));
>+  EXPECT_EQ(milliseconds::max(), absl::ToChronoMilliseconds(inf));
>+  EXPECT_EQ(seconds::min(), absl::ToChronoSeconds(-inf));
>+  EXPECT_EQ(seconds::max(), absl::ToChronoSeconds(inf));
>+  EXPECT_EQ(minutes::min(), absl::ToChronoMinutes(-inf));
>+  EXPECT_EQ(minutes::max(), absl::ToChronoMinutes(inf));
>+  EXPECT_EQ(hours::min(), absl::ToChronoHours(-inf));
>+  EXPECT_EQ(hours::max(), absl::ToChronoHours(inf));
>+}
>+
>+TEST(Duration, FactoryOverloads) {
>+  enum E { kOne = 1 };
>+#define TEST_FACTORY_OVERLOADS(NAME)                                          \
>+  EXPECT_EQ(1, NAME(kOne) / NAME(kOne));                                      \
>+  EXPECT_EQ(1, NAME(static_cast<int8_t>(1)) / NAME(1));                       \
>+  EXPECT_EQ(1, NAME(static_cast<int16_t>(1)) / NAME(1));                      \
>+  EXPECT_EQ(1, NAME(static_cast<int32_t>(1)) / NAME(1));                      \
>+  EXPECT_EQ(1, NAME(static_cast<int64_t>(1)) / NAME(1));                      \
>+  EXPECT_EQ(1, NAME(static_cast<uint8_t>(1)) / NAME(1));                      \
>+  EXPECT_EQ(1, NAME(static_cast<uint16_t>(1)) / NAME(1));                     \
>+  EXPECT_EQ(1, NAME(static_cast<uint32_t>(1)) / NAME(1));                     \
>+  EXPECT_EQ(1, NAME(static_cast<uint64_t>(1)) / NAME(1));                     \
>+  EXPECT_EQ(NAME(1) / 2, NAME(static_cast<float>(0.5)));                      \
>+  EXPECT_EQ(NAME(1) / 2, NAME(static_cast<double>(0.5)));                     \
>+  EXPECT_EQ(1.5, absl::FDivDuration(NAME(static_cast<float>(1.5)), NAME(1))); \
>+  EXPECT_EQ(1.5, absl::FDivDuration(NAME(static_cast<double>(1.5)), NAME(1)));
>+
>+  TEST_FACTORY_OVERLOADS(absl::Nanoseconds);
>+  TEST_FACTORY_OVERLOADS(absl::Microseconds);
>+  TEST_FACTORY_OVERLOADS(absl::Milliseconds);
>+  TEST_FACTORY_OVERLOADS(absl::Seconds);
>+  TEST_FACTORY_OVERLOADS(absl::Minutes);
>+  TEST_FACTORY_OVERLOADS(absl::Hours);
>+
>+#undef TEST_FACTORY_OVERLOADS
>+
>+  EXPECT_EQ(absl::Milliseconds(1500), absl::Seconds(1.5));
>+  EXPECT_LT(absl::Nanoseconds(1), absl::Nanoseconds(1.5));
>+  EXPECT_GT(absl::Nanoseconds(2), absl::Nanoseconds(1.5));
>+
>+  const double dbl_inf = std::numeric_limits<double>::infinity();
>+  EXPECT_EQ(absl::InfiniteDuration(), absl::Nanoseconds(dbl_inf));
>+  EXPECT_EQ(absl::InfiniteDuration(), absl::Microseconds(dbl_inf));
>+  EXPECT_EQ(absl::InfiniteDuration(), absl::Milliseconds(dbl_inf));
>+  EXPECT_EQ(absl::InfiniteDuration(), absl::Seconds(dbl_inf));
>+  EXPECT_EQ(absl::InfiniteDuration(), absl::Minutes(dbl_inf));
>+  EXPECT_EQ(absl::InfiniteDuration(), absl::Hours(dbl_inf));
>+  EXPECT_EQ(-absl::InfiniteDuration(), absl::Nanoseconds(-dbl_inf));
>+  EXPECT_EQ(-absl::InfiniteDuration(), absl::Microseconds(-dbl_inf));
>+  EXPECT_EQ(-absl::InfiniteDuration(), absl::Milliseconds(-dbl_inf));
>+  EXPECT_EQ(-absl::InfiniteDuration(), absl::Seconds(-dbl_inf));
>+  EXPECT_EQ(-absl::InfiniteDuration(), absl::Minutes(-dbl_inf));
>+  EXPECT_EQ(-absl::InfiniteDuration(), absl::Hours(-dbl_inf));
>+}
>+
>+TEST(Duration, InfinityExamples) {
>+  // These examples are used in the documentation in time.h. They are
>+  // written so that they can be copy-n-pasted easily.
>+
>+  constexpr absl::Duration inf = absl::InfiniteDuration();
>+  constexpr absl::Duration d = absl::Seconds(1);  // Any finite duration
>+
>+  EXPECT_TRUE(inf == inf + inf);
>+  EXPECT_TRUE(inf == inf + d);
>+  EXPECT_TRUE(inf == inf - inf);
>+  EXPECT_TRUE(-inf == d - inf);
>+
>+  EXPECT_TRUE(inf == d * 1e100);
>+  EXPECT_TRUE(0 == d / inf);  // NOLINT(readability/check)
>+
>+  // Division by zero returns infinity, or kint64min/MAX where necessary.
>+  EXPECT_TRUE(inf == d / 0);
>+  EXPECT_TRUE(kint64max == d / absl::ZeroDuration());
>+}
>+
>+TEST(Duration, InfinityComparison) {
>+  const absl::Duration inf = absl::InfiniteDuration();
>+  const absl::Duration any_dur = absl::Seconds(1);
>+
>+  // Equality
>+  EXPECT_EQ(inf, inf);
>+  EXPECT_EQ(-inf, -inf);
>+  EXPECT_NE(inf, -inf);
>+  EXPECT_NE(any_dur, inf);
>+  EXPECT_NE(any_dur, -inf);
>+
>+  // Relational
>+  EXPECT_GT(inf, any_dur);
>+  EXPECT_LT(-inf, any_dur);
>+  EXPECT_LT(-inf, inf);
>+  EXPECT_GT(inf, -inf);
>+}
>+
>+TEST(Duration, InfinityAddition) {
>+  const absl::Duration sec_max = absl::Seconds(kint64max);
>+  const absl::Duration sec_min = absl::Seconds(kint64min);
>+  const absl::Duration any_dur = absl::Seconds(1);
>+  const absl::Duration inf = absl::InfiniteDuration();
>+
>+  // Addition
>+  EXPECT_EQ(inf, inf + inf);
>+  EXPECT_EQ(inf, inf + -inf);
>+  EXPECT_EQ(-inf, -inf + inf);
>+  EXPECT_EQ(-inf, -inf + -inf);
>+
>+  EXPECT_EQ(inf, inf + any_dur);
>+  EXPECT_EQ(inf, any_dur + inf);
>+  EXPECT_EQ(-inf, -inf + any_dur);
>+  EXPECT_EQ(-inf, any_dur + -inf);
>+
>+  // Interesting case
>+  absl::Duration almost_inf = sec_max + absl::Nanoseconds(999999999);
>+  EXPECT_GT(inf, almost_inf);
>+  almost_inf += -absl::Nanoseconds(999999999);
>+  EXPECT_GT(inf, almost_inf);
>+
>+  // Addition overflow/underflow
>+  EXPECT_EQ(inf, sec_max + absl::Seconds(1));
>+  EXPECT_EQ(inf, sec_max + sec_max);
>+  EXPECT_EQ(-inf, sec_min + -absl::Seconds(1));
>+  EXPECT_EQ(-inf, sec_min + -sec_max);
>+
>+  // For reference: IEEE 754 behavior
>+  const double dbl_inf = std::numeric_limits<double>::infinity();
>+  EXPECT_TRUE(std::isinf(dbl_inf + dbl_inf));
>+  EXPECT_TRUE(std::isnan(dbl_inf + -dbl_inf));  // We return inf
>+  EXPECT_TRUE(std::isnan(-dbl_inf + dbl_inf));  // We return inf
>+  EXPECT_TRUE(std::isinf(-dbl_inf + -dbl_inf));
>+}
>+
>+TEST(Duration, InfinitySubtraction) {
>+  const absl::Duration sec_max = absl::Seconds(kint64max);
>+  const absl::Duration sec_min = absl::Seconds(kint64min);
>+  const absl::Duration any_dur = absl::Seconds(1);
>+  const absl::Duration inf = absl::InfiniteDuration();
>+
>+  // Subtraction
>+  EXPECT_EQ(inf, inf - inf);
>+  EXPECT_EQ(inf, inf - -inf);
>+  EXPECT_EQ(-inf, -inf - inf);
>+  EXPECT_EQ(-inf, -inf - -inf);
>+
>+  EXPECT_EQ(inf, inf - any_dur);
>+  EXPECT_EQ(-inf, any_dur - inf);
>+  EXPECT_EQ(-inf, -inf - any_dur);
>+  EXPECT_EQ(inf, any_dur - -inf);
>+
>+  // Subtraction overflow/underflow
>+  EXPECT_EQ(inf, sec_max - -absl::Seconds(1));
>+  EXPECT_EQ(inf, sec_max - -sec_max);
>+  EXPECT_EQ(-inf, sec_min - absl::Seconds(1));
>+  EXPECT_EQ(-inf, sec_min - sec_max);
>+
>+  // Interesting case
>+  absl::Duration almost_neg_inf = sec_min;
>+  EXPECT_LT(-inf, almost_neg_inf);
>+  almost_neg_inf -= -absl::Nanoseconds(1);
>+  EXPECT_LT(-inf, almost_neg_inf);
>+
>+  // For reference: IEEE 754 behavior
>+  const double dbl_inf = std::numeric_limits<double>::infinity();
>+  EXPECT_TRUE(std::isnan(dbl_inf - dbl_inf));  // We return inf
>+  EXPECT_TRUE(std::isinf(dbl_inf - -dbl_inf));
>+  EXPECT_TRUE(std::isinf(-dbl_inf - dbl_inf));
>+  EXPECT_TRUE(std::isnan(-dbl_inf - -dbl_inf));  // We return inf
>+}
>+
>+TEST(Duration, InfinityMultiplication) {
>+  const absl::Duration sec_max = absl::Seconds(kint64max);
>+  const absl::Duration sec_min = absl::Seconds(kint64min);
>+  const absl::Duration inf = absl::InfiniteDuration();
>+
>+#define TEST_INF_MUL_WITH_TYPE(T)                                     \
>+  EXPECT_EQ(inf, inf * static_cast<T>(2));                            \
>+  EXPECT_EQ(-inf, inf * static_cast<T>(-2));                          \
>+  EXPECT_EQ(-inf, -inf * static_cast<T>(2));                          \
>+  EXPECT_EQ(inf, -inf * static_cast<T>(-2));                          \
>+  EXPECT_EQ(inf, inf * static_cast<T>(0));                            \
>+  EXPECT_EQ(-inf, -inf * static_cast<T>(0));                          \
>+  EXPECT_EQ(inf, sec_max * static_cast<T>(2));                        \
>+  EXPECT_EQ(inf, sec_min * static_cast<T>(-2));                       \
>+  EXPECT_EQ(inf, (sec_max / static_cast<T>(2)) * static_cast<T>(3));  \
>+  EXPECT_EQ(-inf, sec_max * static_cast<T>(-2));                      \
>+  EXPECT_EQ(-inf, sec_min * static_cast<T>(2));                       \
>+  EXPECT_EQ(-inf, (sec_min / static_cast<T>(2)) * static_cast<T>(3));
>+
>+  TEST_INF_MUL_WITH_TYPE(int64_t);  // NOLINT(readability/function)
>+  TEST_INF_MUL_WITH_TYPE(double);   // NOLINT(readability/function)
>+
>+#undef TEST_INF_MUL_WITH_TYPE
>+
>+  const double dbl_inf = std::numeric_limits<double>::infinity();
>+  EXPECT_EQ(inf, inf * dbl_inf);
>+  EXPECT_EQ(-inf, -inf * dbl_inf);
>+  EXPECT_EQ(-inf, inf * -dbl_inf);
>+  EXPECT_EQ(inf, -inf * -dbl_inf);
>+
>+  const absl::Duration any_dur = absl::Seconds(1);
>+  EXPECT_EQ(inf, any_dur * dbl_inf);
>+  EXPECT_EQ(-inf, -any_dur * dbl_inf);
>+  EXPECT_EQ(-inf, any_dur * -dbl_inf);
>+  EXPECT_EQ(inf, -any_dur * -dbl_inf);
>+
>+  // Fixed-point multiplication will produce a finite value, whereas floating
>+  // point fuzziness will overflow to inf.
>+  EXPECT_NE(absl::InfiniteDuration(), absl::Seconds(1) * kint64max);
>+  EXPECT_EQ(inf, absl::Seconds(1) * static_cast<double>(kint64max));
>+  EXPECT_NE(-absl::InfiniteDuration(), absl::Seconds(1) * kint64min);
>+  EXPECT_EQ(-inf, absl::Seconds(1) * static_cast<double>(kint64min));
>+
>+  // Note that sec_max * or / by 1.0 overflows to inf due to the 53-bit
>+  // limitations of double.
>+  EXPECT_NE(inf, sec_max);
>+  EXPECT_NE(inf, sec_max / 1);
>+  EXPECT_EQ(inf, sec_max / 1.0);
>+  EXPECT_NE(inf, sec_max * 1);
>+  EXPECT_EQ(inf, sec_max * 1.0);
>+}
>+
>+TEST(Duration, InfinityDivision) {
>+  const absl::Duration sec_max = absl::Seconds(kint64max);
>+  const absl::Duration sec_min = absl::Seconds(kint64min);
>+  const absl::Duration inf = absl::InfiniteDuration();
>+
>+  // Division of Duration by a double
>+#define TEST_INF_DIV_WITH_TYPE(T)            \
>+  EXPECT_EQ(inf, inf / static_cast<T>(2));   \
>+  EXPECT_EQ(-inf, inf / static_cast<T>(-2)); \
>+  EXPECT_EQ(-inf, -inf / static_cast<T>(2)); \
>+  EXPECT_EQ(inf, -inf / static_cast<T>(-2));
>+
>+  TEST_INF_DIV_WITH_TYPE(int64_t);  // NOLINT(readability/function)
>+  TEST_INF_DIV_WITH_TYPE(double);   // NOLINT(readability/function)
>+
>+#undef TEST_INF_DIV_WITH_TYPE
>+
>+  // Division of Duration by a double overflow/underflow
>+  EXPECT_EQ(inf, sec_max / 0.5);
>+  EXPECT_EQ(inf, sec_min / -0.5);
>+  EXPECT_EQ(inf, ((sec_max / 0.5) + absl::Seconds(1)) / 0.5);
>+  EXPECT_EQ(-inf, sec_max / -0.5);
>+  EXPECT_EQ(-inf, sec_min / 0.5);
>+  EXPECT_EQ(-inf, ((sec_min / 0.5) - absl::Seconds(1)) / 0.5);
>+
>+  const double dbl_inf = std::numeric_limits<double>::infinity();
>+  EXPECT_EQ(inf, inf / dbl_inf);
>+  EXPECT_EQ(-inf, inf / -dbl_inf);
>+  EXPECT_EQ(-inf, -inf / dbl_inf);
>+  EXPECT_EQ(inf, -inf / -dbl_inf);
>+
>+  const absl::Duration any_dur = absl::Seconds(1);
>+  EXPECT_EQ(absl::ZeroDuration(), any_dur / dbl_inf);
>+  EXPECT_EQ(absl::ZeroDuration(), any_dur / -dbl_inf);
>+  EXPECT_EQ(absl::ZeroDuration(), -any_dur / dbl_inf);
>+  EXPECT_EQ(absl::ZeroDuration(), -any_dur / -dbl_inf);
>+}
>+
>+TEST(Duration, InfinityModulus) {
>+  const absl::Duration sec_max = absl::Seconds(kint64max);
>+  const absl::Duration any_dur = absl::Seconds(1);
>+  const absl::Duration inf = absl::InfiniteDuration();
>+
>+  EXPECT_EQ(inf, inf % inf);
>+  EXPECT_EQ(inf, inf % -inf);
>+  EXPECT_EQ(-inf, -inf % -inf);
>+  EXPECT_EQ(-inf, -inf % inf);
>+
>+  EXPECT_EQ(any_dur, any_dur % inf);
>+  EXPECT_EQ(any_dur, any_dur % -inf);
>+  EXPECT_EQ(-any_dur, -any_dur % inf);
>+  EXPECT_EQ(-any_dur, -any_dur % -inf);
>+
>+  EXPECT_EQ(inf, inf % -any_dur);
>+  EXPECT_EQ(inf, inf % any_dur);
>+  EXPECT_EQ(-inf, -inf % -any_dur);
>+  EXPECT_EQ(-inf, -inf % any_dur);
>+
>+  // Remainder isn't affected by overflow.
>+  EXPECT_EQ(absl::ZeroDuration(), sec_max % absl::Seconds(1));
>+  EXPECT_EQ(absl::ZeroDuration(), sec_max % absl::Milliseconds(1));
>+  EXPECT_EQ(absl::ZeroDuration(), sec_max % absl::Microseconds(1));
>+  EXPECT_EQ(absl::ZeroDuration(), sec_max % absl::Nanoseconds(1));
>+  EXPECT_EQ(absl::ZeroDuration(), sec_max % absl::Nanoseconds(1) / 4);
>+}
>+
>+TEST(Duration, InfinityIDiv) {
>+  const absl::Duration sec_max = absl::Seconds(kint64max);
>+  const absl::Duration any_dur = absl::Seconds(1);
>+  const absl::Duration inf = absl::InfiniteDuration();
>+  const double dbl_inf = std::numeric_limits<double>::infinity();
>+
>+  // IDivDuration (int64_t return value + a remainer)
>+  absl::Duration rem = absl::ZeroDuration();
>+  EXPECT_EQ(kint64max, absl::IDivDuration(inf, inf, &rem));
>+  EXPECT_EQ(inf, rem);
>+
>+  rem = absl::ZeroDuration();
>+  EXPECT_EQ(kint64max, absl::IDivDuration(-inf, -inf, &rem));
>+  EXPECT_EQ(-inf, rem);
>+
>+  rem = absl::ZeroDuration();
>+  EXPECT_EQ(kint64max, absl::IDivDuration(inf, any_dur, &rem));
>+  EXPECT_EQ(inf, rem);
>+
>+  rem = absl::ZeroDuration();
>+  EXPECT_EQ(0, absl::IDivDuration(any_dur, inf, &rem));
>+  EXPECT_EQ(any_dur, rem);
>+
>+  rem = absl::ZeroDuration();
>+  EXPECT_EQ(kint64max, absl::IDivDuration(-inf, -any_dur, &rem));
>+  EXPECT_EQ(-inf, rem);
>+
>+  rem = absl::ZeroDuration();
>+  EXPECT_EQ(0, absl::IDivDuration(-any_dur, -inf, &rem));
>+  EXPECT_EQ(-any_dur, rem);
>+
>+  rem = absl::ZeroDuration();
>+  EXPECT_EQ(kint64min, absl::IDivDuration(-inf, inf, &rem));
>+  EXPECT_EQ(-inf, rem);
>+
>+  rem = absl::ZeroDuration();
>+  EXPECT_EQ(kint64min, absl::IDivDuration(inf, -inf, &rem));
>+  EXPECT_EQ(inf, rem);
>+
>+  rem = absl::ZeroDuration();
>+  EXPECT_EQ(kint64min, absl::IDivDuration(-inf, any_dur, &rem));
>+  EXPECT_EQ(-inf, rem);
>+
>+  rem = absl::ZeroDuration();
>+  EXPECT_EQ(0, absl::IDivDuration(-any_dur, inf, &rem));
>+  EXPECT_EQ(-any_dur, rem);
>+
>+  rem = absl::ZeroDuration();
>+  EXPECT_EQ(kint64min, absl::IDivDuration(inf, -any_dur, &rem));
>+  EXPECT_EQ(inf, rem);
>+
>+  rem = absl::ZeroDuration();
>+  EXPECT_EQ(0, absl::IDivDuration(any_dur, -inf, &rem));
>+  EXPECT_EQ(any_dur, rem);
>+
>+  // IDivDuration overflow/underflow
>+  rem = any_dur;
>+  EXPECT_EQ(kint64max,
>+            absl::IDivDuration(sec_max, absl::Nanoseconds(1) / 4, &rem));
>+  EXPECT_EQ(sec_max - absl::Nanoseconds(kint64max) / 4, rem);
>+
>+  rem = any_dur;
>+  EXPECT_EQ(kint64max,
>+            absl::IDivDuration(sec_max, absl::Milliseconds(1), &rem));
>+  EXPECT_EQ(sec_max - absl::Milliseconds(kint64max), rem);
>+
>+  rem = any_dur;
>+  EXPECT_EQ(kint64max,
>+            absl::IDivDuration(-sec_max, -absl::Milliseconds(1), &rem));
>+  EXPECT_EQ(-sec_max + absl::Milliseconds(kint64max), rem);
>+
>+  rem = any_dur;
>+  EXPECT_EQ(kint64min,
>+            absl::IDivDuration(-sec_max, absl::Milliseconds(1), &rem));
>+  EXPECT_EQ(-sec_max - absl::Milliseconds(kint64min), rem);
>+
>+  rem = any_dur;
>+  EXPECT_EQ(kint64min,
>+            absl::IDivDuration(sec_max, -absl::Milliseconds(1), &rem));
>+  EXPECT_EQ(sec_max + absl::Milliseconds(kint64min), rem);
>+
>+  //
>+  // operator/(Duration, Duration) is a wrapper for IDivDuration().
>+  //
>+
>+  // IEEE 754 says inf / inf should be nan, but int64_t doesn't have
>+  // nan so we'll return kint64max/kint64min instead.
>+  EXPECT_TRUE(std::isnan(dbl_inf / dbl_inf));
>+  EXPECT_EQ(kint64max, inf / inf);
>+  EXPECT_EQ(kint64max, -inf / -inf);
>+  EXPECT_EQ(kint64min, -inf / inf);
>+  EXPECT_EQ(kint64min, inf / -inf);
>+
>+  EXPECT_TRUE(std::isinf(dbl_inf / 2.0));
>+  EXPECT_EQ(kint64max, inf / any_dur);
>+  EXPECT_EQ(kint64max, -inf / -any_dur);
>+  EXPECT_EQ(kint64min, -inf / any_dur);
>+  EXPECT_EQ(kint64min, inf / -any_dur);
>+
>+  EXPECT_EQ(0.0, 2.0 / dbl_inf);
>+  EXPECT_EQ(0, any_dur / inf);
>+  EXPECT_EQ(0, any_dur / -inf);
>+  EXPECT_EQ(0, -any_dur / inf);
>+  EXPECT_EQ(0, -any_dur / -inf);
>+  EXPECT_EQ(0, absl::ZeroDuration() / inf);
>+
>+  // Division of Duration by a Duration overflow/underflow
>+  EXPECT_EQ(kint64max, sec_max / absl::Milliseconds(1));
>+  EXPECT_EQ(kint64max, -sec_max / -absl::Milliseconds(1));
>+  EXPECT_EQ(kint64min, -sec_max / absl::Milliseconds(1));
>+  EXPECT_EQ(kint64min, sec_max / -absl::Milliseconds(1));
>+}
>+
>+TEST(Duration, InfinityFDiv) {
>+  const absl::Duration any_dur = absl::Seconds(1);
>+  const absl::Duration inf = absl::InfiniteDuration();
>+  const double dbl_inf = std::numeric_limits<double>::infinity();
>+
>+  EXPECT_EQ(dbl_inf, absl::FDivDuration(inf, inf));
>+  EXPECT_EQ(dbl_inf, absl::FDivDuration(-inf, -inf));
>+  EXPECT_EQ(dbl_inf, absl::FDivDuration(inf, any_dur));
>+  EXPECT_EQ(0.0, absl::FDivDuration(any_dur, inf));
>+  EXPECT_EQ(dbl_inf, absl::FDivDuration(-inf, -any_dur));
>+  EXPECT_EQ(0.0, absl::FDivDuration(-any_dur, -inf));
>+
>+  EXPECT_EQ(-dbl_inf, absl::FDivDuration(-inf, inf));
>+  EXPECT_EQ(-dbl_inf, absl::FDivDuration(inf, -inf));
>+  EXPECT_EQ(-dbl_inf, absl::FDivDuration(-inf, any_dur));
>+  EXPECT_EQ(0.0, absl::FDivDuration(-any_dur, inf));
>+  EXPECT_EQ(-dbl_inf, absl::FDivDuration(inf, -any_dur));
>+  EXPECT_EQ(0.0, absl::FDivDuration(any_dur, -inf));
>+}
>+
>+TEST(Duration, DivisionByZero) {
>+  const absl::Duration zero = absl::ZeroDuration();
>+  const absl::Duration inf = absl::InfiniteDuration();
>+  const absl::Duration any_dur = absl::Seconds(1);
>+  const double dbl_inf = std::numeric_limits<double>::infinity();
>+  const double dbl_denorm = std::numeric_limits<double>::denorm_min();
>+
>+  // IEEE 754 behavior
>+  double z = 0.0, two = 2.0;
>+  EXPECT_TRUE(std::isinf(two / z));
>+  EXPECT_TRUE(std::isnan(z / z));  // We'll return inf
>+
>+  // Operator/(Duration, double)
>+  EXPECT_EQ(inf, zero / 0.0);
>+  EXPECT_EQ(-inf, zero / -0.0);
>+  EXPECT_EQ(inf, any_dur / 0.0);
>+  EXPECT_EQ(-inf, any_dur / -0.0);
>+  EXPECT_EQ(-inf, -any_dur / 0.0);
>+  EXPECT_EQ(inf, -any_dur / -0.0);
>+
>+  // Tests dividing by a number very close to, but not quite zero.
>+  EXPECT_EQ(zero, zero / dbl_denorm);
>+  EXPECT_EQ(zero, zero / -dbl_denorm);
>+  EXPECT_EQ(inf, any_dur / dbl_denorm);
>+  EXPECT_EQ(-inf, any_dur / -dbl_denorm);
>+  EXPECT_EQ(-inf, -any_dur / dbl_denorm);
>+  EXPECT_EQ(inf, -any_dur / -dbl_denorm);
>+
>+  // IDiv
>+  absl::Duration rem = zero;
>+  EXPECT_EQ(kint64max, absl::IDivDuration(zero, zero, &rem));
>+  EXPECT_EQ(inf, rem);
>+
>+  rem = zero;
>+  EXPECT_EQ(kint64max, absl::IDivDuration(any_dur, zero, &rem));
>+  EXPECT_EQ(inf, rem);
>+
>+  rem = zero;
>+  EXPECT_EQ(kint64min, absl::IDivDuration(-any_dur, zero, &rem));
>+  EXPECT_EQ(-inf, rem);
>+
>+  // Operator/(Duration, Duration)
>+  EXPECT_EQ(kint64max, zero / zero);
>+  EXPECT_EQ(kint64max, any_dur / zero);
>+  EXPECT_EQ(kint64min, -any_dur / zero);
>+
>+  // FDiv
>+  EXPECT_EQ(dbl_inf, absl::FDivDuration(zero, zero));
>+  EXPECT_EQ(dbl_inf, absl::FDivDuration(any_dur, zero));
>+  EXPECT_EQ(-dbl_inf, absl::FDivDuration(-any_dur, zero));
>+}
>+
>+TEST(Duration, Range) {
>+  const absl::Duration range = ApproxYears(100 * 1e9);
>+  const absl::Duration range_future = range;
>+  const absl::Duration range_past = -range;
>+
>+  EXPECT_LT(range_future, absl::InfiniteDuration());
>+  EXPECT_GT(range_past, -absl::InfiniteDuration());
>+
>+  const absl::Duration full_range = range_future - range_past;
>+  EXPECT_GT(full_range, absl::ZeroDuration());
>+  EXPECT_LT(full_range, absl::InfiniteDuration());
>+
>+  const absl::Duration neg_full_range = range_past - range_future;
>+  EXPECT_LT(neg_full_range, absl::ZeroDuration());
>+  EXPECT_GT(neg_full_range, -absl::InfiniteDuration());
>+
>+  EXPECT_LT(neg_full_range, full_range);
>+  EXPECT_EQ(neg_full_range, -full_range);
>+}
>+
>+TEST(Duration, RelationalOperators) {
>+#define TEST_REL_OPS(UNIT)               \
>+  static_assert(UNIT(2) == UNIT(2), ""); \
>+  static_assert(UNIT(1) != UNIT(2), ""); \
>+  static_assert(UNIT(1) < UNIT(2), "");  \
>+  static_assert(UNIT(3) > UNIT(2), "");  \
>+  static_assert(UNIT(1) <= UNIT(2), ""); \
>+  static_assert(UNIT(2) <= UNIT(2), ""); \
>+  static_assert(UNIT(3) >= UNIT(2), ""); \
>+  static_assert(UNIT(2) >= UNIT(2), "");
>+
>+  TEST_REL_OPS(absl::Nanoseconds);
>+  TEST_REL_OPS(absl::Microseconds);
>+  TEST_REL_OPS(absl::Milliseconds);
>+  TEST_REL_OPS(absl::Seconds);
>+  TEST_REL_OPS(absl::Minutes);
>+  TEST_REL_OPS(absl::Hours);
>+
>+#undef TEST_REL_OPS
>+}
>+
>+TEST(Duration, Addition) {
>+#define TEST_ADD_OPS(UNIT)                  \
>+  do {                                      \
>+    EXPECT_EQ(UNIT(2), UNIT(1) + UNIT(1));  \
>+    EXPECT_EQ(UNIT(1), UNIT(2) - UNIT(1));  \
>+    EXPECT_EQ(UNIT(0), UNIT(2) - UNIT(2));  \
>+    EXPECT_EQ(UNIT(-1), UNIT(1) - UNIT(2)); \
>+    EXPECT_EQ(UNIT(-2), UNIT(0) - UNIT(2)); \
>+    EXPECT_EQ(UNIT(-2), UNIT(1) - UNIT(3)); \
>+    absl::Duration a = UNIT(1);             \
>+    a += UNIT(1);                           \
>+    EXPECT_EQ(UNIT(2), a);                  \
>+    a -= UNIT(1);                           \
>+    EXPECT_EQ(UNIT(1), a);                  \
>+  } while (0)
>+
>+  TEST_ADD_OPS(absl::Nanoseconds);
>+  TEST_ADD_OPS(absl::Microseconds);
>+  TEST_ADD_OPS(absl::Milliseconds);
>+  TEST_ADD_OPS(absl::Seconds);
>+  TEST_ADD_OPS(absl::Minutes);
>+  TEST_ADD_OPS(absl::Hours);
>+
>+#undef TEST_ADD_OPS
>+
>+  EXPECT_EQ(absl::Seconds(2), absl::Seconds(3) - 2 * absl::Milliseconds(500));
>+  EXPECT_EQ(absl::Seconds(2) + absl::Milliseconds(500),
>+            absl::Seconds(3) - absl::Milliseconds(500));
>+
>+  EXPECT_EQ(absl::Seconds(1) + absl::Milliseconds(998),
>+            absl::Milliseconds(999) + absl::Milliseconds(999));
>+
>+  EXPECT_EQ(absl::Milliseconds(-1),
>+            absl::Milliseconds(998) - absl::Milliseconds(999));
>+
>+  // Tests fractions of a nanoseconds. These are implementation details only.
>+  EXPECT_GT(absl::Nanoseconds(1), absl::Nanoseconds(1) / 2);
>+  EXPECT_EQ(absl::Nanoseconds(1),
>+            absl::Nanoseconds(1) / 2 + absl::Nanoseconds(1) / 2);
>+  EXPECT_GT(absl::Nanoseconds(1) / 4, absl::Nanoseconds(0));
>+  EXPECT_EQ(absl::Nanoseconds(1) / 8, absl::Nanoseconds(0));
>+
>+  // Tests subtraction that will cause wrap around of the rep_lo_ bits.
>+  absl::Duration d_7_5 = absl::Seconds(7) + absl::Milliseconds(500);
>+  absl::Duration d_3_7 = absl::Seconds(3) + absl::Milliseconds(700);
>+  absl::Duration ans_3_8 = absl::Seconds(3) + absl::Milliseconds(800);
>+  EXPECT_EQ(ans_3_8, d_7_5 - d_3_7);
>+
>+  // Subtracting min_duration
>+  absl::Duration min_dur = absl::Seconds(kint64min);
>+  EXPECT_EQ(absl::Seconds(0), min_dur - min_dur);
>+  EXPECT_EQ(absl::Seconds(kint64max), absl::Seconds(-1) - min_dur);
>+}
>+
>+TEST(Duration, Negation) {
>+  // By storing negations of various values in constexpr variables we
>+  // verify that the initializers are constant expressions.
>+  constexpr absl::Duration negated_zero_duration = -absl::ZeroDuration();
>+  EXPECT_EQ(negated_zero_duration, absl::ZeroDuration());
>+
>+  constexpr absl::Duration negated_infinite_duration =
>+      -absl::InfiniteDuration();
>+  EXPECT_NE(negated_infinite_duration, absl::InfiniteDuration());
>+  EXPECT_EQ(-negated_infinite_duration, absl::InfiniteDuration());
>+
>+  // The public APIs to check if a duration is infinite depend on using
>+  // -InfiniteDuration(), but we're trying to test operator- here, so we
>+  // need to use the lower-level internal query IsInfiniteDuration.
>+  EXPECT_TRUE(
>+      absl::time_internal::IsInfiniteDuration(negated_infinite_duration));
>+
>+  // The largest Duration is kint64max seconds and kTicksPerSecond - 1 ticks.
>+  // Using the absl::time_internal::MakeDuration API is the cleanest way to
>+  // construct that Duration.
>+  constexpr absl::Duration max_duration = absl::time_internal::MakeDuration(
>+      kint64max, absl::time_internal::kTicksPerSecond - 1);
>+  constexpr absl::Duration negated_max_duration = -max_duration;
>+  // The largest negatable value is one tick above the minimum representable;
>+  // it's the negation of max_duration.
>+  constexpr absl::Duration nearly_min_duration =
>+      absl::time_internal::MakeDuration(kint64min, int64_t{1});
>+  constexpr absl::Duration negated_nearly_min_duration = -nearly_min_duration;
>+
>+  EXPECT_EQ(negated_max_duration, nearly_min_duration);
>+  EXPECT_EQ(negated_nearly_min_duration, max_duration);
>+  EXPECT_EQ(-(-max_duration), max_duration);
>+
>+  constexpr absl::Duration min_duration =
>+      absl::time_internal::MakeDuration(kint64min);
>+  constexpr absl::Duration negated_min_duration = -min_duration;
>+  EXPECT_EQ(negated_min_duration, absl::InfiniteDuration());
>+}
>+
>+TEST(Duration, AbsoluteValue) {
>+  EXPECT_EQ(absl::ZeroDuration(), AbsDuration(absl::ZeroDuration()));
>+  EXPECT_EQ(absl::Seconds(1), AbsDuration(absl::Seconds(1)));
>+  EXPECT_EQ(absl::Seconds(1), AbsDuration(absl::Seconds(-1)));
>+
>+  EXPECT_EQ(absl::InfiniteDuration(), AbsDuration(absl::InfiniteDuration()));
>+  EXPECT_EQ(absl::InfiniteDuration(), AbsDuration(-absl::InfiniteDuration()));
>+
>+  absl::Duration max_dur =
>+      absl::Seconds(kint64max) + (absl::Seconds(1) - absl::Nanoseconds(1) / 4);
>+  EXPECT_EQ(max_dur, AbsDuration(max_dur));
>+
>+  absl::Duration min_dur = absl::Seconds(kint64min);
>+  EXPECT_EQ(absl::InfiniteDuration(), AbsDuration(min_dur));
>+  EXPECT_EQ(max_dur, AbsDuration(min_dur + absl::Nanoseconds(1) / 4));
>+}
>+
>+TEST(Duration, Multiplication) {
>+#define TEST_MUL_OPS(UNIT)                                    \
>+  do {                                                        \
>+    EXPECT_EQ(UNIT(5), UNIT(2) * 2.5);                        \
>+    EXPECT_EQ(UNIT(2), UNIT(5) / 2.5);                        \
>+    EXPECT_EQ(UNIT(-5), UNIT(-2) * 2.5);                      \
>+    EXPECT_EQ(UNIT(-5), -UNIT(2) * 2.5);                      \
>+    EXPECT_EQ(UNIT(-5), UNIT(2) * -2.5);                      \
>+    EXPECT_EQ(UNIT(-2), UNIT(-5) / 2.5);                      \
>+    EXPECT_EQ(UNIT(-2), -UNIT(5) / 2.5);                      \
>+    EXPECT_EQ(UNIT(-2), UNIT(5) / -2.5);                      \
>+    EXPECT_EQ(UNIT(2), UNIT(11) % UNIT(3));                   \
>+    absl::Duration a = UNIT(2);                               \
>+    a *= 2.5;                                                 \
>+    EXPECT_EQ(UNIT(5), a);                                    \
>+    a /= 2.5;                                                 \
>+    EXPECT_EQ(UNIT(2), a);                                    \
>+    a %= UNIT(1);                                             \
>+    EXPECT_EQ(UNIT(0), a);                                    \
>+    absl::Duration big = UNIT(1000000000);                    \
>+    big *= 3;                                                 \
>+    big /= 3;                                                 \
>+    EXPECT_EQ(UNIT(1000000000), big);                         \
>+    EXPECT_EQ(-UNIT(2), -UNIT(2));                            \
>+    EXPECT_EQ(-UNIT(2), UNIT(2) * -1);                        \
>+    EXPECT_EQ(-UNIT(2), -1 * UNIT(2));                        \
>+    EXPECT_EQ(-UNIT(-2), UNIT(2));                            \
>+    EXPECT_EQ(2, UNIT(2) / UNIT(1));                          \
>+    absl::Duration rem;                                       \
>+    EXPECT_EQ(2, absl::IDivDuration(UNIT(2), UNIT(1), &rem)); \
>+    EXPECT_EQ(2.0, absl::FDivDuration(UNIT(2), UNIT(1)));     \
>+  } while (0)
>+
>+  TEST_MUL_OPS(absl::Nanoseconds);
>+  TEST_MUL_OPS(absl::Microseconds);
>+  TEST_MUL_OPS(absl::Milliseconds);
>+  TEST_MUL_OPS(absl::Seconds);
>+  TEST_MUL_OPS(absl::Minutes);
>+  TEST_MUL_OPS(absl::Hours);
>+
>+#undef TEST_MUL_OPS
>+
>+  // Ensures that multiplication and division by 1 with a maxed-out durations
>+  // doesn't lose precision.
>+  absl::Duration max_dur =
>+      absl::Seconds(kint64max) + (absl::Seconds(1) - absl::Nanoseconds(1) / 4);
>+  absl::Duration min_dur = absl::Seconds(kint64min);
>+  EXPECT_EQ(max_dur, max_dur * 1);
>+  EXPECT_EQ(max_dur, max_dur / 1);
>+  EXPECT_EQ(min_dur, min_dur * 1);
>+  EXPECT_EQ(min_dur, min_dur / 1);
>+
>+  // Tests division on a Duration with a large number of significant digits.
>+  // Tests when the digits span hi and lo as well as only in hi.
>+  absl::Duration sigfigs = absl::Seconds(2000000000) + absl::Nanoseconds(3);
>+  EXPECT_EQ(absl::Seconds(666666666) + absl::Nanoseconds(666666667) +
>+                absl::Nanoseconds(1) / 2,
>+            sigfigs / 3);
>+  sigfigs = absl::Seconds(7000000000LL);
>+  EXPECT_EQ(absl::Seconds(2333333333) + absl::Nanoseconds(333333333) +
>+                absl::Nanoseconds(1) / 4,
>+            sigfigs / 3);
>+
>+  EXPECT_EQ(absl::Seconds(7) + absl::Milliseconds(500), absl::Seconds(3) * 2.5);
>+  EXPECT_EQ(absl::Seconds(8) * -1 + absl::Milliseconds(300),
>+            (absl::Seconds(2) + absl::Milliseconds(200)) * -3.5);
>+  EXPECT_EQ(-absl::Seconds(8) + absl::Milliseconds(300),
>+            (absl::Seconds(2) + absl::Milliseconds(200)) * -3.5);
>+  EXPECT_EQ(absl::Seconds(1) + absl::Milliseconds(875),
>+            (absl::Seconds(7) + absl::Milliseconds(500)) / 4);
>+  EXPECT_EQ(absl::Seconds(30),
>+            (absl::Seconds(7) + absl::Milliseconds(500)) / 0.25);
>+  EXPECT_EQ(absl::Seconds(3),
>+            (absl::Seconds(7) + absl::Milliseconds(500)) / 2.5);
>+
>+  // Tests division remainder.
>+  EXPECT_EQ(absl::Nanoseconds(0), absl::Nanoseconds(7) % absl::Nanoseconds(1));
>+  EXPECT_EQ(absl::Nanoseconds(0), absl::Nanoseconds(0) % absl::Nanoseconds(10));
>+  EXPECT_EQ(absl::Nanoseconds(2), absl::Nanoseconds(7) % absl::Nanoseconds(5));
>+  EXPECT_EQ(absl::Nanoseconds(2), absl::Nanoseconds(2) % absl::Nanoseconds(5));
>+
>+  EXPECT_EQ(absl::Nanoseconds(1), absl::Nanoseconds(10) % absl::Nanoseconds(3));
>+  EXPECT_EQ(absl::Nanoseconds(1),
>+            absl::Nanoseconds(10) % absl::Nanoseconds(-3));
>+  EXPECT_EQ(absl::Nanoseconds(-1),
>+            absl::Nanoseconds(-10) % absl::Nanoseconds(3));
>+  EXPECT_EQ(absl::Nanoseconds(-1),
>+            absl::Nanoseconds(-10) % absl::Nanoseconds(-3));
>+
>+  EXPECT_EQ(absl::Milliseconds(100),
>+            absl::Seconds(1) % absl::Milliseconds(300));
>+  EXPECT_EQ(
>+      absl::Milliseconds(300),
>+      (absl::Seconds(3) + absl::Milliseconds(800)) % absl::Milliseconds(500));
>+
>+  EXPECT_EQ(absl::Nanoseconds(1), absl::Nanoseconds(1) % absl::Seconds(1));
>+  EXPECT_EQ(absl::Nanoseconds(-1), absl::Nanoseconds(-1) % absl::Seconds(1));
>+  EXPECT_EQ(0, absl::Nanoseconds(-1) / absl::Seconds(1));  // Actual -1e-9
>+
>+  // Tests identity a = (a/b)*b + a%b
>+#define TEST_MOD_IDENTITY(a, b) \
>+  EXPECT_EQ((a), ((a) / (b))*(b) + ((a)%(b)))
>+
>+  TEST_MOD_IDENTITY(absl::Seconds(0), absl::Seconds(2));
>+  TEST_MOD_IDENTITY(absl::Seconds(1), absl::Seconds(1));
>+  TEST_MOD_IDENTITY(absl::Seconds(1), absl::Seconds(2));
>+  TEST_MOD_IDENTITY(absl::Seconds(2), absl::Seconds(1));
>+
>+  TEST_MOD_IDENTITY(absl::Seconds(-2), absl::Seconds(1));
>+  TEST_MOD_IDENTITY(absl::Seconds(2), absl::Seconds(-1));
>+  TEST_MOD_IDENTITY(absl::Seconds(-2), absl::Seconds(-1));
>+
>+  TEST_MOD_IDENTITY(absl::Nanoseconds(0), absl::Nanoseconds(2));
>+  TEST_MOD_IDENTITY(absl::Nanoseconds(1), absl::Nanoseconds(1));
>+  TEST_MOD_IDENTITY(absl::Nanoseconds(1), absl::Nanoseconds(2));
>+  TEST_MOD_IDENTITY(absl::Nanoseconds(2), absl::Nanoseconds(1));
>+
>+  TEST_MOD_IDENTITY(absl::Nanoseconds(-2), absl::Nanoseconds(1));
>+  TEST_MOD_IDENTITY(absl::Nanoseconds(2), absl::Nanoseconds(-1));
>+  TEST_MOD_IDENTITY(absl::Nanoseconds(-2), absl::Nanoseconds(-1));
>+
>+  // Mixed seconds + subseconds
>+  absl::Duration mixed_a = absl::Seconds(1) + absl::Nanoseconds(2);
>+  absl::Duration mixed_b = absl::Seconds(1) + absl::Nanoseconds(3);
>+
>+  TEST_MOD_IDENTITY(absl::Seconds(0), mixed_a);
>+  TEST_MOD_IDENTITY(mixed_a, mixed_a);
>+  TEST_MOD_IDENTITY(mixed_a, mixed_b);
>+  TEST_MOD_IDENTITY(mixed_b, mixed_a);
>+
>+  TEST_MOD_IDENTITY(-mixed_a, mixed_b);
>+  TEST_MOD_IDENTITY(mixed_a, -mixed_b);
>+  TEST_MOD_IDENTITY(-mixed_a, -mixed_b);
>+
>+#undef TEST_MOD_IDENTITY
>+}
>+
>+TEST(Duration, Truncation) {
>+  const absl::Duration d = absl::Nanoseconds(1234567890);
>+  const absl::Duration inf = absl::InfiniteDuration();
>+  for (int unit_sign : {1, -1}) {  // sign shouldn't matter
>+    EXPECT_EQ(absl::Nanoseconds(1234567890),
>+              Trunc(d, unit_sign * absl::Nanoseconds(1)));
>+    EXPECT_EQ(absl::Microseconds(1234567),
>+              Trunc(d, unit_sign * absl::Microseconds(1)));
>+    EXPECT_EQ(absl::Milliseconds(1234),
>+              Trunc(d, unit_sign * absl::Milliseconds(1)));
>+    EXPECT_EQ(absl::Seconds(1), Trunc(d, unit_sign * absl::Seconds(1)));
>+    EXPECT_EQ(inf, Trunc(inf, unit_sign * absl::Seconds(1)));
>+
>+    EXPECT_EQ(absl::Nanoseconds(-1234567890),
>+              Trunc(-d, unit_sign * absl::Nanoseconds(1)));
>+    EXPECT_EQ(absl::Microseconds(-1234567),
>+              Trunc(-d, unit_sign * absl::Microseconds(1)));
>+    EXPECT_EQ(absl::Milliseconds(-1234),
>+              Trunc(-d, unit_sign * absl::Milliseconds(1)));
>+    EXPECT_EQ(absl::Seconds(-1), Trunc(-d, unit_sign * absl::Seconds(1)));
>+    EXPECT_EQ(-inf, Trunc(-inf, unit_sign * absl::Seconds(1)));
>+  }
>+}
>+
>+TEST(Duration, Flooring) {
>+  const absl::Duration d = absl::Nanoseconds(1234567890);
>+  const absl::Duration inf = absl::InfiniteDuration();
>+  for (int unit_sign : {1, -1}) {  // sign shouldn't matter
>+    EXPECT_EQ(absl::Nanoseconds(1234567890),
>+              absl::Floor(d, unit_sign * absl::Nanoseconds(1)));
>+    EXPECT_EQ(absl::Microseconds(1234567),
>+              absl::Floor(d, unit_sign * absl::Microseconds(1)));
>+    EXPECT_EQ(absl::Milliseconds(1234),
>+              absl::Floor(d, unit_sign * absl::Milliseconds(1)));
>+    EXPECT_EQ(absl::Seconds(1), absl::Floor(d, unit_sign * absl::Seconds(1)));
>+    EXPECT_EQ(inf, absl::Floor(inf, unit_sign * absl::Seconds(1)));
>+
>+    EXPECT_EQ(absl::Nanoseconds(-1234567890),
>+              absl::Floor(-d, unit_sign * absl::Nanoseconds(1)));
>+    EXPECT_EQ(absl::Microseconds(-1234568),
>+              absl::Floor(-d, unit_sign * absl::Microseconds(1)));
>+    EXPECT_EQ(absl::Milliseconds(-1235),
>+              absl::Floor(-d, unit_sign * absl::Milliseconds(1)));
>+    EXPECT_EQ(absl::Seconds(-2), absl::Floor(-d, unit_sign * absl::Seconds(1)));
>+    EXPECT_EQ(-inf, absl::Floor(-inf, unit_sign * absl::Seconds(1)));
>+  }
>+}
>+
>+TEST(Duration, Ceiling) {
>+  const absl::Duration d = absl::Nanoseconds(1234567890);
>+  const absl::Duration inf = absl::InfiniteDuration();
>+  for (int unit_sign : {1, -1}) {  // // sign shouldn't matter
>+    EXPECT_EQ(absl::Nanoseconds(1234567890),
>+              absl::Ceil(d, unit_sign * absl::Nanoseconds(1)));
>+    EXPECT_EQ(absl::Microseconds(1234568),
>+              absl::Ceil(d, unit_sign * absl::Microseconds(1)));
>+    EXPECT_EQ(absl::Milliseconds(1235),
>+              absl::Ceil(d, unit_sign * absl::Milliseconds(1)));
>+    EXPECT_EQ(absl::Seconds(2), absl::Ceil(d, unit_sign * absl::Seconds(1)));
>+    EXPECT_EQ(inf, absl::Ceil(inf, unit_sign * absl::Seconds(1)));
>+
>+    EXPECT_EQ(absl::Nanoseconds(-1234567890),
>+              absl::Ceil(-d, unit_sign * absl::Nanoseconds(1)));
>+    EXPECT_EQ(absl::Microseconds(-1234567),
>+              absl::Ceil(-d, unit_sign * absl::Microseconds(1)));
>+    EXPECT_EQ(absl::Milliseconds(-1234),
>+              absl::Ceil(-d, unit_sign * absl::Milliseconds(1)));
>+    EXPECT_EQ(absl::Seconds(-1), absl::Ceil(-d, unit_sign * absl::Seconds(1)));
>+    EXPECT_EQ(-inf, absl::Ceil(-inf, unit_sign * absl::Seconds(1)));
>+  }
>+}
>+
>+TEST(Duration, RoundTripUnits) {
>+  const int kRange = 100000;
>+
>+#define ROUND_TRIP_UNIT(U, LOW, HIGH)          \
>+  do {                                         \
>+    for (int64_t i = LOW; i < HIGH; ++i) {     \
>+      absl::Duration d = absl::U(i);           \
>+      if (d == absl::InfiniteDuration())       \
>+        EXPECT_EQ(kint64max, d / absl::U(1));  \
>+      else if (d == -absl::InfiniteDuration()) \
>+        EXPECT_EQ(kint64min, d / absl::U(1));  \
>+      else                                     \
>+        EXPECT_EQ(i, absl::U(i) / absl::U(1)); \
>+    }                                          \
>+  } while (0)
>+
>+  ROUND_TRIP_UNIT(Nanoseconds, kint64min, kint64min + kRange);
>+  ROUND_TRIP_UNIT(Nanoseconds, -kRange, kRange);
>+  ROUND_TRIP_UNIT(Nanoseconds, kint64max - kRange, kint64max);
>+
>+  ROUND_TRIP_UNIT(Microseconds, kint64min, kint64min + kRange);
>+  ROUND_TRIP_UNIT(Microseconds, -kRange, kRange);
>+  ROUND_TRIP_UNIT(Microseconds, kint64max - kRange, kint64max);
>+
>+  ROUND_TRIP_UNIT(Milliseconds, kint64min, kint64min + kRange);
>+  ROUND_TRIP_UNIT(Milliseconds, -kRange, kRange);
>+  ROUND_TRIP_UNIT(Milliseconds, kint64max - kRange, kint64max);
>+
>+  ROUND_TRIP_UNIT(Seconds, kint64min, kint64min + kRange);
>+  ROUND_TRIP_UNIT(Seconds, -kRange, kRange);
>+  ROUND_TRIP_UNIT(Seconds, kint64max - kRange, kint64max);
>+
>+  ROUND_TRIP_UNIT(Minutes, kint64min / 60, kint64min / 60 + kRange);
>+  ROUND_TRIP_UNIT(Minutes, -kRange, kRange);
>+  ROUND_TRIP_UNIT(Minutes, kint64max / 60 - kRange, kint64max / 60);
>+
>+  ROUND_TRIP_UNIT(Hours, kint64min / 3600, kint64min / 3600 + kRange);
>+  ROUND_TRIP_UNIT(Hours, -kRange, kRange);
>+  ROUND_TRIP_UNIT(Hours, kint64max / 3600 - kRange, kint64max / 3600);
>+
>+#undef ROUND_TRIP_UNIT
>+}
>+
>+TEST(Duration, TruncConversions) {
>+  // Tests ToTimespec()/DurationFromTimespec()
>+  const struct {
>+    absl::Duration d;
>+    timespec ts;
>+  } to_ts[] = {
>+      {absl::Seconds(1) + absl::Nanoseconds(1), {1, 1}},
>+      {absl::Seconds(1) + absl::Nanoseconds(1) / 2, {1, 0}},
>+      {absl::Seconds(1) + absl::Nanoseconds(0), {1, 0}},
>+      {absl::Seconds(0) + absl::Nanoseconds(0), {0, 0}},
>+      {absl::Seconds(0) - absl::Nanoseconds(1) / 2, {0, 0}},
>+      {absl::Seconds(0) - absl::Nanoseconds(1), {-1, 999999999}},
>+      {absl::Seconds(-1) + absl::Nanoseconds(1), {-1, 1}},
>+      {absl::Seconds(-1) + absl::Nanoseconds(1) / 2, {-1, 1}},
>+      {absl::Seconds(-1) + absl::Nanoseconds(0), {-1, 0}},
>+      {absl::Seconds(-1) - absl::Nanoseconds(1) / 2, {-1, 0}},
>+  };
>+  for (const auto& test : to_ts) {
>+    EXPECT_THAT(absl::ToTimespec(test.d), TimespecMatcher(test.ts));
>+  }
>+  const struct {
>+    timespec ts;
>+    absl::Duration d;
>+  } from_ts[] = {
>+      {{1, 1}, absl::Seconds(1) + absl::Nanoseconds(1)},
>+      {{1, 0}, absl::Seconds(1) + absl::Nanoseconds(0)},
>+      {{0, 0}, absl::Seconds(0) + absl::Nanoseconds(0)},
>+      {{0, -1}, absl::Seconds(0) - absl::Nanoseconds(1)},
>+      {{-1, 999999999}, absl::Seconds(0) - absl::Nanoseconds(1)},
>+      {{-1, 1}, absl::Seconds(-1) + absl::Nanoseconds(1)},
>+      {{-1, 0}, absl::Seconds(-1) + absl::Nanoseconds(0)},
>+      {{-1, -1}, absl::Seconds(-1) - absl::Nanoseconds(1)},
>+      {{-2, 999999999}, absl::Seconds(-1) - absl::Nanoseconds(1)},
>+  };
>+  for (const auto& test : from_ts) {
>+    EXPECT_EQ(test.d, absl::DurationFromTimespec(test.ts));
>+  }
>+
>+  // Tests ToTimeval()/DurationFromTimeval() (same as timespec above)
>+  const struct {
>+    absl::Duration d;
>+    timeval tv;
>+  } to_tv[] = {
>+      {absl::Seconds(1) + absl::Microseconds(1), {1, 1}},
>+      {absl::Seconds(1) + absl::Microseconds(1) / 2, {1, 0}},
>+      {absl::Seconds(1) + absl::Microseconds(0), {1, 0}},
>+      {absl::Seconds(0) + absl::Microseconds(0), {0, 0}},
>+      {absl::Seconds(0) - absl::Microseconds(1) / 2, {0, 0}},
>+      {absl::Seconds(0) - absl::Microseconds(1), {-1, 999999}},
>+      {absl::Seconds(-1) + absl::Microseconds(1), {-1, 1}},
>+      {absl::Seconds(-1) + absl::Microseconds(1) / 2, {-1, 1}},
>+      {absl::Seconds(-1) + absl::Microseconds(0), {-1, 0}},
>+      {absl::Seconds(-1) - absl::Microseconds(1) / 2, {-1, 0}},
>+  };
>+  for (const auto& test : to_tv) {
>+    EXPECT_THAT(absl::ToTimeval(test.d), TimevalMatcher(test.tv));
>+  }
>+  const struct {
>+    timeval tv;
>+    absl::Duration d;
>+  } from_tv[] = {
>+      {{1, 1}, absl::Seconds(1) + absl::Microseconds(1)},
>+      {{1, 0}, absl::Seconds(1) + absl::Microseconds(0)},
>+      {{0, 0}, absl::Seconds(0) + absl::Microseconds(0)},
>+      {{0, -1}, absl::Seconds(0) - absl::Microseconds(1)},
>+      {{-1, 999999}, absl::Seconds(0) - absl::Microseconds(1)},
>+      {{-1, 1}, absl::Seconds(-1) + absl::Microseconds(1)},
>+      {{-1, 0}, absl::Seconds(-1) + absl::Microseconds(0)},
>+      {{-1, -1}, absl::Seconds(-1) - absl::Microseconds(1)},
>+      {{-2, 999999}, absl::Seconds(-1) - absl::Microseconds(1)},
>+  };
>+  for (const auto& test : from_tv) {
>+    EXPECT_EQ(test.d, absl::DurationFromTimeval(test.tv));
>+  }
>+}
>+
>+TEST(Duration, SmallConversions) {
>+  // Special tests for conversions of small durations.
>+
>+  EXPECT_EQ(absl::ZeroDuration(), absl::Seconds(0));
>+  // TODO(bww): Is the next one OK?
>+  EXPECT_EQ(absl::ZeroDuration(), absl::Seconds(0.124999999e-9));
>+  EXPECT_EQ(absl::Nanoseconds(1) / 4, absl::Seconds(0.125e-9));
>+  EXPECT_EQ(absl::Nanoseconds(1) / 4, absl::Seconds(0.250e-9));
>+  EXPECT_EQ(absl::Nanoseconds(1) / 2, absl::Seconds(0.375e-9));
>+  EXPECT_EQ(absl::Nanoseconds(1) / 2, absl::Seconds(0.500e-9));
>+  EXPECT_EQ(absl::Nanoseconds(3) / 4, absl::Seconds(0.625e-9));
>+  EXPECT_EQ(absl::Nanoseconds(3) / 4, absl::Seconds(0.750e-9));
>+  EXPECT_EQ(absl::Nanoseconds(1), absl::Seconds(0.875e-9));
>+  EXPECT_EQ(absl::Nanoseconds(1), absl::Seconds(1.000e-9));
>+
>+  EXPECT_EQ(absl::ZeroDuration(), absl::Seconds(-0.124999999e-9));
>+  EXPECT_EQ(-absl::Nanoseconds(1) / 4, absl::Seconds(-0.125e-9));
>+  EXPECT_EQ(-absl::Nanoseconds(1) / 4, absl::Seconds(-0.250e-9));
>+  EXPECT_EQ(-absl::Nanoseconds(1) / 2, absl::Seconds(-0.375e-9));
>+  EXPECT_EQ(-absl::Nanoseconds(1) / 2, absl::Seconds(-0.500e-9));
>+  EXPECT_EQ(-absl::Nanoseconds(3) / 4, absl::Seconds(-0.625e-9));
>+  EXPECT_EQ(-absl::Nanoseconds(3) / 4, absl::Seconds(-0.750e-9));
>+  EXPECT_EQ(-absl::Nanoseconds(1), absl::Seconds(-0.875e-9));
>+  EXPECT_EQ(-absl::Nanoseconds(1), absl::Seconds(-1.000e-9));
>+
>+  timespec ts;
>+  ts.tv_sec = 0;
>+  ts.tv_nsec = 0;
>+  EXPECT_THAT(ToTimespec(absl::Nanoseconds(0)), TimespecMatcher(ts));
>+  // TODO(bww): Are the next three OK?
>+  EXPECT_THAT(ToTimespec(absl::Nanoseconds(1) / 4), TimespecMatcher(ts));
>+  EXPECT_THAT(ToTimespec(absl::Nanoseconds(2) / 4), TimespecMatcher(ts));
>+  EXPECT_THAT(ToTimespec(absl::Nanoseconds(3) / 4), TimespecMatcher(ts));
>+  ts.tv_nsec = 1;
>+  EXPECT_THAT(ToTimespec(absl::Nanoseconds(4) / 4), TimespecMatcher(ts));
>+  EXPECT_THAT(ToTimespec(absl::Nanoseconds(5) / 4), TimespecMatcher(ts));
>+  EXPECT_THAT(ToTimespec(absl::Nanoseconds(6) / 4), TimespecMatcher(ts));
>+  EXPECT_THAT(ToTimespec(absl::Nanoseconds(7) / 4), TimespecMatcher(ts));
>+  ts.tv_nsec = 2;
>+  EXPECT_THAT(ToTimespec(absl::Nanoseconds(8) / 4), TimespecMatcher(ts));
>+
>+  timeval tv;
>+  tv.tv_sec = 0;
>+  tv.tv_usec = 0;
>+  EXPECT_THAT(ToTimeval(absl::Nanoseconds(0)), TimevalMatcher(tv));
>+  // TODO(bww): Is the next one OK?
>+  EXPECT_THAT(ToTimeval(absl::Nanoseconds(999)), TimevalMatcher(tv));
>+  tv.tv_usec = 1;
>+  EXPECT_THAT(ToTimeval(absl::Nanoseconds(1000)), TimevalMatcher(tv));
>+  EXPECT_THAT(ToTimeval(absl::Nanoseconds(1999)), TimevalMatcher(tv));
>+  tv.tv_usec = 2;
>+  EXPECT_THAT(ToTimeval(absl::Nanoseconds(2000)), TimevalMatcher(tv));
>+}
>+
>+void VerifySameAsMul(double time_as_seconds, int* const misses) {
>+  auto direct_seconds = absl::Seconds(time_as_seconds);
>+  auto mul_by_one_second = time_as_seconds * absl::Seconds(1);
>+  if (direct_seconds != mul_by_one_second) {
>+    if (*misses > 10) return;
>+    ASSERT_LE(++(*misses), 10) << "Too many errors, not reporting more.";
>+    EXPECT_EQ(direct_seconds, mul_by_one_second)
>+        << "given double time_as_seconds = " << std::setprecision(17)
>+        << time_as_seconds;
>+  }
>+}
>+
>+// For a variety of interesting durations, we find the exact point
>+// where one double converts to that duration, and the very next double
>+// converts to the next duration.  For both of those points, verify that
>+// Seconds(point) returns the same duration as point * Seconds(1.0)
>+TEST(Duration, ToDoubleSecondsCheckEdgeCases) {
>+  constexpr uint32_t kTicksPerSecond = absl::time_internal::kTicksPerSecond;
>+  constexpr auto duration_tick = absl::time_internal::MakeDuration(0, 1u);
>+  int misses = 0;
>+  for (int64_t seconds = 0; seconds < 99; ++seconds) {
>+    uint32_t tick_vals[] = {0, +999, +999999, +999999999, kTicksPerSecond - 1,
>+                            0, 1000, 1000000, 1000000000, kTicksPerSecond,
>+                            1, 1001, 1000001, 1000000001, kTicksPerSecond + 1,
>+                            2, 1002, 1000002, 1000000002, kTicksPerSecond + 2,
>+                            3, 1003, 1000003, 1000000003, kTicksPerSecond + 3,
>+                            4, 1004, 1000004, 1000000004, kTicksPerSecond + 4,
>+                            5, 6,    7,       8,          9};
>+    for (uint32_t ticks : tick_vals) {
>+      absl::Duration s_plus_t = absl::Seconds(seconds) + ticks * duration_tick;
>+      for (absl::Duration d : {s_plus_t, -s_plus_t}) {
>+        absl::Duration after_d = d + duration_tick;
>+        EXPECT_NE(d, after_d);
>+        EXPECT_EQ(after_d - d, duration_tick);
>+
>+        double low_edge = ToDoubleSeconds(d);
>+        EXPECT_EQ(d, absl::Seconds(low_edge));
>+
>+        double high_edge = ToDoubleSeconds(after_d);
>+        EXPECT_EQ(after_d, absl::Seconds(high_edge));
>+
>+        for (;;) {
>+          double midpoint = low_edge + (high_edge - low_edge) / 2;
>+          if (midpoint == low_edge || midpoint == high_edge) break;
>+          absl::Duration mid_duration = absl::Seconds(midpoint);
>+          if (mid_duration == d) {
>+            low_edge = midpoint;
>+          } else {
>+            EXPECT_EQ(mid_duration, after_d);
>+            high_edge = midpoint;
>+          }
>+        }
>+        // Now low_edge is the highest double that converts to Duration d,
>+        // and high_edge is the lowest double that converts to Duration after_d.
>+        VerifySameAsMul(low_edge, &misses);
>+        VerifySameAsMul(high_edge, &misses);
>+      }
>+    }
>+  }
>+}
>+
>+TEST(Duration, ToDoubleSecondsCheckRandom) {
>+  std::random_device rd;
>+  std::seed_seq seed({rd(), rd(), rd(), rd(), rd(), rd(), rd(), rd()});
>+  std::mt19937_64 gen(seed);
>+  // We want doubles distributed from 1/8ns up to 2^63, where
>+  // as many values are tested from 1ns to 2ns as from 1sec to 2sec,
>+  // so even distribute along a log-scale of those values, and
>+  // exponentiate before using them.  (9.223377e+18 is just slightly
>+  // out of bounds for absl::Duration.)
>+  std::uniform_real_distribution<double> uniform(std::log(0.125e-9),
>+                                                 std::log(9.223377e+18));
>+  int misses = 0;
>+  for (int i = 0; i < 1000000; ++i) {
>+    double d = std::exp(uniform(gen));
>+    VerifySameAsMul(d, &misses);
>+    VerifySameAsMul(-d, &misses);
>+  }
>+}
>+
>+TEST(Duration, ConversionSaturation) {
>+  absl::Duration d;
>+
>+  const auto max_timeval_sec =
>+      std::numeric_limits<decltype(timeval::tv_sec)>::max();
>+  const auto min_timeval_sec =
>+      std::numeric_limits<decltype(timeval::tv_sec)>::min();
>+  timeval tv;
>+  tv.tv_sec = max_timeval_sec;
>+  tv.tv_usec = 999998;
>+  d = absl::DurationFromTimeval(tv);
>+  tv = ToTimeval(d);
>+  EXPECT_EQ(max_timeval_sec, tv.tv_sec);
>+  EXPECT_EQ(999998, tv.tv_usec);
>+  d += absl::Microseconds(1);
>+  tv = ToTimeval(d);
>+  EXPECT_EQ(max_timeval_sec, tv.tv_sec);
>+  EXPECT_EQ(999999, tv.tv_usec);
>+  d += absl::Microseconds(1);  // no effect
>+  tv = ToTimeval(d);
>+  EXPECT_EQ(max_timeval_sec, tv.tv_sec);
>+  EXPECT_EQ(999999, tv.tv_usec);
>+
>+  tv.tv_sec = min_timeval_sec;
>+  tv.tv_usec = 1;
>+  d = absl::DurationFromTimeval(tv);
>+  tv = ToTimeval(d);
>+  EXPECT_EQ(min_timeval_sec, tv.tv_sec);
>+  EXPECT_EQ(1, tv.tv_usec);
>+  d -= absl::Microseconds(1);
>+  tv = ToTimeval(d);
>+  EXPECT_EQ(min_timeval_sec, tv.tv_sec);
>+  EXPECT_EQ(0, tv.tv_usec);
>+  d -= absl::Microseconds(1);  // no effect
>+  tv = ToTimeval(d);
>+  EXPECT_EQ(min_timeval_sec, tv.tv_sec);
>+  EXPECT_EQ(0, tv.tv_usec);
>+
>+  const auto max_timespec_sec =
>+      std::numeric_limits<decltype(timespec::tv_sec)>::max();
>+  const auto min_timespec_sec =
>+      std::numeric_limits<decltype(timespec::tv_sec)>::min();
>+  timespec ts;
>+  ts.tv_sec = max_timespec_sec;
>+  ts.tv_nsec = 999999998;
>+  d = absl::DurationFromTimespec(ts);
>+  ts = absl::ToTimespec(d);
>+  EXPECT_EQ(max_timespec_sec, ts.tv_sec);
>+  EXPECT_EQ(999999998, ts.tv_nsec);
>+  d += absl::Nanoseconds(1);
>+  ts = absl::ToTimespec(d);
>+  EXPECT_EQ(max_timespec_sec, ts.tv_sec);
>+  EXPECT_EQ(999999999, ts.tv_nsec);
>+  d += absl::Nanoseconds(1);  // no effect
>+  ts = absl::ToTimespec(d);
>+  EXPECT_EQ(max_timespec_sec, ts.tv_sec);
>+  EXPECT_EQ(999999999, ts.tv_nsec);
>+
>+  ts.tv_sec = min_timespec_sec;
>+  ts.tv_nsec = 1;
>+  d = absl::DurationFromTimespec(ts);
>+  ts = absl::ToTimespec(d);
>+  EXPECT_EQ(min_timespec_sec, ts.tv_sec);
>+  EXPECT_EQ(1, ts.tv_nsec);
>+  d -= absl::Nanoseconds(1);
>+  ts = absl::ToTimespec(d);
>+  EXPECT_EQ(min_timespec_sec, ts.tv_sec);
>+  EXPECT_EQ(0, ts.tv_nsec);
>+  d -= absl::Nanoseconds(1);  // no effect
>+  ts = absl::ToTimespec(d);
>+  EXPECT_EQ(min_timespec_sec, ts.tv_sec);
>+  EXPECT_EQ(0, ts.tv_nsec);
>+}
>+
>+TEST(Duration, FormatDuration) {
>+  // Example from Go's docs.
>+  EXPECT_EQ("72h3m0.5s",
>+            absl::FormatDuration(absl::Hours(72) + absl::Minutes(3) +
>+                                 absl::Milliseconds(500)));
>+  // Go's largest time: 2540400h10m10.000000000s
>+  EXPECT_EQ("2540400h10m10s",
>+            absl::FormatDuration(absl::Hours(2540400) + absl::Minutes(10) +
>+                                 absl::Seconds(10)));
>+
>+  EXPECT_EQ("0", absl::FormatDuration(absl::ZeroDuration()));
>+  EXPECT_EQ("0", absl::FormatDuration(absl::Seconds(0)));
>+  EXPECT_EQ("0", absl::FormatDuration(absl::Nanoseconds(0)));
>+
>+  EXPECT_EQ("1ns", absl::FormatDuration(absl::Nanoseconds(1)));
>+  EXPECT_EQ("1us", absl::FormatDuration(absl::Microseconds(1)));
>+  EXPECT_EQ("1ms", absl::FormatDuration(absl::Milliseconds(1)));
>+  EXPECT_EQ("1s", absl::FormatDuration(absl::Seconds(1)));
>+  EXPECT_EQ("1m", absl::FormatDuration(absl::Minutes(1)));
>+  EXPECT_EQ("1h", absl::FormatDuration(absl::Hours(1)));
>+
>+  EXPECT_EQ("1h1m", absl::FormatDuration(absl::Hours(1) + absl::Minutes(1)));
>+  EXPECT_EQ("1h1s", absl::FormatDuration(absl::Hours(1) + absl::Seconds(1)));
>+  EXPECT_EQ("1m1s", absl::FormatDuration(absl::Minutes(1) + absl::Seconds(1)));
>+
>+  EXPECT_EQ("1h0.25s",
>+            absl::FormatDuration(absl::Hours(1) + absl::Milliseconds(250)));
>+  EXPECT_EQ("1m0.25s",
>+            absl::FormatDuration(absl::Minutes(1) + absl::Milliseconds(250)));
>+  EXPECT_EQ("1h1m0.25s",
>+            absl::FormatDuration(absl::Hours(1) + absl::Minutes(1) +
>+                                 absl::Milliseconds(250)));
>+  EXPECT_EQ("1h0.0005s",
>+            absl::FormatDuration(absl::Hours(1) + absl::Microseconds(500)));
>+  EXPECT_EQ("1h0.0000005s",
>+            absl::FormatDuration(absl::Hours(1) + absl::Nanoseconds(500)));
>+
>+  // Subsecond special case.
>+  EXPECT_EQ("1.5ns", absl::FormatDuration(absl::Nanoseconds(1) +
>+                                          absl::Nanoseconds(1) / 2));
>+  EXPECT_EQ("1.25ns", absl::FormatDuration(absl::Nanoseconds(1) +
>+                                           absl::Nanoseconds(1) / 4));
>+  EXPECT_EQ("1ns", absl::FormatDuration(absl::Nanoseconds(1) +
>+                                        absl::Nanoseconds(1) / 9));
>+  EXPECT_EQ("1.2us", absl::FormatDuration(absl::Microseconds(1) +
>+                                          absl::Nanoseconds(200)));
>+  EXPECT_EQ("1.2ms", absl::FormatDuration(absl::Milliseconds(1) +
>+                                          absl::Microseconds(200)));
>+  EXPECT_EQ("1.0002ms", absl::FormatDuration(absl::Milliseconds(1) +
>+                                             absl::Nanoseconds(200)));
>+  EXPECT_EQ("1.00001ms", absl::FormatDuration(absl::Milliseconds(1) +
>+                                              absl::Nanoseconds(10)));
>+  EXPECT_EQ("1.000001ms",
>+            absl::FormatDuration(absl::Milliseconds(1) + absl::Nanoseconds(1)));
>+
>+  // Negative durations.
>+  EXPECT_EQ("-1ns", absl::FormatDuration(absl::Nanoseconds(-1)));
>+  EXPECT_EQ("-1us", absl::FormatDuration(absl::Microseconds(-1)));
>+  EXPECT_EQ("-1ms", absl::FormatDuration(absl::Milliseconds(-1)));
>+  EXPECT_EQ("-1s", absl::FormatDuration(absl::Seconds(-1)));
>+  EXPECT_EQ("-1m", absl::FormatDuration(absl::Minutes(-1)));
>+  EXPECT_EQ("-1h", absl::FormatDuration(absl::Hours(-1)));
>+
>+  EXPECT_EQ("-1h1m",
>+            absl::FormatDuration(-(absl::Hours(1) + absl::Minutes(1))));
>+  EXPECT_EQ("-1h1s",
>+            absl::FormatDuration(-(absl::Hours(1) + absl::Seconds(1))));
>+  EXPECT_EQ("-1m1s",
>+            absl::FormatDuration(-(absl::Minutes(1) + absl::Seconds(1))));
>+
>+  EXPECT_EQ("-1ns", absl::FormatDuration(absl::Nanoseconds(-1)));
>+  EXPECT_EQ("-1.2us", absl::FormatDuration(
>+                          -(absl::Microseconds(1) + absl::Nanoseconds(200))));
>+  EXPECT_EQ("-1.2ms", absl::FormatDuration(
>+                          -(absl::Milliseconds(1) + absl::Microseconds(200))));
>+  EXPECT_EQ("-1.0002ms", absl::FormatDuration(-(absl::Milliseconds(1) +
>+                                                absl::Nanoseconds(200))));
>+  EXPECT_EQ("-1.00001ms", absl::FormatDuration(-(absl::Milliseconds(1) +
>+                                                 absl::Nanoseconds(10))));
>+  EXPECT_EQ("-1.000001ms", absl::FormatDuration(-(absl::Milliseconds(1) +
>+                                                  absl::Nanoseconds(1))));
>+
>+  //
>+  // Interesting corner cases.
>+  //
>+
>+  const absl::Duration qns = absl::Nanoseconds(1) / 4;
>+  const absl::Duration max_dur =
>+      absl::Seconds(kint64max) + (absl::Seconds(1) - qns);
>+  const absl::Duration min_dur = absl::Seconds(kint64min);
>+
>+  EXPECT_EQ("0.25ns", absl::FormatDuration(qns));
>+  EXPECT_EQ("-0.25ns", absl::FormatDuration(-qns));
>+  EXPECT_EQ("2562047788015215h30m7.99999999975s",
>+            absl::FormatDuration(max_dur));
>+  EXPECT_EQ("-2562047788015215h30m8s", absl::FormatDuration(min_dur));
>+
>+  // Tests printing full precision from units that print using FDivDuration
>+  EXPECT_EQ("55.00000000025s", absl::FormatDuration(absl::Seconds(55) + qns));
>+  EXPECT_EQ("55.00000025ms",
>+            absl::FormatDuration(absl::Milliseconds(55) + qns));
>+  EXPECT_EQ("55.00025us", absl::FormatDuration(absl::Microseconds(55) + qns));
>+  EXPECT_EQ("55.25ns", absl::FormatDuration(absl::Nanoseconds(55) + qns));
>+
>+  // Formatting infinity
>+  EXPECT_EQ("inf", absl::FormatDuration(absl::InfiniteDuration()));
>+  EXPECT_EQ("-inf", absl::FormatDuration(-absl::InfiniteDuration()));
>+
>+  // Formatting approximately +/- 100 billion years
>+  const absl::Duration huge_range = ApproxYears(100000000000);
>+  EXPECT_EQ("876000000000000h", absl::FormatDuration(huge_range));
>+  EXPECT_EQ("-876000000000000h", absl::FormatDuration(-huge_range));
>+
>+  EXPECT_EQ("876000000000000h0.999999999s",
>+            absl::FormatDuration(huge_range +
>+                                 (absl::Seconds(1) - absl::Nanoseconds(1))));
>+  EXPECT_EQ("876000000000000h0.9999999995s",
>+            absl::FormatDuration(
>+                huge_range + (absl::Seconds(1) - absl::Nanoseconds(1) / 2)));
>+  EXPECT_EQ("876000000000000h0.99999999975s",
>+            absl::FormatDuration(
>+                huge_range + (absl::Seconds(1) - absl::Nanoseconds(1) / 4)));
>+
>+  EXPECT_EQ("-876000000000000h0.999999999s",
>+            absl::FormatDuration(-huge_range -
>+                                 (absl::Seconds(1) - absl::Nanoseconds(1))));
>+  EXPECT_EQ("-876000000000000h0.9999999995s",
>+            absl::FormatDuration(
>+                -huge_range - (absl::Seconds(1) - absl::Nanoseconds(1) / 2)));
>+  EXPECT_EQ("-876000000000000h0.99999999975s",
>+            absl::FormatDuration(
>+                -huge_range - (absl::Seconds(1) - absl::Nanoseconds(1) / 4)));
>+}
>+
>+TEST(Duration, ParseDuration) {
>+  absl::Duration d;
>+
>+  // No specified unit. Should only work for zero and infinity.
>+  EXPECT_TRUE(absl::ParseDuration("0", &d));
>+  EXPECT_EQ(absl::ZeroDuration(), d);
>+  EXPECT_TRUE(absl::ParseDuration("+0", &d));
>+  EXPECT_EQ(absl::ZeroDuration(), d);
>+  EXPECT_TRUE(absl::ParseDuration("-0", &d));
>+  EXPECT_EQ(absl::ZeroDuration(), d);
>+
>+  EXPECT_TRUE(absl::ParseDuration("inf", &d));
>+  EXPECT_EQ(absl::InfiniteDuration(), d);
>+  EXPECT_TRUE(absl::ParseDuration("+inf", &d));
>+  EXPECT_EQ(absl::InfiniteDuration(), d);
>+  EXPECT_TRUE(absl::ParseDuration("-inf", &d));
>+  EXPECT_EQ(-absl::InfiniteDuration(), d);
>+  EXPECT_FALSE(absl::ParseDuration("infBlah", &d));
>+
>+  // Illegal input forms.
>+  EXPECT_FALSE(absl::ParseDuration("", &d));
>+  EXPECT_FALSE(absl::ParseDuration("0.0", &d));
>+  EXPECT_FALSE(absl::ParseDuration(".0", &d));
>+  EXPECT_FALSE(absl::ParseDuration(".", &d));
>+  EXPECT_FALSE(absl::ParseDuration("01", &d));
>+  EXPECT_FALSE(absl::ParseDuration("1", &d));
>+  EXPECT_FALSE(absl::ParseDuration("-1", &d));
>+  EXPECT_FALSE(absl::ParseDuration("2", &d));
>+  EXPECT_FALSE(absl::ParseDuration("2 s", &d));
>+  EXPECT_FALSE(absl::ParseDuration(".s", &d));
>+  EXPECT_FALSE(absl::ParseDuration("-.s", &d));
>+  EXPECT_FALSE(absl::ParseDuration("s", &d));
>+  EXPECT_FALSE(absl::ParseDuration(" 2s", &d));
>+  EXPECT_FALSE(absl::ParseDuration("2s ", &d));
>+  EXPECT_FALSE(absl::ParseDuration(" 2s ", &d));
>+  EXPECT_FALSE(absl::ParseDuration("2mt", &d));
>+  EXPECT_FALSE(absl::ParseDuration("1e3s", &d));
>+
>+  // One unit type.
>+  EXPECT_TRUE(absl::ParseDuration("1ns", &d));
>+  EXPECT_EQ(absl::Nanoseconds(1), d);
>+  EXPECT_TRUE(absl::ParseDuration("1us", &d));
>+  EXPECT_EQ(absl::Microseconds(1), d);
>+  EXPECT_TRUE(absl::ParseDuration("1ms", &d));
>+  EXPECT_EQ(absl::Milliseconds(1), d);
>+  EXPECT_TRUE(absl::ParseDuration("1s", &d));
>+  EXPECT_EQ(absl::Seconds(1), d);
>+  EXPECT_TRUE(absl::ParseDuration("2m", &d));
>+  EXPECT_EQ(absl::Minutes(2), d);
>+  EXPECT_TRUE(absl::ParseDuration("2h", &d));
>+  EXPECT_EQ(absl::Hours(2), d);
>+
>+  // Huge counts of a unit.
>+  EXPECT_TRUE(absl::ParseDuration("9223372036854775807us", &d));
>+  EXPECT_EQ(absl::Microseconds(9223372036854775807), d);
>+  EXPECT_TRUE(absl::ParseDuration("-9223372036854775807us", &d));
>+  EXPECT_EQ(absl::Microseconds(-9223372036854775807), d);
>+
>+  // Multiple units.
>+  EXPECT_TRUE(absl::ParseDuration("2h3m4s", &d));
>+  EXPECT_EQ(absl::Hours(2) + absl::Minutes(3) + absl::Seconds(4), d);
>+  EXPECT_TRUE(absl::ParseDuration("3m4s5us", &d));
>+  EXPECT_EQ(absl::Minutes(3) + absl::Seconds(4) + absl::Microseconds(5), d);
>+  EXPECT_TRUE(absl::ParseDuration("2h3m4s5ms6us7ns", &d));
>+  EXPECT_EQ(absl::Hours(2) + absl::Minutes(3) + absl::Seconds(4) +
>+                absl::Milliseconds(5) + absl::Microseconds(6) +
>+                absl::Nanoseconds(7),
>+            d);
>+
>+  // Multiple units out of order.
>+  EXPECT_TRUE(absl::ParseDuration("2us3m4s5h", &d));
>+  EXPECT_EQ(absl::Hours(5) + absl::Minutes(3) + absl::Seconds(4) +
>+                absl::Microseconds(2),
>+            d);
>+
>+  // Fractional values of units.
>+  EXPECT_TRUE(absl::ParseDuration("1.5ns", &d));
>+  EXPECT_EQ(1.5 * absl::Nanoseconds(1), d);
>+  EXPECT_TRUE(absl::ParseDuration("1.5us", &d));
>+  EXPECT_EQ(1.5 * absl::Microseconds(1), d);
>+  EXPECT_TRUE(absl::ParseDuration("1.5ms", &d));
>+  EXPECT_EQ(1.5 * absl::Milliseconds(1), d);
>+  EXPECT_TRUE(absl::ParseDuration("1.5s", &d));
>+  EXPECT_EQ(1.5 * absl::Seconds(1), d);
>+  EXPECT_TRUE(absl::ParseDuration("1.5m", &d));
>+  EXPECT_EQ(1.5 * absl::Minutes(1), d);
>+  EXPECT_TRUE(absl::ParseDuration("1.5h", &d));
>+  EXPECT_EQ(1.5 * absl::Hours(1), d);
>+
>+  // Huge fractional counts of a unit.
>+  EXPECT_TRUE(absl::ParseDuration("0.4294967295s", &d));
>+  EXPECT_EQ(absl::Nanoseconds(429496729) + absl::Nanoseconds(1) / 2, d);
>+  EXPECT_TRUE(absl::ParseDuration("0.429496729501234567890123456789s", &d));
>+  EXPECT_EQ(absl::Nanoseconds(429496729) + absl::Nanoseconds(1) / 2, d);
>+
>+  // Negative durations.
>+  EXPECT_TRUE(absl::ParseDuration("-1s", &d));
>+  EXPECT_EQ(absl::Seconds(-1), d);
>+  EXPECT_TRUE(absl::ParseDuration("-1m", &d));
>+  EXPECT_EQ(absl::Minutes(-1), d);
>+  EXPECT_TRUE(absl::ParseDuration("-1h", &d));
>+  EXPECT_EQ(absl::Hours(-1), d);
>+
>+  EXPECT_TRUE(absl::ParseDuration("-1h2s", &d));
>+  EXPECT_EQ(-(absl::Hours(1) + absl::Seconds(2)), d);
>+  EXPECT_FALSE(absl::ParseDuration("1h-2s", &d));
>+  EXPECT_FALSE(absl::ParseDuration("-1h-2s", &d));
>+  EXPECT_FALSE(absl::ParseDuration("-1h -2s", &d));
>+}
>+
>+TEST(Duration, FormatParseRoundTrip) {
>+#define TEST_PARSE_ROUNDTRIP(d)                \
>+  do {                                         \
>+    std::string s = absl::FormatDuration(d);        \
>+    absl::Duration dur;                        \
>+    EXPECT_TRUE(absl::ParseDuration(s, &dur)); \
>+    EXPECT_EQ(d, dur);                         \
>+  } while (0)
>+
>+  TEST_PARSE_ROUNDTRIP(absl::Nanoseconds(1));
>+  TEST_PARSE_ROUNDTRIP(absl::Microseconds(1));
>+  TEST_PARSE_ROUNDTRIP(absl::Milliseconds(1));
>+  TEST_PARSE_ROUNDTRIP(absl::Seconds(1));
>+  TEST_PARSE_ROUNDTRIP(absl::Minutes(1));
>+  TEST_PARSE_ROUNDTRIP(absl::Hours(1));
>+  TEST_PARSE_ROUNDTRIP(absl::Hours(1) + absl::Nanoseconds(2));
>+
>+  TEST_PARSE_ROUNDTRIP(absl::Nanoseconds(-1));
>+  TEST_PARSE_ROUNDTRIP(absl::Microseconds(-1));
>+  TEST_PARSE_ROUNDTRIP(absl::Milliseconds(-1));
>+  TEST_PARSE_ROUNDTRIP(absl::Seconds(-1));
>+  TEST_PARSE_ROUNDTRIP(absl::Minutes(-1));
>+  TEST_PARSE_ROUNDTRIP(absl::Hours(-1));
>+
>+  TEST_PARSE_ROUNDTRIP(absl::Hours(-1) + absl::Nanoseconds(2));
>+  TEST_PARSE_ROUNDTRIP(absl::Hours(1) + absl::Nanoseconds(-2));
>+  TEST_PARSE_ROUNDTRIP(absl::Hours(-1) + absl::Nanoseconds(-2));
>+
>+  TEST_PARSE_ROUNDTRIP(absl::Nanoseconds(1) +
>+                       absl::Nanoseconds(1) / 4);  // 1.25ns
>+
>+  const absl::Duration huge_range = ApproxYears(100000000000);
>+  TEST_PARSE_ROUNDTRIP(huge_range);
>+  TEST_PARSE_ROUNDTRIP(huge_range + (absl::Seconds(1) - absl::Nanoseconds(1)));
>+
>+#undef TEST_PARSE_ROUNDTRIP
>+}
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/format.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/format.cc
>new file mode 100644
>index 00000000000..e98e60a372b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/format.cc
>@@ -0,0 +1,139 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include <string.h>
>+#include <cctype>
>+#include <cstdint>
>+
>+#include "absl/time/internal/cctz/include/cctz/time_zone.h"
>+#include "absl/time/time.h"
>+
>+namespace cctz = absl::time_internal::cctz;
>+
>+namespace absl {
>+
>+extern const char RFC3339_full[] = "%Y-%m-%dT%H:%M:%E*S%Ez";
>+extern const char RFC3339_sec[] =  "%Y-%m-%dT%H:%M:%S%Ez";
>+
>+extern const char RFC1123_full[] = "%a, %d %b %E4Y %H:%M:%S %z";
>+extern const char RFC1123_no_wday[] =  "%d %b %E4Y %H:%M:%S %z";
>+
>+namespace {
>+
>+const char kInfiniteFutureStr[] = "infinite-future";
>+const char kInfinitePastStr[] = "infinite-past";
>+
>+struct cctz_parts {
>+  cctz::time_point<cctz::seconds> sec;
>+  cctz::detail::femtoseconds fem;
>+};
>+
>+inline cctz::time_point<cctz::seconds> unix_epoch() {
>+  return std::chrono::time_point_cast<cctz::seconds>(
>+      std::chrono::system_clock::from_time_t(0));
>+}
>+
>+// Splits a Time into seconds and femtoseconds, which can be used with CCTZ.
>+// Requires that 't' is finite. See duration.cc for details about rep_hi and
>+// rep_lo.
>+cctz_parts Split(absl::Time t) {
>+  const auto d = time_internal::ToUnixDuration(t);
>+  const int64_t rep_hi = time_internal::GetRepHi(d);
>+  const int64_t rep_lo = time_internal::GetRepLo(d);
>+  const auto sec = unix_epoch() + cctz::seconds(rep_hi);
>+  const auto fem = cctz::detail::femtoseconds(rep_lo * (1000 * 1000 / 4));
>+  return {sec, fem};
>+}
>+
>+// Joins the given seconds and femtoseconds into a Time. See duration.cc for
>+// details about rep_hi and rep_lo.
>+absl::Time Join(const cctz_parts& parts) {
>+  const int64_t rep_hi = (parts.sec - unix_epoch()).count();
>+  const uint32_t rep_lo = parts.fem.count() / (1000 * 1000 / 4);
>+  const auto d = time_internal::MakeDuration(rep_hi, rep_lo);
>+  return time_internal::FromUnixDuration(d);
>+}
>+
>+}  // namespace
>+
>+std::string FormatTime(const std::string& format, absl::Time t, absl::TimeZone tz) {
>+  if (t == absl::InfiniteFuture()) return kInfiniteFutureStr;
>+  if (t == absl::InfinitePast()) return kInfinitePastStr;
>+  const auto parts = Split(t);
>+  return cctz::detail::format(format, parts.sec, parts.fem,
>+                              cctz::time_zone(tz));
>+}
>+
>+std::string FormatTime(absl::Time t, absl::TimeZone tz) {
>+  return FormatTime(RFC3339_full, t, tz);
>+}
>+
>+std::string FormatTime(absl::Time t) {
>+  return absl::FormatTime(RFC3339_full, t, absl::LocalTimeZone());
>+}
>+
>+bool ParseTime(const std::string& format, const std::string& input, absl::Time* time,
>+               std::string* err) {
>+  return absl::ParseTime(format, input, absl::UTCTimeZone(), time, err);
>+}
>+
>+// If the input std::string does not contain an explicit UTC offset, interpret
>+// the fields with respect to the given TimeZone.
>+bool ParseTime(const std::string& format, const std::string& input, absl::TimeZone tz,
>+               absl::Time* time, std::string* err) {
>+  const char* data = input.c_str();
>+  while (std::isspace(*data)) ++data;
>+
>+  size_t inf_size = strlen(kInfiniteFutureStr);
>+  if (strncmp(data, kInfiniteFutureStr, inf_size) == 0) {
>+    const char* new_data = data + inf_size;
>+    while (std::isspace(*new_data)) ++new_data;
>+    if (*new_data == '\0') {
>+      *time = InfiniteFuture();
>+      return true;
>+    }
>+  }
>+
>+  inf_size = strlen(kInfinitePastStr);
>+  if (strncmp(data, kInfinitePastStr, inf_size) == 0) {
>+    const char* new_data = data + inf_size;
>+    while (std::isspace(*new_data)) ++new_data;
>+    if (*new_data == '\0') {
>+      *time = InfinitePast();
>+      return true;
>+    }
>+  }
>+
>+  std::string error;
>+  cctz_parts parts;
>+  const bool b = cctz::detail::parse(format, input, cctz::time_zone(tz),
>+                                     &parts.sec, &parts.fem, &error);
>+  if (b) {
>+    *time = Join(parts);
>+  } else if (err != nullptr) {
>+    *err = error;
>+  }
>+  return b;
>+}
>+
>+// Functions required to support absl::Time flags.
>+bool ParseFlag(const std::string& text, absl::Time* t, std::string* error) {
>+  return absl::ParseTime(RFC3339_full, text, absl::UTCTimeZone(), t, error);
>+}
>+
>+std::string UnparseFlag(absl::Time t) {
>+  return absl::FormatTime(RFC3339_full, t, absl::UTCTimeZone());
>+}
>+
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/format_benchmark.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/format_benchmark.cc
>new file mode 100644
>index 00000000000..ee53d71c685
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/format_benchmark.cc
>@@ -0,0 +1,63 @@
>+// Copyright 2018 The Abseil Authors.
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include <cstddef>
>+#include <string>
>+
>+#include "absl/time/internal/test_util.h"
>+#include "absl/time/time.h"
>+#include "benchmark/benchmark.h"
>+
>+namespace {
>+
>+namespace {
>+const char* const kFormats[] = {
>+    absl::RFC1123_full,     // 0
>+    absl::RFC1123_no_wday,  // 1
>+    absl::RFC3339_full,     // 2
>+    absl::RFC3339_sec,      // 3
>+    "%Y-%m-%dT%H:%M:%S",    // 4
>+    "%Y-%m-%d",             // 5
>+};
>+const int kNumFormats = sizeof(kFormats) / sizeof(kFormats[0]);
>+}  // namespace
>+
>+void BM_Format_FormatTime(benchmark::State& state) {
>+  const std::string fmt = kFormats[state.range(0)];
>+  state.SetLabel(fmt);
>+  const absl::TimeZone lax =
>+      absl::time_internal::LoadTimeZone("America/Los_Angeles");
>+  const absl::Time t =
>+      absl::FromDateTime(1977, 6, 28, 9, 8, 7, lax) + absl::Nanoseconds(1);
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(absl::FormatTime(fmt, t, lax).length());
>+  }
>+}
>+BENCHMARK(BM_Format_FormatTime)->DenseRange(0, kNumFormats - 1);
>+
>+void BM_Format_ParseTime(benchmark::State& state) {
>+  const std::string fmt = kFormats[state.range(0)];
>+  state.SetLabel(fmt);
>+  const absl::TimeZone lax =
>+      absl::time_internal::LoadTimeZone("America/Los_Angeles");
>+  absl::Time t =
>+      absl::FromDateTime(1977, 6, 28, 9, 8, 7, lax) + absl::Nanoseconds(1);
>+  const std::string when = absl::FormatTime(fmt, t, lax);
>+  std::string err;
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(absl::ParseTime(fmt, when, lax, &t, &err));
>+  }
>+}
>+BENCHMARK(BM_Format_ParseTime)->DenseRange(0, kNumFormats - 1);
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/format_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/format_test.cc
>new file mode 100644
>index 00000000000..7c84c33f1eb
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/format_test.cc
>@@ -0,0 +1,434 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include <cstdint>
>+#include <limits>
>+#include <string>
>+
>+#include "gmock/gmock.h"
>+#include "gtest/gtest.h"
>+#include "absl/time/internal/test_util.h"
>+#include "absl/time/time.h"
>+
>+using testing::HasSubstr;
>+
>+namespace {
>+
>+// A helper that tests the given format specifier by itself, and with leading
>+// and trailing characters.  For example: TestFormatSpecifier(t, "%a", "Thu").
>+void TestFormatSpecifier(absl::Time t, absl::TimeZone tz, const std::string& fmt,
>+                         const std::string& ans) {
>+  EXPECT_EQ(ans, absl::FormatTime(fmt, t, tz));
>+  EXPECT_EQ("xxx " + ans, absl::FormatTime("xxx " + fmt, t, tz));
>+  EXPECT_EQ(ans + " yyy", absl::FormatTime(fmt + " yyy", t, tz));
>+  EXPECT_EQ("xxx " + ans + " yyy",
>+            absl::FormatTime("xxx " + fmt + " yyy", t, tz));
>+}
>+
>+//
>+// Testing FormatTime()
>+//
>+
>+TEST(FormatTime, Basics) {
>+  absl::TimeZone tz = absl::UTCTimeZone();
>+  absl::Time t = absl::FromTimeT(0);
>+
>+  // Starts with a couple basic edge cases.
>+  EXPECT_EQ("", absl::FormatTime("", t, tz));
>+  EXPECT_EQ(" ", absl::FormatTime(" ", t, tz));
>+  EXPECT_EQ("  ", absl::FormatTime("  ", t, tz));
>+  EXPECT_EQ("xxx", absl::FormatTime("xxx", t, tz));
>+  std::string big(128, 'x');
>+  EXPECT_EQ(big, absl::FormatTime(big, t, tz));
>+  // Cause the 1024-byte buffer to grow.
>+  std::string bigger(100000, 'x');
>+  EXPECT_EQ(bigger, absl::FormatTime(bigger, t, tz));
>+
>+  t += absl::Hours(13) + absl::Minutes(4) + absl::Seconds(5);
>+  t += absl::Milliseconds(6) + absl::Microseconds(7) + absl::Nanoseconds(8);
>+  EXPECT_EQ("1970-01-01", absl::FormatTime("%Y-%m-%d", t, tz));
>+  EXPECT_EQ("13:04:05", absl::FormatTime("%H:%M:%S", t, tz));
>+  EXPECT_EQ("13:04:05.006", absl::FormatTime("%H:%M:%E3S", t, tz));
>+  EXPECT_EQ("13:04:05.006007", absl::FormatTime("%H:%M:%E6S", t, tz));
>+  EXPECT_EQ("13:04:05.006007008", absl::FormatTime("%H:%M:%E9S", t, tz));
>+}
>+
>+TEST(FormatTime, LocaleSpecific) {
>+  const absl::TimeZone tz = absl::UTCTimeZone();
>+  absl::Time t = absl::FromTimeT(0);
>+
>+  TestFormatSpecifier(t, tz, "%a", "Thu");
>+  TestFormatSpecifier(t, tz, "%A", "Thursday");
>+  TestFormatSpecifier(t, tz, "%b", "Jan");
>+  TestFormatSpecifier(t, tz, "%B", "January");
>+
>+  // %c should at least produce the numeric year and time-of-day.
>+  const std::string s =
>+      absl::FormatTime("%c", absl::FromTimeT(0), absl::UTCTimeZone());
>+  EXPECT_THAT(s, HasSubstr("1970"));
>+  EXPECT_THAT(s, HasSubstr("00:00:00"));
>+
>+  TestFormatSpecifier(t, tz, "%p", "AM");
>+  TestFormatSpecifier(t, tz, "%x", "01/01/70");
>+  TestFormatSpecifier(t, tz, "%X", "00:00:00");
>+}
>+
>+TEST(FormatTime, ExtendedSeconds) {
>+  const absl::TimeZone tz = absl::UTCTimeZone();
>+
>+  // No subseconds.
>+  absl::Time t = absl::FromTimeT(0) + absl::Seconds(5);
>+  EXPECT_EQ("05", absl::FormatTime("%E*S", t, tz));
>+  EXPECT_EQ("05.000000000000000", absl::FormatTime("%E15S", t, tz));
>+
>+  // With subseconds.
>+  t += absl::Milliseconds(6) + absl::Microseconds(7) + absl::Nanoseconds(8);
>+  EXPECT_EQ("05.006007008", absl::FormatTime("%E*S", t, tz));
>+  EXPECT_EQ("05", absl::FormatTime("%E0S", t, tz));
>+  EXPECT_EQ("05.006007008000000", absl::FormatTime("%E15S", t, tz));
>+
>+  // Times before the Unix epoch.
>+  t = absl::FromUnixMicros(-1);
>+  EXPECT_EQ("1969-12-31 23:59:59.999999",
>+            absl::FormatTime("%Y-%m-%d %H:%M:%E*S", t, tz));
>+
>+  // Here is a "%E*S" case we got wrong for a while.  While the first
>+  // instant below is correctly rendered as "...:07.333304", the second
>+  // one used to appear as "...:07.33330499999999999".
>+  t = absl::FromUnixMicros(1395024427333304);
>+  EXPECT_EQ("2014-03-17 02:47:07.333304",
>+            absl::FormatTime("%Y-%m-%d %H:%M:%E*S", t, tz));
>+  t += absl::Microseconds(1);
>+  EXPECT_EQ("2014-03-17 02:47:07.333305",
>+            absl::FormatTime("%Y-%m-%d %H:%M:%E*S", t, tz));
>+}
>+
>+TEST(FormatTime, RFC1123FormatPadsYear) {  // locale specific
>+  absl::TimeZone tz = absl::UTCTimeZone();
>+
>+  // A year of 77 should be padded to 0077.
>+  absl::Time t = absl::FromDateTime(77, 6, 28, 9, 8, 7, tz);
>+  EXPECT_EQ("Mon, 28 Jun 0077 09:08:07 +0000",
>+            absl::FormatTime(absl::RFC1123_full, t, tz));
>+  EXPECT_EQ("28 Jun 0077 09:08:07 +0000",
>+            absl::FormatTime(absl::RFC1123_no_wday, t, tz));
>+}
>+
>+TEST(FormatTime, InfiniteTime) {
>+  absl::TimeZone tz = absl::time_internal::LoadTimeZone("America/Los_Angeles");
>+
>+  // The format and timezone are ignored.
>+  EXPECT_EQ("infinite-future",
>+            absl::FormatTime("%H:%M blah", absl::InfiniteFuture(), tz));
>+  EXPECT_EQ("infinite-past",
>+            absl::FormatTime("%H:%M blah", absl::InfinitePast(), tz));
>+}
>+
>+//
>+// Testing ParseTime()
>+//
>+
>+TEST(ParseTime, Basics) {
>+  absl::Time t = absl::FromTimeT(1234567890);
>+  std::string err;
>+
>+  // Simple edge cases.
>+  EXPECT_TRUE(absl::ParseTime("", "", &t, &err)) << err;
>+  EXPECT_EQ(absl::UnixEpoch(), t);  // everything defaulted
>+  EXPECT_TRUE(absl::ParseTime(" ", " ", &t, &err)) << err;
>+  EXPECT_TRUE(absl::ParseTime("  ", "  ", &t, &err)) << err;
>+  EXPECT_TRUE(absl::ParseTime("x", "x", &t, &err)) << err;
>+  EXPECT_TRUE(absl::ParseTime("xxx", "xxx", &t, &err)) << err;
>+
>+  EXPECT_TRUE(absl::ParseTime("%Y-%m-%d %H:%M:%S %z",
>+                              "2013-06-28 19:08:09 -0800", &t, &err))
>+      << err;
>+  absl::Time::Breakdown bd = t.In(absl::FixedTimeZone(-8 * 60 * 60));
>+  ABSL_INTERNAL_EXPECT_TIME(bd, 2013, 6, 28, 19, 8, 9, -8 * 60 * 60, false);
>+  EXPECT_EQ(absl::ZeroDuration(), bd.subsecond);
>+}
>+
>+TEST(ParseTime, NullErrorString) {
>+  absl::Time t;
>+  EXPECT_FALSE(absl::ParseTime("%Q", "invalid format", &t, nullptr));
>+  EXPECT_FALSE(absl::ParseTime("%H", "12 trailing data", &t, nullptr));
>+  EXPECT_FALSE(
>+      absl::ParseTime("%H out of range", "42 out of range", &t, nullptr));
>+}
>+
>+TEST(ParseTime, WithTimeZone) {
>+  const absl::TimeZone tz =
>+      absl::time_internal::LoadTimeZone("America/Los_Angeles");
>+  absl::Time t;
>+  std::string e;
>+
>+  // We can parse a std::string without a UTC offset if we supply a timezone.
>+  EXPECT_TRUE(
>+      absl::ParseTime("%Y-%m-%d %H:%M:%S", "2013-06-28 19:08:09", tz, &t, &e))
>+      << e;
>+  absl::Time::Breakdown bd = t.In(tz);
>+  ABSL_INTERNAL_EXPECT_TIME(bd, 2013, 6, 28, 19, 8, 9, -7 * 60 * 60, true);
>+  EXPECT_EQ(absl::ZeroDuration(), bd.subsecond);
>+
>+  // But the timezone is ignored when a UTC offset is present.
>+  EXPECT_TRUE(absl::ParseTime("%Y-%m-%d %H:%M:%S %z",
>+                              "2013-06-28 19:08:09 +0800", tz, &t, &e))
>+      << e;
>+  bd = t.In(absl::FixedTimeZone(8 * 60 * 60));
>+  ABSL_INTERNAL_EXPECT_TIME(bd, 2013, 6, 28, 19, 8, 9, 8 * 60 * 60, false);
>+  EXPECT_EQ(absl::ZeroDuration(), bd.subsecond);
>+}
>+
>+TEST(ParseTime, ErrorCases) {
>+  absl::Time t = absl::FromTimeT(0);
>+  std::string err;
>+
>+  EXPECT_FALSE(absl::ParseTime("%S", "123", &t, &err)) << err;
>+  EXPECT_THAT(err, HasSubstr("Illegal trailing data"));
>+
>+  // Can't parse an illegal format specifier.
>+  err.clear();
>+  EXPECT_FALSE(absl::ParseTime("%Q", "x", &t, &err)) << err;
>+  // Exact contents of "err" are platform-dependent because of
>+  // differences in the strptime implementation between OSX and Linux.
>+  EXPECT_FALSE(err.empty());
>+
>+  // Fails because of trailing, unparsed data "blah".
>+  EXPECT_FALSE(absl::ParseTime("%m-%d", "2-3 blah", &t, &err)) << err;
>+  EXPECT_THAT(err, HasSubstr("Illegal trailing data"));
>+
>+  // Feb 31 requires normalization.
>+  EXPECT_FALSE(absl::ParseTime("%m-%d", "2-31", &t, &err)) << err;
>+  EXPECT_THAT(err, HasSubstr("Out-of-range"));
>+
>+  // Check that we cannot have spaces in UTC offsets.
>+  EXPECT_TRUE(absl::ParseTime("%z", "-0203", &t, &err)) << err;
>+  EXPECT_FALSE(absl::ParseTime("%z", "- 2 3", &t, &err)) << err;
>+  EXPECT_THAT(err, HasSubstr("Failed to parse"));
>+  EXPECT_TRUE(absl::ParseTime("%Ez", "-02:03", &t, &err)) << err;
>+  EXPECT_FALSE(absl::ParseTime("%Ez", "- 2: 3", &t, &err)) << err;
>+  EXPECT_THAT(err, HasSubstr("Failed to parse"));
>+
>+  // Check that we reject other malformed UTC offsets.
>+  EXPECT_FALSE(absl::ParseTime("%Ez", "+-08:00", &t, &err)) << err;
>+  EXPECT_THAT(err, HasSubstr("Failed to parse"));
>+  EXPECT_FALSE(absl::ParseTime("%Ez", "-+08:00", &t, &err)) << err;
>+  EXPECT_THAT(err, HasSubstr("Failed to parse"));
>+
>+  // Check that we do not accept "-0" in fields that allow zero.
>+  EXPECT_FALSE(absl::ParseTime("%Y", "-0", &t, &err)) << err;
>+  EXPECT_THAT(err, HasSubstr("Failed to parse"));
>+  EXPECT_FALSE(absl::ParseTime("%E4Y", "-0", &t, &err)) << err;
>+  EXPECT_THAT(err, HasSubstr("Failed to parse"));
>+  EXPECT_FALSE(absl::ParseTime("%H", "-0", &t, &err)) << err;
>+  EXPECT_THAT(err, HasSubstr("Failed to parse"));
>+  EXPECT_FALSE(absl::ParseTime("%M", "-0", &t, &err)) << err;
>+  EXPECT_THAT(err, HasSubstr("Failed to parse"));
>+  EXPECT_FALSE(absl::ParseTime("%S", "-0", &t, &err)) << err;
>+  EXPECT_THAT(err, HasSubstr("Failed to parse"));
>+  EXPECT_FALSE(absl::ParseTime("%z", "+-000", &t, &err)) << err;
>+  EXPECT_THAT(err, HasSubstr("Failed to parse"));
>+  EXPECT_FALSE(absl::ParseTime("%Ez", "+-0:00", &t, &err)) << err;
>+  EXPECT_THAT(err, HasSubstr("Failed to parse"));
>+  EXPECT_FALSE(absl::ParseTime("%z", "-00-0", &t, &err)) << err;
>+  EXPECT_THAT(err, HasSubstr("Illegal trailing data"));
>+  EXPECT_FALSE(absl::ParseTime("%Ez", "-00:-0", &t, &err)) << err;
>+  EXPECT_THAT(err, HasSubstr("Illegal trailing data"));
>+}
>+
>+TEST(ParseTime, ExtendedSeconds) {
>+  std::string err;
>+  absl::Time t;
>+
>+  // Here is a "%E*S" case we got wrong for a while.  The fractional
>+  // part of the first instant is less than 2^31 and was correctly
>+  // parsed, while the second (and any subsecond field >=2^31) failed.
>+  t = absl::UnixEpoch();
>+  EXPECT_TRUE(absl::ParseTime("%E*S", "0.2147483647", &t, &err)) << err;
>+  EXPECT_EQ(absl::UnixEpoch() + absl::Nanoseconds(214748364) +
>+                absl::Nanoseconds(1) / 2,
>+            t);
>+  t = absl::UnixEpoch();
>+  EXPECT_TRUE(absl::ParseTime("%E*S", "0.2147483648", &t, &err)) << err;
>+  EXPECT_EQ(absl::UnixEpoch() + absl::Nanoseconds(214748364) +
>+                absl::Nanoseconds(3) / 4,
>+            t);
>+
>+  // We should also be able to specify long strings of digits far
>+  // beyond the current resolution and have them convert the same way.
>+  t = absl::UnixEpoch();
>+  EXPECT_TRUE(absl::ParseTime(
>+      "%E*S", "0.214748364801234567890123456789012345678901234567890123456789",
>+      &t, &err))
>+      << err;
>+  EXPECT_EQ(absl::UnixEpoch() + absl::Nanoseconds(214748364) +
>+                absl::Nanoseconds(3) / 4,
>+            t);
>+}
>+
>+TEST(ParseTime, ExtendedOffsetErrors) {
>+  std::string err;
>+  absl::Time t;
>+
>+  // %z against +-HHMM.
>+  EXPECT_FALSE(absl::ParseTime("%z", "-123", &t, &err)) << err;
>+  EXPECT_THAT(err, HasSubstr("Illegal trailing data"));
>+
>+  // %z against +-HH.
>+  EXPECT_FALSE(absl::ParseTime("%z", "-1", &t, &err)) << err;
>+  EXPECT_THAT(err, HasSubstr("Failed to parse"));
>+
>+  // %Ez against +-HH:MM.
>+  EXPECT_FALSE(absl::ParseTime("%Ez", "-12:3", &t, &err)) << err;
>+  EXPECT_THAT(err, HasSubstr("Illegal trailing data"));
>+
>+  // %Ez against +-HHMM.
>+  EXPECT_FALSE(absl::ParseTime("%Ez", "-123", &t, &err)) << err;
>+  EXPECT_THAT(err, HasSubstr("Illegal trailing data"));
>+
>+  // %Ez against +-HH.
>+  EXPECT_FALSE(absl::ParseTime("%Ez", "-1", &t, &err)) << err;
>+  EXPECT_THAT(err, HasSubstr("Failed to parse"));
>+}
>+
>+TEST(ParseTime, InfiniteTime) {
>+  absl::Time t;
>+  std::string err;
>+  EXPECT_TRUE(absl::ParseTime("%H:%M blah", "infinite-future", &t, &err));
>+  EXPECT_EQ(absl::InfiniteFuture(), t);
>+
>+  // Surrounding whitespace.
>+  EXPECT_TRUE(absl::ParseTime("%H:%M blah", "  infinite-future", &t, &err));
>+  EXPECT_EQ(absl::InfiniteFuture(), t);
>+  EXPECT_TRUE(absl::ParseTime("%H:%M blah", "infinite-future  ", &t, &err));
>+  EXPECT_EQ(absl::InfiniteFuture(), t);
>+  EXPECT_TRUE(absl::ParseTime("%H:%M blah", "  infinite-future  ", &t, &err));
>+  EXPECT_EQ(absl::InfiniteFuture(), t);
>+
>+  EXPECT_TRUE(absl::ParseTime("%H:%M blah", "infinite-past", &t, &err));
>+  EXPECT_EQ(absl::InfinitePast(), t);
>+
>+  // Surrounding whitespace.
>+  EXPECT_TRUE(absl::ParseTime("%H:%M blah", "  infinite-past", &t, &err));
>+  EXPECT_EQ(absl::InfinitePast(), t);
>+  EXPECT_TRUE(absl::ParseTime("%H:%M blah", "infinite-past  ", &t, &err));
>+  EXPECT_EQ(absl::InfinitePast(), t);
>+  EXPECT_TRUE(absl::ParseTime("%H:%M blah", "  infinite-past  ", &t, &err));
>+  EXPECT_EQ(absl::InfinitePast(), t);
>+
>+  // "infinite-future" as literal std::string
>+  absl::TimeZone tz = absl::UTCTimeZone();
>+  EXPECT_TRUE(absl::ParseTime("infinite-future %H:%M", "infinite-future 03:04",
>+                              &t, &err));
>+  EXPECT_NE(absl::InfiniteFuture(), t);
>+  EXPECT_EQ(3, t.In(tz).hour);
>+  EXPECT_EQ(4, t.In(tz).minute);
>+
>+  // "infinite-past" as literal std::string
>+  EXPECT_TRUE(
>+      absl::ParseTime("infinite-past %H:%M", "infinite-past 03:04", &t, &err));
>+  EXPECT_NE(absl::InfinitePast(), t);
>+  EXPECT_EQ(3, t.In(tz).hour);
>+  EXPECT_EQ(4, t.In(tz).minute);
>+
>+  // The input doesn't match the format.
>+  EXPECT_FALSE(absl::ParseTime("infinite-future %H:%M", "03:04", &t, &err));
>+  EXPECT_FALSE(absl::ParseTime("infinite-past %H:%M", "03:04", &t, &err));
>+}
>+
>+TEST(ParseTime, FailsOnUnrepresentableTime) {
>+  const absl::TimeZone utc = absl::UTCTimeZone();
>+  absl::Time t;
>+  EXPECT_FALSE(
>+      absl::ParseTime("%Y-%m-%d", "-292277022657-01-27", utc, &t, nullptr));
>+  EXPECT_TRUE(
>+      absl::ParseTime("%Y-%m-%d", "-292277022657-01-28", utc, &t, nullptr));
>+  EXPECT_TRUE(
>+      absl::ParseTime("%Y-%m-%d", "292277026596-12-04", utc, &t, nullptr));
>+  EXPECT_FALSE(
>+      absl::ParseTime("%Y-%m-%d", "292277026596-12-05", utc, &t, nullptr));
>+}
>+
>+//
>+// Roundtrip test for FormatTime()/ParseTime().
>+//
>+
>+TEST(FormatParse, RoundTrip) {
>+  const absl::TimeZone gst =
>+      absl::time_internal::LoadTimeZone("America/Los_Angeles");
>+  const absl::Time in = absl::FromDateTime(1977, 6, 28, 9, 8, 7, gst);
>+  const absl::Duration subseconds = absl::Nanoseconds(654321);
>+  std::string err;
>+
>+  // RFC3339, which renders subseconds.
>+  {
>+    absl::Time out;
>+    const std::string s = absl::FormatTime(absl::RFC3339_full, in + subseconds, gst);
>+    EXPECT_TRUE(absl::ParseTime(absl::RFC3339_full, s, &out, &err))
>+        << s << ": " << err;
>+    EXPECT_EQ(in + subseconds, out);  // RFC3339_full includes %Ez
>+  }
>+
>+  // RFC1123, which only does whole seconds.
>+  {
>+    absl::Time out;
>+    const std::string s = absl::FormatTime(absl::RFC1123_full, in, gst);
>+    EXPECT_TRUE(absl::ParseTime(absl::RFC1123_full, s, &out, &err))
>+        << s << ": " << err;
>+    EXPECT_EQ(in, out);  // RFC1123_full includes %z
>+  }
>+
>+  // `absl::FormatTime()` falls back to strftime() for "%c", which appears to
>+  // work. On Windows, `absl::ParseTime()` falls back to std::get_time() which
>+  // appears to fail on "%c" (or at least on the "%c" text produced by
>+  // `strftime()`). This makes it fail the round-trip test.
>+#ifndef _MSC_VER
>+  // Even though we don't know what %c will produce, it should roundtrip,
>+  // but only in the 0-offset timezone.
>+  {
>+    absl::Time out;
>+    const std::string s = absl::FormatTime("%c", in, absl::UTCTimeZone());
>+    EXPECT_TRUE(absl::ParseTime("%c", s, &out, &err)) << s << ": " << err;
>+    EXPECT_EQ(in, out);
>+  }
>+#endif  // _MSC_VER
>+}
>+
>+TEST(FormatParse, RoundTripDistantFuture) {
>+  const absl::TimeZone tz = absl::UTCTimeZone();
>+  const absl::Time in =
>+      absl::FromUnixSeconds(std::numeric_limits<int64_t>::max());
>+  std::string err;
>+
>+  absl::Time out;
>+  const std::string s = absl::FormatTime(absl::RFC3339_full, in, tz);
>+  EXPECT_TRUE(absl::ParseTime(absl::RFC3339_full, s, &out, &err))
>+      << s << ": " << err;
>+  EXPECT_EQ(in, out);
>+}
>+
>+TEST(FormatParse, RoundTripDistantPast) {
>+  const absl::TimeZone tz = absl::UTCTimeZone();
>+  const absl::Time in =
>+      absl::FromUnixSeconds(std::numeric_limits<int64_t>::min());
>+  std::string err;
>+
>+  absl::Time out;
>+  const std::string s = absl::FormatTime(absl::RFC3339_full, in, tz);
>+  EXPECT_TRUE(absl::ParseTime(absl::RFC3339_full, s, &out, &err))
>+      << s << ": " << err;
>+  EXPECT_EQ(in, out);
>+}
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/BUILD.bazel b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/BUILD.bazel
>new file mode 100644
>index 00000000000..9f1ba21cf7e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/BUILD.bazel
>@@ -0,0 +1,140 @@
>+# Copyright 2016 Google Inc. All Rights Reserved.
>+#
>+# Licensed under the Apache License, Version 2.0 (the "License");
>+# you may not use this file except in compliance with the License.
>+# You may obtain a copy of the License at
>+#
>+#   http://www.apache.org/licenses/LICENSE-2.0
>+#
>+#   Unless required by applicable law or agreed to in writing, software
>+#   distributed under the License is distributed on an "AS IS" BASIS,
>+#   WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+#   See the License for the specific language governing permissions and
>+#   limitations under the License.
>+
>+licenses(["notice"])  # Apache License
>+
>+### libraries
>+
>+cc_library(
>+    name = "includes",
>+    textual_hdrs = [
>+        "include/cctz/civil_time.h",
>+        "include/cctz/civil_time_detail.h",
>+        "include/cctz/time_zone.h",
>+    ],
>+    visibility = ["//absl/time:__pkg__"],
>+)
>+
>+cc_library(
>+    name = "civil_time",
>+    srcs = ["src/civil_time_detail.cc"],
>+    hdrs = [
>+        "include/cctz/civil_time.h",
>+    ],
>+    textual_hdrs = ["include/cctz/civil_time_detail.h"],
>+    visibility = ["//visibility:public"],
>+)
>+
>+cc_library(
>+    name = "time_zone",
>+    srcs = [
>+        "src/time_zone_fixed.cc",
>+        "src/time_zone_fixed.h",
>+        "src/time_zone_format.cc",
>+        "src/time_zone_if.cc",
>+        "src/time_zone_if.h",
>+        "src/time_zone_impl.cc",
>+        "src/time_zone_impl.h",
>+        "src/time_zone_info.cc",
>+        "src/time_zone_info.h",
>+        "src/time_zone_libc.cc",
>+        "src/time_zone_libc.h",
>+        "src/time_zone_lookup.cc",
>+        "src/time_zone_posix.cc",
>+        "src/time_zone_posix.h",
>+        "src/tzfile.h",
>+        "src/zone_info_source.cc",
>+    ],
>+    hdrs = [
>+        "include/cctz/time_zone.h",
>+        "include/cctz/zone_info_source.h",
>+    ],
>+    visibility = ["//visibility:public"],
>+    deps = [":civil_time"],
>+)
>+
>+### tests
>+
>+cc_test(
>+    name = "civil_time_test",
>+    size = "small",
>+    srcs = ["src/civil_time_test.cc"],
>+    deps = [
>+        ":civil_time",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "time_zone_format_test",
>+    size = "small",
>+    srcs = ["src/time_zone_format_test.cc"],
>+    data = [":zoneinfo"],
>+    tags = [
>+        "no_test_android_arm",
>+        "no_test_android_arm64",
>+        "no_test_android_x86",
>+    ],
>+    deps = [
>+        ":civil_time",
>+        ":time_zone",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "time_zone_lookup_test",
>+    size = "small",
>+    srcs = ["src/time_zone_lookup_test.cc"],
>+    data = [":zoneinfo"],
>+    tags = [
>+        "no_test_android_arm",
>+        "no_test_android_arm64",
>+        "no_test_android_x86",
>+    ],
>+    deps = [
>+        ":civil_time",
>+        ":time_zone",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+### benchmarks
>+
>+cc_test(
>+    name = "cctz_benchmark",
>+    srcs = [
>+        "src/cctz_benchmark.cc",
>+        "src/time_zone_if.h",
>+        "src/time_zone_impl.h",
>+        "src/time_zone_info.h",
>+        "src/tzfile.h",
>+    ],
>+    linkstatic = 1,
>+    tags = ["benchmark"],
>+    deps = [
>+        ":civil_time",
>+        ":time_zone",
>+        "@com_github_google_benchmark//:benchmark_main",
>+    ],
>+)
>+
>+### examples
>+
>+### binaries
>+
>+filegroup(
>+    name = "zoneinfo",
>+    srcs = glob(["testdata/zoneinfo/**"]),
>+)
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/BUILD.gn
>new file mode 100644
>index 00000000000..86417e7ce3c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/BUILD.gn
>@@ -0,0 +1,72 @@
>+# Copyright 2018 The Chromium Authors. All rights reserved.
>+# Use of this source code is governed by a BSD-style license that can be
>+# found in the LICENSE file.
>+
>+import("//build_overrides/build.gni")
>+
>+if (build_with_chromium) {
>+  visibility = [
>+    "//third_party/webrtc/*",
>+    "//third_party/abseil-cpp/*",
>+    "//third_party/googletest:gtest",
>+  ]
>+} else {
>+  visibility = [ "*" ]
>+}
>+
>+source_set("includes") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [ "//build/config/compiler:no_chromium_code" ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  public = [
>+    "include/cctz/civil_time.h",
>+    "include/cctz/civil_time_detail.h",
>+    "include/cctz/time_zone.h",
>+  ]
>+  visibility = []
>+  visibility += [ "../time:*" ]
>+}
>+
>+source_set("civil_time") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [ "//build/config/compiler:no_chromium_code" ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  sources = [
>+    "src/civil_time_detail.cc",
>+  ]
>+  public = [
>+    "include/cctz/civil_time.h",
>+    "include/cctz/civil_time_detail.h",
>+  ]
>+}
>+
>+source_set("time_zone") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [ "//build/config/compiler:no_chromium_code" ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  sources = [
>+    "src/time_zone_fixed.cc",
>+    "src/time_zone_fixed.h",
>+    "src/time_zone_format.cc",
>+    "src/time_zone_if.cc",
>+    "src/time_zone_if.h",
>+    "src/time_zone_impl.cc",
>+    "src/time_zone_impl.h",
>+    "src/time_zone_info.cc",
>+    "src/time_zone_info.h",
>+    "src/time_zone_libc.cc",
>+    "src/time_zone_libc.h",
>+    "src/time_zone_lookup.cc",
>+    "src/time_zone_posix.cc",
>+    "src/time_zone_posix.h",
>+    "src/tzfile.h",
>+    "src/zone_info_source.cc",
>+  ]
>+  public = [
>+    "include/cctz/time_zone.h",
>+    "include/cctz/zone_info_source.h",
>+  ]
>+  deps = [
>+    ":civil_time",
>+  ]
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/include/cctz/civil_time.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/include/cctz/civil_time.h
>new file mode 100644
>index 00000000000..898222b4c7a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/include/cctz/civil_time.h
>@@ -0,0 +1,329 @@
>+// Copyright 2016 Google Inc. All Rights Reserved.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//   http://www.apache.org/licenses/LICENSE-2.0
>+//
>+//   Unless required by applicable law or agreed to in writing, software
>+//   distributed under the License is distributed on an "AS IS" BASIS,
>+//   WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+//   See the License for the specific language governing permissions and
>+//   limitations under the License.
>+
>+#ifndef ABSL_TIME_INTERNAL_CCTZ_CIVIL_TIME_H_
>+#define ABSL_TIME_INTERNAL_CCTZ_CIVIL_TIME_H_
>+
>+#include "absl/time/internal/cctz/include/cctz/civil_time_detail.h"
>+
>+namespace absl {
>+namespace time_internal {
>+namespace cctz {
>+
>+// The term "civil time" refers to the legally recognized human-scale time
>+// that is represented by the six fields YYYY-MM-DD hh:mm:ss. Modern-day civil
>+// time follows the Gregorian Calendar and is a time-zone-independent concept.
>+// A "date" is perhaps the most common example of a civil time (represented in
>+// this library as cctz::civil_day). This library provides six classes and a
>+// handful of functions that help with rounding, iterating, and arithmetic on
>+// civil times while avoiding complications like daylight-saving time (DST).
>+//
>+// The following six classes form the core of this civil-time library:
>+//
>+//   * civil_second
>+//   * civil_minute
>+//   * civil_hour
>+//   * civil_day
>+//   * civil_month
>+//   * civil_year
>+//
>+// Each class is a simple value type with the same interface for construction
>+// and the same six accessors for each of the civil fields (year, month, day,
>+// hour, minute, and second, aka YMDHMS). These classes differ only in their
>+// alignment, which is indicated by the type name and specifies the field on
>+// which arithmetic operates.
>+//
>+// Each class can be constructed by passing up to six optional integer
>+// arguments representing the YMDHMS fields (in that order) to the
>+// constructor. Omitted fields are assigned their minimum valid value. Hours,
>+// minutes, and seconds will be set to 0, month and day will be set to 1, and
>+// since there is no minimum valid year, it will be set to 1970. So, a
>+// default-constructed civil-time object will have YMDHMS fields representing
>+// "1970-01-01 00:00:00". Fields that are out-of-range are normalized (e.g.,
>+// October 32 -> November 1) so that all civil-time objects represent valid
>+// values.
>+//
>+// Each civil-time class is aligned to the civil-time field indicated in the
>+// class's name after normalization. Alignment is performed by setting all the
>+// inferior fields to their minimum valid value (as described above). The
>+// following are examples of how each of the six types would align the fields
>+// representing November 22, 2015 at 12:34:56 in the afternoon. (Note: the
>+// std::string format used here is not important; it's just a shorthand way of
>+// showing the six YMDHMS fields.)
>+//
>+//   civil_second  2015-11-22 12:34:56
>+//   civil_minute  2015-11-22 12:34:00
>+//   civil_hour    2015-11-22 12:00:00
>+//   civil_day     2015-11-22 00:00:00
>+//   civil_month   2015-11-01 00:00:00
>+//   civil_year    2015-01-01 00:00:00
>+//
>+// Each civil-time type performs arithmetic on the field to which it is
>+// aligned. This means that adding 1 to a civil_day increments the day field
>+// (normalizing as necessary), and subtracting 7 from a civil_month operates
>+// on the month field (normalizing as necessary). All arithmetic produces a
>+// valid civil time. Difference requires two similarly aligned civil-time
>+// objects and returns the scalar answer in units of the objects' alignment.
>+// For example, the difference between two civil_hour objects will give an
>+// answer in units of civil hours.
>+//
>+// In addition to the six civil-time types just described, there are
>+// a handful of helper functions and algorithms for performing common
>+// calculations. These are described below.
>+//
>+// Note: In C++14 and later, this library is usable in a constexpr context.
>+//
>+// CONSTRUCTION:
>+//
>+// Each of the civil-time types can be constructed in two ways: by directly
>+// passing to the constructor up to six (optional) integers representing the
>+// YMDHMS fields, or by copying the YMDHMS fields from a differently aligned
>+// civil-time type.
>+//
>+//   civil_day default_value;  // 1970-01-01 00:00:00
>+//
>+//   civil_day a(2015, 2, 3);           // 2015-02-03 00:00:00
>+//   civil_day b(2015, 2, 3, 4, 5, 6);  // 2015-02-03 00:00:00
>+//   civil_day c(2015);                 // 2015-01-01 00:00:00
>+//
>+//   civil_second ss(2015, 2, 3, 4, 5, 6);  // 2015-02-03 04:05:06
>+//   civil_minute mm(ss);                   // 2015-02-03 04:05:00
>+//   civil_hour hh(mm);                     // 2015-02-03 04:00:00
>+//   civil_day d(hh);                       // 2015-02-03 00:00:00
>+//   civil_month m(d);                      // 2015-02-01 00:00:00
>+//   civil_year y(m);                       // 2015-01-01 00:00:00
>+//
>+//   m = civil_month(y);     // 2015-01-01 00:00:00
>+//   d = civil_day(m);       // 2015-01-01 00:00:00
>+//   hh = civil_hour(d);     // 2015-01-01 00:00:00
>+//   mm = civil_minute(hh);  // 2015-01-01 00:00:00
>+//   ss = civil_second(mm);  // 2015-01-01 00:00:00
>+//
>+// ALIGNMENT CONVERSION:
>+//
>+// The alignment of a civil-time object cannot change, but the object may be
>+// used to construct a new object with a different alignment. This is referred
>+// to as "realigning". When realigning to a type with the same or more
>+// precision (e.g., civil_day -> civil_second), the conversion may be
>+// performed implicitly since no information is lost. However, if information
>+// could be discarded (e.g., civil_second -> civil_day), the conversion must
>+// be explicit at the call site.
>+//
>+//   void fun(const civil_day& day);
>+//
>+//   civil_second cs;
>+//   fun(cs);  // Won't compile because data may be discarded
>+//   fun(civil_day(cs));  // OK: explicit conversion
>+//
>+//   civil_day cd;
>+//   fun(cd);  // OK: no conversion needed
>+//
>+//   civil_month cm;
>+//   fun(cm);  // OK: implicit conversion to civil_day
>+//
>+// NORMALIZATION:
>+//
>+// Integer arguments passed to the constructor may be out-of-range, in which
>+// case they are normalized to produce a valid civil-time object. This enables
>+// natural arithmetic on constructor arguments without worrying about the
>+// field's range. Normalization guarantees that there are no invalid
>+// civil-time objects.
>+//
>+//   civil_day d(2016, 10, 32);  // Out-of-range day; normalized to 2016-11-01
>+//
>+// Note: If normalization is undesired, you can signal an error by comparing
>+// the constructor arguments to the normalized values returned by the YMDHMS
>+// properties.
>+//
>+// PROPERTIES:
>+//
>+// All civil-time types have accessors for all six of the civil-time fields:
>+// year, month, day, hour, minute, and second. Recall that fields inferior to
>+// the type's aligment will be set to their minimum valid value.
>+//
>+//   civil_day d(2015, 6, 28);
>+//   // d.year() == 2015
>+//   // d.month() == 6
>+//   // d.day() == 28
>+//   // d.hour() == 0
>+//   // d.minute() == 0
>+//   // d.second() == 0
>+//
>+// COMPARISON:
>+//
>+// Comparison always considers all six YMDHMS fields, regardless of the type's
>+// alignment. Comparison between differently aligned civil-time types is
>+// allowed.
>+//
>+//   civil_day feb_3(2015, 2, 3);  // 2015-02-03 00:00:00
>+//   civil_day mar_4(2015, 3, 4);  // 2015-03-04 00:00:00
>+//   // feb_3 < mar_4
>+//   // civil_year(feb_3) == civil_year(mar_4)
>+//
>+//   civil_second feb_3_noon(2015, 2, 3, 12, 0, 0);  // 2015-02-03 12:00:00
>+//   // feb_3 < feb_3_noon
>+//   // feb_3 == civil_day(feb_3_noon)
>+//
>+//   // Iterates all the days of February 2015.
>+//   for (civil_day d(2015, 2, 1); d < civil_month(2015, 3); ++d) {
>+//     // ...
>+//   }
>+//
>+// STREAMING:
>+//
>+// Each civil-time type may be sent to an output stream using operator<<().
>+// The output format follows the pattern "YYYY-MM-DDThh:mm:ss" where fields
>+// inferior to the type's alignment are omitted.
>+//
>+//   civil_second cs(2015, 2, 3, 4, 5, 6);
>+//   std::cout << cs << "\n";  // Outputs: 2015-02-03T04:05:06
>+//
>+//   civil_day cd(cs);
>+//   std::cout << cd << "\n";  // Outputs: 2015-02-03
>+//
>+//   civil_year cy(cs);
>+//   std::cout << cy << "\n";  // Outputs: 2015
>+//
>+// ARITHMETIC:
>+//
>+// Civil-time types support natural arithmetic operators such as addition,
>+// subtraction, and difference. Arithmetic operates on the civil-time field
>+// indicated in the type's name. Difference requires arguments with the same
>+// alignment and returns the answer in units of the alignment.
>+//
>+//   civil_day a(2015, 2, 3);
>+//   ++a;                         // 2015-02-04 00:00:00
>+//   --a;                         // 2015-02-03 00:00:00
>+//   civil_day b = a + 1;         // 2015-02-04 00:00:00
>+//   civil_day c = 1 + b;         // 2015-02-05 00:00:00
>+//   int n = c - a;               // n = 2 (civil days)
>+//   int m = c - civil_month(c);  // Won't compile: different types.
>+//
>+// EXAMPLE: Adding a month to January 31.
>+//
>+// One of the classic questions that arises when considering a civil-time
>+// library (or a date library or a date/time library) is this: "What happens
>+// when you add a month to January 31?" This is an interesting question
>+// because there could be a number of possible answers:
>+//
>+//   1. March 3 (or 2 if a leap year). This may make sense if the operation
>+//      wants the equivalent of February 31.
>+//   2. February 28 (or 29 if a leap year). This may make sense if the operation
>+//      wants the last day of January to go to the last day of February.
>+//   3. Error. The caller may get some error, an exception, an invalid date
>+//      object, or maybe false is returned. This may make sense because there is
>+//      no single unambiguously correct answer to the question.
>+//
>+// Practically speaking, any answer that is not what the programmer intended
>+// is the wrong answer.
>+//
>+// This civil-time library avoids the problem by making it impossible to ask
>+// ambiguous questions. All civil-time objects are aligned to a particular
>+// civil-field boundary (such as aligned to a year, month, day, hour, minute,
>+// or second), and arithmetic operates on the field to which the object is
>+// aligned. This means that in order to "add a month" the object must first be
>+// aligned to a month boundary, which is equivalent to the first day of that
>+// month.
>+//
>+// Of course, there are ways to compute an answer the question at hand using
>+// this civil-time library, but they require the programmer to be explicit
>+// about the answer they expect. To illustrate, let's see how to compute all
>+// three of the above possible answers to the question of "Jan 31 plus 1
>+// month":
>+//
>+//   const civil_day d(2015, 1, 31);
>+//
>+//   // Answer 1:
>+//   // Add 1 to the month field in the constructor, and rely on normalization.
>+//   const auto ans_normalized = civil_day(d.year(), d.month() + 1, d.day());
>+//   // ans_normalized == 2015-03-03 (aka Feb 31)
>+//
>+//   // Answer 2:
>+//   // Add 1 to month field, capping to the end of next month.
>+//   const auto next_month = civil_month(d) + 1;
>+//   const auto last_day_of_next_month = civil_day(next_month + 1) - 1;
>+//   const auto ans_capped = std::min(ans_normalized, last_day_of_next_month);
>+//   // ans_capped == 2015-02-28
>+//
>+//   // Answer 3:
>+//   // Signal an error if the normalized answer is not in next month.
>+//   if (civil_month(ans_normalized) != next_month) {
>+//     // error, month overflow
>+//   }
>+//
>+using civil_year = detail::civil_year;
>+using civil_month = detail::civil_month;
>+using civil_day = detail::civil_day;
>+using civil_hour = detail::civil_hour;
>+using civil_minute = detail::civil_minute;
>+using civil_second = detail::civil_second;
>+
>+// An enum class with members monday, tuesday, wednesday, thursday, friday,
>+// saturday, and sunday. These enum values may be sent to an output stream
>+// using operator<<(). The result is the full weekday name in English with a
>+// leading capital letter.
>+//
>+//   weekday wd = weekday::thursday;
>+//   std::cout << wd << "\n";  // Outputs: Thursday
>+//
>+using detail::weekday;
>+
>+// Returns the weekday for the given civil_day.
>+//
>+//   civil_day a(2015, 8, 13);
>+//   weekday wd = get_weekday(a);  // wd == weekday::thursday
>+//
>+using detail::get_weekday;
>+
>+// Returns the civil_day that strictly follows or precedes the given
>+// civil_day, and that falls on the given weekday.
>+//
>+// For example, given:
>+//
>+//     August 2015
>+// Su Mo Tu We Th Fr Sa
>+//                    1
>+//  2  3  4  5  6  7  8
>+//  9 10 11 12 13 14 15
>+// 16 17 18 19 20 21 22
>+// 23 24 25 26 27 28 29
>+// 30 31
>+//
>+//   civil_day a(2015, 8, 13);  // get_weekday(a) == weekday::thursday
>+//   civil_day b = next_weekday(a, weekday::thursday);  // b = 2015-08-20
>+//   civil_day c = prev_weekday(a, weekday::thursday);  // c = 2015-08-06
>+//
>+//   civil_day d = ...
>+//   // Gets the following Thursday if d is not already Thursday
>+//   civil_day thurs1 = prev_weekday(d, weekday::thursday) + 7;
>+//   // Gets the previous Thursday if d is not already Thursday
>+//   civil_day thurs2 = next_weekday(d, weekday::thursday) - 7;
>+//
>+using detail::next_weekday;
>+using detail::prev_weekday;
>+
>+// Returns the day-of-year for the given civil_day.
>+//
>+//   civil_day a(2015, 1, 1);
>+//   int yd_jan_1 = get_yearday(a);   // yd_jan_1 = 1
>+//   civil_day b(2015, 12, 31);
>+//   int yd_dec_31 = get_yearday(b);  // yd_dec_31 = 365
>+//
>+using detail::get_yearday;
>+
>+}  // namespace cctz
>+}  // namespace time_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_TIME_INTERNAL_CCTZ_CIVIL_TIME_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/include/cctz/civil_time_detail.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/include/cctz/civil_time_detail.h
>new file mode 100644
>index 00000000000..2362a4f4fbc
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/include/cctz/civil_time_detail.h
>@@ -0,0 +1,560 @@
>+// Copyright 2016 Google Inc. All Rights Reserved.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//   http://www.apache.org/licenses/LICENSE-2.0
>+//
>+//   Unless required by applicable law or agreed to in writing, software
>+//   distributed under the License is distributed on an "AS IS" BASIS,
>+//   WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+//   See the License for the specific language governing permissions and
>+//   limitations under the License.
>+
>+#ifndef ABSL_TIME_INTERNAL_CCTZ_CIVIL_TIME_DETAIL_H_
>+#define ABSL_TIME_INTERNAL_CCTZ_CIVIL_TIME_DETAIL_H_
>+
>+#include <cstdint>
>+#include <limits>
>+#include <ostream>
>+#include <type_traits>
>+
>+// Disable constexpr support unless we are in C++14 mode.
>+#if __cpp_constexpr >= 201304 || _MSC_VER >= 1910
>+#define CONSTEXPR_D constexpr  // data
>+#define CONSTEXPR_F constexpr  // function
>+#define CONSTEXPR_M constexpr  // member
>+#else
>+#define CONSTEXPR_D const
>+#define CONSTEXPR_F inline
>+#define CONSTEXPR_M
>+#endif
>+
>+namespace absl {
>+namespace time_internal {
>+namespace cctz {
>+
>+// Support years that at least span the range of 64-bit time_t values.
>+using year_t = std::int_fast64_t;
>+
>+// Type alias that indicates an argument is not normalized (e.g., the
>+// constructor parameters and operands/results of addition/subtraction).
>+using diff_t = std::int_fast64_t;
>+
>+namespace detail {
>+
>+// Type aliases that indicate normalized argument values.
>+using month_t = std::int_fast8_t;   // [1:12]
>+using day_t = std::int_fast8_t;     // [1:31]
>+using hour_t = std::int_fast8_t;    // [0:23]
>+using minute_t = std::int_fast8_t;  // [0:59]
>+using second_t = std::int_fast8_t;  // [0:59]
>+
>+// Normalized civil-time fields: Y-M-D HH:MM:SS.
>+struct fields {
>+  CONSTEXPR_M fields(year_t year, month_t month, day_t day,
>+                     hour_t hour, minute_t minute, second_t second)
>+      : y(year), m(month), d(day), hh(hour), mm(minute), ss(second) {}
>+  std::int_least64_t y;
>+  std::int_least8_t m;
>+  std::int_least8_t d;
>+  std::int_least8_t hh;
>+  std::int_least8_t mm;
>+  std::int_least8_t ss;
>+};
>+
>+struct second_tag {};
>+struct minute_tag : second_tag {};
>+struct hour_tag : minute_tag {};
>+struct day_tag : hour_tag {};
>+struct month_tag : day_tag {};
>+struct year_tag : month_tag {};
>+
>+////////////////////////////////////////////////////////////////////////
>+
>+// Field normalization (without avoidable overflow).
>+
>+namespace impl {
>+
>+CONSTEXPR_F bool is_leap_year(year_t y) noexcept {
>+  return y % 4 == 0 && (y % 100 != 0 || y % 400 == 0);
>+}
>+CONSTEXPR_F int year_index(year_t y, month_t m) noexcept {
>+  return (static_cast<int>((y + (m > 2)) % 400) + 400) % 400;
>+}
>+CONSTEXPR_F int days_per_century(year_t y, month_t m) noexcept {
>+  const int yi = year_index(y, m);
>+  return 36524 + (yi == 0 || yi > 300);
>+}
>+CONSTEXPR_F int days_per_4years(year_t y, month_t m) noexcept {
>+  const int yi = year_index(y, m);
>+  return 1460 + (yi == 0 || yi > 300 || (yi - 1) % 100 < 96);
>+}
>+CONSTEXPR_F int days_per_year(year_t y, month_t m) noexcept {
>+  return is_leap_year(y + (m > 2)) ? 366 : 365;
>+}
>+CONSTEXPR_F int days_per_month(year_t y, month_t m) noexcept {
>+  CONSTEXPR_D int k_days_per_month[1 + 12] = {
>+      -1, 31, 28, 31, 30, 31, 30, 31, 31, 30, 31, 30, 31  // non leap year
>+  };
>+  return k_days_per_month[m] + (m == 2 && is_leap_year(y));
>+}
>+
>+CONSTEXPR_F fields n_day(year_t y, month_t m, diff_t d, diff_t cd,
>+                         hour_t hh, minute_t mm, second_t ss) noexcept {
>+  y += (cd / 146097) * 400;
>+  cd %= 146097;
>+  if (cd < 0) {
>+    y -= 400;
>+    cd += 146097;
>+  }
>+  y += (d / 146097) * 400;
>+  d = d % 146097 + cd;
>+  if (d > 0) {
>+    if (d > 146097) {
>+      y += 400;
>+      d -= 146097;
>+    }
>+  } else {
>+    if (d > -365) {
>+      // We often hit the previous year when stepping a civil time backwards,
>+      // so special case it to avoid counting up by 100/4/1-year chunks.
>+      y -= 1;
>+      d += days_per_year(y, m);
>+    } else {
>+      y -= 400;
>+      d += 146097;
>+    }
>+  }
>+  if (d > 365) {
>+    for (int n = days_per_century(y, m); d > n; n = days_per_century(y, m)) {
>+      d -= n;
>+      y += 100;
>+    }
>+    for (int n = days_per_4years(y, m); d > n; n = days_per_4years(y, m)) {
>+      d -= n;
>+      y += 4;
>+    }
>+    for (int n = days_per_year(y, m); d > n; n = days_per_year(y, m)) {
>+      d -= n;
>+      ++y;
>+    }
>+  }
>+  if (d > 28) {
>+    for (int n = days_per_month(y, m); d > n; n = days_per_month(y, m)) {
>+      d -= n;
>+      if (++m > 12) {
>+        ++y;
>+        m = 1;
>+      }
>+    }
>+  }
>+  return fields(y, m, static_cast<day_t>(d), hh, mm, ss);
>+}
>+CONSTEXPR_F fields n_mon(year_t y, diff_t m, diff_t d, diff_t cd,
>+                         hour_t hh, minute_t mm, second_t ss) noexcept {
>+  if (m != 12) {
>+    y += m / 12;
>+    m %= 12;
>+    if (m <= 0) {
>+      y -= 1;
>+      m += 12;
>+    }
>+  }
>+  return n_day(y, static_cast<month_t>(m), d, cd, hh, mm, ss);
>+}
>+CONSTEXPR_F fields n_hour(year_t y, diff_t m, diff_t d, diff_t cd,
>+                          diff_t hh, minute_t mm, second_t ss) noexcept {
>+  cd += hh / 24;
>+  hh %= 24;
>+  if (hh < 0) {
>+    cd -= 1;
>+    hh += 24;
>+  }
>+  return n_mon(y, m, d, cd, static_cast<hour_t>(hh), mm, ss);
>+}
>+CONSTEXPR_F fields n_min(year_t y, diff_t m, diff_t d, diff_t hh, diff_t ch,
>+                         diff_t mm, second_t ss) noexcept {
>+  ch += mm / 60;
>+  mm %= 60;
>+  if (mm < 0) {
>+    ch -= 1;
>+    mm += 60;
>+  }
>+  return n_hour(y, m, d, hh / 24 + ch / 24, hh % 24 + ch % 24,
>+                static_cast<minute_t>(mm), ss);
>+}
>+CONSTEXPR_F fields n_sec(year_t y, diff_t m, diff_t d, diff_t hh, diff_t mm,
>+                         diff_t ss) noexcept {
>+  // Optimization for when (non-constexpr) fields are already normalized.
>+  if (0 <= ss && ss < 60) {
>+    const second_t nss = static_cast<second_t>(ss);
>+    if (0 <= mm && mm < 60) {
>+      const minute_t nmm = static_cast<minute_t>(mm);
>+      if (0 <= hh && hh < 24) {
>+        const hour_t nhh = static_cast<hour_t>(hh);
>+        if (1 <= d && d <= 28 && 1 <= m && m <= 12) {
>+          const day_t nd = static_cast<day_t>(d);
>+          const month_t nm = static_cast<month_t>(m);
>+          return fields(y, nm, nd, nhh, nmm, nss);
>+        }
>+        return n_mon(y, m, d, 0, nhh, nmm, nss);
>+      }
>+      return n_hour(y, m, d, hh / 24, hh % 24, nmm, nss);
>+    }
>+    return n_min(y, m, d, hh, mm / 60, mm % 60, nss);
>+  }
>+  diff_t cm = ss / 60;
>+  ss %= 60;
>+  if (ss < 0) {
>+    cm -= 1;
>+    ss += 60;
>+  }
>+  return n_min(y, m, d, hh, mm / 60 + cm / 60, mm % 60 + cm % 60,
>+               static_cast<second_t>(ss));
>+}
>+
>+}  // namespace impl
>+
>+////////////////////////////////////////////////////////////////////////
>+
>+// Increments the indicated (normalized) field by "n".
>+CONSTEXPR_F fields step(second_tag, fields f, diff_t n) noexcept {
>+  return impl::n_sec(f.y, f.m, f.d, f.hh, f.mm + n / 60, f.ss + n % 60);
>+}
>+CONSTEXPR_F fields step(minute_tag, fields f, diff_t n) noexcept {
>+  return impl::n_min(f.y, f.m, f.d, f.hh + n / 60, 0, f.mm + n % 60, f.ss);
>+}
>+CONSTEXPR_F fields step(hour_tag, fields f, diff_t n) noexcept {
>+  return impl::n_hour(f.y, f.m, f.d + n / 24, 0, f.hh + n % 24, f.mm, f.ss);
>+}
>+CONSTEXPR_F fields step(day_tag, fields f, diff_t n) noexcept {
>+  return impl::n_day(f.y, f.m, f.d, n, f.hh, f.mm, f.ss);
>+}
>+CONSTEXPR_F fields step(month_tag, fields f, diff_t n) noexcept {
>+  return impl::n_mon(f.y + n / 12, f.m + n % 12, f.d, 0, f.hh, f.mm, f.ss);
>+}
>+CONSTEXPR_F fields step(year_tag, fields f, diff_t n) noexcept {
>+  return fields(f.y + n, f.m, f.d, f.hh, f.mm, f.ss);
>+}
>+
>+////////////////////////////////////////////////////////////////////////
>+
>+namespace impl {
>+
>+// Returns (v * f + a) but avoiding intermediate overflow when possible.
>+CONSTEXPR_F diff_t scale_add(diff_t v, diff_t f, diff_t a) noexcept {
>+  return (v < 0) ? ((v + 1) * f + a) - f : ((v - 1) * f + a) + f;
>+}
>+
>+// Map a (normalized) Y/M/D to the number of days before/after 1970-01-01.
>+// Probably overflows for years outside [-292277022656:292277026595].
>+CONSTEXPR_F diff_t ymd_ord(year_t y, month_t m, day_t d) noexcept {
>+  const diff_t eyear = (m <= 2) ? y - 1 : y;
>+  const diff_t era = (eyear >= 0 ? eyear : eyear - 399) / 400;
>+  const diff_t yoe = eyear - era * 400;
>+  const diff_t doy = (153 * (m + (m > 2 ? -3 : 9)) + 2) / 5 + d - 1;
>+  const diff_t doe = yoe * 365 + yoe / 4 - yoe / 100 + doy;
>+  return era * 146097 + doe - 719468;
>+}
>+
>+// Returns the difference in days between two normalized Y-M-D tuples.
>+// ymd_ord() will encounter integer overflow given extreme year values,
>+// yet the difference between two such extreme values may actually be
>+// small, so we take a little care to avoid overflow when possible by
>+// exploiting the 146097-day cycle.
>+CONSTEXPR_F diff_t day_difference(year_t y1, month_t m1, day_t d1,
>+                                  year_t y2, month_t m2, day_t d2) noexcept {
>+  const diff_t a_c4_off = y1 % 400;
>+  const diff_t b_c4_off = y2 % 400;
>+  diff_t c4_diff = (y1 - a_c4_off) - (y2 - b_c4_off);
>+  diff_t delta = ymd_ord(a_c4_off, m1, d1) - ymd_ord(b_c4_off, m2, d2);
>+  if (c4_diff > 0 && delta < 0) {
>+    delta += 2 * 146097;
>+    c4_diff -= 2 * 400;
>+  } else if (c4_diff < 0 && delta > 0) {
>+    delta -= 2 * 146097;
>+    c4_diff += 2 * 400;
>+  }
>+  return (c4_diff / 400 * 146097) + delta;
>+}
>+
>+}  // namespace impl
>+
>+// Returns the difference between fields structs using the indicated unit.
>+CONSTEXPR_F diff_t difference(year_tag, fields f1, fields f2) noexcept {
>+  return f1.y - f2.y;
>+}
>+CONSTEXPR_F diff_t difference(month_tag, fields f1, fields f2) noexcept {
>+  return impl::scale_add(difference(year_tag{}, f1, f2), 12, (f1.m - f2.m));
>+}
>+CONSTEXPR_F diff_t difference(day_tag, fields f1, fields f2) noexcept {
>+  return impl::day_difference(f1.y, f1.m, f1.d, f2.y, f2.m, f2.d);
>+}
>+CONSTEXPR_F diff_t difference(hour_tag, fields f1, fields f2) noexcept {
>+  return impl::scale_add(difference(day_tag{}, f1, f2), 24, (f1.hh - f2.hh));
>+}
>+CONSTEXPR_F diff_t difference(minute_tag, fields f1, fields f2) noexcept {
>+  return impl::scale_add(difference(hour_tag{}, f1, f2), 60, (f1.mm - f2.mm));
>+}
>+CONSTEXPR_F diff_t difference(second_tag, fields f1, fields f2) noexcept {
>+  return impl::scale_add(difference(minute_tag{}, f1, f2), 60, f1.ss - f2.ss);
>+}
>+
>+////////////////////////////////////////////////////////////////////////
>+
>+// Aligns the (normalized) fields struct to the indicated field.
>+CONSTEXPR_F fields align(second_tag, fields f) noexcept {
>+  return f;
>+}
>+CONSTEXPR_F fields align(minute_tag, fields f) noexcept {
>+  return fields{f.y, f.m, f.d, f.hh, f.mm, 0};
>+}
>+CONSTEXPR_F fields align(hour_tag, fields f) noexcept {
>+  return fields{f.y, f.m, f.d, f.hh, 0, 0};
>+}
>+CONSTEXPR_F fields align(day_tag, fields f) noexcept {
>+  return fields{f.y, f.m, f.d, 0, 0, 0};
>+}
>+CONSTEXPR_F fields align(month_tag, fields f) noexcept {
>+  return fields{f.y, f.m, 1, 0, 0, 0};
>+}
>+CONSTEXPR_F fields align(year_tag, fields f) noexcept {
>+  return fields{f.y, 1, 1, 0, 0, 0};
>+}
>+
>+////////////////////////////////////////////////////////////////////////
>+
>+template <typename T>
>+class civil_time {
>+ public:
>+  explicit CONSTEXPR_M civil_time(year_t y, diff_t m = 1, diff_t d = 1,
>+                                  diff_t hh = 0, diff_t mm = 0,
>+                                  diff_t ss = 0) noexcept
>+      : civil_time(impl::n_sec(y, m, d, hh, mm, ss)) {}
>+
>+  CONSTEXPR_M civil_time() noexcept : f_{1970, 1, 1, 0, 0, 0} {}
>+  civil_time(const civil_time&) = default;
>+  civil_time& operator=(const civil_time&) = default;
>+
>+  // Conversion between civil times of different alignment. Conversion to
>+  // a more precise alignment is allowed implicitly (e.g., day -> hour),
>+  // but conversion where information is discarded must be explicit
>+  // (e.g., second -> minute).
>+  template <typename U, typename S>
>+  using preserves_data =
>+      typename std::enable_if<std::is_base_of<U, S>::value>::type;
>+  template <typename U>
>+  CONSTEXPR_M civil_time(const civil_time<U>& ct,
>+                         preserves_data<T, U>* = nullptr) noexcept
>+      : civil_time(ct.f_) {}
>+  template <typename U>
>+  explicit CONSTEXPR_M civil_time(const civil_time<U>& ct,
>+                                  preserves_data<U, T>* = nullptr) noexcept
>+      : civil_time(ct.f_) {}
>+
>+  // Factories for the maximum/minimum representable civil_time.
>+  static civil_time max() {
>+    const auto max_year = std::numeric_limits<std::int_least64_t>::max();
>+    return civil_time(max_year, 12, 31, 23, 59, 59);
>+  }
>+  static civil_time min() {
>+    const auto min_year = std::numeric_limits<std::int_least64_t>::min();
>+    return civil_time(min_year, 1, 1, 0, 0, 0);
>+  }
>+
>+  // Field accessors.  Note: All but year() return an int.
>+  CONSTEXPR_M year_t year() const noexcept { return f_.y; }
>+  CONSTEXPR_M int month() const noexcept { return f_.m; }
>+  CONSTEXPR_M int day() const noexcept { return f_.d; }
>+  CONSTEXPR_M int hour() const noexcept { return f_.hh; }
>+  CONSTEXPR_M int minute() const noexcept { return f_.mm; }
>+  CONSTEXPR_M int second() const noexcept { return f_.ss; }
>+
>+  // Assigning arithmetic.
>+  CONSTEXPR_M civil_time& operator+=(diff_t n) noexcept {
>+    f_ = step(T{}, f_, n);
>+    return *this;
>+  }
>+  CONSTEXPR_M civil_time& operator-=(diff_t n) noexcept {
>+    if (n != std::numeric_limits<diff_t>::min()) {
>+      f_ = step(T{}, f_, -n);
>+    } else {
>+      f_ = step(T{}, step(T{}, f_, -(n + 1)), 1);
>+    }
>+    return *this;
>+  }
>+  CONSTEXPR_M civil_time& operator++() noexcept {
>+    return *this += 1;
>+  }
>+  CONSTEXPR_M civil_time operator++(int) noexcept {
>+    const civil_time a = *this;
>+    ++*this;
>+    return a;
>+  }
>+  CONSTEXPR_M civil_time& operator--() noexcept {
>+    return *this -= 1;
>+  }
>+  CONSTEXPR_M civil_time operator--(int) noexcept {
>+    const civil_time a = *this;
>+    --*this;
>+    return a;
>+  }
>+
>+  // Binary arithmetic operators.
>+  friend CONSTEXPR_F civil_time operator+(civil_time a, diff_t n) noexcept {
>+    return a += n;
>+  }
>+  friend CONSTEXPR_F civil_time operator+(diff_t n, civil_time a) noexcept {
>+    return a += n;
>+  }
>+  friend CONSTEXPR_F civil_time operator-(civil_time a, diff_t n) noexcept {
>+    return a -= n;
>+  }
>+  friend CONSTEXPR_F diff_t operator-(civil_time lhs, civil_time rhs) noexcept {
>+    return difference(T{}, lhs.f_, rhs.f_);
>+  }
>+
>+ private:
>+  // All instantiations of this template are allowed to call the following
>+  // private constructor and access the private fields member.
>+  template <typename U>
>+  friend class civil_time;
>+
>+  // The designated constructor that all others eventually call.
>+  explicit CONSTEXPR_M civil_time(fields f) noexcept : f_(align(T{}, f)) {}
>+
>+  fields f_;
>+};
>+
>+// Disallows difference between differently aligned types.
>+// auto n = civil_day(...) - civil_hour(...);  // would be confusing.
>+template <typename T, typename U>
>+CONSTEXPR_F diff_t operator-(civil_time<T>, civil_time<U>) = delete;
>+
>+using civil_year = civil_time<year_tag>;
>+using civil_month = civil_time<month_tag>;
>+using civil_day = civil_time<day_tag>;
>+using civil_hour = civil_time<hour_tag>;
>+using civil_minute = civil_time<minute_tag>;
>+using civil_second = civil_time<second_tag>;
>+
>+////////////////////////////////////////////////////////////////////////
>+
>+// Relational operators that work with differently aligned objects.
>+// Always compares all six fields.
>+template <typename T1, typename T2>
>+CONSTEXPR_F bool operator<(const civil_time<T1>& lhs,
>+                           const civil_time<T2>& rhs) noexcept {
>+  return (lhs.year() < rhs.year() ||
>+          (lhs.year() == rhs.year() &&
>+           (lhs.month() < rhs.month() ||
>+            (lhs.month() == rhs.month() &&
>+             (lhs.day() < rhs.day() ||
>+              (lhs.day() == rhs.day() &&
>+               (lhs.hour() < rhs.hour() ||
>+                (lhs.hour() == rhs.hour() &&
>+                 (lhs.minute() < rhs.minute() ||
>+                  (lhs.minute() == rhs.minute() &&
>+                   (lhs.second() < rhs.second())))))))))));
>+}
>+template <typename T1, typename T2>
>+CONSTEXPR_F bool operator<=(const civil_time<T1>& lhs,
>+                            const civil_time<T2>& rhs) noexcept {
>+  return !(rhs < lhs);
>+}
>+template <typename T1, typename T2>
>+CONSTEXPR_F bool operator>=(const civil_time<T1>& lhs,
>+                            const civil_time<T2>& rhs) noexcept {
>+  return !(lhs < rhs);
>+}
>+template <typename T1, typename T2>
>+CONSTEXPR_F bool operator>(const civil_time<T1>& lhs,
>+                           const civil_time<T2>& rhs) noexcept {
>+  return rhs < lhs;
>+}
>+template <typename T1, typename T2>
>+CONSTEXPR_F bool operator==(const civil_time<T1>& lhs,
>+                            const civil_time<T2>& rhs) noexcept {
>+  return lhs.year() == rhs.year() && lhs.month() == rhs.month() &&
>+         lhs.day() == rhs.day() && lhs.hour() == rhs.hour() &&
>+         lhs.minute() == rhs.minute() && lhs.second() == rhs.second();
>+}
>+template <typename T1, typename T2>
>+CONSTEXPR_F bool operator!=(const civil_time<T1>& lhs,
>+                            const civil_time<T2>& rhs) noexcept {
>+  return !(lhs == rhs);
>+}
>+
>+////////////////////////////////////////////////////////////////////////
>+
>+enum class weekday {
>+  monday,
>+  tuesday,
>+  wednesday,
>+  thursday,
>+  friday,
>+  saturday,
>+  sunday,
>+};
>+
>+CONSTEXPR_F weekday get_weekday(const civil_day& cd) noexcept {
>+  CONSTEXPR_D weekday k_weekday_by_sun_off[7] = {
>+      weekday::sunday,     weekday::monday,    weekday::tuesday,
>+      weekday::wednesday,  weekday::thursday,  weekday::friday,
>+      weekday::saturday,
>+  };
>+  CONSTEXPR_D int k_weekday_offsets[1 + 12] = {
>+      -1, 0, 3, 2, 5, 0, 3, 5, 1, 4, 6, 2, 4,
>+  };
>+  year_t wd = cd.year() - (cd.month() < 3);
>+  if (wd >= 0) {
>+    wd += wd / 4 - wd / 100 + wd / 400;
>+  } else {
>+    wd += (wd - 3) / 4 - (wd - 99) / 100 + (wd - 399) / 400;
>+  }
>+  wd += k_weekday_offsets[cd.month()] + cd.day();
>+  return k_weekday_by_sun_off[(wd % 7 + 7) % 7];
>+}
>+
>+////////////////////////////////////////////////////////////////////////
>+
>+CONSTEXPR_F civil_day next_weekday(civil_day cd, weekday wd) noexcept {
>+  do { cd += 1; } while (get_weekday(cd) != wd);
>+  return cd;
>+}
>+
>+CONSTEXPR_F civil_day prev_weekday(civil_day cd, weekday wd) noexcept {
>+  do { cd -= 1; } while (get_weekday(cd) != wd);
>+  return cd;
>+}
>+
>+CONSTEXPR_F int get_yearday(const civil_day& cd) noexcept {
>+  CONSTEXPR_D int k_month_offsets[1 + 12] = {
>+      -1, 0, 31, 59, 90, 120, 151, 181, 212, 243, 273, 304, 334,
>+  };
>+  const int feb29 = (cd.month() > 2 && impl::is_leap_year(cd.year()));
>+  return k_month_offsets[cd.month()] + feb29 + cd.day();
>+}
>+
>+////////////////////////////////////////////////////////////////////////
>+
>+std::ostream& operator<<(std::ostream& os, const civil_year& y);
>+std::ostream& operator<<(std::ostream& os, const civil_month& m);
>+std::ostream& operator<<(std::ostream& os, const civil_day& d);
>+std::ostream& operator<<(std::ostream& os, const civil_hour& h);
>+std::ostream& operator<<(std::ostream& os, const civil_minute& m);
>+std::ostream& operator<<(std::ostream& os, const civil_second& s);
>+std::ostream& operator<<(std::ostream& os, weekday wd);
>+
>+}  // namespace detail
>+}  // namespace cctz
>+}  // namespace time_internal
>+}  // namespace absl
>+
>+#undef CONSTEXPR_M
>+#undef CONSTEXPR_F
>+#undef CONSTEXPR_D
>+
>+#endif  // ABSL_TIME_INTERNAL_CCTZ_CIVIL_TIME_DETAIL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/include/cctz/time_zone.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/include/cctz/time_zone.h
>new file mode 100644
>index 00000000000..0b9764ea72a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/include/cctz/time_zone.h
>@@ -0,0 +1,380 @@
>+// Copyright 2016 Google Inc. All Rights Reserved.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//   http://www.apache.org/licenses/LICENSE-2.0
>+//
>+//   Unless required by applicable law or agreed to in writing, software
>+//   distributed under the License is distributed on an "AS IS" BASIS,
>+//   WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+//   See the License for the specific language governing permissions and
>+//   limitations under the License.
>+
>+// A library for translating between absolute times (represented by
>+// std::chrono::time_points of the std::chrono::system_clock) and civil
>+// times (represented by cctz::civil_second) using the rules defined by
>+// a time zone (cctz::time_zone).
>+
>+#ifndef ABSL_TIME_INTERNAL_CCTZ_TIME_ZONE_H_
>+#define ABSL_TIME_INTERNAL_CCTZ_TIME_ZONE_H_
>+
>+#include <chrono>
>+#include <cstdint>
>+#include <string>
>+#include <utility>
>+
>+#include "absl/time/internal/cctz/include/cctz/civil_time.h"
>+
>+namespace absl {
>+namespace time_internal {
>+namespace cctz {
>+
>+// Convenience aliases. Not intended as public API points.
>+template <typename D>
>+using time_point = std::chrono::time_point<std::chrono::system_clock, D>;
>+using seconds = std::chrono::duration<std::int_fast64_t>;
>+using sys_seconds = seconds;  // Deprecated.  Use cctz::seconds instead.
>+
>+namespace detail {
>+template <typename D>
>+inline std::pair<time_point<seconds>, D>
>+split_seconds(const time_point<D>& tp) {
>+  auto sec = std::chrono::time_point_cast<seconds>(tp);
>+  auto sub = tp - sec;
>+  if (sub.count() < 0) {
>+    sec -= seconds(1);
>+    sub += seconds(1);
>+  }
>+  return {sec, std::chrono::duration_cast<D>(sub)};
>+}
>+inline std::pair<time_point<seconds>, seconds>
>+split_seconds(const time_point<seconds>& tp) {
>+  return {tp, seconds::zero()};
>+}
>+}  // namespace detail
>+
>+// cctz::time_zone is an opaque, small, value-type class representing a
>+// geo-political region within which particular rules are used for mapping
>+// between absolute and civil times. Time zones are named using the TZ
>+// identifiers from the IANA Time Zone Database, such as "America/Los_Angeles"
>+// or "Australia/Sydney". Time zones are created from factory functions such
>+// as load_time_zone(). Note: strings like "PST" and "EDT" are not valid TZ
>+// identifiers.
>+//
>+// Example:
>+//   cctz::time_zone utc = cctz::utc_time_zone();
>+//   cctz::time_zone pst = cctz::fixed_time_zone(std::chrono::hours(-8));
>+//   cctz::time_zone loc = cctz::local_time_zone();
>+//   cctz::time_zone lax;
>+//   if (!cctz::load_time_zone("America/Los_Angeles", &lax)) { ... }
>+//
>+// See also:
>+// - http://www.iana.org/time-zones
>+// - http://en.wikipedia.org/wiki/Zoneinfo
>+class time_zone {
>+ public:
>+  time_zone() : time_zone(nullptr) {}  // Equivalent to UTC
>+  time_zone(const time_zone&) = default;
>+  time_zone& operator=(const time_zone&) = default;
>+
>+  std::string name() const;
>+
>+  // An absolute_lookup represents the civil time (cctz::civil_second) within
>+  // this time_zone at the given absolute time (time_point). There are
>+  // additionally a few other fields that may be useful when working with
>+  // older APIs, such as std::tm.
>+  //
>+  // Example:
>+  //   const cctz::time_zone tz = ...
>+  //   const auto tp = std::chrono::system_clock::now();
>+  //   const cctz::time_zone::absolute_lookup al = tz.lookup(tp);
>+  struct absolute_lookup {
>+    civil_second cs;
>+    // Note: The following fields exist for backward compatibility with older
>+    // APIs. Accessing these fields directly is a sign of imprudent logic in
>+    // the calling code. Modern time-related code should only access this data
>+    // indirectly by way of cctz::format().
>+    int offset;        // civil seconds east of UTC
>+    bool is_dst;       // is offset non-standard?
>+    const char* abbr;  // time-zone abbreviation (e.g., "PST")
>+  };
>+  absolute_lookup lookup(const time_point<seconds>& tp) const;
>+  template <typename D>
>+  absolute_lookup lookup(const time_point<D>& tp) const {
>+    return lookup(detail::split_seconds(tp).first);
>+  }
>+
>+  // A civil_lookup represents the absolute time(s) (time_point) that
>+  // correspond to the given civil time (cctz::civil_second) within this
>+  // time_zone. Usually the given civil time represents a unique instant
>+  // in time, in which case the conversion is unambiguous. However,
>+  // within this time zone, the given civil time may be skipped (e.g.,
>+  // during a positive UTC offset shift), or repeated (e.g., during a
>+  // negative UTC offset shift). To account for these possibilities,
>+  // civil_lookup is richer than just a single time_point.
>+  //
>+  // In all cases the civil_lookup::kind enum will indicate the nature
>+  // of the given civil-time argument, and the pre, trans, and post
>+  // members will give the absolute time answers using the pre-transition
>+  // offset, the transition point itself, and the post-transition offset,
>+  // respectively (all three times are equal if kind == UNIQUE). If any
>+  // of these three absolute times is outside the representable range of a
>+  // time_point<seconds> the field is set to its maximum/minimum value.
>+  //
>+  // Example:
>+  //   cctz::time_zone lax;
>+  //   if (!cctz::load_time_zone("America/Los_Angeles", &lax)) { ... }
>+  //
>+  //   // A unique civil time.
>+  //   auto jan01 = lax.lookup(cctz::civil_second(2011, 1, 1, 0, 0, 0));
>+  //   // jan01.kind == cctz::time_zone::civil_lookup::UNIQUE
>+  //   // jan01.pre    is 2011/01/01 00:00:00 -0800
>+  //   // jan01.trans  is 2011/01/01 00:00:00 -0800
>+  //   // jan01.post   is 2011/01/01 00:00:00 -0800
>+  //
>+  //   // A Spring DST transition, when there is a gap in civil time.
>+  //   auto mar13 = lax.lookup(cctz::civil_second(2011, 3, 13, 2, 15, 0));
>+  //   // mar13.kind == cctz::time_zone::civil_lookup::SKIPPED
>+  //   // mar13.pre   is 2011/03/13 03:15:00 -0700
>+  //   // mar13.trans is 2011/03/13 03:00:00 -0700
>+  //   // mar13.post  is 2011/03/13 01:15:00 -0800
>+  //
>+  //   // A Fall DST transition, when civil times are repeated.
>+  //   auto nov06 = lax.lookup(cctz::civil_second(2011, 11, 6, 1, 15, 0));
>+  //   // nov06.kind == cctz::time_zone::civil_lookup::REPEATED
>+  //   // nov06.pre   is 2011/11/06 01:15:00 -0700
>+  //   // nov06.trans is 2011/11/06 01:00:00 -0800
>+  //   // nov06.post  is 2011/11/06 01:15:00 -0800
>+  struct civil_lookup {
>+    enum civil_kind {
>+      UNIQUE,    // the civil time was singular (pre == trans == post)
>+      SKIPPED,   // the civil time did not exist (pre >= trans > post)
>+      REPEATED,  // the civil time was ambiguous (pre < trans <= post)
>+    } kind;
>+    time_point<seconds> pre;    // uses the pre-transition offset
>+    time_point<seconds> trans;  // instant of civil-offset change
>+    time_point<seconds> post;   // uses the post-transition offset
>+  };
>+  civil_lookup lookup(const civil_second& cs) const;
>+
>+  // Finds the time of the next/previous offset change in this time zone.
>+  //
>+  // By definition, next_transition(tp, &trans) returns false when tp has
>+  // its maximum value, and prev_transition(tp, &trans) returns false
>+  // when tp has its minimum value. If the zone has no transitions, the
>+  // result will also be false no matter what the argument.
>+  //
>+  // Otherwise, when tp has its minimum value, next_transition(tp, &trans)
>+  // returns true and sets trans to the first recorded transition. Chains
>+  // of calls to next_transition()/prev_transition() will eventually return
>+  // false, but it is unspecified exactly when next_transition(tp, &trans)
>+  // jumps to false, or what time is set by prev_transition(tp, &trans) for
>+  // a very distant tp.
>+  //
>+  // Note: Enumeration of time-zone transitions is for informational purposes
>+  // only. Modern time-related code should not care about when offset changes
>+  // occur.
>+  //
>+  // Example:
>+  //   cctz::time_zone nyc;
>+  //   if (!cctz::load_time_zone("America/New_York", &nyc)) { ... }
>+  //   const auto now = std::chrono::system_clock::now();
>+  //   auto tp = cctz::time_point<cctz::seconds>::min();
>+  //   cctz::time_zone::civil_transition trans;
>+  //   while (tp <= now && nyc.next_transition(tp, &trans)) {
>+  //     // transition: trans.from -> trans.to
>+  //     tp = nyc.lookup(trans.to).trans;
>+  //   }
>+  struct civil_transition {
>+    civil_second from;  // the civil time we jump from
>+    civil_second to;    // the civil time we jump to
>+  };
>+  bool next_transition(const time_point<seconds>& tp,
>+                       civil_transition* trans) const;
>+  template <typename D>
>+  bool next_transition(const time_point<D>& tp,
>+                       civil_transition* trans) const {
>+    return next_transition(detail::split_seconds(tp).first, trans);
>+  }
>+  bool prev_transition(const time_point<seconds>& tp,
>+                       civil_transition* trans) const;
>+  template <typename D>
>+  bool prev_transition(const time_point<D>& tp,
>+                       civil_transition* trans) const {
>+    return prev_transition(detail::split_seconds(tp).first, trans);
>+  }
>+
>+  // version() and description() provide additional information about the
>+  // time zone. The content of each of the returned strings is unspecified,
>+  // however, when the IANA Time Zone Database is the underlying data source
>+  // the version() std::string will be in the familar form (e.g, "2018e") or
>+  // empty when unavailable.
>+  //
>+  // Note: These functions are for informational or testing purposes only.
>+  std::string version() const;  // empty when unknown
>+  std::string description() const;
>+
>+  // Relational operators.
>+  friend bool operator==(time_zone lhs, time_zone rhs) {
>+    return &lhs.effective_impl() == &rhs.effective_impl();
>+  }
>+  friend bool operator!=(time_zone lhs, time_zone rhs) {
>+    return !(lhs == rhs);
>+  }
>+
>+  class Impl;
>+
>+ private:
>+  explicit time_zone(const Impl* impl) : impl_(impl) {}
>+  const Impl& effective_impl() const;  // handles implicit UTC
>+  const Impl* impl_;
>+};
>+
>+// Loads the named time zone. May perform I/O on the initial load.
>+// If the name is invalid, or some other kind of error occurs, returns
>+// false and "*tz" is set to the UTC time zone.
>+bool load_time_zone(const std::string& name, time_zone* tz);
>+
>+// Returns a time_zone representing UTC. Cannot fail.
>+time_zone utc_time_zone();
>+
>+// Returns a time zone that is a fixed offset (seconds east) from UTC.
>+// Note: If the absolute value of the offset is greater than 24 hours
>+// you'll get UTC (i.e., zero offset) instead.
>+time_zone fixed_time_zone(const seconds& offset);
>+
>+// Returns a time zone representing the local time zone. Falls back to UTC.
>+// Note: local_time_zone.name() may only be something like "localtime".
>+time_zone local_time_zone();
>+
>+// Returns the civil time (cctz::civil_second) within the given time zone at
>+// the given absolute time (time_point). Since the additional fields provided
>+// by the time_zone::absolute_lookup struct should rarely be needed in modern
>+// code, this convert() function is simpler and should be preferred.
>+template <typename D>
>+inline civil_second convert(const time_point<D>& tp, const time_zone& tz) {
>+  return tz.lookup(tp).cs;
>+}
>+
>+// Returns the absolute time (time_point) that corresponds to the given civil
>+// time within the given time zone. If the civil time is not unique (i.e., if
>+// it was either repeated or non-existent), then the returned time_point is
>+// the best estimate that preserves relative order. That is, this function
>+// guarantees that if cs1 < cs2, then convert(cs1, tz) <= convert(cs2, tz).
>+inline time_point<seconds> convert(const civil_second& cs,
>+                                   const time_zone& tz) {
>+  const time_zone::civil_lookup cl = tz.lookup(cs);
>+  if (cl.kind == time_zone::civil_lookup::SKIPPED) return cl.trans;
>+  return cl.pre;
>+}
>+
>+namespace detail {
>+using femtoseconds = std::chrono::duration<std::int_fast64_t, std::femto>;
>+std::string format(const std::string&, const time_point<seconds>&,
>+                   const femtoseconds&, const time_zone&);
>+bool parse(const std::string&, const std::string&, const time_zone&,
>+           time_point<seconds>*, femtoseconds*, std::string* err = nullptr);
>+}  // namespace detail
>+
>+// Formats the given time_point in the given cctz::time_zone according to
>+// the provided format std::string. Uses strftime()-like formatting options,
>+// with the following extensions:
>+//
>+//   - %Ez  - RFC3339-compatible numeric UTC offset (+hh:mm or -hh:mm)
>+//   - %E*z - Full-resolution numeric UTC offset (+hh:mm:ss or -hh:mm:ss)
>+//   - %E#S - Seconds with # digits of fractional precision
>+//   - %E*S - Seconds with full fractional precision (a literal '*')
>+//   - %E#f - Fractional seconds with # digits of precision
>+//   - %E*f - Fractional seconds with full precision (a literal '*')
>+//   - %E4Y - Four-character years (-999 ... -001, 0000, 0001 ... 9999)
>+//
>+// Note that %E0S behaves like %S, and %E0f produces no characters. In
>+// contrast %E*f always produces at least one digit, which may be '0'.
>+//
>+// Note that %Y produces as many characters as it takes to fully render the
>+// year. A year outside of [-999:9999] when formatted with %E4Y will produce
>+// more than four characters, just like %Y.
>+//
>+// Tip: Format strings should include the UTC offset (e.g., %z, %Ez, or %E*z)
>+// so that the resulting std::string uniquely identifies an absolute time.
>+//
>+// Example:
>+//   cctz::time_zone lax;
>+//   if (!cctz::load_time_zone("America/Los_Angeles", &lax)) { ... }
>+//   auto tp = cctz::convert(cctz::civil_second(2013, 1, 2, 3, 4, 5), lax);
>+//   std::string f = cctz::format("%H:%M:%S", tp, lax);  // "03:04:05"
>+//   f = cctz::format("%H:%M:%E3S", tp, lax);            // "03:04:05.000"
>+template <typename D>
>+inline std::string format(const std::string& fmt, const time_point<D>& tp,
>+                          const time_zone& tz) {
>+  const auto p = detail::split_seconds(tp);
>+  const auto n = std::chrono::duration_cast<detail::femtoseconds>(p.second);
>+  return detail::format(fmt, p.first, n, tz);
>+}
>+
>+// Parses an input std::string according to the provided format std::string and
>+// returns the corresponding time_point. Uses strftime()-like formatting
>+// options, with the same extensions as cctz::format(), but with the
>+// exceptions that %E#S is interpreted as %E*S, and %E#f as %E*f. %Ez
>+// and %E*z also accept the same inputs.
>+//
>+// %Y consumes as many numeric characters as it can, so the matching data
>+// should always be terminated with a non-numeric. %E4Y always consumes
>+// exactly four characters, including any sign.
>+//
>+// Unspecified fields are taken from the default date and time of ...
>+//
>+//   "1970-01-01 00:00:00.0 +0000"
>+//
>+// For example, parsing a std::string of "15:45" (%H:%M) will return a time_point
>+// that represents "1970-01-01 15:45:00.0 +0000".
>+//
>+// Note that parse() returns time instants, so it makes most sense to parse
>+// fully-specified date/time strings that include a UTC offset (%z, %Ez, or
>+// %E*z).
>+//
>+// Note also that parse() only heeds the fields year, month, day, hour,
>+// minute, (fractional) second, and UTC offset. Other fields, like weekday (%a
>+// or %A), while parsed for syntactic validity, are ignored in the conversion.
>+//
>+// Date and time fields that are out-of-range will be treated as errors rather
>+// than normalizing them like cctz::civil_second() would do. For example, it
>+// is an error to parse the date "Oct 32, 2013" because 32 is out of range.
>+//
>+// A second of ":60" is normalized to ":00" of the following minute with
>+// fractional seconds discarded. The following table shows how the given
>+// seconds and subseconds will be parsed:
>+//
>+//   "59.x" -> 59.x  // exact
>+//   "60.x" -> 00.0  // normalized
>+//   "00.x" -> 00.x  // exact
>+//
>+// Errors are indicated by returning false.
>+//
>+// Example:
>+//   const cctz::time_zone tz = ...
>+//   std::chrono::system_clock::time_point tp;
>+//   if (cctz::parse("%Y-%m-%d", "2015-10-09", tz, &tp)) {
>+//     ...
>+//   }
>+template <typename D>
>+inline bool parse(const std::string& fmt, const std::string& input,
>+                  const time_zone& tz, time_point<D>* tpp) {
>+  time_point<seconds> sec;
>+  detail::femtoseconds fs;
>+  const bool b = detail::parse(fmt, input, tz, &sec, &fs);
>+  if (b) {
>+    // TODO: Return false if unrepresentable as a time_point<D>.
>+    *tpp = std::chrono::time_point_cast<D>(sec);
>+    *tpp += std::chrono::duration_cast<D>(fs);
>+  }
>+  return b;
>+}
>+
>+}  // namespace cctz
>+}  // namespace time_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_TIME_INTERNAL_CCTZ_TIME_ZONE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/include/cctz/zone_info_source.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/include/cctz/zone_info_source.h
>new file mode 100644
>index 00000000000..20a76979370
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/include/cctz/zone_info_source.h
>@@ -0,0 +1,96 @@
>+// Copyright 2016 Google Inc. All Rights Reserved.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//   http://www.apache.org/licenses/LICENSE-2.0
>+//
>+//   Unless required by applicable law or agreed to in writing, software
>+//   distributed under the License is distributed on an "AS IS" BASIS,
>+//   WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+//   See the License for the specific language governing permissions and
>+//   limitations under the License.
>+
>+#ifndef ABSL_TIME_INTERNAL_CCTZ_ZONE_INFO_SOURCE_H_
>+#define ABSL_TIME_INTERNAL_CCTZ_ZONE_INFO_SOURCE_H_
>+
>+#include <cstddef>
>+#include <functional>
>+#include <memory>
>+#include <string>
>+
>+namespace absl {
>+namespace time_internal {
>+namespace cctz {
>+
>+// A stdio-like interface for providing zoneinfo data for a particular zone.
>+class ZoneInfoSource {
>+ public:
>+  virtual ~ZoneInfoSource();
>+
>+  virtual std::size_t Read(void* ptr, std::size_t size) = 0;  // like fread()
>+  virtual int Skip(std::size_t offset) = 0;  // like fseek()
>+
>+  // Until the zoneinfo data supports versioning information, we provide
>+  // a way for a ZoneInfoSource to indicate it out-of-band.  The default
>+  // implementation returns an empty std::string.
>+  virtual std::string Version() const;
>+};
>+
>+}  // namespace cctz
>+}  // namespace time_internal
>+}  // namespace absl
>+
>+namespace absl {
>+namespace time_internal {
>+namespace cctz_extension {
>+
>+// A function-pointer type for a factory that returns a ZoneInfoSource
>+// given the name of a time zone and a fallback factory.  Returns null
>+// when the data for the named zone cannot be found.
>+using ZoneInfoSourceFactory =
>+    std::unique_ptr<absl::time_internal::cctz::ZoneInfoSource> (*)(
>+        const std::string&,
>+        const std::function<std::unique_ptr<absl::time_internal::cctz::ZoneInfoSource>(
>+            const std::string&)>&);
>+
>+// The user can control the mapping of zone names to zoneinfo data by
>+// providing a definition for cctz_extension::zone_info_source_factory.
>+// For example, given functions my_factory() and my_other_factory() that
>+// can return a ZoneInfoSource for a named zone, we could inject them into
>+// cctz::load_time_zone() with:
>+//
>+//   namespace cctz_extension {
>+//   namespace {
>+//   std::unique_ptr<cctz::ZoneInfoSource> CustomFactory(
>+//       const std::string& name,
>+//       const std::function<std::unique_ptr<cctz::ZoneInfoSource>(
>+//           const std::string& name)>& fallback_factory) {
>+//     if (auto zip = my_factory(name)) return zip;
>+//     if (auto zip = fallback_factory(name)) return zip;
>+//     if (auto zip = my_other_factory(name)) return zip;
>+//     return nullptr;
>+//   }
>+//   }  // namespace
>+//   ZoneInfoSourceFactory zone_info_source_factory = CustomFactory;
>+//   }  // namespace cctz_extension
>+//
>+// This might be used, say, to use zoneinfo data embedded in the program,
>+// or read from a (possibly compressed) file archive, or both.
>+//
>+// cctz_extension::zone_info_source_factory() will be called:
>+//   (1) from the same thread as the cctz::load_time_zone() call,
>+//   (2) only once for any zone name, and
>+//   (3) serially (i.e., no concurrent execution).
>+//
>+// The fallback factory obtains zoneinfo data by reading files in ${TZDIR},
>+// and it is used automatically when no zone_info_source_factory definition
>+// is linked into the program.
>+extern ZoneInfoSourceFactory zone_info_source_factory;
>+
>+}  // namespace cctz_extension
>+}  // namespace time_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_TIME_INTERNAL_CCTZ_ZONE_INFO_SOURCE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/cctz_benchmark.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/cctz_benchmark.cc
>new file mode 100644
>index 00000000000..c97df78c09c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/cctz_benchmark.cc
>@@ -0,0 +1,980 @@
>+// Copyright 2016 Google Inc. All Rights Reserved.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//   http://www.apache.org/licenses/LICENSE-2.0
>+//
>+//   Unless required by applicable law or agreed to in writing, software
>+//   distributed under the License is distributed on an "AS IS" BASIS,
>+//   WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+//   See the License for the specific language governing permissions and
>+//   limitations under the License.
>+
>+#include <algorithm>
>+#include <cassert>
>+#include <chrono>
>+#include <ctime>
>+#include <random>
>+#include <string>
>+#include <vector>
>+
>+#include "benchmark/benchmark.h"
>+#include "absl/time/internal/cctz/include/cctz/civil_time.h"
>+#include "absl/time/internal/cctz/include/cctz/time_zone.h"
>+#include "time_zone_impl.h"
>+
>+namespace {
>+
>+namespace cctz = absl::time_internal::cctz;
>+
>+void BM_Difference_Days(benchmark::State& state) {
>+  const cctz::civil_day c(2014, 8, 22);
>+  const cctz::civil_day epoch(1970, 1, 1);
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(c - epoch);
>+  }
>+}
>+BENCHMARK(BM_Difference_Days);
>+
>+void BM_Step_Days(benchmark::State& state) {
>+  const cctz::civil_day kStart(2014, 8, 22);
>+  cctz::civil_day c = kStart;
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(++c);
>+  }
>+}
>+BENCHMARK(BM_Step_Days);
>+
>+const char RFC3339_full[] = "%Y-%m-%dT%H:%M:%E*S%Ez";
>+const char RFC3339_sec[] = "%Y-%m-%dT%H:%M:%S%Ez";
>+
>+const char RFC1123_full[] = "%a, %d %b %Y %H:%M:%S %z";
>+const char RFC1123_no_wday[] = "%d %b %Y %H:%M:%S %z";
>+
>+// A list of known time-zone names.
>+// TODO: Refactor with src/time_zone_lookup_test.cc.
>+const char* const kTimeZoneNames[] = {
>+  "Africa/Abidjan",
>+  "Africa/Accra",
>+  "Africa/Addis_Ababa",
>+  "Africa/Algiers",
>+  "Africa/Asmara",
>+  "Africa/Asmera",
>+  "Africa/Bamako",
>+  "Africa/Bangui",
>+  "Africa/Banjul",
>+  "Africa/Bissau",
>+  "Africa/Blantyre",
>+  "Africa/Brazzaville",
>+  "Africa/Bujumbura",
>+  "Africa/Cairo",
>+  "Africa/Casablanca",
>+  "Africa/Ceuta",
>+  "Africa/Conakry",
>+  "Africa/Dakar",
>+  "Africa/Dar_es_Salaam",
>+  "Africa/Djibouti",
>+  "Africa/Douala",
>+  "Africa/El_Aaiun",
>+  "Africa/Freetown",
>+  "Africa/Gaborone",
>+  "Africa/Harare",
>+  "Africa/Johannesburg",
>+  "Africa/Juba",
>+  "Africa/Kampala",
>+  "Africa/Khartoum",
>+  "Africa/Kigali",
>+  "Africa/Kinshasa",
>+  "Africa/Lagos",
>+  "Africa/Libreville",
>+  "Africa/Lome",
>+  "Africa/Luanda",
>+  "Africa/Lubumbashi",
>+  "Africa/Lusaka",
>+  "Africa/Malabo",
>+  "Africa/Maputo",
>+  "Africa/Maseru",
>+  "Africa/Mbabane",
>+  "Africa/Mogadishu",
>+  "Africa/Monrovia",
>+  "Africa/Nairobi",
>+  "Africa/Ndjamena",
>+  "Africa/Niamey",
>+  "Africa/Nouakchott",
>+  "Africa/Ouagadougou",
>+  "Africa/Porto-Novo",
>+  "Africa/Sao_Tome",
>+  "Africa/Timbuktu",
>+  "Africa/Tripoli",
>+  "Africa/Tunis",
>+  "Africa/Windhoek",
>+  "America/Adak",
>+  "America/Anchorage",
>+  "America/Anguilla",
>+  "America/Antigua",
>+  "America/Araguaina",
>+  "America/Argentina/Buenos_Aires",
>+  "America/Argentina/Catamarca",
>+  "America/Argentina/ComodRivadavia",
>+  "America/Argentina/Cordoba",
>+  "America/Argentina/Jujuy",
>+  "America/Argentina/La_Rioja",
>+  "America/Argentina/Mendoza",
>+  "America/Argentina/Rio_Gallegos",
>+  "America/Argentina/Salta",
>+  "America/Argentina/San_Juan",
>+  "America/Argentina/San_Luis",
>+  "America/Argentina/Tucuman",
>+  "America/Argentina/Ushuaia",
>+  "America/Aruba",
>+  "America/Asuncion",
>+  "America/Atikokan",
>+  "America/Atka",
>+  "America/Bahia",
>+  "America/Bahia_Banderas",
>+  "America/Barbados",
>+  "America/Belem",
>+  "America/Belize",
>+  "America/Blanc-Sablon",
>+  "America/Boa_Vista",
>+  "America/Bogota",
>+  "America/Boise",
>+  "America/Buenos_Aires",
>+  "America/Cambridge_Bay",
>+  "America/Campo_Grande",
>+  "America/Cancun",
>+  "America/Caracas",
>+  "America/Catamarca",
>+  "America/Cayenne",
>+  "America/Cayman",
>+  "America/Chicago",
>+  "America/Chihuahua",
>+  "America/Coral_Harbour",
>+  "America/Cordoba",
>+  "America/Costa_Rica",
>+  "America/Creston",
>+  "America/Cuiaba",
>+  "America/Curacao",
>+  "America/Danmarkshavn",
>+  "America/Dawson",
>+  "America/Dawson_Creek",
>+  "America/Denver",
>+  "America/Detroit",
>+  "America/Dominica",
>+  "America/Edmonton",
>+  "America/Eirunepe",
>+  "America/El_Salvador",
>+  "America/Ensenada",
>+  "America/Fort_Nelson",
>+  "America/Fort_Wayne",
>+  "America/Fortaleza",
>+  "America/Glace_Bay",
>+  "America/Godthab",
>+  "America/Goose_Bay",
>+  "America/Grand_Turk",
>+  "America/Grenada",
>+  "America/Guadeloupe",
>+  "America/Guatemala",
>+  "America/Guayaquil",
>+  "America/Guyana",
>+  "America/Halifax",
>+  "America/Havana",
>+  "America/Hermosillo",
>+  "America/Indiana/Indianapolis",
>+  "America/Indiana/Knox",
>+  "America/Indiana/Marengo",
>+  "America/Indiana/Petersburg",
>+  "America/Indiana/Tell_City",
>+  "America/Indiana/Vevay",
>+  "America/Indiana/Vincennes",
>+  "America/Indiana/Winamac",
>+  "America/Indianapolis",
>+  "America/Inuvik",
>+  "America/Iqaluit",
>+  "America/Jamaica",
>+  "America/Jujuy",
>+  "America/Juneau",
>+  "America/Kentucky/Louisville",
>+  "America/Kentucky/Monticello",
>+  "America/Knox_IN",
>+  "America/Kralendijk",
>+  "America/La_Paz",
>+  "America/Lima",
>+  "America/Los_Angeles",
>+  "America/Louisville",
>+  "America/Lower_Princes",
>+  "America/Maceio",
>+  "America/Managua",
>+  "America/Manaus",
>+  "America/Marigot",
>+  "America/Martinique",
>+  "America/Matamoros",
>+  "America/Mazatlan",
>+  "America/Mendoza",
>+  "America/Menominee",
>+  "America/Merida",
>+  "America/Metlakatla",
>+  "America/Mexico_City",
>+  "America/Miquelon",
>+  "America/Moncton",
>+  "America/Monterrey",
>+  "America/Montevideo",
>+  "America/Montreal",
>+  "America/Montserrat",
>+  "America/Nassau",
>+  "America/New_York",
>+  "America/Nipigon",
>+  "America/Nome",
>+  "America/Noronha",
>+  "America/North_Dakota/Beulah",
>+  "America/North_Dakota/Center",
>+  "America/North_Dakota/New_Salem",
>+  "America/Ojinaga",
>+  "America/Panama",
>+  "America/Pangnirtung",
>+  "America/Paramaribo",
>+  "America/Phoenix",
>+  "America/Port-au-Prince",
>+  "America/Port_of_Spain",
>+  "America/Porto_Acre",
>+  "America/Porto_Velho",
>+  "America/Puerto_Rico",
>+  "America/Punta_Arenas",
>+  "America/Rainy_River",
>+  "America/Rankin_Inlet",
>+  "America/Recife",
>+  "America/Regina",
>+  "America/Resolute",
>+  "America/Rio_Branco",
>+  "America/Rosario",
>+  "America/Santa_Isabel",
>+  "America/Santarem",
>+  "America/Santiago",
>+  "America/Santo_Domingo",
>+  "America/Sao_Paulo",
>+  "America/Scoresbysund",
>+  "America/Shiprock",
>+  "America/Sitka",
>+  "America/St_Barthelemy",
>+  "America/St_Johns",
>+  "America/St_Kitts",
>+  "America/St_Lucia",
>+  "America/St_Thomas",
>+  "America/St_Vincent",
>+  "America/Swift_Current",
>+  "America/Tegucigalpa",
>+  "America/Thule",
>+  "America/Thunder_Bay",
>+  "America/Tijuana",
>+  "America/Toronto",
>+  "America/Tortola",
>+  "America/Vancouver",
>+  "America/Virgin",
>+  "America/Whitehorse",
>+  "America/Winnipeg",
>+  "America/Yakutat",
>+  "America/Yellowknife",
>+  "Antarctica/Casey",
>+  "Antarctica/Davis",
>+  "Antarctica/DumontDUrville",
>+  "Antarctica/Macquarie",
>+  "Antarctica/Mawson",
>+  "Antarctica/McMurdo",
>+  "Antarctica/Palmer",
>+  "Antarctica/Rothera",
>+  "Antarctica/South_Pole",
>+  "Antarctica/Syowa",
>+  "Antarctica/Troll",
>+  "Antarctica/Vostok",
>+  "Arctic/Longyearbyen",
>+  "Asia/Aden",
>+  "Asia/Almaty",
>+  "Asia/Amman",
>+  "Asia/Anadyr",
>+  "Asia/Aqtau",
>+  "Asia/Aqtobe",
>+  "Asia/Ashgabat",
>+  "Asia/Ashkhabad",
>+  "Asia/Atyrau",
>+  "Asia/Baghdad",
>+  "Asia/Bahrain",
>+  "Asia/Baku",
>+  "Asia/Bangkok",
>+  "Asia/Barnaul",
>+  "Asia/Beirut",
>+  "Asia/Bishkek",
>+  "Asia/Brunei",
>+  "Asia/Calcutta",
>+  "Asia/Chita",
>+  "Asia/Choibalsan",
>+  "Asia/Chongqing",
>+  "Asia/Chungking",
>+  "Asia/Colombo",
>+  "Asia/Dacca",
>+  "Asia/Damascus",
>+  "Asia/Dhaka",
>+  "Asia/Dili",
>+  "Asia/Dubai",
>+  "Asia/Dushanbe",
>+  "Asia/Famagusta",
>+  "Asia/Gaza",
>+  "Asia/Harbin",
>+  "Asia/Hebron",
>+  "Asia/Ho_Chi_Minh",
>+  "Asia/Hong_Kong",
>+  "Asia/Hovd",
>+  "Asia/Irkutsk",
>+  "Asia/Istanbul",
>+  "Asia/Jakarta",
>+  "Asia/Jayapura",
>+  "Asia/Jerusalem",
>+  "Asia/Kabul",
>+  "Asia/Kamchatka",
>+  "Asia/Karachi",
>+  "Asia/Kashgar",
>+  "Asia/Kathmandu",
>+  "Asia/Katmandu",
>+  "Asia/Khandyga",
>+  "Asia/Kolkata",
>+  "Asia/Krasnoyarsk",
>+  "Asia/Kuala_Lumpur",
>+  "Asia/Kuching",
>+  "Asia/Kuwait",
>+  "Asia/Macao",
>+  "Asia/Macau",
>+  "Asia/Magadan",
>+  "Asia/Makassar",
>+  "Asia/Manila",
>+  "Asia/Muscat",
>+  "Asia/Nicosia",
>+  "Asia/Novokuznetsk",
>+  "Asia/Novosibirsk",
>+  "Asia/Omsk",
>+  "Asia/Oral",
>+  "Asia/Phnom_Penh",
>+  "Asia/Pontianak",
>+  "Asia/Pyongyang",
>+  "Asia/Qatar",
>+  "Asia/Qyzylorda",
>+  "Asia/Rangoon",
>+  "Asia/Riyadh",
>+  "Asia/Saigon",
>+  "Asia/Sakhalin",
>+  "Asia/Samarkand",
>+  "Asia/Seoul",
>+  "Asia/Shanghai",
>+  "Asia/Singapore",
>+  "Asia/Srednekolymsk",
>+  "Asia/Taipei",
>+  "Asia/Tashkent",
>+  "Asia/Tbilisi",
>+  "Asia/Tehran",
>+  "Asia/Tel_Aviv",
>+  "Asia/Thimbu",
>+  "Asia/Thimphu",
>+  "Asia/Tokyo",
>+  "Asia/Tomsk",
>+  "Asia/Ujung_Pandang",
>+  "Asia/Ulaanbaatar",
>+  "Asia/Ulan_Bator",
>+  "Asia/Urumqi",
>+  "Asia/Ust-Nera",
>+  "Asia/Vientiane",
>+  "Asia/Vladivostok",
>+  "Asia/Yakutsk",
>+  "Asia/Yangon",
>+  "Asia/Yekaterinburg",
>+  "Asia/Yerevan",
>+  "Atlantic/Azores",
>+  "Atlantic/Bermuda",
>+  "Atlantic/Canary",
>+  "Atlantic/Cape_Verde",
>+  "Atlantic/Faeroe",
>+  "Atlantic/Faroe",
>+  "Atlantic/Jan_Mayen",
>+  "Atlantic/Madeira",
>+  "Atlantic/Reykjavik",
>+  "Atlantic/South_Georgia",
>+  "Atlantic/St_Helena",
>+  "Atlantic/Stanley",
>+  "Australia/ACT",
>+  "Australia/Adelaide",
>+  "Australia/Brisbane",
>+  "Australia/Broken_Hill",
>+  "Australia/Canberra",
>+  "Australia/Currie",
>+  "Australia/Darwin",
>+  "Australia/Eucla",
>+  "Australia/Hobart",
>+  "Australia/LHI",
>+  "Australia/Lindeman",
>+  "Australia/Lord_Howe",
>+  "Australia/Melbourne",
>+  "Australia/NSW",
>+  "Australia/North",
>+  "Australia/Perth",
>+  "Australia/Queensland",
>+  "Australia/South",
>+  "Australia/Sydney",
>+  "Australia/Tasmania",
>+  "Australia/Victoria",
>+  "Australia/West",
>+  "Australia/Yancowinna",
>+  "Brazil/Acre",
>+  "Brazil/DeNoronha",
>+  "Brazil/East",
>+  "Brazil/West",
>+  "CET",
>+  "CST6CDT",
>+  "Canada/Atlantic",
>+  "Canada/Central",
>+  "Canada/Eastern",
>+  "Canada/Mountain",
>+  "Canada/Newfoundland",
>+  "Canada/Pacific",
>+  "Canada/Saskatchewan",
>+  "Canada/Yukon",
>+  "Chile/Continental",
>+  "Chile/EasterIsland",
>+  "Cuba",
>+  "EET",
>+  "EST",
>+  "EST5EDT",
>+  "Egypt",
>+  "Eire",
>+  "Etc/GMT",
>+  "Etc/GMT+0",
>+  "Etc/GMT+1",
>+  "Etc/GMT+10",
>+  "Etc/GMT+11",
>+  "Etc/GMT+12",
>+  "Etc/GMT+2",
>+  "Etc/GMT+3",
>+  "Etc/GMT+4",
>+  "Etc/GMT+5",
>+  "Etc/GMT+6",
>+  "Etc/GMT+7",
>+  "Etc/GMT+8",
>+  "Etc/GMT+9",
>+  "Etc/GMT-0",
>+  "Etc/GMT-1",
>+  "Etc/GMT-10",
>+  "Etc/GMT-11",
>+  "Etc/GMT-12",
>+  "Etc/GMT-13",
>+  "Etc/GMT-14",
>+  "Etc/GMT-2",
>+  "Etc/GMT-3",
>+  "Etc/GMT-4",
>+  "Etc/GMT-5",
>+  "Etc/GMT-6",
>+  "Etc/GMT-7",
>+  "Etc/GMT-8",
>+  "Etc/GMT-9",
>+  "Etc/GMT0",
>+  "Etc/Greenwich",
>+  "Etc/UCT",
>+  "Etc/UTC",
>+  "Etc/Universal",
>+  "Etc/Zulu",
>+  "Europe/Amsterdam",
>+  "Europe/Andorra",
>+  "Europe/Astrakhan",
>+  "Europe/Athens",
>+  "Europe/Belfast",
>+  "Europe/Belgrade",
>+  "Europe/Berlin",
>+  "Europe/Bratislava",
>+  "Europe/Brussels",
>+  "Europe/Bucharest",
>+  "Europe/Budapest",
>+  "Europe/Busingen",
>+  "Europe/Chisinau",
>+  "Europe/Copenhagen",
>+  "Europe/Dublin",
>+  "Europe/Gibraltar",
>+  "Europe/Guernsey",
>+  "Europe/Helsinki",
>+  "Europe/Isle_of_Man",
>+  "Europe/Istanbul",
>+  "Europe/Jersey",
>+  "Europe/Kaliningrad",
>+  "Europe/Kiev",
>+  "Europe/Kirov",
>+  "Europe/Lisbon",
>+  "Europe/Ljubljana",
>+  "Europe/London",
>+  "Europe/Luxembourg",
>+  "Europe/Madrid",
>+  "Europe/Malta",
>+  "Europe/Mariehamn",
>+  "Europe/Minsk",
>+  "Europe/Monaco",
>+  "Europe/Moscow",
>+  "Europe/Nicosia",
>+  "Europe/Oslo",
>+  "Europe/Paris",
>+  "Europe/Podgorica",
>+  "Europe/Prague",
>+  "Europe/Riga",
>+  "Europe/Rome",
>+  "Europe/Samara",
>+  "Europe/San_Marino",
>+  "Europe/Sarajevo",
>+  "Europe/Saratov",
>+  "Europe/Simferopol",
>+  "Europe/Skopje",
>+  "Europe/Sofia",
>+  "Europe/Stockholm",
>+  "Europe/Tallinn",
>+  "Europe/Tirane",
>+  "Europe/Tiraspol",
>+  "Europe/Ulyanovsk",
>+  "Europe/Uzhgorod",
>+  "Europe/Vaduz",
>+  "Europe/Vatican",
>+  "Europe/Vienna",
>+  "Europe/Vilnius",
>+  "Europe/Volgograd",
>+  "Europe/Warsaw",
>+  "Europe/Zagreb",
>+  "Europe/Zaporozhye",
>+  "Europe/Zurich",
>+  "GB",
>+  "GB-Eire",
>+  "GMT",
>+  "GMT+0",
>+  "GMT-0",
>+  "GMT0",
>+  "Greenwich",
>+  "HST",
>+  "Hongkong",
>+  "Iceland",
>+  "Indian/Antananarivo",
>+  "Indian/Chagos",
>+  "Indian/Christmas",
>+  "Indian/Cocos",
>+  "Indian/Comoro",
>+  "Indian/Kerguelen",
>+  "Indian/Mahe",
>+  "Indian/Maldives",
>+  "Indian/Mauritius",
>+  "Indian/Mayotte",
>+  "Indian/Reunion",
>+  "Iran",
>+  "Israel",
>+  "Jamaica",
>+  "Japan",
>+  "Kwajalein",
>+  "Libya",
>+  "MET",
>+  "MST",
>+  "MST7MDT",
>+  "Mexico/BajaNorte",
>+  "Mexico/BajaSur",
>+  "Mexico/General",
>+  "NZ",
>+  "NZ-CHAT",
>+  "Navajo",
>+  "PRC",
>+  "PST8PDT",
>+  "Pacific/Apia",
>+  "Pacific/Auckland",
>+  "Pacific/Bougainville",
>+  "Pacific/Chatham",
>+  "Pacific/Chuuk",
>+  "Pacific/Easter",
>+  "Pacific/Efate",
>+  "Pacific/Enderbury",
>+  "Pacific/Fakaofo",
>+  "Pacific/Fiji",
>+  "Pacific/Funafuti",
>+  "Pacific/Galapagos",
>+  "Pacific/Gambier",
>+  "Pacific/Guadalcanal",
>+  "Pacific/Guam",
>+  "Pacific/Honolulu",
>+  "Pacific/Johnston",
>+  "Pacific/Kiritimati",
>+  "Pacific/Kosrae",
>+  "Pacific/Kwajalein",
>+  "Pacific/Majuro",
>+  "Pacific/Marquesas",
>+  "Pacific/Midway",
>+  "Pacific/Nauru",
>+  "Pacific/Niue",
>+  "Pacific/Norfolk",
>+  "Pacific/Noumea",
>+  "Pacific/Pago_Pago",
>+  "Pacific/Palau",
>+  "Pacific/Pitcairn",
>+  "Pacific/Pohnpei",
>+  "Pacific/Ponape",
>+  "Pacific/Port_Moresby",
>+  "Pacific/Rarotonga",
>+  "Pacific/Saipan",
>+  "Pacific/Samoa",
>+  "Pacific/Tahiti",
>+  "Pacific/Tarawa",
>+  "Pacific/Tongatapu",
>+  "Pacific/Truk",
>+  "Pacific/Wake",
>+  "Pacific/Wallis",
>+  "Pacific/Yap",
>+  "Poland",
>+  "Portugal",
>+  "ROC",
>+  "ROK",
>+  "Singapore",
>+  "Turkey",
>+  "UCT",
>+  "US/Alaska",
>+  "US/Aleutian",
>+  "US/Arizona",
>+  "US/Central",
>+  "US/East-Indiana",
>+  "US/Eastern",
>+  "US/Hawaii",
>+  "US/Indiana-Starke",
>+  "US/Michigan",
>+  "US/Mountain",
>+  "US/Pacific",
>+  "US/Samoa",
>+  "UTC",
>+  "Universal",
>+  "W-SU",
>+  "WET",
>+  "Zulu",
>+  nullptr
>+};
>+
>+std::vector<std::string> AllTimeZoneNames() {
>+  std::vector<std::string> names;
>+  for (const char* const* namep = kTimeZoneNames; *namep != nullptr; ++namep) {
>+    names.push_back(std::string("file:") + *namep);
>+  }
>+  assert(!names.empty());
>+
>+  std::mt19937 urbg(42);  // a UniformRandomBitGenerator with fixed seed
>+  std::shuffle(names.begin(), names.end(), urbg);
>+  return names;
>+}
>+
>+cctz::time_zone TestTimeZone() {
>+  cctz::time_zone tz;
>+  cctz::load_time_zone("America/Los_Angeles", &tz);
>+  return tz;
>+}
>+
>+void BM_Zone_LoadUTCTimeZoneFirst(benchmark::State& state) {
>+  cctz::time_zone tz;
>+  cctz::load_time_zone("UTC", &tz);  // in case we're first
>+  cctz::time_zone::Impl::ClearTimeZoneMapTestOnly();
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(cctz::load_time_zone("UTC", &tz));
>+  }
>+}
>+BENCHMARK(BM_Zone_LoadUTCTimeZoneFirst);
>+
>+void BM_Zone_LoadUTCTimeZoneLast(benchmark::State& state) {
>+  cctz::time_zone tz;
>+  for (const auto& name : AllTimeZoneNames()) {
>+    cctz::load_time_zone(name, &tz);  // prime cache
>+  }
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(cctz::load_time_zone("UTC", &tz));
>+  }
>+}
>+BENCHMARK(BM_Zone_LoadUTCTimeZoneLast);
>+
>+void BM_Zone_LoadTimeZoneFirst(benchmark::State& state) {
>+  cctz::time_zone tz = cctz::utc_time_zone();  // in case we're first
>+  const std::string name = "file:America/Los_Angeles";
>+  while (state.KeepRunning()) {
>+    state.PauseTiming();
>+    cctz::time_zone::Impl::ClearTimeZoneMapTestOnly();
>+    state.ResumeTiming();
>+    benchmark::DoNotOptimize(cctz::load_time_zone(name, &tz));
>+  }
>+}
>+BENCHMARK(BM_Zone_LoadTimeZoneFirst);
>+
>+void BM_Zone_LoadTimeZoneCached(benchmark::State& state) {
>+  cctz::time_zone tz = cctz::utc_time_zone();  // in case we're first
>+  cctz::time_zone::Impl::ClearTimeZoneMapTestOnly();
>+  const std::string name = "file:America/Los_Angeles";
>+  cctz::load_time_zone(name, &tz);  // prime cache
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(cctz::load_time_zone(name, &tz));
>+  }
>+}
>+BENCHMARK(BM_Zone_LoadTimeZoneCached);
>+
>+void BM_Zone_LoadLocalTimeZoneCached(benchmark::State& state) {
>+  cctz::utc_time_zone();  // in case we're first
>+  cctz::time_zone::Impl::ClearTimeZoneMapTestOnly();
>+  cctz::local_time_zone();  // prime cache
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(cctz::local_time_zone());
>+  }
>+}
>+BENCHMARK(BM_Zone_LoadLocalTimeZoneCached);
>+
>+void BM_Zone_LoadAllTimeZonesFirst(benchmark::State& state) {
>+  cctz::time_zone tz;
>+  const std::vector<std::string> names = AllTimeZoneNames();
>+  for (auto index = names.size(); state.KeepRunning(); ++index) {
>+    if (index == names.size()) {
>+      index = 0;
>+    }
>+    if (index == 0) {
>+      state.PauseTiming();
>+      cctz::time_zone::Impl::ClearTimeZoneMapTestOnly();
>+      state.ResumeTiming();
>+    }
>+    benchmark::DoNotOptimize(cctz::load_time_zone(names[index], &tz));
>+  }
>+}
>+BENCHMARK(BM_Zone_LoadAllTimeZonesFirst);
>+
>+void BM_Zone_LoadAllTimeZonesCached(benchmark::State& state) {
>+  cctz::time_zone tz;
>+  const std::vector<std::string> names = AllTimeZoneNames();
>+  for (const auto& name : names) {
>+    cctz::load_time_zone(name, &tz);  // prime cache
>+  }
>+  for (auto index = names.size(); state.KeepRunning(); ++index) {
>+    if (index == names.size()) {
>+      index = 0;
>+    }
>+    benchmark::DoNotOptimize(cctz::load_time_zone(names[index], &tz));
>+  }
>+}
>+BENCHMARK(BM_Zone_LoadAllTimeZonesCached);
>+
>+void BM_Zone_TimeZoneEqualityImplicit(benchmark::State& state) {
>+  cctz::time_zone tz;  // implicit UTC
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(tz == tz);
>+  }
>+}
>+BENCHMARK(BM_Zone_TimeZoneEqualityImplicit);
>+
>+void BM_Zone_TimeZoneEqualityExplicit(benchmark::State& state) {
>+  cctz::time_zone tz = cctz::utc_time_zone();  // explicit UTC
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(tz == tz);
>+  }
>+}
>+BENCHMARK(BM_Zone_TimeZoneEqualityExplicit);
>+
>+void BM_Zone_UTCTimeZone(benchmark::State& state) {
>+  cctz::time_zone tz;
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(cctz::utc_time_zone());
>+  }
>+}
>+BENCHMARK(BM_Zone_UTCTimeZone);
>+
>+// In each "ToDateTime" benchmark we switch between two instants
>+// separated by at least one transition in order to defeat any
>+// internal caching of previous results (e.g., see local_time_hint_).
>+//
>+// The "UTC" variants use UTC instead of the Google/local time zone.
>+
>+void BM_Time_ToDateTime_CCTZ(benchmark::State& state) {
>+  const cctz::time_zone tz = TestTimeZone();
>+  std::chrono::system_clock::time_point tp =
>+      std::chrono::system_clock::from_time_t(1384569027);
>+  std::chrono::system_clock::time_point tp2 =
>+      std::chrono::system_clock::from_time_t(1418962578);
>+  while (state.KeepRunning()) {
>+    std::swap(tp, tp2);
>+    tp += std::chrono::seconds(1);
>+    benchmark::DoNotOptimize(cctz::convert(tp, tz));
>+  }
>+}
>+BENCHMARK(BM_Time_ToDateTime_CCTZ);
>+
>+void BM_Time_ToDateTime_Libc(benchmark::State& state) {
>+  // No timezone support, so just use localtime.
>+  time_t t = 1384569027;
>+  time_t t2 = 1418962578;
>+  struct tm tm;
>+  while (state.KeepRunning()) {
>+    std::swap(t, t2);
>+    t += 1;
>+#if defined(_WIN32) || defined(_WIN64)
>+    benchmark::DoNotOptimize(localtime_s(&tm, &t));
>+#else
>+    benchmark::DoNotOptimize(localtime_r(&t, &tm));
>+#endif
>+  }
>+}
>+BENCHMARK(BM_Time_ToDateTime_Libc);
>+
>+void BM_Time_ToDateTimeUTC_CCTZ(benchmark::State& state) {
>+  const cctz::time_zone tz = cctz::utc_time_zone();
>+  std::chrono::system_clock::time_point tp =
>+      std::chrono::system_clock::from_time_t(1384569027);
>+  while (state.KeepRunning()) {
>+    tp += std::chrono::seconds(1);
>+    benchmark::DoNotOptimize(cctz::convert(tp, tz));
>+  }
>+}
>+BENCHMARK(BM_Time_ToDateTimeUTC_CCTZ);
>+
>+void BM_Time_ToDateTimeUTC_Libc(benchmark::State& state) {
>+  time_t t = 1384569027;
>+  struct tm tm;
>+  while (state.KeepRunning()) {
>+    t += 1;
>+#if defined(_WIN32) || defined(_WIN64)
>+    benchmark::DoNotOptimize(gmtime_s(&tm, &t));
>+#else
>+    benchmark::DoNotOptimize(gmtime_r(&t, &tm));
>+#endif
>+  }
>+}
>+BENCHMARK(BM_Time_ToDateTimeUTC_Libc);
>+
>+// In each "FromDateTime" benchmark we switch between two YMDhms
>+// values separated by at least one transition in order to defeat any
>+// internal caching of previous results (e.g., see time_local_hint_).
>+//
>+// The "UTC" variants use UTC instead of the Google/local time zone.
>+// The "Day0" variants require normalization of the day of month.
>+
>+void BM_Time_FromDateTime_CCTZ(benchmark::State& state) {
>+  const cctz::time_zone tz = TestTimeZone();
>+  int i = 0;
>+  while (state.KeepRunning()) {
>+    if ((i++ & 1) == 0) {
>+      benchmark::DoNotOptimize(
>+          cctz::convert(cctz::civil_second(2014, 12, 18, 20, 16, 18), tz));
>+    } else {
>+      benchmark::DoNotOptimize(
>+          cctz::convert(cctz::civil_second(2013, 11, 15, 18, 30, 27), tz));
>+    }
>+  }
>+}
>+BENCHMARK(BM_Time_FromDateTime_CCTZ);
>+
>+void BM_Time_FromDateTime_Libc(benchmark::State& state) {
>+  // No timezone support, so just use localtime.
>+  int i = 0;
>+  while (state.KeepRunning()) {
>+    struct tm tm;
>+    if ((i++ & 1) == 0) {
>+      tm.tm_year = 2014 - 1900;
>+      tm.tm_mon = 12 - 1;
>+      tm.tm_mday = 18;
>+      tm.tm_hour = 20;
>+      tm.tm_min = 16;
>+      tm.tm_sec = 18;
>+    } else {
>+      tm.tm_year = 2013 - 1900;
>+      tm.tm_mon = 11 - 1;
>+      tm.tm_mday = 15;
>+      tm.tm_hour = 18;
>+      tm.tm_min = 30;
>+      tm.tm_sec = 27;
>+    }
>+    tm.tm_isdst = -1;
>+    benchmark::DoNotOptimize(mktime(&tm));
>+  }
>+}
>+BENCHMARK(BM_Time_FromDateTime_Libc);
>+
>+void BM_Time_FromDateTimeUTC_CCTZ(benchmark::State& state) {
>+  const cctz::time_zone tz = cctz::utc_time_zone();
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(
>+        cctz::convert(cctz::civil_second(2014, 12, 18, 20, 16, 18), tz));
>+  }
>+}
>+BENCHMARK(BM_Time_FromDateTimeUTC_CCTZ);
>+
>+// There is no BM_Time_FromDateTimeUTC_Libc.
>+
>+void BM_Time_FromDateTimeDay0_CCTZ(benchmark::State& state) {
>+  const cctz::time_zone tz = TestTimeZone();
>+  int i = 0;
>+  while (state.KeepRunning()) {
>+    if ((i++ & 1) == 0) {
>+      benchmark::DoNotOptimize(
>+          cctz::convert(cctz::civil_second(2014, 12, 0, 20, 16, 18), tz));
>+    } else {
>+      benchmark::DoNotOptimize(
>+          cctz::convert(cctz::civil_second(2013, 11, 0, 18, 30, 27), tz));
>+    }
>+  }
>+}
>+BENCHMARK(BM_Time_FromDateTimeDay0_CCTZ);
>+
>+void BM_Time_FromDateTimeDay0_Libc(benchmark::State& state) {
>+  // No timezone support, so just use localtime.
>+  int i = 0;
>+  while (state.KeepRunning()) {
>+    struct tm tm;
>+    if ((i++ & 1) == 0) {
>+      tm.tm_year = 2014 - 1900;
>+      tm.tm_mon = 12 - 1;
>+      tm.tm_mday = 0;
>+      tm.tm_hour = 20;
>+      tm.tm_min = 16;
>+      tm.tm_sec = 18;
>+    } else {
>+      tm.tm_year = 2013 - 1900;
>+      tm.tm_mon = 11 - 1;
>+      tm.tm_mday = 0;
>+      tm.tm_hour = 18;
>+      tm.tm_min = 30;
>+      tm.tm_sec = 27;
>+    }
>+    tm.tm_isdst = -1;
>+    benchmark::DoNotOptimize(mktime(&tm));
>+  }
>+}
>+BENCHMARK(BM_Time_FromDateTimeDay0_Libc);
>+
>+const char* const kFormats[] = {
>+    RFC1123_full,         // 0
>+    RFC1123_no_wday,      // 1
>+    RFC3339_full,         // 2
>+    RFC3339_sec,          // 3
>+    "%Y-%m-%dT%H:%M:%S",  // 4
>+    "%Y-%m-%d",           // 5
>+};
>+const int kNumFormats = sizeof(kFormats) / sizeof(kFormats[0]);
>+
>+void BM_Format_FormatTime(benchmark::State& state) {
>+  const std::string fmt = kFormats[state.range(0)];
>+  state.SetLabel(fmt);
>+  const cctz::time_zone tz = TestTimeZone();
>+  const std::chrono::system_clock::time_point tp =
>+      cctz::convert(cctz::civil_second(1977, 6, 28, 9, 8, 7), tz) +
>+      std::chrono::microseconds(1);
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(cctz::format(fmt, tp, tz));
>+  }
>+}
>+BENCHMARK(BM_Format_FormatTime)->DenseRange(0, kNumFormats - 1);
>+
>+void BM_Format_ParseTime(benchmark::State& state) {
>+  const std::string fmt = kFormats[state.range(0)];
>+  state.SetLabel(fmt);
>+  const cctz::time_zone tz = TestTimeZone();
>+  std::chrono::system_clock::time_point tp =
>+      cctz::convert(cctz::civil_second(1977, 6, 28, 9, 8, 7), tz) +
>+      std::chrono::microseconds(1);
>+  const std::string when = cctz::format(fmt, tp, tz);
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(cctz::parse(fmt, when, tz, &tp));
>+  }
>+}
>+BENCHMARK(BM_Format_ParseTime)->DenseRange(0, kNumFormats - 1);
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/civil_time_detail.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/civil_time_detail.cc
>new file mode 100644
>index 00000000000..780d5c96e8b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/civil_time_detail.cc
>@@ -0,0 +1,90 @@
>+// Copyright 2016 Google Inc. All Rights Reserved.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//   http://www.apache.org/licenses/LICENSE-2.0
>+//
>+//   Unless required by applicable law or agreed to in writing, software
>+//   distributed under the License is distributed on an "AS IS" BASIS,
>+//   WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+//   See the License for the specific language governing permissions and
>+//   limitations under the License.
>+
>+#include "absl/time/internal/cctz/include/cctz/civil_time_detail.h"
>+
>+#include <iomanip>
>+#include <ostream>
>+#include <sstream>
>+
>+namespace absl {
>+namespace time_internal {
>+namespace cctz {
>+namespace detail {
>+
>+// Output stream operators output a format matching YYYY-MM-DDThh:mm:ss,
>+// while omitting fields inferior to the type's alignment. For example,
>+// civil_day is formatted only as YYYY-MM-DD.
>+std::ostream& operator<<(std::ostream& os, const civil_year& y) {
>+  std::stringstream ss;
>+  ss << y.year();  // No padding.
>+  return os << ss.str();
>+}
>+std::ostream& operator<<(std::ostream& os, const civil_month& m) {
>+  std::stringstream ss;
>+  ss << civil_year(m) << '-';
>+  ss << std::setfill('0') << std::setw(2) << m.month();
>+  return os << ss.str();
>+}
>+std::ostream& operator<<(std::ostream& os, const civil_day& d) {
>+  std::stringstream ss;
>+  ss << civil_month(d) << '-';
>+  ss << std::setfill('0') << std::setw(2) << d.day();
>+  return os << ss.str();
>+}
>+std::ostream& operator<<(std::ostream& os, const civil_hour& h) {
>+  std::stringstream ss;
>+  ss << civil_day(h) << 'T';
>+  ss << std::setfill('0') << std::setw(2) << h.hour();
>+  return os << ss.str();
>+}
>+std::ostream& operator<<(std::ostream& os, const civil_minute& m) {
>+  std::stringstream ss;
>+  ss << civil_hour(m) << ':';
>+  ss << std::setfill('0') << std::setw(2) << m.minute();
>+  return os << ss.str();
>+}
>+std::ostream& operator<<(std::ostream& os, const civil_second& s) {
>+  std::stringstream ss;
>+  ss << civil_minute(s) << ':';
>+  ss << std::setfill('0') << std::setw(2) << s.second();
>+  return os << ss.str();
>+}
>+
>+////////////////////////////////////////////////////////////////////////
>+
>+std::ostream& operator<<(std::ostream& os, weekday wd) {
>+  switch (wd) {
>+    case weekday::monday:
>+      return os << "Monday";
>+    case weekday::tuesday:
>+      return os << "Tuesday";
>+    case weekday::wednesday:
>+      return os << "Wednesday";
>+    case weekday::thursday:
>+      return os << "Thursday";
>+    case weekday::friday:
>+      return os << "Friday";
>+    case weekday::saturday:
>+      return os << "Saturday";
>+    case weekday::sunday:
>+      return os << "Sunday";
>+  }
>+  return os;  // Should never get here, but -Wreturn-type may warn without this.
>+}
>+
>+}  // namespace detail
>+}  // namespace cctz
>+}  // namespace time_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/civil_time_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/civil_time_test.cc
>new file mode 100644
>index 00000000000..f6648c8f1f2
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/civil_time_test.cc
>@@ -0,0 +1,1049 @@
>+// Copyright 2016 Google Inc. All Rights Reserved.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//   http://www.apache.org/licenses/LICENSE-2.0
>+//
>+//   Unless required by applicable law or agreed to in writing, software
>+//   distributed under the License is distributed on an "AS IS" BASIS,
>+//   WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+//   See the License for the specific language governing permissions and
>+//   limitations under the License.
>+
>+#include "absl/time/internal/cctz/include/cctz/civil_time.h"
>+
>+#include <iomanip>
>+#include <limits>
>+#include <sstream>
>+#include <string>
>+#include <type_traits>
>+
>+#include "gtest/gtest.h"
>+
>+namespace absl {
>+namespace time_internal {
>+namespace cctz {
>+
>+namespace {
>+
>+template <typename T>
>+std::string Format(const T& t) {
>+  std::stringstream ss;
>+  ss << t;
>+  return ss.str();
>+}
>+
>+}  // namespace
>+
>+#if __cpp_constexpr >= 201304 || _MSC_VER >= 1910
>+// Construction constexpr tests
>+
>+TEST(CivilTime, Normal) {
>+  constexpr civil_second css(2016, 1, 28, 17, 14, 12);
>+  static_assert(css.second() == 12, "Normal.second");
>+  constexpr civil_minute cmm(2016, 1, 28, 17, 14);
>+  static_assert(cmm.minute() == 14, "Normal.minute");
>+  constexpr civil_hour chh(2016, 1, 28, 17);
>+  static_assert(chh.hour() == 17, "Normal.hour");
>+  constexpr civil_day cd(2016, 1, 28);
>+  static_assert(cd.day() == 28, "Normal.day");
>+  constexpr civil_month cm(2016, 1);
>+  static_assert(cm.month() == 1, "Normal.month");
>+  constexpr civil_year cy(2016);
>+  static_assert(cy.year() == 2016, "Normal.year");
>+}
>+
>+TEST(CivilTime, Conversion) {
>+  constexpr civil_year cy(2016);
>+  static_assert(cy.year() == 2016, "Conversion.year");
>+  constexpr civil_month cm(cy);
>+  static_assert(cm.month() == 1, "Conversion.month");
>+  constexpr civil_day cd(cm);
>+  static_assert(cd.day() == 1, "Conversion.day");
>+  constexpr civil_hour chh(cd);
>+  static_assert(chh.hour() == 0, "Conversion.hour");
>+  constexpr civil_minute cmm(chh);
>+  static_assert(cmm.minute() == 0, "Conversion.minute");
>+  constexpr civil_second css(cmm);
>+  static_assert(css.second() == 0, "Conversion.second");
>+}
>+
>+// Normalization constexpr tests
>+
>+TEST(CivilTime, Normalized) {
>+  constexpr civil_second cs(2016, 1, 28, 17, 14, 12);
>+  static_assert(cs.year() == 2016, "Normalized.year");
>+  static_assert(cs.month() == 1, "Normalized.month");
>+  static_assert(cs.day() == 28, "Normalized.day");
>+  static_assert(cs.hour() == 17, "Normalized.hour");
>+  static_assert(cs.minute() == 14, "Normalized.minute");
>+  static_assert(cs.second() == 12, "Normalized.second");
>+}
>+
>+TEST(CivilTime, SecondOverflow) {
>+  constexpr civil_second cs(2016, 1, 28, 17, 14, 121);
>+  static_assert(cs.year() == 2016, "SecondOverflow.year");
>+  static_assert(cs.month() == 1, "SecondOverflow.month");
>+  static_assert(cs.day() == 28, "SecondOverflow.day");
>+  static_assert(cs.hour() == 17, "SecondOverflow.hour");
>+  static_assert(cs.minute() == 16, "SecondOverflow.minute");
>+  static_assert(cs.second() == 1, "SecondOverflow.second");
>+}
>+
>+TEST(CivilTime, SecondUnderflow) {
>+  constexpr civil_second cs(2016, 1, 28, 17, 14, -121);
>+  static_assert(cs.year() == 2016, "SecondUnderflow.year");
>+  static_assert(cs.month() == 1, "SecondUnderflow.month");
>+  static_assert(cs.day() == 28, "SecondUnderflow.day");
>+  static_assert(cs.hour() == 17, "SecondUnderflow.hour");
>+  static_assert(cs.minute() == 11, "SecondUnderflow.minute");
>+  static_assert(cs.second() == 59, "SecondUnderflow.second");
>+}
>+
>+TEST(CivilTime, MinuteOverflow) {
>+  constexpr civil_second cs(2016, 1, 28, 17, 121, 12);
>+  static_assert(cs.year() == 2016, "MinuteOverflow.year");
>+  static_assert(cs.month() == 1, "MinuteOverflow.month");
>+  static_assert(cs.day() == 28, "MinuteOverflow.day");
>+  static_assert(cs.hour() == 19, "MinuteOverflow.hour");
>+  static_assert(cs.minute() == 1, "MinuteOverflow.minute");
>+  static_assert(cs.second() == 12, "MinuteOverflow.second");
>+}
>+
>+TEST(CivilTime, MinuteUnderflow) {
>+  constexpr civil_second cs(2016, 1, 28, 17, -121, 12);
>+  static_assert(cs.year() == 2016, "MinuteUnderflow.year");
>+  static_assert(cs.month() == 1, "MinuteUnderflow.month");
>+  static_assert(cs.day() == 28, "MinuteUnderflow.day");
>+  static_assert(cs.hour() == 14, "MinuteUnderflow.hour");
>+  static_assert(cs.minute() == 59, "MinuteUnderflow.minute");
>+  static_assert(cs.second() == 12, "MinuteUnderflow.second");
>+}
>+
>+TEST(CivilTime, HourOverflow) {
>+  constexpr civil_second cs(2016, 1, 28, 49, 14, 12);
>+  static_assert(cs.year() == 2016, "HourOverflow.year");
>+  static_assert(cs.month() == 1, "HourOverflow.month");
>+  static_assert(cs.day() == 30, "HourOverflow.day");
>+  static_assert(cs.hour() == 1, "HourOverflow.hour");
>+  static_assert(cs.minute() == 14, "HourOverflow.minute");
>+  static_assert(cs.second() == 12, "HourOverflow.second");
>+}
>+
>+TEST(CivilTime, HourUnderflow) {
>+  constexpr civil_second cs(2016, 1, 28, -49, 14, 12);
>+  static_assert(cs.year() == 2016, "HourUnderflow.year");
>+  static_assert(cs.month() == 1, "HourUnderflow.month");
>+  static_assert(cs.day() == 25, "HourUnderflow.day");
>+  static_assert(cs.hour() == 23, "HourUnderflow.hour");
>+  static_assert(cs.minute() == 14, "HourUnderflow.minute");
>+  static_assert(cs.second() == 12, "HourUnderflow.second");
>+}
>+
>+TEST(CivilTime, MonthOverflow) {
>+  constexpr civil_second cs(2016, 25, 28, 17, 14, 12);
>+  static_assert(cs.year() == 2018, "MonthOverflow.year");
>+  static_assert(cs.month() == 1, "MonthOverflow.month");
>+  static_assert(cs.day() == 28, "MonthOverflow.day");
>+  static_assert(cs.hour() == 17, "MonthOverflow.hour");
>+  static_assert(cs.minute() == 14, "MonthOverflow.minute");
>+  static_assert(cs.second() == 12, "MonthOverflow.second");
>+}
>+
>+TEST(CivilTime, MonthUnderflow) {
>+  constexpr civil_second cs(2016, -25, 28, 17, 14, 12);
>+  static_assert(cs.year() == 2013, "MonthUnderflow.year");
>+  static_assert(cs.month() == 11, "MonthUnderflow.month");
>+  static_assert(cs.day() == 28, "MonthUnderflow.day");
>+  static_assert(cs.hour() == 17, "MonthUnderflow.hour");
>+  static_assert(cs.minute() == 14, "MonthUnderflow.minute");
>+  static_assert(cs.second() == 12, "MonthUnderflow.second");
>+}
>+
>+TEST(CivilTime, C4Overflow) {
>+  constexpr civil_second cs(2016, 1, 292195, 17, 14, 12);
>+  static_assert(cs.year() == 2816, "C4Overflow.year");
>+  static_assert(cs.month() == 1, "C4Overflow.month");
>+  static_assert(cs.day() == 1, "C4Overflow.day");
>+  static_assert(cs.hour() == 17, "C4Overflow.hour");
>+  static_assert(cs.minute() == 14, "C4Overflow.minute");
>+  static_assert(cs.second() == 12, "C4Overflow.second");
>+}
>+
>+TEST(CivilTime, C4Underflow) {
>+  constexpr civil_second cs(2016, 1, -292195, 17, 14, 12);
>+  static_assert(cs.year() == 1215, "C4Underflow.year");
>+  static_assert(cs.month() == 12, "C4Underflow.month");
>+  static_assert(cs.day() == 30, "C4Underflow.day");
>+  static_assert(cs.hour() == 17, "C4Underflow.hour");
>+  static_assert(cs.minute() == 14, "C4Underflow.minute");
>+  static_assert(cs.second() == 12, "C4Underflow.second");
>+}
>+
>+TEST(CivilTime, MixedNormalization) {
>+  constexpr civil_second cs(2016, -42, 122, 99, -147, 4949);
>+  static_assert(cs.year() == 2012, "MixedNormalization.year");
>+  static_assert(cs.month() == 10, "MixedNormalization.month");
>+  static_assert(cs.day() == 4, "MixedNormalization.day");
>+  static_assert(cs.hour() == 1, "MixedNormalization.hour");
>+  static_assert(cs.minute() == 55, "MixedNormalization.minute");
>+  static_assert(cs.second() == 29, "MixedNormalization.second");
>+}
>+
>+// Relational constexpr tests
>+
>+TEST(CivilTime, Less) {
>+  constexpr civil_second cs1(2016, 1, 28, 17, 14, 12);
>+  constexpr civil_second cs2(2016, 1, 28, 17, 14, 13);
>+  constexpr bool less = cs1 < cs2;
>+  static_assert(less, "Less");
>+}
>+
>+// Arithmetic constexpr tests
>+
>+TEST(CivilTime, Addition) {
>+  constexpr civil_second cs1(2016, 1, 28, 17, 14, 12);
>+  constexpr civil_second cs2 = cs1 + 50;
>+  static_assert(cs2.year() == 2016, "Addition.year");
>+  static_assert(cs2.month() == 1, "Addition.month");
>+  static_assert(cs2.day() == 28, "Addition.day");
>+  static_assert(cs2.hour() == 17, "Addition.hour");
>+  static_assert(cs2.minute() == 15, "Addition.minute");
>+  static_assert(cs2.second() == 2, "Addition.second");
>+}
>+
>+TEST(CivilTime, Subtraction) {
>+  constexpr civil_second cs1(2016, 1, 28, 17, 14, 12);
>+  constexpr civil_second cs2 = cs1 - 50;
>+  static_assert(cs2.year() == 2016, "Subtraction.year");
>+  static_assert(cs2.month() == 1, "Subtraction.month");
>+  static_assert(cs2.day() == 28, "Subtraction.day");
>+  static_assert(cs2.hour() == 17, "Subtraction.hour");
>+  static_assert(cs2.minute() == 13, "Subtraction.minute");
>+  static_assert(cs2.second() == 22, "Subtraction.second");
>+}
>+
>+TEST(CivilTime, Difference) {
>+  constexpr civil_day cd1(2016, 1, 28);
>+  constexpr civil_day cd2(2015, 1, 28);
>+  constexpr int diff = cd1 - cd2;
>+  static_assert(diff == 365, "Difference");
>+}
>+
>+// NOTE: Run this with --copt=-ftrapv to detect overflow problems.
>+TEST(CivilTime, DifferenceWithHugeYear) {
>+  {
>+    constexpr civil_day d1(9223372036854775807, 1, 1);
>+    constexpr civil_day d2(9223372036854775807, 12, 31);
>+    static_assert(d2 - d1 == 364, "DifferenceWithHugeYear");
>+  }
>+  {
>+    constexpr civil_day d1(-9223372036854775807 - 1, 1, 1);
>+    constexpr civil_day d2(-9223372036854775807 - 1, 12, 31);
>+    static_assert(d2 - d1 == 365, "DifferenceWithHugeYear");
>+  }
>+  {
>+    // Check the limits of the return value at the end of the year range.
>+    constexpr civil_day d1(9223372036854775807, 1, 1);
>+    constexpr civil_day d2(9198119301927009252, 6, 6);
>+    static_assert(d1 - d2 == 9223372036854775807, "DifferenceWithHugeYear");
>+    static_assert((d2 - 1) - d1 == -9223372036854775807 - 1,
>+                  "DifferenceWithHugeYear");
>+  }
>+  {
>+    // Check the limits of the return value at the start of the year range.
>+    constexpr civil_day d1(-9223372036854775807 - 1, 1, 1);
>+    constexpr civil_day d2(-9198119301927009254, 7, 28);
>+    static_assert(d2 - d1 == 9223372036854775807, "DifferenceWithHugeYear");
>+    static_assert(d1 - (d2 + 1) == -9223372036854775807 - 1,
>+                  "DifferenceWithHugeYear");
>+  }
>+  {
>+    // Check the limits of the return value from either side of year 0.
>+    constexpr civil_day d1(-12626367463883278, 9, 3);
>+    constexpr civil_day d2(12626367463883277, 3, 28);
>+    static_assert(d2 - d1 == 9223372036854775807, "DifferenceWithHugeYear");
>+    static_assert(d1 - (d2 + 1) == -9223372036854775807 - 1,
>+                  "DifferenceWithHugeYear");
>+  }
>+}
>+
>+// NOTE: Run this with --copt=-ftrapv to detect overflow problems.
>+TEST(CivilTime, DifferenceNoIntermediateOverflow) {
>+  {
>+    // The difference up to the minute field would be below the minimum
>+    // diff_t, but the 52 extra seconds brings us back to the minimum.
>+    constexpr civil_second s1(-292277022657, 1, 27, 8, 29 - 1, 52);
>+    constexpr civil_second s2(1970, 1, 1, 0, 0 - 1, 0);
>+    static_assert(s1 - s2 == -9223372036854775807 - 1,
>+                  "DifferenceNoIntermediateOverflow");
>+  }
>+  {
>+    // The difference up to the minute field would be above the maximum
>+    // diff_t, but the -53 extra seconds brings us back to the maximum.
>+    constexpr civil_second s1(292277026596, 12, 4, 15, 30, 7 - 7);
>+    constexpr civil_second s2(1970, 1, 1, 0, 0, 0 - 7);
>+    static_assert(s1 - s2 == 9223372036854775807,
>+                  "DifferenceNoIntermediateOverflow");
>+  }
>+}
>+
>+// Helper constexpr tests
>+
>+TEST(CivilTime, WeekDay) {
>+  constexpr civil_day cd(2016, 1, 28);
>+  constexpr weekday wd = get_weekday(cd);
>+  static_assert(wd == weekday::thursday, "Weekday");
>+}
>+
>+TEST(CivilTime, NextWeekDay) {
>+  constexpr civil_day cd(2016, 1, 28);
>+  constexpr civil_day next = next_weekday(cd, weekday::thursday);
>+  static_assert(next.year() == 2016, "NextWeekDay.year");
>+  static_assert(next.month() == 2, "NextWeekDay.month");
>+  static_assert(next.day() == 4, "NextWeekDay.day");
>+}
>+
>+TEST(CivilTime, PrevWeekDay) {
>+  constexpr civil_day cd(2016, 1, 28);
>+  constexpr civil_day prev = prev_weekday(cd, weekday::thursday);
>+  static_assert(prev.year() == 2016, "PrevWeekDay.year");
>+  static_assert(prev.month() == 1, "PrevWeekDay.month");
>+  static_assert(prev.day() == 21, "PrevWeekDay.day");
>+}
>+
>+TEST(CivilTime, YearDay) {
>+  constexpr civil_day cd(2016, 1, 28);
>+  constexpr int yd = get_yearday(cd);
>+  static_assert(yd == 28, "YearDay");
>+}
>+#endif  // __cpp_constexpr >= 201304 || _MSC_VER >= 1910
>+
>+// The remaining tests do not use constexpr.
>+
>+TEST(CivilTime, DefaultConstruction) {
>+  civil_second ss;
>+  EXPECT_EQ("1970-01-01T00:00:00", Format(ss));
>+
>+  civil_minute mm;
>+  EXPECT_EQ("1970-01-01T00:00", Format(mm));
>+
>+  civil_hour hh;
>+  EXPECT_EQ("1970-01-01T00", Format(hh));
>+
>+  civil_day d;
>+  EXPECT_EQ("1970-01-01", Format(d));
>+
>+  civil_month m;
>+  EXPECT_EQ("1970-01", Format(m));
>+
>+  civil_year y;
>+  EXPECT_EQ("1970", Format(y));
>+}
>+
>+TEST(CivilTime, StructMember) {
>+  struct S {
>+    civil_day day;
>+  };
>+  S s = {};
>+  EXPECT_EQ(civil_day{}, s.day);
>+}
>+
>+TEST(CivilTime, FieldsConstruction) {
>+  EXPECT_EQ("2015-01-02T03:04:05", Format(civil_second(2015, 1, 2, 3, 4, 5)));
>+  EXPECT_EQ("2015-01-02T03:04:00", Format(civil_second(2015, 1, 2, 3, 4)));
>+  EXPECT_EQ("2015-01-02T03:00:00", Format(civil_second(2015, 1, 2, 3)));
>+  EXPECT_EQ("2015-01-02T00:00:00", Format(civil_second(2015, 1, 2)));
>+  EXPECT_EQ("2015-01-01T00:00:00", Format(civil_second(2015, 1)));
>+  EXPECT_EQ("2015-01-01T00:00:00", Format(civil_second(2015)));
>+
>+  EXPECT_EQ("2015-01-02T03:04", Format(civil_minute(2015, 1, 2, 3, 4, 5)));
>+  EXPECT_EQ("2015-01-02T03:04", Format(civil_minute(2015, 1, 2, 3, 4)));
>+  EXPECT_EQ("2015-01-02T03:00", Format(civil_minute(2015, 1, 2, 3)));
>+  EXPECT_EQ("2015-01-02T00:00", Format(civil_minute(2015, 1, 2)));
>+  EXPECT_EQ("2015-01-01T00:00", Format(civil_minute(2015, 1)));
>+  EXPECT_EQ("2015-01-01T00:00", Format(civil_minute(2015)));
>+
>+  EXPECT_EQ("2015-01-02T03", Format(civil_hour(2015, 1, 2, 3, 4, 5)));
>+  EXPECT_EQ("2015-01-02T03", Format(civil_hour(2015, 1, 2, 3, 4)));
>+  EXPECT_EQ("2015-01-02T03", Format(civil_hour(2015, 1, 2, 3)));
>+  EXPECT_EQ("2015-01-02T00", Format(civil_hour(2015, 1, 2)));
>+  EXPECT_EQ("2015-01-01T00", Format(civil_hour(2015, 1)));
>+  EXPECT_EQ("2015-01-01T00", Format(civil_hour(2015)));
>+
>+  EXPECT_EQ("2015-01-02", Format(civil_day(2015, 1, 2, 3, 4, 5)));
>+  EXPECT_EQ("2015-01-02", Format(civil_day(2015, 1, 2, 3, 4)));
>+  EXPECT_EQ("2015-01-02", Format(civil_day(2015, 1, 2, 3)));
>+  EXPECT_EQ("2015-01-02", Format(civil_day(2015, 1, 2)));
>+  EXPECT_EQ("2015-01-01", Format(civil_day(2015, 1)));
>+  EXPECT_EQ("2015-01-01", Format(civil_day(2015)));
>+
>+  EXPECT_EQ("2015-01", Format(civil_month(2015, 1, 2, 3, 4, 5)));
>+  EXPECT_EQ("2015-01", Format(civil_month(2015, 1, 2, 3, 4)));
>+  EXPECT_EQ("2015-01", Format(civil_month(2015, 1, 2, 3)));
>+  EXPECT_EQ("2015-01", Format(civil_month(2015, 1, 2)));
>+  EXPECT_EQ("2015-01", Format(civil_month(2015, 1)));
>+  EXPECT_EQ("2015-01", Format(civil_month(2015)));
>+
>+  EXPECT_EQ("2015", Format(civil_year(2015, 1, 2, 3, 4, 5)));
>+  EXPECT_EQ("2015", Format(civil_year(2015, 1, 2, 3, 4)));
>+  EXPECT_EQ("2015", Format(civil_year(2015, 1, 2, 3)));
>+  EXPECT_EQ("2015", Format(civil_year(2015, 1, 2)));
>+  EXPECT_EQ("2015", Format(civil_year(2015, 1)));
>+  EXPECT_EQ("2015", Format(civil_year(2015)));
>+}
>+
>+TEST(CivilTime, FieldsConstructionLimits) {
>+  const int kIntMax = std::numeric_limits<int>::max();
>+  EXPECT_EQ("2038-01-19T03:14:07",
>+            Format(civil_second(1970, 1, 1, 0, 0, kIntMax)));
>+  EXPECT_EQ("6121-02-11T05:21:07",
>+            Format(civil_second(1970, 1, 1, 0, kIntMax, kIntMax)));
>+  EXPECT_EQ("251104-11-20T12:21:07",
>+            Format(civil_second(1970, 1, 1, kIntMax, kIntMax, kIntMax)));
>+  EXPECT_EQ("6130715-05-30T12:21:07",
>+            Format(civil_second(1970, 1, kIntMax, kIntMax, kIntMax, kIntMax)));
>+  EXPECT_EQ(
>+      "185087685-11-26T12:21:07",
>+      Format(civil_second(1970, kIntMax, kIntMax, kIntMax, kIntMax, kIntMax)));
>+
>+  const int kIntMin = std::numeric_limits<int>::min();
>+  EXPECT_EQ("1901-12-13T20:45:52",
>+            Format(civil_second(1970, 1, 1, 0, 0, kIntMin)));
>+  EXPECT_EQ("-2182-11-20T18:37:52",
>+            Format(civil_second(1970, 1, 1, 0, kIntMin, kIntMin)));
>+  EXPECT_EQ("-247165-02-11T10:37:52",
>+            Format(civil_second(1970, 1, 1, kIntMin, kIntMin, kIntMin)));
>+  EXPECT_EQ("-6126776-08-01T10:37:52",
>+            Format(civil_second(1970, 1, kIntMin, kIntMin, kIntMin, kIntMin)));
>+  EXPECT_EQ(
>+      "-185083747-10-31T10:37:52",
>+      Format(civil_second(1970, kIntMin, kIntMin, kIntMin, kIntMin, kIntMin)));
>+}
>+
>+TEST(CivilTime, ImplicitCrossAlignment) {
>+  civil_year year(2015);
>+  civil_month month = year;
>+  civil_day day = month;
>+  civil_hour hour = day;
>+  civil_minute minute = hour;
>+  civil_second second = minute;
>+
>+  second = year;
>+  EXPECT_EQ(second, year);
>+  second = month;
>+  EXPECT_EQ(second, month);
>+  second = day;
>+  EXPECT_EQ(second, day);
>+  second = hour;
>+  EXPECT_EQ(second, hour);
>+  second = minute;
>+  EXPECT_EQ(second, minute);
>+
>+  minute = year;
>+  EXPECT_EQ(minute, year);
>+  minute = month;
>+  EXPECT_EQ(minute, month);
>+  minute = day;
>+  EXPECT_EQ(minute, day);
>+  minute = hour;
>+  EXPECT_EQ(minute, hour);
>+
>+  hour = year;
>+  EXPECT_EQ(hour, year);
>+  hour = month;
>+  EXPECT_EQ(hour, month);
>+  hour = day;
>+  EXPECT_EQ(hour, day);
>+
>+  day = year;
>+  EXPECT_EQ(day, year);
>+  day = month;
>+  EXPECT_EQ(day, month);
>+
>+  month = year;
>+  EXPECT_EQ(month, year);
>+
>+  // Ensures unsafe conversions are not allowed.
>+  EXPECT_FALSE((std::is_convertible<civil_second, civil_minute>::value));
>+  EXPECT_FALSE((std::is_convertible<civil_second, civil_hour>::value));
>+  EXPECT_FALSE((std::is_convertible<civil_second, civil_day>::value));
>+  EXPECT_FALSE((std::is_convertible<civil_second, civil_month>::value));
>+  EXPECT_FALSE((std::is_convertible<civil_second, civil_year>::value));
>+
>+  EXPECT_FALSE((std::is_convertible<civil_minute, civil_hour>::value));
>+  EXPECT_FALSE((std::is_convertible<civil_minute, civil_day>::value));
>+  EXPECT_FALSE((std::is_convertible<civil_minute, civil_month>::value));
>+  EXPECT_FALSE((std::is_convertible<civil_minute, civil_year>::value));
>+
>+  EXPECT_FALSE((std::is_convertible<civil_hour, civil_day>::value));
>+  EXPECT_FALSE((std::is_convertible<civil_hour, civil_month>::value));
>+  EXPECT_FALSE((std::is_convertible<civil_hour, civil_year>::value));
>+
>+  EXPECT_FALSE((std::is_convertible<civil_day, civil_month>::value));
>+  EXPECT_FALSE((std::is_convertible<civil_day, civil_year>::value));
>+
>+  EXPECT_FALSE((std::is_convertible<civil_month, civil_year>::value));
>+}
>+
>+TEST(CivilTime, ExplicitCrossAlignment) {
>+  //
>+  // Assign from smaller units -> larger units
>+  //
>+
>+  civil_second second(2015, 1, 2, 3, 4, 5);
>+  EXPECT_EQ("2015-01-02T03:04:05", Format(second));
>+
>+  civil_minute minute(second);
>+  EXPECT_EQ("2015-01-02T03:04", Format(minute));
>+
>+  civil_hour hour(minute);
>+  EXPECT_EQ("2015-01-02T03", Format(hour));
>+
>+  civil_day day(hour);
>+  EXPECT_EQ("2015-01-02", Format(day));
>+
>+  civil_month month(day);
>+  EXPECT_EQ("2015-01", Format(month));
>+
>+  civil_year year(month);
>+  EXPECT_EQ("2015", Format(year));
>+
>+  //
>+  // Now assign from larger units -> smaller units
>+  //
>+
>+  month = civil_month(year);
>+  EXPECT_EQ("2015-01", Format(month));
>+
>+  day = civil_day(month);
>+  EXPECT_EQ("2015-01-01", Format(day));
>+
>+  hour = civil_hour(day);
>+  EXPECT_EQ("2015-01-01T00", Format(hour));
>+
>+  minute = civil_minute(hour);
>+  EXPECT_EQ("2015-01-01T00:00", Format(minute));
>+
>+  second = civil_second(minute);
>+  EXPECT_EQ("2015-01-01T00:00:00", Format(second));
>+}
>+
>+// Metafunction to test whether difference is allowed between two types.
>+template <typename T1, typename T2>
>+struct HasDifference {
>+  template <typename U1, typename U2>
>+  static std::false_type test(...);
>+  template <typename U1, typename U2>
>+  static std::true_type test(decltype(std::declval<U1>() - std::declval<U2>()));
>+  static constexpr bool value = decltype(test<T1, T2>(0))::value;
>+};
>+
>+TEST(CivilTime, DisallowCrossAlignedDifference) {
>+  // Difference is allowed between types with the same alignment.
>+  static_assert(HasDifference<civil_second, civil_second>::value, "");
>+  static_assert(HasDifference<civil_minute, civil_minute>::value, "");
>+  static_assert(HasDifference<civil_hour, civil_hour>::value, "");
>+  static_assert(HasDifference<civil_day, civil_day>::value, "");
>+  static_assert(HasDifference<civil_month, civil_month>::value, "");
>+  static_assert(HasDifference<civil_year, civil_year>::value, "");
>+
>+  // Difference is disallowed between types with different alignments.
>+  static_assert(!HasDifference<civil_second, civil_minute>::value, "");
>+  static_assert(!HasDifference<civil_second, civil_hour>::value, "");
>+  static_assert(!HasDifference<civil_second, civil_day>::value, "");
>+  static_assert(!HasDifference<civil_second, civil_month>::value, "");
>+  static_assert(!HasDifference<civil_second, civil_year>::value, "");
>+
>+  static_assert(!HasDifference<civil_minute, civil_hour>::value, "");
>+  static_assert(!HasDifference<civil_minute, civil_day>::value, "");
>+  static_assert(!HasDifference<civil_minute, civil_month>::value, "");
>+  static_assert(!HasDifference<civil_minute, civil_year>::value, "");
>+
>+  static_assert(!HasDifference<civil_hour, civil_day>::value, "");
>+  static_assert(!HasDifference<civil_hour, civil_month>::value, "");
>+  static_assert(!HasDifference<civil_hour, civil_year>::value, "");
>+
>+  static_assert(!HasDifference<civil_day, civil_month>::value, "");
>+  static_assert(!HasDifference<civil_day, civil_year>::value, "");
>+
>+  static_assert(!HasDifference<civil_month, civil_year>::value, "");
>+}
>+
>+TEST(CivilTime, ValueSemantics) {
>+  const civil_hour a(2015, 1, 2, 3);
>+  const civil_hour b = a;
>+  const civil_hour c(b);
>+  civil_hour d;
>+  d = c;
>+  EXPECT_EQ("2015-01-02T03", Format(d));
>+}
>+
>+TEST(CivilTime, Relational) {
>+  // Tests that the alignment unit is ignored in comparison.
>+  const civil_year year(2014);
>+  const civil_month month(year);
>+  EXPECT_EQ(year, month);
>+
>+#define TEST_RELATIONAL(OLDER, YOUNGER) \
>+  do {                                  \
>+    EXPECT_FALSE(OLDER < OLDER);        \
>+    EXPECT_FALSE(OLDER > OLDER);        \
>+    EXPECT_TRUE(OLDER >= OLDER);        \
>+    EXPECT_TRUE(OLDER <= OLDER);        \
>+    EXPECT_FALSE(YOUNGER < YOUNGER);    \
>+    EXPECT_FALSE(YOUNGER > YOUNGER);    \
>+    EXPECT_TRUE(YOUNGER >= YOUNGER);    \
>+    EXPECT_TRUE(YOUNGER <= YOUNGER);    \
>+    EXPECT_EQ(OLDER, OLDER);            \
>+    EXPECT_NE(OLDER, YOUNGER);          \
>+    EXPECT_LT(OLDER, YOUNGER);          \
>+    EXPECT_LE(OLDER, YOUNGER);          \
>+    EXPECT_GT(YOUNGER, OLDER);          \
>+    EXPECT_GE(YOUNGER, OLDER);          \
>+  } while (0)
>+
>+  // Alignment is ignored in comparison (verified above), so kSecond is used
>+  // to test comparison in all field positions.
>+  TEST_RELATIONAL(civil_second(2014, 1, 1, 0, 0, 0),
>+                  civil_second(2015, 1, 1, 0, 0, 0));
>+  TEST_RELATIONAL(civil_second(2014, 1, 1, 0, 0, 0),
>+                  civil_second(2014, 2, 1, 0, 0, 0));
>+  TEST_RELATIONAL(civil_second(2014, 1, 1, 0, 0, 0),
>+                  civil_second(2014, 1, 2, 0, 0, 0));
>+  TEST_RELATIONAL(civil_second(2014, 1, 1, 0, 0, 0),
>+                  civil_second(2014, 1, 1, 1, 0, 0));
>+  TEST_RELATIONAL(civil_second(2014, 1, 1, 1, 0, 0),
>+                  civil_second(2014, 1, 1, 1, 1, 0));
>+  TEST_RELATIONAL(civil_second(2014, 1, 1, 1, 1, 0),
>+                  civil_second(2014, 1, 1, 1, 1, 1));
>+
>+  // Tests the relational operators of two different CivilTime types.
>+  TEST_RELATIONAL(civil_day(2014, 1, 1), civil_minute(2014, 1, 1, 1, 1));
>+  TEST_RELATIONAL(civil_day(2014, 1, 1), civil_month(2014, 2));
>+
>+#undef TEST_RELATIONAL
>+}
>+
>+TEST(CivilTime, Arithmetic) {
>+  civil_second second(2015, 1, 2, 3, 4, 5);
>+  EXPECT_EQ("2015-01-02T03:04:06", Format(second += 1));
>+  EXPECT_EQ("2015-01-02T03:04:07", Format(second + 1));
>+  EXPECT_EQ("2015-01-02T03:04:08", Format(2 + second));
>+  EXPECT_EQ("2015-01-02T03:04:05", Format(second - 1));
>+  EXPECT_EQ("2015-01-02T03:04:05", Format(second -= 1));
>+  EXPECT_EQ("2015-01-02T03:04:05", Format(second++));
>+  EXPECT_EQ("2015-01-02T03:04:07", Format(++second));
>+  EXPECT_EQ("2015-01-02T03:04:07", Format(second--));
>+  EXPECT_EQ("2015-01-02T03:04:05", Format(--second));
>+
>+  civil_minute minute(2015, 1, 2, 3, 4);
>+  EXPECT_EQ("2015-01-02T03:05", Format(minute += 1));
>+  EXPECT_EQ("2015-01-02T03:06", Format(minute + 1));
>+  EXPECT_EQ("2015-01-02T03:07", Format(2 + minute));
>+  EXPECT_EQ("2015-01-02T03:04", Format(minute - 1));
>+  EXPECT_EQ("2015-01-02T03:04", Format(minute -= 1));
>+  EXPECT_EQ("2015-01-02T03:04", Format(minute++));
>+  EXPECT_EQ("2015-01-02T03:06", Format(++minute));
>+  EXPECT_EQ("2015-01-02T03:06", Format(minute--));
>+  EXPECT_EQ("2015-01-02T03:04", Format(--minute));
>+
>+  civil_hour hour(2015, 1, 2, 3);
>+  EXPECT_EQ("2015-01-02T04", Format(hour += 1));
>+  EXPECT_EQ("2015-01-02T05", Format(hour + 1));
>+  EXPECT_EQ("2015-01-02T06", Format(2 + hour));
>+  EXPECT_EQ("2015-01-02T03", Format(hour - 1));
>+  EXPECT_EQ("2015-01-02T03", Format(hour -= 1));
>+  EXPECT_EQ("2015-01-02T03", Format(hour++));
>+  EXPECT_EQ("2015-01-02T05", Format(++hour));
>+  EXPECT_EQ("2015-01-02T05", Format(hour--));
>+  EXPECT_EQ("2015-01-02T03", Format(--hour));
>+
>+  civil_day day(2015, 1, 2);
>+  EXPECT_EQ("2015-01-03", Format(day += 1));
>+  EXPECT_EQ("2015-01-04", Format(day + 1));
>+  EXPECT_EQ("2015-01-05", Format(2 + day));
>+  EXPECT_EQ("2015-01-02", Format(day - 1));
>+  EXPECT_EQ("2015-01-02", Format(day -= 1));
>+  EXPECT_EQ("2015-01-02", Format(day++));
>+  EXPECT_EQ("2015-01-04", Format(++day));
>+  EXPECT_EQ("2015-01-04", Format(day--));
>+  EXPECT_EQ("2015-01-02", Format(--day));
>+
>+  civil_month month(2015, 1);
>+  EXPECT_EQ("2015-02", Format(month += 1));
>+  EXPECT_EQ("2015-03", Format(month + 1));
>+  EXPECT_EQ("2015-04", Format(2 + month));
>+  EXPECT_EQ("2015-01", Format(month - 1));
>+  EXPECT_EQ("2015-01", Format(month -= 1));
>+  EXPECT_EQ("2015-01", Format(month++));
>+  EXPECT_EQ("2015-03", Format(++month));
>+  EXPECT_EQ("2015-03", Format(month--));
>+  EXPECT_EQ("2015-01", Format(--month));
>+
>+  civil_year year(2015);
>+  EXPECT_EQ("2016", Format(year += 1));
>+  EXPECT_EQ("2017", Format(year + 1));
>+  EXPECT_EQ("2018", Format(2 + year));
>+  EXPECT_EQ("2015", Format(year - 1));
>+  EXPECT_EQ("2015", Format(year -= 1));
>+  EXPECT_EQ("2015", Format(year++));
>+  EXPECT_EQ("2017", Format(++year));
>+  EXPECT_EQ("2017", Format(year--));
>+  EXPECT_EQ("2015", Format(--year));
>+}
>+
>+TEST(CivilTime, ArithmeticLimits) {
>+  const int kIntMax = std::numeric_limits<int>::max();
>+  const int kIntMin = std::numeric_limits<int>::min();
>+
>+  civil_second second(1970, 1, 1, 0, 0, 0);
>+  second += kIntMax;
>+  EXPECT_EQ("2038-01-19T03:14:07", Format(second));
>+  second -= kIntMax;
>+  EXPECT_EQ("1970-01-01T00:00:00", Format(second));
>+  second += kIntMin;
>+  EXPECT_EQ("1901-12-13T20:45:52", Format(second));
>+  second -= kIntMin;
>+  EXPECT_EQ("1970-01-01T00:00:00", Format(second));
>+
>+  civil_minute minute(1970, 1, 1, 0, 0);
>+  minute += kIntMax;
>+  EXPECT_EQ("6053-01-23T02:07", Format(minute));
>+  minute -= kIntMax;
>+  EXPECT_EQ("1970-01-01T00:00", Format(minute));
>+  minute += kIntMin;
>+  EXPECT_EQ("-2114-12-08T21:52", Format(minute));
>+  minute -= kIntMin;
>+  EXPECT_EQ("1970-01-01T00:00", Format(minute));
>+
>+  civil_hour hour(1970, 1, 1, 0);
>+  hour += kIntMax;
>+  EXPECT_EQ("246953-10-09T07", Format(hour));
>+  hour -= kIntMax;
>+  EXPECT_EQ("1970-01-01T00", Format(hour));
>+  hour += kIntMin;
>+  EXPECT_EQ("-243014-03-24T16", Format(hour));
>+  hour -= kIntMin;
>+  EXPECT_EQ("1970-01-01T00", Format(hour));
>+
>+  civil_day day(1970, 1, 1);
>+  day += kIntMax;
>+  EXPECT_EQ("5881580-07-11", Format(day));
>+  day -= kIntMax;
>+  EXPECT_EQ("1970-01-01", Format(day));
>+  day += kIntMin;
>+  EXPECT_EQ("-5877641-06-23", Format(day));
>+  day -= kIntMin;
>+  EXPECT_EQ("1970-01-01", Format(day));
>+
>+  civil_month month(1970, 1);
>+  month += kIntMax;
>+  EXPECT_EQ("178958940-08", Format(month));
>+  month -= kIntMax;
>+  EXPECT_EQ("1970-01", Format(month));
>+  month += kIntMin;
>+  EXPECT_EQ("-178955001-05", Format(month));
>+  month -= kIntMin;
>+  EXPECT_EQ("1970-01", Format(month));
>+
>+  civil_year year(0);
>+  year += kIntMax;
>+  EXPECT_EQ("2147483647", Format(year));
>+  year -= kIntMax;
>+  EXPECT_EQ("0", Format(year));
>+  year += kIntMin;
>+  EXPECT_EQ("-2147483648", Format(year));
>+  year -= kIntMin;
>+  EXPECT_EQ("0", Format(year));
>+}
>+
>+TEST(CivilTime, ArithmeticDifference) {
>+  civil_second second(2015, 1, 2, 3, 4, 5);
>+  EXPECT_EQ(0, second - second);
>+  EXPECT_EQ(10, (second + 10) - second);
>+  EXPECT_EQ(-10, (second - 10) - second);
>+
>+  civil_minute minute(2015, 1, 2, 3, 4);
>+  EXPECT_EQ(0, minute - minute);
>+  EXPECT_EQ(10, (minute + 10) - minute);
>+  EXPECT_EQ(-10, (minute - 10) - minute);
>+
>+  civil_hour hour(2015, 1, 2, 3);
>+  EXPECT_EQ(0, hour - hour);
>+  EXPECT_EQ(10, (hour + 10) - hour);
>+  EXPECT_EQ(-10, (hour - 10) - hour);
>+
>+  civil_day day(2015, 1, 2);
>+  EXPECT_EQ(0, day - day);
>+  EXPECT_EQ(10, (day + 10) - day);
>+  EXPECT_EQ(-10, (day - 10) - day);
>+
>+  civil_month month(2015, 1);
>+  EXPECT_EQ(0, month - month);
>+  EXPECT_EQ(10, (month + 10) - month);
>+  EXPECT_EQ(-10, (month - 10) - month);
>+
>+  civil_year year(2015);
>+  EXPECT_EQ(0, year - year);
>+  EXPECT_EQ(10, (year + 10) - year);
>+  EXPECT_EQ(-10, (year - 10) - year);
>+}
>+
>+TEST(CivilTime, DifferenceLimits) {
>+  const int kIntMax = std::numeric_limits<int>::max();
>+  const int kIntMin = std::numeric_limits<int>::min();
>+
>+  // Check day arithmetic at the end of the year range.
>+  const civil_day max_day(kIntMax, 12, 31);
>+  EXPECT_EQ(1, max_day - (max_day - 1));
>+  EXPECT_EQ(-1, (max_day - 1) - max_day);
>+
>+  // Check day arithmetic at the end of the year range.
>+  const civil_day min_day(kIntMin, 1, 1);
>+  EXPECT_EQ(1, (min_day + 1) - min_day);
>+  EXPECT_EQ(-1, min_day - (min_day + 1));
>+
>+  // Check the limits of the return value.
>+  const civil_day d1(1970, 1, 1);
>+  const civil_day d2(5881580, 7, 11);
>+  EXPECT_EQ(kIntMax, d2 - d1);
>+  EXPECT_EQ(kIntMin, d1 - (d2 + 1));
>+}
>+
>+TEST(CivilTime, Properties) {
>+  civil_second ss(2015, 2, 3, 4, 5, 6);
>+  EXPECT_EQ(2015, ss.year());
>+  EXPECT_EQ(2, ss.month());
>+  EXPECT_EQ(3, ss.day());
>+  EXPECT_EQ(4, ss.hour());
>+  EXPECT_EQ(5, ss.minute());
>+  EXPECT_EQ(6, ss.second());
>+
>+  civil_minute mm(2015, 2, 3, 4, 5, 6);
>+  EXPECT_EQ(2015, mm.year());
>+  EXPECT_EQ(2, mm.month());
>+  EXPECT_EQ(3, mm.day());
>+  EXPECT_EQ(4, mm.hour());
>+  EXPECT_EQ(5, mm.minute());
>+  EXPECT_EQ(0, mm.second());
>+
>+  civil_hour hh(2015, 2, 3, 4, 5, 6);
>+  EXPECT_EQ(2015, hh.year());
>+  EXPECT_EQ(2, hh.month());
>+  EXPECT_EQ(3, hh.day());
>+  EXPECT_EQ(4, hh.hour());
>+  EXPECT_EQ(0, hh.minute());
>+  EXPECT_EQ(0, hh.second());
>+
>+  civil_day d(2015, 2, 3, 4, 5, 6);
>+  EXPECT_EQ(2015, d.year());
>+  EXPECT_EQ(2, d.month());
>+  EXPECT_EQ(3, d.day());
>+  EXPECT_EQ(0, d.hour());
>+  EXPECT_EQ(0, d.minute());
>+  EXPECT_EQ(0, d.second());
>+  EXPECT_EQ(weekday::tuesday, get_weekday(d));
>+  EXPECT_EQ(34, get_yearday(d));
>+
>+  civil_month m(2015, 2, 3, 4, 5, 6);
>+  EXPECT_EQ(2015, m.year());
>+  EXPECT_EQ(2, m.month());
>+  EXPECT_EQ(1, m.day());
>+  EXPECT_EQ(0, m.hour());
>+  EXPECT_EQ(0, m.minute());
>+  EXPECT_EQ(0, m.second());
>+
>+  civil_year y(2015, 2, 3, 4, 5, 6);
>+  EXPECT_EQ(2015, y.year());
>+  EXPECT_EQ(1, y.month());
>+  EXPECT_EQ(1, y.day());
>+  EXPECT_EQ(0, y.hour());
>+  EXPECT_EQ(0, y.minute());
>+  EXPECT_EQ(0, y.second());
>+}
>+
>+TEST(CivilTime, OutputStream) {
>+  // Tests formatting of civil_year, which does not pad.
>+  EXPECT_EQ("2016", Format(civil_year(2016)));
>+  EXPECT_EQ("123", Format(civil_year(123)));
>+  EXPECT_EQ("0", Format(civil_year(0)));
>+  EXPECT_EQ("-1", Format(civil_year(-1)));
>+
>+  // Tests formatting of sub-year types, which pad to 2 digits
>+  EXPECT_EQ("2016-02", Format(civil_month(2016, 2)));
>+  EXPECT_EQ("2016-02-03", Format(civil_day(2016, 2, 3)));
>+  EXPECT_EQ("2016-02-03T04", Format(civil_hour(2016, 2, 3, 4)));
>+  EXPECT_EQ("2016-02-03T04:05", Format(civil_minute(2016, 2, 3, 4, 5)));
>+  EXPECT_EQ("2016-02-03T04:05:06", Format(civil_second(2016, 2, 3, 4, 5, 6)));
>+
>+  // Tests formatting of weekday.
>+  EXPECT_EQ("Monday", Format(weekday::monday));
>+  EXPECT_EQ("Tuesday", Format(weekday::tuesday));
>+  EXPECT_EQ("Wednesday", Format(weekday::wednesday));
>+  EXPECT_EQ("Thursday", Format(weekday::thursday));
>+  EXPECT_EQ("Friday", Format(weekday::friday));
>+  EXPECT_EQ("Saturday", Format(weekday::saturday));
>+  EXPECT_EQ("Sunday", Format(weekday::sunday));
>+}
>+
>+TEST(CivilTime, OutputStreamLeftFillWidth) {
>+  civil_second cs(2016, 2, 3, 4, 5, 6);
>+  {
>+    std::stringstream ss;
>+    ss << std::left << std::setfill('.');
>+    ss << std::setw(3) << 'X';
>+    ss << std::setw(21) << civil_year(cs);
>+    ss << std::setw(3) << 'X';
>+    EXPECT_EQ("X..2016.................X..", ss.str());
>+  }
>+  {
>+    std::stringstream ss;
>+    ss << std::left << std::setfill('.');
>+    ss << std::setw(3) << 'X';
>+    ss << std::setw(21) << civil_month(cs);
>+    ss << std::setw(3) << 'X';
>+    EXPECT_EQ("X..2016-02..............X..", ss.str());
>+  }
>+  {
>+    std::stringstream ss;
>+    ss << std::left << std::setfill('.');
>+    ss << std::setw(3) << 'X';
>+    ss << std::setw(21) << civil_day(cs);
>+    ss << std::setw(3) << 'X';
>+    EXPECT_EQ("X..2016-02-03...........X..", ss.str());
>+  }
>+  {
>+    std::stringstream ss;
>+    ss << std::left << std::setfill('.');
>+    ss << std::setw(3) << 'X';
>+    ss << std::setw(21) << civil_hour(cs);
>+    ss << std::setw(3) << 'X';
>+    EXPECT_EQ("X..2016-02-03T04........X..", ss.str());
>+  }
>+  {
>+    std::stringstream ss;
>+    ss << std::left << std::setfill('.');
>+    ss << std::setw(3) << 'X';
>+    ss << std::setw(21) << civil_minute(cs);
>+    ss << std::setw(3) << 'X';
>+    EXPECT_EQ("X..2016-02-03T04:05.....X..", ss.str());
>+  }
>+  {
>+    std::stringstream ss;
>+    ss << std::left << std::setfill('.');
>+    ss << std::setw(3) << 'X';
>+    ss << std::setw(21) << civil_second(cs);
>+    ss << std::setw(3) << 'X';
>+    EXPECT_EQ("X..2016-02-03T04:05:06..X..", ss.str());
>+  }
>+}
>+
>+TEST(CivilTime, NextPrevWeekday) {
>+  // Jan 1, 1970 was a Thursday.
>+  const civil_day thursday(1970, 1, 1);
>+  EXPECT_EQ(weekday::thursday, get_weekday(thursday));
>+
>+  // Thursday -> Thursday
>+  civil_day d = next_weekday(thursday, weekday::thursday);
>+  EXPECT_EQ(7, d - thursday) << Format(d);
>+  EXPECT_EQ(d - 14, prev_weekday(thursday, weekday::thursday));
>+
>+  // Thursday -> Friday
>+  d = next_weekday(thursday, weekday::friday);
>+  EXPECT_EQ(1, d - thursday) << Format(d);
>+  EXPECT_EQ(d - 7, prev_weekday(thursday, weekday::friday));
>+
>+  // Thursday -> Saturday
>+  d = next_weekday(thursday, weekday::saturday);
>+  EXPECT_EQ(2, d - thursday) << Format(d);
>+  EXPECT_EQ(d - 7, prev_weekday(thursday, weekday::saturday));
>+
>+  // Thursday -> Sunday
>+  d = next_weekday(thursday, weekday::sunday);
>+  EXPECT_EQ(3, d - thursday) << Format(d);
>+  EXPECT_EQ(d - 7, prev_weekday(thursday, weekday::sunday));
>+
>+  // Thursday -> Monday
>+  d = next_weekday(thursday, weekday::monday);
>+  EXPECT_EQ(4, d - thursday) << Format(d);
>+  EXPECT_EQ(d - 7, prev_weekday(thursday, weekday::monday));
>+
>+  // Thursday -> Tuesday
>+  d = next_weekday(thursday, weekday::tuesday);
>+  EXPECT_EQ(5, d - thursday) << Format(d);
>+  EXPECT_EQ(d - 7, prev_weekday(thursday, weekday::tuesday));
>+
>+  // Thursday -> Wednesday
>+  d = next_weekday(thursday, weekday::wednesday);
>+  EXPECT_EQ(6, d - thursday) << Format(d);
>+  EXPECT_EQ(d - 7, prev_weekday(thursday, weekday::wednesday));
>+}
>+
>+TEST(CivilTime, NormalizeWithHugeYear) {
>+  civil_month c(9223372036854775807, 1);
>+  EXPECT_EQ("9223372036854775807-01", Format(c));
>+  c = c - 1;  // Causes normalization
>+  EXPECT_EQ("9223372036854775806-12", Format(c));
>+
>+  c = civil_month(-9223372036854775807 - 1, 1);
>+  EXPECT_EQ("-9223372036854775808-01", Format(c));
>+  c = c + 12;  // Causes normalization
>+  EXPECT_EQ("-9223372036854775807-01", Format(c));
>+}
>+
>+TEST(CivilTime, LeapYears) {
>+  // Test data for leap years.
>+  const struct {
>+    int year;
>+    int days;
>+    struct {
>+      int month;
>+      int day;
>+    } leap_day;  // The date of the day after Feb 28.
>+  } kLeapYearTable[]{
>+      {1900, 365, {3, 1}},
>+      {1999, 365, {3, 1}},
>+      {2000, 366, {2, 29}},  // leap year
>+      {2001, 365, {3, 1}},
>+      {2002, 365, {3, 1}},
>+      {2003, 365, {3, 1}},
>+      {2004, 366, {2, 29}},  // leap year
>+      {2005, 365, {3, 1}},
>+      {2006, 365, {3, 1}},
>+      {2007, 365, {3, 1}},
>+      {2008, 366, {2, 29}},  // leap year
>+      {2009, 365, {3, 1}},
>+      {2100, 365, {3, 1}},
>+  };
>+
>+  for (const auto& e : kLeapYearTable) {
>+    // Tests incrementing through the leap day.
>+    const civil_day feb28(e.year, 2, 28);
>+    const civil_day next_day = feb28 + 1;
>+    EXPECT_EQ(e.leap_day.month, next_day.month());
>+    EXPECT_EQ(e.leap_day.day, next_day.day());
>+
>+    // Tests difference in days of leap years.
>+    const civil_year year(feb28);
>+    const civil_year next_year = year + 1;
>+    EXPECT_EQ(e.days, civil_day(next_year) - civil_day(year));
>+  }
>+}
>+
>+TEST(CivilTime, FirstThursdayInMonth) {
>+  const civil_day nov1(2014, 11, 1);
>+  const civil_day thursday = prev_weekday(nov1, weekday::thursday) + 7;
>+  EXPECT_EQ("2014-11-06", Format(thursday));
>+
>+  // Bonus: Date of Thanksgiving in the United States
>+  // Rule: Fourth Thursday of November
>+  const civil_day thanksgiving = thursday + 7 * 3;
>+  EXPECT_EQ("2014-11-27", Format(thanksgiving));
>+}
>+
>+}  // namespace cctz
>+}  // namespace time_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_fixed.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_fixed.cc
>new file mode 100644
>index 00000000000..598b08fde42
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_fixed.cc
>@@ -0,0 +1,123 @@
>+// Copyright 2016 Google Inc. All Rights Reserved.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//   http://www.apache.org/licenses/LICENSE-2.0
>+//
>+//   Unless required by applicable law or agreed to in writing, software
>+//   distributed under the License is distributed on an "AS IS" BASIS,
>+//   WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+//   See the License for the specific language governing permissions and
>+//   limitations under the License.
>+
>+#include "time_zone_fixed.h"
>+
>+#include <algorithm>
>+#include <chrono>
>+#include <cstdio>
>+#include <cstring>
>+#include <string>
>+
>+namespace absl {
>+namespace time_internal {
>+namespace cctz {
>+
>+namespace {
>+
>+// The prefix used for the internal names of fixed-offset zones.
>+const char kFixedOffsetPrefix[] = "Fixed/UTC";
>+
>+int Parse02d(const char* p) {
>+  static const char kDigits[] = "0123456789";
>+  if (const char* ap = std::strchr(kDigits, *p)) {
>+    int v = static_cast<int>(ap - kDigits);
>+    if (const char* bp = std::strchr(kDigits, *++p)) {
>+      return (v * 10) + static_cast<int>(bp - kDigits);
>+    }
>+  }
>+  return -1;
>+}
>+
>+}  // namespace
>+
>+bool FixedOffsetFromName(const std::string& name, seconds* offset) {
>+  if (name.compare(0, std::string::npos, "UTC", 3) == 0) {
>+    *offset = seconds::zero();
>+    return true;
>+  }
>+
>+  const std::size_t prefix_len = sizeof(kFixedOffsetPrefix) - 1;
>+  const char* const ep = kFixedOffsetPrefix + prefix_len;
>+  if (name.size() != prefix_len + 9)  // <prefix>+99:99:99
>+    return false;
>+  if (!std::equal(kFixedOffsetPrefix, ep, name.begin()))
>+    return false;
>+  const char* np = name.data() + prefix_len;
>+  if (np[0] != '+' && np[0] != '-')
>+    return false;
>+  if (np[3] != ':' || np[6] != ':')  // see note below about large offsets
>+    return false;
>+
>+  int hours = Parse02d(np + 1);
>+  if (hours == -1) return false;
>+  int mins = Parse02d(np + 4);
>+  if (mins == -1) return false;
>+  int secs = Parse02d(np + 7);
>+  if (secs == -1) return false;
>+
>+  secs += ((hours * 60) + mins) * 60;
>+  if (secs > 24 * 60 * 60) return false;  // outside supported offset range
>+  *offset = seconds(secs * (np[0] == '-' ? -1 : 1));  // "-" means west
>+  return true;
>+}
>+
>+std::string FixedOffsetToName(const seconds& offset) {
>+  if (offset == seconds::zero()) return "UTC";
>+  if (offset < std::chrono::hours(-24) || offset > std::chrono::hours(24)) {
>+    // We don't support fixed-offset zones more than 24 hours
>+    // away from UTC to avoid complications in rendering such
>+    // offsets and to (somewhat) limit the total number of zones.
>+    return "UTC";
>+  }
>+  int seconds = static_cast<int>(offset.count());
>+  const char sign = (seconds < 0 ? '-' : '+');
>+  int minutes = seconds / 60;
>+  seconds %= 60;
>+  if (sign == '-') {
>+    if (seconds > 0) {
>+      seconds -= 60;
>+      minutes += 1;
>+    }
>+    seconds = -seconds;
>+    minutes = -minutes;
>+  }
>+  int hours = minutes / 60;
>+  minutes %= 60;
>+  char buf[sizeof(kFixedOffsetPrefix) + sizeof("-24:00:00")];
>+  snprintf(buf, sizeof(buf), "%s%c%02d:%02d:%02d",
>+           kFixedOffsetPrefix, sign, hours, minutes, seconds);
>+  return buf;
>+}
>+
>+std::string FixedOffsetToAbbr(const seconds& offset) {
>+  std::string abbr = FixedOffsetToName(offset);
>+  const std::size_t prefix_len = sizeof(kFixedOffsetPrefix) - 1;
>+  if (abbr.size() == prefix_len + 9) {         // <prefix>+99:99:99
>+    abbr.erase(0, prefix_len);                 // +99:99:99
>+    abbr.erase(6, 1);                          // +99:9999
>+    abbr.erase(3, 1);                          // +999999
>+    if (abbr[5] == '0' && abbr[6] == '0') {    // +999900
>+      abbr.erase(5, 2);                        // +9999
>+      if (abbr[3] == '0' && abbr[4] == '0') {  // +9900
>+        abbr.erase(3, 2);                      // +99
>+      }
>+    }
>+  }
>+  return abbr;
>+}
>+
>+}  // namespace cctz
>+}  // namespace time_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_fixed.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_fixed.h
>new file mode 100644
>index 00000000000..489b857d5df
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_fixed.h
>@@ -0,0 +1,49 @@
>+// Copyright 2016 Google Inc. All Rights Reserved.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//   http://www.apache.org/licenses/LICENSE-2.0
>+//
>+//   Unless required by applicable law or agreed to in writing, software
>+//   distributed under the License is distributed on an "AS IS" BASIS,
>+//   WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+//   See the License for the specific language governing permissions and
>+//   limitations under the License.
>+
>+#ifndef ABSL_TIME_INTERNAL_CCTZ_TIME_ZONE_FIXED_H_
>+#define ABSL_TIME_INTERNAL_CCTZ_TIME_ZONE_FIXED_H_
>+
>+#include <string>
>+
>+#include "absl/time/internal/cctz/include/cctz/time_zone.h"
>+
>+namespace absl {
>+namespace time_internal {
>+namespace cctz {
>+
>+// Helper functions for dealing with the names and abbreviations
>+// of time zones that are a fixed offset (seconds east) from UTC.
>+// FixedOffsetFromName() extracts the offset from a valid fixed-offset
>+// name, while FixedOffsetToName() and FixedOffsetToAbbr() generate
>+// the canonical zone name and abbreviation respectively for the given
>+// offset.
>+//
>+// A fixed-offset name looks like "Fixed/UTC<+-><hours>:<mins>:<secs>".
>+// Its abbreviation is of the form "UTC(<+->H?H(MM(SS)?)?)?" where the
>+// optional pieces are omitted when their values are zero.  (Note that
>+// the sign is the opposite of that used in a POSIX TZ specification.)
>+//
>+// Note: FixedOffsetFromName() fails on syntax errors or when the parsed
>+// offset exceeds 24 hours.  FixedOffsetToName() and FixedOffsetToAbbr()
>+// both produce "UTC" when the argument offset exceeds 24 hours.
>+bool FixedOffsetFromName(const std::string& name, seconds* offset);
>+std::string FixedOffsetToName(const seconds& offset);
>+std::string FixedOffsetToAbbr(const seconds& offset);
>+
>+}  // namespace cctz
>+}  // namespace time_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_TIME_INTERNAL_CCTZ_TIME_ZONE_FIXED_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_format.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_format.cc
>new file mode 100644
>index 00000000000..1b023848efa
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_format.cc
>@@ -0,0 +1,851 @@
>+// Copyright 2016 Google Inc. All Rights Reserved.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//   http://www.apache.org/licenses/LICENSE-2.0
>+//
>+//   Unless required by applicable law or agreed to in writing, software
>+//   distributed under the License is distributed on an "AS IS" BASIS,
>+//   WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+//   See the License for the specific language governing permissions and
>+//   limitations under the License.
>+
>+#if !defined(HAS_STRPTIME)
>+# if !defined(_MSC_VER)
>+#  define HAS_STRPTIME 1  // assume everyone has strptime() except windows
>+# endif
>+#endif
>+
>+#include "absl/time/internal/cctz/include/cctz/time_zone.h"
>+
>+#include <cctype>
>+#include <chrono>
>+#include <cstddef>
>+#include <cstdint>
>+#include <cstring>
>+#include <ctime>
>+#include <limits>
>+#include <string>
>+#include <vector>
>+#if !HAS_STRPTIME
>+#include <iomanip>
>+#include <sstream>
>+#endif
>+
>+#include "absl/time/internal/cctz/include/cctz/civil_time.h"
>+#include "time_zone_if.h"
>+
>+namespace absl {
>+namespace time_internal {
>+namespace cctz {
>+namespace detail {
>+
>+namespace {
>+
>+#if !HAS_STRPTIME
>+// Build a strptime() using C++11's std::get_time().
>+char* strptime(const char* s, const char* fmt, std::tm* tm) {
>+  std::istringstream input(s);
>+  input >> std::get_time(tm, fmt);
>+  if (input.fail()) return nullptr;
>+  return const_cast<char*>(s) +
>+         (input.eof() ? strlen(s) : static_cast<std::size_t>(input.tellg()));
>+}
>+#endif
>+
>+std::tm ToTM(const time_zone::absolute_lookup& al) {
>+  std::tm tm{};
>+  tm.tm_sec = al.cs.second();
>+  tm.tm_min = al.cs.minute();
>+  tm.tm_hour = al.cs.hour();
>+  tm.tm_mday = al.cs.day();
>+  tm.tm_mon = al.cs.month() - 1;
>+
>+  // Saturate tm.tm_year is cases of over/underflow.
>+  if (al.cs.year() < std::numeric_limits<int>::min() + 1900) {
>+    tm.tm_year = std::numeric_limits<int>::min();
>+  } else if (al.cs.year() - 1900 > std::numeric_limits<int>::max()) {
>+    tm.tm_year = std::numeric_limits<int>::max();
>+  } else {
>+    tm.tm_year = static_cast<int>(al.cs.year() - 1900);
>+  }
>+
>+  switch (get_weekday(civil_day(al.cs))) {
>+    case weekday::sunday:
>+      tm.tm_wday = 0;
>+      break;
>+    case weekday::monday:
>+      tm.tm_wday = 1;
>+      break;
>+    case weekday::tuesday:
>+      tm.tm_wday = 2;
>+      break;
>+    case weekday::wednesday:
>+      tm.tm_wday = 3;
>+      break;
>+    case weekday::thursday:
>+      tm.tm_wday = 4;
>+      break;
>+    case weekday::friday:
>+      tm.tm_wday = 5;
>+      break;
>+    case weekday::saturday:
>+      tm.tm_wday = 6;
>+      break;
>+  }
>+  tm.tm_yday = get_yearday(civil_day(al.cs)) - 1;
>+  tm.tm_isdst = al.is_dst ? 1 : 0;
>+  return tm;
>+}
>+
>+const char kDigits[] = "0123456789";
>+
>+// Formats a 64-bit integer in the given field width.  Note that it is up
>+// to the caller of Format64() [and Format02d()/FormatOffset()] to ensure
>+// that there is sufficient space before ep to hold the conversion.
>+char* Format64(char* ep, int width, std::int_fast64_t v) {
>+  bool neg = false;
>+  if (v < 0) {
>+    --width;
>+    neg = true;
>+    if (v == std::numeric_limits<std::int_fast64_t>::min()) {
>+      // Avoid negating minimum value.
>+      std::int_fast64_t last_digit = -(v % 10);
>+      v /= 10;
>+      if (last_digit < 0) {
>+        ++v;
>+        last_digit += 10;
>+      }
>+      --width;
>+      *--ep = kDigits[last_digit];
>+    }
>+    v = -v;
>+  }
>+  do {
>+    --width;
>+    *--ep = kDigits[v % 10];
>+  } while (v /= 10);
>+  while (--width >= 0) *--ep = '0';  // zero pad
>+  if (neg) *--ep = '-';
>+  return ep;
>+}
>+
>+// Formats [0 .. 99] as %02d.
>+char* Format02d(char* ep, int v) {
>+  *--ep = kDigits[v % 10];
>+  *--ep = kDigits[(v / 10) % 10];
>+  return ep;
>+}
>+
>+// Formats a UTC offset, like +00:00.
>+char* FormatOffset(char* ep, int offset, const char* mode) {
>+  // TODO: Follow the RFC3339 "Unknown Local Offset Convention" and
>+  // generate a "negative zero" when we're formatting a zero offset
>+  // as the result of a failed load_time_zone().
>+  char sign = '+';
>+  if (offset < 0) {
>+    offset = -offset;  // bounded by 24h so no overflow
>+    sign = '-';
>+  }
>+  char sep = mode[0];
>+  if (sep != '\0' && mode[1] == '*') {
>+    ep = Format02d(ep, offset % 60);
>+    *--ep = sep;
>+  }
>+  int minutes = offset / 60;
>+  ep = Format02d(ep, minutes % 60);
>+  if (sep != '\0') *--ep = sep;
>+  ep = Format02d(ep, minutes / 60);
>+  *--ep = sign;
>+  return ep;
>+}
>+
>+// Formats a std::tm using strftime(3).
>+void FormatTM(std::string* out, const std::string& fmt, const std::tm& tm) {
>+  // strftime(3) returns the number of characters placed in the output
>+  // array (which may be 0 characters).  It also returns 0 to indicate
>+  // an error, like the array wasn't large enough.  To accommodate this,
>+  // the following code grows the buffer size from 2x the format std::string
>+  // length up to 32x.
>+  for (std::size_t i = 2; i != 32; i *= 2) {
>+    std::size_t buf_size = fmt.size() * i;
>+    std::vector<char> buf(buf_size);
>+    if (std::size_t len = strftime(&buf[0], buf_size, fmt.c_str(), &tm)) {
>+      out->append(&buf[0], len);
>+      return;
>+    }
>+  }
>+}
>+
>+// Used for %E#S/%E#f specifiers and for data values in parse().
>+template <typename T>
>+const char* ParseInt(const char* dp, int width, T min, T max, T* vp) {
>+  if (dp != nullptr) {
>+    const T kmin = std::numeric_limits<T>::min();
>+    bool erange = false;
>+    bool neg = false;
>+    T value = 0;
>+    if (*dp == '-') {
>+      neg = true;
>+      if (width <= 0 || --width != 0) {
>+        ++dp;
>+      } else {
>+        dp = nullptr;  // width was 1
>+      }
>+    }
>+    if (const char* const bp = dp) {
>+      while (const char* cp = strchr(kDigits, *dp)) {
>+        int d = static_cast<int>(cp - kDigits);
>+        if (d >= 10) break;
>+        if (value < kmin / 10) {
>+          erange = true;
>+          break;
>+        }
>+        value *= 10;
>+        if (value < kmin + d) {
>+          erange = true;
>+          break;
>+        }
>+        value -= d;
>+        dp += 1;
>+        if (width > 0 && --width == 0) break;
>+      }
>+      if (dp != bp && !erange && (neg || value != kmin)) {
>+        if (!neg || value != 0) {
>+          if (!neg) value = -value;  // make positive
>+          if (min <= value && value <= max) {
>+            *vp = value;
>+          } else {
>+            dp = nullptr;
>+          }
>+        } else {
>+          dp = nullptr;
>+        }
>+      } else {
>+        dp = nullptr;
>+      }
>+    }
>+  }
>+  return dp;
>+}
>+
>+// The number of base-10 digits that can be represented by a signed 64-bit
>+// integer.  That is, 10^kDigits10_64 <= 2^63 - 1 < 10^(kDigits10_64 + 1).
>+const int kDigits10_64 = 18;
>+
>+// 10^n for everything that can be represented by a signed 64-bit integer.
>+const std::int_fast64_t kExp10[kDigits10_64 + 1] = {
>+    1,
>+    10,
>+    100,
>+    1000,
>+    10000,
>+    100000,
>+    1000000,
>+    10000000,
>+    100000000,
>+    1000000000,
>+    10000000000,
>+    100000000000,
>+    1000000000000,
>+    10000000000000,
>+    100000000000000,
>+    1000000000000000,
>+    10000000000000000,
>+    100000000000000000,
>+    1000000000000000000,
>+};
>+
>+}  // namespace
>+
>+// Uses strftime(3) to format the given Time.  The following extended format
>+// specifiers are also supported:
>+//
>+//   - %Ez  - RFC3339-compatible numeric UTC offset (+hh:mm or -hh:mm)
>+//   - %E*z - Full-resolution numeric UTC offset (+hh:mm:ss or -hh:mm:ss)
>+//   - %E#S - Seconds with # digits of fractional precision
>+//   - %E*S - Seconds with full fractional precision (a literal '*')
>+//   - %E4Y - Four-character years (-999 ... -001, 0000, 0001 ... 9999)
>+//
>+// The standard specifiers from RFC3339_* (%Y, %m, %d, %H, %M, and %S) are
>+// handled internally for performance reasons.  strftime(3) is slow due to
>+// a POSIX requirement to respect changes to ${TZ}.
>+//
>+// The TZ/GNU %s extension is handled internally because strftime() has
>+// to use mktime() to generate it, and that assumes the local time zone.
>+//
>+// We also handle the %z and %Z specifiers to accommodate platforms that do
>+// not support the tm_gmtoff and tm_zone extensions to std::tm.
>+//
>+// Requires that zero() <= fs < seconds(1).
>+std::string format(const std::string& format, const time_point<seconds>& tp,
>+                   const detail::femtoseconds& fs, const time_zone& tz) {
>+  std::string result;
>+  result.reserve(format.size());  // A reasonable guess for the result size.
>+  const time_zone::absolute_lookup al = tz.lookup(tp);
>+  const std::tm tm = ToTM(al);
>+
>+  // Scratch buffer for internal conversions.
>+  char buf[3 + kDigits10_64];  // enough for longest conversion
>+  char* const ep = buf + sizeof(buf);
>+  char* bp;  // works back from ep
>+
>+  // Maintain three, disjoint subsequences that span format.
>+  //   [format.begin() ... pending) : already formatted into result
>+  //   [pending ... cur) : formatting pending, but no special cases
>+  //   [cur ... format.end()) : unexamined
>+  // Initially, everything is in the unexamined part.
>+  const char* pending = format.c_str();  // NUL terminated
>+  const char* cur = pending;
>+  const char* end = pending + format.length();
>+
>+  while (cur != end) {  // while something is unexamined
>+    // Moves cur to the next percent sign.
>+    const char* start = cur;
>+    while (cur != end && *cur != '%') ++cur;
>+
>+    // If the new pending text is all ordinary, copy it out.
>+    if (cur != start && pending == start) {
>+      result.append(pending, static_cast<std::size_t>(cur - pending));
>+      pending = start = cur;
>+    }
>+
>+    // Span the sequential percent signs.
>+    const char* percent = cur;
>+    while (cur != end && *cur == '%') ++cur;
>+
>+    // If the new pending text is all percents, copy out one
>+    // percent for every matched pair, then skip those pairs.
>+    if (cur != start && pending == start) {
>+      std::size_t escaped = static_cast<std::size_t>(cur - pending) / 2;
>+      result.append(pending, escaped);
>+      pending += escaped * 2;
>+      // Also copy out a single trailing percent.
>+      if (pending != cur && cur == end) {
>+        result.push_back(*pending++);
>+      }
>+    }
>+
>+    // Loop unless we have an unescaped percent.
>+    if (cur == end || (cur - percent) % 2 == 0) continue;
>+
>+    // Simple specifiers that we handle ourselves.
>+    if (strchr("YmdeHMSzZs%", *cur)) {
>+      if (cur - 1 != pending) {
>+        FormatTM(&result, std::string(pending, cur - 1), tm);
>+      }
>+      switch (*cur) {
>+        case 'Y':
>+          // This avoids the tm.tm_year overflow problem for %Y, however
>+          // tm.tm_year will still be used by other specifiers like %D.
>+          bp = Format64(ep, 0, al.cs.year());
>+          result.append(bp, static_cast<std::size_t>(ep - bp));
>+          break;
>+        case 'm':
>+          bp = Format02d(ep, al.cs.month());
>+          result.append(bp, static_cast<std::size_t>(ep - bp));
>+          break;
>+        case 'd':
>+        case 'e':
>+          bp = Format02d(ep, al.cs.day());
>+          if (*cur == 'e' && *bp == '0') *bp = ' ';  // for Windows
>+          result.append(bp, static_cast<std::size_t>(ep - bp));
>+          break;
>+        case 'H':
>+          bp = Format02d(ep, al.cs.hour());
>+          result.append(bp, static_cast<std::size_t>(ep - bp));
>+          break;
>+        case 'M':
>+          bp = Format02d(ep, al.cs.minute());
>+          result.append(bp, static_cast<std::size_t>(ep - bp));
>+          break;
>+        case 'S':
>+          bp = Format02d(ep, al.cs.second());
>+          result.append(bp, static_cast<std::size_t>(ep - bp));
>+          break;
>+        case 'z':
>+          bp = FormatOffset(ep, al.offset, "");
>+          result.append(bp, static_cast<std::size_t>(ep - bp));
>+          break;
>+        case 'Z':
>+          result.append(al.abbr);
>+          break;
>+        case 's':
>+          bp = Format64(ep, 0, ToUnixSeconds(tp));
>+          result.append(bp, static_cast<std::size_t>(ep - bp));
>+          break;
>+        case '%':
>+          result.push_back('%');
>+          break;
>+      }
>+      pending = ++cur;
>+      continue;
>+    }
>+
>+    // Loop if there is no E modifier.
>+    if (*cur != 'E' || ++cur == end) continue;
>+
>+    // Format our extensions.
>+    if (*cur == 'z') {
>+      // Formats %Ez.
>+      if (cur - 2 != pending) {
>+        FormatTM(&result, std::string(pending, cur - 2), tm);
>+      }
>+      bp = FormatOffset(ep, al.offset, ":");
>+      result.append(bp, static_cast<std::size_t>(ep - bp));
>+      pending = ++cur;
>+    } else if (*cur == '*' && cur + 1 != end && *(cur + 1) == 'z') {
>+      // Formats %E*z.
>+      if (cur - 2 != pending) {
>+        FormatTM(&result, std::string(pending, cur - 2), tm);
>+      }
>+      bp = FormatOffset(ep, al.offset, ":*");
>+      result.append(bp, static_cast<std::size_t>(ep - bp));
>+      pending = cur += 2;
>+    } else if (*cur == '*' && cur + 1 != end &&
>+               (*(cur + 1) == 'S' || *(cur + 1) == 'f')) {
>+      // Formats %E*S or %E*F.
>+      if (cur - 2 != pending) {
>+        FormatTM(&result, std::string(pending, cur - 2), tm);
>+      }
>+      char* cp = ep;
>+      bp = Format64(cp, 15, fs.count());
>+      while (cp != bp && cp[-1] == '0') --cp;
>+      switch (*(cur + 1)) {
>+        case 'S':
>+          if (cp != bp) *--bp = '.';
>+          bp = Format02d(bp, al.cs.second());
>+          break;
>+        case 'f':
>+          if (cp == bp) *--bp = '0';
>+          break;
>+      }
>+      result.append(bp, static_cast<std::size_t>(cp - bp));
>+      pending = cur += 2;
>+    } else if (*cur == '4' && cur + 1 != end && *(cur + 1) == 'Y') {
>+      // Formats %E4Y.
>+      if (cur - 2 != pending) {
>+        FormatTM(&result, std::string(pending, cur - 2), tm);
>+      }
>+      bp = Format64(ep, 4, al.cs.year());
>+      result.append(bp, static_cast<std::size_t>(ep - bp));
>+      pending = cur += 2;
>+    } else if (std::isdigit(*cur)) {
>+      // Possibly found %E#S or %E#f.
>+      int n = 0;
>+      if (const char* np = ParseInt(cur, 0, 0, 1024, &n)) {
>+        if (*np == 'S' || *np == 'f') {
>+          // Formats %E#S or %E#f.
>+          if (cur - 2 != pending) {
>+            FormatTM(&result, std::string(pending, cur - 2), tm);
>+          }
>+          bp = ep;
>+          if (n > 0) {
>+            if (n > kDigits10_64) n = kDigits10_64;
>+            bp = Format64(bp, n, (n > 15) ? fs.count() * kExp10[n - 15]
>+                                          : fs.count() / kExp10[15 - n]);
>+            if (*np == 'S') *--bp = '.';
>+          }
>+          if (*np == 'S') bp = Format02d(bp, al.cs.second());
>+          result.append(bp, static_cast<std::size_t>(ep - bp));
>+          pending = cur = ++np;
>+        }
>+      }
>+    }
>+  }
>+
>+  // Formats any remaining data.
>+  if (end != pending) {
>+    FormatTM(&result, std::string(pending, end), tm);
>+  }
>+
>+  return result;
>+}
>+
>+namespace {
>+
>+const char* ParseOffset(const char* dp, const char* mode, int* offset) {
>+  if (dp != nullptr) {
>+    const char first = *dp++;
>+    if (first == '+' || first == '-') {
>+      char sep = mode[0];
>+      int hours = 0;
>+      int minutes = 0;
>+      int seconds = 0;
>+      const char* ap = ParseInt(dp, 2, 0, 23, &hours);
>+      if (ap != nullptr && ap - dp == 2) {
>+        dp = ap;
>+        if (sep != '\0' && *ap == sep) ++ap;
>+        const char* bp = ParseInt(ap, 2, 0, 59, &minutes);
>+        if (bp != nullptr && bp - ap == 2) {
>+          dp = bp;
>+          if (sep != '\0' && *bp == sep) ++bp;
>+          const char* cp = ParseInt(bp, 2, 0, 59, &seconds);
>+          if (cp != nullptr && cp - bp == 2) dp = cp;
>+        }
>+        *offset = ((hours * 60 + minutes) * 60) + seconds;
>+        if (first == '-') *offset = -*offset;
>+      } else {
>+        dp = nullptr;
>+      }
>+    } else if (first == 'Z') {  // Zulu
>+      *offset = 0;
>+    } else {
>+      dp = nullptr;
>+    }
>+  }
>+  return dp;
>+}
>+
>+const char* ParseZone(const char* dp, std::string* zone) {
>+  zone->clear();
>+  if (dp != nullptr) {
>+    while (*dp != '\0' && !std::isspace(*dp)) zone->push_back(*dp++);
>+    if (zone->empty()) dp = nullptr;
>+  }
>+  return dp;
>+}
>+
>+const char* ParseSubSeconds(const char* dp, detail::femtoseconds* subseconds) {
>+  if (dp != nullptr) {
>+    std::int_fast64_t v = 0;
>+    std::int_fast64_t exp = 0;
>+    const char* const bp = dp;
>+    while (const char* cp = strchr(kDigits, *dp)) {
>+      int d = static_cast<int>(cp - kDigits);
>+      if (d >= 10) break;
>+      if (exp < 15) {
>+        exp += 1;
>+        v *= 10;
>+        v += d;
>+      }
>+      ++dp;
>+    }
>+    if (dp != bp) {
>+      v *= kExp10[15 - exp];
>+      *subseconds = detail::femtoseconds(v);
>+    } else {
>+      dp = nullptr;
>+    }
>+  }
>+  return dp;
>+}
>+
>+// Parses a std::string into a std::tm using strptime(3).
>+const char* ParseTM(const char* dp, const char* fmt, std::tm* tm) {
>+  if (dp != nullptr) {
>+    dp = strptime(dp, fmt, tm);
>+  }
>+  return dp;
>+}
>+
>+}  // namespace
>+
>+// Uses strptime(3) to parse the given input.  Supports the same extended
>+// format specifiers as format(), although %E#S and %E*S are treated
>+// identically (and similarly for %E#f and %E*f).  %Ez and %E*z also accept
>+// the same inputs.
>+//
>+// The standard specifiers from RFC3339_* (%Y, %m, %d, %H, %M, and %S) are
>+// handled internally so that we can normally avoid strptime() altogether
>+// (which is particularly helpful when the native implementation is broken).
>+//
>+// The TZ/GNU %s extension is handled internally because strptime() has to
>+// use localtime_r() to generate it, and that assumes the local time zone.
>+//
>+// We also handle the %z specifier to accommodate platforms that do not
>+// support the tm_gmtoff extension to std::tm.  %Z is parsed but ignored.
>+bool parse(const std::string& format, const std::string& input,
>+           const time_zone& tz, time_point<seconds>* sec,
>+           detail::femtoseconds* fs, std::string* err) {
>+  // The unparsed input.
>+  const char* data = input.c_str();  // NUL terminated
>+
>+  // Skips leading whitespace.
>+  while (std::isspace(*data)) ++data;
>+
>+  const year_t kyearmax = std::numeric_limits<year_t>::max();
>+  const year_t kyearmin = std::numeric_limits<year_t>::min();
>+
>+  // Sets default values for unspecified fields.
>+  bool saw_year = false;
>+  year_t year = 1970;
>+  std::tm tm{};
>+  tm.tm_year = 1970 - 1900;
>+  tm.tm_mon = 1 - 1;  // Jan
>+  tm.tm_mday = 1;
>+  tm.tm_hour = 0;
>+  tm.tm_min = 0;
>+  tm.tm_sec = 0;
>+  tm.tm_wday = 4;  // Thu
>+  tm.tm_yday = 0;
>+  tm.tm_isdst = 0;
>+  auto subseconds = detail::femtoseconds::zero();
>+  bool saw_offset = false;
>+  int offset = 0;  // No offset from passed tz.
>+  std::string zone = "UTC";
>+
>+  const char* fmt = format.c_str();  // NUL terminated
>+  bool twelve_hour = false;
>+  bool afternoon = false;
>+
>+  bool saw_percent_s = false;
>+  std::int_fast64_t percent_s = 0;
>+
>+  // Steps through format, one specifier at a time.
>+  while (data != nullptr && *fmt != '\0') {
>+    if (std::isspace(*fmt)) {
>+      while (std::isspace(*data)) ++data;
>+      while (std::isspace(*++fmt)) continue;
>+      continue;
>+    }
>+
>+    if (*fmt != '%') {
>+      if (*data == *fmt) {
>+        ++data;
>+        ++fmt;
>+      } else {
>+        data = nullptr;
>+      }
>+      continue;
>+    }
>+
>+    const char* percent = fmt;
>+    if (*++fmt == '\0') {
>+      data = nullptr;
>+      continue;
>+    }
>+    switch (*fmt++) {
>+      case 'Y':
>+        // Symmetrically with FormatTime(), directly handing %Y avoids the
>+        // tm.tm_year overflow problem.  However, tm.tm_year will still be
>+        // used by other specifiers like %D.
>+        data = ParseInt(data, 0, kyearmin, kyearmax, &year);
>+        if (data != nullptr) saw_year = true;
>+        continue;
>+      case 'm':
>+        data = ParseInt(data, 2, 1, 12, &tm.tm_mon);
>+        if (data != nullptr) tm.tm_mon -= 1;
>+        continue;
>+      case 'd':
>+      case 'e':
>+        data = ParseInt(data, 2, 1, 31, &tm.tm_mday);
>+        continue;
>+      case 'H':
>+        data = ParseInt(data, 2, 0, 23, &tm.tm_hour);
>+        twelve_hour = false;
>+        continue;
>+      case 'M':
>+        data = ParseInt(data, 2, 0, 59, &tm.tm_min);
>+        continue;
>+      case 'S':
>+        data = ParseInt(data, 2, 0, 60, &tm.tm_sec);
>+        continue;
>+      case 'I':
>+      case 'l':
>+      case 'r':  // probably uses %I
>+        twelve_hour = true;
>+        break;
>+      case 'R':  // uses %H
>+      case 'T':  // uses %H
>+      case 'c':  // probably uses %H
>+      case 'X':  // probably uses %H
>+        twelve_hour = false;
>+        break;
>+      case 'z':
>+        data = ParseOffset(data, "", &offset);
>+        if (data != nullptr) saw_offset = true;
>+        continue;
>+      case 'Z':  // ignored; zone abbreviations are ambiguous
>+        data = ParseZone(data, &zone);
>+        continue;
>+      case 's':
>+        data = ParseInt(data, 0,
>+                        std::numeric_limits<std::int_fast64_t>::min(),
>+                        std::numeric_limits<std::int_fast64_t>::max(),
>+                        &percent_s);
>+        if (data != nullptr) saw_percent_s = true;
>+        continue;
>+      case '%':
>+        data = (*data == '%' ? data + 1 : nullptr);
>+        continue;
>+      case 'E':
>+        if (*fmt == 'z' || (*fmt == '*' && *(fmt + 1) == 'z')) {
>+          data = ParseOffset(data, ":", &offset);
>+          if (data != nullptr) saw_offset = true;
>+          fmt += (*fmt == 'z') ? 1 : 2;
>+          continue;
>+        }
>+        if (*fmt == '*' && *(fmt + 1) == 'S') {
>+          data = ParseInt(data, 2, 0, 60, &tm.tm_sec);
>+          if (data != nullptr && *data == '.') {
>+            data = ParseSubSeconds(data + 1, &subseconds);
>+          }
>+          fmt += 2;
>+          continue;
>+        }
>+        if (*fmt == '*' && *(fmt + 1) == 'f') {
>+          if (data != nullptr && std::isdigit(*data)) {
>+            data = ParseSubSeconds(data, &subseconds);
>+          }
>+          fmt += 2;
>+          continue;
>+        }
>+        if (*fmt == '4' && *(fmt + 1) == 'Y') {
>+          const char* bp = data;
>+          data = ParseInt(data, 4, year_t{-999}, year_t{9999}, &year);
>+          if (data != nullptr) {
>+            if (data - bp == 4) {
>+              saw_year = true;
>+            } else {
>+              data = nullptr;  // stopped too soon
>+            }
>+          }
>+          fmt += 2;
>+          continue;
>+        }
>+        if (std::isdigit(*fmt)) {
>+          int n = 0;  // value ignored
>+          if (const char* np = ParseInt(fmt, 0, 0, 1024, &n)) {
>+            if (*np == 'S') {
>+              data = ParseInt(data, 2, 0, 60, &tm.tm_sec);
>+              if (data != nullptr && *data == '.') {
>+                data = ParseSubSeconds(data + 1, &subseconds);
>+              }
>+              fmt = ++np;
>+              continue;
>+            }
>+            if (*np == 'f') {
>+              if (data != nullptr && std::isdigit(*data)) {
>+                data = ParseSubSeconds(data, &subseconds);
>+              }
>+              fmt = ++np;
>+              continue;
>+            }
>+          }
>+        }
>+        if (*fmt == 'c') twelve_hour = false;  // probably uses %H
>+        if (*fmt == 'X') twelve_hour = false;  // probably uses %H
>+        if (*fmt != '\0') ++fmt;
>+        break;
>+      case 'O':
>+        if (*fmt == 'H') twelve_hour = false;
>+        if (*fmt == 'I') twelve_hour = true;
>+        if (*fmt != '\0') ++fmt;
>+        break;
>+    }
>+
>+    // Parses the current specifier.
>+    const char* orig_data = data;
>+    std::string spec(percent, static_cast<std::size_t>(fmt - percent));
>+    data = ParseTM(data, spec.c_str(), &tm);
>+
>+    // If we successfully parsed %p we need to remember whether the result
>+    // was AM or PM so that we can adjust tm_hour before ConvertDateTime().
>+    // So reparse the input with a known AM hour, and check if it is shifted
>+    // to a PM hour.
>+    if (spec == "%p" && data != nullptr) {
>+      std::string test_input = "1";
>+      test_input.append(orig_data, static_cast<std::size_t>(data - orig_data));
>+      const char* test_data = test_input.c_str();
>+      std::tm tmp{};
>+      ParseTM(test_data, "%I%p", &tmp);
>+      afternoon = (tmp.tm_hour == 13);
>+    }
>+  }
>+
>+  // Adjust a 12-hour tm_hour value if it should be in the afternoon.
>+  if (twelve_hour && afternoon && tm.tm_hour < 12) {
>+    tm.tm_hour += 12;
>+  }
>+
>+  if (data == nullptr) {
>+    if (err != nullptr) *err = "Failed to parse input";
>+    return false;
>+  }
>+
>+  // Skip any remaining whitespace.
>+  while (std::isspace(*data)) ++data;
>+
>+  // parse() must consume the entire input std::string.
>+  if (*data != '\0') {
>+    if (err != nullptr) *err = "Illegal trailing data in input string";
>+    return false;
>+  }
>+
>+  // If we saw %s then we ignore anything else and return that time.
>+  if (saw_percent_s) {
>+    *sec = FromUnixSeconds(percent_s);
>+    *fs = detail::femtoseconds::zero();
>+    return true;
>+  }
>+
>+  // If we saw %z, %Ez, or %E*z then we want to interpret the parsed fields
>+  // in UTC and then shift by that offset.  Otherwise we want to interpret
>+  // the fields directly in the passed time_zone.
>+  time_zone ptz = saw_offset ? utc_time_zone() : tz;
>+
>+  // Allows a leap second of 60 to normalize forward to the following ":00".
>+  if (tm.tm_sec == 60) {
>+    tm.tm_sec -= 1;
>+    offset -= 1;
>+    subseconds = detail::femtoseconds::zero();
>+  }
>+
>+  if (!saw_year) {
>+    year = year_t{tm.tm_year};
>+    if (year > kyearmax - 1900) {
>+      // Platform-dependent, maybe unreachable.
>+      if (err != nullptr) *err = "Out-of-range year";
>+      return false;
>+    }
>+    year += 1900;
>+  }
>+
>+  const int month = tm.tm_mon + 1;
>+  civil_second cs(year, month, tm.tm_mday, tm.tm_hour, tm.tm_min, tm.tm_sec);
>+
>+  // parse() should not allow normalization. Due to the restricted field
>+  // ranges above (see ParseInt()), the only possibility is for days to roll
>+  // into months. That is, parsing "Sep 31" should not produce "Oct 1".
>+  if (cs.month() != month || cs.day() != tm.tm_mday) {
>+    if (err != nullptr) *err = "Out-of-range field";
>+    return false;
>+  }
>+
>+  // Accounts for the offset adjustment before converting to absolute time.
>+  if ((offset < 0 && cs > civil_second::max() + offset) ||
>+      (offset > 0 && cs < civil_second::min() + offset)) {
>+    if (err != nullptr) *err = "Out-of-range field";
>+    return false;
>+  }
>+  cs -= offset;
>+
>+  const auto tp = ptz.lookup(cs).pre;
>+  // Checks for overflow/underflow and returns an error as necessary.
>+  if (tp == time_point<seconds>::max()) {
>+    const auto al = ptz.lookup(time_point<seconds>::max());
>+    if (cs > al.cs) {
>+      if (err != nullptr) *err = "Out-of-range field";
>+      return false;
>+    }
>+  }
>+  if (tp == time_point<seconds>::min()) {
>+    const auto al = ptz.lookup(time_point<seconds>::min());
>+    if (cs < al.cs) {
>+      if (err != nullptr) *err = "Out-of-range field";
>+      return false;
>+    }
>+  }
>+
>+  *sec = tp;
>+  *fs = subseconds;
>+  return true;
>+}
>+
>+}  // namespace detail
>+}  // namespace cctz
>+}  // namespace time_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_format_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_format_test.cc
>new file mode 100644
>index 00000000000..a90dda7603a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_format_test.cc
>@@ -0,0 +1,1426 @@
>+// Copyright 2016 Google Inc. All Rights Reserved.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//   http://www.apache.org/licenses/LICENSE-2.0
>+//
>+//   Unless required by applicable law or agreed to in writing, software
>+//   distributed under the License is distributed on an "AS IS" BASIS,
>+//   WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+//   See the License for the specific language governing permissions and
>+//   limitations under the License.
>+
>+#include "absl/time/internal/cctz/include/cctz/time_zone.h"
>+
>+#include <chrono>
>+#include <iomanip>
>+#include <sstream>
>+#include <string>
>+
>+#include "absl/time/internal/cctz/include/cctz/civil_time.h"
>+#include "gmock/gmock.h"
>+#include "gtest/gtest.h"
>+
>+namespace chrono = std::chrono;
>+
>+namespace absl {
>+namespace time_internal {
>+namespace cctz {
>+
>+namespace {
>+
>+// This helper is a macro so that failed expectations show up with the
>+// correct line numbers.
>+#define ExpectTime(tp, tz, y, m, d, hh, mm, ss, off, isdst, zone) \
>+  do {                                                            \
>+    time_zone::absolute_lookup al = tz.lookup(tp);                \
>+    EXPECT_EQ(y, al.cs.year());                                   \
>+    EXPECT_EQ(m, al.cs.month());                                  \
>+    EXPECT_EQ(d, al.cs.day());                                    \
>+    EXPECT_EQ(hh, al.cs.hour());                                  \
>+    EXPECT_EQ(mm, al.cs.minute());                                \
>+    EXPECT_EQ(ss, al.cs.second());                                \
>+    EXPECT_EQ(off, al.offset);                                    \
>+    EXPECT_TRUE(isdst == al.is_dst);                              \
>+    EXPECT_STREQ(zone, al.abbr);                                  \
>+  } while (0)
>+
>+const char RFC3339_full[] = "%Y-%m-%dT%H:%M:%E*S%Ez";
>+const char RFC3339_sec[] =  "%Y-%m-%dT%H:%M:%S%Ez";
>+
>+const char RFC1123_full[] = "%a, %d %b %Y %H:%M:%S %z";
>+const char RFC1123_no_wday[] =  "%d %b %Y %H:%M:%S %z";
>+
>+// A helper that tests the given format specifier by itself, and with leading
>+// and trailing characters.  For example: TestFormatSpecifier(tp, "%a", "Thu").
>+template <typename D>
>+void TestFormatSpecifier(time_point<D> tp, time_zone tz, const std::string& fmt,
>+                         const std::string& ans) {
>+  EXPECT_EQ(ans, format(fmt, tp, tz)) << fmt;
>+  EXPECT_EQ("xxx " + ans, format("xxx " + fmt, tp, tz));
>+  EXPECT_EQ(ans + " yyy", format(fmt + " yyy", tp, tz));
>+  EXPECT_EQ("xxx " + ans + " yyy", format("xxx " + fmt + " yyy", tp, tz));
>+}
>+
>+// These tests sometimes run on platforms that have zoneinfo data so old
>+// that the transition we are attempting to check does not exist, most
>+// notably Android emulators.  Fortunately, AndroidZoneInfoSource supports
>+// time_zone::version() so, in cases where we've learned that it matters,
>+// we can make the check conditionally.
>+int VersionCmp(time_zone tz, const std::string& target) {
>+  std::string version = tz.version();
>+  if (version.empty() && !target.empty()) return 1;  // unknown > known
>+  return version.compare(target);
>+}
>+
>+}  // namespace
>+
>+//
>+// Testing format()
>+//
>+
>+TEST(Format, TimePointResolution) {
>+  const char kFmt[] = "%H:%M:%E*S";
>+  const time_zone utc = utc_time_zone();
>+  const time_point<chrono::nanoseconds> t0 =
>+      chrono::system_clock::from_time_t(1420167845) +
>+      chrono::milliseconds(123) + chrono::microseconds(456) +
>+      chrono::nanoseconds(789);
>+  EXPECT_EQ(
>+      "03:04:05.123456789",
>+      format(kFmt, chrono::time_point_cast<chrono::nanoseconds>(t0), utc));
>+  EXPECT_EQ(
>+      "03:04:05.123456",
>+      format(kFmt, chrono::time_point_cast<chrono::microseconds>(t0), utc));
>+  EXPECT_EQ(
>+      "03:04:05.123",
>+      format(kFmt, chrono::time_point_cast<chrono::milliseconds>(t0), utc));
>+  EXPECT_EQ("03:04:05",
>+            format(kFmt, chrono::time_point_cast<chrono::seconds>(t0), utc));
>+  EXPECT_EQ("03:04:05",
>+            format(kFmt, chrono::time_point_cast<absl::time_internal::cctz::seconds>(t0), utc));
>+  EXPECT_EQ("03:04:00",
>+            format(kFmt, chrono::time_point_cast<chrono::minutes>(t0), utc));
>+  EXPECT_EQ("03:00:00",
>+            format(kFmt, chrono::time_point_cast<chrono::hours>(t0), utc));
>+}
>+
>+TEST(Format, TimePointExtendedResolution) {
>+  const char kFmt[] = "%H:%M:%E*S";
>+  const time_zone utc = utc_time_zone();
>+  const time_point<absl::time_internal::cctz::seconds> tp =
>+      chrono::time_point_cast<absl::time_internal::cctz::seconds>(
>+          chrono::system_clock::from_time_t(0)) +
>+      chrono::hours(12) + chrono::minutes(34) + chrono::seconds(56);
>+
>+  EXPECT_EQ(
>+      "12:34:56.123456789012345",
>+      detail::format(kFmt, tp, detail::femtoseconds(123456789012345), utc));
>+  EXPECT_EQ(
>+      "12:34:56.012345678901234",
>+      detail::format(kFmt, tp, detail::femtoseconds(12345678901234), utc));
>+  EXPECT_EQ(
>+      "12:34:56.001234567890123",
>+      detail::format(kFmt, tp, detail::femtoseconds(1234567890123), utc));
>+  EXPECT_EQ(
>+      "12:34:56.000123456789012",
>+      detail::format(kFmt, tp, detail::femtoseconds(123456789012), utc));
>+
>+  EXPECT_EQ("12:34:56.000000000000123",
>+            detail::format(kFmt, tp, detail::femtoseconds(123), utc));
>+  EXPECT_EQ("12:34:56.000000000000012",
>+            detail::format(kFmt, tp, detail::femtoseconds(12), utc));
>+  EXPECT_EQ("12:34:56.000000000000001",
>+            detail::format(kFmt, tp, detail::femtoseconds(1), utc));
>+}
>+
>+TEST(Format, Basics) {
>+  time_zone tz = utc_time_zone();
>+  time_point<chrono::nanoseconds> tp = chrono::system_clock::from_time_t(0);
>+
>+  // Starts with a couple basic edge cases.
>+  EXPECT_EQ("", format("", tp, tz));
>+  EXPECT_EQ(" ", format(" ", tp, tz));
>+  EXPECT_EQ("  ", format("  ", tp, tz));
>+  EXPECT_EQ("xxx", format("xxx", tp, tz));
>+  std::string big(128, 'x');
>+  EXPECT_EQ(big, format(big, tp, tz));
>+  // Cause the 1024-byte buffer to grow.
>+  std::string bigger(100000, 'x');
>+  EXPECT_EQ(bigger, format(bigger, tp, tz));
>+
>+  tp += chrono::hours(13) + chrono::minutes(4) + chrono::seconds(5);
>+  tp += chrono::milliseconds(6) + chrono::microseconds(7) +
>+        chrono::nanoseconds(8);
>+  EXPECT_EQ("1970-01-01", format("%Y-%m-%d", tp, tz));
>+  EXPECT_EQ("13:04:05", format("%H:%M:%S", tp, tz));
>+  EXPECT_EQ("13:04:05.006", format("%H:%M:%E3S", tp, tz));
>+  EXPECT_EQ("13:04:05.006007", format("%H:%M:%E6S", tp, tz));
>+  EXPECT_EQ("13:04:05.006007008", format("%H:%M:%E9S", tp, tz));
>+}
>+
>+TEST(Format, PosixConversions) {
>+  const time_zone tz = utc_time_zone();
>+  auto tp = chrono::system_clock::from_time_t(0);
>+
>+  TestFormatSpecifier(tp, tz, "%d", "01");
>+  TestFormatSpecifier(tp, tz, "%e", " 1");  // extension but internal support
>+  TestFormatSpecifier(tp, tz, "%H", "00");
>+  TestFormatSpecifier(tp, tz, "%I", "12");
>+  TestFormatSpecifier(tp, tz, "%j", "001");
>+  TestFormatSpecifier(tp, tz, "%m", "01");
>+  TestFormatSpecifier(tp, tz, "%M", "00");
>+  TestFormatSpecifier(tp, tz, "%S", "00");
>+  TestFormatSpecifier(tp, tz, "%U", "00");
>+  TestFormatSpecifier(tp, tz, "%w", "4");  // 4=Thursday
>+  TestFormatSpecifier(tp, tz, "%W", "00");
>+  TestFormatSpecifier(tp, tz, "%y", "70");
>+  TestFormatSpecifier(tp, tz, "%Y", "1970");
>+  TestFormatSpecifier(tp, tz, "%z", "+0000");
>+  TestFormatSpecifier(tp, tz, "%Z", "UTC");
>+  TestFormatSpecifier(tp, tz, "%%", "%");
>+
>+#if defined(__linux__)
>+  // SU/C99/TZ extensions
>+  TestFormatSpecifier(tp, tz, "%C", "19");
>+  TestFormatSpecifier(tp, tz, "%D", "01/01/70");
>+  TestFormatSpecifier(tp, tz, "%F", "1970-01-01");
>+  TestFormatSpecifier(tp, tz, "%g", "70");
>+  TestFormatSpecifier(tp, tz, "%G", "1970");
>+  TestFormatSpecifier(tp, tz, "%k", " 0");
>+  TestFormatSpecifier(tp, tz, "%l", "12");
>+  TestFormatSpecifier(tp, tz, "%n", "\n");
>+  TestFormatSpecifier(tp, tz, "%R", "00:00");
>+  TestFormatSpecifier(tp, tz, "%t", "\t");
>+  TestFormatSpecifier(tp, tz, "%T", "00:00:00");
>+  TestFormatSpecifier(tp, tz, "%u", "4");  // 4=Thursday
>+  TestFormatSpecifier(tp, tz, "%V", "01");
>+  TestFormatSpecifier(tp, tz, "%s", "0");
>+#endif
>+}
>+
>+TEST(Format, LocaleSpecific) {
>+  const time_zone tz = utc_time_zone();
>+  auto tp = chrono::system_clock::from_time_t(0);
>+
>+  TestFormatSpecifier(tp, tz, "%a", "Thu");
>+  TestFormatSpecifier(tp, tz, "%A", "Thursday");
>+  TestFormatSpecifier(tp, tz, "%b", "Jan");
>+  TestFormatSpecifier(tp, tz, "%B", "January");
>+
>+  // %c should at least produce the numeric year and time-of-day.
>+  const std::string s = format("%c", tp, utc_time_zone());
>+  EXPECT_THAT(s, testing::HasSubstr("1970"));
>+  EXPECT_THAT(s, testing::HasSubstr("00:00:00"));
>+
>+  TestFormatSpecifier(tp, tz, "%p", "AM");
>+  TestFormatSpecifier(tp, tz, "%x", "01/01/70");
>+  TestFormatSpecifier(tp, tz, "%X", "00:00:00");
>+
>+#if defined(__linux__)
>+  // SU/C99/TZ extensions
>+  TestFormatSpecifier(tp, tz, "%h", "Jan");  // Same as %b
>+  TestFormatSpecifier(tp, tz, "%P", "am");
>+  TestFormatSpecifier(tp, tz, "%r", "12:00:00 AM");
>+
>+  // Modified conversion specifiers %E_
>+  TestFormatSpecifier(tp, tz, "%Ec", "Thu Jan  1 00:00:00 1970");
>+  TestFormatSpecifier(tp, tz, "%EC", "19");
>+  TestFormatSpecifier(tp, tz, "%Ex", "01/01/70");
>+  TestFormatSpecifier(tp, tz, "%EX", "00:00:00");
>+  TestFormatSpecifier(tp, tz, "%Ey", "70");
>+  TestFormatSpecifier(tp, tz, "%EY", "1970");
>+
>+  // Modified conversion specifiers %O_
>+  TestFormatSpecifier(tp, tz, "%Od", "01");
>+  TestFormatSpecifier(tp, tz, "%Oe", " 1");
>+  TestFormatSpecifier(tp, tz, "%OH", "00");
>+  TestFormatSpecifier(tp, tz, "%OI", "12");
>+  TestFormatSpecifier(tp, tz, "%Om", "01");
>+  TestFormatSpecifier(tp, tz, "%OM", "00");
>+  TestFormatSpecifier(tp, tz, "%OS", "00");
>+  TestFormatSpecifier(tp, tz, "%Ou", "4");  // 4=Thursday
>+  TestFormatSpecifier(tp, tz, "%OU", "00");
>+  TestFormatSpecifier(tp, tz, "%OV", "01");
>+  TestFormatSpecifier(tp, tz, "%Ow", "4");  // 4=Thursday
>+  TestFormatSpecifier(tp, tz, "%OW", "00");
>+  TestFormatSpecifier(tp, tz, "%Oy", "70");
>+#endif
>+}
>+
>+TEST(Format, Escaping) {
>+  const time_zone tz = utc_time_zone();
>+  auto tp = chrono::system_clock::from_time_t(0);
>+
>+  TestFormatSpecifier(tp, tz, "%%", "%");
>+  TestFormatSpecifier(tp, tz, "%%a", "%a");
>+  TestFormatSpecifier(tp, tz, "%%b", "%b");
>+  TestFormatSpecifier(tp, tz, "%%Ea", "%Ea");
>+  TestFormatSpecifier(tp, tz, "%%Es", "%Es");
>+  TestFormatSpecifier(tp, tz, "%%E3S", "%E3S");
>+  TestFormatSpecifier(tp, tz, "%%OS", "%OS");
>+  TestFormatSpecifier(tp, tz, "%%O3S", "%O3S");
>+
>+  // Multiple levels of escaping.
>+  TestFormatSpecifier(tp, tz, "%%%Y", "%1970");
>+  TestFormatSpecifier(tp, tz, "%%%E3S", "%00.000");
>+  TestFormatSpecifier(tp, tz, "%%%%E3S", "%%E3S");
>+}
>+
>+TEST(Format, ExtendedSeconds) {
>+  const time_zone tz = utc_time_zone();
>+
>+  // No subseconds.
>+  time_point<chrono::nanoseconds> tp = chrono::system_clock::from_time_t(0);
>+  tp += chrono::seconds(5);
>+  EXPECT_EQ("05", format("%E*S", tp, tz));
>+  EXPECT_EQ("05", format("%E0S", tp, tz));
>+  EXPECT_EQ("05.0", format("%E1S", tp, tz));
>+  EXPECT_EQ("05.00", format("%E2S", tp, tz));
>+  EXPECT_EQ("05.000", format("%E3S", tp, tz));
>+  EXPECT_EQ("05.0000", format("%E4S", tp, tz));
>+  EXPECT_EQ("05.00000", format("%E5S", tp, tz));
>+  EXPECT_EQ("05.000000", format("%E6S", tp, tz));
>+  EXPECT_EQ("05.0000000", format("%E7S", tp, tz));
>+  EXPECT_EQ("05.00000000", format("%E8S", tp, tz));
>+  EXPECT_EQ("05.000000000", format("%E9S", tp, tz));
>+  EXPECT_EQ("05.0000000000", format("%E10S", tp, tz));
>+  EXPECT_EQ("05.00000000000", format("%E11S", tp, tz));
>+  EXPECT_EQ("05.000000000000", format("%E12S", tp, tz));
>+  EXPECT_EQ("05.0000000000000", format("%E13S", tp, tz));
>+  EXPECT_EQ("05.00000000000000", format("%E14S", tp, tz));
>+  EXPECT_EQ("05.000000000000000", format("%E15S", tp, tz));
>+
>+  // With subseconds.
>+  tp += chrono::milliseconds(6) + chrono::microseconds(7) +
>+        chrono::nanoseconds(8);
>+  EXPECT_EQ("05.006007008", format("%E*S", tp, tz));
>+  EXPECT_EQ("05", format("%E0S", tp, tz));
>+  EXPECT_EQ("05.0", format("%E1S", tp, tz));
>+  EXPECT_EQ("05.00", format("%E2S", tp, tz));
>+  EXPECT_EQ("05.006", format("%E3S", tp, tz));
>+  EXPECT_EQ("05.0060", format("%E4S", tp, tz));
>+  EXPECT_EQ("05.00600", format("%E5S", tp, tz));
>+  EXPECT_EQ("05.006007", format("%E6S", tp, tz));
>+  EXPECT_EQ("05.0060070", format("%E7S", tp, tz));
>+  EXPECT_EQ("05.00600700", format("%E8S", tp, tz));
>+  EXPECT_EQ("05.006007008", format("%E9S", tp, tz));
>+  EXPECT_EQ("05.0060070080", format("%E10S", tp, tz));
>+  EXPECT_EQ("05.00600700800", format("%E11S", tp, tz));
>+  EXPECT_EQ("05.006007008000", format("%E12S", tp, tz));
>+  EXPECT_EQ("05.0060070080000", format("%E13S", tp, tz));
>+  EXPECT_EQ("05.00600700800000", format("%E14S", tp, tz));
>+  EXPECT_EQ("05.006007008000000", format("%E15S", tp, tz));
>+
>+  // Times before the Unix epoch.
>+  tp = chrono::system_clock::from_time_t(0) + chrono::microseconds(-1);
>+  EXPECT_EQ("1969-12-31 23:59:59.999999",
>+            format("%Y-%m-%d %H:%M:%E*S", tp, tz));
>+
>+  // Here is a "%E*S" case we got wrong for a while.  While the first
>+  // instant below is correctly rendered as "...:07.333304", the second
>+  // one used to appear as "...:07.33330499999999999".
>+  tp = chrono::system_clock::from_time_t(0) +
>+       chrono::microseconds(1395024427333304);
>+  EXPECT_EQ("2014-03-17 02:47:07.333304",
>+            format("%Y-%m-%d %H:%M:%E*S", tp, tz));
>+  tp += chrono::microseconds(1);
>+  EXPECT_EQ("2014-03-17 02:47:07.333305",
>+            format("%Y-%m-%d %H:%M:%E*S", tp, tz));
>+}
>+
>+TEST(Format, ExtendedSubeconds) {
>+  const time_zone tz = utc_time_zone();
>+
>+  // No subseconds.
>+  time_point<chrono::nanoseconds> tp = chrono::system_clock::from_time_t(0);
>+  tp += chrono::seconds(5);
>+  EXPECT_EQ("0", format("%E*f", tp, tz));
>+  EXPECT_EQ("", format("%E0f", tp, tz));
>+  EXPECT_EQ("0", format("%E1f", tp, tz));
>+  EXPECT_EQ("00", format("%E2f", tp, tz));
>+  EXPECT_EQ("000", format("%E3f", tp, tz));
>+  EXPECT_EQ("0000", format("%E4f", tp, tz));
>+  EXPECT_EQ("00000", format("%E5f", tp, tz));
>+  EXPECT_EQ("000000", format("%E6f", tp, tz));
>+  EXPECT_EQ("0000000", format("%E7f", tp, tz));
>+  EXPECT_EQ("00000000", format("%E8f", tp, tz));
>+  EXPECT_EQ("000000000", format("%E9f", tp, tz));
>+  EXPECT_EQ("0000000000", format("%E10f", tp, tz));
>+  EXPECT_EQ("00000000000", format("%E11f", tp, tz));
>+  EXPECT_EQ("000000000000", format("%E12f", tp, tz));
>+  EXPECT_EQ("0000000000000", format("%E13f", tp, tz));
>+  EXPECT_EQ("00000000000000", format("%E14f", tp, tz));
>+  EXPECT_EQ("000000000000000", format("%E15f", tp, tz));
>+
>+  // With subseconds.
>+  tp += chrono::milliseconds(6) + chrono::microseconds(7) +
>+        chrono::nanoseconds(8);
>+  EXPECT_EQ("006007008", format("%E*f", tp, tz));
>+  EXPECT_EQ("", format("%E0f", tp, tz));
>+  EXPECT_EQ("0", format("%E1f", tp, tz));
>+  EXPECT_EQ("00", format("%E2f", tp, tz));
>+  EXPECT_EQ("006", format("%E3f", tp, tz));
>+  EXPECT_EQ("0060", format("%E4f", tp, tz));
>+  EXPECT_EQ("00600", format("%E5f", tp, tz));
>+  EXPECT_EQ("006007", format("%E6f", tp, tz));
>+  EXPECT_EQ("0060070", format("%E7f", tp, tz));
>+  EXPECT_EQ("00600700", format("%E8f", tp, tz));
>+  EXPECT_EQ("006007008", format("%E9f", tp, tz));
>+  EXPECT_EQ("0060070080", format("%E10f", tp, tz));
>+  EXPECT_EQ("00600700800", format("%E11f", tp, tz));
>+  EXPECT_EQ("006007008000", format("%E12f", tp, tz));
>+  EXPECT_EQ("0060070080000", format("%E13f", tp, tz));
>+  EXPECT_EQ("00600700800000", format("%E14f", tp, tz));
>+  EXPECT_EQ("006007008000000", format("%E15f", tp, tz));
>+
>+  // Times before the Unix epoch.
>+  tp = chrono::system_clock::from_time_t(0) + chrono::microseconds(-1);
>+  EXPECT_EQ("1969-12-31 23:59:59.999999",
>+            format("%Y-%m-%d %H:%M:%S.%E*f", tp, tz));
>+
>+  // Here is a "%E*S" case we got wrong for a while.  While the first
>+  // instant below is correctly rendered as "...:07.333304", the second
>+  // one used to appear as "...:07.33330499999999999".
>+  tp = chrono::system_clock::from_time_t(0) +
>+       chrono::microseconds(1395024427333304);
>+  EXPECT_EQ("2014-03-17 02:47:07.333304",
>+            format("%Y-%m-%d %H:%M:%S.%E*f", tp, tz));
>+  tp += chrono::microseconds(1);
>+  EXPECT_EQ("2014-03-17 02:47:07.333305",
>+            format("%Y-%m-%d %H:%M:%S.%E*f", tp, tz));
>+}
>+
>+TEST(Format, CompareExtendSecondsVsSubseconds) {
>+  const time_zone tz = utc_time_zone();
>+
>+  // This test case illustrates the differences/similarities between:
>+  //   fmt_A: %E<prec>S
>+  //   fmt_B: %S.%E<prec>f
>+  auto fmt_A = [](const std::string& prec) { return "%E" + prec + "S"; };
>+  auto fmt_B = [](const std::string& prec) { return "%S.%E" + prec + "f"; };
>+
>+  // No subseconds:
>+  time_point<chrono::nanoseconds> tp = chrono::system_clock::from_time_t(0);
>+  tp += chrono::seconds(5);
>+  // ... %E*S and %S.%E*f are different.
>+  EXPECT_EQ("05", format(fmt_A("*"), tp, tz));
>+  EXPECT_EQ("05.0", format(fmt_B("*"), tp, tz));
>+  // ... %E0S and %S.%E0f are different.
>+  EXPECT_EQ("05", format(fmt_A("0"), tp, tz));
>+  EXPECT_EQ("05.", format(fmt_B("0"), tp, tz));
>+  // ... %E<prec>S and %S.%E<prec>f are the same for prec in [1:15].
>+  for (int prec = 1; prec <= 15; ++prec) {
>+    const std::string a = format(fmt_A(std::to_string(prec)), tp, tz);
>+    const std::string b = format(fmt_B(std::to_string(prec)), tp, tz);
>+    EXPECT_EQ(a, b) << "prec=" << prec;
>+  }
>+
>+  // With subseconds:
>+  // ... %E*S and %S.%E*f are the same.
>+  tp += chrono::milliseconds(6) + chrono::microseconds(7) +
>+        chrono::nanoseconds(8);
>+  EXPECT_EQ("05.006007008", format(fmt_A("*"), tp, tz));
>+  EXPECT_EQ("05.006007008", format(fmt_B("*"), tp, tz));
>+  // ... %E0S and %S.%E0f are different.
>+  EXPECT_EQ("05", format(fmt_A("0"), tp, tz));
>+  EXPECT_EQ("05.", format(fmt_B("0"), tp, tz));
>+  // ... %E<prec>S and %S.%E<prec>f are the same for prec in [1:15].
>+  for (int prec = 1; prec <= 15; ++prec) {
>+    const std::string a = format(fmt_A(std::to_string(prec)), tp, tz);
>+    const std::string b = format(fmt_B(std::to_string(prec)), tp, tz);
>+    EXPECT_EQ(a, b) << "prec=" << prec;
>+  }
>+}
>+
>+TEST(Format, ExtendedOffset) {
>+  auto tp = chrono::system_clock::from_time_t(0);
>+
>+  time_zone tz = utc_time_zone();
>+  TestFormatSpecifier(tp, tz, "%Ez", "+00:00");
>+
>+  EXPECT_TRUE(load_time_zone("America/New_York", &tz));
>+  TestFormatSpecifier(tp, tz, "%Ez", "-05:00");
>+
>+  EXPECT_TRUE(load_time_zone("America/Los_Angeles", &tz));
>+  TestFormatSpecifier(tp, tz, "%Ez", "-08:00");
>+
>+  EXPECT_TRUE(load_time_zone("Australia/Sydney", &tz));
>+  TestFormatSpecifier(tp, tz, "%Ez", "+10:00");
>+
>+  EXPECT_TRUE(load_time_zone("Africa/Monrovia", &tz));
>+  // The true offset is -00:44:30 but %z only gives (truncated) minutes.
>+  TestFormatSpecifier(tp, tz, "%z", "-0044");
>+  TestFormatSpecifier(tp, tz, "%Ez", "-00:44");
>+}
>+
>+TEST(Format, ExtendedSecondOffset) {
>+  const time_zone utc = utc_time_zone();
>+  time_point<chrono::seconds> tp;
>+  time_zone tz;
>+
>+  EXPECT_TRUE(load_time_zone("America/New_York", &tz));
>+  tp = convert(civil_second(1883, 11, 18, 16, 59, 59), utc);
>+  if (tz.lookup(tp).offset == -5 * 60 * 60) {
>+    // It looks like the tzdata is only 32 bit (probably macOS),
>+    // which bottoms out at 1901-12-13T20:45:52+00:00.
>+  } else {
>+    TestFormatSpecifier(tp, tz, "%E*z", "-04:56:02");
>+    TestFormatSpecifier(tp, tz, "%Ez", "-04:56");
>+  }
>+  tp += chrono::seconds(1);
>+  TestFormatSpecifier(tp, tz, "%E*z", "-05:00:00");
>+
>+  EXPECT_TRUE(load_time_zone("Europe/Moscow", &tz));
>+  tp = convert(civil_second(1919, 6, 30, 23, 59, 59), utc);
>+  if (VersionCmp(tz, "2016g") >= 0) {
>+    TestFormatSpecifier(tp, tz, "%E*z", "+04:31:19");
>+    TestFormatSpecifier(tp, tz, "%Ez", "+04:31");
>+  }
>+  tp += chrono::seconds(1);
>+  TestFormatSpecifier(tp, tz, "%E*z", "+04:00:00");
>+}
>+
>+TEST(Format, ExtendedYears) {
>+  const time_zone utc = utc_time_zone();
>+  const char e4y_fmt[] = "%E4Y%m%d";  // no separators
>+
>+  // %E4Y zero-pads the year to produce at least 4 chars, including the sign.
>+  auto tp = convert(civil_second(-999, 11, 27, 0, 0, 0), utc);
>+  EXPECT_EQ("-9991127", format(e4y_fmt, tp, utc));
>+  tp = convert(civil_second(-99, 11, 27, 0, 0, 0), utc);
>+  EXPECT_EQ("-0991127", format(e4y_fmt, tp, utc));
>+  tp = convert(civil_second(-9, 11, 27, 0, 0, 0), utc);
>+  EXPECT_EQ("-0091127", format(e4y_fmt, tp, utc));
>+  tp = convert(civil_second(-1, 11, 27, 0, 0, 0), utc);
>+  EXPECT_EQ("-0011127", format(e4y_fmt, tp, utc));
>+  tp = convert(civil_second(0, 11, 27, 0, 0, 0), utc);
>+  EXPECT_EQ("00001127", format(e4y_fmt, tp, utc));
>+  tp = convert(civil_second(1, 11, 27, 0, 0, 0), utc);
>+  EXPECT_EQ("00011127", format(e4y_fmt, tp, utc));
>+  tp = convert(civil_second(9, 11, 27, 0, 0, 0), utc);
>+  EXPECT_EQ("00091127", format(e4y_fmt, tp, utc));
>+  tp = convert(civil_second(99, 11, 27, 0, 0, 0), utc);
>+  EXPECT_EQ("00991127", format(e4y_fmt, tp, utc));
>+  tp = convert(civil_second(999, 11, 27, 0, 0, 0), utc);
>+  EXPECT_EQ("09991127", format(e4y_fmt, tp, utc));
>+  tp = convert(civil_second(9999, 11, 27, 0, 0, 0), utc);
>+  EXPECT_EQ("99991127", format(e4y_fmt, tp, utc));
>+
>+  // When the year is outside [-999:9999], more than 4 chars are produced.
>+  tp = convert(civil_second(-1000, 11, 27, 0, 0, 0), utc);
>+  EXPECT_EQ("-10001127", format(e4y_fmt, tp, utc));
>+  tp = convert(civil_second(10000, 11, 27, 0, 0, 0), utc);
>+  EXPECT_EQ("100001127", format(e4y_fmt, tp, utc));
>+}
>+
>+TEST(Format, RFC3339Format) {
>+  time_zone tz;
>+  EXPECT_TRUE(load_time_zone("America/Los_Angeles", &tz));
>+
>+  time_point<chrono::nanoseconds> tp =
>+      convert(civil_second(1977, 6, 28, 9, 8, 7), tz);
>+  EXPECT_EQ("1977-06-28T09:08:07-07:00", format(RFC3339_full, tp, tz));
>+  EXPECT_EQ("1977-06-28T09:08:07-07:00", format(RFC3339_sec, tp, tz));
>+
>+  tp += chrono::milliseconds(100);
>+  EXPECT_EQ("1977-06-28T09:08:07.1-07:00", format(RFC3339_full, tp, tz));
>+  EXPECT_EQ("1977-06-28T09:08:07-07:00", format(RFC3339_sec, tp, tz));
>+
>+  tp += chrono::milliseconds(20);
>+  EXPECT_EQ("1977-06-28T09:08:07.12-07:00", format(RFC3339_full, tp, tz));
>+  EXPECT_EQ("1977-06-28T09:08:07-07:00", format(RFC3339_sec, tp, tz));
>+
>+  tp += chrono::milliseconds(3);
>+  EXPECT_EQ("1977-06-28T09:08:07.123-07:00", format(RFC3339_full, tp, tz));
>+  EXPECT_EQ("1977-06-28T09:08:07-07:00", format(RFC3339_sec, tp, tz));
>+
>+  tp += chrono::microseconds(400);
>+  EXPECT_EQ("1977-06-28T09:08:07.1234-07:00", format(RFC3339_full, tp, tz));
>+  EXPECT_EQ("1977-06-28T09:08:07-07:00", format(RFC3339_sec, tp, tz));
>+
>+  tp += chrono::microseconds(50);
>+  EXPECT_EQ("1977-06-28T09:08:07.12345-07:00", format(RFC3339_full, tp, tz));
>+  EXPECT_EQ("1977-06-28T09:08:07-07:00", format(RFC3339_sec, tp, tz));
>+
>+  tp += chrono::microseconds(6);
>+  EXPECT_EQ("1977-06-28T09:08:07.123456-07:00", format(RFC3339_full, tp, tz));
>+  EXPECT_EQ("1977-06-28T09:08:07-07:00", format(RFC3339_sec, tp, tz));
>+
>+  tp += chrono::nanoseconds(700);
>+  EXPECT_EQ("1977-06-28T09:08:07.1234567-07:00", format(RFC3339_full, tp, tz));
>+  EXPECT_EQ("1977-06-28T09:08:07-07:00", format(RFC3339_sec, tp, tz));
>+
>+  tp += chrono::nanoseconds(80);
>+  EXPECT_EQ("1977-06-28T09:08:07.12345678-07:00", format(RFC3339_full, tp, tz));
>+  EXPECT_EQ("1977-06-28T09:08:07-07:00", format(RFC3339_sec, tp, tz));
>+
>+  tp += chrono::nanoseconds(9);
>+  EXPECT_EQ("1977-06-28T09:08:07.123456789-07:00",
>+            format(RFC3339_full, tp, tz));
>+  EXPECT_EQ("1977-06-28T09:08:07-07:00", format(RFC3339_sec, tp, tz));
>+}
>+
>+TEST(Format, RFC1123Format) {  // locale specific
>+  time_zone tz;
>+  EXPECT_TRUE(load_time_zone("America/Los_Angeles", &tz));
>+
>+  auto tp = convert(civil_second(1977, 6, 28, 9, 8, 7), tz);
>+  EXPECT_EQ("Tue, 28 Jun 1977 09:08:07 -0700", format(RFC1123_full, tp, tz));
>+  EXPECT_EQ("28 Jun 1977 09:08:07 -0700", format(RFC1123_no_wday, tp, tz));
>+}
>+
>+//
>+// Testing parse()
>+//
>+
>+TEST(Parse, TimePointResolution) {
>+  const char kFmt[] = "%H:%M:%E*S";
>+  const time_zone utc = utc_time_zone();
>+
>+  time_point<chrono::nanoseconds> tp_ns;
>+  EXPECT_TRUE(parse(kFmt, "03:04:05.123456789", utc, &tp_ns));
>+  EXPECT_EQ("03:04:05.123456789", format(kFmt, tp_ns, utc));
>+  EXPECT_TRUE(parse(kFmt, "03:04:05.123456", utc, &tp_ns));
>+  EXPECT_EQ("03:04:05.123456", format(kFmt, tp_ns, utc));
>+
>+  time_point<chrono::microseconds> tp_us;
>+  EXPECT_TRUE(parse(kFmt, "03:04:05.123456789", utc, &tp_us));
>+  EXPECT_EQ("03:04:05.123456", format(kFmt, tp_us, utc));
>+  EXPECT_TRUE(parse(kFmt, "03:04:05.123456", utc, &tp_us));
>+  EXPECT_EQ("03:04:05.123456", format(kFmt, tp_us, utc));
>+  EXPECT_TRUE(parse(kFmt, "03:04:05.123", utc, &tp_us));
>+  EXPECT_EQ("03:04:05.123", format(kFmt, tp_us, utc));
>+
>+  time_point<chrono::milliseconds> tp_ms;
>+  EXPECT_TRUE(parse(kFmt, "03:04:05.123456", utc, &tp_ms));
>+  EXPECT_EQ("03:04:05.123", format(kFmt, tp_ms, utc));
>+  EXPECT_TRUE(parse(kFmt, "03:04:05.123", utc, &tp_ms));
>+  EXPECT_EQ("03:04:05.123", format(kFmt, tp_ms, utc));
>+  EXPECT_TRUE(parse(kFmt, "03:04:05", utc, &tp_ms));
>+  EXPECT_EQ("03:04:05", format(kFmt, tp_ms, utc));
>+
>+  time_point<chrono::seconds> tp_s;
>+  EXPECT_TRUE(parse(kFmt, "03:04:05.123", utc, &tp_s));
>+  EXPECT_EQ("03:04:05", format(kFmt, tp_s, utc));
>+  EXPECT_TRUE(parse(kFmt, "03:04:05", utc, &tp_s));
>+  EXPECT_EQ("03:04:05", format(kFmt, tp_s, utc));
>+
>+  time_point<chrono::minutes> tp_m;
>+  EXPECT_TRUE(parse(kFmt, "03:04:05", utc, &tp_m));
>+  EXPECT_EQ("03:04:00", format(kFmt, tp_m, utc));
>+
>+  time_point<chrono::hours> tp_h;
>+  EXPECT_TRUE(parse(kFmt, "03:04:05", utc, &tp_h));
>+  EXPECT_EQ("03:00:00", format(kFmt, tp_h, utc));
>+}
>+
>+TEST(Parse, TimePointExtendedResolution) {
>+  const char kFmt[] = "%H:%M:%E*S";
>+  const time_zone utc = utc_time_zone();
>+
>+  time_point<absl::time_internal::cctz::seconds> tp;
>+  detail::femtoseconds fs;
>+  EXPECT_TRUE(detail::parse(kFmt, "12:34:56.123456789012345", utc, &tp, &fs));
>+  EXPECT_EQ("12:34:56.123456789012345", detail::format(kFmt, tp, fs, utc));
>+  EXPECT_TRUE(detail::parse(kFmt, "12:34:56.012345678901234", utc, &tp, &fs));
>+  EXPECT_EQ("12:34:56.012345678901234", detail::format(kFmt, tp, fs, utc));
>+  EXPECT_TRUE(detail::parse(kFmt, "12:34:56.001234567890123", utc, &tp, &fs));
>+  EXPECT_EQ("12:34:56.001234567890123", detail::format(kFmt, tp, fs, utc));
>+  EXPECT_TRUE(detail::parse(kFmt, "12:34:56.000000000000123", utc, &tp, &fs));
>+  EXPECT_EQ("12:34:56.000000000000123", detail::format(kFmt, tp, fs, utc));
>+  EXPECT_TRUE(detail::parse(kFmt, "12:34:56.000000000000012", utc, &tp, &fs));
>+  EXPECT_EQ("12:34:56.000000000000012", detail::format(kFmt, tp, fs, utc));
>+  EXPECT_TRUE(detail::parse(kFmt, "12:34:56.000000000000001", utc, &tp, &fs));
>+  EXPECT_EQ("12:34:56.000000000000001", detail::format(kFmt, tp, fs, utc));
>+}
>+
>+TEST(Parse, Basics) {
>+  time_zone tz = utc_time_zone();
>+  time_point<chrono::nanoseconds> tp =
>+      chrono::system_clock::from_time_t(1234567890);
>+
>+  // Simple edge cases.
>+  EXPECT_TRUE(parse("", "", tz, &tp));
>+  EXPECT_EQ(chrono::system_clock::from_time_t(0), tp);  // everything defaulted
>+  EXPECT_TRUE(parse(" ", " ", tz, &tp));
>+  EXPECT_TRUE(parse("  ", "  ", tz, &tp));
>+  EXPECT_TRUE(parse("x", "x", tz, &tp));
>+  EXPECT_TRUE(parse("xxx", "xxx", tz, &tp));
>+
>+  EXPECT_TRUE(
>+      parse("%Y-%m-%d %H:%M:%S %z", "2013-06-28 19:08:09 -0800", tz, &tp));
>+  ExpectTime(tp, tz, 2013, 6, 29, 3, 8, 9, 0, false, "UTC");
>+}
>+
>+TEST(Parse, WithTimeZone) {
>+  time_zone tz;
>+  EXPECT_TRUE(load_time_zone("America/Los_Angeles", &tz));
>+  time_point<chrono::nanoseconds> tp;
>+
>+  // We can parse a std::string without a UTC offset if we supply a timezone.
>+  EXPECT_TRUE(parse("%Y-%m-%d %H:%M:%S", "2013-06-28 19:08:09", tz, &tp));
>+  ExpectTime(tp, tz, 2013, 6, 28, 19, 8, 9, -7 * 60 * 60, true, "PDT");
>+
>+  // But the timezone is ignored when a UTC offset is present.
>+  EXPECT_TRUE(parse("%Y-%m-%d %H:%M:%S %z", "2013-06-28 19:08:09 +0800",
>+                    utc_time_zone(), &tp));
>+  ExpectTime(tp, tz, 2013, 6, 28, 19 - 8 - 7, 8, 9, -7 * 60 * 60, true, "PDT");
>+
>+  // Check a skipped time (a Spring DST transition).  parse() returns
>+  // the preferred-offset result, as defined for ConvertDateTime().
>+  EXPECT_TRUE(parse("%Y-%m-%d %H:%M:%S", "2011-03-13 02:15:00", tz, &tp));
>+  ExpectTime(tp, tz, 2011, 3, 13, 3, 15, 0, -7 * 60 * 60, true, "PDT");
>+
>+  // Check a repeated time (a Fall DST transition).  parse() returns
>+  // the preferred-offset result, as defined for ConvertDateTime().
>+  EXPECT_TRUE(parse("%Y-%m-%d %H:%M:%S", "2011-11-06 01:15:00", tz, &tp));
>+  ExpectTime(tp, tz, 2011, 11, 6, 1, 15, 0, -7 * 60 * 60, true, "PDT");
>+}
>+
>+TEST(Parse, LeapSecond) {
>+  time_zone tz;
>+  EXPECT_TRUE(load_time_zone("America/Los_Angeles", &tz));
>+  time_point<chrono::nanoseconds> tp;
>+
>+  // ":59" -> ":59"
>+  EXPECT_TRUE(parse(RFC3339_full, "2013-06-28T07:08:59-08:00", tz, &tp));
>+  ExpectTime(tp, tz, 2013, 6, 28, 8, 8, 59, -7 * 60 * 60, true, "PDT");
>+
>+  // ":59.5" -> ":59.5"
>+  EXPECT_TRUE(parse(RFC3339_full, "2013-06-28T07:08:59.5-08:00", tz, &tp));
>+  ExpectTime(tp, tz, 2013, 6, 28, 8, 8, 59, -7 * 60 * 60, true, "PDT");
>+
>+  // ":60" -> ":00"
>+  EXPECT_TRUE(parse(RFC3339_full, "2013-06-28T07:08:60-08:00", tz, &tp));
>+  ExpectTime(tp, tz, 2013, 6, 28, 8, 9, 0, -7 * 60 * 60, true, "PDT");
>+
>+  // ":60.5" -> ":00.0"
>+  EXPECT_TRUE(parse(RFC3339_full, "2013-06-28T07:08:60.5-08:00", tz, &tp));
>+  ExpectTime(tp, tz, 2013, 6, 28, 8, 9, 0, -7 * 60 * 60, true, "PDT");
>+
>+  // ":61" -> error
>+  EXPECT_FALSE(parse(RFC3339_full, "2013-06-28T07:08:61-08:00", tz, &tp));
>+}
>+
>+TEST(Parse, ErrorCases) {
>+  const time_zone tz = utc_time_zone();
>+  auto tp = chrono::system_clock::from_time_t(0);
>+
>+  // Illegal trailing data.
>+  EXPECT_FALSE(parse("%S", "123", tz, &tp));
>+
>+  // Can't parse an illegal format specifier.
>+  EXPECT_FALSE(parse("%Q", "x", tz, &tp));
>+
>+  // Fails because of trailing, unparsed data "blah".
>+  EXPECT_FALSE(parse("%m-%d", "2-3 blah", tz, &tp));
>+
>+  // Trailing whitespace is allowed.
>+  EXPECT_TRUE(parse("%m-%d", "2-3  ", tz, &tp));
>+  EXPECT_EQ(2, convert(tp, utc_time_zone()).month());
>+  EXPECT_EQ(3, convert(tp, utc_time_zone()).day());
>+
>+  // Feb 31 requires normalization.
>+  EXPECT_FALSE(parse("%m-%d", "2-31", tz, &tp));
>+
>+  // Check that we cannot have spaces in UTC offsets.
>+  EXPECT_TRUE(parse("%z", "-0203", tz, &tp));
>+  EXPECT_FALSE(parse("%z", "- 2 3", tz, &tp));
>+  EXPECT_TRUE(parse("%Ez", "-02:03", tz, &tp));
>+  EXPECT_FALSE(parse("%Ez", "- 2: 3", tz, &tp));
>+
>+  // Check that we reject other malformed UTC offsets.
>+  EXPECT_FALSE(parse("%Ez", "+-08:00", tz, &tp));
>+  EXPECT_FALSE(parse("%Ez", "-+08:00", tz, &tp));
>+
>+  // Check that we do not accept "-0" in fields that allow zero.
>+  EXPECT_FALSE(parse("%Y", "-0", tz, &tp));
>+  EXPECT_FALSE(parse("%E4Y", "-0", tz, &tp));
>+  EXPECT_FALSE(parse("%H", "-0", tz, &tp));
>+  EXPECT_FALSE(parse("%M", "-0", tz, &tp));
>+  EXPECT_FALSE(parse("%S", "-0", tz, &tp));
>+  EXPECT_FALSE(parse("%z", "+-000", tz, &tp));
>+  EXPECT_FALSE(parse("%Ez", "+-0:00", tz, &tp));
>+  EXPECT_FALSE(parse("%z", "-00-0", tz, &tp));
>+  EXPECT_FALSE(parse("%Ez", "-00:-0", tz, &tp));
>+}
>+
>+TEST(Parse, PosixConversions) {
>+  time_zone tz = utc_time_zone();
>+  auto tp = chrono::system_clock::from_time_t(0);
>+  const auto reset = convert(civil_second(1977, 6, 28, 9, 8, 7), tz);
>+
>+  tp = reset;
>+  EXPECT_TRUE(parse("%d", "15", tz, &tp));
>+  EXPECT_EQ(15, convert(tp, tz).day());
>+
>+  // %e is an extension, but is supported internally.
>+  tp = reset;
>+  EXPECT_TRUE(parse("%e", "15", tz, &tp));
>+  EXPECT_EQ(15, convert(tp, tz).day());  // Equivalent to %d
>+
>+  tp = reset;
>+  EXPECT_TRUE(parse("%H", "17", tz, &tp));
>+  EXPECT_EQ(17, convert(tp, tz).hour());
>+
>+  tp = reset;
>+  EXPECT_TRUE(parse("%I", "5", tz, &tp));
>+  EXPECT_EQ(5, convert(tp, tz).hour());
>+
>+  // %j is parsed but ignored.
>+  EXPECT_TRUE(parse("%j", "32", tz, &tp));
>+
>+  tp = reset;
>+  EXPECT_TRUE(parse("%m", "11", tz, &tp));
>+  EXPECT_EQ(11, convert(tp, tz).month());
>+
>+  tp = reset;
>+  EXPECT_TRUE(parse("%M", "33", tz, &tp));
>+  EXPECT_EQ(33, convert(tp, tz).minute());
>+
>+  tp = reset;
>+  EXPECT_TRUE(parse("%S", "55", tz, &tp));
>+  EXPECT_EQ(55, convert(tp, tz).second());
>+
>+  // %U is parsed but ignored.
>+  EXPECT_TRUE(parse("%U", "15", tz, &tp));
>+
>+  // %w is parsed but ignored.
>+  EXPECT_TRUE(parse("%w", "2", tz, &tp));
>+
>+  // %W is parsed but ignored.
>+  EXPECT_TRUE(parse("%W", "22", tz, &tp));
>+
>+  tp = reset;
>+  EXPECT_TRUE(parse("%y", "04", tz, &tp));
>+  EXPECT_EQ(2004, convert(tp, tz).year());
>+
>+  tp = reset;
>+  EXPECT_TRUE(parse("%Y", "2004", tz, &tp));
>+  EXPECT_EQ(2004, convert(tp, tz).year());
>+
>+  EXPECT_TRUE(parse("%%", "%", tz, &tp));
>+
>+#if defined(__linux__)
>+  // SU/C99/TZ extensions
>+
>+  // Because we handle each (non-internal) specifier in a separate call
>+  // to strptime(), there is no way to group %C and %y together.  So we
>+  // just skip the %C/%y case.
>+#if 0
>+  tp = reset;
>+  EXPECT_TRUE(parse("%C %y", "20 04", tz, &tp));
>+  EXPECT_EQ(2004, convert(tp, tz).year());
>+#endif
>+
>+  tp = reset;
>+  EXPECT_TRUE(parse("%D", "02/03/04", tz, &tp));
>+  EXPECT_EQ(2, convert(tp, tz).month());
>+  EXPECT_EQ(3, convert(tp, tz).day());
>+  EXPECT_EQ(2004, convert(tp, tz).year());
>+
>+  EXPECT_TRUE(parse("%n", "\n", tz, &tp));
>+
>+  tp = reset;
>+  EXPECT_TRUE(parse("%R", "03:44", tz, &tp));
>+  EXPECT_EQ(3, convert(tp, tz).hour());
>+  EXPECT_EQ(44, convert(tp, tz).minute());
>+
>+  EXPECT_TRUE(parse("%t", "\t\v\f\n\r ", tz, &tp));
>+
>+  tp = reset;
>+  EXPECT_TRUE(parse("%T", "03:44:55", tz, &tp));
>+  EXPECT_EQ(3, convert(tp, tz).hour());
>+  EXPECT_EQ(44, convert(tp, tz).minute());
>+  EXPECT_EQ(55, convert(tp, tz).second());
>+
>+  tp = reset;
>+  EXPECT_TRUE(parse("%s", "1234567890", tz, &tp));
>+  EXPECT_EQ(chrono::system_clock::from_time_t(1234567890), tp);
>+
>+  // %s conversion, like %z/%Ez, pays no heed to the optional zone.
>+  time_zone lax;
>+  EXPECT_TRUE(load_time_zone("America/Los_Angeles", &lax));
>+  tp = reset;
>+  EXPECT_TRUE(parse("%s", "1234567890", lax, &tp));
>+  EXPECT_EQ(chrono::system_clock::from_time_t(1234567890), tp);
>+
>+  // This is most important when the time has the same YMDhms
>+  // breakdown in the zone as some other time.  For example, ...
>+  //  1414917000 in US/Pacific -> Sun Nov 2 01:30:00 2014 (PDT)
>+  //  1414920600 in US/Pacific -> Sun Nov 2 01:30:00 2014 (PST)
>+  tp = reset;
>+  EXPECT_TRUE(parse("%s", "1414917000", lax, &tp));
>+  EXPECT_EQ(chrono::system_clock::from_time_t(1414917000), tp);
>+  tp = reset;
>+  EXPECT_TRUE(parse("%s", "1414920600", lax, &tp));
>+  EXPECT_EQ(chrono::system_clock::from_time_t(1414920600), tp);
>+#endif
>+}
>+
>+TEST(Parse, LocaleSpecific) {
>+  time_zone tz = utc_time_zone();
>+  auto tp = chrono::system_clock::from_time_t(0);
>+  const auto reset = convert(civil_second(1977, 6, 28, 9, 8, 7), tz);
>+
>+  // %a is parsed but ignored.
>+  EXPECT_TRUE(parse("%a", "Mon", tz, &tp));
>+
>+  // %A is parsed but ignored.
>+  EXPECT_TRUE(parse("%A", "Monday", tz, &tp));
>+
>+  tp = reset;
>+  EXPECT_TRUE(parse("%b", "Feb", tz, &tp));
>+  EXPECT_EQ(2, convert(tp, tz).month());
>+
>+  tp = reset;
>+  EXPECT_TRUE(parse("%B", "February", tz, &tp));
>+  EXPECT_EQ(2, convert(tp, tz).month());
>+
>+  // %p is parsed but ignored if it's alone.  But it's used with %I.
>+  EXPECT_TRUE(parse("%p", "AM", tz, &tp));
>+  tp = reset;
>+  EXPECT_TRUE(parse("%I %p", "5 PM", tz, &tp));
>+  EXPECT_EQ(17, convert(tp, tz).hour());
>+
>+  tp = reset;
>+  EXPECT_TRUE(parse("%x", "02/03/04", tz, &tp));
>+  if (convert(tp, tz).month() == 2) {
>+    EXPECT_EQ(3, convert(tp, tz).day());
>+  } else {
>+    EXPECT_EQ(2, convert(tp, tz).day());
>+    EXPECT_EQ(3, convert(tp, tz).month());
>+  }
>+  EXPECT_EQ(2004, convert(tp, tz).year());
>+
>+  tp = reset;
>+  EXPECT_TRUE(parse("%X", "15:44:55", tz, &tp));
>+  EXPECT_EQ(15, convert(tp, tz).hour());
>+  EXPECT_EQ(44, convert(tp, tz).minute());
>+  EXPECT_EQ(55, convert(tp, tz).second());
>+
>+#if defined(__linux__)
>+  // SU/C99/TZ extensions
>+
>+  tp = reset;
>+  EXPECT_TRUE(parse("%h", "Feb", tz, &tp));
>+  EXPECT_EQ(2, convert(tp, tz).month());  // Equivalent to %b
>+
>+  tp = reset;
>+  EXPECT_TRUE(parse("%l %p", "5 PM", tz, &tp));
>+  EXPECT_EQ(17, convert(tp, tz).hour());
>+
>+  tp = reset;
>+  EXPECT_TRUE(parse("%r", "03:44:55 PM", tz, &tp));
>+  EXPECT_EQ(15, convert(tp, tz).hour());
>+  EXPECT_EQ(44, convert(tp, tz).minute());
>+  EXPECT_EQ(55, convert(tp, tz).second());
>+
>+  tp = reset;
>+  EXPECT_TRUE(parse("%Ec", "Tue Nov 19 05:06:07 2013", tz, &tp));
>+  EXPECT_EQ(convert(civil_second(2013, 11, 19, 5, 6, 7), tz), tp);
>+
>+  // Modified conversion specifiers %E_
>+
>+  tp = reset;
>+  EXPECT_TRUE(parse("%Ex", "02/03/04", tz, &tp));
>+  EXPECT_EQ(2, convert(tp, tz).month());
>+  EXPECT_EQ(3, convert(tp, tz).day());
>+  EXPECT_EQ(2004, convert(tp, tz).year());
>+
>+  tp = reset;
>+  EXPECT_TRUE(parse("%EX", "15:44:55", tz, &tp));
>+  EXPECT_EQ(15, convert(tp, tz).hour());
>+  EXPECT_EQ(44, convert(tp, tz).minute());
>+  EXPECT_EQ(55, convert(tp, tz).second());
>+
>+  // %Ey, the year offset from %EC, doesn't really make sense alone as there
>+  // is no way to represent it in tm_year (%EC is not simply the century).
>+  // Yet, because we handle each (non-internal) specifier in a separate call
>+  // to strptime(), there is no way to group %EC and %Ey either.  So we just
>+  // skip the %EC and %Ey cases.
>+
>+  tp = reset;
>+  EXPECT_TRUE(parse("%EY", "2004", tz, &tp));
>+  EXPECT_EQ(2004, convert(tp, tz).year());
>+
>+  // Modified conversion specifiers %O_
>+
>+  tp = reset;
>+  EXPECT_TRUE(parse("%Od", "15", tz, &tp));
>+  EXPECT_EQ(15, convert(tp, tz).day());
>+
>+  tp = reset;
>+  EXPECT_TRUE(parse("%Oe", "15", tz, &tp));
>+  EXPECT_EQ(15, convert(tp, tz).day());  // Equivalent to %d
>+
>+  tp = reset;
>+  EXPECT_TRUE(parse("%OH", "17", tz, &tp));
>+  EXPECT_EQ(17, convert(tp, tz).hour());
>+
>+  tp = reset;
>+  EXPECT_TRUE(parse("%OI", "5", tz, &tp));
>+  EXPECT_EQ(5, convert(tp, tz).hour());
>+
>+  tp = reset;
>+  EXPECT_TRUE(parse("%Om", "11", tz, &tp));
>+  EXPECT_EQ(11, convert(tp, tz).month());
>+
>+  tp = reset;
>+  EXPECT_TRUE(parse("%OM", "33", tz, &tp));
>+  EXPECT_EQ(33, convert(tp, tz).minute());
>+
>+  tp = reset;
>+  EXPECT_TRUE(parse("%OS", "55", tz, &tp));
>+  EXPECT_EQ(55, convert(tp, tz).second());
>+
>+  // %OU is parsed but ignored.
>+  EXPECT_TRUE(parse("%OU", "15", tz, &tp));
>+
>+  // %Ow is parsed but ignored.
>+  EXPECT_TRUE(parse("%Ow", "2", tz, &tp));
>+
>+  // %OW is parsed but ignored.
>+  EXPECT_TRUE(parse("%OW", "22", tz, &tp));
>+
>+  tp = reset;
>+  EXPECT_TRUE(parse("%Oy", "04", tz, &tp));
>+  EXPECT_EQ(2004, convert(tp, tz).year());
>+#endif
>+}
>+
>+TEST(Parse, ExtendedSeconds) {
>+  const time_zone tz = utc_time_zone();
>+  const time_point<chrono::nanoseconds> unix_epoch =
>+      chrono::system_clock::from_time_t(0);
>+
>+  // All %E<prec>S cases are treated the same as %E*S on input.
>+  auto precisions = {"*", "0", "1",  "2",  "3",  "4",  "5",  "6", "7",
>+                     "8", "9", "10", "11", "12", "13", "14", "15"};
>+  for (const std::string& prec : precisions) {
>+    const std::string fmt = "%E" + prec + "S";
>+    SCOPED_TRACE(fmt);
>+    time_point<chrono::nanoseconds> tp = unix_epoch;
>+    EXPECT_TRUE(parse(fmt, "5", tz, &tp));
>+    EXPECT_EQ(unix_epoch + chrono::seconds(5), tp);
>+    tp = unix_epoch;
>+    EXPECT_TRUE(parse(fmt, "05", tz, &tp));
>+    EXPECT_EQ(unix_epoch + chrono::seconds(5), tp);
>+    tp = unix_epoch;
>+    EXPECT_TRUE(parse(fmt, "05.0", tz, &tp));
>+    EXPECT_EQ(unix_epoch + chrono::seconds(5), tp);
>+    tp = unix_epoch;
>+    EXPECT_TRUE(parse(fmt, "05.00", tz, &tp));
>+    EXPECT_EQ(unix_epoch + chrono::seconds(5), tp);
>+    tp = unix_epoch;
>+    EXPECT_TRUE(parse(fmt, "05.6", tz, &tp));
>+    EXPECT_EQ(unix_epoch + chrono::seconds(5) + chrono::milliseconds(600), tp);
>+    tp = unix_epoch;
>+    EXPECT_TRUE(parse(fmt, "05.60", tz, &tp));
>+    EXPECT_EQ(unix_epoch + chrono::seconds(5) + chrono::milliseconds(600), tp);
>+    tp = unix_epoch;
>+    EXPECT_TRUE(parse(fmt, "05.600", tz, &tp));
>+    EXPECT_EQ(unix_epoch + chrono::seconds(5) + chrono::milliseconds(600), tp);
>+    tp = unix_epoch;
>+    EXPECT_TRUE(parse(fmt, "05.67", tz, &tp));
>+    EXPECT_EQ(unix_epoch + chrono::seconds(5) + chrono::milliseconds(670), tp);
>+    tp = unix_epoch;
>+    EXPECT_TRUE(parse(fmt, "05.670", tz, &tp));
>+    EXPECT_EQ(unix_epoch + chrono::seconds(5) + chrono::milliseconds(670), tp);
>+    tp = unix_epoch;
>+    EXPECT_TRUE(parse(fmt, "05.678", tz, &tp));
>+    EXPECT_EQ(unix_epoch + chrono::seconds(5) + chrono::milliseconds(678), tp);
>+  }
>+
>+  // Here is a "%E*S" case we got wrong for a while.  The fractional
>+  // part of the first instant is less than 2^31 and was correctly
>+  // parsed, while the second (and any subsecond field >=2^31) failed.
>+  time_point<chrono::nanoseconds> tp = unix_epoch;
>+  EXPECT_TRUE(parse("%E*S", "0.2147483647", tz, &tp));
>+  EXPECT_EQ(unix_epoch + chrono::nanoseconds(214748364), tp);
>+  tp = unix_epoch;
>+  EXPECT_TRUE(parse("%E*S", "0.2147483648", tz, &tp));
>+  EXPECT_EQ(unix_epoch + chrono::nanoseconds(214748364), tp);
>+
>+  // We should also be able to specify long strings of digits far
>+  // beyond the current resolution and have them convert the same way.
>+  tp = unix_epoch;
>+  EXPECT_TRUE(parse(
>+      "%E*S", "0.214748364801234567890123456789012345678901234567890123456789",
>+      tz, &tp));
>+  EXPECT_EQ(unix_epoch + chrono::nanoseconds(214748364), tp);
>+}
>+
>+TEST(Parse, ExtendedSecondsScan) {
>+  const time_zone tz = utc_time_zone();
>+  time_point<chrono::nanoseconds> tp;
>+  for (int ms = 0; ms < 1000; ms += 111) {
>+    for (int us = 0; us < 1000; us += 27) {
>+      const int micros = ms * 1000 + us;
>+      for (int ns = 0; ns < 1000; ns += 9) {
>+        const auto expected = chrono::system_clock::from_time_t(0) +
>+                              chrono::nanoseconds(micros * 1000 + ns);
>+        std::ostringstream oss;
>+        oss << "0." << std::setfill('0') << std::setw(3);
>+        oss << ms << std::setw(3) << us << std::setw(3) << ns;
>+        const std::string input = oss.str();
>+        EXPECT_TRUE(parse("%E*S", input, tz, &tp));
>+        EXPECT_EQ(expected, tp) << input;
>+      }
>+    }
>+  }
>+}
>+
>+TEST(Parse, ExtendedSubeconds) {
>+  const time_zone tz = utc_time_zone();
>+  const time_point<chrono::nanoseconds> unix_epoch =
>+      chrono::system_clock::from_time_t(0);
>+
>+  // All %E<prec>f cases are treated the same as %E*f on input.
>+  auto precisions = {"*", "0", "1",  "2",  "3",  "4",  "5",  "6", "7",
>+                     "8", "9", "10", "11", "12", "13", "14", "15"};
>+  for (const std::string& prec : precisions) {
>+    const std::string fmt = "%E" + prec + "f";
>+    SCOPED_TRACE(fmt);
>+    time_point<chrono::nanoseconds> tp = unix_epoch - chrono::seconds(1);
>+    EXPECT_TRUE(parse(fmt, "", tz, &tp));
>+    EXPECT_EQ(unix_epoch, tp);
>+    tp = unix_epoch;
>+    EXPECT_TRUE(parse(fmt, "6", tz, &tp));
>+    EXPECT_EQ(unix_epoch + chrono::milliseconds(600), tp);
>+    tp = unix_epoch;
>+    EXPECT_TRUE(parse(fmt, "60", tz, &tp));
>+    EXPECT_EQ(unix_epoch + chrono::milliseconds(600), tp);
>+    tp = unix_epoch;
>+    EXPECT_TRUE(parse(fmt, "600", tz, &tp));
>+    EXPECT_EQ(unix_epoch + chrono::milliseconds(600), tp);
>+    tp = unix_epoch;
>+    EXPECT_TRUE(parse(fmt, "67", tz, &tp));
>+    EXPECT_EQ(unix_epoch + chrono::milliseconds(670), tp);
>+    tp = unix_epoch;
>+    EXPECT_TRUE(parse(fmt, "670", tz, &tp));
>+    EXPECT_EQ(unix_epoch + chrono::milliseconds(670), tp);
>+    tp = unix_epoch;
>+    EXPECT_TRUE(parse(fmt, "678", tz, &tp));
>+    EXPECT_EQ(unix_epoch + chrono::milliseconds(678), tp);
>+    tp = unix_epoch;
>+    EXPECT_TRUE(parse(fmt, "6789", tz, &tp));
>+    EXPECT_EQ(
>+        unix_epoch + chrono::milliseconds(678) + chrono::microseconds(900), tp);
>+  }
>+
>+  // Here is a "%E*f" case we got wrong for a while.  The fractional
>+  // part of the first instant is less than 2^31 and was correctly
>+  // parsed, while the second (and any subsecond field >=2^31) failed.
>+  time_point<chrono::nanoseconds> tp = unix_epoch;
>+  EXPECT_TRUE(parse("%E*f", "2147483647", tz, &tp));
>+  EXPECT_EQ(unix_epoch + chrono::nanoseconds(214748364), tp);
>+  tp = unix_epoch;
>+  EXPECT_TRUE(parse("%E*f", "2147483648", tz, &tp));
>+  EXPECT_EQ(unix_epoch + chrono::nanoseconds(214748364), tp);
>+
>+  // We should also be able to specify long strings of digits far
>+  // beyond the current resolution and have them convert the same way.
>+  tp = unix_epoch;
>+  EXPECT_TRUE(parse(
>+      "%E*f", "214748364801234567890123456789012345678901234567890123456789",
>+      tz, &tp));
>+  EXPECT_EQ(unix_epoch + chrono::nanoseconds(214748364), tp);
>+}
>+
>+TEST(Parse, ExtendedSubecondsScan) {
>+  time_point<chrono::nanoseconds> tp;
>+  const time_zone tz = utc_time_zone();
>+  for (int ms = 0; ms < 1000; ms += 111) {
>+    for (int us = 0; us < 1000; us += 27) {
>+      const int micros = ms * 1000 + us;
>+      for (int ns = 0; ns < 1000; ns += 9) {
>+        std::ostringstream oss;
>+        oss << std::setfill('0') << std::setw(3) << ms;
>+        oss << std::setw(3) << us << std::setw(3) << ns;
>+        const std::string nanos = oss.str();
>+        const auto expected = chrono::system_clock::from_time_t(0) +
>+                              chrono::nanoseconds(micros * 1000 + ns);
>+        for (int ps = 0; ps < 1000; ps += 250) {
>+          std::ostringstream oss;
>+          oss << std::setfill('0') << std::setw(3) << ps;
>+          const std::string input = nanos + oss.str() + "999";
>+          EXPECT_TRUE(parse("%E*f", input, tz, &tp));
>+          EXPECT_EQ(expected + chrono::nanoseconds(ps) / 1000, tp) << input;
>+        }
>+      }
>+    }
>+  }
>+}
>+
>+TEST(Parse, ExtendedOffset) {
>+  const time_zone utc = utc_time_zone();
>+  time_point<absl::time_internal::cctz::seconds> tp;
>+
>+  // %z against +-HHMM.
>+  EXPECT_TRUE(parse("%z", "+0000", utc, &tp));
>+  EXPECT_EQ(convert(civil_second(1970, 1, 1, 0, 0, 0), utc), tp);
>+  EXPECT_TRUE(parse("%z", "-1234", utc, &tp));
>+  EXPECT_EQ(convert(civil_second(1970, 1, 1, 12, 34, 0), utc), tp);
>+  EXPECT_TRUE(parse("%z", "+1234", utc, &tp));
>+  EXPECT_EQ(convert(civil_second(1969, 12, 31, 11, 26, 0), utc), tp);
>+  EXPECT_FALSE(parse("%z", "-123", utc, &tp));
>+
>+  // %z against +-HH.
>+  EXPECT_TRUE(parse("%z", "+00", utc, &tp));
>+  EXPECT_EQ(convert(civil_second(1970, 1, 1, 0, 0, 0), utc), tp);
>+  EXPECT_TRUE(parse("%z", "-12", utc, &tp));
>+  EXPECT_EQ(convert(civil_second(1970, 1, 1, 12, 0, 0), utc), tp);
>+  EXPECT_TRUE(parse("%z", "+12", utc, &tp));
>+  EXPECT_EQ(convert(civil_second(1969, 12, 31, 12, 0, 0), utc), tp);
>+  EXPECT_FALSE(parse("%z", "-1", utc, &tp));
>+
>+  // %Ez against +-HH:MM.
>+  EXPECT_TRUE(parse("%Ez", "+00:00", utc, &tp));
>+  EXPECT_EQ(convert(civil_second(1970, 1, 1, 0, 0, 0), utc), tp);
>+  EXPECT_TRUE(parse("%Ez", "-12:34", utc, &tp));
>+  EXPECT_EQ(convert(civil_second(1970, 1, 1, 12, 34, 0), utc), tp);
>+  EXPECT_TRUE(parse("%Ez", "+12:34", utc, &tp));
>+  EXPECT_EQ(convert(civil_second(1969, 12, 31, 11, 26, 0), utc), tp);
>+  EXPECT_FALSE(parse("%Ez", "-12:3", utc, &tp));
>+
>+  // %Ez against +-HHMM.
>+  EXPECT_TRUE(parse("%Ez", "+0000", utc, &tp));
>+  EXPECT_EQ(convert(civil_second(1970, 1, 1, 0, 0, 0), utc), tp);
>+  EXPECT_TRUE(parse("%Ez", "-1234", utc, &tp));
>+  EXPECT_EQ(convert(civil_second(1970, 1, 1, 12, 34, 0), utc), tp);
>+  EXPECT_TRUE(parse("%Ez", "+1234", utc, &tp));
>+  EXPECT_EQ(convert(civil_second(1969, 12, 31, 11, 26, 0), utc), tp);
>+  EXPECT_FALSE(parse("%Ez", "-123", utc, &tp));
>+
>+  // %Ez against +-HH.
>+  EXPECT_TRUE(parse("%Ez", "+00", utc, &tp));
>+  EXPECT_EQ(convert(civil_second(1970, 1, 1, 0, 0, 0), utc), tp);
>+  EXPECT_TRUE(parse("%Ez", "-12", utc, &tp));
>+  EXPECT_EQ(convert(civil_second(1970, 1, 1, 12, 0, 0), utc), tp);
>+  EXPECT_TRUE(parse("%Ez", "+12", utc, &tp));
>+  EXPECT_EQ(convert(civil_second(1969, 12, 31, 12, 0, 0), utc), tp);
>+  EXPECT_FALSE(parse("%Ez", "-1", utc, &tp));
>+}
>+
>+TEST(Parse, ExtendedSecondOffset) {
>+  const time_zone utc = utc_time_zone();
>+  time_point<absl::time_internal::cctz::seconds> tp;
>+
>+  // %Ez against +-HH:MM:SS.
>+  EXPECT_TRUE(parse("%Ez", "+00:00:00", utc, &tp));
>+  EXPECT_EQ(convert(civil_second(1970, 1, 1, 0, 0, 0), utc), tp);
>+  EXPECT_TRUE(parse("%Ez", "-12:34:56", utc, &tp));
>+  EXPECT_EQ(convert(civil_second(1970, 1, 1, 12, 34, 56), utc), tp);
>+  EXPECT_TRUE(parse("%Ez", "+12:34:56", utc, &tp));
>+  EXPECT_EQ(convert(civil_second(1969, 12, 31, 11, 25, 4), utc), tp);
>+  EXPECT_FALSE(parse("%Ez", "-12:34:5", utc, &tp));
>+
>+  // %Ez against +-HHMMSS.
>+  EXPECT_TRUE(parse("%Ez", "+000000", utc, &tp));
>+  EXPECT_EQ(convert(civil_second(1970, 1, 1, 0, 0, 0), utc), tp);
>+  EXPECT_TRUE(parse("%Ez", "-123456", utc, &tp));
>+  EXPECT_EQ(convert(civil_second(1970, 1, 1, 12, 34, 56), utc), tp);
>+  EXPECT_TRUE(parse("%Ez", "+123456", utc, &tp));
>+  EXPECT_EQ(convert(civil_second(1969, 12, 31, 11, 25, 4), utc), tp);
>+  EXPECT_FALSE(parse("%Ez", "-12345", utc, &tp));
>+
>+  // %E*z against +-HH:MM:SS.
>+  EXPECT_TRUE(parse("%E*z", "+00:00:00", utc, &tp));
>+  EXPECT_EQ(convert(civil_second(1970, 1, 1, 0, 0, 0), utc), tp);
>+  EXPECT_TRUE(parse("%E*z", "-12:34:56", utc, &tp));
>+  EXPECT_EQ(convert(civil_second(1970, 1, 1, 12, 34, 56), utc), tp);
>+  EXPECT_TRUE(parse("%E*z", "+12:34:56", utc, &tp));
>+  EXPECT_EQ(convert(civil_second(1969, 12, 31, 11, 25, 4), utc), tp);
>+  EXPECT_FALSE(parse("%E*z", "-12:34:5", utc, &tp));
>+
>+  // %E*z against +-HHMMSS.
>+  EXPECT_TRUE(parse("%E*z", "+000000", utc, &tp));
>+  EXPECT_EQ(convert(civil_second(1970, 1, 1, 0, 0, 0), utc), tp);
>+  EXPECT_TRUE(parse("%E*z", "-123456", utc, &tp));
>+  EXPECT_EQ(convert(civil_second(1970, 1, 1, 12, 34, 56), utc), tp);
>+  EXPECT_TRUE(parse("%E*z", "+123456", utc, &tp));
>+  EXPECT_EQ(convert(civil_second(1969, 12, 31, 11, 25, 4), utc), tp);
>+  EXPECT_FALSE(parse("%E*z", "-12345", utc, &tp));
>+
>+  // %E*z against +-HH:MM.
>+  EXPECT_TRUE(parse("%E*z", "+00:00", utc, &tp));
>+  EXPECT_EQ(convert(civil_second(1970, 1, 1, 0, 0, 0), utc), tp);
>+  EXPECT_TRUE(parse("%E*z", "-12:34", utc, &tp));
>+  EXPECT_EQ(convert(civil_second(1970, 1, 1, 12, 34, 0), utc), tp);
>+  EXPECT_TRUE(parse("%E*z", "+12:34", utc, &tp));
>+  EXPECT_EQ(convert(civil_second(1969, 12, 31, 11, 26, 0), utc), tp);
>+  EXPECT_FALSE(parse("%E*z", "-12:3", utc, &tp));
>+
>+  // %E*z against +-HHMM.
>+  EXPECT_TRUE(parse("%E*z", "+0000", utc, &tp));
>+  EXPECT_EQ(convert(civil_second(1970, 1, 1, 0, 0, 0), utc), tp);
>+  EXPECT_TRUE(parse("%E*z", "-1234", utc, &tp));
>+  EXPECT_EQ(convert(civil_second(1970, 1, 1, 12, 34, 0), utc), tp);
>+  EXPECT_TRUE(parse("%E*z", "+1234", utc, &tp));
>+  EXPECT_EQ(convert(civil_second(1969, 12, 31, 11, 26, 0), utc), tp);
>+  EXPECT_FALSE(parse("%E*z", "-123", utc, &tp));
>+
>+  // %E*z against +-HH.
>+  EXPECT_TRUE(parse("%E*z", "+00", utc, &tp));
>+  EXPECT_EQ(convert(civil_second(1970, 1, 1, 0, 0, 0), utc), tp);
>+  EXPECT_TRUE(parse("%E*z", "-12", utc, &tp));
>+  EXPECT_EQ(convert(civil_second(1970, 1, 1, 12, 0, 0), utc), tp);
>+  EXPECT_TRUE(parse("%E*z", "+12", utc, &tp));
>+  EXPECT_EQ(convert(civil_second(1969, 12, 31, 12, 0, 0), utc), tp);
>+  EXPECT_FALSE(parse("%E*z", "-1", utc, &tp));
>+}
>+
>+TEST(Parse, ExtendedYears) {
>+  const time_zone utc = utc_time_zone();
>+  const char e4y_fmt[] = "%E4Y%m%d";  // no separators
>+  time_point<absl::time_internal::cctz::seconds> tp;
>+
>+  // %E4Y consumes exactly four chars, including any sign.
>+  EXPECT_TRUE(parse(e4y_fmt, "-9991127", utc, &tp));
>+  EXPECT_EQ(convert(civil_second(-999, 11, 27, 0, 0, 0), utc), tp);
>+  EXPECT_TRUE(parse(e4y_fmt, "-0991127", utc, &tp));
>+  EXPECT_EQ(convert(civil_second(-99, 11, 27, 0, 0, 0), utc), tp);
>+  EXPECT_TRUE(parse(e4y_fmt, "-0091127", utc, &tp));
>+  EXPECT_EQ(convert(civil_second(-9, 11, 27, 0, 0, 0), utc), tp);
>+  EXPECT_TRUE(parse(e4y_fmt, "-0011127", utc, &tp));
>+  EXPECT_EQ(convert(civil_second(-1, 11, 27, 0, 0, 0), utc), tp);
>+  EXPECT_TRUE(parse(e4y_fmt, "00001127", utc, &tp));
>+  EXPECT_EQ(convert(civil_second(0, 11, 27, 0, 0, 0), utc), tp);
>+  EXPECT_TRUE(parse(e4y_fmt, "00011127", utc, &tp));
>+  EXPECT_EQ(convert(civil_second(1, 11, 27, 0, 0, 0), utc), tp);
>+  EXPECT_TRUE(parse(e4y_fmt, "00091127", utc, &tp));
>+  EXPECT_EQ(convert(civil_second(9, 11, 27, 0, 0, 0), utc), tp);
>+  EXPECT_TRUE(parse(e4y_fmt, "00991127", utc, &tp));
>+  EXPECT_EQ(convert(civil_second(99, 11, 27, 0, 0, 0), utc), tp);
>+  EXPECT_TRUE(parse(e4y_fmt, "09991127", utc, &tp));
>+  EXPECT_EQ(convert(civil_second(999, 11, 27, 0, 0, 0), utc), tp);
>+  EXPECT_TRUE(parse(e4y_fmt, "99991127", utc, &tp));
>+  EXPECT_EQ(convert(civil_second(9999, 11, 27, 0, 0, 0), utc), tp);
>+
>+  // When the year is outside [-999:9999], the parse fails.
>+  EXPECT_FALSE(parse(e4y_fmt, "-10001127", utc, &tp));
>+  EXPECT_FALSE(parse(e4y_fmt, "100001127", utc, &tp));
>+}
>+
>+TEST(Parse, RFC3339Format) {
>+  const time_zone tz = utc_time_zone();
>+  time_point<chrono::nanoseconds> tp;
>+  EXPECT_TRUE(parse(RFC3339_sec, "2014-02-12T20:21:00+00:00", tz, &tp));
>+  ExpectTime(tp, tz, 2014, 2, 12, 20, 21, 0, 0, false, "UTC");
>+
>+  // Check that %Ez also accepts "Z" as a synonym for "+00:00".
>+  time_point<chrono::nanoseconds> tp2;
>+  EXPECT_TRUE(parse(RFC3339_sec, "2014-02-12T20:21:00Z", tz, &tp2));
>+  EXPECT_EQ(tp, tp2);
>+}
>+
>+TEST(Parse, MaxRange) {
>+  const time_zone utc = utc_time_zone();
>+  time_point<absl::time_internal::cctz::seconds> tp;
>+
>+  // tests the upper limit using +00:00 offset
>+  EXPECT_TRUE(
>+      parse(RFC3339_sec, "292277026596-12-04T15:30:07+00:00", utc, &tp));
>+  EXPECT_EQ(tp, time_point<absl::time_internal::cctz::seconds>::max());
>+  EXPECT_FALSE(
>+      parse(RFC3339_sec, "292277026596-12-04T15:30:08+00:00", utc, &tp));
>+
>+  // tests the upper limit using -01:00 offset
>+  EXPECT_TRUE(
>+      parse(RFC3339_sec, "292277026596-12-04T14:30:07-01:00", utc, &tp));
>+  EXPECT_EQ(tp, time_point<absl::time_internal::cctz::seconds>::max());
>+  EXPECT_FALSE(
>+      parse(RFC3339_sec, "292277026596-12-04T15:30:07-01:00", utc, &tp));
>+
>+  // tests the lower limit using +00:00 offset
>+  EXPECT_TRUE(
>+      parse(RFC3339_sec, "-292277022657-01-27T08:29:52+00:00", utc, &tp));
>+  EXPECT_EQ(tp, time_point<absl::time_internal::cctz::seconds>::min());
>+  EXPECT_FALSE(
>+      parse(RFC3339_sec, "-292277022657-01-27T08:29:51+00:00", utc, &tp));
>+
>+  // tests the lower limit using +01:00 offset
>+  EXPECT_TRUE(
>+      parse(RFC3339_sec, "-292277022657-01-27T09:29:52+01:00", utc, &tp));
>+  EXPECT_EQ(tp, time_point<absl::time_internal::cctz::seconds>::min());
>+  EXPECT_FALSE(
>+      parse(RFC3339_sec, "-292277022657-01-27T08:29:51+01:00", utc, &tp));
>+
>+  // tests max/min civil-second overflow
>+  EXPECT_FALSE(parse(RFC3339_sec, "9223372036854775807-12-31T23:59:59-00:01",
>+                     utc, &tp));
>+  EXPECT_FALSE(parse(RFC3339_sec, "-9223372036854775808-01-01T00:00:00+00:01",
>+                     utc, &tp));
>+
>+  // TODO: Add tests that parsing times with fractional seconds overflow
>+  // appropriately. This can't be done until cctz::parse() properly detects
>+  // overflow when combining the chrono seconds and femto.
>+}
>+
>+//
>+// Roundtrip test for format()/parse().
>+//
>+
>+TEST(FormatParse, RoundTrip) {
>+  time_zone lax;
>+  EXPECT_TRUE(load_time_zone("America/Los_Angeles", &lax));
>+  const auto in = convert(civil_second(1977, 6, 28, 9, 8, 7), lax);
>+  const auto subseconds = chrono::nanoseconds(654321);
>+
>+  // RFC3339, which renders subseconds.
>+  {
>+    time_point<chrono::nanoseconds> out;
>+    const std::string s = format(RFC3339_full, in + subseconds, lax);
>+    EXPECT_TRUE(parse(RFC3339_full, s, lax, &out)) << s;
>+    EXPECT_EQ(in + subseconds, out);  // RFC3339_full includes %Ez
>+  }
>+
>+  // RFC1123, which only does whole seconds.
>+  {
>+    time_point<chrono::nanoseconds> out;
>+    const std::string s = format(RFC1123_full, in, lax);
>+    EXPECT_TRUE(parse(RFC1123_full, s, lax, &out)) << s;
>+    EXPECT_EQ(in, out);  // RFC1123_full includes %z
>+  }
>+
>+#if defined(_WIN32) || defined(_WIN64)
>+  // Initial investigations indicate the %c does not roundtrip on Windows.
>+  // TODO: Figure out what is going on here (perhaps a locale problem).
>+#else
>+  // Even though we don't know what %c will produce, it should roundtrip,
>+  // but only in the 0-offset timezone.
>+  {
>+    time_point<chrono::nanoseconds> out;
>+    time_zone utc = utc_time_zone();
>+    const std::string s = format("%c", in, utc);
>+    EXPECT_TRUE(parse("%c", s, utc, &out)) << s;
>+    EXPECT_EQ(in, out);
>+  }
>+#endif
>+}
>+
>+TEST(FormatParse, RoundTripDistantFuture) {
>+  const time_zone utc = utc_time_zone();
>+  const time_point<absl::time_internal::cctz::seconds> in = time_point<absl::time_internal::cctz::seconds>::max();
>+  const std::string s = format(RFC3339_full, in, utc);
>+  time_point<absl::time_internal::cctz::seconds> out;
>+  EXPECT_TRUE(parse(RFC3339_full, s, utc, &out)) << s;
>+  EXPECT_EQ(in, out);
>+}
>+
>+TEST(FormatParse, RoundTripDistantPast) {
>+  const time_zone utc = utc_time_zone();
>+  const time_point<absl::time_internal::cctz::seconds> in = time_point<absl::time_internal::cctz::seconds>::min();
>+  const std::string s = format(RFC3339_full, in, utc);
>+  time_point<absl::time_internal::cctz::seconds> out;
>+  EXPECT_TRUE(parse(RFC3339_full, s, utc, &out)) << s;
>+  EXPECT_EQ(in, out);
>+}
>+
>+}  // namespace cctz
>+}  // namespace time_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_if.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_if.cc
>new file mode 100644
>index 00000000000..380834a172a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_if.cc
>@@ -0,0 +1,41 @@
>+// Copyright 2016 Google Inc. All Rights Reserved.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//   http://www.apache.org/licenses/LICENSE-2.0
>+//
>+//   Unless required by applicable law or agreed to in writing, software
>+//   distributed under the License is distributed on an "AS IS" BASIS,
>+//   WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+//   See the License for the specific language governing permissions and
>+//   limitations under the License.
>+
>+#include "time_zone_if.h"
>+#include "time_zone_info.h"
>+#include "time_zone_libc.h"
>+
>+namespace absl {
>+namespace time_internal {
>+namespace cctz {
>+
>+std::unique_ptr<TimeZoneIf> TimeZoneIf::Load(const std::string& name) {
>+  // Support "libc:localtime" and "libc:*" to access the legacy
>+  // localtime and UTC support respectively from the C library.
>+  if (name.compare(0, 5, "libc:") == 0) {
>+    return std::unique_ptr<TimeZoneIf>(new TimeZoneLibC(name.substr(5)));
>+  }
>+
>+  // Otherwise use the "zoneinfo" implementation by default.
>+  std::unique_ptr<TimeZoneInfo> tz(new TimeZoneInfo);
>+  if (!tz->Load(name)) tz.reset();
>+  return std::unique_ptr<TimeZoneIf>(tz.release());
>+}
>+
>+// Defined out-of-line to avoid emitting a weak vtable in all TUs.
>+TimeZoneIf::~TimeZoneIf() {}
>+
>+}  // namespace cctz
>+}  // namespace time_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_if.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_if.h
>new file mode 100644
>index 00000000000..e4bd3866a87
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_if.h
>@@ -0,0 +1,72 @@
>+// Copyright 2016 Google Inc. All Rights Reserved.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//   http://www.apache.org/licenses/LICENSE-2.0
>+//
>+//   Unless required by applicable law or agreed to in writing, software
>+//   distributed under the License is distributed on an "AS IS" BASIS,
>+//   WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+//   See the License for the specific language governing permissions and
>+//   limitations under the License.
>+
>+#ifndef ABSL_TIME_INTERNAL_CCTZ_TIME_ZONE_IF_H_
>+#define ABSL_TIME_INTERNAL_CCTZ_TIME_ZONE_IF_H_
>+
>+#include <chrono>
>+#include <cstdint>
>+#include <memory>
>+#include <string>
>+
>+#include "absl/time/internal/cctz/include/cctz/civil_time.h"
>+#include "absl/time/internal/cctz/include/cctz/time_zone.h"
>+
>+namespace absl {
>+namespace time_internal {
>+namespace cctz {
>+
>+// A simple interface used to hide time-zone complexities from time_zone::Impl.
>+// Subclasses implement the functions for civil-time conversions in the zone.
>+class TimeZoneIf {
>+ public:
>+  // A factory function for TimeZoneIf implementations.
>+  static std::unique_ptr<TimeZoneIf> Load(const std::string& name);
>+
>+  virtual ~TimeZoneIf();
>+
>+  virtual time_zone::absolute_lookup BreakTime(
>+      const time_point<seconds>& tp) const = 0;
>+  virtual time_zone::civil_lookup MakeTime(
>+      const civil_second& cs) const = 0;
>+
>+  virtual bool NextTransition(const time_point<seconds>& tp,
>+                              time_zone::civil_transition* trans) const = 0;
>+  virtual bool PrevTransition(const time_point<seconds>& tp,
>+                              time_zone::civil_transition* trans) const = 0;
>+
>+  virtual std::string Version() const = 0;
>+  virtual std::string Description() const = 0;
>+
>+ protected:
>+  TimeZoneIf() {}
>+};
>+
>+// Convert between time_point<seconds> and a count of seconds since the
>+// Unix epoch.  We assume that the std::chrono::system_clock and the
>+// Unix clock are second aligned, but not that they share an epoch.
>+inline std::int_fast64_t ToUnixSeconds(const time_point<seconds>& tp) {
>+  return (tp - std::chrono::time_point_cast<seconds>(
>+                   std::chrono::system_clock::from_time_t(0))).count();
>+}
>+inline time_point<seconds> FromUnixSeconds(std::int_fast64_t t) {
>+  return std::chrono::time_point_cast<seconds>(
>+             std::chrono::system_clock::from_time_t(0)) + seconds(t);
>+}
>+
>+}  // namespace cctz
>+}  // namespace time_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_TIME_INTERNAL_CCTZ_TIME_ZONE_IF_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_impl.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_impl.cc
>new file mode 100644
>index 00000000000..3062ccd3ceb
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_impl.cc
>@@ -0,0 +1,108 @@
>+// Copyright 2016 Google Inc. All Rights Reserved.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//   http://www.apache.org/licenses/LICENSE-2.0
>+//
>+//   Unless required by applicable law or agreed to in writing, software
>+//   distributed under the License is distributed on an "AS IS" BASIS,
>+//   WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+//   See the License for the specific language governing permissions and
>+//   limitations under the License.
>+
>+#include "time_zone_impl.h"
>+
>+#include <mutex>
>+#include <string>
>+#include <unordered_map>
>+#include <utility>
>+
>+#include "time_zone_fixed.h"
>+
>+namespace absl {
>+namespace time_internal {
>+namespace cctz {
>+
>+namespace {
>+
>+// time_zone::Impls are linked into a map to support fast lookup by name.
>+using TimeZoneImplByName =
>+    std::unordered_map<std::string, const time_zone::Impl*>;
>+TimeZoneImplByName* time_zone_map = nullptr;
>+
>+// Mutual exclusion for time_zone_map.
>+std::mutex time_zone_mutex;
>+
>+}  // namespace
>+
>+time_zone time_zone::Impl::UTC() {
>+  return time_zone(UTCImpl());
>+}
>+
>+bool time_zone::Impl::LoadTimeZone(const std::string& name, time_zone* tz) {
>+  const time_zone::Impl* const utc_impl = UTCImpl();
>+
>+  // First check for UTC (which is never a key in time_zone_map).
>+  auto offset = seconds::zero();
>+  if (FixedOffsetFromName(name, &offset) && offset == seconds::zero()) {
>+    *tz = time_zone(utc_impl);
>+    return true;
>+  }
>+
>+  // Then check, under a shared lock, whether the time zone has already
>+  // been loaded. This is the common path. TODO: Move to shared_mutex.
>+  {
>+    std::lock_guard<std::mutex> lock(time_zone_mutex);
>+    if (time_zone_map != nullptr) {
>+      TimeZoneImplByName::const_iterator itr = time_zone_map->find(name);
>+      if (itr != time_zone_map->end()) {
>+        *tz = time_zone(itr->second);
>+        return itr->second != utc_impl;
>+      }
>+    }
>+  }
>+
>+  // Now check again, under an exclusive lock.
>+  std::lock_guard<std::mutex> lock(time_zone_mutex);
>+  if (time_zone_map == nullptr) time_zone_map = new TimeZoneImplByName;
>+  const Impl*& impl = (*time_zone_map)[name];
>+  if (impl == nullptr) {
>+    // The first thread in loads the new time zone.
>+    Impl* new_impl = new Impl(name);
>+    new_impl->zone_ = TimeZoneIf::Load(new_impl->name_);
>+    if (new_impl->zone_ == nullptr) {
>+      delete new_impl;  // free the nascent Impl
>+      impl = utc_impl;  // and fallback to UTC
>+    } else {
>+      impl = new_impl;  // install new time zone
>+    }
>+  }
>+  *tz = time_zone(impl);
>+  return impl != utc_impl;
>+}
>+
>+void time_zone::Impl::ClearTimeZoneMapTestOnly() {
>+  std::lock_guard<std::mutex> lock(time_zone_mutex);
>+  if (time_zone_map != nullptr) {
>+    // Existing time_zone::Impl* entries are in the wild, so we simply
>+    // leak them.  Future requests will result in reloading the data.
>+    time_zone_map->clear();
>+  }
>+}
>+
>+time_zone::Impl::Impl(const std::string& name) : name_(name) {}
>+
>+const time_zone::Impl* time_zone::Impl::UTCImpl() {
>+  static Impl* utc_impl = [] {
>+    Impl* impl = new Impl("UTC");
>+    impl->zone_ = TimeZoneIf::Load(impl->name_);  // never fails
>+    return impl;
>+  }();
>+  return utc_impl;
>+}
>+
>+}  // namespace cctz
>+}  // namespace time_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_impl.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_impl.h
>new file mode 100644
>index 00000000000..14965ef54bf
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_impl.h
>@@ -0,0 +1,90 @@
>+// Copyright 2016 Google Inc. All Rights Reserved.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//   http://www.apache.org/licenses/LICENSE-2.0
>+//
>+//   Unless required by applicable law or agreed to in writing, software
>+//   distributed under the License is distributed on an "AS IS" BASIS,
>+//   WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+//   See the License for the specific language governing permissions and
>+//   limitations under the License.
>+
>+#ifndef ABSL_TIME_INTERNAL_CCTZ_TIME_ZONE_IMPL_H_
>+#define ABSL_TIME_INTERNAL_CCTZ_TIME_ZONE_IMPL_H_
>+
>+#include <memory>
>+#include <string>
>+
>+#include "absl/time/internal/cctz/include/cctz/civil_time.h"
>+#include "absl/time/internal/cctz/include/cctz/time_zone.h"
>+#include "time_zone_if.h"
>+#include "time_zone_info.h"
>+
>+namespace absl {
>+namespace time_internal {
>+namespace cctz {
>+
>+// time_zone::Impl is the internal object referenced by a cctz::time_zone.
>+class time_zone::Impl {
>+ public:
>+  // The UTC time zone. Also used for other time zones that fail to load.
>+  static time_zone UTC();
>+
>+  // Load a named time zone. Returns false if the name is invalid, or if
>+  // some other kind of error occurs. Note that loading "UTC" never fails.
>+  static bool LoadTimeZone(const std::string& name, time_zone* tz);
>+
>+  // Clears the map of cached time zones.  Primarily for use in benchmarks
>+  // that gauge the performance of loading/parsing the time-zone data.
>+  static void ClearTimeZoneMapTestOnly();
>+
>+  // The primary key is the time-zone ID (e.g., "America/New_York").
>+  const std::string& Name() const {
>+    // TODO: It would nice if the zoneinfo data included the zone name.
>+    return name_;
>+  }
>+
>+  // Breaks a time_point down to civil-time components in this time zone.
>+  time_zone::absolute_lookup BreakTime(const time_point<seconds>& tp) const {
>+    return zone_->BreakTime(tp);
>+  }
>+
>+  // Converts the civil-time components in this time zone into a time_point.
>+  // That is, the opposite of BreakTime(). The requested civil time may be
>+  // ambiguous or illegal due to a change of UTC offset.
>+  time_zone::civil_lookup MakeTime(const civil_second& cs) const {
>+    return zone_->MakeTime(cs);
>+  }
>+
>+  // Finds the time of the next/previous offset change in this time zone.
>+  bool NextTransition(const time_point<seconds>& tp,
>+                      time_zone::civil_transition* trans) const {
>+    return zone_->NextTransition(tp, trans);
>+  }
>+  bool PrevTransition(const time_point<seconds>& tp,
>+                      time_zone::civil_transition* trans) const {
>+    return zone_->PrevTransition(tp, trans);
>+  }
>+
>+  // Returns an implementation-defined version std::string for this time zone.
>+  std::string Version() const { return zone_->Version(); }
>+
>+  // Returns an implementation-defined description of this time zone.
>+  std::string Description() const { return zone_->Description(); }
>+
>+ private:
>+  explicit Impl(const std::string& name);
>+  static const Impl* UTCImpl();
>+
>+  const std::string name_;
>+  std::unique_ptr<TimeZoneIf> zone_;
>+};
>+
>+}  // namespace cctz
>+}  // namespace time_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_TIME_INTERNAL_CCTZ_TIME_ZONE_IMPL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_info.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_info.cc
>new file mode 100644
>index 00000000000..bf73635d4c6
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_info.cc
>@@ -0,0 +1,976 @@
>+// Copyright 2016 Google Inc. All Rights Reserved.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//   http://www.apache.org/licenses/LICENSE-2.0
>+//
>+//   Unless required by applicable law or agreed to in writing, software
>+//   distributed under the License is distributed on an "AS IS" BASIS,
>+//   WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+//   See the License for the specific language governing permissions and
>+//   limitations under the License.
>+
>+// This file implements the TimeZoneIf interface using the "zoneinfo"
>+// data provided by the IANA Time Zone Database (i.e., the only real game
>+// in town).
>+//
>+// TimeZoneInfo represents the history of UTC-offset changes within a time
>+// zone. Most changes are due to daylight-saving rules, but occasionally
>+// shifts are made to the time-zone's base offset. The database only attempts
>+// to be definitive for times since 1970, so be wary of local-time conversions
>+// before that. Also, rule and zone-boundary changes are made at the whim
>+// of governments, so the conversion of future times needs to be taken with
>+// a grain of salt.
>+//
>+// For more information see tzfile(5), http://www.iana.org/time-zones, or
>+// http://en.wikipedia.org/wiki/Zoneinfo.
>+//
>+// Note that we assume the proleptic Gregorian calendar and 60-second
>+// minutes throughout.
>+
>+#include "time_zone_info.h"
>+
>+#include <algorithm>
>+#include <cassert>
>+#include <chrono>
>+#include <cstdint>
>+#include <cstdio>
>+#include <cstdlib>
>+#include <cstring>
>+#include <functional>
>+#include <iostream>
>+#include <memory>
>+#include <sstream>
>+#include <string>
>+
>+#include "absl/time/internal/cctz/include/cctz/civil_time.h"
>+#include "time_zone_fixed.h"
>+#include "time_zone_posix.h"
>+
>+namespace absl {
>+namespace time_internal {
>+namespace cctz {
>+
>+namespace {
>+
>+inline bool IsLeap(year_t year) {
>+  return (year % 4) == 0 && ((year % 100) != 0 || (year % 400) == 0);
>+}
>+
>+// The number of days in non-leap and leap years respectively.
>+const std::int_least32_t kDaysPerYear[2] = {365, 366};
>+
>+// The day offsets of the beginning of each (1-based) month in non-leap and
>+// leap years respectively (e.g., 335 days before December in a leap year).
>+const std::int_least16_t kMonthOffsets[2][1 + 12 + 1] = {
>+  {-1, 0, 31, 59, 90, 120, 151, 181, 212, 243, 273, 304, 334, 365},
>+  {-1, 0, 31, 60, 91, 121, 152, 182, 213, 244, 274, 305, 335, 366},
>+};
>+
>+// We reject leap-second encoded zoneinfo and so assume 60-second minutes.
>+const std::int_least32_t kSecsPerDay = 24 * 60 * 60;
>+
>+// 400-year chunks always have 146097 days (20871 weeks).
>+const std::int_least64_t kSecsPer400Years = 146097LL * kSecsPerDay;
>+
>+// Like kDaysPerYear[] but scaled up by a factor of kSecsPerDay.
>+const std::int_least32_t kSecsPerYear[2] = {
>+  365 * kSecsPerDay,
>+  366 * kSecsPerDay,
>+};
>+
>+// Single-byte, unsigned numeric values are encoded directly.
>+inline std::uint_fast8_t Decode8(const char* cp) {
>+  return static_cast<std::uint_fast8_t>(*cp) & 0xff;
>+}
>+
>+// Multi-byte, numeric values are encoded using a MSB first,
>+// twos-complement representation. These helpers decode, from
>+// the given address, 4-byte and 8-byte values respectively.
>+// Note: If int_fastXX_t == intXX_t and this machine is not
>+// twos complement, then there will be at least one input value
>+// we cannot represent.
>+std::int_fast32_t Decode32(const char* cp) {
>+  std::uint_fast32_t v = 0;
>+  for (int i = 0; i != (32 / 8); ++i) v = (v << 8) | Decode8(cp++);
>+  const std::int_fast32_t s32max = 0x7fffffff;
>+  const auto s32maxU = static_cast<std::uint_fast32_t>(s32max);
>+  if (v <= s32maxU) return static_cast<std::int_fast32_t>(v);
>+  return static_cast<std::int_fast32_t>(v - s32maxU - 1) - s32max - 1;
>+}
>+
>+std::int_fast64_t Decode64(const char* cp) {
>+  std::uint_fast64_t v = 0;
>+  for (int i = 0; i != (64 / 8); ++i) v = (v << 8) | Decode8(cp++);
>+  const std::int_fast64_t s64max = 0x7fffffffffffffff;
>+  const auto s64maxU = static_cast<std::uint_fast64_t>(s64max);
>+  if (v <= s64maxU) return static_cast<std::int_fast64_t>(v);
>+  return static_cast<std::int_fast64_t>(v - s64maxU - 1) - s64max - 1;
>+}
>+
>+// Generate a year-relative offset for a PosixTransition.
>+std::int_fast64_t TransOffset(bool leap_year, int jan1_weekday,
>+                              const PosixTransition& pt) {
>+  std::int_fast64_t days = 0;
>+  switch (pt.date.fmt) {
>+    case PosixTransition::J: {
>+      days = pt.date.j.day;
>+      if (!leap_year || days < kMonthOffsets[1][3]) days -= 1;
>+      break;
>+    }
>+    case PosixTransition::N: {
>+      days = pt.date.n.day;
>+      break;
>+    }
>+    case PosixTransition::M: {
>+      const bool last_week = (pt.date.m.week == 5);
>+      days = kMonthOffsets[leap_year][pt.date.m.month + last_week];
>+      const std::int_fast64_t weekday = (jan1_weekday + days) % 7;
>+      if (last_week) {
>+        days -= (weekday + 7 - 1 - pt.date.m.weekday) % 7 + 1;
>+      } else {
>+        days += (pt.date.m.weekday + 7 - weekday) % 7;
>+        days += (pt.date.m.week - 1) * 7;
>+      }
>+      break;
>+    }
>+  }
>+  return (days * kSecsPerDay) + pt.time.offset;
>+}
>+
>+inline time_zone::civil_lookup MakeUnique(const time_point<seconds>& tp) {
>+  time_zone::civil_lookup cl;
>+  cl.kind = time_zone::civil_lookup::UNIQUE;
>+  cl.pre = cl.trans = cl.post = tp;
>+  return cl;
>+}
>+
>+inline time_zone::civil_lookup MakeUnique(std::int_fast64_t unix_time) {
>+  return MakeUnique(FromUnixSeconds(unix_time));
>+}
>+
>+inline time_zone::civil_lookup MakeSkipped(const Transition& tr,
>+                                           const civil_second& cs) {
>+  time_zone::civil_lookup cl;
>+  cl.kind = time_zone::civil_lookup::SKIPPED;
>+  cl.pre = FromUnixSeconds(tr.unix_time - 1 + (cs - tr.prev_civil_sec));
>+  cl.trans = FromUnixSeconds(tr.unix_time);
>+  cl.post = FromUnixSeconds(tr.unix_time - (tr.civil_sec - cs));
>+  return cl;
>+}
>+
>+inline time_zone::civil_lookup MakeRepeated(const Transition& tr,
>+                                            const civil_second& cs) {
>+  time_zone::civil_lookup cl;
>+  cl.kind = time_zone::civil_lookup::REPEATED;
>+  cl.pre = FromUnixSeconds(tr.unix_time - 1 - (tr.prev_civil_sec - cs));
>+  cl.trans = FromUnixSeconds(tr.unix_time);
>+  cl.post = FromUnixSeconds(tr.unix_time + (cs - tr.civil_sec));
>+  return cl;
>+}
>+
>+inline civil_second YearShift(const civil_second& cs, year_t shift) {
>+  return civil_second(cs.year() + shift, cs.month(), cs.day(),
>+                      cs.hour(), cs.minute(), cs.second());
>+}
>+
>+}  // namespace
>+
>+// What (no leap-seconds) UTC+seconds zoneinfo would look like.
>+bool TimeZoneInfo::ResetToBuiltinUTC(const seconds& offset) {
>+  transition_types_.resize(1);
>+  TransitionType& tt(transition_types_.back());
>+  tt.utc_offset = static_cast<std::int_least32_t>(offset.count());
>+  tt.is_dst = false;
>+  tt.abbr_index = 0;
>+
>+  // We temporarily add some redundant, contemporary (2013 through 2023)
>+  // transitions for performance reasons.  See TimeZoneInfo::LocalTime().
>+  // TODO: Fix the performance issue and remove the extra transitions.
>+  transitions_.clear();
>+  transitions_.reserve(12);
>+  for (const std::int_fast64_t unix_time : {
>+           -(1LL << 59),  // BIG_BANG
>+           1356998400LL,  // 2013-01-01T00:00:00+00:00
>+           1388534400LL,  // 2014-01-01T00:00:00+00:00
>+           1420070400LL,  // 2015-01-01T00:00:00+00:00
>+           1451606400LL,  // 2016-01-01T00:00:00+00:00
>+           1483228800LL,  // 2017-01-01T00:00:00+00:00
>+           1514764800LL,  // 2018-01-01T00:00:00+00:00
>+           1546300800LL,  // 2019-01-01T00:00:00+00:00
>+           1577836800LL,  // 2020-01-01T00:00:00+00:00
>+           1609459200LL,  // 2021-01-01T00:00:00+00:00
>+           1640995200LL,  // 2022-01-01T00:00:00+00:00
>+           1672531200LL,  // 2023-01-01T00:00:00+00:00
>+           2147483647LL,  // 2^31 - 1
>+       }) {
>+    Transition& tr(*transitions_.emplace(transitions_.end()));
>+    tr.unix_time = unix_time;
>+    tr.type_index = 0;
>+    tr.civil_sec = LocalTime(tr.unix_time, tt).cs;
>+    tr.prev_civil_sec = tr.civil_sec - 1;
>+  }
>+
>+  default_transition_type_ = 0;
>+  abbreviations_ = FixedOffsetToAbbr(offset);
>+  abbreviations_.append(1, '\0');  // add NUL
>+  future_spec_.clear();  // never needed for a fixed-offset zone
>+  extended_ = false;
>+
>+  tt.civil_max = LocalTime(seconds::max().count(), tt).cs;
>+  tt.civil_min = LocalTime(seconds::min().count(), tt).cs;
>+
>+  transitions_.shrink_to_fit();
>+  return true;
>+}
>+
>+// Builds the in-memory header using the raw bytes from the file.
>+bool TimeZoneInfo::Header::Build(const tzhead& tzh) {
>+  std::int_fast32_t v;
>+  if ((v = Decode32(tzh.tzh_timecnt)) < 0) return false;
>+  timecnt = static_cast<std::size_t>(v);
>+  if ((v = Decode32(tzh.tzh_typecnt)) < 0) return false;
>+  typecnt = static_cast<std::size_t>(v);
>+  if ((v = Decode32(tzh.tzh_charcnt)) < 0) return false;
>+  charcnt = static_cast<std::size_t>(v);
>+  if ((v = Decode32(tzh.tzh_leapcnt)) < 0) return false;
>+  leapcnt = static_cast<std::size_t>(v);
>+  if ((v = Decode32(tzh.tzh_ttisstdcnt)) < 0) return false;
>+  ttisstdcnt = static_cast<std::size_t>(v);
>+  if ((v = Decode32(tzh.tzh_ttisgmtcnt)) < 0) return false;
>+  ttisgmtcnt = static_cast<std::size_t>(v);
>+  return true;
>+}
>+
>+// How many bytes of data are associated with this header. The result
>+// depends upon whether this is a section with 4-byte or 8-byte times.
>+std::size_t TimeZoneInfo::Header::DataLength(std::size_t time_len) const {
>+  std::size_t len = 0;
>+  len += (time_len + 1) * timecnt;  // unix_time + type_index
>+  len += (4 + 1 + 1) * typecnt;     // utc_offset + is_dst + abbr_index
>+  len += 1 * charcnt;               // abbreviations
>+  len += (time_len + 4) * leapcnt;  // leap-time + TAI-UTC
>+  len += 1 * ttisstdcnt;            // UTC/local indicators
>+  len += 1 * ttisgmtcnt;            // standard/wall indicators
>+  return len;
>+}
>+
>+// Check that the TransitionType has the expected offset/is_dst/abbreviation.
>+void TimeZoneInfo::CheckTransition(const std::string& name,
>+                                   const TransitionType& tt,
>+                                   std::int_fast32_t offset, bool is_dst,
>+                                   const std::string& abbr) const {
>+  if (tt.utc_offset != offset || tt.is_dst != is_dst ||
>+      &abbreviations_[tt.abbr_index] != abbr) {
>+    std::clog << name << ": Transition"
>+              << " offset=" << tt.utc_offset << "/"
>+              << (tt.is_dst ? "DST" : "STD")
>+              << "/abbr=" << &abbreviations_[tt.abbr_index]
>+              << " does not match POSIX spec '" << future_spec_ << "'\n";
>+  }
>+}
>+
>+// zic(8) can generate no-op transitions when a zone changes rules at an
>+// instant when there is actually no discontinuity.  So we check whether
>+// two transitions have equivalent types (same offset/is_dst/abbr).
>+bool TimeZoneInfo::EquivTransitions(std::uint_fast8_t tt1_index,
>+                                    std::uint_fast8_t tt2_index) const {
>+  if (tt1_index == tt2_index) return true;
>+  const TransitionType& tt1(transition_types_[tt1_index]);
>+  const TransitionType& tt2(transition_types_[tt2_index]);
>+  if (tt1.is_dst != tt2.is_dst) return false;
>+  if (tt1.utc_offset != tt2.utc_offset) return false;
>+  if (tt1.abbr_index != tt2.abbr_index) return false;
>+  return true;
>+}
>+
>+// Use the POSIX-TZ-environment-variable-style std::string to handle times
>+// in years after the last transition stored in the zoneinfo data.
>+void TimeZoneInfo::ExtendTransitions(const std::string& name,
>+                                     const Header& hdr) {
>+  extended_ = false;
>+  bool extending = !future_spec_.empty();
>+
>+  PosixTimeZone posix;
>+  if (extending && !ParsePosixSpec(future_spec_, &posix)) {
>+    std::clog << name << ": Failed to parse '" << future_spec_ << "'\n";
>+    extending = false;
>+  }
>+
>+  if (extending && posix.dst_abbr.empty()) {  // std only
>+    // The future specification should match the last/default transition,
>+    // and that means that handling the future will fall out naturally.
>+    std::uint_fast8_t index = default_transition_type_;
>+    if (hdr.timecnt != 0) index = transitions_[hdr.timecnt - 1].type_index;
>+    const TransitionType& tt(transition_types_[index]);
>+    CheckTransition(name, tt, posix.std_offset, false, posix.std_abbr);
>+    extending = false;
>+  }
>+
>+  if (extending && hdr.timecnt < 2) {
>+    std::clog << name << ": Too few transitions for POSIX spec\n";
>+    extending = false;
>+  }
>+
>+  if (!extending) {
>+    // Ensure that there is always a transition in the second half of the
>+    // time line (the BIG_BANG transition is in the first half) so that the
>+    // signed difference between a civil_second and the civil_second of its
>+    // previous transition is always representable, without overflow.
>+    const Transition& last(transitions_.back());
>+    if (last.unix_time < 0) {
>+      const std::uint_fast8_t type_index = last.type_index;
>+      Transition& tr(*transitions_.emplace(transitions_.end()));
>+      tr.unix_time = 2147483647;  // 2038-01-19T03:14:07+00:00
>+      tr.type_index = type_index;
>+    }
>+    return;  // last transition wins
>+  }
>+
>+  // Extend the transitions for an additional 400 years using the
>+  // future specification. Years beyond those can be handled by
>+  // mapping back to a cycle-equivalent year within that range.
>+  // zic(8) should probably do this so that we don't have to.
>+  // TODO: Reduce the extension by the number of compatible
>+  // transitions already in place.
>+  transitions_.reserve(hdr.timecnt + 400 * 2 + 1);
>+  transitions_.resize(hdr.timecnt + 400 * 2);
>+  extended_ = true;
>+
>+  // The future specification should match the last two transitions,
>+  // and those transitions should have different is_dst flags.  Note
>+  // that nothing says the UTC offset used by the is_dst transition
>+  // must be greater than that used by the !is_dst transition.  (See
>+  // Europe/Dublin, for example.)
>+  const Transition* tr0 = &transitions_[hdr.timecnt - 1];
>+  const Transition* tr1 = &transitions_[hdr.timecnt - 2];
>+  const TransitionType* tt0 = &transition_types_[tr0->type_index];
>+  const TransitionType* tt1 = &transition_types_[tr1->type_index];
>+  const TransitionType& dst(tt0->is_dst ? *tt0 : *tt1);
>+  const TransitionType& std(tt0->is_dst ? *tt1 : *tt0);
>+  CheckTransition(name, dst, posix.dst_offset, true, posix.dst_abbr);
>+  CheckTransition(name, std, posix.std_offset, false, posix.std_abbr);
>+
>+  // Add the transitions to tr1 and back to tr0 for each extra year.
>+  last_year_ = LocalTime(tr0->unix_time, *tt0).cs.year();
>+  bool leap_year = IsLeap(last_year_);
>+  const civil_day jan1(last_year_, 1, 1);
>+  std::int_fast64_t jan1_time = civil_second(jan1) - civil_second();
>+  int jan1_weekday = (static_cast<int>(get_weekday(jan1)) + 1) % 7;
>+  Transition* tr = &transitions_[hdr.timecnt];  // next trans to fill
>+  if (LocalTime(tr1->unix_time, *tt1).cs.year() != last_year_) {
>+    // Add a single extra transition to align to a calendar year.
>+    transitions_.resize(transitions_.size() + 1);
>+    assert(tr == &transitions_[hdr.timecnt]);  // no reallocation
>+    const PosixTransition& pt1(tt0->is_dst ? posix.dst_end : posix.dst_start);
>+    std::int_fast64_t tr1_offset = TransOffset(leap_year, jan1_weekday, pt1);
>+    tr->unix_time = jan1_time + tr1_offset - tt0->utc_offset;
>+    tr++->type_index = tr1->type_index;
>+    tr0 = &transitions_[hdr.timecnt];
>+    tr1 = &transitions_[hdr.timecnt - 1];
>+    tt0 = &transition_types_[tr0->type_index];
>+    tt1 = &transition_types_[tr1->type_index];
>+  }
>+  const PosixTransition& pt1(tt0->is_dst ? posix.dst_end : posix.dst_start);
>+  const PosixTransition& pt0(tt0->is_dst ? posix.dst_start : posix.dst_end);
>+  for (const year_t limit = last_year_ + 400; last_year_ < limit;) {
>+    last_year_ += 1;  // an additional year of generated transitions
>+    jan1_time += kSecsPerYear[leap_year];
>+    jan1_weekday = (jan1_weekday + kDaysPerYear[leap_year]) % 7;
>+    leap_year = !leap_year && IsLeap(last_year_);
>+    std::int_fast64_t tr1_offset = TransOffset(leap_year, jan1_weekday, pt1);
>+    tr->unix_time = jan1_time + tr1_offset - tt0->utc_offset;
>+    tr++->type_index = tr1->type_index;
>+    std::int_fast64_t tr0_offset = TransOffset(leap_year, jan1_weekday, pt0);
>+    tr->unix_time = jan1_time + tr0_offset - tt1->utc_offset;
>+    tr++->type_index = tr0->type_index;
>+  }
>+  assert(tr == &transitions_[0] + transitions_.size());
>+}
>+
>+bool TimeZoneInfo::Load(const std::string& name, ZoneInfoSource* zip) {
>+  // Read and validate the header.
>+  tzhead tzh;
>+  if (zip->Read(&tzh, sizeof(tzh)) != sizeof(tzh))
>+    return false;
>+  if (strncmp(tzh.tzh_magic, TZ_MAGIC, sizeof(tzh.tzh_magic)) != 0)
>+    return false;
>+  Header hdr;
>+  if (!hdr.Build(tzh))
>+    return false;
>+  std::size_t time_len = 4;
>+  if (tzh.tzh_version[0] != '\0') {
>+    // Skip the 4-byte data.
>+    if (zip->Skip(hdr.DataLength(time_len)) != 0)
>+      return false;
>+    // Read and validate the header for the 8-byte data.
>+    if (zip->Read(&tzh, sizeof(tzh)) != sizeof(tzh))
>+      return false;
>+    if (strncmp(tzh.tzh_magic, TZ_MAGIC, sizeof(tzh.tzh_magic)) != 0)
>+      return false;
>+    if (tzh.tzh_version[0] == '\0')
>+      return false;
>+    if (!hdr.Build(tzh))
>+      return false;
>+    time_len = 8;
>+  }
>+  if (hdr.typecnt == 0)
>+    return false;
>+  if (hdr.leapcnt != 0) {
>+    // This code assumes 60-second minutes so we do not want
>+    // the leap-second encoded zoneinfo. We could reverse the
>+    // compensation, but the "right" encoding is rarely used
>+    // so currently we simply reject such data.
>+    return false;
>+  }
>+  if (hdr.ttisstdcnt != 0 && hdr.ttisstdcnt != hdr.typecnt)
>+    return false;
>+  if (hdr.ttisgmtcnt != 0 && hdr.ttisgmtcnt != hdr.typecnt)
>+    return false;
>+
>+  // Read the data into a local buffer.
>+  std::size_t len = hdr.DataLength(time_len);
>+  std::vector<char> tbuf(len);
>+  if (zip->Read(tbuf.data(), len) != len)
>+    return false;
>+  const char* bp = tbuf.data();
>+
>+  // Decode and validate the transitions.
>+  transitions_.reserve(hdr.timecnt + 2);  // We might add a couple.
>+  transitions_.resize(hdr.timecnt);
>+  for (std::size_t i = 0; i != hdr.timecnt; ++i) {
>+    transitions_[i].unix_time = (time_len == 4) ? Decode32(bp) : Decode64(bp);
>+    bp += time_len;
>+    if (i != 0) {
>+      // Check that the transitions are ordered by time (as zic guarantees).
>+      if (!Transition::ByUnixTime()(transitions_[i - 1], transitions_[i]))
>+        return false;  // out of order
>+    }
>+  }
>+  bool seen_type_0 = false;
>+  for (std::size_t i = 0; i != hdr.timecnt; ++i) {
>+    transitions_[i].type_index = Decode8(bp++);
>+    if (transitions_[i].type_index >= hdr.typecnt)
>+      return false;
>+    if (transitions_[i].type_index == 0)
>+      seen_type_0 = true;
>+  }
>+
>+  // Decode and validate the transition types.
>+  transition_types_.resize(hdr.typecnt);
>+  for (std::size_t i = 0; i != hdr.typecnt; ++i) {
>+    transition_types_[i].utc_offset =
>+        static_cast<std::int_least32_t>(Decode32(bp));
>+    if (transition_types_[i].utc_offset >= kSecsPerDay ||
>+        transition_types_[i].utc_offset <= -kSecsPerDay)
>+      return false;
>+    bp += 4;
>+    transition_types_[i].is_dst = (Decode8(bp++) != 0);
>+    transition_types_[i].abbr_index = Decode8(bp++);
>+    if (transition_types_[i].abbr_index >= hdr.charcnt)
>+      return false;
>+  }
>+
>+  // Determine the before-first-transition type.
>+  default_transition_type_ = 0;
>+  if (seen_type_0 && hdr.timecnt != 0) {
>+    std::uint_fast8_t index = 0;
>+    if (transition_types_[0].is_dst) {
>+      index = transitions_[0].type_index;
>+      while (index != 0 && transition_types_[index].is_dst)
>+        --index;
>+    }
>+    while (index != hdr.typecnt && transition_types_[index].is_dst)
>+      ++index;
>+    if (index != hdr.typecnt)
>+      default_transition_type_ = index;
>+  }
>+
>+  // Copy all the abbreviations.
>+  abbreviations_.assign(bp, hdr.charcnt);
>+  bp += hdr.charcnt;
>+
>+  // Skip the unused portions. We've already dispensed with leap-second
>+  // encoded zoneinfo. The ttisstd/ttisgmt indicators only apply when
>+  // interpreting a POSIX spec that does not include start/end rules, and
>+  // that isn't the case here (see "zic -p").
>+  bp += (8 + 4) * hdr.leapcnt;  // leap-time + TAI-UTC
>+  bp += 1 * hdr.ttisstdcnt;     // UTC/local indicators
>+  bp += 1 * hdr.ttisgmtcnt;     // standard/wall indicators
>+  assert(bp == tbuf.data() + tbuf.size());
>+
>+  future_spec_.clear();
>+  if (tzh.tzh_version[0] != '\0') {
>+    // Snarf up the NL-enclosed future POSIX spec. Note
>+    // that version '3' files utilize an extended format.
>+    auto get_char = [](ZoneInfoSource* zip) -> int {
>+      unsigned char ch;  // all non-EOF results are positive
>+      return (zip->Read(&ch, 1) == 1) ? ch : EOF;
>+    };
>+    if (get_char(zip) != '\n')
>+      return false;
>+    for (int c = get_char(zip); c != '\n'; c = get_char(zip)) {
>+      if (c == EOF)
>+        return false;
>+      future_spec_.push_back(static_cast<char>(c));
>+    }
>+  }
>+
>+  // We don't check for EOF so that we're forwards compatible.
>+
>+  // If we did not find version information during the standard loading
>+  // process (as of tzh_version '3' that is unsupported), then ask the
>+  // ZoneInfoSource for any out-of-bound version std::string it may be privy to.
>+  if (version_.empty()) {
>+    version_ = zip->Version();
>+  }
>+
>+  // Trim redundant transitions. zic may have added these to work around
>+  // differences between the glibc and reference implementations (see
>+  // zic.c:dontmerge) and the Qt library (see zic.c:WORK_AROUND_QTBUG_53071).
>+  // For us, they just get in the way when we do future_spec_ extension.
>+  while (hdr.timecnt > 1) {
>+    if (!EquivTransitions(transitions_[hdr.timecnt - 1].type_index,
>+                          transitions_[hdr.timecnt - 2].type_index)) {
>+      break;
>+    }
>+    hdr.timecnt -= 1;
>+  }
>+  transitions_.resize(hdr.timecnt);
>+
>+  // Ensure that there is always a transition in the first half of the
>+  // time line (the second half is handled in ExtendTransitions()) so that
>+  // the signed difference between a civil_second and the civil_second of
>+  // its previous transition is always representable, without overflow.
>+  // A contemporary zic will usually have already done this for us.
>+  if (transitions_.empty() || transitions_.front().unix_time >= 0) {
>+    Transition& tr(*transitions_.emplace(transitions_.begin()));
>+    tr.unix_time = -(1LL << 59);  // see tz/zic.c "BIG_BANG"
>+    tr.type_index = default_transition_type_;
>+    hdr.timecnt += 1;
>+  }
>+
>+  // Extend the transitions using the future specification.
>+  ExtendTransitions(name, hdr);
>+
>+  // Compute the local civil time for each transition and the preceding
>+  // second. These will be used for reverse conversions in MakeTime().
>+  const TransitionType* ttp = &transition_types_[default_transition_type_];
>+  for (std::size_t i = 0; i != transitions_.size(); ++i) {
>+    Transition& tr(transitions_[i]);
>+    tr.prev_civil_sec = LocalTime(tr.unix_time, *ttp).cs - 1;
>+    ttp = &transition_types_[tr.type_index];
>+    tr.civil_sec = LocalTime(tr.unix_time, *ttp).cs;
>+    if (i != 0) {
>+      // Check that the transitions are ordered by civil time. Essentially
>+      // this means that an offset change cannot cross another such change.
>+      // No one does this in practice, and we depend on it in MakeTime().
>+      if (!Transition::ByCivilTime()(transitions_[i - 1], tr))
>+        return false;  // out of order
>+    }
>+  }
>+
>+  // Compute the maximum/minimum civil times that can be converted to a
>+  // time_point<seconds> for each of the zone's transition types.
>+  for (auto& tt : transition_types_) {
>+    tt.civil_max = LocalTime(seconds::max().count(), tt).cs;
>+    tt.civil_min = LocalTime(seconds::min().count(), tt).cs;
>+  }
>+
>+  transitions_.shrink_to_fit();
>+  return true;
>+}
>+
>+namespace {
>+
>+// fopen(3) adaptor.
>+inline FILE* FOpen(const char* path, const char* mode) {
>+#if defined(_MSC_VER)
>+  FILE* fp;
>+  if (fopen_s(&fp, path, mode) != 0) fp = nullptr;
>+  return fp;
>+#else
>+  return fopen(path, mode);  // TODO: Enable the close-on-exec flag.
>+#endif
>+}
>+
>+// A stdio(3)-backed implementation of ZoneInfoSource.
>+class FileZoneInfoSource : public ZoneInfoSource {
>+ public:
>+  static std::unique_ptr<ZoneInfoSource> Open(const std::string& name);
>+
>+  std::size_t Read(void* ptr, std::size_t size) override {
>+    size = std::min(size, len_);
>+    std::size_t nread = fread(ptr, 1, size, fp_.get());
>+    len_ -= nread;
>+    return nread;
>+  }
>+  int Skip(std::size_t offset) override {
>+    offset = std::min(offset, len_);
>+    int rc = fseek(fp_.get(), static_cast<long>(offset), SEEK_CUR);
>+    if (rc == 0) len_ -= offset;
>+    return rc;
>+  }
>+  std::string Version() const override {
>+    // TODO: It would nice if the zoneinfo data included the tzdb version.
>+    return std::string();
>+  }
>+
>+ protected:
>+  explicit FileZoneInfoSource(
>+      FILE* fp, std::size_t len = std::numeric_limits<std::size_t>::max())
>+      : fp_(fp, fclose), len_(len) {}
>+
>+ private:
>+  std::unique_ptr<FILE, int(*)(FILE*)> fp_;
>+  std::size_t len_;
>+};
>+
>+std::unique_ptr<ZoneInfoSource> FileZoneInfoSource::Open(
>+    const std::string& name) {
>+  // Use of the "file:" prefix is intended for testing purposes only.
>+  if (name.compare(0, 5, "file:") == 0) return Open(name.substr(5));
>+
>+  // Map the time-zone name to a path name.
>+  std::string path;
>+  if (name.empty() || name[0] != '/') {
>+    const char* tzdir = "/usr/share/zoneinfo";
>+    char* tzdir_env = nullptr;
>+#if defined(_MSC_VER)
>+    _dupenv_s(&tzdir_env, nullptr, "TZDIR");
>+#else
>+    tzdir_env = std::getenv("TZDIR");
>+#endif
>+    if (tzdir_env && *tzdir_env) tzdir = tzdir_env;
>+    path += tzdir;
>+    path += '/';
>+#if defined(_MSC_VER)
>+    free(tzdir_env);
>+#endif
>+  }
>+  path += name;
>+
>+  // Open the zoneinfo file.
>+  FILE* fp = FOpen(path.c_str(), "rb");
>+  if (fp == nullptr) return nullptr;
>+  std::size_t length = 0;
>+  if (fseek(fp, 0, SEEK_END) == 0) {
>+    long pos = ftell(fp);
>+    if (pos >= 0) {
>+      length = static_cast<std::size_t>(pos);
>+    }
>+    rewind(fp);
>+  }
>+  return std::unique_ptr<ZoneInfoSource>(new FileZoneInfoSource(fp, length));
>+}
>+
>+class AndroidZoneInfoSource : public FileZoneInfoSource {
>+ public:
>+  static std::unique_ptr<ZoneInfoSource> Open(const std::string& name);
>+  std::string Version() const override { return version_; }
>+
>+ private:
>+  explicit AndroidZoneInfoSource(FILE* fp, std::size_t len, const char* vers)
>+      : FileZoneInfoSource(fp, len), version_(vers) {}
>+  std::string version_;
>+};
>+
>+std::unique_ptr<ZoneInfoSource> AndroidZoneInfoSource::Open(
>+    const std::string& name) {
>+  // Use of the "file:" prefix is intended for testing purposes only.
>+  if (name.compare(0, 5, "file:") == 0) return Open(name.substr(5));
>+
>+#if defined(__ANDROID__)
>+  // See Android's libc/tzcode/bionic.cpp for additional information.
>+  for (const char* tzdata : {"/data/misc/zoneinfo/current/tzdata",
>+                             "/system/usr/share/zoneinfo/tzdata"}) {
>+    std::unique_ptr<FILE, int (*)(FILE*)> fp(FOpen(tzdata, "rb"), fclose);
>+    if (fp.get() == nullptr) continue;
>+
>+    char hbuf[24];  // covers header.zonetab_offset too
>+    if (fread(hbuf, 1, sizeof(hbuf), fp.get()) != sizeof(hbuf)) continue;
>+    if (strncmp(hbuf, "tzdata", 6) != 0) continue;
>+    const char* vers = (hbuf[11] == '\0') ? hbuf + 6 : "";
>+    const std::int_fast32_t index_offset = Decode32(hbuf + 12);
>+    const std::int_fast32_t data_offset = Decode32(hbuf + 16);
>+    if (index_offset < 0 || data_offset < index_offset) continue;
>+    if (fseek(fp.get(), static_cast<long>(index_offset), SEEK_SET) != 0)
>+      continue;
>+
>+    char ebuf[52];  // covers entry.unused too
>+    const std::size_t index_size =
>+        static_cast<std::size_t>(data_offset - index_offset);
>+    const std::size_t zonecnt = index_size / sizeof(ebuf);
>+    if (zonecnt * sizeof(ebuf) != index_size) continue;
>+    for (std::size_t i = 0; i != zonecnt; ++i) {
>+      if (fread(ebuf, 1, sizeof(ebuf), fp.get()) != sizeof(ebuf)) break;
>+      const std::int_fast32_t start = data_offset + Decode32(ebuf + 40);
>+      const std::int_fast32_t length = Decode32(ebuf + 44);
>+      if (start < 0 || length < 0) break;
>+      ebuf[40] = '\0';  // ensure zone name is NUL terminated
>+      if (strcmp(name.c_str(), ebuf) == 0) {
>+        if (fseek(fp.get(), static_cast<long>(start), SEEK_SET) != 0) break;
>+        return std::unique_ptr<ZoneInfoSource>(new AndroidZoneInfoSource(
>+            fp.release(), static_cast<std::size_t>(length), vers));
>+      }
>+    }
>+  }
>+#endif  // __ANDROID__
>+  return nullptr;
>+}
>+
>+}  // namespace
>+
>+bool TimeZoneInfo::Load(const std::string& name) {
>+  // We can ensure that the loading of UTC or any other fixed-offset
>+  // zone never fails because the simple, fixed-offset state can be
>+  // internally generated. Note that this depends on our choice to not
>+  // accept leap-second encoded ("right") zoneinfo.
>+  auto offset = seconds::zero();
>+  if (FixedOffsetFromName(name, &offset)) {
>+    return ResetToBuiltinUTC(offset);
>+  }
>+
>+  // Find and use a ZoneInfoSource to load the named zone.
>+  auto zip = cctz_extension::zone_info_source_factory(
>+      name, [](const std::string& name) -> std::unique_ptr<ZoneInfoSource> {
>+        if (auto zip = FileZoneInfoSource::Open(name)) return zip;
>+        if (auto zip = AndroidZoneInfoSource::Open(name)) return zip;
>+        return nullptr;
>+      });
>+  return zip != nullptr && Load(name, zip.get());
>+}
>+
>+// BreakTime() translation for a particular transition type.
>+time_zone::absolute_lookup TimeZoneInfo::LocalTime(
>+    std::int_fast64_t unix_time, const TransitionType& tt) const {
>+  // A civil time in "+offset" looks like (time+offset) in UTC.
>+  // Note: We perform two additions in the civil_second domain to
>+  // sidestep the chance of overflow in (unix_time + tt.utc_offset).
>+  return {(civil_second() + unix_time) + tt.utc_offset,
>+          tt.utc_offset, tt.is_dst, &abbreviations_[tt.abbr_index]};
>+}
>+
>+// BreakTime() translation for a particular transition.
>+time_zone::absolute_lookup TimeZoneInfo::LocalTime(
>+    std::int_fast64_t unix_time, const Transition& tr) const {
>+  const TransitionType& tt = transition_types_[tr.type_index];
>+  // Note: (unix_time - tr.unix_time) will never overflow as we
>+  // have ensured that there is always a "nearby" transition.
>+  return {tr.civil_sec + (unix_time - tr.unix_time),  // TODO: Optimize.
>+          tt.utc_offset, tt.is_dst, &abbreviations_[tt.abbr_index]};
>+}
>+
>+// MakeTime() translation with a conversion-preserving +N * 400-year shift.
>+time_zone::civil_lookup TimeZoneInfo::TimeLocal(const civil_second& cs,
>+                                                year_t c4_shift) const {
>+  assert(last_year_ - 400 < cs.year() && cs.year() <= last_year_);
>+  time_zone::civil_lookup cl = MakeTime(cs);
>+  if (c4_shift > seconds::max().count() / kSecsPer400Years) {
>+    cl.pre = cl.trans = cl.post = time_point<seconds>::max();
>+  } else {
>+    const auto offset = seconds(c4_shift * kSecsPer400Years);
>+    const auto limit = time_point<seconds>::max() - offset;
>+    for (auto* tp : {&cl.pre, &cl.trans, &cl.post}) {
>+      if (*tp > limit) {
>+        *tp = time_point<seconds>::max();
>+      } else {
>+        *tp += offset;
>+      }
>+    }
>+  }
>+  return cl;
>+}
>+
>+time_zone::absolute_lookup TimeZoneInfo::BreakTime(
>+    const time_point<seconds>& tp) const {
>+  std::int_fast64_t unix_time = ToUnixSeconds(tp);
>+  const std::size_t timecnt = transitions_.size();
>+  assert(timecnt != 0);  // We always add a transition.
>+
>+  if (unix_time < transitions_[0].unix_time) {
>+    return LocalTime(unix_time, transition_types_[default_transition_type_]);
>+  }
>+  if (unix_time >= transitions_[timecnt - 1].unix_time) {
>+    // After the last transition. If we extended the transitions using
>+    // future_spec_, shift back to a supported year using the 400-year
>+    // cycle of calendaric equivalence and then compensate accordingly.
>+    if (extended_) {
>+      const std::int_fast64_t diff =
>+          unix_time - transitions_[timecnt - 1].unix_time;
>+      const year_t shift = diff / kSecsPer400Years + 1;
>+      const auto d = seconds(shift * kSecsPer400Years);
>+      time_zone::absolute_lookup al = BreakTime(tp - d);
>+      al.cs = YearShift(al.cs, shift * 400);
>+      return al;
>+    }
>+    return LocalTime(unix_time, transitions_[timecnt - 1]);
>+  }
>+
>+  const std::size_t hint = local_time_hint_.load(std::memory_order_relaxed);
>+  if (0 < hint && hint < timecnt) {
>+    if (transitions_[hint - 1].unix_time <= unix_time) {
>+      if (unix_time < transitions_[hint].unix_time) {
>+        return LocalTime(unix_time, transitions_[hint - 1]);
>+      }
>+    }
>+  }
>+
>+  const Transition target = {unix_time, 0, civil_second(), civil_second()};
>+  const Transition* begin = &transitions_[0];
>+  const Transition* tr = std::upper_bound(begin, begin + timecnt, target,
>+                                          Transition::ByUnixTime());
>+  local_time_hint_.store(static_cast<std::size_t>(tr - begin),
>+                         std::memory_order_relaxed);
>+  return LocalTime(unix_time, *--tr);
>+}
>+
>+time_zone::civil_lookup TimeZoneInfo::MakeTime(const civil_second& cs) const {
>+  const std::size_t timecnt = transitions_.size();
>+  assert(timecnt != 0);  // We always add a transition.
>+
>+  // Find the first transition after our target civil time.
>+  const Transition* tr = nullptr;
>+  const Transition* begin = &transitions_[0];
>+  const Transition* end = begin + timecnt;
>+  if (cs < begin->civil_sec) {
>+    tr = begin;
>+  } else if (cs >= transitions_[timecnt - 1].civil_sec) {
>+    tr = end;
>+  } else {
>+    const std::size_t hint = time_local_hint_.load(std::memory_order_relaxed);
>+    if (0 < hint && hint < timecnt) {
>+      if (transitions_[hint - 1].civil_sec <= cs) {
>+        if (cs < transitions_[hint].civil_sec) {
>+          tr = begin + hint;
>+        }
>+      }
>+    }
>+    if (tr == nullptr) {
>+      const Transition target = {0, 0, cs, civil_second()};
>+      tr = std::upper_bound(begin, end, target, Transition::ByCivilTime());
>+      time_local_hint_.store(static_cast<std::size_t>(tr - begin),
>+                             std::memory_order_relaxed);
>+    }
>+  }
>+
>+  if (tr == begin) {
>+    if (tr->prev_civil_sec >= cs) {
>+      // Before first transition, so use the default offset.
>+      const TransitionType& tt(transition_types_[default_transition_type_]);
>+      if (cs < tt.civil_min) return MakeUnique(time_point<seconds>::min());
>+      return MakeUnique(cs - (civil_second() + tt.utc_offset));
>+    }
>+    // tr->prev_civil_sec < cs < tr->civil_sec
>+    return MakeSkipped(*tr, cs);
>+  }
>+
>+  if (tr == end) {
>+    if (cs > (--tr)->prev_civil_sec) {
>+      // After the last transition. If we extended the transitions using
>+      // future_spec_, shift back to a supported year using the 400-year
>+      // cycle of calendaric equivalence and then compensate accordingly.
>+      if (extended_ && cs.year() > last_year_) {
>+        const year_t shift = (cs.year() - last_year_ - 1) / 400 + 1;
>+        return TimeLocal(YearShift(cs, shift * -400), shift);
>+      }
>+      const TransitionType& tt(transition_types_[tr->type_index]);
>+      if (cs > tt.civil_max) return MakeUnique(time_point<seconds>::max());
>+      return MakeUnique(tr->unix_time + (cs - tr->civil_sec));
>+    }
>+    // tr->civil_sec <= cs <= tr->prev_civil_sec
>+    return MakeRepeated(*tr, cs);
>+  }
>+
>+  if (tr->prev_civil_sec < cs) {
>+    // tr->prev_civil_sec < cs < tr->civil_sec
>+    return MakeSkipped(*tr, cs);
>+  }
>+
>+  if (cs <= (--tr)->prev_civil_sec) {
>+    // tr->civil_sec <= cs <= tr->prev_civil_sec
>+    return MakeRepeated(*tr, cs);
>+  }
>+
>+  // In between transitions.
>+  return MakeUnique(tr->unix_time + (cs - tr->civil_sec));
>+}
>+
>+std::string TimeZoneInfo::Version() const {
>+  return version_;
>+}
>+
>+std::string TimeZoneInfo::Description() const {
>+  std::ostringstream oss;
>+  oss << "#trans=" << transitions_.size();
>+  oss << " #types=" << transition_types_.size();
>+  oss << " spec='" << future_spec_ << "'";
>+  return oss.str();
>+}
>+
>+bool TimeZoneInfo::NextTransition(const time_point<seconds>& tp,
>+                                  time_zone::civil_transition* trans) const {
>+  if (transitions_.empty()) return false;
>+  const Transition* begin = &transitions_[0];
>+  const Transition* end = begin + transitions_.size();
>+  if (begin->unix_time <= -(1LL << 59)) {
>+    // Do not report the BIG_BANG found in recent zoneinfo data as it is
>+    // really a sentinel, not a transition.  See tz/zic.c.
>+    ++begin;
>+  }
>+  std::int_fast64_t unix_time = ToUnixSeconds(tp);
>+  const Transition target = { unix_time };
>+  const Transition* tr = std::upper_bound(begin, end, target,
>+                                          Transition::ByUnixTime());
>+  for (; tr != end; ++tr) {  // skip no-op transitions
>+    std::uint_fast8_t prev_type_index =
>+        (tr == begin) ? default_transition_type_ : tr[-1].type_index;
>+    if (!EquivTransitions(prev_type_index, tr[0].type_index)) break;
>+  }
>+  // When tr == end we return false, ignoring future_spec_.
>+  if (tr == end) return false;
>+  trans->from = tr->prev_civil_sec + 1;
>+  trans->to = tr->civil_sec;
>+  return true;
>+}
>+
>+bool TimeZoneInfo::PrevTransition(const time_point<seconds>& tp,
>+                                  time_zone::civil_transition* trans) const {
>+  if (transitions_.empty()) return false;
>+  const Transition* begin = &transitions_[0];
>+  const Transition* end = begin + transitions_.size();
>+  if (begin->unix_time <= -(1LL << 59)) {
>+    // Do not report the BIG_BANG found in recent zoneinfo data as it is
>+    // really a sentinel, not a transition.  See tz/zic.c.
>+    ++begin;
>+  }
>+  std::int_fast64_t unix_time = ToUnixSeconds(tp);
>+  if (FromUnixSeconds(unix_time) != tp) {
>+    if (unix_time == std::numeric_limits<std::int_fast64_t>::max()) {
>+      if (end == begin) return false;  // Ignore future_spec_.
>+      trans->from = (--end)->prev_civil_sec + 1;
>+      trans->to = end->civil_sec;
>+      return true;
>+    }
>+    unix_time += 1;  // ceils
>+  }
>+  const Transition target = { unix_time };
>+  const Transition* tr = std::lower_bound(begin, end, target,
>+                                          Transition::ByUnixTime());
>+  for (; tr != begin; --tr) {  // skip no-op transitions
>+    std::uint_fast8_t prev_type_index =
>+        (tr - 1 == begin) ? default_transition_type_ : tr[-2].type_index;
>+    if (!EquivTransitions(prev_type_index, tr[-1].type_index)) break;
>+  }
>+  // When tr == end we return the "last" transition, ignoring future_spec_.
>+  if (tr == begin) return false;
>+  trans->from = (--tr)->prev_civil_sec + 1;
>+  trans->to = tr->civil_sec;
>+  return true;
>+}
>+
>+}  // namespace cctz
>+}  // namespace time_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_info.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_info.h
>new file mode 100644
>index 00000000000..958e9b6bd74
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_info.h
>@@ -0,0 +1,136 @@
>+// Copyright 2016 Google Inc. All Rights Reserved.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//   http://www.apache.org/licenses/LICENSE-2.0
>+//
>+//   Unless required by applicable law or agreed to in writing, software
>+//   distributed under the License is distributed on an "AS IS" BASIS,
>+//   WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+//   See the License for the specific language governing permissions and
>+//   limitations under the License.
>+
>+#ifndef ABSL_TIME_INTERNAL_CCTZ_TIME_ZONE_INFO_H_
>+#define ABSL_TIME_INTERNAL_CCTZ_TIME_ZONE_INFO_H_
>+
>+#include <atomic>
>+#include <cstddef>
>+#include <cstdint>
>+#include <string>
>+#include <vector>
>+
>+#include "absl/time/internal/cctz/include/cctz/civil_time.h"
>+#include "absl/time/internal/cctz/include/cctz/time_zone.h"
>+#include "absl/time/internal/cctz/include/cctz/zone_info_source.h"
>+#include "time_zone_if.h"
>+#include "tzfile.h"
>+
>+namespace absl {
>+namespace time_internal {
>+namespace cctz {
>+
>+// A transition to a new UTC offset.
>+struct Transition {
>+  std::int_least64_t unix_time;   // the instant of this transition
>+  std::uint_least8_t type_index;  // index of the transition type
>+  civil_second civil_sec;         // local civil time of transition
>+  civil_second prev_civil_sec;    // local civil time one second earlier
>+
>+  struct ByUnixTime {
>+    inline bool operator()(const Transition& lhs, const Transition& rhs) const {
>+      return lhs.unix_time < rhs.unix_time;
>+    }
>+  };
>+  struct ByCivilTime {
>+    inline bool operator()(const Transition& lhs, const Transition& rhs) const {
>+      return lhs.civil_sec < rhs.civil_sec;
>+    }
>+  };
>+};
>+
>+// The characteristics of a particular transition.
>+struct TransitionType {
>+  std::int_least32_t utc_offset;  // the new prevailing UTC offset
>+  civil_second civil_max;         // max convertible civil time for offset
>+  civil_second civil_min;         // min convertible civil time for offset
>+  bool is_dst;                    // did we move into daylight-saving time
>+  std::uint_least8_t abbr_index;  // index of the new abbreviation
>+};
>+
>+// A time zone backed by the IANA Time Zone Database (zoneinfo).
>+class TimeZoneInfo : public TimeZoneIf {
>+ public:
>+  TimeZoneInfo() = default;
>+  TimeZoneInfo(const TimeZoneInfo&) = delete;
>+  TimeZoneInfo& operator=(const TimeZoneInfo&) = delete;
>+
>+  // Loads the zoneinfo for the given name, returning true if successful.
>+  bool Load(const std::string& name);
>+
>+  // TimeZoneIf implementations.
>+  time_zone::absolute_lookup BreakTime(
>+      const time_point<seconds>& tp) const override;
>+  time_zone::civil_lookup MakeTime(
>+      const civil_second& cs) const override;
>+  bool NextTransition(const time_point<seconds>& tp,
>+                      time_zone::civil_transition* trans) const override;
>+  bool PrevTransition(const time_point<seconds>& tp,
>+                      time_zone::civil_transition* trans) const override;
>+  std::string Version() const override;
>+  std::string Description() const override;
>+
>+ private:
>+  struct Header {  // counts of:
>+    std::size_t timecnt;     // transition times
>+    std::size_t typecnt;     // transition types
>+    std::size_t charcnt;     // zone abbreviation characters
>+    std::size_t leapcnt;     // leap seconds (we expect none)
>+    std::size_t ttisstdcnt;  // UTC/local indicators (unused)
>+    std::size_t ttisgmtcnt;  // standard/wall indicators (unused)
>+
>+    bool Build(const tzhead& tzh);
>+    std::size_t DataLength(std::size_t time_len) const;
>+  };
>+
>+  void CheckTransition(const std::string& name, const TransitionType& tt,
>+                       std::int_fast32_t offset, bool is_dst,
>+                       const std::string& abbr) const;
>+  bool EquivTransitions(std::uint_fast8_t tt1_index,
>+                        std::uint_fast8_t tt2_index) const;
>+  void ExtendTransitions(const std::string& name, const Header& hdr);
>+
>+  bool ResetToBuiltinUTC(const seconds& offset);
>+  bool Load(const std::string& name, ZoneInfoSource* zip);
>+
>+  // Helpers for BreakTime() and MakeTime().
>+  time_zone::absolute_lookup LocalTime(std::int_fast64_t unix_time,
>+                                       const TransitionType& tt) const;
>+  time_zone::absolute_lookup LocalTime(std::int_fast64_t unix_time,
>+                                       const Transition& tr) const;
>+  time_zone::civil_lookup TimeLocal(const civil_second& cs,
>+                                    year_t c4_shift) const;
>+
>+  std::vector<Transition> transitions_;  // ordered by unix_time and civil_sec
>+  std::vector<TransitionType> transition_types_;  // distinct transition types
>+  std::uint_fast8_t default_transition_type_;  // for before first transition
>+  std::string abbreviations_;  // all the NUL-terminated abbreviations
>+
>+  std::string version_;      // the tzdata version if available
>+  std::string future_spec_;  // for after the last zic transition
>+  bool extended_;            // future_spec_ was used to generate transitions
>+  year_t last_year_;         // the final year of the generated transitions
>+
>+  // We remember the transitions found during the last BreakTime() and
>+  // MakeTime() calls. If the next request is for the same transition we
>+  // will avoid re-searching.
>+  mutable std::atomic<std::size_t> local_time_hint_ = {};  // BreakTime() hint
>+  mutable std::atomic<std::size_t> time_local_hint_ = {};  // MakeTime() hint
>+};
>+
>+}  // namespace cctz
>+}  // namespace time_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_TIME_INTERNAL_CCTZ_TIME_ZONE_INFO_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_libc.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_libc.cc
>new file mode 100644
>index 00000000000..074c8d0a4a4
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_libc.cc
>@@ -0,0 +1,162 @@
>+// Copyright 2016 Google Inc. All Rights Reserved.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//   http://www.apache.org/licenses/LICENSE-2.0
>+//
>+//   Unless required by applicable law or agreed to in writing, software
>+//   distributed under the License is distributed on an "AS IS" BASIS,
>+//   WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+//   See the License for the specific language governing permissions and
>+//   limitations under the License.
>+
>+#if defined(_WIN32) || defined(_WIN64)
>+#define _CRT_SECURE_NO_WARNINGS 1
>+#endif
>+
>+#include "time_zone_libc.h"
>+
>+#include <chrono>
>+#include <ctime>
>+#include <tuple>
>+#include <utility>
>+
>+#include "absl/time/internal/cctz/include/cctz/civil_time.h"
>+#include "absl/time/internal/cctz/include/cctz/time_zone.h"
>+
>+namespace absl {
>+namespace time_internal {
>+namespace cctz {
>+
>+namespace {
>+
>+// .first is seconds east of UTC; .second is the time-zone abbreviation.
>+using OffsetAbbr = std::pair<int, const char*>;
>+
>+// Defines a function that can be called as follows:
>+//
>+//   std::tm tm = ...;
>+//   OffsetAbbr off_abbr = get_offset_abbr(tm);
>+//
>+#if defined(_WIN32) || defined(_WIN64)
>+// Uses the globals: '_timezone', '_dstbias' and '_tzname'.
>+OffsetAbbr get_offset_abbr(const std::tm& tm) {
>+  const bool is_dst = tm.tm_isdst > 0;
>+  const int off = _timezone + (is_dst ? _dstbias : 0);
>+  const char* abbr = _tzname[is_dst];
>+  return {off, abbr};
>+}
>+#elif defined(__sun)
>+// Uses the globals: 'timezone', 'altzone' and 'tzname'.
>+OffsetAbbr get_offset_abbr(const std::tm& tm) {
>+  const bool is_dst = tm.tm_isdst > 0;
>+  const int off = is_dst ? altzone : timezone;
>+  const char* abbr = tzname[is_dst];
>+  return {off, abbr};
>+}
>+#elif defined(__native_client__) || defined(__myriad2__) || \
>+    defined(__EMSCRIPTEN__)
>+// Uses the globals: 'timezone' and 'tzname'.
>+OffsetAbbr get_offset_abbr(const std::tm& tm) {
>+  const bool is_dst = tm.tm_isdst > 0;
>+  const int off = _timezone + (is_dst ? 60 * 60 : 0);
>+  const char* abbr = tzname[is_dst];
>+  return {off, abbr};
>+}
>+#else
>+//
>+// Returns an OffsetAbbr using std::tm fields with various spellings.
>+//
>+#if !defined(tm_gmtoff) && !defined(tm_zone)
>+template <typename T>
>+OffsetAbbr get_offset_abbr(const T& tm, decltype(&T::tm_gmtoff) = nullptr,
>+                           decltype(&T::tm_zone) = nullptr) {
>+  return {tm.tm_gmtoff, tm.tm_zone};
>+}
>+#endif  // !defined(tm_gmtoff) && !defined(tm_zone)
>+#if !defined(__tm_gmtoff) && !defined(__tm_zone)
>+template <typename T>
>+OffsetAbbr get_offset_abbr(const T& tm, decltype(&T::__tm_gmtoff) = nullptr,
>+                           decltype(&T::__tm_zone) = nullptr) {
>+  return {tm.__tm_gmtoff, tm.__tm_zone};
>+}
>+#endif  // !defined(__tm_gmtoff) && !defined(__tm_zone)
>+#endif
>+
>+}  // namespace
>+
>+TimeZoneLibC::TimeZoneLibC(const std::string& name)
>+    : local_(name == "localtime") {}
>+
>+time_zone::absolute_lookup TimeZoneLibC::BreakTime(
>+    const time_point<seconds>& tp) const {
>+  time_zone::absolute_lookup al;
>+  std::time_t t = ToUnixSeconds(tp);
>+  std::tm tm;
>+  if (local_) {
>+#if defined(_WIN32) || defined(_WIN64)
>+    localtime_s(&tm, &t);
>+#else
>+    localtime_r(&t, &tm);
>+#endif
>+    std::tie(al.offset, al.abbr) = get_offset_abbr(tm);
>+  } else {
>+#if defined(_WIN32) || defined(_WIN64)
>+    gmtime_s(&tm, &t);
>+#else
>+    gmtime_r(&t, &tm);
>+#endif
>+    al.offset = 0;
>+    al.abbr = "UTC";
>+  }
>+  al.cs = civil_second(tm.tm_year + 1900, tm.tm_mon + 1, tm.tm_mday,
>+                       tm.tm_hour, tm.tm_min, tm.tm_sec);
>+  al.is_dst = tm.tm_isdst > 0;
>+  return al;
>+}
>+
>+time_zone::civil_lookup TimeZoneLibC::MakeTime(const civil_second& cs) const {
>+  time_zone::civil_lookup cl;
>+  std::time_t t;
>+  if (local_) {
>+    // Does not handle SKIPPED/AMBIGUOUS or huge years.
>+    std::tm tm;
>+    tm.tm_year = static_cast<int>(cs.year() - 1900);
>+    tm.tm_mon = cs.month() - 1;
>+    tm.tm_mday = cs.day();
>+    tm.tm_hour = cs.hour();
>+    tm.tm_min = cs.minute();
>+    tm.tm_sec = cs.second();
>+    tm.tm_isdst = -1;
>+    t = std::mktime(&tm);
>+  } else {
>+    t = cs - civil_second();
>+  }
>+  cl.kind = time_zone::civil_lookup::UNIQUE;
>+  cl.pre = cl.trans = cl.post = FromUnixSeconds(t);
>+  return cl;
>+}
>+
>+bool TimeZoneLibC::NextTransition(const time_point<seconds>& tp,
>+                                  time_zone::civil_transition* trans) const {
>+  return false;
>+}
>+
>+bool TimeZoneLibC::PrevTransition(const time_point<seconds>& tp,
>+                                  time_zone::civil_transition* trans) const {
>+  return false;
>+}
>+
>+std::string TimeZoneLibC::Version() const {
>+  return std::string();  // unknown
>+}
>+
>+std::string TimeZoneLibC::Description() const {
>+  return local_ ? "localtime" : "UTC";
>+}
>+
>+}  // namespace cctz
>+}  // namespace time_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_libc.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_libc.h
>new file mode 100644
>index 00000000000..4e40c61ab24
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_libc.h
>@@ -0,0 +1,53 @@
>+// Copyright 2016 Google Inc. All Rights Reserved.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//   http://www.apache.org/licenses/LICENSE-2.0
>+//
>+//   Unless required by applicable law or agreed to in writing, software
>+//   distributed under the License is distributed on an "AS IS" BASIS,
>+//   WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+//   See the License for the specific language governing permissions and
>+//   limitations under the License.
>+
>+#ifndef ABSL_TIME_INTERNAL_CCTZ_TIME_ZONE_LIBC_H_
>+#define ABSL_TIME_INTERNAL_CCTZ_TIME_ZONE_LIBC_H_
>+
>+#include <string>
>+
>+#include "time_zone_if.h"
>+
>+namespace absl {
>+namespace time_internal {
>+namespace cctz {
>+
>+// A time zone backed by gmtime_r(3), localtime_r(3), and mktime(3),
>+// and which therefore only supports UTC and the local time zone.
>+// TODO: Add support for fixed offsets from UTC.
>+class TimeZoneLibC : public TimeZoneIf {
>+ public:
>+  explicit TimeZoneLibC(const std::string& name);
>+
>+  // TimeZoneIf implementations.
>+  time_zone::absolute_lookup BreakTime(
>+      const time_point<seconds>& tp) const override;
>+  time_zone::civil_lookup MakeTime(
>+      const civil_second& cs) const override;
>+  bool NextTransition(const time_point<seconds>& tp,
>+                      time_zone::civil_transition* trans) const override;
>+  bool PrevTransition(const time_point<seconds>& tp,
>+                      time_zone::civil_transition* trans) const override;
>+  std::string Version() const override;
>+  std::string Description() const override;
>+
>+ private:
>+  const bool local_;  // localtime or UTC
>+};
>+
>+}  // namespace cctz
>+}  // namespace time_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_TIME_INTERNAL_CCTZ_TIME_ZONE_LIBC_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_lookup.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_lookup.cc
>new file mode 100644
>index 00000000000..f2d151e4d5e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_lookup.cc
>@@ -0,0 +1,168 @@
>+// Copyright 2016 Google Inc. All Rights Reserved.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//   http://www.apache.org/licenses/LICENSE-2.0
>+//
>+//   Unless required by applicable law or agreed to in writing, software
>+//   distributed under the License is distributed on an "AS IS" BASIS,
>+//   WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+//   See the License for the specific language governing permissions and
>+//   limitations under the License.
>+
>+#include "absl/time/internal/cctz/include/cctz/time_zone.h"
>+
>+#if defined(__ANDROID__)
>+#include <sys/system_properties.h>
>+#if __ANDROID_API__ >= 21
>+#include <dlfcn.h>
>+#endif
>+#endif
>+#include <cstdlib>
>+#include <cstring>
>+#include <string>
>+
>+#include "time_zone_fixed.h"
>+#include "time_zone_impl.h"
>+
>+namespace absl {
>+namespace time_internal {
>+namespace cctz {
>+
>+#if defined(__ANDROID__) && __ANDROID_API__ >= 21
>+namespace {
>+// Android 'L' removes __system_property_get() from the NDK, however
>+// it is still a hidden symbol in libc so we use dlsym() to access it.
>+// See Chromium's base/sys_info_android.cc for a similar example.
>+
>+using property_get_func = int (*)(const char*, char*);
>+
>+property_get_func LoadSystemPropertyGet() {
>+  int flag = RTLD_LAZY | RTLD_GLOBAL;
>+#if defined(RTLD_NOLOAD)
>+  flag |= RTLD_NOLOAD;  // libc.so should already be resident
>+#endif
>+  if (void* handle = dlopen("libc.so", flag)) {
>+    void* sym = dlsym(handle, "__system_property_get");
>+    dlclose(handle);
>+    return reinterpret_cast<property_get_func>(sym);
>+  }
>+  return nullptr;
>+}
>+
>+int __system_property_get(const char* name, char* value) {
>+  static property_get_func system_property_get = LoadSystemPropertyGet();
>+  return system_property_get ? system_property_get(name, value) : -1;
>+}
>+
>+}  // namespace
>+#endif
>+
>+std::string time_zone::name() const {
>+  return effective_impl().Name();
>+}
>+
>+time_zone::absolute_lookup time_zone::lookup(
>+    const time_point<seconds>& tp) const {
>+  return effective_impl().BreakTime(tp);
>+}
>+
>+time_zone::civil_lookup time_zone::lookup(const civil_second& cs) const {
>+  return effective_impl().MakeTime(cs);
>+}
>+
>+bool time_zone::next_transition(const time_point<seconds>& tp,
>+                                civil_transition* trans) const {
>+  return effective_impl().NextTransition(tp, trans);
>+}
>+
>+bool time_zone::prev_transition(const time_point<seconds>& tp,
>+                                civil_transition* trans) const {
>+  return effective_impl().PrevTransition(tp, trans);
>+}
>+
>+std::string time_zone::version() const {
>+  return effective_impl().Version();
>+}
>+
>+std::string time_zone::description() const {
>+  return effective_impl().Description();
>+}
>+
>+const time_zone::Impl& time_zone::effective_impl() const {
>+  if (impl_ == nullptr) {
>+    // Dereferencing an implicit-UTC time_zone is expected to be
>+    // rare, so we don't mind paying a small synchronization cost.
>+    return *time_zone::Impl::UTC().impl_;
>+  }
>+  return *impl_;
>+}
>+
>+bool load_time_zone(const std::string& name, time_zone* tz) {
>+  return time_zone::Impl::LoadTimeZone(name, tz);
>+}
>+
>+time_zone utc_time_zone() {
>+  return time_zone::Impl::UTC();  // avoid name lookup
>+}
>+
>+time_zone fixed_time_zone(const seconds& offset) {
>+  time_zone tz;
>+  load_time_zone(FixedOffsetToName(offset), &tz);
>+  return tz;
>+}
>+
>+time_zone local_time_zone() {
>+  const char* zone = ":localtime";
>+
>+  // Allow ${TZ} to override to default zone.
>+  char* tz_env = nullptr;
>+#if defined(_MSC_VER)
>+  _dupenv_s(&tz_env, nullptr, "TZ");
>+#else
>+  tz_env = std::getenv("TZ");
>+#endif
>+#if defined(__ANDROID__)
>+  char sysprop[PROP_VALUE_MAX];
>+  if (tz_env == nullptr)
>+    if (__system_property_get("persist.sys.timezone", sysprop) > 0)
>+      tz_env = sysprop;
>+#endif
>+  if (tz_env) zone = tz_env;
>+
>+  // We only support the "[:]<zone-name>" form.
>+  if (*zone == ':') ++zone;
>+
>+  // Map "localtime" to a system-specific name, but
>+  // allow ${LOCALTIME} to override the default name.
>+  char* localtime_env = nullptr;
>+  if (strcmp(zone, "localtime") == 0) {
>+#if defined(_MSC_VER)
>+    // System-specific default is just "localtime".
>+    _dupenv_s(&localtime_env, nullptr, "LOCALTIME");
>+#else
>+    zone = "/etc/localtime";  // System-specific default.
>+    localtime_env = std::getenv("LOCALTIME");
>+#endif
>+    if (localtime_env) zone = localtime_env;
>+  }
>+
>+  const std::string name = zone;
>+#if defined(_MSC_VER)
>+  free(localtime_env);
>+  free(tz_env);
>+#endif
>+
>+  time_zone tz;
>+  load_time_zone(name, &tz);  // Falls back to UTC.
>+  // TODO: Follow the RFC3339 "Unknown Local Offset Convention" and
>+  // arrange for %z to generate "-0000" when we don't know the local
>+  // offset because the load_time_zone() failed and we're using UTC.
>+  return tz;
>+}
>+
>+}  // namespace cctz
>+}  // namespace time_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_lookup_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_lookup_test.cc
>new file mode 100644
>index 00000000000..551292fb55e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_lookup_test.cc
>@@ -0,0 +1,1331 @@
>+// Copyright 2016 Google Inc. All Rights Reserved.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//   http://www.apache.org/licenses/LICENSE-2.0
>+//
>+//   Unless required by applicable law or agreed to in writing, software
>+//   distributed under the License is distributed on an "AS IS" BASIS,
>+//   WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+//   See the License for the specific language governing permissions and
>+//   limitations under the License.
>+
>+#include "absl/time/internal/cctz/include/cctz/time_zone.h"
>+
>+#include <chrono>
>+#include <cstddef>
>+#include <future>
>+#include <string>
>+#include <thread>
>+#include <vector>
>+
>+#include "absl/time/internal/cctz/include/cctz/civil_time.h"
>+#include "gtest/gtest.h"
>+
>+namespace chrono = std::chrono;
>+
>+namespace absl {
>+namespace time_internal {
>+namespace cctz {
>+
>+namespace {
>+
>+// A list of known time-zone names.
>+const char* const kTimeZoneNames[] = {
>+  "Africa/Abidjan",
>+  "Africa/Accra",
>+  "Africa/Addis_Ababa",
>+  "Africa/Algiers",
>+  "Africa/Asmara",
>+  "Africa/Asmera",
>+  "Africa/Bamako",
>+  "Africa/Bangui",
>+  "Africa/Banjul",
>+  "Africa/Bissau",
>+  "Africa/Blantyre",
>+  "Africa/Brazzaville",
>+  "Africa/Bujumbura",
>+  "Africa/Cairo",
>+  "Africa/Casablanca",
>+  "Africa/Ceuta",
>+  "Africa/Conakry",
>+  "Africa/Dakar",
>+  "Africa/Dar_es_Salaam",
>+  "Africa/Djibouti",
>+  "Africa/Douala",
>+  "Africa/El_Aaiun",
>+  "Africa/Freetown",
>+  "Africa/Gaborone",
>+  "Africa/Harare",
>+  "Africa/Johannesburg",
>+  "Africa/Juba",
>+  "Africa/Kampala",
>+  "Africa/Khartoum",
>+  "Africa/Kigali",
>+  "Africa/Kinshasa",
>+  "Africa/Lagos",
>+  "Africa/Libreville",
>+  "Africa/Lome",
>+  "Africa/Luanda",
>+  "Africa/Lubumbashi",
>+  "Africa/Lusaka",
>+  "Africa/Malabo",
>+  "Africa/Maputo",
>+  "Africa/Maseru",
>+  "Africa/Mbabane",
>+  "Africa/Mogadishu",
>+  "Africa/Monrovia",
>+  "Africa/Nairobi",
>+  "Africa/Ndjamena",
>+  "Africa/Niamey",
>+  "Africa/Nouakchott",
>+  "Africa/Ouagadougou",
>+  "Africa/Porto-Novo",
>+  "Africa/Sao_Tome",
>+  "Africa/Timbuktu",
>+  "Africa/Tripoli",
>+  "Africa/Tunis",
>+  "Africa/Windhoek",
>+  "America/Adak",
>+  "America/Anchorage",
>+  "America/Anguilla",
>+  "America/Antigua",
>+  "America/Araguaina",
>+  "America/Argentina/Buenos_Aires",
>+  "America/Argentina/Catamarca",
>+  "America/Argentina/ComodRivadavia",
>+  "America/Argentina/Cordoba",
>+  "America/Argentina/Jujuy",
>+  "America/Argentina/La_Rioja",
>+  "America/Argentina/Mendoza",
>+  "America/Argentina/Rio_Gallegos",
>+  "America/Argentina/Salta",
>+  "America/Argentina/San_Juan",
>+  "America/Argentina/San_Luis",
>+  "America/Argentina/Tucuman",
>+  "America/Argentina/Ushuaia",
>+  "America/Aruba",
>+  "America/Asuncion",
>+  "America/Atikokan",
>+  "America/Atka",
>+  "America/Bahia",
>+  "America/Bahia_Banderas",
>+  "America/Barbados",
>+  "America/Belem",
>+  "America/Belize",
>+  "America/Blanc-Sablon",
>+  "America/Boa_Vista",
>+  "America/Bogota",
>+  "America/Boise",
>+  "America/Buenos_Aires",
>+  "America/Cambridge_Bay",
>+  "America/Campo_Grande",
>+  "America/Cancun",
>+  "America/Caracas",
>+  "America/Catamarca",
>+  "America/Cayenne",
>+  "America/Cayman",
>+  "America/Chicago",
>+  "America/Chihuahua",
>+  "America/Coral_Harbour",
>+  "America/Cordoba",
>+  "America/Costa_Rica",
>+  "America/Creston",
>+  "America/Cuiaba",
>+  "America/Curacao",
>+  "America/Danmarkshavn",
>+  "America/Dawson",
>+  "America/Dawson_Creek",
>+  "America/Denver",
>+  "America/Detroit",
>+  "America/Dominica",
>+  "America/Edmonton",
>+  "America/Eirunepe",
>+  "America/El_Salvador",
>+  "America/Ensenada",
>+  "America/Fort_Nelson",
>+  "America/Fort_Wayne",
>+  "America/Fortaleza",
>+  "America/Glace_Bay",
>+  "America/Godthab",
>+  "America/Goose_Bay",
>+  "America/Grand_Turk",
>+  "America/Grenada",
>+  "America/Guadeloupe",
>+  "America/Guatemala",
>+  "America/Guayaquil",
>+  "America/Guyana",
>+  "America/Halifax",
>+  "America/Havana",
>+  "America/Hermosillo",
>+  "America/Indiana/Indianapolis",
>+  "America/Indiana/Knox",
>+  "America/Indiana/Marengo",
>+  "America/Indiana/Petersburg",
>+  "America/Indiana/Tell_City",
>+  "America/Indiana/Vevay",
>+  "America/Indiana/Vincennes",
>+  "America/Indiana/Winamac",
>+  "America/Indianapolis",
>+  "America/Inuvik",
>+  "America/Iqaluit",
>+  "America/Jamaica",
>+  "America/Jujuy",
>+  "America/Juneau",
>+  "America/Kentucky/Louisville",
>+  "America/Kentucky/Monticello",
>+  "America/Knox_IN",
>+  "America/Kralendijk",
>+  "America/La_Paz",
>+  "America/Lima",
>+  "America/Los_Angeles",
>+  "America/Louisville",
>+  "America/Lower_Princes",
>+  "America/Maceio",
>+  "America/Managua",
>+  "America/Manaus",
>+  "America/Marigot",
>+  "America/Martinique",
>+  "America/Matamoros",
>+  "America/Mazatlan",
>+  "America/Mendoza",
>+  "America/Menominee",
>+  "America/Merida",
>+  "America/Metlakatla",
>+  "America/Mexico_City",
>+  "America/Miquelon",
>+  "America/Moncton",
>+  "America/Monterrey",
>+  "America/Montevideo",
>+  "America/Montreal",
>+  "America/Montserrat",
>+  "America/Nassau",
>+  "America/New_York",
>+  "America/Nipigon",
>+  "America/Nome",
>+  "America/Noronha",
>+  "America/North_Dakota/Beulah",
>+  "America/North_Dakota/Center",
>+  "America/North_Dakota/New_Salem",
>+  "America/Ojinaga",
>+  "America/Panama",
>+  "America/Pangnirtung",
>+  "America/Paramaribo",
>+  "America/Phoenix",
>+  "America/Port-au-Prince",
>+  "America/Port_of_Spain",
>+  "America/Porto_Acre",
>+  "America/Porto_Velho",
>+  "America/Puerto_Rico",
>+  "America/Punta_Arenas",
>+  "America/Rainy_River",
>+  "America/Rankin_Inlet",
>+  "America/Recife",
>+  "America/Regina",
>+  "America/Resolute",
>+  "America/Rio_Branco",
>+  "America/Rosario",
>+  "America/Santa_Isabel",
>+  "America/Santarem",
>+  "America/Santiago",
>+  "America/Santo_Domingo",
>+  "America/Sao_Paulo",
>+  "America/Scoresbysund",
>+  "America/Shiprock",
>+  "America/Sitka",
>+  "America/St_Barthelemy",
>+  "America/St_Johns",
>+  "America/St_Kitts",
>+  "America/St_Lucia",
>+  "America/St_Thomas",
>+  "America/St_Vincent",
>+  "America/Swift_Current",
>+  "America/Tegucigalpa",
>+  "America/Thule",
>+  "America/Thunder_Bay",
>+  "America/Tijuana",
>+  "America/Toronto",
>+  "America/Tortola",
>+  "America/Vancouver",
>+  "America/Virgin",
>+  "America/Whitehorse",
>+  "America/Winnipeg",
>+  "America/Yakutat",
>+  "America/Yellowknife",
>+  "Antarctica/Casey",
>+  "Antarctica/Davis",
>+  "Antarctica/DumontDUrville",
>+  "Antarctica/Macquarie",
>+  "Antarctica/Mawson",
>+  "Antarctica/McMurdo",
>+  "Antarctica/Palmer",
>+  "Antarctica/Rothera",
>+  "Antarctica/South_Pole",
>+  "Antarctica/Syowa",
>+  "Antarctica/Troll",
>+  "Antarctica/Vostok",
>+  "Arctic/Longyearbyen",
>+  "Asia/Aden",
>+  "Asia/Almaty",
>+  "Asia/Amman",
>+  "Asia/Anadyr",
>+  "Asia/Aqtau",
>+  "Asia/Aqtobe",
>+  "Asia/Ashgabat",
>+  "Asia/Ashkhabad",
>+  "Asia/Atyrau",
>+  "Asia/Baghdad",
>+  "Asia/Bahrain",
>+  "Asia/Baku",
>+  "Asia/Bangkok",
>+  "Asia/Barnaul",
>+  "Asia/Beirut",
>+  "Asia/Bishkek",
>+  "Asia/Brunei",
>+  "Asia/Calcutta",
>+  "Asia/Chita",
>+  "Asia/Choibalsan",
>+  "Asia/Chongqing",
>+  "Asia/Chungking",
>+  "Asia/Colombo",
>+  "Asia/Dacca",
>+  "Asia/Damascus",
>+  "Asia/Dhaka",
>+  "Asia/Dili",
>+  "Asia/Dubai",
>+  "Asia/Dushanbe",
>+  "Asia/Famagusta",
>+  "Asia/Gaza",
>+  "Asia/Harbin",
>+  "Asia/Hebron",
>+  "Asia/Ho_Chi_Minh",
>+  "Asia/Hong_Kong",
>+  "Asia/Hovd",
>+  "Asia/Irkutsk",
>+  "Asia/Istanbul",
>+  "Asia/Jakarta",
>+  "Asia/Jayapura",
>+  "Asia/Jerusalem",
>+  "Asia/Kabul",
>+  "Asia/Kamchatka",
>+  "Asia/Karachi",
>+  "Asia/Kashgar",
>+  "Asia/Kathmandu",
>+  "Asia/Katmandu",
>+  "Asia/Khandyga",
>+  "Asia/Kolkata",
>+  "Asia/Krasnoyarsk",
>+  "Asia/Kuala_Lumpur",
>+  "Asia/Kuching",
>+  "Asia/Kuwait",
>+  "Asia/Macao",
>+  "Asia/Macau",
>+  "Asia/Magadan",
>+  "Asia/Makassar",
>+  "Asia/Manila",
>+  "Asia/Muscat",
>+  "Asia/Nicosia",
>+  "Asia/Novokuznetsk",
>+  "Asia/Novosibirsk",
>+  "Asia/Omsk",
>+  "Asia/Oral",
>+  "Asia/Phnom_Penh",
>+  "Asia/Pontianak",
>+  "Asia/Pyongyang",
>+  "Asia/Qatar",
>+  "Asia/Qyzylorda",
>+  "Asia/Rangoon",
>+  "Asia/Riyadh",
>+  "Asia/Saigon",
>+  "Asia/Sakhalin",
>+  "Asia/Samarkand",
>+  "Asia/Seoul",
>+  "Asia/Shanghai",
>+  "Asia/Singapore",
>+  "Asia/Srednekolymsk",
>+  "Asia/Taipei",
>+  "Asia/Tashkent",
>+  "Asia/Tbilisi",
>+  "Asia/Tehran",
>+  "Asia/Tel_Aviv",
>+  "Asia/Thimbu",
>+  "Asia/Thimphu",
>+  "Asia/Tokyo",
>+  "Asia/Tomsk",
>+  "Asia/Ujung_Pandang",
>+  "Asia/Ulaanbaatar",
>+  "Asia/Ulan_Bator",
>+  "Asia/Urumqi",
>+  "Asia/Ust-Nera",
>+  "Asia/Vientiane",
>+  "Asia/Vladivostok",
>+  "Asia/Yakutsk",
>+  "Asia/Yangon",
>+  "Asia/Yekaterinburg",
>+  "Asia/Yerevan",
>+  "Atlantic/Azores",
>+  "Atlantic/Bermuda",
>+  "Atlantic/Canary",
>+  "Atlantic/Cape_Verde",
>+  "Atlantic/Faeroe",
>+  "Atlantic/Faroe",
>+  "Atlantic/Jan_Mayen",
>+  "Atlantic/Madeira",
>+  "Atlantic/Reykjavik",
>+  "Atlantic/South_Georgia",
>+  "Atlantic/St_Helena",
>+  "Atlantic/Stanley",
>+  "Australia/ACT",
>+  "Australia/Adelaide",
>+  "Australia/Brisbane",
>+  "Australia/Broken_Hill",
>+  "Australia/Canberra",
>+  "Australia/Currie",
>+  "Australia/Darwin",
>+  "Australia/Eucla",
>+  "Australia/Hobart",
>+  "Australia/LHI",
>+  "Australia/Lindeman",
>+  "Australia/Lord_Howe",
>+  "Australia/Melbourne",
>+  "Australia/NSW",
>+  "Australia/North",
>+  "Australia/Perth",
>+  "Australia/Queensland",
>+  "Australia/South",
>+  "Australia/Sydney",
>+  "Australia/Tasmania",
>+  "Australia/Victoria",
>+  "Australia/West",
>+  "Australia/Yancowinna",
>+  "Brazil/Acre",
>+  "Brazil/DeNoronha",
>+  "Brazil/East",
>+  "Brazil/West",
>+  "CET",
>+  "CST6CDT",
>+  "Canada/Atlantic",
>+  "Canada/Central",
>+  "Canada/Eastern",
>+  "Canada/Mountain",
>+  "Canada/Newfoundland",
>+  "Canada/Pacific",
>+  "Canada/Saskatchewan",
>+  "Canada/Yukon",
>+  "Chile/Continental",
>+  "Chile/EasterIsland",
>+  "Cuba",
>+  "EET",
>+  "EST",
>+  "EST5EDT",
>+  "Egypt",
>+  "Eire",
>+  "Etc/GMT",
>+  "Etc/GMT+0",
>+  "Etc/GMT+1",
>+  "Etc/GMT+10",
>+  "Etc/GMT+11",
>+  "Etc/GMT+12",
>+  "Etc/GMT+2",
>+  "Etc/GMT+3",
>+  "Etc/GMT+4",
>+  "Etc/GMT+5",
>+  "Etc/GMT+6",
>+  "Etc/GMT+7",
>+  "Etc/GMT+8",
>+  "Etc/GMT+9",
>+  "Etc/GMT-0",
>+  "Etc/GMT-1",
>+  "Etc/GMT-10",
>+  "Etc/GMT-11",
>+  "Etc/GMT-12",
>+  "Etc/GMT-13",
>+  "Etc/GMT-14",
>+  "Etc/GMT-2",
>+  "Etc/GMT-3",
>+  "Etc/GMT-4",
>+  "Etc/GMT-5",
>+  "Etc/GMT-6",
>+  "Etc/GMT-7",
>+  "Etc/GMT-8",
>+  "Etc/GMT-9",
>+  "Etc/GMT0",
>+  "Etc/Greenwich",
>+  "Etc/UCT",
>+  "Etc/UTC",
>+  "Etc/Universal",
>+  "Etc/Zulu",
>+  "Europe/Amsterdam",
>+  "Europe/Andorra",
>+  "Europe/Astrakhan",
>+  "Europe/Athens",
>+  "Europe/Belfast",
>+  "Europe/Belgrade",
>+  "Europe/Berlin",
>+  "Europe/Bratislava",
>+  "Europe/Brussels",
>+  "Europe/Bucharest",
>+  "Europe/Budapest",
>+  "Europe/Busingen",
>+  "Europe/Chisinau",
>+  "Europe/Copenhagen",
>+  "Europe/Dublin",
>+  "Europe/Gibraltar",
>+  "Europe/Guernsey",
>+  "Europe/Helsinki",
>+  "Europe/Isle_of_Man",
>+  "Europe/Istanbul",
>+  "Europe/Jersey",
>+  "Europe/Kaliningrad",
>+  "Europe/Kiev",
>+  "Europe/Kirov",
>+  "Europe/Lisbon",
>+  "Europe/Ljubljana",
>+  "Europe/London",
>+  "Europe/Luxembourg",
>+  "Europe/Madrid",
>+  "Europe/Malta",
>+  "Europe/Mariehamn",
>+  "Europe/Minsk",
>+  "Europe/Monaco",
>+  "Europe/Moscow",
>+  "Europe/Nicosia",
>+  "Europe/Oslo",
>+  "Europe/Paris",
>+  "Europe/Podgorica",
>+  "Europe/Prague",
>+  "Europe/Riga",
>+  "Europe/Rome",
>+  "Europe/Samara",
>+  "Europe/San_Marino",
>+  "Europe/Sarajevo",
>+  "Europe/Saratov",
>+  "Europe/Simferopol",
>+  "Europe/Skopje",
>+  "Europe/Sofia",
>+  "Europe/Stockholm",
>+  "Europe/Tallinn",
>+  "Europe/Tirane",
>+  "Europe/Tiraspol",
>+  "Europe/Ulyanovsk",
>+  "Europe/Uzhgorod",
>+  "Europe/Vaduz",
>+  "Europe/Vatican",
>+  "Europe/Vienna",
>+  "Europe/Vilnius",
>+  "Europe/Volgograd",
>+  "Europe/Warsaw",
>+  "Europe/Zagreb",
>+  "Europe/Zaporozhye",
>+  "Europe/Zurich",
>+  "GB",
>+  "GB-Eire",
>+  "GMT",
>+  "GMT+0",
>+  "GMT-0",
>+  "GMT0",
>+  "Greenwich",
>+  "HST",
>+  "Hongkong",
>+  "Iceland",
>+  "Indian/Antananarivo",
>+  "Indian/Chagos",
>+  "Indian/Christmas",
>+  "Indian/Cocos",
>+  "Indian/Comoro",
>+  "Indian/Kerguelen",
>+  "Indian/Mahe",
>+  "Indian/Maldives",
>+  "Indian/Mauritius",
>+  "Indian/Mayotte",
>+  "Indian/Reunion",
>+  "Iran",
>+  "Israel",
>+  "Jamaica",
>+  "Japan",
>+  "Kwajalein",
>+  "Libya",
>+  "MET",
>+  "MST",
>+  "MST7MDT",
>+  "Mexico/BajaNorte",
>+  "Mexico/BajaSur",
>+  "Mexico/General",
>+  "NZ",
>+  "NZ-CHAT",
>+  "Navajo",
>+  "PRC",
>+  "PST8PDT",
>+  "Pacific/Apia",
>+  "Pacific/Auckland",
>+  "Pacific/Bougainville",
>+  "Pacific/Chatham",
>+  "Pacific/Chuuk",
>+  "Pacific/Easter",
>+  "Pacific/Efate",
>+  "Pacific/Enderbury",
>+  "Pacific/Fakaofo",
>+  "Pacific/Fiji",
>+  "Pacific/Funafuti",
>+  "Pacific/Galapagos",
>+  "Pacific/Gambier",
>+  "Pacific/Guadalcanal",
>+  "Pacific/Guam",
>+  "Pacific/Honolulu",
>+  "Pacific/Johnston",
>+  "Pacific/Kiritimati",
>+  "Pacific/Kosrae",
>+  "Pacific/Kwajalein",
>+  "Pacific/Majuro",
>+  "Pacific/Marquesas",
>+  "Pacific/Midway",
>+  "Pacific/Nauru",
>+  "Pacific/Niue",
>+  "Pacific/Norfolk",
>+  "Pacific/Noumea",
>+  "Pacific/Pago_Pago",
>+  "Pacific/Palau",
>+  "Pacific/Pitcairn",
>+  "Pacific/Pohnpei",
>+  "Pacific/Ponape",
>+  "Pacific/Port_Moresby",
>+  "Pacific/Rarotonga",
>+  "Pacific/Saipan",
>+  "Pacific/Samoa",
>+  "Pacific/Tahiti",
>+  "Pacific/Tarawa",
>+  "Pacific/Tongatapu",
>+  "Pacific/Truk",
>+  "Pacific/Wake",
>+  "Pacific/Wallis",
>+  "Pacific/Yap",
>+  "Poland",
>+  "Portugal",
>+  "ROC",
>+  "ROK",
>+  "Singapore",
>+  "Turkey",
>+  "UCT",
>+  "US/Alaska",
>+  "US/Aleutian",
>+  "US/Arizona",
>+  "US/Central",
>+  "US/East-Indiana",
>+  "US/Eastern",
>+  "US/Hawaii",
>+  "US/Indiana-Starke",
>+  "US/Michigan",
>+  "US/Mountain",
>+  "US/Pacific",
>+  "US/Samoa",
>+  "UTC",
>+  "Universal",
>+  "W-SU",
>+  "WET",
>+  "Zulu",
>+  nullptr
>+};
>+
>+// Helper to return a loaded time zone by value (UTC on error).
>+time_zone LoadZone(const std::string& name) {
>+  time_zone tz;
>+  load_time_zone(name, &tz);
>+  return tz;
>+}
>+
>+// This helper is a macro so that failed expectations show up with the
>+// correct line numbers.
>+#define ExpectTime(tp, tz, y, m, d, hh, mm, ss, off, isdst, zone) \
>+  do {                                                            \
>+    time_zone::absolute_lookup al = tz.lookup(tp);                \
>+    EXPECT_EQ(y, al.cs.year());                                   \
>+    EXPECT_EQ(m, al.cs.month());                                  \
>+    EXPECT_EQ(d, al.cs.day());                                    \
>+    EXPECT_EQ(hh, al.cs.hour());                                  \
>+    EXPECT_EQ(mm, al.cs.minute());                                \
>+    EXPECT_EQ(ss, al.cs.second());                                \
>+    EXPECT_EQ(off, al.offset);                                    \
>+    EXPECT_TRUE(isdst == al.is_dst);                              \
>+    /* EXPECT_STREQ(zone, al.abbr); */                            \
>+  } while (0)
>+
>+// These tests sometimes run on platforms that have zoneinfo data so old
>+// that the transition we are attempting to check does not exist, most
>+// notably Android emulators.  Fortunately, AndroidZoneInfoSource supports
>+// time_zone::version() so, in cases where we've learned that it matters,
>+// we can make the check conditionally.
>+int VersionCmp(time_zone tz, const std::string& target) {
>+  std::string version = tz.version();
>+  if (version.empty() && !target.empty()) return 1;  // unknown > known
>+  return version.compare(target);
>+}
>+
>+}  // namespace
>+
>+TEST(TimeZones, LoadZonesConcurrently) {
>+  std::promise<void> ready_promise;
>+  std::shared_future<void> ready_future(ready_promise.get_future());
>+  auto load_zones = [ready_future](std::promise<void>* started,
>+                                   std::set<std::string>* failures) {
>+    started->set_value();
>+    ready_future.wait();
>+    for (const char* const* np = kTimeZoneNames; *np != nullptr; ++np) {
>+      std::string zone = *np;
>+      time_zone tz;
>+      if (load_time_zone(zone, &tz)) {
>+        EXPECT_EQ(zone, tz.name());
>+      } else {
>+        failures->insert(zone);
>+      }
>+    }
>+  };
>+
>+  const std::size_t n_threads = 128;
>+  std::vector<std::thread> threads;
>+  std::vector<std::set<std::string>> thread_failures(n_threads);
>+  for (std::size_t i = 0; i != n_threads; ++i) {
>+    std::promise<void> started;
>+    threads.emplace_back(load_zones, &started, &thread_failures[i]);
>+    started.get_future().wait();
>+  }
>+  ready_promise.set_value();
>+  for (auto& thread : threads) {
>+    thread.join();
>+  }
>+
>+  // Allow a small number of failures to account for skew between
>+  // the contents of kTimeZoneNames and the zoneinfo data source.
>+#if defined(__ANDROID__)
>+  // Cater to the possibility of using an even older zoneinfo data
>+  // source when running on Android, where it is difficult to override
>+  // the bionic tzdata provided by the test environment.
>+  const std::size_t max_failures = 20;
>+#else
>+  const std::size_t max_failures = 3;
>+#endif
>+  std::set<std::string> failures;
>+  for (const auto& thread_failure : thread_failures) {
>+    failures.insert(thread_failure.begin(), thread_failure.end());
>+  }
>+  EXPECT_LE(failures.size(), max_failures) << testing::PrintToString(failures);
>+}
>+
>+TEST(TimeZone, NamedTimeZones) {
>+  const time_zone utc = utc_time_zone();
>+  EXPECT_EQ("UTC", utc.name());
>+  const time_zone nyc = LoadZone("America/New_York");
>+  EXPECT_EQ("America/New_York", nyc.name());
>+  const time_zone syd = LoadZone("Australia/Sydney");
>+  EXPECT_EQ("Australia/Sydney", syd.name());
>+  const time_zone fixed0 = fixed_time_zone(absl::time_internal::cctz::seconds::zero());
>+  EXPECT_EQ("UTC", fixed0.name());
>+  const time_zone fixed_pos = fixed_time_zone(
>+      chrono::hours(3) + chrono::minutes(25) + chrono::seconds(45));
>+  EXPECT_EQ("Fixed/UTC+03:25:45", fixed_pos.name());
>+  const time_zone fixed_neg = fixed_time_zone(
>+      -(chrono::hours(12) + chrono::minutes(34) + chrono::seconds(56)));
>+  EXPECT_EQ("Fixed/UTC-12:34:56", fixed_neg.name());
>+}
>+
>+TEST(TimeZone, Failures) {
>+  time_zone tz;
>+  EXPECT_FALSE(load_time_zone(":America/Los_Angeles", &tz));
>+
>+  tz = LoadZone("America/Los_Angeles");
>+  EXPECT_FALSE(load_time_zone("Invalid/TimeZone", &tz));
>+  EXPECT_EQ(chrono::system_clock::from_time_t(0),
>+            convert(civil_second(1970, 1, 1, 0, 0, 0), tz));  // UTC
>+
>+  // Ensures that the load still fails on a subsequent attempt.
>+  tz = LoadZone("America/Los_Angeles");
>+  EXPECT_FALSE(load_time_zone("Invalid/TimeZone", &tz));
>+  EXPECT_EQ(chrono::system_clock::from_time_t(0),
>+            convert(civil_second(1970, 1, 1, 0, 0, 0), tz));  // UTC
>+
>+  // Loading an empty std::string timezone should fail.
>+  tz = LoadZone("America/Los_Angeles");
>+  EXPECT_FALSE(load_time_zone("", &tz));
>+  EXPECT_EQ(chrono::system_clock::from_time_t(0),
>+            convert(civil_second(1970, 1, 1, 0, 0, 0), tz));  // UTC
>+}
>+
>+TEST(TimeZone, Equality) {
>+  const time_zone a;
>+  const time_zone b;
>+  EXPECT_EQ(a, b);
>+  EXPECT_EQ(a.name(), b.name());
>+
>+  const time_zone implicit_utc;
>+  const time_zone explicit_utc = utc_time_zone();
>+  EXPECT_EQ(implicit_utc, explicit_utc);
>+  EXPECT_EQ(implicit_utc.name(), explicit_utc.name());
>+
>+  const time_zone fixed_zero = fixed_time_zone(absl::time_internal::cctz::seconds::zero());
>+  EXPECT_EQ(fixed_zero, LoadZone(fixed_zero.name()));
>+  EXPECT_EQ(fixed_zero, explicit_utc);
>+
>+  const time_zone fixed_utc = LoadZone("Fixed/UTC+00:00:00");
>+  EXPECT_EQ(fixed_utc, LoadZone(fixed_utc.name()));
>+  EXPECT_EQ(fixed_utc, explicit_utc);
>+
>+  const time_zone fixed_pos = fixed_time_zone(
>+      chrono::hours(3) + chrono::minutes(25) + chrono::seconds(45));
>+  EXPECT_EQ(fixed_pos, LoadZone(fixed_pos.name()));
>+  EXPECT_NE(fixed_pos, explicit_utc);
>+  const time_zone fixed_neg = fixed_time_zone(
>+      -(chrono::hours(12) + chrono::minutes(34) + chrono::seconds(56)));
>+  EXPECT_EQ(fixed_neg, LoadZone(fixed_neg.name()));
>+  EXPECT_NE(fixed_neg, explicit_utc);
>+
>+  const time_zone fixed_lim = fixed_time_zone(chrono::hours(24));
>+  EXPECT_EQ(fixed_lim, LoadZone(fixed_lim.name()));
>+  EXPECT_NE(fixed_lim, explicit_utc);
>+  const time_zone fixed_ovfl =
>+      fixed_time_zone(chrono::hours(24) + chrono::seconds(1));
>+  EXPECT_EQ(fixed_ovfl, LoadZone(fixed_ovfl.name()));
>+  EXPECT_EQ(fixed_ovfl, explicit_utc);
>+
>+  EXPECT_EQ(fixed_time_zone(chrono::seconds(1)),
>+            fixed_time_zone(chrono::seconds(1)));
>+
>+  const time_zone local = local_time_zone();
>+  EXPECT_EQ(local, LoadZone(local.name()));
>+
>+  time_zone la = LoadZone("America/Los_Angeles");
>+  time_zone nyc = LoadZone("America/New_York");
>+  EXPECT_NE(la, nyc);
>+}
>+
>+TEST(StdChronoTimePoint, TimeTAlignment) {
>+  // Ensures that the Unix epoch and the system clock epoch are an integral
>+  // number of seconds apart. This simplifies conversions to/from time_t.
>+  auto diff = chrono::system_clock::time_point() -
>+              chrono::system_clock::from_time_t(0);
>+  EXPECT_EQ(chrono::system_clock::time_point::duration::zero(),
>+            diff % chrono::seconds(1));
>+}
>+
>+TEST(BreakTime, TimePointResolution) {
>+  const time_zone utc = utc_time_zone();
>+  const auto t0 = chrono::system_clock::from_time_t(0);
>+
>+  ExpectTime(chrono::time_point_cast<chrono::nanoseconds>(t0), utc,
>+             1970, 1, 1, 0, 0, 0, 0, false, "UTC");
>+  ExpectTime(chrono::time_point_cast<chrono::microseconds>(t0), utc,
>+             1970, 1, 1, 0, 0, 0, 0, false, "UTC");
>+  ExpectTime(chrono::time_point_cast<chrono::milliseconds>(t0), utc,
>+             1970, 1, 1, 0, 0, 0, 0, false, "UTC");
>+  ExpectTime(chrono::time_point_cast<chrono::seconds>(t0), utc,
>+             1970, 1, 1, 0, 0, 0, 0, false, "UTC");
>+  ExpectTime(chrono::time_point_cast<absl::time_internal::cctz::seconds>(t0), utc,
>+             1970, 1, 1, 0, 0, 0, 0, false, "UTC");
>+  ExpectTime(chrono::time_point_cast<chrono::minutes>(t0), utc,
>+             1970, 1, 1, 0, 0, 0, 0, false, "UTC");
>+  ExpectTime(chrono::time_point_cast<chrono::hours>(t0), utc,
>+             1970, 1, 1, 0, 0, 0, 0, false, "UTC");
>+}
>+
>+TEST(BreakTime, LocalTimeInUTC) {
>+  const time_zone tz = utc_time_zone();
>+  const auto tp = chrono::system_clock::from_time_t(0);
>+  ExpectTime(tp, tz, 1970, 1, 1, 0, 0, 0, 0, false, "UTC");
>+  EXPECT_EQ(weekday::thursday, get_weekday(civil_day(convert(tp, tz))));
>+}
>+
>+TEST(BreakTime, LocalTimeInUTCUnaligned) {
>+  const time_zone tz = utc_time_zone();
>+  const auto tp =
>+      chrono::system_clock::from_time_t(0) - chrono::milliseconds(500);
>+  ExpectTime(tp, tz, 1969, 12, 31, 23, 59, 59, 0, false, "UTC");
>+  EXPECT_EQ(weekday::wednesday, get_weekday(civil_day(convert(tp, tz))));
>+}
>+
>+TEST(BreakTime, LocalTimePosix) {
>+  // See IEEE Std 1003.1-1988 B.2.3 General Terms, Epoch.
>+  const time_zone tz = utc_time_zone();
>+  const auto tp = chrono::system_clock::from_time_t(536457599);
>+  ExpectTime(tp, tz, 1986, 12, 31, 23, 59, 59, 0, false, "UTC");
>+  EXPECT_EQ(weekday::wednesday, get_weekday(civil_day(convert(tp, tz))));
>+}
>+
>+TEST(TimeZoneImpl, LocalTimeInFixed) {
>+  const absl::time_internal::cctz::seconds offset =
>+      -(chrono::hours(8) + chrono::minutes(33) + chrono::seconds(47));
>+  const time_zone tz = fixed_time_zone(offset);
>+  const auto tp = chrono::system_clock::from_time_t(0);
>+  ExpectTime(tp, tz, 1969, 12, 31, 15, 26, 13, offset.count(), false,
>+             "-083347");
>+  EXPECT_EQ(weekday::wednesday, get_weekday(civil_day(convert(tp, tz))));
>+}
>+
>+TEST(BreakTime, LocalTimeInNewYork) {
>+  const time_zone tz = LoadZone("America/New_York");
>+  const auto tp = chrono::system_clock::from_time_t(45);
>+  ExpectTime(tp, tz, 1969, 12, 31, 19, 0, 45, -5 * 60 * 60, false, "EST");
>+  EXPECT_EQ(weekday::wednesday, get_weekday(civil_day(convert(tp, tz))));
>+}
>+
>+TEST(BreakTime, LocalTimeInMTV) {
>+  const time_zone tz = LoadZone("America/Los_Angeles");
>+  const auto tp = chrono::system_clock::from_time_t(1380855729);
>+  ExpectTime(tp, tz, 2013, 10, 3, 20, 2, 9, -7 * 60 * 60, true, "PDT");
>+  EXPECT_EQ(weekday::thursday, get_weekday(civil_day(convert(tp, tz))));
>+}
>+
>+TEST(BreakTime, LocalTimeInSydney) {
>+  const time_zone tz = LoadZone("Australia/Sydney");
>+  const auto tp = chrono::system_clock::from_time_t(90);
>+  ExpectTime(tp, tz, 1970, 1, 1, 10, 1, 30, 10 * 60 * 60, false, "AEST");
>+  EXPECT_EQ(weekday::thursday, get_weekday(civil_day(convert(tp, tz))));
>+}
>+
>+TEST(MakeTime, TimePointResolution) {
>+  const time_zone utc = utc_time_zone();
>+  const time_point<chrono::nanoseconds> tp_ns =
>+      convert(civil_second(2015, 1, 2, 3, 4, 5), utc);
>+  EXPECT_EQ("04:05", format("%M:%E*S", tp_ns, utc));
>+  const time_point<chrono::microseconds> tp_us =
>+      convert(civil_second(2015, 1, 2, 3, 4, 5), utc);
>+  EXPECT_EQ("04:05", format("%M:%E*S", tp_us, utc));
>+  const time_point<chrono::milliseconds> tp_ms =
>+      convert(civil_second(2015, 1, 2, 3, 4, 5), utc);
>+  EXPECT_EQ("04:05", format("%M:%E*S", tp_ms, utc));
>+  const time_point<chrono::seconds> tp_s =
>+      convert(civil_second(2015, 1, 2, 3, 4, 5), utc);
>+  EXPECT_EQ("04:05", format("%M:%E*S", tp_s, utc));
>+  const time_point<absl::time_internal::cctz::seconds> tp_s64 =
>+      convert(civil_second(2015, 1, 2, 3, 4, 5), utc);
>+  EXPECT_EQ("04:05", format("%M:%E*S", tp_s64, utc));
>+
>+  // These next two require chrono::time_point_cast because the conversion
>+  // from a resolution of seconds (the return value of convert()) to a
>+  // coarser resolution requires an explicit cast.
>+  const time_point<chrono::minutes> tp_m =
>+      chrono::time_point_cast<chrono::minutes>(
>+          convert(civil_second(2015, 1, 2, 3, 4, 5), utc));
>+  EXPECT_EQ("04:00", format("%M:%E*S", tp_m, utc));
>+  const time_point<chrono::hours> tp_h =
>+      chrono::time_point_cast<chrono::hours>(
>+          convert(civil_second(2015, 1, 2, 3, 4, 5), utc));
>+  EXPECT_EQ("00:00", format("%M:%E*S", tp_h, utc));
>+}
>+
>+TEST(MakeTime, Normalization) {
>+  const time_zone tz = LoadZone("America/New_York");
>+  const auto tp = convert(civil_second(2009, 2, 13, 18, 31, 30), tz);
>+  EXPECT_EQ(chrono::system_clock::from_time_t(1234567890), tp);
>+
>+  // Now requests for the same time_point but with out-of-range fields.
>+  EXPECT_EQ(tp, convert(civil_second(2008, 14, 13, 18, 31, 30), tz));  // month
>+  EXPECT_EQ(tp, convert(civil_second(2009, 1, 44, 18, 31, 30), tz));   // day
>+  EXPECT_EQ(tp, convert(civil_second(2009, 2, 12, 42, 31, 30), tz));   // hour
>+  EXPECT_EQ(tp, convert(civil_second(2009, 2, 13, 17, 91, 30), tz));   // minute
>+  EXPECT_EQ(tp, convert(civil_second(2009, 2, 13, 18, 30, 90), tz));   // second
>+}
>+
>+// NOTE: Run this with --copt=-ftrapv to detect overflow problems.
>+TEST(MakeTime, SysSecondsLimits) {
>+  const char RFC3339[] =  "%Y-%m-%dT%H:%M:%S%Ez";
>+  const time_zone utc = utc_time_zone();
>+  const time_zone east = fixed_time_zone(chrono::hours(14));
>+  const time_zone west = fixed_time_zone(-chrono::hours(14));
>+  time_point<absl::time_internal::cctz::seconds> tp;
>+
>+  // Approach the maximal time_point<cctz::seconds> value from below.
>+  tp = convert(civil_second(292277026596, 12, 4, 15, 30, 6), utc);
>+  EXPECT_EQ("292277026596-12-04T15:30:06+00:00", format(RFC3339, tp, utc));
>+  tp = convert(civil_second(292277026596, 12, 4, 15, 30, 7), utc);
>+  EXPECT_EQ("292277026596-12-04T15:30:07+00:00", format(RFC3339, tp, utc));
>+  EXPECT_EQ(time_point<absl::time_internal::cctz::seconds>::max(), tp);
>+  tp = convert(civil_second(292277026596, 12, 4, 15, 30, 8), utc);
>+  EXPECT_EQ(time_point<absl::time_internal::cctz::seconds>::max(), tp);
>+  tp = convert(civil_second::max(), utc);
>+  EXPECT_EQ(time_point<absl::time_internal::cctz::seconds>::max(), tp);
>+
>+  // Checks that we can also get the maximal value for a far-east zone.
>+  tp = convert(civil_second(292277026596, 12, 5, 5, 30, 7), east);
>+  EXPECT_EQ("292277026596-12-05T05:30:07+14:00", format(RFC3339, tp, east));
>+  EXPECT_EQ(time_point<absl::time_internal::cctz::seconds>::max(), tp);
>+  tp = convert(civil_second(292277026596, 12, 5, 5, 30, 8), east);
>+  EXPECT_EQ(time_point<absl::time_internal::cctz::seconds>::max(), tp);
>+  tp = convert(civil_second::max(), east);
>+  EXPECT_EQ(time_point<absl::time_internal::cctz::seconds>::max(), tp);
>+
>+  // Checks that we can also get the maximal value for a far-west zone.
>+  tp = convert(civil_second(292277026596, 12, 4, 1, 30, 7), west);
>+  EXPECT_EQ("292277026596-12-04T01:30:07-14:00", format(RFC3339, tp, west));
>+  EXPECT_EQ(time_point<absl::time_internal::cctz::seconds>::max(), tp);
>+  tp = convert(civil_second(292277026596, 12, 4, 7, 30, 8), west);
>+  EXPECT_EQ(time_point<absl::time_internal::cctz::seconds>::max(), tp);
>+  tp = convert(civil_second::max(), west);
>+  EXPECT_EQ(time_point<absl::time_internal::cctz::seconds>::max(), tp);
>+
>+  // Approach the minimal time_point<cctz::seconds> value from above.
>+  tp = convert(civil_second(-292277022657, 1, 27, 8, 29, 53), utc);
>+  EXPECT_EQ("-292277022657-01-27T08:29:53+00:00", format(RFC3339, tp, utc));
>+  tp = convert(civil_second(-292277022657, 1, 27, 8, 29, 52), utc);
>+  EXPECT_EQ("-292277022657-01-27T08:29:52+00:00", format(RFC3339, tp, utc));
>+  EXPECT_EQ(time_point<absl::time_internal::cctz::seconds>::min(), tp);
>+  tp = convert(civil_second(-292277022657, 1, 27, 8, 29, 51), utc);
>+  EXPECT_EQ(time_point<absl::time_internal::cctz::seconds>::min(), tp);
>+  tp = convert(civil_second::min(), utc);
>+  EXPECT_EQ(time_point<absl::time_internal::cctz::seconds>::min(), tp);
>+
>+  // Checks that we can also get the minimal value for a far-east zone.
>+  tp = convert(civil_second(-292277022657, 1, 27, 22, 29, 52), east);
>+  EXPECT_EQ("-292277022657-01-27T22:29:52+14:00", format(RFC3339, tp, east));
>+  EXPECT_EQ(time_point<absl::time_internal::cctz::seconds>::min(), tp);
>+  tp = convert(civil_second(-292277022657, 1, 27, 22, 29, 51), east);
>+  EXPECT_EQ(time_point<absl::time_internal::cctz::seconds>::min(), tp);
>+  tp = convert(civil_second::min(), east);
>+  EXPECT_EQ(time_point<absl::time_internal::cctz::seconds>::min(), tp);
>+
>+  // Checks that we can also get the minimal value for a far-west zone.
>+  tp = convert(civil_second(-292277022657, 1, 26, 18, 29, 52), west);
>+  EXPECT_EQ("-292277022657-01-26T18:29:52-14:00", format(RFC3339, tp, west));
>+  EXPECT_EQ(time_point<absl::time_internal::cctz::seconds>::min(), tp);
>+  tp = convert(civil_second(-292277022657, 1, 26, 18, 29, 51), west);
>+  EXPECT_EQ(time_point<absl::time_internal::cctz::seconds>::min(), tp);
>+  tp = convert(civil_second::min(), west);
>+  EXPECT_EQ(time_point<absl::time_internal::cctz::seconds>::min(), tp);
>+}
>+
>+TEST(NextTransition, UTC) {
>+  const auto tz = utc_time_zone();
>+  time_zone::civil_transition trans;
>+
>+  auto tp = time_point<absl::time_internal::cctz::seconds>::min();
>+  EXPECT_FALSE(tz.next_transition(tp, &trans));
>+
>+  tp = time_point<absl::time_internal::cctz::seconds>::max();
>+  EXPECT_FALSE(tz.next_transition(tp, &trans));
>+}
>+
>+TEST(PrevTransition, UTC) {
>+  const auto tz = utc_time_zone();
>+  time_zone::civil_transition trans;
>+
>+  auto tp = time_point<absl::time_internal::cctz::seconds>::max();
>+  EXPECT_FALSE(tz.prev_transition(tp, &trans));
>+
>+  tp = time_point<absl::time_internal::cctz::seconds>::min();
>+  EXPECT_FALSE(tz.prev_transition(tp, &trans));
>+}
>+
>+TEST(NextTransition, AmericaNewYork) {
>+  const auto tz = LoadZone("America/New_York");
>+  time_zone::civil_transition trans;
>+
>+  auto tp = convert(civil_second(2018, 6, 30, 0, 0, 0), tz);
>+  EXPECT_TRUE(tz.next_transition(tp, &trans));
>+  EXPECT_EQ(civil_second(2018, 11, 4, 2, 0, 0), trans.from);
>+  EXPECT_EQ(civil_second(2018, 11, 4, 1, 0, 0), trans.to);
>+
>+  tp = time_point<absl::time_internal::cctz::seconds>::max();
>+  EXPECT_FALSE(tz.next_transition(tp, &trans));
>+
>+  tp = time_point<absl::time_internal::cctz::seconds>::min();
>+  EXPECT_TRUE(tz.next_transition(tp, &trans));
>+  if (trans.from == civil_second(1918, 3, 31, 2, 0, 0)) {
>+    // It looks like the tzdata is only 32 bit (probably macOS),
>+    // which bottoms out at 1901-12-13T20:45:52+00:00.
>+    EXPECT_EQ(civil_second(1918, 3, 31, 3, 0, 0), trans.to);
>+  } else {
>+    EXPECT_EQ(civil_second(1883, 11, 18, 12, 3, 58), trans.from);
>+    EXPECT_EQ(civil_second(1883, 11, 18, 12, 0, 0), trans.to);
>+  }
>+}
>+
>+TEST(PrevTransition, AmericaNewYork) {
>+  const auto tz = LoadZone("America/New_York");
>+  time_zone::civil_transition trans;
>+
>+  auto tp = convert(civil_second(2018, 6, 30, 0, 0, 0), tz);
>+  EXPECT_TRUE(tz.prev_transition(tp, &trans));
>+  EXPECT_EQ(civil_second(2018, 3, 11, 2, 0, 0), trans.from);
>+  EXPECT_EQ(civil_second(2018, 3, 11, 3, 0, 0), trans.to);
>+
>+  tp = time_point<absl::time_internal::cctz::seconds>::min();
>+  EXPECT_FALSE(tz.prev_transition(tp, &trans));
>+
>+  tp = time_point<absl::time_internal::cctz::seconds>::max();
>+  EXPECT_TRUE(tz.prev_transition(tp, &trans));
>+  // We have a transition but we don't know which one.
>+}
>+
>+TEST(TimeZoneEdgeCase, AmericaNewYork) {
>+  const time_zone tz = LoadZone("America/New_York");
>+
>+  // Spring 1:59:59 -> 3:00:00
>+  auto tp = convert(civil_second(2013, 3, 10, 1, 59, 59), tz);
>+  ExpectTime(tp, tz, 2013, 3, 10, 1, 59, 59, -5 * 3600, false, "EST");
>+  tp += absl::time_internal::cctz::seconds(1);
>+  ExpectTime(tp, tz, 2013, 3, 10, 3, 0, 0, -4 * 3600, true, "EDT");
>+
>+  // Fall 1:59:59 -> 1:00:00
>+  tp = convert(civil_second(2013, 11, 3, 1, 59, 59), tz);
>+  ExpectTime(tp, tz, 2013, 11, 3, 1, 59, 59, -4 * 3600, true, "EDT");
>+  tp += absl::time_internal::cctz::seconds(1);
>+  ExpectTime(tp, tz, 2013, 11, 3, 1, 0, 0, -5 * 3600, false, "EST");
>+}
>+
>+TEST(TimeZoneEdgeCase, AmericaLosAngeles) {
>+  const time_zone tz = LoadZone("America/Los_Angeles");
>+
>+  // Spring 1:59:59 -> 3:00:00
>+  auto tp = convert(civil_second(2013, 3, 10, 1, 59, 59), tz);
>+  ExpectTime(tp, tz, 2013, 3, 10, 1, 59, 59, -8 * 3600, false, "PST");
>+  tp += absl::time_internal::cctz::seconds(1);
>+  ExpectTime(tp, tz, 2013, 3, 10, 3, 0, 0, -7 * 3600, true, "PDT");
>+
>+  // Fall 1:59:59 -> 1:00:00
>+  tp = convert(civil_second(2013, 11, 3, 1, 59, 59), tz);
>+  ExpectTime(tp, tz, 2013, 11, 3, 1, 59, 59, -7 * 3600, true, "PDT");
>+  tp += absl::time_internal::cctz::seconds(1);
>+  ExpectTime(tp, tz, 2013, 11, 3, 1, 0, 0, -8 * 3600, false, "PST");
>+}
>+
>+TEST(TimeZoneEdgeCase, ArizonaNoTransition) {
>+  const time_zone tz = LoadZone("America/Phoenix");
>+
>+  // No transition in Spring.
>+  auto tp = convert(civil_second(2013, 3, 10, 1, 59, 59), tz);
>+  ExpectTime(tp, tz, 2013, 3, 10, 1, 59, 59, -7 * 3600, false, "MST");
>+  tp += absl::time_internal::cctz::seconds(1);
>+  ExpectTime(tp, tz, 2013, 3, 10, 2, 0, 0, -7 * 3600, false, "MST");
>+
>+  // No transition in Fall.
>+  tp = convert(civil_second(2013, 11, 3, 1, 59, 59), tz);
>+  ExpectTime(tp, tz, 2013, 11, 3, 1, 59, 59, -7 * 3600, false, "MST");
>+  tp += absl::time_internal::cctz::seconds(1);
>+  ExpectTime(tp, tz, 2013, 11, 3, 2, 0, 0, -7 * 3600, false, "MST");
>+}
>+
>+TEST(TimeZoneEdgeCase, AsiaKathmandu) {
>+  const time_zone tz = LoadZone("Asia/Kathmandu");
>+
>+  // A non-DST offset change from +0530 to +0545
>+  //
>+  //   504901799 == Tue, 31 Dec 1985 23:59:59 +0530 (+0530)
>+  //   504901800 == Wed,  1 Jan 1986 00:15:00 +0545 (+0545)
>+  auto tp = convert(civil_second(1985, 12, 31, 23, 59, 59), tz);
>+  ExpectTime(tp, tz, 1985, 12, 31, 23, 59, 59, 5.5 * 3600, false, "+0530");
>+  tp += absl::time_internal::cctz::seconds(1);
>+  ExpectTime(tp, tz, 1986, 1, 1, 0, 15, 0, 5.75 * 3600, false, "+0545");
>+}
>+
>+TEST(TimeZoneEdgeCase, PacificChatham) {
>+  const time_zone tz = LoadZone("Pacific/Chatham");
>+
>+  // One-hour DST offset changes, but at atypical values
>+  //
>+  //   1365256799 == Sun,  7 Apr 2013 03:44:59 +1345 (+1345)
>+  //   1365256800 == Sun,  7 Apr 2013 02:45:00 +1245 (+1245)
>+  auto tp = convert(civil_second(2013, 4, 7, 3, 44, 59), tz);
>+  ExpectTime(tp, tz, 2013, 4, 7, 3, 44, 59, 13.75 * 3600, true, "+1345");
>+  tp += absl::time_internal::cctz::seconds(1);
>+  ExpectTime(tp, tz, 2013, 4, 7, 2, 45, 0, 12.75 * 3600, false, "+1245");
>+
>+  //   1380376799 == Sun, 29 Sep 2013 02:44:59 +1245 (+1245)
>+  //   1380376800 == Sun, 29 Sep 2013 03:45:00 +1345 (+1345)
>+  tp = convert(civil_second(2013, 9, 29, 2, 44, 59), tz);
>+  ExpectTime(tp, tz, 2013, 9, 29, 2, 44, 59, 12.75 * 3600, false, "+1245");
>+  tp += absl::time_internal::cctz::seconds(1);
>+  ExpectTime(tp, tz, 2013, 9, 29, 3, 45, 0, 13.75 * 3600, true, "+1345");
>+}
>+
>+TEST(TimeZoneEdgeCase, AustraliaLordHowe) {
>+  const time_zone tz = LoadZone("Australia/Lord_Howe");
>+
>+  // Half-hour DST offset changes
>+  //
>+  //   1365260399 == Sun,  7 Apr 2013 01:59:59 +1100 (+11)
>+  //   1365260400 == Sun,  7 Apr 2013 01:30:00 +1030 (+1030)
>+  auto tp = convert(civil_second(2013, 4, 7, 1, 59, 59), tz);
>+  ExpectTime(tp, tz, 2013, 4, 7, 1, 59, 59, 11 * 3600, true, "+11");
>+  tp += absl::time_internal::cctz::seconds(1);
>+  ExpectTime(tp, tz, 2013, 4, 7, 1, 30, 0, 10.5 * 3600, false, "+1030");
>+
>+  //   1380986999 == Sun,  6 Oct 2013 01:59:59 +1030 (+1030)
>+  //   1380987000 == Sun,  6 Oct 2013 02:30:00 +1100 (+11)
>+  tp = convert(civil_second(2013, 10, 6, 1, 59, 59), tz);
>+  ExpectTime(tp, tz, 2013, 10, 6, 1, 59, 59, 10.5 * 3600, false, "+1030");
>+  tp += absl::time_internal::cctz::seconds(1);
>+  ExpectTime(tp, tz, 2013, 10, 6, 2, 30, 0, 11 * 3600, true, "+11");
>+}
>+
>+TEST(TimeZoneEdgeCase, PacificApia) {
>+  const time_zone tz = LoadZone("Pacific/Apia");
>+
>+  // At the end of December 2011, Samoa jumped forward by one day,
>+  // skipping 30 December from the local calendar, when the nation
>+  // moved to the west of the International Date Line.
>+  //
>+  // A one-day, non-DST offset change
>+  //
>+  //   1325239199 == Thu, 29 Dec 2011 23:59:59 -1000 (-10)
>+  //   1325239200 == Sat, 31 Dec 2011 00:00:00 +1400 (+14)
>+  auto tp = convert(civil_second(2011, 12, 29, 23, 59, 59), tz);
>+  ExpectTime(tp, tz, 2011, 12, 29, 23, 59, 59, -10 * 3600, true, "-10");
>+  EXPECT_EQ(363, get_yearday(civil_day(convert(tp, tz))));
>+  tp += absl::time_internal::cctz::seconds(1);
>+  ExpectTime(tp, tz, 2011, 12, 31, 0, 0, 0, 14 * 3600, true, "+14");
>+  EXPECT_EQ(365, get_yearday(civil_day(convert(tp, tz))));
>+}
>+
>+TEST(TimeZoneEdgeCase, AfricaCairo) {
>+  const time_zone tz = LoadZone("Africa/Cairo");
>+
>+  if (VersionCmp(tz, "2014c") >= 0) {
>+    // An interesting case of midnight not existing.
>+    //
>+    //   1400191199 == Thu, 15 May 2014 23:59:59 +0200 (EET)
>+    //   1400191200 == Fri, 16 May 2014 01:00:00 +0300 (EEST)
>+    auto tp = convert(civil_second(2014, 5, 15, 23, 59, 59), tz);
>+    ExpectTime(tp, tz, 2014, 5, 15, 23, 59, 59, 2 * 3600, false, "EET");
>+    tp += absl::time_internal::cctz::seconds(1);
>+    ExpectTime(tp, tz, 2014, 5, 16, 1, 0, 0, 3 * 3600, true, "EEST");
>+  }
>+}
>+
>+TEST(TimeZoneEdgeCase, AfricaMonrovia) {
>+  const time_zone tz = LoadZone("Africa/Monrovia");
>+
>+  if (VersionCmp(tz, "2017b") >= 0) {
>+    // Strange offset change -00:44:30 -> +00:00:00 (non-DST)
>+    //
>+    //   63593069 == Thu,  6 Jan 1972 23:59:59 -0044 (MMT)
>+    //   63593070 == Fri,  7 Jan 1972 00:44:30 +0000 (GMT)
>+    auto tp = convert(civil_second(1972, 1, 6, 23, 59, 59), tz);
>+    ExpectTime(tp, tz, 1972, 1, 6, 23, 59, 59, -44.5 * 60, false, "MMT");
>+    tp += absl::time_internal::cctz::seconds(1);
>+    ExpectTime(tp, tz, 1972, 1, 7, 0, 44, 30, 0 * 60, false, "GMT");
>+  }
>+}
>+
>+TEST(TimeZoneEdgeCase, AmericaJamaica) {
>+  // Jamaica discontinued DST transitions in 1983, and is now at a
>+  // constant -0500.  This makes it an interesting edge-case target.
>+  // Note that the 32-bit times used in a (tzh_version == 0) zoneinfo
>+  // file cannot represent the abbreviation-only transition of 1890,
>+  // so we ignore the abbreviation by expecting what we received.
>+  const time_zone tz = LoadZone("America/Jamaica");
>+
>+  // Before the first transition.
>+  if (!tz.version().empty() && VersionCmp(tz, "2018d") >= 0) {
>+    // We avoid the expectations on the -18430 offset below unless we are
>+    // certain we have commit 907241e (Fix off-by-1 error for Jamaica and
>+    // T&C before 1913) from 2018d.  TODO: Remove the "version() not empty"
>+    // part when 2018d is generally available from /usr/share/zoneinfo.
>+    auto tp = convert(civil_second(1889, 12, 31, 0, 0, 0), tz);
>+    ExpectTime(tp, tz, 1889, 12, 31, 0, 0, 0, -18430, false,
>+               tz.lookup(tp).abbr);
>+
>+    // Over the first (abbreviation-change only) transition.
>+    //   -2524503170 == Tue, 31 Dec 1889 23:59:59 -0507 (LMT)
>+    //   -2524503169 == Wed,  1 Jan 1890 00:00:00 -0507 (KMT)
>+    tp = convert(civil_second(1889, 12, 31, 23, 59, 59), tz);
>+    ExpectTime(tp, tz, 1889, 12, 31, 23, 59, 59, -18430, false,
>+               tz.lookup(tp).abbr);
>+    tp += absl::time_internal::cctz::seconds(1);
>+    ExpectTime(tp, tz, 1890, 1, 1, 0, 0, 0, -18430, false, "KMT");
>+  }
>+
>+  // Over the last (DST) transition.
>+  //     436341599 == Sun, 30 Oct 1983 01:59:59 -0400 (EDT)
>+  //     436341600 == Sun, 30 Oct 1983 01:00:00 -0500 (EST)
>+  auto tp = convert(civil_second(1983, 10, 30, 1, 59, 59), tz);
>+  ExpectTime(tp, tz, 1983, 10, 30, 1, 59, 59, -4 * 3600, true, "EDT");
>+  tp += absl::time_internal::cctz::seconds(1);
>+  ExpectTime(tp, tz, 1983, 10, 30, 1, 0, 0, -5 * 3600, false, "EST");
>+
>+  // After the last transition.
>+  tp = convert(civil_second(1983, 12, 31, 23, 59, 59), tz);
>+  ExpectTime(tp, tz, 1983, 12, 31, 23, 59, 59, -5 * 3600, false, "EST");
>+}
>+
>+TEST(TimeZoneEdgeCase, WET) {
>+  // Cover some non-existent times within forward transitions.
>+  const time_zone tz = LoadZone("WET");
>+
>+  // Before the first transition.
>+  auto tp = convert(civil_second(1977, 1, 1, 0, 0, 0), tz);
>+  ExpectTime(tp, tz, 1977, 1, 1, 0, 0, 0, 0, false, "WET");
>+
>+  // Over the first transition.
>+  //     228877199 == Sun,  3 Apr 1977 00:59:59 +0000 (WET)
>+  //     228877200 == Sun,  3 Apr 1977 02:00:00 +0100 (WEST)
>+  tp = convert(civil_second(1977, 4, 3, 0, 59, 59), tz);
>+  ExpectTime(tp, tz, 1977, 4, 3, 0, 59, 59, 0, false, "WET");
>+  tp += absl::time_internal::cctz::seconds(1);
>+  ExpectTime(tp, tz, 1977, 4, 3, 2, 0, 0, 1 * 3600, true, "WEST");
>+
>+  // A non-existent time within the first transition.
>+  time_zone::civil_lookup cl1 = tz.lookup(civil_second(1977, 4, 3, 1, 15, 0));
>+  EXPECT_EQ(time_zone::civil_lookup::SKIPPED, cl1.kind);
>+  ExpectTime(cl1.pre, tz, 1977, 4, 3, 2, 15, 0, 1 * 3600, true, "WEST");
>+  ExpectTime(cl1.trans, tz, 1977, 4, 3, 2, 0, 0, 1 * 3600, true, "WEST");
>+  ExpectTime(cl1.post, tz, 1977, 4, 3, 0, 15, 0, 0 * 3600, false, "WET");
>+
>+  // A non-existent time within the second forward transition.
>+  time_zone::civil_lookup cl2 = tz.lookup(civil_second(1978, 4, 2, 1, 15, 0));
>+  EXPECT_EQ(time_zone::civil_lookup::SKIPPED, cl2.kind);
>+  ExpectTime(cl2.pre, tz, 1978, 4, 2, 2, 15, 0, 1 * 3600, true, "WEST");
>+  ExpectTime(cl2.trans, tz, 1978, 4, 2, 2, 0, 0, 1 * 3600, true, "WEST");
>+  ExpectTime(cl2.post, tz, 1978, 4, 2, 0, 15, 0, 0 * 3600, false, "WET");
>+}
>+
>+TEST(TimeZoneEdgeCase, FixedOffsets) {
>+  const time_zone gmtm5 = LoadZone("Etc/GMT+5");  // -0500
>+  auto tp = convert(civil_second(1970, 1, 1, 0, 0, 0), gmtm5);
>+  ExpectTime(tp, gmtm5, 1970, 1, 1, 0, 0, 0, -5 * 3600, false, "-05");
>+  EXPECT_EQ(chrono::system_clock::from_time_t(5 * 3600), tp);
>+
>+  const time_zone gmtp5 = LoadZone("Etc/GMT-5");  // +0500
>+  tp = convert(civil_second(1970, 1, 1, 0, 0, 0), gmtp5);
>+  ExpectTime(tp, gmtp5, 1970, 1, 1, 0, 0, 0, 5 * 3600, false, "+05");
>+  EXPECT_EQ(chrono::system_clock::from_time_t(-5 * 3600), tp);
>+}
>+
>+TEST(TimeZoneEdgeCase, NegativeYear) {
>+  // Tests transition from year 0 (aka 1BCE) to year -1.
>+  const time_zone tz = utc_time_zone();
>+  auto tp = convert(civil_second(0, 1, 1, 0, 0, 0), tz);
>+  ExpectTime(tp, tz, 0, 1, 1, 0, 0, 0, 0 * 3600, false, "UTC");
>+  EXPECT_EQ(weekday::saturday, get_weekday(civil_day(convert(tp, tz))));
>+  tp -= absl::time_internal::cctz::seconds(1);
>+  ExpectTime(tp, tz, -1, 12, 31, 23, 59, 59, 0 * 3600, false, "UTC");
>+  EXPECT_EQ(weekday::friday, get_weekday(civil_day(convert(tp, tz))));
>+}
>+
>+TEST(TimeZoneEdgeCase, UTC32bitLimit) {
>+  const time_zone tz = utc_time_zone();
>+
>+  // Limits of signed 32-bit time_t
>+  //
>+  //   2147483647 == Tue, 19 Jan 2038 03:14:07 +0000 (UTC)
>+  //   2147483648 == Tue, 19 Jan 2038 03:14:08 +0000 (UTC)
>+  auto tp = convert(civil_second(2038, 1, 19, 3, 14, 7), tz);
>+  ExpectTime(tp, tz, 2038, 1, 19, 3, 14, 7, 0 * 3600, false, "UTC");
>+  tp += absl::time_internal::cctz::seconds(1);
>+  ExpectTime(tp, tz, 2038, 1, 19, 3, 14, 8, 0 * 3600, false, "UTC");
>+}
>+
>+TEST(TimeZoneEdgeCase, UTC5DigitYear) {
>+  const time_zone tz = utc_time_zone();
>+
>+  // Rollover to 5-digit year
>+  //
>+  //   253402300799 == Fri, 31 Dec 9999 23:59:59 +0000 (UTC)
>+  //   253402300800 == Sat,  1 Jan 1000 00:00:00 +0000 (UTC)
>+  auto tp = convert(civil_second(9999, 12, 31, 23, 59, 59), tz);
>+  ExpectTime(tp, tz, 9999, 12, 31, 23, 59, 59, 0 * 3600, false, "UTC");
>+  tp += absl::time_internal::cctz::seconds(1);
>+  ExpectTime(tp, tz, 10000, 1, 1, 0, 0, 0, 0 * 3600, false, "UTC");
>+}
>+
>+}  // namespace cctz
>+}  // namespace time_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_posix.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_posix.cc
>new file mode 100644
>index 00000000000..75ad8bcba93
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_posix.cc
>@@ -0,0 +1,155 @@
>+// Copyright 2016 Google Inc. All Rights Reserved.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//   http://www.apache.org/licenses/LICENSE-2.0
>+//
>+//   Unless required by applicable law or agreed to in writing, software
>+//   distributed under the License is distributed on an "AS IS" BASIS,
>+//   WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+//   See the License for the specific language governing permissions and
>+//   limitations under the License.
>+
>+#include "time_zone_posix.h"
>+
>+#include <cstddef>
>+#include <cstring>
>+#include <limits>
>+#include <string>
>+
>+namespace absl {
>+namespace time_internal {
>+namespace cctz {
>+
>+namespace {
>+
>+const char kDigits[] = "0123456789";
>+
>+const char* ParseInt(const char* p, int min, int max, int* vp) {
>+  int value = 0;
>+  const char* op = p;
>+  const int kMaxInt = std::numeric_limits<int>::max();
>+  for (; const char* dp = strchr(kDigits, *p); ++p) {
>+    int d = static_cast<int>(dp - kDigits);
>+    if (d >= 10) break;  // '\0'
>+    if (value > kMaxInt / 10) return nullptr;
>+    value *= 10;
>+    if (value > kMaxInt - d) return nullptr;
>+    value += d;
>+  }
>+  if (p == op || value < min || value > max) return nullptr;
>+  *vp = value;
>+  return p;
>+}
>+
>+// abbr = <.*?> | [^-+,\d]{3,}
>+const char* ParseAbbr(const char* p, std::string* abbr) {
>+  const char* op = p;
>+  if (*p == '<') {  // special zoneinfo <...> form
>+    while (*++p != '>') {
>+      if (*p == '\0') return nullptr;
>+    }
>+    abbr->assign(op + 1, static_cast<std::size_t>(p - op) - 1);
>+    return ++p;
>+  }
>+  while (*p != '\0') {
>+    if (strchr("-+,", *p)) break;
>+    if (strchr(kDigits, *p)) break;
>+    ++p;
>+  }
>+  if (p - op < 3) return nullptr;
>+  abbr->assign(op, static_cast<std::size_t>(p - op));
>+  return p;
>+}
>+
>+// offset = [+|-]hh[:mm[:ss]] (aggregated into single seconds value)
>+const char* ParseOffset(const char* p, int min_hour, int max_hour, int sign,
>+                        std::int_fast32_t* offset) {
>+  if (p == nullptr) return nullptr;
>+  if (*p == '+' || *p == '-') {
>+    if (*p++ == '-') sign = -sign;
>+  }
>+  int hours = 0;
>+  int minutes = 0;
>+  int seconds = 0;
>+
>+  p = ParseInt(p, min_hour, max_hour, &hours);
>+  if (p == nullptr) return nullptr;
>+  if (*p == ':') {
>+    p = ParseInt(p + 1, 0, 59, &minutes);
>+    if (p == nullptr) return nullptr;
>+    if (*p == ':') {
>+      p = ParseInt(p + 1, 0, 59, &seconds);
>+      if (p == nullptr) return nullptr;
>+    }
>+  }
>+  *offset = sign * ((((hours * 60) + minutes) * 60) + seconds);
>+  return p;
>+}
>+
>+// datetime = ( Jn | n | Mm.w.d ) [ / offset ]
>+const char* ParseDateTime(const char* p, PosixTransition* res) {
>+  if (p != nullptr && *p == ',') {
>+    if (*++p == 'M') {
>+      int month = 0;
>+      if ((p = ParseInt(p + 1, 1, 12, &month)) != nullptr && *p == '.') {
>+        int week = 0;
>+        if ((p = ParseInt(p + 1, 1, 5, &week)) != nullptr && *p == '.') {
>+          int weekday = 0;
>+          if ((p = ParseInt(p + 1, 0, 6, &weekday)) != nullptr) {
>+            res->date.fmt = PosixTransition::M;
>+            res->date.m.month = static_cast<int_fast8_t>(month);
>+            res->date.m.week = static_cast<int_fast8_t>(week);
>+            res->date.m.weekday = static_cast<int_fast8_t>(weekday);
>+          }
>+        }
>+      }
>+    } else if (*p == 'J') {
>+      int day = 0;
>+      if ((p = ParseInt(p + 1, 1, 365, &day)) != nullptr) {
>+        res->date.fmt = PosixTransition::J;
>+        res->date.j.day = static_cast<int_fast16_t>(day);
>+      }
>+    } else {
>+      int day = 0;
>+      if ((p = ParseInt(p, 0, 365, &day)) != nullptr) {
>+        res->date.fmt = PosixTransition::N;
>+        res->date.j.day = static_cast<int_fast16_t>(day);
>+      }
>+    }
>+  }
>+  if (p != nullptr) {
>+    res->time.offset = 2 * 60 * 60;  // default offset is 02:00:00
>+    if (*p == '/') p = ParseOffset(p + 1, -167, 167, 1, &res->time.offset);
>+  }
>+  return p;
>+}
>+
>+}  // namespace
>+
>+// spec = std offset [ dst [ offset ] , datetime , datetime ]
>+bool ParsePosixSpec(const std::string& spec, PosixTimeZone* res) {
>+  const char* p = spec.c_str();
>+  if (*p == ':') return false;
>+
>+  p = ParseAbbr(p, &res->std_abbr);
>+  p = ParseOffset(p, 0, 24, -1, &res->std_offset);
>+  if (p == nullptr) return false;
>+  if (*p == '\0') return true;
>+
>+  p = ParseAbbr(p, &res->dst_abbr);
>+  if (p == nullptr) return false;
>+  res->dst_offset = res->std_offset + (60 * 60);  // default
>+  if (*p != ',') p = ParseOffset(p, 0, 24, -1, &res->dst_offset);
>+
>+  p = ParseDateTime(p, &res->dst_start);
>+  p = ParseDateTime(p, &res->dst_end);
>+
>+  return p != nullptr && *p == '\0';
>+}
>+
>+}  // namespace cctz
>+}  // namespace time_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_posix.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_posix.h
>new file mode 100644
>index 00000000000..6619f27edcf
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/time_zone_posix.h
>@@ -0,0 +1,118 @@
>+// Copyright 2016 Google Inc. All Rights Reserved.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//   http://www.apache.org/licenses/LICENSE-2.0
>+//
>+//   Unless required by applicable law or agreed to in writing, software
>+//   distributed under the License is distributed on an "AS IS" BASIS,
>+//   WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+//   See the License for the specific language governing permissions and
>+//   limitations under the License.
>+
>+// Parsing of a POSIX zone spec as described in the TZ part of section 8.3 in
>+// http://pubs.opengroup.org/onlinepubs/009695399/basedefs/xbd_chap08.html.
>+//
>+// The current POSIX spec for America/Los_Angeles is "PST8PDT,M3.2.0,M11.1.0",
>+// which would be broken down as ...
>+//
>+//   PosixTimeZone {
>+//     std_abbr = "PST"
>+//     std_offset = -28800
>+//     dst_abbr = "PDT"
>+//     dst_offset = -25200
>+//     dst_start = PosixTransition {
>+//       date {
>+//         m {
>+//           month = 3
>+//           week = 2
>+//           weekday = 0
>+//         }
>+//       }
>+//       time {
>+//         offset = 7200
>+//       }
>+//     }
>+//     dst_end = PosixTransition {
>+//       date {
>+//         m {
>+//           month = 11
>+//           week = 1
>+//           weekday = 0
>+//         }
>+//       }
>+//       time {
>+//         offset = 7200
>+//       }
>+//     }
>+//   }
>+
>+#ifndef ABSL_TIME_INTERNAL_CCTZ_TIME_ZONE_POSIX_H_
>+#define ABSL_TIME_INTERNAL_CCTZ_TIME_ZONE_POSIX_H_
>+
>+#include <cstdint>
>+#include <string>
>+
>+namespace absl {
>+namespace time_internal {
>+namespace cctz {
>+
>+// The date/time of the transition. The date is specified as either:
>+// (J) the Nth day of the year (1 <= N <= 365), excluding leap days, or
>+// (N) the Nth day of the year (0 <= N <= 365), including leap days, or
>+// (M) the Nth weekday of a month (e.g., the 2nd Sunday in March).
>+// The time, specified as a day offset, identifies the particular moment
>+// of the transition, and may be negative or >= 24h, and in which case
>+// it would take us to another day, and perhaps week, or even month.
>+struct PosixTransition {
>+  enum DateFormat { J, N, M };
>+  struct {
>+    DateFormat fmt;
>+    union {
>+      struct {
>+        std::int_fast16_t day;  // day of non-leap year [1:365]
>+      } j;
>+      struct {
>+        std::int_fast16_t day;  // day of year [0:365]
>+      } n;
>+      struct {
>+        std::int_fast8_t month;    // month of year [1:12]
>+        std::int_fast8_t week;     // week of month [1:5] (5==last)
>+        std::int_fast8_t weekday;  // 0==Sun, ..., 6=Sat
>+      } m;
>+    };
>+  } date;
>+  struct {
>+    std::int_fast32_t offset;  // seconds before/after 00:00:00
>+  } time;
>+};
>+
>+// The entirety of a POSIX-std::string specified time-zone rule. The standard
>+// abbreviation and offset are always given. If the time zone includes
>+// daylight saving, then the daylight abbrevation is non-empty and the
>+// remaining fields are also valid. Note that the start/end transitions
>+// are not ordered---in the southern hemisphere the transition to end
>+// daylight time occurs first in any particular year.
>+struct PosixTimeZone {
>+  std::string std_abbr;
>+  std::int_fast32_t std_offset;
>+
>+  std::string dst_abbr;
>+  std::int_fast32_t dst_offset;
>+  PosixTransition dst_start;
>+  PosixTransition dst_end;
>+};
>+
>+// Breaks down a POSIX time-zone specification into its constituent pieces,
>+// filling in any missing values (DST offset, or start/end transition times)
>+// with the standard-defined defaults. Returns false if the specification
>+// could not be parsed (although some fields of *res may have been altered).
>+bool ParsePosixSpec(const std::string& spec, PosixTimeZone* res);
>+
>+}  // namespace cctz
>+}  // namespace time_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_TIME_INTERNAL_CCTZ_TIME_ZONE_POSIX_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/tzfile.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/tzfile.h
>new file mode 100644
>index 00000000000..90cfc0c4298
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/tzfile.h
>@@ -0,0 +1,117 @@
>+#ifndef TZFILE_H
>+
>+#define TZFILE_H
>+
>+/*
>+** This file is in the public domain, so clarified as of
>+** 1996-06-05 by Arthur David Olson.
>+*/
>+
>+/*
>+** This header is for use ONLY with the time conversion code.
>+** There is no guarantee that it will remain unchanged,
>+** or that it will remain at all.
>+** Do NOT copy it to any system include directory.
>+** Thank you!
>+*/
>+
>+/*
>+** Information about time zone files.
>+*/
>+
>+#ifndef TZDIR
>+#define TZDIR	"/usr/share/zoneinfo" /* Time zone object file directory */
>+#endif /* !defined TZDIR */
>+
>+#ifndef TZDEFAULT
>+#define TZDEFAULT	"/etc/localtime"
>+#endif /* !defined TZDEFAULT */
>+
>+#ifndef TZDEFRULES
>+#define TZDEFRULES	"posixrules"
>+#endif /* !defined TZDEFRULES */
>+
>+/*
>+** Each file begins with. . .
>+*/
>+
>+#define	TZ_MAGIC	"TZif"
>+
>+struct tzhead {
>+	char	tzh_magic[4];		/* TZ_MAGIC */
>+	char	tzh_version[1];		/* '\0' or '2' or '3' as of 2013 */
>+	char	tzh_reserved[15];	/* reserved; must be zero */
>+	char	tzh_ttisgmtcnt[4];	/* coded number of trans. time flags */
>+	char	tzh_ttisstdcnt[4];	/* coded number of trans. time flags */
>+	char	tzh_leapcnt[4];		/* coded number of leap seconds */
>+	char	tzh_timecnt[4];		/* coded number of transition times */
>+	char	tzh_typecnt[4];		/* coded number of local time types */
>+	char	tzh_charcnt[4];		/* coded number of abbr. chars */
>+};
>+
>+/*
>+** . . .followed by. . .
>+**
>+**	tzh_timecnt (char [4])s		coded transition times a la time(2)
>+**	tzh_timecnt (unsigned char)s	types of local time starting at above
>+**	tzh_typecnt repetitions of
>+**		one (char [4])		coded UT offset in seconds
>+**		one (unsigned char)	used to set tm_isdst
>+**		one (unsigned char)	that's an abbreviation list index
>+**	tzh_charcnt (char)s		'\0'-terminated zone abbreviations
>+**	tzh_leapcnt repetitions of
>+**		one (char [4])		coded leap second transition times
>+**		one (char [4])		total correction after above
>+**	tzh_ttisstdcnt (char)s		indexed by type; if 1, transition
>+**					time is standard time, if 0,
>+**					transition time is wall clock time
>+**					if absent, transition times are
>+**					assumed to be wall clock time
>+**	tzh_ttisgmtcnt (char)s		indexed by type; if 1, transition
>+**					time is UT, if 0,
>+**					transition time is local time
>+**					if absent, transition times are
>+**					assumed to be local time
>+*/
>+
>+/*
>+** If tzh_version is '2' or greater, the above is followed by a second instance
>+** of tzhead and a second instance of the data in which each coded transition
>+** time uses 8 rather than 4 chars,
>+** then a POSIX-TZ-environment-variable-style std::string for use in handling
>+** instants after the last transition time stored in the file
>+** (with nothing between the newlines if there is no POSIX representation for
>+** such instants).
>+**
>+** If tz_version is '3' or greater, the above is extended as follows.
>+** First, the POSIX TZ std::string's hour offset may range from -167
>+** through 167 as compared to the POSIX-required 0 through 24.
>+** Second, its DST start time may be January 1 at 00:00 and its stop
>+** time December 31 at 24:00 plus the difference between DST and
>+** standard time, indicating DST all year.
>+*/
>+
>+/*
>+** In the current implementation, "tzset()" refuses to deal with files that
>+** exceed any of the limits below.
>+*/
>+
>+#ifndef TZ_MAX_TIMES
>+#define TZ_MAX_TIMES	2000
>+#endif /* !defined TZ_MAX_TIMES */
>+
>+#ifndef TZ_MAX_TYPES
>+/* This must be at least 17 for Europe/Samara and Europe/Vilnius.  */
>+#define TZ_MAX_TYPES	256 /* Limited by what (unsigned char)'s can hold */
>+#endif /* !defined TZ_MAX_TYPES */
>+
>+#ifndef TZ_MAX_CHARS
>+#define TZ_MAX_CHARS	50	/* Maximum number of abbreviation characters */
>+				/* (limited by what unsigned chars can hold) */
>+#endif /* !defined TZ_MAX_CHARS */
>+
>+#ifndef TZ_MAX_LEAPS
>+#define TZ_MAX_LEAPS	50	/* Maximum number of leap second corrections */
>+#endif /* !defined TZ_MAX_LEAPS */
>+
>+#endif /* !defined TZFILE_H */
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/zone_info_source.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/zone_info_source.cc
>new file mode 100644
>index 00000000000..bf2d2d2d2b5
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/cctz/src/zone_info_source.cc
>@@ -0,0 +1,79 @@
>+// Copyright 2016 Google Inc. All Rights Reserved.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//   http://www.apache.org/licenses/LICENSE-2.0
>+//
>+//   Unless required by applicable law or agreed to in writing, software
>+//   distributed under the License is distributed on an "AS IS" BASIS,
>+//   WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+//   See the License for the specific language governing permissions and
>+//   limitations under the License.
>+
>+#include "absl/time/internal/cctz/include/cctz/zone_info_source.h"
>+
>+namespace absl {
>+namespace time_internal {
>+namespace cctz {
>+
>+// Defined out-of-line to avoid emitting a weak vtable in all TUs.
>+ZoneInfoSource::~ZoneInfoSource() {}
>+std::string ZoneInfoSource::Version() const { return std::string(); }
>+
>+}  // namespace cctz
>+}  // namespace time_internal
>+}  // namespace absl
>+
>+namespace absl {
>+namespace time_internal {
>+namespace cctz_extension {
>+
>+namespace {
>+
>+// A default for cctz_extension::zone_info_source_factory, which simply
>+// defers to the fallback factory.
>+std::unique_ptr<absl::time_internal::cctz::ZoneInfoSource> DefaultFactory(
>+    const std::string& name,
>+    const std::function<std::unique_ptr<absl::time_internal::cctz::ZoneInfoSource>(
>+        const std::string& name)>& fallback_factory) {
>+  return fallback_factory(name);
>+}
>+
>+}  // namespace
>+
>+// A "weak" definition for cctz_extension::zone_info_source_factory.
>+// The user may override this with their own "strong" definition (see
>+// zone_info_source.h).
>+#if defined(_MSC_VER)
>+extern ZoneInfoSourceFactory zone_info_source_factory;
>+extern ZoneInfoSourceFactory default_factory;
>+ZoneInfoSourceFactory default_factory = DefaultFactory;
>+#if defined(_M_IX86)
>+#pragma comment( \
>+    linker,      \
>+    "/alternatename:?zone_info_source_factory@cctz_extension@time_internal@absl@@3P6A?AV?$unique_ptr@VZoneInfoSource@cctz@time_internal@absl@@U?$default_delete@VZoneInfoSource@cctz@time_internal@absl@@@std@@@std@@ABV?$basic_string@DU?$char_traits@D@std@@V?$allocator@D@2@@5@ABV?$function@$$A6A?AV?$unique_ptr@VZoneInfoSource@cctz@time_internal@absl@@U?$default_delete@VZoneInfoSource@cctz@time_internal@absl@@@std@@@std@@ABV?$basic_string@DU?$char_traits@D@std@@V?$allocator@D@2@@2@@Z@5@@ZA=?default_factory@cctz_extension@time_internal@absl@@3P6A?AV?$unique_ptr@VZoneInfoSource@cctz@time_internal@absl@@U?$default_delete@VZoneInfoSource@cctz@time_internal@absl@@@std@@@std@@ABV?$basic_string@DU?$char_traits@D@std@@V?$allocator@D@2@@5@ABV?$function@$$A6A?AV?$unique_ptr@VZoneInfoSource@cctz@time_internal@absl@@U?$default_delete@VZoneInfoSource@cctz@time_internal@absl@@@std@@@std@@ABV?$basic_string@DU?$char_traits@D@std@@V?$allocator@D@2@@2@@Z@5@@ZA")
>+#elif defined(_M_IA_64) || defined(_M_AMD64)
>+#pragma comment( \
>+    linker,      \
>+    "/alternatename:?zone_info_source_factory@cctz_extension@time_internal@absl@@3P6A?AV?$unique_ptr@VZoneInfoSource@cctz@time_internal@absl@@U?$default_delete@VZoneInfoSource@cctz@time_internal@absl@@@std@@@std@@AEBV?$basic_string@DU?$char_traits@D@std@@V?$allocator@D@2@@5@AEBV?$function@$$A6A?AV?$unique_ptr@VZoneInfoSource@cctz@time_internal@absl@@U?$default_delete@VZoneInfoSource@cctz@time_internal@absl@@@std@@@std@@AEBV?$basic_string@DU?$char_traits@D@std@@V?$allocator@D@2@@2@@Z@5@@ZEA=?default_factory@cctz_extension@time_internal@absl@@3P6A?AV?$unique_ptr@VZoneInfoSource@cctz@time_internal@absl@@U?$default_delete@VZoneInfoSource@cctz@time_internal@absl@@@std@@@std@@AEBV?$basic_string@DU?$char_traits@D@std@@V?$allocator@D@2@@5@AEBV?$function@$$A6A?AV?$unique_ptr@VZoneInfoSource@cctz@time_internal@absl@@U?$default_delete@VZoneInfoSource@cctz@time_internal@absl@@@std@@@std@@AEBV?$basic_string@DU?$char_traits@D@std@@V?$allocator@D@2@@2@@Z@5@@ZEA")
>+#else
>+#error Unsupported MSVC platform
>+#endif
>+#else  // _MSC_VER
>+#if !defined(__has_attribute)
>+#define __has_attribute(x) 0
>+#endif
>+#if __has_attribute(weak) || defined(__GNUC__)
>+ZoneInfoSourceFactory zone_info_source_factory
>+    __attribute__((weak)) = DefaultFactory;
>+#else
>+// Make it a "strong" definition if we have no other choice.
>+ZoneInfoSourceFactory zone_info_source_factory = DefaultFactory;
>+#endif
>+#endif  // _MSC_VER
>+
>+}  // namespace cctz_extension
>+}  // namespace time_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/get_current_time_chrono.inc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/get_current_time_chrono.inc
>new file mode 100644
>index 00000000000..cf884a10ed9
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/get_current_time_chrono.inc
>@@ -0,0 +1,29 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include <chrono>
>+#include <cstdint>
>+
>+namespace absl {
>+namespace time_internal {
>+
>+static int64_t GetCurrentTimeNanosFromSystem() {
>+  return std::chrono::duration_cast<std::chrono::nanoseconds>(
>+             std::chrono::system_clock::now() -
>+             std::chrono::system_clock::from_time_t(0))
>+      .count();
>+}
>+
>+}  // namespace time_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/get_current_time_posix.inc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/get_current_time_posix.inc
>new file mode 100644
>index 00000000000..65474ca6da1
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/get_current_time_posix.inc
>@@ -0,0 +1,22 @@
>+#include "absl/time/clock.h"
>+
>+#include <sys/time.h>
>+#include <ctime>
>+#include <cstdint>
>+
>+#include "absl/base/internal/raw_logging.h"
>+
>+namespace absl {
>+namespace time_internal {
>+
>+static int64_t GetCurrentTimeNanosFromSystem() {
>+  const int64_t kNanosPerSecond = 1000 * 1000 * 1000;
>+  struct timespec ts;
>+  ABSL_RAW_CHECK(clock_gettime(CLOCK_REALTIME, &ts) == 0,
>+                 "Failed to read real-time clock.");
>+  return (int64_t{ts.tv_sec} * kNanosPerSecond +
>+          int64_t{ts.tv_nsec});
>+}
>+
>+}  // namespace time_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/test_util.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/test_util.cc
>new file mode 100644
>index 00000000000..bbbef7da70c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/test_util.cc
>@@ -0,0 +1,129 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/time/internal/test_util.h"
>+
>+#include <algorithm>
>+#include <cstddef>
>+#include <cstring>
>+
>+#include "absl/base/internal/raw_logging.h"
>+#include "absl/time/internal/cctz/include/cctz/zone_info_source.h"
>+
>+namespace cctz = absl::time_internal::cctz;
>+
>+namespace absl {
>+namespace time_internal {
>+
>+#if GTEST_USES_SIMPLE_RE
>+extern const char kZoneAbbrRE[] = ".*";  // just punt
>+#else
>+extern const char kZoneAbbrRE[] = "[A-Za-z]{3,4}|[-+][0-9]{2}([0-9]{2})?";
>+#endif
>+
>+TimeZone LoadTimeZone(const std::string& name) {
>+  TimeZone tz;
>+  ABSL_RAW_CHECK(LoadTimeZone(name, &tz), name.c_str());
>+  return tz;
>+}
>+
>+}  // namespace time_internal
>+}  // namespace absl
>+
>+namespace absl {
>+namespace time_internal {
>+namespace cctz_extension {
>+namespace {
>+
>+// Embed the zoneinfo data for time zones used during tests and benchmarks.
>+// The data was generated using "xxd -i zoneinfo-file".  There is no need
>+// to update the data as long as the tests do not depend on recent changes
>+// (and the past rules remain the same).
>+#include "absl/time/internal/zoneinfo.inc"
>+
>+const struct ZoneInfo {
>+  const char* name;
>+  const char* data;
>+  std::size_t length;
>+} kZoneInfo[] = {
>+    // The three real time zones used by :time_test and :time_benchmark.
>+    {"America/Los_Angeles",  //
>+     reinterpret_cast<char*>(America_Los_Angeles), America_Los_Angeles_len},
>+    {"America/New_York",  //
>+     reinterpret_cast<char*>(America_New_York), America_New_York_len},
>+    {"Australia/Sydney",  //
>+     reinterpret_cast<char*>(Australia_Sydney), Australia_Sydney_len},
>+
>+    // Other zones named in tests but which should fail to load.
>+    {"Invalid/TimeZone", nullptr, 0},
>+    {"", nullptr, 0},
>+
>+    // Also allow for loading the local time zone under TZ=US/Pacific.
>+    {"US/Pacific",  //
>+     reinterpret_cast<char*>(America_Los_Angeles), America_Los_Angeles_len},
>+
>+    // Allows use of the local time zone from a system-specific location.
>+#ifdef _MSC_VER
>+    {"localtime",  //
>+     reinterpret_cast<char*>(America_Los_Angeles), America_Los_Angeles_len},
>+#else
>+    {"/etc/localtime",  //
>+     reinterpret_cast<char*>(America_Los_Angeles), America_Los_Angeles_len},
>+#endif
>+};
>+
>+class TestZoneInfoSource : public cctz::ZoneInfoSource {
>+ public:
>+  TestZoneInfoSource(const char* data, std::size_t size)
>+      : data_(data), end_(data + size) {}
>+
>+  std::size_t Read(void* ptr, std::size_t size) override {
>+    const std::size_t len = std::min<std::size_t>(size, end_ - data_);
>+    memcpy(ptr, data_, len);
>+    data_ += len;
>+    return len;
>+  }
>+
>+  int Skip(std::size_t offset) override {
>+    data_ += std::min<std::size_t>(offset, end_ - data_);
>+    return 0;
>+  }
>+
>+ private:
>+  const char* data_;
>+  const char* const end_;
>+};
>+
>+std::unique_ptr<cctz::ZoneInfoSource> TestFactory(
>+    const std::string& name,
>+    const std::function<std::unique_ptr<cctz::ZoneInfoSource>(
>+        const std::string& name)>& /*fallback_factory*/) {
>+  for (const ZoneInfo& zoneinfo : kZoneInfo) {
>+    if (name == zoneinfo.name) {
>+      if (zoneinfo.data == nullptr) return nullptr;
>+      return std::unique_ptr<cctz::ZoneInfoSource>(
>+          new TestZoneInfoSource(zoneinfo.data, zoneinfo.length));
>+    }
>+  }
>+  ABSL_RAW_LOG(FATAL, "Unexpected time zone \"%s\" in test", name.c_str());
>+  return nullptr;
>+}
>+
>+}  // namespace
>+
>+ZoneInfoSourceFactory zone_info_source_factory = TestFactory;
>+
>+}  // namespace cctz_extension
>+}  // namespace time_internal
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/test_util.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/test_util.h
>new file mode 100644
>index 00000000000..8fd5fb9fd03
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/test_util.h
>@@ -0,0 +1,55 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#ifndef ABSL_TIME_INTERNAL_TEST_UTIL_H_
>+#define ABSL_TIME_INTERNAL_TEST_UTIL_H_
>+
>+#include <string>
>+
>+#include "gmock/gmock.h"
>+#include "gtest/gtest.h"
>+#include "absl/time/time.h"
>+
>+// This helper is a macro so that failed expectations show up with the
>+// correct line numbers.
>+//
>+// This is for internal testing of the Base Time library itself. This is not
>+// part of a public API.
>+#define ABSL_INTERNAL_EXPECT_TIME(bd, y, m, d, h, min, s, off, isdst)     \
>+  do {                                                                    \
>+    EXPECT_EQ(y, bd.year);                                                \
>+    EXPECT_EQ(m, bd.month);                                               \
>+    EXPECT_EQ(d, bd.day);                                                 \
>+    EXPECT_EQ(h, bd.hour);                                                \
>+    EXPECT_EQ(min, bd.minute);                                            \
>+    EXPECT_EQ(s, bd.second);                                              \
>+    EXPECT_EQ(off, bd.offset);                                            \
>+    EXPECT_EQ(isdst, bd.is_dst);                                          \
>+    EXPECT_THAT(bd.zone_abbr,                                             \
>+                testing::MatchesRegex(absl::time_internal::kZoneAbbrRE)); \
>+  } while (0)
>+
>+namespace absl {
>+namespace time_internal {
>+
>+// A regular expression that matches all zone abbreviations (%Z).
>+extern const char kZoneAbbrRE[];
>+
>+// Loads the named timezone, but dies on any failure.
>+absl::TimeZone LoadTimeZone(const std::string& name);
>+
>+}  // namespace time_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_TIME_INTERNAL_TEST_UTIL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/zoneinfo.inc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/zoneinfo.inc
>new file mode 100644
>index 00000000000..bfed82990dd
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/internal/zoneinfo.inc
>@@ -0,0 +1,729 @@
>+unsigned char America_Los_Angeles[] = {
>+  0x54, 0x5a, 0x69, 0x66, 0x32, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
>+  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x05,
>+  0x00, 0x00, 0x00, 0x05, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0xba,
>+  0x00, 0x00, 0x00, 0x05, 0x00, 0x00, 0x00, 0x14, 0x80, 0x00, 0x00, 0x00,
>+  0x9e, 0xa6, 0x48, 0xa0, 0x9f, 0xbb, 0x15, 0x90, 0xa0, 0x86, 0x2a, 0xa0,
>+  0xa1, 0x9a, 0xf7, 0x90, 0xcb, 0x89, 0x1a, 0xa0, 0xd2, 0x23, 0xf4, 0x70,
>+  0xd2, 0x61, 0x26, 0x10, 0xd6, 0xfe, 0x74, 0x5c, 0xd8, 0x80, 0xad, 0x90,
>+  0xda, 0xfe, 0xc3, 0x90, 0xdb, 0xc0, 0x90, 0x10, 0xdc, 0xde, 0xa5, 0x90,
>+  0xdd, 0xa9, 0xac, 0x90, 0xde, 0xbe, 0x87, 0x90, 0xdf, 0x89, 0x8e, 0x90,
>+  0xe0, 0x9e, 0x69, 0x90, 0xe1, 0x69, 0x70, 0x90, 0xe2, 0x7e, 0x4b, 0x90,
>+  0xe3, 0x49, 0x52, 0x90, 0xe4, 0x5e, 0x2d, 0x90, 0xe5, 0x29, 0x34, 0x90,
>+  0xe6, 0x47, 0x4a, 0x10, 0xe7, 0x12, 0x51, 0x10, 0xe8, 0x27, 0x2c, 0x10,
>+  0xe8, 0xf2, 0x33, 0x10, 0xea, 0x07, 0x0e, 0x10, 0xea, 0xd2, 0x15, 0x10,
>+  0xeb, 0xe6, 0xf0, 0x10, 0xec, 0xb1, 0xf7, 0x10, 0xed, 0xc6, 0xd2, 0x10,
>+  0xee, 0x91, 0xd9, 0x10, 0xef, 0xaf, 0xee, 0x90, 0xf0, 0x71, 0xbb, 0x10,
>+  0xf1, 0x8f, 0xd0, 0x90, 0xf2, 0x7f, 0xc1, 0x90, 0xf3, 0x6f, 0xb2, 0x90,
>+  0xf4, 0x5f, 0xa3, 0x90, 0xf5, 0x4f, 0x94, 0x90, 0xf6, 0x3f, 0x85, 0x90,
>+  0xf7, 0x2f, 0x76, 0x90, 0xf8, 0x28, 0xa2, 0x10, 0xf9, 0x0f, 0x58, 0x90,
>+  0xfa, 0x08, 0x84, 0x10, 0xfa, 0xf8, 0x83, 0x20, 0xfb, 0xe8, 0x66, 0x10,
>+  0xfc, 0xd8, 0x65, 0x20, 0xfd, 0xc8, 0x48, 0x10, 0xfe, 0xb8, 0x47, 0x20,
>+  0xff, 0xa8, 0x2a, 0x10, 0x00, 0x98, 0x29, 0x20, 0x01, 0x88, 0x0c, 0x10,
>+  0x02, 0x78, 0x0b, 0x20, 0x03, 0x71, 0x28, 0x90, 0x04, 0x61, 0x27, 0xa0,
>+  0x05, 0x51, 0x0a, 0x90, 0x06, 0x41, 0x09, 0xa0, 0x07, 0x30, 0xec, 0x90,
>+  0x07, 0x8d, 0x43, 0xa0, 0x09, 0x10, 0xce, 0x90, 0x09, 0xad, 0xbf, 0x20,
>+  0x0a, 0xf0, 0xb0, 0x90, 0x0b, 0xe0, 0xaf, 0xa0, 0x0c, 0xd9, 0xcd, 0x10,
>+  0x0d, 0xc0, 0x91, 0xa0, 0x0e, 0xb9, 0xaf, 0x10, 0x0f, 0xa9, 0xae, 0x20,
>+  0x10, 0x99, 0x91, 0x10, 0x11, 0x89, 0x90, 0x20, 0x12, 0x79, 0x73, 0x10,
>+  0x13, 0x69, 0x72, 0x20, 0x14, 0x59, 0x55, 0x10, 0x15, 0x49, 0x54, 0x20,
>+  0x16, 0x39, 0x37, 0x10, 0x17, 0x29, 0x36, 0x20, 0x18, 0x22, 0x53, 0x90,
>+  0x19, 0x09, 0x18, 0x20, 0x1a, 0x02, 0x35, 0x90, 0x1a, 0xf2, 0x34, 0xa0,
>+  0x1b, 0xe2, 0x17, 0x90, 0x1c, 0xd2, 0x16, 0xa0, 0x1d, 0xc1, 0xf9, 0x90,
>+  0x1e, 0xb1, 0xf8, 0xa0, 0x1f, 0xa1, 0xdb, 0x90, 0x20, 0x76, 0x2b, 0x20,
>+  0x21, 0x81, 0xbd, 0x90, 0x22, 0x56, 0x0d, 0x20, 0x23, 0x6a, 0xda, 0x10,
>+  0x24, 0x35, 0xef, 0x20, 0x25, 0x4a, 0xbc, 0x10, 0x26, 0x15, 0xd1, 0x20,
>+  0x27, 0x2a, 0x9e, 0x10, 0x27, 0xfe, 0xed, 0xa0, 0x29, 0x0a, 0x80, 0x10,
>+  0x29, 0xde, 0xcf, 0xa0, 0x2a, 0xea, 0x62, 0x10, 0x2b, 0xbe, 0xb1, 0xa0,
>+  0x2c, 0xd3, 0x7e, 0x90, 0x2d, 0x9e, 0x93, 0xa0, 0x2e, 0xb3, 0x60, 0x90,
>+  0x2f, 0x7e, 0x75, 0xa0, 0x30, 0x93, 0x42, 0x90, 0x31, 0x67, 0x92, 0x20,
>+  0x32, 0x73, 0x24, 0x90, 0x33, 0x47, 0x74, 0x20, 0x34, 0x53, 0x06, 0x90,
>+  0x35, 0x27, 0x56, 0x20, 0x36, 0x32, 0xe8, 0x90, 0x37, 0x07, 0x38, 0x20,
>+  0x38, 0x1c, 0x05, 0x10, 0x38, 0xe7, 0x1a, 0x20, 0x39, 0xfb, 0xe7, 0x10,
>+  0x3a, 0xc6, 0xfc, 0x20, 0x3b, 0xdb, 0xc9, 0x10, 0x3c, 0xb0, 0x18, 0xa0,
>+  0x3d, 0xbb, 0xab, 0x10, 0x3e, 0x8f, 0xfa, 0xa0, 0x3f, 0x9b, 0x8d, 0x10,
>+  0x40, 0x6f, 0xdc, 0xa0, 0x41, 0x84, 0xa9, 0x90, 0x42, 0x4f, 0xbe, 0xa0,
>+  0x43, 0x64, 0x8b, 0x90, 0x44, 0x2f, 0xa0, 0xa0, 0x45, 0x44, 0x6d, 0x90,
>+  0x45, 0xf3, 0xd3, 0x20, 0x47, 0x2d, 0x8a, 0x10, 0x47, 0xd3, 0xb5, 0x20,
>+  0x49, 0x0d, 0x6c, 0x10, 0x49, 0xb3, 0x97, 0x20, 0x4a, 0xed, 0x4e, 0x10,
>+  0x4b, 0x9c, 0xb3, 0xa0, 0x4c, 0xd6, 0x6a, 0x90, 0x4d, 0x7c, 0x95, 0xa0,
>+  0x4e, 0xb6, 0x4c, 0x90, 0x4f, 0x5c, 0x77, 0xa0, 0x50, 0x96, 0x2e, 0x90,
>+  0x51, 0x3c, 0x59, 0xa0, 0x52, 0x76, 0x10, 0x90, 0x53, 0x1c, 0x3b, 0xa0,
>+  0x54, 0x55, 0xf2, 0x90, 0x54, 0xfc, 0x1d, 0xa0, 0x56, 0x35, 0xd4, 0x90,
>+  0x56, 0xe5, 0x3a, 0x20, 0x58, 0x1e, 0xf1, 0x10, 0x58, 0xc5, 0x1c, 0x20,
>+  0x59, 0xfe, 0xd3, 0x10, 0x5a, 0xa4, 0xfe, 0x20, 0x5b, 0xde, 0xb5, 0x10,
>+  0x5c, 0x84, 0xe0, 0x20, 0x5d, 0xbe, 0x97, 0x10, 0x5e, 0x64, 0xc2, 0x20,
>+  0x5f, 0x9e, 0x79, 0x10, 0x60, 0x4d, 0xde, 0xa0, 0x61, 0x87, 0x95, 0x90,
>+  0x62, 0x2d, 0xc0, 0xa0, 0x63, 0x67, 0x77, 0x90, 0x64, 0x0d, 0xa2, 0xa0,
>+  0x65, 0x47, 0x59, 0x90, 0x65, 0xed, 0x84, 0xa0, 0x67, 0x27, 0x3b, 0x90,
>+  0x67, 0xcd, 0x66, 0xa0, 0x69, 0x07, 0x1d, 0x90, 0x69, 0xad, 0x48, 0xa0,
>+  0x6a, 0xe6, 0xff, 0x90, 0x6b, 0x96, 0x65, 0x20, 0x6c, 0xd0, 0x1c, 0x10,
>+  0x6d, 0x76, 0x47, 0x20, 0x6e, 0xaf, 0xfe, 0x10, 0x6f, 0x56, 0x29, 0x20,
>+  0x70, 0x8f, 0xe0, 0x10, 0x71, 0x36, 0x0b, 0x20, 0x72, 0x6f, 0xc2, 0x10,
>+  0x73, 0x15, 0xed, 0x20, 0x74, 0x4f, 0xa4, 0x10, 0x74, 0xff, 0x09, 0xa0,
>+  0x76, 0x38, 0xc0, 0x90, 0x76, 0xde, 0xeb, 0xa0, 0x78, 0x18, 0xa2, 0x90,
>+  0x78, 0xbe, 0xcd, 0xa0, 0x79, 0xf8, 0x84, 0x90, 0x7a, 0x9e, 0xaf, 0xa0,
>+  0x7b, 0xd8, 0x66, 0x90, 0x7c, 0x7e, 0x91, 0xa0, 0x7d, 0xb8, 0x48, 0x90,
>+  0x7e, 0x5e, 0x73, 0xa0, 0x7f, 0x98, 0x2a, 0x90, 0x02, 0x01, 0x02, 0x01,
>+  0x02, 0x03, 0x04, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02,
>+  0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02,
>+  0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02,
>+  0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02,
>+  0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02,
>+  0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02,
>+  0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02,
>+  0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02,
>+  0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02,
>+  0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02,
>+  0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02,
>+  0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02,
>+  0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02,
>+  0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02,
>+  0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02,
>+  0x01, 0x02, 0xff, 0xff, 0x91, 0x26, 0x00, 0x00, 0xff, 0xff, 0x9d, 0x90,
>+  0x01, 0x04, 0xff, 0xff, 0x8f, 0x80, 0x00, 0x08, 0xff, 0xff, 0x9d, 0x90,
>+  0x01, 0x0c, 0xff, 0xff, 0x9d, 0x90, 0x01, 0x10, 0x4c, 0x4d, 0x54, 0x00,
>+  0x50, 0x44, 0x54, 0x00, 0x50, 0x53, 0x54, 0x00, 0x50, 0x57, 0x54, 0x00,
>+  0x50, 0x50, 0x54, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x00, 0x00, 0x00,
>+  0x00, 0x01, 0x54, 0x5a, 0x69, 0x66, 0x32, 0x00, 0x00, 0x00, 0x00, 0x00,
>+  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
>+  0x00, 0x05, 0x00, 0x00, 0x00, 0x05, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
>+  0x00, 0xbb, 0x00, 0x00, 0x00, 0x05, 0x00, 0x00, 0x00, 0x14, 0xf8, 0x00,
>+  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0xff, 0xff, 0xff, 0xff, 0x5e, 0x04,
>+  0x1a, 0xc0, 0xff, 0xff, 0xff, 0xff, 0x9e, 0xa6, 0x48, 0xa0, 0xff, 0xff,
>+  0xff, 0xff, 0x9f, 0xbb, 0x15, 0x90, 0xff, 0xff, 0xff, 0xff, 0xa0, 0x86,
>+  0x2a, 0xa0, 0xff, 0xff, 0xff, 0xff, 0xa1, 0x9a, 0xf7, 0x90, 0xff, 0xff,
>+  0xff, 0xff, 0xcb, 0x89, 0x1a, 0xa0, 0xff, 0xff, 0xff, 0xff, 0xd2, 0x23,
>+  0xf4, 0x70, 0xff, 0xff, 0xff, 0xff, 0xd2, 0x61, 0x26, 0x10, 0xff, 0xff,
>+  0xff, 0xff, 0xd6, 0xfe, 0x74, 0x5c, 0xff, 0xff, 0xff, 0xff, 0xd8, 0x80,
>+  0xad, 0x90, 0xff, 0xff, 0xff, 0xff, 0xda, 0xfe, 0xc3, 0x90, 0xff, 0xff,
>+  0xff, 0xff, 0xdb, 0xc0, 0x90, 0x10, 0xff, 0xff, 0xff, 0xff, 0xdc, 0xde,
>+  0xa5, 0x90, 0xff, 0xff, 0xff, 0xff, 0xdd, 0xa9, 0xac, 0x90, 0xff, 0xff,
>+  0xff, 0xff, 0xde, 0xbe, 0x87, 0x90, 0xff, 0xff, 0xff, 0xff, 0xdf, 0x89,
>+  0x8e, 0x90, 0xff, 0xff, 0xff, 0xff, 0xe0, 0x9e, 0x69, 0x90, 0xff, 0xff,
>+  0xff, 0xff, 0xe1, 0x69, 0x70, 0x90, 0xff, 0xff, 0xff, 0xff, 0xe2, 0x7e,
>+  0x4b, 0x90, 0xff, 0xff, 0xff, 0xff, 0xe3, 0x49, 0x52, 0x90, 0xff, 0xff,
>+  0xff, 0xff, 0xe4, 0x5e, 0x2d, 0x90, 0xff, 0xff, 0xff, 0xff, 0xe5, 0x29,
>+  0x34, 0x90, 0xff, 0xff, 0xff, 0xff, 0xe6, 0x47, 0x4a, 0x10, 0xff, 0xff,
>+  0xff, 0xff, 0xe7, 0x12, 0x51, 0x10, 0xff, 0xff, 0xff, 0xff, 0xe8, 0x27,
>+  0x2c, 0x10, 0xff, 0xff, 0xff, 0xff, 0xe8, 0xf2, 0x33, 0x10, 0xff, 0xff,
>+  0xff, 0xff, 0xea, 0x07, 0x0e, 0x10, 0xff, 0xff, 0xff, 0xff, 0xea, 0xd2,
>+  0x15, 0x10, 0xff, 0xff, 0xff, 0xff, 0xeb, 0xe6, 0xf0, 0x10, 0xff, 0xff,
>+  0xff, 0xff, 0xec, 0xb1, 0xf7, 0x10, 0xff, 0xff, 0xff, 0xff, 0xed, 0xc6,
>+  0xd2, 0x10, 0xff, 0xff, 0xff, 0xff, 0xee, 0x91, 0xd9, 0x10, 0xff, 0xff,
>+  0xff, 0xff, 0xef, 0xaf, 0xee, 0x90, 0xff, 0xff, 0xff, 0xff, 0xf0, 0x71,
>+  0xbb, 0x10, 0xff, 0xff, 0xff, 0xff, 0xf1, 0x8f, 0xd0, 0x90, 0xff, 0xff,
>+  0xff, 0xff, 0xf2, 0x7f, 0xc1, 0x90, 0xff, 0xff, 0xff, 0xff, 0xf3, 0x6f,
>+  0xb2, 0x90, 0xff, 0xff, 0xff, 0xff, 0xf4, 0x5f, 0xa3, 0x90, 0xff, 0xff,
>+  0xff, 0xff, 0xf5, 0x4f, 0x94, 0x90, 0xff, 0xff, 0xff, 0xff, 0xf6, 0x3f,
>+  0x85, 0x90, 0xff, 0xff, 0xff, 0xff, 0xf7, 0x2f, 0x76, 0x90, 0xff, 0xff,
>+  0xff, 0xff, 0xf8, 0x28, 0xa2, 0x10, 0xff, 0xff, 0xff, 0xff, 0xf9, 0x0f,
>+  0x58, 0x90, 0xff, 0xff, 0xff, 0xff, 0xfa, 0x08, 0x84, 0x10, 0xff, 0xff,
>+  0xff, 0xff, 0xfa, 0xf8, 0x83, 0x20, 0xff, 0xff, 0xff, 0xff, 0xfb, 0xe8,
>+  0x66, 0x10, 0xff, 0xff, 0xff, 0xff, 0xfc, 0xd8, 0x65, 0x20, 0xff, 0xff,
>+  0xff, 0xff, 0xfd, 0xc8, 0x48, 0x10, 0xff, 0xff, 0xff, 0xff, 0xfe, 0xb8,
>+  0x47, 0x20, 0xff, 0xff, 0xff, 0xff, 0xff, 0xa8, 0x2a, 0x10, 0x00, 0x00,
>+  0x00, 0x00, 0x00, 0x98, 0x29, 0x20, 0x00, 0x00, 0x00, 0x00, 0x01, 0x88,
>+  0x0c, 0x10, 0x00, 0x00, 0x00, 0x00, 0x02, 0x78, 0x0b, 0x20, 0x00, 0x00,
>+  0x00, 0x00, 0x03, 0x71, 0x28, 0x90, 0x00, 0x00, 0x00, 0x00, 0x04, 0x61,
>+  0x27, 0xa0, 0x00, 0x00, 0x00, 0x00, 0x05, 0x51, 0x0a, 0x90, 0x00, 0x00,
>+  0x00, 0x00, 0x06, 0x41, 0x09, 0xa0, 0x00, 0x00, 0x00, 0x00, 0x07, 0x30,
>+  0xec, 0x90, 0x00, 0x00, 0x00, 0x00, 0x07, 0x8d, 0x43, 0xa0, 0x00, 0x00,
>+  0x00, 0x00, 0x09, 0x10, 0xce, 0x90, 0x00, 0x00, 0x00, 0x00, 0x09, 0xad,
>+  0xbf, 0x20, 0x00, 0x00, 0x00, 0x00, 0x0a, 0xf0, 0xb0, 0x90, 0x00, 0x00,
>+  0x00, 0x00, 0x0b, 0xe0, 0xaf, 0xa0, 0x00, 0x00, 0x00, 0x00, 0x0c, 0xd9,
>+  0xcd, 0x10, 0x00, 0x00, 0x00, 0x00, 0x0d, 0xc0, 0x91, 0xa0, 0x00, 0x00,
>+  0x00, 0x00, 0x0e, 0xb9, 0xaf, 0x10, 0x00, 0x00, 0x00, 0x00, 0x0f, 0xa9,
>+  0xae, 0x20, 0x00, 0x00, 0x00, 0x00, 0x10, 0x99, 0x91, 0x10, 0x00, 0x00,
>+  0x00, 0x00, 0x11, 0x89, 0x90, 0x20, 0x00, 0x00, 0x00, 0x00, 0x12, 0x79,
>+  0x73, 0x10, 0x00, 0x00, 0x00, 0x00, 0x13, 0x69, 0x72, 0x20, 0x00, 0x00,
>+  0x00, 0x00, 0x14, 0x59, 0x55, 0x10, 0x00, 0x00, 0x00, 0x00, 0x15, 0x49,
>+  0x54, 0x20, 0x00, 0x00, 0x00, 0x00, 0x16, 0x39, 0x37, 0x10, 0x00, 0x00,
>+  0x00, 0x00, 0x17, 0x29, 0x36, 0x20, 0x00, 0x00, 0x00, 0x00, 0x18, 0x22,
>+  0x53, 0x90, 0x00, 0x00, 0x00, 0x00, 0x19, 0x09, 0x18, 0x20, 0x00, 0x00,
>+  0x00, 0x00, 0x1a, 0x02, 0x35, 0x90, 0x00, 0x00, 0x00, 0x00, 0x1a, 0xf2,
>+  0x34, 0xa0, 0x00, 0x00, 0x00, 0x00, 0x1b, 0xe2, 0x17, 0x90, 0x00, 0x00,
>+  0x00, 0x00, 0x1c, 0xd2, 0x16, 0xa0, 0x00, 0x00, 0x00, 0x00, 0x1d, 0xc1,
>+  0xf9, 0x90, 0x00, 0x00, 0x00, 0x00, 0x1e, 0xb1, 0xf8, 0xa0, 0x00, 0x00,
>+  0x00, 0x00, 0x1f, 0xa1, 0xdb, 0x90, 0x00, 0x00, 0x00, 0x00, 0x20, 0x76,
>+  0x2b, 0x20, 0x00, 0x00, 0x00, 0x00, 0x21, 0x81, 0xbd, 0x90, 0x00, 0x00,
>+  0x00, 0x00, 0x22, 0x56, 0x0d, 0x20, 0x00, 0x00, 0x00, 0x00, 0x23, 0x6a,
>+  0xda, 0x10, 0x00, 0x00, 0x00, 0x00, 0x24, 0x35, 0xef, 0x20, 0x00, 0x00,
>+  0x00, 0x00, 0x25, 0x4a, 0xbc, 0x10, 0x00, 0x00, 0x00, 0x00, 0x26, 0x15,
>+  0xd1, 0x20, 0x00, 0x00, 0x00, 0x00, 0x27, 0x2a, 0x9e, 0x10, 0x00, 0x00,
>+  0x00, 0x00, 0x27, 0xfe, 0xed, 0xa0, 0x00, 0x00, 0x00, 0x00, 0x29, 0x0a,
>+  0x80, 0x10, 0x00, 0x00, 0x00, 0x00, 0x29, 0xde, 0xcf, 0xa0, 0x00, 0x00,
>+  0x00, 0x00, 0x2a, 0xea, 0x62, 0x10, 0x00, 0x00, 0x00, 0x00, 0x2b, 0xbe,
>+  0xb1, 0xa0, 0x00, 0x00, 0x00, 0x00, 0x2c, 0xd3, 0x7e, 0x90, 0x00, 0x00,
>+  0x00, 0x00, 0x2d, 0x9e, 0x93, 0xa0, 0x00, 0x00, 0x00, 0x00, 0x2e, 0xb3,
>+  0x60, 0x90, 0x00, 0x00, 0x00, 0x00, 0x2f, 0x7e, 0x75, 0xa0, 0x00, 0x00,
>+  0x00, 0x00, 0x30, 0x93, 0x42, 0x90, 0x00, 0x00, 0x00, 0x00, 0x31, 0x67,
>+  0x92, 0x20, 0x00, 0x00, 0x00, 0x00, 0x32, 0x73, 0x24, 0x90, 0x00, 0x00,
>+  0x00, 0x00, 0x33, 0x47, 0x74, 0x20, 0x00, 0x00, 0x00, 0x00, 0x34, 0x53,
>+  0x06, 0x90, 0x00, 0x00, 0x00, 0x00, 0x35, 0x27, 0x56, 0x20, 0x00, 0x00,
>+  0x00, 0x00, 0x36, 0x32, 0xe8, 0x90, 0x00, 0x00, 0x00, 0x00, 0x37, 0x07,
>+  0x38, 0x20, 0x00, 0x00, 0x00, 0x00, 0x38, 0x1c, 0x05, 0x10, 0x00, 0x00,
>+  0x00, 0x00, 0x38, 0xe7, 0x1a, 0x20, 0x00, 0x00, 0x00, 0x00, 0x39, 0xfb,
>+  0xe7, 0x10, 0x00, 0x00, 0x00, 0x00, 0x3a, 0xc6, 0xfc, 0x20, 0x00, 0x00,
>+  0x00, 0x00, 0x3b, 0xdb, 0xc9, 0x10, 0x00, 0x00, 0x00, 0x00, 0x3c, 0xb0,
>+  0x18, 0xa0, 0x00, 0x00, 0x00, 0x00, 0x3d, 0xbb, 0xab, 0x10, 0x00, 0x00,
>+  0x00, 0x00, 0x3e, 0x8f, 0xfa, 0xa0, 0x00, 0x00, 0x00, 0x00, 0x3f, 0x9b,
>+  0x8d, 0x10, 0x00, 0x00, 0x00, 0x00, 0x40, 0x6f, 0xdc, 0xa0, 0x00, 0x00,
>+  0x00, 0x00, 0x41, 0x84, 0xa9, 0x90, 0x00, 0x00, 0x00, 0x00, 0x42, 0x4f,
>+  0xbe, 0xa0, 0x00, 0x00, 0x00, 0x00, 0x43, 0x64, 0x8b, 0x90, 0x00, 0x00,
>+  0x00, 0x00, 0x44, 0x2f, 0xa0, 0xa0, 0x00, 0x00, 0x00, 0x00, 0x45, 0x44,
>+  0x6d, 0x90, 0x00, 0x00, 0x00, 0x00, 0x45, 0xf3, 0xd3, 0x20, 0x00, 0x00,
>+  0x00, 0x00, 0x47, 0x2d, 0x8a, 0x10, 0x00, 0x00, 0x00, 0x00, 0x47, 0xd3,
>+  0xb5, 0x20, 0x00, 0x00, 0x00, 0x00, 0x49, 0x0d, 0x6c, 0x10, 0x00, 0x00,
>+  0x00, 0x00, 0x49, 0xb3, 0x97, 0x20, 0x00, 0x00, 0x00, 0x00, 0x4a, 0xed,
>+  0x4e, 0x10, 0x00, 0x00, 0x00, 0x00, 0x4b, 0x9c, 0xb3, 0xa0, 0x00, 0x00,
>+  0x00, 0x00, 0x4c, 0xd6, 0x6a, 0x90, 0x00, 0x00, 0x00, 0x00, 0x4d, 0x7c,
>+  0x95, 0xa0, 0x00, 0x00, 0x00, 0x00, 0x4e, 0xb6, 0x4c, 0x90, 0x00, 0x00,
>+  0x00, 0x00, 0x4f, 0x5c, 0x77, 0xa0, 0x00, 0x00, 0x00, 0x00, 0x50, 0x96,
>+  0x2e, 0x90, 0x00, 0x00, 0x00, 0x00, 0x51, 0x3c, 0x59, 0xa0, 0x00, 0x00,
>+  0x00, 0x00, 0x52, 0x76, 0x10, 0x90, 0x00, 0x00, 0x00, 0x00, 0x53, 0x1c,
>+  0x3b, 0xa0, 0x00, 0x00, 0x00, 0x00, 0x54, 0x55, 0xf2, 0x90, 0x00, 0x00,
>+  0x00, 0x00, 0x54, 0xfc, 0x1d, 0xa0, 0x00, 0x00, 0x00, 0x00, 0x56, 0x35,
>+  0xd4, 0x90, 0x00, 0x00, 0x00, 0x00, 0x56, 0xe5, 0x3a, 0x20, 0x00, 0x00,
>+  0x00, 0x00, 0x58, 0x1e, 0xf1, 0x10, 0x00, 0x00, 0x00, 0x00, 0x58, 0xc5,
>+  0x1c, 0x20, 0x00, 0x00, 0x00, 0x00, 0x59, 0xfe, 0xd3, 0x10, 0x00, 0x00,
>+  0x00, 0x00, 0x5a, 0xa4, 0xfe, 0x20, 0x00, 0x00, 0x00, 0x00, 0x5b, 0xde,
>+  0xb5, 0x10, 0x00, 0x00, 0x00, 0x00, 0x5c, 0x84, 0xe0, 0x20, 0x00, 0x00,
>+  0x00, 0x00, 0x5d, 0xbe, 0x97, 0x10, 0x00, 0x00, 0x00, 0x00, 0x5e, 0x64,
>+  0xc2, 0x20, 0x00, 0x00, 0x00, 0x00, 0x5f, 0x9e, 0x79, 0x10, 0x00, 0x00,
>+  0x00, 0x00, 0x60, 0x4d, 0xde, 0xa0, 0x00, 0x00, 0x00, 0x00, 0x61, 0x87,
>+  0x95, 0x90, 0x00, 0x00, 0x00, 0x00, 0x62, 0x2d, 0xc0, 0xa0, 0x00, 0x00,
>+  0x00, 0x00, 0x63, 0x67, 0x77, 0x90, 0x00, 0x00, 0x00, 0x00, 0x64, 0x0d,
>+  0xa2, 0xa0, 0x00, 0x00, 0x00, 0x00, 0x65, 0x47, 0x59, 0x90, 0x00, 0x00,
>+  0x00, 0x00, 0x65, 0xed, 0x84, 0xa0, 0x00, 0x00, 0x00, 0x00, 0x67, 0x27,
>+  0x3b, 0x90, 0x00, 0x00, 0x00, 0x00, 0x67, 0xcd, 0x66, 0xa0, 0x00, 0x00,
>+  0x00, 0x00, 0x69, 0x07, 0x1d, 0x90, 0x00, 0x00, 0x00, 0x00, 0x69, 0xad,
>+  0x48, 0xa0, 0x00, 0x00, 0x00, 0x00, 0x6a, 0xe6, 0xff, 0x90, 0x00, 0x00,
>+  0x00, 0x00, 0x6b, 0x96, 0x65, 0x20, 0x00, 0x00, 0x00, 0x00, 0x6c, 0xd0,
>+  0x1c, 0x10, 0x00, 0x00, 0x00, 0x00, 0x6d, 0x76, 0x47, 0x20, 0x00, 0x00,
>+  0x00, 0x00, 0x6e, 0xaf, 0xfe, 0x10, 0x00, 0x00, 0x00, 0x00, 0x6f, 0x56,
>+  0x29, 0x20, 0x00, 0x00, 0x00, 0x00, 0x70, 0x8f, 0xe0, 0x10, 0x00, 0x00,
>+  0x00, 0x00, 0x71, 0x36, 0x0b, 0x20, 0x00, 0x00, 0x00, 0x00, 0x72, 0x6f,
>+  0xc2, 0x10, 0x00, 0x00, 0x00, 0x00, 0x73, 0x15, 0xed, 0x20, 0x00, 0x00,
>+  0x00, 0x00, 0x74, 0x4f, 0xa4, 0x10, 0x00, 0x00, 0x00, 0x00, 0x74, 0xff,
>+  0x09, 0xa0, 0x00, 0x00, 0x00, 0x00, 0x76, 0x38, 0xc0, 0x90, 0x00, 0x00,
>+  0x00, 0x00, 0x76, 0xde, 0xeb, 0xa0, 0x00, 0x00, 0x00, 0x00, 0x78, 0x18,
>+  0xa2, 0x90, 0x00, 0x00, 0x00, 0x00, 0x78, 0xbe, 0xcd, 0xa0, 0x00, 0x00,
>+  0x00, 0x00, 0x79, 0xf8, 0x84, 0x90, 0x00, 0x00, 0x00, 0x00, 0x7a, 0x9e,
>+  0xaf, 0xa0, 0x00, 0x00, 0x00, 0x00, 0x7b, 0xd8, 0x66, 0x90, 0x00, 0x00,
>+  0x00, 0x00, 0x7c, 0x7e, 0x91, 0xa0, 0x00, 0x00, 0x00, 0x00, 0x7d, 0xb8,
>+  0x48, 0x90, 0x00, 0x00, 0x00, 0x00, 0x7e, 0x5e, 0x73, 0xa0, 0x00, 0x00,
>+  0x00, 0x00, 0x7f, 0x98, 0x2a, 0x90, 0x00, 0x02, 0x01, 0x02, 0x01, 0x02,
>+  0x03, 0x04, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01,
>+  0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01,
>+  0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01,
>+  0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01,
>+  0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01,
>+  0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01,
>+  0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01,
>+  0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01,
>+  0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01,
>+  0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01,
>+  0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01,
>+  0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01,
>+  0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01,
>+  0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01,
>+  0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01,
>+  0x02, 0xff, 0xff, 0x91, 0x26, 0x00, 0x00, 0xff, 0xff, 0x9d, 0x90, 0x01,
>+  0x04, 0xff, 0xff, 0x8f, 0x80, 0x00, 0x08, 0xff, 0xff, 0x9d, 0x90, 0x01,
>+  0x0c, 0xff, 0xff, 0x9d, 0x90, 0x01, 0x10, 0x4c, 0x4d, 0x54, 0x00, 0x50,
>+  0x44, 0x54, 0x00, 0x50, 0x53, 0x54, 0x00, 0x50, 0x57, 0x54, 0x00, 0x50,
>+  0x50, 0x54, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00,
>+  0x01, 0x0a, 0x50, 0x53, 0x54, 0x38, 0x50, 0x44, 0x54, 0x2c, 0x4d, 0x33,
>+  0x2e, 0x32, 0x2e, 0x30, 0x2c, 0x4d, 0x31, 0x31, 0x2e, 0x31, 0x2e, 0x30,
>+  0x0a
>+};
>+unsigned int America_Los_Angeles_len = 2845;
>+unsigned char America_New_York[] = {
>+  0x54, 0x5a, 0x69, 0x66, 0x32, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
>+  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x05,
>+  0x00, 0x00, 0x00, 0x05, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0xec,
>+  0x00, 0x00, 0x00, 0x05, 0x00, 0x00, 0x00, 0x14, 0x80, 0x00, 0x00, 0x00,
>+  0x9e, 0xa6, 0x1e, 0x70, 0x9f, 0xba, 0xeb, 0x60, 0xa0, 0x86, 0x00, 0x70,
>+  0xa1, 0x9a, 0xcd, 0x60, 0xa2, 0x65, 0xe2, 0x70, 0xa3, 0x83, 0xe9, 0xe0,
>+  0xa4, 0x6a, 0xae, 0x70, 0xa5, 0x35, 0xa7, 0x60, 0xa6, 0x53, 0xca, 0xf0,
>+  0xa7, 0x15, 0x89, 0x60, 0xa8, 0x33, 0xac, 0xf0, 0xa8, 0xfe, 0xa5, 0xe0,
>+  0xaa, 0x13, 0x8e, 0xf0, 0xaa, 0xde, 0x87, 0xe0, 0xab, 0xf3, 0x70, 0xf0,
>+  0xac, 0xbe, 0x69, 0xe0, 0xad, 0xd3, 0x52, 0xf0, 0xae, 0x9e, 0x4b, 0xe0,
>+  0xaf, 0xb3, 0x34, 0xf0, 0xb0, 0x7e, 0x2d, 0xe0, 0xb1, 0x9c, 0x51, 0x70,
>+  0xb2, 0x67, 0x4a, 0x60, 0xb3, 0x7c, 0x33, 0x70, 0xb4, 0x47, 0x2c, 0x60,
>+  0xb5, 0x5c, 0x15, 0x70, 0xb6, 0x27, 0x0e, 0x60, 0xb7, 0x3b, 0xf7, 0x70,
>+  0xb8, 0x06, 0xf0, 0x60, 0xb9, 0x1b, 0xd9, 0x70, 0xb9, 0xe6, 0xd2, 0x60,
>+  0xbb, 0x04, 0xf5, 0xf0, 0xbb, 0xc6, 0xb4, 0x60, 0xbc, 0xe4, 0xd7, 0xf0,
>+  0xbd, 0xaf, 0xd0, 0xe0, 0xbe, 0xc4, 0xb9, 0xf0, 0xbf, 0x8f, 0xb2, 0xe0,
>+  0xc0, 0xa4, 0x9b, 0xf0, 0xc1, 0x6f, 0x94, 0xe0, 0xc2, 0x84, 0x7d, 0xf0,
>+  0xc3, 0x4f, 0x76, 0xe0, 0xc4, 0x64, 0x5f, 0xf0, 0xc5, 0x2f, 0x58, 0xe0,
>+  0xc6, 0x4d, 0x7c, 0x70, 0xc7, 0x0f, 0x3a, 0xe0, 0xc8, 0x2d, 0x5e, 0x70,
>+  0xc8, 0xf8, 0x57, 0x60, 0xca, 0x0d, 0x40, 0x70, 0xca, 0xd8, 0x39, 0x60,
>+  0xcb, 0x88, 0xf0, 0x70, 0xd2, 0x23, 0xf4, 0x70, 0xd2, 0x60, 0xfb, 0xe0,
>+  0xd3, 0x75, 0xe4, 0xf0, 0xd4, 0x40, 0xdd, 0xe0, 0xd5, 0x55, 0xc6, 0xf0,
>+  0xd6, 0x20, 0xbf, 0xe0, 0xd7, 0x35, 0xa8, 0xf0, 0xd8, 0x00, 0xa1, 0xe0,
>+  0xd9, 0x15, 0x8a, 0xf0, 0xd9, 0xe0, 0x83, 0xe0, 0xda, 0xfe, 0xa7, 0x70,
>+  0xdb, 0xc0, 0x65, 0xe0, 0xdc, 0xde, 0x89, 0x70, 0xdd, 0xa9, 0x82, 0x60,
>+  0xde, 0xbe, 0x6b, 0x70, 0xdf, 0x89, 0x64, 0x60, 0xe0, 0x9e, 0x4d, 0x70,
>+  0xe1, 0x69, 0x46, 0x60, 0xe2, 0x7e, 0x2f, 0x70, 0xe3, 0x49, 0x28, 0x60,
>+  0xe4, 0x5e, 0x11, 0x70, 0xe5, 0x57, 0x2e, 0xe0, 0xe6, 0x47, 0x2d, 0xf0,
>+  0xe7, 0x37, 0x10, 0xe0, 0xe8, 0x27, 0x0f, 0xf0, 0xe9, 0x16, 0xf2, 0xe0,
>+  0xea, 0x06, 0xf1, 0xf0, 0xea, 0xf6, 0xd4, 0xe0, 0xeb, 0xe6, 0xd3, 0xf0,
>+  0xec, 0xd6, 0xb6, 0xe0, 0xed, 0xc6, 0xb5, 0xf0, 0xee, 0xbf, 0xd3, 0x60,
>+  0xef, 0xaf, 0xd2, 0x70, 0xf0, 0x9f, 0xb5, 0x60, 0xf1, 0x8f, 0xb4, 0x70,
>+  0xf2, 0x7f, 0x97, 0x60, 0xf3, 0x6f, 0x96, 0x70, 0xf4, 0x5f, 0x79, 0x60,
>+  0xf5, 0x4f, 0x78, 0x70, 0xf6, 0x3f, 0x5b, 0x60, 0xf7, 0x2f, 0x5a, 0x70,
>+  0xf8, 0x28, 0x77, 0xe0, 0xf9, 0x0f, 0x3c, 0x70, 0xfa, 0x08, 0x59, 0xe0,
>+  0xfa, 0xf8, 0x58, 0xf0, 0xfb, 0xe8, 0x3b, 0xe0, 0xfc, 0xd8, 0x3a, 0xf0,
>+  0xfd, 0xc8, 0x1d, 0xe0, 0xfe, 0xb8, 0x1c, 0xf0, 0xff, 0xa7, 0xff, 0xe0,
>+  0x00, 0x97, 0xfe, 0xf0, 0x01, 0x87, 0xe1, 0xe0, 0x02, 0x77, 0xe0, 0xf0,
>+  0x03, 0x70, 0xfe, 0x60, 0x04, 0x60, 0xfd, 0x70, 0x05, 0x50, 0xe0, 0x60,
>+  0x06, 0x40, 0xdf, 0x70, 0x07, 0x30, 0xc2, 0x60, 0x07, 0x8d, 0x19, 0x70,
>+  0x09, 0x10, 0xa4, 0x60, 0x09, 0xad, 0x94, 0xf0, 0x0a, 0xf0, 0x86, 0x60,
>+  0x0b, 0xe0, 0x85, 0x70, 0x0c, 0xd9, 0xa2, 0xe0, 0x0d, 0xc0, 0x67, 0x70,
>+  0x0e, 0xb9, 0x84, 0xe0, 0x0f, 0xa9, 0x83, 0xf0, 0x10, 0x99, 0x66, 0xe0,
>+  0x11, 0x89, 0x65, 0xf0, 0x12, 0x79, 0x48, 0xe0, 0x13, 0x69, 0x47, 0xf0,
>+  0x14, 0x59, 0x2a, 0xe0, 0x15, 0x49, 0x29, 0xf0, 0x16, 0x39, 0x0c, 0xe0,
>+  0x17, 0x29, 0x0b, 0xf0, 0x18, 0x22, 0x29, 0x60, 0x19, 0x08, 0xed, 0xf0,
>+  0x1a, 0x02, 0x0b, 0x60, 0x1a, 0xf2, 0x0a, 0x70, 0x1b, 0xe1, 0xed, 0x60,
>+  0x1c, 0xd1, 0xec, 0x70, 0x1d, 0xc1, 0xcf, 0x60, 0x1e, 0xb1, 0xce, 0x70,
>+  0x1f, 0xa1, 0xb1, 0x60, 0x20, 0x76, 0x00, 0xf0, 0x21, 0x81, 0x93, 0x60,
>+  0x22, 0x55, 0xe2, 0xf0, 0x23, 0x6a, 0xaf, 0xe0, 0x24, 0x35, 0xc4, 0xf0,
>+  0x25, 0x4a, 0x91, 0xe0, 0x26, 0x15, 0xa6, 0xf0, 0x27, 0x2a, 0x73, 0xe0,
>+  0x27, 0xfe, 0xc3, 0x70, 0x29, 0x0a, 0x55, 0xe0, 0x29, 0xde, 0xa5, 0x70,
>+  0x2a, 0xea, 0x37, 0xe0, 0x2b, 0xbe, 0x87, 0x70, 0x2c, 0xd3, 0x54, 0x60,
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>+  0x29, 0x0a, 0x55, 0xe0, 0x00, 0x00, 0x00, 0x00, 0x29, 0xde, 0xa5, 0x70,
>+  0x00, 0x00, 0x00, 0x00, 0x2a, 0xea, 0x37, 0xe0, 0x00, 0x00, 0x00, 0x00,
>+  0x2b, 0xbe, 0x87, 0x70, 0x00, 0x00, 0x00, 0x00, 0x2c, 0xd3, 0x54, 0x60,
>+  0x00, 0x00, 0x00, 0x00, 0x2d, 0x9e, 0x69, 0x70, 0x00, 0x00, 0x00, 0x00,
>+  0x2e, 0xb3, 0x36, 0x60, 0x00, 0x00, 0x00, 0x00, 0x2f, 0x7e, 0x4b, 0x70,
>+  0x00, 0x00, 0x00, 0x00, 0x30, 0x93, 0x18, 0x60, 0x00, 0x00, 0x00, 0x00,
>+  0x31, 0x67, 0x67, 0xf0, 0x00, 0x00, 0x00, 0x00, 0x32, 0x72, 0xfa, 0x60,
>+  0x00, 0x00, 0x00, 0x00, 0x33, 0x47, 0x49, 0xf0, 0x00, 0x00, 0x00, 0x00,
>+  0x34, 0x52, 0xdc, 0x60, 0x00, 0x00, 0x00, 0x00, 0x35, 0x27, 0x2b, 0xf0,
>+  0x00, 0x00, 0x00, 0x00, 0x36, 0x32, 0xbe, 0x60, 0x00, 0x00, 0x00, 0x00,
>+  0x37, 0x07, 0x0d, 0xf0, 0x00, 0x00, 0x00, 0x00, 0x38, 0x1b, 0xda, 0xe0,
>+  0x00, 0x00, 0x00, 0x00, 0x38, 0xe6, 0xef, 0xf0, 0x00, 0x00, 0x00, 0x00,
>+  0x39, 0xfb, 0xbc, 0xe0, 0x00, 0x00, 0x00, 0x00, 0x3a, 0xc6, 0xd1, 0xf0,
>+  0x00, 0x00, 0x00, 0x00, 0x3b, 0xdb, 0x9e, 0xe0, 0x00, 0x00, 0x00, 0x00,
>+  0x3c, 0xaf, 0xee, 0x70, 0x00, 0x00, 0x00, 0x00, 0x3d, 0xbb, 0x80, 0xe0,
>+  0x00, 0x00, 0x00, 0x00, 0x3e, 0x8f, 0xd0, 0x70, 0x00, 0x00, 0x00, 0x00,
>+  0x3f, 0x9b, 0x62, 0xe0, 0x00, 0x00, 0x00, 0x00, 0x40, 0x6f, 0xb2, 0x70,
>+  0x00, 0x00, 0x00, 0x00, 0x41, 0x84, 0x7f, 0x60, 0x00, 0x00, 0x00, 0x00,
>+  0x42, 0x4f, 0x94, 0x70, 0x00, 0x00, 0x00, 0x00, 0x43, 0x64, 0x61, 0x60,
>+  0x00, 0x00, 0x00, 0x00, 0x44, 0x2f, 0x76, 0x70, 0x00, 0x00, 0x00, 0x00,
>+  0x45, 0x44, 0x43, 0x60, 0x00, 0x00, 0x00, 0x00, 0x45, 0xf3, 0xa8, 0xf0,
>+  0x00, 0x00, 0x00, 0x00, 0x47, 0x2d, 0x5f, 0xe0, 0x00, 0x00, 0x00, 0x00,
>+  0x47, 0xd3, 0x8a, 0xf0, 0x00, 0x00, 0x00, 0x00, 0x49, 0x0d, 0x41, 0xe0,
>+  0x00, 0x00, 0x00, 0x00, 0x49, 0xb3, 0x6c, 0xf0, 0x00, 0x00, 0x00, 0x00,
>+  0x4a, 0xed, 0x23, 0xe0, 0x00, 0x00, 0x00, 0x00, 0x4b, 0x9c, 0x89, 0x70,
>+  0x00, 0x00, 0x00, 0x00, 0x4c, 0xd6, 0x40, 0x60, 0x00, 0x00, 0x00, 0x00,
>+  0x4d, 0x7c, 0x6b, 0x70, 0x00, 0x00, 0x00, 0x00, 0x4e, 0xb6, 0x22, 0x60,
>+  0x00, 0x00, 0x00, 0x00, 0x4f, 0x5c, 0x4d, 0x70, 0x00, 0x00, 0x00, 0x00,
>+  0x50, 0x96, 0x04, 0x60, 0x00, 0x00, 0x00, 0x00, 0x51, 0x3c, 0x2f, 0x70,
>+  0x00, 0x00, 0x00, 0x00, 0x52, 0x75, 0xe6, 0x60, 0x00, 0x00, 0x00, 0x00,
>+  0x53, 0x1c, 0x11, 0x70, 0x00, 0x00, 0x00, 0x00, 0x54, 0x55, 0xc8, 0x60,
>+  0x00, 0x00, 0x00, 0x00, 0x54, 0xfb, 0xf3, 0x70, 0x00, 0x00, 0x00, 0x00,
>+  0x56, 0x35, 0xaa, 0x60, 0x00, 0x00, 0x00, 0x00, 0x56, 0xe5, 0x0f, 0xf0,
>+  0x00, 0x00, 0x00, 0x00, 0x58, 0x1e, 0xc6, 0xe0, 0x00, 0x00, 0x00, 0x00,
>+  0x58, 0xc4, 0xf1, 0xf0, 0x00, 0x00, 0x00, 0x00, 0x59, 0xfe, 0xa8, 0xe0,
>+  0x00, 0x00, 0x00, 0x00, 0x5a, 0xa4, 0xd3, 0xf0, 0x00, 0x00, 0x00, 0x00,
>+  0x5b, 0xde, 0x8a, 0xe0, 0x00, 0x00, 0x00, 0x00, 0x5c, 0x84, 0xb5, 0xf0,
>+  0x00, 0x00, 0x00, 0x00, 0x5d, 0xbe, 0x6c, 0xe0, 0x00, 0x00, 0x00, 0x00,
>+  0x5e, 0x64, 0x97, 0xf0, 0x00, 0x00, 0x00, 0x00, 0x5f, 0x9e, 0x4e, 0xe0,
>+  0x00, 0x00, 0x00, 0x00, 0x60, 0x4d, 0xb4, 0x70, 0x00, 0x00, 0x00, 0x00,
>+  0x61, 0x87, 0x6b, 0x60, 0x00, 0x00, 0x00, 0x00, 0x62, 0x2d, 0x96, 0x70,
>+  0x00, 0x00, 0x00, 0x00, 0x63, 0x67, 0x4d, 0x60, 0x00, 0x00, 0x00, 0x00,
>+  0x64, 0x0d, 0x78, 0x70, 0x00, 0x00, 0x00, 0x00, 0x65, 0x47, 0x2f, 0x60,
>+  0x00, 0x00, 0x00, 0x00, 0x65, 0xed, 0x5a, 0x70, 0x00, 0x00, 0x00, 0x00,
>+  0x67, 0x27, 0x11, 0x60, 0x00, 0x00, 0x00, 0x00, 0x67, 0xcd, 0x3c, 0x70,
>+  0x00, 0x00, 0x00, 0x00, 0x69, 0x06, 0xf3, 0x60, 0x00, 0x00, 0x00, 0x00,
>+  0x69, 0xad, 0x1e, 0x70, 0x00, 0x00, 0x00, 0x00, 0x6a, 0xe6, 0xd5, 0x60,
>+  0x00, 0x00, 0x00, 0x00, 0x6b, 0x96, 0x3a, 0xf0, 0x00, 0x00, 0x00, 0x00,
>+  0x6c, 0xcf, 0xf1, 0xe0, 0x00, 0x00, 0x00, 0x00, 0x6d, 0x76, 0x1c, 0xf0,
>+  0x00, 0x00, 0x00, 0x00, 0x6e, 0xaf, 0xd3, 0xe0, 0x00, 0x00, 0x00, 0x00,
>+  0x6f, 0x55, 0xfe, 0xf0, 0x00, 0x00, 0x00, 0x00, 0x70, 0x8f, 0xb5, 0xe0,
>+  0x00, 0x00, 0x00, 0x00, 0x71, 0x35, 0xe0, 0xf0, 0x00, 0x00, 0x00, 0x00,
>+  0x72, 0x6f, 0x97, 0xe0, 0x00, 0x00, 0x00, 0x00, 0x73, 0x15, 0xc2, 0xf0,
>+  0x00, 0x00, 0x00, 0x00, 0x74, 0x4f, 0x79, 0xe0, 0x00, 0x00, 0x00, 0x00,
>+  0x74, 0xfe, 0xdf, 0x70, 0x00, 0x00, 0x00, 0x00, 0x76, 0x38, 0x96, 0x60,
>+  0x00, 0x00, 0x00, 0x00, 0x76, 0xde, 0xc1, 0x70, 0x00, 0x00, 0x00, 0x00,
>+  0x78, 0x18, 0x78, 0x60, 0x00, 0x00, 0x00, 0x00, 0x78, 0xbe, 0xa3, 0x70,
>+  0x00, 0x00, 0x00, 0x00, 0x79, 0xf8, 0x5a, 0x60, 0x00, 0x00, 0x00, 0x00,
>+  0x7a, 0x9e, 0x85, 0x70, 0x00, 0x00, 0x00, 0x00, 0x7b, 0xd8, 0x3c, 0x60,
>+  0x00, 0x00, 0x00, 0x00, 0x7c, 0x7e, 0x67, 0x70, 0x00, 0x00, 0x00, 0x00,
>+  0x7d, 0xb8, 0x1e, 0x60, 0x00, 0x00, 0x00, 0x00, 0x7e, 0x5e, 0x49, 0x70,
>+  0x00, 0x00, 0x00, 0x00, 0x7f, 0x98, 0x00, 0x60, 0x00, 0x02, 0x01, 0x02,
>+  0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02,
>+  0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02,
>+  0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02,
>+  0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x03, 0x04,
>+  0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01,
>+  0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01,
>+  0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01,
>+  0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01,
>+  0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01,
>+  0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01,
>+  0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01,
>+  0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01,
>+  0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01,
>+  0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01,
>+  0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01,
>+  0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01,
>+  0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01,
>+  0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01,
>+  0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01,
>+  0x02, 0x01, 0x02, 0x01, 0x02, 0xff, 0xff, 0xba, 0x9e, 0x00, 0x00, 0xff,
>+  0xff, 0xc7, 0xc0, 0x01, 0x04, 0xff, 0xff, 0xb9, 0xb0, 0x00, 0x08, 0xff,
>+  0xff, 0xc7, 0xc0, 0x01, 0x0c, 0xff, 0xff, 0xc7, 0xc0, 0x01, 0x10, 0x4c,
>+  0x4d, 0x54, 0x00, 0x45, 0x44, 0x54, 0x00, 0x45, 0x53, 0x54, 0x00, 0x45,
>+  0x57, 0x54, 0x00, 0x45, 0x50, 0x54, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01,
>+  0x00, 0x00, 0x00, 0x00, 0x01, 0x0a, 0x45, 0x53, 0x54, 0x35, 0x45, 0x44,
>+  0x54, 0x2c, 0x4d, 0x33, 0x2e, 0x32, 0x2e, 0x30, 0x2c, 0x4d, 0x31, 0x31,
>+  0x2e, 0x31, 0x2e, 0x30, 0x0a
>+};
>+unsigned int America_New_York_len = 3545;
>+unsigned char Australia_Sydney[] = {
>+  0x54, 0x5a, 0x69, 0x66, 0x32, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
>+  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x05,
>+  0x00, 0x00, 0x00, 0x05, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x8e,
>+  0x00, 0x00, 0x00, 0x05, 0x00, 0x00, 0x00, 0x0e, 0x80, 0x00, 0x00, 0x00,
>+  0x9c, 0x4e, 0xa6, 0x9c, 0x9c, 0xbc, 0x20, 0xf0, 0xcb, 0x54, 0xb3, 0x00,
>+  0xcb, 0xc7, 0x57, 0x70, 0xcc, 0xb7, 0x56, 0x80, 0xcd, 0xa7, 0x39, 0x70,
>+  0xce, 0xa0, 0x73, 0x00, 0xcf, 0x87, 0x1b, 0x70, 0x03, 0x70, 0x39, 0x80,
>+  0x04, 0x0d, 0x1c, 0x00, 0x05, 0x50, 0x1b, 0x80, 0x05, 0xf6, 0x38, 0x80,
>+  0x07, 0x2f, 0xfd, 0x80, 0x07, 0xd6, 0x1a, 0x80, 0x09, 0x0f, 0xdf, 0x80,
>+  0x09, 0xb5, 0xfc, 0x80, 0x0a, 0xef, 0xc1, 0x80, 0x0b, 0x9f, 0x19, 0x00,
>+  0x0c, 0xd8, 0xde, 0x00, 0x0d, 0x7e, 0xfb, 0x00, 0x0e, 0xb8, 0xc0, 0x00,
>+  0x0f, 0x5e, 0xdd, 0x00, 0x10, 0x98, 0xa2, 0x00, 0x11, 0x3e, 0xbf, 0x00,
>+  0x12, 0x78, 0x84, 0x00, 0x13, 0x1e, 0xa1, 0x00, 0x14, 0x58, 0x66, 0x00,
>+  0x14, 0xfe, 0x83, 0x00, 0x16, 0x38, 0x48, 0x00, 0x17, 0x0c, 0x89, 0x80,
>+  0x18, 0x21, 0x64, 0x80, 0x18, 0xc7, 0x81, 0x80, 0x1a, 0x01, 0x46, 0x80,
>+  0x1a, 0xa7, 0x63, 0x80, 0x1b, 0xe1, 0x28, 0x80, 0x1c, 0x87, 0x45, 0x80,
>+  0x1d, 0xc1, 0x0a, 0x80, 0x1e, 0x79, 0x9c, 0x80, 0x1f, 0x97, 0xb2, 0x00,
>+  0x20, 0x59, 0x7e, 0x80, 0x21, 0x80, 0xce, 0x80, 0x22, 0x42, 0x9b, 0x00,
>+  0x23, 0x69, 0xeb, 0x00, 0x24, 0x22, 0x7d, 0x00, 0x25, 0x49, 0xcd, 0x00,
>+  0x25, 0xef, 0xea, 0x00, 0x27, 0x29, 0xaf, 0x00, 0x27, 0xcf, 0xcc, 0x00,
>+  0x29, 0x09, 0x91, 0x00, 0x29, 0xaf, 0xae, 0x00, 0x2a, 0xe9, 0x73, 0x00,
>+  0x2b, 0x98, 0xca, 0x80, 0x2c, 0xd2, 0x8f, 0x80, 0x2d, 0x78, 0xac, 0x80,
>+  0x2e, 0xb2, 0x71, 0x80, 0x2f, 0x58, 0x8e, 0x80, 0x30, 0x92, 0x53, 0x80,
>+  0x31, 0x5d, 0x5a, 0x80, 0x32, 0x72, 0x35, 0x80, 0x33, 0x3d, 0x3c, 0x80,
>+  0x34, 0x52, 0x17, 0x80, 0x35, 0x1d, 0x1e, 0x80, 0x36, 0x31, 0xf9, 0x80,
>+  0x36, 0xfd, 0x00, 0x80, 0x38, 0x1b, 0x16, 0x00, 0x38, 0xdc, 0xe2, 0x80,
>+  0x39, 0xa7, 0xe9, 0x80, 0x3a, 0xbc, 0xc4, 0x80, 0x3b, 0xda, 0xda, 0x00,
>+  0x3c, 0xa5, 0xe1, 0x00, 0x3d, 0xba, 0xbc, 0x00, 0x3e, 0x85, 0xc3, 0x00,
>+  0x3f, 0x9a, 0x9e, 0x00, 0x40, 0x65, 0xa5, 0x00, 0x41, 0x83, 0xba, 0x80,
>+  0x42, 0x45, 0x87, 0x00, 0x43, 0x63, 0x9c, 0x80, 0x44, 0x2e, 0xa3, 0x80,
>+  0x45, 0x43, 0x7e, 0x80, 0x46, 0x05, 0x4b, 0x00, 0x47, 0x23, 0x60, 0x80,
>+  0x47, 0xf7, 0xa2, 0x00, 0x48, 0xe7, 0x93, 0x00, 0x49, 0xd7, 0x84, 0x00,
>+  0x4a, 0xc7, 0x75, 0x00, 0x4b, 0xb7, 0x66, 0x00, 0x4c, 0xa7, 0x57, 0x00,
>+  0x4d, 0x97, 0x48, 0x00, 0x4e, 0x87, 0x39, 0x00, 0x4f, 0x77, 0x2a, 0x00,
>+  0x50, 0x70, 0x55, 0x80, 0x51, 0x60, 0x46, 0x80, 0x52, 0x50, 0x37, 0x80,
>+  0x53, 0x40, 0x28, 0x80, 0x54, 0x30, 0x19, 0x80, 0x55, 0x20, 0x0a, 0x80,
>+  0x56, 0x0f, 0xfb, 0x80, 0x56, 0xff, 0xec, 0x80, 0x57, 0xef, 0xdd, 0x80,
>+  0x58, 0xdf, 0xce, 0x80, 0x59, 0xcf, 0xbf, 0x80, 0x5a, 0xbf, 0xb0, 0x80,
>+  0x5b, 0xb8, 0xdc, 0x00, 0x5c, 0xa8, 0xcd, 0x00, 0x5d, 0x98, 0xbe, 0x00,
>+  0x5e, 0x88, 0xaf, 0x00, 0x5f, 0x78, 0xa0, 0x00, 0x60, 0x68, 0x91, 0x00,
>+  0x61, 0x58, 0x82, 0x00, 0x62, 0x48, 0x73, 0x00, 0x63, 0x38, 0x64, 0x00,
>+  0x64, 0x28, 0x55, 0x00, 0x65, 0x18, 0x46, 0x00, 0x66, 0x11, 0x71, 0x80,
>+  0x67, 0x01, 0x62, 0x80, 0x67, 0xf1, 0x53, 0x80, 0x68, 0xe1, 0x44, 0x80,
>+  0x69, 0xd1, 0x35, 0x80, 0x6a, 0xc1, 0x26, 0x80, 0x6b, 0xb1, 0x17, 0x80,
>+  0x6c, 0xa1, 0x08, 0x80, 0x6d, 0x90, 0xf9, 0x80, 0x6e, 0x80, 0xea, 0x80,
>+  0x6f, 0x70, 0xdb, 0x80, 0x70, 0x6a, 0x07, 0x00, 0x71, 0x59, 0xf8, 0x00,
>+  0x72, 0x49, 0xe9, 0x00, 0x73, 0x39, 0xda, 0x00, 0x74, 0x29, 0xcb, 0x00,
>+  0x75, 0x19, 0xbc, 0x00, 0x76, 0x09, 0xad, 0x00, 0x76, 0xf9, 0x9e, 0x00,
>+  0x77, 0xe9, 0x8f, 0x00, 0x78, 0xd9, 0x80, 0x00, 0x79, 0xc9, 0x71, 0x00,
>+  0x7a, 0xb9, 0x62, 0x00, 0x7b, 0xb2, 0x8d, 0x80, 0x7c, 0xa2, 0x7e, 0x80,
>+  0x7d, 0x92, 0x6f, 0x80, 0x7e, 0x82, 0x60, 0x80, 0x7f, 0x72, 0x51, 0x80,
>+  0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02, 0x03, 0x04, 0x03,
>+  0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03,
>+  0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03,
>+  0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03,
>+  0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03,
>+  0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03,
>+  0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03,
>+  0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03,
>+  0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03,
>+  0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03,
>+  0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03,
>+  0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x00, 0x00,
>+  0x8d, 0xc4, 0x00, 0x00, 0x00, 0x00, 0x9a, 0xb0, 0x01, 0x04, 0x00, 0x00,
>+  0x8c, 0xa0, 0x00, 0x09, 0x00, 0x00, 0x9a, 0xb0, 0x01, 0x04, 0x00, 0x00,
>+  0x8c, 0xa0, 0x00, 0x09, 0x4c, 0x4d, 0x54, 0x00, 0x41, 0x45, 0x44, 0x54,
>+  0x00, 0x41, 0x45, 0x53, 0x54, 0x00, 0x00, 0x00, 0x00, 0x01, 0x01, 0x00,
>+  0x00, 0x00, 0x00, 0x00, 0x54, 0x5a, 0x69, 0x66, 0x32, 0x00, 0x00, 0x00,
>+  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
>+  0x00, 0x00, 0x00, 0x05, 0x00, 0x00, 0x00, 0x05, 0x00, 0x00, 0x00, 0x00,
>+  0x00, 0x00, 0x00, 0x8f, 0x00, 0x00, 0x00, 0x05, 0x00, 0x00, 0x00, 0x0e,
>+  0xf8, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0xff, 0xff, 0xff, 0xff,
>+  0x73, 0x16, 0x7f, 0x3c, 0xff, 0xff, 0xff, 0xff, 0x9c, 0x4e, 0xa6, 0x9c,
>+  0xff, 0xff, 0xff, 0xff, 0x9c, 0xbc, 0x20, 0xf0, 0xff, 0xff, 0xff, 0xff,
>+  0xcb, 0x54, 0xb3, 0x00, 0xff, 0xff, 0xff, 0xff, 0xcb, 0xc7, 0x57, 0x70,
>+  0xff, 0xff, 0xff, 0xff, 0xcc, 0xb7, 0x56, 0x80, 0xff, 0xff, 0xff, 0xff,
>+  0xcd, 0xa7, 0x39, 0x70, 0xff, 0xff, 0xff, 0xff, 0xce, 0xa0, 0x73, 0x00,
>+  0xff, 0xff, 0xff, 0xff, 0xcf, 0x87, 0x1b, 0x70, 0x00, 0x00, 0x00, 0x00,
>+  0x03, 0x70, 0x39, 0x80, 0x00, 0x00, 0x00, 0x00, 0x04, 0x0d, 0x1c, 0x00,
>+  0x00, 0x00, 0x00, 0x00, 0x05, 0x50, 0x1b, 0x80, 0x00, 0x00, 0x00, 0x00,
>+  0x05, 0xf6, 0x38, 0x80, 0x00, 0x00, 0x00, 0x00, 0x07, 0x2f, 0xfd, 0x80,
>+  0x00, 0x00, 0x00, 0x00, 0x07, 0xd6, 0x1a, 0x80, 0x00, 0x00, 0x00, 0x00,
>+  0x09, 0x0f, 0xdf, 0x80, 0x00, 0x00, 0x00, 0x00, 0x09, 0xb5, 0xfc, 0x80,
>+  0x00, 0x00, 0x00, 0x00, 0x0a, 0xef, 0xc1, 0x80, 0x00, 0x00, 0x00, 0x00,
>+  0x0b, 0x9f, 0x19, 0x00, 0x00, 0x00, 0x00, 0x00, 0x0c, 0xd8, 0xde, 0x00,
>+  0x00, 0x00, 0x00, 0x00, 0x0d, 0x7e, 0xfb, 0x00, 0x00, 0x00, 0x00, 0x00,
>+  0x0e, 0xb8, 0xc0, 0x00, 0x00, 0x00, 0x00, 0x00, 0x0f, 0x5e, 0xdd, 0x00,
>+  0x00, 0x00, 0x00, 0x00, 0x10, 0x98, 0xa2, 0x00, 0x00, 0x00, 0x00, 0x00,
>+  0x11, 0x3e, 0xbf, 0x00, 0x00, 0x00, 0x00, 0x00, 0x12, 0x78, 0x84, 0x00,
>+  0x00, 0x00, 0x00, 0x00, 0x13, 0x1e, 0xa1, 0x00, 0x00, 0x00, 0x00, 0x00,
>+  0x14, 0x58, 0x66, 0x00, 0x00, 0x00, 0x00, 0x00, 0x14, 0xfe, 0x83, 0x00,
>+  0x00, 0x00, 0x00, 0x00, 0x16, 0x38, 0x48, 0x00, 0x00, 0x00, 0x00, 0x00,
>+  0x17, 0x0c, 0x89, 0x80, 0x00, 0x00, 0x00, 0x00, 0x18, 0x21, 0x64, 0x80,
>+  0x00, 0x00, 0x00, 0x00, 0x18, 0xc7, 0x81, 0x80, 0x00, 0x00, 0x00, 0x00,
>+  0x1a, 0x01, 0x46, 0x80, 0x00, 0x00, 0x00, 0x00, 0x1a, 0xa7, 0x63, 0x80,
>+  0x00, 0x00, 0x00, 0x00, 0x1b, 0xe1, 0x28, 0x80, 0x00, 0x00, 0x00, 0x00,
>+  0x1c, 0x87, 0x45, 0x80, 0x00, 0x00, 0x00, 0x00, 0x1d, 0xc1, 0x0a, 0x80,
>+  0x00, 0x00, 0x00, 0x00, 0x1e, 0x79, 0x9c, 0x80, 0x00, 0x00, 0x00, 0x00,
>+  0x1f, 0x97, 0xb2, 0x00, 0x00, 0x00, 0x00, 0x00, 0x20, 0x59, 0x7e, 0x80,
>+  0x00, 0x00, 0x00, 0x00, 0x21, 0x80, 0xce, 0x80, 0x00, 0x00, 0x00, 0x00,
>+  0x22, 0x42, 0x9b, 0x00, 0x00, 0x00, 0x00, 0x00, 0x23, 0x69, 0xeb, 0x00,
>+  0x00, 0x00, 0x00, 0x00, 0x24, 0x22, 0x7d, 0x00, 0x00, 0x00, 0x00, 0x00,
>+  0x25, 0x49, 0xcd, 0x00, 0x00, 0x00, 0x00, 0x00, 0x25, 0xef, 0xea, 0x00,
>+  0x00, 0x00, 0x00, 0x00, 0x27, 0x29, 0xaf, 0x00, 0x00, 0x00, 0x00, 0x00,
>+  0x27, 0xcf, 0xcc, 0x00, 0x00, 0x00, 0x00, 0x00, 0x29, 0x09, 0x91, 0x00,
>+  0x00, 0x00, 0x00, 0x00, 0x29, 0xaf, 0xae, 0x00, 0x00, 0x00, 0x00, 0x00,
>+  0x2a, 0xe9, 0x73, 0x00, 0x00, 0x00, 0x00, 0x00, 0x2b, 0x98, 0xca, 0x80,
>+  0x00, 0x00, 0x00, 0x00, 0x2c, 0xd2, 0x8f, 0x80, 0x00, 0x00, 0x00, 0x00,
>+  0x2d, 0x78, 0xac, 0x80, 0x00, 0x00, 0x00, 0x00, 0x2e, 0xb2, 0x71, 0x80,
>+  0x00, 0x00, 0x00, 0x00, 0x2f, 0x58, 0x8e, 0x80, 0x00, 0x00, 0x00, 0x00,
>+  0x30, 0x92, 0x53, 0x80, 0x00, 0x00, 0x00, 0x00, 0x31, 0x5d, 0x5a, 0x80,
>+  0x00, 0x00, 0x00, 0x00, 0x32, 0x72, 0x35, 0x80, 0x00, 0x00, 0x00, 0x00,
>+  0x33, 0x3d, 0x3c, 0x80, 0x00, 0x00, 0x00, 0x00, 0x34, 0x52, 0x17, 0x80,
>+  0x00, 0x00, 0x00, 0x00, 0x35, 0x1d, 0x1e, 0x80, 0x00, 0x00, 0x00, 0x00,
>+  0x36, 0x31, 0xf9, 0x80, 0x00, 0x00, 0x00, 0x00, 0x36, 0xfd, 0x00, 0x80,
>+  0x00, 0x00, 0x00, 0x00, 0x38, 0x1b, 0x16, 0x00, 0x00, 0x00, 0x00, 0x00,
>+  0x38, 0xdc, 0xe2, 0x80, 0x00, 0x00, 0x00, 0x00, 0x39, 0xa7, 0xe9, 0x80,
>+  0x00, 0x00, 0x00, 0x00, 0x3a, 0xbc, 0xc4, 0x80, 0x00, 0x00, 0x00, 0x00,
>+  0x3b, 0xda, 0xda, 0x00, 0x00, 0x00, 0x00, 0x00, 0x3c, 0xa5, 0xe1, 0x00,
>+  0x00, 0x00, 0x00, 0x00, 0x3d, 0xba, 0xbc, 0x00, 0x00, 0x00, 0x00, 0x00,
>+  0x3e, 0x85, 0xc3, 0x00, 0x00, 0x00, 0x00, 0x00, 0x3f, 0x9a, 0x9e, 0x00,
>+  0x00, 0x00, 0x00, 0x00, 0x40, 0x65, 0xa5, 0x00, 0x00, 0x00, 0x00, 0x00,
>+  0x41, 0x83, 0xba, 0x80, 0x00, 0x00, 0x00, 0x00, 0x42, 0x45, 0x87, 0x00,
>+  0x00, 0x00, 0x00, 0x00, 0x43, 0x63, 0x9c, 0x80, 0x00, 0x00, 0x00, 0x00,
>+  0x44, 0x2e, 0xa3, 0x80, 0x00, 0x00, 0x00, 0x00, 0x45, 0x43, 0x7e, 0x80,
>+  0x00, 0x00, 0x00, 0x00, 0x46, 0x05, 0x4b, 0x00, 0x00, 0x00, 0x00, 0x00,
>+  0x47, 0x23, 0x60, 0x80, 0x00, 0x00, 0x00, 0x00, 0x47, 0xf7, 0xa2, 0x00,
>+  0x00, 0x00, 0x00, 0x00, 0x48, 0xe7, 0x93, 0x00, 0x00, 0x00, 0x00, 0x00,
>+  0x49, 0xd7, 0x84, 0x00, 0x00, 0x00, 0x00, 0x00, 0x4a, 0xc7, 0x75, 0x00,
>+  0x00, 0x00, 0x00, 0x00, 0x4b, 0xb7, 0x66, 0x00, 0x00, 0x00, 0x00, 0x00,
>+  0x4c, 0xa7, 0x57, 0x00, 0x00, 0x00, 0x00, 0x00, 0x4d, 0x97, 0x48, 0x00,
>+  0x00, 0x00, 0x00, 0x00, 0x4e, 0x87, 0x39, 0x00, 0x00, 0x00, 0x00, 0x00,
>+  0x4f, 0x77, 0x2a, 0x00, 0x00, 0x00, 0x00, 0x00, 0x50, 0x70, 0x55, 0x80,
>+  0x00, 0x00, 0x00, 0x00, 0x51, 0x60, 0x46, 0x80, 0x00, 0x00, 0x00, 0x00,
>+  0x52, 0x50, 0x37, 0x80, 0x00, 0x00, 0x00, 0x00, 0x53, 0x40, 0x28, 0x80,
>+  0x00, 0x00, 0x00, 0x00, 0x54, 0x30, 0x19, 0x80, 0x00, 0x00, 0x00, 0x00,
>+  0x55, 0x20, 0x0a, 0x80, 0x00, 0x00, 0x00, 0x00, 0x56, 0x0f, 0xfb, 0x80,
>+  0x00, 0x00, 0x00, 0x00, 0x56, 0xff, 0xec, 0x80, 0x00, 0x00, 0x00, 0x00,
>+  0x57, 0xef, 0xdd, 0x80, 0x00, 0x00, 0x00, 0x00, 0x58, 0xdf, 0xce, 0x80,
>+  0x00, 0x00, 0x00, 0x00, 0x59, 0xcf, 0xbf, 0x80, 0x00, 0x00, 0x00, 0x00,
>+  0x5a, 0xbf, 0xb0, 0x80, 0x00, 0x00, 0x00, 0x00, 0x5b, 0xb8, 0xdc, 0x00,
>+  0x00, 0x00, 0x00, 0x00, 0x5c, 0xa8, 0xcd, 0x00, 0x00, 0x00, 0x00, 0x00,
>+  0x5d, 0x98, 0xbe, 0x00, 0x00, 0x00, 0x00, 0x00, 0x5e, 0x88, 0xaf, 0x00,
>+  0x00, 0x00, 0x00, 0x00, 0x5f, 0x78, 0xa0, 0x00, 0x00, 0x00, 0x00, 0x00,
>+  0x60, 0x68, 0x91, 0x00, 0x00, 0x00, 0x00, 0x00, 0x61, 0x58, 0x82, 0x00,
>+  0x00, 0x00, 0x00, 0x00, 0x62, 0x48, 0x73, 0x00, 0x00, 0x00, 0x00, 0x00,
>+  0x63, 0x38, 0x64, 0x00, 0x00, 0x00, 0x00, 0x00, 0x64, 0x28, 0x55, 0x00,
>+  0x00, 0x00, 0x00, 0x00, 0x65, 0x18, 0x46, 0x00, 0x00, 0x00, 0x00, 0x00,
>+  0x66, 0x11, 0x71, 0x80, 0x00, 0x00, 0x00, 0x00, 0x67, 0x01, 0x62, 0x80,
>+  0x00, 0x00, 0x00, 0x00, 0x67, 0xf1, 0x53, 0x80, 0x00, 0x00, 0x00, 0x00,
>+  0x68, 0xe1, 0x44, 0x80, 0x00, 0x00, 0x00, 0x00, 0x69, 0xd1, 0x35, 0x80,
>+  0x00, 0x00, 0x00, 0x00, 0x6a, 0xc1, 0x26, 0x80, 0x00, 0x00, 0x00, 0x00,
>+  0x6b, 0xb1, 0x17, 0x80, 0x00, 0x00, 0x00, 0x00, 0x6c, 0xa1, 0x08, 0x80,
>+  0x00, 0x00, 0x00, 0x00, 0x6d, 0x90, 0xf9, 0x80, 0x00, 0x00, 0x00, 0x00,
>+  0x6e, 0x80, 0xea, 0x80, 0x00, 0x00, 0x00, 0x00, 0x6f, 0x70, 0xdb, 0x80,
>+  0x00, 0x00, 0x00, 0x00, 0x70, 0x6a, 0x07, 0x00, 0x00, 0x00, 0x00, 0x00,
>+  0x71, 0x59, 0xf8, 0x00, 0x00, 0x00, 0x00, 0x00, 0x72, 0x49, 0xe9, 0x00,
>+  0x00, 0x00, 0x00, 0x00, 0x73, 0x39, 0xda, 0x00, 0x00, 0x00, 0x00, 0x00,
>+  0x74, 0x29, 0xcb, 0x00, 0x00, 0x00, 0x00, 0x00, 0x75, 0x19, 0xbc, 0x00,
>+  0x00, 0x00, 0x00, 0x00, 0x76, 0x09, 0xad, 0x00, 0x00, 0x00, 0x00, 0x00,
>+  0x76, 0xf9, 0x9e, 0x00, 0x00, 0x00, 0x00, 0x00, 0x77, 0xe9, 0x8f, 0x00,
>+  0x00, 0x00, 0x00, 0x00, 0x78, 0xd9, 0x80, 0x00, 0x00, 0x00, 0x00, 0x00,
>+  0x79, 0xc9, 0x71, 0x00, 0x00, 0x00, 0x00, 0x00, 0x7a, 0xb9, 0x62, 0x00,
>+  0x00, 0x00, 0x00, 0x00, 0x7b, 0xb2, 0x8d, 0x80, 0x00, 0x00, 0x00, 0x00,
>+  0x7c, 0xa2, 0x7e, 0x80, 0x00, 0x00, 0x00, 0x00, 0x7d, 0x92, 0x6f, 0x80,
>+  0x00, 0x00, 0x00, 0x00, 0x7e, 0x82, 0x60, 0x80, 0x00, 0x00, 0x00, 0x00,
>+  0x7f, 0x72, 0x51, 0x80, 0x00, 0x02, 0x01, 0x02, 0x01, 0x02, 0x01, 0x02,
>+  0x01, 0x02, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04,
>+  0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04,
>+  0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04,
>+  0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04,
>+  0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04,
>+  0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04,
>+  0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04,
>+  0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04,
>+  0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04,
>+  0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04,
>+  0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04, 0x03, 0x04,
>+  0x03, 0x04, 0x03, 0x00, 0x00, 0x8d, 0xc4, 0x00, 0x00, 0x00, 0x00, 0x9a,
>+  0xb0, 0x01, 0x04, 0x00, 0x00, 0x8c, 0xa0, 0x00, 0x09, 0x00, 0x00, 0x9a,
>+  0xb0, 0x01, 0x04, 0x00, 0x00, 0x8c, 0xa0, 0x00, 0x09, 0x4c, 0x4d, 0x54,
>+  0x00, 0x41, 0x45, 0x44, 0x54, 0x00, 0x41, 0x45, 0x53, 0x54, 0x00, 0x00,
>+  0x00, 0x00, 0x01, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x0a, 0x41, 0x45,
>+  0x53, 0x54, 0x2d, 0x31, 0x30, 0x41, 0x45, 0x44, 0x54, 0x2c, 0x4d, 0x31,
>+  0x30, 0x2e, 0x31, 0x2e, 0x30, 0x2c, 0x4d, 0x34, 0x2e, 0x31, 0x2e, 0x30,
>+  0x2f, 0x33, 0x0a
>+};
>+unsigned int Australia_Sydney_len = 2223;
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/time.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/time.cc
>new file mode 100644
>index 00000000000..71fd8ee6e74
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/time.cc
>@@ -0,0 +1,384 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+// The implementation of the absl::Time class, which is declared in
>+// //absl/time.h.
>+//
>+// The representation for an absl::Time is an absl::Duration offset from the
>+// epoch.  We use the traditional Unix epoch (1970-01-01 00:00:00 +0000)
>+// for convenience, but this is not exposed in the API and could be changed.
>+//
>+// NOTE: To keep type verbosity to a minimum, the following variable naming
>+// conventions are used throughout this file.
>+//
>+// cz: A cctz::time_zone
>+// tz: An absl::TimeZone
>+// cl: A cctz::time_zone::civil_lookup
>+// al: A cctz::time_zone::absolute_lookup
>+// cd: A cctz::civil_day
>+// cs: A cctz::civil_second
>+// bd: An absl::Time::Breakdown
>+
>+#include "absl/time/time.h"
>+
>+#include <cstring>
>+#include <ctime>
>+#include <limits>
>+
>+#include "absl/time/internal/cctz/include/cctz/civil_time.h"
>+#include "absl/time/internal/cctz/include/cctz/time_zone.h"
>+
>+namespace cctz = absl::time_internal::cctz;
>+namespace absl {
>+
>+namespace {
>+
>+inline cctz::time_point<cctz::seconds> unix_epoch() {
>+  return std::chrono::time_point_cast<cctz::seconds>(
>+      std::chrono::system_clock::from_time_t(0));
>+}
>+
>+// Floors d to the next unit boundary closer to negative infinity.
>+inline int64_t FloorToUnit(absl::Duration d, absl::Duration unit) {
>+  absl::Duration rem;
>+  int64_t q = absl::IDivDuration(d, unit, &rem);
>+  return (q > 0 ||
>+          rem >= ZeroDuration() ||
>+          q == std::numeric_limits<int64_t>::min()) ? q : q - 1;
>+}
>+
>+inline absl::Time::Breakdown InfiniteFutureBreakdown() {
>+  absl::Time::Breakdown bd;
>+  bd.year = std::numeric_limits<int64_t>::max();
>+  bd.month = 12;
>+  bd.day = 31;
>+  bd.hour = 23;
>+  bd.minute = 59;
>+  bd.second = 59;
>+  bd.subsecond = absl::InfiniteDuration();
>+  bd.weekday = 4;
>+  bd.yearday = 365;
>+  bd.offset = 0;
>+  bd.is_dst = false;
>+  bd.zone_abbr = "-00";
>+  return bd;
>+}
>+
>+inline Time::Breakdown InfinitePastBreakdown() {
>+  Time::Breakdown bd;
>+  bd.year = std::numeric_limits<int64_t>::min();
>+  bd.month = 1;
>+  bd.day = 1;
>+  bd.hour = 0;
>+  bd.minute = 0;
>+  bd.second = 0;
>+  bd.subsecond = -absl::InfiniteDuration();
>+  bd.weekday = 7;
>+  bd.yearday = 1;
>+  bd.offset = 0;
>+  bd.is_dst = false;
>+  bd.zone_abbr = "-00";
>+  return bd;
>+}
>+
>+inline absl::TimeConversion InfiniteFutureTimeConversion() {
>+  absl::TimeConversion tc;
>+  tc.pre = tc.trans = tc.post = absl::InfiniteFuture();
>+  tc.kind = absl::TimeConversion::UNIQUE;
>+  tc.normalized = true;
>+  return tc;
>+}
>+
>+inline TimeConversion InfinitePastTimeConversion() {
>+  absl::TimeConversion tc;
>+  tc.pre = tc.trans = tc.post = absl::InfinitePast();
>+  tc.kind = absl::TimeConversion::UNIQUE;
>+  tc.normalized = true;
>+  return tc;
>+}
>+
>+// Makes a Time from sec, overflowing to InfiniteFuture/InfinitePast as
>+// necessary. If sec is min/max, then consult cs+tz to check for overlow.
>+Time MakeTimeWithOverflow(const cctz::time_point<cctz::seconds>& sec,
>+                          const cctz::civil_second& cs,
>+                          const cctz::time_zone& tz,
>+                          bool* normalized = nullptr) {
>+  const auto max = cctz::time_point<cctz::seconds>::max();
>+  const auto min = cctz::time_point<cctz::seconds>::min();
>+  if (sec == max) {
>+    const auto al = tz.lookup(max);
>+    if (cs > al.cs) {
>+      if (normalized) *normalized = true;
>+      return absl::InfiniteFuture();
>+    }
>+  }
>+  if (sec == min) {
>+    const auto al = tz.lookup(min);
>+    if (cs < al.cs) {
>+      if (normalized) *normalized = true;
>+      return absl::InfinitePast();
>+    }
>+  }
>+  const auto hi = (sec - unix_epoch()).count();
>+  return time_internal::FromUnixDuration(time_internal::MakeDuration(hi));
>+}
>+
>+inline absl::TimeConversion::Kind MapKind(
>+    const cctz::time_zone::civil_lookup::civil_kind& kind) {
>+  switch (kind) {
>+    case cctz::time_zone::civil_lookup::UNIQUE:
>+      return absl::TimeConversion::UNIQUE;
>+    case cctz::time_zone::civil_lookup::SKIPPED:
>+      return absl::TimeConversion::SKIPPED;
>+    case cctz::time_zone::civil_lookup::REPEATED:
>+      return absl::TimeConversion::REPEATED;
>+  }
>+  return absl::TimeConversion::UNIQUE;
>+}
>+
>+// Returns Mon=1..Sun=7.
>+inline int MapWeekday(const cctz::weekday& wd) {
>+  switch (wd) {
>+    case cctz::weekday::monday:
>+      return 1;
>+    case cctz::weekday::tuesday:
>+      return 2;
>+    case cctz::weekday::wednesday:
>+      return 3;
>+    case cctz::weekday::thursday:
>+      return 4;
>+    case cctz::weekday::friday:
>+      return 5;
>+    case cctz::weekday::saturday:
>+      return 6;
>+    case cctz::weekday::sunday:
>+      return 7;
>+  }
>+  return 1;
>+}
>+
>+}  // namespace
>+
>+absl::Time::Breakdown Time::In(absl::TimeZone tz) const {
>+  if (*this == absl::InfiniteFuture()) return absl::InfiniteFutureBreakdown();
>+  if (*this == absl::InfinitePast()) return absl::InfinitePastBreakdown();
>+
>+  const auto tp = unix_epoch() + cctz::seconds(time_internal::GetRepHi(rep_));
>+  const auto al = cctz::time_zone(tz).lookup(tp);
>+  const auto cs = al.cs;
>+  const auto cd = cctz::civil_day(cs);
>+
>+  absl::Time::Breakdown bd;
>+  bd.year = cs.year();
>+  bd.month = cs.month();
>+  bd.day = cs.day();
>+  bd.hour = cs.hour();
>+  bd.minute = cs.minute();
>+  bd.second = cs.second();
>+  bd.subsecond = time_internal::MakeDuration(0, time_internal::GetRepLo(rep_));
>+  bd.weekday = MapWeekday(get_weekday(cd));
>+  bd.yearday = get_yearday(cd);
>+  bd.offset = al.offset;
>+  bd.is_dst = al.is_dst;
>+  bd.zone_abbr = al.abbr;
>+  return bd;
>+}
>+
>+absl::Time FromTM(const struct tm& tm, absl::TimeZone tz) {
>+  const auto cz = cctz::time_zone(tz);
>+  const auto cs =
>+      cctz::civil_second(tm.tm_year + 1900, tm.tm_mon + 1, tm.tm_mday,
>+                         tm.tm_hour, tm.tm_min, tm.tm_sec);
>+  const auto cl = cz.lookup(cs);
>+  const auto tp = tm.tm_isdst == 0 ? cl.post : cl.pre;
>+  return MakeTimeWithOverflow(tp, cs, cz);
>+}
>+
>+struct tm ToTM(absl::Time t, absl::TimeZone tz) {
>+  const absl::Time::Breakdown bd = t.In(tz);
>+  struct tm tm;
>+  std::memset(&tm, 0, sizeof(tm));
>+  tm.tm_sec = bd.second;
>+  tm.tm_min = bd.minute;
>+  tm.tm_hour = bd.hour;
>+  tm.tm_mday = bd.day;
>+  tm.tm_mon = bd.month - 1;
>+
>+  // Saturates tm.tm_year in cases of over/underflow, accounting for the fact
>+  // that tm.tm_year is years since 1900.
>+  if (bd.year < std::numeric_limits<int>::min() + 1900) {
>+    tm.tm_year = std::numeric_limits<int>::min();
>+  } else if (bd.year > std::numeric_limits<int>::max()) {
>+    tm.tm_year = std::numeric_limits<int>::max() - 1900;
>+  } else {
>+    tm.tm_year = static_cast<int>(bd.year - 1900);
>+  }
>+
>+  tm.tm_wday = bd.weekday % 7;
>+  tm.tm_yday = bd.yearday - 1;
>+  tm.tm_isdst = bd.is_dst ? 1 : 0;
>+
>+  return tm;
>+}
>+
>+//
>+// Factory functions.
>+//
>+
>+absl::TimeConversion ConvertDateTime(int64_t year, int mon, int day, int hour,
>+                                     int min, int sec, TimeZone tz) {
>+  // Avoids years that are too extreme for civil_second to normalize.
>+  if (year > 300000000000) return InfiniteFutureTimeConversion();
>+  if (year < -300000000000) return InfinitePastTimeConversion();
>+  const auto cz = cctz::time_zone(tz);
>+  const auto cs = cctz::civil_second(year, mon, day, hour, min, sec);
>+  absl::TimeConversion tc;
>+  tc.normalized = year != cs.year() || mon != cs.month() || day != cs.day() ||
>+                  hour != cs.hour() || min != cs.minute() || sec != cs.second();
>+  const auto cl = cz.lookup(cs);
>+  // Converts the civil_lookup struct to a TimeConversion.
>+  tc.pre = MakeTimeWithOverflow(cl.pre, cs, cz, &tc.normalized);
>+  tc.trans = MakeTimeWithOverflow(cl.trans, cs, cz, &tc.normalized);
>+  tc.post = MakeTimeWithOverflow(cl.post, cs, cz, &tc.normalized);
>+  tc.kind = MapKind(cl.kind);
>+  return tc;
>+}
>+
>+absl::Time FromDateTime(int64_t year, int mon, int day, int hour, int min,
>+                        int sec, TimeZone tz) {
>+  if (year > 300000000000) return InfiniteFuture();
>+  if (year < -300000000000) return InfinitePast();
>+  const auto cz = cctz::time_zone(tz);
>+  const auto cs = cctz::civil_second(year, mon, day, hour, min, sec);
>+  const auto cl = cz.lookup(cs);
>+  return MakeTimeWithOverflow(cl.pre, cs, cz);
>+}
>+
>+absl::Time TimeFromTimespec(timespec ts) {
>+  return time_internal::FromUnixDuration(absl::DurationFromTimespec(ts));
>+}
>+
>+absl::Time TimeFromTimeval(timeval tv) {
>+  return time_internal::FromUnixDuration(absl::DurationFromTimeval(tv));
>+}
>+
>+absl::Time FromUDate(double udate) {
>+  return time_internal::FromUnixDuration(absl::Milliseconds(udate));
>+}
>+
>+absl::Time FromUniversal(int64_t universal) {
>+  return absl::UniversalEpoch() + 100 * absl::Nanoseconds(universal);
>+}
>+
>+//
>+// Conversion to other time types.
>+//
>+
>+int64_t ToUnixNanos(Time t) {
>+  if (time_internal::GetRepHi(time_internal::ToUnixDuration(t)) >= 0 &&
>+      time_internal::GetRepHi(time_internal::ToUnixDuration(t)) >> 33 == 0) {
>+    return (time_internal::GetRepHi(time_internal::ToUnixDuration(t)) *
>+            1000 * 1000 * 1000) +
>+           (time_internal::GetRepLo(time_internal::ToUnixDuration(t)) / 4);
>+  }
>+  return FloorToUnit(time_internal::ToUnixDuration(t), absl::Nanoseconds(1));
>+}
>+
>+int64_t ToUnixMicros(Time t) {
>+  if (time_internal::GetRepHi(time_internal::ToUnixDuration(t)) >= 0 &&
>+      time_internal::GetRepHi(time_internal::ToUnixDuration(t)) >> 43 == 0) {
>+    return (time_internal::GetRepHi(time_internal::ToUnixDuration(t)) *
>+            1000 * 1000) +
>+           (time_internal::GetRepLo(time_internal::ToUnixDuration(t)) / 4000);
>+  }
>+  return FloorToUnit(time_internal::ToUnixDuration(t), absl::Microseconds(1));
>+}
>+
>+int64_t ToUnixMillis(Time t) {
>+  if (time_internal::GetRepHi(time_internal::ToUnixDuration(t)) >= 0 &&
>+      time_internal::GetRepHi(time_internal::ToUnixDuration(t)) >> 53 == 0) {
>+    return (time_internal::GetRepHi(time_internal::ToUnixDuration(t)) * 1000) +
>+           (time_internal::GetRepLo(time_internal::ToUnixDuration(t)) /
>+            (4000 * 1000));
>+  }
>+  return FloorToUnit(time_internal::ToUnixDuration(t), absl::Milliseconds(1));
>+}
>+
>+int64_t ToUnixSeconds(Time t) {
>+  return time_internal::GetRepHi(time_internal::ToUnixDuration(t));
>+}
>+
>+time_t ToTimeT(Time t) { return absl::ToTimespec(t).tv_sec; }
>+
>+timespec ToTimespec(Time t) {
>+  timespec ts;
>+  absl::Duration d = time_internal::ToUnixDuration(t);
>+  if (!time_internal::IsInfiniteDuration(d)) {
>+    ts.tv_sec = time_internal::GetRepHi(d);
>+    if (ts.tv_sec == time_internal::GetRepHi(d)) {  // no time_t narrowing
>+      ts.tv_nsec = time_internal::GetRepLo(d) / 4;  // floor
>+      return ts;
>+    }
>+  }
>+  if (d >= absl::ZeroDuration()) {
>+    ts.tv_sec = std::numeric_limits<time_t>::max();
>+    ts.tv_nsec = 1000 * 1000 * 1000 - 1;
>+  } else {
>+    ts.tv_sec = std::numeric_limits<time_t>::min();
>+    ts.tv_nsec = 0;
>+  }
>+  return ts;
>+}
>+
>+timeval ToTimeval(Time t) {
>+  timeval tv;
>+  timespec ts = absl::ToTimespec(t);
>+  tv.tv_sec = ts.tv_sec;
>+  if (tv.tv_sec != ts.tv_sec) {  // narrowing
>+    if (ts.tv_sec < 0) {
>+      tv.tv_sec = std::numeric_limits<decltype(tv.tv_sec)>::min();
>+      tv.tv_usec = 0;
>+    } else {
>+      tv.tv_sec = std::numeric_limits<decltype(tv.tv_sec)>::max();
>+      tv.tv_usec = 1000 * 1000 - 1;
>+    }
>+    return tv;
>+  }
>+  tv.tv_usec = static_cast<int>(ts.tv_nsec / 1000);  // suseconds_t
>+  return tv;
>+}
>+
>+double ToUDate(Time t) {
>+  return absl::FDivDuration(time_internal::ToUnixDuration(t),
>+                            absl::Milliseconds(1));
>+}
>+
>+int64_t ToUniversal(absl::Time t) {
>+  return absl::FloorToUnit(t - absl::UniversalEpoch(), absl::Nanoseconds(100));
>+}
>+
>+Time FromChrono(const std::chrono::system_clock::time_point& tp) {
>+  return time_internal::FromUnixDuration(time_internal::FromChrono(
>+      tp - std::chrono::system_clock::from_time_t(0)));
>+}
>+
>+std::chrono::system_clock::time_point ToChronoTime(absl::Time t) {
>+  using D = std::chrono::system_clock::duration;
>+  auto d = time_internal::ToUnixDuration(t);
>+  if (d < ZeroDuration()) d = Floor(d, FromChrono(D{1}));
>+  return std::chrono::system_clock::from_time_t(0) +
>+         time_internal::ToChronoDuration<D>(d);
>+}
>+
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/time.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/time.h
>new file mode 100644
>index 00000000000..c41cb89c5ef
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/time.h
>@@ -0,0 +1,1349 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// -----------------------------------------------------------------------------
>+// File: time.h
>+// -----------------------------------------------------------------------------
>+//
>+// This header file defines abstractions for computing with absolute points
>+// in time, durations of time, and formatting and parsing time within a given
>+// time zone. The following abstractions are defined:
>+//
>+//  * `absl::Time` defines an absolute, specific instance in time
>+//  * `absl::Duration` defines a signed, fixed-length span of time
>+//  * `absl::TimeZone` defines geopolitical time zone regions (as collected
>+//     within the IANA Time Zone database (https://www.iana.org/time-zones)).
>+//
>+// Example:
>+//
>+//   absl::TimeZone nyc;
>+//
>+//   // LoadTimeZone may fail so it's always better to check for success.
>+//   if (!absl::LoadTimeZone("America/New_York", &nyc)) {
>+//      // handle error case
>+//   }
>+//
>+//   // My flight leaves NYC on Jan 2, 2017 at 03:04:05
>+//   absl::Time takeoff = absl::FromDateTime(2017, 1, 2, 3, 4, 5, nyc);
>+//   absl::Duration flight_duration = absl::Hours(21) + absl::Minutes(35);
>+//   absl::Time landing = takeoff + flight_duration;
>+//
>+//   absl::TimeZone syd;
>+//   if (!absl::LoadTimeZone("Australia/Sydney", &syd)) {
>+//      // handle error case
>+//   }
>+//   std::string s = absl::FormatTime(
>+//       "My flight will land in Sydney on %Y-%m-%d at %H:%M:%S",
>+//       landing, syd);
>+//
>+#ifndef ABSL_TIME_TIME_H_
>+#define ABSL_TIME_TIME_H_
>+
>+#if !defined(_MSC_VER)
>+#include <sys/time.h>
>+#else
>+#include <winsock2.h>
>+#endif
>+#include <chrono>  // NOLINT(build/c++11)
>+#include <cstdint>
>+#include <ctime>
>+#include <ostream>
>+#include <string>
>+#include <type_traits>
>+#include <utility>
>+
>+#include "absl/base/port.h"  // Needed for string vs std::string
>+#include "absl/strings/string_view.h"
>+#include "absl/time/internal/cctz/include/cctz/time_zone.h"
>+
>+namespace absl {
>+
>+class Duration;  // Defined below
>+class Time;      // Defined below
>+class TimeZone;  // Defined below
>+
>+namespace time_internal {
>+int64_t IDivDuration(bool satq, Duration num, Duration den, Duration* rem);
>+constexpr Time FromUnixDuration(Duration d);
>+constexpr Duration ToUnixDuration(Time t);
>+constexpr int64_t GetRepHi(Duration d);
>+constexpr uint32_t GetRepLo(Duration d);
>+constexpr Duration MakeDuration(int64_t hi, uint32_t lo);
>+constexpr Duration MakeDuration(int64_t hi, int64_t lo);
>+inline Duration MakePosDoubleDuration(double n);
>+constexpr int64_t kTicksPerNanosecond = 4;
>+constexpr int64_t kTicksPerSecond = 1000 * 1000 * 1000 * kTicksPerNanosecond;
>+template <std::intmax_t N>
>+constexpr Duration FromInt64(int64_t v, std::ratio<1, N>);
>+constexpr Duration FromInt64(int64_t v, std::ratio<60>);
>+constexpr Duration FromInt64(int64_t v, std::ratio<3600>);
>+template <typename T>
>+using EnableIfIntegral = typename std::enable_if<
>+    std::is_integral<T>::value || std::is_enum<T>::value, int>::type;
>+template <typename T>
>+using EnableIfFloat =
>+    typename std::enable_if<std::is_floating_point<T>::value, int>::type;
>+}  // namespace time_internal
>+
>+// Duration
>+//
>+// The `absl::Duration` class represents a signed, fixed-length span of time.
>+// A `Duration` is generated using a unit-specific factory function, or is
>+// the result of subtracting one `absl::Time` from another. Durations behave
>+// like unit-safe integers and they support all the natural integer-like
>+// arithmetic operations. Arithmetic overflows and saturates at +/- infinity.
>+// `Duration` should be passed by value rather than const reference.
>+//
>+// Factory functions `Nanoseconds()`, `Microseconds()`, `Milliseconds()`,
>+// `Seconds()`, `Minutes()`, `Hours()` and `InfiniteDuration()` allow for
>+// creation of constexpr `Duration` values
>+//
>+// Examples:
>+//
>+//   constexpr absl::Duration ten_ns = absl::Nanoseconds(10);
>+//   constexpr absl::Duration min = absl::Minutes(1);
>+//   constexpr absl::Duration hour = absl::Hours(1);
>+//   absl::Duration dur = 60 * min;  // dur == hour
>+//   absl::Duration half_sec = absl::Milliseconds(500);
>+//   absl::Duration quarter_sec = 0.25 * absl::Seconds(1);
>+//
>+// `Duration` values can be easily converted to an integral number of units
>+// using the division operator.
>+//
>+// Example:
>+//
>+//   constexpr absl::Duration dur = absl::Milliseconds(1500);
>+//   int64_t ns = dur / absl::Nanoseconds(1);   // ns == 1500000000
>+//   int64_t ms = dur / absl::Milliseconds(1);  // ms == 1500
>+//   int64_t sec = dur / absl::Seconds(1);    // sec == 1 (subseconds truncated)
>+//   int64_t min = dur / absl::Minutes(1);    // min == 0
>+//
>+// See the `IDivDuration()` and `FDivDuration()` functions below for details on
>+// how to access the fractional parts of the quotient.
>+//
>+// Alternatively, conversions can be performed using helpers such as
>+// `ToInt64Microseconds()` and `ToDoubleSeconds()`.
>+class Duration {
>+ public:
>+  // Value semantics.
>+  constexpr Duration() : rep_hi_(0), rep_lo_(0) {}  // zero-length duration
>+
>+  // Compound assignment operators.
>+  Duration& operator+=(Duration d);
>+  Duration& operator-=(Duration d);
>+  Duration& operator*=(int64_t r);
>+  Duration& operator*=(double r);
>+  Duration& operator/=(int64_t r);
>+  Duration& operator/=(double r);
>+  Duration& operator%=(Duration rhs);
>+
>+  // Overloads that forward to either the int64_t or double overloads above.
>+  template <typename T>
>+  Duration& operator*=(T r) {
>+    int64_t x = r;
>+    return *this *= x;
>+  }
>+  template <typename T>
>+  Duration& operator/=(T r) {
>+    int64_t x = r;
>+    return *this /= x;
>+  }
>+  Duration& operator*=(float r) { return *this *= static_cast<double>(r); }
>+  Duration& operator/=(float r) { return *this /= static_cast<double>(r); }
>+
>+ private:
>+  friend constexpr int64_t time_internal::GetRepHi(Duration d);
>+  friend constexpr uint32_t time_internal::GetRepLo(Duration d);
>+  friend constexpr Duration time_internal::MakeDuration(int64_t hi,
>+                                                        uint32_t lo);
>+  constexpr Duration(int64_t hi, uint32_t lo) : rep_hi_(hi), rep_lo_(lo) {}
>+  int64_t rep_hi_;
>+  uint32_t rep_lo_;
>+};
>+
>+// Relational Operators
>+constexpr bool operator<(Duration lhs, Duration rhs);
>+constexpr bool operator>(Duration lhs, Duration rhs) { return rhs < lhs; }
>+constexpr bool operator>=(Duration lhs, Duration rhs) { return !(lhs < rhs); }
>+constexpr bool operator<=(Duration lhs, Duration rhs) { return !(rhs < lhs); }
>+constexpr bool operator==(Duration lhs, Duration rhs);
>+constexpr bool operator!=(Duration lhs, Duration rhs) { return !(lhs == rhs); }
>+
>+// Additive Operators
>+constexpr Duration operator-(Duration d);
>+inline Duration operator+(Duration lhs, Duration rhs) { return lhs += rhs; }
>+inline Duration operator-(Duration lhs, Duration rhs) { return lhs -= rhs; }
>+
>+// Multiplicative Operators
>+template <typename T>
>+Duration operator*(Duration lhs, T rhs) {
>+  return lhs *= rhs;
>+}
>+template <typename T>
>+Duration operator*(T lhs, Duration rhs) {
>+  return rhs *= lhs;
>+}
>+template <typename T>
>+Duration operator/(Duration lhs, T rhs) {
>+  return lhs /= rhs;
>+}
>+inline int64_t operator/(Duration lhs, Duration rhs) {
>+  return time_internal::IDivDuration(true, lhs, rhs,
>+                                     &lhs);  // trunc towards zero
>+}
>+inline Duration operator%(Duration lhs, Duration rhs) { return lhs %= rhs; }
>+
>+// IDivDuration()
>+//
>+// Divides a numerator `Duration` by a denominator `Duration`, returning the
>+// quotient and remainder. The remainder always has the same sign as the
>+// numerator. The returned quotient and remainder respect the identity:
>+//
>+//   numerator = denominator * quotient + remainder
>+//
>+// Returned quotients are capped to the range of `int64_t`, with the difference
>+// spilling into the remainder to uphold the above identity. This means that the
>+// remainder returned could differ from the remainder returned by
>+// `Duration::operator%` for huge quotients.
>+//
>+// See also the notes on `InfiniteDuration()` below regarding the behavior of
>+// division involving zero and infinite durations.
>+//
>+// Example:
>+//
>+//   constexpr absl::Duration a =
>+//       absl::Seconds(std::numeric_limits<int64_t>::max());  // big
>+//   constexpr absl::Duration b = absl::Nanoseconds(1);       // small
>+//
>+//   absl::Duration rem = a % b;
>+//   // rem == absl::ZeroDuration()
>+//
>+//   // Here, q would overflow int64_t, so rem accounts for the difference.
>+//   int64_t q = absl::IDivDuration(a, b, &rem);
>+//   // q == std::numeric_limits<int64_t>::max(), rem == a - b * q
>+inline int64_t IDivDuration(Duration num, Duration den, Duration* rem) {
>+  return time_internal::IDivDuration(true, num, den,
>+                                     rem);  // trunc towards zero
>+}
>+
>+// FDivDuration()
>+//
>+// Divides a `Duration` numerator into a fractional number of units of a
>+// `Duration` denominator.
>+//
>+// See also the notes on `InfiniteDuration()` below regarding the behavior of
>+// division involving zero and infinite durations.
>+//
>+// Example:
>+//
>+//   double d = absl::FDivDuration(absl::Milliseconds(1500), absl::Seconds(1));
>+//   // d == 1.5
>+double FDivDuration(Duration num, Duration den);
>+
>+// ZeroDuration()
>+//
>+// Returns a zero-length duration. This function behaves just like the default
>+// constructor, but the name helps make the semantics clear at call sites.
>+constexpr Duration ZeroDuration() { return Duration(); }
>+
>+// AbsDuration()
>+//
>+// Returns the absolute value of a duration.
>+inline Duration AbsDuration(Duration d) {
>+  return (d < ZeroDuration()) ? -d : d;
>+}
>+
>+// Trunc()
>+//
>+// Truncates a duration (toward zero) to a multiple of a non-zero unit.
>+//
>+// Example:
>+//
>+//   absl::Duration d = absl::Nanoseconds(123456789);
>+//   absl::Duration a = absl::Trunc(d, absl::Microseconds(1));  // 123456us
>+Duration Trunc(Duration d, Duration unit);
>+
>+// Floor()
>+//
>+// Floors a duration using the passed duration unit to its largest value not
>+// greater than the duration.
>+//
>+// Example:
>+//
>+//   absl::Duration d = absl::Nanoseconds(123456789);
>+//   absl::Duration b = absl::Floor(d, absl::Microseconds(1));  // 123456us
>+Duration Floor(Duration d, Duration unit);
>+
>+// Ceil()
>+//
>+// Returns the ceiling of a duration using the passed duration unit to its
>+// smallest value not less than the duration.
>+//
>+// Example:
>+//
>+//   absl::Duration d = absl::Nanoseconds(123456789);
>+//   absl::Duration c = absl::Ceil(d, absl::Microseconds(1));   // 123457us
>+Duration Ceil(Duration d, Duration unit);
>+
>+// InfiniteDuration()
>+//
>+// Returns an infinite `Duration`.  To get a `Duration` representing negative
>+// infinity, use `-InfiniteDuration()`.
>+//
>+// Duration arithmetic overflows to +/- infinity and saturates. In general,
>+// arithmetic with `Duration` infinities is similar to IEEE 754 infinities
>+// except where IEEE 754 NaN would be involved, in which case +/-
>+// `InfiniteDuration()` is used in place of a "nan" Duration.
>+//
>+// Examples:
>+//
>+//   constexpr absl::Duration inf = absl::InfiniteDuration();
>+//   const absl::Duration d = ... any finite duration ...
>+//
>+//   inf == inf + inf
>+//   inf == inf + d
>+//   inf == inf - inf
>+//   -inf == d - inf
>+//
>+//   inf == d * 1e100
>+//   inf == inf / 2
>+//   0 == d / inf
>+//   INT64_MAX == inf / d
>+//
>+//   // Division by zero returns infinity, or INT64_MIN/MAX where appropriate.
>+//   inf == d / 0
>+//   INT64_MAX == d / absl::ZeroDuration()
>+//
>+// The examples involving the `/` operator above also apply to `IDivDuration()`
>+// and `FDivDuration()`.
>+constexpr Duration InfiniteDuration();
>+
>+// Nanoseconds()
>+// Microseconds()
>+// Milliseconds()
>+// Seconds()
>+// Minutes()
>+// Hours()
>+//
>+// Factory functions for constructing `Duration` values from an integral number
>+// of the unit indicated by the factory function's name.
>+//
>+// Note: no "Days()" factory function exists because "a day" is ambiguous. Civil
>+// days are not always 24 hours long, and a 24-hour duration often does not
>+// correspond with a civil day. If a 24-hour duration is needed, use
>+// `absl::Hours(24)`.
>+//
>+//
>+// Example:
>+//
>+//   absl::Duration a = absl::Seconds(60);
>+//   absl::Duration b = absl::Minutes(1);  // b == a
>+constexpr Duration Nanoseconds(int64_t n);
>+constexpr Duration Microseconds(int64_t n);
>+constexpr Duration Milliseconds(int64_t n);
>+constexpr Duration Seconds(int64_t n);
>+constexpr Duration Minutes(int64_t n);
>+constexpr Duration Hours(int64_t n);
>+
>+// Factory overloads for constructing `Duration` values from a floating-point
>+// number of the unit indicated by the factory function's name. These functions
>+// exist for convenience, but they are not as efficient as the integral
>+// factories, which should be preferred.
>+//
>+// Example:
>+//   auto a = absl::Seconds(1.5);        // OK
>+//   auto b = absl::Milliseconds(1500);  // BETTER
>+template <typename T, time_internal::EnableIfFloat<T> = 0>
>+Duration Nanoseconds(T n) {
>+  return n * Nanoseconds(1);
>+}
>+template <typename T, time_internal::EnableIfFloat<T> = 0>
>+Duration Microseconds(T n) {
>+  return n * Microseconds(1);
>+}
>+template <typename T, time_internal::EnableIfFloat<T> = 0>
>+Duration Milliseconds(T n) {
>+  return n * Milliseconds(1);
>+}
>+template <typename T, time_internal::EnableIfFloat<T> = 0>
>+Duration Seconds(T n) {
>+  if (n >= 0) {
>+    if (n >= std::numeric_limits<int64_t>::max()) return InfiniteDuration();
>+    return time_internal::MakePosDoubleDuration(n);
>+  } else {
>+    if (n <= std::numeric_limits<int64_t>::min()) return -InfiniteDuration();
>+    return -time_internal::MakePosDoubleDuration(-n);
>+  }
>+}
>+template <typename T, time_internal::EnableIfFloat<T> = 0>
>+Duration Minutes(T n) {
>+  return n * Minutes(1);
>+}
>+template <typename T, time_internal::EnableIfFloat<T> = 0>
>+Duration Hours(T n) {
>+  return n * Hours(1);
>+}
>+
>+// ToInt64Nanoseconds()
>+// ToInt64Microseconds()
>+// ToInt64Milliseconds()
>+// ToInt64Seconds()
>+// ToInt64Minutes()
>+// ToInt64Hours()
>+//
>+// Helper functions that convert a Duration to an integral count of the
>+// indicated unit. These functions are shorthand for the `IDivDuration()`
>+// function above; see its documentation for details about overflow, etc.
>+//
>+// Example:
>+//
>+//   absl::Duration d = absl::Milliseconds(1500);
>+//   int64_t isec = absl::ToInt64Seconds(d);  // isec == 1
>+int64_t ToInt64Nanoseconds(Duration d);
>+int64_t ToInt64Microseconds(Duration d);
>+int64_t ToInt64Milliseconds(Duration d);
>+int64_t ToInt64Seconds(Duration d);
>+int64_t ToInt64Minutes(Duration d);
>+int64_t ToInt64Hours(Duration d);
>+
>+// ToDoubleNanoSeconds()
>+// ToDoubleMicroseconds()
>+// ToDoubleMilliseconds()
>+// ToDoubleSeconds()
>+// ToDoubleMinutes()
>+// ToDoubleHours()
>+//
>+// Helper functions that convert a Duration to a floating point count of the
>+// indicated unit. These functions are shorthand for the `FDivDuration()`
>+// function above; see its documentation for details about overflow, etc.
>+//
>+// Example:
>+//
>+//   absl::Duration d = absl::Milliseconds(1500);
>+//   double dsec = absl::ToDoubleSeconds(d);  // dsec == 1.5
>+double ToDoubleNanoseconds(Duration d);
>+double ToDoubleMicroseconds(Duration d);
>+double ToDoubleMilliseconds(Duration d);
>+double ToDoubleSeconds(Duration d);
>+double ToDoubleMinutes(Duration d);
>+double ToDoubleHours(Duration d);
>+
>+// FromChrono()
>+//
>+// Converts any of the pre-defined std::chrono durations to an absl::Duration.
>+//
>+// Example:
>+//
>+//   std::chrono::milliseconds ms(123);
>+//   absl::Duration d = absl::FromChrono(ms);
>+constexpr Duration FromChrono(const std::chrono::nanoseconds& d);
>+constexpr Duration FromChrono(const std::chrono::microseconds& d);
>+constexpr Duration FromChrono(const std::chrono::milliseconds& d);
>+constexpr Duration FromChrono(const std::chrono::seconds& d);
>+constexpr Duration FromChrono(const std::chrono::minutes& d);
>+constexpr Duration FromChrono(const std::chrono::hours& d);
>+
>+// ToChronoNanoseconds()
>+// ToChronoMicroseconds()
>+// ToChronoMilliseconds()
>+// ToChronoSeconds()
>+// ToChronoMinutes()
>+// ToChronoHours()
>+//
>+// Converts an absl::Duration to any of the pre-defined std::chrono durations.
>+// If overflow would occur, the returned value will saturate at the min/max
>+// chrono duration value instead.
>+//
>+// Example:
>+//
>+//   absl::Duration d = absl::Microseconds(123);
>+//   auto x = absl::ToChronoMicroseconds(d);
>+//   auto y = absl::ToChronoNanoseconds(d);  // x == y
>+//   auto z = absl::ToChronoSeconds(absl::InfiniteDuration());
>+//   // z == std::chrono::seconds::max()
>+std::chrono::nanoseconds ToChronoNanoseconds(Duration d);
>+std::chrono::microseconds ToChronoMicroseconds(Duration d);
>+std::chrono::milliseconds ToChronoMilliseconds(Duration d);
>+std::chrono::seconds ToChronoSeconds(Duration d);
>+std::chrono::minutes ToChronoMinutes(Duration d);
>+std::chrono::hours ToChronoHours(Duration d);
>+
>+// FormatDuration()
>+//
>+// Returns a std::string representing the duration in the form "72h3m0.5s".
>+// Returns "inf" or "-inf" for +/- `InfiniteDuration()`.
>+std::string FormatDuration(Duration d);
>+
>+// Output stream operator.
>+inline std::ostream& operator<<(std::ostream& os, Duration d) {
>+  return os << FormatDuration(d);
>+}
>+
>+// ParseDuration()
>+//
>+// Parses a duration std::string consisting of a possibly signed sequence of
>+// decimal numbers, each with an optional fractional part and a unit
>+// suffix.  The valid suffixes are "ns", "us" "ms", "s", "m", and "h".
>+// Simple examples include "300ms", "-1.5h", and "2h45m".  Parses "0" as
>+// `ZeroDuration()`.  Parses "inf" and "-inf" as +/- `InfiniteDuration()`.
>+bool ParseDuration(const std::string& dur_string, Duration* d);
>+
>+// Support for flag values of type Duration. Duration flags must be specified
>+// in a format that is valid input for absl::ParseDuration().
>+bool ParseFlag(const std::string& text, Duration* dst, std::string* error);
>+std::string UnparseFlag(Duration d);
>+
>+// Time
>+//
>+// An `absl::Time` represents a specific instant in time. Arithmetic operators
>+// are provided for naturally expressing time calculations. Instances are
>+// created using `absl::Now()` and the `absl::From*()` factory functions that
>+// accept the gamut of other time representations. Formatting and parsing
>+// functions are provided for conversion to and from strings.  `absl::Time`
>+// should be passed by value rather than const reference.
>+//
>+// `absl::Time` assumes there are 60 seconds in a minute, which means the
>+// underlying time scales must be "smeared" to eliminate leap seconds.
>+// See https://developers.google.com/time/smear.
>+//
>+// Even though `absl::Time` supports a wide range of timestamps, exercise
>+// caution when using values in the distant past. `absl::Time` uses the
>+// Proleptic Gregorian calendar, which extends the Gregorian calendar backward
>+// to dates before its introduction in 1582.
>+// See https://en.wikipedia.org/wiki/Proleptic_Gregorian_calendar
>+// for more information. Use the ICU calendar classes to convert a date in
>+// some other calendar (http://userguide.icu-project.org/datetime/calendar).
>+//
>+// Similarly, standardized time zones are a reasonably recent innovation, with
>+// the Greenwich prime meridian being established in 1884. The TZ database
>+// itself does not profess accurate offsets for timestamps prior to 1970. The
>+// breakdown of future timestamps is subject to the whim of regional
>+// governments.
>+//
>+// The `absl::Time` class represents an instant in time as a count of clock
>+// ticks of some granularity (resolution) from some starting point (epoch).
>+//
>+//
>+// `absl::Time` uses a resolution that is high enough to avoid loss in
>+// precision, and a range that is wide enough to avoid overflow, when
>+// converting between tick counts in most Google time scales (i.e., precision
>+// of at least one nanosecond, and range +/-100 billion years).  Conversions
>+// between the time scales are performed by truncating (towards negative
>+// infinity) to the nearest representable point.
>+//
>+// Examples:
>+//
>+//   absl::Time t1 = ...;
>+//   absl::Time t2 = t1 + absl::Minutes(2);
>+//   absl::Duration d = t2 - t1;  // == absl::Minutes(2)
>+//   absl::Time::Breakdown bd = t1.In(absl::LocalTimeZone());
>+//
>+class Time {
>+ public:
>+  // Value semantics.
>+
>+  // Returns the Unix epoch.  However, those reading your code may not know
>+  // or expect the Unix epoch as the default value, so make your code more
>+  // readable by explicitly initializing all instances before use.
>+  //
>+  // Example:
>+  //   absl::Time t = absl::UnixEpoch();
>+  //   absl::Time t = absl::Now();
>+  //   absl::Time t = absl::TimeFromTimeval(tv);
>+  //   absl::Time t = absl::InfinitePast();
>+  constexpr Time() {}
>+
>+  // Assignment operators.
>+  Time& operator+=(Duration d) {
>+    rep_ += d;
>+    return *this;
>+  }
>+  Time& operator-=(Duration d) {
>+    rep_ -= d;
>+    return *this;
>+  }
>+
>+  // Time::Breakdown
>+  //
>+  // The calendar and wall-clock (aka "civil time") components of an
>+  // `absl::Time` in a certain `absl::TimeZone`. This struct is not
>+  // intended to represent an instant in time. So, rather than passing
>+  // a `Time::Breakdown` to a function, pass an `absl::Time` and an
>+  // `absl::TimeZone`.
>+  struct Breakdown {
>+    int64_t year;          // year (e.g., 2013)
>+    int month;           // month of year [1:12]
>+    int day;             // day of month [1:31]
>+    int hour;            // hour of day [0:23]
>+    int minute;          // minute of hour [0:59]
>+    int second;          // second of minute [0:59]
>+    Duration subsecond;  // [Seconds(0):Seconds(1)) if finite
>+    int weekday;         // 1==Mon, ..., 7=Sun
>+    int yearday;         // day of year [1:366]
>+
>+    // Note: The following fields exist for backward compatibility
>+    // with older APIs.  Accessing these fields directly is a sign of
>+    // imprudent logic in the calling code.  Modern time-related code
>+    // should only access this data indirectly by way of FormatTime().
>+    // These fields are undefined for InfiniteFuture() and InfinitePast().
>+    int offset;             // seconds east of UTC
>+    bool is_dst;            // is offset non-standard?
>+    const char* zone_abbr;  // time-zone abbreviation (e.g., "PST")
>+  };
>+
>+  // Time::In()
>+  //
>+  // Returns the breakdown of this instant in the given TimeZone.
>+  Breakdown In(TimeZone tz) const;
>+
>+ private:
>+  friend constexpr Time time_internal::FromUnixDuration(Duration d);
>+  friend constexpr Duration time_internal::ToUnixDuration(Time t);
>+  friend constexpr bool operator<(Time lhs, Time rhs);
>+  friend constexpr bool operator==(Time lhs, Time rhs);
>+  friend Duration operator-(Time lhs, Time rhs);
>+  friend constexpr Time UniversalEpoch();
>+  friend constexpr Time InfiniteFuture();
>+  friend constexpr Time InfinitePast();
>+  constexpr explicit Time(Duration rep) : rep_(rep) {}
>+  Duration rep_;
>+};
>+
>+// Relational Operators
>+constexpr bool operator<(Time lhs, Time rhs) { return lhs.rep_ < rhs.rep_; }
>+constexpr bool operator>(Time lhs, Time rhs) { return rhs < lhs; }
>+constexpr bool operator>=(Time lhs, Time rhs) { return !(lhs < rhs); }
>+constexpr bool operator<=(Time lhs, Time rhs) { return !(rhs < lhs); }
>+constexpr bool operator==(Time lhs, Time rhs) { return lhs.rep_ == rhs.rep_; }
>+constexpr bool operator!=(Time lhs, Time rhs) { return !(lhs == rhs); }
>+
>+// Additive Operators
>+inline Time operator+(Time lhs, Duration rhs) { return lhs += rhs; }
>+inline Time operator+(Duration lhs, Time rhs) { return rhs += lhs; }
>+inline Time operator-(Time lhs, Duration rhs) { return lhs -= rhs; }
>+inline Duration operator-(Time lhs, Time rhs) { return lhs.rep_ - rhs.rep_; }
>+
>+// UnixEpoch()
>+//
>+// Returns the `absl::Time` representing "1970-01-01 00:00:00.0 +0000".
>+constexpr Time UnixEpoch() { return Time(); }
>+
>+// UniversalEpoch()
>+//
>+// Returns the `absl::Time` representing "0001-01-01 00:00:00.0 +0000", the
>+// epoch of the ICU Universal Time Scale.
>+constexpr Time UniversalEpoch() {
>+  // 719162 is the number of days from 0001-01-01 to 1970-01-01,
>+  // assuming the Gregorian calendar.
>+  return Time(time_internal::MakeDuration(-24 * 719162 * int64_t{3600}, 0U));
>+}
>+
>+// InfiniteFuture()
>+//
>+// Returns an `absl::Time` that is infinitely far in the future.
>+constexpr Time InfiniteFuture() {
>+  return Time(
>+      time_internal::MakeDuration(std::numeric_limits<int64_t>::max(), ~0U));
>+}
>+
>+// InfinitePast()
>+//
>+// Returns an `absl::Time` that is infinitely far in the past.
>+constexpr Time InfinitePast() {
>+  return Time(
>+      time_internal::MakeDuration(std::numeric_limits<int64_t>::min(), ~0U));
>+}
>+
>+// TimeConversion
>+//
>+// An `absl::TimeConversion` represents the conversion of year, month, day,
>+// hour, minute, and second values (i.e., a civil time), in a particular
>+// `absl::TimeZone`, to a time instant (an absolute time), as returned by
>+// `absl::ConvertDateTime()`. (Subseconds must be handled separately.)
>+//
>+// It is possible, though, for a caller to try to convert values that
>+// do not represent an actual or unique instant in time (due to a shift
>+// in UTC offset in the `absl::TimeZone`, which results in a discontinuity in
>+// the civil-time components). For example, a daylight-saving-time
>+// transition skips or repeats civil times---in the United States, March
>+// 13, 2011 02:15 never occurred, while November 6, 2011 01:15 occurred
>+// twice---so requests for such times are not well-defined.
>+//
>+// To account for these possibilities, `absl::TimeConversion` is richer
>+// than just a single `absl::Time`. When the civil time is skipped or
>+// repeated, `absl::ConvertDateTime()` returns times calculated using the
>+// pre-transition and post-transition UTC offsets, plus the transition
>+// time itself.
>+//
>+// Examples:
>+//
>+//   absl::TimeZone lax;
>+//   if (!absl::LoadTimeZone("America/Los_Angeles", &lax)) { ... }
>+//
>+//   // A unique civil time
>+//   absl::TimeConversion jan01 =
>+//       absl::ConvertDateTime(2011, 1, 1, 0, 0, 0, lax);
>+//   // jan01.kind == TimeConversion::UNIQUE
>+//   // jan01.pre    is 2011/01/01 00:00:00 -0800
>+//   // jan01.trans  is 2011/01/01 00:00:00 -0800
>+//   // jan01.post   is 2011/01/01 00:00:00 -0800
>+//
>+//   // A Spring DST transition, when there is a gap in civil time
>+//   absl::TimeConversion mar13 =
>+//       absl::ConvertDateTime(2011, 3, 13, 2, 15, 0, lax);
>+//   // mar13.kind == TimeConversion::SKIPPED
>+//   // mar13.pre   is 2011/03/13 03:15:00 -0700
>+//   // mar13.trans is 2011/03/13 03:00:00 -0700
>+//   // mar13.post  is 2011/03/13 01:15:00 -0800
>+//
>+//   // A Fall DST transition, when civil times are repeated
>+//   absl::TimeConversion nov06 =
>+//       absl::ConvertDateTime(2011, 11, 6, 1, 15, 0, lax);
>+//   // nov06.kind == TimeConversion::REPEATED
>+//   // nov06.pre   is 2011/11/06 01:15:00 -0700
>+//   // nov06.trans is 2011/11/06 01:00:00 -0800
>+//   // nov06.post  is 2011/11/06 01:15:00 -0800
>+//
>+// The input month, day, hour, minute, and second values can also be
>+// outside of their valid ranges, in which case they will be "normalized"
>+// during the conversion.
>+//
>+// Example:
>+//
>+//   // "October 32" normalizes to "November 1".
>+//   absl::TimeZone tz = absl::LocalTimeZone();
>+//   absl::TimeConversion tc =
>+//       absl::ConvertDateTime(2013, 10, 32, 8, 30, 0, tz);
>+//   // tc.kind == TimeConversion::UNIQUE && tc.normalized == true
>+//   // tc.pre.In(tz).month == 11 && tc.pre.In(tz).day == 1
>+struct TimeConversion {
>+  Time pre;    // time calculated using the pre-transition offset
>+  Time trans;  // when the civil-time discontinuity occurred
>+  Time post;   // time calculated using the post-transition offset
>+
>+  enum Kind {
>+    UNIQUE,    // the civil time was singular (pre == trans == post)
>+    SKIPPED,   // the civil time did not exist
>+    REPEATED,  // the civil time was ambiguous
>+  };
>+  Kind kind;
>+
>+  bool normalized;  // input values were outside their valid ranges
>+};
>+
>+// ConvertDateTime()
>+//
>+// The full generality of a civil time to absl::Time conversion.
>+TimeConversion ConvertDateTime(int64_t year, int mon, int day, int hour,
>+                               int min, int sec, TimeZone tz);
>+
>+// FromDateTime()
>+//
>+// A convenience wrapper for `absl::ConvertDateTime()` that simply returns the
>+// "pre" `absl::Time`.  That is, the unique result, or the instant that
>+// is correct using the pre-transition offset (as if the transition
>+// never happened). This is typically the answer that humans expected when
>+// faced with non-unique times, such as near daylight-saving time transitions.
>+//
>+// Example:
>+//
>+//   absl::TimeZone seattle;
>+//   if (!absl::LoadTimeZone("America/Los_Angeles", &seattle)) { ... }
>+//   absl::Time t =  absl::FromDateTime(2017, 9, 26, 9, 30, 0, seattle);
>+Time FromDateTime(int64_t year, int mon, int day, int hour, int min, int sec,
>+                  TimeZone tz);
>+
>+// FromTM()
>+//
>+// Converts the `tm_year`, `tm_mon`, `tm_mday`, `tm_hour`, `tm_min`, and
>+// `tm_sec` fields to an `absl::Time` using the given time zone. See ctime(3)
>+// for a description of the expected values of the tm fields. IFF the indicated
>+// time instant is not unique (see `absl::ConvertDateTime()` above), the
>+// `tm_isdst` field is consulted to select the desired instant (`tm_isdst` > 0
>+// means DST, `tm_isdst` == 0 means no DST, `tm_isdst` < 0 means use the default
>+// like `absl::FromDateTime()`).
>+Time FromTM(const struct tm& tm, TimeZone tz);
>+
>+// ToTM()
>+//
>+// Converts the given `absl::Time` to a struct tm using the given time zone.
>+// See ctime(3) for a description of the values of the tm fields.
>+struct tm ToTM(Time t, TimeZone tz);
>+
>+// FromUnixNanos()
>+// FromUnixMicros()
>+// FromUnixMillis()
>+// FromUnixSeconds()
>+// FromTimeT()
>+// FromUDate()
>+// FromUniversal()
>+//
>+// Creates an `absl::Time` from a variety of other representations.
>+constexpr Time FromUnixNanos(int64_t ns);
>+constexpr Time FromUnixMicros(int64_t us);
>+constexpr Time FromUnixMillis(int64_t ms);
>+constexpr Time FromUnixSeconds(int64_t s);
>+constexpr Time FromTimeT(time_t t);
>+Time FromUDate(double udate);
>+Time FromUniversal(int64_t universal);
>+
>+// ToUnixNanos()
>+// ToUnixMicros()
>+// ToUnixMillis()
>+// ToUnixSeconds()
>+// ToTimeT()
>+// ToUDate()
>+// ToUniversal()
>+//
>+// Converts an `absl::Time` to a variety of other representations.  Note that
>+// these operations round down toward negative infinity where necessary to
>+// adjust to the resolution of the result type.  Beware of possible time_t
>+// over/underflow in ToTime{T,val,spec}() on 32-bit platforms.
>+int64_t ToUnixNanos(Time t);
>+int64_t ToUnixMicros(Time t);
>+int64_t ToUnixMillis(Time t);
>+int64_t ToUnixSeconds(Time t);
>+time_t ToTimeT(Time t);
>+double ToUDate(Time t);
>+int64_t ToUniversal(Time t);
>+
>+// DurationFromTimespec()
>+// DurationFromTimeval()
>+// ToTimespec()
>+// ToTimeval()
>+// TimeFromTimespec()
>+// TimeFromTimeval()
>+// ToTimespec()
>+// ToTimeval()
>+//
>+// Some APIs use a timespec or a timeval as a Duration (e.g., nanosleep(2)
>+// and select(2)), while others use them as a Time (e.g. clock_gettime(2)
>+// and gettimeofday(2)), so conversion functions are provided for both cases.
>+// The "to timespec/val" direction is easily handled via overloading, but
>+// for "from timespec/val" the desired type is part of the function name.
>+Duration DurationFromTimespec(timespec ts);
>+Duration DurationFromTimeval(timeval tv);
>+timespec ToTimespec(Duration d);
>+timeval ToTimeval(Duration d);
>+Time TimeFromTimespec(timespec ts);
>+Time TimeFromTimeval(timeval tv);
>+timespec ToTimespec(Time t);
>+timeval ToTimeval(Time t);
>+
>+// FromChrono()
>+//
>+// Converts a std::chrono::system_clock::time_point to an absl::Time.
>+//
>+// Example:
>+//
>+//   auto tp = std::chrono::system_clock::from_time_t(123);
>+//   absl::Time t = absl::FromChrono(tp);
>+//   // t == absl::FromTimeT(123)
>+Time FromChrono(const std::chrono::system_clock::time_point& tp);
>+
>+// ToChronoTime()
>+//
>+// Converts an absl::Time to a std::chrono::system_clock::time_point. If
>+// overflow would occur, the returned value will saturate at the min/max time
>+// point value instead.
>+//
>+// Example:
>+//
>+//   absl::Time t = absl::FromTimeT(123);
>+//   auto tp = absl::ToChronoTime(t);
>+//   // tp == std::chrono::system_clock::from_time_t(123);
>+std::chrono::system_clock::time_point ToChronoTime(Time);
>+
>+// RFC3339_full
>+// RFC3339_sec
>+//
>+// FormatTime()/ParseTime() format specifiers for RFC3339 date/time strings,
>+// with trailing zeros trimmed or with fractional seconds omitted altogether.
>+//
>+// Note that RFC3339_sec[] matches an ISO 8601 extended format for date
>+// and time with UTC offset.
>+extern const char RFC3339_full[];  // %Y-%m-%dT%H:%M:%E*S%Ez
>+extern const char RFC3339_sec[];   // %Y-%m-%dT%H:%M:%S%Ez
>+
>+// RFC1123_full
>+// RFC1123_no_wday
>+//
>+// FormatTime()/ParseTime() format specifiers for RFC1123 date/time strings.
>+extern const char RFC1123_full[];     // %a, %d %b %E4Y %H:%M:%S %z
>+extern const char RFC1123_no_wday[];  // %d %b %E4Y %H:%M:%S %z
>+
>+// FormatTime()
>+//
>+// Formats the given `absl::Time` in the `absl::TimeZone` according to the
>+// provided format std::string. Uses strftime()-like formatting options, with
>+// the following extensions:
>+//
>+//   - %Ez  - RFC3339-compatible numeric UTC offset (+hh:mm or -hh:mm)
>+//   - %E*z - Full-resolution numeric UTC offset (+hh:mm:ss or -hh:mm:ss)
>+//   - %E#S - Seconds with # digits of fractional precision
>+//   - %E*S - Seconds with full fractional precision (a literal '*')
>+//   - %E#f - Fractional seconds with # digits of precision
>+//   - %E*f - Fractional seconds with full precision (a literal '*')
>+//   - %E4Y - Four-character years (-999 ... -001, 0000, 0001 ... 9999)
>+//
>+// Note that %E0S behaves like %S, and %E0f produces no characters.  In
>+// contrast %E*f always produces at least one digit, which may be '0'.
>+//
>+// Note that %Y produces as many characters as it takes to fully render the
>+// year.  A year outside of [-999:9999] when formatted with %E4Y will produce
>+// more than four characters, just like %Y.
>+//
>+// We recommend that format strings include the UTC offset (%z, %Ez, or %E*z)
>+// so that the result uniquely identifies a time instant.
>+//
>+// Example:
>+//
>+//   absl::TimeZone lax;
>+//   if (!absl::LoadTimeZone("America/Los_Angeles", &lax)) { ... }
>+//   absl::Time t = absl::FromDateTime(2013, 1, 2, 3, 4, 5, lax);
>+//
>+//   std::string f = absl::FormatTime("%H:%M:%S", t, lax);  // "03:04:05"
>+//   f = absl::FormatTime("%H:%M:%E3S", t, lax);  // "03:04:05.000"
>+//
>+// Note: If the given `absl::Time` is `absl::InfiniteFuture()`, the returned
>+// std::string will be exactly "infinite-future". If the given `absl::Time` is
>+// `absl::InfinitePast()`, the returned std::string will be exactly "infinite-past".
>+// In both cases the given format std::string and `absl::TimeZone` are ignored.
>+//
>+std::string FormatTime(const std::string& format, Time t, TimeZone tz);
>+
>+// Convenience functions that format the given time using the RFC3339_full
>+// format.  The first overload uses the provided TimeZone, while the second
>+// uses LocalTimeZone().
>+std::string FormatTime(Time t, TimeZone tz);
>+std::string FormatTime(Time t);
>+
>+// Output stream operator.
>+inline std::ostream& operator<<(std::ostream& os, Time t) {
>+  return os << FormatTime(t);
>+}
>+
>+// ParseTime()
>+//
>+// Parses an input std::string according to the provided format std::string and
>+// returns the corresponding `absl::Time`. Uses strftime()-like formatting
>+// options, with the same extensions as FormatTime(), but with the
>+// exceptions that %E#S is interpreted as %E*S, and %E#f as %E*f.  %Ez
>+// and %E*z also accept the same inputs.
>+//
>+// %Y consumes as many numeric characters as it can, so the matching data
>+// should always be terminated with a non-numeric.  %E4Y always consumes
>+// exactly four characters, including any sign.
>+//
>+// Unspecified fields are taken from the default date and time of ...
>+//
>+//   "1970-01-01 00:00:00.0 +0000"
>+//
>+// For example, parsing a std::string of "15:45" (%H:%M) will return an absl::Time
>+// that represents "1970-01-01 15:45:00.0 +0000".
>+//
>+// Note that since ParseTime() returns time instants, it makes the most sense
>+// to parse fully-specified date/time strings that include a UTC offset (%z,
>+// %Ez, or %E*z).
>+//
>+// Note also that `absl::ParseTime()` only heeds the fields year, month, day,
>+// hour, minute, (fractional) second, and UTC offset.  Other fields, like
>+// weekday (%a or %A), while parsed for syntactic validity, are ignored
>+// in the conversion.
>+//
>+// Date and time fields that are out-of-range will be treated as errors
>+// rather than normalizing them like `absl::FromDateTime()` does.  For example,
>+// it is an error to parse the date "Oct 32, 2013" because 32 is out of range.
>+//
>+// A leap second of ":60" is normalized to ":00" of the following minute
>+// with fractional seconds discarded.  The following table shows how the
>+// given seconds and subseconds will be parsed:
>+//
>+//   "59.x" -> 59.x  // exact
>+//   "60.x" -> 00.0  // normalized
>+//   "00.x" -> 00.x  // exact
>+//
>+// Errors are indicated by returning false and assigning an error message
>+// to the "err" out param if it is non-null.
>+//
>+// Note: If the input std::string is exactly "infinite-future", the returned
>+// `absl::Time` will be `absl::InfiniteFuture()` and `true` will be returned.
>+// If the input std::string is "infinite-past", the returned `absl::Time` will be
>+// `absl::InfinitePast()` and `true` will be returned.
>+//
>+bool ParseTime(const std::string& format, const std::string& input, Time* time,
>+               std::string* err);
>+
>+// Like ParseTime() above, but if the format std::string does not contain a UTC
>+// offset specification (%z/%Ez/%E*z) then the input is interpreted in the
>+// given TimeZone.  This means that the input, by itself, does not identify a
>+// unique instant.  Being time-zone dependent, it also admits the possibility
>+// of ambiguity or non-existence, in which case the "pre" time (as defined
>+// for ConvertDateTime()) is returned.  For these reasons we recommend that
>+// all date/time strings include a UTC offset so they're context independent.
>+bool ParseTime(const std::string& format, const std::string& input, TimeZone tz,
>+               Time* time, std::string* err);
>+
>+// Support for flag values of type Time. Time flags must be specified in a
>+// format that matches absl::RFC3339_full. For example:
>+//
>+//   --start_time=2016-01-02T03:04:05.678+08:00
>+//
>+// Note: A UTC offset (or 'Z' indicating a zero-offset from UTC) is required.
>+//
>+// Additionally, if you'd like to specify a time as a count of
>+// seconds/milliseconds/etc from the Unix epoch, use an absl::Duration flag
>+// and add that duration to absl::UnixEpoch() to get an absl::Time.
>+bool ParseFlag(const std::string& text, Time* t, std::string* error);
>+std::string UnparseFlag(Time t);
>+
>+// TimeZone
>+//
>+// The `absl::TimeZone` is an opaque, small, value-type class representing a
>+// geo-political region within which particular rules are used for converting
>+// between absolute and civil times (see https://git.io/v59Ly). `absl::TimeZone`
>+// values are named using the TZ identifiers from the IANA Time Zone Database,
>+// such as "America/Los_Angeles" or "Australia/Sydney". `absl::TimeZone` values
>+// are created from factory functions such as `absl::LoadTimeZone()`. Note:
>+// strings like "PST" and "EDT" are not valid TZ identifiers. Prefer to pass by
>+// value rather than const reference.
>+//
>+// For more on the fundamental concepts of time zones, absolute times, and civil
>+// times, see https://github.com/google/cctz#fundamental-concepts
>+//
>+// Examples:
>+//
>+//   absl::TimeZone utc = absl::UTCTimeZone();
>+//   absl::TimeZone pst = absl::FixedTimeZone(-8 * 60 * 60);
>+//   absl::TimeZone loc = absl::LocalTimeZone();
>+//   absl::TimeZone lax;
>+//   if (!absl::LoadTimeZone("America/Los_Angeles", &lax)) { ... }
>+//
>+// See also:
>+// - https://github.com/google/cctz
>+// - http://www.iana.org/time-zones
>+// - http://en.wikipedia.org/wiki/Zoneinfo
>+class TimeZone {
>+ public:
>+  explicit TimeZone(time_internal::cctz::time_zone tz) : cz_(tz) {}
>+  TimeZone() = default;  // UTC, but prefer UTCTimeZone() to be explicit.
>+  TimeZone(const TimeZone&) = default;
>+  TimeZone& operator=(const TimeZone&) = default;
>+
>+  explicit operator time_internal::cctz::time_zone() const { return cz_; }
>+
>+  std::string name() const { return cz_.name(); }
>+
>+ private:
>+  friend bool operator==(TimeZone a, TimeZone b) { return a.cz_ == b.cz_; }
>+  friend bool operator!=(TimeZone a, TimeZone b) { return a.cz_ != b.cz_; }
>+  friend std::ostream& operator<<(std::ostream& os, TimeZone tz) {
>+    return os << tz.name();
>+  }
>+
>+  time_internal::cctz::time_zone cz_;
>+};
>+
>+// LoadTimeZone()
>+//
>+// Loads the named zone. May perform I/O on the initial load of the named
>+// zone. If the name is invalid, or some other kind of error occurs, returns
>+// `false` and `*tz` is set to the UTC time zone.
>+inline bool LoadTimeZone(const std::string& name, TimeZone* tz) {
>+  if (name == "localtime") {
>+    *tz = TimeZone(time_internal::cctz::local_time_zone());
>+    return true;
>+  }
>+  time_internal::cctz::time_zone cz;
>+  const bool b = time_internal::cctz::load_time_zone(name, &cz);
>+  *tz = TimeZone(cz);
>+  return b;
>+}
>+
>+// FixedTimeZone()
>+//
>+// Returns a TimeZone that is a fixed offset (seconds east) from UTC.
>+// Note: If the absolute value of the offset is greater than 24 hours
>+// you'll get UTC (i.e., no offset) instead.
>+inline TimeZone FixedTimeZone(int seconds) {
>+  return TimeZone(
>+      time_internal::cctz::fixed_time_zone(std::chrono::seconds(seconds)));
>+}
>+
>+// UTCTimeZone()
>+//
>+// Convenience method returning the UTC time zone.
>+inline TimeZone UTCTimeZone() {
>+  return TimeZone(time_internal::cctz::utc_time_zone());
>+}
>+
>+// LocalTimeZone()
>+//
>+// Convenience method returning the local time zone, or UTC if there is
>+// no configured local zone.  Warning: Be wary of using LocalTimeZone(),
>+// and particularly so in a server process, as the zone configured for the
>+// local machine should be irrelevant.  Prefer an explicit zone name.
>+inline TimeZone LocalTimeZone() {
>+  return TimeZone(time_internal::cctz::local_time_zone());
>+}
>+
>+// ============================================================================
>+// Implementation Details Follow
>+// ============================================================================
>+
>+namespace time_internal {
>+
>+// Creates a Duration with a given representation.
>+// REQUIRES: hi,lo is a valid representation of a Duration as specified
>+// in time/duration.cc.
>+constexpr Duration MakeDuration(int64_t hi, uint32_t lo = 0) {
>+  return Duration(hi, lo);
>+}
>+
>+constexpr Duration MakeDuration(int64_t hi, int64_t lo) {
>+  return MakeDuration(hi, static_cast<uint32_t>(lo));
>+}
>+
>+// Make a Duration value from a floating-point number, as long as that number
>+// is in the range [ 0 .. numeric_limits<int64_t>::max ), that is, as long as
>+// it's positive and can be converted to int64_t without risk of UB.
>+inline Duration MakePosDoubleDuration(double n) {
>+  const int64_t int_secs = static_cast<int64_t>(n);
>+  const uint32_t ticks =
>+      static_cast<uint32_t>((n - int_secs) * kTicksPerSecond + 0.5);
>+  return ticks < kTicksPerSecond
>+             ? MakeDuration(int_secs, ticks)
>+             : MakeDuration(int_secs + 1, ticks - kTicksPerSecond);
>+}
>+
>+// Creates a normalized Duration from an almost-normalized (sec,ticks)
>+// pair. sec may be positive or negative.  ticks must be in the range
>+// -kTicksPerSecond < *ticks < kTicksPerSecond.  If ticks is negative it
>+// will be normalized to a positive value in the resulting Duration.
>+constexpr Duration MakeNormalizedDuration(int64_t sec, int64_t ticks) {
>+  return (ticks < 0) ? MakeDuration(sec - 1, ticks + kTicksPerSecond)
>+                     : MakeDuration(sec, ticks);
>+}
>+// Provide access to the Duration representation.
>+constexpr int64_t GetRepHi(Duration d) { return d.rep_hi_; }
>+constexpr uint32_t GetRepLo(Duration d) { return d.rep_lo_; }
>+constexpr bool IsInfiniteDuration(Duration d) { return GetRepLo(d) == ~0U; }
>+
>+// Returns an infinite Duration with the opposite sign.
>+// REQUIRES: IsInfiniteDuration(d)
>+constexpr Duration OppositeInfinity(Duration d) {
>+  return GetRepHi(d) < 0
>+             ? MakeDuration(std::numeric_limits<int64_t>::max(), ~0U)
>+             : MakeDuration(std::numeric_limits<int64_t>::min(), ~0U);
>+}
>+
>+// Returns (-n)-1 (equivalently -(n+1)) without avoidable overflow.
>+constexpr int64_t NegateAndSubtractOne(int64_t n) {
>+  // Note: Good compilers will optimize this expression to ~n when using
>+  // a two's-complement representation (which is required for int64_t).
>+  return (n < 0) ? -(n + 1) : (-n) - 1;
>+}
>+
>+// Map between a Time and a Duration since the Unix epoch.  Note that these
>+// functions depend on the above mentioned choice of the Unix epoch for the
>+// Time representation (and both need to be Time friends).  Without this
>+// knowledge, we would need to add-in/subtract-out UnixEpoch() respectively.
>+constexpr Time FromUnixDuration(Duration d) { return Time(d); }
>+constexpr Duration ToUnixDuration(Time t) { return t.rep_; }
>+
>+template <std::intmax_t N>
>+constexpr Duration FromInt64(int64_t v, std::ratio<1, N>) {
>+  static_assert(0 < N && N <= 1000 * 1000 * 1000, "Unsupported ratio");
>+  // Subsecond ratios cannot overflow.
>+  return MakeNormalizedDuration(
>+      v / N, v % N * kTicksPerNanosecond * 1000 * 1000 * 1000 / N);
>+}
>+constexpr Duration FromInt64(int64_t v, std::ratio<60>) {
>+  return (v <= std::numeric_limits<int64_t>::max() / 60 &&
>+          v >= std::numeric_limits<int64_t>::min() / 60)
>+             ? MakeDuration(v * 60)
>+             : v > 0 ? InfiniteDuration() : -InfiniteDuration();
>+}
>+constexpr Duration FromInt64(int64_t v, std::ratio<3600>) {
>+  return (v <= std::numeric_limits<int64_t>::max() / 3600 &&
>+          v >= std::numeric_limits<int64_t>::min() / 3600)
>+             ? MakeDuration(v * 3600)
>+             : v > 0 ? InfiniteDuration() : -InfiniteDuration();
>+}
>+
>+// IsValidRep64<T>(0) is true if the expression `int64_t{std::declval<T>()}` is
>+// valid. That is, if a T can be assigned to an int64_t without narrowing.
>+template <typename T>
>+constexpr auto IsValidRep64(int)
>+    -> decltype(int64_t{std::declval<T>()}, bool()) {
>+  return true;
>+}
>+template <typename T>
>+constexpr auto IsValidRep64(char) -> bool {
>+  return false;
>+}
>+
>+// Converts a std::chrono::duration to an absl::Duration.
>+template <typename Rep, typename Period>
>+constexpr Duration FromChrono(const std::chrono::duration<Rep, Period>& d) {
>+  static_assert(IsValidRep64<Rep>(0), "duration::rep is invalid");
>+  return FromInt64(int64_t{d.count()}, Period{});
>+}
>+
>+template <typename Ratio>
>+int64_t ToInt64(Duration d, Ratio) {
>+  // Note: This may be used on MSVC, which may have a system_clock period of
>+  // std::ratio<1, 10 * 1000 * 1000>
>+  return ToInt64Seconds(d * Ratio::den / Ratio::num);
>+}
>+// Fastpath implementations for the 6 common duration units.
>+inline int64_t ToInt64(Duration d, std::nano) {
>+  return ToInt64Nanoseconds(d);
>+}
>+inline int64_t ToInt64(Duration d, std::micro) {
>+  return ToInt64Microseconds(d);
>+}
>+inline int64_t ToInt64(Duration d, std::milli) {
>+  return ToInt64Milliseconds(d);
>+}
>+inline int64_t ToInt64(Duration d, std::ratio<1>) {
>+  return ToInt64Seconds(d);
>+}
>+inline int64_t ToInt64(Duration d, std::ratio<60>) {
>+  return ToInt64Minutes(d);
>+}
>+inline int64_t ToInt64(Duration d, std::ratio<3600>) {
>+  return ToInt64Hours(d);
>+}
>+
>+// Converts an absl::Duration to a chrono duration of type T.
>+template <typename T>
>+T ToChronoDuration(Duration d) {
>+  using Rep = typename T::rep;
>+  using Period = typename T::period;
>+  static_assert(IsValidRep64<Rep>(0), "duration::rep is invalid");
>+  if (time_internal::IsInfiniteDuration(d))
>+    return d < ZeroDuration() ? T::min() : T::max();
>+  const auto v = ToInt64(d, Period{});
>+  if (v > std::numeric_limits<Rep>::max()) return T::max();
>+  if (v < std::numeric_limits<Rep>::min()) return T::min();
>+  return T{v};
>+}
>+
>+}  // namespace time_internal
>+constexpr Duration Nanoseconds(int64_t n) {
>+  return time_internal::FromInt64(n, std::nano{});
>+}
>+constexpr Duration Microseconds(int64_t n) {
>+  return time_internal::FromInt64(n, std::micro{});
>+}
>+constexpr Duration Milliseconds(int64_t n) {
>+  return time_internal::FromInt64(n, std::milli{});
>+}
>+constexpr Duration Seconds(int64_t n) {
>+  return time_internal::FromInt64(n, std::ratio<1>{});
>+}
>+constexpr Duration Minutes(int64_t n) {
>+  return time_internal::FromInt64(n, std::ratio<60>{});
>+}
>+constexpr Duration Hours(int64_t n) {
>+  return time_internal::FromInt64(n, std::ratio<3600>{});
>+}
>+
>+constexpr bool operator<(Duration lhs, Duration rhs) {
>+  return time_internal::GetRepHi(lhs) != time_internal::GetRepHi(rhs)
>+             ? time_internal::GetRepHi(lhs) < time_internal::GetRepHi(rhs)
>+             : time_internal::GetRepHi(lhs) == std::numeric_limits<int64_t>::min()
>+                   ? time_internal::GetRepLo(lhs) + 1 <
>+                         time_internal::GetRepLo(rhs) + 1
>+                   : time_internal::GetRepLo(lhs) <
>+                         time_internal::GetRepLo(rhs);
>+}
>+
>+constexpr bool operator==(Duration lhs, Duration rhs) {
>+  return time_internal::GetRepHi(lhs) == time_internal::GetRepHi(rhs) &&
>+         time_internal::GetRepLo(lhs) == time_internal::GetRepLo(rhs);
>+}
>+
>+constexpr Duration operator-(Duration d) {
>+  // This is a little interesting because of the special cases.
>+  //
>+  // If rep_lo_ is zero, we have it easy; it's safe to negate rep_hi_, we're
>+  // dealing with an integral number of seconds, and the only special case is
>+  // the maximum negative finite duration, which can't be negated.
>+  //
>+  // Infinities stay infinite, and just change direction.
>+  //
>+  // Finally we're in the case where rep_lo_ is non-zero, and we can borrow
>+  // a second's worth of ticks and avoid overflow (as negating int64_t-min + 1
>+  // is safe).
>+  return time_internal::GetRepLo(d) == 0
>+             ? time_internal::GetRepHi(d) == std::numeric_limits<int64_t>::min()
>+                   ? InfiniteDuration()
>+                   : time_internal::MakeDuration(-time_internal::GetRepHi(d))
>+             : time_internal::IsInfiniteDuration(d)
>+                   ? time_internal::OppositeInfinity(d)
>+                   : time_internal::MakeDuration(
>+                         time_internal::NegateAndSubtractOne(
>+                             time_internal::GetRepHi(d)),
>+                         time_internal::kTicksPerSecond -
>+                             time_internal::GetRepLo(d));
>+}
>+
>+constexpr Duration InfiniteDuration() {
>+  return time_internal::MakeDuration(std::numeric_limits<int64_t>::max(), ~0U);
>+}
>+
>+constexpr Duration FromChrono(const std::chrono::nanoseconds& d) {
>+  return time_internal::FromChrono(d);
>+}
>+constexpr Duration FromChrono(const std::chrono::microseconds& d) {
>+  return time_internal::FromChrono(d);
>+}
>+constexpr Duration FromChrono(const std::chrono::milliseconds& d) {
>+  return time_internal::FromChrono(d);
>+}
>+constexpr Duration FromChrono(const std::chrono::seconds& d) {
>+  return time_internal::FromChrono(d);
>+}
>+constexpr Duration FromChrono(const std::chrono::minutes& d) {
>+  return time_internal::FromChrono(d);
>+}
>+constexpr Duration FromChrono(const std::chrono::hours& d) {
>+  return time_internal::FromChrono(d);
>+}
>+
>+constexpr Time FromUnixNanos(int64_t ns) {
>+  return time_internal::FromUnixDuration(Nanoseconds(ns));
>+}
>+
>+constexpr Time FromUnixMicros(int64_t us) {
>+  return time_internal::FromUnixDuration(Microseconds(us));
>+}
>+
>+constexpr Time FromUnixMillis(int64_t ms) {
>+  return time_internal::FromUnixDuration(Milliseconds(ms));
>+}
>+
>+constexpr Time FromUnixSeconds(int64_t s) {
>+  return time_internal::FromUnixDuration(Seconds(s));
>+}
>+
>+constexpr Time FromTimeT(time_t t) {
>+  return time_internal::FromUnixDuration(Seconds(t));
>+}
>+
>+}  // namespace absl
>+
>+#endif  // ABSL_TIME_TIME_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/time_benchmark.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/time_benchmark.cc
>new file mode 100644
>index 00000000000..e1009946cd5
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/time_benchmark.cc
>@@ -0,0 +1,316 @@
>+// Copyright 2018 The Abseil Authors.
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/time/time.h"
>+
>+#if !defined(_WIN32)
>+#include <sys/time.h>
>+#endif  // _WIN32
>+#include <algorithm>
>+#include <cmath>
>+#include <cstddef>
>+#include <cstring>
>+#include <ctime>
>+#include <memory>
>+#include <string>
>+
>+#include "absl/time/clock.h"
>+#include "absl/time/internal/test_util.h"
>+#include "benchmark/benchmark.h"
>+
>+namespace {
>+
>+//
>+// Addition/Subtraction of a duration
>+//
>+
>+void BM_Time_Arithmetic(benchmark::State& state) {
>+  const absl::Duration nano = absl::Nanoseconds(1);
>+  const absl::Duration sec = absl::Seconds(1);
>+  absl::Time t = absl::UnixEpoch();
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(t += nano);
>+    benchmark::DoNotOptimize(t -= sec);
>+  }
>+}
>+BENCHMARK(BM_Time_Arithmetic);
>+
>+//
>+// Time difference
>+//
>+
>+void BM_Time_Difference(benchmark::State& state) {
>+  absl::Time start = absl::Now();
>+  absl::Time end = start + absl::Nanoseconds(1);
>+  absl::Duration diff;
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(diff += end - start);
>+  }
>+}
>+BENCHMARK(BM_Time_Difference);
>+
>+//
>+// ToDateTime
>+//
>+// In each "ToDateTime" benchmark we switch between two instants
>+// separated by at least one transition in order to defeat any
>+// internal caching of previous results (e.g., see local_time_hint_).
>+//
>+// The "UTC" variants use UTC instead of the Google/local time zone.
>+//
>+
>+void BM_Time_ToDateTime_Absl(benchmark::State& state) {
>+  const absl::TimeZone tz =
>+      absl::time_internal::LoadTimeZone("America/Los_Angeles");
>+  absl::Time t = absl::FromUnixSeconds(1384569027);
>+  absl::Time t2 = absl::FromUnixSeconds(1418962578);
>+  while (state.KeepRunning()) {
>+    std::swap(t, t2);
>+    t += absl::Seconds(1);
>+    benchmark::DoNotOptimize(t.In(tz));
>+  }
>+}
>+BENCHMARK(BM_Time_ToDateTime_Absl);
>+
>+void BM_Time_ToDateTime_Libc(benchmark::State& state) {
>+  // No timezone support, so just use localtime.
>+  time_t t = 1384569027;
>+  time_t t2 = 1418962578;
>+  while (state.KeepRunning()) {
>+    std::swap(t, t2);
>+    t += 1;
>+    struct tm tm;
>+#if !defined(_WIN32)
>+    benchmark::DoNotOptimize(localtime_r(&t, &tm));
>+#else   // _WIN32
>+    benchmark::DoNotOptimize(localtime_s(&tm, &t));
>+#endif  // _WIN32
>+  }
>+}
>+BENCHMARK(BM_Time_ToDateTime_Libc);
>+
>+void BM_Time_ToDateTimeUTC_Absl(benchmark::State& state) {
>+  const absl::TimeZone tz = absl::UTCTimeZone();
>+  absl::Time t = absl::FromUnixSeconds(1384569027);
>+  while (state.KeepRunning()) {
>+    t += absl::Seconds(1);
>+    benchmark::DoNotOptimize(t.In(tz));
>+  }
>+}
>+BENCHMARK(BM_Time_ToDateTimeUTC_Absl);
>+
>+void BM_Time_ToDateTimeUTC_Libc(benchmark::State& state) {
>+  time_t t = 1384569027;
>+  while (state.KeepRunning()) {
>+    t += 1;
>+    struct tm tm;
>+#if !defined(_WIN32)
>+    benchmark::DoNotOptimize(gmtime_r(&t, &tm));
>+#else   // _WIN32
>+    benchmark::DoNotOptimize(gmtime_s(&tm, &t));
>+#endif  // _WIN32
>+  }
>+}
>+BENCHMARK(BM_Time_ToDateTimeUTC_Libc);
>+
>+//
>+// FromUnixMicros
>+//
>+
>+void BM_Time_FromUnixMicros(benchmark::State& state) {
>+  int i = 0;
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(absl::FromUnixMicros(i));
>+    ++i;
>+  }
>+}
>+BENCHMARK(BM_Time_FromUnixMicros);
>+
>+void BM_Time_ToUnixNanos(benchmark::State& state) {
>+  const absl::Time t = absl::UnixEpoch() + absl::Seconds(123);
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(ToUnixNanos(t));
>+  }
>+}
>+BENCHMARK(BM_Time_ToUnixNanos);
>+
>+void BM_Time_ToUnixMicros(benchmark::State& state) {
>+  const absl::Time t = absl::UnixEpoch() + absl::Seconds(123);
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(ToUnixMicros(t));
>+  }
>+}
>+BENCHMARK(BM_Time_ToUnixMicros);
>+
>+void BM_Time_ToUnixMillis(benchmark::State& state) {
>+  const absl::Time t = absl::UnixEpoch() + absl::Seconds(123);
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(ToUnixMillis(t));
>+  }
>+}
>+BENCHMARK(BM_Time_ToUnixMillis);
>+
>+void BM_Time_ToUnixSeconds(benchmark::State& state) {
>+  const absl::Time t = absl::UnixEpoch() + absl::Seconds(123);
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(absl::ToUnixSeconds(t));
>+  }
>+}
>+BENCHMARK(BM_Time_ToUnixSeconds);
>+
>+//
>+// FromDateTime
>+//
>+// In each "FromDateTime" benchmark we switch between two YMDhms
>+// values separated by at least one transition in order to defeat any
>+// internal caching of previous results (e.g., see time_local_hint_).
>+//
>+// The "UTC" variants use UTC instead of the Google/local time zone.
>+// The "Day0" variants require normalization of the day of month.
>+//
>+
>+void BM_Time_FromDateTime_Absl(benchmark::State& state) {
>+  const absl::TimeZone tz =
>+      absl::time_internal::LoadTimeZone("America/Los_Angeles");
>+  int i = 0;
>+  while (state.KeepRunning()) {
>+    if ((i & 1) == 0) {
>+      absl::FromDateTime(2014, 12, 18, 20, 16, 18, tz);
>+    } else {
>+      absl::FromDateTime(2013, 11, 15, 18, 30, 27, tz);
>+    }
>+    ++i;
>+  }
>+}
>+BENCHMARK(BM_Time_FromDateTime_Absl);
>+
>+void BM_Time_FromDateTime_Libc(benchmark::State& state) {
>+  // No timezone support, so just use localtime.
>+  int i = 0;
>+  while (state.KeepRunning()) {
>+    struct tm tm;
>+    if ((i & 1) == 0) {
>+      tm.tm_year = 2014 - 1900;
>+      tm.tm_mon = 12 - 1;
>+      tm.tm_mday = 18;
>+      tm.tm_hour = 20;
>+      tm.tm_min = 16;
>+      tm.tm_sec = 18;
>+    } else {
>+      tm.tm_year = 2013 - 1900;
>+      tm.tm_mon = 11 - 1;
>+      tm.tm_mday = 15;
>+      tm.tm_hour = 18;
>+      tm.tm_min = 30;
>+      tm.tm_sec = 27;
>+    }
>+    tm.tm_isdst = -1;
>+    mktime(&tm);
>+    ++i;
>+  }
>+}
>+BENCHMARK(BM_Time_FromDateTime_Libc);
>+
>+void BM_Time_FromDateTimeUTC_Absl(benchmark::State& state) {
>+  const absl::TimeZone tz = absl::UTCTimeZone();
>+  while (state.KeepRunning()) {
>+    FromDateTime(2014, 12, 18, 20, 16, 18, tz);
>+  }
>+}
>+BENCHMARK(BM_Time_FromDateTimeUTC_Absl);
>+
>+void BM_Time_FromDateTimeDay0_Absl(benchmark::State& state) {
>+  const absl::TimeZone tz =
>+      absl::time_internal::LoadTimeZone("America/Los_Angeles");
>+  int i = 0;
>+  while (state.KeepRunning()) {
>+    if ((i & 1) == 0) {
>+      absl::FromDateTime(2014, 12, 0, 20, 16, 18, tz);
>+    } else {
>+      absl::FromDateTime(2013, 11, 0, 18, 30, 27, tz);
>+    }
>+    ++i;
>+  }
>+}
>+BENCHMARK(BM_Time_FromDateTimeDay0_Absl);
>+
>+void BM_Time_FromDateTimeDay0_Libc(benchmark::State& state) {
>+  // No timezone support, so just use localtime.
>+  int i = 0;
>+  while (state.KeepRunning()) {
>+    struct tm tm;
>+    if ((i & 1) == 0) {
>+      tm.tm_year = 2014 - 1900;
>+      tm.tm_mon = 12 - 1;
>+      tm.tm_mday = 0;
>+      tm.tm_hour = 20;
>+      tm.tm_min = 16;
>+      tm.tm_sec = 18;
>+    } else {
>+      tm.tm_year = 2013 - 1900;
>+      tm.tm_mon = 11 - 1;
>+      tm.tm_mday = 0;
>+      tm.tm_hour = 18;
>+      tm.tm_min = 30;
>+      tm.tm_sec = 27;
>+    }
>+    tm.tm_isdst = -1;
>+    mktime(&tm);
>+    ++i;
>+  }
>+}
>+BENCHMARK(BM_Time_FromDateTimeDay0_Libc);
>+
>+//
>+// To/FromTimespec
>+//
>+
>+void BM_Time_ToTimespec(benchmark::State& state) {
>+  absl::Time now = absl::Now();
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(absl::ToTimespec(now));
>+  }
>+}
>+BENCHMARK(BM_Time_ToTimespec);
>+
>+void BM_Time_FromTimespec(benchmark::State& state) {
>+  timespec ts = absl::ToTimespec(absl::Now());
>+  while (state.KeepRunning()) {
>+    if (++ts.tv_nsec == 1000 * 1000 * 1000) {
>+      ++ts.tv_sec;
>+      ts.tv_nsec = 0;
>+    }
>+    benchmark::DoNotOptimize(absl::TimeFromTimespec(ts));
>+  }
>+}
>+BENCHMARK(BM_Time_FromTimespec);
>+
>+//
>+// Comparison with InfiniteFuture/Past
>+//
>+
>+void BM_Time_InfiniteFuture(benchmark::State& state) {
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(absl::InfiniteFuture());
>+  }
>+}
>+BENCHMARK(BM_Time_InfiniteFuture);
>+
>+void BM_Time_InfinitePast(benchmark::State& state) {
>+  while (state.KeepRunning()) {
>+    benchmark::DoNotOptimize(absl::InfinitePast());
>+  }
>+}
>+BENCHMARK(BM_Time_InfinitePast);
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/time_norm_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/time_norm_test.cc
>new file mode 100644
>index 00000000000..4436242e412
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/time_norm_test.cc
>@@ -0,0 +1,306 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+// This file contains tests for FromDateTime() normalization, which is
>+// time-zone independent so we just use UTC throughout.
>+
>+#include <cstdint>
>+#include <limits>
>+
>+#include "gmock/gmock.h"
>+#include "gtest/gtest.h"
>+#include "absl/time/internal/test_util.h"
>+#include "absl/time/time.h"
>+
>+namespace {
>+
>+TEST(TimeNormCase, SimpleOverflow) {
>+  const absl::TimeZone utc = absl::UTCTimeZone();
>+
>+  absl::TimeConversion tc =
>+      absl::ConvertDateTime(2013, 11, 15, 16, 32, 59 + 1, utc);
>+  EXPECT_TRUE(tc.normalized);
>+  EXPECT_EQ(absl::TimeConversion::UNIQUE, tc.kind);
>+  absl::Time::Breakdown bd = tc.pre.In(utc);
>+  ABSL_INTERNAL_EXPECT_TIME(bd, 2013, 11, 15, 16, 33, 0, 0, false);
>+
>+  tc = absl::ConvertDateTime(2013, 11, 15, 16, 59 + 1, 14, utc);
>+  EXPECT_TRUE(tc.normalized);
>+  EXPECT_EQ(absl::TimeConversion::UNIQUE, tc.kind);
>+  bd = tc.pre.In(utc);
>+  ABSL_INTERNAL_EXPECT_TIME(bd, 2013, 11, 15, 17, 0, 14, 0, false);
>+
>+  tc = absl::ConvertDateTime(2013, 11, 15, 23 + 1, 32, 14, utc);
>+  EXPECT_TRUE(tc.normalized);
>+  EXPECT_EQ(absl::TimeConversion::UNIQUE, tc.kind);
>+  bd = tc.pre.In(utc);
>+  ABSL_INTERNAL_EXPECT_TIME(bd, 2013, 11, 16, 0, 32, 14, 0, false);
>+
>+  tc = absl::ConvertDateTime(2013, 11, 30 + 1, 16, 32, 14, utc);
>+  EXPECT_TRUE(tc.normalized);
>+  EXPECT_EQ(absl::TimeConversion::UNIQUE, tc.kind);
>+  bd = tc.pre.In(utc);
>+  ABSL_INTERNAL_EXPECT_TIME(bd, 2013, 12, 1, 16, 32, 14, 0, false);
>+
>+  tc = absl::ConvertDateTime(2013, 12 + 1, 15, 16, 32, 14, utc);
>+  EXPECT_TRUE(tc.normalized);
>+  EXPECT_EQ(absl::TimeConversion::UNIQUE, tc.kind);
>+  bd = tc.pre.In(utc);
>+  ABSL_INTERNAL_EXPECT_TIME(bd, 2014, 1, 15, 16, 32, 14, 0, false);
>+}
>+
>+TEST(TimeNormCase, SimpleUnderflow) {
>+  const absl::TimeZone utc = absl::UTCTimeZone();
>+
>+  absl::TimeConversion tc = ConvertDateTime(2013, 11, 15, 16, 32, 0 - 1, utc);
>+  EXPECT_TRUE(tc.normalized);
>+  EXPECT_EQ(absl::TimeConversion::UNIQUE, tc.kind);
>+  absl::Time::Breakdown bd = tc.pre.In(utc);
>+  ABSL_INTERNAL_EXPECT_TIME(bd, 2013, 11, 15, 16, 31, 59, 0, false);
>+
>+  tc = ConvertDateTime(2013, 11, 15, 16, 0 - 1, 14, utc);
>+  EXPECT_TRUE(tc.normalized);
>+  EXPECT_EQ(absl::TimeConversion::UNIQUE, tc.kind);
>+  bd = tc.pre.In(utc);
>+  ABSL_INTERNAL_EXPECT_TIME(bd, 2013, 11, 15, 15, 59, 14, 0, false);
>+
>+  tc = ConvertDateTime(2013, 11, 15, 0 - 1, 32, 14, utc);
>+  EXPECT_TRUE(tc.normalized);
>+  EXPECT_EQ(absl::TimeConversion::UNIQUE, tc.kind);
>+  bd = tc.pre.In(utc);
>+  ABSL_INTERNAL_EXPECT_TIME(bd, 2013, 11, 14, 23, 32, 14, 0, false);
>+
>+  tc = ConvertDateTime(2013, 11, 1 - 1, 16, 32, 14, utc);
>+  EXPECT_TRUE(tc.normalized);
>+  EXPECT_EQ(absl::TimeConversion::UNIQUE, tc.kind);
>+  bd = tc.pre.In(utc);
>+  ABSL_INTERNAL_EXPECT_TIME(bd, 2013, 10, 31, 16, 32, 14, 0, false);
>+
>+  tc = ConvertDateTime(2013, 1 - 1, 15, 16, 32, 14, utc);
>+  EXPECT_TRUE(tc.normalized);
>+  EXPECT_EQ(absl::TimeConversion::UNIQUE, tc.kind);
>+  bd = tc.pre.In(utc);
>+  ABSL_INTERNAL_EXPECT_TIME(bd, 2012, 12, 15, 16, 32, 14, 0, false);
>+}
>+
>+TEST(TimeNormCase, MultipleOverflow) {
>+  const absl::TimeZone utc = absl::UTCTimeZone();
>+  absl::TimeConversion tc = ConvertDateTime(2013, 12, 31, 23, 59, 59 + 1, utc);
>+  EXPECT_TRUE(tc.normalized);
>+  EXPECT_EQ(absl::TimeConversion::UNIQUE, tc.kind);
>+  absl::Time::Breakdown bd = tc.pre.In(utc);
>+  ABSL_INTERNAL_EXPECT_TIME(bd, 2014, 1, 1, 0, 0, 0, 0, false);
>+}
>+
>+TEST(TimeNormCase, MultipleUnderflow) {
>+  const absl::TimeZone utc = absl::UTCTimeZone();
>+  absl::TimeConversion tc = absl::ConvertDateTime(2014, 1, 1, 0, 0, 0 - 1, utc);
>+  EXPECT_TRUE(tc.normalized);
>+  EXPECT_EQ(absl::TimeConversion::UNIQUE, tc.kind);
>+  absl::Time::Breakdown bd = tc.pre.In(utc);
>+  ABSL_INTERNAL_EXPECT_TIME(bd, 2013, 12, 31, 23, 59, 59, 0, false);
>+}
>+
>+TEST(TimeNormCase, OverflowLimits) {
>+  const absl::TimeZone utc = absl::UTCTimeZone();
>+  absl::TimeConversion tc;
>+  absl::Time::Breakdown bd;
>+
>+  const int kintmax = std::numeric_limits<int>::max();
>+  tc = absl::ConvertDateTime(0, kintmax, kintmax, kintmax, kintmax, kintmax,
>+                             utc);
>+  EXPECT_TRUE(tc.normalized);
>+  EXPECT_EQ(absl::TimeConversion::UNIQUE, tc.kind);
>+  bd = tc.pre.In(utc);
>+  ABSL_INTERNAL_EXPECT_TIME(bd, 185085715, 11, 27, 12, 21, 7, 0, false);
>+
>+  const int kintmin = std::numeric_limits<int>::min();
>+  tc = absl::ConvertDateTime(0, kintmin, kintmin, kintmin, kintmin, kintmin,
>+                             utc);
>+  EXPECT_TRUE(tc.normalized);
>+  EXPECT_EQ(absl::TimeConversion::UNIQUE, tc.kind);
>+  bd = tc.pre.In(utc);
>+  ABSL_INTERNAL_EXPECT_TIME(bd, -185085717, 10, 31, 10, 37, 52, 0, false);
>+
>+  const int64_t max_year = std::numeric_limits<int64_t>::max();
>+  tc = absl::ConvertDateTime(max_year, 12, 31, 23, 59, 59, utc);
>+  EXPECT_TRUE(tc.normalized);
>+  EXPECT_EQ(absl::TimeConversion::UNIQUE, tc.kind);
>+  EXPECT_EQ(absl::InfiniteFuture(), tc.pre);
>+
>+  const int64_t min_year = std::numeric_limits<int64_t>::min();
>+  tc = absl::ConvertDateTime(min_year, 1, 1, 0, 0, 0, utc);
>+  EXPECT_TRUE(tc.normalized);
>+  EXPECT_EQ(absl::TimeConversion::UNIQUE, tc.kind);
>+  EXPECT_EQ(absl::InfinitePast(), tc.pre);
>+}
>+
>+TEST(TimeNormCase, ComplexOverflow) {
>+  const absl::TimeZone utc = absl::UTCTimeZone();
>+
>+  absl::TimeConversion tc =
>+      ConvertDateTime(2013, 11, 15, 16, 32, 14 + 123456789, utc);
>+  EXPECT_TRUE(tc.normalized);
>+  EXPECT_EQ(absl::TimeConversion::UNIQUE, tc.kind);
>+  absl::Time::Breakdown bd = tc.pre.In(utc);
>+  ABSL_INTERNAL_EXPECT_TIME(bd, 2017, 10, 14, 14, 5, 23, 0, false);
>+
>+  tc = absl::ConvertDateTime(2013, 11, 15, 16, 32 + 1234567, 14, utc);
>+  EXPECT_TRUE(tc.normalized);
>+  EXPECT_EQ(absl::TimeConversion::UNIQUE, tc.kind);
>+  bd = tc.pre.In(utc);
>+  ABSL_INTERNAL_EXPECT_TIME(bd, 2016, 3, 22, 0, 39, 14, 0, false);
>+
>+  tc = absl::ConvertDateTime(2013, 11, 15, 16 + 123456, 32, 14, utc);
>+  EXPECT_TRUE(tc.normalized);
>+  EXPECT_EQ(absl::TimeConversion::UNIQUE, tc.kind);
>+  bd = tc.pre.In(utc);
>+  ABSL_INTERNAL_EXPECT_TIME(bd, 2027, 12, 16, 16, 32, 14, 0, false);
>+
>+  tc = absl::ConvertDateTime(2013, 11, 15 + 1234, 16, 32, 14, utc);
>+  EXPECT_TRUE(tc.normalized);
>+  EXPECT_EQ(absl::TimeConversion::UNIQUE, tc.kind);
>+  bd = tc.pre.In(utc);
>+  ABSL_INTERNAL_EXPECT_TIME(bd, 2017, 4, 2, 16, 32, 14, 0, false);
>+
>+  tc = absl::ConvertDateTime(2013, 11 + 123, 15, 16, 32, 14, utc);
>+  EXPECT_TRUE(tc.normalized);
>+  EXPECT_EQ(absl::TimeConversion::UNIQUE, tc.kind);
>+  bd = tc.pre.In(utc);
>+  ABSL_INTERNAL_EXPECT_TIME(bd, 2024, 2, 15, 16, 32, 14, 0, false);
>+}
>+
>+TEST(TimeNormCase, ComplexUnderflow) {
>+  const absl::TimeZone utc = absl::UTCTimeZone();
>+
>+  absl::TimeConversion tc =
>+      absl::ConvertDateTime(1999, 3, 0, 0, 0, 0, utc);  // year 400
>+  EXPECT_TRUE(tc.normalized);
>+  EXPECT_EQ(absl::TimeConversion::UNIQUE, tc.kind);
>+  absl::Time::Breakdown bd = tc.pre.In(utc);
>+  ABSL_INTERNAL_EXPECT_TIME(bd, 1999, 2, 28, 0, 0, 0, 0, false);
>+
>+  tc = absl::ConvertDateTime(2013, 11, 15, 16, 32, 14 - 123456789, utc);
>+  EXPECT_TRUE(tc.normalized);
>+  EXPECT_EQ(absl::TimeConversion::UNIQUE, tc.kind);
>+  bd = tc.pre.In(utc);
>+  ABSL_INTERNAL_EXPECT_TIME(bd, 2009, 12, 17, 18, 59, 5, 0, false);
>+
>+  tc = absl::ConvertDateTime(2013, 11, 15, 16, 32 - 1234567, 14, utc);
>+  EXPECT_TRUE(tc.normalized);
>+  EXPECT_EQ(absl::TimeConversion::UNIQUE, tc.kind);
>+  bd = tc.pre.In(utc);
>+  ABSL_INTERNAL_EXPECT_TIME(bd, 2011, 7, 12, 8, 25, 14, 0, false);
>+
>+  tc = absl::ConvertDateTime(2013, 11, 15, 16 - 123456, 32, 14, utc);
>+  EXPECT_TRUE(tc.normalized);
>+  EXPECT_EQ(absl::TimeConversion::UNIQUE, tc.kind);
>+  bd = tc.pre.In(utc);
>+  ABSL_INTERNAL_EXPECT_TIME(bd, 1999, 10, 16, 16, 32, 14, 0, false);
>+
>+  tc = absl::ConvertDateTime(2013, 11, 15 - 1234, 16, 32, 14, utc);
>+  EXPECT_TRUE(tc.normalized);
>+  EXPECT_EQ(absl::TimeConversion::UNIQUE, tc.kind);
>+  bd = tc.pre.In(utc);
>+  ABSL_INTERNAL_EXPECT_TIME(bd, 2010, 6, 30, 16, 32, 14, 0, false);
>+
>+  tc = absl::ConvertDateTime(2013, 11 - 123, 15, 16, 32, 14, utc);
>+  EXPECT_TRUE(tc.normalized);
>+  EXPECT_EQ(absl::TimeConversion::UNIQUE, tc.kind);
>+  bd = tc.pre.In(utc);
>+  ABSL_INTERNAL_EXPECT_TIME(bd, 2003, 8, 15, 16, 32, 14, 0, false);
>+}
>+
>+TEST(TimeNormCase, Mishmash) {
>+  const absl::TimeZone utc = absl::UTCTimeZone();
>+
>+  absl::TimeConversion tc =
>+      absl::ConvertDateTime(2013, 11 - 123, 15 + 1234, 16 - 123456,
>+                            32 + 1234567, 14 - 123456789, utc);
>+  EXPECT_TRUE(tc.normalized);
>+  EXPECT_EQ(absl::TimeConversion::UNIQUE, tc.kind);
>+  absl::Time::Breakdown bd = tc.pre.In(utc);
>+  ABSL_INTERNAL_EXPECT_TIME(bd, 1991, 5, 9, 3, 6, 5, 0, false);
>+
>+  tc = absl::ConvertDateTime(2013, 11 + 123, 15 - 1234, 16 + 123456,
>+                             32 - 1234567, 14 + 123456789, utc);
>+  EXPECT_TRUE(tc.normalized);
>+  EXPECT_EQ(absl::TimeConversion::UNIQUE, tc.kind);
>+  bd = tc.pre.In(utc);
>+  ABSL_INTERNAL_EXPECT_TIME(bd, 2036, 5, 24, 5, 58, 23, 0, false);
>+
>+  // Here is a normalization case we got wrong for a while.  Because the
>+  // day is converted to "1" within a 400-year (146097-day) period, we
>+  // didn't need to roll the month and so we didn't mark it as normalized.
>+  tc = absl::ConvertDateTime(2013, 11, -146097 + 1, 16, 32, 14, utc);
>+  EXPECT_TRUE(tc.normalized);
>+  EXPECT_EQ(absl::TimeConversion::UNIQUE, tc.kind);
>+  bd = tc.pre.In(utc);
>+  ABSL_INTERNAL_EXPECT_TIME(bd, 1613, 11, 1, 16, 32, 14, 0, false);
>+
>+  // Even though the month overflow compensates for the day underflow,
>+  // this should still be marked as normalized.
>+  tc = absl::ConvertDateTime(2013, 11 + 400 * 12, -146097 + 1, 16, 32, 14, utc);
>+  EXPECT_TRUE(tc.normalized);
>+  EXPECT_EQ(absl::TimeConversion::UNIQUE, tc.kind);
>+  bd = tc.pre.In(utc);
>+  ABSL_INTERNAL_EXPECT_TIME(bd, 2013, 11, 1, 16, 32, 14, 0, false);
>+}
>+
>+TEST(TimeNormCase, LeapYears) {
>+  const absl::TimeZone utc = absl::UTCTimeZone();
>+
>+  absl::TimeConversion tc =
>+      absl::ConvertDateTime(2013, 2, 28 + 1, 0, 0, 0, utc);
>+  EXPECT_TRUE(tc.normalized);
>+  EXPECT_EQ(absl::TimeConversion::UNIQUE, tc.kind);
>+  absl::Time::Breakdown bd = tc.pre.In(utc);
>+  ABSL_INTERNAL_EXPECT_TIME(bd, 2013, 3, 1, 0, 0, 0, 0, false);
>+
>+  tc = absl::ConvertDateTime(2012, 2, 28 + 1, 0, 0, 0, utc);
>+  EXPECT_FALSE(tc.normalized);
>+  EXPECT_EQ(absl::TimeConversion::UNIQUE, tc.kind);
>+  bd = tc.pre.In(utc);
>+  ABSL_INTERNAL_EXPECT_TIME(bd, 2012, 2, 29, 0, 0, 0, 0, false);
>+
>+  tc = absl::ConvertDateTime(2000, 2, 28 + 1, 0, 0, 0, utc);
>+  EXPECT_FALSE(tc.normalized);
>+  EXPECT_EQ(absl::TimeConversion::UNIQUE, tc.kind);
>+  bd = tc.pre.In(utc);
>+  ABSL_INTERNAL_EXPECT_TIME(bd, 2000, 2, 29, 0, 0, 0, 0, false);
>+
>+  tc = absl::ConvertDateTime(1900, 2, 28 + 1, 0, 0, 0, utc);
>+  EXPECT_TRUE(tc.normalized);
>+  EXPECT_EQ(absl::TimeConversion::UNIQUE, tc.kind);
>+  bd = tc.pre.In(utc);
>+  ABSL_INTERNAL_EXPECT_TIME(bd, 1900, 3, 1, 0, 0, 0, 0, false);
>+}
>+
>+// Convert all the days from 1970-1-1 to 1970-1-146097 (aka 2369-12-31)
>+// and check that they normalize to the expected time.  146097 days span
>+// the 400-year Gregorian cycle used during normalization.
>+TEST(TimeNormCase, AllTheDays) {
>+  const absl::TimeZone utc = absl::UTCTimeZone();
>+  absl::Time exp_time = absl::UnixEpoch();
>+
>+  for (int day = 1; day <= 146097; ++day) {
>+    absl::TimeConversion tc = absl::ConvertDateTime(1970, 1, day, 0, 0, 0, utc);
>+    EXPECT_EQ(day > 31, tc.normalized);
>+    EXPECT_EQ(absl::TimeConversion::UNIQUE, tc.kind);
>+    EXPECT_EQ(exp_time, tc.pre);
>+    exp_time += absl::Hours(24);
>+  }
>+}
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/time_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/time_test.cc
>new file mode 100644
>index 00000000000..4f8f58a6ef3
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/time_test.cc
>@@ -0,0 +1,1088 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/time/time.h"
>+
>+#include <chrono>  // NOLINT(build/c++11)
>+#include <cstring>
>+#include <ctime>
>+#include <iomanip>
>+#include <limits>
>+#include <string>
>+
>+#include "gmock/gmock.h"
>+#include "gtest/gtest.h"
>+#include "absl/time/clock.h"
>+#include "absl/time/internal/test_util.h"
>+
>+namespace {
>+
>+// A gMock matcher to match timespec values. Use this matcher like:
>+// timespec ts1, ts2;
>+// EXPECT_THAT(ts1, TimespecMatcher(ts2));
>+MATCHER_P(TimespecMatcher, ts, "") {
>+  if (ts.tv_sec == arg.tv_sec && ts.tv_nsec == arg.tv_nsec)
>+    return true;
>+  *result_listener << "expected: {" << ts.tv_sec << ", " << ts.tv_nsec << "} ";
>+  *result_listener << "actual: {" << arg.tv_sec << ", " << arg.tv_nsec << "}";
>+  return false;
>+}
>+
>+// A gMock matcher to match timeval values. Use this matcher like:
>+// timeval tv1, tv2;
>+// EXPECT_THAT(tv1, TimevalMatcher(tv2));
>+MATCHER_P(TimevalMatcher, tv, "") {
>+  if (tv.tv_sec == arg.tv_sec && tv.tv_usec == arg.tv_usec)
>+    return true;
>+  *result_listener << "expected: {" << tv.tv_sec << ", " << tv.tv_usec << "} ";
>+  *result_listener << "actual: {" << arg.tv_sec << ", " << arg.tv_usec << "}";
>+  return false;
>+}
>+
>+TEST(Time, ConstExpr) {
>+  constexpr absl::Time t0 = absl::UnixEpoch();
>+  static_assert(t0 == absl::Time(), "UnixEpoch");
>+  constexpr absl::Time t1 = absl::InfiniteFuture();
>+  static_assert(t1 != absl::Time(), "InfiniteFuture");
>+  constexpr absl::Time t2 = absl::InfinitePast();
>+  static_assert(t2 != absl::Time(), "InfinitePast");
>+  constexpr absl::Time t3 = absl::FromUnixNanos(0);
>+  static_assert(t3 == absl::Time(), "FromUnixNanos");
>+  constexpr absl::Time t4 = absl::FromUnixMicros(0);
>+  static_assert(t4 == absl::Time(), "FromUnixMicros");
>+  constexpr absl::Time t5 = absl::FromUnixMillis(0);
>+  static_assert(t5 == absl::Time(), "FromUnixMillis");
>+  constexpr absl::Time t6 = absl::FromUnixSeconds(0);
>+  static_assert(t6 == absl::Time(), "FromUnixSeconds");
>+  constexpr absl::Time t7 = absl::FromTimeT(0);
>+  static_assert(t7 == absl::Time(), "FromTimeT");
>+}
>+
>+TEST(Time, ValueSemantics) {
>+  absl::Time a;      // Default construction
>+  absl::Time b = a;  // Copy construction
>+  EXPECT_EQ(a, b);
>+  absl::Time c(a);  // Copy construction (again)
>+  EXPECT_EQ(a, b);
>+  EXPECT_EQ(a, c);
>+  EXPECT_EQ(b, c);
>+  b = c;       // Assignment
>+  EXPECT_EQ(a, b);
>+  EXPECT_EQ(a, c);
>+  EXPECT_EQ(b, c);
>+}
>+
>+TEST(Time, UnixEpoch) {
>+  absl::Time::Breakdown bd = absl::UnixEpoch().In(absl::UTCTimeZone());
>+  ABSL_INTERNAL_EXPECT_TIME(bd, 1970, 1, 1, 0, 0, 0, 0, false);
>+  EXPECT_EQ(absl::ZeroDuration(), bd.subsecond);
>+  EXPECT_EQ(4, bd.weekday);  // Thursday
>+}
>+
>+TEST(Time, Breakdown) {
>+  absl::TimeZone tz = absl::time_internal::LoadTimeZone("America/New_York");
>+  absl::Time t = absl::UnixEpoch();
>+
>+  // The Unix epoch as seen in NYC.
>+  absl::Time::Breakdown bd = t.In(tz);
>+  ABSL_INTERNAL_EXPECT_TIME(bd, 1969, 12, 31, 19, 0, 0, -18000, false);
>+  EXPECT_EQ(absl::ZeroDuration(), bd.subsecond);
>+  EXPECT_EQ(3, bd.weekday);  // Wednesday
>+
>+  // Just before the epoch.
>+  t -= absl::Nanoseconds(1);
>+  bd = t.In(tz);
>+  ABSL_INTERNAL_EXPECT_TIME(bd, 1969, 12, 31, 18, 59, 59, -18000, false);
>+  EXPECT_EQ(absl::Nanoseconds(999999999), bd.subsecond);
>+  EXPECT_EQ(3, bd.weekday);  // Wednesday
>+
>+  // Some time later.
>+  t += absl::Hours(24) * 2735;
>+  t += absl::Hours(18) + absl::Minutes(30) + absl::Seconds(15) +
>+       absl::Nanoseconds(9);
>+  bd = t.In(tz);
>+  ABSL_INTERNAL_EXPECT_TIME(bd, 1977, 6, 28, 14, 30, 15, -14400, true);
>+  EXPECT_EQ(8, bd.subsecond / absl::Nanoseconds(1));
>+  EXPECT_EQ(2, bd.weekday);  // Tuesday
>+}
>+
>+TEST(Time, AdditiveOperators) {
>+  const absl::Duration d = absl::Nanoseconds(1);
>+  const absl::Time t0;
>+  const absl::Time t1 = t0 + d;
>+
>+  EXPECT_EQ(d, t1 - t0);
>+  EXPECT_EQ(-d, t0 - t1);
>+  EXPECT_EQ(t0, t1 - d);
>+
>+  absl::Time t(t0);
>+  EXPECT_EQ(t0, t);
>+  t += d;
>+  EXPECT_EQ(t0 + d, t);
>+  EXPECT_EQ(d, t - t0);
>+  t -= d;
>+  EXPECT_EQ(t0, t);
>+
>+  // Tests overflow between subseconds and seconds.
>+  t = absl::UnixEpoch();
>+  t += absl::Milliseconds(500);
>+  EXPECT_EQ(absl::UnixEpoch() + absl::Milliseconds(500), t);
>+  t += absl::Milliseconds(600);
>+  EXPECT_EQ(absl::UnixEpoch() + absl::Milliseconds(1100), t);
>+  t -= absl::Milliseconds(600);
>+  EXPECT_EQ(absl::UnixEpoch() + absl::Milliseconds(500), t);
>+  t -= absl::Milliseconds(500);
>+  EXPECT_EQ(absl::UnixEpoch(), t);
>+}
>+
>+TEST(Time, RelationalOperators) {
>+  constexpr absl::Time t1 = absl::FromUnixNanos(0);
>+  constexpr absl::Time t2 = absl::FromUnixNanos(1);
>+  constexpr absl::Time t3 = absl::FromUnixNanos(2);
>+
>+  static_assert(absl::Time() == t1, "");
>+  static_assert(t1 == t1, "");
>+  static_assert(t2 == t2, "");
>+  static_assert(t3 == t3, "");
>+
>+  static_assert(t1 < t2, "");
>+  static_assert(t2 < t3, "");
>+  static_assert(t1 < t3, "");
>+
>+  static_assert(t1 <= t1, "");
>+  static_assert(t1 <= t2, "");
>+  static_assert(t2 <= t2, "");
>+  static_assert(t2 <= t3, "");
>+  static_assert(t3 <= t3, "");
>+  static_assert(t1 <= t3, "");
>+
>+  static_assert(t2 > t1, "");
>+  static_assert(t3 > t2, "");
>+  static_assert(t3 > t1, "");
>+
>+  static_assert(t2 >= t2, "");
>+  static_assert(t2 >= t1, "");
>+  static_assert(t3 >= t3, "");
>+  static_assert(t3 >= t2, "");
>+  static_assert(t1 >= t1, "");
>+  static_assert(t3 >= t1, "");
>+}
>+
>+TEST(Time, Infinity) {
>+  constexpr absl::Time ifuture = absl::InfiniteFuture();
>+  constexpr absl::Time ipast = absl::InfinitePast();
>+
>+  static_assert(ifuture == ifuture, "");
>+  static_assert(ipast == ipast, "");
>+  static_assert(ipast < ifuture, "");
>+  static_assert(ifuture > ipast, "");
>+
>+  // Arithmetic saturates
>+  EXPECT_EQ(ifuture, ifuture + absl::Seconds(1));
>+  EXPECT_EQ(ifuture, ifuture - absl::Seconds(1));
>+  EXPECT_EQ(ipast, ipast + absl::Seconds(1));
>+  EXPECT_EQ(ipast, ipast - absl::Seconds(1));
>+
>+  EXPECT_EQ(absl::InfiniteDuration(), ifuture - ifuture);
>+  EXPECT_EQ(absl::InfiniteDuration(), ifuture - ipast);
>+  EXPECT_EQ(-absl::InfiniteDuration(), ipast - ifuture);
>+  EXPECT_EQ(-absl::InfiniteDuration(), ipast - ipast);
>+
>+  constexpr absl::Time t = absl::UnixEpoch();  // Any finite time.
>+  static_assert(t < ifuture, "");
>+  static_assert(t > ipast, "");
>+}
>+
>+TEST(Time, FloorConversion) {
>+#define TEST_FLOOR_CONVERSION(TO, FROM) \
>+  EXPECT_EQ(1, TO(FROM(1001)));         \
>+  EXPECT_EQ(1, TO(FROM(1000)));         \
>+  EXPECT_EQ(0, TO(FROM(999)));          \
>+  EXPECT_EQ(0, TO(FROM(1)));            \
>+  EXPECT_EQ(0, TO(FROM(0)));            \
>+  EXPECT_EQ(-1, TO(FROM(-1)));          \
>+  EXPECT_EQ(-1, TO(FROM(-999)));        \
>+  EXPECT_EQ(-1, TO(FROM(-1000)));       \
>+  EXPECT_EQ(-2, TO(FROM(-1001)));
>+
>+  TEST_FLOOR_CONVERSION(absl::ToUnixMicros, absl::FromUnixNanos);
>+  TEST_FLOOR_CONVERSION(absl::ToUnixMillis, absl::FromUnixMicros);
>+  TEST_FLOOR_CONVERSION(absl::ToUnixSeconds, absl::FromUnixMillis);
>+  TEST_FLOOR_CONVERSION(absl::ToTimeT, absl::FromUnixMillis);
>+
>+#undef TEST_FLOOR_CONVERSION
>+
>+  // Tests ToUnixNanos.
>+  EXPECT_EQ(1, absl::ToUnixNanos(absl::UnixEpoch() + absl::Nanoseconds(3) / 2));
>+  EXPECT_EQ(1, absl::ToUnixNanos(absl::UnixEpoch() + absl::Nanoseconds(1)));
>+  EXPECT_EQ(0, absl::ToUnixNanos(absl::UnixEpoch() + absl::Nanoseconds(1) / 2));
>+  EXPECT_EQ(0, absl::ToUnixNanos(absl::UnixEpoch() + absl::Nanoseconds(0)));
>+  EXPECT_EQ(-1,
>+            absl::ToUnixNanos(absl::UnixEpoch() - absl::Nanoseconds(1) / 2));
>+  EXPECT_EQ(-1, absl::ToUnixNanos(absl::UnixEpoch() - absl::Nanoseconds(1)));
>+  EXPECT_EQ(-2,
>+            absl::ToUnixNanos(absl::UnixEpoch() - absl::Nanoseconds(3) / 2));
>+
>+  // Tests ToUniversal, which uses a different epoch than the tests above.
>+  EXPECT_EQ(1,
>+            absl::ToUniversal(absl::UniversalEpoch() + absl::Nanoseconds(101)));
>+  EXPECT_EQ(1,
>+            absl::ToUniversal(absl::UniversalEpoch() + absl::Nanoseconds(100)));
>+  EXPECT_EQ(0,
>+            absl::ToUniversal(absl::UniversalEpoch() + absl::Nanoseconds(99)));
>+  EXPECT_EQ(0,
>+            absl::ToUniversal(absl::UniversalEpoch() + absl::Nanoseconds(1)));
>+  EXPECT_EQ(0,
>+            absl::ToUniversal(absl::UniversalEpoch() + absl::Nanoseconds(0)));
>+  EXPECT_EQ(-1,
>+            absl::ToUniversal(absl::UniversalEpoch() + absl::Nanoseconds(-1)));
>+  EXPECT_EQ(-1,
>+            absl::ToUniversal(absl::UniversalEpoch() + absl::Nanoseconds(-99)));
>+  EXPECT_EQ(
>+      -1, absl::ToUniversal(absl::UniversalEpoch() + absl::Nanoseconds(-100)));
>+  EXPECT_EQ(
>+      -2, absl::ToUniversal(absl::UniversalEpoch() + absl::Nanoseconds(-101)));
>+
>+  // Tests ToTimespec()/TimeFromTimespec()
>+  const struct {
>+    absl::Time t;
>+    timespec ts;
>+  } to_ts[] = {
>+      {absl::FromUnixSeconds(1) + absl::Nanoseconds(1), {1, 1}},
>+      {absl::FromUnixSeconds(1) + absl::Nanoseconds(1) / 2, {1, 0}},
>+      {absl::FromUnixSeconds(1) + absl::Nanoseconds(0), {1, 0}},
>+      {absl::FromUnixSeconds(0) + absl::Nanoseconds(0), {0, 0}},
>+      {absl::FromUnixSeconds(0) - absl::Nanoseconds(1) / 2, {-1, 999999999}},
>+      {absl::FromUnixSeconds(0) - absl::Nanoseconds(1), {-1, 999999999}},
>+      {absl::FromUnixSeconds(-1) + absl::Nanoseconds(1), {-1, 1}},
>+      {absl::FromUnixSeconds(-1) + absl::Nanoseconds(1) / 2, {-1, 0}},
>+      {absl::FromUnixSeconds(-1) + absl::Nanoseconds(0), {-1, 0}},
>+      {absl::FromUnixSeconds(-1) - absl::Nanoseconds(1) / 2, {-2, 999999999}},
>+  };
>+  for (const auto& test : to_ts) {
>+    EXPECT_THAT(absl::ToTimespec(test.t), TimespecMatcher(test.ts));
>+  }
>+  const struct {
>+    timespec ts;
>+    absl::Time t;
>+  } from_ts[] = {
>+      {{1, 1}, absl::FromUnixSeconds(1) + absl::Nanoseconds(1)},
>+      {{1, 0}, absl::FromUnixSeconds(1) + absl::Nanoseconds(0)},
>+      {{0, 0}, absl::FromUnixSeconds(0) + absl::Nanoseconds(0)},
>+      {{0, -1}, absl::FromUnixSeconds(0) - absl::Nanoseconds(1)},
>+      {{-1, 999999999}, absl::FromUnixSeconds(0) - absl::Nanoseconds(1)},
>+      {{-1, 1}, absl::FromUnixSeconds(-1) + absl::Nanoseconds(1)},
>+      {{-1, 0}, absl::FromUnixSeconds(-1) + absl::Nanoseconds(0)},
>+      {{-1, -1}, absl::FromUnixSeconds(-1) - absl::Nanoseconds(1)},
>+      {{-2, 999999999}, absl::FromUnixSeconds(-1) - absl::Nanoseconds(1)},
>+  };
>+  for (const auto& test : from_ts) {
>+    EXPECT_EQ(test.t, absl::TimeFromTimespec(test.ts));
>+  }
>+
>+  // Tests ToTimeval()/TimeFromTimeval() (same as timespec above)
>+  const struct {
>+    absl::Time t;
>+    timeval tv;
>+  } to_tv[] = {
>+      {absl::FromUnixSeconds(1) + absl::Microseconds(1), {1, 1}},
>+      {absl::FromUnixSeconds(1) + absl::Microseconds(1) / 2, {1, 0}},
>+      {absl::FromUnixSeconds(1) + absl::Microseconds(0), {1, 0}},
>+      {absl::FromUnixSeconds(0) + absl::Microseconds(0), {0, 0}},
>+      {absl::FromUnixSeconds(0) - absl::Microseconds(1) / 2, {-1, 999999}},
>+      {absl::FromUnixSeconds(0) - absl::Microseconds(1), {-1, 999999}},
>+      {absl::FromUnixSeconds(-1) + absl::Microseconds(1), {-1, 1}},
>+      {absl::FromUnixSeconds(-1) + absl::Microseconds(1) / 2, {-1, 0}},
>+      {absl::FromUnixSeconds(-1) + absl::Microseconds(0), {-1, 0}},
>+      {absl::FromUnixSeconds(-1) - absl::Microseconds(1) / 2, {-2, 999999}},
>+  };
>+  for (const auto& test : to_tv) {
>+    EXPECT_THAT(ToTimeval(test.t), TimevalMatcher(test.tv));
>+  }
>+  const struct {
>+    timeval tv;
>+    absl::Time t;
>+  } from_tv[] = {
>+      {{1, 1}, absl::FromUnixSeconds(1) + absl::Microseconds(1)},
>+      {{1, 0}, absl::FromUnixSeconds(1) + absl::Microseconds(0)},
>+      {{0, 0}, absl::FromUnixSeconds(0) + absl::Microseconds(0)},
>+      {{0, -1}, absl::FromUnixSeconds(0) - absl::Microseconds(1)},
>+      {{-1, 999999}, absl::FromUnixSeconds(0) - absl::Microseconds(1)},
>+      {{-1, 1}, absl::FromUnixSeconds(-1) + absl::Microseconds(1)},
>+      {{-1, 0}, absl::FromUnixSeconds(-1) + absl::Microseconds(0)},
>+      {{-1, -1}, absl::FromUnixSeconds(-1) - absl::Microseconds(1)},
>+      {{-2, 999999}, absl::FromUnixSeconds(-1) - absl::Microseconds(1)},
>+  };
>+  for (const auto& test : from_tv) {
>+    EXPECT_EQ(test.t, absl::TimeFromTimeval(test.tv));
>+  }
>+
>+  // Tests flooring near negative infinity.
>+  const int64_t min_plus_1 = std::numeric_limits<int64_t>::min() + 1;
>+  EXPECT_EQ(min_plus_1, absl::ToUnixSeconds(absl::FromUnixSeconds(min_plus_1)));
>+  EXPECT_EQ(std::numeric_limits<int64_t>::min(),
>+            absl::ToUnixSeconds(
>+                absl::FromUnixSeconds(min_plus_1) - absl::Nanoseconds(1) / 2));
>+
>+  // Tests flooring near positive infinity.
>+  EXPECT_EQ(std::numeric_limits<int64_t>::max(),
>+            absl::ToUnixSeconds(absl::FromUnixSeconds(
>+                std::numeric_limits<int64_t>::max()) + absl::Nanoseconds(1) / 2));
>+  EXPECT_EQ(std::numeric_limits<int64_t>::max(),
>+            absl::ToUnixSeconds(
>+                absl::FromUnixSeconds(std::numeric_limits<int64_t>::max())));
>+  EXPECT_EQ(std::numeric_limits<int64_t>::max() - 1,
>+            absl::ToUnixSeconds(absl::FromUnixSeconds(
>+                std::numeric_limits<int64_t>::max()) - absl::Nanoseconds(1) / 2));
>+}
>+
>+TEST(Time, RoundtripConversion) {
>+#define TEST_CONVERSION_ROUND_TRIP(SOURCE, FROM, TO, MATCHER) \
>+  EXPECT_THAT(TO(FROM(SOURCE)), MATCHER(SOURCE))
>+
>+  // FromUnixNanos() and ToUnixNanos()
>+  int64_t now_ns = absl::GetCurrentTimeNanos();
>+  TEST_CONVERSION_ROUND_TRIP(-1, absl::FromUnixNanos, absl::ToUnixNanos,
>+                             testing::Eq);
>+  TEST_CONVERSION_ROUND_TRIP(0, absl::FromUnixNanos, absl::ToUnixNanos,
>+                             testing::Eq);
>+  TEST_CONVERSION_ROUND_TRIP(1, absl::FromUnixNanos, absl::ToUnixNanos,
>+                             testing::Eq);
>+  TEST_CONVERSION_ROUND_TRIP(now_ns, absl::FromUnixNanos, absl::ToUnixNanos,
>+                             testing::Eq)
>+      << now_ns;
>+
>+  // FromUnixMicros() and ToUnixMicros()
>+  int64_t now_us = absl::GetCurrentTimeNanos() / 1000;
>+  TEST_CONVERSION_ROUND_TRIP(-1, absl::FromUnixMicros, absl::ToUnixMicros,
>+                             testing::Eq);
>+  TEST_CONVERSION_ROUND_TRIP(0, absl::FromUnixMicros, absl::ToUnixMicros,
>+                             testing::Eq);
>+  TEST_CONVERSION_ROUND_TRIP(1, absl::FromUnixMicros, absl::ToUnixMicros,
>+                             testing::Eq);
>+  TEST_CONVERSION_ROUND_TRIP(now_us, absl::FromUnixMicros, absl::ToUnixMicros,
>+                             testing::Eq)
>+      << now_us;
>+
>+  // FromUnixMillis() and ToUnixMillis()
>+  int64_t now_ms = absl::GetCurrentTimeNanos() / 1000000;
>+  TEST_CONVERSION_ROUND_TRIP(-1, absl::FromUnixMillis, absl::ToUnixMillis,
>+                             testing::Eq);
>+  TEST_CONVERSION_ROUND_TRIP(0, absl::FromUnixMillis, absl::ToUnixMillis,
>+                             testing::Eq);
>+  TEST_CONVERSION_ROUND_TRIP(1, absl::FromUnixMillis, absl::ToUnixMillis,
>+                             testing::Eq);
>+  TEST_CONVERSION_ROUND_TRIP(now_ms, absl::FromUnixMillis, absl::ToUnixMillis,
>+                             testing::Eq)
>+      << now_ms;
>+
>+  // FromUnixSeconds() and ToUnixSeconds()
>+  int64_t now_s = std::time(nullptr);
>+  TEST_CONVERSION_ROUND_TRIP(-1, absl::FromUnixSeconds, absl::ToUnixSeconds,
>+                             testing::Eq);
>+  TEST_CONVERSION_ROUND_TRIP(0, absl::FromUnixSeconds, absl::ToUnixSeconds,
>+                             testing::Eq);
>+  TEST_CONVERSION_ROUND_TRIP(1, absl::FromUnixSeconds, absl::ToUnixSeconds,
>+                             testing::Eq);
>+  TEST_CONVERSION_ROUND_TRIP(now_s, absl::FromUnixSeconds, absl::ToUnixSeconds,
>+                             testing::Eq)
>+      << now_s;
>+
>+  // FromTimeT() and ToTimeT()
>+  time_t now_time_t = std::time(nullptr);
>+  TEST_CONVERSION_ROUND_TRIP(-1, absl::FromTimeT, absl::ToTimeT, testing::Eq);
>+  TEST_CONVERSION_ROUND_TRIP(0, absl::FromTimeT, absl::ToTimeT, testing::Eq);
>+  TEST_CONVERSION_ROUND_TRIP(1, absl::FromTimeT, absl::ToTimeT, testing::Eq);
>+  TEST_CONVERSION_ROUND_TRIP(now_time_t, absl::FromTimeT, absl::ToTimeT,
>+                             testing::Eq)
>+      << now_time_t;
>+
>+  // TimeFromTimeval() and ToTimeval()
>+  timeval tv;
>+  tv.tv_sec = -1;
>+  tv.tv_usec = 0;
>+  TEST_CONVERSION_ROUND_TRIP(tv, absl::TimeFromTimeval, absl::ToTimeval,
>+                             TimevalMatcher);
>+  tv.tv_sec = -1;
>+  tv.tv_usec = 999999;
>+  TEST_CONVERSION_ROUND_TRIP(tv, absl::TimeFromTimeval, absl::ToTimeval,
>+                             TimevalMatcher);
>+  tv.tv_sec = 0;
>+  tv.tv_usec = 0;
>+  TEST_CONVERSION_ROUND_TRIP(tv, absl::TimeFromTimeval, absl::ToTimeval,
>+                             TimevalMatcher);
>+  tv.tv_sec = 0;
>+  tv.tv_usec = 1;
>+  TEST_CONVERSION_ROUND_TRIP(tv, absl::TimeFromTimeval, absl::ToTimeval,
>+                             TimevalMatcher);
>+  tv.tv_sec = 1;
>+  tv.tv_usec = 0;
>+  TEST_CONVERSION_ROUND_TRIP(tv, absl::TimeFromTimeval, absl::ToTimeval,
>+                             TimevalMatcher);
>+
>+  // TimeFromTimespec() and ToTimespec()
>+  timespec ts;
>+  ts.tv_sec = -1;
>+  ts.tv_nsec = 0;
>+  TEST_CONVERSION_ROUND_TRIP(ts, absl::TimeFromTimespec, absl::ToTimespec,
>+                             TimespecMatcher);
>+  ts.tv_sec = -1;
>+  ts.tv_nsec = 999999999;
>+  TEST_CONVERSION_ROUND_TRIP(ts, absl::TimeFromTimespec, absl::ToTimespec,
>+                             TimespecMatcher);
>+  ts.tv_sec = 0;
>+  ts.tv_nsec = 0;
>+  TEST_CONVERSION_ROUND_TRIP(ts, absl::TimeFromTimespec, absl::ToTimespec,
>+                             TimespecMatcher);
>+  ts.tv_sec = 0;
>+  ts.tv_nsec = 1;
>+  TEST_CONVERSION_ROUND_TRIP(ts, absl::TimeFromTimespec, absl::ToTimespec,
>+                             TimespecMatcher);
>+  ts.tv_sec = 1;
>+  ts.tv_nsec = 0;
>+  TEST_CONVERSION_ROUND_TRIP(ts, absl::TimeFromTimespec, absl::ToTimespec,
>+                             TimespecMatcher);
>+
>+  // FromUDate() and ToUDate()
>+  double now_ud = absl::GetCurrentTimeNanos() / 1000000;
>+  TEST_CONVERSION_ROUND_TRIP(-1.5, absl::FromUDate, absl::ToUDate,
>+                             testing::DoubleEq);
>+  TEST_CONVERSION_ROUND_TRIP(-1, absl::FromUDate, absl::ToUDate,
>+                             testing::DoubleEq);
>+  TEST_CONVERSION_ROUND_TRIP(-0.5, absl::FromUDate, absl::ToUDate,
>+                             testing::DoubleEq);
>+  TEST_CONVERSION_ROUND_TRIP(0, absl::FromUDate, absl::ToUDate,
>+                             testing::DoubleEq);
>+  TEST_CONVERSION_ROUND_TRIP(0.5, absl::FromUDate, absl::ToUDate,
>+                             testing::DoubleEq);
>+  TEST_CONVERSION_ROUND_TRIP(1, absl::FromUDate, absl::ToUDate,
>+                             testing::DoubleEq);
>+  TEST_CONVERSION_ROUND_TRIP(1.5, absl::FromUDate, absl::ToUDate,
>+                             testing::DoubleEq);
>+  TEST_CONVERSION_ROUND_TRIP(now_ud, absl::FromUDate, absl::ToUDate,
>+                             testing::DoubleEq)
>+      << std::fixed << std::setprecision(17) << now_ud;
>+
>+  // FromUniversal() and ToUniversal()
>+  int64_t now_uni = ((719162LL * (24 * 60 * 60)) * (1000 * 1000 * 10)) +
>+                    (absl::GetCurrentTimeNanos() / 100);
>+  TEST_CONVERSION_ROUND_TRIP(-1, absl::FromUniversal, absl::ToUniversal,
>+                             testing::Eq);
>+  TEST_CONVERSION_ROUND_TRIP(0, absl::FromUniversal, absl::ToUniversal,
>+                             testing::Eq);
>+  TEST_CONVERSION_ROUND_TRIP(1, absl::FromUniversal, absl::ToUniversal,
>+                             testing::Eq);
>+  TEST_CONVERSION_ROUND_TRIP(now_uni, absl::FromUniversal, absl::ToUniversal,
>+                             testing::Eq)
>+      << now_uni;
>+
>+#undef TEST_CONVERSION_ROUND_TRIP
>+}
>+
>+template <typename Duration>
>+std::chrono::system_clock::time_point MakeChronoUnixTime(const Duration& d) {
>+  return std::chrono::system_clock::from_time_t(0) + d;
>+}
>+
>+TEST(Time, FromChrono) {
>+  EXPECT_EQ(absl::FromTimeT(-1),
>+            absl::FromChrono(std::chrono::system_clock::from_time_t(-1)));
>+  EXPECT_EQ(absl::FromTimeT(0),
>+            absl::FromChrono(std::chrono::system_clock::from_time_t(0)));
>+  EXPECT_EQ(absl::FromTimeT(1),
>+            absl::FromChrono(std::chrono::system_clock::from_time_t(1)));
>+
>+  EXPECT_EQ(
>+      absl::FromUnixMillis(-1),
>+      absl::FromChrono(MakeChronoUnixTime(std::chrono::milliseconds(-1))));
>+  EXPECT_EQ(absl::FromUnixMillis(0),
>+            absl::FromChrono(MakeChronoUnixTime(std::chrono::milliseconds(0))));
>+  EXPECT_EQ(absl::FromUnixMillis(1),
>+            absl::FromChrono(MakeChronoUnixTime(std::chrono::milliseconds(1))));
>+
>+  // Chrono doesn't define exactly its range and precision (neither does
>+  // absl::Time), so let's simply test +/- ~100 years to make sure things work.
>+  const auto century_sec = 60 * 60 * 24 * 365 * int64_t{100};
>+  const auto century = std::chrono::seconds(century_sec);
>+  const auto chrono_future = MakeChronoUnixTime(century);
>+  const auto chrono_past = MakeChronoUnixTime(-century);
>+  EXPECT_EQ(absl::FromUnixSeconds(century_sec),
>+            absl::FromChrono(chrono_future));
>+  EXPECT_EQ(absl::FromUnixSeconds(-century_sec), absl::FromChrono(chrono_past));
>+
>+  // Roundtrip them both back to chrono.
>+  EXPECT_EQ(chrono_future,
>+            absl::ToChronoTime(absl::FromUnixSeconds(century_sec)));
>+  EXPECT_EQ(chrono_past,
>+            absl::ToChronoTime(absl::FromUnixSeconds(-century_sec)));
>+}
>+
>+TEST(Time, ToChronoTime) {
>+  EXPECT_EQ(std::chrono::system_clock::from_time_t(-1),
>+            absl::ToChronoTime(absl::FromTimeT(-1)));
>+  EXPECT_EQ(std::chrono::system_clock::from_time_t(0),
>+            absl::ToChronoTime(absl::FromTimeT(0)));
>+  EXPECT_EQ(std::chrono::system_clock::from_time_t(1),
>+            absl::ToChronoTime(absl::FromTimeT(1)));
>+
>+  EXPECT_EQ(MakeChronoUnixTime(std::chrono::milliseconds(-1)),
>+            absl::ToChronoTime(absl::FromUnixMillis(-1)));
>+  EXPECT_EQ(MakeChronoUnixTime(std::chrono::milliseconds(0)),
>+            absl::ToChronoTime(absl::FromUnixMillis(0)));
>+  EXPECT_EQ(MakeChronoUnixTime(std::chrono::milliseconds(1)),
>+            absl::ToChronoTime(absl::FromUnixMillis(1)));
>+
>+  // Time before the Unix epoch should floor, not trunc.
>+  const auto tick = absl::Nanoseconds(1) / 4;
>+  EXPECT_EQ(std::chrono::system_clock::from_time_t(0) -
>+                std::chrono::system_clock::duration(1),
>+            absl::ToChronoTime(absl::UnixEpoch() - tick));
>+}
>+
>+TEST(Time, ConvertDateTime) {
>+  const absl::TimeZone utc = absl::UTCTimeZone();
>+  const absl::TimeZone goog =
>+      absl::time_internal::LoadTimeZone("America/Los_Angeles");
>+  const absl::TimeZone nyc =
>+      absl::time_internal::LoadTimeZone("America/New_York");
>+  const std::string fmt = "%a, %e %b %Y %H:%M:%S %z (%Z)";
>+
>+  // A simple case of normalization.
>+  absl::TimeConversion oct32 = ConvertDateTime(2013, 10, 32, 8, 30, 0, goog);
>+  EXPECT_TRUE(oct32.normalized);
>+  EXPECT_EQ(absl::TimeConversion::UNIQUE, oct32.kind);
>+  absl::TimeConversion nov01 = ConvertDateTime(2013, 11, 1, 8, 30, 0, goog);
>+  EXPECT_FALSE(nov01.normalized);
>+  EXPECT_EQ(absl::TimeConversion::UNIQUE, nov01.kind);
>+  EXPECT_EQ(oct32.pre, nov01.pre);
>+  EXPECT_EQ("Fri,  1 Nov 2013 08:30:00 -0700 (PDT)",
>+            absl::FormatTime(fmt, nov01.pre, goog));
>+
>+  // A Spring DST transition, when there is a gap in civil time
>+  // and we prefer the later of the possible interpretations of a
>+  // non-existent time.
>+  absl::TimeConversion mar13 = ConvertDateTime(2011, 3, 13, 2, 15, 0, nyc);
>+  EXPECT_FALSE(mar13.normalized);
>+  EXPECT_EQ(absl::TimeConversion::SKIPPED, mar13.kind);
>+  EXPECT_EQ("Sun, 13 Mar 2011 03:15:00 -0400 (EDT)",
>+            absl::FormatTime(fmt, mar13.pre, nyc));
>+  EXPECT_EQ("Sun, 13 Mar 2011 03:00:00 -0400 (EDT)",
>+            absl::FormatTime(fmt, mar13.trans, nyc));
>+  EXPECT_EQ("Sun, 13 Mar 2011 01:15:00 -0500 (EST)",
>+            absl::FormatTime(fmt, mar13.post, nyc));
>+  EXPECT_EQ(mar13.pre, absl::FromDateTime(2011, 3, 13, 2, 15, 0, nyc));
>+
>+  // A Fall DST transition, when civil times are repeated and
>+  // we prefer the earlier of the possible interpretations of an
>+  // ambiguous time.
>+  absl::TimeConversion nov06 = ConvertDateTime(2011, 11, 6, 1, 15, 0, nyc);
>+  EXPECT_FALSE(nov06.normalized);
>+  EXPECT_EQ(absl::TimeConversion::REPEATED, nov06.kind);
>+  EXPECT_EQ("Sun,  6 Nov 2011 01:15:00 -0400 (EDT)",
>+            absl::FormatTime(fmt, nov06.pre, nyc));
>+  EXPECT_EQ("Sun,  6 Nov 2011 01:00:00 -0500 (EST)",
>+            absl::FormatTime(fmt, nov06.trans, nyc));
>+  EXPECT_EQ("Sun,  6 Nov 2011 01:15:00 -0500 (EST)",
>+            absl::FormatTime(fmt, nov06.post, nyc));
>+  EXPECT_EQ(nov06.pre, absl::FromDateTime(2011, 11, 6, 1, 15, 0, nyc));
>+
>+  // Check that (time_t) -1 is handled correctly.
>+  absl::TimeConversion minus1 = ConvertDateTime(1969, 12, 31, 18, 59, 59, nyc);
>+  EXPECT_FALSE(minus1.normalized);
>+  EXPECT_EQ(absl::TimeConversion::UNIQUE, minus1.kind);
>+  EXPECT_EQ(-1, absl::ToTimeT(minus1.pre));
>+  EXPECT_EQ("Wed, 31 Dec 1969 18:59:59 -0500 (EST)",
>+            absl::FormatTime(fmt, minus1.pre, nyc));
>+  EXPECT_EQ("Wed, 31 Dec 1969 23:59:59 +0000 (UTC)",
>+            absl::FormatTime(fmt, minus1.pre, utc));
>+}
>+
>+// FromDateTime(year, mon, day, hour, min, sec, UTCTimeZone()) has
>+// a specialized fastpath implementation which we exercise here.
>+TEST(Time, FromDateTimeUTC) {
>+  const absl::TimeZone utc = absl::UTCTimeZone();
>+  const std::string fmt = "%a, %e %b %Y %H:%M:%S %z (%Z)";
>+  const int kMax = std::numeric_limits<int>::max();
>+  const int kMin = std::numeric_limits<int>::min();
>+  absl::Time t;
>+
>+  // 292091940881 is the last positive year to use the fastpath.
>+  t = absl::FromDateTime(292091940881, kMax, kMax, kMax, kMax, kMax, utc);
>+  EXPECT_EQ("Fri, 25 Nov 292277026596 12:21:07 +0000 (UTC)",
>+            absl::FormatTime(fmt, t, utc));
>+  t = absl::FromDateTime(292091940882, kMax, kMax, kMax, kMax, kMax, utc);
>+  EXPECT_EQ("infinite-future", absl::FormatTime(fmt, t, utc));  // no overflow
>+  t = absl::FromDateTime(
>+      std::numeric_limits<int64_t>::max(), kMax, kMax, kMax, kMax, kMax, utc);
>+  EXPECT_EQ("infinite-future", absl::FormatTime(fmt, t, utc));  // no overflow
>+
>+  // -292091936940 is the last negative year to use the fastpath.
>+  t = absl::FromDateTime(-292091936940, kMin, kMin, kMin, kMin, kMin, utc);
>+  EXPECT_EQ("Fri,  1 Nov -292277022657 10:37:52 +0000 (UTC)",
>+            absl::FormatTime(fmt, t, utc));
>+  t = absl::FromDateTime(-292091936941, kMin, kMin, kMin, kMin, kMin, utc);
>+  EXPECT_EQ("infinite-past", absl::FormatTime(fmt, t, utc));  // no underflow
>+  t = absl::FromDateTime(
>+      std::numeric_limits<int64_t>::min(), kMin, kMin, kMin, kMin, kMin, utc);
>+  EXPECT_EQ("infinite-past", absl::FormatTime(fmt, t, utc));  // no overflow
>+
>+  // Check that we're counting leap years correctly.
>+  t = absl::FromDateTime(1900, 2, 28, 23, 59, 59, utc);
>+  EXPECT_EQ("Wed, 28 Feb 1900 23:59:59 +0000 (UTC)",
>+            absl::FormatTime(fmt, t, utc));
>+  t = absl::FromDateTime(1900, 3, 1, 0, 0, 0, utc);
>+  EXPECT_EQ("Thu,  1 Mar 1900 00:00:00 +0000 (UTC)",
>+            absl::FormatTime(fmt, t, utc));
>+  t = absl::FromDateTime(2000, 2, 29, 23, 59, 59, utc);
>+  EXPECT_EQ("Tue, 29 Feb 2000 23:59:59 +0000 (UTC)",
>+            absl::FormatTime(fmt, t, utc));
>+  t = absl::FromDateTime(2000, 3, 1, 0, 0, 0, utc);
>+  EXPECT_EQ("Wed,  1 Mar 2000 00:00:00 +0000 (UTC)",
>+            absl::FormatTime(fmt, t, utc));
>+
>+  // Check normalization.
>+  const std::string ymdhms = "%Y-%m-%d %H:%M:%S";
>+  t = absl::FromDateTime(2015, 1, 1, 0, 0, 60, utc);
>+  EXPECT_EQ("2015-01-01 00:01:00", absl::FormatTime(ymdhms, t, utc));
>+  t = absl::FromDateTime(2015, 1, 1, 0, 60, 0, utc);
>+  EXPECT_EQ("2015-01-01 01:00:00", absl::FormatTime(ymdhms, t, utc));
>+  t = absl::FromDateTime(2015, 1, 1, 24, 0, 0, utc);
>+  EXPECT_EQ("2015-01-02 00:00:00", absl::FormatTime(ymdhms, t, utc));
>+  t = absl::FromDateTime(2015, 1, 32, 0, 0, 0, utc);
>+  EXPECT_EQ("2015-02-01 00:00:00", absl::FormatTime(ymdhms, t, utc));
>+  t = absl::FromDateTime(2015, 13, 1, 0, 0, 0, utc);
>+  EXPECT_EQ("2016-01-01 00:00:00", absl::FormatTime(ymdhms, t, utc));
>+  t = absl::FromDateTime(2015, 13, 32, 60, 60, 60, utc);
>+  EXPECT_EQ("2016-02-03 13:01:00", absl::FormatTime(ymdhms, t, utc));
>+  t = absl::FromDateTime(2015, 1, 1, 0, 0, -1, utc);
>+  EXPECT_EQ("2014-12-31 23:59:59", absl::FormatTime(ymdhms, t, utc));
>+  t = absl::FromDateTime(2015, 1, 1, 0, -1, 0, utc);
>+  EXPECT_EQ("2014-12-31 23:59:00", absl::FormatTime(ymdhms, t, utc));
>+  t = absl::FromDateTime(2015, 1, 1, -1, 0, 0, utc);
>+  EXPECT_EQ("2014-12-31 23:00:00", absl::FormatTime(ymdhms, t, utc));
>+  t = absl::FromDateTime(2015, 1, -1, 0, 0, 0, utc);
>+  EXPECT_EQ("2014-12-30 00:00:00", absl::FormatTime(ymdhms, t, utc));
>+  t = absl::FromDateTime(2015, -1, 1, 0, 0, 0, utc);
>+  EXPECT_EQ("2014-11-01 00:00:00", absl::FormatTime(ymdhms, t, utc));
>+  t = absl::FromDateTime(2015, -1, -1, -1, -1, -1, utc);
>+  EXPECT_EQ("2014-10-29 22:58:59", absl::FormatTime(ymdhms, t, utc));
>+}
>+
>+TEST(Time, ToTM) {
>+  const absl::TimeZone utc = absl::UTCTimeZone();
>+
>+  // Compares the results of ToTM() to gmtime_r() for lots of times over the
>+  // course of a few days.
>+  const absl::Time start = absl::FromDateTime(2014, 1, 2, 3, 4, 5, utc);
>+  const absl::Time end = absl::FromDateTime(2014, 1, 5, 3, 4, 5, utc);
>+  for (absl::Time t = start; t < end; t += absl::Seconds(30)) {
>+    const struct tm tm_bt = ToTM(t, utc);
>+    const time_t tt = absl::ToTimeT(t);
>+    struct tm tm_lc;
>+#ifdef _WIN32
>+    gmtime_s(&tm_lc, &tt);
>+#else
>+    gmtime_r(&tt, &tm_lc);
>+#endif
>+    EXPECT_EQ(tm_lc.tm_year, tm_bt.tm_year);
>+    EXPECT_EQ(tm_lc.tm_mon, tm_bt.tm_mon);
>+    EXPECT_EQ(tm_lc.tm_mday, tm_bt.tm_mday);
>+    EXPECT_EQ(tm_lc.tm_hour, tm_bt.tm_hour);
>+    EXPECT_EQ(tm_lc.tm_min, tm_bt.tm_min);
>+    EXPECT_EQ(tm_lc.tm_sec, tm_bt.tm_sec);
>+    EXPECT_EQ(tm_lc.tm_wday, tm_bt.tm_wday);
>+    EXPECT_EQ(tm_lc.tm_yday, tm_bt.tm_yday);
>+    EXPECT_EQ(tm_lc.tm_isdst, tm_bt.tm_isdst);
>+
>+    ASSERT_FALSE(HasFailure());
>+  }
>+
>+  // Checks that the tm_isdst field is correct when in standard time.
>+  const absl::TimeZone nyc =
>+      absl::time_internal::LoadTimeZone("America/New_York");
>+  absl::Time t = absl::FromDateTime(2014, 3, 1, 0, 0, 0, nyc);
>+  struct tm tm = ToTM(t, nyc);
>+  EXPECT_FALSE(tm.tm_isdst);
>+
>+  // Checks that the tm_isdst field is correct when in daylight time.
>+  t = absl::FromDateTime(2014, 4, 1, 0, 0, 0, nyc);
>+  tm = ToTM(t, nyc);
>+  EXPECT_TRUE(tm.tm_isdst);
>+
>+  // Checks overflow.
>+  tm = ToTM(absl::InfiniteFuture(), nyc);
>+  EXPECT_EQ(std::numeric_limits<int>::max() - 1900, tm.tm_year);
>+  EXPECT_EQ(11, tm.tm_mon);
>+  EXPECT_EQ(31, tm.tm_mday);
>+  EXPECT_EQ(23, tm.tm_hour);
>+  EXPECT_EQ(59, tm.tm_min);
>+  EXPECT_EQ(59, tm.tm_sec);
>+  EXPECT_EQ(4, tm.tm_wday);
>+  EXPECT_EQ(364, tm.tm_yday);
>+  EXPECT_FALSE(tm.tm_isdst);
>+
>+  // Checks underflow.
>+  tm = ToTM(absl::InfinitePast(), nyc);
>+  EXPECT_EQ(std::numeric_limits<int>::min(), tm.tm_year);
>+  EXPECT_EQ(0, tm.tm_mon);
>+  EXPECT_EQ(1, tm.tm_mday);
>+  EXPECT_EQ(0, tm.tm_hour);
>+  EXPECT_EQ(0, tm.tm_min);
>+  EXPECT_EQ(0, tm.tm_sec);
>+  EXPECT_EQ(0, tm.tm_wday);
>+  EXPECT_EQ(0, tm.tm_yday);
>+  EXPECT_FALSE(tm.tm_isdst);
>+}
>+
>+TEST(Time, FromTM) {
>+  const absl::TimeZone nyc =
>+      absl::time_internal::LoadTimeZone("America/New_York");
>+
>+  // Verifies that tm_isdst doesn't affect anything when the time is unique.
>+  struct tm tm;
>+  std::memset(&tm, 0, sizeof(tm));
>+  tm.tm_year = 2014 - 1900;
>+  tm.tm_mon = 6 - 1;
>+  tm.tm_mday = 28;
>+  tm.tm_hour = 1;
>+  tm.tm_min = 2;
>+  tm.tm_sec = 3;
>+  tm.tm_isdst = -1;
>+  absl::Time t = FromTM(tm, nyc);
>+  EXPECT_EQ("2014-06-28T01:02:03-04:00", absl::FormatTime(t, nyc));  // DST
>+  tm.tm_isdst = 0;
>+  t = FromTM(tm, nyc);
>+  EXPECT_EQ("2014-06-28T01:02:03-04:00", absl::FormatTime(t, nyc));  // DST
>+  tm.tm_isdst = 1;
>+  t = FromTM(tm, nyc);
>+  EXPECT_EQ("2014-06-28T01:02:03-04:00", absl::FormatTime(t, nyc));  // DST
>+
>+  // Adjusts tm to refer to an ambiguous time.
>+  tm.tm_year = 2014 - 1900;
>+  tm.tm_mon = 11 - 1;
>+  tm.tm_mday = 2;
>+  tm.tm_hour = 1;
>+  tm.tm_min = 30;
>+  tm.tm_sec = 42;
>+  tm.tm_isdst = -1;
>+  t = FromTM(tm, nyc);
>+  EXPECT_EQ("2014-11-02T01:30:42-04:00", absl::FormatTime(t, nyc));  // DST
>+  tm.tm_isdst = 0;
>+  t = FromTM(tm, nyc);
>+  EXPECT_EQ("2014-11-02T01:30:42-05:00", absl::FormatTime(t, nyc));  // STD
>+  tm.tm_isdst = 1;
>+  t = FromTM(tm, nyc);
>+  EXPECT_EQ("2014-11-02T01:30:42-04:00", absl::FormatTime(t, nyc));  // DST
>+
>+  // Adjusts tm to refer to a skipped time.
>+  tm.tm_year = 2014 - 1900;
>+  tm.tm_mon = 3 - 1;
>+  tm.tm_mday = 9;
>+  tm.tm_hour = 2;
>+  tm.tm_min = 30;
>+  tm.tm_sec = 42;
>+  tm.tm_isdst = -1;
>+  t = FromTM(tm, nyc);
>+  EXPECT_EQ("2014-03-09T03:30:42-04:00", absl::FormatTime(t, nyc));  // DST
>+  tm.tm_isdst = 0;
>+  t = FromTM(tm, nyc);
>+  EXPECT_EQ("2014-03-09T01:30:42-05:00", absl::FormatTime(t, nyc));  // STD
>+  tm.tm_isdst = 1;
>+  t = FromTM(tm, nyc);
>+  EXPECT_EQ("2014-03-09T03:30:42-04:00", absl::FormatTime(t, nyc));  // DST
>+}
>+
>+TEST(Time, TMRoundTrip) {
>+  const absl::TimeZone nyc =
>+      absl::time_internal::LoadTimeZone("America/New_York");
>+
>+  // Test round-tripping across a skipped transition
>+  absl::Time start = absl::FromDateTime(2014, 3, 9, 0, 0, 0, nyc);
>+  absl::Time end = absl::FromDateTime(2014, 3, 9, 4, 0, 0, nyc);
>+  for (absl::Time t = start; t < end; t += absl::Minutes(1)) {
>+    struct tm tm = ToTM(t, nyc);
>+    absl::Time rt = FromTM(tm, nyc);
>+    EXPECT_EQ(rt, t);
>+  }
>+
>+  // Test round-tripping across an ambiguous transition
>+  start = absl::FromDateTime(2014, 11, 2, 0, 0, 0, nyc);
>+  end = absl::FromDateTime(2014, 11, 2, 4, 0, 0, nyc);
>+  for (absl::Time t = start; t < end; t += absl::Minutes(1)) {
>+    struct tm tm = ToTM(t, nyc);
>+    absl::Time rt = FromTM(tm, nyc);
>+    EXPECT_EQ(rt, t);
>+  }
>+
>+  // Test round-tripping of unique instants crossing a day boundary
>+  start = absl::FromDateTime(2014, 6, 27, 22, 0, 0, nyc);
>+  end = absl::FromDateTime(2014, 6, 28, 4, 0, 0, nyc);
>+  for (absl::Time t = start; t < end; t += absl::Minutes(1)) {
>+    struct tm tm = ToTM(t, nyc);
>+    absl::Time rt = FromTM(tm, nyc);
>+    EXPECT_EQ(rt, t);
>+  }
>+}
>+
>+TEST(Time, Range) {
>+  // The API's documented range is +/- 100 billion years.
>+  const absl::Duration range = absl::Hours(24) * 365.2425 * 100000000000;
>+
>+  // Arithmetic and comparison still works at +/-range around base values.
>+  absl::Time bases[2] = {absl::UnixEpoch(), absl::Now()};
>+  for (const auto base : bases) {
>+    absl::Time bottom = base - range;
>+    EXPECT_GT(bottom, bottom - absl::Nanoseconds(1));
>+    EXPECT_LT(bottom, bottom + absl::Nanoseconds(1));
>+    absl::Time top = base + range;
>+    EXPECT_GT(top, top - absl::Nanoseconds(1));
>+    EXPECT_LT(top, top + absl::Nanoseconds(1));
>+    absl::Duration full_range = 2 * range;
>+    EXPECT_EQ(full_range, top - bottom);
>+    EXPECT_EQ(-full_range, bottom - top);
>+  }
>+}
>+
>+TEST(Time, Limits) {
>+  // It is an implementation detail that Time().rep_ == ZeroDuration(),
>+  // and that the resolution of a Duration is 1/4 of a nanosecond.
>+  const absl::Time zero;
>+  const absl::Time max =
>+      zero + absl::Seconds(std::numeric_limits<int64_t>::max()) +
>+      absl::Nanoseconds(999999999) + absl::Nanoseconds(3) / 4;
>+  const absl::Time min =
>+      zero + absl::Seconds(std::numeric_limits<int64_t>::min());
>+
>+  // Some simple max/min bounds checks.
>+  EXPECT_LT(max, absl::InfiniteFuture());
>+  EXPECT_GT(min, absl::InfinitePast());
>+  EXPECT_LT(zero, max);
>+  EXPECT_GT(zero, min);
>+  EXPECT_GE(absl::UnixEpoch(), min);
>+  EXPECT_LT(absl::UnixEpoch(), max);
>+
>+  // Check sign of Time differences.
>+  EXPECT_LT(absl::ZeroDuration(), max - zero);
>+  EXPECT_LT(absl::ZeroDuration(),
>+            zero - absl::Nanoseconds(1) / 4 - min);  // avoid zero - min
>+
>+  // Arithmetic works at max - 0.25ns and min + 0.25ns.
>+  EXPECT_GT(max, max - absl::Nanoseconds(1) / 4);
>+  EXPECT_LT(min, min + absl::Nanoseconds(1) / 4);
>+}
>+
>+TEST(Time, ConversionSaturation) {
>+  const absl::TimeZone utc = absl::UTCTimeZone();
>+  absl::Time t;
>+
>+  const auto max_time_t = std::numeric_limits<time_t>::max();
>+  const auto min_time_t = std::numeric_limits<time_t>::min();
>+  time_t tt = max_time_t - 1;
>+  t = absl::FromTimeT(tt);
>+  tt = absl::ToTimeT(t);
>+  EXPECT_EQ(max_time_t - 1, tt);
>+  t += absl::Seconds(1);
>+  tt = absl::ToTimeT(t);
>+  EXPECT_EQ(max_time_t, tt);
>+  t += absl::Seconds(1);  // no effect
>+  tt = absl::ToTimeT(t);
>+  EXPECT_EQ(max_time_t, tt);
>+
>+  tt = min_time_t + 1;
>+  t = absl::FromTimeT(tt);
>+  tt = absl::ToTimeT(t);
>+  EXPECT_EQ(min_time_t + 1, tt);
>+  t -= absl::Seconds(1);
>+  tt = absl::ToTimeT(t);
>+  EXPECT_EQ(min_time_t, tt);
>+  t -= absl::Seconds(1);  // no effect
>+  tt = absl::ToTimeT(t);
>+  EXPECT_EQ(min_time_t, tt);
>+
>+  const auto max_timeval_sec =
>+      std::numeric_limits<decltype(timeval::tv_sec)>::max();
>+  const auto min_timeval_sec =
>+      std::numeric_limits<decltype(timeval::tv_sec)>::min();
>+  timeval tv;
>+  tv.tv_sec = max_timeval_sec;
>+  tv.tv_usec = 999998;
>+  t = absl::TimeFromTimeval(tv);
>+  tv = ToTimeval(t);
>+  EXPECT_EQ(max_timeval_sec, tv.tv_sec);
>+  EXPECT_EQ(999998, tv.tv_usec);
>+  t += absl::Microseconds(1);
>+  tv = ToTimeval(t);
>+  EXPECT_EQ(max_timeval_sec, tv.tv_sec);
>+  EXPECT_EQ(999999, tv.tv_usec);
>+  t += absl::Microseconds(1);  // no effect
>+  tv = ToTimeval(t);
>+  EXPECT_EQ(max_timeval_sec, tv.tv_sec);
>+  EXPECT_EQ(999999, tv.tv_usec);
>+
>+  tv.tv_sec = min_timeval_sec;
>+  tv.tv_usec = 1;
>+  t = absl::TimeFromTimeval(tv);
>+  tv = ToTimeval(t);
>+  EXPECT_EQ(min_timeval_sec, tv.tv_sec);
>+  EXPECT_EQ(1, tv.tv_usec);
>+  t -= absl::Microseconds(1);
>+  tv = ToTimeval(t);
>+  EXPECT_EQ(min_timeval_sec, tv.tv_sec);
>+  EXPECT_EQ(0, tv.tv_usec);
>+  t -= absl::Microseconds(1);  // no effect
>+  tv = ToTimeval(t);
>+  EXPECT_EQ(min_timeval_sec, tv.tv_sec);
>+  EXPECT_EQ(0, tv.tv_usec);
>+
>+  const auto max_timespec_sec =
>+      std::numeric_limits<decltype(timespec::tv_sec)>::max();
>+  const auto min_timespec_sec =
>+      std::numeric_limits<decltype(timespec::tv_sec)>::min();
>+  timespec ts;
>+  ts.tv_sec = max_timespec_sec;
>+  ts.tv_nsec = 999999998;
>+  t = absl::TimeFromTimespec(ts);
>+  ts = absl::ToTimespec(t);
>+  EXPECT_EQ(max_timespec_sec, ts.tv_sec);
>+  EXPECT_EQ(999999998, ts.tv_nsec);
>+  t += absl::Nanoseconds(1);
>+  ts = absl::ToTimespec(t);
>+  EXPECT_EQ(max_timespec_sec, ts.tv_sec);
>+  EXPECT_EQ(999999999, ts.tv_nsec);
>+  t += absl::Nanoseconds(1);  // no effect
>+  ts = absl::ToTimespec(t);
>+  EXPECT_EQ(max_timespec_sec, ts.tv_sec);
>+  EXPECT_EQ(999999999, ts.tv_nsec);
>+
>+  ts.tv_sec = min_timespec_sec;
>+  ts.tv_nsec = 1;
>+  t = absl::TimeFromTimespec(ts);
>+  ts = absl::ToTimespec(t);
>+  EXPECT_EQ(min_timespec_sec, ts.tv_sec);
>+  EXPECT_EQ(1, ts.tv_nsec);
>+  t -= absl::Nanoseconds(1);
>+  ts = absl::ToTimespec(t);
>+  EXPECT_EQ(min_timespec_sec, ts.tv_sec);
>+  EXPECT_EQ(0, ts.tv_nsec);
>+  t -= absl::Nanoseconds(1);  // no effect
>+  ts = absl::ToTimespec(t);
>+  EXPECT_EQ(min_timespec_sec, ts.tv_sec);
>+  EXPECT_EQ(0, ts.tv_nsec);
>+
>+  // Checks how Time::In() saturates on infinities.
>+  absl::Time::Breakdown bd = absl::InfiniteFuture().In(utc);
>+  ABSL_INTERNAL_EXPECT_TIME(bd, std::numeric_limits<int64_t>::max(), 12, 31, 23,
>+                            59, 59, 0, false);
>+  EXPECT_EQ(absl::InfiniteDuration(), bd.subsecond);
>+  EXPECT_EQ(4, bd.weekday);  // Thursday
>+  EXPECT_EQ(365, bd.yearday);
>+  EXPECT_STREQ("-00", bd.zone_abbr);  // artifact of absl::Time::In()
>+  bd = absl::InfinitePast().In(utc);
>+  ABSL_INTERNAL_EXPECT_TIME(bd, std::numeric_limits<int64_t>::min(), 1, 1, 0, 0,
>+                            0, 0, false);
>+  EXPECT_EQ(-absl::InfiniteDuration(), bd.subsecond);
>+  EXPECT_EQ(7, bd.weekday);  // Sunday
>+  EXPECT_EQ(1, bd.yearday);
>+  EXPECT_STREQ("-00", bd.zone_abbr);  // artifact of absl::Time::In()
>+
>+  // Approach the maximal Time value from below.
>+  t = absl::FromDateTime(292277026596, 12, 4, 15, 30, 6, utc);
>+  EXPECT_EQ("292277026596-12-04T15:30:06+00:00",
>+            absl::FormatTime(absl::RFC3339_full, t, utc));
>+  t = absl::FromDateTime(292277026596, 12, 4, 15, 30, 7, utc);
>+  EXPECT_EQ("292277026596-12-04T15:30:07+00:00",
>+            absl::FormatTime(absl::RFC3339_full, t, utc));
>+  EXPECT_EQ(
>+      absl::UnixEpoch() + absl::Seconds(std::numeric_limits<int64_t>::max()), t);
>+
>+  // Checks that we can also get the maximal Time value for a far-east zone.
>+  const absl::TimeZone plus14 = absl::FixedTimeZone(14 * 60 * 60);
>+  t = absl::FromDateTime(292277026596, 12, 5, 5, 30, 7, plus14);
>+  EXPECT_EQ("292277026596-12-05T05:30:07+14:00",
>+            absl::FormatTime(absl::RFC3339_full, t, plus14));
>+  EXPECT_EQ(
>+      absl::UnixEpoch() + absl::Seconds(std::numeric_limits<int64_t>::max()), t);
>+
>+  // One second later should push us to infinity.
>+  t = absl::FromDateTime(292277026596, 12, 4, 15, 30, 8, utc);
>+  EXPECT_EQ("infinite-future", absl::FormatTime(absl::RFC3339_full, t, utc));
>+
>+  // Approach the minimal Time value from above.
>+  t = absl::FromDateTime(-292277022657, 1, 27, 8, 29, 53, utc);
>+  EXPECT_EQ("-292277022657-01-27T08:29:53+00:00",
>+            absl::FormatTime(absl::RFC3339_full, t, utc));
>+  t = absl::FromDateTime(-292277022657, 1, 27, 8, 29, 52, utc);
>+  EXPECT_EQ("-292277022657-01-27T08:29:52+00:00",
>+            absl::FormatTime(absl::RFC3339_full, t, utc));
>+  EXPECT_EQ(
>+      absl::UnixEpoch() + absl::Seconds(std::numeric_limits<int64_t>::min()), t);
>+
>+  // Checks that we can also get the minimal Time value for a far-west zone.
>+  const absl::TimeZone minus12 = absl::FixedTimeZone(-12 * 60 * 60);
>+  t = absl::FromDateTime(-292277022657, 1, 26, 20, 29, 52, minus12);
>+  EXPECT_EQ("-292277022657-01-26T20:29:52-12:00",
>+            absl::FormatTime(absl::RFC3339_full, t, minus12));
>+  EXPECT_EQ(
>+      absl::UnixEpoch() + absl::Seconds(std::numeric_limits<int64_t>::min()), t);
>+
>+  // One second before should push us to -infinity.
>+  t = absl::FromDateTime(-292277022657, 1, 27, 8, 29, 51, utc);
>+  EXPECT_EQ("infinite-past", absl::FormatTime(absl::RFC3339_full, t, utc));
>+}
>+
>+// In zones with POSIX-style recurring rules we use special logic to
>+// handle conversions in the distant future.  Here we check the limits
>+// of those conversions, particularly with respect to integer overflow.
>+TEST(Time, ExtendedConversionSaturation) {
>+  const absl::TimeZone syd =
>+      absl::time_internal::LoadTimeZone("Australia/Sydney");
>+  const absl::TimeZone nyc =
>+      absl::time_internal::LoadTimeZone("America/New_York");
>+  const absl::Time max =
>+      absl::FromUnixSeconds(std::numeric_limits<int64_t>::max());
>+  absl::Time::Breakdown bd;
>+  absl::Time t;
>+
>+  // The maximal time converted in each zone.
>+  bd = max.In(syd);
>+  ABSL_INTERNAL_EXPECT_TIME(bd, 292277026596, 12, 5, 2, 30, 7, 39600, true);
>+  t = absl::FromDateTime(292277026596, 12, 5, 2, 30, 7, syd);
>+  EXPECT_EQ(max, t);
>+  bd = max.In(nyc);
>+  ABSL_INTERNAL_EXPECT_TIME(bd, 292277026596, 12, 4, 10, 30, 7, -18000, false);
>+  t = absl::FromDateTime(292277026596, 12, 4, 10, 30, 7, nyc);
>+  EXPECT_EQ(max, t);
>+
>+  // One second later should push us to infinity.
>+  t = absl::FromDateTime(292277026596, 12, 5, 2, 30, 8, syd);
>+  EXPECT_EQ(absl::InfiniteFuture(), t);
>+  t = absl::FromDateTime(292277026596, 12, 4, 10, 30, 8, nyc);
>+  EXPECT_EQ(absl::InfiniteFuture(), t);
>+
>+  // And we should stick there.
>+  t = absl::FromDateTime(292277026596, 12, 5, 2, 30, 9, syd);
>+  EXPECT_EQ(absl::InfiniteFuture(), t);
>+  t = absl::FromDateTime(292277026596, 12, 4, 10, 30, 9, nyc);
>+  EXPECT_EQ(absl::InfiniteFuture(), t);
>+
>+  // All the way up to a saturated date/time, without overflow.
>+  t = absl::FromDateTime(
>+      std::numeric_limits<int64_t>::max(), 12, 31, 23, 59, 59, syd);
>+  EXPECT_EQ(absl::InfiniteFuture(), t);
>+  t = absl::FromDateTime(
>+      std::numeric_limits<int64_t>::max(), 12, 31, 23, 59, 59, nyc);
>+  EXPECT_EQ(absl::InfiniteFuture(), t);
>+}
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/time_zone_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/time_zone_test.cc
>new file mode 100644
>index 00000000000..43d91904044
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/time/time_zone_test.cc
>@@ -0,0 +1,97 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/time/internal/cctz/include/cctz/time_zone.h"
>+
>+#include "gtest/gtest.h"
>+#include "absl/time/internal/test_util.h"
>+#include "absl/time/time.h"
>+
>+namespace cctz = absl::time_internal::cctz;
>+
>+namespace {
>+
>+TEST(TimeZone, ValueSemantics) {
>+  absl::TimeZone tz;
>+  absl::TimeZone tz2 = tz;  // Copy-construct
>+  EXPECT_EQ(tz, tz2);
>+  tz2 = tz;  // Copy-assign
>+  EXPECT_EQ(tz, tz2);
>+}
>+
>+TEST(TimeZone, Equality) {
>+  absl::TimeZone a, b;
>+  EXPECT_EQ(a, b);
>+  EXPECT_EQ(a.name(), b.name());
>+
>+  absl::TimeZone implicit_utc;
>+  absl::TimeZone explicit_utc = absl::UTCTimeZone();
>+  EXPECT_EQ(implicit_utc, explicit_utc);
>+  EXPECT_EQ(implicit_utc.name(), explicit_utc.name());
>+
>+  absl::TimeZone la = absl::time_internal::LoadTimeZone("America/Los_Angeles");
>+  absl::TimeZone nyc = absl::time_internal::LoadTimeZone("America/New_York");
>+  EXPECT_NE(la, nyc);
>+}
>+
>+TEST(TimeZone, CCTZConversion) {
>+  const cctz::time_zone cz = cctz::utc_time_zone();
>+  const absl::TimeZone tz(cz);
>+  EXPECT_EQ(cz, cctz::time_zone(tz));
>+}
>+
>+TEST(TimeZone, DefaultTimeZones) {
>+  absl::TimeZone tz;
>+  EXPECT_EQ("UTC", absl::TimeZone().name());
>+  EXPECT_EQ("UTC", absl::UTCTimeZone().name());
>+}
>+
>+TEST(TimeZone, FixedTimeZone) {
>+  const absl::TimeZone tz = absl::FixedTimeZone(123);
>+  const cctz::time_zone cz = cctz::fixed_time_zone(cctz::seconds(123));
>+  EXPECT_EQ(tz, absl::TimeZone(cz));
>+}
>+
>+TEST(TimeZone, LocalTimeZone) {
>+  const absl::TimeZone local_tz = absl::LocalTimeZone();
>+  absl::TimeZone tz = absl::time_internal::LoadTimeZone("localtime");
>+  EXPECT_EQ(tz, local_tz);
>+}
>+
>+TEST(TimeZone, NamedTimeZones) {
>+  absl::TimeZone nyc = absl::time_internal::LoadTimeZone("America/New_York");
>+  EXPECT_EQ("America/New_York", nyc.name());
>+  absl::TimeZone syd = absl::time_internal::LoadTimeZone("Australia/Sydney");
>+  EXPECT_EQ("Australia/Sydney", syd.name());
>+  absl::TimeZone fixed = absl::FixedTimeZone((((3 * 60) + 25) * 60) + 45);
>+  EXPECT_EQ("Fixed/UTC+03:25:45", fixed.name());
>+}
>+
>+TEST(TimeZone, Failures) {
>+  absl::TimeZone tz = absl::time_internal::LoadTimeZone("America/Los_Angeles");
>+  EXPECT_FALSE(LoadTimeZone("Invalid/TimeZone", &tz));
>+  EXPECT_EQ(absl::UTCTimeZone(), tz);  // guaranteed fallback to UTC
>+
>+  // Ensures that the load still fails on a subsequent attempt.
>+  tz = absl::time_internal::LoadTimeZone("America/Los_Angeles");
>+  EXPECT_FALSE(LoadTimeZone("Invalid/TimeZone", &tz));
>+  EXPECT_EQ(absl::UTCTimeZone(), tz);  // guaranteed fallback to UTC
>+
>+  // Loading an empty std::string timezone should fail.
>+  tz = absl::time_internal::LoadTimeZone("America/Los_Angeles");
>+  EXPECT_FALSE(LoadTimeZone("", &tz));
>+  EXPECT_EQ(absl::UTCTimeZone(), tz);  // guaranteed fallback to UTC
>+}
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/BUILD.bazel b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/BUILD.bazel
>new file mode 100644
>index 00000000000..096c119e638
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/BUILD.bazel
>@@ -0,0 +1,280 @@
>+#
>+# Copyright 2017 The Abseil Authors.
>+#
>+# Licensed under the Apache License, Version 2.0 (the "License");
>+# you may not use this file except in compliance with the License.
>+# You may obtain a copy of the License at
>+#
>+#      http://www.apache.org/licenses/LICENSE-2.0
>+#
>+# Unless required by applicable law or agreed to in writing, software
>+# distributed under the License is distributed on an "AS IS" BASIS,
>+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+# See the License for the specific language governing permissions and
>+# limitations under the License.
>+#
>+
>+load(
>+    "//absl:copts.bzl",
>+    "ABSL_DEFAULT_COPTS",
>+    "ABSL_TEST_COPTS",
>+    "ABSL_EXCEPTIONS_FLAG",
>+)
>+
>+package(default_visibility = ["//visibility:public"])
>+
>+licenses(["notice"])  # Apache 2.0
>+
>+cc_library(
>+    name = "any",
>+    hdrs = ["any.h"],
>+    copts = ABSL_DEFAULT_COPTS,
>+    deps = [
>+        ":bad_any_cast",
>+        "//absl/base:config",
>+        "//absl/base:core_headers",
>+        "//absl/meta:type_traits",
>+        "//absl/utility",
>+    ],
>+)
>+
>+cc_library(
>+    name = "bad_any_cast",
>+    hdrs = ["bad_any_cast.h"],
>+    copts = ABSL_DEFAULT_COPTS,
>+    deps = [
>+        ":bad_any_cast_impl",
>+        "//absl/base:config",
>+    ],
>+)
>+
>+cc_library(
>+    name = "bad_any_cast_impl",
>+    srcs = [
>+        "bad_any_cast.cc",
>+        "bad_any_cast.h",
>+    ],
>+    copts = ABSL_EXCEPTIONS_FLAG + ABSL_DEFAULT_COPTS,
>+    visibility = ["//visibility:private"],
>+    deps = [
>+        "//absl/base",
>+        "//absl/base:config",
>+    ],
>+)
>+
>+cc_test(
>+    name = "any_test",
>+    size = "small",
>+    srcs = [
>+        "any_test.cc",
>+    ],
>+    copts = ABSL_TEST_COPTS + ABSL_EXCEPTIONS_FLAG,
>+    deps = [
>+        ":any",
>+        "//absl/base",
>+        "//absl/base:config",
>+        "//absl/base:exception_testing",
>+        "//absl/container:test_instance_tracker",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "any_test_noexceptions",
>+    size = "small",
>+    srcs = [
>+        "any_test.cc",
>+    ],
>+    copts = ABSL_TEST_COPTS,
>+    deps = [
>+        ":any",
>+        "//absl/base",
>+        "//absl/base:config",
>+        "//absl/base:exception_testing",
>+        "//absl/container:test_instance_tracker",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "any_exception_safety_test",
>+    srcs = ["any_exception_safety_test.cc"],
>+    copts = ABSL_TEST_COPTS + ABSL_EXCEPTIONS_FLAG,
>+    deps = [
>+        ":any",
>+        "//absl/base:exception_safety_testing",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_library(
>+    name = "span",
>+    hdrs = ["span.h"],
>+    copts = ABSL_DEFAULT_COPTS,
>+    deps = [
>+        "//absl/algorithm",
>+        "//absl/base:core_headers",
>+        "//absl/base:throw_delegate",
>+        "//absl/meta:type_traits",
>+    ],
>+)
>+
>+cc_test(
>+    name = "span_test",
>+    size = "small",
>+    srcs = ["span_test.cc"],
>+    copts = ABSL_TEST_COPTS + ABSL_EXCEPTIONS_FLAG,
>+    deps = [
>+        ":span",
>+        "//absl/base:config",
>+        "//absl/base:core_headers",
>+        "//absl/base:exception_testing",
>+        "//absl/container:fixed_array",
>+        "//absl/container:inlined_vector",
>+        "//absl/strings",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "span_test_noexceptions",
>+    size = "small",
>+    srcs = ["span_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    deps = [
>+        ":span",
>+        "//absl/base:config",
>+        "//absl/base:core_headers",
>+        "//absl/base:exception_testing",
>+        "//absl/container:fixed_array",
>+        "//absl/container:inlined_vector",
>+        "//absl/strings",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_library(
>+    name = "optional",
>+    srcs = ["optional.cc"],
>+    hdrs = ["optional.h"],
>+    copts = ABSL_DEFAULT_COPTS,
>+    deps = [
>+        ":bad_optional_access",
>+        "//absl/base:config",
>+        "//absl/memory",
>+        "//absl/meta:type_traits",
>+        "//absl/utility",
>+    ],
>+)
>+
>+cc_library(
>+    name = "bad_optional_access",
>+    srcs = ["bad_optional_access.cc"],
>+    hdrs = ["bad_optional_access.h"],
>+    copts = ABSL_DEFAULT_COPTS + ABSL_EXCEPTIONS_FLAG,
>+    deps = [
>+        "//absl/base",
>+        "//absl/base:config",
>+    ],
>+)
>+
>+cc_library(
>+    name = "bad_variant_access",
>+    srcs = ["bad_variant_access.cc"],
>+    hdrs = ["bad_variant_access.h"],
>+    copts = ABSL_EXCEPTIONS_FLAG + ABSL_DEFAULT_COPTS,
>+    deps = [
>+        "//absl/base",
>+        "//absl/base:config",
>+    ],
>+)
>+
>+cc_test(
>+    name = "optional_test",
>+    size = "small",
>+    srcs = [
>+        "optional_test.cc",
>+    ],
>+    copts = ABSL_TEST_COPTS + ABSL_EXCEPTIONS_FLAG,
>+    deps = [
>+        ":optional",
>+        "//absl/base",
>+        "//absl/base:config",
>+        "//absl/meta:type_traits",
>+        "//absl/strings",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "optional_exception_safety_test",
>+    srcs = [
>+        "optional_exception_safety_test.cc",
>+    ],
>+    copts = ABSL_TEST_COPTS + ABSL_EXCEPTIONS_FLAG,
>+    deps = [
>+        ":optional",
>+        "//absl/base:exception_safety_testing",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_library(
>+    name = "variant",
>+    srcs = ["internal/variant.h"],
>+    hdrs = ["variant.h"],
>+    copts = ABSL_DEFAULT_COPTS,
>+    deps = [
>+        ":bad_variant_access",
>+        "//absl/base:base_internal",
>+        "//absl/base:config",
>+        "//absl/base:core_headers",
>+        "//absl/meta:type_traits",
>+        "//absl/utility",
>+    ],
>+)
>+
>+cc_test(
>+    name = "variant_test",
>+    size = "small",
>+    srcs = ["variant_test.cc"],
>+    copts = ABSL_TEST_COPTS + ABSL_EXCEPTIONS_FLAG,
>+    deps = [
>+        ":variant",
>+        "//absl/base:config",
>+        "//absl/base:core_headers",
>+        "//absl/memory",
>+        "//absl/meta:type_traits",
>+        "//absl/strings",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "variant_benchmark",
>+    srcs = [
>+        "variant_benchmark.cc",
>+    ],
>+    copts = ABSL_TEST_COPTS,
>+    tags = ["benchmark"],
>+    deps = [
>+        ":variant",
>+        "//absl/utility",
>+        "@com_github_google_benchmark//:benchmark_main",
>+    ],
>+)
>+
>+cc_test(
>+    name = "variant_exception_safety_test",
>+    size = "small",
>+    srcs = [
>+        "variant_exception_safety_test.cc",
>+    ],
>+    copts = ABSL_TEST_COPTS + ABSL_EXCEPTIONS_FLAG,
>+    deps = [
>+        ":variant",
>+        "//absl/base:exception_safety_testing",
>+        "//absl/memory",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/BUILD.gn
>new file mode 100644
>index 00000000000..fd7e89dee34
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/BUILD.gn
>@@ -0,0 +1,172 @@
>+# Copyright 2018 The Chromium Authors. All rights reserved.
>+# Use of this source code is governed by a BSD-style license that can be
>+# found in the LICENSE file.
>+
>+import("//build_overrides/build.gni")
>+
>+if (build_with_chromium) {
>+  visibility = [
>+    "//third_party/webrtc/*",
>+    "//third_party/abseil-cpp/*",
>+    "//third_party/googletest:gtest",
>+  ]
>+} else {
>+  visibility = [ "*" ]
>+}
>+
>+source_set("any") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  public = [
>+    "any.h",
>+  ]
>+  deps = [
>+    ":bad_any_cast",
>+    "../base:config",
>+    "../base:core_headers",
>+    "../meta:type_traits",
>+    "../utility",
>+  ]
>+}
>+
>+source_set("bad_any_cast") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  public = [
>+    "bad_any_cast.h",
>+  ]
>+  deps = [
>+    ":bad_any_cast_impl",
>+    "../base:config",
>+  ]
>+}
>+
>+source_set("bad_any_cast_impl") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  sources = [
>+    "bad_any_cast.cc",
>+  ]
>+  public = [
>+    "bad_any_cast.h",
>+  ]
>+  deps = [
>+    "../base",
>+    "../base:config",
>+  ]
>+  visibility = []
>+  visibility += [ ":*" ]
>+}
>+
>+source_set("span") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  public = [
>+    "span.h",
>+  ]
>+  deps = [
>+    "../algorithm",
>+    "../base:core_headers",
>+    "../base:throw_delegate",
>+    "../meta:type_traits",
>+  ]
>+}
>+
>+source_set("optional") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  sources = [
>+    "optional.cc",
>+  ]
>+  public = [
>+    "optional.h",
>+  ]
>+  deps = [
>+    ":bad_optional_access",
>+    "../base:config",
>+    "../memory",
>+    "../meta:type_traits",
>+    "../utility",
>+  ]
>+}
>+
>+source_set("bad_optional_access") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  sources = [
>+    "bad_optional_access.cc",
>+  ]
>+  public = [
>+    "bad_optional_access.h",
>+  ]
>+  deps = [
>+    "../base",
>+    "../base:config",
>+  ]
>+}
>+
>+source_set("bad_variant_access") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  sources = [
>+    "bad_variant_access.cc",
>+  ]
>+  public = [
>+    "bad_variant_access.h",
>+  ]
>+  deps = [
>+    "../base",
>+    "../base:config",
>+  ]
>+}
>+
>+source_set("variant") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  sources = [
>+    "internal/variant.h",
>+  ]
>+  public = [
>+    "variant.h",
>+  ]
>+  deps = [
>+    ":bad_variant_access",
>+    "../base:base_internal",
>+    "../base:config",
>+    "../base:core_headers",
>+    "../meta:type_traits",
>+    "../utility",
>+  ]
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/CMakeLists.txt b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/CMakeLists.txt
>new file mode 100644
>index 00000000000..2f2e3a77808
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/CMakeLists.txt
>@@ -0,0 +1,228 @@
>+#
>+# Copyright 2017 The Abseil Authors.
>+#
>+# Licensed under the Apache License, Version 2.0 (the "License");
>+# you may not use this file except in compliance with the License.
>+# You may obtain a copy of the License at
>+#
>+#      http://www.apache.org/licenses/LICENSE-2.0
>+#
>+# Unless required by applicable law or agreed to in writing, software
>+# distributed under the License is distributed on an "AS IS" BASIS,
>+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+# See the License for the specific language governing permissions and
>+# limitations under the License.
>+#
>+
>+list(APPEND TYPES_PUBLIC_HEADERS
>+  "any.h"
>+  "bad_any_cast.h"
>+  "bad_optional_access.h"
>+  "optional.h"
>+  "span.h"
>+  "variant.h"
>+)
>+
>+
>+# any library
>+absl_header_library(
>+  TARGET
>+    absl_any
>+  PUBLIC_LIBRARIES
>+    absl::bad_any_cast
>+    absl::base
>+    absl::meta
>+    absl::utility
>+  PRIVATE_COMPILE_FLAGS
>+    ${ABSL_EXCEPTIONS_FLAG}
>+  EXPORT_NAME
>+    any
>+)
>+
>+# span library
>+absl_header_library(
>+  TARGET
>+    absl_span
>+  PUBLIC_LIBRARIES
>+    absl::utility
>+  EXPORT_NAME
>+    span
>+)
>+
>+
>+# bad_any_cast library
>+list(APPEND BAD_ANY_CAST_SRC
>+  "bad_any_cast.cc"
>+  ${TYPES_PUBLIC_HEADERS}
>+)
>+
>+absl_library(
>+  TARGET
>+    absl_bad_any_cast
>+  SOURCES
>+    ${BAD_ANY_CAST_SRC}
>+  PUBLIC_LIBRARIES
>+  EXPORT_NAME
>+    bad_any_cast
>+)
>+
>+
>+# optional library
>+list(APPEND OPTIONAL_SRC
>+  "optional.cc"
>+)
>+
>+absl_library(
>+  TARGET
>+    absl_optional
>+  SOURCES
>+    ${OPTIONAL_SRC}
>+  PUBLIC_LIBRARIES
>+    absl::bad_optional_access
>+    absl::base
>+    absl::memory
>+    absl::meta
>+    absl::utility
>+  EXPORT_NAME
>+    optional
>+)
>+
>+
>+set(BAD_OPTIONAL_ACCESS_SRC "bad_optional_access.cc")
>+set(BAD_OPTIONAL_ACCESS_LIBRARIES absl::base)
>+
>+absl_library(
>+  TARGET
>+    absl_bad_optional_access
>+  SOURCES
>+    ${BAD_OPTIONAL_ACCESS_SRC}
>+  PUBLIC_LIBRARIES
>+    ${BAD_OPTIONAL_ACCESS_PUBLIC_LIBRARIES}
>+  EXPORT_NAME
>+    bad_optional_access
>+)
>+
>+# variant library
>+absl_library(
>+  TARGET
>+    absl_variant
>+  SOURCES
>+    "bad_variant_access.h" "bad_variant_access.cc" "variant.h" "internal/variant.h"
>+  PUBLIC_LIBRARIES
>+    absl::base absl::meta absl::utility
>+  PRIVATE_COMPILE_FLAGS
>+    ${ABSL_EXCEPTIONS_FLAG}
>+  EXPORT_NAME
>+    variant
>+)
>+
>+#
>+## TESTS
>+#
>+
>+
>+# test any_test
>+set(ANY_TEST_SRC "any_test.cc")
>+set(ANY_TEST_PUBLIC_LIBRARIES absl::base absl::throw_delegate absl::any absl::bad_any_cast test_instance_tracker_lib)
>+
>+absl_test(
>+  TARGET
>+    any_test
>+  SOURCES
>+    ${ANY_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${ANY_TEST_PUBLIC_LIBRARIES}
>+  PRIVATE_COMPILE_FLAGS
>+    ${ABSL_EXCEPTIONS_FLAG}
>+)
>+
>+
>+# test any_test_noexceptions
>+absl_test(
>+  TARGET
>+    any_test_noexceptions
>+  SOURCES
>+    ${ANY_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${ANY_TEST_PUBLIC_LIBRARIES}
>+)
>+
>+# test any_exception_safety_test
>+set(ANY_EXCEPTION_SAFETY_TEST_SRC "any_exception_safety_test.cc")
>+set(ANY_EXCEPTION_SAFETY_TEST_PUBLIC_LIBRARIES
>+  absl::any
>+  absl::base
>+  absl_base_internal_exception_safety_testing
>+)
>+
>+absl_test(
>+  TARGET
>+    any_exception_safety_test
>+  SOURCES
>+    ${ANY_EXCEPTION_SAFETY_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${ANY_EXCEPTION_SAFETY_TEST_PUBLIC_LIBRARIES}
>+  PRIVATE_COMPILE_FLAGS
>+    ${ABSL_EXCEPTIONS_FLAG}
>+)
>+
>+
>+# test span_test
>+set(SPAN_TEST_SRC "span_test.cc")
>+set(SPAN_TEST_PUBLIC_LIBRARIES absl::base absl::strings absl::throw_delegate absl::span test_instance_tracker_lib)
>+
>+absl_test(
>+  TARGET
>+    span_test
>+  SOURCES
>+    ${SPAN_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${SPAN_TEST_PUBLIC_LIBRARIES}
>+  PRIVATE_COMPILE_FLAGS
>+    ${ABSL_EXCEPTIONS_FLAG}
>+)
>+
>+
>+# test span_test_noexceptions
>+absl_test(
>+  TARGET
>+    span_test_noexceptions
>+  SOURCES
>+    ${SPAN_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${SPAN_TEST_PUBLIC_LIBRARIES}
>+)
>+
>+
>+
>+# test optional_test
>+set(OPTIONAL_TEST_SRC "optional_test.cc")
>+set(OPTIONAL_TEST_PUBLIC_LIBRARIES absl::base absl::throw_delegate absl::optional absl_bad_optional_access)
>+
>+absl_test(
>+  TARGET
>+    optional_test
>+  SOURCES
>+    ${OPTIONAL_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${OPTIONAL_TEST_PUBLIC_LIBRARIES}
>+)
>+
>+
>+# test optional_exception_safety_test
>+set(OPTIONAL_EXCEPTION_SAFETY_TEST_SRC "optional_exception_safety_test.cc")
>+set(OPTIONAL_EXCEPTION_SAFETY_TEST_PUBLIC_LIBRARIES
>+  absl::optional
>+  absl_base_internal_exception_safety_testing
>+)
>+
>+absl_test(
>+  TARGET
>+    optional_exception_safety_test
>+  SOURCES
>+    ${OPTIONAL_EXCEPTION_SAFETY_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${OPTIONAL_EXCEPTION_SAFETY_TEST_PUBLIC_LIBRARIES}
>+  PRIVATE_COMPILE_FLAGS
>+    ${ABSL_EXCEPTIONS_FLAG}
>+)
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/any.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/any.h
>new file mode 100644
>index 00000000000..a973c6dab7e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/any.h
>@@ -0,0 +1,539 @@
>+//
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// -----------------------------------------------------------------------------
>+// any.h
>+// -----------------------------------------------------------------------------
>+//
>+// This header file define the `absl::any` type for holding a type-safe value
>+// of any type. The 'absl::any` type is useful for providing a way to hold
>+// something that is, as yet, unspecified. Such unspecified types
>+// traditionally are passed between API boundaries until they are later cast to
>+// their "destination" types. To cast to such a destination type, use
>+// `absl::any_cast()`. Note that when casting an `absl::any`, you must cast it
>+// to an explicit type; implicit conversions will throw.
>+//
>+// Example:
>+//
>+//   auto a = absl::any(65);
>+//   absl::any_cast<int>(a);         // 65
>+//   absl::any_cast<char>(a);        // throws absl::bad_any_cast
>+//   absl::any_cast<std::string>(a); // throws absl::bad_any_cast
>+//
>+// `absl::any` is a C++11 compatible version of the C++17 `std::any` abstraction
>+// and is designed to be a drop-in replacement for code compliant with C++17.
>+//
>+// Traditionally, the behavior of casting to a temporary unspecified type has
>+// been accomplished with the `void *` paradigm, where the pointer was to some
>+// other unspecified type. `absl::any` provides an "owning" version of `void *`
>+// that avoids issues of pointer management.
>+//
>+// Note: just as in the case of `void *`, use of `absl::any` (and its C++17
>+// version `std::any`) is a code smell indicating that your API might not be
>+// constructed correctly. We have seen that most uses of `any` are unwarranted,
>+// and `absl::any`, like `std::any`, is difficult to use properly. Before using
>+// this abstraction, make sure that you should not instead be rewriting your
>+// code to be more specific.
>+//
>+// Abseil expects to release an `absl::variant` type shortly (a C++11 compatible
>+// version of the C++17 `std::variant), which is generally preferred for use
>+// over `absl::any`.
>+#ifndef ABSL_TYPES_ANY_H_
>+#define ABSL_TYPES_ANY_H_
>+
>+#include "absl/base/config.h"
>+#include "absl/utility/utility.h"
>+
>+#ifdef ABSL_HAVE_STD_ANY
>+
>+#include <any>
>+
>+namespace absl {
>+using std::any;
>+using std::any_cast;
>+using std::bad_any_cast;
>+using std::make_any;
>+}  // namespace absl
>+
>+#else  // ABSL_HAVE_STD_ANY
>+
>+#include <algorithm>
>+#include <cstddef>
>+#include <initializer_list>
>+#include <memory>
>+#include <stdexcept>
>+#include <type_traits>
>+#include <typeinfo>
>+#include <utility>
>+
>+#include "absl/base/macros.h"
>+#include "absl/meta/type_traits.h"
>+#include "absl/types/bad_any_cast.h"
>+
>+// NOTE: This macro is an implementation detail that is undefined at the bottom
>+// of the file. It is not intended for expansion directly from user code.
>+#ifdef ABSL_ANY_DETAIL_HAS_RTTI
>+#error ABSL_ANY_DETAIL_HAS_RTTI cannot be directly set
>+#elif !defined(__GNUC__) || defined(__GXX_RTTI)
>+#define ABSL_ANY_DETAIL_HAS_RTTI 1
>+#endif  // !defined(__GNUC__) || defined(__GXX_RTTI)
>+
>+namespace absl {
>+
>+namespace any_internal {
>+
>+template <typename Type>
>+struct TypeTag {
>+  constexpr static char dummy_var = 0;
>+};
>+
>+template <typename Type>
>+constexpr char TypeTag<Type>::dummy_var;
>+
>+// FastTypeId<Type>() evaluates at compile/link-time to a unique pointer for the
>+// passed in type. These are meant to be good match for keys into maps or
>+// straight up comparisons.
>+template<typename Type>
>+constexpr inline const void* FastTypeId() {
>+  return &TypeTag<Type>::dummy_var;
>+}
>+
>+}  // namespace any_internal
>+
>+class any;
>+
>+// swap()
>+//
>+// Swaps two `absl::any` values. Equivalent to `x.swap(y) where `x` and `y` are
>+// `absl::any` types.
>+void swap(any& x, any& y) noexcept;
>+
>+// make_any()
>+//
>+// Constructs an `absl::any` of type `T` with the given arguments.
>+template <typename T, typename... Args>
>+any make_any(Args&&... args);
>+
>+// Overload of `absl::make_any()` for constructing an `absl::any` type from an
>+// initializer list.
>+template <typename T, typename U, typename... Args>
>+any make_any(std::initializer_list<U> il, Args&&... args);
>+
>+// any_cast()
>+//
>+// Statically casts the value of a `const absl::any` type to the given type.
>+// This function will throw `absl::bad_any_cast` if the stored value type of the
>+// `absl::any` does not match the cast.
>+//
>+// `any_cast()` can also be used to get a reference to the internal storage iff
>+// a reference type is passed as its `ValueType`:
>+//
>+// Example:
>+//
>+//   absl::any my_any = std::vector<int>();
>+//   absl::any_cast<std::vector<int>&>(my_any).push_back(42);
>+template <typename ValueType>
>+ValueType any_cast(const any& operand);
>+
>+// Overload of `any_cast()` to statically cast the value of a non-const
>+// `absl::any` type to the given type. This function will throw
>+// `absl::bad_any_cast` if the stored value type of the `absl::any` does not
>+// match the cast.
>+template <typename ValueType>
>+ValueType any_cast(any& operand);  // NOLINT(runtime/references)
>+
>+// Overload of `any_cast()` to statically cast the rvalue of an `absl::any`
>+// type. This function will throw `absl::bad_any_cast` if the stored value type
>+// of the `absl::any` does not match the cast.
>+template <typename ValueType>
>+ValueType any_cast(any&& operand);
>+
>+// Overload of `any_cast()` to statically cast the value of a const pointer
>+// `absl::any` type to the given pointer type, or `nullptr` if the stored value
>+// type of the `absl::any` does not match the cast.
>+template <typename ValueType>
>+const ValueType* any_cast(const any* operand) noexcept;
>+
>+// Overload of `any_cast()` to statically cast the value of a pointer
>+// `absl::any` type to the given pointer type, or `nullptr` if the stored value
>+// type of the `absl::any` does not match the cast.
>+template <typename ValueType>
>+ValueType* any_cast(any* operand) noexcept;
>+
>+// -----------------------------------------------------------------------------
>+// absl::any
>+// -----------------------------------------------------------------------------
>+//
>+// An `absl::any` object provides the facility to either store an instance of a
>+// type, known as the "contained object", or no value. An `absl::any` is used to
>+// store values of types that are unknown at compile time. The `absl::any`
>+// object, when containing a value, must contain a value type; storing a
>+// reference type is neither desired nor supported.
>+//
>+// An `absl::any` can only store a type that is copy-constructable; move-only
>+// types are not allowed within an `any` object.
>+//
>+// Example:
>+//
>+//   auto a = absl::any(65);                 // Literal, copyable
>+//   auto b = absl::any(std::vector<int>()); // Default-initialized, copyable
>+//   std::unique_ptr<Foo> my_foo;
>+//   auto c = absl::any(std::move(my_foo));  // Error, not copy-constructable
>+//
>+// Note that `absl::any` makes use of decayed types (`absl::decay_t` in this
>+// context) to remove const-volatile qualifiers (known as "cv qualifiers"),
>+// decay functions to function pointers, etc. We essentially "decay" a given
>+// type into its essential type.
>+//
>+// `absl::any` makes use of decayed types when determining the basic type `T` of
>+// the value to store in the any's contained object. In the documentation below,
>+// we explicitly denote this by using the phrase "a decayed type of `T`".
>+//
>+// Example:
>+//
>+//   const int a = 4;
>+//   absl::any foo(a);  // Decay ensures we store an "int", not a "const int&".
>+//
>+//   void my_function() {}
>+//   absl::any bar(my_function);  // Decay ensures we store a function pointer.
>+//
>+// `absl::any` is a C++11 compatible version of the C++17 `std::any` abstraction
>+// and is designed to be a drop-in replacement for code compliant with C++17.
>+class any {
>+ private:
>+  template <typename T>
>+  struct IsInPlaceType;
>+
>+ public:
>+  // Constructors
>+
>+  // Constructs an empty `absl::any` object (`any::has_value()` will return
>+  // `false`).
>+  constexpr any() noexcept;
>+
>+  // Copy constructs an `absl::any` object with a "contained object" of the
>+  // passed type of `other` (or an empty `absl::any` if `other.has_value()` is
>+  // `false`.
>+  any(const any& other)
>+      : obj_(other.has_value() ? other.obj_->Clone()
>+                               : std::unique_ptr<ObjInterface>()) {}
>+
>+  // Move constructs an `absl::any` object with a "contained object" of the
>+  // passed type of `other` (or an empty `absl::any` if `other.has_value()` is
>+  // `false`).
>+  any(any&& other) noexcept = default;
>+
>+  // Constructs an `absl::any` object with a "contained object" of the decayed
>+  // type of `T`, which is initialized via `std::forward<T>(value)`.
>+  //
>+  // This constructor will not participate in overload resolution if the
>+  // decayed type of `T` is not copy-constructible.
>+  template <
>+      typename T, typename VT = absl::decay_t<T>,
>+      absl::enable_if_t<!absl::disjunction<
>+          std::is_same<any, VT>, IsInPlaceType<VT>,
>+          absl::negation<std::is_copy_constructible<VT> > >::value>* = nullptr>
>+  any(T&& value) : obj_(new Obj<VT>(in_place, std::forward<T>(value))) {}
>+
>+  // Constructs an `absl::any` object with a "contained object" of the decayed
>+  // type of `T`, which is initialized via `std::forward<T>(value)`.
>+  template <typename T, typename... Args, typename VT = absl::decay_t<T>,
>+            absl::enable_if_t<absl::conjunction<
>+                std::is_copy_constructible<VT>,
>+                std::is_constructible<VT, Args...>>::value>* = nullptr>
>+  explicit any(in_place_type_t<T> /*tag*/, Args&&... args)
>+      : obj_(new Obj<VT>(in_place, std::forward<Args>(args)...)) {}
>+
>+  // Constructs an `absl::any` object with a "contained object" of the passed
>+  // type `VT` as a decayed type of `T`. `VT` is initialized as if
>+  // direct-non-list-initializing an object of type `VT` with the arguments
>+  // `initializer_list, std::forward<Args>(args)...`.
>+  template <
>+      typename T, typename U, typename... Args, typename VT = absl::decay_t<T>,
>+      absl::enable_if_t<
>+          absl::conjunction<std::is_copy_constructible<VT>,
>+                            std::is_constructible<VT, std::initializer_list<U>&,
>+                                                  Args...>>::value>* = nullptr>
>+  explicit any(in_place_type_t<T> /*tag*/, std::initializer_list<U> ilist,
>+               Args&&... args)
>+      : obj_(new Obj<VT>(in_place, ilist, std::forward<Args>(args)...)) {}
>+
>+  // Assignment operators
>+
>+  // Copy assigns an `absl::any` object with a "contained object" of the
>+  // passed type.
>+  any& operator=(const any& rhs) {
>+    any(rhs).swap(*this);
>+    return *this;
>+  }
>+
>+  // Move assigns an `absl::any` object with a "contained object" of the
>+  // passed type. `rhs` is left in a valid but otherwise unspecified state.
>+  any& operator=(any&& rhs) noexcept {
>+    any(std::move(rhs)).swap(*this);
>+    return *this;
>+  }
>+
>+  // Assigns an `absl::any` object with a "contained object" of the passed type.
>+  template <typename T, typename VT = absl::decay_t<T>,
>+            absl::enable_if_t<absl::conjunction<
>+                absl::negation<std::is_same<VT, any>>,
>+                std::is_copy_constructible<VT>>::value>* = nullptr>
>+  any& operator=(T&& rhs) {
>+    any tmp(in_place_type_t<VT>(), std::forward<T>(rhs));
>+    tmp.swap(*this);
>+    return *this;
>+  }
>+
>+  // Modifiers
>+
>+  // any::emplace()
>+  //
>+  // Emplaces a value within an `absl::any` object by calling `any::reset()`,
>+  // initializing the contained value as if direct-non-list-initializing an
>+  // object of type `VT` with the arguments `std::forward<Args>(args)...`, and
>+  // returning a reference to the new contained value.
>+  //
>+  // Note: If an exception is thrown during the call to `VT`'s constructor,
>+  // `*this` does not contain a value, and any previously contained value has
>+  // been destroyed.
>+  template <
>+      typename T, typename... Args, typename VT = absl::decay_t<T>,
>+      absl::enable_if_t<std::is_copy_constructible<VT>::value &&
>+                        std::is_constructible<VT, Args...>::value>* = nullptr>
>+  VT& emplace(Args&&... args) {
>+    reset();  // NOTE: reset() is required here even in the world of exceptions.
>+    Obj<VT>* const object_ptr =
>+        new Obj<VT>(in_place, std::forward<Args>(args)...);
>+    obj_ = std::unique_ptr<ObjInterface>(object_ptr);
>+    return object_ptr->value;
>+  }
>+
>+  // Overload of `any::emplace()` to emplace a value within an `absl::any`
>+  // object by calling `any::reset()`, initializing the contained value as if
>+  // direct-non-list-initializing an object of type `VT` with the arguments
>+  // `initializer_list, std::forward<Args>(args)...`, and returning a reference
>+  // to the new contained value.
>+  //
>+  // Note: If an exception is thrown during the call to `VT`'s constructor,
>+  // `*this` does not contain a value, and any previously contained value has
>+  // been destroyed. The function shall not participate in overload resolution
>+  // unless `is_copy_constructible_v<VT>` is `true` and
>+  // `is_constructible_v<VT, initializer_list<U>&, Args...>` is `true`.
>+  template <
>+      typename T, typename U, typename... Args, typename VT = absl::decay_t<T>,
>+      absl::enable_if_t<std::is_copy_constructible<VT>::value &&
>+                        std::is_constructible<VT, std::initializer_list<U>&,
>+                                              Args...>::value>* = nullptr>
>+  VT& emplace(std::initializer_list<U> ilist, Args&&... args) {
>+    reset();  // NOTE: reset() is required here even in the world of exceptions.
>+    Obj<VT>* const object_ptr =
>+        new Obj<VT>(in_place, ilist, std::forward<Args>(args)...);
>+    obj_ = std::unique_ptr<ObjInterface>(object_ptr);
>+    return object_ptr->value;
>+  }
>+
>+  // any::reset()
>+  //
>+  // Resets the state of the `absl::any` object, destroying the contained object
>+  // if present.
>+  void reset() noexcept { obj_ = nullptr; }
>+
>+  // any::swap()
>+  //
>+  // Swaps the passed value and the value of this `absl::any` object.
>+  void swap(any& other) noexcept { obj_.swap(other.obj_); }
>+
>+  // Observers
>+
>+  // any::has_value()
>+  //
>+  // Returns `true` if the `any` object has a contained value, otherwise
>+  // returns `false`.
>+  bool has_value() const noexcept { return obj_ != nullptr; }
>+
>+#if ABSL_ANY_DETAIL_HAS_RTTI
>+  // Returns: typeid(T) if *this has a contained object of type T, otherwise
>+  // typeid(void).
>+  const std::type_info& type() const noexcept {
>+    if (has_value()) {
>+      return obj_->Type();
>+    }
>+
>+    return typeid(void);
>+  }
>+#endif  // ABSL_ANY_DETAIL_HAS_RTTI
>+
>+ private:
>+  // Tagged type-erased abstraction for holding a cloneable object.
>+  class ObjInterface {
>+   public:
>+    virtual ~ObjInterface() = default;
>+    virtual std::unique_ptr<ObjInterface> Clone() const = 0;
>+    virtual const void* ObjTypeId() const noexcept = 0;
>+#if ABSL_ANY_DETAIL_HAS_RTTI
>+    virtual const std::type_info& Type() const noexcept = 0;
>+#endif  // ABSL_ANY_DETAIL_HAS_RTTI
>+  };
>+
>+  // Hold a value of some queryable type, with an ability to Clone it.
>+  template <typename T>
>+  class Obj : public ObjInterface {
>+   public:
>+    template <typename... Args>
>+    explicit Obj(in_place_t /*tag*/, Args&&... args)
>+        : value(std::forward<Args>(args)...) {}
>+
>+    std::unique_ptr<ObjInterface> Clone() const final {
>+      return std::unique_ptr<ObjInterface>(new Obj(in_place, value));
>+    }
>+
>+    const void* ObjTypeId() const noexcept final { return IdForType<T>(); }
>+
>+#if ABSL_ANY_DETAIL_HAS_RTTI
>+    const std::type_info& Type() const noexcept final { return typeid(T); }
>+#endif  // ABSL_ANY_DETAIL_HAS_RTTI
>+
>+    T value;
>+  };
>+
>+  std::unique_ptr<ObjInterface> CloneObj() const {
>+    if (!obj_) return nullptr;
>+    return obj_->Clone();
>+  }
>+
>+  template <typename T>
>+  constexpr static const void* IdForType() {
>+    // Note: This type dance is to make the behavior consistent with typeid.
>+    using NormalizedType =
>+        typename std::remove_cv<typename std::remove_reference<T>::type>::type;
>+
>+    return any_internal::FastTypeId<NormalizedType>();
>+  }
>+
>+  const void* GetObjTypeId() const {
>+    return obj_ ? obj_->ObjTypeId() : any_internal::FastTypeId<void>();
>+  }
>+
>+  // `absl::any` nonmember functions //
>+
>+  // Description at the declaration site (top of file).
>+  template <typename ValueType>
>+  friend ValueType any_cast(const any& operand);
>+
>+  // Description at the declaration site (top of file).
>+  template <typename ValueType>
>+  friend ValueType any_cast(any& operand);  // NOLINT(runtime/references)
>+
>+  // Description at the declaration site (top of file).
>+  template <typename T>
>+  friend const T* any_cast(const any* operand) noexcept;
>+
>+  // Description at the declaration site (top of file).
>+  template <typename T>
>+  friend T* any_cast(any* operand) noexcept;
>+
>+  std::unique_ptr<ObjInterface> obj_;
>+};
>+
>+// -----------------------------------------------------------------------------
>+// Implementation Details
>+// -----------------------------------------------------------------------------
>+
>+constexpr any::any() noexcept = default;
>+
>+template <typename T>
>+struct any::IsInPlaceType : std::false_type {};
>+
>+template <typename T>
>+struct any::IsInPlaceType<in_place_type_t<T>> : std::true_type {};
>+
>+inline void swap(any& x, any& y) noexcept { x.swap(y); }
>+
>+// Description at the declaration site (top of file).
>+template <typename T, typename... Args>
>+any make_any(Args&&... args) {
>+  return any(in_place_type_t<T>(), std::forward<Args>(args)...);
>+}
>+
>+// Description at the declaration site (top of file).
>+template <typename T, typename U, typename... Args>
>+any make_any(std::initializer_list<U> il, Args&&... args) {
>+  return any(in_place_type_t<T>(), il, std::forward<Args>(args)...);
>+}
>+
>+// Description at the declaration site (top of file).
>+template <typename ValueType>
>+ValueType any_cast(const any& operand) {
>+  using U = typename std::remove_cv<
>+      typename std::remove_reference<ValueType>::type>::type;
>+  static_assert(std::is_constructible<ValueType, const U&>::value,
>+                "Invalid ValueType");
>+  auto* const result = (any_cast<U>)(&operand);
>+  if (result == nullptr) {
>+    any_internal::ThrowBadAnyCast();
>+  }
>+  return static_cast<ValueType>(*result);
>+}
>+
>+// Description at the declaration site (top of file).
>+template <typename ValueType>
>+ValueType any_cast(any& operand) {  // NOLINT(runtime/references)
>+  using U = typename std::remove_cv<
>+      typename std::remove_reference<ValueType>::type>::type;
>+  static_assert(std::is_constructible<ValueType, U&>::value,
>+                "Invalid ValueType");
>+  auto* result = (any_cast<U>)(&operand);
>+  if (result == nullptr) {
>+    any_internal::ThrowBadAnyCast();
>+  }
>+  return static_cast<ValueType>(*result);
>+}
>+
>+// Description at the declaration site (top of file).
>+template <typename ValueType>
>+ValueType any_cast(any&& operand) {
>+  using U = typename std::remove_cv<
>+      typename std::remove_reference<ValueType>::type>::type;
>+  static_assert(std::is_constructible<ValueType, U>::value,
>+                "Invalid ValueType");
>+  return static_cast<ValueType>(std::move((any_cast<U&>)(operand)));
>+}
>+
>+// Description at the declaration site (top of file).
>+template <typename T>
>+const T* any_cast(const any* operand) noexcept {
>+  return operand && operand->GetObjTypeId() == any::IdForType<T>()
>+             ? std::addressof(
>+                   static_cast<const any::Obj<T>*>(operand->obj_.get())->value)
>+             : nullptr;
>+}
>+
>+// Description at the declaration site (top of file).
>+template <typename T>
>+T* any_cast(any* operand) noexcept {
>+  return operand && operand->GetObjTypeId() == any::IdForType<T>()
>+             ? std::addressof(
>+                   static_cast<any::Obj<T>*>(operand->obj_.get())->value)
>+             : nullptr;
>+}
>+
>+}  // namespace absl
>+
>+#undef ABSL_ANY_DETAIL_HAS_RTTI
>+
>+#endif  // ABSL_HAVE_STD_ANY
>+
>+#endif  // ABSL_TYPES_ANY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/any_exception_safety_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/any_exception_safety_test.cc
>new file mode 100644
>index 00000000000..36955f6c5c3
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/any_exception_safety_test.cc
>@@ -0,0 +1,167 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/types/any.h"
>+
>+#include <typeinfo>
>+#include <vector>
>+
>+#include "gtest/gtest.h"
>+#include "absl/base/internal/exception_safety_testing.h"
>+
>+using Thrower = testing::ThrowingValue<>;
>+using NoThrowMoveThrower =
>+    testing::ThrowingValue<testing::TypeSpec::kNoThrowMove>;
>+using ThrowerList = std::initializer_list<Thrower>;
>+using ThrowerVec = std::vector<Thrower>;
>+using ThrowingAlloc = testing::ThrowingAllocator<Thrower>;
>+using ThrowingThrowerVec = std::vector<Thrower, ThrowingAlloc>;
>+
>+namespace {
>+
>+testing::AssertionResult AnyInvariants(absl::any* a) {
>+  using testing::AssertionFailure;
>+  using testing::AssertionSuccess;
>+
>+  if (a->has_value()) {
>+    if (a->type() == typeid(void)) {
>+      return AssertionFailure()
>+             << "A non-empty any should not have type `void`";
>+    }
>+  } else {
>+    if (a->type() != typeid(void)) {
>+      return AssertionFailure()
>+             << "An empty any should have type void, but has type "
>+             << a->type().name();
>+    }
>+  }
>+
>+  //  Make sure that reset() changes any to a valid state.
>+  a->reset();
>+  if (a->has_value()) {
>+    return AssertionFailure() << "A reset `any` should be valueless";
>+  }
>+  if (a->type() != typeid(void)) {
>+    return AssertionFailure() << "A reset `any` should have type() of `void`, "
>+                                 "but instead has type "
>+                              << a->type().name();
>+  }
>+  try {
>+    auto unused = absl::any_cast<Thrower>(*a);
>+    static_cast<void>(unused);
>+    return AssertionFailure()
>+           << "A reset `any` should not be able to be any_cast";
>+  } catch (absl::bad_any_cast) {
>+  } catch (...) {
>+    return AssertionFailure()
>+           << "Unexpected exception thrown from absl::any_cast";
>+  }
>+  return AssertionSuccess();
>+}
>+
>+testing::AssertionResult AnyIsEmpty(absl::any* a) {
>+  if (!a->has_value()) {
>+    return testing::AssertionSuccess();
>+  }
>+  return testing::AssertionFailure()
>+         << "a should be empty, but instead has value "
>+         << absl::any_cast<Thrower>(*a).Get();
>+}
>+
>+TEST(AnyExceptionSafety, Ctors) {
>+  Thrower val(1);
>+  testing::TestThrowingCtor<absl::any>(val);
>+
>+  Thrower copy(val);
>+  testing::TestThrowingCtor<absl::any>(copy);
>+
>+  testing::TestThrowingCtor<absl::any>(absl::in_place_type_t<Thrower>(), 1);
>+
>+  testing::TestThrowingCtor<absl::any>(absl::in_place_type_t<ThrowerVec>(),
>+                                       ThrowerList{val});
>+
>+  testing::TestThrowingCtor<absl::any,
>+                            absl::in_place_type_t<ThrowingThrowerVec>,
>+                            ThrowerList, ThrowingAlloc>(
>+      absl::in_place_type_t<ThrowingThrowerVec>(), {val}, ThrowingAlloc());
>+}
>+
>+TEST(AnyExceptionSafety, Assignment) {
>+  auto original =
>+      absl::any(absl::in_place_type_t<Thrower>(), 1, testing::nothrow_ctor);
>+  auto any_is_strong = [original](absl::any* ap) {
>+    return testing::AssertionResult(ap->has_value() &&
>+                                    absl::any_cast<Thrower>(original) ==
>+                                        absl::any_cast<Thrower>(*ap));
>+  };
>+  auto any_strong_tester = testing::MakeExceptionSafetyTester()
>+                               .WithInitialValue(original)
>+                               .WithInvariants(AnyInvariants, any_is_strong);
>+
>+  Thrower val(2);
>+  absl::any any_val(val);
>+  NoThrowMoveThrower mv_val(2);
>+
>+  auto assign_any = [&any_val](absl::any* ap) { *ap = any_val; };
>+  auto assign_val = [&val](absl::any* ap) { *ap = val; };
>+  auto move = [&val](absl::any* ap) { *ap = std::move(val); };
>+  auto move_movable = [&mv_val](absl::any* ap) { *ap = std::move(mv_val); };
>+
>+  EXPECT_TRUE(any_strong_tester.Test(assign_any));
>+  EXPECT_TRUE(any_strong_tester.Test(assign_val));
>+  EXPECT_TRUE(any_strong_tester.Test(move));
>+  EXPECT_TRUE(any_strong_tester.Test(move_movable));
>+
>+  auto empty_any_is_strong = [](absl::any* ap) {
>+    return testing::AssertionResult{!ap->has_value()};
>+  };
>+  auto strong_empty_any_tester =
>+      testing::MakeExceptionSafetyTester()
>+          .WithInitialValue(absl::any{})
>+          .WithInvariants(AnyInvariants, empty_any_is_strong);
>+
>+  EXPECT_TRUE(strong_empty_any_tester.Test(assign_any));
>+  EXPECT_TRUE(strong_empty_any_tester.Test(assign_val));
>+  EXPECT_TRUE(strong_empty_any_tester.Test(move));
>+}
>+// libstdc++ std::any fails this test
>+#if !defined(ABSL_HAVE_STD_ANY)
>+TEST(AnyExceptionSafety, Emplace) {
>+  auto initial_val =
>+      absl::any{absl::in_place_type_t<Thrower>(), 1, testing::nothrow_ctor};
>+  auto one_tester = testing::MakeExceptionSafetyTester()
>+                        .WithInitialValue(initial_val)
>+                        .WithInvariants(AnyInvariants, AnyIsEmpty);
>+
>+  auto emp_thrower = [](absl::any* ap) { ap->emplace<Thrower>(2); };
>+  auto emp_throwervec = [](absl::any* ap) {
>+    std::initializer_list<Thrower> il{Thrower(2, testing::nothrow_ctor)};
>+    ap->emplace<ThrowerVec>(il);
>+  };
>+  auto emp_movethrower = [](absl::any* ap) {
>+    ap->emplace<NoThrowMoveThrower>(2);
>+  };
>+
>+  EXPECT_TRUE(one_tester.Test(emp_thrower));
>+  EXPECT_TRUE(one_tester.Test(emp_throwervec));
>+  EXPECT_TRUE(one_tester.Test(emp_movethrower));
>+
>+  auto empty_tester = one_tester.WithInitialValue(absl::any{});
>+
>+  EXPECT_TRUE(empty_tester.Test(emp_thrower));
>+  EXPECT_TRUE(empty_tester.Test(emp_throwervec));
>+}
>+#endif  // ABSL_HAVE_STD_ANY
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/any_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/any_test.cc
>new file mode 100644
>index 00000000000..115e78df7af
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/any_test.cc
>@@ -0,0 +1,724 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/types/any.h"
>+
>+#include <initializer_list>
>+#include <type_traits>
>+#include <utility>
>+#include <vector>
>+
>+#include "gtest/gtest.h"
>+#include "absl/base/config.h"
>+#include "absl/base/internal/exception_testing.h"
>+#include "absl/base/internal/raw_logging.h"
>+#include "absl/container/internal/test_instance_tracker.h"
>+
>+namespace {
>+using absl::test_internal::CopyableOnlyInstance;
>+using absl::test_internal::InstanceTracker;
>+
>+template <typename T>
>+const T& AsConst(const T& t) {
>+  return t;
>+}
>+
>+struct MoveOnly {
>+  MoveOnly() = default;
>+  explicit MoveOnly(int value) : value(value) {}
>+  MoveOnly(MoveOnly&&) = default;
>+  MoveOnly& operator=(MoveOnly&&) = default;
>+
>+  int value = 0;
>+};
>+
>+struct CopyOnly {
>+  CopyOnly() = default;
>+  explicit CopyOnly(int value) : value(value) {}
>+  CopyOnly(CopyOnly&&) = delete;
>+  CopyOnly& operator=(CopyOnly&&) = delete;
>+  CopyOnly(const CopyOnly&) = default;
>+  CopyOnly& operator=(const CopyOnly&) = default;
>+
>+  int value = 0;
>+};
>+
>+struct MoveOnlyWithListConstructor {
>+  MoveOnlyWithListConstructor() = default;
>+  explicit MoveOnlyWithListConstructor(std::initializer_list<int> /*ilist*/,
>+                                       int value)
>+      : value(value) {}
>+  MoveOnlyWithListConstructor(MoveOnlyWithListConstructor&&) = default;
>+  MoveOnlyWithListConstructor& operator=(MoveOnlyWithListConstructor&&) =
>+      default;
>+
>+  int value = 0;
>+};
>+
>+struct IntMoveOnlyCopyOnly {
>+  IntMoveOnlyCopyOnly(int value, MoveOnly /*move_only*/, CopyOnly /*copy_only*/)
>+      : value(value) {}
>+
>+  int value;
>+};
>+
>+struct ListMoveOnlyCopyOnly {
>+  ListMoveOnlyCopyOnly(std::initializer_list<int> ilist, MoveOnly /*move_only*/,
>+                       CopyOnly /*copy_only*/)
>+      : values(ilist) {}
>+
>+  std::vector<int> values;
>+};
>+
>+using FunctionType = void();
>+void FunctionToEmplace() {}
>+
>+using ArrayType = int[2];
>+using DecayedArray = absl::decay_t<ArrayType>;
>+
>+TEST(AnyTest, Noexcept) {
>+  static_assert(std::is_nothrow_default_constructible<absl::any>(), "");
>+  static_assert(std::is_nothrow_move_constructible<absl::any>(), "");
>+  static_assert(std::is_nothrow_move_assignable<absl::any>(), "");
>+  static_assert(noexcept(std::declval<absl::any&>().has_value()), "");
>+  static_assert(noexcept(std::declval<absl::any&>().type()), "");
>+  static_assert(noexcept(absl::any_cast<int>(std::declval<absl::any*>())), "");
>+  static_assert(
>+      noexcept(std::declval<absl::any&>().swap(std::declval<absl::any&>())),
>+      "");
>+
>+  using std::swap;
>+  static_assert(
>+      noexcept(swap(std::declval<absl::any&>(), std::declval<absl::any&>())),
>+      "");
>+}
>+
>+TEST(AnyTest, HasValue) {
>+  absl::any o;
>+  EXPECT_FALSE(o.has_value());
>+  o.emplace<int>();
>+  EXPECT_TRUE(o.has_value());
>+  o.reset();
>+  EXPECT_FALSE(o.has_value());
>+}
>+
>+TEST(AnyTest, Type) {
>+  absl::any o;
>+  EXPECT_EQ(typeid(void), o.type());
>+  o.emplace<int>(5);
>+  EXPECT_EQ(typeid(int), o.type());
>+  o.emplace<float>(5.f);
>+  EXPECT_EQ(typeid(float), o.type());
>+  o.reset();
>+  EXPECT_EQ(typeid(void), o.type());
>+}
>+
>+TEST(AnyTest, EmptyPointerCast) {
>+  // pointer-to-unqualified overload
>+  {
>+    absl::any o;
>+    EXPECT_EQ(nullptr, absl::any_cast<int>(&o));
>+    o.emplace<int>();
>+    EXPECT_NE(nullptr, absl::any_cast<int>(&o));
>+    o.reset();
>+    EXPECT_EQ(nullptr, absl::any_cast<int>(&o));
>+  }
>+
>+  // pointer-to-const overload
>+  {
>+    absl::any o;
>+    EXPECT_EQ(nullptr, absl::any_cast<int>(&AsConst(o)));
>+    o.emplace<int>();
>+    EXPECT_NE(nullptr, absl::any_cast<int>(&AsConst(o)));
>+    o.reset();
>+    EXPECT_EQ(nullptr, absl::any_cast<int>(&AsConst(o)));
>+  }
>+}
>+
>+TEST(AnyTest, InPlaceConstruction) {
>+  const CopyOnly copy_only{};
>+  absl::any o(absl::in_place_type_t<IntMoveOnlyCopyOnly>(), 5, MoveOnly(),
>+              copy_only);
>+  IntMoveOnlyCopyOnly& v = absl::any_cast<IntMoveOnlyCopyOnly&>(o);
>+  EXPECT_EQ(5, v.value);
>+}
>+
>+TEST(AnyTest, InPlaceConstructionWithCV) {
>+  const CopyOnly copy_only{};
>+  absl::any o(absl::in_place_type_t<const volatile IntMoveOnlyCopyOnly>(), 5,
>+              MoveOnly(), copy_only);
>+  IntMoveOnlyCopyOnly& v = absl::any_cast<IntMoveOnlyCopyOnly&>(o);
>+  EXPECT_EQ(5, v.value);
>+}
>+
>+TEST(AnyTest, InPlaceConstructionWithFunction) {
>+  absl::any o(absl::in_place_type_t<FunctionType>(), FunctionToEmplace);
>+  FunctionType*& construction_result = absl::any_cast<FunctionType*&>(o);
>+  EXPECT_EQ(&FunctionToEmplace, construction_result);
>+}
>+
>+TEST(AnyTest, InPlaceConstructionWithArray) {
>+  ArrayType ar = {5, 42};
>+  absl::any o(absl::in_place_type_t<ArrayType>(), ar);
>+  DecayedArray& construction_result = absl::any_cast<DecayedArray&>(o);
>+  EXPECT_EQ(&ar[0], construction_result);
>+}
>+
>+TEST(AnyTest, InPlaceConstructionIlist) {
>+  const CopyOnly copy_only{};
>+  absl::any o(absl::in_place_type_t<ListMoveOnlyCopyOnly>(), {1, 2, 3, 4},
>+              MoveOnly(), copy_only);
>+  ListMoveOnlyCopyOnly& v = absl::any_cast<ListMoveOnlyCopyOnly&>(o);
>+  std::vector<int> expected_values = {1, 2, 3, 4};
>+  EXPECT_EQ(expected_values, v.values);
>+}
>+
>+TEST(AnyTest, InPlaceConstructionIlistWithCV) {
>+  const CopyOnly copy_only{};
>+  absl::any o(absl::in_place_type_t<const volatile ListMoveOnlyCopyOnly>(),
>+              {1, 2, 3, 4}, MoveOnly(), copy_only);
>+  ListMoveOnlyCopyOnly& v = absl::any_cast<ListMoveOnlyCopyOnly&>(o);
>+  std::vector<int> expected_values = {1, 2, 3, 4};
>+  EXPECT_EQ(expected_values, v.values);
>+}
>+
>+TEST(AnyTest, InPlaceNoArgs) {
>+  absl::any o(absl::in_place_type_t<int>{});
>+  EXPECT_EQ(0, absl::any_cast<int&>(o));
>+}
>+
>+template <typename Enabler, typename T, typename... Args>
>+struct CanEmplaceAnyImpl : std::false_type {};
>+
>+template <typename T, typename... Args>
>+struct CanEmplaceAnyImpl<
>+    absl::void_t<decltype(
>+        std::declval<absl::any&>().emplace<T>(std::declval<Args>()...))>,
>+    T, Args...> : std::true_type {};
>+
>+template <typename T, typename... Args>
>+using CanEmplaceAny = CanEmplaceAnyImpl<void, T, Args...>;
>+
>+TEST(AnyTest, Emplace) {
>+  const CopyOnly copy_only{};
>+  absl::any o;
>+  EXPECT_TRUE((std::is_same<decltype(o.emplace<IntMoveOnlyCopyOnly>(
>+                                5, MoveOnly(), copy_only)),
>+                            IntMoveOnlyCopyOnly&>::value));
>+  IntMoveOnlyCopyOnly& emplace_result =
>+      o.emplace<IntMoveOnlyCopyOnly>(5, MoveOnly(), copy_only);
>+  EXPECT_EQ(5, emplace_result.value);
>+  IntMoveOnlyCopyOnly& v = absl::any_cast<IntMoveOnlyCopyOnly&>(o);
>+  EXPECT_EQ(5, v.value);
>+  EXPECT_EQ(&emplace_result, &v);
>+
>+  static_assert(!CanEmplaceAny<int, int, int>::value, "");
>+  static_assert(!CanEmplaceAny<MoveOnly, MoveOnly>::value, "");
>+}
>+
>+TEST(AnyTest, EmplaceWithCV) {
>+  const CopyOnly copy_only{};
>+  absl::any o;
>+  EXPECT_TRUE(
>+      (std::is_same<decltype(o.emplace<const volatile IntMoveOnlyCopyOnly>(
>+                        5, MoveOnly(), copy_only)),
>+                    IntMoveOnlyCopyOnly&>::value));
>+  IntMoveOnlyCopyOnly& emplace_result =
>+      o.emplace<const volatile IntMoveOnlyCopyOnly>(5, MoveOnly(), copy_only);
>+  EXPECT_EQ(5, emplace_result.value);
>+  IntMoveOnlyCopyOnly& v = absl::any_cast<IntMoveOnlyCopyOnly&>(o);
>+  EXPECT_EQ(5, v.value);
>+  EXPECT_EQ(&emplace_result, &v);
>+}
>+
>+TEST(AnyTest, EmplaceWithFunction) {
>+  absl::any o;
>+  EXPECT_TRUE(
>+      (std::is_same<decltype(o.emplace<FunctionType>(FunctionToEmplace)),
>+                    FunctionType*&>::value));
>+  FunctionType*& emplace_result = o.emplace<FunctionType>(FunctionToEmplace);
>+  EXPECT_EQ(&FunctionToEmplace, emplace_result);
>+}
>+
>+TEST(AnyTest, EmplaceWithArray) {
>+  absl::any o;
>+  ArrayType ar = {5, 42};
>+  EXPECT_TRUE(
>+      (std::is_same<decltype(o.emplace<ArrayType>(ar)), DecayedArray&>::value));
>+  DecayedArray& emplace_result = o.emplace<ArrayType>(ar);
>+  EXPECT_EQ(&ar[0], emplace_result);
>+}
>+
>+TEST(AnyTest, EmplaceIlist) {
>+  const CopyOnly copy_only{};
>+  absl::any o;
>+  EXPECT_TRUE((std::is_same<decltype(o.emplace<ListMoveOnlyCopyOnly>(
>+                                {1, 2, 3, 4}, MoveOnly(), copy_only)),
>+                            ListMoveOnlyCopyOnly&>::value));
>+  ListMoveOnlyCopyOnly& emplace_result =
>+      o.emplace<ListMoveOnlyCopyOnly>({1, 2, 3, 4}, MoveOnly(), copy_only);
>+  ListMoveOnlyCopyOnly& v = absl::any_cast<ListMoveOnlyCopyOnly&>(o);
>+  EXPECT_EQ(&v, &emplace_result);
>+  std::vector<int> expected_values = {1, 2, 3, 4};
>+  EXPECT_EQ(expected_values, v.values);
>+
>+  static_assert(!CanEmplaceAny<int, std::initializer_list<int>>::value, "");
>+  static_assert(!CanEmplaceAny<MoveOnlyWithListConstructor,
>+                               std::initializer_list<int>, int>::value,
>+                "");
>+}
>+
>+TEST(AnyTest, EmplaceIlistWithCV) {
>+  const CopyOnly copy_only{};
>+  absl::any o;
>+  EXPECT_TRUE(
>+      (std::is_same<decltype(o.emplace<const volatile ListMoveOnlyCopyOnly>(
>+                        {1, 2, 3, 4}, MoveOnly(), copy_only)),
>+                    ListMoveOnlyCopyOnly&>::value));
>+  ListMoveOnlyCopyOnly& emplace_result =
>+      o.emplace<const volatile ListMoveOnlyCopyOnly>({1, 2, 3, 4}, MoveOnly(),
>+                                                     copy_only);
>+  ListMoveOnlyCopyOnly& v = absl::any_cast<ListMoveOnlyCopyOnly&>(o);
>+  EXPECT_EQ(&v, &emplace_result);
>+  std::vector<int> expected_values = {1, 2, 3, 4};
>+  EXPECT_EQ(expected_values, v.values);
>+}
>+
>+TEST(AnyTest, EmplaceNoArgs) {
>+  absl::any o;
>+  o.emplace<int>();
>+  EXPECT_EQ(0, absl::any_cast<int>(o));
>+}
>+
>+TEST(AnyTest, ConversionConstruction) {
>+  {
>+    absl::any o = 5;
>+    EXPECT_EQ(5, absl::any_cast<int>(o));
>+  }
>+
>+  {
>+    const CopyOnly copy_only(5);
>+    absl::any o = copy_only;
>+    EXPECT_EQ(5, absl::any_cast<CopyOnly&>(o).value);
>+  }
>+
>+  static_assert(!std::is_convertible<MoveOnly, absl::any>::value, "");
>+}
>+
>+TEST(AnyTest, ConversionAssignment) {
>+  {
>+    absl::any o;
>+    o = 5;
>+    EXPECT_EQ(5, absl::any_cast<int>(o));
>+  }
>+
>+  {
>+    const CopyOnly copy_only(5);
>+    absl::any o;
>+    o = copy_only;
>+    EXPECT_EQ(5, absl::any_cast<CopyOnly&>(o).value);
>+  }
>+
>+  static_assert(!std::is_assignable<MoveOnly, absl::any>::value, "");
>+}
>+
>+// Suppress MSVC warnings.
>+// 4521: multiple copy constructors specified
>+// We wrote multiple of them to test that the correct overloads are selected.
>+#ifdef _MSC_VER
>+#pragma warning( push )
>+#pragma warning( disable : 4521)
>+#endif
>+
>+// Weird type for testing, only used to make sure we "properly" perfect-forward
>+// when being placed into an absl::any (use the l-value constructor if given an
>+// l-value rather than use the copy constructor).
>+struct WeirdConstructor42 {
>+  explicit WeirdConstructor42(int value) : value(value) {}
>+
>+  // Copy-constructor
>+  WeirdConstructor42(const WeirdConstructor42& other) : value(other.value) {}
>+
>+  // L-value "weird" constructor (used when given an l-value)
>+  WeirdConstructor42(
>+      WeirdConstructor42& /*other*/)  // NOLINT(runtime/references)
>+      : value(42) {}
>+
>+  int value;
>+};
>+#ifdef _MSC_VER
>+#pragma warning( pop )
>+#endif
>+
>+TEST(AnyTest, WeirdConversionConstruction) {
>+  {
>+    const WeirdConstructor42 source(5);
>+    absl::any o = source;  // Actual copy
>+    EXPECT_EQ(5, absl::any_cast<WeirdConstructor42&>(o).value);
>+  }
>+
>+  {
>+    WeirdConstructor42 source(5);
>+    absl::any o = source;  // Weird "conversion"
>+    EXPECT_EQ(42, absl::any_cast<WeirdConstructor42&>(o).value);
>+  }
>+}
>+
>+TEST(AnyTest, WeirdConversionAssignment) {
>+  {
>+    const WeirdConstructor42 source(5);
>+    absl::any o;
>+    o = source;  // Actual copy
>+    EXPECT_EQ(5, absl::any_cast<WeirdConstructor42&>(o).value);
>+  }
>+
>+  {
>+    WeirdConstructor42 source(5);
>+    absl::any o;
>+    o = source;  // Weird "conversion"
>+    EXPECT_EQ(42, absl::any_cast<WeirdConstructor42&>(o).value);
>+  }
>+}
>+
>+struct Value {};
>+
>+TEST(AnyTest, AnyCastValue) {
>+  {
>+    absl::any o;
>+    o.emplace<int>(5);
>+    EXPECT_EQ(5, absl::any_cast<int>(o));
>+    EXPECT_EQ(5, absl::any_cast<int>(AsConst(o)));
>+    static_assert(
>+        std::is_same<decltype(absl::any_cast<Value>(o)), Value>::value, "");
>+  }
>+
>+  {
>+    absl::any o;
>+    o.emplace<int>(5);
>+    EXPECT_EQ(5, absl::any_cast<const int>(o));
>+    EXPECT_EQ(5, absl::any_cast<const int>(AsConst(o)));
>+    static_assert(std::is_same<decltype(absl::any_cast<const Value>(o)),
>+                               const Value>::value,
>+                  "");
>+  }
>+}
>+
>+TEST(AnyTest, AnyCastReference) {
>+  {
>+    absl::any o;
>+    o.emplace<int>(5);
>+    EXPECT_EQ(5, absl::any_cast<int&>(o));
>+    EXPECT_EQ(5, absl::any_cast<const int&>(AsConst(o)));
>+    static_assert(
>+        std::is_same<decltype(absl::any_cast<Value&>(o)), Value&>::value, "");
>+  }
>+
>+  {
>+    absl::any o;
>+    o.emplace<int>(5);
>+    EXPECT_EQ(5, absl::any_cast<const int>(o));
>+    EXPECT_EQ(5, absl::any_cast<const int>(AsConst(o)));
>+    static_assert(std::is_same<decltype(absl::any_cast<const Value&>(o)),
>+                               const Value&>::value,
>+                  "");
>+  }
>+
>+  {
>+    absl::any o;
>+    o.emplace<int>(5);
>+    EXPECT_EQ(5, absl::any_cast<int&&>(std::move(o)));
>+    static_assert(std::is_same<decltype(absl::any_cast<Value&&>(std::move(o))),
>+                               Value&&>::value,
>+                  "");
>+  }
>+
>+  {
>+    absl::any o;
>+    o.emplace<int>(5);
>+    EXPECT_EQ(5, absl::any_cast<const int>(std::move(o)));
>+    static_assert(
>+        std::is_same<decltype(absl::any_cast<const Value&&>(std::move(o))),
>+                     const Value&&>::value,
>+        "");
>+  }
>+}
>+
>+TEST(AnyTest, AnyCastPointer) {
>+  {
>+    absl::any o;
>+    EXPECT_EQ(nullptr, absl::any_cast<char>(&o));
>+    o.emplace<int>(5);
>+    EXPECT_EQ(nullptr, absl::any_cast<char>(&o));
>+    o.emplace<char>('a');
>+    EXPECT_EQ('a', *absl::any_cast<char>(&o));
>+    static_assert(
>+        std::is_same<decltype(absl::any_cast<Value>(&o)), Value*>::value, "");
>+  }
>+
>+  {
>+    absl::any o;
>+    EXPECT_EQ(nullptr, absl::any_cast<const char>(&o));
>+    o.emplace<int>(5);
>+    EXPECT_EQ(nullptr, absl::any_cast<const char>(&o));
>+    o.emplace<char>('a');
>+    EXPECT_EQ('a', *absl::any_cast<const char>(&o));
>+    static_assert(std::is_same<decltype(absl::any_cast<const Value>(&o)),
>+                               const Value*>::value,
>+                  "");
>+  }
>+}
>+
>+TEST(AnyTest, MakeAny) {
>+  const CopyOnly copy_only{};
>+  auto o = absl::make_any<IntMoveOnlyCopyOnly>(5, MoveOnly(), copy_only);
>+  static_assert(std::is_same<decltype(o), absl::any>::value, "");
>+  EXPECT_EQ(5, absl::any_cast<IntMoveOnlyCopyOnly&>(o).value);
>+}
>+
>+TEST(AnyTest, MakeAnyIList) {
>+  const CopyOnly copy_only{};
>+  auto o =
>+      absl::make_any<ListMoveOnlyCopyOnly>({1, 2, 3}, MoveOnly(), copy_only);
>+  static_assert(std::is_same<decltype(o), absl::any>::value, "");
>+  ListMoveOnlyCopyOnly& v = absl::any_cast<ListMoveOnlyCopyOnly&>(o);
>+  std::vector<int> expected_values = {1, 2, 3};
>+  EXPECT_EQ(expected_values, v.values);
>+}
>+
>+// Test the use of copy constructor and operator=
>+TEST(AnyTest, Copy) {
>+  InstanceTracker tracker_raii;
>+
>+  {
>+    absl::any o(absl::in_place_type_t<CopyableOnlyInstance>{}, 123);
>+    CopyableOnlyInstance* f1 = absl::any_cast<CopyableOnlyInstance>(&o);
>+
>+    absl::any o2(o);
>+    const CopyableOnlyInstance* f2 = absl::any_cast<CopyableOnlyInstance>(&o2);
>+    EXPECT_EQ(123, f2->value());
>+    EXPECT_NE(f1, f2);
>+
>+    absl::any o3;
>+    o3 = o2;
>+    const CopyableOnlyInstance* f3 = absl::any_cast<CopyableOnlyInstance>(&o3);
>+    EXPECT_EQ(123, f3->value());
>+    EXPECT_NE(f2, f3);
>+
>+    const absl::any o4(4);
>+    // copy construct from const lvalue ref.
>+    absl::any o5 = o4;
>+    EXPECT_EQ(4, absl::any_cast<int>(o4));
>+    EXPECT_EQ(4, absl::any_cast<int>(o5));
>+
>+    // Copy construct from const rvalue ref.
>+    absl::any o6 = std::move(o4);  // NOLINT
>+    EXPECT_EQ(4, absl::any_cast<int>(o4));
>+    EXPECT_EQ(4, absl::any_cast<int>(o6));
>+  }
>+}
>+
>+TEST(AnyTest, Move) {
>+  InstanceTracker tracker_raii;
>+
>+  absl::any any1;
>+  any1.emplace<CopyableOnlyInstance>(5);
>+
>+  // This is a copy, so copy count increases to 1.
>+  absl::any any2 = any1;
>+  EXPECT_EQ(5, absl::any_cast<CopyableOnlyInstance&>(any1).value());
>+  EXPECT_EQ(5, absl::any_cast<CopyableOnlyInstance&>(any2).value());
>+  EXPECT_EQ(1, tracker_raii.copies());
>+
>+  // This isn't a copy, so copy count doesn't increase.
>+  absl::any any3 = std::move(any2);
>+  EXPECT_EQ(5, absl::any_cast<CopyableOnlyInstance&>(any3).value());
>+  EXPECT_EQ(1, tracker_raii.copies());
>+
>+  absl::any any4;
>+  any4 = std::move(any3);
>+  EXPECT_EQ(5, absl::any_cast<CopyableOnlyInstance&>(any4).value());
>+  EXPECT_EQ(1, tracker_raii.copies());
>+
>+  absl::any tmp4(4);
>+  absl::any o4(std::move(tmp4));  // move construct
>+  EXPECT_EQ(4, absl::any_cast<int>(o4));
>+  o4 = *&o4;  // self assign
>+  EXPECT_EQ(4, absl::any_cast<int>(o4));
>+  EXPECT_TRUE(o4.has_value());
>+
>+  absl::any o5;
>+  absl::any tmp5(5);
>+  o5 = std::move(tmp5);  // move assign
>+  EXPECT_EQ(5, absl::any_cast<int>(o5));
>+}
>+
>+// Reset the ObjectOwner with an object of a different type
>+TEST(AnyTest, Reset) {
>+  absl::any o;
>+  o.emplace<int>();
>+
>+  o.reset();
>+  EXPECT_FALSE(o.has_value());
>+
>+  o.emplace<char>();
>+  EXPECT_TRUE(o.has_value());
>+}
>+
>+TEST(AnyTest, ConversionConstructionCausesOneCopy) {
>+  InstanceTracker tracker_raii;
>+  CopyableOnlyInstance counter(5);
>+  absl::any o(counter);
>+  EXPECT_EQ(5, absl::any_cast<CopyableOnlyInstance&>(o).value());
>+  EXPECT_EQ(1, tracker_raii.copies());
>+}
>+
>+//////////////////////////////////
>+// Tests for Exception Behavior //
>+//////////////////////////////////
>+
>+#if defined(ABSL_HAVE_STD_ANY)
>+
>+// If using a std `any` implementation, we can't check for a specific message.
>+#define ABSL_ANY_TEST_EXPECT_BAD_ANY_CAST(...)                      \
>+  ABSL_BASE_INTERNAL_EXPECT_FAIL((__VA_ARGS__), absl::bad_any_cast, \
>+                                 "")
>+
>+#else
>+
>+// If using the absl `any` implementation, we can rely on a specific message.
>+#define ABSL_ANY_TEST_EXPECT_BAD_ANY_CAST(...)                      \
>+  ABSL_BASE_INTERNAL_EXPECT_FAIL((__VA_ARGS__), absl::bad_any_cast, \
>+                                 "Bad any cast")
>+
>+#endif  // defined(ABSL_HAVE_STD_ANY)
>+
>+TEST(AnyTest, ThrowBadAlloc) {
>+  {
>+    absl::any a;
>+    ABSL_ANY_TEST_EXPECT_BAD_ANY_CAST(absl::any_cast<int&>(a));
>+    ABSL_ANY_TEST_EXPECT_BAD_ANY_CAST(absl::any_cast<const int&>(a));
>+    ABSL_ANY_TEST_EXPECT_BAD_ANY_CAST(absl::any_cast<int&&>(absl::any{}));
>+    ABSL_ANY_TEST_EXPECT_BAD_ANY_CAST(absl::any_cast<const int&&>(absl::any{}));
>+    ABSL_ANY_TEST_EXPECT_BAD_ANY_CAST(absl::any_cast<int>(a));
>+    ABSL_ANY_TEST_EXPECT_BAD_ANY_CAST(absl::any_cast<const int>(a));
>+    ABSL_ANY_TEST_EXPECT_BAD_ANY_CAST(absl::any_cast<int>(absl::any{}));
>+    ABSL_ANY_TEST_EXPECT_BAD_ANY_CAST(absl::any_cast<const int>(absl::any{}));
>+
>+    // const absl::any operand
>+    ABSL_ANY_TEST_EXPECT_BAD_ANY_CAST(absl::any_cast<const int&>(AsConst(a)));
>+    ABSL_ANY_TEST_EXPECT_BAD_ANY_CAST(absl::any_cast<int>(AsConst(a)));
>+    ABSL_ANY_TEST_EXPECT_BAD_ANY_CAST(absl::any_cast<const int>(AsConst(a)));
>+  }
>+
>+  {
>+    absl::any a(absl::in_place_type_t<int>{});
>+    ABSL_ANY_TEST_EXPECT_BAD_ANY_CAST(absl::any_cast<float&>(a));
>+    ABSL_ANY_TEST_EXPECT_BAD_ANY_CAST(absl::any_cast<const float&>(a));
>+    ABSL_ANY_TEST_EXPECT_BAD_ANY_CAST(absl::any_cast<float&&>(absl::any{}));
>+    ABSL_ANY_TEST_EXPECT_BAD_ANY_CAST(
>+        absl::any_cast<const float&&>(absl::any{}));
>+    ABSL_ANY_TEST_EXPECT_BAD_ANY_CAST(absl::any_cast<float>(a));
>+    ABSL_ANY_TEST_EXPECT_BAD_ANY_CAST(absl::any_cast<const float>(a));
>+    ABSL_ANY_TEST_EXPECT_BAD_ANY_CAST(absl::any_cast<float>(absl::any{}));
>+    ABSL_ANY_TEST_EXPECT_BAD_ANY_CAST(absl::any_cast<const float>(absl::any{}));
>+
>+    // const absl::any operand
>+    ABSL_ANY_TEST_EXPECT_BAD_ANY_CAST(absl::any_cast<const float&>(AsConst(a)));
>+    ABSL_ANY_TEST_EXPECT_BAD_ANY_CAST(absl::any_cast<float>(AsConst(a)));
>+    ABSL_ANY_TEST_EXPECT_BAD_ANY_CAST(absl::any_cast<const float>(AsConst(a)));
>+  }
>+}
>+
>+class BadCopy {};
>+
>+struct BadCopyable {
>+  BadCopyable() = default;
>+  BadCopyable(BadCopyable&&) = default;
>+  BadCopyable(const BadCopyable&) {
>+#ifdef ABSL_HAVE_EXCEPTIONS
>+    throw BadCopy();
>+#else
>+    ABSL_RAW_LOG(FATAL, "Bad copy");
>+#endif
>+  }
>+};
>+
>+#define ABSL_ANY_TEST_EXPECT_BAD_COPY(...) \
>+  ABSL_BASE_INTERNAL_EXPECT_FAIL((__VA_ARGS__), BadCopy, "Bad copy")
>+
>+// Test the guarantees regarding exceptions in copy/assign.
>+TEST(AnyTest, FailedCopy) {
>+  {
>+    const BadCopyable bad{};
>+    ABSL_ANY_TEST_EXPECT_BAD_COPY(absl::any{bad});
>+  }
>+
>+  {
>+    absl::any src(absl::in_place_type_t<BadCopyable>{});
>+    ABSL_ANY_TEST_EXPECT_BAD_COPY(absl::any{src});
>+  }
>+
>+  {
>+    BadCopyable bad;
>+    absl::any target;
>+    ABSL_ANY_TEST_EXPECT_BAD_COPY(target = bad);
>+  }
>+
>+  {
>+    BadCopyable bad;
>+    absl::any target(absl::in_place_type_t<BadCopyable>{});
>+    ABSL_ANY_TEST_EXPECT_BAD_COPY(target = bad);
>+    EXPECT_TRUE(target.has_value());
>+  }
>+
>+  {
>+    absl::any src(absl::in_place_type_t<BadCopyable>{});
>+    absl::any target;
>+    ABSL_ANY_TEST_EXPECT_BAD_COPY(target = src);
>+    EXPECT_FALSE(target.has_value());
>+  }
>+
>+  {
>+    absl::any src(absl::in_place_type_t<BadCopyable>{});
>+    absl::any target(absl::in_place_type_t<BadCopyable>{});
>+    ABSL_ANY_TEST_EXPECT_BAD_COPY(target = src);
>+    EXPECT_TRUE(target.has_value());
>+  }
>+}
>+
>+// Test the guarantees regarding exceptions in emplace.
>+TEST(AnyTest, FailedEmplace) {
>+  {
>+    BadCopyable bad;
>+    absl::any target;
>+    ABSL_ANY_TEST_EXPECT_BAD_COPY(target.emplace<BadCopyable>(bad));
>+  }
>+
>+  {
>+    BadCopyable bad;
>+    absl::any target(absl::in_place_type_t<int>{});
>+    ABSL_ANY_TEST_EXPECT_BAD_COPY(target.emplace<BadCopyable>(bad));
>+#if defined(ABSL_HAVE_STD_ANY) && defined(__GLIBCXX__)
>+    // libstdc++ std::any::emplace() implementation (as of 7.2) has a bug: if an
>+    // exception is thrown, *this contains a value.
>+#define ABSL_GLIBCXX_ANY_EMPLACE_EXCEPTION_BUG 1
>+#endif
>+#if defined(ABSL_HAVE_EXCEPTIONS) && \
>+    !defined(ABSL_GLIBCXX_ANY_EMPLACE_EXCEPTION_BUG)
>+    EXPECT_FALSE(target.has_value());
>+#endif
>+  }
>+}
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/bad_any_cast.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/bad_any_cast.cc
>new file mode 100644
>index 00000000000..2e2fd29a337
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/bad_any_cast.cc
>@@ -0,0 +1,44 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/types/bad_any_cast.h"
>+
>+#ifndef ABSL_HAVE_STD_ANY
>+
>+#include <cstdlib>
>+
>+#include "absl/base/config.h"
>+#include "absl/base/internal/raw_logging.h"
>+
>+namespace absl {
>+
>+bad_any_cast::~bad_any_cast() = default;
>+
>+const char* bad_any_cast::what() const noexcept { return "Bad any cast"; }
>+
>+namespace any_internal {
>+
>+void ThrowBadAnyCast() {
>+#ifdef ABSL_HAVE_EXCEPTIONS
>+  throw bad_any_cast();
>+#else
>+  ABSL_RAW_LOG(FATAL, "Bad any cast");
>+  std::abort();
>+#endif
>+}
>+
>+}  // namespace any_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_HAVE_STD_ANY
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/bad_any_cast.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/bad_any_cast.h
>new file mode 100644
>index 00000000000..603901324af
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/bad_any_cast.h
>@@ -0,0 +1,71 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// -----------------------------------------------------------------------------
>+// bad_any_cast.h
>+// -----------------------------------------------------------------------------
>+//
>+// This header file defines the `absl::bad_any_cast` type.
>+
>+#ifndef ABSL_TYPES_BAD_ANY_CAST_H_
>+#define ABSL_TYPES_BAD_ANY_CAST_H_
>+
>+#include <typeinfo>
>+
>+#include "absl/base/config.h"
>+
>+#ifdef ABSL_HAVE_STD_ANY
>+
>+#include <any>
>+
>+namespace absl {
>+using std::bad_any_cast;
>+}  // namespace absl
>+
>+#else  // ABSL_HAVE_STD_ANY
>+
>+namespace absl {
>+
>+// -----------------------------------------------------------------------------
>+// bad_any_cast
>+// -----------------------------------------------------------------------------
>+//
>+// An `absl::bad_any_cast` type is an exception type that is thrown when
>+// failing to successfully cast the return value of an `absl::any` object.
>+//
>+// Example:
>+//
>+//   auto a = absl::any(65);
>+//   absl::any_cast<int>(a);         // 65
>+//   try {
>+//     absl::any_cast<char>(a);
>+//   } catch(const absl::bad_any_cast& e) {
>+//     std::cout << "Bad any cast: " << e.what() << '\n';
>+//   }
>+class bad_any_cast : public std::bad_cast {
>+ public:
>+  ~bad_any_cast() override;
>+  const char* what() const noexcept override;
>+};
>+
>+namespace any_internal {
>+
>+[[noreturn]] void ThrowBadAnyCast();
>+
>+}  // namespace any_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_HAVE_STD_ANY
>+
>+#endif  // ABSL_TYPES_BAD_ANY_CAST_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/bad_optional_access.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/bad_optional_access.cc
>new file mode 100644
>index 00000000000..558707760e9
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/bad_optional_access.cc
>@@ -0,0 +1,46 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/types/bad_optional_access.h"
>+
>+#ifndef ABSL_HAVE_STD_OPTIONAL
>+
>+#include <cstdlib>
>+
>+#include "absl/base/config.h"
>+#include "absl/base/internal/raw_logging.h"
>+
>+namespace absl {
>+
>+bad_optional_access::~bad_optional_access() = default;
>+
>+const char* bad_optional_access::what() const noexcept {
>+  return "optional has no value";
>+}
>+
>+namespace optional_internal {
>+
>+void throw_bad_optional_access() {
>+#ifdef ABSL_HAVE_EXCEPTIONS
>+  throw bad_optional_access();
>+#else
>+  ABSL_RAW_LOG(FATAL, "Bad optional access");
>+  abort();
>+#endif
>+}
>+
>+}  // namespace optional_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_HAVE_STD_OPTIONAL
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/bad_optional_access.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/bad_optional_access.h
>new file mode 100644
>index 00000000000..c6c27460a76
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/bad_optional_access.h
>@@ -0,0 +1,74 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// -----------------------------------------------------------------------------
>+// bad_optional_access.h
>+// -----------------------------------------------------------------------------
>+//
>+// This header file defines the `absl::bad_optional_access` type.
>+
>+#ifndef ABSL_TYPES_BAD_OPTIONAL_ACCESS_H_
>+#define ABSL_TYPES_BAD_OPTIONAL_ACCESS_H_
>+
>+#include <stdexcept>
>+
>+#include "absl/base/config.h"
>+
>+#ifdef ABSL_HAVE_STD_OPTIONAL
>+
>+#include <optional>
>+
>+namespace absl {
>+using std::bad_optional_access;
>+}  // namespace absl
>+
>+#else  // ABSL_HAVE_STD_OPTIONAL
>+
>+namespace absl {
>+
>+// -----------------------------------------------------------------------------
>+// bad_optional_access
>+// -----------------------------------------------------------------------------
>+//
>+// An `absl::bad_optional_access` type is an exception type that is thrown when
>+// attempting to access an `absl::optional` object that does not contain a
>+// value.
>+//
>+// Example:
>+//
>+//   absl::optional<int> o;
>+//
>+//   try {
>+//     int n = o.value();
>+//   } catch(const absl::bad_optional_access& e) {
>+//     std::cout << "Bad optional access: " << e.what() << '\n';
>+//   }
>+class bad_optional_access : public std::exception {
>+ public:
>+  bad_optional_access() = default;
>+  ~bad_optional_access() override;
>+  const char* what() const noexcept override;
>+};
>+
>+namespace optional_internal {
>+
>+// throw delegator
>+[[noreturn]] void throw_bad_optional_access();
>+
>+}  // namespace optional_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_HAVE_STD_OPTIONAL
>+
>+#endif  // ABSL_TYPES_BAD_OPTIONAL_ACCESS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/bad_variant_access.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/bad_variant_access.cc
>new file mode 100644
>index 00000000000..d27d7756179
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/bad_variant_access.cc
>@@ -0,0 +1,62 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/types/bad_variant_access.h"
>+
>+#ifndef ABSL_HAVE_STD_VARIANT
>+
>+#include <cstdlib>
>+#include <stdexcept>
>+
>+#include "absl/base/config.h"
>+#include "absl/base/internal/raw_logging.h"
>+
>+namespace absl {
>+
>+//////////////////////////
>+// [variant.bad.access] //
>+//////////////////////////
>+
>+bad_variant_access::~bad_variant_access() = default;
>+
>+const char* bad_variant_access::what() const noexcept {
>+  return "Bad variant access";
>+}
>+
>+namespace variant_internal {
>+
>+void ThrowBadVariantAccess() {
>+#ifdef ABSL_HAVE_EXCEPTIONS
>+  throw bad_variant_access();
>+#else
>+  ABSL_RAW_LOG(FATAL, "Bad variant access");
>+  abort();  // TODO(calabrese) Remove once RAW_LOG FATAL is noreturn.
>+#endif
>+}
>+
>+void Rethrow() {
>+#ifdef ABSL_HAVE_EXCEPTIONS
>+  throw;
>+#else
>+  ABSL_RAW_LOG(FATAL,
>+               "Internal error in absl::variant implementation. Attempted to "
>+               "rethrow an exception when building with exceptions disabled.");
>+  abort();  // TODO(calabrese) Remove once RAW_LOG FATAL is noreturn.
>+#endif
>+}
>+
>+}  // namespace variant_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_HAVE_STD_VARIANT
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/bad_variant_access.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/bad_variant_access.h
>new file mode 100644
>index 00000000000..e7355a5a74c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/bad_variant_access.h
>@@ -0,0 +1,78 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// -----------------------------------------------------------------------------
>+// bad_variant_access.h
>+// -----------------------------------------------------------------------------
>+//
>+// This header file defines the `absl::bad_variant_access` type.
>+
>+#ifndef ABSL_TYPES_BAD_VARIANT_ACCESS_H_
>+#define ABSL_TYPES_BAD_VARIANT_ACCESS_H_
>+
>+#include <stdexcept>
>+
>+#include "absl/base/config.h"
>+
>+#ifdef ABSL_HAVE_STD_VARIANT
>+
>+#include <variant>
>+
>+namespace absl {
>+using std::bad_variant_access;
>+}  // namespace absl
>+
>+#else  // ABSL_HAVE_STD_VARIANT
>+
>+namespace absl {
>+
>+// -----------------------------------------------------------------------------
>+// bad_variant_access
>+// -----------------------------------------------------------------------------
>+//
>+// An `absl::bad_variant_access` type is an exception type that is thrown in
>+// the following cases:
>+//
>+//   * Calling `absl::get(absl::variant) with an index or type that does not
>+//     match the currently selected alternative type
>+//   * Calling `absl::visit on an `absl::variant` that is in the
>+//     `variant::valueless_by_exception` state.
>+//
>+// Example:
>+//
>+//   absl::variant<int, std::string> v;
>+//   v = 1;
>+//   try {
>+//     absl::get<std::string>(v);
>+//   } catch(const absl::bad_variant_access& e) {
>+//     std::cout << "Bad variant access: " << e.what() << '\n';
>+//   }
>+class bad_variant_access : public std::exception {
>+ public:
>+  bad_variant_access() noexcept = default;
>+  ~bad_variant_access() override;
>+  const char* what() const noexcept override;
>+};
>+
>+namespace variant_internal {
>+
>+[[noreturn]] void ThrowBadVariantAccess();
>+[[noreturn]] void Rethrow();
>+
>+}  // namespace variant_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_HAVE_STD_VARIANT
>+
>+#endif  // ABSL_TYPES_BAD_VARIANT_ACCESS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/internal/variant.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/internal/variant.h
>new file mode 100644
>index 00000000000..7708e67cb14
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/internal/variant.h
>@@ -0,0 +1,1615 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// Implementation details of absl/types/variant.h, pulled into a
>+// separate file to avoid cluttering the top of the API header with
>+// implementation details.
>+
>+#ifndef ABSL_TYPES_variant_internal_H_
>+#define ABSL_TYPES_variant_internal_H_
>+
>+#include <cassert>
>+#include <cstddef>
>+#include <cstdlib>
>+#include <memory>
>+#include <stdexcept>
>+#include <tuple>
>+#include <type_traits>
>+
>+#include "absl/base/config.h"
>+#include "absl/base/internal/identity.h"
>+#include "absl/base/internal/inline_variable.h"
>+#include "absl/base/internal/invoke.h"
>+#include "absl/base/macros.h"
>+#include "absl/base/optimization.h"
>+#include "absl/meta/type_traits.h"
>+#include "absl/types/bad_variant_access.h"
>+#include "absl/utility/utility.h"
>+
>+namespace absl {
>+
>+template <class... Types>
>+class variant;
>+
>+ABSL_INTERNAL_INLINE_CONSTEXPR(size_t, variant_npos, -1);
>+
>+template <class T>
>+struct variant_size;
>+
>+template <std::size_t I, class T>
>+struct variant_alternative;
>+
>+namespace variant_internal {
>+
>+// NOTE: See specializations below for details.
>+template <std::size_t I, class T>
>+struct VariantAlternativeSfinae {};
>+
>+// Requires: I < variant_size_v<T>.
>+//
>+// Value: The Ith type of Types...
>+template <std::size_t I, class T0, class... Tn>
>+struct VariantAlternativeSfinae<I, variant<T0, Tn...>>
>+    : VariantAlternativeSfinae<I - 1, variant<Tn...>> {};
>+
>+// Value: T0
>+template <class T0, class... Ts>
>+struct VariantAlternativeSfinae<0, variant<T0, Ts...>> {
>+  using type = T0;
>+};
>+
>+template <std::size_t I, class T>
>+using VariantAlternativeSfinaeT = typename VariantAlternativeSfinae<I, T>::type;
>+
>+// NOTE: Requires T to be a reference type.
>+template <class T, class U>
>+struct GiveQualsTo;
>+
>+template <class T, class U>
>+struct GiveQualsTo<T&, U> {
>+  using type = U&;
>+};
>+
>+template <class T, class U>
>+struct GiveQualsTo<T&&, U> {
>+  using type = U&&;
>+};
>+
>+template <class T, class U>
>+struct GiveQualsTo<const T&, U> {
>+  using type = const U&;
>+};
>+
>+template <class T, class U>
>+struct GiveQualsTo<const T&&, U> {
>+  using type = const U&&;
>+};
>+
>+template <class T, class U>
>+struct GiveQualsTo<volatile T&, U> {
>+  using type = volatile U&;
>+};
>+
>+template <class T, class U>
>+struct GiveQualsTo<volatile T&&, U> {
>+  using type = volatile U&&;
>+};
>+
>+template <class T, class U>
>+struct GiveQualsTo<volatile const T&, U> {
>+  using type = volatile const U&;
>+};
>+
>+template <class T, class U>
>+struct GiveQualsTo<volatile const T&&, U> {
>+  using type = volatile const U&&;
>+};
>+
>+template <class T, class U>
>+using GiveQualsToT = typename GiveQualsTo<T, U>::type;
>+
>+// Convenience alias, since size_t integral_constant is used a lot in this file.
>+template <std::size_t I>
>+using SizeT = std::integral_constant<std::size_t, I>;
>+
>+using NPos = SizeT<variant_npos>;
>+
>+template <class Variant, class T, class = void>
>+struct IndexOfConstructedType {};
>+
>+template <std::size_t I, class Variant>
>+struct VariantAccessResultImpl;
>+
>+template <std::size_t I, template <class...> class Variantemplate, class... T>
>+struct VariantAccessResultImpl<I, Variantemplate<T...>&> {
>+  using type = typename absl::variant_alternative<I, variant<T...>>::type&;
>+};
>+
>+template <std::size_t I, template <class...> class Variantemplate, class... T>
>+struct VariantAccessResultImpl<I, const Variantemplate<T...>&> {
>+  using type =
>+      const typename absl::variant_alternative<I, variant<T...>>::type&;
>+};
>+
>+template <std::size_t I, template <class...> class Variantemplate, class... T>
>+struct VariantAccessResultImpl<I, Variantemplate<T...>&&> {
>+  using type = typename absl::variant_alternative<I, variant<T...>>::type&&;
>+};
>+
>+template <std::size_t I, template <class...> class Variantemplate, class... T>
>+struct VariantAccessResultImpl<I, const Variantemplate<T...>&&> {
>+  using type =
>+      const typename absl::variant_alternative<I, variant<T...>>::type&&;
>+};
>+
>+template <std::size_t I, class Variant>
>+using VariantAccessResult =
>+    typename VariantAccessResultImpl<I, Variant&&>::type;
>+
>+// NOTE: This is used instead of std::array to reduce instantiation overhead.
>+template <class T, std::size_t Size>
>+struct SimpleArray {
>+  static_assert(Size != 0, "");
>+  T value[Size];
>+};
>+
>+template <class T>
>+struct AccessedType {
>+  using type = T;
>+};
>+
>+template <class T>
>+using AccessedTypeT = typename AccessedType<T>::type;
>+
>+template <class T, std::size_t Size>
>+struct AccessedType<SimpleArray<T, Size>> {
>+  using type = AccessedTypeT<T>;
>+};
>+
>+template <class T>
>+constexpr T AccessSimpleArray(const T& value) {
>+  return value;
>+}
>+
>+template <class T, std::size_t Size, class... SizeT>
>+constexpr AccessedTypeT<T> AccessSimpleArray(const SimpleArray<T, Size>& table,
>+                                             std::size_t head_index,
>+                                             SizeT... tail_indices) {
>+  return AccessSimpleArray(table.value[head_index], tail_indices...);
>+}
>+
>+// Note: Intentionally is an alias.
>+template <class T>
>+using AlwaysZero = SizeT<0>;
>+
>+template <class Op, class... Vs>
>+struct VisitIndicesResultImpl {
>+  using type = absl::result_of_t<Op(AlwaysZero<Vs>...)>;
>+};
>+
>+template <class Op, class... Vs>
>+using VisitIndicesResultT = typename VisitIndicesResultImpl<Op, Vs...>::type;
>+
>+template <class ReturnType, class FunctionObject, class EndIndices,
>+          std::size_t... BoundIndices>
>+struct MakeVisitationMatrix;
>+
>+template <class ReturnType, class FunctionObject, std::size_t... Indices>
>+constexpr ReturnType call_with_indices(FunctionObject&& function) {
>+  static_assert(
>+      std::is_same<ReturnType, decltype(std::declval<FunctionObject>()(
>+                                   SizeT<Indices>()...))>::value,
>+      "Not all visitation overloads have the same return type.");
>+  return absl::forward<FunctionObject>(function)(SizeT<Indices>()...);
>+}
>+
>+template <class ReturnType, class FunctionObject, std::size_t... BoundIndices>
>+struct MakeVisitationMatrix<ReturnType, FunctionObject, index_sequence<>,
>+                            BoundIndices...> {
>+  using ResultType = ReturnType (*)(FunctionObject&&);
>+  static constexpr ResultType Run() {
>+    return &call_with_indices<ReturnType, FunctionObject,
>+                              (BoundIndices - 1)...>;
>+  }
>+};
>+
>+template <class ReturnType, class FunctionObject, class EndIndices,
>+          class CurrIndices, std::size_t... BoundIndices>
>+struct MakeVisitationMatrixImpl;
>+
>+template <class ReturnType, class FunctionObject, std::size_t... EndIndices,
>+          std::size_t... CurrIndices, std::size_t... BoundIndices>
>+struct MakeVisitationMatrixImpl<
>+    ReturnType, FunctionObject, index_sequence<EndIndices...>,
>+    index_sequence<CurrIndices...>, BoundIndices...> {
>+  using ResultType = SimpleArray<
>+      typename MakeVisitationMatrix<ReturnType, FunctionObject,
>+                                    index_sequence<EndIndices...>>::ResultType,
>+      sizeof...(CurrIndices)>;
>+
>+  static constexpr ResultType Run() {
>+    return {{MakeVisitationMatrix<ReturnType, FunctionObject,
>+                                  index_sequence<EndIndices...>,
>+                                  BoundIndices..., CurrIndices>::Run()...}};
>+  }
>+};
>+
>+template <class ReturnType, class FunctionObject, std::size_t HeadEndIndex,
>+          std::size_t... TailEndIndices, std::size_t... BoundIndices>
>+struct MakeVisitationMatrix<ReturnType, FunctionObject,
>+                            index_sequence<HeadEndIndex, TailEndIndices...>,
>+                            BoundIndices...>
>+    : MakeVisitationMatrixImpl<
>+          ReturnType, FunctionObject, index_sequence<TailEndIndices...>,
>+          absl::make_index_sequence<HeadEndIndex>, BoundIndices...> {};
>+
>+struct UnreachableSwitchCase {
>+  template <class Op>
>+  [[noreturn]] static VisitIndicesResultT<Op, std::size_t> Run(
>+      Op&& /*ignored*/) {
>+#if ABSL_HAVE_BUILTIN(__builtin_unreachable) || \
>+    (defined(__GNUC__) && !defined(__clang__))
>+    __builtin_unreachable();
>+#elif defined(_MSC_VER)
>+    __assume(false);
>+#else
>+    // Try to use assert of false being identified as an unreachable intrinsic.
>+    // NOTE: We use assert directly to increase chances of exploiting an assume
>+    //       intrinsic.
>+    assert(false);  // NOLINT
>+
>+    // Hack to silence potential no return warning -- cause an infinite loop.
>+    return Run(absl::forward<Op>(op));
>+#endif  // Checks for __builtin_unreachable
>+  }
>+};
>+
>+template <class Op, std::size_t I>
>+struct ReachableSwitchCase {
>+  static VisitIndicesResultT<Op, std::size_t> Run(Op&& op) {
>+    return absl::base_internal::Invoke(absl::forward<Op>(op), SizeT<I>());
>+  }
>+};
>+
>+// The number 33 is just a guess at a reasonable maximum to our switch. It is
>+// not based on any analysis. The reason it is a power of 2 plus 1 instead of a
>+// power of 2 is because the number was picked to correspond to a power of 2
>+// amount of "normal" alternatives, plus one for the possibility of the user
>+// providing "monostate" in addition to the more natural alternatives.
>+ABSL_INTERNAL_INLINE_CONSTEXPR(std::size_t, MaxUnrolledVisitCases, 33);
>+
>+// Note: The default-definition is for unreachable cases.
>+template <bool IsReachable>
>+struct PickCaseImpl {
>+  template <class Op, std::size_t I>
>+  using Apply = UnreachableSwitchCase;
>+};
>+
>+template <>
>+struct PickCaseImpl</*IsReachable =*/true> {
>+  template <class Op, std::size_t I>
>+  using Apply = ReachableSwitchCase<Op, I>;
>+};
>+
>+// Note: This form of dance with template aliases is to make sure that we
>+//       instantiate a number of templates proportional to the number of variant
>+//       alternatives rather than a number of templates proportional to our
>+//       maximum unrolled amount of visitation cases (aliases are effectively
>+//       "free" whereas other template instantiations are costly).
>+template <class Op, std::size_t I, std::size_t EndIndex>
>+using PickCase = typename PickCaseImpl<(I < EndIndex)>::template Apply<Op, I>;
>+
>+template <class ReturnType>
>+[[noreturn]] ReturnType TypedThrowBadVariantAccess() {
>+  absl::variant_internal::ThrowBadVariantAccess();
>+}
>+
>+// Given N variant sizes, determine the number of cases there would need to be
>+// in a single switch-statement that would cover every possibility in the
>+// corresponding N-ary visit operation.
>+template <std::size_t... NumAlternatives>
>+struct NumCasesOfSwitch;
>+
>+template <std::size_t HeadNumAlternatives, std::size_t... TailNumAlternatives>
>+struct NumCasesOfSwitch<HeadNumAlternatives, TailNumAlternatives...> {
>+  static constexpr std::size_t value =
>+      (HeadNumAlternatives + 1) *
>+      NumCasesOfSwitch<TailNumAlternatives...>::value;
>+};
>+
>+template <>
>+struct NumCasesOfSwitch<> {
>+  static constexpr std::size_t value = 1;
>+};
>+
>+// A switch statement optimizes better than the table of function pointers.
>+template <std::size_t EndIndex>
>+struct VisitIndicesSwitch {
>+  static_assert(EndIndex <= MaxUnrolledVisitCases,
>+                "Maximum unrolled switch size exceeded.");
>+
>+  template <class Op>
>+  static VisitIndicesResultT<Op, std::size_t> Run(Op&& op, std::size_t i) {
>+    switch (i) {
>+      case 0:
>+        return PickCase<Op, 0, EndIndex>::Run(absl::forward<Op>(op));
>+      case 1:
>+        return PickCase<Op, 1, EndIndex>::Run(absl::forward<Op>(op));
>+      case 2:
>+        return PickCase<Op, 2, EndIndex>::Run(absl::forward<Op>(op));
>+      case 3:
>+        return PickCase<Op, 3, EndIndex>::Run(absl::forward<Op>(op));
>+      case 4:
>+        return PickCase<Op, 4, EndIndex>::Run(absl::forward<Op>(op));
>+      case 5:
>+        return PickCase<Op, 5, EndIndex>::Run(absl::forward<Op>(op));
>+      case 6:
>+        return PickCase<Op, 6, EndIndex>::Run(absl::forward<Op>(op));
>+      case 7:
>+        return PickCase<Op, 7, EndIndex>::Run(absl::forward<Op>(op));
>+      case 8:
>+        return PickCase<Op, 8, EndIndex>::Run(absl::forward<Op>(op));
>+      case 9:
>+        return PickCase<Op, 9, EndIndex>::Run(absl::forward<Op>(op));
>+      case 10:
>+        return PickCase<Op, 10, EndIndex>::Run(absl::forward<Op>(op));
>+      case 11:
>+        return PickCase<Op, 11, EndIndex>::Run(absl::forward<Op>(op));
>+      case 12:
>+        return PickCase<Op, 12, EndIndex>::Run(absl::forward<Op>(op));
>+      case 13:
>+        return PickCase<Op, 13, EndIndex>::Run(absl::forward<Op>(op));
>+      case 14:
>+        return PickCase<Op, 14, EndIndex>::Run(absl::forward<Op>(op));
>+      case 15:
>+        return PickCase<Op, 15, EndIndex>::Run(absl::forward<Op>(op));
>+      case 16:
>+        return PickCase<Op, 16, EndIndex>::Run(absl::forward<Op>(op));
>+      case 17:
>+        return PickCase<Op, 17, EndIndex>::Run(absl::forward<Op>(op));
>+      case 18:
>+        return PickCase<Op, 18, EndIndex>::Run(absl::forward<Op>(op));
>+      case 19:
>+        return PickCase<Op, 19, EndIndex>::Run(absl::forward<Op>(op));
>+      case 20:
>+        return PickCase<Op, 20, EndIndex>::Run(absl::forward<Op>(op));
>+      case 21:
>+        return PickCase<Op, 21, EndIndex>::Run(absl::forward<Op>(op));
>+      case 22:
>+        return PickCase<Op, 22, EndIndex>::Run(absl::forward<Op>(op));
>+      case 23:
>+        return PickCase<Op, 23, EndIndex>::Run(absl::forward<Op>(op));
>+      case 24:
>+        return PickCase<Op, 24, EndIndex>::Run(absl::forward<Op>(op));
>+      case 25:
>+        return PickCase<Op, 25, EndIndex>::Run(absl::forward<Op>(op));
>+      case 26:
>+        return PickCase<Op, 26, EndIndex>::Run(absl::forward<Op>(op));
>+      case 27:
>+        return PickCase<Op, 27, EndIndex>::Run(absl::forward<Op>(op));
>+      case 28:
>+        return PickCase<Op, 28, EndIndex>::Run(absl::forward<Op>(op));
>+      case 29:
>+        return PickCase<Op, 29, EndIndex>::Run(absl::forward<Op>(op));
>+      case 30:
>+        return PickCase<Op, 30, EndIndex>::Run(absl::forward<Op>(op));
>+      case 31:
>+        return PickCase<Op, 31, EndIndex>::Run(absl::forward<Op>(op));
>+      case 32:
>+        return PickCase<Op, 32, EndIndex>::Run(absl::forward<Op>(op));
>+      default:
>+        ABSL_ASSERT(i == variant_npos);
>+        return absl::base_internal::Invoke(absl::forward<Op>(op), NPos());
>+    }
>+  }
>+};
>+
>+template <std::size_t... EndIndices>
>+struct VisitIndicesFallback {
>+  template <class Op, class... SizeT>
>+  static VisitIndicesResultT<Op, SizeT...> Run(Op&& op, SizeT... indices) {
>+    return AccessSimpleArray(
>+        MakeVisitationMatrix<VisitIndicesResultT<Op, SizeT...>, Op,
>+                             index_sequence<(EndIndices + 1)...>>::Run(),
>+        (indices + 1)...)(absl::forward<Op>(op));
>+  }
>+};
>+
>+// Take an N-dimensional series of indices and convert them into a single index
>+// without loss of information. The purpose of this is to be able to convert an
>+// N-ary visit operation into a single switch statement.
>+template <std::size_t...>
>+struct FlattenIndices;
>+
>+template <std::size_t HeadSize, std::size_t... TailSize>
>+struct FlattenIndices<HeadSize, TailSize...> {
>+  template<class... SizeType>
>+  static constexpr std::size_t Run(std::size_t head, SizeType... tail) {
>+    return head + HeadSize * FlattenIndices<TailSize...>::Run(tail...);
>+  }
>+};
>+
>+template <>
>+struct FlattenIndices<> {
>+  static constexpr std::size_t Run() { return 0; }
>+};
>+
>+// Take a single "flattened" index (flattened by FlattenIndices) and determine
>+// the value of the index of one of the logically represented dimensions.
>+template <std::size_t I, std::size_t IndexToGet, std::size_t HeadSize,
>+          std::size_t... TailSize>
>+struct UnflattenIndex {
>+  static constexpr std::size_t value =
>+      UnflattenIndex<I / HeadSize, IndexToGet - 1, TailSize...>::value;
>+};
>+
>+template <std::size_t I, std::size_t HeadSize, std::size_t... TailSize>
>+struct UnflattenIndex<I, 0, HeadSize, TailSize...> {
>+  static constexpr std::size_t value = (I % HeadSize);
>+};
>+
>+// The backend for converting an N-ary visit operation into a unary visit.
>+template <class IndexSequence, std::size_t... EndIndices>
>+struct VisitIndicesVariadicImpl;
>+
>+template <std::size_t... N, std::size_t... EndIndices>
>+struct VisitIndicesVariadicImpl<absl::index_sequence<N...>, EndIndices...> {
>+  // A type that can take an N-ary function object and converts it to a unary
>+  // function object that takes a single, flattened index, and "unflattens" it
>+  // into its individual dimensions when forwarding to the wrapped object.
>+  template <class Op>
>+  struct FlattenedOp {
>+    template <std::size_t I>
>+    VisitIndicesResultT<Op, decltype(EndIndices)...> operator()(
>+        SizeT<I> /*index*/) && {
>+      return base_internal::Invoke(
>+          absl::forward<Op>(op),
>+          SizeT<UnflattenIndex<I, N, (EndIndices + 1)...>::value -
>+                std::size_t{1}>()...);
>+    }
>+
>+    Op&& op;
>+  };
>+
>+  template <class Op, class... SizeType>
>+  static VisitIndicesResultT<Op, decltype(EndIndices)...> Run(
>+      Op&& op, SizeType... i) {
>+    return VisitIndicesSwitch<NumCasesOfSwitch<EndIndices...>::value>::Run(
>+        FlattenedOp<Op>{absl::forward<Op>(op)},
>+        FlattenIndices<(EndIndices + std::size_t{1})...>::Run(
>+            (i + std::size_t{1})...));
>+  }
>+};
>+
>+template <std::size_t... EndIndices>
>+struct VisitIndicesVariadic
>+    : VisitIndicesVariadicImpl<absl::make_index_sequence<sizeof...(EndIndices)>,
>+                               EndIndices...> {};
>+
>+// This implementation will flatten N-ary visit operations into a single switch
>+// statement when the number of cases would be less than our maximum specified
>+// switch-statement size.
>+// TODO(calabrese)
>+//   Based on benchmarks, determine whether the function table approach actually
>+//   does optimize better than a chain of switch statements and possibly update
>+//   the implementation accordingly. Also consider increasing the maximum switch
>+//   size.
>+template <std::size_t... EndIndices>
>+struct VisitIndices
>+    : absl::conditional_t<(NumCasesOfSwitch<EndIndices...>::value <=
>+                           MaxUnrolledVisitCases),
>+                          VisitIndicesVariadic<EndIndices...>,
>+                          VisitIndicesFallback<EndIndices...>> {};
>+
>+template <std::size_t EndIndex>
>+struct VisitIndices<EndIndex>
>+    : absl::conditional_t<(EndIndex <= MaxUnrolledVisitCases),
>+                          VisitIndicesSwitch<EndIndex>,
>+                          VisitIndicesFallback<EndIndex>> {};
>+
>+// Suppress bogus warning on MSVC: MSVC complains that the `reinterpret_cast`
>+// below is returning the address of a temporary or local object.
>+#ifdef _MSC_VER
>+#pragma warning(push)
>+#pragma warning(disable : 4172)
>+#endif  // _MSC_VER
>+
>+// TODO(calabrese) std::launder
>+// TODO(calabrese) constexpr
>+// NOTE: DO NOT REMOVE the `inline` keyword as it is necessary to work around a
>+// MSVC bug. See https://github.com/abseil/abseil-cpp/issues/129 for details.
>+template <class Self, std::size_t I>
>+inline VariantAccessResult<I, Self> AccessUnion(Self&& self, SizeT<I> /*i*/) {
>+  return reinterpret_cast<VariantAccessResult<I, Self>>(self);
>+}
>+
>+#ifdef _MSC_VER
>+#pragma warning(pop)
>+#endif  // _MSC_VER
>+
>+template <class T>
>+void DeducedDestroy(T& self) {  // NOLINT
>+  self.~T();
>+}
>+
>+// NOTE: This type exists as a single entity for variant and its bases to
>+// befriend. It contains helper functionality that manipulates the state of the
>+// variant, such as the implementation of things like assignment and emplace
>+// operations.
>+struct VariantCoreAccess {
>+  template <class VariantType>
>+  static typename VariantType::Variant& Derived(VariantType& self) {  // NOLINT
>+    return static_cast<typename VariantType::Variant&>(self);
>+  }
>+
>+  template <class VariantType>
>+  static const typename VariantType::Variant& Derived(
>+      const VariantType& self) {  // NOLINT
>+    return static_cast<const typename VariantType::Variant&>(self);
>+  }
>+
>+  template <class VariantType>
>+  static void Destroy(VariantType& self) {  // NOLINT
>+    Derived(self).destroy();
>+    self.index_ = absl::variant_npos;
>+  }
>+
>+  template <class Variant>
>+  static void SetIndex(Variant& self, std::size_t i) {  // NOLINT
>+    self.index_ = i;
>+  }
>+
>+  template <class Variant>
>+  static void InitFrom(Variant& self, Variant&& other) {  // NOLINT
>+    VisitIndices<absl::variant_size<Variant>::value>::Run(
>+        InitFromVisitor<Variant, Variant&&>{&self,
>+                                            std::forward<Variant>(other)},
>+        other.index());
>+    self.index_ = other.index();
>+  }
>+
>+  // Access a variant alternative, assuming the index is correct.
>+  template <std::size_t I, class Variant>
>+  static VariantAccessResult<I, Variant> Access(Variant&& self) {
>+    // This cast instead of invocation of AccessUnion with an rvalue is a
>+    // workaround for msvc. Without this there is a runtime failure when dealing
>+    // with rvalues.
>+    // TODO(calabrese) Reduce test case and find a simpler workaround.
>+    return static_cast<VariantAccessResult<I, Variant>>(
>+        variant_internal::AccessUnion(self.state_, SizeT<I>()));
>+  }
>+
>+  // Access a variant alternative, throwing if the index is incorrect.
>+  template <std::size_t I, class Variant>
>+  static VariantAccessResult<I, Variant> CheckedAccess(Variant&& self) {
>+    if (ABSL_PREDICT_FALSE(self.index_ != I)) {
>+      TypedThrowBadVariantAccess<VariantAccessResult<I, Variant>>();
>+    }
>+
>+    return Access<I>(absl::forward<Variant>(self));
>+  }
>+
>+  // The implementation of the move-assignment operation for a variant.
>+  template <class VType>
>+  struct MoveAssignVisitor {
>+    using DerivedType = typename VType::Variant;
>+    template <std::size_t NewIndex>
>+    void operator()(SizeT<NewIndex> /*new_i*/) const {
>+      if (left->index_ == NewIndex) {
>+        Access<NewIndex>(*left) = std::move(Access<NewIndex>(*right));
>+      } else {
>+        Derived(*left).template emplace<NewIndex>(
>+            std::move(Access<NewIndex>(*right)));
>+      }
>+    }
>+
>+    void operator()(SizeT<absl::variant_npos> /*new_i*/) const {
>+      Destroy(*left);
>+    }
>+
>+    VType* left;
>+    VType* right;
>+  };
>+
>+  template <class VType>
>+  static MoveAssignVisitor<VType> MakeMoveAssignVisitor(VType* left,
>+                                                        VType* other) {
>+    return {left, other};
>+  }
>+
>+  // The implementation of the assignment operation for a variant.
>+  template <class VType>
>+  struct CopyAssignVisitor {
>+    using DerivedType = typename VType::Variant;
>+    template <std::size_t NewIndex>
>+    void operator()(SizeT<NewIndex> /*new_i*/) const {
>+      using New =
>+          typename absl::variant_alternative<NewIndex, DerivedType>::type;
>+
>+      if (left->index_ == NewIndex) {
>+        Access<NewIndex>(*left) = Access<NewIndex>(*right);
>+      } else if (std::is_nothrow_copy_constructible<New>::value ||
>+                 !std::is_nothrow_move_constructible<New>::value) {
>+        Derived(*left).template emplace<NewIndex>(Access<NewIndex>(*right));
>+      } else {
>+        Derived(*left) = DerivedType(Derived(*right));
>+      }
>+    }
>+
>+    void operator()(SizeT<absl::variant_npos> /*new_i*/) const {
>+      Destroy(*left);
>+    }
>+
>+    VType* left;
>+    const VType* right;
>+  };
>+
>+  template <class VType>
>+  static CopyAssignVisitor<VType> MakeCopyAssignVisitor(VType* left,
>+                                                        const VType& other) {
>+    return {left, &other};
>+  }
>+
>+  // The implementation of conversion-assignment operations for variant.
>+  template <class Left, class QualifiedNew>
>+  struct ConversionAssignVisitor {
>+    using NewIndex =
>+        variant_internal::IndexOfConstructedType<Left, QualifiedNew>;
>+
>+    void operator()(SizeT<NewIndex::value> /*old_i*/
>+                    ) const {
>+      Access<NewIndex::value>(*left) = absl::forward<QualifiedNew>(other);
>+    }
>+
>+    template <std::size_t OldIndex>
>+    void operator()(SizeT<OldIndex> /*old_i*/
>+                    ) const {
>+      using New =
>+          typename absl::variant_alternative<NewIndex::value, Left>::type;
>+      if (std::is_nothrow_constructible<New, QualifiedNew>::value ||
>+          !std::is_nothrow_move_constructible<New>::value) {
>+        left->template emplace<NewIndex::value>(
>+            absl::forward<QualifiedNew>(other));
>+      } else {
>+        // the standard says "equivalent to
>+        // operator=(variant(std::forward<T>(t)))", but we use `emplace` here
>+        // because the variant's move assignment operator could be deleted.
>+        left->template emplace<NewIndex::value>(
>+            New(absl::forward<QualifiedNew>(other)));
>+      }
>+    }
>+
>+    Left* left;
>+    QualifiedNew&& other;
>+  };
>+
>+  template <class Left, class QualifiedNew>
>+  static ConversionAssignVisitor<Left, QualifiedNew>
>+  MakeConversionAssignVisitor(Left* left, QualifiedNew&& qual) {
>+    return {left, absl::forward<QualifiedNew>(qual)};
>+  }
>+
>+  // Backend for operations for `emplace()` which destructs `*self` then
>+  // construct a new alternative with `Args...`.
>+  template <std::size_t NewIndex, class Self, class... Args>
>+  static typename absl::variant_alternative<NewIndex, Self>::type& Replace(
>+      Self* self, Args&&... args) {
>+    Destroy(*self);
>+    using New = typename absl::variant_alternative<NewIndex, Self>::type;
>+    New* const result = ::new (static_cast<void*>(&self->state_))
>+        New(absl::forward<Args>(args)...);
>+    self->index_ = NewIndex;
>+    return *result;
>+  }
>+
>+  template <class LeftVariant, class QualifiedRightVariant>
>+  struct InitFromVisitor {
>+    template <std::size_t NewIndex>
>+    void operator()(SizeT<NewIndex> /*new_i*/) const {
>+      using Alternative =
>+          typename variant_alternative<NewIndex, LeftVariant>::type;
>+      ::new (static_cast<void*>(&left->state_)) Alternative(
>+          Access<NewIndex>(std::forward<QualifiedRightVariant>(right)));
>+    }
>+
>+    void operator()(SizeT<absl::variant_npos> /*new_i*/) const {
>+      // This space intentionally left blank.
>+    }
>+    LeftVariant* left;
>+    QualifiedRightVariant&& right;
>+  };
>+};
>+
>+template <class Expected, class... T>
>+struct IndexOfImpl;
>+
>+template <class Expected>
>+struct IndexOfImpl<Expected> {
>+  using IndexFromEnd = SizeT<0>;
>+  using MatchedIndexFromEnd = IndexFromEnd;
>+  using MultipleMatches = std::false_type;
>+};
>+
>+template <class Expected, class Head, class... Tail>
>+struct IndexOfImpl<Expected, Head, Tail...> : IndexOfImpl<Expected, Tail...> {
>+  using IndexFromEnd =
>+      SizeT<IndexOfImpl<Expected, Tail...>::IndexFromEnd::value + 1>;
>+};
>+
>+template <class Expected, class... Tail>
>+struct IndexOfImpl<Expected, Expected, Tail...>
>+    : IndexOfImpl<Expected, Tail...> {
>+  using IndexFromEnd =
>+      SizeT<IndexOfImpl<Expected, Tail...>::IndexFromEnd::value + 1>;
>+  using MatchedIndexFromEnd = IndexFromEnd;
>+  using MultipleMatches = std::integral_constant<
>+      bool, IndexOfImpl<Expected, Tail...>::MatchedIndexFromEnd::value != 0>;
>+};
>+
>+template <class Expected, class... Types>
>+struct IndexOfMeta {
>+  using Results = IndexOfImpl<Expected, Types...>;
>+  static_assert(!Results::MultipleMatches::value,
>+                "Attempted to access a variant by specifying a type that "
>+                "matches more than one alternative.");
>+  static_assert(Results::MatchedIndexFromEnd::value != 0,
>+                "Attempted to access a variant by specifying a type that does "
>+                "not match any alternative.");
>+  using type = SizeT<sizeof...(Types) - Results::MatchedIndexFromEnd::value>;
>+};
>+
>+template <class Expected, class... Types>
>+using IndexOf = typename IndexOfMeta<Expected, Types...>::type;
>+
>+template <class Variant, class T, std::size_t CurrIndex>
>+struct UnambiguousIndexOfImpl;
>+
>+// Terminating case encountered once we've checked all of the alternatives
>+template <class T, std::size_t CurrIndex>
>+struct UnambiguousIndexOfImpl<variant<>, T, CurrIndex> : SizeT<CurrIndex> {};
>+
>+// Case where T is not Head
>+template <class Head, class... Tail, class T, std::size_t CurrIndex>
>+struct UnambiguousIndexOfImpl<variant<Head, Tail...>, T, CurrIndex>
>+    : UnambiguousIndexOfImpl<variant<Tail...>, T, CurrIndex + 1>::type {};
>+
>+// Case where T is Head
>+template <class Head, class... Tail, std::size_t CurrIndex>
>+struct UnambiguousIndexOfImpl<variant<Head, Tail...>, Head, CurrIndex>
>+    : SizeT<UnambiguousIndexOfImpl<variant<Tail...>, Head, 0>::value ==
>+                    sizeof...(Tail)
>+                ? CurrIndex
>+                : CurrIndex + sizeof...(Tail) + 1> {};
>+
>+template <class Variant, class T>
>+struct UnambiguousIndexOf;
>+
>+struct NoMatch {
>+  struct type {};
>+};
>+
>+template <class... Alts, class T>
>+struct UnambiguousIndexOf<variant<Alts...>, T>
>+    : std::conditional<UnambiguousIndexOfImpl<variant<Alts...>, T, 0>::value !=
>+                           sizeof...(Alts),
>+                       UnambiguousIndexOfImpl<variant<Alts...>, T, 0>,
>+                       NoMatch>::type::type {};
>+
>+template <class T, std::size_t /*Dummy*/>
>+using UnambiguousTypeOfImpl = T;
>+
>+template <class Variant, class T>
>+using UnambiguousTypeOfT =
>+    UnambiguousTypeOfImpl<T, UnambiguousIndexOf<Variant, T>::value>;
>+
>+template <class H, class... T>
>+class VariantStateBase;
>+
>+// This is an implementation of the "imaginary function" that is described in
>+// [variant.ctor]
>+// It is used in order to determine which alternative to construct during
>+// initialization from some type T.
>+template <class Variant, std::size_t I = 0>
>+struct ImaginaryFun;
>+
>+template <std::size_t I>
>+struct ImaginaryFun<variant<>, I> {
>+  static void Run() = delete;
>+};
>+
>+template <class H, class... T, std::size_t I>
>+struct ImaginaryFun<variant<H, T...>, I> : ImaginaryFun<variant<T...>, I + 1> {
>+  using ImaginaryFun<variant<T...>, I + 1>::Run;
>+
>+  // NOTE: const& and && are used instead of by-value due to lack of guaranteed
>+  // move elision of C++17. This may have other minor differences, but tests
>+  // pass.
>+  static SizeT<I> Run(const H&);
>+  static SizeT<I> Run(H&&);
>+};
>+
>+// The following metafunctions are used in constructor and assignment
>+// constraints.
>+template <class Self, class T>
>+struct IsNeitherSelfNorInPlace : std::true_type {};
>+
>+template <class Self>
>+struct IsNeitherSelfNorInPlace<Self, Self> : std::false_type {};
>+
>+template <class Self, class T>
>+struct IsNeitherSelfNorInPlace<Self, in_place_type_t<T>> : std::false_type {};
>+
>+template <class Self, std::size_t I>
>+struct IsNeitherSelfNorInPlace<Self, in_place_index_t<I>> : std::false_type {};
>+
>+template <class Variant, class T, class = void>
>+struct ConversionIsPossibleImpl : std::false_type {};
>+
>+template <class Variant, class T>
>+struct ConversionIsPossibleImpl<
>+    Variant, T, void_t<decltype(ImaginaryFun<Variant>::Run(std::declval<T>()))>>
>+    : std::true_type {};
>+
>+template <class Variant, class T>
>+struct ConversionIsPossible : ConversionIsPossibleImpl<Variant, T>::type {};
>+
>+template <class Variant, class T>
>+struct IndexOfConstructedType<
>+    Variant, T, void_t<decltype(ImaginaryFun<Variant>::Run(std::declval<T>()))>>
>+    : decltype(ImaginaryFun<Variant>::Run(std::declval<T>())) {};
>+
>+template <std::size_t... Is>
>+struct ContainsVariantNPos
>+    : absl::negation<std::is_same<  // NOLINT
>+          absl::integer_sequence<bool, 0 <= Is...>,
>+          absl::integer_sequence<bool, Is != absl::variant_npos...>>> {};
>+
>+template <class Op, class... QualifiedVariants>
>+using RawVisitResult =
>+    absl::result_of_t<Op(VariantAccessResult<0, QualifiedVariants>...)>;
>+
>+// NOTE: The spec requires that all return-paths yield the same type and is not
>+// SFINAE-friendly, so we can deduce the return type by examining the first
>+// result. If it's not callable, then we get an error, but are compliant and
>+// fast to compile.
>+// TODO(calabrese) Possibly rewrite in a way that yields better compile errors
>+// at the cost of longer compile-times.
>+template <class Op, class... QualifiedVariants>
>+struct VisitResultImpl {
>+  using type =
>+      absl::result_of_t<Op(VariantAccessResult<0, QualifiedVariants>...)>;
>+};
>+
>+// Done in two steps intentionally so that we don't cause substitution to fail.
>+template <class Op, class... QualifiedVariants>
>+using VisitResult = typename VisitResultImpl<Op, QualifiedVariants...>::type;
>+
>+template <class Op, class... QualifiedVariants>
>+struct PerformVisitation {
>+  using ReturnType = VisitResult<Op, QualifiedVariants...>;
>+
>+  template <std::size_t... Is>
>+  constexpr ReturnType operator()(SizeT<Is>... indices) const {
>+    return Run(typename ContainsVariantNPos<Is...>::type{},
>+               absl::index_sequence_for<QualifiedVariants...>(), indices...);
>+  }
>+
>+  template <std::size_t... TupIs, std::size_t... Is>
>+  constexpr ReturnType Run(std::false_type /*has_valueless*/,
>+                           index_sequence<TupIs...>, SizeT<Is>...) const {
>+    return absl::base_internal::Invoke(
>+        absl::forward<Op>(op),
>+        VariantCoreAccess::Access<Is>(
>+            absl::forward<QualifiedVariants>(std::get<TupIs>(variant_tup)))...);
>+  }
>+
>+  template <std::size_t... TupIs, std::size_t... Is>
>+  [[noreturn]] ReturnType Run(std::true_type /*has_valueless*/,
>+                              index_sequence<TupIs...>, SizeT<Is>...) const {
>+    absl::variant_internal::ThrowBadVariantAccess();
>+  }
>+
>+  // TODO(calabrese) Avoid using a tuple, which causes lots of instantiations
>+  // Attempts using lambda variadic captures fail on current GCC.
>+  std::tuple<QualifiedVariants&&...> variant_tup;
>+  Op&& op;
>+};
>+
>+template <class... T>
>+union Union;
>+
>+// We want to allow for variant<> to be trivial. For that, we need the default
>+// constructor to be trivial, which means we can't define it ourselves.
>+// Instead, we use a non-default constructor that takes NoopConstructorTag
>+// that doesn't affect the triviality of the types.
>+struct NoopConstructorTag {};
>+
>+template <std::size_t I>
>+struct EmplaceTag {};
>+
>+template <>
>+union Union<> {
>+  constexpr explicit Union(NoopConstructorTag) noexcept {}
>+};
>+
>+// Suppress bogus warning on MSVC: MSVC complains that Union<T...> has a defined
>+// deleted destructor from the `std::is_destructible` check below.
>+#ifdef _MSC_VER
>+#pragma warning(push)
>+#pragma warning(disable : 4624)
>+#endif  // _MSC_VER
>+
>+template <class Head, class... Tail>
>+union Union<Head, Tail...> {
>+  using TailUnion = Union<Tail...>;
>+
>+  explicit constexpr Union(NoopConstructorTag /*tag*/) noexcept
>+      : tail(NoopConstructorTag()) {}
>+
>+  template <class... P>
>+  explicit constexpr Union(EmplaceTag<0>, P&&... args)
>+      : head(absl::forward<P>(args)...) {}
>+
>+  template <std::size_t I, class... P>
>+  explicit constexpr Union(EmplaceTag<I>, P&&... args)
>+      : tail(EmplaceTag<I - 1>{}, absl::forward<P>(args)...) {}
>+
>+  Head head;
>+  TailUnion tail;
>+};
>+
>+#ifdef _MSC_VER
>+#pragma warning(pop)
>+#endif  // _MSC_VER
>+
>+// TODO(calabrese) Just contain a Union in this union (certain configs fail).
>+template <class... T>
>+union DestructibleUnionImpl;
>+
>+template <>
>+union DestructibleUnionImpl<> {
>+  constexpr explicit DestructibleUnionImpl(NoopConstructorTag) noexcept {}
>+};
>+
>+template <class Head, class... Tail>
>+union DestructibleUnionImpl<Head, Tail...> {
>+  using TailUnion = DestructibleUnionImpl<Tail...>;
>+
>+  explicit constexpr DestructibleUnionImpl(NoopConstructorTag /*tag*/) noexcept
>+      : tail(NoopConstructorTag()) {}
>+
>+  template <class... P>
>+  explicit constexpr DestructibleUnionImpl(EmplaceTag<0>, P&&... args)
>+      : head(absl::forward<P>(args)...) {}
>+
>+  template <std::size_t I, class... P>
>+  explicit constexpr DestructibleUnionImpl(EmplaceTag<I>, P&&... args)
>+      : tail(EmplaceTag<I - 1>{}, absl::forward<P>(args)...) {}
>+
>+  ~DestructibleUnionImpl() {}
>+
>+  Head head;
>+  TailUnion tail;
>+};
>+
>+// This union type is destructible even if one or more T are not trivially
>+// destructible. In the case that all T are trivially destructible, then so is
>+// this resultant type.
>+template <class... T>
>+using DestructibleUnion =
>+    absl::conditional_t<std::is_destructible<Union<T...>>::value, Union<T...>,
>+                        DestructibleUnionImpl<T...>>;
>+
>+// Deepest base, containing the actual union and the discriminator
>+template <class H, class... T>
>+class VariantStateBase {
>+ protected:
>+  using Variant = variant<H, T...>;
>+
>+  template <class LazyH = H,
>+            class ConstructibleH = absl::enable_if_t<
>+                std::is_default_constructible<LazyH>::value, LazyH>>
>+  constexpr VariantStateBase() noexcept(
>+      std::is_nothrow_default_constructible<ConstructibleH>::value)
>+      : state_(EmplaceTag<0>()), index_(0) {}
>+
>+  template <std::size_t I, class... P>
>+  explicit constexpr VariantStateBase(EmplaceTag<I> tag, P&&... args)
>+      : state_(tag, absl::forward<P>(args)...), index_(I) {}
>+
>+  explicit constexpr VariantStateBase(NoopConstructorTag)
>+      : state_(NoopConstructorTag()), index_(variant_npos) {}
>+
>+  void destroy() {}  // Does nothing (shadowed in child if non-trivial)
>+
>+  DestructibleUnion<H, T...> state_;
>+  std::size_t index_;
>+};
>+
>+using absl::internal::identity;
>+
>+// OverloadSet::Overload() is a unary function which is overloaded to
>+// take any of the element types of the variant, by reference-to-const.
>+// The return type of the overload on T is identity<T>, so that you
>+// can statically determine which overload was called.
>+//
>+// Overload() is not defined, so it can only be called in unevaluated
>+// contexts.
>+template <typename... Ts>
>+struct OverloadSet;
>+
>+template <typename T, typename... Ts>
>+struct OverloadSet<T, Ts...> : OverloadSet<Ts...> {
>+  using Base = OverloadSet<Ts...>;
>+  static identity<T> Overload(const T&);
>+  using Base::Overload;
>+};
>+
>+template <>
>+struct OverloadSet<> {
>+  // For any case not handled above.
>+  static void Overload(...);
>+};
>+
>+template <class T>
>+using LessThanResult = decltype(std::declval<T>() < std::declval<T>());
>+
>+template <class T>
>+using GreaterThanResult = decltype(std::declval<T>() > std::declval<T>());
>+
>+template <class T>
>+using LessThanOrEqualResult = decltype(std::declval<T>() <= std::declval<T>());
>+
>+template <class T>
>+using GreaterThanOrEqualResult =
>+    decltype(std::declval<T>() >= std::declval<T>());
>+
>+template <class T>
>+using EqualResult = decltype(std::declval<T>() == std::declval<T>());
>+
>+template <class T>
>+using NotEqualResult = decltype(std::declval<T>() != std::declval<T>());
>+
>+using type_traits_internal::is_detected_convertible;
>+
>+template <class... T>
>+using RequireAllHaveEqualT = absl::enable_if_t<
>+    absl::conjunction<is_detected_convertible<bool, EqualResult, T>...>::value,
>+    bool>;
>+
>+template <class... T>
>+using RequireAllHaveNotEqualT =
>+    absl::enable_if_t<absl::conjunction<is_detected_convertible<
>+                          bool, NotEqualResult, T>...>::value,
>+                      bool>;
>+
>+template <class... T>
>+using RequireAllHaveLessThanT =
>+    absl::enable_if_t<absl::conjunction<is_detected_convertible<
>+                          bool, LessThanResult, T>...>::value,
>+                      bool>;
>+
>+template <class... T>
>+using RequireAllHaveLessThanOrEqualT =
>+    absl::enable_if_t<absl::conjunction<is_detected_convertible<
>+                          bool, LessThanOrEqualResult, T>...>::value,
>+                      bool>;
>+
>+template <class... T>
>+using RequireAllHaveGreaterThanOrEqualT =
>+    absl::enable_if_t<absl::conjunction<is_detected_convertible<
>+                          bool, GreaterThanOrEqualResult, T>...>::value,
>+                      bool>;
>+
>+template <class... T>
>+using RequireAllHaveGreaterThanT =
>+    absl::enable_if_t<absl::conjunction<is_detected_convertible<
>+                          bool, GreaterThanResult, T>...>::value,
>+                      bool>;
>+
>+// Helper template containing implementations details of variant that can't go
>+// in the private section. For convenience, this takes the variant type as a
>+// single template parameter.
>+template <typename T>
>+struct VariantHelper;
>+
>+template <typename... Ts>
>+struct VariantHelper<variant<Ts...>> {
>+  // Type metafunction which returns the element type selected if
>+  // OverloadSet::Overload() is well-formed when called with argument type U.
>+  template <typename U>
>+  using BestMatch = decltype(
>+      variant_internal::OverloadSet<Ts...>::Overload(std::declval<U>()));
>+
>+  // Type metafunction which returns true if OverloadSet::Overload() is
>+  // well-formed when called with argument type U.
>+  // CanAccept can't be just an alias because there is a MSVC bug on parameter
>+  // pack expansion involving decltype.
>+  template <typename U>
>+  struct CanAccept :
>+      std::integral_constant<bool, !std::is_void<BestMatch<U>>::value> {};
>+
>+  // Type metafunction which returns true if Other is an instantiation of
>+  // variant, and variants's converting constructor from Other will be
>+  // well-formed. We will use this to remove constructors that would be
>+  // ill-formed from the overload set.
>+  template <typename Other>
>+  struct CanConvertFrom;
>+
>+  template <typename... Us>
>+  struct CanConvertFrom<variant<Us...>>
>+      : public absl::conjunction<CanAccept<Us>...> {};
>+};
>+
>+// A type with nontrivial copy ctor and trivial move ctor.
>+struct TrivialMoveOnly {
>+  TrivialMoveOnly(TrivialMoveOnly&&) = default;
>+};
>+
>+// Trait class to detect whether a type is trivially move constructible.
>+// A union's defaulted copy/move constructor is deleted if any variant member's
>+// copy/move constructor is nontrivial.
>+template <typename T>
>+struct IsTriviallyMoveConstructible:
>+  std::is_move_constructible<Union<T, TrivialMoveOnly>> {};
>+
>+// To guarantee triviality of all special-member functions that can be trivial,
>+// we use a chain of conditional bases for each one.
>+// The order of inheritance of bases from child to base are logically:
>+//
>+// variant
>+// VariantCopyAssignBase
>+// VariantMoveAssignBase
>+// VariantCopyBase
>+// VariantMoveBase
>+// VariantStateBaseDestructor
>+// VariantStateBase
>+//
>+// Note that there is a separate branch at each base that is dependent on
>+// whether or not that corresponding special-member-function can be trivial in
>+// the resultant variant type.
>+
>+template <class... T>
>+class VariantStateBaseDestructorNontrivial;
>+
>+template <class... T>
>+class VariantMoveBaseNontrivial;
>+
>+template <class... T>
>+class VariantCopyBaseNontrivial;
>+
>+template <class... T>
>+class VariantMoveAssignBaseNontrivial;
>+
>+template <class... T>
>+class VariantCopyAssignBaseNontrivial;
>+
>+// Base that is dependent on whether or not the destructor can be trivial.
>+template <class... T>
>+using VariantStateBaseDestructor =
>+    absl::conditional_t<std::is_destructible<Union<T...>>::value,
>+                        VariantStateBase<T...>,
>+                        VariantStateBaseDestructorNontrivial<T...>>;
>+
>+// Base that is dependent on whether or not the move-constructor can be
>+// implicitly generated by the compiler (trivial or deleted).
>+// Previously we were using `std::is_move_constructible<Union<T...>>` to check
>+// whether all Ts have trivial move constructor, but it ran into a GCC bug:
>+// https://gcc.gnu.org/bugzilla/show_bug.cgi?id=84866
>+// So we have to use a different approach (i.e. `HasTrivialMoveConstructor`) to
>+// work around the bug.
>+template <class... T>
>+using VariantMoveBase = absl::conditional_t<
>+    absl::disjunction<
>+        absl::negation<absl::conjunction<std::is_move_constructible<T>...>>,
>+        absl::conjunction<IsTriviallyMoveConstructible<T>...>>::value,
>+    VariantStateBaseDestructor<T...>, VariantMoveBaseNontrivial<T...>>;
>+
>+// Base that is dependent on whether or not the copy-constructor can be trivial.
>+template <class... T>
>+using VariantCopyBase = absl::conditional_t<
>+    absl::disjunction<
>+        absl::negation<absl::conjunction<std::is_copy_constructible<T>...>>,
>+        std::is_copy_constructible<Union<T...>>>::value,
>+    VariantMoveBase<T...>, VariantCopyBaseNontrivial<T...>>;
>+
>+// Base that is dependent on whether or not the move-assign can be trivial.
>+template <class... T>
>+using VariantMoveAssignBase = absl::conditional_t<
>+    absl::disjunction<absl::conjunction<std::is_move_assignable<Union<T...>>,
>+                                        std::is_move_constructible<Union<T...>>,
>+                                        std::is_destructible<Union<T...>>>,
>+                      absl::negation<absl::conjunction<
>+                          std::is_move_constructible<T>...,
>+                          std::is_move_assignable<T>...>>>::value,
>+    VariantCopyBase<T...>, VariantMoveAssignBaseNontrivial<T...>>;
>+
>+// Base that is dependent on whether or not the copy-assign can be trivial.
>+template <class... T>
>+using VariantCopyAssignBase = absl::conditional_t<
>+    absl::disjunction<absl::conjunction<std::is_copy_assignable<Union<T...>>,
>+                                        std::is_copy_constructible<Union<T...>>,
>+                                        std::is_destructible<Union<T...>>>,
>+                      absl::negation<absl::conjunction<
>+                          std::is_copy_constructible<T>...,
>+                          std::is_copy_assignable<T>...>>>::value,
>+    VariantMoveAssignBase<T...>, VariantCopyAssignBaseNontrivial<T...>>;
>+
>+template <class... T>
>+using VariantBase = VariantCopyAssignBase<T...>;
>+
>+template <class... T>
>+class VariantStateBaseDestructorNontrivial : protected VariantStateBase<T...> {
>+ private:
>+  using Base = VariantStateBase<T...>;
>+
>+ protected:
>+  using Base::Base;
>+
>+  VariantStateBaseDestructorNontrivial() = default;
>+  VariantStateBaseDestructorNontrivial(VariantStateBaseDestructorNontrivial&&) =
>+      default;
>+  VariantStateBaseDestructorNontrivial(
>+      const VariantStateBaseDestructorNontrivial&) = default;
>+  VariantStateBaseDestructorNontrivial& operator=(
>+      VariantStateBaseDestructorNontrivial&&) = default;
>+  VariantStateBaseDestructorNontrivial& operator=(
>+      const VariantStateBaseDestructorNontrivial&) = default;
>+
>+  struct Destroyer {
>+    template <std::size_t I>
>+    void operator()(SizeT<I> i) const {
>+      using Alternative =
>+          typename absl::variant_alternative<I, variant<T...>>::type;
>+      variant_internal::AccessUnion(self->state_, i).~Alternative();
>+    }
>+
>+    void operator()(SizeT<absl::variant_npos> /*i*/) const {
>+      // This space intentionally left blank
>+    }
>+
>+    VariantStateBaseDestructorNontrivial* self;
>+  };
>+
>+  void destroy() { VisitIndices<sizeof...(T)>::Run(Destroyer{this}, index_); }
>+
>+  ~VariantStateBaseDestructorNontrivial() { destroy(); }
>+
>+ protected:
>+  using Base::index_;
>+  using Base::state_;
>+};
>+
>+template <class... T>
>+class VariantMoveBaseNontrivial : protected VariantStateBaseDestructor<T...> {
>+ private:
>+  using Base = VariantStateBaseDestructor<T...>;
>+
>+ protected:
>+  using Base::Base;
>+
>+  struct Construct {
>+    template <std::size_t I>
>+    void operator()(SizeT<I> i) const {
>+      using Alternative =
>+          typename absl::variant_alternative<I, variant<T...>>::type;
>+      ::new (static_cast<void*>(&self->state_)) Alternative(
>+          variant_internal::AccessUnion(absl::move(other->state_), i));
>+    }
>+
>+    void operator()(SizeT<absl::variant_npos> /*i*/) const {}
>+
>+    VariantMoveBaseNontrivial* self;
>+    VariantMoveBaseNontrivial* other;
>+  };
>+
>+  VariantMoveBaseNontrivial() = default;
>+  VariantMoveBaseNontrivial(VariantMoveBaseNontrivial&& other) noexcept(
>+      absl::conjunction<std::is_nothrow_move_constructible<T>...>::value)
>+      : Base(NoopConstructorTag()) {
>+    VisitIndices<sizeof...(T)>::Run(Construct{this, &other}, other.index_);
>+    index_ = other.index_;
>+  }
>+
>+  VariantMoveBaseNontrivial(VariantMoveBaseNontrivial const&) = default;
>+
>+  VariantMoveBaseNontrivial& operator=(VariantMoveBaseNontrivial&&) = default;
>+  VariantMoveBaseNontrivial& operator=(VariantMoveBaseNontrivial const&) =
>+      default;
>+
>+ protected:
>+  using Base::index_;
>+  using Base::state_;
>+};
>+
>+template <class... T>
>+class VariantCopyBaseNontrivial : protected VariantMoveBase<T...> {
>+ private:
>+  using Base = VariantMoveBase<T...>;
>+
>+ protected:
>+  using Base::Base;
>+
>+  VariantCopyBaseNontrivial() = default;
>+  VariantCopyBaseNontrivial(VariantCopyBaseNontrivial&&) = default;
>+
>+  struct Construct {
>+    template <std::size_t I>
>+    void operator()(SizeT<I> i) const {
>+      using Alternative =
>+          typename absl::variant_alternative<I, variant<T...>>::type;
>+      ::new (static_cast<void*>(&self->state_))
>+          Alternative(variant_internal::AccessUnion(other->state_, i));
>+    }
>+
>+    void operator()(SizeT<absl::variant_npos> /*i*/) const {}
>+
>+    VariantCopyBaseNontrivial* self;
>+    const VariantCopyBaseNontrivial* other;
>+  };
>+
>+  VariantCopyBaseNontrivial(VariantCopyBaseNontrivial const& other)
>+      : Base(NoopConstructorTag()) {
>+    VisitIndices<sizeof...(T)>::Run(Construct{this, &other}, other.index_);
>+    index_ = other.index_;
>+  }
>+
>+  VariantCopyBaseNontrivial& operator=(VariantCopyBaseNontrivial&&) = default;
>+  VariantCopyBaseNontrivial& operator=(VariantCopyBaseNontrivial const&) =
>+      default;
>+
>+ protected:
>+  using Base::index_;
>+  using Base::state_;
>+};
>+
>+template <class... T>
>+class VariantMoveAssignBaseNontrivial : protected VariantCopyBase<T...> {
>+  friend struct VariantCoreAccess;
>+
>+ private:
>+  using Base = VariantCopyBase<T...>;
>+
>+ protected:
>+  using Base::Base;
>+
>+  VariantMoveAssignBaseNontrivial() = default;
>+  VariantMoveAssignBaseNontrivial(VariantMoveAssignBaseNontrivial&&) = default;
>+  VariantMoveAssignBaseNontrivial(const VariantMoveAssignBaseNontrivial&) =
>+      default;
>+  VariantMoveAssignBaseNontrivial& operator=(
>+      VariantMoveAssignBaseNontrivial const&) = default;
>+
>+    VariantMoveAssignBaseNontrivial&
>+    operator=(VariantMoveAssignBaseNontrivial&& other) noexcept(
>+        absl::conjunction<std::is_nothrow_move_constructible<T>...,
>+                          std::is_nothrow_move_assignable<T>...>::value) {
>+      VisitIndices<sizeof...(T)>::Run(
>+          VariantCoreAccess::MakeMoveAssignVisitor(this, &other), other.index_);
>+      return *this;
>+    }
>+
>+ protected:
>+  using Base::index_;
>+  using Base::state_;
>+};
>+
>+template <class... T>
>+class VariantCopyAssignBaseNontrivial : protected VariantMoveAssignBase<T...> {
>+  friend struct VariantCoreAccess;
>+
>+ private:
>+  using Base = VariantMoveAssignBase<T...>;
>+
>+ protected:
>+  using Base::Base;
>+
>+  VariantCopyAssignBaseNontrivial() = default;
>+  VariantCopyAssignBaseNontrivial(VariantCopyAssignBaseNontrivial&&) = default;
>+  VariantCopyAssignBaseNontrivial(const VariantCopyAssignBaseNontrivial&) =
>+      default;
>+  VariantCopyAssignBaseNontrivial& operator=(
>+      VariantCopyAssignBaseNontrivial&&) = default;
>+
>+    VariantCopyAssignBaseNontrivial& operator=(
>+        const VariantCopyAssignBaseNontrivial& other) {
>+      VisitIndices<sizeof...(T)>::Run(
>+          VariantCoreAccess::MakeCopyAssignVisitor(this, other), other.index_);
>+      return *this;
>+    }
>+
>+ protected:
>+  using Base::index_;
>+  using Base::state_;
>+};
>+
>+////////////////////////////////////////
>+// Visitors for Comparison Operations //
>+////////////////////////////////////////
>+
>+template <class... Types>
>+struct EqualsOp {
>+  const variant<Types...>* v;
>+  const variant<Types...>* w;
>+
>+  constexpr bool operator()(SizeT<absl::variant_npos> /*v_i*/) const {
>+    return true;
>+  }
>+
>+  template <std::size_t I>
>+  constexpr bool operator()(SizeT<I> /*v_i*/) const {
>+    return VariantCoreAccess::Access<I>(*v) == VariantCoreAccess::Access<I>(*w);
>+  }
>+};
>+
>+template <class... Types>
>+struct NotEqualsOp {
>+  const variant<Types...>* v;
>+  const variant<Types...>* w;
>+
>+  constexpr bool operator()(SizeT<absl::variant_npos> /*v_i*/) const {
>+    return false;
>+  }
>+
>+  template <std::size_t I>
>+  constexpr bool operator()(SizeT<I> /*v_i*/) const {
>+    return VariantCoreAccess::Access<I>(*v) != VariantCoreAccess::Access<I>(*w);
>+  }
>+};
>+
>+template <class... Types>
>+struct LessThanOp {
>+  const variant<Types...>* v;
>+  const variant<Types...>* w;
>+
>+  constexpr bool operator()(SizeT<absl::variant_npos> /*v_i*/) const {
>+    return false;
>+  }
>+
>+  template <std::size_t I>
>+  constexpr bool operator()(SizeT<I> /*v_i*/) const {
>+    return VariantCoreAccess::Access<I>(*v) < VariantCoreAccess::Access<I>(*w);
>+  }
>+};
>+
>+template <class... Types>
>+struct GreaterThanOp {
>+  const variant<Types...>* v;
>+  const variant<Types...>* w;
>+
>+  constexpr bool operator()(SizeT<absl::variant_npos> /*v_i*/) const {
>+    return false;
>+  }
>+
>+  template <std::size_t I>
>+  constexpr bool operator()(SizeT<I> /*v_i*/) const {
>+    return VariantCoreAccess::Access<I>(*v) > VariantCoreAccess::Access<I>(*w);
>+  }
>+};
>+
>+template <class... Types>
>+struct LessThanOrEqualsOp {
>+  const variant<Types...>* v;
>+  const variant<Types...>* w;
>+
>+  constexpr bool operator()(SizeT<absl::variant_npos> /*v_i*/) const {
>+    return true;
>+  }
>+
>+  template <std::size_t I>
>+  constexpr bool operator()(SizeT<I> /*v_i*/) const {
>+    return VariantCoreAccess::Access<I>(*v) <= VariantCoreAccess::Access<I>(*w);
>+  }
>+};
>+
>+template <class... Types>
>+struct GreaterThanOrEqualsOp {
>+  const variant<Types...>* v;
>+  const variant<Types...>* w;
>+
>+  constexpr bool operator()(SizeT<absl::variant_npos> /*v_i*/) const {
>+    return true;
>+  }
>+
>+  template <std::size_t I>
>+  constexpr bool operator()(SizeT<I> /*v_i*/) const {
>+    return VariantCoreAccess::Access<I>(*v) >= VariantCoreAccess::Access<I>(*w);
>+  }
>+};
>+
>+// Precondition: v.index() == w.index();
>+template <class... Types>
>+struct SwapSameIndex {
>+  variant<Types...>* v;
>+  variant<Types...>* w;
>+  template <std::size_t I>
>+  void operator()(SizeT<I>) const {
>+    using std::swap;
>+    swap(VariantCoreAccess::Access<I>(*v), VariantCoreAccess::Access<I>(*w));
>+  }
>+
>+  void operator()(SizeT<variant_npos>) const {}
>+};
>+
>+// TODO(calabrese) do this from a different namespace for proper adl usage
>+template <class... Types>
>+struct Swap {
>+  variant<Types...>* v;
>+  variant<Types...>* w;
>+
>+  void generic_swap() const {
>+    variant<Types...> tmp(std::move(*w));
>+    VariantCoreAccess::Destroy(*w);
>+    VariantCoreAccess::InitFrom(*w, std::move(*v));
>+    VariantCoreAccess::Destroy(*v);
>+    VariantCoreAccess::InitFrom(*v, std::move(tmp));
>+  }
>+
>+  void operator()(SizeT<absl::variant_npos> /*w_i*/) const {
>+    if (!v->valueless_by_exception()) {
>+      generic_swap();
>+    }
>+  }
>+
>+  template <std::size_t Wi>
>+  void operator()(SizeT<Wi> /*w_i*/) {
>+    if (v->index() == Wi) {
>+      VisitIndices<sizeof...(Types)>::Run(SwapSameIndex<Types...>{v, w}, Wi);
>+    } else {
>+      generic_swap();
>+    }
>+  }
>+};
>+
>+template <typename Variant, typename = void, typename... Ts>
>+struct VariantHashBase {
>+  VariantHashBase() = delete;
>+  VariantHashBase(const VariantHashBase&) = delete;
>+  VariantHashBase(VariantHashBase&&) = delete;
>+  VariantHashBase& operator=(const VariantHashBase&) = delete;
>+  VariantHashBase& operator=(VariantHashBase&&) = delete;
>+};
>+
>+struct VariantHashVisitor {
>+  template <typename T>
>+  size_t operator()(const T& t) {
>+    return std::hash<T>{}(t);
>+  }
>+};
>+
>+template <typename Variant, typename... Ts>
>+struct VariantHashBase<Variant,
>+                       absl::enable_if_t<absl::conjunction<
>+                           type_traits_internal::IsHashEnabled<Ts>...>::value>,
>+                       Ts...> {
>+  using argument_type = Variant;
>+  using result_type = size_t;
>+  size_t operator()(const Variant& var) const {
>+    if (var.valueless_by_exception()) {
>+      return 239799884;
>+    }
>+    size_t result = VisitIndices<variant_size<Variant>::value>::Run(
>+        PerformVisitation<VariantHashVisitor, const Variant&>{
>+            std::forward_as_tuple(var), VariantHashVisitor{}},
>+        var.index());
>+    // Combine the index and the hash result in order to distinguish
>+    // std::variant<int, int> holding the same value as different alternative.
>+    return result ^ var.index();
>+  }
>+};
>+
>+}  // namespace variant_internal
>+}  // namespace absl
>+
>+#endif  // ABSL_TYPES_variant_internal_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/optional.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/optional.cc
>new file mode 100644
>index 00000000000..ef272904199
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/optional.cc
>@@ -0,0 +1,24 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/types/optional.h"
>+
>+#ifndef ABSL_HAVE_STD_OPTIONAL
>+namespace absl {
>+
>+nullopt_t::init_t nullopt_t::init;
>+extern const nullopt_t nullopt{nullopt_t::init};
>+
>+}  // namespace absl
>+#endif  // ABSL_HAVE_STD_OPTIONAL
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/optional.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/optional.h
>new file mode 100644
>index 00000000000..c837cddeef4
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/optional.h
>@@ -0,0 +1,1141 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// -----------------------------------------------------------------------------
>+// optional.h
>+// -----------------------------------------------------------------------------
>+//
>+// This header file defines the `absl::optional` type for holding a value which
>+// may or may not be present. This type is useful for providing value semantics
>+// for operations that may either wish to return or hold "something-or-nothing".
>+//
>+// Example:
>+//
>+//   // A common way to signal operation failure is to provide an output
>+//   // parameter and a bool return type:
>+//   bool AcquireResource(const Input&, Resource * out);
>+//
>+//   // Providing an absl::optional return type provides a cleaner API:
>+//   absl::optional<Resource> AcquireResource(const Input&);
>+//
>+// `absl::optional` is a C++11 compatible version of the C++17 `std::optional`
>+// abstraction and is designed to be a drop-in replacement for code compliant
>+// with C++17.
>+#ifndef ABSL_TYPES_OPTIONAL_H_
>+#define ABSL_TYPES_OPTIONAL_H_
>+
>+#include "absl/base/config.h"
>+#include "absl/utility/utility.h"
>+
>+#ifdef ABSL_HAVE_STD_OPTIONAL
>+
>+#include <optional>
>+
>+namespace absl {
>+using std::bad_optional_access;
>+using std::optional;
>+using std::make_optional;
>+using std::nullopt_t;
>+using std::nullopt;
>+}  // namespace absl
>+
>+#else  // ABSL_HAVE_STD_OPTIONAL
>+
>+#include <cassert>
>+#include <functional>
>+#include <initializer_list>
>+#include <new>
>+#include <type_traits>
>+#include <utility>
>+
>+#include "absl/memory/memory.h"
>+#include "absl/meta/type_traits.h"
>+#include "absl/types/bad_optional_access.h"
>+
>+// ABSL_OPTIONAL_USE_INHERITING_CONSTRUCTORS
>+//
>+// Inheriting constructors is supported in GCC 4.8+, Clang 3.3+ and MSVC 2015.
>+// __cpp_inheriting_constructors is a predefined macro and a recommended way to
>+// check for this language feature, but GCC doesn't support it until 5.0 and
>+// Clang doesn't support it until 3.6.
>+// Also, MSVC 2015 has a bug: it doesn't inherit the constexpr template
>+// constructor. For example, the following code won't work on MSVC 2015 Update3:
>+// struct Base {
>+//   int t;
>+//   template <typename T>
>+//   constexpr Base(T t_) : t(t_) {}
>+// };
>+// struct Foo : Base {
>+//   using Base::Base;
>+// }
>+// constexpr Foo foo(0);  // doesn't work on MSVC 2015
>+#if defined(__clang__)
>+#if __has_feature(cxx_inheriting_constructors)
>+#define ABSL_OPTIONAL_USE_INHERITING_CONSTRUCTORS 1
>+#endif
>+#elif (defined(__GNUC__) &&                                       \
>+       (__GNUC__ > 4 || __GNUC__ == 4 && __GNUC_MINOR__ >= 8)) || \
>+    (__cpp_inheriting_constructors >= 200802) ||                  \
>+    (defined(_MSC_VER) && _MSC_VER >= 1910)
>+#define ABSL_OPTIONAL_USE_INHERITING_CONSTRUCTORS 1
>+#endif
>+
>+namespace absl {
>+
>+// -----------------------------------------------------------------------------
>+// absl::optional
>+// -----------------------------------------------------------------------------
>+//
>+// A value of type `absl::optional<T>` holds either a value of `T` or an
>+// "empty" value.  When it holds a value of `T`, it stores it as a direct
>+// sub-object, so `sizeof(optional<T>)` is approximately
>+// `sizeof(T) + sizeof(bool)`.
>+//
>+// This implementation is based on the specification in the latest draft of the
>+// C++17 `std::optional` specification as of May 2017, section 20.6.
>+//
>+// Differences between `absl::optional<T>` and `std::optional<T>` include:
>+//
>+//    * `constexpr` is not used for non-const member functions.
>+//      (dependency on some differences between C++11 and C++14.)
>+//    * `absl::nullopt` and `absl::in_place` are not declared `constexpr`. We
>+//      need the inline variable support in C++17 for external linkage.
>+//    * Throws `absl::bad_optional_access` instead of
>+//      `std::bad_optional_access`.
>+//    * `optional::swap()` and `absl::swap()` relies on
>+//      `std::is_(nothrow_)swappable()`, which has been introduced in C++17.
>+//      As a workaround, we assume `is_swappable()` is always `true`
>+//      and `is_nothrow_swappable()` is the same as `std::is_trivial()`.
>+//    * `make_optional()` cannot be declared `constexpr` due to the absence of
>+//      guaranteed copy elision.
>+//    * The move constructor's `noexcept` specification is stronger, i.e. if the
>+//      default allocator is non-throwing (via setting
>+//      `ABSL_ALLOCATOR_NOTHROW`), it evaluates to `noexcept(true)`, because
>+//      we assume
>+//       a) move constructors should only throw due to allocation failure and
>+//       b) if T's move constructor allocates, it uses the same allocation
>+//          function as the default allocator.
>+template <typename T>
>+class optional;
>+
>+// nullopt_t
>+//
>+// Class type for `absl::nullopt` used to indicate an `absl::optional<T>` type
>+// that does not contain a value.
>+struct nullopt_t {
>+  struct init_t {};
>+  static init_t init;
>+
>+  // It must not be default-constructible to avoid ambiguity for opt = {}.
>+  // Note the non-const reference, which is to eliminate ambiguity for code
>+  // like:
>+  //
>+  // struct S { int value; };
>+  //
>+  // void Test() {
>+  //   optional<S> opt;
>+  //   opt = {{}};
>+  // }
>+  explicit constexpr nullopt_t(init_t& /*unused*/) {}
>+};
>+
>+// nullopt
>+//
>+// A tag constant of type `absl::nullopt_t` used to indicate an empty
>+// `absl::optional` in certain functions, such as construction or assignment.
>+extern const nullopt_t nullopt;
>+
>+namespace optional_internal {
>+
>+struct empty_struct {};
>+// This class stores the data in optional<T>.
>+// It is specialized based on whether T is trivially destructible.
>+// This is the specialization for non trivially destructible type.
>+template <typename T, bool = std::is_trivially_destructible<T>::value>
>+class optional_data_dtor_base {
>+  struct dummy_type {
>+    static_assert(sizeof(T) % sizeof(empty_struct) == 0, "");
>+    // Use an array to avoid GCC 6 placement-new warning.
>+    empty_struct data[sizeof(T) / sizeof(empty_struct)];
>+  };
>+
>+ protected:
>+  // Whether there is data or not.
>+  bool engaged_;
>+  // Data storage
>+  union {
>+    dummy_type dummy_;
>+    T data_;
>+  };
>+
>+  void destruct() noexcept {
>+    if (engaged_) {
>+      data_.~T();
>+      engaged_ = false;
>+    }
>+  }
>+
>+  // dummy_ must be initialized for constexpr constructor.
>+  constexpr optional_data_dtor_base() noexcept : engaged_(false), dummy_{{}} {}
>+
>+  template <typename... Args>
>+  constexpr explicit optional_data_dtor_base(in_place_t, Args&&... args)
>+      : engaged_(true), data_(absl::forward<Args>(args)...) {}
>+
>+  ~optional_data_dtor_base() { destruct(); }
>+};
>+
>+// Specialization for trivially destructible type.
>+template <typename T>
>+class optional_data_dtor_base<T, true> {
>+  struct dummy_type {
>+    static_assert(sizeof(T) % sizeof(empty_struct) == 0, "");
>+    // Use array to avoid GCC 6 placement-new warning.
>+    empty_struct data[sizeof(T) / sizeof(empty_struct)];
>+  };
>+
>+ protected:
>+  // Whether there is data or not.
>+  bool engaged_;
>+  // Data storage
>+  union {
>+    dummy_type dummy_;
>+    T data_;
>+  };
>+  void destruct() noexcept { engaged_ = false; }
>+
>+  // dummy_ must be initialized for constexpr constructor.
>+  constexpr optional_data_dtor_base() noexcept : engaged_(false), dummy_{{}} {}
>+
>+  template <typename... Args>
>+  constexpr explicit optional_data_dtor_base(in_place_t, Args&&... args)
>+      : engaged_(true), data_(absl::forward<Args>(args)...) {}
>+};
>+
>+template <typename T>
>+class optional_data_base : public optional_data_dtor_base<T> {
>+ protected:
>+  using base = optional_data_dtor_base<T>;
>+#if ABSL_OPTIONAL_USE_INHERITING_CONSTRUCTORS
>+  using base::base;
>+#else
>+  optional_data_base() = default;
>+
>+  template <typename... Args>
>+  constexpr explicit optional_data_base(in_place_t t, Args&&... args)
>+      : base(t, absl::forward<Args>(args)...) {}
>+#endif
>+
>+  template <typename... Args>
>+  void construct(Args&&... args) {
>+    // Use dummy_'s address to work around casting cv-qualified T* to void*.
>+    ::new (static_cast<void*>(&this->dummy_)) T(std::forward<Args>(args)...);
>+    this->engaged_ = true;
>+  }
>+
>+  template <typename U>
>+  void assign(U&& u) {
>+    if (this->engaged_) {
>+      this->data_ = std::forward<U>(u);
>+    } else {
>+      construct(std::forward<U>(u));
>+    }
>+  }
>+};
>+
>+// TODO(absl-team): Add another class using
>+// std::is_trivially_move_constructible trait when available to match
>+// http://cplusplus.github.io/LWG/lwg-defects.html#2900, for types that
>+// have trivial move but nontrivial copy.
>+// Also, we should be checking is_trivially_copyable here, which is not
>+// supported now, so we use is_trivially_* traits instead.
>+template <typename T, bool = absl::is_trivially_copy_constructible<T>::value&&
>+                          absl::is_trivially_copy_assignable<
>+                              typename std::remove_cv<T>::type>::value&&
>+                              std::is_trivially_destructible<T>::value>
>+class optional_data;
>+
>+// Trivially copyable types
>+template <typename T>
>+class optional_data<T, true> : public optional_data_base<T> {
>+ protected:
>+#if ABSL_OPTIONAL_USE_INHERITING_CONSTRUCTORS
>+  using optional_data_base<T>::optional_data_base;
>+#else
>+  optional_data() = default;
>+
>+  template <typename... Args>
>+  constexpr explicit optional_data(in_place_t t, Args&&... args)
>+      : optional_data_base<T>(t, absl::forward<Args>(args)...) {}
>+#endif
>+};
>+
>+template <typename T>
>+class optional_data<T, false> : public optional_data_base<T> {
>+ protected:
>+#if ABSL_OPTIONAL_USE_INHERITING_CONSTRUCTORS
>+  using optional_data_base<T>::optional_data_base;
>+#else
>+  template <typename... Args>
>+  constexpr explicit optional_data(in_place_t t, Args&&... args)
>+      : optional_data_base<T>(t, absl::forward<Args>(args)...) {}
>+#endif
>+
>+  optional_data() = default;
>+
>+  optional_data(const optional_data& rhs) {
>+    if (rhs.engaged_) {
>+      this->construct(rhs.data_);
>+    }
>+  }
>+
>+  optional_data(optional_data&& rhs) noexcept(
>+      absl::default_allocator_is_nothrow::value ||
>+      std::is_nothrow_move_constructible<T>::value) {
>+    if (rhs.engaged_) {
>+      this->construct(std::move(rhs.data_));
>+    }
>+  }
>+
>+  optional_data& operator=(const optional_data& rhs) {
>+    if (rhs.engaged_) {
>+      this->assign(rhs.data_);
>+    } else {
>+      this->destruct();
>+    }
>+    return *this;
>+  }
>+
>+  optional_data& operator=(optional_data&& rhs) noexcept(
>+      std::is_nothrow_move_assignable<T>::value&&
>+          std::is_nothrow_move_constructible<T>::value) {
>+    if (rhs.engaged_) {
>+      this->assign(std::move(rhs.data_));
>+    } else {
>+      this->destruct();
>+    }
>+    return *this;
>+  }
>+};
>+
>+// Ordered by level of restriction, from low to high.
>+// Copyable implies movable.
>+enum class copy_traits { copyable = 0, movable = 1, non_movable = 2 };
>+
>+// Base class for enabling/disabling copy/move constructor.
>+template <copy_traits>
>+class optional_ctor_base;
>+
>+template <>
>+class optional_ctor_base<copy_traits::copyable> {
>+ public:
>+  constexpr optional_ctor_base() = default;
>+  optional_ctor_base(const optional_ctor_base&) = default;
>+  optional_ctor_base(optional_ctor_base&&) = default;
>+  optional_ctor_base& operator=(const optional_ctor_base&) = default;
>+  optional_ctor_base& operator=(optional_ctor_base&&) = default;
>+};
>+
>+template <>
>+class optional_ctor_base<copy_traits::movable> {
>+ public:
>+  constexpr optional_ctor_base() = default;
>+  optional_ctor_base(const optional_ctor_base&) = delete;
>+  optional_ctor_base(optional_ctor_base&&) = default;
>+  optional_ctor_base& operator=(const optional_ctor_base&) = default;
>+  optional_ctor_base& operator=(optional_ctor_base&&) = default;
>+};
>+
>+template <>
>+class optional_ctor_base<copy_traits::non_movable> {
>+ public:
>+  constexpr optional_ctor_base() = default;
>+  optional_ctor_base(const optional_ctor_base&) = delete;
>+  optional_ctor_base(optional_ctor_base&&) = delete;
>+  optional_ctor_base& operator=(const optional_ctor_base&) = default;
>+  optional_ctor_base& operator=(optional_ctor_base&&) = default;
>+};
>+
>+// Base class for enabling/disabling copy/move assignment.
>+template <copy_traits>
>+class optional_assign_base;
>+
>+template <>
>+class optional_assign_base<copy_traits::copyable> {
>+ public:
>+  constexpr optional_assign_base() = default;
>+  optional_assign_base(const optional_assign_base&) = default;
>+  optional_assign_base(optional_assign_base&&) = default;
>+  optional_assign_base& operator=(const optional_assign_base&) = default;
>+  optional_assign_base& operator=(optional_assign_base&&) = default;
>+};
>+
>+template <>
>+class optional_assign_base<copy_traits::movable> {
>+ public:
>+  constexpr optional_assign_base() = default;
>+  optional_assign_base(const optional_assign_base&) = default;
>+  optional_assign_base(optional_assign_base&&) = default;
>+  optional_assign_base& operator=(const optional_assign_base&) = delete;
>+  optional_assign_base& operator=(optional_assign_base&&) = default;
>+};
>+
>+template <>
>+class optional_assign_base<copy_traits::non_movable> {
>+ public:
>+  constexpr optional_assign_base() = default;
>+  optional_assign_base(const optional_assign_base&) = default;
>+  optional_assign_base(optional_assign_base&&) = default;
>+  optional_assign_base& operator=(const optional_assign_base&) = delete;
>+  optional_assign_base& operator=(optional_assign_base&&) = delete;
>+};
>+
>+template <typename T>
>+constexpr copy_traits get_ctor_copy_traits() {
>+  return std::is_copy_constructible<T>::value
>+             ? copy_traits::copyable
>+             : std::is_move_constructible<T>::value ? copy_traits::movable
>+                                                    : copy_traits::non_movable;
>+}
>+
>+template <typename T>
>+constexpr copy_traits get_assign_copy_traits() {
>+  return std::is_copy_assignable<T>::value &&
>+                 std::is_copy_constructible<T>::value
>+             ? copy_traits::copyable
>+             : std::is_move_assignable<T>::value &&
>+                       std::is_move_constructible<T>::value
>+                   ? copy_traits::movable
>+                   : copy_traits::non_movable;
>+}
>+
>+// Whether T is constructible or convertible from optional<U>.
>+template <typename T, typename U>
>+struct is_constructible_convertible_from_optional
>+    : std::integral_constant<
>+          bool, std::is_constructible<T, optional<U>&>::value ||
>+                    std::is_constructible<T, optional<U>&&>::value ||
>+                    std::is_constructible<T, const optional<U>&>::value ||
>+                    std::is_constructible<T, const optional<U>&&>::value ||
>+                    std::is_convertible<optional<U>&, T>::value ||
>+                    std::is_convertible<optional<U>&&, T>::value ||
>+                    std::is_convertible<const optional<U>&, T>::value ||
>+                    std::is_convertible<const optional<U>&&, T>::value> {};
>+
>+// Whether T is constructible or convertible or assignable from optional<U>.
>+template <typename T, typename U>
>+struct is_constructible_convertible_assignable_from_optional
>+    : std::integral_constant<
>+          bool, is_constructible_convertible_from_optional<T, U>::value ||
>+                    std::is_assignable<T&, optional<U>&>::value ||
>+                    std::is_assignable<T&, optional<U>&&>::value ||
>+                    std::is_assignable<T&, const optional<U>&>::value ||
>+                    std::is_assignable<T&, const optional<U>&&>::value> {};
>+
>+// Helper function used by [optional.relops], [optional.comp_with_t],
>+// for checking whether an expression is convertible to bool.
>+bool convertible_to_bool(bool);
>+
>+// Base class for std::hash<absl::optional<T>>:
>+// If std::hash<std::remove_const_t<T>> is enabled, it provides operator() to
>+// compute the hash; Otherwise, it is disabled.
>+// Reference N4659 23.14.15 [unord.hash].
>+template <typename T, typename = size_t>
>+struct optional_hash_base {
>+  optional_hash_base() = delete;
>+  optional_hash_base(const optional_hash_base&) = delete;
>+  optional_hash_base(optional_hash_base&&) = delete;
>+  optional_hash_base& operator=(const optional_hash_base&) = delete;
>+  optional_hash_base& operator=(optional_hash_base&&) = delete;
>+};
>+
>+template <typename T>
>+struct optional_hash_base<T, decltype(std::hash<absl::remove_const_t<T> >()(
>+                                 std::declval<absl::remove_const_t<T> >()))> {
>+  using argument_type = absl::optional<T>;
>+  using result_type = size_t;
>+  size_t operator()(const absl::optional<T>& opt) const {
>+    if (opt) {
>+      return std::hash<absl::remove_const_t<T> >()(*opt);
>+    } else {
>+      return static_cast<size_t>(0x297814aaad196e6dULL);
>+    }
>+  }
>+};
>+
>+}  // namespace optional_internal
>+
>+// -----------------------------------------------------------------------------
>+// absl::optional class definition
>+// -----------------------------------------------------------------------------
>+
>+template <typename T>
>+class optional : private optional_internal::optional_data<T>,
>+                 private optional_internal::optional_ctor_base<
>+                     optional_internal::get_ctor_copy_traits<T>()>,
>+                 private optional_internal::optional_assign_base<
>+                     optional_internal::get_assign_copy_traits<T>()> {
>+  using data_base = optional_internal::optional_data<T>;
>+
>+ public:
>+  typedef T value_type;
>+
>+  // Constructors
>+
>+  // Constructs an `optional` holding an empty value, NOT a default constructed
>+  // `T`.
>+  constexpr optional() noexcept {}
>+
>+  // Constructs an `optional` initialized with `nullopt` to hold an empty value.
>+  constexpr optional(nullopt_t) noexcept {}  // NOLINT(runtime/explicit)
>+
>+  // Copy constructor, standard semantics
>+  optional(const optional& src) = default;
>+
>+  // Move constructor, standard semantics
>+  optional(optional&& src) = default;
>+
>+  // Constructs a non-empty `optional` direct-initialized value of type `T` from
>+  // the arguments `std::forward<Args>(args)...`  within the `optional`.
>+  // (The `in_place_t` is a tag used to indicate that the contained object
>+  // should be constructed in-place.)
>+  //
>+  // TODO(absl-team): Add std::is_constructible<T, Args&&...> SFINAE.
>+  template <typename... Args>
>+  constexpr explicit optional(in_place_t, Args&&... args)
>+      : data_base(in_place_t(), absl::forward<Args>(args)...) {}
>+
>+  // Constructs a non-empty `optional` direct-initialized value of type `T` from
>+  // the arguments of an initializer_list and `std::forward<Args>(args)...`.
>+  // (The `in_place_t` is a tag used to indicate that the contained object
>+  // should be constructed in-place.)
>+  template <typename U, typename... Args,
>+            typename = typename std::enable_if<std::is_constructible<
>+                T, std::initializer_list<U>&, Args&&...>::value>::type>
>+  constexpr explicit optional(in_place_t, std::initializer_list<U> il,
>+                              Args&&... args)
>+      : data_base(in_place_t(), il, absl::forward<Args>(args)...) {
>+  }
>+
>+  // Value constructor (implicit)
>+  template <
>+      typename U = T,
>+      typename std::enable_if<
>+          absl::conjunction<absl::negation<std::is_same<
>+                                in_place_t, typename std::decay<U>::type> >,
>+                            absl::negation<std::is_same<
>+                                optional<T>, typename std::decay<U>::type> >,
>+                            std::is_convertible<U&&, T>,
>+                            std::is_constructible<T, U&&> >::value,
>+          bool>::type = false>
>+  constexpr optional(U&& v) : data_base(in_place_t(), absl::forward<U>(v)) {}
>+
>+  // Value constructor (explicit)
>+  template <
>+      typename U = T,
>+      typename std::enable_if<
>+          absl::conjunction<absl::negation<std::is_same<
>+                                in_place_t, typename std::decay<U>::type>>,
>+                            absl::negation<std::is_same<
>+                                optional<T>, typename std::decay<U>::type>>,
>+                            absl::negation<std::is_convertible<U&&, T>>,
>+                            std::is_constructible<T, U&&>>::value,
>+          bool>::type = false>
>+  explicit constexpr optional(U&& v)
>+      : data_base(in_place_t(), absl::forward<U>(v)) {}
>+
>+  // Converting copy constructor (implicit)
>+  template <typename U,
>+            typename std::enable_if<
>+                absl::conjunction<
>+                    absl::negation<std::is_same<T, U> >,
>+                    std::is_constructible<T, const U&>,
>+                    absl::negation<
>+                        optional_internal::
>+                            is_constructible_convertible_from_optional<T, U> >,
>+                    std::is_convertible<const U&, T> >::value,
>+                bool>::type = false>
>+  optional(const optional<U>& rhs) {
>+    if (rhs) {
>+      this->construct(*rhs);
>+    }
>+  }
>+
>+  // Converting copy constructor (explicit)
>+  template <typename U,
>+            typename std::enable_if<
>+                absl::conjunction<
>+                    absl::negation<std::is_same<T, U>>,
>+                    std::is_constructible<T, const U&>,
>+                    absl::negation<
>+                        optional_internal::
>+                            is_constructible_convertible_from_optional<T, U>>,
>+                    absl::negation<std::is_convertible<const U&, T>>>::value,
>+                bool>::type = false>
>+  explicit optional(const optional<U>& rhs) {
>+    if (rhs) {
>+      this->construct(*rhs);
>+    }
>+  }
>+
>+  // Converting move constructor (implicit)
>+  template <typename U,
>+            typename std::enable_if<
>+                absl::conjunction<
>+                    absl::negation<std::is_same<T, U> >,
>+                    std::is_constructible<T, U&&>,
>+                    absl::negation<
>+                        optional_internal::
>+                            is_constructible_convertible_from_optional<T, U> >,
>+                    std::is_convertible<U&&, T> >::value,
>+                bool>::type = false>
>+  optional(optional<U>&& rhs) {
>+    if (rhs) {
>+      this->construct(std::move(*rhs));
>+    }
>+  }
>+
>+  // Converting move constructor (explicit)
>+  template <
>+      typename U,
>+      typename std::enable_if<
>+          absl::conjunction<
>+              absl::negation<std::is_same<T, U>>, std::is_constructible<T, U&&>,
>+              absl::negation<
>+                  optional_internal::is_constructible_convertible_from_optional<
>+                      T, U>>,
>+              absl::negation<std::is_convertible<U&&, T>>>::value,
>+          bool>::type = false>
>+  explicit optional(optional<U>&& rhs) {
>+    if (rhs) {
>+      this->construct(std::move(*rhs));
>+    }
>+  }
>+
>+  // Destructor. Trivial if `T` is trivially destructible.
>+  ~optional() = default;
>+
>+  // Assignment Operators
>+
>+  // Assignment from `nullopt`
>+  //
>+  // Example:
>+  //
>+  //   struct S { int value; };
>+  //   optional<S> opt = absl::nullopt;  // Could also use opt = { };
>+  optional& operator=(nullopt_t) noexcept {
>+    this->destruct();
>+    return *this;
>+  }
>+
>+  // Copy assignment operator, standard semantics
>+  optional& operator=(const optional& src) = default;
>+
>+  // Move assignment operator, standard semantics
>+  optional& operator=(optional&& src) = default;
>+
>+  // Value assignment operators
>+  template <
>+      typename U = T,
>+      typename = typename std::enable_if<absl::conjunction<
>+          absl::negation<
>+              std::is_same<optional<T>, typename std::decay<U>::type>>,
>+          absl::negation<
>+              absl::conjunction<std::is_scalar<T>,
>+                                std::is_same<T, typename std::decay<U>::type>>>,
>+          std::is_constructible<T, U>, std::is_assignable<T&, U>>::value>::type>
>+  optional& operator=(U&& v) {
>+    this->assign(std::forward<U>(v));
>+    return *this;
>+  }
>+
>+  template <
>+      typename U,
>+      typename = typename std::enable_if<absl::conjunction<
>+          absl::negation<std::is_same<T, U>>,
>+          std::is_constructible<T, const U&>, std::is_assignable<T&, const U&>,
>+          absl::negation<
>+              optional_internal::
>+                  is_constructible_convertible_assignable_from_optional<
>+                      T, U>>>::value>::type>
>+  optional& operator=(const optional<U>& rhs) {
>+    if (rhs) {
>+      this->assign(*rhs);
>+    } else {
>+      this->destruct();
>+    }
>+    return *this;
>+  }
>+
>+  template <typename U,
>+            typename = typename std::enable_if<absl::conjunction<
>+                absl::negation<std::is_same<T, U>>, std::is_constructible<T, U>,
>+                std::is_assignable<T&, U>,
>+                absl::negation<
>+                    optional_internal::
>+                        is_constructible_convertible_assignable_from_optional<
>+                            T, U>>>::value>::type>
>+  optional& operator=(optional<U>&& rhs) {
>+    if (rhs) {
>+      this->assign(std::move(*rhs));
>+    } else {
>+      this->destruct();
>+    }
>+    return *this;
>+  }
>+
>+  // Modifiers
>+
>+  // optional::reset()
>+  //
>+  // Destroys the inner `T` value of an `absl::optional` if one is present.
>+  void reset() noexcept { this->destruct(); }
>+
>+  // optional::emplace()
>+  //
>+  // (Re)constructs the underlying `T` in-place with the given forwarded
>+  // arguments.
>+  //
>+  // Example:
>+  //
>+  //   optional<Foo> opt;
>+  //   opt.emplace(arg1,arg2,arg3);  // Constructs Foo(arg1,arg2,arg3)
>+  //
>+  // If the optional is non-empty, and the `args` refer to subobjects of the
>+  // current object, then behaviour is undefined, because the current object
>+  // will be destructed before the new object is constructed with `args`.
>+  template <typename... Args,
>+            typename = typename std::enable_if<
>+                std::is_constructible<T, Args&&...>::value>::type>
>+  T& emplace(Args&&... args) {
>+    this->destruct();
>+    this->construct(std::forward<Args>(args)...);
>+    return reference();
>+  }
>+
>+  // Emplace reconstruction overload for an initializer list and the given
>+  // forwarded arguments.
>+  //
>+  // Example:
>+  //
>+  //   struct Foo {
>+  //     Foo(std::initializer_list<int>);
>+  //   };
>+  //
>+  //   optional<Foo> opt;
>+  //   opt.emplace({1,2,3});  // Constructs Foo({1,2,3})
>+  template <typename U, typename... Args,
>+            typename = typename std::enable_if<std::is_constructible<
>+                T, std::initializer_list<U>&, Args&&...>::value>::type>
>+  T& emplace(std::initializer_list<U> il, Args&&... args) {
>+    this->destruct();
>+    this->construct(il, std::forward<Args>(args)...);
>+    return reference();
>+  }
>+
>+  // Swaps
>+
>+  // Swap, standard semantics
>+  void swap(optional& rhs) noexcept(
>+      std::is_nothrow_move_constructible<T>::value&&
>+          std::is_trivial<T>::value) {
>+    if (*this) {
>+      if (rhs) {
>+        using std::swap;
>+        swap(**this, *rhs);
>+      } else {
>+        rhs.construct(std::move(**this));
>+        this->destruct();
>+      }
>+    } else {
>+      if (rhs) {
>+        this->construct(std::move(*rhs));
>+        rhs.destruct();
>+      } else {
>+        // No effect (swap(disengaged, disengaged)).
>+      }
>+    }
>+  }
>+
>+  // Observers
>+
>+  // optional::operator->()
>+  //
>+  // Accesses the underlying `T` value's member `m` of an `optional`. If the
>+  // `optional` is empty, behavior is undefined.
>+  //
>+  // If you need myOpt->foo in constexpr, use (*myOpt).foo instead.
>+  const T* operator->() const {
>+    assert(this->engaged_);
>+    return std::addressof(this->data_);
>+  }
>+  T* operator->() {
>+    assert(this->engaged_);
>+    return std::addressof(this->data_);
>+  }
>+
>+  // optional::operator*()
>+  //
>+  // Accesses the underlying `T` value of an `optional`. If the `optional` is
>+  // empty, behavior is undefined.
>+  constexpr const T& operator*() const & { return reference(); }
>+  T& operator*() & {
>+    assert(this->engaged_);
>+    return reference();
>+  }
>+  constexpr const T&& operator*() const && {
>+    return absl::move(reference());
>+  }
>+  T&& operator*() && {
>+    assert(this->engaged_);
>+    return std::move(reference());
>+  }
>+
>+  // optional::operator bool()
>+  //
>+  // Returns false if and only if the `optional` is empty.
>+  //
>+  //   if (opt) {
>+  //     // do something with opt.value();
>+  //   } else {
>+  //     // opt is empty.
>+  //   }
>+  //
>+  constexpr explicit operator bool() const noexcept { return this->engaged_; }
>+
>+  // optional::has_value()
>+  //
>+  // Determines whether the `optional` contains a value. Returns `false` if and
>+  // only if `*this` is empty.
>+  constexpr bool has_value() const noexcept { return this->engaged_; }
>+
>+// Suppress bogus warning on MSVC: MSVC complains call to reference() after
>+// throw_bad_optional_access() is unreachable.
>+#ifdef _MSC_VER
>+#pragma warning(push)
>+#pragma warning(disable : 4702)
>+#endif  // _MSC_VER
>+  // optional::value()
>+  //
>+  // Returns a reference to an `optional`s underlying value. The constness
>+  // and lvalue/rvalue-ness of the `optional` is preserved to the view of
>+  // the `T` sub-object. Throws `absl::bad_optional_access` when the `optional`
>+  // is empty.
>+  constexpr const T& value() const & {
>+    return static_cast<bool>(*this)
>+               ? reference()
>+               : (optional_internal::throw_bad_optional_access(), reference());
>+  }
>+  T& value() & {
>+    return static_cast<bool>(*this)
>+               ? reference()
>+               : (optional_internal::throw_bad_optional_access(), reference());
>+  }
>+  T&& value() && {  // NOLINT(build/c++11)
>+    return std::move(
>+        static_cast<bool>(*this)
>+            ? reference()
>+            : (optional_internal::throw_bad_optional_access(), reference()));
>+  }
>+  constexpr const T&& value() const && {  // NOLINT(build/c++11)
>+    return absl::move(
>+        static_cast<bool>(*this)
>+            ? reference()
>+            : (optional_internal::throw_bad_optional_access(), reference()));
>+  }
>+#ifdef _MSC_VER
>+#pragma warning(pop)
>+#endif  // _MSC_VER
>+
>+  // optional::value_or()
>+  //
>+  // Returns either the value of `T` or a passed default `v` if the `optional`
>+  // is empty.
>+  template <typename U>
>+  constexpr T value_or(U&& v) const& {
>+    static_assert(std::is_copy_constructible<value_type>::value,
>+                  "optional<T>::value_or: T must by copy constructible");
>+    static_assert(std::is_convertible<U&&, value_type>::value,
>+                  "optional<T>::value_or: U must be convertible to T");
>+    return static_cast<bool>(*this)
>+               ? **this
>+               : static_cast<T>(absl::forward<U>(v));
>+  }
>+  template <typename U>
>+  T value_or(U&& v) && {  // NOLINT(build/c++11)
>+    static_assert(std::is_move_constructible<value_type>::value,
>+                  "optional<T>::value_or: T must by copy constructible");
>+    static_assert(std::is_convertible<U&&, value_type>::value,
>+                  "optional<T>::value_or: U must be convertible to T");
>+    return static_cast<bool>(*this) ? std::move(**this)
>+                                    : static_cast<T>(std::forward<U>(v));
>+  }
>+
>+ private:
>+  // Private accessors for internal storage viewed as reference to T.
>+  constexpr const T& reference() const { return this->data_; }
>+  T& reference() { return this->data_; }
>+
>+  // T constraint checks.  You can't have an optional of nullopt_t, in_place_t
>+  // or a reference.
>+  static_assert(
>+      !std::is_same<nullopt_t, typename std::remove_cv<T>::type>::value,
>+      "optional<nullopt_t> is not allowed.");
>+  static_assert(
>+      !std::is_same<in_place_t, typename std::remove_cv<T>::type>::value,
>+      "optional<in_place_t> is not allowed.");
>+  static_assert(!std::is_reference<T>::value,
>+                "optional<reference> is not allowed.");
>+};
>+
>+// Non-member functions
>+
>+// swap()
>+//
>+// Performs a swap between two `absl::optional` objects, using standard
>+// semantics.
>+//
>+// NOTE: we assume `is_swappable()` is always `true`. A compile error will
>+// result if this is not the case.
>+template <typename T,
>+          typename std::enable_if<std::is_move_constructible<T>::value,
>+                                  bool>::type = false>
>+void swap(optional<T>& a, optional<T>& b) noexcept(noexcept(a.swap(b))) {
>+  a.swap(b);
>+}
>+
>+// make_optional()
>+//
>+// Creates a non-empty `optional<T>` where the type of `T` is deduced. An
>+// `absl::optional` can also be explicitly instantiated with
>+// `make_optional<T>(v)`.
>+//
>+// Note: `make_optional()` constructions may be declared `constexpr` for
>+// trivially copyable types `T`. Non-trivial types require copy elision
>+// support in C++17 for `make_optional` to support `constexpr` on such
>+// non-trivial types.
>+//
>+// Example:
>+//
>+//   constexpr absl::optional<int> opt = absl::make_optional(1);
>+//   static_assert(opt.value() == 1, "");
>+template <typename T>
>+constexpr optional<typename std::decay<T>::type> make_optional(T&& v) {
>+  return optional<typename std::decay<T>::type>(absl::forward<T>(v));
>+}
>+
>+template <typename T, typename... Args>
>+constexpr optional<T> make_optional(Args&&... args) {
>+  return optional<T>(in_place_t(), absl::forward<Args>(args)...);
>+}
>+
>+template <typename T, typename U, typename... Args>
>+constexpr optional<T> make_optional(std::initializer_list<U> il,
>+                                    Args&&... args) {
>+  return optional<T>(in_place_t(), il,
>+                     absl::forward<Args>(args)...);
>+}
>+
>+// Relational operators [optional.relops]
>+
>+// Empty optionals are considered equal to each other and less than non-empty
>+// optionals. Supports relations between optional<T> and optional<U>, between
>+// optional<T> and U, and between optional<T> and nullopt.
>+//
>+// Note: We're careful to support T having non-bool relationals.
>+
>+// Requires: The expression, e.g. "*x == *y" shall be well-formed and its result
>+// shall be convertible to bool.
>+// The C++17 (N4606) "Returns:" statements are translated into
>+// code in an obvious way here, and the original text retained as function docs.
>+// Returns: If bool(x) != bool(y), false; otherwise if bool(x) == false, true;
>+// otherwise *x == *y.
>+template <typename T, typename U>
>+constexpr auto operator==(const optional<T>& x, const optional<U>& y)
>+    -> decltype(optional_internal::convertible_to_bool(*x == *y)) {
>+  return static_cast<bool>(x) != static_cast<bool>(y)
>+             ? false
>+             : static_cast<bool>(x) == false ? true
>+                                             : static_cast<bool>(*x == *y);
>+}
>+
>+// Returns: If bool(x) != bool(y), true; otherwise, if bool(x) == false, false;
>+// otherwise *x != *y.
>+template <typename T, typename U>
>+constexpr auto operator!=(const optional<T>& x, const optional<U>& y)
>+    -> decltype(optional_internal::convertible_to_bool(*x != *y)) {
>+  return static_cast<bool>(x) != static_cast<bool>(y)
>+             ? true
>+             : static_cast<bool>(x) == false ? false
>+                                             : static_cast<bool>(*x != *y);
>+}
>+// Returns: If !y, false; otherwise, if !x, true; otherwise *x < *y.
>+template <typename T, typename U>
>+constexpr auto operator<(const optional<T>& x, const optional<U>& y)
>+    -> decltype(optional_internal::convertible_to_bool(*x < *y)) {
>+  return !y ? false : !x ? true : static_cast<bool>(*x < *y);
>+}
>+// Returns: If !x, false; otherwise, if !y, true; otherwise *x > *y.
>+template <typename T, typename U>
>+constexpr auto operator>(const optional<T>& x, const optional<U>& y)
>+    -> decltype(optional_internal::convertible_to_bool(*x > *y)) {
>+  return !x ? false : !y ? true : static_cast<bool>(*x > *y);
>+}
>+// Returns: If !x, true; otherwise, if !y, false; otherwise *x <= *y.
>+template <typename T, typename U>
>+constexpr auto operator<=(const optional<T>& x, const optional<U>& y)
>+    -> decltype(optional_internal::convertible_to_bool(*x <= *y)) {
>+  return !x ? true : !y ? false : static_cast<bool>(*x <= *y);
>+}
>+// Returns: If !y, true; otherwise, if !x, false; otherwise *x >= *y.
>+template <typename T, typename U>
>+constexpr auto operator>=(const optional<T>& x, const optional<U>& y)
>+    -> decltype(optional_internal::convertible_to_bool(*x >= *y)) {
>+  return !y ? true : !x ? false : static_cast<bool>(*x >= *y);
>+}
>+
>+// Comparison with nullopt [optional.nullops]
>+// The C++17 (N4606) "Returns:" statements are used directly here.
>+template <typename T>
>+constexpr bool operator==(const optional<T>& x, nullopt_t) noexcept {
>+  return !x;
>+}
>+template <typename T>
>+constexpr bool operator==(nullopt_t, const optional<T>& x) noexcept {
>+  return !x;
>+}
>+template <typename T>
>+constexpr bool operator!=(const optional<T>& x, nullopt_t) noexcept {
>+  return static_cast<bool>(x);
>+}
>+template <typename T>
>+constexpr bool operator!=(nullopt_t, const optional<T>& x) noexcept {
>+  return static_cast<bool>(x);
>+}
>+template <typename T>
>+constexpr bool operator<(const optional<T>&, nullopt_t) noexcept {
>+  return false;
>+}
>+template <typename T>
>+constexpr bool operator<(nullopt_t, const optional<T>& x) noexcept {
>+  return static_cast<bool>(x);
>+}
>+template <typename T>
>+constexpr bool operator<=(const optional<T>& x, nullopt_t) noexcept {
>+  return !x;
>+}
>+template <typename T>
>+constexpr bool operator<=(nullopt_t, const optional<T>&) noexcept {
>+  return true;
>+}
>+template <typename T>
>+constexpr bool operator>(const optional<T>& x, nullopt_t) noexcept {
>+  return static_cast<bool>(x);
>+}
>+template <typename T>
>+constexpr bool operator>(nullopt_t, const optional<T>&) noexcept {
>+  return false;
>+}
>+template <typename T>
>+constexpr bool operator>=(const optional<T>&, nullopt_t) noexcept {
>+  return true;
>+}
>+template <typename T>
>+constexpr bool operator>=(nullopt_t, const optional<T>& x) noexcept {
>+  return !x;
>+}
>+
>+// Comparison with T [optional.comp_with_t]
>+
>+// Requires: The expression, e.g. "*x == v" shall be well-formed and its result
>+// shall be convertible to bool.
>+// The C++17 (N4606) "Equivalent to:" statements are used directly here.
>+template <typename T, typename U>
>+constexpr auto operator==(const optional<T>& x, const U& v)
>+    -> decltype(optional_internal::convertible_to_bool(*x == v)) {
>+  return static_cast<bool>(x) ? static_cast<bool>(*x == v) : false;
>+}
>+template <typename T, typename U>
>+constexpr auto operator==(const U& v, const optional<T>& x)
>+    -> decltype(optional_internal::convertible_to_bool(v == *x)) {
>+  return static_cast<bool>(x) ? static_cast<bool>(v == *x) : false;
>+}
>+template <typename T, typename U>
>+constexpr auto operator!=(const optional<T>& x, const U& v)
>+    -> decltype(optional_internal::convertible_to_bool(*x != v)) {
>+  return static_cast<bool>(x) ? static_cast<bool>(*x != v) : true;
>+}
>+template <typename T, typename U>
>+constexpr auto operator!=(const U& v, const optional<T>& x)
>+    -> decltype(optional_internal::convertible_to_bool(v != *x)) {
>+  return static_cast<bool>(x) ? static_cast<bool>(v != *x) : true;
>+}
>+template <typename T, typename U>
>+constexpr auto operator<(const optional<T>& x, const U& v)
>+    -> decltype(optional_internal::convertible_to_bool(*x < v)) {
>+  return static_cast<bool>(x) ? static_cast<bool>(*x < v) : true;
>+}
>+template <typename T, typename U>
>+constexpr auto operator<(const U& v, const optional<T>& x)
>+    -> decltype(optional_internal::convertible_to_bool(v < *x)) {
>+  return static_cast<bool>(x) ? static_cast<bool>(v < *x) : false;
>+}
>+template <typename T, typename U>
>+constexpr auto operator<=(const optional<T>& x, const U& v)
>+    -> decltype(optional_internal::convertible_to_bool(*x <= v)) {
>+  return static_cast<bool>(x) ? static_cast<bool>(*x <= v) : true;
>+}
>+template <typename T, typename U>
>+constexpr auto operator<=(const U& v, const optional<T>& x)
>+    -> decltype(optional_internal::convertible_to_bool(v <= *x)) {
>+  return static_cast<bool>(x) ? static_cast<bool>(v <= *x) : false;
>+}
>+template <typename T, typename U>
>+constexpr auto operator>(const optional<T>& x, const U& v)
>+    -> decltype(optional_internal::convertible_to_bool(*x > v)) {
>+  return static_cast<bool>(x) ? static_cast<bool>(*x > v) : false;
>+}
>+template <typename T, typename U>
>+constexpr auto operator>(const U& v, const optional<T>& x)
>+    -> decltype(optional_internal::convertible_to_bool(v > *x)) {
>+  return static_cast<bool>(x) ? static_cast<bool>(v > *x) : true;
>+}
>+template <typename T, typename U>
>+constexpr auto operator>=(const optional<T>& x, const U& v)
>+    -> decltype(optional_internal::convertible_to_bool(*x >= v)) {
>+  return static_cast<bool>(x) ? static_cast<bool>(*x >= v) : false;
>+}
>+template <typename T, typename U>
>+constexpr auto operator>=(const U& v, const optional<T>& x)
>+    -> decltype(optional_internal::convertible_to_bool(v >= *x)) {
>+  return static_cast<bool>(x) ? static_cast<bool>(v >= *x) : true;
>+}
>+
>+}  // namespace absl
>+
>+namespace std {
>+
>+// std::hash specialization for absl::optional.
>+template <typename T>
>+struct hash<absl::optional<T> >
>+    : absl::optional_internal::optional_hash_base<T> {};
>+
>+}  // namespace std
>+
>+#undef ABSL_OPTIONAL_USE_INHERITING_CONSTRUCTORS
>+#undef ABSL_MSVC_CONSTEXPR_BUG_IN_UNION_LIKE_CLASS
>+
>+#endif  // ABSL_HAVE_STD_OPTIONAL
>+
>+#endif  // ABSL_TYPES_OPTIONAL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/optional_exception_safety_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/optional_exception_safety_test.cc
>new file mode 100644
>index 00000000000..d2ef04b8d1b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/optional_exception_safety_test.cc
>@@ -0,0 +1,282 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/types/optional.h"
>+
>+#include "gtest/gtest.h"
>+#include "absl/base/internal/exception_safety_testing.h"
>+
>+namespace absl {
>+
>+namespace {
>+
>+using ::testing::AssertionFailure;
>+using ::testing::AssertionResult;
>+using ::testing::AssertionSuccess;
>+using ::testing::MakeExceptionSafetyTester;
>+
>+using Thrower = testing::ThrowingValue<testing::TypeSpec::kEverythingThrows>;
>+using Optional = absl::optional<Thrower>;
>+
>+using MoveThrower = testing::ThrowingValue<testing::TypeSpec::kNoThrowMove>;
>+using MoveOptional = absl::optional<MoveThrower>;
>+
>+constexpr int kInitialInteger = 5;
>+constexpr int kUpdatedInteger = 10;
>+
>+template <typename OptionalT>
>+bool ValueThrowsBadOptionalAccess(const OptionalT& optional) try {
>+  return (static_cast<void>(optional.value()), false);
>+} catch (absl::bad_optional_access) {
>+  return true;
>+}
>+
>+template <typename OptionalT>
>+AssertionResult CheckInvariants(OptionalT* optional_ptr) {
>+  // Check the current state post-throw for validity
>+  auto& optional = *optional_ptr;
>+
>+  if (optional.has_value() && ValueThrowsBadOptionalAccess(optional)) {
>+    return AssertionFailure()
>+           << "Optional with value should not throw bad_optional_access when "
>+              "accessing the value.";
>+  }
>+  if (!optional.has_value() && !ValueThrowsBadOptionalAccess(optional)) {
>+    return AssertionFailure()
>+           << "Optional without a value should throw bad_optional_access when "
>+              "accessing the value.";
>+  }
>+
>+  // Reset to a known state
>+  optional.reset();
>+
>+  // Confirm that the known post-reset state is valid
>+  if (optional.has_value()) {
>+    return AssertionFailure()
>+           << "Optional should not contain a value after being reset.";
>+  }
>+  if (!ValueThrowsBadOptionalAccess(optional)) {
>+    return AssertionFailure() << "Optional should throw bad_optional_access "
>+                                 "when accessing the value after being reset.";
>+  }
>+
>+  return AssertionSuccess();
>+}
>+
>+template <typename OptionalT>
>+AssertionResult CheckDisengaged(OptionalT* optional_ptr) {
>+  auto& optional = *optional_ptr;
>+
>+  if (optional.has_value()) {
>+    return AssertionFailure()
>+           << "Expected optional to not contain a value but a value was found.";
>+  }
>+
>+  return AssertionSuccess();
>+}
>+
>+template <typename OptionalT>
>+AssertionResult CheckEngaged(OptionalT* optional_ptr) {
>+  auto& optional = *optional_ptr;
>+
>+  if (!optional.has_value()) {
>+    return AssertionFailure()
>+           << "Expected optional to contain a value but no value was found.";
>+  }
>+
>+  return AssertionSuccess();
>+}
>+
>+TEST(OptionalExceptionSafety, ThrowingConstructors) {
>+  auto thrower_nonempty = Optional(Thrower(kInitialInteger));
>+  testing::TestThrowingCtor<Optional>(thrower_nonempty);
>+
>+  auto integer_nonempty = absl::optional<int>(kInitialInteger);
>+  testing::TestThrowingCtor<Optional>(integer_nonempty);
>+  testing::TestThrowingCtor<Optional>(std::move(integer_nonempty));  // NOLINT
>+
>+  testing::TestThrowingCtor<Optional>(kInitialInteger);
>+  using ThrowerVec = std::vector<Thrower, testing::ThrowingAllocator<Thrower>>;
>+  testing::TestThrowingCtor<absl::optional<ThrowerVec>>(
>+      absl::in_place,
>+      std::initializer_list<Thrower>{Thrower(), Thrower(), Thrower()},
>+      testing::ThrowingAllocator<Thrower>());
>+}
>+
>+TEST(OptionalExceptionSafety, NothrowConstructors) {
>+  // This constructor is marked noexcept. If it throws, the program will
>+  // terminate.
>+  testing::TestThrowingCtor<MoveOptional>(MoveOptional(kUpdatedInteger));
>+}
>+
>+TEST(OptionalExceptionSafety, Emplace) {
>+  // Test the basic guarantee plus test the result of optional::has_value()
>+  // is false in all cases
>+  auto disengaged_test = MakeExceptionSafetyTester().WithInvariants(
>+      CheckInvariants<Optional>, CheckDisengaged<Optional>);
>+  auto disengaged_test_empty = disengaged_test.WithInitialValue(Optional());
>+  auto disengaged_test_nonempty =
>+      disengaged_test.WithInitialValue(Optional(kInitialInteger));
>+
>+  auto emplace_thrower_directly = [](Optional* optional_ptr) {
>+    optional_ptr->emplace(kUpdatedInteger);
>+  };
>+  EXPECT_TRUE(disengaged_test_empty.Test(emplace_thrower_directly));
>+  EXPECT_TRUE(disengaged_test_nonempty.Test(emplace_thrower_directly));
>+
>+  auto emplace_thrower_copy = [](Optional* optional_ptr) {
>+    auto thrower = Thrower(kUpdatedInteger, testing::nothrow_ctor);
>+    optional_ptr->emplace(thrower);
>+  };
>+  EXPECT_TRUE(disengaged_test_empty.Test(emplace_thrower_copy));
>+  EXPECT_TRUE(disengaged_test_nonempty.Test(emplace_thrower_copy));
>+}
>+
>+TEST(OptionalExceptionSafety, EverythingThrowsSwap) {
>+  // Test the basic guarantee plus test the result of optional::has_value()
>+  // remains the same
>+  auto test =
>+      MakeExceptionSafetyTester().WithInvariants(CheckInvariants<Optional>);
>+  auto disengaged_test_empty = test.WithInitialValue(Optional())
>+                                   .WithInvariants(CheckDisengaged<Optional>);
>+  auto engaged_test_nonempty = test.WithInitialValue(Optional(kInitialInteger))
>+                                   .WithInvariants(CheckEngaged<Optional>);
>+
>+  auto swap_empty = [](Optional* optional_ptr) {
>+    auto empty = Optional();
>+    optional_ptr->swap(empty);
>+  };
>+  EXPECT_TRUE(engaged_test_nonempty.Test(swap_empty));
>+
>+  auto swap_nonempty = [](Optional* optional_ptr) {
>+    auto nonempty =
>+        Optional(absl::in_place, kUpdatedInteger, testing::nothrow_ctor);
>+    optional_ptr->swap(nonempty);
>+  };
>+  EXPECT_TRUE(disengaged_test_empty.Test(swap_nonempty));
>+  EXPECT_TRUE(engaged_test_nonempty.Test(swap_nonempty));
>+}
>+
>+TEST(OptionalExceptionSafety, NoThrowMoveSwap) {
>+  // Tests the nothrow guarantee for optional of T with non-throwing move
>+  {
>+    auto empty = MoveOptional();
>+    auto nonempty = MoveOptional(kInitialInteger);
>+    EXPECT_TRUE(testing::TestNothrowOp([&]() { nonempty.swap(empty); }));
>+  }
>+  {
>+    auto nonempty = MoveOptional(kUpdatedInteger);
>+    auto empty = MoveOptional();
>+    EXPECT_TRUE(testing::TestNothrowOp([&]() { empty.swap(nonempty); }));
>+  }
>+  {
>+    auto nonempty_from = MoveOptional(kUpdatedInteger);
>+    auto nonempty_to = MoveOptional(kInitialInteger);
>+    EXPECT_TRUE(
>+        testing::TestNothrowOp([&]() { nonempty_to.swap(nonempty_from); }));
>+  }
>+}
>+
>+TEST(OptionalExceptionSafety, CopyAssign) {
>+  // Test the basic guarantee plus test the result of optional::has_value()
>+  // remains the same
>+  auto test =
>+      MakeExceptionSafetyTester().WithInvariants(CheckInvariants<Optional>);
>+  auto disengaged_test_empty = test.WithInitialValue(Optional())
>+                                   .WithInvariants(CheckDisengaged<Optional>);
>+  auto engaged_test_nonempty = test.WithInitialValue(Optional(kInitialInteger))
>+                                   .WithInvariants(CheckEngaged<Optional>);
>+
>+  auto copyassign_nonempty = [](Optional* optional_ptr) {
>+    auto nonempty =
>+        Optional(absl::in_place, kUpdatedInteger, testing::nothrow_ctor);
>+    *optional_ptr = nonempty;
>+  };
>+  EXPECT_TRUE(disengaged_test_empty.Test(copyassign_nonempty));
>+  EXPECT_TRUE(engaged_test_nonempty.Test(copyassign_nonempty));
>+
>+  auto copyassign_thrower = [](Optional* optional_ptr) {
>+    auto thrower = Thrower(kUpdatedInteger, testing::nothrow_ctor);
>+    *optional_ptr = thrower;
>+  };
>+  EXPECT_TRUE(disengaged_test_empty.Test(copyassign_thrower));
>+  EXPECT_TRUE(engaged_test_nonempty.Test(copyassign_thrower));
>+}
>+
>+TEST(OptionalExceptionSafety, MoveAssign) {
>+  // Test the basic guarantee plus test the result of optional::has_value()
>+  // remains the same
>+  auto test =
>+      MakeExceptionSafetyTester().WithInvariants(CheckInvariants<Optional>);
>+  auto disengaged_test_empty = test.WithInitialValue(Optional())
>+                                   .WithInvariants(CheckDisengaged<Optional>);
>+  auto engaged_test_nonempty = test.WithInitialValue(Optional(kInitialInteger))
>+                                   .WithInvariants(CheckEngaged<Optional>);
>+
>+  auto moveassign_empty = [](Optional* optional_ptr) {
>+    auto empty = Optional();
>+    *optional_ptr = std::move(empty);
>+  };
>+  EXPECT_TRUE(engaged_test_nonempty.Test(moveassign_empty));
>+
>+  auto moveassign_nonempty = [](Optional* optional_ptr) {
>+    auto nonempty =
>+        Optional(absl::in_place, kUpdatedInteger, testing::nothrow_ctor);
>+    *optional_ptr = std::move(nonempty);
>+  };
>+  EXPECT_TRUE(disengaged_test_empty.Test(moveassign_nonempty));
>+  EXPECT_TRUE(engaged_test_nonempty.Test(moveassign_nonempty));
>+
>+  auto moveassign_thrower = [](Optional* optional_ptr) {
>+    auto thrower = Thrower(kUpdatedInteger, testing::nothrow_ctor);
>+    *optional_ptr = std::move(thrower);
>+  };
>+  EXPECT_TRUE(disengaged_test_empty.Test(moveassign_thrower));
>+  EXPECT_TRUE(engaged_test_nonempty.Test(moveassign_thrower));
>+}
>+
>+TEST(OptionalExceptionSafety, NothrowMoveAssign) {
>+  // Tests the nothrow guarantee for optional of T with non-throwing move
>+  {
>+    auto empty = MoveOptional();
>+    auto nonempty = MoveOptional(kInitialInteger);
>+    EXPECT_TRUE(testing::TestNothrowOp([&]() { nonempty = std::move(empty); }));
>+  }
>+  {
>+    auto nonempty = MoveOptional(kInitialInteger);
>+    auto empty = MoveOptional();
>+    EXPECT_TRUE(testing::TestNothrowOp([&]() { empty = std::move(nonempty); }));
>+  }
>+  {
>+    auto nonempty_from = MoveOptional(kUpdatedInteger);
>+    auto nonempty_to = MoveOptional(kInitialInteger);
>+    EXPECT_TRUE(testing::TestNothrowOp(
>+        [&]() { nonempty_to = std::move(nonempty_from); }));
>+  }
>+  {
>+    auto thrower = MoveThrower(kUpdatedInteger);
>+    auto empty = MoveOptional();
>+    EXPECT_TRUE(testing::TestNothrowOp([&]() { empty = std::move(thrower); }));
>+  }
>+  {
>+    auto thrower = MoveThrower(kUpdatedInteger);
>+    auto nonempty = MoveOptional(kInitialInteger);
>+    EXPECT_TRUE(
>+        testing::TestNothrowOp([&]() { nonempty = std::move(thrower); }));
>+  }
>+}
>+
>+}  // namespace
>+
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/optional_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/optional_test.cc
>new file mode 100644
>index 00000000000..179bfd66d2f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/optional_test.cc
>@@ -0,0 +1,1625 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/types/optional.h"
>+
>+#include <string>
>+#include <type_traits>
>+#include <utility>
>+
>+#include "gtest/gtest.h"
>+#include "absl/base/config.h"
>+#include "absl/base/internal/raw_logging.h"
>+#include "absl/meta/type_traits.h"
>+#include "absl/strings/string_view.h"
>+
>+struct Hashable {};
>+
>+namespace std {
>+template <>
>+struct hash<Hashable> {
>+  size_t operator()(const Hashable&) { return 0; }
>+};
>+}  // namespace std
>+
>+struct NonHashable {};
>+
>+namespace {
>+
>+std::string TypeQuals(std::string&) { return "&"; }
>+std::string TypeQuals(std::string&&) { return "&&"; }
>+std::string TypeQuals(const std::string&) { return "c&"; }
>+std::string TypeQuals(const std::string&&) { return "c&&"; }
>+
>+struct StructorListener {
>+  int construct0 = 0;
>+  int construct1 = 0;
>+  int construct2 = 0;
>+  int listinit = 0;
>+  int copy = 0;
>+  int move = 0;
>+  int copy_assign = 0;
>+  int move_assign = 0;
>+  int destruct = 0;
>+  int volatile_copy = 0;
>+  int volatile_move = 0;
>+  int volatile_copy_assign = 0;
>+  int volatile_move_assign = 0;
>+};
>+
>+// Suppress MSVC warnings.
>+// 4521: multiple copy constructors specified
>+// 4522: multiple assignment operators specified
>+// We wrote multiple of them to test that the correct overloads are selected.
>+#ifdef _MSC_VER
>+#pragma warning( push )
>+#pragma warning( disable : 4521)
>+#pragma warning( disable : 4522)
>+#endif
>+struct Listenable {
>+  static StructorListener* listener;
>+
>+  Listenable() { ++listener->construct0; }
>+  explicit Listenable(int /*unused*/) { ++listener->construct1; }
>+  Listenable(int /*unused*/, int /*unused*/) { ++listener->construct2; }
>+  Listenable(std::initializer_list<int> /*unused*/) { ++listener->listinit; }
>+  Listenable(const Listenable& /*unused*/) { ++listener->copy; }
>+  Listenable(const volatile Listenable& /*unused*/) {
>+    ++listener->volatile_copy;
>+  }
>+  Listenable(volatile Listenable&& /*unused*/) { ++listener->volatile_move; }
>+  Listenable(Listenable&& /*unused*/) { ++listener->move; }
>+  Listenable& operator=(const Listenable& /*unused*/) {
>+    ++listener->copy_assign;
>+    return *this;
>+  }
>+  Listenable& operator=(Listenable&& /*unused*/) {
>+    ++listener->move_assign;
>+    return *this;
>+  }
>+  // use void return type instead of volatile T& to work around GCC warning
>+  // when the assignment's returned reference is ignored.
>+  void operator=(const volatile Listenable& /*unused*/) volatile {
>+    ++listener->volatile_copy_assign;
>+  }
>+  void operator=(volatile Listenable&& /*unused*/) volatile {
>+    ++listener->volatile_move_assign;
>+  }
>+  ~Listenable() { ++listener->destruct; }
>+};
>+#ifdef _MSC_VER
>+#pragma warning( pop )
>+#endif
>+
>+StructorListener* Listenable::listener = nullptr;
>+
>+// ABSL_HAVE_NO_CONSTEXPR_INITIALIZER_LIST is defined to 1 when the standard
>+// library implementation doesn't marked initializer_list's default constructor
>+// constexpr. The C++11 standard doesn't specify constexpr on it, but C++14
>+// added it. However, libstdc++ 4.7 marked it constexpr.
>+#if defined(_LIBCPP_VERSION) && \
>+    (_LIBCPP_STD_VER <= 11 || defined(_LIBCPP_HAS_NO_CXX14_CONSTEXPR))
>+#define ABSL_HAVE_NO_CONSTEXPR_INITIALIZER_LIST 1
>+#endif
>+
>+struct ConstexprType {
>+  enum CtorTypes {
>+    kCtorDefault,
>+    kCtorInt,
>+    kCtorInitializerList,
>+    kCtorConstChar
>+  };
>+  constexpr ConstexprType() : x(kCtorDefault) {}
>+  constexpr explicit ConstexprType(int i) : x(kCtorInt) {}
>+#ifndef ABSL_HAVE_NO_CONSTEXPR_INITIALIZER_LIST
>+  constexpr ConstexprType(std::initializer_list<int> il)
>+      : x(kCtorInitializerList) {}
>+#endif
>+  constexpr ConstexprType(const char*)  // NOLINT(runtime/explicit)
>+      : x(kCtorConstChar) {}
>+  int x;
>+};
>+
>+struct Copyable {
>+  Copyable() {}
>+  Copyable(const Copyable&) {}
>+  Copyable& operator=(const Copyable&) { return *this; }
>+};
>+
>+struct MoveableThrow {
>+  MoveableThrow() {}
>+  MoveableThrow(MoveableThrow&&) {}
>+  MoveableThrow& operator=(MoveableThrow&&) { return *this; }
>+};
>+
>+struct MoveableNoThrow {
>+  MoveableNoThrow() {}
>+  MoveableNoThrow(MoveableNoThrow&&) noexcept {}
>+  MoveableNoThrow& operator=(MoveableNoThrow&&) noexcept { return *this; }
>+};
>+
>+struct NonMovable {
>+  NonMovable() {}
>+  NonMovable(const NonMovable&) = delete;
>+  NonMovable& operator=(const NonMovable&) = delete;
>+  NonMovable(NonMovable&&) = delete;
>+  NonMovable& operator=(NonMovable&&) = delete;
>+};
>+
>+TEST(optionalTest, DefaultConstructor) {
>+  absl::optional<int> empty;
>+  EXPECT_FALSE(empty);
>+  constexpr absl::optional<int> cempty;
>+  static_assert(!cempty.has_value(), "");
>+  EXPECT_TRUE(
>+      std::is_nothrow_default_constructible<absl::optional<int>>::value);
>+}
>+
>+TEST(optionalTest, nulloptConstructor) {
>+  absl::optional<int> empty(absl::nullopt);
>+  EXPECT_FALSE(empty);
>+
>+#ifdef ABSL_HAVE_STD_OPTIONAL
>+  constexpr absl::optional<int> cempty{absl::nullopt};
>+#else
>+  // Creating a temporary absl::nullopt_t object instead of using absl::nullopt
>+  // because absl::nullopt cannot be constexpr and have external linkage at the
>+  // same time.
>+  constexpr absl::optional<int> cempty{absl::nullopt_t(absl::nullopt_t::init)};
>+#endif
>+  static_assert(!cempty.has_value(), "");
>+  EXPECT_TRUE((std::is_nothrow_constructible<absl::optional<int>,
>+                                             absl::nullopt_t>::value));
>+}
>+
>+TEST(optionalTest, CopyConstructor) {
>+  {
>+    absl::optional<int> empty, opt42 = 42;
>+    absl::optional<int> empty_copy(empty);
>+    EXPECT_FALSE(empty_copy);
>+    absl::optional<int> opt42_copy(opt42);
>+    EXPECT_TRUE(opt42_copy);
>+    EXPECT_EQ(42, *opt42_copy);
>+  }
>+  {
>+    absl::optional<const int> empty, opt42 = 42;
>+    absl::optional<const int> empty_copy(empty);
>+    EXPECT_FALSE(empty_copy);
>+    absl::optional<const int> opt42_copy(opt42);
>+    EXPECT_TRUE(opt42_copy);
>+    EXPECT_EQ(42, *opt42_copy);
>+  }
>+  {
>+    absl::optional<volatile int> empty, opt42 = 42;
>+    absl::optional<volatile int> empty_copy(empty);
>+    EXPECT_FALSE(empty_copy);
>+    absl::optional<volatile int> opt42_copy(opt42);
>+    EXPECT_TRUE(opt42_copy);
>+    EXPECT_EQ(42, *opt42_copy);
>+  }
>+  // test copyablility
>+  EXPECT_TRUE(std::is_copy_constructible<absl::optional<int>>::value);
>+  EXPECT_TRUE(std::is_copy_constructible<absl::optional<Copyable>>::value);
>+  EXPECT_FALSE(
>+      std::is_copy_constructible<absl::optional<MoveableThrow>>::value);
>+  EXPECT_FALSE(
>+      std::is_copy_constructible<absl::optional<MoveableNoThrow>>::value);
>+  EXPECT_FALSE(std::is_copy_constructible<absl::optional<NonMovable>>::value);
>+
>+  EXPECT_FALSE(
>+      absl::is_trivially_copy_constructible<absl::optional<Copyable>>::value);
>+#if defined(ABSL_HAVE_STD_OPTIONAL) && defined(__GLIBCXX__)
>+  // libstdc++ std::optional implementation (as of 7.2) has a bug: when T is
>+  // trivially copyable, optional<T> is not trivially copyable (due to one of
>+  // its base class is unconditionally nontrivial).
>+#define ABSL_GLIBCXX_OPTIONAL_TRIVIALITY_BUG 1
>+#endif
>+#ifndef ABSL_GLIBCXX_OPTIONAL_TRIVIALITY_BUG
>+  EXPECT_TRUE(
>+      absl::is_trivially_copy_constructible<absl::optional<int>>::value);
>+  EXPECT_TRUE(
>+      absl::is_trivially_copy_constructible<absl::optional<const int>>::value);
>+#ifndef _MSC_VER
>+  // See defect report "Trivial copy/move constructor for class with volatile
>+  // member" at
>+  // http://www.open-std.org/jtc1/sc22/wg21/docs/cwg_defects.html#2094
>+  // A class with non-static data member of volatile-qualified type should still
>+  // have a trivial copy constructor if the data member is trivial.
>+  // Also a cv-qualified scalar type should be trivially copyable.
>+  EXPECT_TRUE(absl::is_trivially_copy_constructible<
>+              absl::optional<volatile int>>::value);
>+#endif  // _MSC_VER
>+#endif  // ABSL_GLIBCXX_OPTIONAL_TRIVIALITY_BUG
>+
>+  // constexpr copy constructor for trivially copyable types
>+  {
>+    constexpr absl::optional<int> o1;
>+    constexpr absl::optional<int> o2 = o1;
>+    static_assert(!o2, "");
>+  }
>+  {
>+    constexpr absl::optional<int> o1 = 42;
>+    constexpr absl::optional<int> o2 = o1;
>+    static_assert(o2, "");
>+    static_assert(*o2 == 42, "");
>+  }
>+  {
>+    struct TrivialCopyable {
>+      constexpr TrivialCopyable() : x(0) {}
>+      constexpr explicit TrivialCopyable(int i) : x(i) {}
>+      int x;
>+    };
>+    constexpr absl::optional<TrivialCopyable> o1(42);
>+    constexpr absl::optional<TrivialCopyable> o2 = o1;
>+    static_assert(o2, "");
>+    static_assert((*o2).x == 42, "");
>+#ifndef ABSL_GLIBCXX_OPTIONAL_TRIVIALITY_BUG
>+    EXPECT_TRUE(absl::is_trivially_copy_constructible<
>+                absl::optional<TrivialCopyable>>::value);
>+    EXPECT_TRUE(absl::is_trivially_copy_constructible<
>+                absl::optional<const TrivialCopyable>>::value);
>+#endif
>+    // When testing with VS 2017 15.3, there seems to be a bug in MSVC
>+    // std::optional when T is volatile-qualified. So skipping this test.
>+    // Bug report:
>+    // https://connect.microsoft.com/VisualStudio/feedback/details/3142534
>+#if defined(ABSL_HAVE_STD_OPTIONAL) && defined(_MSC_VER) && _MSC_VER >= 1911
>+#define ABSL_MSVC_OPTIONAL_VOLATILE_COPY_BUG 1
>+#endif
>+#ifndef ABSL_MSVC_OPTIONAL_VOLATILE_COPY_BUG
>+    EXPECT_FALSE(std::is_copy_constructible<
>+                 absl::optional<volatile TrivialCopyable>>::value);
>+#endif
>+  }
>+}
>+
>+TEST(optionalTest, MoveConstructor) {
>+  absl::optional<int> empty, opt42 = 42;
>+  absl::optional<int> empty_move(std::move(empty));
>+  EXPECT_FALSE(empty_move);
>+  absl::optional<int> opt42_move(std::move(opt42));
>+  EXPECT_TRUE(opt42_move);
>+  EXPECT_EQ(42, opt42_move);
>+  // test movability
>+  EXPECT_TRUE(std::is_move_constructible<absl::optional<int>>::value);
>+  EXPECT_TRUE(std::is_move_constructible<absl::optional<Copyable>>::value);
>+  EXPECT_TRUE(std::is_move_constructible<absl::optional<MoveableThrow>>::value);
>+  EXPECT_TRUE(
>+      std::is_move_constructible<absl::optional<MoveableNoThrow>>::value);
>+  EXPECT_FALSE(std::is_move_constructible<absl::optional<NonMovable>>::value);
>+  // test noexcept
>+  EXPECT_TRUE(std::is_nothrow_move_constructible<absl::optional<int>>::value);
>+#ifndef ABSL_HAVE_STD_OPTIONAL
>+  EXPECT_EQ(
>+      absl::default_allocator_is_nothrow::value,
>+      std::is_nothrow_move_constructible<absl::optional<MoveableThrow>>::value);
>+#endif
>+  EXPECT_TRUE(std::is_nothrow_move_constructible<
>+              absl::optional<MoveableNoThrow>>::value);
>+}
>+
>+TEST(optionalTest, Destructor) {
>+  struct Trivial {};
>+
>+  struct NonTrivial {
>+    NonTrivial(const NonTrivial&) {}
>+    NonTrivial& operator=(const NonTrivial&) { return *this; }
>+    ~NonTrivial() {}
>+  };
>+
>+  EXPECT_TRUE(std::is_trivially_destructible<absl::optional<int>>::value);
>+  EXPECT_TRUE(std::is_trivially_destructible<absl::optional<Trivial>>::value);
>+  EXPECT_FALSE(
>+      std::is_trivially_destructible<absl::optional<NonTrivial>>::value);
>+}
>+
>+TEST(optionalTest, InPlaceConstructor) {
>+  constexpr absl::optional<ConstexprType> opt0{absl::in_place_t()};
>+  static_assert(opt0, "");
>+  static_assert((*opt0).x == ConstexprType::kCtorDefault, "");
>+  constexpr absl::optional<ConstexprType> opt1{absl::in_place_t(), 1};
>+  static_assert(opt1, "");
>+  static_assert((*opt1).x == ConstexprType::kCtorInt, "");
>+#ifndef ABSL_HAVE_NO_CONSTEXPR_INITIALIZER_LIST
>+  constexpr absl::optional<ConstexprType> opt2{absl::in_place_t(), {1, 2}};
>+  static_assert(opt2, "");
>+  static_assert((*opt2).x == ConstexprType::kCtorInitializerList, "");
>+#endif
>+
>+  // TODO(absl-team): uncomment these when std::is_constructible<T, Args&&...>
>+  // SFINAE is added to optional::optional(absl::in_place_t, Args&&...).
>+  // struct I {
>+  //   I(absl::in_place_t);
>+  // };
>+
>+  // EXPECT_FALSE((std::is_constructible<absl::optional<I>,
>+  // absl::in_place_t>::value));
>+  // EXPECT_FALSE((std::is_constructible<absl::optional<I>, const
>+  // absl::in_place_t&>::value));
>+}
>+
>+// template<U=T> optional(U&&);
>+TEST(optionalTest, ValueConstructor) {
>+  constexpr absl::optional<int> opt0(0);
>+  static_assert(opt0, "");
>+  static_assert(*opt0 == 0, "");
>+  EXPECT_TRUE((std::is_convertible<int, absl::optional<int>>::value));
>+  // Copy initialization ( = "abc") won't work due to optional(optional&&)
>+  // is not constexpr. Use list initialization instead. This invokes
>+  // absl::optional<ConstexprType>::absl::optional<U>(U&&), with U = const char
>+  // (&) [4], which direct-initializes the ConstexprType value held by the
>+  // optional via ConstexprType::ConstexprType(const char*).
>+  constexpr absl::optional<ConstexprType> opt1 = {"abc"};
>+  static_assert(opt1, "");
>+  static_assert(ConstexprType::kCtorConstChar == (*opt1).x, "");
>+  EXPECT_TRUE(
>+      (std::is_convertible<const char*, absl::optional<ConstexprType>>::value));
>+  // direct initialization
>+  constexpr absl::optional<ConstexprType> opt2{2};
>+  static_assert(opt2, "");
>+  static_assert(ConstexprType::kCtorInt == (*opt2).x, "");
>+  EXPECT_FALSE(
>+      (std::is_convertible<int, absl::optional<ConstexprType>>::value));
>+
>+  // this invokes absl::optional<int>::optional(int&&)
>+  // NOTE: this has different behavior than assignment, e.g.
>+  // "opt3 = {};" clears the optional rather than setting the value to 0
>+  // According to C++17 standard N4659 [over.ics.list] 16.3.3.1.5, (9.2)- "if
>+  // the initializer list has no elements, the implicit conversion is the
>+  // identity conversion", so `optional(int&&)` should be a better match than
>+  // `optional(optional&&)` which is a user-defined conversion.
>+  // Note: GCC 7 has a bug with this overload selection when compiled with
>+  // `-std=c++17`.
>+#if defined(__GNUC__) && !defined(__clang__) && __GNUC__ == 7 && \
>+    __cplusplus == 201703L
>+#define ABSL_GCC7_OVER_ICS_LIST_BUG 1
>+#endif
>+#ifndef ABSL_GCC7_OVER_ICS_LIST_BUG
>+  constexpr absl::optional<int> opt3({});
>+  static_assert(opt3, "");
>+  static_assert(*opt3 == 0, "");
>+#endif
>+
>+  // this invokes the move constructor with a default constructed optional
>+  // because non-template function is a better match than template function.
>+  absl::optional<ConstexprType> opt4({});
>+  EXPECT_FALSE(opt4);
>+}
>+
>+struct Implicit {};
>+
>+struct Explicit {};
>+
>+struct Convert {
>+  Convert(const Implicit&)  // NOLINT(runtime/explicit)
>+      : implicit(true), move(false) {}
>+  Convert(Implicit&&)  // NOLINT(runtime/explicit)
>+      : implicit(true), move(true) {}
>+  explicit Convert(const Explicit&) : implicit(false), move(false) {}
>+  explicit Convert(Explicit&&) : implicit(false), move(true) {}
>+
>+  bool implicit;
>+  bool move;
>+};
>+
>+struct ConvertFromOptional {
>+  ConvertFromOptional(const Implicit&)  // NOLINT(runtime/explicit)
>+      : implicit(true), move(false), from_optional(false) {}
>+  ConvertFromOptional(Implicit&&)  // NOLINT(runtime/explicit)
>+      : implicit(true), move(true), from_optional(false) {}
>+  ConvertFromOptional(
>+      const absl::optional<Implicit>&)  // NOLINT(runtime/explicit)
>+      : implicit(true), move(false), from_optional(true) {}
>+  ConvertFromOptional(absl::optional<Implicit>&&)  // NOLINT(runtime/explicit)
>+      : implicit(true), move(true), from_optional(true) {}
>+  explicit ConvertFromOptional(const Explicit&)
>+      : implicit(false), move(false), from_optional(false) {}
>+  explicit ConvertFromOptional(Explicit&&)
>+      : implicit(false), move(true), from_optional(false) {}
>+  explicit ConvertFromOptional(const absl::optional<Explicit>&)
>+      : implicit(false), move(false), from_optional(true) {}
>+  explicit ConvertFromOptional(absl::optional<Explicit>&&)
>+      : implicit(false), move(true), from_optional(true) {}
>+
>+  bool implicit;
>+  bool move;
>+  bool from_optional;
>+};
>+
>+TEST(optionalTest, ConvertingConstructor) {
>+  absl::optional<Implicit> i_empty;
>+  absl::optional<Implicit> i(absl::in_place);
>+  absl::optional<Explicit> e_empty;
>+  absl::optional<Explicit> e(absl::in_place);
>+  {
>+    // implicitly constructing absl::optional<Convert> from
>+    // absl::optional<Implicit>
>+    absl::optional<Convert> empty = i_empty;
>+    EXPECT_FALSE(empty);
>+    absl::optional<Convert> opt_copy = i;
>+    EXPECT_TRUE(opt_copy);
>+    EXPECT_TRUE(opt_copy->implicit);
>+    EXPECT_FALSE(opt_copy->move);
>+    absl::optional<Convert> opt_move = absl::optional<Implicit>(absl::in_place);
>+    EXPECT_TRUE(opt_move);
>+    EXPECT_TRUE(opt_move->implicit);
>+    EXPECT_TRUE(opt_move->move);
>+  }
>+  {
>+    // explicitly constructing absl::optional<Convert> from
>+    // absl::optional<Explicit>
>+    absl::optional<Convert> empty(e_empty);
>+    EXPECT_FALSE(empty);
>+    absl::optional<Convert> opt_copy(e);
>+    EXPECT_TRUE(opt_copy);
>+    EXPECT_FALSE(opt_copy->implicit);
>+    EXPECT_FALSE(opt_copy->move);
>+    EXPECT_FALSE((std::is_convertible<const absl::optional<Explicit>&,
>+                                      absl::optional<Convert>>::value));
>+    absl::optional<Convert> opt_move{absl::optional<Explicit>(absl::in_place)};
>+    EXPECT_TRUE(opt_move);
>+    EXPECT_FALSE(opt_move->implicit);
>+    EXPECT_TRUE(opt_move->move);
>+    EXPECT_FALSE((std::is_convertible<absl::optional<Explicit>&&,
>+                                      absl::optional<Convert>>::value));
>+  }
>+  {
>+    // implicitly constructing absl::optional<ConvertFromOptional> from
>+    // absl::optional<Implicit> via
>+    // ConvertFromOptional(absl::optional<Implicit>&&) check that
>+    // ConvertFromOptional(Implicit&&) is NOT called
>+    static_assert(
>+        std::is_convertible<absl::optional<Implicit>,
>+                            absl::optional<ConvertFromOptional>>::value,
>+        "");
>+    absl::optional<ConvertFromOptional> opt0 = i_empty;
>+    EXPECT_TRUE(opt0);
>+    EXPECT_TRUE(opt0->implicit);
>+    EXPECT_FALSE(opt0->move);
>+    EXPECT_TRUE(opt0->from_optional);
>+    absl::optional<ConvertFromOptional> opt1 = absl::optional<Implicit>();
>+    EXPECT_TRUE(opt1);
>+    EXPECT_TRUE(opt1->implicit);
>+    EXPECT_TRUE(opt1->move);
>+    EXPECT_TRUE(opt1->from_optional);
>+  }
>+  {
>+    // implicitly constructing absl::optional<ConvertFromOptional> from
>+    // absl::optional<Explicit> via
>+    // ConvertFromOptional(absl::optional<Explicit>&&) check that
>+    // ConvertFromOptional(Explicit&&) is NOT called
>+    absl::optional<ConvertFromOptional> opt0(e_empty);
>+    EXPECT_TRUE(opt0);
>+    EXPECT_FALSE(opt0->implicit);
>+    EXPECT_FALSE(opt0->move);
>+    EXPECT_TRUE(opt0->from_optional);
>+    EXPECT_FALSE(
>+        (std::is_convertible<const absl::optional<Explicit>&,
>+                             absl::optional<ConvertFromOptional>>::value));
>+    absl::optional<ConvertFromOptional> opt1{absl::optional<Explicit>()};
>+    EXPECT_TRUE(opt1);
>+    EXPECT_FALSE(opt1->implicit);
>+    EXPECT_TRUE(opt1->move);
>+    EXPECT_TRUE(opt1->from_optional);
>+    EXPECT_FALSE(
>+        (std::is_convertible<absl::optional<Explicit>&&,
>+                             absl::optional<ConvertFromOptional>>::value));
>+  }
>+}
>+
>+TEST(optionalTest, StructorBasic) {
>+  StructorListener listener;
>+  Listenable::listener = &listener;
>+  {
>+    absl::optional<Listenable> empty;
>+    EXPECT_FALSE(empty);
>+    absl::optional<Listenable> opt0(absl::in_place);
>+    EXPECT_TRUE(opt0);
>+    absl::optional<Listenable> opt1(absl::in_place, 1);
>+    EXPECT_TRUE(opt1);
>+    absl::optional<Listenable> opt2(absl::in_place, 1, 2);
>+    EXPECT_TRUE(opt2);
>+  }
>+  EXPECT_EQ(1, listener.construct0);
>+  EXPECT_EQ(1, listener.construct1);
>+  EXPECT_EQ(1, listener.construct2);
>+  EXPECT_EQ(3, listener.destruct);
>+}
>+
>+TEST(optionalTest, CopyMoveStructor) {
>+  StructorListener listener;
>+  Listenable::listener = &listener;
>+  absl::optional<Listenable> original(absl::in_place);
>+  EXPECT_EQ(1, listener.construct0);
>+  EXPECT_EQ(0, listener.copy);
>+  EXPECT_EQ(0, listener.move);
>+  absl::optional<Listenable> copy(original);
>+  EXPECT_EQ(1, listener.construct0);
>+  EXPECT_EQ(1, listener.copy);
>+  EXPECT_EQ(0, listener.move);
>+  absl::optional<Listenable> move(std::move(original));
>+  EXPECT_EQ(1, listener.construct0);
>+  EXPECT_EQ(1, listener.copy);
>+  EXPECT_EQ(1, listener.move);
>+}
>+
>+TEST(optionalTest, ListInit) {
>+  StructorListener listener;
>+  Listenable::listener = &listener;
>+  absl::optional<Listenable> listinit1(absl::in_place, {1});
>+  absl::optional<Listenable> listinit2(absl::in_place, {1, 2});
>+  EXPECT_EQ(2, listener.listinit);
>+}
>+
>+TEST(optionalTest, AssignFromNullopt) {
>+  absl::optional<int> opt(1);
>+  opt = absl::nullopt;
>+  EXPECT_FALSE(opt);
>+
>+  StructorListener listener;
>+  Listenable::listener = &listener;
>+  absl::optional<Listenable> opt1(absl::in_place);
>+  opt1 = absl::nullopt;
>+  EXPECT_FALSE(opt1);
>+  EXPECT_EQ(1, listener.construct0);
>+  EXPECT_EQ(1, listener.destruct);
>+
>+  EXPECT_TRUE((
>+      std::is_nothrow_assignable<absl::optional<int>, absl::nullopt_t>::value));
>+  EXPECT_TRUE((std::is_nothrow_assignable<absl::optional<Listenable>,
>+                                          absl::nullopt_t>::value));
>+}
>+
>+TEST(optionalTest, CopyAssignment) {
>+  const absl::optional<int> empty, opt1 = 1, opt2 = 2;
>+  absl::optional<int> empty_to_opt1, opt1_to_opt2, opt2_to_empty;
>+
>+  EXPECT_FALSE(empty_to_opt1);
>+  empty_to_opt1 = empty;
>+  EXPECT_FALSE(empty_to_opt1);
>+  empty_to_opt1 = opt1;
>+  EXPECT_TRUE(empty_to_opt1);
>+  EXPECT_EQ(1, empty_to_opt1.value());
>+
>+  EXPECT_FALSE(opt1_to_opt2);
>+  opt1_to_opt2 = opt1;
>+  EXPECT_TRUE(opt1_to_opt2);
>+  EXPECT_EQ(1, opt1_to_opt2.value());
>+  opt1_to_opt2 = opt2;
>+  EXPECT_TRUE(opt1_to_opt2);
>+  EXPECT_EQ(2, opt1_to_opt2.value());
>+
>+  EXPECT_FALSE(opt2_to_empty);
>+  opt2_to_empty = opt2;
>+  EXPECT_TRUE(opt2_to_empty);
>+  EXPECT_EQ(2, opt2_to_empty.value());
>+  opt2_to_empty = empty;
>+  EXPECT_FALSE(opt2_to_empty);
>+
>+  EXPECT_FALSE(std::is_copy_assignable<absl::optional<const int>>::value);
>+  EXPECT_TRUE(std::is_copy_assignable<absl::optional<Copyable>>::value);
>+  EXPECT_FALSE(std::is_copy_assignable<absl::optional<MoveableThrow>>::value);
>+  EXPECT_FALSE(std::is_copy_assignable<absl::optional<MoveableNoThrow>>::value);
>+  EXPECT_FALSE(std::is_copy_assignable<absl::optional<NonMovable>>::value);
>+
>+  EXPECT_TRUE(absl::is_trivially_copy_assignable<int>::value);
>+  EXPECT_TRUE(absl::is_trivially_copy_assignable<volatile int>::value);
>+
>+  struct Trivial {
>+    int i;
>+  };
>+  struct NonTrivial {
>+    NonTrivial& operator=(const NonTrivial&) { return *this; }
>+    int i;
>+  };
>+
>+  EXPECT_TRUE(absl::is_trivially_copy_assignable<Trivial>::value);
>+  EXPECT_FALSE(std::is_copy_assignable<const Trivial>::value);
>+  EXPECT_FALSE(std::is_copy_assignable<volatile Trivial>::value);
>+  EXPECT_TRUE(std::is_copy_assignable<NonTrivial>::value);
>+  EXPECT_FALSE(absl::is_trivially_copy_assignable<NonTrivial>::value);
>+
>+  // std::optional doesn't support volatile nontrivial types.
>+#ifndef ABSL_HAVE_STD_OPTIONAL
>+  {
>+    StructorListener listener;
>+    Listenable::listener = &listener;
>+
>+    absl::optional<volatile Listenable> empty, set(absl::in_place);
>+    EXPECT_EQ(1, listener.construct0);
>+    absl::optional<volatile Listenable> empty_to_empty, empty_to_set,
>+        set_to_empty(absl::in_place), set_to_set(absl::in_place);
>+    EXPECT_EQ(3, listener.construct0);
>+    empty_to_empty = empty;  // no effect
>+    empty_to_set = set;      // copy construct
>+    set_to_empty = empty;    // destruct
>+    set_to_set = set;        // copy assign
>+    EXPECT_EQ(1, listener.volatile_copy);
>+    EXPECT_EQ(0, listener.volatile_move);
>+    EXPECT_EQ(1, listener.destruct);
>+    EXPECT_EQ(1, listener.volatile_copy_assign);
>+  }
>+#endif  // ABSL_HAVE_STD_OPTIONAL
>+}
>+
>+TEST(optionalTest, MoveAssignment) {
>+  {
>+    StructorListener listener;
>+    Listenable::listener = &listener;
>+
>+    absl::optional<Listenable> empty1, empty2, set1(absl::in_place),
>+        set2(absl::in_place);
>+    EXPECT_EQ(2, listener.construct0);
>+    absl::optional<Listenable> empty_to_empty, empty_to_set,
>+        set_to_empty(absl::in_place), set_to_set(absl::in_place);
>+    EXPECT_EQ(4, listener.construct0);
>+    empty_to_empty = std::move(empty1);
>+    empty_to_set = std::move(set1);
>+    set_to_empty = std::move(empty2);
>+    set_to_set = std::move(set2);
>+    EXPECT_EQ(0, listener.copy);
>+    EXPECT_EQ(1, listener.move);
>+    EXPECT_EQ(1, listener.destruct);
>+    EXPECT_EQ(1, listener.move_assign);
>+  }
>+  // std::optional doesn't support volatile nontrivial types.
>+#ifndef ABSL_HAVE_STD_OPTIONAL
>+  {
>+    StructorListener listener;
>+    Listenable::listener = &listener;
>+
>+    absl::optional<volatile Listenable> empty1, empty2, set1(absl::in_place),
>+        set2(absl::in_place);
>+    EXPECT_EQ(2, listener.construct0);
>+    absl::optional<volatile Listenable> empty_to_empty, empty_to_set,
>+        set_to_empty(absl::in_place), set_to_set(absl::in_place);
>+    EXPECT_EQ(4, listener.construct0);
>+    empty_to_empty = std::move(empty1);  // no effect
>+    empty_to_set = std::move(set1);      // move construct
>+    set_to_empty = std::move(empty2);    // destruct
>+    set_to_set = std::move(set2);        // move assign
>+    EXPECT_EQ(0, listener.volatile_copy);
>+    EXPECT_EQ(1, listener.volatile_move);
>+    EXPECT_EQ(1, listener.destruct);
>+    EXPECT_EQ(1, listener.volatile_move_assign);
>+  }
>+#endif  // ABSL_HAVE_STD_OPTIONAL
>+  EXPECT_FALSE(std::is_move_assignable<absl::optional<const int>>::value);
>+  EXPECT_TRUE(std::is_move_assignable<absl::optional<Copyable>>::value);
>+  EXPECT_TRUE(std::is_move_assignable<absl::optional<MoveableThrow>>::value);
>+  EXPECT_TRUE(std::is_move_assignable<absl::optional<MoveableNoThrow>>::value);
>+  EXPECT_FALSE(std::is_move_assignable<absl::optional<NonMovable>>::value);
>+
>+  EXPECT_FALSE(
>+      std::is_nothrow_move_assignable<absl::optional<MoveableThrow>>::value);
>+  EXPECT_TRUE(
>+      std::is_nothrow_move_assignable<absl::optional<MoveableNoThrow>>::value);
>+}
>+
>+struct NoConvertToOptional {
>+  // disable implicit conversion from const NoConvertToOptional&
>+  // to absl::optional<NoConvertToOptional>.
>+  NoConvertToOptional(const NoConvertToOptional&) = delete;
>+};
>+
>+struct CopyConvert {
>+  CopyConvert(const NoConvertToOptional&);
>+  CopyConvert& operator=(const CopyConvert&) = delete;
>+  CopyConvert& operator=(const NoConvertToOptional&);
>+};
>+
>+struct CopyConvertFromOptional {
>+  CopyConvertFromOptional(const NoConvertToOptional&);
>+  CopyConvertFromOptional(const absl::optional<NoConvertToOptional>&);
>+  CopyConvertFromOptional& operator=(const CopyConvertFromOptional&) = delete;
>+  CopyConvertFromOptional& operator=(const NoConvertToOptional&);
>+  CopyConvertFromOptional& operator=(
>+      const absl::optional<NoConvertToOptional>&);
>+};
>+
>+struct MoveConvert {
>+  MoveConvert(NoConvertToOptional&&);
>+  MoveConvert& operator=(const MoveConvert&) = delete;
>+  MoveConvert& operator=(NoConvertToOptional&&);
>+};
>+
>+struct MoveConvertFromOptional {
>+  MoveConvertFromOptional(NoConvertToOptional&&);
>+  MoveConvertFromOptional(absl::optional<NoConvertToOptional>&&);
>+  MoveConvertFromOptional& operator=(const MoveConvertFromOptional&) = delete;
>+  MoveConvertFromOptional& operator=(NoConvertToOptional&&);
>+  MoveConvertFromOptional& operator=(absl::optional<NoConvertToOptional>&&);
>+};
>+
>+// template <typename U = T> absl::optional<T>& operator=(U&& v);
>+TEST(optionalTest, ValueAssignment) {
>+  absl::optional<int> opt;
>+  EXPECT_FALSE(opt);
>+  opt = 42;
>+  EXPECT_TRUE(opt);
>+  EXPECT_EQ(42, opt.value());
>+  opt = absl::nullopt;
>+  EXPECT_FALSE(opt);
>+  opt = 42;
>+  EXPECT_TRUE(opt);
>+  EXPECT_EQ(42, opt.value());
>+  opt = 43;
>+  EXPECT_TRUE(opt);
>+  EXPECT_EQ(43, opt.value());
>+  opt = {};  // this should clear optional
>+  EXPECT_FALSE(opt);
>+
>+  opt = {44};
>+  EXPECT_TRUE(opt);
>+  EXPECT_EQ(44, opt.value());
>+
>+  // U = const NoConvertToOptional&
>+  EXPECT_TRUE((std::is_assignable<absl::optional<CopyConvert>&,
>+                                  const NoConvertToOptional&>::value));
>+  // U = const absl::optional<NoConvertToOptional>&
>+  EXPECT_TRUE((std::is_assignable<absl::optional<CopyConvertFromOptional>&,
>+                                  const NoConvertToOptional&>::value));
>+  // U = const NoConvertToOptional& triggers SFINAE because
>+  // std::is_constructible_v<MoveConvert, const NoConvertToOptional&> is false
>+  EXPECT_FALSE((std::is_assignable<absl::optional<MoveConvert>&,
>+                                   const NoConvertToOptional&>::value));
>+  // U = NoConvertToOptional
>+  EXPECT_TRUE((std::is_assignable<absl::optional<MoveConvert>&,
>+                                  NoConvertToOptional&&>::value));
>+  // U = const NoConvertToOptional& triggers SFINAE because
>+  // std::is_constructible_v<MoveConvertFromOptional, const
>+  // NoConvertToOptional&> is false
>+  EXPECT_FALSE((std::is_assignable<absl::optional<MoveConvertFromOptional>&,
>+                                   const NoConvertToOptional&>::value));
>+  // U = NoConvertToOptional
>+  EXPECT_TRUE((std::is_assignable<absl::optional<MoveConvertFromOptional>&,
>+                                  NoConvertToOptional&&>::value));
>+  // U = const absl::optional<NoConvertToOptional>&
>+  EXPECT_TRUE(
>+      (std::is_assignable<absl::optional<CopyConvertFromOptional>&,
>+                          const absl::optional<NoConvertToOptional>&>::value));
>+  // U = absl::optional<NoConvertToOptional>
>+  EXPECT_TRUE(
>+      (std::is_assignable<absl::optional<MoveConvertFromOptional>&,
>+                          absl::optional<NoConvertToOptional>&&>::value));
>+}
>+
>+// template <typename U> absl::optional<T>& operator=(const absl::optional<U>&
>+// rhs); template <typename U> absl::optional<T>& operator=(absl::optional<U>&&
>+// rhs);
>+TEST(optionalTest, ConvertingAssignment) {
>+  absl::optional<int> opt_i;
>+  absl::optional<char> opt_c('c');
>+  opt_i = opt_c;
>+  EXPECT_TRUE(opt_i);
>+  EXPECT_EQ(*opt_c, *opt_i);
>+  opt_i = absl::optional<char>();
>+  EXPECT_FALSE(opt_i);
>+  opt_i = absl::optional<char>('d');
>+  EXPECT_TRUE(opt_i);
>+  EXPECT_EQ('d', *opt_i);
>+
>+  absl::optional<std::string> opt_str;
>+  absl::optional<const char*> opt_cstr("abc");
>+  opt_str = opt_cstr;
>+  EXPECT_TRUE(opt_str);
>+  EXPECT_EQ(std::string("abc"), *opt_str);
>+  opt_str = absl::optional<const char*>();
>+  EXPECT_FALSE(opt_str);
>+  opt_str = absl::optional<const char*>("def");
>+  EXPECT_TRUE(opt_str);
>+  EXPECT_EQ(std::string("def"), *opt_str);
>+
>+  // operator=(const absl::optional<U>&) with U = NoConvertToOptional
>+  EXPECT_TRUE(
>+      (std::is_assignable<absl::optional<CopyConvert>,
>+                          const absl::optional<NoConvertToOptional>&>::value));
>+  // operator=(const absl::optional<U>&) with U = NoConvertToOptional
>+  // triggers SFINAE because
>+  // std::is_constructible_v<MoveConvert, const NoConvertToOptional&> is false
>+  EXPECT_FALSE(
>+      (std::is_assignable<absl::optional<MoveConvert>&,
>+                          const absl::optional<NoConvertToOptional>&>::value));
>+  // operator=(absl::optional<U>&&) with U = NoConvertToOptional
>+  EXPECT_TRUE(
>+      (std::is_assignable<absl::optional<MoveConvert>&,
>+                          absl::optional<NoConvertToOptional>&&>::value));
>+  // operator=(const absl::optional<U>&) with U = NoConvertToOptional triggers
>+  // SFINAE because std::is_constructible_v<MoveConvertFromOptional, const
>+  // NoConvertToOptional&> is false. operator=(U&&) with U = const
>+  // absl::optional<NoConverToOptional>& triggers SFINAE because
>+  // std::is_constructible<MoveConvertFromOptional,
>+  // absl::optional<NoConvertToOptional>&&> is true.
>+  EXPECT_FALSE(
>+      (std::is_assignable<absl::optional<MoveConvertFromOptional>&,
>+                          const absl::optional<NoConvertToOptional>&>::value));
>+}
>+
>+TEST(optionalTest, ResetAndHasValue) {
>+  StructorListener listener;
>+  Listenable::listener = &listener;
>+  absl::optional<Listenable> opt;
>+  EXPECT_FALSE(opt);
>+  EXPECT_FALSE(opt.has_value());
>+  opt.emplace();
>+  EXPECT_TRUE(opt);
>+  EXPECT_TRUE(opt.has_value());
>+  opt.reset();
>+  EXPECT_FALSE(opt);
>+  EXPECT_FALSE(opt.has_value());
>+  EXPECT_EQ(1, listener.destruct);
>+  opt.reset();
>+  EXPECT_FALSE(opt);
>+  EXPECT_FALSE(opt.has_value());
>+
>+  constexpr absl::optional<int> empty;
>+  static_assert(!empty.has_value(), "");
>+  constexpr absl::optional<int> nonempty(1);
>+  static_assert(nonempty.has_value(), "");
>+}
>+
>+TEST(optionalTest, Emplace) {
>+  StructorListener listener;
>+  Listenable::listener = &listener;
>+  absl::optional<Listenable> opt;
>+  EXPECT_FALSE(opt);
>+  opt.emplace(1);
>+  EXPECT_TRUE(opt);
>+  opt.emplace(1, 2);
>+  EXPECT_EQ(1, listener.construct1);
>+  EXPECT_EQ(1, listener.construct2);
>+  EXPECT_EQ(1, listener.destruct);
>+
>+  absl::optional<std::string> o;
>+  EXPECT_TRUE((std::is_same<std::string&, decltype(o.emplace("abc"))>::value));
>+  std::string& ref = o.emplace("abc");
>+  EXPECT_EQ(&ref, &o.value());
>+}
>+
>+TEST(optionalTest, ListEmplace) {
>+  StructorListener listener;
>+  Listenable::listener = &listener;
>+  absl::optional<Listenable> opt;
>+  EXPECT_FALSE(opt);
>+  opt.emplace({1});
>+  EXPECT_TRUE(opt);
>+  opt.emplace({1, 2});
>+  EXPECT_EQ(2, listener.listinit);
>+  EXPECT_EQ(1, listener.destruct);
>+
>+  absl::optional<Listenable> o;
>+  EXPECT_TRUE((std::is_same<Listenable&, decltype(o.emplace({1}))>::value));
>+  Listenable& ref = o.emplace({1});
>+  EXPECT_EQ(&ref, &o.value());
>+}
>+
>+TEST(optionalTest, Swap) {
>+  absl::optional<int> opt_empty, opt1 = 1, opt2 = 2;
>+  EXPECT_FALSE(opt_empty);
>+  EXPECT_TRUE(opt1);
>+  EXPECT_EQ(1, opt1.value());
>+  EXPECT_TRUE(opt2);
>+  EXPECT_EQ(2, opt2.value());
>+  swap(opt_empty, opt1);
>+  EXPECT_FALSE(opt1);
>+  EXPECT_TRUE(opt_empty);
>+  EXPECT_EQ(1, opt_empty.value());
>+  EXPECT_TRUE(opt2);
>+  EXPECT_EQ(2, opt2.value());
>+  swap(opt_empty, opt1);
>+  EXPECT_FALSE(opt_empty);
>+  EXPECT_TRUE(opt1);
>+  EXPECT_EQ(1, opt1.value());
>+  EXPECT_TRUE(opt2);
>+  EXPECT_EQ(2, opt2.value());
>+  swap(opt1, opt2);
>+  EXPECT_FALSE(opt_empty);
>+  EXPECT_TRUE(opt1);
>+  EXPECT_EQ(2, opt1.value());
>+  EXPECT_TRUE(opt2);
>+  EXPECT_EQ(1, opt2.value());
>+
>+  EXPECT_TRUE(noexcept(opt1.swap(opt2)));
>+  EXPECT_TRUE(noexcept(swap(opt1, opt2)));
>+}
>+
>+template <int v>
>+struct DeletedOpAddr {
>+  constexpr static const int value = v;
>+  constexpr DeletedOpAddr() = default;
>+  constexpr const DeletedOpAddr<v>* operator&() const = delete;  // NOLINT
>+  DeletedOpAddr<v>* operator&() = delete;                        // NOLINT
>+};
>+
>+// The static_assert featuring a constexpr call to operator->() is commented out
>+// to document the fact that the current implementation of absl::optional<T>
>+// expects such usecases to be malformed and not compile.
>+TEST(optionalTest, OperatorAddr) {
>+  constexpr const int v = -1;
>+  {  // constexpr
>+    constexpr const absl::optional<DeletedOpAddr<v>> opt(absl::in_place_t{});
>+    static_assert(opt.has_value(), "");
>+    // static_assert(opt->value == v, "");
>+    static_assert((*opt).value == v, "");
>+  }
>+  {  // non-constexpr
>+    const absl::optional<DeletedOpAddr<v>> opt(absl::in_place_t{});
>+    EXPECT_TRUE(opt.has_value());
>+    EXPECT_TRUE(opt->value == v);
>+    EXPECT_TRUE((*opt).value == v);
>+  }
>+}
>+
>+TEST(optionalTest, PointerStuff) {
>+  absl::optional<std::string> opt(absl::in_place, "foo");
>+  EXPECT_EQ("foo", *opt);
>+  const auto& opt_const = opt;
>+  EXPECT_EQ("foo", *opt_const);
>+  EXPECT_EQ(opt->size(), 3);
>+  EXPECT_EQ(opt_const->size(), 3);
>+
>+  constexpr absl::optional<ConstexprType> opt1(1);
>+  static_assert((*opt1).x == ConstexprType::kCtorInt, "");
>+}
>+
>+// gcc has a bug pre 4.9.1 where it doesn't do correct overload resolution
>+// when overloads are const-qualified and *this is an raluve.
>+// Skip that test to make the build green again when using the old compiler.
>+// https://gcc.gnu.org/bugzilla/show_bug.cgi?id=59296 is fixed in 4.9.1.
>+#if defined(__GNUC__) && !defined(__clang__)
>+#define GCC_VERSION (__GNUC__ * 10000 \
>+                     + __GNUC_MINOR__ * 100 \
>+                     + __GNUC_PATCHLEVEL__)
>+#if GCC_VERSION < 40901
>+#define ABSL_SKIP_OVERLOAD_TEST_DUE_TO_GCC_BUG
>+#endif
>+#endif
>+
>+// MSVC has a bug with "cv-qualifiers in class construction", fixed in 2017. See
>+// https://docs.microsoft.com/en-us/cpp/cpp-conformance-improvements-2017#bug-fixes
>+// The compiler some incorrectly ingores the cv-qualifier when generating a
>+// class object via a constructor call. For example:
>+//
>+// class optional {
>+//   constexpr T&& value() &&;
>+//   constexpr const T&& value() const &&;
>+// }
>+//
>+// using COI = const absl::optional<int>;
>+// static_assert(2 == COI(2).value(), "");  // const &&
>+//
>+// This should invoke the "const &&" overload but since it ignores the const
>+// qualifier it finds the "&&" overload the best candidate.
>+#if defined(_MSC_VER) && _MSC_VER < 1910
>+#define ABSL_SKIP_OVERLOAD_TEST_DUE_TO_MSVC_BUG
>+#endif
>+
>+TEST(optionalTest, Value) {
>+  using O = absl::optional<std::string>;
>+  using CO = const absl::optional<std::string>;
>+  using OC = absl::optional<const std::string>;
>+  O lvalue(absl::in_place, "lvalue");
>+  CO clvalue(absl::in_place, "clvalue");
>+  OC lvalue_c(absl::in_place, "lvalue_c");
>+  EXPECT_EQ("lvalue", lvalue.value());
>+  EXPECT_EQ("clvalue", clvalue.value());
>+  EXPECT_EQ("lvalue_c", lvalue_c.value());
>+  EXPECT_EQ("xvalue", O(absl::in_place, "xvalue").value());
>+  EXPECT_EQ("xvalue_c", OC(absl::in_place, "xvalue_c").value());
>+#ifndef ABSL_SKIP_OVERLOAD_TEST_DUE_TO_GCC_BUG
>+  EXPECT_EQ("cxvalue", CO(absl::in_place, "cxvalue").value());
>+#endif
>+  EXPECT_EQ("&", TypeQuals(lvalue.value()));
>+  EXPECT_EQ("c&", TypeQuals(clvalue.value()));
>+  EXPECT_EQ("c&", TypeQuals(lvalue_c.value()));
>+  EXPECT_EQ("&&", TypeQuals(O(absl::in_place, "xvalue").value()));
>+#if !defined(ABSL_SKIP_OVERLOAD_TEST_DUE_TO_MSVC_BUG) && \
>+    !defined(ABSL_SKIP_OVERLOAD_TEST_DUE_TO_GCC_BUG)
>+  EXPECT_EQ("c&&", TypeQuals(CO(absl::in_place, "cxvalue").value()));
>+#endif
>+  EXPECT_EQ("c&&", TypeQuals(OC(absl::in_place, "xvalue_c").value()));
>+
>+  // test on volatile type
>+  using OV = absl::optional<volatile int>;
>+  OV lvalue_v(absl::in_place, 42);
>+  EXPECT_EQ(42, lvalue_v.value());
>+  EXPECT_EQ(42, OV(42).value());
>+  EXPECT_TRUE((std::is_same<volatile int&, decltype(lvalue_v.value())>::value));
>+  EXPECT_TRUE((std::is_same<volatile int&&, decltype(OV(42).value())>::value));
>+
>+  // test exception throw on value()
>+  absl::optional<int> empty;
>+#ifdef ABSL_HAVE_EXCEPTIONS
>+  EXPECT_THROW(empty.value(), absl::bad_optional_access);
>+#else
>+  EXPECT_DEATH(empty.value(), "Bad optional access");
>+#endif
>+
>+  // test constexpr value()
>+  constexpr absl::optional<int> o1(1);
>+  static_assert(1 == o1.value(), "");  // const &
>+#if !defined(ABSL_SKIP_OVERLOAD_TEST_DUE_TO_MSVC_BUG) && \
>+    !defined(ABSL_SKIP_OVERLOAD_TEST_DUE_TO_GCC_BUG)
>+  using COI = const absl::optional<int>;
>+  static_assert(2 == COI(2).value(), "");  // const &&
>+#endif
>+}
>+
>+TEST(optionalTest, DerefOperator) {
>+  using O = absl::optional<std::string>;
>+  using CO = const absl::optional<std::string>;
>+  using OC = absl::optional<const std::string>;
>+  O lvalue(absl::in_place, "lvalue");
>+  CO clvalue(absl::in_place, "clvalue");
>+  OC lvalue_c(absl::in_place, "lvalue_c");
>+  EXPECT_EQ("lvalue", *lvalue);
>+  EXPECT_EQ("clvalue", *clvalue);
>+  EXPECT_EQ("lvalue_c", *lvalue_c);
>+  EXPECT_EQ("xvalue", *O(absl::in_place, "xvalue"));
>+  EXPECT_EQ("xvalue_c", *OC(absl::in_place, "xvalue_c"));
>+#ifndef ABSL_SKIP_OVERLOAD_TEST_DUE_TO_GCC_BUG
>+  EXPECT_EQ("cxvalue", *CO(absl::in_place, "cxvalue"));
>+#endif
>+  EXPECT_EQ("&", TypeQuals(*lvalue));
>+  EXPECT_EQ("c&", TypeQuals(*clvalue));
>+  EXPECT_EQ("&&", TypeQuals(*O(absl::in_place, "xvalue")));
>+#if !defined(ABSL_SKIP_OVERLOAD_TEST_DUE_TO_MSVC_BUG) && \
>+    !defined(ABSL_SKIP_OVERLOAD_TEST_DUE_TO_GCC_BUG)
>+  EXPECT_EQ("c&&", TypeQuals(*CO(absl::in_place, "cxvalue")));
>+#endif
>+  EXPECT_EQ("c&&", TypeQuals(*OC(absl::in_place, "xvalue_c")));
>+
>+  // test on volatile type
>+  using OV = absl::optional<volatile int>;
>+  OV lvalue_v(absl::in_place, 42);
>+  EXPECT_EQ(42, *lvalue_v);
>+  EXPECT_EQ(42, *OV(42));
>+  EXPECT_TRUE((std::is_same<volatile int&, decltype(*lvalue_v)>::value));
>+  EXPECT_TRUE((std::is_same<volatile int&&, decltype(*OV(42))>::value));
>+
>+  constexpr absl::optional<int> opt1(1);
>+  static_assert(*opt1 == 1, "");
>+#if !defined(ABSL_SKIP_OVERLOAD_TEST_DUE_TO_MSVC_BUG) && \
>+    !defined(ABSL_SKIP_OVERLOAD_TEST_DUE_TO_GCC_BUG)
>+  using COI = const absl::optional<int>;
>+  static_assert(*COI(2) == 2, "");
>+#endif
>+}
>+
>+TEST(optionalTest, ValueOr) {
>+  absl::optional<double> opt_empty, opt_set = 1.2;
>+  EXPECT_EQ(42.0, opt_empty.value_or(42));
>+  EXPECT_EQ(1.2, opt_set.value_or(42));
>+  EXPECT_EQ(42.0, absl::optional<double>().value_or(42));
>+  EXPECT_EQ(1.2, absl::optional<double>(1.2).value_or(42));
>+
>+  constexpr absl::optional<double> copt_empty, copt_set = {1.2};
>+  static_assert(42.0 == copt_empty.value_or(42), "");
>+  static_assert(1.2 == copt_set.value_or(42), "");
>+#ifndef ABSL_SKIP_OVERLOAD_TEST_DUE_TO_MSVC_BUG
>+  using COD = const absl::optional<double>;
>+  static_assert(42.0 == COD().value_or(42), "");
>+  static_assert(1.2 == COD(1.2).value_or(42), "");
>+#endif
>+}
>+
>+// make_optional cannot be constexpr until C++17
>+TEST(optionalTest, make_optional) {
>+  auto opt_int = absl::make_optional(42);
>+  EXPECT_TRUE((std::is_same<decltype(opt_int), absl::optional<int>>::value));
>+  EXPECT_EQ(42, opt_int);
>+
>+  StructorListener listener;
>+  Listenable::listener = &listener;
>+
>+  absl::optional<Listenable> opt0 = absl::make_optional<Listenable>();
>+  EXPECT_EQ(1, listener.construct0);
>+  absl::optional<Listenable> opt1 = absl::make_optional<Listenable>(1);
>+  EXPECT_EQ(1, listener.construct1);
>+  absl::optional<Listenable> opt2 = absl::make_optional<Listenable>(1, 2);
>+  EXPECT_EQ(1, listener.construct2);
>+  absl::optional<Listenable> opt3 = absl::make_optional<Listenable>({1});
>+  absl::optional<Listenable> opt4 = absl::make_optional<Listenable>({1, 2});
>+  EXPECT_EQ(2, listener.listinit);
>+
>+  // Constexpr tests on trivially copyable types
>+  // optional<T> has trivial copy/move ctors when T is trivially copyable.
>+  // For nontrivial types with constexpr constructors, we need copy elision in
>+  // C++17 for make_optional to be constexpr.
>+  {
>+    constexpr absl::optional<int> c_opt = absl::make_optional(42);
>+    static_assert(c_opt.value() == 42, "");
>+  }
>+  {
>+    struct TrivialCopyable {
>+      constexpr TrivialCopyable() : x(0) {}
>+      constexpr explicit TrivialCopyable(int i) : x(i) {}
>+      int x;
>+    };
>+
>+    constexpr TrivialCopyable v;
>+    constexpr absl::optional<TrivialCopyable> c_opt0 = absl::make_optional(v);
>+    static_assert((*c_opt0).x == 0, "");
>+    constexpr absl::optional<TrivialCopyable> c_opt1 =
>+        absl::make_optional<TrivialCopyable>();
>+    static_assert((*c_opt1).x == 0, "");
>+    constexpr absl::optional<TrivialCopyable> c_opt2 =
>+        absl::make_optional<TrivialCopyable>(42);
>+    static_assert((*c_opt2).x == 42, "");
>+  }
>+}
>+
>+template <typename T, typename U>
>+void optionalTest_Comparisons_EXPECT_LESS(T x, U y) {
>+  EXPECT_FALSE(x == y);
>+  EXPECT_TRUE(x != y);
>+  EXPECT_TRUE(x < y);
>+  EXPECT_FALSE(x > y);
>+  EXPECT_TRUE(x <= y);
>+  EXPECT_FALSE(x >= y);
>+}
>+
>+template <typename T, typename U>
>+void optionalTest_Comparisons_EXPECT_SAME(T x, U y) {
>+  EXPECT_TRUE(x == y);
>+  EXPECT_FALSE(x != y);
>+  EXPECT_FALSE(x < y);
>+  EXPECT_FALSE(x > y);
>+  EXPECT_TRUE(x <= y);
>+  EXPECT_TRUE(x >= y);
>+}
>+
>+template <typename T, typename U>
>+void optionalTest_Comparisons_EXPECT_GREATER(T x, U y) {
>+  EXPECT_FALSE(x == y);
>+  EXPECT_TRUE(x != y);
>+  EXPECT_FALSE(x < y);
>+  EXPECT_TRUE(x > y);
>+  EXPECT_FALSE(x <= y);
>+  EXPECT_TRUE(x >= y);
>+}
>+
>+
>+template <typename T, typename U, typename V>
>+void TestComparisons() {
>+  absl::optional<T> ae, a2{2}, a4{4};
>+  absl::optional<U> be, b2{2}, b4{4};
>+  V v3 = 3;
>+
>+  // LHS: absl::nullopt, ae, a2, v3, a4
>+  // RHS: absl::nullopt, be, b2, v3, b4
>+
>+  // optionalTest_Comparisons_EXPECT_NOT_TO_WORK(absl::nullopt,absl::nullopt);
>+  optionalTest_Comparisons_EXPECT_SAME(absl::nullopt, be);
>+  optionalTest_Comparisons_EXPECT_LESS(absl::nullopt, b2);
>+  // optionalTest_Comparisons_EXPECT_NOT_TO_WORK(absl::nullopt,v3);
>+  optionalTest_Comparisons_EXPECT_LESS(absl::nullopt, b4);
>+
>+  optionalTest_Comparisons_EXPECT_SAME(ae, absl::nullopt);
>+  optionalTest_Comparisons_EXPECT_SAME(ae, be);
>+  optionalTest_Comparisons_EXPECT_LESS(ae, b2);
>+  optionalTest_Comparisons_EXPECT_LESS(ae, v3);
>+  optionalTest_Comparisons_EXPECT_LESS(ae, b4);
>+
>+  optionalTest_Comparisons_EXPECT_GREATER(a2, absl::nullopt);
>+  optionalTest_Comparisons_EXPECT_GREATER(a2, be);
>+  optionalTest_Comparisons_EXPECT_SAME(a2, b2);
>+  optionalTest_Comparisons_EXPECT_LESS(a2, v3);
>+  optionalTest_Comparisons_EXPECT_LESS(a2, b4);
>+
>+  // optionalTest_Comparisons_EXPECT_NOT_TO_WORK(v3,absl::nullopt);
>+  optionalTest_Comparisons_EXPECT_GREATER(v3, be);
>+  optionalTest_Comparisons_EXPECT_GREATER(v3, b2);
>+  optionalTest_Comparisons_EXPECT_SAME(v3, v3);
>+  optionalTest_Comparisons_EXPECT_LESS(v3, b4);
>+
>+  optionalTest_Comparisons_EXPECT_GREATER(a4, absl::nullopt);
>+  optionalTest_Comparisons_EXPECT_GREATER(a4, be);
>+  optionalTest_Comparisons_EXPECT_GREATER(a4, b2);
>+  optionalTest_Comparisons_EXPECT_GREATER(a4, v3);
>+  optionalTest_Comparisons_EXPECT_SAME(a4, b4);
>+}
>+
>+struct Int1 {
>+  Int1() = default;
>+  Int1(int i) : i(i) {}  // NOLINT(runtime/explicit)
>+  int i;
>+};
>+
>+struct Int2 {
>+  Int2() = default;
>+  Int2(int i) : i(i) {}  // NOLINT(runtime/explicit)
>+  int i;
>+};
>+
>+// comparison between Int1 and Int2
>+constexpr bool operator==(const Int1& lhs, const Int2& rhs) {
>+  return lhs.i == rhs.i;
>+}
>+constexpr bool operator!=(const Int1& lhs, const Int2& rhs) {
>+  return !(lhs == rhs);
>+}
>+constexpr bool operator<(const Int1& lhs, const Int2& rhs) {
>+  return lhs.i < rhs.i;
>+}
>+constexpr bool operator<=(const Int1& lhs, const Int2& rhs) {
>+  return lhs < rhs || lhs == rhs;
>+}
>+constexpr bool operator>(const Int1& lhs, const Int2& rhs) {
>+  return !(lhs <= rhs);
>+}
>+constexpr bool operator>=(const Int1& lhs, const Int2& rhs) {
>+  return !(lhs < rhs);
>+}
>+
>+TEST(optionalTest, Comparisons) {
>+  TestComparisons<int, int, int>();
>+  TestComparisons<const int, int, int>();
>+  TestComparisons<Int1, int, int>();
>+  TestComparisons<int, Int2, int>();
>+  TestComparisons<Int1, Int2, int>();
>+
>+  // compare absl::optional<std::string> with const char*
>+  absl::optional<std::string> opt_str = "abc";
>+  const char* cstr = "abc";
>+  EXPECT_TRUE(opt_str == cstr);
>+  // compare absl::optional<std::string> with absl::optional<const char*>
>+  absl::optional<const char*> opt_cstr = cstr;
>+  EXPECT_TRUE(opt_str == opt_cstr);
>+  // compare absl::optional<std::string> with absl::optional<absl::string_view>
>+  absl::optional<absl::string_view> e1;
>+  absl::optional<std::string> e2;
>+  EXPECT_TRUE(e1 == e2);
>+}
>+
>+
>+TEST(optionalTest, SwapRegression) {
>+  StructorListener listener;
>+  Listenable::listener = &listener;
>+
>+  {
>+    absl::optional<Listenable> a;
>+    absl::optional<Listenable> b(absl::in_place);
>+    a.swap(b);
>+  }
>+
>+  EXPECT_EQ(1, listener.construct0);
>+  EXPECT_EQ(1, listener.move);
>+  EXPECT_EQ(2, listener.destruct);
>+
>+  {
>+    absl::optional<Listenable> a(absl::in_place);
>+    absl::optional<Listenable> b;
>+    a.swap(b);
>+  }
>+
>+  EXPECT_EQ(2, listener.construct0);
>+  EXPECT_EQ(2, listener.move);
>+  EXPECT_EQ(4, listener.destruct);
>+}
>+
>+TEST(optionalTest, BigStringLeakCheck) {
>+  constexpr size_t n = 1 << 16;
>+
>+  using OS = absl::optional<std::string>;
>+
>+  OS a;
>+  OS b = absl::nullopt;
>+  OS c = std::string(n, 'c');
>+  std::string sd(n, 'd');
>+  OS d = sd;
>+  OS e(absl::in_place, n, 'e');
>+  OS f;
>+  f.emplace(n, 'f');
>+
>+  OS ca(a);
>+  OS cb(b);
>+  OS cc(c);
>+  OS cd(d);
>+  OS ce(e);
>+
>+  OS oa;
>+  OS ob = absl::nullopt;
>+  OS oc = std::string(n, 'c');
>+  std::string sod(n, 'd');
>+  OS od = sod;
>+  OS oe(absl::in_place, n, 'e');
>+  OS of;
>+  of.emplace(n, 'f');
>+
>+  OS ma(std::move(oa));
>+  OS mb(std::move(ob));
>+  OS mc(std::move(oc));
>+  OS md(std::move(od));
>+  OS me(std::move(oe));
>+  OS mf(std::move(of));
>+
>+  OS aa1;
>+  OS ab1 = absl::nullopt;
>+  OS ac1 = std::string(n, 'c');
>+  std::string sad1(n, 'd');
>+  OS ad1 = sad1;
>+  OS ae1(absl::in_place, n, 'e');
>+  OS af1;
>+  af1.emplace(n, 'f');
>+
>+  OS aa2;
>+  OS ab2 = absl::nullopt;
>+  OS ac2 = std::string(n, 'c');
>+  std::string sad2(n, 'd');
>+  OS ad2 = sad2;
>+  OS ae2(absl::in_place, n, 'e');
>+  OS af2;
>+  af2.emplace(n, 'f');
>+
>+  aa1 = af2;
>+  ab1 = ae2;
>+  ac1 = ad2;
>+  ad1 = ac2;
>+  ae1 = ab2;
>+  af1 = aa2;
>+
>+  OS aa3;
>+  OS ab3 = absl::nullopt;
>+  OS ac3 = std::string(n, 'c');
>+  std::string sad3(n, 'd');
>+  OS ad3 = sad3;
>+  OS ae3(absl::in_place, n, 'e');
>+  OS af3;
>+  af3.emplace(n, 'f');
>+
>+  aa3 = absl::nullopt;
>+  ab3 = absl::nullopt;
>+  ac3 = absl::nullopt;
>+  ad3 = absl::nullopt;
>+  ae3 = absl::nullopt;
>+  af3 = absl::nullopt;
>+
>+  OS aa4;
>+  OS ab4 = absl::nullopt;
>+  OS ac4 = std::string(n, 'c');
>+  std::string sad4(n, 'd');
>+  OS ad4 = sad4;
>+  OS ae4(absl::in_place, n, 'e');
>+  OS af4;
>+  af4.emplace(n, 'f');
>+
>+  aa4 = OS(absl::in_place, n, 'a');
>+  ab4 = OS(absl::in_place, n, 'b');
>+  ac4 = OS(absl::in_place, n, 'c');
>+  ad4 = OS(absl::in_place, n, 'd');
>+  ae4 = OS(absl::in_place, n, 'e');
>+  af4 = OS(absl::in_place, n, 'f');
>+
>+  OS aa5;
>+  OS ab5 = absl::nullopt;
>+  OS ac5 = std::string(n, 'c');
>+  std::string sad5(n, 'd');
>+  OS ad5 = sad5;
>+  OS ae5(absl::in_place, n, 'e');
>+  OS af5;
>+  af5.emplace(n, 'f');
>+
>+  std::string saa5(n, 'a');
>+  std::string sab5(n, 'a');
>+  std::string sac5(n, 'a');
>+  std::string sad52(n, 'a');
>+  std::string sae5(n, 'a');
>+  std::string saf5(n, 'a');
>+
>+  aa5 = saa5;
>+  ab5 = sab5;
>+  ac5 = sac5;
>+  ad5 = sad52;
>+  ae5 = sae5;
>+  af5 = saf5;
>+
>+  OS aa6;
>+  OS ab6 = absl::nullopt;
>+  OS ac6 = std::string(n, 'c');
>+  std::string sad6(n, 'd');
>+  OS ad6 = sad6;
>+  OS ae6(absl::in_place, n, 'e');
>+  OS af6;
>+  af6.emplace(n, 'f');
>+
>+  aa6 = std::string(n, 'a');
>+  ab6 = std::string(n, 'b');
>+  ac6 = std::string(n, 'c');
>+  ad6 = std::string(n, 'd');
>+  ae6 = std::string(n, 'e');
>+  af6 = std::string(n, 'f');
>+
>+  OS aa7;
>+  OS ab7 = absl::nullopt;
>+  OS ac7 = std::string(n, 'c');
>+  std::string sad7(n, 'd');
>+  OS ad7 = sad7;
>+  OS ae7(absl::in_place, n, 'e');
>+  OS af7;
>+  af7.emplace(n, 'f');
>+
>+  aa7.emplace(n, 'A');
>+  ab7.emplace(n, 'B');
>+  ac7.emplace(n, 'C');
>+  ad7.emplace(n, 'D');
>+  ae7.emplace(n, 'E');
>+  af7.emplace(n, 'F');
>+}
>+
>+TEST(optionalTest, MoveAssignRegression) {
>+  StructorListener listener;
>+  Listenable::listener = &listener;
>+
>+  {
>+    absl::optional<Listenable> a;
>+    Listenable b;
>+    a = std::move(b);
>+  }
>+
>+  EXPECT_EQ(1, listener.construct0);
>+  EXPECT_EQ(1, listener.move);
>+  EXPECT_EQ(2, listener.destruct);
>+}
>+
>+TEST(optionalTest, ValueType) {
>+  EXPECT_TRUE((std::is_same<absl::optional<int>::value_type, int>::value));
>+  EXPECT_TRUE(
>+      (std::is_same<absl::optional<std::string>::value_type, std::string>::value));
>+  EXPECT_FALSE(
>+      (std::is_same<absl::optional<int>::value_type, absl::nullopt_t>::value));
>+}
>+
>+template <typename T>
>+struct is_hash_enabled_for {
>+  template <typename U, typename = decltype(std::hash<U>()(std::declval<U>()))>
>+  static std::true_type test(int);
>+
>+  template <typename U>
>+  static std::false_type test(...);
>+
>+  static constexpr bool value = decltype(test<T>(0))::value;
>+};
>+
>+TEST(optionalTest, Hash) {
>+  std::hash<absl::optional<int>> hash;
>+  std::set<size_t> hashcodes;
>+  hashcodes.insert(hash(absl::nullopt));
>+  for (int i = 0; i < 100; ++i) {
>+    hashcodes.insert(hash(i));
>+  }
>+  EXPECT_GT(hashcodes.size(), 90);
>+
>+  static_assert(is_hash_enabled_for<absl::optional<int>>::value, "");
>+  static_assert(is_hash_enabled_for<absl::optional<Hashable>>::value, "");
>+
>+#if defined(_MSC_VER) || (defined(_LIBCPP_VERSION) && \
>+                          _LIBCPP_VERSION < 4000 && _LIBCPP_STD_VER > 11)
>+  // For MSVC and libc++ (< 4.0 and c++14), std::hash primary template has a
>+  // static_assert to catch any user-defined type that doesn't provide a hash
>+  // specialization. So instantiating std::hash<absl::optional<T>> will result
>+  // in a hard error which is not SFINAE friendly.
>+#define ABSL_STD_HASH_NOT_SFINAE_FRIENDLY 1
>+#endif
>+
>+#ifndef ABSL_STD_HASH_NOT_SFINAE_FRIENDLY
>+  static_assert(!is_hash_enabled_for<absl::optional<NonHashable>>::value, "");
>+#endif
>+
>+  // libstdc++ std::optional is missing remove_const_t, i.e. it's using
>+  // std::hash<T> rather than std::hash<std::remove_const_t<T>>.
>+  // Reference: https://gcc.gnu.org/bugzilla/show_bug.cgi?id=82262
>+#ifndef __GLIBCXX__
>+  static_assert(is_hash_enabled_for<absl::optional<const int>>::value, "");
>+  static_assert(is_hash_enabled_for<absl::optional<const Hashable>>::value, "");
>+  std::hash<absl::optional<const int>> c_hash;
>+  for (int i = 0; i < 100; ++i) {
>+    EXPECT_EQ(hash(i), c_hash(i));
>+  }
>+#endif
>+}
>+
>+struct MoveMeNoThrow {
>+  MoveMeNoThrow() : x(0) {}
>+  [[noreturn]] MoveMeNoThrow(const MoveMeNoThrow& other) : x(other.x) {
>+    ABSL_RAW_LOG(FATAL, "Should not be called.");
>+    abort();
>+  }
>+  MoveMeNoThrow(MoveMeNoThrow&& other) noexcept : x(other.x) {}
>+  int x;
>+};
>+
>+struct MoveMeThrow {
>+  MoveMeThrow() : x(0) {}
>+  MoveMeThrow(const MoveMeThrow& other) : x(other.x) {}
>+  MoveMeThrow(MoveMeThrow&& other) : x(other.x) {}
>+  int x;
>+};
>+
>+TEST(optionalTest, NoExcept) {
>+  static_assert(
>+      std::is_nothrow_move_constructible<absl::optional<MoveMeNoThrow>>::value,
>+      "");
>+#ifndef ABSL_HAVE_STD_OPTIONAL
>+  static_assert(absl::default_allocator_is_nothrow::value ==
>+                    std::is_nothrow_move_constructible<
>+                        absl::optional<MoveMeThrow>>::value,
>+                "");
>+#endif
>+  std::vector<absl::optional<MoveMeNoThrow>> v;
>+  for (int i = 0; i < 10; ++i) v.emplace_back();
>+}
>+
>+struct AnyLike {
>+  AnyLike(AnyLike&&) = default;
>+  AnyLike(const AnyLike&) = default;
>+
>+  template <typename ValueType,
>+            typename T = typename std::decay<ValueType>::type,
>+            typename std::enable_if<
>+                !absl::disjunction<
>+                    std::is_same<AnyLike, T>,
>+                    absl::negation<std::is_copy_constructible<T>>>::value,
>+                int>::type = 0>
>+  AnyLike(ValueType&&) {}  // NOLINT(runtime/explicit)
>+
>+  AnyLike& operator=(AnyLike&&) = default;
>+  AnyLike& operator=(const AnyLike&) = default;
>+
>+  template <typename ValueType,
>+            typename T = typename std::decay<ValueType>::type>
>+  typename std::enable_if<
>+      absl::conjunction<absl::negation<std::is_same<AnyLike, T>>,
>+                        std::is_copy_constructible<T>>::value,
>+      AnyLike&>::type
>+  operator=(ValueType&& /* rhs */) {
>+    return *this;
>+  }
>+};
>+
>+TEST(optionalTest, ConstructionConstraints) {
>+  EXPECT_TRUE((std::is_constructible<AnyLike, absl::optional<AnyLike>>::value));
>+
>+  EXPECT_TRUE(
>+      (std::is_constructible<AnyLike, const absl::optional<AnyLike>&>::value));
>+
>+  EXPECT_TRUE((std::is_constructible<absl::optional<AnyLike>, AnyLike>::value));
>+  EXPECT_TRUE(
>+      (std::is_constructible<absl::optional<AnyLike>, const AnyLike&>::value));
>+
>+  EXPECT_TRUE((std::is_convertible<absl::optional<AnyLike>, AnyLike>::value));
>+
>+  EXPECT_TRUE(
>+      (std::is_convertible<const absl::optional<AnyLike>&, AnyLike>::value));
>+
>+  EXPECT_TRUE((std::is_convertible<AnyLike, absl::optional<AnyLike>>::value));
>+  EXPECT_TRUE(
>+      (std::is_convertible<const AnyLike&, absl::optional<AnyLike>>::value));
>+
>+  EXPECT_TRUE(std::is_move_constructible<absl::optional<AnyLike>>::value);
>+  EXPECT_TRUE(std::is_copy_constructible<absl::optional<AnyLike>>::value);
>+}
>+
>+TEST(optionalTest, AssignmentConstraints) {
>+  EXPECT_TRUE((std::is_assignable<AnyLike&, absl::optional<AnyLike>>::value));
>+  EXPECT_TRUE(
>+      (std::is_assignable<AnyLike&, const absl::optional<AnyLike>&>::value));
>+  EXPECT_TRUE((std::is_assignable<absl::optional<AnyLike>&, AnyLike>::value));
>+  EXPECT_TRUE(
>+      (std::is_assignable<absl::optional<AnyLike>&, const AnyLike&>::value));
>+  EXPECT_TRUE(std::is_move_assignable<absl::optional<AnyLike>>::value);
>+  EXPECT_TRUE(std::is_copy_assignable<absl::optional<AnyLike>>::value);
>+}
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/span.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/span.h
>new file mode 100644
>index 00000000000..76be819ecca
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/span.h
>@@ -0,0 +1,749 @@
>+//
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// -----------------------------------------------------------------------------
>+// span.h
>+// -----------------------------------------------------------------------------
>+//
>+// This header file defines a `Span<T>` type for holding a view of an existing
>+// array of data. The `Span` object, much like the `absl::string_view` object,
>+// does not own such data itself. A span provides a lightweight way to pass
>+// around view of such data.
>+//
>+// Additionally, this header file defines `MakeSpan()` and `MakeConstSpan()`
>+// factory functions, for clearly creating spans of type `Span<T>` or read-only
>+// `Span<const T>` when such types may be difficult to identify due to issues
>+// with implicit conversion.
>+//
>+// The C++ standards committee currently has a proposal for a `std::span` type,
>+// (http://wg21.link/p0122), which is not yet part of the standard (though may
>+// become part of C++20). As of August 2017, the differences between
>+// `absl::Span` and this proposal are:
>+//    * `absl::Span` uses `size_t` for `size_type`
>+//    * `absl::Span` has no `operator()`
>+//    * `absl::Span` has no constructors for `std::unique_ptr` or
>+//      `std::shared_ptr`
>+//    * `absl::Span` has the factory functions `MakeSpan()` and
>+//      `MakeConstSpan()`
>+//    * `absl::Span` has `front()` and `back()` methods
>+//    * bounds-checked access to `absl::Span` is accomplished with `at()`
>+//    * `absl::Span` has compiler-provided move and copy constructors and
>+//      assignment. This is due to them being specified as `constexpr`, but that
>+//      implies const in C++11.
>+//    * `absl::Span` has no `element_type` or `index_type` typedefs
>+//    * A read-only `absl::Span<const T>` can be implicitly constructed from an
>+//      initializer list.
>+//    * `absl::Span` has no `bytes()`, `size_bytes()`, `as_bytes()`, or
>+//      `as_mutable_bytes()` methods
>+//    * `absl::Span` has no static extent template parameter, nor constructors
>+//      which exist only because of the static extent parameter.
>+//    * `absl::Span` has an explicit mutable-reference constructor
>+//
>+// For more information, see the class comments below.
>+#ifndef ABSL_TYPES_SPAN_H_
>+#define ABSL_TYPES_SPAN_H_
>+
>+#include <algorithm>
>+#include <cassert>
>+#include <cstddef>
>+#include <initializer_list>
>+#include <iterator>
>+#include <string>
>+#include <type_traits>
>+#include <utility>
>+
>+#include "absl/algorithm/algorithm.h"
>+#include "absl/base/internal/throw_delegate.h"
>+#include "absl/base/macros.h"
>+#include "absl/base/optimization.h"
>+#include "absl/base/port.h"
>+#include "absl/meta/type_traits.h"
>+
>+namespace absl {
>+
>+template <typename T>
>+class Span;
>+
>+namespace span_internal {
>+// A constexpr min function
>+constexpr size_t Min(size_t a, size_t b) noexcept { return a < b ? a : b; }
>+
>+// Wrappers for access to container data pointers.
>+template <typename C>
>+constexpr auto GetDataImpl(C& c, char) noexcept  // NOLINT(runtime/references)
>+    -> decltype(c.data()) {
>+  return c.data();
>+}
>+
>+// Before C++17, std::string::data returns a const char* in all cases.
>+inline char* GetDataImpl(std::string& s,  // NOLINT(runtime/references)
>+                         int) noexcept {
>+  return &s[0];
>+}
>+
>+template <typename C>
>+constexpr auto GetData(C& c) noexcept  // NOLINT(runtime/references)
>+    -> decltype(GetDataImpl(c, 0)) {
>+  return GetDataImpl(c, 0);
>+}
>+
>+// Detection idioms for size() and data().
>+template <typename C>
>+using HasSize =
>+    std::is_integral<absl::decay_t<decltype(std::declval<C&>().size())>>;
>+
>+// We want to enable conversion from vector<T*> to Span<const T* const> but
>+// disable conversion from vector<Derived> to Span<Base>. Here we use
>+// the fact that U** is convertible to Q* const* if and only if Q is the same
>+// type or a more cv-qualified version of U.  We also decay the result type of
>+// data() to avoid problems with classes which have a member function data()
>+// which returns a reference.
>+template <typename T, typename C>
>+using HasData =
>+    std::is_convertible<absl::decay_t<decltype(GetData(std::declval<C&>()))>*,
>+                        T* const*>;
>+
>+// Extracts value type from a Container
>+template <typename C>
>+struct ElementType {
>+  using type = typename absl::remove_reference_t<C>::value_type;
>+};
>+
>+template <typename T, size_t N>
>+struct ElementType<T (&)[N]> {
>+  using type = T;
>+};
>+
>+template <typename C>
>+using ElementT = typename ElementType<C>::type;
>+
>+template <typename T>
>+using EnableIfMutable =
>+    typename std::enable_if<!std::is_const<T>::value, int>::type;
>+
>+template <typename T>
>+bool EqualImpl(Span<T> a, Span<T> b) {
>+  static_assert(std::is_const<T>::value, "");
>+  return absl::equal(a.begin(), a.end(), b.begin(), b.end());
>+}
>+
>+template <typename T>
>+bool LessThanImpl(Span<T> a, Span<T> b) {
>+  static_assert(std::is_const<T>::value, "");
>+  return std::lexicographical_compare(a.begin(), a.end(), b.begin(), b.end());
>+}
>+
>+// The `IsConvertible` classes here are needed because of the
>+// `std::is_convertible` bug in libcxx when compiled with GCC. This build
>+// configuration is used by Android NDK toolchain. Reference link:
>+// https://bugs.llvm.org/show_bug.cgi?id=27538.
>+template <typename From, typename To>
>+struct IsConvertibleHelper {
>+ private:
>+  static std::true_type testval(To);
>+  static std::false_type testval(...);
>+
>+ public:
>+  using type = decltype(testval(std::declval<From>()));
>+};
>+
>+template <typename From, typename To>
>+struct IsConvertible : IsConvertibleHelper<From, To>::type {};
>+
>+// TODO(zhangxy): replace `IsConvertible` with `std::is_convertible` once the
>+// older version of libcxx is not supported.
>+template <typename From, typename To>
>+using EnableIfConvertibleToSpanConst =
>+    typename std::enable_if<IsConvertible<From, Span<const To>>::value>::type;
>+}  // namespace span_internal
>+
>+//------------------------------------------------------------------------------
>+// Span
>+//------------------------------------------------------------------------------
>+//
>+// A `Span` is an "array view" type for holding a view of a contiguous data
>+// array; the `Span` object does not and cannot own such data itself. A span
>+// provides an easy way to provide overloads for anything operating on
>+// contiguous sequences without needing to manage pointers and array lengths
>+// manually.
>+
>+// A span is conceptually a pointer (ptr) and a length (size) into an already
>+// existing array of contiguous memory; the array it represents references the
>+// elements "ptr[0] .. ptr[size-1]". Passing a properly-constructed `Span`
>+// instead of raw pointers avoids many issues related to index out of bounds
>+// errors.
>+//
>+// Spans may also be constructed from containers holding contiguous sequences.
>+// Such containers must supply `data()` and `size() const` methods (e.g
>+// `std::vector<T>`, `absl::InlinedVector<T, N>`). All implicit conversions to
>+// `absl::Span` from such containers will create spans of type `const T`;
>+// spans which can mutate their values (of type `T`) must use explicit
>+// constructors.
>+//
>+// A `Span<T>` is somewhat analogous to an `absl::string_view`, but for an array
>+// of elements of type `T`. A user of `Span` must ensure that the data being
>+// pointed to outlives the `Span` itself.
>+//
>+// You can construct a `Span<T>` in several ways:
>+//
>+//   * Explicitly from a reference to a container type
>+//   * Explicitly from a pointer and size
>+//   * Implicitly from a container type (but only for spans of type `const T`)
>+//   * Using the `MakeSpan()` or `MakeConstSpan()` factory functions.
>+//
>+// Examples:
>+//
>+//   // Construct a Span explicitly from a container:
>+//   std::vector<int> v = {1, 2, 3, 4, 5};
>+//   auto span = absl::Span<const int>(v);
>+//
>+//   // Construct a Span explicitly from a C-style array:
>+//   int a[5] =  {1, 2, 3, 4, 5};
>+//   auto span = absl::Span<const int>(a);
>+//
>+//   // Construct a Span implicitly from a container
>+//   void MyRoutine(absl::Span<const int> a) {
>+//     ...
>+//   }
>+//   std::vector v = {1,2,3,4,5};
>+//   MyRoutine(v)                     // convert to Span<const T>
>+//
>+// Note that `Span` objects, in addition to requiring that the memory they
>+// point to remains alive, must also ensure that such memory does not get
>+// reallocated. Therefore, to avoid undefined behavior, containers with
>+// associated span views should not invoke operations that may reallocate memory
>+// (such as resizing) or invalidate iterators into the container.
>+//
>+// One common use for a `Span` is when passing arguments to a routine that can
>+// accept a variety of array types (e.g. a `std::vector`, `absl::InlinedVector`,
>+// a C-style array, etc.). Instead of creating overloads for each case, you
>+// can simply specify a `Span` as the argument to such a routine.
>+//
>+// Example:
>+//
>+//   void MyRoutine(absl::Span<const int> a) {
>+//     ...
>+//   }
>+//
>+//   std::vector v = {1,2,3,4,5};
>+//   MyRoutine(v);
>+//
>+//   absl::InlinedVector<int, 4> my_inline_vector;
>+//   MyRoutine(my_inline_vector);
>+//
>+//   // Explicit constructor from pointer,size
>+//   int* my_array = new int[10];
>+//   MyRoutine(absl::Span<const int>(my_array, 10));
>+template <typename T>
>+class Span {
>+ private:
>+  // Used to determine whether a Span can be constructed from a container of
>+  // type C.
>+  template <typename C>
>+  using EnableIfConvertibleFrom =
>+      typename std::enable_if<span_internal::HasData<T, C>::value &&
>+                              span_internal::HasSize<C>::value>::type;
>+
>+  // Used to SFINAE-enable a function when the slice elements are const.
>+  template <typename U>
>+  using EnableIfConstView =
>+      typename std::enable_if<std::is_const<T>::value, U>::type;
>+
>+  // Used to SFINAE-enable a function when the slice elements are mutable.
>+  template <typename U>
>+  using EnableIfMutableView =
>+      typename std::enable_if<!std::is_const<T>::value, U>::type;
>+
>+ public:
>+  using value_type = absl::remove_cv_t<T>;
>+  using pointer = T*;
>+  using const_pointer = const T*;
>+  using reference = T&;
>+  using const_reference = const T&;
>+  using iterator = pointer;
>+  using const_iterator = const_pointer;
>+  using reverse_iterator = std::reverse_iterator<iterator>;
>+  using const_reverse_iterator = std::reverse_iterator<const_iterator>;
>+  using size_type = size_t;
>+  using difference_type = ptrdiff_t;
>+
>+  static const size_type npos = ~(size_type(0));
>+
>+  constexpr Span() noexcept : Span(nullptr, 0) {}
>+  constexpr Span(pointer array, size_type length) noexcept
>+      : ptr_(array), len_(length) {}
>+
>+  // Implicit conversion constructors
>+  template <size_t N>
>+  constexpr Span(T (&a)[N]) noexcept  // NOLINT(runtime/explicit)
>+      : Span(a, N) {}
>+
>+  // Explicit reference constructor for a mutable `Span<T>` type. Can be
>+  // replaced with MakeSpan() to infer the type parameter.
>+  template <typename V, typename = EnableIfConvertibleFrom<V>,
>+            typename = EnableIfMutableView<V>>
>+  explicit Span(V& v) noexcept  // NOLINT(runtime/references)
>+      : Span(span_internal::GetData(v), v.size()) {}
>+
>+  // Implicit reference constructor for a read-only `Span<const T>` type
>+  template <typename V, typename = EnableIfConvertibleFrom<V>,
>+            typename = EnableIfConstView<V>>
>+  constexpr Span(const V& v) noexcept  // NOLINT(runtime/explicit)
>+      : Span(span_internal::GetData(v), v.size()) {}
>+
>+  // Implicit constructor from an initializer list, making it possible to pass a
>+  // brace-enclosed initializer list to a function expecting a `Span`. Such
>+  // spans constructed from an initializer list must be of type `Span<const T>`.
>+  //
>+  //   void Process(absl::Span<const int> x);
>+  //   Process({1, 2, 3});
>+  //
>+  // Note that as always the array referenced by the span must outlive the span.
>+  // Since an initializer list constructor acts as if it is fed a temporary
>+  // array (cf. C++ standard [dcl.init.list]/5), it's safe to use this
>+  // constructor only when the `std::initializer_list` itself outlives the span.
>+  // In order to meet this requirement it's sufficient to ensure that neither
>+  // the span nor a copy of it is used outside of the expression in which it's
>+  // created:
>+  //
>+  //   // Assume that this function uses the array directly, not retaining any
>+  //   // copy of the span or pointer to any of its elements.
>+  //   void Process(absl::Span<const int> ints);
>+  //
>+  //   // Okay: the std::initializer_list<int> will reference a temporary array
>+  //   // that isn't destroyed until after the call to Process returns.
>+  //   Process({ 17, 19 });
>+  //
>+  //   // Not okay: the storage used by the std::initializer_list<int> is not
>+  //   // allowed to be referenced after the first line.
>+  //   absl::Span<const int> ints = { 17, 19 };
>+  //   Process(ints);
>+  //
>+  //   // Not okay for the same reason as above: even when the elements of the
>+  //   // initializer list expression are not temporaries the underlying array
>+  //   // is, so the initializer list must still outlive the span.
>+  //   const int foo = 17;
>+  //   absl::Span<const int> ints = { foo };
>+  //   Process(ints);
>+  //
>+  template <typename LazyT = T,
>+            typename = EnableIfConstView<LazyT>>
>+  Span(
>+      std::initializer_list<value_type> v) noexcept  // NOLINT(runtime/explicit)
>+      : Span(v.begin(), v.size()) {}
>+
>+  // Accessors
>+
>+  // Span::data()
>+  //
>+  // Returns a pointer to the span's underlying array of data (which is held
>+  // outside the span).
>+  constexpr pointer data() const noexcept { return ptr_; }
>+
>+  // Span::size()
>+  //
>+  // Returns the size of this span.
>+  constexpr size_type size() const noexcept { return len_; }
>+
>+  // Span::length()
>+  //
>+  // Returns the length (size) of this span.
>+  constexpr size_type length() const noexcept { return size(); }
>+
>+  // Span::empty()
>+  //
>+  // Returns a boolean indicating whether or not this span is considered empty.
>+  constexpr bool empty() const noexcept { return size() == 0; }
>+
>+  // Span::operator[]
>+  //
>+  // Returns a reference to the i'th element of this span.
>+  constexpr reference operator[](size_type i) const noexcept {
>+    // MSVC 2015 accepts this as constexpr, but not ptr_[i]
>+    return *(data() + i);
>+  }
>+
>+  // Span::at()
>+  //
>+  // Returns a reference to the i'th element of this span.
>+  constexpr reference at(size_type i) const {
>+    return ABSL_PREDICT_TRUE(i < size())
>+               ? ptr_[i]
>+               : (base_internal::ThrowStdOutOfRange(
>+                      "Span::at failed bounds check"),
>+                  ptr_[i]);
>+  }
>+
>+  // Span::front()
>+  //
>+  // Returns a reference to the first element of this span.
>+  reference front() const noexcept { return ABSL_ASSERT(size() > 0), ptr_[0]; }
>+
>+  // Span::back()
>+  //
>+  // Returns a reference to the last element of this span.
>+  reference back() const noexcept {
>+    return ABSL_ASSERT(size() > 0), ptr_[size() - 1];
>+  }
>+
>+  // Span::begin()
>+  //
>+  // Returns an iterator to the first element of this span.
>+  constexpr iterator begin() const noexcept { return ptr_; }
>+
>+  // Span::cbegin()
>+  //
>+  // Returns a const iterator to the first element of this span.
>+  constexpr const_iterator cbegin() const noexcept { return ptr_; }
>+
>+  // Span::end()
>+  //
>+  // Returns an iterator to the last element of this span.
>+  iterator end() const noexcept { return ptr_ + len_; }
>+
>+  // Span::cend()
>+  //
>+  // Returns a const iterator to the last element of this span.
>+  const_iterator cend() const noexcept { return end(); }
>+
>+  // Span::rbegin()
>+  //
>+  // Returns a reverse iterator starting at the last element of this span.
>+  reverse_iterator rbegin() const noexcept { return reverse_iterator(end()); }
>+
>+  // Span::crbegin()
>+  //
>+  // Returns a reverse const iterator starting at the last element of this span.
>+  const_reverse_iterator crbegin() const noexcept { return rbegin(); }
>+
>+  // Span::rend()
>+  //
>+  // Returns a reverse iterator starting at the first element of this span.
>+  reverse_iterator rend() const noexcept { return reverse_iterator(begin()); }
>+
>+  // Span::crend()
>+  //
>+  // Returns a reverse iterator starting at the first element of this span.
>+  const_reverse_iterator crend() const noexcept { return rend(); }
>+
>+  // Span mutations
>+
>+  // Span::remove_prefix()
>+  //
>+  // Removes the first `n` elements from the span.
>+  void remove_prefix(size_type n) noexcept {
>+    assert(len_ >= n);
>+    ptr_ += n;
>+    len_ -= n;
>+  }
>+
>+  // Span::remove_suffix()
>+  //
>+  // Removes the last `n` elements from the span.
>+  void remove_suffix(size_type n) noexcept {
>+    assert(len_ >= n);
>+    len_ -= n;
>+  }
>+
>+  // Span::subspan()
>+  //
>+  // Returns a `Span` starting at element `pos` and of length `len`. Both `pos`
>+  // and `len` are of type `size_type` and thus non-negative. Parameter `pos`
>+  // must be <= size(). Any `len` value that points past the end of the span
>+  // will be trimmed to at most size() - `pos`. A default `len` value of `npos`
>+  // ensures the returned subspan continues until the end of the span.
>+  //
>+  // Examples:
>+  //
>+  //   std::vector<int> vec = {10, 11, 12, 13};
>+  //   absl::MakeSpan(vec).subspan(1, 2);  // {11, 12}
>+  //   absl::MakeSpan(vec).subspan(2, 8);  // {12, 13}
>+  //   absl::MakeSpan(vec).subspan(1);     // {11, 12, 13}
>+  //   absl::MakeSpan(vec).subspan(4);     // {}
>+  //   absl::MakeSpan(vec).subspan(5);     // throws std::out_of_range
>+  constexpr Span subspan(size_type pos = 0, size_type len = npos) const {
>+    return (pos <= len_)
>+               ? Span(ptr_ + pos, span_internal::Min(len_ - pos, len))
>+               : (base_internal::ThrowStdOutOfRange("pos > size()"), Span());
>+  }
>+
>+ private:
>+  pointer ptr_;
>+  size_type len_;
>+};
>+
>+template <typename T>
>+const typename Span<T>::size_type Span<T>::npos;
>+
>+// Span relationals
>+
>+// Equality is compared element-by-element, while ordering is lexicographical.
>+// We provide three overloads for each operator to cover any combination on the
>+// left or right hand side of mutable Span<T>, read-only Span<const T>, and
>+// convertible-to-read-only Span<T>.
>+// TODO(zhangxy): Due to MSVC overload resolution bug with partial ordering
>+// template functions, 5 overloads per operator is needed as a workaround. We
>+// should update them to 3 overloads per operator using non-deduced context like
>+// string_view, i.e.
>+// - (Span<T>, Span<T>)
>+// - (Span<T>, non_deduced<Span<const T>>)
>+// - (non_deduced<Span<const T>>, Span<T>)
>+
>+// operator==
>+template <typename T>
>+bool operator==(Span<T> a, Span<T> b) {
>+  return span_internal::EqualImpl<const T>(a, b);
>+}
>+template <typename T>
>+bool operator==(Span<const T> a, Span<T> b) {
>+  return span_internal::EqualImpl<const T>(a, b);
>+}
>+template <typename T>
>+bool operator==(Span<T> a, Span<const T> b) {
>+  return span_internal::EqualImpl<const T>(a, b);
>+}
>+template <typename T, typename U,
>+          typename = span_internal::EnableIfConvertibleToSpanConst<U, T>>
>+bool operator==(const U& a, Span<T> b) {
>+  return span_internal::EqualImpl<const T>(a, b);
>+}
>+template <typename T, typename U,
>+          typename = span_internal::EnableIfConvertibleToSpanConst<U, T>>
>+bool operator==(Span<T> a, const U& b) {
>+  return span_internal::EqualImpl<const T>(a, b);
>+}
>+
>+// operator!=
>+template <typename T>
>+bool operator!=(Span<T> a, Span<T> b) {
>+  return !(a == b);
>+}
>+template <typename T>
>+bool operator!=(Span<const T> a, Span<T> b) {
>+  return !(a == b);
>+}
>+template <typename T>
>+bool operator!=(Span<T> a, Span<const T> b) {
>+  return !(a == b);
>+}
>+template <typename T, typename U,
>+          typename = span_internal::EnableIfConvertibleToSpanConst<U, T>>
>+bool operator!=(const U& a, Span<T> b) {
>+  return !(a == b);
>+}
>+template <typename T, typename U,
>+          typename = span_internal::EnableIfConvertibleToSpanConst<U, T>>
>+bool operator!=(Span<T> a, const U& b) {
>+  return !(a == b);
>+}
>+
>+// operator<
>+template <typename T>
>+bool operator<(Span<T> a, Span<T> b) {
>+  return span_internal::LessThanImpl<const T>(a, b);
>+}
>+template <typename T>
>+bool operator<(Span<const T> a, Span<T> b) {
>+  return span_internal::LessThanImpl<const T>(a, b);
>+}
>+template <typename T>
>+bool operator<(Span<T> a, Span<const T> b) {
>+  return span_internal::LessThanImpl<const T>(a, b);
>+}
>+template <typename T, typename U,
>+          typename = span_internal::EnableIfConvertibleToSpanConst<U, T>>
>+bool operator<(const U& a, Span<T> b) {
>+  return span_internal::LessThanImpl<const T>(a, b);
>+}
>+template <typename T, typename U,
>+          typename = span_internal::EnableIfConvertibleToSpanConst<U, T>>
>+bool operator<(Span<T> a, const U& b) {
>+  return span_internal::LessThanImpl<const T>(a, b);
>+}
>+
>+// operator>
>+template <typename T>
>+bool operator>(Span<T> a, Span<T> b) {
>+  return b < a;
>+}
>+template <typename T>
>+bool operator>(Span<const T> a, Span<T> b) {
>+  return b < a;
>+}
>+template <typename T>
>+bool operator>(Span<T> a, Span<const T> b) {
>+  return b < a;
>+}
>+template <typename T, typename U,
>+          typename = span_internal::EnableIfConvertibleToSpanConst<U, T>>
>+bool operator>(const U& a, Span<T> b) {
>+  return b < a;
>+}
>+template <typename T, typename U,
>+          typename = span_internal::EnableIfConvertibleToSpanConst<U, T>>
>+bool operator>(Span<T> a, const U& b) {
>+  return b < a;
>+}
>+
>+// operator<=
>+template <typename T>
>+bool operator<=(Span<T> a, Span<T> b) {
>+  return !(b < a);
>+}
>+template <typename T>
>+bool operator<=(Span<const T> a, Span<T> b) {
>+  return !(b < a);
>+}
>+template <typename T>
>+bool operator<=(Span<T> a, Span<const T> b) {
>+  return !(b < a);
>+}
>+template <typename T, typename U,
>+          typename = span_internal::EnableIfConvertibleToSpanConst<U, T>>
>+bool operator<=(const U& a, Span<T> b) {
>+  return !(b < a);
>+}
>+template <typename T, typename U,
>+          typename = span_internal::EnableIfConvertibleToSpanConst<U, T>>
>+bool operator<=(Span<T> a, const U& b) {
>+  return !(b < a);
>+}
>+
>+// operator>=
>+template <typename T>
>+bool operator>=(Span<T> a, Span<T> b) {
>+  return !(a < b);
>+}
>+template <typename T>
>+bool operator>=(Span<const T> a, Span<T> b) {
>+  return !(a < b);
>+}
>+template <typename T>
>+bool operator>=(Span<T> a, Span<const T> b) {
>+  return !(a < b);
>+}
>+template <typename T, typename U,
>+          typename = span_internal::EnableIfConvertibleToSpanConst<U, T>>
>+bool operator>=(const U& a, Span<T> b) {
>+  return !(a < b);
>+}
>+template <typename T, typename U,
>+          typename = span_internal::EnableIfConvertibleToSpanConst<U, T>>
>+bool operator>=(Span<T> a, const U& b) {
>+  return !(a < b);
>+}
>+
>+// MakeSpan()
>+//
>+// Constructs a mutable `Span<T>`, deducing `T` automatically from either a
>+// container or pointer+size.
>+//
>+// Because a read-only `Span<const T>` is implicitly constructed from container
>+// types regardless of whether the container itself is a const container,
>+// constructing mutable spans of type `Span<T>` from containers requires
>+// explicit constructors. The container-accepting version of `MakeSpan()`
>+// deduces the type of `T` by the constness of the pointer received from the
>+// container's `data()` member. Similarly, the pointer-accepting version returns
>+// a `Span<const T>` if `T` is `const`, and a `Span<T>` otherwise.
>+//
>+// Examples:
>+//
>+//   void MyRoutine(absl::Span<MyComplicatedType> a) {
>+//     ...
>+//   };
>+//   // my_vector is a container of non-const types
>+//   std::vector<MyComplicatedType> my_vector;
>+//
>+//   // Constructing a Span implicitly attempts to create a Span of type
>+//   // `Span<const T>`
>+//   MyRoutine(my_vector);                // error, type mismatch
>+//
>+//   // Explicitly constructing the Span is verbose
>+//   MyRoutine(absl::Span<MyComplicatedType>(my_vector));
>+//
>+//   // Use MakeSpan() to make an absl::Span<T>
>+//   MyRoutine(absl::MakeSpan(my_vector));
>+//
>+//   // Construct a span from an array ptr+size
>+//   absl::Span<T> my_span() {
>+//     return absl::MakeSpan(&array[0], num_elements_);
>+//   }
>+//
>+template <int&... ExplicitArgumentBarrier, typename T>
>+constexpr Span<T> MakeSpan(T* ptr, size_t size) noexcept {
>+  return Span<T>(ptr, size);
>+}
>+
>+template <int&... ExplicitArgumentBarrier, typename T>
>+Span<T> MakeSpan(T* begin, T* end) noexcept {
>+  return ABSL_ASSERT(begin <= end), Span<T>(begin, end - begin);
>+}
>+
>+template <int&... ExplicitArgumentBarrier, typename C>
>+constexpr auto MakeSpan(C& c) noexcept  // NOLINT(runtime/references)
>+    -> decltype(absl::MakeSpan(span_internal::GetData(c), c.size())) {
>+  return MakeSpan(span_internal::GetData(c), c.size());
>+}
>+
>+template <int&... ExplicitArgumentBarrier, typename T, size_t N>
>+constexpr Span<T> MakeSpan(T (&array)[N]) noexcept {
>+  return Span<T>(array, N);
>+}
>+
>+// MakeConstSpan()
>+//
>+// Constructs a `Span<const T>` as with `MakeSpan`, deducing `T` automatically,
>+// but always returning a `Span<const T>`.
>+//
>+// Examples:
>+//
>+//   void ProcessInts(absl::Span<const int> some_ints);
>+//
>+//   // Call with a pointer and size.
>+//   int array[3] = { 0, 0, 0 };
>+//   ProcessInts(absl::MakeConstSpan(&array[0], 3));
>+//
>+//   // Call with a [begin, end) pair.
>+//   ProcessInts(absl::MakeConstSpan(&array[0], &array[3]));
>+//
>+//   // Call directly with an array.
>+//   ProcessInts(absl::MakeConstSpan(array));
>+//
>+//   // Call with a contiguous container.
>+//   std::vector<int> some_ints = ...;
>+//   ProcessInts(absl::MakeConstSpan(some_ints));
>+//   ProcessInts(absl::MakeConstSpan(std::vector<int>{ 0, 0, 0 }));
>+//
>+template <int&... ExplicitArgumentBarrier, typename T>
>+constexpr Span<const T> MakeConstSpan(T* ptr, size_t size) noexcept {
>+  return Span<const T>(ptr, size);
>+}
>+
>+template <int&... ExplicitArgumentBarrier, typename T>
>+Span<const T> MakeConstSpan(T* begin, T* end) noexcept {
>+  return ABSL_ASSERT(begin <= end), Span<const T>(begin, end - begin);
>+}
>+
>+template <int&... ExplicitArgumentBarrier, typename C>
>+constexpr auto MakeConstSpan(const C& c) noexcept -> decltype(MakeSpan(c)) {
>+  return MakeSpan(c);
>+}
>+
>+template <int&... ExplicitArgumentBarrier, typename T, size_t N>
>+constexpr Span<const T> MakeConstSpan(const T (&array)[N]) noexcept {
>+  return Span<const T>(array, N);
>+}
>+}  // namespace absl
>+#endif  // ABSL_TYPES_SPAN_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/span_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/span_test.cc
>new file mode 100644
>index 00000000000..fbce7e87479
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/span_test.cc
>@@ -0,0 +1,781 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/types/span.h"
>+
>+#include <array>
>+#include <initializer_list>
>+#include <numeric>
>+#include <stdexcept>
>+#include <string>
>+#include <type_traits>
>+#include <vector>
>+
>+#include "gmock/gmock.h"
>+#include "gtest/gtest.h"
>+#include "absl/base/attributes.h"
>+#include "absl/base/config.h"
>+#include "absl/base/internal/exception_testing.h"
>+#include "absl/container/fixed_array.h"
>+#include "absl/container/inlined_vector.h"
>+#include "absl/strings/str_cat.h"
>+
>+namespace {
>+
>+MATCHER_P(DataIs, data,
>+          absl::StrCat("data() is ", negation ? "is " : "isn't ",
>+                       testing::PrintToString(data))) {
>+  return arg.data() == data;
>+}
>+
>+template <typename T>
>+auto SpanIs(T data, size_t size)
>+    -> decltype(testing::AllOf(DataIs(data), testing::SizeIs(size))) {
>+  return testing::AllOf(DataIs(data), testing::SizeIs(size));
>+}
>+
>+template <typename Container>
>+auto SpanIs(const Container& c) -> decltype(SpanIs(c.data(), c.size())) {
>+  return SpanIs(c.data(), c.size());
>+}
>+
>+std::vector<int> MakeRamp(int len, int offset = 0) {
>+  std::vector<int> v(len);
>+  std::iota(v.begin(), v.end(), offset);
>+  return v;
>+}
>+
>+TEST(IntSpan, EmptyCtors) {
>+  absl::Span<int> s;
>+  EXPECT_THAT(s, SpanIs(nullptr, 0));
>+}
>+
>+TEST(IntSpan, PtrLenCtor) {
>+  int a[] = {1, 2, 3};
>+  absl::Span<int> s(&a[0], 2);
>+  EXPECT_THAT(s, SpanIs(a, 2));
>+}
>+
>+TEST(IntSpan, ArrayCtor) {
>+  int a[] = {1, 2, 3};
>+  absl::Span<int> s(a);
>+  EXPECT_THAT(s, SpanIs(a, 3));
>+
>+  EXPECT_TRUE((std::is_constructible<absl::Span<const int>, int[3]>::value));
>+  EXPECT_TRUE(
>+      (std::is_constructible<absl::Span<const int>, const int[3]>::value));
>+  EXPECT_FALSE((std::is_constructible<absl::Span<int>, const int[3]>::value));
>+  EXPECT_TRUE((std::is_convertible<int[3], absl::Span<const int>>::value));
>+  EXPECT_TRUE(
>+      (std::is_convertible<const int[3], absl::Span<const int>>::value));
>+}
>+
>+template <typename T>
>+void TakesGenericSpan(absl::Span<T>) {}
>+
>+TEST(IntSpan, ContainerCtor) {
>+  std::vector<int> empty;
>+  absl::Span<int> s_empty(empty);
>+  EXPECT_THAT(s_empty, SpanIs(empty));
>+
>+  std::vector<int> filled{1, 2, 3};
>+  absl::Span<int> s_filled(filled);
>+  EXPECT_THAT(s_filled, SpanIs(filled));
>+
>+  absl::Span<int> s_from_span(filled);
>+  EXPECT_THAT(s_from_span, SpanIs(s_filled));
>+
>+  absl::Span<const int> const_filled = filled;
>+  EXPECT_THAT(const_filled, SpanIs(filled));
>+
>+  absl::Span<const int> const_from_span = s_filled;
>+  EXPECT_THAT(const_from_span, SpanIs(s_filled));
>+
>+  EXPECT_TRUE(
>+      (std::is_convertible<std::vector<int>&, absl::Span<const int>>::value));
>+  EXPECT_TRUE(
>+      (std::is_convertible<absl::Span<int>&, absl::Span<const int>>::value));
>+
>+  TakesGenericSpan(absl::Span<int>(filled));
>+}
>+
>+// A struct supplying shallow data() const.
>+struct ContainerWithShallowConstData {
>+  std::vector<int> storage;
>+  int* data() const { return const_cast<int*>(storage.data()); }
>+  int size() const { return storage.size(); }
>+};
>+
>+TEST(IntSpan, ShallowConstness) {
>+  const ContainerWithShallowConstData c{MakeRamp(20)};
>+  absl::Span<int> s(
>+      c);  // We should be able to do this even though data() is const.
>+  s[0] = -1;
>+  EXPECT_EQ(c.storage[0], -1);
>+}
>+
>+TEST(CharSpan, StringCtor) {
>+  std::string empty = "";
>+  absl::Span<char> s_empty(empty);
>+  EXPECT_THAT(s_empty, SpanIs(empty));
>+
>+  std::string abc = "abc";
>+  absl::Span<char> s_abc(abc);
>+  EXPECT_THAT(s_abc, SpanIs(abc));
>+
>+  absl::Span<const char> s_const_abc = abc;
>+  EXPECT_THAT(s_const_abc, SpanIs(abc));
>+
>+  EXPECT_FALSE((std::is_constructible<absl::Span<int>, std::string>::value));
>+  EXPECT_FALSE((std::is_constructible<absl::Span<const int>, std::string>::value));
>+  EXPECT_TRUE((std::is_convertible<std::string, absl::Span<const char>>::value));
>+}
>+
>+TEST(IntSpan, FromConstPointer) {
>+  EXPECT_TRUE((std::is_constructible<absl::Span<const int* const>,
>+                                     std::vector<int*>>::value));
>+  EXPECT_TRUE((std::is_constructible<absl::Span<const int* const>,
>+                                     std::vector<const int*>>::value));
>+  EXPECT_FALSE((
>+      std::is_constructible<absl::Span<const int*>, std::vector<int*>>::value));
>+  EXPECT_FALSE((
>+      std::is_constructible<absl::Span<int*>, std::vector<const int*>>::value));
>+}
>+
>+struct TypeWithMisleadingData {
>+  int& data() { return i; }
>+  int size() { return 1; }
>+  int i;
>+};
>+
>+struct TypeWithMisleadingSize {
>+  int* data() { return &i; }
>+  const char* size() { return "1"; }
>+  int i;
>+};
>+
>+TEST(IntSpan, EvilTypes) {
>+  EXPECT_FALSE(
>+      (std::is_constructible<absl::Span<int>, TypeWithMisleadingData&>::value));
>+  EXPECT_FALSE(
>+      (std::is_constructible<absl::Span<int>, TypeWithMisleadingSize&>::value));
>+}
>+
>+struct Base {
>+  int* data() { return &i; }
>+  int size() { return 1; }
>+  int i;
>+};
>+struct Derived : Base {};
>+
>+TEST(IntSpan, SpanOfDerived) {
>+  EXPECT_TRUE((std::is_constructible<absl::Span<int>, Base&>::value));
>+  EXPECT_TRUE((std::is_constructible<absl::Span<int>, Derived&>::value));
>+  EXPECT_FALSE(
>+      (std::is_constructible<absl::Span<Base>, std::vector<Derived>>::value));
>+}
>+
>+void TestInitializerList(absl::Span<const int> s, const std::vector<int>& v) {
>+  EXPECT_TRUE(absl::equal(s.begin(), s.end(), v.begin(), v.end()));
>+}
>+
>+TEST(ConstIntSpan, InitializerListConversion) {
>+  TestInitializerList({}, {});
>+  TestInitializerList({1}, {1});
>+  TestInitializerList({1, 2, 3}, {1, 2, 3});
>+
>+  EXPECT_FALSE((std::is_constructible<absl::Span<int>,
>+                                      std::initializer_list<int>>::value));
>+  EXPECT_FALSE((
>+      std::is_convertible<absl::Span<int>, std::initializer_list<int>>::value));
>+}
>+
>+TEST(IntSpan, Data) {
>+  int i;
>+  absl::Span<int> s(&i, 1);
>+  EXPECT_EQ(&i, s.data());
>+}
>+
>+TEST(IntSpan, SizeLengthEmpty) {
>+  absl::Span<int> empty;
>+  EXPECT_EQ(empty.size(), 0);
>+  EXPECT_TRUE(empty.empty());
>+  EXPECT_EQ(empty.size(), empty.length());
>+
>+  auto v = MakeRamp(10);
>+  absl::Span<int> s(v);
>+  EXPECT_EQ(s.size(), 10);
>+  EXPECT_FALSE(s.empty());
>+  EXPECT_EQ(s.size(), s.length());
>+}
>+
>+TEST(IntSpan, ElementAccess) {
>+  auto v = MakeRamp(10);
>+  absl::Span<int> s(v);
>+  for (int i = 0; i < s.size(); ++i) {
>+    EXPECT_EQ(s[i], s.at(i));
>+  }
>+
>+  EXPECT_EQ(s.front(), s[0]);
>+  EXPECT_EQ(s.back(), s[9]);
>+}
>+
>+TEST(IntSpan, AtThrows) {
>+  auto v = MakeRamp(10);
>+  absl::Span<int> s(v);
>+
>+  EXPECT_EQ(s.at(9), 9);
>+  ABSL_BASE_INTERNAL_EXPECT_FAIL(s.at(10), std::out_of_range,
>+                                 "failed bounds check");
>+}
>+
>+TEST(IntSpan, RemovePrefixAndSuffix) {
>+  auto v = MakeRamp(20, 1);
>+  absl::Span<int> s(v);
>+  EXPECT_EQ(s.size(), 20);
>+
>+  s.remove_suffix(0);
>+  s.remove_prefix(0);
>+  EXPECT_EQ(s.size(), 20);
>+
>+  s.remove_prefix(1);
>+  EXPECT_EQ(s.size(), 19);
>+  EXPECT_EQ(s[0], 2);
>+
>+  s.remove_suffix(1);
>+  EXPECT_EQ(s.size(), 18);
>+  EXPECT_EQ(s.back(), 19);
>+
>+  s.remove_prefix(7);
>+  EXPECT_EQ(s.size(), 11);
>+  EXPECT_EQ(s[0], 9);
>+
>+  s.remove_suffix(11);
>+  EXPECT_EQ(s.size(), 0);
>+
>+  EXPECT_EQ(v, MakeRamp(20, 1));
>+}
>+
>+TEST(IntSpan, Subspan) {
>+  std::vector<int> empty;
>+  EXPECT_EQ(absl::MakeSpan(empty).subspan(), empty);
>+  EXPECT_THAT(absl::MakeSpan(empty).subspan(0, 0), SpanIs(empty));
>+  EXPECT_THAT(absl::MakeSpan(empty).subspan(0, absl::Span<const int>::npos),
>+              SpanIs(empty));
>+
>+  auto ramp = MakeRamp(10);
>+  EXPECT_THAT(absl::MakeSpan(ramp).subspan(), SpanIs(ramp));
>+  EXPECT_THAT(absl::MakeSpan(ramp).subspan(0, 10), SpanIs(ramp));
>+  EXPECT_THAT(absl::MakeSpan(ramp).subspan(0, absl::Span<const int>::npos),
>+              SpanIs(ramp));
>+  EXPECT_THAT(absl::MakeSpan(ramp).subspan(0, 3), SpanIs(ramp.data(), 3));
>+  EXPECT_THAT(absl::MakeSpan(ramp).subspan(5, absl::Span<const int>::npos),
>+              SpanIs(ramp.data() + 5, 5));
>+  EXPECT_THAT(absl::MakeSpan(ramp).subspan(3, 3), SpanIs(ramp.data() + 3, 3));
>+  EXPECT_THAT(absl::MakeSpan(ramp).subspan(10, 5), SpanIs(ramp.data() + 10, 0));
>+
>+#ifdef ABSL_HAVE_EXCEPTIONS
>+  EXPECT_THROW(absl::MakeSpan(ramp).subspan(11, 5), std::out_of_range);
>+#else
>+  EXPECT_DEATH_IF_SUPPORTED(absl::MakeSpan(ramp).subspan(11, 5), "");
>+#endif
>+}
>+
>+TEST(IntSpan, MakeSpanPtrLength) {
>+  std::vector<int> empty;
>+  auto s_empty = absl::MakeSpan(empty.data(), empty.size());
>+  EXPECT_THAT(s_empty, SpanIs(empty));
>+
>+  std::array<int, 3> a{{1, 2, 3}};
>+  auto s = absl::MakeSpan(a.data(), a.size());
>+  EXPECT_THAT(s, SpanIs(a));
>+
>+  EXPECT_THAT(absl::MakeConstSpan(empty.data(), empty.size()), SpanIs(s_empty));
>+  EXPECT_THAT(absl::MakeConstSpan(a.data(), a.size()), SpanIs(s));
>+}
>+
>+TEST(IntSpan, MakeSpanTwoPtrs) {
>+  std::vector<int> empty;
>+  auto s_empty = absl::MakeSpan(empty.data(), empty.data());
>+  EXPECT_THAT(s_empty, SpanIs(empty));
>+
>+  std::vector<int> v{1, 2, 3};
>+  auto s = absl::MakeSpan(v.data(), v.data() + 1);
>+  EXPECT_THAT(s, SpanIs(v.data(), 1));
>+
>+  EXPECT_THAT(absl::MakeConstSpan(empty.data(), empty.data()), SpanIs(s_empty));
>+  EXPECT_THAT(absl::MakeConstSpan(v.data(), v.data() + 1), SpanIs(s));
>+}
>+
>+TEST(IntSpan, MakeSpanContainer) {
>+  std::vector<int> empty;
>+  auto s_empty = absl::MakeSpan(empty);
>+  EXPECT_THAT(s_empty, SpanIs(empty));
>+
>+  std::vector<int> v{1, 2, 3};
>+  auto s = absl::MakeSpan(v);
>+  EXPECT_THAT(s, SpanIs(v));
>+
>+  EXPECT_THAT(absl::MakeConstSpan(empty), SpanIs(s_empty));
>+  EXPECT_THAT(absl::MakeConstSpan(v), SpanIs(s));
>+
>+  EXPECT_THAT(absl::MakeSpan(s), SpanIs(s));
>+  EXPECT_THAT(absl::MakeConstSpan(s), SpanIs(s));
>+}
>+
>+TEST(CharSpan, MakeSpanString) {
>+  std::string empty = "";
>+  auto s_empty = absl::MakeSpan(empty);
>+  EXPECT_THAT(s_empty, SpanIs(empty));
>+
>+  std::string str = "abc";
>+  auto s_str = absl::MakeSpan(str);
>+  EXPECT_THAT(s_str, SpanIs(str));
>+
>+  EXPECT_THAT(absl::MakeConstSpan(empty), SpanIs(s_empty));
>+  EXPECT_THAT(absl::MakeConstSpan(str), SpanIs(s_str));
>+}
>+
>+TEST(IntSpan, MakeSpanArray) {
>+  int a[] = {1, 2, 3};
>+  auto s = absl::MakeSpan(a);
>+  EXPECT_THAT(s, SpanIs(a, 3));
>+
>+  const int ca[] = {1, 2, 3};
>+  auto s_ca = absl::MakeSpan(ca);
>+  EXPECT_THAT(s_ca, SpanIs(ca, 3));
>+
>+  EXPECT_THAT(absl::MakeConstSpan(a), SpanIs(s));
>+  EXPECT_THAT(absl::MakeConstSpan(ca), SpanIs(s_ca));
>+}
>+
>+// Compile-asserts that the argument has the expected decayed type.
>+template <typename Expected, typename T>
>+void CheckType(const T& /* value */) {
>+  testing::StaticAssertTypeEq<Expected, T>();
>+}
>+
>+TEST(IntSpan, MakeSpanTypes) {
>+  std::vector<int> vec;
>+  const std::vector<int> cvec;
>+  int a[1];
>+  const int ca[] = {1};
>+  int* ip = a;
>+  const int* cip = ca;
>+  std::string s = "";
>+  const std::string cs = "";
>+  CheckType<absl::Span<int>>(absl::MakeSpan(vec));
>+  CheckType<absl::Span<const int>>(absl::MakeSpan(cvec));
>+  CheckType<absl::Span<int>>(absl::MakeSpan(ip, ip + 1));
>+  CheckType<absl::Span<int>>(absl::MakeSpan(ip, 1));
>+  CheckType<absl::Span<const int>>(absl::MakeSpan(cip, cip + 1));
>+  CheckType<absl::Span<const int>>(absl::MakeSpan(cip, 1));
>+  CheckType<absl::Span<int>>(absl::MakeSpan(a));
>+  CheckType<absl::Span<int>>(absl::MakeSpan(a, a + 1));
>+  CheckType<absl::Span<int>>(absl::MakeSpan(a, 1));
>+  CheckType<absl::Span<const int>>(absl::MakeSpan(ca));
>+  CheckType<absl::Span<const int>>(absl::MakeSpan(ca, ca + 1));
>+  CheckType<absl::Span<const int>>(absl::MakeSpan(ca, 1));
>+  CheckType<absl::Span<char>>(absl::MakeSpan(s));
>+  CheckType<absl::Span<const char>>(absl::MakeSpan(cs));
>+}
>+
>+TEST(ConstIntSpan, MakeConstSpanTypes) {
>+  std::vector<int> vec;
>+  const std::vector<int> cvec;
>+  int array[1];
>+  const int carray[] = {0};
>+  int* ptr = array;
>+  const int* cptr = carray;
>+  std::string s = "";
>+  std::string cs = "";
>+  CheckType<absl::Span<const int>>(absl::MakeConstSpan(vec));
>+  CheckType<absl::Span<const int>>(absl::MakeConstSpan(cvec));
>+  CheckType<absl::Span<const int>>(absl::MakeConstSpan(ptr, ptr + 1));
>+  CheckType<absl::Span<const int>>(absl::MakeConstSpan(ptr, 1));
>+  CheckType<absl::Span<const int>>(absl::MakeConstSpan(cptr, cptr + 1));
>+  CheckType<absl::Span<const int>>(absl::MakeConstSpan(cptr, 1));
>+  CheckType<absl::Span<const int>>(absl::MakeConstSpan(array));
>+  CheckType<absl::Span<const int>>(absl::MakeConstSpan(carray));
>+  CheckType<absl::Span<const char>>(absl::MakeConstSpan(s));
>+  CheckType<absl::Span<const char>>(absl::MakeConstSpan(cs));
>+}
>+
>+TEST(IntSpan, Equality) {
>+  const int arr1[] = {1, 2, 3, 4, 5};
>+  int arr2[] = {1, 2, 3, 4, 5};
>+  std::vector<int> vec1(std::begin(arr1), std::end(arr1));
>+  std::vector<int> vec2 = vec1;
>+  std::vector<int> other_vec = {2, 4, 6, 8, 10};
>+  // These two slices are from different vectors, but have the same size and
>+  // have the same elements (right now).  They should compare equal. Test both
>+  // == and !=.
>+  const absl::Span<const int> from1 = vec1;
>+  const absl::Span<const int> from2 = vec2;
>+  EXPECT_EQ(from1, from1);
>+  EXPECT_FALSE(from1 != from1);
>+  EXPECT_EQ(from1, from2);
>+  EXPECT_FALSE(from1 != from2);
>+
>+  // These two slices have different underlying vector values. They should be
>+  // considered not equal. Test both == and !=.
>+  const absl::Span<const int> from_other = other_vec;
>+  EXPECT_NE(from1, from_other);
>+  EXPECT_FALSE(from1 == from_other);
>+
>+  // Comparison between a vector and its slice should be equal. And vice-versa.
>+  // This ensures implicit conversion to Span works on both sides of ==.
>+  EXPECT_EQ(vec1, from1);
>+  EXPECT_FALSE(vec1 != from1);
>+  EXPECT_EQ(from1, vec1);
>+  EXPECT_FALSE(from1 != vec1);
>+
>+  // This verifies that absl::Span<T> can be compared freely with
>+  // absl::Span<const T>.
>+  const absl::Span<int> mutable_from1(vec1);
>+  const absl::Span<int> mutable_from2(vec2);
>+  EXPECT_EQ(from1, mutable_from1);
>+  EXPECT_EQ(mutable_from1, from1);
>+  EXPECT_EQ(mutable_from1, mutable_from2);
>+  EXPECT_EQ(mutable_from2, mutable_from1);
>+
>+  // Comparison between a vector and its slice should be equal for mutable
>+  // Spans as well.
>+  EXPECT_EQ(vec1, mutable_from1);
>+  EXPECT_FALSE(vec1 != mutable_from1);
>+  EXPECT_EQ(mutable_from1, vec1);
>+  EXPECT_FALSE(mutable_from1 != vec1);
>+
>+  // Comparison between convertible-to-Span-of-const and Span-of-mutable. Arrays
>+  // are used because they're the only value type which converts to a
>+  // Span-of-mutable. EXPECT_TRUE is used instead of EXPECT_EQ to avoid
>+  // array-to-pointer decay.
>+  EXPECT_TRUE(arr1 == mutable_from1);
>+  EXPECT_FALSE(arr1 != mutable_from1);
>+  EXPECT_TRUE(mutable_from1 == arr1);
>+  EXPECT_FALSE(mutable_from1 != arr1);
>+
>+  // Comparison between convertible-to-Span-of-mutable and Span-of-const
>+  EXPECT_TRUE(arr2 == from1);
>+  EXPECT_FALSE(arr2 != from1);
>+  EXPECT_TRUE(from1 == arr2);
>+  EXPECT_FALSE(from1 != arr2);
>+
>+  // With a different size, the array slices should not be equal.
>+  EXPECT_NE(from1, absl::Span<const int>(from1).subspan(0, from1.size() - 1));
>+
>+  // With different contents, the array slices should not be equal.
>+  ++vec2.back();
>+  EXPECT_NE(from1, from2);
>+}
>+
>+class IntSpanOrderComparisonTest : public testing::Test {
>+ public:
>+  IntSpanOrderComparisonTest()
>+      : arr_before_{1, 2, 3},
>+        arr_after_{1, 2, 4},
>+        carr_after_{1, 2, 4},
>+        vec_before_(std::begin(arr_before_), std::end(arr_before_)),
>+        vec_after_(std::begin(arr_after_), std::end(arr_after_)),
>+        before_(vec_before_),
>+        after_(vec_after_),
>+        cbefore_(vec_before_),
>+        cafter_(vec_after_) {}
>+
>+ protected:
>+  int arr_before_[3], arr_after_[3];
>+  const int carr_after_[3];
>+  std::vector<int> vec_before_, vec_after_;
>+  absl::Span<int> before_, after_;
>+  absl::Span<const int> cbefore_, cafter_;
>+};
>+
>+TEST_F(IntSpanOrderComparisonTest, CompareSpans) {
>+  EXPECT_TRUE(cbefore_ < cafter_);
>+  EXPECT_TRUE(cbefore_ <= cafter_);
>+  EXPECT_TRUE(cafter_ > cbefore_);
>+  EXPECT_TRUE(cafter_ >= cbefore_);
>+
>+  EXPECT_FALSE(cbefore_ > cafter_);
>+  EXPECT_FALSE(cafter_ < cbefore_);
>+
>+  EXPECT_TRUE(before_ < after_);
>+  EXPECT_TRUE(before_ <= after_);
>+  EXPECT_TRUE(after_ > before_);
>+  EXPECT_TRUE(after_ >= before_);
>+
>+  EXPECT_FALSE(before_ > after_);
>+  EXPECT_FALSE(after_ < before_);
>+
>+  EXPECT_TRUE(cbefore_ < after_);
>+  EXPECT_TRUE(cbefore_ <= after_);
>+  EXPECT_TRUE(after_ > cbefore_);
>+  EXPECT_TRUE(after_ >= cbefore_);
>+
>+  EXPECT_FALSE(cbefore_ > after_);
>+  EXPECT_FALSE(after_ < cbefore_);
>+}
>+
>+TEST_F(IntSpanOrderComparisonTest, SpanOfConstAndContainer) {
>+  EXPECT_TRUE(cbefore_ < vec_after_);
>+  EXPECT_TRUE(cbefore_ <= vec_after_);
>+  EXPECT_TRUE(vec_after_ > cbefore_);
>+  EXPECT_TRUE(vec_after_ >= cbefore_);
>+
>+  EXPECT_FALSE(cbefore_ > vec_after_);
>+  EXPECT_FALSE(vec_after_ < cbefore_);
>+
>+  EXPECT_TRUE(arr_before_ < cafter_);
>+  EXPECT_TRUE(arr_before_ <= cafter_);
>+  EXPECT_TRUE(cafter_ > arr_before_);
>+  EXPECT_TRUE(cafter_ >= arr_before_);
>+
>+  EXPECT_FALSE(arr_before_ > cafter_);
>+  EXPECT_FALSE(cafter_ < arr_before_);
>+}
>+
>+TEST_F(IntSpanOrderComparisonTest, SpanOfMutableAndContainer) {
>+  EXPECT_TRUE(vec_before_ < after_);
>+  EXPECT_TRUE(vec_before_ <= after_);
>+  EXPECT_TRUE(after_ > vec_before_);
>+  EXPECT_TRUE(after_ >= vec_before_);
>+
>+  EXPECT_FALSE(vec_before_ > after_);
>+  EXPECT_FALSE(after_ < vec_before_);
>+
>+  EXPECT_TRUE(before_ < carr_after_);
>+  EXPECT_TRUE(before_ <= carr_after_);
>+  EXPECT_TRUE(carr_after_ > before_);
>+  EXPECT_TRUE(carr_after_ >= before_);
>+
>+  EXPECT_FALSE(before_ > carr_after_);
>+  EXPECT_FALSE(carr_after_ < before_);
>+}
>+
>+TEST_F(IntSpanOrderComparisonTest, EqualSpans) {
>+  EXPECT_FALSE(before_ < before_);
>+  EXPECT_TRUE(before_ <= before_);
>+  EXPECT_FALSE(before_ > before_);
>+  EXPECT_TRUE(before_ >= before_);
>+}
>+
>+TEST_F(IntSpanOrderComparisonTest, Subspans) {
>+  auto subspan = before_.subspan(0, 1);
>+  EXPECT_TRUE(subspan < before_);
>+  EXPECT_TRUE(subspan <= before_);
>+  EXPECT_TRUE(before_ > subspan);
>+  EXPECT_TRUE(before_ >= subspan);
>+
>+  EXPECT_FALSE(subspan > before_);
>+  EXPECT_FALSE(before_ < subspan);
>+}
>+
>+TEST_F(IntSpanOrderComparisonTest, EmptySpans) {
>+  absl::Span<int> empty;
>+  EXPECT_FALSE(empty < empty);
>+  EXPECT_TRUE(empty <= empty);
>+  EXPECT_FALSE(empty > empty);
>+  EXPECT_TRUE(empty >= empty);
>+
>+  EXPECT_TRUE(empty < before_);
>+  EXPECT_TRUE(empty <= before_);
>+  EXPECT_TRUE(before_ > empty);
>+  EXPECT_TRUE(before_ >= empty);
>+
>+  EXPECT_FALSE(empty > before_);
>+  EXPECT_FALSE(before_ < empty);
>+}
>+
>+TEST(IntSpan, ExposesContainerTypesAndConsts) {
>+  absl::Span<int> slice;
>+  CheckType<absl::Span<int>::iterator>(slice.begin());
>+  EXPECT_TRUE((std::is_convertible<decltype(slice.begin()),
>+                                   absl::Span<int>::const_iterator>::value));
>+  CheckType<absl::Span<int>::const_iterator>(slice.cbegin());
>+  EXPECT_TRUE((std::is_convertible<decltype(slice.end()),
>+                                   absl::Span<int>::const_iterator>::value));
>+  CheckType<absl::Span<int>::const_iterator>(slice.cend());
>+  CheckType<absl::Span<int>::reverse_iterator>(slice.rend());
>+  EXPECT_TRUE(
>+      (std::is_convertible<decltype(slice.rend()),
>+                           absl::Span<int>::const_reverse_iterator>::value));
>+  CheckType<absl::Span<int>::const_reverse_iterator>(slice.crend());
>+  testing::StaticAssertTypeEq<int, absl::Span<int>::value_type>();
>+  testing::StaticAssertTypeEq<int, absl::Span<const int>::value_type>();
>+  testing::StaticAssertTypeEq<int*, absl::Span<int>::pointer>();
>+  testing::StaticAssertTypeEq<const int*, absl::Span<const int>::pointer>();
>+  testing::StaticAssertTypeEq<int&, absl::Span<int>::reference>();
>+  testing::StaticAssertTypeEq<const int&, absl::Span<const int>::reference>();
>+  testing::StaticAssertTypeEq<const int&, absl::Span<int>::const_reference>();
>+  testing::StaticAssertTypeEq<const int&,
>+                              absl::Span<const int>::const_reference>();
>+  EXPECT_EQ(static_cast<absl::Span<int>::size_type>(-1), absl::Span<int>::npos);
>+}
>+
>+TEST(IntSpan, IteratorsAndReferences) {
>+  auto accept_pointer = [](int*) {};
>+  auto accept_reference = [](int&) {};
>+  auto accept_iterator = [](absl::Span<int>::iterator) {};
>+  auto accept_const_iterator = [](absl::Span<int>::const_iterator) {};
>+  auto accept_reverse_iterator = [](absl::Span<int>::reverse_iterator) {};
>+  auto accept_const_reverse_iterator =
>+      [](absl::Span<int>::const_reverse_iterator) {};
>+
>+  int a[1];
>+  absl::Span<int> s = a;
>+
>+  accept_pointer(s.data());
>+  accept_iterator(s.begin());
>+  accept_const_iterator(s.begin());
>+  accept_const_iterator(s.cbegin());
>+  accept_iterator(s.end());
>+  accept_const_iterator(s.end());
>+  accept_const_iterator(s.cend());
>+  accept_reverse_iterator(s.rbegin());
>+  accept_const_reverse_iterator(s.rbegin());
>+  accept_const_reverse_iterator(s.crbegin());
>+  accept_reverse_iterator(s.rend());
>+  accept_const_reverse_iterator(s.rend());
>+  accept_const_reverse_iterator(s.crend());
>+
>+  accept_reference(s[0]);
>+  accept_reference(s.at(0));
>+  accept_reference(s.front());
>+  accept_reference(s.back());
>+}
>+
>+TEST(IntSpan, IteratorsAndReferences_Const) {
>+  auto accept_pointer = [](int*) {};
>+  auto accept_reference = [](int&) {};
>+  auto accept_iterator = [](absl::Span<int>::iterator) {};
>+  auto accept_const_iterator = [](absl::Span<int>::const_iterator) {};
>+  auto accept_reverse_iterator = [](absl::Span<int>::reverse_iterator) {};
>+  auto accept_const_reverse_iterator =
>+      [](absl::Span<int>::const_reverse_iterator) {};
>+
>+  int a[1];
>+  const absl::Span<int> s = a;
>+
>+  accept_pointer(s.data());
>+  accept_iterator(s.begin());
>+  accept_const_iterator(s.begin());
>+  accept_const_iterator(s.cbegin());
>+  accept_iterator(s.end());
>+  accept_const_iterator(s.end());
>+  accept_const_iterator(s.cend());
>+  accept_reverse_iterator(s.rbegin());
>+  accept_const_reverse_iterator(s.rbegin());
>+  accept_const_reverse_iterator(s.crbegin());
>+  accept_reverse_iterator(s.rend());
>+  accept_const_reverse_iterator(s.rend());
>+  accept_const_reverse_iterator(s.crend());
>+
>+  accept_reference(s[0]);
>+  accept_reference(s.at(0));
>+  accept_reference(s.front());
>+  accept_reference(s.back());
>+}
>+
>+TEST(IntSpan, NoexceptTest) {
>+  int a[] = {1, 2, 3};
>+  std::vector<int> v;
>+  EXPECT_TRUE(noexcept(absl::Span<const int>()));
>+  EXPECT_TRUE(noexcept(absl::Span<const int>(a, 2)));
>+  EXPECT_TRUE(noexcept(absl::Span<const int>(a)));
>+  EXPECT_TRUE(noexcept(absl::Span<const int>(v)));
>+  EXPECT_TRUE(noexcept(absl::Span<int>(v)));
>+  EXPECT_TRUE(noexcept(absl::Span<const int>({1, 2, 3})));
>+  EXPECT_TRUE(noexcept(absl::MakeSpan(v)));
>+  EXPECT_TRUE(noexcept(absl::MakeSpan(a)));
>+  EXPECT_TRUE(noexcept(absl::MakeSpan(a, 2)));
>+  EXPECT_TRUE(noexcept(absl::MakeSpan(a, a + 1)));
>+  EXPECT_TRUE(noexcept(absl::MakeConstSpan(v)));
>+  EXPECT_TRUE(noexcept(absl::MakeConstSpan(a)));
>+  EXPECT_TRUE(noexcept(absl::MakeConstSpan(a, 2)));
>+  EXPECT_TRUE(noexcept(absl::MakeConstSpan(a, a + 1)));
>+
>+  absl::Span<int> s(v);
>+  EXPECT_TRUE(noexcept(s.data()));
>+  EXPECT_TRUE(noexcept(s.size()));
>+  EXPECT_TRUE(noexcept(s.length()));
>+  EXPECT_TRUE(noexcept(s.empty()));
>+  EXPECT_TRUE(noexcept(s[0]));
>+  EXPECT_TRUE(noexcept(s.front()));
>+  EXPECT_TRUE(noexcept(s.back()));
>+  EXPECT_TRUE(noexcept(s.begin()));
>+  EXPECT_TRUE(noexcept(s.cbegin()));
>+  EXPECT_TRUE(noexcept(s.end()));
>+  EXPECT_TRUE(noexcept(s.cend()));
>+  EXPECT_TRUE(noexcept(s.rbegin()));
>+  EXPECT_TRUE(noexcept(s.crbegin()));
>+  EXPECT_TRUE(noexcept(s.rend()));
>+  EXPECT_TRUE(noexcept(s.crend()));
>+  EXPECT_TRUE(noexcept(s.remove_prefix(0)));
>+  EXPECT_TRUE(noexcept(s.remove_suffix(0)));
>+}
>+
>+// ConstexprTester exercises expressions in a constexpr context. Simply placing
>+// the expression in a constexpr function is not enough, as some compilers will
>+// simply compile the constexpr function as runtime code. Using template
>+// parameters forces compile-time execution.
>+template <int i>
>+struct ConstexprTester {};
>+
>+#define ABSL_TEST_CONSTEXPR(expr)                       \
>+  do {                                                  \
>+    ABSL_ATTRIBUTE_UNUSED ConstexprTester<(expr, 1)> t; \
>+  } while (0)
>+
>+struct ContainerWithConstexprMethods {
>+  constexpr int size() const { return 1; }
>+  constexpr const int* data() const { return &i; }
>+  const int i;
>+};
>+
>+TEST(ConstIntSpan, ConstexprTest) {
>+  static constexpr int a[] = {1, 2, 3};
>+  static constexpr int sized_arr[2] = {1, 2};
>+  static constexpr ContainerWithConstexprMethods c{1};
>+  ABSL_TEST_CONSTEXPR(absl::Span<const int>());
>+  ABSL_TEST_CONSTEXPR(absl::Span<const int>(a, 2));
>+  ABSL_TEST_CONSTEXPR(absl::Span<const int>(sized_arr));
>+  ABSL_TEST_CONSTEXPR(absl::Span<const int>(c));
>+  ABSL_TEST_CONSTEXPR(absl::MakeSpan(&a[0], 1));
>+  ABSL_TEST_CONSTEXPR(absl::MakeSpan(c));
>+  ABSL_TEST_CONSTEXPR(absl::MakeSpan(a));
>+  ABSL_TEST_CONSTEXPR(absl::MakeConstSpan(&a[0], 1));
>+  ABSL_TEST_CONSTEXPR(absl::MakeConstSpan(c));
>+  ABSL_TEST_CONSTEXPR(absl::MakeConstSpan(a));
>+
>+  constexpr absl::Span<const int> span = c;
>+  ABSL_TEST_CONSTEXPR(span.data());
>+  ABSL_TEST_CONSTEXPR(span.size());
>+  ABSL_TEST_CONSTEXPR(span.length());
>+  ABSL_TEST_CONSTEXPR(span.empty());
>+  ABSL_TEST_CONSTEXPR(span.begin());
>+  ABSL_TEST_CONSTEXPR(span.cbegin());
>+  ABSL_TEST_CONSTEXPR(span.subspan(0, 0));
>+  ABSL_TEST_CONSTEXPR(span[0]);
>+}
>+
>+struct BigStruct {
>+  char bytes[10000];
>+};
>+
>+TEST(Span, SpanSize) {
>+  EXPECT_LE(sizeof(absl::Span<int>), 2 * sizeof(void*));
>+  EXPECT_LE(sizeof(absl::Span<BigStruct>), 2 * sizeof(void*));
>+}
>+
>+}  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/variant.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/variant.h
>new file mode 100644
>index 00000000000..17e0634de03
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/variant.h
>@@ -0,0 +1,849 @@
>+// Copyright 2018 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// -----------------------------------------------------------------------------
>+// variant.h
>+// -----------------------------------------------------------------------------
>+//
>+// This header file defines an `absl::variant` type for holding a type-safe
>+// value of some prescribed set of types (noted as alternative types), and
>+// associated functions for managing variants.
>+//
>+// The `absl::variant` type is a form of type-safe union. An `absl::variant`
>+// should always hold a value of one of its alternative types (except in the
>+// "valueless by exception state" -- see below). A default-constructed
>+// `absl::variant` will hold the value of its first alternative type, provided
>+// it is default-constructable.
>+//
>+// In exceptional cases due to error, an `absl::variant` can hold no
>+// value (known as a "valueless by exception" state), though this is not the
>+// norm.
>+//
>+// As with `absl::optional`, an `absl::variant` -- when it holds a value --
>+// allocates a value of that type directly within the `variant` itself; it
>+// cannot hold a reference, array, or the type `void`; it can, however, hold a
>+// pointer to externally managed memory.
>+//
>+// `absl::variant` is a C++11 compatible version of the C++17 `std::variant`
>+// abstraction and is designed to be a drop-in replacement for code compliant
>+// with C++17.
>+
>+#ifndef ABSL_TYPES_VARIANT_H_
>+#define ABSL_TYPES_VARIANT_H_
>+
>+#include "absl/base/config.h"
>+#include "absl/utility/utility.h"
>+
>+#ifdef ABSL_HAVE_STD_VARIANT
>+
>+#include <variant>
>+
>+namespace absl {
>+using std::bad_variant_access;
>+using std::get;
>+using std::get_if;
>+using std::holds_alternative;
>+using std::monostate;
>+using std::variant;
>+using std::variant_alternative;
>+using std::variant_alternative_t;
>+using std::variant_npos;
>+using std::variant_size;
>+using std::variant_size_v;
>+using std::visit;
>+}  // namespace absl
>+
>+#else  // ABSL_HAVE_STD_VARIANT
>+
>+#include <functional>
>+#include <new>
>+#include <type_traits>
>+#include <utility>
>+
>+#include "absl/base/macros.h"
>+#include "absl/base/port.h"
>+#include "absl/meta/type_traits.h"
>+#include "absl/types/internal/variant.h"
>+
>+namespace absl {
>+
>+// -----------------------------------------------------------------------------
>+// absl::variant
>+// -----------------------------------------------------------------------------
>+//
>+// An 'absl::variant` type is a form of type-safe union. An `absl::variant` --
>+// except in exceptional cases -- always holds a value of one of its alternative
>+// types.
>+//
>+// Example:
>+//
>+//   // Construct a variant that holds either an integer or a std::string and
>+//   // assign it to a std::string.
>+//   absl::variant<int, std::string> v = std::string("abc");
>+//
>+//   // A default-contructed variant will hold a value-initialized value of
>+//   // the first alternative type.
>+//   auto a = absl::variant<int, std::string>();   // Holds an int of value '0'.
>+//
>+//   // variants are assignable.
>+//
>+//   // copy assignment
>+//   auto v1 = absl::variant<int, std::string>("abc");
>+//   auto v2 = absl::variant<int, std::string>(10);
>+//   v2 = v1;  // copy assign
>+//
>+//   // move assignment
>+//   auto v1 = absl::variant<int, std::string>("abc");
>+//   v1 = absl::variant<int, std::string>(10);
>+//
>+//   // assignment through type conversion
>+//   a = 128;         // variant contains int
>+//   a = "128";       // variant contains std::string
>+//
>+// An `absl::variant` holding a value of one of its alternative types `T` holds
>+// an allocation of `T` directly within the variant itself. An `absl::variant`
>+// is not allowed to allocate additional storage, such as dynamic memory, to
>+// allocate the contained value. The contained value shall be allocated in a
>+// region of the variant storage suitably aligned for all alternative types.
>+template <typename... Ts>
>+class variant;
>+
>+// swap()
>+//
>+// Swaps two `absl::variant` values. This function is equivalent to `v.swap(w)`
>+// where `v` and `w` are `absl::variant` types.
>+//
>+// Note that this function requires all alternative types to be both swappable
>+// and move-constructible, because any two variants may refer to either the same
>+// type (in which case, they will be swapped) or to two different types (in
>+// which case the values will need to be moved).
>+//
>+template <typename... Ts>
>+void swap(variant<Ts...>& v, variant<Ts...>& w) noexcept(noexcept(v.swap(w))) {
>+  v.swap(w);
>+}
>+
>+// variant_size
>+//
>+// Returns the number of alterative types available for a given `absl::variant`
>+// type as a compile-time constant expression. As this is a class template, it
>+// is not generally useful for accessing the number of alternative types of
>+// any given `absl::variant` instance.
>+//
>+// Example:
>+//
>+//   auto a = absl::variant<int, std::string>;
>+//   constexpr int num_types =
>+//       absl::variant_size<absl::variant<int, std::string>>();
>+//
>+//   // You can also use the member constant `value`.
>+//   constexpr int num_types =
>+//       absl::variant_size<absl::variant<int, std::string>>::value;
>+//
>+//   // `absl::variant_size` is more valuable for use in generic code:
>+//   template <typename Variant>
>+//   constexpr bool IsVariantMultivalue() {
>+//       return absl::variant_size<Variant>() > 1;
>+//   }
>+//
>+// Note that the set of cv-qualified specializations of `variant_size` are
>+// provided to ensure that those specializations compile (especially when passed
>+// within template logic).
>+template <class T>
>+struct variant_size;
>+
>+template <class... Ts>
>+struct variant_size<variant<Ts...>>
>+    : std::integral_constant<std::size_t, sizeof...(Ts)> {};
>+
>+// Specialization of `variant_size` for const qualified variants.
>+template <class T>
>+struct variant_size<const T> : variant_size<T>::type {};
>+
>+// Specialization of `variant_size` for volatile qualified variants.
>+template <class T>
>+struct variant_size<volatile T> : variant_size<T>::type {};
>+
>+// Specialization of `variant_size` for const volatile qualified variants.
>+template <class T>
>+struct variant_size<const volatile T> : variant_size<T>::type {};
>+
>+// variant_alternative
>+//
>+// Returns the alternative type for a given `absl::variant` at the passed
>+// index value as a compile-time constant expression. As this is a class
>+// template resulting in a type, it is not useful for access of the run-time
>+// value of any given `absl::variant` variable.
>+//
>+// Example:
>+//
>+//   // The type of the 0th alternative is "int".
>+//   using alternative_type_0
>+//     = absl::variant_alternative<0, absl::variant<int, std::string>>::type;
>+//
>+//   static_assert(std::is_same<alternative_type_0, int>::value, "");
>+//
>+//   // `absl::variant_alternative` is more valuable for use in generic code:
>+//   template <typename Variant>
>+//   constexpr bool IsFirstElementTrivial() {
>+//       return std::is_trivial_v<variant_alternative<0, Variant>::type>;
>+//   }
>+//
>+// Note that the set of cv-qualified specializations of `variant_alternative`
>+// are provided to ensure that those specializations compile (especially when
>+// passed within template logic).
>+template <std::size_t I, class T>
>+struct variant_alternative;
>+
>+template <std::size_t I, class... Types>
>+struct variant_alternative<I, variant<Types...>> {
>+  using type =
>+      variant_internal::VariantAlternativeSfinaeT<I, variant<Types...>>;
>+};
>+
>+// Specialization of `variant_alternative` for const qualified variants.
>+template <std::size_t I, class T>
>+struct variant_alternative<I, const T> {
>+  using type = const typename variant_alternative<I, T>::type;
>+};
>+
>+// Specialization of `variant_alternative` for volatile qualified variants.
>+template <std::size_t I, class T>
>+struct variant_alternative<I, volatile T> {
>+  using type = volatile typename variant_alternative<I, T>::type;
>+};
>+
>+// Specialization of `variant_alternative` for const volatile qualified
>+// variants.
>+template <std::size_t I, class T>
>+struct variant_alternative<I, const volatile T> {
>+  using type = const volatile typename variant_alternative<I, T>::type;
>+};
>+
>+// Template type alias for variant_alternative<I, T>::type.
>+//
>+// Example:
>+//
>+//   using alternative_type_0
>+//     = absl::variant_alternative_t<0, absl::variant<int, std::string>>;
>+//   static_assert(std::is_same<alternative_type_0, int>::value, "");
>+template <std::size_t I, class T>
>+using variant_alternative_t = typename variant_alternative<I, T>::type;
>+
>+// holds_alternative()
>+//
>+// Checks whether the given variant currently holds a given alternative type,
>+// returning `true` if so.
>+//
>+// Example:
>+//
>+//   absl::variant<int, std::string> foo = 42;
>+//   if (absl::holds_alternative<int>(foo)) {
>+//       std::cout << "The variant holds an integer";
>+//   }
>+template <class T, class... Types>
>+constexpr bool holds_alternative(const variant<Types...>& v) noexcept {
>+  static_assert(
>+      variant_internal::UnambiguousIndexOfImpl<variant<Types...>, T,
>+                                               0>::value != sizeof...(Types),
>+      "The type T must occur exactly once in Types...");
>+  return v.index() ==
>+         variant_internal::UnambiguousIndexOf<variant<Types...>, T>::value;
>+}
>+
>+// get()
>+//
>+// Returns a reference to the value currently within a given variant, using
>+// either a unique alternative type amongst the variant's set of alternative
>+// types, or the variant's index value. Attempting to get a variant's value
>+// using a type that is not unique within the variant's set of alternative types
>+// is a compile-time error. If the index of the alternative being specified is
>+// different from the index of the alternative that is currently stored, throws
>+// `absl::bad_variant_access`.
>+//
>+// Example:
>+//
>+//   auto a = absl::variant<int, std::string>;
>+//
>+//   // Get the value by type (if unique).
>+//   int i = absl::get<int>(a);
>+//
>+//   auto b = absl::variant<int, int>;
>+//
>+//   // Getting the value by a type that is not unique is ill-formed.
>+//   int j = absl::get<int>(b);     // Compile Error!
>+//
>+//   // Getting value by index not ambiguous and allowed.
>+//   int k = absl::get<1>(b);
>+
>+// Overload for getting a variant's lvalue by type.
>+template <class T, class... Types>
>+constexpr T& get(variant<Types...>& v) {  // NOLINT
>+  return variant_internal::VariantCoreAccess::CheckedAccess<
>+      variant_internal::IndexOf<T, Types...>::value>(v);
>+}
>+
>+// Overload for getting a variant's rvalue by type.
>+// Note: `absl::move()` is required to allow use of constexpr in C++11.
>+template <class T, class... Types>
>+constexpr T&& get(variant<Types...>&& v) {
>+  return variant_internal::VariantCoreAccess::CheckedAccess<
>+      variant_internal::IndexOf<T, Types...>::value>(absl::move(v));
>+}
>+
>+// Overload for getting a variant's const lvalue by type.
>+template <class T, class... Types>
>+constexpr const T& get(const variant<Types...>& v) {
>+  return variant_internal::VariantCoreAccess::CheckedAccess<
>+      variant_internal::IndexOf<T, Types...>::value>(v);
>+}
>+
>+// Overload for getting a variant's const rvalue by type.
>+// Note: `absl::move()` is required to allow use of constexpr in C++11.
>+template <class T, class... Types>
>+constexpr const T&& get(const variant<Types...>&& v) {
>+  return variant_internal::VariantCoreAccess::CheckedAccess<
>+      variant_internal::IndexOf<T, Types...>::value>(absl::move(v));
>+}
>+
>+// Overload for getting a variant's lvalue by index.
>+template <std::size_t I, class... Types>
>+constexpr variant_alternative_t<I, variant<Types...>>& get(
>+    variant<Types...>& v) {  // NOLINT
>+  return variant_internal::VariantCoreAccess::CheckedAccess<I>(v);
>+}
>+
>+// Overload for getting a variant's rvalue by index.
>+// Note: `absl::move()` is required to allow use of constexpr in C++11.
>+template <std::size_t I, class... Types>
>+constexpr variant_alternative_t<I, variant<Types...>>&& get(
>+    variant<Types...>&& v) {
>+  return variant_internal::VariantCoreAccess::CheckedAccess<I>(absl::move(v));
>+}
>+
>+// Overload for getting a variant's const lvalue by index.
>+template <std::size_t I, class... Types>
>+constexpr const variant_alternative_t<I, variant<Types...>>& get(
>+    const variant<Types...>& v) {
>+  return variant_internal::VariantCoreAccess::CheckedAccess<I>(v);
>+}
>+
>+// Overload for getting a variant's const rvalue by index.
>+// Note: `absl::move()` is required to allow use of constexpr in C++11.
>+template <std::size_t I, class... Types>
>+constexpr const variant_alternative_t<I, variant<Types...>>&& get(
>+    const variant<Types...>&& v) {
>+  return variant_internal::VariantCoreAccess::CheckedAccess<I>(absl::move(v));
>+}
>+
>+// get_if()
>+//
>+// Returns a pointer to the value currently stored within a given variant, if
>+// present, using either a unique alternative type amongst the variant's set of
>+// alternative types, or the variant's index value. If such a value does not
>+// exist, returns `nullptr`.
>+//
>+// As with `get`, attempting to get a variant's value using a type that is not
>+// unique within the variant's set of alternative types is a compile-time error.
>+
>+// Overload for getting a pointer to the value stored in the given variant by
>+// index.
>+template <std::size_t I, class... Types>
>+constexpr absl::add_pointer_t<variant_alternative_t<I, variant<Types...>>>
>+get_if(variant<Types...>* v) noexcept {
>+  return (v != nullptr && v->index() == I)
>+             ? std::addressof(
>+                   variant_internal::VariantCoreAccess::Access<I>(*v))
>+             : nullptr;
>+}
>+
>+// Overload for getting a pointer to the const value stored in the given
>+// variant by index.
>+template <std::size_t I, class... Types>
>+constexpr absl::add_pointer_t<const variant_alternative_t<I, variant<Types...>>>
>+get_if(const variant<Types...>* v) noexcept {
>+  return (v != nullptr && v->index() == I)
>+             ? std::addressof(
>+                   variant_internal::VariantCoreAccess::Access<I>(*v))
>+             : nullptr;
>+}
>+
>+// Overload for getting a pointer to the value stored in the given variant by
>+// type.
>+template <class T, class... Types>
>+constexpr absl::add_pointer_t<T> get_if(variant<Types...>* v) noexcept {
>+  return absl::get_if<variant_internal::IndexOf<T, Types...>::value>(v);
>+}
>+
>+// Overload for getting a pointer to the const value stored in the given variant
>+// by type.
>+template <class T, class... Types>
>+constexpr absl::add_pointer_t<const T> get_if(
>+    const variant<Types...>* v) noexcept {
>+  return absl::get_if<variant_internal::IndexOf<T, Types...>::value>(v);
>+}
>+
>+// visit()
>+//
>+// Calls a provided functor on a given set of variants. `absl::visit()` is
>+// commonly used to conditionally inspect the state of a given variant (or set
>+// of variants).
>+// Requires: The expression in the Effects: element shall be a valid expression
>+// of the same type and value category, for all combinations of alternative
>+// types of all variants. Otherwise, the program is ill-formed.
>+//
>+// Example:
>+//
>+//   // Define a visitor functor
>+//   struct GetVariant {
>+//       template<typename T>
>+//       void operator()(const T& i) const {
>+//         std::cout << "The variant's value is: " << i;
>+//       }
>+//   };
>+//
>+//   // Declare our variant, and call `absl::visit()` on it.
>+//   std::variant<int, std::string> foo = std::string("foo");
>+//   GetVariant visitor;
>+//   std::visit(visitor, foo);  // Prints `The variant's value is: foo'
>+template <typename Visitor, typename... Variants>
>+variant_internal::VisitResult<Visitor, Variants...> visit(Visitor&& vis,
>+                                                          Variants&&... vars) {
>+  return variant_internal::
>+      VisitIndices<variant_size<absl::decay_t<Variants> >::value...>::Run(
>+          variant_internal::PerformVisitation<Visitor, Variants...>{
>+              std::forward_as_tuple(absl::forward<Variants>(vars)...),
>+              absl::forward<Visitor>(vis)},
>+          vars.index()...);
>+}
>+
>+// monostate
>+//
>+// The monostate class serves as a first alternative type for a variant for
>+// which the first variant type is otherwise not default-constructible.
>+struct monostate {};
>+
>+// `absl::monostate` Relational Operators
>+
>+constexpr bool operator<(monostate, monostate) noexcept { return false; }
>+constexpr bool operator>(monostate, monostate) noexcept { return false; }
>+constexpr bool operator<=(monostate, monostate) noexcept { return true; }
>+constexpr bool operator>=(monostate, monostate) noexcept { return true; }
>+constexpr bool operator==(monostate, monostate) noexcept { return true; }
>+constexpr bool operator!=(monostate, monostate) noexcept { return false; }
>+
>+
>+//------------------------------------------------------------------------------
>+// `absl::variant` Template Definition
>+//------------------------------------------------------------------------------
>+template <typename T0, typename... Tn>
>+class variant<T0, Tn...> : private variant_internal::VariantBase<T0, Tn...> {
>+  static_assert(absl::conjunction<std::is_object<T0>,
>+                                  std::is_object<Tn>...>::value,
>+                "Attempted to instantiate a variant containing a non-object "
>+                "type.");
>+  // Intentionally not qualifing `negation` with `absl::` to work around a bug
>+  // in MSVC 2015 with inline namespace and variadic template.
>+  static_assert(absl::conjunction<negation<std::is_array<T0> >,
>+                                  negation<std::is_array<Tn> >...>::value,
>+                "Attempted to instantiate a variant containing an array type.");
>+  static_assert(absl::conjunction<std::is_nothrow_destructible<T0>,
>+                                  std::is_nothrow_destructible<Tn>...>::value,
>+                "Attempted to instantiate a variant containing a non-nothrow "
>+                "destructible type.");
>+
>+  friend struct variant_internal::VariantCoreAccess;
>+
>+ private:
>+  using Base = variant_internal::VariantBase<T0, Tn...>;
>+
>+ public:
>+  // Constructors
>+
>+  // Constructs a variant holding a default-initialized value of the first
>+  // alternative type.
>+  constexpr variant() /*noexcept(see 111above)*/ = default;
>+
>+  // Copy constructor, standard semantics
>+  variant(const variant& other) = default;
>+
>+  // Move constructor, standard semantics
>+  variant(variant&& other) /*noexcept(see above)*/ = default;
>+
>+  // Constructs a variant of an alternative type specified by overload
>+  // resolution of the provided forwarding arguments through
>+  // direct-initialization.
>+  //
>+  // Note: If the selected constructor is a constexpr constructor, this
>+  // constructor shall be a constexpr constructor.
>+  //
>+  // NOTE: http://www.open-std.org/jtc1/sc22/wg21/docs/papers/2018/p0608r1.html
>+  // has been voted passed the design phase in the C++ standard meeting in Mar
>+  // 2018. It will be implemented and integrated into `absl::variant`.
>+  template <
>+      class T,
>+      std::size_t I = std::enable_if<
>+          variant_internal::IsNeitherSelfNorInPlace<variant,
>+                                                    absl::decay_t<T>>::value,
>+          variant_internal::IndexOfConstructedType<variant, T>>::type::value,
>+      class Tj = absl::variant_alternative_t<I, variant>,
>+      absl::enable_if_t<std::is_constructible<Tj, T>::value>* =
>+          nullptr>
>+  constexpr variant(T&& t) noexcept(std::is_nothrow_constructible<Tj, T>::value)
>+      : Base(variant_internal::EmplaceTag<I>(), absl::forward<T>(t)) {}
>+
>+  // Constructs a variant of an alternative type from the arguments through
>+  // direct-initialization.
>+  //
>+  // Note: If the selected constructor is a constexpr constructor, this
>+  // constructor shall be a constexpr constructor.
>+  template <class T, class... Args,
>+            typename std::enable_if<std::is_constructible<
>+                variant_internal::UnambiguousTypeOfT<variant, T>,
>+                Args...>::value>::type* = nullptr>
>+  constexpr explicit variant(in_place_type_t<T>, Args&&... args)
>+      : Base(variant_internal::EmplaceTag<
>+                 variant_internal::UnambiguousIndexOf<variant, T>::value>(),
>+             absl::forward<Args>(args)...) {}
>+
>+  // Constructs a variant of an alternative type from an initializer list
>+  // and other arguments through direct-initialization.
>+  //
>+  // Note: If the selected constructor is a constexpr constructor, this
>+  // constructor shall be a constexpr constructor.
>+  template <class T, class U, class... Args,
>+            typename std::enable_if<std::is_constructible<
>+                variant_internal::UnambiguousTypeOfT<variant, T>,
>+                std::initializer_list<U>&, Args...>::value>::type* = nullptr>
>+  constexpr explicit variant(in_place_type_t<T>, std::initializer_list<U> il,
>+                             Args&&... args)
>+      : Base(variant_internal::EmplaceTag<
>+                 variant_internal::UnambiguousIndexOf<variant, T>::value>(),
>+             il, absl::forward<Args>(args)...) {}
>+
>+  // Constructs a variant of an alternative type from a provided index,
>+  // through value-initialization using the provided forwarded arguments.
>+  template <std::size_t I, class... Args,
>+            typename std::enable_if<std::is_constructible<
>+                variant_internal::VariantAlternativeSfinaeT<I, variant>,
>+                Args...>::value>::type* = nullptr>
>+  constexpr explicit variant(in_place_index_t<I>, Args&&... args)
>+      : Base(variant_internal::EmplaceTag<I>(), absl::forward<Args>(args)...) {}
>+
>+  // Constructs a variant of an alternative type from a provided index,
>+  // through value-initialization of an initializer list and the provided
>+  // forwarded arguments.
>+  template <std::size_t I, class U, class... Args,
>+            typename std::enable_if<std::is_constructible<
>+                variant_internal::VariantAlternativeSfinaeT<I, variant>,
>+                std::initializer_list<U>&, Args...>::value>::type* = nullptr>
>+  constexpr explicit variant(in_place_index_t<I>, std::initializer_list<U> il,
>+                             Args&&... args)
>+      : Base(variant_internal::EmplaceTag<I>(), il,
>+             absl::forward<Args>(args)...) {}
>+
>+  // Destructors
>+
>+  // Destroys the variant's currently contained value, provided that
>+  // `absl::valueless_by_exception()` is false.
>+  ~variant() = default;
>+
>+  // Assignment Operators
>+
>+  // Copy assignement operator
>+  variant& operator=(const variant& other) = default;
>+
>+  // Move assignment operator
>+  variant& operator=(variant&& other) /*noexcept(see above)*/ = default;
>+
>+  // Converting assignment operator
>+  //
>+  // NOTE: http://www.open-std.org/jtc1/sc22/wg21/docs/papers/2018/p0608r1.html
>+  // has been voted passed the design phase in the C++ standard meeting in Mar
>+  // 2018. It will be implemented and integrated into `absl::variant`.
>+  template <
>+      class T,
>+      std::size_t I = std::enable_if<
>+          !std::is_same<absl::decay_t<T>, variant>::value,
>+          variant_internal::IndexOfConstructedType<variant, T>>::type::value,
>+      class Tj = absl::variant_alternative_t<I, variant>,
>+      typename std::enable_if<std::is_assignable<Tj&, T>::value &&
>+                              std::is_constructible<Tj, T>::value>::type* =
>+          nullptr>
>+  variant& operator=(T&& t) noexcept(
>+      std::is_nothrow_assignable<Tj&, T>::value&&
>+          std::is_nothrow_constructible<Tj, T>::value) {
>+    variant_internal::VisitIndices<sizeof...(Tn) + 1>::Run(
>+        variant_internal::VariantCoreAccess::MakeConversionAssignVisitor(
>+            this, absl::forward<T>(t)),
>+        index());
>+
>+    return *this;
>+  }
>+
>+
>+  // emplace() Functions
>+
>+  // Constructs a value of the given alternative type T within the variant.
>+  //
>+  // Example:
>+  //
>+  //   absl::variant<std::vector<int>, int, std::string> v;
>+  //   v.emplace<int>(99);
>+  //   v.emplace<std::string>("abc");
>+  template <
>+      class T, class... Args,
>+      typename std::enable_if<std::is_constructible<
>+          absl::variant_alternative_t<
>+              variant_internal::UnambiguousIndexOf<variant, T>::value, variant>,
>+          Args...>::value>::type* = nullptr>
>+  T& emplace(Args&&... args) {
>+    return variant_internal::VariantCoreAccess::Replace<
>+        variant_internal::UnambiguousIndexOf<variant, T>::value>(
>+        this, absl::forward<Args>(args)...);
>+  }
>+
>+  // Constructs a value of the given alternative type T within the variant using
>+  // an initializer list.
>+  //
>+  // Example:
>+  //
>+  //   absl::variant<std::vector<int>, int, std::string> v;
>+  //   v.emplace<std::vector<int>>({0, 1, 2});
>+  template <
>+      class T, class U, class... Args,
>+      typename std::enable_if<std::is_constructible<
>+          absl::variant_alternative_t<
>+              variant_internal::UnambiguousIndexOf<variant, T>::value, variant>,
>+          std::initializer_list<U>&, Args...>::value>::type* = nullptr>
>+  T& emplace(std::initializer_list<U> il, Args&&... args) {
>+    return variant_internal::VariantCoreAccess::Replace<
>+        variant_internal::UnambiguousIndexOf<variant, T>::value>(
>+        this, il, absl::forward<Args>(args)...);
>+  }
>+
>+  // Destroys the current value of the variant (provided that
>+  // `absl::valueless_by_exception()` is false, and constructs a new value at
>+  // the given index.
>+  //
>+  // Example:
>+  //
>+  //   absl::variant<std::vector<int>, int, int> v;
>+  //   v.emplace<1>(99);
>+  //   v.emplace<2>(98);
>+  //   v.emplace<int>(99);  // Won't compile. 'int' isn't a unique type.
>+  template <std::size_t I, class... Args,
>+            typename std::enable_if<
>+                std::is_constructible<absl::variant_alternative_t<I, variant>,
>+                                      Args...>::value>::type* = nullptr>
>+  absl::variant_alternative_t<I, variant>& emplace(Args&&... args) {
>+    return variant_internal::VariantCoreAccess::Replace<I>(
>+        this, absl::forward<Args>(args)...);
>+  }
>+
>+  // Destroys the current value of the variant (provided that
>+  // `absl::valueless_by_exception()` is false, and constructs a new value at
>+  // the given index using an initializer list and the provided arguments.
>+  //
>+  // Example:
>+  //
>+  //   absl::variant<std::vector<int>, int, int> v;
>+  //   v.emplace<0>({0, 1, 2});
>+  template <std::size_t I, class U, class... Args,
>+            typename std::enable_if<std::is_constructible<
>+                absl::variant_alternative_t<I, variant>,
>+                std::initializer_list<U>&, Args...>::value>::type* = nullptr>
>+  absl::variant_alternative_t<I, variant>& emplace(std::initializer_list<U> il,
>+                                                   Args&&... args) {
>+    return variant_internal::VariantCoreAccess::Replace<I>(
>+        this, il, absl::forward<Args>(args)...);
>+  }
>+
>+  // variant::valueless_by_exception()
>+  //
>+  // Returns false if and only if the variant currently holds a valid value.
>+  constexpr bool valueless_by_exception() const noexcept {
>+    return this->index_ == absl::variant_npos;
>+  }
>+
>+  // variant::index()
>+  //
>+  // Returns the index value of the variant's currently selected alternative
>+  // type.
>+  constexpr std::size_t index() const noexcept { return this->index_; }
>+
>+  // variant::swap()
>+  //
>+  // Swaps the values of two variant objects.
>+  //
>+  // TODO(calabrese)
>+  //   `variant::swap()` and `swap()` rely on `std::is_(nothrow)_swappable()`
>+  //   which is introduced in C++17. So we assume `is_swappable()` is always
>+  //   true and `is_nothrow_swappable()` is same as `std::is_trivial()`.
>+  void swap(variant& rhs) noexcept(
>+      absl::conjunction<std::is_trivial<T0>, std::is_trivial<Tn>...>::value) {
>+    return variant_internal::VisitIndices<sizeof...(Tn) + 1>::Run(
>+        variant_internal::Swap<T0, Tn...>{this, &rhs}, rhs.index());
>+  }
>+};
>+
>+// We need a valid declaration of variant<> for SFINAE and overload resolution
>+// to work properly above, but we don't need a full declaration since this type
>+// will never be constructed. This declaration, though incomplete, suffices.
>+template <>
>+class variant<>;
>+
>+//------------------------------------------------------------------------------
>+// Relational Operators
>+//------------------------------------------------------------------------------
>+//
>+// If neither operand is in the `variant::valueless_by_exception` state:
>+//
>+//   * If the index of both variants is the same, the relational operator
>+//     returns the result of the corresponding relational operator for the
>+//     corresponding alternative type.
>+//   * If the index of both variants is not the same, the relational operator
>+//     returns the result of that operation applied to the value of the left
>+//     operand's index and the value of the right operand's index.
>+//   * If at least one operand is in the valueless_by_exception state:
>+//     - A variant in the valueless_by_exception state is only considered equal
>+//       to another variant in the valueless_by_exception state.
>+//     - If exactly one operand is in the valueless_by_exception state, the
>+//       variant in the valueless_by_exception state is less than the variant
>+//       that is not in the valueless_by_exception state.
>+//
>+// Note: The value 1 is added to each index in the relational comparisons such
>+// that the index corresponding to the valueless_by_exception state wraps around
>+// to 0 (the lowest value for the index type), and the remaining indices stay in
>+// the same relative order.
>+
>+// Equal-to operator
>+template <typename... Types>
>+constexpr variant_internal::RequireAllHaveEqualT<Types...> operator==(
>+    const variant<Types...>& a, const variant<Types...>& b) {
>+  return (a.index() == b.index()) &&
>+         variant_internal::VisitIndices<sizeof...(Types)>::Run(
>+             variant_internal::EqualsOp<Types...>{&a, &b}, a.index());
>+}
>+
>+// Not equal operator
>+template <typename... Types>
>+constexpr variant_internal::RequireAllHaveNotEqualT<Types...> operator!=(
>+    const variant<Types...>& a, const variant<Types...>& b) {
>+  return (a.index() != b.index()) ||
>+         variant_internal::VisitIndices<sizeof...(Types)>::Run(
>+             variant_internal::NotEqualsOp<Types...>{&a, &b}, a.index());
>+}
>+
>+// Less-than operator
>+template <typename... Types>
>+constexpr variant_internal::RequireAllHaveLessThanT<Types...> operator<(
>+    const variant<Types...>& a, const variant<Types...>& b) {
>+  return (a.index() != b.index())
>+             ? (a.index() + 1) < (b.index() + 1)
>+             : variant_internal::VisitIndices<sizeof...(Types)>::Run(
>+                   variant_internal::LessThanOp<Types...>{&a, &b}, a.index());
>+}
>+
>+// Greater-than operator
>+template <typename... Types>
>+constexpr variant_internal::RequireAllHaveGreaterThanT<Types...> operator>(
>+    const variant<Types...>& a, const variant<Types...>& b) {
>+  return (a.index() != b.index())
>+             ? (a.index() + 1) > (b.index() + 1)
>+             : variant_internal::VisitIndices<sizeof...(Types)>::Run(
>+                   variant_internal::GreaterThanOp<Types...>{&a, &b},
>+                   a.index());
>+}
>+
>+// Less-than or equal-to operator
>+template <typename... Types>
>+constexpr variant_internal::RequireAllHaveLessThanOrEqualT<Types...> operator<=(
>+    const variant<Types...>& a, const variant<Types...>& b) {
>+  return (a.index() != b.index())
>+             ? (a.index() + 1) < (b.index() + 1)
>+             : variant_internal::VisitIndices<sizeof...(Types)>::Run(
>+                   variant_internal::LessThanOrEqualsOp<Types...>{&a, &b},
>+                   a.index());
>+}
>+
>+// Greater-than or equal-to operator
>+template <typename... Types>
>+constexpr variant_internal::RequireAllHaveGreaterThanOrEqualT<Types...>
>+operator>=(const variant<Types...>& a, const variant<Types...>& b) {
>+  return (a.index() != b.index())
>+             ? (a.index() + 1) > (b.index() + 1)
>+             : variant_internal::VisitIndices<sizeof...(Types)>::Run(
>+                   variant_internal::GreaterThanOrEqualsOp<Types...>{&a, &b},
>+                   a.index());
>+}
>+
>+}  // namespace absl
>+
>+namespace std {
>+
>+// hash()
>+template <>  // NOLINT
>+struct hash<absl::monostate> {
>+  std::size_t operator()(absl::monostate) const { return 0; }
>+};
>+
>+template <class... T>  // NOLINT
>+struct hash<absl::variant<T...>>
>+    : absl::variant_internal::VariantHashBase<absl::variant<T...>, void,
>+                                              absl::remove_const_t<T>...> {};
>+
>+}  // namespace std
>+
>+#endif  // ABSL_HAVE_STD_VARIANT
>+
>+namespace absl {
>+namespace variant_internal {
>+
>+// Helper visitor for converting a variant<Ts...>` into another type (mostly
>+// variant) that can be constructed from any type.
>+template <typename To>
>+struct ConversionVisitor {
>+  template <typename T>
>+  To operator()(T&& v) const {
>+    return To(std::forward<T>(v));
>+  }
>+};
>+
>+}  // namespace variant_internal
>+
>+// ConvertVariantTo()
>+//
>+// Helper functions to convert an `absl::variant` to a variant of another set of
>+// types, provided that the alternative type of the new variant type can be
>+// converted from any type in the source variant.
>+//
>+// Example:
>+//
>+//   absl::variant<name1, name2, float> InternalReq(const Req&);
>+//
>+//   // name1 and name2 are convertible to name
>+//   absl::variant<name, float> ExternalReq(const Req& req) {
>+//     return absl::ConvertVariantTo<absl::variant<name, float>>(
>+//              InternalReq(req));
>+//   }
>+template <typename To, typename Variant>
>+To ConvertVariantTo(Variant&& variant) {
>+  return absl::visit(variant_internal::ConversionVisitor<To>{},
>+                     std::forward<Variant>(variant));
>+}
>+
>+}  // namespace absl
>+
>+#endif  // ABSL_TYPES_VARIANT_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/variant_benchmark.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/variant_benchmark.cc
>new file mode 100644
>index 00000000000..99658ac70c4
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/variant_benchmark.cc
>@@ -0,0 +1,220 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+// Unit tests for the variant template. The 'is' and 'IsEmpty' methods
>+// of variant are not explicitly tested because they are used repeatedly
>+// in building other tests. All other public variant methods should have
>+// explicit tests.
>+
>+#include "absl/types/variant.h"
>+
>+#include <cstddef>
>+#include <cstdlib>
>+#include <string>
>+#include <tuple>
>+
>+#include "benchmark/benchmark.h"
>+#include "absl/utility/utility.h"
>+
>+namespace absl {
>+namespace {
>+
>+template <std::size_t I>
>+struct VariantAlternative {
>+  char member;
>+};
>+
>+template <class Indices>
>+struct VariantOfAlternativesImpl;
>+
>+template <std::size_t... Indices>
>+struct VariantOfAlternativesImpl<absl::index_sequence<Indices...>> {
>+  using type = absl::variant<VariantAlternative<Indices>...>;
>+};
>+
>+template <std::size_t NumAlternatives>
>+using VariantOfAlternatives = typename VariantOfAlternativesImpl<
>+    absl::make_index_sequence<NumAlternatives>>::type;
>+
>+struct Empty {};
>+
>+template <class... T>
>+void Ignore(T...) noexcept {}
>+
>+template <class T>
>+Empty DoNotOptimizeAndReturnEmpty(T&& arg) noexcept {
>+  benchmark::DoNotOptimize(arg);
>+  return {};
>+}
>+
>+struct VisitorApplier {
>+  struct Visitor {
>+    template <class... T>
>+    void operator()(T&&... args) const noexcept {
>+      Ignore(DoNotOptimizeAndReturnEmpty(args)...);
>+    }
>+  };
>+
>+  template <class... Vars>
>+  void operator()(const Vars&... vars) const noexcept {
>+    absl::visit(Visitor(), vars...);
>+  }
>+};
>+
>+template <std::size_t NumIndices, std::size_t CurrIndex = NumIndices - 1>
>+struct MakeWithIndex {
>+  using Variant = VariantOfAlternatives<NumIndices>;
>+
>+  static Variant Run(std::size_t index) {
>+    return index == CurrIndex
>+               ? Variant(absl::in_place_index_t<CurrIndex>())
>+               : MakeWithIndex<NumIndices, CurrIndex - 1>::Run(index);
>+  }
>+};
>+
>+template <std::size_t NumIndices>
>+struct MakeWithIndex<NumIndices, 0> {
>+  using Variant = VariantOfAlternatives<NumIndices>;
>+
>+  static Variant Run(std::size_t /*index*/) { return Variant(); }
>+};
>+
>+template <std::size_t NumIndices, class Dimensions>
>+struct MakeVariantTuple;
>+
>+template <class T, std::size_t /*I*/>
>+using always_t = T;
>+
>+template <std::size_t NumIndices>
>+VariantOfAlternatives<NumIndices> MakeVariant(std::size_t dimension,
>+                                              std::size_t index) {
>+  return dimension == 0
>+             ? MakeWithIndex<NumIndices>::Run(index % NumIndices)
>+             : MakeVariant<NumIndices>(dimension - 1, index / NumIndices);
>+}
>+
>+template <std::size_t NumIndices, std::size_t... Dimensions>
>+struct MakeVariantTuple<NumIndices, absl::index_sequence<Dimensions...>> {
>+  using VariantTuple =
>+      std::tuple<always_t<VariantOfAlternatives<NumIndices>, Dimensions>...>;
>+
>+  static VariantTuple Run(int index) {
>+    return std::make_tuple(MakeVariant<NumIndices>(Dimensions, index)...);
>+  }
>+};
>+
>+constexpr std::size_t integral_pow(std::size_t base, std::size_t power) {
>+  return power == 0 ? 1 : base * integral_pow(base, power - 1);
>+}
>+
>+template <std::size_t End, std::size_t I = 0>
>+struct VisitTestBody {
>+  template <class Vars, class State>
>+  static bool Run(Vars& vars, State& state) {
>+    if (state.KeepRunning()) {
>+      absl::apply(VisitorApplier(), vars[I]);
>+      return VisitTestBody<End, I + 1>::Run(vars, state);
>+    }
>+    return false;
>+  }
>+};
>+
>+template <std::size_t End>
>+struct VisitTestBody<End, End> {
>+  template <class Vars, class State>
>+  static bool Run(Vars& /*vars*/, State& /*state*/) {
>+    return true;
>+  }
>+};
>+
>+// Visit operations where branch prediction is likely to give a boost.
>+template <std::size_t NumIndices, std::size_t NumDimensions = 1>
>+void BM_RedundantVisit(benchmark::State& state) {
>+  auto vars =
>+      MakeVariantTuple<NumIndices, absl::make_index_sequence<NumDimensions>>::
>+          Run(static_cast<std::size_t>(state.range(0)));
>+
>+  for (auto _ : state) {  // NOLINT
>+    benchmark::DoNotOptimize(vars);
>+    absl::apply(VisitorApplier(), vars);
>+  }
>+}
>+
>+// Visit operations where branch prediction is unlikely to give a boost.
>+template <std::size_t NumIndices, std::size_t NumDimensions = 1>
>+void BM_Visit(benchmark::State& state) {
>+  constexpr std::size_t num_possibilities =
>+      integral_pow(NumIndices, NumDimensions);
>+
>+  using VariantTupleMaker =
>+      MakeVariantTuple<NumIndices, absl::make_index_sequence<NumDimensions>>;
>+  using Tuple = typename VariantTupleMaker::VariantTuple;
>+
>+  Tuple vars[num_possibilities];
>+  for (std::size_t i = 0; i < num_possibilities; ++i)
>+    vars[i] = VariantTupleMaker::Run(i);
>+
>+  while (VisitTestBody<num_possibilities>::Run(vars, state)) {
>+  }
>+}
>+
>+// Visitation
>+//   Each visit is on a different variant with a different active alternative)
>+
>+// Unary visit
>+BENCHMARK_TEMPLATE(BM_Visit, 1);
>+BENCHMARK_TEMPLATE(BM_Visit, 2);
>+BENCHMARK_TEMPLATE(BM_Visit, 3);
>+BENCHMARK_TEMPLATE(BM_Visit, 4);
>+BENCHMARK_TEMPLATE(BM_Visit, 5);
>+BENCHMARK_TEMPLATE(BM_Visit, 6);
>+BENCHMARK_TEMPLATE(BM_Visit, 7);
>+BENCHMARK_TEMPLATE(BM_Visit, 8);
>+BENCHMARK_TEMPLATE(BM_Visit, 16);
>+BENCHMARK_TEMPLATE(BM_Visit, 32);
>+BENCHMARK_TEMPLATE(BM_Visit, 64);
>+
>+// Binary visit
>+BENCHMARK_TEMPLATE(BM_Visit, 1, 2);
>+BENCHMARK_TEMPLATE(BM_Visit, 2, 2);
>+BENCHMARK_TEMPLATE(BM_Visit, 3, 2);
>+BENCHMARK_TEMPLATE(BM_Visit, 4, 2);
>+BENCHMARK_TEMPLATE(BM_Visit, 5, 2);
>+
>+// Ternary visit
>+BENCHMARK_TEMPLATE(BM_Visit, 1, 3);
>+BENCHMARK_TEMPLATE(BM_Visit, 2, 3);
>+BENCHMARK_TEMPLATE(BM_Visit, 3, 3);
>+
>+// Quaternary visit
>+BENCHMARK_TEMPLATE(BM_Visit, 1, 4);
>+BENCHMARK_TEMPLATE(BM_Visit, 2, 4);
>+
>+// Redundant Visitation
>+//   Each visit consistently has the same alternative active
>+
>+// Unary visit
>+BENCHMARK_TEMPLATE(BM_RedundantVisit, 1)->Arg(0);
>+BENCHMARK_TEMPLATE(BM_RedundantVisit, 2)->DenseRange(0, 1);
>+BENCHMARK_TEMPLATE(BM_RedundantVisit, 8)->DenseRange(0, 7);
>+
>+// Binary visit
>+BENCHMARK_TEMPLATE(BM_RedundantVisit, 1, 2)->Arg(0);
>+BENCHMARK_TEMPLATE(BM_RedundantVisit, 2, 2)
>+    ->DenseRange(0, integral_pow(2, 2) - 1);
>+BENCHMARK_TEMPLATE(BM_RedundantVisit, 4, 2)
>+    ->DenseRange(0, integral_pow(4, 2) - 1);
>+
>+}  // namespace
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/variant_exception_safety_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/variant_exception_safety_test.cc
>new file mode 100644
>index 00000000000..27c0b96ca6f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/variant_exception_safety_test.cc
>@@ -0,0 +1,508 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+#include "absl/types/variant.h"
>+
>+#include <iostream>
>+#include <memory>
>+#include <utility>
>+#include <vector>
>+
>+#include "gmock/gmock.h"
>+#include "gtest/gtest.h"
>+#include "absl/base/internal/exception_safety_testing.h"
>+#include "absl/memory/memory.h"
>+
>+namespace absl {
>+namespace {
>+
>+using ::testing::MakeExceptionSafetyTester;
>+using ::testing::strong_guarantee;
>+using ::testing::TestNothrowOp;
>+using ::testing::TestThrowingCtor;
>+
>+using Thrower = testing::ThrowingValue<>;
>+using CopyNothrow = testing::ThrowingValue<testing::TypeSpec::kNoThrowCopy>;
>+using MoveNothrow = testing::ThrowingValue<testing::TypeSpec::kNoThrowMove>;
>+using ThrowingAlloc = testing::ThrowingAllocator<Thrower>;
>+using ThrowerVec = std::vector<Thrower, ThrowingAlloc>;
>+using ThrowingVariant =
>+    absl::variant<Thrower, CopyNothrow, MoveNothrow, ThrowerVec>;
>+
>+struct ConversionException {};
>+
>+template <class T>
>+struct ExceptionOnConversion {
>+  operator T() const {  // NOLINT
>+    throw ConversionException();
>+  }
>+};
>+
>+// Forces a variant into the valueless by exception state.
>+void ToValuelessByException(ThrowingVariant& v) {  // NOLINT
>+  try {
>+    v.emplace<Thrower>();
>+    v.emplace<Thrower>(ExceptionOnConversion<Thrower>());
>+  } catch (ConversionException& /*e*/) {
>+    // This space intentionally left blank.
>+  }
>+}
>+
>+// Check that variant is still in a usable state after an exception is thrown.
>+testing::AssertionResult CheckInvariants(ThrowingVariant* v) {
>+  using testing::AssertionFailure;
>+  using testing::AssertionSuccess;
>+
>+  // Try using the active alternative
>+  if (absl::holds_alternative<Thrower>(*v)) {
>+    auto& t = absl::get<Thrower>(*v);
>+    t = Thrower{-100};
>+    if (t.Get() != -100) {
>+      return AssertionFailure() << "Thrower should be assigned -100";
>+    }
>+  } else if (absl::holds_alternative<ThrowerVec>(*v)) {
>+    auto& tv = absl::get<ThrowerVec>(*v);
>+    tv.clear();
>+    tv.emplace_back(-100);
>+    if (tv.size() != 1 || tv[0].Get() != -100) {
>+      return AssertionFailure() << "ThrowerVec should be {Thrower{-100}}";
>+    }
>+  } else if (absl::holds_alternative<CopyNothrow>(*v)) {
>+    auto& t = absl::get<CopyNothrow>(*v);
>+    t = CopyNothrow{-100};
>+    if (t.Get() != -100) {
>+      return AssertionFailure() << "CopyNothrow should be assigned -100";
>+    }
>+  } else if (absl::holds_alternative<MoveNothrow>(*v)) {
>+    auto& t = absl::get<MoveNothrow>(*v);
>+    t = MoveNothrow{-100};
>+    if (t.Get() != -100) {
>+      return AssertionFailure() << "MoveNothrow should be assigned -100";
>+    }
>+  }
>+
>+  // Try making variant valueless_by_exception
>+  if (!v->valueless_by_exception()) ToValuelessByException(*v);
>+  if (!v->valueless_by_exception()) {
>+    return AssertionFailure() << "Variant should be valueless_by_exception";
>+  }
>+  try {
>+    auto unused = absl::get<Thrower>(*v);
>+    static_cast<void>(unused);
>+    return AssertionFailure() << "Variant should not contain Thrower";
>+  } catch (absl::bad_variant_access) {
>+  } catch (...) {
>+    return AssertionFailure() << "Unexpected exception throw from absl::get";
>+  }
>+
>+  // Try using the variant
>+  v->emplace<Thrower>(100);
>+  if (!absl::holds_alternative<Thrower>(*v) ||
>+      absl::get<Thrower>(*v) != Thrower(100)) {
>+    return AssertionFailure() << "Variant should contain Thrower(100)";
>+  }
>+  v->emplace<ThrowerVec>({Thrower(100)});
>+  if (!absl::holds_alternative<ThrowerVec>(*v) ||
>+      absl::get<ThrowerVec>(*v)[0] != Thrower(100)) {
>+    return AssertionFailure()
>+           << "Variant should contain ThrowerVec{Thrower(100)}";
>+  }
>+  return AssertionSuccess();
>+}
>+
>+template <typename... Args>
>+Thrower ExpectedThrower(Args&&... args) {
>+  return Thrower(42, args...);
>+}
>+
>+ThrowerVec ExpectedThrowerVec() { return {Thrower(100), Thrower(200)}; }
>+ThrowingVariant ValuelessByException() {
>+  ThrowingVariant v;
>+  ToValuelessByException(v);
>+  return v;
>+}
>+ThrowingVariant WithThrower() { return Thrower(39); }
>+ThrowingVariant WithThrowerVec() {
>+  return ThrowerVec{Thrower(1), Thrower(2), Thrower(3)};
>+}
>+ThrowingVariant WithCopyNoThrow() { return CopyNothrow(39); }
>+ThrowingVariant WithMoveNoThrow() { return MoveNothrow(39); }
>+
>+TEST(VariantExceptionSafetyTest, DefaultConstructor) {
>+  TestThrowingCtor<ThrowingVariant>();
>+}
>+
>+TEST(VariantExceptionSafetyTest, CopyConstructor) {
>+  {
>+    ThrowingVariant v(ExpectedThrower());
>+    TestThrowingCtor<ThrowingVariant>(v);
>+  }
>+  {
>+    ThrowingVariant v(ExpectedThrowerVec());
>+    TestThrowingCtor<ThrowingVariant>(v);
>+  }
>+  {
>+    ThrowingVariant v(ValuelessByException());
>+    TestThrowingCtor<ThrowingVariant>(v);
>+  }
>+}
>+
>+TEST(VariantExceptionSafetyTest, MoveConstructor) {
>+  {
>+    ThrowingVariant v(ExpectedThrower());
>+    TestThrowingCtor<ThrowingVariant>(std::move(v));
>+  }
>+  {
>+    ThrowingVariant v(ExpectedThrowerVec());
>+    TestThrowingCtor<ThrowingVariant>(std::move(v));
>+  }
>+  {
>+    ThrowingVariant v(ValuelessByException());
>+    TestThrowingCtor<ThrowingVariant>(std::move(v));
>+  }
>+}
>+
>+TEST(VariantExceptionSafetyTest, ValueConstructor) {
>+  TestThrowingCtor<ThrowingVariant>(ExpectedThrower());
>+  TestThrowingCtor<ThrowingVariant>(ExpectedThrowerVec());
>+}
>+
>+TEST(VariantExceptionSafetyTest, InPlaceTypeConstructor) {
>+  TestThrowingCtor<ThrowingVariant>(absl::in_place_type_t<Thrower>{},
>+                                    ExpectedThrower());
>+  TestThrowingCtor<ThrowingVariant>(absl::in_place_type_t<ThrowerVec>{},
>+                                    ExpectedThrowerVec());
>+}
>+
>+TEST(VariantExceptionSafetyTest, InPlaceIndexConstructor) {
>+  TestThrowingCtor<ThrowingVariant>(absl::in_place_index_t<0>{},
>+                                    ExpectedThrower());
>+  TestThrowingCtor<ThrowingVariant>(absl::in_place_index_t<3>{},
>+                                    ExpectedThrowerVec());
>+}
>+
>+TEST(VariantExceptionSafetyTest, CopyAssign) {
>+  // variant& operator=(const variant& rhs);
>+  // Let j be rhs.index()
>+  {
>+    // - neither *this nor rhs holds a value
>+    const ThrowingVariant rhs = ValuelessByException();
>+    ThrowingVariant lhs = ValuelessByException();
>+    EXPECT_TRUE(TestNothrowOp([&]() { lhs = rhs; }));
>+  }
>+  {
>+    // - *this holds a value but rhs does not
>+    const ThrowingVariant rhs = ValuelessByException();
>+    ThrowingVariant lhs = WithThrower();
>+    EXPECT_TRUE(TestNothrowOp([&]() { lhs = rhs; }));
>+  }
>+  // - index() == j
>+  {
>+    const ThrowingVariant rhs(ExpectedThrower());
>+    auto tester =
>+        MakeExceptionSafetyTester()
>+            .WithInitialValue(WithThrower())
>+            .WithOperation([&rhs](ThrowingVariant* lhs) { *lhs = rhs; });
>+    EXPECT_TRUE(tester.WithInvariants(CheckInvariants).Test());
>+    EXPECT_FALSE(tester.WithInvariants(strong_guarantee).Test());
>+  }
>+  {
>+    const ThrowingVariant rhs(ExpectedThrowerVec());
>+    auto tester =
>+        MakeExceptionSafetyTester()
>+            .WithInitialValue(WithThrowerVec())
>+            .WithOperation([&rhs](ThrowingVariant* lhs) { *lhs = rhs; });
>+    EXPECT_TRUE(tester.WithInvariants(CheckInvariants).Test());
>+    EXPECT_FALSE(tester.WithInvariants(strong_guarantee).Test());
>+  }
>+  // libstdc++ std::variant has bugs on copy assignment regarding exception
>+  // safety.
>+#if !(defined(ABSL_HAVE_STD_VARIANT) && defined(__GLIBCXX__))
>+  // index() != j
>+  // if is_nothrow_copy_constructible_v<Tj> or
>+  // !is_nothrow_move_constructible<Tj> is true, equivalent to
>+  // emplace<j>(get<j>(rhs))
>+  {
>+    // is_nothrow_copy_constructible_v<Tj> == true
>+    // should not throw because emplace() invokes Tj's copy ctor
>+    // which should not throw.
>+    const ThrowingVariant rhs(CopyNothrow{});
>+    ThrowingVariant lhs = WithThrower();
>+    EXPECT_TRUE(TestNothrowOp([&]() { lhs = rhs; }));
>+  }
>+  {
>+    // is_nothrow_copy_constructible<Tj> == false &&
>+    // is_nothrow_move_constructible<Tj> == false
>+    // should provide basic guarantee because emplace() invokes Tj's copy ctor
>+    // which may throw.
>+    const ThrowingVariant rhs(ExpectedThrower());
>+    auto tester =
>+        MakeExceptionSafetyTester()
>+            .WithInitialValue(WithCopyNoThrow())
>+            .WithOperation([&rhs](ThrowingVariant* lhs) { *lhs = rhs; });
>+    EXPECT_TRUE(tester
>+                    .WithInvariants(CheckInvariants,
>+                                    [](ThrowingVariant* lhs) {
>+                                      return lhs->valueless_by_exception();
>+                                    })
>+                    .Test());
>+    EXPECT_FALSE(tester.WithInvariants(strong_guarantee).Test());
>+  }
>+#endif  // !(defined(ABSL_HAVE_STD_VARIANT) && defined(__GLIBCXX__))
>+  {
>+    // is_nothrow_copy_constructible_v<Tj> == false &&
>+    // is_nothrow_move_constructible_v<Tj> == true
>+    // should provide strong guarantee because it is equivalent to
>+    // operator=(variant(rhs)) which creates a temporary then invoke the move
>+    // ctor which shouldn't throw.
>+    const ThrowingVariant rhs(MoveNothrow{});
>+    EXPECT_TRUE(MakeExceptionSafetyTester()
>+                    .WithInitialValue(WithThrower())
>+                    .WithInvariants(CheckInvariants, strong_guarantee)
>+                    .Test([&rhs](ThrowingVariant* lhs) { *lhs = rhs; }));
>+  }
>+}
>+
>+TEST(VariantExceptionSafetyTest, MoveAssign) {
>+  // variant& operator=(variant&& rhs);
>+  // Let j be rhs.index()
>+  {
>+    // - neither *this nor rhs holds a value
>+    ThrowingVariant rhs = ValuelessByException();
>+    ThrowingVariant lhs = ValuelessByException();
>+    EXPECT_TRUE(TestNothrowOp([&]() { lhs = std::move(rhs); }));
>+  }
>+  {
>+    // - *this holds a value but rhs does not
>+    ThrowingVariant rhs = ValuelessByException();
>+    ThrowingVariant lhs = WithThrower();
>+    EXPECT_TRUE(TestNothrowOp([&]() { lhs = std::move(rhs); }));
>+  }
>+  {
>+    // - index() == j
>+    // assign get<j>(std::move(rhs)) to the value contained in *this.
>+    // If an exception is thrown during call to Tj's move assignment, the state
>+    // of the contained value is as defined by the exception safety guarantee of
>+    // Tj's move assignment; index() will be j.
>+    ThrowingVariant rhs(ExpectedThrower());
>+    size_t j = rhs.index();
>+    // Since Thrower's move assignment has basic guarantee, so should variant's.
>+    auto tester = MakeExceptionSafetyTester()
>+                      .WithInitialValue(WithThrower())
>+                      .WithOperation([&](ThrowingVariant* lhs) {
>+                        auto copy = rhs;
>+                        *lhs = std::move(copy);
>+                      });
>+    EXPECT_TRUE(tester
>+                    .WithInvariants(
>+                        CheckInvariants,
>+                        [&](ThrowingVariant* lhs) { return lhs->index() == j; })
>+                    .Test());
>+    EXPECT_FALSE(tester.WithInvariants(strong_guarantee).Test());
>+  }
>+  {
>+    // - otherwise (index() != j), equivalent to
>+    // emplace<j>(get<j>(std::move(rhs)))
>+    // - If an exception is thrown during the call to Tj's move construction
>+    // (with j being rhs.index()), the variant will hold no value.
>+    ThrowingVariant rhs(CopyNothrow{});
>+    EXPECT_TRUE(MakeExceptionSafetyTester()
>+                    .WithInitialValue(WithThrower())
>+                    .WithInvariants(CheckInvariants,
>+                                    [](ThrowingVariant* lhs) {
>+                                      return lhs->valueless_by_exception();
>+                                    })
>+                    .Test([&](ThrowingVariant* lhs) {
>+                      auto copy = rhs;
>+                      *lhs = std::move(copy);
>+                    }));
>+  }
>+}
>+
>+TEST(VariantExceptionSafetyTest, ValueAssign) {
>+  // template<class T> variant& operator=(T&& t);
>+  // Let Tj be the type that is selected by overload resolution to be assigned.
>+  {
>+    // If *this holds a Tj, assigns std::forward<T>(t) to the value contained in
>+    // *this. If  an exception is thrown during the assignment of
>+    // std::forward<T>(t) to the value contained in *this, the state of the
>+    // contained value and t are as defined by the exception safety guarantee of
>+    // the assignment expression; valueless_by_exception() will be false.
>+    // Since Thrower's copy/move assignment has basic guarantee, so should
>+    // variant's.
>+    Thrower rhs = ExpectedThrower();
>+    // copy assign
>+    auto copy_tester =
>+        MakeExceptionSafetyTester()
>+            .WithInitialValue(WithThrower())
>+            .WithOperation([rhs](ThrowingVariant* lhs) { *lhs = rhs; });
>+    EXPECT_TRUE(copy_tester
>+                    .WithInvariants(CheckInvariants,
>+                                    [](ThrowingVariant* lhs) {
>+                                      return !lhs->valueless_by_exception();
>+                                    })
>+                    .Test());
>+    EXPECT_FALSE(copy_tester.WithInvariants(strong_guarantee).Test());
>+    // move assign
>+    auto move_tester = MakeExceptionSafetyTester()
>+                           .WithInitialValue(WithThrower())
>+                           .WithOperation([&](ThrowingVariant* lhs) {
>+                             auto copy = rhs;
>+                             *lhs = std::move(copy);
>+                           });
>+    EXPECT_TRUE(move_tester
>+                    .WithInvariants(CheckInvariants,
>+                                    [](ThrowingVariant* lhs) {
>+                                      return !lhs->valueless_by_exception();
>+                                    })
>+                    .Test());
>+
>+    EXPECT_FALSE(move_tester.WithInvariants(strong_guarantee).Test());
>+  }
>+  // Otherwise (*this holds something else), if is_nothrow_constructible_v<Tj,
>+  // T> || !is_nothrow_move_constructible_v<Tj> is true, equivalent to
>+  // emplace<j>(std::forward<T>(t)).
>+  // We simplify the test by letting T = `const Tj&`  or `Tj&&`, so we can reuse
>+  // the CopyNothrow and MoveNothrow types.
>+
>+  // if is_nothrow_constructible_v<Tj, T>
>+  // (i.e. is_nothrow_copy/move_constructible_v<Tj>) is true, emplace() just
>+  // invokes the copy/move constructor and it should not throw.
>+  {
>+    const CopyNothrow rhs;
>+    ThrowingVariant lhs = WithThrower();
>+    EXPECT_TRUE(TestNothrowOp([&]() { lhs = rhs; }));
>+  }
>+  {
>+    MoveNothrow rhs;
>+    ThrowingVariant lhs = WithThrower();
>+    EXPECT_TRUE(TestNothrowOp([&]() { lhs = std::move(rhs); }));
>+  }
>+  // if is_nothrow_constructible_v<Tj, T> == false &&
>+  // is_nothrow_move_constructible<Tj> == false
>+  // emplace() invokes the copy/move constructor which may throw so it should
>+  // provide basic guarantee and variant object might not hold a value.
>+  {
>+    Thrower rhs = ExpectedThrower();
>+    // copy
>+    auto copy_tester =
>+        MakeExceptionSafetyTester()
>+            .WithInitialValue(WithCopyNoThrow())
>+            .WithOperation([&rhs](ThrowingVariant* lhs) { *lhs = rhs; });
>+    EXPECT_TRUE(copy_tester
>+                    .WithInvariants(CheckInvariants,
>+                                    [](ThrowingVariant* lhs) {
>+                                      return lhs->valueless_by_exception();
>+                                    })
>+                    .Test());
>+    EXPECT_FALSE(copy_tester.WithInvariants(strong_guarantee).Test());
>+    // move
>+    auto move_tester = MakeExceptionSafetyTester()
>+                           .WithInitialValue(WithCopyNoThrow())
>+                           .WithOperation([](ThrowingVariant* lhs) {
>+                             *lhs = ExpectedThrower(testing::nothrow_ctor);
>+                           });
>+    EXPECT_TRUE(move_tester
>+                    .WithInvariants(CheckInvariants,
>+                                    [](ThrowingVariant* lhs) {
>+                                      return lhs->valueless_by_exception();
>+                                    })
>+                    .Test());
>+    EXPECT_FALSE(move_tester.WithInvariants(strong_guarantee).Test());
>+  }
>+  // Otherwise (if is_nothrow_constructible_v<Tj, T> == false &&
>+  // is_nothrow_move_constructible<Tj> == true),
>+  // equivalent to operator=(variant(std::forward<T>(t)))
>+  // This should have strong guarantee because it creates a temporary variant
>+  // and operator=(variant&&) invokes Tj's move ctor which doesn't throw.
>+  // libstdc++ std::variant has bugs on conversion assignment regarding
>+  // exception safety.
>+#if !(defined(ABSL_HAVE_STD_VARIANT) && defined(__GLIBCXX__))
>+  {
>+    MoveNothrow rhs;
>+    EXPECT_TRUE(MakeExceptionSafetyTester()
>+                    .WithInitialValue(WithThrower())
>+                    .WithInvariants(CheckInvariants, strong_guarantee)
>+                    .Test([&rhs](ThrowingVariant* lhs) { *lhs = rhs; }));
>+  }
>+#endif  // !(defined(ABSL_HAVE_STD_VARIANT) && defined(__GLIBCXX__))
>+}
>+
>+TEST(VariantExceptionSafetyTest, Emplace) {
>+  // If an exception during the initialization of the contained value, the
>+  // variant might not hold a value. The standard requires emplace() to provide
>+  // only basic guarantee.
>+  {
>+    Thrower args = ExpectedThrower();
>+    auto tester = MakeExceptionSafetyTester()
>+                      .WithInitialValue(WithThrower())
>+                      .WithOperation([&args](ThrowingVariant* v) {
>+                        v->emplace<Thrower>(args);
>+                      });
>+    EXPECT_TRUE(tester
>+                    .WithInvariants(CheckInvariants,
>+                                    [](ThrowingVariant* v) {
>+                                      return v->valueless_by_exception();
>+                                    })
>+                    .Test());
>+    EXPECT_FALSE(tester.WithInvariants(strong_guarantee).Test());
>+  }
>+}
>+
>+TEST(VariantExceptionSafetyTest, Swap) {
>+  // if both are valueless_by_exception(), no effect
>+  {
>+    ThrowingVariant rhs = ValuelessByException();
>+    ThrowingVariant lhs = ValuelessByException();
>+    EXPECT_TRUE(TestNothrowOp([&]() { lhs.swap(rhs); }));
>+  }
>+  // if index() == rhs.index(), calls swap(get<i>(*this), get<i>(rhs))
>+  // where i is index().
>+  {
>+    ThrowingVariant rhs = ExpectedThrower();
>+    EXPECT_TRUE(MakeExceptionSafetyTester()
>+                    .WithInitialValue(WithThrower())
>+                    .WithInvariants(CheckInvariants)
>+                    .Test([&](ThrowingVariant* lhs) {
>+                      auto copy = rhs;
>+                      lhs->swap(copy);
>+                    }));
>+  }
>+  // Otherwise, exchanges the value of rhs and *this. The exception safety
>+  // involves variant in moved-from state which is not specified in the
>+  // standard, and since swap is 3-step it's impossible for it to provide a
>+  // overall strong guarantee. So, we are only checking basic guarantee here.
>+  {
>+    ThrowingVariant rhs = ExpectedThrower();
>+    EXPECT_TRUE(MakeExceptionSafetyTester()
>+                    .WithInitialValue(WithCopyNoThrow())
>+                    .WithInvariants(CheckInvariants)
>+                    .Test([&](ThrowingVariant* lhs) {
>+                      auto copy = rhs;
>+                      lhs->swap(copy);
>+                    }));
>+  }
>+  {
>+    ThrowingVariant rhs = ExpectedThrower();
>+    EXPECT_TRUE(MakeExceptionSafetyTester()
>+                    .WithInitialValue(WithCopyNoThrow())
>+                    .WithInvariants(CheckInvariants)
>+                    .Test([&](ThrowingVariant* lhs) {
>+                      auto copy = rhs;
>+                      copy.swap(*lhs);
>+                    }));
>+  }
>+}
>+
>+}  // namespace
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/variant_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/variant_test.cc
>new file mode 100644
>index 00000000000..262bd9446c1
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/types/variant_test.cc
>@@ -0,0 +1,2612 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+// Unit tests for the variant template. The 'is' and 'IsEmpty' methods
>+// of variant are not explicitly tested because they are used repeatedly
>+// in building other tests. All other public variant methods should have
>+// explicit tests.
>+
>+#include "absl/types/variant.h"
>+
>+#include <algorithm>
>+#include <cstddef>
>+#include <functional>
>+#include <initializer_list>
>+#include <memory>
>+#include <ostream>
>+#include <queue>
>+#include <type_traits>
>+#include <unordered_set>
>+#include <utility>
>+#include <vector>
>+
>+#include "gmock/gmock.h"
>+#include "gtest/gtest.h"
>+#include "absl/base/config.h"
>+#include "absl/base/port.h"
>+#include "absl/memory/memory.h"
>+#include "absl/meta/type_traits.h"
>+#include "absl/strings/string_view.h"
>+
>+#ifdef ABSL_HAVE_EXCEPTIONS
>+
>+#define ABSL_VARIANT_TEST_EXPECT_FAIL(expr, exception_t, text) \
>+  EXPECT_THROW(expr, exception_t)
>+
>+#else
>+
>+#define ABSL_VARIANT_TEST_EXPECT_FAIL(expr, exception_t, text) \
>+  EXPECT_DEATH(expr, text)
>+
>+#endif  // ABSL_HAVE_EXCEPTIONS
>+
>+#define ABSL_VARIANT_TEST_EXPECT_BAD_VARIANT_ACCESS(...)                 \
>+  ABSL_VARIANT_TEST_EXPECT_FAIL((__VA_ARGS__), absl::bad_variant_access, \
>+                                "Bad variant access")
>+
>+struct Hashable {};
>+
>+namespace std {
>+template <>
>+struct hash<Hashable> {
>+  size_t operator()(const Hashable&);
>+};
>+}  // namespace std
>+
>+struct NonHashable {};
>+
>+namespace absl {
>+namespace {
>+
>+using ::testing::DoubleEq;
>+using ::testing::Pointee;
>+using ::testing::VariantWith;
>+
>+struct MoveCanThrow {
>+  MoveCanThrow() : v(0) {}
>+  MoveCanThrow(int v) : v(v) {}  // NOLINT(runtime/explicit)
>+  MoveCanThrow(const MoveCanThrow& other) : v(other.v) {}
>+  MoveCanThrow& operator=(const MoveCanThrow& /*other*/) { return *this; }
>+  int v;
>+};
>+
>+bool operator==(MoveCanThrow lhs, MoveCanThrow rhs) { return lhs.v == rhs.v; }
>+bool operator!=(MoveCanThrow lhs, MoveCanThrow rhs) { return lhs.v != rhs.v; }
>+bool operator<(MoveCanThrow lhs, MoveCanThrow rhs) { return lhs.v < rhs.v; }
>+bool operator<=(MoveCanThrow lhs, MoveCanThrow rhs) { return lhs.v <= rhs.v; }
>+bool operator>=(MoveCanThrow lhs, MoveCanThrow rhs) { return lhs.v >= rhs.v; }
>+bool operator>(MoveCanThrow lhs, MoveCanThrow rhs) { return lhs.v > rhs.v; }
>+
>+// This helper class allows us to determine if it was swapped with std::swap()
>+// or with its friend swap() function.
>+struct SpecialSwap {
>+  explicit SpecialSwap(int i) : i(i) {}
>+  friend void swap(SpecialSwap& a, SpecialSwap& b) {
>+    a.special_swap = b.special_swap = true;
>+    std::swap(a.i, b.i);
>+  }
>+  bool operator==(SpecialSwap other) const { return i == other.i; }
>+  int i;
>+  bool special_swap = false;
>+};
>+
>+struct MoveOnlyWithListConstructor {
>+  MoveOnlyWithListConstructor() = default;
>+  explicit MoveOnlyWithListConstructor(std::initializer_list<int> /*ilist*/,
>+                                       int value)
>+      : value(value) {}
>+  MoveOnlyWithListConstructor(MoveOnlyWithListConstructor&&) = default;
>+  MoveOnlyWithListConstructor& operator=(MoveOnlyWithListConstructor&&) =
>+      default;
>+
>+  int value = 0;
>+};
>+
>+#ifdef ABSL_HAVE_EXCEPTIONS
>+
>+struct ConversionException {};
>+
>+template <class T>
>+struct ExceptionOnConversion {
>+  operator T() const {  // NOLINT(runtime/explicit)
>+    throw ConversionException();
>+  }
>+};
>+
>+// Forces a variant into the valueless by exception state.
>+template <class H, class... T>
>+void ToValuelessByException(absl::variant<H, T...>& v) {  // NOLINT
>+  try {
>+    v.template emplace<0>(ExceptionOnConversion<H>());
>+  } catch (ConversionException& /*e*/) {
>+    // This space intentionally left blank.
>+  }
>+}
>+
>+#endif  // ABSL_HAVE_EXCEPTIONS
>+
>+// An indexed sequence of distinct structures holding a single
>+// value of type T
>+template<typename T, size_t N>
>+struct ValueHolder {
>+  explicit ValueHolder(const T& x) : value(x) {}
>+  typedef T value_type;
>+  value_type value;
>+  static const size_t kIndex = N;
>+};
>+template<typename T, size_t N>
>+const size_t ValueHolder<T, N>::kIndex;
>+
>+// The following three functions make ValueHolder compatible with
>+// EXPECT_EQ and EXPECT_NE
>+template<typename T, size_t N>
>+inline bool operator==(const ValueHolder<T, N>& left,
>+                       const ValueHolder<T, N>& right) {
>+  return left.value == right.value;
>+}
>+
>+template<typename T, size_t N>
>+inline bool operator!=(const ValueHolder<T, N>& left,
>+                       const ValueHolder<T, N>& right) {
>+  return left.value != right.value;
>+}
>+
>+template<typename T, size_t N>
>+inline std::ostream& operator<<(
>+    std::ostream& stream, const ValueHolder<T, N>& object) {
>+  return stream << object.value;
>+}
>+
>+// Makes a variant holding twelve uniquely typed T wrappers.
>+template<typename T>
>+struct VariantFactory {
>+  typedef variant<ValueHolder<T, 1>, ValueHolder<T, 2>, ValueHolder<T, 3>,
>+                  ValueHolder<T, 4>>
>+      Type;
>+};
>+
>+// A typelist in 1:1 with VariantFactory, to use type driven unit tests.
>+typedef ::testing::Types<ValueHolder<size_t, 1>, ValueHolder<size_t, 2>,
>+                         ValueHolder<size_t, 3>,
>+                         ValueHolder<size_t, 4>> VariantTypes;
>+
>+// Increments the provided counter pointer in the destructor
>+struct IncrementInDtor {
>+  explicit IncrementInDtor(int* counter) : counter(counter) {}
>+  ~IncrementInDtor() { *counter += 1; }
>+  int* counter;
>+};
>+
>+struct IncrementInDtorCopyCanThrow {
>+  explicit IncrementInDtorCopyCanThrow(int* counter) : counter(counter) {}
>+  IncrementInDtorCopyCanThrow(IncrementInDtorCopyCanThrow&& other) noexcept =
>+      default;
>+  IncrementInDtorCopyCanThrow(const IncrementInDtorCopyCanThrow& other)
>+      : counter(other.counter) {}
>+  IncrementInDtorCopyCanThrow& operator=(
>+      IncrementInDtorCopyCanThrow&&) noexcept = default;
>+  IncrementInDtorCopyCanThrow& operator=(
>+      IncrementInDtorCopyCanThrow const& other) {
>+    counter = other.counter;
>+    return *this;
>+  }
>+  ~IncrementInDtorCopyCanThrow() { *counter += 1; }
>+  int* counter;
>+};
>+
>+// This is defined so operator== for ValueHolder<IncrementInDtor> will
>+// return true if two IncrementInDtor objects increment the same
>+// counter
>+inline bool operator==(const IncrementInDtor& left,
>+                       const IncrementInDtor& right) {
>+  return left.counter == right.counter;
>+}
>+
>+// This is defined so EXPECT_EQ can work with IncrementInDtor
>+inline std::ostream& operator<<(
>+    std::ostream& stream, const IncrementInDtor& object) {
>+  return stream << object.counter;
>+}
>+
>+// A class that can be copied, but not assigned.
>+class CopyNoAssign {
>+ public:
>+  explicit CopyNoAssign(int value) : foo(value) {}
>+  CopyNoAssign(const CopyNoAssign& other) : foo(other.foo) {}
>+  int foo;
>+ private:
>+  const CopyNoAssign& operator=(const CopyNoAssign&);
>+};
>+
>+// A class that can neither be copied nor assigned. We provide
>+// overloads for the constructor with up to four parameters so we can
>+// test the overloads of variant::emplace.
>+class NonCopyable {
>+ public:
>+  NonCopyable()
>+      : value(0) {}
>+  explicit NonCopyable(int value1)
>+      : value(value1) {}
>+
>+  NonCopyable(int value1, int value2)
>+      : value(value1 + value2) {}
>+
>+  NonCopyable(int value1, int value2, int value3)
>+      : value(value1 + value2 + value3) {}
>+
>+  NonCopyable(int value1, int value2, int value3, int value4)
>+      : value(value1 + value2 + value3 + value4) {}
>+  NonCopyable(const NonCopyable&) = delete;
>+  NonCopyable& operator=(const NonCopyable&) = delete;
>+  int value;
>+};
>+
>+// A typed test and typed test case over the VariantTypes typelist,
>+// from which we derive a number of tests that will execute for one of
>+// each type.
>+template <typename T>
>+class VariantTypesTest : public ::testing::Test {};
>+TYPED_TEST_CASE(VariantTypesTest, VariantTypes);
>+
>+////////////////////
>+// [variant.ctor] //
>+////////////////////
>+
>+struct NonNoexceptDefaultConstructible {
>+  NonNoexceptDefaultConstructible() {}
>+  int value = 5;
>+};
>+
>+struct NonDefaultConstructible {
>+  NonDefaultConstructible() = delete;
>+};
>+
>+TEST(VariantTest, TestDefaultConstructor) {
>+  {
>+    using X = variant<int>;
>+    constexpr variant<int> x{};
>+    ASSERT_FALSE(x.valueless_by_exception());
>+    ASSERT_EQ(0, x.index());
>+    EXPECT_EQ(0, absl::get<0>(x));
>+    EXPECT_TRUE(std::is_nothrow_default_constructible<X>::value);
>+  }
>+
>+  {
>+    using X = variant<NonNoexceptDefaultConstructible>;
>+    X x{};
>+    ASSERT_FALSE(x.valueless_by_exception());
>+    ASSERT_EQ(0, x.index());
>+    EXPECT_EQ(5, absl::get<0>(x).value);
>+    EXPECT_FALSE(std::is_nothrow_default_constructible<X>::value);
>+  }
>+
>+  {
>+    using X = variant<int, NonNoexceptDefaultConstructible>;
>+    X x{};
>+    ASSERT_FALSE(x.valueless_by_exception());
>+    ASSERT_EQ(0, x.index());
>+    EXPECT_EQ(0, absl::get<0>(x));
>+    EXPECT_TRUE(std::is_nothrow_default_constructible<X>::value);
>+  }
>+
>+  {
>+    using X = variant<NonNoexceptDefaultConstructible, int>;
>+    X x{};
>+    ASSERT_FALSE(x.valueless_by_exception());
>+    ASSERT_EQ(0, x.index());
>+    EXPECT_EQ(5, absl::get<0>(x).value);
>+    EXPECT_FALSE(std::is_nothrow_default_constructible<X>::value);
>+  }
>+  EXPECT_FALSE(
>+      std::is_default_constructible<variant<NonDefaultConstructible>>::value);
>+  EXPECT_FALSE((std::is_default_constructible<
>+                variant<NonDefaultConstructible, int>>::value));
>+  EXPECT_TRUE((std::is_default_constructible<
>+               variant<int, NonDefaultConstructible>>::value));
>+}
>+
>+// Test that for each slot, copy constructing a variant with that type
>+// produces a sensible object that correctly reports its type, and
>+// that copies the provided value.
>+TYPED_TEST(VariantTypesTest, TestCopyCtor) {
>+  typedef typename VariantFactory<typename TypeParam::value_type>::Type Variant;
>+  using value_type1 = absl::variant_alternative_t<0, Variant>;
>+  using value_type2 = absl::variant_alternative_t<1, Variant>;
>+  using value_type3 = absl::variant_alternative_t<2, Variant>;
>+  using value_type4 = absl::variant_alternative_t<3, Variant>;
>+  const TypeParam value(TypeParam::kIndex);
>+  Variant original(value);
>+  Variant copied(original);
>+  EXPECT_TRUE(absl::holds_alternative<value_type1>(copied) ||
>+              TypeParam::kIndex != 1);
>+  EXPECT_TRUE(absl::holds_alternative<value_type2>(copied) ||
>+              TypeParam::kIndex != 2);
>+  EXPECT_TRUE(absl::holds_alternative<value_type3>(copied) ||
>+              TypeParam::kIndex != 3);
>+  EXPECT_TRUE(absl::holds_alternative<value_type4>(copied) ||
>+              TypeParam::kIndex != 4);
>+  EXPECT_TRUE((absl::get_if<value_type1>(&original) ==
>+               absl::get_if<value_type1>(&copied)) ||
>+              TypeParam::kIndex == 1);
>+  EXPECT_TRUE((absl::get_if<value_type2>(&original) ==
>+               absl::get_if<value_type2>(&copied)) ||
>+              TypeParam::kIndex == 2);
>+  EXPECT_TRUE((absl::get_if<value_type3>(&original) ==
>+               absl::get_if<value_type3>(&copied)) ||
>+              TypeParam::kIndex == 3);
>+  EXPECT_TRUE((absl::get_if<value_type4>(&original) ==
>+               absl::get_if<value_type4>(&copied)) ||
>+              TypeParam::kIndex == 4);
>+  EXPECT_TRUE((absl::get_if<value_type1>(&original) ==
>+               absl::get_if<value_type1>(&copied)) ||
>+              TypeParam::kIndex == 1);
>+  EXPECT_TRUE((absl::get_if<value_type2>(&original) ==
>+               absl::get_if<value_type2>(&copied)) ||
>+              TypeParam::kIndex == 2);
>+  EXPECT_TRUE((absl::get_if<value_type3>(&original) ==
>+               absl::get_if<value_type3>(&copied)) ||
>+              TypeParam::kIndex == 3);
>+  EXPECT_TRUE((absl::get_if<value_type4>(&original) ==
>+               absl::get_if<value_type4>(&copied)) ||
>+              TypeParam::kIndex == 4);
>+  const TypeParam* ovalptr = absl::get_if<TypeParam>(&original);
>+  const TypeParam* cvalptr = absl::get_if<TypeParam>(&copied);
>+  ASSERT_TRUE(ovalptr != nullptr);
>+  ASSERT_TRUE(cvalptr != nullptr);
>+  EXPECT_EQ(*ovalptr, *cvalptr);
>+  TypeParam* mutable_ovalptr = absl::get_if<TypeParam>(&original);
>+  TypeParam* mutable_cvalptr = absl::get_if<TypeParam>(&copied);
>+  ASSERT_TRUE(mutable_ovalptr != nullptr);
>+  ASSERT_TRUE(mutable_cvalptr != nullptr);
>+  EXPECT_EQ(*mutable_ovalptr, *mutable_cvalptr);
>+}
>+
>+template <class>
>+struct MoveOnly {
>+  MoveOnly() = default;
>+  explicit MoveOnly(int value) : value(value) {}
>+  MoveOnly(MoveOnly&&) = default;
>+  MoveOnly& operator=(MoveOnly&&) = default;
>+  int value = 5;
>+};
>+
>+TEST(VariantTest, TestMoveConstruct) {
>+  using V = variant<MoveOnly<class A>, MoveOnly<class B>, MoveOnly<class C>>;
>+
>+  V v(in_place_index_t<1>{}, 10);
>+  V v2 = absl::move(v);
>+  EXPECT_EQ(10, absl::get<1>(v2).value);
>+}
>+
>+// Used internally to emulate missing triviality traits for tests.
>+template <class T>
>+union SingleUnion {
>+  T member;
>+};
>+
>+// NOTE: These don't work with types that can't be union members.
>+//       They are just for testing.
>+template <class T>
>+struct is_trivially_move_constructible
>+    : std::is_move_constructible<SingleUnion<T>>::type {};
>+
>+template <class T>
>+struct is_trivially_move_assignable
>+    : std::is_move_assignable<SingleUnion<T>>::type {};
>+
>+TEST(VariantTest, NothrowMoveConstructible) {
>+  // Verify that variant is nothrow move constructible iff its template
>+  // arguments are.
>+  using U = std::unique_ptr<int>;
>+  struct E {
>+    E(E&&) {}
>+  };
>+  static_assert(std::is_nothrow_move_constructible<variant<U>>::value, "");
>+  static_assert(std::is_nothrow_move_constructible<variant<U, int>>::value, "");
>+  static_assert(!std::is_nothrow_move_constructible<variant<U, E>>::value, "");
>+}
>+
>+// Test that for each slot, constructing a variant with that type
>+// produces a sensible object that correctly reports its type, and
>+// that copies the provided value.
>+TYPED_TEST(VariantTypesTest, TestValueCtor) {
>+  typedef typename VariantFactory<typename TypeParam::value_type>::Type Variant;
>+  using value_type1 = absl::variant_alternative_t<0, Variant>;
>+  using value_type2 = absl::variant_alternative_t<1, Variant>;
>+  using value_type3 = absl::variant_alternative_t<2, Variant>;
>+  using value_type4 = absl::variant_alternative_t<3, Variant>;
>+  const TypeParam value(TypeParam::kIndex);
>+  Variant v(value);
>+  EXPECT_TRUE(absl::holds_alternative<value_type1>(v) ||
>+              TypeParam::kIndex != 1);
>+  EXPECT_TRUE(absl::holds_alternative<value_type2>(v) ||
>+              TypeParam::kIndex != 2);
>+  EXPECT_TRUE(absl::holds_alternative<value_type3>(v) ||
>+              TypeParam::kIndex != 3);
>+  EXPECT_TRUE(absl::holds_alternative<value_type4>(v) ||
>+              TypeParam::kIndex != 4);
>+  EXPECT_TRUE(nullptr != absl::get_if<value_type1>(&v) ||
>+              TypeParam::kIndex != 1);
>+  EXPECT_TRUE(nullptr != absl::get_if<value_type2>(&v) ||
>+              TypeParam::kIndex != 2);
>+  EXPECT_TRUE(nullptr != absl::get_if<value_type3>(&v) ||
>+              TypeParam::kIndex != 3);
>+  EXPECT_TRUE(nullptr != absl::get_if<value_type4>(&v) ||
>+              TypeParam::kIndex != 4);
>+  EXPECT_TRUE(nullptr != absl::get_if<value_type1>(&v) ||
>+              TypeParam::kIndex != 1);
>+  EXPECT_TRUE(nullptr != absl::get_if<value_type2>(&v) ||
>+              TypeParam::kIndex != 2);
>+  EXPECT_TRUE(nullptr != absl::get_if<value_type3>(&v) ||
>+              TypeParam::kIndex != 3);
>+  EXPECT_TRUE(nullptr != absl::get_if<value_type4>(&v) ||
>+              TypeParam::kIndex != 4);
>+  const TypeParam* valptr = absl::get_if<TypeParam>(&v);
>+  ASSERT_TRUE(nullptr != valptr);
>+  EXPECT_EQ(value.value, valptr->value);
>+  const TypeParam* mutable_valptr = absl::get_if<TypeParam>(&v);
>+  ASSERT_TRUE(nullptr != mutable_valptr);
>+  EXPECT_EQ(value.value, mutable_valptr->value);
>+}
>+
>+TEST(VariantTest, InPlaceType) {
>+  using Var = variant<int, std::string, NonCopyable, std::vector<int>>;
>+
>+  Var v1(in_place_type_t<int>(), 7);
>+  ASSERT_TRUE(absl::holds_alternative<int>(v1));
>+  EXPECT_EQ(7, absl::get<int>(v1));
>+
>+  Var v2(in_place_type_t<std::string>(), "ABC");
>+  ASSERT_TRUE(absl::holds_alternative<std::string>(v2));
>+  EXPECT_EQ("ABC", absl::get<std::string>(v2));
>+
>+  Var v3(in_place_type_t<std::string>(), "ABC", 2);
>+  ASSERT_TRUE(absl::holds_alternative<std::string>(v3));
>+  EXPECT_EQ("AB", absl::get<std::string>(v3));
>+
>+  Var v4(in_place_type_t<NonCopyable>{});
>+  ASSERT_TRUE(absl::holds_alternative<NonCopyable>(v4));
>+
>+  Var v5(in_place_type_t<std::vector<int>>(), {1, 2, 3});
>+  ASSERT_TRUE(absl::holds_alternative<std::vector<int>>(v5));
>+  EXPECT_THAT(absl::get<std::vector<int>>(v5), ::testing::ElementsAre(1, 2, 3));
>+}
>+
>+TEST(VariantTest, InPlaceTypeInitializerList) {
>+  using Var = variant<int, std::string, NonCopyable, MoveOnlyWithListConstructor>;
>+
>+  Var v1(in_place_type_t<MoveOnlyWithListConstructor>(), {1, 2, 3, 4, 5}, 6);
>+  ASSERT_TRUE(absl::holds_alternative<MoveOnlyWithListConstructor>(v1));
>+  EXPECT_EQ(6, absl::get<MoveOnlyWithListConstructor>(v1).value);
>+}
>+
>+TEST(VariantTest, InPlaceIndex) {
>+  using Var = variant<int, std::string, NonCopyable, std::vector<int>>;
>+
>+  Var v1(in_place_index_t<0>(), 7);
>+  ASSERT_TRUE(absl::holds_alternative<int>(v1));
>+  EXPECT_EQ(7, absl::get<int>(v1));
>+
>+  Var v2(in_place_index_t<1>(), "ABC");
>+  ASSERT_TRUE(absl::holds_alternative<std::string>(v2));
>+  EXPECT_EQ("ABC", absl::get<std::string>(v2));
>+
>+  Var v3(in_place_index_t<1>(), "ABC", 2);
>+  ASSERT_TRUE(absl::holds_alternative<std::string>(v3));
>+  EXPECT_EQ("AB", absl::get<std::string>(v3));
>+
>+  Var v4(in_place_index_t<2>{});
>+  EXPECT_TRUE(absl::holds_alternative<NonCopyable>(v4));
>+
>+  // Verify that a variant with only non-copyables can still be constructed.
>+  EXPECT_TRUE(absl::holds_alternative<NonCopyable>(
>+      variant<NonCopyable>(in_place_index_t<0>{})));
>+
>+  Var v5(in_place_index_t<3>(), {1, 2, 3});
>+  ASSERT_TRUE(absl::holds_alternative<std::vector<int>>(v5));
>+  EXPECT_THAT(absl::get<std::vector<int>>(v5), ::testing::ElementsAre(1, 2, 3));
>+}
>+
>+TEST(VariantTest, InPlaceIndexInitializerList) {
>+  using Var = variant<int, std::string, NonCopyable, MoveOnlyWithListConstructor>;
>+
>+  Var v1(in_place_index_t<3>(), {1, 2, 3, 4, 5}, 6);
>+  ASSERT_TRUE(absl::holds_alternative<MoveOnlyWithListConstructor>(v1));
>+  EXPECT_EQ(6, absl::get<MoveOnlyWithListConstructor>(v1).value);
>+}
>+
>+////////////////////
>+// [variant.dtor] //
>+////////////////////
>+
>+// Make sure that the destructor destroys the contained value
>+TEST(VariantTest, TestDtor) {
>+  typedef VariantFactory<IncrementInDtor>::Type Variant;
>+  using value_type1 = absl::variant_alternative_t<0, Variant>;
>+  using value_type2 = absl::variant_alternative_t<1, Variant>;
>+  using value_type3 = absl::variant_alternative_t<2, Variant>;
>+  using value_type4 = absl::variant_alternative_t<3, Variant>;
>+  int counter = 0;
>+  IncrementInDtor counter_adjuster(&counter);
>+  EXPECT_EQ(0, counter);
>+
>+  value_type1 value1(counter_adjuster);
>+  { Variant object(value1); }
>+  EXPECT_EQ(1, counter);
>+
>+  value_type2 value2(counter_adjuster);
>+  { Variant object(value2); }
>+  EXPECT_EQ(2, counter);
>+
>+  value_type3 value3(counter_adjuster);
>+  { Variant object(value3); }
>+  EXPECT_EQ(3, counter);
>+
>+  value_type4 value4(counter_adjuster);
>+  { Variant object(value4); }
>+  EXPECT_EQ(4, counter);
>+}
>+
>+#ifdef ABSL_HAVE_EXCEPTIONS
>+
>+// Test destruction when in the valueless_by_exception state.
>+TEST(VariantTest, TestDtorValuelessByException) {
>+  int counter = 0;
>+  IncrementInDtor counter_adjuster(&counter);
>+
>+  {
>+    using Variant = VariantFactory<IncrementInDtor>::Type;
>+
>+    Variant v(in_place_index_t<0>(), counter_adjuster);
>+    EXPECT_EQ(0, counter);
>+
>+    ToValuelessByException(v);
>+    ASSERT_TRUE(v.valueless_by_exception());
>+    EXPECT_EQ(1, counter);
>+  }
>+  EXPECT_EQ(1, counter);
>+}
>+
>+#endif  // ABSL_HAVE_EXCEPTIONS
>+
>+//////////////////////
>+// [variant.assign] //
>+//////////////////////
>+
>+// Test that self-assignment doesn't destroy the current value
>+TEST(VariantTest, TestSelfAssignment) {
>+  typedef VariantFactory<IncrementInDtor>::Type Variant;
>+  int counter = 0;
>+  IncrementInDtor counter_adjuster(&counter);
>+  absl::variant_alternative_t<0, Variant> value(counter_adjuster);
>+  Variant object(value);
>+  object.operator=(object);
>+  EXPECT_EQ(0, counter);
>+
>+  // A std::string long enough that it's likely to defeat any inline representation
>+  // optimization.
>+  const std::string long_str(128, 'a');
>+
>+  std::string foo = long_str;
>+  foo = *&foo;
>+  EXPECT_EQ(long_str, foo);
>+
>+  variant<int, std::string> so = long_str;
>+  ASSERT_EQ(1, so.index());
>+  EXPECT_EQ(long_str, absl::get<1>(so));
>+  so = *&so;
>+
>+  ASSERT_EQ(1, so.index());
>+  EXPECT_EQ(long_str, absl::get<1>(so));
>+}
>+
>+// Test that assigning a variant<..., T, ...> to a variant<..., T, ...> produces
>+// a variant<..., T, ...> with the correct value.
>+TYPED_TEST(VariantTypesTest, TestAssignmentCopiesValueSameTypes) {
>+  typedef typename VariantFactory<typename TypeParam::value_type>::Type Variant;
>+  const TypeParam value(TypeParam::kIndex);
>+  const Variant source(value);
>+  Variant target(TypeParam(value.value + 1));
>+  ASSERT_TRUE(absl::holds_alternative<TypeParam>(source));
>+  ASSERT_TRUE(absl::holds_alternative<TypeParam>(target));
>+  ASSERT_NE(absl::get<TypeParam>(source), absl::get<TypeParam>(target));
>+  target = source;
>+  ASSERT_TRUE(absl::holds_alternative<TypeParam>(source));
>+  ASSERT_TRUE(absl::holds_alternative<TypeParam>(target));
>+  EXPECT_EQ(absl::get<TypeParam>(source), absl::get<TypeParam>(target));
>+}
>+
>+// Test that assisnging a variant<..., T, ...> to a variant<1, ...>
>+// produces a variant<..., T, ...> with the correct value.
>+TYPED_TEST(VariantTypesTest, TestAssignmentCopiesValuesVaryingSourceType) {
>+  typedef typename VariantFactory<typename TypeParam::value_type>::Type Variant;
>+  using value_type1 = absl::variant_alternative_t<0, Variant>;
>+  const TypeParam value(TypeParam::kIndex);
>+  const Variant source(value);
>+  ASSERT_TRUE(absl::holds_alternative<TypeParam>(source));
>+  Variant target(value_type1(1));
>+  ASSERT_TRUE(absl::holds_alternative<value_type1>(target));
>+  target = source;
>+  EXPECT_TRUE(absl::holds_alternative<TypeParam>(source));
>+  EXPECT_TRUE(absl::holds_alternative<TypeParam>(target));
>+  EXPECT_EQ(absl::get<TypeParam>(source), absl::get<TypeParam>(target));
>+}
>+
>+// Test that assigning a variant<1, ...> to a variant<..., T, ...>
>+// produces a variant<1, ...> with the correct value.
>+TYPED_TEST(VariantTypesTest, TestAssignmentCopiesValuesVaryingTargetType) {
>+  typedef typename VariantFactory<typename TypeParam::value_type>::Type Variant;
>+  using value_type1 = absl::variant_alternative_t<0, Variant>;
>+  const Variant source(value_type1(1));
>+  ASSERT_TRUE(absl::holds_alternative<value_type1>(source));
>+  const TypeParam value(TypeParam::kIndex);
>+  Variant target(value);
>+  ASSERT_TRUE(absl::holds_alternative<TypeParam>(target));
>+  target = source;
>+  EXPECT_TRUE(absl::holds_alternative<value_type1>(target));
>+  EXPECT_TRUE(absl::holds_alternative<value_type1>(source));
>+  EXPECT_EQ(absl::get<value_type1>(source), absl::get<value_type1>(target));
>+}
>+
>+// Test that operator=<T> works, that assigning a new value destroys
>+// the old and that assigning the new value again does not redestroy
>+// the old
>+TEST(VariantTest, TestAssign) {
>+  typedef VariantFactory<IncrementInDtor>::Type Variant;
>+  using value_type1 = absl::variant_alternative_t<0, Variant>;
>+  using value_type2 = absl::variant_alternative_t<1, Variant>;
>+  using value_type3 = absl::variant_alternative_t<2, Variant>;
>+  using value_type4 = absl::variant_alternative_t<3, Variant>;
>+
>+  const int kSize = 4;
>+  int counter[kSize];
>+  std::unique_ptr<IncrementInDtor> counter_adjustor[kSize];
>+  for (int i = 0; i != kSize; i++) {
>+    counter[i] = 0;
>+    counter_adjustor[i] = absl::make_unique<IncrementInDtor>(&counter[i]);
>+  }
>+
>+  value_type1 v1(*counter_adjustor[0]);
>+  value_type2 v2(*counter_adjustor[1]);
>+  value_type3 v3(*counter_adjustor[2]);
>+  value_type4 v4(*counter_adjustor[3]);
>+
>+  // Test that reassignment causes destruction of old value
>+  {
>+    Variant object(v1);
>+    object = v2;
>+    object = v3;
>+    object = v4;
>+    object = v1;
>+  }
>+
>+  EXPECT_EQ(2, counter[0]);
>+  EXPECT_EQ(1, counter[1]);
>+  EXPECT_EQ(1, counter[2]);
>+  EXPECT_EQ(1, counter[3]);
>+
>+  std::fill(std::begin(counter), std::end(counter), 0);
>+
>+  // Test that self-assignment does not cause destruction of old value
>+  {
>+    Variant object(v1);
>+    object.operator=(object);
>+    EXPECT_EQ(0, counter[0]);
>+  }
>+  {
>+    Variant object(v2);
>+    object.operator=(object);
>+    EXPECT_EQ(0, counter[1]);
>+  }
>+  {
>+    Variant object(v3);
>+    object.operator=(object);
>+    EXPECT_EQ(0, counter[2]);
>+  }
>+  {
>+    Variant object(v4);
>+    object.operator=(object);
>+    EXPECT_EQ(0, counter[3]);
>+  }
>+
>+  EXPECT_EQ(1, counter[0]);
>+  EXPECT_EQ(1, counter[1]);
>+  EXPECT_EQ(1, counter[2]);
>+  EXPECT_EQ(1, counter[3]);
>+}
>+
>+// This tests that we perform a backup if the copy-assign can throw but the move
>+// cannot throw.
>+TEST(VariantTest, TestBackupAssign) {
>+  typedef VariantFactory<IncrementInDtorCopyCanThrow>::Type Variant;
>+  using value_type1 = absl::variant_alternative_t<0, Variant>;
>+  using value_type2 = absl::variant_alternative_t<1, Variant>;
>+  using value_type3 = absl::variant_alternative_t<2, Variant>;
>+  using value_type4 = absl::variant_alternative_t<3, Variant>;
>+
>+  const int kSize = 4;
>+  int counter[kSize];
>+  std::unique_ptr<IncrementInDtorCopyCanThrow> counter_adjustor[kSize];
>+  for (int i = 0; i != kSize; i++) {
>+    counter[i] = 0;
>+    counter_adjustor[i].reset(new IncrementInDtorCopyCanThrow(&counter[i]));
>+  }
>+
>+  value_type1 v1(*counter_adjustor[0]);
>+  value_type2 v2(*counter_adjustor[1]);
>+  value_type3 v3(*counter_adjustor[2]);
>+  value_type4 v4(*counter_adjustor[3]);
>+
>+  // Test that reassignment causes destruction of old value
>+  {
>+    Variant object(v1);
>+    object = v2;
>+    object = v3;
>+    object = v4;
>+    object = v1;
>+  }
>+
>+  // libstdc++ doesn't pass this test
>+#if !(defined(ABSL_HAVE_STD_VARIANT) && defined(__GLIBCXX__))
>+  EXPECT_EQ(3, counter[0]);
>+  EXPECT_EQ(2, counter[1]);
>+  EXPECT_EQ(2, counter[2]);
>+  EXPECT_EQ(2, counter[3]);
>+#endif
>+
>+  std::fill(std::begin(counter), std::end(counter), 0);
>+
>+  // Test that self-assignment does not cause destruction of old value
>+  {
>+    Variant object(v1);
>+    object.operator=(object);
>+    EXPECT_EQ(0, counter[0]);
>+  }
>+  {
>+    Variant object(v2);
>+    object.operator=(object);
>+    EXPECT_EQ(0, counter[1]);
>+  }
>+  {
>+    Variant object(v3);
>+    object.operator=(object);
>+    EXPECT_EQ(0, counter[2]);
>+  }
>+  {
>+    Variant object(v4);
>+    object.operator=(object);
>+    EXPECT_EQ(0, counter[3]);
>+  }
>+
>+  EXPECT_EQ(1, counter[0]);
>+  EXPECT_EQ(1, counter[1]);
>+  EXPECT_EQ(1, counter[2]);
>+  EXPECT_EQ(1, counter[3]);
>+}
>+
>+///////////////////
>+// [variant.mod] //
>+///////////////////
>+
>+TEST(VariantTest, TestEmplaceBasic) {
>+  using Variant = variant<int, char>;
>+
>+  Variant v(absl::in_place_index_t<0>{}, 0);
>+
>+  {
>+    char& emplace_result = v.emplace<char>();
>+    ASSERT_TRUE(absl::holds_alternative<char>(v));
>+    EXPECT_EQ(absl::get<char>(v), 0);
>+    EXPECT_EQ(&emplace_result, &absl::get<char>(v));
>+  }
>+
>+  // Make sure that another emplace does zero-initialization
>+  absl::get<char>(v) = 'a';
>+  v.emplace<char>('b');
>+  ASSERT_TRUE(absl::holds_alternative<char>(v));
>+  EXPECT_EQ(absl::get<char>(v), 'b');
>+
>+  {
>+    int& emplace_result = v.emplace<int>();
>+    EXPECT_TRUE(absl::holds_alternative<int>(v));
>+    EXPECT_EQ(absl::get<int>(v), 0);
>+    EXPECT_EQ(&emplace_result, &absl::get<int>(v));
>+  }
>+}
>+
>+TEST(VariantTest, TestEmplaceInitializerList) {
>+  using Var = variant<int, std::string, NonCopyable, MoveOnlyWithListConstructor>;
>+
>+  Var v1(absl::in_place_index_t<0>{}, 555);
>+  MoveOnlyWithListConstructor& emplace_result =
>+      v1.emplace<MoveOnlyWithListConstructor>({1, 2, 3, 4, 5}, 6);
>+  ASSERT_TRUE(absl::holds_alternative<MoveOnlyWithListConstructor>(v1));
>+  EXPECT_EQ(6, absl::get<MoveOnlyWithListConstructor>(v1).value);
>+  EXPECT_EQ(&emplace_result, &absl::get<MoveOnlyWithListConstructor>(v1));
>+}
>+
>+TEST(VariantTest, TestEmplaceIndex) {
>+  using Variant = variant<int, char>;
>+
>+  Variant v(absl::in_place_index_t<0>{}, 555);
>+
>+  {
>+    char& emplace_result = v.emplace<1>();
>+    ASSERT_TRUE(absl::holds_alternative<char>(v));
>+    EXPECT_EQ(absl::get<char>(v), 0);
>+    EXPECT_EQ(&emplace_result, &absl::get<char>(v));
>+  }
>+
>+  // Make sure that another emplace does zero-initialization
>+  absl::get<char>(v) = 'a';
>+  v.emplace<1>('b');
>+  ASSERT_TRUE(absl::holds_alternative<char>(v));
>+  EXPECT_EQ(absl::get<char>(v), 'b');
>+
>+  {
>+    int& emplace_result = v.emplace<0>();
>+    EXPECT_TRUE(absl::holds_alternative<int>(v));
>+    EXPECT_EQ(absl::get<int>(v), 0);
>+    EXPECT_EQ(&emplace_result, &absl::get<int>(v));
>+  }
>+}
>+
>+TEST(VariantTest, TestEmplaceIndexInitializerList) {
>+  using Var = variant<int, std::string, NonCopyable, MoveOnlyWithListConstructor>;
>+
>+  Var v1(absl::in_place_index_t<0>{}, 555);
>+  MoveOnlyWithListConstructor& emplace_result =
>+      v1.emplace<3>({1, 2, 3, 4, 5}, 6);
>+  ASSERT_TRUE(absl::holds_alternative<MoveOnlyWithListConstructor>(v1));
>+  EXPECT_EQ(6, absl::get<MoveOnlyWithListConstructor>(v1).value);
>+  EXPECT_EQ(&emplace_result, &absl::get<MoveOnlyWithListConstructor>(v1));
>+}
>+
>+//////////////////////
>+// [variant.status] //
>+//////////////////////
>+
>+TEST(VariantTest, Index) {
>+  using Var = variant<int, std::string, double>;
>+
>+  Var v = 1;
>+  EXPECT_EQ(0, v.index());
>+  v = "str";
>+  EXPECT_EQ(1, v.index());
>+  v = 0.;
>+  EXPECT_EQ(2, v.index());
>+
>+  Var v2 = v;
>+  EXPECT_EQ(2, v2.index());
>+  v2.emplace<int>(3);
>+  EXPECT_EQ(0, v2.index());
>+}
>+
>+TEST(VariantTest, NotValuelessByException) {
>+  using Var = variant<int, std::string, double>;
>+
>+  Var v = 1;
>+  EXPECT_FALSE(v.valueless_by_exception());
>+  v = "str";
>+  EXPECT_FALSE(v.valueless_by_exception());
>+  v = 0.;
>+  EXPECT_FALSE(v.valueless_by_exception());
>+
>+  Var v2 = v;
>+  EXPECT_FALSE(v.valueless_by_exception());
>+  v2.emplace<int>(3);
>+  EXPECT_FALSE(v.valueless_by_exception());
>+}
>+
>+#ifdef ABSL_HAVE_EXCEPTIONS
>+
>+TEST(VariantTest, IndexValuelessByException) {
>+  using Var = variant<MoveCanThrow, std::string, double>;
>+
>+  Var v(absl::in_place_index_t<0>{});
>+  EXPECT_EQ(0, v.index());
>+  ToValuelessByException(v);
>+  EXPECT_EQ(absl::variant_npos, v.index());
>+  v = "str";
>+  EXPECT_EQ(1, v.index());
>+}
>+
>+TEST(VariantTest, ValuelessByException) {
>+  using Var = variant<MoveCanThrow, std::string, double>;
>+
>+  Var v(absl::in_place_index_t<0>{});
>+  EXPECT_FALSE(v.valueless_by_exception());
>+  ToValuelessByException(v);
>+  EXPECT_TRUE(v.valueless_by_exception());
>+  v = "str";
>+  EXPECT_FALSE(v.valueless_by_exception());
>+}
>+
>+#endif  // ABSL_HAVE_EXCEPTIONS
>+
>+////////////////////
>+// [variant.swap] //
>+////////////////////
>+
>+TEST(VariantTest, MemberSwap) {
>+  SpecialSwap v1(3);
>+  SpecialSwap v2(7);
>+
>+  variant<SpecialSwap> a = v1, b = v2;
>+
>+  EXPECT_THAT(a, VariantWith<SpecialSwap>(v1));
>+  EXPECT_THAT(b, VariantWith<SpecialSwap>(v2));
>+
>+  a.swap(b);
>+  EXPECT_THAT(a, VariantWith<SpecialSwap>(v2));
>+  EXPECT_THAT(b, VariantWith<SpecialSwap>(v1));
>+  EXPECT_TRUE(absl::get<SpecialSwap>(a).special_swap);
>+
>+  using V = variant<MoveCanThrow, std::string, int>;
>+  int i = 33;
>+  std::string s = "abc";
>+  V valueless(in_place_index_t<0>{});
>+  ToValuelessByException(valueless);
>+  {
>+    // lhs and rhs holds different alternative
>+    V lhs(i), rhs(s);
>+    lhs.swap(rhs);
>+    EXPECT_THAT(lhs, VariantWith<std::string>(s));
>+    EXPECT_THAT(rhs, VariantWith<int>(i));
>+  }
>+  {
>+    // lhs is valueless
>+    V lhs(valueless), rhs(i);
>+    lhs.swap(rhs);
>+    EXPECT_THAT(lhs, VariantWith<int>(i));
>+    EXPECT_TRUE(rhs.valueless_by_exception());
>+  }
>+  {
>+    // rhs is valueless
>+    V lhs(s), rhs(valueless);
>+    lhs.swap(rhs);
>+    EXPECT_THAT(rhs, VariantWith<std::string>(s));
>+    EXPECT_TRUE(lhs.valueless_by_exception());
>+  }
>+  {
>+    // both are valueless
>+    V lhs(valueless), rhs(valueless);
>+    lhs.swap(rhs);
>+    EXPECT_TRUE(lhs.valueless_by_exception());
>+    EXPECT_TRUE(rhs.valueless_by_exception());
>+  }
>+}
>+
>+//////////////////////
>+// [variant.helper] //
>+//////////////////////
>+
>+TEST(VariantTest, VariantSize) {
>+  {
>+    using Size1Variant = absl::variant<int>;
>+    EXPECT_EQ(1, absl::variant_size<Size1Variant>::value);
>+    EXPECT_EQ(1, absl::variant_size<const Size1Variant>::value);
>+    EXPECT_EQ(1, absl::variant_size<volatile Size1Variant>::value);
>+    EXPECT_EQ(1, absl::variant_size<const volatile Size1Variant>::value);
>+  }
>+
>+  {
>+    using Size3Variant = absl::variant<int, float, int>;
>+    EXPECT_EQ(3, absl::variant_size<Size3Variant>::value);
>+    EXPECT_EQ(3, absl::variant_size<const Size3Variant>::value);
>+    EXPECT_EQ(3, absl::variant_size<volatile Size3Variant>::value);
>+    EXPECT_EQ(3, absl::variant_size<const volatile Size3Variant>::value);
>+  }
>+}
>+
>+TEST(VariantTest, VariantAlternative) {
>+  {
>+    using V = absl::variant<float, int, const char*>;
>+    EXPECT_TRUE(
>+        (std::is_same<float, absl::variant_alternative_t<0, V>>::value));
>+    EXPECT_TRUE((std::is_same<const float,
>+                              absl::variant_alternative_t<0, const V>>::value));
>+    EXPECT_TRUE(
>+        (std::is_same<volatile float,
>+                      absl::variant_alternative_t<0, volatile V>>::value));
>+    EXPECT_TRUE((
>+        std::is_same<const volatile float,
>+                     absl::variant_alternative_t<0, const volatile V>>::value));
>+
>+    EXPECT_TRUE((std::is_same<int, absl::variant_alternative_t<1, V>>::value));
>+    EXPECT_TRUE((std::is_same<const int,
>+                              absl::variant_alternative_t<1, const V>>::value));
>+    EXPECT_TRUE(
>+        (std::is_same<volatile int,
>+                      absl::variant_alternative_t<1, volatile V>>::value));
>+    EXPECT_TRUE((
>+        std::is_same<const volatile int,
>+                     absl::variant_alternative_t<1, const volatile V>>::value));
>+
>+    EXPECT_TRUE(
>+        (std::is_same<const char*, absl::variant_alternative_t<2, V>>::value));
>+    EXPECT_TRUE((std::is_same<const char* const,
>+                              absl::variant_alternative_t<2, const V>>::value));
>+    EXPECT_TRUE(
>+        (std::is_same<const char* volatile,
>+                      absl::variant_alternative_t<2, volatile V>>::value));
>+    EXPECT_TRUE((
>+        std::is_same<const char* const volatile,
>+                     absl::variant_alternative_t<2, const volatile V>>::value));
>+  }
>+
>+  {
>+    using V = absl::variant<float, volatile int, const char*>;
>+    EXPECT_TRUE(
>+        (std::is_same<float, absl::variant_alternative_t<0, V>>::value));
>+    EXPECT_TRUE((std::is_same<const float,
>+                              absl::variant_alternative_t<0, const V>>::value));
>+    EXPECT_TRUE(
>+        (std::is_same<volatile float,
>+                      absl::variant_alternative_t<0, volatile V>>::value));
>+    EXPECT_TRUE((
>+        std::is_same<const volatile float,
>+                     absl::variant_alternative_t<0, const volatile V>>::value));
>+
>+    EXPECT_TRUE(
>+        (std::is_same<volatile int, absl::variant_alternative_t<1, V>>::value));
>+    EXPECT_TRUE((std::is_same<const volatile int,
>+                              absl::variant_alternative_t<1, const V>>::value));
>+    EXPECT_TRUE(
>+        (std::is_same<volatile int,
>+                      absl::variant_alternative_t<1, volatile V>>::value));
>+    EXPECT_TRUE((
>+        std::is_same<const volatile int,
>+                     absl::variant_alternative_t<1, const volatile V>>::value));
>+
>+    EXPECT_TRUE(
>+        (std::is_same<const char*, absl::variant_alternative_t<2, V>>::value));
>+    EXPECT_TRUE((std::is_same<const char* const,
>+                              absl::variant_alternative_t<2, const V>>::value));
>+    EXPECT_TRUE(
>+        (std::is_same<const char* volatile,
>+                      absl::variant_alternative_t<2, volatile V>>::value));
>+    EXPECT_TRUE((
>+        std::is_same<const char* const volatile,
>+                     absl::variant_alternative_t<2, const volatile V>>::value));
>+  }
>+}
>+
>+///////////////////
>+// [variant.get] //
>+///////////////////
>+
>+TEST(VariantTest, HoldsAlternative) {
>+  using Var = variant<int, std::string, double>;
>+
>+  Var v = 1;
>+  EXPECT_TRUE(absl::holds_alternative<int>(v));
>+  EXPECT_FALSE(absl::holds_alternative<std::string>(v));
>+  EXPECT_FALSE(absl::holds_alternative<double>(v));
>+  v = "str";
>+  EXPECT_FALSE(absl::holds_alternative<int>(v));
>+  EXPECT_TRUE(absl::holds_alternative<std::string>(v));
>+  EXPECT_FALSE(absl::holds_alternative<double>(v));
>+  v = 0.;
>+  EXPECT_FALSE(absl::holds_alternative<int>(v));
>+  EXPECT_FALSE(absl::holds_alternative<std::string>(v));
>+  EXPECT_TRUE(absl::holds_alternative<double>(v));
>+
>+  Var v2 = v;
>+  EXPECT_FALSE(absl::holds_alternative<int>(v2));
>+  EXPECT_FALSE(absl::holds_alternative<std::string>(v2));
>+  EXPECT_TRUE(absl::holds_alternative<double>(v2));
>+  v2.emplace<int>(3);
>+  EXPECT_TRUE(absl::holds_alternative<int>(v2));
>+  EXPECT_FALSE(absl::holds_alternative<std::string>(v2));
>+  EXPECT_FALSE(absl::holds_alternative<double>(v2));
>+}
>+
>+TEST(VariantTest, GetIndex) {
>+  using Var = variant<int, std::string, double, int>;
>+
>+  {
>+    Var v(absl::in_place_index_t<0>{}, 0);
>+
>+    using LValueGetType = decltype(absl::get<0>(v));
>+    using RValueGetType = decltype(absl::get<0>(absl::move(v)));
>+
>+    EXPECT_TRUE((std::is_same<LValueGetType, int&>::value));
>+    EXPECT_TRUE((std::is_same<RValueGetType, int&&>::value));
>+    EXPECT_EQ(absl::get<0>(v), 0);
>+    EXPECT_EQ(absl::get<0>(absl::move(v)), 0);
>+
>+    const Var& const_v = v;
>+    using ConstLValueGetType = decltype(absl::get<0>(const_v));
>+    using ConstRValueGetType = decltype(absl::get<0>(absl::move(const_v)));
>+    EXPECT_TRUE((std::is_same<ConstLValueGetType, const int&>::value));
>+    EXPECT_TRUE((std::is_same<ConstRValueGetType, const int&&>::value));
>+    EXPECT_EQ(absl::get<0>(const_v), 0);
>+    EXPECT_EQ(absl::get<0>(absl::move(const_v)), 0);
>+  }
>+
>+  {
>+    Var v = std::string("Hello");
>+
>+    using LValueGetType = decltype(absl::get<1>(v));
>+    using RValueGetType = decltype(absl::get<1>(absl::move(v)));
>+
>+    EXPECT_TRUE((std::is_same<LValueGetType, std::string&>::value));
>+    EXPECT_TRUE((std::is_same<RValueGetType, std::string&&>::value));
>+    EXPECT_EQ(absl::get<1>(v), "Hello");
>+    EXPECT_EQ(absl::get<1>(absl::move(v)), "Hello");
>+
>+    const Var& const_v = v;
>+    using ConstLValueGetType = decltype(absl::get<1>(const_v));
>+    using ConstRValueGetType = decltype(absl::get<1>(absl::move(const_v)));
>+    EXPECT_TRUE((std::is_same<ConstLValueGetType, const std::string&>::value));
>+    EXPECT_TRUE((std::is_same<ConstRValueGetType, const std::string&&>::value));
>+    EXPECT_EQ(absl::get<1>(const_v), "Hello");
>+    EXPECT_EQ(absl::get<1>(absl::move(const_v)), "Hello");
>+  }
>+
>+  {
>+    Var v = 2.0;
>+
>+    using LValueGetType = decltype(absl::get<2>(v));
>+    using RValueGetType = decltype(absl::get<2>(absl::move(v)));
>+
>+    EXPECT_TRUE((std::is_same<LValueGetType, double&>::value));
>+    EXPECT_TRUE((std::is_same<RValueGetType, double&&>::value));
>+    EXPECT_EQ(absl::get<2>(v), 2.);
>+    EXPECT_EQ(absl::get<2>(absl::move(v)), 2.);
>+
>+    const Var& const_v = v;
>+    using ConstLValueGetType = decltype(absl::get<2>(const_v));
>+    using ConstRValueGetType = decltype(absl::get<2>(absl::move(const_v)));
>+    EXPECT_TRUE((std::is_same<ConstLValueGetType, const double&>::value));
>+    EXPECT_TRUE((std::is_same<ConstRValueGetType, const double&&>::value));
>+    EXPECT_EQ(absl::get<2>(const_v), 2.);
>+    EXPECT_EQ(absl::get<2>(absl::move(const_v)), 2.);
>+  }
>+
>+  {
>+    Var v(absl::in_place_index_t<0>{}, 0);
>+    v.emplace<3>(1);
>+
>+    using LValueGetType = decltype(absl::get<3>(v));
>+    using RValueGetType = decltype(absl::get<3>(absl::move(v)));
>+
>+    EXPECT_TRUE((std::is_same<LValueGetType, int&>::value));
>+    EXPECT_TRUE((std::is_same<RValueGetType, int&&>::value));
>+    EXPECT_EQ(absl::get<3>(v), 1);
>+    EXPECT_EQ(absl::get<3>(absl::move(v)), 1);
>+
>+    const Var& const_v = v;
>+    using ConstLValueGetType = decltype(absl::get<3>(const_v));
>+    using ConstRValueGetType = decltype(absl::get<3>(absl::move(const_v)));
>+    EXPECT_TRUE((std::is_same<ConstLValueGetType, const int&>::value));
>+    EXPECT_TRUE((std::is_same<ConstRValueGetType, const int&&>::value));
>+    EXPECT_EQ(absl::get<3>(const_v), 1);
>+    EXPECT_EQ(absl::get<3>(absl::move(const_v)), 1);  // NOLINT
>+  }
>+}
>+
>+TEST(VariantTest, BadGetIndex) {
>+  using Var = variant<int, std::string, double>;
>+
>+  {
>+    Var v = 1;
>+
>+    ABSL_VARIANT_TEST_EXPECT_BAD_VARIANT_ACCESS(absl::get<1>(v));
>+    ABSL_VARIANT_TEST_EXPECT_BAD_VARIANT_ACCESS(absl::get<1>(std::move(v)));
>+
>+    const Var& const_v = v;
>+    ABSL_VARIANT_TEST_EXPECT_BAD_VARIANT_ACCESS(absl::get<1>(const_v));
>+    ABSL_VARIANT_TEST_EXPECT_BAD_VARIANT_ACCESS(
>+        absl::get<1>(std::move(const_v)));  // NOLINT
>+  }
>+
>+  {
>+    Var v = std::string("Hello");
>+
>+    ABSL_VARIANT_TEST_EXPECT_BAD_VARIANT_ACCESS(absl::get<0>(v));
>+    ABSL_VARIANT_TEST_EXPECT_BAD_VARIANT_ACCESS(absl::get<0>(std::move(v)));
>+
>+    const Var& const_v = v;
>+    ABSL_VARIANT_TEST_EXPECT_BAD_VARIANT_ACCESS(absl::get<0>(const_v));
>+    ABSL_VARIANT_TEST_EXPECT_BAD_VARIANT_ACCESS(
>+        absl::get<0>(std::move(const_v)));  // NOLINT
>+  }
>+}
>+
>+TEST(VariantTest, GetType) {
>+  using Var = variant<int, std::string, double>;
>+
>+  {
>+    Var v = 1;
>+
>+    using LValueGetType = decltype(absl::get<int>(v));
>+    using RValueGetType = decltype(absl::get<int>(absl::move(v)));
>+
>+    EXPECT_TRUE((std::is_same<LValueGetType, int&>::value));
>+    EXPECT_TRUE((std::is_same<RValueGetType, int&&>::value));
>+    EXPECT_EQ(absl::get<int>(v), 1);
>+    EXPECT_EQ(absl::get<int>(absl::move(v)), 1);
>+
>+    const Var& const_v = v;
>+    using ConstLValueGetType = decltype(absl::get<int>(const_v));
>+    using ConstRValueGetType = decltype(absl::get<int>(absl::move(const_v)));
>+    EXPECT_TRUE((std::is_same<ConstLValueGetType, const int&>::value));
>+    EXPECT_TRUE((std::is_same<ConstRValueGetType, const int&&>::value));
>+    EXPECT_EQ(absl::get<int>(const_v), 1);
>+    EXPECT_EQ(absl::get<int>(absl::move(const_v)), 1);
>+  }
>+
>+  {
>+    Var v = std::string("Hello");
>+
>+    using LValueGetType = decltype(absl::get<1>(v));
>+    using RValueGetType = decltype(absl::get<1>(absl::move(v)));
>+
>+    EXPECT_TRUE((std::is_same<LValueGetType, std::string&>::value));
>+    EXPECT_TRUE((std::is_same<RValueGetType, std::string&&>::value));
>+    EXPECT_EQ(absl::get<std::string>(v), "Hello");
>+    EXPECT_EQ(absl::get<std::string>(absl::move(v)), "Hello");
>+
>+    const Var& const_v = v;
>+    using ConstLValueGetType = decltype(absl::get<1>(const_v));
>+    using ConstRValueGetType = decltype(absl::get<1>(absl::move(const_v)));
>+    EXPECT_TRUE((std::is_same<ConstLValueGetType, const std::string&>::value));
>+    EXPECT_TRUE((std::is_same<ConstRValueGetType, const std::string&&>::value));
>+    EXPECT_EQ(absl::get<std::string>(const_v), "Hello");
>+    EXPECT_EQ(absl::get<std::string>(absl::move(const_v)), "Hello");
>+  }
>+
>+  {
>+    Var v = 2.0;
>+
>+    using LValueGetType = decltype(absl::get<2>(v));
>+    using RValueGetType = decltype(absl::get<2>(absl::move(v)));
>+
>+    EXPECT_TRUE((std::is_same<LValueGetType, double&>::value));
>+    EXPECT_TRUE((std::is_same<RValueGetType, double&&>::value));
>+    EXPECT_EQ(absl::get<double>(v), 2.);
>+    EXPECT_EQ(absl::get<double>(absl::move(v)), 2.);
>+
>+    const Var& const_v = v;
>+    using ConstLValueGetType = decltype(absl::get<2>(const_v));
>+    using ConstRValueGetType = decltype(absl::get<2>(absl::move(const_v)));
>+    EXPECT_TRUE((std::is_same<ConstLValueGetType, const double&>::value));
>+    EXPECT_TRUE((std::is_same<ConstRValueGetType, const double&&>::value));
>+    EXPECT_EQ(absl::get<double>(const_v), 2.);
>+    EXPECT_EQ(absl::get<double>(absl::move(const_v)), 2.);
>+  }
>+}
>+
>+TEST(VariantTest, BadGetType) {
>+  using Var = variant<int, std::string, double>;
>+
>+  {
>+    Var v = 1;
>+
>+    ABSL_VARIANT_TEST_EXPECT_BAD_VARIANT_ACCESS(absl::get<std::string>(v));
>+    ABSL_VARIANT_TEST_EXPECT_BAD_VARIANT_ACCESS(
>+        absl::get<std::string>(std::move(v)));
>+
>+    const Var& const_v = v;
>+    ABSL_VARIANT_TEST_EXPECT_BAD_VARIANT_ACCESS(absl::get<std::string>(const_v));
>+    ABSL_VARIANT_TEST_EXPECT_BAD_VARIANT_ACCESS(
>+        absl::get<std::string>(std::move(const_v)));  // NOLINT
>+  }
>+
>+  {
>+    Var v = std::string("Hello");
>+
>+    ABSL_VARIANT_TEST_EXPECT_BAD_VARIANT_ACCESS(absl::get<int>(v));
>+    ABSL_VARIANT_TEST_EXPECT_BAD_VARIANT_ACCESS(absl::get<int>(std::move(v)));
>+
>+    const Var& const_v = v;
>+    ABSL_VARIANT_TEST_EXPECT_BAD_VARIANT_ACCESS(absl::get<int>(const_v));
>+    ABSL_VARIANT_TEST_EXPECT_BAD_VARIANT_ACCESS(
>+        absl::get<int>(std::move(const_v)));  // NOLINT
>+  }
>+}
>+
>+TEST(VariantTest, GetIfIndex) {
>+  using Var = variant<int, std::string, double, int>;
>+
>+  {
>+    Var v(absl::in_place_index_t<0>{}, 0);
>+    EXPECT_TRUE(noexcept(absl::get_if<0>(&v)));
>+
>+    {
>+      auto* elem = absl::get_if<0>(&v);
>+      EXPECT_TRUE((std::is_same<decltype(elem), int*>::value));
>+      ASSERT_NE(elem, nullptr);
>+      EXPECT_EQ(*elem, 0);
>+      {
>+        auto* bad_elem = absl::get_if<1>(&v);
>+        EXPECT_TRUE((std::is_same<decltype(bad_elem), std::string*>::value));
>+        EXPECT_EQ(bad_elem, nullptr);
>+      }
>+      {
>+        auto* bad_elem = absl::get_if<2>(&v);
>+        EXPECT_TRUE((std::is_same<decltype(bad_elem), double*>::value));
>+        EXPECT_EQ(bad_elem, nullptr);
>+      }
>+      {
>+        auto* bad_elem = absl::get_if<3>(&v);
>+        EXPECT_TRUE((std::is_same<decltype(bad_elem), int*>::value));
>+        EXPECT_EQ(bad_elem, nullptr);
>+      }
>+    }
>+
>+    const Var& const_v = v;
>+    EXPECT_TRUE(noexcept(absl::get_if<0>(&const_v)));
>+
>+    {
>+      auto* elem = absl::get_if<0>(&const_v);
>+      EXPECT_TRUE((std::is_same<decltype(elem), const int*>::value));
>+      ASSERT_NE(elem, nullptr);
>+      EXPECT_EQ(*elem, 0);
>+      {
>+        auto* bad_elem = absl::get_if<1>(&const_v);
>+        EXPECT_TRUE((std::is_same<decltype(bad_elem), const std::string*>::value));
>+        EXPECT_EQ(bad_elem, nullptr);
>+      }
>+      {
>+        auto* bad_elem = absl::get_if<2>(&const_v);
>+        EXPECT_TRUE((std::is_same<decltype(bad_elem), const double*>::value));
>+        EXPECT_EQ(bad_elem, nullptr);
>+      }
>+      {
>+        auto* bad_elem = absl::get_if<3>(&const_v);
>+        EXPECT_EQ(bad_elem, nullptr);
>+        EXPECT_TRUE((std::is_same<decltype(bad_elem), const int*>::value));
>+      }
>+    }
>+  }
>+
>+  {
>+    Var v = std::string("Hello");
>+    EXPECT_TRUE(noexcept(absl::get_if<1>(&v)));
>+
>+    {
>+      auto* elem = absl::get_if<1>(&v);
>+      EXPECT_TRUE((std::is_same<decltype(elem), std::string*>::value));
>+      ASSERT_NE(elem, nullptr);
>+      EXPECT_EQ(*elem, "Hello");
>+      {
>+        auto* bad_elem = absl::get_if<0>(&v);
>+        EXPECT_TRUE((std::is_same<decltype(bad_elem), int*>::value));
>+        EXPECT_EQ(bad_elem, nullptr);
>+      }
>+      {
>+        auto* bad_elem = absl::get_if<2>(&v);
>+        EXPECT_TRUE((std::is_same<decltype(bad_elem), double*>::value));
>+        EXPECT_EQ(bad_elem, nullptr);
>+      }
>+      {
>+        auto* bad_elem = absl::get_if<3>(&v);
>+        EXPECT_TRUE((std::is_same<decltype(bad_elem), int*>::value));
>+        EXPECT_EQ(bad_elem, nullptr);
>+      }
>+    }
>+
>+    const Var& const_v = v;
>+    EXPECT_TRUE(noexcept(absl::get_if<1>(&const_v)));
>+
>+    {
>+      auto* elem = absl::get_if<1>(&const_v);
>+      EXPECT_TRUE((std::is_same<decltype(elem), const std::string*>::value));
>+      ASSERT_NE(elem, nullptr);
>+      EXPECT_EQ(*elem, "Hello");
>+      {
>+        auto* bad_elem = absl::get_if<0>(&const_v);
>+        EXPECT_TRUE((std::is_same<decltype(bad_elem), const int*>::value));
>+        EXPECT_EQ(bad_elem, nullptr);
>+      }
>+      {
>+        auto* bad_elem = absl::get_if<2>(&const_v);
>+        EXPECT_TRUE((std::is_same<decltype(bad_elem), const double*>::value));
>+        EXPECT_EQ(bad_elem, nullptr);
>+      }
>+      {
>+        auto* bad_elem = absl::get_if<3>(&const_v);
>+        EXPECT_EQ(bad_elem, nullptr);
>+        EXPECT_TRUE((std::is_same<decltype(bad_elem), const int*>::value));
>+      }
>+    }
>+  }
>+
>+  {
>+    Var v = 2.0;
>+    EXPECT_TRUE(noexcept(absl::get_if<2>(&v)));
>+
>+    {
>+      auto* elem = absl::get_if<2>(&v);
>+      EXPECT_TRUE((std::is_same<decltype(elem), double*>::value));
>+      ASSERT_NE(elem, nullptr);
>+      EXPECT_EQ(*elem, 2.0);
>+      {
>+        auto* bad_elem = absl::get_if<0>(&v);
>+        EXPECT_TRUE((std::is_same<decltype(bad_elem), int*>::value));
>+        EXPECT_EQ(bad_elem, nullptr);
>+      }
>+      {
>+        auto* bad_elem = absl::get_if<1>(&v);
>+        EXPECT_TRUE((std::is_same<decltype(bad_elem), std::string*>::value));
>+        EXPECT_EQ(bad_elem, nullptr);
>+      }
>+      {
>+        auto* bad_elem = absl::get_if<3>(&v);
>+        EXPECT_TRUE((std::is_same<decltype(bad_elem), int*>::value));
>+        EXPECT_EQ(bad_elem, nullptr);
>+      }
>+    }
>+
>+    const Var& const_v = v;
>+    EXPECT_TRUE(noexcept(absl::get_if<2>(&const_v)));
>+
>+    {
>+      auto* elem = absl::get_if<2>(&const_v);
>+      EXPECT_TRUE((std::is_same<decltype(elem), const double*>::value));
>+      ASSERT_NE(elem, nullptr);
>+      EXPECT_EQ(*elem, 2.0);
>+      {
>+        auto* bad_elem = absl::get_if<0>(&const_v);
>+        EXPECT_TRUE((std::is_same<decltype(bad_elem), const int*>::value));
>+        EXPECT_EQ(bad_elem, nullptr);
>+      }
>+      {
>+        auto* bad_elem = absl::get_if<1>(&const_v);
>+        EXPECT_TRUE((std::is_same<decltype(bad_elem), const std::string*>::value));
>+        EXPECT_EQ(bad_elem, nullptr);
>+      }
>+      {
>+        auto* bad_elem = absl::get_if<3>(&const_v);
>+        EXPECT_EQ(bad_elem, nullptr);
>+        EXPECT_TRUE((std::is_same<decltype(bad_elem), const int*>::value));
>+      }
>+    }
>+  }
>+
>+  {
>+    Var v(absl::in_place_index_t<0>{}, 0);
>+    v.emplace<3>(1);
>+    EXPECT_TRUE(noexcept(absl::get_if<3>(&v)));
>+
>+    {
>+      auto* elem = absl::get_if<3>(&v);
>+      EXPECT_TRUE((std::is_same<decltype(elem), int*>::value));
>+      ASSERT_NE(elem, nullptr);
>+      EXPECT_EQ(*elem, 1);
>+      {
>+        auto* bad_elem = absl::get_if<0>(&v);
>+        EXPECT_TRUE((std::is_same<decltype(bad_elem), int*>::value));
>+        EXPECT_EQ(bad_elem, nullptr);
>+      }
>+      {
>+        auto* bad_elem = absl::get_if<1>(&v);
>+        EXPECT_TRUE((std::is_same<decltype(bad_elem), std::string*>::value));
>+        EXPECT_EQ(bad_elem, nullptr);
>+      }
>+      {
>+        auto* bad_elem = absl::get_if<2>(&v);
>+        EXPECT_TRUE((std::is_same<decltype(bad_elem), double*>::value));
>+        EXPECT_EQ(bad_elem, nullptr);
>+      }
>+    }
>+
>+    const Var& const_v = v;
>+    EXPECT_TRUE(noexcept(absl::get_if<3>(&const_v)));
>+
>+    {
>+      auto* elem = absl::get_if<3>(&const_v);
>+      EXPECT_TRUE((std::is_same<decltype(elem), const int*>::value));
>+      ASSERT_NE(elem, nullptr);
>+      EXPECT_EQ(*elem, 1);
>+      {
>+        auto* bad_elem = absl::get_if<0>(&const_v);
>+        EXPECT_TRUE((std::is_same<decltype(bad_elem), const int*>::value));
>+        EXPECT_EQ(bad_elem, nullptr);
>+      }
>+      {
>+        auto* bad_elem = absl::get_if<1>(&const_v);
>+        EXPECT_TRUE((std::is_same<decltype(bad_elem), const std::string*>::value));
>+        EXPECT_EQ(bad_elem, nullptr);
>+      }
>+      {
>+        auto* bad_elem = absl::get_if<2>(&const_v);
>+        EXPECT_EQ(bad_elem, nullptr);
>+        EXPECT_TRUE((std::is_same<decltype(bad_elem), const double*>::value));
>+      }
>+    }
>+  }
>+}
>+
>+//////////////////////
>+// [variant.relops] //
>+//////////////////////
>+
>+TEST(VariantTest, OperatorEquals) {
>+  variant<int, std::string> a(1), b(1);
>+  EXPECT_TRUE(a == b);
>+  EXPECT_TRUE(b == a);
>+  EXPECT_FALSE(a != b);
>+  EXPECT_FALSE(b != a);
>+
>+  b = "str";
>+  EXPECT_FALSE(a == b);
>+  EXPECT_FALSE(b == a);
>+  EXPECT_TRUE(a != b);
>+  EXPECT_TRUE(b != a);
>+
>+  b = 0;
>+  EXPECT_FALSE(a == b);
>+  EXPECT_FALSE(b == a);
>+  EXPECT_TRUE(a != b);
>+  EXPECT_TRUE(b != a);
>+
>+  a = b = "foo";
>+  EXPECT_TRUE(a == b);
>+  EXPECT_TRUE(b == a);
>+  EXPECT_FALSE(a != b);
>+  EXPECT_FALSE(b != a);
>+
>+  a = "bar";
>+  EXPECT_FALSE(a == b);
>+  EXPECT_FALSE(b == a);
>+  EXPECT_TRUE(a != b);
>+  EXPECT_TRUE(b != a);
>+}
>+
>+TEST(VariantTest, OperatorRelational) {
>+  variant<int, std::string> a(1), b(1);
>+  EXPECT_FALSE(a < b);
>+  EXPECT_FALSE(b < a);
>+  EXPECT_FALSE(a > b);
>+  EXPECT_FALSE(b > a);
>+  EXPECT_TRUE(a <= b);
>+  EXPECT_TRUE(b <= a);
>+  EXPECT_TRUE(a >= b);
>+  EXPECT_TRUE(b >= a);
>+
>+  b = "str";
>+  EXPECT_TRUE(a < b);
>+  EXPECT_FALSE(b < a);
>+  EXPECT_FALSE(a > b);
>+  EXPECT_TRUE(b > a);
>+  EXPECT_TRUE(a <= b);
>+  EXPECT_FALSE(b <= a);
>+  EXPECT_FALSE(a >= b);
>+  EXPECT_TRUE(b >= a);
>+
>+  b = 0;
>+  EXPECT_FALSE(a < b);
>+  EXPECT_TRUE(b < a);
>+  EXPECT_TRUE(a > b);
>+  EXPECT_FALSE(b > a);
>+  EXPECT_FALSE(a <= b);
>+  EXPECT_TRUE(b <= a);
>+  EXPECT_TRUE(a >= b);
>+  EXPECT_FALSE(b >= a);
>+
>+  a = b = "foo";
>+  EXPECT_FALSE(a < b);
>+  EXPECT_FALSE(b < a);
>+  EXPECT_FALSE(a > b);
>+  EXPECT_FALSE(b > a);
>+  EXPECT_TRUE(a <= b);
>+  EXPECT_TRUE(b <= a);
>+  EXPECT_TRUE(a >= b);
>+  EXPECT_TRUE(b >= a);
>+
>+  a = "bar";
>+  EXPECT_TRUE(a < b);
>+  EXPECT_FALSE(b < a);
>+  EXPECT_FALSE(a > b);
>+  EXPECT_TRUE(b > a);
>+  EXPECT_TRUE(a <= b);
>+  EXPECT_FALSE(b <= a);
>+  EXPECT_FALSE(a >= b);
>+  EXPECT_TRUE(b >= a);
>+}
>+
>+#ifdef ABSL_HAVE_EXCEPTIONS
>+
>+TEST(VariantTest, ValuelessOperatorEquals) {
>+  variant<MoveCanThrow, std::string> int_v(1), string_v("Hello"),
>+      valueless(absl::in_place_index_t<0>{}),
>+      other_valueless(absl::in_place_index_t<0>{});
>+  ToValuelessByException(valueless);
>+  ToValuelessByException(other_valueless);
>+
>+  EXPECT_TRUE(valueless == other_valueless);
>+  EXPECT_TRUE(other_valueless == valueless);
>+  EXPECT_FALSE(valueless == int_v);
>+  EXPECT_FALSE(valueless == string_v);
>+  EXPECT_FALSE(int_v == valueless);
>+  EXPECT_FALSE(string_v == valueless);
>+
>+  EXPECT_FALSE(valueless != other_valueless);
>+  EXPECT_FALSE(other_valueless != valueless);
>+  EXPECT_TRUE(valueless != int_v);
>+  EXPECT_TRUE(valueless != string_v);
>+  EXPECT_TRUE(int_v != valueless);
>+  EXPECT_TRUE(string_v != valueless);
>+}
>+
>+TEST(VariantTest, ValuelessOperatorRelational) {
>+  variant<MoveCanThrow, std::string> int_v(1), string_v("Hello"),
>+      valueless(absl::in_place_index_t<0>{}),
>+      other_valueless(absl::in_place_index_t<0>{});
>+  ToValuelessByException(valueless);
>+  ToValuelessByException(other_valueless);
>+
>+  EXPECT_FALSE(valueless < other_valueless);
>+  EXPECT_FALSE(other_valueless < valueless);
>+  EXPECT_TRUE(valueless < int_v);
>+  EXPECT_TRUE(valueless < string_v);
>+  EXPECT_FALSE(int_v < valueless);
>+  EXPECT_FALSE(string_v < valueless);
>+
>+  EXPECT_TRUE(valueless <= other_valueless);
>+  EXPECT_TRUE(other_valueless <= valueless);
>+  EXPECT_TRUE(valueless <= int_v);
>+  EXPECT_TRUE(valueless <= string_v);
>+  EXPECT_FALSE(int_v <= valueless);
>+  EXPECT_FALSE(string_v <= valueless);
>+
>+  EXPECT_TRUE(valueless >= other_valueless);
>+  EXPECT_TRUE(other_valueless >= valueless);
>+  EXPECT_FALSE(valueless >= int_v);
>+  EXPECT_FALSE(valueless >= string_v);
>+  EXPECT_TRUE(int_v >= valueless);
>+  EXPECT_TRUE(string_v >= valueless);
>+
>+  EXPECT_FALSE(valueless > other_valueless);
>+  EXPECT_FALSE(other_valueless > valueless);
>+  EXPECT_FALSE(valueless > int_v);
>+  EXPECT_FALSE(valueless > string_v);
>+  EXPECT_TRUE(int_v > valueless);
>+  EXPECT_TRUE(string_v > valueless);
>+}
>+
>+#endif
>+
>+/////////////////////
>+// [variant.visit] //
>+/////////////////////
>+
>+template <typename T>
>+struct ConvertTo {
>+  template <typename U>
>+  T operator()(const U& u) const {
>+    return u;
>+  }
>+};
>+
>+TEST(VariantTest, VisitSimple) {
>+  variant<std::string, const char*> v = "A";
>+
>+  std::string str = absl::visit(ConvertTo<std::string>{}, v);
>+  EXPECT_EQ("A", str);
>+
>+  v = std::string("B");
>+
>+  absl::string_view piece = absl::visit(ConvertTo<absl::string_view>{}, v);
>+  EXPECT_EQ("B", piece);
>+
>+  struct StrLen {
>+    int operator()(const std::string& s) const { return s.size(); }
>+    int operator()(const char* s) const { return strlen(s); }
>+  };
>+
>+  v = "SomeStr";
>+  EXPECT_EQ(7, absl::visit(StrLen{}, v));
>+  v = std::string("VeryLargeThisTime");
>+  EXPECT_EQ(17, absl::visit(StrLen{}, v));
>+}
>+
>+TEST(VariantTest, VisitRValue) {
>+  variant<std::string> v = std::string("X");
>+  struct Visitor {
>+    bool operator()(const std::string&) const { return false; }
>+    bool operator()(std::string&&) const { return true; }  // NOLINT
>+
>+    int operator()(const std::string&, const std::string&) const { return 0; }
>+    int operator()(const std::string&, std::string&&) const { return 1; }  // NOLINT
>+    int operator()(std::string&&, const std::string&) const { return 2; }  // NOLINT
>+    int operator()(std::string&&, std::string&&) const { return 3; }       // NOLINT
>+  };
>+  EXPECT_FALSE(absl::visit(Visitor{}, v));
>+  EXPECT_TRUE(absl::visit(Visitor{}, absl::move(v)));
>+
>+  // Also test the variadic overload.
>+  EXPECT_EQ(0, absl::visit(Visitor{}, v, v));
>+  EXPECT_EQ(1, absl::visit(Visitor{}, v, absl::move(v)));
>+  EXPECT_EQ(2, absl::visit(Visitor{}, absl::move(v), v));
>+  EXPECT_EQ(3, absl::visit(Visitor{}, absl::move(v), absl::move(v)));
>+}
>+
>+TEST(VariantTest, VisitRValueVisitor) {
>+  variant<std::string> v = std::string("X");
>+  struct Visitor {
>+    bool operator()(const std::string&) const& { return false; }
>+    bool operator()(const std::string&) && { return true; }
>+  };
>+  Visitor visitor;
>+  EXPECT_FALSE(absl::visit(visitor, v));
>+  EXPECT_TRUE(absl::visit(Visitor{}, v));
>+}
>+
>+TEST(VariantTest, VisitResultTypeDifferent) {
>+  variant<std::string> v = std::string("X");
>+  struct LValue_LValue {};
>+  struct RValue_LValue {};
>+  struct LValue_RValue {};
>+  struct RValue_RValue {};
>+  struct Visitor {
>+    LValue_LValue operator()(const std::string&) const& { return {}; }
>+    RValue_LValue operator()(std::string&&) const& { return {}; }  // NOLINT
>+    LValue_RValue operator()(const std::string&) && { return {}; }
>+    RValue_RValue operator()(std::string&&) && { return {}; }  // NOLINT
>+  } visitor;
>+
>+  EXPECT_TRUE(
>+      (std::is_same<LValue_LValue, decltype(absl::visit(visitor, v))>::value));
>+  EXPECT_TRUE(
>+      (std::is_same<RValue_LValue,
>+                    decltype(absl::visit(visitor, absl::move(v)))>::value));
>+  EXPECT_TRUE((
>+      std::is_same<LValue_RValue, decltype(absl::visit(Visitor{}, v))>::value));
>+  EXPECT_TRUE(
>+      (std::is_same<RValue_RValue,
>+                    decltype(absl::visit(Visitor{}, absl::move(v)))>::value));
>+}
>+
>+TEST(VariantTest, VisitVariadic) {
>+  using A = variant<int, std::string>;
>+  using B = variant<std::unique_ptr<int>, absl::string_view>;
>+
>+  struct Visitor {
>+    std::pair<int, int> operator()(int a, std::unique_ptr<int> b) const {
>+      return {a, *b};
>+    }
>+    std::pair<int, int> operator()(absl::string_view a,
>+                                   std::unique_ptr<int> b) const {
>+      return {static_cast<int>(a.size()), static_cast<int>(*b)};
>+    }
>+    std::pair<int, int> operator()(int a, absl::string_view b) const {
>+      return {a, static_cast<int>(b.size())};
>+    }
>+    std::pair<int, int> operator()(absl::string_view a,
>+                                   absl::string_view b) const {
>+      return {static_cast<int>(a.size()), static_cast<int>(b.size())};
>+    }
>+  };
>+
>+  EXPECT_THAT(absl::visit(Visitor(), A(1), B(std::unique_ptr<int>(new int(7)))),
>+              ::testing::Pair(1, 7));
>+  EXPECT_THAT(absl::visit(Visitor(), A(1), B(absl::string_view("ABC"))),
>+              ::testing::Pair(1, 3));
>+  EXPECT_THAT(absl::visit(Visitor(), A(std::string("BBBBB")),
>+                          B(std::unique_ptr<int>(new int(7)))),
>+              ::testing::Pair(5, 7));
>+  EXPECT_THAT(
>+      absl::visit(Visitor(), A(std::string("BBBBB")), B(absl::string_view("ABC"))),
>+      ::testing::Pair(5, 3));
>+}
>+
>+TEST(VariantTest, VisitNoArgs) {
>+  EXPECT_EQ(5, absl::visit([] { return 5; }));
>+}
>+
>+struct ConstFunctor {
>+  int operator()(int a, int b) const { return a - b; }
>+};
>+
>+struct MutableFunctor {
>+  int operator()(int a, int b) { return a - b; }
>+};
>+
>+struct Class {
>+  int Method(int a, int b) { return a - b; }
>+  int ConstMethod(int a, int b) const { return a - b; }
>+
>+  int member;
>+};
>+
>+TEST(VariantTest, VisitReferenceWrapper) {
>+  ConstFunctor cf;
>+  MutableFunctor mf;
>+  absl::variant<int> three = 3;
>+  absl::variant<int> two = 2;
>+
>+  EXPECT_EQ(1, absl::visit(std::cref(cf), three, two));
>+  EXPECT_EQ(1, absl::visit(std::ref(cf), three, two));
>+  EXPECT_EQ(1, absl::visit(std::ref(mf), three, two));
>+}
>+
>+// libstdc++ std::variant doesn't support the INVOKE semantics.
>+#if !(defined(ABSL_HAVE_STD_VARIANT) && defined(__GLIBCXX__))
>+TEST(VariantTest, VisitMemberFunction) {
>+  absl::variant<std::unique_ptr<Class>> p(absl::make_unique<Class>());
>+  absl::variant<std::unique_ptr<const Class>> cp(
>+      absl::make_unique<const Class>());
>+  absl::variant<int> three = 3;
>+  absl::variant<int> two = 2;
>+
>+  EXPECT_EQ(1, absl::visit(&Class::Method, p, three, two));
>+  EXPECT_EQ(1, absl::visit(&Class::ConstMethod, p, three, two));
>+  EXPECT_EQ(1, absl::visit(&Class::ConstMethod, cp, three, two));
>+}
>+
>+TEST(VariantTest, VisitDataMember) {
>+  absl::variant<std::unique_ptr<Class>> p(absl::make_unique<Class>(Class{42}));
>+  absl::variant<std::unique_ptr<const Class>> cp(
>+      absl::make_unique<const Class>(Class{42}));
>+  EXPECT_EQ(42, absl::visit(&Class::member, p));
>+
>+  absl::visit(&Class::member, p) = 5;
>+  EXPECT_EQ(5, absl::visit(&Class::member, p));
>+
>+  EXPECT_EQ(42, absl::visit(&Class::member, cp));
>+}
>+#endif  // !(defined(ABSL_HAVE_STD_VARIANT) && defined(__GLIBCXX__))
>+
>+/////////////////////////
>+// [variant.monostate] //
>+/////////////////////////
>+
>+TEST(VariantTest, MonostateBasic) {
>+  absl::monostate mono;
>+  (void)mono;
>+
>+  // TODO(mattcalabrese) Expose move triviality metafunctions in absl.
>+  EXPECT_TRUE(absl::is_trivially_default_constructible<absl::monostate>::value);
>+  EXPECT_TRUE(is_trivially_move_constructible<absl::monostate>::value);
>+  EXPECT_TRUE(absl::is_trivially_copy_constructible<absl::monostate>::value);
>+  EXPECT_TRUE(is_trivially_move_assignable<absl::monostate>::value);
>+  EXPECT_TRUE(absl::is_trivially_copy_assignable<absl::monostate>::value);
>+  EXPECT_TRUE(absl::is_trivially_destructible<absl::monostate>::value);
>+}
>+
>+TEST(VariantTest, VariantMonostateDefaultConstruction) {
>+  absl::variant<absl::monostate, NonDefaultConstructible> var;
>+  EXPECT_EQ(var.index(), 0);
>+}
>+
>+////////////////////////////////
>+// [variant.monostate.relops] //
>+////////////////////////////////
>+
>+TEST(VariantTest, MonostateComparisons) {
>+  absl::monostate lhs, rhs;
>+
>+  EXPECT_EQ(lhs, lhs);
>+  EXPECT_EQ(lhs, rhs);
>+
>+  EXPECT_FALSE(lhs != lhs);
>+  EXPECT_FALSE(lhs != rhs);
>+  EXPECT_FALSE(lhs < lhs);
>+  EXPECT_FALSE(lhs < rhs);
>+  EXPECT_FALSE(lhs > lhs);
>+  EXPECT_FALSE(lhs > rhs);
>+
>+  EXPECT_LE(lhs, lhs);
>+  EXPECT_LE(lhs, rhs);
>+  EXPECT_GE(lhs, lhs);
>+  EXPECT_GE(lhs, rhs);
>+
>+  EXPECT_TRUE(noexcept(std::declval<absl::monostate>() ==
>+                       std::declval<absl::monostate>()));
>+  EXPECT_TRUE(noexcept(std::declval<absl::monostate>() !=
>+                       std::declval<absl::monostate>()));
>+  EXPECT_TRUE(noexcept(std::declval<absl::monostate>() <
>+                       std::declval<absl::monostate>()));
>+  EXPECT_TRUE(noexcept(std::declval<absl::monostate>() >
>+                       std::declval<absl::monostate>()));
>+  EXPECT_TRUE(noexcept(std::declval<absl::monostate>() <=
>+                       std::declval<absl::monostate>()));
>+  EXPECT_TRUE(noexcept(std::declval<absl::monostate>() >=
>+                       std::declval<absl::monostate>()));
>+}
>+
>+///////////////////////
>+// [variant.specalg] //
>+///////////////////////
>+
>+TEST(VariantTest, NonmemberSwap) {
>+  using std::swap;
>+
>+  SpecialSwap v1(3);
>+  SpecialSwap v2(7);
>+
>+  variant<SpecialSwap> a = v1, b = v2;
>+
>+  EXPECT_THAT(a, VariantWith<SpecialSwap>(v1));
>+  EXPECT_THAT(b, VariantWith<SpecialSwap>(v2));
>+
>+  std::swap(a, b);
>+  EXPECT_THAT(a, VariantWith<SpecialSwap>(v2));
>+  EXPECT_THAT(b, VariantWith<SpecialSwap>(v1));
>+#ifndef ABSL_HAVE_STD_VARIANT
>+  EXPECT_FALSE(absl::get<SpecialSwap>(a).special_swap);
>+#endif
>+
>+  swap(a, b);
>+  EXPECT_THAT(a, VariantWith<SpecialSwap>(v1));
>+  EXPECT_THAT(b, VariantWith<SpecialSwap>(v2));
>+  EXPECT_TRUE(absl::get<SpecialSwap>(b).special_swap);
>+}
>+
>+//////////////////////////
>+// [variant.bad.access] //
>+//////////////////////////
>+
>+TEST(VariantTest, BadAccess) {
>+  EXPECT_TRUE(noexcept(absl::bad_variant_access()));
>+  absl::bad_variant_access exception_obj;
>+  std::exception* base = &exception_obj;
>+  (void)base;
>+}
>+
>+////////////////////
>+// [variant.hash] //
>+////////////////////
>+
>+TEST(VariantTest, MonostateHash) {
>+  absl::monostate mono, other_mono;
>+  std::hash<absl::monostate> const hasher{};
>+  static_assert(std::is_same<decltype(hasher(mono)), std::size_t>::value, "");
>+  EXPECT_EQ(hasher(mono), hasher(other_mono));
>+}
>+
>+TEST(VariantTest, Hash) {
>+  static_assert(type_traits_internal::IsHashEnabled<variant<int>>::value, "");
>+  static_assert(type_traits_internal::IsHashEnabled<variant<Hashable>>::value,
>+                "");
>+  static_assert(
>+      type_traits_internal::IsHashEnabled<variant<int, Hashable>>::value, "");
>+
>+#if defined(_MSC_VER) ||                                   \
>+    (defined(_LIBCPP_VERSION) && _LIBCPP_VERSION < 4000 && \
>+     _LIBCPP_STD_VER > 11) ||                              \
>+    defined(__APPLE__)
>+  // For MSVC and libc++ (< 4.0 and c++14), std::hash primary template has a
>+  // static_assert to catch any user-defined type T that doesn't provide a hash
>+  // specialization. So instantiating std::hash<variant<T>> will result
>+  // in a hard error which is not SFINAE friendly.
>+#define ABSL_STD_HASH_NOT_SFINAE_FRIENDLY 1
>+#endif
>+
>+#ifndef ABSL_STD_HASH_NOT_SFINAE_FRIENDLY
>+  static_assert(
>+      !type_traits_internal::IsHashEnabled<variant<NonHashable>>::value, "");
>+  static_assert(!type_traits_internal::IsHashEnabled<
>+                    variant<Hashable, NonHashable>>::value,
>+                "");
>+#endif
>+
>+// MSVC std::hash<std::variant> does not use the index, thus produce the same
>+// result on the same value as different alternative.
>+#if !(defined(_MSC_VER) && defined(ABSL_HAVE_STD_VARIANT))
>+  {
>+    // same value as different alternative
>+    variant<int, int> v0(in_place_index_t<0>{}, 42);
>+    variant<int, int> v1(in_place_index_t<1>{}, 42);
>+    std::hash<variant<int, int>> hash;
>+    EXPECT_NE(hash(v0), hash(v1));
>+  }
>+#endif  // !(defined(_MSC_VER) && defined(ABSL_HAVE_STD_VARIANT))
>+
>+  {
>+    std::hash<variant<int>> hash;
>+    std::set<size_t> hashcodes;
>+    for (int i = 0; i < 100; ++i) {
>+      hashcodes.insert(hash(i));
>+    }
>+    EXPECT_GT(hashcodes.size(), 90);
>+
>+    // test const-qualified
>+    static_assert(
>+        type_traits_internal::IsHashEnabled<variant<const int>>::value, "");
>+    static_assert(
>+        type_traits_internal::IsHashEnabled<variant<const Hashable>>::value,
>+        "");
>+    std::hash<absl::variant<const int>> c_hash;
>+    for (int i = 0; i < 100; ++i) {
>+      EXPECT_EQ(hash(i), c_hash(i));
>+    }
>+  }
>+}
>+
>+////////////////////////////////////////
>+// Miscellaneous and deprecated tests //
>+////////////////////////////////////////
>+
>+// Test that a set requiring a basic type conversion works correctly.
>+TEST(VariantTest, TestConvertingSet) {
>+  typedef variant<double> Variant;
>+  Variant v(1.0);
>+  const int two = 2;
>+  v = two;
>+  EXPECT_TRUE(absl::holds_alternative<double>(v));
>+  ASSERT_TRUE(nullptr != absl::get_if<double>(&v));
>+  EXPECT_DOUBLE_EQ(2, absl::get<double>(v));
>+}
>+
>+// Test that a vector of variants behaves reasonably.
>+TEST(VariantTest, Container) {
>+  typedef variant<int, float> Variant;
>+
>+  // Creation of vector should work
>+  std::vector<Variant> vec;
>+  vec.push_back(Variant(10));
>+  vec.push_back(Variant(20.0f));
>+
>+  // Vector resizing should work if we supply a value for new slots
>+  vec.resize(10, Variant(0));
>+}
>+
>+// Test that a variant with a non-copyable type can be constructed and
>+// manipulated to some degree.
>+TEST(VariantTest, TestVariantWithNonCopyableType) {
>+  typedef variant<int, NonCopyable> Variant;
>+  const int kValue = 1;
>+  Variant v(kValue);
>+  ASSERT_TRUE(absl::holds_alternative<int>(v));
>+  EXPECT_EQ(kValue, absl::get<int>(v));
>+}
>+
>+// Test that a variant with a non-copyable type can be transformed to
>+// the non-copyable type with a call to `emplace` for different numbers
>+// of arguments. We do not need to test this for each of T1 ... T8
>+// because `emplace` does not overload on T1 ... to T8, so if this
>+// works for any one of T1 ... T8, then it works for all of them. We
>+// do need to test that it works with varying numbers of parameters
>+// though.
>+TEST(VariantTest, TestEmplace) {
>+  typedef variant<int, NonCopyable> Variant;
>+  const int kValue = 1;
>+  Variant v(kValue);
>+  ASSERT_TRUE(absl::holds_alternative<int>(v));
>+  EXPECT_EQ(kValue, absl::get<int>(v));
>+
>+  // emplace with zero arguments, then back to 'int'
>+  v.emplace<NonCopyable>();
>+  ASSERT_TRUE(absl::holds_alternative<NonCopyable>(v));
>+  EXPECT_EQ(0, absl::get<NonCopyable>(v).value);
>+  v = kValue;
>+  ASSERT_TRUE(absl::holds_alternative<int>(v));
>+
>+  // emplace with one argument:
>+  v.emplace<NonCopyable>(1);
>+  ASSERT_TRUE(absl::holds_alternative<NonCopyable>(v));
>+  EXPECT_EQ(1, absl::get<NonCopyable>(v).value);
>+  v = kValue;
>+  ASSERT_TRUE(absl::holds_alternative<int>(v));
>+
>+  // emplace with two arguments:
>+  v.emplace<NonCopyable>(1, 2);
>+  ASSERT_TRUE(absl::holds_alternative<NonCopyable>(v));
>+  EXPECT_EQ(3, absl::get<NonCopyable>(v).value);
>+  v = kValue;
>+  ASSERT_TRUE(absl::holds_alternative<int>(v));
>+
>+  // emplace with three arguments
>+  v.emplace<NonCopyable>(1, 2, 3);
>+  ASSERT_TRUE(absl::holds_alternative<NonCopyable>(v));
>+  EXPECT_EQ(6, absl::get<NonCopyable>(v).value);
>+  v = kValue;
>+  ASSERT_TRUE(absl::holds_alternative<int>(v));
>+
>+  // emplace with four arguments
>+  v.emplace<NonCopyable>(1, 2, 3, 4);
>+  ASSERT_TRUE(absl::holds_alternative<NonCopyable>(v));
>+  EXPECT_EQ(10, absl::get<NonCopyable>(v).value);
>+  v = kValue;
>+  ASSERT_TRUE(absl::holds_alternative<int>(v));
>+}
>+
>+TEST(VariantTest, TestEmplaceDestroysCurrentValue) {
>+  typedef variant<int, IncrementInDtor, NonCopyable> Variant;
>+  int counter = 0;
>+  Variant v(0);
>+  ASSERT_TRUE(absl::holds_alternative<int>(v));
>+  v.emplace<IncrementInDtor>(&counter);
>+  ASSERT_TRUE(absl::holds_alternative<IncrementInDtor>(v));
>+  ASSERT_EQ(0, counter);
>+  v.emplace<NonCopyable>();
>+  ASSERT_TRUE(absl::holds_alternative<NonCopyable>(v));
>+  EXPECT_EQ(1, counter);
>+}
>+
>+TEST(VariantTest, TestMoveSemantics) {
>+  typedef variant<std::unique_ptr<int>, std::unique_ptr<std::string>> Variant;
>+
>+  // Construct a variant by moving from an element value.
>+  Variant v(absl::WrapUnique(new int(10)));
>+  EXPECT_TRUE(absl::holds_alternative<std::unique_ptr<int>>(v));
>+
>+  // Construct a variant by moving from another variant.
>+  Variant v2(absl::move(v));
>+  ASSERT_TRUE(absl::holds_alternative<std::unique_ptr<int>>(v2));
>+  ASSERT_NE(nullptr, absl::get<std::unique_ptr<int>>(v2));
>+  EXPECT_EQ(10, *absl::get<std::unique_ptr<int>>(v2));
>+
>+  // Moving from a variant object leaves it holding moved-from value of the
>+  // same element type.
>+  EXPECT_TRUE(absl::holds_alternative<std::unique_ptr<int>>(v));
>+  ASSERT_NE(nullptr, absl::get_if<std::unique_ptr<int>>(&v));
>+  EXPECT_EQ(nullptr, absl::get<std::unique_ptr<int>>(v));
>+
>+  // Assign a variant from an element value by move.
>+  v = absl::make_unique<std::string>("foo");
>+  ASSERT_TRUE(absl::holds_alternative<std::unique_ptr<std::string>>(v));
>+  EXPECT_EQ("foo", *absl::get<std::unique_ptr<std::string>>(v));
>+
>+  // Move-assign a variant.
>+  v2 = absl::move(v);
>+  ASSERT_TRUE(absl::holds_alternative<std::unique_ptr<std::string>>(v2));
>+  EXPECT_EQ("foo", *absl::get<std::unique_ptr<std::string>>(v2));
>+  EXPECT_TRUE(absl::holds_alternative<std::unique_ptr<std::string>>(v));
>+}
>+
>+variant<int, std::string> PassThrough(const variant<int, std::string>& arg) {
>+  return arg;
>+}
>+
>+TEST(VariantTest, TestImplicitConversion) {
>+  EXPECT_TRUE(absl::holds_alternative<int>(PassThrough(0)));
>+
>+  // We still need the explicit cast for std::string, because C++ won't apply
>+  // two user-defined implicit conversions in a row.
>+  EXPECT_TRUE(absl::holds_alternative<std::string>(PassThrough(std::string("foo"))));
>+}
>+
>+struct Convertible2;
>+struct Convertible1 {
>+  Convertible1() {}
>+  Convertible1(const Convertible1&) {}
>+  Convertible1& operator=(const Convertible1&) { return *this; }
>+
>+  // implicit conversion from Convertible2
>+  Convertible1(const Convertible2&) {}  // NOLINT(runtime/explicit)
>+};
>+
>+struct Convertible2 {
>+  Convertible2() {}
>+  Convertible2(const Convertible2&) {}
>+  Convertible2& operator=(const Convertible2&) { return *this; }
>+
>+  // implicit conversion from Convertible1
>+  Convertible2(const Convertible1&) {}  // NOLINT(runtime/explicit)
>+};
>+
>+TEST(VariantTest, TestRvalueConversion) {
>+  variant<double, std::string> var(
>+      ConvertVariantTo<variant<double, std::string>>(variant<std::string, int>(0)));
>+  ASSERT_TRUE(absl::holds_alternative<double>(var));
>+  EXPECT_EQ(0.0, absl::get<double>(var));
>+
>+  var = ConvertVariantTo<variant<double, std::string>>(
>+      variant<const char*, float>("foo"));
>+  ASSERT_TRUE(absl::holds_alternative<std::string>(var));
>+  EXPECT_EQ("foo", absl::get<std::string>(var));
>+
>+  variant<double> singleton(
>+      ConvertVariantTo<variant<double>>(variant<int, float>(42)));
>+  ASSERT_TRUE(absl::holds_alternative<double>(singleton));
>+  EXPECT_EQ(42.0, absl::get<double>(singleton));
>+
>+  singleton = ConvertVariantTo<variant<double>>(variant<int, float>(3.14f));
>+  ASSERT_TRUE(absl::holds_alternative<double>(singleton));
>+  EXPECT_FLOAT_EQ(3.14f, static_cast<float>(absl::get<double>(singleton)));
>+
>+  singleton = ConvertVariantTo<variant<double>>(variant<int>(0));
>+  ASSERT_TRUE(absl::holds_alternative<double>(singleton));
>+  EXPECT_EQ(0.0, absl::get<double>(singleton));
>+
>+  variant<int32_t, uint32_t> variant2(
>+      ConvertVariantTo<variant<int32_t, uint32_t>>(variant<int32_t>(42)));
>+  ASSERT_TRUE(absl::holds_alternative<int32_t>(variant2));
>+  EXPECT_EQ(42, absl::get<int32_t>(variant2));
>+
>+  variant2 = ConvertVariantTo<variant<int32_t, uint32_t>>(variant<uint32_t>(42));
>+  ASSERT_TRUE(absl::holds_alternative<uint32_t>(variant2));
>+  EXPECT_EQ(42, absl::get<uint32_t>(variant2));
>+
>+  variant<Convertible1, Convertible2> variant3(
>+      ConvertVariantTo<variant<Convertible1, Convertible2>>(
>+          (variant<Convertible2, Convertible1>(Convertible1()))));
>+  ASSERT_TRUE(absl::holds_alternative<Convertible1>(variant3));
>+
>+  variant3 = ConvertVariantTo<variant<Convertible1, Convertible2>>(
>+      variant<Convertible2, Convertible1>(Convertible2()));
>+  ASSERT_TRUE(absl::holds_alternative<Convertible2>(variant3));
>+}
>+
>+TEST(VariantTest, TestLvalueConversion) {
>+  variant<std::string, int> source1 = 0;
>+  variant<double, std::string> destination(
>+      ConvertVariantTo<variant<double, std::string>>(source1));
>+  ASSERT_TRUE(absl::holds_alternative<double>(destination));
>+  EXPECT_EQ(0.0, absl::get<double>(destination));
>+
>+  variant<const char*, float> source2 = "foo";
>+  destination = ConvertVariantTo<variant<double, std::string>>(source2);
>+  ASSERT_TRUE(absl::holds_alternative<std::string>(destination));
>+  EXPECT_EQ("foo", absl::get<std::string>(destination));
>+
>+  variant<int, float> source3(42);
>+  variant<double> singleton(ConvertVariantTo<variant<double>>(source3));
>+  ASSERT_TRUE(absl::holds_alternative<double>(singleton));
>+  EXPECT_EQ(42.0, absl::get<double>(singleton));
>+
>+  source3 = 3.14f;
>+  singleton = ConvertVariantTo<variant<double>>(source3);
>+  ASSERT_TRUE(absl::holds_alternative<double>(singleton));
>+  EXPECT_FLOAT_EQ(3.14f, static_cast<float>(absl::get<double>(singleton)));
>+
>+  variant<int> source4(0);
>+  singleton = ConvertVariantTo<variant<double>>(source4);
>+  ASSERT_TRUE(absl::holds_alternative<double>(singleton));
>+  EXPECT_EQ(0.0, absl::get<double>(singleton));
>+
>+  variant<int32_t> source5(42);
>+  variant<int32_t, uint32_t> variant2(
>+      ConvertVariantTo<variant<int32_t, uint32_t>>(source5));
>+  ASSERT_TRUE(absl::holds_alternative<int32_t>(variant2));
>+  EXPECT_EQ(42, absl::get<int32_t>(variant2));
>+
>+  variant<uint32_t> source6(42);
>+  variant2 = ConvertVariantTo<variant<int32_t, uint32_t>>(source6);
>+  ASSERT_TRUE(absl::holds_alternative<uint32_t>(variant2));
>+  EXPECT_EQ(42, absl::get<uint32_t>(variant2));
>+
>+  variant<Convertible2, Convertible1> source7((Convertible1()));
>+  variant<Convertible1, Convertible2> variant3(
>+      ConvertVariantTo<variant<Convertible1, Convertible2>>(source7));
>+  ASSERT_TRUE(absl::holds_alternative<Convertible1>(variant3));
>+
>+  source7 = Convertible2();
>+  variant3 = ConvertVariantTo<variant<Convertible1, Convertible2>>(source7);
>+  ASSERT_TRUE(absl::holds_alternative<Convertible2>(variant3));
>+}
>+
>+TEST(VariantTest, TestMoveConversion) {
>+  using Variant =
>+      variant<std::unique_ptr<const int>, std::unique_ptr<const std::string>>;
>+  using OtherVariant = variant<std::unique_ptr<int>, std::unique_ptr<std::string>>;
>+
>+  Variant var(
>+      ConvertVariantTo<Variant>(OtherVariant{absl::make_unique<int>(0)}));
>+  ASSERT_TRUE(absl::holds_alternative<std::unique_ptr<const int>>(var));
>+  ASSERT_NE(absl::get<std::unique_ptr<const int>>(var), nullptr);
>+  EXPECT_EQ(0, *absl::get<std::unique_ptr<const int>>(var));
>+
>+  var =
>+      ConvertVariantTo<Variant>(OtherVariant(absl::make_unique<std::string>("foo")));
>+  ASSERT_TRUE(absl::holds_alternative<std::unique_ptr<const std::string>>(var));
>+  EXPECT_EQ("foo", *absl::get<std::unique_ptr<const std::string>>(var));
>+}
>+
>+TEST(VariantTest, DoesNotMoveFromLvalues) {
>+  // We use shared_ptr here because it's both copyable and movable, and
>+  // a moved-from shared_ptr is guaranteed to be null, so we can detect
>+  // whether moving or copying has occurred.
>+  using Variant =
>+      variant<std::shared_ptr<const int>, std::shared_ptr<const std::string>>;
>+  using OtherVariant = variant<std::shared_ptr<int>, std::shared_ptr<std::string>>;
>+
>+  Variant v1(std::make_shared<const int>(0));
>+
>+  // Test copy constructor
>+  Variant v2(v1);
>+  EXPECT_EQ(absl::get<std::shared_ptr<const int>>(v1),
>+            absl::get<std::shared_ptr<const int>>(v2));
>+
>+  // Test copy-assignment operator
>+  v1 = std::make_shared<const std::string>("foo");
>+  v2 = v1;
>+  EXPECT_EQ(absl::get<std::shared_ptr<const std::string>>(v1),
>+            absl::get<std::shared_ptr<const std::string>>(v2));
>+
>+  // Test converting copy constructor
>+  OtherVariant other(std::make_shared<int>(0));
>+  Variant v3(ConvertVariantTo<Variant>(other));
>+  EXPECT_EQ(absl::get<std::shared_ptr<int>>(other),
>+            absl::get<std::shared_ptr<const int>>(v3));
>+
>+  other = std::make_shared<std::string>("foo");
>+  v3 = ConvertVariantTo<Variant>(other);
>+  EXPECT_EQ(absl::get<std::shared_ptr<std::string>>(other),
>+            absl::get<std::shared_ptr<const std::string>>(v3));
>+}
>+
>+TEST(VariantTest, TestRvalueConversionViaConvertVariantTo) {
>+  variant<double, std::string> var(
>+      ConvertVariantTo<variant<double, std::string>>(variant<std::string, int>(3)));
>+  EXPECT_THAT(absl::get_if<double>(&var), Pointee(3.0));
>+
>+  var = ConvertVariantTo<variant<double, std::string>>(
>+      variant<const char*, float>("foo"));
>+  EXPECT_THAT(absl::get_if<std::string>(&var), Pointee(std::string("foo")));
>+
>+  variant<double> singleton(
>+      ConvertVariantTo<variant<double>>(variant<int, float>(42)));
>+  EXPECT_THAT(absl::get_if<double>(&singleton), Pointee(42.0));
>+
>+  singleton = ConvertVariantTo<variant<double>>(variant<int, float>(3.14f));
>+  EXPECT_THAT(absl::get_if<double>(&singleton), Pointee(DoubleEq(3.14f)));
>+
>+  singleton = ConvertVariantTo<variant<double>>(variant<int>(3));
>+  EXPECT_THAT(absl::get_if<double>(&singleton), Pointee(3.0));
>+
>+  variant<int32_t, uint32_t> variant2(
>+      ConvertVariantTo<variant<int32_t, uint32_t>>(variant<int32_t>(42)));
>+  EXPECT_THAT(absl::get_if<int32_t>(&variant2), Pointee(42));
>+
>+  variant2 = ConvertVariantTo<variant<int32_t, uint32_t>>(variant<uint32_t>(42));
>+  EXPECT_THAT(absl::get_if<uint32_t>(&variant2), Pointee(42));
>+
>+  variant<Convertible1, Convertible2> variant3(
>+      ConvertVariantTo<variant<Convertible1, Convertible2>>(
>+          (variant<Convertible2, Convertible1>(Convertible1()))));
>+  ASSERT_TRUE(absl::holds_alternative<Convertible1>(variant3));
>+
>+  variant3 = ConvertVariantTo<variant<Convertible1, Convertible2>>(
>+      variant<Convertible2, Convertible1>(Convertible2()));
>+  ASSERT_TRUE(absl::holds_alternative<Convertible2>(variant3));
>+}
>+
>+TEST(VariantTest, TestLvalueConversionViaConvertVariantTo) {
>+  variant<std::string, int> source1 = 3;
>+  variant<double, std::string> destination(
>+      ConvertVariantTo<variant<double, std::string>>(source1));
>+  EXPECT_THAT(absl::get_if<double>(&destination), Pointee(3.0));
>+
>+  variant<const char*, float> source2 = "foo";
>+  destination = ConvertVariantTo<variant<double, std::string>>(source2);
>+  EXPECT_THAT(absl::get_if<std::string>(&destination), Pointee(std::string("foo")));
>+
>+  variant<int, float> source3(42);
>+  variant<double> singleton(ConvertVariantTo<variant<double>>(source3));
>+  EXPECT_THAT(absl::get_if<double>(&singleton), Pointee(42.0));
>+
>+  source3 = 3.14f;
>+  singleton = ConvertVariantTo<variant<double>>(source3);
>+  EXPECT_FLOAT_EQ(3.14f, static_cast<float>(absl::get<double>(singleton)));
>+  EXPECT_THAT(absl::get_if<double>(&singleton), Pointee(DoubleEq(3.14f)));
>+
>+  variant<int> source4(3);
>+  singleton = ConvertVariantTo<variant<double>>(source4);
>+  EXPECT_THAT(absl::get_if<double>(&singleton), Pointee(3.0));
>+
>+  variant<int32_t> source5(42);
>+  variant<int32_t, uint32_t> variant2(
>+      ConvertVariantTo<variant<int32_t, uint32_t>>(source5));
>+  EXPECT_THAT(absl::get_if<int32_t>(&variant2), Pointee(42));
>+
>+  variant<uint32_t> source6(42);
>+  variant2 = ConvertVariantTo<variant<int32_t, uint32_t>>(source6);
>+  EXPECT_THAT(absl::get_if<uint32_t>(&variant2), Pointee(42));
>+
>+  variant<Convertible2, Convertible1> source7((Convertible1()));
>+  variant<Convertible1, Convertible2> variant3(
>+      ConvertVariantTo<variant<Convertible1, Convertible2>>(source7));
>+  ASSERT_TRUE(absl::holds_alternative<Convertible1>(variant3));
>+
>+  source7 = Convertible2();
>+  variant3 = ConvertVariantTo<variant<Convertible1, Convertible2>>(source7);
>+  ASSERT_TRUE(absl::holds_alternative<Convertible2>(variant3));
>+}
>+
>+TEST(VariantTest, TestMoveConversionViaConvertVariantTo) {
>+  using Variant =
>+      variant<std::unique_ptr<const int>, std::unique_ptr<const std::string>>;
>+  using OtherVariant = variant<std::unique_ptr<int>, std::unique_ptr<std::string>>;
>+
>+  Variant var(
>+      ConvertVariantTo<Variant>(OtherVariant{absl::make_unique<int>(3)}));
>+  EXPECT_THAT(absl::get_if<std::unique_ptr<const int>>(&var),
>+              Pointee(Pointee(3)));
>+
>+  var =
>+      ConvertVariantTo<Variant>(OtherVariant(absl::make_unique<std::string>("foo")));
>+  EXPECT_THAT(absl::get_if<std::unique_ptr<const std::string>>(&var),
>+              Pointee(Pointee(std::string("foo"))));
>+}
>+
>+// If all alternatives are trivially copy/move constructible, variant should
>+// also be trivially copy/move constructible. This is not required by the
>+// standard and we know that libstdc++ variant doesn't have this feature.
>+// For more details see the paper:
>+// http://open-std.org/JTC1/SC22/WG21/docs/papers/2017/p0602r0.html
>+#if !(defined(ABSL_HAVE_STD_VARIANT) && defined(__GLIBCXX__))
>+#define ABSL_VARIANT_PROPAGATE_COPY_MOVE_TRIVIALITY 1
>+#endif
>+
>+TEST(VariantTest, TestCopyAndMoveTypeTraits) {
>+  EXPECT_TRUE(std::is_copy_constructible<variant<std::string>>::value);
>+  EXPECT_TRUE(std::is_copy_assignable<variant<std::string>>::value);
>+  EXPECT_TRUE(std::is_move_constructible<variant<std::string>>::value);
>+  EXPECT_TRUE(std::is_move_assignable<variant<std::string>>::value);
>+  EXPECT_TRUE(std::is_move_constructible<variant<std::unique_ptr<int>>>::value);
>+  EXPECT_TRUE(std::is_move_assignable<variant<std::unique_ptr<int>>>::value);
>+  EXPECT_FALSE(
>+      std::is_copy_constructible<variant<std::unique_ptr<int>>>::value);
>+  EXPECT_FALSE(std::is_copy_assignable<variant<std::unique_ptr<int>>>::value);
>+
>+  EXPECT_FALSE(
>+      absl::is_trivially_copy_constructible<variant<std::string>>::value);
>+  EXPECT_FALSE(absl::is_trivially_copy_assignable<variant<std::string>>::value);
>+#if ABSL_VARIANT_PROPAGATE_COPY_MOVE_TRIVIALITY
>+  EXPECT_TRUE(absl::is_trivially_copy_constructible<variant<int>>::value);
>+  EXPECT_TRUE(absl::is_trivially_copy_assignable<variant<int>>::value);
>+  EXPECT_TRUE(is_trivially_move_constructible<variant<int>>::value);
>+  EXPECT_TRUE(is_trivially_move_assignable<variant<int>>::value);
>+#endif  // ABSL_VARIANT_PROPAGATE_COPY_MOVE_TRIVIALITY
>+}
>+
>+TEST(VariantTest, TestVectorOfMoveonlyVariant) {
>+  // Verify that variant<MoveonlyType> works correctly as a std::vector element.
>+  std::vector<variant<std::unique_ptr<int>, std::string>> vec;
>+  vec.push_back(absl::make_unique<int>(42));
>+  vec.emplace_back("Hello");
>+  vec.reserve(3);
>+  auto another_vec = absl::move(vec);
>+  // As a sanity check, verify vector contents.
>+  ASSERT_EQ(2, another_vec.size());
>+  EXPECT_EQ(42, *absl::get<std::unique_ptr<int>>(another_vec[0]));
>+  EXPECT_EQ("Hello", absl::get<std::string>(another_vec[1]));
>+}
>+
>+TEST(VariantTest, NestedVariant) {
>+#if ABSL_VARIANT_PROPAGATE_COPY_MOVE_TRIVIALITY
>+  static_assert(absl::is_trivially_copy_constructible<variant<int>>(), "");
>+  static_assert(absl::is_trivially_copy_assignable<variant<int>>(), "");
>+  static_assert(is_trivially_move_constructible<variant<int>>(), "");
>+  static_assert(is_trivially_move_assignable<variant<int>>(), "");
>+
>+  static_assert(absl::is_trivially_copy_constructible<variant<variant<int>>>(),
>+                "");
>+  static_assert(absl::is_trivially_copy_assignable<variant<variant<int>>>(),
>+                "");
>+  static_assert(is_trivially_move_constructible<variant<variant<int>>>(), "");
>+  static_assert(is_trivially_move_assignable<variant<variant<int>>>(), "");
>+#endif  // ABSL_VARIANT_PROPAGATE_COPY_MOVE_TRIVIALITY
>+
>+  variant<int> x(42);
>+  variant<variant<int>> y(x);
>+  variant<variant<int>> z(y);
>+  EXPECT_TRUE(absl::holds_alternative<variant<int>>(z));
>+  EXPECT_EQ(x, absl::get<variant<int>>(z));
>+}
>+
>+struct TriviallyDestructible {
>+  TriviallyDestructible(TriviallyDestructible&&) {}
>+  TriviallyDestructible(const TriviallyDestructible&) {}
>+  TriviallyDestructible& operator=(TriviallyDestructible&&) { return *this; }
>+  TriviallyDestructible& operator=(const TriviallyDestructible&) {
>+    return *this;
>+  }
>+};
>+
>+struct TriviallyMovable {
>+  TriviallyMovable(TriviallyMovable&&) = default;
>+  TriviallyMovable(TriviallyMovable const&) {}
>+  TriviallyMovable& operator=(const TriviallyMovable&) { return *this; }
>+};
>+
>+struct TriviallyCopyable {
>+  TriviallyCopyable(const TriviallyCopyable&) = default;
>+  TriviallyCopyable& operator=(const TriviallyCopyable&) { return *this; }
>+};
>+
>+struct TriviallyMoveAssignable {
>+  TriviallyMoveAssignable(TriviallyMoveAssignable&&) = default;
>+  TriviallyMoveAssignable(const TriviallyMoveAssignable&) {}
>+  TriviallyMoveAssignable& operator=(TriviallyMoveAssignable&&) = default;
>+  TriviallyMoveAssignable& operator=(const TriviallyMoveAssignable&) {
>+    return *this;
>+  }
>+};
>+
>+struct TriviallyCopyAssignable {};
>+
>+#if ABSL_VARIANT_PROPAGATE_COPY_MOVE_TRIVIALITY
>+TEST(VariantTest, TestTriviality) {
>+  {
>+    using TrivDestVar = absl::variant<TriviallyDestructible>;
>+
>+    EXPECT_FALSE(is_trivially_move_constructible<TrivDestVar>::value);
>+    EXPECT_FALSE(absl::is_trivially_copy_constructible<TrivDestVar>::value);
>+    EXPECT_FALSE(is_trivially_move_assignable<TrivDestVar>::value);
>+    EXPECT_FALSE(absl::is_trivially_copy_assignable<TrivDestVar>::value);
>+    EXPECT_TRUE(absl::is_trivially_destructible<TrivDestVar>::value);
>+  }
>+
>+  {
>+    using TrivMoveVar = absl::variant<TriviallyMovable>;
>+
>+    EXPECT_TRUE(is_trivially_move_constructible<TrivMoveVar>::value);
>+    EXPECT_FALSE(absl::is_trivially_copy_constructible<TrivMoveVar>::value);
>+    EXPECT_FALSE(is_trivially_move_assignable<TrivMoveVar>::value);
>+    EXPECT_FALSE(absl::is_trivially_copy_assignable<TrivMoveVar>::value);
>+    EXPECT_TRUE(absl::is_trivially_destructible<TrivMoveVar>::value);
>+  }
>+
>+  {
>+    using TrivCopyVar = absl::variant<TriviallyCopyable>;
>+
>+    EXPECT_TRUE(is_trivially_move_constructible<TrivCopyVar>::value);
>+    EXPECT_TRUE(absl::is_trivially_copy_constructible<TrivCopyVar>::value);
>+    EXPECT_FALSE(is_trivially_move_assignable<TrivCopyVar>::value);
>+    EXPECT_FALSE(absl::is_trivially_copy_assignable<TrivCopyVar>::value);
>+    EXPECT_TRUE(absl::is_trivially_destructible<TrivCopyVar>::value);
>+  }
>+
>+  {
>+    using TrivMoveAssignVar = absl::variant<TriviallyMoveAssignable>;
>+
>+    EXPECT_TRUE(is_trivially_move_constructible<TrivMoveAssignVar>::value);
>+    EXPECT_FALSE(
>+        absl::is_trivially_copy_constructible<TrivMoveAssignVar>::value);
>+    EXPECT_TRUE(is_trivially_move_assignable<TrivMoveAssignVar>::value);
>+    EXPECT_FALSE(absl::is_trivially_copy_assignable<TrivMoveAssignVar>::value);
>+    EXPECT_TRUE(absl::is_trivially_destructible<TrivMoveAssignVar>::value);
>+  }
>+
>+  {
>+    using TrivCopyAssignVar = absl::variant<TriviallyCopyAssignable>;
>+
>+    EXPECT_TRUE(is_trivially_move_constructible<TrivCopyAssignVar>::value);
>+    EXPECT_TRUE(
>+        absl::is_trivially_copy_constructible<TrivCopyAssignVar>::value);
>+    EXPECT_TRUE(is_trivially_move_assignable<TrivCopyAssignVar>::value);
>+    EXPECT_TRUE(absl::is_trivially_copy_assignable<TrivCopyAssignVar>::value);
>+    EXPECT_TRUE(absl::is_trivially_destructible<TrivCopyAssignVar>::value);
>+  }
>+}
>+#endif  // ABSL_VARIANT_PROPAGATE_COPY_MOVE_TRIVIALITY
>+
>+// To verify that absl::variant correctly use the nontrivial move ctor of its
>+// member rather than use the trivial copy constructor.
>+TEST(VariantTest, MoveCtorBug) {
>+  // To simulate std::tuple in libstdc++.
>+  struct TrivialCopyNontrivialMove {
>+    TrivialCopyNontrivialMove() = default;
>+    TrivialCopyNontrivialMove(const TrivialCopyNontrivialMove&) = default;
>+    TrivialCopyNontrivialMove(TrivialCopyNontrivialMove&&) { called = true; }
>+    bool called = false;
>+  };
>+  {
>+    using V = absl::variant<TrivialCopyNontrivialMove, int>;
>+    V v1(absl::in_place_index_t<0>{});
>+    // this should invoke the move ctor, rather than the trivial copy ctor.
>+    V v2(std::move(v1));
>+    EXPECT_TRUE(absl::get<0>(v2).called);
>+  }
>+  {
>+    // this case failed to compile before our fix due to a GCC bug.
>+    using V = absl::variant<int, TrivialCopyNontrivialMove>;
>+    V v1(absl::in_place_index_t<1>{});
>+    // this should invoke the move ctor, rather than the trivial copy ctor.
>+    V v2(std::move(v1));
>+    EXPECT_TRUE(absl::get<1>(v2).called);
>+  }
>+}
>+
>+}  // namespace
>+}  // namespace absl
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/utility/BUILD.bazel b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/utility/BUILD.bazel
>new file mode 100644
>index 00000000000..c01b49bc97d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/utility/BUILD.bazel
>@@ -0,0 +1,33 @@
>+load(
>+    "//absl:copts.bzl",
>+    "ABSL_DEFAULT_COPTS",
>+    "ABSL_TEST_COPTS",
>+)
>+
>+package(default_visibility = ["//visibility:public"])
>+
>+licenses(["notice"])  # Apache 2.0
>+
>+cc_library(
>+    name = "utility",
>+    hdrs = ["utility.h"],
>+    copts = ABSL_DEFAULT_COPTS,
>+    deps = [
>+        "//absl/base:base_internal",
>+        "//absl/base:config",
>+        "//absl/meta:type_traits",
>+    ],
>+)
>+
>+cc_test(
>+    name = "utility_test",
>+    srcs = ["utility_test.cc"],
>+    copts = ABSL_TEST_COPTS,
>+    deps = [
>+        ":utility",
>+        "//absl/base:core_headers",
>+        "//absl/memory",
>+        "//absl/strings",
>+        "@com_google_googletest//:gtest_main",
>+    ],
>+)
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/utility/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/utility/BUILD.gn
>new file mode 100644
>index 00000000000..4cf3c810e65
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/utility/BUILD.gn
>@@ -0,0 +1,32 @@
>+# Copyright 2018 The Chromium Authors. All rights reserved.
>+# Use of this source code is governed by a BSD-style license that can be
>+# found in the LICENSE file.
>+
>+import("//build_overrides/build.gni")
>+
>+if (build_with_chromium) {
>+  visibility = [
>+    "//third_party/webrtc/*",
>+    "//third_party/abseil-cpp/*",
>+    "//third_party/googletest:gtest",
>+  ]
>+} else {
>+  visibility = [ "*" ]
>+}
>+
>+source_set("utility") {
>+  configs -= [ "//build/config/compiler:chromium_code" ]
>+  configs += [
>+    "//build/config/compiler:no_chromium_code",
>+    "//third_party/abseil-cpp:absl_default_cflags_cc",
>+  ]
>+  public_configs = [ "//third_party/abseil-cpp:absl_include_config" ]
>+  public = [
>+    "utility.h",
>+  ]
>+  deps = [
>+    "../base:base_internal",
>+    "../base:config",
>+    "../meta:type_traits",
>+  ]
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/utility/CMakeLists.txt b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/utility/CMakeLists.txt
>new file mode 100644
>index 00000000000..dc3a6319054
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/utility/CMakeLists.txt
>@@ -0,0 +1,52 @@
>+#
>+# Copyright 2017 The Abseil Authors.
>+#
>+# Licensed under the Apache License, Version 2.0 (the "License");
>+# you may not use this file except in compliance with the License.
>+# You may obtain a copy of the License at
>+#
>+#      http://www.apache.org/licenses/LICENSE-2.0
>+#
>+# Unless required by applicable law or agreed to in writing, software
>+# distributed under the License is distributed on an "AS IS" BASIS,
>+# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+# See the License for the specific language governing permissions and
>+# limitations under the License.
>+#
>+
>+
>+list(APPEND UTILITY_PUBLIC_HEADERS
>+  "utility.h"
>+)
>+
>+absl_header_library(
>+  TARGET
>+    absl_utility
>+  PUBLIC_LIBRARIES
>+    absl::base
>+  EXPORT_NAME
>+    utility
>+)
>+
>+
>+#
>+## TESTS
>+#
>+
>+# test utility_test
>+set(UTILITY_TEST_SRC "utility_test.cc")
>+set(UTILITY_TEST_PUBLIC_LIBRARIES
>+  absl::base
>+  absl::memory
>+  absl::strings
>+  absl::utility
>+)
>+
>+absl_test(
>+  TARGET
>+    utility_test
>+  SOURCES
>+    ${UTILITY_TEST_SRC}
>+  PUBLIC_LIBRARIES
>+    ${UTILITY_TEST_PUBLIC_LIBRARIES}
>+)
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/utility/utility.h b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/utility/utility.h
>new file mode 100644
>index 00000000000..d73602c47d3
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/utility/utility.h
>@@ -0,0 +1,291 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+//
>+// This header file contains C++11 versions of standard <utility> header
>+// abstractions available within C++14 and C++17, and are designed to be drop-in
>+// replacement for code compliant with C++14 and C++17.
>+//
>+// The following abstractions are defined:
>+//
>+//   * integer_sequence<T, Ints...>  == std::integer_sequence<T, Ints...>
>+//   * index_sequence<Ints...>       == std::index_sequence<Ints...>
>+//   * make_integer_sequence<T, N>   == std::make_integer_sequence<T, N>
>+//   * make_index_sequence<N>        == std::make_index_sequence<N>
>+//   * index_sequence_for<Ts...>     == std::index_sequence_for<Ts...>
>+//   * apply<Functor, Tuple>         == std::apply<Functor, Tuple>
>+//   * exchange<T>                   == std::exchange<T>
>+//
>+// This header file also provides the tag types `in_place_t`, `in_place_type_t`,
>+// and `in_place_index_t`, as well as the constant `in_place`, and
>+// `constexpr` `std::move()` and `std::forward()` implementations in C++11.
>+//
>+// References:
>+//
>+//  http://en.cppreference.com/w/cpp/utility/integer_sequence
>+//  http://en.cppreference.com/w/cpp/utility/apply
>+//  http://www.open-std.org/jtc1/sc22/wg21/docs/papers/2013/n3658.html
>+//
>+
>+#ifndef ABSL_UTILITY_UTILITY_H_
>+#define ABSL_UTILITY_UTILITY_H_
>+
>+#include <cstddef>
>+#include <cstdlib>
>+#include <tuple>
>+#include <utility>
>+
>+#include "absl/base/config.h"
>+#include "absl/base/internal/inline_variable.h"
>+#include "absl/base/internal/invoke.h"
>+#include "absl/meta/type_traits.h"
>+
>+namespace absl {
>+
>+// integer_sequence
>+//
>+// Class template representing a compile-time integer sequence. An instantiation
>+// of `integer_sequence<T, Ints...>` has a sequence of integers encoded in its
>+// type through its template arguments (which is a common need when
>+// working with C++11 variadic templates). `absl::integer_sequence` is designed
>+// to be a drop-in replacement for C++14's `std::integer_sequence`.
>+//
>+// Example:
>+//
>+//   template< class T, T... Ints >
>+//   void user_function(integer_sequence<T, Ints...>);
>+//
>+//   int main()
>+//   {
>+//     // user_function's `T` will be deduced to `int` and `Ints...`
>+//     // will be deduced to `0, 1, 2, 3, 4`.
>+//     user_function(make_integer_sequence<int, 5>());
>+//   }
>+template <typename T, T... Ints>
>+struct integer_sequence {
>+  using value_type = T;
>+  static constexpr size_t size() noexcept { return sizeof...(Ints); }
>+};
>+
>+// index_sequence
>+//
>+// A helper template for an `integer_sequence` of `size_t`,
>+// `absl::index_sequence` is designed to be a drop-in replacement for C++14's
>+// `std::index_sequence`.
>+template <size_t... Ints>
>+using index_sequence = integer_sequence<size_t, Ints...>;
>+
>+namespace utility_internal {
>+
>+template <typename Seq, size_t SeqSize, size_t Rem>
>+struct Extend;
>+
>+// Note that SeqSize == sizeof...(Ints). It's passed explicitly for efficiency.
>+template <typename T, T... Ints, size_t SeqSize>
>+struct Extend<integer_sequence<T, Ints...>, SeqSize, 0> {
>+  using type = integer_sequence<T, Ints..., (Ints + SeqSize)...>;
>+};
>+
>+template <typename T, T... Ints, size_t SeqSize>
>+struct Extend<integer_sequence<T, Ints...>, SeqSize, 1> {
>+  using type = integer_sequence<T, Ints..., (Ints + SeqSize)..., 2 * SeqSize>;
>+};
>+
>+// Recursion helper for 'make_integer_sequence<T, N>'.
>+// 'Gen<T, N>::type' is an alias for 'integer_sequence<T, 0, 1, ... N-1>'.
>+template <typename T, size_t N>
>+struct Gen {
>+  using type =
>+      typename Extend<typename Gen<T, N / 2>::type, N / 2, N % 2>::type;
>+};
>+
>+template <typename T>
>+struct Gen<T, 0> {
>+  using type = integer_sequence<T>;
>+};
>+
>+}  // namespace utility_internal
>+
>+// Compile-time sequences of integers
>+
>+// make_integer_sequence
>+//
>+// This template alias is equivalent to
>+// `integer_sequence<int, 0, 1, ..., N-1>`, and is designed to be a drop-in
>+// replacement for C++14's `std::make_integer_sequence`.
>+template <typename T, T N>
>+using make_integer_sequence = typename utility_internal::Gen<T, N>::type;
>+
>+// make_index_sequence
>+//
>+// This template alias is equivalent to `index_sequence<0, 1, ..., N-1>`,
>+// and is designed to be a drop-in replacement for C++14's
>+// `std::make_index_sequence`.
>+template <size_t N>
>+using make_index_sequence = make_integer_sequence<size_t, N>;
>+
>+// index_sequence_for
>+//
>+// Converts a typename pack into an index sequence of the same length, and
>+// is designed to be a drop-in replacement for C++14's
>+// `std::index_sequence_for()`
>+template <typename... Ts>
>+using index_sequence_for = make_index_sequence<sizeof...(Ts)>;
>+
>+// Tag types
>+
>+#ifdef ABSL_HAVE_STD_OPTIONAL
>+
>+using std::in_place_t;
>+using std::in_place;
>+
>+#else  // ABSL_HAVE_STD_OPTIONAL
>+
>+// in_place_t
>+//
>+// Tag type used to specify in-place construction, such as with
>+// `absl::optional`, designed to be a drop-in replacement for C++17's
>+// `std::in_place_t`.
>+struct in_place_t {};
>+
>+ABSL_INTERNAL_INLINE_CONSTEXPR(in_place_t, in_place, {});
>+
>+#endif  // ABSL_HAVE_STD_OPTIONAL
>+
>+#if defined(ABSL_HAVE_STD_ANY) || defined(ABSL_HAVE_STD_VARIANT)
>+using std::in_place_type_t;
>+#else
>+
>+// in_place_type_t
>+//
>+// Tag type used for in-place construction when the type to construct needs to
>+// be specified, such as with `absl::any`, designed to be a drop-in replacement
>+// for C++17's `std::in_place_type_t`.
>+template <typename T>
>+struct in_place_type_t {};
>+#endif  // ABSL_HAVE_STD_ANY || ABSL_HAVE_STD_VARIANT
>+
>+#ifdef ABSL_HAVE_STD_VARIANT
>+using std::in_place_index_t;
>+#else
>+
>+// in_place_index_t
>+//
>+// Tag type used for in-place construction when the type to construct needs to
>+// be specified, such as with `absl::any`, designed to be a drop-in replacement
>+// for C++17's `std::in_place_index_t`.
>+template <size_t I>
>+struct in_place_index_t {};
>+#endif  // ABSL_HAVE_STD_VARIANT
>+
>+// Constexpr move and forward
>+
>+// move()
>+//
>+// A constexpr version of `std::move()`, designed to be a drop-in replacement
>+// for C++14's `std::move()`.
>+template <typename T>
>+constexpr absl::remove_reference_t<T>&& move(T&& t) noexcept {
>+  return static_cast<absl::remove_reference_t<T>&&>(t);
>+}
>+
>+// forward()
>+//
>+// A constexpr version of `std::forward()`, designed to be a drop-in replacement
>+// for C++14's `std::forward()`.
>+template <typename T>
>+constexpr T&& forward(
>+    absl::remove_reference_t<T>& t) noexcept {  // NOLINT(runtime/references)
>+  return static_cast<T&&>(t);
>+}
>+
>+namespace utility_internal {
>+// Helper method for expanding tuple into a called method.
>+template <typename Functor, typename Tuple, std::size_t... Indexes>
>+auto apply_helper(Functor&& functor, Tuple&& t, index_sequence<Indexes...>)
>+    -> decltype(absl::base_internal::Invoke(
>+        absl::forward<Functor>(functor),
>+        std::get<Indexes>(absl::forward<Tuple>(t))...)) {
>+  return absl::base_internal::Invoke(
>+      absl::forward<Functor>(functor),
>+      std::get<Indexes>(absl::forward<Tuple>(t))...);
>+}
>+
>+}  // namespace utility_internal
>+
>+// apply
>+//
>+// Invokes a Callable using elements of a tuple as its arguments.
>+// Each element of the tuple corresponds to an argument of the call (in order).
>+// Both the Callable argument and the tuple argument are perfect-forwarded.
>+// For member-function Callables, the first tuple element acts as the `this`
>+// pointer. `absl::apply` is designed to be a drop-in replacement for C++17's
>+// `std::apply`. Unlike C++17's `std::apply`, this is not currently `constexpr`.
>+//
>+// Example:
>+//
>+//   class Foo{void Bar(int);};
>+//   void user_function(int, std::string);
>+//   void user_function(std::unique_ptr<Foo>);
>+//
>+//   int main()
>+//   {
>+//       std::tuple<int, std::string> tuple1(42, "bar");
>+//       // Invokes the user function overload on int, std::string.
>+//       absl::apply(&user_function, tuple1);
>+//
>+//       auto foo = absl::make_unique<Foo>();
>+//       std::tuple<Foo*, int> tuple2(foo.get(), 42);
>+//       // Invokes the method Bar on foo with one argument 42.
>+//       absl::apply(&Foo::Bar, foo.get(), 42);
>+//
>+//       std::tuple<std::unique_ptr<Foo>> tuple3(absl::make_unique<Foo>());
>+//       // Invokes the user function that takes ownership of the unique
>+//       // pointer.
>+//       absl::apply(&user_function, std::move(tuple));
>+//   }
>+template <typename Functor, typename Tuple>
>+auto apply(Functor&& functor, Tuple&& t)
>+    -> decltype(utility_internal::apply_helper(
>+        absl::forward<Functor>(functor), absl::forward<Tuple>(t),
>+        absl::make_index_sequence<std::tuple_size<
>+            typename std::remove_reference<Tuple>::type>::value>{})) {
>+  return utility_internal::apply_helper(
>+      absl::forward<Functor>(functor), absl::forward<Tuple>(t),
>+      absl::make_index_sequence<std::tuple_size<
>+          typename std::remove_reference<Tuple>::type>::value>{});
>+}
>+
>+// exchange
>+//
>+// Replaces the value of `obj` with `new_value` and returns the old value of
>+// `obj`.  `absl::exchange` is designed to be a drop-in replacement for C++14's
>+// `std::exchange`.
>+//
>+// Example:
>+//
>+//   Foo& operator=(Foo&& other) {
>+//     ptr1_ = absl::exchange(other.ptr1_, nullptr);
>+//     int1_ = absl::exchange(other.int1_, -1);
>+//     return *this;
>+//   }
>+template <typename T, typename U = T>
>+T exchange(T& obj, U&& new_value) {
>+  T old_value = absl::move(obj);
>+  obj = absl::forward<U>(new_value);
>+  return old_value;
>+}
>+
>+}  // namespace absl
>+
>+#endif  // ABSL_UTILITY_UTILITY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/utility/utility_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/utility/utility_test.cc
>new file mode 100644
>index 00000000000..3c447b20bb0
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/absl/utility/utility_test.cc
>@@ -0,0 +1,345 @@
>+// Copyright 2017 The Abseil Authors.
>+//
>+// Licensed under the Apache License, Version 2.0 (the "License");
>+// you may not use this file except in compliance with the License.
>+// You may obtain a copy of the License at
>+//
>+//      http://www.apache.org/licenses/LICENSE-2.0
>+//
>+// Unless required by applicable law or agreed to in writing, software
>+// distributed under the License is distributed on an "AS IS" BASIS,
>+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
>+// See the License for the specific language governing permissions and
>+// limitations under the License.
>+
>+#include "absl/utility/utility.h"
>+
>+#include <sstream>
>+#include <string>
>+#include <tuple>
>+#include <type_traits>
>+#include <vector>
>+
>+#include "gmock/gmock.h"
>+#include "gtest/gtest.h"
>+#include "absl/base/attributes.h"
>+#include "absl/memory/memory.h"
>+#include "absl/strings/str_cat.h"
>+
>+namespace {
>+
>+#ifdef _MSC_VER
>+// Warnings for unused variables in this test are false positives.  On other
>+// platforms, they are suppressed by ABSL_ATTRIBUTE_UNUSED, but that doesn't
>+// work on MSVC.
>+// Both the unused variables and the name length warnings are due to calls
>+// to absl::make_index_sequence with very large values, creating very long type
>+// names. The resulting warnings are so long they make build output unreadable.
>+#pragma warning( push )
>+#pragma warning( disable : 4503 )  // decorated name length exceeded
>+#pragma warning( disable : 4101 )  // unreferenced local variable
>+#endif  // _MSC_VER
>+
>+using ::testing::ElementsAre;
>+using ::testing::Pointee;
>+using ::testing::StaticAssertTypeEq;
>+
>+TEST(IntegerSequenceTest, ValueType) {
>+  StaticAssertTypeEq<int, absl::integer_sequence<int>::value_type>();
>+  StaticAssertTypeEq<char, absl::integer_sequence<char>::value_type>();
>+}
>+
>+TEST(IntegerSequenceTest, Size) {
>+  EXPECT_EQ(0, (absl::integer_sequence<int>::size()));
>+  EXPECT_EQ(1, (absl::integer_sequence<int, 0>::size()));
>+  EXPECT_EQ(1, (absl::integer_sequence<int, 1>::size()));
>+  EXPECT_EQ(2, (absl::integer_sequence<int, 1, 2>::size()));
>+  EXPECT_EQ(3, (absl::integer_sequence<int, 0, 1, 2>::size()));
>+  EXPECT_EQ(3, (absl::integer_sequence<int, -123, 123, 456>::size()));
>+  constexpr size_t sz = absl::integer_sequence<int, 0, 1>::size();
>+  EXPECT_EQ(2, sz);
>+}
>+
>+TEST(IntegerSequenceTest, MakeIndexSequence) {
>+  StaticAssertTypeEq<absl::index_sequence<>, absl::make_index_sequence<0>>();
>+  StaticAssertTypeEq<absl::index_sequence<0>, absl::make_index_sequence<1>>();
>+  StaticAssertTypeEq<absl::index_sequence<0, 1>,
>+                     absl::make_index_sequence<2>>();
>+  StaticAssertTypeEq<absl::index_sequence<0, 1, 2>,
>+                     absl::make_index_sequence<3>>();
>+}
>+
>+TEST(IntegerSequenceTest, MakeIntegerSequence) {
>+  StaticAssertTypeEq<absl::integer_sequence<int>,
>+                     absl::make_integer_sequence<int, 0>>();
>+  StaticAssertTypeEq<absl::integer_sequence<int, 0>,
>+                     absl::make_integer_sequence<int, 1>>();
>+  StaticAssertTypeEq<absl::integer_sequence<int, 0, 1>,
>+                     absl::make_integer_sequence<int, 2>>();
>+  StaticAssertTypeEq<absl::integer_sequence<int, 0, 1, 2>,
>+                     absl::make_integer_sequence<int, 3>>();
>+}
>+
>+template <typename... Ts>
>+class Counter {};
>+
>+template <size_t... Is>
>+void CountAll(absl::index_sequence<Is...>) {
>+  // We only need an alias here, but instantiate a variable to silence warnings
>+  // for unused typedefs in some compilers.
>+  ABSL_ATTRIBUTE_UNUSED Counter<absl::make_index_sequence<Is>...> seq;
>+}
>+
>+// This test verifies that absl::make_index_sequence can handle large arguments
>+// without blowing up template instantiation stack, going OOM or taking forever
>+// to compile (there is hard 15 minutes limit imposed by forge).
>+TEST(IntegerSequenceTest, MakeIndexSequencePerformance) {
>+  // O(log N) template instantiations.
>+  // We only need an alias here, but instantiate a variable to silence warnings
>+  // for unused typedefs in some compilers.
>+  ABSL_ATTRIBUTE_UNUSED absl::make_index_sequence<(1 << 16) - 1> seq;
>+  // O(N) template instantiations.
>+  CountAll(absl::make_index_sequence<(1 << 8) - 1>());
>+}
>+
>+template <typename F, typename Tup, size_t... Is>
>+auto ApplyFromTupleImpl(F f, const Tup& tup, absl::index_sequence<Is...>)
>+    -> decltype(f(std::get<Is>(tup)...)) {
>+  return f(std::get<Is>(tup)...);
>+}
>+
>+template <typename Tup>
>+using TupIdxSeq = absl::make_index_sequence<std::tuple_size<Tup>::value>;
>+
>+template <typename F, typename Tup>
>+auto ApplyFromTuple(F f, const Tup& tup)
>+    -> decltype(ApplyFromTupleImpl(f, tup, TupIdxSeq<Tup>{})) {
>+  return ApplyFromTupleImpl(f, tup, TupIdxSeq<Tup>{});
>+}
>+
>+template <typename T>
>+std::string Fmt(const T& x) {
>+  std::ostringstream os;
>+  os << x;
>+  return os.str();
>+}
>+
>+struct PoorStrCat {
>+  template <typename... Args>
>+  std::string operator()(const Args&... args) const {
>+    std::string r;
>+    for (const auto& e : {Fmt(args)...}) r += e;
>+    return r;
>+  }
>+};
>+
>+template <typename Tup, size_t... Is>
>+std::vector<std::string> TupStringVecImpl(const Tup& tup,
>+                                     absl::index_sequence<Is...>) {
>+  return {Fmt(std::get<Is>(tup))...};
>+}
>+
>+template <typename... Ts>
>+std::vector<std::string> TupStringVec(const std::tuple<Ts...>& tup) {
>+  return TupStringVecImpl(tup, absl::index_sequence_for<Ts...>());
>+}
>+
>+TEST(MakeIndexSequenceTest, ApplyFromTupleExample) {
>+  PoorStrCat f{};
>+  EXPECT_EQ("12abc3.14", f(12, "abc", 3.14));
>+  EXPECT_EQ("12abc3.14", ApplyFromTuple(f, std::make_tuple(12, "abc", 3.14)));
>+}
>+
>+TEST(IndexSequenceForTest, Basic) {
>+  StaticAssertTypeEq<absl::index_sequence<>, absl::index_sequence_for<>>();
>+  StaticAssertTypeEq<absl::index_sequence<0>, absl::index_sequence_for<int>>();
>+  StaticAssertTypeEq<absl::index_sequence<0, 1, 2, 3>,
>+                     absl::index_sequence_for<int, void, char, int>>();
>+}
>+
>+TEST(IndexSequenceForTest, Example) {
>+  EXPECT_THAT(TupStringVec(std::make_tuple(12, "abc", 3.14)),
>+              ElementsAre("12", "abc", "3.14"));
>+}
>+
>+int Function(int a, int b) { return a - b; }
>+
>+int Sink(std::unique_ptr<int> p) { return *p; }
>+
>+std::unique_ptr<int> Factory(int n) { return absl::make_unique<int>(n); }
>+
>+void NoOp() {}
>+
>+struct ConstFunctor {
>+  int operator()(int a, int b) const { return a - b; }
>+};
>+
>+struct MutableFunctor {
>+  int operator()(int a, int b) { return a - b; }
>+};
>+
>+struct EphemeralFunctor {
>+  EphemeralFunctor() {}
>+  EphemeralFunctor(const EphemeralFunctor&) {}
>+  EphemeralFunctor(EphemeralFunctor&&) {}
>+  int operator()(int a, int b) && { return a - b; }
>+};
>+
>+struct OverloadedFunctor {
>+  OverloadedFunctor() {}
>+  OverloadedFunctor(const OverloadedFunctor&) {}
>+  OverloadedFunctor(OverloadedFunctor&&) {}
>+  template <typename... Args>
>+  std::string operator()(const Args&... args) & {
>+    return absl::StrCat("&", args...);
>+  }
>+  template <typename... Args>
>+  std::string operator()(const Args&... args) const& {
>+    return absl::StrCat("const&", args...);
>+  }
>+  template <typename... Args>
>+  std::string operator()(const Args&... args) && {
>+    return absl::StrCat("&&", args...);
>+  }
>+};
>+
>+struct Class {
>+  int Method(int a, int b) { return a - b; }
>+  int ConstMethod(int a, int b) const { return a - b; }
>+
>+  int member;
>+};
>+
>+struct FlipFlop {
>+  int ConstMethod() const { return member; }
>+  FlipFlop operator*() const { return {-member}; }
>+
>+  int member;
>+};
>+
>+TEST(ApplyTest, Function) {
>+  EXPECT_EQ(1, absl::apply(Function, std::make_tuple(3, 2)));
>+  EXPECT_EQ(1, absl::apply(&Function, std::make_tuple(3, 2)));
>+}
>+
>+TEST(ApplyTest, NonCopyableArgument) {
>+  EXPECT_EQ(42, absl::apply(Sink, std::make_tuple(absl::make_unique<int>(42))));
>+}
>+
>+TEST(ApplyTest, NonCopyableResult) {
>+  EXPECT_THAT(absl::apply(Factory, std::make_tuple(42)),
>+              ::testing::Pointee(42));
>+}
>+
>+TEST(ApplyTest, VoidResult) { absl::apply(NoOp, std::tuple<>()); }
>+
>+TEST(ApplyTest, ConstFunctor) {
>+  EXPECT_EQ(1, absl::apply(ConstFunctor(), std::make_tuple(3, 2)));
>+}
>+
>+TEST(ApplyTest, MutableFunctor) {
>+  MutableFunctor f;
>+  EXPECT_EQ(1, absl::apply(f, std::make_tuple(3, 2)));
>+  EXPECT_EQ(1, absl::apply(MutableFunctor(), std::make_tuple(3, 2)));
>+}
>+TEST(ApplyTest, EphemeralFunctor) {
>+  EphemeralFunctor f;
>+  EXPECT_EQ(1, absl::apply(std::move(f), std::make_tuple(3, 2)));
>+  EXPECT_EQ(1, absl::apply(EphemeralFunctor(), std::make_tuple(3, 2)));
>+}
>+TEST(ApplyTest, OverloadedFunctor) {
>+  OverloadedFunctor f;
>+  const OverloadedFunctor& cf = f;
>+
>+  EXPECT_EQ("&", absl::apply(f, std::tuple<>{}));
>+  EXPECT_EQ("& 42", absl::apply(f, std::make_tuple(" 42")));
>+
>+  EXPECT_EQ("const&", absl::apply(cf, std::tuple<>{}));
>+  EXPECT_EQ("const& 42", absl::apply(cf, std::make_tuple(" 42")));
>+
>+  EXPECT_EQ("&&", absl::apply(std::move(f), std::tuple<>{}));
>+  OverloadedFunctor f2;
>+  EXPECT_EQ("&& 42", absl::apply(std::move(f2), std::make_tuple(" 42")));
>+}
>+
>+TEST(ApplyTest, ReferenceWrapper) {
>+  ConstFunctor cf;
>+  MutableFunctor mf;
>+  EXPECT_EQ(1, absl::apply(std::cref(cf), std::make_tuple(3, 2)));
>+  EXPECT_EQ(1, absl::apply(std::ref(cf), std::make_tuple(3, 2)));
>+  EXPECT_EQ(1, absl::apply(std::ref(mf), std::make_tuple(3, 2)));
>+}
>+
>+TEST(ApplyTest, MemberFunction) {
>+  std::unique_ptr<Class> p(new Class);
>+  std::unique_ptr<const Class> cp(new Class);
>+  EXPECT_EQ(
>+      1, absl::apply(&Class::Method,
>+                     std::tuple<std::unique_ptr<Class>&, int, int>(p, 3, 2)));
>+  EXPECT_EQ(1, absl::apply(&Class::Method,
>+                           std::tuple<Class*, int, int>(p.get(), 3, 2)));
>+  EXPECT_EQ(
>+      1, absl::apply(&Class::Method, std::tuple<Class&, int, int>(*p, 3, 2)));
>+
>+  EXPECT_EQ(
>+      1, absl::apply(&Class::ConstMethod,
>+                     std::tuple<std::unique_ptr<Class>&, int, int>(p, 3, 2)));
>+  EXPECT_EQ(1, absl::apply(&Class::ConstMethod,
>+                           std::tuple<Class*, int, int>(p.get(), 3, 2)));
>+  EXPECT_EQ(1, absl::apply(&Class::ConstMethod,
>+                           std::tuple<Class&, int, int>(*p, 3, 2)));
>+
>+  EXPECT_EQ(1, absl::apply(&Class::ConstMethod,
>+                           std::tuple<std::unique_ptr<const Class>&, int, int>(
>+                               cp, 3, 2)));
>+  EXPECT_EQ(1, absl::apply(&Class::ConstMethod,
>+                           std::tuple<const Class*, int, int>(cp.get(), 3, 2)));
>+  EXPECT_EQ(1, absl::apply(&Class::ConstMethod,
>+                           std::tuple<const Class&, int, int>(*cp, 3, 2)));
>+
>+  EXPECT_EQ(1, absl::apply(&Class::Method,
>+                           std::make_tuple(absl::make_unique<Class>(), 3, 2)));
>+  EXPECT_EQ(1, absl::apply(&Class::ConstMethod,
>+                           std::make_tuple(absl::make_unique<Class>(), 3, 2)));
>+  EXPECT_EQ(
>+      1, absl::apply(&Class::ConstMethod,
>+                     std::make_tuple(absl::make_unique<const Class>(), 3, 2)));
>+}
>+
>+TEST(ApplyTest, DataMember) {
>+  std::unique_ptr<Class> p(new Class{42});
>+  std::unique_ptr<const Class> cp(new Class{42});
>+  EXPECT_EQ(
>+      42, absl::apply(&Class::member, std::tuple<std::unique_ptr<Class>&>(p)));
>+  EXPECT_EQ(42, absl::apply(&Class::member, std::tuple<Class&>(*p)));
>+  EXPECT_EQ(42, absl::apply(&Class::member, std::tuple<Class*>(p.get())));
>+
>+  absl::apply(&Class::member, std::tuple<std::unique_ptr<Class>&>(p)) = 42;
>+  absl::apply(&Class::member, std::tuple<Class*>(p.get())) = 42;
>+  absl::apply(&Class::member, std::tuple<Class&>(*p)) = 42;
>+
>+  EXPECT_EQ(42, absl::apply(&Class::member,
>+                            std::tuple<std::unique_ptr<const Class>&>(cp)));
>+  EXPECT_EQ(42, absl::apply(&Class::member, std::tuple<const Class&>(*cp)));
>+  EXPECT_EQ(42,
>+            absl::apply(&Class::member, std::tuple<const Class*>(cp.get())));
>+}
>+
>+TEST(ApplyTest, FlipFlop) {
>+  FlipFlop obj = {42};
>+  // This call could resolve to (obj.*&FlipFlop::ConstMethod)() or
>+  // ((*obj).*&FlipFlop::ConstMethod)(). We verify that it's the former.
>+  EXPECT_EQ(42, absl::apply(&FlipFlop::ConstMethod, std::make_tuple(obj)));
>+  EXPECT_EQ(42, absl::apply(&FlipFlop::member, std::make_tuple(obj)));
>+}
>+
>+TEST(ExchangeTest, MoveOnly) {
>+  auto a = Factory(1);
>+  EXPECT_EQ(1, *a);
>+  auto b = absl::exchange(a, Factory(2));
>+  EXPECT_EQ(2, *a);
>+  EXPECT_EQ(1, *b);
>+}
>+
>+}  // namespace
>+
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/rename_dynamic_annotations.sh b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/rename_dynamic_annotations.sh
>new file mode 100755
>index 00000000000..b39017f9585
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/abseil-cpp/rename_dynamic_annotations.sh
>@@ -0,0 +1,127 @@
>+#!/bin/bash
>+
>+# This script renames all the functions and the macros defined in
>+# absl/base/dynamic_annotations.{h,cc}.
>+#
>+# Chromium's dynamic_annotations live in //base/third_party/dynamic_annotations
>+# and the are in conflict with Abseil's dynamic_annotations (ODR violations and
>+# macro clashing).
>+# In order to avoid problems in Chromium, this copy of Abseil has its own
>+# dynamic_annotations renamed.
>+
>+for w in \
>+  AnnotateBarrierDestroy \
>+  AnnotateBarrierInit \
>+  AnnotateBarrierWaitAfter \
>+  AnnotateBarrierWaitBefore \
>+  AnnotateBenignRace \
>+  AnnotateBenignRaceSized \
>+  AnnotateCondVarSignal \
>+  AnnotateCondVarSignalAll \
>+  AnnotateCondVarWait \
>+  AnnotateEnableRaceDetection \
>+  AnnotateExpectRace \
>+  AnnotateFlushExpectedRaces \
>+  AnnotateFlushState \
>+  AnnotateHappensAfter \
>+  AnnotateHappensBefore \
>+  AnnotateIgnoreReadsBegin \
>+  AnnotateIgnoreReadsEnd \
>+  AnnotateIgnoreSyncBegin \
>+  AnnotateIgnoreSyncEnd \
>+  AnnotateIgnoreWritesBegin \
>+  AnnotateIgnoreWritesEnd \
>+  AnnotateMemoryIsInitialized \
>+  AnnotateMemoryIsUninitialized \
>+  AnnotateMutexIsNotPHB \
>+  AnnotateMutexIsUsedAsCondVar \
>+  AnnotateNewMemory \
>+  AnnotateNoOp \
>+  AnnotatePCQCreate \
>+  AnnotatePCQDestroy \
>+  AnnotatePCQGet \
>+  AnnotatePCQPut \
>+  AnnotatePublishMemoryRange \
>+  AnnotateRWLockAcquired \
>+  AnnotateRWLockCreate \
>+  AnnotateRWLockCreateStatic \
>+  AnnotateRWLockDestroy \
>+  AnnotateRWLockReleased \
>+  AnnotateThreadName \
>+  AnnotateTraceMemory \
>+  AnnotateUnpublishMemoryRange \
>+  GetRunningOnValgrind \
>+  RunningOnValgrind \
>+  StaticAnnotateIgnoreReadsBegin \
>+  StaticAnnotateIgnoreReadsEnd \
>+  StaticAnnotateIgnoreWritesBegin \
>+  StaticAnnotateIgnoreWritesEnd \
>+  ValgrindSlowdown \
>+; do
>+  find absl/ -type f -exec sed -i "s/\b$w\b/Absl$w/g" {} \;
>+done
>+
>+for w in \
>+  ADDRESS_SANITIZER_REDZONE \
>+  ANNOTALYSIS_ENABLED \
>+  ANNOTATE_BARRIER_DESTROY \
>+  ANNOTATE_BARRIER_INIT \
>+  ANNOTATE_BARRIER_WAIT_AFTER \
>+  ANNOTATE_BARRIER_WAIT_BEFORE \
>+  ANNOTATE_BENIGN_RACE \
>+  ANNOTATE_BENIGN_RACE_SIZED \
>+  ANNOTATE_BENIGN_RACE_STATIC \
>+  ANNOTATE_CONDVAR_LOCK_WAIT \
>+  ANNOTATE_CONDVAR_SIGNAL \
>+  ANNOTATE_CONDVAR_SIGNAL_ALL \
>+  ANNOTATE_CONDVAR_WAIT \
>+  ANNOTATE_CONTIGUOUS_CONTAINER \
>+  ANNOTATE_ENABLE_RACE_DETECTION \
>+  ANNOTATE_EXPECT_RACE \
>+  ANNOTATE_FLUSH_EXPECTED_RACES \
>+  ANNOTATE_FLUSH_STATE \
>+  ANNOTATE_HAPPENS_AFTER \
>+  ANNOTATE_HAPPENS_BEFORE \
>+  ANNOTATE_IGNORE_READS_AND_WRITES_BEGIN ANNOTATE_IGNORE_READS_AND_WRITES_END \
>+  ANNOTATE_IGNORE_READS_BEGIN \
>+  ANNOTATE_IGNORE_READS_END \
>+  ANNOTATE_IGNORE_SYNC_BEGIN \
>+  ANNOTATE_IGNORE_SYNC_END \
>+  ANNOTATE_IGNORE_WRITES_BEGIN \
>+  ANNOTATE_IGNORE_WRITES_END \
>+  ANNOTATE_MEMORY_IS_INITIALIZED \
>+  ANNOTATE_MEMORY_IS_UNINITIALIZED \
>+  ANNOTATE_MUTEX_IS_USED_AS_CONDVAR \
>+  ANNOTATE_NEW_MEMORY \
>+  ANNOTATE_NOT_HAPPENS_BEFORE_MUTEX \
>+  ANNOTATE_NO_OP \
>+  ANNOTATE_PCQ_CREATE ANNOTATE_PCQ_DESTROY \
>+  ANNOTATE_PCQ_GET ANNOTATE_PCQ_PUT \
>+  ANNOTATE_PUBLISH_MEMORY_RANGE \
>+  ANNOTATE_PURE_HAPPENS_BEFORE_MUTEX \
>+  ANNOTATE_RWLOCK_ACQUIRED \
>+  ANNOTATE_RWLOCK_CREATE \
>+  ANNOTATE_RWLOCK_CREATE_STATIC \
>+  ANNOTATE_RWLOCK_DESTROY \
>+  ANNOTATE_RWLOCK_RELEASED \
>+  ANNOTATE_SWAP_MEMORY_RANGE \
>+  ANNOTATE_THREAD_NAME \
>+  ANNOTATE_TRACE_MEMORY \
>+  ANNOTATE_UNPROTECTED_READ \
>+  ANNOTATE_UNPUBLISH_MEMORY_RANGE \
>+  ANNOTATIONS_ENABLED \
>+  ATTRIBUTE_IGNORE_READS_BEGIN \
>+  ATTRIBUTE_IGNORE_READS_END \
>+  DYNAMIC_ANNOTATIONS_ATTRIBUTE_WEAK \
>+  DYNAMIC_ANNOTATIONS_ENABLED \
>+  DYNAMIC_ANNOTATIONS_EXTERNAL_IMPL \
>+  DYNAMIC_ANNOTATIONS_GLUE \
>+  DYNAMIC_ANNOTATIONS_GLUE0 \
>+  DYNAMIC_ANNOTATIONS_IMPL \
>+  DYNAMIC_ANNOTATIONS_NAME \
>+  DYNAMIC_ANNOTATIONS_PREFIX \
>+  DYNAMIC_ANNOTATIONS_PROVIDE_RUNNING_ON_VALGRIND \
>+  DYNAMIC_ANNOTATIONS_WANT_ATTRIBUTE_WEAK \
>+; do
>+  find absl/ -type f -exec sed -i "s/\b$w\b/ABSL_$w/g" {} \;
>+done
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/.gn b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/.gn
>index 015d0f0e5e8..26a4a62d199 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/.gn
>@@ -6,6 +6,8 @@
> # in the file PATENTS. All contributing project authors may
> # be found in the AUTHORS file in the root of the source tree.
> 
>+import("//build/dotfile_settings.gni")
>+
> # The location of the build configuration file.
> buildconfig = "//build/config/BUILDCONFIG.gn"
> 
>@@ -23,29 +25,8 @@ check_targets = [ "//libyuv/*" ]
> # These are the list of GN files that run exec_script. This whitelist exists
> # to force additional review for new uses of exec_script, which is strongly
> # discouraged except for gypi_to_gn calls.
>-exec_script_whitelist = [
>-  "//build/config/android/BUILD.gn",
>-  "//build/config/android/config.gni",
>-  "//build/config/android/internal_rules.gni",
>-  "//build/config/android/rules.gni",
>-  "//build/config/BUILD.gn",
>-  "//build/config/compiler/BUILD.gn",
>-  "//build/config/gcc/gcc_version.gni",
>-  "//build/config/ios/ios_sdk.gni",
>-  "//build/config/linux/BUILD.gn",
>-  "//build/config/linux/pkg_config.gni",
>-  "//build/config/mac/mac_sdk.gni",
>-  "//build/config/posix/BUILD.gn",
>-  "//build/config/sysroot.gni",
>-  "//build/config/win/BUILD.gn",
>-  "//build/config/win/visual_studio_version.gni",
>-  "//build/gn_helpers.py",
>-  "//build/gypi_to_gn.py",
>-  "//build/toolchain/concurrent_links.gni",
>-  "//build/toolchain/gcc_toolchain.gni",
>-  "//build/toolchain/mac/BUILD.gn",
>-  "//build/toolchain/win/BUILD.gn",
>-]
>+exec_script_whitelist = build_dotfile_settings.exec_script_whitelist +
>+                        [ "//build_overrides/build.gni" ]
> 
> default_args = {
>   mac_sdk_min = "10.11"
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/Android.bp b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/Android.bp
>index a3d8d834ac7..7d95a7865ab 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/Android.bp
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/Android.bp
>@@ -24,14 +24,12 @@ cc_library {
>         "source/rotate_any.cc",
>         "source/rotate_argb.cc",
>         "source/rotate_common.cc",
>-        "source/rotate_dspr2.cc",
>         "source/rotate_gcc.cc",
>         "source/rotate_msa.cc",
>         "source/rotate_neon.cc",
>         "source/rotate_neon64.cc",
>         "source/row_any.cc",
>         "source/row_common.cc",
>-        "source/row_dspr2.cc",
>         "source/row_gcc.cc",
>         "source/row_msa.cc",
>         "source/row_neon.cc",
>@@ -40,7 +38,6 @@ cc_library {
>         "source/scale_any.cc",
>         "source/scale_argb.cc",
>         "source/scale_common.cc",
>-        "source/scale_dspr2.cc",
>         "source/scale_gcc.cc",
>         "source/scale_msa.cc",
>         "source/scale_neon.cc",
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/Android.mk b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/Android.mk
>index 854020610ea..dbc6cad37ab 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/Android.mk
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/Android.mk
>@@ -24,14 +24,12 @@ LOCAL_SRC_FILES := \
>     source/rotate_any.cc        \
>     source/rotate_argb.cc       \
>     source/rotate_common.cc     \
>-    source/rotate_dspr2.cc      \
>     source/rotate_gcc.cc        \
>     source/rotate_msa.cc        \
>     source/rotate_neon.cc       \
>     source/rotate_neon64.cc     \
>     source/row_any.cc           \
>     source/row_common.cc        \
>-    source/row_dspr2.cc         \
>     source/row_gcc.cc           \
>     source/row_msa.cc           \
>     source/row_neon.cc          \
>@@ -40,7 +38,6 @@ LOCAL_SRC_FILES := \
>     source/scale_any.cc         \
>     source/scale_argb.cc        \
>     source/scale_common.cc      \
>-    source/scale_dspr2.cc       \
>     source/scale_gcc.cc         \
>     source/scale_msa.cc         \
>     source/scale_neon.cc        \
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/BUILD.gn
>index 9badf08c846..03ce499e6d9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/BUILD.gn
>@@ -12,6 +12,11 @@ import("//testing/test.gni")
> declare_args() {
>   # Set to false to disable building with gflags.
>   libyuv_use_gflags = true
>+
>+  # When building a shared library using a target in WebRTC or
>+  # Chromium projects that depends on libyuv, setting this flag
>+  # to true makes libyuv symbols visible inside that library.
>+  libyuv_symbols_visible = false
> }
> 
> config("libyuv_config") {
>@@ -42,7 +47,8 @@ group("default") {
> }
> 
> group("libyuv") {
>-  public_configs = [ ":libyuv_config" ]
>+  all_dependent_configs = [ ":libyuv_config" ]
>+  deps = []
> 
>   if (is_win && target_cpu == "x64") {
>     # Compile with clang in order to get inline assembly
>@@ -55,13 +61,19 @@ group("libyuv") {
>     ]
>   }
> 
>+  if (libyuv_use_neon) {
>+    deps += [ ":libyuv_neon" ]
>+  }
>+
>+  if (libyuv_use_msa) {
>+    deps += [ ":libyuv_msa" ]
>+  }
>+
>   if (!is_ios) {
>     # Make sure that clients of libyuv link with libjpeg. This can't go in
>     # libyuv_internal because in Windows x64 builds that will generate a clang
>     # build of libjpeg, and we don't want two copies.
>-    deps = [
>-      "//third_party:jpeg",
>-    ]
>+    deps += [ "//third_party:jpeg" ]
>   }
> }
> 
>@@ -110,19 +122,16 @@ static_library("libyuv_internal") {
>     "source/rotate_any.cc",
>     "source/rotate_argb.cc",
>     "source/rotate_common.cc",
>-    "source/rotate_dspr2.cc",
>     "source/rotate_gcc.cc",
>     "source/rotate_win.cc",
>     "source/row_any.cc",
>     "source/row_common.cc",
>-    "source/row_dspr2.cc",
>     "source/row_gcc.cc",
>     "source/row_win.cc",
>     "source/scale.cc",
>     "source/scale_any.cc",
>     "source/scale_argb.cc",
>     "source/scale_common.cc",
>-    "source/scale_dspr2.cc",
>     "source/scale_gcc.cc",
>     "source/scale_win.cc",
>     "source/video_common.cc",
>@@ -132,6 +141,11 @@ static_library("libyuv_internal") {
>   defines = []
>   deps = []
> 
>+  if (libyuv_symbols_visible) {
>+    configs -= [ "//build/config/gcc:symbol_visibility_hidden" ]
>+    configs += [ "//build/config/gcc:symbol_visibility_default" ]
>+  }
>+
>   if (!is_ios) {
>     defines += [ "HAVE_JPEG" ]
> 
>@@ -140,14 +154,6 @@ static_library("libyuv_internal") {
>     deps += [ "//third_party:jpeg_includes" ]
>   }
> 
>-  if (libyuv_use_neon) {
>-    deps += [ ":libyuv_neon" ]
>-  }
>-
>-  if (libyuv_use_msa) {
>-    deps += [ ":libyuv_msa" ]
>-  }
>-
>   # Always enable optimization for Release and NaCl builds (to workaround
>   # crbug.com/538243).
>   if (!is_debug || is_nacl) {
>@@ -167,6 +173,7 @@ static_library("libyuv_internal") {
>     ]
>   }
> }
>+
> if (libyuv_use_neon) {
>   static_library("libyuv_neon") {
>     sources = [
>@@ -181,6 +188,10 @@ if (libyuv_use_neon) {
>       "source/scale_neon64.cc",
>     ]
> 
>+    deps = [
>+      ":libyuv_internal",
>+    ]
>+
>     public_configs = [ ":libyuv_config" ]
> 
>     # Always enable optimization for Release and NaCl builds (to workaround
>@@ -210,6 +221,10 @@ if (libyuv_use_msa) {
>       "source/scale_msa.cc",
>     ]
> 
>+    deps = [
>+      ":libyuv_internal",
>+    ]
>+
>     public_configs = [ ":libyuv_config" ]
>   }
> }
>@@ -302,7 +317,6 @@ if (libyuv_include_tests) {
>       # Enable the following 3 macros to turn off assembly for specified CPU.
>       # "LIBYUV_DISABLE_X86",
>       # "LIBYUV_DISABLE_NEON",
>-      # "LIBYUV_DISABLE_DSPR2",
>       # Enable the following macro to build libyuv as a shared library (dll).
>       # "LIBYUV_USING_SHARED_LIBRARY"
>     ]
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/DEPS b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/DEPS
>index fdb133c7ac1..266327470b7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/DEPS
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/DEPS
>@@ -1,38 +1,38 @@
> vars = {
>   'chromium_git': 'https://chromium.googlesource.com',
>-  'chromium_revision': 'ff3b31782d552b03104a6d831c7530605b52b13f',
>-  'swarming_revision': '5e8001d9a710121ce7a68efd0804430a34b4f9e4',
>+  'chromium_revision': '80f0c5570649c35a869429b2ab8c381a0a7246cb',
>+  'swarming_revision': '88229872dd17e71658fe96763feaa77915d8cbd6',
>   # Three lines of non-changing comments so that
>   # the commit queue can handle CLs rolling lss
>   # and whatever else without interference from each other.
>-  'lss_revision': '63f24c8221a229f677d26ebe8f3d1528a9d787ac',
>+  'lss_revision': 'e6527b0cd469e3ff5764785dadcb39bf7d787154',
>   # Three lines of non-changing comments so that
>   # the commit queue can handle CLs rolling catapult
>   # and whatever else without interference from each other.
>-  'catapult_revision': 'aa736cc76ee5e35215abcfb83a8c354f12d0c684',
>+  'catapult_revision': 'e7298f36f7912f2caa122086cfbe71734d04b73f',
> }
> 
> deps = {
>   'src/build':
>-    Var('chromium_git') + '/chromium/src/build' + '@' + '156ba982d749902e3403c242e23ded87fd316494',
>+    Var('chromium_git') + '/chromium/src/build' + '@' + '39738e75b27f39d4c0030a0b11d5d2ddd34715f7',
>   'src/buildtools':
>-    Var('chromium_git') + '/chromium/buildtools.git' + '@' + 'f6d165d9d842ddd29056c127a5f3a3c5d8e0d2e3',
>+    Var('chromium_git') + '/chromium/buildtools.git' + '@' + 'a09e064635a49f08e585e3b173d5fbc3dd3f485e',
>   'src/testing':
>-    Var('chromium_git') + '/chromium/src/testing' + '@' + 'cc96d3d66b5b9613fd0fe055509cfec5eb54b19c',
>+    Var('chromium_git') + '/chromium/src/testing' + '@' + '5f7e36cad6434fd3d65674af96653a23ecc9f694',
>   'src/third_party':
>-    Var('chromium_git') + '/chromium/src/third_party' + '@' + '72c52c224cdd3c377f7caff8ffed0f5749e79549',
>+    Var('chromium_git') + '/chromium/src/third_party' + '@' + '2c0ced3ddbf84ce1c0759b277d9538da42f23650',
>   'src/third_party/catapult':
>-   Var('chromium_git') + '/external/github.com/catapult-project/catapult.git' + '@' + Var('catapult_revision'),
>+    Var('chromium_git') + '/catapult.git' + '@' + Var('catapult_revision'),
>   'src/third_party/colorama/src':
>     Var('chromium_git') + '/external/colorama.git' + '@' + '799604a1041e9b3bc5d2789ecbd7e8db2e18e6b8',
>   'src/third_party/googletest/src':
>-    Var('chromium_git') + '/external/github.com/google/googletest.git' + '@' + '7f8fefabedf2965980585be8c2bff97458f28e0b',
>+    Var('chromium_git') + '/external/github.com/google/googletest.git' + '@' + '0062e4869f07a3ef235703ddf63af604b712446c',
>   'src/third_party/libjpeg_turbo':
>     Var('chromium_git') + '/chromium/deps/libjpeg_turbo.git' + '@' + 'a1750dbc79a8792dde3d3f7d7d8ac28ba01ac9dd',
>   'src/third_party/yasm/source/patched-yasm':
>     Var('chromium_git') + '/chromium/deps/yasm/patched-yasm.git' + '@' + 'b98114e18d8b9b84586b10d24353ab8616d4c5fc',
>   'src/tools':
>-    Var('chromium_git') + '/chromium/src/tools' + '@' + 'eceb2c420b20350a2d2ba261953109280968647a',
>+    Var('chromium_git') + '/chromium/src/tools' + '@' + '6202b67fc46a9984097caf237e12e3b8f7a9f7da',
>   'src/tools/gyp':
>     Var('chromium_git') + '/external/gyp.git' + '@' + 'd61a9397e668fa9843c4aa7da9e79460fe590bfb',
>    'src/tools/swarming_client':
>@@ -45,45 +45,84 @@ deps = {
>     Var('chromium_git') + '/external/github.com/gflags/gflags' + '@' + '03bebcb065c83beff83d50ae025a55a4bf94dfca',
>   'src/third_party/gtest-parallel':
>     Var('chromium_git') + '/external/webrtc/deps/third_party/gtest-parallel' + '@' + '1dad0e9f6d82ff994130b529d7d814b40eb32b0e',
>-}
> 
>-deps_os = {
>-  'android': {
>-    'src/base':
>-      Var('chromium_git') + '/chromium/src/base' + '@' + '9b543d487c7c38be191c6180001ff9ce186ae326',
>-    'src/third_party/android_tools':
>-      Var('chromium_git') + '/android_tools.git' + '@' + 'aadb2fed04af8606545b0afe4e3060bc1a15fad7',
>-    'src/third_party/ced/src':
>-      Var('chromium_git') + '/external/github.com/google/compact_enc_det.git' + '@' + '94c367a1fe3a13207f4b22604fcfd1d9f9ddf6d9',
>-    'src/third_party/icu':
>-      Var('chromium_git') + '/chromium/deps/icu.git' + '@' + '08cb956852a5ccdba7f9c941728bb833529ba3c6',
>-    'src/third_party/jsr-305/src':
>-      Var('chromium_git') + '/external/jsr-305.git' + '@' + '642c508235471f7220af6d5df2d3210e3bfc0919',
>-    'src/third_party/junit/src':
>-      Var('chromium_git') + '/external/junit.git' + '@' + '64155f8a9babcfcf4263cf4d08253a1556e75481',
>-    'src/third_party/lss':
>-      Var('chromium_git') + '/linux-syscall-support.git' + '@' + Var('lss_revision'),
>-    'src/third_party/mockito/src':
>-      Var('chromium_git') + '/external/mockito/mockito.git' + '@' + 'de83ad4598ad4cf5ea53c69a8a8053780b04b850',
>-    'src/third_party/requests/src':
>-      Var('chromium_git') + '/external/github.com/kennethreitz/requests.git' + '@' + 'f172b30356d821d180fa4ecfa3e71c7274a32de4',
>-    'src/third_party/robolectric/robolectric':
>-      Var('chromium_git') + '/external/robolectric.git' + '@' + 'b02c65cc6d7465f58f0de48a39914aa905692afa',
>-    'src/third_party/ub-uiautomator/lib':
>-      Var('chromium_git') + '/chromium/third_party/ub-uiautomator.git' + '@' + '00270549ce3161ae72ceb24712618ea28b4f9434',
>-  },
>-  'ios': {
>-    'src/ios':
>-      Var('chromium_git') + '/chromium/src/ios' + '@' + '39c4b2fcf73f5b1e82af3b9c57267c17217d6a30',
>-  },
>-  'unix': {
>-    'src/third_party/lss':
>-      Var('chromium_git') + '/linux-syscall-support.git' + '@' + Var('lss_revision'),
>-  },
>-  'win': {
>-    # Dependencies used by libjpeg-turbo
>-    'src/third_party/yasm/binaries':
>-      Var('chromium_git') + '/chromium/deps/yasm/binaries.git' + '@' + '52f9b3f4b0aa06da24ef8b123058bb61ee468881',
>+  'src/third_party/lss': {
>+    'url': Var('chromium_git') + '/linux-syscall-support.git' + '@' + Var('lss_revision'),
>+    'condition': 'checkout_android or checkout_linux',
>+  },
>+
>+  # Android deps:
>+  'src/third_party/auto/src': {
>+    'url': Var('chromium_git') + '/external/github.com/google/auto.git' + '@' + '8a81a858ae7b78a1aef71ac3905fade0bbd64e82',
>+    'condition': 'checkout_android',
>+  },
>+  'src/third_party/auto/src': {
>+    'url': Var('chromium_git') + '/external/github.com/google/auto.git' + '@' + '8a81a858ae7b78a1aef71ac3905fade0bbd64e82',
>+    'condition': 'checkout_android',
>+  },
>+  'src/base': {
>+    'url': Var('chromium_git') + '/chromium/src/base' + '@' + 'ac3d2b81181b085a9952cb83dba748420eefe691',
>+    'condition': 'checkout_android',
>+  },
>+  'src/third_party/android_ndk': {
>+    'url': Var('chromium_git') + '/android_ndk.git' + '@' + 'e951c37287c7d8cd915bf8d4149fd4a06d808b55',
>+    'condition': 'checkout_android',
>+  },
>+  'src/third_party/android_tools': {
>+    'url': Var('chromium_git') + '/android_tools.git' + '@' + '9a70d48fcdd68cd0e7e968f342bd767ee6323bd1',
>+    'condition': 'checkout_android',
>+  },
>+  'src/third_party/ced/src': {
>+    'url': Var('chromium_git') + '/external/github.com/google/compact_enc_det.git' + '@' + '94c367a1fe3a13207f4b22604fcfd1d9f9ddf6d9',
>+    'condition': 'checkout_android',
>+  },
>+  'src/third_party/errorprone/lib': {
>+      'url': Var('chromium_git') + '/chromium/third_party/errorprone.git' + '@' + 'ecc57c2b00627667874744b9ad8efe10734d97a8',
>+      'condition': 'checkout_android',
>+  },
>+  'src/third_party/findbugs': {
>+    'url': Var('chromium_git') + '/chromium/deps/findbugs.git' + '@' + '4275d9ac8610db6b1bc9a5e887f97e41b33fac67',
>+    'condition': 'checkout_android',
>+  },
>+  'src/third_party/icu': {
>+    'url': Var('chromium_git') + '/chromium/deps/icu.git' + '@' + 'd888fd2a1be890f4d35e43f68d6d79f42519a357',
>+  },
>+  'src/third_party/jsr-305/src': {
>+    'url': Var('chromium_git') + '/external/jsr-305.git' + '@' + '642c508235471f7220af6d5df2d3210e3bfc0919',
>+    'condition': 'checkout_android',
>+  },
>+  'src/third_party/junit/src': {
>+    'url': Var('chromium_git') + '/external/junit.git' + '@' + '64155f8a9babcfcf4263cf4d08253a1556e75481',
>+    'condition': 'checkout_android',
>+  },
>+  'src/third_party/mockito/src': {
>+    'url': Var('chromium_git') + '/external/mockito/mockito.git' + '@' + 'de83ad4598ad4cf5ea53c69a8a8053780b04b850',
>+    'condition': 'checkout_android',
>+  },
>+  'src/third_party/requests/src': {
>+    'url': Var('chromium_git') + '/external/github.com/kennethreitz/requests.git' + '@' + 'f172b30356d821d180fa4ecfa3e71c7274a32de4',
>+    'condition': 'checkout_android',
>+  },
>+  'src/third_party/robolectric/robolectric': {
>+    'url': Var('chromium_git') + '/external/robolectric.git' + '@' + '7e067f1112e1502caa742f7be72d37b5678d3403',
>+    'condition': 'checkout_android',
>+  },
>+  'src/third_party/ub-uiautomator/lib': {
>+    'url': Var('chromium_git') + '/chromium/third_party/ub-uiautomator.git' + '@' + '00270549ce3161ae72ceb24712618ea28b4f9434',
>+    'condition': 'checkout_android',
>+  },
>+
>+  # iOS deps:
>+  'src/ios': {
>+    'url': Var('chromium_git') + '/chromium/src/ios' + '@' + '299ef76e844a74a1f2f4ce7f06d101861fb49aba',
>+    'condition': 'checkout_ios'
>+  },
>+
>+  # Win deps:
>+  # Dependencies used by libjpeg-turbo
>+  'src/third_party/yasm/binaries': {
>+    'url': Var('chromium_git') + '/chromium/deps/yasm/binaries.git' + '@' + '52f9b3f4b0aa06da24ef8b123058bb61ee468881',
>+    'condition': 'checkout_win',
>   },
> }
> 
>@@ -116,200 +155,41 @@ hooks = [
>         'src',
>     ],
>   },
>-  # Android dependencies. Many are downloaded using Google Storage these days.
>-  # They're copied from https://cs.chromium.org/chromium/src/DEPS for all
>-  # such dependencies we share with Chromium.
>-  {
>-    # This downloads SDK extras and puts them in the
>-    # third_party/android_tools/sdk/extras directory.
>-    'name': 'sdkextras',
>-    'pattern': '.',
>-    # When adding a new sdk extras package to download, add the package
>-    # directory and zip file to .gitignore in third_party/android_tools.
>-    'action': ['python',
>-               'src/build/android/play_services/update.py',
>-               'download'
>-    ],
>-  },
>-  {
>-    'name': 'intellij',
>-    'pattern': '.',
>-    'action': ['python',
>-               'src/build/android/update_deps/update_third_party_deps.py',
>-               'download',
>-               '-b', 'chromium-intellij',
>-               '-l', 'third_party/intellij'
>-    ],
>-  },
>-  {
>-    'name': 'javax_inject',
>-    'pattern': '.',
>-    'action': ['python',
>-               'src/build/android/update_deps/update_third_party_deps.py',
>-               'download',
>-               '-b', 'chromium-javax-inject',
>-               '-l', 'third_party/javax_inject'
>-    ],
>-  },
>-  {
>-    'name': 'hamcrest',
>-    'pattern': '.',
>-    'action': ['python',
>-               'src/build/android/update_deps/update_third_party_deps.py',
>-               'download',
>-               '-b', 'chromium-hamcrest',
>-               '-l', 'third_party/hamcrest'
>-    ],
>-  },
>-  {
>-    'name': 'guava',
>-    'pattern': '.',
>-    'action': ['python',
>-               'src/build/android/update_deps/update_third_party_deps.py',
>-               'download',
>-               '-b', 'chromium-guava',
>-               '-l', 'third_party/guava'
>-    ],
>-  },
>-  {
>-    'name': 'android_support_test_runner',
>-    'pattern': '.',
>-    'action': ['python',
>-               'src/build/android/update_deps/update_third_party_deps.py',
>-               'download',
>-               '-b', 'chromium-android-support-test-runner',
>-               '-l', 'third_party/android_support_test_runner'
>-    ],
>-  },
>-  {
>-    'name': 'byte_buddy',
>-    'pattern': '.',
>-    'action': ['python',
>-               'src/build/android/update_deps/update_third_party_deps.py',
>-               'download',
>-               '-b', 'chromium-byte-buddy',
>-               '-l', 'third_party/byte_buddy'
>-    ],
>-  },
>+  # Downloads the current stable linux sysroot to build/linux/ if needed.
>   {
>-    'name': 'espresso',
>+    'name': 'sysroot_arm',
>     'pattern': '.',
>-    'action': ['python',
>-               'src/build/android/update_deps/update_third_party_deps.py',
>-               'download',
>-               '-b', 'chromium-espresso',
>-               '-l', 'third_party/espresso'
>-    ],
>-  },
>-  {
>-    'name': 'robolectric_libs',
>-    'pattern': '.',
>-    'action': ['python',
>-               'src/build/android/update_deps/update_third_party_deps.py',
>-               'download',
>-               '-b', 'chromium-robolectric',
>-               '-l', 'third_party/robolectric'
>-    ],
>-  },
>-  {
>-    'name': 'apache_velocity',
>-    'pattern': '.',
>-    'action': ['python',
>-               'src/build/android/update_deps/update_third_party_deps.py',
>-               'download',
>-               '-b', 'chromium-apache-velocity',
>-               '-l', 'third_party/apache_velocity'
>-    ],
>-  },
>-  {
>-    'name': 'ow2_asm',
>-    'pattern': '.',
>-    'action': ['python',
>-               'src/build/android/update_deps/update_third_party_deps.py',
>-               'download',
>-               '-b', 'chromium-ow2-asm',
>-               '-l', 'third_party/ow2_asm'
>-    ],
>-  },
>-  {
>-    'name': 'desugar',
>-    'pattern': '.',
>-    'action': ['python',
>-               'src/build/android/update_deps/update_third_party_deps.py',
>-               'download',
>-               '-b', 'chromium-android-tools/bazel/desugar',
>-               '-l', 'third_party/bazel/desugar'
>-    ],
>-  },
>-  {
>-    'name': 'icu4j',
>-    'pattern': '.',
>-    'action': ['python',
>-               'src/build/android/update_deps/update_third_party_deps.py',
>-               'download',
>-               '-b', 'chromium-icu4j',
>-               '-l', 'third_party/icu4j'
>-    ],
>-  },
>-  {
>-    'name': 'accessibility_test_framework',
>-    'pattern': '.',
>-    'action': ['python',
>-               'src/build/android/update_deps/update_third_party_deps.py',
>-               'download',
>-               '-b', 'chromium-accessibility-test-framework',
>-               '-l', 'third_party/accessibility_test_framework'
>-    ],
>-  },
>-  {
>-    'name': 'bouncycastle',
>-    'pattern': '.',
>-    'action': ['python',
>-               'src/build/android/update_deps/update_third_party_deps.py',
>-               'download',
>-               '-b', 'chromium-bouncycastle',
>-               '-l', 'third_party/bouncycastle'
>-    ],
>+    'condition': 'checkout_linux and checkout_arm',
>+    'action': ['python', 'src/build/linux/sysroot_scripts/install-sysroot.py',
>+               '--arch=arm'],
>   },
>   {
>-    'name': 'sqlite4java',
>+    'name': 'sysroot_arm64',
>     'pattern': '.',
>-    'action': ['python',
>-               'src/build/android/update_deps/update_third_party_deps.py',
>-               'download',
>-               '-b', 'chromium-sqlite4java',
>-               '-l', 'third_party/sqlite4java'
>-    ],
>+    'condition': 'checkout_linux and checkout_arm64',
>+    'action': ['python', 'src/build/linux/sysroot_scripts/install-sysroot.py',
>+               '--arch=arm64'],
>   },
>   {
>-    'name': 'xstream',
>+    'name': 'sysroot_x86',
>     'pattern': '.',
>-    'action': ['python',
>-               'src/build/android/update_deps/update_third_party_deps.py',
>-               'download',
>-               '-b', 'chromium-robolectric',
>-               '-l', 'third_party/xstream'
>-    ],
>+    'condition': 'checkout_linux and (checkout_x86 or checkout_x64)',
>+    'action': ['python', 'src/build/linux/sysroot_scripts/install-sysroot.py',
>+               '--arch=x86'],
>   },
>   {
>-    'name': 'objenesis',
>+    'name': 'sysroot_mips',
>     'pattern': '.',
>-    'action': ['python',
>-               'src/build/android/update_deps/update_third_party_deps.py',
>-               'download',
>-               '-b', 'chromium-objenesis',
>-               '-l', 'third_party/objenesis'
>-    ],
>+    'condition': 'checkout_linux and checkout_mips',
>+    'action': ['python', 'src/build/linux/sysroot_scripts/install-sysroot.py',
>+               '--arch=mips'],
>   },
>   {
>-    # Downloads the current stable linux sysroot to build/linux/ if needed.
>-    # This sysroot updates at about the same rate that the chrome build deps
>-    # change. This script is a no-op except for linux users who are doing
>-    # official chrome builds or cross compiling.
>-    'name': 'sysroot',
>+    'name': 'sysroot_x64',
>     'pattern': '.',
>+    'condition': 'checkout_linux and checkout_x64',
>     'action': ['python', 'src/build/linux/sysroot_scripts/install-sysroot.py',
>-               '--running-as-hook'],
>+               '--arch=x64'],
>   },
>   {
>     # Update the Windows toolchain if necessary.
>@@ -444,6 +324,30 @@ hooks = [
>                 '-d', 'src/tools/luci-go/linux64',
>     ],
>   },
>+  {
>+    'name': 'Android CIPD Ensure',
>+    'pattern': '.',
>+    'condition': 'checkout_android',
>+    'action': ['src/build/cipd/cipd_wrapper.py',
>+               '--chromium-root', 'src',
>+               '--ensure-file', 'src/build/cipd/android/android.ensure',
>+    ],
>+  },
>+  # Android dependencies. Many are downloaded using Google Storage these days.
>+  # They're copied from https://cs.chromium.org/chromium/src/DEPS for all
>+  # such dependencies we share with Chromium.
>+  {
>+    # This downloads SDK extras and puts them in the
>+    # third_party/android_tools/sdk/extras directory.
>+    'name': 'sdkextras',
>+    'pattern': '.',
>+    # When adding a new sdk extras package to download, add the package
>+    # directory and zip file to .gitignore in third_party/android_tools.
>+    'action': ['python',
>+               'src/build/android/play_services/update.py',
>+               'download'
>+    ],
>+  },
> ]
> 
> recursedeps = [
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/README.chromium b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/README.chromium
>index 88c069734ac..c25373e23ce 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/README.chromium
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/README.chromium
>@@ -1,6 +1,6 @@
> Name: libyuv
> URL: http://code.google.com/p/libyuv/
>-Version: 1678
>+Version: 1703
> License: BSD
> License File: LICENSE
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/README.md b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/README.md
>index b59b71c5d2b..7b6619220b8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/README.md
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/README.md
>@@ -1,12 +1,12 @@
> **libyuv** is an open source project that includes YUV scaling and conversion functionality.
> 
> * Scale YUV to prepare content for compression, with point, bilinear or box filter.
>-* Convert to YUV from webcam formats.
>-* Convert from YUV to formats for rendering/effects.
>+* Convert to YUV from webcam formats for compression.
>+* Convert to RGB formats for rendering/effects.
> * Rotate by 90/180/270 degrees to adjust for mobile devices in portrait mode.
>-* Optimized for SSE2/SSSE3/AVX2 on x86/x64.
>+* Optimized for SSSE3/AVX2 on x86/x64.
> * Optimized for Neon on Arm.
>-* Optimized for DSP R2 on Mips.
>+* Optimized for MSA on Mips.
> 
> ### Development
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/docs/deprecated_builds.md b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/docs/deprecated_builds.md
>index d54a0282c15..29e0bf9bc30 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/docs/deprecated_builds.md
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/docs/deprecated_builds.md
>@@ -165,11 +165,11 @@ mipsel
> 
> arm32 disassembly:
> 
>-    third_party/android_tools/ndk/toolchains/arm-linux-androideabi-4.9/prebuilt/linux-x86_64/bin/arm-linux-androideabi-objdump -d out/Release/obj/source/libyuv.row_neon.o
>+    third_party/android_ndk/toolchains/arm-linux-androideabi-4.9/prebuilt/linux-x86_64/bin/arm-linux-androideabi-objdump -d out/Release/obj/source/libyuv.row_neon.o
> 
> arm64 disassembly:
> 
>-    third_party/android_tools/ndk/toolchains/aarch64-linux-android-4.9/prebuilt/linux-x86_64/bin/aarch64-linux-android-objdump -d out/Release/obj/source/libyuv.row_neon64.o
>+    third_party/android_ndk/toolchains/aarch64-linux-android-4.9/prebuilt/linux-x86_64/bin/aarch64-linux-android-objdump -d out/Release/obj/source/libyuv.row_neon64.o
> 
> Running tests:
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/docs/environment_variables.md b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/docs/environment_variables.md
>index 9071c54de23..c28d83e7dc1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/docs/environment_variables.md
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/docs/environment_variables.md
>@@ -17,7 +17,7 @@ By default the cpu is detected and the most advanced form of SIMD is used.  But
>     LIBYUV_DISABLE_AVX512BW
>     LIBYUV_DISABLE_ERMS
>     LIBYUV_DISABLE_FMA3
>-    LIBYUV_DISABLE_DSPR2
>+    LIBYUV_DISABLE_MSA
>     LIBYUV_DISABLE_NEON
> 
> # Test Width/Height/Repeat
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/docs/formats.md b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/docs/formats.md
>index 2b75d31ac75..f78f57bb4c4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/docs/formats.md
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/docs/formats.md
>@@ -35,9 +35,8 @@ This is how OSX formats map to libyuv
> # FOURCC (Four Charactacter Code) List
> 
> The following is extracted from video_common.h as a complete list of formats supported by libyuv.
>-
>     enum FourCC {
>-      // 8 Primary YUV formats: 5 planar, 2 biplanar, 2 packed.
>+      // 9 Primary YUV formats: 5 planar, 2 biplanar, 2 packed.
>       FOURCC_I420 = FOURCC('I', '4', '2', '0'),
>       FOURCC_I422 = FOURCC('I', '4', '2', '2'),
>       FOURCC_I444 = FOURCC('I', '4', '4', '4'),
>@@ -46,37 +45,35 @@ The following is extracted from video_common.h as a complete list of formats sup
>       FOURCC_NV12 = FOURCC('N', 'V', '1', '2'),
>       FOURCC_YUY2 = FOURCC('Y', 'U', 'Y', '2'),
>       FOURCC_UYVY = FOURCC('U', 'Y', 'V', 'Y'),
>+      FOURCC_H010 = FOURCC('H', '0', '1', '0'),  // unofficial fourcc. 10 bit lsb
> 
>-      // 1 Secondary YUV formats: row biplanar.
>+      // 1 Secondary YUV format: row biplanar.
>       FOURCC_M420 = FOURCC('M', '4', '2', '0'),
> 
>-      // 9 Primary RGB formats: 4 32 bpp, 2 24 bpp, 3 16 bpp.
>+      // 11 Primary RGB formats: 4 32 bpp, 2 24 bpp, 3 16 bpp, 1 10 bpc
>       FOURCC_ARGB = FOURCC('A', 'R', 'G', 'B'),
>       FOURCC_BGRA = FOURCC('B', 'G', 'R', 'A'),
>       FOURCC_ABGR = FOURCC('A', 'B', 'G', 'R'),
>+      FOURCC_AR30 = FOURCC('A', 'R', '3', '0'),  // 10 bit per channel. 2101010.
>+      FOURCC_AB30 = FOURCC('A', 'B', '3', '0'),  // ABGR version of 10 bit
>       FOURCC_24BG = FOURCC('2', '4', 'B', 'G'),
>-      FOURCC_RAW  = FOURCC('r', 'a', 'w', ' '),
>+      FOURCC_RAW = FOURCC('r', 'a', 'w', ' '),
>       FOURCC_RGBA = FOURCC('R', 'G', 'B', 'A'),
>       FOURCC_RGBP = FOURCC('R', 'G', 'B', 'P'),  // rgb565 LE.
>       FOURCC_RGBO = FOURCC('R', 'G', 'B', 'O'),  // argb1555 LE.
>       FOURCC_R444 = FOURCC('R', '4', '4', '4'),  // argb4444 LE.
> 
>-      // 4 Secondary RGB formats: 4 Bayer Patterns.
>-      FOURCC_RGGB = FOURCC('R', 'G', 'G', 'B'),
>-      FOURCC_BGGR = FOURCC('B', 'G', 'G', 'R'),
>-      FOURCC_GRBG = FOURCC('G', 'R', 'B', 'G'),
>-      FOURCC_GBRG = FOURCC('G', 'B', 'R', 'G'),
>-
>       // 1 Primary Compressed YUV format.
>       FOURCC_MJPG = FOURCC('M', 'J', 'P', 'G'),
> 
>-      // 5 Auxiliary YUV variations: 3 with U and V planes are swapped, 1 Alias.
>+      // 7 Auxiliary YUV variations: 3 with U and V planes are swapped, 1 Alias.
>       FOURCC_YV12 = FOURCC('Y', 'V', '1', '2'),
>       FOURCC_YV16 = FOURCC('Y', 'V', '1', '6'),
>       FOURCC_YV24 = FOURCC('Y', 'V', '2', '4'),
>       FOURCC_YU12 = FOURCC('Y', 'U', '1', '2'),  // Linux version of I420.
>       FOURCC_J420 = FOURCC('J', '4', '2', '0'),
>-      FOURCC_J400 = FOURCC('J', '4', '0', '0'),
>+      FOURCC_J400 = FOURCC('J', '4', '0', '0'),  // unofficial fourcc
>+      FOURCC_H420 = FOURCC('H', '4', '2', '0'),  // unofficial fourcc
> 
>       // 14 Auxiliary aliases.  CanonicalFourCC() maps these to canonical fourcc.
>       FOURCC_IYUV = FOURCC('I', 'Y', 'U', 'V'),  // Alias for I420.
>@@ -97,9 +94,6 @@ The following is extracted from video_common.h as a complete list of formats sup
>       FOURCC_L565 = FOURCC('L', '5', '6', '5'),  // Alias for RGBP.
>       FOURCC_5551 = FOURCC('5', '5', '5', '1'),  // Alias for RGBO.
> 
>-      // 1 Auxiliary compressed YUV format set aside for capturer.
>-      FOURCC_H264 = FOURCC('H', '2', '6', '4'),
>-
> # Planar YUV
>       The following formats contains a full size Y plane followed by 1 or 2
>         planes for UV: I420, I422, I444, I400, NV21, NV12, I400
>@@ -145,3 +139,24 @@ There are 2 RGB layouts - RGB24 (aka 24BG) and RAW
> 
> RGB24 is B,G,R in memory
> RAW is R,G,B in memory
>+
>+# AR30 and XR30
>+
>+AR30 is 2 10 10 10 ARGB stored in little endian order.
>+The 2 bit alpha has 4 values.  Here are the comparable 8 bit alpha values.
>+0 - 0.    00000000b = 0x00 = 0
>+1 - 33%.  01010101b = 0x55 = 85
>+2 - 66%.  10101010b = 0xaa = 170
>+3 - 100%. 11111111b = 0xff = 255
>+The 10 bit RGB values range from 0 to 1023.
>+XR30 is the same as AR30 but with no alpha channel.
>+
>+# NV12 and NV21
>+
>+NV12 is a biplanar format with a full sized Y plane followed by a single
>+chroma plane with weaved U and V values.
>+NV21 is the same but with weaved V and U values.
>+The 12 in NV12 refers to 12 bits per pixel.  NV12 has a half width and half
>+height chroma channel, and therefore is a 420 subsampling.
>+NV16 is 16 bits per pixel, with half width and full height.  aka 422.
>+NV24 is 24 bits per pixel with full sized chroma channel. aka 444.
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/docs/getting_started.md b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/docs/getting_started.md
>index 58e05f3cbcb..09297b66a5c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/docs/getting_started.md
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/docs/getting_started.md
>@@ -129,25 +129,20 @@ ia32
>     ninja -v -C out/Debug libyuv_unittest
>     ninja -v -C out/Release libyuv_unittest
> 
>-mipsel
>+mips
> 
>-    gn gen out/Release "--args=is_debug=false target_os=\"android\" target_cpu=\"mipsel\" mips_arch_variant=\"r6\" mips_use_msa=true is_component_build=true is_clang=false"
>-    gn gen out/Debug "--args=is_debug=true target_os=\"android\" target_cpu=\"mipsel\" mips_arch_variant=\"r6\" mips_use_msa=true is_component_build=true is_clang=false"
>-    ninja -v -C out/Debug libyuv_unittest
>-    ninja -v -C out/Release libyuv_unittest
>-
>-    gn gen out/Release "--args=is_debug=false target_os=\"android\" target_cpu=\"mips64el\" mips_arch_variant=\"r6\" mips_use_msa=true is_component_build=true is_clang=false"
>-    gn gen out/Debug "--args=is_debug=true target_os=\"android\" target_cpu=\"mips64el\" mips_arch_variant=\"r6\" mips_use_msa=true is_component_build=true is_clang=false"
>+    gn gen out/Release "--args=is_debug=false target_os=\"android\" target_cpu=\"mips64el\" mips_arch_variant=\"r6\" mips_use_msa=true is_component_build=true is_clang=true"
>+    gn gen out/Debug "--args=is_debug=true target_os=\"android\" target_cpu=\"mips64el\" mips_arch_variant=\"r6\" mips_use_msa=true is_component_build=true is_clang=true"
>     ninja -v -C out/Debug libyuv_unittest
>     ninja -v -C out/Release libyuv_unittest
> 
> arm disassembly:
> 
>-    third_party/android_tools/ndk/toolchains/aarch64-linux-android-4.9/prebuilt/linux-x86_64/bin/aarch64-linux-android-objdump -d ./out/Release/obj/libyuv/row_common.o >row_common.txt
>+    third_party/android_ndk/toolchains/aarch64-linux-android-4.9/prebuilt/linux-x86_64/bin/aarch64-linux-android-objdump -d ./out/Release/obj/libyuv/row_common.o >row_common.txt
> 
>-    third_party/android_tools/ndk/toolchains/aarch64-linux-android-4.9/prebuilt/linux-x86_64/bin/aarch64-linux-android-objdump -d ./out/Release/obj/libyuv_neon/row_neon.o >row_neon.txt
>+    third_party/android_ndk/toolchains/aarch64-linux-android-4.9/prebuilt/linux-x86_64/bin/aarch64-linux-android-objdump -d ./out/Release/obj/libyuv_neon/row_neon.o >row_neon.txt
> 
>-    third_party/android_tools/ndk/toolchains/aarch64-linux-android-4.9/prebuilt/linux-x86_64/bin/aarch64-linux-android-objdump -d ./out/Release/obj/libyuv_neon/row_neon64.o >row_neon64.txt
>+    third_party/android_ndk/toolchains/aarch64-linux-android-4.9/prebuilt/linux-x86_64/bin/aarch64-linux-android-objdump -d ./out/Release/obj/libyuv_neon/row_neon64.o >row_neon64.txt
> 
> Running tests:
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/basic_types.h b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/basic_types.h
>index 7d98bb93f0e..01d9dfc7736 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/basic_types.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/basic_types.h
>@@ -11,79 +11,33 @@
> #ifndef INCLUDE_LIBYUV_BASIC_TYPES_H_
> #define INCLUDE_LIBYUV_BASIC_TYPES_H_
> 
>-#include <stddef.h>  // for NULL, size_t
>+#include <stddef.h>  // For size_t and NULL
>+
>+#if !defined(INT_TYPES_DEFINED) && !defined(GG_LONGLONG)
>+#define INT_TYPES_DEFINED
> 
> #if defined(_MSC_VER) && (_MSC_VER < 1600)
> #include <sys/types.h>  // for uintptr_t on x86
>+typedef unsigned __int64 uint64_t;
>+typedef __int64 int64_t;
>+typedef unsigned int uint32_t;
>+typedef int int32_t;
>+typedef unsigned short uint16_t;
>+typedef short int16_t;
>+typedef unsigned char uint8_t;
>+typedef signed char int8_t;
> #else
>-#include <stdint.h>  // for uintptr_t
>-#endif
>-
>-#ifndef GG_LONGLONG
>-#ifndef INT_TYPES_DEFINED
>-#define INT_TYPES_DEFINED
>-#ifdef COMPILER_MSVC
>-typedef unsigned __int64 uint64;
>-typedef __int64 int64;
>-#ifndef INT64_C
>-#define INT64_C(x) x##I64
>-#endif
>-#ifndef UINT64_C
>-#define UINT64_C(x) x##UI64
>-#endif
>-#define INT64_F "I64"
>-#else  // COMPILER_MSVC
>-#if defined(__LP64__) && !defined(__OpenBSD__) && !defined(__APPLE__)
>-typedef unsigned long uint64;  // NOLINT
>-typedef long int64;            // NOLINT
>-#ifndef INT64_C
>-#define INT64_C(x) x##L
>-#endif
>-#ifndef UINT64_C
>-#define UINT64_C(x) x##UL
>-#endif
>-#define INT64_F "l"
>-#else  // defined(__LP64__) && !defined(__OpenBSD__) && !defined(__APPLE__)
>-typedef unsigned long long uint64;  // NOLINT
>-typedef long long int64;            // NOLINT
>-#ifndef INT64_C
>-#define INT64_C(x) x##LL
>-#endif
>-#ifndef UINT64_C
>-#define UINT64_C(x) x##ULL
>-#endif
>-#define INT64_F "ll"
>-#endif  // __LP64__
>-#endif  // COMPILER_MSVC
>-typedef unsigned int uint32;
>-typedef int int32;
>-typedef unsigned short uint16;  // NOLINT
>-typedef short int16;            // NOLINT
>-typedef unsigned char uint8;
>-typedef signed char int8;
>+#include <stdint.h>  // for uintptr_t and C99 types
>+#endif               // defined(_MSC_VER) && (_MSC_VER < 1600)
>+typedef uint64_t uint64;
>+typedef int64_t int64;
>+typedef uint32_t uint32;
>+typedef int32_t int32;
>+typedef uint16_t uint16;
>+typedef int16_t int16;
>+typedef uint8_t uint8;
>+typedef int8_t int8;
> #endif  // INT_TYPES_DEFINED
>-#endif  // GG_LONGLONG
>-
>-// Detect compiler is for x86 or x64.
>-#if defined(__x86_64__) || defined(_M_X64) || defined(__i386__) || \
>-    defined(_M_IX86)
>-#define CPU_X86 1
>-#endif
>-// Detect compiler is for ARM.
>-#if defined(__arm__) || defined(_M_ARM)
>-#define CPU_ARM 1
>-#endif
>-
>-#ifndef ALIGNP
>-#ifdef __cplusplus
>-#define ALIGNP(p, t)        \
>-  reinterpret_cast<uint8*>( \
>-      ((reinterpret_cast<uintptr_t>(p) + ((t)-1)) & ~((t)-1)))
>-#else
>-#define ALIGNP(p, t) \
>-  (uint8*)((((uintptr_t)(p) + ((t)-1)) & ~((t)-1))) /* NOLINT */
>-#endif
>-#endif
> 
> #if !defined(LIBYUV_API)
> #if defined(_WIN32) || defined(__CYGWIN__)
>@@ -103,15 +57,9 @@ typedef signed char int8;
> #endif  // __GNUC__
> #endif  // LIBYUV_API
> 
>+// TODO(fbarchard): Remove bool macros.
> #define LIBYUV_BOOL int
> #define LIBYUV_FALSE 0
> #define LIBYUV_TRUE 1
> 
>-// Visual C x86 or GCC little endian.
>-#if defined(__x86_64__) || defined(_M_X64) || defined(__i386__) || \
>-    defined(_M_IX86) || defined(__arm__) || defined(_M_ARM) ||     \
>-    (defined(__BYTE_ORDER__) && __BYTE_ORDER__ == __ORDER_LITTLE_ENDIAN__)
>-#define LIBYUV_LITTLE_ENDIAN
>-#endif
>-
> #endif  // INCLUDE_LIBYUV_BASIC_TYPES_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/compare.h b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/compare.h
>index a06eff2066f..3353ad71c68 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/compare.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/compare.h
>@@ -20,80 +20,85 @@ extern "C" {
> 
> // Compute a hash for specified memory. Seed of 5381 recommended.
> LIBYUV_API
>-uint32 HashDjb2(const uint8* src, uint64 count, uint32 seed);
>+uint32_t HashDjb2(const uint8_t* src, uint64_t count, uint32_t seed);
> 
> // Hamming Distance
> LIBYUV_API
>-uint64 ComputeHammingDistance(const uint8* src_a,
>-                              const uint8* src_b,
>-                              int count);
>+uint64_t ComputeHammingDistance(const uint8_t* src_a,
>+                                const uint8_t* src_b,
>+                                int count);
> 
> // Scan an opaque argb image and return fourcc based on alpha offset.
> // Returns FOURCC_ARGB, FOURCC_BGRA, or 0 if unknown.
> LIBYUV_API
>-uint32 ARGBDetect(const uint8* argb, int stride_argb, int width, int height);
>+uint32_t ARGBDetect(const uint8_t* argb,
>+                    int stride_argb,
>+                    int width,
>+                    int height);
> 
> // Sum Square Error - used to compute Mean Square Error or PSNR.
> LIBYUV_API
>-uint64 ComputeSumSquareError(const uint8* src_a, const uint8* src_b, int count);
>+uint64_t ComputeSumSquareError(const uint8_t* src_a,
>+                               const uint8_t* src_b,
>+                               int count);
> 
> LIBYUV_API
>-uint64 ComputeSumSquareErrorPlane(const uint8* src_a,
>-                                  int stride_a,
>-                                  const uint8* src_b,
>-                                  int stride_b,
>-                                  int width,
>-                                  int height);
>+uint64_t ComputeSumSquareErrorPlane(const uint8_t* src_a,
>+                                    int stride_a,
>+                                    const uint8_t* src_b,
>+                                    int stride_b,
>+                                    int width,
>+                                    int height);
> 
> static const int kMaxPsnr = 128;
> 
> LIBYUV_API
>-double SumSquareErrorToPsnr(uint64 sse, uint64 count);
>+double SumSquareErrorToPsnr(uint64_t sse, uint64_t count);
> 
> LIBYUV_API
>-double CalcFramePsnr(const uint8* src_a,
>+double CalcFramePsnr(const uint8_t* src_a,
>                      int stride_a,
>-                     const uint8* src_b,
>+                     const uint8_t* src_b,
>                      int stride_b,
>                      int width,
>                      int height);
> 
> LIBYUV_API
>-double I420Psnr(const uint8* src_y_a,
>+double I420Psnr(const uint8_t* src_y_a,
>                 int stride_y_a,
>-                const uint8* src_u_a,
>+                const uint8_t* src_u_a,
>                 int stride_u_a,
>-                const uint8* src_v_a,
>+                const uint8_t* src_v_a,
>                 int stride_v_a,
>-                const uint8* src_y_b,
>+                const uint8_t* src_y_b,
>                 int stride_y_b,
>-                const uint8* src_u_b,
>+                const uint8_t* src_u_b,
>                 int stride_u_b,
>-                const uint8* src_v_b,
>+                const uint8_t* src_v_b,
>                 int stride_v_b,
>                 int width,
>                 int height);
> 
> LIBYUV_API
>-double CalcFrameSsim(const uint8* src_a,
>+double CalcFrameSsim(const uint8_t* src_a,
>                      int stride_a,
>-                     const uint8* src_b,
>+                     const uint8_t* src_b,
>                      int stride_b,
>                      int width,
>                      int height);
> 
> LIBYUV_API
>-double I420Ssim(const uint8* src_y_a,
>+double I420Ssim(const uint8_t* src_y_a,
>                 int stride_y_a,
>-                const uint8* src_u_a,
>+                const uint8_t* src_u_a,
>                 int stride_u_a,
>-                const uint8* src_v_a,
>+                const uint8_t* src_v_a,
>                 int stride_v_a,
>-                const uint8* src_y_b,
>+                const uint8_t* src_y_b,
>                 int stride_y_b,
>-                const uint8* src_u_b,
>+                const uint8_t* src_u_b,
>                 int stride_u_b,
>-                const uint8* src_v_b,
>+                const uint8_t* src_v_b,
>                 int stride_v_b,
>                 int width,
>                 int height);
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/compare_row.h b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/compare_row.h
>index 2e5ebe508d1..72ee740600a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/compare_row.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/compare_row.h
>@@ -18,17 +18,20 @@ namespace libyuv {
> extern "C" {
> #endif
> 
>-#if defined(__pnacl__) || defined(__CLR_VER) || \
>+#if defined(__pnacl__) || defined(__CLR_VER) ||            \
>+    (defined(__native_client__) && defined(__x86_64__)) || \
>     (defined(__i386__) && !defined(__SSE__) && !defined(__clang__))
> #define LIBYUV_DISABLE_X86
> #endif
>+#if defined(__native_client__)
>+#define LIBYUV_DISABLE_NEON
>+#endif
> // MemorySanitizer does not support assembly code yet. http://crbug.com/344505
> #if defined(__has_feature)
> #if __has_feature(memory_sanitizer)
> #define LIBYUV_DISABLE_X86
> #endif
> #endif
>-
> // Visual C 2012 required for AVX2.
> #if defined(_M_IX86) && !defined(__clang__) && defined(_MSC_VER) && \
>     _MSC_VER >= 1700
>@@ -87,22 +90,44 @@ extern "C" {
> #define HAS_SUMSQUAREERROR_MSA
> #endif
> 
>-uint32 HammingDistance_C(const uint8* src_a, const uint8* src_b, int count);
>-uint32 HammingDistance_SSE42(const uint8* src_a, const uint8* src_b, int count);
>-uint32 HammingDistance_SSSE3(const uint8* src_a, const uint8* src_b, int count);
>-uint32 HammingDistance_AVX2(const uint8* src_a, const uint8* src_b, int count);
>-uint32 HammingDistance_NEON(const uint8* src_a, const uint8* src_b, int count);
>-uint32 HammingDistance_MSA(const uint8* src_a, const uint8* src_b, int count);
>-
>-uint32 SumSquareError_C(const uint8* src_a, const uint8* src_b, int count);
>-uint32 SumSquareError_SSE2(const uint8* src_a, const uint8* src_b, int count);
>-uint32 SumSquareError_AVX2(const uint8* src_a, const uint8* src_b, int count);
>-uint32 SumSquareError_NEON(const uint8* src_a, const uint8* src_b, int count);
>-uint32 SumSquareError_MSA(const uint8* src_a, const uint8* src_b, int count);
>-
>-uint32 HashDjb2_C(const uint8* src, int count, uint32 seed);
>-uint32 HashDjb2_SSE41(const uint8* src, int count, uint32 seed);
>-uint32 HashDjb2_AVX2(const uint8* src, int count, uint32 seed);
>+uint32_t HammingDistance_C(const uint8_t* src_a,
>+                           const uint8_t* src_b,
>+                           int count);
>+uint32_t HammingDistance_SSE42(const uint8_t* src_a,
>+                               const uint8_t* src_b,
>+                               int count);
>+uint32_t HammingDistance_SSSE3(const uint8_t* src_a,
>+                               const uint8_t* src_b,
>+                               int count);
>+uint32_t HammingDistance_AVX2(const uint8_t* src_a,
>+                              const uint8_t* src_b,
>+                              int count);
>+uint32_t HammingDistance_NEON(const uint8_t* src_a,
>+                              const uint8_t* src_b,
>+                              int count);
>+uint32_t HammingDistance_MSA(const uint8_t* src_a,
>+                             const uint8_t* src_b,
>+                             int count);
>+
>+uint32_t SumSquareError_C(const uint8_t* src_a,
>+                          const uint8_t* src_b,
>+                          int count);
>+uint32_t SumSquareError_SSE2(const uint8_t* src_a,
>+                             const uint8_t* src_b,
>+                             int count);
>+uint32_t SumSquareError_AVX2(const uint8_t* src_a,
>+                             const uint8_t* src_b,
>+                             int count);
>+uint32_t SumSquareError_NEON(const uint8_t* src_a,
>+                             const uint8_t* src_b,
>+                             int count);
>+uint32_t SumSquareError_MSA(const uint8_t* src_a,
>+                            const uint8_t* src_b,
>+                            int count);
>+
>+uint32_t HashDjb2_C(const uint8_t* src, int count, uint32_t seed);
>+uint32_t HashDjb2_SSE41(const uint8_t* src, int count, uint32_t seed);
>+uint32_t HashDjb2_AVX2(const uint8_t* src, int count, uint32_t seed);
> 
> #ifdef __cplusplus
> }  // extern "C"
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/convert.h b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/convert.h
>index f096d193a74..d12ef24f799 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/convert.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/convert.h
>@@ -27,34 +27,34 @@ extern "C" {
> 
> // Convert I444 to I420.
> LIBYUV_API
>-int I444ToI420(const uint8* src_y,
>+int I444ToI420(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_u,
>+               uint8_t* dst_u,
>                int dst_stride_u,
>-               uint8* dst_v,
>+               uint8_t* dst_v,
>                int dst_stride_v,
>                int width,
>                int height);
> 
> // Convert I422 to I420.
> LIBYUV_API
>-int I422ToI420(const uint8* src_y,
>+int I422ToI420(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_u,
>+               uint8_t* dst_u,
>                int dst_stride_u,
>-               uint8* dst_v,
>+               uint8_t* dst_v,
>                int dst_stride_v,
>                int width,
>                int height);
>@@ -62,30 +62,67 @@ int I422ToI420(const uint8* src_y,
> // Copy I420 to I420.
> #define I420ToI420 I420Copy
> LIBYUV_API
>-int I420Copy(const uint8* src_y,
>+int I420Copy(const uint8_t* src_y,
>              int src_stride_y,
>-             const uint8* src_u,
>+             const uint8_t* src_u,
>              int src_stride_u,
>-             const uint8* src_v,
>+             const uint8_t* src_v,
>              int src_stride_v,
>-             uint8* dst_y,
>+             uint8_t* dst_y,
>              int dst_stride_y,
>-             uint8* dst_u,
>+             uint8_t* dst_u,
>              int dst_stride_u,
>-             uint8* dst_v,
>+             uint8_t* dst_v,
>              int dst_stride_v,
>              int width,
>              int height);
> 
>+// Copy I010 to I010
>+#define I010ToI010 I010Copy
>+#define H010ToH010 I010Copy
>+LIBYUV_API
>+int I010Copy(const uint16_t* src_y,
>+             int src_stride_y,
>+             const uint16_t* src_u,
>+             int src_stride_u,
>+             const uint16_t* src_v,
>+             int src_stride_v,
>+             uint16_t* dst_y,
>+             int dst_stride_y,
>+             uint16_t* dst_u,
>+             int dst_stride_u,
>+             uint16_t* dst_v,
>+             int dst_stride_v,
>+             int width,
>+             int height);
>+
>+// Convert 10 bit YUV to 8 bit
>+#define H010ToH420 I010ToI420
>+LIBYUV_API
>+int I010ToI420(const uint16_t* src_y,
>+               int src_stride_y,
>+               const uint16_t* src_u,
>+               int src_stride_u,
>+               const uint16_t* src_v,
>+               int src_stride_v,
>+               uint8_t* dst_y,
>+               int dst_stride_y,
>+               uint8_t* dst_u,
>+               int dst_stride_u,
>+               uint8_t* dst_v,
>+               int dst_stride_v,
>+               int width,
>+               int height);
>+
> // Convert I400 (grey) to I420.
> LIBYUV_API
>-int I400ToI420(const uint8* src_y,
>+int I400ToI420(const uint8_t* src_y,
>                int src_stride_y,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_u,
>+               uint8_t* dst_u,
>                int dst_stride_u,
>-               uint8* dst_v,
>+               uint8_t* dst_v,
>                int dst_stride_v,
>                int width,
>                int height);
>@@ -94,204 +131,204 @@ int I400ToI420(const uint8* src_y,
> 
> // Convert NV12 to I420.
> LIBYUV_API
>-int NV12ToI420(const uint8* src_y,
>+int NV12ToI420(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_uv,
>+               const uint8_t* src_uv,
>                int src_stride_uv,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_u,
>+               uint8_t* dst_u,
>                int dst_stride_u,
>-               uint8* dst_v,
>+               uint8_t* dst_v,
>                int dst_stride_v,
>                int width,
>                int height);
> 
> // Convert NV21 to I420.
> LIBYUV_API
>-int NV21ToI420(const uint8* src_y,
>+int NV21ToI420(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_vu,
>+               const uint8_t* src_vu,
>                int src_stride_vu,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_u,
>+               uint8_t* dst_u,
>                int dst_stride_u,
>-               uint8* dst_v,
>+               uint8_t* dst_v,
>                int dst_stride_v,
>                int width,
>                int height);
> 
> // Convert YUY2 to I420.
> LIBYUV_API
>-int YUY2ToI420(const uint8* src_yuy2,
>+int YUY2ToI420(const uint8_t* src_yuy2,
>                int src_stride_yuy2,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_u,
>+               uint8_t* dst_u,
>                int dst_stride_u,
>-               uint8* dst_v,
>+               uint8_t* dst_v,
>                int dst_stride_v,
>                int width,
>                int height);
> 
> // Convert UYVY to I420.
> LIBYUV_API
>-int UYVYToI420(const uint8* src_uyvy,
>+int UYVYToI420(const uint8_t* src_uyvy,
>                int src_stride_uyvy,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_u,
>+               uint8_t* dst_u,
>                int dst_stride_u,
>-               uint8* dst_v,
>+               uint8_t* dst_v,
>                int dst_stride_v,
>                int width,
>                int height);
> 
> // Convert M420 to I420.
> LIBYUV_API
>-int M420ToI420(const uint8* src_m420,
>+int M420ToI420(const uint8_t* src_m420,
>                int src_stride_m420,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_u,
>+               uint8_t* dst_u,
>                int dst_stride_u,
>-               uint8* dst_v,
>+               uint8_t* dst_v,
>                int dst_stride_v,
>                int width,
>                int height);
> 
> // Convert Android420 to I420.
> LIBYUV_API
>-int Android420ToI420(const uint8* src_y,
>+int Android420ToI420(const uint8_t* src_y,
>                      int src_stride_y,
>-                     const uint8* src_u,
>+                     const uint8_t* src_u,
>                      int src_stride_u,
>-                     const uint8* src_v,
>+                     const uint8_t* src_v,
>                      int src_stride_v,
>-                     int pixel_stride_uv,
>-                     uint8* dst_y,
>+                     int src_pixel_stride_uv,
>+                     uint8_t* dst_y,
>                      int dst_stride_y,
>-                     uint8* dst_u,
>+                     uint8_t* dst_u,
>                      int dst_stride_u,
>-                     uint8* dst_v,
>+                     uint8_t* dst_v,
>                      int dst_stride_v,
>                      int width,
>                      int height);
> 
> // ARGB little endian (bgra in memory) to I420.
> LIBYUV_API
>-int ARGBToI420(const uint8* src_frame,
>-               int src_stride_frame,
>-               uint8* dst_y,
>+int ARGBToI420(const uint8_t* src_argb,
>+               int src_stride_argb,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_u,
>+               uint8_t* dst_u,
>                int dst_stride_u,
>-               uint8* dst_v,
>+               uint8_t* dst_v,
>                int dst_stride_v,
>                int width,
>                int height);
> 
> // BGRA little endian (argb in memory) to I420.
> LIBYUV_API
>-int BGRAToI420(const uint8* src_frame,
>-               int src_stride_frame,
>-               uint8* dst_y,
>+int BGRAToI420(const uint8_t* src_bgra,
>+               int src_stride_bgra,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_u,
>+               uint8_t* dst_u,
>                int dst_stride_u,
>-               uint8* dst_v,
>+               uint8_t* dst_v,
>                int dst_stride_v,
>                int width,
>                int height);
> 
> // ABGR little endian (rgba in memory) to I420.
> LIBYUV_API
>-int ABGRToI420(const uint8* src_frame,
>-               int src_stride_frame,
>-               uint8* dst_y,
>+int ABGRToI420(const uint8_t* src_abgr,
>+               int src_stride_abgr,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_u,
>+               uint8_t* dst_u,
>                int dst_stride_u,
>-               uint8* dst_v,
>+               uint8_t* dst_v,
>                int dst_stride_v,
>                int width,
>                int height);
> 
> // RGBA little endian (abgr in memory) to I420.
> LIBYUV_API
>-int RGBAToI420(const uint8* src_frame,
>-               int src_stride_frame,
>-               uint8* dst_y,
>+int RGBAToI420(const uint8_t* src_rgba,
>+               int src_stride_rgba,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_u,
>+               uint8_t* dst_u,
>                int dst_stride_u,
>-               uint8* dst_v,
>+               uint8_t* dst_v,
>                int dst_stride_v,
>                int width,
>                int height);
> 
> // RGB little endian (bgr in memory) to I420.
> LIBYUV_API
>-int RGB24ToI420(const uint8* src_frame,
>-                int src_stride_frame,
>-                uint8* dst_y,
>+int RGB24ToI420(const uint8_t* src_rgb24,
>+                int src_stride_rgb24,
>+                uint8_t* dst_y,
>                 int dst_stride_y,
>-                uint8* dst_u,
>+                uint8_t* dst_u,
>                 int dst_stride_u,
>-                uint8* dst_v,
>+                uint8_t* dst_v,
>                 int dst_stride_v,
>                 int width,
>                 int height);
> 
> // RGB big endian (rgb in memory) to I420.
> LIBYUV_API
>-int RAWToI420(const uint8* src_frame,
>-              int src_stride_frame,
>-              uint8* dst_y,
>+int RAWToI420(const uint8_t* src_raw,
>+              int src_stride_raw,
>+              uint8_t* dst_y,
>               int dst_stride_y,
>-              uint8* dst_u,
>+              uint8_t* dst_u,
>               int dst_stride_u,
>-              uint8* dst_v,
>+              uint8_t* dst_v,
>               int dst_stride_v,
>               int width,
>               int height);
> 
> // RGB16 (RGBP fourcc) little endian to I420.
> LIBYUV_API
>-int RGB565ToI420(const uint8* src_frame,
>-                 int src_stride_frame,
>-                 uint8* dst_y,
>+int RGB565ToI420(const uint8_t* src_rgb565,
>+                 int src_stride_rgb565,
>+                 uint8_t* dst_y,
>                  int dst_stride_y,
>-                 uint8* dst_u,
>+                 uint8_t* dst_u,
>                  int dst_stride_u,
>-                 uint8* dst_v,
>+                 uint8_t* dst_v,
>                  int dst_stride_v,
>                  int width,
>                  int height);
> 
> // RGB15 (RGBO fourcc) little endian to I420.
> LIBYUV_API
>-int ARGB1555ToI420(const uint8* src_frame,
>-                   int src_stride_frame,
>-                   uint8* dst_y,
>+int ARGB1555ToI420(const uint8_t* src_argb1555,
>+                   int src_stride_argb1555,
>+                   uint8_t* dst_y,
>                    int dst_stride_y,
>-                   uint8* dst_u,
>+                   uint8_t* dst_u,
>                    int dst_stride_u,
>-                   uint8* dst_v,
>+                   uint8_t* dst_v,
>                    int dst_stride_v,
>                    int width,
>                    int height);
> 
> // RGB12 (R444 fourcc) little endian to I420.
> LIBYUV_API
>-int ARGB4444ToI420(const uint8* src_frame,
>-                   int src_stride_frame,
>-                   uint8* dst_y,
>+int ARGB4444ToI420(const uint8_t* src_argb4444,
>+                   int src_stride_argb4444,
>+                   uint8_t* dst_y,
>                    int dst_stride_y,
>-                   uint8* dst_u,
>+                   uint8_t* dst_u,
>                    int dst_stride_u,
>-                   uint8* dst_v,
>+                   uint8_t* dst_v,
>                    int dst_stride_v,
>                    int width,
>                    int height);
>@@ -300,13 +337,13 @@ int ARGB4444ToI420(const uint8* src_frame,
> // src_width/height provided by capture.
> // dst_width/height for clipping determine final size.
> LIBYUV_API
>-int MJPGToI420(const uint8* sample,
>+int MJPGToI420(const uint8_t* sample,
>                size_t sample_size,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_u,
>+               uint8_t* dst_u,
>                int dst_stride_u,
>-               uint8* dst_v,
>+               uint8_t* dst_v,
>                int dst_stride_v,
>                int src_width,
>                int src_height,
>@@ -315,7 +352,10 @@ int MJPGToI420(const uint8* sample,
> 
> // Query size of MJPG in pixels.
> LIBYUV_API
>-int MJPGSize(const uint8* sample, size_t sample_size, int* width, int* height);
>+int MJPGSize(const uint8_t* sample,
>+             size_t sample_size,
>+             int* width,
>+             int* height);
> #endif
> 
> // Convert camera sample to I420 with cropping, rotation and vertical flip.
>@@ -338,16 +378,16 @@ int MJPGSize(const uint8* sample, size_t sample_size, int* width, int* height);
> //    Must be less than or equal to src_width/src_height
> //    Cropping parameters are pre-rotation.
> // "rotation" can be 0, 90, 180 or 270.
>-// "format" is a fourcc. ie 'I420', 'YUY2'
>+// "fourcc" is a fourcc. ie 'I420', 'YUY2'
> // Returns 0 for successful; -1 for invalid parameter. Non-zero for failure.
> LIBYUV_API
>-int ConvertToI420(const uint8* src_frame,
>-                  size_t src_size,
>-                  uint8* dst_y,
>+int ConvertToI420(const uint8_t* sample,
>+                  size_t sample_size,
>+                  uint8_t* dst_y,
>                   int dst_stride_y,
>-                  uint8* dst_u,
>+                  uint8_t* dst_u,
>                   int dst_stride_u,
>-                  uint8* dst_v,
>+                  uint8_t* dst_v,
>                   int dst_stride_v,
>                   int crop_x,
>                   int crop_y,
>@@ -356,7 +396,7 @@ int ConvertToI420(const uint8* src_frame,
>                   int crop_width,
>                   int crop_height,
>                   enum RotationMode rotation,
>-                  uint32 format);
>+                  uint32_t fourcc);
> 
> #ifdef __cplusplus
> }  // extern "C"
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/convert_argb.h b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/convert_argb.h
>index f43a5060bdb..ab772b6c323 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/convert_argb.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/convert_argb.h
>@@ -30,102 +30,167 @@ extern "C" {
> 
> // Copy ARGB to ARGB.
> LIBYUV_API
>-int ARGBCopy(const uint8* src_argb,
>+int ARGBCopy(const uint8_t* src_argb,
>              int src_stride_argb,
>-             uint8* dst_argb,
>+             uint8_t* dst_argb,
>              int dst_stride_argb,
>              int width,
>              int height);
> 
> // Convert I420 to ARGB.
> LIBYUV_API
>-int I420ToARGB(const uint8* src_y,
>+int I420ToARGB(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_argb,
>+               uint8_t* dst_argb,
>                int dst_stride_argb,
>                int width,
>                int height);
> 
> // Duplicate prototype for function in convert_from.h for remoting.
> LIBYUV_API
>-int I420ToABGR(const uint8* src_y,
>+int I420ToABGR(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_argb,
>+               uint8_t* dst_abgr,
>+               int dst_stride_abgr,
>+               int width,
>+               int height);
>+
>+// Convert I010 to ARGB.
>+LIBYUV_API
>+int I010ToARGB(const uint16_t* src_y,
>+               int src_stride_y,
>+               const uint16_t* src_u,
>+               int src_stride_u,
>+               const uint16_t* src_v,
>+               int src_stride_v,
>+               uint8_t* dst_argb,
>                int dst_stride_argb,
>                int width,
>                int height);
> 
>+// Convert I010 to ARGB.
>+LIBYUV_API
>+int I010ToARGB(const uint16_t* src_y,
>+               int src_stride_y,
>+               const uint16_t* src_u,
>+               int src_stride_u,
>+               const uint16_t* src_v,
>+               int src_stride_v,
>+               uint8_t* dst_argb,
>+               int dst_stride_argb,
>+               int width,
>+               int height);
>+
>+// Convert I010 to ABGR.
>+LIBYUV_API
>+int I010ToABGR(const uint16_t* src_y,
>+               int src_stride_y,
>+               const uint16_t* src_u,
>+               int src_stride_u,
>+               const uint16_t* src_v,
>+               int src_stride_v,
>+               uint8_t* dst_abgr,
>+               int dst_stride_abgr,
>+               int width,
>+               int height);
>+
>+// Convert H010 to ARGB.
>+LIBYUV_API
>+int H010ToARGB(const uint16_t* src_y,
>+               int src_stride_y,
>+               const uint16_t* src_u,
>+               int src_stride_u,
>+               const uint16_t* src_v,
>+               int src_stride_v,
>+               uint8_t* dst_argb,
>+               int dst_stride_argb,
>+               int width,
>+               int height);
>+
>+// Convert H010 to ABGR.
>+LIBYUV_API
>+int H010ToABGR(const uint16_t* src_y,
>+               int src_stride_y,
>+               const uint16_t* src_u,
>+               int src_stride_u,
>+               const uint16_t* src_v,
>+               int src_stride_v,
>+               uint8_t* dst_abgr,
>+               int dst_stride_abgr,
>+               int width,
>+               int height);
>+
> // Convert I422 to ARGB.
> LIBYUV_API
>-int I422ToARGB(const uint8* src_y,
>+int I422ToARGB(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_argb,
>+               uint8_t* dst_argb,
>                int dst_stride_argb,
>                int width,
>                int height);
> 
> // Convert I444 to ARGB.
> LIBYUV_API
>-int I444ToARGB(const uint8* src_y,
>+int I444ToARGB(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_argb,
>+               uint8_t* dst_argb,
>                int dst_stride_argb,
>                int width,
>                int height);
> 
> // Convert J444 to ARGB.
> LIBYUV_API
>-int J444ToARGB(const uint8* src_y,
>+int J444ToARGB(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_argb,
>+               uint8_t* dst_argb,
>                int dst_stride_argb,
>                int width,
>                int height);
> 
> // Convert I444 to ABGR.
> LIBYUV_API
>-int I444ToABGR(const uint8* src_y,
>+int I444ToABGR(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_abgr,
>+               uint8_t* dst_abgr,
>                int dst_stride_abgr,
>                int width,
>                int height);
> 
> // Convert I420 with Alpha to preattenuated ARGB.
> LIBYUV_API
>-int I420AlphaToARGB(const uint8* src_y,
>+int I420AlphaToARGB(const uint8_t* src_y,
>                     int src_stride_y,
>-                    const uint8* src_u,
>+                    const uint8_t* src_u,
>                     int src_stride_u,
>-                    const uint8* src_v,
>+                    const uint8_t* src_v,
>                     int src_stride_v,
>-                    const uint8* src_a,
>+                    const uint8_t* src_a,
>                     int src_stride_a,
>-                    uint8* dst_argb,
>+                    uint8_t* dst_argb,
>                     int dst_stride_argb,
>                     int width,
>                     int height,
>@@ -133,15 +198,15 @@ int I420AlphaToARGB(const uint8* src_y,
> 
> // Convert I420 with Alpha to preattenuated ABGR.
> LIBYUV_API
>-int I420AlphaToABGR(const uint8* src_y,
>+int I420AlphaToABGR(const uint8_t* src_y,
>                     int src_stride_y,
>-                    const uint8* src_u,
>+                    const uint8_t* src_u,
>                     int src_stride_u,
>-                    const uint8* src_v,
>+                    const uint8_t* src_v,
>                     int src_stride_v,
>-                    const uint8* src_a,
>+                    const uint8_t* src_a,
>                     int src_stride_a,
>-                    uint8* dst_abgr,
>+                    uint8_t* dst_abgr,
>                     int dst_stride_abgr,
>                     int width,
>                     int height,
>@@ -149,18 +214,18 @@ int I420AlphaToABGR(const uint8* src_y,
> 
> // Convert I400 (grey) to ARGB.  Reverse of ARGBToI400.
> LIBYUV_API
>-int I400ToARGB(const uint8* src_y,
>+int I400ToARGB(const uint8_t* src_y,
>                int src_stride_y,
>-               uint8* dst_argb,
>+               uint8_t* dst_argb,
>                int dst_stride_argb,
>                int width,
>                int height);
> 
> // Convert J400 (jpeg grey) to ARGB.
> LIBYUV_API
>-int J400ToARGB(const uint8* src_y,
>+int J400ToARGB(const uint8_t* src_y,
>                int src_stride_y,
>-               uint8* dst_argb,
>+               uint8_t* dst_argb,
>                int dst_stride_argb,
>                int width,
>                int height);
>@@ -170,180 +235,288 @@ int J400ToARGB(const uint8* src_y,
> 
> // Convert NV12 to ARGB.
> LIBYUV_API
>-int NV12ToARGB(const uint8* src_y,
>+int NV12ToARGB(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_uv,
>+               const uint8_t* src_uv,
>                int src_stride_uv,
>-               uint8* dst_argb,
>+               uint8_t* dst_argb,
>                int dst_stride_argb,
>                int width,
>                int height);
> 
> // Convert NV21 to ARGB.
> LIBYUV_API
>-int NV21ToARGB(const uint8* src_y,
>+int NV21ToARGB(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_vu,
>+               const uint8_t* src_vu,
>                int src_stride_vu,
>-               uint8* dst_argb,
>+               uint8_t* dst_argb,
>                int dst_stride_argb,
>                int width,
>                int height);
> 
>+// Convert NV12 to ABGR.
>+int NV12ToABGR(const uint8_t* src_y,
>+               int src_stride_y,
>+               const uint8_t* src_uv,
>+               int src_stride_uv,
>+               uint8_t* dst_abgr,
>+               int dst_stride_abgr,
>+               int width,
>+               int height);
>+
>+// Convert NV21 to ABGR.
>+LIBYUV_API
>+int NV21ToABGR(const uint8_t* src_y,
>+               int src_stride_y,
>+               const uint8_t* src_vu,
>+               int src_stride_vu,
>+               uint8_t* dst_abgr,
>+               int dst_stride_abgr,
>+               int width,
>+               int height);
>+
>+// Convert NV12 to RGB24.
>+LIBYUV_API
>+int NV12ToRGB24(const uint8_t* src_y,
>+                int src_stride_y,
>+                const uint8_t* src_uv,
>+                int src_stride_uv,
>+                uint8_t* dst_rgb24,
>+                int dst_stride_rgb24,
>+                int width,
>+                int height);
>+
>+// Convert NV21 to RGB24.
>+LIBYUV_API
>+int NV21ToRGB24(const uint8_t* src_y,
>+                int src_stride_y,
>+                const uint8_t* src_vu,
>+                int src_stride_vu,
>+                uint8_t* dst_rgb24,
>+                int dst_stride_rgb24,
>+                int width,
>+                int height);
>+
> // Convert M420 to ARGB.
> LIBYUV_API
>-int M420ToARGB(const uint8* src_m420,
>+int M420ToARGB(const uint8_t* src_m420,
>                int src_stride_m420,
>-               uint8* dst_argb,
>+               uint8_t* dst_argb,
>                int dst_stride_argb,
>                int width,
>                int height);
> 
> // Convert YUY2 to ARGB.
> LIBYUV_API
>-int YUY2ToARGB(const uint8* src_yuy2,
>+int YUY2ToARGB(const uint8_t* src_yuy2,
>                int src_stride_yuy2,
>-               uint8* dst_argb,
>+               uint8_t* dst_argb,
>                int dst_stride_argb,
>                int width,
>                int height);
> 
> // Convert UYVY to ARGB.
> LIBYUV_API
>-int UYVYToARGB(const uint8* src_uyvy,
>+int UYVYToARGB(const uint8_t* src_uyvy,
>                int src_stride_uyvy,
>-               uint8* dst_argb,
>+               uint8_t* dst_argb,
>                int dst_stride_argb,
>                int width,
>                int height);
> 
> // Convert J420 to ARGB.
> LIBYUV_API
>-int J420ToARGB(const uint8* src_y,
>+int J420ToARGB(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_argb,
>+               uint8_t* dst_argb,
>                int dst_stride_argb,
>                int width,
>                int height);
> 
> // Convert J422 to ARGB.
> LIBYUV_API
>-int J422ToARGB(const uint8* src_y,
>+int J422ToARGB(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_argb,
>+               uint8_t* dst_argb,
>                int dst_stride_argb,
>                int width,
>                int height);
> 
> // Convert J420 to ABGR.
> LIBYUV_API
>-int J420ToABGR(const uint8* src_y,
>+int J420ToABGR(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_abgr,
>+               uint8_t* dst_abgr,
>                int dst_stride_abgr,
>                int width,
>                int height);
> 
> // Convert J422 to ABGR.
> LIBYUV_API
>-int J422ToABGR(const uint8* src_y,
>+int J422ToABGR(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_abgr,
>+               uint8_t* dst_abgr,
>                int dst_stride_abgr,
>                int width,
>                int height);
> 
> // Convert H420 to ARGB.
> LIBYUV_API
>-int H420ToARGB(const uint8* src_y,
>+int H420ToARGB(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_argb,
>+               uint8_t* dst_argb,
>                int dst_stride_argb,
>                int width,
>                int height);
> 
> // Convert H422 to ARGB.
> LIBYUV_API
>-int H422ToARGB(const uint8* src_y,
>+int H422ToARGB(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_argb,
>+               uint8_t* dst_argb,
>                int dst_stride_argb,
>                int width,
>                int height);
> 
> // Convert H420 to ABGR.
> LIBYUV_API
>-int H420ToABGR(const uint8* src_y,
>+int H420ToABGR(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_abgr,
>+               uint8_t* dst_abgr,
>                int dst_stride_abgr,
>                int width,
>                int height);
> 
> // Convert H422 to ABGR.
> LIBYUV_API
>-int H422ToABGR(const uint8* src_y,
>+int H422ToABGR(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_abgr,
>+               uint8_t* dst_abgr,
>                int dst_stride_abgr,
>                int width,
>                int height);
> 
>+// Convert H010 to ARGB.
>+LIBYUV_API
>+int H010ToARGB(const uint16_t* src_y,
>+               int src_stride_y,
>+               const uint16_t* src_u,
>+               int src_stride_u,
>+               const uint16_t* src_v,
>+               int src_stride_v,
>+               uint8_t* dst_argb,
>+               int dst_stride_argb,
>+               int width,
>+               int height);
>+
>+// Convert I010 to AR30.
>+LIBYUV_API
>+int I010ToAR30(const uint16_t* src_y,
>+               int src_stride_y,
>+               const uint16_t* src_u,
>+               int src_stride_u,
>+               const uint16_t* src_v,
>+               int src_stride_v,
>+               uint8_t* dst_ar30,
>+               int dst_stride_ar30,
>+               int width,
>+               int height);
>+
>+// Convert H010 to AR30.
>+LIBYUV_API
>+int H010ToAR30(const uint16_t* src_y,
>+               int src_stride_y,
>+               const uint16_t* src_u,
>+               int src_stride_u,
>+               const uint16_t* src_v,
>+               int src_stride_v,
>+               uint8_t* dst_ar30,
>+               int dst_stride_ar30,
>+               int width,
>+               int height);
>+
>+// Convert I010 to AB30.
>+LIBYUV_API
>+int I010ToAB30(const uint16_t* src_y,
>+               int src_stride_y,
>+               const uint16_t* src_u,
>+               int src_stride_u,
>+               const uint16_t* src_v,
>+               int src_stride_v,
>+               uint8_t* dst_ab30,
>+               int dst_stride_ab30,
>+               int width,
>+               int height);
>+
>+// Convert H010 to AB30.
>+LIBYUV_API
>+int H010ToAB30(const uint16_t* src_y,
>+               int src_stride_y,
>+               const uint16_t* src_u,
>+               int src_stride_u,
>+               const uint16_t* src_v,
>+               int src_stride_v,
>+               uint8_t* dst_ab30,
>+               int dst_stride_ab30,
>+               int width,
>+               int height);
>+
> // BGRA little endian (argb in memory) to ARGB.
> LIBYUV_API
>-int BGRAToARGB(const uint8* src_frame,
>-               int src_stride_frame,
>-               uint8* dst_argb,
>+int BGRAToARGB(const uint8_t* src_bgra,
>+               int src_stride_bgra,
>+               uint8_t* dst_argb,
>                int dst_stride_argb,
>                int width,
>                int height);
> 
> // ABGR little endian (rgba in memory) to ARGB.
> LIBYUV_API
>-int ABGRToARGB(const uint8* src_frame,
>-               int src_stride_frame,
>-               uint8* dst_argb,
>+int ABGRToARGB(const uint8_t* src_abgr,
>+               int src_stride_abgr,
>+               uint8_t* dst_argb,
>                int dst_stride_argb,
>                int width,
>                int height);
> 
> // RGBA little endian (abgr in memory) to ARGB.
> LIBYUV_API
>-int RGBAToARGB(const uint8* src_frame,
>-               int src_stride_frame,
>-               uint8* dst_argb,
>+int RGBAToARGB(const uint8_t* src_rgba,
>+               int src_stride_rgba,
>+               uint8_t* dst_argb,
>                int dst_stride_argb,
>                int width,
>                int height);
>@@ -353,56 +526,88 @@ int RGBAToARGB(const uint8* src_frame,
> 
> // RGB little endian (bgr in memory) to ARGB.
> LIBYUV_API
>-int RGB24ToARGB(const uint8* src_frame,
>-                int src_stride_frame,
>-                uint8* dst_argb,
>+int RGB24ToARGB(const uint8_t* src_rgb24,
>+                int src_stride_rgb24,
>+                uint8_t* dst_argb,
>                 int dst_stride_argb,
>                 int width,
>                 int height);
> 
> // RGB big endian (rgb in memory) to ARGB.
> LIBYUV_API
>-int RAWToARGB(const uint8* src_frame,
>-              int src_stride_frame,
>-              uint8* dst_argb,
>+int RAWToARGB(const uint8_t* src_raw,
>+              int src_stride_raw,
>+              uint8_t* dst_argb,
>               int dst_stride_argb,
>               int width,
>               int height);
> 
> // RGB16 (RGBP fourcc) little endian to ARGB.
> LIBYUV_API
>-int RGB565ToARGB(const uint8* src_frame,
>-                 int src_stride_frame,
>-                 uint8* dst_argb,
>+int RGB565ToARGB(const uint8_t* src_rgb565,
>+                 int src_stride_rgb565,
>+                 uint8_t* dst_argb,
>                  int dst_stride_argb,
>                  int width,
>                  int height);
> 
> // RGB15 (RGBO fourcc) little endian to ARGB.
> LIBYUV_API
>-int ARGB1555ToARGB(const uint8* src_frame,
>-                   int src_stride_frame,
>-                   uint8* dst_argb,
>+int ARGB1555ToARGB(const uint8_t* src_argb1555,
>+                   int src_stride_argb1555,
>+                   uint8_t* dst_argb,
>                    int dst_stride_argb,
>                    int width,
>                    int height);
> 
> // RGB12 (R444 fourcc) little endian to ARGB.
> LIBYUV_API
>-int ARGB4444ToARGB(const uint8* src_frame,
>-                   int src_stride_frame,
>-                   uint8* dst_argb,
>+int ARGB4444ToARGB(const uint8_t* src_argb4444,
>+                   int src_stride_argb4444,
>+                   uint8_t* dst_argb,
>                    int dst_stride_argb,
>                    int width,
>                    int height);
> 
>+// Aliases
>+#define AB30ToARGB AR30ToABGR
>+#define AB30ToABGR AR30ToARGB
>+#define AB30ToAR30 AR30ToAB30
>+
>+// Convert AR30 To ARGB.
>+LIBYUV_API
>+int AR30ToARGB(const uint8_t* src_ar30,
>+               int src_stride_ar30,
>+               uint8_t* dst_argb,
>+               int dst_stride_argb,
>+               int width,
>+               int height);
>+
>+// Convert AR30 To ABGR.
>+LIBYUV_API
>+int AR30ToABGR(const uint8_t* src_ar30,
>+               int src_stride_ar30,
>+               uint8_t* dst_abgr,
>+               int dst_stride_abgr,
>+               int width,
>+               int height);
>+
>+// Convert AR30 To AB30.
>+LIBYUV_API
>+int AR30ToAB30(const uint8_t* src_ar30,
>+               int src_stride_ar30,
>+               uint8_t* dst_ab30,
>+               int dst_stride_ab30,
>+               int width,
>+               int height);
>+
> #ifdef HAVE_JPEG
> // src_width/height provided by capture
> // dst_width/height for clipping determine final size.
> LIBYUV_API
>-int MJPGToARGB(const uint8* sample,
>+int MJPGToARGB(const uint8_t* sample,
>                size_t sample_size,
>-               uint8* dst_argb,
>+               uint8_t* dst_argb,
>                int dst_stride_argb,
>                int src_width,
>                int src_height,
>@@ -410,8 +615,36 @@ int MJPGToARGB(const uint8* sample,
>                int dst_height);
> #endif
> 
>+// Convert Android420 to ARGB.
>+LIBYUV_API
>+int Android420ToARGB(const uint8_t* src_y,
>+                     int src_stride_y,
>+                     const uint8_t* src_u,
>+                     int src_stride_u,
>+                     const uint8_t* src_v,
>+                     int src_stride_v,
>+                     int src_pixel_stride_uv,
>+                     uint8_t* dst_argb,
>+                     int dst_stride_argb,
>+                     int width,
>+                     int height);
>+
>+// Convert Android420 to ABGR.
>+LIBYUV_API
>+int Android420ToABGR(const uint8_t* src_y,
>+                     int src_stride_y,
>+                     const uint8_t* src_u,
>+                     int src_stride_u,
>+                     const uint8_t* src_v,
>+                     int src_stride_v,
>+                     int src_pixel_stride_uv,
>+                     uint8_t* dst_abgr,
>+                     int dst_stride_abgr,
>+                     int width,
>+                     int height);
>+
> // Convert camera sample to ARGB with cropping, rotation and vertical flip.
>-// "src_size" is needed to parse MJPG.
>+// "sample_size" is needed to parse MJPG.
> // "dst_stride_argb" number of bytes in a row of the dst_argb plane.
> //   Normally this would be the same as dst_width, with recommended alignment
> //   to 16 bytes for better efficiency.
>@@ -430,12 +663,12 @@ int MJPGToARGB(const uint8* sample,
> //    Must be less than or equal to src_width/src_height
> //    Cropping parameters are pre-rotation.
> // "rotation" can be 0, 90, 180 or 270.
>-// "format" is a fourcc. ie 'I420', 'YUY2'
>+// "fourcc" is a fourcc. ie 'I420', 'YUY2'
> // Returns 0 for successful; -1 for invalid parameter. Non-zero for failure.
> LIBYUV_API
>-int ConvertToARGB(const uint8* src_frame,
>-                  size_t src_size,
>-                  uint8* dst_argb,
>+int ConvertToARGB(const uint8_t* sample,
>+                  size_t sample_size,
>+                  uint8_t* dst_argb,
>                   int dst_stride_argb,
>                   int crop_x,
>                   int crop_y,
>@@ -444,7 +677,7 @@ int ConvertToARGB(const uint8* src_frame,
>                   int crop_width,
>                   int crop_height,
>                   enum RotationMode rotation,
>-                  uint32 format);
>+                  uint32_t fourcc);
> 
> #ifdef __cplusplus
> }  // extern "C"
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/convert_from.h b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/convert_from.h
>index 237f68f57aa..5cd8a4bfc04 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/convert_from.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/convert_from.h
>@@ -21,218 +21,233 @@ extern "C" {
> 
> // See Also convert.h for conversions from formats to I420.
> 
>-// I420Copy in convert to I420ToI420.
>+// Convert 8 bit YUV to 10 bit.
>+#define H420ToH010 I420ToI010
>+int I420ToI010(const uint8_t* src_y,
>+               int src_stride_y,
>+               const uint8_t* src_u,
>+               int src_stride_u,
>+               const uint8_t* src_v,
>+               int src_stride_v,
>+               uint16_t* dst_y,
>+               int dst_stride_y,
>+               uint16_t* dst_u,
>+               int dst_stride_u,
>+               uint16_t* dst_v,
>+               int dst_stride_v,
>+               int width,
>+               int height);
> 
> LIBYUV_API
>-int I420ToI422(const uint8* src_y,
>+int I420ToI422(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_u,
>+               uint8_t* dst_u,
>                int dst_stride_u,
>-               uint8* dst_v,
>+               uint8_t* dst_v,
>                int dst_stride_v,
>                int width,
>                int height);
> 
> LIBYUV_API
>-int I420ToI444(const uint8* src_y,
>+int I420ToI444(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_u,
>+               uint8_t* dst_u,
>                int dst_stride_u,
>-               uint8* dst_v,
>+               uint8_t* dst_v,
>                int dst_stride_v,
>                int width,
>                int height);
> 
> // Copy to I400. Source can be I420, I422, I444, I400, NV12 or NV21.
> LIBYUV_API
>-int I400Copy(const uint8* src_y,
>+int I400Copy(const uint8_t* src_y,
>              int src_stride_y,
>-             uint8* dst_y,
>+             uint8_t* dst_y,
>              int dst_stride_y,
>              int width,
>              int height);
> 
> LIBYUV_API
>-int I420ToNV12(const uint8* src_y,
>+int I420ToNV12(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_uv,
>+               uint8_t* dst_uv,
>                int dst_stride_uv,
>                int width,
>                int height);
> 
> LIBYUV_API
>-int I420ToNV21(const uint8* src_y,
>+int I420ToNV21(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_vu,
>+               uint8_t* dst_vu,
>                int dst_stride_vu,
>                int width,
>                int height);
> 
> LIBYUV_API
>-int I420ToYUY2(const uint8* src_y,
>+int I420ToYUY2(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_frame,
>-               int dst_stride_frame,
>+               uint8_t* dst_yuy2,
>+               int dst_stride_yuy2,
>                int width,
>                int height);
> 
> LIBYUV_API
>-int I420ToUYVY(const uint8* src_y,
>+int I420ToUYVY(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_frame,
>-               int dst_stride_frame,
>+               uint8_t* dst_uyvy,
>+               int dst_stride_uyvy,
>                int width,
>                int height);
> 
> LIBYUV_API
>-int I420ToARGB(const uint8* src_y,
>+int I420ToARGB(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_argb,
>+               uint8_t* dst_argb,
>                int dst_stride_argb,
>                int width,
>                int height);
> 
> LIBYUV_API
>-int I420ToBGRA(const uint8* src_y,
>+int I420ToBGRA(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_argb,
>-               int dst_stride_argb,
>+               uint8_t* dst_bgra,
>+               int dst_stride_bgra,
>                int width,
>                int height);
> 
> LIBYUV_API
>-int I420ToABGR(const uint8* src_y,
>+int I420ToABGR(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_argb,
>-               int dst_stride_argb,
>+               uint8_t* dst_abgr,
>+               int dst_stride_abgr,
>                int width,
>                int height);
> 
> LIBYUV_API
>-int I420ToRGBA(const uint8* src_y,
>+int I420ToRGBA(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_rgba,
>+               uint8_t* dst_rgba,
>                int dst_stride_rgba,
>                int width,
>                int height);
> 
> LIBYUV_API
>-int I420ToRGB24(const uint8* src_y,
>+int I420ToRGB24(const uint8_t* src_y,
>                 int src_stride_y,
>-                const uint8* src_u,
>+                const uint8_t* src_u,
>                 int src_stride_u,
>-                const uint8* src_v,
>+                const uint8_t* src_v,
>                 int src_stride_v,
>-                uint8* dst_frame,
>-                int dst_stride_frame,
>+                uint8_t* dst_rgb24,
>+                int dst_stride_rgb24,
>                 int width,
>                 int height);
> 
> LIBYUV_API
>-int I420ToRAW(const uint8* src_y,
>+int I420ToRAW(const uint8_t* src_y,
>               int src_stride_y,
>-              const uint8* src_u,
>+              const uint8_t* src_u,
>               int src_stride_u,
>-              const uint8* src_v,
>+              const uint8_t* src_v,
>               int src_stride_v,
>-              uint8* dst_frame,
>-              int dst_stride_frame,
>+              uint8_t* dst_raw,
>+              int dst_stride_raw,
>               int width,
>               int height);
> 
> LIBYUV_API
>-int H420ToRGB24(const uint8* src_y,
>+int H420ToRGB24(const uint8_t* src_y,
>                 int src_stride_y,
>-                const uint8* src_u,
>+                const uint8_t* src_u,
>                 int src_stride_u,
>-                const uint8* src_v,
>+                const uint8_t* src_v,
>                 int src_stride_v,
>-                uint8* dst_frame,
>-                int dst_stride_frame,
>+                uint8_t* dst_rgb24,
>+                int dst_stride_rgb24,
>                 int width,
>                 int height);
> 
> LIBYUV_API
>-int H420ToRAW(const uint8* src_y,
>+int H420ToRAW(const uint8_t* src_y,
>               int src_stride_y,
>-              const uint8* src_u,
>+              const uint8_t* src_u,
>               int src_stride_u,
>-              const uint8* src_v,
>+              const uint8_t* src_v,
>               int src_stride_v,
>-              uint8* dst_frame,
>-              int dst_stride_frame,
>+              uint8_t* dst_raw,
>+              int dst_stride_raw,
>               int width,
>               int height);
> 
> LIBYUV_API
>-int I420ToRGB565(const uint8* src_y,
>+int I420ToRGB565(const uint8_t* src_y,
>                  int src_stride_y,
>-                 const uint8* src_u,
>+                 const uint8_t* src_u,
>                  int src_stride_u,
>-                 const uint8* src_v,
>+                 const uint8_t* src_v,
>                  int src_stride_v,
>-                 uint8* dst_frame,
>-                 int dst_stride_frame,
>+                 uint8_t* dst_rgb565,
>+                 int dst_stride_rgb565,
>                  int width,
>                  int height);
> 
> LIBYUV_API
>-int I422ToRGB565(const uint8* src_y,
>+int I422ToRGB565(const uint8_t* src_y,
>                  int src_stride_y,
>-                 const uint8* src_u,
>+                 const uint8_t* src_u,
>                  int src_stride_u,
>-                 const uint8* src_v,
>+                 const uint8_t* src_v,
>                  int src_stride_v,
>-                 uint8* dst_frame,
>-                 int dst_stride_frame,
>+                 uint8_t* dst_rgb565,
>+                 int dst_stride_rgb565,
>                  int width,
>                  int height);
> 
>@@ -241,57 +256,83 @@ int I422ToRGB565(const uint8* src_y,
> // The order of the dither matrix is first byte is upper left.
> 
> LIBYUV_API
>-int I420ToRGB565Dither(const uint8* src_y,
>+int I420ToRGB565Dither(const uint8_t* src_y,
>                        int src_stride_y,
>-                       const uint8* src_u,
>+                       const uint8_t* src_u,
>                        int src_stride_u,
>-                       const uint8* src_v,
>+                       const uint8_t* src_v,
>                        int src_stride_v,
>-                       uint8* dst_frame,
>-                       int dst_stride_frame,
>-                       const uint8* dither4x4,
>+                       uint8_t* dst_rgb565,
>+                       int dst_stride_rgb565,
>+                       const uint8_t* dither4x4,
>                        int width,
>                        int height);
> 
> LIBYUV_API
>-int I420ToARGB1555(const uint8* src_y,
>+int I420ToARGB1555(const uint8_t* src_y,
>                    int src_stride_y,
>-                   const uint8* src_u,
>+                   const uint8_t* src_u,
>                    int src_stride_u,
>-                   const uint8* src_v,
>+                   const uint8_t* src_v,
>                    int src_stride_v,
>-                   uint8* dst_frame,
>-                   int dst_stride_frame,
>+                   uint8_t* dst_argb1555,
>+                   int dst_stride_argb1555,
>                    int width,
>                    int height);
> 
> LIBYUV_API
>-int I420ToARGB4444(const uint8* src_y,
>+int I420ToARGB4444(const uint8_t* src_y,
>                    int src_stride_y,
>-                   const uint8* src_u,
>+                   const uint8_t* src_u,
>                    int src_stride_u,
>-                   const uint8* src_v,
>+                   const uint8_t* src_v,
>                    int src_stride_v,
>-                   uint8* dst_frame,
>-                   int dst_stride_frame,
>+                   uint8_t* dst_argb4444,
>+                   int dst_stride_argb4444,
>                    int width,
>                    int height);
> 
>+// Convert I420 to AR30.
>+LIBYUV_API
>+int I420ToAR30(const uint8_t* src_y,
>+               int src_stride_y,
>+               const uint8_t* src_u,
>+               int src_stride_u,
>+               const uint8_t* src_v,
>+               int src_stride_v,
>+               uint8_t* dst_ar30,
>+               int dst_stride_ar30,
>+               int width,
>+               int height);
>+
>+// Convert H420 to AR30.
>+LIBYUV_API
>+int H420ToAR30(const uint8_t* src_y,
>+               int src_stride_y,
>+               const uint8_t* src_u,
>+               int src_stride_u,
>+               const uint8_t* src_v,
>+               int src_stride_v,
>+               uint8_t* dst_ar30,
>+               int dst_stride_ar30,
>+               int width,
>+               int height);
>+
> // Convert I420 to specified format.
> // "dst_sample_stride" is bytes in a row for the destination. Pass 0 if the
> //    buffer has contiguous rows. Can be negative. A multiple of 16 is optimal.
> LIBYUV_API
>-int ConvertFromI420(const uint8* y,
>+int ConvertFromI420(const uint8_t* y,
>                     int y_stride,
>-                    const uint8* u,
>+                    const uint8_t* u,
>                     int u_stride,
>-                    const uint8* v,
>+                    const uint8_t* v,
>                     int v_stride,
>-                    uint8* dst_sample,
>+                    uint8_t* dst_sample,
>                     int dst_sample_stride,
>                     int width,
>                     int height,
>-                    uint32 format);
>+                    uint32_t fourcc);
> 
> #ifdef __cplusplus
> }  // extern "C"
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/convert_from_argb.h b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/convert_from_argb.h
>index 50722d76102..05c815a093e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/convert_from_argb.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/convert_from_argb.h
>@@ -21,63 +21,85 @@ extern "C" {
> // Copy ARGB to ARGB.
> #define ARGBToARGB ARGBCopy
> LIBYUV_API
>-int ARGBCopy(const uint8* src_argb,
>+int ARGBCopy(const uint8_t* src_argb,
>              int src_stride_argb,
>-             uint8* dst_argb,
>+             uint8_t* dst_argb,
>              int dst_stride_argb,
>              int width,
>              int height);
> 
> // Convert ARGB To BGRA.
> LIBYUV_API
>-int ARGBToBGRA(const uint8* src_argb,
>+int ARGBToBGRA(const uint8_t* src_argb,
>                int src_stride_argb,
>-               uint8* dst_bgra,
>+               uint8_t* dst_bgra,
>                int dst_stride_bgra,
>                int width,
>                int height);
> 
> // Convert ARGB To ABGR.
> LIBYUV_API
>-int ARGBToABGR(const uint8* src_argb,
>+int ARGBToABGR(const uint8_t* src_argb,
>                int src_stride_argb,
>-               uint8* dst_abgr,
>+               uint8_t* dst_abgr,
>                int dst_stride_abgr,
>                int width,
>                int height);
> 
> // Convert ARGB To RGBA.
> LIBYUV_API
>-int ARGBToRGBA(const uint8* src_argb,
>+int ARGBToRGBA(const uint8_t* src_argb,
>                int src_stride_argb,
>-               uint8* dst_rgba,
>+               uint8_t* dst_rgba,
>                int dst_stride_rgba,
>                int width,
>                int height);
> 
>+// Aliases
>+#define ARGBToAB30 ABGRToAR30
>+#define ABGRToAB30 ARGBToAR30
>+
>+// Convert ABGR To AR30.
>+LIBYUV_API
>+int ABGRToAR30(const uint8_t* src_abgr,
>+               int src_stride_abgr,
>+               uint8_t* dst_ar30,
>+               int dst_stride_ar30,
>+               int width,
>+               int height);
>+
>+// Convert ARGB To AR30.
>+LIBYUV_API
>+int ARGBToAR30(const uint8_t* src_argb,
>+               int src_stride_argb,
>+               uint8_t* dst_ar30,
>+               int dst_stride_ar30,
>+               int width,
>+               int height);
>+
> // Convert ARGB To RGB24.
> LIBYUV_API
>-int ARGBToRGB24(const uint8* src_argb,
>+int ARGBToRGB24(const uint8_t* src_argb,
>                 int src_stride_argb,
>-                uint8* dst_rgb24,
>+                uint8_t* dst_rgb24,
>                 int dst_stride_rgb24,
>                 int width,
>                 int height);
> 
> // Convert ARGB To RAW.
> LIBYUV_API
>-int ARGBToRAW(const uint8* src_argb,
>+int ARGBToRAW(const uint8_t* src_argb,
>               int src_stride_argb,
>-              uint8* dst_rgb,
>-              int dst_stride_rgb,
>+              uint8_t* dst_raw,
>+              int dst_stride_raw,
>               int width,
>               int height);
> 
> // Convert ARGB To RGB565.
> LIBYUV_API
>-int ARGBToRGB565(const uint8* src_argb,
>+int ARGBToRGB565(const uint8_t* src_argb,
>                  int src_stride_argb,
>-                 uint8* dst_rgb565,
>+                 uint8_t* dst_rgb565,
>                  int dst_stride_rgb565,
>                  int width,
>                  int height);
>@@ -86,173 +108,173 @@ int ARGBToRGB565(const uint8* src_argb,
> // Values in dither matrix from 0 to 7 recommended.
> // The order of the dither matrix is first byte is upper left.
> // TODO(fbarchard): Consider pointer to 2d array for dither4x4.
>-// const uint8(*dither)[4][4];
>+// const uint8_t(*dither)[4][4];
> LIBYUV_API
>-int ARGBToRGB565Dither(const uint8* src_argb,
>+int ARGBToRGB565Dither(const uint8_t* src_argb,
>                        int src_stride_argb,
>-                       uint8* dst_rgb565,
>+                       uint8_t* dst_rgb565,
>                        int dst_stride_rgb565,
>-                       const uint8* dither4x4,
>+                       const uint8_t* dither4x4,
>                        int width,
>                        int height);
> 
> // Convert ARGB To ARGB1555.
> LIBYUV_API
>-int ARGBToARGB1555(const uint8* src_argb,
>+int ARGBToARGB1555(const uint8_t* src_argb,
>                    int src_stride_argb,
>-                   uint8* dst_argb1555,
>+                   uint8_t* dst_argb1555,
>                    int dst_stride_argb1555,
>                    int width,
>                    int height);
> 
> // Convert ARGB To ARGB4444.
> LIBYUV_API
>-int ARGBToARGB4444(const uint8* src_argb,
>+int ARGBToARGB4444(const uint8_t* src_argb,
>                    int src_stride_argb,
>-                   uint8* dst_argb4444,
>+                   uint8_t* dst_argb4444,
>                    int dst_stride_argb4444,
>                    int width,
>                    int height);
> 
> // Convert ARGB To I444.
> LIBYUV_API
>-int ARGBToI444(const uint8* src_argb,
>+int ARGBToI444(const uint8_t* src_argb,
>                int src_stride_argb,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_u,
>+               uint8_t* dst_u,
>                int dst_stride_u,
>-               uint8* dst_v,
>+               uint8_t* dst_v,
>                int dst_stride_v,
>                int width,
>                int height);
> 
> // Convert ARGB To I422.
> LIBYUV_API
>-int ARGBToI422(const uint8* src_argb,
>+int ARGBToI422(const uint8_t* src_argb,
>                int src_stride_argb,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_u,
>+               uint8_t* dst_u,
>                int dst_stride_u,
>-               uint8* dst_v,
>+               uint8_t* dst_v,
>                int dst_stride_v,
>                int width,
>                int height);
> 
> // Convert ARGB To I420. (also in convert.h)
> LIBYUV_API
>-int ARGBToI420(const uint8* src_argb,
>+int ARGBToI420(const uint8_t* src_argb,
>                int src_stride_argb,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_u,
>+               uint8_t* dst_u,
>                int dst_stride_u,
>-               uint8* dst_v,
>+               uint8_t* dst_v,
>                int dst_stride_v,
>                int width,
>                int height);
> 
> // Convert ARGB to J420. (JPeg full range I420).
> LIBYUV_API
>-int ARGBToJ420(const uint8* src_argb,
>+int ARGBToJ420(const uint8_t* src_argb,
>                int src_stride_argb,
>-               uint8* dst_yj,
>+               uint8_t* dst_yj,
>                int dst_stride_yj,
>-               uint8* dst_u,
>+               uint8_t* dst_u,
>                int dst_stride_u,
>-               uint8* dst_v,
>+               uint8_t* dst_v,
>                int dst_stride_v,
>                int width,
>                int height);
> 
> // Convert ARGB to J422.
> LIBYUV_API
>-int ARGBToJ422(const uint8* src_argb,
>+int ARGBToJ422(const uint8_t* src_argb,
>                int src_stride_argb,
>-               uint8* dst_yj,
>+               uint8_t* dst_yj,
>                int dst_stride_yj,
>-               uint8* dst_u,
>+               uint8_t* dst_u,
>                int dst_stride_u,
>-               uint8* dst_v,
>+               uint8_t* dst_v,
>                int dst_stride_v,
>                int width,
>                int height);
> 
> // Convert ARGB to J400. (JPeg full range).
> LIBYUV_API
>-int ARGBToJ400(const uint8* src_argb,
>+int ARGBToJ400(const uint8_t* src_argb,
>                int src_stride_argb,
>-               uint8* dst_yj,
>+               uint8_t* dst_yj,
>                int dst_stride_yj,
>                int width,
>                int height);
> 
> // Convert ARGB to I400.
> LIBYUV_API
>-int ARGBToI400(const uint8* src_argb,
>+int ARGBToI400(const uint8_t* src_argb,
>                int src_stride_argb,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>                int width,
>                int height);
> 
> // Convert ARGB to G. (Reverse of J400toARGB, which replicates G back to ARGB)
> LIBYUV_API
>-int ARGBToG(const uint8* src_argb,
>+int ARGBToG(const uint8_t* src_argb,
>             int src_stride_argb,
>-            uint8* dst_g,
>+            uint8_t* dst_g,
>             int dst_stride_g,
>             int width,
>             int height);
> 
> // Convert ARGB To NV12.
> LIBYUV_API
>-int ARGBToNV12(const uint8* src_argb,
>+int ARGBToNV12(const uint8_t* src_argb,
>                int src_stride_argb,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_uv,
>+               uint8_t* dst_uv,
>                int dst_stride_uv,
>                int width,
>                int height);
> 
> // Convert ARGB To NV21.
> LIBYUV_API
>-int ARGBToNV21(const uint8* src_argb,
>+int ARGBToNV21(const uint8_t* src_argb,
>                int src_stride_argb,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_vu,
>+               uint8_t* dst_vu,
>                int dst_stride_vu,
>                int width,
>                int height);
> 
> // Convert ARGB To NV21.
> LIBYUV_API
>-int ARGBToNV21(const uint8* src_argb,
>+int ARGBToNV21(const uint8_t* src_argb,
>                int src_stride_argb,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_vu,
>+               uint8_t* dst_vu,
>                int dst_stride_vu,
>                int width,
>                int height);
> 
> // Convert ARGB To YUY2.
> LIBYUV_API
>-int ARGBToYUY2(const uint8* src_argb,
>+int ARGBToYUY2(const uint8_t* src_argb,
>                int src_stride_argb,
>-               uint8* dst_yuy2,
>+               uint8_t* dst_yuy2,
>                int dst_stride_yuy2,
>                int width,
>                int height);
> 
> // Convert ARGB To UYVY.
> LIBYUV_API
>-int ARGBToUYVY(const uint8* src_argb,
>+int ARGBToUYVY(const uint8_t* src_argb,
>                int src_stride_argb,
>-               uint8* dst_uyvy,
>+               uint8_t* dst_uyvy,
>                int dst_stride_uyvy,
>                int width,
>                int height);
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/cpu_id.h b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/cpu_id.h
>index c2e9bbbd954..91480c68b01 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/cpu_id.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/cpu_id.h
>@@ -47,8 +47,7 @@ static const int kCpuHasAVX512VPOPCNTDQ = 0x100000;
> 
> // These flags are only valid on MIPS processors.
> static const int kCpuHasMIPS = 0x200000;
>-static const int kCpuHasDSPR2 = 0x400000;
>-static const int kCpuHasMSA = 0x800000;
>+static const int kCpuHasMSA = 0x400000;
> 
> // Optional init function. TestCpuFlag does an auto-init.
> // Returns cpu_info flags.
>@@ -85,7 +84,7 @@ int MaskCpuFlags(int enable_flags);
> // eax is the info type that you want.
> // ecx is typically the cpu number, and should normally be zero.
> LIBYUV_API
>-void CpuId(int eax, int ecx, int* cpu_info);
>+void CpuId(int info_eax, int info_ecx, int* cpu_info);
> 
> #ifdef __cplusplus
> }  // extern "C"
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/macros_msa.h b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/macros_msa.h
>index 61be352e3af..921eb0714d6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/macros_msa.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/macros_msa.h
>@@ -16,38 +16,38 @@
> #include <stdint.h>
> 
> #if (__mips_isa_rev >= 6)
>-#define LW(psrc)                                    \
>-  ({                                                \
>-    uint8* psrc_lw_m = (uint8*)(psrc); /* NOLINT */ \
>-    uint32 val_m;                                   \
>-    asm volatile("lw  %[val_m],  %[psrc_lw_m]  \n"  \
>-                 : [val_m] "=r"(val_m)              \
>-                 : [psrc_lw_m] "m"(*psrc_lw_m));    \
>-    val_m;                                          \
>+#define LW(psrc)                                        \
>+  ({                                                    \
>+    uint8_t* psrc_lw_m = (uint8_t*)(psrc); /* NOLINT */ \
>+    uint32_t val_m;                                     \
>+    asm volatile("lw  %[val_m],  %[psrc_lw_m]  \n"      \
>+                 : [val_m] "=r"(val_m)                  \
>+                 : [psrc_lw_m] "m"(*psrc_lw_m));        \
>+    val_m;                                              \
>   })
> 
> #if (__mips == 64)
>-#define LD(psrc)                                    \
>-  ({                                                \
>-    uint8* psrc_ld_m = (uint8*)(psrc); /* NOLINT */ \
>-    uint64 val_m = 0;                               \
>-    asm volatile("ld  %[val_m],  %[psrc_ld_m]  \n"  \
>-                 : [val_m] "=r"(val_m)              \
>-                 : [psrc_ld_m] "m"(*psrc_ld_m));    \
>-    val_m;                                          \
>+#define LD(psrc)                                        \
>+  ({                                                    \
>+    uint8_t* psrc_ld_m = (uint8_t*)(psrc); /* NOLINT */ \
>+    uint64_t val_m = 0;                                 \
>+    asm volatile("ld  %[val_m],  %[psrc_ld_m]  \n"      \
>+                 : [val_m] "=r"(val_m)                  \
>+                 : [psrc_ld_m] "m"(*psrc_ld_m));        \
>+    val_m;                                              \
>   })
> #else  // !(__mips == 64)
>-#define LD(psrc)                                                       \
>-  ({                                                                   \
>-    uint8* psrc_ld_m = (uint8*)(psrc); /* NOLINT */                    \
>-    uint32 val0_m, val1_m;                                             \
>-    uint64 val_m = 0;                                                  \
>-    val0_m = LW(psrc_ld_m);                                            \
>-    val1_m = LW(psrc_ld_m + 4);                                        \
>-    val_m = (uint64)(val1_m);                             /* NOLINT */ \
>-    val_m = (uint64)((val_m << 32) & 0xFFFFFFFF00000000); /* NOLINT */ \
>-    val_m = (uint64)(val_m | (uint64)val0_m);             /* NOLINT */ \
>-    val_m;                                                             \
>+#define LD(psrc)                                                         \
>+  ({                                                                     \
>+    uint8_t* psrc_ld_m = (uint8_t*)(psrc); /* NOLINT */                  \
>+    uint32_t val0_m, val1_m;                                             \
>+    uint64_t val_m = 0;                                                  \
>+    val0_m = LW(psrc_ld_m);                                              \
>+    val1_m = LW(psrc_ld_m + 4);                                          \
>+    val_m = (uint64_t)(val1_m);                             /* NOLINT */ \
>+    val_m = (uint64_t)((val_m << 32) & 0xFFFFFFFF00000000); /* NOLINT */ \
>+    val_m = (uint64_t)(val_m | (uint64_t)val0_m);           /* NOLINT */ \
>+    val_m;                                                               \
>   })
> #endif  // (__mips == 64)
> 
>@@ -81,38 +81,38 @@
>   })
> #endif  // !(__mips == 64)
> #else   // !(__mips_isa_rev >= 6)
>-#define LW(psrc)                                    \
>-  ({                                                \
>-    uint8* psrc_lw_m = (uint8*)(psrc); /* NOLINT */ \
>-    uint32 val_m;                                   \
>-    asm volatile("ulw  %[val_m],  %[psrc_lw_m]  \n" \
>-                 : [val_m] "=r"(val_m)              \
>-                 : [psrc_lw_m] "m"(*psrc_lw_m));    \
>-    val_m;                                          \
>+#define LW(psrc)                                        \
>+  ({                                                    \
>+    uint8_t* psrc_lw_m = (uint8_t*)(psrc); /* NOLINT */ \
>+    uint32_t val_m;                                     \
>+    asm volatile("ulw  %[val_m],  %[psrc_lw_m]  \n"     \
>+                 : [val_m] "=r"(val_m)                  \
>+                 : [psrc_lw_m] "m"(*psrc_lw_m));        \
>+    val_m;                                              \
>   })
> 
> #if (__mips == 64)
>-#define LD(psrc)                                    \
>-  ({                                                \
>-    uint8* psrc_ld_m = (uint8*)(psrc); /* NOLINT */ \
>-    uint64 val_m = 0;                               \
>-    asm volatile("uld  %[val_m],  %[psrc_ld_m]  \n" \
>-                 : [val_m] "=r"(val_m)              \
>-                 : [psrc_ld_m] "m"(*psrc_ld_m));    \
>-    val_m;                                          \
>+#define LD(psrc)                                        \
>+  ({                                                    \
>+    uint8_t* psrc_ld_m = (uint8_t*)(psrc); /* NOLINT */ \
>+    uint64_t val_m = 0;                                 \
>+    asm volatile("uld  %[val_m],  %[psrc_ld_m]  \n"     \
>+                 : [val_m] "=r"(val_m)                  \
>+                 : [psrc_ld_m] "m"(*psrc_ld_m));        \
>+    val_m;                                              \
>   })
> #else  // !(__mips == 64)
>-#define LD(psrc)                                                       \
>-  ({                                                                   \
>-    uint8* psrc_ld_m = (uint8*)(psrc); /* NOLINT */                    \
>-    uint32 val0_m, val1_m;                                             \
>-    uint64 val_m = 0;                                                  \
>-    val0_m = LW(psrc_ld_m);                                            \
>-    val1_m = LW(psrc_ld_m + 4);                                        \
>-    val_m = (uint64)(val1_m);                             /* NOLINT */ \
>-    val_m = (uint64)((val_m << 32) & 0xFFFFFFFF00000000); /* NOLINT */ \
>-    val_m = (uint64)(val_m | (uint64)val0_m);             /* NOLINT */ \
>-    val_m;                                                             \
>+#define LD(psrc)                                                         \
>+  ({                                                                     \
>+    uint8_t* psrc_ld_m = (uint8_t*)(psrc); /* NOLINT */                  \
>+    uint32_t val0_m, val1_m;                                             \
>+    uint64_t val_m = 0;                                                  \
>+    val0_m = LW(psrc_ld_m);                                              \
>+    val1_m = LW(psrc_ld_m + 4);                                          \
>+    val_m = (uint64_t)(val1_m);                             /* NOLINT */ \
>+    val_m = (uint64_t)((val_m << 32) & 0xFFFFFFFF00000000); /* NOLINT */ \
>+    val_m = (uint64_t)(val_m | (uint64_t)val0_m);           /* NOLINT */ \
>+    val_m;                                                               \
>   })
> #endif  // (__mips == 64)
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/mjpeg_decoder.h b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/mjpeg_decoder.h
>index 8a4f282205b..6c12633387f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/mjpeg_decoder.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/mjpeg_decoder.h
>@@ -26,13 +26,13 @@ namespace libyuv {
> extern "C" {
> #endif
> 
>-LIBYUV_BOOL ValidateJpeg(const uint8* sample, size_t sample_size);
>+LIBYUV_BOOL ValidateJpeg(const uint8_t* sample, size_t sample_size);
> 
> #ifdef __cplusplus
> }  // extern "C"
> #endif
> 
>-static const uint32 kUnknownDataSize = 0xFFFFFFFF;
>+static const uint32_t kUnknownDataSize = 0xFFFFFFFF;
> 
> enum JpegSubsamplingType {
>   kJpegYuv420,
>@@ -43,7 +43,7 @@ enum JpegSubsamplingType {
> };
> 
> struct Buffer {
>-  const uint8* data;
>+  const uint8_t* data;
>   int len;
> };
> 
>@@ -65,7 +65,7 @@ struct SetJmpErrorMgr;
> class LIBYUV_API MJpegDecoder {
>  public:
>   typedef void (*CallbackFunction)(void* opaque,
>-                                   const uint8* const* data,
>+                                   const uint8_t* const* data,
>                                    const int* strides,
>                                    int rows);
> 
>@@ -85,7 +85,7 @@ class LIBYUV_API MJpegDecoder {
>   // If return value is LIBYUV_TRUE, then the values for all the following
>   // getters are populated.
>   // src_len is the size of the compressed mjpeg frame in bytes.
>-  LIBYUV_BOOL LoadFrame(const uint8* src, size_t src_len);
>+  LIBYUV_BOOL LoadFrame(const uint8_t* src, size_t src_len);
> 
>   // Returns width of the last loaded frame in pixels.
>   int GetWidth();
>@@ -138,7 +138,7 @@ class LIBYUV_API MJpegDecoder {
>   // at least GetComponentSize(i). The pointers in planes are incremented
>   // to point to after the end of the written data.
>   // TODO(fbarchard): Add dst_x, dst_y to allow specific rect to be decoded.
>-  LIBYUV_BOOL DecodeToBuffers(uint8** planes, int dst_width, int dst_height);
>+  LIBYUV_BOOL DecodeToBuffers(uint8_t** planes, int dst_width, int dst_height);
> 
>   // Decodes the entire image and passes the data via repeated calls to a
>   // callback function. Each call will get the data for a whole number of
>@@ -162,14 +162,14 @@ class LIBYUV_API MJpegDecoder {
>   LIBYUV_BOOL StartDecode();
>   LIBYUV_BOOL FinishDecode();
> 
>-  void SetScanlinePointers(uint8** data);
>+  void SetScanlinePointers(uint8_t** data);
>   LIBYUV_BOOL DecodeImcuRow();
> 
>   int GetComponentScanlinePadding(int component);
> 
>   // A buffer holding the input data for a frame.
>-  Buffer buf_;
>-  BufferVector buf_vec_;
>+  Buffer buf_{};
>+  BufferVector buf_vec_{};
> 
>   jpeg_decompress_struct* decompress_struct_;
>   jpeg_source_mgr* source_mgr_;
>@@ -181,12 +181,12 @@ class LIBYUV_API MJpegDecoder {
> 
>   // Temporaries used to point to scanline outputs.
>   int num_outbufs_;  // Outermost size of all arrays below.
>-  uint8*** scanlines_;
>-  int* scanlines_sizes_;
>+  uint8_t*** scanlines_{};
>+  int* scanlines_sizes_{};
>   // Temporary buffer used for decoding when we can't decode directly to the
>   // output buffers. Large enough for just one iMCU row.
>-  uint8** databuf_;
>-  int* databuf_strides_;
>+  uint8_t** databuf_{};
>+  int* databuf_strides_{};
> };
> 
> }  // namespace libyuv
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/planar_functions.h b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/planar_functions.h
>index c91501a9c2c..91137baba25 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/planar_functions.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/planar_functions.h
>@@ -22,84 +22,120 @@ namespace libyuv {
> extern "C" {
> #endif
> 
>+// TODO(fbarchard): Move cpu macros to row.h
>+#if defined(__pnacl__) || defined(__CLR_VER) ||            \
>+    (defined(__native_client__) && defined(__x86_64__)) || \
>+    (defined(__i386__) && !defined(__SSE__) && !defined(__clang__))
>+#define LIBYUV_DISABLE_X86
>+#endif
>+// MemorySanitizer does not support assembly code yet. http://crbug.com/344505
>+#if defined(__has_feature)
>+#if __has_feature(memory_sanitizer)
>+#define LIBYUV_DISABLE_X86
>+#endif
>+#endif
>+// The following are available on all x86 platforms:
>+#if !defined(LIBYUV_DISABLE_X86) && \
>+    (defined(_M_IX86) || defined(__x86_64__) || defined(__i386__))
>+#define HAS_ARGBAFFINEROW_SSE2
>+#endif
>+
> // Copy a plane of data.
> LIBYUV_API
>-void CopyPlane(const uint8* src_y,
>+void CopyPlane(const uint8_t* src_y,
>                int src_stride_y,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>                int width,
>                int height);
> 
> LIBYUV_API
>-void CopyPlane_16(const uint16* src_y,
>+void CopyPlane_16(const uint16_t* src_y,
>                   int src_stride_y,
>-                  uint16* dst_y,
>+                  uint16_t* dst_y,
>                   int dst_stride_y,
>                   int width,
>                   int height);
> 
>+LIBYUV_API
>+void Convert16To8Plane(const uint16_t* src_y,
>+                       int src_stride_y,
>+                       uint8_t* dst_y,
>+                       int dst_stride_y,
>+                       int scale,  // 16384 for 10 bits
>+                       int width,
>+                       int height);
>+
>+LIBYUV_API
>+void Convert8To16Plane(const uint8_t* src_y,
>+                       int src_stride_y,
>+                       uint16_t* dst_y,
>+                       int dst_stride_y,
>+                       int scale,  // 1024 for 10 bits
>+                       int width,
>+                       int height);
>+
> // Set a plane of data to a 32 bit value.
> LIBYUV_API
>-void SetPlane(uint8* dst_y,
>+void SetPlane(uint8_t* dst_y,
>               int dst_stride_y,
>               int width,
>               int height,
>-              uint32 value);
>+              uint32_t value);
> 
> // Split interleaved UV plane into separate U and V planes.
> LIBYUV_API
>-void SplitUVPlane(const uint8* src_uv,
>+void SplitUVPlane(const uint8_t* src_uv,
>                   int src_stride_uv,
>-                  uint8* dst_u,
>+                  uint8_t* dst_u,
>                   int dst_stride_u,
>-                  uint8* dst_v,
>+                  uint8_t* dst_v,
>                   int dst_stride_v,
>                   int width,
>                   int height);
> 
> // Merge separate U and V planes into one interleaved UV plane.
> LIBYUV_API
>-void MergeUVPlane(const uint8* src_u,
>+void MergeUVPlane(const uint8_t* src_u,
>                   int src_stride_u,
>-                  const uint8* src_v,
>+                  const uint8_t* src_v,
>                   int src_stride_v,
>-                  uint8* dst_uv,
>+                  uint8_t* dst_uv,
>                   int dst_stride_uv,
>                   int width,
>                   int height);
> 
> // Split interleaved RGB plane into separate R, G and B planes.
> LIBYUV_API
>-void SplitRGBPlane(const uint8* src_rgb,
>+void SplitRGBPlane(const uint8_t* src_rgb,
>                    int src_stride_rgb,
>-                   uint8* dst_r,
>+                   uint8_t* dst_r,
>                    int dst_stride_r,
>-                   uint8* dst_g,
>+                   uint8_t* dst_g,
>                    int dst_stride_g,
>-                   uint8* dst_b,
>+                   uint8_t* dst_b,
>                    int dst_stride_b,
>                    int width,
>                    int height);
> 
> // Merge separate R, G and B planes into one interleaved RGB plane.
> LIBYUV_API
>-void MergeRGBPlane(const uint8* src_r,
>+void MergeRGBPlane(const uint8_t* src_r,
>                    int src_stride_r,
>-                   const uint8* src_g,
>+                   const uint8_t* src_g,
>                    int src_stride_g,
>-                   const uint8* src_b,
>+                   const uint8_t* src_b,
>                    int src_stride_b,
>-                   uint8* dst_rgb,
>+                   uint8_t* dst_rgb,
>                    int dst_stride_rgb,
>                    int width,
>                    int height);
> 
> // Copy I400.  Supports inverting.
> LIBYUV_API
>-int I400ToI400(const uint8* src_y,
>+int I400ToI400(const uint8_t* src_y,
>                int src_stride_y,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>                int width,
>                int height);
>@@ -109,17 +145,17 @@ int I400ToI400(const uint8* src_y,
> // Copy I422 to I422.
> #define I422ToI422 I422Copy
> LIBYUV_API
>-int I422Copy(const uint8* src_y,
>+int I422Copy(const uint8_t* src_y,
>              int src_stride_y,
>-             const uint8* src_u,
>+             const uint8_t* src_u,
>              int src_stride_u,
>-             const uint8* src_v,
>+             const uint8_t* src_v,
>              int src_stride_v,
>-             uint8* dst_y,
>+             uint8_t* dst_y,
>              int dst_stride_y,
>-             uint8* dst_u,
>+             uint8_t* dst_u,
>              int dst_stride_u,
>-             uint8* dst_v,
>+             uint8_t* dst_v,
>              int dst_stride_v,
>              int width,
>              int height);
>@@ -127,84 +163,84 @@ int I422Copy(const uint8* src_y,
> // Copy I444 to I444.
> #define I444ToI444 I444Copy
> LIBYUV_API
>-int I444Copy(const uint8* src_y,
>+int I444Copy(const uint8_t* src_y,
>              int src_stride_y,
>-             const uint8* src_u,
>+             const uint8_t* src_u,
>              int src_stride_u,
>-             const uint8* src_v,
>+             const uint8_t* src_v,
>              int src_stride_v,
>-             uint8* dst_y,
>+             uint8_t* dst_y,
>              int dst_stride_y,
>-             uint8* dst_u,
>+             uint8_t* dst_u,
>              int dst_stride_u,
>-             uint8* dst_v,
>+             uint8_t* dst_v,
>              int dst_stride_v,
>              int width,
>              int height);
> 
> // Convert YUY2 to I422.
> LIBYUV_API
>-int YUY2ToI422(const uint8* src_yuy2,
>+int YUY2ToI422(const uint8_t* src_yuy2,
>                int src_stride_yuy2,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_u,
>+               uint8_t* dst_u,
>                int dst_stride_u,
>-               uint8* dst_v,
>+               uint8_t* dst_v,
>                int dst_stride_v,
>                int width,
>                int height);
> 
> // Convert UYVY to I422.
> LIBYUV_API
>-int UYVYToI422(const uint8* src_uyvy,
>+int UYVYToI422(const uint8_t* src_uyvy,
>                int src_stride_uyvy,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_u,
>+               uint8_t* dst_u,
>                int dst_stride_u,
>-               uint8* dst_v,
>+               uint8_t* dst_v,
>                int dst_stride_v,
>                int width,
>                int height);
> 
> LIBYUV_API
>-int YUY2ToNV12(const uint8* src_yuy2,
>+int YUY2ToNV12(const uint8_t* src_yuy2,
>                int src_stride_yuy2,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_uv,
>+               uint8_t* dst_uv,
>                int dst_stride_uv,
>                int width,
>                int height);
> 
> LIBYUV_API
>-int UYVYToNV12(const uint8* src_uyvy,
>+int UYVYToNV12(const uint8_t* src_uyvy,
>                int src_stride_uyvy,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_uv,
>+               uint8_t* dst_uv,
>                int dst_stride_uv,
>                int width,
>                int height);
> 
> LIBYUV_API
>-int YUY2ToY(const uint8* src_yuy2,
>+int YUY2ToY(const uint8_t* src_yuy2,
>             int src_stride_yuy2,
>-            uint8* dst_y,
>+            uint8_t* dst_y,
>             int dst_stride_y,
>             int width,
>             int height);
> 
> // Convert I420 to I400. (calls CopyPlane ignoring u/v).
> LIBYUV_API
>-int I420ToI400(const uint8* src_y,
>+int I420ToI400(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>                int width,
>                int height);
>@@ -215,17 +251,17 @@ int I420ToI400(const uint8* src_y,
> 
> // I420 mirror.
> LIBYUV_API
>-int I420Mirror(const uint8* src_y,
>+int I420Mirror(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_u,
>+               uint8_t* dst_u,
>                int dst_stride_u,
>-               uint8* dst_v,
>+               uint8_t* dst_v,
>                int dst_stride_v,
>                int width,
>                int height);
>@@ -236,9 +272,9 @@ int I420Mirror(const uint8* src_y,
> // I400 mirror.  A single plane is mirrored horizontally.
> // Pass negative height to achieve 180 degree rotation.
> LIBYUV_API
>-int I400Mirror(const uint8* src_y,
>+int I400Mirror(const uint8_t* src_y,
>                int src_stride_y,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>                int width,
>                int height);
>@@ -248,20 +284,20 @@ int I400Mirror(const uint8* src_y,
> 
> // ARGB mirror.
> LIBYUV_API
>-int ARGBMirror(const uint8* src_argb,
>+int ARGBMirror(const uint8_t* src_argb,
>                int src_stride_argb,
>-               uint8* dst_argb,
>+               uint8_t* dst_argb,
>                int dst_stride_argb,
>                int width,
>                int height);
> 
> // Convert NV12 to RGB565.
> LIBYUV_API
>-int NV12ToRGB565(const uint8* src_y,
>+int NV12ToRGB565(const uint8_t* src_y,
>                  int src_stride_y,
>-                 const uint8* src_uv,
>+                 const uint8_t* src_uv,
>                  int src_stride_uv,
>-                 uint8* dst_rgb565,
>+                 uint8_t* dst_rgb565,
>                  int dst_stride_rgb565,
>                  int width,
>                  int height);
>@@ -269,39 +305,39 @@ int NV12ToRGB565(const uint8* src_y,
> // I422ToARGB is in convert_argb.h
> // Convert I422 to BGRA.
> LIBYUV_API
>-int I422ToBGRA(const uint8* src_y,
>+int I422ToBGRA(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_bgra,
>+               uint8_t* dst_bgra,
>                int dst_stride_bgra,
>                int width,
>                int height);
> 
> // Convert I422 to ABGR.
> LIBYUV_API
>-int I422ToABGR(const uint8* src_y,
>+int I422ToABGR(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_abgr,
>+               uint8_t* dst_abgr,
>                int dst_stride_abgr,
>                int width,
>                int height);
> 
> // Convert I422 to RGBA.
> LIBYUV_API
>-int I422ToRGBA(const uint8* src_y,
>+int I422ToRGBA(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_rgba,
>+               uint8_t* dst_rgba,
>                int dst_stride_rgba,
>                int width,
>                int height);
>@@ -310,20 +346,20 @@ int I422ToRGBA(const uint8* src_y,
> #define RGB24ToRAW RAWToRGB24
> 
> LIBYUV_API
>-int RAWToRGB24(const uint8* src_raw,
>+int RAWToRGB24(const uint8_t* src_raw,
>                int src_stride_raw,
>-               uint8* dst_rgb24,
>+               uint8_t* dst_rgb24,
>                int dst_stride_rgb24,
>                int width,
>                int height);
> 
> // Draw a rectangle into I420.
> LIBYUV_API
>-int I420Rect(uint8* dst_y,
>+int I420Rect(uint8_t* dst_y,
>              int dst_stride_y,
>-             uint8* dst_u,
>+             uint8_t* dst_u,
>              int dst_stride_u,
>-             uint8* dst_v,
>+             uint8_t* dst_v,
>              int dst_stride_v,
>              int x,
>              int y,
>@@ -335,38 +371,38 @@ int I420Rect(uint8* dst_y,
> 
> // Draw a rectangle into ARGB.
> LIBYUV_API
>-int ARGBRect(uint8* dst_argb,
>+int ARGBRect(uint8_t* dst_argb,
>              int dst_stride_argb,
>-             int x,
>-             int y,
>+             int dst_x,
>+             int dst_y,
>              int width,
>              int height,
>-             uint32 value);
>+             uint32_t value);
> 
> // Convert ARGB to gray scale ARGB.
> LIBYUV_API
>-int ARGBGrayTo(const uint8* src_argb,
>+int ARGBGrayTo(const uint8_t* src_argb,
>                int src_stride_argb,
>-               uint8* dst_argb,
>+               uint8_t* dst_argb,
>                int dst_stride_argb,
>                int width,
>                int height);
> 
> // Make a rectangle of ARGB gray scale.
> LIBYUV_API
>-int ARGBGray(uint8* dst_argb,
>+int ARGBGray(uint8_t* dst_argb,
>              int dst_stride_argb,
>-             int x,
>-             int y,
>+             int dst_x,
>+             int dst_y,
>              int width,
>              int height);
> 
> // Make a rectangle of ARGB Sepia tone.
> LIBYUV_API
>-int ARGBSepia(uint8* dst_argb,
>+int ARGBSepia(uint8_t* dst_argb,
>               int dst_stride_argb,
>-              int x,
>-              int y,
>+              int dst_x,
>+              int dst_y,
>               int width,
>               int height);
> 
>@@ -377,11 +413,11 @@ int ARGBSepia(uint8* dst_argb,
> // The next 4 coefficients apply to B, G, R, A and produce R of the output.
> // The last 4 coefficients apply to B, G, R, A and produce A of the output.
> LIBYUV_API
>-int ARGBColorMatrix(const uint8* src_argb,
>+int ARGBColorMatrix(const uint8_t* src_argb,
>                     int src_stride_argb,
>-                    uint8* dst_argb,
>+                    uint8_t* dst_argb,
>                     int dst_stride_argb,
>-                    const int8* matrix_argb,
>+                    const int8_t* matrix_argb,
>                     int width,
>                     int height);
> 
>@@ -392,33 +428,33 @@ int ARGBColorMatrix(const uint8* src_argb,
> // The next 4 coefficients apply to B, G, R, A and produce G of the output.
> // The last 4 coefficients apply to B, G, R, A and produce R of the output.
> LIBYUV_API
>-int RGBColorMatrix(uint8* dst_argb,
>+int RGBColorMatrix(uint8_t* dst_argb,
>                    int dst_stride_argb,
>-                   const int8* matrix_rgb,
>-                   int x,
>-                   int y,
>+                   const int8_t* matrix_rgb,
>+                   int dst_x,
>+                   int dst_y,
>                    int width,
>                    int height);
> 
> // Apply a color table each ARGB pixel.
> // Table contains 256 ARGB values.
> LIBYUV_API
>-int ARGBColorTable(uint8* dst_argb,
>+int ARGBColorTable(uint8_t* dst_argb,
>                    int dst_stride_argb,
>-                   const uint8* table_argb,
>-                   int x,
>-                   int y,
>+                   const uint8_t* table_argb,
>+                   int dst_x,
>+                   int dst_y,
>                    int width,
>                    int height);
> 
> // Apply a color table each ARGB pixel but preserve destination alpha.
> // Table contains 256 ARGB values.
> LIBYUV_API
>-int RGBColorTable(uint8* dst_argb,
>+int RGBColorTable(uint8_t* dst_argb,
>                   int dst_stride_argb,
>-                  const uint8* table_argb,
>-                  int x,
>-                  int y,
>+                  const uint8_t* table_argb,
>+                  int dst_x,
>+                  int dst_y,
>                   int width,
>                   int height);
> 
>@@ -426,11 +462,11 @@ int RGBColorTable(uint8* dst_argb,
> // Table contains 32768 values indexed by [Y][C] where 7 it 7 bit luma from
> // RGB (YJ style) and C is an 8 bit color component (R, G or B).
> LIBYUV_API
>-int ARGBLumaColorTable(const uint8* src_argb,
>+int ARGBLumaColorTable(const uint8_t* src_argb,
>                        int src_stride_argb,
>-                       uint8* dst_argb,
>+                       uint8_t* dst_argb,
>                        int dst_stride_argb,
>-                       const uint8* luma_rgb_table,
>+                       const uint8_t* luma,
>                        int width,
>                        int height);
> 
>@@ -443,9 +479,9 @@ int ARGBLumaColorTable(const uint8* src_argb,
> // A polynomial approximation can be dirived using software such as 'R'.
> 
> LIBYUV_API
>-int ARGBPolynomial(const uint8* src_argb,
>+int ARGBPolynomial(const uint8_t* src_argb,
>                    int src_stride_argb,
>-                   uint8* dst_argb,
>+                   uint8_t* dst_argb,
>                    int dst_stride_argb,
>                    const float* poly,
>                    int width,
>@@ -454,68 +490,72 @@ int ARGBPolynomial(const uint8* src_argb,
> // Convert plane of 16 bit shorts to half floats.
> // Source values are multiplied by scale before storing as half float.
> LIBYUV_API
>-int HalfFloatPlane(const uint16* src_y,
>+int HalfFloatPlane(const uint16_t* src_y,
>                    int src_stride_y,
>-                   uint16* dst_y,
>+                   uint16_t* dst_y,
>                    int dst_stride_y,
>                    float scale,
>                    int width,
>                    int height);
> 
>+// Convert a buffer of bytes to floats, scale the values and store as floats.
>+LIBYUV_API
>+int ByteToFloat(const uint8_t* src_y, float* dst_y, float scale, int width);
>+
> // Quantize a rectangle of ARGB. Alpha unaffected.
> // scale is a 16 bit fractional fixed point scaler between 0 and 65535.
> // interval_size should be a value between 1 and 255.
> // interval_offset should be a value between 0 and 255.
> LIBYUV_API
>-int ARGBQuantize(uint8* dst_argb,
>+int ARGBQuantize(uint8_t* dst_argb,
>                  int dst_stride_argb,
>                  int scale,
>                  int interval_size,
>                  int interval_offset,
>-                 int x,
>-                 int y,
>+                 int dst_x,
>+                 int dst_y,
>                  int width,
>                  int height);
> 
> // Copy ARGB to ARGB.
> LIBYUV_API
>-int ARGBCopy(const uint8* src_argb,
>+int ARGBCopy(const uint8_t* src_argb,
>              int src_stride_argb,
>-             uint8* dst_argb,
>+             uint8_t* dst_argb,
>              int dst_stride_argb,
>              int width,
>              int height);
> 
> // Copy Alpha channel of ARGB to alpha of ARGB.
> LIBYUV_API
>-int ARGBCopyAlpha(const uint8* src_argb,
>+int ARGBCopyAlpha(const uint8_t* src_argb,
>                   int src_stride_argb,
>-                  uint8* dst_argb,
>+                  uint8_t* dst_argb,
>                   int dst_stride_argb,
>                   int width,
>                   int height);
> 
> // Extract the alpha channel from ARGB.
> LIBYUV_API
>-int ARGBExtractAlpha(const uint8* src_argb,
>+int ARGBExtractAlpha(const uint8_t* src_argb,
>                      int src_stride_argb,
>-                     uint8* dst_a,
>+                     uint8_t* dst_a,
>                      int dst_stride_a,
>                      int width,
>                      int height);
> 
> // Copy Y channel to Alpha of ARGB.
> LIBYUV_API
>-int ARGBCopyYToAlpha(const uint8* src_y,
>+int ARGBCopyYToAlpha(const uint8_t* src_y,
>                      int src_stride_y,
>-                     uint8* dst_argb,
>+                     uint8_t* dst_argb,
>                      int dst_stride_argb,
>                      int width,
>                      int height);
> 
>-typedef void (*ARGBBlendRow)(const uint8* src_argb0,
>-                             const uint8* src_argb1,
>-                             uint8* dst_argb,
>+typedef void (*ARGBBlendRow)(const uint8_t* src_argb0,
>+                             const uint8_t* src_argb1,
>+                             uint8_t* dst_argb,
>                              int width);
> 
> // Get function to Alpha Blend ARGB pixels and store to destination.
>@@ -526,11 +566,11 @@ ARGBBlendRow GetARGBBlend();
> // Source is pre-multiplied by alpha using ARGBAttenuate.
> // Alpha of destination is set to 255.
> LIBYUV_API
>-int ARGBBlend(const uint8* src_argb0,
>+int ARGBBlend(const uint8_t* src_argb0,
>               int src_stride_argb0,
>-              const uint8* src_argb1,
>+              const uint8_t* src_argb1,
>               int src_stride_argb1,
>-              uint8* dst_argb,
>+              uint8_t* dst_argb,
>               int dst_stride_argb,
>               int width,
>               int height);
>@@ -538,13 +578,13 @@ int ARGBBlend(const uint8* src_argb0,
> // Alpha Blend plane and store to destination.
> // Source is not pre-multiplied by alpha.
> LIBYUV_API
>-int BlendPlane(const uint8* src_y0,
>+int BlendPlane(const uint8_t* src_y0,
>                int src_stride_y0,
>-               const uint8* src_y1,
>+               const uint8_t* src_y1,
>                int src_stride_y1,
>-               const uint8* alpha,
>+               const uint8_t* alpha,
>                int alpha_stride,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>                int width,
>                int height);
>@@ -553,102 +593,102 @@ int BlendPlane(const uint8* src_y0,
> // Source is not pre-multiplied by alpha.
> // Alpha is full width x height and subsampled to half size to apply to UV.
> LIBYUV_API
>-int I420Blend(const uint8* src_y0,
>+int I420Blend(const uint8_t* src_y0,
>               int src_stride_y0,
>-              const uint8* src_u0,
>+              const uint8_t* src_u0,
>               int src_stride_u0,
>-              const uint8* src_v0,
>+              const uint8_t* src_v0,
>               int src_stride_v0,
>-              const uint8* src_y1,
>+              const uint8_t* src_y1,
>               int src_stride_y1,
>-              const uint8* src_u1,
>+              const uint8_t* src_u1,
>               int src_stride_u1,
>-              const uint8* src_v1,
>+              const uint8_t* src_v1,
>               int src_stride_v1,
>-              const uint8* alpha,
>+              const uint8_t* alpha,
>               int alpha_stride,
>-              uint8* dst_y,
>+              uint8_t* dst_y,
>               int dst_stride_y,
>-              uint8* dst_u,
>+              uint8_t* dst_u,
>               int dst_stride_u,
>-              uint8* dst_v,
>+              uint8_t* dst_v,
>               int dst_stride_v,
>               int width,
>               int height);
> 
> // Multiply ARGB image by ARGB image. Shifted down by 8. Saturates to 255.
> LIBYUV_API
>-int ARGBMultiply(const uint8* src_argb0,
>+int ARGBMultiply(const uint8_t* src_argb0,
>                  int src_stride_argb0,
>-                 const uint8* src_argb1,
>+                 const uint8_t* src_argb1,
>                  int src_stride_argb1,
>-                 uint8* dst_argb,
>+                 uint8_t* dst_argb,
>                  int dst_stride_argb,
>                  int width,
>                  int height);
> 
> // Add ARGB image with ARGB image. Saturates to 255.
> LIBYUV_API
>-int ARGBAdd(const uint8* src_argb0,
>+int ARGBAdd(const uint8_t* src_argb0,
>             int src_stride_argb0,
>-            const uint8* src_argb1,
>+            const uint8_t* src_argb1,
>             int src_stride_argb1,
>-            uint8* dst_argb,
>+            uint8_t* dst_argb,
>             int dst_stride_argb,
>             int width,
>             int height);
> 
> // Subtract ARGB image (argb1) from ARGB image (argb0). Saturates to 0.
> LIBYUV_API
>-int ARGBSubtract(const uint8* src_argb0,
>+int ARGBSubtract(const uint8_t* src_argb0,
>                  int src_stride_argb0,
>-                 const uint8* src_argb1,
>+                 const uint8_t* src_argb1,
>                  int src_stride_argb1,
>-                 uint8* dst_argb,
>+                 uint8_t* dst_argb,
>                  int dst_stride_argb,
>                  int width,
>                  int height);
> 
> // Convert I422 to YUY2.
> LIBYUV_API
>-int I422ToYUY2(const uint8* src_y,
>+int I422ToYUY2(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_frame,
>-               int dst_stride_frame,
>+               uint8_t* dst_yuy2,
>+               int dst_stride_yuy2,
>                int width,
>                int height);
> 
> // Convert I422 to UYVY.
> LIBYUV_API
>-int I422ToUYVY(const uint8* src_y,
>+int I422ToUYVY(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_frame,
>-               int dst_stride_frame,
>+               uint8_t* dst_uyvy,
>+               int dst_stride_uyvy,
>                int width,
>                int height);
> 
> // Convert unattentuated ARGB to preattenuated ARGB.
> LIBYUV_API
>-int ARGBAttenuate(const uint8* src_argb,
>+int ARGBAttenuate(const uint8_t* src_argb,
>                   int src_stride_argb,
>-                  uint8* dst_argb,
>+                  uint8_t* dst_argb,
>                   int dst_stride_argb,
>                   int width,
>                   int height);
> 
> // Convert preattentuated ARGB to unattenuated ARGB.
> LIBYUV_API
>-int ARGBUnattenuate(const uint8* src_argb,
>+int ARGBUnattenuate(const uint8_t* src_argb,
>                     int src_stride_argb,
>-                    uint8* dst_argb,
>+                    uint8_t* dst_argb,
>                     int dst_stride_argb,
>                     int width,
>                     int height);
>@@ -657,9 +697,9 @@ int ARGBUnattenuate(const uint8* src_argb,
> // Computes table of cumulative sum for image where the value is the sum
> // of all values above and to the left of the entry. Used by ARGBBlur.
> LIBYUV_API
>-int ARGBComputeCumulativeSum(const uint8* src_argb,
>+int ARGBComputeCumulativeSum(const uint8_t* src_argb,
>                              int src_stride_argb,
>-                             int32* dst_cumsum,
>+                             int32_t* dst_cumsum,
>                              int dst_stride32_cumsum,
>                              int width,
>                              int height);
>@@ -671,11 +711,11 @@ int ARGBComputeCumulativeSum(const uint8* src_argb,
> // radius is number of pixels around the center.  e.g. 1 = 3x3. 2=5x5.
> // Blur is optimized for radius of 5 (11x11) or less.
> LIBYUV_API
>-int ARGBBlur(const uint8* src_argb,
>+int ARGBBlur(const uint8_t* src_argb,
>              int src_stride_argb,
>-             uint8* dst_argb,
>+             uint8_t* dst_argb,
>              int dst_stride_argb,
>-             int32* dst_cumsum,
>+             int32_t* dst_cumsum,
>              int dst_stride32_cumsum,
>              int width,
>              int height,
>@@ -683,24 +723,24 @@ int ARGBBlur(const uint8* src_argb,
> 
> // Multiply ARGB image by ARGB value.
> LIBYUV_API
>-int ARGBShade(const uint8* src_argb,
>+int ARGBShade(const uint8_t* src_argb,
>               int src_stride_argb,
>-              uint8* dst_argb,
>+              uint8_t* dst_argb,
>               int dst_stride_argb,
>               int width,
>               int height,
>-              uint32 value);
>+              uint32_t value);
> 
> // Interpolate between two images using specified amount of interpolation
> // (0 to 255) and store to destination.
> // 'interpolation' is specified as 8 bit fraction where 0 means 100% src0
> // and 255 means 1% src0 and 99% src1.
> LIBYUV_API
>-int InterpolatePlane(const uint8* src0,
>+int InterpolatePlane(const uint8_t* src0,
>                      int src_stride0,
>-                     const uint8* src1,
>+                     const uint8_t* src1,
>                      int src_stride1,
>-                     uint8* dst,
>+                     uint8_t* dst,
>                      int dst_stride,
>                      int width,
>                      int height,
>@@ -709,11 +749,11 @@ int InterpolatePlane(const uint8* src0,
> // Interpolate between two ARGB images using specified amount of interpolation
> // Internally calls InterpolatePlane with width * 4 (bpp).
> LIBYUV_API
>-int ARGBInterpolate(const uint8* src_argb0,
>+int ARGBInterpolate(const uint8_t* src_argb0,
>                     int src_stride_argb0,
>-                    const uint8* src_argb1,
>+                    const uint8_t* src_argb1,
>                     int src_stride_argb1,
>-                    uint8* dst_argb,
>+                    uint8_t* dst_argb,
>                     int dst_stride_argb,
>                     int width,
>                     int height,
>@@ -723,93 +763,78 @@ int ARGBInterpolate(const uint8* src_argb0,
> // Internally calls InterpolatePlane on each plane where the U and V planes
> // are half width and half height.
> LIBYUV_API
>-int I420Interpolate(const uint8* src0_y,
>+int I420Interpolate(const uint8_t* src0_y,
>                     int src0_stride_y,
>-                    const uint8* src0_u,
>+                    const uint8_t* src0_u,
>                     int src0_stride_u,
>-                    const uint8* src0_v,
>+                    const uint8_t* src0_v,
>                     int src0_stride_v,
>-                    const uint8* src1_y,
>+                    const uint8_t* src1_y,
>                     int src1_stride_y,
>-                    const uint8* src1_u,
>+                    const uint8_t* src1_u,
>                     int src1_stride_u,
>-                    const uint8* src1_v,
>+                    const uint8_t* src1_v,
>                     int src1_stride_v,
>-                    uint8* dst_y,
>+                    uint8_t* dst_y,
>                     int dst_stride_y,
>-                    uint8* dst_u,
>+                    uint8_t* dst_u,
>                     int dst_stride_u,
>-                    uint8* dst_v,
>+                    uint8_t* dst_v,
>                     int dst_stride_v,
>                     int width,
>                     int height,
>                     int interpolation);
> 
>-#if defined(__pnacl__) || defined(__CLR_VER) || \
>-    (defined(__i386__) && !defined(__SSE__) && !defined(__clang__))
>-#define LIBYUV_DISABLE_X86
>-#endif
>-// MemorySanitizer does not support assembly code yet. http://crbug.com/344505
>-#if defined(__has_feature)
>-#if __has_feature(memory_sanitizer)
>-#define LIBYUV_DISABLE_X86
>-#endif
>-#endif
>-// The following are available on all x86 platforms:
>-#if !defined(LIBYUV_DISABLE_X86) && \
>-    (defined(_M_IX86) || defined(__x86_64__) || defined(__i386__))
>-#define HAS_ARGBAFFINEROW_SSE2
>-#endif
>-
> // Row function for copying pixels from a source with a slope to a row
> // of destination. Useful for scaling, rotation, mirror, texture mapping.
> LIBYUV_API
>-void ARGBAffineRow_C(const uint8* src_argb,
>+void ARGBAffineRow_C(const uint8_t* src_argb,
>                      int src_argb_stride,
>-                     uint8* dst_argb,
>+                     uint8_t* dst_argb,
>                      const float* uv_dudv,
>                      int width);
>+// TODO(fbarchard): Move ARGBAffineRow_SSE2 to row.h
> LIBYUV_API
>-void ARGBAffineRow_SSE2(const uint8* src_argb,
>+void ARGBAffineRow_SSE2(const uint8_t* src_argb,
>                         int src_argb_stride,
>-                        uint8* dst_argb,
>+                        uint8_t* dst_argb,
>                         const float* uv_dudv,
>                         int width);
> 
> // Shuffle ARGB channel order.  e.g. BGRA to ARGB.
> // shuffler is 16 bytes and must be aligned.
> LIBYUV_API
>-int ARGBShuffle(const uint8* src_bgra,
>+int ARGBShuffle(const uint8_t* src_bgra,
>                 int src_stride_bgra,
>-                uint8* dst_argb,
>+                uint8_t* dst_argb,
>                 int dst_stride_argb,
>-                const uint8* shuffler,
>+                const uint8_t* shuffler,
>                 int width,
>                 int height);
> 
> // Sobel ARGB effect with planar output.
> LIBYUV_API
>-int ARGBSobelToPlane(const uint8* src_argb,
>+int ARGBSobelToPlane(const uint8_t* src_argb,
>                      int src_stride_argb,
>-                     uint8* dst_y,
>+                     uint8_t* dst_y,
>                      int dst_stride_y,
>                      int width,
>                      int height);
> 
> // Sobel ARGB effect.
> LIBYUV_API
>-int ARGBSobel(const uint8* src_argb,
>+int ARGBSobel(const uint8_t* src_argb,
>               int src_stride_argb,
>-              uint8* dst_argb,
>+              uint8_t* dst_argb,
>               int dst_stride_argb,
>               int width,
>               int height);
> 
> // Sobel ARGB effect w/ Sobel X, Sobel, Sobel Y in ARGB.
> LIBYUV_API
>-int ARGBSobelXY(const uint8* src_argb,
>+int ARGBSobelXY(const uint8_t* src_argb,
>                 int src_stride_argb,
>-                uint8* dst_argb,
>+                uint8_t* dst_argb,
>                 int dst_stride_argb,
>                 int width,
>                 int height);
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/rotate.h b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/rotate.h
>index b9f7154a51d..76b692be8b0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/rotate.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/rotate.h
>@@ -33,79 +33,79 @@ typedef enum RotationMode {
> 
> // Rotate I420 frame.
> LIBYUV_API
>-int I420Rotate(const uint8* src_y,
>+int I420Rotate(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_u,
>+               uint8_t* dst_u,
>                int dst_stride_u,
>-               uint8* dst_v,
>+               uint8_t* dst_v,
>                int dst_stride_v,
>-               int src_width,
>-               int src_height,
>+               int width,
>+               int height,
>                enum RotationMode mode);
> 
> // Rotate NV12 input and store in I420.
> LIBYUV_API
>-int NV12ToI420Rotate(const uint8* src_y,
>+int NV12ToI420Rotate(const uint8_t* src_y,
>                      int src_stride_y,
>-                     const uint8* src_uv,
>+                     const uint8_t* src_uv,
>                      int src_stride_uv,
>-                     uint8* dst_y,
>+                     uint8_t* dst_y,
>                      int dst_stride_y,
>-                     uint8* dst_u,
>+                     uint8_t* dst_u,
>                      int dst_stride_u,
>-                     uint8* dst_v,
>+                     uint8_t* dst_v,
>                      int dst_stride_v,
>-                     int src_width,
>-                     int src_height,
>+                     int width,
>+                     int height,
>                      enum RotationMode mode);
> 
> // Rotate a plane by 0, 90, 180, or 270.
> LIBYUV_API
>-int RotatePlane(const uint8* src,
>+int RotatePlane(const uint8_t* src,
>                 int src_stride,
>-                uint8* dst,
>+                uint8_t* dst,
>                 int dst_stride,
>-                int src_width,
>-                int src_height,
>+                int width,
>+                int height,
>                 enum RotationMode mode);
> 
> // Rotate planes by 90, 180, 270. Deprecated.
> LIBYUV_API
>-void RotatePlane90(const uint8* src,
>+void RotatePlane90(const uint8_t* src,
>                    int src_stride,
>-                   uint8* dst,
>+                   uint8_t* dst,
>                    int dst_stride,
>                    int width,
>                    int height);
> 
> LIBYUV_API
>-void RotatePlane180(const uint8* src,
>+void RotatePlane180(const uint8_t* src,
>                     int src_stride,
>-                    uint8* dst,
>+                    uint8_t* dst,
>                     int dst_stride,
>                     int width,
>                     int height);
> 
> LIBYUV_API
>-void RotatePlane270(const uint8* src,
>+void RotatePlane270(const uint8_t* src,
>                     int src_stride,
>-                    uint8* dst,
>+                    uint8_t* dst,
>                     int dst_stride,
>                     int width,
>                     int height);
> 
> LIBYUV_API
>-void RotateUV90(const uint8* src,
>+void RotateUV90(const uint8_t* src,
>                 int src_stride,
>-                uint8* dst_a,
>+                uint8_t* dst_a,
>                 int dst_stride_a,
>-                uint8* dst_b,
>+                uint8_t* dst_b,
>                 int dst_stride_b,
>                 int width,
>                 int height);
>@@ -115,21 +115,21 @@ void RotateUV90(const uint8* src,
> // split the data into two buffers while
> // rotating them. Deprecated.
> LIBYUV_API
>-void RotateUV180(const uint8* src,
>+void RotateUV180(const uint8_t* src,
>                  int src_stride,
>-                 uint8* dst_a,
>+                 uint8_t* dst_a,
>                  int dst_stride_a,
>-                 uint8* dst_b,
>+                 uint8_t* dst_b,
>                  int dst_stride_b,
>                  int width,
>                  int height);
> 
> LIBYUV_API
>-void RotateUV270(const uint8* src,
>+void RotateUV270(const uint8_t* src,
>                  int src_stride,
>-                 uint8* dst_a,
>+                 uint8_t* dst_a,
>                  int dst_stride_a,
>-                 uint8* dst_b,
>+                 uint8_t* dst_b,
>                  int dst_stride_b,
>                  int width,
>                  int height);
>@@ -139,19 +139,19 @@ void RotateUV270(const uint8* src,
> // order will result in a rotation by +- 90 degrees.
> // Deprecated.
> LIBYUV_API
>-void TransposePlane(const uint8* src,
>+void TransposePlane(const uint8_t* src,
>                     int src_stride,
>-                    uint8* dst,
>+                    uint8_t* dst,
>                     int dst_stride,
>                     int width,
>                     int height);
> 
> LIBYUV_API
>-void TransposeUV(const uint8* src,
>+void TransposeUV(const uint8_t* src,
>                  int src_stride,
>-                 uint8* dst_a,
>+                 uint8_t* dst_a,
>                  int dst_stride_a,
>-                 uint8* dst_b,
>+                 uint8_t* dst_b,
>                  int dst_stride_b,
>                  int width,
>                  int height);
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/rotate_argb.h b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/rotate_argb.h
>index be0190c1787..20432949ab4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/rotate_argb.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/rotate_argb.h
>@@ -21,9 +21,9 @@ extern "C" {
> 
> // Rotate ARGB frame
> LIBYUV_API
>-int ARGBRotate(const uint8* src_argb,
>+int ARGBRotate(const uint8_t* src_argb,
>                int src_stride_argb,
>-               uint8* dst_argb,
>+               uint8_t* dst_argb,
>                int dst_stride_argb,
>                int src_width,
>                int src_height,
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/rotate_row.h b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/rotate_row.h
>index 973fc15284f..5edc0fcf13a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/rotate_row.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/rotate_row.h
>@@ -18,10 +18,14 @@ namespace libyuv {
> extern "C" {
> #endif
> 
>-#if defined(__pnacl__) || defined(__CLR_VER) || \
>+#if defined(__pnacl__) || defined(__CLR_VER) ||            \
>+    (defined(__native_client__) && defined(__x86_64__)) || \
>     (defined(__i386__) && !defined(__SSE__) && !defined(__clang__))
> #define LIBYUV_DISABLE_X86
> #endif
>+#if defined(__native_client__)
>+#define LIBYUV_DISABLE_NEON
>+#endif
> // MemorySanitizer does not support assembly code yet. http://crbug.com/344505
> #if defined(__has_feature)
> #if __has_feature(memory_sanitizer)
>@@ -34,189 +38,151 @@ extern "C" {
> #define HAS_TRANSPOSEUVWX8_SSE2
> #endif
> 
>-// The following are available for GCC 32 or 64 bit but not NaCL for 64 bit:
>-#if !defined(LIBYUV_DISABLE_X86) && \
>-    (defined(__i386__) ||           \
>-     (defined(__x86_64__) && !defined(__native_client__)))
>+// The following are available for GCC 32 or 64 bit:
>+#if !defined(LIBYUV_DISABLE_X86) && (defined(__i386__) || defined(__x86_64__))
> #define HAS_TRANSPOSEWX8_SSSE3
> #endif
> 
>-// The following are available for 64 bit GCC but not NaCL:
>-#if !defined(LIBYUV_DISABLE_X86) && !defined(__native_client__) && \
>-    defined(__x86_64__)
>+// The following are available for 64 bit GCC:
>+#if !defined(LIBYUV_DISABLE_X86) && defined(__x86_64__)
> #define HAS_TRANSPOSEWX8_FAST_SSSE3
> #define HAS_TRANSPOSEUVWX8_SSE2
> #endif
> 
>-#if !defined(LIBYUV_DISABLE_NEON) && !defined(__native_client__) && \
>+#if !defined(LIBYUV_DISABLE_NEON) && \
>     (defined(__ARM_NEON__) || defined(LIBYUV_NEON) || defined(__aarch64__))
> #define HAS_TRANSPOSEWX8_NEON
> #define HAS_TRANSPOSEUVWX8_NEON
> #endif
> 
>-#if !defined(LIBYUV_DISABLE_DSPR2) && !defined(__native_client__) && \
>-    defined(__mips__) && defined(__mips_dsp) && (__mips_dsp_rev >= 2)
>-#define HAS_TRANSPOSEWX8_DSPR2
>-#define HAS_TRANSPOSEUVWX8_DSPR2
>-#endif  // defined(__mips__)
>-
> #if !defined(LIBYUV_DISABLE_MSA) && defined(__mips_msa)
> #define HAS_TRANSPOSEWX16_MSA
> #define HAS_TRANSPOSEUVWX16_MSA
> #endif
> 
>-void TransposeWxH_C(const uint8* src,
>+void TransposeWxH_C(const uint8_t* src,
>                     int src_stride,
>-                    uint8* dst,
>+                    uint8_t* dst,
>                     int dst_stride,
>                     int width,
>                     int height);
> 
>-void TransposeWx8_C(const uint8* src,
>+void TransposeWx8_C(const uint8_t* src,
>                     int src_stride,
>-                    uint8* dst,
>+                    uint8_t* dst,
>                     int dst_stride,
>                     int width);
>-void TransposeWx16_C(const uint8* src,
>+void TransposeWx16_C(const uint8_t* src,
>                      int src_stride,
>-                     uint8* dst,
>+                     uint8_t* dst,
>                      int dst_stride,
>                      int width);
>-void TransposeWx8_NEON(const uint8* src,
>+void TransposeWx8_NEON(const uint8_t* src,
>                        int src_stride,
>-                       uint8* dst,
>+                       uint8_t* dst,
>                        int dst_stride,
>                        int width);
>-void TransposeWx8_SSSE3(const uint8* src,
>+void TransposeWx8_SSSE3(const uint8_t* src,
>                         int src_stride,
>-                        uint8* dst,
>+                        uint8_t* dst,
>                         int dst_stride,
>                         int width);
>-void TransposeWx8_Fast_SSSE3(const uint8* src,
>+void TransposeWx8_Fast_SSSE3(const uint8_t* src,
>                              int src_stride,
>-                             uint8* dst,
>+                             uint8_t* dst,
>                              int dst_stride,
>                              int width);
>-void TransposeWx8_DSPR2(const uint8* src,
>-                        int src_stride,
>-                        uint8* dst,
>-                        int dst_stride,
>-                        int width);
>-void TransposeWx8_Fast_DSPR2(const uint8* src,
>-                             int src_stride,
>-                             uint8* dst,
>-                             int dst_stride,
>-                             int width);
>-void TransposeWx16_MSA(const uint8* src,
>+void TransposeWx16_MSA(const uint8_t* src,
>                        int src_stride,
>-                       uint8* dst,
>+                       uint8_t* dst,
>                        int dst_stride,
>                        int width);
> 
>-void TransposeWx8_Any_NEON(const uint8* src,
>+void TransposeWx8_Any_NEON(const uint8_t* src,
>                            int src_stride,
>-                           uint8* dst,
>+                           uint8_t* dst,
>                            int dst_stride,
>                            int width);
>-void TransposeWx8_Any_SSSE3(const uint8* src,
>+void TransposeWx8_Any_SSSE3(const uint8_t* src,
>                             int src_stride,
>-                            uint8* dst,
>+                            uint8_t* dst,
>                             int dst_stride,
>                             int width);
>-void TransposeWx8_Fast_Any_SSSE3(const uint8* src,
>+void TransposeWx8_Fast_Any_SSSE3(const uint8_t* src,
>                                  int src_stride,
>-                                 uint8* dst,
>+                                 uint8_t* dst,
>                                  int dst_stride,
>                                  int width);
>-void TransposeWx8_Any_DSPR2(const uint8* src,
>-                            int src_stride,
>-                            uint8* dst,
>-                            int dst_stride,
>-                            int width);
>-void TransposeWx16_Any_MSA(const uint8* src,
>+void TransposeWx16_Any_MSA(const uint8_t* src,
>                            int src_stride,
>-                           uint8* dst,
>+                           uint8_t* dst,
>                            int dst_stride,
>                            int width);
> 
>-void TransposeUVWxH_C(const uint8* src,
>+void TransposeUVWxH_C(const uint8_t* src,
>                       int src_stride,
>-                      uint8* dst_a,
>+                      uint8_t* dst_a,
>                       int dst_stride_a,
>-                      uint8* dst_b,
>+                      uint8_t* dst_b,
>                       int dst_stride_b,
>                       int width,
>                       int height);
> 
>-void TransposeUVWx8_C(const uint8* src,
>+void TransposeUVWx8_C(const uint8_t* src,
>                       int src_stride,
>-                      uint8* dst_a,
>+                      uint8_t* dst_a,
>                       int dst_stride_a,
>-                      uint8* dst_b,
>+                      uint8_t* dst_b,
>                       int dst_stride_b,
>                       int width);
>-void TransposeUVWx16_C(const uint8* src,
>+void TransposeUVWx16_C(const uint8_t* src,
>                        int src_stride,
>-                       uint8* dst_a,
>+                       uint8_t* dst_a,
>                        int dst_stride_a,
>-                       uint8* dst_b,
>+                       uint8_t* dst_b,
>                        int dst_stride_b,
>                        int width);
>-void TransposeUVWx8_SSE2(const uint8* src,
>+void TransposeUVWx8_SSE2(const uint8_t* src,
>                          int src_stride,
>-                         uint8* dst_a,
>+                         uint8_t* dst_a,
>                          int dst_stride_a,
>-                         uint8* dst_b,
>+                         uint8_t* dst_b,
>                          int dst_stride_b,
>                          int width);
>-void TransposeUVWx8_NEON(const uint8* src,
>+void TransposeUVWx8_NEON(const uint8_t* src,
>                          int src_stride,
>-                         uint8* dst_a,
>+                         uint8_t* dst_a,
>                          int dst_stride_a,
>-                         uint8* dst_b,
>+                         uint8_t* dst_b,
>                          int dst_stride_b,
>                          int width);
>-void TransposeUVWx8_DSPR2(const uint8* src,
>-                          int src_stride,
>-                          uint8* dst_a,
>-                          int dst_stride_a,
>-                          uint8* dst_b,
>-                          int dst_stride_b,
>-                          int width);
>-void TransposeUVWx16_MSA(const uint8* src,
>+void TransposeUVWx16_MSA(const uint8_t* src,
>                          int src_stride,
>-                         uint8* dst_a,
>+                         uint8_t* dst_a,
>                          int dst_stride_a,
>-                         uint8* dst_b,
>+                         uint8_t* dst_b,
>                          int dst_stride_b,
>                          int width);
> 
>-void TransposeUVWx8_Any_SSE2(const uint8* src,
>+void TransposeUVWx8_Any_SSE2(const uint8_t* src,
>                              int src_stride,
>-                             uint8* dst_a,
>+                             uint8_t* dst_a,
>                              int dst_stride_a,
>-                             uint8* dst_b,
>+                             uint8_t* dst_b,
>                              int dst_stride_b,
>                              int width);
>-void TransposeUVWx8_Any_NEON(const uint8* src,
>+void TransposeUVWx8_Any_NEON(const uint8_t* src,
>                              int src_stride,
>-                             uint8* dst_a,
>+                             uint8_t* dst_a,
>                              int dst_stride_a,
>-                             uint8* dst_b,
>+                             uint8_t* dst_b,
>                              int dst_stride_b,
>                              int width);
>-void TransposeUVWx8_Any_DSPR2(const uint8* src,
>-                              int src_stride,
>-                              uint8* dst_a,
>-                              int dst_stride_a,
>-                              uint8* dst_b,
>-                              int dst_stride_b,
>-                              int width);
>-void TransposeUVWx16_Any_MSA(const uint8* src,
>+void TransposeUVWx16_Any_MSA(const uint8_t* src,
>                              int src_stride,
>-                             uint8* dst_a,
>+                             uint8_t* dst_a,
>                              int dst_stride_a,
>-                             uint8* dst_b,
>+                             uint8_t* dst_b,
>                              int dst_stride_b,
>                              int width);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/row.h b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/row.h
>index 34d727641a8..b5a42d0e62c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/row.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/row.h
>@@ -20,34 +20,20 @@ namespace libyuv {
> extern "C" {
> #endif
> 
>-#define IS_ALIGNED(p, a) (!((uintptr_t)(p) & ((a)-1)))
>-
>-#define align_buffer_64(var, size)                                       \
>-  uint8* var##_mem = (uint8*)(malloc((size) + 63));         /* NOLINT */ \
>-  uint8* var = (uint8*)(((intptr_t)(var##_mem) + 63) & ~63) /* NOLINT */
>-
>-#define free_aligned_buffer_64(var) \
>-  free(var##_mem);                  \
>-  var = 0
>-
>-#if defined(__pnacl__) || defined(__CLR_VER) || \
>+#if defined(__pnacl__) || defined(__CLR_VER) ||            \
>+    (defined(__native_client__) && defined(__x86_64__)) || \
>     (defined(__i386__) && !defined(__SSE__) && !defined(__clang__))
> #define LIBYUV_DISABLE_X86
> #endif
>+#if defined(__native_client__)
>+#define LIBYUV_DISABLE_NEON
>+#endif
> // MemorySanitizer does not support assembly code yet. http://crbug.com/344505
> #if defined(__has_feature)
> #if __has_feature(memory_sanitizer)
> #define LIBYUV_DISABLE_X86
> #endif
> #endif
>-// True if compiling for SSSE3 as a requirement.
>-#if defined(__SSSE3__) || (defined(_M_IX86_FP) && (_M_IX86_FP >= 3))
>-#define LIBYUV_SSSE3_ONLY
>-#endif
>-
>-#if defined(__native_client__)
>-#define LIBYUV_DISABLE_NEON
>-#endif
> // clang >= 3.5.0 required for Arm64.
> #if defined(__clang__) && defined(__aarch64__) && !defined(LIBYUV_DISABLE_NEON)
> #if (__clang_major__ < 3) || (__clang_major__ == 3 && (__clang_minor__ < 5))
>@@ -85,7 +71,6 @@ extern "C" {
> #define HAS_ARGB4444TOARGBROW_SSE2
> #define HAS_ARGBEXTRACTALPHAROW_SSE2
> #define HAS_ARGBSETROW_X86
>-#define HAS_ARGBSHUFFLEROW_SSE2
> #define HAS_ARGBSHUFFLEROW_SSSE3
> #define HAS_ARGBTOARGB1555ROW_SSE2
> #define HAS_ARGBTOARGB4444ROW_SSE2
>@@ -184,7 +169,6 @@ extern "C" {
> 
> // The following are available on all x86 platforms, but
> // require VS2012, clang 3.4 or gcc 4.7.
>-// The code supports NaCL but requires a new compiler and validator.
> #if !defined(LIBYUV_DISABLE_X86) &&                          \
>     (defined(VISUALC_HAS_AVX2) || defined(CLANG_HAS_AVX2) || \
>      defined(GCC_HAS_AVX2))
>@@ -268,6 +252,14 @@ extern "C" {
> // TODO(fbarchard): Port to Visual C
> #if !defined(LIBYUV_DISABLE_X86) && \
>     (defined(__x86_64__) || (defined(__i386__) && !defined(_MSC_VER)))
>+#define HAS_ABGRTOAR30ROW_SSSE3
>+#define HAS_ARGBTOAR30ROW_SSSE3
>+#define HAS_CONVERT16TO8ROW_SSSE3
>+#define HAS_CONVERT8TO16ROW_SSE2
>+// I210 is for H010.  2 = 422.  I for 601 vs H for 709.
>+#define HAS_I210TOAR30ROW_SSSE3
>+#define HAS_I210TOARGBROW_SSSE3
>+#define HAS_I422TOAR30ROW_SSSE3
> #define HAS_MERGERGBROW_SSSE3
> #define HAS_SPLITRGBROW_SSSE3
> #endif
>@@ -277,6 +269,15 @@ extern "C" {
> #if !defined(LIBYUV_DISABLE_X86) &&                                       \
>     (defined(__x86_64__) || (defined(__i386__) && !defined(_MSC_VER))) && \
>     (defined(CLANG_HAS_AVX2) || defined(GCC_HAS_AVX2))
>+#define HAS_ABGRTOAR30ROW_AVX2
>+#define HAS_ARGBTOAR30ROW_AVX2
>+#define HAS_CONVERT16TO8ROW_AVX2
>+#define HAS_CONVERT8TO16ROW_AVX2
>+#define HAS_I210TOARGBROW_AVX2
>+#define HAS_I210TOAR30ROW_AVX2
>+#define HAS_I422TOAR30ROW_AVX2
>+#define HAS_I422TOUYVYROW_AVX2
>+#define HAS_I422TOYUY2ROW_AVX2
> #define HAS_MERGEUVROW_16_AVX2
> #define HAS_MULTIPLYROW_16_AVX2
> #endif
>@@ -307,6 +308,7 @@ extern "C" {
> #define HAS_ARGBTOYROW_NEON
> #define HAS_BGRATOUVROW_NEON
> #define HAS_BGRATOYROW_NEON
>+#define HAS_BYTETOFLOATROW_NEON
> #define HAS_COPYROW_NEON
> #define HAS_HALFFLOATROW_NEON
> #define HAS_I400TOARGBROW_NEON
>@@ -325,8 +327,10 @@ extern "C" {
> #define HAS_MIRRORROW_NEON
> #define HAS_MIRRORUVROW_NEON
> #define HAS_NV12TOARGBROW_NEON
>+#define HAS_NV12TORGB24ROW_NEON
> #define HAS_NV12TORGB565ROW_NEON
> #define HAS_NV21TOARGBROW_NEON
>+#define HAS_NV21TORGB24ROW_NEON
> #define HAS_RAWTOARGBROW_NEON
> #define HAS_RAWTORGB24ROW_NEON
> #define HAS_RAWTOUVROW_NEON
>@@ -376,37 +380,6 @@ extern "C" {
> #if !defined(LIBYUV_DISABLE_NEON) && defined(__aarch64__)
> #define HAS_SCALESUMSAMPLES_NEON
> #endif
>-
>-// The following are available on Mips platforms:
>-#if !defined(LIBYUV_DISABLE_DSPR2) && defined(__mips__) && \
>-    (_MIPS_SIM == _MIPS_SIM_ABI32) && (__mips_isa_rev < 6)
>-#define HAS_COPYROW_MIPS
>-#if defined(__mips_dsp) && (__mips_dsp_rev >= 2)
>-#define HAS_I422TOARGBROW_DSPR2
>-#define HAS_INTERPOLATEROW_DSPR2
>-#define HAS_MIRRORROW_DSPR2
>-#define HAS_MIRRORUVROW_DSPR2
>-#define HAS_SPLITUVROW_DSPR2
>-#define HAS_RGB24TOARGBROW_DSPR2
>-#define HAS_RAWTOARGBROW_DSPR2
>-#define HAS_RGB565TOARGBROW_DSPR2
>-#define HAS_ARGB1555TOARGBROW_DSPR2
>-#define HAS_ARGB4444TOARGBROW_DSPR2
>-#define HAS_I444TOARGBROW_DSPR2
>-#define HAS_I422TOARGB4444ROW_DSPR2
>-#define HAS_I422TOARGB1555ROW_DSPR2
>-#define HAS_NV12TOARGBROW_DSPR2
>-#define HAS_BGRATOUVROW_DSPR2
>-#define HAS_BGRATOYROW_DSPR2
>-#define HAS_ABGRTOUVROW_DSPR2
>-#define HAS_ARGBTOYROW_DSPR2
>-#define HAS_ABGRTOYROW_DSPR2
>-#define HAS_RGBATOUVROW_DSPR2
>-#define HAS_RGBATOYROW_DSPR2
>-#define HAS_ARGBTOUVROW_DSPR2
>-#endif
>-#endif
>-
> #if !defined(LIBYUV_DISABLE_MSA) && defined(__mips_msa)
> #define HAS_ABGRTOUVROW_MSA
> #define HAS_ABGRTOYROW_MSA
>@@ -492,18 +465,18 @@ extern "C" {
> #else
> #define SIMD_ALIGNED(var) __declspec(align(16)) var
> #endif
>-typedef __declspec(align(16)) int16 vec16[8];
>-typedef __declspec(align(16)) int32 vec32[4];
>-typedef __declspec(align(16)) int8 vec8[16];
>-typedef __declspec(align(16)) uint16 uvec16[8];
>-typedef __declspec(align(16)) uint32 uvec32[4];
>-typedef __declspec(align(16)) uint8 uvec8[16];
>-typedef __declspec(align(32)) int16 lvec16[16];
>-typedef __declspec(align(32)) int32 lvec32[8];
>-typedef __declspec(align(32)) int8 lvec8[32];
>-typedef __declspec(align(32)) uint16 ulvec16[16];
>-typedef __declspec(align(32)) uint32 ulvec32[8];
>-typedef __declspec(align(32)) uint8 ulvec8[32];
>+typedef __declspec(align(16)) int16_t vec16[8];
>+typedef __declspec(align(16)) int32_t vec32[4];
>+typedef __declspec(align(16)) int8_t vec8[16];
>+typedef __declspec(align(16)) uint16_t uvec16[8];
>+typedef __declspec(align(16)) uint32_t uvec32[4];
>+typedef __declspec(align(16)) uint8_t uvec8[16];
>+typedef __declspec(align(32)) int16_t lvec16[16];
>+typedef __declspec(align(32)) int32_t lvec32[8];
>+typedef __declspec(align(32)) int8_t lvec8[32];
>+typedef __declspec(align(32)) uint16_t ulvec16[16];
>+typedef __declspec(align(32)) uint32_t ulvec32[8];
>+typedef __declspec(align(32)) uint8_t ulvec8[32];
> #elif !defined(__pnacl__) && (defined(__GNUC__) || defined(__clang__))
> // Caveat GCC 4.2 to 4.7 have a known issue using vectors with const.
> #if defined(CLANG_HAS_AVX2) || defined(GCC_HAS_AVX2)
>@@ -511,32 +484,32 @@ typedef __declspec(align(32)) uint8 ulvec8[32];
> #else
> #define SIMD_ALIGNED(var) var __attribute__((aligned(16)))
> #endif
>-typedef int16 __attribute__((vector_size(16))) vec16;
>-typedef int32 __attribute__((vector_size(16))) vec32;
>-typedef int8 __attribute__((vector_size(16))) vec8;
>-typedef uint16 __attribute__((vector_size(16))) uvec16;
>-typedef uint32 __attribute__((vector_size(16))) uvec32;
>-typedef uint8 __attribute__((vector_size(16))) uvec8;
>-typedef int16 __attribute__((vector_size(32))) lvec16;
>-typedef int32 __attribute__((vector_size(32))) lvec32;
>-typedef int8 __attribute__((vector_size(32))) lvec8;
>-typedef uint16 __attribute__((vector_size(32))) ulvec16;
>-typedef uint32 __attribute__((vector_size(32))) ulvec32;
>-typedef uint8 __attribute__((vector_size(32))) ulvec8;
>+typedef int16_t __attribute__((vector_size(16))) vec16;
>+typedef int32_t __attribute__((vector_size(16))) vec32;
>+typedef int8_t __attribute__((vector_size(16))) vec8;
>+typedef uint16_t __attribute__((vector_size(16))) uvec16;
>+typedef uint32_t __attribute__((vector_size(16))) uvec32;
>+typedef uint8_t __attribute__((vector_size(16))) uvec8;
>+typedef int16_t __attribute__((vector_size(32))) lvec16;
>+typedef int32_t __attribute__((vector_size(32))) lvec32;
>+typedef int8_t __attribute__((vector_size(32))) lvec8;
>+typedef uint16_t __attribute__((vector_size(32))) ulvec16;
>+typedef uint32_t __attribute__((vector_size(32))) ulvec32;
>+typedef uint8_t __attribute__((vector_size(32))) ulvec8;
> #else
> #define SIMD_ALIGNED(var) var
>-typedef int16 vec16[8];
>-typedef int32 vec32[4];
>-typedef int8 vec8[16];
>-typedef uint16 uvec16[8];
>-typedef uint32 uvec32[4];
>-typedef uint8 uvec8[16];
>-typedef int16 lvec16[16];
>-typedef int32 lvec32[8];
>-typedef int8 lvec8[32];
>-typedef uint16 ulvec16[16];
>-typedef uint32 ulvec32[8];
>-typedef uint8 ulvec8[32];
>+typedef int16_t vec16[8];
>+typedef int32_t vec32[4];
>+typedef int8_t vec8[16];
>+typedef uint16_t uvec16[8];
>+typedef uint32_t uvec32[4];
>+typedef uint8_t uvec8[16];
>+typedef int16_t lvec16[16];
>+typedef int32_t lvec32[8];
>+typedef int8_t lvec8[32];
>+typedef uint16_t ulvec16[16];
>+typedef uint32_t ulvec32[8];
>+typedef uint8_t ulvec8[32];
> #endif
> 
> #if defined(__aarch64__)
>@@ -560,13 +533,13 @@ struct YuvConstants {
> #else
> // This struct is for Intel color conversion.
> struct YuvConstants {
>-  int8 kUVToB[32];
>-  int8 kUVToG[32];
>-  int8 kUVToR[32];
>-  int16 kUVBiasB[16];
>-  int16 kUVBiasG[16];
>-  int16 kUVBiasR[16];
>-  int16 kYToRgb[16];
>+  int8_t kUVToB[32];
>+  int8_t kUVToG[32];
>+  int8_t kUVToR[32];
>+  int16_t kUVBiasB[16];
>+  int16_t kUVBiasG[16];
>+  int16_t kUVBiasR[16];
>+  int16_t kYToRgb[16];
> };
> 
> // Offsets into YuvConstants structure
>@@ -589,6 +562,16 @@ extern const struct YuvConstants SIMD_ALIGNED(kYvuI601Constants);  // BT.601
> extern const struct YuvConstants SIMD_ALIGNED(kYvuJPEGConstants);  // JPeg
> extern const struct YuvConstants SIMD_ALIGNED(kYvuH709Constants);  // BT.709
> 
>+#define IS_ALIGNED(p, a) (!((uintptr_t)(p) & ((a)-1)))
>+
>+#define align_buffer_64(var, size)                                           \
>+  uint8_t* var##_mem = (uint8_t*)(malloc((size) + 63));         /* NOLINT */ \
>+  uint8_t* var = (uint8_t*)(((intptr_t)(var##_mem) + 63) & ~63) /* NOLINT */
>+
>+#define free_aligned_buffer_64(var) \
>+  free(var##_mem);                  \
>+  var = 0
>+
> #if defined(__APPLE__) || defined(__x86_64__) || defined(__llvm__)
> #define OMITFP
> #else
>@@ -601,62 +584,6 @@ extern const struct YuvConstants SIMD_ALIGNED(kYvuH709Constants);  // BT.709
> #else
> #define LABELALIGN
> #endif
>-#if defined(__native_client__) && defined(__x86_64__)
>-// r14 is used for MEMOP macros.
>-#define NACL_R14 "r14",
>-#define BUNDLELOCK ".bundle_lock\n"
>-#define BUNDLEUNLOCK ".bundle_unlock\n"
>-#define MEMACCESS(base) "%%nacl:(%%r15,%q" #base ")"
>-#define MEMACCESS2(offset, base) "%%nacl:" #offset "(%%r15,%q" #base ")"
>-#define MEMLEA(offset, base) #offset "(%q" #base ")"
>-#define MEMLEA3(offset, index, scale) #offset "(,%q" #index "," #scale ")"
>-#define MEMLEA4(offset, base, index, scale) \
>-  #offset "(%q" #base ",%q" #index "," #scale ")"
>-#define MEMMOVESTRING(s, d) "%%nacl:(%q" #s "),%%nacl:(%q" #d "), %%r15"
>-#define MEMSTORESTRING(reg, d) "%%" #reg ",%%nacl:(%q" #d "), %%r15"
>-#define MEMOPREG(opcode, offset, base, index, scale, reg)                 \
>-  BUNDLELOCK                                                              \
>-  "lea " #offset "(%q" #base ",%q" #index "," #scale "),%%r14d\n" #opcode \
>-  " (%%r15,%%r14),%%" #reg "\n" BUNDLEUNLOCK
>-#define MEMOPMEM(opcode, reg, offset, base, index, scale)                 \
>-  BUNDLELOCK                                                              \
>-  "lea " #offset "(%q" #base ",%q" #index "," #scale "),%%r14d\n" #opcode \
>-  " %%" #reg ",(%%r15,%%r14)\n" BUNDLEUNLOCK
>-#define MEMOPARG(opcode, offset, base, index, scale, arg)                 \
>-  BUNDLELOCK                                                              \
>-  "lea " #offset "(%q" #base ",%q" #index "," #scale "),%%r14d\n" #opcode \
>-  " (%%r15,%%r14),%" #arg "\n" BUNDLEUNLOCK
>-#define VMEMOPREG(opcode, offset, base, index, scale, reg1, reg2)         \
>-  BUNDLELOCK                                                              \
>-  "lea " #offset "(%q" #base ",%q" #index "," #scale "),%%r14d\n" #opcode \
>-  " (%%r15,%%r14),%%" #reg1 ",%%" #reg2 "\n" BUNDLEUNLOCK
>-#define VEXTOPMEM(op, sel, reg, offset, base, index, scale)           \
>-  BUNDLELOCK                                                          \
>-  "lea " #offset "(%q" #base ",%q" #index "," #scale "),%%r14d\n" #op \
>-  " $" #sel ",%%" #reg ",(%%r15,%%r14)\n" BUNDLEUNLOCK
>-#else  // defined(__native_client__) && defined(__x86_64__)
>-#define NACL_R14
>-#define BUNDLEALIGN
>-#define MEMACCESS(base) "(%" #base ")"
>-#define MEMACCESS2(offset, base) #offset "(%" #base ")"
>-#define MEMLEA(offset, base) #offset "(%" #base ")"
>-#define MEMLEA3(offset, index, scale) #offset "(,%" #index "," #scale ")"
>-#define MEMLEA4(offset, base, index, scale) \
>-  #offset "(%" #base ",%" #index "," #scale ")"
>-#define MEMMOVESTRING(s, d)
>-#define MEMSTORESTRING(reg, d)
>-#define MEMOPREG(opcode, offset, base, index, scale, reg) \
>-  #opcode " " #offset "(%" #base ",%" #index "," #scale "),%%" #reg "\n"
>-#define MEMOPMEM(opcode, reg, offset, base, index, scale) \
>-  #opcode " %%" #reg "," #offset "(%" #base ",%" #index "," #scale ")\n"
>-#define MEMOPARG(opcode, offset, base, index, scale, arg) \
>-  #opcode " " #offset "(%" #base ",%" #index "," #scale "),%" #arg "\n"
>-#define VMEMOPREG(opcode, offset, base, index, scale, reg1, reg2)    \
>-  #opcode " " #offset "(%" #base ",%" #index "," #scale "),%%" #reg1 \
>-          ",%%" #reg2 "\n"
>-#define VEXTOPMEM(op, sel, reg, offset, base, index, scale) \
>-  #op " $" #sel ",%%" #reg "," #offset "(%" #base ",%" #index "," #scale ")\n"
>-#endif  // defined(__native_client__) && defined(__x86_64__)
> 
> // Intel Code Analizer markers.  Insert IACA_START IACA_END around code to be
> // measured and then run with iaca -64 libyuv_unittest.
>@@ -709,2583 +636,2743 @@ extern const struct YuvConstants SIMD_ALIGNED(kYvuH709Constants);  // BT.709
>     IACA_UD_BYTES      \
>   }
> 
>-void I444ToARGBRow_NEON(const uint8* src_y,
>-                        const uint8* src_u,
>-                        const uint8* src_v,
>-                        uint8* dst_argb,
>+void I444ToARGBRow_NEON(const uint8_t* src_y,
>+                        const uint8_t* src_u,
>+                        const uint8_t* src_v,
>+                        uint8_t* dst_argb,
>                         const struct YuvConstants* yuvconstants,
>                         int width);
>-void I422ToARGBRow_NEON(const uint8* src_y,
>-                        const uint8* src_u,
>-                        const uint8* src_v,
>-                        uint8* dst_argb,
>+void I422ToARGBRow_NEON(const uint8_t* src_y,
>+                        const uint8_t* src_u,
>+                        const uint8_t* src_v,
>+                        uint8_t* dst_argb,
>                         const struct YuvConstants* yuvconstants,
>                         int width);
>-void I422AlphaToARGBRow_NEON(const uint8* y_buf,
>-                             const uint8* u_buf,
>-                             const uint8* v_buf,
>-                             const uint8* a_buf,
>-                             uint8* dst_argb,
>+void I422AlphaToARGBRow_NEON(const uint8_t* src_y,
>+                             const uint8_t* src_u,
>+                             const uint8_t* src_v,
>+                             const uint8_t* src_a,
>+                             uint8_t* dst_argb,
>                              const struct YuvConstants* yuvconstants,
>                              int width);
>-void I422ToARGBRow_NEON(const uint8* src_y,
>-                        const uint8* src_u,
>-                        const uint8* src_v,
>-                        uint8* dst_argb,
>+void I422ToARGBRow_NEON(const uint8_t* src_y,
>+                        const uint8_t* src_u,
>+                        const uint8_t* src_v,
>+                        uint8_t* dst_argb,
>                         const struct YuvConstants* yuvconstants,
>                         int width);
>-void I422ToRGBARow_NEON(const uint8* src_y,
>-                        const uint8* src_u,
>-                        const uint8* src_v,
>-                        uint8* dst_rgba,
>+void I422ToRGBARow_NEON(const uint8_t* src_y,
>+                        const uint8_t* src_u,
>+                        const uint8_t* src_v,
>+                        uint8_t* dst_rgba,
>                         const struct YuvConstants* yuvconstants,
>                         int width);
>-void I422ToRGB24Row_NEON(const uint8* src_y,
>-                         const uint8* src_u,
>-                         const uint8* src_v,
>-                         uint8* dst_rgb24,
>+void I422ToRGB24Row_NEON(const uint8_t* src_y,
>+                         const uint8_t* src_u,
>+                         const uint8_t* src_v,
>+                         uint8_t* dst_rgb24,
>                          const struct YuvConstants* yuvconstants,
>                          int width);
>-void I422ToRGB565Row_NEON(const uint8* src_y,
>-                          const uint8* src_u,
>-                          const uint8* src_v,
>-                          uint8* dst_rgb565,
>+void I422ToRGB565Row_NEON(const uint8_t* src_y,
>+                          const uint8_t* src_u,
>+                          const uint8_t* src_v,
>+                          uint8_t* dst_rgb565,
>                           const struct YuvConstants* yuvconstants,
>                           int width);
>-void I422ToARGB1555Row_NEON(const uint8* src_y,
>-                            const uint8* src_u,
>-                            const uint8* src_v,
>-                            uint8* dst_argb1555,
>+void I422ToARGB1555Row_NEON(const uint8_t* src_y,
>+                            const uint8_t* src_u,
>+                            const uint8_t* src_v,
>+                            uint8_t* dst_argb1555,
>                             const struct YuvConstants* yuvconstants,
>                             int width);
>-void I422ToARGB4444Row_NEON(const uint8* src_y,
>-                            const uint8* src_u,
>-                            const uint8* src_v,
>-                            uint8* dst_argb4444,
>+void I422ToARGB4444Row_NEON(const uint8_t* src_y,
>+                            const uint8_t* src_u,
>+                            const uint8_t* src_v,
>+                            uint8_t* dst_argb4444,
>                             const struct YuvConstants* yuvconstants,
>                             int width);
>-void NV12ToARGBRow_NEON(const uint8* src_y,
>-                        const uint8* src_uv,
>-                        uint8* dst_argb,
>+void NV12ToARGBRow_NEON(const uint8_t* src_y,
>+                        const uint8_t* src_uv,
>+                        uint8_t* dst_argb,
>                         const struct YuvConstants* yuvconstants,
>                         int width);
>-void NV12ToRGB565Row_NEON(const uint8* src_y,
>-                          const uint8* src_uv,
>-                          uint8* dst_rgb565,
>+void NV12ToRGB565Row_NEON(const uint8_t* src_y,
>+                          const uint8_t* src_uv,
>+                          uint8_t* dst_rgb565,
>                           const struct YuvConstants* yuvconstants,
>                           int width);
>-void NV21ToARGBRow_NEON(const uint8* src_y,
>-                        const uint8* src_vu,
>-                        uint8* dst_argb,
>+void NV21ToARGBRow_NEON(const uint8_t* src_y,
>+                        const uint8_t* src_vu,
>+                        uint8_t* dst_argb,
>                         const struct YuvConstants* yuvconstants,
>                         int width);
>-void YUY2ToARGBRow_NEON(const uint8* src_yuy2,
>-                        uint8* dst_argb,
>+void NV12ToRGB24Row_NEON(const uint8_t* src_y,
>+                         const uint8_t* src_uv,
>+                         uint8_t* dst_rgb24,
>+                         const struct YuvConstants* yuvconstants,
>+                         int width);
>+void NV21ToRGB24Row_NEON(const uint8_t* src_y,
>+                         const uint8_t* src_vu,
>+                         uint8_t* dst_rgb24,
>+                         const struct YuvConstants* yuvconstants,
>+                         int width);
>+void YUY2ToARGBRow_NEON(const uint8_t* src_yuy2,
>+                        uint8_t* dst_argb,
>                         const struct YuvConstants* yuvconstants,
>                         int width);
>-void UYVYToARGBRow_NEON(const uint8* src_uyvy,
>-                        uint8* dst_argb,
>+void UYVYToARGBRow_NEON(const uint8_t* src_uyvy,
>+                        uint8_t* dst_argb,
>                         const struct YuvConstants* yuvconstants,
>                         int width);
>-void I444ToARGBRow_MSA(const uint8* src_y,
>-                       const uint8* src_u,
>-                       const uint8* src_v,
>-                       uint8* dst_argb,
>+void I444ToARGBRow_MSA(const uint8_t* src_y,
>+                       const uint8_t* src_u,
>+                       const uint8_t* src_v,
>+                       uint8_t* dst_argb,
>                        const struct YuvConstants* yuvconstants,
>                        int width);
>-void I444ToARGBRow_DSPR2(const uint8* src_y,
>-                         const uint8* src_u,
>-                         const uint8* src_v,
>-                         uint8* dst_argb,
>-                         const struct YuvConstants* yuvconstants,
>-                         int width);
>-void I422ToARGB4444Row_DSPR2(const uint8* src_y,
>-                             const uint8* src_u,
>-                             const uint8* src_v,
>-                             uint8* dst_argb4444,
>-                             const struct YuvConstants* yuvconstants,
>-                             int width);
>-void I422ToARGB1555Row_DSPR2(const uint8* src_y,
>-                             const uint8* src_u,
>-                             const uint8* src_v,
>-                             uint8* dst_argb1555,
>-                             const struct YuvConstants* yuvconstants,
>-                             int width);
>-void NV12ToARGBRow_DSPR2(const uint8* src_y,
>-                         const uint8* src_uv,
>-                         uint8* dst_argb,
>-                         const struct YuvConstants* yuvconstants,
>-                         int width);
> 
>-void I422ToARGBRow_MSA(const uint8* src_y,
>-                       const uint8* src_u,
>-                       const uint8* src_v,
>-                       uint8* dst_argb,
>+void I422ToARGBRow_MSA(const uint8_t* src_y,
>+                       const uint8_t* src_u,
>+                       const uint8_t* src_v,
>+                       uint8_t* dst_argb,
>                        const struct YuvConstants* yuvconstants,
>                        int width);
>-void I422ToRGBARow_MSA(const uint8* src_y,
>-                       const uint8* src_u,
>-                       const uint8* src_v,
>-                       uint8* dst_rgba,
>+void I422ToRGBARow_MSA(const uint8_t* src_y,
>+                       const uint8_t* src_u,
>+                       const uint8_t* src_v,
>+                       uint8_t* dst_argb,
>                        const struct YuvConstants* yuvconstants,
>                        int width);
>-void I422AlphaToARGBRow_MSA(const uint8* y_buf,
>-                            const uint8* u_buf,
>-                            const uint8* v_buf,
>-                            const uint8* a_buf,
>-                            uint8* dst_argb,
>+void I422AlphaToARGBRow_MSA(const uint8_t* src_y,
>+                            const uint8_t* src_u,
>+                            const uint8_t* src_v,
>+                            const uint8_t* src_a,
>+                            uint8_t* dst_argb,
>                             const struct YuvConstants* yuvconstants,
>                             int width);
>-void I422ToRGB24Row_MSA(const uint8* src_y,
>-                        const uint8* src_u,
>-                        const uint8* src_v,
>-                        uint8* dst_rgb24,
>+void I422ToRGB24Row_MSA(const uint8_t* src_y,
>+                        const uint8_t* src_u,
>+                        const uint8_t* src_v,
>+                        uint8_t* dst_argb,
>                         const struct YuvConstants* yuvconstants,
>                         int width);
>-void I422ToRGB565Row_MSA(const uint8* src_y,
>-                         const uint8* src_u,
>-                         const uint8* src_v,
>-                         uint8* dst_rgb565,
>+void I422ToRGB565Row_MSA(const uint8_t* src_y,
>+                         const uint8_t* src_u,
>+                         const uint8_t* src_v,
>+                         uint8_t* dst_rgb565,
>                          const struct YuvConstants* yuvconstants,
>                          int width);
>-void I422ToARGB4444Row_MSA(const uint8* src_y,
>-                           const uint8* src_u,
>-                           const uint8* src_v,
>-                           uint8* dst_argb4444,
>+void I422ToARGB4444Row_MSA(const uint8_t* src_y,
>+                           const uint8_t* src_u,
>+                           const uint8_t* src_v,
>+                           uint8_t* dst_argb4444,
>                            const struct YuvConstants* yuvconstants,
>                            int width);
>-void I422ToARGB1555Row_MSA(const uint8* src_y,
>-                           const uint8* src_u,
>-                           const uint8* src_v,
>-                           uint8* dst_argb1555,
>+void I422ToARGB1555Row_MSA(const uint8_t* src_y,
>+                           const uint8_t* src_u,
>+                           const uint8_t* src_v,
>+                           uint8_t* dst_argb1555,
>                            const struct YuvConstants* yuvconstants,
>                            int width);
>-void NV12ToARGBRow_MSA(const uint8* src_y,
>-                       const uint8* src_uv,
>-                       uint8* dst_argb,
>+void NV12ToARGBRow_MSA(const uint8_t* src_y,
>+                       const uint8_t* src_uv,
>+                       uint8_t* dst_argb,
>                        const struct YuvConstants* yuvconstants,
>                        int width);
>-void NV12ToRGB565Row_MSA(const uint8* src_y,
>-                         const uint8* src_uv,
>-                         uint8* dst_rgb565,
>+void NV12ToRGB565Row_MSA(const uint8_t* src_y,
>+                         const uint8_t* src_uv,
>+                         uint8_t* dst_rgb565,
>                          const struct YuvConstants* yuvconstants,
>                          int width);
>-void NV21ToARGBRow_MSA(const uint8* src_y,
>-                       const uint8* src_vu,
>-                       uint8* dst_argb,
>+void NV21ToARGBRow_MSA(const uint8_t* src_y,
>+                       const uint8_t* src_vu,
>+                       uint8_t* dst_argb,
>                        const struct YuvConstants* yuvconstants,
>                        int width);
>-void YUY2ToARGBRow_MSA(const uint8* src_yuy2,
>-                       uint8* dst_argb,
>+void YUY2ToARGBRow_MSA(const uint8_t* src_yuy2,
>+                       uint8_t* dst_argb,
>                        const struct YuvConstants* yuvconstants,
>                        int width);
>-void UYVYToARGBRow_MSA(const uint8* src_uyvy,
>-                       uint8* dst_argb,
>+void UYVYToARGBRow_MSA(const uint8_t* src_uyvy,
>+                       uint8_t* dst_argb,
>                        const struct YuvConstants* yuvconstants,
>                        int width);
> 
>-void ARGBToYRow_AVX2(const uint8* src_argb, uint8* dst_y, int width);
>-void ARGBToYRow_Any_AVX2(const uint8* src_argb, uint8* dst_y, int width);
>-void ARGBToYRow_SSSE3(const uint8* src_argb, uint8* dst_y, int width);
>-void ARGBToYJRow_AVX2(const uint8* src_argb, uint8* dst_y, int width);
>-void ARGBToYJRow_Any_AVX2(const uint8* src_argb, uint8* dst_y, int width);
>-void ARGBToYJRow_SSSE3(const uint8* src_argb, uint8* dst_y, int width);
>-void BGRAToYRow_SSSE3(const uint8* src_bgra, uint8* dst_y, int width);
>-void ABGRToYRow_SSSE3(const uint8* src_abgr, uint8* dst_y, int width);
>-void RGBAToYRow_SSSE3(const uint8* src_rgba, uint8* dst_y, int width);
>-void RGB24ToYRow_SSSE3(const uint8* src_rgb24, uint8* dst_y, int width);
>-void RAWToYRow_SSSE3(const uint8* src_raw, uint8* dst_y, int width);
>-void ARGBToYRow_NEON(const uint8* src_argb, uint8* dst_y, int width);
>-void ARGBToYJRow_NEON(const uint8* src_argb, uint8* dst_y, int width);
>-void ARGBToYRow_MSA(const uint8* src_argb, uint8* dst_y, int width);
>-void ARGBToYJRow_MSA(const uint8* src_argb, uint8* dst_y, int width);
>-void ARGBToUV444Row_NEON(const uint8* src_argb,
>-                         uint8* dst_u,
>-                         uint8* dst_v,
>-                         int width);
>-void ARGBToUVRow_NEON(const uint8* src_argb,
>+void ARGBToYRow_AVX2(const uint8_t* src_argb, uint8_t* dst_y, int width);
>+void ARGBToYRow_Any_AVX2(const uint8_t* src_ptr, uint8_t* dst_ptr, int width);
>+void ARGBToYRow_SSSE3(const uint8_t* src_argb, uint8_t* dst_y, int width);
>+void ARGBToYJRow_AVX2(const uint8_t* src_argb, uint8_t* dst_y, int width);
>+void ARGBToYJRow_Any_AVX2(const uint8_t* src_ptr, uint8_t* dst_ptr, int width);
>+void ARGBToYJRow_SSSE3(const uint8_t* src_argb, uint8_t* dst_y, int width);
>+void BGRAToYRow_SSSE3(const uint8_t* src_bgra, uint8_t* dst_y, int width);
>+void ABGRToYRow_SSSE3(const uint8_t* src_abgr, uint8_t* dst_y, int width);
>+void RGBAToYRow_SSSE3(const uint8_t* src_rgba, uint8_t* dst_y, int width);
>+void RGB24ToYRow_SSSE3(const uint8_t* src_rgb24, uint8_t* dst_y, int width);
>+void RAWToYRow_SSSE3(const uint8_t* src_raw, uint8_t* dst_y, int width);
>+void ARGBToYRow_NEON(const uint8_t* src_argb, uint8_t* dst_y, int width);
>+void ARGBToYJRow_NEON(const uint8_t* src_argb, uint8_t* dst_y, int width);
>+void ARGBToYRow_MSA(const uint8_t* src_argb0, uint8_t* dst_y, int width);
>+void ARGBToYJRow_MSA(const uint8_t* src_argb0, uint8_t* dst_y, int width);
>+void ARGBToUV444Row_NEON(const uint8_t* src_argb,
>+                         uint8_t* dst_u,
>+                         uint8_t* dst_v,
>+                         int width);
>+void ARGBToUVRow_NEON(const uint8_t* src_argb,
>                       int src_stride_argb,
>-                      uint8* dst_u,
>-                      uint8* dst_v,
>+                      uint8_t* dst_u,
>+                      uint8_t* dst_v,
>                       int width);
>-void ARGBToUV444Row_MSA(const uint8* src_argb,
>-                        uint8* dst_u,
>-                        uint8* dst_v,
>+void ARGBToUV444Row_MSA(const uint8_t* src_argb,
>+                        uint8_t* dst_u,
>+                        uint8_t* dst_v,
>                         int width);
>-void ARGBToUVRow_MSA(const uint8* src_argb,
>+void ARGBToUVRow_MSA(const uint8_t* src_argb0,
>                      int src_stride_argb,
>-                     uint8* dst_u,
>-                     uint8* dst_v,
>+                     uint8_t* dst_u,
>+                     uint8_t* dst_v,
>                      int width);
>-void ARGBToUVJRow_NEON(const uint8* src_argb,
>+void ARGBToUVJRow_NEON(const uint8_t* src_argb,
>                        int src_stride_argb,
>-                       uint8* dst_u,
>-                       uint8* dst_v,
>+                       uint8_t* dst_u,
>+                       uint8_t* dst_v,
>                        int width);
>-void BGRAToUVRow_NEON(const uint8* src_bgra,
>+void BGRAToUVRow_NEON(const uint8_t* src_bgra,
>                       int src_stride_bgra,
>-                      uint8* dst_u,
>-                      uint8* dst_v,
>+                      uint8_t* dst_u,
>+                      uint8_t* dst_v,
>                       int width);
>-void ABGRToUVRow_NEON(const uint8* src_abgr,
>+void ABGRToUVRow_NEON(const uint8_t* src_abgr,
>                       int src_stride_abgr,
>-                      uint8* dst_u,
>-                      uint8* dst_v,
>+                      uint8_t* dst_u,
>+                      uint8_t* dst_v,
>                       int width);
>-void RGBAToUVRow_NEON(const uint8* src_rgba,
>+void RGBAToUVRow_NEON(const uint8_t* src_rgba,
>                       int src_stride_rgba,
>-                      uint8* dst_u,
>-                      uint8* dst_v,
>+                      uint8_t* dst_u,
>+                      uint8_t* dst_v,
>                       int width);
>-void RGB24ToUVRow_NEON(const uint8* src_rgb24,
>+void RGB24ToUVRow_NEON(const uint8_t* src_rgb24,
>                        int src_stride_rgb24,
>-                       uint8* dst_u,
>-                       uint8* dst_v,
>+                       uint8_t* dst_u,
>+                       uint8_t* dst_v,
>                        int width);
>-void RAWToUVRow_NEON(const uint8* src_raw,
>+void RAWToUVRow_NEON(const uint8_t* src_raw,
>                      int src_stride_raw,
>-                     uint8* dst_u,
>-                     uint8* dst_v,
>+                     uint8_t* dst_u,
>+                     uint8_t* dst_v,
>                      int width);
>-void RGB565ToUVRow_NEON(const uint8* src_rgb565,
>+void RGB565ToUVRow_NEON(const uint8_t* src_rgb565,
>                         int src_stride_rgb565,
>-                        uint8* dst_u,
>-                        uint8* dst_v,
>+                        uint8_t* dst_u,
>+                        uint8_t* dst_v,
>                         int width);
>-void ARGB1555ToUVRow_NEON(const uint8* src_argb1555,
>+void ARGB1555ToUVRow_NEON(const uint8_t* src_argb1555,
>                           int src_stride_argb1555,
>-                          uint8* dst_u,
>-                          uint8* dst_v,
>+                          uint8_t* dst_u,
>+                          uint8_t* dst_v,
>                           int width);
>-void ARGB4444ToUVRow_NEON(const uint8* src_argb4444,
>+void ARGB4444ToUVRow_NEON(const uint8_t* src_argb4444,
>                           int src_stride_argb4444,
>-                          uint8* dst_u,
>-                          uint8* dst_v,
>+                          uint8_t* dst_u,
>+                          uint8_t* dst_v,
>                           int width);
>-void ARGBToUVJRow_MSA(const uint8* src_argb,
>-                      int src_stride_argb,
>-                      uint8* dst_u,
>-                      uint8* dst_v,
>+void ARGBToUVJRow_MSA(const uint8_t* src_rgb0,
>+                      int src_stride_rgb,
>+                      uint8_t* dst_u,
>+                      uint8_t* dst_v,
>                       int width);
>-void BGRAToUVRow_MSA(const uint8* src_bgra,
>-                     int src_stride_bgra,
>-                     uint8* dst_u,
>-                     uint8* dst_v,
>+void BGRAToUVRow_MSA(const uint8_t* src_rgb0,
>+                     int src_stride_rgb,
>+                     uint8_t* dst_u,
>+                     uint8_t* dst_v,
>                      int width);
>-void ABGRToUVRow_MSA(const uint8* src_abgr,
>-                     int src_stride_abgr,
>-                     uint8* dst_u,
>-                     uint8* dst_v,
>+void ABGRToUVRow_MSA(const uint8_t* src_rgb0,
>+                     int src_stride_rgb,
>+                     uint8_t* dst_u,
>+                     uint8_t* dst_v,
>                      int width);
>-void RGBAToUVRow_MSA(const uint8* src_rgba,
>-                     int src_stride_rgba,
>-                     uint8* dst_u,
>-                     uint8* dst_v,
>+void RGBAToUVRow_MSA(const uint8_t* src_rgb0,
>+                     int src_stride_rgb,
>+                     uint8_t* dst_u,
>+                     uint8_t* dst_v,
>                      int width);
>-void RGB24ToUVRow_MSA(const uint8* src_rgb24,
>-                      int src_stride_rgb24,
>-                      uint8* dst_u,
>-                      uint8* dst_v,
>+void RGB24ToUVRow_MSA(const uint8_t* src_rgb0,
>+                      int src_stride_rgb,
>+                      uint8_t* dst_u,
>+                      uint8_t* dst_v,
>                       int width);
>-void RAWToUVRow_MSA(const uint8* src_raw,
>-                    int src_stride_raw,
>-                    uint8* dst_u,
>-                    uint8* dst_v,
>+void RAWToUVRow_MSA(const uint8_t* src_rgb0,
>+                    int src_stride_rgb,
>+                    uint8_t* dst_u,
>+                    uint8_t* dst_v,
>                     int width);
>-void RGB565ToUVRow_MSA(const uint8* src_rgb565,
>+void RGB565ToUVRow_MSA(const uint8_t* src_rgb565,
>                        int src_stride_rgb565,
>-                       uint8* dst_u,
>-                       uint8* dst_v,
>+                       uint8_t* dst_u,
>+                       uint8_t* dst_v,
>                        int width);
>-void ARGB1555ToUVRow_MSA(const uint8* src_argb1555,
>+void ARGB1555ToUVRow_MSA(const uint8_t* src_argb1555,
>                          int src_stride_argb1555,
>-                         uint8* dst_u,
>-                         uint8* dst_v,
>-                         int width);
>-void BGRAToYRow_NEON(const uint8* src_bgra, uint8* dst_y, int width);
>-void ABGRToYRow_NEON(const uint8* src_abgr, uint8* dst_y, int width);
>-void RGBAToYRow_NEON(const uint8* src_rgba, uint8* dst_y, int width);
>-void RGB24ToYRow_NEON(const uint8* src_rgb24, uint8* dst_y, int width);
>-void RAWToYRow_NEON(const uint8* src_raw, uint8* dst_y, int width);
>-void RGB565ToYRow_NEON(const uint8* src_rgb565, uint8* dst_y, int width);
>-void ARGB1555ToYRow_NEON(const uint8* src_argb1555, uint8* dst_y, int width);
>-void ARGB4444ToYRow_NEON(const uint8* src_argb4444, uint8* dst_y, int width);
>-void BGRAToYRow_MSA(const uint8* src_bgra, uint8* dst_y, int width);
>-void ABGRToYRow_MSA(const uint8* src_abgr, uint8* dst_y, int width);
>-void RGBAToYRow_MSA(const uint8* src_rgba, uint8* dst_y, int width);
>-void RGB24ToYRow_MSA(const uint8* src_rgb24, uint8* dst_y, int width);
>-void RAWToYRow_MSA(const uint8* src_raw, uint8* dst_y, int width);
>-void RGB565ToYRow_MSA(const uint8* src_rgb565, uint8* dst_y, int width);
>-void ARGB1555ToYRow_MSA(const uint8* src_argb1555, uint8* dst_y, int width);
>-void BGRAToUVRow_DSPR2(const uint8* src_bgra,
>-                       int src_stride_bgra,
>-                       uint8* dst_u,
>-                       uint8* dst_v,
>-                       int width);
>-void BGRAToYRow_DSPR2(const uint8* src_bgra, uint8* dst_y, int width);
>-void ABGRToUVRow_DSPR2(const uint8* src_abgr,
>-                       int src_stride_abgr,
>-                       uint8* dst_u,
>-                       uint8* dst_v,
>-                       int width);
>-void ARGBToYRow_DSPR2(const uint8* src_argb, uint8* dst_y, int width);
>-void ABGRToYRow_DSPR2(const uint8* src_abgr, uint8* dst_y, int width);
>-void RGBAToUVRow_DSPR2(const uint8* src_rgba,
>-                       int src_stride_rgba,
>-                       uint8* dst_u,
>-                       uint8* dst_v,
>-                       int width);
>-void RGBAToYRow_DSPR2(const uint8* src_rgba, uint8* dst_y, int width);
>-void ARGBToUVRow_DSPR2(const uint8* src_argb,
>-                       int src_stride_argb,
>-                       uint8* dst_u,
>-                       uint8* dst_v,
>-                       int width);
>-void ARGBToYRow_C(const uint8* src_argb, uint8* dst_y, int width);
>-void ARGBToYJRow_C(const uint8* src_argb, uint8* dst_y, int width);
>-void BGRAToYRow_C(const uint8* src_bgra, uint8* dst_y, int width);
>-void ABGRToYRow_C(const uint8* src_abgr, uint8* dst_y, int width);
>-void RGBAToYRow_C(const uint8* src_rgba, uint8* dst_y, int width);
>-void RGB24ToYRow_C(const uint8* src_rgb24, uint8* dst_y, int width);
>-void RAWToYRow_C(const uint8* src_raw, uint8* dst_y, int width);
>-void RGB565ToYRow_C(const uint8* src_rgb565, uint8* dst_y, int width);
>-void ARGB1555ToYRow_C(const uint8* src_argb1555, uint8* dst_y, int width);
>-void ARGB4444ToYRow_C(const uint8* src_argb4444, uint8* dst_y, int width);
>-void ARGBToYRow_Any_SSSE3(const uint8* src_argb, uint8* dst_y, int width);
>-void ARGBToYJRow_Any_SSSE3(const uint8* src_argb, uint8* dst_y, int width);
>-void BGRAToYRow_Any_SSSE3(const uint8* src_bgra, uint8* dst_y, int width);
>-void ABGRToYRow_Any_SSSE3(const uint8* src_abgr, uint8* dst_y, int width);
>-void RGBAToYRow_Any_SSSE3(const uint8* src_rgba, uint8* dst_y, int width);
>-void RGB24ToYRow_Any_SSSE3(const uint8* src_rgb24, uint8* dst_y, int width);
>-void RAWToYRow_Any_SSSE3(const uint8* src_raw, uint8* dst_y, int width);
>-void ARGBToYRow_Any_NEON(const uint8* src_argb, uint8* dst_y, int width);
>-void ARGBToYJRow_Any_NEON(const uint8* src_argb, uint8* dst_y, int width);
>-void BGRAToYRow_Any_NEON(const uint8* src_bgra, uint8* dst_y, int width);
>-void ABGRToYRow_Any_NEON(const uint8* src_abgr, uint8* dst_y, int width);
>-void RGBAToYRow_Any_NEON(const uint8* src_rgba, uint8* dst_y, int width);
>-void RGB24ToYRow_Any_NEON(const uint8* src_rgb24, uint8* dst_y, int width);
>-void RAWToYRow_Any_NEON(const uint8* src_raw, uint8* dst_y, int width);
>-void RGB565ToYRow_Any_NEON(const uint8* src_rgb565, uint8* dst_y, int width);
>-void ARGB1555ToYRow_Any_NEON(const uint8* src_argb1555,
>-                             uint8* dst_y,
>+                         uint8_t* dst_u,
>+                         uint8_t* dst_v,
>+                         int width);
>+void BGRAToYRow_NEON(const uint8_t* src_bgra, uint8_t* dst_y, int width);
>+void ABGRToYRow_NEON(const uint8_t* src_abgr, uint8_t* dst_y, int width);
>+void RGBAToYRow_NEON(const uint8_t* src_rgba, uint8_t* dst_y, int width);
>+void RGB24ToYRow_NEON(const uint8_t* src_rgb24, uint8_t* dst_y, int width);
>+void RAWToYRow_NEON(const uint8_t* src_raw, uint8_t* dst_y, int width);
>+void RGB565ToYRow_NEON(const uint8_t* src_rgb565, uint8_t* dst_y, int width);
>+void ARGB1555ToYRow_NEON(const uint8_t* src_argb1555,
>+                         uint8_t* dst_y,
>+                         int width);
>+void ARGB4444ToYRow_NEON(const uint8_t* src_argb4444,
>+                         uint8_t* dst_y,
>+                         int width);
>+void BGRAToYRow_MSA(const uint8_t* src_argb0, uint8_t* dst_y, int width);
>+void ABGRToYRow_MSA(const uint8_t* src_argb0, uint8_t* dst_y, int width);
>+void RGBAToYRow_MSA(const uint8_t* src_argb0, uint8_t* dst_y, int width);
>+void RGB24ToYRow_MSA(const uint8_t* src_argb0, uint8_t* dst_y, int width);
>+void RAWToYRow_MSA(const uint8_t* src_argb0, uint8_t* dst_y, int width);
>+void RGB565ToYRow_MSA(const uint8_t* src_rgb565, uint8_t* dst_y, int width);
>+void ARGB1555ToYRow_MSA(const uint8_t* src_argb1555, uint8_t* dst_y, int width);
>+void ARGBToYRow_C(const uint8_t* src_argb0, uint8_t* dst_y, int width);
>+void ARGBToYJRow_C(const uint8_t* src_argb0, uint8_t* dst_y, int width);
>+void BGRAToYRow_C(const uint8_t* src_argb0, uint8_t* dst_y, int width);
>+void ABGRToYRow_C(const uint8_t* src_argb0, uint8_t* dst_y, int width);
>+void RGBAToYRow_C(const uint8_t* src_argb0, uint8_t* dst_y, int width);
>+void RGB24ToYRow_C(const uint8_t* src_argb0, uint8_t* dst_y, int width);
>+void RAWToYRow_C(const uint8_t* src_argb0, uint8_t* dst_y, int width);
>+void RGB565ToYRow_C(const uint8_t* src_rgb565, uint8_t* dst_y, int width);
>+void ARGB1555ToYRow_C(const uint8_t* src_argb1555, uint8_t* dst_y, int width);
>+void ARGB4444ToYRow_C(const uint8_t* src_argb4444, uint8_t* dst_y, int width);
>+void ARGBToYRow_Any_SSSE3(const uint8_t* src_ptr, uint8_t* dst_ptr, int width);
>+void ARGBToYJRow_Any_SSSE3(const uint8_t* src_ptr, uint8_t* dst_ptr, int width);
>+void BGRAToYRow_Any_SSSE3(const uint8_t* src_ptr, uint8_t* dst_ptr, int width);
>+void ABGRToYRow_Any_SSSE3(const uint8_t* src_ptr, uint8_t* dst_ptr, int width);
>+void RGBAToYRow_Any_SSSE3(const uint8_t* src_ptr, uint8_t* dst_ptr, int width);
>+void RGB24ToYRow_Any_SSSE3(const uint8_t* src_rgb24, uint8_t* dst_y, int width);
>+void RAWToYRow_Any_SSSE3(const uint8_t* src_raw, uint8_t* dst_y, int width);
>+void ARGBToYRow_Any_NEON(const uint8_t* src_ptr, uint8_t* dst_ptr, int width);
>+void ARGBToYJRow_Any_NEON(const uint8_t* src_ptr, uint8_t* dst_ptr, int width);
>+void BGRAToYRow_Any_NEON(const uint8_t* src_ptr, uint8_t* dst_ptr, int width);
>+void ABGRToYRow_Any_NEON(const uint8_t* src_ptr, uint8_t* dst_ptr, int width);
>+void RGBAToYRow_Any_NEON(const uint8_t* src_ptr, uint8_t* dst_ptr, int width);
>+void RGB24ToYRow_Any_NEON(const uint8_t* src_ptr, uint8_t* dst_ptr, int width);
>+void RAWToYRow_Any_NEON(const uint8_t* src_ptr, uint8_t* dst_ptr, int width);
>+void RGB565ToYRow_Any_NEON(const uint8_t* src_ptr, uint8_t* dst_ptr, int width);
>+void ARGB1555ToYRow_Any_NEON(const uint8_t* src_ptr,
>+                             uint8_t* dst_ptr,
>                              int width);
>-void BGRAToYRow_Any_DSPR2(const uint8* src_bgra, uint8* dst_y, int width);
>-void ARGBToYRow_Any_DSPR2(const uint8* src_argb, uint8* dst_y, int width);
>-void ABGRToYRow_Any_DSPR2(const uint8* src_abgr, uint8* dst_y, int width);
>-void RGBAToYRow_Any_DSPR2(const uint8* src_rgba, uint8* dst_y, int width);
>-void ARGB4444ToYRow_Any_NEON(const uint8* src_argb4444,
>-                             uint8* dst_y,
>+void ARGB4444ToYRow_Any_NEON(const uint8_t* src_ptr,
>+                             uint8_t* dst_ptr,
>                              int width);
>-void BGRAToYRow_Any_MSA(const uint8* src_bgra, uint8* dst_y, int width);
>-void ABGRToYRow_Any_MSA(const uint8* src_abgr, uint8* dst_y, int width);
>-void RGBAToYRow_Any_MSA(const uint8* src_rgba, uint8* dst_y, int width);
>-void ARGBToYJRow_Any_MSA(const uint8* src_argb, uint8* dst_y, int width);
>-void ARGBToYRow_Any_MSA(const uint8* src_argb, uint8* dst_y, int width);
>-void RGB24ToYRow_Any_MSA(const uint8* src_rgb24, uint8* dst_y, int width);
>-void RAWToYRow_Any_MSA(const uint8* src_raw, uint8* dst_y, int width);
>-void RGB565ToYRow_Any_MSA(const uint8* src_rgb565, uint8* dst_y, int width);
>-void ARGB1555ToYRow_Any_MSA(const uint8* src_argb1555, uint8* dst_y, int width);
>-
>-void ARGBToUVRow_AVX2(const uint8* src_argb,
>+void BGRAToYRow_Any_MSA(const uint8_t* src_ptr, uint8_t* dst_ptr, int width);
>+void ABGRToYRow_Any_MSA(const uint8_t* src_ptr, uint8_t* dst_ptr, int width);
>+void RGBAToYRow_Any_MSA(const uint8_t* src_ptr, uint8_t* dst_ptr, int width);
>+void ARGBToYJRow_Any_MSA(const uint8_t* src_ptr, uint8_t* dst_ptr, int width);
>+void ARGBToYRow_Any_MSA(const uint8_t* src_ptr, uint8_t* dst_ptr, int width);
>+void RGB24ToYRow_Any_MSA(const uint8_t* src_ptr, uint8_t* dst_ptr, int width);
>+void RAWToYRow_Any_MSA(const uint8_t* src_ptr, uint8_t* dst_ptr, int width);
>+void RGB565ToYRow_Any_MSA(const uint8_t* src_ptr, uint8_t* dst_ptr, int width);
>+void ARGB1555ToYRow_Any_MSA(const uint8_t* src_ptr,
>+                            uint8_t* dst_ptr,
>+                            int width);
>+
>+void ARGBToUVRow_AVX2(const uint8_t* src_argb0,
>                       int src_stride_argb,
>-                      uint8* dst_u,
>-                      uint8* dst_v,
>+                      uint8_t* dst_u,
>+                      uint8_t* dst_v,
>                       int width);
>-void ARGBToUVJRow_AVX2(const uint8* src_argb,
>+void ARGBToUVJRow_AVX2(const uint8_t* src_argb0,
>                        int src_stride_argb,
>-                       uint8* dst_u,
>-                       uint8* dst_v,
>+                       uint8_t* dst_u,
>+                       uint8_t* dst_v,
>                        int width);
>-void ARGBToUVRow_SSSE3(const uint8* src_argb,
>+void ARGBToUVRow_SSSE3(const uint8_t* src_argb0,
>                        int src_stride_argb,
>-                       uint8* dst_u,
>-                       uint8* dst_v,
>+                       uint8_t* dst_u,
>+                       uint8_t* dst_v,
>                        int width);
>-void ARGBToUVJRow_SSSE3(const uint8* src_argb,
>+void ARGBToUVJRow_SSSE3(const uint8_t* src_argb0,
>                         int src_stride_argb,
>-                        uint8* dst_u,
>-                        uint8* dst_v,
>+                        uint8_t* dst_u,
>+                        uint8_t* dst_v,
>                         int width);
>-void BGRAToUVRow_SSSE3(const uint8* src_bgra,
>+void BGRAToUVRow_SSSE3(const uint8_t* src_bgra0,
>                        int src_stride_bgra,
>-                       uint8* dst_u,
>-                       uint8* dst_v,
>+                       uint8_t* dst_u,
>+                       uint8_t* dst_v,
>                        int width);
>-void ABGRToUVRow_SSSE3(const uint8* src_abgr,
>+void ABGRToUVRow_SSSE3(const uint8_t* src_abgr0,
>                        int src_stride_abgr,
>-                       uint8* dst_u,
>-                       uint8* dst_v,
>+                       uint8_t* dst_u,
>+                       uint8_t* dst_v,
>                        int width);
>-void RGBAToUVRow_SSSE3(const uint8* src_rgba,
>+void RGBAToUVRow_SSSE3(const uint8_t* src_rgba0,
>                        int src_stride_rgba,
>-                       uint8* dst_u,
>-                       uint8* dst_v,
>+                       uint8_t* dst_u,
>+                       uint8_t* dst_v,
>                        int width);
>-void ARGBToUVRow_Any_AVX2(const uint8* src_argb,
>-                          int src_stride_argb,
>-                          uint8* dst_u,
>-                          uint8* dst_v,
>+void ARGBToUVRow_Any_AVX2(const uint8_t* src_ptr,
>+                          int src_stride_ptr,
>+                          uint8_t* dst_u,
>+                          uint8_t* dst_v,
>                           int width);
>-void ARGBToUVJRow_Any_AVX2(const uint8* src_argb,
>-                           int src_stride_argb,
>-                           uint8* dst_u,
>-                           uint8* dst_v,
>+void ARGBToUVJRow_Any_AVX2(const uint8_t* src_ptr,
>+                           int src_stride_ptr,
>+                           uint8_t* dst_u,
>+                           uint8_t* dst_v,
>                            int width);
>-void ARGBToUVRow_Any_SSSE3(const uint8* src_argb,
>-                           int src_stride_argb,
>-                           uint8* dst_u,
>-                           uint8* dst_v,
>+void ARGBToUVRow_Any_SSSE3(const uint8_t* src_ptr,
>+                           int src_stride_ptr,
>+                           uint8_t* dst_u,
>+                           uint8_t* dst_v,
>                            int width);
>-void ARGBToUVJRow_Any_SSSE3(const uint8* src_argb,
>-                            int src_stride_argb,
>-                            uint8* dst_u,
>-                            uint8* dst_v,
>-                            int width);
>-void BGRAToUVRow_Any_SSSE3(const uint8* src_bgra,
>-                           int src_stride_bgra,
>-                           uint8* dst_u,
>-                           uint8* dst_v,
>+void ARGBToUVJRow_Any_SSSE3(const uint8_t* src_ptr,
>+                            int src_stride_ptr,
>+                            uint8_t* dst_u,
>+                            uint8_t* dst_v,
>+                            int width);
>+void BGRAToUVRow_Any_SSSE3(const uint8_t* src_ptr,
>+                           int src_stride_ptr,
>+                           uint8_t* dst_u,
>+                           uint8_t* dst_v,
>                            int width);
>-void ABGRToUVRow_Any_SSSE3(const uint8* src_abgr,
>-                           int src_stride_abgr,
>-                           uint8* dst_u,
>-                           uint8* dst_v,
>+void ABGRToUVRow_Any_SSSE3(const uint8_t* src_ptr,
>+                           int src_stride_ptr,
>+                           uint8_t* dst_u,
>+                           uint8_t* dst_v,
>                            int width);
>-void RGBAToUVRow_Any_SSSE3(const uint8* src_rgba,
>-                           int src_stride_rgba,
>-                           uint8* dst_u,
>-                           uint8* dst_v,
>+void RGBAToUVRow_Any_SSSE3(const uint8_t* src_ptr,
>+                           int src_stride_ptr,
>+                           uint8_t* dst_u,
>+                           uint8_t* dst_v,
>                            int width);
>-void ARGBToUV444Row_Any_NEON(const uint8* src_argb,
>-                             uint8* dst_u,
>-                             uint8* dst_v,
>+void ARGBToUV444Row_Any_NEON(const uint8_t* src_ptr,
>+                             uint8_t* dst_u,
>+                             uint8_t* dst_v,
>                              int width);
>-void ARGBToUVRow_Any_NEON(const uint8* src_argb,
>-                          int src_stride_argb,
>-                          uint8* dst_u,
>-                          uint8* dst_v,
>+void ARGBToUVRow_Any_NEON(const uint8_t* src_ptr,
>+                          int src_stride_ptr,
>+                          uint8_t* dst_u,
>+                          uint8_t* dst_v,
>                           int width);
>-void ARGBToUV444Row_Any_MSA(const uint8* src_argb,
>-                            uint8* dst_u,
>-                            uint8* dst_v,
>-                            int width);
>-void ARGBToUVRow_Any_MSA(const uint8* src_argb,
>-                         int src_stride_argb,
>-                         uint8* dst_u,
>-                         uint8* dst_v,
>-                         int width);
>-void ARGBToUVJRow_Any_NEON(const uint8* src_argb,
>-                           int src_stride_argb,
>-                           uint8* dst_u,
>-                           uint8* dst_v,
>+void ARGBToUV444Row_Any_MSA(const uint8_t* src_ptr,
>+                            uint8_t* dst_u,
>+                            uint8_t* dst_v,
>+                            int width);
>+void ARGBToUVRow_Any_MSA(const uint8_t* src_ptr,
>+                         int src_stride_ptr,
>+                         uint8_t* dst_u,
>+                         uint8_t* dst_v,
>+                         int width);
>+void ARGBToUVJRow_Any_NEON(const uint8_t* src_ptr,
>+                           int src_stride_ptr,
>+                           uint8_t* dst_u,
>+                           uint8_t* dst_v,
>                            int width);
>-void BGRAToUVRow_Any_NEON(const uint8* src_bgra,
>-                          int src_stride_bgra,
>-                          uint8* dst_u,
>-                          uint8* dst_v,
>+void BGRAToUVRow_Any_NEON(const uint8_t* src_ptr,
>+                          int src_stride_ptr,
>+                          uint8_t* dst_u,
>+                          uint8_t* dst_v,
>                           int width);
>-void ABGRToUVRow_Any_NEON(const uint8* src_abgr,
>-                          int src_stride_abgr,
>-                          uint8* dst_u,
>-                          uint8* dst_v,
>+void ABGRToUVRow_Any_NEON(const uint8_t* src_ptr,
>+                          int src_stride_ptr,
>+                          uint8_t* dst_u,
>+                          uint8_t* dst_v,
>                           int width);
>-void RGBAToUVRow_Any_NEON(const uint8* src_rgba,
>-                          int src_stride_rgba,
>-                          uint8* dst_u,
>-                          uint8* dst_v,
>+void RGBAToUVRow_Any_NEON(const uint8_t* src_ptr,
>+                          int src_stride_ptr,
>+                          uint8_t* dst_u,
>+                          uint8_t* dst_v,
>                           int width);
>-void RGB24ToUVRow_Any_NEON(const uint8* src_rgb24,
>-                           int src_stride_rgb24,
>-                           uint8* dst_u,
>-                           uint8* dst_v,
>+void RGB24ToUVRow_Any_NEON(const uint8_t* src_ptr,
>+                           int src_stride_ptr,
>+                           uint8_t* dst_u,
>+                           uint8_t* dst_v,
>                            int width);
>-void RAWToUVRow_Any_NEON(const uint8* src_raw,
>-                         int src_stride_raw,
>-                         uint8* dst_u,
>-                         uint8* dst_v,
>-                         int width);
>-void RGB565ToUVRow_Any_NEON(const uint8* src_rgb565,
>-                            int src_stride_rgb565,
>-                            uint8* dst_u,
>-                            uint8* dst_v,
>-                            int width);
>-void ARGB1555ToUVRow_Any_NEON(const uint8* src_argb1555,
>-                              int src_stride_argb1555,
>-                              uint8* dst_u,
>-                              uint8* dst_v,
>+void RAWToUVRow_Any_NEON(const uint8_t* src_ptr,
>+                         int src_stride_ptr,
>+                         uint8_t* dst_u,
>+                         uint8_t* dst_v,
>+                         int width);
>+void RGB565ToUVRow_Any_NEON(const uint8_t* src_ptr,
>+                            int src_stride_ptr,
>+                            uint8_t* dst_u,
>+                            uint8_t* dst_v,
>+                            int width);
>+void ARGB1555ToUVRow_Any_NEON(const uint8_t* src_ptr,
>+                              int src_stride_ptr,
>+                              uint8_t* dst_u,
>+                              uint8_t* dst_v,
>                               int width);
>-void ARGB4444ToUVRow_Any_NEON(const uint8* src_argb4444,
>-                              int src_stride_argb4444,
>-                              uint8* dst_u,
>-                              uint8* dst_v,
>+void ARGB4444ToUVRow_Any_NEON(const uint8_t* src_ptr,
>+                              int src_stride_ptr,
>+                              uint8_t* dst_u,
>+                              uint8_t* dst_v,
>                               int width);
>-void ARGBToUVJRow_Any_MSA(const uint8* src_argb,
>-                          int src_stride_argb,
>-                          uint8* dst_u,
>-                          uint8* dst_v,
>+void ARGBToUVJRow_Any_MSA(const uint8_t* src_ptr,
>+                          int src_stride_ptr,
>+                          uint8_t* dst_u,
>+                          uint8_t* dst_v,
>                           int width);
>-void BGRAToUVRow_Any_MSA(const uint8* src_bgra,
>-                         int src_stride_bgra,
>-                         uint8* dst_u,
>-                         uint8* dst_v,
>-                         int width);
>-void ABGRToUVRow_Any_MSA(const uint8* src_abgr,
>-                         int src_stride_abgr,
>-                         uint8* dst_u,
>-                         uint8* dst_v,
>-                         int width);
>-void RGBAToUVRow_Any_MSA(const uint8* src_rgba,
>-                         int src_stride_rgba,
>-                         uint8* dst_u,
>-                         uint8* dst_v,
>-                         int width);
>-void RGB24ToUVRow_Any_MSA(const uint8* src_rgb24,
>-                          int src_stride_rgb24,
>-                          uint8* dst_u,
>-                          uint8* dst_v,
>+void BGRAToUVRow_Any_MSA(const uint8_t* src_ptr,
>+                         int src_stride_ptr,
>+                         uint8_t* dst_u,
>+                         uint8_t* dst_v,
>+                         int width);
>+void ABGRToUVRow_Any_MSA(const uint8_t* src_ptr,
>+                         int src_stride_ptr,
>+                         uint8_t* dst_u,
>+                         uint8_t* dst_v,
>+                         int width);
>+void RGBAToUVRow_Any_MSA(const uint8_t* src_ptr,
>+                         int src_stride_ptr,
>+                         uint8_t* dst_u,
>+                         uint8_t* dst_v,
>+                         int width);
>+void RGB24ToUVRow_Any_MSA(const uint8_t* src_ptr,
>+                          int src_stride_ptr,
>+                          uint8_t* dst_u,
>+                          uint8_t* dst_v,
>                           int width);
>-void RAWToUVRow_Any_MSA(const uint8* src_raw,
>-                        int src_stride_raw,
>-                        uint8* dst_u,
>-                        uint8* dst_v,
>+void RAWToUVRow_Any_MSA(const uint8_t* src_ptr,
>+                        int src_stride_ptr,
>+                        uint8_t* dst_u,
>+                        uint8_t* dst_v,
>                         int width);
>-void RGB565ToUVRow_Any_MSA(const uint8* src_rgb565,
>-                           int src_stride_rgb565,
>-                           uint8* dst_u,
>-                           uint8* dst_v,
>+void RGB565ToUVRow_Any_MSA(const uint8_t* src_ptr,
>+                           int src_stride_ptr,
>+                           uint8_t* dst_u,
>+                           uint8_t* dst_v,
>                            int width);
>-void ARGB1555ToUVRow_Any_MSA(const uint8* src_argb1555,
>-                             int src_stride_argb1555,
>-                             uint8* dst_u,
>-                             uint8* dst_v,
>+void ARGB1555ToUVRow_Any_MSA(const uint8_t* src_ptr,
>+                             int src_stride_ptr,
>+                             uint8_t* dst_u,
>+                             uint8_t* dst_v,
>                              int width);
>-void BGRAToUVRow_Any_DSPR2(const uint8* src_bgra,
>-                           int src_stride_bgra,
>-                           uint8* dst_u,
>-                           uint8* dst_v,
>-                           int width);
>-void ABGRToUVRow_Any_DSPR2(const uint8* src_abgr,
>-                           int src_stride_abgr,
>-                           uint8* dst_u,
>-                           uint8* dst_v,
>-                           int width);
>-void RGBAToUVRow_Any_DSPR2(const uint8* src_rgba,
>-                           int src_stride_rgba,
>-                           uint8* dst_u,
>-                           uint8* dst_v,
>-                           int width);
>-void ARGBToUVRow_Any_DSPR2(const uint8* src_argb,
>-                           int src_stride_argb,
>-                           uint8* dst_u,
>-                           uint8* dst_v,
>-                           int width);
>-void ARGBToUVRow_C(const uint8* src_argb,
>-                   int src_stride_argb,
>-                   uint8* dst_u,
>-                   uint8* dst_v,
>+void ARGBToUVRow_C(const uint8_t* src_rgb0,
>+                   int src_stride_rgb,
>+                   uint8_t* dst_u,
>+                   uint8_t* dst_v,
>                    int width);
>-void ARGBToUVJRow_C(const uint8* src_argb,
>-                    int src_stride_argb,
>-                    uint8* dst_u,
>-                    uint8* dst_v,
>+void ARGBToUVJRow_C(const uint8_t* src_rgb0,
>+                    int src_stride_rgb,
>+                    uint8_t* dst_u,
>+                    uint8_t* dst_v,
>                     int width);
>-void ARGBToUVRow_C(const uint8* src_argb,
>-                   int src_stride_argb,
>-                   uint8* dst_u,
>-                   uint8* dst_v,
>+void ARGBToUVRow_C(const uint8_t* src_rgb0,
>+                   int src_stride_rgb,
>+                   uint8_t* dst_u,
>+                   uint8_t* dst_v,
>                    int width);
>-void ARGBToUVJRow_C(const uint8* src_argb,
>-                    int src_stride_argb,
>-                    uint8* dst_u,
>-                    uint8* dst_v,
>+void ARGBToUVJRow_C(const uint8_t* src_rgb0,
>+                    int src_stride_rgb,
>+                    uint8_t* dst_u,
>+                    uint8_t* dst_v,
>                     int width);
>-void BGRAToUVRow_C(const uint8* src_bgra,
>-                   int src_stride_bgra,
>-                   uint8* dst_u,
>-                   uint8* dst_v,
>+void BGRAToUVRow_C(const uint8_t* src_rgb0,
>+                   int src_stride_rgb,
>+                   uint8_t* dst_u,
>+                   uint8_t* dst_v,
>                    int width);
>-void ABGRToUVRow_C(const uint8* src_abgr,
>-                   int src_stride_abgr,
>-                   uint8* dst_u,
>-                   uint8* dst_v,
>+void ABGRToUVRow_C(const uint8_t* src_rgb0,
>+                   int src_stride_rgb,
>+                   uint8_t* dst_u,
>+                   uint8_t* dst_v,
>                    int width);
>-void RGBAToUVRow_C(const uint8* src_rgba,
>-                   int src_stride_rgba,
>-                   uint8* dst_u,
>-                   uint8* dst_v,
>+void RGBAToUVRow_C(const uint8_t* src_rgb0,
>+                   int src_stride_rgb,
>+                   uint8_t* dst_u,
>+                   uint8_t* dst_v,
>                    int width);
>-void RGB24ToUVRow_C(const uint8* src_rgb24,
>-                    int src_stride_rgb24,
>-                    uint8* dst_u,
>-                    uint8* dst_v,
>+void RGB24ToUVRow_C(const uint8_t* src_rgb0,
>+                    int src_stride_rgb,
>+                    uint8_t* dst_u,
>+                    uint8_t* dst_v,
>                     int width);
>-void RAWToUVRow_C(const uint8* src_raw,
>-                  int src_stride_raw,
>-                  uint8* dst_u,
>-                  uint8* dst_v,
>+void RAWToUVRow_C(const uint8_t* src_rgb0,
>+                  int src_stride_rgb,
>+                  uint8_t* dst_u,
>+                  uint8_t* dst_v,
>                   int width);
>-void RGB565ToUVRow_C(const uint8* src_rgb565,
>+void RGB565ToUVRow_C(const uint8_t* src_rgb565,
>                      int src_stride_rgb565,
>-                     uint8* dst_u,
>-                     uint8* dst_v,
>+                     uint8_t* dst_u,
>+                     uint8_t* dst_v,
>                      int width);
>-void ARGB1555ToUVRow_C(const uint8* src_argb1555,
>+void ARGB1555ToUVRow_C(const uint8_t* src_argb1555,
>                        int src_stride_argb1555,
>-                       uint8* dst_u,
>-                       uint8* dst_v,
>+                       uint8_t* dst_u,
>+                       uint8_t* dst_v,
>                        int width);
>-void ARGB4444ToUVRow_C(const uint8* src_argb4444,
>+void ARGB4444ToUVRow_C(const uint8_t* src_argb4444,
>                        int src_stride_argb4444,
>-                       uint8* dst_u,
>-                       uint8* dst_v,
>+                       uint8_t* dst_u,
>+                       uint8_t* dst_v,
>                        int width);
> 
>-void ARGBToUV444Row_SSSE3(const uint8* src_argb,
>-                          uint8* dst_u,
>-                          uint8* dst_v,
>+void ARGBToUV444Row_SSSE3(const uint8_t* src_argb,
>+                          uint8_t* dst_u,
>+                          uint8_t* dst_v,
>                           int width);
>-void ARGBToUV444Row_Any_SSSE3(const uint8* src_argb,
>-                              uint8* dst_u,
>-                              uint8* dst_v,
>+void ARGBToUV444Row_Any_SSSE3(const uint8_t* src_ptr,
>+                              uint8_t* dst_u,
>+                              uint8_t* dst_v,
>                               int width);
> 
>-void ARGBToUV444Row_C(const uint8* src_argb,
>-                      uint8* dst_u,
>-                      uint8* dst_v,
>+void ARGBToUV444Row_C(const uint8_t* src_argb,
>+                      uint8_t* dst_u,
>+                      uint8_t* dst_v,
>                       int width);
> 
>-void MirrorRow_AVX2(const uint8* src, uint8* dst, int width);
>-void MirrorRow_SSSE3(const uint8* src, uint8* dst, int width);
>-void MirrorRow_NEON(const uint8* src, uint8* dst, int width);
>-void MirrorRow_DSPR2(const uint8* src, uint8* dst, int width);
>-void MirrorRow_MSA(const uint8* src, uint8* dst, int width);
>-void MirrorRow_C(const uint8* src, uint8* dst, int width);
>-void MirrorRow_Any_AVX2(const uint8* src, uint8* dst, int width);
>-void MirrorRow_Any_SSSE3(const uint8* src, uint8* dst, int width);
>-void MirrorRow_Any_SSE2(const uint8* src, uint8* dst, int width);
>-void MirrorRow_Any_NEON(const uint8* src, uint8* dst, int width);
>-void MirrorRow_Any_MSA(const uint8* src, uint8* dst, int width);
>-
>-void MirrorUVRow_SSSE3(const uint8* src_uv,
>-                       uint8* dst_u,
>-                       uint8* dst_v,
>+void MirrorRow_AVX2(const uint8_t* src, uint8_t* dst, int width);
>+void MirrorRow_SSSE3(const uint8_t* src, uint8_t* dst, int width);
>+void MirrorRow_NEON(const uint8_t* src, uint8_t* dst, int width);
>+void MirrorRow_MSA(const uint8_t* src, uint8_t* dst, int width);
>+void MirrorRow_C(const uint8_t* src, uint8_t* dst, int width);
>+void MirrorRow_Any_AVX2(const uint8_t* src_ptr, uint8_t* dst_ptr, int width);
>+void MirrorRow_Any_SSSE3(const uint8_t* src_ptr, uint8_t* dst_ptr, int width);
>+void MirrorRow_Any_SSE2(const uint8_t* src, uint8_t* dst, int width);
>+void MirrorRow_Any_NEON(const uint8_t* src_ptr, uint8_t* dst_ptr, int width);
>+void MirrorRow_Any_MSA(const uint8_t* src_ptr, uint8_t* dst_ptr, int width);
>+
>+void MirrorUVRow_SSSE3(const uint8_t* src,
>+                       uint8_t* dst_u,
>+                       uint8_t* dst_v,
>                        int width);
>-void MirrorUVRow_NEON(const uint8* src_uv,
>-                      uint8* dst_u,
>-                      uint8* dst_v,
>+void MirrorUVRow_NEON(const uint8_t* src_uv,
>+                      uint8_t* dst_u,
>+                      uint8_t* dst_v,
>                       int width);
>-void MirrorUVRow_DSPR2(const uint8* src_uv,
>-                       uint8* dst_u,
>-                       uint8* dst_v,
>-                       int width);
>-void MirrorUVRow_MSA(const uint8* src_uv,
>-                     uint8* dst_u,
>-                     uint8* dst_v,
>+void MirrorUVRow_MSA(const uint8_t* src_uv,
>+                     uint8_t* dst_u,
>+                     uint8_t* dst_v,
>                      int width);
>-void MirrorUVRow_C(const uint8* src_uv, uint8* dst_u, uint8* dst_v, int width);
>-
>-void ARGBMirrorRow_AVX2(const uint8* src, uint8* dst, int width);
>-void ARGBMirrorRow_SSE2(const uint8* src, uint8* dst, int width);
>-void ARGBMirrorRow_NEON(const uint8* src, uint8* dst, int width);
>-void ARGBMirrorRow_MSA(const uint8* src, uint8* dst, int width);
>-void ARGBMirrorRow_C(const uint8* src, uint8* dst, int width);
>-void ARGBMirrorRow_Any_AVX2(const uint8* src, uint8* dst, int width);
>-void ARGBMirrorRow_Any_SSE2(const uint8* src, uint8* dst, int width);
>-void ARGBMirrorRow_Any_NEON(const uint8* src, uint8* dst, int width);
>-void ARGBMirrorRow_Any_MSA(const uint8* src, uint8* dst, int width);
>-
>-void SplitUVRow_C(const uint8* src_uv, uint8* dst_u, uint8* dst_v, int width);
>-void SplitUVRow_SSE2(const uint8* src_uv,
>-                     uint8* dst_u,
>-                     uint8* dst_v,
>+void MirrorUVRow_C(const uint8_t* src_uv,
>+                   uint8_t* dst_u,
>+                   uint8_t* dst_v,
>+                   int width);
>+
>+void ARGBMirrorRow_AVX2(const uint8_t* src, uint8_t* dst, int width);
>+void ARGBMirrorRow_SSE2(const uint8_t* src, uint8_t* dst, int width);
>+void ARGBMirrorRow_NEON(const uint8_t* src, uint8_t* dst, int width);
>+void ARGBMirrorRow_MSA(const uint8_t* src, uint8_t* dst, int width);
>+void ARGBMirrorRow_C(const uint8_t* src, uint8_t* dst, int width);
>+void ARGBMirrorRow_Any_AVX2(const uint8_t* src_ptr,
>+                            uint8_t* dst_ptr,
>+                            int width);
>+void ARGBMirrorRow_Any_SSE2(const uint8_t* src_ptr,
>+                            uint8_t* dst_ptr,
>+                            int width);
>+void ARGBMirrorRow_Any_NEON(const uint8_t* src_ptr,
>+                            uint8_t* dst_ptr,
>+                            int width);
>+void ARGBMirrorRow_Any_MSA(const uint8_t* src_ptr, uint8_t* dst_ptr, int width);
>+
>+void SplitUVRow_C(const uint8_t* src_uv,
>+                  uint8_t* dst_u,
>+                  uint8_t* dst_v,
>+                  int width);
>+void SplitUVRow_SSE2(const uint8_t* src_uv,
>+                     uint8_t* dst_u,
>+                     uint8_t* dst_v,
>                      int width);
>-void SplitUVRow_AVX2(const uint8* src_uv,
>-                     uint8* dst_u,
>-                     uint8* dst_v,
>+void SplitUVRow_AVX2(const uint8_t* src_uv,
>+                     uint8_t* dst_u,
>+                     uint8_t* dst_v,
>                      int width);
>-void SplitUVRow_NEON(const uint8* src_uv,
>-                     uint8* dst_u,
>-                     uint8* dst_v,
>+void SplitUVRow_NEON(const uint8_t* src_uv,
>+                     uint8_t* dst_u,
>+                     uint8_t* dst_v,
>                      int width);
>-void SplitUVRow_DSPR2(const uint8* src_uv,
>-                      uint8* dst_u,
>-                      uint8* dst_v,
>-                      int width);
>-void SplitUVRow_MSA(const uint8* src_uv, uint8* dst_u, uint8* dst_v, int width);
>-void SplitUVRow_Any_SSE2(const uint8* src_uv,
>-                         uint8* dst_u,
>-                         uint8* dst_v,
>-                         int width);
>-void SplitUVRow_Any_AVX2(const uint8* src_uv,
>-                         uint8* dst_u,
>-                         uint8* dst_v,
>-                         int width);
>-void SplitUVRow_Any_NEON(const uint8* src_uv,
>-                         uint8* dst_u,
>-                         uint8* dst_v,
>-                         int width);
>-void SplitUVRow_Any_DSPR2(const uint8* src_uv,
>-                          uint8* dst_u,
>-                          uint8* dst_v,
>-                          int width);
>-void SplitUVRow_Any_MSA(const uint8* src_uv,
>-                        uint8* dst_u,
>-                        uint8* dst_v,
>+void SplitUVRow_MSA(const uint8_t* src_uv,
>+                    uint8_t* dst_u,
>+                    uint8_t* dst_v,
>+                    int width);
>+void SplitUVRow_Any_SSE2(const uint8_t* src_ptr,
>+                         uint8_t* dst_u,
>+                         uint8_t* dst_v,
>+                         int width);
>+void SplitUVRow_Any_AVX2(const uint8_t* src_ptr,
>+                         uint8_t* dst_u,
>+                         uint8_t* dst_v,
>+                         int width);
>+void SplitUVRow_Any_NEON(const uint8_t* src_ptr,
>+                         uint8_t* dst_u,
>+                         uint8_t* dst_v,
>+                         int width);
>+void SplitUVRow_Any_MSA(const uint8_t* src_ptr,
>+                        uint8_t* dst_u,
>+                        uint8_t* dst_v,
>                         int width);
> 
>-void MergeUVRow_C(const uint8* src_u,
>-                  const uint8* src_v,
>-                  uint8* dst_uv,
>+void MergeUVRow_C(const uint8_t* src_u,
>+                  const uint8_t* src_v,
>+                  uint8_t* dst_uv,
>                   int width);
>-void MergeUVRow_SSE2(const uint8* src_u,
>-                     const uint8* src_v,
>-                     uint8* dst_uv,
>+void MergeUVRow_SSE2(const uint8_t* src_u,
>+                     const uint8_t* src_v,
>+                     uint8_t* dst_uv,
>                      int width);
>-void MergeUVRow_AVX2(const uint8* src_u,
>-                     const uint8* src_v,
>-                     uint8* dst_uv,
>+void MergeUVRow_AVX2(const uint8_t* src_u,
>+                     const uint8_t* src_v,
>+                     uint8_t* dst_uv,
>                      int width);
>-void MergeUVRow_NEON(const uint8* src_u,
>-                     const uint8* src_v,
>-                     uint8* dst_uv,
>+void MergeUVRow_NEON(const uint8_t* src_u,
>+                     const uint8_t* src_v,
>+                     uint8_t* dst_uv,
>                      int width);
>-void MergeUVRow_MSA(const uint8* src_u,
>-                    const uint8* src_v,
>-                    uint8* dst_uv,
>+void MergeUVRow_MSA(const uint8_t* src_u,
>+                    const uint8_t* src_v,
>+                    uint8_t* dst_uv,
>                     int width);
>-void MergeUVRow_Any_SSE2(const uint8* src_u,
>-                         const uint8* src_v,
>-                         uint8* dst_uv,
>+void MergeUVRow_Any_SSE2(const uint8_t* y_buf,
>+                         const uint8_t* uv_buf,
>+                         uint8_t* dst_ptr,
>                          int width);
>-void MergeUVRow_Any_AVX2(const uint8* src_u,
>-                         const uint8* src_v,
>-                         uint8* dst_uv,
>+void MergeUVRow_Any_AVX2(const uint8_t* y_buf,
>+                         const uint8_t* uv_buf,
>+                         uint8_t* dst_ptr,
>                          int width);
>-void MergeUVRow_Any_NEON(const uint8* src_u,
>-                         const uint8* src_v,
>-                         uint8* dst_uv,
>+void MergeUVRow_Any_NEON(const uint8_t* y_buf,
>+                         const uint8_t* uv_buf,
>+                         uint8_t* dst_ptr,
>                          int width);
>-void MergeUVRow_Any_MSA(const uint8* src_u,
>-                        const uint8* src_v,
>-                        uint8* dst_uv,
>+void MergeUVRow_Any_MSA(const uint8_t* y_buf,
>+                        const uint8_t* uv_buf,
>+                        uint8_t* dst_ptr,
>                         int width);
> 
>-void SplitRGBRow_C(const uint8* src_rgb,
>-                   uint8* dst_r,
>-                   uint8* dst_g,
>-                   uint8* dst_b,
>+void SplitRGBRow_C(const uint8_t* src_rgb,
>+                   uint8_t* dst_r,
>+                   uint8_t* dst_g,
>+                   uint8_t* dst_b,
>                    int width);
>-void SplitRGBRow_SSSE3(const uint8* src_rgb,
>-                       uint8* dst_r,
>-                       uint8* dst_g,
>-                       uint8* dst_b,
>+void SplitRGBRow_SSSE3(const uint8_t* src_rgb,
>+                       uint8_t* dst_r,
>+                       uint8_t* dst_g,
>+                       uint8_t* dst_b,
>                        int width);
>-void SplitRGBRow_NEON(const uint8* src_rgb,
>-                      uint8* dst_r,
>-                      uint8* dst_g,
>-                      uint8* dst_b,
>+void SplitRGBRow_NEON(const uint8_t* src_rgb,
>+                      uint8_t* dst_r,
>+                      uint8_t* dst_g,
>+                      uint8_t* dst_b,
>                       int width);
>-void SplitRGBRow_Any_SSSE3(const uint8* src_rgb,
>-                           uint8* dst_r,
>-                           uint8* dst_g,
>-                           uint8* dst_b,
>+void SplitRGBRow_Any_SSSE3(const uint8_t* src_ptr,
>+                           uint8_t* dst_r,
>+                           uint8_t* dst_g,
>+                           uint8_t* dst_b,
>                            int width);
>-void SplitRGBRow_Any_NEON(const uint8* src_rgb,
>-                          uint8* dst_r,
>-                          uint8* dst_g,
>-                          uint8* dst_b,
>+void SplitRGBRow_Any_NEON(const uint8_t* src_ptr,
>+                          uint8_t* dst_r,
>+                          uint8_t* dst_g,
>+                          uint8_t* dst_b,
>                           int width);
> 
>-void MergeRGBRow_C(const uint8* src_r,
>-                   const uint8* src_g,
>-                   const uint8* src_b,
>-                   uint8* dst_rgb,
>+void MergeRGBRow_C(const uint8_t* src_r,
>+                   const uint8_t* src_g,
>+                   const uint8_t* src_b,
>+                   uint8_t* dst_rgb,
>                    int width);
>-void MergeRGBRow_SSSE3(const uint8* src_r,
>-                       const uint8* src_g,
>-                       const uint8* src_b,
>-                       uint8* dst_rgb,
>+void MergeRGBRow_SSSE3(const uint8_t* src_r,
>+                       const uint8_t* src_g,
>+                       const uint8_t* src_b,
>+                       uint8_t* dst_rgb,
>                        int width);
>-void MergeRGBRow_NEON(const uint8* src_r,
>-                      const uint8* src_g,
>-                      const uint8* src_b,
>-                      uint8* dst_rgb,
>+void MergeRGBRow_NEON(const uint8_t* src_r,
>+                      const uint8_t* src_g,
>+                      const uint8_t* src_b,
>+                      uint8_t* dst_rgb,
>                       int width);
>-void MergeRGBRow_Any_SSSE3(const uint8* src_r,
>-                           const uint8* src_g,
>-                           const uint8* src_b,
>-                           uint8* dst_rgb,
>+void MergeRGBRow_Any_SSSE3(const uint8_t* y_buf,
>+                           const uint8_t* u_buf,
>+                           const uint8_t* v_buf,
>+                           uint8_t* dst_ptr,
>                            int width);
>-void MergeRGBRow_Any_NEON(const uint8* src_r,
>-                          const uint8* src_g,
>-                          const uint8* src_b,
>-                          uint8* dst_rgb,
>+void MergeRGBRow_Any_NEON(const uint8_t* src_r,
>+                          const uint8_t* src_g,
>+                          const uint8_t* src_b,
>+                          uint8_t* dst_rgb,
>                           int width);
> 
>-void MergeUVRow_16_C(const uint16* src_u,
>-                     const uint16* src_v,
>-                     uint16* dst_uv,
>+void MergeUVRow_16_C(const uint16_t* src_u,
>+                     const uint16_t* src_v,
>+                     uint16_t* dst_uv,
>                      int scale, /* 64 for 10 bit */
>                      int width);
>-void MergeUVRow_16_AVX2(const uint16* src_u,
>-                        const uint16* src_v,
>-                        uint16* dst_uv,
>+void MergeUVRow_16_AVX2(const uint16_t* src_u,
>+                        const uint16_t* src_v,
>+                        uint16_t* dst_uv,
>                         int scale,
>                         int width);
> 
>-void MultiplyRow_16_AVX2(const uint16* src_y,
>-                         uint16* dst_y,
>+void MultiplyRow_16_AVX2(const uint16_t* src_y,
>+                         uint16_t* dst_y,
>                          int scale,
>                          int width);
>-void MultiplyRow_16_C(const uint16* src_y, uint16* dst_y, int scale, int width);
>-
>-void CopyRow_SSE2(const uint8* src, uint8* dst, int count);
>-void CopyRow_AVX(const uint8* src, uint8* dst, int count);
>-void CopyRow_ERMS(const uint8* src, uint8* dst, int count);
>-void CopyRow_NEON(const uint8* src, uint8* dst, int count);
>-void CopyRow_MIPS(const uint8* src, uint8* dst, int count);
>-void CopyRow_C(const uint8* src, uint8* dst, int count);
>-void CopyRow_Any_SSE2(const uint8* src, uint8* dst, int count);
>-void CopyRow_Any_AVX(const uint8* src, uint8* dst, int count);
>-void CopyRow_Any_NEON(const uint8* src, uint8* dst, int count);
>-
>-void CopyRow_16_C(const uint16* src, uint16* dst, int count);
>-
>-void ARGBCopyAlphaRow_C(const uint8* src_argb, uint8* dst_argb, int width);
>-void ARGBCopyAlphaRow_SSE2(const uint8* src_argb, uint8* dst_argb, int width);
>-void ARGBCopyAlphaRow_AVX2(const uint8* src_argb, uint8* dst_argb, int width);
>-void ARGBCopyAlphaRow_Any_SSE2(const uint8* src_argb,
>-                               uint8* dst_argb,
>+void MultiplyRow_16_C(const uint16_t* src_y,
>+                      uint16_t* dst_y,
>+                      int scale,
>+                      int width);
>+
>+void Convert8To16Row_C(const uint8_t* src_y,
>+                       uint16_t* dst_y,
>+                       int scale,
>+                       int width);
>+void Convert8To16Row_SSE2(const uint8_t* src_y,
>+                          uint16_t* dst_y,
>+                          int scale,
>+                          int width);
>+void Convert8To16Row_AVX2(const uint8_t* src_y,
>+                          uint16_t* dst_y,
>+                          int scale,
>+                          int width);
>+void Convert8To16Row_Any_SSE2(const uint8_t* src_ptr,
>+                              uint16_t* dst_ptr,
>+                              int scale,
>+                              int width);
>+void Convert8To16Row_Any_AVX2(const uint8_t* src_ptr,
>+                              uint16_t* dst_ptr,
>+                              int scale,
>+                              int width);
>+
>+void Convert16To8Row_C(const uint16_t* src_y,
>+                       uint8_t* dst_y,
>+                       int scale,
>+                       int width);
>+void Convert16To8Row_SSSE3(const uint16_t* src_y,
>+                           uint8_t* dst_y,
>+                           int scale,
>+                           int width);
>+void Convert16To8Row_AVX2(const uint16_t* src_y,
>+                          uint8_t* dst_y,
>+                          int scale,
>+                          int width);
>+void Convert16To8Row_Any_SSSE3(const uint16_t* src_ptr,
>+                               uint8_t* dst_ptr,
>+                               int scale,
>                                int width);
>-void ARGBCopyAlphaRow_Any_AVX2(const uint8* src_argb,
>-                               uint8* dst_argb,
>+void Convert16To8Row_Any_AVX2(const uint16_t* src_ptr,
>+                              uint8_t* dst_ptr,
>+                              int scale,
>+                              int width);
>+
>+void CopyRow_SSE2(const uint8_t* src, uint8_t* dst, int width);
>+void CopyRow_AVX(const uint8_t* src, uint8_t* dst, int width);
>+void CopyRow_ERMS(const uint8_t* src, uint8_t* dst, int width);
>+void CopyRow_NEON(const uint8_t* src, uint8_t* dst, int width);
>+void CopyRow_MIPS(const uint8_t* src, uint8_t* dst, int count);
>+void CopyRow_C(const uint8_t* src, uint8_t* dst, int count);
>+void CopyRow_Any_SSE2(const uint8_t* src_ptr, uint8_t* dst_ptr, int width);
>+void CopyRow_Any_AVX(const uint8_t* src_ptr, uint8_t* dst_ptr, int width);
>+void CopyRow_Any_NEON(const uint8_t* src_ptr, uint8_t* dst_ptr, int width);
>+
>+void CopyRow_16_C(const uint16_t* src, uint16_t* dst, int count);
>+
>+void ARGBCopyAlphaRow_C(const uint8_t* src, uint8_t* dst, int width);
>+void ARGBCopyAlphaRow_SSE2(const uint8_t* src, uint8_t* dst, int width);
>+void ARGBCopyAlphaRow_AVX2(const uint8_t* src, uint8_t* dst, int width);
>+void ARGBCopyAlphaRow_Any_SSE2(const uint8_t* src_ptr,
>+                               uint8_t* dst_ptr,
>+                               int width);
>+void ARGBCopyAlphaRow_Any_AVX2(const uint8_t* src_ptr,
>+                               uint8_t* dst_ptr,
>                                int width);
> 
>-void ARGBExtractAlphaRow_C(const uint8* src_argb, uint8* dst_a, int width);
>-void ARGBExtractAlphaRow_SSE2(const uint8* src_argb, uint8* dst_a, int width);
>-void ARGBExtractAlphaRow_AVX2(const uint8* src_argb, uint8* dst_a, int width);
>-void ARGBExtractAlphaRow_NEON(const uint8* src_argb, uint8* dst_a, int width);
>-void ARGBExtractAlphaRow_MSA(const uint8* src_argb, uint8* dst_a, int width);
>-void ARGBExtractAlphaRow_Any_SSE2(const uint8* src_argb,
>-                                  uint8* dst_a,
>+void ARGBExtractAlphaRow_C(const uint8_t* src_argb, uint8_t* dst_a, int width);
>+void ARGBExtractAlphaRow_SSE2(const uint8_t* src_argb,
>+                              uint8_t* dst_a,
>+                              int width);
>+void ARGBExtractAlphaRow_AVX2(const uint8_t* src_argb,
>+                              uint8_t* dst_a,
>+                              int width);
>+void ARGBExtractAlphaRow_NEON(const uint8_t* src_argb,
>+                              uint8_t* dst_a,
>+                              int width);
>+void ARGBExtractAlphaRow_MSA(const uint8_t* src_argb,
>+                             uint8_t* dst_a,
>+                             int width);
>+void ARGBExtractAlphaRow_Any_SSE2(const uint8_t* src_ptr,
>+                                  uint8_t* dst_ptr,
>                                   int width);
>-void ARGBExtractAlphaRow_Any_AVX2(const uint8* src_argb,
>-                                  uint8* dst_a,
>+void ARGBExtractAlphaRow_Any_AVX2(const uint8_t* src_ptr,
>+                                  uint8_t* dst_ptr,
>                                   int width);
>-void ARGBExtractAlphaRow_Any_NEON(const uint8* src_argb,
>-                                  uint8* dst_a,
>+void ARGBExtractAlphaRow_Any_NEON(const uint8_t* src_ptr,
>+                                  uint8_t* dst_ptr,
>                                   int width);
>-void ARGBExtractAlphaRow_Any_MSA(const uint8* src_argb,
>-                                 uint8* dst_a,
>+void ARGBExtractAlphaRow_Any_MSA(const uint8_t* src_ptr,
>+                                 uint8_t* dst_ptr,
>                                  int width);
> 
>-void ARGBCopyYToAlphaRow_C(const uint8* src_y, uint8* dst_argb, int width);
>-void ARGBCopyYToAlphaRow_SSE2(const uint8* src_y, uint8* dst_argb, int width);
>-void ARGBCopyYToAlphaRow_AVX2(const uint8* src_y, uint8* dst_argb, int width);
>-void ARGBCopyYToAlphaRow_Any_SSE2(const uint8* src_y,
>-                                  uint8* dst_argb,
>+void ARGBCopyYToAlphaRow_C(const uint8_t* src, uint8_t* dst, int width);
>+void ARGBCopyYToAlphaRow_SSE2(const uint8_t* src, uint8_t* dst, int width);
>+void ARGBCopyYToAlphaRow_AVX2(const uint8_t* src, uint8_t* dst, int width);
>+void ARGBCopyYToAlphaRow_Any_SSE2(const uint8_t* src_ptr,
>+                                  uint8_t* dst_ptr,
>                                   int width);
>-void ARGBCopyYToAlphaRow_Any_AVX2(const uint8* src_y,
>-                                  uint8* dst_argb,
>+void ARGBCopyYToAlphaRow_Any_AVX2(const uint8_t* src_ptr,
>+                                  uint8_t* dst_ptr,
>                                   int width);
> 
>-void SetRow_C(uint8* dst, uint8 v8, int count);
>-void SetRow_MSA(uint8* dst, uint8 v8, int count);
>-void SetRow_X86(uint8* dst, uint8 v8, int count);
>-void SetRow_ERMS(uint8* dst, uint8 v8, int count);
>-void SetRow_NEON(uint8* dst, uint8 v8, int count);
>-void SetRow_Any_X86(uint8* dst, uint8 v8, int count);
>-void SetRow_Any_NEON(uint8* dst, uint8 v8, int count);
>-
>-void ARGBSetRow_C(uint8* dst_argb, uint32 v32, int count);
>-void ARGBSetRow_X86(uint8* dst_argb, uint32 v32, int count);
>-void ARGBSetRow_NEON(uint8* dst_argb, uint32 v32, int count);
>-void ARGBSetRow_Any_NEON(uint8* dst_argb, uint32 v32, int count);
>-void ARGBSetRow_MSA(uint8* dst_argb, uint32 v32, int count);
>-void ARGBSetRow_Any_MSA(uint8* dst_argb, uint32 v32, int count);
>+void SetRow_C(uint8_t* dst, uint8_t v8, int width);
>+void SetRow_MSA(uint8_t* dst, uint8_t v8, int width);
>+void SetRow_X86(uint8_t* dst, uint8_t v8, int width);
>+void SetRow_ERMS(uint8_t* dst, uint8_t v8, int width);
>+void SetRow_NEON(uint8_t* dst, uint8_t v8, int width);
>+void SetRow_Any_X86(uint8_t* dst_ptr, uint8_t v32, int width);
>+void SetRow_Any_NEON(uint8_t* dst_ptr, uint8_t v32, int width);
>+
>+void ARGBSetRow_C(uint8_t* dst_argb, uint32_t v32, int width);
>+void ARGBSetRow_X86(uint8_t* dst_argb, uint32_t v32, int width);
>+void ARGBSetRow_NEON(uint8_t* dst, uint32_t v32, int width);
>+void ARGBSetRow_Any_NEON(uint8_t* dst_ptr, uint32_t v32, int width);
>+void ARGBSetRow_MSA(uint8_t* dst_argb, uint32_t v32, int width);
>+void ARGBSetRow_Any_MSA(uint8_t* dst_ptr, uint32_t v32, int width);
> 
> // ARGBShufflers for BGRAToARGB etc.
>-void ARGBShuffleRow_C(const uint8* src_argb,
>-                      uint8* dst_argb,
>-                      const uint8* shuffler,
>+void ARGBShuffleRow_C(const uint8_t* src_argb,
>+                      uint8_t* dst_argb,
>+                      const uint8_t* shuffler,
>                       int width);
>-void ARGBShuffleRow_SSE2(const uint8* src_argb,
>-                         uint8* dst_argb,
>-                         const uint8* shuffler,
>-                         int width);
>-void ARGBShuffleRow_SSSE3(const uint8* src_argb,
>-                          uint8* dst_argb,
>-                          const uint8* shuffler,
>+void ARGBShuffleRow_SSSE3(const uint8_t* src_argb,
>+                          uint8_t* dst_argb,
>+                          const uint8_t* shuffler,
>                           int width);
>-void ARGBShuffleRow_AVX2(const uint8* src_argb,
>-                         uint8* dst_argb,
>-                         const uint8* shuffler,
>+void ARGBShuffleRow_AVX2(const uint8_t* src_argb,
>+                         uint8_t* dst_argb,
>+                         const uint8_t* shuffler,
>                          int width);
>-void ARGBShuffleRow_NEON(const uint8* src_argb,
>-                         uint8* dst_argb,
>-                         const uint8* shuffler,
>+void ARGBShuffleRow_NEON(const uint8_t* src_argb,
>+                         uint8_t* dst_argb,
>+                         const uint8_t* shuffler,
>                          int width);
>-void ARGBShuffleRow_MSA(const uint8* src_argb,
>-                        uint8* dst_argb,
>-                        const uint8* shuffler,
>+void ARGBShuffleRow_MSA(const uint8_t* src_argb,
>+                        uint8_t* dst_argb,
>+                        const uint8_t* shuffler,
>                         int width);
>-void ARGBShuffleRow_Any_SSE2(const uint8* src_argb,
>-                             uint8* dst_argb,
>-                             const uint8* shuffler,
>-                             int width);
>-void ARGBShuffleRow_Any_SSSE3(const uint8* src_argb,
>-                              uint8* dst_argb,
>-                              const uint8* shuffler,
>+void ARGBShuffleRow_Any_SSSE3(const uint8_t* src_ptr,
>+                              uint8_t* dst_ptr,
>+                              const uint8_t* param,
>                               int width);
>-void ARGBShuffleRow_Any_AVX2(const uint8* src_argb,
>-                             uint8* dst_argb,
>-                             const uint8* shuffler,
>+void ARGBShuffleRow_Any_AVX2(const uint8_t* src_ptr,
>+                             uint8_t* dst_ptr,
>+                             const uint8_t* param,
>                              int width);
>-void ARGBShuffleRow_Any_NEON(const uint8* src_argb,
>-                             uint8* dst_argb,
>-                             const uint8* shuffler,
>+void ARGBShuffleRow_Any_NEON(const uint8_t* src_ptr,
>+                             uint8_t* dst_ptr,
>+                             const uint8_t* param,
>                              int width);
>-void ARGBShuffleRow_Any_MSA(const uint8* src_argb,
>-                            uint8* dst_argb,
>-                            const uint8* shuffler,
>-                            int width);
>-
>-void RGB24ToARGBRow_SSSE3(const uint8* src_rgb24, uint8* dst_argb, int width);
>-void RAWToARGBRow_SSSE3(const uint8* src_raw, uint8* dst_argb, int width);
>-void RAWToRGB24Row_SSSE3(const uint8* src_raw, uint8* dst_rgb24, int width);
>-void RGB565ToARGBRow_SSE2(const uint8* src_rgb565, uint8* dst_argb, int width);
>-void ARGB1555ToARGBRow_SSE2(const uint8* src_argb1555,
>-                            uint8* dst_argb,
>-                            int width);
>-void ARGB4444ToARGBRow_SSE2(const uint8* src_argb4444,
>-                            uint8* dst_argb,
>-                            int width);
>-void RGB565ToARGBRow_AVX2(const uint8* src_rgb565, uint8* dst_argb, int width);
>-void ARGB1555ToARGBRow_AVX2(const uint8* src_argb1555,
>-                            uint8* dst_argb,
>-                            int width);
>-void ARGB4444ToARGBRow_AVX2(const uint8* src_argb4444,
>-                            uint8* dst_argb,
>-                            int width);
>-
>-void RGB24ToARGBRow_NEON(const uint8* src_rgb24, uint8* dst_argb, int width);
>-void RGB24ToARGBRow_MSA(const uint8* src_rgb24, uint8* dst_argb, int width);
>-void RAWToARGBRow_NEON(const uint8* src_raw, uint8* dst_argb, int width);
>-void RAWToARGBRow_MSA(const uint8* src_raw, uint8* dst_argb, int width);
>-void RAWToRGB24Row_NEON(const uint8* src_raw, uint8* dst_rgb24, int width);
>-void RAWToRGB24Row_MSA(const uint8* src_raw, uint8* dst_rgb24, int width);
>-void RGB565ToARGBRow_NEON(const uint8* src_rgb565, uint8* dst_argb, int width);
>-void RGB565ToARGBRow_MSA(const uint8* src_rgb565, uint8* dst_argb, int width);
>-void ARGB1555ToARGBRow_NEON(const uint8* src_argb1555,
>-                            uint8* dst_argb,
>-                            int width);
>-void ARGB1555ToARGBRow_MSA(const uint8* src_argb1555,
>-                           uint8* dst_argb,
>+void ARGBShuffleRow_Any_MSA(const uint8_t* src_ptr,
>+                            uint8_t* dst_ptr,
>+                            const uint8_t* param,
>+                            int width);
>+
>+void RGB24ToARGBRow_SSSE3(const uint8_t* src_rgb24,
>+                          uint8_t* dst_argb,
>+                          int width);
>+void RAWToARGBRow_SSSE3(const uint8_t* src_raw, uint8_t* dst_argb, int width);
>+void RAWToRGB24Row_SSSE3(const uint8_t* src_raw, uint8_t* dst_rgb24, int width);
>+void RGB565ToARGBRow_SSE2(const uint8_t* src, uint8_t* dst, int width);
>+void ARGB1555ToARGBRow_SSE2(const uint8_t* src, uint8_t* dst, int width);
>+void ARGB4444ToARGBRow_SSE2(const uint8_t* src, uint8_t* dst, int width);
>+void RGB565ToARGBRow_AVX2(const uint8_t* src_rgb565,
>+                          uint8_t* dst_argb,
>+                          int width);
>+void ARGB1555ToARGBRow_AVX2(const uint8_t* src_argb1555,
>+                            uint8_t* dst_argb,
>+                            int width);
>+void ARGB4444ToARGBRow_AVX2(const uint8_t* src_argb4444,
>+                            uint8_t* dst_argb,
>+                            int width);
>+
>+void RGB24ToARGBRow_NEON(const uint8_t* src_rgb24,
>+                         uint8_t* dst_argb,
>+                         int width);
>+void RGB24ToARGBRow_MSA(const uint8_t* src_rgb24, uint8_t* dst_argb, int width);
>+void RAWToARGBRow_NEON(const uint8_t* src_raw, uint8_t* dst_argb, int width);
>+void RAWToARGBRow_MSA(const uint8_t* src_raw, uint8_t* dst_argb, int width);
>+void RAWToRGB24Row_NEON(const uint8_t* src_raw, uint8_t* dst_rgb24, int width);
>+void RAWToRGB24Row_MSA(const uint8_t* src_raw, uint8_t* dst_rgb24, int width);
>+void RGB565ToARGBRow_NEON(const uint8_t* src_rgb565,
>+                          uint8_t* dst_argb,
>+                          int width);
>+void RGB565ToARGBRow_MSA(const uint8_t* src_rgb565,
>+                         uint8_t* dst_argb,
>+                         int width);
>+void ARGB1555ToARGBRow_NEON(const uint8_t* src_argb1555,
>+                            uint8_t* dst_argb,
>+                            int width);
>+void ARGB1555ToARGBRow_MSA(const uint8_t* src_argb1555,
>+                           uint8_t* dst_argb,
>                            int width);
>-void ARGB4444ToARGBRow_NEON(const uint8* src_argb4444,
>-                            uint8* dst_argb,
>-                            int width);
>-void RGB24ToARGBRow_DSPR2(const uint8* src_rgb24, uint8* dst_argb, int width);
>-void RAWToARGBRow_DSPR2(const uint8* src_raw, uint8* dst_argb, int width);
>-void RGB565ToARGBRow_DSPR2(const uint8* src_rgb565, uint8* dst_argb, int width);
>-void ARGB1555ToARGBRow_DSPR2(const uint8* src_argb1555,
>-                             uint8* dst_argb,
>-                             int width);
>-void ARGB4444ToARGBRow_DSPR2(const uint8* src_argb4444,
>-                             uint8* dst_argb,
>-                             int width);
>-void ARGB4444ToARGBRow_MSA(const uint8* src_argb4444,
>-                           uint8* dst_argb,
>+void ARGB4444ToARGBRow_NEON(const uint8_t* src_argb4444,
>+                            uint8_t* dst_argb,
>+                            int width);
>+void ARGB4444ToARGBRow_MSA(const uint8_t* src_argb4444,
>+                           uint8_t* dst_argb,
>                            int width);
>-void RGB24ToARGBRow_C(const uint8* src_rgb24, uint8* dst_argb, int width);
>-void RAWToARGBRow_C(const uint8* src_raw, uint8* dst_argb, int width);
>-void RAWToRGB24Row_C(const uint8* src_raw, uint8* dst_rgb24, int width);
>-void RGB565ToARGBRow_C(const uint8* src_rgb, uint8* dst_argb, int width);
>-void ARGB1555ToARGBRow_C(const uint8* src_argb, uint8* dst_argb, int width);
>-void ARGB4444ToARGBRow_C(const uint8* src_argb, uint8* dst_argb, int width);
>-void RGB24ToARGBRow_Any_SSSE3(const uint8* src_rgb24,
>-                              uint8* dst_argb,
>+void RGB24ToARGBRow_C(const uint8_t* src_rgb24, uint8_t* dst_argb, int width);
>+void RAWToARGBRow_C(const uint8_t* src_raw, uint8_t* dst_argb, int width);
>+void RAWToRGB24Row_C(const uint8_t* src_raw, uint8_t* dst_rgb24, int width);
>+void RGB565ToARGBRow_C(const uint8_t* src_rgb565, uint8_t* dst_argb, int width);
>+void ARGB1555ToARGBRow_C(const uint8_t* src_argb1555,
>+                         uint8_t* dst_argb,
>+                         int width);
>+void ARGB4444ToARGBRow_C(const uint8_t* src_argb4444,
>+                         uint8_t* dst_argb,
>+                         int width);
>+void AR30ToARGBRow_C(const uint8_t* src_ar30, uint8_t* dst_argb, int width);
>+void AR30ToABGRRow_C(const uint8_t* src_ar30, uint8_t* dst_abgr, int width);
>+void ARGBToAR30Row_C(const uint8_t* src_argb, uint8_t* dst_ar30, int width);
>+void AR30ToAB30Row_C(const uint8_t* src_ar30, uint8_t* dst_ab30, int width);
>+
>+void RGB24ToARGBRow_Any_SSSE3(const uint8_t* src_ptr,
>+                              uint8_t* dst_ptr,
>                               int width);
>-void RAWToARGBRow_Any_SSSE3(const uint8* src_raw, uint8* dst_argb, int width);
>-void RAWToRGB24Row_Any_SSSE3(const uint8* src_raw, uint8* dst_rgb24, int width);
>+void RAWToARGBRow_Any_SSSE3(const uint8_t* src_ptr,
>+                            uint8_t* dst_ptr,
>+                            int width);
>+void RAWToRGB24Row_Any_SSSE3(const uint8_t* src_ptr,
>+                             uint8_t* dst_ptr,
>+                             int width);
> 
>-void RGB565ToARGBRow_Any_SSE2(const uint8* src_rgb565,
>-                              uint8* dst_argb,
>+void RGB565ToARGBRow_Any_SSE2(const uint8_t* src_ptr,
>+                              uint8_t* dst_ptr,
>                               int width);
>-void ARGB1555ToARGBRow_Any_SSE2(const uint8* src_argb1555,
>-                                uint8* dst_argb,
>+void ARGB1555ToARGBRow_Any_SSE2(const uint8_t* src_ptr,
>+                                uint8_t* dst_ptr,
>                                 int width);
>-void ARGB4444ToARGBRow_Any_SSE2(const uint8* src_argb4444,
>-                                uint8* dst_argb,
>+void ARGB4444ToARGBRow_Any_SSE2(const uint8_t* src_ptr,
>+                                uint8_t* dst_ptr,
>                                 int width);
>-void RGB565ToARGBRow_Any_AVX2(const uint8* src_rgb565,
>-                              uint8* dst_argb,
>+void RGB565ToARGBRow_Any_AVX2(const uint8_t* src_ptr,
>+                              uint8_t* dst_ptr,
>                               int width);
>-void ARGB1555ToARGBRow_Any_AVX2(const uint8* src_argb1555,
>-                                uint8* dst_argb,
>+void ARGB1555ToARGBRow_Any_AVX2(const uint8_t* src_ptr,
>+                                uint8_t* dst_ptr,
>                                 int width);
>-void ARGB4444ToARGBRow_Any_AVX2(const uint8* src_argb4444,
>-                                uint8* dst_argb,
>+void ARGB4444ToARGBRow_Any_AVX2(const uint8_t* src_ptr,
>+                                uint8_t* dst_ptr,
>                                 int width);
> 
>-void RGB24ToARGBRow_Any_NEON(const uint8* src_rgb24,
>-                             uint8* dst_argb,
>+void RGB24ToARGBRow_Any_NEON(const uint8_t* src_ptr,
>+                             uint8_t* dst_ptr,
>                              int width);
>-void RGB24ToARGBRow_Any_MSA(const uint8* src_rgb24, uint8* dst_argb, int width);
>-void RAWToARGBRow_Any_NEON(const uint8* src_raw, uint8* dst_argb, int width);
>-void RAWToARGBRow_Any_MSA(const uint8* src_raw, uint8* dst_argb, int width);
>-void RAWToRGB24Row_Any_NEON(const uint8* src_raw, uint8* dst_rgb24, int width);
>-void RAWToRGB24Row_Any_MSA(const uint8* src_raw, uint8* dst_rgb24, int width);
>-void RGB565ToARGBRow_Any_NEON(const uint8* src_rgb565,
>-                              uint8* dst_argb,
>+void RGB24ToARGBRow_Any_MSA(const uint8_t* src_ptr,
>+                            uint8_t* dst_ptr,
>+                            int width);
>+void RAWToARGBRow_Any_NEON(const uint8_t* src_ptr, uint8_t* dst_ptr, int width);
>+void RAWToARGBRow_Any_MSA(const uint8_t* src_ptr, uint8_t* dst_ptr, int width);
>+void RAWToRGB24Row_Any_NEON(const uint8_t* src_ptr,
>+                            uint8_t* dst_ptr,
>+                            int width);
>+void RAWToRGB24Row_Any_MSA(const uint8_t* src_ptr, uint8_t* dst_ptr, int width);
>+void RGB565ToARGBRow_Any_NEON(const uint8_t* src_ptr,
>+                              uint8_t* dst_ptr,
>                               int width);
>-void RGB565ToARGBRow_Any_MSA(const uint8* src_rgb565,
>-                             uint8* dst_argb,
>+void RGB565ToARGBRow_Any_MSA(const uint8_t* src_ptr,
>+                             uint8_t* dst_ptr,
>                              int width);
>-void ARGB1555ToARGBRow_Any_NEON(const uint8* src_argb1555,
>-                                uint8* dst_argb,
>+void ARGB1555ToARGBRow_Any_NEON(const uint8_t* src_ptr,
>+                                uint8_t* dst_ptr,
>                                 int width);
>-void ARGB1555ToARGBRow_Any_MSA(const uint8* src_argb1555,
>-                               uint8* dst_argb,
>+void ARGB1555ToARGBRow_Any_MSA(const uint8_t* src_ptr,
>+                               uint8_t* dst_ptr,
>                                int width);
>-void ARGB4444ToARGBRow_Any_NEON(const uint8* src_argb4444,
>-                                uint8* dst_argb,
>+void ARGB4444ToARGBRow_Any_NEON(const uint8_t* src_ptr,
>+                                uint8_t* dst_ptr,
>                                 int width);
>-void RGB24ToARGBRow_Any_DSPR2(const uint8* src_rgb24,
>-                              uint8* dst_argb,
>-                              int width);
>-void RAWToARGBRow_Any_DSPR2(const uint8* src_raw, uint8* dst_argb, int width);
>-void RGB565ToARGBRow_Any_DSPR2(const uint8* src_rgb565,
>-                               uint8* dst_argb,
>-                               int width);
>-void ARGB1555ToARGBRow_Any_DSPR2(const uint8* src_argb1555,
>-                                 uint8* dst_argb,
>-                                 int width);
>-void ARGB4444ToARGBRow_Any_DSPR2(const uint8* src_argb4444,
>-                                 uint8* dst_argb,
>-                                 int width);
> 
>-void ARGB4444ToARGBRow_Any_MSA(const uint8* src_argb4444,
>-                               uint8* dst_argb,
>+void ARGB4444ToARGBRow_Any_MSA(const uint8_t* src_ptr,
>+                               uint8_t* dst_ptr,
>                                int width);
> 
>-void ARGBToRGB24Row_SSSE3(const uint8* src_argb, uint8* dst_rgb, int width);
>-void ARGBToRAWRow_SSSE3(const uint8* src_argb, uint8* dst_rgb, int width);
>-void ARGBToRGB565Row_SSE2(const uint8* src_argb, uint8* dst_rgb, int width);
>-void ARGBToARGB1555Row_SSE2(const uint8* src_argb, uint8* dst_rgb, int width);
>-void ARGBToARGB4444Row_SSE2(const uint8* src_argb, uint8* dst_rgb, int width);
>-
>-void ARGBToRGB565DitherRow_C(const uint8* src_argb,
>-                             uint8* dst_rgb,
>-                             const uint32 dither4,
>+void ARGBToRGB24Row_SSSE3(const uint8_t* src, uint8_t* dst, int width);
>+void ARGBToRAWRow_SSSE3(const uint8_t* src, uint8_t* dst, int width);
>+void ARGBToRGB565Row_SSE2(const uint8_t* src, uint8_t* dst, int width);
>+void ARGBToARGB1555Row_SSE2(const uint8_t* src, uint8_t* dst, int width);
>+void ARGBToARGB4444Row_SSE2(const uint8_t* src, uint8_t* dst, int width);
>+void ABGRToAR30Row_SSSE3(const uint8_t* src, uint8_t* dst, int width);
>+void ARGBToAR30Row_SSSE3(const uint8_t* src, uint8_t* dst, int width);
>+
>+void ARGBToRGB565DitherRow_C(const uint8_t* src_argb,
>+                             uint8_t* dst_rgb,
>+                             const uint32_t dither4,
>                              int width);
>-void ARGBToRGB565DitherRow_SSE2(const uint8* src_argb,
>-                                uint8* dst_rgb,
>-                                const uint32 dither4,
>+void ARGBToRGB565DitherRow_SSE2(const uint8_t* src,
>+                                uint8_t* dst,
>+                                const uint32_t dither4,
>                                 int width);
>-void ARGBToRGB565DitherRow_AVX2(const uint8* src_argb,
>-                                uint8* dst_rgb,
>-                                const uint32 dither4,
>+void ARGBToRGB565DitherRow_AVX2(const uint8_t* src,
>+                                uint8_t* dst,
>+                                const uint32_t dither4,
>                                 int width);
> 
>-void ARGBToRGB565Row_AVX2(const uint8* src_argb, uint8* dst_rgb, int width);
>-void ARGBToARGB1555Row_AVX2(const uint8* src_argb, uint8* dst_rgb, int width);
>-void ARGBToARGB4444Row_AVX2(const uint8* src_argb, uint8* dst_rgb, int width);
>-
>-void ARGBToRGB24Row_NEON(const uint8* src_argb, uint8* dst_rgb, int width);
>-void ARGBToRAWRow_NEON(const uint8* src_argb, uint8* dst_rgb, int width);
>-void ARGBToRGB565Row_NEON(const uint8* src_argb, uint8* dst_rgb, int width);
>-void ARGBToARGB1555Row_NEON(const uint8* src_argb, uint8* dst_rgb, int width);
>-void ARGBToARGB4444Row_NEON(const uint8* src_argb, uint8* dst_rgb, int width);
>-void ARGBToRGB565DitherRow_NEON(const uint8* src_argb,
>-                                uint8* dst_rgb,
>-                                const uint32 dither4,
>+void ARGBToRGB565Row_AVX2(const uint8_t* src_argb, uint8_t* dst_rgb, int width);
>+void ARGBToARGB1555Row_AVX2(const uint8_t* src_argb,
>+                            uint8_t* dst_rgb,
>+                            int width);
>+void ARGBToARGB4444Row_AVX2(const uint8_t* src_argb,
>+                            uint8_t* dst_rgb,
>+                            int width);
>+void ABGRToAR30Row_AVX2(const uint8_t* src, uint8_t* dst, int width);
>+void ARGBToAR30Row_AVX2(const uint8_t* src, uint8_t* dst, int width);
>+
>+void ARGBToRGB24Row_NEON(const uint8_t* src_argb,
>+                         uint8_t* dst_rgb24,
>+                         int width);
>+void ARGBToRAWRow_NEON(const uint8_t* src_argb, uint8_t* dst_raw, int width);
>+void ARGBToRGB565Row_NEON(const uint8_t* src_argb,
>+                          uint8_t* dst_rgb565,
>+                          int width);
>+void ARGBToARGB1555Row_NEON(const uint8_t* src_argb,
>+                            uint8_t* dst_argb1555,
>+                            int width);
>+void ARGBToARGB4444Row_NEON(const uint8_t* src_argb,
>+                            uint8_t* dst_argb4444,
>+                            int width);
>+void ARGBToRGB565DitherRow_NEON(const uint8_t* src_argb,
>+                                uint8_t* dst_rgb,
>+                                const uint32_t dither4,
>                                 int width);
>-void ARGBToRGB24Row_MSA(const uint8* src_argb, uint8* dst_rgb, int width);
>-void ARGBToRAWRow_MSA(const uint8* src_argb, uint8* dst_rgb, int width);
>-void ARGBToRGB565Row_MSA(const uint8* src_argb, uint8* dst_rgb, int width);
>-void ARGBToARGB1555Row_MSA(const uint8* src_argb, uint8* dst_rgb, int width);
>-void ARGBToARGB4444Row_MSA(const uint8* src_argb, uint8* dst_rgb, int width);
>-void ARGBToRGB565DitherRow_MSA(const uint8* src_argb,
>-                               uint8* dst_rgb,
>-                               const uint32 dither4,
>+void ARGBToRGB24Row_MSA(const uint8_t* src_argb, uint8_t* dst_rgb, int width);
>+void ARGBToRAWRow_MSA(const uint8_t* src_argb, uint8_t* dst_rgb, int width);
>+void ARGBToRGB565Row_MSA(const uint8_t* src_argb, uint8_t* dst_rgb, int width);
>+void ARGBToARGB1555Row_MSA(const uint8_t* src_argb,
>+                           uint8_t* dst_rgb,
>+                           int width);
>+void ARGBToARGB4444Row_MSA(const uint8_t* src_argb,
>+                           uint8_t* dst_rgb,
>+                           int width);
>+void ARGBToRGB565DitherRow_MSA(const uint8_t* src_argb,
>+                               uint8_t* dst_rgb,
>+                               const uint32_t dither4,
>                                int width);
> 
>-void ARGBToRGBARow_C(const uint8* src_argb, uint8* dst_rgb, int width);
>-void ARGBToRGB24Row_C(const uint8* src_argb, uint8* dst_rgb, int width);
>-void ARGBToRAWRow_C(const uint8* src_argb, uint8* dst_rgb, int width);
>-void ARGBToRGB565Row_C(const uint8* src_argb, uint8* dst_rgb, int width);
>-void ARGBToARGB1555Row_C(const uint8* src_argb, uint8* dst_rgb, int width);
>-void ARGBToARGB4444Row_C(const uint8* src_argb, uint8* dst_rgb, int width);
>-
>-void J400ToARGBRow_SSE2(const uint8* src_y, uint8* dst_argb, int width);
>-void J400ToARGBRow_AVX2(const uint8* src_y, uint8* dst_argb, int width);
>-void J400ToARGBRow_NEON(const uint8* src_y, uint8* dst_argb, int width);
>-void J400ToARGBRow_MSA(const uint8* src_y, uint8* dst_argb, int width);
>-void J400ToARGBRow_C(const uint8* src_y, uint8* dst_argb, int width);
>-void J400ToARGBRow_Any_SSE2(const uint8* src_y, uint8* dst_argb, int width);
>-void J400ToARGBRow_Any_AVX2(const uint8* src_y, uint8* dst_argb, int width);
>-void J400ToARGBRow_Any_NEON(const uint8* src_y, uint8* dst_argb, int width);
>-void J400ToARGBRow_Any_MSA(const uint8* src_y, uint8* dst_argb, int width);
>-
>-void I444ToARGBRow_C(const uint8* src_y,
>-                     const uint8* src_u,
>-                     const uint8* src_v,
>-                     uint8* dst_argb,
>+void ARGBToRGBARow_C(const uint8_t* src_argb, uint8_t* dst_rgb, int width);
>+void ARGBToRGB24Row_C(const uint8_t* src_argb, uint8_t* dst_rgb, int width);
>+void ARGBToRAWRow_C(const uint8_t* src_argb, uint8_t* dst_rgb, int width);
>+void ARGBToRGB565Row_C(const uint8_t* src_argb, uint8_t* dst_rgb, int width);
>+void ARGBToARGB1555Row_C(const uint8_t* src_argb, uint8_t* dst_rgb, int width);
>+void ARGBToARGB4444Row_C(const uint8_t* src_argb, uint8_t* dst_rgb, int width);
>+void ABGRToAR30Row_C(const uint8_t* src_abgr, uint8_t* dst_ar30, int width);
>+void ARGBToAR30Row_C(const uint8_t* src_argb, uint8_t* dst_ar30, int width);
>+
>+void J400ToARGBRow_SSE2(const uint8_t* src_y, uint8_t* dst_argb, int width);
>+void J400ToARGBRow_AVX2(const uint8_t* src_y, uint8_t* dst_argb, int width);
>+void J400ToARGBRow_NEON(const uint8_t* src_y, uint8_t* dst_argb, int width);
>+void J400ToARGBRow_MSA(const uint8_t* src_y, uint8_t* dst_argb, int width);
>+void J400ToARGBRow_C(const uint8_t* src_y, uint8_t* dst_argb, int width);
>+void J400ToARGBRow_Any_SSE2(const uint8_t* src_ptr,
>+                            uint8_t* dst_ptr,
>+                            int width);
>+void J400ToARGBRow_Any_AVX2(const uint8_t* src_ptr,
>+                            uint8_t* dst_ptr,
>+                            int width);
>+void J400ToARGBRow_Any_NEON(const uint8_t* src_ptr,
>+                            uint8_t* dst_ptr,
>+                            int width);
>+void J400ToARGBRow_Any_MSA(const uint8_t* src_ptr, uint8_t* dst_ptr, int width);
>+
>+void I444ToARGBRow_C(const uint8_t* src_y,
>+                     const uint8_t* src_u,
>+                     const uint8_t* src_v,
>+                     uint8_t* rgb_buf,
>+                     const struct YuvConstants* yuvconstants,
>+                     int width);
>+void I422ToARGBRow_C(const uint8_t* src_y,
>+                     const uint8_t* src_u,
>+                     const uint8_t* src_v,
>+                     uint8_t* rgb_buf,
>+                     const struct YuvConstants* yuvconstants,
>+                     int width);
>+void I422ToAR30Row_C(const uint8_t* src_y,
>+                     const uint8_t* src_u,
>+                     const uint8_t* src_v,
>+                     uint8_t* rgb_buf,
>                      const struct YuvConstants* yuvconstants,
>                      int width);
>-void I422ToARGBRow_C(const uint8* src_y,
>-                     const uint8* src_u,
>-                     const uint8* src_v,
>-                     uint8* dst_argb,
>+void I210ToAR30Row_C(const uint16_t* src_y,
>+                     const uint16_t* src_u,
>+                     const uint16_t* src_v,
>+                     uint8_t* rgb_buf,
>                      const struct YuvConstants* yuvconstants,
>                      int width);
>-void I422ToARGBRow_C(const uint8* src_y,
>-                     const uint8* src_u,
>-                     const uint8* src_v,
>-                     uint8* dst_argb,
>+void I210ToARGBRow_C(const uint16_t* src_y,
>+                     const uint16_t* src_u,
>+                     const uint16_t* src_v,
>+                     uint8_t* rgb_buf,
>                      const struct YuvConstants* yuvconstants,
>                      int width);
>-void I422AlphaToARGBRow_C(const uint8* y_buf,
>-                          const uint8* u_buf,
>-                          const uint8* v_buf,
>-                          const uint8* a_buf,
>-                          uint8* dst_argb,
>+void I422AlphaToARGBRow_C(const uint8_t* src_y,
>+                          const uint8_t* src_u,
>+                          const uint8_t* src_v,
>+                          const uint8_t* src_a,
>+                          uint8_t* rgb_buf,
>                           const struct YuvConstants* yuvconstants,
>                           int width);
>-void NV12ToARGBRow_C(const uint8* src_y,
>-                     const uint8* src_uv,
>-                     uint8* dst_argb,
>+void NV12ToARGBRow_C(const uint8_t* src_y,
>+                     const uint8_t* src_uv,
>+                     uint8_t* rgb_buf,
>                      const struct YuvConstants* yuvconstants,
>                      int width);
>-void NV12ToRGB565Row_C(const uint8* src_y,
>-                       const uint8* src_uv,
>-                       uint8* dst_argb,
>+void NV12ToRGB565Row_C(const uint8_t* src_y,
>+                       const uint8_t* src_uv,
>+                       uint8_t* dst_rgb565,
>                        const struct YuvConstants* yuvconstants,
>                        int width);
>-void NV21ToARGBRow_C(const uint8* src_y,
>-                     const uint8* src_uv,
>-                     uint8* dst_argb,
>+void NV21ToARGBRow_C(const uint8_t* src_y,
>+                     const uint8_t* src_vu,
>+                     uint8_t* rgb_buf,
>                      const struct YuvConstants* yuvconstants,
>                      int width);
>-void YUY2ToARGBRow_C(const uint8* src_yuy2,
>-                     uint8* dst_argb,
>+void NV12ToRGB24Row_C(const uint8_t* src_y,
>+                      const uint8_t* src_uv,
>+                      uint8_t* rgb_buf,
>+                      const struct YuvConstants* yuvconstants,
>+                      int width);
>+void NV21ToRGB24Row_C(const uint8_t* src_y,
>+                      const uint8_t* src_vu,
>+                      uint8_t* rgb_buf,
>+                      const struct YuvConstants* yuvconstants,
>+                      int width);
>+void YUY2ToARGBRow_C(const uint8_t* src_yuy2,
>+                     uint8_t* rgb_buf,
>                      const struct YuvConstants* yuvconstants,
>                      int width);
>-void UYVYToARGBRow_C(const uint8* src_uyvy,
>-                     uint8* dst_argb,
>+void UYVYToARGBRow_C(const uint8_t* src_uyvy,
>+                     uint8_t* rgb_buf,
>                      const struct YuvConstants* yuvconstants,
>                      int width);
>-void I422ToRGBARow_C(const uint8* src_y,
>-                     const uint8* src_u,
>-                     const uint8* src_v,
>-                     uint8* dst_rgba,
>+void I422ToRGBARow_C(const uint8_t* src_y,
>+                     const uint8_t* src_u,
>+                     const uint8_t* src_v,
>+                     uint8_t* rgb_buf,
>                      const struct YuvConstants* yuvconstants,
>                      int width);
>-void I422ToRGB24Row_C(const uint8* src_y,
>-                      const uint8* src_u,
>-                      const uint8* src_v,
>-                      uint8* dst_rgb24,
>+void I422ToRGB24Row_C(const uint8_t* src_y,
>+                      const uint8_t* src_u,
>+                      const uint8_t* src_v,
>+                      uint8_t* rgb_buf,
>                       const struct YuvConstants* yuvconstants,
>                       int width);
>-void I422ToARGB4444Row_C(const uint8* src_y,
>-                         const uint8* src_u,
>-                         const uint8* src_v,
>-                         uint8* dst_argb4444,
>+void I422ToARGB4444Row_C(const uint8_t* src_y,
>+                         const uint8_t* src_u,
>+                         const uint8_t* src_v,
>+                         uint8_t* dst_argb4444,
>                          const struct YuvConstants* yuvconstants,
>                          int width);
>-void I422ToARGB1555Row_C(const uint8* src_y,
>-                         const uint8* src_u,
>-                         const uint8* src_v,
>-                         uint8* dst_argb4444,
>+void I422ToARGB1555Row_C(const uint8_t* src_y,
>+                         const uint8_t* src_u,
>+                         const uint8_t* src_v,
>+                         uint8_t* dst_argb1555,
>                          const struct YuvConstants* yuvconstants,
>                          int width);
>-void I422ToRGB565Row_C(const uint8* src_y,
>-                       const uint8* src_u,
>-                       const uint8* src_v,
>-                       uint8* dst_rgb565,
>+void I422ToRGB565Row_C(const uint8_t* src_y,
>+                       const uint8_t* src_u,
>+                       const uint8_t* src_v,
>+                       uint8_t* dst_rgb565,
>                        const struct YuvConstants* yuvconstants,
>                        int width);
>-void I422ToARGBRow_AVX2(const uint8* src_y,
>-                        const uint8* src_u,
>-                        const uint8* src_v,
>-                        uint8* dst_argb,
>-                        const struct YuvConstants* yuvconstants,
>-                        int width);
>-void I422ToARGBRow_AVX2(const uint8* src_y,
>-                        const uint8* src_u,
>-                        const uint8* src_v,
>-                        uint8* dst_argb,
>+void I422ToARGBRow_AVX2(const uint8_t* y_buf,
>+                        const uint8_t* u_buf,
>+                        const uint8_t* v_buf,
>+                        uint8_t* dst_argb,
>                         const struct YuvConstants* yuvconstants,
>                         int width);
>-void I422ToRGBARow_AVX2(const uint8* src_y,
>-                        const uint8* src_u,
>-                        const uint8* src_v,
>-                        uint8* dst_argb,
>+void I422ToRGBARow_AVX2(const uint8_t* y_buf,
>+                        const uint8_t* u_buf,
>+                        const uint8_t* v_buf,
>+                        uint8_t* dst_argb,
>                         const struct YuvConstants* yuvconstants,
>                         int width);
>-void I444ToARGBRow_SSSE3(const uint8* src_y,
>-                         const uint8* src_u,
>-                         const uint8* src_v,
>-                         uint8* dst_argb,
>+void I444ToARGBRow_SSSE3(const uint8_t* y_buf,
>+                         const uint8_t* u_buf,
>+                         const uint8_t* v_buf,
>+                         uint8_t* dst_argb,
>                          const struct YuvConstants* yuvconstants,
>                          int width);
>-void I444ToARGBRow_AVX2(const uint8* src_y,
>-                        const uint8* src_u,
>-                        const uint8* src_v,
>-                        uint8* dst_argb,
>+void I444ToARGBRow_AVX2(const uint8_t* y_buf,
>+                        const uint8_t* u_buf,
>+                        const uint8_t* v_buf,
>+                        uint8_t* dst_argb,
>                         const struct YuvConstants* yuvconstants,
>                         int width);
>-void I444ToARGBRow_SSSE3(const uint8* src_y,
>-                         const uint8* src_u,
>-                         const uint8* src_v,
>-                         uint8* dst_argb,
>+void I444ToARGBRow_SSSE3(const uint8_t* y_buf,
>+                         const uint8_t* u_buf,
>+                         const uint8_t* v_buf,
>+                         uint8_t* dst_argb,
>                          const struct YuvConstants* yuvconstants,
>                          int width);
>-void I444ToARGBRow_AVX2(const uint8* src_y,
>-                        const uint8* src_u,
>-                        const uint8* src_v,
>-                        uint8* dst_argb,
>+void I444ToARGBRow_AVX2(const uint8_t* y_buf,
>+                        const uint8_t* u_buf,
>+                        const uint8_t* v_buf,
>+                        uint8_t* dst_argb,
>                         const struct YuvConstants* yuvconstants,
>                         int width);
>-void I422ToARGBRow_SSSE3(const uint8* src_y,
>-                         const uint8* src_u,
>-                         const uint8* src_v,
>-                         uint8* dst_argb,
>+void I422ToARGBRow_SSSE3(const uint8_t* y_buf,
>+                         const uint8_t* u_buf,
>+                         const uint8_t* v_buf,
>+                         uint8_t* dst_argb,
>+                         const struct YuvConstants* yuvconstants,
>+                         int width);
>+
>+void I422ToAR30Row_SSSE3(const uint8_t* y_buf,
>+                         const uint8_t* u_buf,
>+                         const uint8_t* v_buf,
>+                         uint8_t* dst_ar30,
>                          const struct YuvConstants* yuvconstants,
>                          int width);
>-void I422AlphaToARGBRow_SSSE3(const uint8* y_buf,
>-                              const uint8* u_buf,
>-                              const uint8* v_buf,
>-                              const uint8* a_buf,
>-                              uint8* dst_argb,
>+void I210ToAR30Row_SSSE3(const uint16_t* y_buf,
>+                         const uint16_t* u_buf,
>+                         const uint16_t* v_buf,
>+                         uint8_t* dst_ar30,
>+                         const struct YuvConstants* yuvconstants,
>+                         int width);
>+void I210ToARGBRow_SSSE3(const uint16_t* y_buf,
>+                         const uint16_t* u_buf,
>+                         const uint16_t* v_buf,
>+                         uint8_t* dst_argb,
>+                         const struct YuvConstants* yuvconstants,
>+                         int width);
>+void I422ToAR30Row_AVX2(const uint8_t* y_buf,
>+                        const uint8_t* u_buf,
>+                        const uint8_t* v_buf,
>+                        uint8_t* dst_ar30,
>+                        const struct YuvConstants* yuvconstants,
>+                        int width);
>+void I210ToARGBRow_AVX2(const uint16_t* y_buf,
>+                        const uint16_t* u_buf,
>+                        const uint16_t* v_buf,
>+                        uint8_t* dst_argb,
>+                        const struct YuvConstants* yuvconstants,
>+                        int width);
>+void I210ToAR30Row_AVX2(const uint16_t* y_buf,
>+                        const uint16_t* u_buf,
>+                        const uint16_t* v_buf,
>+                        uint8_t* dst_ar30,
>+                        const struct YuvConstants* yuvconstants,
>+                        int width);
>+void I422AlphaToARGBRow_SSSE3(const uint8_t* y_buf,
>+                              const uint8_t* u_buf,
>+                              const uint8_t* v_buf,
>+                              const uint8_t* a_buf,
>+                              uint8_t* dst_argb,
>                               const struct YuvConstants* yuvconstants,
>                               int width);
>-void I422AlphaToARGBRow_AVX2(const uint8* y_buf,
>-                             const uint8* u_buf,
>-                             const uint8* v_buf,
>-                             const uint8* a_buf,
>-                             uint8* dst_argb,
>+void I422AlphaToARGBRow_AVX2(const uint8_t* y_buf,
>+                             const uint8_t* u_buf,
>+                             const uint8_t* v_buf,
>+                             const uint8_t* a_buf,
>+                             uint8_t* dst_argb,
>                              const struct YuvConstants* yuvconstants,
>                              int width);
>-void I422ToARGBRow_SSSE3(const uint8* src_y,
>-                         const uint8* src_u,
>-                         const uint8* src_v,
>-                         uint8* dst_argb,
>-                         const struct YuvConstants* yuvconstants,
>-                         int width);
>-void NV12ToARGBRow_SSSE3(const uint8* src_y,
>-                         const uint8* src_uv,
>-                         uint8* dst_argb,
>+void NV12ToARGBRow_SSSE3(const uint8_t* y_buf,
>+                         const uint8_t* uv_buf,
>+                         uint8_t* dst_argb,
>                          const struct YuvConstants* yuvconstants,
>                          int width);
>-void NV12ToARGBRow_AVX2(const uint8* src_y,
>-                        const uint8* src_uv,
>-                        uint8* dst_argb,
>+void NV12ToARGBRow_AVX2(const uint8_t* y_buf,
>+                        const uint8_t* uv_buf,
>+                        uint8_t* dst_argb,
>                         const struct YuvConstants* yuvconstants,
>                         int width);
>-void NV12ToRGB565Row_SSSE3(const uint8* src_y,
>-                           const uint8* src_uv,
>-                           uint8* dst_argb,
>+void NV12ToRGB565Row_SSSE3(const uint8_t* src_y,
>+                           const uint8_t* src_uv,
>+                           uint8_t* dst_rgb565,
>                            const struct YuvConstants* yuvconstants,
>                            int width);
>-void NV12ToRGB565Row_AVX2(const uint8* src_y,
>-                          const uint8* src_uv,
>-                          uint8* dst_argb,
>+void NV12ToRGB565Row_AVX2(const uint8_t* src_y,
>+                          const uint8_t* src_uv,
>+                          uint8_t* dst_rgb565,
>                           const struct YuvConstants* yuvconstants,
>                           int width);
>-void NV21ToARGBRow_SSSE3(const uint8* src_y,
>-                         const uint8* src_uv,
>-                         uint8* dst_argb,
>+void NV21ToARGBRow_SSSE3(const uint8_t* y_buf,
>+                         const uint8_t* vu_buf,
>+                         uint8_t* dst_argb,
>                          const struct YuvConstants* yuvconstants,
>                          int width);
>-void NV21ToARGBRow_AVX2(const uint8* src_y,
>-                        const uint8* src_uv,
>-                        uint8* dst_argb,
>+void NV21ToARGBRow_AVX2(const uint8_t* y_buf,
>+                        const uint8_t* vu_buf,
>+                        uint8_t* dst_argb,
>                         const struct YuvConstants* yuvconstants,
>                         int width);
>-void YUY2ToARGBRow_SSSE3(const uint8* src_yuy2,
>-                         uint8* dst_argb,
>+void YUY2ToARGBRow_SSSE3(const uint8_t* yuy2_buf,
>+                         uint8_t* dst_argb,
>                          const struct YuvConstants* yuvconstants,
>                          int width);
>-void UYVYToARGBRow_SSSE3(const uint8* src_uyvy,
>-                         uint8* dst_argb,
>+void UYVYToARGBRow_SSSE3(const uint8_t* uyvy_buf,
>+                         uint8_t* dst_argb,
>                          const struct YuvConstants* yuvconstants,
>                          int width);
>-void YUY2ToARGBRow_AVX2(const uint8* src_yuy2,
>-                        uint8* dst_argb,
>+void YUY2ToARGBRow_AVX2(const uint8_t* yuy2_buf,
>+                        uint8_t* dst_argb,
>                         const struct YuvConstants* yuvconstants,
>                         int width);
>-void UYVYToARGBRow_AVX2(const uint8* src_uyvy,
>-                        uint8* dst_argb,
>+void UYVYToARGBRow_AVX2(const uint8_t* uyvy_buf,
>+                        uint8_t* dst_argb,
>                         const struct YuvConstants* yuvconstants,
>                         int width);
>-void I422ToRGBARow_SSSE3(const uint8* src_y,
>-                         const uint8* src_u,
>-                         const uint8* src_v,
>-                         uint8* dst_rgba,
>+void I422ToRGBARow_SSSE3(const uint8_t* y_buf,
>+                         const uint8_t* u_buf,
>+                         const uint8_t* v_buf,
>+                         uint8_t* dst_rgba,
>                          const struct YuvConstants* yuvconstants,
>                          int width);
>-void I422ToARGB4444Row_SSSE3(const uint8* src_y,
>-                             const uint8* src_u,
>-                             const uint8* src_v,
>-                             uint8* dst_argb,
>+void I422ToARGB4444Row_SSSE3(const uint8_t* src_y,
>+                             const uint8_t* src_u,
>+                             const uint8_t* src_v,
>+                             uint8_t* dst_argb4444,
>                              const struct YuvConstants* yuvconstants,
>                              int width);
>-void I422ToARGB4444Row_AVX2(const uint8* src_y,
>-                            const uint8* src_u,
>-                            const uint8* src_v,
>-                            uint8* dst_argb,
>+void I422ToARGB4444Row_AVX2(const uint8_t* src_y,
>+                            const uint8_t* src_u,
>+                            const uint8_t* src_v,
>+                            uint8_t* dst_argb4444,
>                             const struct YuvConstants* yuvconstants,
>                             int width);
>-void I422ToARGB1555Row_SSSE3(const uint8* src_y,
>-                             const uint8* src_u,
>-                             const uint8* src_v,
>-                             uint8* dst_argb,
>+void I422ToARGB1555Row_SSSE3(const uint8_t* src_y,
>+                             const uint8_t* src_u,
>+                             const uint8_t* src_v,
>+                             uint8_t* dst_argb1555,
>                              const struct YuvConstants* yuvconstants,
>                              int width);
>-void I422ToARGB1555Row_AVX2(const uint8* src_y,
>-                            const uint8* src_u,
>-                            const uint8* src_v,
>-                            uint8* dst_argb,
>+void I422ToARGB1555Row_AVX2(const uint8_t* src_y,
>+                            const uint8_t* src_u,
>+                            const uint8_t* src_v,
>+                            uint8_t* dst_argb1555,
>                             const struct YuvConstants* yuvconstants,
>                             int width);
>-void I422ToRGB565Row_SSSE3(const uint8* src_y,
>-                           const uint8* src_u,
>-                           const uint8* src_v,
>-                           uint8* dst_argb,
>+void I422ToRGB565Row_SSSE3(const uint8_t* src_y,
>+                           const uint8_t* src_u,
>+                           const uint8_t* src_v,
>+                           uint8_t* dst_rgb565,
>                            const struct YuvConstants* yuvconstants,
>                            int width);
>-void I422ToRGB565Row_AVX2(const uint8* src_y,
>-                          const uint8* src_u,
>-                          const uint8* src_v,
>-                          uint8* dst_argb,
>+void I422ToRGB565Row_AVX2(const uint8_t* src_y,
>+                          const uint8_t* src_u,
>+                          const uint8_t* src_v,
>+                          uint8_t* dst_rgb565,
>                           const struct YuvConstants* yuvconstants,
>                           int width);
>-void I422ToRGB24Row_SSSE3(const uint8* src_y,
>-                          const uint8* src_u,
>-                          const uint8* src_v,
>-                          uint8* dst_rgb24,
>+void I422ToRGB24Row_SSSE3(const uint8_t* y_buf,
>+                          const uint8_t* u_buf,
>+                          const uint8_t* v_buf,
>+                          uint8_t* dst_rgb24,
>                           const struct YuvConstants* yuvconstants,
>                           int width);
>-void I422ToRGB24Row_AVX2(const uint8* src_y,
>-                         const uint8* src_u,
>-                         const uint8* src_v,
>-                         uint8* dst_rgb24,
>+void I422ToRGB24Row_AVX2(const uint8_t* src_y,
>+                         const uint8_t* src_u,
>+                         const uint8_t* src_v,
>+                         uint8_t* dst_rgb24,
>                          const struct YuvConstants* yuvconstants,
>                          int width);
>-void I422ToARGBRow_Any_AVX2(const uint8* src_y,
>-                            const uint8* src_u,
>-                            const uint8* src_v,
>-                            uint8* dst_argb,
>+void I422ToARGBRow_Any_AVX2(const uint8_t* y_buf,
>+                            const uint8_t* u_buf,
>+                            const uint8_t* v_buf,
>+                            uint8_t* dst_ptr,
>                             const struct YuvConstants* yuvconstants,
>                             int width);
>-void I422ToRGBARow_Any_AVX2(const uint8* src_y,
>-                            const uint8* src_u,
>-                            const uint8* src_v,
>-                            uint8* dst_argb,
>+void I422ToRGBARow_Any_AVX2(const uint8_t* y_buf,
>+                            const uint8_t* u_buf,
>+                            const uint8_t* v_buf,
>+                            uint8_t* dst_ptr,
>                             const struct YuvConstants* yuvconstants,
>                             int width);
>-void I444ToARGBRow_Any_SSSE3(const uint8* src_y,
>-                             const uint8* src_u,
>-                             const uint8* src_v,
>-                             uint8* dst_argb,
>+void I444ToARGBRow_Any_SSSE3(const uint8_t* y_buf,
>+                             const uint8_t* u_buf,
>+                             const uint8_t* v_buf,
>+                             uint8_t* dst_ptr,
>                              const struct YuvConstants* yuvconstants,
>                              int width);
>-void I444ToARGBRow_Any_AVX2(const uint8* src_y,
>-                            const uint8* src_u,
>-                            const uint8* src_v,
>-                            uint8* dst_argb,
>+void I444ToARGBRow_Any_AVX2(const uint8_t* y_buf,
>+                            const uint8_t* u_buf,
>+                            const uint8_t* v_buf,
>+                            uint8_t* dst_ptr,
>                             const struct YuvConstants* yuvconstants,
>                             int width);
>-void I422ToARGBRow_Any_SSSE3(const uint8* src_y,
>-                             const uint8* src_u,
>-                             const uint8* src_v,
>-                             uint8* dst_argb,
>+void I422ToARGBRow_Any_SSSE3(const uint8_t* y_buf,
>+                             const uint8_t* u_buf,
>+                             const uint8_t* v_buf,
>+                             uint8_t* dst_ptr,
>+                             const struct YuvConstants* yuvconstants,
>+                             int width);
>+void I422ToAR30Row_Any_SSSE3(const uint8_t* y_buf,
>+                             const uint8_t* u_buf,
>+                             const uint8_t* v_buf,
>+                             uint8_t* dst_ptr,
>+                             const struct YuvConstants* yuvconstants,
>+                             int width);
>+void I210ToAR30Row_Any_SSSE3(const uint16_t* y_buf,
>+                             const uint16_t* u_buf,
>+                             const uint16_t* v_buf,
>+                             uint8_t* dst_ptr,
>                              const struct YuvConstants* yuvconstants,
>                              int width);
>-void I422AlphaToARGBRow_Any_SSSE3(const uint8* y_buf,
>-                                  const uint8* u_buf,
>-                                  const uint8* v_buf,
>-                                  const uint8* a_buf,
>-                                  uint8* dst_argb,
>+void I210ToARGBRow_Any_SSSE3(const uint16_t* y_buf,
>+                             const uint16_t* u_buf,
>+                             const uint16_t* v_buf,
>+                             uint8_t* dst_ptr,
>+                             const struct YuvConstants* yuvconstants,
>+                             int width);
>+void I422ToAR30Row_Any_AVX2(const uint8_t* y_buf,
>+                            const uint8_t* u_buf,
>+                            const uint8_t* v_buf,
>+                            uint8_t* dst_ptr,
>+                            const struct YuvConstants* yuvconstants,
>+                            int width);
>+void I210ToARGBRow_Any_AVX2(const uint16_t* y_buf,
>+                            const uint16_t* u_buf,
>+                            const uint16_t* v_buf,
>+                            uint8_t* dst_ptr,
>+                            const struct YuvConstants* yuvconstants,
>+                            int width);
>+void I210ToAR30Row_Any_AVX2(const uint16_t* y_buf,
>+                            const uint16_t* u_buf,
>+                            const uint16_t* v_buf,
>+                            uint8_t* dst_ptr,
>+                            const struct YuvConstants* yuvconstants,
>+                            int width);
>+void I422AlphaToARGBRow_Any_SSSE3(const uint8_t* y_buf,
>+                                  const uint8_t* u_buf,
>+                                  const uint8_t* v_buf,
>+                                  const uint8_t* a_buf,
>+                                  uint8_t* dst_ptr,
>                                   const struct YuvConstants* yuvconstants,
>                                   int width);
>-void I422AlphaToARGBRow_Any_AVX2(const uint8* y_buf,
>-                                 const uint8* u_buf,
>-                                 const uint8* v_buf,
>-                                 const uint8* a_buf,
>-                                 uint8* dst_argb,
>+void I422AlphaToARGBRow_Any_AVX2(const uint8_t* y_buf,
>+                                 const uint8_t* u_buf,
>+                                 const uint8_t* v_buf,
>+                                 const uint8_t* a_buf,
>+                                 uint8_t* dst_ptr,
>                                  const struct YuvConstants* yuvconstants,
>                                  int width);
>-void NV12ToARGBRow_Any_SSSE3(const uint8* src_y,
>-                             const uint8* src_uv,
>-                             uint8* dst_argb,
>+void NV12ToARGBRow_Any_SSSE3(const uint8_t* y_buf,
>+                             const uint8_t* uv_buf,
>+                             uint8_t* dst_ptr,
>                              const struct YuvConstants* yuvconstants,
>                              int width);
>-void NV12ToARGBRow_Any_AVX2(const uint8* src_y,
>-                            const uint8* src_uv,
>-                            uint8* dst_argb,
>+void NV12ToARGBRow_Any_AVX2(const uint8_t* y_buf,
>+                            const uint8_t* uv_buf,
>+                            uint8_t* dst_ptr,
>                             const struct YuvConstants* yuvconstants,
>                             int width);
>-void NV21ToARGBRow_Any_SSSE3(const uint8* src_y,
>-                             const uint8* src_vu,
>-                             uint8* dst_argb,
>+void NV21ToARGBRow_Any_SSSE3(const uint8_t* y_buf,
>+                             const uint8_t* uv_buf,
>+                             uint8_t* dst_ptr,
>                              const struct YuvConstants* yuvconstants,
>                              int width);
>-void NV21ToARGBRow_Any_AVX2(const uint8* src_y,
>-                            const uint8* src_vu,
>-                            uint8* dst_argb,
>+void NV21ToARGBRow_Any_AVX2(const uint8_t* y_buf,
>+                            const uint8_t* uv_buf,
>+                            uint8_t* dst_ptr,
>                             const struct YuvConstants* yuvconstants,
>                             int width);
>-void NV12ToRGB565Row_Any_SSSE3(const uint8* src_y,
>-                               const uint8* src_uv,
>-                               uint8* dst_argb,
>+void NV12ToRGB565Row_Any_SSSE3(const uint8_t* y_buf,
>+                               const uint8_t* uv_buf,
>+                               uint8_t* dst_ptr,
>                                const struct YuvConstants* yuvconstants,
>                                int width);
>-void NV12ToRGB565Row_Any_AVX2(const uint8* src_y,
>-                              const uint8* src_uv,
>-                              uint8* dst_argb,
>+void NV12ToRGB565Row_Any_AVX2(const uint8_t* y_buf,
>+                              const uint8_t* uv_buf,
>+                              uint8_t* dst_ptr,
>                               const struct YuvConstants* yuvconstants,
>                               int width);
>-void YUY2ToARGBRow_Any_SSSE3(const uint8* src_yuy2,
>-                             uint8* dst_argb,
>+void YUY2ToARGBRow_Any_SSSE3(const uint8_t* src_ptr,
>+                             uint8_t* dst_ptr,
>                              const struct YuvConstants* yuvconstants,
>                              int width);
>-void UYVYToARGBRow_Any_SSSE3(const uint8* src_uyvy,
>-                             uint8* dst_argb,
>+void UYVYToARGBRow_Any_SSSE3(const uint8_t* src_ptr,
>+                             uint8_t* dst_ptr,
>                              const struct YuvConstants* yuvconstants,
>                              int width);
>-void YUY2ToARGBRow_Any_AVX2(const uint8* src_yuy2,
>-                            uint8* dst_argb,
>+void YUY2ToARGBRow_Any_AVX2(const uint8_t* src_ptr,
>+                            uint8_t* dst_ptr,
>                             const struct YuvConstants* yuvconstants,
>                             int width);
>-void UYVYToARGBRow_Any_AVX2(const uint8* src_uyvy,
>-                            uint8* dst_argb,
>+void UYVYToARGBRow_Any_AVX2(const uint8_t* src_ptr,
>+                            uint8_t* dst_ptr,
>                             const struct YuvConstants* yuvconstants,
>                             int width);
>-void I422ToRGBARow_Any_SSSE3(const uint8* src_y,
>-                             const uint8* src_u,
>-                             const uint8* src_v,
>-                             uint8* dst_rgba,
>+void I422ToRGBARow_Any_SSSE3(const uint8_t* y_buf,
>+                             const uint8_t* u_buf,
>+                             const uint8_t* v_buf,
>+                             uint8_t* dst_ptr,
>                              const struct YuvConstants* yuvconstants,
>                              int width);
>-void I422ToARGB4444Row_Any_SSSE3(const uint8* src_y,
>-                                 const uint8* src_u,
>-                                 const uint8* src_v,
>-                                 uint8* dst_rgba,
>+void I422ToARGB4444Row_Any_SSSE3(const uint8_t* y_buf,
>+                                 const uint8_t* u_buf,
>+                                 const uint8_t* v_buf,
>+                                 uint8_t* dst_ptr,
>                                  const struct YuvConstants* yuvconstants,
>                                  int width);
>-void I422ToARGB4444Row_Any_AVX2(const uint8* src_y,
>-                                const uint8* src_u,
>-                                const uint8* src_v,
>-                                uint8* dst_rgba,
>+void I422ToARGB4444Row_Any_AVX2(const uint8_t* y_buf,
>+                                const uint8_t* u_buf,
>+                                const uint8_t* v_buf,
>+                                uint8_t* dst_ptr,
>                                 const struct YuvConstants* yuvconstants,
>                                 int width);
>-void I422ToARGB1555Row_Any_SSSE3(const uint8* src_y,
>-                                 const uint8* src_u,
>-                                 const uint8* src_v,
>-                                 uint8* dst_rgba,
>+void I422ToARGB1555Row_Any_SSSE3(const uint8_t* y_buf,
>+                                 const uint8_t* u_buf,
>+                                 const uint8_t* v_buf,
>+                                 uint8_t* dst_ptr,
>                                  const struct YuvConstants* yuvconstants,
>                                  int width);
>-void I422ToARGB1555Row_Any_AVX2(const uint8* src_y,
>-                                const uint8* src_u,
>-                                const uint8* src_v,
>-                                uint8* dst_rgba,
>+void I422ToARGB1555Row_Any_AVX2(const uint8_t* y_buf,
>+                                const uint8_t* u_buf,
>+                                const uint8_t* v_buf,
>+                                uint8_t* dst_ptr,
>                                 const struct YuvConstants* yuvconstants,
>                                 int width);
>-void I422ToRGB565Row_Any_SSSE3(const uint8* src_y,
>-                               const uint8* src_u,
>-                               const uint8* src_v,
>-                               uint8* dst_rgba,
>+void I422ToRGB565Row_Any_SSSE3(const uint8_t* y_buf,
>+                               const uint8_t* u_buf,
>+                               const uint8_t* v_buf,
>+                               uint8_t* dst_ptr,
>                                const struct YuvConstants* yuvconstants,
>                                int width);
>-void I422ToRGB565Row_Any_AVX2(const uint8* src_y,
>-                              const uint8* src_u,
>-                              const uint8* src_v,
>-                              uint8* dst_rgba,
>+void I422ToRGB565Row_Any_AVX2(const uint8_t* y_buf,
>+                              const uint8_t* u_buf,
>+                              const uint8_t* v_buf,
>+                              uint8_t* dst_ptr,
>                               const struct YuvConstants* yuvconstants,
>                               int width);
>-void I422ToRGB24Row_Any_SSSE3(const uint8* src_y,
>-                              const uint8* src_u,
>-                              const uint8* src_v,
>-                              uint8* dst_argb,
>+void I422ToRGB24Row_Any_SSSE3(const uint8_t* y_buf,
>+                              const uint8_t* u_buf,
>+                              const uint8_t* v_buf,
>+                              uint8_t* dst_ptr,
>                               const struct YuvConstants* yuvconstants,
>                               int width);
>-void I422ToRGB24Row_Any_AVX2(const uint8* src_y,
>-                             const uint8* src_u,
>-                             const uint8* src_v,
>-                             uint8* dst_argb,
>+void I422ToRGB24Row_Any_AVX2(const uint8_t* y_buf,
>+                             const uint8_t* u_buf,
>+                             const uint8_t* v_buf,
>+                             uint8_t* dst_ptr,
>                              const struct YuvConstants* yuvconstants,
>                              int width);
> 
>-void I400ToARGBRow_C(const uint8* src_y, uint8* dst_argb, int width);
>-void I400ToARGBRow_SSE2(const uint8* src_y, uint8* dst_argb, int width);
>-void I400ToARGBRow_AVX2(const uint8* src_y, uint8* dst_argb, int width);
>-void I400ToARGBRow_NEON(const uint8* src_y, uint8* dst_argb, int width);
>-void I400ToARGBRow_MSA(const uint8* src_y, uint8* dst_argb, int width);
>-void I400ToARGBRow_Any_SSE2(const uint8* src_y, uint8* dst_argb, int width);
>-void I400ToARGBRow_Any_AVX2(const uint8* src_y, uint8* dst_argb, int width);
>-void I400ToARGBRow_Any_NEON(const uint8* src_y, uint8* dst_argb, int width);
>-void I400ToARGBRow_Any_MSA(const uint8* src_y, uint8* dst_argb, int width);
>+void I400ToARGBRow_C(const uint8_t* src_y, uint8_t* rgb_buf, int width);
>+void I400ToARGBRow_SSE2(const uint8_t* y_buf, uint8_t* dst_argb, int width);
>+void I400ToARGBRow_AVX2(const uint8_t* y_buf, uint8_t* dst_argb, int width);
>+void I400ToARGBRow_NEON(const uint8_t* src_y, uint8_t* dst_argb, int width);
>+void I400ToARGBRow_MSA(const uint8_t* src_y, uint8_t* dst_argb, int width);
>+void I400ToARGBRow_Any_SSE2(const uint8_t* src_ptr,
>+                            uint8_t* dst_ptr,
>+                            int width);
>+void I400ToARGBRow_Any_AVX2(const uint8_t* src_ptr,
>+                            uint8_t* dst_ptr,
>+                            int width);
>+void I400ToARGBRow_Any_NEON(const uint8_t* src_ptr,
>+                            uint8_t* dst_ptr,
>+                            int width);
>+void I400ToARGBRow_Any_MSA(const uint8_t* src_ptr, uint8_t* dst_ptr, int width);
> 
> // ARGB preattenuated alpha blend.
>-void ARGBBlendRow_SSSE3(const uint8* src_argb,
>-                        const uint8* src_argb1,
>-                        uint8* dst_argb,
>+void ARGBBlendRow_SSSE3(const uint8_t* src_argb0,
>+                        const uint8_t* src_argb1,
>+                        uint8_t* dst_argb,
>                         int width);
>-void ARGBBlendRow_NEON(const uint8* src_argb,
>-                       const uint8* src_argb1,
>-                       uint8* dst_argb,
>+void ARGBBlendRow_NEON(const uint8_t* src_argb0,
>+                       const uint8_t* src_argb1,
>+                       uint8_t* dst_argb,
>                        int width);
>-void ARGBBlendRow_MSA(const uint8* src_argb,
>-                      const uint8* src_argb1,
>-                      uint8* dst_argb,
>+void ARGBBlendRow_MSA(const uint8_t* src_argb0,
>+                      const uint8_t* src_argb1,
>+                      uint8_t* dst_argb,
>                       int width);
>-void ARGBBlendRow_C(const uint8* src_argb,
>-                    const uint8* src_argb1,
>-                    uint8* dst_argb,
>+void ARGBBlendRow_C(const uint8_t* src_argb0,
>+                    const uint8_t* src_argb1,
>+                    uint8_t* dst_argb,
>                     int width);
> 
> // Unattenuated planar alpha blend.
>-void BlendPlaneRow_SSSE3(const uint8* src0,
>-                         const uint8* src1,
>-                         const uint8* alpha,
>-                         uint8* dst,
>-                         int width);
>-void BlendPlaneRow_Any_SSSE3(const uint8* src0,
>-                             const uint8* src1,
>-                             const uint8* alpha,
>-                             uint8* dst,
>+void BlendPlaneRow_SSSE3(const uint8_t* src0,
>+                         const uint8_t* src1,
>+                         const uint8_t* alpha,
>+                         uint8_t* dst,
>+                         int width);
>+void BlendPlaneRow_Any_SSSE3(const uint8_t* y_buf,
>+                             const uint8_t* u_buf,
>+                             const uint8_t* v_buf,
>+                             uint8_t* dst_ptr,
>                              int width);
>-void BlendPlaneRow_AVX2(const uint8* src0,
>-                        const uint8* src1,
>-                        const uint8* alpha,
>-                        uint8* dst,
>+void BlendPlaneRow_AVX2(const uint8_t* src0,
>+                        const uint8_t* src1,
>+                        const uint8_t* alpha,
>+                        uint8_t* dst,
>                         int width);
>-void BlendPlaneRow_Any_AVX2(const uint8* src0,
>-                            const uint8* src1,
>-                            const uint8* alpha,
>-                            uint8* dst,
>-                            int width);
>-void BlendPlaneRow_C(const uint8* src0,
>-                     const uint8* src1,
>-                     const uint8* alpha,
>-                     uint8* dst,
>+void BlendPlaneRow_Any_AVX2(const uint8_t* y_buf,
>+                            const uint8_t* u_buf,
>+                            const uint8_t* v_buf,
>+                            uint8_t* dst_ptr,
>+                            int width);
>+void BlendPlaneRow_C(const uint8_t* src0,
>+                     const uint8_t* src1,
>+                     const uint8_t* alpha,
>+                     uint8_t* dst,
>                      int width);
> 
> // ARGB multiply images. Same API as Blend, but these require
> // pointer and width alignment for SSE2.
>-void ARGBMultiplyRow_C(const uint8* src_argb,
>-                       const uint8* src_argb1,
>-                       uint8* dst_argb,
>+void ARGBMultiplyRow_C(const uint8_t* src_argb0,
>+                       const uint8_t* src_argb1,
>+                       uint8_t* dst_argb,
>                        int width);
>-void ARGBMultiplyRow_SSE2(const uint8* src_argb,
>-                          const uint8* src_argb1,
>-                          uint8* dst_argb,
>+void ARGBMultiplyRow_SSE2(const uint8_t* src_argb0,
>+                          const uint8_t* src_argb1,
>+                          uint8_t* dst_argb,
>                           int width);
>-void ARGBMultiplyRow_Any_SSE2(const uint8* src_argb,
>-                              const uint8* src_argb1,
>-                              uint8* dst_argb,
>+void ARGBMultiplyRow_Any_SSE2(const uint8_t* y_buf,
>+                              const uint8_t* uv_buf,
>+                              uint8_t* dst_ptr,
>                               int width);
>-void ARGBMultiplyRow_AVX2(const uint8* src_argb,
>-                          const uint8* src_argb1,
>-                          uint8* dst_argb,
>+void ARGBMultiplyRow_AVX2(const uint8_t* src_argb0,
>+                          const uint8_t* src_argb1,
>+                          uint8_t* dst_argb,
>                           int width);
>-void ARGBMultiplyRow_Any_AVX2(const uint8* src_argb,
>-                              const uint8* src_argb1,
>-                              uint8* dst_argb,
>+void ARGBMultiplyRow_Any_AVX2(const uint8_t* y_buf,
>+                              const uint8_t* uv_buf,
>+                              uint8_t* dst_ptr,
>                               int width);
>-void ARGBMultiplyRow_NEON(const uint8* src_argb,
>-                          const uint8* src_argb1,
>-                          uint8* dst_argb,
>+void ARGBMultiplyRow_NEON(const uint8_t* src_argb0,
>+                          const uint8_t* src_argb1,
>+                          uint8_t* dst_argb,
>                           int width);
>-void ARGBMultiplyRow_Any_NEON(const uint8* src_argb,
>-                              const uint8* src_argb1,
>-                              uint8* dst_argb,
>+void ARGBMultiplyRow_Any_NEON(const uint8_t* y_buf,
>+                              const uint8_t* uv_buf,
>+                              uint8_t* dst_ptr,
>                               int width);
>-void ARGBMultiplyRow_MSA(const uint8* src_argb,
>-                         const uint8* src_argb1,
>-                         uint8* dst_argb,
>+void ARGBMultiplyRow_MSA(const uint8_t* src_argb0,
>+                         const uint8_t* src_argb1,
>+                         uint8_t* dst_argb,
>                          int width);
>-void ARGBMultiplyRow_Any_MSA(const uint8* src_argb,
>-                             const uint8* src_argb1,
>-                             uint8* dst_argb,
>+void ARGBMultiplyRow_Any_MSA(const uint8_t* y_buf,
>+                             const uint8_t* uv_buf,
>+                             uint8_t* dst_ptr,
>                              int width);
> 
> // ARGB add images.
>-void ARGBAddRow_C(const uint8* src_argb,
>-                  const uint8* src_argb1,
>-                  uint8* dst_argb,
>+void ARGBAddRow_C(const uint8_t* src_argb0,
>+                  const uint8_t* src_argb1,
>+                  uint8_t* dst_argb,
>                   int width);
>-void ARGBAddRow_SSE2(const uint8* src_argb,
>-                     const uint8* src_argb1,
>-                     uint8* dst_argb,
>+void ARGBAddRow_SSE2(const uint8_t* src_argb0,
>+                     const uint8_t* src_argb1,
>+                     uint8_t* dst_argb,
>                      int width);
>-void ARGBAddRow_Any_SSE2(const uint8* src_argb,
>-                         const uint8* src_argb1,
>-                         uint8* dst_argb,
>+void ARGBAddRow_Any_SSE2(const uint8_t* y_buf,
>+                         const uint8_t* uv_buf,
>+                         uint8_t* dst_ptr,
>                          int width);
>-void ARGBAddRow_AVX2(const uint8* src_argb,
>-                     const uint8* src_argb1,
>-                     uint8* dst_argb,
>+void ARGBAddRow_AVX2(const uint8_t* src_argb0,
>+                     const uint8_t* src_argb1,
>+                     uint8_t* dst_argb,
>                      int width);
>-void ARGBAddRow_Any_AVX2(const uint8* src_argb,
>-                         const uint8* src_argb1,
>-                         uint8* dst_argb,
>+void ARGBAddRow_Any_AVX2(const uint8_t* y_buf,
>+                         const uint8_t* uv_buf,
>+                         uint8_t* dst_ptr,
>                          int width);
>-void ARGBAddRow_NEON(const uint8* src_argb,
>-                     const uint8* src_argb1,
>-                     uint8* dst_argb,
>+void ARGBAddRow_NEON(const uint8_t* src_argb0,
>+                     const uint8_t* src_argb1,
>+                     uint8_t* dst_argb,
>                      int width);
>-void ARGBAddRow_Any_NEON(const uint8* src_argb,
>-                         const uint8* src_argb1,
>-                         uint8* dst_argb,
>+void ARGBAddRow_Any_NEON(const uint8_t* y_buf,
>+                         const uint8_t* uv_buf,
>+                         uint8_t* dst_ptr,
>                          int width);
>-void ARGBAddRow_MSA(const uint8* src_argb,
>-                    const uint8* src_argb1,
>-                    uint8* dst_argb,
>+void ARGBAddRow_MSA(const uint8_t* src_argb0,
>+                    const uint8_t* src_argb1,
>+                    uint8_t* dst_argb,
>                     int width);
>-void ARGBAddRow_Any_MSA(const uint8* src_argb,
>-                        const uint8* src_argb1,
>-                        uint8* dst_argb,
>+void ARGBAddRow_Any_MSA(const uint8_t* y_buf,
>+                        const uint8_t* uv_buf,
>+                        uint8_t* dst_ptr,
>                         int width);
> 
> // ARGB subtract images. Same API as Blend, but these require
> // pointer and width alignment for SSE2.
>-void ARGBSubtractRow_C(const uint8* src_argb,
>-                       const uint8* src_argb1,
>-                       uint8* dst_argb,
>+void ARGBSubtractRow_C(const uint8_t* src_argb0,
>+                       const uint8_t* src_argb1,
>+                       uint8_t* dst_argb,
>                        int width);
>-void ARGBSubtractRow_SSE2(const uint8* src_argb,
>-                          const uint8* src_argb1,
>-                          uint8* dst_argb,
>+void ARGBSubtractRow_SSE2(const uint8_t* src_argb0,
>+                          const uint8_t* src_argb1,
>+                          uint8_t* dst_argb,
>                           int width);
>-void ARGBSubtractRow_Any_SSE2(const uint8* src_argb,
>-                              const uint8* src_argb1,
>-                              uint8* dst_argb,
>+void ARGBSubtractRow_Any_SSE2(const uint8_t* y_buf,
>+                              const uint8_t* uv_buf,
>+                              uint8_t* dst_ptr,
>                               int width);
>-void ARGBSubtractRow_AVX2(const uint8* src_argb,
>-                          const uint8* src_argb1,
>-                          uint8* dst_argb,
>+void ARGBSubtractRow_AVX2(const uint8_t* src_argb0,
>+                          const uint8_t* src_argb1,
>+                          uint8_t* dst_argb,
>                           int width);
>-void ARGBSubtractRow_Any_AVX2(const uint8* src_argb,
>-                              const uint8* src_argb1,
>-                              uint8* dst_argb,
>+void ARGBSubtractRow_Any_AVX2(const uint8_t* y_buf,
>+                              const uint8_t* uv_buf,
>+                              uint8_t* dst_ptr,
>                               int width);
>-void ARGBSubtractRow_NEON(const uint8* src_argb,
>-                          const uint8* src_argb1,
>-                          uint8* dst_argb,
>+void ARGBSubtractRow_NEON(const uint8_t* src_argb0,
>+                          const uint8_t* src_argb1,
>+                          uint8_t* dst_argb,
>                           int width);
>-void ARGBSubtractRow_Any_NEON(const uint8* src_argb,
>-                              const uint8* src_argb1,
>-                              uint8* dst_argb,
>+void ARGBSubtractRow_Any_NEON(const uint8_t* y_buf,
>+                              const uint8_t* uv_buf,
>+                              uint8_t* dst_ptr,
>                               int width);
>-void ARGBSubtractRow_MSA(const uint8* src_argb,
>-                         const uint8* src_argb1,
>-                         uint8* dst_argb,
>+void ARGBSubtractRow_MSA(const uint8_t* src_argb0,
>+                         const uint8_t* src_argb1,
>+                         uint8_t* dst_argb,
>                          int width);
>-void ARGBSubtractRow_Any_MSA(const uint8* src_argb,
>-                             const uint8* src_argb1,
>-                             uint8* dst_argb,
>+void ARGBSubtractRow_Any_MSA(const uint8_t* y_buf,
>+                             const uint8_t* uv_buf,
>+                             uint8_t* dst_ptr,
>                              int width);
> 
>-void ARGBToRGB24Row_Any_SSSE3(const uint8* src_argb, uint8* dst_rgb, int width);
>-void ARGBToRAWRow_Any_SSSE3(const uint8* src_argb, uint8* dst_rgb, int width);
>-void ARGBToRGB565Row_Any_SSE2(const uint8* src_argb, uint8* dst_rgb, int width);
>-void ARGBToARGB1555Row_Any_SSE2(const uint8* src_argb,
>-                                uint8* dst_rgb,
>+void ARGBToRGB24Row_Any_SSSE3(const uint8_t* src_ptr,
>+                              uint8_t* dst_ptr,
>+                              int width);
>+void ARGBToRAWRow_Any_SSSE3(const uint8_t* src_ptr,
>+                            uint8_t* dst_ptr,
>+                            int width);
>+void ARGBToRGB565Row_Any_SSE2(const uint8_t* src_ptr,
>+                              uint8_t* dst_ptr,
>+                              int width);
>+void ARGBToARGB1555Row_Any_SSE2(const uint8_t* src_ptr,
>+                                uint8_t* dst_ptr,
>                                 int width);
>-void ARGBToARGB4444Row_Any_SSE2(const uint8* src_argb,
>-                                uint8* dst_rgb,
>+void ARGBToARGB4444Row_Any_SSE2(const uint8_t* src_ptr,
>+                                uint8_t* dst_ptr,
>                                 int width);
>+void ABGRToAR30Row_Any_SSSE3(const uint8_t* src_ptr,
>+                             uint8_t* dst_ptr,
>+                             int width);
>+void ARGBToAR30Row_Any_SSSE3(const uint8_t* src_ptr,
>+                             uint8_t* dst_ptr,
>+                             int width);
> 
>-void ARGBToRGB565DitherRow_Any_SSE2(const uint8* src_argb,
>-                                    uint8* dst_rgb,
>-                                    const uint32 dither4,
>+void ARGBToRGB565DitherRow_Any_SSE2(const uint8_t* src_ptr,
>+                                    uint8_t* dst_ptr,
>+                                    const uint32_t param,
>                                     int width);
>-void ARGBToRGB565DitherRow_Any_AVX2(const uint8* src_argb,
>-                                    uint8* dst_rgb,
>-                                    const uint32 dither4,
>+void ARGBToRGB565DitherRow_Any_AVX2(const uint8_t* src_ptr,
>+                                    uint8_t* dst_ptr,
>+                                    const uint32_t param,
>                                     int width);
> 
>-void ARGBToRGB565Row_Any_AVX2(const uint8* src_argb, uint8* dst_rgb, int width);
>-void ARGBToARGB1555Row_Any_AVX2(const uint8* src_argb,
>-                                uint8* dst_rgb,
>+void ARGBToRGB565Row_Any_AVX2(const uint8_t* src_ptr,
>+                              uint8_t* dst_ptr,
>+                              int width);
>+void ARGBToARGB1555Row_Any_AVX2(const uint8_t* src_ptr,
>+                                uint8_t* dst_ptr,
>                                 int width);
>-void ARGBToARGB4444Row_Any_AVX2(const uint8* src_argb,
>-                                uint8* dst_rgb,
>+void ARGBToARGB4444Row_Any_AVX2(const uint8_t* src_ptr,
>+                                uint8_t* dst_ptr,
>                                 int width);
>+void ABGRToAR30Row_Any_AVX2(const uint8_t* src_ptr,
>+                            uint8_t* dst_ptr,
>+                            int width);
>+void ARGBToAR30Row_Any_AVX2(const uint8_t* src_ptr,
>+                            uint8_t* dst_ptr,
>+                            int width);
> 
>-void ARGBToRGB24Row_Any_NEON(const uint8* src_argb, uint8* dst_rgb, int width);
>-void ARGBToRAWRow_Any_NEON(const uint8* src_argb, uint8* dst_rgb, int width);
>-void ARGBToRGB565Row_Any_NEON(const uint8* src_argb, uint8* dst_rgb, int width);
>-void ARGBToARGB1555Row_Any_NEON(const uint8* src_argb,
>-                                uint8* dst_rgb,
>+void ARGBToRGB24Row_Any_NEON(const uint8_t* src_ptr,
>+                             uint8_t* dst_ptr,
>+                             int width);
>+void ARGBToRAWRow_Any_NEON(const uint8_t* src_ptr, uint8_t* dst_ptr, int width);
>+void ARGBToRGB565Row_Any_NEON(const uint8_t* src_ptr,
>+                              uint8_t* dst_ptr,
>+                              int width);
>+void ARGBToARGB1555Row_Any_NEON(const uint8_t* src_ptr,
>+                                uint8_t* dst_ptr,
>                                 int width);
>-void ARGBToARGB4444Row_Any_NEON(const uint8* src_argb,
>-                                uint8* dst_rgb,
>+void ARGBToARGB4444Row_Any_NEON(const uint8_t* src_ptr,
>+                                uint8_t* dst_ptr,
>                                 int width);
>-void ARGBToRGB565DitherRow_Any_NEON(const uint8* src_argb,
>-                                    uint8* dst_rgb,
>-                                    const uint32 dither4,
>+void ARGBToRGB565DitherRow_Any_NEON(const uint8_t* src_ptr,
>+                                    uint8_t* dst_ptr,
>+                                    const uint32_t param,
>                                     int width);
>-void ARGBToRGB24Row_Any_MSA(const uint8* src_argb, uint8* dst_rgb, int width);
>-void ARGBToRAWRow_Any_MSA(const uint8* src_argb, uint8* dst_rgb, int width);
>-void ARGBToRGB565Row_Any_MSA(const uint8* src_argb, uint8* dst_rgb, int width);
>-void ARGBToARGB1555Row_Any_MSA(const uint8* src_argb,
>-                               uint8* dst_rgb,
>+void ARGBToRGB24Row_Any_MSA(const uint8_t* src_ptr,
>+                            uint8_t* dst_ptr,
>+                            int width);
>+void ARGBToRAWRow_Any_MSA(const uint8_t* src_ptr, uint8_t* dst_ptr, int width);
>+void ARGBToRGB565Row_Any_MSA(const uint8_t* src_ptr,
>+                             uint8_t* dst_ptr,
>+                             int width);
>+void ARGBToARGB1555Row_Any_MSA(const uint8_t* src_ptr,
>+                               uint8_t* dst_ptr,
>                                int width);
>-void ARGBToARGB4444Row_Any_MSA(const uint8* src_argb,
>-                               uint8* dst_rgb,
>+void ARGBToARGB4444Row_Any_MSA(const uint8_t* src_ptr,
>+                               uint8_t* dst_ptr,
>                                int width);
>-void ARGBToRGB565DitherRow_Any_MSA(const uint8* src_argb,
>-                                   uint8* dst_rgb,
>-                                   const uint32 dither4,
>+void ARGBToRGB565DitherRow_Any_MSA(const uint8_t* src_ptr,
>+                                   uint8_t* dst_ptr,
>+                                   const uint32_t param,
>                                    int width);
> 
>-void I444ToARGBRow_Any_NEON(const uint8* src_y,
>-                            const uint8* src_u,
>-                            const uint8* src_v,
>-                            uint8* dst_argb,
>+void I444ToARGBRow_Any_NEON(const uint8_t* y_buf,
>+                            const uint8_t* u_buf,
>+                            const uint8_t* v_buf,
>+                            uint8_t* dst_ptr,
>                             const struct YuvConstants* yuvconstants,
>                             int width);
>-void I422ToARGBRow_Any_NEON(const uint8* src_y,
>-                            const uint8* src_u,
>-                            const uint8* src_v,
>-                            uint8* dst_argb,
>+void I422ToARGBRow_Any_NEON(const uint8_t* y_buf,
>+                            const uint8_t* u_buf,
>+                            const uint8_t* v_buf,
>+                            uint8_t* dst_ptr,
>                             const struct YuvConstants* yuvconstants,
>                             int width);
>-void I422AlphaToARGBRow_Any_NEON(const uint8* src_y,
>-                                 const uint8* src_u,
>-                                 const uint8* src_v,
>-                                 const uint8* src_a,
>-                                 uint8* dst_argb,
>+void I422AlphaToARGBRow_Any_NEON(const uint8_t* y_buf,
>+                                 const uint8_t* u_buf,
>+                                 const uint8_t* v_buf,
>+                                 const uint8_t* a_buf,
>+                                 uint8_t* dst_ptr,
>                                  const struct YuvConstants* yuvconstants,
>                                  int width);
>-void I422ToRGBARow_Any_NEON(const uint8* src_y,
>-                            const uint8* src_u,
>-                            const uint8* src_v,
>-                            uint8* dst_argb,
>+void I422ToRGBARow_Any_NEON(const uint8_t* y_buf,
>+                            const uint8_t* u_buf,
>+                            const uint8_t* v_buf,
>+                            uint8_t* dst_ptr,
>                             const struct YuvConstants* yuvconstants,
>                             int width);
>-void I422ToRGB24Row_Any_NEON(const uint8* src_y,
>-                             const uint8* src_u,
>-                             const uint8* src_v,
>-                             uint8* dst_argb,
>+void I422ToRGB24Row_Any_NEON(const uint8_t* y_buf,
>+                             const uint8_t* u_buf,
>+                             const uint8_t* v_buf,
>+                             uint8_t* dst_ptr,
>                              const struct YuvConstants* yuvconstants,
>                              int width);
>-void I422ToARGB4444Row_Any_NEON(const uint8* src_y,
>-                                const uint8* src_u,
>-                                const uint8* src_v,
>-                                uint8* dst_argb,
>+void I422ToARGB4444Row_Any_NEON(const uint8_t* y_buf,
>+                                const uint8_t* u_buf,
>+                                const uint8_t* v_buf,
>+                                uint8_t* dst_ptr,
>                                 const struct YuvConstants* yuvconstants,
>                                 int width);
>-void I422ToARGB1555Row_Any_NEON(const uint8* src_y,
>-                                const uint8* src_u,
>-                                const uint8* src_v,
>-                                uint8* dst_argb,
>+void I422ToARGB1555Row_Any_NEON(const uint8_t* y_buf,
>+                                const uint8_t* u_buf,
>+                                const uint8_t* v_buf,
>+                                uint8_t* dst_ptr,
>                                 const struct YuvConstants* yuvconstants,
>                                 int width);
>-void I422ToRGB565Row_Any_NEON(const uint8* src_y,
>-                              const uint8* src_u,
>-                              const uint8* src_v,
>-                              uint8* dst_argb,
>+void I422ToRGB565Row_Any_NEON(const uint8_t* y_buf,
>+                              const uint8_t* u_buf,
>+                              const uint8_t* v_buf,
>+                              uint8_t* dst_ptr,
>                               const struct YuvConstants* yuvconstants,
>                               int width);
>-void NV12ToARGBRow_Any_NEON(const uint8* src_y,
>-                            const uint8* src_uv,
>-                            uint8* dst_argb,
>+void NV12ToARGBRow_Any_NEON(const uint8_t* y_buf,
>+                            const uint8_t* uv_buf,
>+                            uint8_t* dst_ptr,
>                             const struct YuvConstants* yuvconstants,
>                             int width);
>-void NV21ToARGBRow_Any_NEON(const uint8* src_y,
>-                            const uint8* src_vu,
>-                            uint8* dst_argb,
>+void NV21ToARGBRow_Any_NEON(const uint8_t* y_buf,
>+                            const uint8_t* uv_buf,
>+                            uint8_t* dst_ptr,
>                             const struct YuvConstants* yuvconstants,
>                             int width);
>-void NV12ToRGB565Row_Any_NEON(const uint8* src_y,
>-                              const uint8* src_uv,
>-                              uint8* dst_argb,
>+void NV12ToRGB24Row_Any_NEON(const uint8_t* y_buf,
>+                             const uint8_t* uv_buf,
>+                             uint8_t* dst_ptr,
>+                             const struct YuvConstants* yuvconstants,
>+                             int width);
>+void NV21ToRGB24Row_Any_NEON(const uint8_t* y_buf,
>+                             const uint8_t* uv_buf,
>+                             uint8_t* dst_ptr,
>+                             const struct YuvConstants* yuvconstants,
>+                             int width);
>+void NV12ToRGB565Row_Any_NEON(const uint8_t* y_buf,
>+                              const uint8_t* uv_buf,
>+                              uint8_t* dst_ptr,
>                               const struct YuvConstants* yuvconstants,
>                               int width);
>-void YUY2ToARGBRow_Any_NEON(const uint8* src_yuy2,
>-                            uint8* dst_argb,
>+void YUY2ToARGBRow_Any_NEON(const uint8_t* src_ptr,
>+                            uint8_t* dst_ptr,
>                             const struct YuvConstants* yuvconstants,
>                             int width);
>-void UYVYToARGBRow_Any_NEON(const uint8* src_uyvy,
>-                            uint8* dst_argb,
>+void UYVYToARGBRow_Any_NEON(const uint8_t* src_ptr,
>+                            uint8_t* dst_ptr,
>                             const struct YuvConstants* yuvconstants,
>                             int width);
>-void I444ToARGBRow_Any_DSPR2(const uint8* src_y,
>-                             const uint8* src_u,
>-                             const uint8* src_v,
>-                             uint8* dst_argb,
>-                             const struct YuvConstants* yuvconstants,
>-                             int width);
>-void I422ToARGB4444Row_Any_DSPR2(const uint8* src_y,
>-                                 const uint8* src_u,
>-                                 const uint8* src_v,
>-                                 uint8* dst_argb,
>-                                 const struct YuvConstants* yuvconstants,
>-                                 int width);
>-void I422ToARGBRow_Any_DSPR2(const uint8* src_y,
>-                             const uint8* src_u,
>-                             const uint8* src_v,
>-                             uint8* dst_argb,
>-                             const struct YuvConstants* yuvconstants,
>-                             int width);
>-void I422ToARGBRow_DSPR2(const uint8* src_y,
>-                         const uint8* src_u,
>-                         const uint8* src_v,
>-                         uint8* dst_argb,
>-                         const struct YuvConstants* yuvconstants,
>-                         int width);
>-void I422ToARGB1555Row_Any_DSPR2(const uint8* src_y,
>-                                 const uint8* src_u,
>-                                 const uint8* src_v,
>-                                 uint8* dst_argb,
>-                                 const struct YuvConstants* yuvconstants,
>-                                 int width);
>-void I411ToARGBRow_Any_DSPR2(const uint8* src_y,
>-                             const uint8* src_u,
>-                             const uint8* src_v,
>-                             uint8* dst_argb,
>-                             const struct YuvConstants* yuvconstants,
>-                             int width);
>-void NV12ToARGBRow_Any_DSPR2(const uint8* src_y,
>-                             const uint8* src_uv,
>-                             uint8* dst_argb,
>-                             const struct YuvConstants* yuvconstants,
>-                             int width);
>-void I422ToARGBRow_DSPR2(const uint8* src_y,
>-                         const uint8* src_u,
>-                         const uint8* src_v,
>-                         uint8* dst_argb,
>-                         const struct YuvConstants* yuvconstants,
>-                         int width);
>-void I444ToARGBRow_Any_MSA(const uint8* src_y,
>-                           const uint8* src_u,
>-                           const uint8* src_v,
>-                           uint8* dst_argb,
>+void I444ToARGBRow_Any_MSA(const uint8_t* y_buf,
>+                           const uint8_t* u_buf,
>+                           const uint8_t* v_buf,
>+                           uint8_t* dst_ptr,
>                            const struct YuvConstants* yuvconstants,
>                            int width);
>-void I422ToARGBRow_Any_MSA(const uint8* src_y,
>-                           const uint8* src_u,
>-                           const uint8* src_v,
>-                           uint8* dst_argb,
>+void I422ToARGBRow_Any_MSA(const uint8_t* y_buf,
>+                           const uint8_t* u_buf,
>+                           const uint8_t* v_buf,
>+                           uint8_t* dst_ptr,
>                            const struct YuvConstants* yuvconstants,
>                            int width);
>-void I422ToRGBARow_Any_MSA(const uint8* src_y,
>-                           const uint8* src_u,
>-                           const uint8* src_v,
>-                           uint8* dst_argb,
>+void I422ToRGBARow_Any_MSA(const uint8_t* y_buf,
>+                           const uint8_t* u_buf,
>+                           const uint8_t* v_buf,
>+                           uint8_t* dst_ptr,
>                            const struct YuvConstants* yuvconstants,
>                            int width);
>-void I422AlphaToARGBRow_Any_MSA(const uint8* src_y,
>-                                const uint8* src_u,
>-                                const uint8* src_v,
>-                                const uint8* src_a,
>-                                uint8* dst_argb,
>+void I422AlphaToARGBRow_Any_MSA(const uint8_t* y_buf,
>+                                const uint8_t* u_buf,
>+                                const uint8_t* v_buf,
>+                                const uint8_t* a_buf,
>+                                uint8_t* dst_ptr,
>                                 const struct YuvConstants* yuvconstants,
>                                 int width);
>-void I422ToRGB24Row_Any_MSA(const uint8* src_y,
>-                            const uint8* src_u,
>-                            const uint8* src_v,
>-                            uint8* dst_rgb24,
>+void I422ToRGB24Row_Any_MSA(const uint8_t* y_buf,
>+                            const uint8_t* u_buf,
>+                            const uint8_t* v_buf,
>+                            uint8_t* dst_ptr,
>                             const struct YuvConstants* yuvconstants,
>                             int width);
>-void I422ToRGB565Row_Any_MSA(const uint8* src_y,
>-                             const uint8* src_u,
>-                             const uint8* src_v,
>-                             uint8* dst_rgb565,
>+void I422ToRGB565Row_Any_MSA(const uint8_t* y_buf,
>+                             const uint8_t* u_buf,
>+                             const uint8_t* v_buf,
>+                             uint8_t* dst_ptr,
>                              const struct YuvConstants* yuvconstants,
>                              int width);
>-void I422ToARGB4444Row_Any_MSA(const uint8* src_y,
>-                               const uint8* src_u,
>-                               const uint8* src_v,
>-                               uint8* dst_argb4444,
>+void I422ToARGB4444Row_Any_MSA(const uint8_t* y_buf,
>+                               const uint8_t* u_buf,
>+                               const uint8_t* v_buf,
>+                               uint8_t* dst_ptr,
>                                const struct YuvConstants* yuvconstants,
>                                int width);
>-void I422ToARGB1555Row_Any_MSA(const uint8* src_y,
>-                               const uint8* src_u,
>-                               const uint8* src_v,
>-                               uint8* dst_argb1555,
>+void I422ToARGB1555Row_Any_MSA(const uint8_t* y_buf,
>+                               const uint8_t* u_buf,
>+                               const uint8_t* v_buf,
>+                               uint8_t* dst_ptr,
>                                const struct YuvConstants* yuvconstants,
>                                int width);
>-void NV12ToARGBRow_Any_MSA(const uint8* src_y,
>-                           const uint8* src_uv,
>-                           uint8* dst_argb,
>+void NV12ToARGBRow_Any_MSA(const uint8_t* y_buf,
>+                           const uint8_t* uv_buf,
>+                           uint8_t* dst_ptr,
>                            const struct YuvConstants* yuvconstants,
>                            int width);
>-void NV12ToRGB565Row_Any_MSA(const uint8* src_y,
>-                             const uint8* src_uv,
>-                             uint8* dst_argb,
>+void NV12ToRGB565Row_Any_MSA(const uint8_t* y_buf,
>+                             const uint8_t* uv_buf,
>+                             uint8_t* dst_ptr,
>                              const struct YuvConstants* yuvconstants,
>                              int width);
>-void NV21ToARGBRow_Any_MSA(const uint8* src_y,
>-                           const uint8* src_vu,
>-                           uint8* dst_argb,
>+void NV21ToARGBRow_Any_MSA(const uint8_t* y_buf,
>+                           const uint8_t* uv_buf,
>+                           uint8_t* dst_ptr,
>                            const struct YuvConstants* yuvconstants,
>                            int width);
>-void YUY2ToARGBRow_Any_MSA(const uint8* src_yuy2,
>-                           uint8* dst_argb,
>+void YUY2ToARGBRow_Any_MSA(const uint8_t* src_ptr,
>+                           uint8_t* dst_ptr,
>                            const struct YuvConstants* yuvconstants,
>                            int width);
>-void UYVYToARGBRow_Any_MSA(const uint8* src_uyvy,
>-                           uint8* dst_argb,
>+void UYVYToARGBRow_Any_MSA(const uint8_t* src_ptr,
>+                           uint8_t* dst_ptr,
>                            const struct YuvConstants* yuvconstants,
>                            int width);
> 
>-void YUY2ToYRow_AVX2(const uint8* src_yuy2, uint8* dst_y, int width);
>-void YUY2ToUVRow_AVX2(const uint8* src_yuy2,
>+void YUY2ToYRow_AVX2(const uint8_t* src_yuy2, uint8_t* dst_y, int width);
>+void YUY2ToUVRow_AVX2(const uint8_t* src_yuy2,
>                       int stride_yuy2,
>-                      uint8* dst_u,
>-                      uint8* dst_v,
>+                      uint8_t* dst_u,
>+                      uint8_t* dst_v,
>                       int width);
>-void YUY2ToUV422Row_AVX2(const uint8* src_yuy2,
>-                         uint8* dst_u,
>-                         uint8* dst_v,
>+void YUY2ToUV422Row_AVX2(const uint8_t* src_yuy2,
>+                         uint8_t* dst_u,
>+                         uint8_t* dst_v,
>                          int width);
>-void YUY2ToYRow_SSE2(const uint8* src_yuy2, uint8* dst_y, int width);
>-void YUY2ToUVRow_SSE2(const uint8* src_yuy2,
>+void YUY2ToYRow_SSE2(const uint8_t* src_yuy2, uint8_t* dst_y, int width);
>+void YUY2ToUVRow_SSE2(const uint8_t* src_yuy2,
>                       int stride_yuy2,
>-                      uint8* dst_u,
>-                      uint8* dst_v,
>+                      uint8_t* dst_u,
>+                      uint8_t* dst_v,
>                       int width);
>-void YUY2ToUV422Row_SSE2(const uint8* src_yuy2,
>-                         uint8* dst_u,
>-                         uint8* dst_v,
>+void YUY2ToUV422Row_SSE2(const uint8_t* src_yuy2,
>+                         uint8_t* dst_u,
>+                         uint8_t* dst_v,
>                          int width);
>-void YUY2ToYRow_NEON(const uint8* src_yuy2, uint8* dst_y, int width);
>-void YUY2ToUVRow_NEON(const uint8* src_yuy2,
>+void YUY2ToYRow_NEON(const uint8_t* src_yuy2, uint8_t* dst_y, int width);
>+void YUY2ToUVRow_NEON(const uint8_t* src_yuy2,
>                       int stride_yuy2,
>-                      uint8* dst_u,
>-                      uint8* dst_v,
>+                      uint8_t* dst_u,
>+                      uint8_t* dst_v,
>                       int width);
>-void YUY2ToUV422Row_NEON(const uint8* src_yuy2,
>-                         uint8* dst_u,
>-                         uint8* dst_v,
>-                         int width);
>-void YUY2ToYRow_MSA(const uint8* src_yuy2, uint8* dst_y, int width);
>-void YUY2ToUVRow_MSA(const uint8* src_yuy2,
>-                     int stride_yuy2,
>-                     uint8* dst_u,
>-                     uint8* dst_v,
>+void YUY2ToUV422Row_NEON(const uint8_t* src_yuy2,
>+                         uint8_t* dst_u,
>+                         uint8_t* dst_v,
>+                         int width);
>+void YUY2ToYRow_MSA(const uint8_t* src_yuy2, uint8_t* dst_y, int width);
>+void YUY2ToUVRow_MSA(const uint8_t* src_yuy2,
>+                     int src_stride_yuy2,
>+                     uint8_t* dst_u,
>+                     uint8_t* dst_v,
>                      int width);
>-void YUY2ToUV422Row_MSA(const uint8* src_yuy2,
>-                        uint8* dst_u,
>-                        uint8* dst_v,
>+void YUY2ToUV422Row_MSA(const uint8_t* src_yuy2,
>+                        uint8_t* dst_u,
>+                        uint8_t* dst_v,
>                         int width);
>-void YUY2ToYRow_C(const uint8* src_yuy2, uint8* dst_y, int width);
>-void YUY2ToUVRow_C(const uint8* src_yuy2,
>-                   int stride_yuy2,
>-                   uint8* dst_u,
>-                   uint8* dst_v,
>+void YUY2ToYRow_C(const uint8_t* src_yuy2, uint8_t* dst_y, int width);
>+void YUY2ToUVRow_C(const uint8_t* src_yuy2,
>+                   int src_stride_yuy2,
>+                   uint8_t* dst_u,
>+                   uint8_t* dst_v,
>                    int width);
>-void YUY2ToUV422Row_C(const uint8* src_yuy2,
>-                      uint8* dst_u,
>-                      uint8* dst_v,
>+void YUY2ToUV422Row_C(const uint8_t* src_yuy2,
>+                      uint8_t* dst_u,
>+                      uint8_t* dst_v,
>                       int width);
>-void YUY2ToYRow_Any_AVX2(const uint8* src_yuy2, uint8* dst_y, int width);
>-void YUY2ToUVRow_Any_AVX2(const uint8* src_yuy2,
>-                          int stride_yuy2,
>-                          uint8* dst_u,
>-                          uint8* dst_v,
>+void YUY2ToYRow_Any_AVX2(const uint8_t* src_ptr, uint8_t* dst_ptr, int width);
>+void YUY2ToUVRow_Any_AVX2(const uint8_t* src_ptr,
>+                          int src_stride_ptr,
>+                          uint8_t* dst_u,
>+                          uint8_t* dst_v,
>                           int width);
>-void YUY2ToUV422Row_Any_AVX2(const uint8* src_yuy2,
>-                             uint8* dst_u,
>-                             uint8* dst_v,
>+void YUY2ToUV422Row_Any_AVX2(const uint8_t* src_ptr,
>+                             uint8_t* dst_u,
>+                             uint8_t* dst_v,
>                              int width);
>-void YUY2ToYRow_Any_SSE2(const uint8* src_yuy2, uint8* dst_y, int width);
>-void YUY2ToUVRow_Any_SSE2(const uint8* src_yuy2,
>-                          int stride_yuy2,
>-                          uint8* dst_u,
>-                          uint8* dst_v,
>+void YUY2ToYRow_Any_SSE2(const uint8_t* src_ptr, uint8_t* dst_ptr, int width);
>+void YUY2ToUVRow_Any_SSE2(const uint8_t* src_ptr,
>+                          int src_stride_ptr,
>+                          uint8_t* dst_u,
>+                          uint8_t* dst_v,
>                           int width);
>-void YUY2ToUV422Row_Any_SSE2(const uint8* src_yuy2,
>-                             uint8* dst_u,
>-                             uint8* dst_v,
>+void YUY2ToUV422Row_Any_SSE2(const uint8_t* src_ptr,
>+                             uint8_t* dst_u,
>+                             uint8_t* dst_v,
>                              int width);
>-void YUY2ToYRow_Any_NEON(const uint8* src_yuy2, uint8* dst_y, int width);
>-void YUY2ToUVRow_Any_NEON(const uint8* src_yuy2,
>-                          int stride_yuy2,
>-                          uint8* dst_u,
>-                          uint8* dst_v,
>+void YUY2ToYRow_Any_NEON(const uint8_t* src_ptr, uint8_t* dst_ptr, int width);
>+void YUY2ToUVRow_Any_NEON(const uint8_t* src_ptr,
>+                          int src_stride_ptr,
>+                          uint8_t* dst_u,
>+                          uint8_t* dst_v,
>                           int width);
>-void YUY2ToUV422Row_Any_NEON(const uint8* src_yuy2,
>-                             uint8* dst_u,
>-                             uint8* dst_v,
>+void YUY2ToUV422Row_Any_NEON(const uint8_t* src_ptr,
>+                             uint8_t* dst_u,
>+                             uint8_t* dst_v,
>                              int width);
>-void YUY2ToYRow_Any_MSA(const uint8* src_yuy2, uint8* dst_y, int width);
>-void YUY2ToUVRow_Any_MSA(const uint8* src_yuy2,
>-                         int stride_yuy2,
>-                         uint8* dst_u,
>-                         uint8* dst_v,
>-                         int width);
>-void YUY2ToUV422Row_Any_MSA(const uint8* src_yuy2,
>-                            uint8* dst_u,
>-                            uint8* dst_v,
>-                            int width);
>-void UYVYToYRow_AVX2(const uint8* src_uyvy, uint8* dst_y, int width);
>-void UYVYToUVRow_AVX2(const uint8* src_uyvy,
>+void YUY2ToYRow_Any_MSA(const uint8_t* src_ptr, uint8_t* dst_ptr, int width);
>+void YUY2ToUVRow_Any_MSA(const uint8_t* src_ptr,
>+                         int src_stride_ptr,
>+                         uint8_t* dst_u,
>+                         uint8_t* dst_v,
>+                         int width);
>+void YUY2ToUV422Row_Any_MSA(const uint8_t* src_ptr,
>+                            uint8_t* dst_u,
>+                            uint8_t* dst_v,
>+                            int width);
>+void UYVYToYRow_AVX2(const uint8_t* src_uyvy, uint8_t* dst_y, int width);
>+void UYVYToUVRow_AVX2(const uint8_t* src_uyvy,
>                       int stride_uyvy,
>-                      uint8* dst_u,
>-                      uint8* dst_v,
>+                      uint8_t* dst_u,
>+                      uint8_t* dst_v,
>                       int width);
>-void UYVYToUV422Row_AVX2(const uint8* src_uyvy,
>-                         uint8* dst_u,
>-                         uint8* dst_v,
>+void UYVYToUV422Row_AVX2(const uint8_t* src_uyvy,
>+                         uint8_t* dst_u,
>+                         uint8_t* dst_v,
>                          int width);
>-void UYVYToYRow_SSE2(const uint8* src_uyvy, uint8* dst_y, int width);
>-void UYVYToUVRow_SSE2(const uint8* src_uyvy,
>+void UYVYToYRow_SSE2(const uint8_t* src_uyvy, uint8_t* dst_y, int width);
>+void UYVYToUVRow_SSE2(const uint8_t* src_uyvy,
>                       int stride_uyvy,
>-                      uint8* dst_u,
>-                      uint8* dst_v,
>+                      uint8_t* dst_u,
>+                      uint8_t* dst_v,
>                       int width);
>-void UYVYToUV422Row_SSE2(const uint8* src_uyvy,
>-                         uint8* dst_u,
>-                         uint8* dst_v,
>+void UYVYToUV422Row_SSE2(const uint8_t* src_uyvy,
>+                         uint8_t* dst_u,
>+                         uint8_t* dst_v,
>                          int width);
>-void UYVYToYRow_AVX2(const uint8* src_uyvy, uint8* dst_y, int width);
>-void UYVYToUVRow_AVX2(const uint8* src_uyvy,
>+void UYVYToYRow_AVX2(const uint8_t* src_uyvy, uint8_t* dst_y, int width);
>+void UYVYToUVRow_AVX2(const uint8_t* src_uyvy,
>                       int stride_uyvy,
>-                      uint8* dst_u,
>-                      uint8* dst_v,
>+                      uint8_t* dst_u,
>+                      uint8_t* dst_v,
>                       int width);
>-void UYVYToUV422Row_AVX2(const uint8* src_uyvy,
>-                         uint8* dst_u,
>-                         uint8* dst_v,
>+void UYVYToUV422Row_AVX2(const uint8_t* src_uyvy,
>+                         uint8_t* dst_u,
>+                         uint8_t* dst_v,
>                          int width);
>-void UYVYToYRow_NEON(const uint8* src_uyvy, uint8* dst_y, int width);
>-void UYVYToUVRow_NEON(const uint8* src_uyvy,
>+void UYVYToYRow_NEON(const uint8_t* src_uyvy, uint8_t* dst_y, int width);
>+void UYVYToUVRow_NEON(const uint8_t* src_uyvy,
>                       int stride_uyvy,
>-                      uint8* dst_u,
>-                      uint8* dst_v,
>+                      uint8_t* dst_u,
>+                      uint8_t* dst_v,
>                       int width);
>-void UYVYToUV422Row_NEON(const uint8* src_uyvy,
>-                         uint8* dst_u,
>-                         uint8* dst_v,
>-                         int width);
>-void UYVYToYRow_MSA(const uint8* src_uyvy, uint8* dst_y, int width);
>-void UYVYToUVRow_MSA(const uint8* src_uyvy,
>-                     int stride_uyvy,
>-                     uint8* dst_u,
>-                     uint8* dst_v,
>+void UYVYToUV422Row_NEON(const uint8_t* src_uyvy,
>+                         uint8_t* dst_u,
>+                         uint8_t* dst_v,
>+                         int width);
>+void UYVYToYRow_MSA(const uint8_t* src_uyvy, uint8_t* dst_y, int width);
>+void UYVYToUVRow_MSA(const uint8_t* src_uyvy,
>+                     int src_stride_uyvy,
>+                     uint8_t* dst_u,
>+                     uint8_t* dst_v,
>                      int width);
>-void UYVYToUV422Row_MSA(const uint8* src_uyvy,
>-                        uint8* dst_u,
>-                        uint8* dst_v,
>+void UYVYToUV422Row_MSA(const uint8_t* src_uyvy,
>+                        uint8_t* dst_u,
>+                        uint8_t* dst_v,
>                         int width);
> 
>-void UYVYToYRow_C(const uint8* src_uyvy, uint8* dst_y, int width);
>-void UYVYToUVRow_C(const uint8* src_uyvy,
>-                   int stride_uyvy,
>-                   uint8* dst_u,
>-                   uint8* dst_v,
>+void UYVYToYRow_C(const uint8_t* src_uyvy, uint8_t* dst_y, int width);
>+void UYVYToUVRow_C(const uint8_t* src_uyvy,
>+                   int src_stride_uyvy,
>+                   uint8_t* dst_u,
>+                   uint8_t* dst_v,
>                    int width);
>-void UYVYToUV422Row_C(const uint8* src_uyvy,
>-                      uint8* dst_u,
>-                      uint8* dst_v,
>+void UYVYToUV422Row_C(const uint8_t* src_uyvy,
>+                      uint8_t* dst_u,
>+                      uint8_t* dst_v,
>                       int width);
>-void UYVYToYRow_Any_AVX2(const uint8* src_uyvy, uint8* dst_y, int width);
>-void UYVYToUVRow_Any_AVX2(const uint8* src_uyvy,
>-                          int stride_uyvy,
>-                          uint8* dst_u,
>-                          uint8* dst_v,
>+void UYVYToYRow_Any_AVX2(const uint8_t* src_ptr, uint8_t* dst_ptr, int width);
>+void UYVYToUVRow_Any_AVX2(const uint8_t* src_ptr,
>+                          int src_stride_ptr,
>+                          uint8_t* dst_u,
>+                          uint8_t* dst_v,
>                           int width);
>-void UYVYToUV422Row_Any_AVX2(const uint8* src_uyvy,
>-                             uint8* dst_u,
>-                             uint8* dst_v,
>+void UYVYToUV422Row_Any_AVX2(const uint8_t* src_ptr,
>+                             uint8_t* dst_u,
>+                             uint8_t* dst_v,
>                              int width);
>-void UYVYToYRow_Any_SSE2(const uint8* src_uyvy, uint8* dst_y, int width);
>-void UYVYToUVRow_Any_SSE2(const uint8* src_uyvy,
>-                          int stride_uyvy,
>-                          uint8* dst_u,
>-                          uint8* dst_v,
>+void UYVYToYRow_Any_SSE2(const uint8_t* src_ptr, uint8_t* dst_ptr, int width);
>+void UYVYToUVRow_Any_SSE2(const uint8_t* src_ptr,
>+                          int src_stride_ptr,
>+                          uint8_t* dst_u,
>+                          uint8_t* dst_v,
>                           int width);
>-void UYVYToUV422Row_Any_SSE2(const uint8* src_uyvy,
>-                             uint8* dst_u,
>-                             uint8* dst_v,
>+void UYVYToUV422Row_Any_SSE2(const uint8_t* src_ptr,
>+                             uint8_t* dst_u,
>+                             uint8_t* dst_v,
>                              int width);
>-void UYVYToYRow_Any_NEON(const uint8* src_uyvy, uint8* dst_y, int width);
>-void UYVYToUVRow_Any_NEON(const uint8* src_uyvy,
>-                          int stride_uyvy,
>-                          uint8* dst_u,
>-                          uint8* dst_v,
>+void UYVYToYRow_Any_NEON(const uint8_t* src_ptr, uint8_t* dst_ptr, int width);
>+void UYVYToUVRow_Any_NEON(const uint8_t* src_ptr,
>+                          int src_stride_ptr,
>+                          uint8_t* dst_u,
>+                          uint8_t* dst_v,
>                           int width);
>-void UYVYToUV422Row_Any_NEON(const uint8* src_uyvy,
>-                             uint8* dst_u,
>-                             uint8* dst_v,
>+void UYVYToUV422Row_Any_NEON(const uint8_t* src_ptr,
>+                             uint8_t* dst_u,
>+                             uint8_t* dst_v,
>                              int width);
>-void UYVYToYRow_Any_MSA(const uint8* src_uyvy, uint8* dst_y, int width);
>-void UYVYToUVRow_Any_MSA(const uint8* src_uyvy,
>-                         int stride_uyvy,
>-                         uint8* dst_u,
>-                         uint8* dst_v,
>+void UYVYToYRow_Any_MSA(const uint8_t* src_ptr, uint8_t* dst_ptr, int width);
>+void UYVYToUVRow_Any_MSA(const uint8_t* src_ptr,
>+                         int src_stride_ptr,
>+                         uint8_t* dst_u,
>+                         uint8_t* dst_v,
>                          int width);
>-void UYVYToUV422Row_Any_MSA(const uint8* src_uyvy,
>-                            uint8* dst_u,
>-                            uint8* dst_v,
>+void UYVYToUV422Row_Any_MSA(const uint8_t* src_ptr,
>+                            uint8_t* dst_u,
>+                            uint8_t* dst_v,
>                             int width);
> 
>-void I422ToYUY2Row_C(const uint8* src_y,
>-                     const uint8* src_u,
>-                     const uint8* src_v,
>-                     uint8* dst_yuy2,
>+void I422ToYUY2Row_C(const uint8_t* src_y,
>+                     const uint8_t* src_u,
>+                     const uint8_t* src_v,
>+                     uint8_t* dst_frame,
>                      int width);
>-void I422ToUYVYRow_C(const uint8* src_y,
>-                     const uint8* src_u,
>-                     const uint8* src_v,
>-                     uint8* dst_uyvy,
>+void I422ToUYVYRow_C(const uint8_t* src_y,
>+                     const uint8_t* src_u,
>+                     const uint8_t* src_v,
>+                     uint8_t* dst_frame,
>                      int width);
>-void I422ToYUY2Row_SSE2(const uint8* src_y,
>-                        const uint8* src_u,
>-                        const uint8* src_v,
>-                        uint8* dst_yuy2,
>+void I422ToYUY2Row_SSE2(const uint8_t* src_y,
>+                        const uint8_t* src_u,
>+                        const uint8_t* src_v,
>+                        uint8_t* dst_yuy2,
>                         int width);
>-void I422ToUYVYRow_SSE2(const uint8* src_y,
>-                        const uint8* src_u,
>-                        const uint8* src_v,
>-                        uint8* dst_uyvy,
>+void I422ToUYVYRow_SSE2(const uint8_t* src_y,
>+                        const uint8_t* src_u,
>+                        const uint8_t* src_v,
>+                        uint8_t* dst_uyvy,
>                         int width);
>-void I422ToYUY2Row_Any_SSE2(const uint8* src_y,
>-                            const uint8* src_u,
>-                            const uint8* src_v,
>-                            uint8* dst_yuy2,
>-                            int width);
>-void I422ToUYVYRow_Any_SSE2(const uint8* src_y,
>-                            const uint8* src_u,
>-                            const uint8* src_v,
>-                            uint8* dst_uyvy,
>-                            int width);
>-void I422ToYUY2Row_NEON(const uint8* src_y,
>-                        const uint8* src_u,
>-                        const uint8* src_v,
>-                        uint8* dst_yuy2,
>+void I422ToYUY2Row_Any_SSE2(const uint8_t* y_buf,
>+                            const uint8_t* u_buf,
>+                            const uint8_t* v_buf,
>+                            uint8_t* dst_ptr,
>+                            int width);
>+void I422ToUYVYRow_Any_SSE2(const uint8_t* y_buf,
>+                            const uint8_t* u_buf,
>+                            const uint8_t* v_buf,
>+                            uint8_t* dst_ptr,
>+                            int width);
>+void I422ToYUY2Row_AVX2(const uint8_t* src_y,
>+                        const uint8_t* src_u,
>+                        const uint8_t* src_v,
>+                        uint8_t* dst_yuy2,
>+                        int width);
>+void I422ToUYVYRow_AVX2(const uint8_t* src_y,
>+                        const uint8_t* src_u,
>+                        const uint8_t* src_v,
>+                        uint8_t* dst_uyvy,
>+                        int width);
>+void I422ToYUY2Row_Any_AVX2(const uint8_t* y_buf,
>+                            const uint8_t* u_buf,
>+                            const uint8_t* v_buf,
>+                            uint8_t* dst_ptr,
>+                            int width);
>+void I422ToUYVYRow_Any_AVX2(const uint8_t* y_buf,
>+                            const uint8_t* u_buf,
>+                            const uint8_t* v_buf,
>+                            uint8_t* dst_ptr,
>+                            int width);
>+void I422ToYUY2Row_NEON(const uint8_t* src_y,
>+                        const uint8_t* src_u,
>+                        const uint8_t* src_v,
>+                        uint8_t* dst_yuy2,
>                         int width);
>-void I422ToUYVYRow_NEON(const uint8* src_y,
>-                        const uint8* src_u,
>-                        const uint8* src_v,
>-                        uint8* dst_uyvy,
>+void I422ToUYVYRow_NEON(const uint8_t* src_y,
>+                        const uint8_t* src_u,
>+                        const uint8_t* src_v,
>+                        uint8_t* dst_uyvy,
>                         int width);
>-void I422ToYUY2Row_Any_NEON(const uint8* src_y,
>-                            const uint8* src_u,
>-                            const uint8* src_v,
>-                            uint8* dst_yuy2,
>-                            int width);
>-void I422ToUYVYRow_Any_NEON(const uint8* src_y,
>-                            const uint8* src_u,
>-                            const uint8* src_v,
>-                            uint8* dst_uyvy,
>-                            int width);
>-void I422ToYUY2Row_MSA(const uint8* src_y,
>-                       const uint8* src_u,
>-                       const uint8* src_v,
>-                       uint8* dst_yuy2,
>+void I422ToYUY2Row_Any_NEON(const uint8_t* y_buf,
>+                            const uint8_t* u_buf,
>+                            const uint8_t* v_buf,
>+                            uint8_t* dst_ptr,
>+                            int width);
>+void I422ToUYVYRow_Any_NEON(const uint8_t* y_buf,
>+                            const uint8_t* u_buf,
>+                            const uint8_t* v_buf,
>+                            uint8_t* dst_ptr,
>+                            int width);
>+void I422ToYUY2Row_MSA(const uint8_t* src_y,
>+                       const uint8_t* src_u,
>+                       const uint8_t* src_v,
>+                       uint8_t* dst_yuy2,
>                        int width);
>-void I422ToUYVYRow_MSA(const uint8* src_y,
>-                       const uint8* src_u,
>-                       const uint8* src_v,
>-                       uint8* dst_uyvy,
>+void I422ToUYVYRow_MSA(const uint8_t* src_y,
>+                       const uint8_t* src_u,
>+                       const uint8_t* src_v,
>+                       uint8_t* dst_uyvy,
>                        int width);
>-void I422ToYUY2Row_Any_MSA(const uint8* src_y,
>-                           const uint8* src_u,
>-                           const uint8* src_v,
>-                           uint8* dst_yuy2,
>+void I422ToYUY2Row_Any_MSA(const uint8_t* y_buf,
>+                           const uint8_t* u_buf,
>+                           const uint8_t* v_buf,
>+                           uint8_t* dst_ptr,
>                            int width);
>-void I422ToUYVYRow_Any_MSA(const uint8* src_y,
>-                           const uint8* src_u,
>-                           const uint8* src_v,
>-                           uint8* dst_uyvy,
>+void I422ToUYVYRow_Any_MSA(const uint8_t* y_buf,
>+                           const uint8_t* u_buf,
>+                           const uint8_t* v_buf,
>+                           uint8_t* dst_ptr,
>                            int width);
> 
> // Effects related row functions.
>-void ARGBAttenuateRow_C(const uint8* src_argb, uint8* dst_argb, int width);
>-void ARGBAttenuateRow_SSSE3(const uint8* src_argb, uint8* dst_argb, int width);
>-void ARGBAttenuateRow_AVX2(const uint8* src_argb, uint8* dst_argb, int width);
>-void ARGBAttenuateRow_NEON(const uint8* src_argb, uint8* dst_argb, int width);
>-void ARGBAttenuateRow_MSA(const uint8* src_argb, uint8* dst_argb, int width);
>-void ARGBAttenuateRow_Any_SSE2(const uint8* src_argb,
>-                               uint8* dst_argb,
>-                               int width);
>-void ARGBAttenuateRow_Any_SSSE3(const uint8* src_argb,
>-                                uint8* dst_argb,
>+void ARGBAttenuateRow_C(const uint8_t* src_argb, uint8_t* dst_argb, int width);
>+void ARGBAttenuateRow_SSSE3(const uint8_t* src_argb,
>+                            uint8_t* dst_argb,
>+                            int width);
>+void ARGBAttenuateRow_AVX2(const uint8_t* src_argb,
>+                           uint8_t* dst_argb,
>+                           int width);
>+void ARGBAttenuateRow_NEON(const uint8_t* src_argb,
>+                           uint8_t* dst_argb,
>+                           int width);
>+void ARGBAttenuateRow_MSA(const uint8_t* src_argb,
>+                          uint8_t* dst_argb,
>+                          int width);
>+void ARGBAttenuateRow_Any_SSSE3(const uint8_t* src_ptr,
>+                                uint8_t* dst_ptr,
>                                 int width);
>-void ARGBAttenuateRow_Any_AVX2(const uint8* src_argb,
>-                               uint8* dst_argb,
>+void ARGBAttenuateRow_Any_AVX2(const uint8_t* src_ptr,
>+                               uint8_t* dst_ptr,
>                                int width);
>-void ARGBAttenuateRow_Any_NEON(const uint8* src_argb,
>-                               uint8* dst_argb,
>+void ARGBAttenuateRow_Any_NEON(const uint8_t* src_ptr,
>+                               uint8_t* dst_ptr,
>                                int width);
>-void ARGBAttenuateRow_Any_MSA(const uint8* src_argb,
>-                              uint8* dst_argb,
>+void ARGBAttenuateRow_Any_MSA(const uint8_t* src_ptr,
>+                              uint8_t* dst_ptr,
>                               int width);
> 
> // Inverse table for unattenuate, shared by C and SSE2.
>-extern const uint32 fixed_invtbl8[256];
>-void ARGBUnattenuateRow_C(const uint8* src_argb, uint8* dst_argb, int width);
>-void ARGBUnattenuateRow_SSE2(const uint8* src_argb, uint8* dst_argb, int width);
>-void ARGBUnattenuateRow_AVX2(const uint8* src_argb, uint8* dst_argb, int width);
>-void ARGBUnattenuateRow_Any_SSE2(const uint8* src_argb,
>-                                 uint8* dst_argb,
>+extern const uint32_t fixed_invtbl8[256];
>+void ARGBUnattenuateRow_C(const uint8_t* src_argb,
>+                          uint8_t* dst_argb,
>+                          int width);
>+void ARGBUnattenuateRow_SSE2(const uint8_t* src_argb,
>+                             uint8_t* dst_argb,
>+                             int width);
>+void ARGBUnattenuateRow_AVX2(const uint8_t* src_argb,
>+                             uint8_t* dst_argb,
>+                             int width);
>+void ARGBUnattenuateRow_Any_SSE2(const uint8_t* src_ptr,
>+                                 uint8_t* dst_ptr,
>                                  int width);
>-void ARGBUnattenuateRow_Any_AVX2(const uint8* src_argb,
>-                                 uint8* dst_argb,
>+void ARGBUnattenuateRow_Any_AVX2(const uint8_t* src_ptr,
>+                                 uint8_t* dst_ptr,
>                                  int width);
> 
>-void ARGBGrayRow_C(const uint8* src_argb, uint8* dst_argb, int width);
>-void ARGBGrayRow_SSSE3(const uint8* src_argb, uint8* dst_argb, int width);
>-void ARGBGrayRow_NEON(const uint8* src_argb, uint8* dst_argb, int width);
>-void ARGBGrayRow_MSA(const uint8* src_argb, uint8* dst_argb, int width);
>+void ARGBGrayRow_C(const uint8_t* src_argb, uint8_t* dst_argb, int width);
>+void ARGBGrayRow_SSSE3(const uint8_t* src_argb, uint8_t* dst_argb, int width);
>+void ARGBGrayRow_NEON(const uint8_t* src_argb, uint8_t* dst_argb, int width);
>+void ARGBGrayRow_MSA(const uint8_t* src_argb, uint8_t* dst_argb, int width);
> 
>-void ARGBSepiaRow_C(uint8* dst_argb, int width);
>-void ARGBSepiaRow_SSSE3(uint8* dst_argb, int width);
>-void ARGBSepiaRow_NEON(uint8* dst_argb, int width);
>-void ARGBSepiaRow_MSA(uint8* dst_argb, int width);
>+void ARGBSepiaRow_C(uint8_t* dst_argb, int width);
>+void ARGBSepiaRow_SSSE3(uint8_t* dst_argb, int width);
>+void ARGBSepiaRow_NEON(uint8_t* dst_argb, int width);
>+void ARGBSepiaRow_MSA(uint8_t* dst_argb, int width);
> 
>-void ARGBColorMatrixRow_C(const uint8* src_argb,
>-                          uint8* dst_argb,
>-                          const int8* matrix_argb,
>+void ARGBColorMatrixRow_C(const uint8_t* src_argb,
>+                          uint8_t* dst_argb,
>+                          const int8_t* matrix_argb,
>                           int width);
>-void ARGBColorMatrixRow_SSSE3(const uint8* src_argb,
>-                              uint8* dst_argb,
>-                              const int8* matrix_argb,
>+void ARGBColorMatrixRow_SSSE3(const uint8_t* src_argb,
>+                              uint8_t* dst_argb,
>+                              const int8_t* matrix_argb,
>                               int width);
>-void ARGBColorMatrixRow_NEON(const uint8* src_argb,
>-                             uint8* dst_argb,
>-                             const int8* matrix_argb,
>+void ARGBColorMatrixRow_NEON(const uint8_t* src_argb,
>+                             uint8_t* dst_argb,
>+                             const int8_t* matrix_argb,
>                              int width);
>-void ARGBColorMatrixRow_MSA(const uint8* src_argb,
>-                            uint8* dst_argb,
>-                            const int8* matrix_argb,
>+void ARGBColorMatrixRow_MSA(const uint8_t* src_argb,
>+                            uint8_t* dst_argb,
>+                            const int8_t* matrix_argb,
>                             int width);
> 
>-void ARGBColorTableRow_C(uint8* dst_argb, const uint8* table_argb, int width);
>-void ARGBColorTableRow_X86(uint8* dst_argb, const uint8* table_argb, int width);
>+void ARGBColorTableRow_C(uint8_t* dst_argb,
>+                         const uint8_t* table_argb,
>+                         int width);
>+void ARGBColorTableRow_X86(uint8_t* dst_argb,
>+                           const uint8_t* table_argb,
>+                           int width);
> 
>-void RGBColorTableRow_C(uint8* dst_argb, const uint8* table_argb, int width);
>-void RGBColorTableRow_X86(uint8* dst_argb, const uint8* table_argb, int width);
>+void RGBColorTableRow_C(uint8_t* dst_argb,
>+                        const uint8_t* table_argb,
>+                        int width);
>+void RGBColorTableRow_X86(uint8_t* dst_argb,
>+                          const uint8_t* table_argb,
>+                          int width);
> 
>-void ARGBQuantizeRow_C(uint8* dst_argb,
>+void ARGBQuantizeRow_C(uint8_t* dst_argb,
>                        int scale,
>                        int interval_size,
>                        int interval_offset,
>                        int width);
>-void ARGBQuantizeRow_SSE2(uint8* dst_argb,
>+void ARGBQuantizeRow_SSE2(uint8_t* dst_argb,
>                           int scale,
>                           int interval_size,
>                           int interval_offset,
>                           int width);
>-void ARGBQuantizeRow_NEON(uint8* dst_argb,
>+void ARGBQuantizeRow_NEON(uint8_t* dst_argb,
>                           int scale,
>                           int interval_size,
>                           int interval_offset,
>                           int width);
>-void ARGBQuantizeRow_MSA(uint8* dst_argb,
>+void ARGBQuantizeRow_MSA(uint8_t* dst_argb,
>                          int scale,
>                          int interval_size,
>                          int interval_offset,
>                          int width);
> 
>-void ARGBShadeRow_C(const uint8* src_argb,
>-                    uint8* dst_argb,
>+void ARGBShadeRow_C(const uint8_t* src_argb,
>+                    uint8_t* dst_argb,
>                     int width,
>-                    uint32 value);
>-void ARGBShadeRow_SSE2(const uint8* src_argb,
>-                       uint8* dst_argb,
>+                    uint32_t value);
>+void ARGBShadeRow_SSE2(const uint8_t* src_argb,
>+                       uint8_t* dst_argb,
>                        int width,
>-                       uint32 value);
>-void ARGBShadeRow_NEON(const uint8* src_argb,
>-                       uint8* dst_argb,
>+                       uint32_t value);
>+void ARGBShadeRow_NEON(const uint8_t* src_argb,
>+                       uint8_t* dst_argb,
>                        int width,
>-                       uint32 value);
>-void ARGBShadeRow_MSA(const uint8* src_argb,
>-                      uint8* dst_argb,
>+                       uint32_t value);
>+void ARGBShadeRow_MSA(const uint8_t* src_argb,
>+                      uint8_t* dst_argb,
>                       int width,
>-                      uint32 value);
>+                      uint32_t value);
> 
> // Used for blur.
>-void CumulativeSumToAverageRow_SSE2(const int32* topleft,
>-                                    const int32* botleft,
>+void CumulativeSumToAverageRow_SSE2(const int32_t* topleft,
>+                                    const int32_t* botleft,
>                                     int width,
>                                     int area,
>-                                    uint8* dst,
>+                                    uint8_t* dst,
>                                     int count);
>-void ComputeCumulativeSumRow_SSE2(const uint8* row,
>-                                  int32* cumsum,
>-                                  const int32* previous_cumsum,
>+void ComputeCumulativeSumRow_SSE2(const uint8_t* row,
>+                                  int32_t* cumsum,
>+                                  const int32_t* previous_cumsum,
>                                   int width);
> 
>-void CumulativeSumToAverageRow_C(const int32* topleft,
>-                                 const int32* botleft,
>-                                 int width,
>+void CumulativeSumToAverageRow_C(const int32_t* tl,
>+                                 const int32_t* bl,
>+                                 int w,
>                                  int area,
>-                                 uint8* dst,
>+                                 uint8_t* dst,
>                                  int count);
>-void ComputeCumulativeSumRow_C(const uint8* row,
>-                               int32* cumsum,
>-                               const int32* previous_cumsum,
>+void ComputeCumulativeSumRow_C(const uint8_t* row,
>+                               int32_t* cumsum,
>+                               const int32_t* previous_cumsum,
>                                int width);
> 
> LIBYUV_API
>-void ARGBAffineRow_C(const uint8* src_argb,
>+void ARGBAffineRow_C(const uint8_t* src_argb,
>                      int src_argb_stride,
>-                     uint8* dst_argb,
>+                     uint8_t* dst_argb,
>                      const float* uv_dudv,
>                      int width);
> LIBYUV_API
>-void ARGBAffineRow_SSE2(const uint8* src_argb,
>+void ARGBAffineRow_SSE2(const uint8_t* src_argb,
>                         int src_argb_stride,
>-                        uint8* dst_argb,
>-                        const float* uv_dudv,
>+                        uint8_t* dst_argb,
>+                        const float* src_dudv,
>                         int width);
> 
> // Used for I420Scale, ARGBScale, and ARGBInterpolate.
>-void InterpolateRow_C(uint8* dst_ptr,
>-                      const uint8* src_ptr,
>-                      ptrdiff_t src_stride_ptr,
>+void InterpolateRow_C(uint8_t* dst_ptr,
>+                      const uint8_t* src_ptr,
>+                      ptrdiff_t src_stride,
>                       int width,
>                       int source_y_fraction);
>-void InterpolateRow_SSSE3(uint8* dst_ptr,
>-                          const uint8* src_ptr,
>-                          ptrdiff_t src_stride_ptr,
>-                          int width,
>+void InterpolateRow_SSSE3(uint8_t* dst_ptr,
>+                          const uint8_t* src_ptr,
>+                          ptrdiff_t src_stride,
>+                          int dst_width,
>                           int source_y_fraction);
>-void InterpolateRow_AVX2(uint8* dst_ptr,
>-                         const uint8* src_ptr,
>-                         ptrdiff_t src_stride_ptr,
>-                         int width,
>+void InterpolateRow_AVX2(uint8_t* dst_ptr,
>+                         const uint8_t* src_ptr,
>+                         ptrdiff_t src_stride,
>+                         int dst_width,
>                          int source_y_fraction);
>-void InterpolateRow_NEON(uint8* dst_ptr,
>-                         const uint8* src_ptr,
>-                         ptrdiff_t src_stride_ptr,
>-                         int width,
>+void InterpolateRow_NEON(uint8_t* dst_ptr,
>+                         const uint8_t* src_ptr,
>+                         ptrdiff_t src_stride,
>+                         int dst_width,
>                          int source_y_fraction);
>-void InterpolateRow_DSPR2(uint8* dst_ptr,
>-                          const uint8* src_ptr,
>-                          ptrdiff_t src_stride_ptr,
>-                          int width,
>-                          int source_y_fraction);
>-void InterpolateRow_MSA(uint8* dst_ptr,
>-                        const uint8* src_ptr,
>-                        ptrdiff_t src_stride_ptr,
>+void InterpolateRow_MSA(uint8_t* dst_ptr,
>+                        const uint8_t* src_ptr,
>+                        ptrdiff_t src_stride,
>                         int width,
>                         int source_y_fraction);
>-void InterpolateRow_Any_NEON(uint8* dst_ptr,
>-                             const uint8* src_ptr,
>+void InterpolateRow_Any_NEON(uint8_t* dst_ptr,
>+                             const uint8_t* src_ptr,
>                              ptrdiff_t src_stride_ptr,
>                              int width,
>                              int source_y_fraction);
>-void InterpolateRow_Any_SSSE3(uint8* dst_ptr,
>-                              const uint8* src_ptr,
>+void InterpolateRow_Any_SSSE3(uint8_t* dst_ptr,
>+                              const uint8_t* src_ptr,
>                               ptrdiff_t src_stride_ptr,
>                               int width,
>                               int source_y_fraction);
>-void InterpolateRow_Any_AVX2(uint8* dst_ptr,
>-                             const uint8* src_ptr,
>+void InterpolateRow_Any_AVX2(uint8_t* dst_ptr,
>+                             const uint8_t* src_ptr,
>                              ptrdiff_t src_stride_ptr,
>                              int width,
>                              int source_y_fraction);
>-void InterpolateRow_Any_DSPR2(uint8* dst_ptr,
>-                              const uint8* src_ptr,
>-                              ptrdiff_t src_stride_ptr,
>-                              int width,
>-                              int source_y_fraction);
>-void InterpolateRow_Any_MSA(uint8* dst_ptr,
>-                            const uint8* src_ptr,
>+void InterpolateRow_Any_MSA(uint8_t* dst_ptr,
>+                            const uint8_t* src_ptr,
>                             ptrdiff_t src_stride_ptr,
>                             int width,
>                             int source_y_fraction);
> 
>-void InterpolateRow_16_C(uint16* dst_ptr,
>-                         const uint16* src_ptr,
>-                         ptrdiff_t src_stride_ptr,
>+void InterpolateRow_16_C(uint16_t* dst_ptr,
>+                         const uint16_t* src_ptr,
>+                         ptrdiff_t src_stride,
>                          int width,
>                          int source_y_fraction);
> 
> // Sobel images.
>-void SobelXRow_C(const uint8* src_y0,
>-                 const uint8* src_y1,
>-                 const uint8* src_y2,
>-                 uint8* dst_sobelx,
>+void SobelXRow_C(const uint8_t* src_y0,
>+                 const uint8_t* src_y1,
>+                 const uint8_t* src_y2,
>+                 uint8_t* dst_sobelx,
>                  int width);
>-void SobelXRow_SSE2(const uint8* src_y0,
>-                    const uint8* src_y1,
>-                    const uint8* src_y2,
>-                    uint8* dst_sobelx,
>+void SobelXRow_SSE2(const uint8_t* src_y0,
>+                    const uint8_t* src_y1,
>+                    const uint8_t* src_y2,
>+                    uint8_t* dst_sobelx,
>                     int width);
>-void SobelXRow_NEON(const uint8* src_y0,
>-                    const uint8* src_y1,
>-                    const uint8* src_y2,
>-                    uint8* dst_sobelx,
>+void SobelXRow_NEON(const uint8_t* src_y0,
>+                    const uint8_t* src_y1,
>+                    const uint8_t* src_y2,
>+                    uint8_t* dst_sobelx,
>                     int width);
>-void SobelXRow_MSA(const uint8* src_y0,
>-                   const uint8* src_y1,
>-                   const uint8* src_y2,
>-                   uint8* dst_sobelx,
>+void SobelXRow_MSA(const uint8_t* src_y0,
>+                   const uint8_t* src_y1,
>+                   const uint8_t* src_y2,
>+                   uint8_t* dst_sobelx,
>                    int width);
>-void SobelYRow_C(const uint8* src_y0,
>-                 const uint8* src_y1,
>-                 uint8* dst_sobely,
>+void SobelYRow_C(const uint8_t* src_y0,
>+                 const uint8_t* src_y1,
>+                 uint8_t* dst_sobely,
>                  int width);
>-void SobelYRow_SSE2(const uint8* src_y0,
>-                    const uint8* src_y1,
>-                    uint8* dst_sobely,
>+void SobelYRow_SSE2(const uint8_t* src_y0,
>+                    const uint8_t* src_y1,
>+                    uint8_t* dst_sobely,
>                     int width);
>-void SobelYRow_NEON(const uint8* src_y0,
>-                    const uint8* src_y1,
>-                    uint8* dst_sobely,
>+void SobelYRow_NEON(const uint8_t* src_y0,
>+                    const uint8_t* src_y1,
>+                    uint8_t* dst_sobely,
>                     int width);
>-void SobelYRow_MSA(const uint8* src_y0,
>-                   const uint8* src_y1,
>-                   uint8* dst_sobely,
>+void SobelYRow_MSA(const uint8_t* src_y0,
>+                   const uint8_t* src_y1,
>+                   uint8_t* dst_sobely,
>                    int width);
>-void SobelRow_C(const uint8* src_sobelx,
>-                const uint8* src_sobely,
>-                uint8* dst_argb,
>+void SobelRow_C(const uint8_t* src_sobelx,
>+                const uint8_t* src_sobely,
>+                uint8_t* dst_argb,
>                 int width);
>-void SobelRow_SSE2(const uint8* src_sobelx,
>-                   const uint8* src_sobely,
>-                   uint8* dst_argb,
>+void SobelRow_SSE2(const uint8_t* src_sobelx,
>+                   const uint8_t* src_sobely,
>+                   uint8_t* dst_argb,
>                    int width);
>-void SobelRow_NEON(const uint8* src_sobelx,
>-                   const uint8* src_sobely,
>-                   uint8* dst_argb,
>+void SobelRow_NEON(const uint8_t* src_sobelx,
>+                   const uint8_t* src_sobely,
>+                   uint8_t* dst_argb,
>                    int width);
>-void SobelRow_MSA(const uint8* src_sobelx,
>-                  const uint8* src_sobely,
>-                  uint8* dst_argb,
>+void SobelRow_MSA(const uint8_t* src_sobelx,
>+                  const uint8_t* src_sobely,
>+                  uint8_t* dst_argb,
>                   int width);
>-void SobelToPlaneRow_C(const uint8* src_sobelx,
>-                       const uint8* src_sobely,
>-                       uint8* dst_y,
>+void SobelToPlaneRow_C(const uint8_t* src_sobelx,
>+                       const uint8_t* src_sobely,
>+                       uint8_t* dst_y,
>                        int width);
>-void SobelToPlaneRow_SSE2(const uint8* src_sobelx,
>-                          const uint8* src_sobely,
>-                          uint8* dst_y,
>+void SobelToPlaneRow_SSE2(const uint8_t* src_sobelx,
>+                          const uint8_t* src_sobely,
>+                          uint8_t* dst_y,
>                           int width);
>-void SobelToPlaneRow_NEON(const uint8* src_sobelx,
>-                          const uint8* src_sobely,
>-                          uint8* dst_y,
>+void SobelToPlaneRow_NEON(const uint8_t* src_sobelx,
>+                          const uint8_t* src_sobely,
>+                          uint8_t* dst_y,
>                           int width);
>-void SobelToPlaneRow_MSA(const uint8* src_sobelx,
>-                         const uint8* src_sobely,
>-                         uint8* dst_y,
>+void SobelToPlaneRow_MSA(const uint8_t* src_sobelx,
>+                         const uint8_t* src_sobely,
>+                         uint8_t* dst_y,
>                          int width);
>-void SobelXYRow_C(const uint8* src_sobelx,
>-                  const uint8* src_sobely,
>-                  uint8* dst_argb,
>+void SobelXYRow_C(const uint8_t* src_sobelx,
>+                  const uint8_t* src_sobely,
>+                  uint8_t* dst_argb,
>                   int width);
>-void SobelXYRow_SSE2(const uint8* src_sobelx,
>-                     const uint8* src_sobely,
>-                     uint8* dst_argb,
>+void SobelXYRow_SSE2(const uint8_t* src_sobelx,
>+                     const uint8_t* src_sobely,
>+                     uint8_t* dst_argb,
>                      int width);
>-void SobelXYRow_NEON(const uint8* src_sobelx,
>-                     const uint8* src_sobely,
>-                     uint8* dst_argb,
>+void SobelXYRow_NEON(const uint8_t* src_sobelx,
>+                     const uint8_t* src_sobely,
>+                     uint8_t* dst_argb,
>                      int width);
>-void SobelXYRow_MSA(const uint8* src_sobelx,
>-                    const uint8* src_sobely,
>-                    uint8* dst_argb,
>+void SobelXYRow_MSA(const uint8_t* src_sobelx,
>+                    const uint8_t* src_sobely,
>+                    uint8_t* dst_argb,
>                     int width);
>-void SobelRow_Any_SSE2(const uint8* src_sobelx,
>-                       const uint8* src_sobely,
>-                       uint8* dst_argb,
>+void SobelRow_Any_SSE2(const uint8_t* y_buf,
>+                       const uint8_t* uv_buf,
>+                       uint8_t* dst_ptr,
>                        int width);
>-void SobelRow_Any_NEON(const uint8* src_sobelx,
>-                       const uint8* src_sobely,
>-                       uint8* dst_argb,
>+void SobelRow_Any_NEON(const uint8_t* y_buf,
>+                       const uint8_t* uv_buf,
>+                       uint8_t* dst_ptr,
>                        int width);
>-void SobelRow_Any_MSA(const uint8* src_sobelx,
>-                      const uint8* src_sobely,
>-                      uint8* dst_argb,
>+void SobelRow_Any_MSA(const uint8_t* y_buf,
>+                      const uint8_t* uv_buf,
>+                      uint8_t* dst_ptr,
>                       int width);
>-void SobelToPlaneRow_Any_SSE2(const uint8* src_sobelx,
>-                              const uint8* src_sobely,
>-                              uint8* dst_y,
>+void SobelToPlaneRow_Any_SSE2(const uint8_t* y_buf,
>+                              const uint8_t* uv_buf,
>+                              uint8_t* dst_ptr,
>                               int width);
>-void SobelToPlaneRow_Any_NEON(const uint8* src_sobelx,
>-                              const uint8* src_sobely,
>-                              uint8* dst_y,
>+void SobelToPlaneRow_Any_NEON(const uint8_t* y_buf,
>+                              const uint8_t* uv_buf,
>+                              uint8_t* dst_ptr,
>                               int width);
>-void SobelToPlaneRow_Any_MSA(const uint8* src_sobelx,
>-                             const uint8* src_sobely,
>-                             uint8* dst_y,
>+void SobelToPlaneRow_Any_MSA(const uint8_t* y_buf,
>+                             const uint8_t* uv_buf,
>+                             uint8_t* dst_ptr,
>                              int width);
>-void SobelXYRow_Any_SSE2(const uint8* src_sobelx,
>-                         const uint8* src_sobely,
>-                         uint8* dst_argb,
>+void SobelXYRow_Any_SSE2(const uint8_t* y_buf,
>+                         const uint8_t* uv_buf,
>+                         uint8_t* dst_ptr,
>                          int width);
>-void SobelXYRow_Any_NEON(const uint8* src_sobelx,
>-                         const uint8* src_sobely,
>-                         uint8* dst_argb,
>+void SobelXYRow_Any_NEON(const uint8_t* y_buf,
>+                         const uint8_t* uv_buf,
>+                         uint8_t* dst_ptr,
>                          int width);
>-void SobelXYRow_Any_MSA(const uint8* src_sobelx,
>-                        const uint8* src_sobely,
>-                        uint8* dst_argb,
>+void SobelXYRow_Any_MSA(const uint8_t* y_buf,
>+                        const uint8_t* uv_buf,
>+                        uint8_t* dst_ptr,
>                         int width);
> 
>-void ARGBPolynomialRow_C(const uint8* src_argb,
>-                         uint8* dst_argb,
>+void ARGBPolynomialRow_C(const uint8_t* src_argb,
>+                         uint8_t* dst_argb,
>                          const float* poly,
>                          int width);
>-void ARGBPolynomialRow_SSE2(const uint8* src_argb,
>-                            uint8* dst_argb,
>+void ARGBPolynomialRow_SSE2(const uint8_t* src_argb,
>+                            uint8_t* dst_argb,
>                             const float* poly,
>                             int width);
>-void ARGBPolynomialRow_AVX2(const uint8* src_argb,
>-                            uint8* dst_argb,
>+void ARGBPolynomialRow_AVX2(const uint8_t* src_argb,
>+                            uint8_t* dst_argb,
>                             const float* poly,
>                             int width);
> 
> // Scale and convert to half float.
>-void HalfFloatRow_C(const uint16* src, uint16* dst, float scale, int width);
>-void HalfFloatRow_SSE2(const uint16* src, uint16* dst, float scale, int width);
>-void HalfFloatRow_Any_SSE2(const uint16* src,
>-                           uint16* dst,
>-                           float scale,
>+void HalfFloatRow_C(const uint16_t* src, uint16_t* dst, float scale, int width);
>+void HalfFloatRow_SSE2(const uint16_t* src,
>+                       uint16_t* dst,
>+                       float scale,
>+                       int width);
>+void HalfFloatRow_Any_SSE2(const uint16_t* src_ptr,
>+                           uint16_t* dst_ptr,
>+                           float param,
>                            int width);
>-void HalfFloatRow_AVX2(const uint16* src, uint16* dst, float scale, int width);
>-void HalfFloatRow_Any_AVX2(const uint16* src,
>-                           uint16* dst,
>-                           float scale,
>+void HalfFloatRow_AVX2(const uint16_t* src,
>+                       uint16_t* dst,
>+                       float scale,
>+                       int width);
>+void HalfFloatRow_Any_AVX2(const uint16_t* src_ptr,
>+                           uint16_t* dst_ptr,
>+                           float param,
>                            int width);
>-void HalfFloatRow_F16C(const uint16* src, uint16* dst, float scale, int width);
>-void HalfFloatRow_Any_F16C(const uint16* src,
>-                           uint16* dst,
>+void HalfFloatRow_F16C(const uint16_t* src,
>+                       uint16_t* dst,
>+                       float scale,
>+                       int width);
>+void HalfFloatRow_Any_F16C(const uint16_t* src,
>+                           uint16_t* dst,
>                            float scale,
>                            int width);
>-void HalfFloat1Row_F16C(const uint16* src, uint16* dst, float scale, int width);
>-void HalfFloat1Row_Any_F16C(const uint16* src,
>-                            uint16* dst,
>+void HalfFloat1Row_F16C(const uint16_t* src,
>+                        uint16_t* dst,
>+                        float scale,
>+                        int width);
>+void HalfFloat1Row_Any_F16C(const uint16_t* src,
>+                            uint16_t* dst,
>                             float scale,
>                             int width);
>-void HalfFloatRow_NEON(const uint16* src, uint16* dst, float scale, int width);
>-void HalfFloatRow_Any_NEON(const uint16* src,
>-                           uint16* dst,
>-                           float scale,
>+void HalfFloatRow_NEON(const uint16_t* src,
>+                       uint16_t* dst,
>+                       float scale,
>+                       int width);
>+void HalfFloatRow_Any_NEON(const uint16_t* src_ptr,
>+                           uint16_t* dst_ptr,
>+                           float param,
>                            int width);
>-void HalfFloat1Row_NEON(const uint16* src, uint16* dst, float scale, int width);
>-void HalfFloat1Row_Any_NEON(const uint16* src,
>-                            uint16* dst,
>-                            float scale,
>+void HalfFloat1Row_NEON(const uint16_t* src,
>+                        uint16_t* dst,
>+                        float scale,
>+                        int width);
>+void HalfFloat1Row_Any_NEON(const uint16_t* src_ptr,
>+                            uint16_t* dst_ptr,
>+                            float param,
>                             int width);
>-void HalfFloatRow_MSA(const uint16* src, uint16* dst, float scale, int width);
>-void HalfFloatRow_Any_MSA(const uint16* src,
>-                          uint16* dst,
>-                          float scale,
>+void HalfFloatRow_MSA(const uint16_t* src,
>+                      uint16_t* dst,
>+                      float scale,
>+                      int width);
>+void HalfFloatRow_Any_MSA(const uint16_t* src_ptr,
>+                          uint16_t* dst_ptr,
>+                          float param,
>                           int width);
>+void ByteToFloatRow_C(const uint8_t* src, float* dst, float scale, int width);
>+void ByteToFloatRow_NEON(const uint8_t* src,
>+                         float* dst,
>+                         float scale,
>+                         int width);
>+void ByteToFloatRow_Any_NEON(const uint8_t* src,
>+                             float* dst,
>+                             float scale,
>+                             int width);
> 
>-void ARGBLumaColorTableRow_C(const uint8* src_argb,
>-                             uint8* dst_argb,
>+void ARGBLumaColorTableRow_C(const uint8_t* src_argb,
>+                             uint8_t* dst_argb,
>                              int width,
>-                             const uint8* luma,
>-                             uint32 lumacoeff);
>-void ARGBLumaColorTableRow_SSSE3(const uint8* src_argb,
>-                                 uint8* dst_argb,
>+                             const uint8_t* luma,
>+                             uint32_t lumacoeff);
>+void ARGBLumaColorTableRow_SSSE3(const uint8_t* src_argb,
>+                                 uint8_t* dst_argb,
>                                  int width,
>-                                 const uint8* luma,
>-                                 uint32 lumacoeff);
>+                                 const uint8_t* luma,
>+                                 uint32_t lumacoeff);
> 
> float ScaleMaxSamples_C(const float* src, float* dst, float scale, int width);
> float ScaleMaxSamples_NEON(const float* src,
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/scale.h b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/scale.h
>index 6d6b9a8583a..b937d348cab 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/scale.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/scale.h
>@@ -28,22 +28,22 @@ typedef enum FilterMode {
> 
> // Scale a YUV plane.
> LIBYUV_API
>-void ScalePlane(const uint8* src,
>+void ScalePlane(const uint8_t* src,
>                 int src_stride,
>                 int src_width,
>                 int src_height,
>-                uint8* dst,
>+                uint8_t* dst,
>                 int dst_stride,
>                 int dst_width,
>                 int dst_height,
>                 enum FilterMode filtering);
> 
> LIBYUV_API
>-void ScalePlane_16(const uint16* src,
>+void ScalePlane_16(const uint16_t* src,
>                    int src_stride,
>                    int src_width,
>                    int src_height,
>-                   uint16* dst,
>+                   uint16_t* dst,
>                    int dst_stride,
>                    int dst_width,
>                    int dst_height,
>@@ -60,38 +60,38 @@ void ScalePlane_16(const uint16* src,
> // Returns 0 if successful.
> 
> LIBYUV_API
>-int I420Scale(const uint8* src_y,
>+int I420Scale(const uint8_t* src_y,
>               int src_stride_y,
>-              const uint8* src_u,
>+              const uint8_t* src_u,
>               int src_stride_u,
>-              const uint8* src_v,
>+              const uint8_t* src_v,
>               int src_stride_v,
>               int src_width,
>               int src_height,
>-              uint8* dst_y,
>+              uint8_t* dst_y,
>               int dst_stride_y,
>-              uint8* dst_u,
>+              uint8_t* dst_u,
>               int dst_stride_u,
>-              uint8* dst_v,
>+              uint8_t* dst_v,
>               int dst_stride_v,
>               int dst_width,
>               int dst_height,
>               enum FilterMode filtering);
> 
> LIBYUV_API
>-int I420Scale_16(const uint16* src_y,
>+int I420Scale_16(const uint16_t* src_y,
>                  int src_stride_y,
>-                 const uint16* src_u,
>+                 const uint16_t* src_u,
>                  int src_stride_u,
>-                 const uint16* src_v,
>+                 const uint16_t* src_v,
>                  int src_stride_v,
>                  int src_width,
>                  int src_height,
>-                 uint16* dst_y,
>+                 uint16_t* dst_y,
>                  int dst_stride_y,
>-                 uint16* dst_u,
>+                 uint16_t* dst_u,
>                  int dst_stride_u,
>-                 uint16* dst_v,
>+                 uint16_t* dst_v,
>                  int dst_stride_v,
>                  int dst_width,
>                  int dst_height,
>@@ -100,17 +100,17 @@ int I420Scale_16(const uint16* src_y,
> #ifdef __cplusplus
> // Legacy API.  Deprecated.
> LIBYUV_API
>-int Scale(const uint8* src_y,
>-          const uint8* src_u,
>-          const uint8* src_v,
>+int Scale(const uint8_t* src_y,
>+          const uint8_t* src_u,
>+          const uint8_t* src_v,
>           int src_stride_y,
>           int src_stride_u,
>           int src_stride_v,
>           int src_width,
>           int src_height,
>-          uint8* dst_y,
>-          uint8* dst_u,
>-          uint8* dst_v,
>+          uint8_t* dst_y,
>+          uint8_t* dst_u,
>+          uint8_t* dst_v,
>           int dst_stride_y,
>           int dst_stride_u,
>           int dst_stride_v,
>@@ -118,17 +118,6 @@ int Scale(const uint8* src_y,
>           int dst_height,
>           LIBYUV_BOOL interpolate);
> 
>-// Legacy API.  Deprecated.
>-LIBYUV_API
>-int ScaleOffset(const uint8* src_i420,
>-                int src_width,
>-                int src_height,
>-                uint8* dst_i420,
>-                int dst_width,
>-                int dst_height,
>-                int dst_yoffset,
>-                LIBYUV_BOOL interpolate);
>-
> // For testing, allow disabling of specialized scalers.
> LIBYUV_API
> void SetUseReferenceImpl(LIBYUV_BOOL use);
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/scale_argb.h b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/scale_argb.h
>index 3d25e579cde..7641f18e341 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/scale_argb.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/scale_argb.h
>@@ -20,11 +20,11 @@ extern "C" {
> #endif
> 
> LIBYUV_API
>-int ARGBScale(const uint8* src_argb,
>+int ARGBScale(const uint8_t* src_argb,
>               int src_stride_argb,
>               int src_width,
>               int src_height,
>-              uint8* dst_argb,
>+              uint8_t* dst_argb,
>               int dst_stride_argb,
>               int dst_width,
>               int dst_height,
>@@ -32,11 +32,11 @@ int ARGBScale(const uint8* src_argb,
> 
> // Clipped scale takes destination rectangle coordinates for clip values.
> LIBYUV_API
>-int ARGBScaleClip(const uint8* src_argb,
>+int ARGBScaleClip(const uint8_t* src_argb,
>                   int src_stride_argb,
>                   int src_width,
>                   int src_height,
>-                  uint8* dst_argb,
>+                  uint8_t* dst_argb,
>                   int dst_stride_argb,
>                   int dst_width,
>                   int dst_height,
>@@ -48,18 +48,18 @@ int ARGBScaleClip(const uint8* src_argb,
> 
> // Scale with YUV conversion to ARGB and clipping.
> LIBYUV_API
>-int YUVToARGBScaleClip(const uint8* src_y,
>+int YUVToARGBScaleClip(const uint8_t* src_y,
>                        int src_stride_y,
>-                       const uint8* src_u,
>+                       const uint8_t* src_u,
>                        int src_stride_u,
>-                       const uint8* src_v,
>+                       const uint8_t* src_v,
>                        int src_stride_v,
>-                       uint32 src_fourcc,
>+                       uint32_t src_fourcc,
>                        int src_width,
>                        int src_height,
>-                       uint8* dst_argb,
>+                       uint8_t* dst_argb,
>                        int dst_stride_argb,
>-                       uint32 dst_fourcc,
>+                       uint32_t dst_fourcc,
>                        int dst_width,
>                        int dst_height,
>                        int clip_x,
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/scale_row.h b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/scale_row.h
>index c4a66aa07b1..7194ba09f84 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/scale_row.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/scale_row.h
>@@ -19,17 +19,20 @@ namespace libyuv {
> extern "C" {
> #endif
> 
>-#if defined(__pnacl__) || defined(__CLR_VER) || \
>+#if defined(__pnacl__) || defined(__CLR_VER) ||            \
>+    (defined(__native_client__) && defined(__x86_64__)) || \
>     (defined(__i386__) && !defined(__SSE__) && !defined(__clang__))
> #define LIBYUV_DISABLE_X86
> #endif
>+#if defined(__native_client__)
>+#define LIBYUV_DISABLE_NEON
>+#endif
> // MemorySanitizer does not support assembly code yet. http://crbug.com/344505
> #if defined(__has_feature)
> #if __has_feature(memory_sanitizer)
> #define LIBYUV_DISABLE_X86
> #endif
> #endif
>-
> // GCC >= 4.7.0 required for AVX2.
> #if defined(__GNUC__) && (defined(__x86_64__) || defined(__i386__))
> #if (__GNUC__ > 4) || (__GNUC__ == 4 && (__GNUC_MINOR__ >= 7))
>@@ -81,7 +84,7 @@ extern "C" {
> #endif
> 
> // The following are available on Neon platforms:
>-#if !defined(LIBYUV_DISABLE_NEON) && !defined(__native_client__) && \
>+#if !defined(LIBYUV_DISABLE_NEON) && \
>     (defined(__ARM_NEON__) || defined(LIBYUV_NEON) || defined(__aarch64__))
> #define HAS_SCALEARGBCOLS_NEON
> #define HAS_SCALEARGBROWDOWN2_NEON
>@@ -94,16 +97,6 @@ extern "C" {
> #define HAS_SCALEARGBFILTERCOLS_NEON
> #endif
> 
>-// The following are available on Mips platforms:
>-#if !defined(LIBYUV_DISABLE_DSPR2) && !defined(__native_client__) && \
>-    defined(__mips__) && defined(__mips_dsp) && (__mips_dsp_rev >= 2)
>-#define HAS_SCALEROWDOWN2_DSPR2
>-#define HAS_SCALEROWDOWN4_DSPR2
>-#define HAS_SCALEROWDOWN34_DSPR2
>-#define HAS_SCALEROWDOWN38_DSPR2
>-#define HAS_SCALEADDROW_DSPR2
>-#endif
>-
> #if !defined(LIBYUV_DISABLE_MSA) && defined(__mips_msa)
> #define HAS_SCALEADDROW_MSA
> #define HAS_SCALEARGBCOLS_MSA
>@@ -123,8 +116,8 @@ void ScalePlaneVertical(int src_height,
>                         int dst_height,
>                         int src_stride,
>                         int dst_stride,
>-                        const uint8* src_argb,
>-                        uint8* dst_argb,
>+                        const uint8_t* src_argb,
>+                        uint8_t* dst_argb,
>                         int x,
>                         int y,
>                         int dy,
>@@ -136,8 +129,8 @@ void ScalePlaneVertical_16(int src_height,
>                            int dst_height,
>                            int src_stride,
>                            int dst_stride,
>-                           const uint16* src_argb,
>-                           uint16* dst_argb,
>+                           const uint16_t* src_argb,
>+                           uint16_t* dst_argb,
>                            int x,
>                            int y,
>                            int dy,
>@@ -176,425 +169,431 @@ void ScaleSlope(int src_width,
>                 int* dx,
>                 int* dy);
> 
>-void ScaleRowDown2_C(const uint8* src_ptr,
>+void ScaleRowDown2_C(const uint8_t* src_ptr,
>                      ptrdiff_t src_stride,
>-                     uint8* dst,
>+                     uint8_t* dst,
>                      int dst_width);
>-void ScaleRowDown2_16_C(const uint16* src_ptr,
>+void ScaleRowDown2_16_C(const uint16_t* src_ptr,
>                         ptrdiff_t src_stride,
>-                        uint16* dst,
>+                        uint16_t* dst,
>                         int dst_width);
>-void ScaleRowDown2Linear_C(const uint8* src_ptr,
>+void ScaleRowDown2Linear_C(const uint8_t* src_ptr,
>                            ptrdiff_t src_stride,
>-                           uint8* dst,
>+                           uint8_t* dst,
>                            int dst_width);
>-void ScaleRowDown2Linear_16_C(const uint16* src_ptr,
>+void ScaleRowDown2Linear_16_C(const uint16_t* src_ptr,
>                               ptrdiff_t src_stride,
>-                              uint16* dst,
>+                              uint16_t* dst,
>                               int dst_width);
>-void ScaleRowDown2Box_C(const uint8* src_ptr,
>+void ScaleRowDown2Box_C(const uint8_t* src_ptr,
>                         ptrdiff_t src_stride,
>-                        uint8* dst,
>+                        uint8_t* dst,
>                         int dst_width);
>-void ScaleRowDown2Box_Odd_C(const uint8* src_ptr,
>+void ScaleRowDown2Box_Odd_C(const uint8_t* src_ptr,
>                             ptrdiff_t src_stride,
>-                            uint8* dst,
>+                            uint8_t* dst,
>                             int dst_width);
>-void ScaleRowDown2Box_16_C(const uint16* src_ptr,
>+void ScaleRowDown2Box_16_C(const uint16_t* src_ptr,
>                            ptrdiff_t src_stride,
>-                           uint16* dst,
>+                           uint16_t* dst,
>                            int dst_width);
>-void ScaleRowDown4_C(const uint8* src_ptr,
>+void ScaleRowDown4_C(const uint8_t* src_ptr,
>                      ptrdiff_t src_stride,
>-                     uint8* dst,
>+                     uint8_t* dst,
>                      int dst_width);
>-void ScaleRowDown4_16_C(const uint16* src_ptr,
>+void ScaleRowDown4_16_C(const uint16_t* src_ptr,
>                         ptrdiff_t src_stride,
>-                        uint16* dst,
>+                        uint16_t* dst,
>                         int dst_width);
>-void ScaleRowDown4Box_C(const uint8* src_ptr,
>+void ScaleRowDown4Box_C(const uint8_t* src_ptr,
>                         ptrdiff_t src_stride,
>-                        uint8* dst,
>+                        uint8_t* dst,
>                         int dst_width);
>-void ScaleRowDown4Box_16_C(const uint16* src_ptr,
>+void ScaleRowDown4Box_16_C(const uint16_t* src_ptr,
>                            ptrdiff_t src_stride,
>-                           uint16* dst,
>+                           uint16_t* dst,
>                            int dst_width);
>-void ScaleRowDown34_C(const uint8* src_ptr,
>+void ScaleRowDown34_C(const uint8_t* src_ptr,
>                       ptrdiff_t src_stride,
>-                      uint8* dst,
>+                      uint8_t* dst,
>                       int dst_width);
>-void ScaleRowDown34_16_C(const uint16* src_ptr,
>+void ScaleRowDown34_16_C(const uint16_t* src_ptr,
>                          ptrdiff_t src_stride,
>-                         uint16* dst,
>+                         uint16_t* dst,
>                          int dst_width);
>-void ScaleRowDown34_0_Box_C(const uint8* src_ptr,
>+void ScaleRowDown34_0_Box_C(const uint8_t* src_ptr,
>                             ptrdiff_t src_stride,
>-                            uint8* d,
>+                            uint8_t* d,
>                             int dst_width);
>-void ScaleRowDown34_0_Box_16_C(const uint16* src_ptr,
>+void ScaleRowDown34_0_Box_16_C(const uint16_t* src_ptr,
>                                ptrdiff_t src_stride,
>-                               uint16* d,
>+                               uint16_t* d,
>                                int dst_width);
>-void ScaleRowDown34_1_Box_C(const uint8* src_ptr,
>+void ScaleRowDown34_1_Box_C(const uint8_t* src_ptr,
>                             ptrdiff_t src_stride,
>-                            uint8* d,
>+                            uint8_t* d,
>                             int dst_width);
>-void ScaleRowDown34_1_Box_16_C(const uint16* src_ptr,
>+void ScaleRowDown34_1_Box_16_C(const uint16_t* src_ptr,
>                                ptrdiff_t src_stride,
>-                               uint16* d,
>+                               uint16_t* d,
>                                int dst_width);
>-void ScaleCols_C(uint8* dst_ptr,
>-                 const uint8* src_ptr,
>+void ScaleCols_C(uint8_t* dst_ptr,
>+                 const uint8_t* src_ptr,
>                  int dst_width,
>                  int x,
>                  int dx);
>-void ScaleCols_16_C(uint16* dst_ptr,
>-                    const uint16* src_ptr,
>+void ScaleCols_16_C(uint16_t* dst_ptr,
>+                    const uint16_t* src_ptr,
>                     int dst_width,
>                     int x,
>                     int dx);
>-void ScaleColsUp2_C(uint8* dst_ptr,
>-                    const uint8* src_ptr,
>+void ScaleColsUp2_C(uint8_t* dst_ptr,
>+                    const uint8_t* src_ptr,
>                     int dst_width,
>                     int,
>                     int);
>-void ScaleColsUp2_16_C(uint16* dst_ptr,
>-                       const uint16* src_ptr,
>+void ScaleColsUp2_16_C(uint16_t* dst_ptr,
>+                       const uint16_t* src_ptr,
>                        int dst_width,
>                        int,
>                        int);
>-void ScaleFilterCols_C(uint8* dst_ptr,
>-                       const uint8* src_ptr,
>+void ScaleFilterCols_C(uint8_t* dst_ptr,
>+                       const uint8_t* src_ptr,
>                        int dst_width,
>                        int x,
>                        int dx);
>-void ScaleFilterCols_16_C(uint16* dst_ptr,
>-                          const uint16* src_ptr,
>+void ScaleFilterCols_16_C(uint16_t* dst_ptr,
>+                          const uint16_t* src_ptr,
>                           int dst_width,
>                           int x,
>                           int dx);
>-void ScaleFilterCols64_C(uint8* dst_ptr,
>-                         const uint8* src_ptr,
>+void ScaleFilterCols64_C(uint8_t* dst_ptr,
>+                         const uint8_t* src_ptr,
>                          int dst_width,
>-                         int x,
>+                         int x32,
>                          int dx);
>-void ScaleFilterCols64_16_C(uint16* dst_ptr,
>-                            const uint16* src_ptr,
>+void ScaleFilterCols64_16_C(uint16_t* dst_ptr,
>+                            const uint16_t* src_ptr,
>                             int dst_width,
>-                            int x,
>+                            int x32,
>                             int dx);
>-void ScaleRowDown38_C(const uint8* src_ptr,
>+void ScaleRowDown38_C(const uint8_t* src_ptr,
>                       ptrdiff_t src_stride,
>-                      uint8* dst,
>+                      uint8_t* dst,
>                       int dst_width);
>-void ScaleRowDown38_16_C(const uint16* src_ptr,
>+void ScaleRowDown38_16_C(const uint16_t* src_ptr,
>                          ptrdiff_t src_stride,
>-                         uint16* dst,
>+                         uint16_t* dst,
>                          int dst_width);
>-void ScaleRowDown38_3_Box_C(const uint8* src_ptr,
>+void ScaleRowDown38_3_Box_C(const uint8_t* src_ptr,
>                             ptrdiff_t src_stride,
>-                            uint8* dst_ptr,
>+                            uint8_t* dst_ptr,
>                             int dst_width);
>-void ScaleRowDown38_3_Box_16_C(const uint16* src_ptr,
>+void ScaleRowDown38_3_Box_16_C(const uint16_t* src_ptr,
>                                ptrdiff_t src_stride,
>-                               uint16* dst_ptr,
>+                               uint16_t* dst_ptr,
>                                int dst_width);
>-void ScaleRowDown38_2_Box_C(const uint8* src_ptr,
>+void ScaleRowDown38_2_Box_C(const uint8_t* src_ptr,
>                             ptrdiff_t src_stride,
>-                            uint8* dst_ptr,
>+                            uint8_t* dst_ptr,
>                             int dst_width);
>-void ScaleRowDown38_2_Box_16_C(const uint16* src_ptr,
>+void ScaleRowDown38_2_Box_16_C(const uint16_t* src_ptr,
>                                ptrdiff_t src_stride,
>-                               uint16* dst_ptr,
>+                               uint16_t* dst_ptr,
>                                int dst_width);
>-void ScaleAddRow_C(const uint8* src_ptr, uint16* dst_ptr, int src_width);
>-void ScaleAddRow_16_C(const uint16* src_ptr, uint32* dst_ptr, int src_width);
>-void ScaleARGBRowDown2_C(const uint8* src_argb,
>+void ScaleAddRow_C(const uint8_t* src_ptr, uint16_t* dst_ptr, int src_width);
>+void ScaleAddRow_16_C(const uint16_t* src_ptr,
>+                      uint32_t* dst_ptr,
>+                      int src_width);
>+void ScaleARGBRowDown2_C(const uint8_t* src_argb,
>                          ptrdiff_t src_stride,
>-                         uint8* dst_argb,
>+                         uint8_t* dst_argb,
>                          int dst_width);
>-void ScaleARGBRowDown2Linear_C(const uint8* src_argb,
>+void ScaleARGBRowDown2Linear_C(const uint8_t* src_argb,
>                                ptrdiff_t src_stride,
>-                               uint8* dst_argb,
>+                               uint8_t* dst_argb,
>                                int dst_width);
>-void ScaleARGBRowDown2Box_C(const uint8* src_argb,
>+void ScaleARGBRowDown2Box_C(const uint8_t* src_argb,
>                             ptrdiff_t src_stride,
>-                            uint8* dst_argb,
>+                            uint8_t* dst_argb,
>                             int dst_width);
>-void ScaleARGBRowDownEven_C(const uint8* src_argb,
>+void ScaleARGBRowDownEven_C(const uint8_t* src_argb,
>                             ptrdiff_t src_stride,
>                             int src_stepx,
>-                            uint8* dst_argb,
>+                            uint8_t* dst_argb,
>                             int dst_width);
>-void ScaleARGBRowDownEvenBox_C(const uint8* src_argb,
>+void ScaleARGBRowDownEvenBox_C(const uint8_t* src_argb,
>                                ptrdiff_t src_stride,
>                                int src_stepx,
>-                               uint8* dst_argb,
>+                               uint8_t* dst_argb,
>                                int dst_width);
>-void ScaleARGBCols_C(uint8* dst_argb,
>-                     const uint8* src_argb,
>+void ScaleARGBCols_C(uint8_t* dst_argb,
>+                     const uint8_t* src_argb,
>                      int dst_width,
>                      int x,
>                      int dx);
>-void ScaleARGBCols64_C(uint8* dst_argb,
>-                       const uint8* src_argb,
>+void ScaleARGBCols64_C(uint8_t* dst_argb,
>+                       const uint8_t* src_argb,
>                        int dst_width,
>-                       int x,
>+                       int x32,
>                        int dx);
>-void ScaleARGBColsUp2_C(uint8* dst_argb,
>-                        const uint8* src_argb,
>+void ScaleARGBColsUp2_C(uint8_t* dst_argb,
>+                        const uint8_t* src_argb,
>                         int dst_width,
>                         int,
>                         int);
>-void ScaleARGBFilterCols_C(uint8* dst_argb,
>-                           const uint8* src_argb,
>+void ScaleARGBFilterCols_C(uint8_t* dst_argb,
>+                           const uint8_t* src_argb,
>                            int dst_width,
>                            int x,
>                            int dx);
>-void ScaleARGBFilterCols64_C(uint8* dst_argb,
>-                             const uint8* src_argb,
>+void ScaleARGBFilterCols64_C(uint8_t* dst_argb,
>+                             const uint8_t* src_argb,
>                              int dst_width,
>-                             int x,
>+                             int x32,
>                              int dx);
> 
> // Specialized scalers for x86.
>-void ScaleRowDown2_SSSE3(const uint8* src_ptr,
>+void ScaleRowDown2_SSSE3(const uint8_t* src_ptr,
>                          ptrdiff_t src_stride,
>-                         uint8* dst_ptr,
>+                         uint8_t* dst_ptr,
>                          int dst_width);
>-void ScaleRowDown2Linear_SSSE3(const uint8* src_ptr,
>+void ScaleRowDown2Linear_SSSE3(const uint8_t* src_ptr,
>                                ptrdiff_t src_stride,
>-                               uint8* dst_ptr,
>+                               uint8_t* dst_ptr,
>                                int dst_width);
>-void ScaleRowDown2Box_SSSE3(const uint8* src_ptr,
>+void ScaleRowDown2Box_SSSE3(const uint8_t* src_ptr,
>                             ptrdiff_t src_stride,
>-                            uint8* dst_ptr,
>+                            uint8_t* dst_ptr,
>                             int dst_width);
>-void ScaleRowDown2_AVX2(const uint8* src_ptr,
>+void ScaleRowDown2_AVX2(const uint8_t* src_ptr,
>                         ptrdiff_t src_stride,
>-                        uint8* dst_ptr,
>+                        uint8_t* dst_ptr,
>                         int dst_width);
>-void ScaleRowDown2Linear_AVX2(const uint8* src_ptr,
>+void ScaleRowDown2Linear_AVX2(const uint8_t* src_ptr,
>                               ptrdiff_t src_stride,
>-                              uint8* dst_ptr,
>+                              uint8_t* dst_ptr,
>                               int dst_width);
>-void ScaleRowDown2Box_AVX2(const uint8* src_ptr,
>+void ScaleRowDown2Box_AVX2(const uint8_t* src_ptr,
>                            ptrdiff_t src_stride,
>-                           uint8* dst_ptr,
>+                           uint8_t* dst_ptr,
>                            int dst_width);
>-void ScaleRowDown4_SSSE3(const uint8* src_ptr,
>+void ScaleRowDown4_SSSE3(const uint8_t* src_ptr,
>                          ptrdiff_t src_stride,
>-                         uint8* dst_ptr,
>+                         uint8_t* dst_ptr,
>                          int dst_width);
>-void ScaleRowDown4Box_SSSE3(const uint8* src_ptr,
>+void ScaleRowDown4Box_SSSE3(const uint8_t* src_ptr,
>                             ptrdiff_t src_stride,
>-                            uint8* dst_ptr,
>+                            uint8_t* dst_ptr,
>                             int dst_width);
>-void ScaleRowDown4_AVX2(const uint8* src_ptr,
>+void ScaleRowDown4_AVX2(const uint8_t* src_ptr,
>                         ptrdiff_t src_stride,
>-                        uint8* dst_ptr,
>+                        uint8_t* dst_ptr,
>                         int dst_width);
>-void ScaleRowDown4Box_AVX2(const uint8* src_ptr,
>+void ScaleRowDown4Box_AVX2(const uint8_t* src_ptr,
>                            ptrdiff_t src_stride,
>-                           uint8* dst_ptr,
>+                           uint8_t* dst_ptr,
>                            int dst_width);
> 
>-void ScaleRowDown34_SSSE3(const uint8* src_ptr,
>+void ScaleRowDown34_SSSE3(const uint8_t* src_ptr,
>                           ptrdiff_t src_stride,
>-                          uint8* dst_ptr,
>+                          uint8_t* dst_ptr,
>                           int dst_width);
>-void ScaleRowDown34_1_Box_SSSE3(const uint8* src_ptr,
>+void ScaleRowDown34_1_Box_SSSE3(const uint8_t* src_ptr,
>                                 ptrdiff_t src_stride,
>-                                uint8* dst_ptr,
>+                                uint8_t* dst_ptr,
>                                 int dst_width);
>-void ScaleRowDown34_0_Box_SSSE3(const uint8* src_ptr,
>+void ScaleRowDown34_0_Box_SSSE3(const uint8_t* src_ptr,
>                                 ptrdiff_t src_stride,
>-                                uint8* dst_ptr,
>+                                uint8_t* dst_ptr,
>                                 int dst_width);
>-void ScaleRowDown38_SSSE3(const uint8* src_ptr,
>+void ScaleRowDown38_SSSE3(const uint8_t* src_ptr,
>                           ptrdiff_t src_stride,
>-                          uint8* dst_ptr,
>+                          uint8_t* dst_ptr,
>                           int dst_width);
>-void ScaleRowDown38_3_Box_SSSE3(const uint8* src_ptr,
>+void ScaleRowDown38_3_Box_SSSE3(const uint8_t* src_ptr,
>                                 ptrdiff_t src_stride,
>-                                uint8* dst_ptr,
>+                                uint8_t* dst_ptr,
>                                 int dst_width);
>-void ScaleRowDown38_2_Box_SSSE3(const uint8* src_ptr,
>+void ScaleRowDown38_2_Box_SSSE3(const uint8_t* src_ptr,
>                                 ptrdiff_t src_stride,
>-                                uint8* dst_ptr,
>+                                uint8_t* dst_ptr,
>                                 int dst_width);
>-void ScaleRowDown2_Any_SSSE3(const uint8* src_ptr,
>+void ScaleRowDown2_Any_SSSE3(const uint8_t* src_ptr,
>                              ptrdiff_t src_stride,
>-                             uint8* dst_ptr,
>+                             uint8_t* dst_ptr,
>                              int dst_width);
>-void ScaleRowDown2Linear_Any_SSSE3(const uint8* src_ptr,
>+void ScaleRowDown2Linear_Any_SSSE3(const uint8_t* src_ptr,
>                                    ptrdiff_t src_stride,
>-                                   uint8* dst_ptr,
>+                                   uint8_t* dst_ptr,
>                                    int dst_width);
>-void ScaleRowDown2Box_Any_SSSE3(const uint8* src_ptr,
>+void ScaleRowDown2Box_Any_SSSE3(const uint8_t* src_ptr,
>                                 ptrdiff_t src_stride,
>-                                uint8* dst_ptr,
>+                                uint8_t* dst_ptr,
>                                 int dst_width);
>-void ScaleRowDown2Box_Odd_SSSE3(const uint8* src_ptr,
>+void ScaleRowDown2Box_Odd_SSSE3(const uint8_t* src_ptr,
>                                 ptrdiff_t src_stride,
>-                                uint8* dst_ptr,
>+                                uint8_t* dst_ptr,
>                                 int dst_width);
>-void ScaleRowDown2_Any_AVX2(const uint8* src_ptr,
>+void ScaleRowDown2_Any_AVX2(const uint8_t* src_ptr,
>                             ptrdiff_t src_stride,
>-                            uint8* dst_ptr,
>+                            uint8_t* dst_ptr,
>                             int dst_width);
>-void ScaleRowDown2Linear_Any_AVX2(const uint8* src_ptr,
>+void ScaleRowDown2Linear_Any_AVX2(const uint8_t* src_ptr,
>                                   ptrdiff_t src_stride,
>-                                  uint8* dst_ptr,
>+                                  uint8_t* dst_ptr,
>                                   int dst_width);
>-void ScaleRowDown2Box_Any_AVX2(const uint8* src_ptr,
>+void ScaleRowDown2Box_Any_AVX2(const uint8_t* src_ptr,
>                                ptrdiff_t src_stride,
>-                               uint8* dst_ptr,
>+                               uint8_t* dst_ptr,
>                                int dst_width);
>-void ScaleRowDown2Box_Odd_AVX2(const uint8* src_ptr,
>+void ScaleRowDown2Box_Odd_AVX2(const uint8_t* src_ptr,
>                                ptrdiff_t src_stride,
>-                               uint8* dst_ptr,
>+                               uint8_t* dst_ptr,
>                                int dst_width);
>-void ScaleRowDown4_Any_SSSE3(const uint8* src_ptr,
>+void ScaleRowDown4_Any_SSSE3(const uint8_t* src_ptr,
>                              ptrdiff_t src_stride,
>-                             uint8* dst_ptr,
>+                             uint8_t* dst_ptr,
>                              int dst_width);
>-void ScaleRowDown4Box_Any_SSSE3(const uint8* src_ptr,
>+void ScaleRowDown4Box_Any_SSSE3(const uint8_t* src_ptr,
>                                 ptrdiff_t src_stride,
>-                                uint8* dst_ptr,
>+                                uint8_t* dst_ptr,
>                                 int dst_width);
>-void ScaleRowDown4_Any_AVX2(const uint8* src_ptr,
>+void ScaleRowDown4_Any_AVX2(const uint8_t* src_ptr,
>                             ptrdiff_t src_stride,
>-                            uint8* dst_ptr,
>+                            uint8_t* dst_ptr,
>                             int dst_width);
>-void ScaleRowDown4Box_Any_AVX2(const uint8* src_ptr,
>+void ScaleRowDown4Box_Any_AVX2(const uint8_t* src_ptr,
>                                ptrdiff_t src_stride,
>-                               uint8* dst_ptr,
>+                               uint8_t* dst_ptr,
>                                int dst_width);
> 
>-void ScaleRowDown34_Any_SSSE3(const uint8* src_ptr,
>+void ScaleRowDown34_Any_SSSE3(const uint8_t* src_ptr,
>                               ptrdiff_t src_stride,
>-                              uint8* dst_ptr,
>+                              uint8_t* dst_ptr,
>                               int dst_width);
>-void ScaleRowDown34_1_Box_Any_SSSE3(const uint8* src_ptr,
>+void ScaleRowDown34_1_Box_Any_SSSE3(const uint8_t* src_ptr,
>                                     ptrdiff_t src_stride,
>-                                    uint8* dst_ptr,
>+                                    uint8_t* dst_ptr,
>                                     int dst_width);
>-void ScaleRowDown34_0_Box_Any_SSSE3(const uint8* src_ptr,
>+void ScaleRowDown34_0_Box_Any_SSSE3(const uint8_t* src_ptr,
>                                     ptrdiff_t src_stride,
>-                                    uint8* dst_ptr,
>+                                    uint8_t* dst_ptr,
>                                     int dst_width);
>-void ScaleRowDown38_Any_SSSE3(const uint8* src_ptr,
>+void ScaleRowDown38_Any_SSSE3(const uint8_t* src_ptr,
>                               ptrdiff_t src_stride,
>-                              uint8* dst_ptr,
>+                              uint8_t* dst_ptr,
>                               int dst_width);
>-void ScaleRowDown38_3_Box_Any_SSSE3(const uint8* src_ptr,
>+void ScaleRowDown38_3_Box_Any_SSSE3(const uint8_t* src_ptr,
>                                     ptrdiff_t src_stride,
>-                                    uint8* dst_ptr,
>+                                    uint8_t* dst_ptr,
>                                     int dst_width);
>-void ScaleRowDown38_2_Box_Any_SSSE3(const uint8* src_ptr,
>+void ScaleRowDown38_2_Box_Any_SSSE3(const uint8_t* src_ptr,
>                                     ptrdiff_t src_stride,
>-                                    uint8* dst_ptr,
>+                                    uint8_t* dst_ptr,
>                                     int dst_width);
> 
>-void ScaleAddRow_SSE2(const uint8* src_ptr, uint16* dst_ptr, int src_width);
>-void ScaleAddRow_AVX2(const uint8* src_ptr, uint16* dst_ptr, int src_width);
>-void ScaleAddRow_Any_SSE2(const uint8* src_ptr, uint16* dst_ptr, int src_width);
>-void ScaleAddRow_Any_AVX2(const uint8* src_ptr, uint16* dst_ptr, int src_width);
>-
>-void ScaleFilterCols_SSSE3(uint8* dst_ptr,
>-                           const uint8* src_ptr,
>+void ScaleAddRow_SSE2(const uint8_t* src_ptr, uint16_t* dst_ptr, int src_width);
>+void ScaleAddRow_AVX2(const uint8_t* src_ptr, uint16_t* dst_ptr, int src_width);
>+void ScaleAddRow_Any_SSE2(const uint8_t* src_ptr,
>+                          uint16_t* dst_ptr,
>+                          int src_width);
>+void ScaleAddRow_Any_AVX2(const uint8_t* src_ptr,
>+                          uint16_t* dst_ptr,
>+                          int src_width);
>+
>+void ScaleFilterCols_SSSE3(uint8_t* dst_ptr,
>+                           const uint8_t* src_ptr,
>                            int dst_width,
>                            int x,
>                            int dx);
>-void ScaleColsUp2_SSE2(uint8* dst_ptr,
>-                       const uint8* src_ptr,
>+void ScaleColsUp2_SSE2(uint8_t* dst_ptr,
>+                       const uint8_t* src_ptr,
>                        int dst_width,
>                        int x,
>                        int dx);
> 
> // ARGB Column functions
>-void ScaleARGBCols_SSE2(uint8* dst_argb,
>-                        const uint8* src_argb,
>+void ScaleARGBCols_SSE2(uint8_t* dst_argb,
>+                        const uint8_t* src_argb,
>                         int dst_width,
>                         int x,
>                         int dx);
>-void ScaleARGBFilterCols_SSSE3(uint8* dst_argb,
>-                               const uint8* src_argb,
>+void ScaleARGBFilterCols_SSSE3(uint8_t* dst_argb,
>+                               const uint8_t* src_argb,
>                                int dst_width,
>                                int x,
>                                int dx);
>-void ScaleARGBColsUp2_SSE2(uint8* dst_argb,
>-                           const uint8* src_argb,
>+void ScaleARGBColsUp2_SSE2(uint8_t* dst_argb,
>+                           const uint8_t* src_argb,
>                            int dst_width,
>                            int x,
>                            int dx);
>-void ScaleARGBFilterCols_NEON(uint8* dst_argb,
>-                              const uint8* src_argb,
>+void ScaleARGBFilterCols_NEON(uint8_t* dst_argb,
>+                              const uint8_t* src_argb,
>                               int dst_width,
>                               int x,
>                               int dx);
>-void ScaleARGBCols_NEON(uint8* dst_argb,
>-                        const uint8* src_argb,
>+void ScaleARGBCols_NEON(uint8_t* dst_argb,
>+                        const uint8_t* src_argb,
>                         int dst_width,
>                         int x,
>                         int dx);
>-void ScaleARGBFilterCols_Any_NEON(uint8* dst_argb,
>-                                  const uint8* src_argb,
>+void ScaleARGBFilterCols_Any_NEON(uint8_t* dst_ptr,
>+                                  const uint8_t* src_ptr,
>                                   int dst_width,
>                                   int x,
>                                   int dx);
>-void ScaleARGBCols_Any_NEON(uint8* dst_argb,
>-                            const uint8* src_argb,
>+void ScaleARGBCols_Any_NEON(uint8_t* dst_ptr,
>+                            const uint8_t* src_ptr,
>                             int dst_width,
>                             int x,
>                             int dx);
>-void ScaleARGBFilterCols_MSA(uint8* dst_argb,
>-                             const uint8* src_argb,
>+void ScaleARGBFilterCols_MSA(uint8_t* dst_argb,
>+                             const uint8_t* src_argb,
>                              int dst_width,
>                              int x,
>                              int dx);
>-void ScaleARGBCols_MSA(uint8* dst_argb,
>-                       const uint8* src_argb,
>+void ScaleARGBCols_MSA(uint8_t* dst_argb,
>+                       const uint8_t* src_argb,
>                        int dst_width,
>                        int x,
>                        int dx);
>-void ScaleARGBFilterCols_Any_MSA(uint8* dst_argb,
>-                                 const uint8* src_argb,
>+void ScaleARGBFilterCols_Any_MSA(uint8_t* dst_ptr,
>+                                 const uint8_t* src_ptr,
>                                  int dst_width,
>                                  int x,
>                                  int dx);
>-void ScaleARGBCols_Any_MSA(uint8* dst_argb,
>-                           const uint8* src_argb,
>+void ScaleARGBCols_Any_MSA(uint8_t* dst_ptr,
>+                           const uint8_t* src_ptr,
>                            int dst_width,
>                            int x,
>                            int dx);
> 
> // ARGB Row functions
>-void ScaleARGBRowDown2_SSE2(const uint8* src_argb,
>+void ScaleARGBRowDown2_SSE2(const uint8_t* src_argb,
>                             ptrdiff_t src_stride,
>-                            uint8* dst_argb,
>+                            uint8_t* dst_argb,
>                             int dst_width);
>-void ScaleARGBRowDown2Linear_SSE2(const uint8* src_argb,
>+void ScaleARGBRowDown2Linear_SSE2(const uint8_t* src_argb,
>                                   ptrdiff_t src_stride,
>-                                  uint8* dst_argb,
>+                                  uint8_t* dst_argb,
>                                   int dst_width);
>-void ScaleARGBRowDown2Box_SSE2(const uint8* src_argb,
>+void ScaleARGBRowDown2Box_SSE2(const uint8_t* src_argb,
>                                ptrdiff_t src_stride,
>-                               uint8* dst_argb,
>+                               uint8_t* dst_argb,
>                                int dst_width);
>-void ScaleARGBRowDown2_NEON(const uint8* src_ptr,
>+void ScaleARGBRowDown2_NEON(const uint8_t* src_ptr,
>                             ptrdiff_t src_stride,
>-                            uint8* dst,
>+                            uint8_t* dst,
>                             int dst_width);
>-void ScaleARGBRowDown2Linear_NEON(const uint8* src_argb,
>+void ScaleARGBRowDown2Linear_NEON(const uint8_t* src_argb,
>                                   ptrdiff_t src_stride,
>-                                  uint8* dst_argb,
>+                                  uint8_t* dst_argb,
>                                   int dst_width);
>-void ScaleARGBRowDown2Box_NEON(const uint8* src_ptr,
>+void ScaleARGBRowDown2Box_NEON(const uint8_t* src_ptr,
>                                ptrdiff_t src_stride,
>-                               uint8* dst,
>+                               uint8_t* dst,
>                                int dst_width);
> void ScaleARGBRowDown2_MSA(const uint8_t* src_argb,
>                            ptrdiff_t src_stride,
>@@ -608,274 +607,231 @@ void ScaleARGBRowDown2Box_MSA(const uint8_t* src_argb,
>                               ptrdiff_t src_stride,
>                               uint8_t* dst_argb,
>                               int dst_width);
>-void ScaleARGBRowDown2_Any_SSE2(const uint8* src_argb,
>+void ScaleARGBRowDown2_Any_SSE2(const uint8_t* src_ptr,
>                                 ptrdiff_t src_stride,
>-                                uint8* dst_argb,
>+                                uint8_t* dst_ptr,
>                                 int dst_width);
>-void ScaleARGBRowDown2Linear_Any_SSE2(const uint8* src_argb,
>+void ScaleARGBRowDown2Linear_Any_SSE2(const uint8_t* src_ptr,
>                                       ptrdiff_t src_stride,
>-                                      uint8* dst_argb,
>+                                      uint8_t* dst_ptr,
>                                       int dst_width);
>-void ScaleARGBRowDown2Box_Any_SSE2(const uint8* src_argb,
>+void ScaleARGBRowDown2Box_Any_SSE2(const uint8_t* src_ptr,
>                                    ptrdiff_t src_stride,
>-                                   uint8* dst_argb,
>+                                   uint8_t* dst_ptr,
>                                    int dst_width);
>-void ScaleARGBRowDown2_Any_NEON(const uint8* src_ptr,
>+void ScaleARGBRowDown2_Any_NEON(const uint8_t* src_ptr,
>                                 ptrdiff_t src_stride,
>-                                uint8* dst,
>+                                uint8_t* dst_ptr,
>                                 int dst_width);
>-void ScaleARGBRowDown2Linear_Any_NEON(const uint8* src_argb,
>+void ScaleARGBRowDown2Linear_Any_NEON(const uint8_t* src_ptr,
>                                       ptrdiff_t src_stride,
>-                                      uint8* dst_argb,
>+                                      uint8_t* dst_ptr,
>                                       int dst_width);
>-void ScaleARGBRowDown2Box_Any_NEON(const uint8* src_ptr,
>+void ScaleARGBRowDown2Box_Any_NEON(const uint8_t* src_ptr,
>                                    ptrdiff_t src_stride,
>-                                   uint8* dst,
>+                                   uint8_t* dst_ptr,
>                                    int dst_width);
>-void ScaleARGBRowDown2_Any_MSA(const uint8_t* src_argb,
>+void ScaleARGBRowDown2_Any_MSA(const uint8_t* src_ptr,
>                                ptrdiff_t src_stride,
>-                               uint8_t* dst_argb,
>+                               uint8_t* dst_ptr,
>                                int dst_width);
>-void ScaleARGBRowDown2Linear_Any_MSA(const uint8_t* src_argb,
>+void ScaleARGBRowDown2Linear_Any_MSA(const uint8_t* src_ptr,
>                                      ptrdiff_t src_stride,
>-                                     uint8_t* dst_argb,
>+                                     uint8_t* dst_ptr,
>                                      int dst_width);
>-void ScaleARGBRowDown2Box_Any_MSA(const uint8_t* src_argb,
>+void ScaleARGBRowDown2Box_Any_MSA(const uint8_t* src_ptr,
>                                   ptrdiff_t src_stride,
>-                                  uint8_t* dst_argb,
>+                                  uint8_t* dst_ptr,
>                                   int dst_width);
> 
>-void ScaleARGBRowDownEven_SSE2(const uint8* src_argb,
>+void ScaleARGBRowDownEven_SSE2(const uint8_t* src_argb,
>                                ptrdiff_t src_stride,
>                                int src_stepx,
>-                               uint8* dst_argb,
>+                               uint8_t* dst_argb,
>                                int dst_width);
>-void ScaleARGBRowDownEvenBox_SSE2(const uint8* src_argb,
>+void ScaleARGBRowDownEvenBox_SSE2(const uint8_t* src_argb,
>                                   ptrdiff_t src_stride,
>                                   int src_stepx,
>-                                  uint8* dst_argb,
>+                                  uint8_t* dst_argb,
>                                   int dst_width);
>-void ScaleARGBRowDownEven_NEON(const uint8* src_argb,
>+void ScaleARGBRowDownEven_NEON(const uint8_t* src_argb,
>                                ptrdiff_t src_stride,
>                                int src_stepx,
>-                               uint8* dst_argb,
>+                               uint8_t* dst_argb,
>                                int dst_width);
>-void ScaleARGBRowDownEvenBox_NEON(const uint8* src_argb,
>+void ScaleARGBRowDownEvenBox_NEON(const uint8_t* src_argb,
>                                   ptrdiff_t src_stride,
>                                   int src_stepx,
>-                                  uint8* dst_argb,
>+                                  uint8_t* dst_argb,
>                                   int dst_width);
> void ScaleARGBRowDownEven_MSA(const uint8_t* src_argb,
>                               ptrdiff_t src_stride,
>                               int32_t src_stepx,
>                               uint8_t* dst_argb,
>                               int dst_width);
>-void ScaleARGBRowDownEvenBox_MSA(const uint8* src_argb,
>+void ScaleARGBRowDownEvenBox_MSA(const uint8_t* src_argb,
>                                  ptrdiff_t src_stride,
>                                  int src_stepx,
>-                                 uint8* dst_argb,
>+                                 uint8_t* dst_argb,
>                                  int dst_width);
>-void ScaleARGBRowDownEven_Any_SSE2(const uint8* src_argb,
>+void ScaleARGBRowDownEven_Any_SSE2(const uint8_t* src_ptr,
>                                    ptrdiff_t src_stride,
>                                    int src_stepx,
>-                                   uint8* dst_argb,
>+                                   uint8_t* dst_ptr,
>                                    int dst_width);
>-void ScaleARGBRowDownEvenBox_Any_SSE2(const uint8* src_argb,
>+void ScaleARGBRowDownEvenBox_Any_SSE2(const uint8_t* src_ptr,
>                                       ptrdiff_t src_stride,
>                                       int src_stepx,
>-                                      uint8* dst_argb,
>+                                      uint8_t* dst_ptr,
>                                       int dst_width);
>-void ScaleARGBRowDownEven_Any_NEON(const uint8* src_argb,
>+void ScaleARGBRowDownEven_Any_NEON(const uint8_t* src_ptr,
>                                    ptrdiff_t src_stride,
>                                    int src_stepx,
>-                                   uint8* dst_argb,
>+                                   uint8_t* dst_ptr,
>                                    int dst_width);
>-void ScaleARGBRowDownEvenBox_Any_NEON(const uint8* src_argb,
>+void ScaleARGBRowDownEvenBox_Any_NEON(const uint8_t* src_ptr,
>                                       ptrdiff_t src_stride,
>                                       int src_stepx,
>-                                      uint8* dst_argb,
>+                                      uint8_t* dst_ptr,
>                                       int dst_width);
>-void ScaleARGBRowDownEven_Any_MSA(const uint8_t* src_argb,
>+void ScaleARGBRowDownEven_Any_MSA(const uint8_t* src_ptr,
>                                   ptrdiff_t src_stride,
>                                   int32_t src_stepx,
>-                                  uint8_t* dst_argb,
>+                                  uint8_t* dst_ptr,
>                                   int dst_width);
>-void ScaleARGBRowDownEvenBox_Any_MSA(const uint8* src_argb,
>+void ScaleARGBRowDownEvenBox_Any_MSA(const uint8_t* src_ptr,
>                                      ptrdiff_t src_stride,
>                                      int src_stepx,
>-                                     uint8* dst_argb,
>+                                     uint8_t* dst_ptr,
>                                      int dst_width);
> 
> // ScaleRowDown2Box also used by planar functions
> // NEON downscalers with interpolation.
> 
> // Note - not static due to reuse in convert for 444 to 420.
>-void ScaleRowDown2_NEON(const uint8* src_ptr,
>+void ScaleRowDown2_NEON(const uint8_t* src_ptr,
>                         ptrdiff_t src_stride,
>-                        uint8* dst,
>+                        uint8_t* dst,
>                         int dst_width);
>-void ScaleRowDown2Linear_NEON(const uint8* src_ptr,
>+void ScaleRowDown2Linear_NEON(const uint8_t* src_ptr,
>                               ptrdiff_t src_stride,
>-                              uint8* dst,
>+                              uint8_t* dst,
>                               int dst_width);
>-void ScaleRowDown2Box_NEON(const uint8* src_ptr,
>+void ScaleRowDown2Box_NEON(const uint8_t* src_ptr,
>                            ptrdiff_t src_stride,
>-                           uint8* dst,
>+                           uint8_t* dst,
>                            int dst_width);
> 
>-void ScaleRowDown4_NEON(const uint8* src_ptr,
>+void ScaleRowDown4_NEON(const uint8_t* src_ptr,
>                         ptrdiff_t src_stride,
>-                        uint8* dst_ptr,
>+                        uint8_t* dst_ptr,
>                         int dst_width);
>-void ScaleRowDown4Box_NEON(const uint8* src_ptr,
>+void ScaleRowDown4Box_NEON(const uint8_t* src_ptr,
>                            ptrdiff_t src_stride,
>-                           uint8* dst_ptr,
>+                           uint8_t* dst_ptr,
>                            int dst_width);
> 
> // Down scale from 4 to 3 pixels. Use the neon multilane read/write
> //  to load up the every 4th pixel into a 4 different registers.
> // Point samples 32 pixels to 24 pixels.
>-void ScaleRowDown34_NEON(const uint8* src_ptr,
>+void ScaleRowDown34_NEON(const uint8_t* src_ptr,
>                          ptrdiff_t src_stride,
>-                         uint8* dst_ptr,
>+                         uint8_t* dst_ptr,
>                          int dst_width);
>-void ScaleRowDown34_0_Box_NEON(const uint8* src_ptr,
>+void ScaleRowDown34_0_Box_NEON(const uint8_t* src_ptr,
>                                ptrdiff_t src_stride,
>-                               uint8* dst_ptr,
>+                               uint8_t* dst_ptr,
>                                int dst_width);
>-void ScaleRowDown34_1_Box_NEON(const uint8* src_ptr,
>+void ScaleRowDown34_1_Box_NEON(const uint8_t* src_ptr,
>                                ptrdiff_t src_stride,
>-                               uint8* dst_ptr,
>+                               uint8_t* dst_ptr,
>                                int dst_width);
> 
> // 32 -> 12
>-void ScaleRowDown38_NEON(const uint8* src_ptr,
>+void ScaleRowDown38_NEON(const uint8_t* src_ptr,
>                          ptrdiff_t src_stride,
>-                         uint8* dst_ptr,
>+                         uint8_t* dst_ptr,
>                          int dst_width);
> // 32x3 -> 12x1
>-void ScaleRowDown38_3_Box_NEON(const uint8* src_ptr,
>+void ScaleRowDown38_3_Box_NEON(const uint8_t* src_ptr,
>                                ptrdiff_t src_stride,
>-                               uint8* dst_ptr,
>+                               uint8_t* dst_ptr,
>                                int dst_width);
> // 32x2 -> 12x1
>-void ScaleRowDown38_2_Box_NEON(const uint8* src_ptr,
>+void ScaleRowDown38_2_Box_NEON(const uint8_t* src_ptr,
>                                ptrdiff_t src_stride,
>-                               uint8* dst_ptr,
>+                               uint8_t* dst_ptr,
>                                int dst_width);
> 
>-void ScaleRowDown2_Any_NEON(const uint8* src_ptr,
>+void ScaleRowDown2_Any_NEON(const uint8_t* src_ptr,
>                             ptrdiff_t src_stride,
>-                            uint8* dst,
>+                            uint8_t* dst_ptr,
>                             int dst_width);
>-void ScaleRowDown2Linear_Any_NEON(const uint8* src_ptr,
>+void ScaleRowDown2Linear_Any_NEON(const uint8_t* src_ptr,
>                                   ptrdiff_t src_stride,
>-                                  uint8* dst,
>+                                  uint8_t* dst_ptr,
>                                   int dst_width);
>-void ScaleRowDown2Box_Any_NEON(const uint8* src_ptr,
>+void ScaleRowDown2Box_Any_NEON(const uint8_t* src_ptr,
>                                ptrdiff_t src_stride,
>-                               uint8* dst,
>+                               uint8_t* dst_ptr,
>                                int dst_width);
>-void ScaleRowDown2Box_Odd_NEON(const uint8* src_ptr,
>+void ScaleRowDown2Box_Odd_NEON(const uint8_t* src_ptr,
>                                ptrdiff_t src_stride,
>-                               uint8* dst,
>+                               uint8_t* dst_ptr,
>                                int dst_width);
>-void ScaleRowDown4_Any_NEON(const uint8* src_ptr,
>+void ScaleRowDown4_Any_NEON(const uint8_t* src_ptr,
>                             ptrdiff_t src_stride,
>-                            uint8* dst_ptr,
>+                            uint8_t* dst_ptr,
>                             int dst_width);
>-void ScaleRowDown4Box_Any_NEON(const uint8* src_ptr,
>+void ScaleRowDown4Box_Any_NEON(const uint8_t* src_ptr,
>                                ptrdiff_t src_stride,
>-                               uint8* dst_ptr,
>+                               uint8_t* dst_ptr,
>                                int dst_width);
>-void ScaleRowDown34_Any_NEON(const uint8* src_ptr,
>+void ScaleRowDown34_Any_NEON(const uint8_t* src_ptr,
>                              ptrdiff_t src_stride,
>-                             uint8* dst_ptr,
>+                             uint8_t* dst_ptr,
>                              int dst_width);
>-void ScaleRowDown34_0_Box_Any_NEON(const uint8* src_ptr,
>+void ScaleRowDown34_0_Box_Any_NEON(const uint8_t* src_ptr,
>                                    ptrdiff_t src_stride,
>-                                   uint8* dst_ptr,
>+                                   uint8_t* dst_ptr,
>                                    int dst_width);
>-void ScaleRowDown34_1_Box_Any_NEON(const uint8* src_ptr,
>+void ScaleRowDown34_1_Box_Any_NEON(const uint8_t* src_ptr,
>                                    ptrdiff_t src_stride,
>-                                   uint8* dst_ptr,
>+                                   uint8_t* dst_ptr,
>                                    int dst_width);
> // 32 -> 12
>-void ScaleRowDown38_Any_NEON(const uint8* src_ptr,
>+void ScaleRowDown38_Any_NEON(const uint8_t* src_ptr,
>                              ptrdiff_t src_stride,
>-                             uint8* dst_ptr,
>+                             uint8_t* dst_ptr,
>                              int dst_width);
> // 32x3 -> 12x1
>-void ScaleRowDown38_3_Box_Any_NEON(const uint8* src_ptr,
>+void ScaleRowDown38_3_Box_Any_NEON(const uint8_t* src_ptr,
>                                    ptrdiff_t src_stride,
>-                                   uint8* dst_ptr,
>+                                   uint8_t* dst_ptr,
>                                    int dst_width);
> // 32x2 -> 12x1
>-void ScaleRowDown38_2_Box_Any_NEON(const uint8* src_ptr,
>+void ScaleRowDown38_2_Box_Any_NEON(const uint8_t* src_ptr,
>                                    ptrdiff_t src_stride,
>-                                   uint8* dst_ptr,
>+                                   uint8_t* dst_ptr,
>                                    int dst_width);
> 
>-void ScaleAddRow_NEON(const uint8* src_ptr, uint16* dst_ptr, int src_width);
>-void ScaleAddRow_Any_NEON(const uint8* src_ptr, uint16* dst_ptr, int src_width);
>+void ScaleAddRow_NEON(const uint8_t* src_ptr, uint16_t* dst_ptr, int src_width);
>+void ScaleAddRow_Any_NEON(const uint8_t* src_ptr,
>+                          uint16_t* dst_ptr,
>+                          int src_width);
> 
>-void ScaleFilterCols_NEON(uint8* dst_ptr,
>-                          const uint8* src_ptr,
>+void ScaleFilterCols_NEON(uint8_t* dst_ptr,
>+                          const uint8_t* src_ptr,
>                           int dst_width,
>                           int x,
>                           int dx);
> 
>-void ScaleFilterCols_Any_NEON(uint8* dst_ptr,
>-                              const uint8* src_ptr,
>+void ScaleFilterCols_Any_NEON(uint8_t* dst_ptr,
>+                              const uint8_t* src_ptr,
>                               int dst_width,
>                               int x,
>                               int dx);
> 
>-void ScaleRowDown2_DSPR2(const uint8* src_ptr,
>-                         ptrdiff_t src_stride,
>-                         uint8* dst,
>-                         int dst_width);
>-void ScaleRowDown2Box_DSPR2(const uint8* src_ptr,
>-                            ptrdiff_t src_stride,
>-                            uint8* dst,
>-                            int dst_width);
>-void ScaleRowDown4_DSPR2(const uint8* src_ptr,
>-                         ptrdiff_t src_stride,
>-                         uint8* dst,
>-                         int dst_width);
>-void ScaleRowDown4Box_DSPR2(const uint8* src_ptr,
>-                            ptrdiff_t src_stride,
>-                            uint8* dst,
>-                            int dst_width);
>-void ScaleRowDown34_DSPR2(const uint8* src_ptr,
>-                          ptrdiff_t src_stride,
>-                          uint8* dst,
>-                          int dst_width);
>-void ScaleRowDown34_0_Box_DSPR2(const uint8* src_ptr,
>-                                ptrdiff_t src_stride,
>-                                uint8* d,
>-                                int dst_width);
>-void ScaleRowDown34_1_Box_DSPR2(const uint8* src_ptr,
>-                                ptrdiff_t src_stride,
>-                                uint8* d,
>-                                int dst_width);
>-void ScaleRowDown38_DSPR2(const uint8* src_ptr,
>-                          ptrdiff_t src_stride,
>-                          uint8* dst,
>-                          int dst_width);
>-void ScaleRowDown38_2_Box_DSPR2(const uint8* src_ptr,
>-                                ptrdiff_t src_stride,
>-                                uint8* dst_ptr,
>-                                int dst_width);
>-void ScaleRowDown38_3_Box_DSPR2(const uint8* src_ptr,
>-                                ptrdiff_t src_stride,
>-                                uint8* dst_ptr,
>-                                int dst_width);
>-void ScaleAddRow_DSPR2(const uint8* src_ptr, uint16* dst_ptr, int src_width);
>-void ScaleAddRow_Any_DSPR2(const uint8* src_ptr,
>-                           uint16* dst_ptr,
>-                           int src_width);
>-
> void ScaleRowDown2_MSA(const uint8_t* src_ptr,
>                        ptrdiff_t src_stride,
>                        uint8_t* dst,
>@@ -909,47 +865,47 @@ void ScaleRowDown38_3_Box_MSA(const uint8_t* src_ptr,
>                               uint8_t* dst_ptr,
>                               int dst_width);
> void ScaleAddRow_MSA(const uint8_t* src_ptr, uint16_t* dst_ptr, int src_width);
>-void ScaleFilterCols_MSA(uint8* dst_ptr,
>-                         const uint8* src_ptr,
>+void ScaleFilterCols_MSA(uint8_t* dst_ptr,
>+                         const uint8_t* src_ptr,
>                          int dst_width,
>                          int x,
>                          int dx);
>-void ScaleRowDown34_MSA(const uint8* src_ptr,
>+void ScaleRowDown34_MSA(const uint8_t* src_ptr,
>                         ptrdiff_t src_stride,
>-                        uint8* dst_ptr,
>+                        uint8_t* dst,
>                         int dst_width);
>-void ScaleRowDown34_0_Box_MSA(const uint8* src_ptr,
>+void ScaleRowDown34_0_Box_MSA(const uint8_t* src_ptr,
>                               ptrdiff_t src_stride,
>-                              uint8* dst_ptr,
>+                              uint8_t* d,
>                               int dst_width);
>-void ScaleRowDown34_1_Box_MSA(const uint8* src_ptr,
>+void ScaleRowDown34_1_Box_MSA(const uint8_t* src_ptr,
>                               ptrdiff_t src_stride,
>-                              uint8* dst_ptr,
>+                              uint8_t* d,
>                               int dst_width);
> 
> void ScaleRowDown2_Any_MSA(const uint8_t* src_ptr,
>                            ptrdiff_t src_stride,
>-                           uint8_t* dst,
>+                           uint8_t* dst_ptr,
>                            int dst_width);
> void ScaleRowDown2Linear_Any_MSA(const uint8_t* src_ptr,
>                                  ptrdiff_t src_stride,
>-                                 uint8_t* dst,
>+                                 uint8_t* dst_ptr,
>                                  int dst_width);
> void ScaleRowDown2Box_Any_MSA(const uint8_t* src_ptr,
>                               ptrdiff_t src_stride,
>-                              uint8_t* dst,
>+                              uint8_t* dst_ptr,
>                               int dst_width);
> void ScaleRowDown4_Any_MSA(const uint8_t* src_ptr,
>                            ptrdiff_t src_stride,
>-                           uint8_t* dst,
>+                           uint8_t* dst_ptr,
>                            int dst_width);
> void ScaleRowDown4Box_Any_MSA(const uint8_t* src_ptr,
>                               ptrdiff_t src_stride,
>-                              uint8_t* dst,
>+                              uint8_t* dst_ptr,
>                               int dst_width);
> void ScaleRowDown38_Any_MSA(const uint8_t* src_ptr,
>                             ptrdiff_t src_stride,
>-                            uint8_t* dst,
>+                            uint8_t* dst_ptr,
>                             int dst_width);
> void ScaleRowDown38_2_Box_Any_MSA(const uint8_t* src_ptr,
>                                   ptrdiff_t src_stride,
>@@ -962,22 +918,22 @@ void ScaleRowDown38_3_Box_Any_MSA(const uint8_t* src_ptr,
> void ScaleAddRow_Any_MSA(const uint8_t* src_ptr,
>                          uint16_t* dst_ptr,
>                          int src_width);
>-void ScaleFilterCols_Any_MSA(uint8* dst_ptr,
>-                             const uint8* src_ptr,
>+void ScaleFilterCols_Any_MSA(uint8_t* dst_ptr,
>+                             const uint8_t* src_ptr,
>                              int dst_width,
>                              int x,
>                              int dx);
>-void ScaleRowDown34_Any_MSA(const uint8* src_ptr,
>+void ScaleRowDown34_Any_MSA(const uint8_t* src_ptr,
>                             ptrdiff_t src_stride,
>-                            uint8* dst_ptr,
>+                            uint8_t* dst_ptr,
>                             int dst_width);
>-void ScaleRowDown34_0_Box_Any_MSA(const uint8* src_ptr,
>+void ScaleRowDown34_0_Box_Any_MSA(const uint8_t* src_ptr,
>                                   ptrdiff_t src_stride,
>-                                  uint8* dst_ptr,
>+                                  uint8_t* dst_ptr,
>                                   int dst_width);
>-void ScaleRowDown34_1_Box_Any_MSA(const uint8* src_ptr,
>+void ScaleRowDown34_1_Box_Any_MSA(const uint8_t* src_ptr,
>                                   ptrdiff_t src_stride,
>-                                  uint8* dst_ptr,
>+                                  uint8_t* dst_ptr,
>                                   int dst_width);
> 
> #ifdef __cplusplus
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/version.h b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/version.h
>index 838c70f1349..21522cf351c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/version.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/version.h
>@@ -11,6 +11,6 @@
> #ifndef INCLUDE_LIBYUV_VERSION_H_
> #define INCLUDE_LIBYUV_VERSION_H_
> 
>-#define LIBYUV_VERSION 1678
>+#define LIBYUV_VERSION 1703
> 
> #endif  // INCLUDE_LIBYUV_VERSION_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/video_common.h b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/video_common.h
>index f3711c423ba..bcef378b5a4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/video_common.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/include/libyuv/video_common.h
>@@ -28,13 +28,13 @@ extern "C" {
> // Needs to be a macro otherwise the OS X compiler complains when the kFormat*
> // constants are used in a switch.
> #ifdef __cplusplus
>-#define FOURCC(a, b, c, d)                                    \
>-  ((static_cast<uint32>(a)) | (static_cast<uint32>(b) << 8) | \
>-   (static_cast<uint32>(c) << 16) | (static_cast<uint32>(d) << 24))
>+#define FOURCC(a, b, c, d)                                        \
>+  ((static_cast<uint32_t>(a)) | (static_cast<uint32_t>(b) << 8) | \
>+   (static_cast<uint32_t>(c) << 16) | (static_cast<uint32_t>(d) << 24))
> #else
>-#define FOURCC(a, b, c, d)                                 \
>-  (((uint32)(a)) | ((uint32)(b) << 8) |       /* NOLINT */ \
>-   ((uint32)(c) << 16) | ((uint32)(d) << 24)) /* NOLINT */
>+#define FOURCC(a, b, c, d)                                     \
>+  (((uint32_t)(a)) | ((uint32_t)(b) << 8) |       /* NOLINT */ \
>+   ((uint32_t)(c) << 16) | ((uint32_t)(d) << 24)) /* NOLINT */
> #endif
> 
> // Some pages discussing FourCC codes:
>@@ -49,25 +49,26 @@ extern "C" {
> // Secondary formats are converted in 2 steps.
> // Auxilliary formats call primary converters.
> enum FourCC {
>-  // 8 Primary YUV formats: 5 planar, 2 biplanar, 2 packed.
>+  // 9 Primary YUV formats: 5 planar, 2 biplanar, 2 packed.
>   FOURCC_I420 = FOURCC('I', '4', '2', '0'),
>   FOURCC_I422 = FOURCC('I', '4', '2', '2'),
>   FOURCC_I444 = FOURCC('I', '4', '4', '4'),
>-  FOURCC_I411 = FOURCC('I', '4', '1', '1'),  // deprecated.
>   FOURCC_I400 = FOURCC('I', '4', '0', '0'),
>   FOURCC_NV21 = FOURCC('N', 'V', '2', '1'),
>   FOURCC_NV12 = FOURCC('N', 'V', '1', '2'),
>   FOURCC_YUY2 = FOURCC('Y', 'U', 'Y', '2'),
>   FOURCC_UYVY = FOURCC('U', 'Y', 'V', 'Y'),
>+  FOURCC_H010 = FOURCC('H', '0', '1', '0'),  // unofficial fourcc. 10 bit lsb
> 
>   // 1 Secondary YUV format: row biplanar.
>   FOURCC_M420 = FOURCC('M', '4', '2', '0'),
>-  FOURCC_Q420 = FOURCC('Q', '4', '2', '0'),  // deprecated.
> 
>-  // 9 Primary RGB formats: 4 32 bpp, 2 24 bpp, 3 16 bpp.
>+  // 11 Primary RGB formats: 4 32 bpp, 2 24 bpp, 3 16 bpp, 1 10 bpc
>   FOURCC_ARGB = FOURCC('A', 'R', 'G', 'B'),
>   FOURCC_BGRA = FOURCC('B', 'G', 'R', 'A'),
>   FOURCC_ABGR = FOURCC('A', 'B', 'G', 'R'),
>+  FOURCC_AR30 = FOURCC('A', 'R', '3', '0'),  // 10 bit per channel. 2101010.
>+  FOURCC_AB30 = FOURCC('A', 'B', '3', '0'),  // ABGR version of 10 bit
>   FOURCC_24BG = FOURCC('2', '4', 'B', 'G'),
>   FOURCC_RAW = FOURCC('r', 'a', 'w', ' '),
>   FOURCC_RGBA = FOURCC('R', 'G', 'B', 'A'),
>@@ -75,16 +76,10 @@ enum FourCC {
>   FOURCC_RGBO = FOURCC('R', 'G', 'B', 'O'),  // argb1555 LE.
>   FOURCC_R444 = FOURCC('R', '4', '4', '4'),  // argb4444 LE.
> 
>-  // 4 Secondary RGB formats: 4 Bayer Patterns. deprecated.
>-  FOURCC_RGGB = FOURCC('R', 'G', 'G', 'B'),
>-  FOURCC_BGGR = FOURCC('B', 'G', 'G', 'R'),
>-  FOURCC_GRBG = FOURCC('G', 'R', 'B', 'G'),
>-  FOURCC_GBRG = FOURCC('G', 'B', 'R', 'G'),
>-
>   // 1 Primary Compressed YUV format.
>   FOURCC_MJPG = FOURCC('M', 'J', 'P', 'G'),
> 
>-  // 5 Auxiliary YUV variations: 3 with U and V planes are swapped, 1 Alias.
>+  // 7 Auxiliary YUV variations: 3 with U and V planes are swapped, 1 Alias.
>   FOURCC_YV12 = FOURCC('Y', 'V', '1', '2'),
>   FOURCC_YV16 = FOURCC('Y', 'V', '1', '6'),
>   FOURCC_YV24 = FOURCC('Y', 'V', '2', '4'),
>@@ -112,7 +107,13 @@ enum FourCC {
>   FOURCC_L565 = FOURCC('L', '5', '6', '5'),  // Alias for RGBP.
>   FOURCC_5551 = FOURCC('5', '5', '5', '1'),  // Alias for RGBO.
> 
>-  // 1 Auxiliary compressed YUV format set aside for capturer.
>+  // deprecated formats.  Not supported, but defined for backward compatibility.
>+  FOURCC_I411 = FOURCC('I', '4', '1', '1'),
>+  FOURCC_Q420 = FOURCC('Q', '4', '2', '0'),
>+  FOURCC_RGGB = FOURCC('R', 'G', 'G', 'B'),
>+  FOURCC_BGGR = FOURCC('B', 'G', 'G', 'R'),
>+  FOURCC_GRBG = FOURCC('G', 'R', 'B', 'G'),
>+  FOURCC_GBRG = FOURCC('G', 'B', 'R', 'G'),
>   FOURCC_H264 = FOURCC('H', '2', '6', '4'),
> 
>   // Match any fourcc.
>@@ -136,6 +137,8 @@ enum FourCCBpp {
>   FOURCC_BPP_BGRA = 32,
>   FOURCC_BPP_ABGR = 32,
>   FOURCC_BPP_RGBA = 32,
>+  FOURCC_BPP_AR30 = 32,
>+  FOURCC_BPP_AB30 = 32,
>   FOURCC_BPP_24BG = 24,
>   FOURCC_BPP_RAW = 24,
>   FOURCC_BPP_RGBP = 16,
>@@ -152,6 +155,7 @@ enum FourCCBpp {
>   FOURCC_BPP_J420 = 12,
>   FOURCC_BPP_J400 = 8,
>   FOURCC_BPP_H420 = 12,
>+  FOURCC_BPP_H010 = 24,
>   FOURCC_BPP_MJPG = 0,  // 0 means unknown.
>   FOURCC_BPP_H264 = 0,
>   FOURCC_BPP_IYUV = 12,
>@@ -174,7 +178,7 @@ enum FourCCBpp {
> };
> 
> // Converts fourcc aliases into canonical ones.
>-LIBYUV_API uint32 CanonicalFourCC(uint32 fourcc);
>+LIBYUV_API uint32_t CanonicalFourCC(uint32_t fourcc);
> 
> #ifdef __cplusplus
> }  // extern "C"
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/libyuv.gyp b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/libyuv.gyp
>index f73a1a4b745..e853ba3197a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/libyuv.gyp
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/libyuv.gyp
>@@ -121,7 +121,6 @@
>         # Enable the following 3 macros to turn off assembly for specified CPU.
>         # 'LIBYUV_DISABLE_X86',
>         # 'LIBYUV_DISABLE_NEON',
>-        # 'LIBYUV_DISABLE_DSPR2',
>         # Enable the following macro to build libyuv as a shared library (dll).
>         # 'LIBYUV_USING_SHARED_LIBRARY',
>         # TODO(fbarchard): Make these into gyp defines.
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/libyuv.gypi b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/libyuv.gypi
>index ec81bc9bbab..9467adfc515 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/libyuv.gypi
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/libyuv.gypi
>@@ -55,7 +55,6 @@
>       'source/rotate_argb.cc',
>       'source/rotate_common.cc',
>       'source/rotate_gcc.cc',
>-      'source/rotate_dspr2.cc',
>       'source/rotate_msa.cc',
>       'source/rotate_neon.cc',
>       'source/rotate_neon64.cc',
>@@ -63,7 +62,6 @@
>       'source/row_any.cc',
>       'source/row_common.cc',
>       'source/row_gcc.cc',
>-      'source/row_dspr2.cc',
>       'source/row_msa.cc',
>       'source/row_neon.cc',
>       'source/row_neon64.cc',
>@@ -73,7 +71,6 @@
>       'source/scale_argb.cc',
>       'source/scale_common.cc',
>       'source/scale_gcc.cc',
>-      'source/scale_dspr2.cc',
>       'source/scale_msa.cc',
>       'source/scale_neon.cc',
>       'source/scale_neon64.cc',
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/libyuv_test.gyp b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/libyuv_test.gyp
>index 4222cf26da7..5fe154c610a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/libyuv_test.gyp
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/libyuv_test.gyp
>@@ -100,7 +100,6 @@
>         # Enable the following 3 macros to turn off assembly for specified CPU.
>         # 'LIBYUV_DISABLE_X86',
>         # 'LIBYUV_DISABLE_NEON',
>-        # 'LIBYUV_DISABLE_DSPR2',
>         # Enable the following macro to build libyuv as a shared library (dll).
>         # 'LIBYUV_USING_SHARED_LIBRARY',
>       ],
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/linux.mk b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/linux.mk
>index 7e9aa5e4e8b..b84c89f912e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/linux.mk
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/linux.mk
>@@ -32,14 +32,12 @@ LOCAL_OBJ_FILES := \
> 	source/rotate.o            \
> 	source/rotate_common.o     \
> 	source/rotate_gcc.o        \
>-	source/rotate_dspr2.o      \
> 	source/rotate_neon64.o     \
> 	source/rotate_neon.o       \
> 	source/rotate_win.o        \
> 	source/row_any.o           \
> 	source/row_common.o        \
> 	source/row_gcc.o           \
>-	source/row_dspr2.o         \
> 	source/row_neon64.o        \
> 	source/row_neon.o          \
> 	source/row_win.o           \
>@@ -48,7 +46,6 @@ LOCAL_OBJ_FILES := \
> 	source/scale.o             \
> 	source/scale_common.o      \
> 	source/scale_gcc.o         \
>-	source/scale_dspr2.o       \
> 	source/scale_neon64.o      \
> 	source/scale_neon.o        \
> 	source/scale_win.o         \
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/compare.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/compare.cc
>index 8c379b59cb8..50e3abd0556 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/compare.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/compare.cc
>@@ -29,10 +29,11 @@ extern "C" {
> 
> // hash seed of 5381 recommended.
> LIBYUV_API
>-uint32 HashDjb2(const uint8* src, uint64 count, uint32 seed) {
>+uint32_t HashDjb2(const uint8_t* src, uint64_t count, uint32_t seed) {
>   const int kBlockSize = 1 << 15;  // 32768;
>   int remainder;
>-  uint32 (*HashDjb2_SSE)(const uint8* src, int count, uint32 seed) = HashDjb2_C;
>+  uint32_t (*HashDjb2_SSE)(const uint8_t* src, int count, uint32_t seed) =
>+      HashDjb2_C;
> #if defined(HAS_HASHDJB2_SSE41)
>   if (TestCpuFlag(kCpuHasSSE41)) {
>     HashDjb2_SSE = HashDjb2_SSE41;
>@@ -44,7 +45,7 @@ uint32 HashDjb2(const uint8* src, uint64 count, uint32 seed) {
>   }
> #endif
> 
>-  while (count >= (uint64)(kBlockSize)) {
>+  while (count >= (uint64_t)(kBlockSize)) {
>     seed = HashDjb2_SSE(src, kBlockSize, seed);
>     src += kBlockSize;
>     count -= kBlockSize;
>@@ -62,7 +63,7 @@ uint32 HashDjb2(const uint8* src, uint64 count, uint32 seed) {
>   return seed;
> }
> 
>-static uint32 ARGBDetectRow_C(const uint8* argb, int width) {
>+static uint32_t ARGBDetectRow_C(const uint8_t* argb, int width) {
>   int x;
>   for (x = 0; x < width - 1; x += 2) {
>     if (argb[0] != 255) {  // First byte is not Alpha of 255, so not ARGB.
>@@ -93,8 +94,11 @@ static uint32 ARGBDetectRow_C(const uint8* argb, int width) {
> // Scan an opaque argb image and return fourcc based on alpha offset.
> // Returns FOURCC_ARGB, FOURCC_BGRA, or 0 if unknown.
> LIBYUV_API
>-uint32 ARGBDetect(const uint8* argb, int stride_argb, int width, int height) {
>-  uint32 fourcc = 0;
>+uint32_t ARGBDetect(const uint8_t* argb,
>+                    int stride_argb,
>+                    int width,
>+                    int height) {
>+  uint32_t fourcc = 0;
>   int h;
> 
>   // Coalesce rows.
>@@ -114,17 +118,17 @@ uint32 ARGBDetect(const uint8* argb, int stride_argb, int width, int height) {
> // So actual maximum is 1 less loop, which is 64436 - 32 bytes.
> 
> LIBYUV_API
>-uint64 ComputeHammingDistance(const uint8* src_a,
>-                              const uint8* src_b,
>-                              int count) {
>+uint64_t ComputeHammingDistance(const uint8_t* src_a,
>+                                const uint8_t* src_b,
>+                                int count) {
>   const int kBlockSize = 1 << 15;  // 32768;
>   const int kSimdSize = 64;
>   // SIMD for multiple of 64, and C for remainder
>   int remainder = count & (kBlockSize - 1) & ~(kSimdSize - 1);
>-  uint64 diff = 0;
>+  uint64_t diff = 0;
>   int i;
>-  uint32 (*HammingDistance)(const uint8* src_a, const uint8* src_b, int count) =
>-      HammingDistance_C;
>+  uint32_t (*HammingDistance)(const uint8_t* src_a, const uint8_t* src_b,
>+                              int count) = HammingDistance_C;
> #if defined(HAS_HAMMINGDISTANCE_NEON)
>   if (TestCpuFlag(kCpuHasNEON)) {
>     HammingDistance = HammingDistance_NEON;
>@@ -172,18 +176,18 @@ uint64 ComputeHammingDistance(const uint8* src_a,
> 
> // TODO(fbarchard): Refactor into row function.
> LIBYUV_API
>-uint64 ComputeSumSquareError(const uint8* src_a,
>-                             const uint8* src_b,
>-                             int count) {
>+uint64_t ComputeSumSquareError(const uint8_t* src_a,
>+                               const uint8_t* src_b,
>+                               int count) {
>   // SumSquareError returns values 0 to 65535 for each squared difference.
>-  // Up to 65536 of those can be summed and remain within a uint32.
>-  // After each block of 65536 pixels, accumulate into a uint64.
>+  // Up to 65536 of those can be summed and remain within a uint32_t.
>+  // After each block of 65536 pixels, accumulate into a uint64_t.
>   const int kBlockSize = 65536;
>   int remainder = count & (kBlockSize - 1) & ~31;
>-  uint64 sse = 0;
>+  uint64_t sse = 0;
>   int i;
>-  uint32 (*SumSquareError)(const uint8* src_a, const uint8* src_b, int count) =
>-      SumSquareError_C;
>+  uint32_t (*SumSquareError)(const uint8_t* src_a, const uint8_t* src_b,
>+                             int count) = SumSquareError_C;
> #if defined(HAS_SUMSQUAREERROR_NEON)
>   if (TestCpuFlag(kCpuHasNEON)) {
>     SumSquareError = SumSquareError_NEON;
>@@ -227,13 +231,13 @@ uint64 ComputeSumSquareError(const uint8* src_a,
> }
> 
> LIBYUV_API
>-uint64 ComputeSumSquareErrorPlane(const uint8* src_a,
>-                                  int stride_a,
>-                                  const uint8* src_b,
>-                                  int stride_b,
>-                                  int width,
>-                                  int height) {
>-  uint64 sse = 0;
>+uint64_t ComputeSumSquareErrorPlane(const uint8_t* src_a,
>+                                    int stride_a,
>+                                    const uint8_t* src_b,
>+                                    int stride_b,
>+                                    int width,
>+                                    int height) {
>+  uint64_t sse = 0;
>   int h;
>   // Coalesce rows.
>   if (stride_a == width && stride_b == width) {
>@@ -250,7 +254,7 @@ uint64 ComputeSumSquareErrorPlane(const uint8* src_a,
> }
> 
> LIBYUV_API
>-double SumSquareErrorToPsnr(uint64 sse, uint64 count) {
>+double SumSquareErrorToPsnr(uint64_t sse, uint64_t count) {
>   double psnr;
>   if (sse > 0) {
>     double mse = (double)count / (double)sse;
>@@ -259,65 +263,67 @@ double SumSquareErrorToPsnr(uint64 sse, uint64 count) {
>     psnr = kMaxPsnr;  // Limit to prevent divide by 0
>   }
> 
>-  if (psnr > kMaxPsnr)
>+  if (psnr > kMaxPsnr) {
>     psnr = kMaxPsnr;
>+  }
> 
>   return psnr;
> }
> 
> LIBYUV_API
>-double CalcFramePsnr(const uint8* src_a,
>+double CalcFramePsnr(const uint8_t* src_a,
>                      int stride_a,
>-                     const uint8* src_b,
>+                     const uint8_t* src_b,
>                      int stride_b,
>                      int width,
>                      int height) {
>-  const uint64 samples = width * height;
>-  const uint64 sse = ComputeSumSquareErrorPlane(src_a, stride_a, src_b,
>-                                                stride_b, width, height);
>+  const uint64_t samples = (uint64_t)width * (uint64_t)height;
>+  const uint64_t sse = ComputeSumSquareErrorPlane(src_a, stride_a, src_b,
>+                                                  stride_b, width, height);
>   return SumSquareErrorToPsnr(sse, samples);
> }
> 
> LIBYUV_API
>-double I420Psnr(const uint8* src_y_a,
>+double I420Psnr(const uint8_t* src_y_a,
>                 int stride_y_a,
>-                const uint8* src_u_a,
>+                const uint8_t* src_u_a,
>                 int stride_u_a,
>-                const uint8* src_v_a,
>+                const uint8_t* src_v_a,
>                 int stride_v_a,
>-                const uint8* src_y_b,
>+                const uint8_t* src_y_b,
>                 int stride_y_b,
>-                const uint8* src_u_b,
>+                const uint8_t* src_u_b,
>                 int stride_u_b,
>-                const uint8* src_v_b,
>+                const uint8_t* src_v_b,
>                 int stride_v_b,
>                 int width,
>                 int height) {
>-  const uint64 sse_y = ComputeSumSquareErrorPlane(src_y_a, stride_y_a, src_y_b,
>-                                                  stride_y_b, width, height);
>+  const uint64_t sse_y = ComputeSumSquareErrorPlane(
>+      src_y_a, stride_y_a, src_y_b, stride_y_b, width, height);
>   const int width_uv = (width + 1) >> 1;
>   const int height_uv = (height + 1) >> 1;
>-  const uint64 sse_u = ComputeSumSquareErrorPlane(
>+  const uint64_t sse_u = ComputeSumSquareErrorPlane(
>       src_u_a, stride_u_a, src_u_b, stride_u_b, width_uv, height_uv);
>-  const uint64 sse_v = ComputeSumSquareErrorPlane(
>+  const uint64_t sse_v = ComputeSumSquareErrorPlane(
>       src_v_a, stride_v_a, src_v_b, stride_v_b, width_uv, height_uv);
>-  const uint64 samples = width * height + 2 * (width_uv * height_uv);
>-  const uint64 sse = sse_y + sse_u + sse_v;
>+  const uint64_t samples = (uint64_t)width * (uint64_t)height +
>+                           2 * ((uint64_t)width_uv * (uint64_t)height_uv);
>+  const uint64_t sse = sse_y + sse_u + sse_v;
>   return SumSquareErrorToPsnr(sse, samples);
> }
> 
>-static const int64 cc1 = 26634;   // (64^2*(.01*255)^2
>-static const int64 cc2 = 239708;  // (64^2*(.03*255)^2
>+static const int64_t cc1 = 26634;   // (64^2*(.01*255)^2
>+static const int64_t cc2 = 239708;  // (64^2*(.03*255)^2
> 
>-static double Ssim8x8_C(const uint8* src_a,
>+static double Ssim8x8_C(const uint8_t* src_a,
>                         int stride_a,
>-                        const uint8* src_b,
>+                        const uint8_t* src_b,
>                         int stride_b) {
>-  int64 sum_a = 0;
>-  int64 sum_b = 0;
>-  int64 sum_sq_a = 0;
>-  int64 sum_sq_b = 0;
>-  int64 sum_axb = 0;
>+  int64_t sum_a = 0;
>+  int64_t sum_b = 0;
>+  int64_t sum_sq_a = 0;
>+  int64_t sum_sq_b = 0;
>+  int64_t sum_axb = 0;
> 
>   int i;
>   for (i = 0; i < 8; ++i) {
>@@ -335,20 +341,20 @@ static double Ssim8x8_C(const uint8* src_a,
>   }
> 
>   {
>-    const int64 count = 64;
>+    const int64_t count = 64;
>     // scale the constants by number of pixels
>-    const int64 c1 = (cc1 * count * count) >> 12;
>-    const int64 c2 = (cc2 * count * count) >> 12;
>+    const int64_t c1 = (cc1 * count * count) >> 12;
>+    const int64_t c2 = (cc2 * count * count) >> 12;
> 
>-    const int64 sum_a_x_sum_b = sum_a * sum_b;
>+    const int64_t sum_a_x_sum_b = sum_a * sum_b;
> 
>-    const int64 ssim_n = (2 * sum_a_x_sum_b + c1) *
>-                         (2 * count * sum_axb - 2 * sum_a_x_sum_b + c2);
>+    const int64_t ssim_n = (2 * sum_a_x_sum_b + c1) *
>+                           (2 * count * sum_axb - 2 * sum_a_x_sum_b + c2);
> 
>-    const int64 sum_a_sq = sum_a * sum_a;
>-    const int64 sum_b_sq = sum_b * sum_b;
>+    const int64_t sum_a_sq = sum_a * sum_a;
>+    const int64_t sum_b_sq = sum_b * sum_b;
> 
>-    const int64 ssim_d =
>+    const int64_t ssim_d =
>         (sum_a_sq + sum_b_sq + c1) *
>         (count * sum_sq_a - sum_a_sq + count * sum_sq_b - sum_b_sq + c2);
> 
>@@ -363,15 +369,15 @@ static double Ssim8x8_C(const uint8* src_a,
> // on the 4x4 pixel grid. Such arrangement allows the windows to overlap
> // block boundaries to penalize blocking artifacts.
> LIBYUV_API
>-double CalcFrameSsim(const uint8* src_a,
>+double CalcFrameSsim(const uint8_t* src_a,
>                      int stride_a,
>-                     const uint8* src_b,
>+                     const uint8_t* src_b,
>                      int stride_b,
>                      int width,
>                      int height) {
>   int samples = 0;
>   double ssim_total = 0;
>-  double (*Ssim8x8)(const uint8* src_a, int stride_a, const uint8* src_b,
>+  double (*Ssim8x8)(const uint8_t* src_a, int stride_a, const uint8_t* src_b,
>                     int stride_b) = Ssim8x8_C;
> 
>   // sample point start with each 4x4 location
>@@ -392,17 +398,17 @@ double CalcFrameSsim(const uint8* src_a,
> }
> 
> LIBYUV_API
>-double I420Ssim(const uint8* src_y_a,
>+double I420Ssim(const uint8_t* src_y_a,
>                 int stride_y_a,
>-                const uint8* src_u_a,
>+                const uint8_t* src_u_a,
>                 int stride_u_a,
>-                const uint8* src_v_a,
>+                const uint8_t* src_v_a,
>                 int stride_v_a,
>-                const uint8* src_y_b,
>+                const uint8_t* src_y_b,
>                 int stride_y_b,
>-                const uint8* src_u_b,
>+                const uint8_t* src_u_b,
>                 int stride_u_b,
>-                const uint8* src_v_b,
>+                const uint8_t* src_v_b,
>                 int stride_v_b,
>                 int width,
>                 int height) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/compare_common.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/compare_common.cc
>index 83564a1bcb5..633466addb5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/compare_common.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/compare_common.cc
>@@ -18,8 +18,10 @@ extern "C" {
> #endif
> 
> #if ORIGINAL_OPT
>-uint32 HammingDistance_C1(const uint8* src_a, const uint8* src_b, int count) {
>-  uint32 diff = 0u;
>+uint32_t HammingDistance_C1(const uint8_t* src_a,
>+                            const uint8_t* src_b,
>+                            int count) {
>+  uint32_t diff = 0u;
> 
>   int i;
>   for (i = 0; i < count; ++i) {
>@@ -46,13 +48,15 @@ uint32 HammingDistance_C1(const uint8* src_a, const uint8* src_b, int count) {
> #endif
> 
> // Hakmem method for hamming distance.
>-uint32 HammingDistance_C(const uint8* src_a, const uint8* src_b, int count) {
>-  uint32 diff = 0u;
>+uint32_t HammingDistance_C(const uint8_t* src_a,
>+                           const uint8_t* src_b,
>+                           int count) {
>+  uint32_t diff = 0u;
> 
>   int i;
>   for (i = 0; i < count - 3; i += 4) {
>-    uint32 x = *((uint32*)src_a) ^ *((uint32*)src_b);
>-    uint32 u = x - ((x >> 1) & 0x55555555);
>+    uint32_t x = *((uint32_t*)src_a) ^ *((uint32_t*)src_b);  // NOLINT
>+    uint32_t u = x - ((x >> 1) & 0x55555555);
>     u = ((u >> 2) & 0x33333333) + (u & 0x33333333);
>     diff += ((((u + (u >> 4)) & 0x0f0f0f0f) * 0x01010101) >> 24);
>     src_a += 4;
>@@ -60,8 +64,8 @@ uint32 HammingDistance_C(const uint8* src_a, const uint8* src_b, int count) {
>   }
> 
>   for (; i < count; ++i) {
>-    uint32 x = *src_a ^ *src_b;
>-    uint32 u = x - ((x >> 1) & 0x55);
>+    uint32_t x = *src_a ^ *src_b;
>+    uint32_t u = x - ((x >> 1) & 0x55);
>     u = ((u >> 2) & 0x33) + (u & 0x33);
>     diff += (u + (u >> 4)) & 0x0f;
>     src_a += 1;
>@@ -71,20 +75,22 @@ uint32 HammingDistance_C(const uint8* src_a, const uint8* src_b, int count) {
>   return diff;
> }
> 
>-uint32 SumSquareError_C(const uint8* src_a, const uint8* src_b, int count) {
>-  uint32 sse = 0u;
>+uint32_t SumSquareError_C(const uint8_t* src_a,
>+                          const uint8_t* src_b,
>+                          int count) {
>+  uint32_t sse = 0u;
>   int i;
>   for (i = 0; i < count; ++i) {
>     int diff = src_a[i] - src_b[i];
>-    sse += (uint32)(diff * diff);
>+    sse += (uint32_t)(diff * diff);
>   }
>   return sse;
> }
> 
> // hash seed of 5381 recommended.
> // Internal C version of HashDjb2 with int sized count for efficiency.
>-uint32 HashDjb2_C(const uint8* src, int count, uint32 seed) {
>-  uint32 hash = seed;
>+uint32_t HashDjb2_C(const uint8_t* src, int count, uint32_t seed) {
>+  uint32_t hash = seed;
>   int i;
>   for (i = 0; i < count; ++i) {
>     hash += (hash << 5) + src[i];
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/compare_gcc.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/compare_gcc.cc
>index 595c8ec4ae2..676527c1b1b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/compare_gcc.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/compare_gcc.cc
>@@ -23,10 +23,10 @@ extern "C" {
>     (defined(__x86_64__) || (defined(__i386__) && !defined(_MSC_VER)))
> 
> #if defined(__x86_64__)
>-uint32 HammingDistance_SSE42(const uint8* src_a,
>-                             const uint8* src_b,
>-                             int count) {
>-  uint64 diff = 0u;
>+uint32_t HammingDistance_SSE42(const uint8_t* src_a,
>+                               const uint8_t* src_b,
>+                               int count) {
>+  uint64_t diff = 0u;
> 
>   asm volatile(
>       "xor        %3,%3                          \n"
>@@ -68,13 +68,13 @@ uint32 HammingDistance_SSE42(const uint8* src_a,
>       :
>       : "memory", "cc", "rcx", "rdx", "rsi", "rdi", "r8", "r9", "r10");
> 
>-  return static_cast<uint32>(diff);
>+  return static_cast<uint32_t>(diff);
> }
> #else
>-uint32 HammingDistance_SSE42(const uint8* src_a,
>-                             const uint8* src_b,
>-                             int count) {
>-  uint32 diff = 0u;
>+uint32_t HammingDistance_SSE42(const uint8_t* src_a,
>+                               const uint8_t* src_b,
>+                               int count) {
>+  uint32_t diff = 0u;
> 
>   asm volatile(
>       // Process 16 bytes per loop.
>@@ -111,14 +111,14 @@ uint32 HammingDistance_SSE42(const uint8* src_a,
> }
> #endif
> 
>-static vec8 kNibbleMask = {15, 15, 15, 15, 15, 15, 15, 15,
>-                           15, 15, 15, 15, 15, 15, 15, 15};
>-static vec8 kBitCount = {0, 1, 1, 2, 1, 2, 2, 3, 1, 2, 2, 3, 2, 3, 3, 4};
>+static const vec8 kNibbleMask = {15, 15, 15, 15, 15, 15, 15, 15,
>+                                 15, 15, 15, 15, 15, 15, 15, 15};
>+static const vec8 kBitCount = {0, 1, 1, 2, 1, 2, 2, 3, 1, 2, 2, 3, 2, 3, 3, 4};
> 
>-uint32 HammingDistance_SSSE3(const uint8* src_a,
>-                             const uint8* src_b,
>-                             int count) {
>-  uint32 diff = 0u;
>+uint32_t HammingDistance_SSSE3(const uint8_t* src_a,
>+                               const uint8_t* src_b,
>+                               int count) {
>+  uint32_t diff = 0u;
> 
>   asm volatile(
>       "movdqa     %4,%%xmm2                      \n"
>@@ -174,8 +174,10 @@ uint32 HammingDistance_SSSE3(const uint8* src_a,
> }
> 
> #ifdef HAS_HAMMINGDISTANCE_AVX2
>-uint32 HammingDistance_AVX2(const uint8* src_a, const uint8* src_b, int count) {
>-  uint32 diff = 0u;
>+uint32_t HammingDistance_AVX2(const uint8_t* src_a,
>+                              const uint8_t* src_b,
>+                              int count) {
>+  uint32_t diff = 0u;
> 
>   asm volatile(
>       "vbroadcastf128 %4,%%ymm2                  \n"
>@@ -227,124 +229,127 @@ uint32 HammingDistance_AVX2(const uint8* src_a, const uint8* src_b, int count) {
> }
> #endif  // HAS_HAMMINGDISTANCE_AVX2
> 
>-uint32 SumSquareError_SSE2(const uint8* src_a, const uint8* src_b, int count) {
>-  uint32 sse;
>-  asm volatile (
>-    "pxor      %%xmm0,%%xmm0                   \n"
>-    "pxor      %%xmm5,%%xmm5                   \n"
>-    LABELALIGN
>-  "1:                                          \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm1         \n"
>-    "lea       " MEMLEA(0x10, 0) ",%0          \n"
>-    "movdqu    " MEMACCESS(1) ",%%xmm2         \n"
>-    "lea       " MEMLEA(0x10, 1) ",%1          \n"
>-    "movdqa    %%xmm1,%%xmm3                   \n"
>-    "psubusb   %%xmm2,%%xmm1                   \n"
>-    "psubusb   %%xmm3,%%xmm2                   \n"
>-    "por       %%xmm2,%%xmm1                   \n"
>-    "movdqa    %%xmm1,%%xmm2                   \n"
>-    "punpcklbw %%xmm5,%%xmm1                   \n"
>-    "punpckhbw %%xmm5,%%xmm2                   \n"
>-    "pmaddwd   %%xmm1,%%xmm1                   \n"
>-    "pmaddwd   %%xmm2,%%xmm2                   \n"
>-    "paddd     %%xmm1,%%xmm0                   \n"
>-    "paddd     %%xmm2,%%xmm0                   \n"
>-    "sub       $0x10,%2                        \n"
>-    "jg        1b                              \n"
>+uint32_t SumSquareError_SSE2(const uint8_t* src_a,
>+                             const uint8_t* src_b,
>+                             int count) {
>+  uint32_t sse;
>+  asm volatile(
>+      "pxor      %%xmm0,%%xmm0                   \n"
>+      "pxor      %%xmm5,%%xmm5                   \n"
> 
>-    "pshufd    $0xee,%%xmm0,%%xmm1             \n"
>-    "paddd     %%xmm1,%%xmm0                   \n"
>-    "pshufd    $0x1,%%xmm0,%%xmm1              \n"
>-    "paddd     %%xmm1,%%xmm0                   \n"
>-    "movd      %%xmm0,%3                       \n"
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm1                     \n"
>+      "lea       0x10(%0),%0                     \n"
>+      "movdqu    (%1),%%xmm2                     \n"
>+      "lea       0x10(%1),%1                     \n"
>+      "movdqa    %%xmm1,%%xmm3                   \n"
>+      "psubusb   %%xmm2,%%xmm1                   \n"
>+      "psubusb   %%xmm3,%%xmm2                   \n"
>+      "por       %%xmm2,%%xmm1                   \n"
>+      "movdqa    %%xmm1,%%xmm2                   \n"
>+      "punpcklbw %%xmm5,%%xmm1                   \n"
>+      "punpckhbw %%xmm5,%%xmm2                   \n"
>+      "pmaddwd   %%xmm1,%%xmm1                   \n"
>+      "pmaddwd   %%xmm2,%%xmm2                   \n"
>+      "paddd     %%xmm1,%%xmm0                   \n"
>+      "paddd     %%xmm2,%%xmm0                   \n"
>+      "sub       $0x10,%2                        \n"
>+      "jg        1b                              \n"
> 
>-  : "+r"(src_a),      // %0
>-    "+r"(src_b),      // %1
>-    "+r"(count),      // %2
>-    "=g"(sse)         // %3
>-  :: "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm5"
>-  );
>+      "pshufd    $0xee,%%xmm0,%%xmm1             \n"
>+      "paddd     %%xmm1,%%xmm0                   \n"
>+      "pshufd    $0x1,%%xmm0,%%xmm1              \n"
>+      "paddd     %%xmm1,%%xmm0                   \n"
>+      "movd      %%xmm0,%3                       \n"
>+
>+      : "+r"(src_a),  // %0
>+        "+r"(src_b),  // %1
>+        "+r"(count),  // %2
>+        "=g"(sse)     // %3
>+        ::"memory",
>+        "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm5");
>   return sse;
> }
> 
>-static uvec32 kHash16x33 = {0x92d9e201, 0, 0, 0};  // 33 ^ 16
>-static uvec32 kHashMul0 = {
>+static const uvec32 kHash16x33 = {0x92d9e201, 0, 0, 0};  // 33 ^ 16
>+static const uvec32 kHashMul0 = {
>     0x0c3525e1,  // 33 ^ 15
>     0xa3476dc1,  // 33 ^ 14
>     0x3b4039a1,  // 33 ^ 13
>     0x4f5f0981,  // 33 ^ 12
> };
>-static uvec32 kHashMul1 = {
>+static const uvec32 kHashMul1 = {
>     0x30f35d61,  // 33 ^ 11
>     0x855cb541,  // 33 ^ 10
>     0x040a9121,  // 33 ^ 9
>     0x747c7101,  // 33 ^ 8
> };
>-static uvec32 kHashMul2 = {
>+static const uvec32 kHashMul2 = {
>     0xec41d4e1,  // 33 ^ 7
>     0x4cfa3cc1,  // 33 ^ 6
>     0x025528a1,  // 33 ^ 5
>     0x00121881,  // 33 ^ 4
> };
>-static uvec32 kHashMul3 = {
>+static const uvec32 kHashMul3 = {
>     0x00008c61,  // 33 ^ 3
>     0x00000441,  // 33 ^ 2
>     0x00000021,  // 33 ^ 1
>     0x00000001,  // 33 ^ 0
> };
> 
>-uint32 HashDjb2_SSE41(const uint8* src, int count, uint32 seed) {
>-  uint32 hash;
>-  asm volatile (
>-    "movd      %2,%%xmm0                       \n"
>-    "pxor      %%xmm7,%%xmm7                   \n"
>-    "movdqa    %4,%%xmm6                       \n"
>-    LABELALIGN
>-  "1:                                          \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm1         \n"
>-    "lea       " MEMLEA(0x10, 0) ",%0          \n"
>-    "pmulld    %%xmm6,%%xmm0                   \n"
>-    "movdqa    %5,%%xmm5                       \n"
>-    "movdqa    %%xmm1,%%xmm2                   \n"
>-    "punpcklbw %%xmm7,%%xmm2                   \n"
>-    "movdqa    %%xmm2,%%xmm3                   \n"
>-    "punpcklwd %%xmm7,%%xmm3                   \n"
>-    "pmulld    %%xmm5,%%xmm3                   \n"
>-    "movdqa    %6,%%xmm5                       \n"
>-    "movdqa    %%xmm2,%%xmm4                   \n"
>-    "punpckhwd %%xmm7,%%xmm4                   \n"
>-    "pmulld    %%xmm5,%%xmm4                   \n"
>-    "movdqa    %7,%%xmm5                       \n"
>-    "punpckhbw %%xmm7,%%xmm1                   \n"
>-    "movdqa    %%xmm1,%%xmm2                   \n"
>-    "punpcklwd %%xmm7,%%xmm2                   \n"
>-    "pmulld    %%xmm5,%%xmm2                   \n"
>-    "movdqa    %8,%%xmm5                       \n"
>-    "punpckhwd %%xmm7,%%xmm1                   \n"
>-    "pmulld    %%xmm5,%%xmm1                   \n"
>-    "paddd     %%xmm4,%%xmm3                   \n"
>-    "paddd     %%xmm2,%%xmm1                   \n"
>-    "paddd     %%xmm3,%%xmm1                   \n"
>-    "pshufd    $0xe,%%xmm1,%%xmm2              \n"
>-    "paddd     %%xmm2,%%xmm1                   \n"
>-    "pshufd    $0x1,%%xmm1,%%xmm2              \n"
>-    "paddd     %%xmm2,%%xmm1                   \n"
>-    "paddd     %%xmm1,%%xmm0                   \n"
>-    "sub       $0x10,%1                        \n"
>-    "jg        1b                              \n"
>-    "movd      %%xmm0,%3                       \n"
>-  : "+r"(src),        // %0
>-    "+r"(count),      // %1
>-    "+rm"(seed),      // %2
>-    "=g"(hash)        // %3
>-  : "m"(kHash16x33),  // %4
>-    "m"(kHashMul0),   // %5
>-    "m"(kHashMul1),   // %6
>-    "m"(kHashMul2),   // %7
>-    "m"(kHashMul3)    // %8
>-  : "memory", "cc"
>-    , "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6", "xmm7"
>-  );
>+uint32_t HashDjb2_SSE41(const uint8_t* src, int count, uint32_t seed) {
>+  uint32_t hash;
>+  asm volatile(
>+      "movd      %2,%%xmm0                       \n"
>+      "pxor      %%xmm7,%%xmm7                   \n"
>+      "movdqa    %4,%%xmm6                       \n"
>+
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm1                     \n"
>+      "lea       0x10(%0),%0                     \n"
>+      "pmulld    %%xmm6,%%xmm0                   \n"
>+      "movdqa    %5,%%xmm5                       \n"
>+      "movdqa    %%xmm1,%%xmm2                   \n"
>+      "punpcklbw %%xmm7,%%xmm2                   \n"
>+      "movdqa    %%xmm2,%%xmm3                   \n"
>+      "punpcklwd %%xmm7,%%xmm3                   \n"
>+      "pmulld    %%xmm5,%%xmm3                   \n"
>+      "movdqa    %6,%%xmm5                       \n"
>+      "movdqa    %%xmm2,%%xmm4                   \n"
>+      "punpckhwd %%xmm7,%%xmm4                   \n"
>+      "pmulld    %%xmm5,%%xmm4                   \n"
>+      "movdqa    %7,%%xmm5                       \n"
>+      "punpckhbw %%xmm7,%%xmm1                   \n"
>+      "movdqa    %%xmm1,%%xmm2                   \n"
>+      "punpcklwd %%xmm7,%%xmm2                   \n"
>+      "pmulld    %%xmm5,%%xmm2                   \n"
>+      "movdqa    %8,%%xmm5                       \n"
>+      "punpckhwd %%xmm7,%%xmm1                   \n"
>+      "pmulld    %%xmm5,%%xmm1                   \n"
>+      "paddd     %%xmm4,%%xmm3                   \n"
>+      "paddd     %%xmm2,%%xmm1                   \n"
>+      "paddd     %%xmm3,%%xmm1                   \n"
>+      "pshufd    $0xe,%%xmm1,%%xmm2              \n"
>+      "paddd     %%xmm2,%%xmm1                   \n"
>+      "pshufd    $0x1,%%xmm1,%%xmm2              \n"
>+      "paddd     %%xmm2,%%xmm1                   \n"
>+      "paddd     %%xmm1,%%xmm0                   \n"
>+      "sub       $0x10,%1                        \n"
>+      "jg        1b                              \n"
>+      "movd      %%xmm0,%3                       \n"
>+      : "+r"(src),        // %0
>+        "+r"(count),      // %1
>+        "+rm"(seed),      // %2
>+        "=g"(hash)        // %3
>+      : "m"(kHash16x33),  // %4
>+        "m"(kHashMul0),   // %5
>+        "m"(kHashMul1),   // %6
>+        "m"(kHashMul2),   // %7
>+        "m"(kHashMul3)    // %8
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6",
>+        "xmm7");
>   return hash;
> }
> #endif  // defined(__x86_64__) || (defined(__i386__) && !defined(__pic__)))
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/compare_msa.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/compare_msa.cc
>index efb5185aac7..e944235f020 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/compare_msa.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/compare_msa.cc
>@@ -22,8 +22,10 @@ namespace libyuv {
> extern "C" {
> #endif
> 
>-uint32 HammingDistance_MSA(const uint8* src_a, const uint8* src_b, int count) {
>-  uint32 diff = 0u;
>+uint32_t HammingDistance_MSA(const uint8_t* src_a,
>+                             const uint8_t* src_b,
>+                             int count) {
>+  uint32_t diff = 0u;
>   int i;
>   v16u8 src0, src1, src2, src3;
>   v2i64 vec0 = {0}, vec1 = {0};
>@@ -42,13 +44,15 @@ uint32 HammingDistance_MSA(const uint8* src_a, const uint8* src_b, int count) {
>   }
> 
>   vec0 += vec1;
>-  diff = (uint32)__msa_copy_u_w((v4i32)vec0, 0);
>-  diff += (uint32)__msa_copy_u_w((v4i32)vec0, 2);
>+  diff = (uint32_t)__msa_copy_u_w((v4i32)vec0, 0);
>+  diff += (uint32_t)__msa_copy_u_w((v4i32)vec0, 2);
>   return diff;
> }
> 
>-uint32 SumSquareError_MSA(const uint8* src_a, const uint8* src_b, int count) {
>-  uint32 sse = 0u;
>+uint32_t SumSquareError_MSA(const uint8_t* src_a,
>+                            const uint8_t* src_b,
>+                            int count) {
>+  uint32_t sse = 0u;
>   int i;
>   v16u8 src0, src1, src2, src3;
>   v8i16 vec0, vec1, vec2, vec3;
>@@ -80,8 +84,8 @@ uint32 SumSquareError_MSA(const uint8* src_a, const uint8* src_b, int count) {
>   reg2 += reg3;
>   reg0 += reg2;
>   tmp0 = __msa_hadd_s_d(reg0, reg0);
>-  sse = (uint32)__msa_copy_u_w((v4i32)tmp0, 0);
>-  sse += (uint32)__msa_copy_u_w((v4i32)tmp0, 2);
>+  sse = (uint32_t)__msa_copy_u_w((v4i32)tmp0, 0);
>+  sse += (uint32_t)__msa_copy_u_w((v4i32)tmp0, 2);
>   return sse;
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/compare_neon.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/compare_neon.cc
>index 5dfa71edcbf..2a2181e0cb3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/compare_neon.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/compare_neon.cc
>@@ -23,8 +23,10 @@ extern "C" {
> 
> // 256 bits at a time
> // uses short accumulator which restricts count to 131 KB
>-uint32 HammingDistance_NEON(const uint8* src_a, const uint8* src_b, int count) {
>-  uint32 diff;
>+uint32_t HammingDistance_NEON(const uint8_t* src_a,
>+                              const uint8_t* src_b,
>+                              int count) {
>+  uint32_t diff;
> 
>   asm volatile(
>       "vmov.u16   q4, #0                         \n"  // accumulator
>@@ -52,8 +54,10 @@ uint32 HammingDistance_NEON(const uint8* src_a, const uint8* src_b, int count) {
>   return diff;
> }
> 
>-uint32 SumSquareError_NEON(const uint8* src_a, const uint8* src_b, int count) {
>-  uint32 sse;
>+uint32_t SumSquareError_NEON(const uint8_t* src_a,
>+                             const uint8_t* src_b,
>+                             int count) {
>+  uint32_t sse;
>   asm volatile(
>       "vmov.u8    q8, #0                         \n"
>       "vmov.u8    q10, #0                        \n"
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/compare_neon64.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/compare_neon64.cc
>index ddf98fa68b2..6e8f672ab73 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/compare_neon64.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/compare_neon64.cc
>@@ -22,8 +22,10 @@ extern "C" {
> 
> // 256 bits at a time
> // uses short accumulator which restricts count to 131 KB
>-uint32 HammingDistance_NEON(const uint8* src_a, const uint8* src_b, int count) {
>-  uint32 diff;
>+uint32_t HammingDistance_NEON(const uint8_t* src_a,
>+                              const uint8_t* src_b,
>+                              int count) {
>+  uint32_t diff;
>   asm volatile(
>       "movi       v4.8h, #0                      \n"
> 
>@@ -47,8 +49,10 @@ uint32 HammingDistance_NEON(const uint8* src_a, const uint8* src_b, int count) {
>   return diff;
> }
> 
>-uint32 SumSquareError_NEON(const uint8* src_a, const uint8* src_b, int count) {
>-  uint32 sse;
>+uint32_t SumSquareError_NEON(const uint8_t* src_a,
>+                             const uint8_t* src_b,
>+                             int count) {
>+  uint32_t sse;
>   asm volatile(
>       "eor        v16.16b, v16.16b, v16.16b      \n"
>       "eor        v18.16b, v18.16b, v18.16b      \n"
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/compare_win.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/compare_win.cc
>index bcd6a88ebbb..d57d3d9d1c8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/compare_win.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/compare_win.cc
>@@ -25,14 +25,14 @@ extern "C" {
> // This module is for 32 bit Visual C x86 and clangcl
> #if !defined(LIBYUV_DISABLE_X86) && defined(_M_IX86) && defined(_MSC_VER)
> 
>-uint32 HammingDistance_SSE42(const uint8* src_a,
>-                             const uint8* src_b,
>-                             int count) {
>-  uint32 diff = 0u;
>+uint32_t HammingDistance_SSE42(const uint8_t* src_a,
>+                               const uint8_t* src_b,
>+                               int count) {
>+  uint32_t diff = 0u;
> 
>   int i;
>   for (i = 0; i < count - 3; i += 4) {
>-    uint32 x = *((uint32*)src_a) ^ *((uint32*)src_b);
>+    uint32_t x = *((uint32_t*)src_a) ^ *((uint32_t*)src_b);  // NOLINT
>     src_a += 4;
>     src_b += 4;
>     diff += __popcnt(x);
>@@ -40,8 +40,8 @@ uint32 HammingDistance_SSE42(const uint8* src_a,
>   return diff;
> }
> 
>-__declspec(naked) uint32
>-    SumSquareError_SSE2(const uint8* src_a, const uint8* src_b, int count) {
>+__declspec(naked) uint32_t
>+    SumSquareError_SSE2(const uint8_t* src_a, const uint8_t* src_b, int count) {
>   __asm {
>     mov        eax, [esp + 4]  // src_a
>     mov        edx, [esp + 8]  // src_b
>@@ -81,8 +81,8 @@ __declspec(naked) uint32
> #if _MSC_VER >= 1700
> // C4752: found Intel(R) Advanced Vector Extensions; consider using /arch:AVX.
> #pragma warning(disable : 4752)
>-__declspec(naked) uint32
>-    SumSquareError_AVX2(const uint8* src_a, const uint8* src_b, int count) {
>+__declspec(naked) uint32_t
>+    SumSquareError_AVX2(const uint8_t* src_a, const uint8_t* src_b, int count) {
>   __asm {
>     mov        eax, [esp + 4]  // src_a
>     mov        edx, [esp + 8]  // src_b
>@@ -146,8 +146,8 @@ uvec32 kHashMul3 = {
>     0x00000001,  // 33 ^ 0
> };
> 
>-__declspec(naked) uint32
>-    HashDjb2_SSE41(const uint8* src, int count, uint32 seed) {
>+__declspec(naked) uint32_t
>+    HashDjb2_SSE41(const uint8_t* src, int count, uint32_t seed) {
>   __asm {
>     mov        eax, [esp + 4]  // src
>     mov        ecx, [esp + 8]  // count
>@@ -197,8 +197,8 @@ __declspec(naked) uint32
> 
> // Visual C 2012 required for AVX2.
> #if _MSC_VER >= 1700
>-__declspec(naked) uint32
>-    HashDjb2_AVX2(const uint8* src, int count, uint32 seed) {
>+__declspec(naked) uint32_t
>+    HashDjb2_AVX2(const uint8_t* src, int count, uint32_t seed) {
>   __asm {
>     mov        eax, [esp + 4]  // src
>     mov        ecx, [esp + 8]  // count
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/convert.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/convert.cc
>index dfa83a5a6d8..375cc732c1d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/convert.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/convert.cc
>@@ -28,17 +28,17 @@ static __inline int Abs(int v) {
> }
> 
> // Any I4xx To I420 format with mirroring.
>-static int I4xxToI420(const uint8* src_y,
>+static int I4xxToI420(const uint8_t* src_y,
>                       int src_stride_y,
>-                      const uint8* src_u,
>+                      const uint8_t* src_u,
>                       int src_stride_u,
>-                      const uint8* src_v,
>+                      const uint8_t* src_v,
>                       int src_stride_v,
>-                      uint8* dst_y,
>+                      uint8_t* dst_y,
>                       int dst_stride_y,
>-                      uint8* dst_u,
>+                      uint8_t* dst_u,
>                       int dst_stride_u,
>-                      uint8* dst_v,
>+                      uint8_t* dst_v,
>                       int dst_stride_v,
>                       int src_y_width,
>                       int src_y_height,
>@@ -62,21 +62,21 @@ static int I4xxToI420(const uint8* src_y,
>   return 0;
> }
> 
>-// Copy I420 with optional flipping
>+// Copy I420 with optional flipping.
> // TODO(fbarchard): Use Scale plane which supports mirroring, but ensure
> // is does row coalescing.
> LIBYUV_API
>-int I420Copy(const uint8* src_y,
>+int I420Copy(const uint8_t* src_y,
>              int src_stride_y,
>-             const uint8* src_u,
>+             const uint8_t* src_u,
>              int src_stride_u,
>-             const uint8* src_v,
>+             const uint8_t* src_v,
>              int src_stride_v,
>-             uint8* dst_y,
>+             uint8_t* dst_y,
>              int dst_stride_y,
>-             uint8* dst_u,
>+             uint8_t* dst_u,
>              int dst_stride_u,
>-             uint8* dst_v,
>+             uint8_t* dst_v,
>              int dst_stride_v,
>              int width,
>              int height) {
>@@ -106,20 +106,106 @@ int I420Copy(const uint8* src_y,
>   return 0;
> }
> 
>+// Copy I010 with optional flipping.
>+LIBYUV_API
>+int I010Copy(const uint16_t* src_y,
>+             int src_stride_y,
>+             const uint16_t* src_u,
>+             int src_stride_u,
>+             const uint16_t* src_v,
>+             int src_stride_v,
>+             uint16_t* dst_y,
>+             int dst_stride_y,
>+             uint16_t* dst_u,
>+             int dst_stride_u,
>+             uint16_t* dst_v,
>+             int dst_stride_v,
>+             int width,
>+             int height) {
>+  int halfwidth = (width + 1) >> 1;
>+  int halfheight = (height + 1) >> 1;
>+  if (!src_u || !src_v || !dst_u || !dst_v || width <= 0 || height == 0) {
>+    return -1;
>+  }
>+  // Negative height means invert the image.
>+  if (height < 0) {
>+    height = -height;
>+    halfheight = (height + 1) >> 1;
>+    src_y = src_y + (height - 1) * src_stride_y;
>+    src_u = src_u + (halfheight - 1) * src_stride_u;
>+    src_v = src_v + (halfheight - 1) * src_stride_v;
>+    src_stride_y = -src_stride_y;
>+    src_stride_u = -src_stride_u;
>+    src_stride_v = -src_stride_v;
>+  }
>+
>+  if (dst_y) {
>+    CopyPlane_16(src_y, src_stride_y, dst_y, dst_stride_y, width, height);
>+  }
>+  // Copy UV planes.
>+  CopyPlane_16(src_u, src_stride_u, dst_u, dst_stride_u, halfwidth, halfheight);
>+  CopyPlane_16(src_v, src_stride_v, dst_v, dst_stride_v, halfwidth, halfheight);
>+  return 0;
>+}
>+
>+// Convert 10 bit YUV to 8 bit.
>+LIBYUV_API
>+int I010ToI420(const uint16_t* src_y,
>+               int src_stride_y,
>+               const uint16_t* src_u,
>+               int src_stride_u,
>+               const uint16_t* src_v,
>+               int src_stride_v,
>+               uint8_t* dst_y,
>+               int dst_stride_y,
>+               uint8_t* dst_u,
>+               int dst_stride_u,
>+               uint8_t* dst_v,
>+               int dst_stride_v,
>+               int width,
>+               int height) {
>+  int halfwidth = (width + 1) >> 1;
>+  int halfheight = (height + 1) >> 1;
>+  if (!src_u || !src_v || !dst_u || !dst_v || width <= 0 || height == 0) {
>+    return -1;
>+  }
>+  // Negative height means invert the image.
>+  if (height < 0) {
>+    height = -height;
>+    halfheight = (height + 1) >> 1;
>+    src_y = src_y + (height - 1) * src_stride_y;
>+    src_u = src_u + (halfheight - 1) * src_stride_u;
>+    src_v = src_v + (halfheight - 1) * src_stride_v;
>+    src_stride_y = -src_stride_y;
>+    src_stride_u = -src_stride_u;
>+    src_stride_v = -src_stride_v;
>+  }
>+
>+  // Convert Y plane.
>+  Convert16To8Plane(src_y, src_stride_y, dst_y, dst_stride_y, 16384, width,
>+                    height);
>+  // Convert UV planes.
>+  Convert16To8Plane(src_u, src_stride_u, dst_u, dst_stride_u, 16384, halfwidth,
>+                    halfheight);
>+  Convert16To8Plane(src_v, src_stride_v, dst_v, dst_stride_v, 16384, halfwidth,
>+                    halfheight);
>+  return 0;
>+}
>+
> // 422 chroma is 1/2 width, 1x height
> // 420 chroma is 1/2 width, 1/2 height
> LIBYUV_API
>-int I422ToI420(const uint8* src_y,
>+int I422ToI420(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_u,
>+               uint8_t* dst_u,
>                int dst_stride_u,
>-               uint8* dst_v,
>+               uint8_t* dst_v,
>                int dst_stride_v,
>                int width,
>                int height) {
>@@ -132,17 +218,17 @@ int I422ToI420(const uint8* src_y,
> // 444 chroma is 1x width, 1x height
> // 420 chroma is 1/2 width, 1/2 height
> LIBYUV_API
>-int I444ToI420(const uint8* src_y,
>+int I444ToI420(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_u,
>+               uint8_t* dst_u,
>                int dst_stride_u,
>-               uint8* dst_v,
>+               uint8_t* dst_v,
>                int dst_stride_v,
>                int width,
>                int height) {
>@@ -153,13 +239,13 @@ int I444ToI420(const uint8* src_y,
> 
> // I400 is greyscale typically used in MJPG
> LIBYUV_API
>-int I400ToI420(const uint8* src_y,
>+int I400ToI420(const uint8_t* src_y,
>                int src_stride_y,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_u,
>+               uint8_t* dst_u,
>                int dst_stride_u,
>-               uint8* dst_v,
>+               uint8_t* dst_v,
>                int dst_stride_v,
>                int width,
>                int height) {
>@@ -183,15 +269,15 @@ int I400ToI420(const uint8* src_y,
>   return 0;
> }
> 
>-static void CopyPlane2(const uint8* src,
>+static void CopyPlane2(const uint8_t* src,
>                        int src_stride_0,
>                        int src_stride_1,
>-                       uint8* dst,
>+                       uint8_t* dst,
>                        int dst_stride,
>                        int width,
>                        int height) {
>   int y;
>-  void (*CopyRow)(const uint8* src, uint8* dst, int width) = CopyRow_C;
>+  void (*CopyRow)(const uint8_t* src, uint8_t* dst, int width) = CopyRow_C;
> #if defined(HAS_COPYROW_SSE2)
>   if (TestCpuFlag(kCpuHasSSE2)) {
>     CopyRow = IS_ALIGNED(width, 32) ? CopyRow_SSE2 : CopyRow_Any_SSE2;
>@@ -212,11 +298,6 @@ static void CopyPlane2(const uint8* src,
>     CopyRow = IS_ALIGNED(width, 32) ? CopyRow_NEON : CopyRow_Any_NEON;
>   }
> #endif
>-#if defined(HAS_COPYROW_MIPS)
>-  if (TestCpuFlag(kCpuHasMIPS)) {
>-    CopyRow = CopyRow_MIPS;
>-  }
>-#endif
> 
>   // Copy plane
>   for (y = 0; y < height - 1; y += 2) {
>@@ -239,16 +320,16 @@ static void CopyPlane2(const uint8* src,
> // src_stride_m420 is row planar. Normally this will be the width in pixels.
> //   The UV plane is half width, but 2 values, so src_stride_m420 applies to
> //   this as well as the two Y planes.
>-static int X420ToI420(const uint8* src_y,
>+static int X420ToI420(const uint8_t* src_y,
>                       int src_stride_y0,
>                       int src_stride_y1,
>-                      const uint8* src_uv,
>+                      const uint8_t* src_uv,
>                       int src_stride_uv,
>-                      uint8* dst_y,
>+                      uint8_t* dst_y,
>                       int dst_stride_y,
>-                      uint8* dst_u,
>+                      uint8_t* dst_u,
>                       int dst_stride_u,
>-                      uint8* dst_v,
>+                      uint8_t* dst_v,
>                       int dst_stride_v,
>                       int width,
>                       int height) {
>@@ -303,15 +384,15 @@ static int X420ToI420(const uint8* src_y,
> 
> // Convert NV12 to I420.
> LIBYUV_API
>-int NV12ToI420(const uint8* src_y,
>+int NV12ToI420(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_uv,
>+               const uint8_t* src_uv,
>                int src_stride_uv,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_u,
>+               uint8_t* dst_u,
>                int dst_stride_u,
>-               uint8* dst_v,
>+               uint8_t* dst_v,
>                int dst_stride_v,
>                int width,
>                int height) {
>@@ -322,15 +403,15 @@ int NV12ToI420(const uint8* src_y,
> 
> // Convert NV21 to I420.  Same as NV12 but u and v pointers swapped.
> LIBYUV_API
>-int NV21ToI420(const uint8* src_y,
>+int NV21ToI420(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_vu,
>+               const uint8_t* src_vu,
>                int src_stride_vu,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_u,
>+               uint8_t* dst_u,
>                int dst_stride_u,
>-               uint8* dst_v,
>+               uint8_t* dst_v,
>                int dst_stride_v,
>                int width,
>                int height) {
>@@ -341,13 +422,13 @@ int NV21ToI420(const uint8* src_y,
> 
> // Convert M420 to I420.
> LIBYUV_API
>-int M420ToI420(const uint8* src_m420,
>+int M420ToI420(const uint8_t* src_m420,
>                int src_stride_m420,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_u,
>+               uint8_t* dst_u,
>                int dst_stride_u,
>-               uint8* dst_v,
>+               uint8_t* dst_v,
>                int dst_stride_v,
>                int width,
>                int height) {
>@@ -359,20 +440,21 @@ int M420ToI420(const uint8* src_m420,
> 
> // Convert YUY2 to I420.
> LIBYUV_API
>-int YUY2ToI420(const uint8* src_yuy2,
>+int YUY2ToI420(const uint8_t* src_yuy2,
>                int src_stride_yuy2,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_u,
>+               uint8_t* dst_u,
>                int dst_stride_u,
>-               uint8* dst_v,
>+               uint8_t* dst_v,
>                int dst_stride_v,
>                int width,
>                int height) {
>   int y;
>-  void (*YUY2ToUVRow)(const uint8* src_yuy2, int src_stride_yuy2, uint8* dst_u,
>-                      uint8* dst_v, int width) = YUY2ToUVRow_C;
>-  void (*YUY2ToYRow)(const uint8* src_yuy2, uint8* dst_y, int width) =
>+  void (*YUY2ToUVRow)(const uint8_t* src_yuy2, int src_stride_yuy2,
>+                      uint8_t* dst_u, uint8_t* dst_v, int width) =
>+      YUY2ToUVRow_C;
>+  void (*YUY2ToYRow)(const uint8_t* src_yuy2, uint8_t* dst_y, int width) =
>       YUY2ToYRow_C;
>   // Negative height means invert the image.
>   if (height < 0) {
>@@ -439,20 +521,21 @@ int YUY2ToI420(const uint8* src_yuy2,
> 
> // Convert UYVY to I420.
> LIBYUV_API
>-int UYVYToI420(const uint8* src_uyvy,
>+int UYVYToI420(const uint8_t* src_uyvy,
>                int src_stride_uyvy,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_u,
>+               uint8_t* dst_u,
>                int dst_stride_u,
>-               uint8* dst_v,
>+               uint8_t* dst_v,
>                int dst_stride_v,
>                int width,
>                int height) {
>   int y;
>-  void (*UYVYToUVRow)(const uint8* src_uyvy, int src_stride_uyvy, uint8* dst_u,
>-                      uint8* dst_v, int width) = UYVYToUVRow_C;
>-  void (*UYVYToYRow)(const uint8* src_uyvy, uint8* dst_y, int width) =
>+  void (*UYVYToUVRow)(const uint8_t* src_uyvy, int src_stride_uyvy,
>+                      uint8_t* dst_u, uint8_t* dst_v, int width) =
>+      UYVYToUVRow_C;
>+  void (*UYVYToYRow)(const uint8_t* src_uyvy, uint8_t* dst_y, int width) =
>       UYVYToYRow_C;
>   // Negative height means invert the image.
>   if (height < 0) {
>@@ -519,20 +602,21 @@ int UYVYToI420(const uint8* src_uyvy,
> 
> // Convert ARGB to I420.
> LIBYUV_API
>-int ARGBToI420(const uint8* src_argb,
>+int ARGBToI420(const uint8_t* src_argb,
>                int src_stride_argb,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_u,
>+               uint8_t* dst_u,
>                int dst_stride_u,
>-               uint8* dst_v,
>+               uint8_t* dst_v,
>                int dst_stride_v,
>                int width,
>                int height) {
>   int y;
>-  void (*ARGBToUVRow)(const uint8* src_argb0, int src_stride_argb, uint8* dst_u,
>-                      uint8* dst_v, int width) = ARGBToUVRow_C;
>-  void (*ARGBToYRow)(const uint8* src_argb, uint8* dst_y, int width) =
>+  void (*ARGBToUVRow)(const uint8_t* src_argb0, int src_stride_argb,
>+                      uint8_t* dst_u, uint8_t* dst_v, int width) =
>+      ARGBToUVRow_C;
>+  void (*ARGBToYRow)(const uint8_t* src_argb, uint8_t* dst_y, int width) =
>       ARGBToYRow_C;
>   if (!src_argb || !dst_y || !dst_u || !dst_v || width <= 0 || height == 0) {
>     return -1;
>@@ -579,14 +663,6 @@ int ARGBToI420(const uint8* src_argb,
>     }
>   }
> #endif
>-#if defined(HAS_ARGBTOYROW_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2)) {
>-    ARGBToYRow = ARGBToYRow_Any_DSPR2;
>-    if (IS_ALIGNED(width, 8)) {
>-      ARGBToYRow = ARGBToYRow_DSPR2;
>-    }
>-  }
>-#endif
> #if defined(HAS_ARGBTOYROW_MSA)
>   if (TestCpuFlag(kCpuHasMSA)) {
>     ARGBToYRow = ARGBToYRow_Any_MSA;
>@@ -595,14 +671,6 @@ int ARGBToI420(const uint8* src_argb,
>     }
>   }
> #endif
>-#if defined(HAS_ARGBTOUVROW_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2)) {
>-    ARGBToUVRow = ARGBToUVRow_Any_DSPR2;
>-    if (IS_ALIGNED(width, 16)) {
>-      ARGBToUVRow = ARGBToUVRow_DSPR2;
>-    }
>-  }
>-#endif
> #if defined(HAS_ARGBTOUVROW_MSA)
>   if (TestCpuFlag(kCpuHasMSA)) {
>     ARGBToUVRow = ARGBToUVRow_Any_MSA;
>@@ -630,20 +698,21 @@ int ARGBToI420(const uint8* src_argb,
> 
> // Convert BGRA to I420.
> LIBYUV_API
>-int BGRAToI420(const uint8* src_bgra,
>+int BGRAToI420(const uint8_t* src_bgra,
>                int src_stride_bgra,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_u,
>+               uint8_t* dst_u,
>                int dst_stride_u,
>-               uint8* dst_v,
>+               uint8_t* dst_v,
>                int dst_stride_v,
>                int width,
>                int height) {
>   int y;
>-  void (*BGRAToUVRow)(const uint8* src_bgra0, int src_stride_bgra, uint8* dst_u,
>-                      uint8* dst_v, int width) = BGRAToUVRow_C;
>-  void (*BGRAToYRow)(const uint8* src_bgra, uint8* dst_y, int width) =
>+  void (*BGRAToUVRow)(const uint8_t* src_bgra0, int src_stride_bgra,
>+                      uint8_t* dst_u, uint8_t* dst_v, int width) =
>+      BGRAToUVRow_C;
>+  void (*BGRAToYRow)(const uint8_t* src_bgra, uint8_t* dst_y, int width) =
>       BGRAToYRow_C;
>   if (!src_bgra || !dst_y || !dst_u || !dst_v || width <= 0 || height == 0) {
>     return -1;
>@@ -680,22 +749,6 @@ int BGRAToI420(const uint8* src_bgra,
>     }
>   }
> #endif
>-#if defined(HAS_BGRATOYROW_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2)) {
>-    BGRAToYRow = BGRAToYRow_Any_DSPR2;
>-    if (IS_ALIGNED(width, 8)) {
>-      BGRAToYRow = BGRAToYRow_DSPR2;
>-    }
>-  }
>-#endif
>-#if defined(HAS_BGRATOUVROW_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2)) {
>-    BGRAToUVRow = BGRAToUVRow_Any_DSPR2;
>-    if (IS_ALIGNED(width, 16)) {
>-      BGRAToUVRow = BGRAToUVRow_DSPR2;
>-    }
>-  }
>-#endif
> #if defined(HAS_BGRATOYROW_MSA)
>   if (TestCpuFlag(kCpuHasMSA)) {
>     BGRAToYRow = BGRAToYRow_Any_MSA;
>@@ -731,20 +784,21 @@ int BGRAToI420(const uint8* src_bgra,
> 
> // Convert ABGR to I420.
> LIBYUV_API
>-int ABGRToI420(const uint8* src_abgr,
>+int ABGRToI420(const uint8_t* src_abgr,
>                int src_stride_abgr,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_u,
>+               uint8_t* dst_u,
>                int dst_stride_u,
>-               uint8* dst_v,
>+               uint8_t* dst_v,
>                int dst_stride_v,
>                int width,
>                int height) {
>   int y;
>-  void (*ABGRToUVRow)(const uint8* src_abgr0, int src_stride_abgr, uint8* dst_u,
>-                      uint8* dst_v, int width) = ABGRToUVRow_C;
>-  void (*ABGRToYRow)(const uint8* src_abgr, uint8* dst_y, int width) =
>+  void (*ABGRToUVRow)(const uint8_t* src_abgr0, int src_stride_abgr,
>+                      uint8_t* dst_u, uint8_t* dst_v, int width) =
>+      ABGRToUVRow_C;
>+  void (*ABGRToYRow)(const uint8_t* src_abgr, uint8_t* dst_y, int width) =
>       ABGRToYRow_C;
>   if (!src_abgr || !dst_y || !dst_u || !dst_v || width <= 0 || height == 0) {
>     return -1;
>@@ -781,22 +835,6 @@ int ABGRToI420(const uint8* src_abgr,
>     }
>   }
> #endif
>-#if defined(HAS_ABGRTOYROW_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2)) {
>-    ABGRToYRow = ABGRToYRow_Any_DSPR2;
>-    if (IS_ALIGNED(width, 8)) {
>-      ABGRToYRow = ABGRToYRow_DSPR2;
>-    }
>-  }
>-#endif
>-#if defined(HAS_ABGRTOUVROW_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2)) {
>-    ABGRToUVRow = ABGRToUVRow_Any_DSPR2;
>-    if (IS_ALIGNED(width, 16)) {
>-      ABGRToUVRow = ABGRToUVRow_DSPR2;
>-    }
>-  }
>-#endif
> #if defined(HAS_ABGRTOYROW_MSA)
>   if (TestCpuFlag(kCpuHasMSA)) {
>     ABGRToYRow = ABGRToYRow_Any_MSA;
>@@ -832,20 +870,21 @@ int ABGRToI420(const uint8* src_abgr,
> 
> // Convert RGBA to I420.
> LIBYUV_API
>-int RGBAToI420(const uint8* src_rgba,
>+int RGBAToI420(const uint8_t* src_rgba,
>                int src_stride_rgba,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_u,
>+               uint8_t* dst_u,
>                int dst_stride_u,
>-               uint8* dst_v,
>+               uint8_t* dst_v,
>                int dst_stride_v,
>                int width,
>                int height) {
>   int y;
>-  void (*RGBAToUVRow)(const uint8* src_rgba0, int src_stride_rgba, uint8* dst_u,
>-                      uint8* dst_v, int width) = RGBAToUVRow_C;
>-  void (*RGBAToYRow)(const uint8* src_rgba, uint8* dst_y, int width) =
>+  void (*RGBAToUVRow)(const uint8_t* src_rgba0, int src_stride_rgba,
>+                      uint8_t* dst_u, uint8_t* dst_v, int width) =
>+      RGBAToUVRow_C;
>+  void (*RGBAToYRow)(const uint8_t* src_rgba, uint8_t* dst_y, int width) =
>       RGBAToYRow_C;
>   if (!src_rgba || !dst_y || !dst_u || !dst_v || width <= 0 || height == 0) {
>     return -1;
>@@ -882,22 +921,6 @@ int RGBAToI420(const uint8* src_rgba,
>     }
>   }
> #endif
>-#if defined(HAS_RGBATOYROW_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2)) {
>-    RGBAToYRow = RGBAToYRow_Any_DSPR2;
>-    if (IS_ALIGNED(width, 8)) {
>-      RGBAToYRow = RGBAToYRow_DSPR2;
>-    }
>-  }
>-#endif
>-#if defined(HAS_RGBATOUVROW_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2)) {
>-    RGBAToUVRow = RGBAToUVRow_Any_DSPR2;
>-    if (IS_ALIGNED(width, 16)) {
>-      RGBAToUVRow = RGBAToUVRow_DSPR2;
>-    }
>-  }
>-#endif
> #if defined(HAS_RGBATOYROW_MSA)
>   if (TestCpuFlag(kCpuHasMSA)) {
>     RGBAToYRow = RGBAToYRow_Any_MSA;
>@@ -933,28 +956,30 @@ int RGBAToI420(const uint8* src_rgba,
> 
> // Convert RGB24 to I420.
> LIBYUV_API
>-int RGB24ToI420(const uint8* src_rgb24,
>+int RGB24ToI420(const uint8_t* src_rgb24,
>                 int src_stride_rgb24,
>-                uint8* dst_y,
>+                uint8_t* dst_y,
>                 int dst_stride_y,
>-                uint8* dst_u,
>+                uint8_t* dst_u,
>                 int dst_stride_u,
>-                uint8* dst_v,
>+                uint8_t* dst_v,
>                 int dst_stride_v,
>                 int width,
>                 int height) {
>   int y;
> #if (defined(HAS_RGB24TOYROW_NEON) || defined(HAS_RGB24TOYROW_MSA))
>-  void (*RGB24ToUVRow)(const uint8* src_rgb24, int src_stride_rgb24,
>-                       uint8* dst_u, uint8* dst_v, int width) = RGB24ToUVRow_C;
>-  void (*RGB24ToYRow)(const uint8* src_rgb24, uint8* dst_y, int width) =
>+  void (*RGB24ToUVRow)(const uint8_t* src_rgb24, int src_stride_rgb24,
>+                       uint8_t* dst_u, uint8_t* dst_v, int width) =
>+      RGB24ToUVRow_C;
>+  void (*RGB24ToYRow)(const uint8_t* src_rgb24, uint8_t* dst_y, int width) =
>       RGB24ToYRow_C;
> #else
>-  void (*RGB24ToARGBRow)(const uint8* src_rgb, uint8* dst_argb, int width) =
>+  void (*RGB24ToARGBRow)(const uint8_t* src_rgb, uint8_t* dst_argb, int width) =
>       RGB24ToARGBRow_C;
>-  void (*ARGBToUVRow)(const uint8* src_argb0, int src_stride_argb, uint8* dst_u,
>-                      uint8* dst_v, int width) = ARGBToUVRow_C;
>-  void (*ARGBToYRow)(const uint8* src_argb, uint8* dst_y, int width) =
>+  void (*ARGBToUVRow)(const uint8_t* src_argb0, int src_stride_argb,
>+                      uint8_t* dst_u, uint8_t* dst_v, int width) =
>+      ARGBToUVRow_C;
>+  void (*ARGBToYRow)(const uint8_t* src_argb, uint8_t* dst_y, int width) =
>       ARGBToYRow_C;
> #endif
>   if (!src_rgb24 || !dst_y || !dst_u || !dst_v || width <= 0 || height == 0) {
>@@ -1063,28 +1088,29 @@ int RGB24ToI420(const uint8* src_rgb24,
> 
> // Convert RAW to I420.
> LIBYUV_API
>-int RAWToI420(const uint8* src_raw,
>+int RAWToI420(const uint8_t* src_raw,
>               int src_stride_raw,
>-              uint8* dst_y,
>+              uint8_t* dst_y,
>               int dst_stride_y,
>-              uint8* dst_u,
>+              uint8_t* dst_u,
>               int dst_stride_u,
>-              uint8* dst_v,
>+              uint8_t* dst_v,
>               int dst_stride_v,
>               int width,
>               int height) {
>   int y;
> #if (defined(HAS_RAWTOYROW_NEON) || defined(HAS_RAWTOYROW_MSA))
>-  void (*RAWToUVRow)(const uint8* src_raw, int src_stride_raw, uint8* dst_u,
>-                     uint8* dst_v, int width) = RAWToUVRow_C;
>-  void (*RAWToYRow)(const uint8* src_raw, uint8* dst_y, int width) =
>+  void (*RAWToUVRow)(const uint8_t* src_raw, int src_stride_raw, uint8_t* dst_u,
>+                     uint8_t* dst_v, int width) = RAWToUVRow_C;
>+  void (*RAWToYRow)(const uint8_t* src_raw, uint8_t* dst_y, int width) =
>       RAWToYRow_C;
> #else
>-  void (*RAWToARGBRow)(const uint8* src_rgb, uint8* dst_argb, int width) =
>+  void (*RAWToARGBRow)(const uint8_t* src_rgb, uint8_t* dst_argb, int width) =
>       RAWToARGBRow_C;
>-  void (*ARGBToUVRow)(const uint8* src_argb0, int src_stride_argb, uint8* dst_u,
>-                      uint8* dst_v, int width) = ARGBToUVRow_C;
>-  void (*ARGBToYRow)(const uint8* src_argb, uint8* dst_y, int width) =
>+  void (*ARGBToUVRow)(const uint8_t* src_argb0, int src_stride_argb,
>+                      uint8_t* dst_u, uint8_t* dst_v, int width) =
>+      ARGBToUVRow_C;
>+  void (*ARGBToYRow)(const uint8_t* src_argb, uint8_t* dst_y, int width) =
>       ARGBToYRow_C;
> #endif
>   if (!src_raw || !dst_y || !dst_u || !dst_v || width <= 0 || height == 0) {
>@@ -1193,29 +1219,30 @@ int RAWToI420(const uint8* src_raw,
> 
> // Convert RGB565 to I420.
> LIBYUV_API
>-int RGB565ToI420(const uint8* src_rgb565,
>+int RGB565ToI420(const uint8_t* src_rgb565,
>                  int src_stride_rgb565,
>-                 uint8* dst_y,
>+                 uint8_t* dst_y,
>                  int dst_stride_y,
>-                 uint8* dst_u,
>+                 uint8_t* dst_u,
>                  int dst_stride_u,
>-                 uint8* dst_v,
>+                 uint8_t* dst_v,
>                  int dst_stride_v,
>                  int width,
>                  int height) {
>   int y;
> #if (defined(HAS_RGB565TOYROW_NEON) || defined(HAS_RGB565TOYROW_MSA))
>-  void (*RGB565ToUVRow)(const uint8* src_rgb565, int src_stride_rgb565,
>-                        uint8* dst_u, uint8* dst_v, int width) =
>+  void (*RGB565ToUVRow)(const uint8_t* src_rgb565, int src_stride_rgb565,
>+                        uint8_t* dst_u, uint8_t* dst_v, int width) =
>       RGB565ToUVRow_C;
>-  void (*RGB565ToYRow)(const uint8* src_rgb565, uint8* dst_y, int width) =
>+  void (*RGB565ToYRow)(const uint8_t* src_rgb565, uint8_t* dst_y, int width) =
>       RGB565ToYRow_C;
> #else
>-  void (*RGB565ToARGBRow)(const uint8* src_rgb, uint8* dst_argb, int width) =
>-      RGB565ToARGBRow_C;
>-  void (*ARGBToUVRow)(const uint8* src_argb0, int src_stride_argb, uint8* dst_u,
>-                      uint8* dst_v, int width) = ARGBToUVRow_C;
>-  void (*ARGBToYRow)(const uint8* src_argb, uint8* dst_y, int width) =
>+  void (*RGB565ToARGBRow)(const uint8_t* src_rgb, uint8_t* dst_argb,
>+                          int width) = RGB565ToARGBRow_C;
>+  void (*ARGBToUVRow)(const uint8_t* src_argb0, int src_stride_argb,
>+                      uint8_t* dst_u, uint8_t* dst_v, int width) =
>+      ARGBToUVRow_C;
>+  void (*ARGBToYRow)(const uint8_t* src_argb, uint8_t* dst_y, int width) =
>       ARGBToYRow_C;
> #endif
>   if (!src_rgb565 || !dst_y || !dst_u || !dst_v || width <= 0 || height == 0) {
>@@ -1287,14 +1314,6 @@ int RGB565ToI420(const uint8* src_rgb565,
>     }
>   }
> #endif
>-#if defined(HAS_RGB565TOARGBROW_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2)) {
>-    RGB565ToARGBRow = RGB565ToARGBRow_Any_DSPR2;
>-    if (IS_ALIGNED(width, 8)) {
>-      RGB565ToARGBRow = RGB565ToARGBRow_DSPR2;
>-    }
>-  }
>-#endif
> #endif
>   {
> #if !(defined(HAS_RGB565TOYROW_NEON) || defined(HAS_RGB565TOYROW_MSA))
>@@ -1338,29 +1357,30 @@ int RGB565ToI420(const uint8* src_rgb565,
> 
> // Convert ARGB1555 to I420.
> LIBYUV_API
>-int ARGB1555ToI420(const uint8* src_argb1555,
>+int ARGB1555ToI420(const uint8_t* src_argb1555,
>                    int src_stride_argb1555,
>-                   uint8* dst_y,
>+                   uint8_t* dst_y,
>                    int dst_stride_y,
>-                   uint8* dst_u,
>+                   uint8_t* dst_u,
>                    int dst_stride_u,
>-                   uint8* dst_v,
>+                   uint8_t* dst_v,
>                    int dst_stride_v,
>                    int width,
>                    int height) {
>   int y;
> #if (defined(HAS_ARGB1555TOYROW_NEON) || defined(HAS_ARGB1555TOYROW_MSA))
>-  void (*ARGB1555ToUVRow)(const uint8* src_argb1555, int src_stride_argb1555,
>-                          uint8* dst_u, uint8* dst_v, int width) =
>+  void (*ARGB1555ToUVRow)(const uint8_t* src_argb1555, int src_stride_argb1555,
>+                          uint8_t* dst_u, uint8_t* dst_v, int width) =
>       ARGB1555ToUVRow_C;
>-  void (*ARGB1555ToYRow)(const uint8* src_argb1555, uint8* dst_y, int width) =
>-      ARGB1555ToYRow_C;
>+  void (*ARGB1555ToYRow)(const uint8_t* src_argb1555, uint8_t* dst_y,
>+                         int width) = ARGB1555ToYRow_C;
> #else
>-  void (*ARGB1555ToARGBRow)(const uint8* src_rgb, uint8* dst_argb, int width) =
>-      ARGB1555ToARGBRow_C;
>-  void (*ARGBToUVRow)(const uint8* src_argb0, int src_stride_argb, uint8* dst_u,
>-                      uint8* dst_v, int width) = ARGBToUVRow_C;
>-  void (*ARGBToYRow)(const uint8* src_argb, uint8* dst_y, int width) =
>+  void (*ARGB1555ToARGBRow)(const uint8_t* src_rgb, uint8_t* dst_argb,
>+                            int width) = ARGB1555ToARGBRow_C;
>+  void (*ARGBToUVRow)(const uint8_t* src_argb0, int src_stride_argb,
>+                      uint8_t* dst_u, uint8_t* dst_v, int width) =
>+      ARGBToUVRow_C;
>+  void (*ARGBToYRow)(const uint8_t* src_argb, uint8_t* dst_y, int width) =
>       ARGBToYRow_C;
> #endif
>   if (!src_argb1555 || !dst_y || !dst_u || !dst_v || width <= 0 ||
>@@ -1479,29 +1499,30 @@ int ARGB1555ToI420(const uint8* src_argb1555,
> 
> // Convert ARGB4444 to I420.
> LIBYUV_API
>-int ARGB4444ToI420(const uint8* src_argb4444,
>+int ARGB4444ToI420(const uint8_t* src_argb4444,
>                    int src_stride_argb4444,
>-                   uint8* dst_y,
>+                   uint8_t* dst_y,
>                    int dst_stride_y,
>-                   uint8* dst_u,
>+                   uint8_t* dst_u,
>                    int dst_stride_u,
>-                   uint8* dst_v,
>+                   uint8_t* dst_v,
>                    int dst_stride_v,
>                    int width,
>                    int height) {
>   int y;
> #if defined(HAS_ARGB4444TOYROW_NEON)
>-  void (*ARGB4444ToUVRow)(const uint8* src_argb4444, int src_stride_argb4444,
>-                          uint8* dst_u, uint8* dst_v, int width) =
>+  void (*ARGB4444ToUVRow)(const uint8_t* src_argb4444, int src_stride_argb4444,
>+                          uint8_t* dst_u, uint8_t* dst_v, int width) =
>       ARGB4444ToUVRow_C;
>-  void (*ARGB4444ToYRow)(const uint8* src_argb4444, uint8* dst_y, int width) =
>-      ARGB4444ToYRow_C;
>+  void (*ARGB4444ToYRow)(const uint8_t* src_argb4444, uint8_t* dst_y,
>+                         int width) = ARGB4444ToYRow_C;
> #else
>-  void (*ARGB4444ToARGBRow)(const uint8* src_rgb, uint8* dst_argb, int width) =
>-      ARGB4444ToARGBRow_C;
>-  void (*ARGBToUVRow)(const uint8* src_argb0, int src_stride_argb, uint8* dst_u,
>-                      uint8* dst_v, int width) = ARGBToUVRow_C;
>-  void (*ARGBToYRow)(const uint8* src_argb, uint8* dst_y, int width) =
>+  void (*ARGB4444ToARGBRow)(const uint8_t* src_rgb, uint8_t* dst_argb,
>+                            int width) = ARGB4444ToARGBRow_C;
>+  void (*ARGBToUVRow)(const uint8_t* src_argb0, int src_stride_argb,
>+                      uint8_t* dst_u, uint8_t* dst_v, int width) =
>+      ARGBToUVRow_C;
>+  void (*ARGBToYRow)(const uint8_t* src_argb, uint8_t* dst_y, int width) =
>       ARGBToYRow_C;
> #endif
>   if (!src_argb4444 || !dst_y || !dst_u || !dst_v || width <= 0 ||
>@@ -1630,9 +1651,9 @@ int ARGB4444ToI420(const uint8* src_argb4444,
>   return 0;
> }
> 
>-static void SplitPixels(const uint8* src_u,
>+static void SplitPixels(const uint8_t* src_u,
>                         int src_pixel_stride_uv,
>-                        uint8* dst_u,
>+                        uint8_t* dst_u,
>                         int width) {
>   int i;
>   for (i = 0; i < width; ++i) {
>@@ -1644,18 +1665,18 @@ static void SplitPixels(const uint8* src_u,
> 
> // Convert Android420 to I420.
> LIBYUV_API
>-int Android420ToI420(const uint8* src_y,
>+int Android420ToI420(const uint8_t* src_y,
>                      int src_stride_y,
>-                     const uint8* src_u,
>+                     const uint8_t* src_u,
>                      int src_stride_u,
>-                     const uint8* src_v,
>+                     const uint8_t* src_v,
>                      int src_stride_v,
>                      int src_pixel_stride_uv,
>-                     uint8* dst_y,
>+                     uint8_t* dst_y,
>                      int dst_stride_y,
>-                     uint8* dst_u,
>+                     uint8_t* dst_u,
>                      int dst_stride_u,
>-                     uint8* dst_v,
>+                     uint8_t* dst_v,
>                      int dst_stride_v,
>                      int width,
>                      int height) {
>@@ -1688,14 +1709,15 @@ int Android420ToI420(const uint8* src_y,
>     CopyPlane(src_v, src_stride_v, dst_v, dst_stride_v, halfwidth, halfheight);
>     return 0;
>     // Split UV planes - NV21
>-  } else if (src_pixel_stride_uv == 2 && vu_off == -1 &&
>-             src_stride_u == src_stride_v) {
>+  }
>+  if (src_pixel_stride_uv == 2 && vu_off == -1 &&
>+      src_stride_u == src_stride_v) {
>     SplitUVPlane(src_v, src_stride_v, dst_v, dst_stride_v, dst_u, dst_stride_u,
>                  halfwidth, halfheight);
>     return 0;
>     // Split UV planes - NV12
>-  } else if (src_pixel_stride_uv == 2 && vu_off == 1 &&
>-             src_stride_u == src_stride_v) {
>+  }
>+  if (src_pixel_stride_uv == 2 && vu_off == 1 && src_stride_u == src_stride_v) {
>     SplitUVPlane(src_u, src_stride_u, dst_u, dst_stride_u, dst_v, dst_stride_v,
>                  halfwidth, halfheight);
>     return 0;
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/convert_argb.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/convert_argb.cc
>index 5007bdb9708..4c317ae30c6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/convert_argb.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/convert_argb.cc
>@@ -26,9 +26,9 @@ extern "C" {
> 
> // Copy ARGB with optional flipping
> LIBYUV_API
>-int ARGBCopy(const uint8* src_argb,
>+int ARGBCopy(const uint8_t* src_argb,
>              int src_stride_argb,
>-             uint8* dst_argb,
>+             uint8_t* dst_argb,
>              int dst_stride_argb,
>              int width,
>              int height) {
>@@ -47,21 +47,21 @@ int ARGBCopy(const uint8* src_argb,
>   return 0;
> }
> 
>-// Convert I422 to ARGB with matrix
>-static int I420ToARGBMatrix(const uint8* src_y,
>+// Convert I420 to ARGB with matrix
>+static int I420ToARGBMatrix(const uint8_t* src_y,
>                             int src_stride_y,
>-                            const uint8* src_u,
>+                            const uint8_t* src_u,
>                             int src_stride_u,
>-                            const uint8* src_v,
>+                            const uint8_t* src_v,
>                             int src_stride_v,
>-                            uint8* dst_argb,
>+                            uint8_t* dst_argb,
>                             int dst_stride_argb,
>                             const struct YuvConstants* yuvconstants,
>                             int width,
>                             int height) {
>   int y;
>-  void (*I422ToARGBRow)(const uint8* y_buf, const uint8* u_buf,
>-                        const uint8* v_buf, uint8* rgb_buf,
>+  void (*I422ToARGBRow)(const uint8_t* y_buf, const uint8_t* u_buf,
>+                        const uint8_t* v_buf, uint8_t* rgb_buf,
>                         const struct YuvConstants* yuvconstants, int width) =
>       I422ToARGBRow_C;
>   if (!src_y || !src_u || !src_v || !dst_argb || width <= 0 || height == 0) {
>@@ -97,15 +97,6 @@ static int I420ToARGBMatrix(const uint8* src_y,
>     }
>   }
> #endif
>-#if defined(HAS_I422TOARGBROW_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2) && IS_ALIGNED(width, 4) &&
>-      IS_ALIGNED(src_y, 4) && IS_ALIGNED(src_stride_y, 4) &&
>-      IS_ALIGNED(src_u, 2) && IS_ALIGNED(src_stride_u, 2) &&
>-      IS_ALIGNED(src_v, 2) && IS_ALIGNED(src_stride_v, 2) &&
>-      IS_ALIGNED(dst_argb, 4) && IS_ALIGNED(dst_stride_argb, 4)) {
>-    I422ToARGBRow = I422ToARGBRow_DSPR2;
>-  }
>-#endif
> #if defined(HAS_I422TOARGBROW_MSA)
>   if (TestCpuFlag(kCpuHasMSA)) {
>     I422ToARGBRow = I422ToARGBRow_Any_MSA;
>@@ -129,13 +120,13 @@ static int I420ToARGBMatrix(const uint8* src_y,
> 
> // Convert I420 to ARGB.
> LIBYUV_API
>-int I420ToARGB(const uint8* src_y,
>+int I420ToARGB(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_argb,
>+               uint8_t* dst_argb,
>                int dst_stride_argb,
>                int width,
>                int height) {
>@@ -146,13 +137,13 @@ int I420ToARGB(const uint8* src_y,
> 
> // Convert I420 to ABGR.
> LIBYUV_API
>-int I420ToABGR(const uint8* src_y,
>+int I420ToABGR(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_abgr,
>+               uint8_t* dst_abgr,
>                int dst_stride_abgr,
>                int width,
>                int height) {
>@@ -165,13 +156,13 @@ int I420ToABGR(const uint8* src_y,
> 
> // Convert J420 to ARGB.
> LIBYUV_API
>-int J420ToARGB(const uint8* src_y,
>+int J420ToARGB(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_argb,
>+               uint8_t* dst_argb,
>                int dst_stride_argb,
>                int width,
>                int height) {
>@@ -182,13 +173,13 @@ int J420ToARGB(const uint8* src_y,
> 
> // Convert J420 to ABGR.
> LIBYUV_API
>-int J420ToABGR(const uint8* src_y,
>+int J420ToABGR(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_abgr,
>+               uint8_t* dst_abgr,
>                int dst_stride_abgr,
>                int width,
>                int height) {
>@@ -201,13 +192,13 @@ int J420ToABGR(const uint8* src_y,
> 
> // Convert H420 to ARGB.
> LIBYUV_API
>-int H420ToARGB(const uint8* src_y,
>+int H420ToARGB(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_argb,
>+               uint8_t* dst_argb,
>                int dst_stride_argb,
>                int width,
>                int height) {
>@@ -218,13 +209,13 @@ int H420ToARGB(const uint8* src_y,
> 
> // Convert H420 to ABGR.
> LIBYUV_API
>-int H420ToABGR(const uint8* src_y,
>+int H420ToABGR(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_abgr,
>+               uint8_t* dst_abgr,
>                int dst_stride_abgr,
>                int width,
>                int height) {
>@@ -236,20 +227,20 @@ int H420ToABGR(const uint8* src_y,
> }
> 
> // Convert I422 to ARGB with matrix
>-static int I422ToARGBMatrix(const uint8* src_y,
>+static int I422ToARGBMatrix(const uint8_t* src_y,
>                             int src_stride_y,
>-                            const uint8* src_u,
>+                            const uint8_t* src_u,
>                             int src_stride_u,
>-                            const uint8* src_v,
>+                            const uint8_t* src_v,
>                             int src_stride_v,
>-                            uint8* dst_argb,
>+                            uint8_t* dst_argb,
>                             int dst_stride_argb,
>                             const struct YuvConstants* yuvconstants,
>                             int width,
>                             int height) {
>   int y;
>-  void (*I422ToARGBRow)(const uint8* y_buf, const uint8* u_buf,
>-                        const uint8* v_buf, uint8* rgb_buf,
>+  void (*I422ToARGBRow)(const uint8_t* y_buf, const uint8_t* u_buf,
>+                        const uint8_t* v_buf, uint8_t* rgb_buf,
>                         const struct YuvConstants* yuvconstants, int width) =
>       I422ToARGBRow_C;
>   if (!src_y || !src_u || !src_v || !dst_argb || width <= 0 || height == 0) {
>@@ -292,15 +283,6 @@ static int I422ToARGBMatrix(const uint8* src_y,
>     }
>   }
> #endif
>-#if defined(HAS_I422TOARGBROW_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2) && IS_ALIGNED(width, 4) &&
>-      IS_ALIGNED(src_y, 4) && IS_ALIGNED(src_stride_y, 4) &&
>-      IS_ALIGNED(src_u, 2) && IS_ALIGNED(src_stride_u, 2) &&
>-      IS_ALIGNED(src_v, 2) && IS_ALIGNED(src_stride_v, 2) &&
>-      IS_ALIGNED(dst_argb, 4) && IS_ALIGNED(dst_stride_argb, 4)) {
>-    I422ToARGBRow = I422ToARGBRow_DSPR2;
>-  }
>-#endif
> #if defined(HAS_I422TOARGBROW_MSA)
>   if (TestCpuFlag(kCpuHasMSA)) {
>     I422ToARGBRow = I422ToARGBRow_Any_MSA;
>@@ -322,13 +304,13 @@ static int I422ToARGBMatrix(const uint8* src_y,
> 
> // Convert I422 to ARGB.
> LIBYUV_API
>-int I422ToARGB(const uint8* src_y,
>+int I422ToARGB(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_argb,
>+               uint8_t* dst_argb,
>                int dst_stride_argb,
>                int width,
>                int height) {
>@@ -339,13 +321,13 @@ int I422ToARGB(const uint8* src_y,
> 
> // Convert I422 to ABGR.
> LIBYUV_API
>-int I422ToABGR(const uint8* src_y,
>+int I422ToABGR(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_abgr,
>+               uint8_t* dst_abgr,
>                int dst_stride_abgr,
>                int width,
>                int height) {
>@@ -358,13 +340,13 @@ int I422ToABGR(const uint8* src_y,
> 
> // Convert J422 to ARGB.
> LIBYUV_API
>-int J422ToARGB(const uint8* src_y,
>+int J422ToARGB(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_argb,
>+               uint8_t* dst_argb,
>                int dst_stride_argb,
>                int width,
>                int height) {
>@@ -375,13 +357,13 @@ int J422ToARGB(const uint8* src_y,
> 
> // Convert J422 to ABGR.
> LIBYUV_API
>-int J422ToABGR(const uint8* src_y,
>+int J422ToABGR(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_abgr,
>+               uint8_t* dst_abgr,
>                int dst_stride_abgr,
>                int width,
>                int height) {
>@@ -394,13 +376,13 @@ int J422ToABGR(const uint8* src_y,
> 
> // Convert H422 to ARGB.
> LIBYUV_API
>-int H422ToARGB(const uint8* src_y,
>+int H422ToARGB(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_argb,
>+               uint8_t* dst_argb,
>                int dst_stride_argb,
>                int width,
>                int height) {
>@@ -411,13 +393,13 @@ int H422ToARGB(const uint8* src_y,
> 
> // Convert H422 to ABGR.
> LIBYUV_API
>-int H422ToABGR(const uint8* src_y,
>+int H422ToABGR(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_abgr,
>+               uint8_t* dst_abgr,
>                int dst_stride_abgr,
>                int width,
>                int height) {
>@@ -428,21 +410,271 @@ int H422ToABGR(const uint8* src_y,
>                           width, height);
> }
> 
>+// Convert 10 bit YUV to ARGB with matrix
>+// TODO(fbarchard): Consider passing scale multiplier to I210ToARGB to
>+// multiply 10 bit yuv into high bits to allow any number of bits.
>+static int I010ToAR30Matrix(const uint16_t* src_y,
>+                            int src_stride_y,
>+                            const uint16_t* src_u,
>+                            int src_stride_u,
>+                            const uint16_t* src_v,
>+                            int src_stride_v,
>+                            uint8_t* dst_ar30,
>+                            int dst_stride_ar30,
>+                            const struct YuvConstants* yuvconstants,
>+                            int width,
>+                            int height) {
>+  int y;
>+  void (*I210ToAR30Row)(const uint16_t* y_buf, const uint16_t* u_buf,
>+                        const uint16_t* v_buf, uint8_t* rgb_buf,
>+                        const struct YuvConstants* yuvconstants, int width) =
>+      I210ToAR30Row_C;
>+  if (!src_y || !src_u || !src_v || !dst_ar30 || width <= 0 || height == 0) {
>+    return -1;
>+  }
>+  // Negative height means invert the image.
>+  if (height < 0) {
>+    height = -height;
>+    dst_ar30 = dst_ar30 + (height - 1) * dst_stride_ar30;
>+    dst_stride_ar30 = -dst_stride_ar30;
>+  }
>+#if defined(HAS_I210TOAR30ROW_SSSE3)
>+  if (TestCpuFlag(kCpuHasSSSE3)) {
>+    I210ToAR30Row = I210ToAR30Row_Any_SSSE3;
>+    if (IS_ALIGNED(width, 8)) {
>+      I210ToAR30Row = I210ToAR30Row_SSSE3;
>+    }
>+  }
>+#endif
>+#if defined(HAS_I210TOAR30ROW_AVX2)
>+  if (TestCpuFlag(kCpuHasAVX2)) {
>+    I210ToAR30Row = I210ToAR30Row_Any_AVX2;
>+    if (IS_ALIGNED(width, 16)) {
>+      I210ToAR30Row = I210ToAR30Row_AVX2;
>+    }
>+  }
>+#endif
>+  for (y = 0; y < height; ++y) {
>+    I210ToAR30Row(src_y, src_u, src_v, dst_ar30, yuvconstants, width);
>+    dst_ar30 += dst_stride_ar30;
>+    src_y += src_stride_y;
>+    if (y & 1) {
>+      src_u += src_stride_u;
>+      src_v += src_stride_v;
>+    }
>+  }
>+  return 0;
>+}
>+
>+// Convert I010 to AR30.
>+LIBYUV_API
>+int I010ToAR30(const uint16_t* src_y,
>+               int src_stride_y,
>+               const uint16_t* src_u,
>+               int src_stride_u,
>+               const uint16_t* src_v,
>+               int src_stride_v,
>+               uint8_t* dst_ar30,
>+               int dst_stride_ar30,
>+               int width,
>+               int height) {
>+  return I010ToAR30Matrix(src_y, src_stride_y, src_u, src_stride_u, src_v,
>+                          src_stride_v, dst_ar30, dst_stride_ar30,
>+                          &kYuvI601Constants, width, height);
>+}
>+
>+// Convert H010 to AR30.
>+LIBYUV_API
>+int H010ToAR30(const uint16_t* src_y,
>+               int src_stride_y,
>+               const uint16_t* src_u,
>+               int src_stride_u,
>+               const uint16_t* src_v,
>+               int src_stride_v,
>+               uint8_t* dst_ar30,
>+               int dst_stride_ar30,
>+               int width,
>+               int height) {
>+  return I010ToAR30Matrix(src_y, src_stride_y, src_u, src_stride_u, src_v,
>+                          src_stride_v, dst_ar30, dst_stride_ar30,
>+                          &kYuvH709Constants, width, height);
>+}
>+
>+// Convert I010 to AB30.
>+LIBYUV_API
>+int I010ToAB30(const uint16_t* src_y,
>+               int src_stride_y,
>+               const uint16_t* src_u,
>+               int src_stride_u,
>+               const uint16_t* src_v,
>+               int src_stride_v,
>+               uint8_t* dst_ab30,
>+               int dst_stride_ab30,
>+               int width,
>+               int height) {
>+  return I010ToAR30Matrix(src_y, src_stride_y, src_v, src_stride_v, src_u,
>+                          src_stride_u, dst_ab30, dst_stride_ab30,
>+                          &kYvuI601Constants, width, height);
>+}
>+
>+// Convert H010 to AB30.
>+LIBYUV_API
>+int H010ToAB30(const uint16_t* src_y,
>+               int src_stride_y,
>+               const uint16_t* src_u,
>+               int src_stride_u,
>+               const uint16_t* src_v,
>+               int src_stride_v,
>+               uint8_t* dst_ab30,
>+               int dst_stride_ab30,
>+               int width,
>+               int height) {
>+  return I010ToAR30Matrix(src_y, src_stride_y, src_v, src_stride_v, src_u,
>+                          src_stride_u, dst_ab30, dst_stride_ab30,
>+                          &kYvuH709Constants, width, height);
>+}
>+
>+// Convert 10 bit YUV to ARGB with matrix
>+static int I010ToARGBMatrix(const uint16_t* src_y,
>+                            int src_stride_y,
>+                            const uint16_t* src_u,
>+                            int src_stride_u,
>+                            const uint16_t* src_v,
>+                            int src_stride_v,
>+                            uint8_t* dst_argb,
>+                            int dst_stride_argb,
>+                            const struct YuvConstants* yuvconstants,
>+                            int width,
>+                            int height) {
>+  int y;
>+  void (*I210ToARGBRow)(const uint16_t* y_buf, const uint16_t* u_buf,
>+                        const uint16_t* v_buf, uint8_t* rgb_buf,
>+                        const struct YuvConstants* yuvconstants, int width) =
>+      I210ToARGBRow_C;
>+  if (!src_y || !src_u || !src_v || !dst_argb || width <= 0 || height == 0) {
>+    return -1;
>+  }
>+  // Negative height means invert the image.
>+  if (height < 0) {
>+    height = -height;
>+    dst_argb = dst_argb + (height - 1) * dst_stride_argb;
>+    dst_stride_argb = -dst_stride_argb;
>+  }
>+#if defined(HAS_I210TOARGBROW_SSSE3)
>+  if (TestCpuFlag(kCpuHasSSSE3)) {
>+    I210ToARGBRow = I210ToARGBRow_Any_SSSE3;
>+    if (IS_ALIGNED(width, 8)) {
>+      I210ToARGBRow = I210ToARGBRow_SSSE3;
>+    }
>+  }
>+#endif
>+#if defined(HAS_I210TOARGBROW_AVX2)
>+  if (TestCpuFlag(kCpuHasAVX2)) {
>+    I210ToARGBRow = I210ToARGBRow_Any_AVX2;
>+    if (IS_ALIGNED(width, 16)) {
>+      I210ToARGBRow = I210ToARGBRow_AVX2;
>+    }
>+  }
>+#endif
>+  for (y = 0; y < height; ++y) {
>+    I210ToARGBRow(src_y, src_u, src_v, dst_argb, yuvconstants, width);
>+    dst_argb += dst_stride_argb;
>+    src_y += src_stride_y;
>+    if (y & 1) {
>+      src_u += src_stride_u;
>+      src_v += src_stride_v;
>+    }
>+  }
>+  return 0;
>+}
>+
>+// Convert I010 to ARGB.
>+LIBYUV_API
>+int I010ToARGB(const uint16_t* src_y,
>+               int src_stride_y,
>+               const uint16_t* src_u,
>+               int src_stride_u,
>+               const uint16_t* src_v,
>+               int src_stride_v,
>+               uint8_t* dst_argb,
>+               int dst_stride_argb,
>+               int width,
>+               int height) {
>+  return I010ToARGBMatrix(src_y, src_stride_y, src_u, src_stride_u, src_v,
>+                          src_stride_v, dst_argb, dst_stride_argb,
>+                          &kYuvI601Constants, width, height);
>+}
>+
>+// Convert I010 to ABGR.
>+LIBYUV_API
>+int I010ToABGR(const uint16_t* src_y,
>+               int src_stride_y,
>+               const uint16_t* src_u,
>+               int src_stride_u,
>+               const uint16_t* src_v,
>+               int src_stride_v,
>+               uint8_t* dst_abgr,
>+               int dst_stride_abgr,
>+               int width,
>+               int height) {
>+  return I010ToARGBMatrix(src_y, src_stride_y, src_v,
>+                          src_stride_v,  // Swap U and V
>+                          src_u, src_stride_u, dst_abgr, dst_stride_abgr,
>+                          &kYvuI601Constants,  // Use Yvu matrix
>+                          width, height);
>+}
>+
>+// Convert H010 to ARGB.
>+LIBYUV_API
>+int H010ToARGB(const uint16_t* src_y,
>+               int src_stride_y,
>+               const uint16_t* src_u,
>+               int src_stride_u,
>+               const uint16_t* src_v,
>+               int src_stride_v,
>+               uint8_t* dst_argb,
>+               int dst_stride_argb,
>+               int width,
>+               int height) {
>+  return I010ToARGBMatrix(src_y, src_stride_y, src_u, src_stride_u, src_v,
>+                          src_stride_v, dst_argb, dst_stride_argb,
>+                          &kYuvH709Constants, width, height);
>+}
>+
>+// Convert H010 to ABGR.
>+LIBYUV_API
>+int H010ToABGR(const uint16_t* src_y,
>+               int src_stride_y,
>+               const uint16_t* src_u,
>+               int src_stride_u,
>+               const uint16_t* src_v,
>+               int src_stride_v,
>+               uint8_t* dst_abgr,
>+               int dst_stride_abgr,
>+               int width,
>+               int height) {
>+  return I010ToARGBMatrix(src_y, src_stride_y, src_v,
>+                          src_stride_v,  // Swap U and V
>+                          src_u, src_stride_u, dst_abgr, dst_stride_abgr,
>+                          &kYvuH709Constants,  // Use Yvu matrix
>+                          width, height);
>+}
>+
> // Convert I444 to ARGB with matrix
>-static int I444ToARGBMatrix(const uint8* src_y,
>+static int I444ToARGBMatrix(const uint8_t* src_y,
>                             int src_stride_y,
>-                            const uint8* src_u,
>+                            const uint8_t* src_u,
>                             int src_stride_u,
>-                            const uint8* src_v,
>+                            const uint8_t* src_v,
>                             int src_stride_v,
>-                            uint8* dst_argb,
>+                            uint8_t* dst_argb,
>                             int dst_stride_argb,
>                             const struct YuvConstants* yuvconstants,
>                             int width,
>                             int height) {
>   int y;
>-  void (*I444ToARGBRow)(const uint8* y_buf, const uint8* u_buf,
>-                        const uint8* v_buf, uint8* rgb_buf,
>+  void (*I444ToARGBRow)(const uint8_t* y_buf, const uint8_t* u_buf,
>+                        const uint8_t* v_buf, uint8_t* rgb_buf,
>                         const struct YuvConstants* yuvconstants, int width) =
>       I444ToARGBRow_C;
>   if (!src_y || !src_u || !src_v || !dst_argb || width <= 0 || height == 0) {
>@@ -485,14 +717,6 @@ static int I444ToARGBMatrix(const uint8* src_y,
>     }
>   }
> #endif
>-#if defined(HAS_I444TOARGBROW_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2)) {
>-    I444ToARGBRow = I444ToARGBRow_Any_DSPR2;
>-    if (IS_ALIGNED(width, 8)) {
>-      I444ToARGBRow = I444ToARGBRow_DSPR2;
>-    }
>-  }
>-#endif
> #if defined(HAS_I444TOARGBROW_MSA)
>   if (TestCpuFlag(kCpuHasMSA)) {
>     I444ToARGBRow = I444ToARGBRow_Any_MSA;
>@@ -514,13 +738,13 @@ static int I444ToARGBMatrix(const uint8* src_y,
> 
> // Convert I444 to ARGB.
> LIBYUV_API
>-int I444ToARGB(const uint8* src_y,
>+int I444ToARGB(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_argb,
>+               uint8_t* dst_argb,
>                int dst_stride_argb,
>                int width,
>                int height) {
>@@ -531,13 +755,13 @@ int I444ToARGB(const uint8* src_y,
> 
> // Convert I444 to ABGR.
> LIBYUV_API
>-int I444ToABGR(const uint8* src_y,
>+int I444ToABGR(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_abgr,
>+               uint8_t* dst_abgr,
>                int dst_stride_abgr,
>                int width,
>                int height) {
>@@ -550,13 +774,13 @@ int I444ToABGR(const uint8* src_y,
> 
> // Convert J444 to ARGB.
> LIBYUV_API
>-int J444ToARGB(const uint8* src_y,
>+int J444ToARGB(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_argb,
>+               uint8_t* dst_argb,
>                int dst_stride_argb,
>                int width,
>                int height) {
>@@ -566,28 +790,28 @@ int J444ToARGB(const uint8* src_y,
> }
> 
> // Convert I420 with Alpha to preattenuated ARGB.
>-static int I420AlphaToARGBMatrix(const uint8* src_y,
>+static int I420AlphaToARGBMatrix(const uint8_t* src_y,
>                                  int src_stride_y,
>-                                 const uint8* src_u,
>+                                 const uint8_t* src_u,
>                                  int src_stride_u,
>-                                 const uint8* src_v,
>+                                 const uint8_t* src_v,
>                                  int src_stride_v,
>-                                 const uint8* src_a,
>+                                 const uint8_t* src_a,
>                                  int src_stride_a,
>-                                 uint8* dst_argb,
>+                                 uint8_t* dst_argb,
>                                  int dst_stride_argb,
>                                  const struct YuvConstants* yuvconstants,
>                                  int width,
>                                  int height,
>                                  int attenuate) {
>   int y;
>-  void (*I422AlphaToARGBRow)(const uint8* y_buf, const uint8* u_buf,
>-                             const uint8* v_buf, const uint8* a_buf,
>-                             uint8* dst_argb,
>+  void (*I422AlphaToARGBRow)(const uint8_t* y_buf, const uint8_t* u_buf,
>+                             const uint8_t* v_buf, const uint8_t* a_buf,
>+                             uint8_t* dst_argb,
>                              const struct YuvConstants* yuvconstants,
>                              int width) = I422AlphaToARGBRow_C;
>-  void (*ARGBAttenuateRow)(const uint8* src_argb, uint8* dst_argb, int width) =
>-      ARGBAttenuateRow_C;
>+  void (*ARGBAttenuateRow)(const uint8_t* src_argb, uint8_t* dst_argb,
>+                           int width) = ARGBAttenuateRow_C;
>   if (!src_y || !src_u || !src_v || !dst_argb || width <= 0 || height == 0) {
>     return -1;
>   }
>@@ -621,15 +845,6 @@ static int I420AlphaToARGBMatrix(const uint8* src_y,
>     }
>   }
> #endif
>-#if defined(HAS_I422ALPHATOARGBROW_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2) && IS_ALIGNED(width, 4) &&
>-      IS_ALIGNED(src_y, 4) && IS_ALIGNED(src_stride_y, 4) &&
>-      IS_ALIGNED(src_u, 2) && IS_ALIGNED(src_stride_u, 2) &&
>-      IS_ALIGNED(src_v, 2) && IS_ALIGNED(src_stride_v, 2) &&
>-      IS_ALIGNED(dst_argb, 4) && IS_ALIGNED(dst_stride_argb, 4)) {
>-    I422AlphaToARGBRow = I422AlphaToARGBRow_DSPR2;
>-  }
>-#endif
> #if defined(HAS_I422ALPHATOARGBROW_MSA)
>   if (TestCpuFlag(kCpuHasMSA)) {
>     I422AlphaToARGBRow = I422AlphaToARGBRow_Any_MSA;
>@@ -690,15 +905,15 @@ static int I420AlphaToARGBMatrix(const uint8* src_y,
> 
> // Convert I420 with Alpha to ARGB.
> LIBYUV_API
>-int I420AlphaToARGB(const uint8* src_y,
>+int I420AlphaToARGB(const uint8_t* src_y,
>                     int src_stride_y,
>-                    const uint8* src_u,
>+                    const uint8_t* src_u,
>                     int src_stride_u,
>-                    const uint8* src_v,
>+                    const uint8_t* src_v,
>                     int src_stride_v,
>-                    const uint8* src_a,
>+                    const uint8_t* src_a,
>                     int src_stride_a,
>-                    uint8* dst_argb,
>+                    uint8_t* dst_argb,
>                     int dst_stride_argb,
>                     int width,
>                     int height,
>@@ -711,15 +926,15 @@ int I420AlphaToARGB(const uint8* src_y,
> 
> // Convert I420 with Alpha to ABGR.
> LIBYUV_API
>-int I420AlphaToABGR(const uint8* src_y,
>+int I420AlphaToABGR(const uint8_t* src_y,
>                     int src_stride_y,
>-                    const uint8* src_u,
>+                    const uint8_t* src_u,
>                     int src_stride_u,
>-                    const uint8* src_v,
>+                    const uint8_t* src_v,
>                     int src_stride_v,
>-                    const uint8* src_a,
>+                    const uint8_t* src_a,
>                     int src_stride_a,
>-                    uint8* dst_abgr,
>+                    uint8_t* dst_abgr,
>                     int dst_stride_abgr,
>                     int width,
>                     int height,
>@@ -733,14 +948,14 @@ int I420AlphaToABGR(const uint8* src_y,
> 
> // Convert I400 to ARGB.
> LIBYUV_API
>-int I400ToARGB(const uint8* src_y,
>+int I400ToARGB(const uint8_t* src_y,
>                int src_stride_y,
>-               uint8* dst_argb,
>+               uint8_t* dst_argb,
>                int dst_stride_argb,
>                int width,
>                int height) {
>   int y;
>-  void (*I400ToARGBRow)(const uint8* y_buf, uint8* rgb_buf, int width) =
>+  void (*I400ToARGBRow)(const uint8_t* y_buf, uint8_t* rgb_buf, int width) =
>       I400ToARGBRow_C;
>   if (!src_y || !dst_argb || width <= 0 || height == 0) {
>     return -1;
>@@ -800,14 +1015,14 @@ int I400ToARGB(const uint8* src_y,
> 
> // Convert J400 to ARGB.
> LIBYUV_API
>-int J400ToARGB(const uint8* src_y,
>+int J400ToARGB(const uint8_t* src_y,
>                int src_stride_y,
>-               uint8* dst_argb,
>+               uint8_t* dst_argb,
>                int dst_stride_argb,
>                int width,
>                int height) {
>   int y;
>-  void (*J400ToARGBRow)(const uint8* src_y, uint8* dst_argb, int width) =
>+  void (*J400ToARGBRow)(const uint8_t* src_y, uint8_t* dst_argb, int width) =
>       J400ToARGBRow_C;
>   if (!src_y || !dst_argb || width <= 0 || height == 0) {
>     return -1;
>@@ -865,87 +1080,87 @@ int J400ToARGB(const uint8* src_y,
> }
> 
> // Shuffle table for converting BGRA to ARGB.
>-static uvec8 kShuffleMaskBGRAToARGB = {3u,  2u,  1u, 0u, 7u,  6u,  5u,  4u,
>-                                       11u, 10u, 9u, 8u, 15u, 14u, 13u, 12u};
>+static const uvec8 kShuffleMaskBGRAToARGB = {
>+    3u, 2u, 1u, 0u, 7u, 6u, 5u, 4u, 11u, 10u, 9u, 8u, 15u, 14u, 13u, 12u};
> 
> // Shuffle table for converting ABGR to ARGB.
>-static uvec8 kShuffleMaskABGRToARGB = {2u,  1u, 0u, 3u,  6u,  5u,  4u,  7u,
>-                                       10u, 9u, 8u, 11u, 14u, 13u, 12u, 15u};
>+static const uvec8 kShuffleMaskABGRToARGB = {
>+    2u, 1u, 0u, 3u, 6u, 5u, 4u, 7u, 10u, 9u, 8u, 11u, 14u, 13u, 12u, 15u};
> 
> // Shuffle table for converting RGBA to ARGB.
>-static uvec8 kShuffleMaskRGBAToARGB = {1u, 2u,  3u,  0u, 5u,  6u,  7u,  4u,
>-                                       9u, 10u, 11u, 8u, 13u, 14u, 15u, 12u};
>+static const uvec8 kShuffleMaskRGBAToARGB = {
>+    1u, 2u, 3u, 0u, 5u, 6u, 7u, 4u, 9u, 10u, 11u, 8u, 13u, 14u, 15u, 12u};
> 
> // Convert BGRA to ARGB.
> LIBYUV_API
>-int BGRAToARGB(const uint8* src_bgra,
>+int BGRAToARGB(const uint8_t* src_bgra,
>                int src_stride_bgra,
>-               uint8* dst_argb,
>+               uint8_t* dst_argb,
>                int dst_stride_argb,
>                int width,
>                int height) {
>   return ARGBShuffle(src_bgra, src_stride_bgra, dst_argb, dst_stride_argb,
>-                     (const uint8*)(&kShuffleMaskBGRAToARGB), width, height);
>+                     (const uint8_t*)(&kShuffleMaskBGRAToARGB), width, height);
> }
> 
> // Convert ARGB to BGRA (same as BGRAToARGB).
> LIBYUV_API
>-int ARGBToBGRA(const uint8* src_bgra,
>+int ARGBToBGRA(const uint8_t* src_bgra,
>                int src_stride_bgra,
>-               uint8* dst_argb,
>+               uint8_t* dst_argb,
>                int dst_stride_argb,
>                int width,
>                int height) {
>   return ARGBShuffle(src_bgra, src_stride_bgra, dst_argb, dst_stride_argb,
>-                     (const uint8*)(&kShuffleMaskBGRAToARGB), width, height);
>+                     (const uint8_t*)(&kShuffleMaskBGRAToARGB), width, height);
> }
> 
> // Convert ABGR to ARGB.
> LIBYUV_API
>-int ABGRToARGB(const uint8* src_abgr,
>+int ABGRToARGB(const uint8_t* src_abgr,
>                int src_stride_abgr,
>-               uint8* dst_argb,
>+               uint8_t* dst_argb,
>                int dst_stride_argb,
>                int width,
>                int height) {
>   return ARGBShuffle(src_abgr, src_stride_abgr, dst_argb, dst_stride_argb,
>-                     (const uint8*)(&kShuffleMaskABGRToARGB), width, height);
>+                     (const uint8_t*)(&kShuffleMaskABGRToARGB), width, height);
> }
> 
> // Convert ARGB to ABGR to (same as ABGRToARGB).
> LIBYUV_API
>-int ARGBToABGR(const uint8* src_abgr,
>+int ARGBToABGR(const uint8_t* src_abgr,
>                int src_stride_abgr,
>-               uint8* dst_argb,
>+               uint8_t* dst_argb,
>                int dst_stride_argb,
>                int width,
>                int height) {
>   return ARGBShuffle(src_abgr, src_stride_abgr, dst_argb, dst_stride_argb,
>-                     (const uint8*)(&kShuffleMaskABGRToARGB), width, height);
>+                     (const uint8_t*)(&kShuffleMaskABGRToARGB), width, height);
> }
> 
> // Convert RGBA to ARGB.
> LIBYUV_API
>-int RGBAToARGB(const uint8* src_rgba,
>+int RGBAToARGB(const uint8_t* src_rgba,
>                int src_stride_rgba,
>-               uint8* dst_argb,
>+               uint8_t* dst_argb,
>                int dst_stride_argb,
>                int width,
>                int height) {
>   return ARGBShuffle(src_rgba, src_stride_rgba, dst_argb, dst_stride_argb,
>-                     (const uint8*)(&kShuffleMaskRGBAToARGB), width, height);
>+                     (const uint8_t*)(&kShuffleMaskRGBAToARGB), width, height);
> }
> 
> // Convert RGB24 to ARGB.
> LIBYUV_API
>-int RGB24ToARGB(const uint8* src_rgb24,
>+int RGB24ToARGB(const uint8_t* src_rgb24,
>                 int src_stride_rgb24,
>-                uint8* dst_argb,
>+                uint8_t* dst_argb,
>                 int dst_stride_argb,
>                 int width,
>                 int height) {
>   int y;
>-  void (*RGB24ToARGBRow)(const uint8* src_rgb, uint8* dst_argb, int width) =
>+  void (*RGB24ToARGBRow)(const uint8_t* src_rgb, uint8_t* dst_argb, int width) =
>       RGB24ToARGBRow_C;
>   if (!src_rgb24 || !dst_argb || width <= 0 || height == 0) {
>     return -1;
>@@ -978,14 +1193,6 @@ int RGB24ToARGB(const uint8* src_rgb24,
>     }
>   }
> #endif
>-#if defined(HAS_RGB24TOARGBROW_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2)) {
>-    RGB24ToARGBRow = RGB24ToARGBRow_Any_DSPR2;
>-    if (IS_ALIGNED(width, 8)) {
>-      RGB24ToARGBRow = RGB24ToARGBRow_DSPR2;
>-    }
>-  }
>-#endif
> #if defined(HAS_RGB24TOARGBROW_MSA)
>   if (TestCpuFlag(kCpuHasMSA)) {
>     RGB24ToARGBRow = RGB24ToARGBRow_Any_MSA;
>@@ -1005,14 +1212,14 @@ int RGB24ToARGB(const uint8* src_rgb24,
> 
> // Convert RAW to ARGB.
> LIBYUV_API
>-int RAWToARGB(const uint8* src_raw,
>+int RAWToARGB(const uint8_t* src_raw,
>               int src_stride_raw,
>-              uint8* dst_argb,
>+              uint8_t* dst_argb,
>               int dst_stride_argb,
>               int width,
>               int height) {
>   int y;
>-  void (*RAWToARGBRow)(const uint8* src_rgb, uint8* dst_argb, int width) =
>+  void (*RAWToARGBRow)(const uint8_t* src_rgb, uint8_t* dst_argb, int width) =
>       RAWToARGBRow_C;
>   if (!src_raw || !dst_argb || width <= 0 || height == 0) {
>     return -1;
>@@ -1045,14 +1252,6 @@ int RAWToARGB(const uint8* src_raw,
>     }
>   }
> #endif
>-#if defined(HAS_RAWTOARGBROW_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2)) {
>-    RAWToARGBRow = RAWToARGBRow_Any_DSPR2;
>-    if (IS_ALIGNED(width, 8)) {
>-      RAWToARGBRow = RAWToARGBRow_DSPR2;
>-    }
>-  }
>-#endif
> #if defined(HAS_RAWTOARGBROW_MSA)
>   if (TestCpuFlag(kCpuHasMSA)) {
>     RAWToARGBRow = RAWToARGBRow_Any_MSA;
>@@ -1072,15 +1271,15 @@ int RAWToARGB(const uint8* src_raw,
> 
> // Convert RGB565 to ARGB.
> LIBYUV_API
>-int RGB565ToARGB(const uint8* src_rgb565,
>+int RGB565ToARGB(const uint8_t* src_rgb565,
>                  int src_stride_rgb565,
>-                 uint8* dst_argb,
>+                 uint8_t* dst_argb,
>                  int dst_stride_argb,
>                  int width,
>                  int height) {
>   int y;
>-  void (*RGB565ToARGBRow)(const uint8* src_rgb565, uint8* dst_argb, int width) =
>-      RGB565ToARGBRow_C;
>+  void (*RGB565ToARGBRow)(const uint8_t* src_rgb565, uint8_t* dst_argb,
>+                          int width) = RGB565ToARGBRow_C;
>   if (!src_rgb565 || !dst_argb || width <= 0 || height == 0) {
>     return -1;
>   }
>@@ -1120,14 +1319,6 @@ int RGB565ToARGB(const uint8* src_rgb565,
>     }
>   }
> #endif
>-#if defined(HAS_RGB565TOARGBROW_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2)) {
>-    RGB565ToARGBRow = RGB565ToARGBRow_Any_DSPR2;
>-    if (IS_ALIGNED(width, 8)) {
>-      RGB565ToARGBRow = RGB565ToARGBRow_DSPR2;
>-    }
>-  }
>-#endif
> #if defined(HAS_RGB565TOARGBROW_MSA)
>   if (TestCpuFlag(kCpuHasMSA)) {
>     RGB565ToARGBRow = RGB565ToARGBRow_Any_MSA;
>@@ -1147,14 +1338,14 @@ int RGB565ToARGB(const uint8* src_rgb565,
> 
> // Convert ARGB1555 to ARGB.
> LIBYUV_API
>-int ARGB1555ToARGB(const uint8* src_argb1555,
>+int ARGB1555ToARGB(const uint8_t* src_argb1555,
>                    int src_stride_argb1555,
>-                   uint8* dst_argb,
>+                   uint8_t* dst_argb,
>                    int dst_stride_argb,
>                    int width,
>                    int height) {
>   int y;
>-  void (*ARGB1555ToARGBRow)(const uint8* src_argb1555, uint8* dst_argb,
>+  void (*ARGB1555ToARGBRow)(const uint8_t* src_argb1555, uint8_t* dst_argb,
>                             int width) = ARGB1555ToARGBRow_C;
>   if (!src_argb1555 || !dst_argb || width <= 0 || height == 0) {
>     return -1;
>@@ -1195,14 +1386,6 @@ int ARGB1555ToARGB(const uint8* src_argb1555,
>     }
>   }
> #endif
>-#if defined(HAS_ARGB1555TOARGBROW_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2)) {
>-    ARGB1555ToARGBRow = ARGB1555ToARGBRow_Any_DSPR2;
>-    if (IS_ALIGNED(width, 4)) {
>-      ARGB1555ToARGBRow = ARGB1555ToARGBRow_DSPR2;
>-    }
>-  }
>-#endif
> #if defined(HAS_ARGB1555TOARGBROW_MSA)
>   if (TestCpuFlag(kCpuHasMSA)) {
>     ARGB1555ToARGBRow = ARGB1555ToARGBRow_Any_MSA;
>@@ -1222,14 +1405,14 @@ int ARGB1555ToARGB(const uint8* src_argb1555,
> 
> // Convert ARGB4444 to ARGB.
> LIBYUV_API
>-int ARGB4444ToARGB(const uint8* src_argb4444,
>+int ARGB4444ToARGB(const uint8_t* src_argb4444,
>                    int src_stride_argb4444,
>-                   uint8* dst_argb,
>+                   uint8_t* dst_argb,
>                    int dst_stride_argb,
>                    int width,
>                    int height) {
>   int y;
>-  void (*ARGB4444ToARGBRow)(const uint8* src_argb4444, uint8* dst_argb,
>+  void (*ARGB4444ToARGBRow)(const uint8_t* src_argb4444, uint8_t* dst_argb,
>                             int width) = ARGB4444ToARGBRow_C;
>   if (!src_argb4444 || !dst_argb || width <= 0 || height == 0) {
>     return -1;
>@@ -1270,14 +1453,6 @@ int ARGB4444ToARGB(const uint8* src_argb4444,
>     }
>   }
> #endif
>-#if defined(HAS_ARGB4444TOARGBROW_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2)) {
>-    ARGB4444ToARGBRow = ARGB4444ToARGBRow_Any_DSPR2;
>-    if (IS_ALIGNED(width, 4)) {
>-      ARGB4444ToARGBRow = ARGB4444ToARGBRow_DSPR2;
>-    }
>-  }
>-#endif
> #if defined(HAS_ARGB4444TOARGBROW_MSA)
>   if (TestCpuFlag(kCpuHasMSA)) {
>     ARGB4444ToARGBRow = ARGB4444ToARGBRow_Any_MSA;
>@@ -1295,20 +1470,116 @@ int ARGB4444ToARGB(const uint8* src_argb4444,
>   return 0;
> }
> 
>-// Convert NV12 to ARGB.
>+// Convert AR30 to ARGB.
> LIBYUV_API
>-int NV12ToARGB(const uint8* src_y,
>-               int src_stride_y,
>-               const uint8* src_uv,
>-               int src_stride_uv,
>-               uint8* dst_argb,
>+int AR30ToARGB(const uint8_t* src_ar30,
>+               int src_stride_ar30,
>+               uint8_t* dst_argb,
>                int dst_stride_argb,
>                int width,
>                int height) {
>   int y;
>-  void (*NV12ToARGBRow)(const uint8* y_buf, const uint8* uv_buf, uint8* rgb_buf,
>-                        const struct YuvConstants* yuvconstants, int width) =
>-      NV12ToARGBRow_C;
>+  if (!src_ar30 || !dst_argb || width <= 0 || height == 0) {
>+    return -1;
>+  }
>+  // Negative height means invert the image.
>+  if (height < 0) {
>+    height = -height;
>+    src_ar30 = src_ar30 + (height - 1) * src_stride_ar30;
>+    src_stride_ar30 = -src_stride_ar30;
>+  }
>+  // Coalesce rows.
>+  if (src_stride_ar30 == width * 4 && dst_stride_argb == width * 4) {
>+    width *= height;
>+    height = 1;
>+    src_stride_ar30 = dst_stride_argb = 0;
>+  }
>+  for (y = 0; y < height; ++y) {
>+    AR30ToARGBRow_C(src_ar30, dst_argb, width);
>+    src_ar30 += src_stride_ar30;
>+    dst_argb += dst_stride_argb;
>+  }
>+  return 0;
>+}
>+
>+// Convert AR30 to ABGR.
>+LIBYUV_API
>+int AR30ToABGR(const uint8_t* src_ar30,
>+               int src_stride_ar30,
>+               uint8_t* dst_abgr,
>+               int dst_stride_abgr,
>+               int width,
>+               int height) {
>+  int y;
>+  if (!src_ar30 || !dst_abgr || width <= 0 || height == 0) {
>+    return -1;
>+  }
>+  // Negative height means invert the image.
>+  if (height < 0) {
>+    height = -height;
>+    src_ar30 = src_ar30 + (height - 1) * src_stride_ar30;
>+    src_stride_ar30 = -src_stride_ar30;
>+  }
>+  // Coalesce rows.
>+  if (src_stride_ar30 == width * 4 && dst_stride_abgr == width * 4) {
>+    width *= height;
>+    height = 1;
>+    src_stride_ar30 = dst_stride_abgr = 0;
>+  }
>+  for (y = 0; y < height; ++y) {
>+    AR30ToABGRRow_C(src_ar30, dst_abgr, width);
>+    src_ar30 += src_stride_ar30;
>+    dst_abgr += dst_stride_abgr;
>+  }
>+  return 0;
>+}
>+
>+// Convert AR30 to AB30.
>+LIBYUV_API
>+int AR30ToAB30(const uint8_t* src_ar30,
>+               int src_stride_ar30,
>+               uint8_t* dst_ab30,
>+               int dst_stride_ab30,
>+               int width,
>+               int height) {
>+  int y;
>+  if (!src_ar30 || !dst_ab30 || width <= 0 || height == 0) {
>+    return -1;
>+  }
>+  // Negative height means invert the image.
>+  if (height < 0) {
>+    height = -height;
>+    src_ar30 = src_ar30 + (height - 1) * src_stride_ar30;
>+    src_stride_ar30 = -src_stride_ar30;
>+  }
>+  // Coalesce rows.
>+  if (src_stride_ar30 == width * 4 && dst_stride_ab30 == width * 4) {
>+    width *= height;
>+    height = 1;
>+    src_stride_ar30 = dst_stride_ab30 = 0;
>+  }
>+  for (y = 0; y < height; ++y) {
>+    AR30ToAB30Row_C(src_ar30, dst_ab30, width);
>+    src_ar30 += src_stride_ar30;
>+    dst_ab30 += dst_stride_ab30;
>+  }
>+  return 0;
>+}
>+
>+// Convert NV12 to ARGB with matrix
>+static int NV12ToARGBMatrix(const uint8_t* src_y,
>+                            int src_stride_y,
>+                            const uint8_t* src_uv,
>+                            int src_stride_uv,
>+                            uint8_t* dst_argb,
>+                            int dst_stride_argb,
>+                            const struct YuvConstants* yuvconstants,
>+                            int width,
>+                            int height) {
>+  int y;
>+  void (*NV12ToARGBRow)(
>+      const uint8_t* y_buf, const uint8_t* uv_buf, uint8_t* rgb_buf,
>+      const struct YuvConstants* yuvconstants, int width) = NV12ToARGBRow_C;
>   if (!src_y || !src_uv || !dst_argb || width <= 0 || height == 0) {
>     return -1;
>   }
>@@ -1342,14 +1613,6 @@ int NV12ToARGB(const uint8* src_y,
>     }
>   }
> #endif
>-#if defined(HAS_NV12TOARGBROW_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2)) {
>-    NV12ToARGBRow = NV12ToARGBRow_Any_DSPR2;
>-    if (IS_ALIGNED(width, 8)) {
>-      NV12ToARGBRow = NV12ToARGBRow_DSPR2;
>-    }
>-  }
>-#endif
> #if defined(HAS_NV12TOARGBROW_MSA)
>   if (TestCpuFlag(kCpuHasMSA)) {
>     NV12ToARGBRow = NV12ToARGBRow_Any_MSA;
>@@ -1360,7 +1623,7 @@ int NV12ToARGB(const uint8* src_y,
> #endif
> 
>   for (y = 0; y < height; ++y) {
>-    NV12ToARGBRow(src_y, src_uv, dst_argb, &kYuvI601Constants, width);
>+    NV12ToARGBRow(src_y, src_uv, dst_argb, yuvconstants, width);
>     dst_argb += dst_stride_argb;
>     src_y += src_stride_y;
>     if (y & 1) {
>@@ -1370,21 +1633,21 @@ int NV12ToARGB(const uint8* src_y,
>   return 0;
> }
> 
>-// Convert NV21 to ARGB.
>-LIBYUV_API
>-int NV21ToARGB(const uint8* src_y,
>-               int src_stride_y,
>-               const uint8* src_uv,
>-               int src_stride_uv,
>-               uint8* dst_argb,
>-               int dst_stride_argb,
>-               int width,
>-               int height) {
>+// Convert NV21 to ARGB with matrix
>+static int NV21ToARGBMatrix(const uint8_t* src_y,
>+                            int src_stride_y,
>+                            const uint8_t* src_vu,
>+                            int src_stride_vu,
>+                            uint8_t* dst_argb,
>+                            int dst_stride_argb,
>+                            const struct YuvConstants* yuvconstants,
>+                            int width,
>+                            int height) {
>   int y;
>-  void (*NV21ToARGBRow)(const uint8* y_buf, const uint8* uv_buf, uint8* rgb_buf,
>-                        const struct YuvConstants* yuvconstants, int width) =
>-      NV21ToARGBRow_C;
>-  if (!src_y || !src_uv || !dst_argb || width <= 0 || height == 0) {
>+  void (*NV21ToARGBRow)(
>+      const uint8_t* y_buf, const uint8_t* uv_buf, uint8_t* rgb_buf,
>+      const struct YuvConstants* yuvconstants, int width) = NV21ToARGBRow_C;
>+  if (!src_y || !src_vu || !dst_argb || width <= 0 || height == 0) {
>     return -1;
>   }
>   // Negative height means invert the image.
>@@ -1427,9 +1690,110 @@ int NV21ToARGB(const uint8* src_y,
> #endif
> 
>   for (y = 0; y < height; ++y) {
>-    NV21ToARGBRow(src_y, src_uv, dst_argb, &kYuvI601Constants, width);
>+    NV21ToARGBRow(src_y, src_vu, dst_argb, yuvconstants, width);
>     dst_argb += dst_stride_argb;
>     src_y += src_stride_y;
>+    if (y & 1) {
>+      src_vu += src_stride_vu;
>+    }
>+  }
>+  return 0;
>+}
>+
>+// Convert NV12 to ARGB.
>+LIBYUV_API
>+int NV12ToARGB(const uint8_t* src_y,
>+               int src_stride_y,
>+               const uint8_t* src_uv,
>+               int src_stride_uv,
>+               uint8_t* dst_argb,
>+               int dst_stride_argb,
>+               int width,
>+               int height) {
>+  return NV12ToARGBMatrix(src_y, src_stride_y, src_uv, src_stride_uv, dst_argb,
>+                          dst_stride_argb, &kYuvI601Constants, width, height);
>+}
>+
>+// Convert NV21 to ARGB.
>+LIBYUV_API
>+int NV21ToARGB(const uint8_t* src_y,
>+               int src_stride_y,
>+               const uint8_t* src_vu,
>+               int src_stride_vu,
>+               uint8_t* dst_argb,
>+               int dst_stride_argb,
>+               int width,
>+               int height) {
>+  return NV21ToARGBMatrix(src_y, src_stride_y, src_vu, src_stride_vu, dst_argb,
>+                          dst_stride_argb, &kYuvI601Constants, width, height);
>+}
>+
>+// Convert NV12 to ABGR.
>+// To output ABGR instead of ARGB swap the UV and use a mirrrored yuc matrix.
>+// To swap the UV use NV12 instead of NV21.LIBYUV_API
>+int NV12ToABGR(const uint8_t* src_y,
>+               int src_stride_y,
>+               const uint8_t* src_uv,
>+               int src_stride_uv,
>+               uint8_t* dst_abgr,
>+               int dst_stride_abgr,
>+               int width,
>+               int height) {
>+  return NV21ToARGBMatrix(src_y, src_stride_y, src_uv, src_stride_uv, dst_abgr,
>+                          dst_stride_abgr, &kYvuI601Constants, width, height);
>+}
>+
>+// Convert NV21 to ABGR.
>+LIBYUV_API
>+int NV21ToABGR(const uint8_t* src_y,
>+               int src_stride_y,
>+               const uint8_t* src_vu,
>+               int src_stride_vu,
>+               uint8_t* dst_abgr,
>+               int dst_stride_abgr,
>+               int width,
>+               int height) {
>+  return NV12ToARGBMatrix(src_y, src_stride_y, src_vu, src_stride_vu, dst_abgr,
>+                          dst_stride_abgr, &kYvuI601Constants, width, height);
>+}
>+
>+// TODO(fbarchard): Consider SSSE3 2 step conversion.
>+// Convert NV12 to RGB24 with matrix
>+static int NV12ToRGB24Matrix(const uint8_t* src_y,
>+                             int src_stride_y,
>+                             const uint8_t* src_uv,
>+                             int src_stride_uv,
>+                             uint8_t* dst_rgb24,
>+                             int dst_stride_rgb24,
>+                             const struct YuvConstants* yuvconstants,
>+                             int width,
>+                             int height) {
>+  int y;
>+  void (*NV12ToRGB24Row)(
>+      const uint8_t* y_buf, const uint8_t* uv_buf, uint8_t* rgb_buf,
>+      const struct YuvConstants* yuvconstants, int width) = NV12ToRGB24Row_C;
>+  if (!src_y || !src_uv || !dst_rgb24 || width <= 0 || height == 0) {
>+    return -1;
>+  }
>+  // Negative height means invert the image.
>+  if (height < 0) {
>+    height = -height;
>+    dst_rgb24 = dst_rgb24 + (height - 1) * dst_stride_rgb24;
>+    dst_stride_rgb24 = -dst_stride_rgb24;
>+  }
>+#if defined(HAS_NV12TORGB24ROW_NEON)
>+  if (TestCpuFlag(kCpuHasNEON)) {
>+    NV12ToRGB24Row = NV12ToRGB24Row_Any_NEON;
>+    if (IS_ALIGNED(width, 8)) {
>+      NV12ToRGB24Row = NV12ToRGB24Row_NEON;
>+    }
>+  }
>+#endif
>+
>+  for (y = 0; y < height; ++y) {
>+    NV12ToRGB24Row(src_y, src_uv, dst_rgb24, yuvconstants, width);
>+    dst_rgb24 += dst_stride_rgb24;
>+    src_y += src_stride_y;
>     if (y & 1) {
>       src_uv += src_stride_uv;
>     }
>@@ -1437,18 +1801,92 @@ int NV21ToARGB(const uint8* src_y,
>   return 0;
> }
> 
>+// Convert NV21 to RGB24 with matrix
>+static int NV21ToRGB24Matrix(const uint8_t* src_y,
>+                             int src_stride_y,
>+                             const uint8_t* src_vu,
>+                             int src_stride_vu,
>+                             uint8_t* dst_rgb24,
>+                             int dst_stride_rgb24,
>+                             const struct YuvConstants* yuvconstants,
>+                             int width,
>+                             int height) {
>+  int y;
>+  void (*NV21ToRGB24Row)(
>+      const uint8_t* y_buf, const uint8_t* uv_buf, uint8_t* rgb_buf,
>+      const struct YuvConstants* yuvconstants, int width) = NV21ToRGB24Row_C;
>+  if (!src_y || !src_vu || !dst_rgb24 || width <= 0 || height == 0) {
>+    return -1;
>+  }
>+  // Negative height means invert the image.
>+  if (height < 0) {
>+    height = -height;
>+    dst_rgb24 = dst_rgb24 + (height - 1) * dst_stride_rgb24;
>+    dst_stride_rgb24 = -dst_stride_rgb24;
>+  }
>+#if defined(HAS_NV21TORGB24ROW_NEON)
>+  if (TestCpuFlag(kCpuHasNEON)) {
>+    NV21ToRGB24Row = NV21ToRGB24Row_Any_NEON;
>+    if (IS_ALIGNED(width, 8)) {
>+      NV21ToRGB24Row = NV21ToRGB24Row_NEON;
>+    }
>+  }
>+#endif
>+
>+  for (y = 0; y < height; ++y) {
>+    NV21ToRGB24Row(src_y, src_vu, dst_rgb24, yuvconstants, width);
>+    dst_rgb24 += dst_stride_rgb24;
>+    src_y += src_stride_y;
>+    if (y & 1) {
>+      src_vu += src_stride_vu;
>+    }
>+  }
>+  return 0;
>+}
>+
>+// TODO(fbarchard): \(fbarchard): NV12ToRAW can be implemented by mirrored
>+// matrix. Convert NV12 to RGB24.
>+LIBYUV_API
>+int NV12ToRGB24(const uint8_t* src_y,
>+                int src_stride_y,
>+                const uint8_t* src_uv,
>+                int src_stride_uv,
>+                uint8_t* dst_rgb24,
>+                int dst_stride_rgb24,
>+                int width,
>+                int height) {
>+  return NV12ToRGB24Matrix(src_y, src_stride_y, src_uv, src_stride_uv,
>+                           dst_rgb24, dst_stride_rgb24, &kYuvI601Constants,
>+                           width, height);
>+}
>+
>+// Convert NV21 to RGB24.
>+LIBYUV_API
>+int NV21ToRGB24(const uint8_t* src_y,
>+                int src_stride_y,
>+                const uint8_t* src_vu,
>+                int src_stride_vu,
>+                uint8_t* dst_rgb24,
>+                int dst_stride_rgb24,
>+                int width,
>+                int height) {
>+  return NV21ToRGB24Matrix(src_y, src_stride_y, src_vu, src_stride_vu,
>+                           dst_rgb24, dst_stride_rgb24, &kYuvI601Constants,
>+                           width, height);
>+}
>+
> // Convert M420 to ARGB.
> LIBYUV_API
>-int M420ToARGB(const uint8* src_m420,
>+int M420ToARGB(const uint8_t* src_m420,
>                int src_stride_m420,
>-               uint8* dst_argb,
>+               uint8_t* dst_argb,
>                int dst_stride_argb,
>                int width,
>                int height) {
>   int y;
>-  void (*NV12ToARGBRow)(const uint8* y_buf, const uint8* uv_buf, uint8* rgb_buf,
>-                        const struct YuvConstants* yuvconstants, int width) =
>-      NV12ToARGBRow_C;
>+  void (*NV12ToARGBRow)(
>+      const uint8_t* y_buf, const uint8_t* uv_buf, uint8_t* rgb_buf,
>+      const struct YuvConstants* yuvconstants, int width) = NV12ToARGBRow_C;
>   if (!src_m420 || !dst_argb || width <= 0 || height == 0) {
>     return -1;
>   }
>@@ -1482,14 +1920,6 @@ int M420ToARGB(const uint8* src_m420,
>     }
>   }
> #endif
>-#if defined(HAS_NV12TOARGBROW_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2)) {
>-    NV12ToARGBRow = NV12ToARGBRow_Any_DSPR2;
>-    if (IS_ALIGNED(width, 8)) {
>-      NV12ToARGBRow = NV12ToARGBRow_DSPR2;
>-    }
>-  }
>-#endif
> #if defined(HAS_NV12TOARGBROW_MSA)
>   if (TestCpuFlag(kCpuHasMSA)) {
>     NV12ToARGBRow = NV12ToARGBRow_Any_MSA;
>@@ -1516,14 +1946,14 @@ int M420ToARGB(const uint8* src_m420,
> 
> // Convert YUY2 to ARGB.
> LIBYUV_API
>-int YUY2ToARGB(const uint8* src_yuy2,
>+int YUY2ToARGB(const uint8_t* src_yuy2,
>                int src_stride_yuy2,
>-               uint8* dst_argb,
>+               uint8_t* dst_argb,
>                int dst_stride_argb,
>                int width,
>                int height) {
>   int y;
>-  void (*YUY2ToARGBRow)(const uint8* src_yuy2, uint8* dst_argb,
>+  void (*YUY2ToARGBRow)(const uint8_t* src_yuy2, uint8_t* dst_argb,
>                         const struct YuvConstants* yuvconstants, int width) =
>       YUY2ToARGBRow_C;
>   if (!src_yuy2 || !dst_argb || width <= 0 || height == 0) {
>@@ -1583,14 +2013,14 @@ int YUY2ToARGB(const uint8* src_yuy2,
> 
> // Convert UYVY to ARGB.
> LIBYUV_API
>-int UYVYToARGB(const uint8* src_uyvy,
>+int UYVYToARGB(const uint8_t* src_uyvy,
>                int src_stride_uyvy,
>-               uint8* dst_argb,
>+               uint8_t* dst_argb,
>                int dst_stride_argb,
>                int width,
>                int height) {
>   int y;
>-  void (*UYVYToARGBRow)(const uint8* src_uyvy, uint8* dst_argb,
>+  void (*UYVYToARGBRow)(const uint8_t* src_uyvy, uint8_t* dst_argb,
>                         const struct YuvConstants* yuvconstants, int width) =
>       UYVYToARGBRow_C;
>   if (!src_uyvy || !dst_argb || width <= 0 || height == 0) {
>@@ -1647,6 +2077,121 @@ int UYVYToARGB(const uint8* src_uyvy,
>   }
>   return 0;
> }
>+static void WeavePixels(const uint8_t* src_u,
>+                        const uint8_t* src_v,
>+                        int src_pixel_stride_uv,
>+                        uint8_t* dst_uv,
>+                        int width) {
>+  int i;
>+  for (i = 0; i < width; ++i) {
>+    dst_uv[0] = *src_u;
>+    dst_uv[1] = *src_v;
>+    dst_uv += 2;
>+    src_u += src_pixel_stride_uv;
>+    src_v += src_pixel_stride_uv;
>+  }
>+}
>+
>+// Convert Android420 to ARGB.
>+LIBYUV_API
>+int Android420ToARGBMatrix(const uint8_t* src_y,
>+                           int src_stride_y,
>+                           const uint8_t* src_u,
>+                           int src_stride_u,
>+                           const uint8_t* src_v,
>+                           int src_stride_v,
>+                           int src_pixel_stride_uv,
>+                           uint8_t* dst_argb,
>+                           int dst_stride_argb,
>+                           const struct YuvConstants* yuvconstants,
>+                           int width,
>+                           int height) {
>+  int y;
>+  uint8_t* dst_uv;
>+  const ptrdiff_t vu_off = src_v - src_u;
>+  int halfwidth = (width + 1) >> 1;
>+  int halfheight = (height + 1) >> 1;
>+  if (!src_y || !src_u || !src_v || !dst_argb || width <= 0 || height == 0) {
>+    return -1;
>+  }
>+  // Negative height means invert the image.
>+  if (height < 0) {
>+    height = -height;
>+    halfheight = (height + 1) >> 1;
>+    dst_argb = dst_argb + (height - 1) * dst_stride_argb;
>+    dst_stride_argb = -dst_stride_argb;
>+  }
>+
>+  // I420
>+  if (src_pixel_stride_uv == 1) {
>+    return I420ToARGBMatrix(src_y, src_stride_y, src_u, src_stride_u, src_v,
>+                            src_stride_v, dst_argb, dst_stride_argb,
>+                            yuvconstants, width, height);
>+    // NV21
>+  }
>+  if (src_pixel_stride_uv == 2 && vu_off == -1 &&
>+      src_stride_u == src_stride_v) {
>+    return NV21ToARGBMatrix(src_y, src_stride_y, src_v, src_stride_v, dst_argb,
>+                            dst_stride_argb, yuvconstants, width, height);
>+    // NV12
>+  }
>+  if (src_pixel_stride_uv == 2 && vu_off == 1 && src_stride_u == src_stride_v) {
>+    return NV12ToARGBMatrix(src_y, src_stride_y, src_u, src_stride_u, dst_argb,
>+                            dst_stride_argb, yuvconstants, width, height);
>+  }
>+
>+  // General case fallback creates NV12
>+  align_buffer_64(plane_uv, halfwidth * 2 * halfheight);
>+  dst_uv = plane_uv;
>+  for (y = 0; y < halfheight; ++y) {
>+    WeavePixels(src_u, src_v, src_pixel_stride_uv, dst_uv, halfwidth);
>+    src_u += src_stride_u;
>+    src_v += src_stride_v;
>+    dst_uv += halfwidth * 2;
>+  }
>+  NV12ToARGBMatrix(src_y, src_stride_y, plane_uv, halfwidth * 2, dst_argb,
>+                   dst_stride_argb, yuvconstants, width, height);
>+  free_aligned_buffer_64(plane_uv);
>+  return 0;
>+}
>+
>+// Convert Android420 to ARGB.
>+LIBYUV_API
>+int Android420ToARGB(const uint8_t* src_y,
>+                     int src_stride_y,
>+                     const uint8_t* src_u,
>+                     int src_stride_u,
>+                     const uint8_t* src_v,
>+                     int src_stride_v,
>+                     int src_pixel_stride_uv,
>+                     uint8_t* dst_argb,
>+                     int dst_stride_argb,
>+                     int width,
>+                     int height) {
>+  return Android420ToARGBMatrix(src_y, src_stride_y, src_u, src_stride_u, src_v,
>+                                src_stride_v, src_pixel_stride_uv, dst_argb,
>+                                dst_stride_argb, &kYuvI601Constants, width,
>+                                height);
>+}
>+
>+// Convert Android420 to ABGR.
>+LIBYUV_API
>+int Android420ToABGR(const uint8_t* src_y,
>+                     int src_stride_y,
>+                     const uint8_t* src_u,
>+                     int src_stride_u,
>+                     const uint8_t* src_v,
>+                     int src_stride_v,
>+                     int src_pixel_stride_uv,
>+                     uint8_t* dst_abgr,
>+                     int dst_stride_abgr,
>+                     int width,
>+                     int height) {
>+  return Android420ToARGBMatrix(src_y, src_stride_y, src_v, src_stride_v, src_u,
>+                                src_stride_u, src_pixel_stride_uv, dst_abgr,
>+                                dst_stride_abgr, &kYvuI601Constants, width,
>+                                height);
>+}
> 
> #ifdef __cplusplus
> }  // extern "C"
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/convert_from.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/convert_from.cc
>index 0f52f9ef9e0..b5587ced625 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/convert_from.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/convert_from.cc
>@@ -30,17 +30,17 @@ static __inline int Abs(int v) {
> }
> 
> // I420 To any I4xx YUV format with mirroring.
>-static int I420ToI4xx(const uint8* src_y,
>+static int I420ToI4xx(const uint8_t* src_y,
>                       int src_stride_y,
>-                      const uint8* src_u,
>+                      const uint8_t* src_u,
>                       int src_stride_u,
>-                      const uint8* src_v,
>+                      const uint8_t* src_v,
>                       int src_stride_v,
>-                      uint8* dst_y,
>+                      uint8_t* dst_y,
>                       int dst_stride_y,
>-                      uint8* dst_u,
>+                      uint8_t* dst_u,
>                       int dst_stride_u,
>-                      uint8* dst_v,
>+                      uint8_t* dst_v,
>                       int dst_stride_v,
>                       int src_y_width,
>                       int src_y_height,
>@@ -65,20 +65,64 @@ static int I420ToI4xx(const uint8* src_y,
>   return 0;
> }
> 
>+// Convert 8 bit YUV to 10 bit.
>+LIBYUV_API
>+int I420ToI010(const uint8_t* src_y,
>+               int src_stride_y,
>+               const uint8_t* src_u,
>+               int src_stride_u,
>+               const uint8_t* src_v,
>+               int src_stride_v,
>+               uint16_t* dst_y,
>+               int dst_stride_y,
>+               uint16_t* dst_u,
>+               int dst_stride_u,
>+               uint16_t* dst_v,
>+               int dst_stride_v,
>+               int width,
>+               int height) {
>+  int halfwidth = (width + 1) >> 1;
>+  int halfheight = (height + 1) >> 1;
>+  if (!src_u || !src_v || !dst_u || !dst_v || width <= 0 || height == 0) {
>+    return -1;
>+  }
>+  // Negative height means invert the image.
>+  if (height < 0) {
>+    height = -height;
>+    halfheight = (height + 1) >> 1;
>+    src_y = src_y + (height - 1) * src_stride_y;
>+    src_u = src_u + (halfheight - 1) * src_stride_u;
>+    src_v = src_v + (halfheight - 1) * src_stride_v;
>+    src_stride_y = -src_stride_y;
>+    src_stride_u = -src_stride_u;
>+    src_stride_v = -src_stride_v;
>+  }
>+
>+  // Convert Y plane.
>+  Convert8To16Plane(src_y, src_stride_y, dst_y, dst_stride_y, 1024, width,
>+                    height);
>+  // Convert UV planes.
>+  Convert8To16Plane(src_u, src_stride_u, dst_u, dst_stride_u, 1024, halfwidth,
>+                    halfheight);
>+  Convert8To16Plane(src_v, src_stride_v, dst_v, dst_stride_v, 1024, halfwidth,
>+                    halfheight);
>+  return 0;
>+}
>+
> // 420 chroma is 1/2 width, 1/2 height
> // 422 chroma is 1/2 width, 1x height
> LIBYUV_API
>-int I420ToI422(const uint8* src_y,
>+int I420ToI422(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_u,
>+               uint8_t* dst_u,
>                int dst_stride_u,
>-               uint8* dst_v,
>+               uint8_t* dst_v,
>                int dst_stride_v,
>                int width,
>                int height) {
>@@ -93,17 +137,17 @@ int I420ToI422(const uint8* src_y,
> // 420 chroma is 1/2 width, 1/2 height
> // 444 chroma is 1x width, 1x height
> LIBYUV_API
>-int I420ToI444(const uint8* src_y,
>+int I420ToI444(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_u,
>+               uint8_t* dst_u,
>                int dst_stride_u,
>-               uint8* dst_v,
>+               uint8_t* dst_v,
>                int dst_stride_v,
>                int width,
>                int height) {
>@@ -117,9 +161,9 @@ int I420ToI444(const uint8* src_y,
> 
> // Copy to I400. Source can be I420,422,444,400,NV12,NV21
> LIBYUV_API
>-int I400Copy(const uint8* src_y,
>+int I400Copy(const uint8_t* src_y,
>              int src_stride_y,
>-             uint8* dst_y,
>+             uint8_t* dst_y,
>              int dst_stride_y,
>              int width,
>              int height) {
>@@ -137,19 +181,19 @@ int I400Copy(const uint8* src_y,
> }
> 
> LIBYUV_API
>-int I422ToYUY2(const uint8* src_y,
>+int I422ToYUY2(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_yuy2,
>+               uint8_t* dst_yuy2,
>                int dst_stride_yuy2,
>                int width,
>                int height) {
>   int y;
>-  void (*I422ToYUY2Row)(const uint8* src_y, const uint8* src_u,
>-                        const uint8* src_v, uint8* dst_yuy2, int width) =
>+  void (*I422ToYUY2Row)(const uint8_t* src_y, const uint8_t* src_u,
>+                        const uint8_t* src_v, uint8_t* dst_yuy2, int width) =
>       I422ToYUY2Row_C;
>   if (!src_y || !src_u || !src_v || !dst_yuy2 || width <= 0 || height == 0) {
>     return -1;
>@@ -175,6 +219,14 @@ int I422ToYUY2(const uint8* src_y,
>     }
>   }
> #endif
>+#if defined(HAS_I422TOYUY2ROW_AVX2)
>+  if (TestCpuFlag(kCpuHasAVX2)) {
>+    I422ToYUY2Row = I422ToYUY2Row_Any_AVX2;
>+    if (IS_ALIGNED(width, 32)) {
>+      I422ToYUY2Row = I422ToYUY2Row_AVX2;
>+    }
>+  }
>+#endif
> #if defined(HAS_I422TOYUY2ROW_NEON)
>   if (TestCpuFlag(kCpuHasNEON)) {
>     I422ToYUY2Row = I422ToYUY2Row_Any_NEON;
>@@ -195,19 +247,19 @@ int I422ToYUY2(const uint8* src_y,
> }
> 
> LIBYUV_API
>-int I420ToYUY2(const uint8* src_y,
>+int I420ToYUY2(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_yuy2,
>+               uint8_t* dst_yuy2,
>                int dst_stride_yuy2,
>                int width,
>                int height) {
>   int y;
>-  void (*I422ToYUY2Row)(const uint8* src_y, const uint8* src_u,
>-                        const uint8* src_v, uint8* dst_yuy2, int width) =
>+  void (*I422ToYUY2Row)(const uint8_t* src_y, const uint8_t* src_u,
>+                        const uint8_t* src_v, uint8_t* dst_yuy2, int width) =
>       I422ToYUY2Row_C;
>   if (!src_y || !src_u || !src_v || !dst_yuy2 || width <= 0 || height == 0) {
>     return -1;
>@@ -226,6 +278,14 @@ int I420ToYUY2(const uint8* src_y,
>     }
>   }
> #endif
>+#if defined(HAS_I422TOYUY2ROW_AVX2)
>+  if (TestCpuFlag(kCpuHasAVX2)) {
>+    I422ToYUY2Row = I422ToYUY2Row_Any_AVX2;
>+    if (IS_ALIGNED(width, 32)) {
>+      I422ToYUY2Row = I422ToYUY2Row_AVX2;
>+    }
>+  }
>+#endif
> #if defined(HAS_I422TOYUY2ROW_NEON)
>   if (TestCpuFlag(kCpuHasNEON)) {
>     I422ToYUY2Row = I422ToYUY2Row_Any_NEON;
>@@ -259,19 +319,19 @@ int I420ToYUY2(const uint8* src_y,
> }
> 
> LIBYUV_API
>-int I422ToUYVY(const uint8* src_y,
>+int I422ToUYVY(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_uyvy,
>+               uint8_t* dst_uyvy,
>                int dst_stride_uyvy,
>                int width,
>                int height) {
>   int y;
>-  void (*I422ToUYVYRow)(const uint8* src_y, const uint8* src_u,
>-                        const uint8* src_v, uint8* dst_uyvy, int width) =
>+  void (*I422ToUYVYRow)(const uint8_t* src_y, const uint8_t* src_u,
>+                        const uint8_t* src_v, uint8_t* dst_uyvy, int width) =
>       I422ToUYVYRow_C;
>   if (!src_y || !src_u || !src_v || !dst_uyvy || width <= 0 || height == 0) {
>     return -1;
>@@ -297,6 +357,14 @@ int I422ToUYVY(const uint8* src_y,
>     }
>   }
> #endif
>+#if defined(HAS_I422TOUYVYROW_AVX2)
>+  if (TestCpuFlag(kCpuHasAVX2)) {
>+    I422ToUYVYRow = I422ToUYVYRow_Any_AVX2;
>+    if (IS_ALIGNED(width, 32)) {
>+      I422ToUYVYRow = I422ToUYVYRow_AVX2;
>+    }
>+  }
>+#endif
> #if defined(HAS_I422TOUYVYROW_NEON)
>   if (TestCpuFlag(kCpuHasNEON)) {
>     I422ToUYVYRow = I422ToUYVYRow_Any_NEON;
>@@ -325,19 +393,19 @@ int I422ToUYVY(const uint8* src_y,
> }
> 
> LIBYUV_API
>-int I420ToUYVY(const uint8* src_y,
>+int I420ToUYVY(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_uyvy,
>+               uint8_t* dst_uyvy,
>                int dst_stride_uyvy,
>                int width,
>                int height) {
>   int y;
>-  void (*I422ToUYVYRow)(const uint8* src_y, const uint8* src_u,
>-                        const uint8* src_v, uint8* dst_uyvy, int width) =
>+  void (*I422ToUYVYRow)(const uint8_t* src_y, const uint8_t* src_u,
>+                        const uint8_t* src_v, uint8_t* dst_uyvy, int width) =
>       I422ToUYVYRow_C;
>   if (!src_y || !src_u || !src_v || !dst_uyvy || width <= 0 || height == 0) {
>     return -1;
>@@ -356,6 +424,14 @@ int I420ToUYVY(const uint8* src_y,
>     }
>   }
> #endif
>+#if defined(HAS_I422TOUYVYROW_AVX2)
>+  if (TestCpuFlag(kCpuHasAVX2)) {
>+    I422ToUYVYRow = I422ToUYVYRow_Any_AVX2;
>+    if (IS_ALIGNED(width, 32)) {
>+      I422ToUYVYRow = I422ToUYVYRow_AVX2;
>+    }
>+  }
>+#endif
> #if defined(HAS_I422TOUYVYROW_NEON)
>   if (TestCpuFlag(kCpuHasNEON)) {
>     I422ToUYVYRow = I422ToUYVYRow_Any_NEON;
>@@ -390,15 +466,15 @@ int I420ToUYVY(const uint8* src_y,
> 
> // TODO(fbarchard): test negative height for invert.
> LIBYUV_API
>-int I420ToNV12(const uint8* src_y,
>+int I420ToNV12(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_uv,
>+               uint8_t* dst_uv,
>                int dst_stride_uv,
>                int width,
>                int height) {
>@@ -417,15 +493,15 @@ int I420ToNV12(const uint8* src_y,
> }
> 
> LIBYUV_API
>-int I420ToNV21(const uint8* src_y,
>+int I420ToNV21(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_vu,
>+               uint8_t* dst_vu,
>                int dst_stride_vu,
>                int width,
>                int height) {
>@@ -435,20 +511,20 @@ int I420ToNV21(const uint8* src_y,
> }
> 
> // Convert I422 to RGBA with matrix
>-static int I420ToRGBAMatrix(const uint8* src_y,
>+static int I420ToRGBAMatrix(const uint8_t* src_y,
>                             int src_stride_y,
>-                            const uint8* src_u,
>+                            const uint8_t* src_u,
>                             int src_stride_u,
>-                            const uint8* src_v,
>+                            const uint8_t* src_v,
>                             int src_stride_v,
>-                            uint8* dst_rgba,
>+                            uint8_t* dst_rgba,
>                             int dst_stride_rgba,
>                             const struct YuvConstants* yuvconstants,
>                             int width,
>                             int height) {
>   int y;
>-  void (*I422ToRGBARow)(const uint8* y_buf, const uint8* u_buf,
>-                        const uint8* v_buf, uint8* rgb_buf,
>+  void (*I422ToRGBARow)(const uint8_t* y_buf, const uint8_t* u_buf,
>+                        const uint8_t* v_buf, uint8_t* rgb_buf,
>                         const struct YuvConstants* yuvconstants, int width) =
>       I422ToRGBARow_C;
>   if (!src_y || !src_u || !src_v || !dst_rgba || width <= 0 || height == 0) {
>@@ -484,15 +560,6 @@ static int I420ToRGBAMatrix(const uint8* src_y,
>     }
>   }
> #endif
>-#if defined(HAS_I422TORGBAROW_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2) && IS_ALIGNED(width, 4) &&
>-      IS_ALIGNED(src_y, 4) && IS_ALIGNED(src_stride_y, 4) &&
>-      IS_ALIGNED(src_u, 2) && IS_ALIGNED(src_stride_u, 2) &&
>-      IS_ALIGNED(src_v, 2) && IS_ALIGNED(src_stride_v, 2) &&
>-      IS_ALIGNED(dst_rgba, 4) && IS_ALIGNED(dst_stride_rgba, 4)) {
>-    I422ToRGBARow = I422ToRGBARow_DSPR2;
>-  }
>-#endif
> #if defined(HAS_I422TORGBAROW_MSA)
>   if (TestCpuFlag(kCpuHasMSA)) {
>     I422ToRGBARow = I422ToRGBARow_Any_MSA;
>@@ -516,13 +583,13 @@ static int I420ToRGBAMatrix(const uint8* src_y,
> 
> // Convert I420 to RGBA.
> LIBYUV_API
>-int I420ToRGBA(const uint8* src_y,
>+int I420ToRGBA(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_rgba,
>+               uint8_t* dst_rgba,
>                int dst_stride_rgba,
>                int width,
>                int height) {
>@@ -533,13 +600,13 @@ int I420ToRGBA(const uint8* src_y,
> 
> // Convert I420 to BGRA.
> LIBYUV_API
>-int I420ToBGRA(const uint8* src_y,
>+int I420ToBGRA(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_bgra,
>+               uint8_t* dst_bgra,
>                int dst_stride_bgra,
>                int width,
>                int height) {
>@@ -551,20 +618,20 @@ int I420ToBGRA(const uint8* src_y,
> }
> 
> // Convert I420 to RGB24 with matrix
>-static int I420ToRGB24Matrix(const uint8* src_y,
>+static int I420ToRGB24Matrix(const uint8_t* src_y,
>                              int src_stride_y,
>-                             const uint8* src_u,
>+                             const uint8_t* src_u,
>                              int src_stride_u,
>-                             const uint8* src_v,
>+                             const uint8_t* src_v,
>                              int src_stride_v,
>-                             uint8* dst_rgb24,
>+                             uint8_t* dst_rgb24,
>                              int dst_stride_rgb24,
>                              const struct YuvConstants* yuvconstants,
>                              int width,
>                              int height) {
>   int y;
>-  void (*I422ToRGB24Row)(const uint8* y_buf, const uint8* u_buf,
>-                         const uint8* v_buf, uint8* rgb_buf,
>+  void (*I422ToRGB24Row)(const uint8_t* y_buf, const uint8_t* u_buf,
>+                         const uint8_t* v_buf, uint8_t* rgb_buf,
>                          const struct YuvConstants* yuvconstants, int width) =
>       I422ToRGB24Row_C;
>   if (!src_y || !src_u || !src_v || !dst_rgb24 || width <= 0 || height == 0) {
>@@ -623,13 +690,13 @@ static int I420ToRGB24Matrix(const uint8* src_y,
> 
> // Convert I420 to RGB24.
> LIBYUV_API
>-int I420ToRGB24(const uint8* src_y,
>+int I420ToRGB24(const uint8_t* src_y,
>                 int src_stride_y,
>-                const uint8* src_u,
>+                const uint8_t* src_u,
>                 int src_stride_u,
>-                const uint8* src_v,
>+                const uint8_t* src_v,
>                 int src_stride_v,
>-                uint8* dst_rgb24,
>+                uint8_t* dst_rgb24,
>                 int dst_stride_rgb24,
>                 int width,
>                 int height) {
>@@ -640,13 +707,13 @@ int I420ToRGB24(const uint8* src_y,
> 
> // Convert I420 to RAW.
> LIBYUV_API
>-int I420ToRAW(const uint8* src_y,
>+int I420ToRAW(const uint8_t* src_y,
>               int src_stride_y,
>-              const uint8* src_u,
>+              const uint8_t* src_u,
>               int src_stride_u,
>-              const uint8* src_v,
>+              const uint8_t* src_v,
>               int src_stride_v,
>-              uint8* dst_raw,
>+              uint8_t* dst_raw,
>               int dst_stride_raw,
>               int width,
>               int height) {
>@@ -659,13 +726,13 @@ int I420ToRAW(const uint8* src_y,
> 
> // Convert H420 to RGB24.
> LIBYUV_API
>-int H420ToRGB24(const uint8* src_y,
>+int H420ToRGB24(const uint8_t* src_y,
>                 int src_stride_y,
>-                const uint8* src_u,
>+                const uint8_t* src_u,
>                 int src_stride_u,
>-                const uint8* src_v,
>+                const uint8_t* src_v,
>                 int src_stride_v,
>-                uint8* dst_rgb24,
>+                uint8_t* dst_rgb24,
>                 int dst_stride_rgb24,
>                 int width,
>                 int height) {
>@@ -676,13 +743,13 @@ int H420ToRGB24(const uint8* src_y,
> 
> // Convert H420 to RAW.
> LIBYUV_API
>-int H420ToRAW(const uint8* src_y,
>+int H420ToRAW(const uint8_t* src_y,
>               int src_stride_y,
>-              const uint8* src_u,
>+              const uint8_t* src_u,
>               int src_stride_u,
>-              const uint8* src_v,
>+              const uint8_t* src_v,
>               int src_stride_v,
>-              uint8* dst_raw,
>+              uint8_t* dst_raw,
>               int dst_stride_raw,
>               int width,
>               int height) {
>@@ -695,19 +762,19 @@ int H420ToRAW(const uint8* src_y,
> 
> // Convert I420 to ARGB1555.
> LIBYUV_API
>-int I420ToARGB1555(const uint8* src_y,
>+int I420ToARGB1555(const uint8_t* src_y,
>                    int src_stride_y,
>-                   const uint8* src_u,
>+                   const uint8_t* src_u,
>                    int src_stride_u,
>-                   const uint8* src_v,
>+                   const uint8_t* src_v,
>                    int src_stride_v,
>-                   uint8* dst_argb1555,
>+                   uint8_t* dst_argb1555,
>                    int dst_stride_argb1555,
>                    int width,
>                    int height) {
>   int y;
>-  void (*I422ToARGB1555Row)(const uint8* y_buf, const uint8* u_buf,
>-                            const uint8* v_buf, uint8* rgb_buf,
>+  void (*I422ToARGB1555Row)(const uint8_t* y_buf, const uint8_t* u_buf,
>+                            const uint8_t* v_buf, uint8_t* rgb_buf,
>                             const struct YuvConstants* yuvconstants,
>                             int width) = I422ToARGB1555Row_C;
>   if (!src_y || !src_u || !src_v || !dst_argb1555 || width <= 0 ||
>@@ -744,14 +811,6 @@ int I420ToARGB1555(const uint8* src_y,
>     }
>   }
> #endif
>-#if defined(HAS_I422TOARGB1555ROW_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2)) {
>-    I422ToARGB1555Row = I422ToARGB1555Row_Any_DSPR2;
>-    if (IS_ALIGNED(width, 4)) {
>-      I422ToARGB1555Row = I422ToARGB1555Row_DSPR2;
>-    }
>-  }
>-#endif
> #if defined(HAS_I422TOARGB1555ROW_MSA)
>   if (TestCpuFlag(kCpuHasMSA)) {
>     I422ToARGB1555Row = I422ToARGB1555Row_Any_MSA;
>@@ -776,19 +835,19 @@ int I420ToARGB1555(const uint8* src_y,
> 
> // Convert I420 to ARGB4444.
> LIBYUV_API
>-int I420ToARGB4444(const uint8* src_y,
>+int I420ToARGB4444(const uint8_t* src_y,
>                    int src_stride_y,
>-                   const uint8* src_u,
>+                   const uint8_t* src_u,
>                    int src_stride_u,
>-                   const uint8* src_v,
>+                   const uint8_t* src_v,
>                    int src_stride_v,
>-                   uint8* dst_argb4444,
>+                   uint8_t* dst_argb4444,
>                    int dst_stride_argb4444,
>                    int width,
>                    int height) {
>   int y;
>-  void (*I422ToARGB4444Row)(const uint8* y_buf, const uint8* u_buf,
>-                            const uint8* v_buf, uint8* rgb_buf,
>+  void (*I422ToARGB4444Row)(const uint8_t* y_buf, const uint8_t* u_buf,
>+                            const uint8_t* v_buf, uint8_t* rgb_buf,
>                             const struct YuvConstants* yuvconstants,
>                             int width) = I422ToARGB4444Row_C;
>   if (!src_y || !src_u || !src_v || !dst_argb4444 || width <= 0 ||
>@@ -825,14 +884,6 @@ int I420ToARGB4444(const uint8* src_y,
>     }
>   }
> #endif
>-#if defined(HAS_I422TOARGB4444ROW_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2)) {
>-    I422ToARGB4444Row = I422ToARGB4444Row_Any_DSPR2;
>-    if (IS_ALIGNED(width, 4)) {
>-      I422ToARGB4444Row = I422ToARGB4444Row_DSPR2;
>-    }
>-  }
>-#endif
> #if defined(HAS_I422TOARGB4444ROW_MSA)
>   if (TestCpuFlag(kCpuHasMSA)) {
>     I422ToARGB4444Row = I422ToARGB4444Row_Any_MSA;
>@@ -857,19 +908,19 @@ int I420ToARGB4444(const uint8* src_y,
> 
> // Convert I420 to RGB565.
> LIBYUV_API
>-int I420ToRGB565(const uint8* src_y,
>+int I420ToRGB565(const uint8_t* src_y,
>                  int src_stride_y,
>-                 const uint8* src_u,
>+                 const uint8_t* src_u,
>                  int src_stride_u,
>-                 const uint8* src_v,
>+                 const uint8_t* src_v,
>                  int src_stride_v,
>-                 uint8* dst_rgb565,
>+                 uint8_t* dst_rgb565,
>                  int dst_stride_rgb565,
>                  int width,
>                  int height) {
>   int y;
>-  void (*I422ToRGB565Row)(const uint8* y_buf, const uint8* u_buf,
>-                          const uint8* v_buf, uint8* rgb_buf,
>+  void (*I422ToRGB565Row)(const uint8_t* y_buf, const uint8_t* u_buf,
>+                          const uint8_t* v_buf, uint8_t* rgb_buf,
>                           const struct YuvConstants* yuvconstants, int width) =
>       I422ToRGB565Row_C;
>   if (!src_y || !src_u || !src_v || !dst_rgb565 || width <= 0 || height == 0) {
>@@ -928,19 +979,19 @@ int I420ToRGB565(const uint8* src_y,
> 
> // Convert I422 to RGB565.
> LIBYUV_API
>-int I422ToRGB565(const uint8* src_y,
>+int I422ToRGB565(const uint8_t* src_y,
>                  int src_stride_y,
>-                 const uint8* src_u,
>+                 const uint8_t* src_u,
>                  int src_stride_u,
>-                 const uint8* src_v,
>+                 const uint8_t* src_v,
>                  int src_stride_v,
>-                 uint8* dst_rgb565,
>+                 uint8_t* dst_rgb565,
>                  int dst_stride_rgb565,
>                  int width,
>                  int height) {
>   int y;
>-  void (*I422ToRGB565Row)(const uint8* y_buf, const uint8* u_buf,
>-                          const uint8* v_buf, uint8* rgb_buf,
>+  void (*I422ToRGB565Row)(const uint8_t* y_buf, const uint8_t* u_buf,
>+                          const uint8_t* v_buf, uint8_t* rgb_buf,
>                           const struct YuvConstants* yuvconstants, int width) =
>       I422ToRGB565Row_C;
>   if (!src_y || !src_u || !src_v || !dst_rgb565 || width <= 0 || height == 0) {
>@@ -996,30 +1047,30 @@ int I422ToRGB565(const uint8* src_y,
> }
> 
> // Ordered 8x8 dither for 888 to 565.  Values from 0 to 7.
>-static const uint8 kDither565_4x4[16] = {
>+static const uint8_t kDither565_4x4[16] = {
>     0, 4, 1, 5, 6, 2, 7, 3, 1, 5, 0, 4, 7, 3, 6, 2,
> };
> 
> // Convert I420 to RGB565 with dithering.
> LIBYUV_API
>-int I420ToRGB565Dither(const uint8* src_y,
>+int I420ToRGB565Dither(const uint8_t* src_y,
>                        int src_stride_y,
>-                       const uint8* src_u,
>+                       const uint8_t* src_u,
>                        int src_stride_u,
>-                       const uint8* src_v,
>+                       const uint8_t* src_v,
>                        int src_stride_v,
>-                       uint8* dst_rgb565,
>+                       uint8_t* dst_rgb565,
>                        int dst_stride_rgb565,
>-                       const uint8* dither4x4,
>+                       const uint8_t* dither4x4,
>                        int width,
>                        int height) {
>   int y;
>-  void (*I422ToARGBRow)(const uint8* y_buf, const uint8* u_buf,
>-                        const uint8* v_buf, uint8* rgb_buf,
>+  void (*I422ToARGBRow)(const uint8_t* y_buf, const uint8_t* u_buf,
>+                        const uint8_t* v_buf, uint8_t* rgb_buf,
>                         const struct YuvConstants* yuvconstants, int width) =
>       I422ToARGBRow_C;
>-  void (*ARGBToRGB565DitherRow)(const uint8* src_argb, uint8* dst_rgb,
>-                                const uint32 dither4, int width) =
>+  void (*ARGBToRGB565DitherRow)(const uint8_t* src_argb, uint8_t* dst_rgb,
>+                                const uint32_t dither4, int width) =
>       ARGBToRGB565DitherRow_C;
>   if (!src_y || !src_u || !src_v || !dst_rgb565 || width <= 0 || height == 0) {
>     return -1;
>@@ -1057,14 +1108,6 @@ int I420ToRGB565Dither(const uint8* src_y,
>     }
>   }
> #endif
>-#if defined(HAS_I422TOARGBROW_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2) && IS_ALIGNED(width, 4) &&
>-      IS_ALIGNED(src_y, 4) && IS_ALIGNED(src_stride_y, 4) &&
>-      IS_ALIGNED(src_u, 2) && IS_ALIGNED(src_stride_u, 2) &&
>-      IS_ALIGNED(src_v, 2) && IS_ALIGNED(src_stride_v, 2)) {
>-    I422ToARGBRow = I422ToARGBRow_DSPR2;
>-  }
>-#endif
> #if defined(HAS_I422TOARGBROW_MSA)
>   if (TestCpuFlag(kCpuHasMSA)) {
>     I422ToARGBRow = I422ToARGBRow_Any_MSA;
>@@ -1111,8 +1154,8 @@ int I420ToRGB565Dither(const uint8* src_y,
>     for (y = 0; y < height; ++y) {
>       I422ToARGBRow(src_y, src_u, src_v, row_argb, &kYuvI601Constants, width);
>       ARGBToRGB565DitherRow(row_argb, dst_rgb565,
>-                            *(uint32*)(dither4x4 + ((y & 3) << 2)),  // NOLINT
>-                            width);                                  // NOLINT
>+                            *(uint32_t*)(dither4x4 + ((y & 3) << 2)),  // NOLINT
>+                            width);                                    // NOLINT
>       dst_rgb565 += dst_stride_rgb565;
>       src_y += src_stride_y;
>       if (y & 1) {
>@@ -1125,20 +1168,111 @@ int I420ToRGB565Dither(const uint8* src_y,
>   return 0;
> }
> 
>+// Convert I420 to AR30 with matrix
>+static int I420ToAR30Matrix(const uint8_t* src_y,
>+                            int src_stride_y,
>+                            const uint8_t* src_u,
>+                            int src_stride_u,
>+                            const uint8_t* src_v,
>+                            int src_stride_v,
>+                            uint8_t* dst_ar30,
>+                            int dst_stride_ar30,
>+                            const struct YuvConstants* yuvconstants,
>+                            int width,
>+                            int height) {
>+  int y;
>+  void (*I422ToAR30Row)(const uint8_t* y_buf, const uint8_t* u_buf,
>+                        const uint8_t* v_buf, uint8_t* rgb_buf,
>+                        const struct YuvConstants* yuvconstants, int width) =
>+      I422ToAR30Row_C;
>+
>+  if (!src_y || !src_u || !src_v || !dst_ar30 || width <= 0 || height == 0) {
>+    return -1;
>+  }
>+  // Negative height means invert the image.
>+  if (height < 0) {
>+    height = -height;
>+    dst_ar30 = dst_ar30 + (height - 1) * dst_stride_ar30;
>+    dst_stride_ar30 = -dst_stride_ar30;
>+  }
>+
>+#if defined(HAS_I422TOAR30ROW_SSSE3)
>+  if (TestCpuFlag(kCpuHasSSSE3)) {
>+    I422ToAR30Row = I422ToAR30Row_Any_SSSE3;
>+    if (IS_ALIGNED(width, 8)) {
>+      I422ToAR30Row = I422ToAR30Row_SSSE3;
>+    }
>+  }
>+#endif
>+#if defined(HAS_I422TOAR30ROW_AVX2)
>+  if (TestCpuFlag(kCpuHasAVX2)) {
>+    I422ToAR30Row = I422ToAR30Row_Any_AVX2;
>+    if (IS_ALIGNED(width, 16)) {
>+      I422ToAR30Row = I422ToAR30Row_AVX2;
>+    }
>+  }
>+#endif
>+
>+  for (y = 0; y < height; ++y) {
>+    I422ToAR30Row(src_y, src_u, src_v, dst_ar30, yuvconstants, width);
>+    dst_ar30 += dst_stride_ar30;
>+    src_y += src_stride_y;
>+    if (y & 1) {
>+      src_u += src_stride_u;
>+      src_v += src_stride_v;
>+    }
>+  }
>+  return 0;
>+}
>+
>+// Convert I420 to AR30.
>+LIBYUV_API
>+int I420ToAR30(const uint8_t* src_y,
>+               int src_stride_y,
>+               const uint8_t* src_u,
>+               int src_stride_u,
>+               const uint8_t* src_v,
>+               int src_stride_v,
>+               uint8_t* dst_ar30,
>+               int dst_stride_ar30,
>+               int width,
>+               int height) {
>+  return I420ToAR30Matrix(src_y, src_stride_y, src_u, src_stride_u, src_v,
>+                          src_stride_v, dst_ar30, dst_stride_ar30,
>+                          &kYuvI601Constants, width, height);
>+}
>+
>+// Convert H420 to AR30.
>+LIBYUV_API
>+int H420ToAR30(const uint8_t* src_y,
>+               int src_stride_y,
>+               const uint8_t* src_u,
>+               int src_stride_u,
>+               const uint8_t* src_v,
>+               int src_stride_v,
>+               uint8_t* dst_ar30,
>+               int dst_stride_ar30,
>+               int width,
>+               int height) {
>+  return I420ToAR30Matrix(src_y, src_stride_y, src_u, src_stride_u, src_v,
>+                          src_stride_v, dst_ar30, dst_stride_ar30,
>+                          &kYvuH709Constants, width, height);
>+}
>+
> // Convert I420 to specified format
> LIBYUV_API
>-int ConvertFromI420(const uint8* y,
>+int ConvertFromI420(const uint8_t* y,
>                     int y_stride,
>-                    const uint8* u,
>+                    const uint8_t* u,
>                     int u_stride,
>-                    const uint8* v,
>+                    const uint8_t* v,
>                     int v_stride,
>-                    uint8* dst_sample,
>+                    uint8_t* dst_sample,
>                     int dst_sample_stride,
>                     int width,
>                     int height,
>-                    uint32 fourcc) {
>-  uint32 format = CanonicalFourCC(fourcc);
>+                    uint32_t fourcc) {
>+  uint32_t format = CanonicalFourCC(fourcc);
>   int r = 0;
>   if (!y || !u || !v || !dst_sample || width <= 0 || height == 0) {
>     return -1;
>@@ -1200,13 +1334,18 @@ int ConvertFromI420(const uint8* y,
>                      dst_sample_stride ? dst_sample_stride : width * 4, width,
>                      height);
>       break;
>+    case FOURCC_AR30:
>+      r = I420ToAR30(y, y_stride, u, u_stride, v, v_stride, dst_sample,
>+                     dst_sample_stride ? dst_sample_stride : width * 4, width,
>+                     height);
>+      break;
>     case FOURCC_I400:
>       r = I400Copy(y, y_stride, dst_sample,
>                    dst_sample_stride ? dst_sample_stride : width, width,
>                    height);
>       break;
>     case FOURCC_NV12: {
>-      uint8* dst_uv = dst_sample + width * height;
>+      uint8_t* dst_uv = dst_sample + width * height;
>       r = I420ToNV12(y, y_stride, u, u_stride, v, v_stride, dst_sample,
>                      dst_sample_stride ? dst_sample_stride : width, dst_uv,
>                      dst_sample_stride ? dst_sample_stride : width, width,
>@@ -1214,7 +1353,7 @@ int ConvertFromI420(const uint8* y,
>       break;
>     }
>     case FOURCC_NV21: {
>-      uint8* dst_vu = dst_sample + width * height;
>+      uint8_t* dst_vu = dst_sample + width * height;
>       r = I420ToNV21(y, y_stride, u, u_stride, v, v_stride, dst_sample,
>                      dst_sample_stride ? dst_sample_stride : width, dst_vu,
>                      dst_sample_stride ? dst_sample_stride : width, width,
>@@ -1228,8 +1367,8 @@ int ConvertFromI420(const uint8* y,
>       dst_sample_stride = dst_sample_stride ? dst_sample_stride : width;
>       int halfstride = (dst_sample_stride + 1) / 2;
>       int halfheight = (height + 1) / 2;
>-      uint8* dst_u;
>-      uint8* dst_v;
>+      uint8_t* dst_u;
>+      uint8_t* dst_v;
>       if (format == FOURCC_YV12) {
>         dst_v = dst_sample + dst_sample_stride * height;
>         dst_u = dst_v + halfstride * halfheight;
>@@ -1246,8 +1385,8 @@ int ConvertFromI420(const uint8* y,
>     case FOURCC_YV16: {
>       dst_sample_stride = dst_sample_stride ? dst_sample_stride : width;
>       int halfstride = (dst_sample_stride + 1) / 2;
>-      uint8* dst_u;
>-      uint8* dst_v;
>+      uint8_t* dst_u;
>+      uint8_t* dst_v;
>       if (format == FOURCC_YV16) {
>         dst_v = dst_sample + dst_sample_stride * height;
>         dst_u = dst_v + halfstride * height;
>@@ -1263,8 +1402,8 @@ int ConvertFromI420(const uint8* y,
>     case FOURCC_I444:
>     case FOURCC_YV24: {
>       dst_sample_stride = dst_sample_stride ? dst_sample_stride : width;
>-      uint8* dst_u;
>-      uint8* dst_v;
>+      uint8_t* dst_u;
>+      uint8_t* dst_v;
>       if (format == FOURCC_YV24) {
>         dst_v = dst_sample + dst_sample_stride * height;
>         dst_u = dst_v + dst_sample_stride * height;
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/convert_from_argb.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/convert_from_argb.cc
>index 88f38279ace..16b838458f0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/convert_from_argb.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/convert_from_argb.cc
>@@ -22,21 +22,21 @@ extern "C" {
> 
> // ARGB little endian (bgra in memory) to I444
> LIBYUV_API
>-int ARGBToI444(const uint8* src_argb,
>+int ARGBToI444(const uint8_t* src_argb,
>                int src_stride_argb,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_u,
>+               uint8_t* dst_u,
>                int dst_stride_u,
>-               uint8* dst_v,
>+               uint8_t* dst_v,
>                int dst_stride_v,
>                int width,
>                int height) {
>   int y;
>-  void (*ARGBToYRow)(const uint8* src_argb, uint8* dst_y, int width) =
>+  void (*ARGBToYRow)(const uint8_t* src_argb, uint8_t* dst_y, int width) =
>       ARGBToYRow_C;
>-  void (*ARGBToUV444Row)(const uint8* src_argb, uint8* dst_u, uint8* dst_v,
>-                         int width) = ARGBToUV444Row_C;
>+  void (*ARGBToUV444Row)(const uint8_t* src_argb, uint8_t* dst_u,
>+                         uint8_t* dst_v, int width) = ARGBToUV444Row_C;
>   if (!src_argb || !dst_y || !dst_u || !dst_v || width <= 0 || height == 0) {
>     return -1;
>   }
>@@ -100,14 +100,6 @@ int ARGBToI444(const uint8* src_argb,
>     }
>   }
> #endif
>-#if defined(HAS_ARGBTOYROW_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2)) {
>-    ARGBToYRow = ARGBToYRow_Any_DSPR2;
>-    if (IS_ALIGNED(width, 8)) {
>-      ARGBToYRow = ARGBToYRow_DSPR2;
>-    }
>-  }
>-#endif
> #if defined(HAS_ARGBTOYROW_MSA)
>   if (TestCpuFlag(kCpuHasMSA)) {
>     ARGBToYRow = ARGBToYRow_Any_MSA;
>@@ -130,20 +122,21 @@ int ARGBToI444(const uint8* src_argb,
> 
> // ARGB little endian (bgra in memory) to I422
> LIBYUV_API
>-int ARGBToI422(const uint8* src_argb,
>+int ARGBToI422(const uint8_t* src_argb,
>                int src_stride_argb,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_u,
>+               uint8_t* dst_u,
>                int dst_stride_u,
>-               uint8* dst_v,
>+               uint8_t* dst_v,
>                int dst_stride_v,
>                int width,
>                int height) {
>   int y;
>-  void (*ARGBToUVRow)(const uint8* src_argb0, int src_stride_argb, uint8* dst_u,
>-                      uint8* dst_v, int width) = ARGBToUVRow_C;
>-  void (*ARGBToYRow)(const uint8* src_argb, uint8* dst_y, int width) =
>+  void (*ARGBToUVRow)(const uint8_t* src_argb0, int src_stride_argb,
>+                      uint8_t* dst_u, uint8_t* dst_v, int width) =
>+      ARGBToUVRow_C;
>+  void (*ARGBToYRow)(const uint8_t* src_argb, uint8_t* dst_y, int width) =
>       ARGBToYRow_C;
>   if (!src_argb || !dst_y || !dst_u || !dst_v || width <= 0 || height == 0) {
>     return -1;
>@@ -197,22 +190,6 @@ int ARGBToI422(const uint8* src_argb,
>     }
>   }
> #endif
>-#if defined(HAS_ARGBTOYROW_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2)) {
>-    ARGBToYRow = ARGBToYRow_Any_DSPR2;
>-    if (IS_ALIGNED(width, 8)) {
>-      ARGBToYRow = ARGBToYRow_DSPR2;
>-    }
>-  }
>-#endif
>-#if defined(HAS_ARGBTOUVROW_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2)) {
>-    ARGBToUVRow = ARGBToUVRow_Any_DSPR2;
>-    if (IS_ALIGNED(width, 16)) {
>-      ARGBToUVRow = ARGBToUVRow_DSPR2;
>-    }
>-  }
>-#endif
> 
> #if defined(HAS_ARGBTOYROW_MSA)
>   if (TestCpuFlag(kCpuHasMSA)) {
>@@ -243,22 +220,23 @@ int ARGBToI422(const uint8* src_argb,
> }
> 
> LIBYUV_API
>-int ARGBToNV12(const uint8* src_argb,
>+int ARGBToNV12(const uint8_t* src_argb,
>                int src_stride_argb,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_uv,
>+               uint8_t* dst_uv,
>                int dst_stride_uv,
>                int width,
>                int height) {
>   int y;
>   int halfwidth = (width + 1) >> 1;
>-  void (*ARGBToUVRow)(const uint8* src_argb0, int src_stride_argb, uint8* dst_u,
>-                      uint8* dst_v, int width) = ARGBToUVRow_C;
>-  void (*ARGBToYRow)(const uint8* src_argb, uint8* dst_y, int width) =
>+  void (*ARGBToUVRow)(const uint8_t* src_argb0, int src_stride_argb,
>+                      uint8_t* dst_u, uint8_t* dst_v, int width) =
>+      ARGBToUVRow_C;
>+  void (*ARGBToYRow)(const uint8_t* src_argb, uint8_t* dst_y, int width) =
>       ARGBToYRow_C;
>-  void (*MergeUVRow_)(const uint8* src_u, const uint8* src_v, uint8* dst_uv,
>-                      int width) = MergeUVRow_C;
>+  void (*MergeUVRow_)(const uint8_t* src_u, const uint8_t* src_v,
>+                      uint8_t* dst_uv, int width) = MergeUVRow_C;
>   if (!src_argb || !dst_y || !dst_uv || width <= 0 || height == 0) {
>     return -1;
>   }
>@@ -344,22 +322,6 @@ int ARGBToNV12(const uint8* src_argb,
>     }
>   }
> #endif
>-#if defined(HAS_ARGBTOYROW_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2)) {
>-    ARGBToYRow = ARGBToYRow_Any_DSPR2;
>-    if (IS_ALIGNED(width, 8)) {
>-      ARGBToYRow = ARGBToYRow_DSPR2;
>-    }
>-  }
>-#endif
>-#if defined(HAS_ARGBTOUVROW_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2)) {
>-    ARGBToUVRow = ARGBToUVRow_Any_DSPR2;
>-    if (IS_ALIGNED(width, 16)) {
>-      ARGBToUVRow = ARGBToUVRow_DSPR2;
>-    }
>-  }
>-#endif
> #if defined(HAS_MERGEUVROW_MSA)
>   if (TestCpuFlag(kCpuHasMSA)) {
>     MergeUVRow_ = MergeUVRow_Any_MSA;
>@@ -371,7 +333,7 @@ int ARGBToNV12(const uint8* src_argb,
>   {
>     // Allocate a rows of uv.
>     align_buffer_64(row_u, ((halfwidth + 31) & ~31) * 2);
>-    uint8* row_v = row_u + ((halfwidth + 31) & ~31);
>+    uint8_t* row_v = row_u + ((halfwidth + 31) & ~31);
> 
>     for (y = 0; y < height - 1; y += 2) {
>       ARGBToUVRow(src_argb, src_stride_argb, row_u, row_v, width);
>@@ -394,23 +356,24 @@ int ARGBToNV12(const uint8* src_argb,
> 
> // Same as NV12 but U and V swapped.
> LIBYUV_API
>-int ARGBToNV21(const uint8* src_argb,
>+int ARGBToNV21(const uint8_t* src_argb,
>                int src_stride_argb,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_uv,
>-               int dst_stride_uv,
>+               uint8_t* dst_vu,
>+               int dst_stride_vu,
>                int width,
>                int height) {
>   int y;
>   int halfwidth = (width + 1) >> 1;
>-  void (*ARGBToUVRow)(const uint8* src_argb0, int src_stride_argb, uint8* dst_u,
>-                      uint8* dst_v, int width) = ARGBToUVRow_C;
>-  void (*ARGBToYRow)(const uint8* src_argb, uint8* dst_y, int width) =
>+  void (*ARGBToUVRow)(const uint8_t* src_argb0, int src_stride_argb,
>+                      uint8_t* dst_u, uint8_t* dst_v, int width) =
>+      ARGBToUVRow_C;
>+  void (*ARGBToYRow)(const uint8_t* src_argb, uint8_t* dst_y, int width) =
>       ARGBToYRow_C;
>-  void (*MergeUVRow_)(const uint8* src_u, const uint8* src_v, uint8* dst_uv,
>-                      int width) = MergeUVRow_C;
>-  if (!src_argb || !dst_y || !dst_uv || width <= 0 || height == 0) {
>+  void (*MergeUVRow_)(const uint8_t* src_u, const uint8_t* src_v,
>+                      uint8_t* dst_vu, int width) = MergeUVRow_C;
>+  if (!src_argb || !dst_y || !dst_vu || width <= 0 || height == 0) {
>     return -1;
>   }
>   // Negative height means invert the image.
>@@ -495,22 +458,6 @@ int ARGBToNV21(const uint8* src_argb,
>     }
>   }
> #endif
>-#if defined(HAS_ARGBTOYROW_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2)) {
>-    ARGBToYRow = ARGBToYRow_Any_DSPR2;
>-    if (IS_ALIGNED(width, 8)) {
>-      ARGBToYRow = ARGBToYRow_DSPR2;
>-    }
>-  }
>-#endif
>-#if defined(HAS_ARGBTOUVROW_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2)) {
>-    ARGBToUVRow = ARGBToUVRow_Any_DSPR2;
>-    if (IS_ALIGNED(width, 16)) {
>-      ARGBToUVRow = ARGBToUVRow_DSPR2;
>-    }
>-  }
>-#endif
> #if defined(HAS_MERGEUVROW_MSA)
>   if (TestCpuFlag(kCpuHasMSA)) {
>     MergeUVRow_ = MergeUVRow_Any_MSA;
>@@ -522,20 +469,20 @@ int ARGBToNV21(const uint8* src_argb,
>   {
>     // Allocate a rows of uv.
>     align_buffer_64(row_u, ((halfwidth + 31) & ~31) * 2);
>-    uint8* row_v = row_u + ((halfwidth + 31) & ~31);
>+    uint8_t* row_v = row_u + ((halfwidth + 31) & ~31);
> 
>     for (y = 0; y < height - 1; y += 2) {
>       ARGBToUVRow(src_argb, src_stride_argb, row_u, row_v, width);
>-      MergeUVRow_(row_v, row_u, dst_uv, halfwidth);
>+      MergeUVRow_(row_v, row_u, dst_vu, halfwidth);
>       ARGBToYRow(src_argb, dst_y, width);
>       ARGBToYRow(src_argb + src_stride_argb, dst_y + dst_stride_y, width);
>       src_argb += src_stride_argb * 2;
>       dst_y += dst_stride_y * 2;
>-      dst_uv += dst_stride_uv;
>+      dst_vu += dst_stride_vu;
>     }
>     if (height & 1) {
>       ARGBToUVRow(src_argb, 0, row_u, row_v, width);
>-      MergeUVRow_(row_v, row_u, dst_uv, halfwidth);
>+      MergeUVRow_(row_v, row_u, dst_vu, halfwidth);
>       ARGBToYRow(src_argb, dst_y, width);
>     }
>     free_aligned_buffer_64(row_u);
>@@ -545,19 +492,20 @@ int ARGBToNV21(const uint8* src_argb,
> 
> // Convert ARGB to YUY2.
> LIBYUV_API
>-int ARGBToYUY2(const uint8* src_argb,
>+int ARGBToYUY2(const uint8_t* src_argb,
>                int src_stride_argb,
>-               uint8* dst_yuy2,
>+               uint8_t* dst_yuy2,
>                int dst_stride_yuy2,
>                int width,
>                int height) {
>   int y;
>-  void (*ARGBToUVRow)(const uint8* src_argb, int src_stride_argb, uint8* dst_u,
>-                      uint8* dst_v, int width) = ARGBToUVRow_C;
>-  void (*ARGBToYRow)(const uint8* src_argb, uint8* dst_y, int width) =
>+  void (*ARGBToUVRow)(const uint8_t* src_argb, int src_stride_argb,
>+                      uint8_t* dst_u, uint8_t* dst_v, int width) =
>+      ARGBToUVRow_C;
>+  void (*ARGBToYRow)(const uint8_t* src_argb, uint8_t* dst_y, int width) =
>       ARGBToYRow_C;
>-  void (*I422ToYUY2Row)(const uint8* src_y, const uint8* src_u,
>-                        const uint8* src_v, uint8* dst_yuy2, int width) =
>+  void (*I422ToYUY2Row)(const uint8_t* src_y, const uint8_t* src_u,
>+                        const uint8_t* src_v, uint8_t* dst_yuy2, int width) =
>       I422ToYUY2Row_C;
> 
>   if (!src_argb || !dst_yuy2 || width <= 0 || height == 0) {
>@@ -635,6 +583,14 @@ int ARGBToYUY2(const uint8* src_argb,
>     }
>   }
> #endif
>+#if defined(HAS_I422TOYUY2ROW_AVX2)
>+  if (TestCpuFlag(kCpuHasAVX2)) {
>+    I422ToYUY2Row = I422ToYUY2Row_Any_AVX2;
>+    if (IS_ALIGNED(width, 32)) {
>+      I422ToYUY2Row = I422ToYUY2Row_AVX2;
>+    }
>+  }
>+#endif
> #if defined(HAS_I422TOYUY2ROW_NEON)
>   if (TestCpuFlag(kCpuHasNEON)) {
>     I422ToYUY2Row = I422ToYUY2Row_Any_NEON;
>@@ -643,22 +599,6 @@ int ARGBToYUY2(const uint8* src_argb,
>     }
>   }
> #endif
>-#if defined(HAS_ARGBTOYROW_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2)) {
>-    ARGBToYRow = ARGBToYRow_Any_DSPR2;
>-    if (IS_ALIGNED(width, 8)) {
>-      ARGBToYRow = ARGBToYRow_DSPR2;
>-    }
>-  }
>-#endif
>-#if defined(HAS_ARGBTOUVROW_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2)) {
>-    ARGBToUVRow = ARGBToUVRow_Any_DSPR2;
>-    if (IS_ALIGNED(width, 16)) {
>-      ARGBToUVRow = ARGBToUVRow_DSPR2;
>-    }
>-  }
>-#endif
> #if defined(HAS_I422TOYUY2ROW_MSA)
>   if (TestCpuFlag(kCpuHasMSA)) {
>     I422ToYUY2Row = I422ToYUY2Row_Any_MSA;
>@@ -671,8 +611,8 @@ int ARGBToYUY2(const uint8* src_argb,
>   {
>     // Allocate a rows of yuv.
>     align_buffer_64(row_y, ((width + 63) & ~63) * 2);
>-    uint8* row_u = row_y + ((width + 63) & ~63);
>-    uint8* row_v = row_u + ((width + 63) & ~63) / 2;
>+    uint8_t* row_u = row_y + ((width + 63) & ~63);
>+    uint8_t* row_v = row_u + ((width + 63) & ~63) / 2;
> 
>     for (y = 0; y < height; ++y) {
>       ARGBToUVRow(src_argb, 0, row_u, row_v, width);
>@@ -689,19 +629,20 @@ int ARGBToYUY2(const uint8* src_argb,
> 
> // Convert ARGB to UYVY.
> LIBYUV_API
>-int ARGBToUYVY(const uint8* src_argb,
>+int ARGBToUYVY(const uint8_t* src_argb,
>                int src_stride_argb,
>-               uint8* dst_uyvy,
>+               uint8_t* dst_uyvy,
>                int dst_stride_uyvy,
>                int width,
>                int height) {
>   int y;
>-  void (*ARGBToUVRow)(const uint8* src_argb, int src_stride_argb, uint8* dst_u,
>-                      uint8* dst_v, int width) = ARGBToUVRow_C;
>-  void (*ARGBToYRow)(const uint8* src_argb, uint8* dst_y, int width) =
>+  void (*ARGBToUVRow)(const uint8_t* src_argb, int src_stride_argb,
>+                      uint8_t* dst_u, uint8_t* dst_v, int width) =
>+      ARGBToUVRow_C;
>+  void (*ARGBToYRow)(const uint8_t* src_argb, uint8_t* dst_y, int width) =
>       ARGBToYRow_C;
>-  void (*I422ToUYVYRow)(const uint8* src_y, const uint8* src_u,
>-                        const uint8* src_v, uint8* dst_uyvy, int width) =
>+  void (*I422ToUYVYRow)(const uint8_t* src_y, const uint8_t* src_u,
>+                        const uint8_t* src_v, uint8_t* dst_uyvy, int width) =
>       I422ToUYVYRow_C;
> 
>   if (!src_argb || !dst_uyvy || width <= 0 || height == 0) {
>@@ -779,6 +720,14 @@ int ARGBToUYVY(const uint8* src_argb,
>     }
>   }
> #endif
>+#if defined(HAS_I422TOUYVYROW_AVX2)
>+  if (TestCpuFlag(kCpuHasAVX2)) {
>+    I422ToUYVYRow = I422ToUYVYRow_Any_AVX2;
>+    if (IS_ALIGNED(width, 32)) {
>+      I422ToUYVYRow = I422ToUYVYRow_AVX2;
>+    }
>+  }
>+#endif
> #if defined(HAS_I422TOUYVYROW_NEON)
>   if (TestCpuFlag(kCpuHasNEON)) {
>     I422ToUYVYRow = I422ToUYVYRow_Any_NEON;
>@@ -787,22 +736,6 @@ int ARGBToUYVY(const uint8* src_argb,
>     }
>   }
> #endif
>-#if defined(HAS_ARGBTOYROW_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2)) {
>-    ARGBToYRow = ARGBToYRow_Any_DSPR2;
>-    if (IS_ALIGNED(width, 8)) {
>-      ARGBToYRow = ARGBToYRow_DSPR2;
>-    }
>-  }
>-#endif
>-#if defined(HAS_ARGBTOUVROW_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2)) {
>-    ARGBToUVRow = ARGBToUVRow_Any_DSPR2;
>-    if (IS_ALIGNED(width, 16)) {
>-      ARGBToUVRow = ARGBToUVRow_DSPR2;
>-    }
>-  }
>-#endif
> #if defined(HAS_I422TOUYVYROW_MSA)
>   if (TestCpuFlag(kCpuHasMSA)) {
>     I422ToUYVYRow = I422ToUYVYRow_Any_MSA;
>@@ -815,8 +748,8 @@ int ARGBToUYVY(const uint8* src_argb,
>   {
>     // Allocate a rows of yuv.
>     align_buffer_64(row_y, ((width + 63) & ~63) * 2);
>-    uint8* row_u = row_y + ((width + 63) & ~63);
>-    uint8* row_v = row_u + ((width + 63) & ~63) / 2;
>+    uint8_t* row_u = row_y + ((width + 63) & ~63);
>+    uint8_t* row_v = row_u + ((width + 63) & ~63) / 2;
> 
>     for (y = 0; y < height; ++y) {
>       ARGBToUVRow(src_argb, 0, row_u, row_v, width);
>@@ -833,14 +766,14 @@ int ARGBToUYVY(const uint8* src_argb,
> 
> // Convert ARGB to I400.
> LIBYUV_API
>-int ARGBToI400(const uint8* src_argb,
>+int ARGBToI400(const uint8_t* src_argb,
>                int src_stride_argb,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>                int width,
>                int height) {
>   int y;
>-  void (*ARGBToYRow)(const uint8* src_argb, uint8* dst_y, int width) =
>+  void (*ARGBToYRow)(const uint8_t* src_argb, uint8_t* dst_y, int width) =
>       ARGBToYRow_C;
>   if (!src_argb || !dst_y || width <= 0 || height == 0) {
>     return -1;
>@@ -880,14 +813,6 @@ int ARGBToI400(const uint8* src_argb,
>     }
>   }
> #endif
>-#if defined(HAS_ARGBTOYROW_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2)) {
>-    ARGBToYRow = ARGBToYRow_Any_DSPR2;
>-    if (IS_ALIGNED(width, 8)) {
>-      ARGBToYRow = ARGBToYRow_DSPR2;
>-    }
>-  }
>-#endif
> #if defined(HAS_ARGBTOYROW_MSA)
>   if (TestCpuFlag(kCpuHasMSA)) {
>     ARGBToYRow = ARGBToYRow_Any_MSA;
>@@ -906,31 +831,31 @@ int ARGBToI400(const uint8* src_argb,
> }
> 
> // Shuffle table for converting ARGB to RGBA.
>-static uvec8 kShuffleMaskARGBToRGBA = {3u,  0u, 1u, 2u,  7u,  4u,  5u,  6u,
>-                                       11u, 8u, 9u, 10u, 15u, 12u, 13u, 14u};
>+static const uvec8 kShuffleMaskARGBToRGBA = {
>+    3u, 0u, 1u, 2u, 7u, 4u, 5u, 6u, 11u, 8u, 9u, 10u, 15u, 12u, 13u, 14u};
> 
> // Convert ARGB to RGBA.
> LIBYUV_API
>-int ARGBToRGBA(const uint8* src_argb,
>+int ARGBToRGBA(const uint8_t* src_argb,
>                int src_stride_argb,
>-               uint8* dst_rgba,
>+               uint8_t* dst_rgba,
>                int dst_stride_rgba,
>                int width,
>                int height) {
>   return ARGBShuffle(src_argb, src_stride_argb, dst_rgba, dst_stride_rgba,
>-                     (const uint8*)(&kShuffleMaskARGBToRGBA), width, height);
>+                     (const uint8_t*)(&kShuffleMaskARGBToRGBA), width, height);
> }
> 
> // Convert ARGB To RGB24.
> LIBYUV_API
>-int ARGBToRGB24(const uint8* src_argb,
>+int ARGBToRGB24(const uint8_t* src_argb,
>                 int src_stride_argb,
>-                uint8* dst_rgb24,
>+                uint8_t* dst_rgb24,
>                 int dst_stride_rgb24,
>                 int width,
>                 int height) {
>   int y;
>-  void (*ARGBToRGB24Row)(const uint8* src_argb, uint8* dst_rgb, int width) =
>+  void (*ARGBToRGB24Row)(const uint8_t* src_argb, uint8_t* dst_rgb, int width) =
>       ARGBToRGB24Row_C;
>   if (!src_argb || !dst_rgb24 || width <= 0 || height == 0) {
>     return -1;
>@@ -981,14 +906,14 @@ int ARGBToRGB24(const uint8* src_argb,
> 
> // Convert ARGB To RAW.
> LIBYUV_API
>-int ARGBToRAW(const uint8* src_argb,
>+int ARGBToRAW(const uint8_t* src_argb,
>               int src_stride_argb,
>-              uint8* dst_raw,
>+              uint8_t* dst_raw,
>               int dst_stride_raw,
>               int width,
>               int height) {
>   int y;
>-  void (*ARGBToRAWRow)(const uint8* src_argb, uint8* dst_rgb, int width) =
>+  void (*ARGBToRAWRow)(const uint8_t* src_argb, uint8_t* dst_rgb, int width) =
>       ARGBToRAWRow_C;
>   if (!src_argb || !dst_raw || width <= 0 || height == 0) {
>     return -1;
>@@ -1038,22 +963,22 @@ int ARGBToRAW(const uint8* src_argb,
> }
> 
> // Ordered 8x8 dither for 888 to 565.  Values from 0 to 7.
>-static const uint8 kDither565_4x4[16] = {
>+static const uint8_t kDither565_4x4[16] = {
>     0, 4, 1, 5, 6, 2, 7, 3, 1, 5, 0, 4, 7, 3, 6, 2,
> };
> 
> // Convert ARGB To RGB565 with 4x4 dither matrix (16 bytes).
> LIBYUV_API
>-int ARGBToRGB565Dither(const uint8* src_argb,
>+int ARGBToRGB565Dither(const uint8_t* src_argb,
>                        int src_stride_argb,
>-                       uint8* dst_rgb565,
>+                       uint8_t* dst_rgb565,
>                        int dst_stride_rgb565,
>-                       const uint8* dither4x4,
>+                       const uint8_t* dither4x4,
>                        int width,
>                        int height) {
>   int y;
>-  void (*ARGBToRGB565DitherRow)(const uint8* src_argb, uint8* dst_rgb,
>-                                const uint32 dither4, int width) =
>+  void (*ARGBToRGB565DitherRow)(const uint8_t* src_argb, uint8_t* dst_rgb,
>+                                const uint32_t dither4, int width) =
>       ARGBToRGB565DitherRow_C;
>   if (!src_argb || !dst_rgb565 || width <= 0 || height == 0) {
>     return -1;
>@@ -1101,7 +1026,7 @@ int ARGBToRGB565Dither(const uint8* src_argb,
> 
>   for (y = 0; y < height; ++y) {
>     ARGBToRGB565DitherRow(src_argb, dst_rgb565,
>-                          *(uint32*)(dither4x4 + ((y & 3) << 2)),
>+                          *(uint32_t*)(dither4x4 + ((y & 3) << 2)),  // NOLINT
>                           width); /* NOLINT */
>     src_argb += src_stride_argb;
>     dst_rgb565 += dst_stride_rgb565;
>@@ -1112,15 +1037,15 @@ int ARGBToRGB565Dither(const uint8* src_argb,
> // Convert ARGB To RGB565.
> // TODO(fbarchard): Consider using dither function low level with zeros.
> LIBYUV_API
>-int ARGBToRGB565(const uint8* src_argb,
>+int ARGBToRGB565(const uint8_t* src_argb,
>                  int src_stride_argb,
>-                 uint8* dst_rgb565,
>+                 uint8_t* dst_rgb565,
>                  int dst_stride_rgb565,
>                  int width,
>                  int height) {
>   int y;
>-  void (*ARGBToRGB565Row)(const uint8* src_argb, uint8* dst_rgb, int width) =
>-      ARGBToRGB565Row_C;
>+  void (*ARGBToRGB565Row)(const uint8_t* src_argb, uint8_t* dst_rgb,
>+                          int width) = ARGBToRGB565Row_C;
>   if (!src_argb || !dst_rgb565 || width <= 0 || height == 0) {
>     return -1;
>   }
>@@ -1178,15 +1103,15 @@ int ARGBToRGB565(const uint8* src_argb,
> 
> // Convert ARGB To ARGB1555.
> LIBYUV_API
>-int ARGBToARGB1555(const uint8* src_argb,
>+int ARGBToARGB1555(const uint8_t* src_argb,
>                    int src_stride_argb,
>-                   uint8* dst_argb1555,
>+                   uint8_t* dst_argb1555,
>                    int dst_stride_argb1555,
>                    int width,
>                    int height) {
>   int y;
>-  void (*ARGBToARGB1555Row)(const uint8* src_argb, uint8* dst_rgb, int width) =
>-      ARGBToARGB1555Row_C;
>+  void (*ARGBToARGB1555Row)(const uint8_t* src_argb, uint8_t* dst_rgb,
>+                            int width) = ARGBToARGB1555Row_C;
>   if (!src_argb || !dst_argb1555 || width <= 0 || height == 0) {
>     return -1;
>   }
>@@ -1244,15 +1169,15 @@ int ARGBToARGB1555(const uint8* src_argb,
> 
> // Convert ARGB To ARGB4444.
> LIBYUV_API
>-int ARGBToARGB4444(const uint8* src_argb,
>+int ARGBToARGB4444(const uint8_t* src_argb,
>                    int src_stride_argb,
>-                   uint8* dst_argb4444,
>+                   uint8_t* dst_argb4444,
>                    int dst_stride_argb4444,
>                    int width,
>                    int height) {
>   int y;
>-  void (*ARGBToARGB4444Row)(const uint8* src_argb, uint8* dst_rgb, int width) =
>-      ARGBToARGB4444Row_C;
>+  void (*ARGBToARGB4444Row)(const uint8_t* src_argb, uint8_t* dst_rgb,
>+                            int width) = ARGBToARGB4444Row_C;
>   if (!src_argb || !dst_argb4444 || width <= 0 || height == 0) {
>     return -1;
>   }
>@@ -1308,22 +1233,121 @@ int ARGBToARGB4444(const uint8* src_argb,
>   return 0;
> }
> 
>+// Convert ABGR To AR30.
>+LIBYUV_API
>+int ABGRToAR30(const uint8_t* src_abgr,
>+               int src_stride_abgr,
>+               uint8_t* dst_ar30,
>+               int dst_stride_ar30,
>+               int width,
>+               int height) {
>+  int y;
>+  void (*ABGRToAR30Row)(const uint8_t* src_abgr, uint8_t* dst_rgb, int width) =
>+      ABGRToAR30Row_C;
>+  if (!src_abgr || !dst_ar30 || width <= 0 || height == 0) {
>+    return -1;
>+  }
>+  if (height < 0) {
>+    height = -height;
>+    src_abgr = src_abgr + (height - 1) * src_stride_abgr;
>+    src_stride_abgr = -src_stride_abgr;
>+  }
>+  // Coalesce rows.
>+  if (src_stride_abgr == width * 4 && dst_stride_ar30 == width * 4) {
>+    width *= height;
>+    height = 1;
>+    src_stride_abgr = dst_stride_ar30 = 0;
>+  }
>+#if defined(HAS_ABGRTOAR30ROW_SSSE3)
>+  if (TestCpuFlag(kCpuHasSSSE3)) {
>+    ABGRToAR30Row = ABGRToAR30Row_Any_SSSE3;
>+    if (IS_ALIGNED(width, 4)) {
>+      ABGRToAR30Row = ABGRToAR30Row_SSSE3;
>+    }
>+  }
>+#endif
>+#if defined(HAS_ABGRTOAR30ROW_AVX2)
>+  if (TestCpuFlag(kCpuHasAVX2)) {
>+    ABGRToAR30Row = ABGRToAR30Row_Any_AVX2;
>+    if (IS_ALIGNED(width, 8)) {
>+      ABGRToAR30Row = ABGRToAR30Row_AVX2;
>+    }
>+  }
>+#endif
>+  for (y = 0; y < height; ++y) {
>+    ABGRToAR30Row(src_abgr, dst_ar30, width);
>+    src_abgr += src_stride_abgr;
>+    dst_ar30 += dst_stride_ar30;
>+  }
>+  return 0;
>+}
>+
>+// Convert ARGB To AR30.
>+LIBYUV_API
>+int ARGBToAR30(const uint8_t* src_argb,
>+               int src_stride_argb,
>+               uint8_t* dst_ar30,
>+               int dst_stride_ar30,
>+               int width,
>+               int height) {
>+  int y;
>+  void (*ARGBToAR30Row)(const uint8_t* src_argb, uint8_t* dst_rgb, int width) =
>+      ARGBToAR30Row_C;
>+  if (!src_argb || !dst_ar30 || width <= 0 || height == 0) {
>+    return -1;
>+  }
>+  if (height < 0) {
>+    height = -height;
>+    src_argb = src_argb + (height - 1) * src_stride_argb;
>+    src_stride_argb = -src_stride_argb;
>+  }
>+  // Coalesce rows.
>+  if (src_stride_argb == width * 4 && dst_stride_ar30 == width * 4) {
>+    width *= height;
>+    height = 1;
>+    src_stride_argb = dst_stride_ar30 = 0;
>+  }
>+#if defined(HAS_ARGBTOAR30ROW_SSSE3)
>+  if (TestCpuFlag(kCpuHasSSSE3)) {
>+    ARGBToAR30Row = ARGBToAR30Row_Any_SSSE3;
>+    if (IS_ALIGNED(width, 4)) {
>+      ARGBToAR30Row = ARGBToAR30Row_SSSE3;
>+    }
>+  }
>+#endif
>+#if defined(HAS_ARGBTOAR30ROW_AVX2)
>+  if (TestCpuFlag(kCpuHasAVX2)) {
>+    ARGBToAR30Row = ARGBToAR30Row_Any_AVX2;
>+    if (IS_ALIGNED(width, 8)) {
>+      ARGBToAR30Row = ARGBToAR30Row_AVX2;
>+    }
>+  }
>+#endif
>+  for (y = 0; y < height; ++y) {
>+    ARGBToAR30Row(src_argb, dst_ar30, width);
>+    src_argb += src_stride_argb;
>+    dst_ar30 += dst_stride_ar30;
>+  }
>+  return 0;
>+}
>+
> // Convert ARGB to J420. (JPeg full range I420).
> LIBYUV_API
>-int ARGBToJ420(const uint8* src_argb,
>+int ARGBToJ420(const uint8_t* src_argb,
>                int src_stride_argb,
>-               uint8* dst_yj,
>+               uint8_t* dst_yj,
>                int dst_stride_yj,
>-               uint8* dst_u,
>+               uint8_t* dst_u,
>                int dst_stride_u,
>-               uint8* dst_v,
>+               uint8_t* dst_v,
>                int dst_stride_v,
>                int width,
>                int height) {
>   int y;
>-  void (*ARGBToUVJRow)(const uint8* src_argb0, int src_stride_argb,
>-                       uint8* dst_u, uint8* dst_v, int width) = ARGBToUVJRow_C;
>-  void (*ARGBToYJRow)(const uint8* src_argb, uint8* dst_yj, int width) =
>+  void (*ARGBToUVJRow)(const uint8_t* src_argb0, int src_stride_argb,
>+                       uint8_t* dst_u, uint8_t* dst_v, int width) =
>+      ARGBToUVJRow_C;
>+  void (*ARGBToYJRow)(const uint8_t* src_argb, uint8_t* dst_yj, int width) =
>       ARGBToYJRow_C;
>   if (!src_argb || !dst_yj || !dst_u || !dst_v || width <= 0 || height == 0) {
>     return -1;
>@@ -1403,20 +1427,21 @@ int ARGBToJ420(const uint8* src_argb,
> 
> // Convert ARGB to J422. (JPeg full range I422).
> LIBYUV_API
>-int ARGBToJ422(const uint8* src_argb,
>+int ARGBToJ422(const uint8_t* src_argb,
>                int src_stride_argb,
>-               uint8* dst_yj,
>+               uint8_t* dst_yj,
>                int dst_stride_yj,
>-               uint8* dst_u,
>+               uint8_t* dst_u,
>                int dst_stride_u,
>-               uint8* dst_v,
>+               uint8_t* dst_v,
>                int dst_stride_v,
>                int width,
>                int height) {
>   int y;
>-  void (*ARGBToUVJRow)(const uint8* src_argb0, int src_stride_argb,
>-                       uint8* dst_u, uint8* dst_v, int width) = ARGBToUVJRow_C;
>-  void (*ARGBToYJRow)(const uint8* src_argb, uint8* dst_yj, int width) =
>+  void (*ARGBToUVJRow)(const uint8_t* src_argb0, int src_stride_argb,
>+                       uint8_t* dst_u, uint8_t* dst_v, int width) =
>+      ARGBToUVJRow_C;
>+  void (*ARGBToYJRow)(const uint8_t* src_argb, uint8_t* dst_yj, int width) =
>       ARGBToYJRow_C;
>   if (!src_argb || !dst_yj || !dst_u || !dst_v || width <= 0 || height == 0) {
>     return -1;
>@@ -1498,14 +1523,14 @@ int ARGBToJ422(const uint8* src_argb,
> 
> // Convert ARGB to J400.
> LIBYUV_API
>-int ARGBToJ400(const uint8* src_argb,
>+int ARGBToJ400(const uint8_t* src_argb,
>                int src_stride_argb,
>-               uint8* dst_yj,
>+               uint8_t* dst_yj,
>                int dst_stride_yj,
>                int width,
>                int height) {
>   int y;
>-  void (*ARGBToYJRow)(const uint8* src_argb, uint8* dst_yj, int width) =
>+  void (*ARGBToYJRow)(const uint8_t* src_argb, uint8_t* dst_yj, int width) =
>       ARGBToYJRow_C;
>   if (!src_argb || !dst_yj || width <= 0 || height == 0) {
>     return -1;
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/convert_jpeg.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/convert_jpeg.cc
>index 216a9f26d87..ae3cc18cd24 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/convert_jpeg.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/convert_jpeg.cc
>@@ -22,18 +22,18 @@ extern "C" {
> 
> #ifdef HAVE_JPEG
> struct I420Buffers {
>-  uint8* y;
>+  uint8_t* y;
>   int y_stride;
>-  uint8* u;
>+  uint8_t* u;
>   int u_stride;
>-  uint8* v;
>+  uint8_t* v;
>   int v_stride;
>   int w;
>   int h;
> };
> 
> static void JpegCopyI420(void* opaque,
>-                         const uint8* const* data,
>+                         const uint8_t* const* data,
>                          const int* strides,
>                          int rows) {
>   I420Buffers* dest = (I420Buffers*)(opaque);
>@@ -47,7 +47,7 @@ static void JpegCopyI420(void* opaque,
> }
> 
> static void JpegI422ToI420(void* opaque,
>-                           const uint8* const* data,
>+                           const uint8_t* const* data,
>                            const int* strides,
>                            int rows) {
>   I420Buffers* dest = (I420Buffers*)(opaque);
>@@ -61,7 +61,7 @@ static void JpegI422ToI420(void* opaque,
> }
> 
> static void JpegI444ToI420(void* opaque,
>-                           const uint8* const* data,
>+                           const uint8_t* const* data,
>                            const int* strides,
>                            int rows) {
>   I420Buffers* dest = (I420Buffers*)(opaque);
>@@ -75,7 +75,7 @@ static void JpegI444ToI420(void* opaque,
> }
> 
> static void JpegI400ToI420(void* opaque,
>-                           const uint8* const* data,
>+                           const uint8_t* const* data,
>                            const int* strides,
>                            int rows) {
>   I420Buffers* dest = (I420Buffers*)(opaque);
>@@ -89,7 +89,10 @@ static void JpegI400ToI420(void* opaque,
> 
> // Query size of MJPG in pixels.
> LIBYUV_API
>-int MJPGSize(const uint8* sample, size_t sample_size, int* width, int* height) {
>+int MJPGSize(const uint8_t* sample,
>+             size_t sample_size,
>+             int* width,
>+             int* height) {
>   MJpegDecoder mjpeg_decoder;
>   LIBYUV_BOOL ret = mjpeg_decoder.LoadFrame(sample, sample_size);
>   if (ret) {
>@@ -101,20 +104,21 @@ int MJPGSize(const uint8* sample, size_t sample_size, int* width, int* height) {
> }
> 
> // MJPG (Motion JPeg) to I420
>-// TODO(fbarchard): review w and h requirement. dw and dh may be enough.
>+// TODO(fbarchard): review src_width and src_height requirement. dst_width and
>+// dst_height may be enough.
> LIBYUV_API
>-int MJPGToI420(const uint8* sample,
>+int MJPGToI420(const uint8_t* sample,
>                size_t sample_size,
>-               uint8* y,
>-               int y_stride,
>-               uint8* u,
>-               int u_stride,
>-               uint8* v,
>-               int v_stride,
>-               int w,
>-               int h,
>-               int dw,
>-               int dh) {
>+               uint8_t* dst_y,
>+               int dst_stride_y,
>+               uint8_t* dst_u,
>+               int dst_stride_u,
>+               uint8_t* dst_v,
>+               int dst_stride_v,
>+               int src_width,
>+               int src_height,
>+               int dst_width,
>+               int dst_height) {
>   if (sample_size == kUnknownDataSize) {
>     // ERROR: MJPEG frame size unknown
>     return -1;
>@@ -123,14 +127,15 @@ int MJPGToI420(const uint8* sample,
>   // TODO(fbarchard): Port MJpeg to C.
>   MJpegDecoder mjpeg_decoder;
>   LIBYUV_BOOL ret = mjpeg_decoder.LoadFrame(sample, sample_size);
>-  if (ret &&
>-      (mjpeg_decoder.GetWidth() != w || mjpeg_decoder.GetHeight() != h)) {
>+  if (ret && (mjpeg_decoder.GetWidth() != src_width ||
>+              mjpeg_decoder.GetHeight() != src_height)) {
>     // ERROR: MJPEG frame has unexpected dimensions
>     mjpeg_decoder.UnloadFrame();
>     return 1;  // runtime failure
>   }
>   if (ret) {
>-    I420Buffers bufs = {y, y_stride, u, u_stride, v, v_stride, dw, dh};
>+    I420Buffers bufs = {dst_y, dst_stride_y, dst_u,     dst_stride_u,
>+                        dst_v, dst_stride_v, dst_width, dst_height};
>     // YUV420
>     if (mjpeg_decoder.GetColorSpace() == MJpegDecoder::kColorSpaceYCbCr &&
>         mjpeg_decoder.GetNumComponents() == 3 &&
>@@ -140,7 +145,8 @@ int MJPGToI420(const uint8* sample,
>         mjpeg_decoder.GetHorizSampFactor(1) == 1 &&
>         mjpeg_decoder.GetVertSampFactor(2) == 1 &&
>         mjpeg_decoder.GetHorizSampFactor(2) == 1) {
>-      ret = mjpeg_decoder.DecodeToCallback(&JpegCopyI420, &bufs, dw, dh);
>+      ret = mjpeg_decoder.DecodeToCallback(&JpegCopyI420, &bufs, dst_width,
>+                                           dst_height);
>       // YUV422
>     } else if (mjpeg_decoder.GetColorSpace() ==
>                    MJpegDecoder::kColorSpaceYCbCr &&
>@@ -151,7 +157,8 @@ int MJPGToI420(const uint8* sample,
>                mjpeg_decoder.GetHorizSampFactor(1) == 1 &&
>                mjpeg_decoder.GetVertSampFactor(2) == 1 &&
>                mjpeg_decoder.GetHorizSampFactor(2) == 1) {
>-      ret = mjpeg_decoder.DecodeToCallback(&JpegI422ToI420, &bufs, dw, dh);
>+      ret = mjpeg_decoder.DecodeToCallback(&JpegI422ToI420, &bufs, dst_width,
>+                                           dst_height);
>       // YUV444
>     } else if (mjpeg_decoder.GetColorSpace() ==
>                    MJpegDecoder::kColorSpaceYCbCr &&
>@@ -162,14 +169,16 @@ int MJPGToI420(const uint8* sample,
>                mjpeg_decoder.GetHorizSampFactor(1) == 1 &&
>                mjpeg_decoder.GetVertSampFactor(2) == 1 &&
>                mjpeg_decoder.GetHorizSampFactor(2) == 1) {
>-      ret = mjpeg_decoder.DecodeToCallback(&JpegI444ToI420, &bufs, dw, dh);
>+      ret = mjpeg_decoder.DecodeToCallback(&JpegI444ToI420, &bufs, dst_width,
>+                                           dst_height);
>       // YUV400
>     } else if (mjpeg_decoder.GetColorSpace() ==
>                    MJpegDecoder::kColorSpaceGrayscale &&
>                mjpeg_decoder.GetNumComponents() == 1 &&
>                mjpeg_decoder.GetVertSampFactor(0) == 1 &&
>                mjpeg_decoder.GetHorizSampFactor(0) == 1) {
>-      ret = mjpeg_decoder.DecodeToCallback(&JpegI400ToI420, &bufs, dw, dh);
>+      ret = mjpeg_decoder.DecodeToCallback(&JpegI400ToI420, &bufs, dst_width,
>+                                           dst_height);
>     } else {
>       // TODO(fbarchard): Implement conversion for any other colorspace/sample
>       // factors that occur in practice.
>@@ -183,14 +192,14 @@ int MJPGToI420(const uint8* sample,
> 
> #ifdef HAVE_JPEG
> struct ARGBBuffers {
>-  uint8* argb;
>+  uint8_t* argb;
>   int argb_stride;
>   int w;
>   int h;
> };
> 
> static void JpegI420ToARGB(void* opaque,
>-                           const uint8* const* data,
>+                           const uint8_t* const* data,
>                            const int* strides,
>                            int rows) {
>   ARGBBuffers* dest = (ARGBBuffers*)(opaque);
>@@ -201,7 +210,7 @@ static void JpegI420ToARGB(void* opaque,
> }
> 
> static void JpegI422ToARGB(void* opaque,
>-                           const uint8* const* data,
>+                           const uint8_t* const* data,
>                            const int* strides,
>                            int rows) {
>   ARGBBuffers* dest = (ARGBBuffers*)(opaque);
>@@ -212,7 +221,7 @@ static void JpegI422ToARGB(void* opaque,
> }
> 
> static void JpegI444ToARGB(void* opaque,
>-                           const uint8* const* data,
>+                           const uint8_t* const* data,
>                            const int* strides,
>                            int rows) {
>   ARGBBuffers* dest = (ARGBBuffers*)(opaque);
>@@ -223,7 +232,7 @@ static void JpegI444ToARGB(void* opaque,
> }
> 
> static void JpegI400ToARGB(void* opaque,
>-                           const uint8* const* data,
>+                           const uint8_t* const* data,
>                            const int* strides,
>                            int rows) {
>   ARGBBuffers* dest = (ARGBBuffers*)(opaque);
>@@ -233,16 +242,17 @@ static void JpegI400ToARGB(void* opaque,
> }
> 
> // MJPG (Motion JPeg) to ARGB
>-// TODO(fbarchard): review w and h requirement. dw and dh may be enough.
>+// TODO(fbarchard): review src_width and src_height requirement. dst_width and
>+// dst_height may be enough.
> LIBYUV_API
>-int MJPGToARGB(const uint8* sample,
>+int MJPGToARGB(const uint8_t* sample,
>                size_t sample_size,
>-               uint8* argb,
>-               int argb_stride,
>-               int w,
>-               int h,
>-               int dw,
>-               int dh) {
>+               uint8_t* dst_argb,
>+               int dst_stride_argb,
>+               int src_width,
>+               int src_height,
>+               int dst_width,
>+               int dst_height) {
>   if (sample_size == kUnknownDataSize) {
>     // ERROR: MJPEG frame size unknown
>     return -1;
>@@ -251,14 +261,14 @@ int MJPGToARGB(const uint8* sample,
>   // TODO(fbarchard): Port MJpeg to C.
>   MJpegDecoder mjpeg_decoder;
>   LIBYUV_BOOL ret = mjpeg_decoder.LoadFrame(sample, sample_size);
>-  if (ret &&
>-      (mjpeg_decoder.GetWidth() != w || mjpeg_decoder.GetHeight() != h)) {
>+  if (ret && (mjpeg_decoder.GetWidth() != src_width ||
>+              mjpeg_decoder.GetHeight() != src_height)) {
>     // ERROR: MJPEG frame has unexpected dimensions
>     mjpeg_decoder.UnloadFrame();
>     return 1;  // runtime failure
>   }
>   if (ret) {
>-    ARGBBuffers bufs = {argb, argb_stride, dw, dh};
>+    ARGBBuffers bufs = {dst_argb, dst_stride_argb, dst_width, dst_height};
>     // YUV420
>     if (mjpeg_decoder.GetColorSpace() == MJpegDecoder::kColorSpaceYCbCr &&
>         mjpeg_decoder.GetNumComponents() == 3 &&
>@@ -268,7 +278,8 @@ int MJPGToARGB(const uint8* sample,
>         mjpeg_decoder.GetHorizSampFactor(1) == 1 &&
>         mjpeg_decoder.GetVertSampFactor(2) == 1 &&
>         mjpeg_decoder.GetHorizSampFactor(2) == 1) {
>-      ret = mjpeg_decoder.DecodeToCallback(&JpegI420ToARGB, &bufs, dw, dh);
>+      ret = mjpeg_decoder.DecodeToCallback(&JpegI420ToARGB, &bufs, dst_width,
>+                                           dst_height);
>       // YUV422
>     } else if (mjpeg_decoder.GetColorSpace() ==
>                    MJpegDecoder::kColorSpaceYCbCr &&
>@@ -279,7 +290,8 @@ int MJPGToARGB(const uint8* sample,
>                mjpeg_decoder.GetHorizSampFactor(1) == 1 &&
>                mjpeg_decoder.GetVertSampFactor(2) == 1 &&
>                mjpeg_decoder.GetHorizSampFactor(2) == 1) {
>-      ret = mjpeg_decoder.DecodeToCallback(&JpegI422ToARGB, &bufs, dw, dh);
>+      ret = mjpeg_decoder.DecodeToCallback(&JpegI422ToARGB, &bufs, dst_width,
>+                                           dst_height);
>       // YUV444
>     } else if (mjpeg_decoder.GetColorSpace() ==
>                    MJpegDecoder::kColorSpaceYCbCr &&
>@@ -290,14 +302,16 @@ int MJPGToARGB(const uint8* sample,
>                mjpeg_decoder.GetHorizSampFactor(1) == 1 &&
>                mjpeg_decoder.GetVertSampFactor(2) == 1 &&
>                mjpeg_decoder.GetHorizSampFactor(2) == 1) {
>-      ret = mjpeg_decoder.DecodeToCallback(&JpegI444ToARGB, &bufs, dw, dh);
>+      ret = mjpeg_decoder.DecodeToCallback(&JpegI444ToARGB, &bufs, dst_width,
>+                                           dst_height);
>       // YUV400
>     } else if (mjpeg_decoder.GetColorSpace() ==
>                    MJpegDecoder::kColorSpaceGrayscale &&
>                mjpeg_decoder.GetNumComponents() == 1 &&
>                mjpeg_decoder.GetVertSampFactor(0) == 1 &&
>                mjpeg_decoder.GetHorizSampFactor(0) == 1) {
>-      ret = mjpeg_decoder.DecodeToCallback(&JpegI400ToARGB, &bufs, dw, dh);
>+      ret = mjpeg_decoder.DecodeToCallback(&JpegI400ToARGB, &bufs, dst_width,
>+                                           dst_height);
>     } else {
>       // TODO(fbarchard): Implement conversion for any other colorspace/sample
>       // factors that occur in practice.
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/convert_to_argb.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/convert_to_argb.cc
>index 63a5104b3c7..677e5d56fcc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/convert_to_argb.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/convert_to_argb.cc
>@@ -29,10 +29,10 @@ extern "C" {
> // sample_size is measured in bytes and is the size of the frame.
> //   With MJPEG it is the compressed size of the frame.
> LIBYUV_API
>-int ConvertToARGB(const uint8* sample,
>+int ConvertToARGB(const uint8_t* sample,
>                   size_t sample_size,
>-                  uint8* crop_argb,
>-                  int argb_stride,
>+                  uint8_t* dst_argb,
>+                  int dst_stride_argb,
>                   int crop_x,
>                   int crop_y,
>                   int src_width,
>@@ -40,11 +40,11 @@ int ConvertToARGB(const uint8* sample,
>                   int crop_width,
>                   int crop_height,
>                   enum RotationMode rotation,
>-                  uint32 fourcc) {
>-  uint32 format = CanonicalFourCC(fourcc);
>+                  uint32_t fourcc) {
>+  uint32_t format = CanonicalFourCC(fourcc);
>   int aligned_src_width = (src_width + 1) & ~1;
>-  const uint8* src;
>-  const uint8* src_uv;
>+  const uint8_t* src;
>+  const uint8_t* src_uv;
>   int abs_src_height = (src_height < 0) ? -src_height : src_height;
>   int inv_crop_height = (crop_height < 0) ? -crop_height : crop_height;
>   int r = 0;
>@@ -52,17 +52,17 @@ int ConvertToARGB(const uint8* sample,
>   // One pass rotation is available for some formats. For the rest, convert
>   // to ARGB (with optional vertical flipping) into a temporary ARGB buffer,
>   // and then rotate the ARGB to the final destination buffer.
>-  // For in-place conversion, if destination crop_argb is same as source sample,
>+  // For in-place conversion, if destination dst_argb is same as source sample,
>   // also enable temporary buffer.
>   LIBYUV_BOOL need_buf =
>-      (rotation && format != FOURCC_ARGB) || crop_argb == sample;
>-  uint8* dest_argb = crop_argb;
>-  int dest_argb_stride = argb_stride;
>-  uint8* rotate_buffer = NULL;
>+      (rotation && format != FOURCC_ARGB) || dst_argb == sample;
>+  uint8_t* dest_argb = dst_argb;
>+  int dest_dst_stride_argb = dst_stride_argb;
>+  uint8_t* rotate_buffer = NULL;
>   int abs_crop_height = (crop_height < 0) ? -crop_height : crop_height;
> 
>-  if (crop_argb == NULL || sample == NULL || src_width <= 0 ||
>-      crop_width <= 0 || src_height == 0 || crop_height == 0) {
>+  if (dst_argb == NULL || sample == NULL || src_width <= 0 || crop_width <= 0 ||
>+      src_height == 0 || crop_height == 0) {
>     return -1;
>   }
>   if (src_height < 0) {
>@@ -71,76 +71,76 @@ int ConvertToARGB(const uint8* sample,
> 
>   if (need_buf) {
>     int argb_size = crop_width * 4 * abs_crop_height;
>-    rotate_buffer = (uint8*)malloc(argb_size); /* NOLINT */
>+    rotate_buffer = (uint8_t*)malloc(argb_size); /* NOLINT */
>     if (!rotate_buffer) {
>       return 1;  // Out of memory runtime error.
>     }
>-    crop_argb = rotate_buffer;
>-    argb_stride = crop_width * 4;
>+    dst_argb = rotate_buffer;
>+    dst_stride_argb = crop_width * 4;
>   }
> 
>   switch (format) {
>     // Single plane formats
>     case FOURCC_YUY2:
>       src = sample + (aligned_src_width * crop_y + crop_x) * 2;
>-      r = YUY2ToARGB(src, aligned_src_width * 2, crop_argb, argb_stride,
>+      r = YUY2ToARGB(src, aligned_src_width * 2, dst_argb, dst_stride_argb,
>                      crop_width, inv_crop_height);
>       break;
>     case FOURCC_UYVY:
>       src = sample + (aligned_src_width * crop_y + crop_x) * 2;
>-      r = UYVYToARGB(src, aligned_src_width * 2, crop_argb, argb_stride,
>+      r = UYVYToARGB(src, aligned_src_width * 2, dst_argb, dst_stride_argb,
>                      crop_width, inv_crop_height);
>       break;
>     case FOURCC_24BG:
>       src = sample + (src_width * crop_y + crop_x) * 3;
>-      r = RGB24ToARGB(src, src_width * 3, crop_argb, argb_stride, crop_width,
>+      r = RGB24ToARGB(src, src_width * 3, dst_argb, dst_stride_argb, crop_width,
>                       inv_crop_height);
>       break;
>     case FOURCC_RAW:
>       src = sample + (src_width * crop_y + crop_x) * 3;
>-      r = RAWToARGB(src, src_width * 3, crop_argb, argb_stride, crop_width,
>+      r = RAWToARGB(src, src_width * 3, dst_argb, dst_stride_argb, crop_width,
>                     inv_crop_height);
>       break;
>     case FOURCC_ARGB:
>       if (!need_buf && !rotation) {
>         src = sample + (src_width * crop_y + crop_x) * 4;
>-        r = ARGBToARGB(src, src_width * 4, crop_argb, argb_stride, crop_width,
>-                       inv_crop_height);
>+        r = ARGBToARGB(src, src_width * 4, dst_argb, dst_stride_argb,
>+                       crop_width, inv_crop_height);
>       }
>       break;
>     case FOURCC_BGRA:
>       src = sample + (src_width * crop_y + crop_x) * 4;
>-      r = BGRAToARGB(src, src_width * 4, crop_argb, argb_stride, crop_width,
>+      r = BGRAToARGB(src, src_width * 4, dst_argb, dst_stride_argb, crop_width,
>                      inv_crop_height);
>       break;
>     case FOURCC_ABGR:
>       src = sample + (src_width * crop_y + crop_x) * 4;
>-      r = ABGRToARGB(src, src_width * 4, crop_argb, argb_stride, crop_width,
>+      r = ABGRToARGB(src, src_width * 4, dst_argb, dst_stride_argb, crop_width,
>                      inv_crop_height);
>       break;
>     case FOURCC_RGBA:
>       src = sample + (src_width * crop_y + crop_x) * 4;
>-      r = RGBAToARGB(src, src_width * 4, crop_argb, argb_stride, crop_width,
>+      r = RGBAToARGB(src, src_width * 4, dst_argb, dst_stride_argb, crop_width,
>                      inv_crop_height);
>       break;
>     case FOURCC_RGBP:
>       src = sample + (src_width * crop_y + crop_x) * 2;
>-      r = RGB565ToARGB(src, src_width * 2, crop_argb, argb_stride, crop_width,
>-                       inv_crop_height);
>+      r = RGB565ToARGB(src, src_width * 2, dst_argb, dst_stride_argb,
>+                       crop_width, inv_crop_height);
>       break;
>     case FOURCC_RGBO:
>       src = sample + (src_width * crop_y + crop_x) * 2;
>-      r = ARGB1555ToARGB(src, src_width * 2, crop_argb, argb_stride, crop_width,
>-                         inv_crop_height);
>+      r = ARGB1555ToARGB(src, src_width * 2, dst_argb, dst_stride_argb,
>+                         crop_width, inv_crop_height);
>       break;
>     case FOURCC_R444:
>       src = sample + (src_width * crop_y + crop_x) * 2;
>-      r = ARGB4444ToARGB(src, src_width * 2, crop_argb, argb_stride, crop_width,
>-                         inv_crop_height);
>+      r = ARGB4444ToARGB(src, src_width * 2, dst_argb, dst_stride_argb,
>+                         crop_width, inv_crop_height);
>       break;
>     case FOURCC_I400:
>       src = sample + src_width * crop_y + crop_x;
>-      r = I400ToARGB(src, src_width, crop_argb, argb_stride, crop_width,
>+      r = I400ToARGB(src, src_width, dst_argb, dst_stride_argb, crop_width,
>                      inv_crop_height);
>       break;
> 
>@@ -148,27 +148,27 @@ int ConvertToARGB(const uint8* sample,
>     case FOURCC_NV12:
>       src = sample + (src_width * crop_y + crop_x);
>       src_uv = sample + aligned_src_width * (src_height + crop_y / 2) + crop_x;
>-      r = NV12ToARGB(src, src_width, src_uv, aligned_src_width, crop_argb,
>-                     argb_stride, crop_width, inv_crop_height);
>+      r = NV12ToARGB(src, src_width, src_uv, aligned_src_width, dst_argb,
>+                     dst_stride_argb, crop_width, inv_crop_height);
>       break;
>     case FOURCC_NV21:
>       src = sample + (src_width * crop_y + crop_x);
>       src_uv = sample + aligned_src_width * (src_height + crop_y / 2) + crop_x;
>       // Call NV12 but with u and v parameters swapped.
>-      r = NV21ToARGB(src, src_width, src_uv, aligned_src_width, crop_argb,
>-                     argb_stride, crop_width, inv_crop_height);
>+      r = NV21ToARGB(src, src_width, src_uv, aligned_src_width, dst_argb,
>+                     dst_stride_argb, crop_width, inv_crop_height);
>       break;
>     case FOURCC_M420:
>       src = sample + (src_width * crop_y) * 12 / 8 + crop_x;
>-      r = M420ToARGB(src, src_width, crop_argb, argb_stride, crop_width,
>+      r = M420ToARGB(src, src_width, dst_argb, dst_stride_argb, crop_width,
>                      inv_crop_height);
>       break;
>     // Triplanar formats
>     case FOURCC_I420:
>     case FOURCC_YV12: {
>-      const uint8* src_y = sample + (src_width * crop_y + crop_x);
>-      const uint8* src_u;
>-      const uint8* src_v;
>+      const uint8_t* src_y = sample + (src_width * crop_y + crop_x);
>+      const uint8_t* src_u;
>+      const uint8_t* src_v;
>       int halfwidth = (src_width + 1) / 2;
>       int halfheight = (abs_src_height + 1) / 2;
>       if (format == FOURCC_YV12) {
>@@ -183,14 +183,14 @@ int ConvertToARGB(const uint8* sample,
>                 halfwidth * (halfheight + crop_y / 2) + crop_x / 2;
>       }
>       r = I420ToARGB(src_y, src_width, src_u, halfwidth, src_v, halfwidth,
>-                     crop_argb, argb_stride, crop_width, inv_crop_height);
>+                     dst_argb, dst_stride_argb, crop_width, inv_crop_height);
>       break;
>     }
> 
>     case FOURCC_J420: {
>-      const uint8* src_y = sample + (src_width * crop_y + crop_x);
>-      const uint8* src_u;
>-      const uint8* src_v;
>+      const uint8_t* src_y = sample + (src_width * crop_y + crop_x);
>+      const uint8_t* src_u;
>+      const uint8_t* src_v;
>       int halfwidth = (src_width + 1) / 2;
>       int halfheight = (abs_src_height + 1) / 2;
>       src_u = sample + src_width * abs_src_height +
>@@ -198,15 +198,15 @@ int ConvertToARGB(const uint8* sample,
>       src_v = sample + src_width * abs_src_height +
>               halfwidth * (halfheight + crop_y / 2) + crop_x / 2;
>       r = J420ToARGB(src_y, src_width, src_u, halfwidth, src_v, halfwidth,
>-                     crop_argb, argb_stride, crop_width, inv_crop_height);
>+                     dst_argb, dst_stride_argb, crop_width, inv_crop_height);
>       break;
>     }
> 
>     case FOURCC_I422:
>     case FOURCC_YV16: {
>-      const uint8* src_y = sample + src_width * crop_y + crop_x;
>-      const uint8* src_u;
>-      const uint8* src_v;
>+      const uint8_t* src_y = sample + src_width * crop_y + crop_x;
>+      const uint8_t* src_u;
>+      const uint8_t* src_v;
>       int halfwidth = (src_width + 1) / 2;
>       if (format == FOURCC_YV16) {
>         src_v = sample + src_width * abs_src_height + halfwidth * crop_y +
>@@ -220,14 +220,14 @@ int ConvertToARGB(const uint8* sample,
>                 halfwidth * (abs_src_height + crop_y) + crop_x / 2;
>       }
>       r = I422ToARGB(src_y, src_width, src_u, halfwidth, src_v, halfwidth,
>-                     crop_argb, argb_stride, crop_width, inv_crop_height);
>+                     dst_argb, dst_stride_argb, crop_width, inv_crop_height);
>       break;
>     }
>     case FOURCC_I444:
>     case FOURCC_YV24: {
>-      const uint8* src_y = sample + src_width * crop_y + crop_x;
>-      const uint8* src_u;
>-      const uint8* src_v;
>+      const uint8_t* src_y = sample + src_width * crop_y + crop_x;
>+      const uint8_t* src_u;
>+      const uint8_t* src_v;
>       if (format == FOURCC_YV24) {
>         src_v = sample + src_width * (abs_src_height + crop_y) + crop_x;
>         src_u = sample + src_width * (abs_src_height * 2 + crop_y) + crop_x;
>@@ -236,12 +236,12 @@ int ConvertToARGB(const uint8* sample,
>         src_v = sample + src_width * (abs_src_height * 2 + crop_y) + crop_x;
>       }
>       r = I444ToARGB(src_y, src_width, src_u, src_width, src_v, src_width,
>-                     crop_argb, argb_stride, crop_width, inv_crop_height);
>+                     dst_argb, dst_stride_argb, crop_width, inv_crop_height);
>       break;
>     }
> #ifdef HAVE_JPEG
>     case FOURCC_MJPG:
>-      r = MJPGToARGB(sample, sample_size, crop_argb, argb_stride, src_width,
>+      r = MJPGToARGB(sample, sample_size, dst_argb, dst_stride_argb, src_width,
>                      abs_src_height, crop_width, inv_crop_height);
>       break;
> #endif
>@@ -251,13 +251,13 @@ int ConvertToARGB(const uint8* sample,
> 
>   if (need_buf) {
>     if (!r) {
>-      r = ARGBRotate(crop_argb, argb_stride, dest_argb, dest_argb_stride,
>+      r = ARGBRotate(dst_argb, dst_stride_argb, dest_argb, dest_dst_stride_argb,
>                      crop_width, abs_crop_height, rotation);
>     }
>     free(rotate_buffer);
>   } else if (rotation) {
>     src = sample + (src_width * crop_y + crop_x) * 4;
>-    r = ARGBRotate(src, src_width * 4, crop_argb, argb_stride, crop_width,
>+    r = ARGBRotate(src, src_width * 4, dst_argb, dst_stride_argb, crop_width,
>                    inv_crop_height, rotation);
>   }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/convert_to_i420.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/convert_to_i420.cc
>index a50689db949..1bed9d6440d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/convert_to_i420.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/convert_to_i420.cc
>@@ -25,14 +25,14 @@ extern "C" {
> // sample_size is measured in bytes and is the size of the frame.
> //   With MJPEG it is the compressed size of the frame.
> LIBYUV_API
>-int ConvertToI420(const uint8* sample,
>+int ConvertToI420(const uint8_t* sample,
>                   size_t sample_size,
>-                  uint8* y,
>-                  int y_stride,
>-                  uint8* u,
>-                  int u_stride,
>-                  uint8* v,
>-                  int v_stride,
>+                  uint8_t* dst_y,
>+                  int dst_stride_y,
>+                  uint8_t* dst_u,
>+                  int dst_stride_u,
>+                  uint8_t* dst_v,
>+                  int dst_stride_v,
>                   int crop_x,
>                   int crop_y,
>                   int src_width,
>@@ -40,11 +40,11 @@ int ConvertToI420(const uint8* sample,
>                   int crop_width,
>                   int crop_height,
>                   enum RotationMode rotation,
>-                  uint32 fourcc) {
>-  uint32 format = CanonicalFourCC(fourcc);
>+                  uint32_t fourcc) {
>+  uint32_t format = CanonicalFourCC(fourcc);
>   int aligned_src_width = (src_width + 1) & ~1;
>-  const uint8* src;
>-  const uint8* src_uv;
>+  const uint8_t* src;
>+  const uint8_t* src_uv;
>   const int abs_src_height = (src_height < 0) ? -src_height : src_height;
>   // TODO(nisse): Why allow crop_height < 0?
>   const int abs_crop_height = (crop_height < 0) ? -crop_height : crop_height;
>@@ -52,132 +52,143 @@ int ConvertToI420(const uint8* sample,
>   LIBYUV_BOOL need_buf =
>       (rotation && format != FOURCC_I420 && format != FOURCC_NV12 &&
>        format != FOURCC_NV21 && format != FOURCC_YV12) ||
>-      y == sample;
>-  uint8* tmp_y = y;
>-  uint8* tmp_u = u;
>-  uint8* tmp_v = v;
>-  int tmp_y_stride = y_stride;
>-  int tmp_u_stride = u_stride;
>-  int tmp_v_stride = v_stride;
>-  uint8* rotate_buffer = NULL;
>+      dst_y == sample;
>+  uint8_t* tmp_y = dst_y;
>+  uint8_t* tmp_u = dst_u;
>+  uint8_t* tmp_v = dst_v;
>+  int tmp_y_stride = dst_stride_y;
>+  int tmp_u_stride = dst_stride_u;
>+  int tmp_v_stride = dst_stride_v;
>+  uint8_t* rotate_buffer = NULL;
>   const int inv_crop_height =
>       (src_height < 0) ? -abs_crop_height : abs_crop_height;
> 
>-  if (!y || !u || !v || !sample || src_width <= 0 || crop_width <= 0 ||
>-      src_height == 0 || crop_height == 0) {
>+  if (!dst_y || !dst_u || !dst_v || !sample || src_width <= 0 ||
>+      crop_width <= 0 || src_height == 0 || crop_height == 0) {
>     return -1;
>   }
> 
>   // One pass rotation is available for some formats. For the rest, convert
>   // to I420 (with optional vertical flipping) into a temporary I420 buffer,
>   // and then rotate the I420 to the final destination buffer.
>-  // For in-place conversion, if destination y is same as source sample,
>+  // For in-place conversion, if destination dst_y is same as source sample,
>   // also enable temporary buffer.
>   if (need_buf) {
>     int y_size = crop_width * abs_crop_height;
>     int uv_size = ((crop_width + 1) / 2) * ((abs_crop_height + 1) / 2);
>-    rotate_buffer = (uint8*)malloc(y_size + uv_size * 2); /* NOLINT */
>+    rotate_buffer = (uint8_t*)malloc(y_size + uv_size * 2); /* NOLINT */
>     if (!rotate_buffer) {
>       return 1;  // Out of memory runtime error.
>     }
>-    y = rotate_buffer;
>-    u = y + y_size;
>-    v = u + uv_size;
>-    y_stride = crop_width;
>-    u_stride = v_stride = ((crop_width + 1) / 2);
>+    dst_y = rotate_buffer;
>+    dst_u = dst_y + y_size;
>+    dst_v = dst_u + uv_size;
>+    dst_stride_y = crop_width;
>+    dst_stride_u = dst_stride_v = ((crop_width + 1) / 2);
>   }
> 
>   switch (format) {
>     // Single plane formats
>     case FOURCC_YUY2:
>       src = sample + (aligned_src_width * crop_y + crop_x) * 2;
>-      r = YUY2ToI420(src, aligned_src_width * 2, y, y_stride, u, u_stride, v,
>-                     v_stride, crop_width, inv_crop_height);
>+      r = YUY2ToI420(src, aligned_src_width * 2, dst_y, dst_stride_y, dst_u,
>+                     dst_stride_u, dst_v, dst_stride_v, crop_width,
>+                     inv_crop_height);
>       break;
>     case FOURCC_UYVY:
>       src = sample + (aligned_src_width * crop_y + crop_x) * 2;
>-      r = UYVYToI420(src, aligned_src_width * 2, y, y_stride, u, u_stride, v,
>-                     v_stride, crop_width, inv_crop_height);
>+      r = UYVYToI420(src, aligned_src_width * 2, dst_y, dst_stride_y, dst_u,
>+                     dst_stride_u, dst_v, dst_stride_v, crop_width,
>+                     inv_crop_height);
>       break;
>     case FOURCC_RGBP:
>       src = sample + (src_width * crop_y + crop_x) * 2;
>-      r = RGB565ToI420(src, src_width * 2, y, y_stride, u, u_stride, v,
>-                       v_stride, crop_width, inv_crop_height);
>+      r = RGB565ToI420(src, src_width * 2, dst_y, dst_stride_y, dst_u,
>+                       dst_stride_u, dst_v, dst_stride_v, crop_width,
>+                       inv_crop_height);
>       break;
>     case FOURCC_RGBO:
>       src = sample + (src_width * crop_y + crop_x) * 2;
>-      r = ARGB1555ToI420(src, src_width * 2, y, y_stride, u, u_stride, v,
>-                         v_stride, crop_width, inv_crop_height);
>+      r = ARGB1555ToI420(src, src_width * 2, dst_y, dst_stride_y, dst_u,
>+                         dst_stride_u, dst_v, dst_stride_v, crop_width,
>+                         inv_crop_height);
>       break;
>     case FOURCC_R444:
>       src = sample + (src_width * crop_y + crop_x) * 2;
>-      r = ARGB4444ToI420(src, src_width * 2, y, y_stride, u, u_stride, v,
>-                         v_stride, crop_width, inv_crop_height);
>+      r = ARGB4444ToI420(src, src_width * 2, dst_y, dst_stride_y, dst_u,
>+                         dst_stride_u, dst_v, dst_stride_v, crop_width,
>+                         inv_crop_height);
>       break;
>     case FOURCC_24BG:
>       src = sample + (src_width * crop_y + crop_x) * 3;
>-      r = RGB24ToI420(src, src_width * 3, y, y_stride, u, u_stride, v, v_stride,
>-                      crop_width, inv_crop_height);
>+      r = RGB24ToI420(src, src_width * 3, dst_y, dst_stride_y, dst_u,
>+                      dst_stride_u, dst_v, dst_stride_v, crop_width,
>+                      inv_crop_height);
>       break;
>     case FOURCC_RAW:
>       src = sample + (src_width * crop_y + crop_x) * 3;
>-      r = RAWToI420(src, src_width * 3, y, y_stride, u, u_stride, v, v_stride,
>-                    crop_width, inv_crop_height);
>+      r = RAWToI420(src, src_width * 3, dst_y, dst_stride_y, dst_u,
>+                    dst_stride_u, dst_v, dst_stride_v, crop_width,
>+                    inv_crop_height);
>       break;
>     case FOURCC_ARGB:
>       src = sample + (src_width * crop_y + crop_x) * 4;
>-      r = ARGBToI420(src, src_width * 4, y, y_stride, u, u_stride, v, v_stride,
>-                     crop_width, inv_crop_height);
>+      r = ARGBToI420(src, src_width * 4, dst_y, dst_stride_y, dst_u,
>+                     dst_stride_u, dst_v, dst_stride_v, crop_width,
>+                     inv_crop_height);
>       break;
>     case FOURCC_BGRA:
>       src = sample + (src_width * crop_y + crop_x) * 4;
>-      r = BGRAToI420(src, src_width * 4, y, y_stride, u, u_stride, v, v_stride,
>-                     crop_width, inv_crop_height);
>+      r = BGRAToI420(src, src_width * 4, dst_y, dst_stride_y, dst_u,
>+                     dst_stride_u, dst_v, dst_stride_v, crop_width,
>+                     inv_crop_height);
>       break;
>     case FOURCC_ABGR:
>       src = sample + (src_width * crop_y + crop_x) * 4;
>-      r = ABGRToI420(src, src_width * 4, y, y_stride, u, u_stride, v, v_stride,
>-                     crop_width, inv_crop_height);
>+      r = ABGRToI420(src, src_width * 4, dst_y, dst_stride_y, dst_u,
>+                     dst_stride_u, dst_v, dst_stride_v, crop_width,
>+                     inv_crop_height);
>       break;
>     case FOURCC_RGBA:
>       src = sample + (src_width * crop_y + crop_x) * 4;
>-      r = RGBAToI420(src, src_width * 4, y, y_stride, u, u_stride, v, v_stride,
>-                     crop_width, inv_crop_height);
>+      r = RGBAToI420(src, src_width * 4, dst_y, dst_stride_y, dst_u,
>+                     dst_stride_u, dst_v, dst_stride_v, crop_width,
>+                     inv_crop_height);
>       break;
>     case FOURCC_I400:
>       src = sample + src_width * crop_y + crop_x;
>-      r = I400ToI420(src, src_width, y, y_stride, u, u_stride, v, v_stride,
>-                     crop_width, inv_crop_height);
>+      r = I400ToI420(src, src_width, dst_y, dst_stride_y, dst_u, dst_stride_u,
>+                     dst_v, dst_stride_v, crop_width, inv_crop_height);
>       break;
>     // Biplanar formats
>     case FOURCC_NV12:
>       src = sample + (src_width * crop_y + crop_x);
>       src_uv = sample + (src_width * src_height) +
>                ((crop_y / 2) * aligned_src_width) + ((crop_x / 2) * 2);
>-      r = NV12ToI420Rotate(src, src_width, src_uv, aligned_src_width, y,
>-                           y_stride, u, u_stride, v, v_stride, crop_width,
>-                           inv_crop_height, rotation);
>+      r = NV12ToI420Rotate(src, src_width, src_uv, aligned_src_width, dst_y,
>+                           dst_stride_y, dst_u, dst_stride_u, dst_v,
>+                           dst_stride_v, crop_width, inv_crop_height, rotation);
>       break;
>     case FOURCC_NV21:
>       src = sample + (src_width * crop_y + crop_x);
>       src_uv = sample + (src_width * src_height) +
>                ((crop_y / 2) * aligned_src_width) + ((crop_x / 2) * 2);
>-      // Call NV12 but with u and v parameters swapped.
>-      r = NV12ToI420Rotate(src, src_width, src_uv, aligned_src_width, y,
>-                           y_stride, v, v_stride, u, u_stride, crop_width,
>-                           inv_crop_height, rotation);
>+      // Call NV12 but with dst_u and dst_v parameters swapped.
>+      r = NV12ToI420Rotate(src, src_width, src_uv, aligned_src_width, dst_y,
>+                           dst_stride_y, dst_v, dst_stride_v, dst_u,
>+                           dst_stride_u, crop_width, inv_crop_height, rotation);
>       break;
>     case FOURCC_M420:
>       src = sample + (src_width * crop_y) * 12 / 8 + crop_x;
>-      r = M420ToI420(src, src_width, y, y_stride, u, u_stride, v, v_stride,
>-                     crop_width, inv_crop_height);
>+      r = M420ToI420(src, src_width, dst_y, dst_stride_y, dst_u, dst_stride_u,
>+                     dst_v, dst_stride_v, crop_width, inv_crop_height);
>       break;
>     // Triplanar formats
>     case FOURCC_I420:
>     case FOURCC_YV12: {
>-      const uint8* src_y = sample + (src_width * crop_y + crop_x);
>-      const uint8* src_u;
>-      const uint8* src_v;
>+      const uint8_t* src_y = sample + (src_width * crop_y + crop_x);
>+      const uint8_t* src_u;
>+      const uint8_t* src_v;
>       int halfwidth = (src_width + 1) / 2;
>       int halfheight = (abs_src_height + 1) / 2;
>       if (format == FOURCC_YV12) {
>@@ -191,16 +202,16 @@ int ConvertToI420(const uint8* sample,
>         src_v = sample + src_width * abs_src_height +
>                 halfwidth * (halfheight + crop_y / 2) + crop_x / 2;
>       }
>-      r = I420Rotate(src_y, src_width, src_u, halfwidth, src_v, halfwidth, y,
>-                     y_stride, u, u_stride, v, v_stride, crop_width,
>-                     inv_crop_height, rotation);
>+      r = I420Rotate(src_y, src_width, src_u, halfwidth, src_v, halfwidth,
>+                     dst_y, dst_stride_y, dst_u, dst_stride_u, dst_v,
>+                     dst_stride_v, crop_width, inv_crop_height, rotation);
>       break;
>     }
>     case FOURCC_I422:
>     case FOURCC_YV16: {
>-      const uint8* src_y = sample + src_width * crop_y + crop_x;
>-      const uint8* src_u;
>-      const uint8* src_v;
>+      const uint8_t* src_y = sample + src_width * crop_y + crop_x;
>+      const uint8_t* src_u;
>+      const uint8_t* src_v;
>       int halfwidth = (src_width + 1) / 2;
>       if (format == FOURCC_YV16) {
>         src_v = sample + src_width * abs_src_height + halfwidth * crop_y +
>@@ -213,16 +224,16 @@ int ConvertToI420(const uint8* sample,
>         src_v = sample + src_width * abs_src_height +
>                 halfwidth * (abs_src_height + crop_y) + crop_x / 2;
>       }
>-      r = I422ToI420(src_y, src_width, src_u, halfwidth, src_v, halfwidth, y,
>-                     y_stride, u, u_stride, v, v_stride, crop_width,
>-                     inv_crop_height);
>+      r = I422ToI420(src_y, src_width, src_u, halfwidth, src_v, halfwidth,
>+                     dst_y, dst_stride_y, dst_u, dst_stride_u, dst_v,
>+                     dst_stride_v, crop_width, inv_crop_height);
>       break;
>     }
>     case FOURCC_I444:
>     case FOURCC_YV24: {
>-      const uint8* src_y = sample + src_width * crop_y + crop_x;
>-      const uint8* src_u;
>-      const uint8* src_v;
>+      const uint8_t* src_y = sample + src_width * crop_y + crop_x;
>+      const uint8_t* src_u;
>+      const uint8_t* src_v;
>       if (format == FOURCC_YV24) {
>         src_v = sample + src_width * (abs_src_height + crop_y) + crop_x;
>         src_u = sample + src_width * (abs_src_height * 2 + crop_y) + crop_x;
>@@ -230,15 +241,16 @@ int ConvertToI420(const uint8* sample,
>         src_u = sample + src_width * (abs_src_height + crop_y) + crop_x;
>         src_v = sample + src_width * (abs_src_height * 2 + crop_y) + crop_x;
>       }
>-      r = I444ToI420(src_y, src_width, src_u, src_width, src_v, src_width, y,
>-                     y_stride, u, u_stride, v, v_stride, crop_width,
>-                     inv_crop_height);
>+      r = I444ToI420(src_y, src_width, src_u, src_width, src_v, src_width,
>+                     dst_y, dst_stride_y, dst_u, dst_stride_u, dst_v,
>+                     dst_stride_v, crop_width, inv_crop_height);
>       break;
>     }
> #ifdef HAVE_JPEG
>     case FOURCC_MJPG:
>-      r = MJPGToI420(sample, sample_size, y, y_stride, u, u_stride, v, v_stride,
>-                     src_width, abs_src_height, crop_width, inv_crop_height);
>+      r = MJPGToI420(sample, sample_size, dst_y, dst_stride_y, dst_u,
>+                     dst_stride_u, dst_v, dst_stride_v, src_width,
>+                     abs_src_height, crop_width, inv_crop_height);
>       break;
> #endif
>     default:
>@@ -247,9 +259,10 @@ int ConvertToI420(const uint8* sample,
> 
>   if (need_buf) {
>     if (!r) {
>-      r = I420Rotate(y, y_stride, u, u_stride, v, v_stride, tmp_y, tmp_y_stride,
>-                     tmp_u, tmp_u_stride, tmp_v, tmp_v_stride, crop_width,
>-                     abs_crop_height, rotation);
>+      r = I420Rotate(dst_y, dst_stride_y, dst_u, dst_stride_u, dst_v,
>+                     dst_stride_v, tmp_y, tmp_y_stride, tmp_u, tmp_u_stride,
>+                     tmp_v, tmp_v_stride, crop_width, abs_crop_height,
>+                     rotation);
>     }
>     free(rotate_buffer);
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/cpu_id.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/cpu_id.cc
>index 344f3c06a2b..446aad12078 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/cpu_id.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/cpu_id.cc
>@@ -27,8 +27,6 @@
> #include <stdio.h>
> #include <string.h>
> 
>-#include "libyuv/basic_types.h"  // For CPU_X86
>-
> #ifdef __cplusplus
> namespace libyuv {
> extern "C" {
>@@ -179,7 +177,7 @@ LIBYUV_API SAFEBUFFERS int MipsCpuCaps(const char* cpuinfo_name,
>     if (strcmp(ase, " msa") == 0) {
>       return kCpuHasMSA;
>     }
>-    return kCpuHasDSPR2;
>+    return 0;
>   }
>   while (fgets(cpuinfo_line, sizeof(cpuinfo_line) - 1, f)) {
>     if (memcmp(cpuinfo_line, "ASEs implemented", 16) == 0) {
>@@ -189,7 +187,7 @@ LIBYUV_API SAFEBUFFERS int MipsCpuCaps(const char* cpuinfo_name,
>         if (strcmp(ase, " msa") == 0) {
>           return kCpuHasMSA;
>         }
>-        return kCpuHasDSPR2;
>+        return 0;
>       }
>     }
>   }
>@@ -218,7 +216,9 @@ static LIBYUV_BOOL TestEnv(const char*) {
> 
> static SAFEBUFFERS int GetCpuFlags(void) {
>   int cpu_info = 0;
>-#if !defined(__pnacl__) && !defined(__CLR_VER) && defined(CPU_X86)
>+#if !defined(__pnacl__) && !defined(__CLR_VER) &&                   \
>+    (defined(__x86_64__) || defined(_M_X64) || defined(__i386__) || \
>+     defined(_M_IX86))
>   int cpu_info0[4] = {0, 0, 0, 0};
>   int cpu_info1[4] = {0, 0, 0, 0};
>   int cpu_info7[4] = {0, 0, 0, 0};
>@@ -290,16 +290,10 @@ static SAFEBUFFERS int GetCpuFlags(void) {
> 
> #endif
> #if defined(__mips__) && defined(__linux__)
>-#if defined(__mips_dspr2)
>-  cpu_info |= kCpuHasDSPR2;
>-#endif
> #if defined(__mips_msa)
>   cpu_info = MipsCpuCaps("/proc/cpuinfo", " msa");
> #endif
>   cpu_info |= kCpuHasMIPS;
>-  if (getenv("LIBYUV_DISABLE_DSPR2")) {
>-    cpu_info &= ~kCpuHasDSPR2;
>-  }
>   if (getenv("LIBYUV_DISABLE_MSA")) {
>     cpu_info &= ~kCpuHasMSA;
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/mjpeg_decoder.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/mjpeg_decoder.cc
>index b43c008bdd2..eaf2530130b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/mjpeg_decoder.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/mjpeg_decoder.cc
>@@ -102,7 +102,7 @@ MJpegDecoder::~MJpegDecoder() {
>   DestroyOutputBuffers();
> }
> 
>-LIBYUV_BOOL MJpegDecoder::LoadFrame(const uint8* src, size_t src_len) {
>+LIBYUV_BOOL MJpegDecoder::LoadFrame(const uint8_t* src, size_t src_len) {
>   if (!ValidateJpeg(src, src_len)) {
>     return LIBYUV_FALSE;
>   }
>@@ -129,7 +129,7 @@ LIBYUV_BOOL MJpegDecoder::LoadFrame(const uint8* src, size_t src_len) {
>       if (scanlines_[i]) {
>         delete scanlines_[i];
>       }
>-      scanlines_[i] = new uint8*[scanlines_size];
>+      scanlines_[i] = new uint8_t*[scanlines_size];
>       scanlines_sizes_[i] = scanlines_size;
>     }
> 
>@@ -145,7 +145,7 @@ LIBYUV_BOOL MJpegDecoder::LoadFrame(const uint8* src, size_t src_len) {
>       if (databuf_[i]) {
>         delete databuf_[i];
>       }
>-      databuf_[i] = new uint8[databuf_size];
>+      databuf_[i] = new uint8_t[databuf_size];
>       databuf_strides_[i] = databuf_stride;
>     }
> 
>@@ -243,7 +243,7 @@ LIBYUV_BOOL MJpegDecoder::UnloadFrame() {
> }
> 
> // TODO(fbarchard): Allow rectangle to be specified: x, y, width, height.
>-LIBYUV_BOOL MJpegDecoder::DecodeToBuffers(uint8** planes,
>+LIBYUV_BOOL MJpegDecoder::DecodeToBuffers(uint8_t** planes,
>                                           int dst_width,
>                                           int dst_height) {
>   if (dst_width != GetWidth() || dst_height > GetHeight()) {
>@@ -469,9 +469,9 @@ void MJpegDecoder::AllocOutputBuffers(int num_outbufs) {
>     // it.
>     DestroyOutputBuffers();
> 
>-    scanlines_ = new uint8**[num_outbufs];
>+    scanlines_ = new uint8_t**[num_outbufs];
>     scanlines_sizes_ = new int[num_outbufs];
>-    databuf_ = new uint8*[num_outbufs];
>+    databuf_ = new uint8_t*[num_outbufs];
>     databuf_strides_ = new int[num_outbufs];
> 
>     for (int i = 0; i < num_outbufs; ++i) {
>@@ -527,9 +527,9 @@ LIBYUV_BOOL MJpegDecoder::FinishDecode() {
>   return LIBYUV_TRUE;
> }
> 
>-void MJpegDecoder::SetScanlinePointers(uint8** data) {
>+void MJpegDecoder::SetScanlinePointers(uint8_t** data) {
>   for (int i = 0; i < num_outbufs_; ++i) {
>-    uint8* data_i = data[i];
>+    uint8_t* data_i = data[i];
>     for (int j = 0; j < scanlines_sizes_[i]; ++j) {
>       scanlines_[i][j] = data_i;
>       data_i += GetComponentStride(i);
>@@ -552,13 +552,13 @@ JpegSubsamplingType MJpegDecoder::JpegSubsamplingTypeHelper(
>     if (subsample_x[0] == 1 && subsample_y[0] == 1 && subsample_x[1] == 2 &&
>         subsample_y[1] == 2 && subsample_x[2] == 2 && subsample_y[2] == 2) {
>       return kJpegYuv420;
>-    } else if (subsample_x[0] == 1 && subsample_y[0] == 1 &&
>-               subsample_x[1] == 2 && subsample_y[1] == 1 &&
>-               subsample_x[2] == 2 && subsample_y[2] == 1) {
>+    }
>+    if (subsample_x[0] == 1 && subsample_y[0] == 1 && subsample_x[1] == 2 &&
>+        subsample_y[1] == 1 && subsample_x[2] == 2 && subsample_y[2] == 1) {
>       return kJpegYuv422;
>-    } else if (subsample_x[0] == 1 && subsample_y[0] == 1 &&
>-               subsample_x[1] == 1 && subsample_y[1] == 1 &&
>-               subsample_x[2] == 1 && subsample_y[2] == 1) {
>+    }
>+    if (subsample_x[0] == 1 && subsample_y[0] == 1 && subsample_x[1] == 1 &&
>+        subsample_y[1] == 1 && subsample_x[2] == 1 && subsample_y[2] == 1) {
>       return kJpegYuv444;
>     }
>   } else if (number_of_components == 1) {  // Grey-scale images.
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/mjpeg_validate.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/mjpeg_validate.cc
>index bd760425359..80c2cc0cb9b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/mjpeg_validate.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/mjpeg_validate.cc
>@@ -18,13 +18,13 @@ extern "C" {
> #endif
> 
> // Helper function to scan for EOI marker (0xff 0xd9).
>-static LIBYUV_BOOL ScanEOI(const uint8* sample, size_t sample_size) {
>+static LIBYUV_BOOL ScanEOI(const uint8_t* sample, size_t sample_size) {
>   if (sample_size >= 2) {
>-    const uint8* end = sample + sample_size - 1;
>-    const uint8* it = sample;
>+    const uint8_t* end = sample + sample_size - 1;
>+    const uint8_t* it = sample;
>     while (it < end) {
>       // TODO(fbarchard): scan for 0xd9 instead.
>-      it = (const uint8*)(memchr(it, 0xff, end - it));
>+      it = (const uint8_t*)(memchr(it, 0xff, end - it));
>       if (it == NULL) {
>         break;
>       }
>@@ -39,7 +39,7 @@ static LIBYUV_BOOL ScanEOI(const uint8* sample, size_t sample_size) {
> }
> 
> // Helper function to validate the jpeg appears intact.
>-LIBYUV_BOOL ValidateJpeg(const uint8* sample, size_t sample_size) {
>+LIBYUV_BOOL ValidateJpeg(const uint8_t* sample, size_t sample_size) {
>   // Maximum size that ValidateJpeg will consider valid.
>   const size_t kMaxJpegSize = 0x7fffffffull;
>   const size_t kBackSearchSize = 1024;
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/planar_functions.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/planar_functions.cc
>index dd311d1f08e..5eae3f763a7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/planar_functions.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/planar_functions.cc
>@@ -26,14 +26,14 @@ extern "C" {
> 
> // Copy a plane of data
> LIBYUV_API
>-void CopyPlane(const uint8* src_y,
>+void CopyPlane(const uint8_t* src_y,
>                int src_stride_y,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>                int width,
>                int height) {
>   int y;
>-  void (*CopyRow)(const uint8* src, uint8* dst, int width) = CopyRow_C;
>+  void (*CopyRow)(const uint8_t* src, uint8_t* dst, int width) = CopyRow_C;
>   // Negative height means invert the image.
>   if (height < 0) {
>     height = -height;
>@@ -50,6 +50,7 @@ void CopyPlane(const uint8* src_y,
>   if (src_y == dst_y && src_stride_y == dst_stride_y) {
>     return;
>   }
>+
> #if defined(HAS_COPYROW_SSE2)
>   if (TestCpuFlag(kCpuHasSSE2)) {
>     CopyRow = IS_ALIGNED(width, 32) ? CopyRow_SSE2 : CopyRow_Any_SSE2;
>@@ -70,11 +71,6 @@ void CopyPlane(const uint8* src_y,
>     CopyRow = IS_ALIGNED(width, 32) ? CopyRow_NEON : CopyRow_Any_NEON;
>   }
> #endif
>-#if defined(HAS_COPYROW_MIPS)
>-  if (TestCpuFlag(kCpuHasMIPS)) {
>-    CopyRow = CopyRow_MIPS;
>-  }
>-#endif
> 
>   // Copy plane
>   for (y = 0; y < height; ++y) {
>@@ -87,14 +83,14 @@ void CopyPlane(const uint8* src_y,
> // TODO(fbarchard): Consider support for negative height.
> // TODO(fbarchard): Consider stride measured in bytes.
> LIBYUV_API
>-void CopyPlane_16(const uint16* src_y,
>+void CopyPlane_16(const uint16_t* src_y,
>                   int src_stride_y,
>-                  uint16* dst_y,
>+                  uint16_t* dst_y,
>                   int dst_stride_y,
>                   int width,
>                   int height) {
>   int y;
>-  void (*CopyRow)(const uint16* src, uint16* dst, int width) = CopyRow_16_C;
>+  void (*CopyRow)(const uint16_t* src, uint16_t* dst, int width) = CopyRow_16_C;
>   // Coalesce rows.
>   if (src_stride_y == width && dst_stride_y == width) {
>     width *= height;
>@@ -116,11 +112,6 @@ void CopyPlane_16(const uint16* src_y,
>     CopyRow = CopyRow_16_NEON;
>   }
> #endif
>-#if defined(HAS_COPYROW_16_MIPS)
>-  if (TestCpuFlag(kCpuHasMIPS)) {
>-    CopyRow = CopyRow_16_MIPS;
>-  }
>-#endif
> 
>   // Copy plane
>   for (y = 0; y < height; ++y) {
>@@ -130,19 +121,119 @@ void CopyPlane_16(const uint16* src_y,
>   }
> }
> 
>+// Convert a plane of 16 bit data to 8 bit
>+LIBYUV_API
>+void Convert16To8Plane(const uint16_t* src_y,
>+                       int src_stride_y,
>+                       uint8_t* dst_y,
>+                       int dst_stride_y,
>+                       int scale,  // 16384 for 10 bits
>+                       int width,
>+                       int height) {
>+  int y;
>+  void (*Convert16To8Row)(const uint16_t* src_y, uint8_t* dst_y, int scale,
>+                          int width) = Convert16To8Row_C;
>+
>+  // Negative height means invert the image.
>+  if (height < 0) {
>+    height = -height;
>+    dst_y = dst_y + (height - 1) * dst_stride_y;
>+    dst_stride_y = -dst_stride_y;
>+  }
>+  // Coalesce rows.
>+  if (src_stride_y == width && dst_stride_y == width) {
>+    width *= height;
>+    height = 1;
>+    src_stride_y = dst_stride_y = 0;
>+  }
>+#if defined(HAS_CONVERT16TO8ROW_SSSE3)
>+  if (TestCpuFlag(kCpuHasSSSE3)) {
>+    Convert16To8Row = Convert16To8Row_Any_SSSE3;
>+    if (IS_ALIGNED(width, 16)) {
>+      Convert16To8Row = Convert16To8Row_SSSE3;
>+    }
>+  }
>+#endif
>+#if defined(HAS_CONVERT16TO8ROW_AVX2)
>+  if (TestCpuFlag(kCpuHasAVX2)) {
>+    Convert16To8Row = Convert16To8Row_Any_AVX2;
>+    if (IS_ALIGNED(width, 32)) {
>+      Convert16To8Row = Convert16To8Row_AVX2;
>+    }
>+  }
>+#endif
>+
>+  // Convert plane
>+  for (y = 0; y < height; ++y) {
>+    Convert16To8Row(src_y, dst_y, scale, width);
>+    src_y += src_stride_y;
>+    dst_y += dst_stride_y;
>+  }
>+}
>+
>+// Convert a plane of 8 bit data to 16 bit
>+LIBYUV_API
>+void Convert8To16Plane(const uint8_t* src_y,
>+                       int src_stride_y,
>+                       uint16_t* dst_y,
>+                       int dst_stride_y,
>+                       int scale,  // 16384 for 10 bits
>+                       int width,
>+                       int height) {
>+  int y;
>+  void (*Convert8To16Row)(const uint8_t* src_y, uint16_t* dst_y, int scale,
>+                          int width) = Convert8To16Row_C;
>+
>+  // Negative height means invert the image.
>+  if (height < 0) {
>+    height = -height;
>+    dst_y = dst_y + (height - 1) * dst_stride_y;
>+    dst_stride_y = -dst_stride_y;
>+  }
>+  // Coalesce rows.
>+  if (src_stride_y == width && dst_stride_y == width) {
>+    width *= height;
>+    height = 1;
>+    src_stride_y = dst_stride_y = 0;
>+  }
>+#if defined(HAS_CONVERT8TO16ROW_SSE2)
>+  if (TestCpuFlag(kCpuHasSSE2)) {
>+    Convert8To16Row = Convert8To16Row_Any_SSE2;
>+    if (IS_ALIGNED(width, 16)) {
>+      Convert8To16Row = Convert8To16Row_SSE2;
>+    }
>+  }
>+#endif
>+#if defined(HAS_CONVERT8TO16ROW_AVX2)
>+  if (TestCpuFlag(kCpuHasAVX2)) {
>+    Convert8To16Row = Convert8To16Row_Any_AVX2;
>+    if (IS_ALIGNED(width, 32)) {
>+      Convert8To16Row = Convert8To16Row_AVX2;
>+    }
>+  }
>+#endif
>+
>+  // Convert plane
>+  for (y = 0; y < height; ++y) {
>+    Convert8To16Row(src_y, dst_y, scale, width);
>+    src_y += src_stride_y;
>+    dst_y += dst_stride_y;
>+  }
>+}
>+
> // Copy I422.
> LIBYUV_API
>-int I422Copy(const uint8* src_y,
>+int I422Copy(const uint8_t* src_y,
>              int src_stride_y,
>-             const uint8* src_u,
>+             const uint8_t* src_u,
>              int src_stride_u,
>-             const uint8* src_v,
>+             const uint8_t* src_v,
>              int src_stride_v,
>-             uint8* dst_y,
>+             uint8_t* dst_y,
>              int dst_stride_y,
>-             uint8* dst_u,
>+             uint8_t* dst_u,
>              int dst_stride_u,
>-             uint8* dst_v,
>+             uint8_t* dst_v,
>              int dst_stride_v,
>              int width,
>              int height) {
>@@ -171,17 +262,17 @@ int I422Copy(const uint8* src_y,
> 
> // Copy I444.
> LIBYUV_API
>-int I444Copy(const uint8* src_y,
>+int I444Copy(const uint8_t* src_y,
>              int src_stride_y,
>-             const uint8* src_u,
>+             const uint8_t* src_u,
>              int src_stride_u,
>-             const uint8* src_v,
>+             const uint8_t* src_v,
>              int src_stride_v,
>-             uint8* dst_y,
>+             uint8_t* dst_y,
>              int dst_stride_y,
>-             uint8* dst_u,
>+             uint8_t* dst_u,
>              int dst_stride_u,
>-             uint8* dst_v,
>+             uint8_t* dst_v,
>              int dst_stride_v,
>              int width,
>              int height) {
>@@ -209,9 +300,9 @@ int I444Copy(const uint8* src_y,
> 
> // Copy I400.
> LIBYUV_API
>-int I400ToI400(const uint8* src_y,
>+int I400ToI400(const uint8_t* src_y,
>                int src_stride_y,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>                int width,
>                int height) {
>@@ -230,13 +321,13 @@ int I400ToI400(const uint8* src_y,
> 
> // Convert I420 to I400.
> LIBYUV_API
>-int I420ToI400(const uint8* src_y,
>+int I420ToI400(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>                int width,
>                int height) {
>@@ -261,16 +352,16 @@ int I420ToI400(const uint8* src_y,
> // Support function for NV12 etc UV channels.
> // Width and height are plane sizes (typically half pixel width).
> LIBYUV_API
>-void SplitUVPlane(const uint8* src_uv,
>+void SplitUVPlane(const uint8_t* src_uv,
>                   int src_stride_uv,
>-                  uint8* dst_u,
>+                  uint8_t* dst_u,
>                   int dst_stride_u,
>-                  uint8* dst_v,
>+                  uint8_t* dst_v,
>                   int dst_stride_v,
>                   int width,
>                   int height) {
>   int y;
>-  void (*SplitUVRow)(const uint8* src_uv, uint8* dst_u, uint8* dst_v,
>+  void (*SplitUVRow)(const uint8_t* src_uv, uint8_t* dst_u, uint8_t* dst_v,
>                      int width) = SplitUVRow_C;
>   // Negative height means invert the image.
>   if (height < 0) {
>@@ -311,16 +402,6 @@ void SplitUVPlane(const uint8* src_uv,
>     }
>   }
> #endif
>-#if defined(HAS_SPLITUVROW_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2) && IS_ALIGNED(dst_u, 4) &&
>-      IS_ALIGNED(dst_stride_u, 4) && IS_ALIGNED(dst_v, 4) &&
>-      IS_ALIGNED(dst_stride_v, 4)) {
>-    SplitUVRow = SplitUVRow_Any_DSPR2;
>-    if (IS_ALIGNED(width, 16)) {
>-      SplitUVRow = SplitUVRow_DSPR2;
>-    }
>-  }
>-#endif
> #if defined(HAS_SPLITUVROW_MSA)
>   if (TestCpuFlag(kCpuHasMSA)) {
>     SplitUVRow = SplitUVRow_Any_MSA;
>@@ -340,17 +421,17 @@ void SplitUVPlane(const uint8* src_uv,
> }
> 
> LIBYUV_API
>-void MergeUVPlane(const uint8* src_u,
>+void MergeUVPlane(const uint8_t* src_u,
>                   int src_stride_u,
>-                  const uint8* src_v,
>+                  const uint8_t* src_v,
>                   int src_stride_v,
>-                  uint8* dst_uv,
>+                  uint8_t* dst_uv,
>                   int dst_stride_uv,
>                   int width,
>                   int height) {
>   int y;
>-  void (*MergeUVRow)(const uint8* src_u, const uint8* src_v, uint8* dst_uv,
>-                     int width) = MergeUVRow_C;
>+  void (*MergeUVRow)(const uint8_t* src_u, const uint8_t* src_v,
>+                     uint8_t* dst_uv, int width) = MergeUVRow_C;
>   // Coalesce rows.
>   // Negative height means invert the image.
>   if (height < 0) {
>@@ -410,19 +491,19 @@ void MergeUVPlane(const uint8* src_u,
> // Support function for NV12 etc RGB channels.
> // Width and height are plane sizes (typically half pixel width).
> LIBYUV_API
>-void SplitRGBPlane(const uint8* src_rgb,
>+void SplitRGBPlane(const uint8_t* src_rgb,
>                    int src_stride_rgb,
>-                   uint8* dst_r,
>+                   uint8_t* dst_r,
>                    int dst_stride_r,
>-                   uint8* dst_g,
>+                   uint8_t* dst_g,
>                    int dst_stride_g,
>-                   uint8* dst_b,
>+                   uint8_t* dst_b,
>                    int dst_stride_b,
>                    int width,
>                    int height) {
>   int y;
>-  void (*SplitRGBRow)(const uint8* src_rgb, uint8* dst_r, uint8* dst_g,
>-                      uint8* dst_b, int width) = SplitRGBRow_C;
>+  void (*SplitRGBRow)(const uint8_t* src_rgb, uint8_t* dst_r, uint8_t* dst_g,
>+                      uint8_t* dst_b, int width) = SplitRGBRow_C;
>   // Negative height means invert the image.
>   if (height < 0) {
>     height = -height;
>@@ -468,19 +549,19 @@ void SplitRGBPlane(const uint8* src_rgb,
> }
> 
> LIBYUV_API
>-void MergeRGBPlane(const uint8* src_r,
>+void MergeRGBPlane(const uint8_t* src_r,
>                    int src_stride_r,
>-                   const uint8* src_g,
>+                   const uint8_t* src_g,
>                    int src_stride_g,
>-                   const uint8* src_b,
>+                   const uint8_t* src_b,
>                    int src_stride_b,
>-                   uint8* dst_rgb,
>+                   uint8_t* dst_rgb,
>                    int dst_stride_rgb,
>                    int width,
>                    int height) {
>   int y;
>-  void (*MergeRGBRow)(const uint8* src_r, const uint8* src_g,
>-                      const uint8* src_b, uint8* dst_rgb, int width) =
>+  void (*MergeRGBRow)(const uint8_t* src_r, const uint8_t* src_g,
>+                      const uint8_t* src_b, uint8_t* dst_rgb, int width) =
>       MergeRGBRow_C;
>   // Coalesce rows.
>   // Negative height means invert the image.
>@@ -524,14 +605,14 @@ void MergeRGBPlane(const uint8* src_r,
> }
> 
> // Mirror a plane of data.
>-void MirrorPlane(const uint8* src_y,
>+void MirrorPlane(const uint8_t* src_y,
>                  int src_stride_y,
>-                 uint8* dst_y,
>+                 uint8_t* dst_y,
>                  int dst_stride_y,
>                  int width,
>                  int height) {
>   int y;
>-  void (*MirrorRow)(const uint8* src, uint8* dst, int width) = MirrorRow_C;
>+  void (*MirrorRow)(const uint8_t* src, uint8_t* dst, int width) = MirrorRow_C;
>   // Negative height means invert the image.
>   if (height < 0) {
>     height = -height;
>@@ -562,14 +643,6 @@ void MirrorPlane(const uint8* src_y,
>     }
>   }
> #endif
>-// TODO(fbarchard): Mirror on mips handle unaligned memory.
>-#if defined(HAS_MIRRORROW_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2) && IS_ALIGNED(src_y, 4) &&
>-      IS_ALIGNED(src_stride_y, 4) && IS_ALIGNED(dst_y, 4) &&
>-      IS_ALIGNED(dst_stride_y, 4)) {
>-    MirrorRow = MirrorRow_DSPR2;
>-  }
>-#endif
> #if defined(HAS_MIRRORROW_MSA)
>   if (TestCpuFlag(kCpuHasMSA)) {
>     MirrorRow = MirrorRow_Any_MSA;
>@@ -589,20 +662,20 @@ void MirrorPlane(const uint8* src_y,
> 
> // Convert YUY2 to I422.
> LIBYUV_API
>-int YUY2ToI422(const uint8* src_yuy2,
>+int YUY2ToI422(const uint8_t* src_yuy2,
>                int src_stride_yuy2,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_u,
>+               uint8_t* dst_u,
>                int dst_stride_u,
>-               uint8* dst_v,
>+               uint8_t* dst_v,
>                int dst_stride_v,
>                int width,
>                int height) {
>   int y;
>-  void (*YUY2ToUV422Row)(const uint8* src_yuy2, uint8* dst_u, uint8* dst_v,
>-                         int width) = YUY2ToUV422Row_C;
>-  void (*YUY2ToYRow)(const uint8* src_yuy2, uint8* dst_y, int width) =
>+  void (*YUY2ToUV422Row)(const uint8_t* src_yuy2, uint8_t* dst_u,
>+                         uint8_t* dst_v, int width) = YUY2ToUV422Row_C;
>+  void (*YUY2ToYRow)(const uint8_t* src_yuy2, uint8_t* dst_y, int width) =
>       YUY2ToYRow_C;
>   if (!src_yuy2 || !dst_y || !dst_u || !dst_v || width <= 0 || height == 0) {
>     return -1;
>@@ -675,20 +748,20 @@ int YUY2ToI422(const uint8* src_yuy2,
> 
> // Convert UYVY to I422.
> LIBYUV_API
>-int UYVYToI422(const uint8* src_uyvy,
>+int UYVYToI422(const uint8_t* src_uyvy,
>                int src_stride_uyvy,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_u,
>+               uint8_t* dst_u,
>                int dst_stride_u,
>-               uint8* dst_v,
>+               uint8_t* dst_v,
>                int dst_stride_v,
>                int width,
>                int height) {
>   int y;
>-  void (*UYVYToUV422Row)(const uint8* src_uyvy, uint8* dst_u, uint8* dst_v,
>-                         int width) = UYVYToUV422Row_C;
>-  void (*UYVYToYRow)(const uint8* src_uyvy, uint8* dst_y, int width) =
>+  void (*UYVYToUV422Row)(const uint8_t* src_uyvy, uint8_t* dst_u,
>+                         uint8_t* dst_v, int width) = UYVYToUV422Row_C;
>+  void (*UYVYToYRow)(const uint8_t* src_uyvy, uint8_t* dst_y, int width) =
>       UYVYToYRow_C;
>   if (!src_uyvy || !dst_y || !dst_u || !dst_v || width <= 0 || height == 0) {
>     return -1;
>@@ -761,14 +834,14 @@ int UYVYToI422(const uint8* src_uyvy,
> 
> // Convert YUY2 to Y.
> LIBYUV_API
>-int YUY2ToY(const uint8* src_yuy2,
>+int YUY2ToY(const uint8_t* src_yuy2,
>             int src_stride_yuy2,
>-            uint8* dst_y,
>+            uint8_t* dst_y,
>             int dst_stride_y,
>             int width,
>             int height) {
>   int y;
>-  void (*YUY2ToYRow)(const uint8* src_yuy2, uint8* dst_y, int width) =
>+  void (*YUY2ToYRow)(const uint8_t* src_yuy2, uint8_t* dst_y, int width) =
>       YUY2ToYRow_C;
>   if (!src_yuy2 || !dst_y || width <= 0 || height == 0) {
>     return -1;
>@@ -828,9 +901,9 @@ int YUY2ToY(const uint8* src_yuy2,
> 
> // Mirror I400 with optional flipping
> LIBYUV_API
>-int I400Mirror(const uint8* src_y,
>+int I400Mirror(const uint8_t* src_y,
>                int src_stride_y,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>                int width,
>                int height) {
>@@ -850,17 +923,17 @@ int I400Mirror(const uint8* src_y,
> 
> // Mirror I420 with optional flipping
> LIBYUV_API
>-int I420Mirror(const uint8* src_y,
>+int I420Mirror(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_u,
>+               uint8_t* dst_u,
>                int dst_stride_u,
>-               uint8* dst_v,
>+               uint8_t* dst_v,
>                int dst_stride_v,
>                int width,
>                int height) {
>@@ -892,14 +965,14 @@ int I420Mirror(const uint8* src_y,
> 
> // ARGB mirror.
> LIBYUV_API
>-int ARGBMirror(const uint8* src_argb,
>+int ARGBMirror(const uint8_t* src_argb,
>                int src_stride_argb,
>-               uint8* dst_argb,
>+               uint8_t* dst_argb,
>                int dst_stride_argb,
>                int width,
>                int height) {
>   int y;
>-  void (*ARGBMirrorRow)(const uint8* src, uint8* dst, int width) =
>+  void (*ARGBMirrorRow)(const uint8_t* src, uint8_t* dst, int width) =
>       ARGBMirrorRow_C;
>   if (!src_argb || !dst_argb || width <= 0 || height == 0) {
>     return -1;
>@@ -957,8 +1030,8 @@ int ARGBMirror(const uint8* src_argb,
> // the same blend function for all pixels if possible.
> LIBYUV_API
> ARGBBlendRow GetARGBBlend() {
>-  void (*ARGBBlendRow)(const uint8* src_argb, const uint8* src_argb1,
>-                       uint8* dst_argb, int width) = ARGBBlendRow_C;
>+  void (*ARGBBlendRow)(const uint8_t* src_argb, const uint8_t* src_argb1,
>+                       uint8_t* dst_argb, int width) = ARGBBlendRow_C;
> #if defined(HAS_ARGBBLENDROW_SSSE3)
>   if (TestCpuFlag(kCpuHasSSSE3)) {
>     ARGBBlendRow = ARGBBlendRow_SSSE3;
>@@ -980,17 +1053,17 @@ ARGBBlendRow GetARGBBlend() {
> 
> // Alpha Blend 2 ARGB images and store to destination.
> LIBYUV_API
>-int ARGBBlend(const uint8* src_argb0,
>+int ARGBBlend(const uint8_t* src_argb0,
>               int src_stride_argb0,
>-              const uint8* src_argb1,
>+              const uint8_t* src_argb1,
>               int src_stride_argb1,
>-              uint8* dst_argb,
>+              uint8_t* dst_argb,
>               int dst_stride_argb,
>               int width,
>               int height) {
>   int y;
>-  void (*ARGBBlendRow)(const uint8* src_argb, const uint8* src_argb1,
>-                       uint8* dst_argb, int width) = GetARGBBlend();
>+  void (*ARGBBlendRow)(const uint8_t* src_argb, const uint8_t* src_argb1,
>+                       uint8_t* dst_argb, int width) = GetARGBBlend();
>   if (!src_argb0 || !src_argb1 || !dst_argb || width <= 0 || height == 0) {
>     return -1;
>   }
>@@ -1019,19 +1092,19 @@ int ARGBBlend(const uint8* src_argb0,
> 
> // Alpha Blend plane and store to destination.
> LIBYUV_API
>-int BlendPlane(const uint8* src_y0,
>+int BlendPlane(const uint8_t* src_y0,
>                int src_stride_y0,
>-               const uint8* src_y1,
>+               const uint8_t* src_y1,
>                int src_stride_y1,
>-               const uint8* alpha,
>+               const uint8_t* alpha,
>                int alpha_stride,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>                int width,
>                int height) {
>   int y;
>-  void (*BlendPlaneRow)(const uint8* src0, const uint8* src1,
>-                        const uint8* alpha, uint8* dst, int width) =
>+  void (*BlendPlaneRow)(const uint8_t* src0, const uint8_t* src1,
>+                        const uint8_t* alpha, uint8_t* dst, int width) =
>       BlendPlaneRow_C;
>   if (!src_y0 || !src_y1 || !alpha || !dst_y || width <= 0 || height == 0) {
>     return -1;
>@@ -1081,36 +1154,36 @@ int BlendPlane(const uint8* src_y0,
> #define MAXTWIDTH 2048
> // Alpha Blend YUV images and store to destination.
> LIBYUV_API
>-int I420Blend(const uint8* src_y0,
>+int I420Blend(const uint8_t* src_y0,
>               int src_stride_y0,
>-              const uint8* src_u0,
>+              const uint8_t* src_u0,
>               int src_stride_u0,
>-              const uint8* src_v0,
>+              const uint8_t* src_v0,
>               int src_stride_v0,
>-              const uint8* src_y1,
>+              const uint8_t* src_y1,
>               int src_stride_y1,
>-              const uint8* src_u1,
>+              const uint8_t* src_u1,
>               int src_stride_u1,
>-              const uint8* src_v1,
>+              const uint8_t* src_v1,
>               int src_stride_v1,
>-              const uint8* alpha,
>+              const uint8_t* alpha,
>               int alpha_stride,
>-              uint8* dst_y,
>+              uint8_t* dst_y,
>               int dst_stride_y,
>-              uint8* dst_u,
>+              uint8_t* dst_u,
>               int dst_stride_u,
>-              uint8* dst_v,
>+              uint8_t* dst_v,
>               int dst_stride_v,
>               int width,
>               int height) {
>   int y;
>   // Half width/height for UV.
>   int halfwidth = (width + 1) >> 1;
>-  void (*BlendPlaneRow)(const uint8* src0, const uint8* src1,
>-                        const uint8* alpha, uint8* dst, int width) =
>+  void (*BlendPlaneRow)(const uint8_t* src0, const uint8_t* src1,
>+                        const uint8_t* alpha, uint8_t* dst, int width) =
>       BlendPlaneRow_C;
>-  void (*ScaleRowDown2)(const uint8* src_ptr, ptrdiff_t src_stride,
>-                        uint8* dst_ptr, int dst_width) = ScaleRowDown2Box_C;
>+  void (*ScaleRowDown2)(const uint8_t* src_ptr, ptrdiff_t src_stride,
>+                        uint8_t* dst_ptr, int dst_width) = ScaleRowDown2Box_C;
>   if (!src_y0 || !src_u0 || !src_v0 || !src_y1 || !src_u1 || !src_v1 ||
>       !alpha || !dst_y || !dst_u || !dst_v || width <= 0 || height == 0) {
>     return -1;
>@@ -1205,17 +1278,17 @@ int I420Blend(const uint8* src_y0,
> 
> // Multiply 2 ARGB images and store to destination.
> LIBYUV_API
>-int ARGBMultiply(const uint8* src_argb0,
>+int ARGBMultiply(const uint8_t* src_argb0,
>                  int src_stride_argb0,
>-                 const uint8* src_argb1,
>+                 const uint8_t* src_argb1,
>                  int src_stride_argb1,
>-                 uint8* dst_argb,
>+                 uint8_t* dst_argb,
>                  int dst_stride_argb,
>                  int width,
>                  int height) {
>   int y;
>-  void (*ARGBMultiplyRow)(const uint8* src0, const uint8* src1, uint8* dst,
>-                          int width) = ARGBMultiplyRow_C;
>+  void (*ARGBMultiplyRow)(const uint8_t* src0, const uint8_t* src1,
>+                          uint8_t* dst, int width) = ARGBMultiplyRow_C;
>   if (!src_argb0 || !src_argb1 || !dst_argb || width <= 0 || height == 0) {
>     return -1;
>   }
>@@ -1277,16 +1350,16 @@ int ARGBMultiply(const uint8* src_argb0,
> 
> // Add 2 ARGB images and store to destination.
> LIBYUV_API
>-int ARGBAdd(const uint8* src_argb0,
>+int ARGBAdd(const uint8_t* src_argb0,
>             int src_stride_argb0,
>-            const uint8* src_argb1,
>+            const uint8_t* src_argb1,
>             int src_stride_argb1,
>-            uint8* dst_argb,
>+            uint8_t* dst_argb,
>             int dst_stride_argb,
>             int width,
>             int height) {
>   int y;
>-  void (*ARGBAddRow)(const uint8* src0, const uint8* src1, uint8* dst,
>+  void (*ARGBAddRow)(const uint8_t* src0, const uint8_t* src1, uint8_t* dst,
>                      int width) = ARGBAddRow_C;
>   if (!src_argb0 || !src_argb1 || !dst_argb || width <= 0 || height == 0) {
>     return -1;
>@@ -1354,17 +1427,17 @@ int ARGBAdd(const uint8* src_argb0,
> 
> // Subtract 2 ARGB images and store to destination.
> LIBYUV_API
>-int ARGBSubtract(const uint8* src_argb0,
>+int ARGBSubtract(const uint8_t* src_argb0,
>                  int src_stride_argb0,
>-                 const uint8* src_argb1,
>+                 const uint8_t* src_argb1,
>                  int src_stride_argb1,
>-                 uint8* dst_argb,
>+                 uint8_t* dst_argb,
>                  int dst_stride_argb,
>                  int width,
>                  int height) {
>   int y;
>-  void (*ARGBSubtractRow)(const uint8* src0, const uint8* src1, uint8* dst,
>-                          int width) = ARGBSubtractRow_C;
>+  void (*ARGBSubtractRow)(const uint8_t* src0, const uint8_t* src1,
>+                          uint8_t* dst, int width) = ARGBSubtractRow_C;
>   if (!src_argb0 || !src_argb1 || !dst_argb || width <= 0 || height == 0) {
>     return -1;
>   }
>@@ -1424,20 +1497,20 @@ int ARGBSubtract(const uint8* src_argb0,
>   return 0;
> }
> // Convert I422 to RGBA with matrix
>-static int I422ToRGBAMatrix(const uint8* src_y,
>+static int I422ToRGBAMatrix(const uint8_t* src_y,
>                             int src_stride_y,
>-                            const uint8* src_u,
>+                            const uint8_t* src_u,
>                             int src_stride_u,
>-                            const uint8* src_v,
>+                            const uint8_t* src_v,
>                             int src_stride_v,
>-                            uint8* dst_rgba,
>+                            uint8_t* dst_rgba,
>                             int dst_stride_rgba,
>                             const struct YuvConstants* yuvconstants,
>                             int width,
>                             int height) {
>   int y;
>-  void (*I422ToRGBARow)(const uint8* y_buf, const uint8* u_buf,
>-                        const uint8* v_buf, uint8* rgb_buf,
>+  void (*I422ToRGBARow)(const uint8_t* y_buf, const uint8_t* u_buf,
>+                        const uint8_t* v_buf, uint8_t* rgb_buf,
>                         const struct YuvConstants* yuvconstants, int width) =
>       I422ToRGBARow_C;
>   if (!src_y || !src_u || !src_v || !dst_rgba || width <= 0 || height == 0) {
>@@ -1473,15 +1546,6 @@ static int I422ToRGBAMatrix(const uint8* src_y,
>     }
>   }
> #endif
>-#if defined(HAS_I422TORGBAROW_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2) && IS_ALIGNED(width, 4) &&
>-      IS_ALIGNED(src_y, 4) && IS_ALIGNED(src_stride_y, 4) &&
>-      IS_ALIGNED(src_u, 2) && IS_ALIGNED(src_stride_u, 2) &&
>-      IS_ALIGNED(src_v, 2) && IS_ALIGNED(src_stride_v, 2) &&
>-      IS_ALIGNED(dst_rgba, 4) && IS_ALIGNED(dst_stride_rgba, 4)) {
>-    I422ToRGBARow = I422ToRGBARow_DSPR2;
>-  }
>-#endif
> #if defined(HAS_I422TORGBAROW_MSA)
>   if (TestCpuFlag(kCpuHasMSA)) {
>     I422ToRGBARow = I422ToRGBARow_Any_MSA;
>@@ -1503,13 +1567,13 @@ static int I422ToRGBAMatrix(const uint8* src_y,
> 
> // Convert I422 to RGBA.
> LIBYUV_API
>-int I422ToRGBA(const uint8* src_y,
>+int I422ToRGBA(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_rgba,
>+               uint8_t* dst_rgba,
>                int dst_stride_rgba,
>                int width,
>                int height) {
>@@ -1520,13 +1584,13 @@ int I422ToRGBA(const uint8* src_y,
> 
> // Convert I422 to BGRA.
> LIBYUV_API
>-int I422ToBGRA(const uint8* src_y,
>+int I422ToBGRA(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_bgra,
>+               uint8_t* dst_bgra,
>                int dst_stride_bgra,
>                int width,
>                int height) {
>@@ -1539,17 +1603,17 @@ int I422ToBGRA(const uint8* src_y,
> 
> // Convert NV12 to RGB565.
> LIBYUV_API
>-int NV12ToRGB565(const uint8* src_y,
>+int NV12ToRGB565(const uint8_t* src_y,
>                  int src_stride_y,
>-                 const uint8* src_uv,
>+                 const uint8_t* src_uv,
>                  int src_stride_uv,
>-                 uint8* dst_rgb565,
>+                 uint8_t* dst_rgb565,
>                  int dst_stride_rgb565,
>                  int width,
>                  int height) {
>   int y;
>   void (*NV12ToRGB565Row)(
>-      const uint8* y_buf, const uint8* uv_buf, uint8* rgb_buf,
>+      const uint8_t* y_buf, const uint8_t* uv_buf, uint8_t* rgb_buf,
>       const struct YuvConstants* yuvconstants, int width) = NV12ToRGB565Row_C;
>   if (!src_y || !src_uv || !dst_rgb565 || width <= 0 || height == 0) {
>     return -1;
>@@ -1606,14 +1670,14 @@ int NV12ToRGB565(const uint8* src_y,
> 
> // Convert RAW to RGB24.
> LIBYUV_API
>-int RAWToRGB24(const uint8* src_raw,
>+int RAWToRGB24(const uint8_t* src_raw,
>                int src_stride_raw,
>-               uint8* dst_rgb24,
>+               uint8_t* dst_rgb24,
>                int dst_stride_rgb24,
>                int width,
>                int height) {
>   int y;
>-  void (*RAWToRGB24Row)(const uint8* src_rgb, uint8* dst_rgb24, int width) =
>+  void (*RAWToRGB24Row)(const uint8_t* src_rgb, uint8_t* dst_rgb24, int width) =
>       RAWToRGB24Row_C;
>   if (!src_raw || !dst_rgb24 || width <= 0 || height == 0) {
>     return -1;
>@@ -1664,13 +1728,13 @@ int RAWToRGB24(const uint8* src_raw,
> }
> 
> LIBYUV_API
>-void SetPlane(uint8* dst_y,
>+void SetPlane(uint8_t* dst_y,
>               int dst_stride_y,
>               int width,
>               int height,
>-              uint32 value) {
>+              uint32_t value) {
>   int y;
>-  void (*SetRow)(uint8 * dst, uint8 value, int width) = SetRow_C;
>+  void (*SetRow)(uint8_t * dst, uint8_t value, int width) = SetRow_C;
>   if (height < 0) {
>     height = -height;
>     dst_y = dst_y + (height - 1) * dst_stride_y;
>@@ -1718,11 +1782,11 @@ void SetPlane(uint8* dst_y,
> 
> // Draw a rectangle into I420
> LIBYUV_API
>-int I420Rect(uint8* dst_y,
>+int I420Rect(uint8_t* dst_y,
>              int dst_stride_y,
>-             uint8* dst_u,
>+             uint8_t* dst_u,
>              int dst_stride_u,
>-             uint8* dst_v,
>+             uint8_t* dst_v,
>              int dst_stride_v,
>              int x,
>              int y,
>@@ -1733,9 +1797,9 @@ int I420Rect(uint8* dst_y,
>              int value_v) {
>   int halfwidth = (width + 1) >> 1;
>   int halfheight = (height + 1) >> 1;
>-  uint8* start_y = dst_y + y * dst_stride_y + x;
>-  uint8* start_u = dst_u + (y / 2) * dst_stride_u + (x / 2);
>-  uint8* start_v = dst_v + (y / 2) * dst_stride_v + (x / 2);
>+  uint8_t* start_y = dst_y + y * dst_stride_y + x;
>+  uint8_t* start_u = dst_u + (y / 2) * dst_stride_u + (x / 2);
>+  uint8_t* start_v = dst_v + (y / 2) * dst_stride_v + (x / 2);
>   if (!dst_y || !dst_u || !dst_v || width <= 0 || height == 0 || x < 0 ||
>       y < 0 || value_y < 0 || value_y > 255 || value_u < 0 || value_u > 255 ||
>       value_v < 0 || value_v > 255) {
>@@ -1750,15 +1814,16 @@ int I420Rect(uint8* dst_y,
> 
> // Draw a rectangle into ARGB
> LIBYUV_API
>-int ARGBRect(uint8* dst_argb,
>+int ARGBRect(uint8_t* dst_argb,
>              int dst_stride_argb,
>              int dst_x,
>              int dst_y,
>              int width,
>              int height,
>-             uint32 value) {
>+             uint32_t value) {
>   int y;
>-  void (*ARGBSetRow)(uint8 * dst_argb, uint32 value, int width) = ARGBSetRow_C;
>+  void (*ARGBSetRow)(uint8_t * dst_argb, uint32_t value, int width) =
>+      ARGBSetRow_C;
>   if (!dst_argb || width <= 0 || height == 0 || dst_x < 0 || dst_y < 0) {
>     return -1;
>   }
>@@ -1819,15 +1884,15 @@ int ARGBRect(uint8* dst_argb,
> //   f is foreground pixel premultiplied by alpha
> 
> LIBYUV_API
>-int ARGBAttenuate(const uint8* src_argb,
>+int ARGBAttenuate(const uint8_t* src_argb,
>                   int src_stride_argb,
>-                  uint8* dst_argb,
>+                  uint8_t* dst_argb,
>                   int dst_stride_argb,
>                   int width,
>                   int height) {
>   int y;
>-  void (*ARGBAttenuateRow)(const uint8* src_argb, uint8* dst_argb, int width) =
>-      ARGBAttenuateRow_C;
>+  void (*ARGBAttenuateRow)(const uint8_t* src_argb, uint8_t* dst_argb,
>+                           int width) = ARGBAttenuateRow_C;
>   if (!src_argb || !dst_argb || width <= 0 || height == 0) {
>     return -1;
>   }
>@@ -1885,14 +1950,14 @@ int ARGBAttenuate(const uint8* src_argb,
> 
> // Convert preattentuated ARGB to unattenuated ARGB.
> LIBYUV_API
>-int ARGBUnattenuate(const uint8* src_argb,
>+int ARGBUnattenuate(const uint8_t* src_argb,
>                     int src_stride_argb,
>-                    uint8* dst_argb,
>+                    uint8_t* dst_argb,
>                     int dst_stride_argb,
>                     int width,
>                     int height) {
>   int y;
>-  void (*ARGBUnattenuateRow)(const uint8* src_argb, uint8* dst_argb,
>+  void (*ARGBUnattenuateRow)(const uint8_t* src_argb, uint8_t* dst_argb,
>                              int width) = ARGBUnattenuateRow_C;
>   if (!src_argb || !dst_argb || width <= 0 || height == 0) {
>     return -1;
>@@ -1936,14 +2001,14 @@ int ARGBUnattenuate(const uint8* src_argb,
> 
> // Convert ARGB to Grayed ARGB.
> LIBYUV_API
>-int ARGBGrayTo(const uint8* src_argb,
>+int ARGBGrayTo(const uint8_t* src_argb,
>                int src_stride_argb,
>-               uint8* dst_argb,
>+               uint8_t* dst_argb,
>                int dst_stride_argb,
>                int width,
>                int height) {
>   int y;
>-  void (*ARGBGrayRow)(const uint8* src_argb, uint8* dst_argb, int width) =
>+  void (*ARGBGrayRow)(const uint8_t* src_argb, uint8_t* dst_argb, int width) =
>       ARGBGrayRow_C;
>   if (!src_argb || !dst_argb || width <= 0 || height == 0) {
>     return -1;
>@@ -1985,16 +2050,16 @@ int ARGBGrayTo(const uint8* src_argb,
> 
> // Make a rectangle of ARGB gray scale.
> LIBYUV_API
>-int ARGBGray(uint8* dst_argb,
>+int ARGBGray(uint8_t* dst_argb,
>              int dst_stride_argb,
>              int dst_x,
>              int dst_y,
>              int width,
>              int height) {
>   int y;
>-  void (*ARGBGrayRow)(const uint8* src_argb, uint8* dst_argb, int width) =
>+  void (*ARGBGrayRow)(const uint8_t* src_argb, uint8_t* dst_argb, int width) =
>       ARGBGrayRow_C;
>-  uint8* dst = dst_argb + dst_y * dst_stride_argb + dst_x * 4;
>+  uint8_t* dst = dst_argb + dst_y * dst_stride_argb + dst_x * 4;
>   if (!dst_argb || width <= 0 || height <= 0 || dst_x < 0 || dst_y < 0) {
>     return -1;
>   }
>@@ -2029,15 +2094,15 @@ int ARGBGray(uint8* dst_argb,
> 
> // Make a rectangle of ARGB Sepia tone.
> LIBYUV_API
>-int ARGBSepia(uint8* dst_argb,
>+int ARGBSepia(uint8_t* dst_argb,
>               int dst_stride_argb,
>               int dst_x,
>               int dst_y,
>               int width,
>               int height) {
>   int y;
>-  void (*ARGBSepiaRow)(uint8 * dst_argb, int width) = ARGBSepiaRow_C;
>-  uint8* dst = dst_argb + dst_y * dst_stride_argb + dst_x * 4;
>+  void (*ARGBSepiaRow)(uint8_t * dst_argb, int width) = ARGBSepiaRow_C;
>+  uint8_t* dst = dst_argb + dst_y * dst_stride_argb + dst_x * 4;
>   if (!dst_argb || width <= 0 || height <= 0 || dst_x < 0 || dst_y < 0) {
>     return -1;
>   }
>@@ -2073,16 +2138,16 @@ int ARGBSepia(uint8* dst_argb,
> // Apply a 4x4 matrix to each ARGB pixel.
> // Note: Normally for shading, but can be used to swizzle or invert.
> LIBYUV_API
>-int ARGBColorMatrix(const uint8* src_argb,
>+int ARGBColorMatrix(const uint8_t* src_argb,
>                     int src_stride_argb,
>-                    uint8* dst_argb,
>+                    uint8_t* dst_argb,
>                     int dst_stride_argb,
>-                    const int8* matrix_argb,
>+                    const int8_t* matrix_argb,
>                     int width,
>                     int height) {
>   int y;
>-  void (*ARGBColorMatrixRow)(const uint8* src_argb, uint8* dst_argb,
>-                             const int8* matrix_argb, int width) =
>+  void (*ARGBColorMatrixRow)(const uint8_t* src_argb, uint8_t* dst_argb,
>+                             const int8_t* matrix_argb, int width) =
>       ARGBColorMatrixRow_C;
>   if (!src_argb || !dst_argb || !matrix_argb || width <= 0 || height == 0) {
>     return -1;
>@@ -2124,15 +2189,15 @@ int ARGBColorMatrix(const uint8* src_argb,
> // Apply a 4x3 matrix to each ARGB pixel.
> // Deprecated.
> LIBYUV_API
>-int RGBColorMatrix(uint8* dst_argb,
>+int RGBColorMatrix(uint8_t* dst_argb,
>                    int dst_stride_argb,
>-                   const int8* matrix_rgb,
>+                   const int8_t* matrix_rgb,
>                    int dst_x,
>                    int dst_y,
>                    int width,
>                    int height) {
>-  SIMD_ALIGNED(int8 matrix_argb[16]);
>-  uint8* dst = dst_argb + dst_y * dst_stride_argb + dst_x * 4;
>+  SIMD_ALIGNED(int8_t matrix_argb[16]);
>+  uint8_t* dst = dst_argb + dst_y * dst_stride_argb + dst_x * 4;
>   if (!dst_argb || !matrix_rgb || width <= 0 || height <= 0 || dst_x < 0 ||
>       dst_y < 0) {
>     return -1;
>@@ -2154,24 +2219,24 @@ int RGBColorMatrix(uint8* dst_argb,
>   matrix_argb[14] = matrix_argb[13] = matrix_argb[12] = 0;
>   matrix_argb[15] = 64;  // 1.0
> 
>-  return ARGBColorMatrix((const uint8*)(dst), dst_stride_argb, dst,
>+  return ARGBColorMatrix((const uint8_t*)(dst), dst_stride_argb, dst,
>                          dst_stride_argb, &matrix_argb[0], width, height);
> }
> 
> // Apply a color table each ARGB pixel.
> // Table contains 256 ARGB values.
> LIBYUV_API
>-int ARGBColorTable(uint8* dst_argb,
>+int ARGBColorTable(uint8_t* dst_argb,
>                    int dst_stride_argb,
>-                   const uint8* table_argb,
>+                   const uint8_t* table_argb,
>                    int dst_x,
>                    int dst_y,
>                    int width,
>                    int height) {
>   int y;
>-  void (*ARGBColorTableRow)(uint8 * dst_argb, const uint8* table_argb,
>+  void (*ARGBColorTableRow)(uint8_t * dst_argb, const uint8_t* table_argb,
>                             int width) = ARGBColorTableRow_C;
>-  uint8* dst = dst_argb + dst_y * dst_stride_argb + dst_x * 4;
>+  uint8_t* dst = dst_argb + dst_y * dst_stride_argb + dst_x * 4;
>   if (!dst_argb || !table_argb || width <= 0 || height <= 0 || dst_x < 0 ||
>       dst_y < 0) {
>     return -1;
>@@ -2197,17 +2262,17 @@ int ARGBColorTable(uint8* dst_argb,
> // Apply a color table each ARGB pixel but preserve destination alpha.
> // Table contains 256 ARGB values.
> LIBYUV_API
>-int RGBColorTable(uint8* dst_argb,
>+int RGBColorTable(uint8_t* dst_argb,
>                   int dst_stride_argb,
>-                  const uint8* table_argb,
>+                  const uint8_t* table_argb,
>                   int dst_x,
>                   int dst_y,
>                   int width,
>                   int height) {
>   int y;
>-  void (*RGBColorTableRow)(uint8 * dst_argb, const uint8* table_argb,
>+  void (*RGBColorTableRow)(uint8_t * dst_argb, const uint8_t* table_argb,
>                            int width) = RGBColorTableRow_C;
>-  uint8* dst = dst_argb + dst_y * dst_stride_argb + dst_x * 4;
>+  uint8_t* dst = dst_argb + dst_y * dst_stride_argb + dst_x * 4;
>   if (!dst_argb || !table_argb || width <= 0 || height <= 0 || dst_x < 0 ||
>       dst_y < 0) {
>     return -1;
>@@ -2240,7 +2305,7 @@ int RGBColorTable(uint8* dst_argb,
> // Caveat - although SSE2 saturates, the C function does not and should be used
> // with care if doing anything but quantization.
> LIBYUV_API
>-int ARGBQuantize(uint8* dst_argb,
>+int ARGBQuantize(uint8_t* dst_argb,
>                  int dst_stride_argb,
>                  int scale,
>                  int interval_size,
>@@ -2250,9 +2315,9 @@ int ARGBQuantize(uint8* dst_argb,
>                  int width,
>                  int height) {
>   int y;
>-  void (*ARGBQuantizeRow)(uint8 * dst_argb, int scale, int interval_size,
>+  void (*ARGBQuantizeRow)(uint8_t * dst_argb, int scale, int interval_size,
>                           int interval_offset, int width) = ARGBQuantizeRow_C;
>-  uint8* dst = dst_argb + dst_y * dst_stride_argb + dst_x * 4;
>+  uint8_t* dst = dst_argb + dst_y * dst_stride_argb + dst_x * 4;
>   if (!dst_argb || width <= 0 || height <= 0 || dst_x < 0 || dst_y < 0 ||
>       interval_size < 1 || interval_size > 255) {
>     return -1;
>@@ -2288,17 +2353,17 @@ int ARGBQuantize(uint8* dst_argb,
> // Computes table of cumulative sum for image where the value is the sum
> // of all values above and to the left of the entry. Used by ARGBBlur.
> LIBYUV_API
>-int ARGBComputeCumulativeSum(const uint8* src_argb,
>+int ARGBComputeCumulativeSum(const uint8_t* src_argb,
>                              int src_stride_argb,
>-                             int32* dst_cumsum,
>+                             int32_t* dst_cumsum,
>                              int dst_stride32_cumsum,
>                              int width,
>                              int height) {
>   int y;
>-  void (*ComputeCumulativeSumRow)(const uint8* row, int32* cumsum,
>-                                  const int32* previous_cumsum, int width) =
>+  void (*ComputeCumulativeSumRow)(const uint8_t* row, int32_t* cumsum,
>+                                  const int32_t* previous_cumsum, int width) =
>       ComputeCumulativeSumRow_C;
>-  int32* previous_cumsum = dst_cumsum;
>+  int32_t* previous_cumsum = dst_cumsum;
>   if (!dst_cumsum || !src_argb || width <= 0 || height <= 0) {
>     return -1;
>   }
>@@ -2322,25 +2387,25 @@ int ARGBComputeCumulativeSum(const uint8* src_argb,
> // aligned to 16 byte boundary. height can be radius * 2 + 2 to save memory
> // as the buffer is treated as circular.
> LIBYUV_API
>-int ARGBBlur(const uint8* src_argb,
>+int ARGBBlur(const uint8_t* src_argb,
>              int src_stride_argb,
>-             uint8* dst_argb,
>+             uint8_t* dst_argb,
>              int dst_stride_argb,
>-             int32* dst_cumsum,
>+             int32_t* dst_cumsum,
>              int dst_stride32_cumsum,
>              int width,
>              int height,
>              int radius) {
>   int y;
>-  void (*ComputeCumulativeSumRow)(const uint8* row, int32* cumsum,
>-                                  const int32* previous_cumsum, int width) =
>+  void (*ComputeCumulativeSumRow)(const uint8_t* row, int32_t* cumsum,
>+                                  const int32_t* previous_cumsum, int width) =
>       ComputeCumulativeSumRow_C;
>-  void (*CumulativeSumToAverageRow)(const int32* topleft, const int32* botleft,
>-                                    int width, int area, uint8* dst,
>-                                    int count) = CumulativeSumToAverageRow_C;
>-  int32* cumsum_bot_row;
>-  int32* max_cumsum_bot_row;
>-  int32* cumsum_top_row;
>+  void (*CumulativeSumToAverageRow)(
>+      const int32_t* topleft, const int32_t* botleft, int width, int area,
>+      uint8_t* dst, int count) = CumulativeSumToAverageRow_C;
>+  int32_t* cumsum_bot_row;
>+  int32_t* max_cumsum_bot_row;
>+  int32_t* cumsum_top_row;
> 
>   if (!src_argb || !dst_argb || width <= 0 || height == 0) {
>     return -1;
>@@ -2394,7 +2459,7 @@ int ARGBBlur(const uint8* src_argb,
>     // Increment cumsum_bot_row pointer with circular buffer wrap around and
>     // then fill in a row of CumulativeSum.
>     if ((y + radius) < height) {
>-      const int32* prev_cumsum_bot_row = cumsum_bot_row;
>+      const int32_t* prev_cumsum_bot_row = cumsum_bot_row;
>       cumsum_bot_row += dst_stride32_cumsum;
>       if (cumsum_bot_row >= max_cumsum_bot_row) {
>         cumsum_bot_row = dst_cumsum;
>@@ -2432,16 +2497,16 @@ int ARGBBlur(const uint8* src_argb,
> 
> // Multiply ARGB image by a specified ARGB value.
> LIBYUV_API
>-int ARGBShade(const uint8* src_argb,
>+int ARGBShade(const uint8_t* src_argb,
>               int src_stride_argb,
>-              uint8* dst_argb,
>+              uint8_t* dst_argb,
>               int dst_stride_argb,
>               int width,
>               int height,
>-              uint32 value) {
>+              uint32_t value) {
>   int y;
>-  void (*ARGBShadeRow)(const uint8* src_argb, uint8* dst_argb, int width,
>-                       uint32 value) = ARGBShadeRow_C;
>+  void (*ARGBShadeRow)(const uint8_t* src_argb, uint8_t* dst_argb, int width,
>+                       uint32_t value) = ARGBShadeRow_C;
>   if (!src_argb || !dst_argb || width <= 0 || height == 0 || value == 0u) {
>     return -1;
>   }
>@@ -2482,17 +2547,17 @@ int ARGBShade(const uint8* src_argb,
> 
> // Interpolate 2 planes by specified amount (0 to 255).
> LIBYUV_API
>-int InterpolatePlane(const uint8* src0,
>+int InterpolatePlane(const uint8_t* src0,
>                      int src_stride0,
>-                     const uint8* src1,
>+                     const uint8_t* src1,
>                      int src_stride1,
>-                     uint8* dst,
>+                     uint8_t* dst,
>                      int dst_stride,
>                      int width,
>                      int height,
>                      int interpolation) {
>   int y;
>-  void (*InterpolateRow)(uint8 * dst_ptr, const uint8* src_ptr,
>+  void (*InterpolateRow)(uint8_t * dst_ptr, const uint8_t* src_ptr,
>                          ptrdiff_t src_stride, int dst_width,
>                          int source_y_fraction) = InterpolateRow_C;
>   if (!src0 || !src1 || !dst || width <= 0 || height == 0) {
>@@ -2534,14 +2599,6 @@ int InterpolatePlane(const uint8* src0,
>     }
>   }
> #endif
>-#if defined(HAS_INTERPOLATEROW_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2) && IS_ALIGNED(src0, 4) &&
>-      IS_ALIGNED(src_stride0, 4) && IS_ALIGNED(src1, 4) &&
>-      IS_ALIGNED(src_stride1, 4) && IS_ALIGNED(dst, 4) &&
>-      IS_ALIGNED(dst_stride, 4) && IS_ALIGNED(width, 4)) {
>-    InterpolateRow = InterpolateRow_DSPR2;
>-  }
>-#endif
> #if defined(HAS_INTERPOLATEROW_MSA)
>   if (TestCpuFlag(kCpuHasMSA)) {
>     InterpolateRow = InterpolateRow_Any_MSA;
>@@ -2562,11 +2619,11 @@ int InterpolatePlane(const uint8* src0,
> 
> // Interpolate 2 ARGB images by specified amount (0 to 255).
> LIBYUV_API
>-int ARGBInterpolate(const uint8* src_argb0,
>+int ARGBInterpolate(const uint8_t* src_argb0,
>                     int src_stride_argb0,
>-                    const uint8* src_argb1,
>+                    const uint8_t* src_argb1,
>                     int src_stride_argb1,
>-                    uint8* dst_argb,
>+                    uint8_t* dst_argb,
>                     int dst_stride_argb,
>                     int width,
>                     int height,
>@@ -2578,23 +2635,23 @@ int ARGBInterpolate(const uint8* src_argb0,
> 
> // Interpolate 2 YUV images by specified amount (0 to 255).
> LIBYUV_API
>-int I420Interpolate(const uint8* src0_y,
>+int I420Interpolate(const uint8_t* src0_y,
>                     int src0_stride_y,
>-                    const uint8* src0_u,
>+                    const uint8_t* src0_u,
>                     int src0_stride_u,
>-                    const uint8* src0_v,
>+                    const uint8_t* src0_v,
>                     int src0_stride_v,
>-                    const uint8* src1_y,
>+                    const uint8_t* src1_y,
>                     int src1_stride_y,
>-                    const uint8* src1_u,
>+                    const uint8_t* src1_u,
>                     int src1_stride_u,
>-                    const uint8* src1_v,
>+                    const uint8_t* src1_v,
>                     int src1_stride_v,
>-                    uint8* dst_y,
>+                    uint8_t* dst_y,
>                     int dst_stride_y,
>-                    uint8* dst_u,
>+                    uint8_t* dst_u,
>                     int dst_stride_u,
>-                    uint8* dst_v,
>+                    uint8_t* dst_v,
>                     int dst_stride_v,
>                     int width,
>                     int height,
>@@ -2616,16 +2673,16 @@ int I420Interpolate(const uint8* src0_y,
> 
> // Shuffle ARGB channel order.  e.g. BGRA to ARGB.
> LIBYUV_API
>-int ARGBShuffle(const uint8* src_bgra,
>+int ARGBShuffle(const uint8_t* src_bgra,
>                 int src_stride_bgra,
>-                uint8* dst_argb,
>+                uint8_t* dst_argb,
>                 int dst_stride_argb,
>-                const uint8* shuffler,
>+                const uint8_t* shuffler,
>                 int width,
>                 int height) {
>   int y;
>-  void (*ARGBShuffleRow)(const uint8* src_bgra, uint8* dst_argb,
>-                         const uint8* shuffler, int width) = ARGBShuffleRow_C;
>+  void (*ARGBShuffleRow)(const uint8_t* src_bgra, uint8_t* dst_argb,
>+                         const uint8_t* shuffler, int width) = ARGBShuffleRow_C;
>   if (!src_bgra || !dst_argb || width <= 0 || height == 0) {
>     return -1;
>   }
>@@ -2641,14 +2698,6 @@ int ARGBShuffle(const uint8* src_bgra,
>     height = 1;
>     src_stride_bgra = dst_stride_argb = 0;
>   }
>-#if defined(HAS_ARGBSHUFFLEROW_SSE2)
>-  if (TestCpuFlag(kCpuHasSSE2)) {
>-    ARGBShuffleRow = ARGBShuffleRow_Any_SSE2;
>-    if (IS_ALIGNED(width, 4)) {
>-      ARGBShuffleRow = ARGBShuffleRow_SSE2;
>-    }
>-  }
>-#endif
> #if defined(HAS_ARGBSHUFFLEROW_SSSE3)
>   if (TestCpuFlag(kCpuHasSSSE3)) {
>     ARGBShuffleRow = ARGBShuffleRow_Any_SSSE3;
>@@ -2691,23 +2740,23 @@ int ARGBShuffle(const uint8* src_bgra,
> }
> 
> // Sobel ARGB effect.
>-static int ARGBSobelize(const uint8* src_argb,
>+static int ARGBSobelize(const uint8_t* src_argb,
>                         int src_stride_argb,
>-                        uint8* dst_argb,
>+                        uint8_t* dst_argb,
>                         int dst_stride_argb,
>                         int width,
>                         int height,
>-                        void (*SobelRow)(const uint8* src_sobelx,
>-                                         const uint8* src_sobely,
>-                                         uint8* dst,
>+                        void (*SobelRow)(const uint8_t* src_sobelx,
>+                                         const uint8_t* src_sobely,
>+                                         uint8_t* dst,
>                                          int width)) {
>   int y;
>-  void (*ARGBToYJRow)(const uint8* src_argb, uint8* dst_g, int width) =
>+  void (*ARGBToYJRow)(const uint8_t* src_argb, uint8_t* dst_g, int width) =
>       ARGBToYJRow_C;
>-  void (*SobelYRow)(const uint8* src_y0, const uint8* src_y1, uint8* dst_sobely,
>-                    int width) = SobelYRow_C;
>-  void (*SobelXRow)(const uint8* src_y0, const uint8* src_y1,
>-                    const uint8* src_y2, uint8* dst_sobely, int width) =
>+  void (*SobelYRow)(const uint8_t* src_y0, const uint8_t* src_y1,
>+                    uint8_t* dst_sobely, int width) = SobelYRow_C;
>+  void (*SobelXRow)(const uint8_t* src_y0, const uint8_t* src_y1,
>+                    const uint8_t* src_y2, uint8_t* dst_sobely, int width) =
>       SobelXRow_C;
>   const int kEdge = 16;  // Extra pixels at start of row for extrude/align.
>   if (!src_argb || !dst_argb || width <= 0 || height == 0) {
>@@ -2787,14 +2836,14 @@ static int ARGBSobelize(const uint8* src_argb,
>     // 3 rows with edges before/after.
>     const int kRowSize = (width + kEdge + 31) & ~31;
>     align_buffer_64(rows, kRowSize * 2 + (kEdge + kRowSize * 3 + kEdge));
>-    uint8* row_sobelx = rows;
>-    uint8* row_sobely = rows + kRowSize;
>-    uint8* row_y = rows + kRowSize * 2;
>+    uint8_t* row_sobelx = rows;
>+    uint8_t* row_sobely = rows + kRowSize;
>+    uint8_t* row_y = rows + kRowSize * 2;
> 
>     // Convert first row.
>-    uint8* row_y0 = row_y + kEdge;
>-    uint8* row_y1 = row_y0 + kRowSize;
>-    uint8* row_y2 = row_y1 + kRowSize;
>+    uint8_t* row_y0 = row_y + kEdge;
>+    uint8_t* row_y1 = row_y0 + kRowSize;
>+    uint8_t* row_y2 = row_y1 + kRowSize;
>     ARGBToYJRow(src_argb, row_y0, width);
>     row_y0[-1] = row_y0[0];
>     memset(row_y0 + width, row_y0[width - 1], 16);  // Extrude 16 for valgrind.
>@@ -2818,7 +2867,7 @@ static int ARGBSobelize(const uint8* src_argb,
> 
>       // Cycle thru circular queue of 3 row_y buffers.
>       {
>-        uint8* row_yt = row_y0;
>+        uint8_t* row_yt = row_y0;
>         row_y0 = row_y1;
>         row_y1 = row_y2;
>         row_y2 = row_yt;
>@@ -2833,14 +2882,14 @@ static int ARGBSobelize(const uint8* src_argb,
> 
> // Sobel ARGB effect.
> LIBYUV_API
>-int ARGBSobel(const uint8* src_argb,
>+int ARGBSobel(const uint8_t* src_argb,
>               int src_stride_argb,
>-              uint8* dst_argb,
>+              uint8_t* dst_argb,
>               int dst_stride_argb,
>               int width,
>               int height) {
>-  void (*SobelRow)(const uint8* src_sobelx, const uint8* src_sobely,
>-                   uint8* dst_argb, int width) = SobelRow_C;
>+  void (*SobelRow)(const uint8_t* src_sobelx, const uint8_t* src_sobely,
>+                   uint8_t* dst_argb, int width) = SobelRow_C;
> #if defined(HAS_SOBELROW_SSE2)
>   if (TestCpuFlag(kCpuHasSSE2)) {
>     SobelRow = SobelRow_Any_SSE2;
>@@ -2871,14 +2920,14 @@ int ARGBSobel(const uint8* src_argb,
> 
> // Sobel ARGB effect with planar output.
> LIBYUV_API
>-int ARGBSobelToPlane(const uint8* src_argb,
>+int ARGBSobelToPlane(const uint8_t* src_argb,
>                      int src_stride_argb,
>-                     uint8* dst_y,
>+                     uint8_t* dst_y,
>                      int dst_stride_y,
>                      int width,
>                      int height) {
>-  void (*SobelToPlaneRow)(const uint8* src_sobelx, const uint8* src_sobely,
>-                          uint8* dst_, int width) = SobelToPlaneRow_C;
>+  void (*SobelToPlaneRow)(const uint8_t* src_sobelx, const uint8_t* src_sobely,
>+                          uint8_t* dst_, int width) = SobelToPlaneRow_C;
> #if defined(HAS_SOBELTOPLANEROW_SSE2)
>   if (TestCpuFlag(kCpuHasSSE2)) {
>     SobelToPlaneRow = SobelToPlaneRow_Any_SSE2;
>@@ -2910,14 +2959,14 @@ int ARGBSobelToPlane(const uint8* src_argb,
> // SobelXY ARGB effect.
> // Similar to Sobel, but also stores Sobel X in R and Sobel Y in B.  G = Sobel.
> LIBYUV_API
>-int ARGBSobelXY(const uint8* src_argb,
>+int ARGBSobelXY(const uint8_t* src_argb,
>                 int src_stride_argb,
>-                uint8* dst_argb,
>+                uint8_t* dst_argb,
>                 int dst_stride_argb,
>                 int width,
>                 int height) {
>-  void (*SobelXYRow)(const uint8* src_sobelx, const uint8* src_sobely,
>-                     uint8* dst_argb, int width) = SobelXYRow_C;
>+  void (*SobelXYRow)(const uint8_t* src_sobelx, const uint8_t* src_sobely,
>+                     uint8_t* dst_argb, int width) = SobelXYRow_C;
> #if defined(HAS_SOBELXYROW_SSE2)
>   if (TestCpuFlag(kCpuHasSSE2)) {
>     SobelXYRow = SobelXYRow_Any_SSE2;
>@@ -2948,15 +2997,15 @@ int ARGBSobelXY(const uint8* src_argb,
> 
> // Apply a 4x4 polynomial to each ARGB pixel.
> LIBYUV_API
>-int ARGBPolynomial(const uint8* src_argb,
>+int ARGBPolynomial(const uint8_t* src_argb,
>                    int src_stride_argb,
>-                   uint8* dst_argb,
>+                   uint8_t* dst_argb,
>                    int dst_stride_argb,
>                    const float* poly,
>                    int width,
>                    int height) {
>   int y;
>-  void (*ARGBPolynomialRow)(const uint8* src_argb, uint8* dst_argb,
>+  void (*ARGBPolynomialRow)(const uint8_t* src_argb, uint8_t* dst_argb,
>                             const float* poly, int width) = ARGBPolynomialRow_C;
>   if (!src_argb || !dst_argb || !poly || width <= 0 || height == 0) {
>     return -1;
>@@ -2996,16 +3045,16 @@ int ARGBPolynomial(const uint8* src_argb,
> // Convert plane of 16 bit shorts to half floats.
> // Source values are multiplied by scale before storing as half float.
> LIBYUV_API
>-int HalfFloatPlane(const uint16* src_y,
>+int HalfFloatPlane(const uint16_t* src_y,
>                    int src_stride_y,
>-                   uint16* dst_y,
>+                   uint16_t* dst_y,
>                    int dst_stride_y,
>                    float scale,
>                    int width,
>                    int height) {
>   int y;
>-  void (*HalfFloatRow)(const uint16* src, uint16* dst, float scale, int width) =
>-      HalfFloatRow_C;
>+  void (*HalfFloatRow)(const uint16_t* src, uint16_t* dst, float scale,
>+                       int width) = HalfFloatRow_C;
>   if (!src_y || !dst_y || width <= 0 || height == 0) {
>     return -1;
>   }
>@@ -3074,19 +3123,40 @@ int HalfFloatPlane(const uint16* src_y,
>   return 0;
> }
> 
>+// Convert a buffer of bytes to floats, scale the values and store as floats.
>+LIBYUV_API
>+int ByteToFloat(const uint8_t* src_y, float* dst_y, float scale, int width) {
>+  void (*ByteToFloatRow)(const uint8_t* src, float* dst, float scale,
>+                         int width) = ByteToFloatRow_C;
>+  if (!src_y || !dst_y || width <= 0) {
>+    return -1;
>+  }
>+#if defined(HAS_BYTETOFLOATROW_NEON)
>+  if (TestCpuFlag(kCpuHasNEON)) {
>+    ByteToFloatRow = ByteToFloatRow_Any_NEON;
>+    if (IS_ALIGNED(width, 8)) {
>+      ByteToFloatRow = ByteToFloatRow_NEON;
>+    }
>+  }
>+#endif
>+
>+  ByteToFloatRow(src_y, dst_y, scale, width);
>+  return 0;
>+}
>+
> // Apply a lumacolortable to each ARGB pixel.
> LIBYUV_API
>-int ARGBLumaColorTable(const uint8* src_argb,
>+int ARGBLumaColorTable(const uint8_t* src_argb,
>                        int src_stride_argb,
>-                       uint8* dst_argb,
>+                       uint8_t* dst_argb,
>                        int dst_stride_argb,
>-                       const uint8* luma,
>+                       const uint8_t* luma,
>                        int width,
>                        int height) {
>   int y;
>   void (*ARGBLumaColorTableRow)(
>-      const uint8* src_argb, uint8* dst_argb, int width, const uint8* luma,
>-      const uint32 lumacoeff) = ARGBLumaColorTableRow_C;
>+      const uint8_t* src_argb, uint8_t* dst_argb, int width,
>+      const uint8_t* luma, const uint32_t lumacoeff) = ARGBLumaColorTableRow_C;
>   if (!src_argb || !dst_argb || !luma || width <= 0 || height == 0) {
>     return -1;
>   }
>@@ -3118,15 +3188,15 @@ int ARGBLumaColorTable(const uint8* src_argb,
> 
> // Copy Alpha from one ARGB image to another.
> LIBYUV_API
>-int ARGBCopyAlpha(const uint8* src_argb,
>+int ARGBCopyAlpha(const uint8_t* src_argb,
>                   int src_stride_argb,
>-                  uint8* dst_argb,
>+                  uint8_t* dst_argb,
>                   int dst_stride_argb,
>                   int width,
>                   int height) {
>   int y;
>-  void (*ARGBCopyAlphaRow)(const uint8* src_argb, uint8* dst_argb, int width) =
>-      ARGBCopyAlphaRow_C;
>+  void (*ARGBCopyAlphaRow)(const uint8_t* src_argb, uint8_t* dst_argb,
>+                           int width) = ARGBCopyAlphaRow_C;
>   if (!src_argb || !dst_argb || width <= 0 || height == 0) {
>     return -1;
>   }
>@@ -3169,10 +3239,10 @@ int ARGBCopyAlpha(const uint8* src_argb,
> 
> // Extract just the alpha channel from ARGB.
> LIBYUV_API
>-int ARGBExtractAlpha(const uint8* src_argb,
>-                     int src_stride,
>-                     uint8* dst_a,
>-                     int dst_stride,
>+int ARGBExtractAlpha(const uint8_t* src_argb,
>+                     int src_stride_argb,
>+                     uint8_t* dst_a,
>+                     int dst_stride_a,
>                      int width,
>                      int height) {
>   if (!src_argb || !dst_a || width <= 0 || height == 0) {
>@@ -3181,17 +3251,17 @@ int ARGBExtractAlpha(const uint8* src_argb,
>   // Negative height means invert the image.
>   if (height < 0) {
>     height = -height;
>-    src_argb += (height - 1) * src_stride;
>-    src_stride = -src_stride;
>+    src_argb += (height - 1) * src_stride_argb;
>+    src_stride_argb = -src_stride_argb;
>   }
>   // Coalesce rows.
>-  if (src_stride == width * 4 && dst_stride == width) {
>+  if (src_stride_argb == width * 4 && dst_stride_a == width) {
>     width *= height;
>     height = 1;
>-    src_stride = dst_stride = 0;
>+    src_stride_argb = dst_stride_a = 0;
>   }
>-  void (*ARGBExtractAlphaRow)(const uint8* src_argb, uint8* dst_a, int width) =
>-      ARGBExtractAlphaRow_C;
>+  void (*ARGBExtractAlphaRow)(const uint8_t* src_argb, uint8_t* dst_a,
>+                              int width) = ARGBExtractAlphaRow_C;
> #if defined(HAS_ARGBEXTRACTALPHAROW_SSE2)
>   if (TestCpuFlag(kCpuHasSSE2)) {
>     ARGBExtractAlphaRow = IS_ALIGNED(width, 8) ? ARGBExtractAlphaRow_SSE2
>@@ -3219,23 +3289,23 @@ int ARGBExtractAlpha(const uint8* src_argb,
> 
>   for (int y = 0; y < height; ++y) {
>     ARGBExtractAlphaRow(src_argb, dst_a, width);
>-    src_argb += src_stride;
>-    dst_a += dst_stride;
>+    src_argb += src_stride_argb;
>+    dst_a += dst_stride_a;
>   }
>   return 0;
> }
> 
> // Copy a planar Y channel to the alpha channel of a destination ARGB image.
> LIBYUV_API
>-int ARGBCopyYToAlpha(const uint8* src_y,
>+int ARGBCopyYToAlpha(const uint8_t* src_y,
>                      int src_stride_y,
>-                     uint8* dst_argb,
>+                     uint8_t* dst_argb,
>                      int dst_stride_argb,
>                      int width,
>                      int height) {
>   int y;
>-  void (*ARGBCopyYToAlphaRow)(const uint8* src_y, uint8* dst_argb, int width) =
>-      ARGBCopyYToAlphaRow_C;
>+  void (*ARGBCopyYToAlphaRow)(const uint8_t* src_y, uint8_t* dst_argb,
>+                              int width) = ARGBCopyYToAlphaRow_C;
>   if (!src_y || !dst_argb || width <= 0 || height == 0) {
>     return -1;
>   }
>@@ -3280,19 +3350,19 @@ int ARGBCopyYToAlpha(const uint8* src_y,
> // directly. A SplitUVRow_Odd function could copy the remaining chroma.
> 
> LIBYUV_API
>-int YUY2ToNV12(const uint8* src_yuy2,
>+int YUY2ToNV12(const uint8_t* src_yuy2,
>                int src_stride_yuy2,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_uv,
>+               uint8_t* dst_uv,
>                int dst_stride_uv,
>                int width,
>                int height) {
>   int y;
>   int halfwidth = (width + 1) >> 1;
>-  void (*SplitUVRow)(const uint8* src_uv, uint8* dst_u, uint8* dst_v,
>+  void (*SplitUVRow)(const uint8_t* src_uv, uint8_t* dst_u, uint8_t* dst_v,
>                      int width) = SplitUVRow_C;
>-  void (*InterpolateRow)(uint8 * dst_ptr, const uint8* src_ptr,
>+  void (*InterpolateRow)(uint8_t * dst_ptr, const uint8_t* src_ptr,
>                          ptrdiff_t src_stride, int dst_width,
>                          int source_y_fraction) = InterpolateRow_C;
>   if (!src_yuy2 || !dst_y || !dst_uv || width <= 0 || height == 0) {
>@@ -3396,19 +3466,19 @@ int YUY2ToNV12(const uint8* src_yuy2,
> }
> 
> LIBYUV_API
>-int UYVYToNV12(const uint8* src_uyvy,
>+int UYVYToNV12(const uint8_t* src_uyvy,
>                int src_stride_uyvy,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_uv,
>+               uint8_t* dst_uv,
>                int dst_stride_uv,
>                int width,
>                int height) {
>   int y;
>   int halfwidth = (width + 1) >> 1;
>-  void (*SplitUVRow)(const uint8* src_uv, uint8* dst_u, uint8* dst_v,
>+  void (*SplitUVRow)(const uint8_t* src_uv, uint8_t* dst_u, uint8_t* dst_v,
>                      int width) = SplitUVRow_C;
>-  void (*InterpolateRow)(uint8 * dst_ptr, const uint8* src_ptr,
>+  void (*InterpolateRow)(uint8_t * dst_ptr, const uint8_t* src_ptr,
>                          ptrdiff_t src_stride, int dst_width,
>                          int source_y_fraction) = InterpolateRow_C;
>   if (!src_uyvy || !dst_y || !dst_uv || width <= 0 || height == 0) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/rotate.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/rotate.cc
>index b16af507185..f2bed85b755 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/rotate.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/rotate.cc
>@@ -22,18 +22,18 @@ extern "C" {
> #endif
> 
> LIBYUV_API
>-void TransposePlane(const uint8* src,
>+void TransposePlane(const uint8_t* src,
>                     int src_stride,
>-                    uint8* dst,
>+                    uint8_t* dst,
>                     int dst_stride,
>                     int width,
>                     int height) {
>   int i = height;
> #if defined(HAS_TRANSPOSEWX16_MSA)
>-  void (*TransposeWx16)(const uint8* src, int src_stride, uint8* dst,
>+  void (*TransposeWx16)(const uint8_t* src, int src_stride, uint8_t* dst,
>                         int dst_stride, int width) = TransposeWx16_C;
> #else
>-  void (*TransposeWx8)(const uint8* src, int src_stride, uint8* dst,
>+  void (*TransposeWx8)(const uint8_t* src, int src_stride, uint8_t* dst,
>                        int dst_stride, int width) = TransposeWx8_C;
> #endif
> #if defined(HAS_TRANSPOSEWX8_NEON)
>@@ -57,16 +57,6 @@ void TransposePlane(const uint8* src,
>     }
>   }
> #endif
>-#if defined(HAS_TRANSPOSEWX8_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2)) {
>-    if (IS_ALIGNED(width, 4) && IS_ALIGNED(src, 4) &&
>-        IS_ALIGNED(src_stride, 4)) {
>-      TransposeWx8 = TransposeWx8_Fast_DSPR2;
>-    } else {
>-      TransposeWx8 = TransposeWx8_DSPR2;
>-    }
>-  }
>-#endif
> #if defined(HAS_TRANSPOSEWX16_MSA)
>   if (TestCpuFlag(kCpuHasMSA)) {
>     TransposeWx16 = TransposeWx16_Any_MSA;
>@@ -100,9 +90,9 @@ void TransposePlane(const uint8* src,
> }
> 
> LIBYUV_API
>-void RotatePlane90(const uint8* src,
>+void RotatePlane90(const uint8_t* src,
>                    int src_stride,
>-                   uint8* dst,
>+                   uint8_t* dst,
>                    int dst_stride,
>                    int width,
>                    int height) {
>@@ -115,9 +105,9 @@ void RotatePlane90(const uint8* src,
> }
> 
> LIBYUV_API
>-void RotatePlane270(const uint8* src,
>+void RotatePlane270(const uint8_t* src,
>                     int src_stride,
>-                    uint8* dst,
>+                    uint8_t* dst,
>                     int dst_stride,
>                     int width,
>                     int height) {
>@@ -130,20 +120,20 @@ void RotatePlane270(const uint8* src,
> }
> 
> LIBYUV_API
>-void RotatePlane180(const uint8* src,
>+void RotatePlane180(const uint8_t* src,
>                     int src_stride,
>-                    uint8* dst,
>+                    uint8_t* dst,
>                     int dst_stride,
>                     int width,
>                     int height) {
>   // Swap first and last row and mirror the content. Uses a temporary row.
>   align_buffer_64(row, width);
>-  const uint8* src_bot = src + src_stride * (height - 1);
>-  uint8* dst_bot = dst + dst_stride * (height - 1);
>+  const uint8_t* src_bot = src + src_stride * (height - 1);
>+  uint8_t* dst_bot = dst + dst_stride * (height - 1);
>   int half_height = (height + 1) >> 1;
>   int y;
>-  void (*MirrorRow)(const uint8* src, uint8* dst, int width) = MirrorRow_C;
>-  void (*CopyRow)(const uint8* src, uint8* dst, int width) = CopyRow_C;
>+  void (*MirrorRow)(const uint8_t* src, uint8_t* dst, int width) = MirrorRow_C;
>+  void (*CopyRow)(const uint8_t* src, uint8_t* dst, int width) = CopyRow_C;
> #if defined(HAS_MIRRORROW_NEON)
>   if (TestCpuFlag(kCpuHasNEON)) {
>     MirrorRow = MirrorRow_Any_NEON;
>@@ -168,14 +158,6 @@ void RotatePlane180(const uint8* src,
>     }
>   }
> #endif
>-// TODO(fbarchard): Mirror on mips handle unaligned memory.
>-#if defined(HAS_MIRRORROW_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2) && IS_ALIGNED(src, 4) &&
>-      IS_ALIGNED(src_stride, 4) && IS_ALIGNED(dst, 4) &&
>-      IS_ALIGNED(dst_stride, 4)) {
>-    MirrorRow = MirrorRow_DSPR2;
>-  }
>-#endif
> #if defined(HAS_MIRRORROW_MSA)
>   if (TestCpuFlag(kCpuHasMSA)) {
>     MirrorRow = MirrorRow_Any_MSA;
>@@ -204,11 +186,6 @@ void RotatePlane180(const uint8* src,
>     CopyRow = IS_ALIGNED(width, 32) ? CopyRow_NEON : CopyRow_Any_NEON;
>   }
> #endif
>-#if defined(HAS_COPYROW_MIPS)
>-  if (TestCpuFlag(kCpuHasMIPS)) {
>-    CopyRow = CopyRow_MIPS;
>-  }
>-#endif
> 
>   // Odd height will harmlessly mirror the middle row twice.
>   for (y = 0; y < half_height; ++y) {
>@@ -224,22 +201,22 @@ void RotatePlane180(const uint8* src,
> }
> 
> LIBYUV_API
>-void TransposeUV(const uint8* src,
>+void TransposeUV(const uint8_t* src,
>                  int src_stride,
>-                 uint8* dst_a,
>+                 uint8_t* dst_a,
>                  int dst_stride_a,
>-                 uint8* dst_b,
>+                 uint8_t* dst_b,
>                  int dst_stride_b,
>                  int width,
>                  int height) {
>   int i = height;
> #if defined(HAS_TRANSPOSEUVWX16_MSA)
>-  void (*TransposeUVWx16)(const uint8* src, int src_stride, uint8* dst_a,
>-                          int dst_stride_a, uint8* dst_b, int dst_stride_b,
>+  void (*TransposeUVWx16)(const uint8_t* src, int src_stride, uint8_t* dst_a,
>+                          int dst_stride_a, uint8_t* dst_b, int dst_stride_b,
>                           int width) = TransposeUVWx16_C;
> #else
>-  void (*TransposeUVWx8)(const uint8* src, int src_stride, uint8* dst_a,
>-                         int dst_stride_a, uint8* dst_b, int dst_stride_b,
>+  void (*TransposeUVWx8)(const uint8_t* src, int src_stride, uint8_t* dst_a,
>+                         int dst_stride_a, uint8_t* dst_b, int dst_stride_b,
>                          int width) = TransposeUVWx8_C;
> #endif
> #if defined(HAS_TRANSPOSEUVWX8_NEON)
>@@ -255,12 +232,6 @@ void TransposeUV(const uint8* src,
>     }
>   }
> #endif
>-#if defined(HAS_TRANSPOSEUVWX8_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2) && IS_ALIGNED(width, 2) && IS_ALIGNED(src, 4) &&
>-      IS_ALIGNED(src_stride, 4)) {
>-    TransposeUVWx8 = TransposeUVWx8_DSPR2;
>-  }
>-#endif
> #if defined(HAS_TRANSPOSEUVWX16_MSA)
>   if (TestCpuFlag(kCpuHasMSA)) {
>     TransposeUVWx16 = TransposeUVWx16_Any_MSA;
>@@ -299,11 +270,11 @@ void TransposeUV(const uint8* src,
> }
> 
> LIBYUV_API
>-void RotateUV90(const uint8* src,
>+void RotateUV90(const uint8_t* src,
>                 int src_stride,
>-                uint8* dst_a,
>+                uint8_t* dst_a,
>                 int dst_stride_a,
>-                uint8* dst_b,
>+                uint8_t* dst_b,
>                 int dst_stride_b,
>                 int width,
>                 int height) {
>@@ -315,11 +286,11 @@ void RotateUV90(const uint8* src,
> }
> 
> LIBYUV_API
>-void RotateUV270(const uint8* src,
>+void RotateUV270(const uint8_t* src,
>                  int src_stride,
>-                 uint8* dst_a,
>+                 uint8_t* dst_a,
>                  int dst_stride_a,
>-                 uint8* dst_b,
>+                 uint8_t* dst_b,
>                  int dst_stride_b,
>                  int width,
>                  int height) {
>@@ -334,17 +305,17 @@ void RotateUV270(const uint8* src,
> 
> // Rotate 180 is a horizontal and vertical flip.
> LIBYUV_API
>-void RotateUV180(const uint8* src,
>+void RotateUV180(const uint8_t* src,
>                  int src_stride,
>-                 uint8* dst_a,
>+                 uint8_t* dst_a,
>                  int dst_stride_a,
>-                 uint8* dst_b,
>+                 uint8_t* dst_b,
>                  int dst_stride_b,
>                  int width,
>                  int height) {
>   int i;
>-  void (*MirrorUVRow)(const uint8* src, uint8* dst_u, uint8* dst_v, int width) =
>-      MirrorUVRow_C;
>+  void (*MirrorUVRow)(const uint8_t* src, uint8_t* dst_u, uint8_t* dst_v,
>+                      int width) = MirrorUVRow_C;
> #if defined(HAS_MIRRORUVROW_NEON)
>   if (TestCpuFlag(kCpuHasNEON) && IS_ALIGNED(width, 8)) {
>     MirrorUVRow = MirrorUVRow_NEON;
>@@ -355,12 +326,6 @@ void RotateUV180(const uint8* src,
>     MirrorUVRow = MirrorUVRow_SSSE3;
>   }
> #endif
>-#if defined(HAS_MIRRORUVROW_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2) && IS_ALIGNED(src, 4) &&
>-      IS_ALIGNED(src_stride, 4)) {
>-    MirrorUVRow = MirrorUVRow_DSPR2;
>-  }
>-#endif
> #if defined(HAS_MIRRORUVROW_MSA)
>   if (TestCpuFlag(kCpuHasMSA) && IS_ALIGNED(width, 32)) {
>     MirrorUVRow = MirrorUVRow_MSA;
>@@ -379,9 +344,9 @@ void RotateUV180(const uint8* src,
> }
> 
> LIBYUV_API
>-int RotatePlane(const uint8* src,
>+int RotatePlane(const uint8_t* src,
>                 int src_stride,
>-                uint8* dst,
>+                uint8_t* dst,
>                 int dst_stride,
>                 int width,
>                 int height,
>@@ -418,17 +383,17 @@ int RotatePlane(const uint8* src,
> }
> 
> LIBYUV_API
>-int I420Rotate(const uint8* src_y,
>+int I420Rotate(const uint8_t* src_y,
>                int src_stride_y,
>-               const uint8* src_u,
>+               const uint8_t* src_u,
>                int src_stride_u,
>-               const uint8* src_v,
>+               const uint8_t* src_v,
>                int src_stride_v,
>-               uint8* dst_y,
>+               uint8_t* dst_y,
>                int dst_stride_y,
>-               uint8* dst_u,
>+               uint8_t* dst_u,
>                int dst_stride_u,
>-               uint8* dst_v,
>+               uint8_t* dst_v,
>                int dst_stride_v,
>                int width,
>                int height,
>@@ -486,15 +451,15 @@ int I420Rotate(const uint8* src_y,
> }
> 
> LIBYUV_API
>-int NV12ToI420Rotate(const uint8* src_y,
>+int NV12ToI420Rotate(const uint8_t* src_y,
>                      int src_stride_y,
>-                     const uint8* src_uv,
>+                     const uint8_t* src_uv,
>                      int src_stride_uv,
>-                     uint8* dst_y,
>+                     uint8_t* dst_y,
>                      int dst_stride_y,
>-                     uint8* dst_u,
>+                     uint8_t* dst_u,
>                      int dst_stride_u,
>-                     uint8* dst_v,
>+                     uint8_t* dst_v,
>                      int dst_stride_v,
>                      int width,
>                      int height,
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/rotate_any.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/rotate_any.cc
>index 562096b926c..c2752e6222c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/rotate_any.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/rotate_any.cc
>@@ -19,8 +19,8 @@ extern "C" {
> #endif
> 
> #define TANY(NAMEANY, TPOS_SIMD, MASK)                                        \
>-  void NAMEANY(const uint8* src, int src_stride, uint8* dst, int dst_stride,  \
>-               int width) {                                                   \
>+  void NAMEANY(const uint8_t* src, int src_stride, uint8_t* dst,              \
>+               int dst_stride, int width) {                                   \
>     int r = width & MASK;                                                     \
>     int n = width - r;                                                        \
>     if (n > 0) {                                                              \
>@@ -38,17 +38,15 @@ TANY(TransposeWx8_Any_SSSE3, TransposeWx8_SSSE3, 7)
> #ifdef HAS_TRANSPOSEWX8_FAST_SSSE3
> TANY(TransposeWx8_Fast_Any_SSSE3, TransposeWx8_Fast_SSSE3, 15)
> #endif
>-#ifdef HAS_TRANSPOSEWX8_DSPR2
>-TANY(TransposeWx8_Any_DSPR2, TransposeWx8_DSPR2, 7)
>-#endif
> #ifdef HAS_TRANSPOSEWX16_MSA
> TANY(TransposeWx16_Any_MSA, TransposeWx16_MSA, 15)
> #endif
> #undef TANY
> 
> #define TUVANY(NAMEANY, TPOS_SIMD, MASK)                                       \
>-  void NAMEANY(const uint8* src, int src_stride, uint8* dst_a,                 \
>-               int dst_stride_a, uint8* dst_b, int dst_stride_b, int width) {  \
>+  void NAMEANY(const uint8_t* src, int src_stride, uint8_t* dst_a,             \
>+               int dst_stride_a, uint8_t* dst_b, int dst_stride_b,             \
>+               int width) {                                                    \
>     int r = width & MASK;                                                      \
>     int n = width - r;                                                         \
>     if (n > 0) {                                                               \
>@@ -64,9 +62,6 @@ TUVANY(TransposeUVWx8_Any_NEON, TransposeUVWx8_NEON, 7)
> #ifdef HAS_TRANSPOSEUVWX8_SSE2
> TUVANY(TransposeUVWx8_Any_SSE2, TransposeUVWx8_SSE2, 7)
> #endif
>-#ifdef HAS_TRANSPOSEUVWX8_DSPR2
>-TUVANY(TransposeUVWx8_Any_DSPR2, TransposeUVWx8_DSPR2, 7)
>-#endif
> #ifdef HAS_TRANSPOSEUVWX16_MSA
> TUVANY(TransposeUVWx16_Any_MSA, TransposeUVWx16_MSA, 7)
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/rotate_argb.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/rotate_argb.cc
>index ede4eafa6c2..5a6e05376f1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/rotate_argb.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/rotate_argb.cc
>@@ -14,113 +14,102 @@
> #include "libyuv/cpu_id.h"
> #include "libyuv/planar_functions.h"
> #include "libyuv/row.h"
>+#include "libyuv/scale_row.h" /* for ScaleARGBRowDownEven_ */
> 
> #ifdef __cplusplus
> namespace libyuv {
> extern "C" {
> #endif
> 
>-// ARGBScale has a function to copy pixels to a row, striding each source
>-// pixel by a constant.
>-#if !defined(LIBYUV_DISABLE_X86) &&                          \
>-    (defined(_M_IX86) ||                                     \
>-     (defined(__x86_64__) && !defined(__native_client__)) || \
>-     defined(__i386__))
>-#define HAS_SCALEARGBROWDOWNEVEN_SSE2
>-void ScaleARGBRowDownEven_SSE2(const uint8* src_ptr,
>-                               int src_stride,
>-                               int src_stepx,
>-                               uint8* dst_ptr,
>-                               int dst_width);
>-#endif
>-#if !defined(LIBYUV_DISABLE_NEON) && !defined(__native_client__) && \
>-    (defined(__ARM_NEON__) || defined(LIBYUV_NEON) || defined(__aarch64__))
>-#define HAS_SCALEARGBROWDOWNEVEN_NEON
>-void ScaleARGBRowDownEven_NEON(const uint8* src_ptr,
>-                               int src_stride,
>-                               int src_stepx,
>-                               uint8* dst_ptr,
>-                               int dst_width);
>-#endif
>-
>-void ScaleARGBRowDownEven_C(const uint8* src_ptr,
>-                            int,
>-                            int src_stepx,
>-                            uint8* dst_ptr,
>-                            int dst_width);
>-
>-static void ARGBTranspose(const uint8* src,
>-                          int src_stride,
>-                          uint8* dst,
>-                          int dst_stride,
>+static void ARGBTranspose(const uint8_t* src_argb,
>+                          int src_stride_argb,
>+                          uint8_t* dst_argb,
>+                          int dst_stride_argb,
>                           int width,
>                           int height) {
>   int i;
>-  int src_pixel_step = src_stride >> 2;
>-  void (*ScaleARGBRowDownEven)(const uint8* src_ptr, int src_stride,
>-                               int src_step, uint8* dst_ptr, int dst_width) =
>-      ScaleARGBRowDownEven_C;
>+  int src_pixel_step = src_stride_argb >> 2;
>+  void (*ScaleARGBRowDownEven)(
>+      const uint8_t* src_argb, ptrdiff_t src_stride_argb, int src_step,
>+      uint8_t* dst_argb, int dst_width) = ScaleARGBRowDownEven_C;
> #if defined(HAS_SCALEARGBROWDOWNEVEN_SSE2)
>-  if (TestCpuFlag(kCpuHasSSE2) && IS_ALIGNED(height, 4)) {  // Width of dest.
>-    ScaleARGBRowDownEven = ScaleARGBRowDownEven_SSE2;
>+  if (TestCpuFlag(kCpuHasSSE2)) {
>+    ScaleARGBRowDownEven = ScaleARGBRowDownEven_Any_SSE2;
>+    if (IS_ALIGNED(height, 4)) {  // Width of dest.
>+      ScaleARGBRowDownEven = ScaleARGBRowDownEven_SSE2;
>+    }
>   }
> #endif
> #if defined(HAS_SCALEARGBROWDOWNEVEN_NEON)
>-  if (TestCpuFlag(kCpuHasNEON) && IS_ALIGNED(height, 4)) {  // Width of dest.
>-    ScaleARGBRowDownEven = ScaleARGBRowDownEven_NEON;
>+  if (TestCpuFlag(kCpuHasNEON)) {
>+    ScaleARGBRowDownEven = ScaleARGBRowDownEven_Any_NEON;
>+    if (IS_ALIGNED(height, 4)) {  // Width of dest.
>+      ScaleARGBRowDownEven = ScaleARGBRowDownEven_NEON;
>+    }
>+  }
>+#endif
>+#if defined(HAS_SCALEARGBROWDOWNEVEN_MSA)
>+  if (TestCpuFlag(kCpuHasMSA)) {
>+    ScaleARGBRowDownEven = ScaleARGBRowDownEven_Any_MSA;
>+    if (IS_ALIGNED(height, 4)) {  // Width of dest.
>+      ScaleARGBRowDownEven = ScaleARGBRowDownEven_MSA;
>+    }
>   }
> #endif
> 
>   for (i = 0; i < width; ++i) {  // column of source to row of dest.
>-    ScaleARGBRowDownEven(src, 0, src_pixel_step, dst, height);
>-    dst += dst_stride;
>-    src += 4;
>+    ScaleARGBRowDownEven(src_argb, 0, src_pixel_step, dst_argb, height);
>+    dst_argb += dst_stride_argb;
>+    src_argb += 4;
>   }
> }
> 
>-void ARGBRotate90(const uint8* src,
>-                  int src_stride,
>-                  uint8* dst,
>-                  int dst_stride,
>+void ARGBRotate90(const uint8_t* src_argb,
>+                  int src_stride_argb,
>+                  uint8_t* dst_argb,
>+                  int dst_stride_argb,
>                   int width,
>                   int height) {
>   // Rotate by 90 is a ARGBTranspose with the source read
>   // from bottom to top. So set the source pointer to the end
>   // of the buffer and flip the sign of the source stride.
>-  src += src_stride * (height - 1);
>-  src_stride = -src_stride;
>-  ARGBTranspose(src, src_stride, dst, dst_stride, width, height);
>+  src_argb += src_stride_argb * (height - 1);
>+  src_stride_argb = -src_stride_argb;
>+  ARGBTranspose(src_argb, src_stride_argb, dst_argb, dst_stride_argb, width,
>+                height);
> }
> 
>-void ARGBRotate270(const uint8* src,
>-                   int src_stride,
>-                   uint8* dst,
>-                   int dst_stride,
>+void ARGBRotate270(const uint8_t* src_argb,
>+                   int src_stride_argb,
>+                   uint8_t* dst_argb,
>+                   int dst_stride_argb,
>                    int width,
>                    int height) {
>   // Rotate by 270 is a ARGBTranspose with the destination written
>   // from bottom to top. So set the destination pointer to the end
>   // of the buffer and flip the sign of the destination stride.
>-  dst += dst_stride * (width - 1);
>-  dst_stride = -dst_stride;
>-  ARGBTranspose(src, src_stride, dst, dst_stride, width, height);
>+  dst_argb += dst_stride_argb * (width - 1);
>+  dst_stride_argb = -dst_stride_argb;
>+  ARGBTranspose(src_argb, src_stride_argb, dst_argb, dst_stride_argb, width,
>+                height);
> }
> 
>-void ARGBRotate180(const uint8* src,
>-                   int src_stride,
>-                   uint8* dst,
>-                   int dst_stride,
>+void ARGBRotate180(const uint8_t* src_argb,
>+                   int src_stride_argb,
>+                   uint8_t* dst_argb,
>+                   int dst_stride_argb,
>                    int width,
>                    int height) {
>   // Swap first and last row and mirror the content. Uses a temporary row.
>   align_buffer_64(row, width * 4);
>-  const uint8* src_bot = src + src_stride * (height - 1);
>-  uint8* dst_bot = dst + dst_stride * (height - 1);
>+  const uint8_t* src_bot = src_argb + src_stride_argb * (height - 1);
>+  uint8_t* dst_bot = dst_argb + dst_stride_argb * (height - 1);
>   int half_height = (height + 1) >> 1;
>   int y;
>-  void (*ARGBMirrorRow)(const uint8* src, uint8* dst, int width) =
>+  void (*ARGBMirrorRow)(const uint8_t* src_argb, uint8_t* dst_argb, int width) =
>       ARGBMirrorRow_C;
>-  void (*CopyRow)(const uint8* src, uint8* dst, int width) = CopyRow_C;
>+  void (*CopyRow)(const uint8_t* src_argb, uint8_t* dst_argb, int width) =
>+      CopyRow_C;
> #if defined(HAS_ARGBMIRRORROW_NEON)
>   if (TestCpuFlag(kCpuHasNEON)) {
>     ARGBMirrorRow = ARGBMirrorRow_Any_NEON;
>@@ -173,29 +162,24 @@ void ARGBRotate180(const uint8* src,
>     CopyRow = IS_ALIGNED(width * 4, 32) ? CopyRow_NEON : CopyRow_Any_NEON;
>   }
> #endif
>-#if defined(HAS_COPYROW_MIPS)
>-  if (TestCpuFlag(kCpuHasMIPS)) {
>-    CopyRow = CopyRow_MIPS;
>-  }
>-#endif
> 
>   // Odd height will harmlessly mirror the middle row twice.
>   for (y = 0; y < half_height; ++y) {
>-    ARGBMirrorRow(src, row, width);      // Mirror first row into a buffer
>-    ARGBMirrorRow(src_bot, dst, width);  // Mirror last row into first row
>-    CopyRow(row, dst_bot, width * 4);    // Copy first mirrored row into last
>-    src += src_stride;
>-    dst += dst_stride;
>-    src_bot -= src_stride;
>-    dst_bot -= dst_stride;
>+    ARGBMirrorRow(src_argb, row, width);      // Mirror first row into a buffer
>+    ARGBMirrorRow(src_bot, dst_argb, width);  // Mirror last row into first row
>+    CopyRow(row, dst_bot, width * 4);  // Copy first mirrored row into last
>+    src_argb += src_stride_argb;
>+    dst_argb += dst_stride_argb;
>+    src_bot -= src_stride_argb;
>+    dst_bot -= dst_stride_argb;
>   }
>   free_aligned_buffer_64(row);
> }
> 
> LIBYUV_API
>-int ARGBRotate(const uint8* src_argb,
>+int ARGBRotate(const uint8_t* src_argb,
>                int src_stride_argb,
>-               uint8* dst_argb,
>+               uint8_t* dst_argb,
>                int dst_stride_argb,
>                int width,
>                int height,
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/rotate_common.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/rotate_common.cc
>index 89357e732d2..ff212adebc4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/rotate_common.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/rotate_common.cc
>@@ -16,9 +16,9 @@ namespace libyuv {
> extern "C" {
> #endif
> 
>-void TransposeWx8_C(const uint8* src,
>+void TransposeWx8_C(const uint8_t* src,
>                     int src_stride,
>-                    uint8* dst,
>+                    uint8_t* dst,
>                     int dst_stride,
>                     int width) {
>   int i;
>@@ -36,11 +36,11 @@ void TransposeWx8_C(const uint8* src,
>   }
> }
> 
>-void TransposeUVWx8_C(const uint8* src,
>+void TransposeUVWx8_C(const uint8_t* src,
>                       int src_stride,
>-                      uint8* dst_a,
>+                      uint8_t* dst_a,
>                       int dst_stride_a,
>-                      uint8* dst_b,
>+                      uint8_t* dst_b,
>                       int dst_stride_b,
>                       int width) {
>   int i;
>@@ -67,9 +67,9 @@ void TransposeUVWx8_C(const uint8* src,
>   }
> }
> 
>-void TransposeWxH_C(const uint8* src,
>+void TransposeWxH_C(const uint8_t* src,
>                     int src_stride,
>-                    uint8* dst,
>+                    uint8_t* dst,
>                     int dst_stride,
>                     int width,
>                     int height) {
>@@ -82,11 +82,11 @@ void TransposeWxH_C(const uint8* src,
>   }
> }
> 
>-void TransposeUVWxH_C(const uint8* src,
>+void TransposeUVWxH_C(const uint8_t* src,
>                       int src_stride,
>-                      uint8* dst_a,
>+                      uint8_t* dst_a,
>                       int dst_stride_a,
>-                      uint8* dst_b,
>+                      uint8_t* dst_b,
>                       int dst_stride_b,
>                       int width,
>                       int height) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/rotate_dspr2.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/rotate_dspr2.cc
>deleted file mode 100644
>index 5d2338deff6..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/rotate_dspr2.cc
>+++ /dev/null
>@@ -1,475 +0,0 @@
>-/*
>- *  Copyright 2011 The LibYuv Project Authors. All rights reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS. All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "libyuv/rotate_row.h"
>-#include "libyuv/row.h"
>-
>-#include "libyuv/basic_types.h"
>-
>-#ifdef __cplusplus
>-namespace libyuv {
>-extern "C" {
>-#endif
>-
>-#if !defined(LIBYUV_DISABLE_DSPR2) && defined(__mips_dsp) && \
>-    (__mips_dsp_rev >= 2) && (_MIPS_SIM == _MIPS_SIM_ABI32)
>-
>-void TransposeWx8_DSPR2(const uint8* src,
>-                        int src_stride,
>-                        uint8* dst,
>-                        int dst_stride,
>-                        int width) {
>-  __asm__ __volatile__(
>-      ".set push                                         \n"
>-      ".set noreorder                                    \n"
>-      "sll              $t2, %[src_stride], 0x1          \n"  // src_stride x 2
>-      "sll              $t4, %[src_stride], 0x2          \n"  // src_stride x 4
>-      "sll              $t9, %[src_stride], 0x3          \n"  // src_stride x 8
>-      "addu             $t3, $t2, %[src_stride]          \n"
>-      "addu             $t5, $t4, %[src_stride]          \n"
>-      "addu             $t6, $t2, $t4                    \n"
>-      "andi             $t0, %[dst], 0x3                 \n"
>-      "andi             $t1, %[dst_stride], 0x3          \n"
>-      "or               $t0, $t0, $t1                    \n"
>-      "bnez             $t0, 11f                         \n"
>-      " subu            $t7, $t9, %[src_stride]          \n"
>-      // dst + dst_stride word aligned
>-      "1:                                                \n"
>-      "lbu              $t0, 0(%[src])                   \n"
>-      "lbux             $t1, %[src_stride](%[src])       \n"
>-      "lbux             $t8, $t2(%[src])                 \n"
>-      "lbux             $t9, $t3(%[src])                 \n"
>-      "sll              $t1, $t1, 16                     \n"
>-      "sll              $t9, $t9, 16                     \n"
>-      "or               $t0, $t0, $t1                    \n"
>-      "or               $t8, $t8, $t9                    \n"
>-      "precr.qb.ph      $s0, $t8, $t0                    \n"
>-      "lbux             $t0, $t4(%[src])                 \n"
>-      "lbux             $t1, $t5(%[src])                 \n"
>-      "lbux             $t8, $t6(%[src])                 \n"
>-      "lbux             $t9, $t7(%[src])                 \n"
>-      "sll              $t1, $t1, 16                     \n"
>-      "sll              $t9, $t9, 16                     \n"
>-      "or               $t0, $t0, $t1                    \n"
>-      "or               $t8, $t8, $t9                    \n"
>-      "precr.qb.ph      $s1, $t8, $t0                    \n"
>-      "sw               $s0, 0(%[dst])                   \n"
>-      "addiu            %[width], -1                     \n"
>-      "addiu            %[src], 1                        \n"
>-      "sw               $s1, 4(%[dst])                   \n"
>-      "bnez             %[width], 1b                     \n"
>-      " addu            %[dst], %[dst], %[dst_stride]    \n"
>-      "b                2f                               \n"
>-      // dst + dst_stride unaligned
>-      "11:                                               \n"
>-      "lbu              $t0, 0(%[src])                   \n"
>-      "lbux             $t1, %[src_stride](%[src])       \n"
>-      "lbux             $t8, $t2(%[src])                 \n"
>-      "lbux             $t9, $t3(%[src])                 \n"
>-      "sll              $t1, $t1, 16                     \n"
>-      "sll              $t9, $t9, 16                     \n"
>-      "or               $t0, $t0, $t1                    \n"
>-      "or               $t8, $t8, $t9                    \n"
>-      "precr.qb.ph      $s0, $t8, $t0                    \n"
>-      "lbux             $t0, $t4(%[src])                 \n"
>-      "lbux             $t1, $t5(%[src])                 \n"
>-      "lbux             $t8, $t6(%[src])                 \n"
>-      "lbux             $t9, $t7(%[src])                 \n"
>-      "sll              $t1, $t1, 16                     \n"
>-      "sll              $t9, $t9, 16                     \n"
>-      "or               $t0, $t0, $t1                    \n"
>-      "or               $t8, $t8, $t9                    \n"
>-      "precr.qb.ph      $s1, $t8, $t0                    \n"
>-      "swr              $s0, 0(%[dst])                   \n"
>-      "swl              $s0, 3(%[dst])                   \n"
>-      "addiu            %[width], -1                     \n"
>-      "addiu            %[src], 1                        \n"
>-      "swr              $s1, 4(%[dst])                   \n"
>-      "swl              $s1, 7(%[dst])                   \n"
>-      "bnez             %[width], 11b                    \n"
>-      "addu             %[dst], %[dst], %[dst_stride]    \n"
>-      "2:                                                \n"
>-      ".set pop                                          \n"
>-      : [src] "+r"(src), [dst] "+r"(dst), [width] "+r"(width)
>-      : [src_stride] "r"(src_stride), [dst_stride] "r"(dst_stride)
>-      : "t0", "t1", "t2", "t3", "t4", "t5", "t6", "t7", "t8", "t9", "s0", "s1");
>-}
>-
>-void TransposeWx8_Fast_DSPR2(const uint8* src,
>-                             int src_stride,
>-                             uint8* dst,
>-                             int dst_stride,
>-                             int width) {
>-  __asm__ __volatile__(
>-      ".set noat                                         \n"
>-      ".set push                                         \n"
>-      ".set noreorder                                    \n"
>-      "beqz             %[width], 2f                     \n"
>-      " sll             $t2, %[src_stride], 0x1          \n"  // src_stride x 2
>-      "sll              $t4, %[src_stride], 0x2          \n"  // src_stride x 4
>-      "sll              $t9, %[src_stride], 0x3          \n"  // src_stride x 8
>-      "addu             $t3, $t2, %[src_stride]          \n"
>-      "addu             $t5, $t4, %[src_stride]          \n"
>-      "addu             $t6, $t2, $t4                    \n"
>-
>-      "srl              $AT, %[width], 0x2               \n"
>-      "andi             $t0, %[dst], 0x3                 \n"
>-      "andi             $t1, %[dst_stride], 0x3          \n"
>-      "or               $t0, $t0, $t1                    \n"
>-      "bnez             $t0, 11f                         \n"
>-      " subu            $t7, $t9, %[src_stride]          \n"
>-      // dst + dst_stride word aligned
>-      "1:                                                \n"
>-      "lw               $t0, 0(%[src])                   \n"
>-      "lwx              $t1, %[src_stride](%[src])       \n"
>-      "lwx              $t8, $t2(%[src])                 \n"
>-      "lwx              $t9, $t3(%[src])                 \n"
>-
>-      // t0 = | 30 | 20 | 10 | 00 |
>-      // t1 = | 31 | 21 | 11 | 01 |
>-      // t8 = | 32 | 22 | 12 | 02 |
>-      // t9 = | 33 | 23 | 13 | 03 |
>-
>-      "precr.qb.ph     $s0, $t1, $t0                     \n"
>-      "precr.qb.ph     $s1, $t9, $t8                     \n"
>-      "precrq.qb.ph    $s2, $t1, $t0                     \n"
>-      "precrq.qb.ph    $s3, $t9, $t8                     \n"
>-
>-      // s0 = | 21 | 01 | 20 | 00 |
>-      // s1 = | 23 | 03 | 22 | 02 |
>-      // s2 = | 31 | 11 | 30 | 10 |
>-      // s3 = | 33 | 13 | 32 | 12 |
>-
>-      "precr.qb.ph     $s4, $s1, $s0                     \n"
>-      "precrq.qb.ph    $s5, $s1, $s0                     \n"
>-      "precr.qb.ph     $s6, $s3, $s2                     \n"
>-      "precrq.qb.ph    $s7, $s3, $s2                     \n"
>-
>-      // s4 = | 03 | 02 | 01 | 00 |
>-      // s5 = | 23 | 22 | 21 | 20 |
>-      // s6 = | 13 | 12 | 11 | 10 |
>-      // s7 = | 33 | 32 | 31 | 30 |
>-
>-      "lwx              $t0, $t4(%[src])                 \n"
>-      "lwx              $t1, $t5(%[src])                 \n"
>-      "lwx              $t8, $t6(%[src])                 \n"
>-      "lwx              $t9, $t7(%[src])                 \n"
>-
>-      // t0 = | 34 | 24 | 14 | 04 |
>-      // t1 = | 35 | 25 | 15 | 05 |
>-      // t8 = | 36 | 26 | 16 | 06 |
>-      // t9 = | 37 | 27 | 17 | 07 |
>-
>-      "precr.qb.ph     $s0, $t1, $t0                     \n"
>-      "precr.qb.ph     $s1, $t9, $t8                     \n"
>-      "precrq.qb.ph    $s2, $t1, $t0                     \n"
>-      "precrq.qb.ph    $s3, $t9, $t8                     \n"
>-
>-      // s0 = | 25 | 05 | 24 | 04 |
>-      // s1 = | 27 | 07 | 26 | 06 |
>-      // s2 = | 35 | 15 | 34 | 14 |
>-      // s3 = | 37 | 17 | 36 | 16 |
>-
>-      "precr.qb.ph     $t0, $s1, $s0                     \n"
>-      "precrq.qb.ph    $t1, $s1, $s0                     \n"
>-      "precr.qb.ph     $t8, $s3, $s2                     \n"
>-      "precrq.qb.ph    $t9, $s3, $s2                     \n"
>-
>-      // t0 = | 07 | 06 | 05 | 04 |
>-      // t1 = | 27 | 26 | 25 | 24 |
>-      // t8 = | 17 | 16 | 15 | 14 |
>-      // t9 = | 37 | 36 | 35 | 34 |
>-
>-      "addu            $s0, %[dst], %[dst_stride]        \n"
>-      "addu            $s1, $s0, %[dst_stride]           \n"
>-      "addu            $s2, $s1, %[dst_stride]           \n"
>-
>-      "sw              $s4, 0(%[dst])                    \n"
>-      "sw              $t0, 4(%[dst])                    \n"
>-      "sw              $s6, 0($s0)                       \n"
>-      "sw              $t8, 4($s0)                       \n"
>-      "sw              $s5, 0($s1)                       \n"
>-      "sw              $t1, 4($s1)                       \n"
>-      "sw              $s7, 0($s2)                       \n"
>-      "sw              $t9, 4($s2)                       \n"
>-
>-      "addiu            $AT, -1                          \n"
>-      "addiu            %[src], 4                        \n"
>-
>-      "bnez             $AT, 1b                          \n"
>-      " addu            %[dst], $s2, %[dst_stride]       \n"
>-      "b                2f                               \n"
>-      // dst + dst_stride unaligned
>-      "11:                                               \n"
>-      "lw               $t0, 0(%[src])                   \n"
>-      "lwx              $t1, %[src_stride](%[src])       \n"
>-      "lwx              $t8, $t2(%[src])                 \n"
>-      "lwx              $t9, $t3(%[src])                 \n"
>-
>-      // t0 = | 30 | 20 | 10 | 00 |
>-      // t1 = | 31 | 21 | 11 | 01 |
>-      // t8 = | 32 | 22 | 12 | 02 |
>-      // t9 = | 33 | 23 | 13 | 03 |
>-
>-      "precr.qb.ph     $s0, $t1, $t0                     \n"
>-      "precr.qb.ph     $s1, $t9, $t8                     \n"
>-      "precrq.qb.ph    $s2, $t1, $t0                     \n"
>-      "precrq.qb.ph    $s3, $t9, $t8                     \n"
>-
>-      // s0 = | 21 | 01 | 20 | 00 |
>-      // s1 = | 23 | 03 | 22 | 02 |
>-      // s2 = | 31 | 11 | 30 | 10 |
>-      // s3 = | 33 | 13 | 32 | 12 |
>-
>-      "precr.qb.ph     $s4, $s1, $s0                     \n"
>-      "precrq.qb.ph    $s5, $s1, $s0                     \n"
>-      "precr.qb.ph     $s6, $s3, $s2                     \n"
>-      "precrq.qb.ph    $s7, $s3, $s2                     \n"
>-
>-      // s4 = | 03 | 02 | 01 | 00 |
>-      // s5 = | 23 | 22 | 21 | 20 |
>-      // s6 = | 13 | 12 | 11 | 10 |
>-      // s7 = | 33 | 32 | 31 | 30 |
>-
>-      "lwx              $t0, $t4(%[src])                 \n"
>-      "lwx              $t1, $t5(%[src])                 \n"
>-      "lwx              $t8, $t6(%[src])                 \n"
>-      "lwx              $t9, $t7(%[src])                 \n"
>-
>-      // t0 = | 34 | 24 | 14 | 04 |
>-      // t1 = | 35 | 25 | 15 | 05 |
>-      // t8 = | 36 | 26 | 16 | 06 |
>-      // t9 = | 37 | 27 | 17 | 07 |
>-
>-      "precr.qb.ph     $s0, $t1, $t0                     \n"
>-      "precr.qb.ph     $s1, $t9, $t8                     \n"
>-      "precrq.qb.ph    $s2, $t1, $t0                     \n"
>-      "precrq.qb.ph    $s3, $t9, $t8                     \n"
>-
>-      // s0 = | 25 | 05 | 24 | 04 |
>-      // s1 = | 27 | 07 | 26 | 06 |
>-      // s2 = | 35 | 15 | 34 | 14 |
>-      // s3 = | 37 | 17 | 36 | 16 |
>-
>-      "precr.qb.ph     $t0, $s1, $s0                     \n"
>-      "precrq.qb.ph    $t1, $s1, $s0                     \n"
>-      "precr.qb.ph     $t8, $s3, $s2                     \n"
>-      "precrq.qb.ph    $t9, $s3, $s2                     \n"
>-
>-      // t0 = | 07 | 06 | 05 | 04 |
>-      // t1 = | 27 | 26 | 25 | 24 |
>-      // t8 = | 17 | 16 | 15 | 14 |
>-      // t9 = | 37 | 36 | 35 | 34 |
>-
>-      "addu            $s0, %[dst], %[dst_stride]        \n"
>-      "addu            $s1, $s0, %[dst_stride]           \n"
>-      "addu            $s2, $s1, %[dst_stride]           \n"
>-
>-      "swr              $s4, 0(%[dst])                   \n"
>-      "swl              $s4, 3(%[dst])                   \n"
>-      "swr              $t0, 4(%[dst])                   \n"
>-      "swl              $t0, 7(%[dst])                   \n"
>-      "swr              $s6, 0($s0)                      \n"
>-      "swl              $s6, 3($s0)                      \n"
>-      "swr              $t8, 4($s0)                      \n"
>-      "swl              $t8, 7($s0)                      \n"
>-      "swr              $s5, 0($s1)                      \n"
>-      "swl              $s5, 3($s1)                      \n"
>-      "swr              $t1, 4($s1)                      \n"
>-      "swl              $t1, 7($s1)                      \n"
>-      "swr              $s7, 0($s2)                      \n"
>-      "swl              $s7, 3($s2)                      \n"
>-      "swr              $t9, 4($s2)                      \n"
>-      "swl              $t9, 7($s2)                      \n"
>-
>-      "addiu            $AT, -1                          \n"
>-      "addiu            %[src], 4                        \n"
>-
>-      "bnez             $AT, 11b                         \n"
>-      " addu            %[dst], $s2, %[dst_stride]       \n"
>-      "2:                                                \n"
>-      ".set pop                                          \n"
>-      ".set at                                           \n"
>-      : [src] "+r"(src), [dst] "+r"(dst), [width] "+r"(width)
>-      : [src_stride] "r"(src_stride), [dst_stride] "r"(dst_stride)
>-      : "t0", "t1", "t2", "t3", "t4", "t5", "t6", "t7", "t8", "t9", "s0", "s1",
>-        "s2", "s3", "s4", "s5", "s6", "s7");
>-}
>-
>-void TransposeUVWx8_DSPR2(const uint8* src,
>-                          int src_stride,
>-                          uint8* dst_a,
>-                          int dst_stride_a,
>-                          uint8* dst_b,
>-                          int dst_stride_b,
>-                          int width) {
>-  __asm__ __volatile__(
>-      ".set push                                         \n"
>-      ".set noreorder                                    \n"
>-      "beqz            %[width], 2f                      \n"
>-      " sll            $t2, %[src_stride], 0x1           \n"  // src_stride x 2
>-      "sll             $t4, %[src_stride], 0x2           \n"  // src_stride x 4
>-      "sll             $t9, %[src_stride], 0x3           \n"  // src_stride x 8
>-      "addu            $t3, $t2, %[src_stride]           \n"
>-      "addu            $t5, $t4, %[src_stride]           \n"
>-      "addu            $t6, $t2, $t4                     \n"
>-      "subu            $t7, $t9, %[src_stride]           \n"
>-      "srl             $t1, %[width], 1                  \n"
>-
>-      // check word aligment for dst_a, dst_b, dst_stride_a and dst_stride_b
>-      "andi            $t0, %[dst_a], 0x3                \n"
>-      "andi            $t8, %[dst_b], 0x3                \n"
>-      "or              $t0, $t0, $t8                     \n"
>-      "andi            $t8, %[dst_stride_a], 0x3         \n"
>-      "andi            $s5, %[dst_stride_b], 0x3         \n"
>-      "or              $t8, $t8, $s5                     \n"
>-      "or              $t0, $t0, $t8                     \n"
>-      "bnez            $t0, 11f                          \n"
>-      " nop                                              \n"
>-      // dst + dst_stride word aligned (both, a & b dst addresses)
>-      "1:                                                \n"
>-      "lw              $t0, 0(%[src])                    \n"  // |B0|A0|b0|a0|
>-      "lwx             $t8, %[src_stride](%[src])        \n"  // |B1|A1|b1|a1|
>-      "addu            $s5, %[dst_a], %[dst_stride_a]    \n"
>-      "lwx             $t9, $t2(%[src])                  \n"  // |B2|A2|b2|a2|
>-      "lwx             $s0, $t3(%[src])                  \n"  // |B3|A3|b3|a3|
>-      "addu            $s6, %[dst_b], %[dst_stride_b]    \n"
>-
>-      "precrq.ph.w     $s1, $t8, $t0                     \n"  // |B1|A1|B0|A0|
>-      "precrq.ph.w     $s2, $s0, $t9                     \n"  // |B3|A3|B2|A2|
>-      "precr.qb.ph     $s3, $s2, $s1                     \n"  // |A3|A2|A1|A0|
>-      "precrq.qb.ph    $s4, $s2, $s1                     \n"  // |B3|B2|B1|B0|
>-
>-      "sll             $t0, $t0, 16                      \n"
>-      "packrl.ph       $s1, $t8, $t0                     \n"  // |b1|a1|b0|a0|
>-      "sll             $t9, $t9, 16                      \n"
>-      "packrl.ph       $s2, $s0, $t9                     \n"  // |b3|a3|b2|a2|
>-
>-      "sw              $s3, 0($s5)                       \n"
>-      "sw              $s4, 0($s6)                       \n"
>-
>-      "precr.qb.ph     $s3, $s2, $s1                     \n"  // |a3|a2|a1|a0|
>-      "precrq.qb.ph    $s4, $s2, $s1                     \n"  // |b3|b2|b1|b0|
>-
>-      "lwx             $t0, $t4(%[src])                  \n"  // |B4|A4|b4|a4|
>-      "lwx             $t8, $t5(%[src])                  \n"  // |B5|A5|b5|a5|
>-      "lwx             $t9, $t6(%[src])                  \n"  // |B6|A6|b6|a6|
>-      "lwx             $s0, $t7(%[src])                  \n"  // |B7|A7|b7|a7|
>-      "sw              $s3, 0(%[dst_a])                  \n"
>-      "sw              $s4, 0(%[dst_b])                  \n"
>-
>-      "precrq.ph.w     $s1, $t8, $t0                     \n"  // |B5|A5|B4|A4|
>-      "precrq.ph.w     $s2, $s0, $t9                     \n"  // |B6|A6|B7|A7|
>-      "precr.qb.ph     $s3, $s2, $s1                     \n"  // |A7|A6|A5|A4|
>-      "precrq.qb.ph    $s4, $s2, $s1                     \n"  // |B7|B6|B5|B4|
>-
>-      "sll             $t0, $t0, 16                      \n"
>-      "packrl.ph       $s1, $t8, $t0                     \n"  // |b5|a5|b4|a4|
>-      "sll             $t9, $t9, 16                      \n"
>-      "packrl.ph       $s2, $s0, $t9                     \n"  // |b7|a7|b6|a6|
>-      "sw              $s3, 4($s5)                       \n"
>-      "sw              $s4, 4($s6)                       \n"
>-
>-      "precr.qb.ph     $s3, $s2, $s1                     \n"  // |a7|a6|a5|a4|
>-      "precrq.qb.ph    $s4, $s2, $s1                     \n"  // |b7|b6|b5|b4|
>-
>-      "addiu           %[src], 4                         \n"
>-      "addiu           $t1, -1                           \n"
>-      "sll             $t0, %[dst_stride_a], 1           \n"
>-      "sll             $t8, %[dst_stride_b], 1           \n"
>-      "sw              $s3, 4(%[dst_a])                  \n"
>-      "sw              $s4, 4(%[dst_b])                  \n"
>-      "addu            %[dst_a], %[dst_a], $t0           \n"
>-      "bnez            $t1, 1b                           \n"
>-      " addu           %[dst_b], %[dst_b], $t8           \n"
>-      "b               2f                                \n"
>-      " nop                                              \n"
>-
>-      // dst_a or dst_b or dst_stride_a or dst_stride_b not word aligned
>-      "11:                                               \n"
>-      "lw              $t0, 0(%[src])                    \n"  // |B0|A0|b0|a0|
>-      "lwx             $t8, %[src_stride](%[src])        \n"  // |B1|A1|b1|a1|
>-      "addu            $s5, %[dst_a], %[dst_stride_a]    \n"
>-      "lwx             $t9, $t2(%[src])                  \n"  // |B2|A2|b2|a2|
>-      "lwx             $s0, $t3(%[src])                  \n"  // |B3|A3|b3|a3|
>-      "addu            $s6, %[dst_b], %[dst_stride_b]    \n"
>-
>-      "precrq.ph.w     $s1, $t8, $t0                     \n"  // |B1|A1|B0|A0|
>-      "precrq.ph.w     $s2, $s0, $t9                     \n"  // |B3|A3|B2|A2|
>-      "precr.qb.ph     $s3, $s2, $s1                     \n"  // |A3|A2|A1|A0|
>-      "precrq.qb.ph    $s4, $s2, $s1                     \n"  // |B3|B2|B1|B0|
>-
>-      "sll             $t0, $t0, 16                      \n"
>-      "packrl.ph       $s1, $t8, $t0                     \n"  // |b1|a1|b0|a0|
>-      "sll             $t9, $t9, 16                      \n"
>-      "packrl.ph       $s2, $s0, $t9                     \n"  // |b3|a3|b2|a2|
>-
>-      "swr             $s3, 0($s5)                       \n"
>-      "swl             $s3, 3($s5)                       \n"
>-      "swr             $s4, 0($s6)                       \n"
>-      "swl             $s4, 3($s6)                       \n"
>-
>-      "precr.qb.ph     $s3, $s2, $s1                     \n"  // |a3|a2|a1|a0|
>-      "precrq.qb.ph    $s4, $s2, $s1                     \n"  // |b3|b2|b1|b0|
>-
>-      "lwx             $t0, $t4(%[src])                  \n"  // |B4|A4|b4|a4|
>-      "lwx             $t8, $t5(%[src])                  \n"  // |B5|A5|b5|a5|
>-      "lwx             $t9, $t6(%[src])                  \n"  // |B6|A6|b6|a6|
>-      "lwx             $s0, $t7(%[src])                  \n"  // |B7|A7|b7|a7|
>-      "swr             $s3, 0(%[dst_a])                  \n"
>-      "swl             $s3, 3(%[dst_a])                  \n"
>-      "swr             $s4, 0(%[dst_b])                  \n"
>-      "swl             $s4, 3(%[dst_b])                  \n"
>-
>-      "precrq.ph.w     $s1, $t8, $t0                     \n"  // |B5|A5|B4|A4|
>-      "precrq.ph.w     $s2, $s0, $t9                     \n"  // |B6|A6|B7|A7|
>-      "precr.qb.ph     $s3, $s2, $s1                     \n"  // |A7|A6|A5|A4|
>-      "precrq.qb.ph    $s4, $s2, $s1                     \n"  // |B7|B6|B5|B4|
>-
>-      "sll             $t0, $t0, 16                      \n"
>-      "packrl.ph       $s1, $t8, $t0                     \n"  // |b5|a5|b4|a4|
>-      "sll             $t9, $t9, 16                      \n"
>-      "packrl.ph       $s2, $s0, $t9                     \n"  // |b7|a7|b6|a6|
>-
>-      "swr             $s3, 4($s5)                       \n"
>-      "swl             $s3, 7($s5)                       \n"
>-      "swr             $s4, 4($s6)                       \n"
>-      "swl             $s4, 7($s6)                       \n"
>-
>-      "precr.qb.ph     $s3, $s2, $s1                     \n"  // |a7|a6|a5|a4|
>-      "precrq.qb.ph    $s4, $s2, $s1                     \n"  // |b7|b6|b5|b4|
>-
>-      "addiu           %[src], 4                         \n"
>-      "addiu           $t1, -1                           \n"
>-      "sll             $t0, %[dst_stride_a], 1           \n"
>-      "sll             $t8, %[dst_stride_b], 1           \n"
>-      "swr             $s3, 4(%[dst_a])                  \n"
>-      "swl             $s3, 7(%[dst_a])                  \n"
>-      "swr             $s4, 4(%[dst_b])                  \n"
>-      "swl             $s4, 7(%[dst_b])                  \n"
>-      "addu            %[dst_a], %[dst_a], $t0           \n"
>-      "bnez            $t1, 11b                          \n"
>-      " addu           %[dst_b], %[dst_b], $t8           \n"
>-
>-      "2:                                                \n"
>-      ".set pop                                          \n"
>-      : [src] "+r"(src), [dst_a] "+r"(dst_a), [dst_b] "+r"(dst_b),
>-        [width] "+r"(width), [src_stride] "+r"(src_stride)
>-      : [dst_stride_a] "r"(dst_stride_a), [dst_stride_b] "r"(dst_stride_b)
>-      : "t0", "t1", "t2", "t3", "t4", "t5", "t6", "t7", "t8", "t9", "s0", "s1",
>-        "s2", "s3", "s4", "s5", "s6");
>-}
>-
>-#endif  // defined(__mips_dsp) && (__mips_dsp_rev >= 2)
>-
>-#ifdef __cplusplus
>-}  // extern "C"
>-}  // namespace libyuv
>-#endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/rotate_gcc.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/rotate_gcc.cc
>index 74b48ac4084..04e19e29eef 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/rotate_gcc.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/rotate_gcc.cc
>@@ -22,9 +22,9 @@ extern "C" {
> 
> // Transpose 8x8. 32 or 64 bit, but not NaCL for 64 bit.
> #if defined(HAS_TRANSPOSEWX8_SSSE3)
>-void TransposeWx8_SSSE3(const uint8* src,
>+void TransposeWx8_SSSE3(const uint8_t* src,
>                         int src_stride,
>-                        uint8* dst,
>+                        uint8_t* dst,
>                         int dst_stride,
>                         int width) {
>   asm volatile(
>@@ -112,9 +112,9 @@ void TransposeWx8_SSSE3(const uint8* src,
> 
> // Transpose 16x8. 64 bit
> #if defined(HAS_TRANSPOSEWX8_FAST_SSSE3)
>-void TransposeWx8_Fast_SSSE3(const uint8* src,
>+void TransposeWx8_Fast_SSSE3(const uint8_t* src,
>                              int src_stride,
>-                             uint8* dst,
>+                             uint8_t* dst,
>                              int dst_stride,
>                              int width) {
>   asm volatile(
>@@ -255,11 +255,11 @@ void TransposeWx8_Fast_SSSE3(const uint8* src,
> 
> // Transpose UV 8x8.  64 bit.
> #if defined(HAS_TRANSPOSEUVWX8_SSE2)
>-void TransposeUVWx8_SSE2(const uint8* src,
>+void TransposeUVWx8_SSE2(const uint8_t* src,
>                          int src_stride,
>-                         uint8* dst_a,
>+                         uint8_t* dst_a,
>                          int dst_stride_a,
>-                         uint8* dst_b,
>+                         uint8_t* dst_b,
>                          int dst_stride_b,
>                          int width) {
>   asm volatile(
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/rotate_msa.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/rotate_msa.cc
>index 8907765aba7..99bdca65b32 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/rotate_msa.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/rotate_msa.cc
>@@ -51,9 +51,9 @@ extern "C" {
>     out3 = (v16u8)__msa_ilvl_d((v2i64)in3, (v2i64)in2);     \
>   }
> 
>-void TransposeWx16_C(const uint8* src,
>+void TransposeWx16_C(const uint8_t* src,
>                      int src_stride,
>-                     uint8* dst,
>+                     uint8_t* dst,
>                      int dst_stride,
>                      int width) {
>   TransposeWx8_C(src, src_stride, dst, dst_stride, width);
>@@ -61,11 +61,11 @@ void TransposeWx16_C(const uint8* src,
>                  width);
> }
> 
>-void TransposeUVWx16_C(const uint8* src,
>+void TransposeUVWx16_C(const uint8_t* src,
>                        int src_stride,
>-                       uint8* dst_a,
>+                       uint8_t* dst_a,
>                        int dst_stride_a,
>-                       uint8* dst_b,
>+                       uint8_t* dst_b,
>                        int dst_stride_b,
>                        int width) {
>   TransposeUVWx8_C(src, src_stride, dst_a, dst_stride_a, dst_b, dst_stride_b,
>@@ -74,13 +74,13 @@ void TransposeUVWx16_C(const uint8* src,
>                    dst_stride_a, (dst_b + 8), dst_stride_b, width);
> }
> 
>-void TransposeWx16_MSA(const uint8* src,
>+void TransposeWx16_MSA(const uint8_t* src,
>                        int src_stride,
>-                       uint8* dst,
>+                       uint8_t* dst,
>                        int dst_stride,
>                        int width) {
>   int x;
>-  const uint8* s;
>+  const uint8_t* s;
>   v16u8 src0, src1, src2, src3, dst0, dst1, dst2, dst3, vec0, vec1, vec2, vec3;
>   v16u8 reg0, reg1, reg2, reg3, reg4, reg5, reg6, reg7;
>   v16u8 res0, res1, res2, res3, res4, res5, res6, res7, res8, res9;
>@@ -153,15 +153,15 @@ void TransposeWx16_MSA(const uint8* src,
>   }
> }
> 
>-void TransposeUVWx16_MSA(const uint8* src,
>+void TransposeUVWx16_MSA(const uint8_t* src,
>                          int src_stride,
>-                         uint8* dst_a,
>+                         uint8_t* dst_a,
>                          int dst_stride_a,
>-                         uint8* dst_b,
>+                         uint8_t* dst_b,
>                          int dst_stride_b,
>                          int width) {
>   int x;
>-  const uint8* s;
>+  const uint8_t* s;
>   v16u8 src0, src1, src2, src3, dst0, dst1, dst2, dst3, vec0, vec1, vec2, vec3;
>   v16u8 reg0, reg1, reg2, reg3, reg4, reg5, reg6, reg7;
>   v16u8 res0, res1, res2, res3, res4, res5, res6, res7, res8, res9;
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/rotate_neon.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/rotate_neon.cc
>index d9bbc78cdab..fdc0dd476c6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/rotate_neon.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/rotate_neon.cc
>@@ -21,15 +21,15 @@ extern "C" {
> #if !defined(LIBYUV_DISABLE_NEON) && defined(__ARM_NEON__) && \
>     !defined(__aarch64__)
> 
>-static uvec8 kVTbl4x4Transpose = {0, 4, 8,  12, 1, 5, 9,  13,
>-                                  2, 6, 10, 14, 3, 7, 11, 15};
>+static const uvec8 kVTbl4x4Transpose = {0, 4, 8,  12, 1, 5, 9,  13,
>+                                        2, 6, 10, 14, 3, 7, 11, 15};
> 
>-void TransposeWx8_NEON(const uint8* src,
>+void TransposeWx8_NEON(const uint8_t* src,
>                        int src_stride,
>-                       uint8* dst,
>+                       uint8_t* dst,
>                        int dst_stride,
>                        int width) {
>-  const uint8* src_temp;
>+  const uint8_t* src_temp;
>   asm volatile(
>       // loops are on blocks of 8. loop will stop when
>       // counter gets to or below 0. starting the counter
>@@ -189,17 +189,17 @@ void TransposeWx8_NEON(const uint8* src,
>       : "memory", "cc", "q0", "q1", "q2", "q3");
> }
> 
>-static uvec8 kVTbl4x4TransposeDi = {0, 8,  1, 9,  2, 10, 3, 11,
>-                                    4, 12, 5, 13, 6, 14, 7, 15};
>+static const uvec8 kVTbl4x4TransposeDi = {0, 8,  1, 9,  2, 10, 3, 11,
>+                                          4, 12, 5, 13, 6, 14, 7, 15};
> 
>-void TransposeUVWx8_NEON(const uint8* src,
>+void TransposeUVWx8_NEON(const uint8_t* src,
>                          int src_stride,
>-                         uint8* dst_a,
>+                         uint8_t* dst_a,
>                          int dst_stride_a,
>-                         uint8* dst_b,
>+                         uint8_t* dst_b,
>                          int dst_stride_b,
>                          int width) {
>-  const uint8* src_temp;
>+  const uint8_t* src_temp;
>   asm volatile(
>       // loops are on blocks of 8. loop will stop when
>       // counter gets to or below 0. starting the counter
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/rotate_neon64.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/rotate_neon64.cc
>index 59db7f9f3da..f469baacf68 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/rotate_neon64.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/rotate_neon64.cc
>@@ -21,15 +21,15 @@ extern "C" {
> // This module is for GCC Neon armv8 64 bit.
> #if !defined(LIBYUV_DISABLE_NEON) && defined(__aarch64__)
> 
>-static uvec8 kVTbl4x4Transpose = {0, 4, 8,  12, 1, 5, 9,  13,
>-                                  2, 6, 10, 14, 3, 7, 11, 15};
>+static const uvec8 kVTbl4x4Transpose = {0, 4, 8,  12, 1, 5, 9,  13,
>+                                        2, 6, 10, 14, 3, 7, 11, 15};
> 
>-void TransposeWx8_NEON(const uint8* src,
>+void TransposeWx8_NEON(const uint8_t* src,
>                        int src_stride,
>-                       uint8* dst,
>+                       uint8_t* dst,
>                        int dst_stride,
>                        int width) {
>-  const uint8* src_temp;
>+  const uint8_t* src_temp;
>   asm volatile(
>       // loops are on blocks of 8. loop will stop when
>       // counter gets to or below 0. starting the counter
>@@ -196,18 +196,18 @@ void TransposeWx8_NEON(const uint8* src,
>         "v17", "v18", "v19", "v20", "v21", "v22", "v23");
> }
> 
>-static uint8 kVTbl4x4TransposeDi[32] = {
>+static const uint8_t kVTbl4x4TransposeDi[32] = {
>     0, 16, 32, 48, 2, 18, 34, 50, 4, 20, 36, 52, 6, 22, 38, 54,
>     1, 17, 33, 49, 3, 19, 35, 51, 5, 21, 37, 53, 7, 23, 39, 55};
> 
>-void TransposeUVWx8_NEON(const uint8* src,
>+void TransposeUVWx8_NEON(const uint8_t* src,
>                          int src_stride,
>-                         uint8* dst_a,
>+                         uint8_t* dst_a,
>                          int dst_stride_a,
>-                         uint8* dst_b,
>+                         uint8_t* dst_b,
>                          int dst_stride_b,
>                          int width) {
>-  const uint8* src_temp;
>+  const uint8_t* src_temp;
>   asm volatile(
>       // loops are on blocks of 8. loop will stop when
>       // counter gets to or below 0. starting the counter
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/rotate_win.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/rotate_win.cc
>index fb052f65212..e887dd525c7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/rotate_win.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/rotate_win.cc
>@@ -19,9 +19,9 @@ extern "C" {
> // This module is for 32 bit Visual C x86 and clangcl
> #if !defined(LIBYUV_DISABLE_X86) && defined(_M_IX86) && defined(_MSC_VER)
> 
>-__declspec(naked) void TransposeWx8_SSSE3(const uint8* src,
>+__declspec(naked) void TransposeWx8_SSSE3(const uint8_t* src,
>                                           int src_stride,
>-                                          uint8* dst,
>+                                          uint8_t* dst,
>                                           int dst_stride,
>                                           int width) {
>   __asm {
>@@ -112,11 +112,11 @@ __declspec(naked) void TransposeWx8_SSSE3(const uint8* src,
>   }
> }
> 
>-__declspec(naked) void TransposeUVWx8_SSE2(const uint8* src,
>+__declspec(naked) void TransposeUVWx8_SSE2(const uint8_t* src,
>                                            int src_stride,
>-                                           uint8* dst_a,
>+                                           uint8_t* dst_a,
>                                            int dst_stride_a,
>-                                           uint8* dst_b,
>+                                           uint8_t* dst_b,
>                                            int dst_stride_b,
>                                            int w) {
>   __asm {
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/row_any.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/row_any.cc
>index 8b31ac9fccc..cc5914dd29d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/row_any.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/row_any.cc
>@@ -31,25 +31,25 @@ extern "C" {
> #define SS(width, shift) (((width) + (1 << (shift)) - 1) >> (shift))
> 
> // Any 4 planes to 1 with yuvconstants
>-#define ANY41C(NAMEANY, ANY_SIMD, UVSHIFT, DUVSHIFT, BPP, MASK)            \
>-  void NAMEANY(const uint8* y_buf, const uint8* u_buf, const uint8* v_buf, \
>-               const uint8* a_buf, uint8* dst_ptr,                         \
>-               const struct YuvConstants* yuvconstants, int width) {       \
>-    SIMD_ALIGNED(uint8 temp[64 * 5]);                                      \
>-    memset(temp, 0, 64 * 4); /* for msan */                                \
>-    int r = width & MASK;                                                  \
>-    int n = width & ~MASK;                                                 \
>-    if (n > 0) {                                                           \
>-      ANY_SIMD(y_buf, u_buf, v_buf, a_buf, dst_ptr, yuvconstants, n);      \
>-    }                                                                      \
>-    memcpy(temp, y_buf + n, r);                                            \
>-    memcpy(temp + 64, u_buf + (n >> UVSHIFT), SS(r, UVSHIFT));             \
>-    memcpy(temp + 128, v_buf + (n >> UVSHIFT), SS(r, UVSHIFT));            \
>-    memcpy(temp + 192, a_buf + n, r);                                      \
>-    ANY_SIMD(temp, temp + 64, temp + 128, temp + 192, temp + 256,          \
>-             yuvconstants, MASK + 1);                                      \
>-    memcpy(dst_ptr + (n >> DUVSHIFT) * BPP, temp + 256,                    \
>-           SS(r, DUVSHIFT) * BPP);                                         \
>+#define ANY41C(NAMEANY, ANY_SIMD, UVSHIFT, DUVSHIFT, BPP, MASK)              \
>+  void NAMEANY(const uint8_t* y_buf, const uint8_t* u_buf,                   \
>+               const uint8_t* v_buf, const uint8_t* a_buf, uint8_t* dst_ptr, \
>+               const struct YuvConstants* yuvconstants, int width) {         \
>+    SIMD_ALIGNED(uint8_t temp[64 * 5]);                                      \
>+    memset(temp, 0, 64 * 4); /* for msan */                                  \
>+    int r = width & MASK;                                                    \
>+    int n = width & ~MASK;                                                   \
>+    if (n > 0) {                                                             \
>+      ANY_SIMD(y_buf, u_buf, v_buf, a_buf, dst_ptr, yuvconstants, n);        \
>+    }                                                                        \
>+    memcpy(temp, y_buf + n, r);                                              \
>+    memcpy(temp + 64, u_buf + (n >> UVSHIFT), SS(r, UVSHIFT));               \
>+    memcpy(temp + 128, v_buf + (n >> UVSHIFT), SS(r, UVSHIFT));              \
>+    memcpy(temp + 192, a_buf + n, r);                                        \
>+    ANY_SIMD(temp, temp + 64, temp + 128, temp + 192, temp + 256,            \
>+             yuvconstants, MASK + 1);                                        \
>+    memcpy(dst_ptr + (n >> DUVSHIFT) * BPP, temp + 256,                      \
>+           SS(r, DUVSHIFT) * BPP);                                           \
>   }
> 
> #ifdef HAS_I422ALPHATOARGBROW_SSSE3
>@@ -67,22 +67,22 @@ ANY41C(I422AlphaToARGBRow_Any_MSA, I422AlphaToARGBRow_MSA, 1, 0, 4, 7)
> #undef ANY41C
> 
> // Any 3 planes to 1.
>-#define ANY31(NAMEANY, ANY_SIMD, UVSHIFT, DUVSHIFT, BPP, MASK)             \
>-  void NAMEANY(const uint8* y_buf, const uint8* u_buf, const uint8* v_buf, \
>-               uint8* dst_ptr, int width) {                                \
>-    SIMD_ALIGNED(uint8 temp[64 * 4]);                                      \
>-    memset(temp, 0, 64 * 3); /* for YUY2 and msan */                       \
>-    int r = width & MASK;                                                  \
>-    int n = width & ~MASK;                                                 \
>-    if (n > 0) {                                                           \
>-      ANY_SIMD(y_buf, u_buf, v_buf, dst_ptr, n);                           \
>-    }                                                                      \
>-    memcpy(temp, y_buf + n, r);                                            \
>-    memcpy(temp + 64, u_buf + (n >> UVSHIFT), SS(r, UVSHIFT));             \
>-    memcpy(temp + 128, v_buf + (n >> UVSHIFT), SS(r, UVSHIFT));            \
>-    ANY_SIMD(temp, temp + 64, temp + 128, temp + 192, MASK + 1);           \
>-    memcpy(dst_ptr + (n >> DUVSHIFT) * BPP, temp + 192,                    \
>-           SS(r, DUVSHIFT) * BPP);                                         \
>+#define ANY31(NAMEANY, ANY_SIMD, UVSHIFT, DUVSHIFT, BPP, MASK)      \
>+  void NAMEANY(const uint8_t* y_buf, const uint8_t* u_buf,          \
>+               const uint8_t* v_buf, uint8_t* dst_ptr, int width) { \
>+    SIMD_ALIGNED(uint8_t temp[64 * 4]);                             \
>+    memset(temp, 0, 64 * 3); /* for YUY2 and msan */                \
>+    int r = width & MASK;                                           \
>+    int n = width & ~MASK;                                          \
>+    if (n > 0) {                                                    \
>+      ANY_SIMD(y_buf, u_buf, v_buf, dst_ptr, n);                    \
>+    }                                                               \
>+    memcpy(temp, y_buf + n, r);                                     \
>+    memcpy(temp + 64, u_buf + (n >> UVSHIFT), SS(r, UVSHIFT));      \
>+    memcpy(temp + 128, v_buf + (n >> UVSHIFT), SS(r, UVSHIFT));     \
>+    ANY_SIMD(temp, temp + 64, temp + 128, temp + 192, MASK + 1);    \
>+    memcpy(dst_ptr + (n >> DUVSHIFT) * BPP, temp + 192,             \
>+           SS(r, DUVSHIFT) * BPP);                                  \
>   }
> 
> // Merge functions.
>@@ -96,6 +96,10 @@ ANY31(MergeRGBRow_Any_NEON, MergeRGBRow_NEON, 0, 0, 3, 15)
> ANY31(I422ToYUY2Row_Any_SSE2, I422ToYUY2Row_SSE2, 1, 1, 4, 15)
> ANY31(I422ToUYVYRow_Any_SSE2, I422ToUYVYRow_SSE2, 1, 1, 4, 15)
> #endif
>+#ifdef HAS_I422TOYUY2ROW_AVX2
>+ANY31(I422ToYUY2Row_Any_AVX2, I422ToYUY2Row_AVX2, 1, 1, 4, 31)
>+ANY31(I422ToUYVYRow_Any_AVX2, I422ToUYVYRow_AVX2, 1, 1, 4, 31)
>+#endif
> #ifdef HAS_I422TOYUY2ROW_NEON
> ANY31(I422ToYUY2Row_Any_NEON, I422ToYUY2Row_NEON, 1, 1, 4, 15)
> #endif
>@@ -120,10 +124,10 @@ ANY31(BlendPlaneRow_Any_SSSE3, BlendPlaneRow_SSSE3, 0, 0, 1, 7)
> // on arm that subsamples 444 to 422 internally.
> // Any 3 planes to 1 with yuvconstants
> #define ANY31C(NAMEANY, ANY_SIMD, UVSHIFT, DUVSHIFT, BPP, MASK)                \
>-  void NAMEANY(const uint8* y_buf, const uint8* u_buf, const uint8* v_buf,     \
>-               uint8* dst_ptr, const struct YuvConstants* yuvconstants,        \
>-               int width) {                                                    \
>-    SIMD_ALIGNED(uint8 temp[64 * 4]);                                          \
>+  void NAMEANY(const uint8_t* y_buf, const uint8_t* u_buf,                     \
>+               const uint8_t* v_buf, uint8_t* dst_ptr,                         \
>+               const struct YuvConstants* yuvconstants, int width) {           \
>+    SIMD_ALIGNED(uint8_t temp[64 * 4]);                                        \
>     memset(temp, 0, 64 * 3); /* for YUY2 and msan */                           \
>     int r = width & MASK;                                                      \
>     int n = width & ~MASK;                                                     \
>@@ -145,6 +149,12 @@ ANY31(BlendPlaneRow_Any_SSSE3, BlendPlaneRow_SSSE3, 0, 0, 1, 7)
> #ifdef HAS_I422TOARGBROW_SSSE3
> ANY31C(I422ToARGBRow_Any_SSSE3, I422ToARGBRow_SSSE3, 1, 0, 4, 7)
> #endif
>+#ifdef HAS_I422TOAR30ROW_SSSE3
>+ANY31C(I422ToAR30Row_Any_SSSE3, I422ToAR30Row_SSSE3, 1, 0, 4, 7)
>+#endif
>+#ifdef HAS_I422TOAR30ROW_AVX2
>+ANY31C(I422ToAR30Row_Any_AVX2, I422ToAR30Row_AVX2, 1, 0, 4, 15)
>+#endif
> #ifdef HAS_I444TOARGBROW_SSSE3
> ANY31C(I444ToARGBRow_Any_SSSE3, I444ToARGBRow_SSSE3, 0, 0, 4, 7)
> ANY31C(I422ToRGBARow_Any_SSSE3, I422ToRGBARow_SSSE3, 1, 0, 4, 7)
>@@ -183,12 +193,6 @@ ANY31C(I422ToARGB4444Row_Any_NEON, I422ToARGB4444Row_NEON, 1, 0, 2, 7)
> ANY31C(I422ToARGB1555Row_Any_NEON, I422ToARGB1555Row_NEON, 1, 0, 2, 7)
> ANY31C(I422ToRGB565Row_Any_NEON, I422ToRGB565Row_NEON, 1, 0, 2, 7)
> #endif
>-#ifdef HAS_I422TOARGBROW_DSPR2
>-ANY31C(I444ToARGBRow_Any_DSPR2, I444ToARGBRow_DSPR2, 0, 0, 4, 7)
>-ANY31C(I422ToARGBRow_Any_DSPR2, I422ToARGBRow_DSPR2, 1, 0, 4, 7)
>-ANY31C(I422ToARGB4444Row_Any_DSPR2, I422ToARGB4444Row_DSPR2, 1, 0, 2, 7)
>-ANY31C(I422ToARGB1555Row_Any_DSPR2, I422ToARGB1555Row_DSPR2, 1, 0, 2, 7)
>-#endif
> #ifdef HAS_I422TOARGBROW_MSA
> ANY31C(I444ToARGBRow_Any_MSA, I444ToARGBRow_MSA, 0, 0, 4, 7)
> ANY31C(I422ToARGBRow_Any_MSA, I422ToARGBRow_MSA, 1, 0, 4, 7)
>@@ -200,22 +204,57 @@ ANY31C(I422ToRGB565Row_Any_MSA, I422ToRGB565Row_MSA, 1, 0, 2, 7)
> #endif
> #undef ANY31C
> 
>+// Any 3 planes of 16 bit to 1 with yuvconstants
>+// TODO(fbarchard): consider sharing this code with ANY31C
>+#define ANY31CT(NAMEANY, ANY_SIMD, UVSHIFT, DUVSHIFT, T, SBPP, BPP, MASK) \
>+  void NAMEANY(const T* y_buf, const T* u_buf, const T* v_buf,            \
>+               uint8_t* dst_ptr, const struct YuvConstants* yuvconstants, \
>+               int width) {                                               \
>+    SIMD_ALIGNED(T temp[16 * 3]);                                         \
>+    SIMD_ALIGNED(uint8_t out[64]);                                        \
>+    memset(temp, 0, 16 * 3 * SBPP); /* for YUY2 and msan */               \
>+    int r = width & MASK;                                                 \
>+    int n = width & ~MASK;                                                \
>+    if (n > 0) {                                                          \
>+      ANY_SIMD(y_buf, u_buf, v_buf, dst_ptr, yuvconstants, n);            \
>+    }                                                                     \
>+    memcpy(temp, y_buf + n, r * SBPP);                                    \
>+    memcpy(temp + 16, u_buf + (n >> UVSHIFT), SS(r, UVSHIFT) * SBPP);     \
>+    memcpy(temp + 32, v_buf + (n >> UVSHIFT), SS(r, UVSHIFT) * SBPP);     \
>+    ANY_SIMD(temp, temp + 16, temp + 32, out, yuvconstants, MASK + 1);    \
>+    memcpy(dst_ptr + (n >> DUVSHIFT) * BPP, out, SS(r, DUVSHIFT) * BPP);  \
>+  }
>+
>+#ifdef HAS_I210TOAR30ROW_SSSE3
>+ANY31CT(I210ToAR30Row_Any_SSSE3, I210ToAR30Row_SSSE3, 1, 0, uint16_t, 2, 4, 7)
>+#endif
>+#ifdef HAS_I210TOARGBROW_SSSE3
>+ANY31CT(I210ToARGBRow_Any_SSSE3, I210ToARGBRow_SSSE3, 1, 0, uint16_t, 2, 4, 7)
>+#endif
>+#ifdef HAS_I210TOARGBROW_AVX2
>+ANY31CT(I210ToARGBRow_Any_AVX2, I210ToARGBRow_AVX2, 1, 0, uint16_t, 2, 4, 15)
>+#endif
>+#ifdef HAS_I210TOAR30ROW_AVX2
>+ANY31CT(I210ToAR30Row_Any_AVX2, I210ToAR30Row_AVX2, 1, 0, uint16_t, 2, 4, 15)
>+#endif
>+#undef ANY31CT
>+
> // Any 2 planes to 1.
>-#define ANY21(NAMEANY, ANY_SIMD, UVSHIFT, SBPP, SBPP2, BPP, MASK)       \
>-  void NAMEANY(const uint8* y_buf, const uint8* uv_buf, uint8* dst_ptr, \
>-               int width) {                                             \
>-    SIMD_ALIGNED(uint8 temp[64 * 3]);                                   \
>-    memset(temp, 0, 64 * 2); /* for msan */                             \
>-    int r = width & MASK;                                               \
>-    int n = width & ~MASK;                                              \
>-    if (n > 0) {                                                        \
>-      ANY_SIMD(y_buf, uv_buf, dst_ptr, n);                              \
>-    }                                                                   \
>-    memcpy(temp, y_buf + n * SBPP, r * SBPP);                           \
>-    memcpy(temp + 64, uv_buf + (n >> UVSHIFT) * SBPP2,                  \
>-           SS(r, UVSHIFT) * SBPP2);                                     \
>-    ANY_SIMD(temp, temp + 64, temp + 128, MASK + 1);                    \
>-    memcpy(dst_ptr + n * BPP, temp + 128, r * BPP);                     \
>+#define ANY21(NAMEANY, ANY_SIMD, UVSHIFT, SBPP, SBPP2, BPP, MASK)             \
>+  void NAMEANY(const uint8_t* y_buf, const uint8_t* uv_buf, uint8_t* dst_ptr, \
>+               int width) {                                                   \
>+    SIMD_ALIGNED(uint8_t temp[64 * 3]);                                       \
>+    memset(temp, 0, 64 * 2); /* for msan */                                   \
>+    int r = width & MASK;                                                     \
>+    int n = width & ~MASK;                                                    \
>+    if (n > 0) {                                                              \
>+      ANY_SIMD(y_buf, uv_buf, dst_ptr, n);                                    \
>+    }                                                                         \
>+    memcpy(temp, y_buf + n * SBPP, r * SBPP);                                 \
>+    memcpy(temp + 64, uv_buf + (n >> UVSHIFT) * SBPP2,                        \
>+           SS(r, UVSHIFT) * SBPP2);                                           \
>+    ANY_SIMD(temp, temp + 64, temp + 128, MASK + 1);                          \
>+    memcpy(dst_ptr + n * BPP, temp + 128, r * BPP);                           \
>   }
> 
> // Merge functions.
>@@ -299,21 +338,21 @@ ANY21(SobelXYRow_Any_MSA, SobelXYRow_MSA, 0, 1, 1, 4, 15)
> #undef ANY21
> 
> // Any 2 planes to 1 with yuvconstants
>-#define ANY21C(NAMEANY, ANY_SIMD, UVSHIFT, SBPP, SBPP2, BPP, MASK)      \
>-  void NAMEANY(const uint8* y_buf, const uint8* uv_buf, uint8* dst_ptr, \
>-               const struct YuvConstants* yuvconstants, int width) {    \
>-    SIMD_ALIGNED(uint8 temp[64 * 3]);                                   \
>-    memset(temp, 0, 64 * 2); /* for msan */                             \
>-    int r = width & MASK;                                               \
>-    int n = width & ~MASK;                                              \
>-    if (n > 0) {                                                        \
>-      ANY_SIMD(y_buf, uv_buf, dst_ptr, yuvconstants, n);                \
>-    }                                                                   \
>-    memcpy(temp, y_buf + n * SBPP, r * SBPP);                           \
>-    memcpy(temp + 64, uv_buf + (n >> UVSHIFT) * SBPP2,                  \
>-           SS(r, UVSHIFT) * SBPP2);                                     \
>-    ANY_SIMD(temp, temp + 64, temp + 128, yuvconstants, MASK + 1);      \
>-    memcpy(dst_ptr + n * BPP, temp + 128, r * BPP);                     \
>+#define ANY21C(NAMEANY, ANY_SIMD, UVSHIFT, SBPP, SBPP2, BPP, MASK)            \
>+  void NAMEANY(const uint8_t* y_buf, const uint8_t* uv_buf, uint8_t* dst_ptr, \
>+               const struct YuvConstants* yuvconstants, int width) {          \
>+    SIMD_ALIGNED(uint8_t temp[64 * 3]);                                       \
>+    memset(temp, 0, 64 * 2); /* for msan */                                   \
>+    int r = width & MASK;                                                     \
>+    int n = width & ~MASK;                                                    \
>+    if (n > 0) {                                                              \
>+      ANY_SIMD(y_buf, uv_buf, dst_ptr, yuvconstants, n);                      \
>+    }                                                                         \
>+    memcpy(temp, y_buf + n * SBPP, r * SBPP);                                 \
>+    memcpy(temp + 64, uv_buf + (n >> UVSHIFT) * SBPP2,                        \
>+           SS(r, UVSHIFT) * SBPP2);                                           \
>+    ANY_SIMD(temp, temp + 64, temp + 128, yuvconstants, MASK + 1);            \
>+    memcpy(dst_ptr + n * BPP, temp + 128, r * BPP);                           \
>   }
> 
> // Biplanar to RGB.
>@@ -326,9 +365,6 @@ ANY21C(NV12ToARGBRow_Any_AVX2, NV12ToARGBRow_AVX2, 1, 1, 2, 4, 15)
> #ifdef HAS_NV12TOARGBROW_NEON
> ANY21C(NV12ToARGBRow_Any_NEON, NV12ToARGBRow_NEON, 1, 1, 2, 4, 7)
> #endif
>-#ifdef HAS_NV12TOARGBROW_DSPR2
>-ANY21C(NV12ToARGBRow_Any_DSPR2, NV12ToARGBRow_DSPR2, 1, 1, 2, 4, 7)
>-#endif
> #ifdef HAS_NV12TOARGBROW_MSA
> ANY21C(NV12ToARGBRow_Any_MSA, NV12ToARGBRow_MSA, 1, 1, 2, 4, 7)
> #endif
>@@ -344,6 +380,12 @@ ANY21C(NV21ToARGBRow_Any_NEON, NV21ToARGBRow_NEON, 1, 1, 2, 4, 7)
> #ifdef HAS_NV21TOARGBROW_MSA
> ANY21C(NV21ToARGBRow_Any_MSA, NV21ToARGBRow_MSA, 1, 1, 2, 4, 7)
> #endif
>+#ifdef HAS_NV12TORGB24ROW_NEON
>+ANY21C(NV12ToRGB24Row_Any_NEON, NV12ToRGB24Row_NEON, 1, 1, 2, 3, 7)
>+#endif
>+#ifdef HAS_NV21TORGB24ROW_NEON
>+ANY21C(NV21ToRGB24Row_Any_NEON, NV21ToRGB24Row_NEON, 1, 1, 2, 3, 7)
>+#endif
> #ifdef HAS_NV12TORGB565ROW_SSSE3
> ANY21C(NV12ToRGB565Row_Any_SSSE3, NV12ToRGB565Row_SSSE3, 1, 1, 2, 2, 7)
> #endif
>@@ -360,8 +402,8 @@ ANY21C(NV12ToRGB565Row_Any_MSA, NV12ToRGB565Row_MSA, 1, 1, 2, 2, 7)
> 
> // Any 1 to 1.
> #define ANY11(NAMEANY, ANY_SIMD, UVSHIFT, SBPP, BPP, MASK)                \
>-  void NAMEANY(const uint8* src_ptr, uint8* dst_ptr, int width) {         \
>-    SIMD_ALIGNED(uint8 temp[128 * 2]);                                    \
>+  void NAMEANY(const uint8_t* src_ptr, uint8_t* dst_ptr, int width) {     \
>+    SIMD_ALIGNED(uint8_t temp[128 * 2]);                                  \
>     memset(temp, 0, 128); /* for YUY2 and msan */                         \
>     int r = width & MASK;                                                 \
>     int n = width & ~MASK;                                                \
>@@ -396,6 +438,18 @@ ANY11(ARGBToRGB565Row_Any_AVX2, ARGBToRGB565Row_AVX2, 0, 4, 2, 7)
> ANY11(ARGBToARGB1555Row_Any_AVX2, ARGBToARGB1555Row_AVX2, 0, 4, 2, 7)
> ANY11(ARGBToARGB4444Row_Any_AVX2, ARGBToARGB4444Row_AVX2, 0, 4, 2, 7)
> #endif
>+#if defined(HAS_ABGRTOAR30ROW_SSSE3)
>+ANY11(ABGRToAR30Row_Any_SSSE3, ABGRToAR30Row_SSSE3, 0, 4, 4, 3)
>+#endif
>+#if defined(HAS_ARGBTOAR30ROW_SSSE3)
>+ANY11(ARGBToAR30Row_Any_SSSE3, ARGBToAR30Row_SSSE3, 0, 4, 4, 3)
>+#endif
>+#if defined(HAS_ABGRTOAR30ROW_AVX2)
>+ANY11(ABGRToAR30Row_Any_AVX2, ABGRToAR30Row_AVX2, 0, 4, 4, 7)
>+#endif
>+#if defined(HAS_ARGBTOAR30ROW_AVX2)
>+ANY11(ARGBToAR30Row_Any_AVX2, ARGBToAR30Row_AVX2, 0, 4, 4, 7)
>+#endif
> #if defined(HAS_J400TOARGBROW_SSE2)
> ANY11(J400ToARGBRow_Any_SSE2, J400ToARGBRow_SSE2, 0, 1, 4, 7)
> #endif
>@@ -572,33 +626,6 @@ ANY11(ARGB1555ToARGBRow_Any_MSA, ARGB1555ToARGBRow_MSA, 0, 2, 4, 15)
> #ifdef HAS_ARGB4444TOARGBROW_NEON
> ANY11(ARGB4444ToARGBRow_Any_NEON, ARGB4444ToARGBRow_NEON, 0, 2, 4, 7)
> #endif
>-#ifdef HAS_RGB24TOARGBROW_DSPR2
>-ANY11(RGB24ToARGBRow_Any_DSPR2, RGB24ToARGBRow_DSPR2, 0, 3, 4, 7)
>-#endif
>-#ifdef HAS_RAWTOARGBROW_DSPR2
>-ANY11(RAWToARGBRow_Any_DSPR2, RAWToARGBRow_DSPR2, 0, 3, 4, 7)
>-#endif
>-#ifdef HAS_RGB565TOARGBROW_DSPR2
>-ANY11(RGB565ToARGBRow_Any_DSPR2, RGB565ToARGBRow_DSPR2, 0, 2, 4, 7)
>-#endif
>-#ifdef HAS_ARGB1555TOARGBROW_DSPR2
>-ANY11(ARGB1555ToARGBRow_Any_DSPR2, ARGB1555ToARGBRow_DSPR2, 0, 2, 4, 7)
>-#endif
>-#ifdef HAS_ARGB4444TOARGBROW_DSPR2
>-ANY11(ARGB4444ToARGBRow_Any_DSPR2, ARGB4444ToARGBRow_DSPR2, 0, 2, 4, 7)
>-#endif
>-#ifdef HAS_BGRATOYROW_DSPR2
>-ANY11(BGRAToYRow_Any_DSPR2, BGRAToYRow_DSPR2, 0, 4, 1, 7)
>-#endif
>-#ifdef HAS_ARGBTOYROW_DSPR2
>-ANY11(ARGBToYRow_Any_DSPR2, ARGBToYRow_DSPR2, 0, 4, 1, 7)
>-#endif
>-#ifdef HAS_ABGRTOYROW_DSPR2
>-ANY11(ABGRToYRow_Any_DSPR2, ABGRToYRow_DSPR2, 0, 4, 1, 7)
>-#endif
>-#ifdef HAS_RGBATOYROW_DSPR2
>-ANY11(RGBAToYRow_Any_DSPR2, RGBAToYRow_DSPR2, 0, 4, 1, 7)
>-#endif
> #ifdef HAS_ARGB4444TOARGBROW_MSA
> ANY11(ARGB4444ToARGBRow_Any_MSA, ARGB4444ToARGBRow_MSA, 0, 2, 4, 15)
> #endif
>@@ -636,8 +663,8 @@ ANY11(ARGBExtractAlphaRow_Any_MSA, ARGBExtractAlphaRow_MSA, 0, 4, 1, 15)
> 
> // Any 1 to 1 blended.  Destination is read, modify, write.
> #define ANY11B(NAMEANY, ANY_SIMD, UVSHIFT, SBPP, BPP, MASK)               \
>-  void NAMEANY(const uint8* src_ptr, uint8* dst_ptr, int width) {         \
>-    SIMD_ALIGNED(uint8 temp[64 * 2]);                                     \
>+  void NAMEANY(const uint8_t* src_ptr, uint8_t* dst_ptr, int width) {     \
>+    SIMD_ALIGNED(uint8_t temp[64 * 2]);                                   \
>     memset(temp, 0, 64 * 2); /* for msan */                               \
>     int r = width & MASK;                                                 \
>     int n = width & ~MASK;                                                \
>@@ -665,24 +692,24 @@ ANY11B(ARGBCopyYToAlphaRow_Any_SSE2, ARGBCopyYToAlphaRow_SSE2, 0, 1, 4, 7)
> #undef ANY11B
> 
> // Any 1 to 1 with parameter.
>-#define ANY11P(NAMEANY, ANY_SIMD, T, SBPP, BPP, MASK)                      \
>-  void NAMEANY(const uint8* src_ptr, uint8* dst_ptr, T param, int width) { \
>-    SIMD_ALIGNED(uint8 temp[64 * 2]);                                      \
>-    memset(temp, 0, 64); /* for msan */                                    \
>-    int r = width & MASK;                                                  \
>-    int n = width & ~MASK;                                                 \
>-    if (n > 0) {                                                           \
>-      ANY_SIMD(src_ptr, dst_ptr, param, n);                                \
>-    }                                                                      \
>-    memcpy(temp, src_ptr + n * SBPP, r * SBPP);                            \
>-    ANY_SIMD(temp, temp + 64, param, MASK + 1);                            \
>-    memcpy(dst_ptr + n * BPP, temp + 64, r * BPP);                         \
>+#define ANY11P(NAMEANY, ANY_SIMD, T, SBPP, BPP, MASK)                          \
>+  void NAMEANY(const uint8_t* src_ptr, uint8_t* dst_ptr, T param, int width) { \
>+    SIMD_ALIGNED(uint8_t temp[64 * 2]);                                        \
>+    memset(temp, 0, 64); /* for msan */                                        \
>+    int r = width & MASK;                                                      \
>+    int n = width & ~MASK;                                                     \
>+    if (n > 0) {                                                               \
>+      ANY_SIMD(src_ptr, dst_ptr, param, n);                                    \
>+    }                                                                          \
>+    memcpy(temp, src_ptr + n * SBPP, r * SBPP);                                \
>+    ANY_SIMD(temp, temp + 64, param, MASK + 1);                                \
>+    memcpy(dst_ptr + n * BPP, temp + 64, r * BPP);                             \
>   }
> 
> #if defined(HAS_ARGBTORGB565DITHERROW_SSE2)
> ANY11P(ARGBToRGB565DitherRow_Any_SSE2,
>        ARGBToRGB565DitherRow_SSE2,
>-       const uint32,
>+       const uint32_t,
>        4,
>        2,
>        3)
>@@ -690,7 +717,7 @@ ANY11P(ARGBToRGB565DitherRow_Any_SSE2,
> #if defined(HAS_ARGBTORGB565DITHERROW_AVX2)
> ANY11P(ARGBToRGB565DitherRow_Any_AVX2,
>        ARGBToRGB565DitherRow_AVX2,
>-       const uint32,
>+       const uint32_t,
>        4,
>        2,
>        7)
>@@ -698,7 +725,7 @@ ANY11P(ARGBToRGB565DitherRow_Any_AVX2,
> #if defined(HAS_ARGBTORGB565DITHERROW_NEON)
> ANY11P(ARGBToRGB565DitherRow_Any_NEON,
>        ARGBToRGB565DitherRow_NEON,
>-       const uint32,
>+       const uint32_t,
>        4,
>        2,
>        7)
>@@ -706,67 +733,134 @@ ANY11P(ARGBToRGB565DitherRow_Any_NEON,
> #if defined(HAS_ARGBTORGB565DITHERROW_MSA)
> ANY11P(ARGBToRGB565DitherRow_Any_MSA,
>        ARGBToRGB565DitherRow_MSA,
>-       const uint32,
>+       const uint32_t,
>        4,
>        2,
>        7)
> #endif
>-#ifdef HAS_ARGBSHUFFLEROW_SSE2
>-ANY11P(ARGBShuffleRow_Any_SSE2, ARGBShuffleRow_SSE2, const uint8*, 4, 4, 3)
>-#endif
> #ifdef HAS_ARGBSHUFFLEROW_SSSE3
>-ANY11P(ARGBShuffleRow_Any_SSSE3, ARGBShuffleRow_SSSE3, const uint8*, 4, 4, 7)
>+ANY11P(ARGBShuffleRow_Any_SSSE3, ARGBShuffleRow_SSSE3, const uint8_t*, 4, 4, 7)
> #endif
> #ifdef HAS_ARGBSHUFFLEROW_AVX2
>-ANY11P(ARGBShuffleRow_Any_AVX2, ARGBShuffleRow_AVX2, const uint8*, 4, 4, 15)
>+ANY11P(ARGBShuffleRow_Any_AVX2, ARGBShuffleRow_AVX2, const uint8_t*, 4, 4, 15)
> #endif
> #ifdef HAS_ARGBSHUFFLEROW_NEON
>-ANY11P(ARGBShuffleRow_Any_NEON, ARGBShuffleRow_NEON, const uint8*, 4, 4, 3)
>+ANY11P(ARGBShuffleRow_Any_NEON, ARGBShuffleRow_NEON, const uint8_t*, 4, 4, 3)
> #endif
> #ifdef HAS_ARGBSHUFFLEROW_MSA
>-ANY11P(ARGBShuffleRow_Any_MSA, ARGBShuffleRow_MSA, const uint8*, 4, 4, 7)
>+ANY11P(ARGBShuffleRow_Any_MSA, ARGBShuffleRow_MSA, const uint8_t*, 4, 4, 7)
> #endif
> #undef ANY11P
> 
> // Any 1 to 1 with parameter and shorts.  BPP measures in shorts.
>-#define ANY11P16(NAMEANY, ANY_SIMD, T, SBPP, BPP, MASK)                      \
>-  void NAMEANY(const uint16* src_ptr, uint16* dst_ptr, T param, int width) { \
>-    SIMD_ALIGNED(uint16 temp[16 * 2]);                                       \
>-    memset(temp, 0, 32); /* for msan */                                      \
>+#define ANY11C(NAMEANY, ANY_SIMD, SBPP, BPP, STYPE, DTYPE, MASK)             \
>+  void NAMEANY(const STYPE* src_ptr, DTYPE* dst_ptr, int scale, int width) { \
>+    SIMD_ALIGNED(STYPE temp[32]);                                            \
>+    SIMD_ALIGNED(DTYPE out[32]);                                             \
>+    memset(temp, 0, 32 * SBPP); /* for msan */                               \
>     int r = width & MASK;                                                    \
>     int n = width & ~MASK;                                                   \
>     if (n > 0) {                                                             \
>-      ANY_SIMD(src_ptr, dst_ptr, param, n);                                  \
>+      ANY_SIMD(src_ptr, dst_ptr, scale, n);                                  \
>     }                                                                        \
>     memcpy(temp, src_ptr + n, r * SBPP);                                     \
>-    ANY_SIMD(temp, temp + 16, param, MASK + 1);                              \
>-    memcpy(dst_ptr + n, temp + 16, r * BPP);                                 \
>+    ANY_SIMD(temp, out, scale, MASK + 1);                                    \
>+    memcpy(dst_ptr + n, out, r * BPP);                                       \
>+  }
>+
>+#ifdef HAS_CONVERT16TO8ROW_SSSE3
>+ANY11C(Convert16To8Row_Any_SSSE3,
>+       Convert16To8Row_SSSE3,
>+       2,
>+       1,
>+       uint16_t,
>+       uint8_t,
>+       15)
>+#endif
>+#ifdef HAS_CONVERT16TO8ROW_AVX2
>+ANY11C(Convert16To8Row_Any_AVX2,
>+       Convert16To8Row_AVX2,
>+       2,
>+       1,
>+       uint16_t,
>+       uint8_t,
>+       31)
>+#endif
>+#ifdef HAS_CONVERT8TO16ROW_SSE2
>+ANY11C(Convert8To16Row_Any_SSE2,
>+       Convert8To16Row_SSE2,
>+       1,
>+       2,
>+       uint8_t,
>+       uint16_t,
>+       15)
>+#endif
>+#ifdef HAS_CONVERT8TO16ROW_AVX2
>+ANY11C(Convert8To16Row_Any_AVX2,
>+       Convert8To16Row_AVX2,
>+       1,
>+       2,
>+       uint8_t,
>+       uint16_t,
>+       31)
>+#endif
>+#undef ANY11C
>+
>+// Any 1 to 1 with parameter and shorts to byte.  BPP measures in shorts.
>+#define ANY11P16(NAMEANY, ANY_SIMD, ST, T, SBPP, BPP, MASK)             \
>+  void NAMEANY(const ST* src_ptr, T* dst_ptr, float param, int width) { \
>+    SIMD_ALIGNED(ST temp[32]);                                          \
>+    SIMD_ALIGNED(T out[32]);                                            \
>+    memset(temp, 0, SBPP * 32); /* for msan */                          \
>+    int r = width & MASK;                                               \
>+    int n = width & ~MASK;                                              \
>+    if (n > 0) {                                                        \
>+      ANY_SIMD(src_ptr, dst_ptr, param, n);                             \
>+    }                                                                   \
>+    memcpy(temp, src_ptr + n, r * SBPP);                                \
>+    ANY_SIMD(temp, out, param, MASK + 1);                               \
>+    memcpy(dst_ptr + n, out, r * BPP);                                  \
>   }
> 
> #ifdef HAS_HALFFLOATROW_SSE2
>-ANY11P16(HalfFloatRow_Any_SSE2, HalfFloatRow_SSE2, float, 2, 2, 7)
>+ANY11P16(HalfFloatRow_Any_SSE2, HalfFloatRow_SSE2, uint16_t, uint16_t, 2, 2, 7)
> #endif
> #ifdef HAS_HALFFLOATROW_AVX2
>-ANY11P16(HalfFloatRow_Any_AVX2, HalfFloatRow_AVX2, float, 2, 2, 15)
>+ANY11P16(HalfFloatRow_Any_AVX2, HalfFloatRow_AVX2, uint16_t, uint16_t, 2, 2, 15)
> #endif
> #ifdef HAS_HALFFLOATROW_F16C
>-ANY11P16(HalfFloatRow_Any_F16C, HalfFloatRow_F16C, float, 2, 2, 15)
>-ANY11P16(HalfFloat1Row_Any_F16C, HalfFloat1Row_F16C, float, 2, 2, 15)
>+ANY11P16(HalfFloatRow_Any_F16C, HalfFloatRow_F16C, uint16_t, uint16_t, 2, 2, 15)
>+ANY11P16(HalfFloat1Row_Any_F16C,
>+         HalfFloat1Row_F16C,
>+         uint16_t,
>+         uint16_t,
>+         2,
>+         2,
>+         15)
> #endif
> #ifdef HAS_HALFFLOATROW_NEON
>-ANY11P16(HalfFloatRow_Any_NEON, HalfFloatRow_NEON, float, 2, 2, 7)
>-ANY11P16(HalfFloat1Row_Any_NEON, HalfFloat1Row_NEON, float, 2, 2, 7)
>+ANY11P16(HalfFloatRow_Any_NEON, HalfFloatRow_NEON, uint16_t, uint16_t, 2, 2, 7)
>+ANY11P16(HalfFloat1Row_Any_NEON,
>+         HalfFloat1Row_NEON,
>+         uint16_t,
>+         uint16_t,
>+         2,
>+         2,
>+         7)
> #endif
> #ifdef HAS_HALFFLOATROW_MSA
>-ANY11P16(HalfFloatRow_Any_MSA, HalfFloatRow_MSA, float, 2, 2, 31)
>+ANY11P16(HalfFloatRow_Any_MSA, HalfFloatRow_MSA, uint16_t, uint16_t, 2, 2, 31)
>+#endif
>+#ifdef HAS_BYTETOFLOATROW_NEON
>+ANY11P16(ByteToFloatRow_Any_NEON, ByteToFloatRow_NEON, uint8_t, float, 1, 3, 7)
> #endif
> #undef ANY11P16
> 
> // Any 1 to 1 with yuvconstants
> #define ANY11C(NAMEANY, ANY_SIMD, UVSHIFT, SBPP, BPP, MASK)               \
>-  void NAMEANY(const uint8* src_ptr, uint8* dst_ptr,                      \
>+  void NAMEANY(const uint8_t* src_ptr, uint8_t* dst_ptr,                  \
>                const struct YuvConstants* yuvconstants, int width) {      \
>-    SIMD_ALIGNED(uint8 temp[128 * 2]);                                    \
>+    SIMD_ALIGNED(uint8_t temp[128 * 2]);                                  \
>     memset(temp, 0, 128); /* for YUY2 and msan */                         \
>     int r = width & MASK;                                                 \
>     int n = width & ~MASK;                                                \
>@@ -796,20 +890,20 @@ ANY11C(UYVYToARGBRow_Any_MSA, UYVYToARGBRow_MSA, 1, 4, 4, 7)
> #undef ANY11C
> 
> // Any 1 to 1 interpolate.  Takes 2 rows of source via stride.
>-#define ANY11T(NAMEANY, ANY_SIMD, SBPP, BPP, MASK)                             \
>-  void NAMEANY(uint8* dst_ptr, const uint8* src_ptr, ptrdiff_t src_stride_ptr, \
>-               int width, int source_y_fraction) {                             \
>-    SIMD_ALIGNED(uint8 temp[64 * 3]);                                          \
>-    memset(temp, 0, 64 * 2); /* for msan */                                    \
>-    int r = width & MASK;                                                      \
>-    int n = width & ~MASK;                                                     \
>-    if (n > 0) {                                                               \
>-      ANY_SIMD(dst_ptr, src_ptr, src_stride_ptr, n, source_y_fraction);        \
>-    }                                                                          \
>-    memcpy(temp, src_ptr + n * SBPP, r * SBPP);                                \
>-    memcpy(temp + 64, src_ptr + src_stride_ptr + n * SBPP, r * SBPP);          \
>-    ANY_SIMD(temp + 128, temp, 64, MASK + 1, source_y_fraction);               \
>-    memcpy(dst_ptr + n * BPP, temp + 128, r * BPP);                            \
>+#define ANY11T(NAMEANY, ANY_SIMD, SBPP, BPP, MASK)                           \
>+  void NAMEANY(uint8_t* dst_ptr, const uint8_t* src_ptr,                     \
>+               ptrdiff_t src_stride_ptr, int width, int source_y_fraction) { \
>+    SIMD_ALIGNED(uint8_t temp[64 * 3]);                                      \
>+    memset(temp, 0, 64 * 2); /* for msan */                                  \
>+    int r = width & MASK;                                                    \
>+    int n = width & ~MASK;                                                   \
>+    if (n > 0) {                                                             \
>+      ANY_SIMD(dst_ptr, src_ptr, src_stride_ptr, n, source_y_fraction);      \
>+    }                                                                        \
>+    memcpy(temp, src_ptr + n * SBPP, r * SBPP);                              \
>+    memcpy(temp + 64, src_ptr + src_stride_ptr + n * SBPP, r * SBPP);        \
>+    ANY_SIMD(temp + 128, temp, 64, MASK + 1, source_y_fraction);             \
>+    memcpy(dst_ptr + n * BPP, temp + 128, r * BPP);                          \
>   }
> 
> #ifdef HAS_INTERPOLATEROW_AVX2
>@@ -821,9 +915,6 @@ ANY11T(InterpolateRow_Any_SSSE3, InterpolateRow_SSSE3, 1, 1, 15)
> #ifdef HAS_INTERPOLATEROW_NEON
> ANY11T(InterpolateRow_Any_NEON, InterpolateRow_NEON, 1, 1, 15)
> #endif
>-#ifdef HAS_INTERPOLATEROW_DSPR2
>-ANY11T(InterpolateRow_Any_DSPR2, InterpolateRow_DSPR2, 1, 1, 3)
>-#endif
> #ifdef HAS_INTERPOLATEROW_MSA
> ANY11T(InterpolateRow_Any_MSA, InterpolateRow_MSA, 1, 1, 31)
> #endif
>@@ -831,8 +922,8 @@ ANY11T(InterpolateRow_Any_MSA, InterpolateRow_MSA, 1, 1, 31)
> 
> // Any 1 to 1 mirror.
> #define ANY11M(NAMEANY, ANY_SIMD, BPP, MASK)                              \
>-  void NAMEANY(const uint8* src_ptr, uint8* dst_ptr, int width) {         \
>-    SIMD_ALIGNED(uint8 temp[64 * 2]);                                     \
>+  void NAMEANY(const uint8_t* src_ptr, uint8_t* dst_ptr, int width) {     \
>+    SIMD_ALIGNED(uint8_t temp[64 * 2]);                                   \
>     memset(temp, 0, 64); /* for msan */                                   \
>     int r = width & MASK;                                                 \
>     int n = width & ~MASK;                                                \
>@@ -871,46 +962,47 @@ ANY11M(ARGBMirrorRow_Any_MSA, ARGBMirrorRow_MSA, 4, 15)
> #undef ANY11M
> 
> // Any 1 plane. (memset)
>-#define ANY1(NAMEANY, ANY_SIMD, T, BPP, MASK)      \
>-  void NAMEANY(uint8* dst_ptr, T v32, int width) { \
>-    SIMD_ALIGNED(uint8 temp[64]);                  \
>-    int r = width & MASK;                          \
>-    int n = width & ~MASK;                         \
>-    if (n > 0) {                                   \
>-      ANY_SIMD(dst_ptr, v32, n);                   \
>-    }                                              \
>-    ANY_SIMD(temp, v32, MASK + 1);                 \
>-    memcpy(dst_ptr + n * BPP, temp, r * BPP);      \
>+#define ANY1(NAMEANY, ANY_SIMD, T, BPP, MASK)        \
>+  void NAMEANY(uint8_t* dst_ptr, T v32, int width) { \
>+    SIMD_ALIGNED(uint8_t temp[64]);                  \
>+    int r = width & MASK;                            \
>+    int n = width & ~MASK;                           \
>+    if (n > 0) {                                     \
>+      ANY_SIMD(dst_ptr, v32, n);                     \
>+    }                                                \
>+    ANY_SIMD(temp, v32, MASK + 1);                   \
>+    memcpy(dst_ptr + n * BPP, temp, r * BPP);        \
>   }
> 
> #ifdef HAS_SETROW_X86
>-ANY1(SetRow_Any_X86, SetRow_X86, uint8, 1, 3)
>+ANY1(SetRow_Any_X86, SetRow_X86, uint8_t, 1, 3)
> #endif
> #ifdef HAS_SETROW_NEON
>-ANY1(SetRow_Any_NEON, SetRow_NEON, uint8, 1, 15)
>+ANY1(SetRow_Any_NEON, SetRow_NEON, uint8_t, 1, 15)
> #endif
> #ifdef HAS_ARGBSETROW_NEON
>-ANY1(ARGBSetRow_Any_NEON, ARGBSetRow_NEON, uint32, 4, 3)
>+ANY1(ARGBSetRow_Any_NEON, ARGBSetRow_NEON, uint32_t, 4, 3)
> #endif
> #ifdef HAS_ARGBSETROW_MSA
>-ANY1(ARGBSetRow_Any_MSA, ARGBSetRow_MSA, uint32, 4, 3)
>+ANY1(ARGBSetRow_Any_MSA, ARGBSetRow_MSA, uint32_t, 4, 3)
> #endif
> #undef ANY1
> 
> // Any 1 to 2.  Outputs UV planes.
>-#define ANY12(NAMEANY, ANY_SIMD, UVSHIFT, BPP, DUVSHIFT, MASK)                \
>-  void NAMEANY(const uint8* src_ptr, uint8* dst_u, uint8* dst_v, int width) { \
>-    SIMD_ALIGNED(uint8 temp[128 * 3]);                                        \
>-    memset(temp, 0, 128); /* for msan */                                      \
>-    int r = width & MASK;                                                     \
>-    int n = width & ~MASK;                                                    \
>-    if (n > 0) {                                                              \
>-      ANY_SIMD(src_ptr, dst_u, dst_v, n);                                     \
>-    }                                                                         \
>-    memcpy(temp, src_ptr + (n >> UVSHIFT) * BPP, SS(r, UVSHIFT) * BPP);       \
>-    ANY_SIMD(temp, temp + 128, temp + 256, MASK + 1);                         \
>-    memcpy(dst_u + (n >> DUVSHIFT), temp + 128, SS(r, DUVSHIFT));             \
>-    memcpy(dst_v + (n >> DUVSHIFT), temp + 256, SS(r, DUVSHIFT));             \
>+#define ANY12(NAMEANY, ANY_SIMD, UVSHIFT, BPP, DUVSHIFT, MASK)          \
>+  void NAMEANY(const uint8_t* src_ptr, uint8_t* dst_u, uint8_t* dst_v,  \
>+               int width) {                                             \
>+    SIMD_ALIGNED(uint8_t temp[128 * 3]);                                \
>+    memset(temp, 0, 128); /* for msan */                                \
>+    int r = width & MASK;                                               \
>+    int n = width & ~MASK;                                              \
>+    if (n > 0) {                                                        \
>+      ANY_SIMD(src_ptr, dst_u, dst_v, n);                               \
>+    }                                                                   \
>+    memcpy(temp, src_ptr + (n >> UVSHIFT) * BPP, SS(r, UVSHIFT) * BPP); \
>+    ANY_SIMD(temp, temp + 128, temp + 256, MASK + 1);                   \
>+    memcpy(dst_u + (n >> DUVSHIFT), temp + 128, SS(r, DUVSHIFT));       \
>+    memcpy(dst_v + (n >> DUVSHIFT), temp + 256, SS(r, DUVSHIFT));       \
>   }
> 
> #ifdef HAS_SPLITUVROW_SSE2
>@@ -922,9 +1014,6 @@ ANY12(SplitUVRow_Any_AVX2, SplitUVRow_AVX2, 0, 2, 0, 31)
> #ifdef HAS_SPLITUVROW_NEON
> ANY12(SplitUVRow_Any_NEON, SplitUVRow_NEON, 0, 2, 0, 15)
> #endif
>-#ifdef HAS_SPLITUVROW_DSPR2
>-ANY12(SplitUVRow_Any_DSPR2, SplitUVRow_DSPR2, 0, 2, 0, 15)
>-#endif
> #ifdef HAS_SPLITUVROW_MSA
> ANY12(SplitUVRow_Any_MSA, SplitUVRow_MSA, 0, 2, 0, 31)
> #endif
>@@ -952,21 +1041,21 @@ ANY12(UYVYToUV422Row_Any_MSA, UYVYToUV422Row_MSA, 1, 4, 1, 31)
> #undef ANY12
> 
> // Any 1 to 3.  Outputs RGB planes.
>-#define ANY13(NAMEANY, ANY_SIMD, BPP, MASK)                                    \
>-  void NAMEANY(const uint8* src_ptr, uint8* dst_r, uint8* dst_g, uint8* dst_b, \
>-               int width) {                                                    \
>-    SIMD_ALIGNED(uint8 temp[16 * 6]);                                          \
>-    memset(temp, 0, 16 * 3); /* for msan */                                    \
>-    int r = width & MASK;                                                      \
>-    int n = width & ~MASK;                                                     \
>-    if (n > 0) {                                                               \
>-      ANY_SIMD(src_ptr, dst_r, dst_g, dst_b, n);                               \
>-    }                                                                          \
>-    memcpy(temp, src_ptr + n * BPP, r * BPP);                                  \
>-    ANY_SIMD(temp, temp + 16 * 3, temp + 16 * 4, temp + 16 * 5, MASK + 1);     \
>-    memcpy(dst_r + n, temp + 16 * 3, r);                                       \
>-    memcpy(dst_g + n, temp + 16 * 4, r);                                       \
>-    memcpy(dst_b + n, temp + 16 * 5, r);                                       \
>+#define ANY13(NAMEANY, ANY_SIMD, BPP, MASK)                                \
>+  void NAMEANY(const uint8_t* src_ptr, uint8_t* dst_r, uint8_t* dst_g,     \
>+               uint8_t* dst_b, int width) {                                \
>+    SIMD_ALIGNED(uint8_t temp[16 * 6]);                                    \
>+    memset(temp, 0, 16 * 3); /* for msan */                                \
>+    int r = width & MASK;                                                  \
>+    int n = width & ~MASK;                                                 \
>+    if (n > 0) {                                                           \
>+      ANY_SIMD(src_ptr, dst_r, dst_g, dst_b, n);                           \
>+    }                                                                      \
>+    memcpy(temp, src_ptr + n * BPP, r * BPP);                              \
>+    ANY_SIMD(temp, temp + 16 * 3, temp + 16 * 4, temp + 16 * 5, MASK + 1); \
>+    memcpy(dst_r + n, temp + 16 * 3, r);                                   \
>+    memcpy(dst_g + n, temp + 16 * 4, r);                                   \
>+    memcpy(dst_b + n, temp + 16 * 5, r);                                   \
>   }
> 
> #ifdef HAS_SPLITRGBROW_SSSE3
>@@ -979,9 +1068,9 @@ ANY13(SplitRGBRow_Any_NEON, SplitRGBRow_NEON, 3, 15)
> // Any 1 to 2 with source stride (2 rows of source).  Outputs UV planes.
> // 128 byte row allows for 32 avx ARGB pixels.
> #define ANY12S(NAMEANY, ANY_SIMD, UVSHIFT, BPP, MASK)                        \
>-  void NAMEANY(const uint8* src_ptr, int src_stride_ptr, uint8* dst_u,       \
>-               uint8* dst_v, int width) {                                    \
>-    SIMD_ALIGNED(uint8 temp[128 * 4]);                                       \
>+  void NAMEANY(const uint8_t* src_ptr, int src_stride_ptr, uint8_t* dst_u,   \
>+               uint8_t* dst_v, int width) {                                  \
>+    SIMD_ALIGNED(uint8_t temp[128 * 4]);                                     \
>     memset(temp, 0, 128 * 2); /* for msan */                                 \
>     int r = width & MASK;                                                    \
>     int n = width & ~MASK;                                                   \
>@@ -1086,18 +1175,6 @@ ANY12S(YUY2ToUVRow_Any_NEON, YUY2ToUVRow_NEON, 1, 4, 15)
> #ifdef HAS_UYVYTOUVROW_NEON
> ANY12S(UYVYToUVRow_Any_NEON, UYVYToUVRow_NEON, 1, 4, 15)
> #endif
>-#ifdef HAS_BGRATOUVROW_DSPR2
>-ANY12S(BGRAToUVRow_Any_DSPR2, BGRAToUVRow_DSPR2, 0, 4, 15)
>-#endif
>-#ifdef HAS_ABGRTOUVROW_DSPR2
>-ANY12S(ABGRToUVRow_Any_DSPR2, ABGRToUVRow_DSPR2, 0, 4, 15)
>-#endif
>-#ifdef HAS_RGBATOUVROW_DSPR2
>-ANY12S(RGBAToUVRow_Any_DSPR2, RGBAToUVRow_DSPR2, 0, 4, 15)
>-#endif
>-#ifdef HAS_ARGBTOUVROW_DSPR2
>-ANY12S(ARGBToUVRow_Any_DSPR2, ARGBToUVRow_DSPR2, 0, 4, 15)
>-#endif
> #ifdef HAS_YUY2TOUVROW_MSA
> ANY12S(YUY2ToUVRow_Any_MSA, YUY2ToUVRow_MSA, 1, 4, 31)
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/row_common.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/row_common.cc
>index 6ffc4febbf6..da97821f796 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/row_common.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/row_common.cc
>@@ -10,6 +10,7 @@
> 
> #include "libyuv/row.h"
> 
>+#include <stdio.h>
> #include <string.h>  // For memcpy and memset.
> 
> #include "libyuv/basic_types.h"
>@@ -23,59 +24,69 @@ extern "C" {
> 
> #define USE_BRANCHLESS 1
> #if USE_BRANCHLESS
>-static __inline int32 clamp0(int32 v) {
>+static __inline int32_t clamp0(int32_t v) {
>   return ((-(v) >> 31) & (v));
> }
> 
>-static __inline int32 clamp255(int32 v) {
>+static __inline int32_t clamp255(int32_t v) {
>   return (((255 - (v)) >> 31) | (v)) & 255;
> }
> 
>-static __inline uint32 Clamp(int32 val) {
>-  int v = clamp0(val);
>-  return (uint32)(clamp255(v));
>+static __inline int32_t clamp1023(int32_t v) {
>+  return (((1023 - (v)) >> 31) | (v)) & 1023;
> }
> 
>-static __inline uint32 Abs(int32 v) {
>+static __inline uint32_t Abs(int32_t v) {
>   int m = v >> 31;
>   return (v + m) ^ m;
> }
> #else   // USE_BRANCHLESS
>-static __inline int32 clamp0(int32 v) {
>+static __inline int32_t clamp0(int32_t v) {
>   return (v < 0) ? 0 : v;
> }
> 
>-static __inline int32 clamp255(int32 v) {
>+static __inline int32_t clamp255(int32_t v) {
>   return (v > 255) ? 255 : v;
> }
> 
>-static __inline uint32 Clamp(int32 val) {
>-  int v = clamp0(val);
>-  return (uint32)(clamp255(v));
>+static __inline int32_t clamp1023(int32_t v) {
>+  return (v > 1023) ? 1023 : v;
> }
> 
>-static __inline uint32 Abs(int32 v) {
>+static __inline uint32_t Abs(int32_t v) {
>   return (v < 0) ? -v : v;
> }
> #endif  // USE_BRANCHLESS
>+static __inline uint32_t Clamp(int32_t val) {
>+  int v = clamp0(val);
>+  return (uint32_t)(clamp255(v));
>+}
>+
>+static __inline uint32_t Clamp10(int32_t val) {
>+  int v = clamp0(val);
>+  return (uint32_t)(clamp1023(v));
>+}
> 
>-#ifdef LIBYUV_LITTLE_ENDIAN
>-#define WRITEWORD(p, v) *(uint32*)(p) = v
>+// Little Endian
>+#if defined(__x86_64__) || defined(_M_X64) || defined(__i386__) || \
>+    defined(_M_IX86) || defined(__arm__) || defined(_M_ARM) ||     \
>+    (defined(__BYTE_ORDER__) && __BYTE_ORDER__ == __ORDER_LITTLE_ENDIAN__)
>+#define WRITEWORD(p, v) *(uint32_t*)(p) = v
> #else
>-static inline void WRITEWORD(uint8* p, uint32 v) {
>-  p[0] = (uint8)(v & 255);
>-  p[1] = (uint8)((v >> 8) & 255);
>-  p[2] = (uint8)((v >> 16) & 255);
>-  p[3] = (uint8)((v >> 24) & 255);
>+static inline void WRITEWORD(uint8_t* p, uint32_t v) {
>+  p[0] = (uint8_t)(v & 255);
>+  p[1] = (uint8_t)((v >> 8) & 255);
>+  p[2] = (uint8_t)((v >> 16) & 255);
>+  p[3] = (uint8_t)((v >> 24) & 255);
> }
> #endif
> 
>-void RGB24ToARGBRow_C(const uint8* src_rgb24, uint8* dst_argb, int width) {
>+void RGB24ToARGBRow_C(const uint8_t* src_rgb24, uint8_t* dst_argb, int width) {
>   int x;
>   for (x = 0; x < width; ++x) {
>-    uint8 b = src_rgb24[0];
>-    uint8 g = src_rgb24[1];
>-    uint8 r = src_rgb24[2];
>+    uint8_t b = src_rgb24[0];
>+    uint8_t g = src_rgb24[1];
>+    uint8_t r = src_rgb24[2];
>     dst_argb[0] = b;
>     dst_argb[1] = g;
>     dst_argb[2] = r;
>@@ -85,12 +96,12 @@ void RGB24ToARGBRow_C(const uint8* src_rgb24, uint8* dst_argb, int width) {
>   }
> }
> 
>-void RAWToARGBRow_C(const uint8* src_raw, uint8* dst_argb, int width) {
>+void RAWToARGBRow_C(const uint8_t* src_raw, uint8_t* dst_argb, int width) {
>   int x;
>   for (x = 0; x < width; ++x) {
>-    uint8 r = src_raw[0];
>-    uint8 g = src_raw[1];
>-    uint8 b = src_raw[2];
>+    uint8_t r = src_raw[0];
>+    uint8_t g = src_raw[1];
>+    uint8_t b = src_raw[2];
>     dst_argb[0] = b;
>     dst_argb[1] = g;
>     dst_argb[2] = r;
>@@ -100,12 +111,12 @@ void RAWToARGBRow_C(const uint8* src_raw, uint8* dst_argb, int width) {
>   }
> }
> 
>-void RAWToRGB24Row_C(const uint8* src_raw, uint8* dst_rgb24, int width) {
>+void RAWToRGB24Row_C(const uint8_t* src_raw, uint8_t* dst_rgb24, int width) {
>   int x;
>   for (x = 0; x < width; ++x) {
>-    uint8 r = src_raw[0];
>-    uint8 g = src_raw[1];
>-    uint8 b = src_raw[2];
>+    uint8_t r = src_raw[0];
>+    uint8_t g = src_raw[1];
>+    uint8_t b = src_raw[2];
>     dst_rgb24[0] = b;
>     dst_rgb24[1] = g;
>     dst_rgb24[2] = r;
>@@ -114,12 +125,14 @@ void RAWToRGB24Row_C(const uint8* src_raw, uint8* dst_rgb24, int width) {
>   }
> }
> 
>-void RGB565ToARGBRow_C(const uint8* src_rgb565, uint8* dst_argb, int width) {
>+void RGB565ToARGBRow_C(const uint8_t* src_rgb565,
>+                       uint8_t* dst_argb,
>+                       int width) {
>   int x;
>   for (x = 0; x < width; ++x) {
>-    uint8 b = src_rgb565[0] & 0x1f;
>-    uint8 g = (src_rgb565[0] >> 5) | ((src_rgb565[1] & 0x07) << 3);
>-    uint8 r = src_rgb565[1] >> 3;
>+    uint8_t b = src_rgb565[0] & 0x1f;
>+    uint8_t g = (src_rgb565[0] >> 5) | ((src_rgb565[1] & 0x07) << 3);
>+    uint8_t r = src_rgb565[1] >> 3;
>     dst_argb[0] = (b << 3) | (b >> 2);
>     dst_argb[1] = (g << 2) | (g >> 4);
>     dst_argb[2] = (r << 3) | (r >> 2);
>@@ -129,15 +142,15 @@ void RGB565ToARGBRow_C(const uint8* src_rgb565, uint8* dst_argb, int width) {
>   }
> }
> 
>-void ARGB1555ToARGBRow_C(const uint8* src_argb1555,
>-                         uint8* dst_argb,
>+void ARGB1555ToARGBRow_C(const uint8_t* src_argb1555,
>+                         uint8_t* dst_argb,
>                          int width) {
>   int x;
>   for (x = 0; x < width; ++x) {
>-    uint8 b = src_argb1555[0] & 0x1f;
>-    uint8 g = (src_argb1555[0] >> 5) | ((src_argb1555[1] & 0x03) << 3);
>-    uint8 r = (src_argb1555[1] & 0x7c) >> 2;
>-    uint8 a = src_argb1555[1] >> 7;
>+    uint8_t b = src_argb1555[0] & 0x1f;
>+    uint8_t g = (src_argb1555[0] >> 5) | ((src_argb1555[1] & 0x03) << 3);
>+    uint8_t r = (src_argb1555[1] & 0x7c) >> 2;
>+    uint8_t a = src_argb1555[1] >> 7;
>     dst_argb[0] = (b << 3) | (b >> 2);
>     dst_argb[1] = (g << 3) | (g >> 2);
>     dst_argb[2] = (r << 3) | (r >> 2);
>@@ -147,15 +160,15 @@ void ARGB1555ToARGBRow_C(const uint8* src_argb1555,
>   }
> }
> 
>-void ARGB4444ToARGBRow_C(const uint8* src_argb4444,
>-                         uint8* dst_argb,
>+void ARGB4444ToARGBRow_C(const uint8_t* src_argb4444,
>+                         uint8_t* dst_argb,
>                          int width) {
>   int x;
>   for (x = 0; x < width; ++x) {
>-    uint8 b = src_argb4444[0] & 0x0f;
>-    uint8 g = src_argb4444[0] >> 4;
>-    uint8 r = src_argb4444[1] & 0x0f;
>-    uint8 a = src_argb4444[1] >> 4;
>+    uint8_t b = src_argb4444[0] & 0x0f;
>+    uint8_t g = src_argb4444[0] >> 4;
>+    uint8_t r = src_argb4444[1] & 0x0f;
>+    uint8_t a = src_argb4444[1] >> 4;
>     dst_argb[0] = (b << 4) | b;
>     dst_argb[1] = (g << 4) | g;
>     dst_argb[2] = (r << 4) | r;
>@@ -165,12 +178,53 @@ void ARGB4444ToARGBRow_C(const uint8* src_argb4444,
>   }
> }
> 
>-void ARGBToRGB24Row_C(const uint8* src_argb, uint8* dst_rgb, int width) {
>+void AR30ToARGBRow_C(const uint8_t* src_ar30, uint8_t* dst_argb, int width) {
>   int x;
>   for (x = 0; x < width; ++x) {
>-    uint8 b = src_argb[0];
>-    uint8 g = src_argb[1];
>-    uint8 r = src_argb[2];
>+    uint32_t ar30 = *(uint32_t*)src_ar30;
>+    uint32_t b = (ar30 >> 2) & 0xff;
>+    uint32_t g = (ar30 >> 12) & 0xff;
>+    uint32_t r = (ar30 >> 22) & 0xff;
>+    uint32_t a = (ar30 >> 30) * 0x55;  // Replicate 2 bits to 8 bits.
>+    *(uint32_t*)(dst_argb) = b | (g << 8) | (r << 16) | (a << 24);
>+    dst_argb += 4;
>+    src_ar30 += 4;
>+  }
>+}
>+
>+void AR30ToABGRRow_C(const uint8_t* src_ar30, uint8_t* dst_abgr, int width) {
>+  int x;
>+  for (x = 0; x < width; ++x) {
>+    uint32_t ar30 = *(uint32_t*)src_ar30;
>+    uint32_t b = (ar30 >> 2) & 0xff;
>+    uint32_t g = (ar30 >> 12) & 0xff;
>+    uint32_t r = (ar30 >> 22) & 0xff;
>+    uint32_t a = (ar30 >> 30) * 0x55;  // Replicate 2 bits to 8 bits.
>+    *(uint32_t*)(dst_abgr) = r | (g << 8) | (b << 16) | (a << 24);
>+    dst_abgr += 4;
>+    src_ar30 += 4;
>+  }
>+}
>+
>+void AR30ToAB30Row_C(const uint8_t* src_ar30, uint8_t* dst_ab30, int width) {
>+  int x;
>+  for (x = 0; x < width; ++x) {
>+    uint32_t ar30 = *(uint32_t*)src_ar30;
>+    uint32_t b = ar30 & 0x3ff;
>+    uint32_t ga = ar30 & 0xc00ffc00;
>+    uint32_t r = (ar30 >> 20) & 0x3ff;
>+    *(uint32_t*)(dst_ab30) = r | ga | (b << 20);
>+    dst_ab30 += 4;
>+    src_ar30 += 4;
>+  }
>+}
>+
>+void ARGBToRGB24Row_C(const uint8_t* src_argb, uint8_t* dst_rgb, int width) {
>+  int x;
>+  for (x = 0; x < width; ++x) {
>+    uint8_t b = src_argb[0];
>+    uint8_t g = src_argb[1];
>+    uint8_t r = src_argb[2];
>     dst_rgb[0] = b;
>     dst_rgb[1] = g;
>     dst_rgb[2] = r;
>@@ -179,12 +233,12 @@ void ARGBToRGB24Row_C(const uint8* src_argb, uint8* dst_rgb, int width) {
>   }
> }
> 
>-void ARGBToRAWRow_C(const uint8* src_argb, uint8* dst_rgb, int width) {
>+void ARGBToRAWRow_C(const uint8_t* src_argb, uint8_t* dst_rgb, int width) {
>   int x;
>   for (x = 0; x < width; ++x) {
>-    uint8 b = src_argb[0];
>-    uint8 g = src_argb[1];
>-    uint8 r = src_argb[2];
>+    uint8_t b = src_argb[0];
>+    uint8_t g = src_argb[1];
>+    uint8_t r = src_argb[2];
>     dst_rgb[0] = r;
>     dst_rgb[1] = g;
>     dst_rgb[2] = b;
>@@ -193,25 +247,25 @@ void ARGBToRAWRow_C(const uint8* src_argb, uint8* dst_rgb, int width) {
>   }
> }
> 
>-void ARGBToRGB565Row_C(const uint8* src_argb, uint8* dst_rgb, int width) {
>+void ARGBToRGB565Row_C(const uint8_t* src_argb, uint8_t* dst_rgb, int width) {
>   int x;
>   for (x = 0; x < width - 1; x += 2) {
>-    uint8 b0 = src_argb[0] >> 3;
>-    uint8 g0 = src_argb[1] >> 2;
>-    uint8 r0 = src_argb[2] >> 3;
>-    uint8 b1 = src_argb[4] >> 3;
>-    uint8 g1 = src_argb[5] >> 2;
>-    uint8 r1 = src_argb[6] >> 3;
>+    uint8_t b0 = src_argb[0] >> 3;
>+    uint8_t g0 = src_argb[1] >> 2;
>+    uint8_t r0 = src_argb[2] >> 3;
>+    uint8_t b1 = src_argb[4] >> 3;
>+    uint8_t g1 = src_argb[5] >> 2;
>+    uint8_t r1 = src_argb[6] >> 3;
>     WRITEWORD(dst_rgb, b0 | (g0 << 5) | (r0 << 11) | (b1 << 16) | (g1 << 21) |
>                            (r1 << 27));
>     dst_rgb += 4;
>     src_argb += 8;
>   }
>   if (width & 1) {
>-    uint8 b0 = src_argb[0] >> 3;
>-    uint8 g0 = src_argb[1] >> 2;
>-    uint8 r0 = src_argb[2] >> 3;
>-    *(uint16*)(dst_rgb) = b0 | (g0 << 5) | (r0 << 11);
>+    uint8_t b0 = src_argb[0] >> 3;
>+    uint8_t g0 = src_argb[1] >> 2;
>+    uint8_t r0 = src_argb[2] >> 3;
>+    *(uint16_t*)(dst_rgb) = b0 | (g0 << 5) | (r0 << 11);
>   }
> }
> 
>@@ -223,20 +277,20 @@ void ARGBToRGB565Row_C(const uint8* src_argb, uint8* dst_rgb, int width) {
> // endian will not affect order of the original matrix.  But the dither4
> // will containing the first pixel in the lower byte for little endian
> // or the upper byte for big endian.
>-void ARGBToRGB565DitherRow_C(const uint8* src_argb,
>-                             uint8* dst_rgb,
>-                             const uint32 dither4,
>+void ARGBToRGB565DitherRow_C(const uint8_t* src_argb,
>+                             uint8_t* dst_rgb,
>+                             const uint32_t dither4,
>                              int width) {
>   int x;
>   for (x = 0; x < width - 1; x += 2) {
>     int dither0 = ((const unsigned char*)(&dither4))[x & 3];
>     int dither1 = ((const unsigned char*)(&dither4))[(x + 1) & 3];
>-    uint8 b0 = clamp255(src_argb[0] + dither0) >> 3;
>-    uint8 g0 = clamp255(src_argb[1] + dither0) >> 2;
>-    uint8 r0 = clamp255(src_argb[2] + dither0) >> 3;
>-    uint8 b1 = clamp255(src_argb[4] + dither1) >> 3;
>-    uint8 g1 = clamp255(src_argb[5] + dither1) >> 2;
>-    uint8 r1 = clamp255(src_argb[6] + dither1) >> 3;
>+    uint8_t b0 = clamp255(src_argb[0] + dither0) >> 3;
>+    uint8_t g0 = clamp255(src_argb[1] + dither0) >> 2;
>+    uint8_t r0 = clamp255(src_argb[2] + dither0) >> 3;
>+    uint8_t b1 = clamp255(src_argb[4] + dither1) >> 3;
>+    uint8_t g1 = clamp255(src_argb[5] + dither1) >> 2;
>+    uint8_t r1 = clamp255(src_argb[6] + dither1) >> 3;
>     WRITEWORD(dst_rgb, b0 | (g0 << 5) | (r0 << 11) | (b1 << 16) | (g1 << 21) |
>                            (r1 << 27));
>     dst_rgb += 4;
>@@ -244,112 +298,138 @@ void ARGBToRGB565DitherRow_C(const uint8* src_argb,
>   }
>   if (width & 1) {
>     int dither0 = ((const unsigned char*)(&dither4))[(width - 1) & 3];
>-    uint8 b0 = clamp255(src_argb[0] + dither0) >> 3;
>-    uint8 g0 = clamp255(src_argb[1] + dither0) >> 2;
>-    uint8 r0 = clamp255(src_argb[2] + dither0) >> 3;
>-    *(uint16*)(dst_rgb) = b0 | (g0 << 5) | (r0 << 11);
>+    uint8_t b0 = clamp255(src_argb[0] + dither0) >> 3;
>+    uint8_t g0 = clamp255(src_argb[1] + dither0) >> 2;
>+    uint8_t r0 = clamp255(src_argb[2] + dither0) >> 3;
>+    *(uint16_t*)(dst_rgb) = b0 | (g0 << 5) | (r0 << 11);
>   }
> }
> 
>-void ARGBToARGB1555Row_C(const uint8* src_argb, uint8* dst_rgb, int width) {
>+void ARGBToARGB1555Row_C(const uint8_t* src_argb, uint8_t* dst_rgb, int width) {
>   int x;
>   for (x = 0; x < width - 1; x += 2) {
>-    uint8 b0 = src_argb[0] >> 3;
>-    uint8 g0 = src_argb[1] >> 3;
>-    uint8 r0 = src_argb[2] >> 3;
>-    uint8 a0 = src_argb[3] >> 7;
>-    uint8 b1 = src_argb[4] >> 3;
>-    uint8 g1 = src_argb[5] >> 3;
>-    uint8 r1 = src_argb[6] >> 3;
>-    uint8 a1 = src_argb[7] >> 7;
>-    *(uint32*)(dst_rgb) = b0 | (g0 << 5) | (r0 << 10) | (a0 << 15) |
>-                          (b1 << 16) | (g1 << 21) | (r1 << 26) | (a1 << 31);
>+    uint8_t b0 = src_argb[0] >> 3;
>+    uint8_t g0 = src_argb[1] >> 3;
>+    uint8_t r0 = src_argb[2] >> 3;
>+    uint8_t a0 = src_argb[3] >> 7;
>+    uint8_t b1 = src_argb[4] >> 3;
>+    uint8_t g1 = src_argb[5] >> 3;
>+    uint8_t r1 = src_argb[6] >> 3;
>+    uint8_t a1 = src_argb[7] >> 7;
>+    *(uint32_t*)(dst_rgb) = b0 | (g0 << 5) | (r0 << 10) | (a0 << 15) |
>+                            (b1 << 16) | (g1 << 21) | (r1 << 26) | (a1 << 31);
>     dst_rgb += 4;
>     src_argb += 8;
>   }
>   if (width & 1) {
>-    uint8 b0 = src_argb[0] >> 3;
>-    uint8 g0 = src_argb[1] >> 3;
>-    uint8 r0 = src_argb[2] >> 3;
>-    uint8 a0 = src_argb[3] >> 7;
>-    *(uint16*)(dst_rgb) = b0 | (g0 << 5) | (r0 << 10) | (a0 << 15);
>+    uint8_t b0 = src_argb[0] >> 3;
>+    uint8_t g0 = src_argb[1] >> 3;
>+    uint8_t r0 = src_argb[2] >> 3;
>+    uint8_t a0 = src_argb[3] >> 7;
>+    *(uint16_t*)(dst_rgb) = b0 | (g0 << 5) | (r0 << 10) | (a0 << 15);
>   }
> }
> 
>-void ARGBToARGB4444Row_C(const uint8* src_argb, uint8* dst_rgb, int width) {
>+void ARGBToARGB4444Row_C(const uint8_t* src_argb, uint8_t* dst_rgb, int width) {
>   int x;
>   for (x = 0; x < width - 1; x += 2) {
>-    uint8 b0 = src_argb[0] >> 4;
>-    uint8 g0 = src_argb[1] >> 4;
>-    uint8 r0 = src_argb[2] >> 4;
>-    uint8 a0 = src_argb[3] >> 4;
>-    uint8 b1 = src_argb[4] >> 4;
>-    uint8 g1 = src_argb[5] >> 4;
>-    uint8 r1 = src_argb[6] >> 4;
>-    uint8 a1 = src_argb[7] >> 4;
>-    *(uint32*)(dst_rgb) = b0 | (g0 << 4) | (r0 << 8) | (a0 << 12) | (b1 << 16) |
>-                          (g1 << 20) | (r1 << 24) | (a1 << 28);
>+    uint8_t b0 = src_argb[0] >> 4;
>+    uint8_t g0 = src_argb[1] >> 4;
>+    uint8_t r0 = src_argb[2] >> 4;
>+    uint8_t a0 = src_argb[3] >> 4;
>+    uint8_t b1 = src_argb[4] >> 4;
>+    uint8_t g1 = src_argb[5] >> 4;
>+    uint8_t r1 = src_argb[6] >> 4;
>+    uint8_t a1 = src_argb[7] >> 4;
>+    *(uint32_t*)(dst_rgb) = b0 | (g0 << 4) | (r0 << 8) | (a0 << 12) |
>+                            (b1 << 16) | (g1 << 20) | (r1 << 24) | (a1 << 28);
>     dst_rgb += 4;
>     src_argb += 8;
>   }
>   if (width & 1) {
>-    uint8 b0 = src_argb[0] >> 4;
>-    uint8 g0 = src_argb[1] >> 4;
>-    uint8 r0 = src_argb[2] >> 4;
>-    uint8 a0 = src_argb[3] >> 4;
>-    *(uint16*)(dst_rgb) = b0 | (g0 << 4) | (r0 << 8) | (a0 << 12);
>+    uint8_t b0 = src_argb[0] >> 4;
>+    uint8_t g0 = src_argb[1] >> 4;
>+    uint8_t r0 = src_argb[2] >> 4;
>+    uint8_t a0 = src_argb[3] >> 4;
>+    *(uint16_t*)(dst_rgb) = b0 | (g0 << 4) | (r0 << 8) | (a0 << 12);
>+  }
>+}
>+
>+void ABGRToAR30Row_C(const uint8_t* src_abgr, uint8_t* dst_ar30, int width) {
>+  int x;
>+  for (x = 0; x < width; ++x) {
>+    uint32_t b0 = (src_abgr[0] >> 6) | ((uint32_t)(src_abgr[0]) << 2);
>+    uint32_t g0 = (src_abgr[1] >> 6) | ((uint32_t)(src_abgr[1]) << 2);
>+    uint32_t r0 = (src_abgr[2] >> 6) | ((uint32_t)(src_abgr[2]) << 2);
>+    uint32_t a0 = (src_abgr[3] >> 6);
>+    *(uint32_t*)(dst_ar30) = r0 | (g0 << 10) | (b0 << 20) | (a0 << 30);
>+    dst_ar30 += 4;
>+    src_abgr += 4;
>   }
> }
> 
>-static __inline int RGBToY(uint8 r, uint8 g, uint8 b) {
>+void ARGBToAR30Row_C(const uint8_t* src_argb, uint8_t* dst_ar30, int width) {
>+  int x;
>+  for (x = 0; x < width; ++x) {
>+    uint32_t b0 = (src_argb[0] >> 6) | ((uint32_t)(src_argb[0]) << 2);
>+    uint32_t g0 = (src_argb[1] >> 6) | ((uint32_t)(src_argb[1]) << 2);
>+    uint32_t r0 = (src_argb[2] >> 6) | ((uint32_t)(src_argb[2]) << 2);
>+    uint32_t a0 = (src_argb[3] >> 6);
>+    *(uint32_t*)(dst_ar30) = b0 | (g0 << 10) | (r0 << 20) | (a0 << 30);
>+    dst_ar30 += 4;
>+    src_argb += 4;
>+  }
>+}
>+
>+static __inline int RGBToY(uint8_t r, uint8_t g, uint8_t b) {
>   return (66 * r + 129 * g + 25 * b + 0x1080) >> 8;
> }
> 
>-static __inline int RGBToU(uint8 r, uint8 g, uint8 b) {
>+static __inline int RGBToU(uint8_t r, uint8_t g, uint8_t b) {
>   return (112 * b - 74 * g - 38 * r + 0x8080) >> 8;
> }
>-static __inline int RGBToV(uint8 r, uint8 g, uint8 b) {
>+static __inline int RGBToV(uint8_t r, uint8_t g, uint8_t b) {
>   return (112 * r - 94 * g - 18 * b + 0x8080) >> 8;
> }
> 
> // ARGBToY_C and ARGBToUV_C
>-#define MAKEROWY(NAME, R, G, B, BPP)                                     \
>-  void NAME##ToYRow_C(const uint8* src_argb0, uint8* dst_y, int width) { \
>-    int x;                                                               \
>-    for (x = 0; x < width; ++x) {                                        \
>-      dst_y[0] = RGBToY(src_argb0[R], src_argb0[G], src_argb0[B]);       \
>-      src_argb0 += BPP;                                                  \
>-      dst_y += 1;                                                        \
>-    }                                                                    \
>-  }                                                                      \
>-  void NAME##ToUVRow_C(const uint8* src_rgb0, int src_stride_rgb,        \
>-                       uint8* dst_u, uint8* dst_v, int width) {          \
>-    const uint8* src_rgb1 = src_rgb0 + src_stride_rgb;                   \
>-    int x;                                                               \
>-    for (x = 0; x < width - 1; x += 2) {                                 \
>-      uint8 ab = (src_rgb0[B] + src_rgb0[B + BPP] + src_rgb1[B] +        \
>-                  src_rgb1[B + BPP]) >>                                  \
>-                 2;                                                      \
>-      uint8 ag = (src_rgb0[G] + src_rgb0[G + BPP] + src_rgb1[G] +        \
>-                  src_rgb1[G + BPP]) >>                                  \
>-                 2;                                                      \
>-      uint8 ar = (src_rgb0[R] + src_rgb0[R + BPP] + src_rgb1[R] +        \
>-                  src_rgb1[R + BPP]) >>                                  \
>-                 2;                                                      \
>-      dst_u[0] = RGBToU(ar, ag, ab);                                     \
>-      dst_v[0] = RGBToV(ar, ag, ab);                                     \
>-      src_rgb0 += BPP * 2;                                               \
>-      src_rgb1 += BPP * 2;                                               \
>-      dst_u += 1;                                                        \
>-      dst_v += 1;                                                        \
>-    }                                                                    \
>-    if (width & 1) {                                                     \
>-      uint8 ab = (src_rgb0[B] + src_rgb1[B]) >> 1;                       \
>-      uint8 ag = (src_rgb0[G] + src_rgb1[G]) >> 1;                       \
>-      uint8 ar = (src_rgb0[R] + src_rgb1[R]) >> 1;                       \
>-      dst_u[0] = RGBToU(ar, ag, ab);                                     \
>-      dst_v[0] = RGBToV(ar, ag, ab);                                     \
>-    }                                                                    \
>+#define MAKEROWY(NAME, R, G, B, BPP)                                         \
>+  void NAME##ToYRow_C(const uint8_t* src_argb0, uint8_t* dst_y, int width) { \
>+    int x;                                                                   \
>+    for (x = 0; x < width; ++x) {                                            \
>+      dst_y[0] = RGBToY(src_argb0[R], src_argb0[G], src_argb0[B]);           \
>+      src_argb0 += BPP;                                                      \
>+      dst_y += 1;                                                            \
>+    }                                                                        \
>+  }                                                                          \
>+  void NAME##ToUVRow_C(const uint8_t* src_rgb0, int src_stride_rgb,          \
>+                       uint8_t* dst_u, uint8_t* dst_v, int width) {          \
>+    const uint8_t* src_rgb1 = src_rgb0 + src_stride_rgb;                     \
>+    int x;                                                                   \
>+    for (x = 0; x < width - 1; x += 2) {                                     \
>+      uint8_t ab = (src_rgb0[B] + src_rgb0[B + BPP] + src_rgb1[B] +          \
>+                    src_rgb1[B + BPP]) >>                                    \
>+                   2;                                                        \
>+      uint8_t ag = (src_rgb0[G] + src_rgb0[G + BPP] + src_rgb1[G] +          \
>+                    src_rgb1[G + BPP]) >>                                    \
>+                   2;                                                        \
>+      uint8_t ar = (src_rgb0[R] + src_rgb0[R + BPP] + src_rgb1[R] +          \
>+                    src_rgb1[R + BPP]) >>                                    \
>+                   2;                                                        \
>+      dst_u[0] = RGBToU(ar, ag, ab);                                         \
>+      dst_v[0] = RGBToV(ar, ag, ab);                                         \
>+      src_rgb0 += BPP * 2;                                                   \
>+      src_rgb1 += BPP * 2;                                                   \
>+      dst_u += 1;                                                            \
>+      dst_v += 1;                                                            \
>+    }                                                                        \
>+    if (width & 1) {                                                         \
>+      uint8_t ab = (src_rgb0[B] + src_rgb1[B]) >> 1;                         \
>+      uint8_t ag = (src_rgb0[G] + src_rgb1[G]) >> 1;                         \
>+      uint8_t ar = (src_rgb0[R] + src_rgb1[R]) >> 1;                         \
>+      dst_u[0] = RGBToU(ar, ag, ab);                                         \
>+      dst_v[0] = RGBToV(ar, ag, ab);                                         \
>+    }                                                                        \
>   }
> 
> MAKEROWY(ARGB, 2, 1, 0, 4)
>@@ -385,65 +465,65 @@ MAKEROWY(RAW, 0, 1, 2, 3)
> // g -0.41869 * 255 = -106.76595 = -107
> // r  0.50000 * 255 = 127.5 = 127
> 
>-static __inline int RGBToYJ(uint8 r, uint8 g, uint8 b) {
>+static __inline int RGBToYJ(uint8_t r, uint8_t g, uint8_t b) {
>   return (38 * r + 75 * g + 15 * b + 64) >> 7;
> }
> 
>-static __inline int RGBToUJ(uint8 r, uint8 g, uint8 b) {
>+static __inline int RGBToUJ(uint8_t r, uint8_t g, uint8_t b) {
>   return (127 * b - 84 * g - 43 * r + 0x8080) >> 8;
> }
>-static __inline int RGBToVJ(uint8 r, uint8 g, uint8 b) {
>+static __inline int RGBToVJ(uint8_t r, uint8_t g, uint8_t b) {
>   return (127 * r - 107 * g - 20 * b + 0x8080) >> 8;
> }
> 
> #define AVGB(a, b) (((a) + (b) + 1) >> 1)
> 
> // ARGBToYJ_C and ARGBToUVJ_C
>-#define MAKEROWYJ(NAME, R, G, B, BPP)                                     \
>-  void NAME##ToYJRow_C(const uint8* src_argb0, uint8* dst_y, int width) { \
>-    int x;                                                                \
>-    for (x = 0; x < width; ++x) {                                         \
>-      dst_y[0] = RGBToYJ(src_argb0[R], src_argb0[G], src_argb0[B]);       \
>-      src_argb0 += BPP;                                                   \
>-      dst_y += 1;                                                         \
>-    }                                                                     \
>-  }                                                                       \
>-  void NAME##ToUVJRow_C(const uint8* src_rgb0, int src_stride_rgb,        \
>-                        uint8* dst_u, uint8* dst_v, int width) {          \
>-    const uint8* src_rgb1 = src_rgb0 + src_stride_rgb;                    \
>-    int x;                                                                \
>-    for (x = 0; x < width - 1; x += 2) {                                  \
>-      uint8 ab = AVGB(AVGB(src_rgb0[B], src_rgb1[B]),                     \
>-                      AVGB(src_rgb0[B + BPP], src_rgb1[B + BPP]));        \
>-      uint8 ag = AVGB(AVGB(src_rgb0[G], src_rgb1[G]),                     \
>-                      AVGB(src_rgb0[G + BPP], src_rgb1[G + BPP]));        \
>-      uint8 ar = AVGB(AVGB(src_rgb0[R], src_rgb1[R]),                     \
>-                      AVGB(src_rgb0[R + BPP], src_rgb1[R + BPP]));        \
>-      dst_u[0] = RGBToUJ(ar, ag, ab);                                     \
>-      dst_v[0] = RGBToVJ(ar, ag, ab);                                     \
>-      src_rgb0 += BPP * 2;                                                \
>-      src_rgb1 += BPP * 2;                                                \
>-      dst_u += 1;                                                         \
>-      dst_v += 1;                                                         \
>-    }                                                                     \
>-    if (width & 1) {                                                      \
>-      uint8 ab = AVGB(src_rgb0[B], src_rgb1[B]);                          \
>-      uint8 ag = AVGB(src_rgb0[G], src_rgb1[G]);                          \
>-      uint8 ar = AVGB(src_rgb0[R], src_rgb1[R]);                          \
>-      dst_u[0] = RGBToUJ(ar, ag, ab);                                     \
>-      dst_v[0] = RGBToVJ(ar, ag, ab);                                     \
>-    }                                                                     \
>+#define MAKEROWYJ(NAME, R, G, B, BPP)                                         \
>+  void NAME##ToYJRow_C(const uint8_t* src_argb0, uint8_t* dst_y, int width) { \
>+    int x;                                                                    \
>+    for (x = 0; x < width; ++x) {                                             \
>+      dst_y[0] = RGBToYJ(src_argb0[R], src_argb0[G], src_argb0[B]);           \
>+      src_argb0 += BPP;                                                       \
>+      dst_y += 1;                                                             \
>+    }                                                                         \
>+  }                                                                           \
>+  void NAME##ToUVJRow_C(const uint8_t* src_rgb0, int src_stride_rgb,          \
>+                        uint8_t* dst_u, uint8_t* dst_v, int width) {          \
>+    const uint8_t* src_rgb1 = src_rgb0 + src_stride_rgb;                      \
>+    int x;                                                                    \
>+    for (x = 0; x < width - 1; x += 2) {                                      \
>+      uint8_t ab = AVGB(AVGB(src_rgb0[B], src_rgb1[B]),                       \
>+                        AVGB(src_rgb0[B + BPP], src_rgb1[B + BPP]));          \
>+      uint8_t ag = AVGB(AVGB(src_rgb0[G], src_rgb1[G]),                       \
>+                        AVGB(src_rgb0[G + BPP], src_rgb1[G + BPP]));          \
>+      uint8_t ar = AVGB(AVGB(src_rgb0[R], src_rgb1[R]),                       \
>+                        AVGB(src_rgb0[R + BPP], src_rgb1[R + BPP]));          \
>+      dst_u[0] = RGBToUJ(ar, ag, ab);                                         \
>+      dst_v[0] = RGBToVJ(ar, ag, ab);                                         \
>+      src_rgb0 += BPP * 2;                                                    \
>+      src_rgb1 += BPP * 2;                                                    \
>+      dst_u += 1;                                                             \
>+      dst_v += 1;                                                             \
>+    }                                                                         \
>+    if (width & 1) {                                                          \
>+      uint8_t ab = AVGB(src_rgb0[B], src_rgb1[B]);                            \
>+      uint8_t ag = AVGB(src_rgb0[G], src_rgb1[G]);                            \
>+      uint8_t ar = AVGB(src_rgb0[R], src_rgb1[R]);                            \
>+      dst_u[0] = RGBToUJ(ar, ag, ab);                                         \
>+      dst_v[0] = RGBToVJ(ar, ag, ab);                                         \
>+    }                                                                         \
>   }
> 
> MAKEROWYJ(ARGB, 2, 1, 0, 4)
> #undef MAKEROWYJ
> 
>-void RGB565ToYRow_C(const uint8* src_rgb565, uint8* dst_y, int width) {
>+void RGB565ToYRow_C(const uint8_t* src_rgb565, uint8_t* dst_y, int width) {
>   int x;
>   for (x = 0; x < width; ++x) {
>-    uint8 b = src_rgb565[0] & 0x1f;
>-    uint8 g = (src_rgb565[0] >> 5) | ((src_rgb565[1] & 0x07) << 3);
>-    uint8 r = src_rgb565[1] >> 3;
>+    uint8_t b = src_rgb565[0] & 0x1f;
>+    uint8_t g = (src_rgb565[0] >> 5) | ((src_rgb565[1] & 0x07) << 3);
>+    uint8_t r = src_rgb565[1] >> 3;
>     b = (b << 3) | (b >> 2);
>     g = (g << 2) | (g >> 4);
>     r = (r << 3) | (r >> 2);
>@@ -453,12 +533,12 @@ void RGB565ToYRow_C(const uint8* src_rgb565, uint8* dst_y, int width) {
>   }
> }
> 
>-void ARGB1555ToYRow_C(const uint8* src_argb1555, uint8* dst_y, int width) {
>+void ARGB1555ToYRow_C(const uint8_t* src_argb1555, uint8_t* dst_y, int width) {
>   int x;
>   for (x = 0; x < width; ++x) {
>-    uint8 b = src_argb1555[0] & 0x1f;
>-    uint8 g = (src_argb1555[0] >> 5) | ((src_argb1555[1] & 0x03) << 3);
>-    uint8 r = (src_argb1555[1] & 0x7c) >> 2;
>+    uint8_t b = src_argb1555[0] & 0x1f;
>+    uint8_t g = (src_argb1555[0] >> 5) | ((src_argb1555[1] & 0x03) << 3);
>+    uint8_t r = (src_argb1555[1] & 0x7c) >> 2;
>     b = (b << 3) | (b >> 2);
>     g = (g << 3) | (g >> 2);
>     r = (r << 3) | (r >> 2);
>@@ -468,12 +548,12 @@ void ARGB1555ToYRow_C(const uint8* src_argb1555, uint8* dst_y, int width) {
>   }
> }
> 
>-void ARGB4444ToYRow_C(const uint8* src_argb4444, uint8* dst_y, int width) {
>+void ARGB4444ToYRow_C(const uint8_t* src_argb4444, uint8_t* dst_y, int width) {
>   int x;
>   for (x = 0; x < width; ++x) {
>-    uint8 b = src_argb4444[0] & 0x0f;
>-    uint8 g = src_argb4444[0] >> 4;
>-    uint8 r = src_argb4444[1] & 0x0f;
>+    uint8_t b = src_argb4444[0] & 0x0f;
>+    uint8_t g = src_argb4444[0] >> 4;
>+    uint8_t r = src_argb4444[1] & 0x0f;
>     b = (b << 4) | b;
>     g = (g << 4) | g;
>     r = (r << 4) | r;
>@@ -483,29 +563,29 @@ void ARGB4444ToYRow_C(const uint8* src_argb4444, uint8* dst_y, int width) {
>   }
> }
> 
>-void RGB565ToUVRow_C(const uint8* src_rgb565,
>+void RGB565ToUVRow_C(const uint8_t* src_rgb565,
>                      int src_stride_rgb565,
>-                     uint8* dst_u,
>-                     uint8* dst_v,
>+                     uint8_t* dst_u,
>+                     uint8_t* dst_v,
>                      int width) {
>-  const uint8* next_rgb565 = src_rgb565 + src_stride_rgb565;
>+  const uint8_t* next_rgb565 = src_rgb565 + src_stride_rgb565;
>   int x;
>   for (x = 0; x < width - 1; x += 2) {
>-    uint8 b0 = src_rgb565[0] & 0x1f;
>-    uint8 g0 = (src_rgb565[0] >> 5) | ((src_rgb565[1] & 0x07) << 3);
>-    uint8 r0 = src_rgb565[1] >> 3;
>-    uint8 b1 = src_rgb565[2] & 0x1f;
>-    uint8 g1 = (src_rgb565[2] >> 5) | ((src_rgb565[3] & 0x07) << 3);
>-    uint8 r1 = src_rgb565[3] >> 3;
>-    uint8 b2 = next_rgb565[0] & 0x1f;
>-    uint8 g2 = (next_rgb565[0] >> 5) | ((next_rgb565[1] & 0x07) << 3);
>-    uint8 r2 = next_rgb565[1] >> 3;
>-    uint8 b3 = next_rgb565[2] & 0x1f;
>-    uint8 g3 = (next_rgb565[2] >> 5) | ((next_rgb565[3] & 0x07) << 3);
>-    uint8 r3 = next_rgb565[3] >> 3;
>-    uint8 b = (b0 + b1 + b2 + b3);  // 565 * 4 = 787.
>-    uint8 g = (g0 + g1 + g2 + g3);
>-    uint8 r = (r0 + r1 + r2 + r3);
>+    uint8_t b0 = src_rgb565[0] & 0x1f;
>+    uint8_t g0 = (src_rgb565[0] >> 5) | ((src_rgb565[1] & 0x07) << 3);
>+    uint8_t r0 = src_rgb565[1] >> 3;
>+    uint8_t b1 = src_rgb565[2] & 0x1f;
>+    uint8_t g1 = (src_rgb565[2] >> 5) | ((src_rgb565[3] & 0x07) << 3);
>+    uint8_t r1 = src_rgb565[3] >> 3;
>+    uint8_t b2 = next_rgb565[0] & 0x1f;
>+    uint8_t g2 = (next_rgb565[0] >> 5) | ((next_rgb565[1] & 0x07) << 3);
>+    uint8_t r2 = next_rgb565[1] >> 3;
>+    uint8_t b3 = next_rgb565[2] & 0x1f;
>+    uint8_t g3 = (next_rgb565[2] >> 5) | ((next_rgb565[3] & 0x07) << 3);
>+    uint8_t r3 = next_rgb565[3] >> 3;
>+    uint8_t b = (b0 + b1 + b2 + b3);  // 565 * 4 = 787.
>+    uint8_t g = (g0 + g1 + g2 + g3);
>+    uint8_t r = (r0 + r1 + r2 + r3);
>     b = (b << 1) | (b >> 6);  // 787 -> 888.
>     r = (r << 1) | (r >> 6);
>     dst_u[0] = RGBToU(r, g, b);
>@@ -516,15 +596,15 @@ void RGB565ToUVRow_C(const uint8* src_rgb565,
>     dst_v += 1;
>   }
>   if (width & 1) {
>-    uint8 b0 = src_rgb565[0] & 0x1f;
>-    uint8 g0 = (src_rgb565[0] >> 5) | ((src_rgb565[1] & 0x07) << 3);
>-    uint8 r0 = src_rgb565[1] >> 3;
>-    uint8 b2 = next_rgb565[0] & 0x1f;
>-    uint8 g2 = (next_rgb565[0] >> 5) | ((next_rgb565[1] & 0x07) << 3);
>-    uint8 r2 = next_rgb565[1] >> 3;
>-    uint8 b = (b0 + b2);  // 565 * 2 = 676.
>-    uint8 g = (g0 + g2);
>-    uint8 r = (r0 + r2);
>+    uint8_t b0 = src_rgb565[0] & 0x1f;
>+    uint8_t g0 = (src_rgb565[0] >> 5) | ((src_rgb565[1] & 0x07) << 3);
>+    uint8_t r0 = src_rgb565[1] >> 3;
>+    uint8_t b2 = next_rgb565[0] & 0x1f;
>+    uint8_t g2 = (next_rgb565[0] >> 5) | ((next_rgb565[1] & 0x07) << 3);
>+    uint8_t r2 = next_rgb565[1] >> 3;
>+    uint8_t b = (b0 + b2);  // 565 * 2 = 676.
>+    uint8_t g = (g0 + g2);
>+    uint8_t r = (r0 + r2);
>     b = (b << 2) | (b >> 4);  // 676 -> 888
>     g = (g << 1) | (g >> 6);
>     r = (r << 2) | (r >> 4);
>@@ -533,29 +613,29 @@ void RGB565ToUVRow_C(const uint8* src_rgb565,
>   }
> }
> 
>-void ARGB1555ToUVRow_C(const uint8* src_argb1555,
>+void ARGB1555ToUVRow_C(const uint8_t* src_argb1555,
>                        int src_stride_argb1555,
>-                       uint8* dst_u,
>-                       uint8* dst_v,
>+                       uint8_t* dst_u,
>+                       uint8_t* dst_v,
>                        int width) {
>-  const uint8* next_argb1555 = src_argb1555 + src_stride_argb1555;
>+  const uint8_t* next_argb1555 = src_argb1555 + src_stride_argb1555;
>   int x;
>   for (x = 0; x < width - 1; x += 2) {
>-    uint8 b0 = src_argb1555[0] & 0x1f;
>-    uint8 g0 = (src_argb1555[0] >> 5) | ((src_argb1555[1] & 0x03) << 3);
>-    uint8 r0 = (src_argb1555[1] & 0x7c) >> 2;
>-    uint8 b1 = src_argb1555[2] & 0x1f;
>-    uint8 g1 = (src_argb1555[2] >> 5) | ((src_argb1555[3] & 0x03) << 3);
>-    uint8 r1 = (src_argb1555[3] & 0x7c) >> 2;
>-    uint8 b2 = next_argb1555[0] & 0x1f;
>-    uint8 g2 = (next_argb1555[0] >> 5) | ((next_argb1555[1] & 0x03) << 3);
>-    uint8 r2 = (next_argb1555[1] & 0x7c) >> 2;
>-    uint8 b3 = next_argb1555[2] & 0x1f;
>-    uint8 g3 = (next_argb1555[2] >> 5) | ((next_argb1555[3] & 0x03) << 3);
>-    uint8 r3 = (next_argb1555[3] & 0x7c) >> 2;
>-    uint8 b = (b0 + b1 + b2 + b3);  // 555 * 4 = 777.
>-    uint8 g = (g0 + g1 + g2 + g3);
>-    uint8 r = (r0 + r1 + r2 + r3);
>+    uint8_t b0 = src_argb1555[0] & 0x1f;
>+    uint8_t g0 = (src_argb1555[0] >> 5) | ((src_argb1555[1] & 0x03) << 3);
>+    uint8_t r0 = (src_argb1555[1] & 0x7c) >> 2;
>+    uint8_t b1 = src_argb1555[2] & 0x1f;
>+    uint8_t g1 = (src_argb1555[2] >> 5) | ((src_argb1555[3] & 0x03) << 3);
>+    uint8_t r1 = (src_argb1555[3] & 0x7c) >> 2;
>+    uint8_t b2 = next_argb1555[0] & 0x1f;
>+    uint8_t g2 = (next_argb1555[0] >> 5) | ((next_argb1555[1] & 0x03) << 3);
>+    uint8_t r2 = (next_argb1555[1] & 0x7c) >> 2;
>+    uint8_t b3 = next_argb1555[2] & 0x1f;
>+    uint8_t g3 = (next_argb1555[2] >> 5) | ((next_argb1555[3] & 0x03) << 3);
>+    uint8_t r3 = (next_argb1555[3] & 0x7c) >> 2;
>+    uint8_t b = (b0 + b1 + b2 + b3);  // 555 * 4 = 777.
>+    uint8_t g = (g0 + g1 + g2 + g3);
>+    uint8_t r = (r0 + r1 + r2 + r3);
>     b = (b << 1) | (b >> 6);  // 777 -> 888.
>     g = (g << 1) | (g >> 6);
>     r = (r << 1) | (r >> 6);
>@@ -567,15 +647,15 @@ void ARGB1555ToUVRow_C(const uint8* src_argb1555,
>     dst_v += 1;
>   }
>   if (width & 1) {
>-    uint8 b0 = src_argb1555[0] & 0x1f;
>-    uint8 g0 = (src_argb1555[0] >> 5) | ((src_argb1555[1] & 0x03) << 3);
>-    uint8 r0 = (src_argb1555[1] & 0x7c) >> 2;
>-    uint8 b2 = next_argb1555[0] & 0x1f;
>-    uint8 g2 = (next_argb1555[0] >> 5) | ((next_argb1555[1] & 0x03) << 3);
>-    uint8 r2 = next_argb1555[1] >> 3;
>-    uint8 b = (b0 + b2);  // 555 * 2 = 666.
>-    uint8 g = (g0 + g2);
>-    uint8 r = (r0 + r2);
>+    uint8_t b0 = src_argb1555[0] & 0x1f;
>+    uint8_t g0 = (src_argb1555[0] >> 5) | ((src_argb1555[1] & 0x03) << 3);
>+    uint8_t r0 = (src_argb1555[1] & 0x7c) >> 2;
>+    uint8_t b2 = next_argb1555[0] & 0x1f;
>+    uint8_t g2 = (next_argb1555[0] >> 5) | ((next_argb1555[1] & 0x03) << 3);
>+    uint8_t r2 = next_argb1555[1] >> 3;
>+    uint8_t b = (b0 + b2);  // 555 * 2 = 666.
>+    uint8_t g = (g0 + g2);
>+    uint8_t r = (r0 + r2);
>     b = (b << 2) | (b >> 4);  // 666 -> 888.
>     g = (g << 2) | (g >> 4);
>     r = (r << 2) | (r >> 4);
>@@ -584,29 +664,29 @@ void ARGB1555ToUVRow_C(const uint8* src_argb1555,
>   }
> }
> 
>-void ARGB4444ToUVRow_C(const uint8* src_argb4444,
>+void ARGB4444ToUVRow_C(const uint8_t* src_argb4444,
>                        int src_stride_argb4444,
>-                       uint8* dst_u,
>-                       uint8* dst_v,
>+                       uint8_t* dst_u,
>+                       uint8_t* dst_v,
>                        int width) {
>-  const uint8* next_argb4444 = src_argb4444 + src_stride_argb4444;
>+  const uint8_t* next_argb4444 = src_argb4444 + src_stride_argb4444;
>   int x;
>   for (x = 0; x < width - 1; x += 2) {
>-    uint8 b0 = src_argb4444[0] & 0x0f;
>-    uint8 g0 = src_argb4444[0] >> 4;
>-    uint8 r0 = src_argb4444[1] & 0x0f;
>-    uint8 b1 = src_argb4444[2] & 0x0f;
>-    uint8 g1 = src_argb4444[2] >> 4;
>-    uint8 r1 = src_argb4444[3] & 0x0f;
>-    uint8 b2 = next_argb4444[0] & 0x0f;
>-    uint8 g2 = next_argb4444[0] >> 4;
>-    uint8 r2 = next_argb4444[1] & 0x0f;
>-    uint8 b3 = next_argb4444[2] & 0x0f;
>-    uint8 g3 = next_argb4444[2] >> 4;
>-    uint8 r3 = next_argb4444[3] & 0x0f;
>-    uint8 b = (b0 + b1 + b2 + b3);  // 444 * 4 = 666.
>-    uint8 g = (g0 + g1 + g2 + g3);
>-    uint8 r = (r0 + r1 + r2 + r3);
>+    uint8_t b0 = src_argb4444[0] & 0x0f;
>+    uint8_t g0 = src_argb4444[0] >> 4;
>+    uint8_t r0 = src_argb4444[1] & 0x0f;
>+    uint8_t b1 = src_argb4444[2] & 0x0f;
>+    uint8_t g1 = src_argb4444[2] >> 4;
>+    uint8_t r1 = src_argb4444[3] & 0x0f;
>+    uint8_t b2 = next_argb4444[0] & 0x0f;
>+    uint8_t g2 = next_argb4444[0] >> 4;
>+    uint8_t r2 = next_argb4444[1] & 0x0f;
>+    uint8_t b3 = next_argb4444[2] & 0x0f;
>+    uint8_t g3 = next_argb4444[2] >> 4;
>+    uint8_t r3 = next_argb4444[3] & 0x0f;
>+    uint8_t b = (b0 + b1 + b2 + b3);  // 444 * 4 = 666.
>+    uint8_t g = (g0 + g1 + g2 + g3);
>+    uint8_t r = (r0 + r1 + r2 + r3);
>     b = (b << 2) | (b >> 4);  // 666 -> 888.
>     g = (g << 2) | (g >> 4);
>     r = (r << 2) | (r >> 4);
>@@ -618,15 +698,15 @@ void ARGB4444ToUVRow_C(const uint8* src_argb4444,
>     dst_v += 1;
>   }
>   if (width & 1) {
>-    uint8 b0 = src_argb4444[0] & 0x0f;
>-    uint8 g0 = src_argb4444[0] >> 4;
>-    uint8 r0 = src_argb4444[1] & 0x0f;
>-    uint8 b2 = next_argb4444[0] & 0x0f;
>-    uint8 g2 = next_argb4444[0] >> 4;
>-    uint8 r2 = next_argb4444[1] & 0x0f;
>-    uint8 b = (b0 + b2);  // 444 * 2 = 555.
>-    uint8 g = (g0 + g2);
>-    uint8 r = (r0 + r2);
>+    uint8_t b0 = src_argb4444[0] & 0x0f;
>+    uint8_t g0 = src_argb4444[0] >> 4;
>+    uint8_t r0 = src_argb4444[1] & 0x0f;
>+    uint8_t b2 = next_argb4444[0] & 0x0f;
>+    uint8_t g2 = next_argb4444[0] >> 4;
>+    uint8_t r2 = next_argb4444[1] & 0x0f;
>+    uint8_t b = (b0 + b2);  // 444 * 2 = 555.
>+    uint8_t g = (g0 + g2);
>+    uint8_t r = (r0 + r2);
>     b = (b << 3) | (b >> 2);  // 555 -> 888.
>     g = (g << 3) | (g >> 2);
>     r = (r << 3) | (r >> 2);
>@@ -635,15 +715,15 @@ void ARGB4444ToUVRow_C(const uint8* src_argb4444,
>   }
> }
> 
>-void ARGBToUV444Row_C(const uint8* src_argb,
>-                      uint8* dst_u,
>-                      uint8* dst_v,
>+void ARGBToUV444Row_C(const uint8_t* src_argb,
>+                      uint8_t* dst_u,
>+                      uint8_t* dst_v,
>                       int width) {
>   int x;
>   for (x = 0; x < width; ++x) {
>-    uint8 ab = src_argb[0];
>-    uint8 ag = src_argb[1];
>-    uint8 ar = src_argb[2];
>+    uint8_t ab = src_argb[0];
>+    uint8_t ag = src_argb[1];
>+    uint8_t ar = src_argb[2];
>     dst_u[0] = RGBToU(ar, ag, ab);
>     dst_v[0] = RGBToV(ar, ag, ab);
>     src_argb += 4;
>@@ -652,10 +732,10 @@ void ARGBToUV444Row_C(const uint8* src_argb,
>   }
> }
> 
>-void ARGBGrayRow_C(const uint8* src_argb, uint8* dst_argb, int width) {
>+void ARGBGrayRow_C(const uint8_t* src_argb, uint8_t* dst_argb, int width) {
>   int x;
>   for (x = 0; x < width; ++x) {
>-    uint8 y = RGBToYJ(src_argb[2], src_argb[1], src_argb[0]);
>+    uint8_t y = RGBToYJ(src_argb[2], src_argb[1], src_argb[0]);
>     dst_argb[2] = dst_argb[1] = dst_argb[0] = y;
>     dst_argb[3] = src_argb[3];
>     dst_argb += 4;
>@@ -664,7 +744,7 @@ void ARGBGrayRow_C(const uint8* src_argb, uint8* dst_argb, int width) {
> }
> 
> // Convert a row of image to Sepia tone.
>-void ARGBSepiaRow_C(uint8* dst_argb, int width) {
>+void ARGBSepiaRow_C(uint8_t* dst_argb, int width) {
>   int x;
>   for (x = 0; x < width; ++x) {
>     int b = dst_argb[0];
>@@ -683,9 +763,9 @@ void ARGBSepiaRow_C(uint8* dst_argb, int width) {
> 
> // Apply color matrix to a row of image. Matrix is signed.
> // TODO(fbarchard): Consider adding rounding (+32).
>-void ARGBColorMatrixRow_C(const uint8* src_argb,
>-                          uint8* dst_argb,
>-                          const int8* matrix_argb,
>+void ARGBColorMatrixRow_C(const uint8_t* src_argb,
>+                          uint8_t* dst_argb,
>+                          const int8_t* matrix_argb,
>                           int width) {
>   int x;
>   for (x = 0; x < width; ++x) {
>@@ -715,7 +795,9 @@ void ARGBColorMatrixRow_C(const uint8* src_argb,
> }
> 
> // Apply color table to a row of image.
>-void ARGBColorTableRow_C(uint8* dst_argb, const uint8* table_argb, int width) {
>+void ARGBColorTableRow_C(uint8_t* dst_argb,
>+                         const uint8_t* table_argb,
>+                         int width) {
>   int x;
>   for (x = 0; x < width; ++x) {
>     int b = dst_argb[0];
>@@ -731,7 +813,9 @@ void ARGBColorTableRow_C(uint8* dst_argb, const uint8* table_argb, int width) {
> }
> 
> // Apply color table to a row of image.
>-void RGBColorTableRow_C(uint8* dst_argb, const uint8* table_argb, int width) {
>+void RGBColorTableRow_C(uint8_t* dst_argb,
>+                        const uint8_t* table_argb,
>+                        int width) {
>   int x;
>   for (x = 0; x < width; ++x) {
>     int b = dst_argb[0];
>@@ -744,7 +828,7 @@ void RGBColorTableRow_C(uint8* dst_argb, const uint8* table_argb, int width) {
>   }
> }
> 
>-void ARGBQuantizeRow_C(uint8* dst_argb,
>+void ARGBQuantizeRow_C(uint8_t* dst_argb,
>                        int scale,
>                        int interval_size,
>                        int interval_offset,
>@@ -764,21 +848,21 @@ void ARGBQuantizeRow_C(uint8* dst_argb,
> #define REPEAT8(v) (v) | ((v) << 8)
> #define SHADE(f, v) v* f >> 24
> 
>-void ARGBShadeRow_C(const uint8* src_argb,
>-                    uint8* dst_argb,
>+void ARGBShadeRow_C(const uint8_t* src_argb,
>+                    uint8_t* dst_argb,
>                     int width,
>-                    uint32 value) {
>-  const uint32 b_scale = REPEAT8(value & 0xff);
>-  const uint32 g_scale = REPEAT8((value >> 8) & 0xff);
>-  const uint32 r_scale = REPEAT8((value >> 16) & 0xff);
>-  const uint32 a_scale = REPEAT8(value >> 24);
>+                    uint32_t value) {
>+  const uint32_t b_scale = REPEAT8(value & 0xff);
>+  const uint32_t g_scale = REPEAT8((value >> 8) & 0xff);
>+  const uint32_t r_scale = REPEAT8((value >> 16) & 0xff);
>+  const uint32_t a_scale = REPEAT8(value >> 24);
> 
>   int i;
>   for (i = 0; i < width; ++i) {
>-    const uint32 b = REPEAT8(src_argb[0]);
>-    const uint32 g = REPEAT8(src_argb[1]);
>-    const uint32 r = REPEAT8(src_argb[2]);
>-    const uint32 a = REPEAT8(src_argb[3]);
>+    const uint32_t b = REPEAT8(src_argb[0]);
>+    const uint32_t g = REPEAT8(src_argb[1]);
>+    const uint32_t r = REPEAT8(src_argb[2]);
>+    const uint32_t a = REPEAT8(src_argb[3]);
>     dst_argb[0] = SHADE(b, b_scale);
>     dst_argb[1] = SHADE(g, g_scale);
>     dst_argb[2] = SHADE(r, r_scale);
>@@ -793,20 +877,20 @@ void ARGBShadeRow_C(const uint8* src_argb,
> #define REPEAT8(v) (v) | ((v) << 8)
> #define SHADE(f, v) v* f >> 16
> 
>-void ARGBMultiplyRow_C(const uint8* src_argb0,
>-                       const uint8* src_argb1,
>-                       uint8* dst_argb,
>+void ARGBMultiplyRow_C(const uint8_t* src_argb0,
>+                       const uint8_t* src_argb1,
>+                       uint8_t* dst_argb,
>                        int width) {
>   int i;
>   for (i = 0; i < width; ++i) {
>-    const uint32 b = REPEAT8(src_argb0[0]);
>-    const uint32 g = REPEAT8(src_argb0[1]);
>-    const uint32 r = REPEAT8(src_argb0[2]);
>-    const uint32 a = REPEAT8(src_argb0[3]);
>-    const uint32 b_scale = src_argb1[0];
>-    const uint32 g_scale = src_argb1[1];
>-    const uint32 r_scale = src_argb1[2];
>-    const uint32 a_scale = src_argb1[3];
>+    const uint32_t b = REPEAT8(src_argb0[0]);
>+    const uint32_t g = REPEAT8(src_argb0[1]);
>+    const uint32_t r = REPEAT8(src_argb0[2]);
>+    const uint32_t a = REPEAT8(src_argb0[3]);
>+    const uint32_t b_scale = src_argb1[0];
>+    const uint32_t g_scale = src_argb1[1];
>+    const uint32_t r_scale = src_argb1[2];
>+    const uint32_t a_scale = src_argb1[3];
>     dst_argb[0] = SHADE(b, b_scale);
>     dst_argb[1] = SHADE(g, g_scale);
>     dst_argb[2] = SHADE(r, r_scale);
>@@ -821,9 +905,9 @@ void ARGBMultiplyRow_C(const uint8* src_argb0,
> 
> #define SHADE(f, v) clamp255(v + f)
> 
>-void ARGBAddRow_C(const uint8* src_argb0,
>-                  const uint8* src_argb1,
>-                  uint8* dst_argb,
>+void ARGBAddRow_C(const uint8_t* src_argb0,
>+                  const uint8_t* src_argb1,
>+                  uint8_t* dst_argb,
>                   int width) {
>   int i;
>   for (i = 0; i < width; ++i) {
>@@ -848,9 +932,9 @@ void ARGBAddRow_C(const uint8* src_argb0,
> 
> #define SHADE(f, v) clamp0(f - v)
> 
>-void ARGBSubtractRow_C(const uint8* src_argb0,
>-                       const uint8* src_argb1,
>-                       uint8* dst_argb,
>+void ARGBSubtractRow_C(const uint8_t* src_argb0,
>+                       const uint8_t* src_argb1,
>+                       uint8_t* dst_argb,
>                        int width) {
>   int i;
>   for (i = 0; i < width; ++i) {
>@@ -874,10 +958,10 @@ void ARGBSubtractRow_C(const uint8* src_argb0,
> #undef SHADE
> 
> // Sobel functions which mimics SSSE3.
>-void SobelXRow_C(const uint8* src_y0,
>-                 const uint8* src_y1,
>-                 const uint8* src_y2,
>-                 uint8* dst_sobelx,
>+void SobelXRow_C(const uint8_t* src_y0,
>+                 const uint8_t* src_y1,
>+                 const uint8_t* src_y2,
>+                 uint8_t* dst_sobelx,
>                  int width) {
>   int i;
>   for (i = 0; i < width; ++i) {
>@@ -891,13 +975,13 @@ void SobelXRow_C(const uint8* src_y0,
>     int b_diff = b - b_sub;
>     int c_diff = c - c_sub;
>     int sobel = Abs(a_diff + b_diff * 2 + c_diff);
>-    dst_sobelx[i] = (uint8)(clamp255(sobel));
>+    dst_sobelx[i] = (uint8_t)(clamp255(sobel));
>   }
> }
> 
>-void SobelYRow_C(const uint8* src_y0,
>-                 const uint8* src_y1,
>-                 uint8* dst_sobely,
>+void SobelYRow_C(const uint8_t* src_y0,
>+                 const uint8_t* src_y1,
>+                 uint8_t* dst_sobely,
>                  int width) {
>   int i;
>   for (i = 0; i < width; ++i) {
>@@ -911,62 +995,62 @@ void SobelYRow_C(const uint8* src_y0,
>     int b_diff = b - b_sub;
>     int c_diff = c - c_sub;
>     int sobel = Abs(a_diff + b_diff * 2 + c_diff);
>-    dst_sobely[i] = (uint8)(clamp255(sobel));
>+    dst_sobely[i] = (uint8_t)(clamp255(sobel));
>   }
> }
> 
>-void SobelRow_C(const uint8* src_sobelx,
>-                const uint8* src_sobely,
>-                uint8* dst_argb,
>+void SobelRow_C(const uint8_t* src_sobelx,
>+                const uint8_t* src_sobely,
>+                uint8_t* dst_argb,
>                 int width) {
>   int i;
>   for (i = 0; i < width; ++i) {
>     int r = src_sobelx[i];
>     int b = src_sobely[i];
>     int s = clamp255(r + b);
>-    dst_argb[0] = (uint8)(s);
>-    dst_argb[1] = (uint8)(s);
>-    dst_argb[2] = (uint8)(s);
>-    dst_argb[3] = (uint8)(255u);
>+    dst_argb[0] = (uint8_t)(s);
>+    dst_argb[1] = (uint8_t)(s);
>+    dst_argb[2] = (uint8_t)(s);
>+    dst_argb[3] = (uint8_t)(255u);
>     dst_argb += 4;
>   }
> }
> 
>-void SobelToPlaneRow_C(const uint8* src_sobelx,
>-                       const uint8* src_sobely,
>-                       uint8* dst_y,
>+void SobelToPlaneRow_C(const uint8_t* src_sobelx,
>+                       const uint8_t* src_sobely,
>+                       uint8_t* dst_y,
>                        int width) {
>   int i;
>   for (i = 0; i < width; ++i) {
>     int r = src_sobelx[i];
>     int b = src_sobely[i];
>     int s = clamp255(r + b);
>-    dst_y[i] = (uint8)(s);
>+    dst_y[i] = (uint8_t)(s);
>   }
> }
> 
>-void SobelXYRow_C(const uint8* src_sobelx,
>-                  const uint8* src_sobely,
>-                  uint8* dst_argb,
>+void SobelXYRow_C(const uint8_t* src_sobelx,
>+                  const uint8_t* src_sobely,
>+                  uint8_t* dst_argb,
>                   int width) {
>   int i;
>   for (i = 0; i < width; ++i) {
>     int r = src_sobelx[i];
>     int b = src_sobely[i];
>     int g = clamp255(r + b);
>-    dst_argb[0] = (uint8)(b);
>-    dst_argb[1] = (uint8)(g);
>-    dst_argb[2] = (uint8)(r);
>-    dst_argb[3] = (uint8)(255u);
>+    dst_argb[0] = (uint8_t)(b);
>+    dst_argb[1] = (uint8_t)(g);
>+    dst_argb[2] = (uint8_t)(r);
>+    dst_argb[3] = (uint8_t)(255u);
>     dst_argb += 4;
>   }
> }
> 
>-void J400ToARGBRow_C(const uint8* src_y, uint8* dst_argb, int width) {
>+void J400ToARGBRow_C(const uint8_t* src_y, uint8_t* dst_argb, int width) {
>   // Copy a Y to RGB.
>   int x;
>   for (x = 0; x < width; ++x) {
>-    uint8 y = src_y[0];
>+    uint8_t y = src_y[0];
>     dst_argb[2] = dst_argb[1] = dst_argb[0] = y;
>     dst_argb[3] = 255u;
>     dst_argb += 4;
>@@ -1223,12 +1307,14 @@ const struct YuvConstants SIMD_ALIGNED(kYvuH709Constants) = {
> #undef YG
> 
> // C reference code that mimics the YUV assembly.
>-static __inline void YuvPixel(uint8 y,
>-                              uint8 u,
>-                              uint8 v,
>-                              uint8* b,
>-                              uint8* g,
>-                              uint8* r,
>+// Reads 8 bit YUV and leaves result as 16 bit.
>+
>+static __inline void YuvPixel(uint8_t y,
>+                              uint8_t u,
>+                              uint8_t v,
>+                              uint8_t* b,
>+                              uint8_t* g,
>+                              uint8_t* r,
>                               const struct YuvConstants* yuvconstants) {
> #if defined(__aarch64__)
>   int ub = -yuvconstants->kUVToRB[0];
>@@ -1259,10 +1345,117 @@ static __inline void YuvPixel(uint8 y,
>   int yg = yuvconstants->kYToRgb[0];
> #endif
> 
>-  uint32 y1 = (uint32)(y * 0x0101 * yg) >> 16;
>-  *b = Clamp((int32)(-(u * ub) + y1 + bb) >> 6);
>-  *g = Clamp((int32)(-(u * ug + v * vg) + y1 + bg) >> 6);
>-  *r = Clamp((int32)(-(v * vr) + y1 + br) >> 6);
>+  uint32_t y1 = (uint32_t)(y * 0x0101 * yg) >> 16;
>+  *b = Clamp((int32_t)(-(u * ub) + y1 + bb) >> 6);
>+  *g = Clamp((int32_t)(-(u * ug + v * vg) + y1 + bg) >> 6);
>+  *r = Clamp((int32_t)(-(v * vr) + y1 + br) >> 6);
>+}
>+
>+// Reads 8 bit YUV and leaves result as 16 bit.
>+static __inline void YuvPixel8_16(uint8_t y,
>+                                  uint8_t u,
>+                                  uint8_t v,
>+                                  int* b,
>+                                  int* g,
>+                                  int* r,
>+                                  const struct YuvConstants* yuvconstants) {
>+#if defined(__aarch64__)
>+  int ub = -yuvconstants->kUVToRB[0];
>+  int ug = yuvconstants->kUVToG[0];
>+  int vg = yuvconstants->kUVToG[1];
>+  int vr = -yuvconstants->kUVToRB[1];
>+  int bb = yuvconstants->kUVBiasBGR[0];
>+  int bg = yuvconstants->kUVBiasBGR[1];
>+  int br = yuvconstants->kUVBiasBGR[2];
>+  int yg = yuvconstants->kYToRgb[0] / 0x0101;
>+#elif defined(__arm__)
>+  int ub = -yuvconstants->kUVToRB[0];
>+  int ug = yuvconstants->kUVToG[0];
>+  int vg = yuvconstants->kUVToG[4];
>+  int vr = -yuvconstants->kUVToRB[4];
>+  int bb = yuvconstants->kUVBiasBGR[0];
>+  int bg = yuvconstants->kUVBiasBGR[1];
>+  int br = yuvconstants->kUVBiasBGR[2];
>+  int yg = yuvconstants->kYToRgb[0] / 0x0101;
>+#else
>+  int ub = yuvconstants->kUVToB[0];
>+  int ug = yuvconstants->kUVToG[0];
>+  int vg = yuvconstants->kUVToG[1];
>+  int vr = yuvconstants->kUVToR[1];
>+  int bb = yuvconstants->kUVBiasB[0];
>+  int bg = yuvconstants->kUVBiasG[0];
>+  int br = yuvconstants->kUVBiasR[0];
>+  int yg = yuvconstants->kYToRgb[0];
>+#endif
>+
>+  uint32_t y1 = (uint32_t)(y * 0x0101 * yg) >> 16;
>+  *b = (int)(-(u * ub) + y1 + bb);
>+  *g = (int)(-(u * ug + v * vg) + y1 + bg);
>+  *r = (int)(-(v * vr) + y1 + br);
>+}
>+
>+// C reference code that mimics the YUV 16 bit assembly.
>+// Reads 10 bit YUV and leaves result as 16 bit.
>+static __inline void YuvPixel16(int16_t y,
>+                                int16_t u,
>+                                int16_t v,
>+                                int* b,
>+                                int* g,
>+                                int* r,
>+                                const struct YuvConstants* yuvconstants) {
>+#if defined(__aarch64__)
>+  int ub = -yuvconstants->kUVToRB[0];
>+  int ug = yuvconstants->kUVToG[0];
>+  int vg = yuvconstants->kUVToG[1];
>+  int vr = -yuvconstants->kUVToRB[1];
>+  int bb = yuvconstants->kUVBiasBGR[0];
>+  int bg = yuvconstants->kUVBiasBGR[1];
>+  int br = yuvconstants->kUVBiasBGR[2];
>+  int yg = yuvconstants->kYToRgb[0] / 0x0101;
>+#elif defined(__arm__)
>+  int ub = -yuvconstants->kUVToRB[0];
>+  int ug = yuvconstants->kUVToG[0];
>+  int vg = yuvconstants->kUVToG[4];
>+  int vr = -yuvconstants->kUVToRB[4];
>+  int bb = yuvconstants->kUVBiasBGR[0];
>+  int bg = yuvconstants->kUVBiasBGR[1];
>+  int br = yuvconstants->kUVBiasBGR[2];
>+  int yg = yuvconstants->kYToRgb[0] / 0x0101;
>+#else
>+  int ub = yuvconstants->kUVToB[0];
>+  int ug = yuvconstants->kUVToG[0];
>+  int vg = yuvconstants->kUVToG[1];
>+  int vr = yuvconstants->kUVToR[1];
>+  int bb = yuvconstants->kUVBiasB[0];
>+  int bg = yuvconstants->kUVBiasG[0];
>+  int br = yuvconstants->kUVBiasR[0];
>+  int yg = yuvconstants->kYToRgb[0];
>+#endif
>+
>+  uint32_t y1 = (uint32_t)((y << 6) * yg) >> 16;
>+  u = clamp255(u >> 2);
>+  v = clamp255(v >> 2);
>+  *b = (int)(-(u * ub) + y1 + bb);
>+  *g = (int)(-(u * ug + v * vg) + y1 + bg);
>+  *r = (int)(-(v * vr) + y1 + br);
>+}
>+
>+// C reference code that mimics the YUV 10 bit assembly.
>+// Reads 10 bit YUV and clamps down to 8 bit RGB.
>+static __inline void YuvPixel10(uint16_t y,
>+                                uint16_t u,
>+                                uint16_t v,
>+                                uint8_t* b,
>+                                uint8_t* g,
>+                                uint8_t* r,
>+                                const struct YuvConstants* yuvconstants) {
>+  int b16;
>+  int g16;
>+  int r16;
>+  YuvPixel16(y, u, v, &b16, &g16, &r16, yuvconstants);
>+  *b = Clamp(b16 >> 6);
>+  *g = Clamp(g16 >> 6);
>+  *r = Clamp(r16 >> 6);
> }
> 
> // Y contribution to R,G,B.  Scale and bias.
>@@ -1270,11 +1463,11 @@ static __inline void YuvPixel(uint8 y,
> #define YGB -1160 /* 1.164 * 64 * -16 + 64 / 2 */
> 
> // C reference code that mimics the YUV assembly.
>-static __inline void YPixel(uint8 y, uint8* b, uint8* g, uint8* r) {
>-  uint32 y1 = (uint32)(y * 0x0101 * YG) >> 16;
>-  *b = Clamp((int32)(y1 + YGB) >> 6);
>-  *g = Clamp((int32)(y1 + YGB) >> 6);
>-  *r = Clamp((int32)(y1 + YGB) >> 6);
>+static __inline void YPixel(uint8_t y, uint8_t* b, uint8_t* g, uint8_t* r) {
>+  uint32_t y1 = (uint32_t)(y * 0x0101 * YG) >> 16;
>+  *b = Clamp((int32_t)(y1 + YGB) >> 6);
>+  *g = Clamp((int32_t)(y1 + YGB) >> 6);
>+  *r = Clamp((int32_t)(y1 + YGB) >> 6);
> }
> 
> #undef YG
>@@ -1284,16 +1477,16 @@ static __inline void YPixel(uint8 y, uint8* b, uint8* g, uint8* r) {
>     (defined(__ARM_NEON__) || defined(__aarch64__) || defined(LIBYUV_NEON))
> // C mimic assembly.
> // TODO(fbarchard): Remove subsampling from Neon.
>-void I444ToARGBRow_C(const uint8* src_y,
>-                     const uint8* src_u,
>-                     const uint8* src_v,
>-                     uint8* rgb_buf,
>+void I444ToARGBRow_C(const uint8_t* src_y,
>+                     const uint8_t* src_u,
>+                     const uint8_t* src_v,
>+                     uint8_t* rgb_buf,
>                      const struct YuvConstants* yuvconstants,
>                      int width) {
>   int x;
>   for (x = 0; x < width - 1; x += 2) {
>-    uint8 u = (src_u[0] + src_u[1] + 1) >> 1;
>-    uint8 v = (src_v[0] + src_v[1] + 1) >> 1;
>+    uint8_t u = (src_u[0] + src_u[1] + 1) >> 1;
>+    uint8_t v = (src_v[0] + src_v[1] + 1) >> 1;
>     YuvPixel(src_y[0], u, v, rgb_buf + 0, rgb_buf + 1, rgb_buf + 2,
>              yuvconstants);
>     rgb_buf[3] = 255;
>@@ -1312,10 +1505,10 @@ void I444ToARGBRow_C(const uint8* src_y,
>   }
> }
> #else
>-void I444ToARGBRow_C(const uint8* src_y,
>-                     const uint8* src_u,
>-                     const uint8* src_v,
>-                     uint8* rgb_buf,
>+void I444ToARGBRow_C(const uint8_t* src_y,
>+                     const uint8_t* src_u,
>+                     const uint8_t* src_v,
>+                     uint8_t* rgb_buf,
>                      const struct YuvConstants* yuvconstants,
>                      int width) {
>   int x;
>@@ -1332,10 +1525,10 @@ void I444ToARGBRow_C(const uint8* src_y,
> #endif
> 
> // Also used for 420
>-void I422ToARGBRow_C(const uint8* src_y,
>-                     const uint8* src_u,
>-                     const uint8* src_v,
>-                     uint8* rgb_buf,
>+void I422ToARGBRow_C(const uint8_t* src_y,
>+                     const uint8_t* src_u,
>+                     const uint8_t* src_v,
>+                     uint8_t* rgb_buf,
>                      const struct YuvConstants* yuvconstants,
>                      int width) {
>   int x;
>@@ -1358,11 +1551,105 @@ void I422ToARGBRow_C(const uint8* src_y,
>   }
> }
> 
>-void I422AlphaToARGBRow_C(const uint8* src_y,
>-                          const uint8* src_u,
>-                          const uint8* src_v,
>-                          const uint8* src_a,
>-                          uint8* rgb_buf,
>+// 10 bit YUV to ARGB
>+void I210ToARGBRow_C(const uint16_t* src_y,
>+                     const uint16_t* src_u,
>+                     const uint16_t* src_v,
>+                     uint8_t* rgb_buf,
>+                     const struct YuvConstants* yuvconstants,
>+                     int width) {
>+  int x;
>+  for (x = 0; x < width - 1; x += 2) {
>+    YuvPixel10(src_y[0], src_u[0], src_v[0], rgb_buf + 0, rgb_buf + 1,
>+               rgb_buf + 2, yuvconstants);
>+    rgb_buf[3] = 255;
>+    YuvPixel10(src_y[1], src_u[0], src_v[0], rgb_buf + 4, rgb_buf + 5,
>+               rgb_buf + 6, yuvconstants);
>+    rgb_buf[7] = 255;
>+    src_y += 2;
>+    src_u += 1;
>+    src_v += 1;
>+    rgb_buf += 8;  // Advance 2 pixels.
>+  }
>+  if (width & 1) {
>+    YuvPixel10(src_y[0], src_u[0], src_v[0], rgb_buf + 0, rgb_buf + 1,
>+               rgb_buf + 2, yuvconstants);
>+    rgb_buf[3] = 255;
>+  }
>+}
>+
>+static void StoreAR30(uint8_t* rgb_buf, int b, int g, int r) {
>+  uint32_t ar30;
>+  b = b >> 4;  // convert 10.6 to 10 bit.
>+  g = g >> 4;
>+  r = r >> 4;
>+  b = Clamp10(b);
>+  g = Clamp10(g);
>+  r = Clamp10(r);
>+  ar30 = b | ((uint32_t)g << 10) | ((uint32_t)r << 20) | 0xc0000000;
>+  (*(uint32_t*)rgb_buf) = ar30;
>+}
>+
>+// 10 bit YUV to 10 bit AR30
>+void I210ToAR30Row_C(const uint16_t* src_y,
>+                     const uint16_t* src_u,
>+                     const uint16_t* src_v,
>+                     uint8_t* rgb_buf,
>+                     const struct YuvConstants* yuvconstants,
>+                     int width) {
>+  int x;
>+  int b;
>+  int g;
>+  int r;
>+  for (x = 0; x < width - 1; x += 2) {
>+    YuvPixel16(src_y[0], src_u[0], src_v[0], &b, &g, &r, yuvconstants);
>+    StoreAR30(rgb_buf, b, g, r);
>+    YuvPixel16(src_y[1], src_u[0], src_v[0], &b, &g, &r, yuvconstants);
>+    StoreAR30(rgb_buf + 4, b, g, r);
>+    src_y += 2;
>+    src_u += 1;
>+    src_v += 1;
>+    rgb_buf += 8;  // Advance 2 pixels.
>+  }
>+  if (width & 1) {
>+    YuvPixel16(src_y[0], src_u[0], src_v[0], &b, &g, &r, yuvconstants);
>+    StoreAR30(rgb_buf, b, g, r);
>+  }
>+}
>+
>+// 8 bit YUV to 10 bit AR30
>+// Uses same code as 10 bit YUV bit shifts the 8 bit values up to 10 bits.
>+void I422ToAR30Row_C(const uint8_t* src_y,
>+                     const uint8_t* src_u,
>+                     const uint8_t* src_v,
>+                     uint8_t* rgb_buf,
>+                     const struct YuvConstants* yuvconstants,
>+                     int width) {
>+  int x;
>+  int b;
>+  int g;
>+  int r;
>+  for (x = 0; x < width - 1; x += 2) {
>+    YuvPixel8_16(src_y[0], src_u[0], src_v[0], &b, &g, &r, yuvconstants);
>+    StoreAR30(rgb_buf, b, g, r);
>+    YuvPixel8_16(src_y[1], src_u[0], src_v[0], &b, &g, &r, yuvconstants);
>+    StoreAR30(rgb_buf + 4, b, g, r);
>+    src_y += 2;
>+    src_u += 1;
>+    src_v += 1;
>+    rgb_buf += 8;  // Advance 2 pixels.
>+  }
>+  if (width & 1) {
>+    YuvPixel8_16(src_y[0], src_u[0], src_v[0], &b, &g, &r, yuvconstants);
>+    StoreAR30(rgb_buf, b, g, r);
>+  }
>+}
>+
>+void I422AlphaToARGBRow_C(const uint8_t* src_y,
>+                          const uint8_t* src_u,
>+                          const uint8_t* src_v,
>+                          const uint8_t* src_a,
>+                          uint8_t* rgb_buf,
>                           const struct YuvConstants* yuvconstants,
>                           int width) {
>   int x;
>@@ -1386,10 +1673,10 @@ void I422AlphaToARGBRow_C(const uint8* src_y,
>   }
> }
> 
>-void I422ToRGB24Row_C(const uint8* src_y,
>-                      const uint8* src_u,
>-                      const uint8* src_v,
>-                      uint8* rgb_buf,
>+void I422ToRGB24Row_C(const uint8_t* src_y,
>+                      const uint8_t* src_u,
>+                      const uint8_t* src_v,
>+                      uint8_t* rgb_buf,
>                       const struct YuvConstants* yuvconstants,
>                       int width) {
>   int x;
>@@ -1409,18 +1696,18 @@ void I422ToRGB24Row_C(const uint8* src_y,
>   }
> }
> 
>-void I422ToARGB4444Row_C(const uint8* src_y,
>-                         const uint8* src_u,
>-                         const uint8* src_v,
>-                         uint8* dst_argb4444,
>+void I422ToARGB4444Row_C(const uint8_t* src_y,
>+                         const uint8_t* src_u,
>+                         const uint8_t* src_v,
>+                         uint8_t* dst_argb4444,
>                          const struct YuvConstants* yuvconstants,
>                          int width) {
>-  uint8 b0;
>-  uint8 g0;
>-  uint8 r0;
>-  uint8 b1;
>-  uint8 g1;
>-  uint8 r1;
>+  uint8_t b0;
>+  uint8_t g0;
>+  uint8_t r0;
>+  uint8_t b1;
>+  uint8_t g1;
>+  uint8_t r1;
>   int x;
>   for (x = 0; x < width - 1; x += 2) {
>     YuvPixel(src_y[0], src_u[0], src_v[0], &b0, &g0, &r0, yuvconstants);
>@@ -1431,8 +1718,8 @@ void I422ToARGB4444Row_C(const uint8* src_y,
>     b1 = b1 >> 4;
>     g1 = g1 >> 4;
>     r1 = r1 >> 4;
>-    *(uint32*)(dst_argb4444) = b0 | (g0 << 4) | (r0 << 8) | (b1 << 16) |
>-                               (g1 << 20) | (r1 << 24) | 0xf000f000;
>+    *(uint32_t*)(dst_argb4444) = b0 | (g0 << 4) | (r0 << 8) | (b1 << 16) |
>+                                 (g1 << 20) | (r1 << 24) | 0xf000f000;
>     src_y += 2;
>     src_u += 1;
>     src_v += 1;
>@@ -1443,22 +1730,22 @@ void I422ToARGB4444Row_C(const uint8* src_y,
>     b0 = b0 >> 4;
>     g0 = g0 >> 4;
>     r0 = r0 >> 4;
>-    *(uint16*)(dst_argb4444) = b0 | (g0 << 4) | (r0 << 8) | 0xf000;
>+    *(uint16_t*)(dst_argb4444) = b0 | (g0 << 4) | (r0 << 8) | 0xf000;
>   }
> }
> 
>-void I422ToARGB1555Row_C(const uint8* src_y,
>-                         const uint8* src_u,
>-                         const uint8* src_v,
>-                         uint8* dst_argb1555,
>+void I422ToARGB1555Row_C(const uint8_t* src_y,
>+                         const uint8_t* src_u,
>+                         const uint8_t* src_v,
>+                         uint8_t* dst_argb1555,
>                          const struct YuvConstants* yuvconstants,
>                          int width) {
>-  uint8 b0;
>-  uint8 g0;
>-  uint8 r0;
>-  uint8 b1;
>-  uint8 g1;
>-  uint8 r1;
>+  uint8_t b0;
>+  uint8_t g0;
>+  uint8_t r0;
>+  uint8_t b1;
>+  uint8_t g1;
>+  uint8_t r1;
>   int x;
>   for (x = 0; x < width - 1; x += 2) {
>     YuvPixel(src_y[0], src_u[0], src_v[0], &b0, &g0, &r0, yuvconstants);
>@@ -1469,8 +1756,8 @@ void I422ToARGB1555Row_C(const uint8* src_y,
>     b1 = b1 >> 3;
>     g1 = g1 >> 3;
>     r1 = r1 >> 3;
>-    *(uint32*)(dst_argb1555) = b0 | (g0 << 5) | (r0 << 10) | (b1 << 16) |
>-                               (g1 << 21) | (r1 << 26) | 0x80008000;
>+    *(uint32_t*)(dst_argb1555) = b0 | (g0 << 5) | (r0 << 10) | (b1 << 16) |
>+                                 (g1 << 21) | (r1 << 26) | 0x80008000;
>     src_y += 2;
>     src_u += 1;
>     src_v += 1;
>@@ -1481,22 +1768,22 @@ void I422ToARGB1555Row_C(const uint8* src_y,
>     b0 = b0 >> 3;
>     g0 = g0 >> 3;
>     r0 = r0 >> 3;
>-    *(uint16*)(dst_argb1555) = b0 | (g0 << 5) | (r0 << 10) | 0x8000;
>+    *(uint16_t*)(dst_argb1555) = b0 | (g0 << 5) | (r0 << 10) | 0x8000;
>   }
> }
> 
>-void I422ToRGB565Row_C(const uint8* src_y,
>-                       const uint8* src_u,
>-                       const uint8* src_v,
>-                       uint8* dst_rgb565,
>+void I422ToRGB565Row_C(const uint8_t* src_y,
>+                       const uint8_t* src_u,
>+                       const uint8_t* src_v,
>+                       uint8_t* dst_rgb565,
>                        const struct YuvConstants* yuvconstants,
>                        int width) {
>-  uint8 b0;
>-  uint8 g0;
>-  uint8 r0;
>-  uint8 b1;
>-  uint8 g1;
>-  uint8 r1;
>+  uint8_t b0;
>+  uint8_t g0;
>+  uint8_t r0;
>+  uint8_t b1;
>+  uint8_t g1;
>+  uint8_t r1;
>   int x;
>   for (x = 0; x < width - 1; x += 2) {
>     YuvPixel(src_y[0], src_u[0], src_v[0], &b0, &g0, &r0, yuvconstants);
>@@ -1507,7 +1794,7 @@ void I422ToRGB565Row_C(const uint8* src_y,
>     b1 = b1 >> 3;
>     g1 = g1 >> 2;
>     r1 = r1 >> 3;
>-    *(uint32*)(dst_rgb565) =
>+    *(uint32_t*)(dst_rgb565) =
>         b0 | (g0 << 5) | (r0 << 11) | (b1 << 16) | (g1 << 21) | (r1 << 27);
>     src_y += 2;
>     src_u += 1;
>@@ -1519,13 +1806,13 @@ void I422ToRGB565Row_C(const uint8* src_y,
>     b0 = b0 >> 3;
>     g0 = g0 >> 2;
>     r0 = r0 >> 3;
>-    *(uint16*)(dst_rgb565) = b0 | (g0 << 5) | (r0 << 11);
>+    *(uint16_t*)(dst_rgb565) = b0 | (g0 << 5) | (r0 << 11);
>   }
> }
> 
>-void NV12ToARGBRow_C(const uint8* src_y,
>-                     const uint8* src_uv,
>-                     uint8* rgb_buf,
>+void NV12ToARGBRow_C(const uint8_t* src_y,
>+                     const uint8_t* src_uv,
>+                     uint8_t* rgb_buf,
>                      const struct YuvConstants* yuvconstants,
>                      int width) {
>   int x;
>@@ -1547,9 +1834,9 @@ void NV12ToARGBRow_C(const uint8* src_y,
>   }
> }
> 
>-void NV21ToARGBRow_C(const uint8* src_y,
>-                     const uint8* src_vu,
>-                     uint8* rgb_buf,
>+void NV21ToARGBRow_C(const uint8_t* src_y,
>+                     const uint8_t* src_vu,
>+                     uint8_t* rgb_buf,
>                      const struct YuvConstants* yuvconstants,
>                      int width) {
>   int x;
>@@ -1571,17 +1858,59 @@ void NV21ToARGBRow_C(const uint8* src_y,
>   }
> }
> 
>-void NV12ToRGB565Row_C(const uint8* src_y,
>-                       const uint8* src_uv,
>-                       uint8* dst_rgb565,
>+void NV12ToRGB24Row_C(const uint8_t* src_y,
>+                      const uint8_t* src_uv,
>+                      uint8_t* rgb_buf,
>+                      const struct YuvConstants* yuvconstants,
>+                      int width) {
>+  int x;
>+  for (x = 0; x < width - 1; x += 2) {
>+    YuvPixel(src_y[0], src_uv[0], src_uv[1], rgb_buf + 0, rgb_buf + 1,
>+             rgb_buf + 2, yuvconstants);
>+    YuvPixel(src_y[1], src_uv[0], src_uv[1], rgb_buf + 3, rgb_buf + 4,
>+             rgb_buf + 5, yuvconstants);
>+    src_y += 2;
>+    src_uv += 2;
>+    rgb_buf += 6;  // Advance 2 pixels.
>+  }
>+  if (width & 1) {
>+    YuvPixel(src_y[0], src_uv[0], src_uv[1], rgb_buf + 0, rgb_buf + 1,
>+             rgb_buf + 2, yuvconstants);
>+  }
>+}
>+
>+void NV21ToRGB24Row_C(const uint8_t* src_y,
>+                      const uint8_t* src_vu,
>+                      uint8_t* rgb_buf,
>+                      const struct YuvConstants* yuvconstants,
>+                      int width) {
>+  int x;
>+  for (x = 0; x < width - 1; x += 2) {
>+    YuvPixel(src_y[0], src_vu[1], src_vu[0], rgb_buf + 0, rgb_buf + 1,
>+             rgb_buf + 2, yuvconstants);
>+    YuvPixel(src_y[1], src_vu[1], src_vu[0], rgb_buf + 3, rgb_buf + 4,
>+             rgb_buf + 5, yuvconstants);
>+    src_y += 2;
>+    src_vu += 2;
>+    rgb_buf += 6;  // Advance 2 pixels.
>+  }
>+  if (width & 1) {
>+    YuvPixel(src_y[0], src_vu[1], src_vu[0], rgb_buf + 0, rgb_buf + 1,
>+             rgb_buf + 2, yuvconstants);
>+  }
>+}
>+
>+void NV12ToRGB565Row_C(const uint8_t* src_y,
>+                       const uint8_t* src_uv,
>+                       uint8_t* dst_rgb565,
>                        const struct YuvConstants* yuvconstants,
>                        int width) {
>-  uint8 b0;
>-  uint8 g0;
>-  uint8 r0;
>-  uint8 b1;
>-  uint8 g1;
>-  uint8 r1;
>+  uint8_t b0;
>+  uint8_t g0;
>+  uint8_t r0;
>+  uint8_t b1;
>+  uint8_t g1;
>+  uint8_t r1;
>   int x;
>   for (x = 0; x < width - 1; x += 2) {
>     YuvPixel(src_y[0], src_uv[0], src_uv[1], &b0, &g0, &r0, yuvconstants);
>@@ -1592,7 +1921,7 @@ void NV12ToRGB565Row_C(const uint8* src_y,
>     b1 = b1 >> 3;
>     g1 = g1 >> 2;
>     r1 = r1 >> 3;
>-    *(uint32*)(dst_rgb565) =
>+    *(uint32_t*)(dst_rgb565) =
>         b0 | (g0 << 5) | (r0 << 11) | (b1 << 16) | (g1 << 21) | (r1 << 27);
>     src_y += 2;
>     src_uv += 2;
>@@ -1603,12 +1932,12 @@ void NV12ToRGB565Row_C(const uint8* src_y,
>     b0 = b0 >> 3;
>     g0 = g0 >> 2;
>     r0 = r0 >> 3;
>-    *(uint16*)(dst_rgb565) = b0 | (g0 << 5) | (r0 << 11);
>+    *(uint16_t*)(dst_rgb565) = b0 | (g0 << 5) | (r0 << 11);
>   }
> }
> 
>-void YUY2ToARGBRow_C(const uint8* src_yuy2,
>-                     uint8* rgb_buf,
>+void YUY2ToARGBRow_C(const uint8_t* src_yuy2,
>+                     uint8_t* rgb_buf,
>                      const struct YuvConstants* yuvconstants,
>                      int width) {
>   int x;
>@@ -1629,8 +1958,8 @@ void YUY2ToARGBRow_C(const uint8* src_yuy2,
>   }
> }
> 
>-void UYVYToARGBRow_C(const uint8* src_uyvy,
>-                     uint8* rgb_buf,
>+void UYVYToARGBRow_C(const uint8_t* src_uyvy,
>+                     uint8_t* rgb_buf,
>                      const struct YuvConstants* yuvconstants,
>                      int width) {
>   int x;
>@@ -1651,10 +1980,10 @@ void UYVYToARGBRow_C(const uint8* src_uyvy,
>   }
> }
> 
>-void I422ToRGBARow_C(const uint8* src_y,
>-                     const uint8* src_u,
>-                     const uint8* src_v,
>-                     uint8* rgb_buf,
>+void I422ToRGBARow_C(const uint8_t* src_y,
>+                     const uint8_t* src_u,
>+                     const uint8_t* src_v,
>+                     uint8_t* rgb_buf,
>                      const struct YuvConstants* yuvconstants,
>                      int width) {
>   int x;
>@@ -1677,7 +2006,7 @@ void I422ToRGBARow_C(const uint8* src_y,
>   }
> }
> 
>-void I400ToARGBRow_C(const uint8* src_y, uint8* rgb_buf, int width) {
>+void I400ToARGBRow_C(const uint8_t* src_y, uint8_t* rgb_buf, int width) {
>   int x;
>   for (x = 0; x < width - 1; x += 2) {
>     YPixel(src_y[0], rgb_buf + 0, rgb_buf + 1, rgb_buf + 2);
>@@ -1693,7 +2022,7 @@ void I400ToARGBRow_C(const uint8* src_y, uint8* rgb_buf, int width) {
>   }
> }
> 
>-void MirrorRow_C(const uint8* src, uint8* dst, int width) {
>+void MirrorRow_C(const uint8_t* src, uint8_t* dst, int width) {
>   int x;
>   src += width - 1;
>   for (x = 0; x < width - 1; x += 2) {
>@@ -1706,7 +2035,10 @@ void MirrorRow_C(const uint8* src, uint8* dst, int width) {
>   }
> }
> 
>-void MirrorUVRow_C(const uint8* src_uv, uint8* dst_u, uint8* dst_v, int width) {
>+void MirrorUVRow_C(const uint8_t* src_uv,
>+                   uint8_t* dst_u,
>+                   uint8_t* dst_v,
>+                   int width) {
>   int x;
>   src_uv += (width - 1) << 1;
>   for (x = 0; x < width - 1; x += 2) {
>@@ -1722,10 +2054,10 @@ void MirrorUVRow_C(const uint8* src_uv, uint8* dst_u, uint8* dst_v, int width) {
>   }
> }
> 
>-void ARGBMirrorRow_C(const uint8* src, uint8* dst, int width) {
>+void ARGBMirrorRow_C(const uint8_t* src, uint8_t* dst, int width) {
>   int x;
>-  const uint32* src32 = (const uint32*)(src);
>-  uint32* dst32 = (uint32*)(dst);
>+  const uint32_t* src32 = (const uint32_t*)(src);
>+  uint32_t* dst32 = (uint32_t*)(dst);
>   src32 += width - 1;
>   for (x = 0; x < width - 1; x += 2) {
>     dst32[x] = src32[0];
>@@ -1737,7 +2069,10 @@ void ARGBMirrorRow_C(const uint8* src, uint8* dst, int width) {
>   }
> }
> 
>-void SplitUVRow_C(const uint8* src_uv, uint8* dst_u, uint8* dst_v, int width) {
>+void SplitUVRow_C(const uint8_t* src_uv,
>+                  uint8_t* dst_u,
>+                  uint8_t* dst_v,
>+                  int width) {
>   int x;
>   for (x = 0; x < width - 1; x += 2) {
>     dst_u[x] = src_uv[0];
>@@ -1752,9 +2087,9 @@ void SplitUVRow_C(const uint8* src_uv, uint8* dst_u, uint8* dst_v, int width) {
>   }
> }
> 
>-void MergeUVRow_C(const uint8* src_u,
>-                  const uint8* src_v,
>-                  uint8* dst_uv,
>+void MergeUVRow_C(const uint8_t* src_u,
>+                  const uint8_t* src_v,
>+                  uint8_t* dst_uv,
>                   int width) {
>   int x;
>   for (x = 0; x < width - 1; x += 2) {
>@@ -1770,10 +2105,10 @@ void MergeUVRow_C(const uint8* src_u,
>   }
> }
> 
>-void SplitRGBRow_C(const uint8* src_rgb,
>-                   uint8* dst_r,
>-                   uint8* dst_g,
>-                   uint8* dst_b,
>+void SplitRGBRow_C(const uint8_t* src_rgb,
>+                   uint8_t* dst_r,
>+                   uint8_t* dst_g,
>+                   uint8_t* dst_b,
>                    int width) {
>   int x;
>   for (x = 0; x < width; ++x) {
>@@ -1784,10 +2119,10 @@ void SplitRGBRow_C(const uint8* src_rgb,
>   }
> }
> 
>-void MergeRGBRow_C(const uint8* src_r,
>-                   const uint8* src_g,
>-                   const uint8* src_b,
>-                   uint8* dst_rgb,
>+void MergeRGBRow_C(const uint8_t* src_r,
>+                   const uint8_t* src_g,
>+                   const uint8_t* src_b,
>+                   uint8_t* dst_rgb,
>                    int width) {
>   int x;
>   for (x = 0; x < width; ++x) {
>@@ -1798,9 +2133,14 @@ void MergeRGBRow_C(const uint8* src_r,
>   }
> }
> 
>-void MergeUVRow_16_C(const uint16* src_u,
>-                     const uint16* src_v,
>-                     uint16* dst_uv,
>+// Use scale to convert lsb formats to msb, depending how many bits there are:
>+// 128 = 9 bits
>+// 64 = 10 bits
>+// 16 = 12 bits
>+// 1 = 16 bits
>+void MergeUVRow_16_C(const uint16_t* src_u,
>+                     const uint16_t* src_v,
>+                     uint16_t* dst_uv,
>                      int scale,
>                      int width) {
>   int x;
>@@ -1817,8 +2157,8 @@ void MergeUVRow_16_C(const uint16* src_u,
>   }
> }
> 
>-void MultiplyRow_16_C(const uint16* src_y,
>-                      uint16* dst_y,
>+void MultiplyRow_16_C(const uint16_t* src_y,
>+                      uint16_t* dst_y,
>                       int scale,
>                       int width) {
>   int x;
>@@ -1827,20 +2167,48 @@ void MultiplyRow_16_C(const uint16* src_y,
>   }
> }
> 
>-void CopyRow_C(const uint8* src, uint8* dst, int count) {
>+// Use scale to convert lsb formats to msb, depending how many bits there are:
>+// 32768 = 9 bits
>+// 16384 = 10 bits
>+// 4096 = 12 bits
>+// 256 = 16 bits
>+void Convert16To8Row_C(const uint16_t* src_y,
>+                       uint8_t* dst_y,
>+                       int scale,
>+                       int width) {
>+  int x;
>+  for (x = 0; x < width; ++x) {
>+    dst_y[x] = clamp255((src_y[x] * scale) >> 16);
>+  }
>+}
>+
>+// Use scale to convert lsb formats to msb, depending how many bits there are:
>+// 1024 = 10 bits
>+void Convert8To16Row_C(const uint8_t* src_y,
>+                       uint16_t* dst_y,
>+                       int scale,
>+                       int width) {
>+  int x;
>+  scale *= 0x0101;  // replicates the byte.
>+  for (x = 0; x < width; ++x) {
>+    dst_y[x] = (src_y[x] * scale) >> 16;
>+  }
>+}
>+
>+void CopyRow_C(const uint8_t* src, uint8_t* dst, int count) {
>   memcpy(dst, src, count);
> }
> 
>-void CopyRow_16_C(const uint16* src, uint16* dst, int count) {
>+void CopyRow_16_C(const uint16_t* src, uint16_t* dst, int count) {
>   memcpy(dst, src, count * 2);
> }
> 
>-void SetRow_C(uint8* dst, uint8 v8, int width) {
>+void SetRow_C(uint8_t* dst, uint8_t v8, int width) {
>   memset(dst, v8, width);
> }
> 
>-void ARGBSetRow_C(uint8* dst_argb, uint32 v32, int width) {
>-  uint32* d = (uint32*)(dst_argb);
>+void ARGBSetRow_C(uint8_t* dst_argb, uint32_t v32, int width) {
>+  uint32_t* d = (uint32_t*)(dst_argb);
>   int x;
>   for (x = 0; x < width; ++x) {
>     d[x] = v32;
>@@ -1848,10 +2216,10 @@ void ARGBSetRow_C(uint8* dst_argb, uint32 v32, int width) {
> }
> 
> // Filter 2 rows of YUY2 UV's (422) into U and V (420).
>-void YUY2ToUVRow_C(const uint8* src_yuy2,
>+void YUY2ToUVRow_C(const uint8_t* src_yuy2,
>                    int src_stride_yuy2,
>-                   uint8* dst_u,
>-                   uint8* dst_v,
>+                   uint8_t* dst_u,
>+                   uint8_t* dst_v,
>                    int width) {
>   // Output a row of UV values, filtering 2 rows of YUY2.
>   int x;
>@@ -1865,9 +2233,9 @@ void YUY2ToUVRow_C(const uint8* src_yuy2,
> }
> 
> // Copy row of YUY2 UV's (422) into U and V (422).
>-void YUY2ToUV422Row_C(const uint8* src_yuy2,
>-                      uint8* dst_u,
>-                      uint8* dst_v,
>+void YUY2ToUV422Row_C(const uint8_t* src_yuy2,
>+                      uint8_t* dst_u,
>+                      uint8_t* dst_v,
>                       int width) {
>   // Output a row of UV values.
>   int x;
>@@ -1881,7 +2249,7 @@ void YUY2ToUV422Row_C(const uint8* src_yuy2,
> }
> 
> // Copy row of YUY2 Y's (422) into Y (420/422).
>-void YUY2ToYRow_C(const uint8* src_yuy2, uint8* dst_y, int width) {
>+void YUY2ToYRow_C(const uint8_t* src_yuy2, uint8_t* dst_y, int width) {
>   // Output a row of Y values.
>   int x;
>   for (x = 0; x < width - 1; x += 2) {
>@@ -1895,10 +2263,10 @@ void YUY2ToYRow_C(const uint8* src_yuy2, uint8* dst_y, int width) {
> }
> 
> // Filter 2 rows of UYVY UV's (422) into U and V (420).
>-void UYVYToUVRow_C(const uint8* src_uyvy,
>+void UYVYToUVRow_C(const uint8_t* src_uyvy,
>                    int src_stride_uyvy,
>-                   uint8* dst_u,
>-                   uint8* dst_v,
>+                   uint8_t* dst_u,
>+                   uint8_t* dst_v,
>                    int width) {
>   // Output a row of UV values.
>   int x;
>@@ -1912,9 +2280,9 @@ void UYVYToUVRow_C(const uint8* src_uyvy,
> }
> 
> // Copy row of UYVY UV's (422) into U and V (422).
>-void UYVYToUV422Row_C(const uint8* src_uyvy,
>-                      uint8* dst_u,
>-                      uint8* dst_v,
>+void UYVYToUV422Row_C(const uint8_t* src_uyvy,
>+                      uint8_t* dst_u,
>+                      uint8_t* dst_v,
>                       int width) {
>   // Output a row of UV values.
>   int x;
>@@ -1928,7 +2296,7 @@ void UYVYToUV422Row_C(const uint8* src_uyvy,
> }
> 
> // Copy row of UYVY Y's (422) into Y (420/422).
>-void UYVYToYRow_C(const uint8* src_uyvy, uint8* dst_y, int width) {
>+void UYVYToYRow_C(const uint8_t* src_uyvy, uint8_t* dst_y, int width) {
>   // Output a row of Y values.
>   int x;
>   for (x = 0; x < width - 1; x += 2) {
>@@ -1946,19 +2314,19 @@ void UYVYToYRow_C(const uint8* src_uyvy, uint8* dst_y, int width) {
> // Blend src_argb0 over src_argb1 and store to dst_argb.
> // dst_argb may be src_argb0 or src_argb1.
> // This code mimics the SSSE3 version for better testability.
>-void ARGBBlendRow_C(const uint8* src_argb0,
>-                    const uint8* src_argb1,
>-                    uint8* dst_argb,
>+void ARGBBlendRow_C(const uint8_t* src_argb0,
>+                    const uint8_t* src_argb1,
>+                    uint8_t* dst_argb,
>                     int width) {
>   int x;
>   for (x = 0; x < width - 1; x += 2) {
>-    uint32 fb = src_argb0[0];
>-    uint32 fg = src_argb0[1];
>-    uint32 fr = src_argb0[2];
>-    uint32 a = src_argb0[3];
>-    uint32 bb = src_argb1[0];
>-    uint32 bg = src_argb1[1];
>-    uint32 br = src_argb1[2];
>+    uint32_t fb = src_argb0[0];
>+    uint32_t fg = src_argb0[1];
>+    uint32_t fr = src_argb0[2];
>+    uint32_t a = src_argb0[3];
>+    uint32_t bb = src_argb1[0];
>+    uint32_t bg = src_argb1[1];
>+    uint32_t br = src_argb1[2];
>     dst_argb[0] = BLEND(fb, bb, a);
>     dst_argb[1] = BLEND(fg, bg, a);
>     dst_argb[2] = BLEND(fr, br, a);
>@@ -1981,13 +2349,13 @@ void ARGBBlendRow_C(const uint8* src_argb0,
>   }
> 
>   if (width & 1) {
>-    uint32 fb = src_argb0[0];
>-    uint32 fg = src_argb0[1];
>-    uint32 fr = src_argb0[2];
>-    uint32 a = src_argb0[3];
>-    uint32 bb = src_argb1[0];
>-    uint32 bg = src_argb1[1];
>-    uint32 br = src_argb1[2];
>+    uint32_t fb = src_argb0[0];
>+    uint32_t fg = src_argb0[1];
>+    uint32_t fr = src_argb0[2];
>+    uint32_t a = src_argb0[3];
>+    uint32_t bb = src_argb1[0];
>+    uint32_t bg = src_argb1[1];
>+    uint32_t br = src_argb1[2];
>     dst_argb[0] = BLEND(fb, bb, a);
>     dst_argb[1] = BLEND(fg, bg, a);
>     dst_argb[2] = BLEND(fr, br, a);
>@@ -1997,10 +2365,10 @@ void ARGBBlendRow_C(const uint8* src_argb0,
> #undef BLEND
> 
> #define UBLEND(f, b, a) (((a)*f) + ((255 - a) * b) + 255) >> 8
>-void BlendPlaneRow_C(const uint8* src0,
>-                     const uint8* src1,
>-                     const uint8* alpha,
>-                     uint8* dst,
>+void BlendPlaneRow_C(const uint8_t* src0,
>+                     const uint8_t* src1,
>+                     const uint8_t* alpha,
>+                     uint8_t* dst,
>                      int width) {
>   int x;
>   for (x = 0; x < width - 1; x += 2) {
>@@ -2021,13 +2389,13 @@ void BlendPlaneRow_C(const uint8* src0,
> 
> // Multiply source RGB by alpha and store to destination.
> // This code mimics the SSSE3 version for better testability.
>-void ARGBAttenuateRow_C(const uint8* src_argb, uint8* dst_argb, int width) {
>+void ARGBAttenuateRow_C(const uint8_t* src_argb, uint8_t* dst_argb, int width) {
>   int i;
>   for (i = 0; i < width - 1; i += 2) {
>-    uint32 b = src_argb[0];
>-    uint32 g = src_argb[1];
>-    uint32 r = src_argb[2];
>-    uint32 a = src_argb[3];
>+    uint32_t b = src_argb[0];
>+    uint32_t g = src_argb[1];
>+    uint32_t r = src_argb[2];
>+    uint32_t a = src_argb[3];
>     dst_argb[0] = ATTENUATE(b, a);
>     dst_argb[1] = ATTENUATE(g, a);
>     dst_argb[2] = ATTENUATE(r, a);
>@@ -2045,10 +2413,10 @@ void ARGBAttenuateRow_C(const uint8* src_argb, uint8* dst_argb, int width) {
>   }
> 
>   if (width & 1) {
>-    const uint32 b = src_argb[0];
>-    const uint32 g = src_argb[1];
>-    const uint32 r = src_argb[2];
>-    const uint32 a = src_argb[3];
>+    const uint32_t b = src_argb[0];
>+    const uint32_t g = src_argb[1];
>+    const uint32_t r = src_argb[2];
>+    const uint32_t a = src_argb[3];
>     dst_argb[0] = ATTENUATE(b, a);
>     dst_argb[1] = ATTENUATE(g, a);
>     dst_argb[2] = ATTENUATE(r, a);
>@@ -2064,7 +2432,7 @@ void ARGBAttenuateRow_C(const uint8* src_argb, uint8* dst_argb, int width) {
> // Reciprocal method is off by 1 on some values. ie 125
> // 8.8 fixed point inverse table with 1.0 in upper short and 1 / a in lower.
> #define T(a) 0x01000000 + (0x10000 / a)
>-const uint32 fixed_invtbl8[256] = {
>+const uint32_t fixed_invtbl8[256] = {
>     0x01000000, 0x0100ffff, T(0x02), T(0x03),   T(0x04), T(0x05), T(0x06),
>     T(0x07),    T(0x08),    T(0x09), T(0x0a),   T(0x0b), T(0x0c), T(0x0d),
>     T(0x0e),    T(0x0f),    T(0x10), T(0x11),   T(0x12), T(0x13), T(0x14),
>@@ -2104,14 +2472,16 @@ const uint32 fixed_invtbl8[256] = {
>     T(0xfc),    T(0xfd),    T(0xfe), 0x01000100};
> #undef T
> 
>-void ARGBUnattenuateRow_C(const uint8* src_argb, uint8* dst_argb, int width) {
>+void ARGBUnattenuateRow_C(const uint8_t* src_argb,
>+                          uint8_t* dst_argb,
>+                          int width) {
>   int i;
>   for (i = 0; i < width; ++i) {
>-    uint32 b = src_argb[0];
>-    uint32 g = src_argb[1];
>-    uint32 r = src_argb[2];
>-    const uint32 a = src_argb[3];
>-    const uint32 ia = fixed_invtbl8[a] & 0xffff;  // 8.8 fixed point
>+    uint32_t b = src_argb[0];
>+    uint32_t g = src_argb[1];
>+    uint32_t r = src_argb[2];
>+    const uint32_t a = src_argb[3];
>+    const uint32_t ia = fixed_invtbl8[a] & 0xffff;  // 8.8 fixed point
>     b = (b * ia) >> 8;
>     g = (g * ia) >> 8;
>     r = (r * ia) >> 8;
>@@ -2125,11 +2495,11 @@ void ARGBUnattenuateRow_C(const uint8* src_argb, uint8* dst_argb, int width) {
>   }
> }
> 
>-void ComputeCumulativeSumRow_C(const uint8* row,
>-                               int32* cumsum,
>-                               const int32* previous_cumsum,
>+void ComputeCumulativeSumRow_C(const uint8_t* row,
>+                               int32_t* cumsum,
>+                               const int32_t* previous_cumsum,
>                                int width) {
>-  int32 row_sum[4] = {0, 0, 0, 0};
>+  int32_t row_sum[4] = {0, 0, 0, 0};
>   int x;
>   for (x = 0; x < width; ++x) {
>     row_sum[0] += row[x * 4 + 0];
>@@ -2143,19 +2513,19 @@ void ComputeCumulativeSumRow_C(const uint8* row,
>   }
> }
> 
>-void CumulativeSumToAverageRow_C(const int32* tl,
>-                                 const int32* bl,
>+void CumulativeSumToAverageRow_C(const int32_t* tl,
>+                                 const int32_t* bl,
>                                  int w,
>                                  int area,
>-                                 uint8* dst,
>+                                 uint8_t* dst,
>                                  int count) {
>   float ooa = 1.0f / area;
>   int i;
>   for (i = 0; i < count; ++i) {
>-    dst[0] = (uint8)((bl[w + 0] + tl[0] - bl[0] - tl[w + 0]) * ooa);
>-    dst[1] = (uint8)((bl[w + 1] + tl[1] - bl[1] - tl[w + 1]) * ooa);
>-    dst[2] = (uint8)((bl[w + 2] + tl[2] - bl[2] - tl[w + 2]) * ooa);
>-    dst[3] = (uint8)((bl[w + 3] + tl[3] - bl[3] - tl[w + 3]) * ooa);
>+    dst[0] = (uint8_t)((bl[w + 0] + tl[0] - bl[0] - tl[w + 0]) * ooa);
>+    dst[1] = (uint8_t)((bl[w + 1] + tl[1] - bl[1] - tl[w + 1]) * ooa);
>+    dst[2] = (uint8_t)((bl[w + 2] + tl[2] - bl[2] - tl[w + 2]) * ooa);
>+    dst[3] = (uint8_t)((bl[w + 3] + tl[3] - bl[3] - tl[w + 3]) * ooa);
>     dst += 4;
>     tl += 4;
>     bl += 4;
>@@ -2164,9 +2534,9 @@ void CumulativeSumToAverageRow_C(const int32* tl,
> 
> // Copy pixels from rotated source to destination row with a slope.
> LIBYUV_API
>-void ARGBAffineRow_C(const uint8* src_argb,
>+void ARGBAffineRow_C(const uint8_t* src_argb,
>                      int src_argb_stride,
>-                     uint8* dst_argb,
>+                     uint8_t* dst_argb,
>                      const float* uv_dudv,
>                      int width) {
>   int i;
>@@ -2177,8 +2547,8 @@ void ARGBAffineRow_C(const uint8* src_argb,
>   for (i = 0; i < width; ++i) {
>     int x = (int)(uv[0]);
>     int y = (int)(uv[1]);
>-    *(uint32*)(dst_argb) =
>-        *(const uint32*)(src_argb + y * src_argb_stride + x * 4);
>+    *(uint32_t*)(dst_argb) =
>+        *(const uint32_t*)(src_argb + y * src_argb_stride + x * 4);
>     dst_argb += 4;
>     uv[0] += uv_dudv[2];
>     uv[1] += uv_dudv[3];
>@@ -2186,9 +2556,9 @@ void ARGBAffineRow_C(const uint8* src_argb,
> }
> 
> // Blend 2 rows into 1.
>-static void HalfRow_C(const uint8* src_uv,
>+static void HalfRow_C(const uint8_t* src_uv,
>                       ptrdiff_t src_uv_stride,
>-                      uint8* dst_uv,
>+                      uint8_t* dst_uv,
>                       int width) {
>   int x;
>   for (x = 0; x < width; ++x) {
>@@ -2196,9 +2566,9 @@ static void HalfRow_C(const uint8* src_uv,
>   }
> }
> 
>-static void HalfRow_16_C(const uint16* src_uv,
>+static void HalfRow_16_C(const uint16_t* src_uv,
>                          ptrdiff_t src_uv_stride,
>-                         uint16* dst_uv,
>+                         uint16_t* dst_uv,
>                          int width) {
>   int x;
>   for (x = 0; x < width; ++x) {
>@@ -2207,14 +2577,14 @@ static void HalfRow_16_C(const uint16* src_uv,
> }
> 
> // C version 2x2 -> 2x1.
>-void InterpolateRow_C(uint8* dst_ptr,
>-                      const uint8* src_ptr,
>+void InterpolateRow_C(uint8_t* dst_ptr,
>+                      const uint8_t* src_ptr,
>                       ptrdiff_t src_stride,
>                       int width,
>                       int source_y_fraction) {
>   int y1_fraction = source_y_fraction;
>   int y0_fraction = 256 - y1_fraction;
>-  const uint8* src_ptr1 = src_ptr + src_stride;
>+  const uint8_t* src_ptr1 = src_ptr + src_stride;
>   int x;
>   if (y1_fraction == 0) {
>     memcpy(dst_ptr, src_ptr, width);
>@@ -2239,14 +2609,14 @@ void InterpolateRow_C(uint8* dst_ptr,
>   }
> }
> 
>-void InterpolateRow_16_C(uint16* dst_ptr,
>-                         const uint16* src_ptr,
>+void InterpolateRow_16_C(uint16_t* dst_ptr,
>+                         const uint16_t* src_ptr,
>                          ptrdiff_t src_stride,
>                          int width,
>                          int source_y_fraction) {
>   int y1_fraction = source_y_fraction;
>   int y0_fraction = 256 - y1_fraction;
>-  const uint16* src_ptr1 = src_ptr + src_stride;
>+  const uint16_t* src_ptr1 = src_ptr + src_stride;
>   int x;
>   if (source_y_fraction == 0) {
>     memcpy(dst_ptr, src_ptr, width * 2);
>@@ -2269,9 +2639,9 @@ void InterpolateRow_16_C(uint16* dst_ptr,
> }
> 
> // Use first 4 shuffler values to reorder ARGB channels.
>-void ARGBShuffleRow_C(const uint8* src_argb,
>-                      uint8* dst_argb,
>-                      const uint8* shuffler,
>+void ARGBShuffleRow_C(const uint8_t* src_argb,
>+                      uint8_t* dst_argb,
>+                      const uint8_t* shuffler,
>                       int width) {
>   int index0 = shuffler[0];
>   int index1 = shuffler[1];
>@@ -2281,10 +2651,10 @@ void ARGBShuffleRow_C(const uint8* src_argb,
>   int x;
>   for (x = 0; x < width; ++x) {
>     // To support in-place conversion.
>-    uint8 b = src_argb[index0];
>-    uint8 g = src_argb[index1];
>-    uint8 r = src_argb[index2];
>-    uint8 a = src_argb[index3];
>+    uint8_t b = src_argb[index0];
>+    uint8_t g = src_argb[index1];
>+    uint8_t r = src_argb[index2];
>+    uint8_t a = src_argb[index3];
>     dst_argb[0] = b;
>     dst_argb[1] = g;
>     dst_argb[2] = r;
>@@ -2294,10 +2664,10 @@ void ARGBShuffleRow_C(const uint8* src_argb,
>   }
> }
> 
>-void I422ToYUY2Row_C(const uint8* src_y,
>-                     const uint8* src_u,
>-                     const uint8* src_v,
>-                     uint8* dst_frame,
>+void I422ToYUY2Row_C(const uint8_t* src_y,
>+                     const uint8_t* src_u,
>+                     const uint8_t* src_v,
>+                     uint8_t* dst_frame,
>                      int width) {
>   int x;
>   for (x = 0; x < width - 1; x += 2) {
>@@ -2318,10 +2688,10 @@ void I422ToYUY2Row_C(const uint8* src_y,
>   }
> }
> 
>-void I422ToUYVYRow_C(const uint8* src_y,
>-                     const uint8* src_u,
>-                     const uint8* src_v,
>-                     uint8* dst_frame,
>+void I422ToUYVYRow_C(const uint8_t* src_y,
>+                     const uint8_t* src_u,
>+                     const uint8_t* src_v,
>+                     uint8_t* dst_frame,
>                      int width) {
>   int x;
>   for (x = 0; x < width - 1; x += 2) {
>@@ -2342,8 +2712,8 @@ void I422ToUYVYRow_C(const uint8* src_y,
>   }
> }
> 
>-void ARGBPolynomialRow_C(const uint8* src_argb,
>-                         uint8* dst_argb,
>+void ARGBPolynomialRow_C(const uint8_t* src_argb,
>+                         uint8_t* dst_argb,
>                          const float* poly,
>                          int width) {
>   int i;
>@@ -2373,10 +2743,10 @@ void ARGBPolynomialRow_C(const uint8* src_argb,
>     dr += poly[14] * r3;
>     da += poly[15] * a3;
> 
>-    dst_argb[0] = Clamp((int32)(db));
>-    dst_argb[1] = Clamp((int32)(dg));
>-    dst_argb[2] = Clamp((int32)(dr));
>-    dst_argb[3] = Clamp((int32)(da));
>+    dst_argb[0] = Clamp((int32_t)(db));
>+    dst_argb[1] = Clamp((int32_t)(dg));
>+    dst_argb[2] = Clamp((int32_t)(dr));
>+    dst_argb[3] = Clamp((int32_t)(da));
>     src_argb += 4;
>     dst_argb += 4;
>   }
>@@ -2392,31 +2762,42 @@ void ARGBPolynomialRow_C(const uint8* src_argb,
> // simply extract the low bits of the exponent and the high
> // bits of the mantissa from our float and we're done.
> 
>-void HalfFloatRow_C(const uint16* src, uint16* dst, float scale, int width) {
>+void HalfFloatRow_C(const uint16_t* src,
>+                    uint16_t* dst,
>+                    float scale,
>+                    int width) {
>   int i;
>   float mult = 1.9259299444e-34f * scale;
>   for (i = 0; i < width; ++i) {
>     float value = src[i] * mult;
>-    dst[i] = (uint16)((*(uint32_t*)&value) >> 13);
>+    dst[i] = (uint16_t)((*(uint32_t*)&value) >> 13);
>+  }
>+}
>+
>+void ByteToFloatRow_C(const uint8_t* src, float* dst, float scale, int width) {
>+  int i;
>+  for (i = 0; i < width; ++i) {
>+    float value = src[i] * scale;
>+    dst[i] = value;
>   }
> }
> 
>-void ARGBLumaColorTableRow_C(const uint8* src_argb,
>-                             uint8* dst_argb,
>+void ARGBLumaColorTableRow_C(const uint8_t* src_argb,
>+                             uint8_t* dst_argb,
>                              int width,
>-                             const uint8* luma,
>-                             uint32 lumacoeff) {
>-  uint32 bc = lumacoeff & 0xff;
>-  uint32 gc = (lumacoeff >> 8) & 0xff;
>-  uint32 rc = (lumacoeff >> 16) & 0xff;
>+                             const uint8_t* luma,
>+                             uint32_t lumacoeff) {
>+  uint32_t bc = lumacoeff & 0xff;
>+  uint32_t gc = (lumacoeff >> 8) & 0xff;
>+  uint32_t rc = (lumacoeff >> 16) & 0xff;
> 
>   int i;
>   for (i = 0; i < width - 1; i += 2) {
>     // Luminance in rows, color values in columns.
>-    const uint8* luma0 =
>+    const uint8_t* luma0 =
>         ((src_argb[0] * bc + src_argb[1] * gc + src_argb[2] * rc) & 0x7F00u) +
>         luma;
>-    const uint8* luma1;
>+    const uint8_t* luma1;
>     dst_argb[0] = luma0[src_argb[0]];
>     dst_argb[1] = luma0[src_argb[1]];
>     dst_argb[2] = luma0[src_argb[2]];
>@@ -2433,7 +2814,7 @@ void ARGBLumaColorTableRow_C(const uint8* src_argb,
>   }
>   if (width & 1) {
>     // Luminance in rows, color values in columns.
>-    const uint8* luma0 =
>+    const uint8_t* luma0 =
>         ((src_argb[0] * bc + src_argb[1] * gc + src_argb[2] * rc) & 0x7F00u) +
>         luma;
>     dst_argb[0] = luma0[src_argb[0]];
>@@ -2443,7 +2824,7 @@ void ARGBLumaColorTableRow_C(const uint8* src_argb,
>   }
> }
> 
>-void ARGBCopyAlphaRow_C(const uint8* src, uint8* dst, int width) {
>+void ARGBCopyAlphaRow_C(const uint8_t* src, uint8_t* dst, int width) {
>   int i;
>   for (i = 0; i < width - 1; i += 2) {
>     dst[3] = src[3];
>@@ -2456,7 +2837,7 @@ void ARGBCopyAlphaRow_C(const uint8* src, uint8* dst, int width) {
>   }
> }
> 
>-void ARGBExtractAlphaRow_C(const uint8* src_argb, uint8* dst_a, int width) {
>+void ARGBExtractAlphaRow_C(const uint8_t* src_argb, uint8_t* dst_a, int width) {
>   int i;
>   for (i = 0; i < width - 1; i += 2) {
>     dst_a[0] = src_argb[3];
>@@ -2469,7 +2850,7 @@ void ARGBExtractAlphaRow_C(const uint8* src_argb, uint8* dst_a, int width) {
>   }
> }
> 
>-void ARGBCopyYToAlphaRow_C(const uint8* src, uint8* dst, int width) {
>+void ARGBCopyYToAlphaRow_C(const uint8_t* src, uint8_t* dst, int width) {
>   int i;
>   for (i = 0; i < width - 1; i += 2) {
>     dst[3] = src[0];
>@@ -2488,13 +2869,13 @@ void ARGBCopyYToAlphaRow_C(const uint8* src, uint8* dst, int width) {
> #if !(defined(_MSC_VER) && defined(_M_IX86)) && \
>     defined(HAS_I422TORGB565ROW_SSSE3)
> // row_win.cc has asm version, but GCC uses 2 step wrapper.
>-void I422ToRGB565Row_SSSE3(const uint8* src_y,
>-                           const uint8* src_u,
>-                           const uint8* src_v,
>-                           uint8* dst_rgb565,
>+void I422ToRGB565Row_SSSE3(const uint8_t* src_y,
>+                           const uint8_t* src_u,
>+                           const uint8_t* src_v,
>+                           uint8_t* dst_rgb565,
>                            const struct YuvConstants* yuvconstants,
>                            int width) {
>-  SIMD_ALIGNED(uint8 row[MAXTWIDTH * 4]);
>+  SIMD_ALIGNED(uint8_t row[MAXTWIDTH * 4]);
>   while (width > 0) {
>     int twidth = width > MAXTWIDTH ? MAXTWIDTH : width;
>     I422ToARGBRow_SSSE3(src_y, src_u, src_v, row, yuvconstants, twidth);
>@@ -2509,14 +2890,14 @@ void I422ToRGB565Row_SSSE3(const uint8* src_y,
> #endif
> 
> #if defined(HAS_I422TOARGB1555ROW_SSSE3)
>-void I422ToARGB1555Row_SSSE3(const uint8* src_y,
>-                             const uint8* src_u,
>-                             const uint8* src_v,
>-                             uint8* dst_argb1555,
>+void I422ToARGB1555Row_SSSE3(const uint8_t* src_y,
>+                             const uint8_t* src_u,
>+                             const uint8_t* src_v,
>+                             uint8_t* dst_argb1555,
>                              const struct YuvConstants* yuvconstants,
>                              int width) {
>   // Row buffer for intermediate ARGB pixels.
>-  SIMD_ALIGNED(uint8 row[MAXTWIDTH * 4]);
>+  SIMD_ALIGNED(uint8_t row[MAXTWIDTH * 4]);
>   while (width > 0) {
>     int twidth = width > MAXTWIDTH ? MAXTWIDTH : width;
>     I422ToARGBRow_SSSE3(src_y, src_u, src_v, row, yuvconstants, twidth);
>@@ -2531,14 +2912,14 @@ void I422ToARGB1555Row_SSSE3(const uint8* src_y,
> #endif
> 
> #if defined(HAS_I422TOARGB4444ROW_SSSE3)
>-void I422ToARGB4444Row_SSSE3(const uint8* src_y,
>-                             const uint8* src_u,
>-                             const uint8* src_v,
>-                             uint8* dst_argb4444,
>+void I422ToARGB4444Row_SSSE3(const uint8_t* src_y,
>+                             const uint8_t* src_u,
>+                             const uint8_t* src_v,
>+                             uint8_t* dst_argb4444,
>                              const struct YuvConstants* yuvconstants,
>                              int width) {
>   // Row buffer for intermediate ARGB pixels.
>-  SIMD_ALIGNED(uint8 row[MAXTWIDTH * 4]);
>+  SIMD_ALIGNED(uint8_t row[MAXTWIDTH * 4]);
>   while (width > 0) {
>     int twidth = width > MAXTWIDTH ? MAXTWIDTH : width;
>     I422ToARGBRow_SSSE3(src_y, src_u, src_v, row, yuvconstants, twidth);
>@@ -2553,13 +2934,13 @@ void I422ToARGB4444Row_SSSE3(const uint8* src_y,
> #endif
> 
> #if defined(HAS_NV12TORGB565ROW_SSSE3)
>-void NV12ToRGB565Row_SSSE3(const uint8* src_y,
>-                           const uint8* src_uv,
>-                           uint8* dst_rgb565,
>+void NV12ToRGB565Row_SSSE3(const uint8_t* src_y,
>+                           const uint8_t* src_uv,
>+                           uint8_t* dst_rgb565,
>                            const struct YuvConstants* yuvconstants,
>                            int width) {
>   // Row buffer for intermediate ARGB pixels.
>-  SIMD_ALIGNED(uint8 row[MAXTWIDTH * 4]);
>+  SIMD_ALIGNED(uint8_t row[MAXTWIDTH * 4]);
>   while (width > 0) {
>     int twidth = width > MAXTWIDTH ? MAXTWIDTH : width;
>     NV12ToARGBRow_SSSE3(src_y, src_uv, row, yuvconstants, twidth);
>@@ -2573,13 +2954,13 @@ void NV12ToRGB565Row_SSSE3(const uint8* src_y,
> #endif
> 
> #if defined(HAS_I422TORGB565ROW_AVX2)
>-void I422ToRGB565Row_AVX2(const uint8* src_y,
>-                          const uint8* src_u,
>-                          const uint8* src_v,
>-                          uint8* dst_rgb565,
>+void I422ToRGB565Row_AVX2(const uint8_t* src_y,
>+                          const uint8_t* src_u,
>+                          const uint8_t* src_v,
>+                          uint8_t* dst_rgb565,
>                           const struct YuvConstants* yuvconstants,
>                           int width) {
>-  SIMD_ALIGNED(uint8 row[MAXTWIDTH * 4]);
>+  SIMD_ALIGNED(uint8_t row[MAXTWIDTH * 4]);
>   while (width > 0) {
>     int twidth = width > MAXTWIDTH ? MAXTWIDTH : width;
>     I422ToARGBRow_AVX2(src_y, src_u, src_v, row, yuvconstants, twidth);
>@@ -2598,14 +2979,14 @@ void I422ToRGB565Row_AVX2(const uint8* src_y,
> #endif
> 
> #if defined(HAS_I422TOARGB1555ROW_AVX2)
>-void I422ToARGB1555Row_AVX2(const uint8* src_y,
>-                            const uint8* src_u,
>-                            const uint8* src_v,
>-                            uint8* dst_argb1555,
>+void I422ToARGB1555Row_AVX2(const uint8_t* src_y,
>+                            const uint8_t* src_u,
>+                            const uint8_t* src_v,
>+                            uint8_t* dst_argb1555,
>                             const struct YuvConstants* yuvconstants,
>                             int width) {
>   // Row buffer for intermediate ARGB pixels.
>-  SIMD_ALIGNED(uint8 row[MAXTWIDTH * 4]);
>+  SIMD_ALIGNED(uint8_t row[MAXTWIDTH * 4]);
>   while (width > 0) {
>     int twidth = width > MAXTWIDTH ? MAXTWIDTH : width;
>     I422ToARGBRow_AVX2(src_y, src_u, src_v, row, yuvconstants, twidth);
>@@ -2624,14 +3005,14 @@ void I422ToARGB1555Row_AVX2(const uint8* src_y,
> #endif
> 
> #if defined(HAS_I422TOARGB4444ROW_AVX2)
>-void I422ToARGB4444Row_AVX2(const uint8* src_y,
>-                            const uint8* src_u,
>-                            const uint8* src_v,
>-                            uint8* dst_argb4444,
>+void I422ToARGB4444Row_AVX2(const uint8_t* src_y,
>+                            const uint8_t* src_u,
>+                            const uint8_t* src_v,
>+                            uint8_t* dst_argb4444,
>                             const struct YuvConstants* yuvconstants,
>                             int width) {
>   // Row buffer for intermediate ARGB pixels.
>-  SIMD_ALIGNED(uint8 row[MAXTWIDTH * 4]);
>+  SIMD_ALIGNED(uint8_t row[MAXTWIDTH * 4]);
>   while (width > 0) {
>     int twidth = width > MAXTWIDTH ? MAXTWIDTH : width;
>     I422ToARGBRow_AVX2(src_y, src_u, src_v, row, yuvconstants, twidth);
>@@ -2650,14 +3031,14 @@ void I422ToARGB4444Row_AVX2(const uint8* src_y,
> #endif
> 
> #if defined(HAS_I422TORGB24ROW_AVX2)
>-void I422ToRGB24Row_AVX2(const uint8* src_y,
>-                         const uint8* src_u,
>-                         const uint8* src_v,
>-                         uint8* dst_rgb24,
>+void I422ToRGB24Row_AVX2(const uint8_t* src_y,
>+                         const uint8_t* src_u,
>+                         const uint8_t* src_v,
>+                         uint8_t* dst_rgb24,
>                          const struct YuvConstants* yuvconstants,
>                          int width) {
>   // Row buffer for intermediate ARGB pixels.
>-  SIMD_ALIGNED(uint8 row[MAXTWIDTH * 4]);
>+  SIMD_ALIGNED(uint8_t row[MAXTWIDTH * 4]);
>   while (width > 0) {
>     int twidth = width > MAXTWIDTH ? MAXTWIDTH : width;
>     I422ToARGBRow_AVX2(src_y, src_u, src_v, row, yuvconstants, twidth);
>@@ -2673,13 +3054,13 @@ void I422ToRGB24Row_AVX2(const uint8* src_y,
> #endif
> 
> #if defined(HAS_NV12TORGB565ROW_AVX2)
>-void NV12ToRGB565Row_AVX2(const uint8* src_y,
>-                          const uint8* src_uv,
>-                          uint8* dst_rgb565,
>+void NV12ToRGB565Row_AVX2(const uint8_t* src_y,
>+                          const uint8_t* src_uv,
>+                          uint8_t* dst_rgb565,
>                           const struct YuvConstants* yuvconstants,
>                           int width) {
>   // Row buffer for intermediate ARGB pixels.
>-  SIMD_ALIGNED(uint8 row[MAXTWIDTH * 4]);
>+  SIMD_ALIGNED(uint8_t row[MAXTWIDTH * 4]);
>   while (width > 0) {
>     int twidth = width > MAXTWIDTH ? MAXTWIDTH : width;
>     NV12ToARGBRow_AVX2(src_y, src_uv, row, yuvconstants, twidth);
>@@ -2729,7 +3110,7 @@ void ScaleSamples_C(const float* src, float* dst, float scale, int width) {
>   }
> }
> 
>-void GaussRow_C(const uint32* src, uint16* dst, int width) {
>+void GaussRow_C(const uint32_t* src, uint16_t* dst, int width) {
>   int i;
>   for (i = 0; i < width; ++i) {
>     *dst++ =
>@@ -2739,12 +3120,12 @@ void GaussRow_C(const uint32* src, uint16* dst, int width) {
> }
> 
> // filter 5 rows with 1, 4, 6, 4, 1 coefficients to produce 1 row.
>-void GaussCol_C(const uint16* src0,
>-                const uint16* src1,
>-                const uint16* src2,
>-                const uint16* src3,
>-                const uint16* src4,
>-                uint32* dst,
>+void GaussCol_C(const uint16_t* src0,
>+                const uint16_t* src1,
>+                const uint16_t* src2,
>+                const uint16_t* src3,
>+                const uint16_t* src4,
>+                uint32_t* dst,
>                 int width) {
>   int i;
>   for (i = 0; i < width; ++i) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/row_dspr2.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/row_dspr2.cc
>deleted file mode 100644
>index 11f78e0d2f1..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/row_dspr2.cc
>+++ /dev/null
>@@ -1,1721 +0,0 @@
>-/*
>- *  Copyright (c) 2012 The LibYuv project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS. All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "libyuv/row.h"
>-
>-#ifdef __cplusplus
>-namespace libyuv {
>-extern "C" {
>-#endif
>-
>-// The following are available on Mips platforms:
>-#if !defined(LIBYUV_DISABLE_DSPR2) && defined(__mips__) && \
>-    (_MIPS_SIM == _MIPS_SIM_ABI32)
>-
>-#ifdef HAS_COPYROW_MIPS
>-void CopyRow_MIPS(const uint8* src, uint8* dst, int count) {
>-  __asm__ __volatile__(
>-      ".set      noreorder                         \n"
>-      ".set      noat                              \n"
>-      "slti      $at, %[count], 8                  \n"
>-      "bne       $at ,$zero, $last8                \n"
>-      "xor       $t8, %[src], %[dst]               \n"
>-      "andi      $t8, $t8, 0x3                     \n"
>-
>-      "bne       $t8, $zero, unaligned             \n"
>-      "negu      $a3, %[dst]                       \n"
>-      // make dst/src aligned
>-      "andi      $a3, $a3, 0x3                     \n"
>-      "beq       $a3, $zero, $chk16w               \n"
>-      // word-aligned now count is the remining bytes count
>-      "subu     %[count], %[count], $a3            \n"
>-
>-      "lwr       $t8, 0(%[src])                    \n"
>-      "addu      %[src], %[src], $a3               \n"
>-      "swr       $t8, 0(%[dst])                    \n"
>-      "addu      %[dst], %[dst], $a3               \n"
>-
>-      // Now the dst/src are mutually word-aligned with word-aligned addresses
>-      "$chk16w:                                    \n"
>-      "andi      $t8, %[count], 0x3f               \n"  // whole 64-B chunks?
>-      // t8 is the byte count after 64-byte chunks
>-      "beq       %[count], $t8, chk8w              \n"
>-      // There will be at most 1 32-byte chunk after it
>-      "subu      $a3, %[count], $t8                \n"  // the reminder
>-      // Here a3 counts bytes in 16w chunks
>-      "addu      $a3, %[dst], $a3                  \n"
>-      // Now a3 is the final dst after 64-byte chunks
>-      "addu      $t0, %[dst], %[count]             \n"
>-      // t0 is the "past the end" address
>-
>-      // When in the loop we exercise "pref 30,x(a1)", the a1+x should not be
>-      // past
>-      // the "t0-32" address
>-      // This means: for x=128 the last "safe" a1 address is "t0-160"
>-      // Alternatively, for x=64 the last "safe" a1 address is "t0-96"
>-      // we will use "pref 30,128(a1)", so "t0-160" is the limit
>-      "subu      $t9, $t0, 160                     \n"
>-      // t9 is the "last safe pref 30,128(a1)" address
>-      "pref      0, 0(%[src])                      \n"  // first line of src
>-      "pref      0, 32(%[src])                     \n"  // second line of src
>-      "pref      0, 64(%[src])                     \n"
>-      "pref      30, 32(%[dst])                    \n"
>-      // In case the a1 > t9 don't use "pref 30" at all
>-      "sltu      $v1, $t9, %[dst]                  \n"
>-      "bgtz      $v1, $loop16w                     \n"
>-      "nop                                         \n"
>-      // otherwise, start with using pref30
>-      "pref      30, 64(%[dst])                    \n"
>-      "$loop16w:                                    \n"
>-      "pref      0, 96(%[src])                     \n"
>-      "lw        $t0, 0(%[src])                    \n"
>-      "bgtz      $v1, $skip_pref30_96              \n"  // skip
>-      "lw        $t1, 4(%[src])                    \n"
>-      "pref      30, 96(%[dst])                    \n"  // continue
>-      "$skip_pref30_96:                            \n"
>-      "lw        $t2, 8(%[src])                    \n"
>-      "lw        $t3, 12(%[src])                   \n"
>-      "lw        $t4, 16(%[src])                   \n"
>-      "lw        $t5, 20(%[src])                   \n"
>-      "lw        $t6, 24(%[src])                   \n"
>-      "lw        $t7, 28(%[src])                   \n"
>-      "pref      0, 128(%[src])                    \n"
>-      //  bring the next lines of src, addr 128
>-      "sw        $t0, 0(%[dst])                    \n"
>-      "sw        $t1, 4(%[dst])                    \n"
>-      "sw        $t2, 8(%[dst])                    \n"
>-      "sw        $t3, 12(%[dst])                   \n"
>-      "sw        $t4, 16(%[dst])                   \n"
>-      "sw        $t5, 20(%[dst])                   \n"
>-      "sw        $t6, 24(%[dst])                   \n"
>-      "sw        $t7, 28(%[dst])                   \n"
>-      "lw        $t0, 32(%[src])                   \n"
>-      "bgtz      $v1, $skip_pref30_128             \n"  // skip pref 30,128(a1)
>-      "lw        $t1, 36(%[src])                   \n"
>-      "pref      30, 128(%[dst])                   \n"  // set dest, addr 128
>-      "$skip_pref30_128:                           \n"
>-      "lw        $t2, 40(%[src])                   \n"
>-      "lw        $t3, 44(%[src])                   \n"
>-      "lw        $t4, 48(%[src])                   \n"
>-      "lw        $t5, 52(%[src])                   \n"
>-      "lw        $t6, 56(%[src])                   \n"
>-      "lw        $t7, 60(%[src])                   \n"
>-      "pref      0, 160(%[src])                    \n"
>-      // bring the next lines of src, addr 160
>-      "sw        $t0, 32(%[dst])                   \n"
>-      "sw        $t1, 36(%[dst])                   \n"
>-      "sw        $t2, 40(%[dst])                   \n"
>-      "sw        $t3, 44(%[dst])                   \n"
>-      "sw        $t4, 48(%[dst])                   \n"
>-      "sw        $t5, 52(%[dst])                   \n"
>-      "sw        $t6, 56(%[dst])                   \n"
>-      "sw        $t7, 60(%[dst])                   \n"
>-
>-      "addiu     %[dst], %[dst], 64                \n"  // adding 64 to dest
>-      "sltu      $v1, $t9, %[dst]                  \n"
>-      "bne       %[dst], $a3, $loop16w             \n"
>-      " addiu    %[src], %[src], 64                \n"  // adding 64 to src
>-      "move      %[count], $t8                     \n"
>-
>-      // Here we have src and dest word-aligned but less than 64-bytes to go
>-
>-      "chk8w:                                      \n"
>-      "pref      0, 0x0(%[src])                    \n"
>-      "andi      $t8, %[count], 0x1f               \n"  // 32-byte chunk?
>-      // the t8 is the reminder count past 32-bytes
>-      "beq       %[count], $t8, chk1w              \n"
>-      // count=t8,no 32-byte chunk
>-      " nop                                        \n"
>-
>-      "lw        $t0, 0(%[src])                    \n"
>-      "lw        $t1, 4(%[src])                    \n"
>-      "lw        $t2, 8(%[src])                    \n"
>-      "lw        $t3, 12(%[src])                   \n"
>-      "lw        $t4, 16(%[src])                   \n"
>-      "lw        $t5, 20(%[src])                   \n"
>-      "lw        $t6, 24(%[src])                   \n"
>-      "lw        $t7, 28(%[src])                   \n"
>-      "addiu     %[src], %[src], 32                \n"
>-
>-      "sw        $t0, 0(%[dst])                    \n"
>-      "sw        $t1, 4(%[dst])                    \n"
>-      "sw        $t2, 8(%[dst])                    \n"
>-      "sw        $t3, 12(%[dst])                   \n"
>-      "sw        $t4, 16(%[dst])                   \n"
>-      "sw        $t5, 20(%[dst])                   \n"
>-      "sw        $t6, 24(%[dst])                   \n"
>-      "sw        $t7, 28(%[dst])                   \n"
>-      "addiu     %[dst], %[dst], 32                \n"
>-
>-      "chk1w:                                      \n"
>-      "andi      %[count], $t8, 0x3                \n"
>-      // now count is the reminder past 1w chunks
>-      "beq       %[count], $t8, $last8             \n"
>-      " subu     $a3, $t8, %[count]                \n"
>-      // a3 is count of bytes in 1w chunks
>-      "addu      $a3, %[dst], $a3                  \n"
>-      // now a3 is the dst address past the 1w chunks
>-      // copying in words (4-byte chunks)
>-      "$wordCopy_loop:                             \n"
>-      "lw        $t3, 0(%[src])                    \n"
>-      // the first t3 may be equal t0 ... optimize?
>-      "addiu     %[src], %[src],4                  \n"
>-      "addiu     %[dst], %[dst],4                  \n"
>-      "bne       %[dst], $a3,$wordCopy_loop        \n"
>-      " sw       $t3, -4(%[dst])                   \n"
>-
>-      // For the last (<8) bytes
>-      "$last8:                                     \n"
>-      "blez      %[count], leave                   \n"
>-      " addu     $a3, %[dst], %[count]             \n"  // a3 -last dst address
>-      "$last8loop:                                 \n"
>-      "lb        $v1, 0(%[src])                    \n"
>-      "addiu     %[src], %[src], 1                 \n"
>-      "addiu     %[dst], %[dst], 1                 \n"
>-      "bne       %[dst], $a3, $last8loop           \n"
>-      " sb       $v1, -1(%[dst])                   \n"
>-
>-      "leave:                                      \n"
>-      "  j       $ra                               \n"
>-      "  nop                                       \n"
>-
>-      //
>-      // UNALIGNED case
>-      //
>-
>-      "unaligned:                                  \n"
>-      // got here with a3="negu a1"
>-      "andi      $a3, $a3, 0x3                     \n"  // a1 is word aligned?
>-      "beqz      $a3, $ua_chk16w                   \n"
>-      " subu     %[count], %[count], $a3           \n"
>-      // bytes left after initial a3 bytes
>-      "lwr       $v1, 0(%[src])                    \n"
>-      "lwl       $v1, 3(%[src])                    \n"
>-      "addu      %[src], %[src], $a3               \n"  // a3 may be 1, 2 or 3
>-      "swr       $v1, 0(%[dst])                    \n"
>-      "addu      %[dst], %[dst], $a3               \n"
>-      // below the dst will be word aligned (NOTE1)
>-      "$ua_chk16w:                                 \n"
>-      "andi      $t8, %[count], 0x3f               \n"  // whole 64-B chunks?
>-      // t8 is the byte count after 64-byte chunks
>-      "beq       %[count], $t8, ua_chk8w           \n"
>-      // if a2==t8, no 64-byte chunks
>-      // There will be at most 1 32-byte chunk after it
>-      "subu      $a3, %[count], $t8                \n"  // the reminder
>-      // Here a3 counts bytes in 16w chunks
>-      "addu      $a3, %[dst], $a3                  \n"
>-      // Now a3 is the final dst after 64-byte chunks
>-      "addu      $t0, %[dst], %[count]             \n"  // t0 "past the end"
>-      "subu      $t9, $t0, 160                     \n"
>-      // t9 is the "last safe pref 30,128(a1)" address
>-      "pref      0, 0(%[src])                      \n"  // first line of src
>-      "pref      0, 32(%[src])                     \n"  // second line  addr 32
>-      "pref      0, 64(%[src])                     \n"
>-      "pref      30, 32(%[dst])                    \n"
>-      // safe, as we have at least 64 bytes ahead
>-      // In case the a1 > t9 don't use "pref 30" at all
>-      "sltu      $v1, $t9, %[dst]                  \n"
>-      "bgtz      $v1, $ua_loop16w                  \n"
>-      // skip "pref 30,64(a1)" for too short arrays
>-      " nop                                        \n"
>-      // otherwise, start with using pref30
>-      "pref      30, 64(%[dst])                    \n"
>-      "$ua_loop16w:                                \n"
>-      "pref      0, 96(%[src])                     \n"
>-      "lwr       $t0, 0(%[src])                    \n"
>-      "lwl       $t0, 3(%[src])                    \n"
>-      "lwr       $t1, 4(%[src])                    \n"
>-      "bgtz      $v1, $ua_skip_pref30_96           \n"
>-      " lwl      $t1, 7(%[src])                    \n"
>-      "pref      30, 96(%[dst])                    \n"
>-      // continue setting up the dest, addr 96
>-      "$ua_skip_pref30_96:                         \n"
>-      "lwr       $t2, 8(%[src])                    \n"
>-      "lwl       $t2, 11(%[src])                   \n"
>-      "lwr       $t3, 12(%[src])                   \n"
>-      "lwl       $t3, 15(%[src])                   \n"
>-      "lwr       $t4, 16(%[src])                   \n"
>-      "lwl       $t4, 19(%[src])                   \n"
>-      "lwr       $t5, 20(%[src])                   \n"
>-      "lwl       $t5, 23(%[src])                   \n"
>-      "lwr       $t6, 24(%[src])                   \n"
>-      "lwl       $t6, 27(%[src])                   \n"
>-      "lwr       $t7, 28(%[src])                   \n"
>-      "lwl       $t7, 31(%[src])                   \n"
>-      "pref      0, 128(%[src])                    \n"
>-      // bring the next lines of src, addr 128
>-      "sw        $t0, 0(%[dst])                    \n"
>-      "sw        $t1, 4(%[dst])                    \n"
>-      "sw        $t2, 8(%[dst])                    \n"
>-      "sw        $t3, 12(%[dst])                   \n"
>-      "sw        $t4, 16(%[dst])                   \n"
>-      "sw        $t5, 20(%[dst])                   \n"
>-      "sw        $t6, 24(%[dst])                   \n"
>-      "sw        $t7, 28(%[dst])                   \n"
>-      "lwr       $t0, 32(%[src])                   \n"
>-      "lwl       $t0, 35(%[src])                   \n"
>-      "lwr       $t1, 36(%[src])                   \n"
>-      "bgtz      $v1, ua_skip_pref30_128           \n"
>-      " lwl      $t1, 39(%[src])                   \n"
>-      "pref      30, 128(%[dst])                   \n"
>-      // continue setting up the dest, addr 128
>-      "ua_skip_pref30_128:                         \n"
>-
>-      "lwr       $t2, 40(%[src])                   \n"
>-      "lwl       $t2, 43(%[src])                   \n"
>-      "lwr       $t3, 44(%[src])                   \n"
>-      "lwl       $t3, 47(%[src])                   \n"
>-      "lwr       $t4, 48(%[src])                   \n"
>-      "lwl       $t4, 51(%[src])                   \n"
>-      "lwr       $t5, 52(%[src])                   \n"
>-      "lwl       $t5, 55(%[src])                   \n"
>-      "lwr       $t6, 56(%[src])                   \n"
>-      "lwl       $t6, 59(%[src])                   \n"
>-      "lwr       $t7, 60(%[src])                   \n"
>-      "lwl       $t7, 63(%[src])                   \n"
>-      "pref      0, 160(%[src])                    \n"
>-      // bring the next lines of src, addr 160
>-      "sw        $t0, 32(%[dst])                   \n"
>-      "sw        $t1, 36(%[dst])                   \n"
>-      "sw        $t2, 40(%[dst])                   \n"
>-      "sw        $t3, 44(%[dst])                   \n"
>-      "sw        $t4, 48(%[dst])                   \n"
>-      "sw        $t5, 52(%[dst])                   \n"
>-      "sw        $t6, 56(%[dst])                   \n"
>-      "sw        $t7, 60(%[dst])                   \n"
>-
>-      "addiu     %[dst],%[dst],64                  \n"  // adding 64 to dest
>-      "sltu      $v1,$t9,%[dst]                    \n"
>-      "bne       %[dst],$a3,$ua_loop16w            \n"
>-      " addiu    %[src],%[src],64                  \n"  // adding 64 to src
>-      "move      %[count],$t8                      \n"
>-
>-      // Here we have src and dest word-aligned but less than 64-bytes to go
>-
>-      "ua_chk8w:                                   \n"
>-      "pref      0, 0x0(%[src])                    \n"
>-      "andi      $t8, %[count], 0x1f               \n"  // 32-byte chunk?
>-      // the t8 is the reminder count
>-      "beq       %[count], $t8, $ua_chk1w          \n"
>-      // when count==t8, no 32-byte chunk
>-
>-      "lwr       $t0, 0(%[src])                    \n"
>-      "lwl       $t0, 3(%[src])                    \n"
>-      "lwr       $t1, 4(%[src])                    \n"
>-      "lwl       $t1, 7(%[src])                    \n"
>-      "lwr       $t2, 8(%[src])                    \n"
>-      "lwl       $t2, 11(%[src])                   \n"
>-      "lwr       $t3, 12(%[src])                   \n"
>-      "lwl       $t3, 15(%[src])                   \n"
>-      "lwr       $t4, 16(%[src])                   \n"
>-      "lwl       $t4, 19(%[src])                   \n"
>-      "lwr       $t5, 20(%[src])                   \n"
>-      "lwl       $t5, 23(%[src])                   \n"
>-      "lwr       $t6, 24(%[src])                   \n"
>-      "lwl       $t6, 27(%[src])                   \n"
>-      "lwr       $t7, 28(%[src])                   \n"
>-      "lwl       $t7, 31(%[src])                   \n"
>-      "addiu     %[src], %[src], 32                \n"
>-
>-      "sw        $t0, 0(%[dst])                    \n"
>-      "sw        $t1, 4(%[dst])                    \n"
>-      "sw        $t2, 8(%[dst])                    \n"
>-      "sw        $t3, 12(%[dst])                   \n"
>-      "sw        $t4, 16(%[dst])                   \n"
>-      "sw        $t5, 20(%[dst])                   \n"
>-      "sw        $t6, 24(%[dst])                   \n"
>-      "sw        $t7, 28(%[dst])                   \n"
>-      "addiu     %[dst], %[dst], 32                \n"
>-
>-      "$ua_chk1w:                                  \n"
>-      "andi      %[count], $t8, 0x3                \n"
>-      // now count is the reminder past 1w chunks
>-      "beq       %[count], $t8, ua_smallCopy       \n"
>-      "subu      $a3, $t8, %[count]                \n"
>-      // a3 is count of bytes in 1w chunks
>-      "addu      $a3, %[dst], $a3                  \n"
>-      // now a3 is the dst address past the 1w chunks
>-
>-      // copying in words (4-byte chunks)
>-      "$ua_wordCopy_loop:                          \n"
>-      "lwr       $v1, 0(%[src])                    \n"
>-      "lwl       $v1, 3(%[src])                    \n"
>-      "addiu     %[src], %[src], 4                 \n"
>-      "addiu     %[dst], %[dst], 4                 \n"
>-      // note: dst=a1 is word aligned here, see NOTE1
>-      "bne       %[dst], $a3, $ua_wordCopy_loop    \n"
>-      " sw       $v1,-4(%[dst])                    \n"
>-
>-      // Now less than 4 bytes (value in count) left to copy
>-      "ua_smallCopy:                               \n"
>-      "beqz      %[count], leave                   \n"
>-      " addu     $a3, %[dst], %[count]             \n"  // a3 = last dst address
>-      "$ua_smallCopy_loop:                         \n"
>-      "lb        $v1, 0(%[src])                    \n"
>-      "addiu     %[src], %[src], 1                 \n"
>-      "addiu     %[dst], %[dst], 1                 \n"
>-      "bne       %[dst],$a3,$ua_smallCopy_loop     \n"
>-      " sb       $v1, -1(%[dst])                   \n"
>-
>-      "j         $ra                               \n"
>-      " nop                                        \n"
>-      ".set      at                                \n"
>-      ".set      reorder                           \n"
>-      : [dst] "+r"(dst), [src] "+r"(src)
>-      : [count] "r"(count)
>-      : "t0", "t1", "t2", "t3", "t4", "t5", "t6", "t7", "t8", "t9", "a3", "v1",
>-        "at");
>-}
>-#endif  // HAS_COPYROW_MIPS
>-
>-// DSPR2 functions
>-#if !defined(LIBYUV_DISABLE_DSPR2) && defined(__mips_dsp) &&   \
>-    (__mips_dsp_rev >= 2) && (_MIPS_SIM == _MIPS_SIM_ABI32) && \
>-    (__mips_isa_rev < 6)
>-
>-void SplitUVRow_DSPR2(const uint8* src_uv,
>-                      uint8* dst_u,
>-                      uint8* dst_v,
>-                      int width) {
>-  __asm__ __volatile__(
>-      ".set push                                     \n"
>-      ".set noreorder                                \n"
>-      "srl             $t4, %[width], 4              \n"  // multiplies of 16
>-      "blez            $t4, 2f                       \n"
>-      " andi           %[width], %[width], 0xf       \n"  // residual
>-
>-      "1:                                            \n"
>-      "addiu           $t4, $t4, -1                  \n"
>-      "lw              $t0, 0(%[src_uv])             \n"  // V1 | U1 | V0 | U0
>-      "lw              $t1, 4(%[src_uv])             \n"  // V3 | U3 | V2 | U2
>-      "lw              $t2, 8(%[src_uv])             \n"  // V5 | U5 | V4 | U4
>-      "lw              $t3, 12(%[src_uv])            \n"  // V7 | U7 | V6 | U6
>-      "lw              $t5, 16(%[src_uv])            \n"  // V9 | U9 | V8 | U8
>-      "lw              $t6, 20(%[src_uv])            \n"  // V11 | U11 | V10 |
>-                                                          // U10
>-      "lw              $t7, 24(%[src_uv])            \n"  // V13 | U13 | V12 |
>-                                                          // U12
>-      "lw              $t8, 28(%[src_uv])            \n"  // V15 | U15 | V14 |
>-                                                          // U14
>-      "addiu           %[src_uv], %[src_uv], 32      \n"
>-      "precrq.qb.ph    $t9, $t1, $t0                 \n"  // V3 | V2 | V1 | V0
>-      "precr.qb.ph     $t0, $t1, $t0                 \n"  // U3 | U2 | U1 | U0
>-      "precrq.qb.ph    $t1, $t3, $t2                 \n"  // V7 | V6 | V5 | V4
>-      "precr.qb.ph     $t2, $t3, $t2                 \n"  // U7 | U6 | U5 | U4
>-      "precrq.qb.ph    $t3, $t6, $t5                 \n"  // V11 | V10 | V9 | V8
>-      "precr.qb.ph     $t5, $t6, $t5                 \n"  // U11 | U10 | U9 | U8
>-      "precrq.qb.ph    $t6, $t8, $t7                 \n"  // V15 | V14 | V13 |
>-                                                          // V12
>-      "precr.qb.ph     $t7, $t8, $t7                 \n"  // U15 | U14 | U13 |
>-                                                          // U12
>-      "sw              $t9, 0(%[dst_v])              \n"
>-      "sw              $t0, 0(%[dst_u])              \n"
>-      "sw              $t1, 4(%[dst_v])              \n"
>-      "sw              $t2, 4(%[dst_u])              \n"
>-      "sw              $t3, 8(%[dst_v])              \n"
>-      "sw              $t5, 8(%[dst_u])              \n"
>-      "sw              $t6, 12(%[dst_v])             \n"
>-      "sw              $t7, 12(%[dst_u])             \n"
>-      "addiu           %[dst_v], %[dst_v], 16        \n"
>-      "bgtz            $t4, 1b                       \n"
>-      " addiu          %[dst_u], %[dst_u], 16        \n"
>-
>-      "beqz            %[width], 3f                  \n"
>-      " nop                                          \n"
>-
>-      "2:                                              \n"
>-      "lbu             $t0, 0(%[src_uv])             \n"
>-      "lbu             $t1, 1(%[src_uv])             \n"
>-      "addiu           %[src_uv], %[src_uv], 2       \n"
>-      "addiu           %[width], %[width], -1        \n"
>-      "sb              $t0, 0(%[dst_u])              \n"
>-      "sb              $t1, 0(%[dst_v])              \n"
>-      "addiu           %[dst_u], %[dst_u], 1         \n"
>-      "bgtz            %[width], 2b                  \n"
>-      " addiu          %[dst_v], %[dst_v], 1         \n"
>-
>-      "3:                                              \n"
>-      ".set pop                                      \n"
>-      : [src_uv] "+r"(src_uv), [width] "+r"(width), [dst_u] "+r"(dst_u),
>-        [dst_v] "+r"(dst_v)
>-      :
>-      : "t0", "t1", "t2", "t3", "t4", "t5", "t6", "t7", "t8", "t9");
>-}
>-
>-void MirrorRow_DSPR2(const uint8* src, uint8* dst, int width) {
>-  __asm__ __volatile__(
>-      ".set push                             \n"
>-      ".set noreorder                        \n"
>-
>-      "srl       $t4, %[width], 4            \n"  // multiplies of 16
>-      "andi      $t5, %[width], 0xf          \n"
>-      "blez      $t4, 2f                     \n"
>-      " addu     %[src], %[src], %[width]    \n"  // src += width
>-
>-      "1:                                     \n"
>-      "lw        $t0, -16(%[src])            \n"  // |3|2|1|0|
>-      "lw        $t1, -12(%[src])            \n"  // |7|6|5|4|
>-      "lw        $t2, -8(%[src])             \n"  // |11|10|9|8|
>-      "lw        $t3, -4(%[src])             \n"  // |15|14|13|12|
>-      "wsbh      $t0, $t0                    \n"  // |2|3|0|1|
>-      "wsbh      $t1, $t1                    \n"  // |6|7|4|5|
>-      "wsbh      $t2, $t2                    \n"  // |10|11|8|9|
>-      "wsbh      $t3, $t3                    \n"  // |14|15|12|13|
>-      "rotr      $t0, $t0, 16                \n"  // |0|1|2|3|
>-      "rotr      $t1, $t1, 16                \n"  // |4|5|6|7|
>-      "rotr      $t2, $t2, 16                \n"  // |8|9|10|11|
>-      "rotr      $t3, $t3, 16                \n"  // |12|13|14|15|
>-      "addiu     %[src], %[src], -16         \n"
>-      "addiu     $t4, $t4, -1                \n"
>-      "sw        $t3, 0(%[dst])              \n"  // |15|14|13|12|
>-      "sw        $t2, 4(%[dst])              \n"  // |11|10|9|8|
>-      "sw        $t1, 8(%[dst])              \n"  // |7|6|5|4|
>-      "sw        $t0, 12(%[dst])             \n"  // |3|2|1|0|
>-      "bgtz      $t4, 1b                     \n"
>-      " addiu    %[dst], %[dst], 16          \n"
>-      "beqz      $t5, 3f                     \n"
>-      " nop                                  \n"
>-
>-      "2:                                     \n"
>-      "lbu       $t0, -1(%[src])             \n"
>-      "addiu     $t5, $t5, -1                \n"
>-      "addiu     %[src], %[src], -1          \n"
>-      "sb        $t0, 0(%[dst])              \n"
>-      "bgez      $t5, 2b                     \n"
>-      " addiu    %[dst], %[dst], 1           \n"
>-
>-      "3:                                     \n"
>-      ".set pop                              \n"
>-      : [src] "+r"(src), [dst] "+r"(dst)
>-      : [width] "r"(width)
>-      : "t0", "t1", "t2", "t3", "t4", "t5");
>-}
>-
>-void MirrorUVRow_DSPR2(const uint8* src_uv,
>-                       uint8* dst_u,
>-                       uint8* dst_v,
>-                       int width) {
>-  int x;
>-  int y;
>-  __asm__ __volatile__(
>-      ".set push                                    \n"
>-      ".set noreorder                               \n"
>-
>-      "addu            $t4, %[width], %[width]      \n"
>-      "srl             %[x], %[width], 4            \n"
>-      "andi            %[y], %[width], 0xf          \n"
>-      "blez            %[x], 2f                     \n"
>-      " addu           %[src_uv], %[src_uv], $t4    \n"
>-
>-      "1:                                          \n"
>-      "lw              $t0, -32(%[src_uv])          \n"  // |3|2|1|0|
>-      "lw              $t1, -28(%[src_uv])          \n"  // |7|6|5|4|
>-      "lw              $t2, -24(%[src_uv])          \n"  // |11|10|9|8|
>-      "lw              $t3, -20(%[src_uv])          \n"  // |15|14|13|12|
>-      "lw              $t4, -16(%[src_uv])          \n"  // |19|18|17|16|
>-      "lw              $t6, -12(%[src_uv])          \n"  // |23|22|21|20|
>-      "lw              $t7, -8(%[src_uv])           \n"  // |27|26|25|24|
>-      "lw              $t8, -4(%[src_uv])           \n"  // |31|30|29|28|
>-
>-      "rotr            $t0, $t0, 16                 \n"  // |1|0|3|2|
>-      "rotr            $t1, $t1, 16                 \n"  // |5|4|7|6|
>-      "rotr            $t2, $t2, 16                 \n"  // |9|8|11|10|
>-      "rotr            $t3, $t3, 16                 \n"  // |13|12|15|14|
>-      "rotr            $t4, $t4, 16                 \n"  // |17|16|19|18|
>-      "rotr            $t6, $t6, 16                 \n"  // |21|20|23|22|
>-      "rotr            $t7, $t7, 16                 \n"  // |25|24|27|26|
>-      "rotr            $t8, $t8, 16                 \n"  // |29|28|31|30|
>-      "precr.qb.ph     $t9, $t0, $t1                \n"  // |0|2|4|6|
>-      "precrq.qb.ph    $t5, $t0, $t1                \n"  // |1|3|5|7|
>-      "precr.qb.ph     $t0, $t2, $t3                \n"  // |8|10|12|14|
>-      "precrq.qb.ph    $t1, $t2, $t3                \n"  // |9|11|13|15|
>-      "precr.qb.ph     $t2, $t4, $t6                \n"  // |16|18|20|22|
>-      "precrq.qb.ph    $t3, $t4, $t6                \n"  // |17|19|21|23|
>-      "precr.qb.ph     $t4, $t7, $t8                \n"  // |24|26|28|30|
>-      "precrq.qb.ph    $t6, $t7, $t8                \n"  // |25|27|29|31|
>-      "addiu           %[src_uv], %[src_uv], -32    \n"
>-      "addiu           %[x], %[x], -1               \n"
>-      "swr             $t4, 0(%[dst_u])             \n"
>-      "swl             $t4, 3(%[dst_u])             \n"  // |30|28|26|24|
>-      "swr             $t6, 0(%[dst_v])             \n"
>-      "swl             $t6, 3(%[dst_v])             \n"  // |31|29|27|25|
>-      "swr             $t2, 4(%[dst_u])             \n"
>-      "swl             $t2, 7(%[dst_u])             \n"  // |22|20|18|16|
>-      "swr             $t3, 4(%[dst_v])             \n"
>-      "swl             $t3, 7(%[dst_v])             \n"  // |23|21|19|17|
>-      "swr             $t0, 8(%[dst_u])             \n"
>-      "swl             $t0, 11(%[dst_u])            \n"  // |14|12|10|8|
>-      "swr             $t1, 8(%[dst_v])             \n"
>-      "swl             $t1, 11(%[dst_v])            \n"  // |15|13|11|9|
>-      "swr             $t9, 12(%[dst_u])            \n"
>-      "swl             $t9, 15(%[dst_u])            \n"  // |6|4|2|0|
>-      "swr             $t5, 12(%[dst_v])            \n"
>-      "swl             $t5, 15(%[dst_v])            \n"  // |7|5|3|1|
>-      "addiu           %[dst_v], %[dst_v], 16       \n"
>-      "bgtz            %[x], 1b                     \n"
>-      " addiu          %[dst_u], %[dst_u], 16       \n"
>-      "beqz            %[y], 3f                     \n"
>-      " nop                                         \n"
>-      "b               2f                           \n"
>-      " nop                                         \n"
>-
>-      "2:                                            \n"
>-      "lbu             $t0, -2(%[src_uv])           \n"
>-      "lbu             $t1, -1(%[src_uv])           \n"
>-      "addiu           %[src_uv], %[src_uv], -2     \n"
>-      "addiu           %[y], %[y], -1               \n"
>-      "sb              $t0, 0(%[dst_u])             \n"
>-      "sb              $t1, 0(%[dst_v])             \n"
>-      "addiu           %[dst_u], %[dst_u], 1        \n"
>-      "bgtz            %[y], 2b                     \n"
>-      " addiu          %[dst_v], %[dst_v], 1        \n"
>-
>-      "3:                                            \n"
>-      ".set pop                                     \n"
>-      : [src_uv] "+r"(src_uv), [dst_u] "+r"(dst_u), [dst_v] "+r"(dst_v),
>-        [x] "=&r"(x), [y] "=&r"(y)
>-      : [width] "r"(width)
>-      : "t0", "t1", "t2", "t3", "t4", "t5", "t7", "t8", "t9");
>-}
>-
>-void I422ToARGBRow_DSPR2(const uint8* src_y,
>-                         const uint8* src_u,
>-                         const uint8* src_v,
>-                         uint8* rgb_buf,
>-                         const struct YuvConstants* yuvconstants,
>-                         int width) {
>-  int x;
>-  uint32 tmp_ub = yuvconstants->kUVToB[0];
>-  uint32 tmp_ug = yuvconstants->kUVToG[0];
>-  uint32 tmp_vg = yuvconstants->kUVToG[1];
>-  uint32 tmp_vr = yuvconstants->kUVToR[1];
>-  uint32 tmp_bb = yuvconstants->kUVBiasB[0];
>-  uint32 tmp_bg = yuvconstants->kUVBiasG[0];
>-  uint32 tmp_br = yuvconstants->kUVBiasR[0];
>-  uint32 yg = yuvconstants->kYToRgb[0];
>-  uint32 tmp_yg;
>-  uint32 tmp_mask = 0x7fff7fff;
>-  tmp_bb = ((uint)(tmp_bb & 0xffff) << 16) | (tmp_bb & 0xffff);
>-  tmp_bg = ((uint)(tmp_bg & 0xffff) << 16) | (tmp_bg & 0xffff);
>-  tmp_br = ((uint)(tmp_br & 0xffff) << 16) | (tmp_br & 0xffff);
>-  tmp_yg = ((uint)(yg & 0xffff) << 16) | (yg & 0xffff);
>-  tmp_ub = ~(((uint)(tmp_ub & 0xffff) << 16) | (tmp_ub & 0xffff)) + 0x00010001;
>-  tmp_ug = ((uint)(tmp_ug & 0xffff) << 16) | (tmp_ug & 0xffff);
>-  tmp_vg = ((uint)(tmp_vg & 0xffff) << 16) | (tmp_vg & 0xffff);
>-  tmp_vr = ~(((uint)(tmp_vr & 0xffff) << 16) | (tmp_vr & 0xffff)) + 0x00010001;
>-  yg = yg * 0x0101;
>-
>-  for (x = 0; x < width - 1; x += 2) {
>-    uint32 tmp_t1, tmp_t2, tmp_t3, tmp_t4, tmp_t5;
>-    uint32 tmp_t6, tmp_t7, tmp_t8, tmp_t9;
>-    __asm__ __volatile__(
>-        ".set push                                             \n"
>-        ".set noreorder                                        \n"
>-        "lbu              %[tmp_t7], 0(%[src_y])               \n"
>-        "lbu              %[tmp_t1], 1(%[src_y])               \n"
>-        "mul              %[tmp_t7], %[tmp_t7],     %[yg]      \n"
>-        "mul              %[tmp_t1], %[tmp_t1],     %[yg]      \n"
>-        "lbu              %[tmp_t2], 0(%[src_u])               \n"
>-        "lbu              %[tmp_t3], 0(%[src_v])               \n"
>-        "replv.ph         %[tmp_t2], %[tmp_t2]                 \n"
>-        "replv.ph         %[tmp_t3], %[tmp_t3]                 \n"
>-        "mul.ph           %[tmp_t4], %[tmp_t2],     %[tmp_ub]  \n"
>-        "mul.ph           %[tmp_t5], %[tmp_t2],     %[tmp_ug]  \n"
>-        "mul.ph           %[tmp_t6], %[tmp_t3],     %[tmp_vr]  \n"
>-        "mul.ph           %[tmp_t3], %[tmp_t3],     %[tmp_vg]  \n"
>-        "srl              %[tmp_t7], %[tmp_t7],     16         \n"
>-        "ins              %[tmp_t1], %[tmp_t7],     0,      16 \n"
>-        "addq_s.ph        %[tmp_t7], %[tmp_t1],     %[tmp_bb]  \n"
>-        "addq_s.ph        %[tmp_t8], %[tmp_t1],     %[tmp_bg]  \n"
>-        "addq_s.ph        %[tmp_t9], %[tmp_t1],     %[tmp_br]  \n"
>-        "addq_s.ph        %[tmp_t5], %[tmp_t5],     %[tmp_t3]  \n"
>-        "addq_s.ph        %[tmp_t7], %[tmp_t7],     %[tmp_t4]  \n"
>-        "subq_s.ph        %[tmp_t8], %[tmp_t8],     %[tmp_t5]  \n"
>-        "addq_s.ph        %[tmp_t9], %[tmp_t9],     %[tmp_t6]  \n"
>-        "shra.ph          %[tmp_t7], %[tmp_t7],     6          \n"
>-        "shra.ph          %[tmp_t8], %[tmp_t8],     6          \n"
>-        "shra.ph          %[tmp_t9], %[tmp_t9],     6          \n"
>-        "shll_s.ph        %[tmp_t7], %[tmp_t7],     7          \n"
>-        "shll_s.ph        %[tmp_t8], %[tmp_t8],     7          \n"
>-        "shll_s.ph        %[tmp_t9], %[tmp_t9],     7          \n"
>-        "precrqu_s.qb.ph  %[tmp_t8], %[tmp_mask],   %[tmp_t8]  \n"
>-        "precrqu_s.qb.ph  %[tmp_t7], %[tmp_t9],     %[tmp_t7]  \n"
>-        "precrq.ph.w      %[tmp_t9], %[tmp_t8],     %[tmp_t7]  \n"
>-        "ins              %[tmp_t7], %[tmp_t8],     16,     16 \n"
>-        "precr.qb.ph      %[tmp_t8], %[tmp_t9],     %[tmp_t7]  \n"
>-        "precrq.qb.ph     %[tmp_t7], %[tmp_t9],     %[tmp_t7]  \n"
>-        "sw               %[tmp_t8], 0(%[rgb_buf])             \n"
>-        "sw               %[tmp_t7], 4(%[rgb_buf])             \n"
>-        ".set pop                                              \n"
>-        : [tmp_t1] "=&r"(tmp_t1), [tmp_t2] "=&r"(tmp_t2),
>-          [tmp_t3] "=&r"(tmp_t3), [tmp_t4] "=&r"(tmp_t4),
>-          [tmp_t5] "=&r"(tmp_t5), [tmp_t6] "=&r"(tmp_t6),
>-          [tmp_t7] "=&r"(tmp_t7), [tmp_t8] "=&r"(tmp_t8), [tmp_t9] "=&r"(tmp_t9)
>-        : [src_y] "r"(src_y), [src_u] "r"(src_u), [src_v] "r"(src_v),
>-          [tmp_ub] "r"(tmp_ub), [tmp_ug] "r"(tmp_ug), [yg] "r"(yg),
>-          [tmp_vg] "r"(tmp_vg), [tmp_vr] "r"(tmp_vr), [tmp_bb] "r"(tmp_bb),
>-          [tmp_bg] "r"(tmp_bg), [tmp_br] "r"(tmp_br), [tmp_yg] "r"(tmp_yg),
>-          [rgb_buf] "r"(rgb_buf), [tmp_mask] "r"(tmp_mask));
>-    src_y += 2;
>-    src_u += 1;
>-    src_v += 1;
>-    rgb_buf += 8;  // Advance 4 pixels.
>-  }
>-}
>-
>-// Bilinear filter 8x2 -> 8x1
>-void InterpolateRow_DSPR2(uint8* dst_ptr,
>-                          const uint8* src_ptr,
>-                          ptrdiff_t src_stride,
>-                          int dst_width,
>-                          int source_y_fraction) {
>-  int y0_fraction = 256 - source_y_fraction;
>-  const uint8* src_ptr1 = src_ptr + src_stride;
>-
>-  __asm__ __volatile__(
>-      ".set push                                           \n"
>-      ".set noreorder                                      \n"
>-
>-      "replv.ph          $t0, %[y0_fraction]               \n"
>-      "replv.ph          $t1, %[source_y_fraction]         \n"
>-
>-      "1:                                                    \n"
>-      "lw                $t2, 0(%[src_ptr])                \n"
>-      "lw                $t3, 0(%[src_ptr1])               \n"
>-      "lw                $t4, 4(%[src_ptr])                \n"
>-      "lw                $t5, 4(%[src_ptr1])               \n"
>-      "muleu_s.ph.qbl    $t6, $t2, $t0                     \n"
>-      "muleu_s.ph.qbr    $t7, $t2, $t0                     \n"
>-      "muleu_s.ph.qbl    $t8, $t3, $t1                     \n"
>-      "muleu_s.ph.qbr    $t9, $t3, $t1                     \n"
>-      "muleu_s.ph.qbl    $t2, $t4, $t0                     \n"
>-      "muleu_s.ph.qbr    $t3, $t4, $t0                     \n"
>-      "muleu_s.ph.qbl    $t4, $t5, $t1                     \n"
>-      "muleu_s.ph.qbr    $t5, $t5, $t1                     \n"
>-      "addq.ph           $t6, $t6, $t8                     \n"
>-      "addq.ph           $t7, $t7, $t9                     \n"
>-      "addq.ph           $t2, $t2, $t4                     \n"
>-      "addq.ph           $t3, $t3, $t5                     \n"
>-      "shra_r.ph         $t6, $t6, 8                       \n"
>-      "shra_r.ph         $t7, $t7, 8                       \n"
>-      "shra_r.ph         $t2, $t2, 8                       \n"
>-      "shra_r.ph         $t3, $t3, 8                       \n"
>-      "precr.qb.ph       $t6, $t6, $t7                     \n"
>-      "precr.qb.ph       $t2, $t2, $t3                     \n"
>-      "addiu             %[src_ptr], %[src_ptr], 8         \n"
>-      "addiu             %[src_ptr1], %[src_ptr1], 8       \n"
>-      "addiu             %[dst_width], %[dst_width], -8    \n"
>-      "sw                $t6, 0(%[dst_ptr])                \n"
>-      "sw                $t2, 4(%[dst_ptr])                \n"
>-      "bgtz              %[dst_width], 1b                  \n"
>-      " addiu            %[dst_ptr], %[dst_ptr], 8         \n"
>-
>-      ".set pop                                            \n"
>-      : [dst_ptr] "+r"(dst_ptr), [src_ptr1] "+r"(src_ptr1),
>-        [src_ptr] "+r"(src_ptr), [dst_width] "+r"(dst_width)
>-      : [source_y_fraction] "r"(source_y_fraction),
>-        [y0_fraction] "r"(y0_fraction), [src_stride] "r"(src_stride)
>-      : "t0", "t1", "t2", "t3", "t4", "t5", "t6", "t7", "t8", "t9");
>-}
>-#include <stdio.h>
>-void RGB24ToARGBRow_DSPR2(const uint8* src_rgb24, uint8* dst_argb, int width) {
>-  int x;
>-  uint32 tmp_mask = 0xff;
>-  uint32 tmp_t1;
>-  for (x = 0; x < (width - 1); ++x) {
>-    __asm__ __volatile__(
>-        ".set push                                                  \n"
>-        ".set noreorder                                             \n"
>-        "ulw             %[tmp_t1],    0(%[src_rgb24])              \n"
>-        "addiu           %[dst_argb],  %[dst_argb],     4           \n"
>-        "addiu           %[src_rgb24], %[src_rgb24],    3           \n"
>-        "ins             %[tmp_t1],    %[tmp_mask],     24,    8    \n"
>-        "sw              %[tmp_t1],    -4(%[dst_argb])              \n"
>-        ".set pop                                                   \n"
>-        : [src_rgb24] "+r"(src_rgb24), [dst_argb] "+r"(dst_argb),
>-          [tmp_t1] "=&r"(tmp_t1)
>-        : [tmp_mask] "r"(tmp_mask)
>-        : "memory");
>-  }
>-  uint8 b = src_rgb24[0];
>-  uint8 g = src_rgb24[1];
>-  uint8 r = src_rgb24[2];
>-  dst_argb[0] = b;
>-  dst_argb[1] = g;
>-  dst_argb[2] = r;
>-  dst_argb[3] = 255u;
>-}
>-
>-void RAWToARGBRow_DSPR2(const uint8* src_raw, uint8* dst_argb, int width) {
>-  int x;
>-  uint32 tmp_mask = 0xff;
>-  uint32 tmp_t1, tmp_t2;
>-  for (x = 0; x < (width - 1); ++x) {
>-    __asm__ __volatile__(
>-        ".set push                                               \n"
>-        ".set noreorder                                          \n"
>-        "ulw               %[tmp_t1],   0(%[src_raw])            \n"
>-        "addiu             %[dst_argb], %[dst_argb],      4      \n"
>-        "addiu             %[src_raw],  %[src_raw],       3      \n"
>-        "srl               %[tmp_t2],   %[tmp_t1],        16     \n"
>-        "ins               %[tmp_t1],   %[tmp_mask],      24, 8  \n"
>-        "ins               %[tmp_t1],   %[tmp_t1],        16, 8  \n"
>-        "ins               %[tmp_t1],   %[tmp_t2],        0,  8  \n"
>-        "sw                %[tmp_t1],   -4(%[dst_argb])          \n"
>-        ".set pop                                                \n"
>-        : [src_raw] "+r"(src_raw), [dst_argb] "+r"(dst_argb),
>-          [tmp_t1] "=&r"(tmp_t1), [tmp_t2] "=&r"(tmp_t2)
>-        : [tmp_mask] "r"(tmp_mask)
>-        : "memory");
>-  }
>-  uint8 r = src_raw[0];
>-  uint8 g = src_raw[1];
>-  uint8 b = src_raw[2];
>-  dst_argb[0] = b;
>-  dst_argb[1] = g;
>-  dst_argb[2] = r;
>-  dst_argb[3] = 255u;
>-}
>-
>-void RGB565ToARGBRow_DSPR2(const uint8* src_rgb565,
>-                           uint8* dst_argb,
>-                           int width) {
>-  int x;
>-  uint32 tmp_mask = 0xff;
>-  uint32 tmp_t1, tmp_t2, tmp_t3;
>-  for (x = 0; x < width; ++x) {
>-    __asm__ __volatile__(
>-        ".set push                                                   \n"
>-        ".set noreorder                                              \n"
>-        "lhu               %[tmp_t1],     0(%[src_rgb565])           \n"
>-        "addiu             %[dst_argb],   %[dst_argb],      4        \n"
>-        "addiu             %[src_rgb565], %[src_rgb565],    2        \n"
>-        "sll               %[tmp_t2],     %[tmp_t1],        8        \n"
>-        "ins               %[tmp_t2],     %[tmp_mask],      24,8     \n"
>-        "ins               %[tmp_t2],     %[tmp_t1],        3, 16    \n"
>-        "ins               %[tmp_t2],     %[tmp_t1],        5, 11    \n"
>-        "srl               %[tmp_t3],     %[tmp_t1],        9        \n"
>-        "ins               %[tmp_t2],     %[tmp_t3],        8, 2     \n"
>-        "ins               %[tmp_t2],     %[tmp_t1],        3, 5     \n"
>-        "srl               %[tmp_t3],     %[tmp_t1],        2        \n"
>-        "ins               %[tmp_t2],     %[tmp_t3],        0, 3     \n"
>-        "sw                %[tmp_t2],     -4(%[dst_argb])            \n"
>-        ".set pop                                                    \n"
>-        : [tmp_t1] "=&r"(tmp_t1), [tmp_t2] "=&r"(tmp_t2),
>-          [tmp_t3] "=&r"(tmp_t3), [src_rgb565] "+r"(src_rgb565),
>-          [dst_argb] "+r"(dst_argb)
>-        : [tmp_mask] "r"(tmp_mask));
>-  }
>-}
>-
>-void ARGB1555ToARGBRow_DSPR2(const uint8* src_argb1555,
>-                             uint8* dst_argb,
>-                             int width) {
>-  int x;
>-  uint32 tmp_t1, tmp_t2, tmp_t3;
>-  for (x = 0; x < width; ++x) {
>-    __asm__ __volatile__(
>-        ".set push                                                   \n"
>-        ".set noreorder                                              \n"
>-        "lh                %[tmp_t1],       0(%[src_argb1555])       \n"
>-        "addiu             %[dst_argb],     %[dst_argb],      4      \n"
>-        "addiu             %[src_argb1555], %[src_argb1555],  2      \n"
>-        "sll               %[tmp_t2],       %[tmp_t1],        9      \n"
>-        "ins               %[tmp_t2],       %[tmp_t1],        4, 15  \n"
>-        "ins               %[tmp_t2],       %[tmp_t1],        6, 10  \n"
>-        "srl               %[tmp_t3],       %[tmp_t1],        7      \n"
>-        "ins               %[tmp_t2],       %[tmp_t3],        8, 3   \n"
>-        "ins               %[tmp_t2],       %[tmp_t1],        3, 5   \n"
>-        "srl               %[tmp_t3],       %[tmp_t1],        2      \n"
>-        "ins               %[tmp_t2],       %[tmp_t3],        0, 3   \n"
>-        "sw                %[tmp_t2],       -4(%[dst_argb])          \n"
>-        ".set pop                                                    \n"
>-        : [tmp_t1] "=&r"(tmp_t1), [tmp_t2] "=&r"(tmp_t2),
>-          [tmp_t3] "=&r"(tmp_t3), [src_argb1555] "+r"(src_argb1555),
>-          [dst_argb] "+r"(dst_argb)
>-        :);
>-  }
>-}
>-
>-void ARGB4444ToARGBRow_DSPR2(const uint8* src_argb4444,
>-                             uint8* dst_argb,
>-                             int width) {
>-  int x;
>-  uint32 tmp_t1;
>-  for (x = 0; x < width; ++x) {
>-    __asm__ __volatile__(
>-        ".set push                                                    \n"
>-        ".set noreorder                                               \n"
>-        "lh                %[tmp_t1],       0(%[src_argb4444])        \n"
>-        "addiu             %[dst_argb],     %[dst_argb],       4      \n"
>-        "addiu             %[src_argb4444], %[src_argb4444],   2      \n"
>-        "ins               %[tmp_t1],       %[tmp_t1],         16, 16 \n"
>-        "ins               %[tmp_t1],       %[tmp_t1],         12, 16 \n"
>-        "ins               %[tmp_t1],       %[tmp_t1],         8,  12 \n"
>-        "ins               %[tmp_t1],       %[tmp_t1],         4,  8  \n"
>-        "sw                %[tmp_t1],       -4(%[dst_argb])           \n"
>-        ".set pop                                                     \n"
>-        : [src_argb4444] "+r"(src_argb4444), [dst_argb] "+r"(dst_argb),
>-          [tmp_t1] "=&r"(tmp_t1));
>-  }
>-}
>-
>-void I444ToARGBRow_DSPR2(const uint8* y_buf,
>-                         const uint8* u_buf,
>-                         const uint8* v_buf,
>-                         uint8* rgb_buf,
>-                         const struct YuvConstants* yuvconstants,
>-                         int width) {
>-  int x;
>-  uint32 tmp_ub = yuvconstants->kUVToB[0];
>-  uint32 tmp_ug = yuvconstants->kUVToG[0];
>-  uint32 tmp_vg = yuvconstants->kUVToG[1];
>-  uint32 tmp_vr = yuvconstants->kUVToR[1];
>-  uint32 tmp_bb = yuvconstants->kUVBiasB[0];
>-  uint32 tmp_bg = yuvconstants->kUVBiasG[0];
>-  uint32 tmp_br = yuvconstants->kUVBiasR[0];
>-  uint32 yg = yuvconstants->kYToRgb[0];
>-  uint32 tmp_mask = 0x7fff7fff;
>-  uint32 tmp_yg;
>-
>-  tmp_bb = ((uint)(tmp_bb & 0xffff) << 16) | (tmp_bb & 0xffff);
>-  tmp_bg = ((uint)(tmp_bg & 0xffff) << 16) | (tmp_bg & 0xffff);
>-  tmp_br = ((uint)(tmp_br & 0xffff) << 16) | (tmp_br & 0xffff);
>-  tmp_yg = ((uint)(yg & 0xffff) << 16) | (yg & 0xffff);
>-  tmp_ub = ~(((uint)(tmp_ub & 0xffff) << 16) | (tmp_ub & 0xffff)) + 0x00010001;
>-  tmp_ug = ((uint)(tmp_ug & 0xffff) << 16) | (tmp_ug & 0xffff);
>-  tmp_vg = ((uint)(tmp_vg & 0xffff) << 16) | (tmp_vg & 0xffff);
>-  tmp_vr = ~(((uint)(tmp_vr & 0xffff) << 16) | (tmp_vr & 0xffff)) + 0x00010001;
>-  yg = yg * 0x0101;
>-
>-  for (x = 0; x < width - 1; x += 2) {
>-    uint32 tmp_t1, tmp_t2, tmp_t3, tmp_t4, tmp_t5;
>-    uint32 tmp_t6, tmp_t7, tmp_t8, tmp_t9;
>-    __asm__ __volatile__(
>-        ".set push                                              \n"
>-        ".set noreorder                                         \n"
>-        "lbu              %[tmp_t7], 0(%[y_buf])               \n"
>-        "lbu              %[tmp_t1], 1(%[y_buf])               \n"
>-        "mul              %[tmp_t7], %[tmp_t7],     %[yg]      \n"
>-        "mul              %[tmp_t1], %[tmp_t1],     %[yg]      \n"
>-        "lh               %[tmp_t2], 0(%[u_buf])               \n"
>-        "lh               %[tmp_t3], 0(%[v_buf])               \n"
>-        "preceu.ph.qbr    %[tmp_t2], %[tmp_t2]                 \n"
>-        "preceu.ph.qbr    %[tmp_t3], %[tmp_t3]                 \n"
>-        "mul.ph           %[tmp_t4], %[tmp_t2],     %[tmp_ub]  \n"
>-        "mul.ph           %[tmp_t5], %[tmp_t2],     %[tmp_ug]  \n"
>-        "mul.ph           %[tmp_t6], %[tmp_t3],     %[tmp_vr]  \n"
>-        "mul.ph           %[tmp_t3], %[tmp_t3],     %[tmp_vg]  \n"
>-        "srl              %[tmp_t7], %[tmp_t7],     16         \n"
>-        "ins              %[tmp_t1], %[tmp_t7],     0,      16 \n"
>-        "addq_s.ph        %[tmp_t7], %[tmp_t1],     %[tmp_bb]  \n"
>-        "addq_s.ph        %[tmp_t8], %[tmp_t1],     %[tmp_bg]  \n"
>-        "addq_s.ph        %[tmp_t9], %[tmp_t1],     %[tmp_br]  \n"
>-        "addq_s.ph        %[tmp_t5], %[tmp_t5],     %[tmp_t3]  \n"
>-        "addq_s.ph        %[tmp_t7], %[tmp_t7],     %[tmp_t4]  \n"
>-        "subq_s.ph        %[tmp_t8], %[tmp_t8],     %[tmp_t5]  \n"
>-        "addq_s.ph        %[tmp_t9], %[tmp_t9],     %[tmp_t6]  \n"
>-        "shra.ph          %[tmp_t7], %[tmp_t7],     6          \n"
>-        "shra.ph          %[tmp_t8], %[tmp_t8],     6          \n"
>-        "shra.ph          %[tmp_t9], %[tmp_t9],     6          \n"
>-        "shll_s.ph        %[tmp_t7], %[tmp_t7],     7          \n"
>-        "shll_s.ph        %[tmp_t8], %[tmp_t8],     7          \n"
>-        "shll_s.ph        %[tmp_t9], %[tmp_t9],     7          \n"
>-        "precrqu_s.qb.ph  %[tmp_t8], %[tmp_mask],   %[tmp_t8]  \n"
>-        "precrqu_s.qb.ph  %[tmp_t7], %[tmp_t9],     %[tmp_t7]  \n"
>-        "precrq.ph.w      %[tmp_t2], %[tmp_t8],     %[tmp_t7]  \n"
>-        "ins              %[tmp_t7], %[tmp_t8],     16,     16 \n"
>-        "precr.qb.ph      %[tmp_t8], %[tmp_t2],     %[tmp_t7]  \n"
>-        "precrq.qb.ph     %[tmp_t7], %[tmp_t2],     %[tmp_t7]  \n"
>-        "sw               %[tmp_t8], 0(%[rgb_buf])             \n"
>-        "sw               %[tmp_t7], 4(%[rgb_buf])             \n"
>-        ".set pop                                              \n"
>-        : [tmp_t1] "=&r"(tmp_t1), [tmp_t2] "=&r"(tmp_t2),
>-          [tmp_t3] "=&r"(tmp_t3), [tmp_t4] "=&r"(tmp_t4),
>-          [tmp_t5] "=&r"(tmp_t5), [tmp_t6] "=&r"(tmp_t6),
>-          [tmp_t7] "=&r"(tmp_t7), [tmp_t8] "=&r"(tmp_t8), [tmp_t9] "=&r"(tmp_t9)
>-        : [y_buf] "r"(y_buf), [yg] "r"(yg), [u_buf] "r"(u_buf),
>-          [v_buf] "r"(v_buf), [tmp_ub] "r"(tmp_ub), [tmp_ug] "r"(tmp_ug),
>-          [tmp_vg] "r"(tmp_vg), [tmp_vr] "r"(tmp_vr), [tmp_bb] "r"(tmp_bb),
>-          [tmp_bg] "r"(tmp_bg), [tmp_br] "r"(tmp_br), [tmp_yg] "r"(tmp_yg),
>-          [rgb_buf] "r"(rgb_buf), [tmp_mask] "r"(tmp_mask));
>-    y_buf += 2;
>-    u_buf += 2;
>-    v_buf += 2;
>-    rgb_buf += 8;  // Advance 1 pixel.
>-  }
>-}
>-
>-void I422ToARGB4444Row_DSPR2(const uint8* src_y,
>-                             const uint8* src_u,
>-                             const uint8* src_v,
>-                             uint8* dst_argb4444,
>-                             const struct YuvConstants* yuvconstants,
>-                             int width) {
>-  int x;
>-  uint32 tmp_ub = yuvconstants->kUVToB[0];
>-  uint32 tmp_ug = yuvconstants->kUVToG[0];
>-  uint32 tmp_vg = yuvconstants->kUVToG[1];
>-  uint32 tmp_vr = yuvconstants->kUVToR[1];
>-  uint32 tmp_bb = yuvconstants->kUVBiasB[0];
>-  uint32 tmp_bg = yuvconstants->kUVBiasG[0];
>-  uint32 tmp_br = yuvconstants->kUVBiasR[0];
>-  uint32 yg = yuvconstants->kYToRgb[0];
>-  uint32 tmp_yg;
>-  uint32 tmp_mask = 0x7fff7fff;
>-  tmp_bb = ((uint)(tmp_bb & 0xffff) << 16) | (tmp_bb & 0xffff);
>-  tmp_bg = ((uint)(tmp_bg & 0xffff) << 16) | (tmp_bg & 0xffff);
>-  tmp_br = ((uint)(tmp_br & 0xffff) << 16) | (tmp_br & 0xffff);
>-  tmp_yg = ((uint)(yg & 0xffff) << 16) | (yg & 0xffff);
>-  tmp_ub = ~(((uint)(tmp_ub & 0xffff) << 16) | (tmp_ub & 0xffff)) + 0x00010001;
>-  tmp_ug = ((uint)(tmp_ug & 0xffff) << 16) | (tmp_ug & 0xffff);
>-  tmp_vg = ((uint)(tmp_vg & 0xffff) << 16) | (tmp_vg & 0xffff);
>-  tmp_vr = ~(((uint)(tmp_vr & 0xffff) << 16) | (tmp_vr & 0xffff)) + 0x00010001;
>-  yg = yg * 0x0101;
>-
>-  for (x = 0; x < width - 1; x += 2) {
>-    uint32 tmp_t1, tmp_t2, tmp_t3, tmp_t4, tmp_t5;
>-    uint32 tmp_t6, tmp_t7, tmp_t8, tmp_t9;
>-    __asm__ __volatile__(
>-        ".set push                                             \n"
>-        ".set noreorder                                        \n"
>-        "lbu              %[tmp_t7], 0(%[src_y])               \n"
>-        "lbu              %[tmp_t1], 1(%[src_y])               \n"
>-        "mul              %[tmp_t7], %[tmp_t7],     %[yg]      \n"
>-        "mul              %[tmp_t1], %[tmp_t1],     %[yg]      \n"
>-        "lbu              %[tmp_t2], 0(%[src_u])               \n"
>-        "lbu              %[tmp_t3], 0(%[src_v])               \n"
>-        "replv.ph         %[tmp_t2], %[tmp_t2]                 \n"
>-        "replv.ph         %[tmp_t3], %[tmp_t3]                 \n"
>-        "mul.ph           %[tmp_t4], %[tmp_t2],     %[tmp_ub]  \n"
>-        "mul.ph           %[tmp_t5], %[tmp_t2],     %[tmp_ug]  \n"
>-        "mul.ph           %[tmp_t6], %[tmp_t3],     %[tmp_vr]  \n"
>-        "mul.ph           %[tmp_t3], %[tmp_t3],     %[tmp_vg]  \n"
>-        "srl              %[tmp_t7], %[tmp_t7],     16         \n"
>-        "ins              %[tmp_t1], %[tmp_t7],     0,      16 \n"
>-        "addq_s.ph        %[tmp_t7], %[tmp_t1],     %[tmp_bb]  \n"
>-        "addq_s.ph        %[tmp_t8], %[tmp_t1],     %[tmp_bg]  \n"
>-        "addq_s.ph        %[tmp_t9], %[tmp_t1],     %[tmp_br]  \n"
>-        "addq_s.ph        %[tmp_t5], %[tmp_t5],     %[tmp_t3]  \n"
>-        "addq_s.ph        %[tmp_t7], %[tmp_t7],     %[tmp_t4]  \n"
>-        "subq_s.ph        %[tmp_t8], %[tmp_t8],     %[tmp_t5]  \n"
>-        "addq_s.ph        %[tmp_t9], %[tmp_t9],     %[tmp_t6]  \n"
>-        "shra.ph          %[tmp_t7], %[tmp_t7],     6          \n"
>-        "shra.ph          %[tmp_t8], %[tmp_t8],     6          \n"
>-        "shra.ph          %[tmp_t9], %[tmp_t9],     6          \n"
>-        "shll_s.ph        %[tmp_t7], %[tmp_t7],     7          \n"
>-        "shll_s.ph        %[tmp_t8], %[tmp_t8],     7          \n"
>-        "shll_s.ph        %[tmp_t9], %[tmp_t9],     7          \n"
>-        "precrqu_s.qb.ph  %[tmp_t8], %[tmp_mask],   %[tmp_t8]  \n"
>-        "precrqu_s.qb.ph  %[tmp_t7], %[tmp_t9],     %[tmp_t7]  \n"
>-        "precrq.ph.w      %[tmp_t2], %[tmp_t8],     %[tmp_t7]  \n"
>-        "ins              %[tmp_t7], %[tmp_t8],     16,     16 \n"
>-        "precr.qb.ph      %[tmp_t8], %[tmp_t2],     %[tmp_t7]  \n"
>-        "precrq.qb.ph     %[tmp_t7], %[tmp_t2],     %[tmp_t7]  \n"
>-        "shrl.qb          %[tmp_t1], %[tmp_t8],     4          \n"
>-        "shrl.qb          %[tmp_t2], %[tmp_t7],     4          \n"
>-        "shrl.ph          %[tmp_t8], %[tmp_t1],     4          \n"
>-        "shrl.ph          %[tmp_t7], %[tmp_t2],     4          \n"
>-        "or               %[tmp_t8], %[tmp_t8],     %[tmp_t1]  \n"
>-        "or               %[tmp_t7], %[tmp_t7],     %[tmp_t2]  \n"
>-        "precr.qb.ph      %[tmp_t8], %[tmp_t7],     %[tmp_t8]  \n"
>-        "sw               %[tmp_t8], 0(%[dst_argb4444])        \n"
>-        ".set pop                                              \n"
>-        : [tmp_t1] "=&r"(tmp_t1), [tmp_t2] "=&r"(tmp_t2),
>-          [tmp_t3] "=&r"(tmp_t3), [tmp_t4] "=&r"(tmp_t4),
>-          [tmp_t5] "=&r"(tmp_t5), [tmp_t6] "=&r"(tmp_t6),
>-          [tmp_t7] "=&r"(tmp_t7), [tmp_t8] "=&r"(tmp_t8), [tmp_t9] "=&r"(tmp_t9)
>-        : [dst_argb4444] "r"(dst_argb4444), [yg] "r"(yg), [src_u] "r"(src_u),
>-          [src_v] "r"(src_v), [src_y] "r"(src_y), [tmp_ub] "r"(tmp_ub),
>-          [tmp_ug] "r"(tmp_ug), [tmp_vg] "r"(tmp_vg), [tmp_vr] "r"(tmp_vr),
>-          [tmp_bb] "r"(tmp_bb), [tmp_bg] "r"(tmp_bg), [tmp_br] "r"(tmp_br),
>-          [tmp_yg] "r"(tmp_yg), [tmp_mask] "r"(tmp_mask));
>-    src_y += 2;
>-    src_u += 1;
>-    src_v += 1;
>-    dst_argb4444 += 4;  // Advance 2 pixels.
>-  }
>-}
>-
>-void I422ToARGB1555Row_DSPR2(const uint8* src_y,
>-                             const uint8* src_u,
>-                             const uint8* src_v,
>-                             uint8* dst_argb1555,
>-                             const struct YuvConstants* yuvconstants,
>-                             int width) {
>-  int x;
>-  uint32 tmp_ub = yuvconstants->kUVToB[0];
>-  uint32 tmp_ug = yuvconstants->kUVToG[0];
>-  uint32 tmp_vg = yuvconstants->kUVToG[1];
>-  uint32 tmp_vr = yuvconstants->kUVToR[1];
>-  uint32 tmp_bb = yuvconstants->kUVBiasB[0];
>-  uint32 tmp_bg = yuvconstants->kUVBiasG[0];
>-  uint32 tmp_br = yuvconstants->kUVBiasR[0];
>-  uint32 yg = yuvconstants->kYToRgb[0];
>-  uint32 tmp_yg;
>-  uint32 tmp_mask = 0x80008000;
>-  tmp_bb = ((uint)(tmp_bb & 0xffff) << 16) | (tmp_bb & 0xffff);
>-  tmp_bg = ((uint)(tmp_bg & 0xffff) << 16) | (tmp_bg & 0xffff);
>-  tmp_br = ((uint)(tmp_br & 0xffff) << 16) | (tmp_br & 0xffff);
>-  tmp_yg = ((uint)(yg & 0xffff) << 16) | (yg & 0xffff);
>-  tmp_ub = ~(((uint)(tmp_ub & 0xffff) << 16) | (tmp_ub & 0xffff)) + 0x00010001;
>-  tmp_ug = ((uint)(tmp_ug & 0xffff) << 16) | (tmp_ug & 0xffff);
>-  tmp_vg = ((uint)(tmp_vg & 0xffff) << 16) | (tmp_vg & 0xffff);
>-  tmp_vr = ~(((uint)(tmp_vr & 0xffff) << 16) | (tmp_vr & 0xffff)) + 0x00010001;
>-  yg = yg * 0x0101;
>-
>-  for (x = 0; x < width - 1; x += 2) {
>-    uint32 tmp_t1, tmp_t2, tmp_t3, tmp_t4, tmp_t5;
>-    uint32 tmp_t6, tmp_t7, tmp_t8, tmp_t9;
>-    __asm__ __volatile__(
>-        ".set push                                             \n"
>-        ".set noreorder                                        \n"
>-        "lbu              %[tmp_t7], 0(%[src_y])               \n"
>-        "lbu              %[tmp_t1], 1(%[src_y])               \n"
>-        "mul              %[tmp_t7], %[tmp_t7],     %[yg]      \n"
>-        "mul              %[tmp_t1], %[tmp_t1],     %[yg]      \n"
>-        "lbu              %[tmp_t2], 0(%[src_u])               \n"
>-        "lbu              %[tmp_t3], 0(%[src_v])               \n"
>-        "replv.ph         %[tmp_t2], %[tmp_t2]                 \n"
>-        "replv.ph         %[tmp_t3], %[tmp_t3]                 \n"
>-        "mul.ph           %[tmp_t4], %[tmp_t2],     %[tmp_ub]  \n"
>-        "mul.ph           %[tmp_t5], %[tmp_t2],     %[tmp_ug]  \n"
>-        "mul.ph           %[tmp_t6], %[tmp_t3],     %[tmp_vr]  \n"
>-        "mul.ph           %[tmp_t3], %[tmp_t3],     %[tmp_vg]  \n"
>-        "srl              %[tmp_t7], %[tmp_t7],     16         \n"
>-        "ins              %[tmp_t1], %[tmp_t7],     0,      16 \n"
>-        "addq_s.ph        %[tmp_t7], %[tmp_t1],     %[tmp_bb]  \n"
>-        "addq_s.ph        %[tmp_t8], %[tmp_t1],     %[tmp_bg]  \n"
>-        "addq_s.ph        %[tmp_t9], %[tmp_t1],     %[tmp_br]  \n"
>-        "addq_s.ph        %[tmp_t5], %[tmp_t5],     %[tmp_t3]  \n"
>-        "addq_s.ph        %[tmp_t7], %[tmp_t7],     %[tmp_t4]  \n"
>-        "subq_s.ph        %[tmp_t8], %[tmp_t8],     %[tmp_t5]  \n"
>-        "addq_s.ph        %[tmp_t9], %[tmp_t9],     %[tmp_t6]  \n"
>-        "shra.ph          %[tmp_t7], %[tmp_t7],     6          \n"
>-        "shra.ph          %[tmp_t8], %[tmp_t8],     6          \n"
>-        "shra.ph          %[tmp_t9], %[tmp_t9],     6          \n"
>-        "shll_s.ph        %[tmp_t7], %[tmp_t7],     7          \n"
>-        "shll_s.ph        %[tmp_t8], %[tmp_t8],     7          \n"
>-        "shll_s.ph        %[tmp_t9], %[tmp_t9],     7          \n"
>-        "precrqu_s.qb.ph  %[tmp_t8], %[tmp_mask],   %[tmp_t8]  \n"
>-        "precrqu_s.qb.ph  %[tmp_t7], %[tmp_t9],     %[tmp_t7]  \n"
>-        "precrq.ph.w      %[tmp_t2], %[tmp_t8],     %[tmp_t7]  \n"
>-        "ins              %[tmp_t7], %[tmp_t8],     16,     16 \n"
>-        "precr.qb.ph      %[tmp_t8], %[tmp_t2],     %[tmp_t7]  \n"
>-        "precrq.qb.ph     %[tmp_t7], %[tmp_t2],     %[tmp_t7]  \n"
>-        "ins              %[tmp_t3], %[tmp_t8],     7,      24 \n"
>-        "ins              %[tmp_t3], %[tmp_t8],     10,     16 \n"
>-        "ins              %[tmp_t3], %[tmp_t8],     13,     8  \n"
>-        "ins              %[tmp_t4], %[tmp_t7],     7,      24 \n"
>-        "ins              %[tmp_t4], %[tmp_t7],     10,     16 \n"
>-        "ins              %[tmp_t4], %[tmp_t7],     13,     8  \n"
>-        "precrq.ph.w      %[tmp_t8], %[tmp_t4],     %[tmp_t3]  \n"
>-        "or               %[tmp_t8], %[tmp_t8],     %[tmp_mask]\n"
>-        "sw               %[tmp_t8], 0(%[dst_argb1555])        \n"
>-        ".set pop                                              \n"
>-        : [tmp_t1] "=&r"(tmp_t1), [tmp_t2] "=&r"(tmp_t2),
>-          [tmp_t3] "=&r"(tmp_t3), [tmp_t4] "=&r"(tmp_t4),
>-          [tmp_t5] "=&r"(tmp_t5), [tmp_t6] "=&r"(tmp_t6),
>-          [tmp_t7] "=&r"(tmp_t7), [tmp_t8] "=&r"(tmp_t8), [tmp_t9] "=&r"(tmp_t9)
>-        : [dst_argb1555] "r"(dst_argb1555), [yg] "r"(yg), [src_u] "r"(src_u),
>-          [src_v] "r"(src_v), [src_y] "r"(src_y), [tmp_ub] "r"(tmp_ub),
>-          [tmp_ug] "r"(tmp_ug), [tmp_vg] "r"(tmp_vg), [tmp_vr] "r"(tmp_vr),
>-          [tmp_bb] "r"(tmp_bb), [tmp_bg] "r"(tmp_bg), [tmp_br] "r"(tmp_br),
>-          [tmp_yg] "r"(tmp_yg), [tmp_mask] "r"(tmp_mask));
>-    src_y += 2;
>-    src_u += 1;
>-    src_v += 1;
>-    dst_argb1555 += 4;  // Advance 2 pixels.
>-  }
>-}
>-
>-void NV12ToARGBRow_DSPR2(const uint8* src_y,
>-                         const uint8* src_uv,
>-                         uint8* rgb_buf,
>-                         const struct YuvConstants* yuvconstants,
>-                         int width) {
>-  int x;
>-  uint32 tmp_ub = yuvconstants->kUVToB[0];
>-  uint32 tmp_ug = yuvconstants->kUVToG[0];
>-  uint32 tmp_vg = yuvconstants->kUVToG[1];
>-  uint32 tmp_vr = yuvconstants->kUVToR[1];
>-  uint32 tmp_bb = yuvconstants->kUVBiasB[0];
>-  uint32 tmp_bg = yuvconstants->kUVBiasG[0];
>-  uint32 tmp_br = yuvconstants->kUVBiasR[0];
>-  uint32 yg = yuvconstants->kYToRgb[0];
>-  uint32 tmp_mask = 0x7fff7fff;
>-  uint32 tmp_yg;
>-  tmp_bb = ((uint)(tmp_bb & 0xffff) << 16) | (tmp_bb & 0xffff);
>-  tmp_bg = ((uint)(tmp_bg & 0xffff) << 16) | (tmp_bg & 0xffff);
>-  tmp_br = ((uint)(tmp_br & 0xffff) << 16) | (tmp_br & 0xffff);
>-  tmp_yg = ((uint)(yg & 0xffff) << 16) | (yg & 0xffff);
>-  tmp_ub = ~(((uint)(tmp_ub & 0xffff) << 16) | (tmp_ub & 0xffff)) + 0x00010001;
>-  tmp_ug = ((uint)(tmp_ug & 0xffff) << 16) | (tmp_ug & 0xffff);
>-  tmp_vg = ((uint)(tmp_vg & 0xffff) << 16) | (tmp_vg & 0xffff);
>-  tmp_vr = ~(((uint)(tmp_vr & 0xffff) << 16) | (tmp_vr & 0xffff)) + 0x00010001;
>-  yg = yg * 0x0101;
>-
>-  for (x = 0; x < width - 1; x += 2) {
>-    uint32 tmp_t1, tmp_t2, tmp_t3, tmp_t4, tmp_t5;
>-    uint32 tmp_t6, tmp_t7, tmp_t8, tmp_t9;
>-    __asm__ __volatile__(
>-        ".set push                                             \n"
>-        ".set noreorder                                        \n"
>-        "lbu              %[tmp_t7], 0(%[src_y])               \n"
>-        "lbu              %[tmp_t1], 1(%[src_y])               \n"
>-        "mul              %[tmp_t7], %[tmp_t7],     %[yg]      \n"
>-        "mul              %[tmp_t1], %[tmp_t1],     %[yg]      \n"
>-        "lbu              %[tmp_t2], 0(%[src_uv])              \n"
>-        "lbu              %[tmp_t3], 1(%[src_uv])              \n"
>-        "replv.ph         %[tmp_t2], %[tmp_t2]                 \n"
>-        "replv.ph         %[tmp_t3], %[tmp_t3]                 \n"
>-        "mul.ph           %[tmp_t4], %[tmp_t2],     %[tmp_ub]  \n"
>-        "mul.ph           %[tmp_t5], %[tmp_t2],     %[tmp_ug]  \n"
>-        "mul.ph           %[tmp_t6], %[tmp_t3],     %[tmp_vr]  \n"
>-        "mul.ph           %[tmp_t3], %[tmp_t3],     %[tmp_vg]  \n"
>-        "srl              %[tmp_t7], %[tmp_t7],     16         \n"
>-        "ins              %[tmp_t1], %[tmp_t7],     0,      16 \n"
>-        "addq_s.ph        %[tmp_t7], %[tmp_t1],     %[tmp_bb]  \n"
>-        "addq_s.ph        %[tmp_t8], %[tmp_t1],     %[tmp_bg]  \n"
>-        "addq_s.ph        %[tmp_t9], %[tmp_t1],     %[tmp_br]  \n"
>-        "addq_s.ph        %[tmp_t5], %[tmp_t5],     %[tmp_t3]  \n"
>-        "addq_s.ph        %[tmp_t7], %[tmp_t7],     %[tmp_t4]  \n"
>-        "subq_s.ph        %[tmp_t8], %[tmp_t8],     %[tmp_t5]  \n"
>-        "addq_s.ph        %[tmp_t9], %[tmp_t9],     %[tmp_t6]  \n"
>-        "shra.ph          %[tmp_t7], %[tmp_t7],     6          \n"
>-        "shra.ph          %[tmp_t8], %[tmp_t8],     6          \n"
>-        "shra.ph          %[tmp_t9], %[tmp_t9],     6          \n"
>-        "shll_s.ph        %[tmp_t7], %[tmp_t7],     7          \n"
>-        "shll_s.ph        %[tmp_t8], %[tmp_t8],     7          \n"
>-        "shll_s.ph        %[tmp_t9], %[tmp_t9],     7          \n"
>-        "precrqu_s.qb.ph  %[tmp_t8], %[tmp_mask],   %[tmp_t8]  \n"
>-        "precrqu_s.qb.ph  %[tmp_t7], %[tmp_t9],     %[tmp_t7]  \n"
>-        "precrq.ph.w      %[tmp_t2], %[tmp_t8],     %[tmp_t7]  \n"
>-        "ins              %[tmp_t7], %[tmp_t8],     16,     16 \n"
>-        "precr.qb.ph      %[tmp_t8], %[tmp_t2],     %[tmp_t7]  \n"
>-        "precrq.qb.ph     %[tmp_t7], %[tmp_t2],     %[tmp_t7]  \n"
>-        "sw               %[tmp_t8], 0(%[rgb_buf])             \n"
>-        "sw               %[tmp_t7], 4(%[rgb_buf])             \n"
>-        ".set pop                                              \n"
>-        : [tmp_t1] "=&r"(tmp_t1), [tmp_t2] "=&r"(tmp_t2),
>-          [tmp_t3] "=&r"(tmp_t3), [tmp_t4] "=&r"(tmp_t4),
>-          [tmp_t5] "=&r"(tmp_t5), [tmp_t6] "=&r"(tmp_t6),
>-          [tmp_t7] "=&r"(tmp_t7), [tmp_t8] "=&r"(tmp_t8), [tmp_t9] "=&r"(tmp_t9)
>-        : [src_y] "r"(src_y), [src_uv] "r"(src_uv), [yg] "r"(yg),
>-          [tmp_ub] "r"(tmp_ub), [tmp_ug] "r"(tmp_ug), [tmp_vg] "r"(tmp_vg),
>-          [tmp_vr] "r"(tmp_vr), [tmp_bb] "r"(tmp_bb), [tmp_bg] "r"(tmp_bg),
>-          [tmp_br] "r"(tmp_br), [tmp_yg] "r"(tmp_yg), [rgb_buf] "r"(rgb_buf),
>-          [tmp_mask] "r"(tmp_mask));
>-
>-    src_y += 2;
>-    src_uv += 2;
>-    rgb_buf += 8;  // Advance 2 pixels.
>-  }
>-}
>-
>-void BGRAToUVRow_DSPR2(const uint8* src_rgb0,
>-                       int src_stride_rgb,
>-                       uint8* dst_u,
>-                       uint8* dst_v,
>-                       int width) {
>-  const uint8* src_rgb1 = src_rgb0 + src_stride_rgb;
>-  int x;
>-  int const1 = 0xffda0000;
>-  int const2 = 0x0070ffb6;
>-  int const3 = 0x00700000;
>-  int const4 = 0xffeeffa2;
>-  int const5 = 0x100;
>-  for (x = 0; x < width - 1; x += 2) {
>-    int tmp_t1, tmp_t2, tmp_t3, tmp_t4, tmp_t5;
>-    int tmp_t6, tmp_t7, tmp_t8;
>-    __asm__ __volatile__(
>-        ".set push                                                 \n"
>-        ".set noreorder                                            \n"
>-        "lw                %[tmp_t1],   0(%[src_rgb0])             \n"
>-        "lw                %[tmp_t2],   4(%[src_rgb0])             \n"
>-        "lw                %[tmp_t3],   0(%[src_rgb1])             \n"
>-        "lw                %[tmp_t4],   4(%[src_rgb1])             \n"
>-        "preceu.ph.qbr     %[tmp_t5],   %[tmp_t1]                  \n"
>-        "preceu.ph.qbl     %[tmp_t1],   %[tmp_t1]                  \n"
>-        "preceu.ph.qbr     %[tmp_t6],   %[tmp_t2]                  \n"
>-        "preceu.ph.qbl     %[tmp_t2],   %[tmp_t2]                  \n"
>-        "preceu.ph.qbr     %[tmp_t7],   %[tmp_t3]                  \n"
>-        "preceu.ph.qbl     %[tmp_t3],   %[tmp_t3]                  \n"
>-        "preceu.ph.qbr     %[tmp_t8],   %[tmp_t4]                  \n"
>-        "preceu.ph.qbl     %[tmp_t4],   %[tmp_t4]                  \n"
>-        "addu.ph           %[tmp_t5],   %[tmp_t5],     %[tmp_t6]   \n"
>-        "addu.ph           %[tmp_t7],   %[tmp_t7],     %[tmp_t8]   \n"
>-        "addu.ph           %[tmp_t1],   %[tmp_t1],     %[tmp_t2]   \n"
>-        "addu.ph           %[tmp_t3],   %[tmp_t3],     %[tmp_t4]   \n"
>-        "addu.ph           %[tmp_t5],   %[tmp_t5],     %[tmp_t7]   \n"
>-        "addu.ph           %[tmp_t1],   %[tmp_t1],     %[tmp_t3]   \n"
>-        "shrl.ph           %[tmp_t5],   %[tmp_t5],     2           \n"
>-        "shrl.ph           %[tmp_t1],   %[tmp_t1],     2           \n"
>-        "mult              $ac0,        %[const5],     %[const5]   \n"
>-        "mult              $ac1,        %[const5],     %[const5]   \n"
>-        "dpaq_s.w.ph       $ac0,        %[tmp_t5],     %[const1]   \n"
>-        "dpaq_s.w.ph       $ac1,        %[tmp_t5],     %[const3]   \n"
>-        "dpaq_s.w.ph       $ac0,        %[tmp_t1],     %[const2]   \n"
>-        "dpaq_s.w.ph       $ac1,        %[tmp_t1],     %[const4]   \n"
>-        "extr_r.w          %[tmp_t7],   $ac0,          9           \n"
>-        "extr_r.w          %[tmp_t8],   $ac1,          9           \n"
>-        "addiu             %[dst_u],    %[dst_u],    1             \n"
>-        "addiu             %[dst_v],    %[dst_v],    1             \n"
>-        "addiu             %[src_rgb0], %[src_rgb0], 8             \n"
>-        "addiu             %[src_rgb1], %[src_rgb1], 8             \n"
>-        "sb                %[tmp_t7],   -1(%[dst_u])               \n"
>-        "sb                %[tmp_t8],   -1(%[dst_v])               \n"
>-        ".set pop                                                  \n"
>-        : [tmp_t1] "=&r"(tmp_t1), [tmp_t2] "=&r"(tmp_t2),
>-          [tmp_t3] "=&r"(tmp_t3), [tmp_t4] "=&r"(tmp_t4),
>-          [tmp_t5] "=&r"(tmp_t5), [tmp_t6] "=&r"(tmp_t6),
>-          [tmp_t7] "=&r"(tmp_t7), [tmp_t8] "=&r"(tmp_t8),
>-          [src_rgb0] "+r"(src_rgb0), [src_rgb1] "+r"(src_rgb1),
>-          [dst_u] "+r"(dst_u), [dst_v] "+r"(dst_v)
>-        : [const1] "r"(const1), [const2] "r"(const2), [const3] "r"(const3),
>-          [const4] "r"(const4), [const5] "r"(const5)
>-        : "hi", "lo", "$ac1lo", "$ac1hi");
>-  }
>-}
>-
>-void BGRAToYRow_DSPR2(const uint8* src_argb0, uint8* dst_y, int width) {
>-  int x;
>-  int const1 = 0x00420000;
>-  int const2 = 0x00190081;
>-  int const5 = 0x40;
>-  for (x = 0; x < width; x += 4) {
>-    int tmp_t1, tmp_t2, tmp_t3, tmp_t4, tmp_t5;
>-    int tmp_t6, tmp_t7, tmp_t8;
>-    __asm__ __volatile__(
>-        ".set push                                                \n"
>-        ".set noreorder                                           \n"
>-        "lw                %[tmp_t1],   0(%[src_argb0])           \n"
>-        "lw                %[tmp_t2],   4(%[src_argb0])           \n"
>-        "lw                %[tmp_t3],   8(%[src_argb0])           \n"
>-        "lw                %[tmp_t4],   12(%[src_argb0])          \n"
>-        "preceu.ph.qbr     %[tmp_t5],   %[tmp_t1]                 \n"
>-        "preceu.ph.qbl     %[tmp_t1],   %[tmp_t1]                 \n"
>-        "preceu.ph.qbr     %[tmp_t6],   %[tmp_t2]                 \n"
>-        "preceu.ph.qbl     %[tmp_t2],   %[tmp_t2]                 \n"
>-        "preceu.ph.qbr     %[tmp_t7],   %[tmp_t3]                 \n"
>-        "preceu.ph.qbl     %[tmp_t3],   %[tmp_t3]                 \n"
>-        "preceu.ph.qbr     %[tmp_t8],   %[tmp_t4]                 \n"
>-        "preceu.ph.qbl     %[tmp_t4],   %[tmp_t4]                 \n"
>-        "mult              $ac0,        %[const5],     %[const5]  \n"
>-        "mult              $ac1,        %[const5],     %[const5]  \n"
>-        "mult              $ac2,        %[const5],     %[const5]  \n"
>-        "mult              $ac3,        %[const5],     %[const5]  \n"
>-        "dpa.w.ph          $ac0,        %[tmp_t5],     %[const1]  \n"
>-        "dpa.w.ph          $ac1,        %[tmp_t6],     %[const1]  \n"
>-        "dpa.w.ph          $ac2,        %[tmp_t7],     %[const1]  \n"
>-        "dpa.w.ph          $ac3,        %[tmp_t8],     %[const1]  \n"
>-        "dpa.w.ph          $ac0,        %[tmp_t1],     %[const2]  \n"
>-        "dpa.w.ph          $ac1,        %[tmp_t2],     %[const2]  \n"
>-        "dpa.w.ph          $ac2,        %[tmp_t3],     %[const2]  \n"
>-        "dpa.w.ph          $ac3,        %[tmp_t4],     %[const2]  \n"
>-        "extr_r.w          %[tmp_t1],   $ac0,          8          \n"
>-        "extr_r.w          %[tmp_t2],   $ac1,          8          \n"
>-        "extr_r.w          %[tmp_t3],   $ac2,          8          \n"
>-        "extr_r.w          %[tmp_t4],   $ac3,          8          \n"
>-        "addiu             %[src_argb0],%[src_argb0],  16         \n"
>-        "addiu             %[dst_y],    %[dst_y],      4          \n"
>-        "sb                %[tmp_t1],   -4(%[dst_y])              \n"
>-        "sb                %[tmp_t2],   -3(%[dst_y])              \n"
>-        "sb                %[tmp_t3],   -2(%[dst_y])              \n"
>-        "sb                %[tmp_t4],   -1(%[dst_y])              \n"
>-        ".set pop                                                 \n"
>-        : [tmp_t1] "=&r"(tmp_t1), [tmp_t2] "=&r"(tmp_t2),
>-          [tmp_t3] "=&r"(tmp_t3), [tmp_t4] "=&r"(tmp_t4),
>-          [tmp_t5] "=&r"(tmp_t5), [tmp_t6] "=&r"(tmp_t6),
>-          [tmp_t7] "=&r"(tmp_t7), [tmp_t8] "=&r"(tmp_t8),
>-          [src_argb0] "+r"(src_argb0), [dst_y] "+r"(dst_y)
>-        : [const1] "r"(const1), [const2] "r"(const2), [const5] "r"(const5)
>-        : "hi", "lo", "$ac1lo", "$ac1hi", "$ac2lo", "$ac2hi", "$ac3lo",
>-          "$ac3hi");
>-  }
>-}
>-
>-void ABGRToUVRow_DSPR2(const uint8* src_rgb0,
>-                       int src_stride_rgb,
>-                       uint8* dst_u,
>-                       uint8* dst_v,
>-                       int width) {
>-  const uint8* src_rgb1 = src_rgb0 + src_stride_rgb;
>-  int x;
>-  int const1 = 0xffb6ffda;
>-  int const2 = 0x00000070;
>-  int const3 = 0xffa20070;
>-  int const4 = 0x0000ffee;
>-  int const5 = 0x100;
>-
>-  for (x = 0; x < width - 1; x += 2) {
>-    int tmp_t1, tmp_t2, tmp_t3, tmp_t4, tmp_t5;
>-    int tmp_t6, tmp_t7, tmp_t8;
>-    __asm__ __volatile__(
>-        ".set push                                                \n"
>-        ".set noreorder                                           \n"
>-        "lw                %[tmp_t1],   0(%[src_rgb0])            \n"
>-        "lw                %[tmp_t2],   4(%[src_rgb0])            \n"
>-        "lw                %[tmp_t3],   0(%[src_rgb1])            \n"
>-        "lw                %[tmp_t4],   4(%[src_rgb1])            \n"
>-        "preceu.ph.qbr     %[tmp_t5],   %[tmp_t1]                 \n"
>-        "preceu.ph.qbl     %[tmp_t1],   %[tmp_t1]                 \n"
>-        "preceu.ph.qbr     %[tmp_t6],   %[tmp_t2]                 \n"
>-        "preceu.ph.qbl     %[tmp_t2],   %[tmp_t2]                 \n"
>-        "preceu.ph.qbr     %[tmp_t7],   %[tmp_t3]                 \n"
>-        "preceu.ph.qbl     %[tmp_t3],   %[tmp_t3]                 \n"
>-        "preceu.ph.qbr     %[tmp_t8],   %[tmp_t4]                 \n"
>-        "preceu.ph.qbl     %[tmp_t4],   %[tmp_t4]                 \n"
>-        "addu.ph           %[tmp_t5],   %[tmp_t5],     %[tmp_t6]  \n"
>-        "addu.ph           %[tmp_t7],   %[tmp_t7],     %[tmp_t8]  \n"
>-        "addu.ph           %[tmp_t1],   %[tmp_t1],     %[tmp_t2]  \n"
>-        "addu.ph           %[tmp_t3],   %[tmp_t3],     %[tmp_t4]  \n"
>-        "addu.ph           %[tmp_t5],   %[tmp_t5],     %[tmp_t7]  \n"
>-        "addu.ph           %[tmp_t1],   %[tmp_t1],     %[tmp_t3]  \n"
>-        "shrl.ph           %[tmp_t5],   %[tmp_t5],     2          \n"
>-        "shrl.ph           %[tmp_t1],   %[tmp_t1],     2          \n"
>-        "mult              $ac0,        %[const5],     %[const5]  \n"
>-        "mult              $ac1,        %[const5],     %[const5]  \n"
>-        "dpaq_s.w.ph       $ac0,        %[tmp_t5],     %[const1]  \n"
>-        "dpaq_s.w.ph       $ac1,        %[tmp_t5],     %[const3]  \n"
>-        "dpaq_s.w.ph       $ac0,        %[tmp_t1],     %[const2]  \n"
>-        "dpaq_s.w.ph       $ac1,        %[tmp_t1],     %[const4]  \n"
>-        "extr_r.w          %[tmp_t7],   $ac0,          9          \n"
>-        "extr_r.w          %[tmp_t8],   $ac1,          9          \n"
>-        "addiu             %[dst_u],    %[dst_u],    1            \n"
>-        "addiu             %[dst_v],    %[dst_v],    1            \n"
>-        "addiu             %[src_rgb0], %[src_rgb0], 8            \n"
>-        "addiu             %[src_rgb1], %[src_rgb1], 8            \n"
>-        "sb                %[tmp_t7],   -1(%[dst_u])              \n"
>-        "sb                %[tmp_t8],   -1(%[dst_v])              \n"
>-        ".set pop                                                 \n"
>-        : [tmp_t1] "=&r"(tmp_t1), [tmp_t2] "=&r"(tmp_t2),
>-          [tmp_t3] "=&r"(tmp_t3), [tmp_t4] "=&r"(tmp_t4),
>-          [tmp_t5] "=&r"(tmp_t5), [tmp_t6] "=&r"(tmp_t6),
>-          [tmp_t7] "=&r"(tmp_t7), [tmp_t8] "=&r"(tmp_t8),
>-          [src_rgb0] "+r"(src_rgb0), [src_rgb1] "+r"(src_rgb1),
>-          [dst_u] "+r"(dst_u), [dst_v] "+r"(dst_v)
>-        : [const1] "r"(const1), [const2] "r"(const2), [const3] "r"(const3),
>-          [const4] "r"(const4), [const5] "r"(const5)
>-        : "hi", "lo", "$ac1lo", "$ac1hi");
>-  }
>-}
>-
>-void ARGBToYRow_DSPR2(const uint8* src_argb0, uint8* dst_y, int width) {
>-  int x;
>-  int const1 = 0x00810019;
>-  int const2 = 0x00000042;
>-  int const5 = 0x40;
>-  for (x = 0; x < width; x += 4) {
>-    int tmp_t1, tmp_t2, tmp_t3, tmp_t4, tmp_t5;
>-    int tmp_t6, tmp_t7, tmp_t8;
>-    __asm__ __volatile__(
>-        ".set push                                                \n"
>-        ".set noreorder                                           \n"
>-        "lw                %[tmp_t1],   0(%[src_argb0])           \n"
>-        "lw                %[tmp_t2],   4(%[src_argb0])           \n"
>-        "lw                %[tmp_t3],   8(%[src_argb0])           \n"
>-        "lw                %[tmp_t4],   12(%[src_argb0])          \n"
>-        "preceu.ph.qbr     %[tmp_t5],   %[tmp_t1]                 \n"
>-        "preceu.ph.qbl     %[tmp_t1],   %[tmp_t1]                 \n"
>-        "preceu.ph.qbr     %[tmp_t6],   %[tmp_t2]                 \n"
>-        "preceu.ph.qbl     %[tmp_t2],   %[tmp_t2]                 \n"
>-        "preceu.ph.qbr     %[tmp_t7],   %[tmp_t3]                 \n"
>-        "preceu.ph.qbl     %[tmp_t3],   %[tmp_t3]                 \n"
>-        "preceu.ph.qbr     %[tmp_t8],   %[tmp_t4]                 \n"
>-        "preceu.ph.qbl     %[tmp_t4],   %[tmp_t4]                 \n"
>-        "mult              $ac0,        %[const5],     %[const5]  \n"
>-        "mult              $ac1,        %[const5],     %[const5]  \n"
>-        "mult              $ac2,        %[const5],     %[const5]  \n"
>-        "mult              $ac3,        %[const5],     %[const5]  \n"
>-        "dpa.w.ph          $ac0,        %[tmp_t5],     %[const1]  \n"
>-        "dpa.w.ph          $ac1,        %[tmp_t6],     %[const1]  \n"
>-        "dpa.w.ph          $ac2,        %[tmp_t7],     %[const1]  \n"
>-        "dpa.w.ph          $ac3,        %[tmp_t8],     %[const1]  \n"
>-        "dpa.w.ph          $ac0,        %[tmp_t1],     %[const2]  \n"
>-        "dpa.w.ph          $ac1,        %[tmp_t2],     %[const2]  \n"
>-        "dpa.w.ph          $ac2,        %[tmp_t3],     %[const2]  \n"
>-        "dpa.w.ph          $ac3,        %[tmp_t4],     %[const2]  \n"
>-        "extr_r.w          %[tmp_t1],   $ac0,          8          \n"
>-        "extr_r.w          %[tmp_t2],   $ac1,          8          \n"
>-        "extr_r.w          %[tmp_t3],   $ac2,          8          \n"
>-        "extr_r.w          %[tmp_t4],   $ac3,          8          \n"
>-        "addiu             %[dst_y],    %[dst_y],      4          \n"
>-        "addiu             %[src_argb0],%[src_argb0],  16         \n"
>-        "sb                %[tmp_t1],   -4(%[dst_y])              \n"
>-        "sb                %[tmp_t2],   -3(%[dst_y])              \n"
>-        "sb                %[tmp_t3],   -2(%[dst_y])              \n"
>-        "sb                %[tmp_t4],   -1(%[dst_y])              \n"
>-        ".set pop                                                 \n"
>-        : [tmp_t1] "=&r"(tmp_t1), [tmp_t2] "=&r"(tmp_t2),
>-          [tmp_t3] "=&r"(tmp_t3), [tmp_t4] "=&r"(tmp_t4),
>-          [tmp_t5] "=&r"(tmp_t5), [tmp_t6] "=&r"(tmp_t6),
>-          [tmp_t7] "=&r"(tmp_t7), [tmp_t8] "=&r"(tmp_t8),
>-          [src_argb0] "+r"(src_argb0), [dst_y] "+r"(dst_y)
>-        : [const1] "r"(const1), [const2] "r"(const2), [const5] "r"(const5)
>-        : "hi", "lo", "$ac1lo", "$ac1hi", "$ac2lo", "$ac2hi", "$ac3lo",
>-          "$ac3hi");
>-  }
>-}
>-
>-void ABGRToYRow_DSPR2(const uint8* src_argb0, uint8* dst_y, int width) {
>-  int x;
>-  int const1 = 0x00810042;
>-  int const2 = 0x00000019;
>-  int const5 = 0x40;
>-  for (x = 0; x < width; x += 4) {
>-    int tmp_t1, tmp_t2, tmp_t3, tmp_t4, tmp_t5;
>-    int tmp_t6, tmp_t7, tmp_t8;
>-    __asm__ __volatile__(
>-        ".set push                                                \n"
>-        ".set noreorder                                           \n"
>-        "lw                %[tmp_t1],   0(%[src_argb0])           \n"
>-        "lw                %[tmp_t2],   4(%[src_argb0])           \n"
>-        "lw                %[tmp_t3],   8(%[src_argb0])           \n"
>-        "lw                %[tmp_t4],   12(%[src_argb0])          \n"
>-        "preceu.ph.qbr     %[tmp_t5],   %[tmp_t1]                 \n"
>-        "preceu.ph.qbl     %[tmp_t1],   %[tmp_t1]                 \n"
>-        "preceu.ph.qbr     %[tmp_t6],   %[tmp_t2]                 \n"
>-        "preceu.ph.qbl     %[tmp_t2],   %[tmp_t2]                 \n"
>-        "preceu.ph.qbr     %[tmp_t7],   %[tmp_t3]                 \n"
>-        "preceu.ph.qbl     %[tmp_t3],   %[tmp_t3]                 \n"
>-        "preceu.ph.qbr     %[tmp_t8],   %[tmp_t4]                 \n"
>-        "preceu.ph.qbl     %[tmp_t4],   %[tmp_t4]                 \n"
>-        "mult              $ac0,        %[const5],     %[const5]  \n"
>-        "mult              $ac1,        %[const5],     %[const5]  \n"
>-        "mult              $ac2,        %[const5],     %[const5]  \n"
>-        "mult              $ac3,        %[const5],     %[const5]  \n"
>-        "dpa.w.ph          $ac0,        %[tmp_t5],     %[const1]  \n"
>-        "dpa.w.ph          $ac1,        %[tmp_t6],     %[const1]  \n"
>-        "dpa.w.ph          $ac2,        %[tmp_t7],     %[const1]  \n"
>-        "dpa.w.ph          $ac3,        %[tmp_t8],     %[const1]  \n"
>-        "dpa.w.ph          $ac0,        %[tmp_t1],     %[const2]  \n"
>-        "dpa.w.ph          $ac1,        %[tmp_t2],     %[const2]  \n"
>-        "dpa.w.ph          $ac2,        %[tmp_t3],     %[const2]  \n"
>-        "dpa.w.ph          $ac3,        %[tmp_t4],     %[const2]  \n"
>-        "extr_r.w          %[tmp_t1],   $ac0,          8          \n"
>-        "extr_r.w          %[tmp_t2],   $ac1,          8          \n"
>-        "extr_r.w          %[tmp_t3],   $ac2,          8          \n"
>-        "extr_r.w          %[tmp_t4],   $ac3,          8          \n"
>-        "addiu             %[src_argb0],%[src_argb0],  16         \n"
>-        "addiu             %[dst_y],    %[dst_y],      4          \n"
>-        "sb                %[tmp_t1],   -4(%[dst_y])              \n"
>-        "sb                %[tmp_t2],   -3(%[dst_y])              \n"
>-        "sb                %[tmp_t3],   -2(%[dst_y])              \n"
>-        "sb                %[tmp_t4],   -1(%[dst_y])              \n"
>-        ".set pop                                                 \n"
>-        : [tmp_t1] "=&r"(tmp_t1), [tmp_t2] "=&r"(tmp_t2),
>-          [tmp_t3] "=&r"(tmp_t3), [tmp_t4] "=&r"(tmp_t4),
>-          [tmp_t5] "=&r"(tmp_t5), [tmp_t6] "=&r"(tmp_t6),
>-          [tmp_t7] "=&r"(tmp_t7), [tmp_t8] "=&r"(tmp_t8),
>-          [src_argb0] "+r"(src_argb0), [dst_y] "+r"(dst_y)
>-        : [const1] "r"(const1), [const2] "r"(const2), [const5] "r"(const5)
>-        : "hi", "lo", "$ac1lo", "$ac1hi", "$ac2lo", "$ac2hi", "$ac3lo",
>-          "$ac3hi");
>-  }
>-}
>-
>-void RGBAToUVRow_DSPR2(const uint8* src_rgb0,
>-                       int src_stride_rgb,
>-                       uint8* dst_u,
>-                       uint8* dst_v,
>-                       int width) {
>-  const uint8* src_rgb1 = src_rgb0 + src_stride_rgb;
>-  int x;
>-  int const1 = 0xffb60070;
>-  int const2 = 0x0000ffda;
>-  int const3 = 0xffa2ffee;
>-  int const4 = 0x00000070;
>-  int const5 = 0x100;
>-
>-  for (x = 0; x < width - 1; x += 2) {
>-    int tmp_t1, tmp_t2, tmp_t3, tmp_t4, tmp_t5;
>-    int tmp_t6, tmp_t7, tmp_t8;
>-    __asm__ __volatile__(
>-        ".set push                                                \n"
>-        ".set noreorder                                           \n"
>-        "ulw               %[tmp_t1],   0+1(%[src_rgb0])          \n"
>-        "ulw               %[tmp_t2],   4+1(%[src_rgb0])          \n"
>-        "ulw               %[tmp_t3],   0+1(%[src_rgb1])          \n"
>-        "ulw               %[tmp_t4],   4+1(%[src_rgb1])          \n"
>-        "preceu.ph.qbr     %[tmp_t5],   %[tmp_t1]                 \n"
>-        "preceu.ph.qbl     %[tmp_t1],   %[tmp_t1]                 \n"
>-        "preceu.ph.qbr     %[tmp_t6],   %[tmp_t2]                 \n"
>-        "preceu.ph.qbl     %[tmp_t2],   %[tmp_t2]                 \n"
>-        "preceu.ph.qbr     %[tmp_t7],   %[tmp_t3]                 \n"
>-        "preceu.ph.qbl     %[tmp_t3],   %[tmp_t3]                 \n"
>-        "preceu.ph.qbr     %[tmp_t8],   %[tmp_t4]                 \n"
>-        "preceu.ph.qbl     %[tmp_t4],   %[tmp_t4]                 \n"
>-        "addu.ph           %[tmp_t5],   %[tmp_t5],     %[tmp_t6]  \n"
>-        "addu.ph           %[tmp_t7],   %[tmp_t7],     %[tmp_t8]  \n"
>-        "addu.ph           %[tmp_t1],   %[tmp_t1],     %[tmp_t2]  \n"
>-        "addu.ph           %[tmp_t3],   %[tmp_t3],     %[tmp_t4]  \n"
>-        "addu.ph           %[tmp_t5],   %[tmp_t5],     %[tmp_t7]  \n"
>-        "addu.ph           %[tmp_t1],   %[tmp_t1],     %[tmp_t3]  \n"
>-        "shrl.ph           %[tmp_t5],   %[tmp_t5],     2          \n"
>-        "shrl.ph           %[tmp_t1],   %[tmp_t1],     2          \n"
>-        "mult              $ac0,        %[const5],     %[const5]  \n"
>-        "mult              $ac1,        %[const5],     %[const5]  \n"
>-        "dpaq_s.w.ph       $ac0,        %[tmp_t5],     %[const1]  \n"
>-        "dpaq_s.w.ph       $ac1,        %[tmp_t5],     %[const3]  \n"
>-        "dpaq_s.w.ph       $ac0,        %[tmp_t1],     %[const2]  \n"
>-        "dpaq_s.w.ph       $ac1,        %[tmp_t1],     %[const4]  \n"
>-        "extr_r.w          %[tmp_t7],   $ac0,          9          \n"
>-        "extr_r.w          %[tmp_t8],   $ac1,          9          \n"
>-        "addiu             %[src_rgb0], %[src_rgb0], 8            \n"
>-        "addiu             %[src_rgb1], %[src_rgb1], 8            \n"
>-        "addiu             %[dst_u],    %[dst_u],    1            \n"
>-        "addiu             %[dst_v],    %[dst_v],    1            \n"
>-        "sb                %[tmp_t7],   -1(%[dst_u])              \n"
>-        "sb                %[tmp_t8],   -1(%[dst_v])              \n"
>-        ".set pop                                                 \n"
>-        : [tmp_t1] "=&r"(tmp_t1), [tmp_t2] "=&r"(tmp_t2),
>-          [tmp_t3] "=&r"(tmp_t3), [tmp_t4] "=&r"(tmp_t4),
>-          [tmp_t5] "=&r"(tmp_t5), [tmp_t6] "=&r"(tmp_t6),
>-          [tmp_t7] "=&r"(tmp_t7), [tmp_t8] "=&r"(tmp_t8),
>-          [src_rgb0] "+r"(src_rgb0), [src_rgb1] "+r"(src_rgb1),
>-          [dst_u] "+r"(dst_u), [dst_v] "+r"(dst_v)
>-        : [const1] "r"(const1), [const2] "r"(const2), [const3] "r"(const3),
>-          [const4] "r"(const4), [const5] "r"(const5)
>-        : "hi", "lo", "$ac1lo", "$ac1hi");
>-  }
>-}
>-
>-void RGBAToYRow_DSPR2(const uint8* src_argb0, uint8* dst_y, int width) {
>-  int x;
>-  int const1 = 0x00420081;
>-  int const2 = 0x00190000;
>-  int const5 = 0x40;
>-  for (x = 0; x < width; x += 4) {
>-    int tmp_t1, tmp_t2, tmp_t3, tmp_t4, tmp_t5;
>-    int tmp_t6, tmp_t7, tmp_t8;
>-    __asm__ __volatile__(
>-        ".set push                                                \n"
>-        ".set noreorder                                           \n"
>-        "lw                %[tmp_t1],   0(%[src_argb0])           \n"
>-        "lw                %[tmp_t2],   4(%[src_argb0])           \n"
>-        "lw                %[tmp_t3],   8(%[src_argb0])           \n"
>-        "lw                %[tmp_t4],   12(%[src_argb0])          \n"
>-        "preceu.ph.qbl     %[tmp_t5],   %[tmp_t1]                 \n"
>-        "preceu.ph.qbr     %[tmp_t1],   %[tmp_t1]                 \n"
>-        "preceu.ph.qbl     %[tmp_t6],   %[tmp_t2]                 \n"
>-        "preceu.ph.qbr     %[tmp_t2],   %[tmp_t2]                 \n"
>-        "preceu.ph.qbl     %[tmp_t7],   %[tmp_t3]                 \n"
>-        "preceu.ph.qbr     %[tmp_t3],   %[tmp_t3]                 \n"
>-        "preceu.ph.qbl     %[tmp_t8],   %[tmp_t4]                 \n"
>-        "preceu.ph.qbr     %[tmp_t4],   %[tmp_t4]                 \n"
>-        "mult              $ac0,        %[const5],     %[const5]  \n"
>-        "mult              $ac1,        %[const5],     %[const5]  \n"
>-        "mult              $ac2,        %[const5],     %[const5]  \n"
>-        "mult              $ac3,        %[const5],     %[const5]  \n"
>-        "dpa.w.ph          $ac0,        %[tmp_t5],     %[const1]  \n"
>-        "dpa.w.ph          $ac1,        %[tmp_t6],     %[const1]  \n"
>-        "dpa.w.ph          $ac2,        %[tmp_t7],     %[const1]  \n"
>-        "dpa.w.ph          $ac3,        %[tmp_t8],     %[const1]  \n"
>-        "dpa.w.ph          $ac0,        %[tmp_t1],     %[const2]  \n"
>-        "dpa.w.ph          $ac1,        %[tmp_t2],     %[const2]  \n"
>-        "dpa.w.ph          $ac2,        %[tmp_t3],     %[const2]  \n"
>-        "dpa.w.ph          $ac3,        %[tmp_t4],     %[const2]  \n"
>-        "extr_r.w          %[tmp_t1],   $ac0,          8          \n"
>-        "extr_r.w          %[tmp_t2],   $ac1,          8          \n"
>-        "extr_r.w          %[tmp_t3],   $ac2,          8          \n"
>-        "extr_r.w          %[tmp_t4],   $ac3,          8          \n"
>-        "addiu             %[dst_y],    %[dst_y],      4          \n"
>-        "addiu             %[src_argb0],%[src_argb0],  16         \n"
>-        "sb                %[tmp_t1],   -4(%[dst_y])              \n"
>-        "sb                %[tmp_t2],   -3(%[dst_y])              \n"
>-        "sb                %[tmp_t3],   -2(%[dst_y])              \n"
>-        "sb                %[tmp_t4],   -1(%[dst_y])              \n"
>-        ".set pop                                                 \n"
>-        : [tmp_t1] "=&r"(tmp_t1), [tmp_t2] "=&r"(tmp_t2),
>-          [tmp_t3] "=&r"(tmp_t3), [tmp_t4] "=&r"(tmp_t4),
>-          [tmp_t5] "=&r"(tmp_t5), [tmp_t6] "=&r"(tmp_t6),
>-          [tmp_t7] "=&r"(tmp_t7), [tmp_t8] "=&r"(tmp_t8),
>-          [src_argb0] "+r"(src_argb0), [dst_y] "+r"(dst_y)
>-        : [const1] "r"(const1), [const2] "r"(const2), [const5] "r"(const5)
>-        : "hi", "lo", "$ac1lo", "$ac1hi", "$ac2lo", "$ac2hi", "$ac3lo",
>-          "$ac3hi");
>-  }
>-}
>-
>-void ARGBToUVRow_DSPR2(const uint8* src_rgb0,
>-                       int src_stride_rgb,
>-                       uint8* dst_u,
>-                       uint8* dst_v,
>-                       int width) {
>-  const uint8* src_rgb1 = src_rgb0 + src_stride_rgb;
>-  int x;
>-  int const1 = 0xffb60070;
>-  int const2 = 0x0000ffda;
>-  int const3 = 0xffa2ffee;
>-  int const4 = 0x00000070;
>-  int const5 = 0x100;
>-
>-  for (x = 0; x < width - 1; x += 2) {
>-    int tmp_t1, tmp_t2, tmp_t3, tmp_t4, tmp_t5;
>-    int tmp_t6, tmp_t7, tmp_t8;
>-    __asm__ __volatile__(
>-        ".set push                                                \n"
>-        ".set noreorder                                           \n"
>-        "lw                %[tmp_t1],   0(%[src_rgb0])            \n"
>-        "lw                %[tmp_t2],   4(%[src_rgb0])            \n"
>-        "lw                %[tmp_t3],   0(%[src_rgb1])            \n"
>-        "lw                %[tmp_t4],   4(%[src_rgb1])            \n"
>-        "preceu.ph.qbr     %[tmp_t5],   %[tmp_t1]                 \n"
>-        "preceu.ph.qbl     %[tmp_t1],   %[tmp_t1]                 \n"
>-        "preceu.ph.qbr     %[tmp_t6],   %[tmp_t2]                 \n"
>-        "preceu.ph.qbl     %[tmp_t2],   %[tmp_t2]                 \n"
>-        "preceu.ph.qbr     %[tmp_t7],   %[tmp_t3]                 \n"
>-        "preceu.ph.qbl     %[tmp_t3],   %[tmp_t3]                 \n"
>-        "preceu.ph.qbr     %[tmp_t8],   %[tmp_t4]                 \n"
>-        "preceu.ph.qbl     %[tmp_t4],   %[tmp_t4]                 \n"
>-        "addu.ph           %[tmp_t5],   %[tmp_t5],     %[tmp_t6]  \n"
>-        "addu.ph           %[tmp_t7],   %[tmp_t7],     %[tmp_t8]  \n"
>-        "addu.ph           %[tmp_t1],   %[tmp_t1],     %[tmp_t2]  \n"
>-        "addu.ph           %[tmp_t3],   %[tmp_t3],     %[tmp_t4]  \n"
>-        "addu.ph           %[tmp_t5],   %[tmp_t5],     %[tmp_t7]  \n"
>-        "addu.ph           %[tmp_t1],   %[tmp_t1],     %[tmp_t3]  \n"
>-        "shrl.ph           %[tmp_t5],   %[tmp_t5],     2          \n"
>-        "shrl.ph           %[tmp_t1],   %[tmp_t1],     2          \n"
>-        "mult              $ac0,        %[const5],     %[const5]  \n"
>-        "mult              $ac1,        %[const5],     %[const5]  \n"
>-        "dpaq_s.w.ph       $ac0,        %[tmp_t5],     %[const1]  \n"
>-        "dpaq_s.w.ph       $ac1,        %[tmp_t5],     %[const3]  \n"
>-        "dpaq_s.w.ph       $ac0,        %[tmp_t1],     %[const2]  \n"
>-        "dpaq_s.w.ph       $ac1,        %[tmp_t1],     %[const4]  \n"
>-        "extr_r.w          %[tmp_t7],   $ac0,          9          \n"
>-        "extr_r.w          %[tmp_t8],   $ac1,          9          \n"
>-        "addiu             %[src_rgb0], %[src_rgb0], 8            \n"
>-        "addiu             %[src_rgb1], %[src_rgb1], 8            \n"
>-        "addiu             %[dst_u],    %[dst_u],    1            \n"
>-        "addiu             %[dst_v],    %[dst_v],    1            \n"
>-        "sb                %[tmp_t7],   -1(%[dst_u])              \n"
>-        "sb                %[tmp_t8],   -1(%[dst_v])              \n"
>-        ".set pop                                                 \n"
>-        : [tmp_t1] "=&r"(tmp_t1), [tmp_t2] "=&r"(tmp_t2),
>-          [tmp_t3] "=&r"(tmp_t3), [tmp_t4] "=&r"(tmp_t4),
>-          [tmp_t5] "=&r"(tmp_t5), [tmp_t6] "=&r"(tmp_t6),
>-          [tmp_t7] "=&r"(tmp_t7), [tmp_t8] "=&r"(tmp_t8),
>-          [src_rgb0] "+r"(src_rgb0), [src_rgb1] "+r"(src_rgb1),
>-          [dst_u] "+r"(dst_u), [dst_v] "+r"(dst_v)
>-        : [const1] "r"(const1), [const2] "r"(const2), [const3] "r"(const3),
>-          [const4] "r"(const4), [const5] "r"(const5)
>-        : "hi", "lo", "$ac1lo", "$ac1hi");
>-  }
>-}
>-
>-#endif  // __mips_dsp_rev >= 2
>-
>-#endif  // defined(__mips__)
>-
>-#ifdef __cplusplus
>-}  // extern "C"
>-}  // namespace libyuv
>-#endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/row_gcc.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/row_gcc.cc
>index b5c2e65c938..95845c2592f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/row_gcc.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/row_gcc.cc
>@@ -22,80 +22,80 @@ extern "C" {
> #if defined(HAS_ARGBTOYROW_SSSE3) || defined(HAS_ARGBGRAYROW_SSSE3)
> 
> // Constants for ARGB
>-static vec8 kARGBToY = {13, 65, 33, 0, 13, 65, 33, 0,
>-                        13, 65, 33, 0, 13, 65, 33, 0};
>+static const vec8 kARGBToY = {13, 65, 33, 0, 13, 65, 33, 0,
>+                              13, 65, 33, 0, 13, 65, 33, 0};
> 
> // JPeg full range.
>-static vec8 kARGBToYJ = {15, 75, 38, 0, 15, 75, 38, 0,
>-                         15, 75, 38, 0, 15, 75, 38, 0};
>+static const vec8 kARGBToYJ = {15, 75, 38, 0, 15, 75, 38, 0,
>+                               15, 75, 38, 0, 15, 75, 38, 0};
> #endif  // defined(HAS_ARGBTOYROW_SSSE3) || defined(HAS_ARGBGRAYROW_SSSE3)
> 
> #if defined(HAS_ARGBTOYROW_SSSE3) || defined(HAS_I422TOARGBROW_SSSE3)
> 
>-static vec8 kARGBToU = {112, -74, -38, 0, 112, -74, -38, 0,
>-                        112, -74, -38, 0, 112, -74, -38, 0};
>+static const vec8 kARGBToU = {112, -74, -38, 0, 112, -74, -38, 0,
>+                              112, -74, -38, 0, 112, -74, -38, 0};
> 
>-static vec8 kARGBToUJ = {127, -84, -43, 0, 127, -84, -43, 0,
>-                         127, -84, -43, 0, 127, -84, -43, 0};
>+static const vec8 kARGBToUJ = {127, -84, -43, 0, 127, -84, -43, 0,
>+                               127, -84, -43, 0, 127, -84, -43, 0};
> 
>-static vec8 kARGBToV = {-18, -94, 112, 0, -18, -94, 112, 0,
>-                        -18, -94, 112, 0, -18, -94, 112, 0};
>+static const vec8 kARGBToV = {-18, -94, 112, 0, -18, -94, 112, 0,
>+                              -18, -94, 112, 0, -18, -94, 112, 0};
> 
>-static vec8 kARGBToVJ = {-20, -107, 127, 0, -20, -107, 127, 0,
>-                         -20, -107, 127, 0, -20, -107, 127, 0};
>+static const vec8 kARGBToVJ = {-20, -107, 127, 0, -20, -107, 127, 0,
>+                               -20, -107, 127, 0, -20, -107, 127, 0};
> 
> // Constants for BGRA
>-static vec8 kBGRAToY = {0, 33, 65, 13, 0, 33, 65, 13,
>-                        0, 33, 65, 13, 0, 33, 65, 13};
>+static const vec8 kBGRAToY = {0, 33, 65, 13, 0, 33, 65, 13,
>+                              0, 33, 65, 13, 0, 33, 65, 13};
> 
>-static vec8 kBGRAToU = {0, -38, -74, 112, 0, -38, -74, 112,
>-                        0, -38, -74, 112, 0, -38, -74, 112};
>+static const vec8 kBGRAToU = {0, -38, -74, 112, 0, -38, -74, 112,
>+                              0, -38, -74, 112, 0, -38, -74, 112};
> 
>-static vec8 kBGRAToV = {0, 112, -94, -18, 0, 112, -94, -18,
>-                        0, 112, -94, -18, 0, 112, -94, -18};
>+static const vec8 kBGRAToV = {0, 112, -94, -18, 0, 112, -94, -18,
>+                              0, 112, -94, -18, 0, 112, -94, -18};
> 
> // Constants for ABGR
>-static vec8 kABGRToY = {33, 65, 13, 0, 33, 65, 13, 0,
>-                        33, 65, 13, 0, 33, 65, 13, 0};
>+static const vec8 kABGRToY = {33, 65, 13, 0, 33, 65, 13, 0,
>+                              33, 65, 13, 0, 33, 65, 13, 0};
> 
>-static vec8 kABGRToU = {-38, -74, 112, 0, -38, -74, 112, 0,
>-                        -38, -74, 112, 0, -38, -74, 112, 0};
>+static const vec8 kABGRToU = {-38, -74, 112, 0, -38, -74, 112, 0,
>+                              -38, -74, 112, 0, -38, -74, 112, 0};
> 
>-static vec8 kABGRToV = {112, -94, -18, 0, 112, -94, -18, 0,
>-                        112, -94, -18, 0, 112, -94, -18, 0};
>+static const vec8 kABGRToV = {112, -94, -18, 0, 112, -94, -18, 0,
>+                              112, -94, -18, 0, 112, -94, -18, 0};
> 
> // Constants for RGBA.
>-static vec8 kRGBAToY = {0, 13, 65, 33, 0, 13, 65, 33,
>-                        0, 13, 65, 33, 0, 13, 65, 33};
>+static const vec8 kRGBAToY = {0, 13, 65, 33, 0, 13, 65, 33,
>+                              0, 13, 65, 33, 0, 13, 65, 33};
> 
>-static vec8 kRGBAToU = {0, 112, -74, -38, 0, 112, -74, -38,
>-                        0, 112, -74, -38, 0, 112, -74, -38};
>+static const vec8 kRGBAToU = {0, 112, -74, -38, 0, 112, -74, -38,
>+                              0, 112, -74, -38, 0, 112, -74, -38};
> 
>-static vec8 kRGBAToV = {0, -18, -94, 112, 0, -18, -94, 112,
>-                        0, -18, -94, 112, 0, -18, -94, 112};
>+static const vec8 kRGBAToV = {0, -18, -94, 112, 0, -18, -94, 112,
>+                              0, -18, -94, 112, 0, -18, -94, 112};
> 
>-static uvec8 kAddY16 = {16u, 16u, 16u, 16u, 16u, 16u, 16u, 16u,
>-                        16u, 16u, 16u, 16u, 16u, 16u, 16u, 16u};
>+static const uvec8 kAddY16 = {16u, 16u, 16u, 16u, 16u, 16u, 16u, 16u,
>+                              16u, 16u, 16u, 16u, 16u, 16u, 16u, 16u};
> 
> // 7 bit fixed point 0.5.
>-static vec16 kAddYJ64 = {64, 64, 64, 64, 64, 64, 64, 64};
>+static const vec16 kAddYJ64 = {64, 64, 64, 64, 64, 64, 64, 64};
> 
>-static uvec8 kAddUV128 = {128u, 128u, 128u, 128u, 128u, 128u, 128u, 128u,
>-                          128u, 128u, 128u, 128u, 128u, 128u, 128u, 128u};
>+static const uvec8 kAddUV128 = {128u, 128u, 128u, 128u, 128u, 128u, 128u, 128u,
>+                                128u, 128u, 128u, 128u, 128u, 128u, 128u, 128u};
> 
>-static uvec16 kAddUVJ128 = {0x8080u, 0x8080u, 0x8080u, 0x8080u,
>-                            0x8080u, 0x8080u, 0x8080u, 0x8080u};
>+static const uvec16 kAddUVJ128 = {0x8080u, 0x8080u, 0x8080u, 0x8080u,
>+                                  0x8080u, 0x8080u, 0x8080u, 0x8080u};
> #endif  // defined(HAS_ARGBTOYROW_SSSE3) || defined(HAS_I422TOARGBROW_SSSE3)
> 
> #ifdef HAS_RGB24TOARGBROW_SSSE3
> 
> // Shuffle table for converting RGB24 to ARGB.
>-static uvec8 kShuffleMaskRGB24ToARGB = {0u, 1u, 2u, 12u, 3u, 4u,  5u,  13u,
>-                                        6u, 7u, 8u, 14u, 9u, 10u, 11u, 15u};
>+static const uvec8 kShuffleMaskRGB24ToARGB = {
>+    0u, 1u, 2u, 12u, 3u, 4u, 5u, 13u, 6u, 7u, 8u, 14u, 9u, 10u, 11u, 15u};
> 
> // Shuffle table for converting RAW to ARGB.
>-static uvec8 kShuffleMaskRAWToARGB = {2u, 1u, 0u, 12u, 5u,  4u,  3u, 13u,
>-                                      8u, 7u, 6u, 14u, 11u, 10u, 9u, 15u};
>+static const uvec8 kShuffleMaskRAWToARGB = {2u, 1u, 0u, 12u, 5u,  4u,  3u, 13u,
>+                                            8u, 7u, 6u, 14u, 11u, 10u, 9u, 15u};
> 
> // Shuffle table for converting RAW to RGB24.  First 8.
> static const uvec8 kShuffleMaskRAWToRGB24_0 = {
>@@ -113,15 +113,15 @@ static const uvec8 kShuffleMaskRAWToRGB24_2 = {
>     128u, 128u, 128u, 128u, 128u, 128u, 128u, 128u};
> 
> // Shuffle table for converting ARGB to RGB24.
>-static uvec8 kShuffleMaskARGBToRGB24 = {
>+static const uvec8 kShuffleMaskARGBToRGB24 = {
>     0u, 1u, 2u, 4u, 5u, 6u, 8u, 9u, 10u, 12u, 13u, 14u, 128u, 128u, 128u, 128u};
> 
> // Shuffle table for converting ARGB to RAW.
>-static uvec8 kShuffleMaskARGBToRAW = {
>+static const uvec8 kShuffleMaskARGBToRAW = {
>     2u, 1u, 0u, 6u, 5u, 4u, 10u, 9u, 8u, 14u, 13u, 12u, 128u, 128u, 128u, 128u};
> 
> // Shuffle table for converting ARGBToRGB24 for I422ToRGB24.  First 8 + next 4
>-static uvec8 kShuffleMaskARGBToRGB24_0 = {
>+static const uvec8 kShuffleMaskARGBToRGB24_0 = {
>     0u, 1u, 2u, 4u, 5u, 6u, 8u, 9u, 128u, 128u, 128u, 128u, 10u, 12u, 13u, 14u};
> 
> // YUY2 shuf 16 Y to 32 Y.
>@@ -152,392 +152,399 @@ static const lvec8 kShuffleNV21 = {
> #endif  // HAS_RGB24TOARGBROW_SSSE3
> 
> #ifdef HAS_J400TOARGBROW_SSE2
>-void J400ToARGBRow_SSE2(const uint8* src_y, uint8* dst_argb, int width) {
>-  asm volatile (
>-    "pcmpeqb   %%xmm5,%%xmm5                   \n"
>-    "pslld     $0x18,%%xmm5                    \n"
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movq      " MEMACCESS(0) ",%%xmm0         \n"
>-    "lea       " MEMLEA(0x8,0) ",%0            \n"
>-    "punpcklbw %%xmm0,%%xmm0                   \n"
>-    "movdqa    %%xmm0,%%xmm1                   \n"
>-    "punpcklwd %%xmm0,%%xmm0                   \n"
>-    "punpckhwd %%xmm1,%%xmm1                   \n"
>-    "por       %%xmm5,%%xmm0                   \n"
>-    "por       %%xmm5,%%xmm1                   \n"
>-    "movdqu    %%xmm0," MEMACCESS(1) "         \n"
>-    "movdqu    %%xmm1," MEMACCESS2(0x10,1) "   \n"
>-    "lea       " MEMLEA(0x20,1) ",%1           \n"
>-    "sub       $0x8,%2                         \n"
>-    "jg        1b                              \n"
>-  : "+r"(src_y),     // %0
>-    "+r"(dst_argb),  // %1
>-    "+r"(width)        // %2
>-  :: "memory", "cc", "xmm0", "xmm1", "xmm5"
>-  );
>+void J400ToARGBRow_SSE2(const uint8_t* src_y, uint8_t* dst_argb, int width) {
>+  asm volatile(
>+      "pcmpeqb   %%xmm5,%%xmm5                   \n"
>+      "pslld     $0x18,%%xmm5                    \n"
>+
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movq      (%0),%%xmm0                     \n"
>+      "lea       0x8(%0),%0                      \n"
>+      "punpcklbw %%xmm0,%%xmm0                   \n"
>+      "movdqa    %%xmm0,%%xmm1                   \n"
>+      "punpcklwd %%xmm0,%%xmm0                   \n"
>+      "punpckhwd %%xmm1,%%xmm1                   \n"
>+      "por       %%xmm5,%%xmm0                   \n"
>+      "por       %%xmm5,%%xmm1                   \n"
>+      "movdqu    %%xmm0,(%1)                     \n"
>+      "movdqu    %%xmm1,0x10(%1)                 \n"
>+      "lea       0x20(%1),%1                     \n"
>+      "sub       $0x8,%2                         \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_y),     // %0
>+        "+r"(dst_argb),  // %1
>+        "+r"(width)      // %2
>+        ::"memory",
>+        "cc", "xmm0", "xmm1", "xmm5");
> }
> #endif  // HAS_J400TOARGBROW_SSE2
> 
> #ifdef HAS_RGB24TOARGBROW_SSSE3
>-void RGB24ToARGBRow_SSSE3(const uint8* src_rgb24, uint8* dst_argb, int width) {
>-  asm volatile (
>-    "pcmpeqb   %%xmm5,%%xmm5                   \n"  // generate mask 0xff000000
>-    "pslld     $0x18,%%xmm5                    \n"
>-    "movdqa    %3,%%xmm4                       \n"
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    "movdqu    " MEMACCESS2(0x10,0) ",%%xmm1   \n"
>-    "movdqu    " MEMACCESS2(0x20,0) ",%%xmm3   \n"
>-    "lea       " MEMLEA(0x30,0) ",%0           \n"
>-    "movdqa    %%xmm3,%%xmm2                   \n"
>-    "palignr   $0x8,%%xmm1,%%xmm2              \n"
>-    "pshufb    %%xmm4,%%xmm2                   \n"
>-    "por       %%xmm5,%%xmm2                   \n"
>-    "palignr   $0xc,%%xmm0,%%xmm1              \n"
>-    "pshufb    %%xmm4,%%xmm0                   \n"
>-    "movdqu    %%xmm2," MEMACCESS2(0x20,1) "   \n"
>-    "por       %%xmm5,%%xmm0                   \n"
>-    "pshufb    %%xmm4,%%xmm1                   \n"
>-    "movdqu    %%xmm0," MEMACCESS(1) "         \n"
>-    "por       %%xmm5,%%xmm1                   \n"
>-    "palignr   $0x4,%%xmm3,%%xmm3              \n"
>-    "pshufb    %%xmm4,%%xmm3                   \n"
>-    "movdqu    %%xmm1," MEMACCESS2(0x10,1) "   \n"
>-    "por       %%xmm5,%%xmm3                   \n"
>-    "movdqu    %%xmm3," MEMACCESS2(0x30,1) "   \n"
>-    "lea       " MEMLEA(0x40,1) ",%1           \n"
>-    "sub       $0x10,%2                        \n"
>-    "jg        1b                              \n"
>-  : "+r"(src_rgb24),  // %0
>-    "+r"(dst_argb),  // %1
>-    "+r"(width)        // %2
>-  : "m"(kShuffleMaskRGB24ToARGB)  // %3
>-  : "memory", "cc" , "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5"
>-  );
>+void RGB24ToARGBRow_SSSE3(const uint8_t* src_rgb24,
>+                          uint8_t* dst_argb,
>+                          int width) {
>+  asm volatile(
>+      "pcmpeqb   %%xmm5,%%xmm5                   \n"  // 0xff000000
>+      "pslld     $0x18,%%xmm5                    \n"
>+      "movdqa    %3,%%xmm4                       \n"
>+
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "movdqu    0x10(%0),%%xmm1                 \n"
>+      "movdqu    0x20(%0),%%xmm3                 \n"
>+      "lea       0x30(%0),%0                     \n"
>+      "movdqa    %%xmm3,%%xmm2                   \n"
>+      "palignr   $0x8,%%xmm1,%%xmm2              \n"
>+      "pshufb    %%xmm4,%%xmm2                   \n"
>+      "por       %%xmm5,%%xmm2                   \n"
>+      "palignr   $0xc,%%xmm0,%%xmm1              \n"
>+      "pshufb    %%xmm4,%%xmm0                   \n"
>+      "movdqu    %%xmm2,0x20(%1)                 \n"
>+      "por       %%xmm5,%%xmm0                   \n"
>+      "pshufb    %%xmm4,%%xmm1                   \n"
>+      "movdqu    %%xmm0,(%1)                     \n"
>+      "por       %%xmm5,%%xmm1                   \n"
>+      "palignr   $0x4,%%xmm3,%%xmm3              \n"
>+      "pshufb    %%xmm4,%%xmm3                   \n"
>+      "movdqu    %%xmm1,0x10(%1)                 \n"
>+      "por       %%xmm5,%%xmm3                   \n"
>+      "movdqu    %%xmm3,0x30(%1)                 \n"
>+      "lea       0x40(%1),%1                     \n"
>+      "sub       $0x10,%2                        \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_rgb24),              // %0
>+        "+r"(dst_argb),               // %1
>+        "+r"(width)                   // %2
>+      : "m"(kShuffleMaskRGB24ToARGB)  // %3
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5");
> }
> 
>-void RAWToARGBRow_SSSE3(const uint8* src_raw, uint8* dst_argb, int width) {
>-  asm volatile (
>-    "pcmpeqb   %%xmm5,%%xmm5                   \n"  // generate mask 0xff000000
>-    "pslld     $0x18,%%xmm5                    \n"
>-    "movdqa    %3,%%xmm4                       \n"
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    "movdqu    " MEMACCESS2(0x10,0) ",%%xmm1   \n"
>-    "movdqu    " MEMACCESS2(0x20,0) ",%%xmm3   \n"
>-    "lea       " MEMLEA(0x30,0) ",%0           \n"
>-    "movdqa    %%xmm3,%%xmm2                   \n"
>-    "palignr   $0x8,%%xmm1,%%xmm2              \n"
>-    "pshufb    %%xmm4,%%xmm2                   \n"
>-    "por       %%xmm5,%%xmm2                   \n"
>-    "palignr   $0xc,%%xmm0,%%xmm1              \n"
>-    "pshufb    %%xmm4,%%xmm0                   \n"
>-    "movdqu    %%xmm2," MEMACCESS2(0x20,1) "   \n"
>-    "por       %%xmm5,%%xmm0                   \n"
>-    "pshufb    %%xmm4,%%xmm1                   \n"
>-    "movdqu    %%xmm0," MEMACCESS(1) "         \n"
>-    "por       %%xmm5,%%xmm1                   \n"
>-    "palignr   $0x4,%%xmm3,%%xmm3              \n"
>-    "pshufb    %%xmm4,%%xmm3                   \n"
>-    "movdqu    %%xmm1," MEMACCESS2(0x10,1) "   \n"
>-    "por       %%xmm5,%%xmm3                   \n"
>-    "movdqu    %%xmm3," MEMACCESS2(0x30,1) "   \n"
>-    "lea       " MEMLEA(0x40,1) ",%1           \n"
>-    "sub       $0x10,%2                        \n"
>-    "jg        1b                              \n"
>-  : "+r"(src_raw),   // %0
>-    "+r"(dst_argb),  // %1
>-    "+r"(width)        // %2
>-  : "m"(kShuffleMaskRAWToARGB)  // %3
>-  : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5"
>-  );
>+void RAWToARGBRow_SSSE3(const uint8_t* src_raw, uint8_t* dst_argb, int width) {
>+  asm volatile(
>+      "pcmpeqb   %%xmm5,%%xmm5                   \n"  // 0xff000000
>+      "pslld     $0x18,%%xmm5                    \n"
>+      "movdqa    %3,%%xmm4                       \n"
>+
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "movdqu    0x10(%0),%%xmm1                 \n"
>+      "movdqu    0x20(%0),%%xmm3                 \n"
>+      "lea       0x30(%0),%0                     \n"
>+      "movdqa    %%xmm3,%%xmm2                   \n"
>+      "palignr   $0x8,%%xmm1,%%xmm2              \n"
>+      "pshufb    %%xmm4,%%xmm2                   \n"
>+      "por       %%xmm5,%%xmm2                   \n"
>+      "palignr   $0xc,%%xmm0,%%xmm1              \n"
>+      "pshufb    %%xmm4,%%xmm0                   \n"
>+      "movdqu    %%xmm2,0x20(%1)                 \n"
>+      "por       %%xmm5,%%xmm0                   \n"
>+      "pshufb    %%xmm4,%%xmm1                   \n"
>+      "movdqu    %%xmm0,(%1)                     \n"
>+      "por       %%xmm5,%%xmm1                   \n"
>+      "palignr   $0x4,%%xmm3,%%xmm3              \n"
>+      "pshufb    %%xmm4,%%xmm3                   \n"
>+      "movdqu    %%xmm1,0x10(%1)                 \n"
>+      "por       %%xmm5,%%xmm3                   \n"
>+      "movdqu    %%xmm3,0x30(%1)                 \n"
>+      "lea       0x40(%1),%1                     \n"
>+      "sub       $0x10,%2                        \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_raw),              // %0
>+        "+r"(dst_argb),             // %1
>+        "+r"(width)                 // %2
>+      : "m"(kShuffleMaskRAWToARGB)  // %3
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5");
> }
> 
>-void RAWToRGB24Row_SSSE3(const uint8* src_raw, uint8* dst_rgb24, int width) {
>-  asm volatile (
>-   "movdqa     %3,%%xmm3                       \n"
>-   "movdqa     %4,%%xmm4                       \n"
>-   "movdqa     %5,%%xmm5                       \n"
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    "movdqu    " MEMACCESS2(0x4,0) ",%%xmm1    \n"
>-    "movdqu    " MEMACCESS2(0x8,0) ",%%xmm2    \n"
>-    "lea       " MEMLEA(0x18,0) ",%0           \n"
>-    "pshufb    %%xmm3,%%xmm0                   \n"
>-    "pshufb    %%xmm4,%%xmm1                   \n"
>-    "pshufb    %%xmm5,%%xmm2                   \n"
>-    "movq      %%xmm0," MEMACCESS(1) "         \n"
>-    "movq      %%xmm1," MEMACCESS2(0x8,1) "    \n"
>-    "movq      %%xmm2," MEMACCESS2(0x10,1) "   \n"
>-    "lea       " MEMLEA(0x18,1) ",%1           \n"
>-    "sub       $0x8,%2                         \n"
>-    "jg        1b                              \n"
>-  : "+r"(src_raw),    // %0
>-    "+r"(dst_rgb24),  // %1
>-    "+r"(width)       // %2
>-  : "m"(kShuffleMaskRAWToRGB24_0),  // %3
>-    "m"(kShuffleMaskRAWToRGB24_1),  // %4
>-    "m"(kShuffleMaskRAWToRGB24_2)   // %5
>-  : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5"
>-  );
>+void RAWToRGB24Row_SSSE3(const uint8_t* src_raw,
>+                         uint8_t* dst_rgb24,
>+                         int width) {
>+  asm volatile(
>+      "movdqa     %3,%%xmm3                       \n"
>+      "movdqa     %4,%%xmm4                       \n"
>+      "movdqa     %5,%%xmm5                       \n"
>+
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "movdqu    0x4(%0),%%xmm1                  \n"
>+      "movdqu    0x8(%0),%%xmm2                  \n"
>+      "lea       0x18(%0),%0                     \n"
>+      "pshufb    %%xmm3,%%xmm0                   \n"
>+      "pshufb    %%xmm4,%%xmm1                   \n"
>+      "pshufb    %%xmm5,%%xmm2                   \n"
>+      "movq      %%xmm0,(%1)                     \n"
>+      "movq      %%xmm1,0x8(%1)                  \n"
>+      "movq      %%xmm2,0x10(%1)                 \n"
>+      "lea       0x18(%1),%1                     \n"
>+      "sub       $0x8,%2                         \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_raw),                  // %0
>+        "+r"(dst_rgb24),                // %1
>+        "+r"(width)                     // %2
>+      : "m"(kShuffleMaskRAWToRGB24_0),  // %3
>+        "m"(kShuffleMaskRAWToRGB24_1),  // %4
>+        "m"(kShuffleMaskRAWToRGB24_2)   // %5
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5");
> }
> 
>-void RGB565ToARGBRow_SSE2(const uint8* src, uint8* dst, int width) {
>-  asm volatile (
>-    "mov       $0x1080108,%%eax                \n"
>-    "movd      %%eax,%%xmm5                    \n"
>-    "pshufd    $0x0,%%xmm5,%%xmm5              \n"
>-    "mov       $0x20802080,%%eax               \n"
>-    "movd      %%eax,%%xmm6                    \n"
>-    "pshufd    $0x0,%%xmm6,%%xmm6              \n"
>-    "pcmpeqb   %%xmm3,%%xmm3                   \n"
>-    "psllw     $0xb,%%xmm3                     \n"
>-    "pcmpeqb   %%xmm4,%%xmm4                   \n"
>-    "psllw     $0xa,%%xmm4                     \n"
>-    "psrlw     $0x5,%%xmm4                     \n"
>-    "pcmpeqb   %%xmm7,%%xmm7                   \n"
>-    "psllw     $0x8,%%xmm7                     \n"
>-    "sub       %0,%1                           \n"
>-    "sub       %0,%1                           \n"
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    "movdqa    %%xmm0,%%xmm1                   \n"
>-    "movdqa    %%xmm0,%%xmm2                   \n"
>-    "pand      %%xmm3,%%xmm1                   \n"
>-    "psllw     $0xb,%%xmm2                     \n"
>-    "pmulhuw   %%xmm5,%%xmm1                   \n"
>-    "pmulhuw   %%xmm5,%%xmm2                   \n"
>-    "psllw     $0x8,%%xmm1                     \n"
>-    "por       %%xmm2,%%xmm1                   \n"
>-    "pand      %%xmm4,%%xmm0                   \n"
>-    "pmulhuw   %%xmm6,%%xmm0                   \n"
>-    "por       %%xmm7,%%xmm0                   \n"
>-    "movdqa    %%xmm1,%%xmm2                   \n"
>-    "punpcklbw %%xmm0,%%xmm1                   \n"
>-    "punpckhbw %%xmm0,%%xmm2                   \n"
>-    MEMOPMEM(movdqu,xmm1,0x00,1,0,2)           //  movdqu  %%xmm1,(%1,%0,2)
>-    MEMOPMEM(movdqu,xmm2,0x10,1,0,2)           //  movdqu  %%xmm2,0x10(%1,%0,2)
>-    "lea       " MEMLEA(0x10,0) ",%0           \n"
>-    "sub       $0x8,%2                         \n"
>-    "jg        1b                              \n"
>-  : "+r"(src),  // %0
>-    "+r"(dst),  // %1
>-    "+r"(width)   // %2
>-  :
>-  : "memory", "cc", "eax", NACL_R14
>-    "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6", "xmm7"
>-  );
>+void RGB565ToARGBRow_SSE2(const uint8_t* src, uint8_t* dst, int width) {
>+  asm volatile(
>+      "mov       $0x1080108,%%eax                \n"
>+      "movd      %%eax,%%xmm5                    \n"
>+      "pshufd    $0x0,%%xmm5,%%xmm5              \n"
>+      "mov       $0x20802080,%%eax               \n"
>+      "movd      %%eax,%%xmm6                    \n"
>+      "pshufd    $0x0,%%xmm6,%%xmm6              \n"
>+      "pcmpeqb   %%xmm3,%%xmm3                   \n"
>+      "psllw     $0xb,%%xmm3                     \n"
>+      "pcmpeqb   %%xmm4,%%xmm4                   \n"
>+      "psllw     $0xa,%%xmm4                     \n"
>+      "psrlw     $0x5,%%xmm4                     \n"
>+      "pcmpeqb   %%xmm7,%%xmm7                   \n"
>+      "psllw     $0x8,%%xmm7                     \n"
>+      "sub       %0,%1                           \n"
>+      "sub       %0,%1                           \n"
>+
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "movdqa    %%xmm0,%%xmm1                   \n"
>+      "movdqa    %%xmm0,%%xmm2                   \n"
>+      "pand      %%xmm3,%%xmm1                   \n"
>+      "psllw     $0xb,%%xmm2                     \n"
>+      "pmulhuw   %%xmm5,%%xmm1                   \n"
>+      "pmulhuw   %%xmm5,%%xmm2                   \n"
>+      "psllw     $0x8,%%xmm1                     \n"
>+      "por       %%xmm2,%%xmm1                   \n"
>+      "pand      %%xmm4,%%xmm0                   \n"
>+      "pmulhuw   %%xmm6,%%xmm0                   \n"
>+      "por       %%xmm7,%%xmm0                   \n"
>+      "movdqa    %%xmm1,%%xmm2                   \n"
>+      "punpcklbw %%xmm0,%%xmm1                   \n"
>+      "punpckhbw %%xmm0,%%xmm2                   \n"
>+      "movdqu    %%xmm1,0x00(%1,%0,2)            \n"
>+      "movdqu    %%xmm2,0x10(%1,%0,2)            \n"
>+      "lea       0x10(%0),%0                     \n"
>+      "sub       $0x8,%2                         \n"
>+      "jg        1b                              \n"
>+      : "+r"(src),   // %0
>+        "+r"(dst),   // %1
>+        "+r"(width)  // %2
>+      :
>+      : "memory", "cc", "eax", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5",
>+        "xmm6", "xmm7");
> }
> 
>-void ARGB1555ToARGBRow_SSE2(const uint8* src, uint8* dst, int width) {
>-  asm volatile (
>-    "mov       $0x1080108,%%eax                \n"
>-    "movd      %%eax,%%xmm5                    \n"
>-    "pshufd    $0x0,%%xmm5,%%xmm5              \n"
>-    "mov       $0x42004200,%%eax               \n"
>-    "movd      %%eax,%%xmm6                    \n"
>-    "pshufd    $0x0,%%xmm6,%%xmm6              \n"
>-    "pcmpeqb   %%xmm3,%%xmm3                   \n"
>-    "psllw     $0xb,%%xmm3                     \n"
>-    "movdqa    %%xmm3,%%xmm4                   \n"
>-    "psrlw     $0x6,%%xmm4                     \n"
>-    "pcmpeqb   %%xmm7,%%xmm7                   \n"
>-    "psllw     $0x8,%%xmm7                     \n"
>-    "sub       %0,%1                           \n"
>-    "sub       %0,%1                           \n"
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    "movdqa    %%xmm0,%%xmm1                   \n"
>-    "movdqa    %%xmm0,%%xmm2                   \n"
>-    "psllw     $0x1,%%xmm1                     \n"
>-    "psllw     $0xb,%%xmm2                     \n"
>-    "pand      %%xmm3,%%xmm1                   \n"
>-    "pmulhuw   %%xmm5,%%xmm2                   \n"
>-    "pmulhuw   %%xmm5,%%xmm1                   \n"
>-    "psllw     $0x8,%%xmm1                     \n"
>-    "por       %%xmm2,%%xmm1                   \n"
>-    "movdqa    %%xmm0,%%xmm2                   \n"
>-    "pand      %%xmm4,%%xmm0                   \n"
>-    "psraw     $0x8,%%xmm2                     \n"
>-    "pmulhuw   %%xmm6,%%xmm0                   \n"
>-    "pand      %%xmm7,%%xmm2                   \n"
>-    "por       %%xmm2,%%xmm0                   \n"
>-    "movdqa    %%xmm1,%%xmm2                   \n"
>-    "punpcklbw %%xmm0,%%xmm1                   \n"
>-    "punpckhbw %%xmm0,%%xmm2                   \n"
>-    MEMOPMEM(movdqu,xmm1,0x00,1,0,2)           //  movdqu  %%xmm1,(%1,%0,2)
>-    MEMOPMEM(movdqu,xmm2,0x10,1,0,2)           //  movdqu  %%xmm2,0x10(%1,%0,2)
>-    "lea       " MEMLEA(0x10,0) ",%0           \n"
>-    "sub       $0x8,%2                         \n"
>-    "jg        1b                              \n"
>-  : "+r"(src),  // %0
>-    "+r"(dst),  // %1
>-    "+r"(width)   // %2
>-  :
>-  : "memory", "cc", "eax", NACL_R14
>-    "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6", "xmm7"
>-  );
>+void ARGB1555ToARGBRow_SSE2(const uint8_t* src, uint8_t* dst, int width) {
>+  asm volatile(
>+      "mov       $0x1080108,%%eax                \n"
>+      "movd      %%eax,%%xmm5                    \n"
>+      "pshufd    $0x0,%%xmm5,%%xmm5              \n"
>+      "mov       $0x42004200,%%eax               \n"
>+      "movd      %%eax,%%xmm6                    \n"
>+      "pshufd    $0x0,%%xmm6,%%xmm6              \n"
>+      "pcmpeqb   %%xmm3,%%xmm3                   \n"
>+      "psllw     $0xb,%%xmm3                     \n"
>+      "movdqa    %%xmm3,%%xmm4                   \n"
>+      "psrlw     $0x6,%%xmm4                     \n"
>+      "pcmpeqb   %%xmm7,%%xmm7                   \n"
>+      "psllw     $0x8,%%xmm7                     \n"
>+      "sub       %0,%1                           \n"
>+      "sub       %0,%1                           \n"
>+
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "movdqa    %%xmm0,%%xmm1                   \n"
>+      "movdqa    %%xmm0,%%xmm2                   \n"
>+      "psllw     $0x1,%%xmm1                     \n"
>+      "psllw     $0xb,%%xmm2                     \n"
>+      "pand      %%xmm3,%%xmm1                   \n"
>+      "pmulhuw   %%xmm5,%%xmm2                   \n"
>+      "pmulhuw   %%xmm5,%%xmm1                   \n"
>+      "psllw     $0x8,%%xmm1                     \n"
>+      "por       %%xmm2,%%xmm1                   \n"
>+      "movdqa    %%xmm0,%%xmm2                   \n"
>+      "pand      %%xmm4,%%xmm0                   \n"
>+      "psraw     $0x8,%%xmm2                     \n"
>+      "pmulhuw   %%xmm6,%%xmm0                   \n"
>+      "pand      %%xmm7,%%xmm2                   \n"
>+      "por       %%xmm2,%%xmm0                   \n"
>+      "movdqa    %%xmm1,%%xmm2                   \n"
>+      "punpcklbw %%xmm0,%%xmm1                   \n"
>+      "punpckhbw %%xmm0,%%xmm2                   \n"
>+      "movdqu    %%xmm1,0x00(%1,%0,2)            \n"
>+      "movdqu    %%xmm2,0x10(%1,%0,2)            \n"
>+      "lea       0x10(%0),%0                     \n"
>+      "sub       $0x8,%2                         \n"
>+      "jg        1b                              \n"
>+      : "+r"(src),   // %0
>+        "+r"(dst),   // %1
>+        "+r"(width)  // %2
>+      :
>+      : "memory", "cc", "eax", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5",
>+        "xmm6", "xmm7");
> }
> 
>-void ARGB4444ToARGBRow_SSE2(const uint8* src, uint8* dst, int width) {
>-  asm volatile (
>-    "mov       $0xf0f0f0f,%%eax                \n"
>-    "movd      %%eax,%%xmm4                    \n"
>-    "pshufd    $0x0,%%xmm4,%%xmm4              \n"
>-    "movdqa    %%xmm4,%%xmm5                   \n"
>-    "pslld     $0x4,%%xmm5                     \n"
>-    "sub       %0,%1                           \n"
>-    "sub       %0,%1                           \n"
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    "movdqa    %%xmm0,%%xmm2                   \n"
>-    "pand      %%xmm4,%%xmm0                   \n"
>-    "pand      %%xmm5,%%xmm2                   \n"
>-    "movdqa    %%xmm0,%%xmm1                   \n"
>-    "movdqa    %%xmm2,%%xmm3                   \n"
>-    "psllw     $0x4,%%xmm1                     \n"
>-    "psrlw     $0x4,%%xmm3                     \n"
>-    "por       %%xmm1,%%xmm0                   \n"
>-    "por       %%xmm3,%%xmm2                   \n"
>-    "movdqa    %%xmm0,%%xmm1                   \n"
>-    "punpcklbw %%xmm2,%%xmm0                   \n"
>-    "punpckhbw %%xmm2,%%xmm1                   \n"
>-    MEMOPMEM(movdqu,xmm0,0x00,1,0,2)           //  movdqu  %%xmm0,(%1,%0,2)
>-    MEMOPMEM(movdqu,xmm1,0x10,1,0,2)           //  movdqu  %%xmm1,0x10(%1,%0,2)
>-    "lea       " MEMLEA(0x10,0) ",%0           \n"
>-    "sub       $0x8,%2                         \n"
>-    "jg        1b                              \n"
>-  : "+r"(src),  // %0
>-    "+r"(dst),  // %1
>-    "+r"(width)   // %2
>-  :
>-  : "memory", "cc", "eax", NACL_R14
>-    "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5"
>-  );
>+void ARGB4444ToARGBRow_SSE2(const uint8_t* src, uint8_t* dst, int width) {
>+  asm volatile(
>+      "mov       $0xf0f0f0f,%%eax                \n"
>+      "movd      %%eax,%%xmm4                    \n"
>+      "pshufd    $0x0,%%xmm4,%%xmm4              \n"
>+      "movdqa    %%xmm4,%%xmm5                   \n"
>+      "pslld     $0x4,%%xmm5                     \n"
>+      "sub       %0,%1                           \n"
>+      "sub       %0,%1                           \n"
>+
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "movdqa    %%xmm0,%%xmm2                   \n"
>+      "pand      %%xmm4,%%xmm0                   \n"
>+      "pand      %%xmm5,%%xmm2                   \n"
>+      "movdqa    %%xmm0,%%xmm1                   \n"
>+      "movdqa    %%xmm2,%%xmm3                   \n"
>+      "psllw     $0x4,%%xmm1                     \n"
>+      "psrlw     $0x4,%%xmm3                     \n"
>+      "por       %%xmm1,%%xmm0                   \n"
>+      "por       %%xmm3,%%xmm2                   \n"
>+      "movdqa    %%xmm0,%%xmm1                   \n"
>+      "punpcklbw %%xmm2,%%xmm0                   \n"
>+      "punpckhbw %%xmm2,%%xmm1                   \n"
>+      "movdqu    %%xmm0,0x00(%1,%0,2)            \n"
>+      "movdqu    %%xmm1,0x10(%1,%0,2)            \n"
>+      "lea       0x10(%0),%0                     \n"
>+      "sub       $0x8,%2                         \n"
>+      "jg        1b                              \n"
>+      : "+r"(src),   // %0
>+        "+r"(dst),   // %1
>+        "+r"(width)  // %2
>+      :
>+      : "memory", "cc", "eax", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5");
> }
> 
>-void ARGBToRGB24Row_SSSE3(const uint8* src, uint8* dst, int width) {
>-  asm volatile (
>-    "movdqa    %3,%%xmm6                       \n"
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    "movdqu    " MEMACCESS2(0x10,0) ",%%xmm1   \n"
>-    "movdqu    " MEMACCESS2(0x20,0) ",%%xmm2   \n"
>-    "movdqu    " MEMACCESS2(0x30,0) ",%%xmm3   \n"
>-    "lea       " MEMLEA(0x40,0) ",%0           \n"
>-    "pshufb    %%xmm6,%%xmm0                   \n"
>-    "pshufb    %%xmm6,%%xmm1                   \n"
>-    "pshufb    %%xmm6,%%xmm2                   \n"
>-    "pshufb    %%xmm6,%%xmm3                   \n"
>-    "movdqa    %%xmm1,%%xmm4                   \n"
>-    "psrldq    $0x4,%%xmm1                     \n"
>-    "pslldq    $0xc,%%xmm4                     \n"
>-    "movdqa    %%xmm2,%%xmm5                   \n"
>-    "por       %%xmm4,%%xmm0                   \n"
>-    "pslldq    $0x8,%%xmm5                     \n"
>-    "movdqu    %%xmm0," MEMACCESS(1) "         \n"
>-    "por       %%xmm5,%%xmm1                   \n"
>-    "psrldq    $0x8,%%xmm2                     \n"
>-    "pslldq    $0x4,%%xmm3                     \n"
>-    "por       %%xmm3,%%xmm2                   \n"
>-    "movdqu    %%xmm1," MEMACCESS2(0x10,1) "   \n"
>-    "movdqu    %%xmm2," MEMACCESS2(0x20,1) "   \n"
>-    "lea       " MEMLEA(0x30,1) ",%1           \n"
>-    "sub       $0x10,%2                        \n"
>-    "jg        1b                              \n"
>-  : "+r"(src),  // %0
>-    "+r"(dst),  // %1
>-    "+r"(width)   // %2
>-  : "m"(kShuffleMaskARGBToRGB24)  // %3
>-  : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6"
>-  );
>+void ARGBToRGB24Row_SSSE3(const uint8_t* src, uint8_t* dst, int width) {
>+  asm volatile(
>+
>+      "movdqa    %3,%%xmm6                       \n"
>+
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "movdqu    0x10(%0),%%xmm1                 \n"
>+      "movdqu    0x20(%0),%%xmm2                 \n"
>+      "movdqu    0x30(%0),%%xmm3                 \n"
>+      "lea       0x40(%0),%0                     \n"
>+      "pshufb    %%xmm6,%%xmm0                   \n"
>+      "pshufb    %%xmm6,%%xmm1                   \n"
>+      "pshufb    %%xmm6,%%xmm2                   \n"
>+      "pshufb    %%xmm6,%%xmm3                   \n"
>+      "movdqa    %%xmm1,%%xmm4                   \n"
>+      "psrldq    $0x4,%%xmm1                     \n"
>+      "pslldq    $0xc,%%xmm4                     \n"
>+      "movdqa    %%xmm2,%%xmm5                   \n"
>+      "por       %%xmm4,%%xmm0                   \n"
>+      "pslldq    $0x8,%%xmm5                     \n"
>+      "movdqu    %%xmm0,(%1)                     \n"
>+      "por       %%xmm5,%%xmm1                   \n"
>+      "psrldq    $0x8,%%xmm2                     \n"
>+      "pslldq    $0x4,%%xmm3                     \n"
>+      "por       %%xmm3,%%xmm2                   \n"
>+      "movdqu    %%xmm1,0x10(%1)                 \n"
>+      "movdqu    %%xmm2,0x20(%1)                 \n"
>+      "lea       0x30(%1),%1                     \n"
>+      "sub       $0x10,%2                        \n"
>+      "jg        1b                              \n"
>+      : "+r"(src),                    // %0
>+        "+r"(dst),                    // %1
>+        "+r"(width)                   // %2
>+      : "m"(kShuffleMaskARGBToRGB24)  // %3
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6");
> }
> 
>-void ARGBToRAWRow_SSSE3(const uint8* src, uint8* dst, int width) {
>-  asm volatile (
>-    "movdqa    %3,%%xmm6                       \n"
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    "movdqu    " MEMACCESS2(0x10,0) ",%%xmm1   \n"
>-    "movdqu    " MEMACCESS2(0x20,0) ",%%xmm2   \n"
>-    "movdqu    " MEMACCESS2(0x30,0) ",%%xmm3   \n"
>-    "lea       " MEMLEA(0x40,0) ",%0           \n"
>-    "pshufb    %%xmm6,%%xmm0                   \n"
>-    "pshufb    %%xmm6,%%xmm1                   \n"
>-    "pshufb    %%xmm6,%%xmm2                   \n"
>-    "pshufb    %%xmm6,%%xmm3                   \n"
>-    "movdqa    %%xmm1,%%xmm4                   \n"
>-    "psrldq    $0x4,%%xmm1                     \n"
>-    "pslldq    $0xc,%%xmm4                     \n"
>-    "movdqa    %%xmm2,%%xmm5                   \n"
>-    "por       %%xmm4,%%xmm0                   \n"
>-    "pslldq    $0x8,%%xmm5                     \n"
>-    "movdqu    %%xmm0," MEMACCESS(1) "         \n"
>-    "por       %%xmm5,%%xmm1                   \n"
>-    "psrldq    $0x8,%%xmm2                     \n"
>-    "pslldq    $0x4,%%xmm3                     \n"
>-    "por       %%xmm3,%%xmm2                   \n"
>-    "movdqu    %%xmm1," MEMACCESS2(0x10,1) "   \n"
>-    "movdqu    %%xmm2," MEMACCESS2(0x20,1) "   \n"
>-    "lea       " MEMLEA(0x30,1) ",%1           \n"
>-    "sub       $0x10,%2                        \n"
>-    "jg        1b                              \n"
>-  : "+r"(src),  // %0
>-    "+r"(dst),  // %1
>-    "+r"(width)   // %2
>-  : "m"(kShuffleMaskARGBToRAW)  // %3
>-  : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6"
>-  );
>+void ARGBToRAWRow_SSSE3(const uint8_t* src, uint8_t* dst, int width) {
>+  asm volatile(
>+
>+      "movdqa    %3,%%xmm6                       \n"
>+
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "movdqu    0x10(%0),%%xmm1                 \n"
>+      "movdqu    0x20(%0),%%xmm2                 \n"
>+      "movdqu    0x30(%0),%%xmm3                 \n"
>+      "lea       0x40(%0),%0                     \n"
>+      "pshufb    %%xmm6,%%xmm0                   \n"
>+      "pshufb    %%xmm6,%%xmm1                   \n"
>+      "pshufb    %%xmm6,%%xmm2                   \n"
>+      "pshufb    %%xmm6,%%xmm3                   \n"
>+      "movdqa    %%xmm1,%%xmm4                   \n"
>+      "psrldq    $0x4,%%xmm1                     \n"
>+      "pslldq    $0xc,%%xmm4                     \n"
>+      "movdqa    %%xmm2,%%xmm5                   \n"
>+      "por       %%xmm4,%%xmm0                   \n"
>+      "pslldq    $0x8,%%xmm5                     \n"
>+      "movdqu    %%xmm0,(%1)                     \n"
>+      "por       %%xmm5,%%xmm1                   \n"
>+      "psrldq    $0x8,%%xmm2                     \n"
>+      "pslldq    $0x4,%%xmm3                     \n"
>+      "por       %%xmm3,%%xmm2                   \n"
>+      "movdqu    %%xmm1,0x10(%1)                 \n"
>+      "movdqu    %%xmm2,0x20(%1)                 \n"
>+      "lea       0x30(%1),%1                     \n"
>+      "sub       $0x10,%2                        \n"
>+      "jg        1b                              \n"
>+      : "+r"(src),                  // %0
>+        "+r"(dst),                  // %1
>+        "+r"(width)                 // %2
>+      : "m"(kShuffleMaskARGBToRAW)  // %3
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6");
> }
> 
>-void ARGBToRGB565Row_SSE2(const uint8* src, uint8* dst, int width) {
>-  asm volatile (
>-    "pcmpeqb   %%xmm3,%%xmm3                   \n"
>-    "psrld     $0x1b,%%xmm3                    \n"
>-    "pcmpeqb   %%xmm4,%%xmm4                   \n"
>-    "psrld     $0x1a,%%xmm4                    \n"
>-    "pslld     $0x5,%%xmm4                     \n"
>-    "pcmpeqb   %%xmm5,%%xmm5                   \n"
>-    "pslld     $0xb,%%xmm5                     \n"
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    "movdqa    %%xmm0,%%xmm1                   \n"
>-    "movdqa    %%xmm0,%%xmm2                   \n"
>-    "pslld     $0x8,%%xmm0                     \n"
>-    "psrld     $0x3,%%xmm1                     \n"
>-    "psrld     $0x5,%%xmm2                     \n"
>-    "psrad     $0x10,%%xmm0                    \n"
>-    "pand      %%xmm3,%%xmm1                   \n"
>-    "pand      %%xmm4,%%xmm2                   \n"
>-    "pand      %%xmm5,%%xmm0                   \n"
>-    "por       %%xmm2,%%xmm1                   \n"
>-    "por       %%xmm1,%%xmm0                   \n"
>-    "packssdw  %%xmm0,%%xmm0                   \n"
>-    "lea       " MEMLEA(0x10,0) ",%0           \n"
>-    "movq      %%xmm0," MEMACCESS(1) "         \n"
>-    "lea       " MEMLEA(0x8,1) ",%1            \n"
>-    "sub       $0x4,%2                         \n"
>-    "jg        1b                              \n"
>-  : "+r"(src),  // %0
>-    "+r"(dst),  // %1
>-    "+r"(width)   // %2
>-  :: "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5"
>-  );
>+void ARGBToRGB565Row_SSE2(const uint8_t* src, uint8_t* dst, int width) {
>+  asm volatile(
>+      "pcmpeqb   %%xmm3,%%xmm3                   \n"
>+      "psrld     $0x1b,%%xmm3                    \n"
>+      "pcmpeqb   %%xmm4,%%xmm4                   \n"
>+      "psrld     $0x1a,%%xmm4                    \n"
>+      "pslld     $0x5,%%xmm4                     \n"
>+      "pcmpeqb   %%xmm5,%%xmm5                   \n"
>+      "pslld     $0xb,%%xmm5                     \n"
>+
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "movdqa    %%xmm0,%%xmm1                   \n"
>+      "movdqa    %%xmm0,%%xmm2                   \n"
>+      "pslld     $0x8,%%xmm0                     \n"
>+      "psrld     $0x3,%%xmm1                     \n"
>+      "psrld     $0x5,%%xmm2                     \n"
>+      "psrad     $0x10,%%xmm0                    \n"
>+      "pand      %%xmm3,%%xmm1                   \n"
>+      "pand      %%xmm4,%%xmm2                   \n"
>+      "pand      %%xmm5,%%xmm0                   \n"
>+      "por       %%xmm2,%%xmm1                   \n"
>+      "por       %%xmm1,%%xmm0                   \n"
>+      "packssdw  %%xmm0,%%xmm0                   \n"
>+      "lea       0x10(%0),%0                     \n"
>+      "movq      %%xmm0,(%1)                     \n"
>+      "lea       0x8(%1),%1                      \n"
>+      "sub       $0x4,%2                         \n"
>+      "jg        1b                              \n"
>+      : "+r"(src),   // %0
>+        "+r"(dst),   // %1
>+        "+r"(width)  // %2
>+        ::"memory",
>+        "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5");
> }
> 
>-void ARGBToRGB565DitherRow_SSE2(const uint8* src,
>-                                uint8* dst,
>-                                const uint32 dither4,
>+void ARGBToRGB565DitherRow_SSE2(const uint8_t* src,
>+                                uint8_t* dst,
>+                                const uint32_t dither4,
>                                 int width) {
>   asm volatile(
>       "movd       %3,%%xmm6                      \n"
>@@ -583,9 +590,9 @@ void ARGBToRGB565DitherRow_SSE2(const uint8* src,
> }
> 
> #ifdef HAS_ARGBTORGB565DITHERROW_AVX2
>-void ARGBToRGB565DitherRow_AVX2(const uint8* src,
>-                                uint8* dst,
>-                                const uint32 dither4,
>+void ARGBToRGB565DitherRow_AVX2(const uint8_t* src,
>+                                uint8_t* dst,
>+                                const uint32_t dither4,
>                                 int width) {
>   asm volatile(
>       "vbroadcastss %3,%%xmm6                    \n"
>@@ -628,153 +635,335 @@ void ARGBToRGB565DitherRow_AVX2(const uint8* src,
> }
> #endif  // HAS_ARGBTORGB565DITHERROW_AVX2
> 
>-void ARGBToARGB1555Row_SSE2(const uint8* src, uint8* dst, int width) {
>-  asm volatile (
>-    "pcmpeqb   %%xmm4,%%xmm4                   \n"
>-    "psrld     $0x1b,%%xmm4                    \n"
>-    "movdqa    %%xmm4,%%xmm5                   \n"
>-    "pslld     $0x5,%%xmm5                     \n"
>-    "movdqa    %%xmm4,%%xmm6                   \n"
>-    "pslld     $0xa,%%xmm6                     \n"
>-    "pcmpeqb   %%xmm7,%%xmm7                   \n"
>-    "pslld     $0xf,%%xmm7                     \n"
>+void ARGBToARGB1555Row_SSE2(const uint8_t* src, uint8_t* dst, int width) {
>+  asm volatile(
>+      "pcmpeqb   %%xmm4,%%xmm4                   \n"
>+      "psrld     $0x1b,%%xmm4                    \n"
>+      "movdqa    %%xmm4,%%xmm5                   \n"
>+      "pslld     $0x5,%%xmm5                     \n"
>+      "movdqa    %%xmm4,%%xmm6                   \n"
>+      "pslld     $0xa,%%xmm6                     \n"
>+      "pcmpeqb   %%xmm7,%%xmm7                   \n"
>+      "pslld     $0xf,%%xmm7                     \n"
> 
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    "movdqa    %%xmm0,%%xmm1                   \n"
>-    "movdqa    %%xmm0,%%xmm2                   \n"
>-    "movdqa    %%xmm0,%%xmm3                   \n"
>-    "psrad     $0x10,%%xmm0                    \n"
>-    "psrld     $0x3,%%xmm1                     \n"
>-    "psrld     $0x6,%%xmm2                     \n"
>-    "psrld     $0x9,%%xmm3                     \n"
>-    "pand      %%xmm7,%%xmm0                   \n"
>-    "pand      %%xmm4,%%xmm1                   \n"
>-    "pand      %%xmm5,%%xmm2                   \n"
>-    "pand      %%xmm6,%%xmm3                   \n"
>-    "por       %%xmm1,%%xmm0                   \n"
>-    "por       %%xmm3,%%xmm2                   \n"
>-    "por       %%xmm2,%%xmm0                   \n"
>-    "packssdw  %%xmm0,%%xmm0                   \n"
>-    "lea       " MEMLEA(0x10,0) ",%0           \n"
>-    "movq      %%xmm0," MEMACCESS(1) "         \n"
>-    "lea       " MEMLEA(0x8,1) ",%1            \n"
>-    "sub       $0x4,%2                         \n"
>-    "jg        1b                              \n"
>-  : "+r"(src),  // %0
>-    "+r"(dst),  // %1
>-    "+r"(width)   // %2
>-  :: "memory", "cc",
>-    "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6", "xmm7"
>-  );
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "movdqa    %%xmm0,%%xmm1                   \n"
>+      "movdqa    %%xmm0,%%xmm2                   \n"
>+      "movdqa    %%xmm0,%%xmm3                   \n"
>+      "psrad     $0x10,%%xmm0                    \n"
>+      "psrld     $0x3,%%xmm1                     \n"
>+      "psrld     $0x6,%%xmm2                     \n"
>+      "psrld     $0x9,%%xmm3                     \n"
>+      "pand      %%xmm7,%%xmm0                   \n"
>+      "pand      %%xmm4,%%xmm1                   \n"
>+      "pand      %%xmm5,%%xmm2                   \n"
>+      "pand      %%xmm6,%%xmm3                   \n"
>+      "por       %%xmm1,%%xmm0                   \n"
>+      "por       %%xmm3,%%xmm2                   \n"
>+      "por       %%xmm2,%%xmm0                   \n"
>+      "packssdw  %%xmm0,%%xmm0                   \n"
>+      "lea       0x10(%0),%0                     \n"
>+      "movq      %%xmm0,(%1)                     \n"
>+      "lea       0x8(%1),%1                      \n"
>+      "sub       $0x4,%2                         \n"
>+      "jg        1b                              \n"
>+      : "+r"(src),   // %0
>+        "+r"(dst),   // %1
>+        "+r"(width)  // %2
>+        ::"memory",
>+        "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6", "xmm7");
> }
> 
>-void ARGBToARGB4444Row_SSE2(const uint8* src, uint8* dst, int width) {
>-  asm volatile (
>-    "pcmpeqb   %%xmm4,%%xmm4                   \n"
>-    "psllw     $0xc,%%xmm4                     \n"
>-    "movdqa    %%xmm4,%%xmm3                   \n"
>-    "psrlw     $0x8,%%xmm3                     \n"
>+void ARGBToARGB4444Row_SSE2(const uint8_t* src, uint8_t* dst, int width) {
>+  asm volatile(
>+      "pcmpeqb   %%xmm4,%%xmm4                   \n"
>+      "psllw     $0xc,%%xmm4                     \n"
>+      "movdqa    %%xmm4,%%xmm3                   \n"
>+      "psrlw     $0x8,%%xmm3                     \n"
> 
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    "movdqa    %%xmm0,%%xmm1                   \n"
>-    "pand      %%xmm3,%%xmm0                   \n"
>-    "pand      %%xmm4,%%xmm1                   \n"
>-    "psrlq     $0x4,%%xmm0                     \n"
>-    "psrlq     $0x8,%%xmm1                     \n"
>-    "por       %%xmm1,%%xmm0                   \n"
>-    "packuswb  %%xmm0,%%xmm0                   \n"
>-    "lea       " MEMLEA(0x10,0) ",%0           \n"
>-    "movq      %%xmm0," MEMACCESS(1) "         \n"
>-    "lea       " MEMLEA(0x8,1) ",%1            \n"
>-    "sub       $0x4,%2                         \n"
>-    "jg        1b                              \n"
>-  : "+r"(src),  // %0
>-    "+r"(dst),  // %1
>-    "+r"(width)   // %2
>-  :: "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4"
>-  );
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "movdqa    %%xmm0,%%xmm1                   \n"
>+      "pand      %%xmm3,%%xmm0                   \n"
>+      "pand      %%xmm4,%%xmm1                   \n"
>+      "psrlq     $0x4,%%xmm0                     \n"
>+      "psrlq     $0x8,%%xmm1                     \n"
>+      "por       %%xmm1,%%xmm0                   \n"
>+      "packuswb  %%xmm0,%%xmm0                   \n"
>+      "lea       0x10(%0),%0                     \n"
>+      "movq      %%xmm0,(%1)                     \n"
>+      "lea       0x8(%1),%1                      \n"
>+      "sub       $0x4,%2                         \n"
>+      "jg        1b                              \n"
>+      : "+r"(src),   // %0
>+        "+r"(dst),   // %1
>+        "+r"(width)  // %2
>+        ::"memory",
>+        "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4");
> }
> #endif  // HAS_RGB24TOARGBROW_SSSE3
> 
>+/*
>+
>+ARGBToAR30Row:
>+
>+Red Blue
>+With the 8 bit value in the upper bits of a short, vpmulhuw by (1024+4) will
>+produce a 10 bit value in the low 10 bits of each 16 bit value. This is whats
>+wanted for the blue channel. The red needs to be shifted 4 left, so multiply by
>+(1024+4)*16 for red.
>+
>+Alpha Green
>+Alpha and Green are already in the high bits so vpand can zero out the other
>+bits, keeping just 2 upper bits of alpha and 8 bit green. The same multiplier
>+could be used for Green - (1024+4) putting the 10 bit green in the lsb.  Alpha
>+would be a simple multiplier to shift it into position.  It wants a gap of 10
>+above the green.  Green is 10 bits, so there are 6 bits in the low short.  4
>+more are needed, so a multiplier of 4 gets the 2 bits into the upper 16 bits,
>+and then a shift of 4 is a multiply of 16, so (4*16) = 64.  Then shift the
>+result left 10 to position the A and G channels.
>+*/
>+
>+// Shuffle table for converting RAW to RGB24.  Last 8.
>+static const uvec8 kShuffleRB30 = {128u, 0u, 128u, 2u,  128u, 4u,  128u, 6u,
>+                                   128u, 8u, 128u, 10u, 128u, 12u, 128u, 14u};
>+
>+static const uvec8 kShuffleBR30 = {128u, 2u,  128u, 0u, 128u, 6u,  128u, 4u,
>+                                   128u, 10u, 128u, 8u, 128u, 14u, 128u, 12u};
>+
>+static const uint32_t kMulRB10 = 1028 * 16 * 65536 + 1028;
>+static const uint32_t kMaskRB10 = 0x3ff003ff;
>+static const uint32_t kMaskAG10 = 0xc000ff00;
>+static const uint32_t kMulAG10 = 64 * 65536 + 1028;
>+
>+void ARGBToAR30Row_SSSE3(const uint8_t* src, uint8_t* dst, int width) {
>+  asm volatile(
>+      "movdqa     %3,%%xmm2                     \n"  // shuffler for RB
>+      "movd       %4,%%xmm3                     \n"  // multipler for RB
>+      "movd       %5,%%xmm4                     \n"  // mask for R10 B10
>+      "movd       %6,%%xmm5                     \n"  // mask for AG
>+      "movd       %7,%%xmm6                     \n"  // multipler for AG
>+      "pshufd     $0x0,%%xmm3,%%xmm3            \n"
>+      "pshufd     $0x0,%%xmm4,%%xmm4            \n"
>+      "pshufd     $0x0,%%xmm5,%%xmm5            \n"
>+      "pshufd     $0x0,%%xmm6,%%xmm6            \n"
>+      "sub        %0,%1                         \n"
>+
>+      "1:                                       \n"
>+      "movdqu     (%0),%%xmm0                   \n"  // fetch 4 ARGB pixels
>+      "movdqa     %%xmm0,%%xmm1                 \n"
>+      "pshufb     %%xmm2,%%xmm1                 \n"  // R0B0
>+      "pand       %%xmm5,%%xmm0                 \n"  // A0G0
>+      "pmulhuw    %%xmm3,%%xmm1                 \n"  // X2 R16 X4  B10
>+      "pmulhuw    %%xmm6,%%xmm0                 \n"  // X10 A2 X10 G10
>+      "pand       %%xmm4,%%xmm1                 \n"  // X2 R10 X10 B10
>+      "pslld      $10,%%xmm0                    \n"  // A2 x10 G10 x10
>+      "por        %%xmm1,%%xmm0                 \n"  // A2 R10 G10 B10
>+      "movdqu     %%xmm0,(%1,%0)                \n"  // store 4 AR30 pixels
>+      "add        $0x10,%0                      \n"
>+      "sub        $0x4,%2                       \n"
>+      "jg         1b                            \n"
>+
>+      : "+r"(src),          // %0
>+        "+r"(dst),          // %1
>+        "+r"(width)         // %2
>+      : "m"(kShuffleRB30),  // %3
>+        "m"(kMulRB10),      // %4
>+        "m"(kMaskRB10),     // %5
>+        "m"(kMaskAG10),     // %6
>+        "m"(kMulAG10)       // %7
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6");
>+}
>+
>+void ABGRToAR30Row_SSSE3(const uint8_t* src, uint8_t* dst, int width) {
>+  asm volatile(
>+      "movdqa     %3,%%xmm2                     \n"  // shuffler for RB
>+      "movd       %4,%%xmm3                     \n"  // multipler for RB
>+      "movd       %5,%%xmm4                     \n"  // mask for R10 B10
>+      "movd       %6,%%xmm5                     \n"  // mask for AG
>+      "movd       %7,%%xmm6                     \n"  // multipler for AG
>+      "pshufd     $0x0,%%xmm3,%%xmm3            \n"
>+      "pshufd     $0x0,%%xmm4,%%xmm4            \n"
>+      "pshufd     $0x0,%%xmm5,%%xmm5            \n"
>+      "pshufd     $0x0,%%xmm6,%%xmm6            \n"
>+      "sub        %0,%1                         \n"
>+
>+      "1:                                       \n"
>+      "movdqu     (%0),%%xmm0                   \n"  // fetch 4 ABGR pixels
>+      "movdqa     %%xmm0,%%xmm1                 \n"
>+      "pshufb     %%xmm2,%%xmm1                 \n"  // R0B0
>+      "pand       %%xmm5,%%xmm0                 \n"  // A0G0
>+      "pmulhuw    %%xmm3,%%xmm1                 \n"  // X2 R16 X4  B10
>+      "pmulhuw    %%xmm6,%%xmm0                 \n"  // X10 A2 X10 G10
>+      "pand       %%xmm4,%%xmm1                 \n"  // X2 R10 X10 B10
>+      "pslld      $10,%%xmm0                    \n"  // A2 x10 G10 x10
>+      "por        %%xmm1,%%xmm0                 \n"  // A2 R10 G10 B10
>+      "movdqu     %%xmm0,(%1,%0)                \n"  // store 4 AR30 pixels
>+      "add        $0x10,%0                      \n"
>+      "sub        $0x4,%2                       \n"
>+      "jg         1b                            \n"
>+
>+      : "+r"(src),          // %0
>+        "+r"(dst),          // %1
>+        "+r"(width)         // %2
>+      : "m"(kShuffleBR30),  // %3  reversed shuffler
>+        "m"(kMulRB10),      // %4
>+        "m"(kMaskRB10),     // %5
>+        "m"(kMaskAG10),     // %6
>+        "m"(kMulAG10)       // %7
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6");
>+}
>+
>+#ifdef HAS_ARGBTOAR30ROW_AVX2
>+void ARGBToAR30Row_AVX2(const uint8_t* src, uint8_t* dst, int width) {
>+  asm volatile(
>+      "vbroadcastf128 %3,%%ymm2                  \n"  // shuffler for RB
>+      "vbroadcastss  %4,%%ymm3                   \n"  // multipler for RB
>+      "vbroadcastss  %5,%%ymm4                   \n"  // mask for R10 B10
>+      "vbroadcastss  %6,%%ymm5                   \n"  // mask for AG
>+      "vbroadcastss  %7,%%ymm6                   \n"  // multipler for AG
>+      "sub        %0,%1                          \n"
>+
>+      "1:                                        \n"
>+      "vmovdqu    (%0),%%ymm0                    \n"  // fetch 8 ARGB pixels
>+      "vpshufb    %%ymm2,%%ymm0,%%ymm1           \n"  // R0B0
>+      "vpand      %%ymm5,%%ymm0,%%ymm0           \n"  // A0G0
>+      "vpmulhuw   %%ymm3,%%ymm1,%%ymm1           \n"  // X2 R16 X4  B10
>+      "vpmulhuw   %%ymm6,%%ymm0,%%ymm0           \n"  // X10 A2 X10 G10
>+      "vpand      %%ymm4,%%ymm1,%%ymm1           \n"  // X2 R10 X10 B10
>+      "vpslld     $10,%%ymm0,%%ymm0              \n"  // A2 x10 G10 x10
>+      "vpor       %%ymm1,%%ymm0,%%ymm0           \n"  // A2 R10 G10 B10
>+      "vmovdqu    %%ymm0,(%1,%0)                 \n"  // store 8 AR30 pixels
>+      "add        $0x20,%0                       \n"
>+      "sub        $0x8,%2                        \n"
>+      "jg         1b                             \n"
>+      "vzeroupper                                \n"
>+
>+      : "+r"(src),          // %0
>+        "+r"(dst),          // %1
>+        "+r"(width)         // %2
>+      : "m"(kShuffleRB30),  // %3
>+        "m"(kMulRB10),      // %4
>+        "m"(kMaskRB10),     // %5
>+        "m"(kMaskAG10),     // %6
>+        "m"(kMulAG10)       // %7
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6");
>+}
>+#endif
>+
>+#ifdef HAS_ABGRTOAR30ROW_AVX2
>+void ABGRToAR30Row_AVX2(const uint8_t* src, uint8_t* dst, int width) {
>+  asm volatile(
>+      "vbroadcastf128 %3,%%ymm2                  \n"  // shuffler for RB
>+      "vbroadcastss  %4,%%ymm3                   \n"  // multipler for RB
>+      "vbroadcastss  %5,%%ymm4                   \n"  // mask for R10 B10
>+      "vbroadcastss  %6,%%ymm5                   \n"  // mask for AG
>+      "vbroadcastss  %7,%%ymm6                   \n"  // multipler for AG
>+      "sub        %0,%1                          \n"
>+
>+      "1:                                        \n"
>+      "vmovdqu    (%0),%%ymm0                    \n"  // fetch 8 ABGR pixels
>+      "vpshufb    %%ymm2,%%ymm0,%%ymm1           \n"  // R0B0
>+      "vpand      %%ymm5,%%ymm0,%%ymm0           \n"  // A0G0
>+      "vpmulhuw   %%ymm3,%%ymm1,%%ymm1           \n"  // X2 R16 X4  B10
>+      "vpmulhuw   %%ymm6,%%ymm0,%%ymm0           \n"  // X10 A2 X10 G10
>+      "vpand      %%ymm4,%%ymm1,%%ymm1           \n"  // X2 R10 X10 B10
>+      "vpslld     $10,%%ymm0,%%ymm0              \n"  // A2 x10 G10 x10
>+      "vpor       %%ymm1,%%ymm0,%%ymm0           \n"  // A2 R10 G10 B10
>+      "vmovdqu    %%ymm0,(%1,%0)                 \n"  // store 8 AR30 pixels
>+      "add        $0x20,%0                       \n"
>+      "sub        $0x8,%2                        \n"
>+      "jg         1b                             \n"
>+      "vzeroupper                                \n"
>+
>+      : "+r"(src),          // %0
>+        "+r"(dst),          // %1
>+        "+r"(width)         // %2
>+      : "m"(kShuffleBR30),  // %3  reversed shuffler
>+        "m"(kMulRB10),      // %4
>+        "m"(kMaskRB10),     // %5
>+        "m"(kMaskAG10),     // %6
>+        "m"(kMulAG10)       // %7
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6");
>+}
>+#endif
>+
> #ifdef HAS_ARGBTOYROW_SSSE3
> // Convert 16 ARGB pixels (64 bytes) to 16 Y values.
>-void ARGBToYRow_SSSE3(const uint8* src_argb, uint8* dst_y, int width) {
>-  asm volatile (
>-    "movdqa    %3,%%xmm4                       \n"
>-    "movdqa    %4,%%xmm5                       \n"
>+void ARGBToYRow_SSSE3(const uint8_t* src_argb, uint8_t* dst_y, int width) {
>+  asm volatile(
>+      "movdqa    %3,%%xmm4                       \n"
>+      "movdqa    %4,%%xmm5                       \n"
> 
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    "movdqu    " MEMACCESS2(0x10,0) ",%%xmm1   \n"
>-    "movdqu    " MEMACCESS2(0x20,0) ",%%xmm2   \n"
>-    "movdqu    " MEMACCESS2(0x30,0) ",%%xmm3   \n"
>-    "pmaddubsw %%xmm4,%%xmm0                   \n"
>-    "pmaddubsw %%xmm4,%%xmm1                   \n"
>-    "pmaddubsw %%xmm4,%%xmm2                   \n"
>-    "pmaddubsw %%xmm4,%%xmm3                   \n"
>-    "lea       " MEMLEA(0x40,0) ",%0           \n"
>-    "phaddw    %%xmm1,%%xmm0                   \n"
>-    "phaddw    %%xmm3,%%xmm2                   \n"
>-    "psrlw     $0x7,%%xmm0                     \n"
>-    "psrlw     $0x7,%%xmm2                     \n"
>-    "packuswb  %%xmm2,%%xmm0                   \n"
>-    "paddb     %%xmm5,%%xmm0                   \n"
>-    "movdqu    %%xmm0," MEMACCESS(1) "         \n"
>-    "lea       " MEMLEA(0x10,1) ",%1           \n"
>-    "sub       $0x10,%2                        \n"
>-    "jg        1b                              \n"
>-  : "+r"(src_argb),  // %0
>-    "+r"(dst_y),     // %1
>-    "+r"(width)        // %2
>-  : "m"(kARGBToY),   // %3
>-    "m"(kAddY16)     // %4
>-  : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5"
>-  );
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "movdqu    0x10(%0),%%xmm1                 \n"
>+      "movdqu    0x20(%0),%%xmm2                 \n"
>+      "movdqu    0x30(%0),%%xmm3                 \n"
>+      "pmaddubsw %%xmm4,%%xmm0                   \n"
>+      "pmaddubsw %%xmm4,%%xmm1                   \n"
>+      "pmaddubsw %%xmm4,%%xmm2                   \n"
>+      "pmaddubsw %%xmm4,%%xmm3                   \n"
>+      "lea       0x40(%0),%0                     \n"
>+      "phaddw    %%xmm1,%%xmm0                   \n"
>+      "phaddw    %%xmm3,%%xmm2                   \n"
>+      "psrlw     $0x7,%%xmm0                     \n"
>+      "psrlw     $0x7,%%xmm2                     \n"
>+      "packuswb  %%xmm2,%%xmm0                   \n"
>+      "paddb     %%xmm5,%%xmm0                   \n"
>+      "movdqu    %%xmm0,(%1)                     \n"
>+      "lea       0x10(%1),%1                     \n"
>+      "sub       $0x10,%2                        \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_argb),  // %0
>+        "+r"(dst_y),     // %1
>+        "+r"(width)      // %2
>+      : "m"(kARGBToY),   // %3
>+        "m"(kAddY16)     // %4
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5");
> }
> #endif  // HAS_ARGBTOYROW_SSSE3
> 
> #ifdef HAS_ARGBTOYJROW_SSSE3
> // Convert 16 ARGB pixels (64 bytes) to 16 YJ values.
> // Same as ARGBToYRow but different coefficients, no add 16, but do rounding.
>-void ARGBToYJRow_SSSE3(const uint8* src_argb, uint8* dst_y, int width) {
>-  asm volatile (
>-    "movdqa    %3,%%xmm4                       \n"
>-    "movdqa    %4,%%xmm5                       \n"
>+void ARGBToYJRow_SSSE3(const uint8_t* src_argb, uint8_t* dst_y, int width) {
>+  asm volatile(
>+      "movdqa    %3,%%xmm4                       \n"
>+      "movdqa    %4,%%xmm5                       \n"
> 
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    "movdqu    " MEMACCESS2(0x10,0) ",%%xmm1   \n"
>-    "movdqu    " MEMACCESS2(0x20,0) ",%%xmm2   \n"
>-    "movdqu    " MEMACCESS2(0x30,0) ",%%xmm3   \n"
>-    "pmaddubsw %%xmm4,%%xmm0                   \n"
>-    "pmaddubsw %%xmm4,%%xmm1                   \n"
>-    "pmaddubsw %%xmm4,%%xmm2                   \n"
>-    "pmaddubsw %%xmm4,%%xmm3                   \n"
>-    "lea       " MEMLEA(0x40,0) ",%0           \n"
>-    "phaddw    %%xmm1,%%xmm0                   \n"
>-    "phaddw    %%xmm3,%%xmm2                   \n"
>-    "paddw     %%xmm5,%%xmm0                   \n"
>-    "paddw     %%xmm5,%%xmm2                   \n"
>-    "psrlw     $0x7,%%xmm0                     \n"
>-    "psrlw     $0x7,%%xmm2                     \n"
>-    "packuswb  %%xmm2,%%xmm0                   \n"
>-    "movdqu    %%xmm0," MEMACCESS(1) "         \n"
>-    "lea       " MEMLEA(0x10,1) ",%1           \n"
>-    "sub       $0x10,%2                        \n"
>-    "jg        1b                              \n"
>-  : "+r"(src_argb),  // %0
>-    "+r"(dst_y),     // %1
>-    "+r"(width)        // %2
>-  : "m"(kARGBToYJ),  // %3
>-    "m"(kAddYJ64)    // %4
>-  : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5"
>-  );
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "movdqu    0x10(%0),%%xmm1                 \n"
>+      "movdqu    0x20(%0),%%xmm2                 \n"
>+      "movdqu    0x30(%0),%%xmm3                 \n"
>+      "pmaddubsw %%xmm4,%%xmm0                   \n"
>+      "pmaddubsw %%xmm4,%%xmm1                   \n"
>+      "pmaddubsw %%xmm4,%%xmm2                   \n"
>+      "pmaddubsw %%xmm4,%%xmm3                   \n"
>+      "lea       0x40(%0),%0                     \n"
>+      "phaddw    %%xmm1,%%xmm0                   \n"
>+      "phaddw    %%xmm3,%%xmm2                   \n"
>+      "paddw     %%xmm5,%%xmm0                   \n"
>+      "paddw     %%xmm5,%%xmm2                   \n"
>+      "psrlw     $0x7,%%xmm0                     \n"
>+      "psrlw     $0x7,%%xmm2                     \n"
>+      "packuswb  %%xmm2,%%xmm0                   \n"
>+      "movdqu    %%xmm0,(%1)                     \n"
>+      "lea       0x10(%1),%1                     \n"
>+      "sub       $0x10,%2                        \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_argb),  // %0
>+        "+r"(dst_y),     // %1
>+        "+r"(width)      // %2
>+      : "m"(kARGBToYJ),  // %3
>+        "m"(kAddYJ64)    // %4
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5");
> }
> #endif  // HAS_ARGBTOYJROW_SSSE3
> 
>@@ -783,153 +972,149 @@ void ARGBToYJRow_SSSE3(const uint8* src_argb, uint8* dst_y, int width) {
> static const lvec32 kPermdARGBToY_AVX = {0, 4, 1, 5, 2, 6, 3, 7};
> 
> // Convert 32 ARGB pixels (128 bytes) to 32 Y values.
>-void ARGBToYRow_AVX2(const uint8* src_argb, uint8* dst_y, int width) {
>-  asm volatile (
>-    "vbroadcastf128 %3,%%ymm4                  \n"
>-    "vbroadcastf128 %4,%%ymm5                  \n"
>-    "vmovdqu    %5,%%ymm6                      \n"
>+void ARGBToYRow_AVX2(const uint8_t* src_argb, uint8_t* dst_y, int width) {
>+  asm volatile(
>+      "vbroadcastf128 %3,%%ymm4                  \n"
>+      "vbroadcastf128 %4,%%ymm5                  \n"
>+      "vmovdqu    %5,%%ymm6                      \n"
> 
>-    LABELALIGN
>-    "1:                                        \n"
>-    "vmovdqu    " MEMACCESS(0) ",%%ymm0        \n"
>-    "vmovdqu    " MEMACCESS2(0x20,0) ",%%ymm1  \n"
>-    "vmovdqu    " MEMACCESS2(0x40,0) ",%%ymm2  \n"
>-    "vmovdqu    " MEMACCESS2(0x60,0) ",%%ymm3  \n"
>-    "vpmaddubsw %%ymm4,%%ymm0,%%ymm0           \n"
>-    "vpmaddubsw %%ymm4,%%ymm1,%%ymm1           \n"
>-    "vpmaddubsw %%ymm4,%%ymm2,%%ymm2           \n"
>-    "vpmaddubsw %%ymm4,%%ymm3,%%ymm3           \n"
>-    "lea       " MEMLEA(0x80,0) ",%0           \n"
>-    "vphaddw    %%ymm1,%%ymm0,%%ymm0           \n"  // mutates.
>-    "vphaddw    %%ymm3,%%ymm2,%%ymm2           \n"
>-    "vpsrlw     $0x7,%%ymm0,%%ymm0             \n"
>-    "vpsrlw     $0x7,%%ymm2,%%ymm2             \n"
>-    "vpackuswb  %%ymm2,%%ymm0,%%ymm0           \n"  // mutates.
>-    "vpermd     %%ymm0,%%ymm6,%%ymm0           \n"  // unmutate.
>-    "vpaddb     %%ymm5,%%ymm0,%%ymm0           \n"  // add 16 for Y
>-    "vmovdqu    %%ymm0," MEMACCESS(1) "        \n"
>-    "lea       " MEMLEA(0x20,1) ",%1           \n"
>-    "sub       $0x20,%2                        \n"
>-    "jg        1b                              \n"
>-    "vzeroupper                                \n"
>-  : "+r"(src_argb),  // %0
>-    "+r"(dst_y),     // %1
>-    "+r"(width)        // %2
>-  : "m"(kARGBToY),   // %3
>-    "m"(kAddY16),    // %4
>-    "m"(kPermdARGBToY_AVX)  // %5
>-  : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6"
>-  );
>+      LABELALIGN
>+      "1:                                        \n"
>+      "vmovdqu    (%0),%%ymm0                    \n"
>+      "vmovdqu    0x20(%0),%%ymm1                \n"
>+      "vmovdqu    0x40(%0),%%ymm2                \n"
>+      "vmovdqu    0x60(%0),%%ymm3                \n"
>+      "vpmaddubsw %%ymm4,%%ymm0,%%ymm0           \n"
>+      "vpmaddubsw %%ymm4,%%ymm1,%%ymm1           \n"
>+      "vpmaddubsw %%ymm4,%%ymm2,%%ymm2           \n"
>+      "vpmaddubsw %%ymm4,%%ymm3,%%ymm3           \n"
>+      "lea       0x80(%0),%0                     \n"
>+      "vphaddw    %%ymm1,%%ymm0,%%ymm0           \n"  // mutates.
>+      "vphaddw    %%ymm3,%%ymm2,%%ymm2           \n"
>+      "vpsrlw     $0x7,%%ymm0,%%ymm0             \n"
>+      "vpsrlw     $0x7,%%ymm2,%%ymm2             \n"
>+      "vpackuswb  %%ymm2,%%ymm0,%%ymm0           \n"  // mutates.
>+      "vpermd     %%ymm0,%%ymm6,%%ymm0           \n"  // unmutate.
>+      "vpaddb     %%ymm5,%%ymm0,%%ymm0           \n"  // add 16 for Y
>+      "vmovdqu    %%ymm0,(%1)                    \n"
>+      "lea       0x20(%1),%1                     \n"
>+      "sub       $0x20,%2                        \n"
>+      "jg        1b                              \n"
>+      "vzeroupper                                \n"
>+      : "+r"(src_argb),         // %0
>+        "+r"(dst_y),            // %1
>+        "+r"(width)             // %2
>+      : "m"(kARGBToY),          // %3
>+        "m"(kAddY16),           // %4
>+        "m"(kPermdARGBToY_AVX)  // %5
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6");
> }
> #endif  // HAS_ARGBTOYROW_AVX2
> 
> #ifdef HAS_ARGBTOYJROW_AVX2
> // Convert 32 ARGB pixels (128 bytes) to 32 Y values.
>-void ARGBToYJRow_AVX2(const uint8* src_argb, uint8* dst_y, int width) {
>-  asm volatile (
>-    "vbroadcastf128 %3,%%ymm4                  \n"
>-    "vbroadcastf128 %4,%%ymm5                  \n"
>-    "vmovdqu    %5,%%ymm6                      \n"
>+void ARGBToYJRow_AVX2(const uint8_t* src_argb, uint8_t* dst_y, int width) {
>+  asm volatile(
>+      "vbroadcastf128 %3,%%ymm4                  \n"
>+      "vbroadcastf128 %4,%%ymm5                  \n"
>+      "vmovdqu    %5,%%ymm6                      \n"
> 
>-    LABELALIGN
>-    "1:                                        \n"
>-    "vmovdqu    " MEMACCESS(0) ",%%ymm0        \n"
>-    "vmovdqu    " MEMACCESS2(0x20,0) ",%%ymm1  \n"
>-    "vmovdqu    " MEMACCESS2(0x40,0) ",%%ymm2  \n"
>-    "vmovdqu    " MEMACCESS2(0x60,0) ",%%ymm3  \n"
>-    "vpmaddubsw %%ymm4,%%ymm0,%%ymm0           \n"
>-    "vpmaddubsw %%ymm4,%%ymm1,%%ymm1           \n"
>-    "vpmaddubsw %%ymm4,%%ymm2,%%ymm2           \n"
>-    "vpmaddubsw %%ymm4,%%ymm3,%%ymm3           \n"
>-    "lea       " MEMLEA(0x80,0) ",%0           \n"
>-    "vphaddw    %%ymm1,%%ymm0,%%ymm0           \n"  // mutates.
>-    "vphaddw    %%ymm3,%%ymm2,%%ymm2           \n"
>-    "vpaddw     %%ymm5,%%ymm0,%%ymm0           \n"  // Add .5 for rounding.
>-    "vpaddw     %%ymm5,%%ymm2,%%ymm2           \n"
>-    "vpsrlw     $0x7,%%ymm0,%%ymm0             \n"
>-    "vpsrlw     $0x7,%%ymm2,%%ymm2             \n"
>-    "vpackuswb  %%ymm2,%%ymm0,%%ymm0           \n"  // mutates.
>-    "vpermd     %%ymm0,%%ymm6,%%ymm0           \n"  // unmutate.
>-    "vmovdqu    %%ymm0," MEMACCESS(1) "        \n"
>-    "lea       " MEMLEA(0x20,1) ",%1           \n"
>-    "sub       $0x20,%2                        \n"
>-    "jg        1b                              \n"
>-    "vzeroupper                                \n"
>-  : "+r"(src_argb),  // %0
>-    "+r"(dst_y),     // %1
>-    "+r"(width)        // %2
>-  : "m"(kARGBToYJ),   // %3
>-    "m"(kAddYJ64),    // %4
>-    "m"(kPermdARGBToY_AVX)  // %5
>-  : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6"
>-  );
>+      LABELALIGN
>+      "1:                                        \n"
>+      "vmovdqu    (%0),%%ymm0                    \n"
>+      "vmovdqu    0x20(%0),%%ymm1                \n"
>+      "vmovdqu    0x40(%0),%%ymm2                \n"
>+      "vmovdqu    0x60(%0),%%ymm3                \n"
>+      "vpmaddubsw %%ymm4,%%ymm0,%%ymm0           \n"
>+      "vpmaddubsw %%ymm4,%%ymm1,%%ymm1           \n"
>+      "vpmaddubsw %%ymm4,%%ymm2,%%ymm2           \n"
>+      "vpmaddubsw %%ymm4,%%ymm3,%%ymm3           \n"
>+      "lea       0x80(%0),%0                     \n"
>+      "vphaddw    %%ymm1,%%ymm0,%%ymm0           \n"  // mutates.
>+      "vphaddw    %%ymm3,%%ymm2,%%ymm2           \n"
>+      "vpaddw     %%ymm5,%%ymm0,%%ymm0           \n"  // Add .5 for rounding.
>+      "vpaddw     %%ymm5,%%ymm2,%%ymm2           \n"
>+      "vpsrlw     $0x7,%%ymm0,%%ymm0             \n"
>+      "vpsrlw     $0x7,%%ymm2,%%ymm2             \n"
>+      "vpackuswb  %%ymm2,%%ymm0,%%ymm0           \n"  // mutates.
>+      "vpermd     %%ymm0,%%ymm6,%%ymm0           \n"  // unmutate.
>+      "vmovdqu    %%ymm0,(%1)                    \n"
>+      "lea       0x20(%1),%1                     \n"
>+      "sub       $0x20,%2                        \n"
>+      "jg        1b                              \n"
>+      "vzeroupper                                \n"
>+      : "+r"(src_argb),         // %0
>+        "+r"(dst_y),            // %1
>+        "+r"(width)             // %2
>+      : "m"(kARGBToYJ),         // %3
>+        "m"(kAddYJ64),          // %4
>+        "m"(kPermdARGBToY_AVX)  // %5
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6");
> }
> #endif  // HAS_ARGBTOYJROW_AVX2
> 
> #ifdef HAS_ARGBTOUVROW_SSSE3
>-void ARGBToUVRow_SSSE3(const uint8* src_argb0,
>+void ARGBToUVRow_SSSE3(const uint8_t* src_argb0,
>                        int src_stride_argb,
>-                       uint8* dst_u,
>-                       uint8* dst_v,
>+                       uint8_t* dst_u,
>+                       uint8_t* dst_v,
>                        int width) {
>-  asm volatile (
>-    "movdqa    %5,%%xmm3                       \n"
>-    "movdqa    %6,%%xmm4                       \n"
>-    "movdqa    %7,%%xmm5                       \n"
>-    "sub       %1,%2                           \n"
>+  asm volatile(
>+      "movdqa    %5,%%xmm3                       \n"
>+      "movdqa    %6,%%xmm4                       \n"
>+      "movdqa    %7,%%xmm5                       \n"
>+      "sub       %1,%2                           \n"
> 
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    MEMOPREG(movdqu,0x00,0,4,1,xmm7)            //  movdqu (%0,%4,1),%%xmm7
>-    "pavgb     %%xmm7,%%xmm0                   \n"
>-    "movdqu    " MEMACCESS2(0x10,0) ",%%xmm1   \n"
>-    MEMOPREG(movdqu,0x10,0,4,1,xmm7)            //  movdqu 0x10(%0,%4,1),%%xmm7
>-    "pavgb     %%xmm7,%%xmm1                   \n"
>-    "movdqu    " MEMACCESS2(0x20,0) ",%%xmm2   \n"
>-    MEMOPREG(movdqu,0x20,0,4,1,xmm7)            //  movdqu 0x20(%0,%4,1),%%xmm7
>-    "pavgb     %%xmm7,%%xmm2                   \n"
>-    "movdqu    " MEMACCESS2(0x30,0) ",%%xmm6   \n"
>-    MEMOPREG(movdqu,0x30,0,4,1,xmm7)            //  movdqu 0x30(%0,%4,1),%%xmm7
>-    "pavgb     %%xmm7,%%xmm6                   \n"
>-
>-    "lea       " MEMLEA(0x40,0) ",%0           \n"
>-    "movdqa    %%xmm0,%%xmm7                   \n"
>-    "shufps    $0x88,%%xmm1,%%xmm0             \n"
>-    "shufps    $0xdd,%%xmm1,%%xmm7             \n"
>-    "pavgb     %%xmm7,%%xmm0                   \n"
>-    "movdqa    %%xmm2,%%xmm7                   \n"
>-    "shufps    $0x88,%%xmm6,%%xmm2             \n"
>-    "shufps    $0xdd,%%xmm6,%%xmm7             \n"
>-    "pavgb     %%xmm7,%%xmm2                   \n"
>-    "movdqa    %%xmm0,%%xmm1                   \n"
>-    "movdqa    %%xmm2,%%xmm6                   \n"
>-    "pmaddubsw %%xmm4,%%xmm0                   \n"
>-    "pmaddubsw %%xmm4,%%xmm2                   \n"
>-    "pmaddubsw %%xmm3,%%xmm1                   \n"
>-    "pmaddubsw %%xmm3,%%xmm6                   \n"
>-    "phaddw    %%xmm2,%%xmm0                   \n"
>-    "phaddw    %%xmm6,%%xmm1                   \n"
>-    "psraw     $0x8,%%xmm0                     \n"
>-    "psraw     $0x8,%%xmm1                     \n"
>-    "packsswb  %%xmm1,%%xmm0                   \n"
>-    "paddb     %%xmm5,%%xmm0                   \n"
>-    "movlps    %%xmm0," MEMACCESS(1) "         \n"
>-    MEMOPMEM(movhps,xmm0,0x00,1,2,1)           //  movhps    %%xmm0,(%1,%2,1)
>-    "lea       " MEMLEA(0x8,1) ",%1            \n"
>-    "sub       $0x10,%3                        \n"
>-    "jg        1b                              \n"
>-  : "+r"(src_argb0),       // %0
>-    "+r"(dst_u),           // %1
>-    "+r"(dst_v),           // %2
>-    "+rm"(width)           // %3
>-  : "r"((intptr_t)(src_stride_argb)), // %4
>-    "m"(kARGBToV),  // %5
>-    "m"(kARGBToU),  // %6
>-    "m"(kAddUV128)  // %7
>-  : "memory", "cc", NACL_R14
>-    "xmm0", "xmm1", "xmm2", "xmm6", "xmm7"
>-  );
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "movdqu    0x00(%0,%4,1),%%xmm7            \n"
>+      "pavgb     %%xmm7,%%xmm0                   \n"
>+      "movdqu    0x10(%0),%%xmm1                 \n"
>+      "movdqu    0x10(%0,%4,1),%%xmm7            \n"
>+      "pavgb     %%xmm7,%%xmm1                   \n"
>+      "movdqu    0x20(%0),%%xmm2                 \n"
>+      "movdqu    0x20(%0,%4,1),%%xmm7            \n"
>+      "pavgb     %%xmm7,%%xmm2                   \n"
>+      "movdqu    0x30(%0),%%xmm6                 \n"
>+      "movdqu    0x30(%0,%4,1),%%xmm7            \n"
>+      "pavgb     %%xmm7,%%xmm6                   \n"
>+
>+      "lea       0x40(%0),%0                     \n"
>+      "movdqa    %%xmm0,%%xmm7                   \n"
>+      "shufps    $0x88,%%xmm1,%%xmm0             \n"
>+      "shufps    $0xdd,%%xmm1,%%xmm7             \n"
>+      "pavgb     %%xmm7,%%xmm0                   \n"
>+      "movdqa    %%xmm2,%%xmm7                   \n"
>+      "shufps    $0x88,%%xmm6,%%xmm2             \n"
>+      "shufps    $0xdd,%%xmm6,%%xmm7             \n"
>+      "pavgb     %%xmm7,%%xmm2                   \n"
>+      "movdqa    %%xmm0,%%xmm1                   \n"
>+      "movdqa    %%xmm2,%%xmm6                   \n"
>+      "pmaddubsw %%xmm4,%%xmm0                   \n"
>+      "pmaddubsw %%xmm4,%%xmm2                   \n"
>+      "pmaddubsw %%xmm3,%%xmm1                   \n"
>+      "pmaddubsw %%xmm3,%%xmm6                   \n"
>+      "phaddw    %%xmm2,%%xmm0                   \n"
>+      "phaddw    %%xmm6,%%xmm1                   \n"
>+      "psraw     $0x8,%%xmm0                     \n"
>+      "psraw     $0x8,%%xmm1                     \n"
>+      "packsswb  %%xmm1,%%xmm0                   \n"
>+      "paddb     %%xmm5,%%xmm0                   \n"
>+      "movlps    %%xmm0,(%1)                     \n"
>+      "movhps    %%xmm0,0x00(%1,%2,1)            \n"
>+      "lea       0x8(%1),%1                      \n"
>+      "sub       $0x10,%3                        \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_argb0),                   // %0
>+        "+r"(dst_u),                       // %1
>+        "+r"(dst_v),                       // %2
>+        "+rm"(width)                       // %3
>+      : "r"((intptr_t)(src_stride_argb)),  // %4
>+        "m"(kARGBToV),                     // %5
>+        "m"(kARGBToU),                     // %6
>+        "m"(kAddUV128)                     // %7
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm6", "xmm7");
> }
> #endif  // HAS_ARGBTOUVROW_SSSE3
> 
>@@ -938,643 +1123,644 @@ void ARGBToUVRow_SSSE3(const uint8* src_argb0,
> static const lvec8 kShufARGBToUV_AVX = {
>     0, 1, 8, 9, 2, 3, 10, 11, 4, 5, 12, 13, 6, 7, 14, 15,
>     0, 1, 8, 9, 2, 3, 10, 11, 4, 5, 12, 13, 6, 7, 14, 15};
>-void ARGBToUVRow_AVX2(const uint8* src_argb0,
>+void ARGBToUVRow_AVX2(const uint8_t* src_argb0,
>                       int src_stride_argb,
>-                      uint8* dst_u,
>-                      uint8* dst_v,
>+                      uint8_t* dst_u,
>+                      uint8_t* dst_v,
>                       int width) {
>-  asm volatile (
>-    "vbroadcastf128 %5,%%ymm5                  \n"
>-    "vbroadcastf128 %6,%%ymm6                  \n"
>-    "vbroadcastf128 %7,%%ymm7                  \n"
>-    "sub        %1,%2                          \n"
>+  asm volatile(
>+      "vbroadcastf128 %5,%%ymm5                  \n"
>+      "vbroadcastf128 %6,%%ymm6                  \n"
>+      "vbroadcastf128 %7,%%ymm7                  \n"
>+      "sub        %1,%2                          \n"
> 
>-    LABELALIGN
>-    "1:                                        \n"
>-    "vmovdqu    " MEMACCESS(0) ",%%ymm0        \n"
>-    "vmovdqu    " MEMACCESS2(0x20,0) ",%%ymm1  \n"
>-    "vmovdqu    " MEMACCESS2(0x40,0) ",%%ymm2  \n"
>-    "vmovdqu    " MEMACCESS2(0x60,0) ",%%ymm3  \n"
>-    VMEMOPREG(vpavgb,0x00,0,4,1,ymm0,ymm0)     // vpavgb (%0,%4,1),%%ymm0,%%ymm0
>-    VMEMOPREG(vpavgb,0x20,0,4,1,ymm1,ymm1)
>-    VMEMOPREG(vpavgb,0x40,0,4,1,ymm2,ymm2)
>-    VMEMOPREG(vpavgb,0x60,0,4,1,ymm3,ymm3)
>-    "lea        " MEMLEA(0x80,0) ",%0          \n"
>-    "vshufps    $0x88,%%ymm1,%%ymm0,%%ymm4     \n"
>-    "vshufps    $0xdd,%%ymm1,%%ymm0,%%ymm0     \n"
>-    "vpavgb     %%ymm4,%%ymm0,%%ymm0           \n"
>-    "vshufps    $0x88,%%ymm3,%%ymm2,%%ymm4     \n"
>-    "vshufps    $0xdd,%%ymm3,%%ymm2,%%ymm2     \n"
>-    "vpavgb     %%ymm4,%%ymm2,%%ymm2           \n"
>-
>-    "vpmaddubsw %%ymm7,%%ymm0,%%ymm1           \n"
>-    "vpmaddubsw %%ymm7,%%ymm2,%%ymm3           \n"
>-    "vpmaddubsw %%ymm6,%%ymm0,%%ymm0           \n"
>-    "vpmaddubsw %%ymm6,%%ymm2,%%ymm2           \n"
>-    "vphaddw    %%ymm3,%%ymm1,%%ymm1           \n"
>-    "vphaddw    %%ymm2,%%ymm0,%%ymm0           \n"
>-    "vpsraw     $0x8,%%ymm1,%%ymm1             \n"
>-    "vpsraw     $0x8,%%ymm0,%%ymm0             \n"
>-    "vpacksswb  %%ymm0,%%ymm1,%%ymm0           \n"
>-    "vpermq     $0xd8,%%ymm0,%%ymm0            \n"
>-    "vpshufb    %8,%%ymm0,%%ymm0               \n"
>-    "vpaddb     %%ymm5,%%ymm0,%%ymm0           \n"
>-
>-    "vextractf128 $0x0,%%ymm0," MEMACCESS(1) " \n"
>-    VEXTOPMEM(vextractf128,1,ymm0,0x0,1,2,1) // vextractf128 $1,%%ymm0,(%1,%2,1)
>-    "lea        " MEMLEA(0x10,1) ",%1          \n"
>-    "sub        $0x20,%3                       \n"
>-    "jg         1b                             \n"
>-    "vzeroupper                                \n"
>-  : "+r"(src_argb0),       // %0
>-    "+r"(dst_u),           // %1
>-    "+r"(dst_v),           // %2
>-    "+rm"(width)           // %3
>-  : "r"((intptr_t)(src_stride_argb)), // %4
>-    "m"(kAddUV128),  // %5
>-    "m"(kARGBToV),   // %6
>-    "m"(kARGBToU),   // %7
>-    "m"(kShufARGBToUV_AVX)  // %8
>-  : "memory", "cc", NACL_R14
>-    "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6", "xmm7"
>-  );
>-}
>-#endif  // HAS_ARGBTOUVROW_AVX2
>+      LABELALIGN
>+      "1:                                        \n"
>+      "vmovdqu    (%0),%%ymm0                    \n"
>+      "vmovdqu    0x20(%0),%%ymm1                \n"
>+      "vmovdqu    0x40(%0),%%ymm2                \n"
>+      "vmovdqu    0x60(%0),%%ymm3                \n"
>+      "vpavgb    0x00(%0,%4,1),%%ymm0,%%ymm0     \n"
>+      "vpavgb    0x20(%0,%4,1),%%ymm1,%%ymm1     \n"
>+      "vpavgb    0x40(%0,%4,1),%%ymm2,%%ymm2     \n"
>+      "vpavgb    0x60(%0,%4,1),%%ymm3,%%ymm3     \n"
>+      "lea        0x80(%0),%0                    \n"
>+      "vshufps    $0x88,%%ymm1,%%ymm0,%%ymm4     \n"
>+      "vshufps    $0xdd,%%ymm1,%%ymm0,%%ymm0     \n"
>+      "vpavgb     %%ymm4,%%ymm0,%%ymm0           \n"
>+      "vshufps    $0x88,%%ymm3,%%ymm2,%%ymm4     \n"
>+      "vshufps    $0xdd,%%ymm3,%%ymm2,%%ymm2     \n"
>+      "vpavgb     %%ymm4,%%ymm2,%%ymm2           \n"
>+
>+      "vpmaddubsw %%ymm7,%%ymm0,%%ymm1           \n"
>+      "vpmaddubsw %%ymm7,%%ymm2,%%ymm3           \n"
>+      "vpmaddubsw %%ymm6,%%ymm0,%%ymm0           \n"
>+      "vpmaddubsw %%ymm6,%%ymm2,%%ymm2           \n"
>+      "vphaddw    %%ymm3,%%ymm1,%%ymm1           \n"
>+      "vphaddw    %%ymm2,%%ymm0,%%ymm0           \n"
>+      "vpsraw     $0x8,%%ymm1,%%ymm1             \n"
>+      "vpsraw     $0x8,%%ymm0,%%ymm0             \n"
>+      "vpacksswb  %%ymm0,%%ymm1,%%ymm0           \n"
>+      "vpermq     $0xd8,%%ymm0,%%ymm0            \n"
>+      "vpshufb    %8,%%ymm0,%%ymm0               \n"
>+      "vpaddb     %%ymm5,%%ymm0,%%ymm0           \n"
> 
>-#ifdef HAS_ARGBTOUVJROW_AVX2
>-void ARGBToUVJRow_AVX2(const uint8* src_argb0,
>-                       int src_stride_argb,
>-                       uint8* dst_u,
>-                       uint8* dst_v,
>+      "vextractf128 $0x0,%%ymm0,(%1)             \n"
>+      "vextractf128 $0x1,%%ymm0,0x0(%1,%2,1)     \n"
>+      "lea        0x10(%1),%1                    \n"
>+      "sub        $0x20,%3                       \n"
>+      "jg         1b                             \n"
>+      "vzeroupper                                \n"
>+      : "+r"(src_argb0),                   // %0
>+        "+r"(dst_u),                       // %1
>+        "+r"(dst_v),                       // %2
>+        "+rm"(width)                       // %3
>+      : "r"((intptr_t)(src_stride_argb)),  // %4
>+        "m"(kAddUV128),                    // %5
>+        "m"(kARGBToV),                     // %6
>+        "m"(kARGBToU),                     // %7
>+        "m"(kShufARGBToUV_AVX)             // %8
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6",
>+        "xmm7");
>+}
>+#endif  // HAS_ARGBTOUVROW_AVX2
>+
>+#ifdef HAS_ARGBTOUVJROW_AVX2
>+void ARGBToUVJRow_AVX2(const uint8_t* src_argb0,
>+                       int src_stride_argb,
>+                       uint8_t* dst_u,
>+                       uint8_t* dst_v,
>                        int width) {
>-  asm volatile (
>-    "vbroadcastf128 %5,%%ymm5                  \n"
>-    "vbroadcastf128 %6,%%ymm6                  \n"
>-    "vbroadcastf128 %7,%%ymm7                  \n"
>-    "sub        %1,%2                          \n"
>+  asm volatile(
>+      "vbroadcastf128 %5,%%ymm5                  \n"
>+      "vbroadcastf128 %6,%%ymm6                  \n"
>+      "vbroadcastf128 %7,%%ymm7                  \n"
>+      "sub        %1,%2                          \n"
> 
>-    LABELALIGN
>-    "1:                                        \n"
>-    "vmovdqu    " MEMACCESS(0) ",%%ymm0        \n"
>-    "vmovdqu    " MEMACCESS2(0x20,0) ",%%ymm1  \n"
>-    "vmovdqu    " MEMACCESS2(0x40,0) ",%%ymm2  \n"
>-    "vmovdqu    " MEMACCESS2(0x60,0) ",%%ymm3  \n"
>-    VMEMOPREG(vpavgb,0x00,0,4,1,ymm0,ymm0)     // vpavgb (%0,%4,1),%%ymm0,%%ymm0
>-    VMEMOPREG(vpavgb,0x20,0,4,1,ymm1,ymm1)
>-    VMEMOPREG(vpavgb,0x40,0,4,1,ymm2,ymm2)
>-    VMEMOPREG(vpavgb,0x60,0,4,1,ymm3,ymm3)
>-    "lea       " MEMLEA(0x80,0) ",%0           \n"
>-    "vshufps    $0x88,%%ymm1,%%ymm0,%%ymm4     \n"
>-    "vshufps    $0xdd,%%ymm1,%%ymm0,%%ymm0     \n"
>-    "vpavgb     %%ymm4,%%ymm0,%%ymm0           \n"
>-    "vshufps    $0x88,%%ymm3,%%ymm2,%%ymm4     \n"
>-    "vshufps    $0xdd,%%ymm3,%%ymm2,%%ymm2     \n"
>-    "vpavgb     %%ymm4,%%ymm2,%%ymm2           \n"
>-
>-    "vpmaddubsw %%ymm7,%%ymm0,%%ymm1           \n"
>-    "vpmaddubsw %%ymm7,%%ymm2,%%ymm3           \n"
>-    "vpmaddubsw %%ymm6,%%ymm0,%%ymm0           \n"
>-    "vpmaddubsw %%ymm6,%%ymm2,%%ymm2           \n"
>-    "vphaddw    %%ymm3,%%ymm1,%%ymm1           \n"
>-    "vphaddw    %%ymm2,%%ymm0,%%ymm0           \n"
>-    "vpaddw     %%ymm5,%%ymm0,%%ymm0           \n"
>-    "vpaddw     %%ymm5,%%ymm1,%%ymm1           \n"
>-    "vpsraw     $0x8,%%ymm1,%%ymm1             \n"
>-    "vpsraw     $0x8,%%ymm0,%%ymm0             \n"
>-    "vpacksswb  %%ymm0,%%ymm1,%%ymm0           \n"
>-    "vpermq     $0xd8,%%ymm0,%%ymm0            \n"
>-    "vpshufb    %8,%%ymm0,%%ymm0               \n"
>-
>-    "vextractf128 $0x0,%%ymm0," MEMACCESS(1) " \n"
>-    VEXTOPMEM(vextractf128,1,ymm0,0x0,1,2,1) // vextractf128 $1,%%ymm0,(%1,%2,1)
>-    "lea       " MEMLEA(0x10,1) ",%1           \n"
>-    "sub       $0x20,%3                        \n"
>-    "jg        1b                              \n"
>-    "vzeroupper                                \n"
>-  : "+r"(src_argb0),       // %0
>-    "+r"(dst_u),           // %1
>-    "+r"(dst_v),           // %2
>-    "+rm"(width)           // %3
>-  : "r"((intptr_t)(src_stride_argb)), // %4
>-    "m"(kAddUVJ128),  // %5
>-    "m"(kARGBToVJ),  // %6
>-    "m"(kARGBToUJ),  // %7
>-    "m"(kShufARGBToUV_AVX)  // %8
>-  : "memory", "cc", NACL_R14
>-    "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6", "xmm7"
>-  );
>+      LABELALIGN
>+      "1:                                        \n"
>+      "vmovdqu    (%0),%%ymm0                    \n"
>+      "vmovdqu    0x20(%0),%%ymm1                \n"
>+      "vmovdqu    0x40(%0),%%ymm2                \n"
>+      "vmovdqu    0x60(%0),%%ymm3                \n"
>+      "vpavgb    0x00(%0,%4,1),%%ymm0,%%ymm0     \n"
>+      "vpavgb    0x20(%0,%4,1),%%ymm1,%%ymm1     \n"
>+      "vpavgb    0x40(%0,%4,1),%%ymm2,%%ymm2     \n"
>+      "vpavgb    0x60(%0,%4,1),%%ymm3,%%ymm3     \n"
>+      "lea       0x80(%0),%0                     \n"
>+      "vshufps    $0x88,%%ymm1,%%ymm0,%%ymm4     \n"
>+      "vshufps    $0xdd,%%ymm1,%%ymm0,%%ymm0     \n"
>+      "vpavgb     %%ymm4,%%ymm0,%%ymm0           \n"
>+      "vshufps    $0x88,%%ymm3,%%ymm2,%%ymm4     \n"
>+      "vshufps    $0xdd,%%ymm3,%%ymm2,%%ymm2     \n"
>+      "vpavgb     %%ymm4,%%ymm2,%%ymm2           \n"
>+
>+      "vpmaddubsw %%ymm7,%%ymm0,%%ymm1           \n"
>+      "vpmaddubsw %%ymm7,%%ymm2,%%ymm3           \n"
>+      "vpmaddubsw %%ymm6,%%ymm0,%%ymm0           \n"
>+      "vpmaddubsw %%ymm6,%%ymm2,%%ymm2           \n"
>+      "vphaddw    %%ymm3,%%ymm1,%%ymm1           \n"
>+      "vphaddw    %%ymm2,%%ymm0,%%ymm0           \n"
>+      "vpaddw     %%ymm5,%%ymm0,%%ymm0           \n"
>+      "vpaddw     %%ymm5,%%ymm1,%%ymm1           \n"
>+      "vpsraw     $0x8,%%ymm1,%%ymm1             \n"
>+      "vpsraw     $0x8,%%ymm0,%%ymm0             \n"
>+      "vpacksswb  %%ymm0,%%ymm1,%%ymm0           \n"
>+      "vpermq     $0xd8,%%ymm0,%%ymm0            \n"
>+      "vpshufb    %8,%%ymm0,%%ymm0               \n"
>+
>+      "vextractf128 $0x0,%%ymm0,(%1)             \n"
>+      "vextractf128 $0x1,%%ymm0,0x0(%1,%2,1)     \n"
>+      "lea       0x10(%1),%1                     \n"
>+      "sub       $0x20,%3                        \n"
>+      "jg        1b                              \n"
>+      "vzeroupper                                \n"
>+      : "+r"(src_argb0),                   // %0
>+        "+r"(dst_u),                       // %1
>+        "+r"(dst_v),                       // %2
>+        "+rm"(width)                       // %3
>+      : "r"((intptr_t)(src_stride_argb)),  // %4
>+        "m"(kAddUVJ128),                   // %5
>+        "m"(kARGBToVJ),                    // %6
>+        "m"(kARGBToUJ),                    // %7
>+        "m"(kShufARGBToUV_AVX)             // %8
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6",
>+        "xmm7");
> }
> #endif  // HAS_ARGBTOUVJROW_AVX2
> 
> #ifdef HAS_ARGBTOUVJROW_SSSE3
>-void ARGBToUVJRow_SSSE3(const uint8* src_argb0,
>+void ARGBToUVJRow_SSSE3(const uint8_t* src_argb0,
>                         int src_stride_argb,
>-                        uint8* dst_u,
>-                        uint8* dst_v,
>+                        uint8_t* dst_u,
>+                        uint8_t* dst_v,
>                         int width) {
>-  asm volatile (
>-    "movdqa    %5,%%xmm3                       \n"
>-    "movdqa    %6,%%xmm4                       \n"
>-    "movdqa    %7,%%xmm5                       \n"
>-    "sub       %1,%2                           \n"
>+  asm volatile(
>+      "movdqa    %5,%%xmm3                       \n"
>+      "movdqa    %6,%%xmm4                       \n"
>+      "movdqa    %7,%%xmm5                       \n"
>+      "sub       %1,%2                           \n"
> 
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    MEMOPREG(movdqu,0x00,0,4,1,xmm7)            //  movdqu (%0,%4,1),%%xmm7
>-    "pavgb     %%xmm7,%%xmm0                   \n"
>-    "movdqu    " MEMACCESS2(0x10,0) ",%%xmm1   \n"
>-    MEMOPREG(movdqu,0x10,0,4,1,xmm7)            //  movdqu 0x10(%0,%4,1),%%xmm7
>-    "pavgb     %%xmm7,%%xmm1                   \n"
>-    "movdqu    " MEMACCESS2(0x20,0) ",%%xmm2   \n"
>-    MEMOPREG(movdqu,0x20,0,4,1,xmm7)            //  movdqu 0x20(%0,%4,1),%%xmm7
>-    "pavgb     %%xmm7,%%xmm2                   \n"
>-    "movdqu    " MEMACCESS2(0x30,0) ",%%xmm6   \n"
>-    MEMOPREG(movdqu,0x30,0,4,1,xmm7)            //  movdqu 0x30(%0,%4,1),%%xmm7
>-    "pavgb     %%xmm7,%%xmm6                   \n"
>-
>-    "lea       " MEMLEA(0x40,0) ",%0           \n"
>-    "movdqa    %%xmm0,%%xmm7                   \n"
>-    "shufps    $0x88,%%xmm1,%%xmm0             \n"
>-    "shufps    $0xdd,%%xmm1,%%xmm7             \n"
>-    "pavgb     %%xmm7,%%xmm0                   \n"
>-    "movdqa    %%xmm2,%%xmm7                   \n"
>-    "shufps    $0x88,%%xmm6,%%xmm2             \n"
>-    "shufps    $0xdd,%%xmm6,%%xmm7             \n"
>-    "pavgb     %%xmm7,%%xmm2                   \n"
>-    "movdqa    %%xmm0,%%xmm1                   \n"
>-    "movdqa    %%xmm2,%%xmm6                   \n"
>-    "pmaddubsw %%xmm4,%%xmm0                   \n"
>-    "pmaddubsw %%xmm4,%%xmm2                   \n"
>-    "pmaddubsw %%xmm3,%%xmm1                   \n"
>-    "pmaddubsw %%xmm3,%%xmm6                   \n"
>-    "phaddw    %%xmm2,%%xmm0                   \n"
>-    "phaddw    %%xmm6,%%xmm1                   \n"
>-    "paddw     %%xmm5,%%xmm0                   \n"
>-    "paddw     %%xmm5,%%xmm1                   \n"
>-    "psraw     $0x8,%%xmm0                     \n"
>-    "psraw     $0x8,%%xmm1                     \n"
>-    "packsswb  %%xmm1,%%xmm0                   \n"
>-    "movlps    %%xmm0," MEMACCESS(1) "         \n"
>-    MEMOPMEM(movhps,xmm0,0x00,1,2,1)           //  movhps  %%xmm0,(%1,%2,1)
>-    "lea       " MEMLEA(0x8,1) ",%1            \n"
>-    "sub       $0x10,%3                        \n"
>-    "jg        1b                              \n"
>-  : "+r"(src_argb0),       // %0
>-    "+r"(dst_u),           // %1
>-    "+r"(dst_v),           // %2
>-    "+rm"(width)           // %3
>-  : "r"((intptr_t)(src_stride_argb)), // %4
>-    "m"(kARGBToVJ),  // %5
>-    "m"(kARGBToUJ),  // %6
>-    "m"(kAddUVJ128)  // %7
>-  : "memory", "cc", NACL_R14
>-    "xmm0", "xmm1", "xmm2", "xmm6", "xmm7"
>-  );
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "movdqu    0x00(%0,%4,1),%%xmm7            \n"
>+      "pavgb     %%xmm7,%%xmm0                   \n"
>+      "movdqu    0x10(%0),%%xmm1                 \n"
>+      "movdqu    0x10(%0,%4,1),%%xmm7            \n"
>+      "pavgb     %%xmm7,%%xmm1                   \n"
>+      "movdqu    0x20(%0),%%xmm2                 \n"
>+      "movdqu    0x20(%0,%4,1),%%xmm7            \n"
>+      "pavgb     %%xmm7,%%xmm2                   \n"
>+      "movdqu    0x30(%0),%%xmm6                 \n"
>+      "movdqu    0x30(%0,%4,1),%%xmm7            \n"
>+      "pavgb     %%xmm7,%%xmm6                   \n"
>+
>+      "lea       0x40(%0),%0                     \n"
>+      "movdqa    %%xmm0,%%xmm7                   \n"
>+      "shufps    $0x88,%%xmm1,%%xmm0             \n"
>+      "shufps    $0xdd,%%xmm1,%%xmm7             \n"
>+      "pavgb     %%xmm7,%%xmm0                   \n"
>+      "movdqa    %%xmm2,%%xmm7                   \n"
>+      "shufps    $0x88,%%xmm6,%%xmm2             \n"
>+      "shufps    $0xdd,%%xmm6,%%xmm7             \n"
>+      "pavgb     %%xmm7,%%xmm2                   \n"
>+      "movdqa    %%xmm0,%%xmm1                   \n"
>+      "movdqa    %%xmm2,%%xmm6                   \n"
>+      "pmaddubsw %%xmm4,%%xmm0                   \n"
>+      "pmaddubsw %%xmm4,%%xmm2                   \n"
>+      "pmaddubsw %%xmm3,%%xmm1                   \n"
>+      "pmaddubsw %%xmm3,%%xmm6                   \n"
>+      "phaddw    %%xmm2,%%xmm0                   \n"
>+      "phaddw    %%xmm6,%%xmm1                   \n"
>+      "paddw     %%xmm5,%%xmm0                   \n"
>+      "paddw     %%xmm5,%%xmm1                   \n"
>+      "psraw     $0x8,%%xmm0                     \n"
>+      "psraw     $0x8,%%xmm1                     \n"
>+      "packsswb  %%xmm1,%%xmm0                   \n"
>+      "movlps    %%xmm0,(%1)                     \n"
>+      "movhps    %%xmm0,0x00(%1,%2,1)            \n"
>+      "lea       0x8(%1),%1                      \n"
>+      "sub       $0x10,%3                        \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_argb0),                   // %0
>+        "+r"(dst_u),                       // %1
>+        "+r"(dst_v),                       // %2
>+        "+rm"(width)                       // %3
>+      : "r"((intptr_t)(src_stride_argb)),  // %4
>+        "m"(kARGBToVJ),                    // %5
>+        "m"(kARGBToUJ),                    // %6
>+        "m"(kAddUVJ128)                    // %7
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm6", "xmm7");
> }
> #endif  // HAS_ARGBTOUVJROW_SSSE3
> 
> #ifdef HAS_ARGBTOUV444ROW_SSSE3
>-void ARGBToUV444Row_SSSE3(const uint8* src_argb,
>-                          uint8* dst_u,
>-                          uint8* dst_v,
>+void ARGBToUV444Row_SSSE3(const uint8_t* src_argb,
>+                          uint8_t* dst_u,
>+                          uint8_t* dst_v,
>                           int width) {
>-  asm volatile (
>-    "movdqa    %4,%%xmm3                       \n"
>-    "movdqa    %5,%%xmm4                       \n"
>-    "movdqa    %6,%%xmm5                       \n"
>-    "sub       %1,%2                           \n"
>+  asm volatile(
>+      "movdqa    %4,%%xmm3                       \n"
>+      "movdqa    %5,%%xmm4                       \n"
>+      "movdqa    %6,%%xmm5                       \n"
>+      "sub       %1,%2                           \n"
> 
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    "movdqu    " MEMACCESS2(0x10,0) ",%%xmm1   \n"
>-    "movdqu    " MEMACCESS2(0x20,0) ",%%xmm2   \n"
>-    "movdqu    " MEMACCESS2(0x30,0) ",%%xmm6   \n"
>-    "pmaddubsw %%xmm4,%%xmm0                   \n"
>-    "pmaddubsw %%xmm4,%%xmm1                   \n"
>-    "pmaddubsw %%xmm4,%%xmm2                   \n"
>-    "pmaddubsw %%xmm4,%%xmm6                   \n"
>-    "phaddw    %%xmm1,%%xmm0                   \n"
>-    "phaddw    %%xmm6,%%xmm2                   \n"
>-    "psraw     $0x8,%%xmm0                     \n"
>-    "psraw     $0x8,%%xmm2                     \n"
>-    "packsswb  %%xmm2,%%xmm0                   \n"
>-    "paddb     %%xmm5,%%xmm0                   \n"
>-    "movdqu    %%xmm0," MEMACCESS(1) "         \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    "movdqu    " MEMACCESS2(0x10,0) ",%%xmm1   \n"
>-    "movdqu    " MEMACCESS2(0x20,0) ",%%xmm2   \n"
>-    "movdqu    " MEMACCESS2(0x30,0) ",%%xmm6   \n"
>-    "pmaddubsw %%xmm3,%%xmm0                   \n"
>-    "pmaddubsw %%xmm3,%%xmm1                   \n"
>-    "pmaddubsw %%xmm3,%%xmm2                   \n"
>-    "pmaddubsw %%xmm3,%%xmm6                   \n"
>-    "phaddw    %%xmm1,%%xmm0                   \n"
>-    "phaddw    %%xmm6,%%xmm2                   \n"
>-    "psraw     $0x8,%%xmm0                     \n"
>-    "psraw     $0x8,%%xmm2                     \n"
>-    "packsswb  %%xmm2,%%xmm0                   \n"
>-    "paddb     %%xmm5,%%xmm0                   \n"
>-    "lea       " MEMLEA(0x40,0) ",%0           \n"
>-    MEMOPMEM(movdqu,xmm0,0x00,1,2,1)           //  movdqu  %%xmm0,(%1,%2,1)
>-    "lea       " MEMLEA(0x10,1) ",%1           \n"
>-    "sub       $0x10,%3                        \n"
>-    "jg        1b                              \n"
>-  : "+r"(src_argb),        // %0
>-    "+r"(dst_u),           // %1
>-    "+r"(dst_v),           // %2
>-    "+rm"(width)           // %3
>-  : "m"(kARGBToV),  // %4
>-    "m"(kARGBToU),  // %5
>-    "m"(kAddUV128)  // %6
>-  : "memory", "cc", NACL_R14
>-    "xmm0", "xmm1", "xmm2", "xmm6"
>-  );
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "movdqu    0x10(%0),%%xmm1                 \n"
>+      "movdqu    0x20(%0),%%xmm2                 \n"
>+      "movdqu    0x30(%0),%%xmm6                 \n"
>+      "pmaddubsw %%xmm4,%%xmm0                   \n"
>+      "pmaddubsw %%xmm4,%%xmm1                   \n"
>+      "pmaddubsw %%xmm4,%%xmm2                   \n"
>+      "pmaddubsw %%xmm4,%%xmm6                   \n"
>+      "phaddw    %%xmm1,%%xmm0                   \n"
>+      "phaddw    %%xmm6,%%xmm2                   \n"
>+      "psraw     $0x8,%%xmm0                     \n"
>+      "psraw     $0x8,%%xmm2                     \n"
>+      "packsswb  %%xmm2,%%xmm0                   \n"
>+      "paddb     %%xmm5,%%xmm0                   \n"
>+      "movdqu    %%xmm0,(%1)                     \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "movdqu    0x10(%0),%%xmm1                 \n"
>+      "movdqu    0x20(%0),%%xmm2                 \n"
>+      "movdqu    0x30(%0),%%xmm6                 \n"
>+      "pmaddubsw %%xmm3,%%xmm0                   \n"
>+      "pmaddubsw %%xmm3,%%xmm1                   \n"
>+      "pmaddubsw %%xmm3,%%xmm2                   \n"
>+      "pmaddubsw %%xmm3,%%xmm6                   \n"
>+      "phaddw    %%xmm1,%%xmm0                   \n"
>+      "phaddw    %%xmm6,%%xmm2                   \n"
>+      "psraw     $0x8,%%xmm0                     \n"
>+      "psraw     $0x8,%%xmm2                     \n"
>+      "packsswb  %%xmm2,%%xmm0                   \n"
>+      "paddb     %%xmm5,%%xmm0                   \n"
>+      "lea       0x40(%0),%0                     \n"
>+      "movdqu    %%xmm0,0x00(%1,%2,1)            \n"
>+      "lea       0x10(%1),%1                     \n"
>+      "sub       $0x10,%3                        \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_argb),  // %0
>+        "+r"(dst_u),     // %1
>+        "+r"(dst_v),     // %2
>+        "+rm"(width)     // %3
>+      : "m"(kARGBToV),   // %4
>+        "m"(kARGBToU),   // %5
>+        "m"(kAddUV128)   // %6
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm6");
> }
> #endif  // HAS_ARGBTOUV444ROW_SSSE3
> 
>-void BGRAToYRow_SSSE3(const uint8* src_bgra, uint8* dst_y, int width) {
>-  asm volatile (
>-    "movdqa    %4,%%xmm5                       \n"
>-    "movdqa    %3,%%xmm4                       \n"
>+void BGRAToYRow_SSSE3(const uint8_t* src_bgra, uint8_t* dst_y, int width) {
>+  asm volatile(
>+      "movdqa    %4,%%xmm5                       \n"
>+      "movdqa    %3,%%xmm4                       \n"
> 
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    "movdqu    " MEMACCESS2(0x10,0) ",%%xmm1   \n"
>-    "movdqu    " MEMACCESS2(0x20,0) ",%%xmm2   \n"
>-    "movdqu    " MEMACCESS2(0x30,0) ",%%xmm3   \n"
>-    "pmaddubsw %%xmm4,%%xmm0                   \n"
>-    "pmaddubsw %%xmm4,%%xmm1                   \n"
>-    "pmaddubsw %%xmm4,%%xmm2                   \n"
>-    "pmaddubsw %%xmm4,%%xmm3                   \n"
>-    "lea       " MEMLEA(0x40,0) ",%0           \n"
>-    "phaddw    %%xmm1,%%xmm0                   \n"
>-    "phaddw    %%xmm3,%%xmm2                   \n"
>-    "psrlw     $0x7,%%xmm0                     \n"
>-    "psrlw     $0x7,%%xmm2                     \n"
>-    "packuswb  %%xmm2,%%xmm0                   \n"
>-    "paddb     %%xmm5,%%xmm0                   \n"
>-    "movdqu    %%xmm0," MEMACCESS(1) "         \n"
>-    "lea       " MEMLEA(0x10,1) ",%1           \n"
>-    "sub       $0x10,%2                        \n"
>-    "jg        1b                              \n"
>-  : "+r"(src_bgra),  // %0
>-    "+r"(dst_y),     // %1
>-    "+r"(width)        // %2
>-  : "m"(kBGRAToY),   // %3
>-    "m"(kAddY16)     // %4
>-  : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5"
>-  );
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "movdqu    0x10(%0),%%xmm1                 \n"
>+      "movdqu    0x20(%0),%%xmm2                 \n"
>+      "movdqu    0x30(%0),%%xmm3                 \n"
>+      "pmaddubsw %%xmm4,%%xmm0                   \n"
>+      "pmaddubsw %%xmm4,%%xmm1                   \n"
>+      "pmaddubsw %%xmm4,%%xmm2                   \n"
>+      "pmaddubsw %%xmm4,%%xmm3                   \n"
>+      "lea       0x40(%0),%0                     \n"
>+      "phaddw    %%xmm1,%%xmm0                   \n"
>+      "phaddw    %%xmm3,%%xmm2                   \n"
>+      "psrlw     $0x7,%%xmm0                     \n"
>+      "psrlw     $0x7,%%xmm2                     \n"
>+      "packuswb  %%xmm2,%%xmm0                   \n"
>+      "paddb     %%xmm5,%%xmm0                   \n"
>+      "movdqu    %%xmm0,(%1)                     \n"
>+      "lea       0x10(%1),%1                     \n"
>+      "sub       $0x10,%2                        \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_bgra),  // %0
>+        "+r"(dst_y),     // %1
>+        "+r"(width)      // %2
>+      : "m"(kBGRAToY),   // %3
>+        "m"(kAddY16)     // %4
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5");
> }
> 
>-void BGRAToUVRow_SSSE3(const uint8* src_bgra0,
>+void BGRAToUVRow_SSSE3(const uint8_t* src_bgra0,
>                        int src_stride_bgra,
>-                       uint8* dst_u,
>-                       uint8* dst_v,
>+                       uint8_t* dst_u,
>+                       uint8_t* dst_v,
>                        int width) {
>-  asm volatile (
>-    "movdqa    %5,%%xmm3                       \n"
>-    "movdqa    %6,%%xmm4                       \n"
>-    "movdqa    %7,%%xmm5                       \n"
>-    "sub       %1,%2                           \n"
>+  asm volatile(
>+      "movdqa    %5,%%xmm3                       \n"
>+      "movdqa    %6,%%xmm4                       \n"
>+      "movdqa    %7,%%xmm5                       \n"
>+      "sub       %1,%2                           \n"
> 
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    MEMOPREG(movdqu,0x00,0,4,1,xmm7)            //  movdqu (%0,%4,1),%%xmm7
>-    "pavgb     %%xmm7,%%xmm0                   \n"
>-    "movdqu    " MEMACCESS2(0x10,0) ",%%xmm1   \n"
>-    MEMOPREG(movdqu,0x10,0,4,1,xmm7)            //  movdqu 0x10(%0,%4,1),%%xmm7
>-    "pavgb     %%xmm7,%%xmm1                   \n"
>-    "movdqu    " MEMACCESS2(0x20,0) ",%%xmm2   \n"
>-    MEMOPREG(movdqu,0x20,0,4,1,xmm7)            //  movdqu 0x20(%0,%4,1),%%xmm7
>-    "pavgb     %%xmm7,%%xmm2                   \n"
>-    "movdqu    " MEMACCESS2(0x30,0) ",%%xmm6   \n"
>-    MEMOPREG(movdqu,0x30,0,4,1,xmm7)            //  movdqu 0x30(%0,%4,1),%%xmm7
>-    "pavgb     %%xmm7,%%xmm6                   \n"
>-
>-    "lea       " MEMLEA(0x40,0) ",%0           \n"
>-    "movdqa    %%xmm0,%%xmm7                   \n"
>-    "shufps    $0x88,%%xmm1,%%xmm0             \n"
>-    "shufps    $0xdd,%%xmm1,%%xmm7             \n"
>-    "pavgb     %%xmm7,%%xmm0                   \n"
>-    "movdqa    %%xmm2,%%xmm7                   \n"
>-    "shufps    $0x88,%%xmm6,%%xmm2             \n"
>-    "shufps    $0xdd,%%xmm6,%%xmm7             \n"
>-    "pavgb     %%xmm7,%%xmm2                   \n"
>-    "movdqa    %%xmm0,%%xmm1                   \n"
>-    "movdqa    %%xmm2,%%xmm6                   \n"
>-    "pmaddubsw %%xmm4,%%xmm0                   \n"
>-    "pmaddubsw %%xmm4,%%xmm2                   \n"
>-    "pmaddubsw %%xmm3,%%xmm1                   \n"
>-    "pmaddubsw %%xmm3,%%xmm6                   \n"
>-    "phaddw    %%xmm2,%%xmm0                   \n"
>-    "phaddw    %%xmm6,%%xmm1                   \n"
>-    "psraw     $0x8,%%xmm0                     \n"
>-    "psraw     $0x8,%%xmm1                     \n"
>-    "packsswb  %%xmm1,%%xmm0                   \n"
>-    "paddb     %%xmm5,%%xmm0                   \n"
>-    "movlps    %%xmm0," MEMACCESS(1) "         \n"
>-    MEMOPMEM(movhps,xmm0,0x00,1,2,1)           //  movhps  %%xmm0,(%1,%2,1)
>-    "lea       " MEMLEA(0x8,1) ",%1            \n"
>-    "sub       $0x10,%3                        \n"
>-    "jg        1b                              \n"
>-  : "+r"(src_bgra0),       // %0
>-    "+r"(dst_u),           // %1
>-    "+r"(dst_v),           // %2
>-    "+rm"(width)           // %3
>-  : "r"((intptr_t)(src_stride_bgra)), // %4
>-    "m"(kBGRAToV),  // %5
>-    "m"(kBGRAToU),  // %6
>-    "m"(kAddUV128)  // %7
>-  : "memory", "cc", NACL_R14
>-    "xmm0", "xmm1", "xmm2", "xmm6", "xmm7"
>-  );
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "movdqu    0x00(%0,%4,1),%%xmm7            \n"
>+      "pavgb     %%xmm7,%%xmm0                   \n"
>+      "movdqu    0x10(%0),%%xmm1                 \n"
>+      "movdqu    0x10(%0,%4,1),%%xmm7            \n"
>+      "pavgb     %%xmm7,%%xmm1                   \n"
>+      "movdqu    0x20(%0),%%xmm2                 \n"
>+      "movdqu    0x20(%0,%4,1),%%xmm7            \n"
>+      "pavgb     %%xmm7,%%xmm2                   \n"
>+      "movdqu    0x30(%0),%%xmm6                 \n"
>+      "movdqu    0x30(%0,%4,1),%%xmm7            \n"
>+      "pavgb     %%xmm7,%%xmm6                   \n"
>+
>+      "lea       0x40(%0),%0                     \n"
>+      "movdqa    %%xmm0,%%xmm7                   \n"
>+      "shufps    $0x88,%%xmm1,%%xmm0             \n"
>+      "shufps    $0xdd,%%xmm1,%%xmm7             \n"
>+      "pavgb     %%xmm7,%%xmm0                   \n"
>+      "movdqa    %%xmm2,%%xmm7                   \n"
>+      "shufps    $0x88,%%xmm6,%%xmm2             \n"
>+      "shufps    $0xdd,%%xmm6,%%xmm7             \n"
>+      "pavgb     %%xmm7,%%xmm2                   \n"
>+      "movdqa    %%xmm0,%%xmm1                   \n"
>+      "movdqa    %%xmm2,%%xmm6                   \n"
>+      "pmaddubsw %%xmm4,%%xmm0                   \n"
>+      "pmaddubsw %%xmm4,%%xmm2                   \n"
>+      "pmaddubsw %%xmm3,%%xmm1                   \n"
>+      "pmaddubsw %%xmm3,%%xmm6                   \n"
>+      "phaddw    %%xmm2,%%xmm0                   \n"
>+      "phaddw    %%xmm6,%%xmm1                   \n"
>+      "psraw     $0x8,%%xmm0                     \n"
>+      "psraw     $0x8,%%xmm1                     \n"
>+      "packsswb  %%xmm1,%%xmm0                   \n"
>+      "paddb     %%xmm5,%%xmm0                   \n"
>+      "movlps    %%xmm0,(%1)                     \n"
>+      "movhps    %%xmm0,0x00(%1,%2,1)            \n"
>+      "lea       0x8(%1),%1                      \n"
>+      "sub       $0x10,%3                        \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_bgra0),                   // %0
>+        "+r"(dst_u),                       // %1
>+        "+r"(dst_v),                       // %2
>+        "+rm"(width)                       // %3
>+      : "r"((intptr_t)(src_stride_bgra)),  // %4
>+        "m"(kBGRAToV),                     // %5
>+        "m"(kBGRAToU),                     // %6
>+        "m"(kAddUV128)                     // %7
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm6", "xmm7");
> }
> 
>-void ABGRToYRow_SSSE3(const uint8* src_abgr, uint8* dst_y, int width) {
>-  asm volatile (
>-    "movdqa    %4,%%xmm5                       \n"
>-    "movdqa    %3,%%xmm4                       \n"
>+void ABGRToYRow_SSSE3(const uint8_t* src_abgr, uint8_t* dst_y, int width) {
>+  asm volatile(
>+      "movdqa    %4,%%xmm5                       \n"
>+      "movdqa    %3,%%xmm4                       \n"
> 
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    "movdqu    " MEMACCESS2(0x10,0) ",%%xmm1   \n"
>-    "movdqu    " MEMACCESS2(0x20,0) ",%%xmm2   \n"
>-    "movdqu    " MEMACCESS2(0x30,0) ",%%xmm3   \n"
>-    "pmaddubsw %%xmm4,%%xmm0                   \n"
>-    "pmaddubsw %%xmm4,%%xmm1                   \n"
>-    "pmaddubsw %%xmm4,%%xmm2                   \n"
>-    "pmaddubsw %%xmm4,%%xmm3                   \n"
>-    "lea       " MEMLEA(0x40,0) ",%0           \n"
>-    "phaddw    %%xmm1,%%xmm0                   \n"
>-    "phaddw    %%xmm3,%%xmm2                   \n"
>-    "psrlw     $0x7,%%xmm0                     \n"
>-    "psrlw     $0x7,%%xmm2                     \n"
>-    "packuswb  %%xmm2,%%xmm0                   \n"
>-    "paddb     %%xmm5,%%xmm0                   \n"
>-    "movdqu    %%xmm0," MEMACCESS(1) "         \n"
>-    "lea       " MEMLEA(0x10,1) ",%1           \n"
>-    "sub       $0x10,%2                        \n"
>-    "jg        1b                              \n"
>-  : "+r"(src_abgr),  // %0
>-    "+r"(dst_y),     // %1
>-    "+r"(width)        // %2
>-  : "m"(kABGRToY),   // %3
>-    "m"(kAddY16)     // %4
>-  : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5"
>-  );
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "movdqu    0x10(%0),%%xmm1                 \n"
>+      "movdqu    0x20(%0),%%xmm2                 \n"
>+      "movdqu    0x30(%0),%%xmm3                 \n"
>+      "pmaddubsw %%xmm4,%%xmm0                   \n"
>+      "pmaddubsw %%xmm4,%%xmm1                   \n"
>+      "pmaddubsw %%xmm4,%%xmm2                   \n"
>+      "pmaddubsw %%xmm4,%%xmm3                   \n"
>+      "lea       0x40(%0),%0                     \n"
>+      "phaddw    %%xmm1,%%xmm0                   \n"
>+      "phaddw    %%xmm3,%%xmm2                   \n"
>+      "psrlw     $0x7,%%xmm0                     \n"
>+      "psrlw     $0x7,%%xmm2                     \n"
>+      "packuswb  %%xmm2,%%xmm0                   \n"
>+      "paddb     %%xmm5,%%xmm0                   \n"
>+      "movdqu    %%xmm0,(%1)                     \n"
>+      "lea       0x10(%1),%1                     \n"
>+      "sub       $0x10,%2                        \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_abgr),  // %0
>+        "+r"(dst_y),     // %1
>+        "+r"(width)      // %2
>+      : "m"(kABGRToY),   // %3
>+        "m"(kAddY16)     // %4
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5");
> }
> 
>-void RGBAToYRow_SSSE3(const uint8* src_rgba, uint8* dst_y, int width) {
>-  asm volatile (
>-    "movdqa    %4,%%xmm5                       \n"
>-    "movdqa    %3,%%xmm4                       \n"
>+void RGBAToYRow_SSSE3(const uint8_t* src_rgba, uint8_t* dst_y, int width) {
>+  asm volatile(
>+      "movdqa    %4,%%xmm5                       \n"
>+      "movdqa    %3,%%xmm4                       \n"
> 
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    "movdqu    " MEMACCESS2(0x10,0) ",%%xmm1   \n"
>-    "movdqu    " MEMACCESS2(0x20,0) ",%%xmm2   \n"
>-    "movdqu    " MEMACCESS2(0x30,0) ",%%xmm3   \n"
>-    "pmaddubsw %%xmm4,%%xmm0                   \n"
>-    "pmaddubsw %%xmm4,%%xmm1                   \n"
>-    "pmaddubsw %%xmm4,%%xmm2                   \n"
>-    "pmaddubsw %%xmm4,%%xmm3                   \n"
>-    "lea       " MEMLEA(0x40,0) ",%0           \n"
>-    "phaddw    %%xmm1,%%xmm0                   \n"
>-    "phaddw    %%xmm3,%%xmm2                   \n"
>-    "psrlw     $0x7,%%xmm0                     \n"
>-    "psrlw     $0x7,%%xmm2                     \n"
>-    "packuswb  %%xmm2,%%xmm0                   \n"
>-    "paddb     %%xmm5,%%xmm0                   \n"
>-    "movdqu    %%xmm0," MEMACCESS(1) "         \n"
>-    "lea       " MEMLEA(0x10,1) ",%1           \n"
>-    "sub       $0x10,%2                        \n"
>-    "jg        1b                              \n"
>-  : "+r"(src_rgba),  // %0
>-    "+r"(dst_y),     // %1
>-    "+r"(width)        // %2
>-  : "m"(kRGBAToY),   // %3
>-    "m"(kAddY16)     // %4
>-  : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5"
>-  );
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "movdqu    0x10(%0),%%xmm1                 \n"
>+      "movdqu    0x20(%0),%%xmm2                 \n"
>+      "movdqu    0x30(%0),%%xmm3                 \n"
>+      "pmaddubsw %%xmm4,%%xmm0                   \n"
>+      "pmaddubsw %%xmm4,%%xmm1                   \n"
>+      "pmaddubsw %%xmm4,%%xmm2                   \n"
>+      "pmaddubsw %%xmm4,%%xmm3                   \n"
>+      "lea       0x40(%0),%0                     \n"
>+      "phaddw    %%xmm1,%%xmm0                   \n"
>+      "phaddw    %%xmm3,%%xmm2                   \n"
>+      "psrlw     $0x7,%%xmm0                     \n"
>+      "psrlw     $0x7,%%xmm2                     \n"
>+      "packuswb  %%xmm2,%%xmm0                   \n"
>+      "paddb     %%xmm5,%%xmm0                   \n"
>+      "movdqu    %%xmm0,(%1)                     \n"
>+      "lea       0x10(%1),%1                     \n"
>+      "sub       $0x10,%2                        \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_rgba),  // %0
>+        "+r"(dst_y),     // %1
>+        "+r"(width)      // %2
>+      : "m"(kRGBAToY),   // %3
>+        "m"(kAddY16)     // %4
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5");
> }
> 
>-void ABGRToUVRow_SSSE3(const uint8* src_abgr0,
>+void ABGRToUVRow_SSSE3(const uint8_t* src_abgr0,
>                        int src_stride_abgr,
>-                       uint8* dst_u,
>-                       uint8* dst_v,
>+                       uint8_t* dst_u,
>+                       uint8_t* dst_v,
>                        int width) {
>-  asm volatile (
>-    "movdqa    %5,%%xmm3                       \n"
>-    "movdqa    %6,%%xmm4                       \n"
>-    "movdqa    %7,%%xmm5                       \n"
>-    "sub       %1,%2                           \n"
>+  asm volatile(
>+      "movdqa    %5,%%xmm3                       \n"
>+      "movdqa    %6,%%xmm4                       \n"
>+      "movdqa    %7,%%xmm5                       \n"
>+      "sub       %1,%2                           \n"
> 
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    MEMOPREG(movdqu,0x00,0,4,1,xmm7)            //  movdqu (%0,%4,1),%%xmm7
>-    "pavgb     %%xmm7,%%xmm0                   \n"
>-    "movdqu    " MEMACCESS2(0x10,0) ",%%xmm1   \n"
>-    MEMOPREG(movdqu,0x10,0,4,1,xmm7)            //  movdqu 0x10(%0,%4,1),%%xmm7
>-    "pavgb     %%xmm7,%%xmm1                   \n"
>-    "movdqu    " MEMACCESS2(0x20,0) ",%%xmm2   \n"
>-    MEMOPREG(movdqu,0x20,0,4,1,xmm7)            //  movdqu 0x20(%0,%4,1),%%xmm7
>-    "pavgb     %%xmm7,%%xmm2                   \n"
>-    "movdqu    " MEMACCESS2(0x30,0) ",%%xmm6   \n"
>-    MEMOPREG(movdqu,0x30,0,4,1,xmm7)            //  movdqu 0x30(%0,%4,1),%%xmm7
>-    "pavgb     %%xmm7,%%xmm6                   \n"
>-
>-    "lea       " MEMLEA(0x40,0) ",%0           \n"
>-    "movdqa    %%xmm0,%%xmm7                   \n"
>-    "shufps    $0x88,%%xmm1,%%xmm0             \n"
>-    "shufps    $0xdd,%%xmm1,%%xmm7             \n"
>-    "pavgb     %%xmm7,%%xmm0                   \n"
>-    "movdqa    %%xmm2,%%xmm7                   \n"
>-    "shufps    $0x88,%%xmm6,%%xmm2             \n"
>-    "shufps    $0xdd,%%xmm6,%%xmm7             \n"
>-    "pavgb     %%xmm7,%%xmm2                   \n"
>-    "movdqa    %%xmm0,%%xmm1                   \n"
>-    "movdqa    %%xmm2,%%xmm6                   \n"
>-    "pmaddubsw %%xmm4,%%xmm0                   \n"
>-    "pmaddubsw %%xmm4,%%xmm2                   \n"
>-    "pmaddubsw %%xmm3,%%xmm1                   \n"
>-    "pmaddubsw %%xmm3,%%xmm6                   \n"
>-    "phaddw    %%xmm2,%%xmm0                   \n"
>-    "phaddw    %%xmm6,%%xmm1                   \n"
>-    "psraw     $0x8,%%xmm0                     \n"
>-    "psraw     $0x8,%%xmm1                     \n"
>-    "packsswb  %%xmm1,%%xmm0                   \n"
>-    "paddb     %%xmm5,%%xmm0                   \n"
>-    "movlps    %%xmm0," MEMACCESS(1) "         \n"
>-    MEMOPMEM(movhps,xmm0,0x00,1,2,1)           //  movhps  %%xmm0,(%1,%2,1)
>-    "lea       " MEMLEA(0x8,1) ",%1            \n"
>-    "sub       $0x10,%3                        \n"
>-    "jg        1b                              \n"
>-  : "+r"(src_abgr0),       // %0
>-    "+r"(dst_u),           // %1
>-    "+r"(dst_v),           // %2
>-    "+rm"(width)           // %3
>-  : "r"((intptr_t)(src_stride_abgr)), // %4
>-    "m"(kABGRToV),  // %5
>-    "m"(kABGRToU),  // %6
>-    "m"(kAddUV128)  // %7
>-  : "memory", "cc", NACL_R14
>-    "xmm0", "xmm1", "xmm2", "xmm6", "xmm7"
>-  );
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "movdqu    0x00(%0,%4,1),%%xmm7            \n"
>+      "pavgb     %%xmm7,%%xmm0                   \n"
>+      "movdqu    0x10(%0),%%xmm1                 \n"
>+      "movdqu    0x10(%0,%4,1),%%xmm7            \n"
>+      "pavgb     %%xmm7,%%xmm1                   \n"
>+      "movdqu    0x20(%0),%%xmm2                 \n"
>+      "movdqu    0x20(%0,%4,1),%%xmm7            \n"
>+      "pavgb     %%xmm7,%%xmm2                   \n"
>+      "movdqu    0x30(%0),%%xmm6                 \n"
>+      "movdqu    0x30(%0,%4,1),%%xmm7            \n"
>+      "pavgb     %%xmm7,%%xmm6                   \n"
>+
>+      "lea       0x40(%0),%0                     \n"
>+      "movdqa    %%xmm0,%%xmm7                   \n"
>+      "shufps    $0x88,%%xmm1,%%xmm0             \n"
>+      "shufps    $0xdd,%%xmm1,%%xmm7             \n"
>+      "pavgb     %%xmm7,%%xmm0                   \n"
>+      "movdqa    %%xmm2,%%xmm7                   \n"
>+      "shufps    $0x88,%%xmm6,%%xmm2             \n"
>+      "shufps    $0xdd,%%xmm6,%%xmm7             \n"
>+      "pavgb     %%xmm7,%%xmm2                   \n"
>+      "movdqa    %%xmm0,%%xmm1                   \n"
>+      "movdqa    %%xmm2,%%xmm6                   \n"
>+      "pmaddubsw %%xmm4,%%xmm0                   \n"
>+      "pmaddubsw %%xmm4,%%xmm2                   \n"
>+      "pmaddubsw %%xmm3,%%xmm1                   \n"
>+      "pmaddubsw %%xmm3,%%xmm6                   \n"
>+      "phaddw    %%xmm2,%%xmm0                   \n"
>+      "phaddw    %%xmm6,%%xmm1                   \n"
>+      "psraw     $0x8,%%xmm0                     \n"
>+      "psraw     $0x8,%%xmm1                     \n"
>+      "packsswb  %%xmm1,%%xmm0                   \n"
>+      "paddb     %%xmm5,%%xmm0                   \n"
>+      "movlps    %%xmm0,(%1)                     \n"
>+      "movhps    %%xmm0,0x00(%1,%2,1)            \n"
>+      "lea       0x8(%1),%1                      \n"
>+      "sub       $0x10,%3                        \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_abgr0),                   // %0
>+        "+r"(dst_u),                       // %1
>+        "+r"(dst_v),                       // %2
>+        "+rm"(width)                       // %3
>+      : "r"((intptr_t)(src_stride_abgr)),  // %4
>+        "m"(kABGRToV),                     // %5
>+        "m"(kABGRToU),                     // %6
>+        "m"(kAddUV128)                     // %7
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm6", "xmm7");
> }
> 
>-void RGBAToUVRow_SSSE3(const uint8* src_rgba0,
>+void RGBAToUVRow_SSSE3(const uint8_t* src_rgba0,
>                        int src_stride_rgba,
>-                       uint8* dst_u,
>-                       uint8* dst_v,
>+                       uint8_t* dst_u,
>+                       uint8_t* dst_v,
>                        int width) {
>-  asm volatile (
>-    "movdqa    %5,%%xmm3                       \n"
>-    "movdqa    %6,%%xmm4                       \n"
>-    "movdqa    %7,%%xmm5                       \n"
>-    "sub       %1,%2                           \n"
>+  asm volatile(
>+      "movdqa    %5,%%xmm3                       \n"
>+      "movdqa    %6,%%xmm4                       \n"
>+      "movdqa    %7,%%xmm5                       \n"
>+      "sub       %1,%2                           \n"
> 
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    MEMOPREG(movdqu,0x00,0,4,1,xmm7)            //  movdqu (%0,%4,1),%%xmm7
>-    "pavgb     %%xmm7,%%xmm0                   \n"
>-    "movdqu    " MEMACCESS2(0x10,0) ",%%xmm1   \n"
>-    MEMOPREG(movdqu,0x10,0,4,1,xmm7)            //  movdqu 0x10(%0,%4,1),%%xmm7
>-    "pavgb     %%xmm7,%%xmm1                   \n"
>-    "movdqu    " MEMACCESS2(0x20,0) ",%%xmm2   \n"
>-    MEMOPREG(movdqu,0x20,0,4,1,xmm7)            //  movdqu 0x20(%0,%4,1),%%xmm7
>-    "pavgb     %%xmm7,%%xmm2                   \n"
>-    "movdqu    " MEMACCESS2(0x30,0) ",%%xmm6   \n"
>-    MEMOPREG(movdqu,0x30,0,4,1,xmm7)            //  movdqu 0x30(%0,%4,1),%%xmm7
>-    "pavgb     %%xmm7,%%xmm6                   \n"
>-
>-    "lea       " MEMLEA(0x40,0) ",%0           \n"
>-    "movdqa    %%xmm0,%%xmm7                   \n"
>-    "shufps    $0x88,%%xmm1,%%xmm0             \n"
>-    "shufps    $0xdd,%%xmm1,%%xmm7             \n"
>-    "pavgb     %%xmm7,%%xmm0                   \n"
>-    "movdqa    %%xmm2,%%xmm7                   \n"
>-    "shufps    $0x88,%%xmm6,%%xmm2             \n"
>-    "shufps    $0xdd,%%xmm6,%%xmm7             \n"
>-    "pavgb     %%xmm7,%%xmm2                   \n"
>-    "movdqa    %%xmm0,%%xmm1                   \n"
>-    "movdqa    %%xmm2,%%xmm6                   \n"
>-    "pmaddubsw %%xmm4,%%xmm0                   \n"
>-    "pmaddubsw %%xmm4,%%xmm2                   \n"
>-    "pmaddubsw %%xmm3,%%xmm1                   \n"
>-    "pmaddubsw %%xmm3,%%xmm6                   \n"
>-    "phaddw    %%xmm2,%%xmm0                   \n"
>-    "phaddw    %%xmm6,%%xmm1                   \n"
>-    "psraw     $0x8,%%xmm0                     \n"
>-    "psraw     $0x8,%%xmm1                     \n"
>-    "packsswb  %%xmm1,%%xmm0                   \n"
>-    "paddb     %%xmm5,%%xmm0                   \n"
>-    "movlps    %%xmm0," MEMACCESS(1) "         \n"
>-    MEMOPMEM(movhps,xmm0,0x00,1,2,1)           //  movhps  %%xmm0,(%1,%2,1)
>-    "lea       " MEMLEA(0x8,1) ",%1            \n"
>-    "sub       $0x10,%3                        \n"
>-    "jg        1b                              \n"
>-  : "+r"(src_rgba0),       // %0
>-    "+r"(dst_u),           // %1
>-    "+r"(dst_v),           // %2
>-    "+rm"(width)           // %3
>-  : "r"((intptr_t)(src_stride_rgba)), // %4
>-    "m"(kRGBAToV),  // %5
>-    "m"(kRGBAToU),  // %6
>-    "m"(kAddUV128)  // %7
>-  : "memory", "cc", NACL_R14
>-    "xmm0", "xmm1", "xmm2", "xmm6", "xmm7"
>-  );
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "movdqu    0x00(%0,%4,1),%%xmm7            \n"
>+      "pavgb     %%xmm7,%%xmm0                   \n"
>+      "movdqu    0x10(%0),%%xmm1                 \n"
>+      "movdqu    0x10(%0,%4,1),%%xmm7            \n"
>+      "pavgb     %%xmm7,%%xmm1                   \n"
>+      "movdqu    0x20(%0),%%xmm2                 \n"
>+      "movdqu    0x20(%0,%4,1),%%xmm7            \n"
>+      "pavgb     %%xmm7,%%xmm2                   \n"
>+      "movdqu    0x30(%0),%%xmm6                 \n"
>+      "movdqu    0x30(%0,%4,1),%%xmm7            \n"
>+      "pavgb     %%xmm7,%%xmm6                   \n"
>+
>+      "lea       0x40(%0),%0                     \n"
>+      "movdqa    %%xmm0,%%xmm7                   \n"
>+      "shufps    $0x88,%%xmm1,%%xmm0             \n"
>+      "shufps    $0xdd,%%xmm1,%%xmm7             \n"
>+      "pavgb     %%xmm7,%%xmm0                   \n"
>+      "movdqa    %%xmm2,%%xmm7                   \n"
>+      "shufps    $0x88,%%xmm6,%%xmm2             \n"
>+      "shufps    $0xdd,%%xmm6,%%xmm7             \n"
>+      "pavgb     %%xmm7,%%xmm2                   \n"
>+      "movdqa    %%xmm0,%%xmm1                   \n"
>+      "movdqa    %%xmm2,%%xmm6                   \n"
>+      "pmaddubsw %%xmm4,%%xmm0                   \n"
>+      "pmaddubsw %%xmm4,%%xmm2                   \n"
>+      "pmaddubsw %%xmm3,%%xmm1                   \n"
>+      "pmaddubsw %%xmm3,%%xmm6                   \n"
>+      "phaddw    %%xmm2,%%xmm0                   \n"
>+      "phaddw    %%xmm6,%%xmm1                   \n"
>+      "psraw     $0x8,%%xmm0                     \n"
>+      "psraw     $0x8,%%xmm1                     \n"
>+      "packsswb  %%xmm1,%%xmm0                   \n"
>+      "paddb     %%xmm5,%%xmm0                   \n"
>+      "movlps    %%xmm0,(%1)                     \n"
>+      "movhps    %%xmm0,0x00(%1,%2,1)            \n"
>+      "lea       0x8(%1),%1                      \n"
>+      "sub       $0x10,%3                        \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_rgba0),                   // %0
>+        "+r"(dst_u),                       // %1
>+        "+r"(dst_v),                       // %2
>+        "+rm"(width)                       // %3
>+      : "r"((intptr_t)(src_stride_rgba)),  // %4
>+        "m"(kRGBAToV),                     // %5
>+        "m"(kRGBAToU),                     // %6
>+        "m"(kAddUV128)                     // %7
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm6", "xmm7");
> }
> 
> #if defined(HAS_I422TOARGBROW_SSSE3) || defined(HAS_I422TOARGBROW_AVX2)
> 
> // Read 8 UV from 444
>-#define READYUV444 \
>-  "movq       " MEMACCESS([u_buf]) ",%%xmm0                     \n"            \
>-    MEMOPREG(movq, 0x00, [u_buf], [v_buf], 1, xmm1)                            \
>-    "lea        " MEMLEA(0x8, [u_buf]) ",%[u_buf]               \n"            \
>-    "punpcklbw  %%xmm1,%%xmm0                                   \n"            \
>-    "movq       " MEMACCESS([y_buf]) ",%%xmm4                   \n"            \
>-    "punpcklbw  %%xmm4,%%xmm4                                   \n"            \
>-    "lea        " MEMLEA(0x8, [y_buf]) ",%[y_buf]               \n"
>+#define READYUV444                                                \
>+  "movq       (%[u_buf]),%%xmm0                               \n" \
>+  "movq       0x00(%[u_buf],%[v_buf],1),%%xmm1                \n" \
>+  "lea        0x8(%[u_buf]),%[u_buf]                          \n" \
>+  "punpcklbw  %%xmm1,%%xmm0                                   \n" \
>+  "movq       (%[y_buf]),%%xmm4                               \n" \
>+  "punpcklbw  %%xmm4,%%xmm4                                   \n" \
>+  "lea        0x8(%[y_buf]),%[y_buf]                          \n"
> 
> // Read 4 UV from 422, upsample to 8 UV
>-#define READYUV422 \
>-  "movd       " MEMACCESS([u_buf]) ",%%xmm0                     \n"            \
>-    MEMOPREG(movd, 0x00, [u_buf], [v_buf], 1, xmm1)                            \
>-    "lea        " MEMLEA(0x4, [u_buf]) ",%[u_buf]               \n"            \
>-    "punpcklbw  %%xmm1,%%xmm0                                   \n"            \
>-    "punpcklwd  %%xmm0,%%xmm0                                   \n"            \
>-    "movq       " MEMACCESS([y_buf]) ",%%xmm4                   \n"            \
>-    "punpcklbw  %%xmm4,%%xmm4                                   \n"            \
>-    "lea        " MEMLEA(0x8, [y_buf]) ",%[y_buf]               \n"
>+#define READYUV422                                                \
>+  "movd       (%[u_buf]),%%xmm0                               \n" \
>+  "movd       0x00(%[u_buf],%[v_buf],1),%%xmm1                \n" \
>+  "lea        0x4(%[u_buf]),%[u_buf]                          \n" \
>+  "punpcklbw  %%xmm1,%%xmm0                                   \n" \
>+  "punpcklwd  %%xmm0,%%xmm0                                   \n" \
>+  "movq       (%[y_buf]),%%xmm4                               \n" \
>+  "punpcklbw  %%xmm4,%%xmm4                                   \n" \
>+  "lea        0x8(%[y_buf]),%[y_buf]                          \n"
>+
>+// Read 4 UV from 422 10 bit, upsample to 8 UV
>+// TODO(fbarchard): Consider shufb to replace pack/unpack
>+// TODO(fbarchard): Consider pmulhuw to replace psraw
>+// TODO(fbarchard): Consider pmullw to replace psllw and allow different bits.
>+#define READYUV210                                                \
>+  "movq       (%[u_buf]),%%xmm0                               \n" \
>+  "movq       0x00(%[u_buf],%[v_buf],1),%%xmm1                \n" \
>+  "lea        0x8(%[u_buf]),%[u_buf]                          \n" \
>+  "punpcklwd  %%xmm1,%%xmm0                                   \n" \
>+  "psraw      $0x2,%%xmm0                                     \n" \
>+  "packuswb   %%xmm0,%%xmm0                                   \n" \
>+  "punpcklwd  %%xmm0,%%xmm0                                   \n" \
>+  "movdqu     (%[y_buf]),%%xmm4                               \n" \
>+  "psllw      $0x6,%%xmm4                                     \n" \
>+  "lea        0x10(%[y_buf]),%[y_buf]                         \n"
> 
> // Read 4 UV from 422, upsample to 8 UV.  With 8 Alpha.
>-#define READYUVA422 \
>-  "movd       " MEMACCESS([u_buf]) ",%%xmm0                     \n"            \
>-    MEMOPREG(movd, 0x00, [u_buf], [v_buf], 1, xmm1)                            \
>-    "lea        " MEMLEA(0x4, [u_buf]) ",%[u_buf]               \n"            \
>-    "punpcklbw  %%xmm1,%%xmm0                                   \n"            \
>-    "punpcklwd  %%xmm0,%%xmm0                                   \n"            \
>-    "movq       " MEMACCESS([y_buf]) ",%%xmm4                   \n"            \
>-    "punpcklbw  %%xmm4,%%xmm4                                   \n"            \
>-    "lea        " MEMLEA(0x8, [y_buf]) ",%[y_buf]               \n"            \
>-    "movq       " MEMACCESS([a_buf]) ",%%xmm5                   \n"            \
>-    "lea        " MEMLEA(0x8, [a_buf]) ",%[a_buf]               \n"
>+#define READYUVA422                                               \
>+  "movd       (%[u_buf]),%%xmm0                               \n" \
>+  "movd       0x00(%[u_buf],%[v_buf],1),%%xmm1                \n" \
>+  "lea        0x4(%[u_buf]),%[u_buf]                          \n" \
>+  "punpcklbw  %%xmm1,%%xmm0                                   \n" \
>+  "punpcklwd  %%xmm0,%%xmm0                                   \n" \
>+  "movq       (%[y_buf]),%%xmm4                               \n" \
>+  "punpcklbw  %%xmm4,%%xmm4                                   \n" \
>+  "lea        0x8(%[y_buf]),%[y_buf]                          \n" \
>+  "movq       (%[a_buf]),%%xmm5                               \n" \
>+  "lea        0x8(%[a_buf]),%[a_buf]                          \n"
> 
> // Read 4 UV from NV12, upsample to 8 UV
>-#define READNV12 \
>-  "movq       " MEMACCESS([uv_buf]) ",%%xmm0                    \n"            \
>-    "lea        " MEMLEA(0x8, [uv_buf]) ",%[uv_buf]             \n"            \
>-    "punpcklwd  %%xmm0,%%xmm0                                   \n"            \
>-    "movq       " MEMACCESS([y_buf]) ",%%xmm4                   \n"            \
>-    "punpcklbw  %%xmm4,%%xmm4                                   \n"            \
>-    "lea        " MEMLEA(0x8, [y_buf]) ",%[y_buf]               \n"
>+#define READNV12                                                  \
>+  "movq       (%[uv_buf]),%%xmm0                              \n" \
>+  "lea        0x8(%[uv_buf]),%[uv_buf]                        \n" \
>+  "punpcklwd  %%xmm0,%%xmm0                                   \n" \
>+  "movq       (%[y_buf]),%%xmm4                               \n" \
>+  "punpcklbw  %%xmm4,%%xmm4                                   \n" \
>+  "lea        0x8(%[y_buf]),%[y_buf]                          \n"
> 
> // Read 4 VU from NV21, upsample to 8 UV
>-#define READNV21 \
>-  "movq       " MEMACCESS([vu_buf]) ",%%xmm0                    \n"            \
>-    "lea        " MEMLEA(0x8, [vu_buf]) ",%[vu_buf]             \n"            \
>-    "pshufb     %[kShuffleNV21], %%xmm0                         \n"            \
>-    "movq       " MEMACCESS([y_buf]) ",%%xmm4                   \n"            \
>-    "punpcklbw  %%xmm4,%%xmm4                                   \n"            \
>-    "lea        " MEMLEA(0x8, [y_buf]) ",%[y_buf]               \n"
>+#define READNV21                                                  \
>+  "movq       (%[vu_buf]),%%xmm0                              \n" \
>+  "lea        0x8(%[vu_buf]),%[vu_buf]                        \n" \
>+  "pshufb     %[kShuffleNV21], %%xmm0                         \n" \
>+  "movq       (%[y_buf]),%%xmm4                               \n" \
>+  "punpcklbw  %%xmm4,%%xmm4                                   \n" \
>+  "lea        0x8(%[y_buf]),%[y_buf]                          \n"
> 
> // Read 4 YUY2 with 8 Y and update 4 UV to 8 UV.
>-#define READYUY2 \
>-  "movdqu     " MEMACCESS([yuy2_buf]) ",%%xmm4                  \n"            \
>-    "pshufb     %[kShuffleYUY2Y], %%xmm4                        \n"            \
>-    "movdqu     " MEMACCESS([yuy2_buf]) ",%%xmm0                \n"            \
>-    "pshufb     %[kShuffleYUY2UV], %%xmm0                       \n"            \
>-    "lea        " MEMLEA(0x10, [yuy2_buf]) ",%[yuy2_buf]        \n"
>+#define READYUY2                                                  \
>+  "movdqu     (%[yuy2_buf]),%%xmm4                            \n" \
>+  "pshufb     %[kShuffleYUY2Y], %%xmm4                        \n" \
>+  "movdqu     (%[yuy2_buf]),%%xmm0                            \n" \
>+  "pshufb     %[kShuffleYUY2UV], %%xmm0                       \n" \
>+  "lea        0x10(%[yuy2_buf]),%[yuy2_buf]                   \n"
> 
> // Read 4 UYVY with 8 Y and update 4 UV to 8 UV.
>-#define READUYVY \
>-  "movdqu     " MEMACCESS([uyvy_buf]) ",%%xmm4                  \n"            \
>-    "pshufb     %[kShuffleUYVYY], %%xmm4                        \n"            \
>-    "movdqu     " MEMACCESS([uyvy_buf]) ",%%xmm0                \n"            \
>-    "pshufb     %[kShuffleUYVYUV], %%xmm0                       \n"            \
>-    "lea        " MEMLEA(0x10, [uyvy_buf]) ",%[uyvy_buf]        \n"
>+#define READUYVY                                                  \
>+  "movdqu     (%[uyvy_buf]),%%xmm4                            \n" \
>+  "pshufb     %[kShuffleUYVYY], %%xmm4                        \n" \
>+  "movdqu     (%[uyvy_buf]),%%xmm0                            \n" \
>+  "pshufb     %[kShuffleUYVYUV], %%xmm0                       \n" \
>+  "lea        0x10(%[uyvy_buf]),%[uyvy_buf]                   \n"
> 
> #if defined(__x86_64__)
>-#define YUVTORGB_SETUP(yuvconstants) \
>-  "movdqa     " MEMACCESS([yuvconstants]) ",%%xmm8              \n"            \
>-    "movdqa     " MEMACCESS2(32, [yuvconstants]) ",%%xmm9       \n"            \
>-    "movdqa     " MEMACCESS2(64, [yuvconstants]) ",%%xmm10      \n"            \
>-    "movdqa     " MEMACCESS2(96, [yuvconstants]) ",%%xmm11      \n"            \
>-    "movdqa     " MEMACCESS2(128, [yuvconstants]) ",%%xmm12     \n"            \
>-    "movdqa     " MEMACCESS2(160, [yuvconstants]) ",%%xmm13     \n"            \
>-    "movdqa     " MEMACCESS2(192, [yuvconstants]) ",%%xmm14     \n"
>+#define YUVTORGB_SETUP(yuvconstants)                              \
>+  "movdqa     (%[yuvconstants]),%%xmm8                        \n" \
>+  "movdqa     32(%[yuvconstants]),%%xmm9                      \n" \
>+  "movdqa     64(%[yuvconstants]),%%xmm10                     \n" \
>+  "movdqa     96(%[yuvconstants]),%%xmm11                     \n" \
>+  "movdqa     128(%[yuvconstants]),%%xmm12                    \n" \
>+  "movdqa     160(%[yuvconstants]),%%xmm13                    \n" \
>+  "movdqa     192(%[yuvconstants]),%%xmm14                    \n"
> // Convert 8 pixels: 8 UV and 8 Y
>-#define YUVTORGB(yuvconstants)                                    \
>+#define YUVTORGB16(yuvconstants)                                  \
>   "movdqa     %%xmm0,%%xmm1                                   \n" \
>   "movdqa     %%xmm0,%%xmm2                                   \n" \
>   "movdqa     %%xmm0,%%xmm3                                   \n" \
>@@ -1590,72 +1776,95 @@ void RGBAToUVRow_SSSE3(const uint8* src_rgba0,
>   "pmulhuw    %%xmm14,%%xmm4                                  \n" \
>   "paddsw     %%xmm4,%%xmm0                                   \n" \
>   "paddsw     %%xmm4,%%xmm1                                   \n" \
>-  "paddsw     %%xmm4,%%xmm2                                   \n" \
>-  "psraw      $0x6,%%xmm0                                     \n" \
>-  "psraw      $0x6,%%xmm1                                     \n" \
>-  "psraw      $0x6,%%xmm2                                     \n" \
>-  "packuswb   %%xmm0,%%xmm0                                   \n" \
>-  "packuswb   %%xmm1,%%xmm1                                   \n" \
>-  "packuswb   %%xmm2,%%xmm2                                   \n"
>+  "paddsw     %%xmm4,%%xmm2                                   \n"
> #define YUVTORGB_REGS \
>   "xmm8", "xmm9", "xmm10", "xmm11", "xmm12", "xmm13", "xmm14",
> 
> #else
> #define YUVTORGB_SETUP(yuvconstants)
> // Convert 8 pixels: 8 UV and 8 Y
>-#define YUVTORGB(yuvconstants) \
>-  "movdqa     %%xmm0,%%xmm1                                     \n"            \
>-    "movdqa     %%xmm0,%%xmm2                                   \n"            \
>-    "movdqa     %%xmm0,%%xmm3                                   \n"            \
>-    "movdqa     " MEMACCESS2(96, [yuvconstants]) ",%%xmm0       \n"            \
>-    "pmaddubsw  " MEMACCESS([yuvconstants]) ",%%xmm1            \n"            \
>-    "psubw      %%xmm1,%%xmm0                                   \n"            \
>-    "movdqa     " MEMACCESS2(128, [yuvconstants]) ",%%xmm1      \n"            \
>-    "pmaddubsw  " MEMACCESS2(32, [yuvconstants]) ",%%xmm2       \n"            \
>-    "psubw      %%xmm2,%%xmm1                                   \n"            \
>-    "movdqa     " MEMACCESS2(160, [yuvconstants]) ",%%xmm2      \n"            \
>-    "pmaddubsw  " MEMACCESS2(64, [yuvconstants]) ",%%xmm3       \n"            \
>-    "psubw      %%xmm3,%%xmm2                                   \n"            \
>-    "pmulhuw    " MEMACCESS2(192, [yuvconstants]) ",%%xmm4      \n"            \
>-    "paddsw     %%xmm4,%%xmm0                                   \n"            \
>-    "paddsw     %%xmm4,%%xmm1                                   \n"            \
>-    "paddsw     %%xmm4,%%xmm2                                   \n"            \
>-    "psraw      $0x6,%%xmm0                                     \n"            \
>-    "psraw      $0x6,%%xmm1                                     \n"            \
>-    "psraw      $0x6,%%xmm2                                     \n"            \
>-    "packuswb   %%xmm0,%%xmm0                                   \n"            \
>-    "packuswb   %%xmm1,%%xmm1                                   \n"            \
>-    "packuswb   %%xmm2,%%xmm2                                   \n"
>+#define YUVTORGB16(yuvconstants)                                  \
>+  "movdqa     %%xmm0,%%xmm1                                   \n" \
>+  "movdqa     %%xmm0,%%xmm2                                   \n" \
>+  "movdqa     %%xmm0,%%xmm3                                   \n" \
>+  "movdqa     96(%[yuvconstants]),%%xmm0                      \n" \
>+  "pmaddubsw  (%[yuvconstants]),%%xmm1                        \n" \
>+  "psubw      %%xmm1,%%xmm0                                   \n" \
>+  "movdqa     128(%[yuvconstants]),%%xmm1                     \n" \
>+  "pmaddubsw  32(%[yuvconstants]),%%xmm2                      \n" \
>+  "psubw      %%xmm2,%%xmm1                                   \n" \
>+  "movdqa     160(%[yuvconstants]),%%xmm2                     \n" \
>+  "pmaddubsw  64(%[yuvconstants]),%%xmm3                      \n" \
>+  "psubw      %%xmm3,%%xmm2                                   \n" \
>+  "pmulhuw    192(%[yuvconstants]),%%xmm4                     \n" \
>+  "paddsw     %%xmm4,%%xmm0                                   \n" \
>+  "paddsw     %%xmm4,%%xmm1                                   \n" \
>+  "paddsw     %%xmm4,%%xmm2                                   \n"
> #define YUVTORGB_REGS
> #endif
> 
>+#define YUVTORGB(yuvconstants)                                    \
>+  YUVTORGB16(yuvconstants)                                        \
>+  "psraw      $0x6,%%xmm0                                     \n" \
>+  "psraw      $0x6,%%xmm1                                     \n" \
>+  "psraw      $0x6,%%xmm2                                     \n" \
>+  "packuswb   %%xmm0,%%xmm0                                   \n" \
>+  "packuswb   %%xmm1,%%xmm1                                   \n" \
>+  "packuswb   %%xmm2,%%xmm2                                   \n"
>+
> // Store 8 ARGB values.
>-#define STOREARGB \
>-  "punpcklbw  %%xmm1,%%xmm0                                      \n"           \
>-    "punpcklbw  %%xmm5,%%xmm2                                    \n"           \
>-    "movdqa     %%xmm0,%%xmm1                                    \n"           \
>-    "punpcklwd  %%xmm2,%%xmm0                                    \n"           \
>-    "punpckhwd  %%xmm2,%%xmm1                                    \n"           \
>-    "movdqu     %%xmm0," MEMACCESS([dst_argb]) "                 \n"           \
>-    "movdqu     %%xmm1," MEMACCESS2(0x10, [dst_argb]) "          \n"           \
>-    "lea        " MEMLEA(0x20, [dst_argb]) ", %[dst_argb]        \n"
>+#define STOREARGB                                                  \
>+  "punpcklbw  %%xmm1,%%xmm0                                    \n" \
>+  "punpcklbw  %%xmm5,%%xmm2                                    \n" \
>+  "movdqa     %%xmm0,%%xmm1                                    \n" \
>+  "punpcklwd  %%xmm2,%%xmm0                                    \n" \
>+  "punpckhwd  %%xmm2,%%xmm1                                    \n" \
>+  "movdqu     %%xmm0,(%[dst_argb])                             \n" \
>+  "movdqu     %%xmm1,0x10(%[dst_argb])                         \n" \
>+  "lea        0x20(%[dst_argb]), %[dst_argb]                   \n"
> 
> // Store 8 RGBA values.
>-#define STORERGBA \
>-  "pcmpeqb   %%xmm5,%%xmm5                                       \n"           \
>-    "punpcklbw %%xmm2,%%xmm1                                     \n"           \
>-    "punpcklbw %%xmm0,%%xmm5                                     \n"           \
>-    "movdqa    %%xmm5,%%xmm0                                     \n"           \
>-    "punpcklwd %%xmm1,%%xmm5                                     \n"           \
>-    "punpckhwd %%xmm1,%%xmm0                                     \n"           \
>-    "movdqu    %%xmm5," MEMACCESS([dst_rgba]) "                  \n"           \
>-    "movdqu    %%xmm0," MEMACCESS2(0x10, [dst_rgba]) "           \n"           \
>-    "lea       " MEMLEA(0x20, [dst_rgba]) ",%[dst_rgba]          \n"
>-
>-void OMITFP I444ToARGBRow_SSSE3(const uint8* y_buf,
>-                                const uint8* u_buf,
>-                                const uint8* v_buf,
>-                                uint8* dst_argb,
>+#define STORERGBA                                                  \
>+  "pcmpeqb   %%xmm5,%%xmm5                                     \n" \
>+  "punpcklbw %%xmm2,%%xmm1                                     \n" \
>+  "punpcklbw %%xmm0,%%xmm5                                     \n" \
>+  "movdqa    %%xmm5,%%xmm0                                     \n" \
>+  "punpcklwd %%xmm1,%%xmm5                                     \n" \
>+  "punpckhwd %%xmm1,%%xmm0                                     \n" \
>+  "movdqu    %%xmm5,(%[dst_rgba])                              \n" \
>+  "movdqu    %%xmm0,0x10(%[dst_rgba])                          \n" \
>+  "lea       0x20(%[dst_rgba]),%[dst_rgba]                     \n"
>+
>+// Store 8 AR30 values.
>+#define STOREAR30                                                  \
>+  "psraw      $0x4,%%xmm0                                      \n" \
>+  "psraw      $0x4,%%xmm1                                      \n" \
>+  "psraw      $0x4,%%xmm2                                      \n" \
>+  "pminsw     %%xmm7,%%xmm0                                    \n" \
>+  "pminsw     %%xmm7,%%xmm1                                    \n" \
>+  "pminsw     %%xmm7,%%xmm2                                    \n" \
>+  "pmaxsw     %%xmm6,%%xmm0                                    \n" \
>+  "pmaxsw     %%xmm6,%%xmm1                                    \n" \
>+  "pmaxsw     %%xmm6,%%xmm2                                    \n" \
>+  "psllw      $0x4,%%xmm2                                      \n" \
>+  "movdqa     %%xmm0,%%xmm3                                    \n" \
>+  "punpcklwd  %%xmm2,%%xmm0                                    \n" \
>+  "punpckhwd  %%xmm2,%%xmm3                                    \n" \
>+  "movdqa     %%xmm1,%%xmm2                                    \n" \
>+  "punpcklwd  %%xmm5,%%xmm1                                    \n" \
>+  "punpckhwd  %%xmm5,%%xmm2                                    \n" \
>+  "pslld      $0xa,%%xmm1                                      \n" \
>+  "pslld      $0xa,%%xmm2                                      \n" \
>+  "por        %%xmm1,%%xmm0                                    \n" \
>+  "por        %%xmm2,%%xmm3                                    \n" \
>+  "movdqu     %%xmm0,(%[dst_ar30])                             \n" \
>+  "movdqu     %%xmm3,0x10(%[dst_ar30])                         \n" \
>+  "lea        0x20(%[dst_ar30]), %[dst_ar30]                   \n"
>+
>+void OMITFP I444ToARGBRow_SSSE3(const uint8_t* y_buf,
>+                                const uint8_t* u_buf,
>+                                const uint8_t* v_buf,
>+                                uint8_t* dst_argb,
>                                 const struct YuvConstants* yuvconstants,
>                                 int width) {
>   asm volatile (
>@@ -1676,15 +1885,15 @@ void OMITFP I444ToARGBRow_SSSE3(const uint8* y_buf,
>     [dst_argb]"+r"(dst_argb),  // %[dst_argb]
>     [width]"+rm"(width)    // %[width]
>   : [yuvconstants]"r"(yuvconstants)  // %[yuvconstants]
>-  : "memory", "cc", NACL_R14 YUVTORGB_REGS
>+  : "memory", "cc", YUVTORGB_REGS
>     "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5"
>   );
> }
> 
>-void OMITFP I422ToRGB24Row_SSSE3(const uint8* y_buf,
>-                                 const uint8* u_buf,
>-                                 const uint8* v_buf,
>-                                 uint8* dst_rgb24,
>+void OMITFP I422ToRGB24Row_SSSE3(const uint8_t* y_buf,
>+                                 const uint8_t* u_buf,
>+                                 const uint8_t* v_buf,
>+                                 uint8_t* dst_rgb24,
>                                  const struct YuvConstants* yuvconstants,
>                                  int width) {
>   asm volatile (
>@@ -1705,9 +1914,9 @@ void OMITFP I422ToRGB24Row_SSSE3(const uint8* y_buf,
>     "pshufb    %%xmm5,%%xmm0                   \n"
>     "pshufb    %%xmm6,%%xmm1                   \n"
>     "palignr   $0xc,%%xmm0,%%xmm1              \n"
>-    "movq      %%xmm0," MEMACCESS([dst_rgb24]) "\n"
>-    "movdqu    %%xmm1," MEMACCESS2(0x8,[dst_rgb24]) "\n"
>-    "lea       " MEMLEA(0x18,[dst_rgb24]) ",%[dst_rgb24] \n"
>+    "movq      %%xmm0,(%[dst_rgb24])           \n"
>+    "movdqu    %%xmm1,0x8(%[dst_rgb24])        \n"
>+    "lea       0x18(%[dst_rgb24]),%[dst_rgb24] \n"
>     "subl      $0x8,%[width]                   \n"
>     "jg        1b                              \n"
>   : [y_buf]"+r"(y_buf),    // %[y_buf]
>@@ -1722,15 +1931,15 @@ void OMITFP I422ToRGB24Row_SSSE3(const uint8* y_buf,
>   : [yuvconstants]"r"(yuvconstants),  // %[yuvconstants]
>     [kShuffleMaskARGBToRGB24_0]"m"(kShuffleMaskARGBToRGB24_0),
>     [kShuffleMaskARGBToRGB24]"m"(kShuffleMaskARGBToRGB24)
>-  : "memory", "cc", NACL_R14 YUVTORGB_REGS
>+  : "memory", "cc", YUVTORGB_REGS
>     "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6"
>   );
> }
> 
>-void OMITFP I422ToARGBRow_SSSE3(const uint8* y_buf,
>-                                const uint8* u_buf,
>-                                const uint8* v_buf,
>-                                uint8* dst_argb,
>+void OMITFP I422ToARGBRow_SSSE3(const uint8_t* y_buf,
>+                                const uint8_t* u_buf,
>+                                const uint8_t* v_buf,
>+                                uint8_t* dst_argb,
>                                 const struct YuvConstants* yuvconstants,
>                                 int width) {
>   asm volatile (
>@@ -1751,17 +1960,116 @@ void OMITFP I422ToARGBRow_SSSE3(const uint8* y_buf,
>     [dst_argb]"+r"(dst_argb),  // %[dst_argb]
>     [width]"+rm"(width)    // %[width]
>   : [yuvconstants]"r"(yuvconstants)  // %[yuvconstants]
>-  : "memory", "cc", NACL_R14 YUVTORGB_REGS
>+  : "memory", "cc", YUVTORGB_REGS
>+    "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5"
>+  );
>+}
>+
>+void OMITFP I422ToAR30Row_SSSE3(const uint8_t* y_buf,
>+                                const uint8_t* u_buf,
>+                                const uint8_t* v_buf,
>+                                uint8_t* dst_ar30,
>+                                const struct YuvConstants* yuvconstants,
>+                                int width) {
>+  asm volatile (
>+    YUVTORGB_SETUP(yuvconstants)
>+    "sub       %[u_buf],%[v_buf]               \n"
>+    "pcmpeqb   %%xmm5,%%xmm5                   \n"  // AR30 constants
>+    "psrlw     $14,%%xmm5                      \n"
>+    "psllw     $4,%%xmm5                       \n"  // 2 alpha bits
>+    "pxor      %%xmm6,%%xmm6                   \n"
>+    "pcmpeqb   %%xmm7,%%xmm7                   \n"  // 0 for min
>+    "psrlw     $6,%%xmm7                       \n"  // 1023 for max
>+
>+    LABELALIGN
>+    "1:                                        \n"
>+    READYUV422
>+    YUVTORGB16(yuvconstants)
>+    STOREAR30
>+    "sub       $0x8,%[width]                   \n"
>+    "jg        1b                              \n"
>+  : [y_buf]"+r"(y_buf),    // %[y_buf]
>+    [u_buf]"+r"(u_buf),    // %[u_buf]
>+    [v_buf]"+r"(v_buf),    // %[v_buf]
>+    [dst_ar30]"+r"(dst_ar30),  // %[dst_ar30]
>+    [width]"+rm"(width)    // %[width]
>+  : [yuvconstants]"r"(yuvconstants)  // %[yuvconstants]
>+  : "memory", "cc", YUVTORGB_REGS
>+    "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6", "xmm7"
>+  );
>+}
>+
>+// 10 bit YUV to ARGB
>+void OMITFP I210ToARGBRow_SSSE3(const uint16_t* y_buf,
>+                                const uint16_t* u_buf,
>+                                const uint16_t* v_buf,
>+                                uint8_t* dst_argb,
>+                                const struct YuvConstants* yuvconstants,
>+                                int width) {
>+  asm volatile (
>+    YUVTORGB_SETUP(yuvconstants)
>+    "sub       %[u_buf],%[v_buf]               \n"
>+    "pcmpeqb   %%xmm5,%%xmm5                   \n"
>+
>+    LABELALIGN
>+    "1:                                        \n"
>+    READYUV210
>+    YUVTORGB(yuvconstants)
>+    STOREARGB
>+    "sub       $0x8,%[width]                   \n"
>+    "jg        1b                              \n"
>+  : [y_buf]"+r"(y_buf),    // %[y_buf]
>+    [u_buf]"+r"(u_buf),    // %[u_buf]
>+    [v_buf]"+r"(v_buf),    // %[v_buf]
>+    [dst_argb]"+r"(dst_argb),  // %[dst_argb]
>+    [width]"+rm"(width)    // %[width]
>+  : [yuvconstants]"r"(yuvconstants)  // %[yuvconstants]
>+  : "memory", "cc", YUVTORGB_REGS
>     "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5"
>   );
> }
> 
>+// 10 bit YUV to AR30
>+void OMITFP I210ToAR30Row_SSSE3(const uint16_t* y_buf,
>+                                const uint16_t* u_buf,
>+                                const uint16_t* v_buf,
>+                                uint8_t* dst_ar30,
>+                                const struct YuvConstants* yuvconstants,
>+                                int width) {
>+  asm volatile (
>+    YUVTORGB_SETUP(yuvconstants)
>+    "sub       %[u_buf],%[v_buf]               \n"
>+    "pcmpeqb   %%xmm5,%%xmm5                   \n"
>+    "psrlw     $14,%%xmm5                      \n"
>+    "psllw     $4,%%xmm5                       \n"  // 2 alpha bits
>+    "pxor      %%xmm6,%%xmm6                   \n"
>+    "pcmpeqb   %%xmm7,%%xmm7                   \n"  // 0 for min
>+    "psrlw     $6,%%xmm7                       \n"  // 1023 for max
>+
>+    LABELALIGN
>+    "1:                                        \n"
>+    READYUV210
>+    YUVTORGB16(yuvconstants)
>+    STOREAR30
>+    "sub       $0x8,%[width]                   \n"
>+    "jg        1b                              \n"
>+  : [y_buf]"+r"(y_buf),    // %[y_buf]
>+    [u_buf]"+r"(u_buf),    // %[u_buf]
>+    [v_buf]"+r"(v_buf),    // %[v_buf]
>+    [dst_ar30]"+r"(dst_ar30),  // %[dst_ar30]
>+    [width]"+rm"(width)    // %[width]
>+  : [yuvconstants]"r"(yuvconstants)  // %[yuvconstants]
>+  : "memory", "cc", YUVTORGB_REGS
>+    "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6", "xmm7"
>+  );
>+}
>+
> #ifdef HAS_I422ALPHATOARGBROW_SSSE3
>-void OMITFP I422AlphaToARGBRow_SSSE3(const uint8* y_buf,
>-                                     const uint8* u_buf,
>-                                     const uint8* v_buf,
>-                                     const uint8* a_buf,
>-                                     uint8* dst_argb,
>+void OMITFP I422AlphaToARGBRow_SSSE3(const uint8_t* y_buf,
>+                                     const uint8_t* u_buf,
>+                                     const uint8_t* v_buf,
>+                                     const uint8_t* a_buf,
>+                                     uint8_t* dst_argb,
>                                      const struct YuvConstants* yuvconstants,
>                                      int width) {
>   // clang-format off
>@@ -1787,16 +2095,16 @@ void OMITFP I422AlphaToARGBRow_SSSE3(const uint8* y_buf,
>     [width]"+rm"(width)    // %[width]
> #endif
>   : [yuvconstants]"r"(yuvconstants)  // %[yuvconstants]
>-  : "memory", "cc", NACL_R14 YUVTORGB_REGS
>+  : "memory", "cc", YUVTORGB_REGS
>     "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5"
>   );
>   // clang-format on
> }
> #endif  // HAS_I422ALPHATOARGBROW_SSSE3
> 
>-void OMITFP NV12ToARGBRow_SSSE3(const uint8* y_buf,
>-                                const uint8* uv_buf,
>-                                uint8* dst_argb,
>+void OMITFP NV12ToARGBRow_SSSE3(const uint8_t* y_buf,
>+                                const uint8_t* uv_buf,
>+                                uint8_t* dst_argb,
>                                 const struct YuvConstants* yuvconstants,
>                                 int width) {
>   // clang-format off
>@@ -1816,15 +2124,15 @@ void OMITFP NV12ToARGBRow_SSSE3(const uint8* y_buf,
>     [dst_argb]"+r"(dst_argb),  // %[dst_argb]
>     [width]"+rm"(width)    // %[width]
>   : [yuvconstants]"r"(yuvconstants)  // %[yuvconstants]
>-    : "memory", "cc", YUVTORGB_REGS  // Does not use r14.
>+    : "memory", "cc", YUVTORGB_REGS
>       "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5"
>   );
>   // clang-format on
> }
> 
>-void OMITFP NV21ToARGBRow_SSSE3(const uint8* y_buf,
>-                                const uint8* vu_buf,
>-                                uint8* dst_argb,
>+void OMITFP NV21ToARGBRow_SSSE3(const uint8_t* y_buf,
>+                                const uint8_t* vu_buf,
>+                                uint8_t* dst_argb,
>                                 const struct YuvConstants* yuvconstants,
>                                 int width) {
>   // clang-format off
>@@ -1845,14 +2153,14 @@ void OMITFP NV21ToARGBRow_SSSE3(const uint8* y_buf,
>     [width]"+rm"(width)    // %[width]
>   : [yuvconstants]"r"(yuvconstants), // %[yuvconstants]
>     [kShuffleNV21]"m"(kShuffleNV21)
>-    : "memory", "cc", YUVTORGB_REGS  // Does not use r14.
>+    : "memory", "cc", YUVTORGB_REGS
>       "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5"
>   );
>   // clang-format on
> }
> 
>-void OMITFP YUY2ToARGBRow_SSSE3(const uint8* yuy2_buf,
>-                                uint8* dst_argb,
>+void OMITFP YUY2ToARGBRow_SSSE3(const uint8_t* yuy2_buf,
>+                                uint8_t* dst_argb,
>                                 const struct YuvConstants* yuvconstants,
>                                 int width) {
>   // clang-format off
>@@ -1873,14 +2181,14 @@ void OMITFP YUY2ToARGBRow_SSSE3(const uint8* yuy2_buf,
>   : [yuvconstants]"r"(yuvconstants), // %[yuvconstants]
>     [kShuffleYUY2Y]"m"(kShuffleYUY2Y),
>     [kShuffleYUY2UV]"m"(kShuffleYUY2UV)
>-    : "memory", "cc", YUVTORGB_REGS  // Does not use r14.
>+    : "memory", "cc", YUVTORGB_REGS
>       "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5"
>   );
>   // clang-format on
> }
> 
>-void OMITFP UYVYToARGBRow_SSSE3(const uint8* uyvy_buf,
>-                                uint8* dst_argb,
>+void OMITFP UYVYToARGBRow_SSSE3(const uint8_t* uyvy_buf,
>+                                uint8_t* dst_argb,
>                                 const struct YuvConstants* yuvconstants,
>                                 int width) {
>   // clang-format off
>@@ -1901,16 +2209,16 @@ void OMITFP UYVYToARGBRow_SSSE3(const uint8* uyvy_buf,
>   : [yuvconstants]"r"(yuvconstants), // %[yuvconstants]
>     [kShuffleUYVYY]"m"(kShuffleUYVYY),
>     [kShuffleUYVYUV]"m"(kShuffleUYVYUV)
>-    : "memory", "cc", YUVTORGB_REGS  // Does not use r14.
>+    : "memory", "cc", YUVTORGB_REGS
>       "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5"
>   );
>   // clang-format on
> }
> 
>-void OMITFP I422ToRGBARow_SSSE3(const uint8* y_buf,
>-                                const uint8* u_buf,
>-                                const uint8* v_buf,
>-                                uint8* dst_rgba,
>+void OMITFP I422ToRGBARow_SSSE3(const uint8_t* y_buf,
>+                                const uint8_t* u_buf,
>+                                const uint8_t* v_buf,
>+                                uint8_t* dst_rgba,
>                                 const struct YuvConstants* yuvconstants,
>                                 int width) {
>   asm volatile (
>@@ -1931,7 +2239,7 @@ void OMITFP I422ToRGBARow_SSSE3(const uint8* y_buf,
>     [dst_rgba]"+r"(dst_rgba),  // %[dst_rgba]
>     [width]"+rm"(width)    // %[width]
>   : [yuvconstants]"r"(yuvconstants)  // %[yuvconstants]
>-  : "memory", "cc", NACL_R14 YUVTORGB_REGS
>+  : "memory", "cc", YUVTORGB_REGS
>     "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5"
>   );
> }
>@@ -1939,96 +2247,113 @@ void OMITFP I422ToRGBARow_SSSE3(const uint8* y_buf,
> #endif  // HAS_I422TOARGBROW_SSSE3
> 
> // Read 16 UV from 444
>-#define READYUV444_AVX2 \
>-  "vmovdqu    " MEMACCESS([u_buf]) ",%%xmm0                         \n"        \
>-    MEMOPREG(vmovdqu, 0x00, [u_buf], [v_buf], 1, xmm1)                         \
>-    "lea        " MEMLEA(0x10, [u_buf]) ",%[u_buf]                  \n"        \
>-    "vpermq     $0xd8,%%ymm0,%%ymm0                                 \n"        \
>-    "vpermq     $0xd8,%%ymm1,%%ymm1                                 \n"        \
>-    "vpunpcklbw %%ymm1,%%ymm0,%%ymm0                                \n"        \
>-    "vmovdqu    " MEMACCESS([y_buf]) ",%%xmm4                       \n"        \
>-    "vpermq     $0xd8,%%ymm4,%%ymm4                                 \n"        \
>-    "vpunpcklbw %%ymm4,%%ymm4,%%ymm4                                \n"        \
>-    "lea        " MEMLEA(0x10, [y_buf]) ",%[y_buf]                  \n"
>+#define READYUV444_AVX2                                               \
>+  "vmovdqu    (%[u_buf]),%%xmm0                                   \n" \
>+  "vmovdqu    0x00(%[u_buf],%[v_buf],1),%%xmm1                    \n" \
>+  "lea        0x10(%[u_buf]),%[u_buf]                             \n" \
>+  "vpermq     $0xd8,%%ymm0,%%ymm0                                 \n" \
>+  "vpermq     $0xd8,%%ymm1,%%ymm1                                 \n" \
>+  "vpunpcklbw %%ymm1,%%ymm0,%%ymm0                                \n" \
>+  "vmovdqu    (%[y_buf]),%%xmm4                                   \n" \
>+  "vpermq     $0xd8,%%ymm4,%%ymm4                                 \n" \
>+  "vpunpcklbw %%ymm4,%%ymm4,%%ymm4                                \n" \
>+  "lea        0x10(%[y_buf]),%[y_buf]                             \n"
> 
> // Read 8 UV from 422, upsample to 16 UV.
>-#define READYUV422_AVX2 \
>-  "vmovq      " MEMACCESS([u_buf]) ",%%xmm0                         \n"        \
>-    MEMOPREG(vmovq, 0x00, [u_buf], [v_buf], 1, xmm1)                           \
>-    "lea        " MEMLEA(0x8, [u_buf]) ",%[u_buf]                   \n"        \
>-    "vpunpcklbw %%ymm1,%%ymm0,%%ymm0                                \n"        \
>-    "vpermq     $0xd8,%%ymm0,%%ymm0                                 \n"        \
>-    "vpunpcklwd %%ymm0,%%ymm0,%%ymm0                                \n"        \
>-    "vmovdqu    " MEMACCESS([y_buf]) ",%%xmm4                       \n"        \
>-    "vpermq     $0xd8,%%ymm4,%%ymm4                                 \n"        \
>-    "vpunpcklbw %%ymm4,%%ymm4,%%ymm4                                \n"        \
>-    "lea        " MEMLEA(0x10, [y_buf]) ",%[y_buf]                  \n"
>+#define READYUV422_AVX2                                               \
>+  "vmovq      (%[u_buf]),%%xmm0                                   \n" \
>+  "vmovq      0x00(%[u_buf],%[v_buf],1),%%xmm1                    \n" \
>+  "lea        0x8(%[u_buf]),%[u_buf]                              \n" \
>+  "vpunpcklbw %%ymm1,%%ymm0,%%ymm0                                \n" \
>+  "vpermq     $0xd8,%%ymm0,%%ymm0                                 \n" \
>+  "vpunpcklwd %%ymm0,%%ymm0,%%ymm0                                \n" \
>+  "vmovdqu    (%[y_buf]),%%xmm4                                   \n" \
>+  "vpermq     $0xd8,%%ymm4,%%ymm4                                 \n" \
>+  "vpunpcklbw %%ymm4,%%ymm4,%%ymm4                                \n" \
>+  "lea        0x10(%[y_buf]),%[y_buf]                             \n"
>+
>+// Read 8 UV from 210 10 bit, upsample to 16 UV
>+// TODO(fbarchard): Consider vshufb to replace pack/unpack
>+// TODO(fbarchard): Consider vunpcklpd to combine the 2 registers into 1.
>+#define READYUV210_AVX2                                            \
>+  "vmovdqu    (%[u_buf]),%%xmm0                                \n" \
>+  "vmovdqu    0x00(%[u_buf],%[v_buf],1),%%xmm1                 \n" \
>+  "lea        0x10(%[u_buf]),%[u_buf]                          \n" \
>+  "vpermq     $0xd8,%%ymm0,%%ymm0                              \n" \
>+  "vpermq     $0xd8,%%ymm1,%%ymm1                              \n" \
>+  "vpunpcklwd %%ymm1,%%ymm0,%%ymm0                             \n" \
>+  "vpsraw     $0x2,%%ymm0,%%ymm0                               \n" \
>+  "vpackuswb  %%ymm0,%%ymm0,%%ymm0                             \n" \
>+  "vpunpcklwd %%ymm0,%%ymm0,%%ymm0                             \n" \
>+  "vmovdqu    (%[y_buf]),%%ymm4                                \n" \
>+  "vpsllw     $0x6,%%ymm4,%%ymm4                               \n" \
>+  "lea        0x20(%[y_buf]),%[y_buf]                          \n"
> 
> // Read 8 UV from 422, upsample to 16 UV.  With 16 Alpha.
>-#define READYUVA422_AVX2 \
>-  "vmovq      " MEMACCESS([u_buf]) ",%%xmm0                         \n"        \
>-    MEMOPREG(vmovq, 0x00, [u_buf], [v_buf], 1, xmm1)                           \
>-    "lea        " MEMLEA(0x8, [u_buf]) ",%[u_buf]                   \n"        \
>-    "vpunpcklbw %%ymm1,%%ymm0,%%ymm0                                \n"        \
>-    "vpermq     $0xd8,%%ymm0,%%ymm0                                 \n"        \
>-    "vpunpcklwd %%ymm0,%%ymm0,%%ymm0                                \n"        \
>-    "vmovdqu    " MEMACCESS([y_buf]) ",%%xmm4                       \n"        \
>-    "vpermq     $0xd8,%%ymm4,%%ymm4                                 \n"        \
>-    "vpunpcklbw %%ymm4,%%ymm4,%%ymm4                                \n"        \
>-    "lea        " MEMLEA(0x10, [y_buf]) ",%[y_buf]                  \n"        \
>-    "vmovdqu    " MEMACCESS([a_buf]) ",%%xmm5                       \n"        \
>-    "vpermq     $0xd8,%%ymm5,%%ymm5                                 \n"        \
>-    "lea        " MEMLEA(0x10, [a_buf]) ",%[a_buf]                  \n"
>+#define READYUVA422_AVX2                                              \
>+  "vmovq      (%[u_buf]),%%xmm0                                   \n" \
>+  "vmovq      0x00(%[u_buf],%[v_buf],1),%%xmm1                    \n" \
>+  "lea        0x8(%[u_buf]),%[u_buf]                              \n" \
>+  "vpunpcklbw %%ymm1,%%ymm0,%%ymm0                                \n" \
>+  "vpermq     $0xd8,%%ymm0,%%ymm0                                 \n" \
>+  "vpunpcklwd %%ymm0,%%ymm0,%%ymm0                                \n" \
>+  "vmovdqu    (%[y_buf]),%%xmm4                                   \n" \
>+  "vpermq     $0xd8,%%ymm4,%%ymm4                                 \n" \
>+  "vpunpcklbw %%ymm4,%%ymm4,%%ymm4                                \n" \
>+  "lea        0x10(%[y_buf]),%[y_buf]                             \n" \
>+  "vmovdqu    (%[a_buf]),%%xmm5                                   \n" \
>+  "vpermq     $0xd8,%%ymm5,%%ymm5                                 \n" \
>+  "lea        0x10(%[a_buf]),%[a_buf]                             \n"
> 
> // Read 8 UV from NV12, upsample to 16 UV.
>-#define READNV12_AVX2 \
>-  "vmovdqu    " MEMACCESS([uv_buf]) ",%%xmm0                        \n"        \
>-    "lea        " MEMLEA(0x10, [uv_buf]) ",%[uv_buf]                \n"        \
>-    "vpermq     $0xd8,%%ymm0,%%ymm0                                 \n"        \
>-    "vpunpcklwd %%ymm0,%%ymm0,%%ymm0                                \n"        \
>-    "vmovdqu    " MEMACCESS([y_buf]) ",%%xmm4                       \n"        \
>-    "vpermq     $0xd8,%%ymm4,%%ymm4                                 \n"        \
>-    "vpunpcklbw %%ymm4,%%ymm4,%%ymm4                                \n"        \
>-    "lea        " MEMLEA(0x10, [y_buf]) ",%[y_buf]                  \n"
>+#define READNV12_AVX2                                                 \
>+  "vmovdqu    (%[uv_buf]),%%xmm0                                  \n" \
>+  "lea        0x10(%[uv_buf]),%[uv_buf]                           \n" \
>+  "vpermq     $0xd8,%%ymm0,%%ymm0                                 \n" \
>+  "vpunpcklwd %%ymm0,%%ymm0,%%ymm0                                \n" \
>+  "vmovdqu    (%[y_buf]),%%xmm4                                   \n" \
>+  "vpermq     $0xd8,%%ymm4,%%ymm4                                 \n" \
>+  "vpunpcklbw %%ymm4,%%ymm4,%%ymm4                                \n" \
>+  "lea        0x10(%[y_buf]),%[y_buf]                             \n"
> 
> // Read 8 VU from NV21, upsample to 16 UV.
>-#define READNV21_AVX2 \
>-  "vmovdqu    " MEMACCESS([vu_buf]) ",%%xmm0                        \n"        \
>-    "lea        " MEMLEA(0x10, [vu_buf]) ",%[vu_buf]                \n"        \
>-    "vpermq     $0xd8,%%ymm0,%%ymm0                                 \n"        \
>-    "vpshufb     %[kShuffleNV21], %%ymm0, %%ymm0                    \n"        \
>-    "vmovdqu    " MEMACCESS([y_buf]) ",%%xmm4                       \n"        \
>-    "vpermq     $0xd8,%%ymm4,%%ymm4                                 \n"        \
>-    "vpunpcklbw %%ymm4,%%ymm4,%%ymm4                                \n"        \
>-    "lea        " MEMLEA(0x10, [y_buf]) ",%[y_buf]                  \n"
>+#define READNV21_AVX2                                                 \
>+  "vmovdqu    (%[vu_buf]),%%xmm0                                  \n" \
>+  "lea        0x10(%[vu_buf]),%[vu_buf]                           \n" \
>+  "vpermq     $0xd8,%%ymm0,%%ymm0                                 \n" \
>+  "vpshufb     %[kShuffleNV21], %%ymm0, %%ymm0                    \n" \
>+  "vmovdqu    (%[y_buf]),%%xmm4                                   \n" \
>+  "vpermq     $0xd8,%%ymm4,%%ymm4                                 \n" \
>+  "vpunpcklbw %%ymm4,%%ymm4,%%ymm4                                \n" \
>+  "lea        0x10(%[y_buf]),%[y_buf]                             \n"
> 
> // Read 8 YUY2 with 16 Y and upsample 8 UV to 16 UV.
>-#define READYUY2_AVX2 \
>-  "vmovdqu    " MEMACCESS([yuy2_buf]) ",%%ymm4                      \n"        \
>-    "vpshufb    %[kShuffleYUY2Y], %%ymm4, %%ymm4                    \n"        \
>-    "vmovdqu    " MEMACCESS([yuy2_buf]) ",%%ymm0                    \n"        \
>-    "vpshufb    %[kShuffleYUY2UV], %%ymm0, %%ymm0                   \n"        \
>-    "lea        " MEMLEA(0x20, [yuy2_buf]) ",%[yuy2_buf]            \n"
>+#define READYUY2_AVX2                                                 \
>+  "vmovdqu    (%[yuy2_buf]),%%ymm4                                \n" \
>+  "vpshufb    %[kShuffleYUY2Y], %%ymm4, %%ymm4                    \n" \
>+  "vmovdqu    (%[yuy2_buf]),%%ymm0                                \n" \
>+  "vpshufb    %[kShuffleYUY2UV], %%ymm0, %%ymm0                   \n" \
>+  "lea        0x20(%[yuy2_buf]),%[yuy2_buf]                       \n"
> 
> // Read 8 UYVY with 16 Y and upsample 8 UV to 16 UV.
>-#define READUYVY_AVX2 \
>-  "vmovdqu     " MEMACCESS([uyvy_buf]) ",%%ymm4                     \n"        \
>-    "vpshufb     %[kShuffleUYVYY], %%ymm4, %%ymm4                   \n"        \
>-    "vmovdqu     " MEMACCESS([uyvy_buf]) ",%%ymm0                   \n"        \
>-    "vpshufb     %[kShuffleUYVYUV], %%ymm0, %%ymm0                  \n"        \
>-    "lea        " MEMLEA(0x20, [uyvy_buf]) ",%[uyvy_buf]            \n"
>+#define READUYVY_AVX2                                                 \
>+  "vmovdqu    (%[uyvy_buf]),%%ymm4                                \n" \
>+  "vpshufb    %[kShuffleUYVYY], %%ymm4, %%ymm4                    \n" \
>+  "vmovdqu    (%[uyvy_buf]),%%ymm0                                \n" \
>+  "vpshufb    %[kShuffleUYVYUV], %%ymm0, %%ymm0                   \n" \
>+  "lea        0x20(%[uyvy_buf]),%[uyvy_buf]                       \n"
> 
> #if defined(__x86_64__)
>-#define YUVTORGB_SETUP_AVX2(yuvconstants) \
>-  "vmovdqa     " MEMACCESS([yuvconstants]) ",%%ymm8              \n"           \
>-    "vmovdqa     " MEMACCESS2(32, [yuvconstants]) ",%%ymm9       \n"           \
>-    "vmovdqa     " MEMACCESS2(64, [yuvconstants]) ",%%ymm10      \n"           \
>-    "vmovdqa     " MEMACCESS2(96, [yuvconstants]) ",%%ymm11      \n"           \
>-    "vmovdqa     " MEMACCESS2(128, [yuvconstants]) ",%%ymm12     \n"           \
>-    "vmovdqa     " MEMACCESS2(160, [yuvconstants]) ",%%ymm13     \n"           \
>-    "vmovdqa     " MEMACCESS2(192, [yuvconstants]) ",%%ymm14     \n"
>-
>-#define YUVTORGB_AVX2(yuvconstants)                                   \
>+#define YUVTORGB_SETUP_AVX2(yuvconstants)                            \
>+  "vmovdqa     (%[yuvconstants]),%%ymm8                          \n" \
>+  "vmovdqa     32(%[yuvconstants]),%%ymm9                        \n" \
>+  "vmovdqa     64(%[yuvconstants]),%%ymm10                       \n" \
>+  "vmovdqa     96(%[yuvconstants]),%%ymm11                       \n" \
>+  "vmovdqa     128(%[yuvconstants]),%%ymm12                      \n" \
>+  "vmovdqa     160(%[yuvconstants]),%%ymm13                      \n" \
>+  "vmovdqa     192(%[yuvconstants]),%%ymm14                      \n"
>+
>+#define YUVTORGB16_AVX2(yuvconstants)                                 \
>   "vpmaddubsw  %%ymm10,%%ymm0,%%ymm2                              \n" \
>   "vpmaddubsw  %%ymm9,%%ymm0,%%ymm1                               \n" \
>   "vpmaddubsw  %%ymm8,%%ymm0,%%ymm0                               \n" \
>@@ -2038,13 +2363,7 @@ void OMITFP I422ToRGBARow_SSSE3(const uint8* y_buf,
>   "vpmulhuw    %%ymm14,%%ymm4,%%ymm4                              \n" \
>   "vpaddsw     %%ymm4,%%ymm0,%%ymm0                               \n" \
>   "vpaddsw     %%ymm4,%%ymm1,%%ymm1                               \n" \
>-  "vpaddsw     %%ymm4,%%ymm2,%%ymm2                               \n" \
>-  "vpsraw      $0x6,%%ymm0,%%ymm0                                 \n" \
>-  "vpsraw      $0x6,%%ymm1,%%ymm1                                 \n" \
>-  "vpsraw      $0x6,%%ymm2,%%ymm2                                 \n" \
>-  "vpackuswb   %%ymm0,%%ymm0,%%ymm0                               \n" \
>-  "vpackuswb   %%ymm1,%%ymm1,%%ymm1                               \n" \
>-  "vpackuswb   %%ymm2,%%ymm2,%%ymm2                               \n"
>+  "vpaddsw     %%ymm4,%%ymm2,%%ymm2                               \n"
> 
> #define YUVTORGB_REGS_AVX2 \
>   "xmm8", "xmm9", "xmm10", "xmm11", "xmm12", "xmm13", "xmm14",
>@@ -2052,48 +2371,78 @@ void OMITFP I422ToRGBARow_SSSE3(const uint8* y_buf,
> #else  // Convert 16 pixels: 16 UV and 16 Y.
> 
> #define YUVTORGB_SETUP_AVX2(yuvconstants)
>-#define YUVTORGB_AVX2(yuvconstants) \
>-  "vpmaddubsw  " MEMACCESS2(64, [yuvconstants]) ",%%ymm0,%%ymm2     \n"        \
>-    "vpmaddubsw  " MEMACCESS2(32, [yuvconstants]) ",%%ymm0,%%ymm1   \n"        \
>-    "vpmaddubsw  " MEMACCESS([yuvconstants]) ",%%ymm0,%%ymm0        \n"        \
>-    "vmovdqu     " MEMACCESS2(160, [yuvconstants]) ",%%ymm3         \n"        \
>-    "vpsubw      %%ymm2,%%ymm3,%%ymm2                               \n"        \
>-    "vmovdqu     " MEMACCESS2(128, [yuvconstants]) ",%%ymm3         \n"        \
>-    "vpsubw      %%ymm1,%%ymm3,%%ymm1                               \n"        \
>-    "vmovdqu     " MEMACCESS2(96, [yuvconstants]) ",%%ymm3          \n"        \
>-    "vpsubw      %%ymm0,%%ymm3,%%ymm0                               \n"        \
>-    "vpmulhuw    " MEMACCESS2(192, [yuvconstants]) ",%%ymm4,%%ymm4  \n"        \
>-    "vpaddsw     %%ymm4,%%ymm0,%%ymm0                               \n"        \
>-    "vpaddsw     %%ymm4,%%ymm1,%%ymm1                               \n"        \
>-    "vpaddsw     %%ymm4,%%ymm2,%%ymm2                               \n"        \
>-    "vpsraw      $0x6,%%ymm0,%%ymm0                                 \n"        \
>-    "vpsraw      $0x6,%%ymm1,%%ymm1                                 \n"        \
>-    "vpsraw      $0x6,%%ymm2,%%ymm2                                 \n"        \
>-    "vpackuswb   %%ymm0,%%ymm0,%%ymm0                               \n"        \
>-    "vpackuswb   %%ymm1,%%ymm1,%%ymm1                               \n"        \
>-    "vpackuswb   %%ymm2,%%ymm2,%%ymm2                               \n"
>+#define YUVTORGB16_AVX2(yuvconstants)                                 \
>+  "vpmaddubsw  64(%[yuvconstants]),%%ymm0,%%ymm2                  \n" \
>+  "vpmaddubsw  32(%[yuvconstants]),%%ymm0,%%ymm1                  \n" \
>+  "vpmaddubsw  (%[yuvconstants]),%%ymm0,%%ymm0                    \n" \
>+  "vmovdqu     160(%[yuvconstants]),%%ymm3                        \n" \
>+  "vpsubw      %%ymm2,%%ymm3,%%ymm2                               \n" \
>+  "vmovdqu     128(%[yuvconstants]),%%ymm3                        \n" \
>+  "vpsubw      %%ymm1,%%ymm3,%%ymm1                               \n" \
>+  "vmovdqu     96(%[yuvconstants]),%%ymm3                         \n" \
>+  "vpsubw      %%ymm0,%%ymm3,%%ymm0                               \n" \
>+  "vpmulhuw    192(%[yuvconstants]),%%ymm4,%%ymm4                 \n" \
>+  "vpaddsw     %%ymm4,%%ymm0,%%ymm0                               \n" \
>+  "vpaddsw     %%ymm4,%%ymm1,%%ymm1                               \n" \
>+  "vpaddsw     %%ymm4,%%ymm2,%%ymm2                               \n"
> #define YUVTORGB_REGS_AVX2
> #endif
> 
>+#define YUVTORGB_AVX2(yuvconstants)                                   \
>+  YUVTORGB16_AVX2(yuvconstants)                                       \
>+  "vpsraw      $0x6,%%ymm0,%%ymm0                                 \n" \
>+  "vpsraw      $0x6,%%ymm1,%%ymm1                                 \n" \
>+  "vpsraw      $0x6,%%ymm2,%%ymm2                                 \n" \
>+  "vpackuswb   %%ymm0,%%ymm0,%%ymm0                               \n" \
>+  "vpackuswb   %%ymm1,%%ymm1,%%ymm1                               \n" \
>+  "vpackuswb   %%ymm2,%%ymm2,%%ymm2                               \n"
>+
> // Store 16 ARGB values.
>-#define STOREARGB_AVX2 \
>-  "vpunpcklbw %%ymm1,%%ymm0,%%ymm0                                  \n"        \
>-    "vpermq     $0xd8,%%ymm0,%%ymm0                                 \n"        \
>-    "vpunpcklbw %%ymm5,%%ymm2,%%ymm2                                \n"        \
>-    "vpermq     $0xd8,%%ymm2,%%ymm2                                 \n"        \
>-    "vpunpcklwd %%ymm2,%%ymm0,%%ymm1                                \n"        \
>-    "vpunpckhwd %%ymm2,%%ymm0,%%ymm0                                \n"        \
>-    "vmovdqu    %%ymm1," MEMACCESS([dst_argb]) "                    \n"        \
>-    "vmovdqu    %%ymm0," MEMACCESS2(0x20, [dst_argb]) "             \n"        \
>-    "lea       " MEMLEA(0x40, [dst_argb]) ", %[dst_argb]            \n"
>+#define STOREARGB_AVX2                                                \
>+  "vpunpcklbw %%ymm1,%%ymm0,%%ymm0                                \n" \
>+  "vpermq     $0xd8,%%ymm0,%%ymm0                                 \n" \
>+  "vpunpcklbw %%ymm5,%%ymm2,%%ymm2                                \n" \
>+  "vpermq     $0xd8,%%ymm2,%%ymm2                                 \n" \
>+  "vpunpcklwd %%ymm2,%%ymm0,%%ymm1                                \n" \
>+  "vpunpckhwd %%ymm2,%%ymm0,%%ymm0                                \n" \
>+  "vmovdqu    %%ymm1,(%[dst_argb])                                \n" \
>+  "vmovdqu    %%ymm0,0x20(%[dst_argb])                            \n" \
>+  "lea       0x40(%[dst_argb]), %[dst_argb]                       \n"
>+
>+// Store 16 AR30 values.
>+#define STOREAR30_AVX2                                                \
>+  "vpsraw     $0x4,%%ymm0,%%ymm0                                  \n" \
>+  "vpsraw     $0x4,%%ymm1,%%ymm1                                  \n" \
>+  "vpsraw     $0x4,%%ymm2,%%ymm2                                  \n" \
>+  "vpminsw    %%ymm7,%%ymm0,%%ymm0                                \n" \
>+  "vpminsw    %%ymm7,%%ymm1,%%ymm1                                \n" \
>+  "vpminsw    %%ymm7,%%ymm2,%%ymm2                                \n" \
>+  "vpmaxsw    %%ymm6,%%ymm0,%%ymm0                                \n" \
>+  "vpmaxsw    %%ymm6,%%ymm1,%%ymm1                                \n" \
>+  "vpmaxsw    %%ymm6,%%ymm2,%%ymm2                                \n" \
>+  "vpsllw     $0x4,%%ymm2,%%ymm2                                  \n" \
>+  "vpermq     $0xd8,%%ymm0,%%ymm0                                 \n" \
>+  "vpermq     $0xd8,%%ymm1,%%ymm1                                 \n" \
>+  "vpermq     $0xd8,%%ymm2,%%ymm2                                 \n" \
>+  "vpunpckhwd %%ymm2,%%ymm0,%%ymm3                                \n" \
>+  "vpunpcklwd %%ymm2,%%ymm0,%%ymm0                                \n" \
>+  "vpunpckhwd %%ymm5,%%ymm1,%%ymm2                                \n" \
>+  "vpunpcklwd %%ymm5,%%ymm1,%%ymm1                                \n" \
>+  "vpslld     $0xa,%%ymm1,%%ymm1                                  \n" \
>+  "vpslld     $0xa,%%ymm2,%%ymm2                                  \n" \
>+  "vpor       %%ymm1,%%ymm0,%%ymm0                                \n" \
>+  "vpor       %%ymm2,%%ymm3,%%ymm3                                \n" \
>+  "vmovdqu    %%ymm0,(%[dst_ar30])                                \n" \
>+  "vmovdqu    %%ymm3,0x20(%[dst_ar30])                            \n" \
>+  "lea        0x40(%[dst_ar30]), %[dst_ar30]                      \n"
> 
> #ifdef HAS_I444TOARGBROW_AVX2
> // 16 pixels
> // 16 UV values with 16 Y producing 16 ARGB (64 bytes).
>-void OMITFP I444ToARGBRow_AVX2(const uint8* y_buf,
>-                               const uint8* u_buf,
>-                               const uint8* v_buf,
>-                               uint8* dst_argb,
>+void OMITFP I444ToARGBRow_AVX2(const uint8_t* y_buf,
>+                               const uint8_t* u_buf,
>+                               const uint8_t* v_buf,
>+                               uint8_t* dst_argb,
>                                const struct YuvConstants* yuvconstants,
>                                int width) {
>   asm volatile (
>@@ -2115,7 +2464,7 @@ void OMITFP I444ToARGBRow_AVX2(const uint8* y_buf,
>     [dst_argb]"+r"(dst_argb),  // %[dst_argb]
>     [width]"+rm"(width)    // %[width]
>   : [yuvconstants]"r"(yuvconstants)  // %[yuvconstants]
>-  : "memory", "cc", NACL_R14 YUVTORGB_REGS_AVX2
>+  : "memory", "cc", YUVTORGB_REGS_AVX2
>     "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5"
>   );
> }
>@@ -2124,10 +2473,10 @@ void OMITFP I444ToARGBRow_AVX2(const uint8* y_buf,
> #if defined(HAS_I422TOARGBROW_AVX2)
> // 16 pixels
> // 8 UV values upsampled to 16 UV, mixed with 16 Y producing 16 ARGB (64 bytes).
>-void OMITFP I422ToARGBRow_AVX2(const uint8* y_buf,
>-                               const uint8* u_buf,
>-                               const uint8* v_buf,
>-                               uint8* dst_argb,
>+void OMITFP I422ToARGBRow_AVX2(const uint8_t* y_buf,
>+                               const uint8_t* u_buf,
>+                               const uint8_t* v_buf,
>+                               uint8_t* dst_argb,
>                                const struct YuvConstants* yuvconstants,
>                                int width) {
>   asm volatile (
>@@ -2150,84 +2499,74 @@ void OMITFP I422ToARGBRow_AVX2(const uint8* y_buf,
>     [dst_argb]"+r"(dst_argb),  // %[dst_argb]
>     [width]"+rm"(width)    // %[width]
>   : [yuvconstants]"r"(yuvconstants)  // %[yuvconstants]
>-  : "memory", "cc", NACL_R14 YUVTORGB_REGS_AVX2
>+  : "memory", "cc", YUVTORGB_REGS_AVX2
>     "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5"
>   );
> }
> #endif  // HAS_I422TOARGBROW_AVX2
> 
>-#if defined(HAS_I422ALPHATOARGBROW_AVX2)
>+#if defined(HAS_I422TOAR30ROW_AVX2)
> // 16 pixels
>-// 8 UV values upsampled to 16 UV, mixed with 16 Y and 16 A producing 16 ARGB.
>-void OMITFP I422AlphaToARGBRow_AVX2(const uint8* y_buf,
>-                                    const uint8* u_buf,
>-                                    const uint8* v_buf,
>-                                    const uint8* a_buf,
>-                                    uint8* dst_argb,
>-                                    const struct YuvConstants* yuvconstants,
>-                                    int width) {
>-  // clang-format off
>+// 8 UV values upsampled to 16 UV, mixed with 16 Y producing 16 AR30 (64 bytes).
>+void OMITFP I422ToAR30Row_AVX2(const uint8_t* y_buf,
>+                               const uint8_t* u_buf,
>+                               const uint8_t* v_buf,
>+                               uint8_t* dst_ar30,
>+                               const struct YuvConstants* yuvconstants,
>+                               int width) {
>   asm volatile (
>     YUVTORGB_SETUP_AVX2(yuvconstants)
>     "sub       %[u_buf],%[v_buf]               \n"
>+    "vpcmpeqb  %%ymm5,%%ymm5,%%ymm5            \n"  // AR30 constants
>+    "vpsrlw    $14,%%ymm5,%%ymm5               \n"
>+    "vpsllw    $4,%%ymm5,%%ymm5                \n"  // 2 alpha bits
>+    "vpxor     %%ymm6,%%ymm6,%%ymm6            \n"  // 0 for min
>+    "vpcmpeqb  %%ymm7,%%ymm7,%%ymm7            \n"  // 1023 for max
>+    "vpsrlw    $6,%%ymm7,%%ymm7                \n"
> 
>     LABELALIGN
>     "1:                                        \n"
>-    READYUVA422_AVX2
>-    YUVTORGB_AVX2(yuvconstants)
>-    STOREARGB_AVX2
>-    "subl      $0x10,%[width]                  \n"
>+    READYUV422_AVX2
>+    YUVTORGB16_AVX2(yuvconstants)
>+    STOREAR30_AVX2
>+    "sub       $0x10,%[width]                  \n"
>     "jg        1b                              \n"
>+
>     "vzeroupper                                \n"
>   : [y_buf]"+r"(y_buf),    // %[y_buf]
>     [u_buf]"+r"(u_buf),    // %[u_buf]
>     [v_buf]"+r"(v_buf),    // %[v_buf]
>-    [a_buf]"+r"(a_buf),    // %[a_buf]
>-    [dst_argb]"+r"(dst_argb),  // %[dst_argb]
>-#if defined(__i386__)
>-    [width]"+m"(width)     // %[width]
>-#else
>+    [dst_ar30]"+r"(dst_ar30),  // %[dst_ar30]
>     [width]"+rm"(width)    // %[width]
>-#endif
>   : [yuvconstants]"r"(yuvconstants)  // %[yuvconstants]
>-  : "memory", "cc", NACL_R14 YUVTORGB_REGS_AVX2
>-    "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5"
>+  : "memory", "cc", YUVTORGB_REGS_AVX2
>+    "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6", "xmm7"
>   );
>-  // clang-format on
> }
>-#endif  // HAS_I422ALPHATOARGBROW_AVX2
>+#endif  // HAS_I422TOAR30ROW_AVX2
> 
>-#if defined(HAS_I422TORGBAROW_AVX2)
>+#if defined(HAS_I210TOARGBROW_AVX2)
> // 16 pixels
>-// 8 UV values upsampled to 16 UV, mixed with 16 Y producing 16 RGBA (64 bytes).
>-void OMITFP I422ToRGBARow_AVX2(const uint8* y_buf,
>-                               const uint8* u_buf,
>-                               const uint8* v_buf,
>-                               uint8* dst_argb,
>+// 8 UV values upsampled to 16 UV, mixed with 16 Y producing 16 ARGB (64 bytes).
>+void OMITFP I210ToARGBRow_AVX2(const uint16_t* y_buf,
>+                               const uint16_t* u_buf,
>+                               const uint16_t* v_buf,
>+                               uint8_t* dst_argb,
>                                const struct YuvConstants* yuvconstants,
>                                int width) {
>   asm volatile (
>     YUVTORGB_SETUP_AVX2(yuvconstants)
>     "sub       %[u_buf],%[v_buf]               \n"
>-    "vpcmpeqb   %%ymm5,%%ymm5,%%ymm5           \n"
>+    "vpcmpeqb  %%ymm5,%%ymm5,%%ymm5            \n"
> 
>     LABELALIGN
>     "1:                                        \n"
>-    READYUV422_AVX2
>+    READYUV210_AVX2
>     YUVTORGB_AVX2(yuvconstants)
>-
>-    // Step 3: Weave into RGBA
>-    "vpunpcklbw %%ymm2,%%ymm1,%%ymm1           \n"
>-    "vpermq     $0xd8,%%ymm1,%%ymm1            \n"
>-    "vpunpcklbw %%ymm0,%%ymm5,%%ymm2           \n"
>-    "vpermq     $0xd8,%%ymm2,%%ymm2            \n"
>-    "vpunpcklwd %%ymm1,%%ymm2,%%ymm0           \n"
>-    "vpunpckhwd %%ymm1,%%ymm2,%%ymm1           \n"
>-    "vmovdqu    %%ymm0," MEMACCESS([dst_argb]) "\n"
>-    "vmovdqu    %%ymm1," MEMACCESS2(0x20,[dst_argb]) "\n"
>-    "lea       " MEMLEA(0x40,[dst_argb]) ",%[dst_argb] \n"
>+    STOREARGB_AVX2
>     "sub       $0x10,%[width]                  \n"
>     "jg        1b                              \n"
>+
>     "vzeroupper                                \n"
>   : [y_buf]"+r"(y_buf),    // %[y_buf]
>     [u_buf]"+r"(u_buf),    // %[u_buf]
>@@ -2235,18 +2574,143 @@ void OMITFP I422ToRGBARow_AVX2(const uint8* y_buf,
>     [dst_argb]"+r"(dst_argb),  // %[dst_argb]
>     [width]"+rm"(width)    // %[width]
>   : [yuvconstants]"r"(yuvconstants)  // %[yuvconstants]
>-  : "memory", "cc", NACL_R14 YUVTORGB_REGS_AVX2
>+  : "memory", "cc", YUVTORGB_REGS_AVX2
>     "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5"
>   );
> }
>-#endif  // HAS_I422TORGBAROW_AVX2
>+#endif  // HAS_I210TOARGBROW_AVX2
> 
>-#if defined(HAS_NV12TOARGBROW_AVX2)
>-// 16 pixels.
>+#if defined(HAS_I210TOAR30ROW_AVX2)
>+// 16 pixels
>+// 8 UV values upsampled to 16 UV, mixed with 16 Y producing 16 AR30 (64 bytes).
>+void OMITFP I210ToAR30Row_AVX2(const uint16_t* y_buf,
>+                               const uint16_t* u_buf,
>+                               const uint16_t* v_buf,
>+                               uint8_t* dst_ar30,
>+                               const struct YuvConstants* yuvconstants,
>+                               int width) {
>+  asm volatile (
>+    YUVTORGB_SETUP_AVX2(yuvconstants)
>+    "sub       %[u_buf],%[v_buf]               \n"
>+    "vpcmpeqb  %%ymm5,%%ymm5,%%ymm5            \n"  // AR30 constants
>+    "vpsrlw    $14,%%ymm5,%%ymm5               \n"
>+    "vpsllw    $4,%%ymm5,%%ymm5                \n"  // 2 alpha bits
>+    "vpxor     %%ymm6,%%ymm6,%%ymm6            \n"  // 0 for min
>+    "vpcmpeqb  %%ymm7,%%ymm7,%%ymm7            \n"  // 1023 for max
>+    "vpsrlw    $6,%%ymm7,%%ymm7                \n"
>+
>+    LABELALIGN
>+    "1:                                        \n"
>+    READYUV210_AVX2
>+    YUVTORGB16_AVX2(yuvconstants)
>+    STOREAR30_AVX2
>+    "sub       $0x10,%[width]                  \n"
>+    "jg        1b                              \n"
>+
>+    "vzeroupper                                \n"
>+  : [y_buf]"+r"(y_buf),    // %[y_buf]
>+    [u_buf]"+r"(u_buf),    // %[u_buf]
>+    [v_buf]"+r"(v_buf),    // %[v_buf]
>+    [dst_ar30]"+r"(dst_ar30),  // %[dst_ar30]
>+    [width]"+rm"(width)    // %[width]
>+  : [yuvconstants]"r"(yuvconstants)  // %[yuvconstants]
>+  : "memory", "cc", YUVTORGB_REGS_AVX2
>+    "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5"
>+  );
>+}
>+#endif  // HAS_I210TOAR30ROW_AVX2
>+
>+#if defined(HAS_I422ALPHATOARGBROW_AVX2)
>+// 16 pixels
>+// 8 UV values upsampled to 16 UV, mixed with 16 Y and 16 A producing 16 ARGB.
>+void OMITFP I422AlphaToARGBRow_AVX2(const uint8_t* y_buf,
>+                                    const uint8_t* u_buf,
>+                                    const uint8_t* v_buf,
>+                                    const uint8_t* a_buf,
>+                                    uint8_t* dst_argb,
>+                                    const struct YuvConstants* yuvconstants,
>+                                    int width) {
>+  // clang-format off
>+  asm volatile (
>+    YUVTORGB_SETUP_AVX2(yuvconstants)
>+    "sub       %[u_buf],%[v_buf]               \n"
>+
>+    LABELALIGN
>+    "1:                                        \n"
>+    READYUVA422_AVX2
>+    YUVTORGB_AVX2(yuvconstants)
>+    STOREARGB_AVX2
>+    "subl      $0x10,%[width]                  \n"
>+    "jg        1b                              \n"
>+    "vzeroupper                                \n"
>+  : [y_buf]"+r"(y_buf),    // %[y_buf]
>+    [u_buf]"+r"(u_buf),    // %[u_buf]
>+    [v_buf]"+r"(v_buf),    // %[v_buf]
>+    [a_buf]"+r"(a_buf),    // %[a_buf]
>+    [dst_argb]"+r"(dst_argb),  // %[dst_argb]
>+#if defined(__i386__)
>+    [width]"+m"(width)     // %[width]
>+#else
>+    [width]"+rm"(width)    // %[width]
>+#endif
>+  : [yuvconstants]"r"(yuvconstants)  // %[yuvconstants]
>+  : "memory", "cc", YUVTORGB_REGS_AVX2
>+    "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5"
>+  );
>+  // clang-format on
>+}
>+#endif  // HAS_I422ALPHATOARGBROW_AVX2
>+
>+#if defined(HAS_I422TORGBAROW_AVX2)
>+// 16 pixels
>+// 8 UV values upsampled to 16 UV, mixed with 16 Y producing 16 RGBA (64 bytes).
>+void OMITFP I422ToRGBARow_AVX2(const uint8_t* y_buf,
>+                               const uint8_t* u_buf,
>+                               const uint8_t* v_buf,
>+                               uint8_t* dst_argb,
>+                               const struct YuvConstants* yuvconstants,
>+                               int width) {
>+  asm volatile (
>+    YUVTORGB_SETUP_AVX2(yuvconstants)
>+    "sub       %[u_buf],%[v_buf]               \n"
>+    "vpcmpeqb   %%ymm5,%%ymm5,%%ymm5           \n"
>+
>+    LABELALIGN
>+    "1:                                        \n"
>+    READYUV422_AVX2
>+    YUVTORGB_AVX2(yuvconstants)
>+
>+    // Step 3: Weave into RGBA
>+    "vpunpcklbw %%ymm2,%%ymm1,%%ymm1           \n"
>+    "vpermq     $0xd8,%%ymm1,%%ymm1            \n"
>+    "vpunpcklbw %%ymm0,%%ymm5,%%ymm2           \n"
>+    "vpermq     $0xd8,%%ymm2,%%ymm2            \n"
>+    "vpunpcklwd %%ymm1,%%ymm2,%%ymm0           \n"
>+    "vpunpckhwd %%ymm1,%%ymm2,%%ymm1           \n"
>+    "vmovdqu    %%ymm0,(%[dst_argb])           \n"
>+    "vmovdqu    %%ymm1,0x20(%[dst_argb])       \n"
>+    "lea        0x40(%[dst_argb]),%[dst_argb]  \n"
>+    "sub        $0x10,%[width]                 \n"
>+    "jg         1b                             \n"
>+    "vzeroupper                                \n"
>+  : [y_buf]"+r"(y_buf),    // %[y_buf]
>+    [u_buf]"+r"(u_buf),    // %[u_buf]
>+    [v_buf]"+r"(v_buf),    // %[v_buf]
>+    [dst_argb]"+r"(dst_argb),  // %[dst_argb]
>+    [width]"+rm"(width)    // %[width]
>+  : [yuvconstants]"r"(yuvconstants)  // %[yuvconstants]
>+  : "memory", "cc", YUVTORGB_REGS_AVX2
>+    "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5"
>+  );
>+}
>+#endif  // HAS_I422TORGBAROW_AVX2
>+
>+#if defined(HAS_NV12TOARGBROW_AVX2)
>+// 16 pixels.
> // 8 UV values upsampled to 16 UV, mixed with 16 Y producing 16 ARGB (64 bytes).
>-void OMITFP NV12ToARGBRow_AVX2(const uint8* y_buf,
>-                               const uint8* uv_buf,
>-                               uint8* dst_argb,
>+void OMITFP NV12ToARGBRow_AVX2(const uint8_t* y_buf,
>+                               const uint8_t* uv_buf,
>+                               uint8_t* dst_argb,
>                                const struct YuvConstants* yuvconstants,
>                                int width) {
>   // clang-format off
>@@ -2267,7 +2731,7 @@ void OMITFP NV12ToARGBRow_AVX2(const uint8* y_buf,
>     [dst_argb]"+r"(dst_argb),  // %[dst_argb]
>     [width]"+rm"(width)    // %[width]
>   : [yuvconstants]"r"(yuvconstants)  // %[yuvconstants]
>-    : "memory", "cc", YUVTORGB_REGS_AVX2  // Does not use r14.
>+    : "memory", "cc", YUVTORGB_REGS_AVX2
>     "xmm0", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5"
>   );
>   // clang-format on
>@@ -2277,9 +2741,9 @@ void OMITFP NV12ToARGBRow_AVX2(const uint8* y_buf,
> #if defined(HAS_NV21TOARGBROW_AVX2)
> // 16 pixels.
> // 8 VU values upsampled to 16 UV, mixed with 16 Y producing 16 ARGB (64 bytes).
>-void OMITFP NV21ToARGBRow_AVX2(const uint8* y_buf,
>-                               const uint8* vu_buf,
>-                               uint8* dst_argb,
>+void OMITFP NV21ToARGBRow_AVX2(const uint8_t* y_buf,
>+                               const uint8_t* vu_buf,
>+                               uint8_t* dst_argb,
>                                const struct YuvConstants* yuvconstants,
>                                int width) {
>   // clang-format off
>@@ -2301,7 +2765,7 @@ void OMITFP NV21ToARGBRow_AVX2(const uint8* y_buf,
>     [width]"+rm"(width)    // %[width]
>   : [yuvconstants]"r"(yuvconstants), // %[yuvconstants]
>     [kShuffleNV21]"m"(kShuffleNV21)
>-    : "memory", "cc", YUVTORGB_REGS_AVX2  // Does not use r14.
>+    : "memory", "cc", YUVTORGB_REGS_AVX2
>       "xmm0", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5"
>   );
>   // clang-format on
>@@ -2311,8 +2775,8 @@ void OMITFP NV21ToARGBRow_AVX2(const uint8* y_buf,
> #if defined(HAS_YUY2TOARGBROW_AVX2)
> // 16 pixels.
> // 8 YUY2 values with 16 Y and 8 UV producing 16 ARGB (64 bytes).
>-void OMITFP YUY2ToARGBRow_AVX2(const uint8* yuy2_buf,
>-                               uint8* dst_argb,
>+void OMITFP YUY2ToARGBRow_AVX2(const uint8_t* yuy2_buf,
>+                               uint8_t* dst_argb,
>                                const struct YuvConstants* yuvconstants,
>                                int width) {
>   // clang-format off
>@@ -2334,7 +2798,7 @@ void OMITFP YUY2ToARGBRow_AVX2(const uint8* yuy2_buf,
>   : [yuvconstants]"r"(yuvconstants), // %[yuvconstants]
>     [kShuffleYUY2Y]"m"(kShuffleYUY2Y),
>     [kShuffleYUY2UV]"m"(kShuffleYUY2UV)
>-    : "memory", "cc", YUVTORGB_REGS_AVX2  // Does not use r14.
>+    : "memory", "cc", YUVTORGB_REGS_AVX2
>       "xmm0", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5"
>   );
>   // clang-format on
>@@ -2344,8 +2808,8 @@ void OMITFP YUY2ToARGBRow_AVX2(const uint8* yuy2_buf,
> #if defined(HAS_UYVYTOARGBROW_AVX2)
> // 16 pixels.
> // 8 UYVY values with 16 Y and 8 UV producing 16 ARGB (64 bytes).
>-void OMITFP UYVYToARGBRow_AVX2(const uint8* uyvy_buf,
>-                               uint8* dst_argb,
>+void OMITFP UYVYToARGBRow_AVX2(const uint8_t* uyvy_buf,
>+                               uint8_t* dst_argb,
>                                const struct YuvConstants* yuvconstants,
>                                int width) {
>   // clang-format off
>@@ -2367,7 +2831,7 @@ void OMITFP UYVYToARGBRow_AVX2(const uint8* uyvy_buf,
>   : [yuvconstants]"r"(yuvconstants), // %[yuvconstants]
>     [kShuffleUYVYY]"m"(kShuffleUYVYY),
>     [kShuffleUYVYUV]"m"(kShuffleUYVYUV)
>-    : "memory", "cc", YUVTORGB_REGS_AVX2  // Does not use r14.
>+    : "memory", "cc", YUVTORGB_REGS_AVX2
>       "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5"
>   );
>   // clang-format on
>@@ -2375,381 +2839,367 @@ void OMITFP UYVYToARGBRow_AVX2(const uint8* uyvy_buf,
> #endif  // HAS_UYVYTOARGBROW_AVX2
> 
> #ifdef HAS_I400TOARGBROW_SSE2
>-void I400ToARGBRow_SSE2(const uint8* y_buf, uint8* dst_argb, int width) {
>-  asm volatile (
>-    "mov       $0x4a354a35,%%eax               \n"  // 4a35 = 18997 = 1.164
>-    "movd      %%eax,%%xmm2                    \n"
>-    "pshufd    $0x0,%%xmm2,%%xmm2              \n"
>-    "mov       $0x04880488,%%eax               \n"  // 0488 = 1160 = 1.164 * 16
>-    "movd      %%eax,%%xmm3                    \n"
>-    "pshufd    $0x0,%%xmm3,%%xmm3              \n"
>-    "pcmpeqb   %%xmm4,%%xmm4                   \n"
>-    "pslld     $0x18,%%xmm4                    \n"
>+void I400ToARGBRow_SSE2(const uint8_t* y_buf, uint8_t* dst_argb, int width) {
>+  asm volatile(
>+      "mov       $0x4a354a35,%%eax               \n"  // 4a35 = 18997 = 1.164
>+      "movd      %%eax,%%xmm2                    \n"
>+      "pshufd    $0x0,%%xmm2,%%xmm2              \n"
>+      "mov       $0x04880488,%%eax               \n"  // 0488 = 1160 = 1.164 *
>+                                                      // 16
>+      "movd      %%eax,%%xmm3                    \n"
>+      "pshufd    $0x0,%%xmm3,%%xmm3              \n"
>+      "pcmpeqb   %%xmm4,%%xmm4                   \n"
>+      "pslld     $0x18,%%xmm4                    \n"
> 
>-    LABELALIGN
>-    "1:                                        \n"
>-    // Step 1: Scale Y contribution to 8 G values. G = (y - 16) * 1.164
>-    "movq      " MEMACCESS(0) ",%%xmm0         \n"
>-    "lea       " MEMLEA(0x8,0) ",%0            \n"
>-    "punpcklbw %%xmm0,%%xmm0                   \n"
>-    "pmulhuw   %%xmm2,%%xmm0                   \n"
>-    "psubusw   %%xmm3,%%xmm0                   \n"
>-    "psrlw     $6, %%xmm0                      \n"
>-    "packuswb  %%xmm0,%%xmm0                   \n"
>-
>-    // Step 2: Weave into ARGB
>-    "punpcklbw %%xmm0,%%xmm0                   \n"
>-    "movdqa    %%xmm0,%%xmm1                   \n"
>-    "punpcklwd %%xmm0,%%xmm0                   \n"
>-    "punpckhwd %%xmm1,%%xmm1                   \n"
>-    "por       %%xmm4,%%xmm0                   \n"
>-    "por       %%xmm4,%%xmm1                   \n"
>-    "movdqu    %%xmm0," MEMACCESS(1) "         \n"
>-    "movdqu    %%xmm1," MEMACCESS2(0x10,1) "   \n"
>-    "lea       " MEMLEA(0x20,1) ",%1           \n"
>-
>-    "sub       $0x8,%2                         \n"
>-    "jg        1b                              \n"
>-  : "+r"(y_buf),     // %0
>-    "+r"(dst_argb),  // %1
>-    "+rm"(width)     // %2
>-  :
>-  : "memory", "cc", "eax"
>-    , "xmm0", "xmm1", "xmm2", "xmm3", "xmm4"
>-  );
>+      LABELALIGN
>+      "1:                                        \n"
>+      // Step 1: Scale Y contribution to 8 G values. G = (y - 16) * 1.164
>+      "movq      (%0),%%xmm0                     \n"
>+      "lea       0x8(%0),%0                      \n"
>+      "punpcklbw %%xmm0,%%xmm0                   \n"
>+      "pmulhuw   %%xmm2,%%xmm0                   \n"
>+      "psubusw   %%xmm3,%%xmm0                   \n"
>+      "psrlw     $6, %%xmm0                      \n"
>+      "packuswb  %%xmm0,%%xmm0                   \n"
>+
>+      // Step 2: Weave into ARGB
>+      "punpcklbw %%xmm0,%%xmm0                   \n"
>+      "movdqa    %%xmm0,%%xmm1                   \n"
>+      "punpcklwd %%xmm0,%%xmm0                   \n"
>+      "punpckhwd %%xmm1,%%xmm1                   \n"
>+      "por       %%xmm4,%%xmm0                   \n"
>+      "por       %%xmm4,%%xmm1                   \n"
>+      "movdqu    %%xmm0,(%1)                     \n"
>+      "movdqu    %%xmm1,0x10(%1)                 \n"
>+      "lea       0x20(%1),%1                     \n"
>+
>+      "sub       $0x8,%2                         \n"
>+      "jg        1b                              \n"
>+      : "+r"(y_buf),     // %0
>+        "+r"(dst_argb),  // %1
>+        "+rm"(width)     // %2
>+      :
>+      : "memory", "cc", "eax", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4");
> }
> #endif  // HAS_I400TOARGBROW_SSE2
> 
> #ifdef HAS_I400TOARGBROW_AVX2
> // 16 pixels of Y converted to 16 pixels of ARGB (64 bytes).
> // note: vpunpcklbw mutates and vpackuswb unmutates.
>-void I400ToARGBRow_AVX2(const uint8* y_buf, uint8* dst_argb, int width) {
>-  asm volatile (
>-    "mov        $0x4a354a35,%%eax              \n" // 0488 = 1160 = 1.164 * 16
>-    "vmovd      %%eax,%%xmm2                   \n"
>-    "vbroadcastss %%xmm2,%%ymm2                \n"
>-    "mov        $0x4880488,%%eax               \n" // 4a35 = 18997 = 1.164
>-    "vmovd      %%eax,%%xmm3                   \n"
>-    "vbroadcastss %%xmm3,%%ymm3                \n"
>-    "vpcmpeqb   %%ymm4,%%ymm4,%%ymm4           \n"
>-    "vpslld     $0x18,%%ymm4,%%ymm4            \n"
>+void I400ToARGBRow_AVX2(const uint8_t* y_buf, uint8_t* dst_argb, int width) {
>+  asm volatile(
>+      "mov        $0x4a354a35,%%eax              \n"  // 0488 = 1160 = 1.164 *
>+                                                      // 16
>+      "vmovd      %%eax,%%xmm2                   \n"
>+      "vbroadcastss %%xmm2,%%ymm2                \n"
>+      "mov        $0x4880488,%%eax               \n"  // 4a35 = 18997 = 1.164
>+      "vmovd      %%eax,%%xmm3                   \n"
>+      "vbroadcastss %%xmm3,%%ymm3                \n"
>+      "vpcmpeqb   %%ymm4,%%ymm4,%%ymm4           \n"
>+      "vpslld     $0x18,%%ymm4,%%ymm4            \n"
> 
>-    LABELALIGN
>-    "1:                                        \n"
>-    // Step 1: Scale Y contribution to 16 G values. G = (y - 16) * 1.164
>-    "vmovdqu    " MEMACCESS(0) ",%%xmm0        \n"
>-    "lea        " MEMLEA(0x10,0) ",%0          \n"
>-    "vpermq     $0xd8,%%ymm0,%%ymm0            \n"
>-    "vpunpcklbw %%ymm0,%%ymm0,%%ymm0           \n"
>-    "vpmulhuw   %%ymm2,%%ymm0,%%ymm0           \n"
>-    "vpsubusw   %%ymm3,%%ymm0,%%ymm0           \n"
>-    "vpsrlw     $0x6,%%ymm0,%%ymm0             \n"
>-    "vpackuswb  %%ymm0,%%ymm0,%%ymm0           \n"
>-    "vpunpcklbw %%ymm0,%%ymm0,%%ymm1           \n"
>-    "vpermq     $0xd8,%%ymm1,%%ymm1            \n"
>-    "vpunpcklwd %%ymm1,%%ymm1,%%ymm0           \n"
>-    "vpunpckhwd %%ymm1,%%ymm1,%%ymm1           \n"
>-    "vpor       %%ymm4,%%ymm0,%%ymm0           \n"
>-    "vpor       %%ymm4,%%ymm1,%%ymm1           \n"
>-    "vmovdqu    %%ymm0," MEMACCESS(1) "        \n"
>-    "vmovdqu    %%ymm1," MEMACCESS2(0x20,1) "  \n"
>-    "lea       " MEMLEA(0x40,1) ",%1           \n"
>-    "sub        $0x10,%2                       \n"
>-    "jg        1b                              \n"
>-    "vzeroupper                                \n"
>-  : "+r"(y_buf),     // %0
>-    "+r"(dst_argb),  // %1
>-    "+rm"(width)     // %2
>-  :
>-  : "memory", "cc", "eax"
>-    , "xmm0", "xmm1", "xmm2", "xmm3", "xmm4"
>-  );
>+      LABELALIGN
>+      "1:                                        \n"
>+      // Step 1: Scale Y contribution to 16 G values. G = (y - 16) * 1.164
>+      "vmovdqu    (%0),%%xmm0                    \n"
>+      "lea        0x10(%0),%0                    \n"
>+      "vpermq     $0xd8,%%ymm0,%%ymm0            \n"
>+      "vpunpcklbw %%ymm0,%%ymm0,%%ymm0           \n"
>+      "vpmulhuw   %%ymm2,%%ymm0,%%ymm0           \n"
>+      "vpsubusw   %%ymm3,%%ymm0,%%ymm0           \n"
>+      "vpsrlw     $0x6,%%ymm0,%%ymm0             \n"
>+      "vpackuswb  %%ymm0,%%ymm0,%%ymm0           \n"
>+      "vpunpcklbw %%ymm0,%%ymm0,%%ymm1           \n"
>+      "vpermq     $0xd8,%%ymm1,%%ymm1            \n"
>+      "vpunpcklwd %%ymm1,%%ymm1,%%ymm0           \n"
>+      "vpunpckhwd %%ymm1,%%ymm1,%%ymm1           \n"
>+      "vpor       %%ymm4,%%ymm0,%%ymm0           \n"
>+      "vpor       %%ymm4,%%ymm1,%%ymm1           \n"
>+      "vmovdqu    %%ymm0,(%1)                    \n"
>+      "vmovdqu    %%ymm1,0x20(%1)                \n"
>+      "lea       0x40(%1),%1                     \n"
>+      "sub        $0x10,%2                       \n"
>+      "jg        1b                              \n"
>+      "vzeroupper                                \n"
>+      : "+r"(y_buf),     // %0
>+        "+r"(dst_argb),  // %1
>+        "+rm"(width)     // %2
>+      :
>+      : "memory", "cc", "eax", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4");
> }
> #endif  // HAS_I400TOARGBROW_AVX2
> 
> #ifdef HAS_MIRRORROW_SSSE3
> // Shuffle table for reversing the bytes.
>-static uvec8 kShuffleMirror = {15u, 14u, 13u, 12u, 11u, 10u, 9u, 8u,
>-                               7u,  6u,  5u,  4u,  3u,  2u,  1u, 0u};
>+static const uvec8 kShuffleMirror = {15u, 14u, 13u, 12u, 11u, 10u, 9u, 8u,
>+                                     7u,  6u,  5u,  4u,  3u,  2u,  1u, 0u};
> 
>-void MirrorRow_SSSE3(const uint8* src, uint8* dst, int width) {
>+void MirrorRow_SSSE3(const uint8_t* src, uint8_t* dst, int width) {
>   intptr_t temp_width = (intptr_t)(width);
>-  asm volatile (
>-    "movdqa    %3,%%xmm5                       \n"
>+  asm volatile(
> 
>-    LABELALIGN
>-    "1:                                        \n"
>-    MEMOPREG(movdqu,-0x10,0,2,1,xmm0)          //  movdqu -0x10(%0,%2),%%xmm0
>-    "pshufb    %%xmm5,%%xmm0                   \n"
>-    "movdqu    %%xmm0," MEMACCESS(1) "         \n"
>-    "lea       " MEMLEA(0x10,1) ",%1           \n"
>-    "sub       $0x10,%2                        \n"
>-    "jg        1b                              \n"
>-  : "+r"(src),  // %0
>-    "+r"(dst),  // %1
>-    "+r"(temp_width)  // %2
>-  : "m"(kShuffleMirror) // %3
>-  : "memory", "cc", NACL_R14
>-    "xmm0", "xmm5"
>-  );
>+      "movdqa    %3,%%xmm5                       \n"
>+
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    -0x10(%0,%2,1),%%xmm0           \n"
>+      "pshufb    %%xmm5,%%xmm0                   \n"
>+      "movdqu    %%xmm0,(%1)                     \n"
>+      "lea       0x10(%1),%1                     \n"
>+      "sub       $0x10,%2                        \n"
>+      "jg        1b                              \n"
>+      : "+r"(src),           // %0
>+        "+r"(dst),           // %1
>+        "+r"(temp_width)     // %2
>+      : "m"(kShuffleMirror)  // %3
>+      : "memory", "cc", "xmm0", "xmm5");
> }
> #endif  // HAS_MIRRORROW_SSSE3
> 
> #ifdef HAS_MIRRORROW_AVX2
>-void MirrorRow_AVX2(const uint8* src, uint8* dst, int width) {
>+void MirrorRow_AVX2(const uint8_t* src, uint8_t* dst, int width) {
>   intptr_t temp_width = (intptr_t)(width);
>-  asm volatile (
>-    "vbroadcastf128 %3,%%ymm5                  \n"
>+  asm volatile(
> 
>-    LABELALIGN
>-    "1:                                        \n"
>-    MEMOPREG(vmovdqu,-0x20,0,2,1,ymm0)         //  vmovdqu -0x20(%0,%2),%%ymm0
>-    "vpshufb    %%ymm5,%%ymm0,%%ymm0           \n"
>-    "vpermq     $0x4e,%%ymm0,%%ymm0            \n"
>-    "vmovdqu    %%ymm0," MEMACCESS(1) "        \n"
>-    "lea       " MEMLEA(0x20,1) ",%1           \n"
>-    "sub       $0x20,%2                        \n"
>-    "jg        1b                              \n"
>-    "vzeroupper                                \n"
>-  : "+r"(src),  // %0
>-    "+r"(dst),  // %1
>-    "+r"(temp_width)  // %2
>-  : "m"(kShuffleMirror) // %3
>-  : "memory", "cc", NACL_R14
>-    "xmm0", "xmm5"
>-  );
>+      "vbroadcastf128 %3,%%ymm5                  \n"
>+
>+      LABELALIGN
>+      "1:                                        \n"
>+      "vmovdqu    -0x20(%0,%2,1),%%ymm0          \n"
>+      "vpshufb    %%ymm5,%%ymm0,%%ymm0           \n"
>+      "vpermq     $0x4e,%%ymm0,%%ymm0            \n"
>+      "vmovdqu    %%ymm0,(%1)                    \n"
>+      "lea       0x20(%1),%1                     \n"
>+      "sub       $0x20,%2                        \n"
>+      "jg        1b                              \n"
>+      "vzeroupper                                \n"
>+      : "+r"(src),           // %0
>+        "+r"(dst),           // %1
>+        "+r"(temp_width)     // %2
>+      : "m"(kShuffleMirror)  // %3
>+      : "memory", "cc", "xmm0", "xmm5");
> }
> #endif  // HAS_MIRRORROW_AVX2
> 
> #ifdef HAS_MIRRORUVROW_SSSE3
> // Shuffle table for reversing the bytes of UV channels.
>-static uvec8 kShuffleMirrorUV = {14u, 12u, 10u, 8u, 6u, 4u, 2u, 0u,
>-                                 15u, 13u, 11u, 9u, 7u, 5u, 3u, 1u};
>-void MirrorUVRow_SSSE3(const uint8* src,
>-                       uint8* dst_u,
>-                       uint8* dst_v,
>+static const uvec8 kShuffleMirrorUV = {14u, 12u, 10u, 8u, 6u, 4u, 2u, 0u,
>+                                       15u, 13u, 11u, 9u, 7u, 5u, 3u, 1u};
>+void MirrorUVRow_SSSE3(const uint8_t* src,
>+                       uint8_t* dst_u,
>+                       uint8_t* dst_v,
>                        int width) {
>   intptr_t temp_width = (intptr_t)(width);
>-  asm volatile (
>-    "movdqa    %4,%%xmm1                       \n"
>-    "lea       " MEMLEA4(-0x10,0,3,2) ",%0     \n"
>-    "sub       %1,%2                           \n"
>+  asm volatile(
>+      "movdqa    %4,%%xmm1                       \n"
>+      "lea       -0x10(%0,%3,2),%0               \n"
>+      "sub       %1,%2                           \n"
> 
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    "lea       " MEMLEA(-0x10,0) ",%0          \n"
>-    "pshufb    %%xmm1,%%xmm0                   \n"
>-    "movlpd    %%xmm0," MEMACCESS(1) "         \n"
>-    MEMOPMEM(movhpd,xmm0,0x00,1,2,1)           //  movhpd    %%xmm0,(%1,%2)
>-    "lea       " MEMLEA(0x8,1) ",%1            \n"
>-    "sub       $8,%3                           \n"
>-    "jg        1b                              \n"
>-  : "+r"(src),      // %0
>-    "+r"(dst_u),    // %1
>-    "+r"(dst_v),    // %2
>-    "+r"(temp_width)  // %3
>-  : "m"(kShuffleMirrorUV)  // %4
>-  : "memory", "cc", NACL_R14
>-    "xmm0", "xmm1"
>-  );
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "lea       -0x10(%0),%0                    \n"
>+      "pshufb    %%xmm1,%%xmm0                   \n"
>+      "movlpd    %%xmm0,(%1)                     \n"
>+      "movhpd    %%xmm0,0x00(%1,%2,1)            \n"
>+      "lea       0x8(%1),%1                      \n"
>+      "sub       $8,%3                           \n"
>+      "jg        1b                              \n"
>+      : "+r"(src),             // %0
>+        "+r"(dst_u),           // %1
>+        "+r"(dst_v),           // %2
>+        "+r"(temp_width)       // %3
>+      : "m"(kShuffleMirrorUV)  // %4
>+      : "memory", "cc", "xmm0", "xmm1");
> }
> #endif  // HAS_MIRRORUVROW_SSSE3
> 
> #ifdef HAS_ARGBMIRRORROW_SSE2
> 
>-void ARGBMirrorRow_SSE2(const uint8* src, uint8* dst, int width) {
>+void ARGBMirrorRow_SSE2(const uint8_t* src, uint8_t* dst, int width) {
>   intptr_t temp_width = (intptr_t)(width);
>-  asm volatile (
>-    "lea       " MEMLEA4(-0x10,0,2,4) ",%0     \n"
>+  asm volatile(
> 
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    "pshufd    $0x1b,%%xmm0,%%xmm0             \n"
>-    "lea       " MEMLEA(-0x10,0) ",%0          \n"
>-    "movdqu    %%xmm0," MEMACCESS(1) "         \n"
>-    "lea       " MEMLEA(0x10,1) ",%1           \n"
>-    "sub       $0x4,%2                         \n"
>-    "jg        1b                              \n"
>-  : "+r"(src),  // %0
>-    "+r"(dst),  // %1
>-    "+r"(temp_width)  // %2
>-  :
>-  : "memory", "cc"
>-    , "xmm0"
>-  );
>+      "lea       -0x10(%0,%2,4),%0               \n"
>+
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "pshufd    $0x1b,%%xmm0,%%xmm0             \n"
>+      "lea       -0x10(%0),%0                    \n"
>+      "movdqu    %%xmm0,(%1)                     \n"
>+      "lea       0x10(%1),%1                     \n"
>+      "sub       $0x4,%2                         \n"
>+      "jg        1b                              \n"
>+      : "+r"(src),        // %0
>+        "+r"(dst),        // %1
>+        "+r"(temp_width)  // %2
>+      :
>+      : "memory", "cc", "xmm0");
> }
> #endif  // HAS_ARGBMIRRORROW_SSE2
> 
> #ifdef HAS_ARGBMIRRORROW_AVX2
> // Shuffle table for reversing the bytes.
> static const ulvec32 kARGBShuffleMirror_AVX2 = {7u, 6u, 5u, 4u, 3u, 2u, 1u, 0u};
>-void ARGBMirrorRow_AVX2(const uint8* src, uint8* dst, int width) {
>+void ARGBMirrorRow_AVX2(const uint8_t* src, uint8_t* dst, int width) {
>   intptr_t temp_width = (intptr_t)(width);
>-  asm volatile (
>-    "vmovdqu    %3,%%ymm5                      \n"
>+  asm volatile(
> 
>-    LABELALIGN
>-    "1:                                        \n"
>-    VMEMOPREG(vpermd,-0x20,0,2,4,ymm5,ymm0) // vpermd -0x20(%0,%2,4),ymm5,ymm0
>-    "vmovdqu    %%ymm0," MEMACCESS(1) "        \n"
>-    "lea        " MEMLEA(0x20,1) ",%1          \n"
>-    "sub        $0x8,%2                        \n"
>-    "jg         1b                             \n"
>-    "vzeroupper                                \n"
>-  : "+r"(src),  // %0
>-    "+r"(dst),  // %1
>-    "+r"(temp_width)  // %2
>-  : "m"(kARGBShuffleMirror_AVX2) // %3
>-  : "memory", "cc", NACL_R14
>-    "xmm0", "xmm5"
>-  );
>+      "vmovdqu    %3,%%ymm5                      \n"
>+
>+      LABELALIGN
>+      "1:                                        \n"
>+      "vpermd    -0x20(%0,%2,4),%%ymm5,%%ymm0    \n"
>+      "vmovdqu    %%ymm0,(%1)                    \n"
>+      "lea        0x20(%1),%1                    \n"
>+      "sub        $0x8,%2                        \n"
>+      "jg         1b                             \n"
>+      "vzeroupper                                \n"
>+      : "+r"(src),                    // %0
>+        "+r"(dst),                    // %1
>+        "+r"(temp_width)              // %2
>+      : "m"(kARGBShuffleMirror_AVX2)  // %3
>+      : "memory", "cc", "xmm0", "xmm5");
> }
> #endif  // HAS_ARGBMIRRORROW_AVX2
> 
> #ifdef HAS_SPLITUVROW_AVX2
>-void SplitUVRow_AVX2(const uint8* src_uv,
>-                     uint8* dst_u,
>-                     uint8* dst_v,
>+void SplitUVRow_AVX2(const uint8_t* src_uv,
>+                     uint8_t* dst_u,
>+                     uint8_t* dst_v,
>                      int width) {
>-  asm volatile (
>-    "vpcmpeqb   %%ymm5,%%ymm5,%%ymm5           \n"
>-    "vpsrlw     $0x8,%%ymm5,%%ymm5             \n"
>-    "sub        %1,%2                          \n"
>+  asm volatile(
>+      "vpcmpeqb   %%ymm5,%%ymm5,%%ymm5           \n"
>+      "vpsrlw     $0x8,%%ymm5,%%ymm5             \n"
>+      "sub        %1,%2                          \n"
> 
>-    LABELALIGN
>-    "1:                                        \n"
>-    "vmovdqu    " MEMACCESS(0) ",%%ymm0        \n"
>-    "vmovdqu    " MEMACCESS2(0x20,0) ",%%ymm1  \n"
>-    "lea        " MEMLEA(0x40,0) ",%0          \n"
>-    "vpsrlw     $0x8,%%ymm0,%%ymm2             \n"
>-    "vpsrlw     $0x8,%%ymm1,%%ymm3             \n"
>-    "vpand      %%ymm5,%%ymm0,%%ymm0           \n"
>-    "vpand      %%ymm5,%%ymm1,%%ymm1           \n"
>-    "vpackuswb  %%ymm1,%%ymm0,%%ymm0           \n"
>-    "vpackuswb  %%ymm3,%%ymm2,%%ymm2           \n"
>-    "vpermq     $0xd8,%%ymm0,%%ymm0            \n"
>-    "vpermq     $0xd8,%%ymm2,%%ymm2            \n"
>-    "vmovdqu    %%ymm0," MEMACCESS(1) "        \n"
>-    MEMOPMEM(vmovdqu,ymm2,0x00,1,2,1)           //  vmovdqu %%ymm2,(%1,%2)
>-    "lea        " MEMLEA(0x20,1) ",%1          \n"
>-    "sub        $0x20,%3                       \n"
>-    "jg         1b                             \n"
>-    "vzeroupper                                \n"
>-  : "+r"(src_uv),     // %0
>-    "+r"(dst_u),      // %1
>-    "+r"(dst_v),      // %2
>-    "+r"(width)         // %3
>-  :
>-  : "memory", "cc", NACL_R14
>-    "xmm0", "xmm1", "xmm2", "xmm3", "xmm5"
>-  );
>+      LABELALIGN
>+      "1:                                        \n"
>+      "vmovdqu    (%0),%%ymm0                    \n"
>+      "vmovdqu    0x20(%0),%%ymm1                \n"
>+      "lea        0x40(%0),%0                    \n"
>+      "vpsrlw     $0x8,%%ymm0,%%ymm2             \n"
>+      "vpsrlw     $0x8,%%ymm1,%%ymm3             \n"
>+      "vpand      %%ymm5,%%ymm0,%%ymm0           \n"
>+      "vpand      %%ymm5,%%ymm1,%%ymm1           \n"
>+      "vpackuswb  %%ymm1,%%ymm0,%%ymm0           \n"
>+      "vpackuswb  %%ymm3,%%ymm2,%%ymm2           \n"
>+      "vpermq     $0xd8,%%ymm0,%%ymm0            \n"
>+      "vpermq     $0xd8,%%ymm2,%%ymm2            \n"
>+      "vmovdqu    %%ymm0,(%1)                    \n"
>+      "vmovdqu    %%ymm2,0x00(%1,%2,1)            \n"
>+      "lea        0x20(%1),%1                    \n"
>+      "sub        $0x20,%3                       \n"
>+      "jg         1b                             \n"
>+      "vzeroupper                                \n"
>+      : "+r"(src_uv),  // %0
>+        "+r"(dst_u),   // %1
>+        "+r"(dst_v),   // %2
>+        "+r"(width)    // %3
>+      :
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm5");
> }
> #endif  // HAS_SPLITUVROW_AVX2
> 
> #ifdef HAS_SPLITUVROW_SSE2
>-void SplitUVRow_SSE2(const uint8* src_uv,
>-                     uint8* dst_u,
>-                     uint8* dst_v,
>+void SplitUVRow_SSE2(const uint8_t* src_uv,
>+                     uint8_t* dst_u,
>+                     uint8_t* dst_v,
>                      int width) {
>-  asm volatile (
>-    "pcmpeqb    %%xmm5,%%xmm5                  \n"
>-    "psrlw      $0x8,%%xmm5                    \n"
>-    "sub        %1,%2                          \n"
>+  asm volatile(
>+      "pcmpeqb    %%xmm5,%%xmm5                  \n"
>+      "psrlw      $0x8,%%xmm5                    \n"
>+      "sub        %1,%2                          \n"
> 
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movdqu     " MEMACCESS(0) ",%%xmm0        \n"
>-    "movdqu     " MEMACCESS2(0x10,0) ",%%xmm1  \n"
>-    "lea        " MEMLEA(0x20,0) ",%0          \n"
>-    "movdqa     %%xmm0,%%xmm2                  \n"
>-    "movdqa     %%xmm1,%%xmm3                  \n"
>-    "pand       %%xmm5,%%xmm0                  \n"
>-    "pand       %%xmm5,%%xmm1                  \n"
>-    "packuswb   %%xmm1,%%xmm0                  \n"
>-    "psrlw      $0x8,%%xmm2                    \n"
>-    "psrlw      $0x8,%%xmm3                    \n"
>-    "packuswb   %%xmm3,%%xmm2                  \n"
>-    "movdqu     %%xmm0," MEMACCESS(1) "        \n"
>-    MEMOPMEM(movdqu,xmm2,0x00,1,2,1)           //  movdqu     %%xmm2,(%1,%2)
>-    "lea        " MEMLEA(0x10,1) ",%1          \n"
>-    "sub        $0x10,%3                       \n"
>-    "jg         1b                             \n"
>-  : "+r"(src_uv),     // %0
>-    "+r"(dst_u),      // %1
>-    "+r"(dst_v),      // %2
>-    "+r"(width)         // %3
>-  :
>-  : "memory", "cc", NACL_R14
>-    "xmm0", "xmm1", "xmm2", "xmm3", "xmm5"
>-  );
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu     (%0),%%xmm0                    \n"
>+      "movdqu     0x10(%0),%%xmm1                \n"
>+      "lea        0x20(%0),%0                    \n"
>+      "movdqa     %%xmm0,%%xmm2                  \n"
>+      "movdqa     %%xmm1,%%xmm3                  \n"
>+      "pand       %%xmm5,%%xmm0                  \n"
>+      "pand       %%xmm5,%%xmm1                  \n"
>+      "packuswb   %%xmm1,%%xmm0                  \n"
>+      "psrlw      $0x8,%%xmm2                    \n"
>+      "psrlw      $0x8,%%xmm3                    \n"
>+      "packuswb   %%xmm3,%%xmm2                  \n"
>+      "movdqu     %%xmm0,(%1)                    \n"
>+      "movdqu    %%xmm2,0x00(%1,%2,1)            \n"
>+      "lea        0x10(%1),%1                    \n"
>+      "sub        $0x10,%3                       \n"
>+      "jg         1b                             \n"
>+      : "+r"(src_uv),  // %0
>+        "+r"(dst_u),   // %1
>+        "+r"(dst_v),   // %2
>+        "+r"(width)    // %3
>+      :
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm5");
> }
> #endif  // HAS_SPLITUVROW_SSE2
> 
> #ifdef HAS_MERGEUVROW_AVX2
>-void MergeUVRow_AVX2(const uint8* src_u,
>-                     const uint8* src_v,
>-                     uint8* dst_uv,
>+void MergeUVRow_AVX2(const uint8_t* src_u,
>+                     const uint8_t* src_v,
>+                     uint8_t* dst_uv,
>                      int width) {
>-  asm volatile (
>-    "sub       %0,%1                           \n"
>+  asm volatile(
> 
>-    LABELALIGN
>-    "1:                                        \n"
>-    "vmovdqu   " MEMACCESS(0) ",%%ymm0         \n"
>-    MEMOPREG(vmovdqu,0x00,0,1,1,ymm1)           //  vmovdqu (%0,%1,1),%%ymm1
>-    "lea       " MEMLEA(0x20,0) ",%0           \n"
>-    "vpunpcklbw %%ymm1,%%ymm0,%%ymm2           \n"
>-    "vpunpckhbw %%ymm1,%%ymm0,%%ymm0           \n"
>-    "vextractf128 $0x0,%%ymm2," MEMACCESS(2) " \n"
>-    "vextractf128 $0x0,%%ymm0," MEMACCESS2(0x10,2) "\n"
>-    "vextractf128 $0x1,%%ymm2," MEMACCESS2(0x20,2) "\n"
>-    "vextractf128 $0x1,%%ymm0," MEMACCESS2(0x30,2) "\n"
>-    "lea       " MEMLEA(0x40,2) ",%2           \n"
>-    "sub       $0x20,%3                        \n"
>-    "jg        1b                              \n"
>-    "vzeroupper                                \n"
>-  : "+r"(src_u),     // %0
>-    "+r"(src_v),     // %1
>-    "+r"(dst_uv),    // %2
>-    "+r"(width)      // %3
>-  :
>-  : "memory", "cc", NACL_R14
>-    "xmm0", "xmm1", "xmm2"
>-  );
>+      "sub       %0,%1                           \n"
>+
>+      LABELALIGN
>+      "1:                                        \n"
>+      "vmovdqu   (%0),%%ymm0                     \n"
>+      "vmovdqu    0x00(%0,%1,1),%%ymm1           \n"
>+      "lea       0x20(%0),%0                     \n"
>+      "vpunpcklbw %%ymm1,%%ymm0,%%ymm2           \n"
>+      "vpunpckhbw %%ymm1,%%ymm0,%%ymm0           \n"
>+      "vextractf128 $0x0,%%ymm2,(%2)             \n"
>+      "vextractf128 $0x0,%%ymm0,0x10(%2)         \n"
>+      "vextractf128 $0x1,%%ymm2,0x20(%2)         \n"
>+      "vextractf128 $0x1,%%ymm0,0x30(%2)         \n"
>+      "lea       0x40(%2),%2                     \n"
>+      "sub       $0x20,%3                        \n"
>+      "jg        1b                              \n"
>+      "vzeroupper                                \n"
>+      : "+r"(src_u),   // %0
>+        "+r"(src_v),   // %1
>+        "+r"(dst_uv),  // %2
>+        "+r"(width)    // %3
>+      :
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2");
> }
> #endif  // HAS_MERGEUVROW_AVX2
> 
> #ifdef HAS_MERGEUVROW_SSE2
>-void MergeUVRow_SSE2(const uint8* src_u,
>-                     const uint8* src_v,
>-                     uint8* dst_uv,
>+void MergeUVRow_SSE2(const uint8_t* src_u,
>+                     const uint8_t* src_v,
>+                     uint8_t* dst_uv,
>                      int width) {
>-  asm volatile (
>-    "sub       %0,%1                           \n"
>+  asm volatile(
> 
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    MEMOPREG(movdqu,0x00,0,1,1,xmm1)           //  movdqu    (%0,%1,1),%%xmm1
>-    "lea       " MEMLEA(0x10,0) ",%0           \n"
>-    "movdqa    %%xmm0,%%xmm2                   \n"
>-    "punpcklbw %%xmm1,%%xmm0                   \n"
>-    "punpckhbw %%xmm1,%%xmm2                   \n"
>-    "movdqu    %%xmm0," MEMACCESS(2) "         \n"
>-    "movdqu    %%xmm2," MEMACCESS2(0x10,2) "   \n"
>-    "lea       " MEMLEA(0x20,2) ",%2           \n"
>-    "sub       $0x10,%3                        \n"
>-    "jg        1b                              \n"
>-  : "+r"(src_u),     // %0
>-    "+r"(src_v),     // %1
>-    "+r"(dst_uv),    // %2
>-    "+r"(width)      // %3
>-  :
>-  : "memory", "cc", NACL_R14
>-    "xmm0", "xmm1", "xmm2"
>-  );
>+      "sub       %0,%1                           \n"
>+
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "movdqu    0x00(%0,%1,1),%%xmm1            \n"
>+      "lea       0x10(%0),%0                     \n"
>+      "movdqa    %%xmm0,%%xmm2                   \n"
>+      "punpcklbw %%xmm1,%%xmm0                   \n"
>+      "punpckhbw %%xmm1,%%xmm2                   \n"
>+      "movdqu    %%xmm0,(%2)                     \n"
>+      "movdqu    %%xmm2,0x10(%2)                 \n"
>+      "lea       0x20(%2),%2                     \n"
>+      "sub       $0x10,%3                        \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_u),   // %0
>+        "+r"(src_v),   // %1
>+        "+r"(dst_uv),  // %2
>+        "+r"(width)    // %3
>+      :
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2");
> }
> #endif  // HAS_MERGEUVROW_SSE2
> 
>@@ -2759,9 +3209,9 @@ void MergeUVRow_SSE2(const uint8* src_u,
> // 16 = 12 bits
> // 1 = 16 bits
> #ifdef HAS_MERGEUVROW_16_AVX2
>-void MergeUVRow_16_AVX2(const uint16* src_u,
>-                        const uint16* src_v,
>-                        uint16* dst_uv,
>+void MergeUVRow_16_AVX2(const uint16_t* src_u,
>+                        const uint16_t* src_v,
>+                        uint16_t* dst_uv,
>                         int scale,
>                         int width) {
>   // clang-format off
>@@ -2800,9 +3250,14 @@ void MergeUVRow_16_AVX2(const uint16* src_u,
> }
> #endif  // HAS_MERGEUVROW_AVX2
> 
>+// Use scale to convert lsb formats to msb, depending how many bits there are:
>+// 128 = 9 bits
>+// 64 = 10 bits
>+// 16 = 12 bits
>+// 1 = 16 bits
> #ifdef HAS_MULTIPLYROW_16_AVX2
>-void MultiplyRow_16_AVX2(const uint16* src_y,
>-                         uint16* dst_y,
>+void MultiplyRow_16_AVX2(const uint16_t* src_y,
>+                         uint16_t* dst_y,
>                          int scale,
>                          int width) {
>   // clang-format off
>@@ -2834,367 +3289,507 @@ void MultiplyRow_16_AVX2(const uint16* src_y,
> }
> #endif  // HAS_MULTIPLYROW_16_AVX2
> 
>-#ifdef HAS_SPLITRGBROW_SSSE3
>-
>-// Shuffle table for converting RGB to Planar.
>-static uvec8 kShuffleMaskRGBToR0 = {0u,   3u,   6u,   9u,   12u,  15u,
>-                                    128u, 128u, 128u, 128u, 128u, 128u,
>-                                    128u, 128u, 128u, 128u};
>-static uvec8 kShuffleMaskRGBToR1 = {128u, 128u, 128u, 128u, 128u, 128u,
>-                                    2u,   5u,   8u,   11u,  14u,  128u,
>-                                    128u, 128u, 128u, 128u};
>-static uvec8 kShuffleMaskRGBToR2 = {128u, 128u, 128u, 128u, 128u, 128u,
>-                                    128u, 128u, 128u, 128u, 128u, 1u,
>-                                    4u,   7u,   10u,  13u};
>-
>-static uvec8 kShuffleMaskRGBToG0 = {1u,   4u,   7u,   10u,  13u,  128u,
>-                                    128u, 128u, 128u, 128u, 128u, 128u,
>-                                    128u, 128u, 128u, 128u};
>-static uvec8 kShuffleMaskRGBToG1 = {128u, 128u, 128u, 128u, 128u, 0u,
>-                                    3u,   6u,   9u,   12u,  15u,  128u,
>-                                    128u, 128u, 128u, 128u};
>-static uvec8 kShuffleMaskRGBToG2 = {128u, 128u, 128u, 128u, 128u, 128u,
>-                                    128u, 128u, 128u, 128u, 128u, 2u,
>-                                    5u,   8u,   11u,  14u};
>-
>-static uvec8 kShuffleMaskRGBToB0 = {2u,   5u,   8u,   11u,  14u,  128u,
>-                                    128u, 128u, 128u, 128u, 128u, 128u,
>-                                    128u, 128u, 128u, 128u};
>-static uvec8 kShuffleMaskRGBToB1 = {128u, 128u, 128u, 128u, 128u, 1u,
>-                                    4u,   7u,   10u,  13u,  128u, 128u,
>-                                    128u, 128u, 128u, 128u};
>-static uvec8 kShuffleMaskRGBToB2 = {128u, 128u, 128u, 128u, 128u, 128u,
>-                                    128u, 128u, 128u, 128u, 0u,   3u,
>-                                    6u,   9u,   12u,  15u};
>-
>-void SplitRGBRow_SSSE3(const uint8* src_rgb,
>-                       uint8* dst_r,
>-                       uint8* dst_g,
>-                       uint8* dst_b,
>-                       int width) {
>+// Use scale to convert lsb formats to msb, depending how many bits there are:
>+// 32768 = 9 bits
>+// 16384 = 10 bits
>+// 4096 = 12 bits
>+// 256 = 16 bits
>+void Convert16To8Row_SSSE3(const uint16_t* src_y,
>+                           uint8_t* dst_y,
>+                           int scale,
>+                           int width) {
>+  // clang-format off
>   asm volatile (
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movdqu     " MEMACCESS(0) ",%%xmm0        \n"
>-    "movdqu     " MEMACCESS2(0x10,0) ",%%xmm1  \n"
>-    "movdqu     " MEMACCESS2(0x20,0) ",%%xmm2  \n"
>-    "pshufb     %5, %%xmm0                     \n"
>-    "pshufb     %6, %%xmm1                     \n"
>-    "pshufb     %7, %%xmm2                     \n"
>-    "por        %%xmm1,%%xmm0                  \n"
>-    "por        %%xmm2,%%xmm0                  \n"
>-    "movdqu     %%xmm0," MEMACCESS(1) "        \n"
>-    "lea        " MEMLEA(0x10,1) ",%1          \n"
>-
>-    "movdqu     " MEMACCESS(0) ",%%xmm0        \n"
>-    "movdqu     " MEMACCESS2(0x10,0) ",%%xmm1  \n"
>-    "movdqu     " MEMACCESS2(0x20,0) ",%%xmm2  \n"
>-    "pshufb     %8, %%xmm0                     \n"
>-    "pshufb     %9, %%xmm1                     \n"
>-    "pshufb     %10, %%xmm2                    \n"
>-    "por        %%xmm1,%%xmm0                  \n"
>-    "por        %%xmm2,%%xmm0                  \n"
>-    "movdqu     %%xmm0," MEMACCESS(2) "        \n"
>-    "lea        " MEMLEA(0x10,2) ",%2          \n"
>-
>-    "movdqu     " MEMACCESS(0) ",%%xmm0        \n"
>-    "movdqu     " MEMACCESS2(0x10,0) ",%%xmm1  \n"
>-    "movdqu     " MEMACCESS2(0x20,0) ",%%xmm2  \n"
>-    "pshufb     %11, %%xmm0                    \n"
>-    "pshufb     %12, %%xmm1                    \n"
>-    "pshufb     %13, %%xmm2                    \n"
>-    "por        %%xmm1,%%xmm0                  \n"
>-    "por        %%xmm2,%%xmm0                  \n"
>-    "movdqu     %%xmm0," MEMACCESS(3) "        \n"
>-    "lea        " MEMLEA(0x10,3) ",%3          \n"
>-    "lea        " MEMLEA(0x30,0) ",%0          \n"
>-    "sub        $0x10,%4                       \n"
>-    "jg         1b                             \n"
>-  : "+r"(src_rgb),              // %0
>-    "+r"(dst_r),                // %1
>-    "+r"(dst_g),                // %2
>-    "+r"(dst_b),                // %3
>-    "+r"(width)                 // %4
>-  : "m"(kShuffleMaskRGBToR0),   // %5
>-    "m"(kShuffleMaskRGBToR1),   // %6
>-    "m"(kShuffleMaskRGBToR2),   // %7
>-    "m"(kShuffleMaskRGBToG0),   // %8
>-    "m"(kShuffleMaskRGBToG1),   // %9
>-    "m"(kShuffleMaskRGBToG2),   // %10
>-    "m"(kShuffleMaskRGBToB0),   // %11
>-    "m"(kShuffleMaskRGBToB1),   // %12
>-    "m"(kShuffleMaskRGBToB2)    // %13
>-  : "memory", "cc", NACL_R14
>-    "xmm0", "xmm1", "xmm2"
>-  );
>-}
>-#endif  // HAS_SPLITRGBROW_SSSE3
>-
>-#ifdef HAS_MERGERGBROW_SSSE3
>+    "movd      %3,%%xmm2                      \n"
>+    "punpcklwd %%xmm2,%%xmm2                  \n"
>+    "pshufd    $0x0,%%xmm2,%%xmm2             \n"
>+
>+    // 32 pixels per loop.
>+    LABELALIGN
>+    "1:                                       \n"
>+    "movdqu    (%0),%%xmm0                    \n"
>+    "movdqu    0x10(%0),%%xmm1                \n"
>+    "add       $0x20,%0                       \n"
>+    "pmulhuw   %%xmm2,%%xmm0                  \n"
>+    "pmulhuw   %%xmm2,%%xmm1                  \n"
>+    "packuswb  %%xmm1,%%xmm0                  \n"
>+    "movdqu    %%xmm0,(%1)                    \n"
>+    "add       $0x10,%1                       \n"
>+    "sub       $0x10,%2                       \n"
>+    "jg        1b                             \n"
>+  : "+r"(src_y),   // %0
>+    "+r"(dst_y),   // %1
>+    "+r"(width)    // %2
>+  : "r"(scale)     // %3
>+  : "memory", "cc", "xmm0", "xmm1", "xmm2");
>+  // clang-format on
>+}
> 
>-// Shuffle table for converting RGB to Planar.
>-static uvec8 kShuffleMaskRToRGB0 = {0u, 128u, 128u, 1u, 128u, 128u,
>-                                    2u, 128u, 128u, 3u, 128u, 128u,
>-                                    4u, 128u, 128u, 5u};
>-static uvec8 kShuffleMaskGToRGB0 = {128u, 0u, 128u, 128u, 1u, 128u,
>-                                    128u, 2u, 128u, 128u, 3u, 128u,
>-                                    128u, 4u, 128u, 128u};
>-static uvec8 kShuffleMaskBToRGB0 = {128u, 128u, 0u, 128u, 128u, 1u,
>-                                    128u, 128u, 2u, 128u, 128u, 3u,
>-                                    128u, 128u, 4u, 128u};
>-
>-static uvec8 kShuffleMaskGToRGB1 = {5u, 128u, 128u, 6u, 128u, 128u,
>-                                    7u, 128u, 128u, 8u, 128u, 128u,
>-                                    9u, 128u, 128u, 10u};
>-static uvec8 kShuffleMaskBToRGB1 = {128u, 5u, 128u, 128u, 6u, 128u,
>-                                    128u, 7u, 128u, 128u, 8u, 128u,
>-                                    128u, 9u, 128u, 128u};
>-static uvec8 kShuffleMaskRToRGB1 = {128u, 128u, 6u,  128u, 128u, 7u,
>-                                    128u, 128u, 8u,  128u, 128u, 9u,
>-                                    128u, 128u, 10u, 128u};
>-
>-static uvec8 kShuffleMaskBToRGB2 = {10u, 128u, 128u, 11u, 128u, 128u,
>-                                    12u, 128u, 128u, 13u, 128u, 128u,
>-                                    14u, 128u, 128u, 15u};
>-static uvec8 kShuffleMaskRToRGB2 = {128u, 11u, 128u, 128u, 12u, 128u,
>-                                    128u, 13u, 128u, 128u, 14u, 128u,
>-                                    128u, 15u, 128u, 128u};
>-static uvec8 kShuffleMaskGToRGB2 = {128u, 128u, 11u, 128u, 128u, 12u,
>-                                    128u, 128u, 13u, 128u, 128u, 14u,
>-                                    128u, 128u, 15u, 128u};
>-
>-void MergeRGBRow_SSSE3(const uint8* src_r,
>-                       const uint8* src_g,
>-                       const uint8* src_b,
>-                       uint8* dst_rgb,
>-                       int width) {
>+#ifdef HAS_CONVERT16TO8ROW_AVX2
>+void Convert16To8Row_AVX2(const uint16_t* src_y,
>+                          uint8_t* dst_y,
>+                          int scale,
>+                          int width) {
>+  // clang-format off
>   asm volatile (
>+    "vmovd      %3,%%xmm2                      \n"
>+    "vpunpcklwd %%xmm2,%%xmm2,%%xmm2           \n"
>+    "vbroadcastss %%xmm2,%%ymm2                \n"
>+
>+    // 32 pixels per loop.
>     LABELALIGN
>     "1:                                        \n"
>-    "movdqu     " MEMACCESS(0) ",%%xmm0        \n"
>-    "movdqu     " MEMACCESS(1) ",%%xmm1        \n"
>-    "movdqu     " MEMACCESS(2) ",%%xmm2        \n"
>-    "pshufb     %5, %%xmm0                     \n"
>-    "pshufb     %6, %%xmm1                     \n"
>-    "pshufb     %7, %%xmm2                     \n"
>-    "por        %%xmm1,%%xmm0                  \n"
>-    "por        %%xmm2,%%xmm0                  \n"
>-    "movdqu     %%xmm0," MEMACCESS(3) "        \n"
>-
>-    "movdqu     " MEMACCESS(0) ",%%xmm0        \n"
>-    "movdqu     " MEMACCESS(1) ",%%xmm1        \n"
>-    "movdqu     " MEMACCESS(2) ",%%xmm2        \n"
>-    "pshufb     %8, %%xmm0                     \n"
>-    "pshufb     %9, %%xmm1                     \n"
>-    "pshufb     %10, %%xmm2                    \n"
>-    "por        %%xmm1,%%xmm0                  \n"
>-    "por        %%xmm2,%%xmm0                  \n"
>-    "movdqu     %%xmm0," MEMACCESS2(16, 3) "   \n"
>-
>-    "movdqu     " MEMACCESS(0) ",%%xmm0        \n"
>-    "movdqu     " MEMACCESS(1) ",%%xmm1        \n"
>-    "movdqu     " MEMACCESS(2) ",%%xmm2        \n"
>-    "pshufb     %11, %%xmm0                    \n"
>-    "pshufb     %12, %%xmm1                    \n"
>-    "pshufb     %13, %%xmm2                    \n"
>-    "por        %%xmm1,%%xmm0                  \n"
>-    "por        %%xmm2,%%xmm0                  \n"
>-    "movdqu     %%xmm0," MEMACCESS2(32, 3) "   \n"
>-
>-    "lea        " MEMLEA(0x10,0) ",%0          \n"
>-    "lea        " MEMLEA(0x10,1) ",%1          \n"
>-    "lea        " MEMLEA(0x10,2) ",%2          \n"
>-    "lea        " MEMLEA(0x30,3) ",%3          \n"
>-    "sub        $0x10,%4                       \n"
>-    "jg         1b                             \n"
>-  : "+r"(src_r),                // %0
>-    "+r"(src_g),                // %1
>-    "+r"(src_b),                // %2
>-    "+r"(dst_rgb),              // %3
>-    "+r"(width)                 // %4
>-  : "m"(kShuffleMaskRToRGB0),   // %5
>-    "m"(kShuffleMaskGToRGB0),   // %6
>-    "m"(kShuffleMaskBToRGB0),   // %7
>-    "m"(kShuffleMaskRToRGB1),   // %8
>-    "m"(kShuffleMaskGToRGB1),   // %9
>-    "m"(kShuffleMaskBToRGB1),   // %10
>-    "m"(kShuffleMaskRToRGB2),   // %11
>-    "m"(kShuffleMaskGToRGB2),   // %12
>-    "m"(kShuffleMaskBToRGB2)    // %13
>-  : "memory", "cc", NACL_R14
>-    "xmm0", "xmm1", "xmm2"
>-  );
>+    "vmovdqu   (%0),%%ymm0                     \n"
>+    "vmovdqu   0x20(%0),%%ymm1                 \n"
>+    "add       $0x40,%0                        \n"
>+    "vpmulhuw  %%ymm2,%%ymm0,%%ymm0            \n"
>+    "vpmulhuw  %%ymm2,%%ymm1,%%ymm1            \n"
>+    "vpackuswb %%ymm1,%%ymm0,%%ymm0            \n"  // mutates
>+    "vpermq    $0xd8,%%ymm0,%%ymm0             \n"
>+    "vmovdqu   %%ymm0,(%1)                     \n"
>+    "add       $0x20,%1                        \n"
>+    "sub       $0x20,%2                        \n"
>+    "jg        1b                              \n"
>+    "vzeroupper                                \n"
>+  : "+r"(src_y),   // %0
>+    "+r"(dst_y),   // %1
>+    "+r"(width)    // %2
>+  : "r"(scale)     // %3
>+  : "memory", "cc", "xmm0", "xmm1", "xmm2");
>+  // clang-format on
>+}
>+#endif  // HAS_CONVERT16TO8ROW_AVX2
>+
>+// Use scale to convert to lsb formats depending how many bits there are:
>+// 512 = 9 bits
>+// 1024 = 10 bits
>+// 4096 = 12 bits
>+// TODO(fbarchard): reduce to SSE2
>+void Convert8To16Row_SSE2(const uint8_t* src_y,
>+                          uint16_t* dst_y,
>+                          int scale,
>+                          int width) {
>+  // clang-format off
>+  asm volatile (
>+    "movd      %3,%%xmm2                      \n"
>+    "punpcklwd %%xmm2,%%xmm2                  \n"
>+    "pshufd    $0x0,%%xmm2,%%xmm2             \n"
>+
>+    // 32 pixels per loop.
>+    LABELALIGN
>+    "1:                                       \n"
>+    "movdqu    (%0),%%xmm0                    \n"
>+    "movdqa    %%xmm0,%%xmm1                  \n"
>+    "punpcklbw %%xmm0,%%xmm0                  \n"
>+    "punpckhbw %%xmm1,%%xmm1                  \n"
>+    "add       $0x10,%0                       \n"
>+    "pmulhuw   %%xmm2,%%xmm0                  \n"
>+    "pmulhuw   %%xmm2,%%xmm1                  \n"
>+    "movdqu    %%xmm0,(%1)                    \n"
>+    "movdqu    %%xmm1,0x10(%1)                \n"
>+    "add       $0x20,%1                       \n"
>+    "sub       $0x10,%2                       \n"
>+    "jg        1b                             \n"
>+  : "+r"(src_y),   // %0
>+    "+r"(dst_y),   // %1
>+    "+r"(width)    // %2
>+  : "r"(scale)     // %3
>+  : "memory", "cc", "xmm0", "xmm1", "xmm2");
>+  // clang-format on
> }
>-#endif  // HAS_MERGERGBROW_SSSE3
> 
>-#ifdef HAS_COPYROW_SSE2
>-void CopyRow_SSE2(const uint8* src, uint8* dst, int count) {
>+#ifdef HAS_CONVERT8TO16ROW_AVX2
>+void Convert8To16Row_AVX2(const uint8_t* src_y,
>+                          uint16_t* dst_y,
>+                          int scale,
>+                          int width) {
>+  // clang-format off
>   asm volatile (
>-    "test       $0xf,%0                        \n"
>-    "jne        2f                             \n"
>-    "test       $0xf,%1                        \n"
>-    "jne        2f                             \n"
>+    "vmovd      %3,%%xmm2                      \n"
>+    "vpunpcklwd %%xmm2,%%xmm2,%%xmm2           \n"
>+    "vbroadcastss %%xmm2,%%ymm2                \n"
> 
>+    // 32 pixels per loop.
>     LABELALIGN
>     "1:                                        \n"
>-    "movdqa    " MEMACCESS(0) ",%%xmm0         \n"
>-    "movdqa    " MEMACCESS2(0x10,0) ",%%xmm1   \n"
>-    "lea       " MEMLEA(0x20,0) ",%0           \n"
>-    "movdqa    %%xmm0," MEMACCESS(1) "         \n"
>-    "movdqa    %%xmm1," MEMACCESS2(0x10,1) "   \n"
>-    "lea       " MEMLEA(0x20,1) ",%1           \n"
>+    "vmovdqu   (%0),%%ymm0                     \n"
>+    "vpermq    $0xd8,%%ymm0,%%ymm0             \n"
>+    "add       $0x20,%0                        \n"
>+    "vpunpckhbw %%ymm0,%%ymm0,%%ymm1           \n"
>+    "vpunpcklbw %%ymm0,%%ymm0,%%ymm0           \n"
>+    "vpmulhuw  %%ymm2,%%ymm0,%%ymm0            \n"
>+    "vpmulhuw  %%ymm2,%%ymm1,%%ymm1            \n"
>+    "vmovdqu   %%ymm0,(%1)                     \n"
>+    "vmovdqu   %%ymm1,0x20(%1)                 \n"
>+    "add       $0x40,%1                        \n"
>     "sub       $0x20,%2                        \n"
>     "jg        1b                              \n"
>-    "jmp       9f                              \n"
>+    "vzeroupper                                \n"
>+  : "+r"(src_y),   // %0
>+    "+r"(dst_y),   // %1
>+    "+r"(width)    // %2
>+  : "r"(scale)     // %3
>+  : "memory", "cc", "xmm0", "xmm1", "xmm2");
>+  // clang-format on
>+}
>+#endif  // HAS_CONVERT8TO16ROW_AVX2
> 
>-    LABELALIGN
>-  "2:                                          \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    "movdqu    " MEMACCESS2(0x10,0) ",%%xmm1   \n"
>-    "lea       " MEMLEA(0x20,0) ",%0           \n"
>-    "movdqu    %%xmm0," MEMACCESS(1) "         \n"
>-    "movdqu    %%xmm1," MEMACCESS2(0x10,1) "   \n"
>-    "lea       " MEMLEA(0x20,1) ",%1           \n"
>-    "sub       $0x20,%2                        \n"
>-    "jg        2b                              \n"
>-  "9:                                          \n"
>-  : "+r"(src),   // %0
>-    "+r"(dst),   // %1
>-    "+r"(count)  // %2
>-  :
>-  : "memory", "cc"
>-    , "xmm0", "xmm1"
>-  );
>+#ifdef HAS_SPLITRGBROW_SSSE3
>+
>+// Shuffle table for converting RGB to Planar.
>+static const uvec8 kShuffleMaskRGBToR0 = {0u,   3u,   6u,   9u,   12u,  15u,
>+                                          128u, 128u, 128u, 128u, 128u, 128u,
>+                                          128u, 128u, 128u, 128u};
>+static const uvec8 kShuffleMaskRGBToR1 = {128u, 128u, 128u, 128u, 128u, 128u,
>+                                          2u,   5u,   8u,   11u,  14u,  128u,
>+                                          128u, 128u, 128u, 128u};
>+static const uvec8 kShuffleMaskRGBToR2 = {128u, 128u, 128u, 128u, 128u, 128u,
>+                                          128u, 128u, 128u, 128u, 128u, 1u,
>+                                          4u,   7u,   10u,  13u};
>+
>+static const uvec8 kShuffleMaskRGBToG0 = {1u,   4u,   7u,   10u,  13u,  128u,
>+                                          128u, 128u, 128u, 128u, 128u, 128u,
>+                                          128u, 128u, 128u, 128u};
>+static const uvec8 kShuffleMaskRGBToG1 = {128u, 128u, 128u, 128u, 128u, 0u,
>+                                          3u,   6u,   9u,   12u,  15u,  128u,
>+                                          128u, 128u, 128u, 128u};
>+static const uvec8 kShuffleMaskRGBToG2 = {128u, 128u, 128u, 128u, 128u, 128u,
>+                                          128u, 128u, 128u, 128u, 128u, 2u,
>+                                          5u,   8u,   11u,  14u};
>+
>+static const uvec8 kShuffleMaskRGBToB0 = {2u,   5u,   8u,   11u,  14u,  128u,
>+                                          128u, 128u, 128u, 128u, 128u, 128u,
>+                                          128u, 128u, 128u, 128u};
>+static const uvec8 kShuffleMaskRGBToB1 = {128u, 128u, 128u, 128u, 128u, 1u,
>+                                          4u,   7u,   10u,  13u,  128u, 128u,
>+                                          128u, 128u, 128u, 128u};
>+static const uvec8 kShuffleMaskRGBToB2 = {128u, 128u, 128u, 128u, 128u, 128u,
>+                                          128u, 128u, 128u, 128u, 0u,   3u,
>+                                          6u,   9u,   12u,  15u};
>+
>+void SplitRGBRow_SSSE3(const uint8_t* src_rgb,
>+                       uint8_t* dst_r,
>+                       uint8_t* dst_g,
>+                       uint8_t* dst_b,
>+                       int width) {
>+  asm volatile(
>+
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu     (%0),%%xmm0                    \n"
>+      "movdqu     0x10(%0),%%xmm1                \n"
>+      "movdqu     0x20(%0),%%xmm2                \n"
>+      "pshufb     %5, %%xmm0                     \n"
>+      "pshufb     %6, %%xmm1                     \n"
>+      "pshufb     %7, %%xmm2                     \n"
>+      "por        %%xmm1,%%xmm0                  \n"
>+      "por        %%xmm2,%%xmm0                  \n"
>+      "movdqu     %%xmm0,(%1)                    \n"
>+      "lea        0x10(%1),%1                    \n"
>+
>+      "movdqu     (%0),%%xmm0                    \n"
>+      "movdqu     0x10(%0),%%xmm1                \n"
>+      "movdqu     0x20(%0),%%xmm2                \n"
>+      "pshufb     %8, %%xmm0                     \n"
>+      "pshufb     %9, %%xmm1                     \n"
>+      "pshufb     %10, %%xmm2                    \n"
>+      "por        %%xmm1,%%xmm0                  \n"
>+      "por        %%xmm2,%%xmm0                  \n"
>+      "movdqu     %%xmm0,(%2)                    \n"
>+      "lea        0x10(%2),%2                    \n"
>+
>+      "movdqu     (%0),%%xmm0                    \n"
>+      "movdqu     0x10(%0),%%xmm1                \n"
>+      "movdqu     0x20(%0),%%xmm2                \n"
>+      "pshufb     %11, %%xmm0                    \n"
>+      "pshufb     %12, %%xmm1                    \n"
>+      "pshufb     %13, %%xmm2                    \n"
>+      "por        %%xmm1,%%xmm0                  \n"
>+      "por        %%xmm2,%%xmm0                  \n"
>+      "movdqu     %%xmm0,(%3)                    \n"
>+      "lea        0x10(%3),%3                    \n"
>+      "lea        0x30(%0),%0                    \n"
>+      "sub        $0x10,%4                       \n"
>+      "jg         1b                             \n"
>+      : "+r"(src_rgb),             // %0
>+        "+r"(dst_r),               // %1
>+        "+r"(dst_g),               // %2
>+        "+r"(dst_b),               // %3
>+        "+r"(width)                // %4
>+      : "m"(kShuffleMaskRGBToR0),  // %5
>+        "m"(kShuffleMaskRGBToR1),  // %6
>+        "m"(kShuffleMaskRGBToR2),  // %7
>+        "m"(kShuffleMaskRGBToG0),  // %8
>+        "m"(kShuffleMaskRGBToG1),  // %9
>+        "m"(kShuffleMaskRGBToG2),  // %10
>+        "m"(kShuffleMaskRGBToB0),  // %11
>+        "m"(kShuffleMaskRGBToB1),  // %12
>+        "m"(kShuffleMaskRGBToB2)   // %13
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2");
>+}
>+#endif  // HAS_SPLITRGBROW_SSSE3
>+
>+#ifdef HAS_MERGERGBROW_SSSE3
>+
>+// Shuffle table for converting RGB to Planar.
>+static const uvec8 kShuffleMaskRToRGB0 = {0u, 128u, 128u, 1u, 128u, 128u,
>+                                          2u, 128u, 128u, 3u, 128u, 128u,
>+                                          4u, 128u, 128u, 5u};
>+static const uvec8 kShuffleMaskGToRGB0 = {128u, 0u, 128u, 128u, 1u, 128u,
>+                                          128u, 2u, 128u, 128u, 3u, 128u,
>+                                          128u, 4u, 128u, 128u};
>+static const uvec8 kShuffleMaskBToRGB0 = {128u, 128u, 0u, 128u, 128u, 1u,
>+                                          128u, 128u, 2u, 128u, 128u, 3u,
>+                                          128u, 128u, 4u, 128u};
>+
>+static const uvec8 kShuffleMaskGToRGB1 = {5u, 128u, 128u, 6u, 128u, 128u,
>+                                          7u, 128u, 128u, 8u, 128u, 128u,
>+                                          9u, 128u, 128u, 10u};
>+static const uvec8 kShuffleMaskBToRGB1 = {128u, 5u, 128u, 128u, 6u, 128u,
>+                                          128u, 7u, 128u, 128u, 8u, 128u,
>+                                          128u, 9u, 128u, 128u};
>+static const uvec8 kShuffleMaskRToRGB1 = {128u, 128u, 6u,  128u, 128u, 7u,
>+                                          128u, 128u, 8u,  128u, 128u, 9u,
>+                                          128u, 128u, 10u, 128u};
>+
>+static const uvec8 kShuffleMaskBToRGB2 = {10u, 128u, 128u, 11u, 128u, 128u,
>+                                          12u, 128u, 128u, 13u, 128u, 128u,
>+                                          14u, 128u, 128u, 15u};
>+static const uvec8 kShuffleMaskRToRGB2 = {128u, 11u, 128u, 128u, 12u, 128u,
>+                                          128u, 13u, 128u, 128u, 14u, 128u,
>+                                          128u, 15u, 128u, 128u};
>+static const uvec8 kShuffleMaskGToRGB2 = {128u, 128u, 11u, 128u, 128u, 12u,
>+                                          128u, 128u, 13u, 128u, 128u, 14u,
>+                                          128u, 128u, 15u, 128u};
>+
>+void MergeRGBRow_SSSE3(const uint8_t* src_r,
>+                       const uint8_t* src_g,
>+                       const uint8_t* src_b,
>+                       uint8_t* dst_rgb,
>+                       int width) {
>+  asm volatile(
>+
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu     (%0),%%xmm0                    \n"
>+      "movdqu     (%1),%%xmm1                    \n"
>+      "movdqu     (%2),%%xmm2                    \n"
>+      "pshufb     %5, %%xmm0                     \n"
>+      "pshufb     %6, %%xmm1                     \n"
>+      "pshufb     %7, %%xmm2                     \n"
>+      "por        %%xmm1,%%xmm0                  \n"
>+      "por        %%xmm2,%%xmm0                  \n"
>+      "movdqu     %%xmm0,(%3)                    \n"
>+
>+      "movdqu     (%0),%%xmm0                    \n"
>+      "movdqu     (%1),%%xmm1                    \n"
>+      "movdqu     (%2),%%xmm2                    \n"
>+      "pshufb     %8, %%xmm0                     \n"
>+      "pshufb     %9, %%xmm1                     \n"
>+      "pshufb     %10, %%xmm2                    \n"
>+      "por        %%xmm1,%%xmm0                  \n"
>+      "por        %%xmm2,%%xmm0                  \n"
>+      "movdqu     %%xmm0,16(%3)                  \n"
>+
>+      "movdqu     (%0),%%xmm0                    \n"
>+      "movdqu     (%1),%%xmm1                    \n"
>+      "movdqu     (%2),%%xmm2                    \n"
>+      "pshufb     %11, %%xmm0                    \n"
>+      "pshufb     %12, %%xmm1                    \n"
>+      "pshufb     %13, %%xmm2                    \n"
>+      "por        %%xmm1,%%xmm0                  \n"
>+      "por        %%xmm2,%%xmm0                  \n"
>+      "movdqu     %%xmm0,32(%3)                  \n"
>+
>+      "lea        0x10(%0),%0                    \n"
>+      "lea        0x10(%1),%1                    \n"
>+      "lea        0x10(%2),%2                    \n"
>+      "lea        0x30(%3),%3                    \n"
>+      "sub        $0x10,%4                       \n"
>+      "jg         1b                             \n"
>+      : "+r"(src_r),               // %0
>+        "+r"(src_g),               // %1
>+        "+r"(src_b),               // %2
>+        "+r"(dst_rgb),             // %3
>+        "+r"(width)                // %4
>+      : "m"(kShuffleMaskRToRGB0),  // %5
>+        "m"(kShuffleMaskGToRGB0),  // %6
>+        "m"(kShuffleMaskBToRGB0),  // %7
>+        "m"(kShuffleMaskRToRGB1),  // %8
>+        "m"(kShuffleMaskGToRGB1),  // %9
>+        "m"(kShuffleMaskBToRGB1),  // %10
>+        "m"(kShuffleMaskRToRGB2),  // %11
>+        "m"(kShuffleMaskGToRGB2),  // %12
>+        "m"(kShuffleMaskBToRGB2)   // %13
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2");
>+}
>+#endif  // HAS_MERGERGBROW_SSSE3
>+
>+#ifdef HAS_COPYROW_SSE2
>+void CopyRow_SSE2(const uint8_t* src, uint8_t* dst, int width) {
>+  asm volatile(
>+      "test       $0xf,%0                        \n"
>+      "jne        2f                             \n"
>+      "test       $0xf,%1                        \n"
>+      "jne        2f                             \n"
>+
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqa    (%0),%%xmm0                     \n"
>+      "movdqa    0x10(%0),%%xmm1                 \n"
>+      "lea       0x20(%0),%0                     \n"
>+      "movdqa    %%xmm0,(%1)                     \n"
>+      "movdqa    %%xmm1,0x10(%1)                 \n"
>+      "lea       0x20(%1),%1                     \n"
>+      "sub       $0x20,%2                        \n"
>+      "jg        1b                              \n"
>+      "jmp       9f                              \n"
>+
>+      LABELALIGN
>+      "2:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "movdqu    0x10(%0),%%xmm1                 \n"
>+      "lea       0x20(%0),%0                     \n"
>+      "movdqu    %%xmm0,(%1)                     \n"
>+      "movdqu    %%xmm1,0x10(%1)                 \n"
>+      "lea       0x20(%1),%1                     \n"
>+      "sub       $0x20,%2                        \n"
>+      "jg        2b                              \n"
>+
>+      LABELALIGN "9:                             \n"
>+      : "+r"(src),   // %0
>+        "+r"(dst),   // %1
>+        "+r"(width)  // %2
>+      :
>+      : "memory", "cc", "xmm0", "xmm1");
> }
> #endif  // HAS_COPYROW_SSE2
> 
> #ifdef HAS_COPYROW_AVX
>-void CopyRow_AVX(const uint8* src, uint8* dst, int count) {
>-  asm volatile (
>-    LABELALIGN
>-    "1:                                        \n"
>-    "vmovdqu   " MEMACCESS(0) ",%%ymm0         \n"
>-    "vmovdqu   " MEMACCESS2(0x20,0) ",%%ymm1   \n"
>-    "lea       " MEMLEA(0x40,0) ",%0           \n"
>-    "vmovdqu   %%ymm0," MEMACCESS(1) "         \n"
>-    "vmovdqu   %%ymm1," MEMACCESS2(0x20,1) "   \n"
>-    "lea       " MEMLEA(0x40,1) ",%1           \n"
>-    "sub       $0x40,%2                        \n"
>-    "jg        1b                              \n"
>-  : "+r"(src),   // %0
>-    "+r"(dst),   // %1
>-    "+r"(count)  // %2
>-  :
>-  : "memory", "cc"
>-    , "xmm0", "xmm1"
>-  );
>+void CopyRow_AVX(const uint8_t* src, uint8_t* dst, int width) {
>+  asm volatile(
>+
>+      LABELALIGN
>+      "1:                                        \n"
>+      "vmovdqu   (%0),%%ymm0                     \n"
>+      "vmovdqu   0x20(%0),%%ymm1                 \n"
>+      "lea       0x40(%0),%0                     \n"
>+      "vmovdqu   %%ymm0,(%1)                     \n"
>+      "vmovdqu   %%ymm1,0x20(%1)                 \n"
>+      "lea       0x40(%1),%1                     \n"
>+      "sub       $0x40,%2                        \n"
>+      "jg        1b                              \n"
>+      : "+r"(src),   // %0
>+        "+r"(dst),   // %1
>+        "+r"(width)  // %2
>+      :
>+      : "memory", "cc", "xmm0", "xmm1");
> }
> #endif  // HAS_COPYROW_AVX
> 
> #ifdef HAS_COPYROW_ERMS
> // Multiple of 1.
>-void CopyRow_ERMS(const uint8* src, uint8* dst, int width) {
>+void CopyRow_ERMS(const uint8_t* src, uint8_t* dst, int width) {
>   size_t width_tmp = (size_t)(width);
>-  asm volatile("rep movsb " MEMMOVESTRING(0, 1) "          \n"
>-               : "+S"(src),       // %0
>-                 "+D"(dst),       // %1
>-                 "+c"(width_tmp)  // %2
>-               :
>-               : "memory", "cc");
>+  asm volatile(
>+
>+      "rep movsb                      \n"
>+      : "+S"(src),       // %0
>+        "+D"(dst),       // %1
>+        "+c"(width_tmp)  // %2
>+      :
>+      : "memory", "cc");
> }
> #endif  // HAS_COPYROW_ERMS
> 
> #ifdef HAS_ARGBCOPYALPHAROW_SSE2
> // width in pixels
>-void ARGBCopyAlphaRow_SSE2(const uint8* src, uint8* dst, int width) {
>-  asm volatile (
>-    "pcmpeqb   %%xmm0,%%xmm0                   \n"
>-    "pslld     $0x18,%%xmm0                    \n"
>-    "pcmpeqb   %%xmm1,%%xmm1                   \n"
>-    "psrld     $0x8,%%xmm1                     \n"
>+void ARGBCopyAlphaRow_SSE2(const uint8_t* src, uint8_t* dst, int width) {
>+  asm volatile(
>+      "pcmpeqb   %%xmm0,%%xmm0                   \n"
>+      "pslld     $0x18,%%xmm0                    \n"
>+      "pcmpeqb   %%xmm1,%%xmm1                   \n"
>+      "psrld     $0x8,%%xmm1                     \n"
> 
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm2         \n"
>-    "movdqu    " MEMACCESS2(0x10,0) ",%%xmm3   \n"
>-    "lea       " MEMLEA(0x20,0) ",%0           \n"
>-    "movdqu    " MEMACCESS(1) ",%%xmm4         \n"
>-    "movdqu    " MEMACCESS2(0x10,1) ",%%xmm5   \n"
>-    "pand      %%xmm0,%%xmm2                   \n"
>-    "pand      %%xmm0,%%xmm3                   \n"
>-    "pand      %%xmm1,%%xmm4                   \n"
>-    "pand      %%xmm1,%%xmm5                   \n"
>-    "por       %%xmm4,%%xmm2                   \n"
>-    "por       %%xmm5,%%xmm3                   \n"
>-    "movdqu    %%xmm2," MEMACCESS(1) "         \n"
>-    "movdqu    %%xmm3," MEMACCESS2(0x10,1) "   \n"
>-    "lea       " MEMLEA(0x20,1) ",%1           \n"
>-    "sub       $0x8,%2                         \n"
>-    "jg        1b                              \n"
>-  : "+r"(src),   // %0
>-    "+r"(dst),   // %1
>-    "+r"(width)  // %2
>-  :
>-  : "memory", "cc"
>-    , "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5"
>-  );
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm2                     \n"
>+      "movdqu    0x10(%0),%%xmm3                 \n"
>+      "lea       0x20(%0),%0                     \n"
>+      "movdqu    (%1),%%xmm4                     \n"
>+      "movdqu    0x10(%1),%%xmm5                 \n"
>+      "pand      %%xmm0,%%xmm2                   \n"
>+      "pand      %%xmm0,%%xmm3                   \n"
>+      "pand      %%xmm1,%%xmm4                   \n"
>+      "pand      %%xmm1,%%xmm5                   \n"
>+      "por       %%xmm4,%%xmm2                   \n"
>+      "por       %%xmm5,%%xmm3                   \n"
>+      "movdqu    %%xmm2,(%1)                     \n"
>+      "movdqu    %%xmm3,0x10(%1)                 \n"
>+      "lea       0x20(%1),%1                     \n"
>+      "sub       $0x8,%2                         \n"
>+      "jg        1b                              \n"
>+      : "+r"(src),   // %0
>+        "+r"(dst),   // %1
>+        "+r"(width)  // %2
>+      :
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5");
> }
> #endif  // HAS_ARGBCOPYALPHAROW_SSE2
> 
> #ifdef HAS_ARGBCOPYALPHAROW_AVX2
> // width in pixels
>-void ARGBCopyAlphaRow_AVX2(const uint8* src, uint8* dst, int width) {
>-  asm volatile (
>-    "vpcmpeqb  %%ymm0,%%ymm0,%%ymm0            \n"
>-    "vpsrld    $0x8,%%ymm0,%%ymm0              \n"
>+void ARGBCopyAlphaRow_AVX2(const uint8_t* src, uint8_t* dst, int width) {
>+  asm volatile(
>+      "vpcmpeqb  %%ymm0,%%ymm0,%%ymm0            \n"
>+      "vpsrld    $0x8,%%ymm0,%%ymm0              \n"
> 
>-    LABELALIGN
>-    "1:                                        \n"
>-    "vmovdqu   " MEMACCESS(0) ",%%ymm1         \n"
>-    "vmovdqu   " MEMACCESS2(0x20,0) ",%%ymm2   \n"
>-    "lea       " MEMLEA(0x40,0) ",%0           \n"
>-    "vpblendvb %%ymm0," MEMACCESS(1) ",%%ymm1,%%ymm1        \n"
>-    "vpblendvb %%ymm0," MEMACCESS2(0x20,1) ",%%ymm2,%%ymm2  \n"
>-    "vmovdqu   %%ymm1," MEMACCESS(1) "         \n"
>-    "vmovdqu   %%ymm2," MEMACCESS2(0x20,1) "   \n"
>-    "lea       " MEMLEA(0x40,1) ",%1           \n"
>-    "sub       $0x10,%2                        \n"
>-    "jg        1b                              \n"
>-    "vzeroupper                                \n"
>-  : "+r"(src),   // %0
>-    "+r"(dst),   // %1
>-    "+r"(width)  // %2
>-  :
>-  : "memory", "cc"
>-    , "xmm0", "xmm1", "xmm2"
>-  );
>+      LABELALIGN
>+      "1:                                        \n"
>+      "vmovdqu   (%0),%%ymm1                     \n"
>+      "vmovdqu   0x20(%0),%%ymm2                 \n"
>+      "lea       0x40(%0),%0                     \n"
>+      "vpblendvb %%ymm0,(%1),%%ymm1,%%ymm1       \n"
>+      "vpblendvb %%ymm0,0x20(%1),%%ymm2,%%ymm2   \n"
>+      "vmovdqu   %%ymm1,(%1)                     \n"
>+      "vmovdqu   %%ymm2,0x20(%1)                 \n"
>+      "lea       0x40(%1),%1                     \n"
>+      "sub       $0x10,%2                        \n"
>+      "jg        1b                              \n"
>+      "vzeroupper                                \n"
>+      : "+r"(src),   // %0
>+        "+r"(dst),   // %1
>+        "+r"(width)  // %2
>+      :
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2");
> }
> #endif  // HAS_ARGBCOPYALPHAROW_AVX2
> 
> #ifdef HAS_ARGBEXTRACTALPHAROW_SSE2
> // width in pixels
>-void ARGBExtractAlphaRow_SSE2(const uint8* src_argb, uint8* dst_a, int width) {
>-  asm volatile (
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movdqu    " MEMACCESS(0) ", %%xmm0        \n"
>-    "movdqu    " MEMACCESS2(0x10, 0) ", %%xmm1 \n"
>-    "lea       " MEMLEA(0x20, 0) ", %0         \n"
>-    "psrld     $0x18, %%xmm0                   \n"
>-    "psrld     $0x18, %%xmm1                   \n"
>-    "packssdw  %%xmm1, %%xmm0                  \n"
>-    "packuswb  %%xmm0, %%xmm0                  \n"
>-    "movq      %%xmm0," MEMACCESS(1) "         \n"
>-    "lea       " MEMLEA(0x8, 1) ", %1          \n"
>-    "sub       $0x8, %2                        \n"
>-    "jg        1b                              \n"
>-  : "+r"(src_argb),  // %0
>-    "+r"(dst_a),     // %1
>-    "+rm"(width)     // %2
>-  :
>-  : "memory", "cc"
>-    , "xmm0", "xmm1"
>-  );
>+void ARGBExtractAlphaRow_SSE2(const uint8_t* src_argb,
>+                              uint8_t* dst_a,
>+                              int width) {
>+  asm volatile(
>+
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0), %%xmm0                    \n"
>+      "movdqu    0x10(%0), %%xmm1                \n"
>+      "lea       0x20(%0), %0                    \n"
>+      "psrld     $0x18, %%xmm0                   \n"
>+      "psrld     $0x18, %%xmm1                   \n"
>+      "packssdw  %%xmm1, %%xmm0                  \n"
>+      "packuswb  %%xmm0, %%xmm0                  \n"
>+      "movq      %%xmm0,(%1)                     \n"
>+      "lea       0x8(%1), %1                     \n"
>+      "sub       $0x8, %2                        \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_argb),  // %0
>+        "+r"(dst_a),     // %1
>+        "+rm"(width)     // %2
>+      :
>+      : "memory", "cc", "xmm0", "xmm1");
> }
> #endif  // HAS_ARGBEXTRACTALPHAROW_SSE2
> 
>@@ -3203,657 +3798,636 @@ static const uvec8 kShuffleAlphaShort_AVX2 = {
>     3u,  128u, 128u, 128u, 7u,  128u, 128u, 128u,
>     11u, 128u, 128u, 128u, 15u, 128u, 128u, 128u};
> 
>-void ARGBExtractAlphaRow_AVX2(const uint8* src_argb, uint8* dst_a, int width) {
>-  asm volatile (
>-    "vmovdqa    %3,%%ymm4                      \n"
>-    "vbroadcastf128 %4,%%ymm5                  \n"
>+void ARGBExtractAlphaRow_AVX2(const uint8_t* src_argb,
>+                              uint8_t* dst_a,
>+                              int width) {
>+  asm volatile(
>+      "vmovdqa    %3,%%ymm4                      \n"
>+      "vbroadcastf128 %4,%%ymm5                  \n"
> 
>-    LABELALIGN
>-    "1:                                        \n"
>-    "vmovdqu   " MEMACCESS(0) ", %%ymm0        \n"
>-    "vmovdqu   " MEMACCESS2(0x20, 0) ", %%ymm1 \n"
>-    "vpshufb    %%ymm5,%%ymm0,%%ymm0           \n" // vpsrld $0x18, %%ymm0
>-    "vpshufb    %%ymm5,%%ymm1,%%ymm1           \n"
>-    "vmovdqu   " MEMACCESS2(0x40, 0) ", %%ymm2 \n"
>-    "vmovdqu   " MEMACCESS2(0x60, 0) ", %%ymm3 \n"
>-    "lea       " MEMLEA(0x80, 0) ", %0         \n"
>-    "vpackssdw  %%ymm1, %%ymm0, %%ymm0         \n"  // mutates
>-    "vpshufb    %%ymm5,%%ymm2,%%ymm2           \n"
>-    "vpshufb    %%ymm5,%%ymm3,%%ymm3           \n"
>-    "vpackssdw  %%ymm3, %%ymm2, %%ymm2         \n"  // mutates
>-    "vpackuswb  %%ymm2,%%ymm0,%%ymm0           \n"  // mutates.
>-    "vpermd     %%ymm0,%%ymm4,%%ymm0           \n"  // unmutate.
>-    "vmovdqu    %%ymm0," MEMACCESS(1) "        \n"
>-    "lea       " MEMLEA(0x20,1) ",%1           \n"
>-    "sub        $0x20, %2                      \n"
>-    "jg         1b                             \n"
>-    "vzeroupper                                \n"
>-  : "+r"(src_argb),  // %0
>-    "+r"(dst_a),     // %1
>-    "+rm"(width)     // %2
>-  : "m"(kPermdARGBToY_AVX),  // %3
>-    "m"(kShuffleAlphaShort_AVX2)  // %4
>-  : "memory", "cc"
>-    , "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5"
>-  );
>+      LABELALIGN
>+      "1:                                        \n"
>+      "vmovdqu   (%0), %%ymm0                    \n"
>+      "vmovdqu   0x20(%0), %%ymm1                \n"
>+      "vpshufb    %%ymm5,%%ymm0,%%ymm0           \n"  // vpsrld $0x18, %%ymm0
>+      "vpshufb    %%ymm5,%%ymm1,%%ymm1           \n"
>+      "vmovdqu   0x40(%0), %%ymm2                \n"
>+      "vmovdqu   0x60(%0), %%ymm3                \n"
>+      "lea       0x80(%0), %0                    \n"
>+      "vpackssdw  %%ymm1, %%ymm0, %%ymm0         \n"  // mutates
>+      "vpshufb    %%ymm5,%%ymm2,%%ymm2           \n"
>+      "vpshufb    %%ymm5,%%ymm3,%%ymm3           \n"
>+      "vpackssdw  %%ymm3, %%ymm2, %%ymm2         \n"  // mutates
>+      "vpackuswb  %%ymm2,%%ymm0,%%ymm0           \n"  // mutates.
>+      "vpermd     %%ymm0,%%ymm4,%%ymm0           \n"  // unmutate.
>+      "vmovdqu    %%ymm0,(%1)                    \n"
>+      "lea       0x20(%1),%1                     \n"
>+      "sub        $0x20, %2                      \n"
>+      "jg         1b                             \n"
>+      "vzeroupper                                \n"
>+      : "+r"(src_argb),               // %0
>+        "+r"(dst_a),                  // %1
>+        "+rm"(width)                  // %2
>+      : "m"(kPermdARGBToY_AVX),       // %3
>+        "m"(kShuffleAlphaShort_AVX2)  // %4
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5");
> }
> #endif  // HAS_ARGBEXTRACTALPHAROW_AVX2
> 
> #ifdef HAS_ARGBCOPYYTOALPHAROW_SSE2
> // width in pixels
>-void ARGBCopyYToAlphaRow_SSE2(const uint8* src, uint8* dst, int width) {
>-  asm volatile (
>-    "pcmpeqb   %%xmm0,%%xmm0                   \n"
>-    "pslld     $0x18,%%xmm0                    \n"
>-    "pcmpeqb   %%xmm1,%%xmm1                   \n"
>-    "psrld     $0x8,%%xmm1                     \n"
>+void ARGBCopyYToAlphaRow_SSE2(const uint8_t* src, uint8_t* dst, int width) {
>+  asm volatile(
>+      "pcmpeqb   %%xmm0,%%xmm0                   \n"
>+      "pslld     $0x18,%%xmm0                    \n"
>+      "pcmpeqb   %%xmm1,%%xmm1                   \n"
>+      "psrld     $0x8,%%xmm1                     \n"
> 
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movq      " MEMACCESS(0) ",%%xmm2         \n"
>-    "lea       " MEMLEA(0x8,0) ",%0            \n"
>-    "punpcklbw %%xmm2,%%xmm2                   \n"
>-    "punpckhwd %%xmm2,%%xmm3                   \n"
>-    "punpcklwd %%xmm2,%%xmm2                   \n"
>-    "movdqu    " MEMACCESS(1) ",%%xmm4         \n"
>-    "movdqu    " MEMACCESS2(0x10,1) ",%%xmm5   \n"
>-    "pand      %%xmm0,%%xmm2                   \n"
>-    "pand      %%xmm0,%%xmm3                   \n"
>-    "pand      %%xmm1,%%xmm4                   \n"
>-    "pand      %%xmm1,%%xmm5                   \n"
>-    "por       %%xmm4,%%xmm2                   \n"
>-    "por       %%xmm5,%%xmm3                   \n"
>-    "movdqu    %%xmm2," MEMACCESS(1) "         \n"
>-    "movdqu    %%xmm3," MEMACCESS2(0x10,1) "   \n"
>-    "lea       " MEMLEA(0x20,1) ",%1           \n"
>-    "sub       $0x8,%2                         \n"
>-    "jg        1b                              \n"
>-  : "+r"(src),   // %0
>-    "+r"(dst),   // %1
>-    "+r"(width)  // %2
>-  :
>-  : "memory", "cc"
>-    , "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5"
>-  );
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movq      (%0),%%xmm2                     \n"
>+      "lea       0x8(%0),%0                      \n"
>+      "punpcklbw %%xmm2,%%xmm2                   \n"
>+      "punpckhwd %%xmm2,%%xmm3                   \n"
>+      "punpcklwd %%xmm2,%%xmm2                   \n"
>+      "movdqu    (%1),%%xmm4                     \n"
>+      "movdqu    0x10(%1),%%xmm5                 \n"
>+      "pand      %%xmm0,%%xmm2                   \n"
>+      "pand      %%xmm0,%%xmm3                   \n"
>+      "pand      %%xmm1,%%xmm4                   \n"
>+      "pand      %%xmm1,%%xmm5                   \n"
>+      "por       %%xmm4,%%xmm2                   \n"
>+      "por       %%xmm5,%%xmm3                   \n"
>+      "movdqu    %%xmm2,(%1)                     \n"
>+      "movdqu    %%xmm3,0x10(%1)                 \n"
>+      "lea       0x20(%1),%1                     \n"
>+      "sub       $0x8,%2                         \n"
>+      "jg        1b                              \n"
>+      : "+r"(src),   // %0
>+        "+r"(dst),   // %1
>+        "+r"(width)  // %2
>+      :
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5");
> }
> #endif  // HAS_ARGBCOPYYTOALPHAROW_SSE2
> 
> #ifdef HAS_ARGBCOPYYTOALPHAROW_AVX2
> // width in pixels
>-void ARGBCopyYToAlphaRow_AVX2(const uint8* src, uint8* dst, int width) {
>-  asm volatile (
>-    "vpcmpeqb  %%ymm0,%%ymm0,%%ymm0            \n"
>-    "vpsrld    $0x8,%%ymm0,%%ymm0              \n"
>+void ARGBCopyYToAlphaRow_AVX2(const uint8_t* src, uint8_t* dst, int width) {
>+  asm volatile(
>+      "vpcmpeqb  %%ymm0,%%ymm0,%%ymm0            \n"
>+      "vpsrld    $0x8,%%ymm0,%%ymm0              \n"
> 
>-    LABELALIGN
>-    "1:                                        \n"
>-    "vpmovzxbd " MEMACCESS(0) ",%%ymm1         \n"
>-    "vpmovzxbd " MEMACCESS2(0x8,0) ",%%ymm2    \n"
>-    "lea       " MEMLEA(0x10,0) ",%0           \n"
>-    "vpslld    $0x18,%%ymm1,%%ymm1             \n"
>-    "vpslld    $0x18,%%ymm2,%%ymm2             \n"
>-    "vpblendvb %%ymm0," MEMACCESS(1) ",%%ymm1,%%ymm1        \n"
>-    "vpblendvb %%ymm0," MEMACCESS2(0x20,1) ",%%ymm2,%%ymm2  \n"
>-    "vmovdqu   %%ymm1," MEMACCESS(1) "         \n"
>-    "vmovdqu   %%ymm2," MEMACCESS2(0x20,1) "   \n"
>-    "lea       " MEMLEA(0x40,1) ",%1           \n"
>-    "sub       $0x10,%2                        \n"
>-    "jg        1b                              \n"
>-    "vzeroupper                                \n"
>-  : "+r"(src),   // %0
>-    "+r"(dst),   // %1
>-    "+r"(width)  // %2
>-  :
>-  : "memory", "cc"
>-    , "xmm0", "xmm1", "xmm2"
>-  );
>+      LABELALIGN
>+      "1:                                        \n"
>+      "vpmovzxbd (%0),%%ymm1                     \n"
>+      "vpmovzxbd 0x8(%0),%%ymm2                  \n"
>+      "lea       0x10(%0),%0                     \n"
>+      "vpslld    $0x18,%%ymm1,%%ymm1             \n"
>+      "vpslld    $0x18,%%ymm2,%%ymm2             \n"
>+      "vpblendvb %%ymm0,(%1),%%ymm1,%%ymm1       \n"
>+      "vpblendvb %%ymm0,0x20(%1),%%ymm2,%%ymm2   \n"
>+      "vmovdqu   %%ymm1,(%1)                     \n"
>+      "vmovdqu   %%ymm2,0x20(%1)                 \n"
>+      "lea       0x40(%1),%1                     \n"
>+      "sub       $0x10,%2                        \n"
>+      "jg        1b                              \n"
>+      "vzeroupper                                \n"
>+      : "+r"(src),   // %0
>+        "+r"(dst),   // %1
>+        "+r"(width)  // %2
>+      :
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2");
> }
> #endif  // HAS_ARGBCOPYYTOALPHAROW_AVX2
> 
> #ifdef HAS_SETROW_X86
>-void SetRow_X86(uint8* dst, uint8 v8, int width) {
>+void SetRow_X86(uint8_t* dst, uint8_t v8, int width) {
>   size_t width_tmp = (size_t)(width >> 2);
>-  const uint32 v32 = v8 * 0x01010101u;  // Duplicate byte to all bytes.
>-  asm volatile("rep stosl " MEMSTORESTRING(eax, 0) "       \n"
>-               : "+D"(dst),       // %0
>-                 "+c"(width_tmp)  // %1
>-               : "a"(v32)         // %2
>-               : "memory", "cc");
>+  const uint32_t v32 = v8 * 0x01010101u;  // Duplicate byte to all bytes.
>+  asm volatile(
>+
>+      "rep stosl                      \n"
>+      : "+D"(dst),       // %0
>+        "+c"(width_tmp)  // %1
>+      : "a"(v32)         // %2
>+      : "memory", "cc");
> }
> 
>-void SetRow_ERMS(uint8* dst, uint8 v8, int width) {
>+void SetRow_ERMS(uint8_t* dst, uint8_t v8, int width) {
>   size_t width_tmp = (size_t)(width);
>-  asm volatile("rep stosb " MEMSTORESTRING(al, 0) "        \n"
>-               : "+D"(dst),       // %0
>-                 "+c"(width_tmp)  // %1
>-               : "a"(v8)          // %2
>-               : "memory", "cc");
>+  asm volatile(
>+
>+      "rep stosb                      \n"
>+      : "+D"(dst),       // %0
>+        "+c"(width_tmp)  // %1
>+      : "a"(v8)          // %2
>+      : "memory", "cc");
> }
> 
>-void ARGBSetRow_X86(uint8* dst_argb, uint32 v32, int width) {
>+void ARGBSetRow_X86(uint8_t* dst_argb, uint32_t v32, int width) {
>   size_t width_tmp = (size_t)(width);
>-  asm volatile("rep stosl " MEMSTORESTRING(eax, 0) "       \n"
>-               : "+D"(dst_argb),  // %0
>-                 "+c"(width_tmp)  // %1
>-               : "a"(v32)         // %2
>-               : "memory", "cc");
>+  asm volatile(
>+
>+      "rep stosl                      \n"
>+      : "+D"(dst_argb),  // %0
>+        "+c"(width_tmp)  // %1
>+      : "a"(v32)         // %2
>+      : "memory", "cc");
> }
> #endif  // HAS_SETROW_X86
> 
> #ifdef HAS_YUY2TOYROW_SSE2
>-void YUY2ToYRow_SSE2(const uint8* src_yuy2, uint8* dst_y, int width) {
>-  asm volatile (
>-    "pcmpeqb   %%xmm5,%%xmm5                   \n"
>-    "psrlw     $0x8,%%xmm5                     \n"
>+void YUY2ToYRow_SSE2(const uint8_t* src_yuy2, uint8_t* dst_y, int width) {
>+  asm volatile(
>+      "pcmpeqb   %%xmm5,%%xmm5                   \n"
>+      "psrlw     $0x8,%%xmm5                     \n"
> 
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    "movdqu    " MEMACCESS2(0x10,0) ",%%xmm1   \n"
>-    "lea       " MEMLEA(0x20,0) ",%0           \n"
>-    "pand      %%xmm5,%%xmm0                   \n"
>-    "pand      %%xmm5,%%xmm1                   \n"
>-    "packuswb  %%xmm1,%%xmm0                   \n"
>-    "movdqu    %%xmm0," MEMACCESS(1) "         \n"
>-    "lea       " MEMLEA(0x10,1) ",%1           \n"
>-    "sub       $0x10,%2                        \n"
>-    "jg        1b                              \n"
>-  : "+r"(src_yuy2),  // %0
>-    "+r"(dst_y),     // %1
>-    "+r"(width)        // %2
>-  :
>-  : "memory", "cc"
>-    , "xmm0", "xmm1", "xmm5"
>-  );
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "movdqu    0x10(%0),%%xmm1                 \n"
>+      "lea       0x20(%0),%0                     \n"
>+      "pand      %%xmm5,%%xmm0                   \n"
>+      "pand      %%xmm5,%%xmm1                   \n"
>+      "packuswb  %%xmm1,%%xmm0                   \n"
>+      "movdqu    %%xmm0,(%1)                     \n"
>+      "lea       0x10(%1),%1                     \n"
>+      "sub       $0x10,%2                        \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_yuy2),  // %0
>+        "+r"(dst_y),     // %1
>+        "+r"(width)      // %2
>+      :
>+      : "memory", "cc", "xmm0", "xmm1", "xmm5");
> }
> 
>-void YUY2ToUVRow_SSE2(const uint8* src_yuy2,
>+void YUY2ToUVRow_SSE2(const uint8_t* src_yuy2,
>                       int stride_yuy2,
>-                      uint8* dst_u,
>-                      uint8* dst_v,
>+                      uint8_t* dst_u,
>+                      uint8_t* dst_v,
>                       int width) {
>-  asm volatile (
>-    "pcmpeqb   %%xmm5,%%xmm5                   \n"
>-    "psrlw     $0x8,%%xmm5                     \n"
>-    "sub       %1,%2                           \n"
>+  asm volatile(
>+      "pcmpeqb   %%xmm5,%%xmm5                   \n"
>+      "psrlw     $0x8,%%xmm5                     \n"
>+      "sub       %1,%2                           \n"
> 
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    "movdqu    " MEMACCESS2(0x10,0) ",%%xmm1   \n"
>-    MEMOPREG(movdqu,0x00,0,4,1,xmm2)           //  movdqu  (%0,%4,1),%%xmm2
>-    MEMOPREG(movdqu,0x10,0,4,1,xmm3)           //  movdqu  0x10(%0,%4,1),%%xmm3
>-    "lea       " MEMLEA(0x20,0) ",%0           \n"
>-    "pavgb     %%xmm2,%%xmm0                   \n"
>-    "pavgb     %%xmm3,%%xmm1                   \n"
>-    "psrlw     $0x8,%%xmm0                     \n"
>-    "psrlw     $0x8,%%xmm1                     \n"
>-    "packuswb  %%xmm1,%%xmm0                   \n"
>-    "movdqa    %%xmm0,%%xmm1                   \n"
>-    "pand      %%xmm5,%%xmm0                   \n"
>-    "packuswb  %%xmm0,%%xmm0                   \n"
>-    "psrlw     $0x8,%%xmm1                     \n"
>-    "packuswb  %%xmm1,%%xmm1                   \n"
>-    "movq      %%xmm0," MEMACCESS(1) "         \n"
>-    MEMOPMEM(movq,xmm1,0x00,1,2,1)             //  movq    %%xmm1,(%1,%2)
>-    "lea       " MEMLEA(0x8,1) ",%1            \n"
>-    "sub       $0x10,%3                        \n"
>-    "jg        1b                              \n"
>-  : "+r"(src_yuy2),    // %0
>-    "+r"(dst_u),       // %1
>-    "+r"(dst_v),       // %2
>-    "+r"(width)          // %3
>-  : "r"((intptr_t)(stride_yuy2))  // %4
>-  : "memory", "cc", NACL_R14
>-    "xmm0", "xmm1", "xmm2", "xmm3", "xmm5"
>-  );
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "movdqu    0x10(%0),%%xmm1                 \n"
>+      "movdqu    0x00(%0,%4,1),%%xmm2            \n"
>+      "movdqu    0x10(%0,%4,1),%%xmm3            \n"
>+      "lea       0x20(%0),%0                     \n"
>+      "pavgb     %%xmm2,%%xmm0                   \n"
>+      "pavgb     %%xmm3,%%xmm1                   \n"
>+      "psrlw     $0x8,%%xmm0                     \n"
>+      "psrlw     $0x8,%%xmm1                     \n"
>+      "packuswb  %%xmm1,%%xmm0                   \n"
>+      "movdqa    %%xmm0,%%xmm1                   \n"
>+      "pand      %%xmm5,%%xmm0                   \n"
>+      "packuswb  %%xmm0,%%xmm0                   \n"
>+      "psrlw     $0x8,%%xmm1                     \n"
>+      "packuswb  %%xmm1,%%xmm1                   \n"
>+      "movq      %%xmm0,(%1)                     \n"
>+      "movq    %%xmm1,0x00(%1,%2,1)              \n"
>+      "lea       0x8(%1),%1                      \n"
>+      "sub       $0x10,%3                        \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_yuy2),               // %0
>+        "+r"(dst_u),                  // %1
>+        "+r"(dst_v),                  // %2
>+        "+r"(width)                   // %3
>+      : "r"((intptr_t)(stride_yuy2))  // %4
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm5");
> }
> 
>-void YUY2ToUV422Row_SSE2(const uint8* src_yuy2,
>-                         uint8* dst_u,
>-                         uint8* dst_v,
>+void YUY2ToUV422Row_SSE2(const uint8_t* src_yuy2,
>+                         uint8_t* dst_u,
>+                         uint8_t* dst_v,
>                          int width) {
>-  asm volatile (
>-    "pcmpeqb   %%xmm5,%%xmm5                   \n"
>-    "psrlw     $0x8,%%xmm5                     \n"
>-    "sub       %1,%2                           \n"
>+  asm volatile(
>+      "pcmpeqb   %%xmm5,%%xmm5                   \n"
>+      "psrlw     $0x8,%%xmm5                     \n"
>+      "sub       %1,%2                           \n"
> 
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    "movdqu    " MEMACCESS2(0x10,0) ",%%xmm1   \n"
>-    "lea       " MEMLEA(0x20,0) ",%0           \n"
>-    "psrlw     $0x8,%%xmm0                     \n"
>-    "psrlw     $0x8,%%xmm1                     \n"
>-    "packuswb  %%xmm1,%%xmm0                   \n"
>-    "movdqa    %%xmm0,%%xmm1                   \n"
>-    "pand      %%xmm5,%%xmm0                   \n"
>-    "packuswb  %%xmm0,%%xmm0                   \n"
>-    "psrlw     $0x8,%%xmm1                     \n"
>-    "packuswb  %%xmm1,%%xmm1                   \n"
>-    "movq      %%xmm0," MEMACCESS(1) "         \n"
>-    MEMOPMEM(movq,xmm1,0x00,1,2,1)             //  movq    %%xmm1,(%1,%2)
>-    "lea       " MEMLEA(0x8,1) ",%1            \n"
>-    "sub       $0x10,%3                        \n"
>-    "jg        1b                              \n"
>-  : "+r"(src_yuy2),    // %0
>-    "+r"(dst_u),       // %1
>-    "+r"(dst_v),       // %2
>-    "+r"(width)          // %3
>-  :
>-  : "memory", "cc", NACL_R14
>-    "xmm0", "xmm1", "xmm5"
>-  );
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "movdqu    0x10(%0),%%xmm1                 \n"
>+      "lea       0x20(%0),%0                     \n"
>+      "psrlw     $0x8,%%xmm0                     \n"
>+      "psrlw     $0x8,%%xmm1                     \n"
>+      "packuswb  %%xmm1,%%xmm0                   \n"
>+      "movdqa    %%xmm0,%%xmm1                   \n"
>+      "pand      %%xmm5,%%xmm0                   \n"
>+      "packuswb  %%xmm0,%%xmm0                   \n"
>+      "psrlw     $0x8,%%xmm1                     \n"
>+      "packuswb  %%xmm1,%%xmm1                   \n"
>+      "movq      %%xmm0,(%1)                     \n"
>+      "movq    %%xmm1,0x00(%1,%2,1)              \n"
>+      "lea       0x8(%1),%1                      \n"
>+      "sub       $0x10,%3                        \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_yuy2),  // %0
>+        "+r"(dst_u),     // %1
>+        "+r"(dst_v),     // %2
>+        "+r"(width)      // %3
>+      :
>+      : "memory", "cc", "xmm0", "xmm1", "xmm5");
> }
> 
>-void UYVYToYRow_SSE2(const uint8* src_uyvy, uint8* dst_y, int width) {
>-  asm volatile (
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    "movdqu    " MEMACCESS2(0x10,0) ",%%xmm1   \n"
>-    "lea       " MEMLEA(0x20,0) ",%0           \n"
>-    "psrlw     $0x8,%%xmm0                     \n"
>-    "psrlw     $0x8,%%xmm1                     \n"
>-    "packuswb  %%xmm1,%%xmm0                   \n"
>-    "movdqu    %%xmm0," MEMACCESS(1) "         \n"
>-    "lea       " MEMLEA(0x10,1) ",%1           \n"
>-    "sub       $0x10,%2                        \n"
>-    "jg        1b                              \n"
>-  : "+r"(src_uyvy),  // %0
>-    "+r"(dst_y),     // %1
>-    "+r"(width)        // %2
>-  :
>-  : "memory", "cc"
>-    , "xmm0", "xmm1"
>-  );
>+void UYVYToYRow_SSE2(const uint8_t* src_uyvy, uint8_t* dst_y, int width) {
>+  asm volatile(
>+
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "movdqu    0x10(%0),%%xmm1                 \n"
>+      "lea       0x20(%0),%0                     \n"
>+      "psrlw     $0x8,%%xmm0                     \n"
>+      "psrlw     $0x8,%%xmm1                     \n"
>+      "packuswb  %%xmm1,%%xmm0                   \n"
>+      "movdqu    %%xmm0,(%1)                     \n"
>+      "lea       0x10(%1),%1                     \n"
>+      "sub       $0x10,%2                        \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_uyvy),  // %0
>+        "+r"(dst_y),     // %1
>+        "+r"(width)      // %2
>+      :
>+      : "memory", "cc", "xmm0", "xmm1");
> }
> 
>-void UYVYToUVRow_SSE2(const uint8* src_uyvy,
>+void UYVYToUVRow_SSE2(const uint8_t* src_uyvy,
>                       int stride_uyvy,
>-                      uint8* dst_u,
>-                      uint8* dst_v,
>+                      uint8_t* dst_u,
>+                      uint8_t* dst_v,
>                       int width) {
>-  asm volatile (
>-    "pcmpeqb   %%xmm5,%%xmm5                   \n"
>-    "psrlw     $0x8,%%xmm5                     \n"
>-    "sub       %1,%2                           \n"
>+  asm volatile(
>+      "pcmpeqb   %%xmm5,%%xmm5                   \n"
>+      "psrlw     $0x8,%%xmm5                     \n"
>+      "sub       %1,%2                           \n"
> 
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    "movdqu    " MEMACCESS2(0x10,0) ",%%xmm1   \n"
>-    MEMOPREG(movdqu,0x00,0,4,1,xmm2)           //  movdqu  (%0,%4,1),%%xmm2
>-    MEMOPREG(movdqu,0x10,0,4,1,xmm3)           //  movdqu  0x10(%0,%4,1),%%xmm3
>-    "lea       " MEMLEA(0x20,0) ",%0           \n"
>-    "pavgb     %%xmm2,%%xmm0                   \n"
>-    "pavgb     %%xmm3,%%xmm1                   \n"
>-    "pand      %%xmm5,%%xmm0                   \n"
>-    "pand      %%xmm5,%%xmm1                   \n"
>-    "packuswb  %%xmm1,%%xmm0                   \n"
>-    "movdqa    %%xmm0,%%xmm1                   \n"
>-    "pand      %%xmm5,%%xmm0                   \n"
>-    "packuswb  %%xmm0,%%xmm0                   \n"
>-    "psrlw     $0x8,%%xmm1                     \n"
>-    "packuswb  %%xmm1,%%xmm1                   \n"
>-    "movq      %%xmm0," MEMACCESS(1) "         \n"
>-    MEMOPMEM(movq,xmm1,0x00,1,2,1)             //  movq    %%xmm1,(%1,%2)
>-    "lea       " MEMLEA(0x8,1) ",%1            \n"
>-    "sub       $0x10,%3                        \n"
>-    "jg        1b                              \n"
>-  : "+r"(src_uyvy),    // %0
>-    "+r"(dst_u),       // %1
>-    "+r"(dst_v),       // %2
>-    "+r"(width)          // %3
>-  : "r"((intptr_t)(stride_uyvy))  // %4
>-  : "memory", "cc", NACL_R14
>-    "xmm0", "xmm1", "xmm2", "xmm3", "xmm5"
>-  );
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "movdqu    0x10(%0),%%xmm1                 \n"
>+      "movdqu    0x00(%0,%4,1),%%xmm2            \n"
>+      "movdqu    0x10(%0,%4,1),%%xmm3            \n"
>+      "lea       0x20(%0),%0                     \n"
>+      "pavgb     %%xmm2,%%xmm0                   \n"
>+      "pavgb     %%xmm3,%%xmm1                   \n"
>+      "pand      %%xmm5,%%xmm0                   \n"
>+      "pand      %%xmm5,%%xmm1                   \n"
>+      "packuswb  %%xmm1,%%xmm0                   \n"
>+      "movdqa    %%xmm0,%%xmm1                   \n"
>+      "pand      %%xmm5,%%xmm0                   \n"
>+      "packuswb  %%xmm0,%%xmm0                   \n"
>+      "psrlw     $0x8,%%xmm1                     \n"
>+      "packuswb  %%xmm1,%%xmm1                   \n"
>+      "movq      %%xmm0,(%1)                     \n"
>+      "movq    %%xmm1,0x00(%1,%2,1)              \n"
>+      "lea       0x8(%1),%1                      \n"
>+      "sub       $0x10,%3                        \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_uyvy),               // %0
>+        "+r"(dst_u),                  // %1
>+        "+r"(dst_v),                  // %2
>+        "+r"(width)                   // %3
>+      : "r"((intptr_t)(stride_uyvy))  // %4
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm5");
> }
> 
>-void UYVYToUV422Row_SSE2(const uint8* src_uyvy,
>-                         uint8* dst_u,
>-                         uint8* dst_v,
>+void UYVYToUV422Row_SSE2(const uint8_t* src_uyvy,
>+                         uint8_t* dst_u,
>+                         uint8_t* dst_v,
>                          int width) {
>-  asm volatile (
>-    "pcmpeqb   %%xmm5,%%xmm5                   \n"
>-    "psrlw     $0x8,%%xmm5                     \n"
>-    "sub       %1,%2                           \n"
>+  asm volatile(
>+      "pcmpeqb   %%xmm5,%%xmm5                   \n"
>+      "psrlw     $0x8,%%xmm5                     \n"
>+      "sub       %1,%2                           \n"
> 
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    "movdqu    " MEMACCESS2(0x10,0) ",%%xmm1   \n"
>-    "lea       " MEMLEA(0x20,0) ",%0           \n"
>-    "pand      %%xmm5,%%xmm0                   \n"
>-    "pand      %%xmm5,%%xmm1                   \n"
>-    "packuswb  %%xmm1,%%xmm0                   \n"
>-    "movdqa    %%xmm0,%%xmm1                   \n"
>-    "pand      %%xmm5,%%xmm0                   \n"
>-    "packuswb  %%xmm0,%%xmm0                   \n"
>-    "psrlw     $0x8,%%xmm1                     \n"
>-    "packuswb  %%xmm1,%%xmm1                   \n"
>-    "movq      %%xmm0," MEMACCESS(1) "         \n"
>-    MEMOPMEM(movq,xmm1,0x00,1,2,1)             //  movq    %%xmm1,(%1,%2)
>-    "lea       " MEMLEA(0x8,1) ",%1            \n"
>-    "sub       $0x10,%3                        \n"
>-    "jg        1b                              \n"
>-  : "+r"(src_uyvy),    // %0
>-    "+r"(dst_u),       // %1
>-    "+r"(dst_v),       // %2
>-    "+r"(width)          // %3
>-  :
>-  : "memory", "cc", NACL_R14
>-    "xmm0", "xmm1", "xmm5"
>-  );
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "movdqu    0x10(%0),%%xmm1                 \n"
>+      "lea       0x20(%0),%0                     \n"
>+      "pand      %%xmm5,%%xmm0                   \n"
>+      "pand      %%xmm5,%%xmm1                   \n"
>+      "packuswb  %%xmm1,%%xmm0                   \n"
>+      "movdqa    %%xmm0,%%xmm1                   \n"
>+      "pand      %%xmm5,%%xmm0                   \n"
>+      "packuswb  %%xmm0,%%xmm0                   \n"
>+      "psrlw     $0x8,%%xmm1                     \n"
>+      "packuswb  %%xmm1,%%xmm1                   \n"
>+      "movq      %%xmm0,(%1)                     \n"
>+      "movq    %%xmm1,0x00(%1,%2,1)              \n"
>+      "lea       0x8(%1),%1                      \n"
>+      "sub       $0x10,%3                        \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_uyvy),  // %0
>+        "+r"(dst_u),     // %1
>+        "+r"(dst_v),     // %2
>+        "+r"(width)      // %3
>+      :
>+      : "memory", "cc", "xmm0", "xmm1", "xmm5");
> }
> #endif  // HAS_YUY2TOYROW_SSE2
> 
> #ifdef HAS_YUY2TOYROW_AVX2
>-void YUY2ToYRow_AVX2(const uint8* src_yuy2, uint8* dst_y, int width) {
>-  asm volatile (
>-    "vpcmpeqb  %%ymm5,%%ymm5,%%ymm5            \n"
>-    "vpsrlw    $0x8,%%ymm5,%%ymm5              \n"
>+void YUY2ToYRow_AVX2(const uint8_t* src_yuy2, uint8_t* dst_y, int width) {
>+  asm volatile(
>+      "vpcmpeqb  %%ymm5,%%ymm5,%%ymm5            \n"
>+      "vpsrlw    $0x8,%%ymm5,%%ymm5              \n"
> 
>-    LABELALIGN
>-    "1:                                        \n"
>-    "vmovdqu   " MEMACCESS(0) ",%%ymm0         \n"
>-    "vmovdqu   " MEMACCESS2(0x20,0) ",%%ymm1   \n"
>-    "lea       " MEMLEA(0x40,0) ",%0           \n"
>-    "vpand     %%ymm5,%%ymm0,%%ymm0            \n"
>-    "vpand     %%ymm5,%%ymm1,%%ymm1            \n"
>-    "vpackuswb %%ymm1,%%ymm0,%%ymm0            \n"
>-    "vpermq    $0xd8,%%ymm0,%%ymm0             \n"
>-    "vmovdqu   %%ymm0," MEMACCESS(1) "         \n"
>-    "lea      " MEMLEA(0x20,1) ",%1            \n"
>-    "sub       $0x20,%2                        \n"
>-    "jg        1b                              \n"
>-    "vzeroupper                                \n"
>-  : "+r"(src_yuy2),  // %0
>-    "+r"(dst_y),     // %1
>-    "+r"(width)        // %2
>-  :
>-  : "memory", "cc"
>-    , "xmm0", "xmm1", "xmm5"
>-  );
>+      LABELALIGN
>+      "1:                                        \n"
>+      "vmovdqu   (%0),%%ymm0                     \n"
>+      "vmovdqu   0x20(%0),%%ymm1                 \n"
>+      "lea       0x40(%0),%0                     \n"
>+      "vpand     %%ymm5,%%ymm0,%%ymm0            \n"
>+      "vpand     %%ymm5,%%ymm1,%%ymm1            \n"
>+      "vpackuswb %%ymm1,%%ymm0,%%ymm0            \n"
>+      "vpermq    $0xd8,%%ymm0,%%ymm0             \n"
>+      "vmovdqu   %%ymm0,(%1)                     \n"
>+      "lea      0x20(%1),%1                      \n"
>+      "sub       $0x20,%2                        \n"
>+      "jg        1b                              \n"
>+      "vzeroupper                                \n"
>+      : "+r"(src_yuy2),  // %0
>+        "+r"(dst_y),     // %1
>+        "+r"(width)      // %2
>+      :
>+      : "memory", "cc", "xmm0", "xmm1", "xmm5");
> }
> 
>-void YUY2ToUVRow_AVX2(const uint8* src_yuy2,
>+void YUY2ToUVRow_AVX2(const uint8_t* src_yuy2,
>                       int stride_yuy2,
>-                      uint8* dst_u,
>-                      uint8* dst_v,
>+                      uint8_t* dst_u,
>+                      uint8_t* dst_v,
>                       int width) {
>-  asm volatile (
>-    "vpcmpeqb  %%ymm5,%%ymm5,%%ymm5            \n"
>-    "vpsrlw    $0x8,%%ymm5,%%ymm5              \n"
>-    "sub       %1,%2                           \n"
>+  asm volatile(
>+      "vpcmpeqb  %%ymm5,%%ymm5,%%ymm5            \n"
>+      "vpsrlw    $0x8,%%ymm5,%%ymm5              \n"
>+      "sub       %1,%2                           \n"
> 
>-    LABELALIGN
>-    "1:                                        \n"
>-    "vmovdqu   " MEMACCESS(0) ",%%ymm0         \n"
>-    "vmovdqu   " MEMACCESS2(0x20,0) ",%%ymm1   \n"
>-    VMEMOPREG(vpavgb,0x00,0,4,1,ymm0,ymm0)     // vpavgb (%0,%4,1),%%ymm0,%%ymm0
>-    VMEMOPREG(vpavgb,0x20,0,4,1,ymm1,ymm1)
>-    "lea       " MEMLEA(0x40,0) ",%0           \n"
>-    "vpsrlw    $0x8,%%ymm0,%%ymm0              \n"
>-    "vpsrlw    $0x8,%%ymm1,%%ymm1              \n"
>-    "vpackuswb %%ymm1,%%ymm0,%%ymm0            \n"
>-    "vpermq    $0xd8,%%ymm0,%%ymm0             \n"
>-    "vpand     %%ymm5,%%ymm0,%%ymm1            \n"
>-    "vpsrlw    $0x8,%%ymm0,%%ymm0              \n"
>-    "vpackuswb %%ymm1,%%ymm1,%%ymm1            \n"
>-    "vpackuswb %%ymm0,%%ymm0,%%ymm0            \n"
>-    "vpermq    $0xd8,%%ymm1,%%ymm1             \n"
>-    "vpermq    $0xd8,%%ymm0,%%ymm0             \n"
>-    "vextractf128 $0x0,%%ymm1," MEMACCESS(1) " \n"
>-    VEXTOPMEM(vextractf128,0,ymm0,0x00,1,2,1) // vextractf128 $0x0,%%ymm0,(%1,%2,1)
>-    "lea      " MEMLEA(0x10,1) ",%1            \n"
>-    "sub       $0x20,%3                        \n"
>-    "jg        1b                              \n"
>-    "vzeroupper                                \n"
>-  : "+r"(src_yuy2),    // %0
>-    "+r"(dst_u),       // %1
>-    "+r"(dst_v),       // %2
>-    "+r"(width)          // %3
>-  : "r"((intptr_t)(stride_yuy2))  // %4
>-  : "memory", "cc", NACL_R14
>-    "xmm0", "xmm1", "xmm5"
>-  );
>+      LABELALIGN
>+      "1:                                        \n"
>+      "vmovdqu   (%0),%%ymm0                     \n"
>+      "vmovdqu   0x20(%0),%%ymm1                 \n"
>+      "vpavgb    0x00(%0,%4,1),%%ymm0,%%ymm0     \n"
>+      "vpavgb    0x20(%0,%4,1),%%ymm1,%%ymm1     \n"
>+      "lea       0x40(%0),%0                     \n"
>+      "vpsrlw    $0x8,%%ymm0,%%ymm0              \n"
>+      "vpsrlw    $0x8,%%ymm1,%%ymm1              \n"
>+      "vpackuswb %%ymm1,%%ymm0,%%ymm0            \n"
>+      "vpermq    $0xd8,%%ymm0,%%ymm0             \n"
>+      "vpand     %%ymm5,%%ymm0,%%ymm1            \n"
>+      "vpsrlw    $0x8,%%ymm0,%%ymm0              \n"
>+      "vpackuswb %%ymm1,%%ymm1,%%ymm1            \n"
>+      "vpackuswb %%ymm0,%%ymm0,%%ymm0            \n"
>+      "vpermq    $0xd8,%%ymm1,%%ymm1             \n"
>+      "vpermq    $0xd8,%%ymm0,%%ymm0             \n"
>+      "vextractf128 $0x0,%%ymm1,(%1)             \n"
>+      "vextractf128 $0x0,%%ymm0,0x00(%1,%2,1)    \n"
>+      "lea      0x10(%1),%1                      \n"
>+      "sub       $0x20,%3                        \n"
>+      "jg        1b                              \n"
>+      "vzeroupper                                \n"
>+      : "+r"(src_yuy2),               // %0
>+        "+r"(dst_u),                  // %1
>+        "+r"(dst_v),                  // %2
>+        "+r"(width)                   // %3
>+      : "r"((intptr_t)(stride_yuy2))  // %4
>+      : "memory", "cc", "xmm0", "xmm1", "xmm5");
> }
> 
>-void YUY2ToUV422Row_AVX2(const uint8* src_yuy2,
>-                         uint8* dst_u,
>-                         uint8* dst_v,
>+void YUY2ToUV422Row_AVX2(const uint8_t* src_yuy2,
>+                         uint8_t* dst_u,
>+                         uint8_t* dst_v,
>                          int width) {
>-  asm volatile (
>-    "vpcmpeqb  %%ymm5,%%ymm5,%%ymm5            \n"
>-    "vpsrlw    $0x8,%%ymm5,%%ymm5              \n"
>-    "sub       %1,%2                           \n"
>+  asm volatile(
>+      "vpcmpeqb  %%ymm5,%%ymm5,%%ymm5            \n"
>+      "vpsrlw    $0x8,%%ymm5,%%ymm5              \n"
>+      "sub       %1,%2                           \n"
> 
>-    LABELALIGN
>-    "1:                                        \n"
>-    "vmovdqu   " MEMACCESS(0) ",%%ymm0         \n"
>-    "vmovdqu   " MEMACCESS2(0x20,0) ",%%ymm1   \n"
>-    "lea       " MEMLEA(0x40,0) ",%0           \n"
>-    "vpsrlw    $0x8,%%ymm0,%%ymm0              \n"
>-    "vpsrlw    $0x8,%%ymm1,%%ymm1              \n"
>-    "vpackuswb %%ymm1,%%ymm0,%%ymm0            \n"
>-    "vpermq    $0xd8,%%ymm0,%%ymm0             \n"
>-    "vpand     %%ymm5,%%ymm0,%%ymm1            \n"
>-    "vpsrlw    $0x8,%%ymm0,%%ymm0              \n"
>-    "vpackuswb %%ymm1,%%ymm1,%%ymm1            \n"
>-    "vpackuswb %%ymm0,%%ymm0,%%ymm0            \n"
>-    "vpermq    $0xd8,%%ymm1,%%ymm1             \n"
>-    "vpermq    $0xd8,%%ymm0,%%ymm0             \n"
>-    "vextractf128 $0x0,%%ymm1," MEMACCESS(1) " \n"
>-    VEXTOPMEM(vextractf128,0,ymm0,0x00,1,2,1) // vextractf128 $0x0,%%ymm0,(%1,%2,1)
>-    "lea      " MEMLEA(0x10,1) ",%1            \n"
>-    "sub       $0x20,%3                        \n"
>-    "jg        1b                              \n"
>-    "vzeroupper                                \n"
>-  : "+r"(src_yuy2),    // %0
>-    "+r"(dst_u),       // %1
>-    "+r"(dst_v),       // %2
>-    "+r"(width)          // %3
>-  :
>-  : "memory", "cc", NACL_R14
>-    "xmm0", "xmm1", "xmm5"
>-  );
>+      LABELALIGN
>+      "1:                                        \n"
>+      "vmovdqu   (%0),%%ymm0                     \n"
>+      "vmovdqu   0x20(%0),%%ymm1                 \n"
>+      "lea       0x40(%0),%0                     \n"
>+      "vpsrlw    $0x8,%%ymm0,%%ymm0              \n"
>+      "vpsrlw    $0x8,%%ymm1,%%ymm1              \n"
>+      "vpackuswb %%ymm1,%%ymm0,%%ymm0            \n"
>+      "vpermq    $0xd8,%%ymm0,%%ymm0             \n"
>+      "vpand     %%ymm5,%%ymm0,%%ymm1            \n"
>+      "vpsrlw    $0x8,%%ymm0,%%ymm0              \n"
>+      "vpackuswb %%ymm1,%%ymm1,%%ymm1            \n"
>+      "vpackuswb %%ymm0,%%ymm0,%%ymm0            \n"
>+      "vpermq    $0xd8,%%ymm1,%%ymm1             \n"
>+      "vpermq    $0xd8,%%ymm0,%%ymm0             \n"
>+      "vextractf128 $0x0,%%ymm1,(%1)             \n"
>+      "vextractf128 $0x0,%%ymm0,0x00(%1,%2,1)    \n"
>+      "lea      0x10(%1),%1                      \n"
>+      "sub       $0x20,%3                        \n"
>+      "jg        1b                              \n"
>+      "vzeroupper                                \n"
>+      : "+r"(src_yuy2),  // %0
>+        "+r"(dst_u),     // %1
>+        "+r"(dst_v),     // %2
>+        "+r"(width)      // %3
>+      :
>+      : "memory", "cc", "xmm0", "xmm1", "xmm5");
> }
>-
>-void UYVYToYRow_AVX2(const uint8* src_uyvy, uint8* dst_y, int width) {
>-  asm volatile (
>-    LABELALIGN
>-    "1:                                        \n"
>-    "vmovdqu   " MEMACCESS(0) ",%%ymm0         \n"
>-    "vmovdqu   " MEMACCESS2(0x20,0) ",%%ymm1   \n"
>-    "lea       " MEMLEA(0x40,0) ",%0           \n"
>-    "vpsrlw    $0x8,%%ymm0,%%ymm0              \n"
>-    "vpsrlw    $0x8,%%ymm1,%%ymm1              \n"
>-    "vpackuswb %%ymm1,%%ymm0,%%ymm0            \n"
>-    "vpermq    $0xd8,%%ymm0,%%ymm0             \n"
>-    "vmovdqu   %%ymm0," MEMACCESS(1) "         \n"
>-    "lea      " MEMLEA(0x20,1) ",%1            \n"
>-    "sub       $0x20,%2                        \n"
>-    "jg        1b                              \n"
>-    "vzeroupper                                \n"
>-  : "+r"(src_uyvy),  // %0
>-    "+r"(dst_y),     // %1
>-    "+r"(width)        // %2
>-  :
>-  : "memory", "cc"
>-    , "xmm0", "xmm1", "xmm5"
>-  );
>+
>+void UYVYToYRow_AVX2(const uint8_t* src_uyvy, uint8_t* dst_y, int width) {
>+  asm volatile(
>+
>+      LABELALIGN
>+      "1:                                        \n"
>+      "vmovdqu   (%0),%%ymm0                     \n"
>+      "vmovdqu   0x20(%0),%%ymm1                 \n"
>+      "lea       0x40(%0),%0                     \n"
>+      "vpsrlw    $0x8,%%ymm0,%%ymm0              \n"
>+      "vpsrlw    $0x8,%%ymm1,%%ymm1              \n"
>+      "vpackuswb %%ymm1,%%ymm0,%%ymm0            \n"
>+      "vpermq    $0xd8,%%ymm0,%%ymm0             \n"
>+      "vmovdqu   %%ymm0,(%1)                     \n"
>+      "lea      0x20(%1),%1                      \n"
>+      "sub       $0x20,%2                        \n"
>+      "jg        1b                              \n"
>+      "vzeroupper                                \n"
>+      : "+r"(src_uyvy),  // %0
>+        "+r"(dst_y),     // %1
>+        "+r"(width)      // %2
>+      :
>+      : "memory", "cc", "xmm0", "xmm1", "xmm5");
> }
>-void UYVYToUVRow_AVX2(const uint8* src_uyvy,
>+void UYVYToUVRow_AVX2(const uint8_t* src_uyvy,
>                       int stride_uyvy,
>-                      uint8* dst_u,
>-                      uint8* dst_v,
>+                      uint8_t* dst_u,
>+                      uint8_t* dst_v,
>                       int width) {
>-  asm volatile (
>-    "vpcmpeqb  %%ymm5,%%ymm5,%%ymm5            \n"
>-    "vpsrlw    $0x8,%%ymm5,%%ymm5              \n"
>-    "sub       %1,%2                           \n"
>+  asm volatile(
>+      "vpcmpeqb  %%ymm5,%%ymm5,%%ymm5            \n"
>+      "vpsrlw    $0x8,%%ymm5,%%ymm5              \n"
>+      "sub       %1,%2                           \n"
> 
>-    LABELALIGN
>-    "1:                                        \n"
>-    "vmovdqu   " MEMACCESS(0) ",%%ymm0         \n"
>-    "vmovdqu   " MEMACCESS2(0x20,0) ",%%ymm1   \n"
>-    VMEMOPREG(vpavgb,0x00,0,4,1,ymm0,ymm0)     // vpavgb (%0,%4,1),%%ymm0,%%ymm0
>-    VMEMOPREG(vpavgb,0x20,0,4,1,ymm1,ymm1)
>-    "lea       " MEMLEA(0x40,0) ",%0           \n"
>-    "vpand     %%ymm5,%%ymm0,%%ymm0            \n"
>-    "vpand     %%ymm5,%%ymm1,%%ymm1            \n"
>-    "vpackuswb %%ymm1,%%ymm0,%%ymm0            \n"
>-    "vpermq    $0xd8,%%ymm0,%%ymm0             \n"
>-    "vpand     %%ymm5,%%ymm0,%%ymm1            \n"
>-    "vpsrlw    $0x8,%%ymm0,%%ymm0              \n"
>-    "vpackuswb %%ymm1,%%ymm1,%%ymm1            \n"
>-    "vpackuswb %%ymm0,%%ymm0,%%ymm0            \n"
>-    "vpermq    $0xd8,%%ymm1,%%ymm1             \n"
>-    "vpermq    $0xd8,%%ymm0,%%ymm0             \n"
>-    "vextractf128 $0x0,%%ymm1," MEMACCESS(1) " \n"
>-    VEXTOPMEM(vextractf128,0,ymm0,0x00,1,2,1) // vextractf128 $0x0,%%ymm0,(%1,%2,1)
>-    "lea      " MEMLEA(0x10,1) ",%1            \n"
>-    "sub       $0x20,%3                        \n"
>-    "jg        1b                              \n"
>-    "vzeroupper                                \n"
>-  : "+r"(src_uyvy),    // %0
>-    "+r"(dst_u),       // %1
>-    "+r"(dst_v),       // %2
>-    "+r"(width)          // %3
>-  : "r"((intptr_t)(stride_uyvy))  // %4
>-  : "memory", "cc", NACL_R14
>-    "xmm0", "xmm1", "xmm5"
>-  );
>+      LABELALIGN
>+      "1:                                        \n"
>+      "vmovdqu   (%0),%%ymm0                     \n"
>+      "vmovdqu   0x20(%0),%%ymm1                 \n"
>+      "vpavgb    0x00(%0,%4,1),%%ymm0,%%ymm0     \n"
>+      "vpavgb    0x20(%0,%4,1),%%ymm1,%%ymm1     \n"
>+      "lea       0x40(%0),%0                     \n"
>+      "vpand     %%ymm5,%%ymm0,%%ymm0            \n"
>+      "vpand     %%ymm5,%%ymm1,%%ymm1            \n"
>+      "vpackuswb %%ymm1,%%ymm0,%%ymm0            \n"
>+      "vpermq    $0xd8,%%ymm0,%%ymm0             \n"
>+      "vpand     %%ymm5,%%ymm0,%%ymm1            \n"
>+      "vpsrlw    $0x8,%%ymm0,%%ymm0              \n"
>+      "vpackuswb %%ymm1,%%ymm1,%%ymm1            \n"
>+      "vpackuswb %%ymm0,%%ymm0,%%ymm0            \n"
>+      "vpermq    $0xd8,%%ymm1,%%ymm1             \n"
>+      "vpermq    $0xd8,%%ymm0,%%ymm0             \n"
>+      "vextractf128 $0x0,%%ymm1,(%1)             \n"
>+      "vextractf128 $0x0,%%ymm0,0x00(%1,%2,1)    \n"
>+      "lea      0x10(%1),%1                      \n"
>+      "sub       $0x20,%3                        \n"
>+      "jg        1b                              \n"
>+      "vzeroupper                                \n"
>+      : "+r"(src_uyvy),               // %0
>+        "+r"(dst_u),                  // %1
>+        "+r"(dst_v),                  // %2
>+        "+r"(width)                   // %3
>+      : "r"((intptr_t)(stride_uyvy))  // %4
>+      : "memory", "cc", "xmm0", "xmm1", "xmm5");
> }
> 
>-void UYVYToUV422Row_AVX2(const uint8* src_uyvy,
>-                         uint8* dst_u,
>-                         uint8* dst_v,
>+void UYVYToUV422Row_AVX2(const uint8_t* src_uyvy,
>+                         uint8_t* dst_u,
>+                         uint8_t* dst_v,
>                          int width) {
>-  asm volatile (
>-    "vpcmpeqb   %%ymm5,%%ymm5,%%ymm5           \n"
>-    "vpsrlw     $0x8,%%ymm5,%%ymm5             \n"
>-    "sub       %1,%2                           \n"
>+  asm volatile(
>+      "vpcmpeqb   %%ymm5,%%ymm5,%%ymm5           \n"
>+      "vpsrlw     $0x8,%%ymm5,%%ymm5             \n"
>+      "sub       %1,%2                           \n"
> 
>-    LABELALIGN
>-    "1:                                        \n"
>-    "vmovdqu   " MEMACCESS(0) ",%%ymm0         \n"
>-    "vmovdqu   " MEMACCESS2(0x20,0) ",%%ymm1   \n"
>-    "lea       " MEMLEA(0x40,0) ",%0           \n"
>-    "vpand     %%ymm5,%%ymm0,%%ymm0            \n"
>-    "vpand     %%ymm5,%%ymm1,%%ymm1            \n"
>-    "vpackuswb %%ymm1,%%ymm0,%%ymm0            \n"
>-    "vpermq    $0xd8,%%ymm0,%%ymm0             \n"
>-    "vpand     %%ymm5,%%ymm0,%%ymm1            \n"
>-    "vpsrlw    $0x8,%%ymm0,%%ymm0              \n"
>-    "vpackuswb %%ymm1,%%ymm1,%%ymm1            \n"
>-    "vpackuswb %%ymm0,%%ymm0,%%ymm0            \n"
>-    "vpermq    $0xd8,%%ymm1,%%ymm1             \n"
>-    "vpermq    $0xd8,%%ymm0,%%ymm0             \n"
>-    "vextractf128 $0x0,%%ymm1," MEMACCESS(1) " \n"
>-    VEXTOPMEM(vextractf128,0,ymm0,0x00,1,2,1) // vextractf128 $0x0,%%ymm0,(%1,%2,1)
>-    "lea      " MEMLEA(0x10,1) ",%1            \n"
>-    "sub       $0x20,%3                        \n"
>-    "jg        1b                              \n"
>-    "vzeroupper                                \n"
>-  : "+r"(src_uyvy),    // %0
>-    "+r"(dst_u),       // %1
>-    "+r"(dst_v),       // %2
>-    "+r"(width)          // %3
>-  :
>-  : "memory", "cc", NACL_R14
>-    "xmm0", "xmm1", "xmm5"
>-  );
>+      LABELALIGN
>+      "1:                                        \n"
>+      "vmovdqu   (%0),%%ymm0                     \n"
>+      "vmovdqu   0x20(%0),%%ymm1                 \n"
>+      "lea       0x40(%0),%0                     \n"
>+      "vpand     %%ymm5,%%ymm0,%%ymm0            \n"
>+      "vpand     %%ymm5,%%ymm1,%%ymm1            \n"
>+      "vpackuswb %%ymm1,%%ymm0,%%ymm0            \n"
>+      "vpermq    $0xd8,%%ymm0,%%ymm0             \n"
>+      "vpand     %%ymm5,%%ymm0,%%ymm1            \n"
>+      "vpsrlw    $0x8,%%ymm0,%%ymm0              \n"
>+      "vpackuswb %%ymm1,%%ymm1,%%ymm1            \n"
>+      "vpackuswb %%ymm0,%%ymm0,%%ymm0            \n"
>+      "vpermq    $0xd8,%%ymm1,%%ymm1             \n"
>+      "vpermq    $0xd8,%%ymm0,%%ymm0             \n"
>+      "vextractf128 $0x0,%%ymm1,(%1)             \n"
>+      "vextractf128 $0x0,%%ymm0,0x00(%1,%2,1)    \n"
>+      "lea      0x10(%1),%1                      \n"
>+      "sub       $0x20,%3                        \n"
>+      "jg        1b                              \n"
>+      "vzeroupper                                \n"
>+      : "+r"(src_uyvy),  // %0
>+        "+r"(dst_u),     // %1
>+        "+r"(dst_v),     // %2
>+        "+r"(width)      // %3
>+      :
>+      : "memory", "cc", "xmm0", "xmm1", "xmm5");
> }
> #endif  // HAS_YUY2TOYROW_AVX2
> 
> #ifdef HAS_ARGBBLENDROW_SSSE3
> // Shuffle table for isolating alpha.
>-static uvec8 kShuffleAlpha = {3u,  0x80, 3u,  0x80, 7u,  0x80, 7u,  0x80,
>-                              11u, 0x80, 11u, 0x80, 15u, 0x80, 15u, 0x80};
>+static const uvec8 kShuffleAlpha = {3u,  0x80, 3u,  0x80, 7u,  0x80, 7u,  0x80,
>+                                    11u, 0x80, 11u, 0x80, 15u, 0x80, 15u, 0x80};
> 
> // Blend 8 pixels at a time
>-void ARGBBlendRow_SSSE3(const uint8* src_argb0,
>-                        const uint8* src_argb1,
>-                        uint8* dst_argb,
>+void ARGBBlendRow_SSSE3(const uint8_t* src_argb0,
>+                        const uint8_t* src_argb1,
>+                        uint8_t* dst_argb,
>                         int width) {
>-  asm volatile (
>-    "pcmpeqb   %%xmm7,%%xmm7                   \n"
>-    "psrlw     $0xf,%%xmm7                     \n"
>-    "pcmpeqb   %%xmm6,%%xmm6                   \n"
>-    "psrlw     $0x8,%%xmm6                     \n"
>-    "pcmpeqb   %%xmm5,%%xmm5                   \n"
>-    "psllw     $0x8,%%xmm5                     \n"
>-    "pcmpeqb   %%xmm4,%%xmm4                   \n"
>-    "pslld     $0x18,%%xmm4                    \n"
>-    "sub       $0x4,%3                         \n"
>-    "jl        49f                             \n"
>-
>-    // 4 pixel loop.
>-    LABELALIGN
>-  "40:                                         \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm3         \n"
>-    "lea       " MEMLEA(0x10,0) ",%0           \n"
>-    "movdqa    %%xmm3,%%xmm0                   \n"
>-    "pxor      %%xmm4,%%xmm3                   \n"
>-    "movdqu    " MEMACCESS(1) ",%%xmm2         \n"
>-    "pshufb    %4,%%xmm3                       \n"
>-    "pand      %%xmm6,%%xmm2                   \n"
>-    "paddw     %%xmm7,%%xmm3                   \n"
>-    "pmullw    %%xmm3,%%xmm2                   \n"
>-    "movdqu    " MEMACCESS(1) ",%%xmm1         \n"
>-    "lea       " MEMLEA(0x10,1) ",%1           \n"
>-    "psrlw     $0x8,%%xmm1                     \n"
>-    "por       %%xmm4,%%xmm0                   \n"
>-    "pmullw    %%xmm3,%%xmm1                   \n"
>-    "psrlw     $0x8,%%xmm2                     \n"
>-    "paddusb   %%xmm2,%%xmm0                   \n"
>-    "pand      %%xmm5,%%xmm1                   \n"
>-    "paddusb   %%xmm1,%%xmm0                   \n"
>-    "movdqu    %%xmm0," MEMACCESS(2) "         \n"
>-    "lea       " MEMLEA(0x10,2) ",%2           \n"
>-    "sub       $0x4,%3                         \n"
>-    "jge       40b                             \n"
>-
>-  "49:                                         \n"
>-    "add       $0x3,%3                         \n"
>-    "jl        99f                             \n"
>-
>-    // 1 pixel loop.
>-  "91:                                         \n"
>-    "movd      " MEMACCESS(0) ",%%xmm3         \n"
>-    "lea       " MEMLEA(0x4,0) ",%0            \n"
>-    "movdqa    %%xmm3,%%xmm0                   \n"
>-    "pxor      %%xmm4,%%xmm3                   \n"
>-    "movd      " MEMACCESS(1) ",%%xmm2         \n"
>-    "pshufb    %4,%%xmm3                       \n"
>-    "pand      %%xmm6,%%xmm2                   \n"
>-    "paddw     %%xmm7,%%xmm3                   \n"
>-    "pmullw    %%xmm3,%%xmm2                   \n"
>-    "movd      " MEMACCESS(1) ",%%xmm1         \n"
>-    "lea       " MEMLEA(0x4,1) ",%1            \n"
>-    "psrlw     $0x8,%%xmm1                     \n"
>-    "por       %%xmm4,%%xmm0                   \n"
>-    "pmullw    %%xmm3,%%xmm1                   \n"
>-    "psrlw     $0x8,%%xmm2                     \n"
>-    "paddusb   %%xmm2,%%xmm0                   \n"
>-    "pand      %%xmm5,%%xmm1                   \n"
>-    "paddusb   %%xmm1,%%xmm0                   \n"
>-    "movd      %%xmm0," MEMACCESS(2) "         \n"
>-    "lea       " MEMLEA(0x4,2) ",%2            \n"
>-    "sub       $0x1,%3                         \n"
>-    "jge       91b                             \n"
>-  "99:                                         \n"
>-  : "+r"(src_argb0),    // %0
>-    "+r"(src_argb1),    // %1
>-    "+r"(dst_argb),     // %2
>-    "+r"(width)         // %3
>-  : "m"(kShuffleAlpha)  // %4
>-  : "memory", "cc"
>-    , "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6", "xmm7"
>-  );
>+  asm volatile(
>+      "pcmpeqb   %%xmm7,%%xmm7                   \n"
>+      "psrlw     $0xf,%%xmm7                     \n"
>+      "pcmpeqb   %%xmm6,%%xmm6                   \n"
>+      "psrlw     $0x8,%%xmm6                     \n"
>+      "pcmpeqb   %%xmm5,%%xmm5                   \n"
>+      "psllw     $0x8,%%xmm5                     \n"
>+      "pcmpeqb   %%xmm4,%%xmm4                   \n"
>+      "pslld     $0x18,%%xmm4                    \n"
>+      "sub       $0x4,%3                         \n"
>+      "jl        49f                             \n"
>+
>+      // 4 pixel loop.
>+      LABELALIGN
>+      "40:                                       \n"
>+      "movdqu    (%0),%%xmm3                     \n"
>+      "lea       0x10(%0),%0                     \n"
>+      "movdqa    %%xmm3,%%xmm0                   \n"
>+      "pxor      %%xmm4,%%xmm3                   \n"
>+      "movdqu    (%1),%%xmm2                     \n"
>+      "pshufb    %4,%%xmm3                       \n"
>+      "pand      %%xmm6,%%xmm2                   \n"
>+      "paddw     %%xmm7,%%xmm3                   \n"
>+      "pmullw    %%xmm3,%%xmm2                   \n"
>+      "movdqu    (%1),%%xmm1                     \n"
>+      "lea       0x10(%1),%1                     \n"
>+      "psrlw     $0x8,%%xmm1                     \n"
>+      "por       %%xmm4,%%xmm0                   \n"
>+      "pmullw    %%xmm3,%%xmm1                   \n"
>+      "psrlw     $0x8,%%xmm2                     \n"
>+      "paddusb   %%xmm2,%%xmm0                   \n"
>+      "pand      %%xmm5,%%xmm1                   \n"
>+      "paddusb   %%xmm1,%%xmm0                   \n"
>+      "movdqu    %%xmm0,(%2)                     \n"
>+      "lea       0x10(%2),%2                     \n"
>+      "sub       $0x4,%3                         \n"
>+      "jge       40b                             \n"
>+
>+      "49:                                       \n"
>+      "add       $0x3,%3                         \n"
>+      "jl        99f                             \n"
>+
>+      // 1 pixel loop.
>+      "91:                                       \n"
>+      "movd      (%0),%%xmm3                     \n"
>+      "lea       0x4(%0),%0                      \n"
>+      "movdqa    %%xmm3,%%xmm0                   \n"
>+      "pxor      %%xmm4,%%xmm3                   \n"
>+      "movd      (%1),%%xmm2                     \n"
>+      "pshufb    %4,%%xmm3                       \n"
>+      "pand      %%xmm6,%%xmm2                   \n"
>+      "paddw     %%xmm7,%%xmm3                   \n"
>+      "pmullw    %%xmm3,%%xmm2                   \n"
>+      "movd      (%1),%%xmm1                     \n"
>+      "lea       0x4(%1),%1                      \n"
>+      "psrlw     $0x8,%%xmm1                     \n"
>+      "por       %%xmm4,%%xmm0                   \n"
>+      "pmullw    %%xmm3,%%xmm1                   \n"
>+      "psrlw     $0x8,%%xmm2                     \n"
>+      "paddusb   %%xmm2,%%xmm0                   \n"
>+      "pand      %%xmm5,%%xmm1                   \n"
>+      "paddusb   %%xmm1,%%xmm0                   \n"
>+      "movd      %%xmm0,(%2)                     \n"
>+      "lea       0x4(%2),%2                      \n"
>+      "sub       $0x1,%3                         \n"
>+      "jge       91b                             \n"
>+      "99:                                       \n"
>+      : "+r"(src_argb0),    // %0
>+        "+r"(src_argb1),    // %1
>+        "+r"(dst_argb),     // %2
>+        "+r"(width)         // %3
>+      : "m"(kShuffleAlpha)  // %4
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6",
>+        "xmm7");
> }
> #endif  // HAS_ARGBBLENDROW_SSSE3
> 
>@@ -3863,10 +4437,10 @@ void ARGBBlendRow_SSSE3(const uint8* src_argb0,
> // =((A2*C2)+(B2*(255-C2))+255)/256
> // signed version of math
> // =(((A2-128)*C2)+((B2-128)*(255-C2))+32768+127)/256
>-void BlendPlaneRow_SSSE3(const uint8* src0,
>-                         const uint8* src1,
>-                         const uint8* alpha,
>-                         uint8* dst,
>+void BlendPlaneRow_SSSE3(const uint8_t* src0,
>+                         const uint8_t* src1,
>+                         const uint8_t* alpha,
>+                         uint8_t* dst,
>                          int width) {
>   asm volatile(
>       "pcmpeqb    %%xmm5,%%xmm5                  \n"
>@@ -3915,10 +4489,10 @@ void BlendPlaneRow_SSSE3(const uint8* src0,
> // =((A2*C2)+(B2*(255-C2))+255)/256
> // signed version of math
> // =(((A2-128)*C2)+((B2-128)*(255-C2))+32768+127)/256
>-void BlendPlaneRow_AVX2(const uint8* src0,
>-                        const uint8* src1,
>-                        const uint8* alpha,
>-                        uint8* dst,
>+void BlendPlaneRow_AVX2(const uint8_t* src0,
>+                        const uint8_t* src1,
>+                        const uint8_t* alpha,
>+                        uint8_t* dst,
>                         int width) {
>   asm volatile(
>       "vpcmpeqb   %%ymm5,%%ymm5,%%ymm5           \n"
>@@ -3972,50 +4546,50 @@ void BlendPlaneRow_AVX2(const uint8* src0,
> 
> #ifdef HAS_ARGBATTENUATEROW_SSSE3
> // Shuffle table duplicating alpha
>-static uvec8 kShuffleAlpha0 = {3u, 3u, 3u, 3u, 3u, 3u, 128u, 128u,
>-                               7u, 7u, 7u, 7u, 7u, 7u, 128u, 128u};
>-static uvec8 kShuffleAlpha1 = {11u, 11u, 11u, 11u, 11u, 11u, 128u, 128u,
>-                               15u, 15u, 15u, 15u, 15u, 15u, 128u, 128u};
>+static const uvec8 kShuffleAlpha0 = {3u, 3u, 3u, 3u, 3u, 3u, 128u, 128u,
>+                                     7u, 7u, 7u, 7u, 7u, 7u, 128u, 128u};
>+static const uvec8 kShuffleAlpha1 = {11u, 11u, 11u, 11u, 11u, 11u, 128u, 128u,
>+                                     15u, 15u, 15u, 15u, 15u, 15u, 128u, 128u};
> // Attenuate 4 pixels at a time.
>-void ARGBAttenuateRow_SSSE3(const uint8* src_argb, uint8* dst_argb, int width) {
>-  asm volatile (
>-    "pcmpeqb   %%xmm3,%%xmm3                   \n"
>-    "pslld     $0x18,%%xmm3                    \n"
>-    "movdqa    %3,%%xmm4                       \n"
>-    "movdqa    %4,%%xmm5                       \n"
>+void ARGBAttenuateRow_SSSE3(const uint8_t* src_argb,
>+                            uint8_t* dst_argb,
>+                            int width) {
>+  asm volatile(
>+      "pcmpeqb   %%xmm3,%%xmm3                   \n"
>+      "pslld     $0x18,%%xmm3                    \n"
>+      "movdqa    %3,%%xmm4                       \n"
>+      "movdqa    %4,%%xmm5                       \n"
> 
>-    // 4 pixel loop.
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    "pshufb    %%xmm4,%%xmm0                   \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm1         \n"
>-    "punpcklbw %%xmm1,%%xmm1                   \n"
>-    "pmulhuw   %%xmm1,%%xmm0                   \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm1         \n"
>-    "pshufb    %%xmm5,%%xmm1                   \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm2         \n"
>-    "punpckhbw %%xmm2,%%xmm2                   \n"
>-    "pmulhuw   %%xmm2,%%xmm1                   \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm2         \n"
>-    "lea       " MEMLEA(0x10,0) ",%0           \n"
>-    "pand      %%xmm3,%%xmm2                   \n"
>-    "psrlw     $0x8,%%xmm0                     \n"
>-    "psrlw     $0x8,%%xmm1                     \n"
>-    "packuswb  %%xmm1,%%xmm0                   \n"
>-    "por       %%xmm2,%%xmm0                   \n"
>-    "movdqu    %%xmm0," MEMACCESS(1) "         \n"
>-    "lea       " MEMLEA(0x10,1) ",%1           \n"
>-    "sub       $0x4,%2                         \n"
>-    "jg        1b                              \n"
>-  : "+r"(src_argb),    // %0
>-    "+r"(dst_argb),    // %1
>-    "+r"(width)        // %2
>-  : "m"(kShuffleAlpha0),  // %3
>-    "m"(kShuffleAlpha1)  // %4
>-  : "memory", "cc"
>-    , "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5"
>-  );
>+      // 4 pixel loop.
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "pshufb    %%xmm4,%%xmm0                   \n"
>+      "movdqu    (%0),%%xmm1                     \n"
>+      "punpcklbw %%xmm1,%%xmm1                   \n"
>+      "pmulhuw   %%xmm1,%%xmm0                   \n"
>+      "movdqu    (%0),%%xmm1                     \n"
>+      "pshufb    %%xmm5,%%xmm1                   \n"
>+      "movdqu    (%0),%%xmm2                     \n"
>+      "punpckhbw %%xmm2,%%xmm2                   \n"
>+      "pmulhuw   %%xmm2,%%xmm1                   \n"
>+      "movdqu    (%0),%%xmm2                     \n"
>+      "lea       0x10(%0),%0                     \n"
>+      "pand      %%xmm3,%%xmm2                   \n"
>+      "psrlw     $0x8,%%xmm0                     \n"
>+      "psrlw     $0x8,%%xmm1                     \n"
>+      "packuswb  %%xmm1,%%xmm0                   \n"
>+      "por       %%xmm2,%%xmm0                   \n"
>+      "movdqu    %%xmm0,(%1)                     \n"
>+      "lea       0x10(%1),%1                     \n"
>+      "sub       $0x4,%2                         \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_argb),       // %0
>+        "+r"(dst_argb),       // %1
>+        "+r"(width)           // %2
>+      : "m"(kShuffleAlpha0),  // %3
>+        "m"(kShuffleAlpha1)   // %4
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5");
> }
> #endif  // HAS_ARGBATTENUATEROW_SSSE3
> 
>@@ -4025,87 +4599,85 @@ static const uvec8 kShuffleAlpha_AVX2 = {6u,   7u,   6u,   7u,  6u,  7u,
>                                          128u, 128u, 14u,  15u, 14u, 15u,
>                                          14u,  15u,  128u, 128u};
> // Attenuate 8 pixels at a time.
>-void ARGBAttenuateRow_AVX2(const uint8* src_argb, uint8* dst_argb, int width) {
>-  asm volatile (
>-    "vbroadcastf128 %3,%%ymm4                  \n"
>-    "vpcmpeqb   %%ymm5,%%ymm5,%%ymm5           \n"
>-    "vpslld     $0x18,%%ymm5,%%ymm5            \n"
>-    "sub        %0,%1                          \n"
>+void ARGBAttenuateRow_AVX2(const uint8_t* src_argb,
>+                           uint8_t* dst_argb,
>+                           int width) {
>+  asm volatile(
>+      "vbroadcastf128 %3,%%ymm4                  \n"
>+      "vpcmpeqb   %%ymm5,%%ymm5,%%ymm5           \n"
>+      "vpslld     $0x18,%%ymm5,%%ymm5            \n"
>+      "sub        %0,%1                          \n"
> 
>-    // 8 pixel loop.
>-    LABELALIGN
>-    "1:                                        \n"
>-    "vmovdqu    " MEMACCESS(0) ",%%ymm6        \n"
>-    "vpunpcklbw %%ymm6,%%ymm6,%%ymm0           \n"
>-    "vpunpckhbw %%ymm6,%%ymm6,%%ymm1           \n"
>-    "vpshufb    %%ymm4,%%ymm0,%%ymm2           \n"
>-    "vpshufb    %%ymm4,%%ymm1,%%ymm3           \n"
>-    "vpmulhuw   %%ymm2,%%ymm0,%%ymm0           \n"
>-    "vpmulhuw   %%ymm3,%%ymm1,%%ymm1           \n"
>-    "vpand      %%ymm5,%%ymm6,%%ymm6           \n"
>-    "vpsrlw     $0x8,%%ymm0,%%ymm0             \n"
>-    "vpsrlw     $0x8,%%ymm1,%%ymm1             \n"
>-    "vpackuswb  %%ymm1,%%ymm0,%%ymm0           \n"
>-    "vpor       %%ymm6,%%ymm0,%%ymm0           \n"
>-    MEMOPMEM(vmovdqu,ymm0,0x00,0,1,1)          //  vmovdqu %%ymm0,(%0,%1)
>-    "lea       " MEMLEA(0x20,0) ",%0           \n"
>-    "sub        $0x8,%2                        \n"
>-    "jg        1b                              \n"
>-    "vzeroupper                                \n"
>-  : "+r"(src_argb),    // %0
>-    "+r"(dst_argb),    // %1
>-    "+r"(width)        // %2
>-  : "m"(kShuffleAlpha_AVX2)  // %3
>-  : "memory", "cc"
>-    , "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6"
>-  );
>+      // 8 pixel loop.
>+      LABELALIGN
>+      "1:                                        \n"
>+      "vmovdqu    (%0),%%ymm6                    \n"
>+      "vpunpcklbw %%ymm6,%%ymm6,%%ymm0           \n"
>+      "vpunpckhbw %%ymm6,%%ymm6,%%ymm1           \n"
>+      "vpshufb    %%ymm4,%%ymm0,%%ymm2           \n"
>+      "vpshufb    %%ymm4,%%ymm1,%%ymm3           \n"
>+      "vpmulhuw   %%ymm2,%%ymm0,%%ymm0           \n"
>+      "vpmulhuw   %%ymm3,%%ymm1,%%ymm1           \n"
>+      "vpand      %%ymm5,%%ymm6,%%ymm6           \n"
>+      "vpsrlw     $0x8,%%ymm0,%%ymm0             \n"
>+      "vpsrlw     $0x8,%%ymm1,%%ymm1             \n"
>+      "vpackuswb  %%ymm1,%%ymm0,%%ymm0           \n"
>+      "vpor       %%ymm6,%%ymm0,%%ymm0           \n"
>+      "vmovdqu    %%ymm0,0x00(%0,%1,1)           \n"
>+      "lea       0x20(%0),%0                     \n"
>+      "sub        $0x8,%2                        \n"
>+      "jg        1b                              \n"
>+      "vzeroupper                                \n"
>+      : "+r"(src_argb),          // %0
>+        "+r"(dst_argb),          // %1
>+        "+r"(width)              // %2
>+      : "m"(kShuffleAlpha_AVX2)  // %3
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6");
> }
> #endif  // HAS_ARGBATTENUATEROW_AVX2
> 
> #ifdef HAS_ARGBUNATTENUATEROW_SSE2
> // Unattenuate 4 pixels at a time.
>-void ARGBUnattenuateRow_SSE2(const uint8* src_argb,
>-                             uint8* dst_argb,
>+void ARGBUnattenuateRow_SSE2(const uint8_t* src_argb,
>+                             uint8_t* dst_argb,
>                              int width) {
>   uintptr_t alpha;
>-  asm volatile (
>-    // 4 pixel loop.
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    "movzb     " MEMACCESS2(0x03,0) ",%3       \n"
>-    "punpcklbw %%xmm0,%%xmm0                   \n"
>-    MEMOPREG(movd,0x00,4,3,4,xmm2)             //  movd      0x0(%4,%3,4),%%xmm2
>-    "movzb     " MEMACCESS2(0x07,0) ",%3       \n"
>-    MEMOPREG(movd,0x00,4,3,4,xmm3)             //  movd      0x0(%4,%3,4),%%xmm3
>-    "pshuflw   $0x40,%%xmm2,%%xmm2             \n"
>-    "pshuflw   $0x40,%%xmm3,%%xmm3             \n"
>-    "movlhps   %%xmm3,%%xmm2                   \n"
>-    "pmulhuw   %%xmm2,%%xmm0                   \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm1         \n"
>-    "movzb     " MEMACCESS2(0x0b,0) ",%3       \n"
>-    "punpckhbw %%xmm1,%%xmm1                   \n"
>-    MEMOPREG(movd,0x00,4,3,4,xmm2)             //  movd      0x0(%4,%3,4),%%xmm2
>-    "movzb     " MEMACCESS2(0x0f,0) ",%3       \n"
>-    MEMOPREG(movd,0x00,4,3,4,xmm3)             //  movd      0x0(%4,%3,4),%%xmm3
>-    "pshuflw   $0x40,%%xmm2,%%xmm2             \n"
>-    "pshuflw   $0x40,%%xmm3,%%xmm3             \n"
>-    "movlhps   %%xmm3,%%xmm2                   \n"
>-    "pmulhuw   %%xmm2,%%xmm1                   \n"
>-    "lea       " MEMLEA(0x10,0) ",%0           \n"
>-    "packuswb  %%xmm1,%%xmm0                   \n"
>-    "movdqu    %%xmm0," MEMACCESS(1) "         \n"
>-    "lea       " MEMLEA(0x10,1) ",%1           \n"
>-    "sub       $0x4,%2                         \n"
>-    "jg        1b                              \n"
>-  : "+r"(src_argb),     // %0
>-    "+r"(dst_argb),     // %1
>-    "+r"(width),        // %2
>-    "=&r"(alpha)        // %3
>-  : "r"(fixed_invtbl8)  // %4
>-  : "memory", "cc", NACL_R14
>-    "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5"
>-  );
>+  asm volatile(
>+      // 4 pixel loop.
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "movzb     0x03(%0),%3                     \n"
>+      "punpcklbw %%xmm0,%%xmm0                   \n"
>+      "movd      0x00(%4,%3,4),%%xmm2            \n"
>+      "movzb     0x07(%0),%3                     \n"
>+      "movd      0x00(%4,%3,4),%%xmm3            \n"
>+      "pshuflw   $0x40,%%xmm2,%%xmm2             \n"
>+      "pshuflw   $0x40,%%xmm3,%%xmm3             \n"
>+      "movlhps   %%xmm3,%%xmm2                   \n"
>+      "pmulhuw   %%xmm2,%%xmm0                   \n"
>+      "movdqu    (%0),%%xmm1                     \n"
>+      "movzb     0x0b(%0),%3                     \n"
>+      "punpckhbw %%xmm1,%%xmm1                   \n"
>+      "movd      0x00(%4,%3,4),%%xmm2            \n"
>+      "movzb     0x0f(%0),%3                     \n"
>+      "movd      0x00(%4,%3,4),%%xmm3            \n"
>+      "pshuflw   $0x40,%%xmm2,%%xmm2             \n"
>+      "pshuflw   $0x40,%%xmm3,%%xmm3             \n"
>+      "movlhps   %%xmm3,%%xmm2                   \n"
>+      "pmulhuw   %%xmm2,%%xmm1                   \n"
>+      "lea       0x10(%0),%0                     \n"
>+      "packuswb  %%xmm1,%%xmm0                   \n"
>+      "movdqu    %%xmm0,(%1)                     \n"
>+      "lea       0x10(%1),%1                     \n"
>+      "sub       $0x4,%2                         \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_argb),     // %0
>+        "+r"(dst_argb),     // %1
>+        "+r"(width),        // %2
>+        "=&r"(alpha)        // %3
>+      : "r"(fixed_invtbl8)  // %4
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5");
> }
> #endif  // HAS_ARGBUNATTENUATEROW_SSE2
> 
>@@ -4114,114 +4686,111 @@ void ARGBUnattenuateRow_SSE2(const uint8* src_argb,
> static const uvec8 kUnattenShuffleAlpha_AVX2 = {
>     0u, 1u, 0u, 1u, 0u, 1u, 6u, 7u, 8u, 9u, 8u, 9u, 8u, 9u, 14u, 15u};
> // Unattenuate 8 pixels at a time.
>-void ARGBUnattenuateRow_AVX2(const uint8* src_argb,
>-                             uint8* dst_argb,
>+void ARGBUnattenuateRow_AVX2(const uint8_t* src_argb,
>+                             uint8_t* dst_argb,
>                              int width) {
>   uintptr_t alpha;
>-  asm volatile (
>-    "sub        %0,%1                          \n"
>-    "vbroadcastf128 %5,%%ymm5                  \n"
>+  asm volatile(
>+      "sub        %0,%1                          \n"
>+      "vbroadcastf128 %5,%%ymm5                  \n"
> 
>-    // 8 pixel loop.
>-    LABELALIGN
>-    "1:                                        \n"
>-    // replace VPGATHER
>-    "movzb     " MEMACCESS2(0x03,0) ",%3       \n"
>-    MEMOPREG(vmovd,0x00,4,3,4,xmm0)             //  vmovd 0x0(%4,%3,4),%%xmm0
>-    "movzb     " MEMACCESS2(0x07,0) ",%3       \n"
>-    MEMOPREG(vmovd,0x00,4,3,4,xmm1)             //  vmovd 0x0(%4,%3,4),%%xmm1
>-    "movzb     " MEMACCESS2(0x0b,0) ",%3       \n"
>-    "vpunpckldq %%xmm1,%%xmm0,%%xmm6           \n"
>-    MEMOPREG(vmovd,0x00,4,3,4,xmm2)             //  vmovd 0x0(%4,%3,4),%%xmm2
>-    "movzb     " MEMACCESS2(0x0f,0) ",%3       \n"
>-    MEMOPREG(vmovd,0x00,4,3,4,xmm3)             //  vmovd 0x0(%4,%3,4),%%xmm3
>-    "movzb     " MEMACCESS2(0x13,0) ",%3       \n"
>-    "vpunpckldq %%xmm3,%%xmm2,%%xmm7           \n"
>-    MEMOPREG(vmovd,0x00,4,3,4,xmm0)             //  vmovd 0x0(%4,%3,4),%%xmm0
>-    "movzb     " MEMACCESS2(0x17,0) ",%3       \n"
>-    MEMOPREG(vmovd,0x00,4,3,4,xmm1)             //  vmovd 0x0(%4,%3,4),%%xmm1
>-    "movzb     " MEMACCESS2(0x1b,0) ",%3       \n"
>-    "vpunpckldq %%xmm1,%%xmm0,%%xmm0           \n"
>-    MEMOPREG(vmovd,0x00,4,3,4,xmm2)             //  vmovd 0x0(%4,%3,4),%%xmm2
>-    "movzb     " MEMACCESS2(0x1f,0) ",%3       \n"
>-    MEMOPREG(vmovd,0x00,4,3,4,xmm3)             //  vmovd 0x0(%4,%3,4),%%xmm3
>-    "vpunpckldq %%xmm3,%%xmm2,%%xmm2           \n"
>-    "vpunpcklqdq %%xmm7,%%xmm6,%%xmm3          \n"
>-    "vpunpcklqdq %%xmm2,%%xmm0,%%xmm0          \n"
>-    "vinserti128 $0x1,%%xmm0,%%ymm3,%%ymm3     \n"
>-    // end of VPGATHER
>-
>-    "vmovdqu    " MEMACCESS(0) ",%%ymm6        \n"
>-    "vpunpcklbw %%ymm6,%%ymm6,%%ymm0           \n"
>-    "vpunpckhbw %%ymm6,%%ymm6,%%ymm1           \n"
>-    "vpunpcklwd %%ymm3,%%ymm3,%%ymm2           \n"
>-    "vpunpckhwd %%ymm3,%%ymm3,%%ymm3           \n"
>-    "vpshufb    %%ymm5,%%ymm2,%%ymm2           \n"
>-    "vpshufb    %%ymm5,%%ymm3,%%ymm3           \n"
>-    "vpmulhuw   %%ymm2,%%ymm0,%%ymm0           \n"
>-    "vpmulhuw   %%ymm3,%%ymm1,%%ymm1           \n"
>-    "vpackuswb  %%ymm1,%%ymm0,%%ymm0           \n"
>-    MEMOPMEM(vmovdqu,ymm0,0x00,0,1,1)          //  vmovdqu %%ymm0,(%0,%1)
>-    "lea       " MEMLEA(0x20,0) ",%0           \n"
>-    "sub        $0x8,%2                        \n"
>-    "jg        1b                              \n"
>-    "vzeroupper                                \n"
>-  : "+r"(src_argb),      // %0
>-    "+r"(dst_argb),      // %1
>-    "+r"(width),         // %2
>-    "=&r"(alpha)         // %3
>-  : "r"(fixed_invtbl8),  // %4
>-    "m"(kUnattenShuffleAlpha_AVX2)  // %5
>-  : "memory", "cc", NACL_R14
>-    "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6", "xmm7"
>-  );
>+      // 8 pixel loop.
>+      LABELALIGN
>+      "1:                                        \n"
>+      // replace VPGATHER
>+      "movzb     0x03(%0),%3                     \n"
>+      "vmovd     0x00(%4,%3,4),%%xmm0            \n"
>+      "movzb     0x07(%0),%3                     \n"
>+      "vmovd     0x00(%4,%3,4),%%xmm1            \n"
>+      "movzb     0x0b(%0),%3                     \n"
>+      "vpunpckldq %%xmm1,%%xmm0,%%xmm6           \n"
>+      "vmovd     0x00(%4,%3,4),%%xmm2            \n"
>+      "movzb     0x0f(%0),%3                     \n"
>+      "vmovd     0x00(%4,%3,4),%%xmm3            \n"
>+      "movzb     0x13(%0),%3                     \n"
>+      "vpunpckldq %%xmm3,%%xmm2,%%xmm7           \n"
>+      "vmovd     0x00(%4,%3,4),%%xmm0            \n"
>+      "movzb     0x17(%0),%3                     \n"
>+      "vmovd     0x00(%4,%3,4),%%xmm1            \n"
>+      "movzb     0x1b(%0),%3                     \n"
>+      "vpunpckldq %%xmm1,%%xmm0,%%xmm0           \n"
>+      "vmovd     0x00(%4,%3,4),%%xmm2            \n"
>+      "movzb     0x1f(%0),%3                     \n"
>+      "vmovd     0x00(%4,%3,4),%%xmm3            \n"
>+      "vpunpckldq %%xmm3,%%xmm2,%%xmm2           \n"
>+      "vpunpcklqdq %%xmm7,%%xmm6,%%xmm3          \n"
>+      "vpunpcklqdq %%xmm2,%%xmm0,%%xmm0          \n"
>+      "vinserti128 $0x1,%%xmm0,%%ymm3,%%ymm3     \n"
>+      // end of VPGATHER
>+
>+      "vmovdqu    (%0),%%ymm6                    \n"
>+      "vpunpcklbw %%ymm6,%%ymm6,%%ymm0           \n"
>+      "vpunpckhbw %%ymm6,%%ymm6,%%ymm1           \n"
>+      "vpunpcklwd %%ymm3,%%ymm3,%%ymm2           \n"
>+      "vpunpckhwd %%ymm3,%%ymm3,%%ymm3           \n"
>+      "vpshufb    %%ymm5,%%ymm2,%%ymm2           \n"
>+      "vpshufb    %%ymm5,%%ymm3,%%ymm3           \n"
>+      "vpmulhuw   %%ymm2,%%ymm0,%%ymm0           \n"
>+      "vpmulhuw   %%ymm3,%%ymm1,%%ymm1           \n"
>+      "vpackuswb  %%ymm1,%%ymm0,%%ymm0           \n"
>+      "vmovdqu    %%ymm0,0x00(%0,%1,1)           \n"
>+      "lea       0x20(%0),%0                     \n"
>+      "sub        $0x8,%2                        \n"
>+      "jg        1b                              \n"
>+      "vzeroupper                                \n"
>+      : "+r"(src_argb),                 // %0
>+        "+r"(dst_argb),                 // %1
>+        "+r"(width),                    // %2
>+        "=&r"(alpha)                    // %3
>+      : "r"(fixed_invtbl8),             // %4
>+        "m"(kUnattenShuffleAlpha_AVX2)  // %5
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6",
>+        "xmm7");
> }
> #endif  // HAS_ARGBUNATTENUATEROW_AVX2
> 
> #ifdef HAS_ARGBGRAYROW_SSSE3
> // Convert 8 ARGB pixels (64 bytes) to 8 Gray ARGB pixels
>-void ARGBGrayRow_SSSE3(const uint8* src_argb, uint8* dst_argb, int width) {
>-  asm volatile (
>-    "movdqa    %3,%%xmm4                       \n"
>-    "movdqa    %4,%%xmm5                       \n"
>+void ARGBGrayRow_SSSE3(const uint8_t* src_argb, uint8_t* dst_argb, int width) {
>+  asm volatile(
>+      "movdqa    %3,%%xmm4                       \n"
>+      "movdqa    %4,%%xmm5                       \n"
> 
>-    // 8 pixel loop.
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    "movdqu    " MEMACCESS2(0x10,0) ",%%xmm1   \n"
>-    "pmaddubsw %%xmm4,%%xmm0                   \n"
>-    "pmaddubsw %%xmm4,%%xmm1                   \n"
>-    "phaddw    %%xmm1,%%xmm0                   \n"
>-    "paddw     %%xmm5,%%xmm0                   \n"
>-    "psrlw     $0x7,%%xmm0                     \n"
>-    "packuswb  %%xmm0,%%xmm0                   \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm2         \n"
>-    "movdqu    " MEMACCESS2(0x10,0) ",%%xmm3   \n"
>-    "lea       " MEMLEA(0x20,0) ",%0           \n"
>-    "psrld     $0x18,%%xmm2                    \n"
>-    "psrld     $0x18,%%xmm3                    \n"
>-    "packuswb  %%xmm3,%%xmm2                   \n"
>-    "packuswb  %%xmm2,%%xmm2                   \n"
>-    "movdqa    %%xmm0,%%xmm3                   \n"
>-    "punpcklbw %%xmm0,%%xmm0                   \n"
>-    "punpcklbw %%xmm2,%%xmm3                   \n"
>-    "movdqa    %%xmm0,%%xmm1                   \n"
>-    "punpcklwd %%xmm3,%%xmm0                   \n"
>-    "punpckhwd %%xmm3,%%xmm1                   \n"
>-    "movdqu    %%xmm0," MEMACCESS(1) "         \n"
>-    "movdqu    %%xmm1," MEMACCESS2(0x10,1) "   \n"
>-    "lea       " MEMLEA(0x20,1) ",%1           \n"
>-    "sub       $0x8,%2                         \n"
>-    "jg        1b                              \n"
>-  : "+r"(src_argb),   // %0
>-    "+r"(dst_argb),   // %1
>-    "+r"(width)       // %2
>-  : "m"(kARGBToYJ),   // %3
>-    "m"(kAddYJ64)     // %4
>-  : "memory", "cc"
>-    , "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5"
>-  );
>+      // 8 pixel loop.
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "movdqu    0x10(%0),%%xmm1                 \n"
>+      "pmaddubsw %%xmm4,%%xmm0                   \n"
>+      "pmaddubsw %%xmm4,%%xmm1                   \n"
>+      "phaddw    %%xmm1,%%xmm0                   \n"
>+      "paddw     %%xmm5,%%xmm0                   \n"
>+      "psrlw     $0x7,%%xmm0                     \n"
>+      "packuswb  %%xmm0,%%xmm0                   \n"
>+      "movdqu    (%0),%%xmm2                     \n"
>+      "movdqu    0x10(%0),%%xmm3                 \n"
>+      "lea       0x20(%0),%0                     \n"
>+      "psrld     $0x18,%%xmm2                    \n"
>+      "psrld     $0x18,%%xmm3                    \n"
>+      "packuswb  %%xmm3,%%xmm2                   \n"
>+      "packuswb  %%xmm2,%%xmm2                   \n"
>+      "movdqa    %%xmm0,%%xmm3                   \n"
>+      "punpcklbw %%xmm0,%%xmm0                   \n"
>+      "punpcklbw %%xmm2,%%xmm3                   \n"
>+      "movdqa    %%xmm0,%%xmm1                   \n"
>+      "punpcklwd %%xmm3,%%xmm0                   \n"
>+      "punpckhwd %%xmm3,%%xmm1                   \n"
>+      "movdqu    %%xmm0,(%1)                     \n"
>+      "movdqu    %%xmm1,0x10(%1)                 \n"
>+      "lea       0x20(%1),%1                     \n"
>+      "sub       $0x8,%2                         \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_argb),  // %0
>+        "+r"(dst_argb),  // %1
>+        "+r"(width)      // %2
>+      : "m"(kARGBToYJ),  // %3
>+        "m"(kAddYJ64)    // %4
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5");
> }
> #endif  // HAS_ARGBGRAYROW_SSSE3
> 
>@@ -4230,428 +4799,415 @@ void ARGBGrayRow_SSSE3(const uint8* src_argb, uint8* dst_argb, int width) {
> //    g = (r * 45 + g * 88 + b * 22) >> 7
> //    r = (r * 50 + g * 98 + b * 24) >> 7
> // Constant for ARGB color to sepia tone
>-static vec8 kARGBToSepiaB = {17, 68, 35, 0, 17, 68, 35, 0,
>-                             17, 68, 35, 0, 17, 68, 35, 0};
>+static const vec8 kARGBToSepiaB = {17, 68, 35, 0, 17, 68, 35, 0,
>+                                   17, 68, 35, 0, 17, 68, 35, 0};
> 
>-static vec8 kARGBToSepiaG = {22, 88, 45, 0, 22, 88, 45, 0,
>-                             22, 88, 45, 0, 22, 88, 45, 0};
>+static const vec8 kARGBToSepiaG = {22, 88, 45, 0, 22, 88, 45, 0,
>+                                   22, 88, 45, 0, 22, 88, 45, 0};
> 
>-static vec8 kARGBToSepiaR = {24, 98, 50, 0, 24, 98, 50, 0,
>-                             24, 98, 50, 0, 24, 98, 50, 0};
>+static const vec8 kARGBToSepiaR = {24, 98, 50, 0, 24, 98, 50, 0,
>+                                   24, 98, 50, 0, 24, 98, 50, 0};
> 
> // Convert 8 ARGB pixels (32 bytes) to 8 Sepia ARGB pixels.
>-void ARGBSepiaRow_SSSE3(uint8* dst_argb, int width) {
>-  asm volatile (
>-    "movdqa    %2,%%xmm2                       \n"
>-    "movdqa    %3,%%xmm3                       \n"
>-    "movdqa    %4,%%xmm4                       \n"
>+void ARGBSepiaRow_SSSE3(uint8_t* dst_argb, int width) {
>+  asm volatile(
>+      "movdqa    %2,%%xmm2                       \n"
>+      "movdqa    %3,%%xmm3                       \n"
>+      "movdqa    %4,%%xmm4                       \n"
> 
>-    // 8 pixel loop.
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    "movdqu    " MEMACCESS2(0x10,0) ",%%xmm6   \n"
>-    "pmaddubsw %%xmm2,%%xmm0                   \n"
>-    "pmaddubsw %%xmm2,%%xmm6                   \n"
>-    "phaddw    %%xmm6,%%xmm0                   \n"
>-    "psrlw     $0x7,%%xmm0                     \n"
>-    "packuswb  %%xmm0,%%xmm0                   \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm5         \n"
>-    "movdqu    " MEMACCESS2(0x10,0) ",%%xmm1   \n"
>-    "pmaddubsw %%xmm3,%%xmm5                   \n"
>-    "pmaddubsw %%xmm3,%%xmm1                   \n"
>-    "phaddw    %%xmm1,%%xmm5                   \n"
>-    "psrlw     $0x7,%%xmm5                     \n"
>-    "packuswb  %%xmm5,%%xmm5                   \n"
>-    "punpcklbw %%xmm5,%%xmm0                   \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm5         \n"
>-    "movdqu    " MEMACCESS2(0x10,0) ",%%xmm1   \n"
>-    "pmaddubsw %%xmm4,%%xmm5                   \n"
>-    "pmaddubsw %%xmm4,%%xmm1                   \n"
>-    "phaddw    %%xmm1,%%xmm5                   \n"
>-    "psrlw     $0x7,%%xmm5                     \n"
>-    "packuswb  %%xmm5,%%xmm5                   \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm6         \n"
>-    "movdqu    " MEMACCESS2(0x10,0) ",%%xmm1   \n"
>-    "psrld     $0x18,%%xmm6                    \n"
>-    "psrld     $0x18,%%xmm1                    \n"
>-    "packuswb  %%xmm1,%%xmm6                   \n"
>-    "packuswb  %%xmm6,%%xmm6                   \n"
>-    "punpcklbw %%xmm6,%%xmm5                   \n"
>-    "movdqa    %%xmm0,%%xmm1                   \n"
>-    "punpcklwd %%xmm5,%%xmm0                   \n"
>-    "punpckhwd %%xmm5,%%xmm1                   \n"
>-    "movdqu    %%xmm0," MEMACCESS(0) "         \n"
>-    "movdqu    %%xmm1," MEMACCESS2(0x10,0) "   \n"
>-    "lea       " MEMLEA(0x20,0) ",%0           \n"
>-    "sub       $0x8,%1                         \n"
>-    "jg        1b                              \n"
>-  : "+r"(dst_argb),      // %0
>-    "+r"(width)          // %1
>-  : "m"(kARGBToSepiaB),  // %2
>-    "m"(kARGBToSepiaG),  // %3
>-    "m"(kARGBToSepiaR)   // %4
>-  : "memory", "cc"
>-    , "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6"
>-  );
>+      // 8 pixel loop.
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "movdqu    0x10(%0),%%xmm6                 \n"
>+      "pmaddubsw %%xmm2,%%xmm0                   \n"
>+      "pmaddubsw %%xmm2,%%xmm6                   \n"
>+      "phaddw    %%xmm6,%%xmm0                   \n"
>+      "psrlw     $0x7,%%xmm0                     \n"
>+      "packuswb  %%xmm0,%%xmm0                   \n"
>+      "movdqu    (%0),%%xmm5                     \n"
>+      "movdqu    0x10(%0),%%xmm1                 \n"
>+      "pmaddubsw %%xmm3,%%xmm5                   \n"
>+      "pmaddubsw %%xmm3,%%xmm1                   \n"
>+      "phaddw    %%xmm1,%%xmm5                   \n"
>+      "psrlw     $0x7,%%xmm5                     \n"
>+      "packuswb  %%xmm5,%%xmm5                   \n"
>+      "punpcklbw %%xmm5,%%xmm0                   \n"
>+      "movdqu    (%0),%%xmm5                     \n"
>+      "movdqu    0x10(%0),%%xmm1                 \n"
>+      "pmaddubsw %%xmm4,%%xmm5                   \n"
>+      "pmaddubsw %%xmm4,%%xmm1                   \n"
>+      "phaddw    %%xmm1,%%xmm5                   \n"
>+      "psrlw     $0x7,%%xmm5                     \n"
>+      "packuswb  %%xmm5,%%xmm5                   \n"
>+      "movdqu    (%0),%%xmm6                     \n"
>+      "movdqu    0x10(%0),%%xmm1                 \n"
>+      "psrld     $0x18,%%xmm6                    \n"
>+      "psrld     $0x18,%%xmm1                    \n"
>+      "packuswb  %%xmm1,%%xmm6                   \n"
>+      "packuswb  %%xmm6,%%xmm6                   \n"
>+      "punpcklbw %%xmm6,%%xmm5                   \n"
>+      "movdqa    %%xmm0,%%xmm1                   \n"
>+      "punpcklwd %%xmm5,%%xmm0                   \n"
>+      "punpckhwd %%xmm5,%%xmm1                   \n"
>+      "movdqu    %%xmm0,(%0)                     \n"
>+      "movdqu    %%xmm1,0x10(%0)                 \n"
>+      "lea       0x20(%0),%0                     \n"
>+      "sub       $0x8,%1                         \n"
>+      "jg        1b                              \n"
>+      : "+r"(dst_argb),      // %0
>+        "+r"(width)          // %1
>+      : "m"(kARGBToSepiaB),  // %2
>+        "m"(kARGBToSepiaG),  // %3
>+        "m"(kARGBToSepiaR)   // %4
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6");
> }
> #endif  // HAS_ARGBSEPIAROW_SSSE3
> 
> #ifdef HAS_ARGBCOLORMATRIXROW_SSSE3
> // Tranform 8 ARGB pixels (32 bytes) with color matrix.
> // Same as Sepia except matrix is provided.
>-void ARGBColorMatrixRow_SSSE3(const uint8* src_argb,
>-                              uint8* dst_argb,
>-                              const int8* matrix_argb,
>+void ARGBColorMatrixRow_SSSE3(const uint8_t* src_argb,
>+                              uint8_t* dst_argb,
>+                              const int8_t* matrix_argb,
>                               int width) {
>-  asm volatile (
>-    "movdqu    " MEMACCESS(3) ",%%xmm5         \n"
>-    "pshufd    $0x00,%%xmm5,%%xmm2             \n"
>-    "pshufd    $0x55,%%xmm5,%%xmm3             \n"
>-    "pshufd    $0xaa,%%xmm5,%%xmm4             \n"
>-    "pshufd    $0xff,%%xmm5,%%xmm5             \n"
>+  asm volatile(
>+      "movdqu    (%3),%%xmm5                     \n"
>+      "pshufd    $0x00,%%xmm5,%%xmm2             \n"
>+      "pshufd    $0x55,%%xmm5,%%xmm3             \n"
>+      "pshufd    $0xaa,%%xmm5,%%xmm4             \n"
>+      "pshufd    $0xff,%%xmm5,%%xmm5             \n"
> 
>-    // 8 pixel loop.
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    "movdqu    " MEMACCESS2(0x10,0) ",%%xmm7   \n"
>-    "pmaddubsw %%xmm2,%%xmm0                   \n"
>-    "pmaddubsw %%xmm2,%%xmm7                   \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm6         \n"
>-    "movdqu    " MEMACCESS2(0x10,0) ",%%xmm1   \n"
>-    "pmaddubsw %%xmm3,%%xmm6                   \n"
>-    "pmaddubsw %%xmm3,%%xmm1                   \n"
>-    "phaddsw   %%xmm7,%%xmm0                   \n"
>-    "phaddsw   %%xmm1,%%xmm6                   \n"
>-    "psraw     $0x6,%%xmm0                     \n"
>-    "psraw     $0x6,%%xmm6                     \n"
>-    "packuswb  %%xmm0,%%xmm0                   \n"
>-    "packuswb  %%xmm6,%%xmm6                   \n"
>-    "punpcklbw %%xmm6,%%xmm0                   \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm1         \n"
>-    "movdqu    " MEMACCESS2(0x10,0) ",%%xmm7   \n"
>-    "pmaddubsw %%xmm4,%%xmm1                   \n"
>-    "pmaddubsw %%xmm4,%%xmm7                   \n"
>-    "phaddsw   %%xmm7,%%xmm1                   \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm6         \n"
>-    "movdqu    " MEMACCESS2(0x10,0) ",%%xmm7   \n"
>-    "pmaddubsw %%xmm5,%%xmm6                   \n"
>-    "pmaddubsw %%xmm5,%%xmm7                   \n"
>-    "phaddsw   %%xmm7,%%xmm6                   \n"
>-    "psraw     $0x6,%%xmm1                     \n"
>-    "psraw     $0x6,%%xmm6                     \n"
>-    "packuswb  %%xmm1,%%xmm1                   \n"
>-    "packuswb  %%xmm6,%%xmm6                   \n"
>-    "punpcklbw %%xmm6,%%xmm1                   \n"
>-    "movdqa    %%xmm0,%%xmm6                   \n"
>-    "punpcklwd %%xmm1,%%xmm0                   \n"
>-    "punpckhwd %%xmm1,%%xmm6                   \n"
>-    "movdqu    %%xmm0," MEMACCESS(1) "         \n"
>-    "movdqu    %%xmm6," MEMACCESS2(0x10,1) "   \n"
>-    "lea       " MEMLEA(0x20,0) ",%0           \n"
>-    "lea       " MEMLEA(0x20,1) ",%1           \n"
>-    "sub       $0x8,%2                         \n"
>-    "jg        1b                              \n"
>-  : "+r"(src_argb),      // %0
>-    "+r"(dst_argb),      // %1
>-    "+r"(width)          // %2
>-  : "r"(matrix_argb)     // %3
>-  : "memory", "cc"
>-    , "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6", "xmm7"
>-  );
>+      // 8 pixel loop.
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "movdqu    0x10(%0),%%xmm7                 \n"
>+      "pmaddubsw %%xmm2,%%xmm0                   \n"
>+      "pmaddubsw %%xmm2,%%xmm7                   \n"
>+      "movdqu    (%0),%%xmm6                     \n"
>+      "movdqu    0x10(%0),%%xmm1                 \n"
>+      "pmaddubsw %%xmm3,%%xmm6                   \n"
>+      "pmaddubsw %%xmm3,%%xmm1                   \n"
>+      "phaddsw   %%xmm7,%%xmm0                   \n"
>+      "phaddsw   %%xmm1,%%xmm6                   \n"
>+      "psraw     $0x6,%%xmm0                     \n"
>+      "psraw     $0x6,%%xmm6                     \n"
>+      "packuswb  %%xmm0,%%xmm0                   \n"
>+      "packuswb  %%xmm6,%%xmm6                   \n"
>+      "punpcklbw %%xmm6,%%xmm0                   \n"
>+      "movdqu    (%0),%%xmm1                     \n"
>+      "movdqu    0x10(%0),%%xmm7                 \n"
>+      "pmaddubsw %%xmm4,%%xmm1                   \n"
>+      "pmaddubsw %%xmm4,%%xmm7                   \n"
>+      "phaddsw   %%xmm7,%%xmm1                   \n"
>+      "movdqu    (%0),%%xmm6                     \n"
>+      "movdqu    0x10(%0),%%xmm7                 \n"
>+      "pmaddubsw %%xmm5,%%xmm6                   \n"
>+      "pmaddubsw %%xmm5,%%xmm7                   \n"
>+      "phaddsw   %%xmm7,%%xmm6                   \n"
>+      "psraw     $0x6,%%xmm1                     \n"
>+      "psraw     $0x6,%%xmm6                     \n"
>+      "packuswb  %%xmm1,%%xmm1                   \n"
>+      "packuswb  %%xmm6,%%xmm6                   \n"
>+      "punpcklbw %%xmm6,%%xmm1                   \n"
>+      "movdqa    %%xmm0,%%xmm6                   \n"
>+      "punpcklwd %%xmm1,%%xmm0                   \n"
>+      "punpckhwd %%xmm1,%%xmm6                   \n"
>+      "movdqu    %%xmm0,(%1)                     \n"
>+      "movdqu    %%xmm6,0x10(%1)                 \n"
>+      "lea       0x20(%0),%0                     \n"
>+      "lea       0x20(%1),%1                     \n"
>+      "sub       $0x8,%2                         \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_argb),   // %0
>+        "+r"(dst_argb),   // %1
>+        "+r"(width)       // %2
>+      : "r"(matrix_argb)  // %3
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6",
>+        "xmm7");
> }
> #endif  // HAS_ARGBCOLORMATRIXROW_SSSE3
> 
> #ifdef HAS_ARGBQUANTIZEROW_SSE2
> // Quantize 4 ARGB pixels (16 bytes).
>-void ARGBQuantizeRow_SSE2(uint8* dst_argb,
>+void ARGBQuantizeRow_SSE2(uint8_t* dst_argb,
>                           int scale,
>                           int interval_size,
>                           int interval_offset,
>                           int width) {
>-  asm volatile (
>-    "movd      %2,%%xmm2                       \n"
>-    "movd      %3,%%xmm3                       \n"
>-    "movd      %4,%%xmm4                       \n"
>-    "pshuflw   $0x40,%%xmm2,%%xmm2             \n"
>-    "pshufd    $0x44,%%xmm2,%%xmm2             \n"
>-    "pshuflw   $0x40,%%xmm3,%%xmm3             \n"
>-    "pshufd    $0x44,%%xmm3,%%xmm3             \n"
>-    "pshuflw   $0x40,%%xmm4,%%xmm4             \n"
>-    "pshufd    $0x44,%%xmm4,%%xmm4             \n"
>-    "pxor      %%xmm5,%%xmm5                   \n"
>-    "pcmpeqb   %%xmm6,%%xmm6                   \n"
>-    "pslld     $0x18,%%xmm6                    \n"
>-
>-    // 4 pixel loop.
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    "punpcklbw %%xmm5,%%xmm0                   \n"
>-    "pmulhuw   %%xmm2,%%xmm0                   \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm1         \n"
>-    "punpckhbw %%xmm5,%%xmm1                   \n"
>-    "pmulhuw   %%xmm2,%%xmm1                   \n"
>-    "pmullw    %%xmm3,%%xmm0                   \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm7         \n"
>-    "pmullw    %%xmm3,%%xmm1                   \n"
>-    "pand      %%xmm6,%%xmm7                   \n"
>-    "paddw     %%xmm4,%%xmm0                   \n"
>-    "paddw     %%xmm4,%%xmm1                   \n"
>-    "packuswb  %%xmm1,%%xmm0                   \n"
>-    "por       %%xmm7,%%xmm0                   \n"
>-    "movdqu    %%xmm0," MEMACCESS(0) "         \n"
>-    "lea       " MEMLEA(0x10,0) ",%0           \n"
>-    "sub       $0x4,%1                         \n"
>-    "jg        1b                              \n"
>-  : "+r"(dst_argb),       // %0
>-    "+r"(width)           // %1
>-  : "r"(scale),           // %2
>-    "r"(interval_size),   // %3
>-    "r"(interval_offset)  // %4
>-  : "memory", "cc"
>-    , "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6", "xmm7"
>-  );
>+  asm volatile(
>+      "movd      %2,%%xmm2                       \n"
>+      "movd      %3,%%xmm3                       \n"
>+      "movd      %4,%%xmm4                       \n"
>+      "pshuflw   $0x40,%%xmm2,%%xmm2             \n"
>+      "pshufd    $0x44,%%xmm2,%%xmm2             \n"
>+      "pshuflw   $0x40,%%xmm3,%%xmm3             \n"
>+      "pshufd    $0x44,%%xmm3,%%xmm3             \n"
>+      "pshuflw   $0x40,%%xmm4,%%xmm4             \n"
>+      "pshufd    $0x44,%%xmm4,%%xmm4             \n"
>+      "pxor      %%xmm5,%%xmm5                   \n"
>+      "pcmpeqb   %%xmm6,%%xmm6                   \n"
>+      "pslld     $0x18,%%xmm6                    \n"
>+
>+      // 4 pixel loop.
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "punpcklbw %%xmm5,%%xmm0                   \n"
>+      "pmulhuw   %%xmm2,%%xmm0                   \n"
>+      "movdqu    (%0),%%xmm1                     \n"
>+      "punpckhbw %%xmm5,%%xmm1                   \n"
>+      "pmulhuw   %%xmm2,%%xmm1                   \n"
>+      "pmullw    %%xmm3,%%xmm0                   \n"
>+      "movdqu    (%0),%%xmm7                     \n"
>+      "pmullw    %%xmm3,%%xmm1                   \n"
>+      "pand      %%xmm6,%%xmm7                   \n"
>+      "paddw     %%xmm4,%%xmm0                   \n"
>+      "paddw     %%xmm4,%%xmm1                   \n"
>+      "packuswb  %%xmm1,%%xmm0                   \n"
>+      "por       %%xmm7,%%xmm0                   \n"
>+      "movdqu    %%xmm0,(%0)                     \n"
>+      "lea       0x10(%0),%0                     \n"
>+      "sub       $0x4,%1                         \n"
>+      "jg        1b                              \n"
>+      : "+r"(dst_argb),       // %0
>+        "+r"(width)           // %1
>+      : "r"(scale),           // %2
>+        "r"(interval_size),   // %3
>+        "r"(interval_offset)  // %4
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6",
>+        "xmm7");
> }
> #endif  // HAS_ARGBQUANTIZEROW_SSE2
> 
> #ifdef HAS_ARGBSHADEROW_SSE2
> // Shade 4 pixels at a time by specified value.
>-void ARGBShadeRow_SSE2(const uint8* src_argb,
>-                       uint8* dst_argb,
>+void ARGBShadeRow_SSE2(const uint8_t* src_argb,
>+                       uint8_t* dst_argb,
>                        int width,
>-                       uint32 value) {
>-  asm volatile (
>-    "movd      %3,%%xmm2                       \n"
>-    "punpcklbw %%xmm2,%%xmm2                   \n"
>-    "punpcklqdq %%xmm2,%%xmm2                  \n"
>+                       uint32_t value) {
>+  asm volatile(
>+      "movd      %3,%%xmm2                       \n"
>+      "punpcklbw %%xmm2,%%xmm2                   \n"
>+      "punpcklqdq %%xmm2,%%xmm2                  \n"
> 
>-    // 4 pixel loop.
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    "lea       " MEMLEA(0x10,0) ",%0           \n"
>-    "movdqa    %%xmm0,%%xmm1                   \n"
>-    "punpcklbw %%xmm0,%%xmm0                   \n"
>-    "punpckhbw %%xmm1,%%xmm1                   \n"
>-    "pmulhuw   %%xmm2,%%xmm0                   \n"
>-    "pmulhuw   %%xmm2,%%xmm1                   \n"
>-    "psrlw     $0x8,%%xmm0                     \n"
>-    "psrlw     $0x8,%%xmm1                     \n"
>-    "packuswb  %%xmm1,%%xmm0                   \n"
>-    "movdqu    %%xmm0," MEMACCESS(1) "         \n"
>-    "lea       " MEMLEA(0x10,1) ",%1           \n"
>-    "sub       $0x4,%2                         \n"
>-    "jg        1b                              \n"
>-  : "+r"(src_argb),  // %0
>-    "+r"(dst_argb),  // %1
>-    "+r"(width)      // %2
>-  : "r"(value)       // %3
>-  : "memory", "cc"
>-    , "xmm0", "xmm1", "xmm2"
>-  );
>+      // 4 pixel loop.
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "lea       0x10(%0),%0                     \n"
>+      "movdqa    %%xmm0,%%xmm1                   \n"
>+      "punpcklbw %%xmm0,%%xmm0                   \n"
>+      "punpckhbw %%xmm1,%%xmm1                   \n"
>+      "pmulhuw   %%xmm2,%%xmm0                   \n"
>+      "pmulhuw   %%xmm2,%%xmm1                   \n"
>+      "psrlw     $0x8,%%xmm0                     \n"
>+      "psrlw     $0x8,%%xmm1                     \n"
>+      "packuswb  %%xmm1,%%xmm0                   \n"
>+      "movdqu    %%xmm0,(%1)                     \n"
>+      "lea       0x10(%1),%1                     \n"
>+      "sub       $0x4,%2                         \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_argb),  // %0
>+        "+r"(dst_argb),  // %1
>+        "+r"(width)      // %2
>+      : "r"(value)       // %3
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2");
> }
> #endif  // HAS_ARGBSHADEROW_SSE2
> 
> #ifdef HAS_ARGBMULTIPLYROW_SSE2
> // Multiply 2 rows of ARGB pixels together, 4 pixels at a time.
>-void ARGBMultiplyRow_SSE2(const uint8* src_argb0,
>-                          const uint8* src_argb1,
>-                          uint8* dst_argb,
>+void ARGBMultiplyRow_SSE2(const uint8_t* src_argb0,
>+                          const uint8_t* src_argb1,
>+                          uint8_t* dst_argb,
>                           int width) {
>-  asm volatile (
>-    "pxor      %%xmm5,%%xmm5                   \n"
>+  asm volatile(
> 
>-    // 4 pixel loop.
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    "lea       " MEMLEA(0x10,0) ",%0           \n"
>-    "movdqu    " MEMACCESS(1) ",%%xmm2         \n"
>-    "lea       " MEMLEA(0x10,1) ",%1           \n"
>-    "movdqu    %%xmm0,%%xmm1                   \n"
>-    "movdqu    %%xmm2,%%xmm3                   \n"
>-    "punpcklbw %%xmm0,%%xmm0                   \n"
>-    "punpckhbw %%xmm1,%%xmm1                   \n"
>-    "punpcklbw %%xmm5,%%xmm2                   \n"
>-    "punpckhbw %%xmm5,%%xmm3                   \n"
>-    "pmulhuw   %%xmm2,%%xmm0                   \n"
>-    "pmulhuw   %%xmm3,%%xmm1                   \n"
>-    "packuswb  %%xmm1,%%xmm0                   \n"
>-    "movdqu    %%xmm0," MEMACCESS(2) "         \n"
>-    "lea       " MEMLEA(0x10,2) ",%2           \n"
>-    "sub       $0x4,%3                         \n"
>-    "jg        1b                              \n"
>-  : "+r"(src_argb0),  // %0
>-    "+r"(src_argb1),  // %1
>-    "+r"(dst_argb),   // %2
>-    "+r"(width)       // %3
>-  :
>-  : "memory", "cc"
>-    , "xmm0", "xmm1", "xmm2", "xmm3", "xmm5"
>-  );
>+      "pxor      %%xmm5,%%xmm5                   \n"
>+
>+      // 4 pixel loop.
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "lea       0x10(%0),%0                     \n"
>+      "movdqu    (%1),%%xmm2                     \n"
>+      "lea       0x10(%1),%1                     \n"
>+      "movdqu    %%xmm0,%%xmm1                   \n"
>+      "movdqu    %%xmm2,%%xmm3                   \n"
>+      "punpcklbw %%xmm0,%%xmm0                   \n"
>+      "punpckhbw %%xmm1,%%xmm1                   \n"
>+      "punpcklbw %%xmm5,%%xmm2                   \n"
>+      "punpckhbw %%xmm5,%%xmm3                   \n"
>+      "pmulhuw   %%xmm2,%%xmm0                   \n"
>+      "pmulhuw   %%xmm3,%%xmm1                   \n"
>+      "packuswb  %%xmm1,%%xmm0                   \n"
>+      "movdqu    %%xmm0,(%2)                     \n"
>+      "lea       0x10(%2),%2                     \n"
>+      "sub       $0x4,%3                         \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_argb0),  // %0
>+        "+r"(src_argb1),  // %1
>+        "+r"(dst_argb),   // %2
>+        "+r"(width)       // %3
>+      :
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm5");
> }
> #endif  // HAS_ARGBMULTIPLYROW_SSE2
> 
> #ifdef HAS_ARGBMULTIPLYROW_AVX2
> // Multiply 2 rows of ARGB pixels together, 8 pixels at a time.
>-void ARGBMultiplyRow_AVX2(const uint8* src_argb0,
>-                          const uint8* src_argb1,
>-                          uint8* dst_argb,
>+void ARGBMultiplyRow_AVX2(const uint8_t* src_argb0,
>+                          const uint8_t* src_argb1,
>+                          uint8_t* dst_argb,
>                           int width) {
>-  asm volatile (
>-    "vpxor      %%ymm5,%%ymm5,%%ymm5           \n"
>+  asm volatile(
> 
>-    // 4 pixel loop.
>-    LABELALIGN
>-    "1:                                        \n"
>-    "vmovdqu    " MEMACCESS(0) ",%%ymm1        \n"
>-    "lea        " MEMLEA(0x20,0) ",%0          \n"
>-    "vmovdqu    " MEMACCESS(1) ",%%ymm3        \n"
>-    "lea        " MEMLEA(0x20,1) ",%1          \n"
>-    "vpunpcklbw %%ymm1,%%ymm1,%%ymm0           \n"
>-    "vpunpckhbw %%ymm1,%%ymm1,%%ymm1           \n"
>-    "vpunpcklbw %%ymm5,%%ymm3,%%ymm2           \n"
>-    "vpunpckhbw %%ymm5,%%ymm3,%%ymm3           \n"
>-    "vpmulhuw   %%ymm2,%%ymm0,%%ymm0           \n"
>-    "vpmulhuw   %%ymm3,%%ymm1,%%ymm1           \n"
>-    "vpackuswb  %%ymm1,%%ymm0,%%ymm0           \n"
>-    "vmovdqu    %%ymm0," MEMACCESS(2) "        \n"
>-    "lea       " MEMLEA(0x20,2) ",%2           \n"
>-    "sub        $0x8,%3                        \n"
>-    "jg        1b                              \n"
>-    "vzeroupper                                \n"
>-  : "+r"(src_argb0),  // %0
>-    "+r"(src_argb1),  // %1
>-    "+r"(dst_argb),   // %2
>-    "+r"(width)       // %3
>-  :
>-  : "memory", "cc"
>+      "vpxor      %%ymm5,%%ymm5,%%ymm5           \n"
>+
>+      // 4 pixel loop.
>+      LABELALIGN
>+      "1:                                        \n"
>+      "vmovdqu    (%0),%%ymm1                    \n"
>+      "lea        0x20(%0),%0                    \n"
>+      "vmovdqu    (%1),%%ymm3                    \n"
>+      "lea        0x20(%1),%1                    \n"
>+      "vpunpcklbw %%ymm1,%%ymm1,%%ymm0           \n"
>+      "vpunpckhbw %%ymm1,%%ymm1,%%ymm1           \n"
>+      "vpunpcklbw %%ymm5,%%ymm3,%%ymm2           \n"
>+      "vpunpckhbw %%ymm5,%%ymm3,%%ymm3           \n"
>+      "vpmulhuw   %%ymm2,%%ymm0,%%ymm0           \n"
>+      "vpmulhuw   %%ymm3,%%ymm1,%%ymm1           \n"
>+      "vpackuswb  %%ymm1,%%ymm0,%%ymm0           \n"
>+      "vmovdqu    %%ymm0,(%2)                    \n"
>+      "lea       0x20(%2),%2                     \n"
>+      "sub        $0x8,%3                        \n"
>+      "jg        1b                              \n"
>+      "vzeroupper                                \n"
>+      : "+r"(src_argb0),  // %0
>+        "+r"(src_argb1),  // %1
>+        "+r"(dst_argb),   // %2
>+        "+r"(width)       // %3
>+      :
>+      : "memory", "cc"
> #if defined(__AVX2__)
>-    , "xmm0", "xmm1", "xmm2", "xmm3", "xmm5"
>+        ,
>+        "xmm0", "xmm1", "xmm2", "xmm3", "xmm5"
> #endif
>-  );
>+      );
> }
> #endif  // HAS_ARGBMULTIPLYROW_AVX2
> 
> #ifdef HAS_ARGBADDROW_SSE2
> // Add 2 rows of ARGB pixels together, 4 pixels at a time.
>-void ARGBAddRow_SSE2(const uint8* src_argb0,
>-                     const uint8* src_argb1,
>-                     uint8* dst_argb,
>+void ARGBAddRow_SSE2(const uint8_t* src_argb0,
>+                     const uint8_t* src_argb1,
>+                     uint8_t* dst_argb,
>                      int width) {
>-  asm volatile (
>-    // 4 pixel loop.
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    "lea       " MEMLEA(0x10,0) ",%0           \n"
>-    "movdqu    " MEMACCESS(1) ",%%xmm1         \n"
>-    "lea       " MEMLEA(0x10,1) ",%1           \n"
>-    "paddusb   %%xmm1,%%xmm0                   \n"
>-    "movdqu    %%xmm0," MEMACCESS(2) "         \n"
>-    "lea       " MEMLEA(0x10,2) ",%2           \n"
>-    "sub       $0x4,%3                         \n"
>-    "jg        1b                              \n"
>-  : "+r"(src_argb0),  // %0
>-    "+r"(src_argb1),  // %1
>-    "+r"(dst_argb),   // %2
>-    "+r"(width)       // %3
>-  :
>-  : "memory", "cc"
>-    , "xmm0", "xmm1"
>-  );
>+  asm volatile(
>+      // 4 pixel loop.
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "lea       0x10(%0),%0                     \n"
>+      "movdqu    (%1),%%xmm1                     \n"
>+      "lea       0x10(%1),%1                     \n"
>+      "paddusb   %%xmm1,%%xmm0                   \n"
>+      "movdqu    %%xmm0,(%2)                     \n"
>+      "lea       0x10(%2),%2                     \n"
>+      "sub       $0x4,%3                         \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_argb0),  // %0
>+        "+r"(src_argb1),  // %1
>+        "+r"(dst_argb),   // %2
>+        "+r"(width)       // %3
>+      :
>+      : "memory", "cc", "xmm0", "xmm1");
> }
> #endif  // HAS_ARGBADDROW_SSE2
> 
> #ifdef HAS_ARGBADDROW_AVX2
> // Add 2 rows of ARGB pixels together, 4 pixels at a time.
>-void ARGBAddRow_AVX2(const uint8* src_argb0,
>-                     const uint8* src_argb1,
>-                     uint8* dst_argb,
>+void ARGBAddRow_AVX2(const uint8_t* src_argb0,
>+                     const uint8_t* src_argb1,
>+                     uint8_t* dst_argb,
>                      int width) {
>-  asm volatile (
>-    // 4 pixel loop.
>-    LABELALIGN
>-    "1:                                        \n"
>-    "vmovdqu    " MEMACCESS(0) ",%%ymm0        \n"
>-    "lea        " MEMLEA(0x20,0) ",%0          \n"
>-    "vpaddusb   " MEMACCESS(1) ",%%ymm0,%%ymm0 \n"
>-    "lea        " MEMLEA(0x20,1) ",%1          \n"
>-    "vmovdqu    %%ymm0," MEMACCESS(2) "        \n"
>-    "lea        " MEMLEA(0x20,2) ",%2          \n"
>-    "sub        $0x8,%3                        \n"
>-    "jg        1b                              \n"
>-    "vzeroupper                                \n"
>-  : "+r"(src_argb0),  // %0
>-    "+r"(src_argb1),  // %1
>-    "+r"(dst_argb),   // %2
>-    "+r"(width)       // %3
>-  :
>-  : "memory", "cc"
>-    , "xmm0"
>-  );
>+  asm volatile(
>+      // 4 pixel loop.
>+      LABELALIGN
>+      "1:                                        \n"
>+      "vmovdqu    (%0),%%ymm0                    \n"
>+      "lea        0x20(%0),%0                    \n"
>+      "vpaddusb   (%1),%%ymm0,%%ymm0             \n"
>+      "lea        0x20(%1),%1                    \n"
>+      "vmovdqu    %%ymm0,(%2)                    \n"
>+      "lea        0x20(%2),%2                    \n"
>+      "sub        $0x8,%3                        \n"
>+      "jg        1b                              \n"
>+      "vzeroupper                                \n"
>+      : "+r"(src_argb0),  // %0
>+        "+r"(src_argb1),  // %1
>+        "+r"(dst_argb),   // %2
>+        "+r"(width)       // %3
>+      :
>+      : "memory", "cc", "xmm0");
> }
> #endif  // HAS_ARGBADDROW_AVX2
> 
> #ifdef HAS_ARGBSUBTRACTROW_SSE2
> // Subtract 2 rows of ARGB pixels, 4 pixels at a time.
>-void ARGBSubtractRow_SSE2(const uint8* src_argb0,
>-                          const uint8* src_argb1,
>-                          uint8* dst_argb,
>+void ARGBSubtractRow_SSE2(const uint8_t* src_argb0,
>+                          const uint8_t* src_argb1,
>+                          uint8_t* dst_argb,
>                           int width) {
>-  asm volatile (
>-    // 4 pixel loop.
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    "lea       " MEMLEA(0x10,0) ",%0           \n"
>-    "movdqu    " MEMACCESS(1) ",%%xmm1         \n"
>-    "lea       " MEMLEA(0x10,1) ",%1           \n"
>-    "psubusb   %%xmm1,%%xmm0                   \n"
>-    "movdqu    %%xmm0," MEMACCESS(2) "         \n"
>-    "lea       " MEMLEA(0x10,2) ",%2           \n"
>-    "sub       $0x4,%3                         \n"
>-    "jg        1b                              \n"
>-  : "+r"(src_argb0),  // %0
>-    "+r"(src_argb1),  // %1
>-    "+r"(dst_argb),   // %2
>-    "+r"(width)       // %3
>-  :
>-  : "memory", "cc"
>-    , "xmm0", "xmm1"
>-  );
>+  asm volatile(
>+      // 4 pixel loop.
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "lea       0x10(%0),%0                     \n"
>+      "movdqu    (%1),%%xmm1                     \n"
>+      "lea       0x10(%1),%1                     \n"
>+      "psubusb   %%xmm1,%%xmm0                   \n"
>+      "movdqu    %%xmm0,(%2)                     \n"
>+      "lea       0x10(%2),%2                     \n"
>+      "sub       $0x4,%3                         \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_argb0),  // %0
>+        "+r"(src_argb1),  // %1
>+        "+r"(dst_argb),   // %2
>+        "+r"(width)       // %3
>+      :
>+      : "memory", "cc", "xmm0", "xmm1");
> }
> #endif  // HAS_ARGBSUBTRACTROW_SSE2
> 
> #ifdef HAS_ARGBSUBTRACTROW_AVX2
> // Subtract 2 rows of ARGB pixels, 8 pixels at a time.
>-void ARGBSubtractRow_AVX2(const uint8* src_argb0,
>-                          const uint8* src_argb1,
>-                          uint8* dst_argb,
>+void ARGBSubtractRow_AVX2(const uint8_t* src_argb0,
>+                          const uint8_t* src_argb1,
>+                          uint8_t* dst_argb,
>                           int width) {
>-  asm volatile (
>-    // 4 pixel loop.
>-    LABELALIGN
>-    "1:                                        \n"
>-    "vmovdqu    " MEMACCESS(0) ",%%ymm0        \n"
>-    "lea        " MEMLEA(0x20,0) ",%0          \n"
>-    "vpsubusb   " MEMACCESS(1) ",%%ymm0,%%ymm0 \n"
>-    "lea        " MEMLEA(0x20,1) ",%1          \n"
>-    "vmovdqu    %%ymm0," MEMACCESS(2) "        \n"
>-    "lea        " MEMLEA(0x20,2) ",%2          \n"
>-    "sub        $0x8,%3                        \n"
>-    "jg         1b                             \n"
>-    "vzeroupper                                \n"
>-  : "+r"(src_argb0),  // %0
>-    "+r"(src_argb1),  // %1
>-    "+r"(dst_argb),   // %2
>-    "+r"(width)       // %3
>-  :
>-  : "memory", "cc"
>-    , "xmm0"
>-  );
>+  asm volatile(
>+      // 4 pixel loop.
>+      LABELALIGN
>+      "1:                                        \n"
>+      "vmovdqu    (%0),%%ymm0                    \n"
>+      "lea        0x20(%0),%0                    \n"
>+      "vpsubusb   (%1),%%ymm0,%%ymm0             \n"
>+      "lea        0x20(%1),%1                    \n"
>+      "vmovdqu    %%ymm0,(%2)                    \n"
>+      "lea        0x20(%2),%2                    \n"
>+      "sub        $0x8,%3                        \n"
>+      "jg         1b                             \n"
>+      "vzeroupper                                \n"
>+      : "+r"(src_argb0),  // %0
>+        "+r"(src_argb1),  // %1
>+        "+r"(dst_argb),   // %2
>+        "+r"(width)       // %3
>+      :
>+      : "memory", "cc", "xmm0");
> }
> #endif  // HAS_ARGBSUBTRACTROW_AVX2
> 
>@@ -4660,55 +5216,53 @@ void ARGBSubtractRow_AVX2(const uint8* src_argb0,
> // -1  0  1
> // -2  0  2
> // -1  0  1
>-void SobelXRow_SSE2(const uint8* src_y0,
>-                    const uint8* src_y1,
>-                    const uint8* src_y2,
>-                    uint8* dst_sobelx,
>+void SobelXRow_SSE2(const uint8_t* src_y0,
>+                    const uint8_t* src_y1,
>+                    const uint8_t* src_y2,
>+                    uint8_t* dst_sobelx,
>                     int width) {
>-  asm volatile (
>-    "sub       %0,%1                           \n"
>-    "sub       %0,%2                           \n"
>-    "sub       %0,%3                           \n"
>-    "pxor      %%xmm5,%%xmm5                   \n"
>+  asm volatile(
>+      "sub       %0,%1                           \n"
>+      "sub       %0,%2                           \n"
>+      "sub       %0,%3                           \n"
>+      "pxor      %%xmm5,%%xmm5                   \n"
> 
>-    // 8 pixel loop.
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movq      " MEMACCESS(0) ",%%xmm0         \n"
>-    "movq      " MEMACCESS2(0x2,0) ",%%xmm1    \n"
>-    "punpcklbw %%xmm5,%%xmm0                   \n"
>-    "punpcklbw %%xmm5,%%xmm1                   \n"
>-    "psubw     %%xmm1,%%xmm0                   \n"
>-    MEMOPREG(movq,0x00,0,1,1,xmm1)             //  movq      (%0,%1,1),%%xmm1
>-    MEMOPREG(movq,0x02,0,1,1,xmm2)             //  movq      0x2(%0,%1,1),%%xmm2
>-    "punpcklbw %%xmm5,%%xmm1                   \n"
>-    "punpcklbw %%xmm5,%%xmm2                   \n"
>-    "psubw     %%xmm2,%%xmm1                   \n"
>-    MEMOPREG(movq,0x00,0,2,1,xmm2)             //  movq      (%0,%2,1),%%xmm2
>-    MEMOPREG(movq,0x02,0,2,1,xmm3)             //  movq      0x2(%0,%2,1),%%xmm3
>-    "punpcklbw %%xmm5,%%xmm2                   \n"
>-    "punpcklbw %%xmm5,%%xmm3                   \n"
>-    "psubw     %%xmm3,%%xmm2                   \n"
>-    "paddw     %%xmm2,%%xmm0                   \n"
>-    "paddw     %%xmm1,%%xmm0                   \n"
>-    "paddw     %%xmm1,%%xmm0                   \n"
>-    "pxor      %%xmm1,%%xmm1                   \n"
>-    "psubw     %%xmm0,%%xmm1                   \n"
>-    "pmaxsw    %%xmm1,%%xmm0                   \n"
>-    "packuswb  %%xmm0,%%xmm0                   \n"
>-    MEMOPMEM(movq,xmm0,0x00,0,3,1)             //  movq      %%xmm0,(%0,%3,1)
>-    "lea       " MEMLEA(0x8,0) ",%0            \n"
>-    "sub       $0x8,%4                         \n"
>-    "jg        1b                              \n"
>-  : "+r"(src_y0),      // %0
>-    "+r"(src_y1),      // %1
>-    "+r"(src_y2),      // %2
>-    "+r"(dst_sobelx),  // %3
>-    "+r"(width)        // %4
>-  :
>-  : "memory", "cc", NACL_R14
>-    "xmm0", "xmm1", "xmm2", "xmm3", "xmm5"
>-  );
>+      // 8 pixel loop.
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movq      (%0),%%xmm0                     \n"
>+      "movq      0x2(%0),%%xmm1                  \n"
>+      "punpcklbw %%xmm5,%%xmm0                   \n"
>+      "punpcklbw %%xmm5,%%xmm1                   \n"
>+      "psubw     %%xmm1,%%xmm0                   \n"
>+      "movq      0x00(%0,%1,1),%%xmm1            \n"
>+      "movq      0x02(%0,%1,1),%%xmm2            \n"
>+      "punpcklbw %%xmm5,%%xmm1                   \n"
>+      "punpcklbw %%xmm5,%%xmm2                   \n"
>+      "psubw     %%xmm2,%%xmm1                   \n"
>+      "movq      0x00(%0,%2,1),%%xmm2            \n"
>+      "movq      0x02(%0,%2,1),%%xmm3            \n"
>+      "punpcklbw %%xmm5,%%xmm2                   \n"
>+      "punpcklbw %%xmm5,%%xmm3                   \n"
>+      "psubw     %%xmm3,%%xmm2                   \n"
>+      "paddw     %%xmm2,%%xmm0                   \n"
>+      "paddw     %%xmm1,%%xmm0                   \n"
>+      "paddw     %%xmm1,%%xmm0                   \n"
>+      "pxor      %%xmm1,%%xmm1                   \n"
>+      "psubw     %%xmm0,%%xmm1                   \n"
>+      "pmaxsw    %%xmm1,%%xmm0                   \n"
>+      "packuswb  %%xmm0,%%xmm0                   \n"
>+      "movq      %%xmm0,0x00(%0,%3,1)            \n"
>+      "lea       0x8(%0),%0                      \n"
>+      "sub       $0x8,%4                         \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_y0),      // %0
>+        "+r"(src_y1),      // %1
>+        "+r"(src_y2),      // %2
>+        "+r"(dst_sobelx),  // %3
>+        "+r"(width)        // %4
>+      :
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm5");
> }
> #endif  // HAS_SOBELXROW_SSE2
> 
>@@ -4717,52 +5271,50 @@ void SobelXRow_SSE2(const uint8* src_y0,
> // -1 -2 -1
> //  0  0  0
> //  1  2  1
>-void SobelYRow_SSE2(const uint8* src_y0,
>-                    const uint8* src_y1,
>-                    uint8* dst_sobely,
>+void SobelYRow_SSE2(const uint8_t* src_y0,
>+                    const uint8_t* src_y1,
>+                    uint8_t* dst_sobely,
>                     int width) {
>-  asm volatile (
>-    "sub       %0,%1                           \n"
>-    "sub       %0,%2                           \n"
>-    "pxor      %%xmm5,%%xmm5                   \n"
>+  asm volatile(
>+      "sub       %0,%1                           \n"
>+      "sub       %0,%2                           \n"
>+      "pxor      %%xmm5,%%xmm5                   \n"
> 
>-    // 8 pixel loop.
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movq      " MEMACCESS(0) ",%%xmm0         \n"
>-    MEMOPREG(movq,0x00,0,1,1,xmm1)             //  movq      (%0,%1,1),%%xmm1
>-    "punpcklbw %%xmm5,%%xmm0                   \n"
>-    "punpcklbw %%xmm5,%%xmm1                   \n"
>-    "psubw     %%xmm1,%%xmm0                   \n"
>-    "movq      " MEMACCESS2(0x1,0) ",%%xmm1    \n"
>-    MEMOPREG(movq,0x01,0,1,1,xmm2)             //  movq      0x1(%0,%1,1),%%xmm2
>-    "punpcklbw %%xmm5,%%xmm1                   \n"
>-    "punpcklbw %%xmm5,%%xmm2                   \n"
>-    "psubw     %%xmm2,%%xmm1                   \n"
>-    "movq      " MEMACCESS2(0x2,0) ",%%xmm2    \n"
>-    MEMOPREG(movq,0x02,0,1,1,xmm3)             //  movq      0x2(%0,%1,1),%%xmm3
>-    "punpcklbw %%xmm5,%%xmm2                   \n"
>-    "punpcklbw %%xmm5,%%xmm3                   \n"
>-    "psubw     %%xmm3,%%xmm2                   \n"
>-    "paddw     %%xmm2,%%xmm0                   \n"
>-    "paddw     %%xmm1,%%xmm0                   \n"
>-    "paddw     %%xmm1,%%xmm0                   \n"
>-    "pxor      %%xmm1,%%xmm1                   \n"
>-    "psubw     %%xmm0,%%xmm1                   \n"
>-    "pmaxsw    %%xmm1,%%xmm0                   \n"
>-    "packuswb  %%xmm0,%%xmm0                   \n"
>-    MEMOPMEM(movq,xmm0,0x00,0,2,1)             //  movq      %%xmm0,(%0,%2,1)
>-    "lea       " MEMLEA(0x8,0) ",%0            \n"
>-    "sub       $0x8,%3                         \n"
>-    "jg        1b                              \n"
>-  : "+r"(src_y0),      // %0
>-    "+r"(src_y1),      // %1
>-    "+r"(dst_sobely),  // %2
>-    "+r"(width)        // %3
>-  :
>-  : "memory", "cc", NACL_R14
>-    "xmm0", "xmm1", "xmm2", "xmm3", "xmm5"
>-  );
>+      // 8 pixel loop.
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movq      (%0),%%xmm0                     \n"
>+      "movq      0x00(%0,%1,1),%%xmm1            \n"
>+      "punpcklbw %%xmm5,%%xmm0                   \n"
>+      "punpcklbw %%xmm5,%%xmm1                   \n"
>+      "psubw     %%xmm1,%%xmm0                   \n"
>+      "movq      0x1(%0),%%xmm1                  \n"
>+      "movq      0x01(%0,%1,1),%%xmm2            \n"
>+      "punpcklbw %%xmm5,%%xmm1                   \n"
>+      "punpcklbw %%xmm5,%%xmm2                   \n"
>+      "psubw     %%xmm2,%%xmm1                   \n"
>+      "movq      0x2(%0),%%xmm2                  \n"
>+      "movq      0x02(%0,%1,1),%%xmm3            \n"
>+      "punpcklbw %%xmm5,%%xmm2                   \n"
>+      "punpcklbw %%xmm5,%%xmm3                   \n"
>+      "psubw     %%xmm3,%%xmm2                   \n"
>+      "paddw     %%xmm2,%%xmm0                   \n"
>+      "paddw     %%xmm1,%%xmm0                   \n"
>+      "paddw     %%xmm1,%%xmm0                   \n"
>+      "pxor      %%xmm1,%%xmm1                   \n"
>+      "psubw     %%xmm0,%%xmm1                   \n"
>+      "pmaxsw    %%xmm1,%%xmm0                   \n"
>+      "packuswb  %%xmm0,%%xmm0                   \n"
>+      "movq      %%xmm0,0x00(%0,%2,1)            \n"
>+      "lea       0x8(%0),%0                      \n"
>+      "sub       $0x8,%3                         \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_y0),      // %0
>+        "+r"(src_y1),      // %1
>+        "+r"(dst_sobely),  // %2
>+        "+r"(width)        // %3
>+      :
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm5");
> }
> #endif  // HAS_SOBELYROW_SSE2
> 
>@@ -4772,83 +5324,79 @@ void SobelYRow_SSE2(const uint8* src_y0,
> // R = Sobel
> // G = Sobel
> // B = Sobel
>-void SobelRow_SSE2(const uint8* src_sobelx,
>-                   const uint8* src_sobely,
>-                   uint8* dst_argb,
>+void SobelRow_SSE2(const uint8_t* src_sobelx,
>+                   const uint8_t* src_sobely,
>+                   uint8_t* dst_argb,
>                    int width) {
>-  asm volatile (
>-    "sub       %0,%1                           \n"
>-    "pcmpeqb   %%xmm5,%%xmm5                   \n"
>-    "pslld     $0x18,%%xmm5                    \n"
>+  asm volatile(
>+      "sub       %0,%1                           \n"
>+      "pcmpeqb   %%xmm5,%%xmm5                   \n"
>+      "pslld     $0x18,%%xmm5                    \n"
> 
>-    // 8 pixel loop.
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    MEMOPREG(movdqu,0x00,0,1,1,xmm1)           //  movdqu    (%0,%1,1),%%xmm1
>-    "lea       " MEMLEA(0x10,0) ",%0           \n"
>-    "paddusb   %%xmm1,%%xmm0                   \n"
>-    "movdqa    %%xmm0,%%xmm2                   \n"
>-    "punpcklbw %%xmm0,%%xmm2                   \n"
>-    "punpckhbw %%xmm0,%%xmm0                   \n"
>-    "movdqa    %%xmm2,%%xmm1                   \n"
>-    "punpcklwd %%xmm2,%%xmm1                   \n"
>-    "punpckhwd %%xmm2,%%xmm2                   \n"
>-    "por       %%xmm5,%%xmm1                   \n"
>-    "por       %%xmm5,%%xmm2                   \n"
>-    "movdqa    %%xmm0,%%xmm3                   \n"
>-    "punpcklwd %%xmm0,%%xmm3                   \n"
>-    "punpckhwd %%xmm0,%%xmm0                   \n"
>-    "por       %%xmm5,%%xmm3                   \n"
>-    "por       %%xmm5,%%xmm0                   \n"
>-    "movdqu    %%xmm1," MEMACCESS(2) "         \n"
>-    "movdqu    %%xmm2," MEMACCESS2(0x10,2) "   \n"
>-    "movdqu    %%xmm3," MEMACCESS2(0x20,2) "   \n"
>-    "movdqu    %%xmm0," MEMACCESS2(0x30,2) "   \n"
>-    "lea       " MEMLEA(0x40,2) ",%2           \n"
>-    "sub       $0x10,%3                        \n"
>-    "jg        1b                              \n"
>-  : "+r"(src_sobelx),  // %0
>-    "+r"(src_sobely),  // %1
>-    "+r"(dst_argb),    // %2
>-    "+r"(width)        // %3
>-  :
>-  : "memory", "cc", NACL_R14
>-    "xmm0", "xmm1", "xmm2", "xmm3", "xmm5"
>-  );
>+      // 8 pixel loop.
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "movdqu    0x00(%0,%1,1),%%xmm1            \n"
>+      "lea       0x10(%0),%0                     \n"
>+      "paddusb   %%xmm1,%%xmm0                   \n"
>+      "movdqa    %%xmm0,%%xmm2                   \n"
>+      "punpcklbw %%xmm0,%%xmm2                   \n"
>+      "punpckhbw %%xmm0,%%xmm0                   \n"
>+      "movdqa    %%xmm2,%%xmm1                   \n"
>+      "punpcklwd %%xmm2,%%xmm1                   \n"
>+      "punpckhwd %%xmm2,%%xmm2                   \n"
>+      "por       %%xmm5,%%xmm1                   \n"
>+      "por       %%xmm5,%%xmm2                   \n"
>+      "movdqa    %%xmm0,%%xmm3                   \n"
>+      "punpcklwd %%xmm0,%%xmm3                   \n"
>+      "punpckhwd %%xmm0,%%xmm0                   \n"
>+      "por       %%xmm5,%%xmm3                   \n"
>+      "por       %%xmm5,%%xmm0                   \n"
>+      "movdqu    %%xmm1,(%2)                     \n"
>+      "movdqu    %%xmm2,0x10(%2)                 \n"
>+      "movdqu    %%xmm3,0x20(%2)                 \n"
>+      "movdqu    %%xmm0,0x30(%2)                 \n"
>+      "lea       0x40(%2),%2                     \n"
>+      "sub       $0x10,%3                        \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_sobelx),  // %0
>+        "+r"(src_sobely),  // %1
>+        "+r"(dst_argb),    // %2
>+        "+r"(width)        // %3
>+      :
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm5");
> }
> #endif  // HAS_SOBELROW_SSE2
> 
> #ifdef HAS_SOBELTOPLANEROW_SSE2
> // Adds Sobel X and Sobel Y and stores Sobel into a plane.
>-void SobelToPlaneRow_SSE2(const uint8* src_sobelx,
>-                          const uint8* src_sobely,
>-                          uint8* dst_y,
>+void SobelToPlaneRow_SSE2(const uint8_t* src_sobelx,
>+                          const uint8_t* src_sobely,
>+                          uint8_t* dst_y,
>                           int width) {
>-  asm volatile (
>-    "sub       %0,%1                           \n"
>-    "pcmpeqb   %%xmm5,%%xmm5                   \n"
>-    "pslld     $0x18,%%xmm5                    \n"
>+  asm volatile(
>+      "sub       %0,%1                           \n"
>+      "pcmpeqb   %%xmm5,%%xmm5                   \n"
>+      "pslld     $0x18,%%xmm5                    \n"
> 
>-    // 8 pixel loop.
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    MEMOPREG(movdqu,0x00,0,1,1,xmm1)           //  movdqu    (%0,%1,1),%%xmm1
>-    "lea       " MEMLEA(0x10,0) ",%0           \n"
>-    "paddusb   %%xmm1,%%xmm0                   \n"
>-    "movdqu    %%xmm0," MEMACCESS(2) "         \n"
>-    "lea       " MEMLEA(0x10,2) ",%2           \n"
>-    "sub       $0x10,%3                        \n"
>-    "jg        1b                              \n"
>-  : "+r"(src_sobelx),  // %0
>-    "+r"(src_sobely),  // %1
>-    "+r"(dst_y),       // %2
>-    "+r"(width)        // %3
>-  :
>-  : "memory", "cc", NACL_R14
>-    "xmm0", "xmm1"
>-  );
>+      // 8 pixel loop.
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "movdqu    0x00(%0,%1,1),%%xmm1            \n"
>+      "lea       0x10(%0),%0                     \n"
>+      "paddusb   %%xmm1,%%xmm0                   \n"
>+      "movdqu    %%xmm0,(%2)                     \n"
>+      "lea       0x10(%2),%2                     \n"
>+      "sub       $0x10,%3                        \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_sobelx),  // %0
>+        "+r"(src_sobely),  // %1
>+        "+r"(dst_y),       // %2
>+        "+r"(width)        // %3
>+      :
>+      : "memory", "cc", "xmm0", "xmm1");
> }
> #endif  // HAS_SOBELTOPLANEROW_SSE2
> 
>@@ -4858,1179 +5406,1123 @@ void SobelToPlaneRow_SSE2(const uint8* src_sobelx,
> // R = Sobel X
> // G = Sobel
> // B = Sobel Y
>-void SobelXYRow_SSE2(const uint8* src_sobelx,
>-                     const uint8* src_sobely,
>-                     uint8* dst_argb,
>+void SobelXYRow_SSE2(const uint8_t* src_sobelx,
>+                     const uint8_t* src_sobely,
>+                     uint8_t* dst_argb,
>                      int width) {
>-  asm volatile (
>-    "sub       %0,%1                           \n"
>-    "pcmpeqb   %%xmm5,%%xmm5                   \n"
>+  asm volatile(
>+      "sub       %0,%1                           \n"
>+      "pcmpeqb   %%xmm5,%%xmm5                   \n"
> 
>-    // 8 pixel loop.
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    MEMOPREG(movdqu,0x00,0,1,1,xmm1)           //  movdqu    (%0,%1,1),%%xmm1
>-    "lea       " MEMLEA(0x10,0) ",%0           \n"
>-    "movdqa    %%xmm0,%%xmm2                   \n"
>-    "paddusb   %%xmm1,%%xmm2                   \n"
>-    "movdqa    %%xmm0,%%xmm3                   \n"
>-    "punpcklbw %%xmm5,%%xmm3                   \n"
>-    "punpckhbw %%xmm5,%%xmm0                   \n"
>-    "movdqa    %%xmm1,%%xmm4                   \n"
>-    "punpcklbw %%xmm2,%%xmm4                   \n"
>-    "punpckhbw %%xmm2,%%xmm1                   \n"
>-    "movdqa    %%xmm4,%%xmm6                   \n"
>-    "punpcklwd %%xmm3,%%xmm6                   \n"
>-    "punpckhwd %%xmm3,%%xmm4                   \n"
>-    "movdqa    %%xmm1,%%xmm7                   \n"
>-    "punpcklwd %%xmm0,%%xmm7                   \n"
>-    "punpckhwd %%xmm0,%%xmm1                   \n"
>-    "movdqu    %%xmm6," MEMACCESS(2) "         \n"
>-    "movdqu    %%xmm4," MEMACCESS2(0x10,2) "   \n"
>-    "movdqu    %%xmm7," MEMACCESS2(0x20,2) "   \n"
>-    "movdqu    %%xmm1," MEMACCESS2(0x30,2) "   \n"
>-    "lea       " MEMLEA(0x40,2) ",%2           \n"
>-    "sub       $0x10,%3                        \n"
>-    "jg        1b                              \n"
>-  : "+r"(src_sobelx),  // %0
>-    "+r"(src_sobely),  // %1
>-    "+r"(dst_argb),    // %2
>-    "+r"(width)        // %3
>-  :
>-  : "memory", "cc", NACL_R14
>-    "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6", "xmm7"
>-  );
>+      // 8 pixel loop.
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "movdqu    0x00(%0,%1,1),%%xmm1            \n"
>+      "lea       0x10(%0),%0                     \n"
>+      "movdqa    %%xmm0,%%xmm2                   \n"
>+      "paddusb   %%xmm1,%%xmm2                   \n"
>+      "movdqa    %%xmm0,%%xmm3                   \n"
>+      "punpcklbw %%xmm5,%%xmm3                   \n"
>+      "punpckhbw %%xmm5,%%xmm0                   \n"
>+      "movdqa    %%xmm1,%%xmm4                   \n"
>+      "punpcklbw %%xmm2,%%xmm4                   \n"
>+      "punpckhbw %%xmm2,%%xmm1                   \n"
>+      "movdqa    %%xmm4,%%xmm6                   \n"
>+      "punpcklwd %%xmm3,%%xmm6                   \n"
>+      "punpckhwd %%xmm3,%%xmm4                   \n"
>+      "movdqa    %%xmm1,%%xmm7                   \n"
>+      "punpcklwd %%xmm0,%%xmm7                   \n"
>+      "punpckhwd %%xmm0,%%xmm1                   \n"
>+      "movdqu    %%xmm6,(%2)                     \n"
>+      "movdqu    %%xmm4,0x10(%2)                 \n"
>+      "movdqu    %%xmm7,0x20(%2)                 \n"
>+      "movdqu    %%xmm1,0x30(%2)                 \n"
>+      "lea       0x40(%2),%2                     \n"
>+      "sub       $0x10,%3                        \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_sobelx),  // %0
>+        "+r"(src_sobely),  // %1
>+        "+r"(dst_argb),    // %2
>+        "+r"(width)        // %3
>+      :
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6",
>+        "xmm7");
> }
> #endif  // HAS_SOBELXYROW_SSE2
> 
> #ifdef HAS_COMPUTECUMULATIVESUMROW_SSE2
> // Creates a table of cumulative sums where each value is a sum of all values
> // above and to the left of the value, inclusive of the value.
>-void ComputeCumulativeSumRow_SSE2(const uint8* row,
>-                                  int32* cumsum,
>-                                  const int32* previous_cumsum,
>+void ComputeCumulativeSumRow_SSE2(const uint8_t* row,
>+                                  int32_t* cumsum,
>+                                  const int32_t* previous_cumsum,
>                                   int width) {
>-  asm volatile (
>-    "pxor      %%xmm0,%%xmm0                   \n"
>-    "pxor      %%xmm1,%%xmm1                   \n"
>-    "sub       $0x4,%3                         \n"
>-    "jl        49f                             \n"
>-    "test      $0xf,%1                         \n"
>-    "jne       49f                             \n"
>-
>-    // 4 pixel loop.
>-    LABELALIGN
>-    "40:                                       \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm2         \n"
>-    "lea       " MEMLEA(0x10,0) ",%0           \n"
>-    "movdqa    %%xmm2,%%xmm4                   \n"
>-    "punpcklbw %%xmm1,%%xmm2                   \n"
>-    "movdqa    %%xmm2,%%xmm3                   \n"
>-    "punpcklwd %%xmm1,%%xmm2                   \n"
>-    "punpckhwd %%xmm1,%%xmm3                   \n"
>-    "punpckhbw %%xmm1,%%xmm4                   \n"
>-    "movdqa    %%xmm4,%%xmm5                   \n"
>-    "punpcklwd %%xmm1,%%xmm4                   \n"
>-    "punpckhwd %%xmm1,%%xmm5                   \n"
>-    "paddd     %%xmm2,%%xmm0                   \n"
>-    "movdqu    " MEMACCESS(2) ",%%xmm2         \n"
>-    "paddd     %%xmm0,%%xmm2                   \n"
>-    "paddd     %%xmm3,%%xmm0                   \n"
>-    "movdqu    " MEMACCESS2(0x10,2) ",%%xmm3   \n"
>-    "paddd     %%xmm0,%%xmm3                   \n"
>-    "paddd     %%xmm4,%%xmm0                   \n"
>-    "movdqu    " MEMACCESS2(0x20,2) ",%%xmm4   \n"
>-    "paddd     %%xmm0,%%xmm4                   \n"
>-    "paddd     %%xmm5,%%xmm0                   \n"
>-    "movdqu    " MEMACCESS2(0x30,2) ",%%xmm5   \n"
>-    "lea       " MEMLEA(0x40,2) ",%2           \n"
>-    "paddd     %%xmm0,%%xmm5                   \n"
>-    "movdqu    %%xmm2," MEMACCESS(1) "         \n"
>-    "movdqu    %%xmm3," MEMACCESS2(0x10,1) "   \n"
>-    "movdqu    %%xmm4," MEMACCESS2(0x20,1) "   \n"
>-    "movdqu    %%xmm5," MEMACCESS2(0x30,1) "   \n"
>-    "lea       " MEMLEA(0x40,1) ",%1           \n"
>-    "sub       $0x4,%3                         \n"
>-    "jge       40b                             \n"
>-
>-    "49:                                       \n"
>-    "add       $0x3,%3                         \n"
>-    "jl        19f                             \n"
>-
>-    // 1 pixel loop.
>-    LABELALIGN
>-    "10:                                       \n"
>-    "movd      " MEMACCESS(0) ",%%xmm2         \n"
>-    "lea       " MEMLEA(0x4,0) ",%0            \n"
>-    "punpcklbw %%xmm1,%%xmm2                   \n"
>-    "punpcklwd %%xmm1,%%xmm2                   \n"
>-    "paddd     %%xmm2,%%xmm0                   \n"
>-    "movdqu    " MEMACCESS(2) ",%%xmm2         \n"
>-    "lea       " MEMLEA(0x10,2) ",%2           \n"
>-    "paddd     %%xmm0,%%xmm2                   \n"
>-    "movdqu    %%xmm2," MEMACCESS(1) "         \n"
>-    "lea       " MEMLEA(0x10,1) ",%1           \n"
>-    "sub       $0x1,%3                         \n"
>-    "jge       10b                             \n"
>-
>-    "19:                                       \n"
>-  : "+r"(row),  // %0
>-    "+r"(cumsum),  // %1
>-    "+r"(previous_cumsum),  // %2
>-    "+r"(width)  // %3
>-  :
>-  : "memory", "cc"
>-    , "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5"
>-  );
>+  asm volatile(
>+      "pxor      %%xmm0,%%xmm0                   \n"
>+      "pxor      %%xmm1,%%xmm1                   \n"
>+      "sub       $0x4,%3                         \n"
>+      "jl        49f                             \n"
>+      "test      $0xf,%1                         \n"
>+      "jne       49f                             \n"
>+
>+      // 4 pixel loop.
>+      LABELALIGN
>+      "40:                                       \n"
>+      "movdqu    (%0),%%xmm2                     \n"
>+      "lea       0x10(%0),%0                     \n"
>+      "movdqa    %%xmm2,%%xmm4                   \n"
>+      "punpcklbw %%xmm1,%%xmm2                   \n"
>+      "movdqa    %%xmm2,%%xmm3                   \n"
>+      "punpcklwd %%xmm1,%%xmm2                   \n"
>+      "punpckhwd %%xmm1,%%xmm3                   \n"
>+      "punpckhbw %%xmm1,%%xmm4                   \n"
>+      "movdqa    %%xmm4,%%xmm5                   \n"
>+      "punpcklwd %%xmm1,%%xmm4                   \n"
>+      "punpckhwd %%xmm1,%%xmm5                   \n"
>+      "paddd     %%xmm2,%%xmm0                   \n"
>+      "movdqu    (%2),%%xmm2                     \n"
>+      "paddd     %%xmm0,%%xmm2                   \n"
>+      "paddd     %%xmm3,%%xmm0                   \n"
>+      "movdqu    0x10(%2),%%xmm3                 \n"
>+      "paddd     %%xmm0,%%xmm3                   \n"
>+      "paddd     %%xmm4,%%xmm0                   \n"
>+      "movdqu    0x20(%2),%%xmm4                 \n"
>+      "paddd     %%xmm0,%%xmm4                   \n"
>+      "paddd     %%xmm5,%%xmm0                   \n"
>+      "movdqu    0x30(%2),%%xmm5                 \n"
>+      "lea       0x40(%2),%2                     \n"
>+      "paddd     %%xmm0,%%xmm5                   \n"
>+      "movdqu    %%xmm2,(%1)                     \n"
>+      "movdqu    %%xmm3,0x10(%1)                 \n"
>+      "movdqu    %%xmm4,0x20(%1)                 \n"
>+      "movdqu    %%xmm5,0x30(%1)                 \n"
>+      "lea       0x40(%1),%1                     \n"
>+      "sub       $0x4,%3                         \n"
>+      "jge       40b                             \n"
>+
>+      "49:                                       \n"
>+      "add       $0x3,%3                         \n"
>+      "jl        19f                             \n"
>+
>+      // 1 pixel loop.
>+      LABELALIGN
>+      "10:                                       \n"
>+      "movd      (%0),%%xmm2                     \n"
>+      "lea       0x4(%0),%0                      \n"
>+      "punpcklbw %%xmm1,%%xmm2                   \n"
>+      "punpcklwd %%xmm1,%%xmm2                   \n"
>+      "paddd     %%xmm2,%%xmm0                   \n"
>+      "movdqu    (%2),%%xmm2                     \n"
>+      "lea       0x10(%2),%2                     \n"
>+      "paddd     %%xmm0,%%xmm2                   \n"
>+      "movdqu    %%xmm2,(%1)                     \n"
>+      "lea       0x10(%1),%1                     \n"
>+      "sub       $0x1,%3                         \n"
>+      "jge       10b                             \n"
>+
>+      "19:                                       \n"
>+      : "+r"(row),              // %0
>+        "+r"(cumsum),           // %1
>+        "+r"(previous_cumsum),  // %2
>+        "+r"(width)             // %3
>+      :
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5");
> }
> #endif  // HAS_COMPUTECUMULATIVESUMROW_SSE2
> 
> #ifdef HAS_CUMULATIVESUMTOAVERAGEROW_SSE2
>-void CumulativeSumToAverageRow_SSE2(const int32* topleft,
>-                                    const int32* botleft,
>+void CumulativeSumToAverageRow_SSE2(const int32_t* topleft,
>+                                    const int32_t* botleft,
>                                     int width,
>                                     int area,
>-                                    uint8* dst,
>+                                    uint8_t* dst,
>                                     int count) {
>-  asm volatile (
>-    "movd      %5,%%xmm5                       \n"
>-    "cvtdq2ps  %%xmm5,%%xmm5                   \n"
>-    "rcpss     %%xmm5,%%xmm4                   \n"
>-    "pshufd    $0x0,%%xmm4,%%xmm4              \n"
>-    "sub       $0x4,%3                         \n"
>-    "jl        49f                             \n"
>-    "cmpl      $0x80,%5                        \n"
>-    "ja        40f                             \n"
>-
>-    "pshufd    $0x0,%%xmm5,%%xmm5              \n"
>-    "pcmpeqb   %%xmm6,%%xmm6                   \n"
>-    "psrld     $0x10,%%xmm6                    \n"
>-    "cvtdq2ps  %%xmm6,%%xmm6                   \n"
>-    "addps     %%xmm6,%%xmm5                   \n"
>-    "mulps     %%xmm4,%%xmm5                   \n"
>-    "cvtps2dq  %%xmm5,%%xmm5                   \n"
>-    "packssdw  %%xmm5,%%xmm5                   \n"
>-
>-    // 4 pixel small loop.
>-    LABELALIGN
>-  "4:                                         \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    "movdqu    " MEMACCESS2(0x10,0) ",%%xmm1   \n"
>-    "movdqu    " MEMACCESS2(0x20,0) ",%%xmm2   \n"
>-    "movdqu    " MEMACCESS2(0x30,0) ",%%xmm3   \n"
>-    MEMOPREG(psubd,0x00,0,4,4,xmm0)            // psubd    0x00(%0,%4,4),%%xmm0
>-    MEMOPREG(psubd,0x10,0,4,4,xmm1)            // psubd    0x10(%0,%4,4),%%xmm1
>-    MEMOPREG(psubd,0x20,0,4,4,xmm2)            // psubd    0x20(%0,%4,4),%%xmm2
>-    MEMOPREG(psubd,0x30,0,4,4,xmm3)            // psubd    0x30(%0,%4,4),%%xmm3
>-    "lea       " MEMLEA(0x40,0) ",%0           \n"
>-    "psubd     " MEMACCESS(1) ",%%xmm0         \n"
>-    "psubd     " MEMACCESS2(0x10,1) ",%%xmm1   \n"
>-    "psubd     " MEMACCESS2(0x20,1) ",%%xmm2   \n"
>-    "psubd     " MEMACCESS2(0x30,1) ",%%xmm3   \n"
>-    MEMOPREG(paddd,0x00,1,4,4,xmm0)            // paddd    0x00(%1,%4,4),%%xmm0
>-    MEMOPREG(paddd,0x10,1,4,4,xmm1)            // paddd    0x10(%1,%4,4),%%xmm1
>-    MEMOPREG(paddd,0x20,1,4,4,xmm2)            // paddd    0x20(%1,%4,4),%%xmm2
>-    MEMOPREG(paddd,0x30,1,4,4,xmm3)            // paddd    0x30(%1,%4,4),%%xmm3
>-    "lea       " MEMLEA(0x40,1) ",%1           \n"
>-    "packssdw  %%xmm1,%%xmm0                   \n"
>-    "packssdw  %%xmm3,%%xmm2                   \n"
>-    "pmulhuw   %%xmm5,%%xmm0                   \n"
>-    "pmulhuw   %%xmm5,%%xmm2                   \n"
>-    "packuswb  %%xmm2,%%xmm0                   \n"
>-    "movdqu    %%xmm0," MEMACCESS(2) "         \n"
>-    "lea       " MEMLEA(0x10,2) ",%2           \n"
>-    "sub       $0x4,%3                         \n"
>-    "jge       4b                              \n"
>-    "jmp       49f                             \n"
>-
>-  // 4 pixel loop                              \n"
>-    LABELALIGN
>-  "40:                                         \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    "movdqu    " MEMACCESS2(0x10,0) ",%%xmm1   \n"
>-    "movdqu    " MEMACCESS2(0x20,0) ",%%xmm2   \n"
>-    "movdqu    " MEMACCESS2(0x30,0) ",%%xmm3   \n"
>-    MEMOPREG(psubd,0x00,0,4,4,xmm0)            // psubd    0x00(%0,%4,4),%%xmm0
>-    MEMOPREG(psubd,0x10,0,4,4,xmm1)            // psubd    0x10(%0,%4,4),%%xmm1
>-    MEMOPREG(psubd,0x20,0,4,4,xmm2)            // psubd    0x20(%0,%4,4),%%xmm2
>-    MEMOPREG(psubd,0x30,0,4,4,xmm3)            // psubd    0x30(%0,%4,4),%%xmm3
>-    "lea       " MEMLEA(0x40,0) ",%0           \n"
>-    "psubd     " MEMACCESS(1) ",%%xmm0         \n"
>-    "psubd     " MEMACCESS2(0x10,1) ",%%xmm1   \n"
>-    "psubd     " MEMACCESS2(0x20,1) ",%%xmm2   \n"
>-    "psubd     " MEMACCESS2(0x30,1) ",%%xmm3   \n"
>-    MEMOPREG(paddd,0x00,1,4,4,xmm0)            // paddd    0x00(%1,%4,4),%%xmm0
>-    MEMOPREG(paddd,0x10,1,4,4,xmm1)            // paddd    0x10(%1,%4,4),%%xmm1
>-    MEMOPREG(paddd,0x20,1,4,4,xmm2)            // paddd    0x20(%1,%4,4),%%xmm2
>-    MEMOPREG(paddd,0x30,1,4,4,xmm3)            // paddd    0x30(%1,%4,4),%%xmm3
>-    "lea       " MEMLEA(0x40,1) ",%1           \n"
>-    "cvtdq2ps  %%xmm0,%%xmm0                   \n"
>-    "cvtdq2ps  %%xmm1,%%xmm1                   \n"
>-    "mulps     %%xmm4,%%xmm0                   \n"
>-    "mulps     %%xmm4,%%xmm1                   \n"
>-    "cvtdq2ps  %%xmm2,%%xmm2                   \n"
>-    "cvtdq2ps  %%xmm3,%%xmm3                   \n"
>-    "mulps     %%xmm4,%%xmm2                   \n"
>-    "mulps     %%xmm4,%%xmm3                   \n"
>-    "cvtps2dq  %%xmm0,%%xmm0                   \n"
>-    "cvtps2dq  %%xmm1,%%xmm1                   \n"
>-    "cvtps2dq  %%xmm2,%%xmm2                   \n"
>-    "cvtps2dq  %%xmm3,%%xmm3                   \n"
>-    "packssdw  %%xmm1,%%xmm0                   \n"
>-    "packssdw  %%xmm3,%%xmm2                   \n"
>-    "packuswb  %%xmm2,%%xmm0                   \n"
>-    "movdqu    %%xmm0," MEMACCESS(2) "         \n"
>-    "lea       " MEMLEA(0x10,2) ",%2           \n"
>-    "sub       $0x4,%3                         \n"
>-    "jge       40b                             \n"
>-
>-  "49:                                         \n"
>-    "add       $0x3,%3                         \n"
>-    "jl        19f                             \n"
>-
>-  // 1 pixel loop                              \n"
>-    LABELALIGN
>-  "10:                                         \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    MEMOPREG(psubd,0x00,0,4,4,xmm0)            // psubd    0x00(%0,%4,4),%%xmm0
>-    "lea       " MEMLEA(0x10,0) ",%0           \n"
>-    "psubd     " MEMACCESS(1) ",%%xmm0         \n"
>-    MEMOPREG(paddd,0x00,1,4,4,xmm0)            // paddd    0x00(%1,%4,4),%%xmm0
>-    "lea       " MEMLEA(0x10,1) ",%1           \n"
>-    "cvtdq2ps  %%xmm0,%%xmm0                   \n"
>-    "mulps     %%xmm4,%%xmm0                   \n"
>-    "cvtps2dq  %%xmm0,%%xmm0                   \n"
>-    "packssdw  %%xmm0,%%xmm0                   \n"
>-    "packuswb  %%xmm0,%%xmm0                   \n"
>-    "movd      %%xmm0," MEMACCESS(2) "         \n"
>-    "lea       " MEMLEA(0x4,2) ",%2            \n"
>-    "sub       $0x1,%3                         \n"
>-    "jge       10b                             \n"
>-  "19:                                         \n"
>-  : "+r"(topleft),  // %0
>-    "+r"(botleft),  // %1
>-    "+r"(dst),      // %2
>-    "+rm"(count)    // %3
>-  : "r"((intptr_t)(width)),  // %4
>-    "rm"(area)     // %5
>-  : "memory", "cc", NACL_R14
>-    "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6"
>-  );
>+  asm volatile(
>+      "movd      %5,%%xmm5                       \n"
>+      "cvtdq2ps  %%xmm5,%%xmm5                   \n"
>+      "rcpss     %%xmm5,%%xmm4                   \n"
>+      "pshufd    $0x0,%%xmm4,%%xmm4              \n"
>+      "sub       $0x4,%3                         \n"
>+      "jl        49f                             \n"
>+      "cmpl      $0x80,%5                        \n"
>+      "ja        40f                             \n"
>+
>+      "pshufd    $0x0,%%xmm5,%%xmm5              \n"
>+      "pcmpeqb   %%xmm6,%%xmm6                   \n"
>+      "psrld     $0x10,%%xmm6                    \n"
>+      "cvtdq2ps  %%xmm6,%%xmm6                   \n"
>+      "addps     %%xmm6,%%xmm5                   \n"
>+      "mulps     %%xmm4,%%xmm5                   \n"
>+      "cvtps2dq  %%xmm5,%%xmm5                   \n"
>+      "packssdw  %%xmm5,%%xmm5                   \n"
>+
>+      // 4 pixel small loop.
>+      LABELALIGN
>+      "4:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "movdqu    0x10(%0),%%xmm1                 \n"
>+      "movdqu    0x20(%0),%%xmm2                 \n"
>+      "movdqu    0x30(%0),%%xmm3                 \n"
>+      "psubd     0x00(%0,%4,4),%%xmm0            \n"
>+      "psubd     0x10(%0,%4,4),%%xmm1            \n"
>+      "psubd     0x20(%0,%4,4),%%xmm2            \n"
>+      "psubd     0x30(%0,%4,4),%%xmm3            \n"
>+      "lea       0x40(%0),%0                     \n"
>+      "psubd     (%1),%%xmm0                     \n"
>+      "psubd     0x10(%1),%%xmm1                 \n"
>+      "psubd     0x20(%1),%%xmm2                 \n"
>+      "psubd     0x30(%1),%%xmm3                 \n"
>+      "paddd     0x00(%1,%4,4),%%xmm0            \n"
>+      "paddd     0x10(%1,%4,4),%%xmm1            \n"
>+      "paddd     0x20(%1,%4,4),%%xmm2            \n"
>+      "paddd     0x30(%1,%4,4),%%xmm3            \n"
>+      "lea       0x40(%1),%1                     \n"
>+      "packssdw  %%xmm1,%%xmm0                   \n"
>+      "packssdw  %%xmm3,%%xmm2                   \n"
>+      "pmulhuw   %%xmm5,%%xmm0                   \n"
>+      "pmulhuw   %%xmm5,%%xmm2                   \n"
>+      "packuswb  %%xmm2,%%xmm0                   \n"
>+      "movdqu    %%xmm0,(%2)                     \n"
>+      "lea       0x10(%2),%2                     \n"
>+      "sub       $0x4,%3                         \n"
>+      "jge       4b                              \n"
>+      "jmp       49f                             \n"
>+
>+      // 4 pixel loop
>+      LABELALIGN
>+      "40:                                       \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "movdqu    0x10(%0),%%xmm1                 \n"
>+      "movdqu    0x20(%0),%%xmm2                 \n"
>+      "movdqu    0x30(%0),%%xmm3                 \n"
>+      "psubd     0x00(%0,%4,4),%%xmm0            \n"
>+      "psubd     0x10(%0,%4,4),%%xmm1            \n"
>+      "psubd     0x20(%0,%4,4),%%xmm2            \n"
>+      "psubd     0x30(%0,%4,4),%%xmm3            \n"
>+      "lea       0x40(%0),%0                     \n"
>+      "psubd     (%1),%%xmm0                     \n"
>+      "psubd     0x10(%1),%%xmm1                 \n"
>+      "psubd     0x20(%1),%%xmm2                 \n"
>+      "psubd     0x30(%1),%%xmm3                 \n"
>+      "paddd     0x00(%1,%4,4),%%xmm0            \n"
>+      "paddd     0x10(%1,%4,4),%%xmm1            \n"
>+      "paddd     0x20(%1,%4,4),%%xmm2            \n"
>+      "paddd     0x30(%1,%4,4),%%xmm3            \n"
>+      "lea       0x40(%1),%1                     \n"
>+      "cvtdq2ps  %%xmm0,%%xmm0                   \n"
>+      "cvtdq2ps  %%xmm1,%%xmm1                   \n"
>+      "mulps     %%xmm4,%%xmm0                   \n"
>+      "mulps     %%xmm4,%%xmm1                   \n"
>+      "cvtdq2ps  %%xmm2,%%xmm2                   \n"
>+      "cvtdq2ps  %%xmm3,%%xmm3                   \n"
>+      "mulps     %%xmm4,%%xmm2                   \n"
>+      "mulps     %%xmm4,%%xmm3                   \n"
>+      "cvtps2dq  %%xmm0,%%xmm0                   \n"
>+      "cvtps2dq  %%xmm1,%%xmm1                   \n"
>+      "cvtps2dq  %%xmm2,%%xmm2                   \n"
>+      "cvtps2dq  %%xmm3,%%xmm3                   \n"
>+      "packssdw  %%xmm1,%%xmm0                   \n"
>+      "packssdw  %%xmm3,%%xmm2                   \n"
>+      "packuswb  %%xmm2,%%xmm0                   \n"
>+      "movdqu    %%xmm0,(%2)                     \n"
>+      "lea       0x10(%2),%2                     \n"
>+      "sub       $0x4,%3                         \n"
>+      "jge       40b                             \n"
>+
>+      "49:                                       \n"
>+      "add       $0x3,%3                         \n"
>+      "jl        19f                             \n"
>+
>+      // 1 pixel loop
>+      LABELALIGN
>+      "10:                                       \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "psubd     0x00(%0,%4,4),%%xmm0            \n"
>+      "lea       0x10(%0),%0                     \n"
>+      "psubd     (%1),%%xmm0                     \n"
>+      "paddd     0x00(%1,%4,4),%%xmm0            \n"
>+      "lea       0x10(%1),%1                     \n"
>+      "cvtdq2ps  %%xmm0,%%xmm0                   \n"
>+      "mulps     %%xmm4,%%xmm0                   \n"
>+      "cvtps2dq  %%xmm0,%%xmm0                   \n"
>+      "packssdw  %%xmm0,%%xmm0                   \n"
>+      "packuswb  %%xmm0,%%xmm0                   \n"
>+      "movd      %%xmm0,(%2)                     \n"
>+      "lea       0x4(%2),%2                      \n"
>+      "sub       $0x1,%3                         \n"
>+      "jge       10b                             \n"
>+      "19:                                       \n"
>+      : "+r"(topleft),           // %0
>+        "+r"(botleft),           // %1
>+        "+r"(dst),               // %2
>+        "+rm"(count)             // %3
>+      : "r"((intptr_t)(width)),  // %4
>+        "rm"(area)               // %5
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6");
> }
> #endif  // HAS_CUMULATIVESUMTOAVERAGEROW_SSE2
> 
> #ifdef HAS_ARGBAFFINEROW_SSE2
> // Copy ARGB pixels from source image with slope to a row of destination.
> LIBYUV_API
>-void ARGBAffineRow_SSE2(const uint8* src_argb,
>+void ARGBAffineRow_SSE2(const uint8_t* src_argb,
>                         int src_argb_stride,
>-                        uint8* dst_argb,
>+                        uint8_t* dst_argb,
>                         const float* src_dudv,
>                         int width) {
>   intptr_t src_argb_stride_temp = src_argb_stride;
>   intptr_t temp;
>-  asm volatile (
>-    "movq      " MEMACCESS(3) ",%%xmm2         \n"
>-    "movq      " MEMACCESS2(0x08,3) ",%%xmm7   \n"
>-    "shl       $0x10,%1                        \n"
>-    "add       $0x4,%1                         \n"
>-    "movd      %1,%%xmm5                       \n"
>-    "sub       $0x4,%4                         \n"
>-    "jl        49f                             \n"
>-
>-    "pshufd    $0x44,%%xmm7,%%xmm7             \n"
>-    "pshufd    $0x0,%%xmm5,%%xmm5              \n"
>-    "movdqa    %%xmm2,%%xmm0                   \n"
>-    "addps     %%xmm7,%%xmm0                   \n"
>-    "movlhps   %%xmm0,%%xmm2                   \n"
>-    "movdqa    %%xmm7,%%xmm4                   \n"
>-    "addps     %%xmm4,%%xmm4                   \n"
>-    "movdqa    %%xmm2,%%xmm3                   \n"
>-    "addps     %%xmm4,%%xmm3                   \n"
>-    "addps     %%xmm4,%%xmm4                   \n"
>-
>-  // 4 pixel loop                              \n"
>-    LABELALIGN
>-  "40:                                         \n"
>-    "cvttps2dq %%xmm2,%%xmm0                   \n"  // x, y float to int first 2
>-    "cvttps2dq %%xmm3,%%xmm1                   \n"  // x, y float to int next 2
>-    "packssdw  %%xmm1,%%xmm0                   \n"  // x, y as 8 shorts
>-    "pmaddwd   %%xmm5,%%xmm0                   \n"  // off = x * 4 + y * stride
>-    "movd      %%xmm0,%k1                      \n"
>-    "pshufd    $0x39,%%xmm0,%%xmm0             \n"
>-    "movd      %%xmm0,%k5                      \n"
>-    "pshufd    $0x39,%%xmm0,%%xmm0             \n"
>-    MEMOPREG(movd,0x00,0,1,1,xmm1)             //  movd      (%0,%1,1),%%xmm1
>-    MEMOPREG(movd,0x00,0,5,1,xmm6)             //  movd      (%0,%5,1),%%xmm6
>-    "punpckldq %%xmm6,%%xmm1                   \n"
>-    "addps     %%xmm4,%%xmm2                   \n"
>-    "movq      %%xmm1," MEMACCESS(2) "         \n"
>-    "movd      %%xmm0,%k1                      \n"
>-    "pshufd    $0x39,%%xmm0,%%xmm0             \n"
>-    "movd      %%xmm0,%k5                      \n"
>-    MEMOPREG(movd,0x00,0,1,1,xmm0)             //  movd      (%0,%1,1),%%xmm0
>-    MEMOPREG(movd,0x00,0,5,1,xmm6)             //  movd      (%0,%5,1),%%xmm6
>-    "punpckldq %%xmm6,%%xmm0                   \n"
>-    "addps     %%xmm4,%%xmm3                   \n"
>-    "movq      %%xmm0," MEMACCESS2(0x08,2) "   \n"
>-    "lea       " MEMLEA(0x10,2) ",%2           \n"
>-    "sub       $0x4,%4                         \n"
>-    "jge       40b                             \n"
>-
>-  "49:                                         \n"
>-    "add       $0x3,%4                         \n"
>-    "jl        19f                             \n"
>-
>-  // 1 pixel loop                              \n"
>-    LABELALIGN
>-  "10:                                         \n"
>-    "cvttps2dq %%xmm2,%%xmm0                   \n"
>-    "packssdw  %%xmm0,%%xmm0                   \n"
>-    "pmaddwd   %%xmm5,%%xmm0                   \n"
>-    "addps     %%xmm7,%%xmm2                   \n"
>-    "movd      %%xmm0,%k1                      \n"
>-    MEMOPREG(movd,0x00,0,1,1,xmm0)             //  movd      (%0,%1,1),%%xmm0
>-    "movd      %%xmm0," MEMACCESS(2) "         \n"
>-    "lea       " MEMLEA(0x04,2) ",%2           \n"
>-    "sub       $0x1,%4                         \n"
>-    "jge       10b                             \n"
>-  "19:                                         \n"
>-  : "+r"(src_argb),  // %0
>-    "+r"(src_argb_stride_temp),  // %1
>-    "+r"(dst_argb),  // %2
>-    "+r"(src_dudv),  // %3
>-    "+rm"(width),    // %4
>-    "=&r"(temp)      // %5
>-  :
>-  : "memory", "cc", NACL_R14
>-    "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6", "xmm7"
>-  );
>+  asm volatile(
>+      "movq      (%3),%%xmm2                     \n"
>+      "movq      0x08(%3),%%xmm7                 \n"
>+      "shl       $0x10,%1                        \n"
>+      "add       $0x4,%1                         \n"
>+      "movd      %1,%%xmm5                       \n"
>+      "sub       $0x4,%4                         \n"
>+      "jl        49f                             \n"
>+
>+      "pshufd    $0x44,%%xmm7,%%xmm7             \n"
>+      "pshufd    $0x0,%%xmm5,%%xmm5              \n"
>+      "movdqa    %%xmm2,%%xmm0                   \n"
>+      "addps     %%xmm7,%%xmm0                   \n"
>+      "movlhps   %%xmm0,%%xmm2                   \n"
>+      "movdqa    %%xmm7,%%xmm4                   \n"
>+      "addps     %%xmm4,%%xmm4                   \n"
>+      "movdqa    %%xmm2,%%xmm3                   \n"
>+      "addps     %%xmm4,%%xmm3                   \n"
>+      "addps     %%xmm4,%%xmm4                   \n"
>+
>+      // 4 pixel loop
>+      LABELALIGN
>+      "40:                                       \n"
>+      "cvttps2dq %%xmm2,%%xmm0                   \n"  // x,y float->int first 2
>+      "cvttps2dq %%xmm3,%%xmm1                   \n"  // x,y float->int next 2
>+      "packssdw  %%xmm1,%%xmm0                   \n"  // x, y as 8 shorts
>+      "pmaddwd   %%xmm5,%%xmm0                   \n"  // off = x*4 + y*stride
>+      "movd      %%xmm0,%k1                      \n"
>+      "pshufd    $0x39,%%xmm0,%%xmm0             \n"
>+      "movd      %%xmm0,%k5                      \n"
>+      "pshufd    $0x39,%%xmm0,%%xmm0             \n"
>+      "movd      0x00(%0,%1,1),%%xmm1            \n"
>+      "movd      0x00(%0,%5,1),%%xmm6            \n"
>+      "punpckldq %%xmm6,%%xmm1                   \n"
>+      "addps     %%xmm4,%%xmm2                   \n"
>+      "movq      %%xmm1,(%2)                     \n"
>+      "movd      %%xmm0,%k1                      \n"
>+      "pshufd    $0x39,%%xmm0,%%xmm0             \n"
>+      "movd      %%xmm0,%k5                      \n"
>+      "movd      0x00(%0,%1,1),%%xmm0            \n"
>+      "movd      0x00(%0,%5,1),%%xmm6            \n"
>+      "punpckldq %%xmm6,%%xmm0                   \n"
>+      "addps     %%xmm4,%%xmm3                   \n"
>+      "movq      %%xmm0,0x08(%2)                 \n"
>+      "lea       0x10(%2),%2                     \n"
>+      "sub       $0x4,%4                         \n"
>+      "jge       40b                             \n"
>+
>+      "49:                                       \n"
>+      "add       $0x3,%4                         \n"
>+      "jl        19f                             \n"
>+
>+      // 1 pixel loop
>+      LABELALIGN
>+      "10:                                       \n"
>+      "cvttps2dq %%xmm2,%%xmm0                   \n"
>+      "packssdw  %%xmm0,%%xmm0                   \n"
>+      "pmaddwd   %%xmm5,%%xmm0                   \n"
>+      "addps     %%xmm7,%%xmm2                   \n"
>+      "movd      %%xmm0,%k1                      \n"
>+      "movd      0x00(%0,%1,1),%%xmm0            \n"
>+      "movd      %%xmm0,(%2)                     \n"
>+      "lea       0x04(%2),%2                     \n"
>+      "sub       $0x1,%4                         \n"
>+      "jge       10b                             \n"
>+      "19:                                       \n"
>+      : "+r"(src_argb),              // %0
>+        "+r"(src_argb_stride_temp),  // %1
>+        "+r"(dst_argb),              // %2
>+        "+r"(src_dudv),              // %3
>+        "+rm"(width),                // %4
>+        "=&r"(temp)                  // %5
>+      :
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6",
>+        "xmm7");
> }
> #endif  // HAS_ARGBAFFINEROW_SSE2
> 
> #ifdef HAS_INTERPOLATEROW_SSSE3
> // Bilinear filter 16x2 -> 16x1
>-void InterpolateRow_SSSE3(uint8* dst_ptr,
>-                          const uint8* src_ptr,
>+void InterpolateRow_SSSE3(uint8_t* dst_ptr,
>+                          const uint8_t* src_ptr,
>                           ptrdiff_t src_stride,
>                           int dst_width,
>                           int source_y_fraction) {
>-  asm volatile (
>-    "sub       %1,%0                           \n"
>-    "cmp       $0x0,%3                         \n"
>-    "je        100f                            \n"
>-    "cmp       $0x80,%3                        \n"
>-    "je        50f                             \n"
>-
>-    "movd      %3,%%xmm0                       \n"
>-    "neg       %3                              \n"
>-    "add       $0x100,%3                       \n"
>-    "movd      %3,%%xmm5                       \n"
>-    "punpcklbw %%xmm0,%%xmm5                   \n"
>-    "punpcklwd %%xmm5,%%xmm5                   \n"
>-    "pshufd    $0x0,%%xmm5,%%xmm5              \n"
>-    "mov       $0x80808080,%%eax               \n"
>-    "movd      %%eax,%%xmm4                    \n"
>-    "pshufd    $0x0,%%xmm4,%%xmm4              \n"
>-
>-    // General purpose row blend.
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movdqu    " MEMACCESS(1) ",%%xmm0         \n"
>-    MEMOPREG(movdqu,0x00,1,4,1,xmm2)
>-    "movdqa     %%xmm0,%%xmm1                  \n"
>-    "punpcklbw  %%xmm2,%%xmm0                  \n"
>-    "punpckhbw  %%xmm2,%%xmm1                  \n"
>-    "psubb      %%xmm4,%%xmm0                  \n"
>-    "psubb      %%xmm4,%%xmm1                  \n"
>-    "movdqa     %%xmm5,%%xmm2                  \n"
>-    "movdqa     %%xmm5,%%xmm3                  \n"
>-    "pmaddubsw  %%xmm0,%%xmm2                  \n"
>-    "pmaddubsw  %%xmm1,%%xmm3                  \n"
>-    "paddw      %%xmm4,%%xmm2                  \n"
>-    "paddw      %%xmm4,%%xmm3                  \n"
>-    "psrlw      $0x8,%%xmm2                    \n"
>-    "psrlw      $0x8,%%xmm3                    \n"
>-    "packuswb   %%xmm3,%%xmm2                  \n"
>-    MEMOPMEM(movdqu,xmm2,0x00,1,0,1)
>-    "lea       " MEMLEA(0x10,1) ",%1           \n"
>-    "sub       $0x10,%2                        \n"
>-    "jg        1b                              \n"
>-    "jmp       99f                             \n"
>+  asm volatile(
>+      "sub       %1,%0                           \n"
>+      "cmp       $0x0,%3                         \n"
>+      "je        100f                            \n"
>+      "cmp       $0x80,%3                        \n"
>+      "je        50f                             \n"
>+
>+      "movd      %3,%%xmm0                       \n"
>+      "neg       %3                              \n"
>+      "add       $0x100,%3                       \n"
>+      "movd      %3,%%xmm5                       \n"
>+      "punpcklbw %%xmm0,%%xmm5                   \n"
>+      "punpcklwd %%xmm5,%%xmm5                   \n"
>+      "pshufd    $0x0,%%xmm5,%%xmm5              \n"
>+      "mov       $0x80808080,%%eax               \n"
>+      "movd      %%eax,%%xmm4                    \n"
>+      "pshufd    $0x0,%%xmm4,%%xmm4              \n"
>+
>+      // General purpose row blend.
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%1),%%xmm0                     \n"
>+      "movdqu    0x00(%1,%4,1),%%xmm2            \n"
>+      "movdqa     %%xmm0,%%xmm1                  \n"
>+      "punpcklbw  %%xmm2,%%xmm0                  \n"
>+      "punpckhbw  %%xmm2,%%xmm1                  \n"
>+      "psubb      %%xmm4,%%xmm0                  \n"
>+      "psubb      %%xmm4,%%xmm1                  \n"
>+      "movdqa     %%xmm5,%%xmm2                  \n"
>+      "movdqa     %%xmm5,%%xmm3                  \n"
>+      "pmaddubsw  %%xmm0,%%xmm2                  \n"
>+      "pmaddubsw  %%xmm1,%%xmm3                  \n"
>+      "paddw      %%xmm4,%%xmm2                  \n"
>+      "paddw      %%xmm4,%%xmm3                  \n"
>+      "psrlw      $0x8,%%xmm2                    \n"
>+      "psrlw      $0x8,%%xmm3                    \n"
>+      "packuswb   %%xmm3,%%xmm2                  \n"
>+      "movdqu    %%xmm2,0x00(%1,%0,1)            \n"
>+      "lea       0x10(%1),%1                     \n"
>+      "sub       $0x10,%2                        \n"
>+      "jg        1b                              \n"
>+      "jmp       99f                             \n"
> 
>-    // Blend 50 / 50.
>-    LABELALIGN
>-  "50:                                         \n"
>-    "movdqu    " MEMACCESS(1) ",%%xmm0         \n"
>-    MEMOPREG(movdqu,0x00,1,4,1,xmm1)
>-    "pavgb     %%xmm1,%%xmm0                   \n"
>-    MEMOPMEM(movdqu,xmm0,0x00,1,0,1)
>-    "lea       " MEMLEA(0x10,1) ",%1           \n"
>-    "sub       $0x10,%2                        \n"
>-    "jg        50b                             \n"
>-    "jmp       99f                             \n"
>-
>-    // Blend 100 / 0 - Copy row unchanged.
>-    LABELALIGN
>-  "100:                                        \n"
>-    "movdqu    " MEMACCESS(1) ",%%xmm0         \n"
>-    MEMOPMEM(movdqu,xmm0,0x00,1,0,1)
>-    "lea       " MEMLEA(0x10,1) ",%1           \n"
>-    "sub       $0x10,%2                        \n"
>-    "jg        100b                            \n"
>-
>-  "99:                                         \n"
>-  : "+r"(dst_ptr),     // %0
>-    "+r"(src_ptr),     // %1
>-    "+rm"(dst_width),  // %2
>-    "+r"(source_y_fraction)  // %3
>-  : "r"((intptr_t)(src_stride))  // %4
>-  : "memory", "cc", "eax", NACL_R14
>-    "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5"
>-  );
>+      // Blend 50 / 50.
>+      LABELALIGN
>+      "50:                                       \n"
>+      "movdqu    (%1),%%xmm0                     \n"
>+      "movdqu    0x00(%1,%4,1),%%xmm1            \n"
>+      "pavgb     %%xmm1,%%xmm0                   \n"
>+      "movdqu    %%xmm0,0x00(%1,%0,1)            \n"
>+      "lea       0x10(%1),%1                     \n"
>+      "sub       $0x10,%2                        \n"
>+      "jg        50b                             \n"
>+      "jmp       99f                             \n"
>+
>+      // Blend 100 / 0 - Copy row unchanged.
>+      LABELALIGN
>+      "100:                                      \n"
>+      "movdqu    (%1),%%xmm0                     \n"
>+      "movdqu    %%xmm0,0x00(%1,%0,1)            \n"
>+      "lea       0x10(%1),%1                     \n"
>+      "sub       $0x10,%2                        \n"
>+      "jg        100b                            \n"
>+
>+      "99:                                       \n"
>+      : "+r"(dst_ptr),               // %0
>+        "+r"(src_ptr),               // %1
>+        "+rm"(dst_width),            // %2
>+        "+r"(source_y_fraction)      // %3
>+      : "r"((intptr_t)(src_stride))  // %4
>+      : "memory", "cc", "eax", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5");
> }
> #endif  // HAS_INTERPOLATEROW_SSSE3
> 
> #ifdef HAS_INTERPOLATEROW_AVX2
> // Bilinear filter 32x2 -> 32x1
>-void InterpolateRow_AVX2(uint8* dst_ptr,
>-                         const uint8* src_ptr,
>+void InterpolateRow_AVX2(uint8_t* dst_ptr,
>+                         const uint8_t* src_ptr,
>                          ptrdiff_t src_stride,
>                          int dst_width,
>                          int source_y_fraction) {
>-  asm volatile (
>-    "cmp       $0x0,%3                         \n"
>-    "je        100f                            \n"
>-    "sub       %1,%0                           \n"
>-    "cmp       $0x80,%3                        \n"
>-    "je        50f                             \n"
>-
>-    "vmovd      %3,%%xmm0                      \n"
>-    "neg        %3                             \n"
>-    "add        $0x100,%3                      \n"
>-    "vmovd      %3,%%xmm5                      \n"
>-    "vpunpcklbw %%xmm0,%%xmm5,%%xmm5           \n"
>-    "vpunpcklwd %%xmm5,%%xmm5,%%xmm5           \n"
>-    "vbroadcastss %%xmm5,%%ymm5                \n"
>-    "mov        $0x80808080,%%eax              \n"
>-    "vmovd      %%eax,%%xmm4                   \n"
>-    "vbroadcastss %%xmm4,%%ymm4                \n"
>-
>-    // General purpose row blend.
>-    LABELALIGN
>-    "1:                                        \n"
>-    "vmovdqu    " MEMACCESS(1) ",%%ymm0        \n"
>-    MEMOPREG(vmovdqu,0x00,1,4,1,ymm2)
>-    "vpunpckhbw %%ymm2,%%ymm0,%%ymm1           \n"
>-    "vpunpcklbw %%ymm2,%%ymm0,%%ymm0           \n"
>-    "vpsubb     %%ymm4,%%ymm1,%%ymm1           \n"
>-    "vpsubb     %%ymm4,%%ymm0,%%ymm0           \n"
>-    "vpmaddubsw %%ymm1,%%ymm5,%%ymm1           \n"
>-    "vpmaddubsw %%ymm0,%%ymm5,%%ymm0           \n"
>-    "vpaddw     %%ymm4,%%ymm1,%%ymm1           \n"
>-    "vpaddw     %%ymm4,%%ymm0,%%ymm0           \n"
>-    "vpsrlw     $0x8,%%ymm1,%%ymm1             \n"
>-    "vpsrlw     $0x8,%%ymm0,%%ymm0             \n"
>-    "vpackuswb  %%ymm1,%%ymm0,%%ymm0           \n"
>-    MEMOPMEM(vmovdqu,ymm0,0x00,1,0,1)
>-    "lea       " MEMLEA(0x20,1) ",%1           \n"
>-    "sub       $0x20,%2                        \n"
>-    "jg        1b                              \n"
>-    "jmp       99f                             \n"
>+  asm volatile(
>+      "cmp       $0x0,%3                         \n"
>+      "je        100f                            \n"
>+      "sub       %1,%0                           \n"
>+      "cmp       $0x80,%3                        \n"
>+      "je        50f                             \n"
>+
>+      "vmovd      %3,%%xmm0                      \n"
>+      "neg        %3                             \n"
>+      "add        $0x100,%3                      \n"
>+      "vmovd      %3,%%xmm5                      \n"
>+      "vpunpcklbw %%xmm0,%%xmm5,%%xmm5           \n"
>+      "vpunpcklwd %%xmm5,%%xmm5,%%xmm5           \n"
>+      "vbroadcastss %%xmm5,%%ymm5                \n"
>+      "mov        $0x80808080,%%eax              \n"
>+      "vmovd      %%eax,%%xmm4                   \n"
>+      "vbroadcastss %%xmm4,%%ymm4                \n"
> 
>-    // Blend 50 / 50.
>-    LABELALIGN
>-  "50:                                         \n"
>-    "vmovdqu    " MEMACCESS(1) ",%%ymm0        \n"
>-    VMEMOPREG(vpavgb,0x00,1,4,1,ymm0,ymm0)     // vpavgb (%1,%4,1),%%ymm0,%%ymm0
>-    MEMOPMEM(vmovdqu,ymm0,0x00,1,0,1)
>-    "lea       " MEMLEA(0x20,1) ",%1           \n"
>-    "sub       $0x20,%2                        \n"
>-    "jg        50b                             \n"
>-    "jmp       99f                             \n"
>+      // General purpose row blend.
>+      LABELALIGN
>+      "1:                                        \n"
>+      "vmovdqu    (%1),%%ymm0                    \n"
>+      "vmovdqu    0x00(%1,%4,1),%%ymm2           \n"
>+      "vpunpckhbw %%ymm2,%%ymm0,%%ymm1           \n"
>+      "vpunpcklbw %%ymm2,%%ymm0,%%ymm0           \n"
>+      "vpsubb     %%ymm4,%%ymm1,%%ymm1           \n"
>+      "vpsubb     %%ymm4,%%ymm0,%%ymm0           \n"
>+      "vpmaddubsw %%ymm1,%%ymm5,%%ymm1           \n"
>+      "vpmaddubsw %%ymm0,%%ymm5,%%ymm0           \n"
>+      "vpaddw     %%ymm4,%%ymm1,%%ymm1           \n"
>+      "vpaddw     %%ymm4,%%ymm0,%%ymm0           \n"
>+      "vpsrlw     $0x8,%%ymm1,%%ymm1             \n"
>+      "vpsrlw     $0x8,%%ymm0,%%ymm0             \n"
>+      "vpackuswb  %%ymm1,%%ymm0,%%ymm0           \n"
>+      "vmovdqu    %%ymm0,0x00(%1,%0,1)           \n"
>+      "lea        0x20(%1),%1                    \n"
>+      "sub        $0x20,%2                       \n"
>+      "jg         1b                             \n"
>+      "jmp        99f                            \n"
> 
>-    // Blend 100 / 0 - Copy row unchanged.
>-    LABELALIGN
>-  "100:                                        \n"
>-    "rep movsb " MEMMOVESTRING(1,0) "          \n"
>-    "jmp       999f                            \n"
>+      // Blend 50 / 50.
>+      LABELALIGN
>+      "50:                                       \n"
>+      "vmovdqu   (%1),%%ymm0                     \n"
>+      "vpavgb    0x00(%1,%4,1),%%ymm0,%%ymm0     \n"
>+      "vmovdqu   %%ymm0,0x00(%1,%0,1)            \n"
>+      "lea       0x20(%1),%1                     \n"
>+      "sub       $0x20,%2                        \n"
>+      "jg        50b                             \n"
>+      "jmp       99f                             \n"
>+
>+      // Blend 100 / 0 - Copy row unchanged.
>+      LABELALIGN
>+      "100:                                      \n"
>+      "rep movsb                                 \n"
>+      "jmp       999f                            \n"
> 
>-  "99:                                         \n"
>-    "vzeroupper                                \n"
>-  "999:                                        \n"
>-  : "+D"(dst_ptr),    // %0
>-    "+S"(src_ptr),    // %1
>-    "+cm"(dst_width),  // %2
>-    "+r"(source_y_fraction)  // %3
>-  : "r"((intptr_t)(src_stride))  // %4
>-  : "memory", "cc", "eax", NACL_R14
>-    "xmm0", "xmm1", "xmm2", "xmm4", "xmm5"
>-  );
>+      "99:                                       \n"
>+      "vzeroupper                                \n"
>+      "999:                                      \n"
>+      : "+D"(dst_ptr),               // %0
>+        "+S"(src_ptr),               // %1
>+        "+cm"(dst_width),            // %2
>+        "+r"(source_y_fraction)      // %3
>+      : "r"((intptr_t)(src_stride))  // %4
>+      : "memory", "cc", "eax", "xmm0", "xmm1", "xmm2", "xmm4", "xmm5");
> }
> #endif  // HAS_INTERPOLATEROW_AVX2
> 
> #ifdef HAS_ARGBSHUFFLEROW_SSSE3
> // For BGRAToARGB, ABGRToARGB, RGBAToARGB, and ARGBToRGBA.
>-void ARGBShuffleRow_SSSE3(const uint8* src_argb,
>-                          uint8* dst_argb,
>-                          const uint8* shuffler,
>+void ARGBShuffleRow_SSSE3(const uint8_t* src_argb,
>+                          uint8_t* dst_argb,
>+                          const uint8_t* shuffler,
>                           int width) {
>-  asm volatile (
>-    "movdqu    " MEMACCESS(3) ",%%xmm5         \n"
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    "movdqu    " MEMACCESS2(0x10,0) ",%%xmm1   \n"
>-    "lea       " MEMLEA(0x20,0) ",%0           \n"
>-    "pshufb    %%xmm5,%%xmm0                   \n"
>-    "pshufb    %%xmm5,%%xmm1                   \n"
>-    "movdqu    %%xmm0," MEMACCESS(1) "         \n"
>-    "movdqu    %%xmm1," MEMACCESS2(0x10,1) "   \n"
>-    "lea       " MEMLEA(0x20,1) ",%1           \n"
>-    "sub       $0x8,%2                         \n"
>-    "jg        1b                              \n"
>-  : "+r"(src_argb),  // %0
>-    "+r"(dst_argb),  // %1
>-    "+r"(width)        // %2
>-  : "r"(shuffler)    // %3
>-  : "memory", "cc"
>-    , "xmm0", "xmm1", "xmm5"
>-  );
>+  asm volatile(
>+
>+      "movdqu    (%3),%%xmm5                     \n"
>+
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "movdqu    0x10(%0),%%xmm1                 \n"
>+      "lea       0x20(%0),%0                     \n"
>+      "pshufb    %%xmm5,%%xmm0                   \n"
>+      "pshufb    %%xmm5,%%xmm1                   \n"
>+      "movdqu    %%xmm0,(%1)                     \n"
>+      "movdqu    %%xmm1,0x10(%1)                 \n"
>+      "lea       0x20(%1),%1                     \n"
>+      "sub       $0x8,%2                         \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_argb),  // %0
>+        "+r"(dst_argb),  // %1
>+        "+r"(width)      // %2
>+      : "r"(shuffler)    // %3
>+      : "memory", "cc", "xmm0", "xmm1", "xmm5");
> }
> #endif  // HAS_ARGBSHUFFLEROW_SSSE3
> 
> #ifdef HAS_ARGBSHUFFLEROW_AVX2
> // For BGRAToARGB, ABGRToARGB, RGBAToARGB, and ARGBToRGBA.
>-void ARGBShuffleRow_AVX2(const uint8* src_argb,
>-                         uint8* dst_argb,
>-                         const uint8* shuffler,
>+void ARGBShuffleRow_AVX2(const uint8_t* src_argb,
>+                         uint8_t* dst_argb,
>+                         const uint8_t* shuffler,
>                          int width) {
>-  asm volatile (
>-    "vbroadcastf128 " MEMACCESS(3) ",%%ymm5    \n"
>-    LABELALIGN
>-    "1:                                        \n"
>-    "vmovdqu   " MEMACCESS(0) ",%%ymm0         \n"
>-    "vmovdqu   " MEMACCESS2(0x20,0) ",%%ymm1   \n"
>-    "lea       " MEMLEA(0x40,0) ",%0           \n"
>-    "vpshufb   %%ymm5,%%ymm0,%%ymm0            \n"
>-    "vpshufb   %%ymm5,%%ymm1,%%ymm1            \n"
>-    "vmovdqu   %%ymm0," MEMACCESS(1) "         \n"
>-    "vmovdqu   %%ymm1," MEMACCESS2(0x20,1) "   \n"
>-    "lea       " MEMLEA(0x40,1) ",%1           \n"
>-    "sub       $0x10,%2                        \n"
>-    "jg        1b                              \n"
>-    "vzeroupper                                \n"
>-  : "+r"(src_argb),  // %0
>-    "+r"(dst_argb),  // %1
>-    "+r"(width)        // %2
>-  : "r"(shuffler)    // %3
>-  : "memory", "cc"
>-    , "xmm0", "xmm1", "xmm5"
>-  );
>+  asm volatile(
>+
>+      "vbroadcastf128 (%3),%%ymm5                \n"
>+
>+      LABELALIGN
>+      "1:                                        \n"
>+      "vmovdqu   (%0),%%ymm0                     \n"
>+      "vmovdqu   0x20(%0),%%ymm1                 \n"
>+      "lea       0x40(%0),%0                     \n"
>+      "vpshufb   %%ymm5,%%ymm0,%%ymm0            \n"
>+      "vpshufb   %%ymm5,%%ymm1,%%ymm1            \n"
>+      "vmovdqu   %%ymm0,(%1)                     \n"
>+      "vmovdqu   %%ymm1,0x20(%1)                 \n"
>+      "lea       0x40(%1),%1                     \n"
>+      "sub       $0x10,%2                        \n"
>+      "jg        1b                              \n"
>+      "vzeroupper                                \n"
>+      : "+r"(src_argb),  // %0
>+        "+r"(dst_argb),  // %1
>+        "+r"(width)      // %2
>+      : "r"(shuffler)    // %3
>+      : "memory", "cc", "xmm0", "xmm1", "xmm5");
> }
> #endif  // HAS_ARGBSHUFFLEROW_AVX2
> 
>-#ifdef HAS_ARGBSHUFFLEROW_SSE2
>-// For BGRAToARGB, ABGRToARGB, RGBAToARGB, and ARGBToRGBA.
>-void ARGBShuffleRow_SSE2(const uint8* src_argb,
>-                         uint8* dst_argb,
>-                         const uint8* shuffler,
>-                         int width) {
>-  uintptr_t pixel_temp;
>-  asm volatile (
>-    "pxor      %%xmm5,%%xmm5                   \n"
>-    "mov       " MEMACCESS(4) ",%k2            \n"
>-    "cmp       $0x3000102,%k2                  \n"
>-    "je        3012f                           \n"
>-    "cmp       $0x10203,%k2                    \n"
>-    "je        123f                            \n"
>-    "cmp       $0x30201,%k2                    \n"
>-    "je        321f                            \n"
>-    "cmp       $0x2010003,%k2                  \n"
>-    "je        2103f                           \n"
>+#ifdef HAS_I422TOYUY2ROW_SSE2
>+void I422ToYUY2Row_SSE2(const uint8_t* src_y,
>+                        const uint8_t* src_u,
>+                        const uint8_t* src_v,
>+                        uint8_t* dst_yuy2,
>+                        int width) {
>+  asm volatile(
> 
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movzb     " MEMACCESS(4) ",%2             \n"
>-    MEMOPARG(movzb,0x00,0,2,1,2) "             \n"  //  movzb     (%0,%2,1),%2
>-    "mov       %b2," MEMACCESS(1) "            \n"
>-    "movzb     " MEMACCESS2(0x1,4) ",%2        \n"
>-    MEMOPARG(movzb,0x00,0,2,1,2) "             \n"  //  movzb     (%0,%2,1),%2
>-    "mov       %b2," MEMACCESS2(0x1,1) "       \n"
>-    "movzb     " MEMACCESS2(0x2,4) ",%2        \n"
>-    MEMOPARG(movzb,0x00,0,2,1,2) "             \n"  //  movzb     (%0,%2,1),%2
>-    "mov       %b2," MEMACCESS2(0x2,1) "       \n"
>-    "movzb     " MEMACCESS2(0x3,4) ",%2        \n"
>-    MEMOPARG(movzb,0x00,0,2,1,2) "             \n"  //  movzb     (%0,%2,1),%2
>-    "mov       %b2," MEMACCESS2(0x3,1) "       \n"
>-    "lea       " MEMLEA(0x4,0) ",%0            \n"
>-    "lea       " MEMLEA(0x4,1) ",%1            \n"
>-    "sub       $0x1,%3                         \n"
>-    "jg        1b                              \n"
>-    "jmp       99f                             \n"
>+      "sub       %1,%2                             \n"
> 
>-    LABELALIGN
>-  "123:                                        \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    "lea       " MEMLEA(0x10,0) ",%0           \n"
>-    "movdqa    %%xmm0,%%xmm1                   \n"
>-    "punpcklbw %%xmm5,%%xmm0                   \n"
>-    "punpckhbw %%xmm5,%%xmm1                   \n"
>-    "pshufhw   $0x1b,%%xmm0,%%xmm0             \n"
>-    "pshuflw   $0x1b,%%xmm0,%%xmm0             \n"
>-    "pshufhw   $0x1b,%%xmm1,%%xmm1             \n"
>-    "pshuflw   $0x1b,%%xmm1,%%xmm1             \n"
>-    "packuswb  %%xmm1,%%xmm0                   \n"
>-    "movdqu    %%xmm0," MEMACCESS(1) "         \n"
>-    "lea       " MEMLEA(0x10,1) ",%1           \n"
>-    "sub       $0x4,%3                         \n"
>-    "jg        123b                            \n"
>-    "jmp       99f                             \n"
>+      LABELALIGN
>+      "1:                                          \n"
>+      "movq      (%1),%%xmm2                       \n"
>+      "movq      0x00(%1,%2,1),%%xmm1              \n"
>+      "add       $0x8,%1                           \n"
>+      "punpcklbw %%xmm1,%%xmm2                     \n"
>+      "movdqu    (%0),%%xmm0                       \n"
>+      "add       $0x10,%0                          \n"
>+      "movdqa    %%xmm0,%%xmm1                     \n"
>+      "punpcklbw %%xmm2,%%xmm0                     \n"
>+      "punpckhbw %%xmm2,%%xmm1                     \n"
>+      "movdqu    %%xmm0,(%3)                       \n"
>+      "movdqu    %%xmm1,0x10(%3)                   \n"
>+      "lea       0x20(%3),%3                       \n"
>+      "sub       $0x10,%4                          \n"
>+      "jg         1b                               \n"
>+      : "+r"(src_y),     // %0
>+        "+r"(src_u),     // %1
>+        "+r"(src_v),     // %2
>+        "+r"(dst_yuy2),  // %3
>+        "+rm"(width)     // %4
>+      :
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2");
>+}
>+#endif  // HAS_I422TOYUY2ROW_SSE2
> 
>-    LABELALIGN
>-  "321:                                        \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    "lea       " MEMLEA(0x10,0) ",%0           \n"
>-    "movdqa    %%xmm0,%%xmm1                   \n"
>-    "punpcklbw %%xmm5,%%xmm0                   \n"
>-    "punpckhbw %%xmm5,%%xmm1                   \n"
>-    "pshufhw   $0x39,%%xmm0,%%xmm0             \n"
>-    "pshuflw   $0x39,%%xmm0,%%xmm0             \n"
>-    "pshufhw   $0x39,%%xmm1,%%xmm1             \n"
>-    "pshuflw   $0x39,%%xmm1,%%xmm1             \n"
>-    "packuswb  %%xmm1,%%xmm0                   \n"
>-    "movdqu    %%xmm0," MEMACCESS(1) "         \n"
>-    "lea       " MEMLEA(0x10,1) ",%1           \n"
>-    "sub       $0x4,%3                         \n"
>-    "jg        321b                            \n"
>-    "jmp       99f                             \n"
>+#ifdef HAS_I422TOUYVYROW_SSE2
>+void I422ToUYVYRow_SSE2(const uint8_t* src_y,
>+                        const uint8_t* src_u,
>+                        const uint8_t* src_v,
>+                        uint8_t* dst_uyvy,
>+                        int width) {
>+  asm volatile(
> 
>-    LABELALIGN
>-  "2103:                                       \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    "lea       " MEMLEA(0x10,0) ",%0           \n"
>-    "movdqa    %%xmm0,%%xmm1                   \n"
>-    "punpcklbw %%xmm5,%%xmm0                   \n"
>-    "punpckhbw %%xmm5,%%xmm1                   \n"
>-    "pshufhw   $0x93,%%xmm0,%%xmm0             \n"
>-    "pshuflw   $0x93,%%xmm0,%%xmm0             \n"
>-    "pshufhw   $0x93,%%xmm1,%%xmm1             \n"
>-    "pshuflw   $0x93,%%xmm1,%%xmm1             \n"
>-    "packuswb  %%xmm1,%%xmm0                   \n"
>-    "movdqu    %%xmm0," MEMACCESS(1) "         \n"
>-    "lea       " MEMLEA(0x10,1) ",%1           \n"
>-    "sub       $0x4,%3                         \n"
>-    "jg        2103b                           \n"
>-    "jmp       99f                             \n"
>+      "sub        %1,%2                            \n"
> 
>-    LABELALIGN
>-  "3012:                                       \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    "lea       " MEMLEA(0x10,0) ",%0           \n"
>-    "movdqa    %%xmm0,%%xmm1                   \n"
>-    "punpcklbw %%xmm5,%%xmm0                   \n"
>-    "punpckhbw %%xmm5,%%xmm1                   \n"
>-    "pshufhw   $0xc6,%%xmm0,%%xmm0             \n"
>-    "pshuflw   $0xc6,%%xmm0,%%xmm0             \n"
>-    "pshufhw   $0xc6,%%xmm1,%%xmm1             \n"
>-    "pshuflw   $0xc6,%%xmm1,%%xmm1             \n"
>-    "packuswb  %%xmm1,%%xmm0                   \n"
>-    "movdqu    %%xmm0," MEMACCESS(1) "         \n"
>-    "lea       " MEMLEA(0x10,1) ",%1           \n"
>-    "sub       $0x4,%3                         \n"
>-    "jg        3012b                           \n"
>-
>-  "99:                                         \n"
>-  : "+r"(src_argb),     // %0
>-    "+r"(dst_argb),     // %1
>-    "=&d"(pixel_temp),  // %2
>-    "+r"(width)         // %3
>-  : "r"(shuffler)       // %4
>-  : "memory", "cc", NACL_R14
>-    "xmm0", "xmm1", "xmm5"
>-  );
>+      LABELALIGN
>+      "1:                                          \n"
>+      "movq      (%1),%%xmm2                       \n"
>+      "movq      0x00(%1,%2,1),%%xmm1              \n"
>+      "add       $0x8,%1                           \n"
>+      "punpcklbw %%xmm1,%%xmm2                     \n"
>+      "movdqu    (%0),%%xmm0                       \n"
>+      "movdqa    %%xmm2,%%xmm1                     \n"
>+      "add       $0x10,%0                          \n"
>+      "punpcklbw %%xmm0,%%xmm1                     \n"
>+      "punpckhbw %%xmm0,%%xmm2                     \n"
>+      "movdqu    %%xmm1,(%3)                       \n"
>+      "movdqu    %%xmm2,0x10(%3)                   \n"
>+      "lea       0x20(%3),%3                       \n"
>+      "sub       $0x10,%4                          \n"
>+      "jg         1b                               \n"
>+      : "+r"(src_y),     // %0
>+        "+r"(src_u),     // %1
>+        "+r"(src_v),     // %2
>+        "+r"(dst_uyvy),  // %3
>+        "+rm"(width)     // %4
>+      :
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2");
> }
>-#endif  // HAS_ARGBSHUFFLEROW_SSE2
>+#endif  // HAS_I422TOUYVYROW_SSE2
> 
>-#ifdef HAS_I422TOYUY2ROW_SSE2
>-void I422ToYUY2Row_SSE2(const uint8* src_y,
>-                        const uint8* src_u,
>-                        const uint8* src_v,
>-                        uint8* dst_frame,
>+#ifdef HAS_I422TOYUY2ROW_AVX2
>+void I422ToYUY2Row_AVX2(const uint8_t* src_y,
>+                        const uint8_t* src_u,
>+                        const uint8_t* src_v,
>+                        uint8_t* dst_yuy2,
>                         int width) {
>-  asm volatile (
>-    "sub       %1,%2                             \n"
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movq      " MEMACCESS(1) ",%%xmm2           \n"
>-    MEMOPREG(movq,0x00,1,2,1,xmm3)               //  movq    (%1,%2,1),%%xmm3
>-    "lea       " MEMLEA(0x8,1) ",%1              \n"
>-    "punpcklbw %%xmm3,%%xmm2                     \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0           \n"
>-    "lea       " MEMLEA(0x10,0) ",%0             \n"
>-    "movdqa    %%xmm0,%%xmm1                     \n"
>-    "punpcklbw %%xmm2,%%xmm0                     \n"
>-    "punpckhbw %%xmm2,%%xmm1                     \n"
>-    "movdqu    %%xmm0," MEMACCESS(3) "           \n"
>-    "movdqu    %%xmm1," MEMACCESS2(0x10,3) "     \n"
>-    "lea       " MEMLEA(0x20,3) ",%3             \n"
>-    "sub       $0x10,%4                          \n"
>-    "jg         1b                               \n"
>-    : "+r"(src_y),  // %0
>-      "+r"(src_u),  // %1
>-      "+r"(src_v),  // %2
>-      "+r"(dst_frame),  // %3
>-      "+rm"(width)  // %4
>-    :
>-    : "memory", "cc", NACL_R14
>-    "xmm0", "xmm1", "xmm2", "xmm3"
>-  );
>+  asm volatile(
>+
>+      "sub       %1,%2                             \n"
>+
>+      LABELALIGN
>+      "1:                                          \n"
>+      "vpmovzxbw  (%1),%%ymm1                      \n"
>+      "vpmovzxbw  0x00(%1,%2,1),%%ymm2             \n"
>+      "add        $0x10,%1                         \n"
>+      "vpsllw     $0x8,%%ymm2,%%ymm2               \n"
>+      "vpor       %%ymm1,%%ymm2,%%ymm2             \n"
>+      "vmovdqu    (%0),%%ymm0                      \n"
>+      "add        $0x20,%0                         \n"
>+      "vpunpcklbw %%ymm2,%%ymm0,%%ymm1             \n"
>+      "vpunpckhbw %%ymm2,%%ymm0,%%ymm2             \n"
>+      "vextractf128 $0x0,%%ymm1,(%3)               \n"
>+      "vextractf128 $0x0,%%ymm2,0x10(%3)           \n"
>+      "vextractf128 $0x1,%%ymm1,0x20(%3)           \n"
>+      "vextractf128 $0x1,%%ymm2,0x30(%3)           \n"
>+      "lea        0x40(%3),%3                      \n"
>+      "sub        $0x20,%4                         \n"
>+      "jg         1b                               \n"
>+      "vzeroupper                                  \n"
>+      : "+r"(src_y),     // %0
>+        "+r"(src_u),     // %1
>+        "+r"(src_v),     // %2
>+        "+r"(dst_yuy2),  // %3
>+        "+rm"(width)     // %4
>+      :
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2");
> }
>-#endif  // HAS_I422TOYUY2ROW_SSE2
>+#endif  // HAS_I422TOYUY2ROW_AVX2
> 
>-#ifdef HAS_I422TOUYVYROW_SSE2
>-void I422ToUYVYRow_SSE2(const uint8* src_y,
>-                        const uint8* src_u,
>-                        const uint8* src_v,
>-                        uint8* dst_frame,
>+#ifdef HAS_I422TOUYVYROW_AVX2
>+void I422ToUYVYRow_AVX2(const uint8_t* src_y,
>+                        const uint8_t* src_u,
>+                        const uint8_t* src_v,
>+                        uint8_t* dst_uyvy,
>                         int width) {
>-  asm volatile (
>-    "sub        %1,%2                            \n"
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movq      " MEMACCESS(1) ",%%xmm2           \n"
>-    MEMOPREG(movq,0x00,1,2,1,xmm3)               //  movq    (%1,%2,1),%%xmm3
>-    "lea       " MEMLEA(0x8,1) ",%1              \n"
>-    "punpcklbw %%xmm3,%%xmm2                     \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0           \n"
>-    "movdqa    %%xmm2,%%xmm1                     \n"
>-    "lea       " MEMLEA(0x10,0) ",%0             \n"
>-    "punpcklbw %%xmm0,%%xmm1                     \n"
>-    "punpckhbw %%xmm0,%%xmm2                     \n"
>-    "movdqu    %%xmm1," MEMACCESS(3) "           \n"
>-    "movdqu    %%xmm2," MEMACCESS2(0x10,3) "     \n"
>-    "lea       " MEMLEA(0x20,3) ",%3             \n"
>-    "sub       $0x10,%4                          \n"
>-    "jg         1b                               \n"
>-    : "+r"(src_y),  // %0
>-      "+r"(src_u),  // %1
>-      "+r"(src_v),  // %2
>-      "+r"(dst_frame),  // %3
>-      "+rm"(width)  // %4
>-    :
>-    : "memory", "cc", NACL_R14
>-    "xmm0", "xmm1", "xmm2", "xmm3"
>-  );
>+  asm volatile(
>+
>+      "sub        %1,%2                            \n"
>+
>+      LABELALIGN
>+      "1:                                          \n"
>+      "vpmovzxbw  (%1),%%ymm1                      \n"
>+      "vpmovzxbw  0x00(%1,%2,1),%%ymm2             \n"
>+      "add        $0x10,%1                         \n"
>+      "vpsllw     $0x8,%%ymm2,%%ymm2               \n"
>+      "vpor       %%ymm1,%%ymm2,%%ymm2             \n"
>+      "vmovdqu    (%0),%%ymm0                      \n"
>+      "add        $0x20,%0                         \n"
>+      "vpunpcklbw %%ymm0,%%ymm2,%%ymm1             \n"
>+      "vpunpckhbw %%ymm0,%%ymm2,%%ymm2             \n"
>+      "vextractf128 $0x0,%%ymm1,(%3)               \n"
>+      "vextractf128 $0x0,%%ymm2,0x10(%3)           \n"
>+      "vextractf128 $0x1,%%ymm1,0x20(%3)           \n"
>+      "vextractf128 $0x1,%%ymm2,0x30(%3)           \n"
>+      "lea        0x40(%3),%3                      \n"
>+      "sub        $0x20,%4                         \n"
>+      "jg         1b                               \n"
>+      "vzeroupper                                  \n"
>+      : "+r"(src_y),     // %0
>+        "+r"(src_u),     // %1
>+        "+r"(src_v),     // %2
>+        "+r"(dst_uyvy),  // %3
>+        "+rm"(width)     // %4
>+      :
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2");
> }
>-#endif  // HAS_I422TOUYVYROW_SSE2
>+#endif  // HAS_I422TOUYVYROW_AVX2
> 
> #ifdef HAS_ARGBPOLYNOMIALROW_SSE2
>-void ARGBPolynomialRow_SSE2(const uint8* src_argb,
>-                            uint8* dst_argb,
>+void ARGBPolynomialRow_SSE2(const uint8_t* src_argb,
>+                            uint8_t* dst_argb,
>                             const float* poly,
>                             int width) {
>-  asm volatile (
>-    "pxor      %%xmm3,%%xmm3                   \n"
>+  asm volatile(
> 
>-    // 2 pixel loop.
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movq      " MEMACCESS(0) ",%%xmm0         \n"
>-    "lea       " MEMLEA(0x8,0) ",%0            \n"
>-    "punpcklbw %%xmm3,%%xmm0                   \n"
>-    "movdqa    %%xmm0,%%xmm4                   \n"
>-    "punpcklwd %%xmm3,%%xmm0                   \n"
>-    "punpckhwd %%xmm3,%%xmm4                   \n"
>-    "cvtdq2ps  %%xmm0,%%xmm0                   \n"
>-    "cvtdq2ps  %%xmm4,%%xmm4                   \n"
>-    "movdqa    %%xmm0,%%xmm1                   \n"
>-    "movdqa    %%xmm4,%%xmm5                   \n"
>-    "mulps     " MEMACCESS2(0x10,3) ",%%xmm0   \n"
>-    "mulps     " MEMACCESS2(0x10,3) ",%%xmm4   \n"
>-    "addps     " MEMACCESS(3) ",%%xmm0         \n"
>-    "addps     " MEMACCESS(3) ",%%xmm4         \n"
>-    "movdqa    %%xmm1,%%xmm2                   \n"
>-    "movdqa    %%xmm5,%%xmm6                   \n"
>-    "mulps     %%xmm1,%%xmm2                   \n"
>-    "mulps     %%xmm5,%%xmm6                   \n"
>-    "mulps     %%xmm2,%%xmm1                   \n"
>-    "mulps     %%xmm6,%%xmm5                   \n"
>-    "mulps     " MEMACCESS2(0x20,3) ",%%xmm2   \n"
>-    "mulps     " MEMACCESS2(0x20,3) ",%%xmm6   \n"
>-    "mulps     " MEMACCESS2(0x30,3) ",%%xmm1   \n"
>-    "mulps     " MEMACCESS2(0x30,3) ",%%xmm5   \n"
>-    "addps     %%xmm2,%%xmm0                   \n"
>-    "addps     %%xmm6,%%xmm4                   \n"
>-    "addps     %%xmm1,%%xmm0                   \n"
>-    "addps     %%xmm5,%%xmm4                   \n"
>-    "cvttps2dq %%xmm0,%%xmm0                   \n"
>-    "cvttps2dq %%xmm4,%%xmm4                   \n"
>-    "packuswb  %%xmm4,%%xmm0                   \n"
>-    "packuswb  %%xmm0,%%xmm0                   \n"
>-    "movq      %%xmm0," MEMACCESS(1) "         \n"
>-    "lea       " MEMLEA(0x8,1) ",%1            \n"
>-    "sub       $0x2,%2                         \n"
>-    "jg        1b                              \n"
>-  : "+r"(src_argb),  // %0
>-    "+r"(dst_argb),  // %1
>-    "+r"(width)      // %2
>-  : "r"(poly)        // %3
>-  : "memory", "cc"
>-    , "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6"
>-  );
>+      "pxor      %%xmm3,%%xmm3                   \n"
>+
>+      // 2 pixel loop.
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movq      (%0),%%xmm0                     \n"
>+      "lea       0x8(%0),%0                      \n"
>+      "punpcklbw %%xmm3,%%xmm0                   \n"
>+      "movdqa    %%xmm0,%%xmm4                   \n"
>+      "punpcklwd %%xmm3,%%xmm0                   \n"
>+      "punpckhwd %%xmm3,%%xmm4                   \n"
>+      "cvtdq2ps  %%xmm0,%%xmm0                   \n"
>+      "cvtdq2ps  %%xmm4,%%xmm4                   \n"
>+      "movdqa    %%xmm0,%%xmm1                   \n"
>+      "movdqa    %%xmm4,%%xmm5                   \n"
>+      "mulps     0x10(%3),%%xmm0                 \n"
>+      "mulps     0x10(%3),%%xmm4                 \n"
>+      "addps     (%3),%%xmm0                     \n"
>+      "addps     (%3),%%xmm4                     \n"
>+      "movdqa    %%xmm1,%%xmm2                   \n"
>+      "movdqa    %%xmm5,%%xmm6                   \n"
>+      "mulps     %%xmm1,%%xmm2                   \n"
>+      "mulps     %%xmm5,%%xmm6                   \n"
>+      "mulps     %%xmm2,%%xmm1                   \n"
>+      "mulps     %%xmm6,%%xmm5                   \n"
>+      "mulps     0x20(%3),%%xmm2                 \n"
>+      "mulps     0x20(%3),%%xmm6                 \n"
>+      "mulps     0x30(%3),%%xmm1                 \n"
>+      "mulps     0x30(%3),%%xmm5                 \n"
>+      "addps     %%xmm2,%%xmm0                   \n"
>+      "addps     %%xmm6,%%xmm4                   \n"
>+      "addps     %%xmm1,%%xmm0                   \n"
>+      "addps     %%xmm5,%%xmm4                   \n"
>+      "cvttps2dq %%xmm0,%%xmm0                   \n"
>+      "cvttps2dq %%xmm4,%%xmm4                   \n"
>+      "packuswb  %%xmm4,%%xmm0                   \n"
>+      "packuswb  %%xmm0,%%xmm0                   \n"
>+      "movq      %%xmm0,(%1)                     \n"
>+      "lea       0x8(%1),%1                      \n"
>+      "sub       $0x2,%2                         \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_argb),  // %0
>+        "+r"(dst_argb),  // %1
>+        "+r"(width)      // %2
>+      : "r"(poly)        // %3
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6");
> }
> #endif  // HAS_ARGBPOLYNOMIALROW_SSE2
> 
> #ifdef HAS_ARGBPOLYNOMIALROW_AVX2
>-void ARGBPolynomialRow_AVX2(const uint8* src_argb,
>-                            uint8* dst_argb,
>+void ARGBPolynomialRow_AVX2(const uint8_t* src_argb,
>+                            uint8_t* dst_argb,
>                             const float* poly,
>                             int width) {
>-  asm volatile (
>-    "vbroadcastf128 " MEMACCESS(3) ",%%ymm4     \n"
>-    "vbroadcastf128 " MEMACCESS2(0x10,3) ",%%ymm5 \n"
>-    "vbroadcastf128 " MEMACCESS2(0x20,3) ",%%ymm6 \n"
>-    "vbroadcastf128 " MEMACCESS2(0x30,3) ",%%ymm7 \n"
>+  asm volatile(
>+      "vbroadcastf128 (%3),%%ymm4                \n"
>+      "vbroadcastf128 0x10(%3),%%ymm5            \n"
>+      "vbroadcastf128 0x20(%3),%%ymm6            \n"
>+      "vbroadcastf128 0x30(%3),%%ymm7            \n"
> 
>-    // 2 pixel loop.
>-    LABELALIGN
>-    "1:                                        \n"
>-    "vpmovzxbd   " MEMACCESS(0) ",%%ymm0       \n"  // 2 ARGB pixels
>-    "lea         " MEMLEA(0x8,0) ",%0          \n"
>-    "vcvtdq2ps   %%ymm0,%%ymm0                 \n"  // X 8 floats
>-    "vmulps      %%ymm0,%%ymm0,%%ymm2          \n"  // X * X
>-    "vmulps      %%ymm7,%%ymm0,%%ymm3          \n"  // C3 * X
>-    "vfmadd132ps %%ymm5,%%ymm4,%%ymm0          \n"  // result = C0 + C1 * X
>-    "vfmadd231ps %%ymm6,%%ymm2,%%ymm0          \n"  // result += C2 * X * X
>-    "vfmadd231ps %%ymm3,%%ymm2,%%ymm0          \n"  // result += C3 * X * X * X
>-    "vcvttps2dq  %%ymm0,%%ymm0                 \n"
>-    "vpackusdw   %%ymm0,%%ymm0,%%ymm0          \n"
>-    "vpermq      $0xd8,%%ymm0,%%ymm0           \n"
>-    "vpackuswb   %%xmm0,%%xmm0,%%xmm0          \n"
>-    "vmovq       %%xmm0," MEMACCESS(1) "       \n"
>-    "lea         " MEMLEA(0x8,1) ",%1          \n"
>-    "sub         $0x2,%2                       \n"
>-    "jg          1b                            \n"
>-    "vzeroupper                                \n"
>-  : "+r"(src_argb),  // %0
>-    "+r"(dst_argb),  // %1
>-    "+r"(width)      // %2
>-  : "r"(poly)        // %3
>-  : "memory", "cc",
>-    "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6", "xmm7"
>-  );
>+      // 2 pixel loop.
>+      LABELALIGN
>+      "1:                                        \n"
>+      "vpmovzxbd   (%0),%%ymm0                   \n"  // 2 ARGB pixels
>+      "lea         0x8(%0),%0                    \n"
>+      "vcvtdq2ps   %%ymm0,%%ymm0                 \n"  // X 8 floats
>+      "vmulps      %%ymm0,%%ymm0,%%ymm2          \n"  // X * X
>+      "vmulps      %%ymm7,%%ymm0,%%ymm3          \n"  // C3 * X
>+      "vfmadd132ps %%ymm5,%%ymm4,%%ymm0          \n"  // result = C0 + C1 * X
>+      "vfmadd231ps %%ymm6,%%ymm2,%%ymm0          \n"  // result += C2 * X * X
>+      "vfmadd231ps %%ymm3,%%ymm2,%%ymm0          \n"  // result += C3 * X * X *
>+                                                      // X
>+      "vcvttps2dq  %%ymm0,%%ymm0                 \n"
>+      "vpackusdw   %%ymm0,%%ymm0,%%ymm0          \n"
>+      "vpermq      $0xd8,%%ymm0,%%ymm0           \n"
>+      "vpackuswb   %%xmm0,%%xmm0,%%xmm0          \n"
>+      "vmovq       %%xmm0,(%1)                   \n"
>+      "lea         0x8(%1),%1                    \n"
>+      "sub         $0x2,%2                       \n"
>+      "jg          1b                            \n"
>+      "vzeroupper                                \n"
>+      : "+r"(src_argb),  // %0
>+        "+r"(dst_argb),  // %1
>+        "+r"(width)      // %2
>+      : "r"(poly)        // %3
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6",
>+        "xmm7");
> }
> #endif  // HAS_ARGBPOLYNOMIALROW_AVX2
> 
> #ifdef HAS_HALFFLOATROW_SSE2
> static float kScaleBias = 1.9259299444e-34f;
>-void HalfFloatRow_SSE2(const uint16* src, uint16* dst, float scale, int width) {
>+void HalfFloatRow_SSE2(const uint16_t* src,
>+                       uint16_t* dst,
>+                       float scale,
>+                       int width) {
>   scale *= kScaleBias;
>-  asm volatile (
>-    "pshufd      $0x0,%3,%%xmm4                \n"
>-    "pxor        %%xmm5,%%xmm5                 \n"
>-    "sub         %0,%1                         \n"
>+  asm volatile(
>+      "movd        %3,%%xmm4                     \n"
>+      "pshufd      $0x0,%%xmm4,%%xmm4            \n"
>+      "pxor        %%xmm5,%%xmm5                 \n"
>+      "sub         %0,%1                         \n"
> 
>-    // 16 pixel loop.
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movdqu      " MEMACCESS(0) ",%%xmm2       \n"  // 8 shorts
>-    "add         $0x10,%0                      \n"
>-    "movdqa      %%xmm2,%%xmm3                 \n"
>-    "punpcklwd   %%xmm5,%%xmm2                 \n"  // 8 ints in xmm2/1
>-    "cvtdq2ps    %%xmm2,%%xmm2                 \n"  // 8 floats
>-    "punpckhwd   %%xmm5,%%xmm3                 \n"
>-    "cvtdq2ps    %%xmm3,%%xmm3                 \n"
>-    "mulps       %%xmm4,%%xmm2                 \n"
>-    "mulps       %%xmm4,%%xmm3                 \n"
>-    "psrld       $0xd,%%xmm2                   \n"
>-    "psrld       $0xd,%%xmm3                   \n"
>-    "packssdw    %%xmm3,%%xmm2                 \n"
>-    MEMOPMEM(movdqu,xmm2,-0x10,0,1,1)
>-    "sub         $0x8,%2                       \n"
>-    "jg          1b                            \n"
>-  : "+r"(src),    // %0
>-    "+r"(dst),    // %1
>-    "+r"(width)   // %2
>-#if defined(__x86_64__)
>-  : "x"(scale)   // %3
>-#else
>-  : "m"(scale)   // %3
>-#endif
>-  : "memory", "cc",
>-    "xmm2", "xmm3", "xmm4", "xmm5"
>-  );
>+      // 16 pixel loop.
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu      (%0),%%xmm2                   \n"  // 8 shorts
>+      "add         $0x10,%0                      \n"
>+      "movdqa      %%xmm2,%%xmm3                 \n"
>+      "punpcklwd   %%xmm5,%%xmm2                 \n"  // 8 ints in xmm2/1
>+      "cvtdq2ps    %%xmm2,%%xmm2                 \n"  // 8 floats
>+      "punpckhwd   %%xmm5,%%xmm3                 \n"
>+      "cvtdq2ps    %%xmm3,%%xmm3                 \n"
>+      "mulps       %%xmm4,%%xmm2                 \n"
>+      "mulps       %%xmm4,%%xmm3                 \n"
>+      "psrld       $0xd,%%xmm2                   \n"
>+      "psrld       $0xd,%%xmm3                   \n"
>+      "packssdw    %%xmm3,%%xmm2                 \n"
>+      "movdqu      %%xmm2,-0x10(%0,%1,1)         \n"
>+      "sub         $0x8,%2                       \n"
>+      "jg          1b                            \n"
>+      : "+r"(src),   // %0
>+        "+r"(dst),   // %1
>+        "+r"(width)  // %2
>+      : "m"(scale)   // %3
>+      : "memory", "cc", "xmm2", "xmm3", "xmm4", "xmm5");
> }
> #endif  // HAS_HALFFLOATROW_SSE2
> 
> #ifdef HAS_HALFFLOATROW_AVX2
>-void HalfFloatRow_AVX2(const uint16* src, uint16* dst, float scale, int width) {
>+void HalfFloatRow_AVX2(const uint16_t* src,
>+                       uint16_t* dst,
>+                       float scale,
>+                       int width) {
>   scale *= kScaleBias;
>-  asm volatile (
>-    "vbroadcastss  %3, %%ymm4                  \n"
>-    "vpxor      %%ymm5,%%ymm5,%%ymm5           \n"
>-    "sub        %0,%1                          \n"
>+  asm volatile(
>+      "vbroadcastss  %3, %%ymm4                  \n"
>+      "vpxor      %%ymm5,%%ymm5,%%ymm5           \n"
>+      "sub        %0,%1                          \n"
> 
>-    // 16 pixel loop.
>-    LABELALIGN
>-    "1:                                        \n"
>-    "vmovdqu    " MEMACCESS(0) ",%%ymm2        \n"  // 16 shorts
>-    "add        $0x20,%0                       \n"
>-    "vpunpckhwd %%ymm5,%%ymm2,%%ymm3           \n"  // mutates
>-    "vpunpcklwd %%ymm5,%%ymm2,%%ymm2           \n"
>-    "vcvtdq2ps  %%ymm3,%%ymm3                  \n"
>-    "vcvtdq2ps  %%ymm2,%%ymm2                  \n"
>-    "vmulps     %%ymm3,%%ymm4,%%ymm3           \n"
>-    "vmulps     %%ymm2,%%ymm4,%%ymm2           \n"
>-    "vpsrld     $0xd,%%ymm3,%%ymm3             \n"
>-    "vpsrld     $0xd,%%ymm2,%%ymm2             \n"
>-    "vpackssdw  %%ymm3, %%ymm2, %%ymm2         \n"  // unmutates
>-    MEMOPMEM(vmovdqu,ymm2,-0x20,0,1,1)
>-    "sub        $0x10,%2                       \n"
>-    "jg         1b                             \n"
>+      // 16 pixel loop.
>+      LABELALIGN
>+      "1:                                        \n"
>+      "vmovdqu    (%0),%%ymm2                    \n"  // 16 shorts
>+      "add        $0x20,%0                       \n"
>+      "vpunpckhwd %%ymm5,%%ymm2,%%ymm3           \n"  // mutates
>+      "vpunpcklwd %%ymm5,%%ymm2,%%ymm2           \n"
>+      "vcvtdq2ps  %%ymm3,%%ymm3                  \n"
>+      "vcvtdq2ps  %%ymm2,%%ymm2                  \n"
>+      "vmulps     %%ymm3,%%ymm4,%%ymm3           \n"
>+      "vmulps     %%ymm2,%%ymm4,%%ymm2           \n"
>+      "vpsrld     $0xd,%%ymm3,%%ymm3             \n"
>+      "vpsrld     $0xd,%%ymm2,%%ymm2             \n"
>+      "vpackssdw  %%ymm3, %%ymm2, %%ymm2         \n"  // unmutates
>+      "vmovdqu    %%ymm2,-0x20(%0,%1,1)          \n"
>+      "sub        $0x10,%2                       \n"
>+      "jg         1b                             \n"
> 
>-    "vzeroupper                                \n"
>-  : "+r"(src),    // %0
>-    "+r"(dst),    // %1
>-    "+r"(width)   // %2
>+      "vzeroupper                                \n"
>+      : "+r"(src),   // %0
>+        "+r"(dst),   // %1
>+        "+r"(width)  // %2
> #if defined(__x86_64__)
>-  : "x"(scale)   // %3
>+      : "x"(scale)  // %3
> #else
>-  : "m"(scale)   // %3
>+      : "m"(scale)  // %3
> #endif
>-  : "memory", "cc",
>-    "xmm2", "xmm3", "xmm4", "xmm5"
>-  );
>+      : "memory", "cc", "xmm2", "xmm3", "xmm4", "xmm5");
> }
> #endif  // HAS_HALFFLOATROW_AVX2
> 
> #ifdef HAS_HALFFLOATROW_F16C
>-void HalfFloatRow_F16C(const uint16* src, uint16* dst, float scale, int width) {
>-  asm volatile (
>-    "vbroadcastss  %3, %%ymm4                  \n"
>-    "sub        %0,%1                          \n"
>+void HalfFloatRow_F16C(const uint16_t* src,
>+                       uint16_t* dst,
>+                       float scale,
>+                       int width) {
>+  asm volatile(
>+      "vbroadcastss  %3, %%ymm4                  \n"
>+      "sub        %0,%1                          \n"
> 
>-    // 16 pixel loop.
>-    LABELALIGN
>-    "1:                                        \n"
>-    "vpmovzxwd   " MEMACCESS(0) ",%%ymm2       \n"  // 16 shorts -> 16 ints
>-    "vpmovzxwd   " MEMACCESS2(0x10,0) ",%%ymm3 \n"
>-    "vcvtdq2ps   %%ymm2,%%ymm2                 \n"
>-    "vcvtdq2ps   %%ymm3,%%ymm3                 \n"
>-    "vmulps      %%ymm2,%%ymm4,%%ymm2          \n"
>-    "vmulps      %%ymm3,%%ymm4,%%ymm3          \n"
>-    "vcvtps2ph   $3, %%ymm2, %%xmm2            \n"
>-    "vcvtps2ph   $3, %%ymm3, %%xmm3            \n"
>-    MEMOPMEM(vmovdqu,xmm2,0x00,0,1,1)
>-    MEMOPMEM(vmovdqu,xmm3,0x10,0,1,1)
>-    "add         $0x20,%0                      \n"
>-    "sub         $0x10,%2                      \n"
>-    "jg          1b                            \n"
>-    "vzeroupper                                \n"
>-  : "+r"(src),   // %0
>-    "+r"(dst),   // %1
>-    "+r"(width)  // %2
>+      // 16 pixel loop.
>+      LABELALIGN
>+      "1:                                        \n"
>+      "vpmovzxwd   (%0),%%ymm2                   \n"  // 16 shorts -> 16 ints
>+      "vpmovzxwd   0x10(%0),%%ymm3               \n"
>+      "vcvtdq2ps   %%ymm2,%%ymm2                 \n"
>+      "vcvtdq2ps   %%ymm3,%%ymm3                 \n"
>+      "vmulps      %%ymm2,%%ymm4,%%ymm2          \n"
>+      "vmulps      %%ymm3,%%ymm4,%%ymm3          \n"
>+      "vcvtps2ph   $3, %%ymm2, %%xmm2            \n"
>+      "vcvtps2ph   $3, %%ymm3, %%xmm3            \n"
>+      "vmovdqu     %%xmm2,0x00(%0,%1,1)          \n"
>+      "vmovdqu     %%xmm3,0x10(%0,%1,1)          \n"
>+      "add         $0x20,%0                      \n"
>+      "sub         $0x10,%2                      \n"
>+      "jg          1b                            \n"
>+      "vzeroupper                                \n"
>+      : "+r"(src),   // %0
>+        "+r"(dst),   // %1
>+        "+r"(width)  // %2
> #if defined(__x86_64__)
>-  : "x"(scale)   // %3
>+      : "x"(scale)  // %3
> #else
>-  : "m"(scale)   // %3
>+      : "m"(scale)  // %3
> #endif
>-  : "memory", "cc",
>-    "xmm2", "xmm3", "xmm4"
>-  );
>+      : "memory", "cc", "xmm2", "xmm3", "xmm4");
> }
> #endif  // HAS_HALFFLOATROW_F16C
> 
> #ifdef HAS_HALFFLOATROW_F16C
>-void HalfFloat1Row_F16C(const uint16* src, uint16* dst, float, int width) {
>-  asm volatile (
>-    "sub        %0,%1                          \n"
>-    // 16 pixel loop.
>-    LABELALIGN
>-    "1:                                        \n"
>-    "vpmovzxwd   " MEMACCESS(0) ",%%ymm2       \n"  // 16 shorts -> 16 ints
>-    "vpmovzxwd   " MEMACCESS2(0x10,0) ",%%ymm3 \n"
>-    "vcvtdq2ps   %%ymm2,%%ymm2                 \n"
>-    "vcvtdq2ps   %%ymm3,%%ymm3                 \n"
>-    "vcvtps2ph   $3, %%ymm2, %%xmm2            \n"
>-    "vcvtps2ph   $3, %%ymm3, %%xmm3            \n"
>-    MEMOPMEM(vmovdqu,xmm2,0x00,0,1,1)
>-    MEMOPMEM(vmovdqu,xmm3,0x10,0,1,1)
>-    "add         $0x20,%0                      \n"
>-    "sub         $0x10,%2                      \n"
>-    "jg          1b                            \n"
>-    "vzeroupper                                \n"
>-  : "+r"(src),   // %0
>-    "+r"(dst),   // %1
>-    "+r"(width)  // %2
>-  :
>-  : "memory", "cc",
>-    "xmm2", "xmm3"
>-  );
>+void HalfFloat1Row_F16C(const uint16_t* src, uint16_t* dst, float, int width) {
>+  asm volatile(
>+      "sub        %0,%1                          \n"
>+      // 16 pixel loop.
>+      LABELALIGN
>+      "1:                                        \n"
>+      "vpmovzxwd   (%0),%%ymm2                   \n"  // 16 shorts -> 16 ints
>+      "vpmovzxwd   0x10(%0),%%ymm3               \n"
>+      "vcvtdq2ps   %%ymm2,%%ymm2                 \n"
>+      "vcvtdq2ps   %%ymm3,%%ymm3                 \n"
>+      "vcvtps2ph   $3, %%ymm2, %%xmm2            \n"
>+      "vcvtps2ph   $3, %%ymm3, %%xmm3            \n"
>+      "vmovdqu     %%xmm2,0x00(%0,%1,1)          \n"
>+      "vmovdqu     %%xmm3,0x10(%0,%1,1)          \n"
>+      "add         $0x20,%0                      \n"
>+      "sub         $0x10,%2                      \n"
>+      "jg          1b                            \n"
>+      "vzeroupper                                \n"
>+      : "+r"(src),   // %0
>+        "+r"(dst),   // %1
>+        "+r"(width)  // %2
>+      :
>+      : "memory", "cc", "xmm2", "xmm3");
> }
> #endif  // HAS_HALFFLOATROW_F16C
> 
> #ifdef HAS_ARGBCOLORTABLEROW_X86
> // Tranform ARGB pixels with color table.
>-void ARGBColorTableRow_X86(uint8* dst_argb,
>-                           const uint8* table_argb,
>+void ARGBColorTableRow_X86(uint8_t* dst_argb,
>+                           const uint8_t* table_argb,
>                            int width) {
>   uintptr_t pixel_temp;
>-  asm volatile (
>-    // 1 pixel loop.
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movzb     " MEMACCESS(0) ",%1             \n"
>-    "lea       " MEMLEA(0x4,0) ",%0            \n"
>-    MEMOPARG(movzb,0x00,3,1,4,1) "             \n"  // movzb (%3,%1,4),%1
>-    "mov       %b1," MEMACCESS2(-0x4,0) "      \n"
>-    "movzb     " MEMACCESS2(-0x3,0) ",%1       \n"
>-    MEMOPARG(movzb,0x01,3,1,4,1) "             \n"  // movzb 0x1(%3,%1,4),%1
>-    "mov       %b1," MEMACCESS2(-0x3,0) "      \n"
>-    "movzb     " MEMACCESS2(-0x2,0) ",%1       \n"
>-    MEMOPARG(movzb,0x02,3,1,4,1) "             \n"  // movzb 0x2(%3,%1,4),%1
>-    "mov       %b1," MEMACCESS2(-0x2,0) "      \n"
>-    "movzb     " MEMACCESS2(-0x1,0) ",%1       \n"
>-    MEMOPARG(movzb,0x03,3,1,4,1) "             \n"  // movzb 0x3(%3,%1,4),%1
>-    "mov       %b1," MEMACCESS2(-0x1,0) "      \n"
>-    "dec       %2                              \n"
>-    "jg        1b                              \n"
>-  : "+r"(dst_argb),     // %0
>-    "=&d"(pixel_temp),  // %1
>-    "+r"(width)         // %2
>-  : "r"(table_argb)     // %3
>-  : "memory", "cc");
>+  asm volatile(
>+      // 1 pixel loop.
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movzb     (%0),%1                         \n"
>+      "lea       0x4(%0),%0                      \n"
>+      "movzb     0x00(%3,%1,4),%1                \n"
>+      "mov       %b1,-0x4(%0)                    \n"
>+      "movzb     -0x3(%0),%1                     \n"
>+      "movzb     0x01(%3,%1,4),%1                \n"
>+      "mov       %b1,-0x3(%0)                    \n"
>+      "movzb     -0x2(%0),%1                     \n"
>+      "movzb     0x02(%3,%1,4),%1                \n"
>+      "mov       %b1,-0x2(%0)                    \n"
>+      "movzb     -0x1(%0),%1                     \n"
>+      "movzb     0x03(%3,%1,4),%1                \n"
>+      "mov       %b1,-0x1(%0)                    \n"
>+      "dec       %2                              \n"
>+      "jg        1b                              \n"
>+      : "+r"(dst_argb),     // %0
>+        "=&d"(pixel_temp),  // %1
>+        "+r"(width)         // %2
>+      : "r"(table_argb)     // %3
>+      : "memory", "cc");
> }
> #endif  // HAS_ARGBCOLORTABLEROW_X86
> 
> #ifdef HAS_RGBCOLORTABLEROW_X86
> // Tranform RGB pixels with color table.
>-void RGBColorTableRow_X86(uint8* dst_argb, const uint8* table_argb, int width) {
>+void RGBColorTableRow_X86(uint8_t* dst_argb,
>+                          const uint8_t* table_argb,
>+                          int width) {
>   uintptr_t pixel_temp;
>-  asm volatile (
>-    // 1 pixel loop.
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movzb     " MEMACCESS(0) ",%1             \n"
>-    "lea       " MEMLEA(0x4,0) ",%0            \n"
>-    MEMOPARG(movzb,0x00,3,1,4,1) "             \n"  // movzb (%3,%1,4),%1
>-    "mov       %b1," MEMACCESS2(-0x4,0) "      \n"
>-    "movzb     " MEMACCESS2(-0x3,0) ",%1       \n"
>-    MEMOPARG(movzb,0x01,3,1,4,1) "             \n"  // movzb 0x1(%3,%1,4),%1
>-    "mov       %b1," MEMACCESS2(-0x3,0) "      \n"
>-    "movzb     " MEMACCESS2(-0x2,0) ",%1       \n"
>-    MEMOPARG(movzb,0x02,3,1,4,1) "             \n"  // movzb 0x2(%3,%1,4),%1
>-    "mov       %b1," MEMACCESS2(-0x2,0) "      \n"
>-    "dec       %2                              \n"
>-    "jg        1b                              \n"
>-  : "+r"(dst_argb),     // %0
>-    "=&d"(pixel_temp),  // %1
>-    "+r"(width)         // %2
>-  : "r"(table_argb)     // %3
>-  : "memory", "cc");
>+  asm volatile(
>+      // 1 pixel loop.
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movzb     (%0),%1                         \n"
>+      "lea       0x4(%0),%0                      \n"
>+      "movzb     0x00(%3,%1,4),%1                \n"
>+      "mov       %b1,-0x4(%0)                    \n"
>+      "movzb     -0x3(%0),%1                     \n"
>+      "movzb     0x01(%3,%1,4),%1                \n"
>+      "mov       %b1,-0x3(%0)                    \n"
>+      "movzb     -0x2(%0),%1                     \n"
>+      "movzb     0x02(%3,%1,4),%1                \n"
>+      "mov       %b1,-0x2(%0)                    \n"
>+      "dec       %2                              \n"
>+      "jg        1b                              \n"
>+      : "+r"(dst_argb),     // %0
>+        "=&d"(pixel_temp),  // %1
>+        "+r"(width)         // %2
>+      : "r"(table_argb)     // %3
>+      : "memory", "cc");
> }
> #endif  // HAS_RGBCOLORTABLEROW_X86
> 
> #ifdef HAS_ARGBLUMACOLORTABLEROW_SSSE3
> // Tranform RGB pixels with luma table.
>-void ARGBLumaColorTableRow_SSSE3(const uint8* src_argb,
>-                                 uint8* dst_argb,
>+void ARGBLumaColorTableRow_SSSE3(const uint8_t* src_argb,
>+                                 uint8_t* dst_argb,
>                                  int width,
>-                                 const uint8* luma,
>-                                 uint32 lumacoeff) {
>+                                 const uint8_t* luma,
>+                                 uint32_t lumacoeff) {
>   uintptr_t pixel_temp;
>   uintptr_t table_temp;
>-  asm volatile (
>-    "movd      %6,%%xmm3                       \n"
>-    "pshufd    $0x0,%%xmm3,%%xmm3              \n"
>-    "pcmpeqb   %%xmm4,%%xmm4                   \n"
>-    "psllw     $0x8,%%xmm4                     \n"
>-    "pxor      %%xmm5,%%xmm5                   \n"
>+  asm volatile(
>+      "movd      %6,%%xmm3                       \n"
>+      "pshufd    $0x0,%%xmm3,%%xmm3              \n"
>+      "pcmpeqb   %%xmm4,%%xmm4                   \n"
>+      "psllw     $0x8,%%xmm4                     \n"
>+      "pxor      %%xmm5,%%xmm5                   \n"
> 
>-    // 4 pixel loop.
>-    LABELALIGN
>-    "1:                                        \n"
>-    "movdqu    " MEMACCESS(2) ",%%xmm0         \n"
>-    "pmaddubsw %%xmm3,%%xmm0                   \n"
>-    "phaddw    %%xmm0,%%xmm0                   \n"
>-    "pand      %%xmm4,%%xmm0                   \n"
>-    "punpcklwd %%xmm5,%%xmm0                   \n"
>-    "movd      %%xmm0,%k1                      \n"  // 32 bit offset
>-    "add       %5,%1                           \n"
>-    "pshufd    $0x39,%%xmm0,%%xmm0             \n"
>-
>-    "movzb     " MEMACCESS(2) ",%0             \n"
>-    MEMOPARG(movzb,0x00,1,0,1,0) "             \n"  // movzb     (%1,%0,1),%0
>-    "mov       %b0," MEMACCESS(3) "            \n"
>-    "movzb     " MEMACCESS2(0x1,2) ",%0        \n"
>-    MEMOPARG(movzb,0x00,1,0,1,0) "             \n"  // movzb     (%1,%0,1),%0
>-    "mov       %b0," MEMACCESS2(0x1,3) "       \n"
>-    "movzb     " MEMACCESS2(0x2,2) ",%0        \n"
>-    MEMOPARG(movzb,0x00,1,0,1,0) "             \n"  // movzb     (%1,%0,1),%0
>-    "mov       %b0," MEMACCESS2(0x2,3) "       \n"
>-    "movzb     " MEMACCESS2(0x3,2) ",%0        \n"
>-    "mov       %b0," MEMACCESS2(0x3,3) "       \n"
>-
>-    "movd      %%xmm0,%k1                      \n"  // 32 bit offset
>-    "add       %5,%1                           \n"
>-    "pshufd    $0x39,%%xmm0,%%xmm0             \n"
>-
>-    "movzb     " MEMACCESS2(0x4,2) ",%0        \n"
>-    MEMOPARG(movzb,0x00,1,0,1,0) "             \n"  // movzb     (%1,%0,1),%0
>-    "mov       %b0," MEMACCESS2(0x4,3) "       \n"
>-    "movzb     " MEMACCESS2(0x5,2) ",%0        \n"
>-    MEMOPARG(movzb,0x00,1,0,1,0) "             \n"  // movzb     (%1,%0,1),%0
>-    "mov       %b0," MEMACCESS2(0x5,3) "       \n"
>-    "movzb     " MEMACCESS2(0x6,2) ",%0        \n"
>-    MEMOPARG(movzb,0x00,1,0,1,0) "             \n"  // movzb     (%1,%0,1),%0
>-    "mov       %b0," MEMACCESS2(0x6,3) "       \n"
>-    "movzb     " MEMACCESS2(0x7,2) ",%0        \n"
>-    "mov       %b0," MEMACCESS2(0x7,3) "       \n"
>-
>-    "movd      %%xmm0,%k1                      \n"  // 32 bit offset
>-    "add       %5,%1                           \n"
>-    "pshufd    $0x39,%%xmm0,%%xmm0             \n"
>-
>-    "movzb     " MEMACCESS2(0x8,2) ",%0        \n"
>-    MEMOPARG(movzb,0x00,1,0,1,0) "             \n"  // movzb     (%1,%0,1),%0
>-    "mov       %b0," MEMACCESS2(0x8,3) "       \n"
>-    "movzb     " MEMACCESS2(0x9,2) ",%0        \n"
>-    MEMOPARG(movzb,0x00,1,0,1,0) "             \n"  // movzb     (%1,%0,1),%0
>-    "mov       %b0," MEMACCESS2(0x9,3) "       \n"
>-    "movzb     " MEMACCESS2(0xa,2) ",%0        \n"
>-    MEMOPARG(movzb,0x00,1,0,1,0) "             \n"  // movzb     (%1,%0,1),%0
>-    "mov       %b0," MEMACCESS2(0xa,3) "       \n"
>-    "movzb     " MEMACCESS2(0xb,2) ",%0        \n"
>-    "mov       %b0," MEMACCESS2(0xb,3) "       \n"
>-
>-    "movd      %%xmm0,%k1                      \n"  // 32 bit offset
>-    "add       %5,%1                           \n"
>-
>-    "movzb     " MEMACCESS2(0xc,2) ",%0        \n"
>-    MEMOPARG(movzb,0x00,1,0,1,0) "             \n"  // movzb     (%1,%0,1),%0
>-    "mov       %b0," MEMACCESS2(0xc,3) "       \n"
>-    "movzb     " MEMACCESS2(0xd,2) ",%0        \n"
>-    MEMOPARG(movzb,0x00,1,0,1,0) "             \n"  // movzb     (%1,%0,1),%0
>-    "mov       %b0," MEMACCESS2(0xd,3) "       \n"
>-    "movzb     " MEMACCESS2(0xe,2) ",%0        \n"
>-    MEMOPARG(movzb,0x00,1,0,1,0) "             \n"  // movzb     (%1,%0,1),%0
>-    "mov       %b0," MEMACCESS2(0xe,3) "       \n"
>-    "movzb     " MEMACCESS2(0xf,2) ",%0        \n"
>-    "mov       %b0," MEMACCESS2(0xf,3) "       \n"
>-    "lea       " MEMLEA(0x10,2) ",%2           \n"
>-    "lea       " MEMLEA(0x10,3) ",%3           \n"
>-    "sub       $0x4,%4                         \n"
>-    "jg        1b                              \n"
>-  : "=&d"(pixel_temp),  // %0
>-    "=&a"(table_temp),  // %1
>-    "+r"(src_argb),     // %2
>-    "+r"(dst_argb),     // %3
>-    "+rm"(width)        // %4
>-  : "r"(luma),          // %5
>-    "rm"(lumacoeff)     // %6
>-  : "memory", "cc", "xmm0", "xmm3", "xmm4", "xmm5"
>-  );
>+      // 4 pixel loop.
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%2),%%xmm0                     \n"
>+      "pmaddubsw %%xmm3,%%xmm0                   \n"
>+      "phaddw    %%xmm0,%%xmm0                   \n"
>+      "pand      %%xmm4,%%xmm0                   \n"
>+      "punpcklwd %%xmm5,%%xmm0                   \n"
>+      "movd      %%xmm0,%k1                      \n"  // 32 bit offset
>+      "add       %5,%1                           \n"
>+      "pshufd    $0x39,%%xmm0,%%xmm0             \n"
>+
>+      "movzb     (%2),%0                         \n"
>+      "movzb     0x00(%1,%0,1),%0                \n"
>+      "mov       %b0,(%3)                        \n"
>+      "movzb     0x1(%2),%0                      \n"
>+      "movzb     0x00(%1,%0,1),%0                \n"
>+      "mov       %b0,0x1(%3)                     \n"
>+      "movzb     0x2(%2),%0                      \n"
>+      "movzb     0x00(%1,%0,1),%0                \n"
>+      "mov       %b0,0x2(%3)                     \n"
>+      "movzb     0x3(%2),%0                      \n"
>+      "mov       %b0,0x3(%3)                     \n"
>+
>+      "movd      %%xmm0,%k1                      \n"  // 32 bit offset
>+      "add       %5,%1                           \n"
>+      "pshufd    $0x39,%%xmm0,%%xmm0             \n"
>+
>+      "movzb     0x4(%2),%0                      \n"
>+      "movzb     0x00(%1,%0,1),%0                \n"
>+      "mov       %b0,0x4(%3)                     \n"
>+      "movzb     0x5(%2),%0                      \n"
>+      "movzb     0x00(%1,%0,1),%0                \n"
>+      "mov       %b0,0x5(%3)                     \n"
>+      "movzb     0x6(%2),%0                      \n"
>+      "movzb     0x00(%1,%0,1),%0                \n"
>+      "mov       %b0,0x6(%3)                     \n"
>+      "movzb     0x7(%2),%0                      \n"
>+      "mov       %b0,0x7(%3)                     \n"
>+
>+      "movd      %%xmm0,%k1                      \n"  // 32 bit offset
>+      "add       %5,%1                           \n"
>+      "pshufd    $0x39,%%xmm0,%%xmm0             \n"
>+
>+      "movzb     0x8(%2),%0                      \n"
>+      "movzb     0x00(%1,%0,1),%0                \n"
>+      "mov       %b0,0x8(%3)                     \n"
>+      "movzb     0x9(%2),%0                      \n"
>+      "movzb     0x00(%1,%0,1),%0                \n"
>+      "mov       %b0,0x9(%3)                     \n"
>+      "movzb     0xa(%2),%0                      \n"
>+      "movzb     0x00(%1,%0,1),%0                \n"
>+      "mov       %b0,0xa(%3)                     \n"
>+      "movzb     0xb(%2),%0                      \n"
>+      "mov       %b0,0xb(%3)                     \n"
>+
>+      "movd      %%xmm0,%k1                      \n"  // 32 bit offset
>+      "add       %5,%1                           \n"
>+
>+      "movzb     0xc(%2),%0                      \n"
>+      "movzb     0x00(%1,%0,1),%0                \n"
>+      "mov       %b0,0xc(%3)                     \n"
>+      "movzb     0xd(%2),%0                      \n"
>+      "movzb     0x00(%1,%0,1),%0                \n"
>+      "mov       %b0,0xd(%3)                     \n"
>+      "movzb     0xe(%2),%0                      \n"
>+      "movzb     0x00(%1,%0,1),%0                \n"
>+      "mov       %b0,0xe(%3)                     \n"
>+      "movzb     0xf(%2),%0                      \n"
>+      "mov       %b0,0xf(%3)                     \n"
>+      "lea       0x10(%2),%2                     \n"
>+      "lea       0x10(%3),%3                     \n"
>+      "sub       $0x4,%4                         \n"
>+      "jg        1b                              \n"
>+      : "=&d"(pixel_temp),  // %0
>+        "=&a"(table_temp),  // %1
>+        "+r"(src_argb),     // %2
>+        "+r"(dst_argb),     // %3
>+        "+rm"(width)        // %4
>+      : "r"(luma),          // %5
>+        "rm"(lumacoeff)     // %6
>+      : "memory", "cc", "xmm0", "xmm3", "xmm4", "xmm5");
> }
> #endif  // HAS_ARGBLUMACOLORTABLEROW_SSSE3
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/row_msa.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/row_msa.cc
>index 5cc23450a52..66666cefcd9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/row_msa.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/row_msa.cc
>@@ -37,17 +37,17 @@ extern "C" {
>   }
> 
> // Load YUV 422 pixel data
>-#define READYUV422(psrc_y, psrc_u, psrc_v, out_y, out_u, out_v)  \
>-  {                                                              \
>-    uint64 y_m;                                                  \
>-    uint32 u_m, v_m;                                             \
>-    v4i32 zero_m = {0};                                          \
>-    y_m = LD(psrc_y);                                            \
>-    u_m = LW(psrc_u);                                            \
>-    v_m = LW(psrc_v);                                            \
>-    out_y = (v16u8)__msa_insert_d((v2i64)zero_m, 0, (int64)y_m); \
>-    out_u = (v16u8)__msa_insert_w(zero_m, 0, (int32)u_m);        \
>-    out_v = (v16u8)__msa_insert_w(zero_m, 0, (int32)v_m);        \
>+#define READYUV422(psrc_y, psrc_u, psrc_v, out_y, out_u, out_v)    \
>+  {                                                                \
>+    uint64_t y_m;                                                  \
>+    uint32_t u_m, v_m;                                             \
>+    v4i32 zero_m = {0};                                            \
>+    y_m = LD(psrc_y);                                              \
>+    u_m = LW(psrc_u);                                              \
>+    v_m = LW(psrc_v);                                              \
>+    out_y = (v16u8)__msa_insert_d((v2i64)zero_m, 0, (int64_t)y_m); \
>+    out_u = (v16u8)__msa_insert_w(zero_m, 0, (int32_t)u_m);        \
>+    out_v = (v16u8)__msa_insert_w(zero_m, 0, (int32_t)v_m);        \
>   }
> 
> // Clip input vector elements between 0 to 255
>@@ -275,17 +275,17 @@ extern "C" {
> // Load I444 pixel data
> #define READI444(psrc_y, psrc_u, psrc_v, out_y, out_u, out_v) \
>   {                                                           \
>-    uint64 y_m, u_m, v_m;                                     \
>+    uint64_t y_m, u_m, v_m;                                   \
>     v2i64 zero_m = {0};                                       \
>     y_m = LD(psrc_y);                                         \
>     u_m = LD(psrc_u);                                         \
>     v_m = LD(psrc_v);                                         \
>-    out_y = (v16u8)__msa_insert_d(zero_m, 0, (int64)y_m);     \
>-    out_u = (v16u8)__msa_insert_d(zero_m, 0, (int64)u_m);     \
>-    out_v = (v16u8)__msa_insert_d(zero_m, 0, (int64)v_m);     \
>+    out_y = (v16u8)__msa_insert_d(zero_m, 0, (int64_t)y_m);   \
>+    out_u = (v16u8)__msa_insert_d(zero_m, 0, (int64_t)u_m);   \
>+    out_v = (v16u8)__msa_insert_d(zero_m, 0, (int64_t)v_m);   \
>   }
> 
>-void MirrorRow_MSA(const uint8* src, uint8* dst, int width) {
>+void MirrorRow_MSA(const uint8_t* src, uint8_t* dst, int width) {
>   int x;
>   v16u8 src0, src1, src2, src3;
>   v16u8 dst0, dst1, dst2, dst3;
>@@ -302,7 +302,7 @@ void MirrorRow_MSA(const uint8* src, uint8* dst, int width) {
>   }
> }
> 
>-void ARGBMirrorRow_MSA(const uint8* src, uint8* dst, int width) {
>+void ARGBMirrorRow_MSA(const uint8_t* src, uint8_t* dst, int width) {
>   int x;
>   v16u8 src0, src1, src2, src3;
>   v16u8 dst0, dst1, dst2, dst3;
>@@ -319,10 +319,10 @@ void ARGBMirrorRow_MSA(const uint8* src, uint8* dst, int width) {
>   }
> }
> 
>-void I422ToYUY2Row_MSA(const uint8* src_y,
>-                       const uint8* src_u,
>-                       const uint8* src_v,
>-                       uint8* dst_yuy2,
>+void I422ToYUY2Row_MSA(const uint8_t* src_y,
>+                       const uint8_t* src_u,
>+                       const uint8_t* src_v,
>+                       uint8_t* dst_yuy2,
>                        int width) {
>   int x;
>   v16u8 src_u0, src_v0, src_y0, src_y1, vec_uv0, vec_uv1;
>@@ -343,10 +343,10 @@ void I422ToYUY2Row_MSA(const uint8* src_y,
>   }
> }
> 
>-void I422ToUYVYRow_MSA(const uint8* src_y,
>-                       const uint8* src_u,
>-                       const uint8* src_v,
>-                       uint8* dst_uyvy,
>+void I422ToUYVYRow_MSA(const uint8_t* src_y,
>+                       const uint8_t* src_u,
>+                       const uint8_t* src_v,
>+                       uint8_t* dst_uyvy,
>                        int width) {
>   int x;
>   v16u8 src_u0, src_v0, src_y0, src_y1, vec_uv0, vec_uv1;
>@@ -367,10 +367,10 @@ void I422ToUYVYRow_MSA(const uint8* src_y,
>   }
> }
> 
>-void I422ToARGBRow_MSA(const uint8* src_y,
>-                       const uint8* src_u,
>-                       const uint8* src_v,
>-                       uint8* rgb_buf,
>+void I422ToARGBRow_MSA(const uint8_t* src_y,
>+                       const uint8_t* src_u,
>+                       const uint8_t* src_v,
>+                       uint8_t* dst_argb,
>                        const struct YuvConstants* yuvconstants,
>                        int width) {
>   int x;
>@@ -390,18 +390,18 @@ void I422ToARGBRow_MSA(const uint8* src_y,
>     src1 = (v16u8)__msa_ilvr_b((v16i8)src2, (v16i8)src1);
>     YUVTORGB(src0, src1, vec_ubvr, vec_ugvg, vec_bb, vec_bg, vec_br, vec_yg,
>              vec0, vec1, vec2);
>-    STOREARGB(vec0, vec1, vec2, alpha, rgb_buf);
>+    STOREARGB(vec0, vec1, vec2, alpha, dst_argb);
>     src_y += 8;
>     src_u += 4;
>     src_v += 4;
>-    rgb_buf += 32;
>+    dst_argb += 32;
>   }
> }
> 
>-void I422ToRGBARow_MSA(const uint8* src_y,
>-                       const uint8* src_u,
>-                       const uint8* src_v,
>-                       uint8* rgb_buf,
>+void I422ToRGBARow_MSA(const uint8_t* src_y,
>+                       const uint8_t* src_u,
>+                       const uint8_t* src_v,
>+                       uint8_t* dst_argb,
>                        const struct YuvConstants* yuvconstants,
>                        int width) {
>   int x;
>@@ -421,23 +421,23 @@ void I422ToRGBARow_MSA(const uint8* src_y,
>     src1 = (v16u8)__msa_ilvr_b((v16i8)src2, (v16i8)src1);
>     YUVTORGB(src0, src1, vec_ubvr, vec_ugvg, vec_bb, vec_bg, vec_br, vec_yg,
>              vec0, vec1, vec2);
>-    STOREARGB(alpha, vec0, vec1, vec2, rgb_buf);
>+    STOREARGB(alpha, vec0, vec1, vec2, dst_argb);
>     src_y += 8;
>     src_u += 4;
>     src_v += 4;
>-    rgb_buf += 32;
>+    dst_argb += 32;
>   }
> }
> 
>-void I422AlphaToARGBRow_MSA(const uint8* src_y,
>-                            const uint8* src_u,
>-                            const uint8* src_v,
>-                            const uint8* src_a,
>-                            uint8* rgb_buf,
>+void I422AlphaToARGBRow_MSA(const uint8_t* src_y,
>+                            const uint8_t* src_u,
>+                            const uint8_t* src_v,
>+                            const uint8_t* src_a,
>+                            uint8_t* dst_argb,
>                             const struct YuvConstants* yuvconstants,
>                             int width) {
>   int x;
>-  int64 data_a;
>+  int64_t data_a;
>   v16u8 src0, src1, src2, src3;
>   v8i16 vec0, vec1, vec2;
>   v4i32 vec_ub, vec_vr, vec_ug, vec_vg, vec_bb, vec_bg, vec_br, vec_yg;
>@@ -457,23 +457,23 @@ void I422AlphaToARGBRow_MSA(const uint8* src_y,
>     YUVTORGB(src0, src1, vec_ubvr, vec_ugvg, vec_bb, vec_bg, vec_br, vec_yg,
>              vec0, vec1, vec2);
>     src3 = (v16u8)__msa_ilvr_b((v16i8)src3, (v16i8)src3);
>-    STOREARGB(vec0, vec1, vec2, src3, rgb_buf);
>+    STOREARGB(vec0, vec1, vec2, src3, dst_argb);
>     src_y += 8;
>     src_u += 4;
>     src_v += 4;
>     src_a += 8;
>-    rgb_buf += 32;
>+    dst_argb += 32;
>   }
> }
> 
>-void I422ToRGB24Row_MSA(const uint8* src_y,
>-                        const uint8* src_u,
>-                        const uint8* src_v,
>-                        uint8* rgb_buf,
>+void I422ToRGB24Row_MSA(const uint8_t* src_y,
>+                        const uint8_t* src_u,
>+                        const uint8_t* src_v,
>+                        uint8_t* dst_argb,
>                         const struct YuvConstants* yuvconstants,
>-                        int32 width) {
>+                        int32_t width) {
>   int x;
>-  int64 data_u, data_v;
>+  int64_t data_u, data_v;
>   v16u8 src0, src1, src2, src3, src4, dst0, dst1, dst2;
>   v8i16 vec0, vec1, vec2, vec3, vec4, vec5;
>   v4i32 vec_ub, vec_vr, vec_ug, vec_vg, vec_bb, vec_bg, vec_br, vec_yg;
>@@ -510,20 +510,20 @@ void I422ToRGB24Row_MSA(const uint8* src_y,
>     dst0 = (v16u8)__msa_vshf_b(shuffler0, (v16i8)reg3, (v16i8)reg0);
>     dst1 = (v16u8)__msa_vshf_b(shuffler1, (v16i8)reg3, (v16i8)reg1);
>     dst2 = (v16u8)__msa_vshf_b(shuffler2, (v16i8)reg3, (v16i8)reg2);
>-    ST_UB2(dst0, dst1, rgb_buf, 16);
>-    ST_UB(dst2, (rgb_buf + 32));
>+    ST_UB2(dst0, dst1, dst_argb, 16);
>+    ST_UB(dst2, (dst_argb + 32));
>     src_y += 16;
>     src_u += 8;
>     src_v += 8;
>-    rgb_buf += 48;
>+    dst_argb += 48;
>   }
> }
> 
> // TODO(fbarchard): Consider AND instead of shift to isolate 5 upper bits of R.
>-void I422ToRGB565Row_MSA(const uint8* src_y,
>-                         const uint8* src_u,
>-                         const uint8* src_v,
>-                         uint8* dst_rgb565,
>+void I422ToRGB565Row_MSA(const uint8_t* src_y,
>+                         const uint8_t* src_u,
>+                         const uint8_t* src_v,
>+                         uint8_t* dst_rgb565,
>                          const struct YuvConstants* yuvconstants,
>                          int width) {
>   int x;
>@@ -558,10 +558,10 @@ void I422ToRGB565Row_MSA(const uint8* src_y,
> }
> 
> // TODO(fbarchard): Consider AND instead of shift to isolate 4 upper bits of G.
>-void I422ToARGB4444Row_MSA(const uint8* src_y,
>-                           const uint8* src_u,
>-                           const uint8* src_v,
>-                           uint8* dst_argb4444,
>+void I422ToARGB4444Row_MSA(const uint8_t* src_y,
>+                           const uint8_t* src_u,
>+                           const uint8_t* src_v,
>+                           uint8_t* dst_argb4444,
>                            const struct YuvConstants* yuvconstants,
>                            int width) {
>   int x;
>@@ -598,10 +598,10 @@ void I422ToARGB4444Row_MSA(const uint8* src_y,
>   }
> }
> 
>-void I422ToARGB1555Row_MSA(const uint8* src_y,
>-                           const uint8* src_u,
>-                           const uint8* src_v,
>-                           uint8* dst_argb1555,
>+void I422ToARGB1555Row_MSA(const uint8_t* src_y,
>+                           const uint8_t* src_u,
>+                           const uint8_t* src_v,
>+                           uint8_t* dst_argb1555,
>                            const struct YuvConstants* yuvconstants,
>                            int width) {
>   int x;
>@@ -638,7 +638,7 @@ void I422ToARGB1555Row_MSA(const uint8* src_y,
>   }
> }
> 
>-void YUY2ToYRow_MSA(const uint8* src_yuy2, uint8* dst_y, int width) {
>+void YUY2ToYRow_MSA(const uint8_t* src_yuy2, uint8_t* dst_y, int width) {
>   int x;
>   v16u8 src0, src1, src2, src3, dst0, dst1;
> 
>@@ -652,12 +652,12 @@ void YUY2ToYRow_MSA(const uint8* src_yuy2, uint8* dst_y, int width) {
>   }
> }
> 
>-void YUY2ToUVRow_MSA(const uint8* src_yuy2,
>+void YUY2ToUVRow_MSA(const uint8_t* src_yuy2,
>                      int src_stride_yuy2,
>-                     uint8* dst_u,
>-                     uint8* dst_v,
>+                     uint8_t* dst_u,
>+                     uint8_t* dst_v,
>                      int width) {
>-  const uint8* src_yuy2_next = src_yuy2 + src_stride_yuy2;
>+  const uint8_t* src_yuy2_next = src_yuy2 + src_stride_yuy2;
>   int x;
>   v16u8 src0, src1, src2, src3, src4, src5, src6, src7;
>   v16u8 vec0, vec1, dst0, dst1;
>@@ -682,9 +682,9 @@ void YUY2ToUVRow_MSA(const uint8* src_yuy2,
>   }
> }
> 
>-void YUY2ToUV422Row_MSA(const uint8* src_yuy2,
>-                        uint8* dst_u,
>-                        uint8* dst_v,
>+void YUY2ToUV422Row_MSA(const uint8_t* src_yuy2,
>+                        uint8_t* dst_u,
>+                        uint8_t* dst_v,
>                         int width) {
>   int x;
>   v16u8 src0, src1, src2, src3, dst0, dst1;
>@@ -703,7 +703,7 @@ void YUY2ToUV422Row_MSA(const uint8* src_yuy2,
>   }
> }
> 
>-void UYVYToYRow_MSA(const uint8* src_uyvy, uint8* dst_y, int width) {
>+void UYVYToYRow_MSA(const uint8_t* src_uyvy, uint8_t* dst_y, int width) {
>   int x;
>   v16u8 src0, src1, src2, src3, dst0, dst1;
> 
>@@ -717,12 +717,12 @@ void UYVYToYRow_MSA(const uint8* src_uyvy, uint8* dst_y, int width) {
>   }
> }
> 
>-void UYVYToUVRow_MSA(const uint8* src_uyvy,
>+void UYVYToUVRow_MSA(const uint8_t* src_uyvy,
>                      int src_stride_uyvy,
>-                     uint8* dst_u,
>-                     uint8* dst_v,
>+                     uint8_t* dst_u,
>+                     uint8_t* dst_v,
>                      int width) {
>-  const uint8* src_uyvy_next = src_uyvy + src_stride_uyvy;
>+  const uint8_t* src_uyvy_next = src_uyvy + src_stride_uyvy;
>   int x;
>   v16u8 src0, src1, src2, src3, src4, src5, src6, src7;
>   v16u8 vec0, vec1, dst0, dst1;
>@@ -747,9 +747,9 @@ void UYVYToUVRow_MSA(const uint8* src_uyvy,
>   }
> }
> 
>-void UYVYToUV422Row_MSA(const uint8* src_uyvy,
>-                        uint8* dst_u,
>-                        uint8* dst_v,
>+void UYVYToUV422Row_MSA(const uint8_t* src_uyvy,
>+                        uint8_t* dst_u,
>+                        uint8_t* dst_v,
>                         int width) {
>   int x;
>   v16u8 src0, src1, src2, src3, dst0, dst1;
>@@ -768,7 +768,7 @@ void UYVYToUV422Row_MSA(const uint8* src_uyvy,
>   }
> }
> 
>-void ARGBToYRow_MSA(const uint8* src_argb0, uint8* dst_y, int width) {
>+void ARGBToYRow_MSA(const uint8_t* src_argb0, uint8_t* dst_y, int width) {
>   int x;
>   v16u8 src0, src1, src2, src3, vec0, vec1, vec2, vec3, dst0;
>   v8u16 reg0, reg1, reg2, reg3, reg4, reg5;
>@@ -814,13 +814,13 @@ void ARGBToYRow_MSA(const uint8* src_argb0, uint8* dst_y, int width) {
>   }
> }
> 
>-void ARGBToUVRow_MSA(const uint8* src_argb0,
>+void ARGBToUVRow_MSA(const uint8_t* src_argb0,
>                      int src_stride_argb,
>-                     uint8* dst_u,
>-                     uint8* dst_v,
>+                     uint8_t* dst_u,
>+                     uint8_t* dst_v,
>                      int width) {
>   int x;
>-  const uint8* src_argb0_next = src_argb0 + src_stride_argb;
>+  const uint8_t* src_argb0_next = src_argb0 + src_stride_argb;
>   v16u8 src0, src1, src2, src3, src4, src5, src6, src7;
>   v16u8 vec0, vec1, vec2, vec3, vec4, vec5, vec6, vec7, vec8, vec9;
>   v8u16 reg0, reg1, reg2, reg3, reg4, reg5, reg6, reg7, reg8, reg9;
>@@ -932,7 +932,7 @@ void ARGBToUVRow_MSA(const uint8* src_argb0,
>   }
> }
> 
>-void ARGBToRGB24Row_MSA(const uint8* src_argb, uint8* dst_rgb, int width) {
>+void ARGBToRGB24Row_MSA(const uint8_t* src_argb, uint8_t* dst_rgb, int width) {
>   int x;
>   v16u8 src0, src1, src2, src3, dst0, dst1, dst2;
>   v16i8 shuffler0 = {0, 1, 2, 4, 5, 6, 8, 9, 10, 12, 13, 14, 16, 17, 18, 20};
>@@ -956,7 +956,7 @@ void ARGBToRGB24Row_MSA(const uint8* src_argb, uint8* dst_rgb, int width) {
>   }
> }
> 
>-void ARGBToRAWRow_MSA(const uint8* src_argb, uint8* dst_rgb, int width) {
>+void ARGBToRAWRow_MSA(const uint8_t* src_argb, uint8_t* dst_rgb, int width) {
>   int x;
>   v16u8 src0, src1, src2, src3, dst0, dst1, dst2;
>   v16i8 shuffler0 = {2, 1, 0, 6, 5, 4, 10, 9, 8, 14, 13, 12, 18, 17, 16, 22};
>@@ -980,7 +980,7 @@ void ARGBToRAWRow_MSA(const uint8* src_argb, uint8* dst_rgb, int width) {
>   }
> }
> 
>-void ARGBToRGB565Row_MSA(const uint8* src_argb, uint8* dst_rgb, int width) {
>+void ARGBToRGB565Row_MSA(const uint8_t* src_argb, uint8_t* dst_rgb, int width) {
>   int x;
>   v16u8 src0, src1, dst0;
>   v16u8 vec0, vec1, vec2, vec3, vec4, vec5, vec6, vec7;
>@@ -1014,7 +1014,9 @@ void ARGBToRGB565Row_MSA(const uint8* src_argb, uint8* dst_rgb, int width) {
>   }
> }
> 
>-void ARGBToARGB1555Row_MSA(const uint8* src_argb, uint8* dst_rgb, int width) {
>+void ARGBToARGB1555Row_MSA(const uint8_t* src_argb,
>+                           uint8_t* dst_rgb,
>+                           int width) {
>   int x;
>   v16u8 src0, src1, dst0;
>   v16u8 vec0, vec1, vec2, vec3, vec4, vec5, vec6, vec7, vec8, vec9;
>@@ -1054,7 +1056,9 @@ void ARGBToARGB1555Row_MSA(const uint8* src_argb, uint8* dst_rgb, int width) {
>   }
> }
> 
>-void ARGBToARGB4444Row_MSA(const uint8* src_argb, uint8* dst_rgb, int width) {
>+void ARGBToARGB4444Row_MSA(const uint8_t* src_argb,
>+                           uint8_t* dst_rgb,
>+                           int width) {
>   int x;
>   v16u8 src0, src1;
>   v16u8 vec0, vec1;
>@@ -1077,11 +1081,11 @@ void ARGBToARGB4444Row_MSA(const uint8* src_argb, uint8* dst_rgb, int width) {
>   }
> }
> 
>-void ARGBToUV444Row_MSA(const uint8* src_argb,
>-                        uint8* dst_u,
>-                        uint8* dst_v,
>-                        int32 width) {
>-  int32 x;
>+void ARGBToUV444Row_MSA(const uint8_t* src_argb,
>+                        uint8_t* dst_u,
>+                        uint8_t* dst_v,
>+                        int32_t width) {
>+  int32_t x;
>   v16u8 src0, src1, src2, src3, reg0, reg1, reg2, reg3, dst0, dst1;
>   v8u16 vec0, vec1, vec2, vec3, vec4, vec5, vec6, vec7;
>   v8u16 vec8, vec9, vec10, vec11;
>@@ -1149,9 +1153,9 @@ void ARGBToUV444Row_MSA(const uint8* src_argb,
>   }
> }
> 
>-void ARGBMultiplyRow_MSA(const uint8* src_argb0,
>-                         const uint8* src_argb1,
>-                         uint8* dst_argb,
>+void ARGBMultiplyRow_MSA(const uint8_t* src_argb0,
>+                         const uint8_t* src_argb1,
>+                         uint8_t* dst_argb,
>                          int width) {
>   int x;
>   v16u8 src0, src1, dst0;
>@@ -1188,9 +1192,9 @@ void ARGBMultiplyRow_MSA(const uint8* src_argb0,
>   }
> }
> 
>-void ARGBAddRow_MSA(const uint8* src_argb0,
>-                    const uint8* src_argb1,
>-                    uint8* dst_argb,
>+void ARGBAddRow_MSA(const uint8_t* src_argb0,
>+                    const uint8_t* src_argb1,
>+                    uint8_t* dst_argb,
>                     int width) {
>   int x;
>   v16u8 src0, src1, src2, src3, dst0, dst1;
>@@ -1209,9 +1213,9 @@ void ARGBAddRow_MSA(const uint8* src_argb0,
>   }
> }
> 
>-void ARGBSubtractRow_MSA(const uint8* src_argb0,
>-                         const uint8* src_argb1,
>-                         uint8* dst_argb,
>+void ARGBSubtractRow_MSA(const uint8_t* src_argb0,
>+                         const uint8_t* src_argb1,
>+                         uint8_t* dst_argb,
>                          int width) {
>   int x;
>   v16u8 src0, src1, src2, src3, dst0, dst1;
>@@ -1230,7 +1234,9 @@ void ARGBSubtractRow_MSA(const uint8* src_argb0,
>   }
> }
> 
>-void ARGBAttenuateRow_MSA(const uint8* src_argb, uint8* dst_argb, int width) {
>+void ARGBAttenuateRow_MSA(const uint8_t* src_argb,
>+                          uint8_t* dst_argb,
>+                          int width) {
>   int x;
>   v16u8 src0, src1, dst0, dst1;
>   v8u16 vec0, vec1, vec2, vec3, vec4, vec5, vec6, vec7, vec8, vec9;
>@@ -1295,9 +1301,9 @@ void ARGBAttenuateRow_MSA(const uint8* src_argb, uint8* dst_argb, int width) {
>   }
> }
> 
>-void ARGBToRGB565DitherRow_MSA(const uint8* src_argb,
>-                               uint8* dst_rgb,
>-                               uint32 dither4,
>+void ARGBToRGB565DitherRow_MSA(const uint8_t* src_argb,
>+                               uint8_t* dst_rgb,
>+                               uint32_t dither4,
>                                int width) {
>   int x;
>   v16u8 src0, src1, dst0, vec0, vec1;
>@@ -1339,15 +1345,15 @@ void ARGBToRGB565DitherRow_MSA(const uint8* src_argb,
>   }
> }
> 
>-void ARGBShuffleRow_MSA(const uint8* src_argb,
>-                        uint8* dst_argb,
>-                        const uint8* shuffler,
>+void ARGBShuffleRow_MSA(const uint8_t* src_argb,
>+                        uint8_t* dst_argb,
>+                        const uint8_t* shuffler,
>                         int width) {
>   int x;
>   v16u8 src0, src1, dst0, dst1;
>   v16i8 vec0;
>   v16i8 shuffler_vec = {0, 0, 0, 0, 4, 4, 4, 4, 8, 8, 8, 8, 12, 12, 12, 12};
>-  int32 val = LW((int32*)shuffler);
>+  int32_t val = LW((int32_t*)shuffler);
> 
>   vec0 = (v16i8)__msa_fill_w(val);
>   shuffler_vec += vec0;
>@@ -1363,10 +1369,10 @@ void ARGBShuffleRow_MSA(const uint8* src_argb,
>   }
> }
> 
>-void ARGBShadeRow_MSA(const uint8* src_argb,
>-                      uint8* dst_argb,
>+void ARGBShadeRow_MSA(const uint8_t* src_argb,
>+                      uint8_t* dst_argb,
>                       int width,
>-                      uint32 value) {
>+                      uint32_t value) {
>   int x;
>   v16u8 src0, dst0;
>   v8u16 vec0, vec1;
>@@ -1402,7 +1408,7 @@ void ARGBShadeRow_MSA(const uint8* src_argb,
>   }
> }
> 
>-void ARGBGrayRow_MSA(const uint8* src_argb, uint8* dst_argb, int width) {
>+void ARGBGrayRow_MSA(const uint8_t* src_argb, uint8_t* dst_argb, int width) {
>   int x;
>   v16u8 src0, src1, vec0, vec1, dst0, dst1;
>   v8u16 reg0;
>@@ -1427,7 +1433,7 @@ void ARGBGrayRow_MSA(const uint8* src_argb, uint8* dst_argb, int width) {
>   }
> }
> 
>-void ARGBSepiaRow_MSA(uint8* dst_argb, int width) {
>+void ARGBSepiaRow_MSA(uint8_t* dst_argb, int width) {
>   int x;
>   v16u8 src0, src1, dst0, dst1, vec0, vec1, vec2, vec3, vec4, vec5;
>   v8u16 reg0, reg1, reg2;
>@@ -1468,8 +1474,8 @@ void ARGBSepiaRow_MSA(uint8* dst_argb, int width) {
>   }
> }
> 
>-void ARGB4444ToARGBRow_MSA(const uint8* src_argb4444,
>-                           uint8* dst_argb,
>+void ARGB4444ToARGBRow_MSA(const uint8_t* src_argb4444,
>+                           uint8_t* dst_argb,
>                            int width) {
>   int x;
>   v16u8 src0, src1;
>@@ -1497,8 +1503,8 @@ void ARGB4444ToARGBRow_MSA(const uint8* src_argb4444,
>   }
> }
> 
>-void ARGB1555ToARGBRow_MSA(const uint8* src_argb1555,
>-                           uint8* dst_argb,
>+void ARGB1555ToARGBRow_MSA(const uint8_t* src_argb1555,
>+                           uint8_t* dst_argb,
>                            int width) {
>   int x;
>   v8u16 src0, src1;
>@@ -1547,7 +1553,9 @@ void ARGB1555ToARGBRow_MSA(const uint8* src_argb1555,
>   }
> }
> 
>-void RGB565ToARGBRow_MSA(const uint8* src_rgb565, uint8* dst_argb, int width) {
>+void RGB565ToARGBRow_MSA(const uint8_t* src_rgb565,
>+                         uint8_t* dst_argb,
>+                         int width) {
>   int x;
>   v8u16 src0, src1, vec0, vec1, vec2, vec3, vec4, vec5;
>   v8u16 reg0, reg1, reg2, reg3, reg4, reg5;
>@@ -1592,7 +1600,9 @@ void RGB565ToARGBRow_MSA(const uint8* src_rgb565, uint8* dst_argb, int width) {
>   }
> }
> 
>-void RGB24ToARGBRow_MSA(const uint8* src_rgb24, uint8* dst_argb, int width) {
>+void RGB24ToARGBRow_MSA(const uint8_t* src_rgb24,
>+                        uint8_t* dst_argb,
>+                        int width) {
>   int x;
>   v16u8 src0, src1, src2;
>   v16u8 vec0, vec1, vec2;
>@@ -1617,7 +1627,7 @@ void RGB24ToARGBRow_MSA(const uint8* src_rgb24, uint8* dst_argb, int width) {
>   }
> }
> 
>-void RAWToARGBRow_MSA(const uint8* src_raw, uint8* dst_argb, int width) {
>+void RAWToARGBRow_MSA(const uint8_t* src_raw, uint8_t* dst_argb, int width) {
>   int x;
>   v16u8 src0, src1, src2;
>   v16u8 vec0, vec1, vec2;
>@@ -1642,7 +1652,9 @@ void RAWToARGBRow_MSA(const uint8* src_raw, uint8* dst_argb, int width) {
>   }
> }
> 
>-void ARGB1555ToYRow_MSA(const uint8* src_argb1555, uint8* dst_y, int width) {
>+void ARGB1555ToYRow_MSA(const uint8_t* src_argb1555,
>+                        uint8_t* dst_y,
>+                        int width) {
>   int x;
>   v8u16 src0, src1, vec0, vec1, vec2, vec3, vec4, vec5;
>   v8u16 reg0, reg1, reg2, reg3, reg4, reg5;
>@@ -1699,7 +1711,7 @@ void ARGB1555ToYRow_MSA(const uint8* src_argb1555, uint8* dst_y, int width) {
>   }
> }
> 
>-void RGB565ToYRow_MSA(const uint8* src_rgb565, uint8* dst_y, int width) {
>+void RGB565ToYRow_MSA(const uint8_t* src_rgb565, uint8_t* dst_y, int width) {
>   int x;
>   v8u16 src0, src1, vec0, vec1, vec2, vec3, vec4, vec5, vec6, vec7;
>   v8u16 reg0, reg1, reg2, reg3, reg4, reg5;
>@@ -1762,7 +1774,7 @@ void RGB565ToYRow_MSA(const uint8* src_rgb565, uint8* dst_y, int width) {
>   }
> }
> 
>-void RGB24ToYRow_MSA(const uint8* src_argb0, uint8* dst_y, int width) {
>+void RGB24ToYRow_MSA(const uint8_t* src_argb0, uint8_t* dst_y, int width) {
>   int x;
>   v16u8 src0, src1, src2, reg0, reg1, reg2, reg3, dst0;
>   v8u16 vec0, vec1, vec2, vec3;
>@@ -1803,7 +1815,7 @@ void RGB24ToYRow_MSA(const uint8* src_argb0, uint8* dst_y, int width) {
>   }
> }
> 
>-void RAWToYRow_MSA(const uint8* src_argb0, uint8* dst_y, int width) {
>+void RAWToYRow_MSA(const uint8_t* src_argb0, uint8_t* dst_y, int width) {
>   int x;
>   v16u8 src0, src1, src2, reg0, reg1, reg2, reg3, dst0;
>   v8u16 vec0, vec1, vec2, vec3;
>@@ -1844,14 +1856,14 @@ void RAWToYRow_MSA(const uint8* src_argb0, uint8* dst_y, int width) {
>   }
> }
> 
>-void ARGB1555ToUVRow_MSA(const uint8* src_argb1555,
>+void ARGB1555ToUVRow_MSA(const uint8_t* src_argb1555,
>                          int src_stride_argb1555,
>-                         uint8* dst_u,
>-                         uint8* dst_v,
>+                         uint8_t* dst_u,
>+                         uint8_t* dst_v,
>                          int width) {
>   int x;
>-  const uint16* s = (const uint16*)src_argb1555;
>-  const uint16* t = (const uint16*)(src_argb1555 + src_stride_argb1555);
>+  const uint16_t* s = (const uint16_t*)src_argb1555;
>+  const uint16_t* t = (const uint16_t*)(src_argb1555 + src_stride_argb1555);
>   int64_t res0, res1;
>   v8u16 src0, src1, src2, src3, reg0, reg1, reg2, reg3;
>   v8u16 vec0, vec1, vec2, vec3, vec4, vec5, vec6;
>@@ -1925,14 +1937,14 @@ void ARGB1555ToUVRow_MSA(const uint8* src_argb1555,
>   }
> }
> 
>-void RGB565ToUVRow_MSA(const uint8* src_rgb565,
>+void RGB565ToUVRow_MSA(const uint8_t* src_rgb565,
>                        int src_stride_rgb565,
>-                       uint8* dst_u,
>-                       uint8* dst_v,
>+                       uint8_t* dst_u,
>+                       uint8_t* dst_v,
>                        int width) {
>   int x;
>-  const uint16* s = (const uint16*)src_rgb565;
>-  const uint16* t = (const uint16*)(src_rgb565 + src_stride_rgb565);
>+  const uint16_t* s = (const uint16_t*)src_rgb565;
>+  const uint16_t* t = (const uint16_t*)(src_rgb565 + src_stride_rgb565);
>   int64_t res0, res1;
>   v8u16 src0, src1, src2, src3, reg0, reg1, reg2, reg3;
>   v8u16 vec0, vec1, vec2, vec3, vec4, vec5;
>@@ -2005,15 +2017,15 @@ void RGB565ToUVRow_MSA(const uint8* src_rgb565,
>   }
> }
> 
>-void RGB24ToUVRow_MSA(const uint8* src_rgb0,
>+void RGB24ToUVRow_MSA(const uint8_t* src_rgb0,
>                       int src_stride_rgb,
>-                      uint8* dst_u,
>-                      uint8* dst_v,
>+                      uint8_t* dst_u,
>+                      uint8_t* dst_v,
>                       int width) {
>   int x;
>-  const uint8* s = src_rgb0;
>-  const uint8* t = src_rgb0 + src_stride_rgb;
>-  int64 res0, res1;
>+  const uint8_t* s = src_rgb0;
>+  const uint8_t* t = src_rgb0 + src_stride_rgb;
>+  int64_t res0, res1;
>   v16u8 src0, src1, src2, src3, src4, src5, src6, src7;
>   v16u8 inp0, inp1, inp2, inp3, inp4, inp5;
>   v8u16 vec0, vec1, vec2, vec3, vec4, vec5, vec6, vec7;
>@@ -2110,15 +2122,15 @@ void RGB24ToUVRow_MSA(const uint8* src_rgb0,
>   }
> }
> 
>-void RAWToUVRow_MSA(const uint8* src_rgb0,
>+void RAWToUVRow_MSA(const uint8_t* src_rgb0,
>                     int src_stride_rgb,
>-                    uint8* dst_u,
>-                    uint8* dst_v,
>+                    uint8_t* dst_u,
>+                    uint8_t* dst_v,
>                     int width) {
>   int x;
>-  const uint8* s = src_rgb0;
>-  const uint8* t = src_rgb0 + src_stride_rgb;
>-  int64 res0, res1;
>+  const uint8_t* s = src_rgb0;
>+  const uint8_t* t = src_rgb0 + src_stride_rgb;
>+  int64_t res0, res1;
>   v16u8 inp0, inp1, inp2, inp3, inp4, inp5;
>   v16u8 src0, src1, src2, src3, src4, src5, src6, src7;
>   v8u16 vec0, vec1, vec2, vec3, vec4, vec5, vec6, vec7;
>@@ -2215,13 +2227,13 @@ void RAWToUVRow_MSA(const uint8* src_rgb0,
>   }
> }
> 
>-void NV12ToARGBRow_MSA(const uint8* src_y,
>-                       const uint8* src_uv,
>-                       uint8* rgb_buf,
>+void NV12ToARGBRow_MSA(const uint8_t* src_y,
>+                       const uint8_t* src_uv,
>+                       uint8_t* dst_argb,
>                        const struct YuvConstants* yuvconstants,
>                        int width) {
>   int x;
>-  uint64 val0, val1;
>+  uint64_t val0, val1;
>   v16u8 src0, src1, res0, res1, dst0, dst1;
>   v8i16 vec0, vec1, vec2;
>   v4i32 vec_ub, vec_vr, vec_ug, vec_vg, vec_bb, vec_bg, vec_br, vec_yg;
>@@ -2245,20 +2257,20 @@ void NV12ToARGBRow_MSA(const uint8* src_y,
>     res1 = (v16u8)__msa_ilvev_b((v16i8)alpha, (v16i8)vec1);
>     dst0 = (v16u8)__msa_ilvr_b((v16i8)res1, (v16i8)res0);
>     dst1 = (v16u8)__msa_ilvl_b((v16i8)res1, (v16i8)res0);
>-    ST_UB2(dst0, dst1, rgb_buf, 16);
>+    ST_UB2(dst0, dst1, dst_argb, 16);
>     src_y += 8;
>     src_uv += 8;
>-    rgb_buf += 32;
>+    dst_argb += 32;
>   }
> }
> 
>-void NV12ToRGB565Row_MSA(const uint8* src_y,
>-                         const uint8* src_uv,
>-                         uint8* rgb_buf,
>+void NV12ToRGB565Row_MSA(const uint8_t* src_y,
>+                         const uint8_t* src_uv,
>+                         uint8_t* dst_rgb565,
>                          const struct YuvConstants* yuvconstants,
>                          int width) {
>   int x;
>-  uint64 val0, val1;
>+  uint64_t val0, val1;
>   v16u8 src0, src1, dst0;
>   v8i16 vec0, vec1, vec2;
>   v4i32 vec_ub, vec_vr, vec_ug, vec_vg, vec_bb, vec_bg, vec_br, vec_yg;
>@@ -2281,20 +2293,20 @@ void NV12ToRGB565Row_MSA(const uint8* src_y,
>     vec1 = (vec1 >> 2) << 5;
>     vec2 = (vec2 >> 3) << 11;
>     dst0 = (v16u8)(vec0 | vec1 | vec2);
>-    ST_UB(dst0, rgb_buf);
>+    ST_UB(dst0, dst_rgb565);
>     src_y += 8;
>     src_uv += 8;
>-    rgb_buf += 16;
>+    dst_rgb565 += 16;
>   }
> }
> 
>-void NV21ToARGBRow_MSA(const uint8* src_y,
>-                       const uint8* src_vu,
>-                       uint8* rgb_buf,
>+void NV21ToARGBRow_MSA(const uint8_t* src_y,
>+                       const uint8_t* src_vu,
>+                       uint8_t* dst_argb,
>                        const struct YuvConstants* yuvconstants,
>                        int width) {
>   int x;
>-  uint64 val0, val1;
>+  uint64_t val0, val1;
>   v16u8 src0, src1, res0, res1, dst0, dst1;
>   v8i16 vec0, vec1, vec2;
>   v4i32 vec_ub, vec_vr, vec_ug, vec_vg, vec_bb, vec_bg, vec_br, vec_yg;
>@@ -2320,16 +2332,16 @@ void NV21ToARGBRow_MSA(const uint8* src_y,
>     res1 = (v16u8)__msa_ilvev_b((v16i8)alpha, (v16i8)vec1);
>     dst0 = (v16u8)__msa_ilvr_b((v16i8)res1, (v16i8)res0);
>     dst1 = (v16u8)__msa_ilvl_b((v16i8)res1, (v16i8)res0);
>-    ST_UB2(dst0, dst1, rgb_buf, 16);
>+    ST_UB2(dst0, dst1, dst_argb, 16);
>     src_y += 8;
>     src_vu += 8;
>-    rgb_buf += 32;
>+    dst_argb += 32;
>   }
> }
> 
>-void SobelRow_MSA(const uint8* src_sobelx,
>-                  const uint8* src_sobely,
>-                  uint8* dst_argb,
>+void SobelRow_MSA(const uint8_t* src_sobelx,
>+                  const uint8_t* src_sobely,
>+                  uint8_t* dst_argb,
>                   int width) {
>   int x;
>   v16u8 src0, src1, vec0, dst0, dst1, dst2, dst3;
>@@ -2355,9 +2367,9 @@ void SobelRow_MSA(const uint8* src_sobelx,
>   }
> }
> 
>-void SobelToPlaneRow_MSA(const uint8* src_sobelx,
>-                         const uint8* src_sobely,
>-                         uint8* dst_y,
>+void SobelToPlaneRow_MSA(const uint8_t* src_sobelx,
>+                         const uint8_t* src_sobely,
>+                         uint8_t* dst_y,
>                          int width) {
>   int x;
>   v16u8 src0, src1, src2, src3, dst0, dst1;
>@@ -2376,9 +2388,9 @@ void SobelToPlaneRow_MSA(const uint8* src_sobelx,
>   }
> }
> 
>-void SobelXYRow_MSA(const uint8* src_sobelx,
>-                    const uint8* src_sobely,
>-                    uint8* dst_argb,
>+void SobelXYRow_MSA(const uint8_t* src_sobelx,
>+                    const uint8_t* src_sobely,
>+                    uint8_t* dst_argb,
>                     int width) {
>   int x;
>   v16u8 src0, src1, vec0, vec1, vec2;
>@@ -2404,7 +2416,7 @@ void SobelXYRow_MSA(const uint8* src_sobelx,
>   }
> }
> 
>-void ARGBToYJRow_MSA(const uint8* src_argb0, uint8* dst_y, int width) {
>+void ARGBToYJRow_MSA(const uint8_t* src_argb0, uint8_t* dst_y, int width) {
>   int x;
>   v16u8 src0, src1, src2, src3, dst0;
>   v16u8 const_0x4B0F = (v16u8)__msa_fill_h(0x4B0F);
>@@ -2424,7 +2436,7 @@ void ARGBToYJRow_MSA(const uint8* src_argb0, uint8* dst_y, int width) {
>   }
> }
> 
>-void BGRAToYRow_MSA(const uint8* src_argb0, uint8* dst_y, int width) {
>+void BGRAToYRow_MSA(const uint8_t* src_argb0, uint8_t* dst_y, int width) {
>   int x;
>   v16u8 src0, src1, src2, src3, dst0;
>   v16u8 const_0x4200 = (v16u8)__msa_fill_h(0x4200);
>@@ -2444,7 +2456,7 @@ void BGRAToYRow_MSA(const uint8* src_argb0, uint8* dst_y, int width) {
>   }
> }
> 
>-void ABGRToYRow_MSA(const uint8* src_argb0, uint8* dst_y, int width) {
>+void ABGRToYRow_MSA(const uint8_t* src_argb0, uint8_t* dst_y, int width) {
>   int x;
>   v16u8 src0, src1, src2, src3, dst0;
>   v16u8 const_0x8142 = (v16u8)__msa_fill_h(0x8142);
>@@ -2464,7 +2476,7 @@ void ABGRToYRow_MSA(const uint8* src_argb0, uint8* dst_y, int width) {
>   }
> }
> 
>-void RGBAToYRow_MSA(const uint8* src_argb0, uint8* dst_y, int width) {
>+void RGBAToYRow_MSA(const uint8_t* src_argb0, uint8_t* dst_y, int width) {
>   int x;
>   v16u8 src0, src1, src2, src3, dst0;
>   v16u8 const_0x1900 = (v16u8)__msa_fill_h(0x1900);
>@@ -2484,14 +2496,14 @@ void RGBAToYRow_MSA(const uint8* src_argb0, uint8* dst_y, int width) {
>   }
> }
> 
>-void ARGBToUVJRow_MSA(const uint8* src_rgb0,
>+void ARGBToUVJRow_MSA(const uint8_t* src_rgb0,
>                       int src_stride_rgb,
>-                      uint8* dst_u,
>-                      uint8* dst_v,
>+                      uint8_t* dst_u,
>+                      uint8_t* dst_v,
>                       int width) {
>   int x;
>-  const uint8* s = src_rgb0;
>-  const uint8* t = src_rgb0 + src_stride_rgb;
>+  const uint8_t* s = src_rgb0;
>+  const uint8_t* t = src_rgb0 + src_stride_rgb;
>   v16u8 src0, src1, src2, src3, src4, src5, src6, src7;
>   v16u8 vec0, vec1, vec2, vec3;
>   v16u8 dst0, dst1;
>@@ -2554,14 +2566,14 @@ void ARGBToUVJRow_MSA(const uint8* src_rgb0,
>   }
> }
> 
>-void BGRAToUVRow_MSA(const uint8* src_rgb0,
>+void BGRAToUVRow_MSA(const uint8_t* src_rgb0,
>                      int src_stride_rgb,
>-                     uint8* dst_u,
>-                     uint8* dst_v,
>+                     uint8_t* dst_u,
>+                     uint8_t* dst_v,
>                      int width) {
>   int x;
>-  const uint8* s = src_rgb0;
>-  const uint8* t = src_rgb0 + src_stride_rgb;
>+  const uint8_t* s = src_rgb0;
>+  const uint8_t* t = src_rgb0 + src_stride_rgb;
>   v16u8 dst0, dst1, vec0, vec1, vec2, vec3;
>   v16i8 shuffler0 = {0, 1, 4, 5, 8, 9, 12, 13, 16, 17, 20, 21, 24, 25, 28, 29};
>   v16i8 shuffler1 = {2,  3,  6,  7,  10, 11, 14, 15,
>@@ -2587,14 +2599,14 @@ void BGRAToUVRow_MSA(const uint8* src_rgb0,
>   }
> }
> 
>-void ABGRToUVRow_MSA(const uint8* src_rgb0,
>+void ABGRToUVRow_MSA(const uint8_t* src_rgb0,
>                      int src_stride_rgb,
>-                     uint8* dst_u,
>-                     uint8* dst_v,
>+                     uint8_t* dst_u,
>+                     uint8_t* dst_v,
>                      int width) {
>   int x;
>-  const uint8* s = src_rgb0;
>-  const uint8* t = src_rgb0 + src_stride_rgb;
>+  const uint8_t* s = src_rgb0;
>+  const uint8_t* t = src_rgb0 + src_stride_rgb;
>   v16u8 src0, src1, src2, src3;
>   v16u8 dst0, dst1;
>   v16i8 shuffler0 = {0, 1, 4, 5, 8, 9, 12, 13, 16, 17, 20, 21, 24, 25, 28, 29};
>@@ -2621,14 +2633,14 @@ void ABGRToUVRow_MSA(const uint8* src_rgb0,
>   }
> }
> 
>-void RGBAToUVRow_MSA(const uint8* src_rgb0,
>+void RGBAToUVRow_MSA(const uint8_t* src_rgb0,
>                      int src_stride_rgb,
>-                     uint8* dst_u,
>-                     uint8* dst_v,
>+                     uint8_t* dst_u,
>+                     uint8_t* dst_v,
>                      int width) {
>   int x;
>-  const uint8* s = src_rgb0;
>-  const uint8* t = src_rgb0 + src_stride_rgb;
>+  const uint8_t* s = src_rgb0;
>+  const uint8_t* t = src_rgb0 + src_stride_rgb;
>   v16u8 dst0, dst1, vec0, vec1, vec2, vec3;
>   v16i8 shuffler0 = {0, 1, 4, 5, 8, 9, 12, 13, 16, 17, 20, 21, 24, 25, 28, 29};
>   v16i8 shuffler1 = {2,  3,  6,  7,  10, 11, 14, 15,
>@@ -2654,10 +2666,10 @@ void RGBAToUVRow_MSA(const uint8* src_rgb0,
>   }
> }
> 
>-void I444ToARGBRow_MSA(const uint8* src_y,
>-                       const uint8* src_u,
>-                       const uint8* src_v,
>-                       uint8* rgb_buf,
>+void I444ToARGBRow_MSA(const uint8_t* src_y,
>+                       const uint8_t* src_u,
>+                       const uint8_t* src_v,
>+                       uint8_t* dst_argb,
>                        const struct YuvConstants* yuvconstants,
>                        int width) {
>   int x;
>@@ -2714,15 +2726,15 @@ void I444ToARGBRow_MSA(const uint8* src_y,
>     vec1 = (v8u16)__msa_ilvev_b((v16i8)alpha, (v16i8)vec2);
>     dst0 = (v16u8)__msa_ilvr_h((v8i16)vec1, (v8i16)vec0);
>     dst1 = (v16u8)__msa_ilvl_h((v8i16)vec1, (v8i16)vec0);
>-    ST_UB2(dst0, dst1, rgb_buf, 16);
>+    ST_UB2(dst0, dst1, dst_argb, 16);
>     src_y += 8;
>     src_u += 8;
>     src_v += 8;
>-    rgb_buf += 32;
>+    dst_argb += 32;
>   }
> }
> 
>-void I400ToARGBRow_MSA(const uint8* src_y, uint8* rgb_buf, int width) {
>+void I400ToARGBRow_MSA(const uint8_t* src_y, uint8_t* dst_argb, int width) {
>   int x;
>   v16u8 src0, res0, res1, res2, res3, res4, dst0, dst1, dst2, dst3;
>   v8i16 vec0, vec1;
>@@ -2768,13 +2780,13 @@ void I400ToARGBRow_MSA(const uint8* src_y, uint8* rgb_buf, int width) {
>     dst1 = (v16u8)__msa_ilvl_b((v16i8)res3, (v16i8)res1);
>     dst2 = (v16u8)__msa_ilvr_b((v16i8)res4, (v16i8)res2);
>     dst3 = (v16u8)__msa_ilvl_b((v16i8)res4, (v16i8)res2);
>-    ST_UB4(dst0, dst1, dst2, dst3, rgb_buf, 16);
>+    ST_UB4(dst0, dst1, dst2, dst3, dst_argb, 16);
>     src_y += 16;
>-    rgb_buf += 64;
>+    dst_argb += 64;
>   }
> }
> 
>-void J400ToARGBRow_MSA(const uint8* src_y, uint8* dst_argb, int width) {
>+void J400ToARGBRow_MSA(const uint8_t* src_y, uint8_t* dst_argb, int width) {
>   int x;
>   v16u8 src0, vec0, vec1, vec2, vec3, dst0, dst1, dst2, dst3;
>   v16u8 alpha = (v16u8)__msa_ldi_b(ALPHA_VAL);
>@@ -2795,8 +2807,8 @@ void J400ToARGBRow_MSA(const uint8* src_y, uint8* dst_argb, int width) {
>   }
> }
> 
>-void YUY2ToARGBRow_MSA(const uint8* src_yuy2,
>-                       uint8* rgb_buf,
>+void YUY2ToARGBRow_MSA(const uint8_t* src_yuy2,
>+                       uint8_t* dst_argb,
>                        const struct YuvConstants* yuvconstants,
>                        int width) {
>   int x;
>@@ -2817,14 +2829,14 @@ void YUY2ToARGBRow_MSA(const uint8* src_yuy2,
>     src2 = (v16u8)__msa_pckod_b((v16i8)src0, (v16i8)src0);
>     YUVTORGB(src1, src2, vec_ubvr, vec_ugvg, vec_bb, vec_bg, vec_br, vec_yg,
>              vec0, vec1, vec2);
>-    STOREARGB(vec0, vec1, vec2, alpha, rgb_buf);
>+    STOREARGB(vec0, vec1, vec2, alpha, dst_argb);
>     src_yuy2 += 16;
>-    rgb_buf += 32;
>+    dst_argb += 32;
>   }
> }
> 
>-void UYVYToARGBRow_MSA(const uint8* src_uyvy,
>-                       uint8* rgb_buf,
>+void UYVYToARGBRow_MSA(const uint8_t* src_uyvy,
>+                       uint8_t* dst_argb,
>                        const struct YuvConstants* yuvconstants,
>                        int width) {
>   int x;
>@@ -2845,22 +2857,22 @@ void UYVYToARGBRow_MSA(const uint8* src_uyvy,
>     src2 = (v16u8)__msa_pckev_b((v16i8)src0, (v16i8)src0);
>     YUVTORGB(src1, src2, vec_ubvr, vec_ugvg, vec_bb, vec_bg, vec_br, vec_yg,
>              vec0, vec1, vec2);
>-    STOREARGB(vec0, vec1, vec2, alpha, rgb_buf);
>+    STOREARGB(vec0, vec1, vec2, alpha, dst_argb);
>     src_uyvy += 16;
>-    rgb_buf += 32;
>+    dst_argb += 32;
>   }
> }
> 
>-void InterpolateRow_MSA(uint8* dst_ptr,
>-                        const uint8* src_ptr,
>+void InterpolateRow_MSA(uint8_t* dst_ptr,
>+                        const uint8_t* src_ptr,
>                         ptrdiff_t src_stride,
>                         int width,
>-                        int32 source_y_fraction) {
>-  int32 y1_fraction = source_y_fraction;
>-  int32 y0_fraction = 256 - y1_fraction;
>-  uint16 y_fractions;
>-  const uint8* s = src_ptr;
>-  const uint8* t = src_ptr + src_stride;
>+                        int32_t source_y_fraction) {
>+  int32_t y1_fraction = source_y_fraction;
>+  int32_t y0_fraction = 256 - y1_fraction;
>+  uint16_t y_fractions;
>+  const uint8_t* s = src_ptr;
>+  const uint8_t* t = src_ptr + src_stride;
>   int x;
>   v16u8 src0, src1, src2, src3, dst0, dst1;
>   v8u16 vec0, vec1, vec2, vec3, y_frac;
>@@ -2886,7 +2898,7 @@ void InterpolateRow_MSA(uint8* dst_ptr,
>     return;
>   }
> 
>-  y_fractions = (uint16)(y0_fraction + (y1_fraction << 8));
>+  y_fractions = (uint16_t)(y0_fraction + (y1_fraction << 8));
>   y_frac = (v8u16)__msa_fill_h(y_fractions);
> 
>   for (x = 0; x < width; x += 32) {
>@@ -2915,7 +2927,7 @@ void InterpolateRow_MSA(uint8* dst_ptr,
>   }
> }
> 
>-void ARGBSetRow_MSA(uint8* dst_argb, uint32 v32, int width) {
>+void ARGBSetRow_MSA(uint8_t* dst_argb, uint32_t v32, int width) {
>   int x;
>   v4i32 dst0 = __builtin_msa_fill_w(v32);
> 
>@@ -2925,7 +2937,7 @@ void ARGBSetRow_MSA(uint8* dst_argb, uint32 v32, int width) {
>   }
> }
> 
>-void RAWToRGB24Row_MSA(const uint8* src_raw, uint8* dst_rgb24, int width) {
>+void RAWToRGB24Row_MSA(const uint8_t* src_raw, uint8_t* dst_rgb24, int width) {
>   int x;
>   v16u8 src0, src1, src2, src3, src4, dst0, dst1, dst2;
>   v16i8 shuffler0 = {2, 1, 0, 5, 4, 3, 8, 7, 6, 11, 10, 9, 14, 13, 12, 17};
>@@ -2950,9 +2962,9 @@ void RAWToRGB24Row_MSA(const uint8* src_raw, uint8* dst_rgb24, int width) {
>   }
> }
> 
>-void MergeUVRow_MSA(const uint8* src_u,
>-                    const uint8* src_v,
>-                    uint8* dst_uv,
>+void MergeUVRow_MSA(const uint8_t* src_u,
>+                    const uint8_t* src_v,
>+                    uint8_t* dst_uv,
>                     int width) {
>   int x;
>   v16u8 src0, src1, dst0, dst1;
>@@ -2969,7 +2981,9 @@ void MergeUVRow_MSA(const uint8* src_u,
>   }
> }
> 
>-void ARGBExtractAlphaRow_MSA(const uint8* src_argb, uint8* dst_a, int width) {
>+void ARGBExtractAlphaRow_MSA(const uint8_t* src_argb,
>+                             uint8_t* dst_a,
>+                             int width) {
>   int i;
>   v16u8 src0, src1, src2, src3, vec0, vec1, dst0;
> 
>@@ -2987,9 +3001,9 @@ void ARGBExtractAlphaRow_MSA(const uint8* src_argb, uint8* dst_a, int width) {
>   }
> }
> 
>-void ARGBBlendRow_MSA(const uint8* src_argb0,
>-                      const uint8* src_argb1,
>-                      uint8* dst_argb,
>+void ARGBBlendRow_MSA(const uint8_t* src_argb0,
>+                      const uint8_t* src_argb1,
>+                      uint8_t* dst_argb,
>                       int width) {
>   int x;
>   v16u8 src0, src1, src2, src3, dst0, dst1;
>@@ -3052,7 +3066,7 @@ void ARGBBlendRow_MSA(const uint8* src_argb0,
>   }
> }
> 
>-void ARGBQuantizeRow_MSA(uint8* dst_argb,
>+void ARGBQuantizeRow_MSA(uint8_t* dst_argb,
>                          int scale,
>                          int interval_size,
>                          int interval_offset,
>@@ -3158,11 +3172,11 @@ void ARGBQuantizeRow_MSA(uint8* dst_argb,
>   }
> }
> 
>-void ARGBColorMatrixRow_MSA(const uint8* src_argb,
>-                            uint8* dst_argb,
>-                            const int8* matrix_argb,
>+void ARGBColorMatrixRow_MSA(const uint8_t* src_argb,
>+                            uint8_t* dst_argb,
>+                            const int8_t* matrix_argb,
>                             int width) {
>-  int32 x;
>+  int32_t x;
>   v16i8 src0;
>   v16u8 src1, src2, dst0, dst1;
>   v8i16 vec0, vec1, vec2, vec3, vec4, vec5, vec6, vec7, vec8, vec9;
>@@ -3267,9 +3281,9 @@ void ARGBColorMatrixRow_MSA(const uint8* src_argb,
>   }
> }
> 
>-void SplitUVRow_MSA(const uint8* src_uv,
>-                    uint8* dst_u,
>-                    uint8* dst_v,
>+void SplitUVRow_MSA(const uint8_t* src_uv,
>+                    uint8_t* dst_u,
>+                    uint8_t* dst_v,
>                     int width) {
>   int x;
>   v16u8 src0, src1, src2, src3, dst0, dst1, dst2, dst3;
>@@ -3291,7 +3305,7 @@ void SplitUVRow_MSA(const uint8* src_uv,
>   }
> }
> 
>-void SetRow_MSA(uint8* dst, uint8 v8, int width) {
>+void SetRow_MSA(uint8_t* dst, uint8_t v8, int width) {
>   int x;
>   v16u8 dst0 = (v16u8)__msa_fill_b(v8);
> 
>@@ -3301,9 +3315,9 @@ void SetRow_MSA(uint8* dst, uint8 v8, int width) {
>   }
> }
> 
>-void MirrorUVRow_MSA(const uint8* src_uv,
>-                     uint8* dst_u,
>-                     uint8* dst_v,
>+void MirrorUVRow_MSA(const uint8_t* src_uv,
>+                     uint8_t* dst_u,
>+                     uint8_t* dst_v,
>                      int width) {
>   int x;
>   v16u8 src0, src1, src2, src3;
>@@ -3330,11 +3344,11 @@ void MirrorUVRow_MSA(const uint8* src_uv,
>   }
> }
> 
>-void SobelXRow_MSA(const uint8* src_y0,
>-                   const uint8* src_y1,
>-                   const uint8* src_y2,
>-                   uint8* dst_sobelx,
>-                   int32 width) {
>+void SobelXRow_MSA(const uint8_t* src_y0,
>+                   const uint8_t* src_y1,
>+                   const uint8_t* src_y2,
>+                   uint8_t* dst_sobelx,
>+                   int32_t width) {
>   int x;
>   v16u8 src0, src1, src2, src3, src4, src5, dst0;
>   v8i16 vec0, vec1, vec2, vec3, vec4, vec5;
>@@ -3384,10 +3398,10 @@ void SobelXRow_MSA(const uint8* src_y0,
>   }
> }
> 
>-void SobelYRow_MSA(const uint8* src_y0,
>-                   const uint8* src_y1,
>-                   uint8* dst_sobely,
>-                   int32 width) {
>+void SobelYRow_MSA(const uint8_t* src_y0,
>+                   const uint8_t* src_y1,
>+                   uint8_t* dst_sobely,
>+                   int32_t width) {
>   int x;
>   v16u8 src0, src1, dst0;
>   v8i16 vec0, vec1, vec2, vec3, vec4, vec5, vec6;
>@@ -3429,7 +3443,10 @@ void SobelYRow_MSA(const uint8* src_y0,
>   }
> }
> 
>-void HalfFloatRow_MSA(const uint16* src, uint16* dst, float scale, int width) {
>+void HalfFloatRow_MSA(const uint16_t* src,
>+                      uint16_t* dst,
>+                      float scale,
>+                      int width) {
>   int i;
>   v8u16 src0, src1, src2, src3, dst0, dst1, dst2, dst3;
>   v4u32 vec0, vec1, vec2, vec3, vec4, vec5, vec6, vec7;
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/row_neon.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/row_neon.cc
>index 1af828622cd..8b6c1952072 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/row_neon.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/row_neon.cc
>@@ -106,10 +106,10 @@ extern "C" {
>   "vqshrun.s16 d22, q9, #6                   \n" /* R */                      \
>   "vqshrun.s16 d21, q0, #6                   \n" /* G */
> 
>-void I444ToARGBRow_NEON(const uint8* src_y,
>-                        const uint8* src_u,
>-                        const uint8* src_v,
>-                        uint8* dst_argb,
>+void I444ToARGBRow_NEON(const uint8_t* src_y,
>+                        const uint8_t* src_u,
>+                        const uint8_t* src_v,
>+                        uint8_t* dst_argb,
>                         const struct YuvConstants* yuvconstants,
>                         int width) {
>   asm volatile(
>@@ -132,10 +132,10 @@ void I444ToARGBRow_NEON(const uint8* src_y,
>         "q12", "q13", "q14", "q15");
> }
> 
>-void I422ToARGBRow_NEON(const uint8* src_y,
>-                        const uint8* src_u,
>-                        const uint8* src_v,
>-                        uint8* dst_argb,
>+void I422ToARGBRow_NEON(const uint8_t* src_y,
>+                        const uint8_t* src_u,
>+                        const uint8_t* src_v,
>+                        uint8_t* dst_argb,
>                         const struct YuvConstants* yuvconstants,
>                         int width) {
>   asm volatile(
>@@ -158,11 +158,11 @@ void I422ToARGBRow_NEON(const uint8* src_y,
>         "q12", "q13", "q14", "q15");
> }
> 
>-void I422AlphaToARGBRow_NEON(const uint8* src_y,
>-                             const uint8* src_u,
>-                             const uint8* src_v,
>-                             const uint8* src_a,
>-                             uint8* dst_argb,
>+void I422AlphaToARGBRow_NEON(const uint8_t* src_y,
>+                             const uint8_t* src_u,
>+                             const uint8_t* src_v,
>+                             const uint8_t* src_a,
>+                             uint8_t* dst_argb,
>                              const struct YuvConstants* yuvconstants,
>                              int width) {
>   asm volatile(
>@@ -186,18 +186,17 @@ void I422AlphaToARGBRow_NEON(const uint8* src_y,
>         "q12", "q13", "q14", "q15");
> }
> 
>-void I422ToRGBARow_NEON(const uint8* src_y,
>-                        const uint8* src_u,
>-                        const uint8* src_v,
>-                        uint8* dst_rgba,
>+void I422ToRGBARow_NEON(const uint8_t* src_y,
>+                        const uint8_t* src_u,
>+                        const uint8_t* src_v,
>+                        uint8_t* dst_rgba,
>                         const struct YuvConstants* yuvconstants,
>                         int width) {
>   asm volatile(
>       YUVTORGB_SETUP
>       "1:                                        \n" READYUV422 YUVTORGB
>       "subs       %4, %4, #8                     \n"
>-      "vmov.u8    d19, #255                      \n"  // d19 modified by
>-                                                      // YUVTORGB
>+      "vmov.u8    d19, #255                      \n"  // YUVTORGB modified d19
>       "vst4.8     {d19, d20, d21, d22}, [%3]!    \n"
>       "bgt        1b                             \n"
>       : "+r"(src_y),     // %0
>@@ -213,10 +212,10 @@ void I422ToRGBARow_NEON(const uint8* src_y,
>         "q12", "q13", "q14", "q15");
> }
> 
>-void I422ToRGB24Row_NEON(const uint8* src_y,
>-                         const uint8* src_u,
>-                         const uint8* src_v,
>-                         uint8* dst_rgb24,
>+void I422ToRGB24Row_NEON(const uint8_t* src_y,
>+                         const uint8_t* src_u,
>+                         const uint8_t* src_v,
>+                         uint8_t* dst_rgb24,
>                          const struct YuvConstants* yuvconstants,
>                          int width) {
>   asm volatile(
>@@ -245,10 +244,10 @@ void I422ToRGB24Row_NEON(const uint8* src_y,
>   "vsri.16     q0, q8, #5                    \n" /* RG                   */ \
>   "vsri.16     q0, q9, #11                   \n" /* RGB                  */
> 
>-void I422ToRGB565Row_NEON(const uint8* src_y,
>-                          const uint8* src_u,
>-                          const uint8* src_v,
>-                          uint8* dst_rgb565,
>+void I422ToRGB565Row_NEON(const uint8_t* src_y,
>+                          const uint8_t* src_u,
>+                          const uint8_t* src_v,
>+                          uint8_t* dst_rgb565,
>                           const struct YuvConstants* yuvconstants,
>                           int width) {
>   asm volatile(
>@@ -279,10 +278,10 @@ void I422ToRGB565Row_NEON(const uint8* src_y,
>   "vsri.16     q0, q9, #6                    \n" /* ARG                  */ \
>   "vsri.16     q0, q10, #11                  \n" /* ARGB                 */
> 
>-void I422ToARGB1555Row_NEON(const uint8* src_y,
>-                            const uint8* src_u,
>-                            const uint8* src_v,
>-                            uint8* dst_argb1555,
>+void I422ToARGB1555Row_NEON(const uint8_t* src_y,
>+                            const uint8_t* src_u,
>+                            const uint8_t* src_v,
>+                            uint8_t* dst_argb1555,
>                             const struct YuvConstants* yuvconstants,
>                             int width) {
>   asm volatile(
>@@ -291,7 +290,6 @@ void I422ToARGB1555Row_NEON(const uint8* src_y,
>       "subs       %4, %4, #8                     \n"
>       "vmov.u8    d23, #255                      \n" ARGBTOARGB1555
>       "vst1.8     {q0}, [%3]!                    \n"  // store 8 pixels
>-                                                      // ARGB1555.
>       "bgt        1b                             \n"
>       : "+r"(src_y),         // %0
>         "+r"(src_u),         // %1
>@@ -315,21 +313,21 @@ void I422ToARGB1555Row_NEON(const uint8* src_y,
>   "vorr       d1, d22, d23                   \n" /* RA                   */ \
>   "vzip.u8    d0, d1                         \n" /* BGRA                 */
> 
>-void I422ToARGB4444Row_NEON(const uint8* src_y,
>-                            const uint8* src_u,
>-                            const uint8* src_v,
>-                            uint8* dst_argb4444,
>+void I422ToARGB4444Row_NEON(const uint8_t* src_y,
>+                            const uint8_t* src_u,
>+                            const uint8_t* src_v,
>+                            uint8_t* dst_argb4444,
>                             const struct YuvConstants* yuvconstants,
>                             int width) {
>   asm volatile(
>       YUVTORGB_SETUP
>-      "vmov.u8    d4, #0x0f                      \n"  // bits to clear with
>-                                                      // vbic.
>-      "1:                                        \n" READYUV422 YUVTORGB
>+      "vmov.u8    d4, #0x0f                      \n"  // vbic bits to clear
>+      "1:                                        \n"
>+
>+      READYUV422 YUVTORGB
>       "subs       %4, %4, #8                     \n"
>       "vmov.u8    d23, #255                      \n" ARGBTOARGB4444
>       "vst1.8     {q0}, [%3]!                    \n"  // store 8 pixels
>-                                                      // ARGB4444.
>       "bgt        1b                             \n"
>       : "+r"(src_y),         // %0
>         "+r"(src_u),         // %1
>@@ -344,7 +342,7 @@ void I422ToARGB4444Row_NEON(const uint8* src_y,
>         "q12", "q13", "q14", "q15");
> }
> 
>-void I400ToARGBRow_NEON(const uint8* src_y, uint8* dst_argb, int width) {
>+void I400ToARGBRow_NEON(const uint8_t* src_y, uint8_t* dst_argb, int width) {
>   asm volatile(
>       YUVTORGB_SETUP
>       "vmov.u8    d23, #255                      \n"
>@@ -363,7 +361,7 @@ void I400ToARGBRow_NEON(const uint8* src_y, uint8* dst_argb, int width) {
>         "q12", "q13", "q14", "q15");
> }
> 
>-void J400ToARGBRow_NEON(const uint8* src_y, uint8* dst_argb, int width) {
>+void J400ToARGBRow_NEON(const uint8_t* src_y, uint8_t* dst_argb, int width) {
>   asm volatile(
>       "vmov.u8    d23, #255                      \n"
>       "1:                                        \n"
>@@ -380,9 +378,9 @@ void J400ToARGBRow_NEON(const uint8* src_y, uint8* dst_argb, int width) {
>       : "cc", "memory", "d20", "d21", "d22", "d23");
> }
> 
>-void NV12ToARGBRow_NEON(const uint8* src_y,
>-                        const uint8* src_uv,
>-                        uint8* dst_argb,
>+void NV12ToARGBRow_NEON(const uint8_t* src_y,
>+                        const uint8_t* src_uv,
>+                        uint8_t* dst_argb,
>                         const struct YuvConstants* yuvconstants,
>                         int width) {
>   asm volatile(YUVTORGB_SETUP
>@@ -403,9 +401,9 @@ void NV12ToARGBRow_NEON(const uint8* src_y,
>                  "q10", "q11", "q12", "q13", "q14", "q15");
> }
> 
>-void NV21ToARGBRow_NEON(const uint8* src_y,
>-                        const uint8* src_vu,
>-                        uint8* dst_argb,
>+void NV21ToARGBRow_NEON(const uint8_t* src_y,
>+                        const uint8_t* src_vu,
>+                        uint8_t* dst_argb,
>                         const struct YuvConstants* yuvconstants,
>                         int width) {
>   asm volatile(YUVTORGB_SETUP
>@@ -426,9 +424,63 @@ void NV21ToARGBRow_NEON(const uint8* src_y,
>                  "q10", "q11", "q12", "q13", "q14", "q15");
> }
> 
>-void NV12ToRGB565Row_NEON(const uint8* src_y,
>-                          const uint8* src_uv,
>-                          uint8* dst_rgb565,
>+void NV12ToRGB24Row_NEON(const uint8_t* src_y,
>+                         const uint8_t* src_uv,
>+                         uint8_t* dst_rgb24,
>+                         const struct YuvConstants* yuvconstants,
>+                         int width) {
>+  asm volatile(
>+
>+      YUVTORGB_SETUP
>+
>+      "1:                                        \n"
>+
>+      READNV12 YUVTORGB
>+      "subs       %3, %3, #8                     \n"
>+      "vst3.8     {d20, d21, d22}, [%2]!         \n"
>+      "bgt        1b                             \n"
>+      : "+r"(src_y),      // %0
>+        "+r"(src_uv),     // %1
>+        "+r"(dst_rgb24),  // %2
>+        "+r"(width)       // %3
>+      : [kUVToRB] "r"(&yuvconstants->kUVToRB),
>+        [kUVToG] "r"(&yuvconstants->kUVToG),
>+        [kUVBiasBGR] "r"(&yuvconstants->kUVBiasBGR),
>+        [kYToRgb] "r"(&yuvconstants->kYToRgb)
>+      : "cc", "memory", "q0", "q1", "q2", "q3", "q4", "q8", "q9", "q10", "q11",
>+        "q12", "q13", "q14", "q15");
>+}
>+
>+void NV21ToRGB24Row_NEON(const uint8_t* src_y,
>+                         const uint8_t* src_vu,
>+                         uint8_t* dst_rgb24,
>+                         const struct YuvConstants* yuvconstants,
>+                         int width) {
>+  asm volatile(
>+
>+      YUVTORGB_SETUP
>+
>+      "1:                                        \n"
>+
>+      READNV21 YUVTORGB
>+      "subs       %3, %3, #8                     \n"
>+      "vst3.8     {d20, d21, d22}, [%2]!         \n"
>+      "bgt        1b                             \n"
>+      : "+r"(src_y),      // %0
>+        "+r"(src_vu),     // %1
>+        "+r"(dst_rgb24),  // %2
>+        "+r"(width)       // %3
>+      : [kUVToRB] "r"(&yuvconstants->kUVToRB),
>+        [kUVToG] "r"(&yuvconstants->kUVToG),
>+        [kUVBiasBGR] "r"(&yuvconstants->kUVBiasBGR),
>+        [kYToRgb] "r"(&yuvconstants->kYToRgb)
>+      : "cc", "memory", "q0", "q1", "q2", "q3", "q4", "q8", "q9", "q10", "q11",
>+        "q12", "q13", "q14", "q15");
>+}
>+
>+void NV12ToRGB565Row_NEON(const uint8_t* src_y,
>+                          const uint8_t* src_uv,
>+                          uint8_t* dst_rgb565,
>                           const struct YuvConstants* yuvconstants,
>                           int width) {
>   asm volatile(
>@@ -449,8 +501,8 @@ void NV12ToRGB565Row_NEON(const uint8* src_y,
>         "q12", "q13", "q14", "q15");
> }
> 
>-void YUY2ToARGBRow_NEON(const uint8* src_yuy2,
>-                        uint8* dst_argb,
>+void YUY2ToARGBRow_NEON(const uint8_t* src_yuy2,
>+                        uint8_t* dst_argb,
>                         const struct YuvConstants* yuvconstants,
>                         int width) {
>   asm volatile(YUVTORGB_SETUP
>@@ -470,8 +522,8 @@ void YUY2ToARGBRow_NEON(const uint8* src_yuy2,
>                  "q10", "q11", "q12", "q13", "q14", "q15");
> }
> 
>-void UYVYToARGBRow_NEON(const uint8* src_uyvy,
>-                        uint8* dst_argb,
>+void UYVYToARGBRow_NEON(const uint8_t* src_uyvy,
>+                        uint8_t* dst_argb,
>                         const struct YuvConstants* yuvconstants,
>                         int width) {
>   asm volatile(YUVTORGB_SETUP
>@@ -492,9 +544,9 @@ void UYVYToARGBRow_NEON(const uint8* src_uyvy,
> }
> 
> // Reads 16 pairs of UV and write even values to dst_u and odd to dst_v.
>-void SplitUVRow_NEON(const uint8* src_uv,
>-                     uint8* dst_u,
>-                     uint8* dst_v,
>+void SplitUVRow_NEON(const uint8_t* src_uv,
>+                     uint8_t* dst_u,
>+                     uint8_t* dst_v,
>                      int width) {
>   asm volatile(
>       "1:                                        \n"
>@@ -513,9 +565,9 @@ void SplitUVRow_NEON(const uint8* src_uv,
> }
> 
> // Reads 16 U's and V's and writes out 16 pairs of UV.
>-void MergeUVRow_NEON(const uint8* src_u,
>-                     const uint8* src_v,
>-                     uint8* dst_uv,
>+void MergeUVRow_NEON(const uint8_t* src_u,
>+                     const uint8_t* src_v,
>+                     uint8_t* dst_uv,
>                      int width) {
>   asm volatile(
>       "1:                                        \n"
>@@ -534,10 +586,10 @@ void MergeUVRow_NEON(const uint8* src_u,
> }
> 
> // Reads 16 packed RGB and write to planar dst_r, dst_g, dst_b.
>-void SplitRGBRow_NEON(const uint8* src_rgb,
>-                      uint8* dst_r,
>-                      uint8* dst_g,
>-                      uint8* dst_b,
>+void SplitRGBRow_NEON(const uint8_t* src_rgb,
>+                      uint8_t* dst_r,
>+                      uint8_t* dst_g,
>+                      uint8_t* dst_b,
>                       int width) {
>   asm volatile(
>       "1:                                        \n"
>@@ -559,10 +611,10 @@ void SplitRGBRow_NEON(const uint8* src_rgb,
> }
> 
> // Reads 16 planar R's, G's and B's and writes out 16 packed RGB at a time
>-void MergeRGBRow_NEON(const uint8* src_r,
>-                      const uint8* src_g,
>-                      const uint8* src_b,
>-                      uint8* dst_rgb,
>+void MergeRGBRow_NEON(const uint8_t* src_r,
>+                      const uint8_t* src_g,
>+                      const uint8_t* src_b,
>+                      uint8_t* dst_rgb,
>                       int width) {
>   asm volatile(
>       "1:                                        \n"
>@@ -584,7 +636,7 @@ void MergeRGBRow_NEON(const uint8* src_r,
> }
> 
> // Copy multiple of 32.  vld4.8  allow unaligned and is fastest on a15.
>-void CopyRow_NEON(const uint8* src, uint8* dst, int count) {
>+void CopyRow_NEON(const uint8_t* src, uint8_t* dst, int width) {
>   asm volatile(
>       "1:                                        \n"
>       "vld1.8     {d0, d1, d2, d3}, [%0]!        \n"  // load 32
>@@ -593,14 +645,14 @@ void CopyRow_NEON(const uint8* src, uint8* dst, int count) {
>       "bgt        1b                             \n"
>       : "+r"(src),                  // %0
>         "+r"(dst),                  // %1
>-        "+r"(count)                 // %2  // Output registers
>+        "+r"(width)                 // %2  // Output registers
>       :                             // Input registers
>       : "cc", "memory", "q0", "q1"  // Clobber List
>       );
> }
> 
>-// SetRow writes 'count' bytes using an 8 bit value repeated.
>-void SetRow_NEON(uint8* dst, uint8 v8, int count) {
>+// SetRow writes 'width' bytes using an 8 bit value repeated.
>+void SetRow_NEON(uint8_t* dst, uint8_t v8, int width) {
>   asm volatile(
>       "vdup.8    q0, %2                          \n"  // duplicate 16 bytes
>       "1:                                        \n"
>@@ -608,13 +660,13 @@ void SetRow_NEON(uint8* dst, uint8 v8, int count) {
>       "vst1.8    {q0}, [%0]!                     \n"  // store
>       "bgt       1b                              \n"
>       : "+r"(dst),   // %0
>-        "+r"(count)  // %1
>+        "+r"(width)  // %1
>       : "r"(v8)      // %2
>       : "cc", "memory", "q0");
> }
> 
>-// ARGBSetRow writes 'count' pixels using an 32 bit value repeated.
>-void ARGBSetRow_NEON(uint8* dst, uint32 v32, int count) {
>+// ARGBSetRow writes 'width' pixels using an 32 bit value repeated.
>+void ARGBSetRow_NEON(uint8_t* dst, uint32_t v32, int width) {
>   asm volatile(
>       "vdup.u32  q0, %2                          \n"  // duplicate 4 ints
>       "1:                                        \n"
>@@ -622,12 +674,12 @@ void ARGBSetRow_NEON(uint8* dst, uint32 v32, int count) {
>       "vst1.8    {q0}, [%0]!                     \n"  // store
>       "bgt       1b                              \n"
>       : "+r"(dst),   // %0
>-        "+r"(count)  // %1
>+        "+r"(width)  // %1
>       : "r"(v32)     // %2
>       : "cc", "memory", "q0");
> }
> 
>-void MirrorRow_NEON(const uint8* src, uint8* dst, int width) {
>+void MirrorRow_NEON(const uint8_t* src, uint8_t* dst, int width) {
>   asm volatile(
>       // Start at end of source row.
>       "mov        r3, #-16                       \n"
>@@ -648,9 +700,9 @@ void MirrorRow_NEON(const uint8* src, uint8* dst, int width) {
>       : "cc", "memory", "r3", "q0");
> }
> 
>-void MirrorUVRow_NEON(const uint8* src_uv,
>-                      uint8* dst_u,
>-                      uint8* dst_v,
>+void MirrorUVRow_NEON(const uint8_t* src_uv,
>+                      uint8_t* dst_u,
>+                      uint8_t* dst_v,
>                       int width) {
>   asm volatile(
>       // Start at end of source row.
>@@ -673,7 +725,7 @@ void MirrorUVRow_NEON(const uint8* src_uv,
>       : "cc", "memory", "r12", "q0");
> }
> 
>-void ARGBMirrorRow_NEON(const uint8* src, uint8* dst, int width) {
>+void ARGBMirrorRow_NEON(const uint8_t* src, uint8_t* dst, int width) {
>   asm volatile(
>       // Start at end of source row.
>       "mov        r3, #-16                       \n"
>@@ -694,7 +746,9 @@ void ARGBMirrorRow_NEON(const uint8* src, uint8* dst, int width) {
>       : "cc", "memory", "r3", "q0");
> }
> 
>-void RGB24ToARGBRow_NEON(const uint8* src_rgb24, uint8* dst_argb, int width) {
>+void RGB24ToARGBRow_NEON(const uint8_t* src_rgb24,
>+                         uint8_t* dst_argb,
>+                         int width) {
>   asm volatile(
>       "vmov.u8    d4, #255                       \n"  // Alpha
>       "1:                                        \n"
>@@ -710,7 +764,7 @@ void RGB24ToARGBRow_NEON(const uint8* src_rgb24, uint8* dst_argb, int width) {
>       );
> }
> 
>-void RAWToARGBRow_NEON(const uint8* src_raw, uint8* dst_argb, int width) {
>+void RAWToARGBRow_NEON(const uint8_t* src_raw, uint8_t* dst_argb, int width) {
>   asm volatile(
>       "vmov.u8    d4, #255                       \n"  // Alpha
>       "1:                                        \n"
>@@ -727,7 +781,7 @@ void RAWToARGBRow_NEON(const uint8* src_raw, uint8* dst_argb, int width) {
>       );
> }
> 
>-void RAWToRGB24Row_NEON(const uint8* src_raw, uint8* dst_rgb24, int width) {
>+void RAWToRGB24Row_NEON(const uint8_t* src_raw, uint8_t* dst_rgb24, int width) {
>   asm volatile(
>       "1:                                        \n"
>       "vld3.8     {d1, d2, d3}, [%0]!            \n"  // load 8 pixels of RAW.
>@@ -756,7 +810,9 @@ void RAWToRGB24Row_NEON(const uint8* src_raw, uint8* dst_rgb24, int width) {
>   "vorr.u8    d2, d1, d5                     \n" /* R                    */ \
>   "vorr.u8    d1, d4, d6                     \n" /* G                    */
> 
>-void RGB565ToARGBRow_NEON(const uint8* src_rgb565, uint8* dst_argb, int width) {
>+void RGB565ToARGBRow_NEON(const uint8_t* src_rgb565,
>+                          uint8_t* dst_argb,
>+                          int width) {
>   asm volatile(
>       "vmov.u8    d3, #255                       \n"  // Alpha
>       "1:                                        \n"
>@@ -800,8 +856,8 @@ void RGB565ToARGBRow_NEON(const uint8* src_rgb565, uint8* dst_argb, int width) {
>   "vorr.u8    d2, d1, d5                     \n" /* R                    */ \
>   "vorr.u8    d1, d4, d6                     \n" /* G                    */
> 
>-void ARGB1555ToARGBRow_NEON(const uint8* src_argb1555,
>-                            uint8* dst_argb,
>+void ARGB1555ToARGBRow_NEON(const uint8_t* src_argb1555,
>+                            uint8_t* dst_argb,
>                             int width) {
>   asm volatile(
>       "vmov.u8    d3, #255                       \n"  // Alpha
>@@ -829,8 +885,8 @@ void ARGB1555ToARGBRow_NEON(const uint8* src_argb1555,
>   "vorr.u8    q1, q1, q2                     \n" /* G,A GGGGGGGG         */ \
>   "vswp.u8    d1, d2                         \n" /* B,R,G,A -> B,G,R,A   */
> 
>-void ARGB4444ToARGBRow_NEON(const uint8* src_argb4444,
>-                            uint8* dst_argb,
>+void ARGB4444ToARGBRow_NEON(const uint8_t* src_argb4444,
>+                            uint8_t* dst_argb,
>                             int width) {
>   asm volatile(
>       "vmov.u8    d3, #255                       \n"  // Alpha
>@@ -848,7 +904,9 @@ void ARGB4444ToARGBRow_NEON(const uint8* src_argb4444,
>       );
> }
> 
>-void ARGBToRGB24Row_NEON(const uint8* src_argb, uint8* dst_rgb24, int width) {
>+void ARGBToRGB24Row_NEON(const uint8_t* src_argb,
>+                         uint8_t* dst_rgb24,
>+                         int width) {
>   asm volatile(
>       "1:                                        \n"
>       "vld4.8     {d1, d2, d3, d4}, [%0]!        \n"  // load 8 pixels of ARGB.
>@@ -864,7 +922,7 @@ void ARGBToRGB24Row_NEON(const uint8* src_argb, uint8* dst_rgb24, int width) {
>       );
> }
> 
>-void ARGBToRAWRow_NEON(const uint8* src_argb, uint8* dst_raw, int width) {
>+void ARGBToRAWRow_NEON(const uint8_t* src_argb, uint8_t* dst_raw, int width) {
>   asm volatile(
>       "1:                                        \n"
>       "vld4.8     {d1, d2, d3, d4}, [%0]!        \n"  // load 8 pixels of ARGB.
>@@ -880,7 +938,7 @@ void ARGBToRAWRow_NEON(const uint8* src_argb, uint8* dst_raw, int width) {
>       );
> }
> 
>-void YUY2ToYRow_NEON(const uint8* src_yuy2, uint8* dst_y, int width) {
>+void YUY2ToYRow_NEON(const uint8_t* src_yuy2, uint8_t* dst_y, int width) {
>   asm volatile(
>       "1:                                        \n"
>       "vld2.8     {q0, q1}, [%0]!                \n"  // load 16 pixels of YUY2.
>@@ -895,7 +953,7 @@ void YUY2ToYRow_NEON(const uint8* src_yuy2, uint8* dst_y, int width) {
>       );
> }
> 
>-void UYVYToYRow_NEON(const uint8* src_uyvy, uint8* dst_y, int width) {
>+void UYVYToYRow_NEON(const uint8_t* src_uyvy, uint8_t* dst_y, int width) {
>   asm volatile(
>       "1:                                        \n"
>       "vld2.8     {q0, q1}, [%0]!                \n"  // load 16 pixels of UYVY.
>@@ -910,9 +968,9 @@ void UYVYToYRow_NEON(const uint8* src_uyvy, uint8* dst_y, int width) {
>       );
> }
> 
>-void YUY2ToUV422Row_NEON(const uint8* src_yuy2,
>-                         uint8* dst_u,
>-                         uint8* dst_v,
>+void YUY2ToUV422Row_NEON(const uint8_t* src_yuy2,
>+                         uint8_t* dst_u,
>+                         uint8_t* dst_v,
>                          int width) {
>   asm volatile(
>       "1:                                        \n"
>@@ -930,9 +988,9 @@ void YUY2ToUV422Row_NEON(const uint8* src_yuy2,
>       );
> }
> 
>-void UYVYToUV422Row_NEON(const uint8* src_uyvy,
>-                         uint8* dst_u,
>-                         uint8* dst_v,
>+void UYVYToUV422Row_NEON(const uint8_t* src_uyvy,
>+                         uint8_t* dst_u,
>+                         uint8_t* dst_v,
>                          int width) {
>   asm volatile(
>       "1:                                        \n"
>@@ -950,10 +1008,10 @@ void UYVYToUV422Row_NEON(const uint8* src_uyvy,
>       );
> }
> 
>-void YUY2ToUVRow_NEON(const uint8* src_yuy2,
>+void YUY2ToUVRow_NEON(const uint8_t* src_yuy2,
>                       int stride_yuy2,
>-                      uint8* dst_u,
>-                      uint8* dst_v,
>+                      uint8_t* dst_u,
>+                      uint8_t* dst_v,
>                       int width) {
>   asm volatile(
>       "add        %1, %0, %1                     \n"  // stride + src_yuy2
>@@ -977,10 +1035,10 @@ void YUY2ToUVRow_NEON(const uint8* src_yuy2,
>       );
> }
> 
>-void UYVYToUVRow_NEON(const uint8* src_uyvy,
>+void UYVYToUVRow_NEON(const uint8_t* src_uyvy,
>                       int stride_uyvy,
>-                      uint8* dst_u,
>-                      uint8* dst_v,
>+                      uint8_t* dst_u,
>+                      uint8_t* dst_v,
>                       int width) {
>   asm volatile(
>       "add        %1, %0, %1                     \n"  // stride + src_uyvy
>@@ -1005,9 +1063,9 @@ void UYVYToUVRow_NEON(const uint8* src_uyvy,
> }
> 
> // For BGRAToARGB, ABGRToARGB, RGBAToARGB, and ARGBToRGBA.
>-void ARGBShuffleRow_NEON(const uint8* src_argb,
>-                         uint8* dst_argb,
>-                         const uint8* shuffler,
>+void ARGBShuffleRow_NEON(const uint8_t* src_argb,
>+                         uint8_t* dst_argb,
>+                         const uint8_t* shuffler,
>                          int width) {
>   asm volatile(
>       "vld1.8     {q2}, [%3]                     \n"  // shuffler
>@@ -1026,10 +1084,10 @@ void ARGBShuffleRow_NEON(const uint8* src_argb,
>       );
> }
> 
>-void I422ToYUY2Row_NEON(const uint8* src_y,
>-                        const uint8* src_u,
>-                        const uint8* src_v,
>-                        uint8* dst_yuy2,
>+void I422ToYUY2Row_NEON(const uint8_t* src_y,
>+                        const uint8_t* src_u,
>+                        const uint8_t* src_v,
>+                        uint8_t* dst_yuy2,
>                         int width) {
>   asm volatile(
>       "1:                                        \n"
>@@ -1048,10 +1106,10 @@ void I422ToYUY2Row_NEON(const uint8* src_y,
>       : "cc", "memory", "d0", "d1", "d2", "d3");
> }
> 
>-void I422ToUYVYRow_NEON(const uint8* src_y,
>-                        const uint8* src_u,
>-                        const uint8* src_v,
>-                        uint8* dst_uyvy,
>+void I422ToUYVYRow_NEON(const uint8_t* src_y,
>+                        const uint8_t* src_u,
>+                        const uint8_t* src_v,
>+                        uint8_t* dst_uyvy,
>                         int width) {
>   asm volatile(
>       "1:                                        \n"
>@@ -1070,7 +1128,9 @@ void I422ToUYVYRow_NEON(const uint8* src_y,
>       : "cc", "memory", "d0", "d1", "d2", "d3");
> }
> 
>-void ARGBToRGB565Row_NEON(const uint8* src_argb, uint8* dst_rgb565, int width) {
>+void ARGBToRGB565Row_NEON(const uint8_t* src_argb,
>+                          uint8_t* dst_rgb565,
>+                          int width) {
>   asm volatile(
>       "1:                                        \n"
>       "vld4.8     {d20, d21, d22, d23}, [%0]!    \n"  // load 8 pixels of ARGB.
>@@ -1085,9 +1145,9 @@ void ARGBToRGB565Row_NEON(const uint8* src_argb, uint8* dst_rgb565, int width) {
>       : "cc", "memory", "q0", "q8", "q9", "q10", "q11");
> }
> 
>-void ARGBToRGB565DitherRow_NEON(const uint8* src_argb,
>-                                uint8* dst_rgb,
>-                                const uint32 dither4,
>+void ARGBToRGB565DitherRow_NEON(const uint8_t* src_argb,
>+                                uint8_t* dst_rgb,
>+                                const uint32_t dither4,
>                                 int width) {
>   asm volatile(
>       "vdup.32    d2, %2                         \n"  // dither4
>@@ -1107,8 +1167,8 @@ void ARGBToRGB565DitherRow_NEON(const uint8* src_argb,
>       : "cc", "memory", "q0", "q1", "q8", "q9", "q10", "q11");
> }
> 
>-void ARGBToARGB1555Row_NEON(const uint8* src_argb,
>-                            uint8* dst_argb1555,
>+void ARGBToARGB1555Row_NEON(const uint8_t* src_argb,
>+                            uint8_t* dst_argb1555,
>                             int width) {
>   asm volatile(
>       "1:                                        \n"
>@@ -1124,8 +1184,8 @@ void ARGBToARGB1555Row_NEON(const uint8* src_argb,
>       : "cc", "memory", "q0", "q8", "q9", "q10", "q11");
> }
> 
>-void ARGBToARGB4444Row_NEON(const uint8* src_argb,
>-                            uint8* dst_argb4444,
>+void ARGBToARGB4444Row_NEON(const uint8_t* src_argb,
>+                            uint8_t* dst_argb4444,
>                             int width) {
>   asm volatile(
>       "vmov.u8    d4, #0x0f                      \n"  // bits to clear with
>@@ -1143,7 +1203,7 @@ void ARGBToARGB4444Row_NEON(const uint8* src_argb,
>       : "cc", "memory", "q0", "q8", "q9", "q10", "q11");
> }
> 
>-void ARGBToYRow_NEON(const uint8* src_argb, uint8* dst_y, int width) {
>+void ARGBToYRow_NEON(const uint8_t* src_argb, uint8_t* dst_y, int width) {
>   asm volatile(
>       "vmov.u8    d24, #13                       \n"  // B * 0.1016 coefficient
>       "vmov.u8    d25, #65                       \n"  // G * 0.5078 coefficient
>@@ -1166,7 +1226,9 @@ void ARGBToYRow_NEON(const uint8* src_argb, uint8* dst_y, int width) {
>       : "cc", "memory", "q0", "q1", "q2", "q12", "q13");
> }
> 
>-void ARGBExtractAlphaRow_NEON(const uint8* src_argb, uint8* dst_a, int width) {
>+void ARGBExtractAlphaRow_NEON(const uint8_t* src_argb,
>+                              uint8_t* dst_a,
>+                              int width) {
>   asm volatile(
>       "1:                                        \n"
>       "vld4.8     {d0, d2, d4, d6}, [%0]!        \n"  // load 8 ARGB pixels
>@@ -1182,7 +1244,7 @@ void ARGBExtractAlphaRow_NEON(const uint8* src_argb, uint8* dst_a, int width) {
>       );
> }
> 
>-void ARGBToYJRow_NEON(const uint8* src_argb, uint8* dst_y, int width) {
>+void ARGBToYJRow_NEON(const uint8_t* src_argb, uint8_t* dst_y, int width) {
>   asm volatile(
>       "vmov.u8    d24, #15                       \n"  // B * 0.11400 coefficient
>       "vmov.u8    d25, #75                       \n"  // G * 0.58700 coefficient
>@@ -1204,9 +1266,9 @@ void ARGBToYJRow_NEON(const uint8* src_argb, uint8* dst_y, int width) {
> }
> 
> // 8x1 pixels.
>-void ARGBToUV444Row_NEON(const uint8* src_argb,
>-                         uint8* dst_u,
>-                         uint8* dst_v,
>+void ARGBToUV444Row_NEON(const uint8_t* src_argb,
>+                         uint8_t* dst_u,
>+                         uint8_t* dst_v,
>                          int width) {
>   asm volatile(
>       "vmov.u8    d24, #112                      \n"  // UB / VR 0.875
>@@ -1260,10 +1322,10 @@ void ARGBToUV444Row_NEON(const uint8* src_argb,
> // clang-format on
> 
> // TODO(fbarchard): Consider vhadd vertical, then vpaddl horizontal, avoid shr.
>-void ARGBToUVRow_NEON(const uint8* src_argb,
>+void ARGBToUVRow_NEON(const uint8_t* src_argb,
>                       int src_stride_argb,
>-                      uint8* dst_u,
>-                      uint8* dst_v,
>+                      uint8_t* dst_u,
>+                      uint8_t* dst_v,
>                       int width) {
>   asm volatile (
>     "add        %1, %0, %1                     \n"  // src_stride + src_argb
>@@ -1306,10 +1368,10 @@ void ARGBToUVRow_NEON(const uint8* src_argb,
> }
> 
> // TODO(fbarchard): Subsample match C code.
>-void ARGBToUVJRow_NEON(const uint8* src_argb,
>+void ARGBToUVJRow_NEON(const uint8_t* src_argb,
>                        int src_stride_argb,
>-                       uint8* dst_u,
>-                       uint8* dst_v,
>+                       uint8_t* dst_u,
>+                       uint8_t* dst_v,
>                        int width) {
>   asm volatile (
>     "add        %1, %0, %1                     \n"  // src_stride + src_argb
>@@ -1351,10 +1413,10 @@ void ARGBToUVJRow_NEON(const uint8* src_argb,
>   );
> }
> 
>-void BGRAToUVRow_NEON(const uint8* src_bgra,
>+void BGRAToUVRow_NEON(const uint8_t* src_bgra,
>                       int src_stride_bgra,
>-                      uint8* dst_u,
>-                      uint8* dst_v,
>+                      uint8_t* dst_u,
>+                      uint8_t* dst_v,
>                       int width) {
>   asm volatile (
>     "add        %1, %0, %1                     \n"  // src_stride + src_bgra
>@@ -1396,10 +1458,10 @@ void BGRAToUVRow_NEON(const uint8* src_bgra,
>   );
> }
> 
>-void ABGRToUVRow_NEON(const uint8* src_abgr,
>+void ABGRToUVRow_NEON(const uint8_t* src_abgr,
>                       int src_stride_abgr,
>-                      uint8* dst_u,
>-                      uint8* dst_v,
>+                      uint8_t* dst_u,
>+                      uint8_t* dst_v,
>                       int width) {
>   asm volatile (
>     "add        %1, %0, %1                     \n"  // src_stride + src_abgr
>@@ -1441,10 +1503,10 @@ void ABGRToUVRow_NEON(const uint8* src_abgr,
>   );
> }
> 
>-void RGBAToUVRow_NEON(const uint8* src_rgba,
>+void RGBAToUVRow_NEON(const uint8_t* src_rgba,
>                       int src_stride_rgba,
>-                      uint8* dst_u,
>-                      uint8* dst_v,
>+                      uint8_t* dst_u,
>+                      uint8_t* dst_v,
>                       int width) {
>   asm volatile (
>     "add        %1, %0, %1                     \n"  // src_stride + src_rgba
>@@ -1486,10 +1548,10 @@ void RGBAToUVRow_NEON(const uint8* src_rgba,
>   );
> }
> 
>-void RGB24ToUVRow_NEON(const uint8* src_rgb24,
>+void RGB24ToUVRow_NEON(const uint8_t* src_rgb24,
>                        int src_stride_rgb24,
>-                       uint8* dst_u,
>-                       uint8* dst_v,
>+                       uint8_t* dst_u,
>+                       uint8_t* dst_v,
>                        int width) {
>   asm volatile (
>     "add        %1, %0, %1                     \n"  // src_stride + src_rgb24
>@@ -1531,10 +1593,10 @@ void RGB24ToUVRow_NEON(const uint8* src_rgb24,
>   );
> }
> 
>-void RAWToUVRow_NEON(const uint8* src_raw,
>+void RAWToUVRow_NEON(const uint8_t* src_raw,
>                      int src_stride_raw,
>-                     uint8* dst_u,
>-                     uint8* dst_v,
>+                     uint8_t* dst_u,
>+                     uint8_t* dst_v,
>                      int width) {
>   asm volatile (
>     "add        %1, %0, %1                     \n"  // src_stride + src_raw
>@@ -1577,10 +1639,10 @@ void RAWToUVRow_NEON(const uint8* src_raw,
> }
> 
> // 16x2 pixels -> 8x1.  width is number of argb pixels. e.g. 16.
>-void RGB565ToUVRow_NEON(const uint8* src_rgb565,
>+void RGB565ToUVRow_NEON(const uint8_t* src_rgb565,
>                         int src_stride_rgb565,
>-                        uint8* dst_u,
>-                        uint8* dst_v,
>+                        uint8_t* dst_u,
>+                        uint8_t* dst_v,
>                         int width) {
>   asm volatile(
>       "add        %1, %0, %1                     \n"  // src_stride + src_argb
>@@ -1643,10 +1705,10 @@ void RGB565ToUVRow_NEON(const uint8* src_rgb565,
> }
> 
> // 16x2 pixels -> 8x1.  width is number of argb pixels. e.g. 16.
>-void ARGB1555ToUVRow_NEON(const uint8* src_argb1555,
>+void ARGB1555ToUVRow_NEON(const uint8_t* src_argb1555,
>                           int src_stride_argb1555,
>-                          uint8* dst_u,
>-                          uint8* dst_v,
>+                          uint8_t* dst_u,
>+                          uint8_t* dst_v,
>                           int width) {
>   asm volatile(
>       "add        %1, %0, %1                     \n"  // src_stride + src_argb
>@@ -1709,10 +1771,10 @@ void ARGB1555ToUVRow_NEON(const uint8* src_argb1555,
> }
> 
> // 16x2 pixels -> 8x1.  width is number of argb pixels. e.g. 16.
>-void ARGB4444ToUVRow_NEON(const uint8* src_argb4444,
>+void ARGB4444ToUVRow_NEON(const uint8_t* src_argb4444,
>                           int src_stride_argb4444,
>-                          uint8* dst_u,
>-                          uint8* dst_v,
>+                          uint8_t* dst_u,
>+                          uint8_t* dst_v,
>                           int width) {
>   asm volatile(
>       "add        %1, %0, %1                     \n"  // src_stride + src_argb
>@@ -1774,7 +1836,7 @@ void ARGB4444ToUVRow_NEON(const uint8* src_argb4444,
>         "q9", "q10", "q11", "q12", "q13", "q14", "q15");
> }
> 
>-void RGB565ToYRow_NEON(const uint8* src_rgb565, uint8* dst_y, int width) {
>+void RGB565ToYRow_NEON(const uint8_t* src_rgb565, uint8_t* dst_y, int width) {
>   asm volatile(
>       "vmov.u8    d24, #13                       \n"  // B * 0.1016 coefficient
>       "vmov.u8    d25, #65                       \n"  // G * 0.5078 coefficient
>@@ -1798,7 +1860,9 @@ void RGB565ToYRow_NEON(const uint8* src_rgb565, uint8* dst_y, int width) {
>       : "cc", "memory", "q0", "q1", "q2", "q3", "q12", "q13");
> }
> 
>-void ARGB1555ToYRow_NEON(const uint8* src_argb1555, uint8* dst_y, int width) {
>+void ARGB1555ToYRow_NEON(const uint8_t* src_argb1555,
>+                         uint8_t* dst_y,
>+                         int width) {
>   asm volatile(
>       "vmov.u8    d24, #13                       \n"  // B * 0.1016 coefficient
>       "vmov.u8    d25, #65                       \n"  // G * 0.5078 coefficient
>@@ -1822,7 +1886,9 @@ void ARGB1555ToYRow_NEON(const uint8* src_argb1555, uint8* dst_y, int width) {
>       : "cc", "memory", "q0", "q1", "q2", "q3", "q12", "q13");
> }
> 
>-void ARGB4444ToYRow_NEON(const uint8* src_argb4444, uint8* dst_y, int width) {
>+void ARGB4444ToYRow_NEON(const uint8_t* src_argb4444,
>+                         uint8_t* dst_y,
>+                         int width) {
>   asm volatile(
>       "vmov.u8    d24, #13                       \n"  // B * 0.1016 coefficient
>       "vmov.u8    d25, #65                       \n"  // G * 0.5078 coefficient
>@@ -1846,7 +1912,7 @@ void ARGB4444ToYRow_NEON(const uint8* src_argb4444, uint8* dst_y, int width) {
>       : "cc", "memory", "q0", "q1", "q2", "q3", "q12", "q13");
> }
> 
>-void BGRAToYRow_NEON(const uint8* src_bgra, uint8* dst_y, int width) {
>+void BGRAToYRow_NEON(const uint8_t* src_bgra, uint8_t* dst_y, int width) {
>   asm volatile(
>       "vmov.u8    d4, #33                        \n"  // R * 0.2578 coefficient
>       "vmov.u8    d5, #65                        \n"  // G * 0.5078 coefficient
>@@ -1869,7 +1935,7 @@ void BGRAToYRow_NEON(const uint8* src_bgra, uint8* dst_y, int width) {
>       : "cc", "memory", "d0", "d1", "d2", "d3", "d4", "d5", "d6", "d7", "q8");
> }
> 
>-void ABGRToYRow_NEON(const uint8* src_abgr, uint8* dst_y, int width) {
>+void ABGRToYRow_NEON(const uint8_t* src_abgr, uint8_t* dst_y, int width) {
>   asm volatile(
>       "vmov.u8    d4, #33                        \n"  // R * 0.2578 coefficient
>       "vmov.u8    d5, #65                        \n"  // G * 0.5078 coefficient
>@@ -1892,7 +1958,7 @@ void ABGRToYRow_NEON(const uint8* src_abgr, uint8* dst_y, int width) {
>       : "cc", "memory", "d0", "d1", "d2", "d3", "d4", "d5", "d6", "d7", "q8");
> }
> 
>-void RGBAToYRow_NEON(const uint8* src_rgba, uint8* dst_y, int width) {
>+void RGBAToYRow_NEON(const uint8_t* src_rgba, uint8_t* dst_y, int width) {
>   asm volatile(
>       "vmov.u8    d4, #13                        \n"  // B * 0.1016 coefficient
>       "vmov.u8    d5, #65                        \n"  // G * 0.5078 coefficient
>@@ -1915,7 +1981,7 @@ void RGBAToYRow_NEON(const uint8* src_rgba, uint8* dst_y, int width) {
>       : "cc", "memory", "d0", "d1", "d2", "d3", "d4", "d5", "d6", "d7", "q8");
> }
> 
>-void RGB24ToYRow_NEON(const uint8* src_rgb24, uint8* dst_y, int width) {
>+void RGB24ToYRow_NEON(const uint8_t* src_rgb24, uint8_t* dst_y, int width) {
>   asm volatile(
>       "vmov.u8    d4, #13                        \n"  // B * 0.1016 coefficient
>       "vmov.u8    d5, #65                        \n"  // G * 0.5078 coefficient
>@@ -1938,7 +2004,7 @@ void RGB24ToYRow_NEON(const uint8* src_rgb24, uint8* dst_y, int width) {
>       : "cc", "memory", "d0", "d1", "d2", "d3", "d4", "d5", "d6", "d7", "q8");
> }
> 
>-void RAWToYRow_NEON(const uint8* src_raw, uint8* dst_y, int width) {
>+void RAWToYRow_NEON(const uint8_t* src_raw, uint8_t* dst_y, int width) {
>   asm volatile(
>       "vmov.u8    d4, #33                        \n"  // R * 0.2578 coefficient
>       "vmov.u8    d5, #65                        \n"  // G * 0.5078 coefficient
>@@ -1962,8 +2028,8 @@ void RAWToYRow_NEON(const uint8* src_raw, uint8* dst_y, int width) {
> }
> 
> // Bilinear filter 16x2 -> 16x1
>-void InterpolateRow_NEON(uint8* dst_ptr,
>-                         const uint8* src_ptr,
>+void InterpolateRow_NEON(uint8_t* dst_ptr,
>+                         const uint8_t* src_ptr,
>                          ptrdiff_t src_stride,
>                          int dst_width,
>                          int source_y_fraction) {
>@@ -2021,9 +2087,9 @@ void InterpolateRow_NEON(uint8* dst_ptr,
> }
> 
> // dr * (256 - sa) / 256 + sr = dr - dr * sa / 256 + sr
>-void ARGBBlendRow_NEON(const uint8* src_argb0,
>-                       const uint8* src_argb1,
>-                       uint8* dst_argb,
>+void ARGBBlendRow_NEON(const uint8_t* src_argb0,
>+                       const uint8_t* src_argb1,
>+                       uint8_t* dst_argb,
>                        int width) {
>   asm volatile(
>       "subs       %3, #8                         \n"
>@@ -2081,7 +2147,9 @@ void ARGBBlendRow_NEON(const uint8* src_argb0,
> }
> 
> // Attenuate 8 pixels at a time.
>-void ARGBAttenuateRow_NEON(const uint8* src_argb, uint8* dst_argb, int width) {
>+void ARGBAttenuateRow_NEON(const uint8_t* src_argb,
>+                           uint8_t* dst_argb,
>+                           int width) {
>   asm volatile(
>       // Attenuate 8 pixels.
>       "1:                                        \n"
>@@ -2104,7 +2172,7 @@ void ARGBAttenuateRow_NEON(const uint8* src_argb, uint8* dst_argb, int width) {
> 
> // Quantize 8 ARGB pixels (32 bytes).
> // dst = (dst * scale >> 16) * interval_size + interval_offset;
>-void ARGBQuantizeRow_NEON(uint8* dst_argb,
>+void ARGBQuantizeRow_NEON(uint8_t* dst_argb,
>                           int scale,
>                           int interval_size,
>                           int interval_offset,
>@@ -2147,10 +2215,10 @@ void ARGBQuantizeRow_NEON(uint8* dst_argb,
> // Shade 8 pixels at a time by specified value.
> // NOTE vqrdmulh.s16 q10, q10, d0[0] must use a scaler register from 0 to 8.
> // Rounding in vqrdmulh does +1 to high if high bit of low s16 is set.
>-void ARGBShadeRow_NEON(const uint8* src_argb,
>-                       uint8* dst_argb,
>+void ARGBShadeRow_NEON(const uint8_t* src_argb,
>+                       uint8_t* dst_argb,
>                        int width,
>-                       uint32 value) {
>+                       uint32_t value) {
>   asm volatile(
>       "vdup.u32   q0, %3                         \n"  // duplicate scale value.
>       "vzip.u8    d0, d1                         \n"  // d0 aarrggbb.
>@@ -2184,7 +2252,7 @@ void ARGBShadeRow_NEON(const uint8* src_argb,
> // Convert 8 ARGB pixels (64 bytes) to 8 Gray ARGB pixels
> // Similar to ARGBToYJ but stores ARGB.
> // C code is (15 * b + 75 * g + 38 * r + 64) >> 7;
>-void ARGBGrayRow_NEON(const uint8* src_argb, uint8* dst_argb, int width) {
>+void ARGBGrayRow_NEON(const uint8_t* src_argb, uint8_t* dst_argb, int width) {
>   asm volatile(
>       "vmov.u8    d24, #15                       \n"  // B * 0.11400 coefficient
>       "vmov.u8    d25, #75                       \n"  // G * 0.58700 coefficient
>@@ -2211,7 +2279,7 @@ void ARGBGrayRow_NEON(const uint8* src_argb, uint8* dst_argb, int width) {
> //    b = (r * 35 + g * 68 + b * 17) >> 7
> //    g = (r * 45 + g * 88 + b * 22) >> 7
> //    r = (r * 50 + g * 98 + b * 24) >> 7
>-void ARGBSepiaRow_NEON(uint8* dst_argb, int width) {
>+void ARGBSepiaRow_NEON(uint8_t* dst_argb, int width) {
>   asm volatile(
>       "vmov.u8    d20, #17                       \n"  // BB coefficient
>       "vmov.u8    d21, #68                       \n"  // BG coefficient
>@@ -2249,9 +2317,9 @@ void ARGBSepiaRow_NEON(uint8* dst_argb, int width) {
> // Tranform 8 ARGB pixels (32 bytes) with color matrix.
> // TODO(fbarchard): Was same as Sepia except matrix is provided.  This function
> // needs to saturate.  Consider doing a non-saturating version.
>-void ARGBColorMatrixRow_NEON(const uint8* src_argb,
>-                             uint8* dst_argb,
>-                             const int8* matrix_argb,
>+void ARGBColorMatrixRow_NEON(const uint8_t* src_argb,
>+                             uint8_t* dst_argb,
>+                             const int8_t* matrix_argb,
>                              int width) {
>   asm volatile(
>       "vld1.8     {q2}, [%3]                     \n"  // load 3 ARGB vectors.
>@@ -2308,9 +2376,9 @@ void ARGBColorMatrixRow_NEON(const uint8* src_argb,
> }
> 
> // Multiply 2 rows of ARGB pixels together, 8 pixels at a time.
>-void ARGBMultiplyRow_NEON(const uint8* src_argb0,
>-                          const uint8* src_argb1,
>-                          uint8* dst_argb,
>+void ARGBMultiplyRow_NEON(const uint8_t* src_argb0,
>+                          const uint8_t* src_argb1,
>+                          uint8_t* dst_argb,
>                           int width) {
>   asm volatile(
>       // 8 pixel loop.
>@@ -2337,9 +2405,9 @@ void ARGBMultiplyRow_NEON(const uint8* src_argb0,
> }
> 
> // Add 2 rows of ARGB pixels together, 8 pixels at a time.
>-void ARGBAddRow_NEON(const uint8* src_argb0,
>-                     const uint8* src_argb1,
>-                     uint8* dst_argb,
>+void ARGBAddRow_NEON(const uint8_t* src_argb0,
>+                     const uint8_t* src_argb1,
>+                     uint8_t* dst_argb,
>                      int width) {
>   asm volatile(
>       // 8 pixel loop.
>@@ -2360,9 +2428,9 @@ void ARGBAddRow_NEON(const uint8* src_argb0,
> }
> 
> // Subtract 2 rows of ARGB pixels, 8 pixels at a time.
>-void ARGBSubtractRow_NEON(const uint8* src_argb0,
>-                          const uint8* src_argb1,
>-                          uint8* dst_argb,
>+void ARGBSubtractRow_NEON(const uint8_t* src_argb0,
>+                          const uint8_t* src_argb1,
>+                          uint8_t* dst_argb,
>                           int width) {
>   asm volatile(
>       // 8 pixel loop.
>@@ -2387,9 +2455,9 @@ void ARGBSubtractRow_NEON(const uint8* src_argb0,
> // R = Sobel
> // G = Sobel
> // B = Sobel
>-void SobelRow_NEON(const uint8* src_sobelx,
>-                   const uint8* src_sobely,
>-                   uint8* dst_argb,
>+void SobelRow_NEON(const uint8_t* src_sobelx,
>+                   const uint8_t* src_sobely,
>+                   uint8_t* dst_argb,
>                    int width) {
>   asm volatile(
>       "vmov.u8    d3, #255                       \n"  // alpha
>@@ -2412,9 +2480,9 @@ void SobelRow_NEON(const uint8* src_sobelx,
> }
> 
> // Adds Sobel X and Sobel Y and stores Sobel into plane.
>-void SobelToPlaneRow_NEON(const uint8* src_sobelx,
>-                          const uint8* src_sobely,
>-                          uint8* dst_y,
>+void SobelToPlaneRow_NEON(const uint8_t* src_sobelx,
>+                          const uint8_t* src_sobely,
>+                          uint8_t* dst_y,
>                           int width) {
>   asm volatile(
>       // 16 pixel loop.
>@@ -2438,9 +2506,9 @@ void SobelToPlaneRow_NEON(const uint8* src_sobelx,
> // R = Sobel X
> // G = Sobel
> // B = Sobel Y
>-void SobelXYRow_NEON(const uint8* src_sobelx,
>-                     const uint8* src_sobely,
>-                     uint8* dst_argb,
>+void SobelXYRow_NEON(const uint8_t* src_sobelx,
>+                     const uint8_t* src_sobely,
>+                     uint8_t* dst_argb,
>                      int width) {
>   asm volatile(
>       "vmov.u8    d3, #255                       \n"  // alpha
>@@ -2464,10 +2532,10 @@ void SobelXYRow_NEON(const uint8* src_sobelx,
> // -1  0  1
> // -2  0  2
> // -1  0  1
>-void SobelXRow_NEON(const uint8* src_y0,
>-                    const uint8* src_y1,
>-                    const uint8* src_y2,
>-                    uint8* dst_sobelx,
>+void SobelXRow_NEON(const uint8_t* src_y0,
>+                    const uint8_t* src_y1,
>+                    const uint8_t* src_y2,
>+                    uint8_t* dst_sobelx,
>                     int width) {
>   asm volatile(
>       "1:                                        \n"
>@@ -2503,9 +2571,9 @@ void SobelXRow_NEON(const uint8* src_y0,
> // -1 -2 -1
> //  0  0  0
> //  1  2  1
>-void SobelYRow_NEON(const uint8* src_y0,
>-                    const uint8* src_y1,
>-                    uint8* dst_sobely,
>+void SobelYRow_NEON(const uint8_t* src_y0,
>+                    const uint8_t* src_y1,
>+                    uint8_t* dst_sobely,
>                     int width) {
>   asm volatile(
>       "1:                                        \n"
>@@ -2536,7 +2604,10 @@ void SobelYRow_NEON(const uint8* src_y0,
>       );
> }
> 
>-void HalfFloat1Row_NEON(const uint16* src, uint16* dst, float, int width) {
>+void HalfFloat1Row_NEON(const uint16_t* src,
>+                        uint16_t* dst,
>+                        float /*unused*/,
>+                        int width) {
>   asm volatile(
>       "vdup.32    q0, %3                         \n"
> 
>@@ -2561,7 +2632,10 @@ void HalfFloat1Row_NEON(const uint16* src, uint16* dst, float, int width) {
> }
> 
> // TODO(fbarchard): multiply by element.
>-void HalfFloatRow_NEON(const uint16* src, uint16* dst, float scale, int width) {
>+void HalfFloatRow_NEON(const uint16_t* src,
>+                       uint16_t* dst,
>+                       float scale,
>+                       int width) {
>   asm volatile(
>       "vdup.32    q0, %3                         \n"
> 
>@@ -2585,6 +2659,32 @@ void HalfFloatRow_NEON(const uint16* src, uint16* dst, float scale, int width) {
>       : "cc", "memory", "q0", "q1", "q2", "q3");
> }
> 
>+void ByteToFloatRow_NEON(const uint8_t* src,
>+                         float* dst,
>+                         float scale,
>+                         int width) {
>+  asm volatile(
>+      "vdup.32    q0, %3                         \n"
>+
>+      "1:                                        \n"
>+      "vld1.8     {d2}, [%0]!                    \n"  // load 8 bytes
>+      "subs       %2, %2, #8                     \n"  // 8 pixels per loop
>+      "vmovl.u8   q1, d2                         \n"  // 8 shorts
>+      "vmovl.u16  q2, d2                         \n"  // 8 ints
>+      "vmovl.u16  q3, d3                         \n"
>+      "vcvt.f32.u32  q2, q2                      \n"  // 8 floats
>+      "vcvt.f32.u32  q3, q3                      \n"
>+      "vmul.f32   q2, q2, d0[0]                  \n"  // scale
>+      "vmul.f32   q3, q3, d0[0]                  \n"
>+      "vst1.8     {q2, q3}, [%1]!                \n"  // store 8 floats
>+      "bgt        1b                             \n"
>+      : "+r"(src),   // %0
>+        "+r"(dst),   // %1
>+        "+r"(width)  // %2
>+      : "r"(scale)   // %3
>+      : "cc", "memory", "q0", "q1", "q2", "q3");
>+}
>+
> #endif  // !defined(LIBYUV_DISABLE_NEON) && defined(__ARM_NEON__)..
> 
> #ifdef __cplusplus
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/row_neon64.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/row_neon64.cc
>index 5616d8a5b5f..24b4520babc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/row_neon64.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/row_neon64.cc
>@@ -112,10 +112,10 @@ extern "C" {
>   ".8h, #6     \n"                               /* G */            \
>   "sqshrun    " #vR ".8b, " #vR ".8h, #6     \n" /* R */
> 
>-void I444ToARGBRow_NEON(const uint8* src_y,
>-                        const uint8* src_u,
>-                        const uint8* src_v,
>-                        uint8* dst_argb,
>+void I444ToARGBRow_NEON(const uint8_t* src_y,
>+                        const uint8_t* src_u,
>+                        const uint8_t* src_v,
>+                        uint8_t* dst_argb,
>                         const struct YuvConstants* yuvconstants,
>                         int width) {
>   asm volatile (
>@@ -141,10 +141,10 @@ void I444ToARGBRow_NEON(const uint8* src_y,
>   );
> }
> 
>-void I422ToARGBRow_NEON(const uint8* src_y,
>-                        const uint8* src_u,
>-                        const uint8* src_v,
>-                        uint8* dst_argb,
>+void I422ToARGBRow_NEON(const uint8_t* src_y,
>+                        const uint8_t* src_u,
>+                        const uint8_t* src_v,
>+                        uint8_t* dst_argb,
>                         const struct YuvConstants* yuvconstants,
>                         int width) {
>   asm volatile (
>@@ -170,11 +170,11 @@ void I422ToARGBRow_NEON(const uint8* src_y,
>   );
> }
> 
>-void I422AlphaToARGBRow_NEON(const uint8* src_y,
>-                             const uint8* src_u,
>-                             const uint8* src_v,
>-                             const uint8* src_a,
>-                             uint8* dst_argb,
>+void I422AlphaToARGBRow_NEON(const uint8_t* src_y,
>+                             const uint8_t* src_u,
>+                             const uint8_t* src_v,
>+                             const uint8_t* src_a,
>+                             uint8_t* dst_argb,
>                              const struct YuvConstants* yuvconstants,
>                              int width) {
>   asm volatile (
>@@ -201,10 +201,10 @@ void I422AlphaToARGBRow_NEON(const uint8* src_y,
>   );
> }
> 
>-void I422ToRGBARow_NEON(const uint8* src_y,
>-                        const uint8* src_u,
>-                        const uint8* src_v,
>-                        uint8* dst_rgba,
>+void I422ToRGBARow_NEON(const uint8_t* src_y,
>+                        const uint8_t* src_u,
>+                        const uint8_t* src_v,
>+                        uint8_t* dst_rgba,
>                         const struct YuvConstants* yuvconstants,
>                         int width) {
>   asm volatile (
>@@ -230,10 +230,10 @@ void I422ToRGBARow_NEON(const uint8* src_y,
>   );
> }
> 
>-void I422ToRGB24Row_NEON(const uint8* src_y,
>-                         const uint8* src_u,
>-                         const uint8* src_v,
>-                         uint8* dst_rgb24,
>+void I422ToRGB24Row_NEON(const uint8_t* src_y,
>+                         const uint8_t* src_u,
>+                         const uint8_t* src_v,
>+                         uint8_t* dst_rgb24,
>                          const struct YuvConstants* yuvconstants,
>                          int width) {
>   asm volatile (
>@@ -265,10 +265,10 @@ void I422ToRGB24Row_NEON(const uint8* src_y,
>   "sri        v0.8h,  v21.8h, #5             \n" /* RG                   */ \
>   "sri        v0.8h,  v20.8h, #11            \n" /* RGB                  */
> 
>-void I422ToRGB565Row_NEON(const uint8* src_y,
>-                          const uint8* src_u,
>-                          const uint8* src_v,
>-                          uint8* dst_rgb565,
>+void I422ToRGB565Row_NEON(const uint8_t* src_y,
>+                          const uint8_t* src_u,
>+                          const uint8_t* src_v,
>+                          uint8_t* dst_rgb565,
>                           const struct YuvConstants* yuvconstants,
>                           int width) {
>   asm volatile(
>@@ -301,10 +301,10 @@ void I422ToRGB565Row_NEON(const uint8* src_y,
>   "sri        v0.8h,  v21.8h, #6             \n" /* ARG                  */ \
>   "sri        v0.8h,  v20.8h, #11            \n" /* ARGB                 */
> 
>-void I422ToARGB1555Row_NEON(const uint8* src_y,
>-                            const uint8* src_u,
>-                            const uint8* src_v,
>-                            uint8* dst_argb1555,
>+void I422ToARGB1555Row_NEON(const uint8_t* src_y,
>+                            const uint8_t* src_u,
>+                            const uint8_t* src_v,
>+                            uint8_t* dst_argb1555,
>                             const struct YuvConstants* yuvconstants,
>                             int width) {
>   asm volatile(
>@@ -339,10 +339,10 @@ void I422ToARGB1555Row_NEON(const uint8* src_y,
>   "orr        v1.8b,  v22.8b, v23.8b         \n" /* RA                   */  \
>   "zip1       v0.16b, v0.16b, v1.16b         \n" /* BGRA                 */
> 
>-void I422ToARGB4444Row_NEON(const uint8* src_y,
>-                            const uint8* src_u,
>-                            const uint8* src_v,
>-                            uint8* dst_argb4444,
>+void I422ToARGB4444Row_NEON(const uint8_t* src_y,
>+                            const uint8_t* src_u,
>+                            const uint8_t* src_v,
>+                            uint8_t* dst_argb4444,
>                             const struct YuvConstants* yuvconstants,
>                             int width) {
>   asm volatile (
>@@ -370,7 +370,7 @@ void I422ToARGB4444Row_NEON(const uint8* src_y,
>   );
> }
> 
>-void I400ToARGBRow_NEON(const uint8* src_y, uint8* dst_argb, int width) {
>+void I400ToARGBRow_NEON(const uint8_t* src_y, uint8_t* dst_argb, int width) {
>   asm volatile (
>     YUVTORGB_SETUP
>     "movi       v23.8b, #255                   \n"
>@@ -392,7 +392,7 @@ void I400ToARGBRow_NEON(const uint8* src_y, uint8* dst_argb, int width) {
>   );
> }
> 
>-void J400ToARGBRow_NEON(const uint8* src_y, uint8* dst_argb, int width) {
>+void J400ToARGBRow_NEON(const uint8_t* src_y, uint8_t* dst_argb, int width) {
>   asm volatile(
>       "movi       v23.8b, #255                   \n"
>       "1:                                        \n"
>@@ -409,9 +409,9 @@ void J400ToARGBRow_NEON(const uint8* src_y, uint8* dst_argb, int width) {
>       : "cc", "memory", "v20", "v21", "v22", "v23");
> }
> 
>-void NV12ToARGBRow_NEON(const uint8* src_y,
>-                        const uint8* src_uv,
>-                        uint8* dst_argb,
>+void NV12ToARGBRow_NEON(const uint8_t* src_y,
>+                        const uint8_t* src_uv,
>+                        uint8_t* dst_argb,
>                         const struct YuvConstants* yuvconstants,
>                         int width) {
>   asm volatile (
>@@ -436,9 +436,9 @@ void NV12ToARGBRow_NEON(const uint8* src_y,
>   );
> }
> 
>-void NV21ToARGBRow_NEON(const uint8* src_y,
>-                        const uint8* src_vu,
>-                        uint8* dst_argb,
>+void NV21ToARGBRow_NEON(const uint8_t* src_y,
>+                        const uint8_t* src_vu,
>+                        uint8_t* dst_argb,
>                         const struct YuvConstants* yuvconstants,
>                         int width) {
>   asm volatile (
>@@ -463,9 +463,61 @@ void NV21ToARGBRow_NEON(const uint8* src_y,
>   );
> }
> 
>-void NV12ToRGB565Row_NEON(const uint8* src_y,
>-                          const uint8* src_uv,
>-                          uint8* dst_rgb565,
>+void NV12ToRGB24Row_NEON(const uint8_t* src_y,
>+                         const uint8_t* src_uv,
>+                         uint8_t* dst_rgb24,
>+                         const struct YuvConstants* yuvconstants,
>+                         int width) {
>+  asm volatile (
>+    YUVTORGB_SETUP
>+  "1:                                          \n"
>+    READNV12
>+    YUVTORGB(v22, v21, v20)
>+    "subs       %w3, %w3, #8                   \n"
>+    "st3        {v20.8b,v21.8b,v22.8b}, [%2], #24     \n"
>+    "b.gt       1b                             \n"
>+    : "+r"(src_y),     // %0
>+      "+r"(src_uv),    // %1
>+      "+r"(dst_rgb24),  // %2
>+      "+r"(width)      // %3
>+    : [kUVToRB]"r"(&yuvconstants->kUVToRB),
>+      [kUVToG]"r"(&yuvconstants->kUVToG),
>+      [kUVBiasBGR]"r"(&yuvconstants->kUVBiasBGR),
>+      [kYToRgb]"r"(&yuvconstants->kYToRgb)
>+    : "cc", "memory", "v0", "v1", "v2", "v3", "v4", "v5", "v6", "v7", "v20",
>+      "v21", "v22", "v23", "v24", "v25", "v26", "v27", "v28", "v29", "v30"
>+  );
>+}
>+
>+void NV21ToRGB24Row_NEON(const uint8_t* src_y,
>+                         const uint8_t* src_vu,
>+                         uint8_t* dst_rgb24,
>+                         const struct YuvConstants* yuvconstants,
>+                         int width) {
>+  asm volatile (
>+    YUVTORGB_SETUP
>+  "1:                                          \n"
>+    READNV21
>+    YUVTORGB(v22, v21, v20)
>+    "subs       %w3, %w3, #8                   \n"
>+    "st3        {v20.8b,v21.8b,v22.8b}, [%2], #24     \n"
>+    "b.gt       1b                             \n"
>+    : "+r"(src_y),     // %0
>+      "+r"(src_vu),    // %1
>+      "+r"(dst_rgb24),  // %2
>+      "+r"(width)      // %3
>+    : [kUVToRB]"r"(&yuvconstants->kUVToRB),
>+      [kUVToG]"r"(&yuvconstants->kUVToG),
>+      [kUVBiasBGR]"r"(&yuvconstants->kUVBiasBGR),
>+      [kYToRgb]"r"(&yuvconstants->kYToRgb)
>+    : "cc", "memory", "v0", "v1", "v2", "v3", "v4", "v5", "v6", "v7", "v20",
>+      "v21", "v22", "v23", "v24", "v25", "v26", "v27", "v28", "v29", "v30"
>+  );
>+}
>+
>+void NV12ToRGB565Row_NEON(const uint8_t* src_y,
>+                          const uint8_t* src_uv,
>+                          uint8_t* dst_rgb565,
>                           const struct YuvConstants* yuvconstants,
>                           int width) {
>   asm volatile(
>@@ -488,8 +540,8 @@ void NV12ToRGB565Row_NEON(const uint8* src_y,
>         "v21", "v22", "v23", "v24", "v25", "v26", "v27", "v28", "v29", "v30");
> }
> 
>-void YUY2ToARGBRow_NEON(const uint8* src_yuy2,
>-                        uint8* dst_argb,
>+void YUY2ToARGBRow_NEON(const uint8_t* src_yuy2,
>+                        uint8_t* dst_argb,
>                         const struct YuvConstants* yuvconstants,
>                         int width) {
>   asm volatile (
>@@ -513,8 +565,8 @@ void YUY2ToARGBRow_NEON(const uint8* src_yuy2,
>   );
> }
> 
>-void UYVYToARGBRow_NEON(const uint8* src_uyvy,
>-                        uint8* dst_argb,
>+void UYVYToARGBRow_NEON(const uint8_t* src_uyvy,
>+                        uint8_t* dst_argb,
>                         const struct YuvConstants* yuvconstants,
>                         int width) {
>   asm volatile (
>@@ -539,9 +591,9 @@ void UYVYToARGBRow_NEON(const uint8* src_uyvy,
> }
> 
> // Reads 16 pairs of UV and write even values to dst_u and odd to dst_v.
>-void SplitUVRow_NEON(const uint8* src_uv,
>-                     uint8* dst_u,
>-                     uint8* dst_v,
>+void SplitUVRow_NEON(const uint8_t* src_uv,
>+                     uint8_t* dst_u,
>+                     uint8_t* dst_v,
>                      int width) {
>   asm volatile(
>       "1:                                        \n"
>@@ -560,9 +612,9 @@ void SplitUVRow_NEON(const uint8* src_uv,
> }
> 
> // Reads 16 U's and V's and writes out 16 pairs of UV.
>-void MergeUVRow_NEON(const uint8* src_u,
>-                     const uint8* src_v,
>-                     uint8* dst_uv,
>+void MergeUVRow_NEON(const uint8_t* src_u,
>+                     const uint8_t* src_v,
>+                     uint8_t* dst_uv,
>                      int width) {
>   asm volatile(
>       "1:                                        \n"
>@@ -581,10 +633,10 @@ void MergeUVRow_NEON(const uint8* src_u,
> }
> 
> // Reads 16 packed RGB and write to planar dst_r, dst_g, dst_b.
>-void SplitRGBRow_NEON(const uint8* src_rgb,
>-                      uint8* dst_r,
>-                      uint8* dst_g,
>-                      uint8* dst_b,
>+void SplitRGBRow_NEON(const uint8_t* src_rgb,
>+                      uint8_t* dst_r,
>+                      uint8_t* dst_g,
>+                      uint8_t* dst_b,
>                       int width) {
>   asm volatile(
>       "1:                                        \n"
>@@ -605,10 +657,10 @@ void SplitRGBRow_NEON(const uint8* src_rgb,
> }
> 
> // Reads 16 planar R's, G's and B's and writes out 16 packed RGB at a time
>-void MergeRGBRow_NEON(const uint8* src_r,
>-                      const uint8* src_g,
>-                      const uint8* src_b,
>-                      uint8* dst_rgb,
>+void MergeRGBRow_NEON(const uint8_t* src_r,
>+                      const uint8_t* src_g,
>+                      const uint8_t* src_b,
>+                      uint8_t* dst_rgb,
>                       int width) {
>   asm volatile(
>       "1:                                        \n"
>@@ -629,7 +681,7 @@ void MergeRGBRow_NEON(const uint8* src_r,
> }
> 
> // Copy multiple of 32.
>-void CopyRow_NEON(const uint8* src, uint8* dst, int count) {
>+void CopyRow_NEON(const uint8_t* src, uint8_t* dst, int width) {
>   asm volatile(
>       "1:                                        \n"
>       "ldp        q0, q1, [%0], #32              \n"
>@@ -638,14 +690,14 @@ void CopyRow_NEON(const uint8* src, uint8* dst, int count) {
>       "b.gt       1b                             \n"
>       : "+r"(src),                  // %0
>         "+r"(dst),                  // %1
>-        "+r"(count)                 // %2  // Output registers
>+        "+r"(width)                 // %2  // Output registers
>       :                             // Input registers
>       : "cc", "memory", "v0", "v1"  // Clobber List
>       );
> }
> 
>-// SetRow writes 'count' bytes using an 8 bit value repeated.
>-void SetRow_NEON(uint8* dst, uint8 v8, int count) {
>+// SetRow writes 'width' bytes using an 8 bit value repeated.
>+void SetRow_NEON(uint8_t* dst, uint8_t v8, int width) {
>   asm volatile(
>       "dup        v0.16b, %w2                    \n"  // duplicate 16 bytes
>       "1:                                        \n"
>@@ -653,12 +705,12 @@ void SetRow_NEON(uint8* dst, uint8 v8, int count) {
>       "st1        {v0.16b}, [%0], #16            \n"  // store
>       "b.gt       1b                             \n"
>       : "+r"(dst),   // %0
>-        "+r"(count)  // %1
>+        "+r"(width)  // %1
>       : "r"(v8)      // %2
>       : "cc", "memory", "v0");
> }
> 
>-void ARGBSetRow_NEON(uint8* dst, uint32 v32, int count) {
>+void ARGBSetRow_NEON(uint8_t* dst, uint32_t v32, int width) {
>   asm volatile(
>       "dup        v0.4s, %w2                     \n"  // duplicate 4 ints
>       "1:                                        \n"
>@@ -666,12 +718,12 @@ void ARGBSetRow_NEON(uint8* dst, uint32 v32, int count) {
>       "st1        {v0.16b}, [%0], #16            \n"  // store
>       "b.gt       1b                             \n"
>       : "+r"(dst),   // %0
>-        "+r"(count)  // %1
>+        "+r"(width)  // %1
>       : "r"(v32)     // %2
>       : "cc", "memory", "v0");
> }
> 
>-void MirrorRow_NEON(const uint8* src, uint8* dst, int width) {
>+void MirrorRow_NEON(const uint8_t* src, uint8_t* dst, int width) {
>   asm volatile(
>       // Start at end of source row.
>       "add        %0, %0, %w2, sxtw              \n"
>@@ -690,9 +742,9 @@ void MirrorRow_NEON(const uint8* src, uint8* dst, int width) {
>       : "cc", "memory", "v0");
> }
> 
>-void MirrorUVRow_NEON(const uint8* src_uv,
>-                      uint8* dst_u,
>-                      uint8* dst_v,
>+void MirrorUVRow_NEON(const uint8_t* src_uv,
>+                      uint8_t* dst_u,
>+                      uint8_t* dst_v,
>                       int width) {
>   asm volatile(
>       // Start at end of source row.
>@@ -714,7 +766,7 @@ void MirrorUVRow_NEON(const uint8* src_uv,
>       : "cc", "memory", "v0", "v1");
> }
> 
>-void ARGBMirrorRow_NEON(const uint8* src, uint8* dst, int width) {
>+void ARGBMirrorRow_NEON(const uint8_t* src, uint8_t* dst, int width) {
>   asm volatile(
>       // Start at end of source row.
>       "add        %0, %0, %w2, sxtw #2           \n"
>@@ -733,7 +785,9 @@ void ARGBMirrorRow_NEON(const uint8* src, uint8* dst, int width) {
>       : "cc", "memory", "v0");
> }
> 
>-void RGB24ToARGBRow_NEON(const uint8* src_rgb24, uint8* dst_argb, int width) {
>+void RGB24ToARGBRow_NEON(const uint8_t* src_rgb24,
>+                         uint8_t* dst_argb,
>+                         int width) {
>   asm volatile(
>       "movi       v4.8b, #255                    \n"  // Alpha
>       "1:                                        \n"
>@@ -749,7 +803,7 @@ void RGB24ToARGBRow_NEON(const uint8* src_rgb24, uint8* dst_argb, int width) {
>       );
> }
> 
>-void RAWToARGBRow_NEON(const uint8* src_raw, uint8* dst_argb, int width) {
>+void RAWToARGBRow_NEON(const uint8_t* src_raw, uint8_t* dst_argb, int width) {
>   asm volatile(
>       "movi       v5.8b, #255                    \n"  // Alpha
>       "1:                                        \n"
>@@ -767,7 +821,7 @@ void RAWToARGBRow_NEON(const uint8* src_raw, uint8* dst_argb, int width) {
>       );
> }
> 
>-void RAWToRGB24Row_NEON(const uint8* src_raw, uint8* dst_rgb24, int width) {
>+void RAWToRGB24Row_NEON(const uint8_t* src_raw, uint8_t* dst_rgb24, int width) {
>   asm volatile(
>       "1:                                        \n"
>       "ld3        {v0.8b,v1.8b,v2.8b}, [%0], #24 \n"  // read r g b
>@@ -797,7 +851,9 @@ void RAWToRGB24Row_NEON(const uint8* src_raw, uint8* dst_rgb24, int width) {
>   "orr        v0.16b, v0.16b, v2.16b         \n" /* R,B                  */ \
>   "dup        v2.2D, v0.D[1]                 \n" /* R                    */
> 
>-void RGB565ToARGBRow_NEON(const uint8* src_rgb565, uint8* dst_argb, int width) {
>+void RGB565ToARGBRow_NEON(const uint8_t* src_rgb565,
>+                          uint8_t* dst_argb,
>+                          int width) {
>   asm volatile(
>       "movi       v3.8b, #255                    \n"  // Alpha
>       "1:                                        \n"
>@@ -851,8 +907,8 @@ void RGB565ToARGBRow_NEON(const uint8* src_rgb565, uint8* dst_argb, int width) {
>   "orr        v2.16b, v1.16b, v3.16b         \n" /* R                    */ \
>   "dup        v1.2D, v0.D[1]                 \n" /* G */
> 
>-void ARGB1555ToARGBRow_NEON(const uint8* src_argb1555,
>-                            uint8* dst_argb,
>+void ARGB1555ToARGBRow_NEON(const uint8_t* src_argb1555,
>+                            uint8_t* dst_argb,
>                             int width) {
>   asm volatile(
>       "movi       v3.8b, #255                    \n"  // Alpha
>@@ -883,8 +939,8 @@ void ARGB1555ToARGBRow_NEON(const uint8* src_argb1555,
>   "dup        v0.2D, v2.D[1]                 \n"                            \
>   "dup        v1.2D, v3.D[1]                 \n"
> 
>-void ARGB4444ToARGBRow_NEON(const uint8* src_argb4444,
>-                            uint8* dst_argb,
>+void ARGB4444ToARGBRow_NEON(const uint8_t* src_argb4444,
>+                            uint8_t* dst_argb,
>                             int width) {
>   asm volatile(
>       "1:                                        \n"
>@@ -902,7 +958,9 @@ void ARGB4444ToARGBRow_NEON(const uint8* src_argb4444,
>       );
> }
> 
>-void ARGBToRGB24Row_NEON(const uint8* src_argb, uint8* dst_rgb24, int width) {
>+void ARGBToRGB24Row_NEON(const uint8_t* src_argb,
>+                         uint8_t* dst_rgb24,
>+                         int width) {
>   asm volatile(
>       "1:                                        \n"
>       "ld4        {v1.8b,v2.8b,v3.8b,v4.8b}, [%0], #32 \n"  // load 8 ARGB
>@@ -918,7 +976,7 @@ void ARGBToRGB24Row_NEON(const uint8* src_argb, uint8* dst_rgb24, int width) {
>       );
> }
> 
>-void ARGBToRAWRow_NEON(const uint8* src_argb, uint8* dst_raw, int width) {
>+void ARGBToRAWRow_NEON(const uint8_t* src_argb, uint8_t* dst_raw, int width) {
>   asm volatile(
>       "1:                                        \n"
>       "ld4        {v1.8b,v2.8b,v3.8b,v4.8b}, [%0], #32 \n"  // load b g r a
>@@ -935,7 +993,7 @@ void ARGBToRAWRow_NEON(const uint8* src_argb, uint8* dst_raw, int width) {
>       );
> }
> 
>-void YUY2ToYRow_NEON(const uint8* src_yuy2, uint8* dst_y, int width) {
>+void YUY2ToYRow_NEON(const uint8_t* src_yuy2, uint8_t* dst_y, int width) {
>   asm volatile(
>       "1:                                        \n"
>       "ld2        {v0.16b,v1.16b}, [%0], #32     \n"  // load 16 pixels of YUY2.
>@@ -950,7 +1008,7 @@ void YUY2ToYRow_NEON(const uint8* src_yuy2, uint8* dst_y, int width) {
>       );
> }
> 
>-void UYVYToYRow_NEON(const uint8* src_uyvy, uint8* dst_y, int width) {
>+void UYVYToYRow_NEON(const uint8_t* src_uyvy, uint8_t* dst_y, int width) {
>   asm volatile(
>       "1:                                        \n"
>       "ld2        {v0.16b,v1.16b}, [%0], #32     \n"  // load 16 pixels of UYVY.
>@@ -965,9 +1023,9 @@ void UYVYToYRow_NEON(const uint8* src_uyvy, uint8* dst_y, int width) {
>       );
> }
> 
>-void YUY2ToUV422Row_NEON(const uint8* src_yuy2,
>-                         uint8* dst_u,
>-                         uint8* dst_v,
>+void YUY2ToUV422Row_NEON(const uint8_t* src_yuy2,
>+                         uint8_t* dst_u,
>+                         uint8_t* dst_v,
>                          int width) {
>   asm volatile(
>       "1:                                        \n"
>@@ -985,9 +1043,9 @@ void YUY2ToUV422Row_NEON(const uint8* src_yuy2,
>       );
> }
> 
>-void UYVYToUV422Row_NEON(const uint8* src_uyvy,
>-                         uint8* dst_u,
>-                         uint8* dst_v,
>+void UYVYToUV422Row_NEON(const uint8_t* src_uyvy,
>+                         uint8_t* dst_u,
>+                         uint8_t* dst_v,
>                          int width) {
>   asm volatile(
>       "1:                                        \n"
>@@ -1005,12 +1063,12 @@ void UYVYToUV422Row_NEON(const uint8* src_uyvy,
>       );
> }
> 
>-void YUY2ToUVRow_NEON(const uint8* src_yuy2,
>+void YUY2ToUVRow_NEON(const uint8_t* src_yuy2,
>                       int stride_yuy2,
>-                      uint8* dst_u,
>-                      uint8* dst_v,
>+                      uint8_t* dst_u,
>+                      uint8_t* dst_v,
>                       int width) {
>-  const uint8* src_yuy2b = src_yuy2 + stride_yuy2;
>+  const uint8_t* src_yuy2b = src_yuy2 + stride_yuy2;
>   asm volatile(
>       "1:                                        \n"
>       "ld4        {v0.8b,v1.8b,v2.8b,v3.8b}, [%0], #32 \n"  // load 16 pixels
>@@ -1032,12 +1090,12 @@ void YUY2ToUVRow_NEON(const uint8* src_yuy2,
>       );
> }
> 
>-void UYVYToUVRow_NEON(const uint8* src_uyvy,
>+void UYVYToUVRow_NEON(const uint8_t* src_uyvy,
>                       int stride_uyvy,
>-                      uint8* dst_u,
>-                      uint8* dst_v,
>+                      uint8_t* dst_u,
>+                      uint8_t* dst_v,
>                       int width) {
>-  const uint8* src_uyvyb = src_uyvy + stride_uyvy;
>+  const uint8_t* src_uyvyb = src_uyvy + stride_uyvy;
>   asm volatile(
>       "1:                                        \n"
>       "ld4        {v0.8b,v1.8b,v2.8b,v3.8b}, [%0], #32 \n"  // load 16 pixels
>@@ -1060,9 +1118,9 @@ void UYVYToUVRow_NEON(const uint8* src_uyvy,
> }
> 
> // For BGRAToARGB, ABGRToARGB, RGBAToARGB, and ARGBToRGBA.
>-void ARGBShuffleRow_NEON(const uint8* src_argb,
>-                         uint8* dst_argb,
>-                         const uint8* shuffler,
>+void ARGBShuffleRow_NEON(const uint8_t* src_argb,
>+                         uint8_t* dst_argb,
>+                         const uint8_t* shuffler,
>                          int width) {
>   asm volatile(
>       "ld1        {v2.16b}, [%3]                 \n"  // shuffler
>@@ -1080,10 +1138,10 @@ void ARGBShuffleRow_NEON(const uint8* src_argb,
>       );
> }
> 
>-void I422ToYUY2Row_NEON(const uint8* src_y,
>-                        const uint8* src_u,
>-                        const uint8* src_v,
>-                        uint8* dst_yuy2,
>+void I422ToYUY2Row_NEON(const uint8_t* src_y,
>+                        const uint8_t* src_u,
>+                        const uint8_t* src_v,
>+                        uint8_t* dst_yuy2,
>                         int width) {
>   asm volatile(
>       "1:                                        \n"
>@@ -1103,10 +1161,10 @@ void I422ToYUY2Row_NEON(const uint8* src_y,
>       : "cc", "memory", "v0", "v1", "v2", "v3");
> }
> 
>-void I422ToUYVYRow_NEON(const uint8* src_y,
>-                        const uint8* src_u,
>-                        const uint8* src_v,
>-                        uint8* dst_uyvy,
>+void I422ToUYVYRow_NEON(const uint8_t* src_y,
>+                        const uint8_t* src_u,
>+                        const uint8_t* src_v,
>+                        uint8_t* dst_uyvy,
>                         int width) {
>   asm volatile(
>       "1:                                        \n"
>@@ -1126,7 +1184,9 @@ void I422ToUYVYRow_NEON(const uint8* src_y,
>       : "cc", "memory", "v0", "v1", "v2", "v3");
> }
> 
>-void ARGBToRGB565Row_NEON(const uint8* src_argb, uint8* dst_rgb565, int width) {
>+void ARGBToRGB565Row_NEON(const uint8_t* src_argb,
>+                          uint8_t* dst_rgb565,
>+                          int width) {
>   asm volatile(
>       "1:                                        \n"
>       "ld4        {v20.8b,v21.8b,v22.8b,v23.8b}, [%0], #32 \n"  // load 8 pixels
>@@ -1141,9 +1201,9 @@ void ARGBToRGB565Row_NEON(const uint8* src_argb, uint8* dst_rgb565, int width) {
>       : "cc", "memory", "v0", "v20", "v21", "v22", "v23");
> }
> 
>-void ARGBToRGB565DitherRow_NEON(const uint8* src_argb,
>-                                uint8* dst_rgb,
>-                                const uint32 dither4,
>+void ARGBToRGB565DitherRow_NEON(const uint8_t* src_argb,
>+                                uint8_t* dst_rgb,
>+                                const uint32_t dither4,
>                                 int width) {
>   asm volatile(
>       "dup        v1.4s, %w2                     \n"  // dither4
>@@ -1162,8 +1222,8 @@ void ARGBToRGB565DitherRow_NEON(const uint8* src_argb,
>       : "cc", "memory", "v0", "v1", "v20", "v21", "v22", "v23");
> }
> 
>-void ARGBToARGB1555Row_NEON(const uint8* src_argb,
>-                            uint8* dst_argb1555,
>+void ARGBToARGB1555Row_NEON(const uint8_t* src_argb,
>+                            uint8_t* dst_argb1555,
>                             int width) {
>   asm volatile(
>       "1:                                        \n"
>@@ -1180,8 +1240,8 @@ void ARGBToARGB1555Row_NEON(const uint8* src_argb,
>       : "cc", "memory", "v0", "v20", "v21", "v22", "v23");
> }
> 
>-void ARGBToARGB4444Row_NEON(const uint8* src_argb,
>-                            uint8* dst_argb4444,
>+void ARGBToARGB4444Row_NEON(const uint8_t* src_argb,
>+                            uint8_t* dst_argb4444,
>                             int width) {
>   asm volatile(
>       "movi       v4.16b, #0x0f                  \n"  // bits to clear with
>@@ -1200,7 +1260,7 @@ void ARGBToARGB4444Row_NEON(const uint8* src_argb,
>       : "cc", "memory", "v0", "v1", "v4", "v20", "v21", "v22", "v23");
> }
> 
>-void ARGBToYRow_NEON(const uint8* src_argb, uint8* dst_y, int width) {
>+void ARGBToYRow_NEON(const uint8_t* src_argb, uint8_t* dst_y, int width) {
>   asm volatile(
>       "movi       v4.8b, #13                     \n"  // B * 0.1016 coefficient
>       "movi       v5.8b, #65                     \n"  // G * 0.5078 coefficient
>@@ -1223,7 +1283,9 @@ void ARGBToYRow_NEON(const uint8* src_argb, uint8* dst_y, int width) {
>       : "cc", "memory", "v0", "v1", "v2", "v3", "v4", "v5", "v6", "v7");
> }
> 
>-void ARGBExtractAlphaRow_NEON(const uint8* src_argb, uint8* dst_a, int width) {
>+void ARGBExtractAlphaRow_NEON(const uint8_t* src_argb,
>+                              uint8_t* dst_a,
>+                              int width) {
>   asm volatile(
>       "1:                                        \n"
>       "ld4        {v0.16b,v1.16b,v2.16b,v3.16b}, [%0], #64 \n"  // load row 16
>@@ -1239,7 +1301,7 @@ void ARGBExtractAlphaRow_NEON(const uint8* src_argb, uint8* dst_a, int width) {
>       );
> }
> 
>-void ARGBToYJRow_NEON(const uint8* src_argb, uint8* dst_y, int width) {
>+void ARGBToYJRow_NEON(const uint8_t* src_argb, uint8_t* dst_y, int width) {
>   asm volatile(
>       "movi       v4.8b, #15                     \n"  // B * 0.11400 coefficient
>       "movi       v5.8b, #75                     \n"  // G * 0.58700 coefficient
>@@ -1261,9 +1323,9 @@ void ARGBToYJRow_NEON(const uint8* src_argb, uint8* dst_y, int width) {
> }
> 
> // 8x1 pixels.
>-void ARGBToUV444Row_NEON(const uint8* src_argb,
>-                         uint8* dst_u,
>-                         uint8* dst_v,
>+void ARGBToUV444Row_NEON(const uint8_t* src_argb,
>+                         uint8_t* dst_u,
>+                         uint8_t* dst_v,
>                          int width) {
>   asm volatile(
>       "movi       v24.8b, #112                   \n"  // UB / VR 0.875
>@@ -1328,12 +1390,12 @@ void ARGBToUV444Row_NEON(const uint8* src_argb,
> // TODO(fbarchard): Consider vhadd vertical, then vpaddl horizontal, avoid shr.
> // TODO(fbarchard): consider ptrdiff_t for all strides.
> 
>-void ARGBToUVRow_NEON(const uint8* src_argb,
>+void ARGBToUVRow_NEON(const uint8_t* src_argb,
>                       int src_stride_argb,
>-                      uint8* dst_u,
>-                      uint8* dst_v,
>+                      uint8_t* dst_u,
>+                      uint8_t* dst_v,
>                       int width) {
>-  const uint8* src_argb_1 = src_argb + src_stride_argb;
>+  const uint8_t* src_argb_1 = src_argb + src_stride_argb;
>   asm volatile (
>     RGBTOUV_SETUP_REG
>   "1:                                          \n"
>@@ -1368,12 +1430,12 @@ void ARGBToUVRow_NEON(const uint8* src_argb,
> }
> 
> // TODO(fbarchard): Subsample match C code.
>-void ARGBToUVJRow_NEON(const uint8* src_argb,
>+void ARGBToUVJRow_NEON(const uint8_t* src_argb,
>                        int src_stride_argb,
>-                       uint8* dst_u,
>-                       uint8* dst_v,
>+                       uint8_t* dst_u,
>+                       uint8_t* dst_v,
>                        int width) {
>-  const uint8* src_argb_1 = src_argb + src_stride_argb;
>+  const uint8_t* src_argb_1 = src_argb + src_stride_argb;
>   asm volatile (
>     "movi       v20.8h, #63, lsl #0            \n"  // UB/VR coeff (0.500) / 2
>     "movi       v21.8h, #42, lsl #0            \n"  // UG coeff (-0.33126) / 2
>@@ -1411,12 +1473,12 @@ void ARGBToUVJRow_NEON(const uint8* src_argb,
>   );
> }
> 
>-void BGRAToUVRow_NEON(const uint8* src_bgra,
>+void BGRAToUVRow_NEON(const uint8_t* src_bgra,
>                       int src_stride_bgra,
>-                      uint8* dst_u,
>-                      uint8* dst_v,
>+                      uint8_t* dst_u,
>+                      uint8_t* dst_v,
>                       int width) {
>-  const uint8* src_bgra_1 = src_bgra + src_stride_bgra;
>+  const uint8_t* src_bgra_1 = src_bgra + src_stride_bgra;
>   asm volatile (
>     RGBTOUV_SETUP_REG
>   "1:                                          \n"
>@@ -1449,12 +1511,12 @@ void BGRAToUVRow_NEON(const uint8* src_bgra,
>   );
> }
> 
>-void ABGRToUVRow_NEON(const uint8* src_abgr,
>+void ABGRToUVRow_NEON(const uint8_t* src_abgr,
>                       int src_stride_abgr,
>-                      uint8* dst_u,
>-                      uint8* dst_v,
>+                      uint8_t* dst_u,
>+                      uint8_t* dst_v,
>                       int width) {
>-  const uint8* src_abgr_1 = src_abgr + src_stride_abgr;
>+  const uint8_t* src_abgr_1 = src_abgr + src_stride_abgr;
>   asm volatile (
>     RGBTOUV_SETUP_REG
>   "1:                                          \n"
>@@ -1487,12 +1549,12 @@ void ABGRToUVRow_NEON(const uint8* src_abgr,
>   );
> }
> 
>-void RGBAToUVRow_NEON(const uint8* src_rgba,
>+void RGBAToUVRow_NEON(const uint8_t* src_rgba,
>                       int src_stride_rgba,
>-                      uint8* dst_u,
>-                      uint8* dst_v,
>+                      uint8_t* dst_u,
>+                      uint8_t* dst_v,
>                       int width) {
>-  const uint8* src_rgba_1 = src_rgba + src_stride_rgba;
>+  const uint8_t* src_rgba_1 = src_rgba + src_stride_rgba;
>   asm volatile (
>     RGBTOUV_SETUP_REG
>   "1:                                          \n"
>@@ -1525,12 +1587,12 @@ void RGBAToUVRow_NEON(const uint8* src_rgba,
>   );
> }
> 
>-void RGB24ToUVRow_NEON(const uint8* src_rgb24,
>+void RGB24ToUVRow_NEON(const uint8_t* src_rgb24,
>                        int src_stride_rgb24,
>-                       uint8* dst_u,
>-                       uint8* dst_v,
>+                       uint8_t* dst_u,
>+                       uint8_t* dst_v,
>                        int width) {
>-  const uint8* src_rgb24_1 = src_rgb24 + src_stride_rgb24;
>+  const uint8_t* src_rgb24_1 = src_rgb24 + src_stride_rgb24;
>   asm volatile (
>     RGBTOUV_SETUP_REG
>   "1:                                          \n"
>@@ -1563,12 +1625,12 @@ void RGB24ToUVRow_NEON(const uint8* src_rgb24,
>   );
> }
> 
>-void RAWToUVRow_NEON(const uint8* src_raw,
>+void RAWToUVRow_NEON(const uint8_t* src_raw,
>                      int src_stride_raw,
>-                     uint8* dst_u,
>-                     uint8* dst_v,
>+                     uint8_t* dst_u,
>+                     uint8_t* dst_v,
>                      int width) {
>-  const uint8* src_raw_1 = src_raw + src_stride_raw;
>+  const uint8_t* src_raw_1 = src_raw + src_stride_raw;
>   asm volatile (
>     RGBTOUV_SETUP_REG
>   "1:                                          \n"
>@@ -1602,12 +1664,12 @@ void RAWToUVRow_NEON(const uint8* src_raw,
> }
> 
> // 16x2 pixels -> 8x1.  width is number of argb pixels. e.g. 16.
>-void RGB565ToUVRow_NEON(const uint8* src_rgb565,
>+void RGB565ToUVRow_NEON(const uint8_t* src_rgb565,
>                         int src_stride_rgb565,
>-                        uint8* dst_u,
>-                        uint8* dst_v,
>+                        uint8_t* dst_u,
>+                        uint8_t* dst_v,
>                         int width) {
>-  const uint8* src_rgb565_1 = src_rgb565 + src_stride_rgb565;
>+  const uint8_t* src_rgb565_1 = src_rgb565 + src_stride_rgb565;
>   asm volatile(
>       "movi       v22.8h, #56, lsl #0            \n"  // UB / VR coeff (0.875) /
>                                                       // 2
>@@ -1673,12 +1735,12 @@ void RGB565ToUVRow_NEON(const uint8* src_rgb565,
> }
> 
> // 16x2 pixels -> 8x1.  width is number of argb pixels. e.g. 16.
>-void ARGB1555ToUVRow_NEON(const uint8* src_argb1555,
>+void ARGB1555ToUVRow_NEON(const uint8_t* src_argb1555,
>                           int src_stride_argb1555,
>-                          uint8* dst_u,
>-                          uint8* dst_v,
>+                          uint8_t* dst_u,
>+                          uint8_t* dst_v,
>                           int width) {
>-  const uint8* src_argb1555_1 = src_argb1555 + src_stride_argb1555;
>+  const uint8_t* src_argb1555_1 = src_argb1555 + src_stride_argb1555;
>   asm volatile(
>       RGBTOUV_SETUP_REG
>       "1:                                        \n"
>@@ -1738,12 +1800,12 @@ void ARGB1555ToUVRow_NEON(const uint8* src_argb1555,
> }
> 
> // 16x2 pixels -> 8x1.  width is number of argb pixels. e.g. 16.
>-void ARGB4444ToUVRow_NEON(const uint8* src_argb4444,
>+void ARGB4444ToUVRow_NEON(const uint8_t* src_argb4444,
>                           int src_stride_argb4444,
>-                          uint8* dst_u,
>-                          uint8* dst_v,
>+                          uint8_t* dst_u,
>+                          uint8_t* dst_v,
>                           int width) {
>-  const uint8* src_argb4444_1 = src_argb4444 + src_stride_argb4444;
>+  const uint8_t* src_argb4444_1 = src_argb4444 + src_stride_argb4444;
>   asm volatile(
>       RGBTOUV_SETUP_REG
>       "1:                                        \n"
>@@ -1804,7 +1866,7 @@ void ARGB4444ToUVRow_NEON(const uint8* src_argb4444,
>       );
> }
> 
>-void RGB565ToYRow_NEON(const uint8* src_rgb565, uint8* dst_y, int width) {
>+void RGB565ToYRow_NEON(const uint8_t* src_rgb565, uint8_t* dst_y, int width) {
>   asm volatile(
>       "movi       v24.8b, #13                    \n"  // B * 0.1016 coefficient
>       "movi       v25.8b, #65                    \n"  // G * 0.5078 coefficient
>@@ -1829,7 +1891,9 @@ void RGB565ToYRow_NEON(const uint8* src_rgb565, uint8* dst_y, int width) {
>         "v27");
> }
> 
>-void ARGB1555ToYRow_NEON(const uint8* src_argb1555, uint8* dst_y, int width) {
>+void ARGB1555ToYRow_NEON(const uint8_t* src_argb1555,
>+                         uint8_t* dst_y,
>+                         int width) {
>   asm volatile(
>       "movi       v4.8b, #13                     \n"  // B * 0.1016 coefficient
>       "movi       v5.8b, #65                     \n"  // G * 0.5078 coefficient
>@@ -1853,7 +1917,9 @@ void ARGB1555ToYRow_NEON(const uint8* src_argb1555, uint8* dst_y, int width) {
>       : "cc", "memory", "v0", "v1", "v2", "v3", "v4", "v5", "v6", "v7");
> }
> 
>-void ARGB4444ToYRow_NEON(const uint8* src_argb4444, uint8* dst_y, int width) {
>+void ARGB4444ToYRow_NEON(const uint8_t* src_argb4444,
>+                         uint8_t* dst_y,
>+                         int width) {
>   asm volatile(
>       "movi       v24.8b, #13                    \n"  // B * 0.1016 coefficient
>       "movi       v25.8b, #65                    \n"  // G * 0.5078 coefficient
>@@ -1877,7 +1943,7 @@ void ARGB4444ToYRow_NEON(const uint8* src_argb4444, uint8* dst_y, int width) {
>       : "cc", "memory", "v0", "v1", "v2", "v3", "v24", "v25", "v26", "v27");
> }
> 
>-void BGRAToYRow_NEON(const uint8* src_bgra, uint8* dst_y, int width) {
>+void BGRAToYRow_NEON(const uint8_t* src_bgra, uint8_t* dst_y, int width) {
>   asm volatile(
>       "movi       v4.8b, #33                     \n"  // R * 0.2578 coefficient
>       "movi       v5.8b, #65                     \n"  // G * 0.5078 coefficient
>@@ -1900,7 +1966,7 @@ void BGRAToYRow_NEON(const uint8* src_bgra, uint8* dst_y, int width) {
>       : "cc", "memory", "v0", "v1", "v2", "v3", "v4", "v5", "v6", "v7", "v16");
> }
> 
>-void ABGRToYRow_NEON(const uint8* src_abgr, uint8* dst_y, int width) {
>+void ABGRToYRow_NEON(const uint8_t* src_abgr, uint8_t* dst_y, int width) {
>   asm volatile(
>       "movi       v4.8b, #33                     \n"  // R * 0.2578 coefficient
>       "movi       v5.8b, #65                     \n"  // G * 0.5078 coefficient
>@@ -1923,7 +1989,7 @@ void ABGRToYRow_NEON(const uint8* src_abgr, uint8* dst_y, int width) {
>       : "cc", "memory", "v0", "v1", "v2", "v3", "v4", "v5", "v6", "v7", "v16");
> }
> 
>-void RGBAToYRow_NEON(const uint8* src_rgba, uint8* dst_y, int width) {
>+void RGBAToYRow_NEON(const uint8_t* src_rgba, uint8_t* dst_y, int width) {
>   asm volatile(
>       "movi       v4.8b, #13                     \n"  // B * 0.1016 coefficient
>       "movi       v5.8b, #65                     \n"  // G * 0.5078 coefficient
>@@ -1946,7 +2012,7 @@ void RGBAToYRow_NEON(const uint8* src_rgba, uint8* dst_y, int width) {
>       : "cc", "memory", "v0", "v1", "v2", "v3", "v4", "v5", "v6", "v7", "v16");
> }
> 
>-void RGB24ToYRow_NEON(const uint8* src_rgb24, uint8* dst_y, int width) {
>+void RGB24ToYRow_NEON(const uint8_t* src_rgb24, uint8_t* dst_y, int width) {
>   asm volatile(
>       "movi       v4.8b, #13                     \n"  // B * 0.1016 coefficient
>       "movi       v5.8b, #65                     \n"  // G * 0.5078 coefficient
>@@ -1969,7 +2035,7 @@ void RGB24ToYRow_NEON(const uint8* src_rgb24, uint8* dst_y, int width) {
>       : "cc", "memory", "v0", "v1", "v2", "v3", "v4", "v5", "v6", "v7", "v16");
> }
> 
>-void RAWToYRow_NEON(const uint8* src_raw, uint8* dst_y, int width) {
>+void RAWToYRow_NEON(const uint8_t* src_raw, uint8_t* dst_y, int width) {
>   asm volatile(
>       "movi       v4.8b, #33                     \n"  // R * 0.2578 coefficient
>       "movi       v5.8b, #65                     \n"  // G * 0.5078 coefficient
>@@ -1993,14 +2059,14 @@ void RAWToYRow_NEON(const uint8* src_raw, uint8* dst_y, int width) {
> }
> 
> // Bilinear filter 16x2 -> 16x1
>-void InterpolateRow_NEON(uint8* dst_ptr,
>-                         const uint8* src_ptr,
>+void InterpolateRow_NEON(uint8_t* dst_ptr,
>+                         const uint8_t* src_ptr,
>                          ptrdiff_t src_stride,
>                          int dst_width,
>                          int source_y_fraction) {
>   int y1_fraction = source_y_fraction;
>   int y0_fraction = 256 - y1_fraction;
>-  const uint8* src_ptr1 = src_ptr + src_stride;
>+  const uint8_t* src_ptr1 = src_ptr + src_stride;
>   asm volatile(
>       "cmp        %w4, #0                        \n"
>       "b.eq       100f                           \n"
>@@ -2053,9 +2119,9 @@ void InterpolateRow_NEON(uint8* dst_ptr,
> }
> 
> // dr * (256 - sa) / 256 + sr = dr - dr * sa / 256 + sr
>-void ARGBBlendRow_NEON(const uint8* src_argb0,
>-                       const uint8* src_argb1,
>-                       uint8* dst_argb,
>+void ARGBBlendRow_NEON(const uint8_t* src_argb0,
>+                       const uint8_t* src_argb1,
>+                       uint8_t* dst_argb,
>                        int width) {
>   asm volatile(
>       "subs       %w3, %w3, #8                   \n"
>@@ -2121,7 +2187,9 @@ void ARGBBlendRow_NEON(const uint8* src_argb0,
> }
> 
> // Attenuate 8 pixels at a time.
>-void ARGBAttenuateRow_NEON(const uint8* src_argb, uint8* dst_argb, int width) {
>+void ARGBAttenuateRow_NEON(const uint8_t* src_argb,
>+                           uint8_t* dst_argb,
>+                           int width) {
>   asm volatile(
>       // Attenuate 8 pixels.
>       "1:                                        \n"
>@@ -2145,7 +2213,7 @@ void ARGBAttenuateRow_NEON(const uint8* src_argb, uint8* dst_argb, int width) {
> 
> // Quantize 8 ARGB pixels (32 bytes).
> // dst = (dst * scale >> 16) * interval_size + interval_offset;
>-void ARGBQuantizeRow_NEON(uint8* dst_argb,
>+void ARGBQuantizeRow_NEON(uint8_t* dst_argb,
>                           int scale,
>                           int interval_size,
>                           int interval_offset,
>@@ -2188,10 +2256,10 @@ void ARGBQuantizeRow_NEON(uint8* dst_argb,
> // Shade 8 pixels at a time by specified value.
> // NOTE vqrdmulh.s16 q10, q10, d0[0] must use a scaler register from 0 to 8.
> // Rounding in vqrdmulh does +1 to high if high bit of low s16 is set.
>-void ARGBShadeRow_NEON(const uint8* src_argb,
>-                       uint8* dst_argb,
>+void ARGBShadeRow_NEON(const uint8_t* src_argb,
>+                       uint8_t* dst_argb,
>                        int width,
>-                       uint32 value) {
>+                       uint32_t value) {
>   asm volatile(
>       "dup        v0.4s, %w3                     \n"  // duplicate scale value.
>       "zip1       v0.8b, v0.8b, v0.8b            \n"  // v0.8b aarrggbb.
>@@ -2225,7 +2293,7 @@ void ARGBShadeRow_NEON(const uint8* src_argb,
> // Convert 8 ARGB pixels (64 bytes) to 8 Gray ARGB pixels
> // Similar to ARGBToYJ but stores ARGB.
> // C code is (15 * b + 75 * g + 38 * r + 64) >> 7;
>-void ARGBGrayRow_NEON(const uint8* src_argb, uint8* dst_argb, int width) {
>+void ARGBGrayRow_NEON(const uint8_t* src_argb, uint8_t* dst_argb, int width) {
>   asm volatile(
>       "movi       v24.8b, #15                    \n"  // B * 0.11400 coefficient
>       "movi       v25.8b, #75                    \n"  // G * 0.58700 coefficient
>@@ -2253,7 +2321,7 @@ void ARGBGrayRow_NEON(const uint8* src_argb, uint8* dst_argb, int width) {
> //    g = (r * 45 + g * 88 + b * 22) >> 7
> //    r = (r * 50 + g * 98 + b * 24) >> 7
> 
>-void ARGBSepiaRow_NEON(uint8* dst_argb, int width) {
>+void ARGBSepiaRow_NEON(uint8_t* dst_argb, int width) {
>   asm volatile(
>       "movi       v20.8b, #17                    \n"  // BB coefficient
>       "movi       v21.8b, #68                    \n"  // BG coefficient
>@@ -2291,9 +2359,9 @@ void ARGBSepiaRow_NEON(uint8* dst_argb, int width) {
> // Tranform 8 ARGB pixels (32 bytes) with color matrix.
> // TODO(fbarchard): Was same as Sepia except matrix is provided.  This function
> // needs to saturate.  Consider doing a non-saturating version.
>-void ARGBColorMatrixRow_NEON(const uint8* src_argb,
>-                             uint8* dst_argb,
>-                             const int8* matrix_argb,
>+void ARGBColorMatrixRow_NEON(const uint8_t* src_argb,
>+                             uint8_t* dst_argb,
>+                             const int8_t* matrix_argb,
>                              int width) {
>   asm volatile(
>       "ld1        {v2.16b}, [%3]                 \n"  // load 3 ARGB vectors.
>@@ -2351,9 +2419,9 @@ void ARGBColorMatrixRow_NEON(const uint8* src_argb,
> 
> // TODO(fbarchard): fix vqshrun in ARGBMultiplyRow_NEON and reenable.
> // Multiply 2 rows of ARGB pixels together, 8 pixels at a time.
>-void ARGBMultiplyRow_NEON(const uint8* src_argb0,
>-                          const uint8* src_argb1,
>-                          uint8* dst_argb,
>+void ARGBMultiplyRow_NEON(const uint8_t* src_argb0,
>+                          const uint8_t* src_argb1,
>+                          uint8_t* dst_argb,
>                           int width) {
>   asm volatile(
>       // 8 pixel loop.
>@@ -2380,9 +2448,9 @@ void ARGBMultiplyRow_NEON(const uint8* src_argb0,
> }
> 
> // Add 2 rows of ARGB pixels together, 8 pixels at a time.
>-void ARGBAddRow_NEON(const uint8* src_argb0,
>-                     const uint8* src_argb1,
>-                     uint8* dst_argb,
>+void ARGBAddRow_NEON(const uint8_t* src_argb0,
>+                     const uint8_t* src_argb1,
>+                     uint8_t* dst_argb,
>                      int width) {
>   asm volatile(
>       // 8 pixel loop.
>@@ -2405,9 +2473,9 @@ void ARGBAddRow_NEON(const uint8* src_argb0,
> }
> 
> // Subtract 2 rows of ARGB pixels, 8 pixels at a time.
>-void ARGBSubtractRow_NEON(const uint8* src_argb0,
>-                          const uint8* src_argb1,
>-                          uint8* dst_argb,
>+void ARGBSubtractRow_NEON(const uint8_t* src_argb0,
>+                          const uint8_t* src_argb1,
>+                          uint8_t* dst_argb,
>                           int width) {
>   asm volatile(
>       // 8 pixel loop.
>@@ -2434,9 +2502,9 @@ void ARGBSubtractRow_NEON(const uint8* src_argb0,
> // R = Sobel
> // G = Sobel
> // B = Sobel
>-void SobelRow_NEON(const uint8* src_sobelx,
>-                   const uint8* src_sobely,
>-                   uint8* dst_argb,
>+void SobelRow_NEON(const uint8_t* src_sobelx,
>+                   const uint8_t* src_sobely,
>+                   uint8_t* dst_argb,
>                    int width) {
>   asm volatile(
>       "movi       v3.8b, #255                    \n"  // alpha
>@@ -2459,9 +2527,9 @@ void SobelRow_NEON(const uint8* src_sobelx,
> }
> 
> // Adds Sobel X and Sobel Y and stores Sobel into plane.
>-void SobelToPlaneRow_NEON(const uint8* src_sobelx,
>-                          const uint8* src_sobely,
>-                          uint8* dst_y,
>+void SobelToPlaneRow_NEON(const uint8_t* src_sobelx,
>+                          const uint8_t* src_sobely,
>+                          uint8_t* dst_y,
>                           int width) {
>   asm volatile(
>       // 16 pixel loop.
>@@ -2485,9 +2553,9 @@ void SobelToPlaneRow_NEON(const uint8* src_sobelx,
> // R = Sobel X
> // G = Sobel
> // B = Sobel Y
>-void SobelXYRow_NEON(const uint8* src_sobelx,
>-                     const uint8* src_sobely,
>-                     uint8* dst_argb,
>+void SobelXYRow_NEON(const uint8_t* src_sobelx,
>+                     const uint8_t* src_sobely,
>+                     uint8_t* dst_argb,
>                      int width) {
>   asm volatile(
>       "movi       v3.8b, #255                    \n"  // alpha
>@@ -2511,10 +2579,10 @@ void SobelXYRow_NEON(const uint8* src_sobelx,
> // -1  0  1
> // -2  0  2
> // -1  0  1
>-void SobelXRow_NEON(const uint8* src_y0,
>-                    const uint8* src_y1,
>-                    const uint8* src_y2,
>-                    uint8* dst_sobelx,
>+void SobelXRow_NEON(const uint8_t* src_y0,
>+                    const uint8_t* src_y1,
>+                    const uint8_t* src_y2,
>+                    uint8_t* dst_sobelx,
>                     int width) {
>   asm volatile(
>       "1:                                        \n"
>@@ -2550,9 +2618,9 @@ void SobelXRow_NEON(const uint8* src_y0,
> // -1 -2 -1
> //  0  0  0
> //  1  2  1
>-void SobelYRow_NEON(const uint8* src_y0,
>-                    const uint8* src_y1,
>-                    uint8* dst_sobely,
>+void SobelYRow_NEON(const uint8_t* src_y0,
>+                    const uint8_t* src_y1,
>+                    uint8_t* dst_sobely,
>                     int width) {
>   asm volatile(
>       "1:                                        \n"
>@@ -2584,7 +2652,10 @@ void SobelYRow_NEON(const uint8* src_y0,
> }
> 
> // Caveat - rounds float to half float whereas scaling version truncates.
>-void HalfFloat1Row_NEON(const uint16* src, uint16* dst, float, int width) {
>+void HalfFloat1Row_NEON(const uint16_t* src,
>+                        uint16_t* dst,
>+                        float /*unused*/,
>+                        int width) {
>   asm volatile(
>       "1:                                        \n"
>       "ld1        {v1.16b}, [%0], #16            \n"  // load 8 shorts
>@@ -2604,7 +2675,10 @@ void HalfFloat1Row_NEON(const uint16* src, uint16* dst, float, int width) {
>       : "cc", "memory", "v1", "v2", "v3");
> }
> 
>-void HalfFloatRow_NEON(const uint16* src, uint16* dst, float scale, int width) {
>+void HalfFloatRow_NEON(const uint16_t* src,
>+                       uint16_t* dst,
>+                       float scale,
>+                       int width) {
>   asm volatile(
>       "1:                                        \n"
>       "ld1        {v1.16b}, [%0], #16            \n"  // load 8 shorts
>@@ -2626,6 +2700,30 @@ void HalfFloatRow_NEON(const uint16* src, uint16* dst, float scale, int width) {
>       : "cc", "memory", "v1", "v2", "v3");
> }
> 
>+void ByteToFloatRow_NEON(const uint8_t* src,
>+                         float* dst,
>+                         float scale,
>+                         int width) {
>+  asm volatile(
>+      "1:                                        \n"
>+      "ld1        {v1.8b}, [%0], #8              \n"  // load 8 bytes
>+      "subs       %w2, %w2, #8                   \n"  // 8 pixels per loop
>+      "uxtl       v1.8h, v1.8b                   \n"  // 8 shorts
>+      "uxtl       v2.4s, v1.4h                   \n"  // 8 ints
>+      "uxtl2      v3.4s, v1.8h                   \n"
>+      "scvtf      v2.4s, v2.4s                   \n"  // 8 floats
>+      "scvtf      v3.4s, v3.4s                   \n"
>+      "fmul       v2.4s, v2.4s, %3.s[0]          \n"  // scale
>+      "fmul       v3.4s, v3.4s, %3.s[0]          \n"
>+      "st1        {v2.16b, v3.16b}, [%1], #32    \n"  // store 8 floats
>+      "b.gt       1b                             \n"
>+      : "+r"(src),   // %0
>+        "+r"(dst),   // %1
>+        "+r"(width)  // %2
>+      : "w"(scale)   // %3
>+      : "cc", "memory", "v1", "v2", "v3");
>+}
>+
> float ScaleMaxSamples_NEON(const float* src,
>                            float* dst,
>                            float scale,
>@@ -2702,12 +2800,12 @@ void ScaleSamples_NEON(const float* src, float* dst, float scale, int width) {
> }
> 
> // filter 5 rows with 1, 4, 6, 4, 1 coefficients to produce 1 row.
>-void GaussCol_NEON(const uint16* src0,
>-                   const uint16* src1,
>-                   const uint16* src2,
>-                   const uint16* src3,
>-                   const uint16* src4,
>-                   uint32* dst,
>+void GaussCol_NEON(const uint16_t* src0,
>+                   const uint16_t* src1,
>+                   const uint16_t* src2,
>+                   const uint16_t* src3,
>+                   const uint16_t* src4,
>+                   uint32_t* dst,
>                    int width) {
>   asm volatile(
>       "movi       v6.8h, #4                      \n"  // constant 4
>@@ -2742,10 +2840,10 @@ void GaussCol_NEON(const uint16* src0,
> }
> 
> // filter 5 rows with 1, 4, 6, 4, 1 coefficients to produce 1 row.
>-void GaussRow_NEON(const uint32* src, uint16* dst, int width) {
>-  const uint32* src1 = src + 1;
>-  const uint32* src2 = src + 2;
>-  const uint32* src3 = src + 3;
>+void GaussRow_NEON(const uint32_t* src, uint16_t* dst, int width) {
>+  const uint32_t* src1 = src + 1;
>+  const uint32_t* src2 = src + 2;
>+  const uint32_t* src3 = src + 3;
>   asm volatile(
>       "movi       v6.4s, #4                      \n"  // constant 4
>       "movi       v7.4s, #6                      \n"  // constant 6
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/row_win.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/row_win.cc
>index 03a7e9506da..5500d7f5a64 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/row_win.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/row_win.cc
>@@ -28,27 +28,27 @@ extern "C" {
> #if defined(_M_X64)
> 
> // Read 4 UV from 422, upsample to 8 UV.
>-#define READYUV422                                      \
>-  xmm0 = _mm_cvtsi32_si128(*(uint32*)u_buf);            \
>-  xmm1 = _mm_cvtsi32_si128(*(uint32*)(u_buf + offset)); \
>-  xmm0 = _mm_unpacklo_epi8(xmm0, xmm1);                 \
>-  xmm0 = _mm_unpacklo_epi16(xmm0, xmm0);                \
>-  u_buf += 4;                                           \
>-  xmm4 = _mm_loadl_epi64((__m128i*)y_buf);              \
>-  xmm4 = _mm_unpacklo_epi8(xmm4, xmm4);                 \
>+#define READYUV422                                        \
>+  xmm0 = _mm_cvtsi32_si128(*(uint32_t*)u_buf);            \
>+  xmm1 = _mm_cvtsi32_si128(*(uint32_t*)(u_buf + offset)); \
>+  xmm0 = _mm_unpacklo_epi8(xmm0, xmm1);                   \
>+  xmm0 = _mm_unpacklo_epi16(xmm0, xmm0);                  \
>+  u_buf += 4;                                             \
>+  xmm4 = _mm_loadl_epi64((__m128i*)y_buf);                \
>+  xmm4 = _mm_unpacklo_epi8(xmm4, xmm4);                   \
>   y_buf += 8;
> 
> // Read 4 UV from 422, upsample to 8 UV.  With 8 Alpha.
>-#define READYUVA422                                     \
>-  xmm0 = _mm_cvtsi32_si128(*(uint32*)u_buf);            \
>-  xmm1 = _mm_cvtsi32_si128(*(uint32*)(u_buf + offset)); \
>-  xmm0 = _mm_unpacklo_epi8(xmm0, xmm1);                 \
>-  xmm0 = _mm_unpacklo_epi16(xmm0, xmm0);                \
>-  u_buf += 4;                                           \
>-  xmm4 = _mm_loadl_epi64((__m128i*)y_buf);              \
>-  xmm4 = _mm_unpacklo_epi8(xmm4, xmm4);                 \
>-  y_buf += 8;                                           \
>-  xmm5 = _mm_loadl_epi64((__m128i*)a_buf);              \
>+#define READYUVA422                                       \
>+  xmm0 = _mm_cvtsi32_si128(*(uint32_t*)u_buf);            \
>+  xmm1 = _mm_cvtsi32_si128(*(uint32_t*)(u_buf + offset)); \
>+  xmm0 = _mm_unpacklo_epi8(xmm0, xmm1);                   \
>+  xmm0 = _mm_unpacklo_epi16(xmm0, xmm0);                  \
>+  u_buf += 4;                                             \
>+  xmm4 = _mm_loadl_epi64((__m128i*)y_buf);                \
>+  xmm4 = _mm_unpacklo_epi8(xmm4, xmm4);                   \
>+  y_buf += 8;                                             \
>+  xmm5 = _mm_loadl_epi64((__m128i*)a_buf);                \
>   a_buf += 8;
> 
> // Convert 8 pixels: 8 UV and 8 Y.
>@@ -84,15 +84,15 @@ extern "C" {
>   dst_argb += 32;
> 
> #if defined(HAS_I422TOARGBROW_SSSE3)
>-void I422ToARGBRow_SSSE3(const uint8* y_buf,
>-                         const uint8* u_buf,
>-                         const uint8* v_buf,
>-                         uint8* dst_argb,
>+void I422ToARGBRow_SSSE3(const uint8_t* y_buf,
>+                         const uint8_t* u_buf,
>+                         const uint8_t* v_buf,
>+                         uint8_t* dst_argb,
>                          const struct YuvConstants* yuvconstants,
>                          int width) {
>   __m128i xmm0, xmm1, xmm2, xmm4;
>   const __m128i xmm5 = _mm_set1_epi8(-1);
>-  const ptrdiff_t offset = (uint8*)v_buf - (uint8*)u_buf;
>+  const ptrdiff_t offset = (uint8_t*)v_buf - (uint8_t*)u_buf;
>   while (width > 0) {
>     READYUV422
>     YUVTORGB(yuvconstants)
>@@ -103,15 +103,15 @@ void I422ToARGBRow_SSSE3(const uint8* y_buf,
> #endif
> 
> #if defined(HAS_I422ALPHATOARGBROW_SSSE3)
>-void I422AlphaToARGBRow_SSSE3(const uint8* y_buf,
>-                              const uint8* u_buf,
>-                              const uint8* v_buf,
>-                              const uint8* a_buf,
>-                              uint8* dst_argb,
>+void I422AlphaToARGBRow_SSSE3(const uint8_t* y_buf,
>+                              const uint8_t* u_buf,
>+                              const uint8_t* v_buf,
>+                              const uint8_t* a_buf,
>+                              uint8_t* dst_argb,
>                               const struct YuvConstants* yuvconstants,
>                               int width) {
>   __m128i xmm0, xmm1, xmm2, xmm4, xmm5;
>-  const ptrdiff_t offset = (uint8*)v_buf - (uint8*)u_buf;
>+  const ptrdiff_t offset = (uint8_t*)v_buf - (uint8_t*)u_buf;
>   while (width > 0) {
>     READYUVA422
>     YUVTORGB(yuvconstants)
>@@ -255,8 +255,8 @@ static const lvec8 kShuffleNV21 = {
> };
> 
> // Duplicates gray value 3 times and fills in alpha opaque.
>-__declspec(naked) void J400ToARGBRow_SSE2(const uint8* src_y,
>-                                          uint8* dst_argb,
>+__declspec(naked) void J400ToARGBRow_SSE2(const uint8_t* src_y,
>+                                          uint8_t* dst_argb,
>                                           int width) {
>   __asm {
>     mov        eax, [esp + 4]  // src_y
>@@ -285,8 +285,8 @@ __declspec(naked) void J400ToARGBRow_SSE2(const uint8* src_y,
> 
> #ifdef HAS_J400TOARGBROW_AVX2
> // Duplicates gray value 3 times and fills in alpha opaque.
>-__declspec(naked) void J400ToARGBRow_AVX2(const uint8* src_y,
>-                                          uint8* dst_argb,
>+__declspec(naked) void J400ToARGBRow_AVX2(const uint8_t* src_y,
>+                                          uint8_t* dst_argb,
>                                           int width) {
>   __asm {
>     mov         eax, [esp + 4]  // src_y
>@@ -316,8 +316,8 @@ __declspec(naked) void J400ToARGBRow_AVX2(const uint8* src_y,
> }
> #endif  // HAS_J400TOARGBROW_AVX2
> 
>-__declspec(naked) void RGB24ToARGBRow_SSSE3(const uint8* src_rgb24,
>-                                            uint8* dst_argb,
>+__declspec(naked) void RGB24ToARGBRow_SSSE3(const uint8_t* src_rgb24,
>+                                            uint8_t* dst_argb,
>                                             int width) {
>   __asm {
>     mov       eax, [esp + 4]  // src_rgb24
>@@ -355,8 +355,8 @@ __declspec(naked) void RGB24ToARGBRow_SSSE3(const uint8* src_rgb24,
>   }
> }
> 
>-__declspec(naked) void RAWToARGBRow_SSSE3(const uint8* src_raw,
>-                                          uint8* dst_argb,
>+__declspec(naked) void RAWToARGBRow_SSSE3(const uint8_t* src_raw,
>+                                          uint8_t* dst_argb,
>                                           int width) {
>   __asm {
>     mov       eax, [esp + 4]  // src_raw
>@@ -394,8 +394,8 @@ __declspec(naked) void RAWToARGBRow_SSSE3(const uint8* src_raw,
>   }
> }
> 
>-__declspec(naked) void RAWToRGB24Row_SSSE3(const uint8* src_raw,
>-                                           uint8* dst_rgb24,
>+__declspec(naked) void RAWToRGB24Row_SSSE3(const uint8_t* src_raw,
>+                                           uint8_t* dst_rgb24,
>                                            int width) {
>   __asm {
>     mov       eax, [esp + 4]  // src_raw
>@@ -430,8 +430,8 @@ __declspec(naked) void RAWToRGB24Row_SSSE3(const uint8* src_raw,
> // v * (256 + 8)
> // G shift of 5 is incorporated, so shift is 5 + 8 and 5 + 3
> // 20 instructions.
>-__declspec(naked) void RGB565ToARGBRow_SSE2(const uint8* src_rgb565,
>-                                            uint8* dst_argb,
>+__declspec(naked) void RGB565ToARGBRow_SSE2(const uint8_t* src_rgb565,
>+                                            uint8_t* dst_argb,
>                                             int width) {
>   __asm {
>     mov       eax, 0x01080108  // generate multiplier to repeat 5 bits
>@@ -486,8 +486,8 @@ __declspec(naked) void RGB565ToARGBRow_SSE2(const uint8* src_rgb565,
> // v * 256 + v * 8
> // v * (256 + 8)
> // G shift of 5 is incorporated, so shift is 5 + 8 and 5 + 3
>-__declspec(naked) void RGB565ToARGBRow_AVX2(const uint8* src_rgb565,
>-                                            uint8* dst_argb,
>+__declspec(naked) void RGB565ToARGBRow_AVX2(const uint8_t* src_rgb565,
>+                                            uint8_t* dst_argb,
>                                             int width) {
>   __asm {
>     mov        eax, 0x01080108  // generate multiplier to repeat 5 bits
>@@ -537,8 +537,8 @@ __declspec(naked) void RGB565ToARGBRow_AVX2(const uint8* src_rgb565,
> #endif  // HAS_RGB565TOARGBROW_AVX2
> 
> #ifdef HAS_ARGB1555TOARGBROW_AVX2
>-__declspec(naked) void ARGB1555ToARGBRow_AVX2(const uint8* src_argb1555,
>-                                              uint8* dst_argb,
>+__declspec(naked) void ARGB1555ToARGBRow_AVX2(const uint8_t* src_argb1555,
>+                                              uint8_t* dst_argb,
>                                               int width) {
>   __asm {
>     mov        eax, 0x01080108  // generate multiplier to repeat 5 bits
>@@ -589,8 +589,8 @@ __declspec(naked) void ARGB1555ToARGBRow_AVX2(const uint8* src_argb1555,
> #endif  // HAS_ARGB1555TOARGBROW_AVX2
> 
> #ifdef HAS_ARGB4444TOARGBROW_AVX2
>-__declspec(naked) void ARGB4444ToARGBRow_AVX2(const uint8* src_argb4444,
>-                                              uint8* dst_argb,
>+__declspec(naked) void ARGB4444ToARGBRow_AVX2(const uint8_t* src_argb4444,
>+                                              uint8_t* dst_argb,
>                                               int width) {
>   __asm {
>     mov       eax,  0x0f0f0f0f  // generate mask 0x0f0f0f0f
>@@ -627,8 +627,8 @@ __declspec(naked) void ARGB4444ToARGBRow_AVX2(const uint8* src_argb4444,
> #endif  // HAS_ARGB4444TOARGBROW_AVX2
> 
> // 24 instructions
>-__declspec(naked) void ARGB1555ToARGBRow_SSE2(const uint8* src_argb1555,
>-                                              uint8* dst_argb,
>+__declspec(naked) void ARGB1555ToARGBRow_SSE2(const uint8_t* src_argb1555,
>+                                              uint8_t* dst_argb,
>                                               int width) {
>   __asm {
>     mov       eax, 0x01080108  // generate multiplier to repeat 5 bits
>@@ -680,8 +680,8 @@ __declspec(naked) void ARGB1555ToARGBRow_SSE2(const uint8* src_argb1555,
> }
> 
> // 18 instructions.
>-__declspec(naked) void ARGB4444ToARGBRow_SSE2(const uint8* src_argb4444,
>-                                              uint8* dst_argb,
>+__declspec(naked) void ARGB4444ToARGBRow_SSE2(const uint8_t* src_argb4444,
>+                                              uint8_t* dst_argb,
>                                               int width) {
>   __asm {
>     mov       eax, 0x0f0f0f0f  // generate mask 0x0f0f0f0f
>@@ -718,8 +718,8 @@ __declspec(naked) void ARGB4444ToARGBRow_SSE2(const uint8* src_argb4444,
>   }
> }
> 
>-__declspec(naked) void ARGBToRGB24Row_SSSE3(const uint8* src_argb,
>-                                            uint8* dst_rgb,
>+__declspec(naked) void ARGBToRGB24Row_SSSE3(const uint8_t* src_argb,
>+                                            uint8_t* dst_rgb,
>                                             int width) {
>   __asm {
>     mov       eax, [esp + 4]  // src_argb
>@@ -757,8 +757,8 @@ __declspec(naked) void ARGBToRGB24Row_SSSE3(const uint8* src_argb,
>   }
> }
> 
>-__declspec(naked) void ARGBToRAWRow_SSSE3(const uint8* src_argb,
>-                                          uint8* dst_rgb,
>+__declspec(naked) void ARGBToRAWRow_SSSE3(const uint8_t* src_argb,
>+                                          uint8_t* dst_rgb,
>                                           int width) {
>   __asm {
>     mov       eax, [esp + 4]  // src_argb
>@@ -796,8 +796,8 @@ __declspec(naked) void ARGBToRAWRow_SSSE3(const uint8* src_argb,
>   }
> }
> 
>-__declspec(naked) void ARGBToRGB565Row_SSE2(const uint8* src_argb,
>-                                            uint8* dst_rgb,
>+__declspec(naked) void ARGBToRGB565Row_SSE2(const uint8_t* src_argb,
>+                                            uint8_t* dst_rgb,
>                                             int width) {
>   __asm {
>     mov       eax, [esp + 4]  // src_argb
>@@ -834,9 +834,9 @@ __declspec(naked) void ARGBToRGB565Row_SSE2(const uint8* src_argb,
>   }
> }
> 
>-__declspec(naked) void ARGBToRGB565DitherRow_SSE2(const uint8* src_argb,
>-                                                  uint8* dst_rgb,
>-                                                  const uint32 dither4,
>+__declspec(naked) void ARGBToRGB565DitherRow_SSE2(const uint8_t* src_argb,
>+                                                  uint8_t* dst_rgb,
>+                                                  const uint32_t dither4,
>                                                   int width) {
>   __asm {
> 
>@@ -881,9 +881,9 @@ __declspec(naked) void ARGBToRGB565DitherRow_SSE2(const uint8* src_argb,
> }
> 
> #ifdef HAS_ARGBTORGB565DITHERROW_AVX2
>-__declspec(naked) void ARGBToRGB565DitherRow_AVX2(const uint8* src_argb,
>-                                                  uint8* dst_rgb,
>-                                                  const uint32 dither4,
>+__declspec(naked) void ARGBToRGB565DitherRow_AVX2(const uint8_t* src_argb,
>+                                                  uint8_t* dst_rgb,
>+                                                  const uint32_t dither4,
>                                                   int width) {
>   __asm {
>     mov        eax, [esp + 4]  // src_argb
>@@ -925,8 +925,8 @@ __declspec(naked) void ARGBToRGB565DitherRow_AVX2(const uint8* src_argb,
> #endif  // HAS_ARGBTORGB565DITHERROW_AVX2
> 
> // TODO(fbarchard): Improve sign extension/packing.
>-__declspec(naked) void ARGBToARGB1555Row_SSE2(const uint8* src_argb,
>-                                              uint8* dst_rgb,
>+__declspec(naked) void ARGBToARGB1555Row_SSE2(const uint8_t* src_argb,
>+                                              uint8_t* dst_rgb,
>                                               int width) {
>   __asm {
>     mov       eax, [esp + 4]  // src_argb
>@@ -967,8 +967,8 @@ __declspec(naked) void ARGBToARGB1555Row_SSE2(const uint8* src_argb,
>   }
> }
> 
>-__declspec(naked) void ARGBToARGB4444Row_SSE2(const uint8* src_argb,
>-                                              uint8* dst_rgb,
>+__declspec(naked) void ARGBToARGB4444Row_SSE2(const uint8_t* src_argb,
>+                                              uint8_t* dst_rgb,
>                                               int width) {
>   __asm {
>     mov       eax, [esp + 4]  // src_argb
>@@ -998,8 +998,8 @@ __declspec(naked) void ARGBToARGB4444Row_SSE2(const uint8* src_argb,
> }
> 
> #ifdef HAS_ARGBTORGB565ROW_AVX2
>-__declspec(naked) void ARGBToRGB565Row_AVX2(const uint8* src_argb,
>-                                            uint8* dst_rgb,
>+__declspec(naked) void ARGBToRGB565Row_AVX2(const uint8_t* src_argb,
>+                                            uint8_t* dst_rgb,
>                                             int width) {
>   __asm {
>     mov        eax, [esp + 4]  // src_argb
>@@ -1036,8 +1036,8 @@ __declspec(naked) void ARGBToRGB565Row_AVX2(const uint8* src_argb,
> #endif  // HAS_ARGBTORGB565ROW_AVX2
> 
> #ifdef HAS_ARGBTOARGB1555ROW_AVX2
>-__declspec(naked) void ARGBToARGB1555Row_AVX2(const uint8* src_argb,
>-                                              uint8* dst_rgb,
>+__declspec(naked) void ARGBToARGB1555Row_AVX2(const uint8_t* src_argb,
>+                                              uint8_t* dst_rgb,
>                                               int width) {
>   __asm {
>     mov        eax, [esp + 4]  // src_argb
>@@ -1077,8 +1077,8 @@ __declspec(naked) void ARGBToARGB1555Row_AVX2(const uint8* src_argb,
> #endif  // HAS_ARGBTOARGB1555ROW_AVX2
> 
> #ifdef HAS_ARGBTOARGB4444ROW_AVX2
>-__declspec(naked) void ARGBToARGB4444Row_AVX2(const uint8* src_argb,
>-                                              uint8* dst_rgb,
>+__declspec(naked) void ARGBToARGB4444Row_AVX2(const uint8_t* src_argb,
>+                                              uint8_t* dst_rgb,
>                                               int width) {
>   __asm {
>     mov        eax, [esp + 4]  // src_argb
>@@ -1109,8 +1109,8 @@ __declspec(naked) void ARGBToARGB4444Row_AVX2(const uint8* src_argb,
> #endif  // HAS_ARGBTOARGB4444ROW_AVX2
> 
> // Convert 16 ARGB pixels (64 bytes) to 16 Y values.
>-__declspec(naked) void ARGBToYRow_SSSE3(const uint8* src_argb,
>-                                        uint8* dst_y,
>+__declspec(naked) void ARGBToYRow_SSSE3(const uint8_t* src_argb,
>+                                        uint8_t* dst_y,
>                                         int width) {
>   __asm {
>     mov        eax, [esp + 4] /* src_argb */
>@@ -1145,8 +1145,8 @@ __declspec(naked) void ARGBToYRow_SSSE3(const uint8* src_argb,
> 
> // Convert 16 ARGB pixels (64 bytes) to 16 YJ values.
> // Same as ARGBToYRow but different coefficients, no add 16, but do rounding.
>-__declspec(naked) void ARGBToYJRow_SSSE3(const uint8* src_argb,
>-                                         uint8* dst_y,
>+__declspec(naked) void ARGBToYJRow_SSSE3(const uint8_t* src_argb,
>+                                         uint8_t* dst_y,
>                                          int width) {
>   __asm {
>     mov        eax, [esp + 4] /* src_argb */
>@@ -1185,8 +1185,8 @@ __declspec(naked) void ARGBToYJRow_SSSE3(const uint8* src_argb,
> static const lvec32 kPermdARGBToY_AVX = {0, 4, 1, 5, 2, 6, 3, 7};
> 
> // Convert 32 ARGB pixels (128 bytes) to 32 Y values.
>-__declspec(naked) void ARGBToYRow_AVX2(const uint8* src_argb,
>-                                       uint8* dst_y,
>+__declspec(naked) void ARGBToYRow_AVX2(const uint8_t* src_argb,
>+                                       uint8_t* dst_y,
>                                        int width) {
>   __asm {
>     mov        eax, [esp + 4] /* src_argb */
>@@ -1225,8 +1225,8 @@ __declspec(naked) void ARGBToYRow_AVX2(const uint8* src_argb,
> 
> #ifdef HAS_ARGBTOYJROW_AVX2
> // Convert 32 ARGB pixels (128 bytes) to 32 Y values.
>-__declspec(naked) void ARGBToYJRow_AVX2(const uint8* src_argb,
>-                                        uint8* dst_y,
>+__declspec(naked) void ARGBToYJRow_AVX2(const uint8_t* src_argb,
>+                                        uint8_t* dst_y,
>                                         int width) {
>   __asm {
>     mov        eax, [esp + 4] /* src_argb */
>@@ -1265,8 +1265,8 @@ __declspec(naked) void ARGBToYJRow_AVX2(const uint8* src_argb,
> }
> #endif  //  HAS_ARGBTOYJROW_AVX2
> 
>-__declspec(naked) void BGRAToYRow_SSSE3(const uint8* src_argb,
>-                                        uint8* dst_y,
>+__declspec(naked) void BGRAToYRow_SSSE3(const uint8_t* src_argb,
>+                                        uint8_t* dst_y,
>                                         int width) {
>   __asm {
>     mov        eax, [esp + 4] /* src_argb */
>@@ -1299,8 +1299,8 @@ __declspec(naked) void BGRAToYRow_SSSE3(const uint8* src_argb,
>   }
> }
> 
>-__declspec(naked) void ABGRToYRow_SSSE3(const uint8* src_argb,
>-                                        uint8* dst_y,
>+__declspec(naked) void ABGRToYRow_SSSE3(const uint8_t* src_argb,
>+                                        uint8_t* dst_y,
>                                         int width) {
>   __asm {
>     mov        eax, [esp + 4] /* src_argb */
>@@ -1333,8 +1333,8 @@ __declspec(naked) void ABGRToYRow_SSSE3(const uint8* src_argb,
>   }
> }
> 
>-__declspec(naked) void RGBAToYRow_SSSE3(const uint8* src_argb,
>-                                        uint8* dst_y,
>+__declspec(naked) void RGBAToYRow_SSSE3(const uint8_t* src_argb,
>+                                        uint8_t* dst_y,
>                                         int width) {
>   __asm {
>     mov        eax, [esp + 4] /* src_argb */
>@@ -1367,10 +1367,10 @@ __declspec(naked) void RGBAToYRow_SSSE3(const uint8* src_argb,
>   }
> }
> 
>-__declspec(naked) void ARGBToUVRow_SSSE3(const uint8* src_argb0,
>+__declspec(naked) void ARGBToUVRow_SSSE3(const uint8_t* src_argb0,
>                                          int src_stride_argb,
>-                                         uint8* dst_u,
>-                                         uint8* dst_v,
>+                                         uint8_t* dst_u,
>+                                         uint8_t* dst_v,
>                                          int width) {
>   __asm {
>     push       esi
>@@ -1439,10 +1439,10 @@ __declspec(naked) void ARGBToUVRow_SSSE3(const uint8* src_argb0,
>   }
> }
> 
>-__declspec(naked) void ARGBToUVJRow_SSSE3(const uint8* src_argb0,
>+__declspec(naked) void ARGBToUVJRow_SSSE3(const uint8_t* src_argb0,
>                                           int src_stride_argb,
>-                                          uint8* dst_u,
>-                                          uint8* dst_v,
>+                                          uint8_t* dst_u,
>+                                          uint8_t* dst_v,
>                                           int width) {
>   __asm {
>     push       esi
>@@ -1513,10 +1513,10 @@ __declspec(naked) void ARGBToUVJRow_SSSE3(const uint8* src_argb0,
> }
> 
> #ifdef HAS_ARGBTOUVROW_AVX2
>-__declspec(naked) void ARGBToUVRow_AVX2(const uint8* src_argb0,
>+__declspec(naked) void ARGBToUVRow_AVX2(const uint8_t* src_argb0,
>                                         int src_stride_argb,
>-                                        uint8* dst_u,
>-                                        uint8* dst_v,
>+                                        uint8_t* dst_u,
>+                                        uint8_t* dst_v,
>                                         int width) {
>   __asm {
>     push       esi
>@@ -1581,10 +1581,10 @@ __declspec(naked) void ARGBToUVRow_AVX2(const uint8* src_argb0,
> #endif  // HAS_ARGBTOUVROW_AVX2
> 
> #ifdef HAS_ARGBTOUVJROW_AVX2
>-__declspec(naked) void ARGBToUVJRow_AVX2(const uint8* src_argb0,
>+__declspec(naked) void ARGBToUVJRow_AVX2(const uint8_t* src_argb0,
>                                          int src_stride_argb,
>-                                         uint8* dst_u,
>-                                         uint8* dst_v,
>+                                         uint8_t* dst_u,
>+                                         uint8_t* dst_v,
>                                          int width) {
>   __asm {
>     push       esi
>@@ -1649,9 +1649,9 @@ __declspec(naked) void ARGBToUVJRow_AVX2(const uint8* src_argb0,
> }
> #endif  // HAS_ARGBTOUVJROW_AVX2
> 
>-__declspec(naked) void ARGBToUV444Row_SSSE3(const uint8* src_argb0,
>-                                            uint8* dst_u,
>-                                            uint8* dst_v,
>+__declspec(naked) void ARGBToUV444Row_SSSE3(const uint8_t* src_argb0,
>+                                            uint8_t* dst_u,
>+                                            uint8_t* dst_v,
>                                             int width) {
>   __asm {
>     push       edi
>@@ -1707,10 +1707,10 @@ __declspec(naked) void ARGBToUV444Row_SSSE3(const uint8* src_argb0,
>   }
> }
> 
>-__declspec(naked) void BGRAToUVRow_SSSE3(const uint8* src_argb0,
>+__declspec(naked) void BGRAToUVRow_SSSE3(const uint8_t* src_argb0,
>                                          int src_stride_argb,
>-                                         uint8* dst_u,
>-                                         uint8* dst_v,
>+                                         uint8_t* dst_u,
>+                                         uint8_t* dst_v,
>                                          int width) {
>   __asm {
>     push       esi
>@@ -1779,10 +1779,10 @@ __declspec(naked) void BGRAToUVRow_SSSE3(const uint8* src_argb0,
>   }
> }
> 
>-__declspec(naked) void ABGRToUVRow_SSSE3(const uint8* src_argb0,
>+__declspec(naked) void ABGRToUVRow_SSSE3(const uint8_t* src_argb0,
>                                          int src_stride_argb,
>-                                         uint8* dst_u,
>-                                         uint8* dst_v,
>+                                         uint8_t* dst_u,
>+                                         uint8_t* dst_v,
>                                          int width) {
>   __asm {
>     push       esi
>@@ -1851,10 +1851,10 @@ __declspec(naked) void ABGRToUVRow_SSSE3(const uint8* src_argb0,
>   }
> }
> 
>-__declspec(naked) void RGBAToUVRow_SSSE3(const uint8* src_argb0,
>+__declspec(naked) void RGBAToUVRow_SSSE3(const uint8_t* src_argb0,
>                                          int src_stride_argb,
>-                                         uint8* dst_u,
>-                                         uint8* dst_v,
>+                                         uint8_t* dst_u,
>+                                         uint8_t* dst_v,
>                                          int width) {
>   __asm {
>     push       esi
>@@ -2065,10 +2065,10 @@ __declspec(naked) void RGBAToUVRow_SSSE3(const uint8* src_argb0,
> // 16 pixels
> // 8 UV values upsampled to 16 UV, mixed with 16 Y producing 16 ARGB (64 bytes).
> __declspec(naked) void I422ToARGBRow_AVX2(
>-    const uint8* y_buf,
>-    const uint8* u_buf,
>-    const uint8* v_buf,
>-    uint8* dst_argb,
>+    const uint8_t* y_buf,
>+    const uint8_t* u_buf,
>+    const uint8_t* v_buf,
>+    uint8_t* dst_argb,
>     const struct YuvConstants* yuvconstants,
>     int width) {
>   __asm {
>@@ -2105,11 +2105,11 @@ __declspec(naked) void I422ToARGBRow_AVX2(
> // 16 pixels
> // 8 UV values upsampled to 16 UV, mixed with 16 Y and 16 A producing 16 ARGB.
> __declspec(naked) void I422AlphaToARGBRow_AVX2(
>-    const uint8* y_buf,
>-    const uint8* u_buf,
>-    const uint8* v_buf,
>-    const uint8* a_buf,
>-    uint8* dst_argb,
>+    const uint8_t* y_buf,
>+    const uint8_t* u_buf,
>+    const uint8_t* v_buf,
>+    const uint8_t* a_buf,
>+    uint8_t* dst_argb,
>     const struct YuvConstants* yuvconstants,
>     int width) {
>   __asm {
>@@ -2148,10 +2148,10 @@ __declspec(naked) void I422AlphaToARGBRow_AVX2(
> // 16 pixels
> // 16 UV values with 16 Y producing 16 ARGB (64 bytes).
> __declspec(naked) void I444ToARGBRow_AVX2(
>-    const uint8* y_buf,
>-    const uint8* u_buf,
>-    const uint8* v_buf,
>-    uint8* dst_argb,
>+    const uint8_t* y_buf,
>+    const uint8_t* u_buf,
>+    const uint8_t* v_buf,
>+    uint8_t* dst_argb,
>     const struct YuvConstants* yuvconstants,
>     int width) {
>   __asm {
>@@ -2187,9 +2187,9 @@ __declspec(naked) void I444ToARGBRow_AVX2(
> // 16 pixels.
> // 8 UV values upsampled to 16 UV, mixed with 16 Y producing 16 ARGB (64 bytes).
> __declspec(naked) void NV12ToARGBRow_AVX2(
>-    const uint8* y_buf,
>-    const uint8* uv_buf,
>-    uint8* dst_argb,
>+    const uint8_t* y_buf,
>+    const uint8_t* uv_buf,
>+    uint8_t* dst_argb,
>     const struct YuvConstants* yuvconstants,
>     int width) {
>   __asm {
>@@ -2222,9 +2222,9 @@ __declspec(naked) void NV12ToARGBRow_AVX2(
> // 16 pixels.
> // 8 VU values upsampled to 16 UV, mixed with 16 Y producing 16 ARGB (64 bytes).
> __declspec(naked) void NV21ToARGBRow_AVX2(
>-    const uint8* y_buf,
>-    const uint8* vu_buf,
>-    uint8* dst_argb,
>+    const uint8_t* y_buf,
>+    const uint8_t* vu_buf,
>+    uint8_t* dst_argb,
>     const struct YuvConstants* yuvconstants,
>     int width) {
>   __asm {
>@@ -2257,8 +2257,8 @@ __declspec(naked) void NV21ToARGBRow_AVX2(
> // 16 pixels.
> // 8 YUY2 values with 16 Y and 8 UV producing 16 ARGB (64 bytes).
> __declspec(naked) void YUY2ToARGBRow_AVX2(
>-    const uint8* src_yuy2,
>-    uint8* dst_argb,
>+    const uint8_t* src_yuy2,
>+    uint8_t* dst_argb,
>     const struct YuvConstants* yuvconstants,
>     int width) {
>   __asm {
>@@ -2288,8 +2288,8 @@ __declspec(naked) void YUY2ToARGBRow_AVX2(
> // 16 pixels.
> // 8 UYVY values with 16 Y and 8 UV producing 16 ARGB (64 bytes).
> __declspec(naked) void UYVYToARGBRow_AVX2(
>-    const uint8* src_uyvy,
>-    uint8* dst_argb,
>+    const uint8_t* src_uyvy,
>+    uint8_t* dst_argb,
>     const struct YuvConstants* yuvconstants,
>     int width) {
>   __asm {
>@@ -2319,10 +2319,10 @@ __declspec(naked) void UYVYToARGBRow_AVX2(
> // 16 pixels
> // 8 UV values upsampled to 16 UV, mixed with 16 Y producing 16 RGBA (64 bytes).
> __declspec(naked) void I422ToRGBARow_AVX2(
>-    const uint8* y_buf,
>-    const uint8* u_buf,
>-    const uint8* v_buf,
>-    uint8* dst_argb,
>+    const uint8_t* y_buf,
>+    const uint8_t* u_buf,
>+    const uint8_t* v_buf,
>+    uint8_t* dst_argb,
>     const struct YuvConstants* yuvconstants,
>     int width) {
>   __asm {
>@@ -2551,10 +2551,10 @@ __declspec(naked) void I422ToRGBARow_AVX2(
> // 8 pixels.
> // 8 UV values, mixed with 8 Y producing 8 ARGB (32 bytes).
> __declspec(naked) void I444ToARGBRow_SSSE3(
>-    const uint8* y_buf,
>-    const uint8* u_buf,
>-    const uint8* v_buf,
>-    uint8* dst_argb,
>+    const uint8_t* y_buf,
>+    const uint8_t* u_buf,
>+    const uint8_t* v_buf,
>+    uint8_t* dst_argb,
>     const struct YuvConstants* yuvconstants,
>     int width) {
>   __asm {
>@@ -2588,10 +2588,10 @@ __declspec(naked) void I444ToARGBRow_SSSE3(
> // 8 pixels.
> // 4 UV values upsampled to 8 UV, mixed with 8 Y producing 8 RGB24 (24 bytes).
> __declspec(naked) void I422ToRGB24Row_SSSE3(
>-    const uint8* y_buf,
>-    const uint8* u_buf,
>-    const uint8* v_buf,
>-    uint8* dst_rgb24,
>+    const uint8_t* y_buf,
>+    const uint8_t* u_buf,
>+    const uint8_t* v_buf,
>+    uint8_t* dst_rgb24,
>     const struct YuvConstants* yuvconstants,
>     int width) {
>   __asm {
>@@ -2626,10 +2626,10 @@ __declspec(naked) void I422ToRGB24Row_SSSE3(
> // 8 pixels
> // 4 UV values upsampled to 8 UV, mixed with 8 Y producing 8 RGB565 (16 bytes).
> __declspec(naked) void I422ToRGB565Row_SSSE3(
>-    const uint8* y_buf,
>-    const uint8* u_buf,
>-    const uint8* v_buf,
>-    uint8* rgb565_buf,
>+    const uint8_t* y_buf,
>+    const uint8_t* u_buf,
>+    const uint8_t* v_buf,
>+    uint8_t* rgb565_buf,
>     const struct YuvConstants* yuvconstants,
>     int width) {
>   __asm {
>@@ -2669,10 +2669,10 @@ __declspec(naked) void I422ToRGB565Row_SSSE3(
> // 8 pixels.
> // 4 UV values upsampled to 8 UV, mixed with 8 Y producing 8 ARGB (32 bytes).
> __declspec(naked) void I422ToARGBRow_SSSE3(
>-    const uint8* y_buf,
>-    const uint8* u_buf,
>-    const uint8* v_buf,
>-    uint8* dst_argb,
>+    const uint8_t* y_buf,
>+    const uint8_t* u_buf,
>+    const uint8_t* v_buf,
>+    uint8_t* dst_argb,
>     const struct YuvConstants* yuvconstants,
>     int width) {
>   __asm {
>@@ -2706,11 +2706,11 @@ __declspec(naked) void I422ToARGBRow_SSSE3(
> // 8 pixels.
> // 4 UV values upsampled to 8 UV, mixed with 8 Y and 8 A producing 8 ARGB.
> __declspec(naked) void I422AlphaToARGBRow_SSSE3(
>-    const uint8* y_buf,
>-    const uint8* u_buf,
>-    const uint8* v_buf,
>-    const uint8* a_buf,
>-    uint8* dst_argb,
>+    const uint8_t* y_buf,
>+    const uint8_t* u_buf,
>+    const uint8_t* v_buf,
>+    const uint8_t* a_buf,
>+    uint8_t* dst_argb,
>     const struct YuvConstants* yuvconstants,
>     int width) {
>   __asm {
>@@ -2746,9 +2746,9 @@ __declspec(naked) void I422AlphaToARGBRow_SSSE3(
> // 8 pixels.
> // 4 UV values upsampled to 8 UV, mixed with 8 Y producing 8 ARGB (32 bytes).
> __declspec(naked) void NV12ToARGBRow_SSSE3(
>-    const uint8* y_buf,
>-    const uint8* uv_buf,
>-    uint8* dst_argb,
>+    const uint8_t* y_buf,
>+    const uint8_t* uv_buf,
>+    uint8_t* dst_argb,
>     const struct YuvConstants* yuvconstants,
>     int width) {
>   __asm {
>@@ -2778,9 +2778,9 @@ __declspec(naked) void NV12ToARGBRow_SSSE3(
> // 8 pixels.
> // 4 UV values upsampled to 8 UV, mixed with 8 Y producing 8 ARGB (32 bytes).
> __declspec(naked) void NV21ToARGBRow_SSSE3(
>-    const uint8* y_buf,
>-    const uint8* vu_buf,
>-    uint8* dst_argb,
>+    const uint8_t* y_buf,
>+    const uint8_t* vu_buf,
>+    uint8_t* dst_argb,
>     const struct YuvConstants* yuvconstants,
>     int width) {
>   __asm {
>@@ -2810,8 +2810,8 @@ __declspec(naked) void NV21ToARGBRow_SSSE3(
> // 8 pixels.
> // 4 YUY2 values with 8 Y and 4 UV producing 8 ARGB (32 bytes).
> __declspec(naked) void YUY2ToARGBRow_SSSE3(
>-    const uint8* src_yuy2,
>-    uint8* dst_argb,
>+    const uint8_t* src_yuy2,
>+    uint8_t* dst_argb,
>     const struct YuvConstants* yuvconstants,
>     int width) {
>   __asm {
>@@ -2838,8 +2838,8 @@ __declspec(naked) void YUY2ToARGBRow_SSSE3(
> // 8 pixels.
> // 4 UYVY values with 8 Y and 4 UV producing 8 ARGB (32 bytes).
> __declspec(naked) void UYVYToARGBRow_SSSE3(
>-    const uint8* src_uyvy,
>-    uint8* dst_argb,
>+    const uint8_t* src_uyvy,
>+    uint8_t* dst_argb,
>     const struct YuvConstants* yuvconstants,
>     int width) {
>   __asm {
>@@ -2864,10 +2864,10 @@ __declspec(naked) void UYVYToARGBRow_SSSE3(
> }
> 
> __declspec(naked) void I422ToRGBARow_SSSE3(
>-    const uint8* y_buf,
>-    const uint8* u_buf,
>-    const uint8* v_buf,
>-    uint8* dst_rgba,
>+    const uint8_t* y_buf,
>+    const uint8_t* u_buf,
>+    const uint8_t* v_buf,
>+    uint8_t* dst_rgba,
>     const struct YuvConstants* yuvconstants,
>     int width) {
>   __asm {
>@@ -2900,8 +2900,8 @@ __declspec(naked) void I422ToRGBARow_SSSE3(
> 
> #ifdef HAS_I400TOARGBROW_SSE2
> // 8 pixels of Y converted to 8 pixels of ARGB (32 bytes).
>-__declspec(naked) void I400ToARGBRow_SSE2(const uint8* y_buf,
>-                                          uint8* rgb_buf,
>+__declspec(naked) void I400ToARGBRow_SSE2(const uint8_t* y_buf,
>+                                          uint8_t* rgb_buf,
>                                           int width) {
>   __asm {
>     mov        eax, 0x4a354a35  // 4a35 = 18997 = round(1.164 * 64 * 256)
>@@ -2947,8 +2947,8 @@ __declspec(naked) void I400ToARGBRow_SSE2(const uint8* y_buf,
> #ifdef HAS_I400TOARGBROW_AVX2
> // 16 pixels of Y converted to 16 pixels of ARGB (64 bytes).
> // note: vpunpcklbw mutates and vpackuswb unmutates.
>-__declspec(naked) void I400ToARGBRow_AVX2(const uint8* y_buf,
>-                                          uint8* rgb_buf,
>+__declspec(naked) void I400ToARGBRow_AVX2(const uint8_t* y_buf,
>+                                          uint8_t* rgb_buf,
>                                           int width) {
>   __asm {
>     mov        eax, 0x4a354a35  // 4a35 = 18997 = round(1.164 * 64 * 256)
>@@ -3000,8 +3000,8 @@ static const uvec8 kShuffleMirror = {15u, 14u, 13u, 12u, 11u, 10u, 9u, 8u,
>                                      7u,  6u,  5u,  4u,  3u,  2u,  1u, 0u};
> 
> // TODO(fbarchard): Replace lea with -16 offset.
>-__declspec(naked) void MirrorRow_SSSE3(const uint8* src,
>-                                       uint8* dst,
>+__declspec(naked) void MirrorRow_SSSE3(const uint8_t* src,
>+                                       uint8_t* dst,
>                                        int width) {
>   __asm {
>     mov       eax, [esp + 4]  // src
>@@ -3022,7 +3022,9 @@ __declspec(naked) void MirrorRow_SSSE3(const uint8* src,
> #endif  // HAS_MIRRORROW_SSSE3
> 
> #ifdef HAS_MIRRORROW_AVX2
>-__declspec(naked) void MirrorRow_AVX2(const uint8* src, uint8* dst, int width) {
>+__declspec(naked) void MirrorRow_AVX2(const uint8_t* src,
>+                                      uint8_t* dst,
>+                                      int width) {
>   __asm {
>     mov       eax, [esp + 4]  // src
>     mov       edx, [esp + 8]  // dst
>@@ -3048,9 +3050,9 @@ __declspec(naked) void MirrorRow_AVX2(const uint8* src, uint8* dst, int width) {
> static const uvec8 kShuffleMirrorUV = {14u, 12u, 10u, 8u, 6u, 4u, 2u, 0u,
>                                        15u, 13u, 11u, 9u, 7u, 5u, 3u, 1u};
> 
>-__declspec(naked) void MirrorUVRow_SSSE3(const uint8* src,
>-                                         uint8* dst_u,
>-                                         uint8* dst_v,
>+__declspec(naked) void MirrorUVRow_SSSE3(const uint8_t* src,
>+                                         uint8_t* dst_u,
>+                                         uint8_t* dst_v,
>                                          int width) {
>   __asm {
>     push      edi
>@@ -3079,8 +3081,8 @@ __declspec(naked) void MirrorUVRow_SSSE3(const uint8* src,
> #endif  // HAS_MIRRORUVROW_SSSE3
> 
> #ifdef HAS_ARGBMIRRORROW_SSE2
>-__declspec(naked) void ARGBMirrorRow_SSE2(const uint8* src,
>-                                          uint8* dst,
>+__declspec(naked) void ARGBMirrorRow_SSE2(const uint8_t* src,
>+                                          uint8_t* dst,
>                                           int width) {
>   __asm {
>     mov       eax, [esp + 4]  // src
>@@ -3105,8 +3107,8 @@ __declspec(naked) void ARGBMirrorRow_SSE2(const uint8* src,
> // Shuffle table for reversing the bytes.
> static const ulvec32 kARGBShuffleMirror_AVX2 = {7u, 6u, 5u, 4u, 3u, 2u, 1u, 0u};
> 
>-__declspec(naked) void ARGBMirrorRow_AVX2(const uint8* src,
>-                                          uint8* dst,
>+__declspec(naked) void ARGBMirrorRow_AVX2(const uint8_t* src,
>+                                          uint8_t* dst,
>                                           int width) {
>   __asm {
>     mov       eax, [esp + 4]  // src
>@@ -3127,9 +3129,9 @@ __declspec(naked) void ARGBMirrorRow_AVX2(const uint8* src,
> #endif  // HAS_ARGBMIRRORROW_AVX2
> 
> #ifdef HAS_SPLITUVROW_SSE2
>-__declspec(naked) void SplitUVRow_SSE2(const uint8* src_uv,
>-                                       uint8* dst_u,
>-                                       uint8* dst_v,
>+__declspec(naked) void SplitUVRow_SSE2(const uint8_t* src_uv,
>+                                       uint8_t* dst_u,
>+                                       uint8_t* dst_v,
>                                        int width) {
>   __asm {
>     push       edi
>@@ -3167,9 +3169,9 @@ __declspec(naked) void SplitUVRow_SSE2(const uint8* src_uv,
> #endif  // HAS_SPLITUVROW_SSE2
> 
> #ifdef HAS_SPLITUVROW_AVX2
>-__declspec(naked) void SplitUVRow_AVX2(const uint8* src_uv,
>-                                       uint8* dst_u,
>-                                       uint8* dst_v,
>+__declspec(naked) void SplitUVRow_AVX2(const uint8_t* src_uv,
>+                                       uint8_t* dst_u,
>+                                       uint8_t* dst_v,
>                                        int width) {
>   __asm {
>     push       edi
>@@ -3207,9 +3209,9 @@ __declspec(naked) void SplitUVRow_AVX2(const uint8* src_uv,
> #endif  // HAS_SPLITUVROW_AVX2
> 
> #ifdef HAS_MERGEUVROW_SSE2
>-__declspec(naked) void MergeUVRow_SSE2(const uint8* src_u,
>-                                       const uint8* src_v,
>-                                       uint8* dst_uv,
>+__declspec(naked) void MergeUVRow_SSE2(const uint8_t* src_u,
>+                                       const uint8_t* src_v,
>+                                       uint8_t* dst_uv,
>                                        int width) {
>   __asm {
>     push       edi
>@@ -3239,9 +3241,9 @@ __declspec(naked) void MergeUVRow_SSE2(const uint8* src_u,
> #endif  //  HAS_MERGEUVROW_SSE2
> 
> #ifdef HAS_MERGEUVROW_AVX2
>-__declspec(naked) void MergeUVRow_AVX2(const uint8* src_u,
>-                                       const uint8* src_v,
>-                                       uint8* dst_uv,
>+__declspec(naked) void MergeUVRow_AVX2(const uint8_t* src_u,
>+                                       const uint8_t* src_v,
>+                                       uint8_t* dst_uv,
>                                        int width) {
>   __asm {
>     push       edi
>@@ -3273,12 +3275,14 @@ __declspec(naked) void MergeUVRow_AVX2(const uint8* src_u,
> #endif  //  HAS_MERGEUVROW_AVX2
> 
> #ifdef HAS_COPYROW_SSE2
>-// CopyRow copys 'count' bytes using a 16 byte load/store, 32 bytes at time.
>-__declspec(naked) void CopyRow_SSE2(const uint8* src, uint8* dst, int count) {
>+// CopyRow copys 'width' bytes using a 16 byte load/store, 32 bytes at time.
>+__declspec(naked) void CopyRow_SSE2(const uint8_t* src,
>+                                    uint8_t* dst,
>+                                    int width) {
>   __asm {
>     mov        eax, [esp + 4]  // src
>     mov        edx, [esp + 8]  // dst
>-    mov        ecx, [esp + 12]  // count
>+    mov        ecx, [esp + 12]  // width
>     test       eax, 15
>     jne        convertloopu
>     test       edx, 15
>@@ -3310,12 +3314,14 @@ __declspec(naked) void CopyRow_SSE2(const uint8* src, uint8* dst, int count) {
> #endif  // HAS_COPYROW_SSE2
> 
> #ifdef HAS_COPYROW_AVX
>-// CopyRow copys 'count' bytes using a 32 byte load/store, 64 bytes at time.
>-__declspec(naked) void CopyRow_AVX(const uint8* src, uint8* dst, int count) {
>+// CopyRow copys 'width' bytes using a 32 byte load/store, 64 bytes at time.
>+__declspec(naked) void CopyRow_AVX(const uint8_t* src,
>+                                   uint8_t* dst,
>+                                   int width) {
>   __asm {
>     mov        eax, [esp + 4]  // src
>     mov        edx, [esp + 8]  // dst
>-    mov        ecx, [esp + 12]  // count
>+    mov        ecx, [esp + 12]  // width
> 
>   convertloop:
>     vmovdqu    ymm0, [eax]
>@@ -3334,13 +3340,15 @@ __declspec(naked) void CopyRow_AVX(const uint8* src, uint8* dst, int count) {
> #endif  // HAS_COPYROW_AVX
> 
> // Multiple of 1.
>-__declspec(naked) void CopyRow_ERMS(const uint8* src, uint8* dst, int count) {
>+__declspec(naked) void CopyRow_ERMS(const uint8_t* src,
>+                                    uint8_t* dst,
>+                                    int width) {
>   __asm {
>     mov        eax, esi
>     mov        edx, edi
>     mov        esi, [esp + 4]  // src
>     mov        edi, [esp + 8]  // dst
>-    mov        ecx, [esp + 12]  // count
>+    mov        ecx, [esp + 12]  // width
>     rep movsb
>     mov        edi, edx
>     mov        esi, eax
>@@ -3350,13 +3358,13 @@ __declspec(naked) void CopyRow_ERMS(const uint8* src, uint8* dst, int count) {
> 
> #ifdef HAS_ARGBCOPYALPHAROW_SSE2
> // width in pixels
>-__declspec(naked) void ARGBCopyAlphaRow_SSE2(const uint8* src,
>-                                             uint8* dst,
>+__declspec(naked) void ARGBCopyAlphaRow_SSE2(const uint8_t* src,
>+                                             uint8_t* dst,
>                                              int width) {
>   __asm {
>     mov        eax, [esp + 4]  // src
>     mov        edx, [esp + 8]  // dst
>-    mov        ecx, [esp + 12]  // count
>+    mov        ecx, [esp + 12]  // width
>     pcmpeqb    xmm0, xmm0  // generate mask 0xff000000
>     pslld      xmm0, 24
>     pcmpeqb    xmm1, xmm1  // generate mask 0x00ffffff
>@@ -3387,13 +3395,13 @@ __declspec(naked) void ARGBCopyAlphaRow_SSE2(const uint8* src,
> 
> #ifdef HAS_ARGBCOPYALPHAROW_AVX2
> // width in pixels
>-__declspec(naked) void ARGBCopyAlphaRow_AVX2(const uint8* src,
>-                                             uint8* dst,
>+__declspec(naked) void ARGBCopyAlphaRow_AVX2(const uint8_t* src,
>+                                             uint8_t* dst,
>                                              int width) {
>   __asm {
>     mov        eax, [esp + 4]  // src
>     mov        edx, [esp + 8]  // dst
>-    mov        ecx, [esp + 12]  // count
>+    mov        ecx, [esp + 12]  // width
>     vpcmpeqb   ymm0, ymm0, ymm0
>     vpsrld     ymm0, ymm0, 8  // generate mask 0x00ffffff
> 
>@@ -3417,8 +3425,8 @@ __declspec(naked) void ARGBCopyAlphaRow_AVX2(const uint8* src,
> 
> #ifdef HAS_ARGBEXTRACTALPHAROW_SSE2
> // width in pixels
>-__declspec(naked) void ARGBExtractAlphaRow_SSE2(const uint8* src_argb,
>-                                                uint8* dst_a,
>+__declspec(naked) void ARGBExtractAlphaRow_SSE2(const uint8_t* src_argb,
>+                                                uint8_t* dst_a,
>                                                 int width) {
>   __asm {
>     mov        eax, [esp + 4]  // src_argb
>@@ -3445,8 +3453,8 @@ __declspec(naked) void ARGBExtractAlphaRow_SSE2(const uint8* src_argb,
> 
> #ifdef HAS_ARGBEXTRACTALPHAROW_AVX2
> // width in pixels
>-__declspec(naked) void ARGBExtractAlphaRow_AVX2(const uint8* src_argb,
>-                                                uint8* dst_a,
>+__declspec(naked) void ARGBExtractAlphaRow_AVX2(const uint8_t* src_argb,
>+                                                uint8_t* dst_a,
>                                                 int width) {
>   __asm {
>     mov        eax, [esp + 4]  // src_argb
>@@ -3481,13 +3489,13 @@ __declspec(naked) void ARGBExtractAlphaRow_AVX2(const uint8* src_argb,
> 
> #ifdef HAS_ARGBCOPYYTOALPHAROW_SSE2
> // width in pixels
>-__declspec(naked) void ARGBCopyYToAlphaRow_SSE2(const uint8* src,
>-                                                uint8* dst,
>+__declspec(naked) void ARGBCopyYToAlphaRow_SSE2(const uint8_t* src,
>+                                                uint8_t* dst,
>                                                 int width) {
>   __asm {
>     mov        eax, [esp + 4]  // src
>     mov        edx, [esp + 8]  // dst
>-    mov        ecx, [esp + 12]  // count
>+    mov        ecx, [esp + 12]  // width
>     pcmpeqb    xmm0, xmm0  // generate mask 0xff000000
>     pslld      xmm0, 24
>     pcmpeqb    xmm1, xmm1  // generate mask 0x00ffffff
>@@ -3520,13 +3528,13 @@ __declspec(naked) void ARGBCopyYToAlphaRow_SSE2(const uint8* src,
> 
> #ifdef HAS_ARGBCOPYYTOALPHAROW_AVX2
> // width in pixels
>-__declspec(naked) void ARGBCopyYToAlphaRow_AVX2(const uint8* src,
>-                                                uint8* dst,
>+__declspec(naked) void ARGBCopyYToAlphaRow_AVX2(const uint8_t* src,
>+                                                uint8_t* dst,
>                                                 int width) {
>   __asm {
>     mov        eax, [esp + 4]  // src
>     mov        edx, [esp + 8]  // dst
>-    mov        ecx, [esp + 12]  // count
>+    mov        ecx, [esp + 12]  // width
>     vpcmpeqb   ymm0, ymm0, ymm0
>     vpsrld     ymm0, ymm0, 8  // generate mask 0x00ffffff
> 
>@@ -3551,16 +3559,16 @@ __declspec(naked) void ARGBCopyYToAlphaRow_AVX2(const uint8* src,
> #endif  // HAS_ARGBCOPYYTOALPHAROW_AVX2
> 
> #ifdef HAS_SETROW_X86
>-// Write 'count' bytes using an 8 bit value repeated.
>-// Count should be multiple of 4.
>-__declspec(naked) void SetRow_X86(uint8* dst, uint8 v8, int count) {
>+// Write 'width' bytes using an 8 bit value repeated.
>+// width should be multiple of 4.
>+__declspec(naked) void SetRow_X86(uint8_t* dst, uint8_t v8, int width) {
>   __asm {
>     movzx      eax, byte ptr [esp + 8]  // v8
>     mov        edx, 0x01010101  // Duplicate byte to all bytes.
>     mul        edx  // overwrites edx with upper part of result.
>     mov        edx, edi
>     mov        edi, [esp + 4]  // dst
>-    mov        ecx, [esp + 12]  // count
>+    mov        ecx, [esp + 12]  // width
>     shr        ecx, 2
>     rep stosd
>     mov        edi, edx
>@@ -3568,26 +3576,28 @@ __declspec(naked) void SetRow_X86(uint8* dst, uint8 v8, int count) {
>   }
> }
> 
>-// Write 'count' bytes using an 8 bit value repeated.
>-__declspec(naked) void SetRow_ERMS(uint8* dst, uint8 v8, int count) {
>+// Write 'width' bytes using an 8 bit value repeated.
>+__declspec(naked) void SetRow_ERMS(uint8_t* dst, uint8_t v8, int width) {
>   __asm {
>     mov        edx, edi
>     mov        edi, [esp + 4]  // dst
>     mov        eax, [esp + 8]  // v8
>-    mov        ecx, [esp + 12]  // count
>+    mov        ecx, [esp + 12]  // width
>     rep stosb
>     mov        edi, edx
>     ret
>   }
> }
> 
>-// Write 'count' 32 bit values.
>-__declspec(naked) void ARGBSetRow_X86(uint8* dst_argb, uint32 v32, int count) {
>+// Write 'width' 32 bit values.
>+__declspec(naked) void ARGBSetRow_X86(uint8_t* dst_argb,
>+                                      uint32_t v32,
>+                                      int width) {
>   __asm {
>     mov        edx, edi
>     mov        edi, [esp + 4]  // dst
>     mov        eax, [esp + 8]  // v32
>-    mov        ecx, [esp + 12]  // count
>+    mov        ecx, [esp + 12]  // width
>     rep stosd
>     mov        edi, edx
>     ret
>@@ -3596,8 +3606,8 @@ __declspec(naked) void ARGBSetRow_X86(uint8* dst_argb, uint32 v32, int count) {
> #endif  // HAS_SETROW_X86
> 
> #ifdef HAS_YUY2TOYROW_AVX2
>-__declspec(naked) void YUY2ToYRow_AVX2(const uint8* src_yuy2,
>-                                       uint8* dst_y,
>+__declspec(naked) void YUY2ToYRow_AVX2(const uint8_t* src_yuy2,
>+                                       uint8_t* dst_y,
>                                        int width) {
>   __asm {
>     mov        eax, [esp + 4]  // src_yuy2
>@@ -3623,10 +3633,10 @@ __declspec(naked) void YUY2ToYRow_AVX2(const uint8* src_yuy2,
>   }
> }
> 
>-__declspec(naked) void YUY2ToUVRow_AVX2(const uint8* src_yuy2,
>+__declspec(naked) void YUY2ToUVRow_AVX2(const uint8_t* src_yuy2,
>                                         int stride_yuy2,
>-                                        uint8* dst_u,
>-                                        uint8* dst_v,
>+                                        uint8_t* dst_u,
>+                                        uint8_t* dst_v,
>                                         int width) {
>   __asm {
>     push       esi
>@@ -3669,9 +3679,9 @@ __declspec(naked) void YUY2ToUVRow_AVX2(const uint8* src_yuy2,
>   }
> }
> 
>-__declspec(naked) void YUY2ToUV422Row_AVX2(const uint8* src_yuy2,
>-                                           uint8* dst_u,
>-                                           uint8* dst_v,
>+__declspec(naked) void YUY2ToUV422Row_AVX2(const uint8_t* src_yuy2,
>+                                           uint8_t* dst_u,
>+                                           uint8_t* dst_v,
>                                            int width) {
>   __asm {
>     push       edi
>@@ -3709,8 +3719,8 @@ __declspec(naked) void YUY2ToUV422Row_AVX2(const uint8* src_yuy2,
>   }
> }
> 
>-__declspec(naked) void UYVYToYRow_AVX2(const uint8* src_uyvy,
>-                                       uint8* dst_y,
>+__declspec(naked) void UYVYToYRow_AVX2(const uint8_t* src_uyvy,
>+                                       uint8_t* dst_y,
>                                        int width) {
>   __asm {
>     mov        eax, [esp + 4]  // src_uyvy
>@@ -3734,10 +3744,10 @@ __declspec(naked) void UYVYToYRow_AVX2(const uint8* src_uyvy,
>   }
> }
> 
>-__declspec(naked) void UYVYToUVRow_AVX2(const uint8* src_uyvy,
>+__declspec(naked) void UYVYToUVRow_AVX2(const uint8_t* src_uyvy,
>                                         int stride_uyvy,
>-                                        uint8* dst_u,
>-                                        uint8* dst_v,
>+                                        uint8_t* dst_u,
>+                                        uint8_t* dst_v,
>                                         int width) {
>   __asm {
>     push       esi
>@@ -3780,9 +3790,9 @@ __declspec(naked) void UYVYToUVRow_AVX2(const uint8* src_uyvy,
>   }
> }
> 
>-__declspec(naked) void UYVYToUV422Row_AVX2(const uint8* src_uyvy,
>-                                           uint8* dst_u,
>-                                           uint8* dst_v,
>+__declspec(naked) void UYVYToUV422Row_AVX2(const uint8_t* src_uyvy,
>+                                           uint8_t* dst_u,
>+                                           uint8_t* dst_v,
>                                            int width) {
>   __asm {
>     push       edi
>@@ -3822,8 +3832,8 @@ __declspec(naked) void UYVYToUV422Row_AVX2(const uint8* src_uyvy,
> #endif  // HAS_YUY2TOYROW_AVX2
> 
> #ifdef HAS_YUY2TOYROW_SSE2
>-__declspec(naked) void YUY2ToYRow_SSE2(const uint8* src_yuy2,
>-                                       uint8* dst_y,
>+__declspec(naked) void YUY2ToYRow_SSE2(const uint8_t* src_yuy2,
>+                                       uint8_t* dst_y,
>                                        int width) {
>   __asm {
>     mov        eax, [esp + 4]  // src_yuy2
>@@ -3847,10 +3857,10 @@ __declspec(naked) void YUY2ToYRow_SSE2(const uint8* src_yuy2,
>   }
> }
> 
>-__declspec(naked) void YUY2ToUVRow_SSE2(const uint8* src_yuy2,
>+__declspec(naked) void YUY2ToUVRow_SSE2(const uint8_t* src_yuy2,
>                                         int stride_yuy2,
>-                                        uint8* dst_u,
>-                                        uint8* dst_v,
>+                                        uint8_t* dst_u,
>+                                        uint8_t* dst_v,
>                                         int width) {
>   __asm {
>     push       esi
>@@ -3892,9 +3902,9 @@ __declspec(naked) void YUY2ToUVRow_SSE2(const uint8* src_yuy2,
>   }
> }
> 
>-__declspec(naked) void YUY2ToUV422Row_SSE2(const uint8* src_yuy2,
>-                                           uint8* dst_u,
>-                                           uint8* dst_v,
>+__declspec(naked) void YUY2ToUV422Row_SSE2(const uint8_t* src_yuy2,
>+                                           uint8_t* dst_u,
>+                                           uint8_t* dst_v,
>                                            int width) {
>   __asm {
>     push       edi
>@@ -3929,8 +3939,8 @@ __declspec(naked) void YUY2ToUV422Row_SSE2(const uint8* src_yuy2,
>   }
> }
> 
>-__declspec(naked) void UYVYToYRow_SSE2(const uint8* src_uyvy,
>-                                       uint8* dst_y,
>+__declspec(naked) void UYVYToYRow_SSE2(const uint8_t* src_uyvy,
>+                                       uint8_t* dst_y,
>                                        int width) {
>   __asm {
>     mov        eax, [esp + 4]  // src_uyvy
>@@ -3952,10 +3962,10 @@ __declspec(naked) void UYVYToYRow_SSE2(const uint8* src_uyvy,
>   }
> }
> 
>-__declspec(naked) void UYVYToUVRow_SSE2(const uint8* src_uyvy,
>+__declspec(naked) void UYVYToUVRow_SSE2(const uint8_t* src_uyvy,
>                                         int stride_uyvy,
>-                                        uint8* dst_u,
>-                                        uint8* dst_v,
>+                                        uint8_t* dst_u,
>+                                        uint8_t* dst_v,
>                                         int width) {
>   __asm {
>     push       esi
>@@ -3997,9 +4007,9 @@ __declspec(naked) void UYVYToUVRow_SSE2(const uint8* src_uyvy,
>   }
> }
> 
>-__declspec(naked) void UYVYToUV422Row_SSE2(const uint8* src_uyvy,
>-                                           uint8* dst_u,
>-                                           uint8* dst_v,
>+__declspec(naked) void UYVYToUV422Row_SSE2(const uint8_t* src_uyvy,
>+                                           uint8_t* dst_u,
>+                                           uint8_t* dst_v,
>                                            int width) {
>   __asm {
>     push       edi
>@@ -4041,10 +4051,10 @@ __declspec(naked) void UYVYToUV422Row_SSE2(const uint8* src_uyvy,
> // =((A2*C2)+(B2*(255-C2))+255)/256
> // signed version of math
> // =(((A2-128)*C2)+((B2-128)*(255-C2))+32768+127)/256
>-__declspec(naked) void BlendPlaneRow_SSSE3(const uint8* src0,
>-                                           const uint8* src1,
>-                                           const uint8* alpha,
>-                                           uint8* dst,
>+__declspec(naked) void BlendPlaneRow_SSSE3(const uint8_t* src0,
>+                                           const uint8_t* src1,
>+                                           const uint8_t* alpha,
>+                                           uint8_t* dst,
>                                            int width) {
>   __asm {
>     push       esi
>@@ -4098,10 +4108,10 @@ __declspec(naked) void BlendPlaneRow_SSSE3(const uint8* src0,
> // =((A2*C2)+(B2*(255-C2))+255)/256
> // signed version of math
> // =(((A2-128)*C2)+((B2-128)*(255-C2))+32768+127)/256
>-__declspec(naked) void BlendPlaneRow_AVX2(const uint8* src0,
>-                                          const uint8* src1,
>-                                          const uint8* alpha,
>-                                          uint8* dst,
>+__declspec(naked) void BlendPlaneRow_AVX2(const uint8_t* src0,
>+                                          const uint8_t* src1,
>+                                          const uint8_t* alpha,
>+                                          uint8_t* dst,
>                                           int width) {
>   __asm {
>     push        esi
>@@ -4162,9 +4172,9 @@ static const uvec8 kShuffleAlpha = {3u,  0x80, 3u,  0x80, 7u,  0x80, 7u,  0x80,
>                                     11u, 0x80, 11u, 0x80, 15u, 0x80, 15u, 0x80};
> 
> // Blend 8 pixels at a time.
>-__declspec(naked) void ARGBBlendRow_SSSE3(const uint8* src_argb0,
>-                                          const uint8* src_argb1,
>-                                          uint8* dst_argb,
>+__declspec(naked) void ARGBBlendRow_SSSE3(const uint8_t* src_argb0,
>+                                          const uint8_t* src_argb1,
>+                                          uint8_t* dst_argb,
>                                           int width) {
>   __asm {
>     push       esi
>@@ -4253,8 +4263,8 @@ static const uvec8 kShuffleAlpha1 = {
>     11u, 11u, 11u, 11u, 11u, 11u, 128u, 128u,
>     15u, 15u, 15u, 15u, 15u, 15u, 128u, 128u,
> };
>-__declspec(naked) void ARGBAttenuateRow_SSSE3(const uint8* src_argb,
>-                                              uint8* dst_argb,
>+__declspec(naked) void ARGBAttenuateRow_SSSE3(const uint8_t* src_argb,
>+                                              uint8_t* dst_argb,
>                                               int width) {
>   __asm {
>     mov        eax, [esp + 4]  // src_argb0
>@@ -4298,8 +4308,8 @@ __declspec(naked) void ARGBAttenuateRow_SSSE3(const uint8* src_argb,
> static const uvec8 kShuffleAlpha_AVX2 = {6u,   7u,   6u,   7u,  6u,  7u,
>                                          128u, 128u, 14u,  15u, 14u, 15u,
>                                          14u,  15u,  128u, 128u};
>-__declspec(naked) void ARGBAttenuateRow_AVX2(const uint8* src_argb,
>-                                             uint8* dst_argb,
>+__declspec(naked) void ARGBAttenuateRow_AVX2(const uint8_t* src_argb,
>+                                             uint8_t* dst_argb,
>                                              int width) {
>   __asm {
>     mov        eax, [esp + 4]  // src_argb0
>@@ -4336,8 +4346,8 @@ __declspec(naked) void ARGBAttenuateRow_AVX2(const uint8* src_argb,
> 
> #ifdef HAS_ARGBUNATTENUATEROW_SSE2
> // Unattenuate 4 pixels at a time.
>-__declspec(naked) void ARGBUnattenuateRow_SSE2(const uint8* src_argb,
>-                                               uint8* dst_argb,
>+__declspec(naked) void ARGBUnattenuateRow_SSE2(const uint8_t* src_argb,
>+                                               uint8_t* dst_argb,
>                                                int width) {
>   __asm {
>     push       ebx
>@@ -4392,8 +4402,8 @@ static const uvec8 kUnattenShuffleAlpha_AVX2 = {
> // TODO(fbarchard): Enable USE_GATHER for future hardware if faster.
> // USE_GATHER is not on by default, due to being a slow instruction.
> #ifdef USE_GATHER
>-__declspec(naked) void ARGBUnattenuateRow_AVX2(const uint8* src_argb,
>-                                               uint8* dst_argb,
>+__declspec(naked) void ARGBUnattenuateRow_AVX2(const uint8_t* src_argb,
>+                                               uint8_t* dst_argb,
>                                                int width) {
>   __asm {
>     mov        eax, [esp + 4]  // src_argb0
>@@ -4426,8 +4436,8 @@ __declspec(naked) void ARGBUnattenuateRow_AVX2(const uint8* src_argb,
>   }
> }
> #else   // USE_GATHER
>-__declspec(naked) void ARGBUnattenuateRow_AVX2(const uint8* src_argb,
>-                                               uint8* dst_argb,
>+__declspec(naked) void ARGBUnattenuateRow_AVX2(const uint8_t* src_argb,
>+                                               uint8_t* dst_argb,
>                                                int width) {
>   __asm {
> 
>@@ -4495,8 +4505,8 @@ __declspec(naked) void ARGBUnattenuateRow_AVX2(const uint8* src_argb,
> 
> #ifdef HAS_ARGBGRAYROW_SSSE3
> // Convert 8 ARGB pixels (64 bytes) to 8 Gray ARGB pixels.
>-__declspec(naked) void ARGBGrayRow_SSSE3(const uint8* src_argb,
>-                                         uint8* dst_argb,
>+__declspec(naked) void ARGBGrayRow_SSSE3(const uint8_t* src_argb,
>+                                         uint8_t* dst_argb,
>                                          int width) {
>   __asm {
>     mov        eax, [esp + 4] /* src_argb */
>@@ -4552,7 +4562,7 @@ static const vec8 kARGBToSepiaR = {24, 98, 50, 0, 24, 98, 50, 0,
>                                    24, 98, 50, 0, 24, 98, 50, 0};
> 
> // Convert 8 ARGB pixels (32 bytes) to 8 Sepia ARGB pixels.
>-__declspec(naked) void ARGBSepiaRow_SSSE3(uint8* dst_argb, int width) {
>+__declspec(naked) void ARGBSepiaRow_SSSE3(uint8_t* dst_argb, int width) {
>   __asm {
>     mov        eax, [esp + 4] /* dst_argb */
>     mov        ecx, [esp + 8] /* width */
>@@ -4608,9 +4618,9 @@ __declspec(naked) void ARGBSepiaRow_SSSE3(uint8* dst_argb, int width) {
> // Same as Sepia except matrix is provided.
> // TODO(fbarchard): packuswbs only use half of the reg. To make RGBA, combine R
> // and B into a high and low, then G/A, unpackl/hbw and then unpckl/hwd.
>-__declspec(naked) void ARGBColorMatrixRow_SSSE3(const uint8* src_argb,
>-                                                uint8* dst_argb,
>-                                                const int8* matrix_argb,
>+__declspec(naked) void ARGBColorMatrixRow_SSSE3(const uint8_t* src_argb,
>+                                                uint8_t* dst_argb,
>+                                                const int8_t* matrix_argb,
>                                                 int width) {
>   __asm {
>     mov        eax, [esp + 4] /* src_argb */
>@@ -4670,7 +4680,7 @@ __declspec(naked) void ARGBColorMatrixRow_SSSE3(const uint8* src_argb,
> 
> #ifdef HAS_ARGBQUANTIZEROW_SSE2
> // Quantize 4 ARGB pixels (16 bytes).
>-__declspec(naked) void ARGBQuantizeRow_SSE2(uint8* dst_argb,
>+__declspec(naked) void ARGBQuantizeRow_SSE2(uint8_t* dst_argb,
>                                             int scale,
>                                             int interval_size,
>                                             int interval_offset,
>@@ -4717,10 +4727,10 @@ __declspec(naked) void ARGBQuantizeRow_SSE2(uint8* dst_argb,
> 
> #ifdef HAS_ARGBSHADEROW_SSE2
> // Shade 4 pixels at a time by specified value.
>-__declspec(naked) void ARGBShadeRow_SSE2(const uint8* src_argb,
>-                                         uint8* dst_argb,
>+__declspec(naked) void ARGBShadeRow_SSE2(const uint8_t* src_argb,
>+                                         uint8_t* dst_argb,
>                                          int width,
>-                                         uint32 value) {
>+                                         uint32_t value) {
>   __asm {
>     mov        eax, [esp + 4]  // src_argb
>     mov        edx, [esp + 8]  // dst_argb
>@@ -4752,9 +4762,9 @@ __declspec(naked) void ARGBShadeRow_SSE2(const uint8* src_argb,
> 
> #ifdef HAS_ARGBMULTIPLYROW_SSE2
> // Multiply 2 rows of ARGB pixels together, 4 pixels at a time.
>-__declspec(naked) void ARGBMultiplyRow_SSE2(const uint8* src_argb0,
>-                                            const uint8* src_argb1,
>-                                            uint8* dst_argb,
>+__declspec(naked) void ARGBMultiplyRow_SSE2(const uint8_t* src_argb0,
>+                                            const uint8_t* src_argb1,
>+                                            uint8_t* dst_argb,
>                                             int width) {
>   __asm {
>     push       esi
>@@ -4792,9 +4802,9 @@ __declspec(naked) void ARGBMultiplyRow_SSE2(const uint8* src_argb0,
> #ifdef HAS_ARGBADDROW_SSE2
> // Add 2 rows of ARGB pixels together, 4 pixels at a time.
> // TODO(fbarchard): Port this to posix, neon and other math functions.
>-__declspec(naked) void ARGBAddRow_SSE2(const uint8* src_argb0,
>-                                       const uint8* src_argb1,
>-                                       uint8* dst_argb,
>+__declspec(naked) void ARGBAddRow_SSE2(const uint8_t* src_argb0,
>+                                       const uint8_t* src_argb1,
>+                                       uint8_t* dst_argb,
>                                        int width) {
>   __asm {
>     push       esi
>@@ -4841,9 +4851,9 @@ __declspec(naked) void ARGBAddRow_SSE2(const uint8* src_argb0,
> 
> #ifdef HAS_ARGBSUBTRACTROW_SSE2
> // Subtract 2 rows of ARGB pixels together, 4 pixels at a time.
>-__declspec(naked) void ARGBSubtractRow_SSE2(const uint8* src_argb0,
>-                                            const uint8* src_argb1,
>-                                            uint8* dst_argb,
>+__declspec(naked) void ARGBSubtractRow_SSE2(const uint8_t* src_argb0,
>+                                            const uint8_t* src_argb1,
>+                                            uint8_t* dst_argb,
>                                             int width) {
>   __asm {
>     push       esi
>@@ -4871,9 +4881,9 @@ __declspec(naked) void ARGBSubtractRow_SSE2(const uint8* src_argb0,
> 
> #ifdef HAS_ARGBMULTIPLYROW_AVX2
> // Multiply 2 rows of ARGB pixels together, 8 pixels at a time.
>-__declspec(naked) void ARGBMultiplyRow_AVX2(const uint8* src_argb0,
>-                                            const uint8* src_argb1,
>-                                            uint8* dst_argb,
>+__declspec(naked) void ARGBMultiplyRow_AVX2(const uint8_t* src_argb0,
>+                                            const uint8_t* src_argb1,
>+                                            uint8_t* dst_argb,
>                                             int width) {
>   __asm {
>     push       esi
>@@ -4909,9 +4919,9 @@ __declspec(naked) void ARGBMultiplyRow_AVX2(const uint8* src_argb0,
> 
> #ifdef HAS_ARGBADDROW_AVX2
> // Add 2 rows of ARGB pixels together, 8 pixels at a time.
>-__declspec(naked) void ARGBAddRow_AVX2(const uint8* src_argb0,
>-                                       const uint8* src_argb1,
>-                                       uint8* dst_argb,
>+__declspec(naked) void ARGBAddRow_AVX2(const uint8_t* src_argb0,
>+                                       const uint8_t* src_argb1,
>+                                       uint8_t* dst_argb,
>                                        int width) {
>   __asm {
>     push       esi
>@@ -4939,9 +4949,9 @@ __declspec(naked) void ARGBAddRow_AVX2(const uint8* src_argb0,
> 
> #ifdef HAS_ARGBSUBTRACTROW_AVX2
> // Subtract 2 rows of ARGB pixels together, 8 pixels at a time.
>-__declspec(naked) void ARGBSubtractRow_AVX2(const uint8* src_argb0,
>-                                            const uint8* src_argb1,
>-                                            uint8* dst_argb,
>+__declspec(naked) void ARGBSubtractRow_AVX2(const uint8_t* src_argb0,
>+                                            const uint8_t* src_argb1,
>+                                            uint8_t* dst_argb,
>                                             int width) {
>   __asm {
>     push       esi
>@@ -4972,10 +4982,10 @@ __declspec(naked) void ARGBSubtractRow_AVX2(const uint8* src_argb0,
> // -1  0  1
> // -2  0  2
> // -1  0  1
>-__declspec(naked) void SobelXRow_SSE2(const uint8* src_y0,
>-                                      const uint8* src_y1,
>-                                      const uint8* src_y2,
>-                                      uint8* dst_sobelx,
>+__declspec(naked) void SobelXRow_SSE2(const uint8_t* src_y0,
>+                                      const uint8_t* src_y1,
>+                                      const uint8_t* src_y2,
>+                                      uint8_t* dst_sobelx,
>                                       int width) {
>   __asm {
>     push       esi
>@@ -5030,9 +5040,9 @@ __declspec(naked) void SobelXRow_SSE2(const uint8* src_y0,
> // -1 -2 -1
> //  0  0  0
> //  1  2  1
>-__declspec(naked) void SobelYRow_SSE2(const uint8* src_y0,
>-                                      const uint8* src_y1,
>-                                      uint8* dst_sobely,
>+__declspec(naked) void SobelYRow_SSE2(const uint8_t* src_y0,
>+                                      const uint8_t* src_y1,
>+                                      uint8_t* dst_sobely,
>                                       int width) {
>   __asm {
>     push       esi
>@@ -5084,9 +5094,9 @@ __declspec(naked) void SobelYRow_SSE2(const uint8* src_y0,
> // R = Sobel
> // G = Sobel
> // B = Sobel
>-__declspec(naked) void SobelRow_SSE2(const uint8* src_sobelx,
>-                                     const uint8* src_sobely,
>-                                     uint8* dst_argb,
>+__declspec(naked) void SobelRow_SSE2(const uint8_t* src_sobelx,
>+                                     const uint8_t* src_sobely,
>+                                     uint8_t* dst_argb,
>                                      int width) {
>   __asm {
>     push       esi
>@@ -5132,9 +5142,9 @@ __declspec(naked) void SobelRow_SSE2(const uint8* src_sobelx,
> 
> #ifdef HAS_SOBELTOPLANEROW_SSE2
> // Adds Sobel X and Sobel Y and stores Sobel into a plane.
>-__declspec(naked) void SobelToPlaneRow_SSE2(const uint8* src_sobelx,
>-                                            const uint8* src_sobely,
>-                                            uint8* dst_y,
>+__declspec(naked) void SobelToPlaneRow_SSE2(const uint8_t* src_sobelx,
>+                                            const uint8_t* src_sobely,
>+                                            uint8_t* dst_y,
>                                             int width) {
>   __asm {
>     push       esi
>@@ -5166,9 +5176,9 @@ __declspec(naked) void SobelToPlaneRow_SSE2(const uint8* src_sobelx,
> // R = Sobel X
> // G = Sobel
> // B = Sobel Y
>-__declspec(naked) void SobelXYRow_SSE2(const uint8* src_sobelx,
>-                                       const uint8* src_sobely,
>-                                       uint8* dst_argb,
>+__declspec(naked) void SobelXYRow_SSE2(const uint8_t* src_sobelx,
>+                                       const uint8_t* src_sobely,
>+                                       uint8_t* dst_argb,
>                                        int width) {
>   __asm {
>     push       esi
>@@ -5225,11 +5235,11 @@ __declspec(naked) void SobelXYRow_SSE2(const uint8* src_sobelx,
> // count is number of averaged pixels to produce.
> // Does 4 pixels at a time.
> // This function requires alignment on accumulation buffer pointers.
>-void CumulativeSumToAverageRow_SSE2(const int32* topleft,
>-                                    const int32* botleft,
>+void CumulativeSumToAverageRow_SSE2(const int32_t* topleft,
>+                                    const int32_t* botleft,
>                                     int width,
>                                     int area,
>-                                    uint8* dst,
>+                                    uint8_t* dst,
>                                     int count) {
>   __asm {
>     mov        eax, topleft  // eax topleft
>@@ -5375,9 +5385,9 @@ void CumulativeSumToAverageRow_SSE2(const int32* topleft,
> #ifdef HAS_COMPUTECUMULATIVESUMROW_SSE2
> // Creates a table of cumulative sums where each value is a sum of all values
> // above and to the left of the value.
>-void ComputeCumulativeSumRow_SSE2(const uint8* row,
>-                                  int32* cumsum,
>-                                  const int32* previous_cumsum,
>+void ComputeCumulativeSumRow_SSE2(const uint8_t* row,
>+                                  int32_t* cumsum,
>+                                  const int32_t* previous_cumsum,
>                                   int width) {
>   __asm {
>     mov        eax, row
>@@ -5460,9 +5470,9 @@ void ComputeCumulativeSumRow_SSE2(const uint8* row,
> 
> #ifdef HAS_ARGBAFFINEROW_SSE2
> // Copy ARGB pixels from source image with slope to a row of destination.
>-__declspec(naked) LIBYUV_API void ARGBAffineRow_SSE2(const uint8* src_argb,
>+__declspec(naked) LIBYUV_API void ARGBAffineRow_SSE2(const uint8_t* src_argb,
>                                                      int src_argb_stride,
>-                                                     uint8* dst_argb,
>+                                                     uint8_t* dst_argb,
>                                                      const float* uv_dudv,
>                                                      int width) {
>   __asm {
>@@ -5546,8 +5556,8 @@ __declspec(naked) LIBYUV_API void ARGBAffineRow_SSE2(const uint8* src_argb,
> 
> #ifdef HAS_INTERPOLATEROW_AVX2
> // Bilinear filter 32x2 -> 32x1
>-__declspec(naked) void InterpolateRow_AVX2(uint8* dst_ptr,
>-                                           const uint8* src_ptr,
>+__declspec(naked) void InterpolateRow_AVX2(uint8_t* dst_ptr,
>+                                           const uint8_t* src_ptr,
>                                            ptrdiff_t src_stride,
>                                            int dst_width,
>                                            int source_y_fraction) {
>@@ -5623,8 +5633,8 @@ __declspec(naked) void InterpolateRow_AVX2(uint8* dst_ptr,
> 
> // Bilinear filter 16x2 -> 16x1
> // TODO(fbarchard): Consider allowing 256 using memcpy.
>-__declspec(naked) void InterpolateRow_SSSE3(uint8* dst_ptr,
>-                                            const uint8* src_ptr,
>+__declspec(naked) void InterpolateRow_SSSE3(uint8_t* dst_ptr,
>+                                            const uint8_t* src_ptr,
>                                             ptrdiff_t src_stride,
>                                             int dst_width,
>                                             int source_y_fraction) {
>@@ -5705,9 +5715,9 @@ __declspec(naked) void InterpolateRow_SSSE3(uint8* dst_ptr,
> }
> 
> // For BGRAToARGB, ABGRToARGB, RGBAToARGB, and ARGBToRGBA.
>-__declspec(naked) void ARGBShuffleRow_SSSE3(const uint8* src_argb,
>-                                            uint8* dst_argb,
>-                                            const uint8* shuffler,
>+__declspec(naked) void ARGBShuffleRow_SSSE3(const uint8_t* src_argb,
>+                                            uint8_t* dst_argb,
>+                                            const uint8_t* shuffler,
>                                             int width) {
>   __asm {
>     mov        eax, [esp + 4]  // src_argb
>@@ -5732,9 +5742,9 @@ __declspec(naked) void ARGBShuffleRow_SSSE3(const uint8* src_argb,
> }
> 
> #ifdef HAS_ARGBSHUFFLEROW_AVX2
>-__declspec(naked) void ARGBShuffleRow_AVX2(const uint8* src_argb,
>-                                           uint8* dst_argb,
>-                                           const uint8* shuffler,
>+__declspec(naked) void ARGBShuffleRow_AVX2(const uint8_t* src_argb,
>+                                           uint8_t* dst_argb,
>+                                           const uint8_t* shuffler,
>                                            int width) {
>   __asm {
>     mov        eax, [esp + 4]  // src_argb
>@@ -5761,133 +5771,16 @@ __declspec(naked) void ARGBShuffleRow_AVX2(const uint8* src_argb,
> }
> #endif  // HAS_ARGBSHUFFLEROW_AVX2
> 
>-__declspec(naked) void ARGBShuffleRow_SSE2(const uint8* src_argb,
>-                                           uint8* dst_argb,
>-                                           const uint8* shuffler,
>-                                           int width) {
>-  __asm {
>-    push       ebx
>-    push       esi
>-    mov        eax, [esp + 8 + 4]  // src_argb
>-    mov        edx, [esp + 8 + 8]  // dst_argb
>-    mov        esi, [esp + 8 + 12]  // shuffler
>-    mov        ecx, [esp + 8 + 16]  // width
>-    pxor       xmm5, xmm5
>-
>-    mov        ebx, [esi]  // shuffler
>-    cmp        ebx, 0x03000102
>-    je         shuf_3012
>-    cmp        ebx, 0x00010203
>-    je         shuf_0123
>-    cmp        ebx, 0x00030201
>-    je         shuf_0321
>-    cmp        ebx, 0x02010003
>-    je         shuf_2103
>-
>-        // TODO(fbarchard): Use one source pointer and 3 offsets.
>-  shuf_any1:
>-    movzx      ebx, byte ptr [esi]
>-    movzx      ebx, byte ptr [eax + ebx]
>-    mov        [edx], bl
>-    movzx      ebx, byte ptr [esi + 1]
>-    movzx      ebx, byte ptr [eax + ebx]
>-    mov        [edx + 1], bl
>-    movzx      ebx, byte ptr [esi + 2]
>-    movzx      ebx, byte ptr [eax + ebx]
>-    mov        [edx + 2], bl
>-    movzx      ebx, byte ptr [esi + 3]
>-    movzx      ebx, byte ptr [eax + ebx]
>-    mov        [edx + 3], bl
>-    lea        eax, [eax + 4]
>-    lea        edx, [edx + 4]
>-    sub        ecx, 1
>-    jg         shuf_any1
>-    jmp        shuf99
>-
>-  shuf_0123:
>-    movdqu     xmm0, [eax]
>-    lea        eax, [eax + 16]
>-    movdqa     xmm1, xmm0
>-    punpcklbw  xmm0, xmm5
>-    punpckhbw  xmm1, xmm5
>-    pshufhw    xmm0, xmm0, 01Bh  // 1B = 00011011 = 0x0123 = BGRAToARGB
>-    pshuflw    xmm0, xmm0, 01Bh
>-    pshufhw    xmm1, xmm1, 01Bh
>-    pshuflw    xmm1, xmm1, 01Bh
>-    packuswb   xmm0, xmm1
>-    movdqu     [edx], xmm0
>-    lea        edx, [edx + 16]
>-    sub        ecx, 4
>-    jg         shuf_0123
>-    jmp        shuf99
>-
>-  shuf_0321:
>-    movdqu     xmm0, [eax]
>-    lea        eax, [eax + 16]
>-    movdqa     xmm1, xmm0
>-    punpcklbw  xmm0, xmm5
>-    punpckhbw  xmm1, xmm5
>-    pshufhw    xmm0, xmm0, 039h  // 39 = 00111001 = 0x0321 = RGBAToARGB
>-    pshuflw    xmm0, xmm0, 039h
>-    pshufhw    xmm1, xmm1, 039h
>-    pshuflw    xmm1, xmm1, 039h
>-    packuswb   xmm0, xmm1
>-    movdqu     [edx], xmm0
>-    lea        edx, [edx + 16]
>-    sub        ecx, 4
>-    jg         shuf_0321
>-    jmp        shuf99
>-
>-  shuf_2103:
>-    movdqu     xmm0, [eax]
>-    lea        eax, [eax + 16]
>-    movdqa     xmm1, xmm0
>-    punpcklbw  xmm0, xmm5
>-    punpckhbw  xmm1, xmm5
>-    pshufhw    xmm0, xmm0, 093h  // 93 = 10010011 = 0x2103 = ARGBToRGBA
>-    pshuflw    xmm0, xmm0, 093h
>-    pshufhw    xmm1, xmm1, 093h
>-    pshuflw    xmm1, xmm1, 093h
>-    packuswb   xmm0, xmm1
>-    movdqu     [edx], xmm0
>-    lea        edx, [edx + 16]
>-    sub        ecx, 4
>-    jg         shuf_2103
>-    jmp        shuf99
>-
>-  shuf_3012:
>-    movdqu     xmm0, [eax]
>-    lea        eax, [eax + 16]
>-    movdqa     xmm1, xmm0
>-    punpcklbw  xmm0, xmm5
>-    punpckhbw  xmm1, xmm5
>-    pshufhw    xmm0, xmm0, 0C6h  // C6 = 11000110 = 0x3012 = ABGRToARGB
>-    pshuflw    xmm0, xmm0, 0C6h
>-    pshufhw    xmm1, xmm1, 0C6h
>-    pshuflw    xmm1, xmm1, 0C6h
>-    packuswb   xmm0, xmm1
>-    movdqu     [edx], xmm0
>-    lea        edx, [edx + 16]
>-    sub        ecx, 4
>-    jg         shuf_3012
>-
>-  shuf99:
>-    pop        esi
>-    pop        ebx
>-    ret
>-  }
>-}
>-
> // YUY2 - Macro-pixel = 2 image pixels
> // Y0U0Y1V0....Y2U2Y3V2...Y4U4Y5V4....
> 
> // UYVY - Macro-pixel = 2 image pixels
> // U0Y0V0Y1
> 
>-__declspec(naked) void I422ToYUY2Row_SSE2(const uint8* src_y,
>-                                          const uint8* src_u,
>-                                          const uint8* src_v,
>-                                          uint8* dst_frame,
>+__declspec(naked) void I422ToYUY2Row_SSE2(const uint8_t* src_y,
>+                                          const uint8_t* src_u,
>+                                          const uint8_t* src_v,
>+                                          uint8_t* dst_frame,
>                                           int width) {
>   __asm {
>     push       esi
>@@ -5921,10 +5814,10 @@ __declspec(naked) void I422ToYUY2Row_SSE2(const uint8* src_y,
>   }
> }
> 
>-__declspec(naked) void I422ToUYVYRow_SSE2(const uint8* src_y,
>-                                          const uint8* src_u,
>-                                          const uint8* src_v,
>-                                          uint8* dst_frame,
>+__declspec(naked) void I422ToUYVYRow_SSE2(const uint8_t* src_y,
>+                                          const uint8_t* src_u,
>+                                          const uint8_t* src_v,
>+                                          uint8_t* dst_frame,
>                                           int width) {
>   __asm {
>     push       esi
>@@ -5959,8 +5852,8 @@ __declspec(naked) void I422ToUYVYRow_SSE2(const uint8* src_y,
> }
> 
> #ifdef HAS_ARGBPOLYNOMIALROW_SSE2
>-__declspec(naked) void ARGBPolynomialRow_SSE2(const uint8* src_argb,
>-                                              uint8* dst_argb,
>+__declspec(naked) void ARGBPolynomialRow_SSE2(const uint8_t* src_argb,
>+                                              uint8_t* dst_argb,
>                                               const float* poly,
>                                               int width) {
>   __asm {
>@@ -6018,8 +5911,8 @@ __declspec(naked) void ARGBPolynomialRow_SSE2(const uint8* src_argb,
> #endif  // HAS_ARGBPOLYNOMIALROW_SSE2
> 
> #ifdef HAS_ARGBPOLYNOMIALROW_AVX2
>-__declspec(naked) void ARGBPolynomialRow_AVX2(const uint8* src_argb,
>-                                              uint8* dst_argb,
>+__declspec(naked) void ARGBPolynomialRow_AVX2(const uint8_t* src_argb,
>+                                              uint8_t* dst_argb,
>                                               const float* poly,
>                                               int width) {
>   __asm {
>@@ -6058,8 +5951,8 @@ __declspec(naked) void ARGBPolynomialRow_AVX2(const uint8* src_argb,
> 
> #ifdef HAS_HALFFLOATROW_SSE2
> static float kExpBias = 1.9259299444e-34f;
>-__declspec(naked) void HalfFloatRow_SSE2(const uint16* src,
>-                                         uint16* dst,
>+__declspec(naked) void HalfFloatRow_SSE2(const uint16_t* src,
>+                                         uint16_t* dst,
>                                          float scale,
>                                          int width) {
>   __asm {
>@@ -6095,8 +5988,8 @@ __declspec(naked) void HalfFloatRow_SSE2(const uint16* src,
> #endif  // HAS_HALFFLOATROW_SSE2
> 
> #ifdef HAS_HALFFLOATROW_AVX2
>-__declspec(naked) void HalfFloatRow_AVX2(const uint16* src,
>-                                         uint16* dst,
>+__declspec(naked) void HalfFloatRow_AVX2(const uint16_t* src,
>+                                         uint16_t* dst,
>                                          float scale,
>                                          int width) {
>   __asm {
>@@ -6133,8 +6026,8 @@ __declspec(naked) void HalfFloatRow_AVX2(const uint16* src,
> #endif  // HAS_HALFFLOATROW_AVX2
> 
> #ifdef HAS_HALFFLOATROW_F16C
>-__declspec(naked) void HalfFloatRow_F16C(const uint16* src,
>-                                         uint16* dst,
>+__declspec(naked) void HalfFloatRow_F16C(const uint16_t* src,
>+                                         uint16_t* dst,
>                                          float scale,
>                                          int width) {
>   __asm {
>@@ -6167,8 +6060,8 @@ __declspec(naked) void HalfFloatRow_F16C(const uint16* src,
> 
> #ifdef HAS_ARGBCOLORTABLEROW_X86
> // Tranform ARGB pixels with color table.
>-__declspec(naked) void ARGBColorTableRow_X86(uint8* dst_argb,
>-                                             const uint8* table_argb,
>+__declspec(naked) void ARGBColorTableRow_X86(uint8_t* dst_argb,
>+                                             const uint8_t* table_argb,
>                                              int width) {
>   __asm {
>     push       esi
>@@ -6201,8 +6094,8 @@ __declspec(naked) void ARGBColorTableRow_X86(uint8* dst_argb,
> 
> #ifdef HAS_RGBCOLORTABLEROW_X86
> // Tranform RGB pixels with color table.
>-__declspec(naked) void RGBColorTableRow_X86(uint8* dst_argb,
>-                                            const uint8* table_argb,
>+__declspec(naked) void RGBColorTableRow_X86(uint8_t* dst_argb,
>+                                            const uint8_t* table_argb,
>                                             int width) {
>   __asm {
>     push       esi
>@@ -6233,11 +6126,11 @@ __declspec(naked) void RGBColorTableRow_X86(uint8* dst_argb,
> 
> #ifdef HAS_ARGBLUMACOLORTABLEROW_SSSE3
> // Tranform RGB pixels with luma table.
>-__declspec(naked) void ARGBLumaColorTableRow_SSSE3(const uint8* src_argb,
>-                                                   uint8* dst_argb,
>+__declspec(naked) void ARGBLumaColorTableRow_SSSE3(const uint8_t* src_argb,
>+                                                   uint8_t* dst_argb,
>                                                    int width,
>-                                                   const uint8* luma,
>-                                                   uint32 lumacoeff) {
>+                                                   const uint8_t* luma,
>+                                                   uint32_t lumacoeff) {
>   __asm {
>     push       esi
>     push       edi
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/scale.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/scale.cc
>index 9104acb95fc..2cfa1c6cb1c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/scale.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/scale.cc
>@@ -39,12 +39,12 @@ static void ScalePlaneDown2(int src_width,
>                             int dst_height,
>                             int src_stride,
>                             int dst_stride,
>-                            const uint8* src_ptr,
>-                            uint8* dst_ptr,
>+                            const uint8_t* src_ptr,
>+                            uint8_t* dst_ptr,
>                             enum FilterMode filtering) {
>   int y;
>-  void (*ScaleRowDown2)(const uint8* src_ptr, ptrdiff_t src_stride,
>-                        uint8* dst_ptr, int dst_width) =
>+  void (*ScaleRowDown2)(const uint8_t* src_ptr, ptrdiff_t src_stride,
>+                        uint8_t* dst_ptr, int dst_width) =
>       filtering == kFilterNone
>           ? ScaleRowDown2_C
>           : (filtering == kFilterLinear ? ScaleRowDown2Linear_C
>@@ -103,13 +103,6 @@ static void ScalePlaneDown2(int src_width,
>     }
>   }
> #endif
>-#if defined(HAS_SCALEROWDOWN2_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2) && IS_ALIGNED(src_ptr, 4) &&
>-      IS_ALIGNED(src_stride, 4) && IS_ALIGNED(row_stride, 4) &&
>-      IS_ALIGNED(dst_ptr, 4) && IS_ALIGNED(dst_stride, 4)) {
>-    ScaleRowDown2 = filtering ? ScaleRowDown2Box_DSPR2 : ScaleRowDown2_DSPR2;
>-  }
>-#endif
> #if defined(HAS_SCALEROWDOWN2_MSA)
>   if (TestCpuFlag(kCpuHasMSA)) {
>     ScaleRowDown2 =
>@@ -143,12 +136,12 @@ static void ScalePlaneDown2_16(int src_width,
>                                int dst_height,
>                                int src_stride,
>                                int dst_stride,
>-                               const uint16* src_ptr,
>-                               uint16* dst_ptr,
>+                               const uint16_t* src_ptr,
>+                               uint16_t* dst_ptr,
>                                enum FilterMode filtering) {
>   int y;
>-  void (*ScaleRowDown2)(const uint16* src_ptr, ptrdiff_t src_stride,
>-                        uint16* dst_ptr, int dst_width) =
>+  void (*ScaleRowDown2)(const uint16_t* src_ptr, ptrdiff_t src_stride,
>+                        uint16_t* dst_ptr, int dst_width) =
>       filtering == kFilterNone
>           ? ScaleRowDown2_16_C
>           : (filtering == kFilterLinear ? ScaleRowDown2Linear_16_C
>@@ -176,14 +169,6 @@ static void ScalePlaneDown2_16(int src_width,
>                                           : ScaleRowDown2Box_16_SSE2);
>   }
> #endif
>-#if defined(HAS_SCALEROWDOWN2_16_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2) && IS_ALIGNED(src_ptr, 4) &&
>-      IS_ALIGNED(src_stride, 4) && IS_ALIGNED(row_stride, 4) &&
>-      IS_ALIGNED(dst_ptr, 4) && IS_ALIGNED(dst_stride, 4)) {
>-    ScaleRowDown2 =
>-        filtering ? ScaleRowDown2Box_16_DSPR2 : ScaleRowDown2_16_DSPR2;
>-  }
>-#endif
> 
>   if (filtering == kFilterLinear) {
>     src_stride = 0;
>@@ -206,12 +191,12 @@ static void ScalePlaneDown4(int src_width,
>                             int dst_height,
>                             int src_stride,
>                             int dst_stride,
>-                            const uint8* src_ptr,
>-                            uint8* dst_ptr,
>+                            const uint8_t* src_ptr,
>+                            uint8_t* dst_ptr,
>                             enum FilterMode filtering) {
>   int y;
>-  void (*ScaleRowDown4)(const uint8* src_ptr, ptrdiff_t src_stride,
>-                        uint8* dst_ptr, int dst_width) =
>+  void (*ScaleRowDown4)(const uint8_t* src_ptr, ptrdiff_t src_stride,
>+                        uint8_t* dst_ptr, int dst_width) =
>       filtering ? ScaleRowDown4Box_C : ScaleRowDown4_C;
>   int row_stride = src_stride << 2;
>   (void)src_width;
>@@ -247,13 +232,6 @@ static void ScalePlaneDown4(int src_width,
>     }
>   }
> #endif
>-#if defined(HAS_SCALEROWDOWN4_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2) && IS_ALIGNED(row_stride, 4) &&
>-      IS_ALIGNED(src_ptr, 4) && IS_ALIGNED(src_stride, 4) &&
>-      IS_ALIGNED(dst_ptr, 4) && IS_ALIGNED(dst_stride, 4)) {
>-    ScaleRowDown4 = filtering ? ScaleRowDown4Box_DSPR2 : ScaleRowDown4_DSPR2;
>-  }
>-#endif
> #if defined(HAS_SCALEROWDOWN4_MSA)
>   if (TestCpuFlag(kCpuHasMSA)) {
>     ScaleRowDown4 =
>@@ -280,12 +258,12 @@ static void ScalePlaneDown4_16(int src_width,
>                                int dst_height,
>                                int src_stride,
>                                int dst_stride,
>-                               const uint16* src_ptr,
>-                               uint16* dst_ptr,
>+                               const uint16_t* src_ptr,
>+                               uint16_t* dst_ptr,
>                                enum FilterMode filtering) {
>   int y;
>-  void (*ScaleRowDown4)(const uint16* src_ptr, ptrdiff_t src_stride,
>-                        uint16* dst_ptr, int dst_width) =
>+  void (*ScaleRowDown4)(const uint16_t* src_ptr, ptrdiff_t src_stride,
>+                        uint16_t* dst_ptr, int dst_width) =
>       filtering ? ScaleRowDown4Box_16_C : ScaleRowDown4_16_C;
>   int row_stride = src_stride << 2;
>   (void)src_width;
>@@ -306,14 +284,6 @@ static void ScalePlaneDown4_16(int src_width,
>         filtering ? ScaleRowDown4Box_16_SSE2 : ScaleRowDown4_16_SSE2;
>   }
> #endif
>-#if defined(HAS_SCALEROWDOWN4_16_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2) && IS_ALIGNED(row_stride, 4) &&
>-      IS_ALIGNED(src_ptr, 4) && IS_ALIGNED(src_stride, 4) &&
>-      IS_ALIGNED(dst_ptr, 4) && IS_ALIGNED(dst_stride, 4)) {
>-    ScaleRowDown4 =
>-        filtering ? ScaleRowDown4Box_16_DSPR2 : ScaleRowDown4_16_DSPR2;
>-  }
>-#endif
> 
>   if (filtering == kFilterLinear) {
>     src_stride = 0;
>@@ -332,14 +302,14 @@ static void ScalePlaneDown34(int src_width,
>                              int dst_height,
>                              int src_stride,
>                              int dst_stride,
>-                             const uint8* src_ptr,
>-                             uint8* dst_ptr,
>+                             const uint8_t* src_ptr,
>+                             uint8_t* dst_ptr,
>                              enum FilterMode filtering) {
>   int y;
>-  void (*ScaleRowDown34_0)(const uint8* src_ptr, ptrdiff_t src_stride,
>-                           uint8* dst_ptr, int dst_width);
>-  void (*ScaleRowDown34_1)(const uint8* src_ptr, ptrdiff_t src_stride,
>-                           uint8* dst_ptr, int dst_width);
>+  void (*ScaleRowDown34_0)(const uint8_t* src_ptr, ptrdiff_t src_stride,
>+                           uint8_t* dst_ptr, int dst_width);
>+  void (*ScaleRowDown34_1)(const uint8_t* src_ptr, ptrdiff_t src_stride,
>+                           uint8_t* dst_ptr, int dst_width);
>   const int filter_stride = (filtering == kFilterLinear) ? 0 : src_stride;
>   (void)src_width;
>   (void)src_height;
>@@ -411,19 +381,6 @@ static void ScalePlaneDown34(int src_width,
>     }
>   }
> #endif
>-#if defined(HAS_SCALEROWDOWN34_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2) && (dst_width % 24 == 0) &&
>-      IS_ALIGNED(src_ptr, 4) && IS_ALIGNED(src_stride, 4) &&
>-      IS_ALIGNED(dst_ptr, 4) && IS_ALIGNED(dst_stride, 4)) {
>-    if (!filtering) {
>-      ScaleRowDown34_0 = ScaleRowDown34_DSPR2;
>-      ScaleRowDown34_1 = ScaleRowDown34_DSPR2;
>-    } else {
>-      ScaleRowDown34_0 = ScaleRowDown34_0_Box_DSPR2;
>-      ScaleRowDown34_1 = ScaleRowDown34_1_Box_DSPR2;
>-    }
>-  }
>-#endif
> 
>   for (y = 0; y < dst_height - 2; y += 3) {
>     ScaleRowDown34_0(src_ptr, filter_stride, dst_ptr, dst_width);
>@@ -454,14 +411,14 @@ static void ScalePlaneDown34_16(int src_width,
>                                 int dst_height,
>                                 int src_stride,
>                                 int dst_stride,
>-                                const uint16* src_ptr,
>-                                uint16* dst_ptr,
>+                                const uint16_t* src_ptr,
>+                                uint16_t* dst_ptr,
>                                 enum FilterMode filtering) {
>   int y;
>-  void (*ScaleRowDown34_0)(const uint16* src_ptr, ptrdiff_t src_stride,
>-                           uint16* dst_ptr, int dst_width);
>-  void (*ScaleRowDown34_1)(const uint16* src_ptr, ptrdiff_t src_stride,
>-                           uint16* dst_ptr, int dst_width);
>+  void (*ScaleRowDown34_0)(const uint16_t* src_ptr, ptrdiff_t src_stride,
>+                           uint16_t* dst_ptr, int dst_width);
>+  void (*ScaleRowDown34_1)(const uint16_t* src_ptr, ptrdiff_t src_stride,
>+                           uint16_t* dst_ptr, int dst_width);
>   const int filter_stride = (filtering == kFilterLinear) ? 0 : src_stride;
>   (void)src_width;
>   (void)src_height;
>@@ -495,19 +452,6 @@ static void ScalePlaneDown34_16(int src_width,
>     }
>   }
> #endif
>-#if defined(HAS_SCALEROWDOWN34_16_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2) && (dst_width % 24 == 0) &&
>-      IS_ALIGNED(src_ptr, 4) && IS_ALIGNED(src_stride, 4) &&
>-      IS_ALIGNED(dst_ptr, 4) && IS_ALIGNED(dst_stride, 4)) {
>-    if (!filtering) {
>-      ScaleRowDown34_0 = ScaleRowDown34_16_DSPR2;
>-      ScaleRowDown34_1 = ScaleRowDown34_16_DSPR2;
>-    } else {
>-      ScaleRowDown34_0 = ScaleRowDown34_0_Box_16_DSPR2;
>-      ScaleRowDown34_1 = ScaleRowDown34_1_Box_16_DSPR2;
>-    }
>-  }
>-#endif
> 
>   for (y = 0; y < dst_height - 2; y += 3) {
>     ScaleRowDown34_0(src_ptr, filter_stride, dst_ptr, dst_width);
>@@ -553,14 +497,14 @@ static void ScalePlaneDown38(int src_width,
>                              int dst_height,
>                              int src_stride,
>                              int dst_stride,
>-                             const uint8* src_ptr,
>-                             uint8* dst_ptr,
>+                             const uint8_t* src_ptr,
>+                             uint8_t* dst_ptr,
>                              enum FilterMode filtering) {
>   int y;
>-  void (*ScaleRowDown38_3)(const uint8* src_ptr, ptrdiff_t src_stride,
>-                           uint8* dst_ptr, int dst_width);
>-  void (*ScaleRowDown38_2)(const uint8* src_ptr, ptrdiff_t src_stride,
>-                           uint8* dst_ptr, int dst_width);
>+  void (*ScaleRowDown38_3)(const uint8_t* src_ptr, ptrdiff_t src_stride,
>+                           uint8_t* dst_ptr, int dst_width);
>+  void (*ScaleRowDown38_2)(const uint8_t* src_ptr, ptrdiff_t src_stride,
>+                           uint8_t* dst_ptr, int dst_width);
>   const int filter_stride = (filtering == kFilterLinear) ? 0 : src_stride;
>   assert(dst_width % 3 == 0);
>   (void)src_width;
>@@ -612,19 +556,6 @@ static void ScalePlaneDown38(int src_width,
>     }
>   }
> #endif
>-#if defined(HAS_SCALEROWDOWN38_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2) && (dst_width % 12 == 0) &&
>-      IS_ALIGNED(src_ptr, 4) && IS_ALIGNED(src_stride, 4) &&
>-      IS_ALIGNED(dst_ptr, 4) && IS_ALIGNED(dst_stride, 4)) {
>-    if (!filtering) {
>-      ScaleRowDown38_3 = ScaleRowDown38_DSPR2;
>-      ScaleRowDown38_2 = ScaleRowDown38_DSPR2;
>-    } else {
>-      ScaleRowDown38_3 = ScaleRowDown38_3_Box_DSPR2;
>-      ScaleRowDown38_2 = ScaleRowDown38_2_Box_DSPR2;
>-    }
>-  }
>-#endif
> #if defined(HAS_SCALEROWDOWN38_MSA)
>   if (TestCpuFlag(kCpuHasMSA)) {
>     if (!filtering) {
>@@ -675,14 +606,14 @@ static void ScalePlaneDown38_16(int src_width,
>                                 int dst_height,
>                                 int src_stride,
>                                 int dst_stride,
>-                                const uint16* src_ptr,
>-                                uint16* dst_ptr,
>+                                const uint16_t* src_ptr,
>+                                uint16_t* dst_ptr,
>                                 enum FilterMode filtering) {
>   int y;
>-  void (*ScaleRowDown38_3)(const uint16* src_ptr, ptrdiff_t src_stride,
>-                           uint16* dst_ptr, int dst_width);
>-  void (*ScaleRowDown38_2)(const uint16* src_ptr, ptrdiff_t src_stride,
>-                           uint16* dst_ptr, int dst_width);
>+  void (*ScaleRowDown38_3)(const uint16_t* src_ptr, ptrdiff_t src_stride,
>+                           uint16_t* dst_ptr, int dst_width);
>+  void (*ScaleRowDown38_2)(const uint16_t* src_ptr, ptrdiff_t src_stride,
>+                           uint16_t* dst_ptr, int dst_width);
>   const int filter_stride = (filtering == kFilterLinear) ? 0 : src_stride;
>   (void)src_width;
>   (void)src_height;
>@@ -716,19 +647,6 @@ static void ScalePlaneDown38_16(int src_width,
>     }
>   }
> #endif
>-#if defined(HAS_SCALEROWDOWN38_16_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2) && (dst_width % 12 == 0) &&
>-      IS_ALIGNED(src_ptr, 4) && IS_ALIGNED(src_stride, 4) &&
>-      IS_ALIGNED(dst_ptr, 4) && IS_ALIGNED(dst_stride, 4)) {
>-    if (!filtering) {
>-      ScaleRowDown38_3 = ScaleRowDown38_16_DSPR2;
>-      ScaleRowDown38_2 = ScaleRowDown38_16_DSPR2;
>-    } else {
>-      ScaleRowDown38_3 = ScaleRowDown38_3_Box_16_DSPR2;
>-      ScaleRowDown38_2 = ScaleRowDown38_2_Box_16_DSPR2;
>-    }
>-  }
>-#endif
> 
>   for (y = 0; y < dst_height - 2; y += 3) {
>     ScaleRowDown38_3(src_ptr, filter_stride, dst_ptr, dst_width);
>@@ -755,8 +673,8 @@ static void ScalePlaneDown38_16(int src_width,
> 
> #define MIN1(x) ((x) < 1 ? 1 : (x))
> 
>-static __inline uint32 SumPixels(int iboxwidth, const uint16* src_ptr) {
>-  uint32 sum = 0u;
>+static __inline uint32_t SumPixels(int iboxwidth, const uint16_t* src_ptr) {
>+  uint32_t sum = 0u;
>   int x;
>   assert(iboxwidth > 0);
>   for (x = 0; x < iboxwidth; ++x) {
>@@ -765,8 +683,8 @@ static __inline uint32 SumPixels(int iboxwidth, const uint16* src_ptr) {
>   return sum;
> }
> 
>-static __inline uint32 SumPixels_16(int iboxwidth, const uint32* src_ptr) {
>-  uint32 sum = 0u;
>+static __inline uint32_t SumPixels_16(int iboxwidth, const uint32_t* src_ptr) {
>+  uint32_t sum = 0u;
>   int x;
>   assert(iboxwidth > 0);
>   for (x = 0; x < iboxwidth; ++x) {
>@@ -779,8 +697,8 @@ static void ScaleAddCols2_C(int dst_width,
>                             int boxheight,
>                             int x,
>                             int dx,
>-                            const uint16* src_ptr,
>-                            uint8* dst_ptr) {
>+                            const uint16_t* src_ptr,
>+                            uint8_t* dst_ptr) {
>   int i;
>   int scaletbl[2];
>   int minboxwidth = dx >> 16;
>@@ -801,8 +719,8 @@ static void ScaleAddCols2_16_C(int dst_width,
>                                int boxheight,
>                                int x,
>                                int dx,
>-                               const uint32* src_ptr,
>-                               uint16* dst_ptr) {
>+                               const uint32_t* src_ptr,
>+                               uint16_t* dst_ptr) {
>   int i;
>   int scaletbl[2];
>   int minboxwidth = dx >> 16;
>@@ -823,8 +741,8 @@ static void ScaleAddCols0_C(int dst_width,
>                             int boxheight,
>                             int x,
>                             int dx,
>-                            const uint16* src_ptr,
>-                            uint8* dst_ptr) {
>+                            const uint16_t* src_ptr,
>+                            uint8_t* dst_ptr) {
>   int scaleval = 65536 / boxheight;
>   int i;
>   (void)dx;
>@@ -838,8 +756,8 @@ static void ScaleAddCols1_C(int dst_width,
>                             int boxheight,
>                             int x,
>                             int dx,
>-                            const uint16* src_ptr,
>-                            uint8* dst_ptr) {
>+                            const uint16_t* src_ptr,
>+                            uint8_t* dst_ptr) {
>   int boxwidth = MIN1(dx >> 16);
>   int scaleval = 65536 / (boxwidth * boxheight);
>   int i;
>@@ -854,8 +772,8 @@ static void ScaleAddCols1_16_C(int dst_width,
>                                int boxheight,
>                                int x,
>                                int dx,
>-                               const uint32* src_ptr,
>-                               uint16* dst_ptr) {
>+                               const uint32_t* src_ptr,
>+                               uint16_t* dst_ptr) {
>   int boxwidth = MIN1(dx >> 16);
>   int scaleval = 65536 / (boxwidth * boxheight);
>   int i;
>@@ -878,8 +796,8 @@ static void ScalePlaneBox(int src_width,
>                           int dst_height,
>                           int src_stride,
>                           int dst_stride,
>-                          const uint8* src_ptr,
>-                          uint8* dst_ptr) {
>+                          const uint8_t* src_ptr,
>+                          uint8_t* dst_ptr) {
>   int j, k;
>   // Initial source x/y coordinate and step values as 16.16 fixed point.
>   int x = 0;
>@@ -891,14 +809,14 @@ static void ScalePlaneBox(int src_width,
>              &dx, &dy);
>   src_width = Abs(src_width);
>   {
>-    // Allocate a row buffer of uint16.
>+    // Allocate a row buffer of uint16_t.
>     align_buffer_64(row16, src_width * 2);
>     void (*ScaleAddCols)(int dst_width, int boxheight, int x, int dx,
>-                         const uint16* src_ptr, uint8* dst_ptr) =
>+                         const uint16_t* src_ptr, uint8_t* dst_ptr) =
>         (dx & 0xffff) ? ScaleAddCols2_C
>                       : ((dx != 0x10000) ? ScaleAddCols1_C : ScaleAddCols0_C);
>-    void (*ScaleAddRow)(const uint8* src_ptr, uint16* dst_ptr, int src_width) =
>-        ScaleAddRow_C;
>+    void (*ScaleAddRow)(const uint8_t* src_ptr, uint16_t* dst_ptr,
>+                        int src_width) = ScaleAddRow_C;
> #if defined(HAS_SCALEADDROW_SSE2)
>     if (TestCpuFlag(kCpuHasSSE2)) {
>       ScaleAddRow = ScaleAddRow_Any_SSE2;
>@@ -931,19 +849,11 @@ static void ScalePlaneBox(int src_width,
>       }
>     }
> #endif
>-#if defined(HAS_SCALEADDROW_DSPR2)
>-    if (TestCpuFlag(kCpuHasDSPR2)) {
>-      ScaleAddRow = ScaleAddRow_Any_DSPR2;
>-      if (IS_ALIGNED(src_width, 16)) {
>-        ScaleAddRow = ScaleAddRow_DSPR2;
>-      }
>-    }
>-#endif
> 
>     for (j = 0; j < dst_height; ++j) {
>       int boxheight;
>       int iy = y >> 16;
>-      const uint8* src = src_ptr + iy * src_stride;
>+      const uint8_t* src = src_ptr + iy * src_stride;
>       y += dy;
>       if (y > max_y) {
>         y = max_y;
>@@ -951,10 +861,10 @@ static void ScalePlaneBox(int src_width,
>       boxheight = MIN1((y >> 16) - iy);
>       memset(row16, 0, src_width * 2);
>       for (k = 0; k < boxheight; ++k) {
>-        ScaleAddRow(src, (uint16*)(row16), src_width);
>+        ScaleAddRow(src, (uint16_t*)(row16), src_width);
>         src += src_stride;
>       }
>-      ScaleAddCols(dst_width, boxheight, x, dx, (uint16*)(row16), dst_ptr);
>+      ScaleAddCols(dst_width, boxheight, x, dx, (uint16_t*)(row16), dst_ptr);
>       dst_ptr += dst_stride;
>     }
>     free_aligned_buffer_64(row16);
>@@ -967,8 +877,8 @@ static void ScalePlaneBox_16(int src_width,
>                              int dst_height,
>                              int src_stride,
>                              int dst_stride,
>-                             const uint16* src_ptr,
>-                             uint16* dst_ptr) {
>+                             const uint16_t* src_ptr,
>+                             uint16_t* dst_ptr) {
>   int j, k;
>   // Initial source x/y coordinate and step values as 16.16 fixed point.
>   int x = 0;
>@@ -980,13 +890,13 @@ static void ScalePlaneBox_16(int src_width,
>              &dx, &dy);
>   src_width = Abs(src_width);
>   {
>-    // Allocate a row buffer of uint32.
>+    // Allocate a row buffer of uint32_t.
>     align_buffer_64(row32, src_width * 4);
>     void (*ScaleAddCols)(int dst_width, int boxheight, int x, int dx,
>-                         const uint32* src_ptr, uint16* dst_ptr) =
>+                         const uint32_t* src_ptr, uint16_t* dst_ptr) =
>         (dx & 0xffff) ? ScaleAddCols2_16_C : ScaleAddCols1_16_C;
>-    void (*ScaleAddRow)(const uint16* src_ptr, uint32* dst_ptr, int src_width) =
>-        ScaleAddRow_16_C;
>+    void (*ScaleAddRow)(const uint16_t* src_ptr, uint32_t* dst_ptr,
>+                        int src_width) = ScaleAddRow_16_C;
> 
> #if defined(HAS_SCALEADDROW_16_SSE2)
>     if (TestCpuFlag(kCpuHasSSE2) && IS_ALIGNED(src_width, 16)) {
>@@ -997,7 +907,7 @@ static void ScalePlaneBox_16(int src_width,
>     for (j = 0; j < dst_height; ++j) {
>       int boxheight;
>       int iy = y >> 16;
>-      const uint16* src = src_ptr + iy * src_stride;
>+      const uint16_t* src = src_ptr + iy * src_stride;
>       y += dy;
>       if (y > max_y) {
>         y = max_y;
>@@ -1005,10 +915,10 @@ static void ScalePlaneBox_16(int src_width,
>       boxheight = MIN1((y >> 16) - iy);
>       memset(row32, 0, src_width * 4);
>       for (k = 0; k < boxheight; ++k) {
>-        ScaleAddRow(src, (uint32*)(row32), src_width);
>+        ScaleAddRow(src, (uint32_t*)(row32), src_width);
>         src += src_stride;
>       }
>-      ScaleAddCols(dst_width, boxheight, x, dx, (uint32*)(row32), dst_ptr);
>+      ScaleAddCols(dst_width, boxheight, x, dx, (uint32_t*)(row32), dst_ptr);
>       dst_ptr += dst_stride;
>     }
>     free_aligned_buffer_64(row32);
>@@ -1022,8 +932,8 @@ void ScalePlaneBilinearDown(int src_width,
>                             int dst_height,
>                             int src_stride,
>                             int dst_stride,
>-                            const uint8* src_ptr,
>-                            uint8* dst_ptr,
>+                            const uint8_t* src_ptr,
>+                            uint8_t* dst_ptr,
>                             enum FilterMode filtering) {
>   // Initial source x/y coordinate and step values as 16.16 fixed point.
>   int x = 0;
>@@ -1036,10 +946,10 @@ void ScalePlaneBilinearDown(int src_width,
> 
>   const int max_y = (src_height - 1) << 16;
>   int j;
>-  void (*ScaleFilterCols)(uint8 * dst_ptr, const uint8* src_ptr, int dst_width,
>-                          int x, int dx) =
>+  void (*ScaleFilterCols)(uint8_t * dst_ptr, const uint8_t* src_ptr,
>+                          int dst_width, int x, int dx) =
>       (src_width >= 32768) ? ScaleFilterCols64_C : ScaleFilterCols_C;
>-  void (*InterpolateRow)(uint8 * dst_ptr, const uint8* src_ptr,
>+  void (*InterpolateRow)(uint8_t * dst_ptr, const uint8_t* src_ptr,
>                          ptrdiff_t src_stride, int dst_width,
>                          int source_y_fraction) = InterpolateRow_C;
>   ScaleSlope(src_width, src_height, dst_width, dst_height, filtering, &x, &y,
>@@ -1070,14 +980,6 @@ void ScalePlaneBilinearDown(int src_width,
>     }
>   }
> #endif
>-#if defined(HAS_INTERPOLATEROW_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2)) {
>-    InterpolateRow = InterpolateRow_Any_DSPR2;
>-    if (IS_ALIGNED(src_width, 4)) {
>-      InterpolateRow = InterpolateRow_DSPR2;
>-    }
>-  }
>-#endif
> #if defined(HAS_INTERPOLATEROW_MSA)
>   if (TestCpuFlag(kCpuHasMSA)) {
>     InterpolateRow = InterpolateRow_Any_MSA;
>@@ -1114,7 +1016,7 @@ void ScalePlaneBilinearDown(int src_width,
> 
>   for (j = 0; j < dst_height; ++j) {
>     int yi = y >> 16;
>-    const uint8* src = src_ptr + yi * src_stride;
>+    const uint8_t* src = src_ptr + yi * src_stride;
>     if (filtering == kFilterLinear) {
>       ScaleFilterCols(dst_ptr, src, dst_width, x, dx);
>     } else {
>@@ -1137,8 +1039,8 @@ void ScalePlaneBilinearDown_16(int src_width,
>                                int dst_height,
>                                int src_stride,
>                                int dst_stride,
>-                               const uint16* src_ptr,
>-                               uint16* dst_ptr,
>+                               const uint16_t* src_ptr,
>+                               uint16_t* dst_ptr,
>                                enum FilterMode filtering) {
>   // Initial source x/y coordinate and step values as 16.16 fixed point.
>   int x = 0;
>@@ -1151,10 +1053,10 @@ void ScalePlaneBilinearDown_16(int src_width,
> 
>   const int max_y = (src_height - 1) << 16;
>   int j;
>-  void (*ScaleFilterCols)(uint16 * dst_ptr, const uint16* src_ptr,
>+  void (*ScaleFilterCols)(uint16_t * dst_ptr, const uint16_t* src_ptr,
>                           int dst_width, int x, int dx) =
>       (src_width >= 32768) ? ScaleFilterCols64_16_C : ScaleFilterCols_16_C;
>-  void (*InterpolateRow)(uint16 * dst_ptr, const uint16* src_ptr,
>+  void (*InterpolateRow)(uint16_t * dst_ptr, const uint16_t* src_ptr,
>                          ptrdiff_t src_stride, int dst_width,
>                          int source_y_fraction) = InterpolateRow_16_C;
>   ScaleSlope(src_width, src_height, dst_width, dst_height, filtering, &x, &y,
>@@ -1193,14 +1095,6 @@ void ScalePlaneBilinearDown_16(int src_width,
>     }
>   }
> #endif
>-#if defined(HAS_INTERPOLATEROW_16_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2)) {
>-    InterpolateRow = InterpolateRow_Any_16_DSPR2;
>-    if (IS_ALIGNED(src_width, 4)) {
>-      InterpolateRow = InterpolateRow_16_DSPR2;
>-    }
>-  }
>-#endif
> 
> #if defined(HAS_SCALEFILTERCOLS_16_SSSE3)
>   if (TestCpuFlag(kCpuHasSSSE3) && src_width < 32768) {
>@@ -1213,13 +1107,13 @@ void ScalePlaneBilinearDown_16(int src_width,
> 
>   for (j = 0; j < dst_height; ++j) {
>     int yi = y >> 16;
>-    const uint16* src = src_ptr + yi * src_stride;
>+    const uint16_t* src = src_ptr + yi * src_stride;
>     if (filtering == kFilterLinear) {
>       ScaleFilterCols(dst_ptr, src, dst_width, x, dx);
>     } else {
>       int yf = (y >> 8) & 255;
>-      InterpolateRow((uint16*)row, src, src_stride, src_width, yf);
>-      ScaleFilterCols(dst_ptr, (uint16*)row, dst_width, x, dx);
>+      InterpolateRow((uint16_t*)row, src, src_stride, src_width, yf);
>+      ScaleFilterCols(dst_ptr, (uint16_t*)row, dst_width, x, dx);
>     }
>     dst_ptr += dst_stride;
>     y += dy;
>@@ -1237,8 +1131,8 @@ void ScalePlaneBilinearUp(int src_width,
>                           int dst_height,
>                           int src_stride,
>                           int dst_stride,
>-                          const uint8* src_ptr,
>-                          uint8* dst_ptr,
>+                          const uint8_t* src_ptr,
>+                          uint8_t* dst_ptr,
>                           enum FilterMode filtering) {
>   int j;
>   // Initial source x/y coordinate and step values as 16.16 fixed point.
>@@ -1247,11 +1141,11 @@ void ScalePlaneBilinearUp(int src_width,
>   int dx = 0;
>   int dy = 0;
>   const int max_y = (src_height - 1) << 16;
>-  void (*InterpolateRow)(uint8 * dst_ptr, const uint8* src_ptr,
>+  void (*InterpolateRow)(uint8_t * dst_ptr, const uint8_t* src_ptr,
>                          ptrdiff_t src_stride, int dst_width,
>                          int source_y_fraction) = InterpolateRow_C;
>-  void (*ScaleFilterCols)(uint8 * dst_ptr, const uint8* src_ptr, int dst_width,
>-                          int x, int dx) =
>+  void (*ScaleFilterCols)(uint8_t * dst_ptr, const uint8_t* src_ptr,
>+                          int dst_width, int x, int dx) =
>       filtering ? ScaleFilterCols_C : ScaleCols_C;
>   ScaleSlope(src_width, src_height, dst_width, dst_height, filtering, &x, &y,
>              &dx, &dy);
>@@ -1281,14 +1175,6 @@ void ScalePlaneBilinearUp(int src_width,
>     }
>   }
> #endif
>-#if defined(HAS_INTERPOLATEROW_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2)) {
>-    InterpolateRow = InterpolateRow_Any_DSPR2;
>-    if (IS_ALIGNED(dst_width, 4)) {
>-      InterpolateRow = InterpolateRow_DSPR2;
>-    }
>-  }
>-#endif
> 
>   if (filtering && src_width >= 32768) {
>     ScaleFilterCols = ScaleFilterCols64_C;
>@@ -1328,13 +1214,13 @@ void ScalePlaneBilinearUp(int src_width,
>   }
>   {
>     int yi = y >> 16;
>-    const uint8* src = src_ptr + yi * src_stride;
>+    const uint8_t* src = src_ptr + yi * src_stride;
> 
>     // Allocate 2 row buffers.
>     const int kRowSize = (dst_width + 31) & ~31;
>     align_buffer_64(row, kRowSize * 2);
> 
>-    uint8* rowptr = row;
>+    uint8_t* rowptr = row;
>     int rowstride = kRowSize;
>     int lasty = yi;
> 
>@@ -1380,8 +1266,8 @@ void ScalePlaneBilinearUp_16(int src_width,
>                              int dst_height,
>                              int src_stride,
>                              int dst_stride,
>-                             const uint16* src_ptr,
>-                             uint16* dst_ptr,
>+                             const uint16_t* src_ptr,
>+                             uint16_t* dst_ptr,
>                              enum FilterMode filtering) {
>   int j;
>   // Initial source x/y coordinate and step values as 16.16 fixed point.
>@@ -1390,10 +1276,10 @@ void ScalePlaneBilinearUp_16(int src_width,
>   int dx = 0;
>   int dy = 0;
>   const int max_y = (src_height - 1) << 16;
>-  void (*InterpolateRow)(uint16 * dst_ptr, const uint16* src_ptr,
>+  void (*InterpolateRow)(uint16_t * dst_ptr, const uint16_t* src_ptr,
>                          ptrdiff_t src_stride, int dst_width,
>                          int source_y_fraction) = InterpolateRow_16_C;
>-  void (*ScaleFilterCols)(uint16 * dst_ptr, const uint16* src_ptr,
>+  void (*ScaleFilterCols)(uint16_t * dst_ptr, const uint16_t* src_ptr,
>                           int dst_width, int x, int dx) =
>       filtering ? ScaleFilterCols_16_C : ScaleCols_16_C;
>   ScaleSlope(src_width, src_height, dst_width, dst_height, filtering, &x, &y,
>@@ -1432,14 +1318,6 @@ void ScalePlaneBilinearUp_16(int src_width,
>     }
>   }
> #endif
>-#if defined(HAS_INTERPOLATEROW_16_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2)) {
>-    InterpolateRow = InterpolateRow_Any_16_DSPR2;
>-    if (IS_ALIGNED(dst_width, 4)) {
>-      InterpolateRow = InterpolateRow_16_DSPR2;
>-    }
>-  }
>-#endif
> 
>   if (filtering && src_width >= 32768) {
>     ScaleFilterCols = ScaleFilterCols64_16_C;
>@@ -1463,13 +1341,13 @@ void ScalePlaneBilinearUp_16(int src_width,
>   }
>   {
>     int yi = y >> 16;
>-    const uint16* src = src_ptr + yi * src_stride;
>+    const uint16_t* src = src_ptr + yi * src_stride;
> 
>     // Allocate 2 row buffers.
>     const int kRowSize = (dst_width + 31) & ~31;
>     align_buffer_64(row, kRowSize * 4);
> 
>-    uint16* rowptr = (uint16*)row;
>+    uint16_t* rowptr = (uint16_t*)row;
>     int rowstride = kRowSize;
>     int lasty = yi;
> 
>@@ -1520,11 +1398,11 @@ static void ScalePlaneSimple(int src_width,
>                              int dst_height,
>                              int src_stride,
>                              int dst_stride,
>-                             const uint8* src_ptr,
>-                             uint8* dst_ptr) {
>+                             const uint8_t* src_ptr,
>+                             uint8_t* dst_ptr) {
>   int i;
>-  void (*ScaleCols)(uint8 * dst_ptr, const uint8* src_ptr, int dst_width, int x,
>-                    int dx) = ScaleCols_C;
>+  void (*ScaleCols)(uint8_t * dst_ptr, const uint8_t* src_ptr, int dst_width,
>+                    int x, int dx) = ScaleCols_C;
>   // Initial source x/y coordinate and step values as 16.16 fixed point.
>   int x = 0;
>   int y = 0;
>@@ -1556,10 +1434,10 @@ static void ScalePlaneSimple_16(int src_width,
>                                 int dst_height,
>                                 int src_stride,
>                                 int dst_stride,
>-                                const uint16* src_ptr,
>-                                uint16* dst_ptr) {
>+                                const uint16_t* src_ptr,
>+                                uint16_t* dst_ptr) {
>   int i;
>-  void (*ScaleCols)(uint16 * dst_ptr, const uint16* src_ptr, int dst_width,
>+  void (*ScaleCols)(uint16_t * dst_ptr, const uint16_t* src_ptr, int dst_width,
>                     int x, int dx) = ScaleCols_16_C;
>   // Initial source x/y coordinate and step values as 16.16 fixed point.
>   int x = 0;
>@@ -1590,11 +1468,11 @@ static void ScalePlaneSimple_16(int src_width,
> // This function dispatches to a specialized scaler based on scale factor.
> 
> LIBYUV_API
>-void ScalePlane(const uint8* src,
>+void ScalePlane(const uint8_t* src,
>                 int src_stride,
>                 int src_width,
>                 int src_height,
>-                uint8* dst,
>+                uint8_t* dst,
>                 int dst_stride,
>                 int dst_width,
>                 int dst_height,
>@@ -1673,11 +1551,11 @@ void ScalePlane(const uint8* src,
> }
> 
> LIBYUV_API
>-void ScalePlane_16(const uint16* src,
>+void ScalePlane_16(const uint16_t* src,
>                    int src_stride,
>                    int src_width,
>                    int src_height,
>-                   uint16* dst,
>+                   uint16_t* dst,
>                    int dst_stride,
>                    int dst_width,
>                    int dst_height,
>@@ -1759,19 +1637,19 @@ void ScalePlane_16(const uint16* src,
> // This function in turn calls a scaling function for each plane.
> 
> LIBYUV_API
>-int I420Scale(const uint8* src_y,
>+int I420Scale(const uint8_t* src_y,
>               int src_stride_y,
>-              const uint8* src_u,
>+              const uint8_t* src_u,
>               int src_stride_u,
>-              const uint8* src_v,
>+              const uint8_t* src_v,
>               int src_stride_v,
>               int src_width,
>               int src_height,
>-              uint8* dst_y,
>+              uint8_t* dst_y,
>               int dst_stride_y,
>-              uint8* dst_u,
>+              uint8_t* dst_u,
>               int dst_stride_u,
>-              uint8* dst_v,
>+              uint8_t* dst_v,
>               int dst_stride_v,
>               int dst_width,
>               int dst_height,
>@@ -1796,19 +1674,19 @@ int I420Scale(const uint8* src_y,
> }
> 
> LIBYUV_API
>-int I420Scale_16(const uint16* src_y,
>+int I420Scale_16(const uint16_t* src_y,
>                  int src_stride_y,
>-                 const uint16* src_u,
>+                 const uint16_t* src_u,
>                  int src_stride_u,
>-                 const uint16* src_v,
>+                 const uint16_t* src_v,
>                  int src_stride_v,
>                  int src_width,
>                  int src_height,
>-                 uint16* dst_y,
>+                 uint16_t* dst_y,
>                  int dst_stride_y,
>-                 uint16* dst_u,
>+                 uint16_t* dst_u,
>                  int dst_stride_u,
>-                 uint16* dst_v,
>+                 uint16_t* dst_v,
>                  int dst_stride_v,
>                  int dst_width,
>                  int dst_height,
>@@ -1834,17 +1712,17 @@ int I420Scale_16(const uint16* src_y,
> 
> // Deprecated api
> LIBYUV_API
>-int Scale(const uint8* src_y,
>-          const uint8* src_u,
>-          const uint8* src_v,
>+int Scale(const uint8_t* src_y,
>+          const uint8_t* src_u,
>+          const uint8_t* src_v,
>           int src_stride_y,
>           int src_stride_u,
>           int src_stride_v,
>           int src_width,
>           int src_height,
>-          uint8* dst_y,
>-          uint8* dst_u,
>-          uint8* dst_v,
>+          uint8_t* dst_y,
>+          uint8_t* dst_u,
>+          uint8_t* dst_v,
>           int dst_stride_y,
>           int dst_stride_u,
>           int dst_stride_v,
>@@ -1857,43 +1735,6 @@ int Scale(const uint8* src_y,
>                    dst_height, interpolate ? kFilterBox : kFilterNone);
> }
> 
>-// Deprecated api
>-LIBYUV_API
>-int ScaleOffset(const uint8* src,
>-                int src_width,
>-                int src_height,
>-                uint8* dst,
>-                int dst_width,
>-                int dst_height,
>-                int dst_yoffset,
>-                LIBYUV_BOOL interpolate) {
>-  // Chroma requires offset to multiple of 2.
>-  int dst_yoffset_even = dst_yoffset & ~1;
>-  int src_halfwidth = SUBSAMPLE(src_width, 1, 1);
>-  int src_halfheight = SUBSAMPLE(src_height, 1, 1);
>-  int dst_halfwidth = SUBSAMPLE(dst_width, 1, 1);
>-  int dst_halfheight = SUBSAMPLE(dst_height, 1, 1);
>-  int aheight = dst_height - dst_yoffset_even * 2;  // actual output height
>-  const uint8* src_y = src;
>-  const uint8* src_u = src + src_width * src_height;
>-  const uint8* src_v =
>-      src + src_width * src_height + src_halfwidth * src_halfheight;
>-  uint8* dst_y = dst + dst_yoffset_even * dst_width;
>-  uint8* dst_u =
>-      dst + dst_width * dst_height + (dst_yoffset_even >> 1) * dst_halfwidth;
>-  uint8* dst_v = dst + dst_width * dst_height + dst_halfwidth * dst_halfheight +
>-                 (dst_yoffset_even >> 1) * dst_halfwidth;
>-  if (!src || src_width <= 0 || src_height <= 0 || !dst || dst_width <= 0 ||
>-      dst_height <= 0 || dst_yoffset_even < 0 ||
>-      dst_yoffset_even >= dst_height) {
>-    return -1;
>-  }
>-  return I420Scale(src_y, src_width, src_u, src_halfwidth, src_v, src_halfwidth,
>-                   src_width, src_height, dst_y, dst_width, dst_u,
>-                   dst_halfwidth, dst_v, dst_halfwidth, dst_width, aheight,
>-                   interpolate ? kFilterBox : kFilterNone);
>-}
>-
> #ifdef __cplusplus
> }  // extern "C"
> }  // namespace libyuv
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/scale_any.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/scale_any.cc
>index c4d6626ab72..53ad1364049 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/scale_any.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/scale_any.cc
>@@ -19,15 +19,15 @@ extern "C" {
> #endif
> 
> // Definition for ScaleFilterCols, ScaleARGBCols and ScaleARGBFilterCols
>-#define CANY(NAMEANY, TERP_SIMD, TERP_C, BPP, MASK)                        \
>-  void NAMEANY(uint8* dst_ptr, const uint8* src_ptr, int dst_width, int x, \
>-               int dx) {                                                   \
>-    int r = dst_width & MASK;                                              \
>-    int n = dst_width & ~MASK;                                             \
>-    if (n > 0) {                                                           \
>-      TERP_SIMD(dst_ptr, src_ptr, n, x, dx);                               \
>-    }                                                                      \
>-    TERP_C(dst_ptr + n * BPP, src_ptr, r, x + n * dx, dx);                 \
>+#define CANY(NAMEANY, TERP_SIMD, TERP_C, BPP, MASK)                            \
>+  void NAMEANY(uint8_t* dst_ptr, const uint8_t* src_ptr, int dst_width, int x, \
>+               int dx) {                                                       \
>+    int r = dst_width & MASK;                                                  \
>+    int n = dst_width & ~MASK;                                                 \
>+    if (n > 0) {                                                               \
>+      TERP_SIMD(dst_ptr, src_ptr, n, x, dx);                                   \
>+    }                                                                          \
>+    TERP_C(dst_ptr + n * BPP, src_ptr, r, x + n * dx, dx);                     \
>   }
> 
> #ifdef HAS_SCALEFILTERCOLS_NEON
>@@ -60,31 +60,31 @@ CANY(ScaleARGBFilterCols_Any_MSA,
> 
> // Fixed scale down.
> // Mask may be non-power of 2, so use MOD
>-#define SDANY(NAMEANY, SCALEROWDOWN_SIMD, SCALEROWDOWN_C, FACTOR, BPP, MASK) \
>-  void NAMEANY(const uint8* src_ptr, ptrdiff_t src_stride, uint8* dst_ptr,   \
>-               int dst_width) {                                              \
>-    int r = (int)((unsigned int)dst_width % (MASK + 1)); /* NOLINT */        \
>-    int n = dst_width - r;                                                   \
>-    if (n > 0) {                                                             \
>-      SCALEROWDOWN_SIMD(src_ptr, src_stride, dst_ptr, n);                    \
>-    }                                                                        \
>-    SCALEROWDOWN_C(src_ptr + (n * FACTOR) * BPP, src_stride,                 \
>-                   dst_ptr + n * BPP, r);                                    \
>+#define SDANY(NAMEANY, SCALEROWDOWN_SIMD, SCALEROWDOWN_C, FACTOR, BPP, MASK)   \
>+  void NAMEANY(const uint8_t* src_ptr, ptrdiff_t src_stride, uint8_t* dst_ptr, \
>+               int dst_width) {                                                \
>+    int r = (int)((unsigned int)dst_width % (MASK + 1)); /* NOLINT */          \
>+    int n = dst_width - r;                                                     \
>+    if (n > 0) {                                                               \
>+      SCALEROWDOWN_SIMD(src_ptr, src_stride, dst_ptr, n);                      \
>+    }                                                                          \
>+    SCALEROWDOWN_C(src_ptr + (n * FACTOR) * BPP, src_stride,                   \
>+                   dst_ptr + n * BPP, r);                                      \
>   }
> 
> // Fixed scale down for odd source width.  Used by I420Blend subsampling.
> // Since dst_width is (width + 1) / 2, this function scales one less pixel
> // and copies the last pixel.
>-#define SDODD(NAMEANY, SCALEROWDOWN_SIMD, SCALEROWDOWN_C, FACTOR, BPP, MASK) \
>-  void NAMEANY(const uint8* src_ptr, ptrdiff_t src_stride, uint8* dst_ptr,   \
>-               int dst_width) {                                              \
>-    int r = (int)((unsigned int)(dst_width - 1) % (MASK + 1)); /* NOLINT */  \
>-    int n = (dst_width - 1) - r;                                             \
>-    if (n > 0) {                                                             \
>-      SCALEROWDOWN_SIMD(src_ptr, src_stride, dst_ptr, n);                    \
>-    }                                                                        \
>-    SCALEROWDOWN_C(src_ptr + (n * FACTOR) * BPP, src_stride,                 \
>-                   dst_ptr + n * BPP, r + 1);                                \
>+#define SDODD(NAMEANY, SCALEROWDOWN_SIMD, SCALEROWDOWN_C, FACTOR, BPP, MASK)   \
>+  void NAMEANY(const uint8_t* src_ptr, ptrdiff_t src_stride, uint8_t* dst_ptr, \
>+               int dst_width) {                                                \
>+    int r = (int)((unsigned int)(dst_width - 1) % (MASK + 1)); /* NOLINT */    \
>+    int n = (dst_width - 1) - r;                                               \
>+    if (n > 0) {                                                               \
>+      SCALEROWDOWN_SIMD(src_ptr, src_stride, dst_ptr, n);                      \
>+    }                                                                          \
>+    SCALEROWDOWN_C(src_ptr + (n * FACTOR) * BPP, src_stride,                   \
>+                   dst_ptr + n * BPP, r + 1);                                  \
>   }
> 
> #ifdef HAS_SCALEROWDOWN2_SSSE3
>@@ -385,16 +385,16 @@ SDANY(ScaleARGBRowDown2Box_Any_MSA,
> #undef SDANY
> 
> // Scale down by even scale factor.
>-#define SDAANY(NAMEANY, SCALEROWDOWN_SIMD, SCALEROWDOWN_C, BPP, MASK)      \
>-  void NAMEANY(const uint8* src_ptr, ptrdiff_t src_stride, int src_stepx,  \
>-               uint8* dst_ptr, int dst_width) {                            \
>-    int r = dst_width & MASK;                                              \
>-    int n = dst_width & ~MASK;                                             \
>-    if (n > 0) {                                                           \
>-      SCALEROWDOWN_SIMD(src_ptr, src_stride, src_stepx, dst_ptr, n);       \
>-    }                                                                      \
>-    SCALEROWDOWN_C(src_ptr + (n * src_stepx) * BPP, src_stride, src_stepx, \
>-                   dst_ptr + n * BPP, r);                                  \
>+#define SDAANY(NAMEANY, SCALEROWDOWN_SIMD, SCALEROWDOWN_C, BPP, MASK)       \
>+  void NAMEANY(const uint8_t* src_ptr, ptrdiff_t src_stride, int src_stepx, \
>+               uint8_t* dst_ptr, int dst_width) {                           \
>+    int r = dst_width & MASK;                                               \
>+    int n = dst_width & ~MASK;                                              \
>+    if (n > 0) {                                                            \
>+      SCALEROWDOWN_SIMD(src_ptr, src_stride, src_stepx, dst_ptr, n);        \
>+    }                                                                       \
>+    SCALEROWDOWN_C(src_ptr + (n * src_stepx) * BPP, src_stride, src_stepx,  \
>+                   dst_ptr + n * BPP, r);                                   \
>   }
> 
> #ifdef HAS_SCALEARGBROWDOWNEVEN_SSE2
>@@ -435,13 +435,13 @@ SDAANY(ScaleARGBRowDownEvenBox_Any_MSA,
> #endif
> 
> // Add rows box filter scale down.
>-#define SAANY(NAMEANY, SCALEADDROW_SIMD, SCALEADDROW_C, MASK)          \
>-  void NAMEANY(const uint8* src_ptr, uint16* dst_ptr, int src_width) { \
>-    int n = src_width & ~MASK;                                         \
>-    if (n > 0) {                                                       \
>-      SCALEADDROW_SIMD(src_ptr, dst_ptr, n);                           \
>-    }                                                                  \
>-    SCALEADDROW_C(src_ptr + n, dst_ptr + n, src_width & MASK);         \
>+#define SAANY(NAMEANY, SCALEADDROW_SIMD, SCALEADDROW_C, MASK)              \
>+  void NAMEANY(const uint8_t* src_ptr, uint16_t* dst_ptr, int src_width) { \
>+    int n = src_width & ~MASK;                                             \
>+    if (n > 0) {                                                           \
>+      SCALEADDROW_SIMD(src_ptr, dst_ptr, n);                               \
>+    }                                                                      \
>+    SCALEADDROW_C(src_ptr + n, dst_ptr + n, src_width & MASK);             \
>   }
> 
> #ifdef HAS_SCALEADDROW_SSE2
>@@ -456,9 +456,6 @@ SAANY(ScaleAddRow_Any_NEON, ScaleAddRow_NEON, ScaleAddRow_C, 15)
> #ifdef HAS_SCALEADDROW_MSA
> SAANY(ScaleAddRow_Any_MSA, ScaleAddRow_MSA, ScaleAddRow_C, 15)
> #endif
>-#ifdef HAS_SCALEADDROW_DSPR2
>-SAANY(ScaleAddRow_Any_DSPR2, ScaleAddRow_DSPR2, ScaleAddRow_C, 15)
>-#endif
> #undef SAANY
> 
> #ifdef __cplusplus
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/scale_argb.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/scale_argb.cc
>index c3ec7d6bb67..53a22e8b41e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/scale_argb.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/scale_argb.cc
>@@ -36,8 +36,8 @@ static void ScaleARGBDown2(int src_width,
>                            int dst_height,
>                            int src_stride,
>                            int dst_stride,
>-                           const uint8* src_argb,
>-                           uint8* dst_argb,
>+                           const uint8_t* src_argb,
>+                           uint8_t* dst_argb,
>                            int x,
>                            int dx,
>                            int y,
>@@ -45,8 +45,8 @@ static void ScaleARGBDown2(int src_width,
>                            enum FilterMode filtering) {
>   int j;
>   int row_stride = src_stride * (dy >> 16);
>-  void (*ScaleARGBRowDown2)(const uint8* src_argb, ptrdiff_t src_stride,
>-                            uint8* dst_argb, int dst_width) =
>+  void (*ScaleARGBRowDown2)(const uint8_t* src_argb, ptrdiff_t src_stride,
>+                            uint8_t* dst_argb, int dst_width) =
>       filtering == kFilterNone
>           ? ScaleARGBRowDown2_C
>           : (filtering == kFilterLinear ? ScaleARGBRowDown2Linear_C
>@@ -131,8 +131,8 @@ static void ScaleARGBDown4Box(int src_width,
>                               int dst_height,
>                               int src_stride,
>                               int dst_stride,
>-                              const uint8* src_argb,
>-                              uint8* dst_argb,
>+                              const uint8_t* src_argb,
>+                              uint8_t* dst_argb,
>                               int x,
>                               int dx,
>                               int y,
>@@ -142,8 +142,8 @@ static void ScaleARGBDown4Box(int src_width,
>   const int kRowSize = (dst_width * 2 * 4 + 31) & ~31;
>   align_buffer_64(row, kRowSize * 2);
>   int row_stride = src_stride * (dy >> 16);
>-  void (*ScaleARGBRowDown2)(const uint8* src_argb, ptrdiff_t src_stride,
>-                            uint8* dst_argb, int dst_width) =
>+  void (*ScaleARGBRowDown2)(const uint8_t* src_argb, ptrdiff_t src_stride,
>+                            uint8_t* dst_argb, int dst_width) =
>       ScaleARGBRowDown2Box_C;
>   // Advance to odd row, even column.
>   src_argb += (y >> 16) * src_stride + (x >> 16) * 4;
>@@ -189,8 +189,8 @@ static void ScaleARGBDownEven(int src_width,
>                               int dst_height,
>                               int src_stride,
>                               int dst_stride,
>-                              const uint8* src_argb,
>-                              uint8* dst_argb,
>+                              const uint8_t* src_argb,
>+                              uint8_t* dst_argb,
>                               int x,
>                               int dx,
>                               int y,
>@@ -199,8 +199,8 @@ static void ScaleARGBDownEven(int src_width,
>   int j;
>   int col_step = dx >> 16;
>   int row_stride = (dy >> 16) * src_stride;
>-  void (*ScaleARGBRowDownEven)(const uint8* src_argb, ptrdiff_t src_stride,
>-                               int src_step, uint8* dst_argb, int dst_width) =
>+  void (*ScaleARGBRowDownEven)(const uint8_t* src_argb, ptrdiff_t src_stride,
>+                               int src_step, uint8_t* dst_argb, int dst_width) =
>       filtering ? ScaleARGBRowDownEvenBox_C : ScaleARGBRowDownEven_C;
>   (void)src_width;
>   (void)src_height;
>@@ -255,23 +255,23 @@ static void ScaleARGBBilinearDown(int src_width,
>                                   int dst_height,
>                                   int src_stride,
>                                   int dst_stride,
>-                                  const uint8* src_argb,
>-                                  uint8* dst_argb,
>+                                  const uint8_t* src_argb,
>+                                  uint8_t* dst_argb,
>                                   int x,
>                                   int dx,
>                                   int y,
>                                   int dy,
>                                   enum FilterMode filtering) {
>   int j;
>-  void (*InterpolateRow)(uint8 * dst_argb, const uint8* src_argb,
>+  void (*InterpolateRow)(uint8_t * dst_argb, const uint8_t* src_argb,
>                          ptrdiff_t src_stride, int dst_width,
>                          int source_y_fraction) = InterpolateRow_C;
>-  void (*ScaleARGBFilterCols)(uint8 * dst_argb, const uint8* src_argb,
>+  void (*ScaleARGBFilterCols)(uint8_t * dst_argb, const uint8_t* src_argb,
>                               int dst_width, int x, int dx) =
>       (src_width >= 32768) ? ScaleARGBFilterCols64_C : ScaleARGBFilterCols_C;
>-  int64 xlast = x + (int64)(dst_width - 1) * dx;
>-  int64 xl = (dx >= 0) ? x : xlast;
>-  int64 xr = (dx >= 0) ? xlast : x;
>+  int64_t xlast = x + (int64_t)(dst_width - 1) * dx;
>+  int64_t xl = (dx >= 0) ? x : xlast;
>+  int64_t xr = (dx >= 0) ? xlast : x;
>   int clip_src_width;
>   xl = (xl >> 16) & ~3;    // Left edge aligned.
>   xr = (xr >> 16) + 1;     // Right most pixel used.  Bilinear uses 2 pixels.
>@@ -306,15 +306,6 @@ static void ScaleARGBBilinearDown(int src_width,
>     }
>   }
> #endif
>-#if defined(HAS_INTERPOLATEROW_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2) && IS_ALIGNED(src_argb, 4) &&
>-      IS_ALIGNED(src_stride, 4)) {
>-    InterpolateRow = InterpolateRow_Any_DSPR2;
>-    if (IS_ALIGNED(clip_src_width, 4)) {
>-      InterpolateRow = InterpolateRow_DSPR2;
>-    }
>-  }
>-#endif
> #if defined(HAS_INTERPOLATEROW_MSA)
>   if (TestCpuFlag(kCpuHasMSA)) {
>     InterpolateRow = InterpolateRow_Any_MSA;
>@@ -355,7 +346,7 @@ static void ScaleARGBBilinearDown(int src_width,
>     }
>     for (j = 0; j < dst_height; ++j) {
>       int yi = y >> 16;
>-      const uint8* src = src_argb + yi * src_stride;
>+      const uint8_t* src = src_argb + yi * src_stride;
>       if (filtering == kFilterLinear) {
>         ScaleARGBFilterCols(dst_argb, src, dst_width, x, dx);
>       } else {
>@@ -380,18 +371,18 @@ static void ScaleARGBBilinearUp(int src_width,
>                                 int dst_height,
>                                 int src_stride,
>                                 int dst_stride,
>-                                const uint8* src_argb,
>-                                uint8* dst_argb,
>+                                const uint8_t* src_argb,
>+                                uint8_t* dst_argb,
>                                 int x,
>                                 int dx,
>                                 int y,
>                                 int dy,
>                                 enum FilterMode filtering) {
>   int j;
>-  void (*InterpolateRow)(uint8 * dst_argb, const uint8* src_argb,
>+  void (*InterpolateRow)(uint8_t * dst_argb, const uint8_t* src_argb,
>                          ptrdiff_t src_stride, int dst_width,
>                          int source_y_fraction) = InterpolateRow_C;
>-  void (*ScaleARGBFilterCols)(uint8 * dst_argb, const uint8* src_argb,
>+  void (*ScaleARGBFilterCols)(uint8_t * dst_argb, const uint8_t* src_argb,
>                               int dst_width, int x, int dx) =
>       filtering ? ScaleARGBFilterCols_C : ScaleARGBCols_C;
>   const int max_y = (src_height - 1) << 16;
>@@ -419,12 +410,6 @@ static void ScaleARGBBilinearUp(int src_width,
>     }
>   }
> #endif
>-#if defined(HAS_INTERPOLATEROW_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2) && IS_ALIGNED(dst_argb, 4) &&
>-      IS_ALIGNED(dst_stride, 4)) {
>-    InterpolateRow = InterpolateRow_DSPR2;
>-  }
>-#endif
> #if defined(HAS_INTERPOLATEROW_MSA)
>   if (TestCpuFlag(kCpuHasMSA)) {
>     InterpolateRow = InterpolateRow_Any_MSA;
>@@ -494,13 +479,13 @@ static void ScaleARGBBilinearUp(int src_width,
> 
>   {
>     int yi = y >> 16;
>-    const uint8* src = src_argb + yi * src_stride;
>+    const uint8_t* src = src_argb + yi * src_stride;
> 
>     // Allocate 2 rows of ARGB.
>     const int kRowSize = (dst_width * 4 + 31) & ~31;
>     align_buffer_64(row, kRowSize * 2);
> 
>-    uint8* rowptr = row;
>+    uint8_t* rowptr = row;
>     int rowstride = kRowSize;
>     int lasty = yi;
> 
>@@ -550,18 +535,18 @@ static void ScaleYUVToARGBBilinearUp(int src_width,
>                                      int src_stride_u,
>                                      int src_stride_v,
>                                      int dst_stride_argb,
>-                                     const uint8* src_y,
>-                                     const uint8* src_u,
>-                                     const uint8* src_v,
>-                                     uint8* dst_argb,
>+                                     const uint8_t* src_y,
>+                                     const uint8_t* src_u,
>+                                     const uint8_t* src_v,
>+                                     uint8_t* dst_argb,
>                                      int x,
>                                      int dx,
>                                      int y,
>                                      int dy,
>                                      enum FilterMode filtering) {
>   int j;
>-  void (*I422ToARGBRow)(const uint8* y_buf, const uint8* u_buf,
>-                        const uint8* v_buf, uint8* rgb_buf, int width) =
>+  void (*I422ToARGBRow)(const uint8_t* y_buf, const uint8_t* u_buf,
>+                        const uint8_t* v_buf, uint8_t* rgb_buf, int width) =
>       I422ToARGBRow_C;
> #if defined(HAS_I422TOARGBROW_SSSE3)
>   if (TestCpuFlag(kCpuHasSSSE3)) {
>@@ -587,15 +572,6 @@ static void ScaleYUVToARGBBilinearUp(int src_width,
>     }
>   }
> #endif
>-#if defined(HAS_I422TOARGBROW_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2) && IS_ALIGNED(src_width, 4) &&
>-      IS_ALIGNED(src_y, 4) && IS_ALIGNED(src_stride_y, 4) &&
>-      IS_ALIGNED(src_u, 2) && IS_ALIGNED(src_stride_u, 2) &&
>-      IS_ALIGNED(src_v, 2) && IS_ALIGNED(src_stride_v, 2) &&
>-      IS_ALIGNED(dst_argb, 4) && IS_ALIGNED(dst_stride_argb, 4)) {
>-    I422ToARGBRow = I422ToARGBRow_DSPR2;
>-  }
>-#endif
> #if defined(HAS_I422TOARGBROW_MSA)
>   if (TestCpuFlag(kCpuHasMSA)) {
>     I422ToARGBRow = I422ToARGBRow_Any_MSA;
>@@ -605,7 +581,7 @@ static void ScaleYUVToARGBBilinearUp(int src_width,
>   }
> #endif
> 
>-  void (*InterpolateRow)(uint8 * dst_argb, const uint8* src_argb,
>+  void (*InterpolateRow)(uint8_t * dst_argb, const uint8_t* src_argb,
>                          ptrdiff_t src_stride, int dst_width,
>                          int source_y_fraction) = InterpolateRow_C;
> #if defined(HAS_INTERPOLATEROW_SSSE3)
>@@ -632,12 +608,6 @@ static void ScaleYUVToARGBBilinearUp(int src_width,
>     }
>   }
> #endif
>-#if defined(HAS_INTERPOLATEROW_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2) && IS_ALIGNED(dst_argb, 4) &&
>-      IS_ALIGNED(dst_stride_argb, 4)) {
>-    InterpolateRow = InterpolateRow_DSPR2;
>-  }
>-#endif
> #if defined(HAS_INTERPOLATEROW_MSA)
>   if (TestCpuFlag(kCpuHasMSA)) {
>     InterpolateRow = InterpolateRow_Any_MSA;
>@@ -647,7 +617,7 @@ static void ScaleYUVToARGBBilinearUp(int src_width,
>   }
> #endif
> 
>-  void (*ScaleARGBFilterCols)(uint8 * dst_argb, const uint8* src_argb,
>+  void (*ScaleARGBFilterCols)(uint8_t * dst_argb, const uint8_t* src_argb,
>                               int dst_width, int x, int dx) =
>       filtering ? ScaleARGBFilterCols_C : ScaleARGBCols_C;
>   if (src_width >= 32768) {
>@@ -712,9 +682,9 @@ static void ScaleYUVToARGBBilinearUp(int src_width,
>   const int kYShift = 1;  // Shift Y by 1 to convert Y plane to UV coordinate.
>   int yi = y >> 16;
>   int uv_yi = yi >> kYShift;
>-  const uint8* src_row_y = src_y + yi * src_stride_y;
>-  const uint8* src_row_u = src_u + uv_yi * src_stride_u;
>-  const uint8* src_row_v = src_v + uv_yi * src_stride_v;
>+  const uint8_t* src_row_y = src_y + yi * src_stride_y;
>+  const uint8_t* src_row_u = src_u + uv_yi * src_stride_u;
>+  const uint8_t* src_row_v = src_v + uv_yi * src_stride_v;
> 
>   // Allocate 2 rows of ARGB.
>   const int kRowSize = (dst_width * 4 + 31) & ~31;
>@@ -723,7 +693,7 @@ static void ScaleYUVToARGBBilinearUp(int src_width,
>   // Allocate 1 row of ARGB for source conversion.
>   align_buffer_64(argb_row, src_width * 4);
> 
>-  uint8* rowptr = row;
>+  uint8_t* rowptr = row;
>   int rowstride = kRowSize;
>   int lasty = yi;
> 
>@@ -795,15 +765,15 @@ static void ScaleARGBSimple(int src_width,
>                             int dst_height,
>                             int src_stride,
>                             int dst_stride,
>-                            const uint8* src_argb,
>-                            uint8* dst_argb,
>+                            const uint8_t* src_argb,
>+                            uint8_t* dst_argb,
>                             int x,
>                             int dx,
>                             int y,
>                             int dy) {
>   int j;
>-  void (*ScaleARGBCols)(uint8 * dst_argb, const uint8* src_argb, int dst_width,
>-                        int x, int dx) =
>+  void (*ScaleARGBCols)(uint8_t * dst_argb, const uint8_t* src_argb,
>+                        int dst_width, int x, int dx) =
>       (src_width >= 32768) ? ScaleARGBCols64_C : ScaleARGBCols_C;
>   (void)src_height;
> #if defined(HAS_SCALEARGBCOLS_SSE2)
>@@ -847,11 +817,11 @@ static void ScaleARGBSimple(int src_width,
> // ScaleARGB a ARGB.
> // This function in turn calls a scaling function
> // suitable for handling the desired resolutions.
>-static void ScaleARGB(const uint8* src,
>+static void ScaleARGB(const uint8_t* src,
>                       int src_stride,
>                       int src_width,
>                       int src_height,
>-                      uint8* dst,
>+                      uint8_t* dst,
>                       int dst_stride,
>                       int dst_width,
>                       int dst_height,
>@@ -880,13 +850,13 @@ static void ScaleARGB(const uint8* src,
>              &dx, &dy);
>   src_width = Abs(src_width);
>   if (clip_x) {
>-    int64 clipf = (int64)(clip_x)*dx;
>+    int64_t clipf = (int64_t)(clip_x)*dx;
>     x += (clipf & 0xffff);
>     src += (clipf >> 16) * 4;
>     dst += clip_x * 4;
>   }
>   if (clip_y) {
>-    int64 clipf = (int64)(clip_y)*dy;
>+    int64_t clipf = (int64_t)(clip_y)*dy;
>     y += (clipf & 0xffff);
>     src += (clipf >> 16) * src_stride;
>     dst += clip_y * dst_stride;
>@@ -952,11 +922,11 @@ static void ScaleARGB(const uint8* src,
> }
> 
> LIBYUV_API
>-int ARGBScaleClip(const uint8* src_argb,
>+int ARGBScaleClip(const uint8_t* src_argb,
>                   int src_stride_argb,
>                   int src_width,
>                   int src_height,
>-                  uint8* dst_argb,
>+                  uint8_t* dst_argb,
>                   int dst_stride_argb,
>                   int dst_width,
>                   int dst_height,
>@@ -980,11 +950,11 @@ int ARGBScaleClip(const uint8* src_argb,
> 
> // Scale an ARGB image.
> LIBYUV_API
>-int ARGBScale(const uint8* src_argb,
>+int ARGBScale(const uint8_t* src_argb,
>               int src_stride_argb,
>               int src_width,
>               int src_height,
>-              uint8* dst_argb,
>+              uint8_t* dst_argb,
>               int dst_stride_argb,
>               int dst_width,
>               int dst_height,
>@@ -1001,18 +971,18 @@ int ARGBScale(const uint8* src_argb,
> 
> // Scale with YUV conversion to ARGB and clipping.
> LIBYUV_API
>-int YUVToARGBScaleClip(const uint8* src_y,
>+int YUVToARGBScaleClip(const uint8_t* src_y,
>                        int src_stride_y,
>-                       const uint8* src_u,
>+                       const uint8_t* src_u,
>                        int src_stride_u,
>-                       const uint8* src_v,
>+                       const uint8_t* src_v,
>                        int src_stride_v,
>-                       uint32 src_fourcc,
>+                       uint32_t src_fourcc,
>                        int src_width,
>                        int src_height,
>-                       uint8* dst_argb,
>+                       uint8_t* dst_argb,
>                        int dst_stride_argb,
>-                       uint32 dst_fourcc,
>+                       uint32_t dst_fourcc,
>                        int dst_width,
>                        int dst_height,
>                        int clip_x,
>@@ -1020,7 +990,7 @@ int YUVToARGBScaleClip(const uint8* src_y,
>                        int clip_width,
>                        int clip_height,
>                        enum FilterMode filtering) {
>-  uint8* argb_buffer = (uint8*)malloc(src_width * src_height * 4);
>+  uint8_t* argb_buffer = (uint8_t*)malloc(src_width * src_height * 4);
>   int r;
>   (void)src_fourcc;  // TODO(fbarchard): implement and/or assert.
>   (void)dst_fourcc;
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/scale_common.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/scale_common.cc
>index fefb027bf76..b28d7da41fc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/scale_common.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/scale_common.cc
>@@ -28,9 +28,9 @@ static __inline int Abs(int v) {
> }
> 
> // CPU agnostic row functions
>-void ScaleRowDown2_C(const uint8* src_ptr,
>+void ScaleRowDown2_C(const uint8_t* src_ptr,
>                      ptrdiff_t src_stride,
>-                     uint8* dst,
>+                     uint8_t* dst,
>                      int dst_width) {
>   int x;
>   (void)src_stride;
>@@ -45,9 +45,9 @@ void ScaleRowDown2_C(const uint8* src_ptr,
>   }
> }
> 
>-void ScaleRowDown2_16_C(const uint16* src_ptr,
>+void ScaleRowDown2_16_C(const uint16_t* src_ptr,
>                         ptrdiff_t src_stride,
>-                        uint16* dst,
>+                        uint16_t* dst,
>                         int dst_width) {
>   int x;
>   (void)src_stride;
>@@ -62,11 +62,11 @@ void ScaleRowDown2_16_C(const uint16* src_ptr,
>   }
> }
> 
>-void ScaleRowDown2Linear_C(const uint8* src_ptr,
>+void ScaleRowDown2Linear_C(const uint8_t* src_ptr,
>                            ptrdiff_t src_stride,
>-                           uint8* dst,
>+                           uint8_t* dst,
>                            int dst_width) {
>-  const uint8* s = src_ptr;
>+  const uint8_t* s = src_ptr;
>   int x;
>   (void)src_stride;
>   for (x = 0; x < dst_width - 1; x += 2) {
>@@ -80,11 +80,11 @@ void ScaleRowDown2Linear_C(const uint8* src_ptr,
>   }
> }
> 
>-void ScaleRowDown2Linear_16_C(const uint16* src_ptr,
>+void ScaleRowDown2Linear_16_C(const uint16_t* src_ptr,
>                               ptrdiff_t src_stride,
>-                              uint16* dst,
>+                              uint16_t* dst,
>                               int dst_width) {
>-  const uint16* s = src_ptr;
>+  const uint16_t* s = src_ptr;
>   int x;
>   (void)src_stride;
>   for (x = 0; x < dst_width - 1; x += 2) {
>@@ -98,12 +98,12 @@ void ScaleRowDown2Linear_16_C(const uint16* src_ptr,
>   }
> }
> 
>-void ScaleRowDown2Box_C(const uint8* src_ptr,
>+void ScaleRowDown2Box_C(const uint8_t* src_ptr,
>                         ptrdiff_t src_stride,
>-                        uint8* dst,
>+                        uint8_t* dst,
>                         int dst_width) {
>-  const uint8* s = src_ptr;
>-  const uint8* t = src_ptr + src_stride;
>+  const uint8_t* s = src_ptr;
>+  const uint8_t* t = src_ptr + src_stride;
>   int x;
>   for (x = 0; x < dst_width - 1; x += 2) {
>     dst[0] = (s[0] + s[1] + t[0] + t[1] + 2) >> 2;
>@@ -117,12 +117,12 @@ void ScaleRowDown2Box_C(const uint8* src_ptr,
>   }
> }
> 
>-void ScaleRowDown2Box_Odd_C(const uint8* src_ptr,
>+void ScaleRowDown2Box_Odd_C(const uint8_t* src_ptr,
>                             ptrdiff_t src_stride,
>-                            uint8* dst,
>+                            uint8_t* dst,
>                             int dst_width) {
>-  const uint8* s = src_ptr;
>-  const uint8* t = src_ptr + src_stride;
>+  const uint8_t* s = src_ptr;
>+  const uint8_t* t = src_ptr + src_stride;
>   int x;
>   dst_width -= 1;
>   for (x = 0; x < dst_width - 1; x += 2) {
>@@ -141,12 +141,12 @@ void ScaleRowDown2Box_Odd_C(const uint8* src_ptr,
>   dst[0] = (s[0] + t[0] + 1) >> 1;
> }
> 
>-void ScaleRowDown2Box_16_C(const uint16* src_ptr,
>+void ScaleRowDown2Box_16_C(const uint16_t* src_ptr,
>                            ptrdiff_t src_stride,
>-                           uint16* dst,
>+                           uint16_t* dst,
>                            int dst_width) {
>-  const uint16* s = src_ptr;
>-  const uint16* t = src_ptr + src_stride;
>+  const uint16_t* s = src_ptr;
>+  const uint16_t* t = src_ptr + src_stride;
>   int x;
>   for (x = 0; x < dst_width - 1; x += 2) {
>     dst[0] = (s[0] + s[1] + t[0] + t[1] + 2) >> 2;
>@@ -160,9 +160,9 @@ void ScaleRowDown2Box_16_C(const uint16* src_ptr,
>   }
> }
> 
>-void ScaleRowDown4_C(const uint8* src_ptr,
>+void ScaleRowDown4_C(const uint8_t* src_ptr,
>                      ptrdiff_t src_stride,
>-                     uint8* dst,
>+                     uint8_t* dst,
>                      int dst_width) {
>   int x;
>   (void)src_stride;
>@@ -177,9 +177,9 @@ void ScaleRowDown4_C(const uint8* src_ptr,
>   }
> }
> 
>-void ScaleRowDown4_16_C(const uint16* src_ptr,
>+void ScaleRowDown4_16_C(const uint16_t* src_ptr,
>                         ptrdiff_t src_stride,
>-                        uint16* dst,
>+                        uint16_t* dst,
>                         int dst_width) {
>   int x;
>   (void)src_stride;
>@@ -194,9 +194,9 @@ void ScaleRowDown4_16_C(const uint16* src_ptr,
>   }
> }
> 
>-void ScaleRowDown4Box_C(const uint8* src_ptr,
>+void ScaleRowDown4Box_C(const uint8_t* src_ptr,
>                         ptrdiff_t src_stride,
>-                        uint8* dst,
>+                        uint8_t* dst,
>                         int dst_width) {
>   intptr_t stride = src_stride;
>   int x;
>@@ -232,9 +232,9 @@ void ScaleRowDown4Box_C(const uint8* src_ptr,
>   }
> }
> 
>-void ScaleRowDown4Box_16_C(const uint16* src_ptr,
>+void ScaleRowDown4Box_16_C(const uint16_t* src_ptr,
>                            ptrdiff_t src_stride,
>-                           uint16* dst,
>+                           uint16_t* dst,
>                            int dst_width) {
>   intptr_t stride = src_stride;
>   int x;
>@@ -270,9 +270,9 @@ void ScaleRowDown4Box_16_C(const uint16* src_ptr,
>   }
> }
> 
>-void ScaleRowDown34_C(const uint8* src_ptr,
>+void ScaleRowDown34_C(const uint8_t* src_ptr,
>                       ptrdiff_t src_stride,
>-                      uint8* dst,
>+                      uint8_t* dst,
>                       int dst_width) {
>   int x;
>   (void)src_stride;
>@@ -286,9 +286,9 @@ void ScaleRowDown34_C(const uint8* src_ptr,
>   }
> }
> 
>-void ScaleRowDown34_16_C(const uint16* src_ptr,
>+void ScaleRowDown34_16_C(const uint16_t* src_ptr,
>                          ptrdiff_t src_stride,
>-                         uint16* dst,
>+                         uint16_t* dst,
>                          int dst_width) {
>   int x;
>   (void)src_stride;
>@@ -303,21 +303,21 @@ void ScaleRowDown34_16_C(const uint16* src_ptr,
> }
> 
> // Filter rows 0 and 1 together, 3 : 1
>-void ScaleRowDown34_0_Box_C(const uint8* src_ptr,
>+void ScaleRowDown34_0_Box_C(const uint8_t* src_ptr,
>                             ptrdiff_t src_stride,
>-                            uint8* d,
>+                            uint8_t* d,
>                             int dst_width) {
>-  const uint8* s = src_ptr;
>-  const uint8* t = src_ptr + src_stride;
>+  const uint8_t* s = src_ptr;
>+  const uint8_t* t = src_ptr + src_stride;
>   int x;
>   assert((dst_width % 3 == 0) && (dst_width > 0));
>   for (x = 0; x < dst_width; x += 3) {
>-    uint8 a0 = (s[0] * 3 + s[1] * 1 + 2) >> 2;
>-    uint8 a1 = (s[1] * 1 + s[2] * 1 + 1) >> 1;
>-    uint8 a2 = (s[2] * 1 + s[3] * 3 + 2) >> 2;
>-    uint8 b0 = (t[0] * 3 + t[1] * 1 + 2) >> 2;
>-    uint8 b1 = (t[1] * 1 + t[2] * 1 + 1) >> 1;
>-    uint8 b2 = (t[2] * 1 + t[3] * 3 + 2) >> 2;
>+    uint8_t a0 = (s[0] * 3 + s[1] * 1 + 2) >> 2;
>+    uint8_t a1 = (s[1] * 1 + s[2] * 1 + 1) >> 1;
>+    uint8_t a2 = (s[2] * 1 + s[3] * 3 + 2) >> 2;
>+    uint8_t b0 = (t[0] * 3 + t[1] * 1 + 2) >> 2;
>+    uint8_t b1 = (t[1] * 1 + t[2] * 1 + 1) >> 1;
>+    uint8_t b2 = (t[2] * 1 + t[3] * 3 + 2) >> 2;
>     d[0] = (a0 * 3 + b0 + 2) >> 2;
>     d[1] = (a1 * 3 + b1 + 2) >> 2;
>     d[2] = (a2 * 3 + b2 + 2) >> 2;
>@@ -327,21 +327,21 @@ void ScaleRowDown34_0_Box_C(const uint8* src_ptr,
>   }
> }
> 
>-void ScaleRowDown34_0_Box_16_C(const uint16* src_ptr,
>+void ScaleRowDown34_0_Box_16_C(const uint16_t* src_ptr,
>                                ptrdiff_t src_stride,
>-                               uint16* d,
>+                               uint16_t* d,
>                                int dst_width) {
>-  const uint16* s = src_ptr;
>-  const uint16* t = src_ptr + src_stride;
>+  const uint16_t* s = src_ptr;
>+  const uint16_t* t = src_ptr + src_stride;
>   int x;
>   assert((dst_width % 3 == 0) && (dst_width > 0));
>   for (x = 0; x < dst_width; x += 3) {
>-    uint16 a0 = (s[0] * 3 + s[1] * 1 + 2) >> 2;
>-    uint16 a1 = (s[1] * 1 + s[2] * 1 + 1) >> 1;
>-    uint16 a2 = (s[2] * 1 + s[3] * 3 + 2) >> 2;
>-    uint16 b0 = (t[0] * 3 + t[1] * 1 + 2) >> 2;
>-    uint16 b1 = (t[1] * 1 + t[2] * 1 + 1) >> 1;
>-    uint16 b2 = (t[2] * 1 + t[3] * 3 + 2) >> 2;
>+    uint16_t a0 = (s[0] * 3 + s[1] * 1 + 2) >> 2;
>+    uint16_t a1 = (s[1] * 1 + s[2] * 1 + 1) >> 1;
>+    uint16_t a2 = (s[2] * 1 + s[3] * 3 + 2) >> 2;
>+    uint16_t b0 = (t[0] * 3 + t[1] * 1 + 2) >> 2;
>+    uint16_t b1 = (t[1] * 1 + t[2] * 1 + 1) >> 1;
>+    uint16_t b2 = (t[2] * 1 + t[3] * 3 + 2) >> 2;
>     d[0] = (a0 * 3 + b0 + 2) >> 2;
>     d[1] = (a1 * 3 + b1 + 2) >> 2;
>     d[2] = (a2 * 3 + b2 + 2) >> 2;
>@@ -352,21 +352,21 @@ void ScaleRowDown34_0_Box_16_C(const uint16* src_ptr,
> }
> 
> // Filter rows 1 and 2 together, 1 : 1
>-void ScaleRowDown34_1_Box_C(const uint8* src_ptr,
>+void ScaleRowDown34_1_Box_C(const uint8_t* src_ptr,
>                             ptrdiff_t src_stride,
>-                            uint8* d,
>+                            uint8_t* d,
>                             int dst_width) {
>-  const uint8* s = src_ptr;
>-  const uint8* t = src_ptr + src_stride;
>+  const uint8_t* s = src_ptr;
>+  const uint8_t* t = src_ptr + src_stride;
>   int x;
>   assert((dst_width % 3 == 0) && (dst_width > 0));
>   for (x = 0; x < dst_width; x += 3) {
>-    uint8 a0 = (s[0] * 3 + s[1] * 1 + 2) >> 2;
>-    uint8 a1 = (s[1] * 1 + s[2] * 1 + 1) >> 1;
>-    uint8 a2 = (s[2] * 1 + s[3] * 3 + 2) >> 2;
>-    uint8 b0 = (t[0] * 3 + t[1] * 1 + 2) >> 2;
>-    uint8 b1 = (t[1] * 1 + t[2] * 1 + 1) >> 1;
>-    uint8 b2 = (t[2] * 1 + t[3] * 3 + 2) >> 2;
>+    uint8_t a0 = (s[0] * 3 + s[1] * 1 + 2) >> 2;
>+    uint8_t a1 = (s[1] * 1 + s[2] * 1 + 1) >> 1;
>+    uint8_t a2 = (s[2] * 1 + s[3] * 3 + 2) >> 2;
>+    uint8_t b0 = (t[0] * 3 + t[1] * 1 + 2) >> 2;
>+    uint8_t b1 = (t[1] * 1 + t[2] * 1 + 1) >> 1;
>+    uint8_t b2 = (t[2] * 1 + t[3] * 3 + 2) >> 2;
>     d[0] = (a0 + b0 + 1) >> 1;
>     d[1] = (a1 + b1 + 1) >> 1;
>     d[2] = (a2 + b2 + 1) >> 1;
>@@ -376,21 +376,21 @@ void ScaleRowDown34_1_Box_C(const uint8* src_ptr,
>   }
> }
> 
>-void ScaleRowDown34_1_Box_16_C(const uint16* src_ptr,
>+void ScaleRowDown34_1_Box_16_C(const uint16_t* src_ptr,
>                                ptrdiff_t src_stride,
>-                               uint16* d,
>+                               uint16_t* d,
>                                int dst_width) {
>-  const uint16* s = src_ptr;
>-  const uint16* t = src_ptr + src_stride;
>+  const uint16_t* s = src_ptr;
>+  const uint16_t* t = src_ptr + src_stride;
>   int x;
>   assert((dst_width % 3 == 0) && (dst_width > 0));
>   for (x = 0; x < dst_width; x += 3) {
>-    uint16 a0 = (s[0] * 3 + s[1] * 1 + 2) >> 2;
>-    uint16 a1 = (s[1] * 1 + s[2] * 1 + 1) >> 1;
>-    uint16 a2 = (s[2] * 1 + s[3] * 3 + 2) >> 2;
>-    uint16 b0 = (t[0] * 3 + t[1] * 1 + 2) >> 2;
>-    uint16 b1 = (t[1] * 1 + t[2] * 1 + 1) >> 1;
>-    uint16 b2 = (t[2] * 1 + t[3] * 3 + 2) >> 2;
>+    uint16_t a0 = (s[0] * 3 + s[1] * 1 + 2) >> 2;
>+    uint16_t a1 = (s[1] * 1 + s[2] * 1 + 1) >> 1;
>+    uint16_t a2 = (s[2] * 1 + s[3] * 3 + 2) >> 2;
>+    uint16_t b0 = (t[0] * 3 + t[1] * 1 + 2) >> 2;
>+    uint16_t b1 = (t[1] * 1 + t[2] * 1 + 1) >> 1;
>+    uint16_t b2 = (t[2] * 1 + t[3] * 3 + 2) >> 2;
>     d[0] = (a0 + b0 + 1) >> 1;
>     d[1] = (a1 + b1 + 1) >> 1;
>     d[2] = (a2 + b2 + 1) >> 1;
>@@ -401,8 +401,8 @@ void ScaleRowDown34_1_Box_16_C(const uint16* src_ptr,
> }
> 
> // Scales a single row of pixels using point sampling.
>-void ScaleCols_C(uint8* dst_ptr,
>-                 const uint8* src_ptr,
>+void ScaleCols_C(uint8_t* dst_ptr,
>+                 const uint8_t* src_ptr,
>                  int dst_width,
>                  int x,
>                  int dx) {
>@@ -419,8 +419,8 @@ void ScaleCols_C(uint8* dst_ptr,
>   }
> }
> 
>-void ScaleCols_16_C(uint16* dst_ptr,
>-                    const uint16* src_ptr,
>+void ScaleCols_16_C(uint16_t* dst_ptr,
>+                    const uint16_t* src_ptr,
>                     int dst_width,
>                     int x,
>                     int dx) {
>@@ -438,8 +438,8 @@ void ScaleCols_16_C(uint16* dst_ptr,
> }
> 
> // Scales a single row of pixels up by 2x using point sampling.
>-void ScaleColsUp2_C(uint8* dst_ptr,
>-                    const uint8* src_ptr,
>+void ScaleColsUp2_C(uint8_t* dst_ptr,
>+                    const uint8_t* src_ptr,
>                     int dst_width,
>                     int x,
>                     int dx) {
>@@ -456,8 +456,8 @@ void ScaleColsUp2_C(uint8* dst_ptr,
>   }
> }
> 
>-void ScaleColsUp2_16_C(uint16* dst_ptr,
>-                       const uint16* src_ptr,
>+void ScaleColsUp2_16_C(uint16_t* dst_ptr,
>+                       const uint16_t* src_ptr,
>                        int dst_width,
>                        int x,
>                        int dx) {
>@@ -477,15 +477,15 @@ void ScaleColsUp2_16_C(uint16* dst_ptr,
> // (1-f)a + fb can be replaced with a + f(b-a)
> #if defined(__arm__) || defined(__aarch64__)
> #define BLENDER(a, b, f) \
>-  (uint8)((int)(a) + ((((int)((f)) * ((int)(b) - (int)(a))) + 0x8000) >> 16))
>+  (uint8_t)((int)(a) + ((((int)((f)) * ((int)(b) - (int)(a))) + 0x8000) >> 16))
> #else
> // Intel uses 7 bit math with rounding.
> #define BLENDER(a, b, f) \
>-  (uint8)((int)(a) + (((int)((f) >> 9) * ((int)(b) - (int)(a)) + 0x40) >> 7))
>+  (uint8_t)((int)(a) + (((int)((f) >> 9) * ((int)(b) - (int)(a)) + 0x40) >> 7))
> #endif
> 
>-void ScaleFilterCols_C(uint8* dst_ptr,
>-                       const uint8* src_ptr,
>+void ScaleFilterCols_C(uint8_t* dst_ptr,
>+                       const uint8_t* src_ptr,
>                        int dst_width,
>                        int x,
>                        int dx) {
>@@ -511,15 +511,15 @@ void ScaleFilterCols_C(uint8* dst_ptr,
>   }
> }
> 
>-void ScaleFilterCols64_C(uint8* dst_ptr,
>-                         const uint8* src_ptr,
>+void ScaleFilterCols64_C(uint8_t* dst_ptr,
>+                         const uint8_t* src_ptr,
>                          int dst_width,
>                          int x32,
>                          int dx) {
>-  int64 x = (int64)(x32);
>+  int64_t x = (int64_t)(x32);
>   int j;
>   for (j = 0; j < dst_width - 1; j += 2) {
>-    int64 xi = x >> 16;
>+    int64_t xi = x >> 16;
>     int a = src_ptr[xi];
>     int b = src_ptr[xi + 1];
>     dst_ptr[0] = BLENDER(a, b, x & 0xffff);
>@@ -532,7 +532,7 @@ void ScaleFilterCols64_C(uint8* dst_ptr,
>     dst_ptr += 2;
>   }
>   if (dst_width & 1) {
>-    int64 xi = x >> 16;
>+    int64_t xi = x >> 16;
>     int a = src_ptr[xi];
>     int b = src_ptr[xi + 1];
>     dst_ptr[0] = BLENDER(a, b, x & 0xffff);
>@@ -540,12 +540,12 @@ void ScaleFilterCols64_C(uint8* dst_ptr,
> }
> #undef BLENDER
> 
>-// Same as 8 bit arm blender but return is cast to uint16
>+// Same as 8 bit arm blender but return is cast to uint16_t
> #define BLENDER(a, b, f) \
>-  (uint16)((int)(a) + ((((int)((f)) * ((int)(b) - (int)(a))) + 0x8000) >> 16))
>+  (uint16_t)((int)(a) + ((((int)((f)) * ((int)(b) - (int)(a))) + 0x8000) >> 16))
> 
>-void ScaleFilterCols_16_C(uint16* dst_ptr,
>-                          const uint16* src_ptr,
>+void ScaleFilterCols_16_C(uint16_t* dst_ptr,
>+                          const uint16_t* src_ptr,
>                           int dst_width,
>                           int x,
>                           int dx) {
>@@ -571,15 +571,15 @@ void ScaleFilterCols_16_C(uint16* dst_ptr,
>   }
> }
> 
>-void ScaleFilterCols64_16_C(uint16* dst_ptr,
>-                            const uint16* src_ptr,
>+void ScaleFilterCols64_16_C(uint16_t* dst_ptr,
>+                            const uint16_t* src_ptr,
>                             int dst_width,
>                             int x32,
>                             int dx) {
>-  int64 x = (int64)(x32);
>+  int64_t x = (int64_t)(x32);
>   int j;
>   for (j = 0; j < dst_width - 1; j += 2) {
>-    int64 xi = x >> 16;
>+    int64_t xi = x >> 16;
>     int a = src_ptr[xi];
>     int b = src_ptr[xi + 1];
>     dst_ptr[0] = BLENDER(a, b, x & 0xffff);
>@@ -592,7 +592,7 @@ void ScaleFilterCols64_16_C(uint16* dst_ptr,
>     dst_ptr += 2;
>   }
>   if (dst_width & 1) {
>-    int64 xi = x >> 16;
>+    int64_t xi = x >> 16;
>     int a = src_ptr[xi];
>     int b = src_ptr[xi + 1];
>     dst_ptr[0] = BLENDER(a, b, x & 0xffff);
>@@ -600,9 +600,9 @@ void ScaleFilterCols64_16_C(uint16* dst_ptr,
> }
> #undef BLENDER
> 
>-void ScaleRowDown38_C(const uint8* src_ptr,
>+void ScaleRowDown38_C(const uint8_t* src_ptr,
>                       ptrdiff_t src_stride,
>-                      uint8* dst,
>+                      uint8_t* dst,
>                       int dst_width) {
>   int x;
>   (void)src_stride;
>@@ -616,9 +616,9 @@ void ScaleRowDown38_C(const uint8* src_ptr,
>   }
> }
> 
>-void ScaleRowDown38_16_C(const uint16* src_ptr,
>+void ScaleRowDown38_16_C(const uint16_t* src_ptr,
>                          ptrdiff_t src_stride,
>-                         uint16* dst,
>+                         uint16_t* dst,
>                          int dst_width) {
>   int x;
>   (void)src_stride;
>@@ -633,9 +633,9 @@ void ScaleRowDown38_16_C(const uint16* src_ptr,
> }
> 
> // 8x3 -> 3x1
>-void ScaleRowDown38_3_Box_C(const uint8* src_ptr,
>+void ScaleRowDown38_3_Box_C(const uint8_t* src_ptr,
>                             ptrdiff_t src_stride,
>-                            uint8* dst_ptr,
>+                            uint8_t* dst_ptr,
>                             int dst_width) {
>   intptr_t stride = src_stride;
>   int i;
>@@ -663,9 +663,9 @@ void ScaleRowDown38_3_Box_C(const uint8* src_ptr,
>   }
> }
> 
>-void ScaleRowDown38_3_Box_16_C(const uint16* src_ptr,
>+void ScaleRowDown38_3_Box_16_C(const uint16_t* src_ptr,
>                                ptrdiff_t src_stride,
>-                               uint16* dst_ptr,
>+                               uint16_t* dst_ptr,
>                                int dst_width) {
>   intptr_t stride = src_stride;
>   int i;
>@@ -694,9 +694,9 @@ void ScaleRowDown38_3_Box_16_C(const uint16* src_ptr,
> }
> 
> // 8x2 -> 3x1
>-void ScaleRowDown38_2_Box_C(const uint8* src_ptr,
>+void ScaleRowDown38_2_Box_C(const uint8_t* src_ptr,
>                             ptrdiff_t src_stride,
>-                            uint8* dst_ptr,
>+                            uint8_t* dst_ptr,
>                             int dst_width) {
>   intptr_t stride = src_stride;
>   int i;
>@@ -719,9 +719,9 @@ void ScaleRowDown38_2_Box_C(const uint8* src_ptr,
>   }
> }
> 
>-void ScaleRowDown38_2_Box_16_C(const uint16* src_ptr,
>+void ScaleRowDown38_2_Box_16_C(const uint16_t* src_ptr,
>                                ptrdiff_t src_stride,
>-                               uint16* dst_ptr,
>+                               uint16_t* dst_ptr,
>                                int dst_width) {
>   intptr_t stride = src_stride;
>   int i;
>@@ -744,7 +744,7 @@ void ScaleRowDown38_2_Box_16_C(const uint16* src_ptr,
>   }
> }
> 
>-void ScaleAddRow_C(const uint8* src_ptr, uint16* dst_ptr, int src_width) {
>+void ScaleAddRow_C(const uint8_t* src_ptr, uint16_t* dst_ptr, int src_width) {
>   int x;
>   assert(src_width > 0);
>   for (x = 0; x < src_width - 1; x += 2) {
>@@ -758,7 +758,9 @@ void ScaleAddRow_C(const uint8* src_ptr, uint16* dst_ptr, int src_width) {
>   }
> }
> 
>-void ScaleAddRow_16_C(const uint16* src_ptr, uint32* dst_ptr, int src_width) {
>+void ScaleAddRow_16_C(const uint16_t* src_ptr,
>+                      uint32_t* dst_ptr,
>+                      int src_width) {
>   int x;
>   assert(src_width > 0);
>   for (x = 0; x < src_width - 1; x += 2) {
>@@ -772,12 +774,12 @@ void ScaleAddRow_16_C(const uint16* src_ptr, uint32* dst_ptr, int src_width) {
>   }
> }
> 
>-void ScaleARGBRowDown2_C(const uint8* src_argb,
>+void ScaleARGBRowDown2_C(const uint8_t* src_argb,
>                          ptrdiff_t src_stride,
>-                         uint8* dst_argb,
>+                         uint8_t* dst_argb,
>                          int dst_width) {
>-  const uint32* src = (const uint32*)(src_argb);
>-  uint32* dst = (uint32*)(dst_argb);
>+  const uint32_t* src = (const uint32_t*)(src_argb);
>+  uint32_t* dst = (uint32_t*)(dst_argb);
>   int x;
>   (void)src_stride;
>   for (x = 0; x < dst_width - 1; x += 2) {
>@@ -791,9 +793,9 @@ void ScaleARGBRowDown2_C(const uint8* src_argb,
>   }
> }
> 
>-void ScaleARGBRowDown2Linear_C(const uint8* src_argb,
>+void ScaleARGBRowDown2Linear_C(const uint8_t* src_argb,
>                                ptrdiff_t src_stride,
>-                               uint8* dst_argb,
>+                               uint8_t* dst_argb,
>                                int dst_width) {
>   int x;
>   (void)src_stride;
>@@ -807,9 +809,9 @@ void ScaleARGBRowDown2Linear_C(const uint8* src_argb,
>   }
> }
> 
>-void ScaleARGBRowDown2Box_C(const uint8* src_argb,
>+void ScaleARGBRowDown2Box_C(const uint8_t* src_argb,
>                             ptrdiff_t src_stride,
>-                            uint8* dst_argb,
>+                            uint8_t* dst_argb,
>                             int dst_width) {
>   int x;
>   for (x = 0; x < dst_width; ++x) {
>@@ -830,13 +832,13 @@ void ScaleARGBRowDown2Box_C(const uint8* src_argb,
>   }
> }
> 
>-void ScaleARGBRowDownEven_C(const uint8* src_argb,
>+void ScaleARGBRowDownEven_C(const uint8_t* src_argb,
>                             ptrdiff_t src_stride,
>                             int src_stepx,
>-                            uint8* dst_argb,
>+                            uint8_t* dst_argb,
>                             int dst_width) {
>-  const uint32* src = (const uint32*)(src_argb);
>-  uint32* dst = (uint32*)(dst_argb);
>+  const uint32_t* src = (const uint32_t*)(src_argb);
>+  uint32_t* dst = (uint32_t*)(dst_argb);
>   (void)src_stride;
>   int x;
>   for (x = 0; x < dst_width - 1; x += 2) {
>@@ -850,10 +852,10 @@ void ScaleARGBRowDownEven_C(const uint8* src_argb,
>   }
> }
> 
>-void ScaleARGBRowDownEvenBox_C(const uint8* src_argb,
>+void ScaleARGBRowDownEvenBox_C(const uint8_t* src_argb,
>                                ptrdiff_t src_stride,
>                                int src_stepx,
>-                               uint8* dst_argb,
>+                               uint8_t* dst_argb,
>                                int dst_width) {
>   int x;
>   for (x = 0; x < dst_width; ++x) {
>@@ -875,13 +877,13 @@ void ScaleARGBRowDownEvenBox_C(const uint8* src_argb,
> }
> 
> // Scales a single row of pixels using point sampling.
>-void ScaleARGBCols_C(uint8* dst_argb,
>-                     const uint8* src_argb,
>+void ScaleARGBCols_C(uint8_t* dst_argb,
>+                     const uint8_t* src_argb,
>                      int dst_width,
>                      int x,
>                      int dx) {
>-  const uint32* src = (const uint32*)(src_argb);
>-  uint32* dst = (uint32*)(dst_argb);
>+  const uint32_t* src = (const uint32_t*)(src_argb);
>+  uint32_t* dst = (uint32_t*)(dst_argb);
>   int j;
>   for (j = 0; j < dst_width - 1; j += 2) {
>     dst[0] = src[x >> 16];
>@@ -895,14 +897,14 @@ void ScaleARGBCols_C(uint8* dst_argb,
>   }
> }
> 
>-void ScaleARGBCols64_C(uint8* dst_argb,
>-                       const uint8* src_argb,
>+void ScaleARGBCols64_C(uint8_t* dst_argb,
>+                       const uint8_t* src_argb,
>                        int dst_width,
>                        int x32,
>                        int dx) {
>-  int64 x = (int64)(x32);
>-  const uint32* src = (const uint32*)(src_argb);
>-  uint32* dst = (uint32*)(dst_argb);
>+  int64_t x = (int64_t)(x32);
>+  const uint32_t* src = (const uint32_t*)(src_argb);
>+  uint32_t* dst = (uint32_t*)(dst_argb);
>   int j;
>   for (j = 0; j < dst_width - 1; j += 2) {
>     dst[0] = src[x >> 16];
>@@ -917,13 +919,13 @@ void ScaleARGBCols64_C(uint8* dst_argb,
> }
> 
> // Scales a single row of pixels up by 2x using point sampling.
>-void ScaleARGBColsUp2_C(uint8* dst_argb,
>-                        const uint8* src_argb,
>+void ScaleARGBColsUp2_C(uint8_t* dst_argb,
>+                        const uint8_t* src_argb,
>                         int dst_width,
>                         int x,
>                         int dx) {
>-  const uint32* src = (const uint32*)(src_argb);
>-  uint32* dst = (uint32*)(dst_argb);
>+  const uint32_t* src = (const uint32_t*)(src_argb);
>+  uint32_t* dst = (uint32_t*)(dst_argb);
>   int j;
>   (void)x;
>   (void)dx;
>@@ -941,24 +943,24 @@ void ScaleARGBColsUp2_C(uint8* dst_argb,
> // Mimics SSSE3 blender
> #define BLENDER1(a, b, f) ((a) * (0x7f ^ f) + (b)*f) >> 7
> #define BLENDERC(a, b, f, s) \
>-  (uint32)(BLENDER1(((a) >> s) & 255, ((b) >> s) & 255, f) << s)
>+  (uint32_t)(BLENDER1(((a) >> s) & 255, ((b) >> s) & 255, f) << s)
> #define BLENDER(a, b, f)                                                 \
>   BLENDERC(a, b, f, 24) | BLENDERC(a, b, f, 16) | BLENDERC(a, b, f, 8) | \
>       BLENDERC(a, b, f, 0)
> 
>-void ScaleARGBFilterCols_C(uint8* dst_argb,
>-                           const uint8* src_argb,
>+void ScaleARGBFilterCols_C(uint8_t* dst_argb,
>+                           const uint8_t* src_argb,
>                            int dst_width,
>                            int x,
>                            int dx) {
>-  const uint32* src = (const uint32*)(src_argb);
>-  uint32* dst = (uint32*)(dst_argb);
>+  const uint32_t* src = (const uint32_t*)(src_argb);
>+  uint32_t* dst = (uint32_t*)(dst_argb);
>   int j;
>   for (j = 0; j < dst_width - 1; j += 2) {
>     int xi = x >> 16;
>     int xf = (x >> 9) & 0x7f;
>-    uint32 a = src[xi];
>-    uint32 b = src[xi + 1];
>+    uint32_t a = src[xi];
>+    uint32_t b = src[xi + 1];
>     dst[0] = BLENDER(a, b, xf);
>     x += dx;
>     xi = x >> 16;
>@@ -972,26 +974,26 @@ void ScaleARGBFilterCols_C(uint8* dst_argb,
>   if (dst_width & 1) {
>     int xi = x >> 16;
>     int xf = (x >> 9) & 0x7f;
>-    uint32 a = src[xi];
>-    uint32 b = src[xi + 1];
>+    uint32_t a = src[xi];
>+    uint32_t b = src[xi + 1];
>     dst[0] = BLENDER(a, b, xf);
>   }
> }
> 
>-void ScaleARGBFilterCols64_C(uint8* dst_argb,
>-                             const uint8* src_argb,
>+void ScaleARGBFilterCols64_C(uint8_t* dst_argb,
>+                             const uint8_t* src_argb,
>                              int dst_width,
>                              int x32,
>                              int dx) {
>-  int64 x = (int64)(x32);
>-  const uint32* src = (const uint32*)(src_argb);
>-  uint32* dst = (uint32*)(dst_argb);
>+  int64_t x = (int64_t)(x32);
>+  const uint32_t* src = (const uint32_t*)(src_argb);
>+  uint32_t* dst = (uint32_t*)(dst_argb);
>   int j;
>   for (j = 0; j < dst_width - 1; j += 2) {
>-    int64 xi = x >> 16;
>+    int64_t xi = x >> 16;
>     int xf = (x >> 9) & 0x7f;
>-    uint32 a = src[xi];
>-    uint32 b = src[xi + 1];
>+    uint32_t a = src[xi];
>+    uint32_t b = src[xi + 1];
>     dst[0] = BLENDER(a, b, xf);
>     x += dx;
>     xi = x >> 16;
>@@ -1003,10 +1005,10 @@ void ScaleARGBFilterCols64_C(uint8* dst_argb,
>     dst += 2;
>   }
>   if (dst_width & 1) {
>-    int64 xi = x >> 16;
>+    int64_t xi = x >> 16;
>     int xf = (x >> 9) & 0x7f;
>-    uint32 a = src[xi];
>-    uint32 b = src[xi + 1];
>+    uint32_t a = src[xi];
>+    uint32_t b = src[xi + 1];
>     dst[0] = BLENDER(a, b, xf);
>   }
> }
>@@ -1020,8 +1022,8 @@ void ScalePlaneVertical(int src_height,
>                         int dst_height,
>                         int src_stride,
>                         int dst_stride,
>-                        const uint8* src_argb,
>-                        uint8* dst_argb,
>+                        const uint8_t* src_argb,
>+                        uint8_t* dst_argb,
>                         int x,
>                         int y,
>                         int dy,
>@@ -1029,7 +1031,7 @@ void ScalePlaneVertical(int src_height,
>                         enum FilterMode filtering) {
>   // TODO(fbarchard): Allow higher bpp.
>   int dst_width_bytes = dst_width * bpp;
>-  void (*InterpolateRow)(uint8 * dst_argb, const uint8* src_argb,
>+  void (*InterpolateRow)(uint8_t * dst_argb, const uint8_t* src_argb,
>                          ptrdiff_t src_stride, int dst_width,
>                          int source_y_fraction) = InterpolateRow_C;
>   const int max_y = (src_height > 1) ? ((src_height - 1) << 16) - 1 : 0;
>@@ -1063,16 +1065,6 @@ void ScalePlaneVertical(int src_height,
>     }
>   }
> #endif
>-#if defined(HAS_INTERPOLATEROW_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2) && IS_ALIGNED(src_argb, 4) &&
>-      IS_ALIGNED(src_stride, 4) && IS_ALIGNED(dst_argb, 4) &&
>-      IS_ALIGNED(dst_stride, 4)) {
>-    InterpolateRow = InterpolateRow_Any_DSPR2;
>-    if (IS_ALIGNED(dst_width_bytes, 4)) {
>-      InterpolateRow = InterpolateRow_DSPR2;
>-    }
>-  }
>-#endif
> #if defined(HAS_INTERPOLATEROW_MSA)
>   if (TestCpuFlag(kCpuHasMSA)) {
>     InterpolateRow = InterpolateRow_Any_MSA;
>@@ -1100,8 +1092,8 @@ void ScalePlaneVertical_16(int src_height,
>                            int dst_height,
>                            int src_stride,
>                            int dst_stride,
>-                           const uint16* src_argb,
>-                           uint16* dst_argb,
>+                           const uint16_t* src_argb,
>+                           uint16_t* dst_argb,
>                            int x,
>                            int y,
>                            int dy,
>@@ -1109,7 +1101,7 @@ void ScalePlaneVertical_16(int src_height,
>                            enum FilterMode filtering) {
>   // TODO(fbarchard): Allow higher wpp.
>   int dst_width_words = dst_width * wpp;
>-  void (*InterpolateRow)(uint16 * dst_argb, const uint16* src_argb,
>+  void (*InterpolateRow)(uint16_t * dst_argb, const uint16_t* src_argb,
>                          ptrdiff_t src_stride, int dst_width,
>                          int source_y_fraction) = InterpolateRow_16_C;
>   const int max_y = (src_height > 1) ? ((src_height - 1) << 16) - 1 : 0;
>@@ -1150,16 +1142,6 @@ void ScalePlaneVertical_16(int src_height,
>       InterpolateRow = InterpolateRow_16_NEON;
>     }
>   }
>-#endif
>-#if defined(HAS_INTERPOLATEROW_16_DSPR2)
>-  if (TestCpuFlag(kCpuHasDSPR2) && IS_ALIGNED(src_argb, 4) &&
>-      IS_ALIGNED(src_stride, 4) && IS_ALIGNED(dst_argb, 4) &&
>-      IS_ALIGNED(dst_stride, 4)) {
>-    InterpolateRow = InterpolateRow_Any_16_DSPR2;
>-    if (IS_ALIGNED(dst_width_bytes, 4)) {
>-      InterpolateRow = InterpolateRow_16_DSPR2;
>-    }
>-  }
> #endif
>   for (j = 0; j < dst_height; ++j) {
>     int yi;
>@@ -1222,12 +1204,12 @@ enum FilterMode ScaleFilterReduce(int src_width,
> 
> // Divide num by div and return as 16.16 fixed point result.
> int FixedDiv_C(int num, int div) {
>-  return (int)(((int64)(num) << 16) / div);
>+  return (int)(((int64_t)(num) << 16) / div);
> }
> 
> // Divide num by div and return as 16.16 fixed point result.
> int FixedDiv1_C(int num, int div) {
>-  return (int)((((int64)(num) << 16) - 0x00010001) / (div - 1));
>+  return (int)((((int64_t)(num) << 16) - 0x00010001) / (div - 1));
> }
> 
> #define CENTERSTART(dx, s) (dx < 0) ? -((-dx >> 1) + s) : ((dx >> 1) + s)
>@@ -1308,18 +1290,18 @@ void ScaleSlope(int src_width,
> 
> // Read 8x2 upsample with filtering and write 16x1.
> // actually reads an extra pixel, so 9x2.
>-void ScaleRowUp2_16_C(const uint16* src_ptr,
>+void ScaleRowUp2_16_C(const uint16_t* src_ptr,
>                       ptrdiff_t src_stride,
>-                      uint16* dst,
>+                      uint16_t* dst,
>                       int dst_width) {
>-  const uint16* src2 = src_ptr + src_stride;
>+  const uint16_t* src2 = src_ptr + src_stride;
> 
>   int x;
>   for (x = 0; x < dst_width - 1; x += 2) {
>-    uint16 p0 = src_ptr[0];
>-    uint16 p1 = src_ptr[1];
>-    uint16 p2 = src2[0];
>-    uint16 p3 = src2[1];
>+    uint16_t p0 = src_ptr[0];
>+    uint16_t p1 = src_ptr[1];
>+    uint16_t p2 = src2[0];
>+    uint16_t p3 = src2[1];
>     dst[0] = (p0 * 9 + p1 * 3 + p2 * 3 + p3 + 8) >> 4;
>     dst[1] = (p0 * 3 + p1 * 9 + p2 + p3 * 3 + 8) >> 4;
>     ++src_ptr;
>@@ -1327,10 +1309,10 @@ void ScaleRowUp2_16_C(const uint16* src_ptr,
>     dst += 2;
>   }
>   if (dst_width & 1) {
>-    uint16 p0 = src_ptr[0];
>-    uint16 p1 = src_ptr[1];
>-    uint16 p2 = src2[0];
>-    uint16 p3 = src2[1];
>+    uint16_t p0 = src_ptr[0];
>+    uint16_t p1 = src_ptr[1];
>+    uint16_t p2 = src2[0];
>+    uint16_t p3 = src2[1];
>     dst[0] = (p0 * 9 + p1 * 3 + p2 * 3 + p3 + 8) >> 4;
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/scale_dspr2.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/scale_dspr2.cc
>deleted file mode 100644
>index ddedcbf46c2..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/scale_dspr2.cc
>+++ /dev/null
>@@ -1,668 +0,0 @@
>-/*
>- *  Copyright 2012 The LibYuv Project Authors. All rights reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS. All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "libyuv/basic_types.h"
>-#include "libyuv/row.h"
>-
>-#ifdef __cplusplus
>-namespace libyuv {
>-extern "C" {
>-#endif
>-
>-// This module is for GCC MIPS DSPR2
>-#if !defined(LIBYUV_DISABLE_DSPR2) && defined(__mips_dsp) && \
>-    (__mips_dsp_rev >= 2) && (_MIPS_SIM == _MIPS_SIM_ABI32)
>-
>-void ScaleRowDown2_DSPR2(const uint8* src_ptr,
>-                         ptrdiff_t src_stride,
>-                         uint8* dst,
>-                         int dst_width) {
>-  __asm__ __volatile__(
>-      ".set push                                     \n"
>-      ".set noreorder                                \n"
>-
>-      "srl            $t9, %[dst_width], 4           \n"  // iterations -> by 16
>-      "beqz           $t9, 2f                        \n"
>-      " nop                                          \n"
>-
>-      "1:                                            \n"
>-      "lw             $t0, 0(%[src_ptr])             \n"  // |3|2|1|0|
>-      "lw             $t1, 4(%[src_ptr])             \n"  // |7|6|5|4|
>-      "lw             $t2, 8(%[src_ptr])             \n"  // |11|10|9|8|
>-      "lw             $t3, 12(%[src_ptr])            \n"  // |15|14|13|12|
>-      "lw             $t4, 16(%[src_ptr])            \n"  // |19|18|17|16|
>-      "lw             $t5, 20(%[src_ptr])            \n"  // |23|22|21|20|
>-      "lw             $t6, 24(%[src_ptr])            \n"  // |27|26|25|24|
>-      "lw             $t7, 28(%[src_ptr])            \n"  // |31|30|29|28|
>-      // TODO(fbarchard): Use odd pixels instead of even.
>-      "precrq.qb.ph   $t8, $t1, $t0                  \n"  // |7|5|3|1|
>-      "precrq.qb.ph   $t0, $t3, $t2                  \n"  // |15|13|11|9|
>-      "precrq.qb.ph   $t1, $t5, $t4                  \n"  // |23|21|19|17|
>-      "precrq.qb.ph   $t2, $t7, $t6                  \n"  // |31|29|27|25|
>-      "addiu          %[src_ptr], %[src_ptr], 32     \n"
>-      "addiu          $t9, $t9, -1                   \n"
>-      "sw             $t8, 0(%[dst])                 \n"
>-      "sw             $t0, 4(%[dst])                 \n"
>-      "sw             $t1, 8(%[dst])                 \n"
>-      "sw             $t2, 12(%[dst])                \n"
>-      "bgtz           $t9, 1b                        \n"
>-      " addiu         %[dst], %[dst], 16             \n"
>-
>-      "2:                                            \n"
>-      "andi           $t9, %[dst_width], 0xf         \n"  // residue
>-      "beqz           $t9, 3f                        \n"
>-      " nop                                          \n"
>-
>-      "21:                                           \n"
>-      "lbu            $t0, 1(%[src_ptr])             \n"
>-      "addiu          %[src_ptr], %[src_ptr], 2      \n"
>-      "addiu          $t9, $t9, -1                   \n"
>-      "sb             $t0, 0(%[dst])                 \n"
>-      "bgtz           $t9, 21b                       \n"
>-      " addiu         %[dst], %[dst], 1              \n"
>-
>-      "3:                                            \n"
>-      ".set pop                                      \n"
>-      : [src_ptr] "+r"(src_ptr), [dst] "+r"(dst)
>-      : [dst_width] "r"(dst_width)
>-      : "t0", "t1", "t2", "t3", "t4", "t5", "t6", "t7", "t8", "t9");
>-}
>-
>-void ScaleRowDown2Box_DSPR2(const uint8* src_ptr,
>-                            ptrdiff_t src_stride,
>-                            uint8* dst,
>-                            int dst_width) {
>-  const uint8* t = src_ptr + src_stride;
>-
>-  __asm__ __volatile__(
>-      ".set push                                    \n"
>-      ".set noreorder                               \n"
>-
>-      "srl            $t9, %[dst_width], 3          \n"  // iterations -> step 8
>-      "bltz           $t9, 2f                       \n"
>-      " nop                                         \n"
>-
>-      "1:                                           \n"
>-      "lw             $t0, 0(%[src_ptr])            \n"  // |3|2|1|0|
>-      "lw             $t1, 4(%[src_ptr])            \n"  // |7|6|5|4|
>-      "lw             $t2, 8(%[src_ptr])            \n"  // |11|10|9|8|
>-      "lw             $t3, 12(%[src_ptr])           \n"  // |15|14|13|12|
>-      "lw             $t4, 0(%[t])                  \n"  // |19|18|17|16|
>-      "lw             $t5, 4(%[t])                  \n"  // |23|22|21|20|
>-      "lw             $t6, 8(%[t])                  \n"  // |27|26|25|24|
>-      "lw             $t7, 12(%[t])                 \n"  // |31|30|29|28|
>-      "addiu          $t9, $t9, -1                  \n"
>-      "srl            $t8, $t0, 16                  \n"  // |X|X|3|2|
>-      "ins            $t0, $t4, 16, 16              \n"  // |17|16|1|0|
>-      "ins            $t4, $t8, 0, 16               \n"  // |19|18|3|2|
>-      "raddu.w.qb     $t0, $t0                      \n"  // |17+16+1+0|
>-      "raddu.w.qb     $t4, $t4                      \n"  // |19+18+3+2|
>-      "shra_r.w       $t0, $t0, 2                   \n"  // |t0+2|>>2
>-      "shra_r.w       $t4, $t4, 2                   \n"  // |t4+2|>>2
>-      "srl            $t8, $t1, 16                  \n"  // |X|X|7|6|
>-      "ins            $t1, $t5, 16, 16              \n"  // |21|20|5|4|
>-      "ins            $t5, $t8, 0, 16               \n"  // |22|23|7|6|
>-      "raddu.w.qb     $t1, $t1                      \n"  // |21+20+5+4|
>-      "raddu.w.qb     $t5, $t5                      \n"  // |23+22+7+6|
>-      "shra_r.w       $t1, $t1, 2                   \n"  // |t1+2|>>2
>-      "shra_r.w       $t5, $t5, 2                   \n"  // |t5+2|>>2
>-      "srl            $t8, $t2, 16                  \n"  // |X|X|11|10|
>-      "ins            $t2, $t6, 16, 16              \n"  // |25|24|9|8|
>-      "ins            $t6, $t8, 0, 16               \n"  // |27|26|11|10|
>-      "raddu.w.qb     $t2, $t2                      \n"  // |25+24+9+8|
>-      "raddu.w.qb     $t6, $t6                      \n"  // |27+26+11+10|
>-      "shra_r.w       $t2, $t2, 2                   \n"  // |t2+2|>>2
>-      "shra_r.w       $t6, $t6, 2                   \n"  // |t5+2|>>2
>-      "srl            $t8, $t3, 16                  \n"  // |X|X|15|14|
>-      "ins            $t3, $t7, 16, 16              \n"  // |29|28|13|12|
>-      "ins            $t7, $t8, 0, 16               \n"  // |31|30|15|14|
>-      "raddu.w.qb     $t3, $t3                      \n"  // |29+28+13+12|
>-      "raddu.w.qb     $t7, $t7                      \n"  // |31+30+15+14|
>-      "shra_r.w       $t3, $t3, 2                   \n"  // |t3+2|>>2
>-      "shra_r.w       $t7, $t7, 2                   \n"  // |t7+2|>>2
>-      "addiu          %[src_ptr], %[src_ptr], 16    \n"
>-      "addiu          %[t], %[t], 16                \n"
>-      "sb             $t0, 0(%[dst])                \n"
>-      "sb             $t4, 1(%[dst])                \n"
>-      "sb             $t1, 2(%[dst])                \n"
>-      "sb             $t5, 3(%[dst])                \n"
>-      "sb             $t2, 4(%[dst])                \n"
>-      "sb             $t6, 5(%[dst])                \n"
>-      "sb             $t3, 6(%[dst])                \n"
>-      "sb             $t7, 7(%[dst])                \n"
>-      "bgtz           $t9, 1b                       \n"
>-      " addiu         %[dst], %[dst], 8             \n"
>-
>-      "2:                                           \n"
>-      "andi           $t9, %[dst_width], 0x7        \n"  // x = residue
>-      "beqz           $t9, 3f                       \n"
>-      " nop                                         \n"
>-
>-      "21:                                          \n"
>-      "lwr            $t1, 0(%[src_ptr])            \n"
>-      "lwl            $t1, 3(%[src_ptr])            \n"
>-      "lwr            $t2, 0(%[t])                  \n"
>-      "lwl            $t2, 3(%[t])                  \n"
>-      "srl            $t8, $t1, 16                  \n"
>-      "ins            $t1, $t2, 16, 16              \n"
>-      "ins            $t2, $t8, 0, 16               \n"
>-      "raddu.w.qb     $t1, $t1                      \n"
>-      "raddu.w.qb     $t2, $t2                      \n"
>-      "shra_r.w       $t1, $t1, 2                   \n"
>-      "shra_r.w       $t2, $t2, 2                   \n"
>-      "sb             $t1, 0(%[dst])                \n"
>-      "sb             $t2, 1(%[dst])                \n"
>-      "addiu          %[src_ptr], %[src_ptr], 4     \n"
>-      "addiu          $t9, $t9, -2                  \n"
>-      "addiu          %[t], %[t], 4                 \n"
>-      "bgtz           $t9, 21b                      \n"
>-      " addiu         %[dst], %[dst], 2             \n"
>-
>-      "3:                                           \n"
>-      ".set pop                                     \n"
>-
>-      : [src_ptr] "+r"(src_ptr), [dst] "+r"(dst), [t] "+r"(t)
>-      : [dst_width] "r"(dst_width)
>-      : "t0", "t1", "t2", "t3", "t4", "t5", "t6", "t7", "t8", "t9");
>-}
>-
>-void ScaleRowDown4_DSPR2(const uint8* src_ptr,
>-                         ptrdiff_t src_stride,
>-                         uint8* dst,
>-                         int dst_width) {
>-  __asm__ __volatile__(
>-      ".set push                                    \n"
>-      ".set noreorder                               \n"
>-
>-      "srl            $t9, %[dst_width], 3          \n"
>-      "beqz           $t9, 2f                       \n"
>-      " nop                                         \n"
>-
>-      "1:                                           \n"
>-      "lw             $t1, 0(%[src_ptr])            \n"  // |3|2|1|0|
>-      "lw             $t2, 4(%[src_ptr])            \n"  // |7|6|5|4|
>-      "lw             $t3, 8(%[src_ptr])            \n"  // |11|10|9|8|
>-      "lw             $t4, 12(%[src_ptr])           \n"  // |15|14|13|12|
>-      "lw             $t5, 16(%[src_ptr])           \n"  // |19|18|17|16|
>-      "lw             $t6, 20(%[src_ptr])           \n"  // |23|22|21|20|
>-      "lw             $t7, 24(%[src_ptr])           \n"  // |27|26|25|24|
>-      "lw             $t8, 28(%[src_ptr])           \n"  // |31|30|29|28|
>-      "precr.qb.ph    $t1, $t2, $t1                 \n"  // |6|4|2|0|
>-      "precr.qb.ph    $t2, $t4, $t3                 \n"  // |14|12|10|8|
>-      "precr.qb.ph    $t5, $t6, $t5                 \n"  // |22|20|18|16|
>-      "precr.qb.ph    $t6, $t8, $t7                 \n"  // |30|28|26|24|
>-      "precrq.qb.ph   $t1, $t2, $t1                 \n"  // |14|10|6|2|
>-      "precrq.qb.ph   $t5, $t6, $t5                 \n"  // |30|26|22|18|
>-      "addiu          %[src_ptr], %[src_ptr], 32    \n"
>-      "addiu          $t9, $t9, -1                  \n"
>-      "sw             $t1, 0(%[dst])                \n"
>-      "sw             $t5, 4(%[dst])                \n"
>-      "bgtz           $t9, 1b                       \n"
>-      " addiu         %[dst], %[dst], 8             \n"
>-
>-      "2:                                           \n"
>-      "andi           $t9, %[dst_width], 7          \n"  // residue
>-      "beqz           $t9, 3f                       \n"
>-      " nop                                         \n"
>-
>-      "21:                                          \n"
>-      "lbu            $t1, 2(%[src_ptr])            \n"
>-      "addiu          %[src_ptr], %[src_ptr], 4     \n"
>-      "addiu          $t9, $t9, -1                  \n"
>-      "sb             $t1, 0(%[dst])                \n"
>-      "bgtz           $t9, 21b                      \n"
>-      " addiu         %[dst], %[dst], 1             \n"
>-
>-      "3:                                           \n"
>-      ".set pop                                     \n"
>-      : [src_ptr] "+r"(src_ptr), [dst] "+r"(dst)
>-      : [dst_width] "r"(dst_width)
>-      : "t1", "t2", "t3", "t4", "t5", "t6", "t7", "t8", "t9");
>-}
>-
>-void ScaleRowDown4Box_DSPR2(const uint8* src_ptr,
>-                            ptrdiff_t src_stride,
>-                            uint8* dst,
>-                            int dst_width) {
>-  intptr_t stride = src_stride;
>-  const uint8* s1 = src_ptr + stride;
>-  const uint8* s2 = s1 + stride;
>-  const uint8* s3 = s2 + stride;
>-
>-  __asm__ __volatile__(
>-      ".set push                                  \n"
>-      ".set noreorder                             \n"
>-
>-      "srl           $t9, %[dst_width], 1         \n"
>-      "andi          $t8, %[dst_width], 1         \n"
>-
>-      "1:                                         \n"
>-      "lw            $t0, 0(%[src_ptr])           \n"  // |3|2|1|0|
>-      "lw            $t1, 0(%[s1])                \n"  // |7|6|5|4|
>-      "lw            $t2, 0(%[s2])                \n"  // |11|10|9|8|
>-      "lw            $t3, 0(%[s3])                \n"  // |15|14|13|12|
>-      "lw            $t4, 4(%[src_ptr])           \n"  // |19|18|17|16|
>-      "lw            $t5, 4(%[s1])                \n"  // |23|22|21|20|
>-      "lw            $t6, 4(%[s2])                \n"  // |27|26|25|24|
>-      "lw            $t7, 4(%[s3])                \n"  // |31|30|29|28|
>-      "raddu.w.qb    $t0, $t0                     \n"  // |3 + 2 + 1 + 0|
>-      "raddu.w.qb    $t1, $t1                     \n"  // |7 + 6 + 5 + 4|
>-      "raddu.w.qb    $t2, $t2                     \n"  // |11 + 10 + 9 + 8|
>-      "raddu.w.qb    $t3, $t3                     \n"  // |15 + 14 + 13 + 12|
>-      "raddu.w.qb    $t4, $t4                     \n"  // |19 + 18 + 17 + 16|
>-      "raddu.w.qb    $t5, $t5                     \n"  // |23 + 22 + 21 + 20|
>-      "raddu.w.qb    $t6, $t6                     \n"  // |27 + 26 + 25 + 24|
>-      "raddu.w.qb    $t7, $t7                     \n"  // |31 + 30 + 29 + 28|
>-      "add           $t0, $t0, $t1                \n"
>-      "add           $t1, $t2, $t3                \n"
>-      "add           $t0, $t0, $t1                \n"
>-      "add           $t4, $t4, $t5                \n"
>-      "add           $t6, $t6, $t7                \n"
>-      "add           $t4, $t4, $t6                \n"
>-      "shra_r.w      $t0, $t0, 4                  \n"
>-      "shra_r.w      $t4, $t4, 4                  \n"
>-      "sb            $t0, 0(%[dst])               \n"
>-      "sb            $t4, 1(%[dst])               \n"
>-      "addiu         %[src_ptr], %[src_ptr], 8    \n"
>-      "addiu         %[s1], %[s1], 8              \n"
>-      "addiu         %[s2], %[s2], 8              \n"
>-      "addiu         %[s3], %[s3], 8              \n"
>-      "addiu         $t9, $t9, -1                 \n"
>-      "bgtz          $t9, 1b                      \n"
>-      " addiu        %[dst], %[dst], 2            \n"
>-      "beqz          $t8, 2f                      \n"
>-      " nop                                       \n"
>-
>-      "lw            $t0, 0(%[src_ptr])           \n"  // |3|2|1|0|
>-      "lw            $t1, 0(%[s1])                \n"  // |7|6|5|4|
>-      "lw            $t2, 0(%[s2])                \n"  // |11|10|9|8|
>-      "lw            $t3, 0(%[s3])                \n"  // |15|14|13|12|
>-      "raddu.w.qb    $t0, $t0                     \n"  // |3 + 2 + 1 + 0|
>-      "raddu.w.qb    $t1, $t1                     \n"  // |7 + 6 + 5 + 4|
>-      "raddu.w.qb    $t2, $t2                     \n"  // |11 + 10 + 9 + 8|
>-      "raddu.w.qb    $t3, $t3                     \n"  // |15 + 14 + 13 + 12|
>-      "add           $t0, $t0, $t1                \n"
>-      "add           $t1, $t2, $t3                \n"
>-      "add           $t0, $t0, $t1                \n"
>-      "shra_r.w      $t0, $t0, 4                  \n"
>-      "sb            $t0, 0(%[dst])               \n"
>-
>-      "2:                                         \n"
>-      ".set pop                                   \n"
>-
>-      : [src_ptr] "+r"(src_ptr), [dst] "+r"(dst), [s1] "+r"(s1), [s2] "+r"(s2),
>-        [s3] "+r"(s3)
>-      : [dst_width] "r"(dst_width)
>-      : "t0", "t1", "t2", "t3", "t4", "t5", "t6", "t7", "t8", "t9");
>-}
>-
>-void ScaleRowDown34_DSPR2(const uint8* src_ptr,
>-                          ptrdiff_t src_stride,
>-                          uint8* dst,
>-                          int dst_width) {
>-  __asm__ __volatile__(
>-      ".set push                                          \n"
>-      ".set noreorder                                     \n"
>-      "1:                                                 \n"
>-      "lw              $t1, 0(%[src_ptr])                 \n"  // |3|2|1|0|
>-      "lw              $t2, 4(%[src_ptr])                 \n"  // |7|6|5|4|
>-      "lw              $t3, 8(%[src_ptr])                 \n"  // |11|10|9|8|
>-      "lw              $t4, 12(%[src_ptr])                \n"  // |15|14|13|12|
>-      "lw              $t5, 16(%[src_ptr])                \n"  // |19|18|17|16|
>-      "lw              $t6, 20(%[src_ptr])                \n"  // |23|22|21|20|
>-      "lw              $t7, 24(%[src_ptr])                \n"  // |27|26|25|24|
>-      "lw              $t8, 28(%[src_ptr])                \n"  // |31|30|29|28|
>-      "precrq.qb.ph    $t0, $t2, $t4                      \n"  // |7|5|15|13|
>-      "precrq.qb.ph    $t9, $t6, $t8                      \n"  // |23|21|31|30|
>-      "addiu           %[dst_width], %[dst_width], -24    \n"
>-      "ins             $t1, $t1, 8, 16                    \n"  // |3|1|0|X|
>-      "ins             $t4, $t0, 8, 16                    \n"  // |X|15|13|12|
>-      "ins             $t5, $t5, 8, 16                    \n"  // |19|17|16|X|
>-      "ins             $t8, $t9, 8, 16                    \n"  // |X|31|29|28|
>-      "addiu           %[src_ptr], %[src_ptr], 32         \n"
>-      "packrl.ph       $t0, $t3, $t0                      \n"  // |9|8|7|5|
>-      "packrl.ph       $t9, $t7, $t9                      \n"  // |25|24|23|21|
>-      "prepend         $t1, $t2, 8                        \n"  // |4|3|1|0|
>-      "prepend         $t3, $t4, 24                       \n"  // |15|13|12|11|
>-      "prepend         $t5, $t6, 8                        \n"  // |20|19|17|16|
>-      "prepend         $t7, $t8, 24                       \n"  // |31|29|28|27|
>-      "sw              $t1, 0(%[dst])                     \n"
>-      "sw              $t0, 4(%[dst])                     \n"
>-      "sw              $t3, 8(%[dst])                     \n"
>-      "sw              $t5, 12(%[dst])                    \n"
>-      "sw              $t9, 16(%[dst])                    \n"
>-      "sw              $t7, 20(%[dst])                    \n"
>-      "bnez            %[dst_width], 1b                   \n"
>-      " addiu          %[dst], %[dst], 24                 \n"
>-      ".set pop                                           \n"
>-      : [src_ptr] "+r"(src_ptr), [dst] "+r"(dst), [dst_width] "+r"(dst_width)
>-      :
>-      : "t0", "t1", "t2", "t3", "t4", "t5", "t6", "t7", "t8", "t9");
>-}
>-
>-void ScaleRowDown34_0_Box_DSPR2(const uint8* src_ptr,
>-                                ptrdiff_t src_stride,
>-                                uint8* d,
>-                                int dst_width) {
>-  __asm__ __volatile__(
>-      ".set push                                         \n"
>-      ".set noreorder                                    \n"
>-      "repl.ph           $t3, 3                          \n"  // 0x00030003
>-
>-      "1:                                                \n"
>-      "lw                $t0, 0(%[src_ptr])              \n"  // |S3|S2|S1|S0|
>-      "lwx               $t1, %[src_stride](%[src_ptr])  \n"  // |T3|T2|T1|T0|
>-      "rotr              $t2, $t0, 8                     \n"  // |S0|S3|S2|S1|
>-      "rotr              $t6, $t1, 8                     \n"  // |T0|T3|T2|T1|
>-      "muleu_s.ph.qbl    $t4, $t2, $t3                   \n"  // |S0*3|S3*3|
>-      "muleu_s.ph.qbl    $t5, $t6, $t3                   \n"  // |T0*3|T3*3|
>-      "andi              $t0, $t2, 0xFFFF                \n"  // |0|0|S2|S1|
>-      "andi              $t1, $t6, 0xFFFF                \n"  // |0|0|T2|T1|
>-      "raddu.w.qb        $t0, $t0                        \n"
>-      "raddu.w.qb        $t1, $t1                        \n"
>-      "shra_r.w          $t0, $t0, 1                     \n"
>-      "shra_r.w          $t1, $t1, 1                     \n"
>-      "preceu.ph.qbr     $t2, $t2                        \n"  // |0|S2|0|S1|
>-      "preceu.ph.qbr     $t6, $t6                        \n"  // |0|T2|0|T1|
>-      "rotr              $t2, $t2, 16                    \n"  // |0|S1|0|S2|
>-      "rotr              $t6, $t6, 16                    \n"  // |0|T1|0|T2|
>-      "addu.ph           $t2, $t2, $t4                   \n"
>-      "addu.ph           $t6, $t6, $t5                   \n"
>-      "sll               $t5, $t0, 1                     \n"
>-      "add               $t0, $t5, $t0                   \n"
>-      "shra_r.ph         $t2, $t2, 2                     \n"
>-      "shra_r.ph         $t6, $t6, 2                     \n"
>-      "shll.ph           $t4, $t2, 1                     \n"
>-      "addq.ph           $t4, $t4, $t2                   \n"
>-      "addu              $t0, $t0, $t1                   \n"
>-      "addiu             %[src_ptr], %[src_ptr], 4       \n"
>-      "shra_r.w          $t0, $t0, 2                     \n"
>-      "addu.ph           $t6, $t6, $t4                   \n"
>-      "shra_r.ph         $t6, $t6, 2                     \n"
>-      "srl               $t1, $t6, 16                    \n"
>-      "addiu             %[dst_width], %[dst_width], -3  \n"
>-      "sb                $t1, 0(%[d])                    \n"
>-      "sb                $t0, 1(%[d])                    \n"
>-      "sb                $t6, 2(%[d])                    \n"
>-      "bgtz              %[dst_width], 1b                \n"
>-      " addiu            %[d], %[d], 3                   \n"
>-      "3:                                                \n"
>-      ".set pop                                          \n"
>-      : [src_ptr] "+r"(src_ptr), [src_stride] "+r"(src_stride), [d] "+r"(d),
>-        [dst_width] "+r"(dst_width)
>-      :
>-      : "t0", "t1", "t2", "t3", "t4", "t5", "t6");
>-}
>-
>-void ScaleRowDown34_1_Box_DSPR2(const uint8* src_ptr,
>-                                ptrdiff_t src_stride,
>-                                uint8* d,
>-                                int dst_width) {
>-  __asm__ __volatile__(
>-      ".set push                                           \n"
>-      ".set noreorder                                      \n"
>-      "repl.ph           $t2, 3                            \n"  // 0x00030003
>-
>-      "1:                                                  \n"
>-      "lw                $t0, 0(%[src_ptr])                \n"  // |S3|S2|S1|S0|
>-      "lwx               $t1, %[src_stride](%[src_ptr])    \n"  // |T3|T2|T1|T0|
>-      "rotr              $t4, $t0, 8                       \n"  // |S0|S3|S2|S1|
>-      "rotr              $t6, $t1, 8                       \n"  // |T0|T3|T2|T1|
>-      "muleu_s.ph.qbl    $t3, $t4, $t2                     \n"  // |S0*3|S3*3|
>-      "muleu_s.ph.qbl    $t5, $t6, $t2                     \n"  // |T0*3|T3*3|
>-      "andi              $t0, $t4, 0xFFFF                  \n"  // |0|0|S2|S1|
>-      "andi              $t1, $t6, 0xFFFF                  \n"  // |0|0|T2|T1|
>-      "raddu.w.qb        $t0, $t0                          \n"
>-      "raddu.w.qb        $t1, $t1                          \n"
>-      "shra_r.w          $t0, $t0, 1                       \n"
>-      "shra_r.w          $t1, $t1, 1                       \n"
>-      "preceu.ph.qbr     $t4, $t4                          \n"  // |0|S2|0|S1|
>-      "preceu.ph.qbr     $t6, $t6                          \n"  // |0|T2|0|T1|
>-      "rotr              $t4, $t4, 16                      \n"  // |0|S1|0|S2|
>-      "rotr              $t6, $t6, 16                      \n"  // |0|T1|0|T2|
>-      "addu.ph           $t4, $t4, $t3                     \n"
>-      "addu.ph           $t6, $t6, $t5                     \n"
>-      "shra_r.ph         $t6, $t6, 2                       \n"
>-      "shra_r.ph         $t4, $t4, 2                       \n"
>-      "addu.ph           $t6, $t6, $t4                     \n"
>-      "addiu             %[src_ptr], %[src_ptr], 4         \n"
>-      "shra_r.ph         $t6, $t6, 1                       \n"
>-      "addu              $t0, $t0, $t1                     \n"
>-      "addiu             %[dst_width], %[dst_width], -3    \n"
>-      "shra_r.w          $t0, $t0, 1                       \n"
>-      "srl               $t1, $t6, 16                      \n"
>-      "sb                $t1, 0(%[d])                      \n"
>-      "sb                $t0, 1(%[d])                      \n"
>-      "sb                $t6, 2(%[d])                      \n"
>-      "bgtz              %[dst_width], 1b                  \n"
>-      " addiu            %[d], %[d], 3                     \n"
>-      "3:                                                  \n"
>-      ".set pop                                            \n"
>-      : [src_ptr] "+r"(src_ptr), [src_stride] "+r"(src_stride), [d] "+r"(d),
>-        [dst_width] "+r"(dst_width)
>-      :
>-      : "t0", "t1", "t2", "t3", "t4", "t5", "t6");
>-}
>-
>-void ScaleRowDown38_DSPR2(const uint8* src_ptr,
>-                          ptrdiff_t src_stride,
>-                          uint8* dst,
>-                          int dst_width) {
>-  __asm__ __volatile__(
>-      ".set push                                     \n"
>-      ".set noreorder                                \n"
>-
>-      "1:                                            \n"
>-      "lw         $t0, 0(%[src_ptr])                 \n"  // |3|2|1|0|
>-      "lw         $t1, 4(%[src_ptr])                 \n"  // |7|6|5|4|
>-      "lw         $t2, 8(%[src_ptr])                 \n"  // |11|10|9|8|
>-      "lw         $t3, 12(%[src_ptr])                \n"  // |15|14|13|12|
>-      "lw         $t4, 16(%[src_ptr])                \n"  // |19|18|17|16|
>-      "lw         $t5, 20(%[src_ptr])                \n"  // |23|22|21|20|
>-      "lw         $t6, 24(%[src_ptr])                \n"  // |27|26|25|24|
>-      "lw         $t7, 28(%[src_ptr])                \n"  // |31|30|29|28|
>-      "wsbh       $t0, $t0                           \n"  // |2|3|0|1|
>-      "wsbh       $t6, $t6                           \n"  // |26|27|24|25|
>-      "srl        $t0, $t0, 8                        \n"  // |X|2|3|0|
>-      "srl        $t3, $t3, 16                       \n"  // |X|X|15|14|
>-      "srl        $t5, $t5, 16                       \n"  // |X|X|23|22|
>-      "srl        $t7, $t7, 16                       \n"  // |X|X|31|30|
>-      "ins        $t1, $t2, 24, 8                    \n"  // |8|6|5|4|
>-      "ins        $t6, $t5, 0, 8                     \n"  // |26|27|24|22|
>-      "ins        $t1, $t0, 0, 16                    \n"  // |8|6|3|0|
>-      "ins        $t6, $t7, 24, 8                    \n"  // |30|27|24|22|
>-      "prepend    $t2, $t3, 24                       \n"  // |X|15|14|11|
>-      "ins        $t4, $t4, 16, 8                    \n"  // |19|16|17|X|
>-      "ins        $t4, $t2, 0, 16                    \n"  // |19|16|14|11|
>-      "addiu      %[src_ptr], %[src_ptr], 32         \n"
>-      "addiu      %[dst_width], %[dst_width], -12    \n"
>-      "addiu      $t8,%[dst_width], -12              \n"
>-      "sw         $t1, 0(%[dst])                     \n"
>-      "sw         $t4, 4(%[dst])                     \n"
>-      "sw         $t6, 8(%[dst])                     \n"
>-      "bgez       $t8, 1b                            \n"
>-      " addiu     %[dst], %[dst], 12                 \n"
>-      ".set pop                                      \n"
>-      : [src_ptr] "+r"(src_ptr), [dst] "+r"(dst), [dst_width] "+r"(dst_width)
>-      :
>-      : "t0", "t1", "t2", "t3", "t4", "t5", "t6", "t7", "t8");
>-}
>-
>-void ScaleRowDown38_2_Box_DSPR2(const uint8* src_ptr,
>-                                ptrdiff_t src_stride,
>-                                uint8* dst_ptr,
>-                                int dst_width) {
>-  intptr_t stride = src_stride;
>-  const uint8* t = src_ptr + stride;
>-  const int c = 0x2AAA;
>-
>-  __asm__ __volatile__(
>-      ".set push                                         \n"
>-      ".set noreorder                                    \n"
>-
>-      "1:                                                \n"
>-      "lw              $t0, 0(%[src_ptr])                \n"  // |S3|S2|S1|S0|
>-      "lw              $t1, 4(%[src_ptr])                \n"  // |S7|S6|S5|S4|
>-      "lw              $t2, 0(%[t])                      \n"  // |T3|T2|T1|T0|
>-      "lw              $t3, 4(%[t])                      \n"  // |T7|T6|T5|T4|
>-      "rotr            $t1, $t1, 16                      \n"  // |S5|S4|S7|S6|
>-      "packrl.ph       $t4, $t1, $t3                     \n"  // |S7|S6|T7|T6|
>-      "packrl.ph       $t5, $t3, $t1                     \n"  // |T5|T4|S5|S4|
>-      "raddu.w.qb      $t4, $t4                          \n"  // S7+S6+T7+T6
>-      "raddu.w.qb      $t5, $t5                          \n"  // T5+T4+S5+S4
>-      "precrq.qb.ph    $t6, $t0, $t2                     \n"  // |S3|S1|T3|T1|
>-      "precrq.qb.ph    $t6, $t6, $t6                     \n"  // |S3|T3|S3|T3|
>-      "srl             $t4, $t4, 2                       \n"  // t4 / 4
>-      "srl             $t6, $t6, 16                      \n"  // |0|0|S3|T3|
>-      "raddu.w.qb      $t6, $t6                          \n"  // 0+0+S3+T3
>-      "addu            $t6, $t5, $t6                     \n"
>-      "mul             $t6, $t6, %[c]                    \n"  // t6 * 0x2AAA
>-      "sll             $t0, $t0, 8                       \n"  // |S2|S1|S0|0|
>-      "sll             $t2, $t2, 8                       \n"  // |T2|T1|T0|0|
>-      "raddu.w.qb      $t0, $t0                          \n"  // S2+S1+S0+0
>-      "raddu.w.qb      $t2, $t2                          \n"  // T2+T1+T0+0
>-      "addu            $t0, $t0, $t2                     \n"
>-      "mul             $t0, $t0, %[c]                    \n"  // t0 * 0x2AAA
>-      "addiu           %[src_ptr], %[src_ptr], 8         \n"
>-      "addiu           %[t], %[t], 8                     \n"
>-      "addiu           %[dst_width], %[dst_width], -3    \n"
>-      "addiu           %[dst_ptr], %[dst_ptr], 3         \n"
>-      "srl             $t6, $t6, 16                      \n"
>-      "srl             $t0, $t0, 16                      \n"
>-      "sb              $t4, -1(%[dst_ptr])               \n"
>-      "sb              $t6, -2(%[dst_ptr])               \n"
>-      "bgtz            %[dst_width], 1b                  \n"
>-      " sb             $t0, -3(%[dst_ptr])               \n"
>-      ".set pop                                          \n"
>-      : [src_ptr] "+r"(src_ptr), [dst_ptr] "+r"(dst_ptr), [t] "+r"(t),
>-        [dst_width] "+r"(dst_width)
>-      : [c] "r"(c)
>-      : "t0", "t1", "t2", "t3", "t4", "t5", "t6");
>-}
>-
>-void ScaleRowDown38_3_Box_DSPR2(const uint8* src_ptr,
>-                                ptrdiff_t src_stride,
>-                                uint8* dst_ptr,
>-                                int dst_width) {
>-  intptr_t stride = src_stride;
>-  const uint8* s1 = src_ptr + stride;
>-  stride += stride;
>-  const uint8* s2 = src_ptr + stride;
>-  const int c1 = 0x1C71;
>-  const int c2 = 0x2AAA;
>-
>-  __asm__ __volatile__(
>-      ".set push                                         \n"
>-      ".set noreorder                                    \n"
>-
>-      "1:                                                \n"
>-      "lw              $t0, 0(%[src_ptr])                \n"  // |S3|S2|S1|S0|
>-      "lw              $t1, 4(%[src_ptr])                \n"  // |S7|S6|S5|S4|
>-      "lw              $t2, 0(%[s1])                     \n"  // |T3|T2|T1|T0|
>-      "lw              $t3, 4(%[s1])                     \n"  // |T7|T6|T5|T4|
>-      "lw              $t4, 0(%[s2])                     \n"  // |R3|R2|R1|R0|
>-      "lw              $t5, 4(%[s2])                     \n"  // |R7|R6|R5|R4|
>-      "rotr            $t1, $t1, 16                      \n"  // |S5|S4|S7|S6|
>-      "packrl.ph       $t6, $t1, $t3                     \n"  // |S7|S6|T7|T6|
>-      "raddu.w.qb      $t6, $t6                          \n"  // S7+S6+T7+T6
>-      "packrl.ph       $t7, $t3, $t1                     \n"  // |T5|T4|S5|S4|
>-      "raddu.w.qb      $t7, $t7                          \n"  // T5+T4+S5+S4
>-      "sll             $t8, $t5, 16                      \n"  // |R5|R4|0|0|
>-      "raddu.w.qb      $t8, $t8                          \n"  // R5+R4
>-      "addu            $t7, $t7, $t8                     \n"
>-      "srl             $t8, $t5, 16                      \n"  // |0|0|R7|R6|
>-      "raddu.w.qb      $t8, $t8                          \n"  // R7 + R6
>-      "addu            $t6, $t6, $t8                     \n"
>-      "mul             $t6, $t6, %[c2]                   \n"  // t6 * 0x2AAA
>-      "precrq.qb.ph    $t8, $t0, $t2                     \n"  // |S3|S1|T3|T1|
>-      "precrq.qb.ph    $t8, $t8, $t4                     \n"  // |S3|T3|R3|R1|
>-      "srl             $t8, $t8, 8                       \n"  // |0|S3|T3|R3|
>-      "raddu.w.qb      $t8, $t8                          \n"  // S3 + T3 + R3
>-      "addu            $t7, $t7, $t8                     \n"
>-      "mul             $t7, $t7, %[c1]                   \n"  // t7 * 0x1C71
>-      "sll             $t0, $t0, 8                       \n"  // |S2|S1|S0|0|
>-      "sll             $t2, $t2, 8                       \n"  // |T2|T1|T0|0|
>-      "sll             $t4, $t4, 8                       \n"  // |R2|R1|R0|0|
>-      "raddu.w.qb      $t0, $t0                          \n"
>-      "raddu.w.qb      $t2, $t2                          \n"
>-      "raddu.w.qb      $t4, $t4                          \n"
>-      "addu            $t0, $t0, $t2                     \n"
>-      "addu            $t0, $t0, $t4                     \n"
>-      "mul             $t0, $t0, %[c1]                   \n"  // t0 * 0x1C71
>-      "addiu           %[src_ptr], %[src_ptr], 8         \n"
>-      "addiu           %[s1], %[s1], 8                   \n"
>-      "addiu           %[s2], %[s2], 8                   \n"
>-      "addiu           %[dst_width], %[dst_width], -3    \n"
>-      "addiu           %[dst_ptr], %[dst_ptr], 3         \n"
>-      "srl             $t6, $t6, 16                      \n"
>-      "srl             $t7, $t7, 16                      \n"
>-      "srl             $t0, $t0, 16                      \n"
>-      "sb              $t6, -1(%[dst_ptr])               \n"
>-      "sb              $t7, -2(%[dst_ptr])               \n"
>-      "bgtz            %[dst_width], 1b                  \n"
>-      " sb             $t0, -3(%[dst_ptr])               \n"
>-      ".set pop                                          \n"
>-      : [src_ptr] "+r"(src_ptr), [dst_ptr] "+r"(dst_ptr), [s1] "+r"(s1),
>-        [s2] "+r"(s2), [dst_width] "+r"(dst_width)
>-      : [c1] "r"(c1), [c2] "r"(c2)
>-      : "t0", "t1", "t2", "t3", "t4", "t5", "t6", "t7", "t8");
>-}
>-
>-void ScaleAddRow_DSPR2(const uint8* src_ptr, uint16* dst_ptr, int src_width) {
>-  int x;
>-  for (x = 0; x < ((src_width - 1)); x += 8) {
>-    uint32 tmp_t1, tmp_t2, tmp_t3, tmp_t4;
>-    uint32 tmp_t5, tmp_t6, tmp_t7, tmp_t8;
>-    __asm__ __volatile__(
>-        ".set push                                                \n"
>-        ".set noreorder                                           \n"
>-        "lw                %[tmp_t5],   0(%[src_ptr])             \n"
>-        "lw                %[tmp_t6],   4(%[src_ptr])             \n"
>-        "lw                %[tmp_t1],   0(%[dst_ptr])             \n"
>-        "lw                %[tmp_t2],   4(%[dst_ptr])             \n"
>-        "lw                %[tmp_t3],   8(%[dst_ptr])             \n"
>-        "lw                %[tmp_t4],   12(%[dst_ptr])            \n"
>-        "preceu.ph.qbr     %[tmp_t7],   %[tmp_t5]                 \n"
>-        "preceu.ph.qbl     %[tmp_t8],   %[tmp_t5]                 \n"
>-        "addu.ph           %[tmp_t1],   %[tmp_t1],     %[tmp_t7]  \n"
>-        "addu.ph           %[tmp_t2],   %[tmp_t2],     %[tmp_t8]  \n"
>-        "preceu.ph.qbr     %[tmp_t7],   %[tmp_t6]                 \n"
>-        "preceu.ph.qbl     %[tmp_t8],   %[tmp_t6]                 \n"
>-        "addu.ph           %[tmp_t3],   %[tmp_t3],     %[tmp_t7]  \n"
>-        "addu.ph           %[tmp_t4],   %[tmp_t4],     %[tmp_t8]  \n"
>-        "sw                %[tmp_t1],   0(%[dst_ptr])             \n"
>-        "sw                %[tmp_t2],   4(%[dst_ptr])             \n"
>-        "sw                %[tmp_t3],   8(%[dst_ptr])             \n"
>-        "sw                %[tmp_t4],   12(%[dst_ptr])            \n"
>-        ".set pop                                                 \n"
>-        :
>-        [tmp_t1] "=&r"(tmp_t1), [tmp_t2] "=&r"(tmp_t2), [tmp_t3] "=&r"(tmp_t3),
>-        [tmp_t4] "=&r"(tmp_t4), [tmp_t5] "=&r"(tmp_t5), [tmp_t6] "=&r"(tmp_t6),
>-        [tmp_t7] "=&r"(tmp_t7), [tmp_t8] "=&r"(tmp_t8), [src_ptr] "+r"(src_ptr)
>-        : [dst_ptr] "r"(dst_ptr));
>-    src_ptr += 8;
>-    dst_ptr += 8;
>-  }
>-
>-  if ((src_width)&7) {
>-    for (x = 0; x < ((src_width - 1) & 7); x += 1) {
>-      dst_ptr[0] += src_ptr[0];
>-      src_ptr += 1;
>-      dst_ptr += 1;
>-    }
>-  }
>-}
>-
>-#endif  // defined(__mips_dsp) && (__mips_dsp_rev >= 2)
>-
>-#ifdef __cplusplus
>-}  // extern "C"
>-}  // namespace libyuv
>-#endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/scale_gcc.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/scale_gcc.cc
>index f0ac56fcb06..312236d2df8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/scale_gcc.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/scale_gcc.cc
>@@ -21,462 +21,458 @@ extern "C" {
>     (defined(__x86_64__) || (defined(__i386__) && !defined(_MSC_VER)))
> 
> // Offsets for source bytes 0 to 9
>-static uvec8 kShuf0 = {0,   1,   3,   4,   5,   7,   8,   9,
>-                       128, 128, 128, 128, 128, 128, 128, 128};
>+static const uvec8 kShuf0 = {0,   1,   3,   4,   5,   7,   8,   9,
>+                             128, 128, 128, 128, 128, 128, 128, 128};
> 
> // Offsets for source bytes 11 to 20 with 8 subtracted = 3 to 12.
>-static uvec8 kShuf1 = {3,   4,   5,   7,   8,   9,   11,  12,
>-                       128, 128, 128, 128, 128, 128, 128, 128};
>+static const uvec8 kShuf1 = {3,   4,   5,   7,   8,   9,   11,  12,
>+                             128, 128, 128, 128, 128, 128, 128, 128};
> 
> // Offsets for source bytes 21 to 31 with 16 subtracted = 5 to 31.
>-static uvec8 kShuf2 = {5,   7,   8,   9,   11,  12,  13,  15,
>-                       128, 128, 128, 128, 128, 128, 128, 128};
>+static const uvec8 kShuf2 = {5,   7,   8,   9,   11,  12,  13,  15,
>+                             128, 128, 128, 128, 128, 128, 128, 128};
> 
> // Offsets for source bytes 0 to 10
>-static uvec8 kShuf01 = {0, 1, 1, 2, 2, 3, 4, 5, 5, 6, 6, 7, 8, 9, 9, 10};
>+static const uvec8 kShuf01 = {0, 1, 1, 2, 2, 3, 4, 5, 5, 6, 6, 7, 8, 9, 9, 10};
> 
> // Offsets for source bytes 10 to 21 with 8 subtracted = 3 to 13.
>-static uvec8 kShuf11 = {2, 3, 4, 5, 5, 6, 6, 7, 8, 9, 9, 10, 10, 11, 12, 13};
>+static const uvec8 kShuf11 = {2, 3, 4, 5,  5,  6,  6,  7,
>+                              8, 9, 9, 10, 10, 11, 12, 13};
> 
> // Offsets for source bytes 21 to 31 with 16 subtracted = 5 to 31.
>-static uvec8 kShuf21 = {5,  6,  6,  7,  8,  9,  9,  10,
>-                        10, 11, 12, 13, 13, 14, 14, 15};
>+static const uvec8 kShuf21 = {5,  6,  6,  7,  8,  9,  9,  10,
>+                              10, 11, 12, 13, 13, 14, 14, 15};
> 
> // Coefficients for source bytes 0 to 10
>-static uvec8 kMadd01 = {3, 1, 2, 2, 1, 3, 3, 1, 2, 2, 1, 3, 3, 1, 2, 2};
>+static const uvec8 kMadd01 = {3, 1, 2, 2, 1, 3, 3, 1, 2, 2, 1, 3, 3, 1, 2, 2};
> 
> // Coefficients for source bytes 10 to 21
>-static uvec8 kMadd11 = {1, 3, 3, 1, 2, 2, 1, 3, 3, 1, 2, 2, 1, 3, 3, 1};
>+static const uvec8 kMadd11 = {1, 3, 3, 1, 2, 2, 1, 3, 3, 1, 2, 2, 1, 3, 3, 1};
> 
> // Coefficients for source bytes 21 to 31
>-static uvec8 kMadd21 = {2, 2, 1, 3, 3, 1, 2, 2, 1, 3, 3, 1, 2, 2, 1, 3};
>+static const uvec8 kMadd21 = {2, 2, 1, 3, 3, 1, 2, 2, 1, 3, 3, 1, 2, 2, 1, 3};
> 
> // Coefficients for source bytes 21 to 31
>-static vec16 kRound34 = {2, 2, 2, 2, 2, 2, 2, 2};
>+static const vec16 kRound34 = {2, 2, 2, 2, 2, 2, 2, 2};
> 
>-static uvec8 kShuf38a = {0,   3,   6,   8,   11,  14,  128, 128,
>-                         128, 128, 128, 128, 128, 128, 128, 128};
>+static const uvec8 kShuf38a = {0,   3,   6,   8,   11,  14,  128, 128,
>+                               128, 128, 128, 128, 128, 128, 128, 128};
> 
>-static uvec8 kShuf38b = {128, 128, 128, 128, 128, 128, 0,   3,
>-                         6,   8,   11,  14,  128, 128, 128, 128};
>+static const uvec8 kShuf38b = {128, 128, 128, 128, 128, 128, 0,   3,
>+                               6,   8,   11,  14,  128, 128, 128, 128};
> 
> // Arrange words 0,3,6 into 0,1,2
>-static uvec8 kShufAc = {0,   1,   6,   7,   12,  13,  128, 128,
>-                        128, 128, 128, 128, 128, 128, 128, 128};
>+static const uvec8 kShufAc = {0,   1,   6,   7,   12,  13,  128, 128,
>+                              128, 128, 128, 128, 128, 128, 128, 128};
> 
> // Arrange words 0,3,6 into 3,4,5
>-static uvec8 kShufAc3 = {128, 128, 128, 128, 128, 128, 0,   1,
>-                         6,   7,   12,  13,  128, 128, 128, 128};
>+static const uvec8 kShufAc3 = {128, 128, 128, 128, 128, 128, 0,   1,
>+                               6,   7,   12,  13,  128, 128, 128, 128};
> 
> // Scaling values for boxes of 3x3 and 2x3
>-static uvec16 kScaleAc33 = {65536 / 9, 65536 / 9, 65536 / 6, 65536 / 9,
>-                            65536 / 9, 65536 / 6, 0,         0};
>+static const uvec16 kScaleAc33 = {65536 / 9, 65536 / 9, 65536 / 6, 65536 / 9,
>+                                  65536 / 9, 65536 / 6, 0,         0};
> 
> // Arrange first value for pixels 0,1,2,3,4,5
>-static uvec8 kShufAb0 = {0,  128, 3,  128, 6,   128, 8,   128,
>-                         11, 128, 14, 128, 128, 128, 128, 128};
>+static const uvec8 kShufAb0 = {0,  128, 3,  128, 6,   128, 8,   128,
>+                               11, 128, 14, 128, 128, 128, 128, 128};
> 
> // Arrange second value for pixels 0,1,2,3,4,5
>-static uvec8 kShufAb1 = {1,  128, 4,  128, 7,   128, 9,   128,
>-                         12, 128, 15, 128, 128, 128, 128, 128};
>+static const uvec8 kShufAb1 = {1,  128, 4,  128, 7,   128, 9,   128,
>+                               12, 128, 15, 128, 128, 128, 128, 128};
> 
> // Arrange third value for pixels 0,1,2,3,4,5
>-static uvec8 kShufAb2 = {2,  128, 5,   128, 128, 128, 10,  128,
>-                         13, 128, 128, 128, 128, 128, 128, 128};
>+static const uvec8 kShufAb2 = {2,  128, 5,   128, 128, 128, 10,  128,
>+                               13, 128, 128, 128, 128, 128, 128, 128};
> 
> // Scaling values for boxes of 3x2 and 2x2
>-static uvec16 kScaleAb2 = {65536 / 3, 65536 / 3, 65536 / 2, 65536 / 3,
>-                           65536 / 3, 65536 / 2, 0,         0};
>+static const uvec16 kScaleAb2 = {65536 / 3, 65536 / 3, 65536 / 2, 65536 / 3,
>+                                 65536 / 3, 65536 / 2, 0,         0};
> 
> // GCC versions of row functions are verbatim conversions from Visual C.
> // Generated using gcc disassembly on Visual C object file:
> // objdump -D yuvscaler.obj >yuvscaler.txt
> 
>-void ScaleRowDown2_SSSE3(const uint8* src_ptr,
>+void ScaleRowDown2_SSSE3(const uint8_t* src_ptr,
>                          ptrdiff_t src_stride,
>-                         uint8* dst_ptr,
>+                         uint8_t* dst_ptr,
>                          int dst_width) {
>   (void)src_stride;
>-  asm volatile (
>-    LABELALIGN
>-  "1:                                          \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    "movdqu    " MEMACCESS2(0x10,0) ",%%xmm1   \n"
>-    "lea       " MEMLEA(0x20,0) ",%0           \n"
>-    "psrlw     $0x8,%%xmm0                     \n"
>-    "psrlw     $0x8,%%xmm1                     \n"
>-    "packuswb  %%xmm1,%%xmm0                   \n"
>-    "movdqu    %%xmm0," MEMACCESS(1) "         \n"
>-    "lea       " MEMLEA(0x10,1) ",%1           \n"
>-    "sub       $0x10,%2                        \n"
>-    "jg        1b                              \n"
>-  : "+r"(src_ptr),    // %0
>-    "+r"(dst_ptr),    // %1
>-    "+r"(dst_width)   // %2
>-  :: "memory", "cc", "xmm0", "xmm1"
>-  );
>+  asm volatile(
>+      // 16 pixel loop.
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "movdqu    0x10(%0),%%xmm1                 \n"
>+      "lea       0x20(%0),%0                     \n"
>+      "psrlw     $0x8,%%xmm0                     \n"
>+      "psrlw     $0x8,%%xmm1                     \n"
>+      "packuswb  %%xmm1,%%xmm0                   \n"
>+      "movdqu    %%xmm0,(%1)                     \n"
>+      "lea       0x10(%1),%1                     \n"
>+      "sub       $0x10,%2                        \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_ptr),   // %0
>+        "+r"(dst_ptr),   // %1
>+        "+r"(dst_width)  // %2
>+        ::"memory",
>+        "cc", "xmm0", "xmm1");
> }
> 
>-void ScaleRowDown2Linear_SSSE3(const uint8* src_ptr,
>+void ScaleRowDown2Linear_SSSE3(const uint8_t* src_ptr,
>                                ptrdiff_t src_stride,
>-                               uint8* dst_ptr,
>+                               uint8_t* dst_ptr,
>                                int dst_width) {
>   (void)src_stride;
>-  asm volatile (
>-    "pcmpeqb    %%xmm4,%%xmm4                  \n"
>-    "psrlw      $0xf,%%xmm4                    \n"
>-    "packuswb   %%xmm4,%%xmm4                  \n"
>-    "pxor       %%xmm5,%%xmm5                  \n"
>-
>-    LABELALIGN
>-  "1:                                          \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    "movdqu    " MEMACCESS2(0x10, 0) ",%%xmm1  \n"
>-    "lea       " MEMLEA(0x20,0) ",%0           \n"
>-    "pmaddubsw  %%xmm4,%%xmm0                  \n"
>-    "pmaddubsw  %%xmm4,%%xmm1                  \n"
>-    "pavgw      %%xmm5,%%xmm0                  \n"
>-    "pavgw      %%xmm5,%%xmm1                  \n"
>-    "packuswb   %%xmm1,%%xmm0                  \n"
>-    "movdqu    %%xmm0," MEMACCESS(1) "         \n"
>-    "lea       " MEMLEA(0x10,1) ",%1           \n"
>-    "sub       $0x10,%2                        \n"
>-    "jg        1b                              \n"
>-  : "+r"(src_ptr),    // %0
>-    "+r"(dst_ptr),    // %1
>-    "+r"(dst_width)   // %2
>-  :: "memory", "cc", "xmm0", "xmm1", "xmm4", "xmm5"
>-  );
>+  asm volatile(
>+      "pcmpeqb    %%xmm4,%%xmm4                  \n"
>+      "psrlw      $0xf,%%xmm4                    \n"
>+      "packuswb   %%xmm4,%%xmm4                  \n"
>+      "pxor       %%xmm5,%%xmm5                  \n"
>+
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "movdqu    0x10(%0),%%xmm1                 \n"
>+      "lea       0x20(%0),%0                     \n"
>+      "pmaddubsw  %%xmm4,%%xmm0                  \n"
>+      "pmaddubsw  %%xmm4,%%xmm1                  \n"
>+      "pavgw      %%xmm5,%%xmm0                  \n"
>+      "pavgw      %%xmm5,%%xmm1                  \n"
>+      "packuswb   %%xmm1,%%xmm0                  \n"
>+      "movdqu    %%xmm0,(%1)                     \n"
>+      "lea       0x10(%1),%1                     \n"
>+      "sub       $0x10,%2                        \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_ptr),   // %0
>+        "+r"(dst_ptr),   // %1
>+        "+r"(dst_width)  // %2
>+        ::"memory",
>+        "cc", "xmm0", "xmm1", "xmm4", "xmm5");
> }
> 
>-void ScaleRowDown2Box_SSSE3(const uint8* src_ptr,
>+void ScaleRowDown2Box_SSSE3(const uint8_t* src_ptr,
>                             ptrdiff_t src_stride,
>-                            uint8* dst_ptr,
>+                            uint8_t* dst_ptr,
>                             int dst_width) {
>-  asm volatile (
>-    "pcmpeqb    %%xmm4,%%xmm4                  \n"
>-    "psrlw      $0xf,%%xmm4                    \n"
>-    "packuswb   %%xmm4,%%xmm4                  \n"
>-    "pxor       %%xmm5,%%xmm5                  \n"
>-
>-    LABELALIGN
>-  "1:                                          \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    "movdqu    " MEMACCESS2(0x10,0) ",%%xmm1   \n"
>-    MEMOPREG(movdqu,0x00,0,3,1,xmm2)           //  movdqu  (%0,%3,1),%%xmm2
>-    MEMOPREG(movdqu,0x10,0,3,1,xmm3)           //  movdqu  0x10(%0,%3,1),%%xmm3
>-    "lea       " MEMLEA(0x20,0) ",%0           \n"
>-    "pmaddubsw  %%xmm4,%%xmm0                  \n"
>-    "pmaddubsw  %%xmm4,%%xmm1                  \n"
>-    "pmaddubsw  %%xmm4,%%xmm2                  \n"
>-    "pmaddubsw  %%xmm4,%%xmm3                  \n"
>-    "paddw      %%xmm2,%%xmm0                  \n"
>-    "paddw      %%xmm3,%%xmm1                  \n"
>-    "psrlw      $0x1,%%xmm0                    \n"
>-    "psrlw      $0x1,%%xmm1                    \n"
>-    "pavgw      %%xmm5,%%xmm0                  \n"
>-    "pavgw      %%xmm5,%%xmm1                  \n"
>-    "packuswb   %%xmm1,%%xmm0                  \n"
>-    "movdqu    %%xmm0," MEMACCESS(1) "         \n"
>-    "lea       " MEMLEA(0x10,1) ",%1           \n"
>-    "sub       $0x10,%2                        \n"
>-    "jg        1b                              \n"
>-  : "+r"(src_ptr),    // %0
>-    "+r"(dst_ptr),    // %1
>-    "+r"(dst_width)   // %2
>-  : "r"((intptr_t)(src_stride))   // %3
>-  : "memory", "cc", NACL_R14
>-    "xmm0", "xmm1", "xmm2", "xmm3", "xmm5"
>-  );
>+  asm volatile(
>+      "pcmpeqb    %%xmm4,%%xmm4                  \n"
>+      "psrlw      $0xf,%%xmm4                    \n"
>+      "packuswb   %%xmm4,%%xmm4                  \n"
>+      "pxor       %%xmm5,%%xmm5                  \n"
>+
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "movdqu    0x10(%0),%%xmm1                 \n"
>+      "movdqu    0x00(%0,%3,1),%%xmm2            \n"
>+      "movdqu    0x10(%0,%3,1),%%xmm3            \n"
>+      "lea       0x20(%0),%0                     \n"
>+      "pmaddubsw  %%xmm4,%%xmm0                  \n"
>+      "pmaddubsw  %%xmm4,%%xmm1                  \n"
>+      "pmaddubsw  %%xmm4,%%xmm2                  \n"
>+      "pmaddubsw  %%xmm4,%%xmm3                  \n"
>+      "paddw      %%xmm2,%%xmm0                  \n"
>+      "paddw      %%xmm3,%%xmm1                  \n"
>+      "psrlw      $0x1,%%xmm0                    \n"
>+      "psrlw      $0x1,%%xmm1                    \n"
>+      "pavgw      %%xmm5,%%xmm0                  \n"
>+      "pavgw      %%xmm5,%%xmm1                  \n"
>+      "packuswb   %%xmm1,%%xmm0                  \n"
>+      "movdqu    %%xmm0,(%1)                     \n"
>+      "lea       0x10(%1),%1                     \n"
>+      "sub       $0x10,%2                        \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_ptr),               // %0
>+        "+r"(dst_ptr),               // %1
>+        "+r"(dst_width)              // %2
>+      : "r"((intptr_t)(src_stride))  // %3
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm5");
> }
> 
> #ifdef HAS_SCALEROWDOWN2_AVX2
>-void ScaleRowDown2_AVX2(const uint8* src_ptr,
>+void ScaleRowDown2_AVX2(const uint8_t* src_ptr,
>                         ptrdiff_t src_stride,
>-                        uint8* dst_ptr,
>+                        uint8_t* dst_ptr,
>                         int dst_width) {
>   (void)src_stride;
>-  asm volatile (
>-    LABELALIGN
>-  "1:                                          \n"
>-    "vmovdqu    " MEMACCESS(0) ",%%ymm0        \n"
>-    "vmovdqu    " MEMACCESS2(0x20,0) ",%%ymm1  \n"
>-    "lea        " MEMLEA(0x40,0) ",%0          \n"
>-    "vpsrlw     $0x8,%%ymm0,%%ymm0             \n"
>-    "vpsrlw     $0x8,%%ymm1,%%ymm1             \n"
>-    "vpackuswb  %%ymm1,%%ymm0,%%ymm0           \n"
>-    "vpermq     $0xd8,%%ymm0,%%ymm0            \n"
>-    "vmovdqu    %%ymm0," MEMACCESS(1) "        \n"
>-    "lea        " MEMLEA(0x20,1) ",%1          \n"
>-    "sub        $0x20,%2                       \n"
>-    "jg         1b                             \n"
>-    "vzeroupper                                \n"
>-  : "+r"(src_ptr),    // %0
>-    "+r"(dst_ptr),    // %1
>-    "+r"(dst_width)   // %2
>-  :: "memory", "cc", "xmm0", "xmm1"
>-  );
>+  asm volatile(
>+
>+      LABELALIGN
>+      "1:                                        \n"
>+      "vmovdqu    (%0),%%ymm0                    \n"
>+      "vmovdqu    0x20(%0),%%ymm1                \n"
>+      "lea        0x40(%0),%0                    \n"
>+      "vpsrlw     $0x8,%%ymm0,%%ymm0             \n"
>+      "vpsrlw     $0x8,%%ymm1,%%ymm1             \n"
>+      "vpackuswb  %%ymm1,%%ymm0,%%ymm0           \n"
>+      "vpermq     $0xd8,%%ymm0,%%ymm0            \n"
>+      "vmovdqu    %%ymm0,(%1)                    \n"
>+      "lea        0x20(%1),%1                    \n"
>+      "sub        $0x20,%2                       \n"
>+      "jg         1b                             \n"
>+      "vzeroupper                                \n"
>+      : "+r"(src_ptr),   // %0
>+        "+r"(dst_ptr),   // %1
>+        "+r"(dst_width)  // %2
>+        ::"memory",
>+        "cc", "xmm0", "xmm1");
> }
> 
>-void ScaleRowDown2Linear_AVX2(const uint8* src_ptr,
>+void ScaleRowDown2Linear_AVX2(const uint8_t* src_ptr,
>                               ptrdiff_t src_stride,
>-                              uint8* dst_ptr,
>+                              uint8_t* dst_ptr,
>                               int dst_width) {
>   (void)src_stride;
>-  asm volatile (
>-    "vpcmpeqb   %%ymm4,%%ymm4,%%ymm4           \n"
>-    "vpsrlw     $0xf,%%ymm4,%%ymm4             \n"
>-    "vpackuswb  %%ymm4,%%ymm4,%%ymm4           \n"
>-    "vpxor      %%ymm5,%%ymm5,%%ymm5           \n"
>-
>-    LABELALIGN
>-  "1:                                          \n"
>-    "vmovdqu    " MEMACCESS(0) ",%%ymm0        \n"
>-    "vmovdqu    " MEMACCESS2(0x20, 0) ",%%ymm1 \n"
>-    "lea        " MEMLEA(0x40,0) ",%0          \n"
>-    "vpmaddubsw %%ymm4,%%ymm0,%%ymm0           \n"
>-    "vpmaddubsw %%ymm4,%%ymm1,%%ymm1           \n"
>-    "vpavgw     %%ymm5,%%ymm0,%%ymm0           \n"
>-    "vpavgw     %%ymm5,%%ymm1,%%ymm1           \n"
>-    "vpackuswb  %%ymm1,%%ymm0,%%ymm0           \n"
>-    "vpermq     $0xd8,%%ymm0,%%ymm0            \n"
>-    "vmovdqu    %%ymm0," MEMACCESS(1) "        \n"
>-    "lea        " MEMLEA(0x20,1) ",%1          \n"
>-    "sub        $0x20,%2                       \n"
>-    "jg         1b                             \n"
>-    "vzeroupper                                \n"
>-  : "+r"(src_ptr),    // %0
>-    "+r"(dst_ptr),    // %1
>-    "+r"(dst_width)   // %2
>-  :: "memory", "cc", "xmm0", "xmm1", "xmm4", "xmm5"
>-  );
>+  asm volatile(
>+      "vpcmpeqb   %%ymm4,%%ymm4,%%ymm4           \n"
>+      "vpsrlw     $0xf,%%ymm4,%%ymm4             \n"
>+      "vpackuswb  %%ymm4,%%ymm4,%%ymm4           \n"
>+      "vpxor      %%ymm5,%%ymm5,%%ymm5           \n"
>+
>+      LABELALIGN
>+      "1:                                        \n"
>+      "vmovdqu    (%0),%%ymm0                    \n"
>+      "vmovdqu    0x20(%0),%%ymm1                \n"
>+      "lea        0x40(%0),%0                    \n"
>+      "vpmaddubsw %%ymm4,%%ymm0,%%ymm0           \n"
>+      "vpmaddubsw %%ymm4,%%ymm1,%%ymm1           \n"
>+      "vpavgw     %%ymm5,%%ymm0,%%ymm0           \n"
>+      "vpavgw     %%ymm5,%%ymm1,%%ymm1           \n"
>+      "vpackuswb  %%ymm1,%%ymm0,%%ymm0           \n"
>+      "vpermq     $0xd8,%%ymm0,%%ymm0            \n"
>+      "vmovdqu    %%ymm0,(%1)                    \n"
>+      "lea        0x20(%1),%1                    \n"
>+      "sub        $0x20,%2                       \n"
>+      "jg         1b                             \n"
>+      "vzeroupper                                \n"
>+      : "+r"(src_ptr),   // %0
>+        "+r"(dst_ptr),   // %1
>+        "+r"(dst_width)  // %2
>+        ::"memory",
>+        "cc", "xmm0", "xmm1", "xmm4", "xmm5");
> }
> 
>-void ScaleRowDown2Box_AVX2(const uint8* src_ptr,
>+void ScaleRowDown2Box_AVX2(const uint8_t* src_ptr,
>                            ptrdiff_t src_stride,
>-                           uint8* dst_ptr,
>+                           uint8_t* dst_ptr,
>                            int dst_width) {
>-  asm volatile (
>-    "vpcmpeqb   %%ymm4,%%ymm4,%%ymm4           \n"
>-    "vpsrlw     $0xf,%%ymm4,%%ymm4             \n"
>-    "vpackuswb  %%ymm4,%%ymm4,%%ymm4           \n"
>-    "vpxor      %%ymm5,%%ymm5,%%ymm5           \n"
>-
>-    LABELALIGN
>-  "1:                                          \n"
>-    "vmovdqu    " MEMACCESS(0) ",%%ymm0        \n"
>-    "vmovdqu    " MEMACCESS2(0x20,0) ",%%ymm1  \n"
>-    MEMOPREG(vmovdqu,0x00,0,3,1,ymm2)          //  vmovdqu  (%0,%3,1),%%ymm2
>-    MEMOPREG(vmovdqu,0x20,0,3,1,ymm3)          //  vmovdqu  0x20(%0,%3,1),%%ymm3
>-    "lea        " MEMLEA(0x40,0) ",%0          \n"
>-    "vpmaddubsw %%ymm4,%%ymm0,%%ymm0           \n"
>-    "vpmaddubsw %%ymm4,%%ymm1,%%ymm1           \n"
>-    "vpmaddubsw %%ymm4,%%ymm2,%%ymm2           \n"
>-    "vpmaddubsw %%ymm4,%%ymm3,%%ymm3           \n"
>-    "vpaddw     %%ymm2,%%ymm0,%%ymm0           \n"
>-    "vpaddw     %%ymm3,%%ymm1,%%ymm1           \n"
>-    "vpsrlw     $0x1,%%ymm0,%%ymm0             \n"
>-    "vpsrlw     $0x1,%%ymm1,%%ymm1             \n"
>-    "vpavgw     %%ymm5,%%ymm0,%%ymm0           \n"
>-    "vpavgw     %%ymm5,%%ymm1,%%ymm1           \n"
>-    "vpackuswb  %%ymm1,%%ymm0,%%ymm0           \n"
>-    "vpermq     $0xd8,%%ymm0,%%ymm0            \n"
>-    "vmovdqu    %%ymm0," MEMACCESS(1) "        \n"
>-    "lea        " MEMLEA(0x20,1) ",%1          \n"
>-    "sub        $0x20,%2                       \n"
>-    "jg         1b                             \n"
>-    "vzeroupper                                \n"
>-  : "+r"(src_ptr),    // %0
>-    "+r"(dst_ptr),    // %1
>-    "+r"(dst_width)   // %2
>-  : "r"((intptr_t)(src_stride))   // %3
>-  : "memory", "cc", NACL_R14
>-    "xmm0", "xmm1", "xmm2", "xmm3", "xmm5"
>-  );
>+  asm volatile(
>+      "vpcmpeqb   %%ymm4,%%ymm4,%%ymm4           \n"
>+      "vpsrlw     $0xf,%%ymm4,%%ymm4             \n"
>+      "vpackuswb  %%ymm4,%%ymm4,%%ymm4           \n"
>+      "vpxor      %%ymm5,%%ymm5,%%ymm5           \n"
>+
>+      LABELALIGN
>+      "1:                                        \n"
>+      "vmovdqu    (%0),%%ymm0                    \n"
>+      "vmovdqu    0x20(%0),%%ymm1                \n"
>+      "vmovdqu    0x00(%0,%3,1),%%ymm2           \n"
>+      "vmovdqu    0x20(%0,%3,1),%%ymm3           \n"
>+      "lea        0x40(%0),%0                    \n"
>+      "vpmaddubsw %%ymm4,%%ymm0,%%ymm0           \n"
>+      "vpmaddubsw %%ymm4,%%ymm1,%%ymm1           \n"
>+      "vpmaddubsw %%ymm4,%%ymm2,%%ymm2           \n"
>+      "vpmaddubsw %%ymm4,%%ymm3,%%ymm3           \n"
>+      "vpaddw     %%ymm2,%%ymm0,%%ymm0           \n"
>+      "vpaddw     %%ymm3,%%ymm1,%%ymm1           \n"
>+      "vpsrlw     $0x1,%%ymm0,%%ymm0             \n"
>+      "vpsrlw     $0x1,%%ymm1,%%ymm1             \n"
>+      "vpavgw     %%ymm5,%%ymm0,%%ymm0           \n"
>+      "vpavgw     %%ymm5,%%ymm1,%%ymm1           \n"
>+      "vpackuswb  %%ymm1,%%ymm0,%%ymm0           \n"
>+      "vpermq     $0xd8,%%ymm0,%%ymm0            \n"
>+      "vmovdqu    %%ymm0,(%1)                    \n"
>+      "lea        0x20(%1),%1                    \n"
>+      "sub        $0x20,%2                       \n"
>+      "jg         1b                             \n"
>+      "vzeroupper                                \n"
>+      : "+r"(src_ptr),               // %0
>+        "+r"(dst_ptr),               // %1
>+        "+r"(dst_width)              // %2
>+      : "r"((intptr_t)(src_stride))  // %3
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm5");
> }
> #endif  // HAS_SCALEROWDOWN2_AVX2
> 
>-void ScaleRowDown4_SSSE3(const uint8* src_ptr,
>+void ScaleRowDown4_SSSE3(const uint8_t* src_ptr,
>                          ptrdiff_t src_stride,
>-                         uint8* dst_ptr,
>+                         uint8_t* dst_ptr,
>                          int dst_width) {
>   (void)src_stride;
>-  asm volatile (
>-    "pcmpeqb   %%xmm5,%%xmm5                   \n"
>-    "psrld     $0x18,%%xmm5                    \n"
>-    "pslld     $0x10,%%xmm5                    \n"
>-
>-    LABELALIGN
>-  "1:                                          \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    "movdqu    " MEMACCESS2(0x10,0) ",%%xmm1   \n"
>-    "lea       " MEMLEA(0x20,0) ",%0           \n"
>-    "pand      %%xmm5,%%xmm0                   \n"
>-    "pand      %%xmm5,%%xmm1                   \n"
>-    "packuswb  %%xmm1,%%xmm0                   \n"
>-    "psrlw     $0x8,%%xmm0                     \n"
>-    "packuswb  %%xmm0,%%xmm0                   \n"
>-    "movq      %%xmm0," MEMACCESS(1) "         \n"
>-    "lea       " MEMLEA(0x8,1) ",%1            \n"
>-    "sub       $0x8,%2                         \n"
>-    "jg        1b                              \n"
>-  : "+r"(src_ptr),    // %0
>-    "+r"(dst_ptr),    // %1
>-    "+r"(dst_width)   // %2
>-  :: "memory", "cc", "xmm0", "xmm1", "xmm5"
>-  );
>+  asm volatile(
>+      "pcmpeqb   %%xmm5,%%xmm5                   \n"
>+      "psrld     $0x18,%%xmm5                    \n"
>+      "pslld     $0x10,%%xmm5                    \n"
>+
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "movdqu    0x10(%0),%%xmm1                 \n"
>+      "lea       0x20(%0),%0                     \n"
>+      "pand      %%xmm5,%%xmm0                   \n"
>+      "pand      %%xmm5,%%xmm1                   \n"
>+      "packuswb  %%xmm1,%%xmm0                   \n"
>+      "psrlw     $0x8,%%xmm0                     \n"
>+      "packuswb  %%xmm0,%%xmm0                   \n"
>+      "movq      %%xmm0,(%1)                     \n"
>+      "lea       0x8(%1),%1                      \n"
>+      "sub       $0x8,%2                         \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_ptr),   // %0
>+        "+r"(dst_ptr),   // %1
>+        "+r"(dst_width)  // %2
>+        ::"memory",
>+        "cc", "xmm0", "xmm1", "xmm5");
> }
> 
>-void ScaleRowDown4Box_SSSE3(const uint8* src_ptr,
>+void ScaleRowDown4Box_SSSE3(const uint8_t* src_ptr,
>                             ptrdiff_t src_stride,
>-                            uint8* dst_ptr,
>+                            uint8_t* dst_ptr,
>                             int dst_width) {
>   intptr_t stridex3;
>-  asm volatile (
>-    "pcmpeqb    %%xmm4,%%xmm4                  \n"
>-    "psrlw      $0xf,%%xmm4                    \n"
>-    "movdqa     %%xmm4,%%xmm5                  \n"
>-    "packuswb   %%xmm4,%%xmm4                  \n"
>-    "psllw      $0x3,%%xmm5                    \n"
>-    "lea       " MEMLEA4(0x00,4,4,2) ",%3      \n"
>-
>-    LABELALIGN
>-  "1:                                          \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    "movdqu    " MEMACCESS2(0x10,0) ",%%xmm1   \n"
>-    MEMOPREG(movdqu,0x00,0,4,1,xmm2)           //  movdqu  (%0,%4,1),%%xmm2
>-    MEMOPREG(movdqu,0x10,0,4,1,xmm3)           //  movdqu  0x10(%0,%4,1),%%xmm3
>-    "pmaddubsw  %%xmm4,%%xmm0                  \n"
>-    "pmaddubsw  %%xmm4,%%xmm1                  \n"
>-    "pmaddubsw  %%xmm4,%%xmm2                  \n"
>-    "pmaddubsw  %%xmm4,%%xmm3                  \n"
>-    "paddw      %%xmm2,%%xmm0                  \n"
>-    "paddw      %%xmm3,%%xmm1                  \n"
>-    MEMOPREG(movdqu,0x00,0,4,2,xmm2)           //  movdqu  (%0,%4,2),%%xmm2
>-    MEMOPREG(movdqu,0x10,0,4,2,xmm3)           //  movdqu  0x10(%0,%4,2),%%xmm3
>-    "pmaddubsw  %%xmm4,%%xmm2                  \n"
>-    "pmaddubsw  %%xmm4,%%xmm3                  \n"
>-    "paddw      %%xmm2,%%xmm0                  \n"
>-    "paddw      %%xmm3,%%xmm1                  \n"
>-    MEMOPREG(movdqu,0x00,0,3,1,xmm2)           //  movdqu  (%0,%3,1),%%xmm2
>-    MEMOPREG(movdqu,0x10,0,3,1,xmm3)           //  movdqu  0x10(%0,%3,1),%%xmm3
>-    "lea       " MEMLEA(0x20,0) ",%0           \n"
>-    "pmaddubsw  %%xmm4,%%xmm2                  \n"
>-    "pmaddubsw  %%xmm4,%%xmm3                  \n"
>-    "paddw      %%xmm2,%%xmm0                  \n"
>-    "paddw      %%xmm3,%%xmm1                  \n"
>-    "phaddw     %%xmm1,%%xmm0                  \n"
>-    "paddw      %%xmm5,%%xmm0                  \n"
>-    "psrlw      $0x4,%%xmm0                    \n"
>-    "packuswb   %%xmm0,%%xmm0                  \n"
>-    "movq      %%xmm0," MEMACCESS(1) "         \n"
>-    "lea       " MEMLEA(0x8,1) ",%1            \n"
>-    "sub       $0x8,%2                         \n"
>-    "jg        1b                              \n"
>-  : "+r"(src_ptr),     // %0
>-    "+r"(dst_ptr),     // %1
>-    "+r"(dst_width),   // %2
>-    "=&r"(stridex3)    // %3
>-  : "r"((intptr_t)(src_stride))    // %4
>-  : "memory", "cc", NACL_R14
>-    "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5"
>-  );
>+  asm volatile(
>+      "pcmpeqb    %%xmm4,%%xmm4                  \n"
>+      "psrlw      $0xf,%%xmm4                    \n"
>+      "movdqa     %%xmm4,%%xmm5                  \n"
>+      "packuswb   %%xmm4,%%xmm4                  \n"
>+      "psllw      $0x3,%%xmm5                    \n"
>+      "lea       0x00(%4,%4,2),%3                \n"
>+
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "movdqu    0x10(%0),%%xmm1                 \n"
>+      "movdqu    0x00(%0,%4,1),%%xmm2            \n"
>+      "movdqu    0x10(%0,%4,1),%%xmm3            \n"
>+      "pmaddubsw  %%xmm4,%%xmm0                  \n"
>+      "pmaddubsw  %%xmm4,%%xmm1                  \n"
>+      "pmaddubsw  %%xmm4,%%xmm2                  \n"
>+      "pmaddubsw  %%xmm4,%%xmm3                  \n"
>+      "paddw      %%xmm2,%%xmm0                  \n"
>+      "paddw      %%xmm3,%%xmm1                  \n"
>+      "movdqu    0x00(%0,%4,2),%%xmm2            \n"
>+      "movdqu    0x10(%0,%4,2),%%xmm3            \n"
>+      "pmaddubsw  %%xmm4,%%xmm2                  \n"
>+      "pmaddubsw  %%xmm4,%%xmm3                  \n"
>+      "paddw      %%xmm2,%%xmm0                  \n"
>+      "paddw      %%xmm3,%%xmm1                  \n"
>+      "movdqu    0x00(%0,%3,1),%%xmm2            \n"
>+      "movdqu    0x10(%0,%3,1),%%xmm3            \n"
>+      "lea       0x20(%0),%0                     \n"
>+      "pmaddubsw  %%xmm4,%%xmm2                  \n"
>+      "pmaddubsw  %%xmm4,%%xmm3                  \n"
>+      "paddw      %%xmm2,%%xmm0                  \n"
>+      "paddw      %%xmm3,%%xmm1                  \n"
>+      "phaddw     %%xmm1,%%xmm0                  \n"
>+      "paddw      %%xmm5,%%xmm0                  \n"
>+      "psrlw      $0x4,%%xmm0                    \n"
>+      "packuswb   %%xmm0,%%xmm0                  \n"
>+      "movq      %%xmm0,(%1)                     \n"
>+      "lea       0x8(%1),%1                      \n"
>+      "sub       $0x8,%2                         \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_ptr),               // %0
>+        "+r"(dst_ptr),               // %1
>+        "+r"(dst_width),             // %2
>+        "=&r"(stridex3)              // %3
>+      : "r"((intptr_t)(src_stride))  // %4
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5");
> }
> 
> #ifdef HAS_SCALEROWDOWN4_AVX2
>-void ScaleRowDown4_AVX2(const uint8* src_ptr,
>+void ScaleRowDown4_AVX2(const uint8_t* src_ptr,
>                         ptrdiff_t src_stride,
>-                        uint8* dst_ptr,
>+                        uint8_t* dst_ptr,
>                         int dst_width) {
>   (void)src_stride;
>-  asm volatile (
>-    "vpcmpeqb   %%ymm5,%%ymm5,%%ymm5           \n"
>-    "vpsrld     $0x18,%%ymm5,%%ymm5            \n"
>-    "vpslld     $0x10,%%ymm5,%%ymm5            \n"
>-    LABELALIGN
>-  "1:                                          \n"
>-    "vmovdqu    " MEMACCESS(0) ",%%ymm0        \n"
>-    "vmovdqu    " MEMACCESS2(0x20,0) ",%%ymm1  \n"
>-    "lea        " MEMLEA(0x40,0) ",%0          \n"
>-    "vpand      %%ymm5,%%ymm0,%%ymm0           \n"
>-    "vpand      %%ymm5,%%ymm1,%%ymm1           \n"
>-    "vpackuswb  %%ymm1,%%ymm0,%%ymm0           \n"
>-    "vpermq     $0xd8,%%ymm0,%%ymm0            \n"
>-    "vpsrlw     $0x8,%%ymm0,%%ymm0             \n"
>-    "vpackuswb  %%ymm0,%%ymm0,%%ymm0           \n"
>-    "vpermq     $0xd8,%%ymm0,%%ymm0            \n"
>-    "vmovdqu    %%xmm0," MEMACCESS(1) "        \n"
>-    "lea        " MEMLEA(0x10,1) ",%1          \n"
>-    "sub        $0x10,%2                       \n"
>-    "jg         1b                             \n"
>-    "vzeroupper                                \n"
>-  : "+r"(src_ptr),    // %0
>-    "+r"(dst_ptr),    // %1
>-    "+r"(dst_width)   // %2
>-  :: "memory", "cc", "xmm0", "xmm1", "xmm5"
>-  );
>+  asm volatile(
>+      "vpcmpeqb   %%ymm5,%%ymm5,%%ymm5           \n"
>+      "vpsrld     $0x18,%%ymm5,%%ymm5            \n"
>+      "vpslld     $0x10,%%ymm5,%%ymm5            \n"
>+
>+      LABELALIGN
>+      "1:                                        \n"
>+      "vmovdqu    (%0),%%ymm0                    \n"
>+      "vmovdqu    0x20(%0),%%ymm1                \n"
>+      "lea        0x40(%0),%0                    \n"
>+      "vpand      %%ymm5,%%ymm0,%%ymm0           \n"
>+      "vpand      %%ymm5,%%ymm1,%%ymm1           \n"
>+      "vpackuswb  %%ymm1,%%ymm0,%%ymm0           \n"
>+      "vpermq     $0xd8,%%ymm0,%%ymm0            \n"
>+      "vpsrlw     $0x8,%%ymm0,%%ymm0             \n"
>+      "vpackuswb  %%ymm0,%%ymm0,%%ymm0           \n"
>+      "vpermq     $0xd8,%%ymm0,%%ymm0            \n"
>+      "vmovdqu    %%xmm0,(%1)                    \n"
>+      "lea        0x10(%1),%1                    \n"
>+      "sub        $0x10,%2                       \n"
>+      "jg         1b                             \n"
>+      "vzeroupper                                \n"
>+      : "+r"(src_ptr),   // %0
>+        "+r"(dst_ptr),   // %1
>+        "+r"(dst_width)  // %2
>+        ::"memory",
>+        "cc", "xmm0", "xmm1", "xmm5");
> }
> 
>-void ScaleRowDown4Box_AVX2(const uint8* src_ptr,
>+void ScaleRowDown4Box_AVX2(const uint8_t* src_ptr,
>                            ptrdiff_t src_stride,
>-                           uint8* dst_ptr,
>+                           uint8_t* dst_ptr,
>                            int dst_width) {
>-  asm volatile (
>-    "vpcmpeqb   %%ymm4,%%ymm4,%%ymm4           \n"
>-    "vpsrlw     $0xf,%%ymm4,%%ymm4             \n"
>-    "vpsllw     $0x3,%%ymm4,%%ymm5             \n"
>-    "vpackuswb  %%ymm4,%%ymm4,%%ymm4           \n"
>-
>-    LABELALIGN
>-  "1:                                          \n"
>-    "vmovdqu    " MEMACCESS(0) ",%%ymm0        \n"
>-    "vmovdqu    " MEMACCESS2(0x20,0) ",%%ymm1  \n"
>-    MEMOPREG(vmovdqu,0x00,0,3,1,ymm2)          //  vmovdqu  (%0,%3,1),%%ymm2
>-    MEMOPREG(vmovdqu,0x20,0,3,1,ymm3)          //  vmovdqu  0x20(%0,%3,1),%%ymm3
>-    "vpmaddubsw %%ymm4,%%ymm0,%%ymm0           \n"
>-    "vpmaddubsw %%ymm4,%%ymm1,%%ymm1           \n"
>-    "vpmaddubsw %%ymm4,%%ymm2,%%ymm2           \n"
>-    "vpmaddubsw %%ymm4,%%ymm3,%%ymm3           \n"
>-    "vpaddw     %%ymm2,%%ymm0,%%ymm0           \n"
>-    "vpaddw     %%ymm3,%%ymm1,%%ymm1           \n"
>-    MEMOPREG(vmovdqu,0x00,0,3,2,ymm2)          //  vmovdqu  (%0,%3,2),%%ymm2
>-    MEMOPREG(vmovdqu,0x20,0,3,2,ymm3)          //  vmovdqu  0x20(%0,%3,2),%%ymm3
>-    "vpmaddubsw %%ymm4,%%ymm2,%%ymm2           \n"
>-    "vpmaddubsw %%ymm4,%%ymm3,%%ymm3           \n"
>-    "vpaddw     %%ymm2,%%ymm0,%%ymm0           \n"
>-    "vpaddw     %%ymm3,%%ymm1,%%ymm1           \n"
>-    MEMOPREG(vmovdqu,0x00,0,4,1,ymm2)          //  vmovdqu  (%0,%4,1),%%ymm2
>-    MEMOPREG(vmovdqu,0x20,0,4,1,ymm3)          //  vmovdqu  0x20(%0,%4,1),%%ymm3
>-    "lea        " MEMLEA(0x40,0) ",%0          \n"
>-    "vpmaddubsw %%ymm4,%%ymm2,%%ymm2           \n"
>-    "vpmaddubsw %%ymm4,%%ymm3,%%ymm3           \n"
>-    "vpaddw     %%ymm2,%%ymm0,%%ymm0           \n"
>-    "vpaddw     %%ymm3,%%ymm1,%%ymm1           \n"
>-    "vphaddw    %%ymm1,%%ymm0,%%ymm0           \n"
>-    "vpermq     $0xd8,%%ymm0,%%ymm0            \n"
>-    "vpaddw     %%ymm5,%%ymm0,%%ymm0           \n"
>-    "vpsrlw     $0x4,%%ymm0,%%ymm0             \n"
>-    "vpackuswb  %%ymm0,%%ymm0,%%ymm0           \n"
>-    "vpermq     $0xd8,%%ymm0,%%ymm0            \n"
>-    "vmovdqu    %%xmm0," MEMACCESS(1) "        \n"
>-    "lea        " MEMLEA(0x10,1) ",%1          \n"
>-    "sub        $0x10,%2                       \n"
>-    "jg         1b                             \n"
>-    "vzeroupper                                \n"
>-  : "+r"(src_ptr),    // %0
>-    "+r"(dst_ptr),    // %1
>-    "+r"(dst_width)   // %2
>-  : "r"((intptr_t)(src_stride)),  // %3
>-    "r"((intptr_t)(src_stride * 3))   // %4
>-  : "memory", "cc", NACL_R14
>-    "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5"
>-  );
>+  asm volatile(
>+      "vpcmpeqb   %%ymm4,%%ymm4,%%ymm4           \n"
>+      "vpsrlw     $0xf,%%ymm4,%%ymm4             \n"
>+      "vpsllw     $0x3,%%ymm4,%%ymm5             \n"
>+      "vpackuswb  %%ymm4,%%ymm4,%%ymm4           \n"
>+
>+      LABELALIGN
>+      "1:                                        \n"
>+      "vmovdqu    (%0),%%ymm0                    \n"
>+      "vmovdqu    0x20(%0),%%ymm1                \n"
>+      "vmovdqu    0x00(%0,%3,1),%%ymm2           \n"
>+      "vmovdqu    0x20(%0,%3,1),%%ymm3           \n"
>+      "vpmaddubsw %%ymm4,%%ymm0,%%ymm0           \n"
>+      "vpmaddubsw %%ymm4,%%ymm1,%%ymm1           \n"
>+      "vpmaddubsw %%ymm4,%%ymm2,%%ymm2           \n"
>+      "vpmaddubsw %%ymm4,%%ymm3,%%ymm3           \n"
>+      "vpaddw     %%ymm2,%%ymm0,%%ymm0           \n"
>+      "vpaddw     %%ymm3,%%ymm1,%%ymm1           \n"
>+      "vmovdqu    0x00(%0,%3,2),%%ymm2           \n"
>+      "vmovdqu    0x20(%0,%3,2),%%ymm3           \n"
>+      "vpmaddubsw %%ymm4,%%ymm2,%%ymm2           \n"
>+      "vpmaddubsw %%ymm4,%%ymm3,%%ymm3           \n"
>+      "vpaddw     %%ymm2,%%ymm0,%%ymm0           \n"
>+      "vpaddw     %%ymm3,%%ymm1,%%ymm1           \n"
>+      "vmovdqu    0x00(%0,%4,1),%%ymm2           \n"
>+      "vmovdqu    0x20(%0,%4,1),%%ymm3           \n"
>+      "lea        0x40(%0),%0                    \n"
>+      "vpmaddubsw %%ymm4,%%ymm2,%%ymm2           \n"
>+      "vpmaddubsw %%ymm4,%%ymm3,%%ymm3           \n"
>+      "vpaddw     %%ymm2,%%ymm0,%%ymm0           \n"
>+      "vpaddw     %%ymm3,%%ymm1,%%ymm1           \n"
>+      "vphaddw    %%ymm1,%%ymm0,%%ymm0           \n"
>+      "vpermq     $0xd8,%%ymm0,%%ymm0            \n"
>+      "vpaddw     %%ymm5,%%ymm0,%%ymm0           \n"
>+      "vpsrlw     $0x4,%%ymm0,%%ymm0             \n"
>+      "vpackuswb  %%ymm0,%%ymm0,%%ymm0           \n"
>+      "vpermq     $0xd8,%%ymm0,%%ymm0            \n"
>+      "vmovdqu    %%xmm0,(%1)                    \n"
>+      "lea        0x10(%1),%1                    \n"
>+      "sub        $0x10,%2                       \n"
>+      "jg         1b                             \n"
>+      "vzeroupper                                \n"
>+      : "+r"(src_ptr),                   // %0
>+        "+r"(dst_ptr),                   // %1
>+        "+r"(dst_width)                  // %2
>+      : "r"((intptr_t)(src_stride)),     // %3
>+        "r"((intptr_t)(src_stride * 3))  // %4
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5");
> }
> #endif  // HAS_SCALEROWDOWN4_AVX2
> 
>-void ScaleRowDown34_SSSE3(const uint8* src_ptr,
>+void ScaleRowDown34_SSSE3(const uint8_t* src_ptr,
>                           ptrdiff_t src_stride,
>-                          uint8* dst_ptr,
>+                          uint8_t* dst_ptr,
>                           int dst_width) {
>   (void)src_stride;
>   asm volatile(
>@@ -488,33 +484,34 @@ void ScaleRowDown34_SSSE3(const uint8* src_ptr,
>         "m"(kShuf1),  // %1
>         "m"(kShuf2)   // %2
>       );
>-  asm volatile (
>-    LABELALIGN
>-  "1:                                          \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    "movdqu    " MEMACCESS2(0x10,0) ",%%xmm2   \n"
>-    "lea       " MEMLEA(0x20,0) ",%0           \n"
>-    "movdqa    %%xmm2,%%xmm1                   \n"
>-    "palignr   $0x8,%%xmm0,%%xmm1              \n"
>-    "pshufb    %%xmm3,%%xmm0                   \n"
>-    "pshufb    %%xmm4,%%xmm1                   \n"
>-    "pshufb    %%xmm5,%%xmm2                   \n"
>-    "movq      %%xmm0," MEMACCESS(1) "         \n"
>-    "movq      %%xmm1," MEMACCESS2(0x8,1) "    \n"
>-    "movq      %%xmm2," MEMACCESS2(0x10,1) "   \n"
>-    "lea       " MEMLEA(0x18,1) ",%1           \n"
>-    "sub       $0x18,%2                        \n"
>-    "jg        1b                              \n"
>-  : "+r"(src_ptr),   // %0
>-    "+r"(dst_ptr),   // %1
>-    "+r"(dst_width)  // %2
>-  :: "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5"
>-  );
>+  asm volatile(
>+
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "movdqu    0x10(%0),%%xmm2                 \n"
>+      "lea       0x20(%0),%0                     \n"
>+      "movdqa    %%xmm2,%%xmm1                   \n"
>+      "palignr   $0x8,%%xmm0,%%xmm1              \n"
>+      "pshufb    %%xmm3,%%xmm0                   \n"
>+      "pshufb    %%xmm4,%%xmm1                   \n"
>+      "pshufb    %%xmm5,%%xmm2                   \n"
>+      "movq      %%xmm0,(%1)                     \n"
>+      "movq      %%xmm1,0x8(%1)                  \n"
>+      "movq      %%xmm2,0x10(%1)                 \n"
>+      "lea       0x18(%1),%1                     \n"
>+      "sub       $0x18,%2                        \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_ptr),   // %0
>+        "+r"(dst_ptr),   // %1
>+        "+r"(dst_width)  // %2
>+        ::"memory",
>+        "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5");
> }
> 
>-void ScaleRowDown34_1_Box_SSSE3(const uint8* src_ptr,
>+void ScaleRowDown34_1_Box_SSSE3(const uint8_t* src_ptr,
>                                 ptrdiff_t src_stride,
>-                                uint8* dst_ptr,
>+                                uint8_t* dst_ptr,
>                                 int dst_width) {
>   asm volatile(
>       "movdqa    %0,%%xmm2                       \n"  // kShuf01
>@@ -534,53 +531,53 @@ void ScaleRowDown34_1_Box_SSSE3(const uint8* src_ptr,
>         "m"(kMadd11),  // %1
>         "m"(kRound34)  // %2
>       );
>-  asm volatile (
>-    LABELALIGN
>-  "1:                                          \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm6         \n"
>-    MEMOPREG(movdqu,0x00,0,3,1,xmm7)           //  movdqu  (%0,%3),%%xmm7
>-    "pavgb     %%xmm7,%%xmm6                   \n"
>-    "pshufb    %%xmm2,%%xmm6                   \n"
>-    "pmaddubsw %%xmm5,%%xmm6                   \n"
>-    "paddsw    %%xmm1,%%xmm6                   \n"
>-    "psrlw     $0x2,%%xmm6                     \n"
>-    "packuswb  %%xmm6,%%xmm6                   \n"
>-    "movq      %%xmm6," MEMACCESS(1) "         \n"
>-    "movdqu    " MEMACCESS2(0x8,0) ",%%xmm6    \n"
>-    MEMOPREG(movdqu,0x8,0,3,1,xmm7)            //  movdqu  0x8(%0,%3),%%xmm7
>-    "pavgb     %%xmm7,%%xmm6                   \n"
>-    "pshufb    %%xmm3,%%xmm6                   \n"
>-    "pmaddubsw %%xmm0,%%xmm6                   \n"
>-    "paddsw    %%xmm1,%%xmm6                   \n"
>-    "psrlw     $0x2,%%xmm6                     \n"
>-    "packuswb  %%xmm6,%%xmm6                   \n"
>-    "movq      %%xmm6," MEMACCESS2(0x8,1) "    \n"
>-    "movdqu    " MEMACCESS2(0x10,0) ",%%xmm6   \n"
>-    MEMOPREG(movdqu,0x10,0,3,1,xmm7)           //  movdqu  0x10(%0,%3),%%xmm7
>-    "lea       " MEMLEA(0x20,0) ",%0           \n"
>-    "pavgb     %%xmm7,%%xmm6                   \n"
>-    "pshufb    %%xmm4,%%xmm6                   \n"
>-    "pmaddubsw %4,%%xmm6                       \n"
>-    "paddsw    %%xmm1,%%xmm6                   \n"
>-    "psrlw     $0x2,%%xmm6                     \n"
>-    "packuswb  %%xmm6,%%xmm6                   \n"
>-    "movq      %%xmm6," MEMACCESS2(0x10,1) "   \n"
>-    "lea       " MEMLEA(0x18,1) ",%1           \n"
>-    "sub       $0x18,%2                        \n"
>-    "jg        1b                              \n"
>-  : "+r"(src_ptr),   // %0
>-    "+r"(dst_ptr),   // %1
>-    "+r"(dst_width)  // %2
>-  : "r"((intptr_t)(src_stride)),  // %3
>-    "m"(kMadd21)     // %4
>-  : "memory", "cc", NACL_R14
>-    "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6", "xmm7"
>-  );
>+  asm volatile(
>+
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm6                     \n"
>+      "movdqu    0x00(%0,%3,1),%%xmm7            \n"
>+      "pavgb     %%xmm7,%%xmm6                   \n"
>+      "pshufb    %%xmm2,%%xmm6                   \n"
>+      "pmaddubsw %%xmm5,%%xmm6                   \n"
>+      "paddsw    %%xmm1,%%xmm6                   \n"
>+      "psrlw     $0x2,%%xmm6                     \n"
>+      "packuswb  %%xmm6,%%xmm6                   \n"
>+      "movq      %%xmm6,(%1)                     \n"
>+      "movdqu    0x8(%0),%%xmm6                  \n"
>+      "movdqu    0x8(%0,%3,1),%%xmm7             \n"
>+      "pavgb     %%xmm7,%%xmm6                   \n"
>+      "pshufb    %%xmm3,%%xmm6                   \n"
>+      "pmaddubsw %%xmm0,%%xmm6                   \n"
>+      "paddsw    %%xmm1,%%xmm6                   \n"
>+      "psrlw     $0x2,%%xmm6                     \n"
>+      "packuswb  %%xmm6,%%xmm6                   \n"
>+      "movq      %%xmm6,0x8(%1)                  \n"
>+      "movdqu    0x10(%0),%%xmm6                 \n"
>+      "movdqu    0x10(%0,%3,1),%%xmm7            \n"
>+      "lea       0x20(%0),%0                     \n"
>+      "pavgb     %%xmm7,%%xmm6                   \n"
>+      "pshufb    %%xmm4,%%xmm6                   \n"
>+      "pmaddubsw %4,%%xmm6                       \n"
>+      "paddsw    %%xmm1,%%xmm6                   \n"
>+      "psrlw     $0x2,%%xmm6                     \n"
>+      "packuswb  %%xmm6,%%xmm6                   \n"
>+      "movq      %%xmm6,0x10(%1)                 \n"
>+      "lea       0x18(%1),%1                     \n"
>+      "sub       $0x18,%2                        \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_ptr),                // %0
>+        "+r"(dst_ptr),                // %1
>+        "+r"(dst_width)               // %2
>+      : "r"((intptr_t)(src_stride)),  // %3
>+        "m"(kMadd21)                  // %4
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6",
>+        "xmm7");
> }
> 
>-void ScaleRowDown34_0_Box_SSSE3(const uint8* src_ptr,
>+void ScaleRowDown34_0_Box_SSSE3(const uint8_t* src_ptr,
>                                 ptrdiff_t src_stride,
>-                                uint8* dst_ptr,
>+                                uint8_t* dst_ptr,
>                                 int dst_width) {
>   asm volatile(
>       "movdqa    %0,%%xmm2                       \n"  // kShuf01
>@@ -601,88 +598,87 @@ void ScaleRowDown34_0_Box_SSSE3(const uint8* src_ptr,
>         "m"(kRound34)  // %2
>       );
> 
>-  asm volatile (
>-    LABELALIGN
>-  "1:                                          \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm6         \n"
>-    MEMOPREG(movdqu,0x00,0,3,1,xmm7)           //  movdqu  (%0,%3,1),%%xmm7
>-    "pavgb     %%xmm6,%%xmm7                   \n"
>-    "pavgb     %%xmm7,%%xmm6                   \n"
>-    "pshufb    %%xmm2,%%xmm6                   \n"
>-    "pmaddubsw %%xmm5,%%xmm6                   \n"
>-    "paddsw    %%xmm1,%%xmm6                   \n"
>-    "psrlw     $0x2,%%xmm6                     \n"
>-    "packuswb  %%xmm6,%%xmm6                   \n"
>-    "movq      %%xmm6," MEMACCESS(1) "         \n"
>-    "movdqu    " MEMACCESS2(0x8,0) ",%%xmm6    \n"
>-    MEMOPREG(movdqu,0x8,0,3,1,xmm7)            //  movdqu  0x8(%0,%3,1),%%xmm7
>-    "pavgb     %%xmm6,%%xmm7                   \n"
>-    "pavgb     %%xmm7,%%xmm6                   \n"
>-    "pshufb    %%xmm3,%%xmm6                   \n"
>-    "pmaddubsw %%xmm0,%%xmm6                   \n"
>-    "paddsw    %%xmm1,%%xmm6                   \n"
>-    "psrlw     $0x2,%%xmm6                     \n"
>-    "packuswb  %%xmm6,%%xmm6                   \n"
>-    "movq      %%xmm6," MEMACCESS2(0x8,1) "    \n"
>-    "movdqu    " MEMACCESS2(0x10,0) ",%%xmm6   \n"
>-    MEMOPREG(movdqu,0x10,0,3,1,xmm7)           //  movdqu  0x10(%0,%3,1),%%xmm7
>-    "lea       " MEMLEA(0x20,0) ",%0           \n"
>-    "pavgb     %%xmm6,%%xmm7                   \n"
>-    "pavgb     %%xmm7,%%xmm6                   \n"
>-    "pshufb    %%xmm4,%%xmm6                   \n"
>-    "pmaddubsw %4,%%xmm6                       \n"
>-    "paddsw    %%xmm1,%%xmm6                   \n"
>-    "psrlw     $0x2,%%xmm6                     \n"
>-    "packuswb  %%xmm6,%%xmm6                   \n"
>-    "movq      %%xmm6," MEMACCESS2(0x10,1) "   \n"
>-    "lea       " MEMLEA(0x18,1) ",%1           \n"
>-    "sub       $0x18,%2                        \n"
>-    "jg        1b                              \n"
>-    : "+r"(src_ptr),   // %0
>-      "+r"(dst_ptr),   // %1
>-      "+r"(dst_width)  // %2
>-    : "r"((intptr_t)(src_stride)),  // %3
>-      "m"(kMadd21)     // %4
>-    : "memory", "cc", NACL_R14
>-      "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6", "xmm7"
>-  );
>+  asm volatile(
>+
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm6                     \n"
>+      "movdqu    0x00(%0,%3,1),%%xmm7            \n"
>+      "pavgb     %%xmm6,%%xmm7                   \n"
>+      "pavgb     %%xmm7,%%xmm6                   \n"
>+      "pshufb    %%xmm2,%%xmm6                   \n"
>+      "pmaddubsw %%xmm5,%%xmm6                   \n"
>+      "paddsw    %%xmm1,%%xmm6                   \n"
>+      "psrlw     $0x2,%%xmm6                     \n"
>+      "packuswb  %%xmm6,%%xmm6                   \n"
>+      "movq      %%xmm6,(%1)                     \n"
>+      "movdqu    0x8(%0),%%xmm6                  \n"
>+      "movdqu    0x8(%0,%3,1),%%xmm7             \n"
>+      "pavgb     %%xmm6,%%xmm7                   \n"
>+      "pavgb     %%xmm7,%%xmm6                   \n"
>+      "pshufb    %%xmm3,%%xmm6                   \n"
>+      "pmaddubsw %%xmm0,%%xmm6                   \n"
>+      "paddsw    %%xmm1,%%xmm6                   \n"
>+      "psrlw     $0x2,%%xmm6                     \n"
>+      "packuswb  %%xmm6,%%xmm6                   \n"
>+      "movq      %%xmm6,0x8(%1)                  \n"
>+      "movdqu    0x10(%0),%%xmm6                 \n"
>+      "movdqu    0x10(%0,%3,1),%%xmm7            \n"
>+      "lea       0x20(%0),%0                     \n"
>+      "pavgb     %%xmm6,%%xmm7                   \n"
>+      "pavgb     %%xmm7,%%xmm6                   \n"
>+      "pshufb    %%xmm4,%%xmm6                   \n"
>+      "pmaddubsw %4,%%xmm6                       \n"
>+      "paddsw    %%xmm1,%%xmm6                   \n"
>+      "psrlw     $0x2,%%xmm6                     \n"
>+      "packuswb  %%xmm6,%%xmm6                   \n"
>+      "movq      %%xmm6,0x10(%1)                 \n"
>+      "lea       0x18(%1),%1                     \n"
>+      "sub       $0x18,%2                        \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_ptr),                // %0
>+        "+r"(dst_ptr),                // %1
>+        "+r"(dst_width)               // %2
>+      : "r"((intptr_t)(src_stride)),  // %3
>+        "m"(kMadd21)                  // %4
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6",
>+        "xmm7");
> }
> 
>-void ScaleRowDown38_SSSE3(const uint8* src_ptr,
>+void ScaleRowDown38_SSSE3(const uint8_t* src_ptr,
>                           ptrdiff_t src_stride,
>-                          uint8* dst_ptr,
>+                          uint8_t* dst_ptr,
>                           int dst_width) {
>   (void)src_stride;
>-  asm volatile (
>-    "movdqa    %3,%%xmm4                       \n"
>-    "movdqa    %4,%%xmm5                       \n"
>-
>-    LABELALIGN
>-  "1:                                          \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    "movdqu    " MEMACCESS2(0x10,0) ",%%xmm1   \n"
>-    "lea       " MEMLEA(0x20,0) ",%0           \n"
>-    "pshufb    %%xmm4,%%xmm0                   \n"
>-    "pshufb    %%xmm5,%%xmm1                   \n"
>-    "paddusb   %%xmm1,%%xmm0                   \n"
>-    "movq      %%xmm0," MEMACCESS(1) "         \n"
>-    "movhlps   %%xmm0,%%xmm1                   \n"
>-    "movd      %%xmm1," MEMACCESS2(0x8,1) "    \n"
>-    "lea       " MEMLEA(0xc,1) ",%1            \n"
>-    "sub       $0xc,%2                         \n"
>-    "jg        1b                              \n"
>-  : "+r"(src_ptr),   // %0
>-    "+r"(dst_ptr),   // %1
>-    "+r"(dst_width)  // %2
>-  : "m"(kShuf38a),   // %3
>-    "m"(kShuf38b)    // %4
>-  : "memory", "cc", "xmm0", "xmm1", "xmm4", "xmm5"
>-  );
>+  asm volatile(
>+      "movdqa    %3,%%xmm4                       \n"
>+      "movdqa    %4,%%xmm5                       \n"
>+
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "movdqu    0x10(%0),%%xmm1                 \n"
>+      "lea       0x20(%0),%0                     \n"
>+      "pshufb    %%xmm4,%%xmm0                   \n"
>+      "pshufb    %%xmm5,%%xmm1                   \n"
>+      "paddusb   %%xmm1,%%xmm0                   \n"
>+      "movq      %%xmm0,(%1)                     \n"
>+      "movhlps   %%xmm0,%%xmm1                   \n"
>+      "movd      %%xmm1,0x8(%1)                  \n"
>+      "lea       0xc(%1),%1                      \n"
>+      "sub       $0xc,%2                         \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_ptr),   // %0
>+        "+r"(dst_ptr),   // %1
>+        "+r"(dst_width)  // %2
>+      : "m"(kShuf38a),   // %3
>+        "m"(kShuf38b)    // %4
>+      : "memory", "cc", "xmm0", "xmm1", "xmm4", "xmm5");
> }
> 
>-void ScaleRowDown38_2_Box_SSSE3(const uint8* src_ptr,
>+void ScaleRowDown38_2_Box_SSSE3(const uint8_t* src_ptr,
>                                 ptrdiff_t src_stride,
>-                                uint8* dst_ptr,
>+                                uint8_t* dst_ptr,
>                                 int dst_width) {
>   asm volatile(
>       "movdqa    %0,%%xmm2                       \n"
>@@ -695,40 +691,39 @@ void ScaleRowDown38_2_Box_SSSE3(const uint8* src_ptr,
>         "m"(kShufAb2),  // %2
>         "m"(kScaleAb2)  // %3
>       );
>-  asm volatile (
>-    LABELALIGN
>-  "1:                                          \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    MEMOPREG(movdqu,0x00,0,3,1,xmm1)           //  movdqu  (%0,%3,1),%%xmm1
>-    "lea       " MEMLEA(0x10,0) ",%0           \n"
>-    "pavgb     %%xmm1,%%xmm0                   \n"
>-    "movdqa    %%xmm0,%%xmm1                   \n"
>-    "pshufb    %%xmm2,%%xmm1                   \n"
>-    "movdqa    %%xmm0,%%xmm6                   \n"
>-    "pshufb    %%xmm3,%%xmm6                   \n"
>-    "paddusw   %%xmm6,%%xmm1                   \n"
>-    "pshufb    %%xmm4,%%xmm0                   \n"
>-    "paddusw   %%xmm0,%%xmm1                   \n"
>-    "pmulhuw   %%xmm5,%%xmm1                   \n"
>-    "packuswb  %%xmm1,%%xmm1                   \n"
>-    "movd      %%xmm1," MEMACCESS(1) "         \n"
>-    "psrlq     $0x10,%%xmm1                    \n"
>-    "movd      %%xmm1," MEMACCESS2(0x2,1) "    \n"
>-    "lea       " MEMLEA(0x6,1) ",%1            \n"
>-    "sub       $0x6,%2                         \n"
>-    "jg        1b                              \n"
>-  : "+r"(src_ptr),     // %0
>-    "+r"(dst_ptr),     // %1
>-    "+r"(dst_width)    // %2
>-  : "r"((intptr_t)(src_stride))  // %3
>-  : "memory", "cc", NACL_R14
>-    "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6"
>-  );
>+  asm volatile(
>+
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "movdqu    0x00(%0,%3,1),%%xmm1            \n"
>+      "lea       0x10(%0),%0                     \n"
>+      "pavgb     %%xmm1,%%xmm0                   \n"
>+      "movdqa    %%xmm0,%%xmm1                   \n"
>+      "pshufb    %%xmm2,%%xmm1                   \n"
>+      "movdqa    %%xmm0,%%xmm6                   \n"
>+      "pshufb    %%xmm3,%%xmm6                   \n"
>+      "paddusw   %%xmm6,%%xmm1                   \n"
>+      "pshufb    %%xmm4,%%xmm0                   \n"
>+      "paddusw   %%xmm0,%%xmm1                   \n"
>+      "pmulhuw   %%xmm5,%%xmm1                   \n"
>+      "packuswb  %%xmm1,%%xmm1                   \n"
>+      "movd      %%xmm1,(%1)                     \n"
>+      "psrlq     $0x10,%%xmm1                    \n"
>+      "movd      %%xmm1,0x2(%1)                  \n"
>+      "lea       0x6(%1),%1                      \n"
>+      "sub       $0x6,%2                         \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_ptr),               // %0
>+        "+r"(dst_ptr),               // %1
>+        "+r"(dst_width)              // %2
>+      : "r"((intptr_t)(src_stride))  // %3
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6");
> }
> 
>-void ScaleRowDown38_3_Box_SSSE3(const uint8* src_ptr,
>+void ScaleRowDown38_3_Box_SSSE3(const uint8_t* src_ptr,
>                                 ptrdiff_t src_stride,
>-                                uint8* dst_ptr,
>+                                uint8_t* dst_ptr,
>                                 int dst_width) {
>   asm volatile(
>       "movdqa    %0,%%xmm2                       \n"
>@@ -740,529 +735,533 @@ void ScaleRowDown38_3_Box_SSSE3(const uint8* src_ptr,
>         "m"(kShufAc3),   // %1
>         "m"(kScaleAc33)  // %2
>       );
>-  asm volatile (
>-    LABELALIGN
>-  "1:                                          \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    MEMOPREG(movdqu,0x00,0,3,1,xmm6)           //  movdqu  (%0,%3,1),%%xmm6
>-    "movhlps   %%xmm0,%%xmm1                   \n"
>-    "movhlps   %%xmm6,%%xmm7                   \n"
>-    "punpcklbw %%xmm5,%%xmm0                   \n"
>-    "punpcklbw %%xmm5,%%xmm1                   \n"
>-    "punpcklbw %%xmm5,%%xmm6                   \n"
>-    "punpcklbw %%xmm5,%%xmm7                   \n"
>-    "paddusw   %%xmm6,%%xmm0                   \n"
>-    "paddusw   %%xmm7,%%xmm1                   \n"
>-    MEMOPREG(movdqu,0x00,0,3,2,xmm6)           //  movdqu  (%0,%3,2),%%xmm6
>-    "lea       " MEMLEA(0x10,0) ",%0           \n"
>-    "movhlps   %%xmm6,%%xmm7                   \n"
>-    "punpcklbw %%xmm5,%%xmm6                   \n"
>-    "punpcklbw %%xmm5,%%xmm7                   \n"
>-    "paddusw   %%xmm6,%%xmm0                   \n"
>-    "paddusw   %%xmm7,%%xmm1                   \n"
>-    "movdqa    %%xmm0,%%xmm6                   \n"
>-    "psrldq    $0x2,%%xmm0                     \n"
>-    "paddusw   %%xmm0,%%xmm6                   \n"
>-    "psrldq    $0x2,%%xmm0                     \n"
>-    "paddusw   %%xmm0,%%xmm6                   \n"
>-    "pshufb    %%xmm2,%%xmm6                   \n"
>-    "movdqa    %%xmm1,%%xmm7                   \n"
>-    "psrldq    $0x2,%%xmm1                     \n"
>-    "paddusw   %%xmm1,%%xmm7                   \n"
>-    "psrldq    $0x2,%%xmm1                     \n"
>-    "paddusw   %%xmm1,%%xmm7                   \n"
>-    "pshufb    %%xmm3,%%xmm7                   \n"
>-    "paddusw   %%xmm7,%%xmm6                   \n"
>-    "pmulhuw   %%xmm4,%%xmm6                   \n"
>-    "packuswb  %%xmm6,%%xmm6                   \n"
>-    "movd      %%xmm6," MEMACCESS(1) "         \n"
>-    "psrlq     $0x10,%%xmm6                    \n"
>-    "movd      %%xmm6," MEMACCESS2(0x2,1) "    \n"
>-    "lea       " MEMLEA(0x6,1) ",%1            \n"
>-    "sub       $0x6,%2                         \n"
>-    "jg        1b                              \n"
>-  : "+r"(src_ptr),    // %0
>-    "+r"(dst_ptr),    // %1
>-    "+r"(dst_width)   // %2
>-  : "r"((intptr_t)(src_stride))   // %3
>-  : "memory", "cc", NACL_R14
>-    "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6", "xmm7"
>-  );
>+  asm volatile(
>+
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "movdqu    0x00(%0,%3,1),%%xmm6            \n"
>+      "movhlps   %%xmm0,%%xmm1                   \n"
>+      "movhlps   %%xmm6,%%xmm7                   \n"
>+      "punpcklbw %%xmm5,%%xmm0                   \n"
>+      "punpcklbw %%xmm5,%%xmm1                   \n"
>+      "punpcklbw %%xmm5,%%xmm6                   \n"
>+      "punpcklbw %%xmm5,%%xmm7                   \n"
>+      "paddusw   %%xmm6,%%xmm0                   \n"
>+      "paddusw   %%xmm7,%%xmm1                   \n"
>+      "movdqu    0x00(%0,%3,2),%%xmm6            \n"
>+      "lea       0x10(%0),%0                     \n"
>+      "movhlps   %%xmm6,%%xmm7                   \n"
>+      "punpcklbw %%xmm5,%%xmm6                   \n"
>+      "punpcklbw %%xmm5,%%xmm7                   \n"
>+      "paddusw   %%xmm6,%%xmm0                   \n"
>+      "paddusw   %%xmm7,%%xmm1                   \n"
>+      "movdqa    %%xmm0,%%xmm6                   \n"
>+      "psrldq    $0x2,%%xmm0                     \n"
>+      "paddusw   %%xmm0,%%xmm6                   \n"
>+      "psrldq    $0x2,%%xmm0                     \n"
>+      "paddusw   %%xmm0,%%xmm6                   \n"
>+      "pshufb    %%xmm2,%%xmm6                   \n"
>+      "movdqa    %%xmm1,%%xmm7                   \n"
>+      "psrldq    $0x2,%%xmm1                     \n"
>+      "paddusw   %%xmm1,%%xmm7                   \n"
>+      "psrldq    $0x2,%%xmm1                     \n"
>+      "paddusw   %%xmm1,%%xmm7                   \n"
>+      "pshufb    %%xmm3,%%xmm7                   \n"
>+      "paddusw   %%xmm7,%%xmm6                   \n"
>+      "pmulhuw   %%xmm4,%%xmm6                   \n"
>+      "packuswb  %%xmm6,%%xmm6                   \n"
>+      "movd      %%xmm6,(%1)                     \n"
>+      "psrlq     $0x10,%%xmm6                    \n"
>+      "movd      %%xmm6,0x2(%1)                  \n"
>+      "lea       0x6(%1),%1                      \n"
>+      "sub       $0x6,%2                         \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_ptr),               // %0
>+        "+r"(dst_ptr),               // %1
>+        "+r"(dst_width)              // %2
>+      : "r"((intptr_t)(src_stride))  // %3
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6",
>+        "xmm7");
> }
> 
> // Reads 16xN bytes and produces 16 shorts at a time.
>-void ScaleAddRow_SSE2(const uint8* src_ptr, uint16* dst_ptr, int src_width) {
>-  asm volatile (
>-    "pxor      %%xmm5,%%xmm5                   \n"
>-
>-    LABELALIGN
>-  "1:                                          \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm3         \n"
>-    "lea       " MEMLEA(0x10,0) ",%0           \n"  // src_ptr += 16
>-    "movdqu    " MEMACCESS(1) ",%%xmm0         \n"
>-    "movdqu    " MEMACCESS2(0x10,1) ",%%xmm1   \n"
>-    "movdqa    %%xmm3,%%xmm2                   \n"
>-    "punpcklbw %%xmm5,%%xmm2                   \n"
>-    "punpckhbw %%xmm5,%%xmm3                   \n"
>-    "paddusw   %%xmm2,%%xmm0                   \n"
>-    "paddusw   %%xmm3,%%xmm1                   \n"
>-    "movdqu    %%xmm0," MEMACCESS(1) "         \n"
>-    "movdqu    %%xmm1," MEMACCESS2(0x10,1) "   \n"
>-    "lea       " MEMLEA(0x20,1) ",%1           \n"
>-    "sub       $0x10,%2                        \n"
>-    "jg        1b                              \n"
>-  : "+r"(src_ptr),     // %0
>-    "+r"(dst_ptr),     // %1
>-    "+r"(src_width)    // %2
>-  :
>-  : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm5"
>-  );
>+void ScaleAddRow_SSE2(const uint8_t* src_ptr,
>+                      uint16_t* dst_ptr,
>+                      int src_width) {
>+  asm volatile(
>+
>+      "pxor      %%xmm5,%%xmm5                   \n"
>+
>+      // 16 pixel loop.
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm3                     \n"
>+      "lea       0x10(%0),%0                     \n"  // src_ptr += 16
>+      "movdqu    (%1),%%xmm0                     \n"
>+      "movdqu    0x10(%1),%%xmm1                 \n"
>+      "movdqa    %%xmm3,%%xmm2                   \n"
>+      "punpcklbw %%xmm5,%%xmm2                   \n"
>+      "punpckhbw %%xmm5,%%xmm3                   \n"
>+      "paddusw   %%xmm2,%%xmm0                   \n"
>+      "paddusw   %%xmm3,%%xmm1                   \n"
>+      "movdqu    %%xmm0,(%1)                     \n"
>+      "movdqu    %%xmm1,0x10(%1)                 \n"
>+      "lea       0x20(%1),%1                     \n"
>+      "sub       $0x10,%2                        \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_ptr),   // %0
>+        "+r"(dst_ptr),   // %1
>+        "+r"(src_width)  // %2
>+      :
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm5");
> }
> 
> #ifdef HAS_SCALEADDROW_AVX2
> // Reads 32 bytes and accumulates to 32 shorts at a time.
>-void ScaleAddRow_AVX2(const uint8* src_ptr, uint16* dst_ptr, int src_width) {
>-  asm volatile (
>-    "vpxor      %%ymm5,%%ymm5,%%ymm5           \n"
>-
>-    LABELALIGN
>-  "1:                                          \n"
>-    "vmovdqu    " MEMACCESS(0) ",%%ymm3        \n"
>-    "lea        " MEMLEA(0x20,0) ",%0          \n"  // src_ptr += 32
>-    "vpermq     $0xd8,%%ymm3,%%ymm3            \n"
>-    "vpunpcklbw %%ymm5,%%ymm3,%%ymm2           \n"
>-    "vpunpckhbw %%ymm5,%%ymm3,%%ymm3           \n"
>-    "vpaddusw   " MEMACCESS(1) ",%%ymm2,%%ymm0 \n"
>-    "vpaddusw   " MEMACCESS2(0x20,1) ",%%ymm3,%%ymm1 \n"
>-    "vmovdqu    %%ymm0," MEMACCESS(1) "        \n"
>-    "vmovdqu    %%ymm1," MEMACCESS2(0x20,1) "  \n"
>-    "lea       " MEMLEA(0x40,1) ",%1           \n"
>-    "sub       $0x20,%2                        \n"
>-    "jg        1b                              \n"
>-    "vzeroupper                                \n"
>-  : "+r"(src_ptr),     // %0
>-    "+r"(dst_ptr),     // %1
>-    "+r"(src_width)    // %2
>-  :
>-  : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm5"
>-  );
>+void ScaleAddRow_AVX2(const uint8_t* src_ptr,
>+                      uint16_t* dst_ptr,
>+                      int src_width) {
>+  asm volatile(
>+
>+      "vpxor      %%ymm5,%%ymm5,%%ymm5           \n"
>+
>+      LABELALIGN
>+      "1:                                        \n"
>+      "vmovdqu    (%0),%%ymm3                    \n"
>+      "lea        0x20(%0),%0                    \n"  // src_ptr += 32
>+      "vpermq     $0xd8,%%ymm3,%%ymm3            \n"
>+      "vpunpcklbw %%ymm5,%%ymm3,%%ymm2           \n"
>+      "vpunpckhbw %%ymm5,%%ymm3,%%ymm3           \n"
>+      "vpaddusw   (%1),%%ymm2,%%ymm0             \n"
>+      "vpaddusw   0x20(%1),%%ymm3,%%ymm1         \n"
>+      "vmovdqu    %%ymm0,(%1)                    \n"
>+      "vmovdqu    %%ymm1,0x20(%1)                \n"
>+      "lea       0x40(%1),%1                     \n"
>+      "sub       $0x20,%2                        \n"
>+      "jg        1b                              \n"
>+      "vzeroupper                                \n"
>+      : "+r"(src_ptr),   // %0
>+        "+r"(dst_ptr),   // %1
>+        "+r"(src_width)  // %2
>+      :
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm5");
> }
> #endif  // HAS_SCALEADDROW_AVX2
> 
> // Constant for making pixels signed to avoid pmaddubsw
> // saturation.
>-static uvec8 kFsub80 = {0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80,
>-                        0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80};
>+static const uvec8 kFsub80 = {0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80,
>+                              0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80};
> 
> // Constant for making pixels unsigned and adding .5 for rounding.
>-static uvec16 kFadd40 = {0x4040, 0x4040, 0x4040, 0x4040,
>-                         0x4040, 0x4040, 0x4040, 0x4040};
>+static const uvec16 kFadd40 = {0x4040, 0x4040, 0x4040, 0x4040,
>+                               0x4040, 0x4040, 0x4040, 0x4040};
> 
> // Bilinear column filtering. SSSE3 version.
>-void ScaleFilterCols_SSSE3(uint8* dst_ptr,
>-                           const uint8* src_ptr,
>+void ScaleFilterCols_SSSE3(uint8_t* dst_ptr,
>+                           const uint8_t* src_ptr,
>                            int dst_width,
>                            int x,
>                            int dx) {
>   intptr_t x0, x1, temp_pixel;
>-  asm volatile (
>-    "movd      %6,%%xmm2                       \n"
>-    "movd      %7,%%xmm3                       \n"
>-    "movl      $0x04040000,%k2                 \n"
>-    "movd      %k2,%%xmm5                      \n"
>-    "pcmpeqb   %%xmm6,%%xmm6                   \n"
>-    "psrlw     $0x9,%%xmm6                     \n"  // 0x007f007f
>-    "pcmpeqb   %%xmm7,%%xmm7                   \n"
>-    "psrlw     $15,%%xmm7                      \n"  // 0x00010001
>-
>-    "pextrw    $0x1,%%xmm2,%k3                 \n"
>-    "subl      $0x2,%5                         \n"
>-    "jl        29f                             \n"
>-    "movdqa    %%xmm2,%%xmm0                   \n"
>-    "paddd     %%xmm3,%%xmm0                   \n"
>-    "punpckldq %%xmm0,%%xmm2                   \n"
>-    "punpckldq %%xmm3,%%xmm3                   \n"
>-    "paddd     %%xmm3,%%xmm3                   \n"
>-    "pextrw    $0x3,%%xmm2,%k4                 \n"
>-
>-    LABELALIGN
>-  "2:                                          \n"
>-    "movdqa    %%xmm2,%%xmm1                   \n"
>-    "paddd     %%xmm3,%%xmm2                   \n"
>-    MEMOPARG(movzwl,0x00,1,3,1,k2)             //  movzwl  (%1,%3,1),%k2
>-    "movd      %k2,%%xmm0                      \n"
>-    "psrlw     $0x9,%%xmm1                     \n"
>-    MEMOPARG(movzwl,0x00,1,4,1,k2)             //  movzwl  (%1,%4,1),%k2
>-    "movd      %k2,%%xmm4                      \n"
>-    "pshufb    %%xmm5,%%xmm1                   \n"
>-    "punpcklwd %%xmm4,%%xmm0                   \n"
>-    "psubb     %8,%%xmm0                       \n"  // make pixels signed.
>-    "pxor      %%xmm6,%%xmm1                   \n"  // 128 - f = (f ^ 127 ) + 1
>-    "paddusb   %%xmm7,%%xmm1                   \n"
>-    "pmaddubsw %%xmm0,%%xmm1                   \n"
>-    "pextrw    $0x1,%%xmm2,%k3                 \n"
>-    "pextrw    $0x3,%%xmm2,%k4                 \n"
>-    "paddw     %9,%%xmm1                       \n"  // make pixels unsigned.
>-    "psrlw     $0x7,%%xmm1                     \n"
>-    "packuswb  %%xmm1,%%xmm1                   \n"
>-    "movd      %%xmm1,%k2                      \n"
>-    "mov       %w2," MEMACCESS(0) "            \n"
>-    "lea       " MEMLEA(0x2,0) ",%0            \n"
>-    "subl      $0x2,%5                         \n"
>-    "jge       2b                              \n"
>-
>-    LABELALIGN
>-  "29:                                         \n"
>-    "addl      $0x1,%5                         \n"
>-    "jl        99f                             \n"
>-    MEMOPARG(movzwl,0x00,1,3,1,k2)             //  movzwl  (%1,%3,1),%k2
>-    "movd      %k2,%%xmm0                      \n"
>-    "psrlw     $0x9,%%xmm2                     \n"
>-    "pshufb    %%xmm5,%%xmm2                   \n"
>-    "psubb     %8,%%xmm0                       \n"  // make pixels signed.
>-    "pxor      %%xmm6,%%xmm2                   \n"
>-    "paddusb   %%xmm7,%%xmm2                   \n"
>-    "pmaddubsw %%xmm0,%%xmm2                   \n"
>-    "paddw     %9,%%xmm2                       \n"  // make pixels unsigned.
>-    "psrlw     $0x7,%%xmm2                     \n"
>-    "packuswb  %%xmm2,%%xmm2                   \n"
>-    "movd      %%xmm2,%k2                      \n"
>-    "mov       %b2," MEMACCESS(0) "            \n"
>-  "99:                                         \n"
>-  : "+r"(dst_ptr),      // %0
>-    "+r"(src_ptr),      // %1
>-    "=&a"(temp_pixel),  // %2
>-    "=&r"(x0),          // %3
>-    "=&r"(x1),          // %4
>+  asm volatile(
>+      "movd      %6,%%xmm2                       \n"
>+      "movd      %7,%%xmm3                       \n"
>+      "movl      $0x04040000,%k2                 \n"
>+      "movd      %k2,%%xmm5                      \n"
>+      "pcmpeqb   %%xmm6,%%xmm6                   \n"
>+      "psrlw     $0x9,%%xmm6                     \n"  // 0x007f007f
>+      "pcmpeqb   %%xmm7,%%xmm7                   \n"
>+      "psrlw     $15,%%xmm7                      \n"  // 0x00010001
>+
>+      "pextrw    $0x1,%%xmm2,%k3                 \n"
>+      "subl      $0x2,%5                         \n"
>+      "jl        29f                             \n"
>+      "movdqa    %%xmm2,%%xmm0                   \n"
>+      "paddd     %%xmm3,%%xmm0                   \n"
>+      "punpckldq %%xmm0,%%xmm2                   \n"
>+      "punpckldq %%xmm3,%%xmm3                   \n"
>+      "paddd     %%xmm3,%%xmm3                   \n"
>+      "pextrw    $0x3,%%xmm2,%k4                 \n"
>+
>+      LABELALIGN
>+      "2:                                        \n"
>+      "movdqa    %%xmm2,%%xmm1                   \n"
>+      "paddd     %%xmm3,%%xmm2                   \n"
>+      "movzwl    0x00(%1,%3,1),%k2               \n"
>+      "movd      %k2,%%xmm0                      \n"
>+      "psrlw     $0x9,%%xmm1                     \n"
>+      "movzwl    0x00(%1,%4,1),%k2               \n"
>+      "movd      %k2,%%xmm4                      \n"
>+      "pshufb    %%xmm5,%%xmm1                   \n"
>+      "punpcklwd %%xmm4,%%xmm0                   \n"
>+      "psubb     %8,%%xmm0                       \n"  // make pixels signed.
>+      "pxor      %%xmm6,%%xmm1                   \n"  // 128 - f = (f ^ 127 ) +
>+                                                      // 1
>+      "paddusb   %%xmm7,%%xmm1                   \n"
>+      "pmaddubsw %%xmm0,%%xmm1                   \n"
>+      "pextrw    $0x1,%%xmm2,%k3                 \n"
>+      "pextrw    $0x3,%%xmm2,%k4                 \n"
>+      "paddw     %9,%%xmm1                       \n"  // make pixels unsigned.
>+      "psrlw     $0x7,%%xmm1                     \n"
>+      "packuswb  %%xmm1,%%xmm1                   \n"
>+      "movd      %%xmm1,%k2                      \n"
>+      "mov       %w2,(%0)                        \n"
>+      "lea       0x2(%0),%0                      \n"
>+      "subl      $0x2,%5                         \n"
>+      "jge       2b                              \n"
>+
>+      LABELALIGN
>+      "29:                                       \n"
>+      "addl      $0x1,%5                         \n"
>+      "jl        99f                             \n"
>+      "movzwl    0x00(%1,%3,1),%k2               \n"
>+      "movd      %k2,%%xmm0                      \n"
>+      "psrlw     $0x9,%%xmm2                     \n"
>+      "pshufb    %%xmm5,%%xmm2                   \n"
>+      "psubb     %8,%%xmm0                       \n"  // make pixels signed.
>+      "pxor      %%xmm6,%%xmm2                   \n"
>+      "paddusb   %%xmm7,%%xmm2                   \n"
>+      "pmaddubsw %%xmm0,%%xmm2                   \n"
>+      "paddw     %9,%%xmm2                       \n"  // make pixels unsigned.
>+      "psrlw     $0x7,%%xmm2                     \n"
>+      "packuswb  %%xmm2,%%xmm2                   \n"
>+      "movd      %%xmm2,%k2                      \n"
>+      "mov       %b2,(%0)                        \n"
>+      "99:                                       \n"
>+      : "+r"(dst_ptr),      // %0
>+        "+r"(src_ptr),      // %1
>+        "=&a"(temp_pixel),  // %2
>+        "=&r"(x0),          // %3
>+        "=&r"(x1),          // %4
> #if defined(__x86_64__)
>-    "+rm"(dst_width)    // %5
>+        "+rm"(dst_width)  // %5
> #else
>-    "+m"(dst_width)    // %5
>+        "+m"(dst_width)  // %5
> #endif
>-  : "rm"(x),            // %6
>-    "rm"(dx),           // %7
>+      : "rm"(x),   // %6
>+        "rm"(dx),  // %7
> #if defined(__x86_64__)
>-    "x"(kFsub80),       // %8
>-    "x"(kFadd40)        // %9
>+        "x"(kFsub80),  // %8
>+        "x"(kFadd40)   // %9
> #else
>-    "m"(kFsub80),       // %8
>-    "m"(kFadd40)        // %9
>+        "m"(kFsub80),    // %8
>+        "m"(kFadd40)     // %9
> #endif
>-  : "memory", "cc", NACL_R14
>-    "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6", "xmm7"
>-  );
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6",
>+        "xmm7");
> }
> 
> // Reads 4 pixels, duplicates them and writes 8 pixels.
> // Alignment requirement: src_argb 16 byte aligned, dst_argb 16 byte aligned.
>-void ScaleColsUp2_SSE2(uint8* dst_ptr,
>-                       const uint8* src_ptr,
>+void ScaleColsUp2_SSE2(uint8_t* dst_ptr,
>+                       const uint8_t* src_ptr,
>                        int dst_width,
>                        int x,
>                        int dx) {
>   (void)x;
>   (void)dx;
>-  asm volatile (
>-    LABELALIGN
>-  "1:                                          \n"
>-    "movdqu    " MEMACCESS(1) ",%%xmm0         \n"
>-    "lea       " MEMLEA(0x10,1) ",%1           \n"
>-    "movdqa    %%xmm0,%%xmm1                   \n"
>-    "punpcklbw %%xmm0,%%xmm0                   \n"
>-    "punpckhbw %%xmm1,%%xmm1                   \n"
>-    "movdqu    %%xmm0," MEMACCESS(0) "         \n"
>-    "movdqu    %%xmm1," MEMACCESS2(0x10,0) "   \n"
>-    "lea       " MEMLEA(0x20,0) ",%0           \n"
>-    "sub       $0x20,%2                         \n"
>-    "jg        1b                              \n"
>-
>-  : "+r"(dst_ptr),     // %0
>-    "+r"(src_ptr),     // %1
>-    "+r"(dst_width)    // %2
>-  :: "memory", "cc", "xmm0", "xmm1"
>-  );
>+  asm volatile(
>+
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%1),%%xmm0                     \n"
>+      "lea       0x10(%1),%1                     \n"
>+      "movdqa    %%xmm0,%%xmm1                   \n"
>+      "punpcklbw %%xmm0,%%xmm0                   \n"
>+      "punpckhbw %%xmm1,%%xmm1                   \n"
>+      "movdqu    %%xmm0,(%0)                     \n"
>+      "movdqu    %%xmm1,0x10(%0)                 \n"
>+      "lea       0x20(%0),%0                     \n"
>+      "sub       $0x20,%2                        \n"
>+      "jg        1b                              \n"
>+
>+      : "+r"(dst_ptr),   // %0
>+        "+r"(src_ptr),   // %1
>+        "+r"(dst_width)  // %2
>+        ::"memory",
>+        "cc", "xmm0", "xmm1");
> }
> 
>-void ScaleARGBRowDown2_SSE2(const uint8* src_argb,
>+void ScaleARGBRowDown2_SSE2(const uint8_t* src_argb,
>                             ptrdiff_t src_stride,
>-                            uint8* dst_argb,
>+                            uint8_t* dst_argb,
>                             int dst_width) {
>   (void)src_stride;
>-  asm volatile (
>-    LABELALIGN
>-  "1:                                          \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    "movdqu    " MEMACCESS2(0x10,0) ",%%xmm1   \n"
>-    "lea       " MEMLEA(0x20,0) ",%0           \n"
>-    "shufps    $0xdd,%%xmm1,%%xmm0             \n"
>-    "movdqu    %%xmm0," MEMACCESS(1) "         \n"
>-    "lea       " MEMLEA(0x10,1) ",%1           \n"
>-    "sub       $0x4,%2                         \n"
>-    "jg        1b                              \n"
>-  : "+r"(src_argb),  // %0
>-    "+r"(dst_argb),  // %1
>-    "+r"(dst_width)  // %2
>-  :: "memory", "cc", "xmm0", "xmm1"
>-  );
>+  asm volatile(
>+
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "movdqu    0x10(%0),%%xmm1                 \n"
>+      "lea       0x20(%0),%0                     \n"
>+      "shufps    $0xdd,%%xmm1,%%xmm0             \n"
>+      "movdqu    %%xmm0,(%1)                     \n"
>+      "lea       0x10(%1),%1                     \n"
>+      "sub       $0x4,%2                         \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_argb),  // %0
>+        "+r"(dst_argb),  // %1
>+        "+r"(dst_width)  // %2
>+        ::"memory",
>+        "cc", "xmm0", "xmm1");
> }
> 
>-void ScaleARGBRowDown2Linear_SSE2(const uint8* src_argb,
>+void ScaleARGBRowDown2Linear_SSE2(const uint8_t* src_argb,
>                                   ptrdiff_t src_stride,
>-                                  uint8* dst_argb,
>+                                  uint8_t* dst_argb,
>                                   int dst_width) {
>   (void)src_stride;
>-  asm volatile (
>-    LABELALIGN
>-  "1:                                          \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    "movdqu    " MEMACCESS2(0x10,0) ",%%xmm1   \n"
>-    "lea       " MEMLEA(0x20,0) ",%0           \n"
>-    "movdqa    %%xmm0,%%xmm2                   \n"
>-    "shufps    $0x88,%%xmm1,%%xmm0             \n"
>-    "shufps    $0xdd,%%xmm1,%%xmm2             \n"
>-    "pavgb     %%xmm2,%%xmm0                   \n"
>-    "movdqu    %%xmm0," MEMACCESS(1) "         \n"
>-    "lea       " MEMLEA(0x10,1) ",%1           \n"
>-    "sub       $0x4,%2                         \n"
>-    "jg        1b                              \n"
>-  : "+r"(src_argb),  // %0
>-    "+r"(dst_argb),  // %1
>-    "+r"(dst_width)  // %2
>-  :: "memory", "cc", "xmm0", "xmm1"
>-  );
>+  asm volatile(
>+
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "movdqu    0x10(%0),%%xmm1                 \n"
>+      "lea       0x20(%0),%0                     \n"
>+      "movdqa    %%xmm0,%%xmm2                   \n"
>+      "shufps    $0x88,%%xmm1,%%xmm0             \n"
>+      "shufps    $0xdd,%%xmm1,%%xmm2             \n"
>+      "pavgb     %%xmm2,%%xmm0                   \n"
>+      "movdqu    %%xmm0,(%1)                     \n"
>+      "lea       0x10(%1),%1                     \n"
>+      "sub       $0x4,%2                         \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_argb),  // %0
>+        "+r"(dst_argb),  // %1
>+        "+r"(dst_width)  // %2
>+        ::"memory",
>+        "cc", "xmm0", "xmm1");
> }
> 
>-void ScaleARGBRowDown2Box_SSE2(const uint8* src_argb,
>+void ScaleARGBRowDown2Box_SSE2(const uint8_t* src_argb,
>                                ptrdiff_t src_stride,
>-                               uint8* dst_argb,
>+                               uint8_t* dst_argb,
>                                int dst_width) {
>-  asm volatile (
>-    LABELALIGN
>-  "1:                                          \n"
>-    "movdqu    " MEMACCESS(0) ",%%xmm0         \n"
>-    "movdqu    " MEMACCESS2(0x10,0) ",%%xmm1   \n"
>-    MEMOPREG(movdqu,0x00,0,3,1,xmm2)           //  movdqu   (%0,%3,1),%%xmm2
>-    MEMOPREG(movdqu,0x10,0,3,1,xmm3)           //  movdqu   0x10(%0,%3,1),%%xmm3
>-    "lea       " MEMLEA(0x20,0) ",%0           \n"
>-    "pavgb     %%xmm2,%%xmm0                   \n"
>-    "pavgb     %%xmm3,%%xmm1                   \n"
>-    "movdqa    %%xmm0,%%xmm2                   \n"
>-    "shufps    $0x88,%%xmm1,%%xmm0             \n"
>-    "shufps    $0xdd,%%xmm1,%%xmm2             \n"
>-    "pavgb     %%xmm2,%%xmm0                   \n"
>-    "movdqu    %%xmm0," MEMACCESS(1) "         \n"
>-    "lea       " MEMLEA(0x10,1) ",%1           \n"
>-    "sub       $0x4,%2                         \n"
>-    "jg        1b                              \n"
>-  : "+r"(src_argb),   // %0
>-    "+r"(dst_argb),   // %1
>-    "+r"(dst_width)   // %2
>-  : "r"((intptr_t)(src_stride))   // %3
>-  : "memory", "cc", NACL_R14
>-    "xmm0", "xmm1", "xmm2", "xmm3"
>-  );
>+  asm volatile(
>+
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%0),%%xmm0                     \n"
>+      "movdqu    0x10(%0),%%xmm1                 \n"
>+      "movdqu    0x00(%0,%3,1),%%xmm2            \n"
>+      "movdqu    0x10(%0,%3,1),%%xmm3            \n"
>+      "lea       0x20(%0),%0                     \n"
>+      "pavgb     %%xmm2,%%xmm0                   \n"
>+      "pavgb     %%xmm3,%%xmm1                   \n"
>+      "movdqa    %%xmm0,%%xmm2                   \n"
>+      "shufps    $0x88,%%xmm1,%%xmm0             \n"
>+      "shufps    $0xdd,%%xmm1,%%xmm2             \n"
>+      "pavgb     %%xmm2,%%xmm0                   \n"
>+      "movdqu    %%xmm0,(%1)                     \n"
>+      "lea       0x10(%1),%1                     \n"
>+      "sub       $0x4,%2                         \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_argb),              // %0
>+        "+r"(dst_argb),              // %1
>+        "+r"(dst_width)              // %2
>+      : "r"((intptr_t)(src_stride))  // %3
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3");
> }
> 
> // Reads 4 pixels at a time.
> // Alignment requirement: dst_argb 16 byte aligned.
>-void ScaleARGBRowDownEven_SSE2(const uint8* src_argb,
>+void ScaleARGBRowDownEven_SSE2(const uint8_t* src_argb,
>                                ptrdiff_t src_stride,
>                                int src_stepx,
>-                               uint8* dst_argb,
>+                               uint8_t* dst_argb,
>                                int dst_width) {
>   intptr_t src_stepx_x4 = (intptr_t)(src_stepx);
>   intptr_t src_stepx_x12;
>   (void)src_stride;
>-  asm volatile (
>-    "lea       " MEMLEA3(0x00,1,4) ",%1        \n"
>-    "lea       " MEMLEA4(0x00,1,1,2) ",%4      \n"
>-    LABELALIGN
>-  "1:                                          \n"
>-    "movd      " MEMACCESS(0) ",%%xmm0         \n"
>-    MEMOPREG(movd,0x00,0,1,1,xmm1)             //  movd      (%0,%1,1),%%xmm1
>-    "punpckldq %%xmm1,%%xmm0                   \n"
>-    MEMOPREG(movd,0x00,0,1,2,xmm2)             //  movd      (%0,%1,2),%%xmm2
>-    MEMOPREG(movd,0x00,0,4,1,xmm3)             //  movd      (%0,%4,1),%%xmm3
>-    "lea       " MEMLEA4(0x00,0,1,4) ",%0      \n"
>-    "punpckldq %%xmm3,%%xmm2                   \n"
>-    "punpcklqdq %%xmm2,%%xmm0                  \n"
>-    "movdqu    %%xmm0," MEMACCESS(2) "         \n"
>-    "lea       " MEMLEA(0x10,2) ",%2           \n"
>-    "sub       $0x4,%3                         \n"
>-    "jg        1b                              \n"
>-  : "+r"(src_argb),       // %0
>-    "+r"(src_stepx_x4),   // %1
>-    "+r"(dst_argb),       // %2
>-    "+r"(dst_width),      // %3
>-    "=&r"(src_stepx_x12)  // %4
>-  :: "memory", "cc", NACL_R14
>-    "xmm0", "xmm1", "xmm2", "xmm3"
>-  );
>+  asm volatile(
>+      "lea       0x00(,%1,4),%1                  \n"
>+      "lea       0x00(%1,%1,2),%4                \n"
>+
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movd      (%0),%%xmm0                     \n"
>+      "movd      0x00(%0,%1,1),%%xmm1            \n"
>+      "punpckldq %%xmm1,%%xmm0                   \n"
>+      "movd      0x00(%0,%1,2),%%xmm2            \n"
>+      "movd      0x00(%0,%4,1),%%xmm3            \n"
>+      "lea       0x00(%0,%1,4),%0                \n"
>+      "punpckldq %%xmm3,%%xmm2                   \n"
>+      "punpcklqdq %%xmm2,%%xmm0                  \n"
>+      "movdqu    %%xmm0,(%2)                     \n"
>+      "lea       0x10(%2),%2                     \n"
>+      "sub       $0x4,%3                         \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_argb),       // %0
>+        "+r"(src_stepx_x4),   // %1
>+        "+r"(dst_argb),       // %2
>+        "+r"(dst_width),      // %3
>+        "=&r"(src_stepx_x12)  // %4
>+        ::"memory",
>+        "cc", "xmm0", "xmm1", "xmm2", "xmm3");
> }
> 
> // Blends four 2x2 to 4x1.
> // Alignment requirement: dst_argb 16 byte aligned.
>-void ScaleARGBRowDownEvenBox_SSE2(const uint8* src_argb,
>+void ScaleARGBRowDownEvenBox_SSE2(const uint8_t* src_argb,
>                                   ptrdiff_t src_stride,
>                                   int src_stepx,
>-                                  uint8* dst_argb,
>+                                  uint8_t* dst_argb,
>                                   int dst_width) {
>   intptr_t src_stepx_x4 = (intptr_t)(src_stepx);
>   intptr_t src_stepx_x12;
>   intptr_t row1 = (intptr_t)(src_stride);
>-  asm volatile (
>-    "lea       " MEMLEA3(0x00,1,4) ",%1        \n"
>-    "lea       " MEMLEA4(0x00,1,1,2) ",%4      \n"
>-    "lea       " MEMLEA4(0x00,0,5,1) ",%5      \n"
>-
>-    LABELALIGN
>-  "1:                                          \n"
>-    "movq      " MEMACCESS(0) ",%%xmm0         \n"
>-    MEMOPREG(movhps,0x00,0,1,1,xmm0)           //  movhps    (%0,%1,1),%%xmm0
>-    MEMOPREG(movq,0x00,0,1,2,xmm1)             //  movq      (%0,%1,2),%%xmm1
>-    MEMOPREG(movhps,0x00,0,4,1,xmm1)           //  movhps    (%0,%4,1),%%xmm1
>-    "lea       " MEMLEA4(0x00,0,1,4) ",%0      \n"
>-    "movq      " MEMACCESS(5) ",%%xmm2         \n"
>-    MEMOPREG(movhps,0x00,5,1,1,xmm2)           //  movhps    (%5,%1,1),%%xmm2
>-    MEMOPREG(movq,0x00,5,1,2,xmm3)             //  movq      (%5,%1,2),%%xmm3
>-    MEMOPREG(movhps,0x00,5,4,1,xmm3)           //  movhps    (%5,%4,1),%%xmm3
>-    "lea       " MEMLEA4(0x00,5,1,4) ",%5      \n"
>-    "pavgb     %%xmm2,%%xmm0                   \n"
>-    "pavgb     %%xmm3,%%xmm1                   \n"
>-    "movdqa    %%xmm0,%%xmm2                   \n"
>-    "shufps    $0x88,%%xmm1,%%xmm0             \n"
>-    "shufps    $0xdd,%%xmm1,%%xmm2             \n"
>-    "pavgb     %%xmm2,%%xmm0                   \n"
>-    "movdqu    %%xmm0," MEMACCESS(2) "         \n"
>-    "lea       " MEMLEA(0x10,2) ",%2           \n"
>-    "sub       $0x4,%3                         \n"
>-    "jg        1b                              \n"
>-  : "+r"(src_argb),        // %0
>-    "+r"(src_stepx_x4),    // %1
>-    "+r"(dst_argb),        // %2
>-    "+rm"(dst_width),      // %3
>-    "=&r"(src_stepx_x12),  // %4
>-    "+r"(row1)             // %5
>-  :: "memory", "cc", NACL_R14
>-    "xmm0", "xmm1", "xmm2", "xmm3"
>-  );
>+  asm volatile(
>+      "lea       0x00(,%1,4),%1                  \n"
>+      "lea       0x00(%1,%1,2),%4                \n"
>+      "lea       0x00(%0,%5,1),%5                \n"
>+
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movq      (%0),%%xmm0                     \n"
>+      "movhps    0x00(%0,%1,1),%%xmm0            \n"
>+      "movq      0x00(%0,%1,2),%%xmm1            \n"
>+      "movhps    0x00(%0,%4,1),%%xmm1            \n"
>+      "lea       0x00(%0,%1,4),%0                \n"
>+      "movq      (%5),%%xmm2                     \n"
>+      "movhps    0x00(%5,%1,1),%%xmm2            \n"
>+      "movq      0x00(%5,%1,2),%%xmm3            \n"
>+      "movhps    0x00(%5,%4,1),%%xmm3            \n"
>+      "lea       0x00(%5,%1,4),%5                \n"
>+      "pavgb     %%xmm2,%%xmm0                   \n"
>+      "pavgb     %%xmm3,%%xmm1                   \n"
>+      "movdqa    %%xmm0,%%xmm2                   \n"
>+      "shufps    $0x88,%%xmm1,%%xmm0             \n"
>+      "shufps    $0xdd,%%xmm1,%%xmm2             \n"
>+      "pavgb     %%xmm2,%%xmm0                   \n"
>+      "movdqu    %%xmm0,(%2)                     \n"
>+      "lea       0x10(%2),%2                     \n"
>+      "sub       $0x4,%3                         \n"
>+      "jg        1b                              \n"
>+      : "+r"(src_argb),        // %0
>+        "+r"(src_stepx_x4),    // %1
>+        "+r"(dst_argb),        // %2
>+        "+rm"(dst_width),      // %3
>+        "=&r"(src_stepx_x12),  // %4
>+        "+r"(row1)             // %5
>+        ::"memory",
>+        "cc", "xmm0", "xmm1", "xmm2", "xmm3");
> }
> 
>-void ScaleARGBCols_SSE2(uint8* dst_argb,
>-                        const uint8* src_argb,
>+void ScaleARGBCols_SSE2(uint8_t* dst_argb,
>+                        const uint8_t* src_argb,
>                         int dst_width,
>                         int x,
>                         int dx) {
>   intptr_t x0, x1;
>-  asm volatile (
>-    "movd      %5,%%xmm2                       \n"
>-    "movd      %6,%%xmm3                       \n"
>-    "pshufd    $0x0,%%xmm2,%%xmm2              \n"
>-    "pshufd    $0x11,%%xmm3,%%xmm0             \n"
>-    "paddd     %%xmm0,%%xmm2                   \n"
>-    "paddd     %%xmm3,%%xmm3                   \n"
>-    "pshufd    $0x5,%%xmm3,%%xmm0              \n"
>-    "paddd     %%xmm0,%%xmm2                   \n"
>-    "paddd     %%xmm3,%%xmm3                   \n"
>-    "pshufd    $0x0,%%xmm3,%%xmm3              \n"
>-    "pextrw    $0x1,%%xmm2,%k0                 \n"
>-    "pextrw    $0x3,%%xmm2,%k1                 \n"
>-    "cmp       $0x0,%4                         \n"
>-    "jl        99f                             \n"
>-    "sub       $0x4,%4                         \n"
>-    "jl        49f                             \n"
>-
>-    LABELALIGN
>-  "40:                                         \n"
>-    MEMOPREG(movd,0x00,3,0,4,xmm0)             //  movd      (%3,%0,4),%%xmm0
>-    MEMOPREG(movd,0x00,3,1,4,xmm1)             //  movd      (%3,%1,4),%%xmm1
>-    "pextrw    $0x5,%%xmm2,%k0                 \n"
>-    "pextrw    $0x7,%%xmm2,%k1                 \n"
>-    "paddd     %%xmm3,%%xmm2                   \n"
>-    "punpckldq %%xmm1,%%xmm0                   \n"
>-    MEMOPREG(movd,0x00,3,0,4,xmm1)             //  movd      (%3,%0,4),%%xmm1
>-    MEMOPREG(movd,0x00,3,1,4,xmm4)             //  movd      (%3,%1,4),%%xmm4
>-    "pextrw    $0x1,%%xmm2,%k0                 \n"
>-    "pextrw    $0x3,%%xmm2,%k1                 \n"
>-    "punpckldq %%xmm4,%%xmm1                   \n"
>-    "punpcklqdq %%xmm1,%%xmm0                  \n"
>-    "movdqu    %%xmm0," MEMACCESS(2) "         \n"
>-    "lea       " MEMLEA(0x10,2) ",%2           \n"
>-    "sub       $0x4,%4                         \n"
>-    "jge       40b                             \n"
>-
>-  "49:                                         \n"
>-    "test      $0x2,%4                         \n"
>-    "je        29f                             \n"
>-    MEMOPREG(movd,0x00,3,0,4,xmm0)             //  movd      (%3,%0,4),%%xmm0
>-    MEMOPREG(movd,0x00,3,1,4,xmm1)             //  movd      (%3,%1,4),%%xmm1
>-    "pextrw    $0x5,%%xmm2,%k0                 \n"
>-    "punpckldq %%xmm1,%%xmm0                   \n"
>-    "movq      %%xmm0," MEMACCESS(2) "         \n"
>-    "lea       " MEMLEA(0x8,2) ",%2            \n"
>-  "29:                                         \n"
>-    "test      $0x1,%4                         \n"
>-    "je        99f                             \n"
>-    MEMOPREG(movd,0x00,3,0,4,xmm0)             //  movd      (%3,%0,4),%%xmm0
>-    "movd      %%xmm0," MEMACCESS(2) "         \n"
>-  "99:                                         \n"
>-  : "=&a"(x0),         // %0
>-    "=&d"(x1),         // %1
>-    "+r"(dst_argb),    // %2
>-    "+r"(src_argb),    // %3
>-    "+r"(dst_width)    // %4
>-  : "rm"(x),           // %5
>-    "rm"(dx)           // %6
>-  : "memory", "cc", NACL_R14
>-    "xmm0", "xmm1", "xmm2", "xmm3", "xmm4"
>-  );
>+  asm volatile(
>+      "movd      %5,%%xmm2                       \n"
>+      "movd      %6,%%xmm3                       \n"
>+      "pshufd    $0x0,%%xmm2,%%xmm2              \n"
>+      "pshufd    $0x11,%%xmm3,%%xmm0             \n"
>+      "paddd     %%xmm0,%%xmm2                   \n"
>+      "paddd     %%xmm3,%%xmm3                   \n"
>+      "pshufd    $0x5,%%xmm3,%%xmm0              \n"
>+      "paddd     %%xmm0,%%xmm2                   \n"
>+      "paddd     %%xmm3,%%xmm3                   \n"
>+      "pshufd    $0x0,%%xmm3,%%xmm3              \n"
>+      "pextrw    $0x1,%%xmm2,%k0                 \n"
>+      "pextrw    $0x3,%%xmm2,%k1                 \n"
>+      "cmp       $0x0,%4                         \n"
>+      "jl        99f                             \n"
>+      "sub       $0x4,%4                         \n"
>+      "jl        49f                             \n"
>+
>+      LABELALIGN
>+      "40:                                       \n"
>+      "movd      0x00(%3,%0,4),%%xmm0            \n"
>+      "movd      0x00(%3,%1,4),%%xmm1            \n"
>+      "pextrw    $0x5,%%xmm2,%k0                 \n"
>+      "pextrw    $0x7,%%xmm2,%k1                 \n"
>+      "paddd     %%xmm3,%%xmm2                   \n"
>+      "punpckldq %%xmm1,%%xmm0                   \n"
>+      "movd      0x00(%3,%0,4),%%xmm1            \n"
>+      "movd      0x00(%3,%1,4),%%xmm4            \n"
>+      "pextrw    $0x1,%%xmm2,%k0                 \n"
>+      "pextrw    $0x3,%%xmm2,%k1                 \n"
>+      "punpckldq %%xmm4,%%xmm1                   \n"
>+      "punpcklqdq %%xmm1,%%xmm0                  \n"
>+      "movdqu    %%xmm0,(%2)                     \n"
>+      "lea       0x10(%2),%2                     \n"
>+      "sub       $0x4,%4                         \n"
>+      "jge       40b                             \n"
>+
>+      "49:                                       \n"
>+      "test      $0x2,%4                         \n"
>+      "je        29f                             \n"
>+      "movd      0x00(%3,%0,4),%%xmm0            \n"
>+      "movd      0x00(%3,%1,4),%%xmm1            \n"
>+      "pextrw    $0x5,%%xmm2,%k0                 \n"
>+      "punpckldq %%xmm1,%%xmm0                   \n"
>+      "movq      %%xmm0,(%2)                     \n"
>+      "lea       0x8(%2),%2                      \n"
>+      "29:                                       \n"
>+      "test      $0x1,%4                         \n"
>+      "je        99f                             \n"
>+      "movd      0x00(%3,%0,4),%%xmm0            \n"
>+      "movd      %%xmm0,(%2)                     \n"
>+      "99:                                       \n"
>+      : "=&a"(x0),       // %0
>+        "=&d"(x1),       // %1
>+        "+r"(dst_argb),  // %2
>+        "+r"(src_argb),  // %3
>+        "+r"(dst_width)  // %4
>+      : "rm"(x),         // %5
>+        "rm"(dx)         // %6
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4");
> }
> 
> // Reads 4 pixels, duplicates them and writes 8 pixels.
> // Alignment requirement: src_argb 16 byte aligned, dst_argb 16 byte aligned.
>-void ScaleARGBColsUp2_SSE2(uint8* dst_argb,
>-                           const uint8* src_argb,
>+void ScaleARGBColsUp2_SSE2(uint8_t* dst_argb,
>+                           const uint8_t* src_argb,
>                            int dst_width,
>                            int x,
>                            int dx) {
>   (void)x;
>   (void)dx;
>-  asm volatile (
>-    LABELALIGN
>-  "1:                                          \n"
>-    "movdqu    " MEMACCESS(1) ",%%xmm0         \n"
>-    "lea       " MEMLEA(0x10,1) ",%1           \n"
>-    "movdqa    %%xmm0,%%xmm1                   \n"
>-    "punpckldq %%xmm0,%%xmm0                   \n"
>-    "punpckhdq %%xmm1,%%xmm1                   \n"
>-    "movdqu    %%xmm0," MEMACCESS(0) "         \n"
>-    "movdqu    %%xmm1," MEMACCESS2(0x10,0) "   \n"
>-    "lea       " MEMLEA(0x20,0) ",%0           \n"
>-    "sub       $0x8,%2                         \n"
>-    "jg        1b                              \n"
>-
>-  : "+r"(dst_argb),    // %0
>-    "+r"(src_argb),    // %1
>-    "+r"(dst_width)    // %2
>-  :: "memory", "cc", NACL_R14
>-    "xmm0", "xmm1"
>-  );
>+  asm volatile(
>+
>+      LABELALIGN
>+      "1:                                        \n"
>+      "movdqu    (%1),%%xmm0                     \n"
>+      "lea       0x10(%1),%1                     \n"
>+      "movdqa    %%xmm0,%%xmm1                   \n"
>+      "punpckldq %%xmm0,%%xmm0                   \n"
>+      "punpckhdq %%xmm1,%%xmm1                   \n"
>+      "movdqu    %%xmm0,(%0)                     \n"
>+      "movdqu    %%xmm1,0x10(%0)                 \n"
>+      "lea       0x20(%0),%0                     \n"
>+      "sub       $0x8,%2                         \n"
>+      "jg        1b                              \n"
>+
>+      : "+r"(dst_argb),  // %0
>+        "+r"(src_argb),  // %1
>+        "+r"(dst_width)  // %2
>+        ::"memory",
>+        "cc", "xmm0", "xmm1");
> }
> 
> // Shuffle table for arranging 2 pixels into pairs for pmaddubsw
>-static uvec8 kShuffleColARGB = {
>+static const uvec8 kShuffleColARGB = {
>     0u, 4u,  1u, 5u,  2u,  6u,  3u,  7u,  // bbggrraa 1st pixel
>     8u, 12u, 9u, 13u, 10u, 14u, 11u, 15u  // bbggrraa 2nd pixel
> };
> 
> // Shuffle table for duplicating 2 fractions into 8 bytes each
>-static uvec8 kShuffleFractions = {
>+static const uvec8 kShuffleFractions = {
>     0u, 0u, 0u, 0u, 0u, 0u, 0u, 0u, 4u, 4u, 4u, 4u, 4u, 4u, 4u, 4u,
> };
> 
> // Bilinear row filtering combines 4x2 -> 4x1. SSSE3 version
>-void ScaleARGBFilterCols_SSSE3(uint8* dst_argb,
>-                               const uint8* src_argb,
>+void ScaleARGBFilterCols_SSSE3(uint8_t* dst_argb,
>+                               const uint8_t* src_argb,
>                                int dst_width,
>                                int x,
>                                int dx) {
>@@ -1275,67 +1274,65 @@ void ScaleARGBFilterCols_SSSE3(uint8* dst_argb,
>         "m"(kShuffleFractions)  // %1
>       );
> 
>-  asm volatile (
>-    "movd      %5,%%xmm2                       \n"
>-    "movd      %6,%%xmm3                       \n"
>-    "pcmpeqb   %%xmm6,%%xmm6                   \n"
>-    "psrlw     $0x9,%%xmm6                     \n"
>-    "pextrw    $0x1,%%xmm2,%k3                 \n"
>-    "sub       $0x2,%2                         \n"
>-    "jl        29f                             \n"
>-    "movdqa    %%xmm2,%%xmm0                   \n"
>-    "paddd     %%xmm3,%%xmm0                   \n"
>-    "punpckldq %%xmm0,%%xmm2                   \n"
>-    "punpckldq %%xmm3,%%xmm3                   \n"
>-    "paddd     %%xmm3,%%xmm3                   \n"
>-    "pextrw    $0x3,%%xmm2,%k4                 \n"
>-
>-    LABELALIGN
>-  "2:                                          \n"
>-    "movdqa    %%xmm2,%%xmm1                   \n"
>-    "paddd     %%xmm3,%%xmm2                   \n"
>-    MEMOPREG(movq,0x00,1,3,4,xmm0)             //  movq      (%1,%3,4),%%xmm0
>-    "psrlw     $0x9,%%xmm1                     \n"
>-    MEMOPREG(movhps,0x00,1,4,4,xmm0)           //  movhps    (%1,%4,4),%%xmm0
>-    "pshufb    %%xmm5,%%xmm1                   \n"
>-    "pshufb    %%xmm4,%%xmm0                   \n"
>-    "pxor      %%xmm6,%%xmm1                   \n"
>-    "pmaddubsw %%xmm1,%%xmm0                   \n"
>-    "psrlw     $0x7,%%xmm0                     \n"
>-    "pextrw    $0x1,%%xmm2,%k3                 \n"
>-    "pextrw    $0x3,%%xmm2,%k4                 \n"
>-    "packuswb  %%xmm0,%%xmm0                   \n"
>-    "movq      %%xmm0," MEMACCESS(0) "         \n"
>-    "lea       " MEMLEA(0x8,0) ",%0            \n"
>-    "sub       $0x2,%2                         \n"
>-    "jge       2b                              \n"
>-
>-    LABELALIGN
>-  "29:                                         \n"
>-    "add       $0x1,%2                         \n"
>-    "jl        99f                             \n"
>-    "psrlw     $0x9,%%xmm2                     \n"
>-    MEMOPREG(movq,0x00,1,3,4,xmm0)             //  movq      (%1,%3,4),%%xmm0
>-    "pshufb    %%xmm5,%%xmm2                   \n"
>-    "pshufb    %%xmm4,%%xmm0                   \n"
>-    "pxor      %%xmm6,%%xmm2                   \n"
>-    "pmaddubsw %%xmm2,%%xmm0                   \n"
>-    "psrlw     $0x7,%%xmm0                     \n"
>-    "packuswb  %%xmm0,%%xmm0                   \n"
>-    "movd      %%xmm0," MEMACCESS(0) "         \n"
>-
>-    LABELALIGN
>-  "99:                                         \n"
>-  : "+r"(dst_argb),    // %0
>-    "+r"(src_argb),    // %1
>-    "+rm"(dst_width),  // %2
>-    "=&r"(x0),         // %3
>-    "=&r"(x1)          // %4
>-  : "rm"(x),           // %5
>-    "rm"(dx)           // %6
>-  : "memory", "cc", NACL_R14
>-    "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6"
>-  );
>+  asm volatile(
>+      "movd      %5,%%xmm2                       \n"
>+      "movd      %6,%%xmm3                       \n"
>+      "pcmpeqb   %%xmm6,%%xmm6                   \n"
>+      "psrlw     $0x9,%%xmm6                     \n"
>+      "pextrw    $0x1,%%xmm2,%k3                 \n"
>+      "sub       $0x2,%2                         \n"
>+      "jl        29f                             \n"
>+      "movdqa    %%xmm2,%%xmm0                   \n"
>+      "paddd     %%xmm3,%%xmm0                   \n"
>+      "punpckldq %%xmm0,%%xmm2                   \n"
>+      "punpckldq %%xmm3,%%xmm3                   \n"
>+      "paddd     %%xmm3,%%xmm3                   \n"
>+      "pextrw    $0x3,%%xmm2,%k4                 \n"
>+
>+      LABELALIGN
>+      "2:                                        \n"
>+      "movdqa    %%xmm2,%%xmm1                   \n"
>+      "paddd     %%xmm3,%%xmm2                   \n"
>+      "movq      0x00(%1,%3,4),%%xmm0            \n"
>+      "psrlw     $0x9,%%xmm1                     \n"
>+      "movhps    0x00(%1,%4,4),%%xmm0            \n"
>+      "pshufb    %%xmm5,%%xmm1                   \n"
>+      "pshufb    %%xmm4,%%xmm0                   \n"
>+      "pxor      %%xmm6,%%xmm1                   \n"
>+      "pmaddubsw %%xmm1,%%xmm0                   \n"
>+      "psrlw     $0x7,%%xmm0                     \n"
>+      "pextrw    $0x1,%%xmm2,%k3                 \n"
>+      "pextrw    $0x3,%%xmm2,%k4                 \n"
>+      "packuswb  %%xmm0,%%xmm0                   \n"
>+      "movq      %%xmm0,(%0)                     \n"
>+      "lea       0x8(%0),%0                      \n"
>+      "sub       $0x2,%2                         \n"
>+      "jge       2b                              \n"
>+
>+      LABELALIGN
>+      "29:                                       \n"
>+      "add       $0x1,%2                         \n"
>+      "jl        99f                             \n"
>+      "psrlw     $0x9,%%xmm2                     \n"
>+      "movq      0x00(%1,%3,4),%%xmm0            \n"
>+      "pshufb    %%xmm5,%%xmm2                   \n"
>+      "pshufb    %%xmm4,%%xmm0                   \n"
>+      "pxor      %%xmm6,%%xmm2                   \n"
>+      "pmaddubsw %%xmm2,%%xmm0                   \n"
>+      "psrlw     $0x7,%%xmm0                     \n"
>+      "packuswb  %%xmm0,%%xmm0                   \n"
>+      "movd      %%xmm0,(%0)                     \n"
>+
>+      LABELALIGN "99:                            \n"  // clang-format error.
>+
>+      : "+r"(dst_argb),    // %0
>+        "+r"(src_argb),    // %1
>+        "+rm"(dst_width),  // %2
>+        "=&r"(x0),         // %3
>+        "=&r"(x1)          // %4
>+      : "rm"(x),           // %5
>+        "rm"(dx)           // %6
>+      : "memory", "cc", "xmm0", "xmm1", "xmm2", "xmm3", "xmm4", "xmm5", "xmm6");
> }
> 
> // Divide num by div and return as 16.16 fixed point result.
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/scale_msa.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/scale_msa.cc
>index df1f482be6d..482a521f0d2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/scale_msa.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/scale_msa.cc
>@@ -127,13 +127,13 @@ void ScaleARGBRowDownEven_MSA(const uint8_t* src_argb,
>   }
> }
> 
>-void ScaleARGBRowDownEvenBox_MSA(const uint8* src_argb,
>+void ScaleARGBRowDownEvenBox_MSA(const uint8_t* src_argb,
>                                  ptrdiff_t src_stride,
>                                  int src_stepx,
>-                                 uint8* dst_argb,
>+                                 uint8_t* dst_argb,
>                                  int dst_width) {
>   int x;
>-  const uint8* nxt_argb = src_argb + src_stride;
>+  const uint8_t* nxt_argb = src_argb + src_stride;
>   int32_t stepx = src_stepx * 4;
>   int64_t data0, data1, data2, data3;
>   v16u8 src0 = {0}, src1 = {0}, src2 = {0}, src3 = {0};
>@@ -553,8 +553,8 @@ void ScaleAddRow_MSA(const uint8_t* src_ptr, uint16_t* dst_ptr, int src_width) {
>   }
> }
> 
>-void ScaleFilterCols_MSA(uint8* dst_ptr,
>-                         const uint8* src_ptr,
>+void ScaleFilterCols_MSA(uint8_t* dst_ptr,
>+                         const uint8_t* src_ptr,
>                          int dst_width,
>                          int x,
>                          int dx) {
>@@ -630,13 +630,13 @@ void ScaleFilterCols_MSA(uint8* dst_ptr,
>   }
> }
> 
>-void ScaleARGBCols_MSA(uint8* dst_argb,
>-                       const uint8* src_argb,
>+void ScaleARGBCols_MSA(uint8_t* dst_argb,
>+                       const uint8_t* src_argb,
>                        int dst_width,
>                        int x,
>                        int dx) {
>-  const uint32* src = (const uint32*)(src_argb);
>-  uint32* dst = (uint32*)(dst_argb);
>+  const uint32_t* src = (const uint32_t*)(src_argb);
>+  uint32_t* dst = (uint32_t*)(dst_argb);
>   int j;
>   v4i32 x_vec = __msa_fill_w(x);
>   v4i32 dx_vec = __msa_fill_w(dx);
>@@ -657,12 +657,12 @@ void ScaleARGBCols_MSA(uint8* dst_argb,
>   }
> }
> 
>-void ScaleARGBFilterCols_MSA(uint8* dst_argb,
>-                             const uint8* src_argb,
>+void ScaleARGBFilterCols_MSA(uint8_t* dst_argb,
>+                             const uint8_t* src_argb,
>                              int dst_width,
>                              int x,
>                              int dx) {
>-  const uint32* src = (const uint32*)(src_argb);
>+  const uint32_t* src = (const uint32_t*)(src_argb);
>   int j;
>   v4u32 src0, src1, src2, src3;
>   v4u32 vec0, vec1, vec2, vec3;
>@@ -722,9 +722,9 @@ void ScaleARGBFilterCols_MSA(uint8* dst_argb,
>   }
> }
> 
>-void ScaleRowDown34_MSA(const uint8* src_ptr,
>+void ScaleRowDown34_MSA(const uint8_t* src_ptr,
>                         ptrdiff_t src_stride,
>-                        uint8* dst,
>+                        uint8_t* dst,
>                         int dst_width) {
>   int x;
>   (void)src_stride;
>@@ -753,12 +753,12 @@ void ScaleRowDown34_MSA(const uint8* src_ptr,
>   }
> }
> 
>-void ScaleRowDown34_0_Box_MSA(const uint8* src_ptr,
>+void ScaleRowDown34_0_Box_MSA(const uint8_t* src_ptr,
>                               ptrdiff_t src_stride,
>-                              uint8* d,
>+                              uint8_t* d,
>                               int dst_width) {
>-  const uint8* s = src_ptr;
>-  const uint8* t = src_ptr + src_stride;
>+  const uint8_t* s = src_ptr;
>+  const uint8_t* t = src_ptr + src_stride;
>   int x;
>   v16u8 src0, src1, src2, src3, src4, src5, src6, src7, dst0, dst1, dst2;
>   v16u8 vec0, vec1, vec2, vec3, vec4, vec5;
>@@ -847,12 +847,12 @@ void ScaleRowDown34_0_Box_MSA(const uint8* src_ptr,
>   }
> }
> 
>-void ScaleRowDown34_1_Box_MSA(const uint8* src_ptr,
>+void ScaleRowDown34_1_Box_MSA(const uint8_t* src_ptr,
>                               ptrdiff_t src_stride,
>-                              uint8* d,
>+                              uint8_t* d,
>                               int dst_width) {
>-  const uint8* s = src_ptr;
>-  const uint8* t = src_ptr + src_stride;
>+  const uint8_t* s = src_ptr;
>+  const uint8_t* t = src_ptr + src_stride;
>   int x;
>   v16u8 src0, src1, src2, src3, src4, src5, src6, src7, dst0, dst1, dst2;
>   v16u8 vec0, vec1, vec2, vec3, vec4, vec5;
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/scale_neon.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/scale_neon.cc
>index b03a828213d..459a2995dfe 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/scale_neon.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/scale_neon.cc
>@@ -23,9 +23,9 @@ extern "C" {
> // Provided by Fritz Koenig
> 
> // Read 32x1 throw away even pixels, and write 16x1.
>-void ScaleRowDown2_NEON(const uint8* src_ptr,
>+void ScaleRowDown2_NEON(const uint8_t* src_ptr,
>                         ptrdiff_t src_stride,
>-                        uint8* dst,
>+                        uint8_t* dst,
>                         int dst_width) {
>   (void)src_stride;
>   asm volatile(
>@@ -44,9 +44,9 @@ void ScaleRowDown2_NEON(const uint8* src_ptr,
> }
> 
> // Read 32x1 average down and write 16x1.
>-void ScaleRowDown2Linear_NEON(const uint8* src_ptr,
>+void ScaleRowDown2Linear_NEON(const uint8_t* src_ptr,
>                               ptrdiff_t src_stride,
>-                              uint8* dst,
>+                              uint8_t* dst,
>                               int dst_width) {
>   (void)src_stride;
>   asm volatile(
>@@ -65,9 +65,9 @@ void ScaleRowDown2Linear_NEON(const uint8* src_ptr,
> }
> 
> // Read 32x2 average down and write 16x1.
>-void ScaleRowDown2Box_NEON(const uint8* src_ptr,
>+void ScaleRowDown2Box_NEON(const uint8_t* src_ptr,
>                            ptrdiff_t src_stride,
>-                           uint8* dst,
>+                           uint8_t* dst,
>                            int dst_width) {
>   asm volatile(
>       // change the stride to row 2 pointer
>@@ -95,9 +95,9 @@ void ScaleRowDown2Box_NEON(const uint8* src_ptr,
>       );
> }
> 
>-void ScaleRowDown4_NEON(const uint8* src_ptr,
>+void ScaleRowDown4_NEON(const uint8_t* src_ptr,
>                         ptrdiff_t src_stride,
>-                        uint8* dst_ptr,
>+                        uint8_t* dst_ptr,
>                         int dst_width) {
>   (void)src_stride;
>   asm volatile(
>@@ -113,13 +113,13 @@ void ScaleRowDown4_NEON(const uint8* src_ptr,
>       : "q0", "q1", "memory", "cc");
> }
> 
>-void ScaleRowDown4Box_NEON(const uint8* src_ptr,
>+void ScaleRowDown4Box_NEON(const uint8_t* src_ptr,
>                            ptrdiff_t src_stride,
>-                           uint8* dst_ptr,
>+                           uint8_t* dst_ptr,
>                            int dst_width) {
>-  const uint8* src_ptr1 = src_ptr + src_stride;
>-  const uint8* src_ptr2 = src_ptr + src_stride * 2;
>-  const uint8* src_ptr3 = src_ptr + src_stride * 3;
>+  const uint8_t* src_ptr1 = src_ptr + src_stride;
>+  const uint8_t* src_ptr2 = src_ptr + src_stride * 2;
>+  const uint8_t* src_ptr3 = src_ptr + src_stride * 3;
>   asm volatile(
>       "1:                                        \n"
>       "vld1.8     {q0}, [%0]!                    \n"  // load up 16x4
>@@ -149,9 +149,9 @@ void ScaleRowDown4Box_NEON(const uint8* src_ptr,
> // Down scale from 4 to 3 pixels. Use the neon multilane read/write
> // to load up the every 4th pixel into a 4 different registers.
> // Point samples 32 pixels to 24 pixels.
>-void ScaleRowDown34_NEON(const uint8* src_ptr,
>+void ScaleRowDown34_NEON(const uint8_t* src_ptr,
>                          ptrdiff_t src_stride,
>-                         uint8* dst_ptr,
>+                         uint8_t* dst_ptr,
>                          int dst_width) {
>   (void)src_stride;
>   asm volatile(
>@@ -168,9 +168,9 @@ void ScaleRowDown34_NEON(const uint8* src_ptr,
>       : "d0", "d1", "d2", "d3", "memory", "cc");
> }
> 
>-void ScaleRowDown34_0_Box_NEON(const uint8* src_ptr,
>+void ScaleRowDown34_0_Box_NEON(const uint8_t* src_ptr,
>                                ptrdiff_t src_stride,
>-                               uint8* dst_ptr,
>+                               uint8_t* dst_ptr,
>                                int dst_width) {
>   asm volatile(
>       "vmov.u8    d24, #3                        \n"
>@@ -225,9 +225,9 @@ void ScaleRowDown34_0_Box_NEON(const uint8* src_ptr,
>         "cc");
> }
> 
>-void ScaleRowDown34_1_Box_NEON(const uint8* src_ptr,
>+void ScaleRowDown34_1_Box_NEON(const uint8_t* src_ptr,
>                                ptrdiff_t src_stride,
>-                               uint8* dst_ptr,
>+                               uint8_t* dst_ptr,
>                                int dst_width) {
>   asm volatile(
>       "vmov.u8    d24, #3                        \n"
>@@ -264,18 +264,21 @@ void ScaleRowDown34_1_Box_NEON(const uint8* src_ptr,
> }
> 
> #define HAS_SCALEROWDOWN38_NEON
>-static uvec8 kShuf38 = {0, 3, 6, 8, 11, 14, 16, 19, 22, 24, 27, 30, 0, 0, 0, 0};
>-static uvec8 kShuf38_2 = {0,  8, 16, 2,  10, 17, 4, 12,
>-                          18, 6, 14, 19, 0,  0,  0, 0};
>-static vec16 kMult38_Div6 = {65536 / 12, 65536 / 12, 65536 / 12, 65536 / 12,
>-                             65536 / 12, 65536 / 12, 65536 / 12, 65536 / 12};
>-static vec16 kMult38_Div9 = {65536 / 18, 65536 / 18, 65536 / 18, 65536 / 18,
>-                             65536 / 18, 65536 / 18, 65536 / 18, 65536 / 18};
>+static const uvec8 kShuf38 = {0,  3,  6,  8,  11, 14, 16, 19,
>+                              22, 24, 27, 30, 0,  0,  0,  0};
>+static const uvec8 kShuf38_2 = {0,  8, 16, 2,  10, 17, 4, 12,
>+                                18, 6, 14, 19, 0,  0,  0, 0};
>+static const vec16 kMult38_Div6 = {65536 / 12, 65536 / 12, 65536 / 12,
>+                                   65536 / 12, 65536 / 12, 65536 / 12,
>+                                   65536 / 12, 65536 / 12};
>+static const vec16 kMult38_Div9 = {65536 / 18, 65536 / 18, 65536 / 18,
>+                                   65536 / 18, 65536 / 18, 65536 / 18,
>+                                   65536 / 18, 65536 / 18};
> 
> // 32 -> 12
>-void ScaleRowDown38_NEON(const uint8* src_ptr,
>+void ScaleRowDown38_NEON(const uint8_t* src_ptr,
>                          ptrdiff_t src_stride,
>-                         uint8* dst_ptr,
>+                         uint8_t* dst_ptr,
>                          int dst_width) {
>   (void)src_stride;
>   asm volatile(
>@@ -296,11 +299,11 @@ void ScaleRowDown38_NEON(const uint8* src_ptr,
> }
> 
> // 32x3 -> 12x1
>-void OMITFP ScaleRowDown38_3_Box_NEON(const uint8* src_ptr,
>+void OMITFP ScaleRowDown38_3_Box_NEON(const uint8_t* src_ptr,
>                                       ptrdiff_t src_stride,
>-                                      uint8* dst_ptr,
>+                                      uint8_t* dst_ptr,
>                                       int dst_width) {
>-  const uint8* src_ptr1 = src_ptr + src_stride * 2;
>+  const uint8_t* src_ptr1 = src_ptr + src_stride * 2;
> 
>   asm volatile(
>       "vld1.16    {q13}, [%5]                    \n"
>@@ -408,9 +411,9 @@ void OMITFP ScaleRowDown38_3_Box_NEON(const uint8* src_ptr,
> }
> 
> // 32x2 -> 12x1
>-void ScaleRowDown38_2_Box_NEON(const uint8* src_ptr,
>+void ScaleRowDown38_2_Box_NEON(const uint8_t* src_ptr,
>                                ptrdiff_t src_stride,
>-                               uint8* dst_ptr,
>+                               uint8_t* dst_ptr,
>                                int dst_width) {
>   asm volatile(
>       "vld1.16    {q13}, [%4]                    \n"
>@@ -501,12 +504,12 @@ void ScaleRowDown38_2_Box_NEON(const uint8* src_ptr,
>       : "q0", "q1", "q2", "q3", "q13", "q14", "memory", "cc");
> }
> 
>-void ScaleAddRows_NEON(const uint8* src_ptr,
>+void ScaleAddRows_NEON(const uint8_t* src_ptr,
>                        ptrdiff_t src_stride,
>-                       uint16* dst_ptr,
>+                       uint16_t* dst_ptr,
>                        int src_width,
>                        int src_height) {
>-  const uint8* src_tmp;
>+  const uint8_t* src_tmp;
>   asm volatile(
>       "1:                                        \n"
>       "mov       %0, %1                          \n"
>@@ -544,17 +547,17 @@ void ScaleAddRows_NEON(const uint8* src_ptr,
>   "vld2.8     {d6[" #n "], d7[" #n "]}, [%6] \n"
> 
> // The NEON version mimics this formula (from row_common.cc):
>-// #define BLENDER(a, b, f) (uint8)((int)(a) +
>+// #define BLENDER(a, b, f) (uint8_t)((int)(a) +
> //    ((((int)((f)) * ((int)(b) - (int)(a))) + 0x8000) >> 16))
> 
>-void ScaleFilterCols_NEON(uint8* dst_ptr,
>-                          const uint8* src_ptr,
>+void ScaleFilterCols_NEON(uint8_t* dst_ptr,
>+                          const uint8_t* src_ptr,
>                           int dst_width,
>                           int x,
>                           int dx) {
>   int dx_offset[4] = {0, 1, 2, 3};
>   int* tmp = dx_offset;
>-  const uint8* src_tmp = src_ptr;
>+  const uint8_t* src_tmp = src_ptr;
>   asm volatile (
>     "vdup.32    q0, %3                         \n"  // x
>     "vdup.32    q1, %4                         \n"  // dx
>@@ -612,8 +615,8 @@ void ScaleFilterCols_NEON(uint8* dst_ptr,
> #undef LOAD2_DATA8_LANE
> 
> // 16x2 -> 16x1
>-void ScaleFilterRows_NEON(uint8* dst_ptr,
>-                          const uint8* src_ptr,
>+void ScaleFilterRows_NEON(uint8_t* dst_ptr,
>+                          const uint8_t* src_ptr,
>                           ptrdiff_t src_stride,
>                           int dst_width,
>                           int source_y_fraction) {
>@@ -696,9 +699,9 @@ void ScaleFilterRows_NEON(uint8* dst_ptr,
>       : "q0", "q1", "d4", "d5", "q13", "q14", "memory", "cc");
> }
> 
>-void ScaleARGBRowDown2_NEON(const uint8* src_ptr,
>+void ScaleARGBRowDown2_NEON(const uint8_t* src_ptr,
>                             ptrdiff_t src_stride,
>-                            uint8* dst,
>+                            uint8_t* dst,
>                             int dst_width) {
>   (void)src_stride;
>   asm volatile(
>@@ -724,9 +727,9 @@ void ScaleARGBRowDown2_NEON(const uint8* src_ptr,
> //  54:  f942 038d   vst2.32  {d16-d19}, [r2]!
> //  58:  d1f5        bne.n  46 <ScaleARGBRowDown2_C+0x46>
> 
>-void ScaleARGBRowDown2Linear_NEON(const uint8* src_argb,
>+void ScaleARGBRowDown2Linear_NEON(const uint8_t* src_argb,
>                                   ptrdiff_t src_stride,
>-                                  uint8* dst_argb,
>+                                  uint8_t* dst_argb,
>                                   int dst_width) {
>   (void)src_stride;
>   asm volatile(
>@@ -746,9 +749,9 @@ void ScaleARGBRowDown2Linear_NEON(const uint8* src_argb,
>       );
> }
> 
>-void ScaleARGBRowDown2Box_NEON(const uint8* src_ptr,
>+void ScaleARGBRowDown2Box_NEON(const uint8_t* src_ptr,
>                                ptrdiff_t src_stride,
>-                               uint8* dst,
>+                               uint8_t* dst,
>                                int dst_width) {
>   asm volatile(
>       // change the stride to row 2 pointer
>@@ -783,10 +786,10 @@ void ScaleARGBRowDown2Box_NEON(const uint8* src_ptr,
> 
> // Reads 4 pixels at a time.
> // Alignment requirement: src_argb 4 byte aligned.
>-void ScaleARGBRowDownEven_NEON(const uint8* src_argb,
>+void ScaleARGBRowDownEven_NEON(const uint8_t* src_argb,
>                                ptrdiff_t src_stride,
>                                int src_stepx,
>-                               uint8* dst_argb,
>+                               uint8_t* dst_argb,
>                                int dst_width) {
>   (void)src_stride;
>   asm volatile(
>@@ -808,10 +811,10 @@ void ScaleARGBRowDownEven_NEON(const uint8* src_argb,
> 
> // Reads 4 pixels at a time.
> // Alignment requirement: src_argb 4 byte aligned.
>-void ScaleARGBRowDownEvenBox_NEON(const uint8* src_argb,
>+void ScaleARGBRowDownEvenBox_NEON(const uint8_t* src_argb,
>                                   ptrdiff_t src_stride,
>                                   int src_stepx,
>-                                  uint8* dst_argb,
>+                                  uint8_t* dst_argb,
>                                   int dst_width) {
>   asm volatile(
>       "mov        r12, %4, lsl #2                \n"
>@@ -854,13 +857,13 @@ void ScaleARGBRowDownEvenBox_NEON(const uint8* src_argb,
>   "add        %3, %3, %4                     \n" \
>   "vld1.32    {" #dn "[" #n "]}, [%6]        \n"
> 
>-void ScaleARGBCols_NEON(uint8* dst_argb,
>-                        const uint8* src_argb,
>+void ScaleARGBCols_NEON(uint8_t* dst_argb,
>+                        const uint8_t* src_argb,
>                         int dst_width,
>                         int x,
>                         int dx) {
>   int tmp;
>-  const uint8* src_tmp = src_argb;
>+  const uint8_t* src_tmp = src_argb;
>   asm volatile(
>       "1:                                        \n"
>       // clang-format off
>@@ -897,14 +900,14 @@ void ScaleARGBCols_NEON(uint8* dst_argb,
>   "add        %3, %3, %4                                 \n" \
>   "vld2.32    {" #dn1 "[" #n "], " #dn2 "[" #n "]}, [%6] \n"
> 
>-void ScaleARGBFilterCols_NEON(uint8* dst_argb,
>-                              const uint8* src_argb,
>+void ScaleARGBFilterCols_NEON(uint8_t* dst_argb,
>+                              const uint8_t* src_argb,
>                               int dst_width,
>                               int x,
>                               int dx) {
>   int dx_offset[4] = {0, 1, 2, 3};
>   int* tmp = dx_offset;
>-  const uint8* src_tmp = src_argb;
>+  const uint8_t* src_tmp = src_argb;
>   asm volatile (
>     "vdup.32    q0, %3                         \n"  // x
>     "vdup.32    q1, %4                         \n"  // dx
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/scale_neon64.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/scale_neon64.cc
>index 93fe67bf1e6..494a9cfbfbe 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/scale_neon64.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/scale_neon64.cc
>@@ -21,9 +21,9 @@ extern "C" {
> #if !defined(LIBYUV_DISABLE_NEON) && defined(__aarch64__)
> 
> // Read 32x1 throw away even pixels, and write 16x1.
>-void ScaleRowDown2_NEON(const uint8* src_ptr,
>+void ScaleRowDown2_NEON(const uint8_t* src_ptr,
>                         ptrdiff_t src_stride,
>-                        uint8* dst,
>+                        uint8_t* dst,
>                         int dst_width) {
>   (void)src_stride;
>   asm volatile(
>@@ -42,9 +42,9 @@ void ScaleRowDown2_NEON(const uint8* src_ptr,
> }
> 
> // Read 32x1 average down and write 16x1.
>-void ScaleRowDown2Linear_NEON(const uint8* src_ptr,
>+void ScaleRowDown2Linear_NEON(const uint8_t* src_ptr,
>                               ptrdiff_t src_stride,
>-                              uint8* dst,
>+                              uint8_t* dst,
>                               int dst_width) {
>   (void)src_stride;
>   asm volatile(
>@@ -64,9 +64,9 @@ void ScaleRowDown2Linear_NEON(const uint8* src_ptr,
> }
> 
> // Read 32x2 average down and write 16x1.
>-void ScaleRowDown2Box_NEON(const uint8* src_ptr,
>+void ScaleRowDown2Box_NEON(const uint8_t* src_ptr,
>                            ptrdiff_t src_stride,
>-                           uint8* dst,
>+                           uint8_t* dst,
>                            int dst_width) {
>   asm volatile(
>       // change the stride to row 2 pointer
>@@ -92,9 +92,9 @@ void ScaleRowDown2Box_NEON(const uint8* src_ptr,
>       );
> }
> 
>-void ScaleRowDown4_NEON(const uint8* src_ptr,
>+void ScaleRowDown4_NEON(const uint8_t* src_ptr,
>                         ptrdiff_t src_stride,
>-                        uint8* dst_ptr,
>+                        uint8_t* dst_ptr,
>                         int dst_width) {
>   (void)src_stride;
>   asm volatile(
>@@ -110,13 +110,13 @@ void ScaleRowDown4_NEON(const uint8* src_ptr,
>       : "v0", "v1", "v2", "v3", "memory", "cc");
> }
> 
>-void ScaleRowDown4Box_NEON(const uint8* src_ptr,
>+void ScaleRowDown4Box_NEON(const uint8_t* src_ptr,
>                            ptrdiff_t src_stride,
>-                           uint8* dst_ptr,
>+                           uint8_t* dst_ptr,
>                            int dst_width) {
>-  const uint8* src_ptr1 = src_ptr + src_stride;
>-  const uint8* src_ptr2 = src_ptr + src_stride * 2;
>-  const uint8* src_ptr3 = src_ptr + src_stride * 3;
>+  const uint8_t* src_ptr1 = src_ptr + src_stride;
>+  const uint8_t* src_ptr2 = src_ptr + src_stride * 2;
>+  const uint8_t* src_ptr3 = src_ptr + src_stride * 3;
>   asm volatile(
>       "1:                                        \n"
>       "ld1     {v0.16b}, [%0], #16               \n"  // load up 16x4
>@@ -145,9 +145,9 @@ void ScaleRowDown4Box_NEON(const uint8* src_ptr,
> // Down scale from 4 to 3 pixels. Use the neon multilane read/write
> // to load up the every 4th pixel into a 4 different registers.
> // Point samples 32 pixels to 24 pixels.
>-void ScaleRowDown34_NEON(const uint8* src_ptr,
>+void ScaleRowDown34_NEON(const uint8_t* src_ptr,
>                          ptrdiff_t src_stride,
>-                         uint8* dst_ptr,
>+                         uint8_t* dst_ptr,
>                          int dst_width) {
>   (void)src_stride;
>   asm volatile(
>@@ -164,9 +164,9 @@ void ScaleRowDown34_NEON(const uint8* src_ptr,
>       : "v0", "v1", "v2", "v3", "memory", "cc");
> }
> 
>-void ScaleRowDown34_0_Box_NEON(const uint8* src_ptr,
>+void ScaleRowDown34_0_Box_NEON(const uint8_t* src_ptr,
>                                ptrdiff_t src_stride,
>-                               uint8* dst_ptr,
>+                               uint8_t* dst_ptr,
>                                int dst_width) {
>   asm volatile(
>       "movi      v20.8b, #3                              \n"
>@@ -221,9 +221,9 @@ void ScaleRowDown34_0_Box_NEON(const uint8* src_ptr,
>         "v19", "v20", "memory", "cc");
> }
> 
>-void ScaleRowDown34_1_Box_NEON(const uint8* src_ptr,
>+void ScaleRowDown34_1_Box_NEON(const uint8_t* src_ptr,
>                                ptrdiff_t src_stride,
>-                               uint8* dst_ptr,
>+                               uint8_t* dst_ptr,
>                                int dst_width) {
>   asm volatile(
>       "movi      v20.8b, #3                              \n"
>@@ -261,18 +261,21 @@ void ScaleRowDown34_1_Box_NEON(const uint8* src_ptr,
>       : "v0", "v1", "v2", "v3", "v4", "v5", "v6", "v7", "v20", "memory", "cc");
> }
> 
>-static uvec8 kShuf38 = {0, 3, 6, 8, 11, 14, 16, 19, 22, 24, 27, 30, 0, 0, 0, 0};
>-static uvec8 kShuf38_2 = {0,  16, 32, 2,  18, 33, 4, 20,
>-                          34, 6,  22, 35, 0,  0,  0, 0};
>-static vec16 kMult38_Div6 = {65536 / 12, 65536 / 12, 65536 / 12, 65536 / 12,
>-                             65536 / 12, 65536 / 12, 65536 / 12, 65536 / 12};
>-static vec16 kMult38_Div9 = {65536 / 18, 65536 / 18, 65536 / 18, 65536 / 18,
>-                             65536 / 18, 65536 / 18, 65536 / 18, 65536 / 18};
>+static const uvec8 kShuf38 = {0,  3,  6,  8,  11, 14, 16, 19,
>+                              22, 24, 27, 30, 0,  0,  0,  0};
>+static const uvec8 kShuf38_2 = {0,  16, 32, 2,  18, 33, 4, 20,
>+                                34, 6,  22, 35, 0,  0,  0, 0};
>+static const vec16 kMult38_Div6 = {65536 / 12, 65536 / 12, 65536 / 12,
>+                                   65536 / 12, 65536 / 12, 65536 / 12,
>+                                   65536 / 12, 65536 / 12};
>+static const vec16 kMult38_Div9 = {65536 / 18, 65536 / 18, 65536 / 18,
>+                                   65536 / 18, 65536 / 18, 65536 / 18,
>+                                   65536 / 18, 65536 / 18};
> 
> // 32 -> 12
>-void ScaleRowDown38_NEON(const uint8* src_ptr,
>+void ScaleRowDown38_NEON(const uint8_t* src_ptr,
>                          ptrdiff_t src_stride,
>-                         uint8* dst_ptr,
>+                         uint8_t* dst_ptr,
>                          int dst_width) {
>   (void)src_stride;
>   asm volatile(
>@@ -292,11 +295,11 @@ void ScaleRowDown38_NEON(const uint8* src_ptr,
> }
> 
> // 32x3 -> 12x1
>-void OMITFP ScaleRowDown38_3_Box_NEON(const uint8* src_ptr,
>+void OMITFP ScaleRowDown38_3_Box_NEON(const uint8_t* src_ptr,
>                                       ptrdiff_t src_stride,
>-                                      uint8* dst_ptr,
>+                                      uint8_t* dst_ptr,
>                                       int dst_width) {
>-  const uint8* src_ptr1 = src_ptr + src_stride * 2;
>+  const uint8_t* src_ptr1 = src_ptr + src_stride * 2;
>   ptrdiff_t tmp_src_stride = src_stride;
> 
>   asm volatile(
>@@ -412,9 +415,9 @@ void OMITFP ScaleRowDown38_3_Box_NEON(const uint8* src_ptr,
> }
> 
> // 32x2 -> 12x1
>-void ScaleRowDown38_2_Box_NEON(const uint8* src_ptr,
>+void ScaleRowDown38_2_Box_NEON(const uint8_t* src_ptr,
>                                ptrdiff_t src_stride,
>-                               uint8* dst_ptr,
>+                               uint8_t* dst_ptr,
>                                int dst_width) {
>   // TODO(fbarchard): use src_stride directly for clang 3.5+.
>   ptrdiff_t tmp_src_stride = src_stride;
>@@ -512,12 +515,12 @@ void ScaleRowDown38_2_Box_NEON(const uint8* src_ptr,
>         "v19", "v30", "v31", "memory", "cc");
> }
> 
>-void ScaleAddRows_NEON(const uint8* src_ptr,
>+void ScaleAddRows_NEON(const uint8_t* src_ptr,
>                        ptrdiff_t src_stride,
>-                       uint16* dst_ptr,
>+                       uint16_t* dst_ptr,
>                        int src_width,
>                        int src_height) {
>-  const uint8* src_tmp;
>+  const uint8_t* src_tmp;
>   asm volatile(
>       "1:                                        \n"
>       "mov       %0, %1                          \n"
>@@ -555,19 +558,19 @@ void ScaleAddRows_NEON(const uint8* src_ptr,
>   "ld2        {v4.b, v5.b}[" #n "], [%6]     \n"
> 
> // The NEON version mimics this formula (from row_common.cc):
>-// #define BLENDER(a, b, f) (uint8)((int)(a) +
>+// #define BLENDER(a, b, f) (uint8_t)((int)(a) +
> //    ((((int)((f)) * ((int)(b) - (int)(a))) + 0x8000) >> 16))
> 
>-void ScaleFilterCols_NEON(uint8* dst_ptr,
>-                          const uint8* src_ptr,
>+void ScaleFilterCols_NEON(uint8_t* dst_ptr,
>+                          const uint8_t* src_ptr,
>                           int dst_width,
>                           int x,
>                           int dx) {
>   int dx_offset[4] = {0, 1, 2, 3};
>   int* tmp = dx_offset;
>-  const uint8* src_tmp = src_ptr;
>-  int64 x64 = (int64)x;    // NOLINT
>-  int64 dx64 = (int64)dx;  // NOLINT
>+  const uint8_t* src_tmp = src_ptr;
>+  int64_t x64 = (int64_t)x;    // NOLINT
>+  int64_t dx64 = (int64_t)dx;  // NOLINT
>   asm volatile (
>     "dup        v0.4s, %w3                     \n"  // x
>     "dup        v1.4s, %w4                     \n"  // dx
>@@ -625,8 +628,8 @@ void ScaleFilterCols_NEON(uint8* dst_ptr,
> #undef LOAD2_DATA8_LANE
> 
> // 16x2 -> 16x1
>-void ScaleFilterRows_NEON(uint8* dst_ptr,
>-                          const uint8* src_ptr,
>+void ScaleFilterRows_NEON(uint8_t* dst_ptr,
>+                          const uint8_t* src_ptr,
>                           ptrdiff_t src_stride,
>                           int dst_width,
>                           int source_y_fraction) {
>@@ -710,9 +713,9 @@ void ScaleFilterRows_NEON(uint8* dst_ptr,
>       : "v0", "v1", "v2", "v3", "v4", "v5", "v6", "v7", "memory", "cc");
> }
> 
>-void ScaleARGBRowDown2_NEON(const uint8* src_ptr,
>+void ScaleARGBRowDown2_NEON(const uint8_t* src_ptr,
>                             ptrdiff_t src_stride,
>-                            uint8* dst,
>+                            uint8_t* dst,
>                             int dst_width) {
>   (void)src_stride;
>   asm volatile(
>@@ -731,9 +734,9 @@ void ScaleARGBRowDown2_NEON(const uint8* src_ptr,
>       );
> }
> 
>-void ScaleARGBRowDown2Linear_NEON(const uint8* src_argb,
>+void ScaleARGBRowDown2Linear_NEON(const uint8_t* src_argb,
>                                   ptrdiff_t src_stride,
>-                                  uint8* dst_argb,
>+                                  uint8_t* dst_argb,
>                                   int dst_width) {
>   (void)src_stride;
>   asm volatile(
>@@ -754,9 +757,9 @@ void ScaleARGBRowDown2Linear_NEON(const uint8* src_argb,
>       );
> }
> 
>-void ScaleARGBRowDown2Box_NEON(const uint8* src_ptr,
>+void ScaleARGBRowDown2Box_NEON(const uint8_t* src_ptr,
>                                ptrdiff_t src_stride,
>-                               uint8* dst,
>+                               uint8_t* dst,
>                                int dst_width) {
>   asm volatile(
>       // change the stride to row 2 pointer
>@@ -789,10 +792,10 @@ void ScaleARGBRowDown2Box_NEON(const uint8* src_ptr,
> 
> // Reads 4 pixels at a time.
> // Alignment requirement: src_argb 4 byte aligned.
>-void ScaleARGBRowDownEven_NEON(const uint8* src_argb,
>+void ScaleARGBRowDownEven_NEON(const uint8_t* src_argb,
>                                ptrdiff_t src_stride,
>                                int src_stepx,
>-                               uint8* dst_argb,
>+                               uint8_t* dst_argb,
>                                int dst_width) {
>   (void)src_stride;
>   asm volatile(
>@@ -804,10 +807,10 @@ void ScaleARGBRowDownEven_NEON(const uint8* src_argb,
>       "subs       %w2, %w2, #4                   \n"  // 4 pixels per loop.
>       "st1        {v0.16b}, [%1], #16            \n"
>       "b.gt       1b                             \n"
>-      : "+r"(src_argb),              // %0
>-        "+r"(dst_argb),              // %1
>-        "+r"(dst_width)              // %2
>-      : "r"((int64)(src_stepx * 4))  // %3
>+      : "+r"(src_argb),                // %0
>+        "+r"(dst_argb),                // %1
>+        "+r"(dst_width)                // %2
>+      : "r"((int64_t)(src_stepx * 4))  // %3
>       : "memory", "cc", "v0");
> }
> 
>@@ -815,10 +818,10 @@ void ScaleARGBRowDownEven_NEON(const uint8* src_argb,
> // Alignment requirement: src_argb 4 byte aligned.
> // TODO(Yang Zhang): Might be worth another optimization pass in future.
> // It could be upgraded to 8 pixels at a time to start with.
>-void ScaleARGBRowDownEvenBox_NEON(const uint8* src_argb,
>+void ScaleARGBRowDownEvenBox_NEON(const uint8_t* src_argb,
>                                   ptrdiff_t src_stride,
>                                   int src_stepx,
>-                                  uint8* dst_argb,
>+                                  uint8_t* dst_argb,
>                                   int dst_width) {
>   asm volatile(
>       "add        %1, %1, %0                     \n"
>@@ -848,11 +851,11 @@ void ScaleARGBRowDownEvenBox_NEON(const uint8* src_argb,
>       "subs       %w3, %w3, #4                   \n"  // 4 pixels per loop.
>       "st1     {v0.16b}, [%2], #16               \n"
>       "b.gt       1b                             \n"
>-      : "+r"(src_argb),              // %0
>-        "+r"(src_stride),            // %1
>-        "+r"(dst_argb),              // %2
>-        "+r"(dst_width)              // %3
>-      : "r"((int64)(src_stepx * 4))  // %4
>+      : "+r"(src_argb),                // %0
>+        "+r"(src_stride),              // %1
>+        "+r"(dst_argb),                // %2
>+        "+r"(dst_width)                // %3
>+      : "r"((int64_t)(src_stepx * 4))  // %4
>       : "memory", "cc", "v0", "v1", "v2", "v3", "v4", "v5", "v6", "v7", "v16");
> }
> 
>@@ -864,15 +867,15 @@ void ScaleARGBRowDownEvenBox_NEON(const uint8* src_argb,
>   "add        %3, %3, %4                     \n" \
>   "ld1        {" #vn ".s}[" #n "], [%6]      \n"
> 
>-void ScaleARGBCols_NEON(uint8* dst_argb,
>-                        const uint8* src_argb,
>+void ScaleARGBCols_NEON(uint8_t* dst_argb,
>+                        const uint8_t* src_argb,
>                         int dst_width,
>                         int x,
>                         int dx) {
>-  const uint8* src_tmp = src_argb;
>-  int64 x64 = (int64)x;    // NOLINT
>-  int64 dx64 = (int64)dx;  // NOLINT
>-  int64 tmp64;
>+  const uint8_t* src_tmp = src_argb;
>+  int64_t x64 = (int64_t)x;    // NOLINT
>+  int64_t dx64 = (int64_t)dx;  // NOLINT
>+  int64_t tmp64;
>   asm volatile(
>       "1:                                        \n"
>       // clang-format off
>@@ -909,16 +912,16 @@ void ScaleARGBCols_NEON(uint8* dst_argb,
>   "add        %3, %3, %4                            \n" \
>   "ld2        {" #vn1 ".s, " #vn2 ".s}[" #n "], [%6]  \n"
> 
>-void ScaleARGBFilterCols_NEON(uint8* dst_argb,
>-                              const uint8* src_argb,
>+void ScaleARGBFilterCols_NEON(uint8_t* dst_argb,
>+                              const uint8_t* src_argb,
>                               int dst_width,
>                               int x,
>                               int dx) {
>   int dx_offset[4] = {0, 1, 2, 3};
>   int* tmp = dx_offset;
>-  const uint8* src_tmp = src_argb;
>-  int64 x64 = (int64)x;    // NOLINT
>-  int64 dx64 = (int64)dx;  // NOLINT
>+  const uint8_t* src_tmp = src_argb;
>+  int64_t x64 = (int64_t)x;    // NOLINT
>+  int64_t dx64 = (int64_t)dx;  // NOLINT
>   asm volatile (
>     "dup        v0.4s, %w3                     \n"  // x
>     "dup        v1.4s, %w4                     \n"  // dx
>@@ -975,9 +978,9 @@ void ScaleARGBFilterCols_NEON(uint8* dst_argb,
> #undef LOAD2_DATA32_LANE
> 
> // Read 16x2 average down and write 8x1.
>-void ScaleRowDown2Box_16_NEON(const uint16* src_ptr,
>+void ScaleRowDown2Box_16_NEON(const uint16_t* src_ptr,
>                               ptrdiff_t src_stride,
>-                              uint16* dst,
>+                              uint16_t* dst,
>                               int dst_width) {
>   asm volatile(
>       // change the stride to row 2 pointer
>@@ -1005,9 +1008,9 @@ void ScaleRowDown2Box_16_NEON(const uint16* src_ptr,
> 
> // Read 8x2 upsample with filtering and write 16x1.
> // Actually reads an extra pixel, so 9x2.
>-void ScaleRowUp2_16_NEON(const uint16* src_ptr,
>+void ScaleRowUp2_16_NEON(const uint16_t* src_ptr,
>                          ptrdiff_t src_stride,
>-                         uint16* dst,
>+                         uint16_t* dst,
>                          int dst_width) {
>   asm volatile(
>       "add        %1, %0, %1, lsl #1             \n"  // ptr + stide * 2
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/scale_win.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/scale_win.cc
>index b5fd6638262..c5fc86f3e96 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/scale_win.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/scale_win.cc
>@@ -20,77 +20,78 @@ extern "C" {
> #if !defined(LIBYUV_DISABLE_X86) && defined(_M_IX86) && defined(_MSC_VER)
> 
> // Offsets for source bytes 0 to 9
>-static uvec8 kShuf0 = {0,   1,   3,   4,   5,   7,   8,   9,
>-                       128, 128, 128, 128, 128, 128, 128, 128};
>+static const uvec8 kShuf0 = {0,   1,   3,   4,   5,   7,   8,   9,
>+                             128, 128, 128, 128, 128, 128, 128, 128};
> 
> // Offsets for source bytes 11 to 20 with 8 subtracted = 3 to 12.
>-static uvec8 kShuf1 = {3,   4,   5,   7,   8,   9,   11,  12,
>-                       128, 128, 128, 128, 128, 128, 128, 128};
>+static const uvec8 kShuf1 = {3,   4,   5,   7,   8,   9,   11,  12,
>+                             128, 128, 128, 128, 128, 128, 128, 128};
> 
> // Offsets for source bytes 21 to 31 with 16 subtracted = 5 to 31.
>-static uvec8 kShuf2 = {5,   7,   8,   9,   11,  12,  13,  15,
>-                       128, 128, 128, 128, 128, 128, 128, 128};
>+static const uvec8 kShuf2 = {5,   7,   8,   9,   11,  12,  13,  15,
>+                             128, 128, 128, 128, 128, 128, 128, 128};
> 
> // Offsets for source bytes 0 to 10
>-static uvec8 kShuf01 = {0, 1, 1, 2, 2, 3, 4, 5, 5, 6, 6, 7, 8, 9, 9, 10};
>+static const uvec8 kShuf01 = {0, 1, 1, 2, 2, 3, 4, 5, 5, 6, 6, 7, 8, 9, 9, 10};
> 
> // Offsets for source bytes 10 to 21 with 8 subtracted = 3 to 13.
>-static uvec8 kShuf11 = {2, 3, 4, 5, 5, 6, 6, 7, 8, 9, 9, 10, 10, 11, 12, 13};
>+static const uvec8 kShuf11 = {2, 3, 4, 5,  5,  6,  6,  7,
>+                              8, 9, 9, 10, 10, 11, 12, 13};
> 
> // Offsets for source bytes 21 to 31 with 16 subtracted = 5 to 31.
>-static uvec8 kShuf21 = {5,  6,  6,  7,  8,  9,  9,  10,
>-                        10, 11, 12, 13, 13, 14, 14, 15};
>+static const uvec8 kShuf21 = {5,  6,  6,  7,  8,  9,  9,  10,
>+                              10, 11, 12, 13, 13, 14, 14, 15};
> 
> // Coefficients for source bytes 0 to 10
>-static uvec8 kMadd01 = {3, 1, 2, 2, 1, 3, 3, 1, 2, 2, 1, 3, 3, 1, 2, 2};
>+static const uvec8 kMadd01 = {3, 1, 2, 2, 1, 3, 3, 1, 2, 2, 1, 3, 3, 1, 2, 2};
> 
> // Coefficients for source bytes 10 to 21
>-static uvec8 kMadd11 = {1, 3, 3, 1, 2, 2, 1, 3, 3, 1, 2, 2, 1, 3, 3, 1};
>+static const uvec8 kMadd11 = {1, 3, 3, 1, 2, 2, 1, 3, 3, 1, 2, 2, 1, 3, 3, 1};
> 
> // Coefficients for source bytes 21 to 31
>-static uvec8 kMadd21 = {2, 2, 1, 3, 3, 1, 2, 2, 1, 3, 3, 1, 2, 2, 1, 3};
>+static const uvec8 kMadd21 = {2, 2, 1, 3, 3, 1, 2, 2, 1, 3, 3, 1, 2, 2, 1, 3};
> 
> // Coefficients for source bytes 21 to 31
>-static vec16 kRound34 = {2, 2, 2, 2, 2, 2, 2, 2};
>+static const vec16 kRound34 = {2, 2, 2, 2, 2, 2, 2, 2};
> 
>-static uvec8 kShuf38a = {0,   3,   6,   8,   11,  14,  128, 128,
>-                         128, 128, 128, 128, 128, 128, 128, 128};
>+static const uvec8 kShuf38a = {0,   3,   6,   8,   11,  14,  128, 128,
>+                               128, 128, 128, 128, 128, 128, 128, 128};
> 
>-static uvec8 kShuf38b = {128, 128, 128, 128, 128, 128, 0,   3,
>-                         6,   8,   11,  14,  128, 128, 128, 128};
>+static const uvec8 kShuf38b = {128, 128, 128, 128, 128, 128, 0,   3,
>+                               6,   8,   11,  14,  128, 128, 128, 128};
> 
> // Arrange words 0,3,6 into 0,1,2
>-static uvec8 kShufAc = {0,   1,   6,   7,   12,  13,  128, 128,
>-                        128, 128, 128, 128, 128, 128, 128, 128};
>+static const uvec8 kShufAc = {0,   1,   6,   7,   12,  13,  128, 128,
>+                              128, 128, 128, 128, 128, 128, 128, 128};
> 
> // Arrange words 0,3,6 into 3,4,5
>-static uvec8 kShufAc3 = {128, 128, 128, 128, 128, 128, 0,   1,
>-                         6,   7,   12,  13,  128, 128, 128, 128};
>+static const uvec8 kShufAc3 = {128, 128, 128, 128, 128, 128, 0,   1,
>+                               6,   7,   12,  13,  128, 128, 128, 128};
> 
> // Scaling values for boxes of 3x3 and 2x3
>-static uvec16 kScaleAc33 = {65536 / 9, 65536 / 9, 65536 / 6, 65536 / 9,
>-                            65536 / 9, 65536 / 6, 0,         0};
>+static const uvec16 kScaleAc33 = {65536 / 9, 65536 / 9, 65536 / 6, 65536 / 9,
>+                                  65536 / 9, 65536 / 6, 0,         0};
> 
> // Arrange first value for pixels 0,1,2,3,4,5
>-static uvec8 kShufAb0 = {0,  128, 3,  128, 6,   128, 8,   128,
>-                         11, 128, 14, 128, 128, 128, 128, 128};
>+static const uvec8 kShufAb0 = {0,  128, 3,  128, 6,   128, 8,   128,
>+                               11, 128, 14, 128, 128, 128, 128, 128};
> 
> // Arrange second value for pixels 0,1,2,3,4,5
>-static uvec8 kShufAb1 = {1,  128, 4,  128, 7,   128, 9,   128,
>-                         12, 128, 15, 128, 128, 128, 128, 128};
>+static const uvec8 kShufAb1 = {1,  128, 4,  128, 7,   128, 9,   128,
>+                               12, 128, 15, 128, 128, 128, 128, 128};
> 
> // Arrange third value for pixels 0,1,2,3,4,5
>-static uvec8 kShufAb2 = {2,  128, 5,   128, 128, 128, 10,  128,
>-                         13, 128, 128, 128, 128, 128, 128, 128};
>+static const uvec8 kShufAb2 = {2,  128, 5,   128, 128, 128, 10,  128,
>+                               13, 128, 128, 128, 128, 128, 128, 128};
> 
> // Scaling values for boxes of 3x2 and 2x2
>-static uvec16 kScaleAb2 = {65536 / 3, 65536 / 3, 65536 / 2, 65536 / 3,
>-                           65536 / 3, 65536 / 2, 0,         0};
>+static const uvec16 kScaleAb2 = {65536 / 3, 65536 / 3, 65536 / 2, 65536 / 3,
>+                                 65536 / 3, 65536 / 2, 0,         0};
> 
> // Reads 32 pixels, throws half away and writes 16 pixels.
>-__declspec(naked) void ScaleRowDown2_SSSE3(const uint8* src_ptr,
>+__declspec(naked) void ScaleRowDown2_SSSE3(const uint8_t* src_ptr,
>                                            ptrdiff_t src_stride,
>-                                           uint8* dst_ptr,
>+                                           uint8_t* dst_ptr,
>                                            int dst_width) {
>   __asm {
>     mov        eax, [esp + 4]  // src_ptr
>@@ -115,9 +116,9 @@ __declspec(naked) void ScaleRowDown2_SSSE3(const uint8* src_ptr,
> }
> 
> // Blends 32x1 rectangle to 16x1.
>-__declspec(naked) void ScaleRowDown2Linear_SSSE3(const uint8* src_ptr,
>+__declspec(naked) void ScaleRowDown2Linear_SSSE3(const uint8_t* src_ptr,
>                                                  ptrdiff_t src_stride,
>-                                                 uint8* dst_ptr,
>+                                                 uint8_t* dst_ptr,
>                                                  int dst_width) {
>   __asm {
>     mov        eax, [esp + 4]  // src_ptr
>@@ -149,9 +150,9 @@ __declspec(naked) void ScaleRowDown2Linear_SSSE3(const uint8* src_ptr,
> }
> 
> // Blends 32x2 rectangle to 16x1.
>-__declspec(naked) void ScaleRowDown2Box_SSSE3(const uint8* src_ptr,
>+__declspec(naked) void ScaleRowDown2Box_SSSE3(const uint8_t* src_ptr,
>                                               ptrdiff_t src_stride,
>-                                              uint8* dst_ptr,
>+                                              uint8_t* dst_ptr,
>                                               int dst_width) {
>   __asm {
>     push       esi
>@@ -194,9 +195,9 @@ __declspec(naked) void ScaleRowDown2Box_SSSE3(const uint8* src_ptr,
> 
> #ifdef HAS_SCALEROWDOWN2_AVX2
> // Reads 64 pixels, throws half away and writes 32 pixels.
>-__declspec(naked) void ScaleRowDown2_AVX2(const uint8* src_ptr,
>+__declspec(naked) void ScaleRowDown2_AVX2(const uint8_t* src_ptr,
>                                           ptrdiff_t src_stride,
>-                                          uint8* dst_ptr,
>+                                          uint8_t* dst_ptr,
>                                           int dst_width) {
>   __asm {
>     mov        eax, [esp + 4]  // src_ptr
>@@ -223,9 +224,9 @@ __declspec(naked) void ScaleRowDown2_AVX2(const uint8* src_ptr,
> }
> 
> // Blends 64x1 rectangle to 32x1.
>-__declspec(naked) void ScaleRowDown2Linear_AVX2(const uint8* src_ptr,
>+__declspec(naked) void ScaleRowDown2Linear_AVX2(const uint8_t* src_ptr,
>                                                 ptrdiff_t src_stride,
>-                                                uint8* dst_ptr,
>+                                                uint8_t* dst_ptr,
>                                                 int dst_width) {
>   __asm {
>     mov         eax, [esp + 4]  // src_ptr
>@@ -261,9 +262,9 @@ __declspec(naked) void ScaleRowDown2Linear_AVX2(const uint8* src_ptr,
> // For rounding, average = (sum + 2) / 4
> // becomes average((sum >> 1), 0)
> // Blends 64x2 rectangle to 32x1.
>-__declspec(naked) void ScaleRowDown2Box_AVX2(const uint8* src_ptr,
>+__declspec(naked) void ScaleRowDown2Box_AVX2(const uint8_t* src_ptr,
>                                              ptrdiff_t src_stride,
>-                                             uint8* dst_ptr,
>+                                             uint8_t* dst_ptr,
>                                              int dst_width) {
>   __asm {
>     push        esi
>@@ -308,9 +309,9 @@ __declspec(naked) void ScaleRowDown2Box_AVX2(const uint8* src_ptr,
> #endif  // HAS_SCALEROWDOWN2_AVX2
> 
> // Point samples 32 pixels to 8 pixels.
>-__declspec(naked) void ScaleRowDown4_SSSE3(const uint8* src_ptr,
>+__declspec(naked) void ScaleRowDown4_SSSE3(const uint8_t* src_ptr,
>                                            ptrdiff_t src_stride,
>-                                           uint8* dst_ptr,
>+                                           uint8_t* dst_ptr,
>                                            int dst_width) {
>   __asm {
>     mov        eax, [esp + 4]  // src_ptr
>@@ -340,9 +341,9 @@ __declspec(naked) void ScaleRowDown4_SSSE3(const uint8* src_ptr,
> }
> 
> // Blends 32x4 rectangle to 8x1.
>-__declspec(naked) void ScaleRowDown4Box_SSSE3(const uint8* src_ptr,
>+__declspec(naked) void ScaleRowDown4Box_SSSE3(const uint8_t* src_ptr,
>                                               ptrdiff_t src_stride,
>-                                              uint8* dst_ptr,
>+                                              uint8_t* dst_ptr,
>                                               int dst_width) {
>   __asm {
>     push       esi
>@@ -399,9 +400,9 @@ __declspec(naked) void ScaleRowDown4Box_SSSE3(const uint8* src_ptr,
> 
> #ifdef HAS_SCALEROWDOWN4_AVX2
> // Point samples 64 pixels to 16 pixels.
>-__declspec(naked) void ScaleRowDown4_AVX2(const uint8* src_ptr,
>+__declspec(naked) void ScaleRowDown4_AVX2(const uint8_t* src_ptr,
>                                           ptrdiff_t src_stride,
>-                                          uint8* dst_ptr,
>+                                          uint8_t* dst_ptr,
>                                           int dst_width) {
>   __asm {
>     mov         eax, [esp + 4]  // src_ptr
>@@ -434,9 +435,9 @@ __declspec(naked) void ScaleRowDown4_AVX2(const uint8* src_ptr,
> }
> 
> // Blends 64x4 rectangle to 16x1.
>-__declspec(naked) void ScaleRowDown4Box_AVX2(const uint8* src_ptr,
>+__declspec(naked) void ScaleRowDown4Box_AVX2(const uint8_t* src_ptr,
>                                              ptrdiff_t src_stride,
>-                                             uint8* dst_ptr,
>+                                             uint8_t* dst_ptr,
>                                              int dst_width) {
>   __asm {
>     push        esi
>@@ -498,9 +499,9 @@ __declspec(naked) void ScaleRowDown4Box_AVX2(const uint8* src_ptr,
> // Produces three 8 byte values. For each 8 bytes, 16 bytes are read.
> // Then shuffled to do the scaling.
> 
>-__declspec(naked) void ScaleRowDown34_SSSE3(const uint8* src_ptr,
>+__declspec(naked) void ScaleRowDown34_SSSE3(const uint8_t* src_ptr,
>                                             ptrdiff_t src_stride,
>-                                            uint8* dst_ptr,
>+                                            uint8_t* dst_ptr,
>                                             int dst_width) {
>   __asm {
>     mov        eax, [esp + 4]   // src_ptr
>@@ -546,9 +547,9 @@ __declspec(naked) void ScaleRowDown34_SSSE3(const uint8* src_ptr,
> // xmm7 kRound34
> 
> // Note that movdqa+palign may be better than movdqu.
>-__declspec(naked) void ScaleRowDown34_1_Box_SSSE3(const uint8* src_ptr,
>+__declspec(naked) void ScaleRowDown34_1_Box_SSSE3(const uint8_t* src_ptr,
>                                                   ptrdiff_t src_stride,
>-                                                  uint8* dst_ptr,
>+                                                  uint8_t* dst_ptr,
>                                                   int dst_width) {
>   __asm {
>     push       esi
>@@ -603,9 +604,9 @@ __declspec(naked) void ScaleRowDown34_1_Box_SSSE3(const uint8* src_ptr,
> }
> 
> // Note that movdqa+palign may be better than movdqu.
>-__declspec(naked) void ScaleRowDown34_0_Box_SSSE3(const uint8* src_ptr,
>+__declspec(naked) void ScaleRowDown34_0_Box_SSSE3(const uint8_t* src_ptr,
>                                                   ptrdiff_t src_stride,
>-                                                  uint8* dst_ptr,
>+                                                  uint8_t* dst_ptr,
>                                                   int dst_width) {
>   __asm {
>     push       esi
>@@ -665,9 +666,9 @@ __declspec(naked) void ScaleRowDown34_0_Box_SSSE3(const uint8* src_ptr,
> // 3/8 point sampler
> 
> // Scale 32 pixels to 12
>-__declspec(naked) void ScaleRowDown38_SSSE3(const uint8* src_ptr,
>+__declspec(naked) void ScaleRowDown38_SSSE3(const uint8_t* src_ptr,
>                                             ptrdiff_t src_stride,
>-                                            uint8* dst_ptr,
>+                                            uint8_t* dst_ptr,
>                                             int dst_width) {
>   __asm {
>     mov        eax, [esp + 4]  // src_ptr
>@@ -697,9 +698,9 @@ __declspec(naked) void ScaleRowDown38_SSSE3(const uint8* src_ptr,
> }
> 
> // Scale 16x3 pixels to 6x1 with interpolation
>-__declspec(naked) void ScaleRowDown38_3_Box_SSSE3(const uint8* src_ptr,
>+__declspec(naked) void ScaleRowDown38_3_Box_SSSE3(const uint8_t* src_ptr,
>                                                   ptrdiff_t src_stride,
>-                                                  uint8* dst_ptr,
>+                                                  uint8_t* dst_ptr,
>                                                   int dst_width) {
>   __asm {
>     push       esi
>@@ -762,9 +763,9 @@ __declspec(naked) void ScaleRowDown38_3_Box_SSSE3(const uint8* src_ptr,
> }
> 
> // Scale 16x2 pixels to 6x1 with interpolation
>-__declspec(naked) void ScaleRowDown38_2_Box_SSSE3(const uint8* src_ptr,
>+__declspec(naked) void ScaleRowDown38_2_Box_SSSE3(const uint8_t* src_ptr,
>                                                   ptrdiff_t src_stride,
>-                                                  uint8* dst_ptr,
>+                                                  uint8_t* dst_ptr,
>                                                   int dst_width) {
>   __asm {
>     push       esi
>@@ -807,8 +808,8 @@ __declspec(naked) void ScaleRowDown38_2_Box_SSSE3(const uint8* src_ptr,
> }
> 
> // Reads 16 bytes and accumulates to 16 shorts at a time.
>-__declspec(naked) void ScaleAddRow_SSE2(const uint8* src_ptr,
>-                                        uint16* dst_ptr,
>+__declspec(naked) void ScaleAddRow_SSE2(const uint8_t* src_ptr,
>+                                        uint16_t* dst_ptr,
>                                         int src_width) {
>   __asm {
>     mov        eax, [esp + 4]  // src_ptr
>@@ -838,8 +839,8 @@ __declspec(naked) void ScaleAddRow_SSE2(const uint8* src_ptr,
> 
> #ifdef HAS_SCALEADDROW_AVX2
> // Reads 32 bytes and accumulates to 32 shorts at a time.
>-__declspec(naked) void ScaleAddRow_AVX2(const uint8* src_ptr,
>-                                        uint16* dst_ptr,
>+__declspec(naked) void ScaleAddRow_AVX2(const uint8_t* src_ptr,
>+                                        uint16_t* dst_ptr,
>                                         int src_width) {
>   __asm {
>     mov         eax, [esp + 4]  // src_ptr
>@@ -870,16 +871,16 @@ __declspec(naked) void ScaleAddRow_AVX2(const uint8* src_ptr,
> 
> // Constant for making pixels signed to avoid pmaddubsw
> // saturation.
>-static uvec8 kFsub80 = {0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80,
>-                        0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80};
>+static const uvec8 kFsub80 = {0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80,
>+                              0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80, 0x80};
> 
> // Constant for making pixels unsigned and adding .5 for rounding.
>-static uvec16 kFadd40 = {0x4040, 0x4040, 0x4040, 0x4040,
>-                         0x4040, 0x4040, 0x4040, 0x4040};
>+static const uvec16 kFadd40 = {0x4040, 0x4040, 0x4040, 0x4040,
>+                               0x4040, 0x4040, 0x4040, 0x4040};
> 
> // Bilinear column filtering. SSSE3 version.
>-__declspec(naked) void ScaleFilterCols_SSSE3(uint8* dst_ptr,
>-                                             const uint8* src_ptr,
>+__declspec(naked) void ScaleFilterCols_SSSE3(uint8_t* dst_ptr,
>+                                             const uint8_t* src_ptr,
>                                              int dst_width,
>                                              int x,
>                                              int dx) {
>@@ -964,8 +965,8 @@ __declspec(naked) void ScaleFilterCols_SSSE3(uint8* dst_ptr,
> }
> 
> // Reads 16 pixels, duplicates them and writes 32 pixels.
>-__declspec(naked) void ScaleColsUp2_SSE2(uint8* dst_ptr,
>-                                         const uint8* src_ptr,
>+__declspec(naked) void ScaleColsUp2_SSE2(uint8_t* dst_ptr,
>+                                         const uint8_t* src_ptr,
>                                          int dst_width,
>                                          int x,
>                                          int dx) {
>@@ -991,9 +992,9 @@ __declspec(naked) void ScaleColsUp2_SSE2(uint8* dst_ptr,
> }
> 
> // Reads 8 pixels, throws half away and writes 4 even pixels (0, 2, 4, 6)
>-__declspec(naked) void ScaleARGBRowDown2_SSE2(const uint8* src_argb,
>+__declspec(naked) void ScaleARGBRowDown2_SSE2(const uint8_t* src_argb,
>                                               ptrdiff_t src_stride,
>-                                              uint8* dst_argb,
>+                                              uint8_t* dst_argb,
>                                               int dst_width) {
>   __asm {
>     mov        eax, [esp + 4]   // src_argb
>@@ -1016,9 +1017,9 @@ __declspec(naked) void ScaleARGBRowDown2_SSE2(const uint8* src_argb,
> }
> 
> // Blends 8x1 rectangle to 4x1.
>-__declspec(naked) void ScaleARGBRowDown2Linear_SSE2(const uint8* src_argb,
>+__declspec(naked) void ScaleARGBRowDown2Linear_SSE2(const uint8_t* src_argb,
>                                                     ptrdiff_t src_stride,
>-                                                    uint8* dst_argb,
>+                                                    uint8_t* dst_argb,
>                                                     int dst_width) {
>   __asm {
>     mov        eax, [esp + 4]  // src_argb
>@@ -1044,9 +1045,9 @@ __declspec(naked) void ScaleARGBRowDown2Linear_SSE2(const uint8* src_argb,
> }
> 
> // Blends 8x2 rectangle to 4x1.
>-__declspec(naked) void ScaleARGBRowDown2Box_SSE2(const uint8* src_argb,
>+__declspec(naked) void ScaleARGBRowDown2Box_SSE2(const uint8_t* src_argb,
>                                                  ptrdiff_t src_stride,
>-                                                 uint8* dst_argb,
>+                                                 uint8_t* dst_argb,
>                                                  int dst_width) {
>   __asm {
>     push       esi
>@@ -1078,10 +1079,10 @@ __declspec(naked) void ScaleARGBRowDown2Box_SSE2(const uint8* src_argb,
> }
> 
> // Reads 4 pixels at a time.
>-__declspec(naked) void ScaleARGBRowDownEven_SSE2(const uint8* src_argb,
>+__declspec(naked) void ScaleARGBRowDownEven_SSE2(const uint8_t* src_argb,
>                                                  ptrdiff_t src_stride,
>                                                  int src_stepx,
>-                                                 uint8* dst_argb,
>+                                                 uint8_t* dst_argb,
>                                                  int dst_width) {
>   __asm {
>     push       ebx
>@@ -1115,10 +1116,10 @@ __declspec(naked) void ScaleARGBRowDownEven_SSE2(const uint8* src_argb,
> }
> 
> // Blends four 2x2 to 4x1.
>-__declspec(naked) void ScaleARGBRowDownEvenBox_SSE2(const uint8* src_argb,
>+__declspec(naked) void ScaleARGBRowDownEvenBox_SSE2(const uint8_t* src_argb,
>                                                     ptrdiff_t src_stride,
>                                                     int src_stepx,
>-                                                    uint8* dst_argb,
>+                                                    uint8_t* dst_argb,
>                                                     int dst_width) {
>   __asm {
>     push       ebx
>@@ -1163,8 +1164,8 @@ __declspec(naked) void ScaleARGBRowDownEvenBox_SSE2(const uint8* src_argb,
> }
> 
> // Column scaling unfiltered. SSE2 version.
>-__declspec(naked) void ScaleARGBCols_SSE2(uint8* dst_argb,
>-                                          const uint8* src_argb,
>+__declspec(naked) void ScaleARGBCols_SSE2(uint8_t* dst_argb,
>+                                          const uint8_t* src_argb,
>                                           int dst_width,
>                                           int x,
>                                           int dx) {
>@@ -1246,18 +1247,18 @@ __declspec(naked) void ScaleARGBCols_SSE2(uint8* dst_argb,
> // TODO(fbarchard): Port to Neon
> 
> // Shuffle table for arranging 2 pixels into pairs for pmaddubsw
>-static uvec8 kShuffleColARGB = {
>+static const uvec8 kShuffleColARGB = {
>     0u, 4u,  1u, 5u,  2u,  6u,  3u,  7u,  // bbggrraa 1st pixel
>     8u, 12u, 9u, 13u, 10u, 14u, 11u, 15u  // bbggrraa 2nd pixel
> };
> 
> // Shuffle table for duplicating 2 fractions into 8 bytes each
>-static uvec8 kShuffleFractions = {
>+static const uvec8 kShuffleFractions = {
>     0u, 0u, 0u, 0u, 0u, 0u, 0u, 0u, 4u, 4u, 4u, 4u, 4u, 4u, 4u, 4u,
> };
> 
>-__declspec(naked) void ScaleARGBFilterCols_SSSE3(uint8* dst_argb,
>-                                                 const uint8* src_argb,
>+__declspec(naked) void ScaleARGBFilterCols_SSSE3(uint8_t* dst_argb,
>+                                                 const uint8_t* src_argb,
>                                                  int dst_width,
>                                                  int x,
>                                                  int dx) {
>@@ -1329,8 +1330,8 @@ __declspec(naked) void ScaleARGBFilterCols_SSSE3(uint8* dst_argb,
> }
> 
> // Reads 4 pixels, duplicates them and writes 8 pixels.
>-__declspec(naked) void ScaleARGBColsUp2_SSE2(uint8* dst_argb,
>-                                             const uint8* src_argb,
>+__declspec(naked) void ScaleARGBColsUp2_SSE2(uint8_t* dst_argb,
>+                                             const uint8_t* src_argb,
>                                              int dst_width,
>                                              int x,
>                                              int dx) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/video_common.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/video_common.cc
>index 3e9c6a29502..92384c050cd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/video_common.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/source/video_common.cc
>@@ -15,14 +15,13 @@ namespace libyuv {
> extern "C" {
> #endif
> 
>-#define ARRAY_SIZE(x) (int)(sizeof(x) / sizeof(x[0]))
>-
> struct FourCCAliasEntry {
>-  uint32 alias;
>-  uint32 canonical;
>+  uint32_t alias;
>+  uint32_t canonical;
> };
> 
>-static const struct FourCCAliasEntry kFourCCAliases[] = {
>+#define NUM_ALIASES 18
>+static const struct FourCCAliasEntry kFourCCAliases[NUM_ALIASES] = {
>     {FOURCC_IYUV, FOURCC_I420},
>     {FOURCC_YU12, FOURCC_I420},
>     {FOURCC_YU16, FOURCC_I422},
>@@ -46,9 +45,9 @@ static const struct FourCCAliasEntry kFourCCAliases[] = {
> //  {FOURCC_BGRA, FOURCC_ARGB},  // kCMPixelFormat_32BGRA
> 
> LIBYUV_API
>-uint32 CanonicalFourCC(uint32 fourcc) {
>+uint32_t CanonicalFourCC(uint32_t fourcc) {
>   int i;
>-  for (i = 0; i < ARRAY_SIZE(kFourCCAliases); ++i) {
>+  for (i = 0; i < NUM_ALIASES; ++i) {
>     if (kFourCCAliases[i].alias == fourcc) {
>       return kFourCCAliases[i].canonical;
>     }
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/tools_libyuv/autoroller/roll_deps.py b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/tools_libyuv/autoroller/roll_deps.py
>index efea81e07b1..09ddc40e730 100755
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/tools_libyuv/autoroller/roll_deps.py
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/tools_libyuv/autoroller/roll_deps.py
>@@ -207,7 +207,13 @@ def BuildDepsentryDict(deps_dict):
>   """Builds a dict of paths to DepsEntry objects from a raw parsed deps dict."""
>   result = {}
>   def AddDepsEntries(deps_subdict):
>-    for path, deps_url in deps_subdict.iteritems():
>+    for path, deps_url_spec in deps_subdict.iteritems():
>+      # The deps url is either an URL and a condition, or just the URL.
>+      if isinstance(deps_url_spec, dict):
>+        deps_url = deps_url_spec['url']
>+      else:
>+        deps_url = deps_url_spec
>+
>       if not result.has_key(path):
>         url, revision = deps_url.split('@') if deps_url else (None, None)
>         result[path] = DepsEntry(path, url, revision)
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/tools_libyuv/get_landmines.py b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/tools_libyuv/get_landmines.py
>index 3dc78bb973f..c554f04a39a 100755
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/tools_libyuv/get_landmines.py
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/tools_libyuv/get_landmines.py
>@@ -12,20 +12,8 @@ This file emits the list of reasons why a particular build needs to be clobbered
> (or a list of 'landmines').
> """
> 
>-import os
> import sys
> 
>-script_dir = os.path.dirname(os.path.realpath(__file__))
>-checkout_root = os.path.abspath(os.path.join(script_dir, os.pardir))
>-sys.path.insert(0, os.path.join(checkout_root, 'build'))
>-import landmine_utils
>-
>-
>-distributor = landmine_utils.distributor
>-gyp_defines = landmine_utils.gyp_defines
>-gyp_msvs_version = landmine_utils.gyp_msvs_version
>-platform = landmine_utils.platform
>-
> 
> def print_landmines():
>   """
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/tools_libyuv/valgrind/chrome_tests.bat b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/tools_libyuv/valgrind/chrome_tests.bat
>old mode 100644
>new mode 100755
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/tools_libyuv/valgrind/chrome_tests.py b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/tools_libyuv/valgrind/chrome_tests.py
>old mode 100644
>new mode 100755
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/tools_libyuv/valgrind/chrome_tests.sh b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/tools_libyuv/valgrind/chrome_tests.sh
>old mode 100644
>new mode 100755
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/tools_libyuv/valgrind/locate_valgrind.sh b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/tools_libyuv/valgrind/locate_valgrind.sh
>old mode 100644
>new mode 100755
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/tools_libyuv/valgrind/memcheck_analyze.py b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/tools_libyuv/valgrind/memcheck_analyze.py
>old mode 100644
>new mode 100755
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/tools_libyuv/valgrind/valgrind.sh b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/tools_libyuv/valgrind/valgrind.sh
>old mode 100644
>new mode 100755
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/tools_libyuv/valgrind/valgrind_test.py b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/tools_libyuv/valgrind/valgrind_test.py
>old mode 100644
>new mode 100755
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/unit_test/basictypes_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/unit_test/basictypes_test.cc
>index 89f7644d58e..9aaa2dcd989 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/unit_test/basictypes_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/unit_test/basictypes_test.cc
>@@ -13,25 +13,15 @@
> 
> namespace libyuv {
> 
>-TEST_F(LibYUVBaseTest, Endian) {
>-  uint16 v16 = 0x1234u;
>-  uint8 first_byte = *reinterpret_cast<uint8*>(&v16);
>-#if defined(LIBYUV_LITTLE_ENDIAN)
>-  EXPECT_EQ(0x34u, first_byte);
>-#else
>-  EXPECT_EQ(0x12u, first_byte);
>-#endif
>-}
>-
> TEST_F(LibYUVBaseTest, SizeOfTypes) {
>-  int8 i8 = -1;
>-  uint8 u8 = 1u;
>-  int16 i16 = -1;
>-  uint16 u16 = 1u;
>-  int32 i32 = -1;
>-  uint32 u32 = 1u;
>-  int64 i64 = -1;
>-  uint64 u64 = 1u;
>+  int8_t i8 = -1;
>+  uint8_t u8 = 1u;
>+  int16_t i16 = -1;
>+  uint16_t u16 = 1u;
>+  int32_t i32 = -1;
>+  uint32_t u32 = 1u;
>+  int64_t i64 = -1;
>+  uint64_t u64 = 1u;
>   EXPECT_EQ(1u, sizeof(i8));
>   EXPECT_EQ(1u, sizeof(u8));
>   EXPECT_EQ(2u, sizeof(i16));
>@@ -50,11 +40,4 @@ TEST_F(LibYUVBaseTest, SizeOfTypes) {
>   EXPECT_LT(0u, u64);
> }
> 
>-TEST_F(LibYUVBaseTest, SizeOfConstants) {
>-  EXPECT_EQ(8u, sizeof(INT64_C(0)));
>-  EXPECT_EQ(8u, sizeof(UINT64_C(0)));
>-  EXPECT_EQ(8u, sizeof(INT64_C(0x1234567887654321)));
>-  EXPECT_EQ(8u, sizeof(UINT64_C(0x8765432112345678)));
>-}
>-
> }  // namespace libyuv
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/unit_test/color_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/unit_test/color_test.cc
>index 30b6411283f..4bb448d56fe 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/unit_test/color_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/unit_test/color_test.cc
>@@ -63,10 +63,10 @@ namespace libyuv {
>                                                                                \
>     /* The test is overall for color conversion matrix being reversible, so */ \
>     /* this initializes the pixel with 2x2 blocks to eliminate subsampling. */ \
>-    uint8* p = orig_y;                                                         \
>+    uint8_t* p = orig_y;                                                       \
>     for (int y = 0; y < benchmark_height_ - HS1; y += HS) {                    \
>       for (int x = 0; x < benchmark_width_ - 1; x += 2) {                      \
>-        uint8 r = static_cast<uint8>(fastrand());                              \
>+        uint8_t r = static_cast<uint8_t>(fastrand());                          \
>         p[0] = r;                                                              \
>         p[1] = r;                                                              \
>         p[HN] = r;                                                             \
>@@ -74,7 +74,7 @@ namespace libyuv {
>         p += 2;                                                                \
>       }                                                                        \
>       if (benchmark_width_ & 1) {                                              \
>-        uint8 r = static_cast<uint8>(fastrand());                              \
>+        uint8_t r = static_cast<uint8_t>(fastrand());                          \
>         p[0] = r;                                                              \
>         p[HN] = r;                                                             \
>         p += 1;                                                                \
>@@ -83,13 +83,13 @@ namespace libyuv {
>     }                                                                          \
>     if ((benchmark_height_ & 1) && HS == 2) {                                  \
>       for (int x = 0; x < benchmark_width_ - 1; x += 2) {                      \
>-        uint8 r = static_cast<uint8>(fastrand());                              \
>+        uint8_t r = static_cast<uint8_t>(fastrand());                          \
>         p[0] = r;                                                              \
>         p[1] = r;                                                              \
>         p += 2;                                                                \
>       }                                                                        \
>       if (benchmark_width_ & 1) {                                              \
>-        uint8 r = static_cast<uint8>(fastrand());                              \
>+        uint8_t r = static_cast<uint8_t>(fastrand());                          \
>         p[0] = r;                                                              \
>         p += 1;                                                                \
>       }                                                                        \
>@@ -147,10 +147,10 @@ static void YUVToRGB(int y, int u, int v, int* r, int* g, int* b) {
>   const int kPixels = kWidth * kHeight;
>   const int kHalfPixels = ((kWidth + 1) / 2) * ((kHeight + 1) / 2);
> 
>-  SIMD_ALIGNED(uint8 orig_y[16]);
>-  SIMD_ALIGNED(uint8 orig_u[8]);
>-  SIMD_ALIGNED(uint8 orig_v[8]);
>-  SIMD_ALIGNED(uint8 orig_pixels[16 * 4]);
>+  SIMD_ALIGNED(uint8_t orig_y[16]);
>+  SIMD_ALIGNED(uint8_t orig_u[8]);
>+  SIMD_ALIGNED(uint8_t orig_v[8]);
>+  SIMD_ALIGNED(uint8_t orig_pixels[16 * 4]);
>   memset(orig_y, y, kPixels);
>   memset(orig_u, u, kHalfPixels);
>   memset(orig_v, v, kHalfPixels);
>@@ -170,10 +170,10 @@ static void YUVJToRGB(int y, int u, int v, int* r, int* g, int* b) {
>   const int kPixels = kWidth * kHeight;
>   const int kHalfPixels = ((kWidth + 1) / 2) * ((kHeight + 1) / 2);
> 
>-  SIMD_ALIGNED(uint8 orig_y[16]);
>-  SIMD_ALIGNED(uint8 orig_u[8]);
>-  SIMD_ALIGNED(uint8 orig_v[8]);
>-  SIMD_ALIGNED(uint8 orig_pixels[16 * 4]);
>+  SIMD_ALIGNED(uint8_t orig_y[16]);
>+  SIMD_ALIGNED(uint8_t orig_u[8]);
>+  SIMD_ALIGNED(uint8_t orig_v[8]);
>+  SIMD_ALIGNED(uint8_t orig_pixels[16 * 4]);
>   memset(orig_y, y, kPixels);
>   memset(orig_u, u, kHalfPixels);
>   memset(orig_v, v, kHalfPixels);
>@@ -192,8 +192,8 @@ static void YToRGB(int y, int* r, int* g, int* b) {
>   const int kHeight = 1;
>   const int kPixels = kWidth * kHeight;
> 
>-  SIMD_ALIGNED(uint8 orig_y[16]);
>-  SIMD_ALIGNED(uint8 orig_pixels[16 * 4]);
>+  SIMD_ALIGNED(uint8_t orig_y[16]);
>+  SIMD_ALIGNED(uint8_t orig_pixels[16 * 4]);
>   memset(orig_y, y, kPixels);
> 
>   /* YUV converted to ARGB. */
>@@ -209,8 +209,8 @@ static void YJToRGB(int y, int* r, int* g, int* b) {
>   const int kHeight = 1;
>   const int kPixels = kWidth * kHeight;
> 
>-  SIMD_ALIGNED(uint8 orig_y[16]);
>-  SIMD_ALIGNED(uint8 orig_pixels[16 * 4]);
>+  SIMD_ALIGNED(uint8_t orig_y[16]);
>+  SIMD_ALIGNED(uint8_t orig_pixels[16 * 4]);
>   memset(orig_y, y, kPixels);
> 
>   /* YUV converted to ARGB. */
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/unit_test/compare_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/unit_test/compare_test.cc
>index ff39b2b0f60..136254e169b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/unit_test/compare_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/unit_test/compare_test.cc
>@@ -22,8 +22,10 @@
> namespace libyuv {
> 
> // hash seed of 5381 recommended.
>-static uint32 ReferenceHashDjb2(const uint8* src, uint64 count, uint32 seed) {
>-  uint32 hash = seed;
>+static uint32_t ReferenceHashDjb2(const uint8_t* src,
>+                                  uint64_t count,
>+                                  uint32_t seed) {
>+  uint32_t hash = seed;
>   if (count > 0) {
>     do {
>       hash = hash * 33 + *src++;
>@@ -41,8 +43,8 @@ TEST_F(LibYUVCompareTest, Djb2_Test) {
>       "The quick brown fox jumps over the lazy dog"
>       " and feels as if he were in the seventh heaven of typography"
>       " together with Hermann Zapf";
>-  uint32 foxhash = HashDjb2(reinterpret_cast<const uint8*>(fox), 131, 5381);
>-  const uint32 kExpectedFoxHash = 2611006483u;
>+  uint32_t foxhash = HashDjb2(reinterpret_cast<const uint8_t*>(fox), 131, 5381);
>+  const uint32_t kExpectedFoxHash = 2611006483u;
>   EXPECT_EQ(kExpectedFoxHash, foxhash);
> 
>   for (int i = 0; i < kMaxTest; ++i) {
>@@ -50,8 +52,8 @@ TEST_F(LibYUVCompareTest, Djb2_Test) {
>     src_b[i] = (fastrand() & 0xff);
>   }
>   // Compare different buffers. Expect hash is different.
>-  uint32 h1 = HashDjb2(src_a, kMaxTest, 5381);
>-  uint32 h2 = HashDjb2(src_b, kMaxTest, 5381);
>+  uint32_t h1 = HashDjb2(src_a, kMaxTest, 5381);
>+  uint32_t h2 = HashDjb2(src_b, kMaxTest, 5381);
>   EXPECT_NE(h1, h2);
> 
>   // Make last half same. Expect hash is different.
>@@ -124,8 +126,8 @@ TEST_F(LibYUVCompareTest, BenchmarkDjb2_Opt) {
>   for (int i = 0; i < kMaxTest; ++i) {
>     src_a[i] = i;
>   }
>-  uint32 h2 = ReferenceHashDjb2(src_a, kMaxTest, 5381);
>-  uint32 h1;
>+  uint32_t h2 = ReferenceHashDjb2(src_a, kMaxTest, 5381);
>+  uint32_t h1;
>   for (int i = 0; i < benchmark_iterations_; ++i) {
>     h1 = HashDjb2(src_a, kMaxTest, 5381);
>   }
>@@ -139,8 +141,8 @@ TEST_F(LibYUVCompareTest, BenchmarkDjb2_Unaligned) {
>   for (int i = 0; i < kMaxTest; ++i) {
>     src_a[i + 1] = i;
>   }
>-  uint32 h2 = ReferenceHashDjb2(src_a + 1, kMaxTest, 5381);
>-  uint32 h1;
>+  uint32_t h2 = ReferenceHashDjb2(src_a + 1, kMaxTest, 5381);
>+  uint32_t h1;
>   for (int i = 0; i < benchmark_iterations_; ++i) {
>     h1 = HashDjb2(src_a + 1, kMaxTest, 5381);
>   }
>@@ -149,7 +151,7 @@ TEST_F(LibYUVCompareTest, BenchmarkDjb2_Unaligned) {
> }
> 
> TEST_F(LibYUVCompareTest, BenchmarkARGBDetect_Opt) {
>-  uint32 fourcc;
>+  uint32_t fourcc;
>   const int kMaxTest = benchmark_width_ * benchmark_height_ * 4;
>   align_buffer_page_end(src_a, kMaxTest);
>   for (int i = 0; i < kMaxTest; ++i) {
>@@ -159,12 +161,12 @@ TEST_F(LibYUVCompareTest, BenchmarkARGBDetect_Opt) {
>   src_a[0] = 0;
>   fourcc = ARGBDetect(src_a, benchmark_width_ * 4, benchmark_width_,
>                       benchmark_height_);
>-  EXPECT_EQ(static_cast<uint32>(libyuv::FOURCC_BGRA), fourcc);
>+  EXPECT_EQ(static_cast<uint32_t>(libyuv::FOURCC_BGRA), fourcc);
>   src_a[0] = 255;
>   src_a[3] = 0;
>   fourcc = ARGBDetect(src_a, benchmark_width_ * 4, benchmark_width_,
>                       benchmark_height_);
>-  EXPECT_EQ(static_cast<uint32>(libyuv::FOURCC_ARGB), fourcc);
>+  EXPECT_EQ(static_cast<uint32_t>(libyuv::FOURCC_ARGB), fourcc);
>   src_a[3] = 255;
> 
>   for (int i = 0; i < benchmark_iterations_; ++i) {
>@@ -177,7 +179,7 @@ TEST_F(LibYUVCompareTest, BenchmarkARGBDetect_Opt) {
> }
> 
> TEST_F(LibYUVCompareTest, BenchmarkARGBDetect_Unaligned) {
>-  uint32 fourcc;
>+  uint32_t fourcc;
>   const int kMaxTest = benchmark_width_ * benchmark_height_ * 4 + 1;
>   align_buffer_page_end(src_a, kMaxTest);
>   for (int i = 1; i < kMaxTest; ++i) {
>@@ -187,12 +189,12 @@ TEST_F(LibYUVCompareTest, BenchmarkARGBDetect_Unaligned) {
>   src_a[0 + 1] = 0;
>   fourcc = ARGBDetect(src_a + 1, benchmark_width_ * 4, benchmark_width_,
>                       benchmark_height_);
>-  EXPECT_EQ(static_cast<uint32>(libyuv::FOURCC_BGRA), fourcc);
>+  EXPECT_EQ(static_cast<uint32_t>(libyuv::FOURCC_BGRA), fourcc);
>   src_a[0 + 1] = 255;
>   src_a[3 + 1] = 0;
>   fourcc = ARGBDetect(src_a + 1, benchmark_width_ * 4, benchmark_width_,
>                       benchmark_height_);
>-  EXPECT_EQ(static_cast<uint32>(libyuv::FOURCC_ARGB), fourcc);
>+  EXPECT_EQ(static_cast<uint32_t>(libyuv::FOURCC_ARGB), fourcc);
>   src_a[3 + 1] = 255;
> 
>   for (int i = 0; i < benchmark_iterations_; ++i) {
>@@ -214,14 +216,14 @@ TEST_F(LibYUVCompareTest, BenchmarkHammingDistance_Opt) {
>   // Test known value
>   memcpy(src_a, "test0123test4567", 16);
>   memcpy(src_b, "tick0123tock4567", 16);
>-  uint32 h1 = HammingDistance_C(src_a, src_b, 16);
>+  uint32_t h1 = HammingDistance_C(src_a, src_b, 16);
>   EXPECT_EQ(16u, h1);
> 
>   // Test C vs OPT on random buffer
>   MemRandomize(src_a, kMaxWidth);
>   MemRandomize(src_b, kMaxWidth);
> 
>-  uint32 h0 = HammingDistance_C(src_a, src_b, kMaxWidth);
>+  uint32_t h0 = HammingDistance_C(src_a, src_b, kMaxWidth);
> 
>   int count =
>       benchmark_iterations_ *
>@@ -273,14 +275,14 @@ TEST_F(LibYUVCompareTest, BenchmarkHammingDistance_C) {
>   // Test known value
>   memcpy(src_a, "test0123test4567", 16);
>   memcpy(src_b, "tick0123tock4567", 16);
>-  uint32 h1 = HammingDistance_C(src_a, src_b, 16);
>+  uint32_t h1 = HammingDistance_C(src_a, src_b, 16);
>   EXPECT_EQ(16u, h1);
> 
>   // Test C vs OPT on random buffer
>   MemRandomize(src_a, kMaxWidth);
>   MemRandomize(src_b, kMaxWidth);
> 
>-  uint32 h0 = HammingDistance_C(src_a, src_b, kMaxWidth);
>+  uint32_t h0 = HammingDistance_C(src_a, src_b, kMaxWidth);
> 
>   int count =
>       benchmark_iterations_ *
>@@ -304,14 +306,14 @@ TEST_F(LibYUVCompareTest, BenchmarkHammingDistance) {
> 
>   memcpy(src_a, "test0123test4567", 16);
>   memcpy(src_b, "tick0123tock4567", 16);
>-  uint64 h1 = ComputeHammingDistance(src_a, src_b, 16);
>+  uint64_t h1 = ComputeHammingDistance(src_a, src_b, 16);
>   EXPECT_EQ(16u, h1);
> 
>   // Test C vs OPT on random buffer
>   MemRandomize(src_a, kMaxWidth);
>   MemRandomize(src_b, kMaxWidth);
> 
>-  uint32 h0 = HammingDistance_C(src_a, src_b, kMaxWidth);
>+  uint32_t h0 = HammingDistance_C(src_a, src_b, kMaxWidth);
> 
>   int count =
>       benchmark_iterations_ *
>@@ -337,14 +339,14 @@ static const int kMaxOptCount = (1 << (32 - 3)) - 64;  // 536870848
> #endif
> 
> TEST_F(LibYUVCompareTest, TestHammingDistance_Opt) {
>-  uint32 h1 = 0;
>-  const int kMaxWidth = benchmark_width_ * benchmark_height_;
>+  uint32_t h1 = 0;
>+  const int kMaxWidth = (benchmark_width_ * benchmark_height_ + 31) & ~31;
>   align_buffer_page_end(src_a, kMaxWidth);
>   align_buffer_page_end(src_b, kMaxWidth);
>   memset(src_a, 255u, kMaxWidth);
>   memset(src_b, 0u, kMaxWidth);
> 
>-  uint64 h0 = ComputeHammingDistance(src_a, src_b, kMaxWidth);
>+  uint64_t h0 = ComputeHammingDistance(src_a, src_b, kMaxWidth);
>   EXPECT_EQ(kMaxWidth * 8ULL, h0);
> 
>   for (int i = 0; i < benchmark_iterations_; ++i) {
>@@ -385,7 +387,7 @@ TEST_F(LibYUVCompareTest, TestHammingDistance_Opt) {
>   if (kMaxWidth <= kMaxOptCount) {
>     EXPECT_EQ(kMaxWidth * 8U, h1);
>   } else {
>-    if (kMaxWidth * 8ULL != static_cast<uint64>(h1)) {
>+    if (kMaxWidth * 8ULL != static_cast<uint64_t>(h1)) {
>       printf(
>           "warning - HammingDistance_Opt %u does not match %llu "
>           "but length of %u is longer than guaranteed.\n",
>@@ -408,7 +410,7 @@ TEST_F(LibYUVCompareTest, TestHammingDistance) {
>   memset(src_a, 255u, benchmark_width_ * benchmark_height_);
>   memset(src_b, 0, benchmark_width_ * benchmark_height_);
> 
>-  uint64 h1 = 0;
>+  uint64_t h1 = 0;
>   for (int i = 0; i < benchmark_iterations_; ++i) {
>     h1 = ComputeHammingDistance(src_a, src_b,
>                                 benchmark_width_ * benchmark_height_);
>@@ -428,7 +430,7 @@ TEST_F(LibYUVCompareTest, BenchmarkSumSquareError_Opt) {
> 
>   memcpy(src_a, "test0123test4567", 16);
>   memcpy(src_b, "tick0123tock4567", 16);
>-  uint64 h1 = ComputeSumSquareError(src_a, src_b, 16);
>+  uint64_t h1 = ComputeSumSquareError(src_a, src_b, 16);
>   EXPECT_EQ(790u, h1);
> 
>   for (int i = 0; i < kMaxWidth; ++i) {
>@@ -458,7 +460,7 @@ TEST_F(LibYUVCompareTest, SumSquareError) {
>   memset(src_a, 0, kMaxWidth);
>   memset(src_b, 0, kMaxWidth);
> 
>-  uint64 err;
>+  uint64_t err;
>   err = ComputeSumSquareError(src_a, src_b, kMaxWidth);
> 
>   EXPECT_EQ(0u, err);
>@@ -480,10 +482,10 @@ TEST_F(LibYUVCompareTest, SumSquareError) {
>   }
> 
>   MaskCpuFlags(disable_cpu_flags_);
>-  uint64 c_err = ComputeSumSquareError(src_a, src_b, kMaxWidth);
>+  uint64_t c_err = ComputeSumSquareError(src_a, src_b, kMaxWidth);
> 
>   MaskCpuFlags(benchmark_cpu_info_);
>-  uint64 opt_err = ComputeSumSquareError(src_a, src_b, kMaxWidth);
>+  uint64_t opt_err = ComputeSumSquareError(src_a, src_b, kMaxWidth);
> 
>   EXPECT_EQ(c_err, opt_err);
> 
>@@ -502,9 +504,10 @@ TEST_F(LibYUVCompareTest, BenchmarkPsnr_Opt) {
>   MaskCpuFlags(benchmark_cpu_info_);
> 
>   double opt_time = get_time();
>-  for (int i = 0; i < benchmark_iterations_; ++i)
>+  for (int i = 0; i < benchmark_iterations_; ++i) {
>     CalcFramePsnr(src_a, benchmark_width_, src_b, benchmark_width_,
>                   benchmark_width_, benchmark_height_);
>+  }
> 
>   opt_time = (get_time() - opt_time) / benchmark_iterations_;
>   printf("BenchmarkPsnr_Opt - %8.2f us opt\n", opt_time * 1e6);
>@@ -526,9 +529,10 @@ TEST_F(LibYUVCompareTest, BenchmarkPsnr_Unaligned) {
>   MaskCpuFlags(benchmark_cpu_info_);
> 
>   double opt_time = get_time();
>-  for (int i = 0; i < benchmark_iterations_; ++i)
>+  for (int i = 0; i < benchmark_iterations_; ++i) {
>     CalcFramePsnr(src_a + 1, benchmark_width_, src_b, benchmark_width_,
>                   benchmark_width_, benchmark_height_);
>+  }
> 
>   opt_time = (get_time() - opt_time) / benchmark_iterations_;
>   printf("BenchmarkPsnr_Opt - %8.2f us opt\n", opt_time * 1e6);
>@@ -627,9 +631,10 @@ TEST_F(LibYUVCompareTest, DISABLED_BenchmarkSsim_Opt) {
>   MaskCpuFlags(benchmark_cpu_info_);
> 
>   double opt_time = get_time();
>-  for (int i = 0; i < benchmark_iterations_; ++i)
>+  for (int i = 0; i < benchmark_iterations_; ++i) {
>     CalcFrameSsim(src_a, benchmark_width_, src_b, benchmark_width_,
>                   benchmark_width_, benchmark_height_);
>+  }
> 
>   opt_time = (get_time() - opt_time) / benchmark_iterations_;
>   printf("BenchmarkSsim_Opt - %8.2f us opt\n", opt_time * 1e6);
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/unit_test/convert_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/unit_test/convert_test.cc
>index 56b6364e5eb..39281ae0807 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/unit_test/convert_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/unit_test/convert_test.cc
>@@ -8,9 +8,12 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>+#include <assert.h>
> #include <stdlib.h>
> #include <time.h>
> 
>+#include "libyuv/row.h" /* For ARGBToAR30Row_AVX2 */
>+
> #include "libyuv/basic_types.h"
> #include "libyuv/compare.h"
> #include "libyuv/convert.h"
>@@ -26,102 +29,91 @@
> #include "libyuv/rotate.h"
> #include "libyuv/video_common.h"
> 
>+#if defined(__arm__) || defined(__aarch64__)
>+// arm version subsamples by summing 4 pixels then multiplying by matrix with
>+// 4x smaller coefficients which are rounded to nearest integer.
>+#define ARM_YUV_ERROR 4
>+#else
>+#define ARM_YUV_ERROR 0
>+#endif
>+
> namespace libyuv {
> 
>+// Alias to copy pixels as is
>+#define AR30ToAR30 ARGBCopy
>+#define ABGRToABGR ARGBCopy
>+
> #define SUBSAMPLE(v, a) ((((v) + (a)-1)) / (a))
> 
>-#define TESTPLANARTOPI(SRC_FMT_PLANAR, SRC_SUBSAMP_X, SRC_SUBSAMP_Y,          \
>-                       FMT_PLANAR, SUBSAMP_X, SUBSAMP_Y, W1280, N, NEG, OFF)  \
>+// Planar test
>+
>+#define TESTPLANARTOPI(SRC_FMT_PLANAR, SRC_T, SRC_BPC, SRC_SUBSAMP_X,         \
>+                       SRC_SUBSAMP_Y, FMT_PLANAR, DST_T, DST_BPC,             \
>+                       DST_SUBSAMP_X, DST_SUBSAMP_Y, W1280, N, NEG, OFF)      \
>   TEST_F(LibYUVConvertTest, SRC_FMT_PLANAR##To##FMT_PLANAR##N) {              \
>+    static_assert(SRC_BPC == 1 || SRC_BPC == 2, "SRC BPC unsupported");       \
>+    static_assert(DST_BPC == 1 || DST_BPC == 2, "DST BPC unsupported");       \
>+    static_assert(SRC_SUBSAMP_X == 1 || SRC_SUBSAMP_X == 2,                   \
>+                  "DST SRC_SUBSAMP_X unsupported");                           \
>+    static_assert(SRC_SUBSAMP_Y == 1 || SRC_SUBSAMP_Y == 2,                   \
>+                  "DST SRC_SUBSAMP_Y unsupported");                           \
>+    static_assert(DST_SUBSAMP_X == 1 || DST_SUBSAMP_X == 2,                   \
>+                  "DST DST_SUBSAMP_X unsupported");                           \
>+    static_assert(DST_SUBSAMP_Y == 1 || DST_SUBSAMP_Y == 2,                   \
>+                  "DST DST_SUBSAMP_Y unsupported");                           \
>     const int kWidth = ((W1280) > 0) ? (W1280) : 1;                           \
>     const int kHeight = benchmark_height_;                                    \
>-    align_buffer_page_end(src_y, kWidth* kHeight + OFF);                      \
>-    align_buffer_page_end(src_u, SUBSAMPLE(kWidth, SRC_SUBSAMP_X) *           \
>-                                         SUBSAMPLE(kHeight, SRC_SUBSAMP_Y) +  \
>-                                     OFF);                                    \
>-    align_buffer_page_end(src_v, SUBSAMPLE(kWidth, SRC_SUBSAMP_X) *           \
>-                                         SUBSAMPLE(kHeight, SRC_SUBSAMP_Y) +  \
>-                                     OFF);                                    \
>-    align_buffer_page_end(dst_y_c, kWidth* kHeight);                          \
>-    align_buffer_page_end(dst_u_c, SUBSAMPLE(kWidth, SUBSAMP_X) *             \
>-                                       SUBSAMPLE(kHeight, SUBSAMP_Y));        \
>-    align_buffer_page_end(dst_v_c, SUBSAMPLE(kWidth, SUBSAMP_X) *             \
>-                                       SUBSAMPLE(kHeight, SUBSAMP_Y));        \
>-    align_buffer_page_end(dst_y_opt, kWidth* kHeight);                        \
>-    align_buffer_page_end(dst_u_opt, SUBSAMPLE(kWidth, SUBSAMP_X) *           \
>-                                         SUBSAMPLE(kHeight, SUBSAMP_Y));      \
>-    align_buffer_page_end(dst_v_opt, SUBSAMPLE(kWidth, SUBSAMP_X) *           \
>-                                         SUBSAMPLE(kHeight, SUBSAMP_Y));      \
>-    for (int i = 0; i < kHeight; ++i)                                         \
>-      for (int j = 0; j < kWidth; ++j)                                        \
>-        src_y[i * kWidth + j + OFF] = (fastrand() & 0xff);                    \
>-    for (int i = 0; i < SUBSAMPLE(kHeight, SRC_SUBSAMP_Y); ++i) {             \
>-      for (int j = 0; j < SUBSAMPLE(kWidth, SRC_SUBSAMP_X); ++j) {            \
>-        src_u[(i * SUBSAMPLE(kWidth, SRC_SUBSAMP_X)) + j + OFF] =             \
>-            (fastrand() & 0xff);                                              \
>-        src_v[(i * SUBSAMPLE(kWidth, SRC_SUBSAMP_X)) + j + OFF] =             \
>-            (fastrand() & 0xff);                                              \
>-      }                                                                       \
>-    }                                                                         \
>-    memset(dst_y_c, 1, kWidth* kHeight);                                      \
>-    memset(dst_u_c, 2,                                                        \
>-           SUBSAMPLE(kWidth, SUBSAMP_X) * SUBSAMPLE(kHeight, SUBSAMP_Y));     \
>-    memset(dst_v_c, 3,                                                        \
>-           SUBSAMPLE(kWidth, SUBSAMP_X) * SUBSAMPLE(kHeight, SUBSAMP_Y));     \
>-    memset(dst_y_opt, 101, kWidth* kHeight);                                  \
>-    memset(dst_u_opt, 102,                                                    \
>-           SUBSAMPLE(kWidth, SUBSAMP_X) * SUBSAMPLE(kHeight, SUBSAMP_Y));     \
>-    memset(dst_v_opt, 103,                                                    \
>-           SUBSAMPLE(kWidth, SUBSAMP_X) * SUBSAMPLE(kHeight, SUBSAMP_Y));     \
>+    const int kSrcHalfWidth = SUBSAMPLE(kWidth, SRC_SUBSAMP_X);               \
>+    const int kSrcHalfHeight = SUBSAMPLE(kHeight, SRC_SUBSAMP_Y);             \
>+    const int kDstHalfWidth = SUBSAMPLE(kWidth, DST_SUBSAMP_X);               \
>+    const int kDstHalfHeight = SUBSAMPLE(kHeight, DST_SUBSAMP_Y);             \
>+    align_buffer_page_end(src_y, kWidth* kHeight* SRC_BPC + OFF);             \
>+    align_buffer_page_end(src_u,                                              \
>+                          kSrcHalfWidth* kSrcHalfHeight* SRC_BPC + OFF);      \
>+    align_buffer_page_end(src_v,                                              \
>+                          kSrcHalfWidth* kSrcHalfHeight* SRC_BPC + OFF);      \
>+    align_buffer_page_end(dst_y_c, kWidth* kHeight* DST_BPC);                 \
>+    align_buffer_page_end(dst_u_c, kDstHalfWidth* kDstHalfHeight* DST_BPC);   \
>+    align_buffer_page_end(dst_v_c, kDstHalfWidth* kDstHalfHeight* DST_BPC);   \
>+    align_buffer_page_end(dst_y_opt, kWidth* kHeight* DST_BPC);               \
>+    align_buffer_page_end(dst_u_opt, kDstHalfWidth* kDstHalfHeight* DST_BPC); \
>+    align_buffer_page_end(dst_v_opt, kDstHalfWidth* kDstHalfHeight* DST_BPC); \
>+    MemRandomize(src_y + OFF, kWidth * kHeight * SRC_BPC);                    \
>+    MemRandomize(src_u + OFF, kSrcHalfWidth * kSrcHalfHeight * SRC_BPC);      \
>+    MemRandomize(src_v + OFF, kSrcHalfWidth * kSrcHalfHeight * SRC_BPC);      \
>+    memset(dst_y_c, 1, kWidth* kHeight* DST_BPC);                             \
>+    memset(dst_u_c, 2, kDstHalfWidth* kDstHalfHeight* DST_BPC);               \
>+    memset(dst_v_c, 3, kDstHalfWidth* kDstHalfHeight* DST_BPC);               \
>+    memset(dst_y_opt, 101, kWidth* kHeight* DST_BPC);                         \
>+    memset(dst_u_opt, 102, kDstHalfWidth* kDstHalfHeight* DST_BPC);           \
>+    memset(dst_v_opt, 103, kDstHalfWidth* kDstHalfHeight* DST_BPC);           \
>     MaskCpuFlags(disable_cpu_flags_);                                         \
>     SRC_FMT_PLANAR##To##FMT_PLANAR(                                           \
>-        src_y + OFF, kWidth, src_u + OFF, SUBSAMPLE(kWidth, SRC_SUBSAMP_X),   \
>-        src_v + OFF, SUBSAMPLE(kWidth, SRC_SUBSAMP_X), dst_y_c, kWidth,       \
>-        dst_u_c, SUBSAMPLE(kWidth, SUBSAMP_X), dst_v_c,                       \
>-        SUBSAMPLE(kWidth, SUBSAMP_X), kWidth, NEG kHeight);                   \
>+        reinterpret_cast<SRC_T*>(src_y + OFF), kWidth,                        \
>+        reinterpret_cast<SRC_T*>(src_u + OFF), kSrcHalfWidth,                 \
>+        reinterpret_cast<SRC_T*>(src_v + OFF), kSrcHalfWidth,                 \
>+        reinterpret_cast<DST_T*>(dst_y_c), kWidth,                            \
>+        reinterpret_cast<DST_T*>(dst_u_c), kDstHalfWidth,                     \
>+        reinterpret_cast<DST_T*>(dst_v_c), kDstHalfWidth, kWidth,             \
>+        NEG kHeight);                                                         \
>     MaskCpuFlags(benchmark_cpu_info_);                                        \
>     for (int i = 0; i < benchmark_iterations_; ++i) {                         \
>       SRC_FMT_PLANAR##To##FMT_PLANAR(                                         \
>-          src_y + OFF, kWidth, src_u + OFF, SUBSAMPLE(kWidth, SRC_SUBSAMP_X), \
>-          src_v + OFF, SUBSAMPLE(kWidth, SRC_SUBSAMP_X), dst_y_opt, kWidth,   \
>-          dst_u_opt, SUBSAMPLE(kWidth, SUBSAMP_X), dst_v_opt,                 \
>-          SUBSAMPLE(kWidth, SUBSAMP_X), kWidth, NEG kHeight);                 \
>+          reinterpret_cast<SRC_T*>(src_y + OFF), kWidth,                      \
>+          reinterpret_cast<SRC_T*>(src_u + OFF), kSrcHalfWidth,               \
>+          reinterpret_cast<SRC_T*>(src_v + OFF), kSrcHalfWidth,               \
>+          reinterpret_cast<DST_T*>(dst_y_opt), kWidth,                        \
>+          reinterpret_cast<DST_T*>(dst_u_opt), kDstHalfWidth,                 \
>+          reinterpret_cast<DST_T*>(dst_v_opt), kDstHalfWidth, kWidth,         \
>+          NEG kHeight);                                                       \
>     }                                                                         \
>-    int max_diff = 0;                                                         \
>-    for (int i = 0; i < kHeight; ++i) {                                       \
>-      for (int j = 0; j < kWidth; ++j) {                                      \
>-        int abs_diff = abs(static_cast<int>(dst_y_c[i * kWidth + j]) -        \
>-                           static_cast<int>(dst_y_opt[i * kWidth + j]));      \
>-        if (abs_diff > max_diff) {                                            \
>-          max_diff = abs_diff;                                                \
>-        }                                                                     \
>-      }                                                                       \
>-    }                                                                         \
>-    EXPECT_EQ(0, max_diff);                                                   \
>-    for (int i = 0; i < SUBSAMPLE(kHeight, SUBSAMP_Y); ++i) {                 \
>-      for (int j = 0; j < SUBSAMPLE(kWidth, SUBSAMP_X); ++j) {                \
>-        int abs_diff = abs(                                                   \
>-            static_cast<int>(dst_u_c[i * SUBSAMPLE(kWidth, SUBSAMP_X) + j]) - \
>-            static_cast<int>(                                                 \
>-                dst_u_opt[i * SUBSAMPLE(kWidth, SUBSAMP_X) + j]));            \
>-        if (abs_diff > max_diff) {                                            \
>-          max_diff = abs_diff;                                                \
>-        }                                                                     \
>-      }                                                                       \
>+    for (int i = 0; i < kHeight * kWidth * DST_BPC; ++i) {                    \
>+      EXPECT_EQ(dst_y_c[i], dst_y_opt[i]);                                    \
>     }                                                                         \
>-    EXPECT_LE(max_diff, 3);                                                   \
>-    for (int i = 0; i < SUBSAMPLE(kHeight, SUBSAMP_Y); ++i) {                 \
>-      for (int j = 0; j < SUBSAMPLE(kWidth, SUBSAMP_X); ++j) {                \
>-        int abs_diff = abs(                                                   \
>-            static_cast<int>(dst_v_c[i * SUBSAMPLE(kWidth, SUBSAMP_X) + j]) - \
>-            static_cast<int>(                                                 \
>-                dst_v_opt[i * SUBSAMPLE(kWidth, SUBSAMP_X) + j]));            \
>-        if (abs_diff > max_diff) {                                            \
>-          max_diff = abs_diff;                                                \
>-        }                                                                     \
>-      }                                                                       \
>+    for (int i = 0; i < kDstHalfWidth * kDstHalfHeight * DST_BPC; ++i) {      \
>+      EXPECT_EQ(dst_u_c[i], dst_u_opt[i]);                                    \
>+      EXPECT_EQ(dst_v_c[i], dst_v_opt[i]);                                    \
>     }                                                                         \
>-    EXPECT_LE(max_diff, 3);                                                   \
>     free_aligned_buffer_page_end(dst_y_c);                                    \
>     free_aligned_buffer_page_end(dst_u_c);                                    \
>     free_aligned_buffer_page_end(dst_v_c);                                    \
>@@ -133,25 +125,36 @@ namespace libyuv {
>     free_aligned_buffer_page_end(src_v);                                      \
>   }
> 
>-#define TESTPLANARTOP(SRC_FMT_PLANAR, SRC_SUBSAMP_X, SRC_SUBSAMP_Y,        \
>-                      FMT_PLANAR, SUBSAMP_X, SUBSAMP_Y)                    \
>-  TESTPLANARTOPI(SRC_FMT_PLANAR, SRC_SUBSAMP_X, SRC_SUBSAMP_Y, FMT_PLANAR, \
>-                 SUBSAMP_X, SUBSAMP_Y, benchmark_width_ - 4, _Any, +, 0)   \
>-  TESTPLANARTOPI(SRC_FMT_PLANAR, SRC_SUBSAMP_X, SRC_SUBSAMP_Y, FMT_PLANAR, \
>-                 SUBSAMP_X, SUBSAMP_Y, benchmark_width_, _Unaligned, +, 1) \
>-  TESTPLANARTOPI(SRC_FMT_PLANAR, SRC_SUBSAMP_X, SRC_SUBSAMP_Y, FMT_PLANAR, \
>-                 SUBSAMP_X, SUBSAMP_Y, benchmark_width_, _Invert, -, 0)    \
>-  TESTPLANARTOPI(SRC_FMT_PLANAR, SRC_SUBSAMP_X, SRC_SUBSAMP_Y, FMT_PLANAR, \
>-                 SUBSAMP_X, SUBSAMP_Y, benchmark_width_, _Opt, +, 0)
>-
>-TESTPLANARTOP(I420, 2, 2, I420, 2, 2)
>-TESTPLANARTOP(I422, 2, 1, I420, 2, 2)
>-TESTPLANARTOP(I444, 1, 1, I420, 2, 2)
>-TESTPLANARTOP(I420, 2, 2, I422, 2, 1)
>-TESTPLANARTOP(I420, 2, 2, I444, 1, 1)
>-TESTPLANARTOP(I420, 2, 2, I420Mirror, 2, 2)
>-TESTPLANARTOP(I422, 2, 1, I422, 2, 1)
>-TESTPLANARTOP(I444, 1, 1, I444, 1, 1)
>+#define TESTPLANARTOP(SRC_FMT_PLANAR, SRC_T, SRC_BPC, SRC_SUBSAMP_X,           \
>+                      SRC_SUBSAMP_Y, FMT_PLANAR, DST_T, DST_BPC,               \
>+                      DST_SUBSAMP_X, DST_SUBSAMP_Y)                            \
>+  TESTPLANARTOPI(SRC_FMT_PLANAR, SRC_T, SRC_BPC, SRC_SUBSAMP_X, SRC_SUBSAMP_Y, \
>+                 FMT_PLANAR, DST_T, DST_BPC, DST_SUBSAMP_X, DST_SUBSAMP_Y,     \
>+                 benchmark_width_ - 4, _Any, +, 0)                             \
>+  TESTPLANARTOPI(SRC_FMT_PLANAR, SRC_T, SRC_BPC, SRC_SUBSAMP_X, SRC_SUBSAMP_Y, \
>+                 FMT_PLANAR, DST_T, DST_BPC, DST_SUBSAMP_X, DST_SUBSAMP_Y,     \
>+                 benchmark_width_, _Unaligned, +, 1)                           \
>+  TESTPLANARTOPI(SRC_FMT_PLANAR, SRC_T, SRC_BPC, SRC_SUBSAMP_X, SRC_SUBSAMP_Y, \
>+                 FMT_PLANAR, DST_T, DST_BPC, DST_SUBSAMP_X, DST_SUBSAMP_Y,     \
>+                 benchmark_width_, _Invert, -, 0)                              \
>+  TESTPLANARTOPI(SRC_FMT_PLANAR, SRC_T, SRC_BPC, SRC_SUBSAMP_X, SRC_SUBSAMP_Y, \
>+                 FMT_PLANAR, DST_T, DST_BPC, DST_SUBSAMP_X, DST_SUBSAMP_Y,     \
>+                 benchmark_width_, _Opt, +, 0)
>+
>+TESTPLANARTOP(I420, uint8_t, 1, 2, 2, I420, uint8_t, 1, 2, 2)
>+TESTPLANARTOP(I422, uint8_t, 1, 2, 1, I420, uint8_t, 1, 2, 2)
>+TESTPLANARTOP(I444, uint8_t, 1, 1, 1, I420, uint8_t, 1, 2, 2)
>+TESTPLANARTOP(I420, uint8_t, 1, 2, 2, I422, uint8_t, 1, 2, 1)
>+TESTPLANARTOP(I420, uint8_t, 1, 2, 2, I444, uint8_t, 1, 1, 1)
>+TESTPLANARTOP(I420, uint8_t, 1, 2, 2, I420Mirror, uint8_t, 1, 2, 2)
>+TESTPLANARTOP(I422, uint8_t, 1, 2, 1, I422, uint8_t, 1, 2, 1)
>+TESTPLANARTOP(I444, uint8_t, 1, 1, 1, I444, uint8_t, 1, 1, 1)
>+TESTPLANARTOP(I010, uint16_t, 2, 2, 2, I010, uint16_t, 2, 2, 2)
>+TESTPLANARTOP(I010, uint16_t, 2, 2, 2, I420, uint8_t, 1, 2, 2)
>+TESTPLANARTOP(I420, uint8_t, 1, 2, 2, I010, uint16_t, 2, 2, 2)
>+TESTPLANARTOP(H010, uint16_t, 2, 2, 2, H010, uint16_t, 2, 2, 2)
>+TESTPLANARTOP(H010, uint16_t, 2, 2, 2, H420, uint8_t, 1, 2, 2)
>+TESTPLANARTOP(H420, uint8_t, 1, 2, 2, H010, uint16_t, 2, 2, 2)
> 
> // Test Android 420 to I420
> #define TESTAPLANARTOPI(SRC_FMT_PLANAR, PIXEL_STRIDE, SRC_SUBSAMP_X,          \
>@@ -175,8 +178,8 @@ TESTPLANARTOP(I444, 1, 1, I444, 1, 1)
>                                          SUBSAMPLE(kHeight, SUBSAMP_Y));      \
>     align_buffer_page_end(dst_v_opt, SUBSAMPLE(kWidth, SUBSAMP_X) *           \
>                                          SUBSAMPLE(kHeight, SUBSAMP_Y));      \
>-    uint8* src_u = src_uv + OFF_U;                                            \
>-    uint8* src_v = src_uv + (PIXEL_STRIDE == 1 ? kSizeUV : OFF_V);            \
>+    uint8_t* src_u = src_uv + OFF_U;                                          \
>+    uint8_t* src_v = src_uv + (PIXEL_STRIDE == 1 ? kSizeUV : OFF_V);          \
>     int src_stride_uv = SUBSAMPLE(kWidth, SUBSAMP_X) * PIXEL_STRIDE;          \
>     for (int i = 0; i < kHeight; ++i)                                         \
>       for (int j = 0; j < kWidth; ++j)                                        \
>@@ -491,110 +494,100 @@ TESTBIPLANARTOP(NV21, 2, 2, I420, 2, 2)
> 
> #define ALIGNINT(V, ALIGN) (((V) + (ALIGN)-1) / (ALIGN) * (ALIGN))
> 
>-#define TESTPLANARTOBI(FMT_PLANAR, SUBSAMP_X, SUBSAMP_Y, FMT_B, BPP_B, ALIGN,  \
>-                       YALIGN, W1280, DIFF, N, NEG, OFF, FMT_C, BPP_C)         \
>-  TEST_F(LibYUVConvertTest, FMT_PLANAR##To##FMT_B##N) {                        \
>-    const int kWidth = ((W1280) > 0) ? (W1280) : 1;                            \
>-    const int kHeight = ALIGNINT(benchmark_height_, YALIGN);                   \
>-    const int kStrideB = ALIGNINT(kWidth * BPP_B, ALIGN);                      \
>-    const int kStrideUV = SUBSAMPLE(kWidth, SUBSAMP_X);                        \
>-    const int kSizeUV = kStrideUV * SUBSAMPLE(kHeight, SUBSAMP_Y);             \
>-    align_buffer_page_end(src_y, kWidth* kHeight + OFF);                       \
>-    align_buffer_page_end(src_u, kSizeUV + OFF);                               \
>-    align_buffer_page_end(src_v, kSizeUV + OFF);                               \
>-    align_buffer_page_end(dst_argb_c, kStrideB* kHeight + OFF);                \
>-    align_buffer_page_end(dst_argb_opt, kStrideB* kHeight + OFF);              \
>-    for (int i = 0; i < kWidth * kHeight; ++i) {                               \
>-      src_y[i + OFF] = (fastrand() & 0xff);                                    \
>-    }                                                                          \
>-    for (int i = 0; i < kSizeUV; ++i) {                                        \
>-      src_u[i + OFF] = (fastrand() & 0xff);                                    \
>-      src_v[i + OFF] = (fastrand() & 0xff);                                    \
>-    }                                                                          \
>-    memset(dst_argb_c + OFF, 1, kStrideB * kHeight);                           \
>-    memset(dst_argb_opt + OFF, 101, kStrideB * kHeight);                       \
>-    MaskCpuFlags(disable_cpu_flags_);                                          \
>-    FMT_PLANAR##To##FMT_B(src_y + OFF, kWidth, src_u + OFF, kStrideUV,         \
>-                          src_v + OFF, kStrideUV, dst_argb_c + OFF, kStrideB,  \
>-                          kWidth, NEG kHeight);                                \
>-    MaskCpuFlags(benchmark_cpu_info_);                                         \
>-    for (int i = 0; i < benchmark_iterations_; ++i) {                          \
>-      FMT_PLANAR##To##FMT_B(src_y + OFF, kWidth, src_u + OFF, kStrideUV,       \
>-                            src_v + OFF, kStrideUV, dst_argb_opt + OFF,        \
>-                            kStrideB, kWidth, NEG kHeight);                    \
>-    }                                                                          \
>-    int max_diff = 0;                                                          \
>-    /* Convert to ARGB so 565 is expanded to bytes that can be compared. */    \
>-    align_buffer_page_end(dst_argb32_c, kWidth* BPP_C* kHeight);               \
>-    align_buffer_page_end(dst_argb32_opt, kWidth* BPP_C* kHeight);             \
>-    memset(dst_argb32_c, 2, kWidth* BPP_C* kHeight);                           \
>-    memset(dst_argb32_opt, 102, kWidth* BPP_C* kHeight);                       \
>-    FMT_B##To##FMT_C(dst_argb_c + OFF, kStrideB, dst_argb32_c, kWidth * BPP_C, \
>-                     kWidth, kHeight);                                         \
>-    FMT_B##To##FMT_C(dst_argb_opt + OFF, kStrideB, dst_argb32_opt,             \
>-                     kWidth * BPP_C, kWidth, kHeight);                         \
>-    for (int i = 0; i < kWidth * BPP_C * kHeight; ++i) {                       \
>-      int abs_diff = abs(static_cast<int>(dst_argb32_c[i]) -                   \
>-                         static_cast<int>(dst_argb32_opt[i]));                 \
>-      if (abs_diff > max_diff) {                                               \
>-        max_diff = abs_diff;                                                   \
>-      }                                                                        \
>-    }                                                                          \
>-    EXPECT_LE(max_diff, DIFF);                                                 \
>-    free_aligned_buffer_page_end(src_y);                                       \
>-    free_aligned_buffer_page_end(src_u);                                       \
>-    free_aligned_buffer_page_end(src_v);                                       \
>-    free_aligned_buffer_page_end(dst_argb_c);                                  \
>-    free_aligned_buffer_page_end(dst_argb_opt);                                \
>-    free_aligned_buffer_page_end(dst_argb32_c);                                \
>-    free_aligned_buffer_page_end(dst_argb32_opt);                              \
>+#define TESTPLANARTOBI(FMT_PLANAR, SUBSAMP_X, SUBSAMP_Y, FMT_B, BPP_B, ALIGN, \
>+                       YALIGN, W1280, N, NEG, OFF)                            \
>+  TEST_F(LibYUVConvertTest, FMT_PLANAR##To##FMT_B##N) {                       \
>+    const int kWidth = ((W1280) > 0) ? (W1280) : 1;                           \
>+    const int kHeight = ALIGNINT(benchmark_height_, YALIGN);                  \
>+    const int kStrideB = ALIGNINT(kWidth * BPP_B, ALIGN);                     \
>+    const int kStrideUV = SUBSAMPLE(kWidth, SUBSAMP_X);                       \
>+    const int kSizeUV = kStrideUV * SUBSAMPLE(kHeight, SUBSAMP_Y);            \
>+    align_buffer_page_end(src_y, kWidth* kHeight + OFF);                      \
>+    align_buffer_page_end(src_u, kSizeUV + OFF);                              \
>+    align_buffer_page_end(src_v, kSizeUV + OFF);                              \
>+    align_buffer_page_end(dst_argb_c, kStrideB* kHeight + OFF);               \
>+    align_buffer_page_end(dst_argb_opt, kStrideB* kHeight + OFF);             \
>+    for (int i = 0; i < kWidth * kHeight; ++i) {                              \
>+      src_y[i + OFF] = (fastrand() & 0xff);                                   \
>+    }                                                                         \
>+    for (int i = 0; i < kSizeUV; ++i) {                                       \
>+      src_u[i + OFF] = (fastrand() & 0xff);                                   \
>+      src_v[i + OFF] = (fastrand() & 0xff);                                   \
>+    }                                                                         \
>+    memset(dst_argb_c + OFF, 1, kStrideB * kHeight);                          \
>+    memset(dst_argb_opt + OFF, 101, kStrideB * kHeight);                      \
>+    MaskCpuFlags(disable_cpu_flags_);                                         \
>+    double time0 = get_time();                                                \
>+    FMT_PLANAR##To##FMT_B(src_y + OFF, kWidth, src_u + OFF, kStrideUV,        \
>+                          src_v + OFF, kStrideUV, dst_argb_c + OFF, kStrideB, \
>+                          kWidth, NEG kHeight);                               \
>+    double time1 = get_time();                                                \
>+    MaskCpuFlags(benchmark_cpu_info_);                                        \
>+    for (int i = 0; i < benchmark_iterations_; ++i) {                         \
>+      FMT_PLANAR##To##FMT_B(src_y + OFF, kWidth, src_u + OFF, kStrideUV,      \
>+                            src_v + OFF, kStrideUV, dst_argb_opt + OFF,       \
>+                            kStrideB, kWidth, NEG kHeight);                   \
>+    }                                                                         \
>+    double time2 = get_time();                                                \
>+    printf(" %8d us C - %8d us OPT\n",                                        \
>+           static_cast<int>((time1 - time0) * 1e6),                           \
>+           static_cast<int>((time2 - time1) * 1e6 / benchmark_iterations_));  \
>+    for (int i = 0; i < kWidth * BPP_B * kHeight; ++i) {                      \
>+      EXPECT_EQ(dst_argb_c[i + OFF], dst_argb_opt[i + OFF]);                  \
>+    }                                                                         \
>+    free_aligned_buffer_page_end(src_y);                                      \
>+    free_aligned_buffer_page_end(src_u);                                      \
>+    free_aligned_buffer_page_end(src_v);                                      \
>+    free_aligned_buffer_page_end(dst_argb_c);                                 \
>+    free_aligned_buffer_page_end(dst_argb_opt);                               \
>   }
> 
>-#define TESTPLANARTOB(FMT_PLANAR, SUBSAMP_X, SUBSAMP_Y, FMT_B, BPP_B, ALIGN,   \
>-                      YALIGN, DIFF, FMT_C, BPP_C)                              \
>-  TESTPLANARTOBI(FMT_PLANAR, SUBSAMP_X, SUBSAMP_Y, FMT_B, BPP_B, ALIGN,        \
>-                 YALIGN, benchmark_width_ - 4, DIFF, _Any, +, 0, FMT_C, BPP_C) \
>-  TESTPLANARTOBI(FMT_PLANAR, SUBSAMP_X, SUBSAMP_Y, FMT_B, BPP_B, ALIGN,        \
>-                 YALIGN, benchmark_width_, DIFF, _Unaligned, +, 1, FMT_C,      \
>-                 BPP_C)                                                        \
>-  TESTPLANARTOBI(FMT_PLANAR, SUBSAMP_X, SUBSAMP_Y, FMT_B, BPP_B, ALIGN,        \
>-                 YALIGN, benchmark_width_, DIFF, _Invert, -, 0, FMT_C, BPP_C)  \
>-  TESTPLANARTOBI(FMT_PLANAR, SUBSAMP_X, SUBSAMP_Y, FMT_B, BPP_B, ALIGN,        \
>-                 YALIGN, benchmark_width_, DIFF, _Opt, +, 0, FMT_C, BPP_C)
>-
>-TESTPLANARTOB(I420, 2, 2, ARGB, 4, 4, 1, 2, ARGB, 4)
>-TESTPLANARTOB(J420, 2, 2, ARGB, 4, 4, 1, 2, ARGB, 4)
>-TESTPLANARTOB(J420, 2, 2, ABGR, 4, 4, 1, 2, ARGB, 4)
>-TESTPLANARTOB(H420, 2, 2, ARGB, 4, 4, 1, 2, ARGB, 4)
>-TESTPLANARTOB(H420, 2, 2, ABGR, 4, 4, 1, 2, ARGB, 4)
>-TESTPLANARTOB(I420, 2, 2, BGRA, 4, 4, 1, 2, ARGB, 4)
>-TESTPLANARTOB(I420, 2, 2, ABGR, 4, 4, 1, 2, ARGB, 4)
>-TESTPLANARTOB(I420, 2, 2, RGBA, 4, 4, 1, 2, ARGB, 4)
>-TESTPLANARTOB(I420, 2, 2, RAW, 3, 3, 1, 2, ARGB, 4)
>-TESTPLANARTOB(I420, 2, 2, RGB24, 3, 3, 1, 2, ARGB, 4)
>-TESTPLANARTOB(H420, 2, 2, RAW, 3, 3, 1, 2, ARGB, 4)
>-TESTPLANARTOB(H420, 2, 2, RGB24, 3, 3, 1, 2, ARGB, 4)
>-TESTPLANARTOB(I420, 2, 2, RGB565, 2, 2, 1, 9, ARGB, 4)
>-TESTPLANARTOB(I420, 2, 2, ARGB1555, 2, 2, 1, 9, ARGB, 4)
>-TESTPLANARTOB(I420, 2, 2, ARGB4444, 2, 2, 1, 17, ARGB, 4)
>-TESTPLANARTOB(I422, 2, 1, ARGB, 4, 4, 1, 2, ARGB, 4)
>-TESTPLANARTOB(I422, 2, 1, RGB565, 2, 2, 1, 9, ARGB, 4)
>-TESTPLANARTOB(J422, 2, 1, ARGB, 4, 4, 1, 2, ARGB, 4)
>-TESTPLANARTOB(J422, 2, 1, ABGR, 4, 4, 1, 2, ARGB, 4)
>-TESTPLANARTOB(H422, 2, 1, ARGB, 4, 4, 1, 2, ARGB, 4)
>-TESTPLANARTOB(H422, 2, 1, ABGR, 4, 4, 1, 2, ARGB, 4)
>-TESTPLANARTOB(I422, 2, 1, BGRA, 4, 4, 1, 2, ARGB, 4)
>-TESTPLANARTOB(I422, 2, 1, ABGR, 4, 4, 1, 2, ARGB, 4)
>-TESTPLANARTOB(I422, 2, 1, RGBA, 4, 4, 1, 2, ARGB, 4)
>-TESTPLANARTOB(I444, 1, 1, ARGB, 4, 4, 1, 2, ARGB, 4)
>-TESTPLANARTOB(J444, 1, 1, ARGB, 4, 4, 1, 2, ARGB, 4)
>-TESTPLANARTOB(I444, 1, 1, ABGR, 4, 4, 1, 2, ARGB, 4)
>-TESTPLANARTOB(I420, 2, 2, YUY2, 2, 4, 1, 1, ARGB, 4)
>-TESTPLANARTOB(I420, 2, 2, UYVY, 2, 4, 1, 1, ARGB, 4)
>-TESTPLANARTOB(I422, 2, 1, YUY2, 2, 4, 1, 0, ARGB, 4)
>-TESTPLANARTOB(I422, 2, 1, UYVY, 2, 4, 1, 0, ARGB, 4)
>-TESTPLANARTOB(I420, 2, 2, I400, 1, 1, 1, 0, ARGB, 4)
>-TESTPLANARTOB(J420, 2, 2, J400, 1, 1, 1, 0, ARGB, 4)
>+#define TESTPLANARTOB(FMT_PLANAR, SUBSAMP_X, SUBSAMP_Y, FMT_B, BPP_B, ALIGN, \
>+                      YALIGN)                                                \
>+  TESTPLANARTOBI(FMT_PLANAR, SUBSAMP_X, SUBSAMP_Y, FMT_B, BPP_B, ALIGN,      \
>+                 YALIGN, benchmark_width_ - 4, _Any, +, 0)                   \
>+  TESTPLANARTOBI(FMT_PLANAR, SUBSAMP_X, SUBSAMP_Y, FMT_B, BPP_B, ALIGN,      \
>+                 YALIGN, benchmark_width_, _Unaligned, +, 1)                 \
>+  TESTPLANARTOBI(FMT_PLANAR, SUBSAMP_X, SUBSAMP_Y, FMT_B, BPP_B, ALIGN,      \
>+                 YALIGN, benchmark_width_, _Invert, -, 0)                    \
>+  TESTPLANARTOBI(FMT_PLANAR, SUBSAMP_X, SUBSAMP_Y, FMT_B, BPP_B, ALIGN,      \
>+                 YALIGN, benchmark_width_, _Opt, +, 0)
>+
>+TESTPLANARTOB(I420, 2, 2, ARGB, 4, 4, 1)
>+TESTPLANARTOB(J420, 2, 2, ARGB, 4, 4, 1)
>+TESTPLANARTOB(J420, 2, 2, ABGR, 4, 4, 1)
>+TESTPLANARTOB(H420, 2, 2, ARGB, 4, 4, 1)
>+TESTPLANARTOB(H420, 2, 2, ABGR, 4, 4, 1)
>+TESTPLANARTOB(I420, 2, 2, BGRA, 4, 4, 1)
>+TESTPLANARTOB(I420, 2, 2, ABGR, 4, 4, 1)
>+TESTPLANARTOB(I420, 2, 2, RGBA, 4, 4, 1)
>+TESTPLANARTOB(I420, 2, 2, RAW, 3, 3, 1)
>+TESTPLANARTOB(I420, 2, 2, RGB24, 3, 3, 1)
>+TESTPLANARTOB(H420, 2, 2, RAW, 3, 3, 1)
>+TESTPLANARTOB(H420, 2, 2, RGB24, 3, 3, 1)
>+TESTPLANARTOB(I420, 2, 2, RGB565, 2, 2, 1)
>+TESTPLANARTOB(I420, 2, 2, ARGB1555, 2, 2, 1)
>+TESTPLANARTOB(I420, 2, 2, ARGB4444, 2, 2, 1)
>+TESTPLANARTOB(I422, 2, 1, ARGB, 4, 4, 1)
>+TESTPLANARTOB(I422, 2, 1, RGB565, 2, 2, 1)
>+TESTPLANARTOB(J422, 2, 1, ARGB, 4, 4, 1)
>+TESTPLANARTOB(J422, 2, 1, ABGR, 4, 4, 1)
>+TESTPLANARTOB(H422, 2, 1, ARGB, 4, 4, 1)
>+TESTPLANARTOB(H422, 2, 1, ABGR, 4, 4, 1)
>+TESTPLANARTOB(I422, 2, 1, BGRA, 4, 4, 1)
>+TESTPLANARTOB(I422, 2, 1, ABGR, 4, 4, 1)
>+TESTPLANARTOB(I422, 2, 1, RGBA, 4, 4, 1)
>+TESTPLANARTOB(I444, 1, 1, ARGB, 4, 4, 1)
>+TESTPLANARTOB(J444, 1, 1, ARGB, 4, 4, 1)
>+TESTPLANARTOB(I444, 1, 1, ABGR, 4, 4, 1)
>+TESTPLANARTOB(I420, 2, 2, YUY2, 2, 4, 1)
>+TESTPLANARTOB(I420, 2, 2, UYVY, 2, 4, 1)
>+TESTPLANARTOB(I422, 2, 1, YUY2, 2, 4, 1)
>+TESTPLANARTOB(I422, 2, 1, UYVY, 2, 4, 1)
>+TESTPLANARTOB(I420, 2, 2, I400, 1, 1, 1)
>+TESTPLANARTOB(J420, 2, 2, J400, 1, 1, 1)
>+TESTPLANARTOB(I420, 2, 2, AR30, 4, 4, 1)
>+TESTPLANARTOB(H420, 2, 2, AR30, 4, 4, 1)
> 
> #define TESTQPLANARTOBI(FMT_PLANAR, SUBSAMP_X, SUBSAMP_Y, FMT_B, BPP_B, ALIGN, \
>                         YALIGN, W1280, DIFF, N, NEG, OFF, ATTEN)               \
>@@ -737,6 +730,10 @@ TESTQPLANARTOB(I420Alpha, 2, 2, ABGR, 4, 4, 1, 2)
> 
> TESTBIPLANARTOB(NV12, 2, 2, ARGB, 4, 2)
> TESTBIPLANARTOB(NV21, 2, 2, ARGB, 4, 2)
>+TESTBIPLANARTOB(NV12, 2, 2, ABGR, 4, 2)
>+TESTBIPLANARTOB(NV21, 2, 2, ABGR, 4, 2)
>+TESTBIPLANARTOB(NV12, 2, 2, RGB24, 3, 2)
>+TESTBIPLANARTOB(NV21, 2, 2, RGB24, 3, 2)
> TESTBIPLANARTOB(NV12, 2, 2, RGB565, 2, 9)
> 
> #ifdef DO_THREE_PLANES
>@@ -865,15 +862,8 @@ TESTBIPLANARTOB(NV12, 2, 2, RGB565, 2, 9)
>                  benchmark_width_, DIFF, _Opt, +, 0)
> 
> TESTATOPLANAR(ARGB, 4, 1, I420, 2, 2, 4)
>-#if defined(__arm__) || defined(__aarch64__)
>-// arm version subsamples by summing 4 pixels then multiplying by matrix with
>-// 4x smaller coefficients which are rounded to nearest integer.
>-TESTATOPLANAR(ARGB, 4, 1, J420, 2, 2, 4)
>-TESTATOPLANAR(ARGB, 4, 1, J422, 2, 1, 4)
>-#else
>-TESTATOPLANAR(ARGB, 4, 1, J420, 2, 2, 0)
>-TESTATOPLANAR(ARGB, 4, 1, J422, 2, 1, 0)
>-#endif
>+TESTATOPLANAR(ARGB, 4, 1, J420, 2, 2, ARM_YUV_ERROR)
>+TESTATOPLANAR(ARGB, 4, 1, J422, 2, 1, ARM_YUV_ERROR)
> TESTATOPLANAR(BGRA, 4, 1, I420, 2, 2, 4)
> TESTATOPLANAR(ABGR, 4, 1, I420, 2, 2, 4)
> TESTATOPLANAR(RGBA, 4, 1, I420, 2, 2, 4)
>@@ -1069,6 +1059,8 @@ TESTATOB(ARGB, 4, 4, 1, RGB24, 3, 3, 1, 0)
> TESTATOB(ARGB, 4, 4, 1, RGB565, 2, 2, 1, 0)
> TESTATOB(ARGB, 4, 4, 1, ARGB1555, 2, 2, 1, 0)
> TESTATOB(ARGB, 4, 4, 1, ARGB4444, 2, 2, 1, 0)
>+TESTATOB(ABGR, 4, 4, 1, AR30, 4, 4, 1, 0)
>+TESTATOB(ARGB, 4, 4, 1, AR30, 4, 4, 1, 0)
> TESTATOB(ARGB, 4, 4, 1, YUY2, 2, 4, 1, 4)
> TESTATOB(ARGB, 4, 4, 1, UYVY, 2, 4, 1, 4)
> TESTATOB(ARGB, 4, 4, 1, I400, 1, 1, 1, 2)
>@@ -1076,14 +1068,20 @@ TESTATOB(ARGB, 4, 4, 1, J400, 1, 1, 1, 2)
> TESTATOB(BGRA, 4, 4, 1, ARGB, 4, 4, 1, 0)
> TESTATOB(ABGR, 4, 4, 1, ARGB, 4, 4, 1, 0)
> TESTATOB(RGBA, 4, 4, 1, ARGB, 4, 4, 1, 0)
>+TESTATOB(AR30, 4, 4, 1, AR30, 4, 4, 1, 0)
> TESTATOB(RAW, 3, 3, 1, ARGB, 4, 4, 1, 0)
> TESTATOB(RAW, 3, 3, 1, RGB24, 3, 3, 1, 0)
> TESTATOB(RGB24, 3, 3, 1, ARGB, 4, 4, 1, 0)
> TESTATOB(RGB565, 2, 2, 1, ARGB, 4, 4, 1, 0)
> TESTATOB(ARGB1555, 2, 2, 1, ARGB, 4, 4, 1, 0)
> TESTATOB(ARGB4444, 2, 2, 1, ARGB, 4, 4, 1, 0)
>-TESTATOB(YUY2, 2, 4, 1, ARGB, 4, 4, 1, 4)
>-TESTATOB(UYVY, 2, 4, 1, ARGB, 4, 4, 1, 4)
>+TESTATOB(AR30, 4, 4, 1, ARGB, 4, 4, 1, 0)
>+TESTATOB(AR30, 4, 4, 1, ABGR, 4, 4, 1, 0)
>+TESTATOB(AB30, 4, 4, 1, ARGB, 4, 4, 1, 0)
>+TESTATOB(AB30, 4, 4, 1, ABGR, 4, 4, 1, 0)
>+TESTATOB(AR30, 4, 4, 1, AB30, 4, 4, 1, 0)
>+TESTATOB(YUY2, 2, 4, 1, ARGB, 4, 4, 1, ARM_YUV_ERROR)
>+TESTATOB(UYVY, 2, 4, 1, ARGB, 4, 4, 1, ARM_YUV_ERROR)
> TESTATOB(YUY2, 2, 4, 1, Y, 1, 1, 1, 0)
> TESTATOB(I400, 1, 1, 1, ARGB, 4, 4, 1, 0)
> TESTATOB(J400, 1, 1, 1, ARGB, 4, 4, 1, 0)
>@@ -1240,8 +1238,8 @@ TESTSYM(BGRAToARGB, 4, 4, 1)
> TESTSYM(ABGRToARGB, 4, 4, 1)
> 
> TEST_F(LibYUVConvertTest, Test565) {
>-  SIMD_ALIGNED(uint8 orig_pixels[256][4]);
>-  SIMD_ALIGNED(uint8 pixels565[256][2]);
>+  SIMD_ALIGNED(uint8_t orig_pixels[256][4]);
>+  SIMD_ALIGNED(uint8_t pixels565[256][2]);
> 
>   for (int i = 0; i < 256; ++i) {
>     for (int j = 0; j < 4; ++j) {
>@@ -1249,7 +1247,7 @@ TEST_F(LibYUVConvertTest, Test565) {
>     }
>   }
>   ARGBToRGB565(&orig_pixels[0][0], 0, &pixels565[0][0], 0, 256, 1);
>-  uint32 checksum = HashDjb2(&pixels565[0][0], sizeof(pixels565), 5381);
>+  uint32_t checksum = HashDjb2(&pixels565[0][0], sizeof(pixels565), 5381);
>   EXPECT_EQ(610919429u, checksum);
> }
> 
>@@ -1444,7 +1442,7 @@ TEST_F(LibYUVConvertTest, NV12Crop) {
>   const int sample_size =
>       kWidth * kHeight + kStrideUV * SUBSAMPLE(kHeight, SUBSAMP_Y) * 2;
>   align_buffer_page_end(src_y, sample_size);
>-  uint8* src_uv = src_y + kWidth * kHeight;
>+  uint8_t* src_uv = src_y + kWidth * kHeight;
> 
>   align_buffer_page_end(dst_y, kDestWidth * kDestHeight);
>   align_buffer_page_end(dst_u, SUBSAMPLE(kDestWidth, SUBSAMP_X) *
>@@ -1512,13 +1510,13 @@ TEST_F(LibYUVConvertTest, NV12Crop) {
> }
> 
> TEST_F(LibYUVConvertTest, TestYToARGB) {
>-  uint8 y[32];
>-  uint8 expectedg[32];
>+  uint8_t y[32];
>+  uint8_t expectedg[32];
>   for (int i = 0; i < 32; ++i) {
>     y[i] = i * 5 + 17;
>     expectedg[i] = static_cast<int>((y[i] - 16) * 1.164f + 0.5f);
>   }
>-  uint8 argb[32 * 4];
>+  uint8_t argb[32 * 4];
>   YToARGB(y, 0, argb, 0, 32, 1);
> 
>   for (int i = 0; i < 32; ++i) {
>@@ -1530,7 +1528,7 @@ TEST_F(LibYUVConvertTest, TestYToARGB) {
>   }
> }
> 
>-static const uint8 kNoDither4x4[16] = {
>+static const uint8_t kNoDither4x4[16] = {
>     0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
> };
> 
>@@ -1557,7 +1555,7 @@ TEST_F(LibYUVConvertTest, TestNoDither) {
> }
> 
> // Ordered 4x4 dither for 888 to 565.  Values from 0 to 7.
>-static const uint8 kDither565_4x4[16] = {
>+static const uint8_t kDither565_4x4[16] = {
>     0, 4, 1, 5, 6, 2, 7, 3, 1, 5, 0, 4, 7, 3, 6, 2,
> };
> 
>@@ -1728,6 +1726,8 @@ TESTPLANARTOBD(I420, 2, 2, RGB565, 2, 2, 1, 9, ARGB, 4)
> TESTPTOB(TestYUY2ToNV12, YUY2ToI420, YUY2ToNV12)
> TESTPTOB(TestUYVYToNV12, UYVYToI420, UYVYToNV12)
> 
>+// Transitive tests.  A to B to C is same as A to C.
>+
> #define TESTPLANARTOEI(FMT_PLANAR, SUBSAMP_X, SUBSAMP_Y, FMT_B, SUB_B, BPP_B, \
>                        W1280, N, NEG, OFF, FMT_C, BPP_C)                      \
>   TEST_F(LibYUVConvertTest, FMT_PLANAR##To##FMT_B##_##FMT_C##N) {             \
>@@ -1893,6 +1893,64 @@ TESTPLANARTOE(I422, 2, 1, UYVY, 2, 4, ARGB, 4)
> TESTQPLANARTOE(I420Alpha, 2, 2, ARGB, 1, 4, ABGR, 4)
> TESTQPLANARTOE(I420Alpha, 2, 2, ABGR, 1, 4, ARGB, 4)
> 
>+#define TESTPLANETOEI(FMT_A, SUB_A, BPP_A, FMT_B, SUB_B, BPP_B, W1280, N, NEG, \
>+                      OFF, FMT_C, BPP_C)                                       \
>+  TEST_F(LibYUVConvertTest, FMT_A##To##FMT_B##_##FMT_C##N) {                   \
>+    const int kWidth = ((W1280) > 0) ? (W1280) : 1;                            \
>+    const int kHeight = benchmark_height_;                                     \
>+    const int kStrideA = SUBSAMPLE(kWidth, SUB_A) * BPP_A;                     \
>+    const int kStrideB = SUBSAMPLE(kWidth, SUB_B) * BPP_B;                     \
>+    align_buffer_page_end(src_argb_a, kStrideA* kHeight + OFF);                \
>+    align_buffer_page_end(dst_argb_b, kStrideB* kHeight + OFF);                \
>+    MemRandomize(src_argb_a + OFF, kStrideA * kHeight);                        \
>+    memset(dst_argb_b + OFF, 1, kStrideB * kHeight);                           \
>+    for (int i = 0; i < benchmark_iterations_; ++i) {                          \
>+      FMT_A##To##FMT_B(src_argb_a + OFF, kStrideA, dst_argb_b + OFF, kStrideB, \
>+                       kWidth, NEG kHeight);                                   \
>+    }                                                                          \
>+    /* Convert to a 3rd format in 1 step and 2 steps and compare  */           \
>+    const int kStrideC = kWidth * BPP_C;                                       \
>+    align_buffer_page_end(dst_argb_c, kStrideC* kHeight + OFF);                \
>+    align_buffer_page_end(dst_argb_bc, kStrideC* kHeight + OFF);               \
>+    memset(dst_argb_c + OFF, 2, kStrideC * kHeight);                           \
>+    memset(dst_argb_bc + OFF, 3, kStrideC * kHeight);                          \
>+    FMT_A##To##FMT_C(src_argb_a + OFF, kStrideA, dst_argb_c + OFF, kStrideC,   \
>+                     kWidth, NEG kHeight);                                     \
>+    /* Convert B to C */                                                       \
>+    FMT_B##To##FMT_C(dst_argb_b + OFF, kStrideB, dst_argb_bc + OFF, kStrideC,  \
>+                     kWidth, kHeight);                                         \
>+    for (int i = 0; i < kStrideC * kHeight; i += 4) {                          \
>+      EXPECT_EQ(dst_argb_c[i + OFF + 0], dst_argb_bc[i + OFF + 0]);            \
>+      EXPECT_EQ(dst_argb_c[i + OFF + 1], dst_argb_bc[i + OFF + 1]);            \
>+      EXPECT_EQ(dst_argb_c[i + OFF + 2], dst_argb_bc[i + OFF + 2]);            \
>+      EXPECT_NEAR(dst_argb_c[i + OFF + 3], dst_argb_bc[i + OFF + 3], 64);      \
>+    }                                                                          \
>+    free_aligned_buffer_page_end(src_argb_a);                                  \
>+    free_aligned_buffer_page_end(dst_argb_b);                                  \
>+    free_aligned_buffer_page_end(dst_argb_c);                                  \
>+    free_aligned_buffer_page_end(dst_argb_bc);                                 \
>+  }
>+
>+#define TESTPLANETOE(FMT_A, SUB_A, BPP_A, FMT_B, SUB_B, BPP_B, FMT_C, BPP_C) \
>+  TESTPLANETOEI(FMT_A, SUB_A, BPP_A, FMT_B, SUB_B, BPP_B,                    \
>+                benchmark_width_ - 4, _Any, +, 0, FMT_C, BPP_C)              \
>+  TESTPLANETOEI(FMT_A, SUB_A, BPP_A, FMT_B, SUB_B, BPP_B, benchmark_width_,  \
>+                _Unaligned, +, 1, FMT_C, BPP_C)                              \
>+  TESTPLANETOEI(FMT_A, SUB_A, BPP_A, FMT_B, SUB_B, BPP_B, benchmark_width_,  \
>+                _Invert, -, 0, FMT_C, BPP_C)                                 \
>+  TESTPLANETOEI(FMT_A, SUB_A, BPP_A, FMT_B, SUB_B, BPP_B, benchmark_width_,  \
>+                _Opt, +, 0, FMT_C, BPP_C)
>+
>+// Caveat: Destination needs to be 4 bytes
>+TESTPLANETOE(ARGB, 1, 4, AR30, 1, 4, ARGB, 4)
>+TESTPLANETOE(ABGR, 1, 4, AR30, 1, 4, ABGR, 4)
>+TESTPLANETOE(AR30, 1, 4, ARGB, 1, 4, ABGR, 4)
>+TESTPLANETOE(AR30, 1, 4, ABGR, 1, 4, ARGB, 4)
>+TESTPLANETOE(ARGB, 1, 4, AB30, 1, 4, ARGB, 4)
>+TESTPLANETOE(ABGR, 1, 4, AB30, 1, 4, ABGR, 4)
>+TESTPLANETOE(AB30, 1, 4, ARGB, 1, 4, ABGR, 4)
>+TESTPLANETOE(AB30, 1, 4, ABGR, 1, 4, ARGB, 4)
>+
> TEST_F(LibYUVConvertTest, RotateWithARGBSource) {
>   // 2x2 frames
>   uint32_t src[4];
>@@ -1928,4 +1986,476 @@ TEST_F(LibYUVConvertTest, RotateWithARGBSource) {
>   EXPECT_EQ(dst[3], src[1]);
> }
> 
>+#ifdef HAS_ARGBTOAR30ROW_AVX2
>+TEST_F(LibYUVConvertTest, ARGBToAR30Row_Opt) {
>+  // ARGBToAR30Row_AVX2 expects a multiple of 8 pixels.
>+  const int kPixels = (benchmark_width_ * benchmark_height_ + 7) & ~7;
>+  align_buffer_page_end(src, kPixels * 4);
>+  align_buffer_page_end(dst_opt, kPixels * 4);
>+  align_buffer_page_end(dst_c, kPixels * 4);
>+  MemRandomize(src, kPixels * 4);
>+  memset(dst_opt, 0, kPixels * 4);
>+  memset(dst_c, 1, kPixels * 4);
>+
>+  ARGBToAR30Row_C(src, dst_c, kPixels);
>+
>+  int has_avx2 = TestCpuFlag(kCpuHasAVX2);
>+  int has_ssse3 = TestCpuFlag(kCpuHasSSSE3);
>+  for (int i = 0; i < benchmark_iterations_; ++i) {
>+    if (has_avx2) {
>+      ARGBToAR30Row_AVX2(src, dst_opt, kPixels);
>+    } else if (has_ssse3) {
>+      ARGBToAR30Row_SSSE3(src, dst_opt, kPixels);
>+    } else {
>+      ARGBToAR30Row_C(src, dst_opt, kPixels);
>+    }
>+  }
>+  for (int i = 0; i < kPixels * 4; ++i) {
>+    EXPECT_EQ(dst_opt[i], dst_c[i]);
>+  }
>+
>+  free_aligned_buffer_page_end(src);
>+  free_aligned_buffer_page_end(dst_opt);
>+  free_aligned_buffer_page_end(dst_c);
>+}
>+#endif  // HAS_ARGBTOAR30ROW_AVX2
>+
>+#ifdef HAS_ABGRTOAR30ROW_AVX2
>+TEST_F(LibYUVConvertTest, ABGRToAR30Row_Opt) {
>+  // ABGRToAR30Row_AVX2 expects a multiple of 8 pixels.
>+  const int kPixels = (benchmark_width_ * benchmark_height_ + 7) & ~7;
>+  align_buffer_page_end(src, kPixels * 4);
>+  align_buffer_page_end(dst_opt, kPixels * 4);
>+  align_buffer_page_end(dst_c, kPixels * 4);
>+  MemRandomize(src, kPixels * 4);
>+  memset(dst_opt, 0, kPixels * 4);
>+  memset(dst_c, 1, kPixels * 4);
>+
>+  ABGRToAR30Row_C(src, dst_c, kPixels);
>+
>+  int has_avx2 = TestCpuFlag(kCpuHasAVX2);
>+  int has_ssse3 = TestCpuFlag(kCpuHasSSSE3);
>+  for (int i = 0; i < benchmark_iterations_; ++i) {
>+    if (has_avx2) {
>+      ABGRToAR30Row_AVX2(src, dst_opt, kPixels);
>+    } else if (has_ssse3) {
>+      ABGRToAR30Row_SSSE3(src, dst_opt, kPixels);
>+    } else {
>+      ABGRToAR30Row_C(src, dst_opt, kPixels);
>+    }
>+  }
>+  for (int i = 0; i < kPixels * 4; ++i) {
>+    EXPECT_EQ(dst_opt[i], dst_c[i]);
>+  }
>+
>+  free_aligned_buffer_page_end(src);
>+  free_aligned_buffer_page_end(dst_opt);
>+  free_aligned_buffer_page_end(dst_c);
>+}
>+#endif  // HAS_ABGRTOAR30ROW_AVX2
>+
>+// TODO(fbarchard): Fix clamping issue affected by U channel.
>+#define TESTPLANAR16TOBI(FMT_PLANAR, SUBSAMP_X, SUBSAMP_Y, FMT_B, BPP_B,   \
>+                         ALIGN, YALIGN, W1280, DIFF, N, NEG, SOFF, DOFF)   \
>+  TEST_F(LibYUVConvertTest, FMT_PLANAR##To##FMT_B##N) {                    \
>+    const int kWidth = ((W1280) > 0) ? (W1280) : 1;                        \
>+    const int kHeight = ALIGNINT(benchmark_height_, YALIGN);               \
>+    const int kStrideB = ALIGNINT(kWidth * BPP_B, ALIGN);                  \
>+    const int kStrideUV = SUBSAMPLE(kWidth, SUBSAMP_X);                    \
>+    const int kSizeUV = kStrideUV * SUBSAMPLE(kHeight, SUBSAMP_Y);         \
>+    const int kBpc = 2;                                                    \
>+    align_buffer_page_end(src_y, kWidth* kHeight* kBpc + SOFF);            \
>+    align_buffer_page_end(src_u, kSizeUV* kBpc + SOFF);                    \
>+    align_buffer_page_end(src_v, kSizeUV* kBpc + SOFF);                    \
>+    align_buffer_page_end(dst_argb_c, kStrideB* kHeight + DOFF);           \
>+    align_buffer_page_end(dst_argb_opt, kStrideB* kHeight + DOFF);         \
>+    for (int i = 0; i < kWidth * kHeight; ++i) {                           \
>+      reinterpret_cast<uint16_t*>(src_y + SOFF)[i] = (fastrand() & 0x3ff); \
>+    }                                                                      \
>+    for (int i = 0; i < kSizeUV; ++i) {                                    \
>+      reinterpret_cast<uint16_t*>(src_u + SOFF)[i] = (fastrand() & 0x3ff); \
>+      reinterpret_cast<uint16_t*>(src_v + SOFF)[i] = (fastrand() & 0x3ff); \
>+    }                                                                      \
>+    memset(dst_argb_c + DOFF, 1, kStrideB * kHeight);                      \
>+    memset(dst_argb_opt + DOFF, 101, kStrideB * kHeight);                  \
>+    MaskCpuFlags(disable_cpu_flags_);                                      \
>+    FMT_PLANAR##To##FMT_B(                                                 \
>+        reinterpret_cast<uint16_t*>(src_y + SOFF), kWidth,                 \
>+        reinterpret_cast<uint16_t*>(src_u + SOFF), kStrideUV,              \
>+        reinterpret_cast<uint16_t*>(src_v + SOFF), kStrideUV,              \
>+        dst_argb_c + DOFF, kStrideB, kWidth, NEG kHeight);                 \
>+    MaskCpuFlags(benchmark_cpu_info_);                                     \
>+    for (int i = 0; i < benchmark_iterations_; ++i) {                      \
>+      FMT_PLANAR##To##FMT_B(                                               \
>+          reinterpret_cast<uint16_t*>(src_y + SOFF), kWidth,               \
>+          reinterpret_cast<uint16_t*>(src_u + SOFF), kStrideUV,            \
>+          reinterpret_cast<uint16_t*>(src_v + SOFF), kStrideUV,            \
>+          dst_argb_opt + DOFF, kStrideB, kWidth, NEG kHeight);             \
>+    }                                                                      \
>+    int max_diff = 0;                                                      \
>+    for (int i = 0; i < kWidth * BPP_B * kHeight; ++i) {                   \
>+      int abs_diff = abs(static_cast<int>(dst_argb_c[i + DOFF]) -          \
>+                         static_cast<int>(dst_argb_opt[i + DOFF]));        \
>+      if (abs_diff > max_diff) {                                           \
>+        max_diff = abs_diff;                                               \
>+      }                                                                    \
>+    }                                                                      \
>+    EXPECT_LE(max_diff, DIFF);                                             \
>+    free_aligned_buffer_page_end(src_y);                                   \
>+    free_aligned_buffer_page_end(src_u);                                   \
>+    free_aligned_buffer_page_end(src_v);                                   \
>+    free_aligned_buffer_page_end(dst_argb_c);                              \
>+    free_aligned_buffer_page_end(dst_argb_opt);                            \
>+  }
>+
>+#define TESTPLANAR16TOB(FMT_PLANAR, SUBSAMP_X, SUBSAMP_Y, FMT_B, BPP_B, ALIGN, \
>+                        YALIGN, DIFF)                                          \
>+  TESTPLANAR16TOBI(FMT_PLANAR, SUBSAMP_X, SUBSAMP_Y, FMT_B, BPP_B, ALIGN,      \
>+                   YALIGN, benchmark_width_ - 4, DIFF, _Any, +, 0, 0)          \
>+  TESTPLANAR16TOBI(FMT_PLANAR, SUBSAMP_X, SUBSAMP_Y, FMT_B, BPP_B, ALIGN,      \
>+                   YALIGN, benchmark_width_, DIFF, _Unaligned, +, 1, 1)        \
>+  TESTPLANAR16TOBI(FMT_PLANAR, SUBSAMP_X, SUBSAMP_Y, FMT_B, BPP_B, ALIGN,      \
>+                   YALIGN, benchmark_width_, DIFF, _Invert, -, 0, 0)           \
>+  TESTPLANAR16TOBI(FMT_PLANAR, SUBSAMP_X, SUBSAMP_Y, FMT_B, BPP_B, ALIGN,      \
>+                   YALIGN, benchmark_width_, DIFF, _Opt, +, 0, 0)
>+
>+TESTPLANAR16TOB(I010, 2, 2, ARGB, 4, 4, 1, 2)
>+TESTPLANAR16TOB(I010, 2, 2, ABGR, 4, 4, 1, 2)
>+TESTPLANAR16TOB(I010, 2, 2, AR30, 4, 4, 1, 2)
>+TESTPLANAR16TOB(I010, 2, 2, AB30, 4, 4, 1, 2)
>+TESTPLANAR16TOB(H010, 2, 2, ARGB, 4, 4, 1, 2)
>+TESTPLANAR16TOB(H010, 2, 2, ABGR, 4, 4, 1, 2)
>+TESTPLANAR16TOB(H010, 2, 2, AR30, 4, 4, 1, 2)
>+TESTPLANAR16TOB(H010, 2, 2, AB30, 4, 4, 1, 2)
>+
>+static int Clamp(int y) {
>+  if (y < 0) {
>+    y = 0;
>+  }
>+  if (y > 255) {
>+    y = 255;
>+  }
>+  return y;
>+}
>+
>+static int Clamp10(int y) {
>+  if (y < 0) {
>+    y = 0;
>+  }
>+  if (y > 1023) {
>+    y = 1023;
>+  }
>+  return y;
>+}
>+
>+// Test 8 bit YUV to 8 bit RGB
>+TEST_F(LibYUVConvertTest, TestH420ToARGB) {
>+  const int kSize = 256;
>+  int histogram_b[256];
>+  int histogram_g[256];
>+  int histogram_r[256];
>+  memset(histogram_b, 0, sizeof(histogram_b));
>+  memset(histogram_g, 0, sizeof(histogram_g));
>+  memset(histogram_r, 0, sizeof(histogram_r));
>+  align_buffer_page_end(orig_yuv, kSize + kSize / 2 * 2);
>+  align_buffer_page_end(argb_pixels, kSize * 4);
>+  uint8_t* orig_y = orig_yuv;
>+  uint8_t* orig_u = orig_y + kSize;
>+  uint8_t* orig_v = orig_u + kSize / 2;
>+
>+  // Test grey scale
>+  for (int i = 0; i < kSize; ++i) {
>+    orig_y[i] = i;
>+  }
>+  for (int i = 0; i < kSize / 2; ++i) {
>+    orig_u[i] = 128;  // 128 is 0.
>+    orig_v[i] = 128;
>+  }
>+
>+  H420ToARGB(orig_y, 0, orig_u, 0, orig_v, 0, argb_pixels, 0, kSize, 1);
>+
>+  for (int i = 0; i < kSize; ++i) {
>+    int b = argb_pixels[i * 4 + 0];
>+    int g = argb_pixels[i * 4 + 1];
>+    int r = argb_pixels[i * 4 + 2];
>+    int a = argb_pixels[i * 4 + 3];
>+    ++histogram_b[b];
>+    ++histogram_g[g];
>+    ++histogram_r[r];
>+    int expected_y = Clamp(static_cast<int>((i - 16) * 1.164f));
>+    EXPECT_NEAR(b, expected_y, 1);
>+    EXPECT_NEAR(g, expected_y, 1);
>+    EXPECT_NEAR(r, expected_y, 1);
>+    EXPECT_EQ(a, 255);
>+  }
>+
>+  int count_b = 0;
>+  int count_g = 0;
>+  int count_r = 0;
>+  for (int i = 0; i < kSize; ++i) {
>+    if (histogram_b[i]) {
>+      ++count_b;
>+    }
>+    if (histogram_g[i]) {
>+      ++count_g;
>+    }
>+    if (histogram_r[i]) {
>+      ++count_r;
>+    }
>+  }
>+  printf("uniques: B %d, G, %d, R %d\n", count_b, count_g, count_r);
>+
>+  free_aligned_buffer_page_end(orig_yuv);
>+  free_aligned_buffer_page_end(argb_pixels);
>+}
>+
>+// Test 10 bit YUV to 8 bit RGB
>+TEST_F(LibYUVConvertTest, TestH010ToARGB) {
>+  const int kSize = 1024;
>+  int histogram_b[1024];
>+  int histogram_g[1024];
>+  int histogram_r[1024];
>+  memset(histogram_b, 0, sizeof(histogram_b));
>+  memset(histogram_g, 0, sizeof(histogram_g));
>+  memset(histogram_r, 0, sizeof(histogram_r));
>+  align_buffer_page_end(orig_yuv, kSize * 2 + kSize / 2 * 2 * 2);
>+  align_buffer_page_end(argb_pixels, kSize * 4);
>+  uint16_t* orig_y = reinterpret_cast<uint16_t*>(orig_yuv);
>+  uint16_t* orig_u = orig_y + kSize;
>+  uint16_t* orig_v = orig_u + kSize / 2;
>+
>+  // Test grey scale
>+  for (int i = 0; i < kSize; ++i) {
>+    orig_y[i] = i;
>+  }
>+  for (int i = 0; i < kSize / 2; ++i) {
>+    orig_u[i] = 512;  // 512 is 0.
>+    orig_v[i] = 512;
>+  }
>+
>+  H010ToARGB(orig_y, 0, orig_u, 0, orig_v, 0, argb_pixels, 0, kSize, 1);
>+
>+  for (int i = 0; i < kSize; ++i) {
>+    int b = argb_pixels[i * 4 + 0];
>+    int g = argb_pixels[i * 4 + 1];
>+    int r = argb_pixels[i * 4 + 2];
>+    int a = argb_pixels[i * 4 + 3];
>+    ++histogram_b[b];
>+    ++histogram_g[g];
>+    ++histogram_r[r];
>+    int expected_y = Clamp(static_cast<int>((i - 64) * 1.164f / 4));
>+    EXPECT_NEAR(b, expected_y, 1);
>+    EXPECT_NEAR(g, expected_y, 1);
>+    EXPECT_NEAR(r, expected_y, 1);
>+    EXPECT_EQ(a, 255);
>+  }
>+
>+  int count_b = 0;
>+  int count_g = 0;
>+  int count_r = 0;
>+  for (int i = 0; i < kSize; ++i) {
>+    if (histogram_b[i]) {
>+      ++count_b;
>+    }
>+    if (histogram_g[i]) {
>+      ++count_g;
>+    }
>+    if (histogram_r[i]) {
>+      ++count_r;
>+    }
>+  }
>+  printf("uniques: B %d, G, %d, R %d\n", count_b, count_g, count_r);
>+
>+  free_aligned_buffer_page_end(orig_yuv);
>+  free_aligned_buffer_page_end(argb_pixels);
>+}
>+
>+// Test 10 bit YUV to 10 bit RGB
>+// Caveat: Result is near due to float rounding in expected result.
>+TEST_F(LibYUVConvertTest, TestH010ToAR30) {
>+  const int kSize = 1024;
>+  int histogram_b[1024];
>+  int histogram_g[1024];
>+  int histogram_r[1024];
>+  memset(histogram_b, 0, sizeof(histogram_b));
>+  memset(histogram_g, 0, sizeof(histogram_g));
>+  memset(histogram_r, 0, sizeof(histogram_r));
>+
>+  align_buffer_page_end(orig_yuv, kSize * 2 + kSize / 2 * 2 * 2);
>+  align_buffer_page_end(ar30_pixels, kSize * 4);
>+  uint16_t* orig_y = reinterpret_cast<uint16_t*>(orig_yuv);
>+  uint16_t* orig_u = orig_y + kSize;
>+  uint16_t* orig_v = orig_u + kSize / 2;
>+
>+  // Test grey scale
>+  for (int i = 0; i < kSize; ++i) {
>+    orig_y[i] = i;
>+  }
>+  for (int i = 0; i < kSize / 2; ++i) {
>+    orig_u[i] = 512;  // 512 is 0.
>+    orig_v[i] = 512;
>+  }
>+
>+  H010ToAR30(orig_y, 0, orig_u, 0, orig_v, 0, ar30_pixels, 0, kSize, 1);
>+
>+  for (int i = 0; i < kSize; ++i) {
>+    int b10 = reinterpret_cast<uint32_t*>(ar30_pixels)[i] & 1023;
>+    int g10 = (reinterpret_cast<uint32_t*>(ar30_pixels)[i] >> 10) & 1023;
>+    int r10 = (reinterpret_cast<uint32_t*>(ar30_pixels)[i] >> 20) & 1023;
>+    int a2 = (reinterpret_cast<uint32_t*>(ar30_pixels)[i] >> 30) & 3;
>+    ++histogram_b[b10];
>+    ++histogram_g[g10];
>+    ++histogram_r[r10];
>+    int expected_y = Clamp10(static_cast<int>((i - 64) * 1.164f));
>+    EXPECT_NEAR(b10, expected_y, 4);
>+    EXPECT_NEAR(g10, expected_y, 4);
>+    EXPECT_NEAR(r10, expected_y, 4);
>+    EXPECT_EQ(a2, 3);
>+  }
>+
>+  int count_b = 0;
>+  int count_g = 0;
>+  int count_r = 0;
>+  for (int i = 0; i < kSize; ++i) {
>+    if (histogram_b[i]) {
>+      ++count_b;
>+    }
>+    if (histogram_g[i]) {
>+      ++count_g;
>+    }
>+    if (histogram_r[i]) {
>+      ++count_r;
>+    }
>+  }
>+  printf("uniques: B %d, G, %d, R %d\n", count_b, count_g, count_r);
>+
>+  free_aligned_buffer_page_end(orig_yuv);
>+  free_aligned_buffer_page_end(ar30_pixels);
>+}
>+
>+// Test 10 bit YUV to 10 bit RGB
>+// Caveat: Result is near due to float rounding in expected result.
>+TEST_F(LibYUVConvertTest, TestH010ToAB30) {
>+  const int kSize = 1024;
>+  int histogram_b[1024];
>+  int histogram_g[1024];
>+  int histogram_r[1024];
>+  memset(histogram_b, 0, sizeof(histogram_b));
>+  memset(histogram_g, 0, sizeof(histogram_g));
>+  memset(histogram_r, 0, sizeof(histogram_r));
>+
>+  align_buffer_page_end(orig_yuv, kSize * 2 + kSize / 2 * 2 * 2);
>+  align_buffer_page_end(ab30_pixels, kSize * 4);
>+  uint16_t* orig_y = reinterpret_cast<uint16_t*>(orig_yuv);
>+  uint16_t* orig_u = orig_y + kSize;
>+  uint16_t* orig_v = orig_u + kSize / 2;
>+
>+  // Test grey scale
>+  for (int i = 0; i < kSize; ++i) {
>+    orig_y[i] = i;
>+  }
>+  for (int i = 0; i < kSize / 2; ++i) {
>+    orig_u[i] = 512;  // 512 is 0.
>+    orig_v[i] = 512;
>+  }
>+
>+  H010ToAB30(orig_y, 0, orig_u, 0, orig_v, 0, ab30_pixels, 0, kSize, 1);
>+
>+  for (int i = 0; i < kSize; ++i) {
>+    int r10 = reinterpret_cast<uint32_t*>(ab30_pixels)[i] & 1023;
>+    int g10 = (reinterpret_cast<uint32_t*>(ab30_pixels)[i] >> 10) & 1023;
>+    int b10 = (reinterpret_cast<uint32_t*>(ab30_pixels)[i] >> 20) & 1023;
>+    int a2 = (reinterpret_cast<uint32_t*>(ab30_pixels)[i] >> 30) & 3;
>+    ++histogram_b[b10];
>+    ++histogram_g[g10];
>+    ++histogram_r[r10];
>+    int expected_y = Clamp10(static_cast<int>((i - 64) * 1.164f));
>+    EXPECT_NEAR(b10, expected_y, 4);
>+    EXPECT_NEAR(g10, expected_y, 4);
>+    EXPECT_NEAR(r10, expected_y, 4);
>+    EXPECT_EQ(a2, 3);
>+  }
>+
>+  int count_b = 0;
>+  int count_g = 0;
>+  int count_r = 0;
>+  for (int i = 0; i < kSize; ++i) {
>+    if (histogram_b[i]) {
>+      ++count_b;
>+    }
>+    if (histogram_g[i]) {
>+      ++count_g;
>+    }
>+    if (histogram_r[i]) {
>+      ++count_r;
>+    }
>+  }
>+  printf("uniques: B %d, G, %d, R %d\n", count_b, count_g, count_r);
>+
>+  free_aligned_buffer_page_end(orig_yuv);
>+  free_aligned_buffer_page_end(ab30_pixels);
>+}
>+
>+// Test 8 bit YUV to 10 bit RGB
>+TEST_F(LibYUVConvertTest, TestH420ToAR30) {
>+  const int kSize = 256;
>+  const int kHistSize = 1024;
>+  int histogram_b[kHistSize];
>+  int histogram_g[kHistSize];
>+  int histogram_r[kHistSize];
>+  memset(histogram_b, 0, sizeof(histogram_b));
>+  memset(histogram_g, 0, sizeof(histogram_g));
>+  memset(histogram_r, 0, sizeof(histogram_r));
>+  align_buffer_page_end(orig_yuv, kSize + kSize / 2 * 2);
>+  align_buffer_page_end(ar30_pixels, kSize * 4);
>+  uint8_t* orig_y = orig_yuv;
>+  uint8_t* orig_u = orig_y + kSize;
>+  uint8_t* orig_v = orig_u + kSize / 2;
>+
>+  // Test grey scale
>+  for (int i = 0; i < kSize; ++i) {
>+    orig_y[i] = i;
>+  }
>+  for (int i = 0; i < kSize / 2; ++i) {
>+    orig_u[i] = 128;  // 128 is 0.
>+    orig_v[i] = 128;
>+  }
>+
>+  H420ToAR30(orig_y, 0, orig_u, 0, orig_v, 0, ar30_pixels, 0, kSize, 1);
>+
>+  for (int i = 0; i < kSize; ++i) {
>+    int b10 = reinterpret_cast<uint32_t*>(ar30_pixels)[i] & 1023;
>+    int g10 = (reinterpret_cast<uint32_t*>(ar30_pixels)[i] >> 10) & 1023;
>+    int r10 = (reinterpret_cast<uint32_t*>(ar30_pixels)[i] >> 20) & 1023;
>+    int a2 = (reinterpret_cast<uint32_t*>(ar30_pixels)[i] >> 30) & 3;
>+    ++histogram_b[b10];
>+    ++histogram_g[g10];
>+    ++histogram_r[r10];
>+    int expected_y = Clamp10(static_cast<int>((i - 16) * 1.164f * 4.f));
>+    EXPECT_NEAR(b10, expected_y, 4);
>+    EXPECT_NEAR(g10, expected_y, 4);
>+    EXPECT_NEAR(r10, expected_y, 4);
>+    EXPECT_EQ(a2, 3);
>+  }
>+
>+  int count_b = 0;
>+  int count_g = 0;
>+  int count_r = 0;
>+  for (int i = 0; i < kHistSize; ++i) {
>+    if (histogram_b[i]) {
>+      ++count_b;
>+    }
>+    if (histogram_g[i]) {
>+      ++count_g;
>+    }
>+    if (histogram_r[i]) {
>+      ++count_r;
>+    }
>+  }
>+  printf("uniques: B %d, G, %d, R %d\n", count_b, count_g, count_r);
>+
>+  free_aligned_buffer_page_end(orig_yuv);
>+  free_aligned_buffer_page_end(ar30_pixels);
>+}
>+
> }  // namespace libyuv
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/unit_test/cpu_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/unit_test/cpu_test.cc
>index 4e694f55ce5..a8fb4b4ac01 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/unit_test/cpu_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/unit_test/cpu_test.cc
>@@ -65,8 +65,6 @@ TEST_F(LibYUVBaseTest, TestCpuHas) {
> #if defined(__mips__)
>   int has_mips = TestCpuFlag(kCpuHasMIPS);
>   printf("Has MIPS %x\n", has_mips);
>-  int has_dspr2 = TestCpuFlag(kCpuHasDSPR2);
>-  printf("Has DSPR2 %x\n", has_dspr2);
>   int has_msa = TestCpuFlag(kCpuHasMSA);
>   printf("Has MSA %x\n", has_msa);
> #endif
>@@ -147,6 +145,8 @@ static int FileExists(const char* file_name) {
> 
> TEST_F(LibYUVBaseTest, TestLinuxNeon) {
>   if (FileExists("../../unit_test/testdata/arm_v7.txt")) {
>+    printf("Note: testing to load \"../../unit_test/testdata/arm_v7.txt\"\n");
>+
>     EXPECT_EQ(0, ArmCpuCaps("../../unit_test/testdata/arm_v7.txt"));
>     EXPECT_EQ(kCpuHasNEON, ArmCpuCaps("../../unit_test/testdata/tegra3.txt"));
>     EXPECT_EQ(kCpuHasNEON, ArmCpuCaps("../../unit_test/testdata/juno.txt"));
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/unit_test/math_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/unit_test/math_test.cc
>index 2b4b57b1cea..0abbad51321 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/unit_test/math_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/unit_test/math_test.cc
>@@ -65,8 +65,8 @@ TEST_F(LibYUVBaseTest, TestFixedDiv) {
>   }
>   EXPECT_EQ(123 * 65536, libyuv::FixedDiv(123, 1));
> 
>-  MemRandomize(reinterpret_cast<uint8*>(&num[0]), sizeof(num));
>-  MemRandomize(reinterpret_cast<uint8*>(&div[0]), sizeof(div));
>+  MemRandomize(reinterpret_cast<uint8_t*>(&num[0]), sizeof(num));
>+  MemRandomize(reinterpret_cast<uint8_t*>(&div[0]), sizeof(div));
>   for (int j = 0; j < 1280; ++j) {
>     if (div[j] == 0) {
>       div[j] = 1280;
>@@ -90,8 +90,8 @@ TEST_F(LibYUVBaseTest, TestFixedDiv_Opt) {
>   int result_opt[1280];
>   int result_c[1280];
> 
>-  MemRandomize(reinterpret_cast<uint8*>(&num[0]), sizeof(num));
>-  MemRandomize(reinterpret_cast<uint8*>(&div[0]), sizeof(div));
>+  MemRandomize(reinterpret_cast<uint8_t*>(&num[0]), sizeof(num));
>+  MemRandomize(reinterpret_cast<uint8_t*>(&div[0]), sizeof(div));
>   for (int j = 0; j < 1280; ++j) {
>     num[j] &= 4095;  // Make numerator smaller.
>     div[j] &= 4095;  // Make divisor smaller.
>@@ -124,8 +124,8 @@ TEST_F(LibYUVBaseTest, TestFixedDiv1_Opt) {
>   int result_opt[1280];
>   int result_c[1280];
> 
>-  MemRandomize(reinterpret_cast<uint8*>(&num[0]), sizeof(num));
>-  MemRandomize(reinterpret_cast<uint8*>(&div[0]), sizeof(div));
>+  MemRandomize(reinterpret_cast<uint8_t*>(&num[0]), sizeof(num));
>+  MemRandomize(reinterpret_cast<uint8_t*>(&div[0]), sizeof(div));
>   for (int j = 0; j < 1280; ++j) {
>     num[j] &= 4095;  // Make numerator smaller.
>     div[j] &= 4095;  // Make divisor smaller.
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/unit_test/planar_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/unit_test/planar_test.cc
>index f9e6f8abb2f..756089558f7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/unit_test/planar_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/unit_test/planar_test.cc
>@@ -252,8 +252,8 @@ TEST_F(LibYUVPlanarTest, ARGBUnattenuate_Opt) {
> }
> 
> TEST_F(LibYUVPlanarTest, TestARGBComputeCumulativeSum) {
>-  SIMD_ALIGNED(uint8 orig_pixels[16][16][4]);
>-  SIMD_ALIGNED(int32 added_pixels[16][16][4]);
>+  SIMD_ALIGNED(uint8_t orig_pixels[16][16][4]);
>+  SIMD_ALIGNED(int32_t added_pixels[16][16][4]);
> 
>   for (int y = 0; y < 16; ++y) {
>     for (int x = 0; x < 16; ++x) {
>@@ -278,7 +278,7 @@ TEST_F(LibYUVPlanarTest, TestARGBComputeCumulativeSum) {
> }
> 
> TEST_F(LibYUVPlanarTest, TestARGBGray) {
>-  SIMD_ALIGNED(uint8 orig_pixels[1280][4]);
>+  SIMD_ALIGNED(uint8_t orig_pixels[1280][4]);
>   memset(orig_pixels, 0, sizeof(orig_pixels));
> 
>   // Test blue
>@@ -349,8 +349,8 @@ TEST_F(LibYUVPlanarTest, TestARGBGray) {
> }
> 
> TEST_F(LibYUVPlanarTest, TestARGBGrayTo) {
>-  SIMD_ALIGNED(uint8 orig_pixels[1280][4]);
>-  SIMD_ALIGNED(uint8 gray_pixels[1280][4]);
>+  SIMD_ALIGNED(uint8_t orig_pixels[1280][4]);
>+  SIMD_ALIGNED(uint8_t gray_pixels[1280][4]);
>   memset(orig_pixels, 0, sizeof(orig_pixels));
> 
>   // Test blue
>@@ -421,7 +421,7 @@ TEST_F(LibYUVPlanarTest, TestARGBGrayTo) {
> }
> 
> TEST_F(LibYUVPlanarTest, TestARGBSepia) {
>-  SIMD_ALIGNED(uint8 orig_pixels[1280][4]);
>+  SIMD_ALIGNED(uint8_t orig_pixels[1280][4]);
>   memset(orig_pixels, 0, sizeof(orig_pixels));
> 
>   // Test blue
>@@ -493,12 +493,12 @@ TEST_F(LibYUVPlanarTest, TestARGBSepia) {
> }
> 
> TEST_F(LibYUVPlanarTest, TestARGBColorMatrix) {
>-  SIMD_ALIGNED(uint8 orig_pixels[1280][4]);
>-  SIMD_ALIGNED(uint8 dst_pixels_opt[1280][4]);
>-  SIMD_ALIGNED(uint8 dst_pixels_c[1280][4]);
>+  SIMD_ALIGNED(uint8_t orig_pixels[1280][4]);
>+  SIMD_ALIGNED(uint8_t dst_pixels_opt[1280][4]);
>+  SIMD_ALIGNED(uint8_t dst_pixels_c[1280][4]);
> 
>   // Matrix for Sepia.
>-  SIMD_ALIGNED(static const int8 kRGBToSepia[]) = {
>+  SIMD_ALIGNED(static const int8_t kRGBToSepia[]) = {
>       17 / 2, 68 / 2, 35 / 2, 0, 22 / 2, 88 / 2, 45 / 2, 0,
>       24 / 2, 98 / 2, 50 / 2, 0, 0,      0,      0,      64,  // Copy alpha.
>   };
>@@ -569,10 +569,10 @@ TEST_F(LibYUVPlanarTest, TestARGBColorMatrix) {
> }
> 
> TEST_F(LibYUVPlanarTest, TestRGBColorMatrix) {
>-  SIMD_ALIGNED(uint8 orig_pixels[1280][4]);
>+  SIMD_ALIGNED(uint8_t orig_pixels[1280][4]);
> 
>   // Matrix for Sepia.
>-  SIMD_ALIGNED(static const int8 kRGBToSepia[]) = {
>+  SIMD_ALIGNED(static const int8_t kRGBToSepia[]) = {
>       17, 68, 35, 0, 22, 88, 45, 0,
>       24, 98, 50, 0, 0,  0,  0,  0,  // Unused but makes matrix 16 bytes.
>   };
>@@ -629,11 +629,11 @@ TEST_F(LibYUVPlanarTest, TestRGBColorMatrix) {
> }
> 
> TEST_F(LibYUVPlanarTest, TestARGBColorTable) {
>-  SIMD_ALIGNED(uint8 orig_pixels[1280][4]);
>+  SIMD_ALIGNED(uint8_t orig_pixels[1280][4]);
>   memset(orig_pixels, 0, sizeof(orig_pixels));
> 
>   // Matrix for Sepia.
>-  static const uint8 kARGBTable[256 * 4] = {
>+  static const uint8_t kARGBTable[256 * 4] = {
>       1u, 2u, 3u, 4u, 5u, 6u, 7u, 8u, 9u, 10u, 11u, 12u, 13u, 14u, 15u, 16u,
>   };
> 
>@@ -685,11 +685,11 @@ TEST_F(LibYUVPlanarTest, TestARGBColorTable) {
> 
> // Same as TestARGBColorTable except alpha does not change.
> TEST_F(LibYUVPlanarTest, TestRGBColorTable) {
>-  SIMD_ALIGNED(uint8 orig_pixels[1280][4]);
>+  SIMD_ALIGNED(uint8_t orig_pixels[1280][4]);
>   memset(orig_pixels, 0, sizeof(orig_pixels));
> 
>   // Matrix for Sepia.
>-  static const uint8 kARGBTable[256 * 4] = {
>+  static const uint8_t kARGBTable[256 * 4] = {
>       1u, 2u, 3u, 4u, 5u, 6u, 7u, 8u, 9u, 10u, 11u, 12u, 13u, 14u, 15u, 16u,
>   };
> 
>@@ -740,7 +740,7 @@ TEST_F(LibYUVPlanarTest, TestRGBColorTable) {
> }
> 
> TEST_F(LibYUVPlanarTest, TestARGBQuantize) {
>-  SIMD_ALIGNED(uint8 orig_pixels[1280][4]);
>+  SIMD_ALIGNED(uint8_t orig_pixels[1280][4]);
> 
>   for (int i = 0; i < 1280; ++i) {
>     orig_pixels[i][0] = i;
>@@ -764,8 +764,8 @@ TEST_F(LibYUVPlanarTest, TestARGBQuantize) {
> }
> 
> TEST_F(LibYUVPlanarTest, TestARGBMirror) {
>-  SIMD_ALIGNED(uint8 orig_pixels[1280][4]);
>-  SIMD_ALIGNED(uint8 dst_pixels[1280][4]);
>+  SIMD_ALIGNED(uint8_t orig_pixels[1280][4]);
>+  SIMD_ALIGNED(uint8_t dst_pixels[1280][4]);
> 
>   for (int i = 0; i < 1280; ++i) {
>     orig_pixels[i][0] = i;
>@@ -787,8 +787,8 @@ TEST_F(LibYUVPlanarTest, TestARGBMirror) {
> }
> 
> TEST_F(LibYUVPlanarTest, TestShade) {
>-  SIMD_ALIGNED(uint8 orig_pixels[1280][4]);
>-  SIMD_ALIGNED(uint8 shade_pixels[1280][4]);
>+  SIMD_ALIGNED(uint8_t orig_pixels[1280][4]);
>+  SIMD_ALIGNED(uint8_t shade_pixels[1280][4]);
>   memset(orig_pixels, 0, sizeof(orig_pixels));
> 
>   orig_pixels[0][0] = 10u;
>@@ -845,9 +845,9 @@ TEST_F(LibYUVPlanarTest, TestShade) {
> }
> 
> TEST_F(LibYUVPlanarTest, TestARGBInterpolate) {
>-  SIMD_ALIGNED(uint8 orig_pixels_0[1280][4]);
>-  SIMD_ALIGNED(uint8 orig_pixels_1[1280][4]);
>-  SIMD_ALIGNED(uint8 interpolate_pixels[1280][4]);
>+  SIMD_ALIGNED(uint8_t orig_pixels_0[1280][4]);
>+  SIMD_ALIGNED(uint8_t orig_pixels_1[1280][4]);
>+  SIMD_ALIGNED(uint8_t interpolate_pixels[1280][4]);
>   memset(orig_pixels_0, 0, sizeof(orig_pixels_0));
>   memset(orig_pixels_1, 0, sizeof(orig_pixels_1));
> 
>@@ -926,9 +926,9 @@ TEST_F(LibYUVPlanarTest, TestARGBInterpolate) {
> }
> 
> TEST_F(LibYUVPlanarTest, TestInterpolatePlane) {
>-  SIMD_ALIGNED(uint8 orig_pixels_0[1280]);
>-  SIMD_ALIGNED(uint8 orig_pixels_1[1280]);
>-  SIMD_ALIGNED(uint8 interpolate_pixels[1280]);
>+  SIMD_ALIGNED(uint8_t orig_pixels_0[1280]);
>+  SIMD_ALIGNED(uint8_t orig_pixels_1[1280]);
>+  SIMD_ALIGNED(uint8_t interpolate_pixels[1280]);
>   memset(orig_pixels_0, 0, sizeof(orig_pixels_0));
>   memset(orig_pixels_1, 0, sizeof(orig_pixels_1));
> 
>@@ -1192,7 +1192,6 @@ static void TestBlendPlane(int width,
>   free_aligned_buffer_page_end(src_argb_alpha);
>   free_aligned_buffer_page_end(dst_argb_c);
>   free_aligned_buffer_page_end(dst_argb_opt);
>-  return;
> }
> 
> TEST_F(LibYUVPlanarTest, BlendPlane_Opt) {
>@@ -1286,7 +1285,6 @@ static void TestI420Blend(int width,
>   free_aligned_buffer_page_end(dst_y_opt);
>   free_aligned_buffer_page_end(dst_u_opt);
>   free_aligned_buffer_page_end(dst_v_opt);
>-  return;
> }
> 
> TEST_F(LibYUVPlanarTest, I420Blend_Opt) {
>@@ -1309,8 +1307,8 @@ TEST_F(LibYUVPlanarTest, I420Blend_Invert) {
> }
> 
> TEST_F(LibYUVPlanarTest, TestAffine) {
>-  SIMD_ALIGNED(uint8 orig_pixels_0[1280][4]);
>-  SIMD_ALIGNED(uint8 interpolate_pixels_C[1280][4]);
>+  SIMD_ALIGNED(uint8_t orig_pixels_0[1280][4]);
>+  SIMD_ALIGNED(uint8_t interpolate_pixels_C[1280][4]);
> 
>   for (int i = 0; i < 1280; ++i) {
>     for (int j = 0; j < 4; ++j) {
>@@ -1327,7 +1325,7 @@ TEST_F(LibYUVPlanarTest, TestAffine) {
>   EXPECT_EQ(191u, interpolate_pixels_C[255][3]);
> 
> #if defined(HAS_ARGBAFFINEROW_SSE2)
>-  SIMD_ALIGNED(uint8 interpolate_pixels_Opt[1280][4]);
>+  SIMD_ALIGNED(uint8_t interpolate_pixels_Opt[1280][4]);
>   ARGBAffineRow_SSE2(&orig_pixels_0[0][0], 0, &interpolate_pixels_Opt[0][0],
>                      uv_step, 1280);
>   EXPECT_EQ(0, memcmp(interpolate_pixels_Opt, interpolate_pixels_C, 1280 * 4));
>@@ -1367,7 +1365,7 @@ TEST_F(LibYUVPlanarTest, TestCopyPlane) {
> 
>   // Fill destination buffers with random data.
>   for (i = 0; i < y_plane_size; ++i) {
>-    uint8 random_number = fastrand() & 0x7f;
>+    uint8_t random_number = fastrand() & 0x7f;
>     dst_c[i] = random_number;
>     dst_opt[i] = dst_c[i];
>   }
>@@ -1390,8 +1388,9 @@ TEST_F(LibYUVPlanarTest, TestCopyPlane) {
>   }
> 
>   for (i = 0; i < y_plane_size; ++i) {
>-    if (dst_c[i] != dst_opt[i])
>+    if (dst_c[i] != dst_opt[i]) {
>       ++err;
>+    }
>   }
> 
>   free_aligned_buffer_page_end(orig_y);
>@@ -1867,12 +1866,12 @@ static int TestBlur(int width,
> 
>   MaskCpuFlags(disable_cpu_flags);
>   ARGBBlur(src_argb_a + off, kStride, dst_argb_c, kStride,
>-           reinterpret_cast<int32*>(dst_cumsum), width * 4, width,
>+           reinterpret_cast<int32_t*>(dst_cumsum), width * 4, width,
>            invert * height, radius);
>   MaskCpuFlags(benchmark_cpu_info);
>   for (int i = 0; i < benchmark_iterations; ++i) {
>     ARGBBlur(src_argb_a + off, kStride, dst_argb_opt, kStride,
>-             reinterpret_cast<int32*>(dst_cumsum), width * 4, width,
>+             reinterpret_cast<int32_t*>(dst_cumsum), width * 4, width,
>              invert * height, radius);
>   }
>   int max_diff = 0;
>@@ -1949,9 +1948,9 @@ TEST_F(LibYUVPlanarTest, ARGBBlurSmall_Opt) {
> }
> 
> TEST_F(LibYUVPlanarTest, TestARGBPolynomial) {
>-  SIMD_ALIGNED(uint8 orig_pixels[1280][4]);
>-  SIMD_ALIGNED(uint8 dst_pixels_opt[1280][4]);
>-  SIMD_ALIGNED(uint8 dst_pixels_c[1280][4]);
>+  SIMD_ALIGNED(uint8_t orig_pixels[1280][4]);
>+  SIMD_ALIGNED(uint8_t dst_pixels_opt[1280][4]);
>+  SIMD_ALIGNED(uint8_t dst_pixels_c[1280][4]);
>   memset(orig_pixels, 0, sizeof(orig_pixels));
> 
>   SIMD_ALIGNED(static const float kWarmifyPolynomial[16]) = {
>@@ -2046,37 +2045,38 @@ int TestHalfFloatPlane(int benchmark_width,
>   const int y_plane_size = benchmark_width * benchmark_height * 2;
> 
>   align_buffer_page_end(orig_y, y_plane_size * 3);
>-  uint8* dst_opt = orig_y + y_plane_size;
>-  uint8* dst_c = orig_y + y_plane_size * 2;
>+  uint8_t* dst_opt = orig_y + y_plane_size;
>+  uint8_t* dst_c = orig_y + y_plane_size * 2;
> 
>   MemRandomize(orig_y, y_plane_size);
>   memset(dst_c, 0, y_plane_size);
>   memset(dst_opt, 1, y_plane_size);
> 
>   for (i = 0; i < y_plane_size / 2; ++i) {
>-    reinterpret_cast<uint16*>(orig_y)[i] &= mask;
>+    reinterpret_cast<uint16_t*>(orig_y)[i] &= mask;
>   }
> 
>   // Disable all optimizations.
>   MaskCpuFlags(disable_cpu_flags);
>   for (j = 0; j < benchmark_iterations; j++) {
>-    HalfFloatPlane(reinterpret_cast<uint16*>(orig_y), benchmark_width * 2,
>-                   reinterpret_cast<uint16*>(dst_c), benchmark_width * 2, scale,
>-                   benchmark_width, benchmark_height);
>+    HalfFloatPlane(reinterpret_cast<uint16_t*>(orig_y), benchmark_width * 2,
>+                   reinterpret_cast<uint16_t*>(dst_c), benchmark_width * 2,
>+                   scale, benchmark_width, benchmark_height);
>   }
> 
>   // Enable optimizations.
>   MaskCpuFlags(benchmark_cpu_info);
>   for (j = 0; j < benchmark_iterations; j++) {
>-    HalfFloatPlane(reinterpret_cast<uint16*>(orig_y), benchmark_width * 2,
>-                   reinterpret_cast<uint16*>(dst_opt), benchmark_width * 2,
>+    HalfFloatPlane(reinterpret_cast<uint16_t*>(orig_y), benchmark_width * 2,
>+                   reinterpret_cast<uint16_t*>(dst_opt), benchmark_width * 2,
>                    scale, benchmark_width, benchmark_height);
>   }
> 
>   int max_diff = 0;
>   for (i = 0; i < y_plane_size / 2; ++i) {
>-    int abs_diff = abs(static_cast<int>(reinterpret_cast<uint16*>(dst_c)[i]) -
>-                       static_cast<int>(reinterpret_cast<uint16*>(dst_opt)[i]));
>+    int abs_diff =
>+        abs(static_cast<int>(reinterpret_cast<uint16_t*>(dst_c)[i]) -
>+            static_cast<int>(reinterpret_cast<uint16_t*>(dst_opt)[i]));
>     if (abs_diff > max_diff) {
>       max_diff = abs_diff;
>     }
>@@ -2168,10 +2168,56 @@ TEST_F(LibYUVPlanarTest, TestHalfFloatPlane_12bit_One) {
>   EXPECT_LE(diff, 1);
> }
> 
>+float TestByteToFloat(int benchmark_width,
>+                      int benchmark_height,
>+                      int benchmark_iterations,
>+                      int disable_cpu_flags,
>+                      int benchmark_cpu_info,
>+                      float scale) {
>+  int i, j;
>+  const int y_plane_size = benchmark_width * benchmark_height;
>+
>+  align_buffer_page_end(orig_y, y_plane_size * (1 + 4 + 4));
>+  float* dst_opt = reinterpret_cast<float*>(orig_y + y_plane_size);
>+  float* dst_c = reinterpret_cast<float*>(orig_y + y_plane_size * 5);
>+
>+  MemRandomize(orig_y, y_plane_size);
>+  memset(dst_c, 0, y_plane_size * 4);
>+  memset(dst_opt, 1, y_plane_size * 4);
>+
>+  // Disable all optimizations.
>+  MaskCpuFlags(disable_cpu_flags);
>+  ByteToFloat(orig_y, dst_c, scale, y_plane_size);
>+
>+  // Enable optimizations.
>+  MaskCpuFlags(benchmark_cpu_info);
>+  for (j = 0; j < benchmark_iterations; j++) {
>+    ByteToFloat(orig_y, dst_opt, scale, y_plane_size);
>+  }
>+
>+  float max_diff = 0;
>+  for (i = 0; i < y_plane_size; ++i) {
>+    float abs_diff = fabs(dst_c[i] - dst_opt[i]);
>+    if (abs_diff > max_diff) {
>+      max_diff = abs_diff;
>+    }
>+  }
>+
>+  free_aligned_buffer_page_end(orig_y);
>+  return max_diff;
>+}
>+
>+TEST_F(LibYUVPlanarTest, TestByteToFloat) {
>+  float diff = TestByteToFloat(benchmark_width_, benchmark_height_,
>+                               benchmark_iterations_, disable_cpu_flags_,
>+                               benchmark_cpu_info_, 1.0f);
>+  EXPECT_EQ(0.f, diff);
>+}
>+
> TEST_F(LibYUVPlanarTest, TestARGBLumaColorTable) {
>-  SIMD_ALIGNED(uint8 orig_pixels[1280][4]);
>-  SIMD_ALIGNED(uint8 dst_pixels_opt[1280][4]);
>-  SIMD_ALIGNED(uint8 dst_pixels_c[1280][4]);
>+  SIMD_ALIGNED(uint8_t orig_pixels[1280][4]);
>+  SIMD_ALIGNED(uint8_t dst_pixels_opt[1280][4]);
>+  SIMD_ALIGNED(uint8_t dst_pixels_c[1280][4]);
>   memset(orig_pixels, 0, sizeof(orig_pixels));
> 
>   align_buffer_page_end(lumacolortable, 32768);
>@@ -2343,7 +2389,7 @@ static int TestARGBRect(int width,
>   }
>   const int kStride = width * bpp;
>   const int kSize = kStride * height;
>-  const uint32 v32 = fastrand() & (bpp == 4 ? 0xffffffff : 0xff);
>+  const uint32_t v32 = fastrand() & (bpp == 4 ? 0xffffffff : 0xff);
> 
>   align_buffer_page_end(dst_argb_c, kSize + off);
>   align_buffer_page_end(dst_argb_opt, kSize + off);
>@@ -2631,21 +2677,21 @@ TEST_F(LibYUVPlanarTest, MergeUVRow_16_Opt) {
>   memset(dst_pixels_uv_opt, 0, kPixels * 2 * 2);
>   memset(dst_pixels_uv_c, 1, kPixels * 2 * 2);
> 
>-  MergeUVRow_16_C(reinterpret_cast<const uint16*>(src_pixels_u),
>-                  reinterpret_cast<const uint16*>(src_pixels_v),
>-                  reinterpret_cast<uint16*>(dst_pixels_uv_c), 64, kPixels);
>+  MergeUVRow_16_C(reinterpret_cast<const uint16_t*>(src_pixels_u),
>+                  reinterpret_cast<const uint16_t*>(src_pixels_v),
>+                  reinterpret_cast<uint16_t*>(dst_pixels_uv_c), 64, kPixels);
> 
>   int has_avx2 = TestCpuFlag(kCpuHasAVX2);
>   for (int i = 0; i < benchmark_iterations_; ++i) {
>     if (has_avx2) {
>-      MergeUVRow_16_AVX2(reinterpret_cast<const uint16*>(src_pixels_u),
>-                         reinterpret_cast<const uint16*>(src_pixels_v),
>-                         reinterpret_cast<uint16*>(dst_pixels_uv_opt), 64,
>+      MergeUVRow_16_AVX2(reinterpret_cast<const uint16_t*>(src_pixels_u),
>+                         reinterpret_cast<const uint16_t*>(src_pixels_v),
>+                         reinterpret_cast<uint16_t*>(dst_pixels_uv_opt), 64,
>                          kPixels);
>     } else {
>-      MergeUVRow_16_C(reinterpret_cast<const uint16*>(src_pixels_u),
>-                      reinterpret_cast<const uint16*>(src_pixels_v),
>-                      reinterpret_cast<uint16*>(dst_pixels_uv_opt), 64,
>+      MergeUVRow_16_C(reinterpret_cast<const uint16_t*>(src_pixels_u),
>+                      reinterpret_cast<const uint16_t*>(src_pixels_v),
>+                      reinterpret_cast<uint16_t*>(dst_pixels_uv_opt), 64,
>                       kPixels);
>     }
>   }
>@@ -2661,7 +2707,7 @@ TEST_F(LibYUVPlanarTest, MergeUVRow_16_Opt) {
> }
> #endif
> 
>-// TODO(fbarchard): improve test for platforms and cpu detect
>+// TODO(fbarchard): Improve test for more platforms.
> #ifdef HAS_MULTIPLYROW_16_AVX2
> TEST_F(LibYUVPlanarTest, MultiplyRow_16_Opt) {
>   const int kPixels = benchmark_width_ * benchmark_height_;
>@@ -2673,18 +2719,18 @@ TEST_F(LibYUVPlanarTest, MultiplyRow_16_Opt) {
>   memset(dst_pixels_y_opt, 0, kPixels * 2);
>   memset(dst_pixels_y_c, 1, kPixels * 2);
> 
>-  MultiplyRow_16_C(reinterpret_cast<const uint16*>(src_pixels_y),
>-                   reinterpret_cast<uint16*>(dst_pixels_y_c), 64, kPixels);
>+  MultiplyRow_16_C(reinterpret_cast<const uint16_t*>(src_pixels_y),
>+                   reinterpret_cast<uint16_t*>(dst_pixels_y_c), 64, kPixels);
> 
>   int has_avx2 = TestCpuFlag(kCpuHasAVX2);
>   for (int i = 0; i < benchmark_iterations_; ++i) {
>     if (has_avx2) {
>-      MultiplyRow_16_AVX2(reinterpret_cast<const uint16*>(src_pixels_y),
>-                          reinterpret_cast<uint16*>(dst_pixels_y_opt), 64,
>+      MultiplyRow_16_AVX2(reinterpret_cast<const uint16_t*>(src_pixels_y),
>+                          reinterpret_cast<uint16_t*>(dst_pixels_y_opt), 64,
>                           kPixels);
>     } else {
>-      MultiplyRow_16_C(reinterpret_cast<const uint16*>(src_pixels_y),
>-                       reinterpret_cast<uint16*>(dst_pixels_y_opt), 64,
>+      MultiplyRow_16_C(reinterpret_cast<const uint16_t*>(src_pixels_y),
>+                       reinterpret_cast<uint16_t*>(dst_pixels_y_opt), 64,
>                        kPixels);
>     }
>   }
>@@ -2697,7 +2743,160 @@ TEST_F(LibYUVPlanarTest, MultiplyRow_16_Opt) {
>   free_aligned_buffer_page_end(dst_pixels_y_opt);
>   free_aligned_buffer_page_end(dst_pixels_y_c);
> }
>-#endif
>+#endif  // HAS_MULTIPLYROW_16_AVX2
>+
>+TEST_F(LibYUVPlanarTest, Convert16To8Plane) {
>+  const int kPixels = benchmark_width_ * benchmark_height_;
>+  align_buffer_page_end(src_pixels_y, kPixels * 2);
>+  align_buffer_page_end(dst_pixels_y_opt, kPixels);
>+  align_buffer_page_end(dst_pixels_y_c, kPixels);
>+
>+  MemRandomize(src_pixels_y, kPixels * 2);
>+  memset(dst_pixels_y_opt, 0, kPixels);
>+  memset(dst_pixels_y_c, 1, kPixels);
>+
>+  MaskCpuFlags(disable_cpu_flags_);
>+  Convert16To8Plane(reinterpret_cast<const uint16_t*>(src_pixels_y),
>+                    benchmark_width_, dst_pixels_y_c, benchmark_width_, 16384,
>+                    benchmark_width_, benchmark_height_);
>+  MaskCpuFlags(benchmark_cpu_info_);
>+
>+  for (int i = 0; i < benchmark_iterations_; ++i) {
>+    Convert16To8Plane(reinterpret_cast<const uint16_t*>(src_pixels_y),
>+                      benchmark_width_, dst_pixels_y_opt, benchmark_width_,
>+                      16384, benchmark_width_, benchmark_height_);
>+  }
>+
>+  for (int i = 0; i < kPixels; ++i) {
>+    EXPECT_EQ(dst_pixels_y_opt[i], dst_pixels_y_c[i]);
>+  }
>+
>+  free_aligned_buffer_page_end(src_pixels_y);
>+  free_aligned_buffer_page_end(dst_pixels_y_opt);
>+  free_aligned_buffer_page_end(dst_pixels_y_c);
>+}
>+
>+// TODO(fbarchard): Improve test for more platforms.
>+#ifdef HAS_CONVERT16TO8ROW_AVX2
>+TEST_F(LibYUVPlanarTest, Convert16To8Row_Opt) {
>+  // AVX2 does multiple of 32, so round count up
>+  const int kPixels = (benchmark_width_ * benchmark_height_ + 31) & ~31;
>+  align_buffer_page_end(src_pixels_y, kPixels * 2);
>+  align_buffer_page_end(dst_pixels_y_opt, kPixels);
>+  align_buffer_page_end(dst_pixels_y_c, kPixels);
>+
>+  MemRandomize(src_pixels_y, kPixels * 2);
>+  // clamp source range to 10 bits.
>+  for (int i = 0; i < kPixels; ++i) {
>+    reinterpret_cast<uint16_t*>(src_pixels_y)[i] &= 1023;
>+  }
>+
>+  memset(dst_pixels_y_opt, 0, kPixels);
>+  memset(dst_pixels_y_c, 1, kPixels);
>+
>+  Convert16To8Row_C(reinterpret_cast<const uint16_t*>(src_pixels_y),
>+                    dst_pixels_y_c, 16384, kPixels);
>+
>+  int has_avx2 = TestCpuFlag(kCpuHasAVX2);
>+  int has_ssse3 = TestCpuFlag(kCpuHasSSSE3);
>+  for (int i = 0; i < benchmark_iterations_; ++i) {
>+    if (has_avx2) {
>+      Convert16To8Row_AVX2(reinterpret_cast<const uint16_t*>(src_pixels_y),
>+                           dst_pixels_y_opt, 16384, kPixels);
>+    } else if (has_ssse3) {
>+      Convert16To8Row_SSSE3(reinterpret_cast<const uint16_t*>(src_pixels_y),
>+                            dst_pixels_y_opt, 16384, kPixels);
>+    } else {
>+      Convert16To8Row_C(reinterpret_cast<const uint16_t*>(src_pixels_y),
>+                        dst_pixels_y_opt, 16384, kPixels);
>+    }
>+  }
>+
>+  for (int i = 0; i < kPixels; ++i) {
>+    EXPECT_EQ(dst_pixels_y_opt[i], dst_pixels_y_c[i]);
>+  }
>+
>+  free_aligned_buffer_page_end(src_pixels_y);
>+  free_aligned_buffer_page_end(dst_pixels_y_opt);
>+  free_aligned_buffer_page_end(dst_pixels_y_c);
>+}
>+#endif  // HAS_CONVERT16TO8ROW_AVX2
>+
>+TEST_F(LibYUVPlanarTest, Convert8To16Plane) {
>+  const int kPixels = benchmark_width_ * benchmark_height_;
>+  align_buffer_page_end(src_pixels_y, kPixels);
>+  align_buffer_page_end(dst_pixels_y_opt, kPixels * 2);
>+  align_buffer_page_end(dst_pixels_y_c, kPixels * 2);
>+
>+  MemRandomize(src_pixels_y, kPixels);
>+  memset(dst_pixels_y_opt, 0, kPixels * 2);
>+  memset(dst_pixels_y_c, 1, kPixels * 2);
>+
>+  MaskCpuFlags(disable_cpu_flags_);
>+  Convert8To16Plane(src_pixels_y, benchmark_width_,
>+                    reinterpret_cast<uint16_t*>(dst_pixels_y_c),
>+                    benchmark_width_, 1024, benchmark_width_,
>+                    benchmark_height_);
>+  MaskCpuFlags(benchmark_cpu_info_);
>+
>+  for (int i = 0; i < benchmark_iterations_; ++i) {
>+    Convert8To16Plane(src_pixels_y, benchmark_width_,
>+                      reinterpret_cast<uint16_t*>(dst_pixels_y_opt),
>+                      benchmark_width_, 1024, benchmark_width_,
>+                      benchmark_height_);
>+  }
>+
>+  for (int i = 0; i < kPixels * 2; ++i) {
>+    EXPECT_EQ(dst_pixels_y_opt[i], dst_pixels_y_c[i]);
>+  }
>+
>+  free_aligned_buffer_page_end(src_pixels_y);
>+  free_aligned_buffer_page_end(dst_pixels_y_opt);
>+  free_aligned_buffer_page_end(dst_pixels_y_c);
>+}
>+
>+// TODO(fbarchard): Improve test for more platforms.
>+#ifdef HAS_CONVERT8TO16ROW_AVX2
>+TEST_F(LibYUVPlanarTest, Convert8To16Row_Opt) {
>+  const int kPixels = (benchmark_width_ * benchmark_height_ + 31) & ~31;
>+  align_buffer_page_end(src_pixels_y, kPixels);
>+  align_buffer_page_end(dst_pixels_y_opt, kPixels * 2);
>+  align_buffer_page_end(dst_pixels_y_c, kPixels * 2);
>+
>+  MemRandomize(src_pixels_y, kPixels);
>+  memset(dst_pixels_y_opt, 0, kPixels * 2);
>+  memset(dst_pixels_y_c, 1, kPixels * 2);
>+
>+  Convert8To16Row_C(src_pixels_y, reinterpret_cast<uint16_t*>(dst_pixels_y_c),
>+                    1024, kPixels);
>+
>+  int has_avx2 = TestCpuFlag(kCpuHasAVX2);
>+  int has_sse2 = TestCpuFlag(kCpuHasSSE2);
>+  for (int i = 0; i < benchmark_iterations_; ++i) {
>+    if (has_avx2) {
>+      Convert8To16Row_AVX2(src_pixels_y,
>+                           reinterpret_cast<uint16_t*>(dst_pixels_y_opt), 1024,
>+                           kPixels);
>+    } else if (has_sse2) {
>+      Convert8To16Row_SSE2(src_pixels_y,
>+                           reinterpret_cast<uint16_t*>(dst_pixels_y_opt), 1024,
>+                           kPixels);
>+    } else {
>+      Convert8To16Row_C(src_pixels_y,
>+                        reinterpret_cast<uint16_t*>(dst_pixels_y_opt), 1024,
>+                        kPixels);
>+    }
>+  }
>+
>+  for (int i = 0; i < kPixels * 2; ++i) {
>+    EXPECT_EQ(dst_pixels_y_opt[i], dst_pixels_y_c[i]);
>+  }
>+
>+  free_aligned_buffer_page_end(src_pixels_y);
>+  free_aligned_buffer_page_end(dst_pixels_y_opt);
>+  free_aligned_buffer_page_end(dst_pixels_y_c);
>+}
>+#endif  // HAS_CONVERT8TO16ROW_AVX2
> 
> float TestScaleMaxSamples(int benchmark_width,
>                           int benchmark_height,
>@@ -2709,8 +2908,8 @@ float TestScaleMaxSamples(int benchmark_width,
>   // NEON does multiple of 8, so round count up
>   const int kPixels = (benchmark_width * benchmark_height + 7) & ~7;
>   align_buffer_page_end(orig_y, kPixels * 4 * 3 + 48);
>-  uint8* dst_c = orig_y + kPixels * 4 + 16;
>-  uint8* dst_opt = orig_y + kPixels * 4 * 2 + 32;
>+  uint8_t* dst_c = orig_y + kPixels * 4 + 16;
>+  uint8_t* dst_opt = orig_y + kPixels * 4 * 2 + 32;
> 
>   // Randomize works but may contain some denormals affecting performance.
>   // MemRandomize(orig_y, kPixels * 4);
>@@ -2777,8 +2976,8 @@ float TestScaleSumSamples(int benchmark_width,
>   // NEON does multiple of 8, so round count up
>   const int kPixels = (benchmark_width * benchmark_height + 7) & ~7;
>   align_buffer_page_end(orig_y, kPixels * 4 * 3);
>-  uint8* dst_c = orig_y + kPixels * 4;
>-  uint8* dst_opt = orig_y + kPixels * 4 * 2;
>+  uint8_t* dst_c = orig_y + kPixels * 4;
>+  uint8_t* dst_opt = orig_y + kPixels * 4 * 2;
> 
>   // Randomize works but may contain some denormals affecting performance.
>   // MemRandomize(orig_y, kPixels * 4);
>@@ -2855,8 +3054,8 @@ float TestScaleSamples(int benchmark_width,
>   // NEON does multiple of 8, so round count up
>   const int kPixels = (benchmark_width * benchmark_height + 7) & ~7;
>   align_buffer_page_end(orig_y, kPixels * 4 * 3);
>-  uint8* dst_c = orig_y + kPixels * 4;
>-  uint8* dst_opt = orig_y + kPixels * 4 * 2;
>+  uint8_t* dst_c = orig_y + kPixels * 4;
>+  uint8_t* dst_opt = orig_y + kPixels * 4 * 2;
> 
>   // Randomize works but may contain some denormals affecting performance.
>   // MemRandomize(orig_y, kPixels * 4);
>@@ -2918,8 +3117,8 @@ float TestCopySamples(int benchmark_width,
>   // NEON does multiple of 16 floats, so round count up
>   const int kPixels = (benchmark_width * benchmark_height + 15) & ~15;
>   align_buffer_page_end(orig_y, kPixels * 4 * 3);
>-  uint8* dst_c = orig_y + kPixels * 4;
>-  uint8* dst_opt = orig_y + kPixels * 4 * 2;
>+  uint8_t* dst_c = orig_y + kPixels * 4;
>+  uint8_t* dst_opt = orig_y + kPixels * 4 * 2;
> 
>   // Randomize works but may contain some denormals affecting performance.
>   // MemRandomize(orig_y, kPixels * 4);
>@@ -2970,13 +3169,13 @@ TEST_F(LibYUVPlanarTest, TestCopySamples_Opt) {
>   EXPECT_EQ(0, diff);
> }
> 
>-extern "C" void GaussRow_NEON(const uint32* src, uint16* dst, int width);
>-extern "C" void GaussRow_C(const uint32* src, uint16* dst, int width);
>+extern "C" void GaussRow_NEON(const uint32_t* src, uint16_t* dst, int width);
>+extern "C" void GaussRow_C(const uint32_t* src, uint16_t* dst, int width);
> 
> TEST_F(LibYUVPlanarTest, TestGaussRow_Opt) {
>-  SIMD_ALIGNED(uint32 orig_pixels[640 + 4]);
>-  SIMD_ALIGNED(uint16 dst_pixels_c[640]);
>-  SIMD_ALIGNED(uint16 dst_pixels_opt[640]);
>+  SIMD_ALIGNED(uint32_t orig_pixels[640 + 4]);
>+  SIMD_ALIGNED(uint16_t dst_pixels_c[640]);
>+  SIMD_ALIGNED(uint16_t dst_pixels_opt[640]);
> 
>   memset(orig_pixels, 0, sizeof(orig_pixels));
>   memset(dst_pixels_c, 1, sizeof(dst_pixels_c));
>@@ -3004,30 +3203,30 @@ TEST_F(LibYUVPlanarTest, TestGaussRow_Opt) {
>   }
> 
>   EXPECT_EQ(dst_pixels_c[0],
>-            static_cast<uint16>(0 * 1 + 1 * 4 + 2 * 6 + 3 * 4 + 4 * 1));
>-  EXPECT_EQ(dst_pixels_c[639], static_cast<uint16>(10256));
>+            static_cast<uint16_t>(0 * 1 + 1 * 4 + 2 * 6 + 3 * 4 + 4 * 1));
>+  EXPECT_EQ(dst_pixels_c[639], static_cast<uint16_t>(10256));
> }
> 
>-extern "C" void GaussCol_NEON(const uint16* src0,
>-                              const uint16* src1,
>-                              const uint16* src2,
>-                              const uint16* src3,
>-                              const uint16* src4,
>-                              uint32* dst,
>+extern "C" void GaussCol_NEON(const uint16_t* src0,
>+                              const uint16_t* src1,
>+                              const uint16_t* src2,
>+                              const uint16_t* src3,
>+                              const uint16_t* src4,
>+                              uint32_t* dst,
>                               int width);
> 
>-extern "C" void GaussCol_C(const uint16* src0,
>-                           const uint16* src1,
>-                           const uint16* src2,
>-                           const uint16* src3,
>-                           const uint16* src4,
>-                           uint32* dst,
>+extern "C" void GaussCol_C(const uint16_t* src0,
>+                           const uint16_t* src1,
>+                           const uint16_t* src2,
>+                           const uint16_t* src3,
>+                           const uint16_t* src4,
>+                           uint32_t* dst,
>                            int width);
> 
> TEST_F(LibYUVPlanarTest, TestGaussCol_Opt) {
>-  SIMD_ALIGNED(uint16 orig_pixels[640 * 5]);
>-  SIMD_ALIGNED(uint32 dst_pixels_c[640]);
>-  SIMD_ALIGNED(uint32 dst_pixels_opt[640]);
>+  SIMD_ALIGNED(uint16_t orig_pixels[640 * 5]);
>+  SIMD_ALIGNED(uint32_t dst_pixels_c[640]);
>+  SIMD_ALIGNED(uint32_t dst_pixels_opt[640]);
> 
>   memset(orig_pixels, 0, sizeof(orig_pixels));
>   memset(dst_pixels_c, 1, sizeof(dst_pixels_c));
>@@ -3062,9 +3261,10 @@ TEST_F(LibYUVPlanarTest, TestGaussCol_Opt) {
>     EXPECT_EQ(dst_pixels_c[i], dst_pixels_opt[i]);
>   }
> 
>-  EXPECT_EQ(dst_pixels_c[0], static_cast<uint32>(0 * 1 + 640 * 4 + 640 * 2 * 6 +
>-                                                 640 * 3 * 4 + 640 * 4 * 1));
>-  EXPECT_EQ(dst_pixels_c[639], static_cast<uint32>(30704));
>+  EXPECT_EQ(dst_pixels_c[0],
>+            static_cast<uint32_t>(0 * 1 + 640 * 4 + 640 * 2 * 6 + 640 * 3 * 4 +
>+                                  640 * 4 * 1));
>+  EXPECT_EQ(dst_pixels_c[639], static_cast<uint32_t>(30704));
> }
> 
> }  // namespace libyuv
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/unit_test/scale_argb_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/unit_test/scale_argb_test.cc
>index d11aec20476..a1be85b8d8a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/unit_test/scale_argb_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/unit_test/scale_argb_test.cc
>@@ -37,7 +37,7 @@ static int ARGBTestFilter(int src_width,
> 
>   int i, j;
>   const int b = 0;  // 128 to test for padding/stride.
>-  int64 src_argb_plane_size =
>+  int64_t src_argb_plane_size =
>       (Abs(src_width) + b * 2) * (Abs(src_height) + b * 2) * 4LL;
>   int src_stride_argb = (b * 2 + Abs(src_width)) * 4;
> 
>@@ -48,7 +48,8 @@ static int ARGBTestFilter(int src_width,
>   }
>   MemRandomize(src_argb, src_argb_plane_size);
> 
>-  int64 dst_argb_plane_size = (dst_width + b * 2) * (dst_height + b * 2) * 4LL;
>+  int64_t dst_argb_plane_size =
>+      (dst_width + b * 2) * (dst_height + b * 2) * 4LL;
>   int dst_stride_argb = (b * 2 + dst_width) * 4;
> 
>   align_buffer_page_end(dst_argb_c, dst_argb_plane_size);
>@@ -116,11 +117,11 @@ static int ARGBTestFilter(int src_width,
> static const int kTileX = 8;
> static const int kTileY = 8;
> 
>-static int TileARGBScale(const uint8* src_argb,
>+static int TileARGBScale(const uint8_t* src_argb,
>                          int src_stride_argb,
>                          int src_width,
>                          int src_height,
>-                         uint8* dst_argb,
>+                         uint8_t* dst_argb,
>                          int dst_stride_argb,
>                          int dst_width,
>                          int dst_height,
>@@ -157,7 +158,7 @@ static int ARGBClipTestFilter(int src_width,
>   }
> 
>   const int b = 128;
>-  int64 src_argb_plane_size =
>+  int64_t src_argb_plane_size =
>       (Abs(src_width) + b * 2) * (Abs(src_height) + b * 2) * 4;
>   int src_stride_argb = (b * 2 + Abs(src_width)) * 4;
> 
>@@ -168,7 +169,7 @@ static int ARGBClipTestFilter(int src_width,
>   }
>   memset(src_argb, 1, src_argb_plane_size);
> 
>-  int64 dst_argb_plane_size = (dst_width + b * 2) * (dst_height + b * 2) * 4;
>+  int64_t dst_argb_plane_size = (dst_width + b * 2) * (dst_height + b * 2) * 4;
>   int dst_stride_argb = (b * 2 + dst_width) * 4;
> 
>   int i, j;
>@@ -310,19 +311,20 @@ TEST_SCALETO(ARGBScale, 1280, 720)
> #undef TEST_SCALETO
> 
> // Scale with YUV conversion to ARGB and clipping.
>+// TODO(fbarchard): Add fourcc support.  All 4 ARGB formats is easy to support.
> LIBYUV_API
>-int YUVToARGBScaleReference2(const uint8* src_y,
>+int YUVToARGBScaleReference2(const uint8_t* src_y,
>                              int src_stride_y,
>-                             const uint8* src_u,
>+                             const uint8_t* src_u,
>                              int src_stride_u,
>-                             const uint8* src_v,
>+                             const uint8_t* src_v,
>                              int src_stride_v,
>-                             uint32 /* src_fourcc */,  // TODO: Add support.
>+                             uint32 /* src_fourcc */,
>                              int src_width,
>                              int src_height,
>-                             uint8* dst_argb,
>+                             uint8_t* dst_argb,
>                              int dst_stride_argb,
>-                             uint32 /* dst_fourcc */,  // TODO: Add support.
>+                             uint32 /* dst_fourcc */,
>                              int dst_width,
>                              int dst_height,
>                              int clip_x,
>@@ -330,7 +332,8 @@ int YUVToARGBScaleReference2(const uint8* src_y,
>                              int clip_width,
>                              int clip_height,
>                              enum FilterMode filtering) {
>-  uint8* argb_buffer = static_cast<uint8*>(malloc(src_width * src_height * 4));
>+  uint8_t* argb_buffer =
>+      static_cast<uint8_t*>(malloc(src_width * src_height * 4));
>   int r;
>   I420ToARGB(src_y, src_stride_y, src_u, src_stride_u, src_v, src_stride_v,
>              argb_buffer, src_width * 4, src_width, src_height);
>@@ -342,7 +345,12 @@ int YUVToARGBScaleReference2(const uint8* src_y,
>   return r;
> }
> 
>-static void FillRamp(uint8* buf, int width, int height, int v, int dx, int dy) {
>+static void FillRamp(uint8_t* buf,
>+                     int width,
>+                     int height,
>+                     int v,
>+                     int dx,
>+                     int dy) {
>   int rv = v;
>   for (int y = 0; y < height; ++y) {
>     for (int x = 0; x < width; ++x) {
>@@ -369,8 +377,8 @@ static int YUVToARGBTestFilter(int src_width,
>                                int dst_height,
>                                FilterMode f,
>                                int benchmark_iterations) {
>-  int64 src_y_plane_size = Abs(src_width) * Abs(src_height);
>-  int64 src_uv_plane_size =
>+  int64_t src_y_plane_size = Abs(src_width) * Abs(src_height);
>+  int64_t src_uv_plane_size =
>       ((Abs(src_width) + 1) / 2) * ((Abs(src_height) + 1) / 2);
>   int src_stride_y = Abs(src_width);
>   int src_stride_uv = (Abs(src_width) + 1) / 2;
>@@ -379,7 +387,7 @@ static int YUVToARGBTestFilter(int src_width,
>   align_buffer_page_end(src_u, src_uv_plane_size);
>   align_buffer_page_end(src_v, src_uv_plane_size);
> 
>-  int64 dst_argb_plane_size = (dst_width) * (dst_height)*4LL;
>+  int64_t dst_argb_plane_size = (dst_width) * (dst_height)*4LL;
>   int dst_stride_argb = (dst_width)*4;
>   align_buffer_page_end(dst_argb_c, dst_argb_plane_size);
>   align_buffer_page_end(dst_argb_opt, dst_argb_plane_size);
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/unit_test/scale_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/unit_test/scale_test.cc
>index c39211a161b..08b6cffaa26 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/unit_test/scale_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/unit_test/scale_test.cc
>@@ -38,8 +38,8 @@ static int TestFilter(int src_width,
>   int src_width_uv = (Abs(src_width) + 1) >> 1;
>   int src_height_uv = (Abs(src_height) + 1) >> 1;
> 
>-  int64 src_y_plane_size = (Abs(src_width)) * (Abs(src_height));
>-  int64 src_uv_plane_size = (src_width_uv) * (src_height_uv);
>+  int64_t src_y_plane_size = (Abs(src_width)) * (Abs(src_height));
>+  int64_t src_uv_plane_size = (src_width_uv) * (src_height_uv);
> 
>   int src_stride_y = Abs(src_width);
>   int src_stride_uv = src_width_uv;
>@@ -58,8 +58,8 @@ static int TestFilter(int src_width,
>   int dst_width_uv = (dst_width + 1) >> 1;
>   int dst_height_uv = (dst_height + 1) >> 1;
> 
>-  int64 dst_y_plane_size = (dst_width) * (dst_height);
>-  int64 dst_uv_plane_size = (dst_width_uv) * (dst_height_uv);
>+  int64_t dst_y_plane_size = (dst_width) * (dst_height);
>+  int64_t dst_uv_plane_size = (dst_width_uv) * (dst_height_uv);
> 
>   int dst_stride_y = dst_width;
>   int dst_stride_uv = dst_width_uv;
>@@ -157,8 +157,8 @@ static int TestFilter_16(int src_width,
>   int src_width_uv = (Abs(src_width) + 1) >> 1;
>   int src_height_uv = (Abs(src_height) + 1) >> 1;
> 
>-  int64 src_y_plane_size = (Abs(src_width)) * (Abs(src_height));
>-  int64 src_uv_plane_size = (src_width_uv) * (src_height_uv);
>+  int64_t src_y_plane_size = (Abs(src_width)) * (Abs(src_height));
>+  int64_t src_uv_plane_size = (src_width_uv) * (src_height_uv);
> 
>   int src_stride_y = Abs(src_width);
>   int src_stride_uv = src_width_uv;
>@@ -173,9 +173,9 @@ static int TestFilter_16(int src_width,
>     printf("Skipped.  Alloc failed " FILELINESTR(__FILE__, __LINE__) "\n");
>     return 0;
>   }
>-  uint16* p_src_y_16 = reinterpret_cast<uint16*>(src_y_16);
>-  uint16* p_src_u_16 = reinterpret_cast<uint16*>(src_u_16);
>-  uint16* p_src_v_16 = reinterpret_cast<uint16*>(src_v_16);
>+  uint16_t* p_src_y_16 = reinterpret_cast<uint16_t*>(src_y_16);
>+  uint16_t* p_src_u_16 = reinterpret_cast<uint16_t*>(src_u_16);
>+  uint16_t* p_src_v_16 = reinterpret_cast<uint16_t*>(src_v_16);
> 
>   MemRandomize(src_y, src_y_plane_size);
>   MemRandomize(src_u, src_uv_plane_size);
>@@ -205,9 +205,9 @@ static int TestFilter_16(int src_width,
>   align_buffer_page_end(dst_u_16, dst_uv_plane_size * 2);
>   align_buffer_page_end(dst_v_16, dst_uv_plane_size * 2);
> 
>-  uint16* p_dst_y_16 = reinterpret_cast<uint16*>(dst_y_16);
>-  uint16* p_dst_u_16 = reinterpret_cast<uint16*>(dst_u_16);
>-  uint16* p_dst_v_16 = reinterpret_cast<uint16*>(dst_v_16);
>+  uint16_t* p_dst_y_16 = reinterpret_cast<uint16_t*>(dst_y_16);
>+  uint16_t* p_dst_u_16 = reinterpret_cast<uint16_t*>(dst_u_16);
>+  uint16_t* p_dst_v_16 = reinterpret_cast<uint16_t*>(dst_v_16);
> 
>   MaskCpuFlags(disable_cpu_flags);  // Disable all CPU optimization.
>   I420Scale(src_y, src_stride_y, src_u, src_stride_uv, src_v, src_stride_uv,
>@@ -345,9 +345,9 @@ TEST_SCALETO(Scale, 1280, 720)
> 
> #ifdef HAS_SCALEROWDOWN2_SSSE3
> TEST_F(LibYUVScaleTest, TestScaleRowDown2Box_Odd_SSSE3) {
>-  SIMD_ALIGNED(uint8 orig_pixels[128 * 2]);
>-  SIMD_ALIGNED(uint8 dst_pixels_opt[64]);
>-  SIMD_ALIGNED(uint8 dst_pixels_c[64]);
>+  SIMD_ALIGNED(uint8_t orig_pixels[128 * 2]);
>+  SIMD_ALIGNED(uint8_t dst_pixels_opt[64]);
>+  SIMD_ALIGNED(uint8_t dst_pixels_c[64]);
>   memset(orig_pixels, 0, sizeof(orig_pixels));
>   memset(dst_pixels_opt, 0, sizeof(dst_pixels_opt));
>   memset(dst_pixels_c, 0, sizeof(dst_pixels_c));
>@@ -433,19 +433,19 @@ TEST_F(LibYUVScaleTest, TestScaleRowDown2Box_Odd_SSSE3) {
> }
> #endif  // HAS_SCALEROWDOWN2_SSSE3
> 
>-extern "C" void ScaleRowUp2_16_NEON(const uint16* src_ptr,
>+extern "C" void ScaleRowUp2_16_NEON(const uint16_t* src_ptr,
>                                     ptrdiff_t src_stride,
>-                                    uint16* dst,
>+                                    uint16_t* dst,
>                                     int dst_width);
>-extern "C" void ScaleRowUp2_16_C(const uint16* src_ptr,
>+extern "C" void ScaleRowUp2_16_C(const uint16_t* src_ptr,
>                                  ptrdiff_t src_stride,
>-                                 uint16* dst,
>+                                 uint16_t* dst,
>                                  int dst_width);
> 
> TEST_F(LibYUVScaleTest, TestScaleRowUp2_16) {
>-  SIMD_ALIGNED(uint16 orig_pixels[640 * 2 + 1]);  // 2 rows + 1 pixel overrun.
>-  SIMD_ALIGNED(uint16 dst_pixels_opt[1280]);
>-  SIMD_ALIGNED(uint16 dst_pixels_c[1280]);
>+  SIMD_ALIGNED(uint16_t orig_pixels[640 * 2 + 1]);  // 2 rows + 1 pixel overrun.
>+  SIMD_ALIGNED(uint16_t dst_pixels_opt[1280]);
>+  SIMD_ALIGNED(uint16_t dst_pixels_c[1280]);
> 
>   memset(orig_pixels, 0, sizeof(orig_pixels));
>   memset(dst_pixels_opt, 1, sizeof(dst_pixels_opt));
>@@ -475,15 +475,15 @@ TEST_F(LibYUVScaleTest, TestScaleRowUp2_16) {
>   EXPECT_EQ(dst_pixels_c[1279], 800);
> }
> 
>-extern "C" void ScaleRowDown2Box_16_NEON(const uint16* src_ptr,
>+extern "C" void ScaleRowDown2Box_16_NEON(const uint16_t* src_ptr,
>                                          ptrdiff_t src_stride,
>-                                         uint16* dst,
>+                                         uint16_t* dst,
>                                          int dst_width);
> 
> TEST_F(LibYUVScaleTest, TestScaleRowDown2Box_16) {
>-  SIMD_ALIGNED(uint16 orig_pixels[2560 * 2]);
>-  SIMD_ALIGNED(uint16 dst_pixels_c[1280]);
>-  SIMD_ALIGNED(uint16 dst_pixels_opt[1280]);
>+  SIMD_ALIGNED(uint16_t orig_pixels[2560 * 2]);
>+  SIMD_ALIGNED(uint16_t dst_pixels_c[1280]);
>+  SIMD_ALIGNED(uint16_t dst_pixels_opt[1280]);
> 
>   memset(orig_pixels, 0, sizeof(orig_pixels));
>   memset(dst_pixels_c, 1, sizeof(dst_pixels_c));
>@@ -530,7 +530,7 @@ static int TestPlaneFilter_16(int src_width,
>   }
> 
>   int i;
>-  int64 src_y_plane_size = (Abs(src_width)) * (Abs(src_height));
>+  int64_t src_y_plane_size = (Abs(src_width)) * (Abs(src_height));
>   int src_stride_y = Abs(src_width);
>   int dst_y_plane_size = dst_width * dst_height;
>   int dst_stride_y = dst_width;
>@@ -539,8 +539,8 @@ static int TestPlaneFilter_16(int src_width,
>   align_buffer_page_end(src_y_16, src_y_plane_size * 2);
>   align_buffer_page_end(dst_y_8, dst_y_plane_size);
>   align_buffer_page_end(dst_y_16, dst_y_plane_size * 2);
>-  uint16* p_src_y_16 = reinterpret_cast<uint16*>(src_y_16);
>-  uint16* p_dst_y_16 = reinterpret_cast<uint16*>(dst_y_16);
>+  uint16_t* p_src_y_16 = reinterpret_cast<uint16_t*>(src_y_16);
>+  uint16_t* p_dst_y_16 = reinterpret_cast<uint16_t*>(dst_y_16);
> 
>   MemRandomize(src_y, src_y_plane_size);
>   memset(dst_y_8, 0, dst_y_plane_size);
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/unit_test/unit_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/unit_test/unit_test.cc
>index 1ad4dece861..20aadb44e2f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/unit_test/unit_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/unit_test/unit_test.cc
>@@ -19,10 +19,6 @@
> #endif
> #include "libyuv/cpu_id.h"
> 
>-// Change this to 1000 for benchmarking.
>-// TODO(fbarchard): Add command line parsing to pass this as option.
>-#define BENCHMARK_ITERATIONS 1
>-
> unsigned int fastrand_seed = 0xfb;
> 
> #ifdef LIBYUV_USE_GFLAGS
>@@ -35,11 +31,11 @@ DEFINE_int32(libyuv_cpu_info,
>              "cpu flags for benchmark code. 1 = C, -1 = SIMD");
> #else
> // Disable command line parameters if gflags disabled.
>-static const int32 FLAGS_libyuv_width = 0;
>-static const int32 FLAGS_libyuv_height = 0;
>-static const int32 FLAGS_libyuv_repeat = 0;
>-static const int32 FLAGS_libyuv_flags = 0;
>-static const int32 FLAGS_libyuv_cpu_info = 0;
>+static const int32_t FLAGS_libyuv_width = 0;
>+static const int32_t FLAGS_libyuv_height = 0;
>+static const int32_t FLAGS_libyuv_repeat = 0;
>+static const int32_t FLAGS_libyuv_flags = 0;
>+static const int32_t FLAGS_libyuv_cpu_info = 0;
> #endif
> 
> // For quicker unittests, default is 128 x 72.  But when benchmarking,
>@@ -47,7 +43,7 @@ static const int32 FLAGS_libyuv_cpu_info = 0;
> // Set flags to -1 for benchmarking to avoid slower C code.
> 
> LibYUVConvertTest::LibYUVConvertTest()
>-    : benchmark_iterations_(BENCHMARK_ITERATIONS),
>+    : benchmark_iterations_(1),
>       benchmark_width_(128),
>       benchmark_height_(72),
>       disable_cpu_flags_(1),
>@@ -92,12 +88,6 @@ LibYUVConvertTest::LibYUVConvertTest()
>     benchmark_cpu_info_ = FLAGS_libyuv_cpu_info;
>   }
>   libyuv::MaskCpuFlags(benchmark_cpu_info_);
>-  benchmark_pixels_div256_ =
>-      static_cast<int>((static_cast<double>(Abs(benchmark_width_)) *
>-                            static_cast<double>(Abs(benchmark_height_)) *
>-                            static_cast<double>(benchmark_iterations_) +
>-                        255.0) /
>-                       256.0);
>   benchmark_pixels_div1280_ =
>       static_cast<int>((static_cast<double>(Abs(benchmark_width_)) *
>                             static_cast<double>(Abs(benchmark_height_)) *
>@@ -107,7 +97,7 @@ LibYUVConvertTest::LibYUVConvertTest()
> }
> 
> LibYUVColorTest::LibYUVColorTest()
>-    : benchmark_iterations_(BENCHMARK_ITERATIONS),
>+    : benchmark_iterations_(1),
>       benchmark_width_(128),
>       benchmark_height_(72),
>       disable_cpu_flags_(1),
>@@ -152,12 +142,6 @@ LibYUVColorTest::LibYUVColorTest()
>     benchmark_cpu_info_ = FLAGS_libyuv_cpu_info;
>   }
>   libyuv::MaskCpuFlags(benchmark_cpu_info_);
>-  benchmark_pixels_div256_ =
>-      static_cast<int>((static_cast<double>(Abs(benchmark_width_)) *
>-                            static_cast<double>(Abs(benchmark_height_)) *
>-                            static_cast<double>(benchmark_iterations_) +
>-                        255.0) /
>-                       256.0);
>   benchmark_pixels_div1280_ =
>       static_cast<int>((static_cast<double>(Abs(benchmark_width_)) *
>                             static_cast<double>(Abs(benchmark_height_)) *
>@@ -167,7 +151,7 @@ LibYUVColorTest::LibYUVColorTest()
> }
> 
> LibYUVScaleTest::LibYUVScaleTest()
>-    : benchmark_iterations_(BENCHMARK_ITERATIONS),
>+    : benchmark_iterations_(1),
>       benchmark_width_(128),
>       benchmark_height_(72),
>       disable_cpu_flags_(1),
>@@ -212,12 +196,6 @@ LibYUVScaleTest::LibYUVScaleTest()
>     benchmark_cpu_info_ = FLAGS_libyuv_cpu_info;
>   }
>   libyuv::MaskCpuFlags(benchmark_cpu_info_);
>-  benchmark_pixels_div256_ =
>-      static_cast<int>((static_cast<double>(Abs(benchmark_width_)) *
>-                            static_cast<double>(Abs(benchmark_height_)) *
>-                            static_cast<double>(benchmark_iterations_) +
>-                        255.0) /
>-                       256.0);
>   benchmark_pixels_div1280_ =
>       static_cast<int>((static_cast<double>(Abs(benchmark_width_)) *
>                             static_cast<double>(Abs(benchmark_height_)) *
>@@ -227,7 +205,7 @@ LibYUVScaleTest::LibYUVScaleTest()
> }
> 
> LibYUVRotateTest::LibYUVRotateTest()
>-    : benchmark_iterations_(BENCHMARK_ITERATIONS),
>+    : benchmark_iterations_(1),
>       benchmark_width_(128),
>       benchmark_height_(72),
>       disable_cpu_flags_(1),
>@@ -272,12 +250,6 @@ LibYUVRotateTest::LibYUVRotateTest()
>     benchmark_cpu_info_ = FLAGS_libyuv_cpu_info;
>   }
>   libyuv::MaskCpuFlags(benchmark_cpu_info_);
>-  benchmark_pixels_div256_ =
>-      static_cast<int>((static_cast<double>(Abs(benchmark_width_)) *
>-                            static_cast<double>(Abs(benchmark_height_)) *
>-                            static_cast<double>(benchmark_iterations_) +
>-                        255.0) /
>-                       256.0);
>   benchmark_pixels_div1280_ =
>       static_cast<int>((static_cast<double>(Abs(benchmark_width_)) *
>                             static_cast<double>(Abs(benchmark_height_)) *
>@@ -287,7 +259,7 @@ LibYUVRotateTest::LibYUVRotateTest()
> }
> 
> LibYUVPlanarTest::LibYUVPlanarTest()
>-    : benchmark_iterations_(BENCHMARK_ITERATIONS),
>+    : benchmark_iterations_(1),
>       benchmark_width_(128),
>       benchmark_height_(72),
>       disable_cpu_flags_(1),
>@@ -332,12 +304,6 @@ LibYUVPlanarTest::LibYUVPlanarTest()
>     benchmark_cpu_info_ = FLAGS_libyuv_cpu_info;
>   }
>   libyuv::MaskCpuFlags(benchmark_cpu_info_);
>-  benchmark_pixels_div256_ =
>-      static_cast<int>((static_cast<double>(Abs(benchmark_width_)) *
>-                            static_cast<double>(Abs(benchmark_height_)) *
>-                            static_cast<double>(benchmark_iterations_) +
>-                        255.0) /
>-                       256.0);
>   benchmark_pixels_div1280_ =
>       static_cast<int>((static_cast<double>(Abs(benchmark_width_)) *
>                             static_cast<double>(Abs(benchmark_height_)) *
>@@ -347,7 +313,7 @@ LibYUVPlanarTest::LibYUVPlanarTest()
> }
> 
> LibYUVBaseTest::LibYUVBaseTest()
>-    : benchmark_iterations_(BENCHMARK_ITERATIONS),
>+    : benchmark_iterations_(1),
>       benchmark_width_(128),
>       benchmark_height_(72),
>       disable_cpu_flags_(1),
>@@ -392,12 +358,6 @@ LibYUVBaseTest::LibYUVBaseTest()
>     benchmark_cpu_info_ = FLAGS_libyuv_cpu_info;
>   }
>   libyuv::MaskCpuFlags(benchmark_cpu_info_);
>-  benchmark_pixels_div256_ =
>-      static_cast<int>((static_cast<double>(Abs(benchmark_width_)) *
>-                            static_cast<double>(Abs(benchmark_height_)) *
>-                            static_cast<double>(benchmark_iterations_) +
>-                        255.0) /
>-                       256.0);
>   benchmark_pixels_div1280_ =
>       static_cast<int>((static_cast<double>(Abs(benchmark_width_)) *
>                             static_cast<double>(Abs(benchmark_height_)) *
>@@ -407,7 +367,7 @@ LibYUVBaseTest::LibYUVBaseTest()
> }
> 
> LibYUVCompareTest::LibYUVCompareTest()
>-    : benchmark_iterations_(BENCHMARK_ITERATIONS),
>+    : benchmark_iterations_(1),
>       benchmark_width_(128),
>       benchmark_height_(72),
>       disable_cpu_flags_(1),
>@@ -452,12 +412,6 @@ LibYUVCompareTest::LibYUVCompareTest()
>     benchmark_cpu_info_ = FLAGS_libyuv_cpu_info;
>   }
>   libyuv::MaskCpuFlags(benchmark_cpu_info_);
>-  benchmark_pixels_div256_ =
>-      static_cast<int>((static_cast<double>(Abs(benchmark_width_)) *
>-                            static_cast<double>(Abs(benchmark_height_)) *
>-                            static_cast<double>(benchmark_iterations_) +
>-                        255.0) /
>-                       256.0);
>   benchmark_pixels_div1280_ =
>       static_cast<int>((static_cast<double>(Abs(benchmark_width_)) *
>                             static_cast<double>(Abs(benchmark_height_)) *
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/unit_test/unit_test.h b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/unit_test/unit_test.h
>index 6454389d52d..87907fa1603 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/unit_test/unit_test.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/unit_test/unit_test.h
>@@ -69,10 +69,10 @@ static inline bool SizeValid(int src_width,
>   return true;
> }
> 
>-#define align_buffer_page_end(var, size)                              \
>-  uint8* var##_mem =                                                  \
>-      reinterpret_cast<uint8*>(malloc(((size) + 4095 + 63) & ~4095)); \
>-  uint8* var = reinterpret_cast<uint8*>(                              \
>+#define align_buffer_page_end(var, size)                                \
>+  uint8_t* var##_mem =                                                  \
>+      reinterpret_cast<uint8_t*>(malloc(((size) + 4095 + 63) & ~4095)); \
>+  uint8_t* var = reinterpret_cast<uint8_t*>(                            \
>       (intptr_t)(var##_mem + (((size) + 4095 + 63) & ~4095) - (size)) & ~63)
> 
> #define free_aligned_buffer_page_end(var) \
>@@ -111,10 +111,10 @@ inline int fastrand() {
>   return static_cast<int>((fastrand_seed >> 16) & 0xffff);
> }
> 
>-static inline void MemRandomize(uint8* dst, int64 len) {
>-  int64 i;
>+static inline void MemRandomize(uint8_t* dst, int64_t len) {
>+  int64_t i;
>   for (i = 0; i < len - 1; i += 2) {
>-    *reinterpret_cast<uint16*>(dst) = fastrand();
>+    *reinterpret_cast<uint16_t*>(dst) = fastrand();
>     dst += 2;
>   }
>   for (; i < len; ++i) {
>@@ -126,10 +126,9 @@ class LibYUVColorTest : public ::testing::Test {
>  protected:
>   LibYUVColorTest();
> 
>-  int benchmark_iterations_;     // Default 1. Use 1000 for benchmarking.
>-  int benchmark_width_;          // Default 1280.  Use 640 for benchmarking VGA.
>-  int benchmark_height_;         // Default 720.  Use 360 for benchmarking VGA.
>-  int benchmark_pixels_div256_;  // Total pixels to benchmark / 256.
>+  int benchmark_iterations_;  // Default 1. Use 1000 for benchmarking.
>+  int benchmark_width_;       // Default 1280.  Use 640 for benchmarking VGA.
>+  int benchmark_height_;      // Default 720.  Use 360 for benchmarking VGA.
>   int benchmark_pixels_div1280_;  // Total pixels to benchmark / 1280.
>   int disable_cpu_flags_;         // Default 1.  Use -1 for benchmarking.
>   int benchmark_cpu_info_;        // Default -1.  Use 1 to disable SIMD.
>@@ -139,10 +138,9 @@ class LibYUVConvertTest : public ::testing::Test {
>  protected:
>   LibYUVConvertTest();
> 
>-  int benchmark_iterations_;     // Default 1. Use 1000 for benchmarking.
>-  int benchmark_width_;          // Default 1280.  Use 640 for benchmarking VGA.
>-  int benchmark_height_;         // Default 720.  Use 360 for benchmarking VGA.
>-  int benchmark_pixels_div256_;  // Total pixels to benchmark / 256.
>+  int benchmark_iterations_;  // Default 1. Use 1000 for benchmarking.
>+  int benchmark_width_;       // Default 1280.  Use 640 for benchmarking VGA.
>+  int benchmark_height_;      // Default 720.  Use 360 for benchmarking VGA.
>   int benchmark_pixels_div1280_;  // Total pixels to benchmark / 1280.
>   int disable_cpu_flags_;         // Default 1.  Use -1 for benchmarking.
>   int benchmark_cpu_info_;        // Default -1.  Use 1 to disable SIMD.
>@@ -152,10 +150,9 @@ class LibYUVScaleTest : public ::testing::Test {
>  protected:
>   LibYUVScaleTest();
> 
>-  int benchmark_iterations_;     // Default 1. Use 1000 for benchmarking.
>-  int benchmark_width_;          // Default 1280.  Use 640 for benchmarking VGA.
>-  int benchmark_height_;         // Default 720.  Use 360 for benchmarking VGA.
>-  int benchmark_pixels_div256_;  // Total pixels to benchmark / 256.
>+  int benchmark_iterations_;  // Default 1. Use 1000 for benchmarking.
>+  int benchmark_width_;       // Default 1280.  Use 640 for benchmarking VGA.
>+  int benchmark_height_;      // Default 720.  Use 360 for benchmarking VGA.
>   int benchmark_pixels_div1280_;  // Total pixels to benchmark / 1280.
>   int disable_cpu_flags_;         // Default 1.  Use -1 for benchmarking.
>   int benchmark_cpu_info_;        // Default -1.  Use 1 to disable SIMD.
>@@ -165,10 +162,9 @@ class LibYUVRotateTest : public ::testing::Test {
>  protected:
>   LibYUVRotateTest();
> 
>-  int benchmark_iterations_;     // Default 1. Use 1000 for benchmarking.
>-  int benchmark_width_;          // Default 1280.  Use 640 for benchmarking VGA.
>-  int benchmark_height_;         // Default 720.  Use 360 for benchmarking VGA.
>-  int benchmark_pixels_div256_;  // Total pixels to benchmark / 256.
>+  int benchmark_iterations_;  // Default 1. Use 1000 for benchmarking.
>+  int benchmark_width_;       // Default 1280.  Use 640 for benchmarking VGA.
>+  int benchmark_height_;      // Default 720.  Use 360 for benchmarking VGA.
>   int benchmark_pixels_div1280_;  // Total pixels to benchmark / 1280.
>   int disable_cpu_flags_;         // Default 1.  Use -1 for benchmarking.
>   int benchmark_cpu_info_;        // Default -1.  Use 1 to disable SIMD.
>@@ -178,10 +174,9 @@ class LibYUVPlanarTest : public ::testing::Test {
>  protected:
>   LibYUVPlanarTest();
> 
>-  int benchmark_iterations_;     // Default 1. Use 1000 for benchmarking.
>-  int benchmark_width_;          // Default 1280.  Use 640 for benchmarking VGA.
>-  int benchmark_height_;         // Default 720.  Use 360 for benchmarking VGA.
>-  int benchmark_pixels_div256_;  // Total pixels to benchmark / 256.
>+  int benchmark_iterations_;  // Default 1. Use 1000 for benchmarking.
>+  int benchmark_width_;       // Default 1280.  Use 640 for benchmarking VGA.
>+  int benchmark_height_;      // Default 720.  Use 360 for benchmarking VGA.
>   int benchmark_pixels_div1280_;  // Total pixels to benchmark / 1280.
>   int disable_cpu_flags_;         // Default 1.  Use -1 for benchmarking.
>   int benchmark_cpu_info_;        // Default -1.  Use 1 to disable SIMD.
>@@ -191,10 +186,9 @@ class LibYUVBaseTest : public ::testing::Test {
>  protected:
>   LibYUVBaseTest();
> 
>-  int benchmark_iterations_;     // Default 1. Use 1000 for benchmarking.
>-  int benchmark_width_;          // Default 1280.  Use 640 for benchmarking VGA.
>-  int benchmark_height_;         // Default 720.  Use 360 for benchmarking VGA.
>-  int benchmark_pixels_div256_;  // Total pixels to benchmark / 256.
>+  int benchmark_iterations_;  // Default 1. Use 1000 for benchmarking.
>+  int benchmark_width_;       // Default 1280.  Use 640 for benchmarking VGA.
>+  int benchmark_height_;      // Default 720.  Use 360 for benchmarking VGA.
>   int benchmark_pixels_div1280_;  // Total pixels to benchmark / 1280.
>   int disable_cpu_flags_;         // Default 1.  Use -1 for benchmarking.
>   int benchmark_cpu_info_;        // Default -1.  Use 1 to disable SIMD.
>@@ -204,10 +198,9 @@ class LibYUVCompareTest : public ::testing::Test {
>  protected:
>   LibYUVCompareTest();
> 
>-  int benchmark_iterations_;     // Default 1. Use 1000 for benchmarking.
>-  int benchmark_width_;          // Default 1280.  Use 640 for benchmarking VGA.
>-  int benchmark_height_;         // Default 720.  Use 360 for benchmarking VGA.
>-  int benchmark_pixels_div256_;  // Total pixels to benchmark / 256.
>+  int benchmark_iterations_;  // Default 1. Use 1000 for benchmarking.
>+  int benchmark_width_;       // Default 1280.  Use 640 for benchmarking VGA.
>+  int benchmark_height_;      // Default 720.  Use 360 for benchmarking VGA.
>   int benchmark_pixels_div1280_;  // Total pixels to benchmark / 1280.
>   int disable_cpu_flags_;         // Default 1.  Use -1 for benchmarking.
>   int benchmark_cpu_info_;        // Default -1.  Use 1 to disable SIMD.
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/unit_test/video_common_test.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/unit_test/video_common_test.cc
>index f16b6772f95..4d89586e76f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/unit_test/video_common_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/unit_test/video_common_test.cc
>@@ -18,15 +18,12 @@ namespace libyuv {
> 
> // Tests FourCC codes in video common, which are used for ConvertToI420().
> 
>-static bool TestValidChar(uint32 onecc) {
>-  if ((onecc >= '0' && onecc <= '9') || (onecc >= 'A' && onecc <= 'Z') ||
>-      (onecc >= 'a' && onecc <= 'z') || (onecc == ' ') || (onecc == 0xff)) {
>-    return true;
>-  }
>-  return false;
>+static bool TestValidChar(uint32_t onecc) {
>+  return (onecc >= '0' && onecc <= '9') || (onecc >= 'A' && onecc <= 'Z') ||
>+         (onecc >= 'a' && onecc <= 'z') || (onecc == ' ') || (onecc == 0xff);
> }
> 
>-static bool TestValidFourCC(uint32 fourcc, int bpp) {
>+static bool TestValidFourCC(uint32_t fourcc, int bpp) {
>   if (!TestValidChar(fourcc & 0xff) || !TestValidChar((fourcc >> 8) & 0xff) ||
>       !TestValidChar((fourcc >> 16) & 0xff) ||
>       !TestValidChar((fourcc >> 24) & 0xff)) {
>@@ -39,23 +36,23 @@ static bool TestValidFourCC(uint32 fourcc, int bpp) {
> }
> 
> TEST_F(LibYUVBaseTest, TestCanonicalFourCC) {
>-  EXPECT_EQ(static_cast<uint32>(FOURCC_I420), CanonicalFourCC(FOURCC_IYUV));
>-  EXPECT_EQ(static_cast<uint32>(FOURCC_I420), CanonicalFourCC(FOURCC_YU12));
>-  EXPECT_EQ(static_cast<uint32>(FOURCC_I422), CanonicalFourCC(FOURCC_YU16));
>-  EXPECT_EQ(static_cast<uint32>(FOURCC_I444), CanonicalFourCC(FOURCC_YU24));
>-  EXPECT_EQ(static_cast<uint32>(FOURCC_YUY2), CanonicalFourCC(FOURCC_YUYV));
>-  EXPECT_EQ(static_cast<uint32>(FOURCC_YUY2), CanonicalFourCC(FOURCC_YUVS));
>-  EXPECT_EQ(static_cast<uint32>(FOURCC_UYVY), CanonicalFourCC(FOURCC_HDYC));
>-  EXPECT_EQ(static_cast<uint32>(FOURCC_UYVY), CanonicalFourCC(FOURCC_2VUY));
>-  EXPECT_EQ(static_cast<uint32>(FOURCC_MJPG), CanonicalFourCC(FOURCC_JPEG));
>-  EXPECT_EQ(static_cast<uint32>(FOURCC_MJPG), CanonicalFourCC(FOURCC_DMB1));
>-  EXPECT_EQ(static_cast<uint32>(FOURCC_RAW), CanonicalFourCC(FOURCC_RGB3));
>-  EXPECT_EQ(static_cast<uint32>(FOURCC_24BG), CanonicalFourCC(FOURCC_BGR3));
>-  EXPECT_EQ(static_cast<uint32>(FOURCC_BGRA), CanonicalFourCC(FOURCC_CM32));
>-  EXPECT_EQ(static_cast<uint32>(FOURCC_RAW), CanonicalFourCC(FOURCC_CM24));
>-  EXPECT_EQ(static_cast<uint32>(FOURCC_RGBO), CanonicalFourCC(FOURCC_L555));
>-  EXPECT_EQ(static_cast<uint32>(FOURCC_RGBP), CanonicalFourCC(FOURCC_L565));
>-  EXPECT_EQ(static_cast<uint32>(FOURCC_RGBO), CanonicalFourCC(FOURCC_5551));
>+  EXPECT_EQ(static_cast<uint32_t>(FOURCC_I420), CanonicalFourCC(FOURCC_IYUV));
>+  EXPECT_EQ(static_cast<uint32_t>(FOURCC_I420), CanonicalFourCC(FOURCC_YU12));
>+  EXPECT_EQ(static_cast<uint32_t>(FOURCC_I422), CanonicalFourCC(FOURCC_YU16));
>+  EXPECT_EQ(static_cast<uint32_t>(FOURCC_I444), CanonicalFourCC(FOURCC_YU24));
>+  EXPECT_EQ(static_cast<uint32_t>(FOURCC_YUY2), CanonicalFourCC(FOURCC_YUYV));
>+  EXPECT_EQ(static_cast<uint32_t>(FOURCC_YUY2), CanonicalFourCC(FOURCC_YUVS));
>+  EXPECT_EQ(static_cast<uint32_t>(FOURCC_UYVY), CanonicalFourCC(FOURCC_HDYC));
>+  EXPECT_EQ(static_cast<uint32_t>(FOURCC_UYVY), CanonicalFourCC(FOURCC_2VUY));
>+  EXPECT_EQ(static_cast<uint32_t>(FOURCC_MJPG), CanonicalFourCC(FOURCC_JPEG));
>+  EXPECT_EQ(static_cast<uint32_t>(FOURCC_MJPG), CanonicalFourCC(FOURCC_DMB1));
>+  EXPECT_EQ(static_cast<uint32_t>(FOURCC_RAW), CanonicalFourCC(FOURCC_RGB3));
>+  EXPECT_EQ(static_cast<uint32_t>(FOURCC_24BG), CanonicalFourCC(FOURCC_BGR3));
>+  EXPECT_EQ(static_cast<uint32_t>(FOURCC_BGRA), CanonicalFourCC(FOURCC_CM32));
>+  EXPECT_EQ(static_cast<uint32_t>(FOURCC_RAW), CanonicalFourCC(FOURCC_CM24));
>+  EXPECT_EQ(static_cast<uint32_t>(FOURCC_RGBO), CanonicalFourCC(FOURCC_L555));
>+  EXPECT_EQ(static_cast<uint32_t>(FOURCC_RGBP), CanonicalFourCC(FOURCC_L565));
>+  EXPECT_EQ(static_cast<uint32_t>(FOURCC_RGBO), CanonicalFourCC(FOURCC_5551));
> }
> 
> TEST_F(LibYUVBaseTest, TestFourCC) {
>@@ -73,12 +70,16 @@ TEST_F(LibYUVBaseTest, TestFourCC) {
>   EXPECT_TRUE(TestValidFourCC(FOURCC_ARGB, FOURCC_BPP_ARGB));
>   EXPECT_TRUE(TestValidFourCC(FOURCC_BGRA, FOURCC_BPP_BGRA));
>   EXPECT_TRUE(TestValidFourCC(FOURCC_ABGR, FOURCC_BPP_ABGR));
>+  EXPECT_TRUE(TestValidFourCC(FOURCC_AR30, FOURCC_BPP_AR30));
>+  EXPECT_TRUE(TestValidFourCC(FOURCC_AB30, FOURCC_BPP_AB30));
>   EXPECT_TRUE(TestValidFourCC(FOURCC_24BG, FOURCC_BPP_24BG));
>   EXPECT_TRUE(TestValidFourCC(FOURCC_RAW, FOURCC_BPP_RAW));
>   EXPECT_TRUE(TestValidFourCC(FOURCC_RGBA, FOURCC_BPP_RGBA));
>   EXPECT_TRUE(TestValidFourCC(FOURCC_RGBP, FOURCC_BPP_RGBP));
>   EXPECT_TRUE(TestValidFourCC(FOURCC_RGBO, FOURCC_BPP_RGBO));
>   EXPECT_TRUE(TestValidFourCC(FOURCC_R444, FOURCC_BPP_R444));
>+  EXPECT_TRUE(TestValidFourCC(FOURCC_H420, FOURCC_BPP_H420));
>+  EXPECT_TRUE(TestValidFourCC(FOURCC_H010, FOURCC_BPP_H010));
>   EXPECT_TRUE(TestValidFourCC(FOURCC_MJPG, FOURCC_BPP_MJPG));
>   EXPECT_TRUE(TestValidFourCC(FOURCC_YV12, FOURCC_BPP_YV12));
>   EXPECT_TRUE(TestValidFourCC(FOURCC_YV16, FOURCC_BPP_YV16));
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/util/Makefile b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/util/Makefile
>index 6044d2adf63..40e74b6509c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/util/Makefile
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/util/Makefile
>@@ -4,3 +4,6 @@ ifeq ($(CXX),icl)
> else
> 	$(CXX) -msse2 -O3 -fopenmp -static -o psnr psnr.cc ssim.cc psnr_main.cc -Wl,--strip-all
> endif
>+
>+# for MacOS
>+# /usr/local/bin/g++-7 -msse2 -O3 -fopenmp -Bstatic -o psnr psnr.cc ssim.cc psnr_main.cc
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/util/compare.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/util/compare.cc
>index ef0beefafee..a16613ee2f9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/util/compare.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/util/compare.cc
>@@ -29,22 +29,24 @@ int main(int argc, char** argv) {
>   FILE* fin2 = name2 ? fopen(name2, "rb") : NULL;
> 
>   const int kBlockSize = 32768;
>-  uint8 buf1[kBlockSize];
>-  uint8 buf2[kBlockSize];
>-  uint32 hash1 = 5381;
>-  uint32 hash2 = 5381;
>-  uint64 sum_square_err = 0;
>-  uint64 size_min = 0;
>+  uint8_t buf1[kBlockSize];
>+  uint8_t buf2[kBlockSize];
>+  uint32_t hash1 = 5381;
>+  uint32_t hash2 = 5381;
>+  uint64_t sum_square_err = 0;
>+  uint64_t size_min = 0;
>   int amt1 = 0;
>   int amt2 = 0;
>   do {
>     amt1 = static_cast<int>(fread(buf1, 1, kBlockSize, fin1));
>-    if (amt1 > 0)
>+    if (amt1 > 0) {
>       hash1 = libyuv::HashDjb2(buf1, amt1, hash1);
>+    }
>     if (fin2) {
>       amt2 = static_cast<int>(fread(buf2, 1, kBlockSize, fin2));
>-      if (amt2 > 0)
>+      if (amt2 > 0) {
>         hash2 = libyuv::HashDjb2(buf2, amt2, hash2);
>+      }
>       int amt_min = (amt1 < amt2) ? amt1 : amt2;
>       size_min += amt_min;
>       sum_square_err += libyuv::ComputeSumSquareError(buf1, buf2, amt_min);
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/util/cpuid.c b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/util/cpuid.c
>index 9ff618e0d28..59c65d60e0f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/util/cpuid.c
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/util/cpuid.c
>@@ -69,8 +69,8 @@ int main(int argc, const char* argv[]) {
>     printf("Has NEON %x\n", has_neon);
>   }
>   if (has_mips) {
>-    int has_dspr2 = TestCpuFlag(kCpuHasDSPR2);
>-    printf("Has DSPR2 %x\n", has_dspr2);
>+    int has_msa = TestCpuFlag(kCpuHasMSA);
>+    printf("Has MSA %x\n", has_msa);
>   }
>   if (has_x86) {
>     int has_sse2 = TestCpuFlag(kCpuHasSSE2);
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/util/psnr.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/util/psnr.cc
>index 27f876c0b4a..f54015bab82 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/util/psnr.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/util/psnr.cc
>@@ -21,14 +21,14 @@
> extern "C" {
> #endif
> 
>-typedef unsigned int uint32;  // NOLINT
>+typedef unsigned int uint32_t;  // NOLINT
> #ifdef _MSC_VER
>-typedef unsigned __int64 uint64;
>+typedef unsigned __int64 uint64_t;
> #else  // COMPILER_MSVC
> #if defined(__LP64__) && !defined(__OpenBSD__) && !defined(__APPLE__)
>-typedef unsigned long uint64;  // NOLINT
>+typedef unsigned long uint64_t;  // NOLINT
> #else   // defined(__LP64__) && !defined(__OpenBSD__) && !defined(__APPLE__)
>-typedef unsigned long long uint64;  // NOLINT
>+typedef unsigned long long uint64_t;  // NOLINT
> #endif  // __LP64__
> #endif  // _MSC_VER
> 
>@@ -38,10 +38,10 @@ typedef unsigned long long uint64;  // NOLINT
> #if !defined(LIBYUV_DISABLE_NEON) && defined(__ARM_NEON__) && \
>     !defined(__aarch64__)
> #define HAS_SUMSQUAREERROR_NEON
>-static uint32 SumSquareError_NEON(const uint8* src_a,
>-                                  const uint8* src_b,
>-                                  int count) {
>-  volatile uint32 sse;
>+static uint32_t SumSquareError_NEON(const uint8_t* src_a,
>+                                    const uint8_t* src_b,
>+                                    int count) {
>+  volatile uint32_t sse;
>   asm volatile(
>       "vmov.u8    q7, #0                         \n"
>       "vmov.u8    q9, #0                         \n"
>@@ -73,10 +73,10 @@ static uint32 SumSquareError_NEON(const uint8* src_a,
> }
> #elif !defined(LIBYUV_DISABLE_NEON) && defined(__aarch64__)
> #define HAS_SUMSQUAREERROR_NEON
>-static uint32 SumSquareError_NEON(const uint8* src_a,
>-                                  const uint8* src_b,
>-                                  int count) {
>-  volatile uint32 sse;
>+static uint32_t SumSquareError_NEON(const uint8_t* src_a,
>+                                    const uint8_t* src_b,
>+                                    int count) {
>+  volatile uint32_t sse;
>   asm volatile(
>       "eor        v16.16b, v16.16b, v16.16b      \n"
>       "eor        v18.16b, v18.16b, v18.16b      \n"
>@@ -107,9 +107,9 @@ static uint32 SumSquareError_NEON(const uint8* src_a,
> }
> #elif !defined(LIBYUV_DISABLE_X86) && defined(_M_IX86) && defined(_MSC_VER)
> #define HAS_SUMSQUAREERROR_SSE2
>-__declspec(naked) static uint32 SumSquareError_SSE2(const uint8* /*src_a*/,
>-                                                    const uint8* /*src_b*/,
>-                                                    int /*count*/) {
>+__declspec(naked) static uint32_t SumSquareError_SSE2(const uint8_t* /*src_a*/,
>+                                                      const uint8_t* /*src_b*/,
>+                                                      int /*count*/) {
>   __asm {
>     mov        eax, [esp + 4]  // src_a
>     mov        edx, [esp + 8]  // src_b
>@@ -146,10 +146,10 @@ __declspec(naked) static uint32 SumSquareError_SSE2(const uint8* /*src_a*/,
> }
> #elif !defined(LIBYUV_DISABLE_X86) && (defined(__x86_64__) || defined(__i386__))
> #define HAS_SUMSQUAREERROR_SSE2
>-static uint32 SumSquareError_SSE2(const uint8* src_a,
>-                                  const uint8* src_b,
>-                                  int count) {
>-  uint32 sse;
>+static uint32_t SumSquareError_SSE2(const uint8_t* src_a,
>+                                    const uint8_t* src_b,
>+                                    int count) {
>+  uint32_t sse;
>   asm volatile(  // NOLINT
>       "pxor      %%xmm0,%%xmm0                   \n"
>       "pxor      %%xmm5,%%xmm5                   \n"
>@@ -228,22 +228,22 @@ static int CpuHasSSE2() {
> }
> #endif  // HAS_SUMSQUAREERROR_SSE2
> 
>-static uint32 SumSquareError_C(const uint8* src_a,
>-                               const uint8* src_b,
>-                               int count) {
>-  uint32 sse = 0u;
>+static uint32_t SumSquareError_C(const uint8_t* src_a,
>+                                 const uint8_t* src_b,
>+                                 int count) {
>+  uint32_t sse = 0u;
>   for (int x = 0; x < count; ++x) {
>     int diff = src_a[x] - src_b[x];
>-    sse += static_cast<uint32>(diff * diff);
>+    sse += static_cast<uint32_t>(diff * diff);
>   }
>   return sse;
> }
> 
>-double ComputeSumSquareError(const uint8* src_a,
>-                             const uint8* src_b,
>+double ComputeSumSquareError(const uint8_t* src_a,
>+                             const uint8_t* src_b,
>                              int count) {
>-  uint32 (*SumSquareError)(const uint8* src_a, const uint8* src_b, int count) =
>-      SumSquareError_C;
>+  uint32_t (*SumSquareError)(const uint8_t* src_a, const uint8_t* src_b,
>+                             int count) = SumSquareError_C;
> #if defined(HAS_SUMSQUAREERROR_NEON)
>   SumSquareError = SumSquareError_NEON;
> #endif
>@@ -253,7 +253,7 @@ double ComputeSumSquareError(const uint8* src_a,
>   }
> #endif
>   const int kBlockSize = 1 << 15;
>-  uint64 sse = 0;
>+  uint64_t sse = 0;
> #ifdef _OPENMP
> #pragma omp parallel for reduction(+ : sse)
> #endif
>@@ -280,8 +280,9 @@ double ComputeSumSquareError(const uint8* src_a,
> // Returns 128.0 (kMaxPSNR) if sse is 0 (perfect match).
> double ComputePSNR(double sse, double size) {
>   const double kMINSSE = 255.0 * 255.0 * size / pow(10.0, kMaxPSNR / 10.0);
>-  if (sse <= kMINSSE)
>+  if (sse <= kMINSSE) {
>     sse = kMINSSE;  // Produces max PSNR of 128
>+  }
>   return 10.0 * log10(255.0 * 255.0 * size / sse);
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/util/psnr.h b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/util/psnr.h
>index 0816b976001..aac128cbca8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/util/psnr.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/util/psnr.h
>@@ -20,7 +20,7 @@ extern "C" {
> #endif
> 
> #if !defined(INT_TYPES_DEFINED) && !defined(UINT8_TYPE_DEFINED)
>-typedef unsigned char uint8;
>+typedef unsigned char uint8_t;
> #define UINT8_TYPE_DEFINED
> #endif
> 
>@@ -31,7 +31,9 @@ static const double kMaxPSNR = 128.0;
> #if !defined(HAVE_JPEG)
> // Computer Sum of Squared Error (SSE).
> // Pass this to ComputePSNR for final result.
>-double ComputeSumSquareError(const uint8* org, const uint8* rec, int size);
>+double ComputeSumSquareError(const uint8_t* src_a,
>+                             const uint8_t* src_b,
>+                             int count);
> #endif
> 
> // PSNR formula: psnr = 10 * log10 (Peak Signal^2 * size / sse)
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/util/psnr_main.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/util/psnr_main.cc
>index 4d930be4aed..a930b202ecf 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/util/psnr_main.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/util/psnr_main.cc
>@@ -90,9 +90,9 @@ bool ExtractResolutionFromFilename(const char* name,
>   fseek(file_org, 0, SEEK_END);
>   size_t total_size = ftell(file_org);
>   fseek(file_org, 0, SEEK_SET);
>-  uint8* const ch_org = new uint8[total_size];
>+  uint8_t* const ch_org = new uint8_t[total_size];
>   memset(ch_org, 0, total_size);
>-  size_t bytes_org = fread(ch_org, sizeof(uint8), total_size, file_org);
>+  size_t bytes_org = fread(ch_org, sizeof(uint8_t), total_size, file_org);
>   fclose(file_org);
>   if (bytes_org == total_size) {
>     if (0 == libyuv::MJPGSize(ch_org, total_size, width_ptr, height_ptr)) {
>@@ -107,13 +107,15 @@ bool ExtractResolutionFromFilename(const char* name,
> 
> // Scale Y channel from 16..240 to 0..255.
> // This can be useful when comparing codecs that are inconsistant about Y
>-uint8 ScaleY(uint8 y) {
>+uint8_t ScaleY(uint8_t y) {
>   int ny = (y - 16) * 256 / 224;
>-  if (ny < 0)
>+  if (ny < 0) {
>     ny = 0;
>-  if (ny > 255)
>+  }
>+  if (ny > 255) {
>     ny = 255;
>-  return static_cast<uint8>(ny);
>+  }
>+  return static_cast<uint8_t>(ny);
> }
> 
> // MSE = Mean Square Error
>@@ -150,8 +152,9 @@ void PrintHelp(const char* program) {
> }
> 
> void ParseOptions(int argc, const char* argv[]) {
>-  if (argc <= 1)
>+  if (argc <= 1) {
>     PrintHelp(argv[0]);
>+  }
>   for (int c = 1; c < argc; ++c) {
>     if (!strcmp(argv[c], "-v")) {
>       verbose = true;
>@@ -237,8 +240,8 @@ void ParseOptions(int argc, const char* argv[]) {
>   }
> }
> 
>-bool UpdateMetrics(uint8* ch_org,
>-                   uint8* ch_rec,
>+bool UpdateMetrics(uint8_t* ch_org,
>+                   uint8_t* ch_rec,
>                    const int y_size,
>                    const int uv_size,
>                    const size_t total_size,
>@@ -247,10 +250,10 @@ bool UpdateMetrics(uint8* ch_org,
>                    metric* distorted_frame,
>                    bool do_psnr) {
>   const int uv_offset = (do_swap_uv ? uv_size : 0);
>-  const uint8* const u_org = ch_org + y_size + uv_offset;
>-  const uint8* const u_rec = ch_rec + y_size;
>-  const uint8* const v_org = ch_org + y_size + (uv_size - uv_offset);
>-  const uint8* const v_rec = ch_rec + y_size + uv_size;
>+  const uint8_t* const u_org = ch_org + y_size + uv_offset;
>+  const uint8_t* const u_rec = ch_rec + y_size;
>+  const uint8_t* const v_org = ch_org + y_size + (uv_size - uv_offset);
>+  const uint8_t* const v_rec = ch_rec + y_size + uv_size;
>   if (do_psnr) {
> #ifdef HAVE_JPEG
>     double y_err = static_cast<double>(
>@@ -301,12 +304,15 @@ bool UpdateMetrics(uint8* ch_org,
>   cur_distortion_psnr->all += distorted_frame->all;
> 
>   bool ismin = false;
>-  if (distorted_frame->y < cur_distortion_psnr->min_y)
>+  if (distorted_frame->y < cur_distortion_psnr->min_y) {
>     cur_distortion_psnr->min_y = distorted_frame->y;
>-  if (distorted_frame->u < cur_distortion_psnr->min_u)
>+  }
>+  if (distorted_frame->u < cur_distortion_psnr->min_u) {
>     cur_distortion_psnr->min_u = distorted_frame->u;
>-  if (distorted_frame->v < cur_distortion_psnr->min_v)
>+  }
>+  if (distorted_frame->v < cur_distortion_psnr->min_v) {
>     cur_distortion_psnr->min_v = distorted_frame->v;
>+  }
>   if (distorted_frame->all < cur_distortion_psnr->min_all) {
>     cur_distortion_psnr->min_all = distorted_frame->all;
>     cur_distortion_psnr->min_frame = number_of_frames;
>@@ -374,8 +380,8 @@ int main(int argc, const char* argv[]) {
> #endif
>   }
> 
>-  uint8* const ch_org = new uint8[total_size];
>-  uint8* const ch_rec = new uint8[total_size];
>+  uint8_t* const ch_org = new uint8_t[total_size];
>+  uint8_t* const ch_rec = new uint8_t[total_size];
>   if (ch_org == NULL || ch_rec == NULL) {
>     fprintf(stderr, "No memory available\n");
>     fclose(file_org);
>@@ -429,14 +435,15 @@ int main(int argc, const char* argv[]) {
> 
>   int number_of_frames;
>   for (number_of_frames = 0;; ++number_of_frames) {
>-    if (num_frames && number_of_frames >= num_frames)
>+    if (num_frames && number_of_frames >= num_frames) {
>       break;
>+    }
> 
>-    size_t bytes_org = fread(ch_org, sizeof(uint8), total_size, file_org);
>+    size_t bytes_org = fread(ch_org, sizeof(uint8_t), total_size, file_org);
>     if (bytes_org < total_size) {
> #ifdef HAVE_JPEG
>       // Try parsing file as a jpeg.
>-      uint8* const ch_jpeg = new uint8[bytes_org];
>+      uint8_t* const ch_jpeg = new uint8_t[bytes_org];
>       memcpy(ch_jpeg, ch_org, bytes_org);
>       memset(ch_org, 0, total_size);
> 
>@@ -456,11 +463,11 @@ int main(int argc, const char* argv[]) {
> 
>     for (int cur_rec = 0; cur_rec < num_rec; ++cur_rec) {
>       size_t bytes_rec =
>-          fread(ch_rec, sizeof(uint8), total_size, file_rec[cur_rec]);
>+          fread(ch_rec, sizeof(uint8_t), total_size, file_rec[cur_rec]);
>       if (bytes_rec < total_size) {
> #ifdef HAVE_JPEG
>         // Try parsing file as a jpeg.
>-        uint8* const ch_jpeg = new uint8[bytes_rec];
>+        uint8_t* const ch_jpeg = new uint8_t[bytes_rec];
>         memcpy(ch_jpeg, ch_rec, bytes_rec);
>         memset(ch_rec, 0, total_size);
> 
>@@ -482,7 +489,7 @@ int main(int argc, const char* argv[]) {
>         printf("%5d", number_of_frames);
>       }
>       if (do_psnr) {
>-        metric distorted_frame;
>+        metric distorted_frame = {};
>         metric* cur_distortion_psnr = &distortion_psnr[cur_rec];
>         bool ismin = UpdateMetrics(ch_org, ch_rec, y_size, uv_size, total_size,
>                                    number_of_frames, cur_distortion_psnr,
>@@ -496,7 +503,7 @@ int main(int argc, const char* argv[]) {
>         }
>       }
>       if (do_ssim) {
>-        metric distorted_frame;
>+        metric distorted_frame = {};
>         metric* cur_distortion_ssim = &distortion_ssim[cur_rec];
>         bool ismin = UpdateMetrics(ch_org, ch_rec, y_size, uv_size, total_size,
>                                    number_of_frames, cur_distortion_ssim,
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/util/ssim.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/util/ssim.cc
>index 43e725d8210..096fbcf0610 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/util/ssim.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/util/ssim.cc
>@@ -16,8 +16,8 @@
> extern "C" {
> #endif
> 
>-typedef unsigned int uint32;    // NOLINT
>-typedef unsigned short uint16;  // NOLINT
>+typedef unsigned int uint32_t;    // NOLINT
>+typedef unsigned short uint16_t;  // NOLINT
> 
> #if !defined(LIBYUV_DISABLE_X86) && !defined(__SSE2__) && \
>     (defined(_M_X64) || (defined(_M_IX86_FP) && (_M_IX86_FP >= 2)))
>@@ -50,7 +50,7 @@ static const double kiW[KERNEL + 1 + 1] = {
> 
> #if !defined(LIBYUV_DISABLE_X86) && defined(__SSE2__)
> 
>-#define PWEIGHT(A, B) static_cast<uint16>(K[(A)] * K[(B)])  // weight product
>+#define PWEIGHT(A, B) static_cast<uint16_t>(K[(A)] * K[(B)])  // weight product
> #define MAKE_WEIGHT(L)                                                \
>   {                                                                   \
>     {                                                                 \
>@@ -66,7 +66,7 @@ static const double kiW[KERNEL + 1 + 1] = {
> // values. We can't call _mm_set_epi16() for static compile-time initialization.
> static const struct {
>   union {
>-    uint16 i16_[8];
>+    uint16_t i16_[8];
>     __m128i m_;
>   } values_;
> } W0 = MAKE_WEIGHT(0), W1 = MAKE_WEIGHT(1), W2 = MAKE_WEIGHT(2),
>@@ -88,10 +88,12 @@ static double FinalizeSSIM(double iw,
>   double sxx = xxm * iw - iwx * iwx;
>   double syy = yym * iw - iwy * iwy;
>   // small errors are possible, due to rounding. Clamp to zero.
>-  if (sxx < 0.)
>+  if (sxx < 0.) {
>     sxx = 0.;
>-  if (syy < 0.)
>+  }
>+  if (syy < 0.) {
>     syy = 0.;
>+  }
>   const double sxsy = sqrt(sxx * syy);
>   const double sxy = xym * iw - iwx * iwy;
>   static const double C11 = (0.01 * 0.01) * (255 * 255);
>@@ -109,21 +111,22 @@ static double FinalizeSSIM(double iw,
> // Note: worst case of accumulation is a weight of 33 = 11 + 2 * (7 + 3 + 1)
> // with a diff of 255, squared. The maximum error is thus 0x4388241,
> // which fits into 32 bits integers.
>-double GetSSIM(const uint8* org,
>-               const uint8* rec,
>+double GetSSIM(const uint8_t* org,
>+               const uint8_t* rec,
>                int xo,
>                int yo,
>                int W,
>                int H,
>                int stride) {
>-  uint32 ws = 0, xm = 0, ym = 0, xxm = 0, xym = 0, yym = 0;
>+  uint32_t ws = 0, xm = 0, ym = 0, xxm = 0, xym = 0, yym = 0;
>   org += (yo - KERNEL) * stride;
>   org += (xo - KERNEL);
>   rec += (yo - KERNEL) * stride;
>   rec += (xo - KERNEL);
>   for (int y_ = 0; y_ < KERNEL_SIZE; ++y_, org += stride, rec += stride) {
>-    if (((yo - KERNEL + y_) < 0) || ((yo - KERNEL + y_) >= H))
>+    if (((yo - KERNEL + y_) < 0) || ((yo - KERNEL + y_) >= H)) {
>       continue;
>+    }
>     const int Wy = K[y_];
>     for (int x_ = 0; x_ < KERNEL_SIZE; ++x_) {
>       const int Wxy = Wy * K[x_];
>@@ -142,13 +145,13 @@ double GetSSIM(const uint8* org,
>   return FinalizeSSIM(1. / ws, xm, ym, xxm, xym, yym);
> }
> 
>-double GetSSIMFullKernel(const uint8* org,
>-                         const uint8* rec,
>+double GetSSIMFullKernel(const uint8_t* org,
>+                         const uint8_t* rec,
>                          int xo,
>                          int yo,
>                          int stride,
>                          double area_weight) {
>-  uint32 xm = 0, ym = 0, xxm = 0, xym = 0, yym = 0;
>+  uint32_t xm = 0, ym = 0, xxm = 0, xym = 0, yym = 0;
> 
> #if defined(LIBYUV_DISABLE_X86) || !defined(__SSE2__)
> 
>@@ -262,7 +265,7 @@ double GetSSIMFullKernel(const uint8* org,
> 
> #define ADD_AND_STORE_FOUR_EPI32(M, OUT)                    \
>   do {                                                      \
>-    uint32 tmp[4];                                          \
>+    uint32_t tmp[4];                                        \
>     _mm_storeu_si128(reinterpret_cast<__m128i*>(tmp), (M)); \
>     (OUT) = tmp[3] + tmp[2] + tmp[1] + tmp[0];              \
>   } while (0)
>@@ -292,8 +295,8 @@ static int start_max(int x, int y) {
>   return (x > y) ? x : y;
> }
> 
>-double CalcSSIM(const uint8* org,
>-                const uint8* rec,
>+double CalcSSIM(const uint8_t* org,
>+                const uint8_t* rec,
>                 const int image_width,
>                 const int image_height) {
>   double SSIM = 0.;
>@@ -328,8 +331,8 @@ double CalcSSIM(const uint8* org,
>       // NOTE: we could use similar method for the left-most pixels too.
>       const int kScratchWidth = 8;
>       const int kScratchStride = kScratchWidth + KERNEL + 1;
>-      uint8 scratch_org[KERNEL_SIZE * kScratchStride] = {0};
>-      uint8 scratch_rec[KERNEL_SIZE * kScratchStride] = {0};
>+      uint8_t scratch_org[KERNEL_SIZE * kScratchStride] = {0};
>+      uint8_t scratch_rec[KERNEL_SIZE * kScratchStride] = {0};
> 
>       for (int k = 0; k < KERNEL_SIZE; ++k) {
>         const int offset =
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/util/ssim.h b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/util/ssim.h
>index 4647f45de14..a855f1d1233 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/util/ssim.h
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/util/ssim.h
>@@ -20,12 +20,12 @@ extern "C" {
> #endif
> 
> #if !defined(INT_TYPES_DEFINED) && !defined(UINT8_TYPE_DEFINED)
>-typedef unsigned char uint8;
>+typedef unsigned char uint8_t;
> #define UINT8_TYPE_DEFINED
> #endif
> 
>-double CalcSSIM(const uint8* org,
>-                const uint8* rec,
>+double CalcSSIM(const uint8_t* org,
>+                const uint8_t* rec,
>                 const int image_width,
>                 const int image_height);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/util/yuvconvert.cc b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/util/yuvconvert.cc
>index bc01d9ff503..27cdfe9e375 100644
>--- a/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/util/yuvconvert.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/libyuv/util/yuvconvert.cc
>@@ -37,7 +37,7 @@ int num_skip_org = 0;   // Number of frames to skip in original.
> int num_frames = 0;     // Number of frames to convert.
> int filter = 1;         // Bilinear filter for scaling.
> 
>-static __inline uint32 Abs(int32 v) {
>+static __inline uint32_t Abs(int32_t v) {
>   return v >= 0 ? v : -v;
> }
> 
>@@ -79,8 +79,9 @@ void PrintHelp(const char* program) {
> }
> 
> void ParseOptions(int argc, const char* argv[]) {
>-  if (argc <= 1)
>+  if (argc <= 1) {
>     PrintHelp(argv[0]);
>+  }
>   for (int c = 1; c < argc; ++c) {
>     if (!strcmp(argv[c], "-v")) {
>       verbose = true;
>@@ -158,11 +159,11 @@ void ParseOptions(int argc, const char* argv[]) {
> static const int kTileX = 32;
> static const int kTileY = 32;
> 
>-static int TileARGBScale(const uint8* src_argb,
>+static int TileARGBScale(const uint8_t* src_argb,
>                          int src_stride_argb,
>                          int src_width,
>                          int src_height,
>-                         uint8* dst_argb,
>+                         uint8_t* dst_argb,
>                          int dst_stride_argb,
>                          int dst_width,
>                          int dst_height,
>@@ -242,9 +243,9 @@ int main(int argc, const char* argv[]) {
>   fseek(file_org, num_skip_org * total_size, SEEK_SET);
> #endif
> 
>-  uint8* const ch_org = new uint8[org_size];
>-  uint8* const ch_dst = new uint8[dst_size];
>-  uint8* const ch_rec = new uint8[total_size];
>+  uint8_t* const ch_org = new uint8_t[org_size];
>+  uint8_t* const ch_dst = new uint8_t[dst_size];
>+  uint8_t* const ch_rec = new uint8_t[total_size];
>   if (ch_org == NULL || ch_rec == NULL) {
>     fprintf(stderr, "No memory available\n");
>     fclose(file_org);
>@@ -265,14 +266,16 @@ int main(int argc, const char* argv[]) {
> 
>   int number_of_frames;
>   for (number_of_frames = 0;; ++number_of_frames) {
>-    if (num_frames && number_of_frames >= num_frames)
>+    if (num_frames && number_of_frames >= num_frames) {
>       break;
>+    }
> 
>     // Load original YUV or ARGB frame.
>     size_t bytes_org =
>-        fread(ch_org, sizeof(uint8), static_cast<size_t>(org_size), file_org);
>-    if (bytes_org < static_cast<size_t>(org_size))
>+        fread(ch_org, sizeof(uint8_t), static_cast<size_t>(org_size), file_org);
>+    if (bytes_org < static_cast<size_t>(org_size)) {
>       break;
>+    }
> 
>     // TODO(fbarchard): Attenuate doesnt need to know dimensions.
>     // ARGB attenuate frame
>@@ -329,16 +332,18 @@ int main(int argc, const char* argv[]) {
>       // Output YUV or ARGB frame.
>       if (rec_is_yuv) {
>         size_t bytes_rec =
>-            fwrite(ch_rec, sizeof(uint8), static_cast<size_t>(total_size),
>+            fwrite(ch_rec, sizeof(uint8_t), static_cast<size_t>(total_size),
>                    file_rec[cur_rec]);
>-        if (bytes_rec < static_cast<size_t>(total_size))
>+        if (bytes_rec < static_cast<size_t>(total_size)) {
>           break;
>+        }
>       } else {
>         size_t bytes_rec =
>-            fwrite(ch_dst, sizeof(uint8), static_cast<size_t>(dst_size),
>+            fwrite(ch_dst, sizeof(uint8_t), static_cast<size_t>(dst_size),
>                    file_rec[cur_rec]);
>-        if (bytes_rec < static_cast<size_t>(dst_size))
>+        if (bytes_rec < static_cast<size_t>(dst_size)) {
>           break;
>+        }
>       }
>       if (verbose) {
>         printf("%5d", number_of_frames);
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/rnnoise/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/third_party/rnnoise/BUILD.gn
>new file mode 100644
>index 00000000000..c669179a5f0
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/rnnoise/BUILD.gn
>@@ -0,0 +1,38 @@
>+# Copyright 2018 The Chromium Authors. All rights reserved.
>+# Use of this source code is governed by a BSD-style license that can be
>+# found in the LICENSE file.
>+
>+import("//testing/test.gni")
>+
>+group("rnnoise") {
>+  deps = [
>+    ":kiss_fft",
>+    ":rnn_vad",
>+  ]
>+}
>+
>+source_set("kiss_fft") {
>+  sources = [
>+    "src/kiss_fft.cc",
>+    "src/kiss_fft.h",
>+  ]
>+}
>+
>+source_set("rnn_vad") {
>+  sources = [
>+    "src/rnn_activations.h",
>+    "src/rnn_vad_weights.cc",
>+    "src/rnn_vad_weights.h",
>+  ]
>+}
>+
>+test("kiss_fft_unittest") {
>+  sources = [
>+    "src/kiss_fft_unittest.cc",
>+  ]
>+  deps = [
>+    ":kiss_fft",
>+    "//testing/gtest",
>+    "//testing/gtest:gtest_main",
>+  ]
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/rnnoise/COPYING b/Source/ThirdParty/libwebrtc/Source/third_party/rnnoise/COPYING
>new file mode 100644
>index 00000000000..01ea4b1a2ac
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/rnnoise/COPYING
>@@ -0,0 +1,31 @@
>+Copyright (c) 2017, Mozilla
>+Copyright (c) 2007-2017, Jean-Marc Valin
>+Copyright (c) 2005-2017, Xiph.Org Foundation
>+Copyright (c) 2003-2004, Mark Borgerding
>+
>+Redistribution and use in source and binary forms, with or without
>+modification, are permitted provided that the following conditions
>+are met:
>+
>+- Redistributions of source code must retain the above copyright
>+notice, this list of conditions and the following disclaimer.
>+
>+- Redistributions in binary form must reproduce the above copyright
>+notice, this list of conditions and the following disclaimer in the
>+documentation and/or other materials provided with the distribution.
>+
>+- Neither the name of the Xiph.Org Foundation nor the names of its
>+contributors may be used to endorse or promote products derived from
>+this software without specific prior written permission.
>+
>+THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
>+``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
>+LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
>+A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION
>+OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
>+SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
>+LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
>+DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
>+THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
>+(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
>+OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/rnnoise/DEPS b/Source/ThirdParty/libwebrtc/Source/third_party/rnnoise/DEPS
>new file mode 100644
>index 00000000000..fca61fcf64b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/rnnoise/DEPS
>@@ -0,0 +1,3 @@
>+include_rules = [
>+  "+testing/gtest/include/gtest",
>+]
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/rnnoise/OWNERS b/Source/ThirdParty/libwebrtc/Source/third_party/rnnoise/OWNERS
>new file mode 100644
>index 00000000000..b3f35ec172b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/rnnoise/OWNERS
>@@ -0,0 +1,2 @@
>+alessiob@chromium.org
>+aleloi@chromium.org
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/rnnoise/README.chromium b/Source/ThirdParty/libwebrtc/Source/third_party/rnnoise/README.chromium
>new file mode 100644
>index 00000000000..8efe059ca6b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/rnnoise/README.chromium
>@@ -0,0 +1,28 @@
>+Name: Recurrent neural network for audio noise reduction
>+Short Name: rnnoise
>+URL: https://github.com/xiph/rnnoise
>+Version: 91ef401
>+Date: Oct 10, 2017
>+Revision:
>+License: BSD 3-Clause
>+License File: COPYING
>+Security Critical: no
>+License Android Compatible:
>+
>+Description:
>+RNNoise is a noise suppression library based on a recurrent neural network.
>+The library is used for speech processing in WebRTC.
>+
>+Local Modifications:
>+* Only retaining COPYING and from src/ the following files:
>+ - kiss_fft.c, kiss_fft.h
>+ - rnn.c
>+ - rnn_data.c
>+ - tansig_table.h
>+* KissFFT: non-floating point parts removed, code clean, from C to C++,
>+  class wrapper added
>+* BUILD targets and KissFFT unit tests added
>+* rnn_vad_weights.h: output layer sizes + weights scaling factor
>+* removing unwanted extern from constants in rnn_vad_weights.h and using
>+  constants to declare array sizes
>+* Add braces around arrays in unit test.
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/rnnoise/src/kiss_fft.cc b/Source/ThirdParty/libwebrtc/Source/third_party/rnnoise/src/kiss_fft.cc
>new file mode 100644
>index 00000000000..27b4900b918
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/rnnoise/src/kiss_fft.cc
>@@ -0,0 +1,443 @@
>+/*Copyright (c) 2003-2004, Mark Borgerding
>+  Lots of modifications by Jean-Marc Valin
>+  Copyright (c) 2005-2007, Xiph.Org Foundation
>+  Copyright (c) 2008,      Xiph.Org Foundation, CSIRO
>+  Copyright (c) 2018,      The WebRTC project authors
>+
>+  All rights reserved.
>+
>+  Redistribution and use in source and binary forms, with or without
>+   modification, are permitted provided that the following conditions are met:
>+
>+    * Redistributions of source code must retain the above copyright notice,
>+       this list of conditions and the following disclaimer.
>+    * Redistributions in binary form must reproduce the above copyright notice,
>+       this list of conditions and the following disclaimer in the
>+       documentation and/or other materials provided with the distribution.
>+
>+  THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
>+  AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
>+  IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
>+  ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
>+  LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
>+  CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
>+  SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
>+  INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
>+  CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
>+  ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
>+  POSSIBILITY OF SUCH DAMAGE.*/
>+
>+#include "third_party/rnnoise/src/kiss_fft.h"
>+
>+#include <cassert>
>+#include <cmath>
>+#include <utility>
>+
>+namespace rnnoise {
>+namespace {
>+
>+void kf_bfly2(std::complex<float>* Fout, int m, int N) {
>+  if (m == 1) {
>+    for (int i = 0; i < N; i++) {
>+      const std::complex<float> t = Fout[1];
>+      Fout[1] = Fout[0] - t;
>+      Fout[0] += t;
>+      Fout += 2;
>+    }
>+  } else {
>+    constexpr float tw = 0.7071067812f;
>+    // We know that m==4 here because the radix-2 is just after a radix-4.
>+    assert(m == 4);
>+    for (int i = 0; i < N; i++) {
>+      std::complex<float>* Fout2 = Fout + 4;
>+      std::complex<float> t = Fout2[0];
>+
>+      *Fout2 = Fout[0] - t;
>+      Fout[0] += t;
>+
>+      t.real((Fout2[1].real() + Fout2[1].imag()) * tw);
>+      t.imag((Fout2[1].imag() - Fout2[1].real()) * tw);
>+      Fout2[1] = Fout[1] - t;
>+      Fout[1] += t;
>+
>+      t.real(Fout2[2].imag());
>+      t.imag(-Fout2[2].real());
>+      Fout2[2] = Fout[2] - t;
>+      Fout[2] += t;
>+
>+      t.real((Fout2[3].imag() - Fout2[3].real()) * tw);
>+      t.imag(-(Fout2[3].imag() + Fout2[3].real()) * tw);
>+      Fout2[3] = Fout[3] - t;
>+      Fout[3] += t;
>+      Fout += 8;
>+    }
>+  }
>+}
>+
>+void kf_bfly4(std::complex<float>* Fout,
>+              const size_t fstride,
>+              const KissFft::KissFftState* st,
>+              int m,
>+              int N,
>+              int mm) {
>+  assert(Fout);
>+  assert(st);
>+  if (m == 1) {
>+    // Degenerate case where all the twiddles are 1.
>+    for (int i = 0; i < N; i++) {
>+      std::complex<float> scratch0 = Fout[0] - Fout[2];
>+      Fout[0] += Fout[2];
>+      std::complex<float> scratch1 = Fout[1] + Fout[3];
>+      Fout[2] = Fout[0] - scratch1;
>+      Fout[0] += scratch1;
>+      scratch1 = Fout[1] - Fout[3];
>+
>+      Fout[1].real(scratch0.real() + scratch1.imag());
>+      Fout[1].imag(scratch0.imag() - scratch1.real());
>+      Fout[3].real(scratch0.real() - scratch1.imag());
>+      Fout[3].imag(scratch0.imag() + scratch1.real());
>+      Fout += 4;
>+    }
>+  } else {
>+    std::complex<float> scratch[6];
>+    const int m2 = 2 * m;
>+    const int m3 = 3 * m;
>+    std::complex<float>* Fout_beg = Fout;
>+    for (int i = 0; i < N; i++) {
>+      Fout = Fout_beg + i * mm;
>+      const std::complex<float>* tw1;
>+      const std::complex<float>* tw2;
>+      const std::complex<float>* tw3;
>+      tw3 = tw2 = tw1 = &st->twiddles[0];
>+      assert(m % 4 == 0);  // |m| is guaranteed to be a multiple of 4.
>+      for (int j = 0; j < m; j++) {
>+        scratch[0] = Fout[m] * *tw1;
>+        scratch[1] = Fout[m2] * *tw2;
>+        scratch[2] = Fout[m3] * *tw3;
>+
>+        scratch[5] = Fout[0] - scratch[1];
>+        Fout[0] += scratch[1];
>+        scratch[3] = scratch[0] + scratch[2];
>+        scratch[4] = scratch[0] - scratch[2];
>+        Fout[m2] = Fout[0] - scratch[3];
>+
>+        tw1 += fstride;
>+        tw2 += fstride * 2;
>+        tw3 += fstride * 3;
>+        Fout[0] += scratch[3];
>+
>+        Fout[m].real(scratch[5].real() + scratch[4].imag());
>+        Fout[m].imag(scratch[5].imag() - scratch[4].real());
>+        Fout[m3].real(scratch[5].real() - scratch[4].imag());
>+        Fout[m3].imag(scratch[5].imag() + scratch[4].real());
>+        ++Fout;
>+      }
>+    }
>+  }
>+}
>+
>+void kf_bfly3(std::complex<float>* Fout,
>+              const size_t fstride,
>+              const KissFft::KissFftState* st,
>+              int m,
>+              int N,
>+              int mm) {
>+  assert(Fout);
>+  assert(st);
>+  const size_t m2 = 2 * m;
>+  const std::complex<float>*tw1, *tw2;
>+  std::complex<float> scratch[5];
>+  std::complex<float> epi3;
>+
>+  std::complex<float>* Fout_beg = Fout;
>+  epi3 = st->twiddles[fstride * m];
>+  for (int i = 0; i < N; i++) {
>+    Fout = Fout_beg + i * mm;
>+    tw1 = tw2 = &st->twiddles[0];
>+    size_t k = m;
>+    do {
>+      scratch[1] = Fout[m] * *tw1;
>+      scratch[2] = Fout[m2] * *tw2;
>+
>+      scratch[3] = scratch[1] + scratch[2];
>+      scratch[0] = scratch[1] - scratch[2];
>+      tw1 += fstride;
>+      tw2 += fstride * 2;
>+
>+      Fout[m] = Fout[0] - 0.5f * scratch[3];
>+
>+      scratch[0] *= epi3.imag();
>+
>+      Fout[0] += scratch[3];
>+
>+      Fout[m2].real(Fout[m].real() + scratch[0].imag());
>+      Fout[m2].imag(Fout[m].imag() - scratch[0].real());
>+
>+      Fout[m].real(Fout[m].real() - scratch[0].imag());
>+      Fout[m].imag(Fout[m].imag() + scratch[0].real());
>+
>+      ++Fout;
>+    } while (--k);
>+  }
>+}
>+
>+void kf_bfly5(std::complex<float>* Fout,
>+              const size_t fstride,
>+              const KissFft::KissFftState* st,
>+              int m,
>+              int N,
>+              int mm) {
>+  assert(Fout);
>+  assert(st);
>+  std::complex<float> scratch[13];
>+  const std::complex<float>* tw;
>+  std::complex<float> ya, yb;
>+  std::complex<float>* const Fout_beg = Fout;
>+
>+  ya = st->twiddles[fstride * m];
>+  yb = st->twiddles[fstride * 2 * m];
>+  tw = &st->twiddles[0];
>+
>+  for (int i = 0; i < N; i++) {
>+    Fout = Fout_beg + i * mm;
>+    std::complex<float>* Fout0 = Fout;
>+    std::complex<float>* Fout1 = Fout0 + m;
>+    std::complex<float>* Fout2 = Fout0 + 2 * m;
>+    std::complex<float>* Fout3 = Fout0 + 3 * m;
>+    std::complex<float>* Fout4 = Fout0 + 4 * m;
>+
>+    // For non-custom modes, m is guaranteed to be a multiple of 4.
>+    for (int u = 0; u < m; ++u) {
>+      scratch[0] = *Fout0;
>+
>+      scratch[1] = *Fout1 * tw[u * fstride];
>+      scratch[2] = *Fout2 * tw[2 * u * fstride];
>+      scratch[3] = *Fout3 * tw[3 * u * fstride];
>+      scratch[4] = *Fout4 * tw[4 * u * fstride];
>+
>+      scratch[7] = scratch[1] + scratch[4];
>+      scratch[10] = scratch[1] - scratch[4];
>+      scratch[8] = scratch[2] + scratch[3];
>+      scratch[9] = scratch[2] - scratch[3];
>+
>+      Fout0->real(Fout0->real() + scratch[7].real() + scratch[8].real());
>+      Fout0->imag(Fout0->imag() + scratch[7].imag() + scratch[8].imag());
>+
>+      scratch[5].real(scratch[0].real() + scratch[7].real() * ya.real() +
>+                      scratch[8].real() * yb.real());
>+      scratch[5].imag(scratch[0].imag() + scratch[7].imag() * ya.real() +
>+                      scratch[8].imag() * yb.real());
>+
>+      scratch[6].real(scratch[10].imag() * ya.imag() +
>+                      scratch[9].imag() * yb.imag());
>+      scratch[6].imag(
>+          -(scratch[10].real() * ya.imag() + scratch[9].real() * yb.imag()));
>+
>+      *Fout1 = scratch[5] - scratch[6];
>+      *Fout4 = scratch[5] + scratch[6];
>+
>+      scratch[11].real(scratch[0].real() + scratch[7].real() * yb.real() +
>+                       scratch[8].real() * ya.real());
>+      scratch[11].imag(scratch[0].imag() + scratch[7].imag() * yb.real() +
>+                       scratch[8].imag() * ya.real());
>+      scratch[12].real(scratch[9].imag() * ya.imag() -
>+                       scratch[10].imag() * yb.imag());
>+      scratch[12].imag(scratch[10].real() * yb.imag() -
>+                       scratch[9].real() * ya.imag());
>+
>+      *Fout2 = scratch[11] + scratch[12];
>+      *Fout3 = scratch[11] - scratch[12];
>+
>+      ++Fout0;
>+      ++Fout1;
>+      ++Fout2;
>+      ++Fout3;
>+      ++Fout4;
>+    }
>+  }
>+}
>+
>+void compute_bitrev_table(int base_index,
>+                          const size_t stride,
>+                          const int16_t* factors,
>+                          const KissFft::KissFftState* st,
>+                          const int16_t* bitrev_table_last,
>+                          int16_t* bitrev_table) {
>+  const int p = *factors++;  // The radix.
>+  const int m = *factors++;  // Stage's fft length/p.
>+  if (m == 1) {
>+    for (int j = 0; j < p; j++) {
>+      assert(bitrev_table <= bitrev_table_last);
>+      *bitrev_table = base_index + j;
>+      bitrev_table += stride;
>+    }
>+  } else {
>+    for (int j = 0; j < p; j++) {
>+      compute_bitrev_table(base_index, stride * p, factors, st,
>+                           bitrev_table_last, bitrev_table);
>+      bitrev_table += stride;
>+      base_index += m;
>+    }
>+  }
>+}
>+
>+// Populates |facbuf| with p1, m1, p2, m2, ... where p[i] * m[i] = m[i-1] and
>+// m0 = n.
>+bool kf_factor(int n, int16_t* facbuf) {
>+  assert(facbuf);
>+  int p = 4;
>+  int stages = 0;
>+  int nbak = n;
>+
>+  // Factor out powers of 4, powers of 2, then any remaining primes.
>+  do {
>+    while (n % p) {
>+      switch (p) {
>+        case 4:
>+          p = 2;
>+          break;
>+        case 2:
>+          p = 3;
>+          break;
>+        default:
>+          p += 2;
>+          break;
>+      }
>+      if (p > 32000 || p * p > n)
>+        p = n;  // No more factors, skip to end.
>+    }
>+    n /= p;
>+    if (p > 5)
>+      return false;
>+    facbuf[2 * stages] = p;
>+    if (p == 2 && stages > 1) {
>+      facbuf[2 * stages] = 4;
>+      facbuf[2] = 2;
>+    }
>+    stages++;
>+  } while (n > 1);
>+  n = nbak;
>+  // Reverse the order to get the radix 4 at the end, so we can use the
>+  // fast degenerate case. It turns out that reversing the order also
>+  // improves the noise behavior.
>+  for (int i = 0; i < stages / 2; i++)
>+    std::swap(facbuf[2 * i], facbuf[2 * (stages - i - 1)]);
>+  for (int i = 0; i < stages; i++) {
>+    n /= facbuf[2 * i];
>+    facbuf[2 * i + 1] = n;
>+  }
>+  return true;
>+}
>+
>+void compute_twiddles(const int nfft, std::complex<float>* twiddles) {
>+  constexpr double pi = 3.14159265358979323846264338327;
>+  assert(twiddles);
>+  for (int i = 0; i < nfft; ++i) {
>+    const double phase = (-2 * pi / nfft) * i;
>+    twiddles[i].real(std::cos(phase));
>+    twiddles[i].imag(std::sin(phase));
>+  }
>+}
>+
>+void fft_impl(const KissFft::KissFftState* st, std::complex<float>* fout) {
>+  assert(st);
>+  assert(fout);
>+  int m2, m;
>+  int p;
>+  int L;
>+  int fstride[KissFft::kMaxFactors];
>+
>+  fstride[0] = 1;
>+  L = 0;
>+  do {
>+    p = st->factors[2 * L];
>+    m = st->factors[2 * L + 1];
>+    assert(static_cast<size_t>(L + 1) < KissFft::kMaxFactors);
>+    fstride[L + 1] = fstride[L] * p;
>+    L++;
>+  } while (m != 1);
>+  m = st->factors[2 * L - 1];
>+  for (int i = L - 1; i >= 0; i--) {
>+    if (i != 0)
>+      m2 = st->factors[2 * i - 1];
>+    else
>+      m2 = 1;
>+    switch (st->factors[2 * i]) {
>+      case 2:
>+        kf_bfly2(fout, m, fstride[i]);
>+        break;
>+      case 4:
>+        kf_bfly4(fout, fstride[i], st, m, fstride[i], m2);
>+        break;
>+      case 3:
>+        kf_bfly3(fout, fstride[i], st, m, fstride[i], m2);
>+        break;
>+      case 5:
>+        kf_bfly5(fout, fstride[i], st, m, fstride[i], m2);
>+        break;
>+      default:
>+        assert(0);
>+        break;
>+    }
>+    m = m2;
>+  }
>+}
>+
>+}  // namespace
>+
>+KissFft::KissFftState::KissFftState(int num_fft_points)
>+    : nfft(num_fft_points), scale(1.f / nfft) {
>+  // Factorize |nfft|.
>+  // TODO(alessiob): Handle kf_factor fails (invalid nfft).
>+  if (!kf_factor(nfft, factors.data()))
>+    assert(0);
>+  // Twiddles.
>+  twiddles.resize(nfft);
>+  compute_twiddles(nfft, twiddles.data());
>+  // Bit-reverse table.
>+  bitrev.resize(nfft);
>+  compute_bitrev_table(0, 1, factors.data(), this, &bitrev.back(),
>+                       bitrev.data());
>+}
>+
>+KissFft::KissFftState::~KissFftState() = default;
>+
>+KissFft::KissFft(const int nfft) : state_(nfft) {}
>+
>+KissFft::~KissFft() = default;
>+
>+void KissFft::ForwardFft(const size_t in_size,
>+                         const std::complex<float>* in,
>+                         const size_t out_size,
>+                         std::complex<float>* out) {
>+  assert(in);
>+  assert(out);
>+  assert(in != out);  // In-place FFT not supported.
>+  assert(state_.nfft == static_cast<int>(in_size));
>+  assert(state_.nfft == static_cast<int>(out_size));
>+  // Bit-reverse the input.
>+  for (int i = 0; i < state_.nfft; i++)
>+    out[state_.bitrev[i]] = state_.scale * in[i];
>+  fft_impl(&state_, out);
>+}
>+
>+void KissFft::ReverseFft(const size_t in_size,
>+                         const std::complex<float>* in,
>+                         const size_t out_size,
>+                         std::complex<float>* out) {
>+  assert(in);
>+  assert(out);
>+  assert(in != out);  // In-place IFFT not supported.
>+  assert(state_.nfft == static_cast<int>(in_size));
>+  assert(state_.nfft == static_cast<int>(out_size));
>+  // Bit-reverse the input.
>+  for (int i = 0; i < state_.nfft; i++)
>+    out[state_.bitrev[i]] = in[i];
>+  for (int i = 0; i < state_.nfft; i++)
>+    out[i].imag(-out[i].imag());
>+  fft_impl(&state_, out);
>+  for (int i = 0; i < state_.nfft; i++)
>+    out[i].imag(-out[i].imag());
>+}
>+
>+}  // namespace rnnoise
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/rnnoise/src/kiss_fft.h b/Source/ThirdParty/libwebrtc/Source/third_party/rnnoise/src/kiss_fft.h
>new file mode 100644
>index 00000000000..b90f77d1ae3
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/rnnoise/src/kiss_fft.h
>@@ -0,0 +1,78 @@
>+/*Copyright (c) 2003-2004, Mark Borgerding
>+  Lots of modifications by Jean-Marc Valin
>+  Copyright (c) 2005-2007, Xiph.Org Foundation
>+  Copyright (c) 2008,      Xiph.Org Foundation, CSIRO
>+  Copyright (c) 2018,      The WebRTC project authors
>+
>+  All rights reserved.
>+
>+  Redistribution and use in source and binary forms, with or without
>+   modification, are permitted provided that the following conditions are met:
>+
>+    * Redistributions of source code must retain the above copyright notice,
>+       this list of conditions and the following disclaimer.
>+    * Redistributions in binary form must reproduce the above copyright notice,
>+       this list of conditions and the following disclaimer in the
>+       documentation and/or other materials provided with the distribution.
>+
>+  THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS "AS IS"
>+  AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE
>+  IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE
>+  ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR CONTRIBUTORS BE
>+  LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR
>+  CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF
>+  SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS
>+  INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN
>+  CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE)
>+  ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE
>+  POSSIBILITY OF SUCH DAMAGE.*/
>+
>+#ifndef THIRD_PARTY_RNNOISE_SRC_KISS_FFT_H_
>+#define THIRD_PARTY_RNNOISE_SRC_KISS_FFT_H_
>+
>+#include <array>
>+#include <complex>
>+#include <vector>
>+
>+namespace rnnoise {
>+
>+class KissFft {
>+ public:
>+  // Example: an FFT of length 128 has 4 factors as far as kissfft is concerned
>+  // (namely, 4*4*4*2).
>+  static const size_t kMaxFactors = 8;
>+
>+  class KissFftState {
>+   public:
>+    KissFftState(int num_fft_points);
>+    KissFftState(const KissFftState&) = delete;
>+    KissFftState& operator=(const KissFftState&) = delete;
>+    ~KissFftState();
>+
>+    const int nfft;
>+    const float scale;
>+    std::array<int16_t, 2 * kMaxFactors> factors;
>+    std::vector<int16_t> bitrev;
>+    std::vector<std::complex<float>> twiddles;
>+  };
>+
>+  explicit KissFft(const int nfft);
>+  KissFft(const KissFft&) = delete;
>+  KissFft& operator=(const KissFft&) = delete;
>+  ~KissFft();
>+  void ForwardFft(const size_t in_size,
>+                  const std::complex<float>* in,
>+                  const size_t out_size,
>+                  std::complex<float>* out);
>+  void ReverseFft(const size_t in_size,
>+                  const std::complex<float>* in,
>+                  const size_t out_size,
>+                  std::complex<float>* out);
>+
>+ private:
>+  KissFftState state_;
>+};
>+
>+}  // namespace rnnoise
>+
>+#endif  // THIRD_PARTY_RNNOISE_SRC_KISS_FFT_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/rnnoise/src/kiss_fft_unittest.cc b/Source/ThirdParty/libwebrtc/Source/third_party/rnnoise/src/kiss_fft_unittest.cc
>new file mode 100644
>index 00000000000..31324453886
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/rnnoise/src/kiss_fft_unittest.cc
>@@ -0,0 +1,145 @@
>+// Copyright 2018 The Chromium Authors. All rights reserved.
>+// Use of this source code is governed by a BSD-style license that can be
>+// found in the LICENSE file.
>+
>+#include <array>
>+#include <cmath>
>+#include <limits>
>+#include <tuple>
>+#include <vector>
>+
>+#include "testing/gtest/include/gtest/gtest.h"
>+#include "third_party/rnnoise/src/kiss_fft.h"
>+
>+namespace rnnoise {
>+namespace test {
>+namespace {
>+
>+const double kPi = std::acos(-1.0);
>+
>+void FillFftInputBuffer(const size_t num_samples,
>+                        const float* samples,
>+                        std::complex<float>* input_buf) {
>+  for (size_t i = 0; i < num_samples; ++i)
>+    input_buf[i].real(samples[i]);
>+}
>+
>+void CheckFftResult(const size_t num_fft_points,
>+                    const float* expected_real,
>+                    const float* expected_imag,
>+                    const std::complex<float>* computed,
>+                    const float tolerance) {
>+  for (size_t i = 0; i < num_fft_points; ++i) {
>+    SCOPED_TRACE(i);
>+    EXPECT_NEAR(expected_real[i], computed[i].real(), tolerance);
>+    EXPECT_NEAR(expected_imag[i], computed[i].imag(), tolerance);
>+  }
>+}
>+
>+}  // namespace
>+
>+class RnnVadTest
>+    : public testing::Test,
>+      public ::testing::WithParamInterface<std::tuple<size_t, float, float>> {};
>+
>+// Check that IFFT(FFT(x)) == x (tolerating round-off errors).
>+TEST_P(RnnVadTest, KissFftForwardReverseCheckIdentity) {
>+  const auto params = GetParam();
>+  const float amplitude = std::get<0>(params);
>+  const size_t num_fft = std::get<1>(params);
>+  const float tolerance = std::get<2>(params);
>+  std::vector<float> samples;
>+  std::vector<float> zeros;
>+  samples.resize(num_fft);
>+  zeros.resize(num_fft);
>+  for (size_t i = 0; i < num_fft; ++i) {
>+    samples[i] = amplitude * std::sin(2.f * kPi * 10 * i / num_fft);
>+    zeros[i] = 0.f;
>+  }
>+
>+  KissFft fft(num_fft);
>+  std::vector<std::complex<float>> fft_buf_1;
>+  fft_buf_1.resize(num_fft);
>+  std::vector<std::complex<float>> fft_buf_2;
>+  fft_buf_2.resize(num_fft);
>+
>+  FillFftInputBuffer(samples.size(), samples.data(), fft_buf_1.data());
>+  {
>+    // TODO(alessiob): Underflow with non power of 2 frame sizes.
>+    // FloatingPointExceptionObserver fpe_observer;
>+
>+    fft.ForwardFft(fft_buf_1.size(), fft_buf_1.data(), fft_buf_2.size(),
>+                   fft_buf_2.data());
>+    fft.ReverseFft(fft_buf_2.size(), fft_buf_2.data(), fft_buf_1.size(),
>+                   fft_buf_1.data());
>+  }
>+  CheckFftResult(samples.size(), samples.data(), zeros.data(), fft_buf_1.data(),
>+                 tolerance);
>+}
>+
>+INSTANTIATE_TEST_CASE_P(FftPoints,
>+                        RnnVadTest,
>+                        ::testing::Values(std::make_tuple(1.f, 240, 3e-7f),
>+                                          std::make_tuple(1.f, 256, 3e-7f),
>+                                          std::make_tuple(1.f, 480, 3e-7f),
>+                                          std::make_tuple(1.f, 512, 3e-7f),
>+                                          std::make_tuple(1.f, 960, 4e-7f),
>+                                          std::make_tuple(1.f, 1024, 3e-7f),
>+                                          std::make_tuple(30.f, 240, 5e-6f),
>+                                          std::make_tuple(30.f, 256, 5e-6f),
>+                                          std::make_tuple(30.f, 480, 6e-6f),
>+                                          std::make_tuple(30.f, 512, 6e-6f),
>+                                          std::make_tuple(30.f, 960, 8e-6f),
>+                                          std::make_tuple(30.f, 1024, 6e-6f)));
>+
>+TEST(RnnVadTest, KissFftBitExactness) {
>+  constexpr std::array<float, 32> samples = {
>+      {0.3524301946163177490234375f,  0.891803801059722900390625f,
>+       0.07706542313098907470703125f, 0.699530780315399169921875f,
>+       0.3789891898632049560546875f,  0.5438187122344970703125f,
>+       0.332781612873077392578125f,   0.449340641498565673828125f,
>+       0.105229437351226806640625f,   0.722373783588409423828125f,
>+       0.13155306875705718994140625f, 0.340857982635498046875f,
>+       0.970204889774322509765625f,   0.53061950206756591796875f,
>+       0.91507828235626220703125f,    0.830274522304534912109375f,
>+       0.74468600749969482421875f,    0.24320767819881439208984375f,
>+       0.743998110294342041015625f,   0.17574800550937652587890625f,
>+       0.1834825575351715087890625f,  0.63317775726318359375f,
>+       0.11414264142513275146484375f, 0.1612723171710968017578125f,
>+       0.80316197872161865234375f,    0.4979794919490814208984375f,
>+       0.554282128810882568359375f,   0.67189347743988037109375f,
>+       0.06660757958889007568359375f, 0.89568817615509033203125f,
>+       0.29327380657196044921875f,    0.3472573757171630859375f}};
>+  constexpr std::array<float, 17> expected_real = {
>+      {0.4813065826892852783203125f, -0.0246877372264862060546875f,
>+       0.04095232486724853515625f, -0.0401695556938648223876953125f,
>+       0.00500857271254062652587890625f, 0.0160773508250713348388671875f,
>+       -0.011385642923414707183837890625f, -0.008461721241474151611328125f,
>+       0.01383177936077117919921875f, 0.0117270611226558685302734375f,
>+       -0.0164460353553295135498046875f, 0.0585579685866832733154296875f,
>+       0.02038039825856685638427734375f, -0.0209107734262943267822265625f,
>+       0.01046995259821414947509765625f, -0.09019653499126434326171875f,
>+       -0.0583711564540863037109375f}};
>+  constexpr std::array<float, 17> expected_imag = {
>+      {0.f, -0.010482530109584331512451171875f, 0.04762755334377288818359375f,
>+       -0.0558677613735198974609375f, 0.007908363826572895050048828125f,
>+       -0.0071932487189769744873046875f, 0.01322011835873126983642578125f,
>+       -0.011227893643081188201904296875f, -0.0400779247283935546875f,
>+       -0.0290451310575008392333984375f, 0.01519204117357730865478515625f,
>+       -0.09711246192455291748046875f, -0.00136523949913680553436279296875f,
>+       0.038602568209171295166015625f, -0.009693108499050140380859375f,
>+       -0.0183933563530445098876953125f, 0.f}};
>+
>+  KissFft fft(32);
>+  std::array<std::complex<float>, 32> fft_buf_in;
>+  std::array<std::complex<float>, 32> fft_buf_out;
>+  FillFftInputBuffer(samples.size(), samples.data(), fft_buf_in.data());
>+  fft.ForwardFft(fft_buf_in.size(), fft_buf_in.data(), fft_buf_out.size(),
>+                 fft_buf_out.data());
>+  CheckFftResult(expected_real.size(), expected_real.data(),
>+                 expected_imag.data(), fft_buf_out.data(),
>+                 std::numeric_limits<float>::min());
>+}
>+
>+}  // namespace test
>+}  // namespace rnnoise
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/rnnoise/src/rnn_activations.h b/Source/ThirdParty/libwebrtc/Source/third_party/rnnoise/src/rnn_activations.h
>new file mode 100644
>index 00000000000..af810319962
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/rnnoise/src/rnn_activations.h
>@@ -0,0 +1,102 @@
>+/* Copyright (c) 2008-2011 Octasic Inc.
>+                 2012-2017 Jean-Marc Valin */
>+/*
>+   Redistribution and use in source and binary forms, with or without
>+   modification, are permitted provided that the following conditions
>+   are met:
>+
>+   - Redistributions of source code must retain the above copyright
>+   notice, this list of conditions and the following disclaimer.
>+
>+   - Redistributions in binary form must reproduce the above copyright
>+   notice, this list of conditions and the following disclaimer in the
>+   documentation and/or other materials provided with the distribution.
>+
>+   THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
>+   ``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
>+   LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
>+   A PARTICULAR PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE FOUNDATION OR
>+   CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
>+   EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
>+   PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR
>+   PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF
>+   LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING
>+   NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS
>+   SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
>+*/
>+
>+#ifndef THIRD_PARTY_RNNOISE_SRC_RNN_ACTIVATIONS_H_
>+#define THIRD_PARTY_RNNOISE_SRC_RNN_ACTIVATIONS_H_
>+
>+#include <cmath>
>+
>+namespace rnnoise {
>+
>+inline float TansigApproximated(float x) {
>+  static constexpr float kTansigTable[201] = {
>+      0.000000f, 0.039979f, 0.079830f, 0.119427f, 0.158649f, 0.197375f,
>+      0.235496f, 0.272905f, 0.309507f, 0.345214f, 0.379949f, 0.413644f,
>+      0.446244f, 0.477700f, 0.507977f, 0.537050f, 0.564900f, 0.591519f,
>+      0.616909f, 0.641077f, 0.664037f, 0.685809f, 0.706419f, 0.725897f,
>+      0.744277f, 0.761594f, 0.777888f, 0.793199f, 0.807569f, 0.821040f,
>+      0.833655f, 0.845456f, 0.856485f, 0.866784f, 0.876393f, 0.885352f,
>+      0.893698f, 0.901468f, 0.908698f, 0.915420f, 0.921669f, 0.927473f,
>+      0.932862f, 0.937863f, 0.942503f, 0.946806f, 0.950795f, 0.954492f,
>+      0.957917f, 0.961090f, 0.964028f, 0.966747f, 0.969265f, 0.971594f,
>+      0.973749f, 0.975743f, 0.977587f, 0.979293f, 0.980869f, 0.982327f,
>+      0.983675f, 0.984921f, 0.986072f, 0.987136f, 0.988119f, 0.989027f,
>+      0.989867f, 0.990642f, 0.991359f, 0.992020f, 0.992631f, 0.993196f,
>+      0.993718f, 0.994199f, 0.994644f, 0.995055f, 0.995434f, 0.995784f,
>+      0.996108f, 0.996407f, 0.996682f, 0.996937f, 0.997172f, 0.997389f,
>+      0.997590f, 0.997775f, 0.997946f, 0.998104f, 0.998249f, 0.998384f,
>+      0.998508f, 0.998623f, 0.998728f, 0.998826f, 0.998916f, 0.999000f,
>+      0.999076f, 0.999147f, 0.999213f, 0.999273f, 0.999329f, 0.999381f,
>+      0.999428f, 0.999472f, 0.999513f, 0.999550f, 0.999585f, 0.999617f,
>+      0.999646f, 0.999673f, 0.999699f, 0.999722f, 0.999743f, 0.999763f,
>+      0.999781f, 0.999798f, 0.999813f, 0.999828f, 0.999841f, 0.999853f,
>+      0.999865f, 0.999875f, 0.999885f, 0.999893f, 0.999902f, 0.999909f,
>+      0.999916f, 0.999923f, 0.999929f, 0.999934f, 0.999939f, 0.999944f,
>+      0.999948f, 0.999952f, 0.999956f, 0.999959f, 0.999962f, 0.999965f,
>+      0.999968f, 0.999970f, 0.999973f, 0.999975f, 0.999977f, 0.999978f,
>+      0.999980f, 0.999982f, 0.999983f, 0.999984f, 0.999986f, 0.999987f,
>+      0.999988f, 0.999989f, 0.999990f, 0.999990f, 0.999991f, 0.999992f,
>+      0.999992f, 0.999993f, 0.999994f, 0.999994f, 0.999994f, 0.999995f,
>+      0.999995f, 0.999996f, 0.999996f, 0.999996f, 0.999997f, 0.999997f,
>+      0.999997f, 0.999997f, 0.999997f, 0.999998f, 0.999998f, 0.999998f,
>+      0.999998f, 0.999998f, 0.999998f, 0.999999f, 0.999999f, 0.999999f,
>+      0.999999f, 0.999999f, 0.999999f, 0.999999f, 0.999999f, 0.999999f,
>+      0.999999f, 0.999999f, 0.999999f, 0.999999f, 1.000000f, 1.000000f,
>+      1.000000f, 1.000000f, 1.000000f, 1.000000f, 1.000000f, 1.000000f,
>+      1.000000f, 1.000000f, 1.000000f,
>+  };
>+
>+  // Tests are reversed to catch NaNs.
>+  if (!(x < 8.f))
>+    return 1.f;
>+  if (!(x > -8.f))
>+    return -1.f;
>+  float sign = 1.f;
>+  if (x < 0.f) {
>+    x = -x;
>+    sign = -1.f;
>+  }
>+  // Look-up.
>+  int i = static_cast<int>(std::floor(0.5f + 25 * x));
>+  float y = kTansigTable[i];
>+  // Map i back to x's scale (undo 25 factor).
>+  x -= 0.04f * i;
>+  y = y + x * (1.f - y * y) * (1.f - y * x);
>+  return sign * y;
>+}
>+
>+inline float SigmoidApproximated(const float x) {
>+  return 0.5f + 0.5f * TansigApproximated(0.5f * x);
>+}
>+
>+inline float RectifiedLinearUnit(const float x) {
>+  return x < 0.f ? 0.f : x;
>+}
>+
>+}  // namespace rnnoise
>+
>+#endif  // THIRD_PARTY_RNNOISE_SRC_RNN_ACTIVATIONS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/rnnoise/src/rnn_vad_weights.cc b/Source/ThirdParty/libwebrtc/Source/third_party/rnnoise/src/rnn_vad_weights.cc
>new file mode 100644
>index 00000000000..40c184b98b0
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/rnnoise/src/rnn_vad_weights.cc
>@@ -0,0 +1,401 @@
>+#include "third_party/rnnoise/src/rnn_vad_weights.h"
>+
>+namespace rnnoise {
>+
>+const int8_t kInputDenseWeights[kInputLayerWeights] = {
>+    -10,  0,    -3,   1,    -8,   -6,   3,    -13,  1,    0,    -3,   -7,
>+    -5,   -3,   6,    -1,   -6,   0,    -6,   -4,   -1,   -2,   1,    1,
>+    -7,   2,    21,   10,   -5,   -20,  24,   23,   37,   8,    -2,   33,
>+    -6,   22,   13,   -2,   50,   8,    13,   1,    -15,  30,   -10,  30,
>+    0,    3,    5,    27,   1,    4,    -3,   41,   56,   35,   -2,   49,
>+    -13,  11,   13,   -2,   -47,  5,    -16,  -60,  -15,  77,   -17,  26,
>+    -3,   14,   -21,  19,   -5,   -19,  -13,  0,    10,   14,   9,    31,
>+    -13,  -41,  -10,  4,    22,   18,   -48,  -6,   -10,  62,   -3,   -18,
>+    -14,  12,   26,   -28,  3,    14,   25,   -13,  -19,  6,    5,    36,
>+    -3,   -65,  -12,  0,    31,   -7,   -9,   101,  -4,   26,   16,   17,
>+    -12,  -12,  14,   -36,  -3,   5,    -15,  21,   2,    30,   -3,   38,
>+    -4,   1,    -6,   7,    -7,   14,   38,   -22,  -30,  -3,   -7,   3,
>+    -39,  -70,  -126, 25,   34,   94,   -67,  -22,  -33,  83,   -47,  -118,
>+    4,    70,   33,   25,   62,   -128, -76,  -118, -113, 49,   -12,  -100,
>+    -18,  -114, -33,  43,   32,   61,   40,   -9,   -106, 2,    36,   -100,
>+    -40,  -5,   20,   -75,  61,   -51,  -9,   126,  -27,  -52,  5,    -24,
>+    -21,  -126, -114, -12,  15,   106,  -2,   73,   -125, 50,   13,   -120,
>+    35,   35,   4,    -61,  29,   -124, 6,    -53,  -69,  -125, 64,   -89,
>+    36,   -107, -103, -7,   27,   121,  69,   77,   -35,  35,   95,   -125,
>+    -49,  97,   -45,  -43,  -23,  23,   -28,  -65,  -118, 2,    8,    -126,
>+    27,   -97,  92,   5,    55,   82,   17,   -57,  -115, 37,   8,    -106,
>+    -46,  41,   -2,   21,   -44,  8,    -73,  -58,  -39,  34,   89,   -95,
>+    95,   -117, 120,  -58,  31,   123,  1,    -32,  -109, -110, 60,   -120,
>+    -43,  -74,  5,    91,   26,   21,   114,  82,   -83,  -126, 123,  22,
>+    -16,  -67,  25,   -83,  46,   48,   -34,  -121, -124, -63,  -35,  -9,
>+    31,   82,   123,  6,    -3,   117,  93,   -2,   -13,  -36,  124,  -112,
>+    -6,   -102, -5,   -33,  -15,  44,   -69,  -127, -23,  -40,  -34,  -85,
>+    68,   83,   -1,   40,   8,    84,   118,  -58,  -55,  -102, 123,  -55,
>+    -14,  -123, 44,   -63,  -14,  21,   35,   16,   24,   -126, -13,  -114,
>+    35,   20,   -36,  61,   -9,   97,   34,   19,   -32,  -109, 76,   -104,
>+    99,   -119, 45,   -125, -51,  -28,  -8,   -69,  -8,   125,  -45,  -93,
>+    113,  103,  -41,  -82,  52,   7,    126,  0,    -40,  104,  55,   -58,
>+    17,   -124, -93,  -58,  8,    -45,  1,    56,   -123, 108,  -47,  -23,
>+    115,  127,  17,   -68,  -13,  116,  -82,  -44,  45,   67,   -120, -101,
>+    -15,  -125, 120,  -113, 17,   -48,  -73,  126,  -64,  -86,  -118, -19,
>+    112,  -1,   -66,  -27,  -62,  121,  -86,  -58,  50,   89,   -38,  -75,
>+    95,   -111, 12,   -113, 2,    -68,  2,    -94,  -121, 91,   -5,   0,
>+    79,   43,   -7,   -18,  79,   35,   -38,  47,   1,    -45,  83,   -50,
>+    102,  32,   55,   -96,  15,   -122, -69,  45,   -27,  91,   -62,  -30,
>+    46,   -95,  22,   -72,  -97,  -1,   14,   -122, 28,   127,  61,   -126,
>+    121,  9,    68,   -120, 49,   -60,  90,   3,    43,   68,   54,   34,
>+    -10,  28,   21,   -24,  -54,  22,   -113, -12,  82,   -2,   -17,  -9,
>+    127,  8,    116,  -92,  0,    -70,  -33,  123,  66,   116,  -74,  -4,
>+    74,   -72,  -22,  -47,  1,    -83,  -60,  -124, 1,    122,  -57,  -43,
>+    49,   40,   -126, -128, -8,   -29,  28,   -24,  -123, -121, -70,  -93,
>+    -37,  -126, 11,   -125, -37,  11,   -31,  -51,  -124, 116,  -128, 8,
>+    -25,  109,  75,   -12,  7,    8,    10,   117,  124,  -128, -128, 29,
>+    -26,  101,  21,   -128, 87,   8,    -39,  23,   -128, 127,  -127, 74,
>+    -55,  74,   112,  127,  4,    55,   44,   -92,  123,  34,   -93,  47,
>+    -21,  -92,  17,   49,   -121, 92,   7,    -126, -125, 124,  -74,  3,
>+    -59,  18,   -91,  3,    -9,   9,    56,   116,  7,    -29,  33,   87,
>+    -21,  -128, -13,  57,   74,   9,    -29,  -61,  -97,  -21,  -95,  -12,
>+    -114, 16,   82,   125,  -7,   10,   -24,  9,    77,   -128, -102, -25,
>+    3,    -126, 10,   13,   -18,  51,   26,   127,  -79,  35,   51,   12,
>+    -50,  -24,  1,    -7,   22,   81,   65,   120,  -30,  -38,  85,   122,
>+    -4,   -106, -11,  27,   53,   41,   8,    -104, -66,  -38,  -124, 10,
>+    12,   76,   117,  -109, 9,    11,   2,    -18,  3,    113,  -16,  -79,
>+    -39,  -123, -20,  -128, 2,    13,   -33,  -58,  10,   84,   -104, 13,
>+    64,   109,  1,    54,   -12,  28,   24,   63,   -126, 118,  -82,  46,
>+    -12,  -15,  14,   -43,  60,   22,   -32,  -19,  -46,  91,   -107, 24,
>+    -94,  26,   -47,  125,  6,    58,   -15,  -75,  -26,  -38,  -35,  103,
>+    -16,  -17,  -13,  63,   -2,   45,   -45,  -73,  -23,  70,   -87,  51,
>+    -17,  53,   76,   14,   -18,  -31,  -14,  103,  8,    21,   -28,  -33,
>+    -20,  -47,  6,    39,   40,   -30,  7,    -76,  55,   31,   -20,  -21,
>+    -59,  1,    25,   -11,  17,   5,    -13,  -39,  0,    -76,  50,   -33,
>+    -29,  -50,  -16,  -11,  -12,  -1,   -46,  40,   -10,  65,   -19,  21,
>+    -41,  -32,  -83,  -19,  -4,   49,   -60,  118,  -24,  -46,  9,    102,
>+    -20,  8,    -19,  25,   31,   -3,   -37,  0,    25,   7,    29,   2,
>+    -39,  127,  -64,  -20,  64,   115,  -30,  36,   100,  35,   122,  127,
>+    127,  -127, 127,  -127, 19,   127,  -89,  -79,  -32,  39,   -127, 125,
>+    -80,  126,  -127, 26,   8,    98,   -8,   -57,  -90,  -50,  126,  61,
>+    127,  -126, 40,   -106, -68,  104,  -125, -119, 11,   10,   -127, 66,
>+    -56,  -12,  -126, -104, 27,   75,   38,   -124, -126, -125, 84,   -123,
>+    -45,  -114, -128, 127,  103,  -101, -124, 127,  -11,  -23,  -123, 92,
>+    -123, 24,   126,  41,   -2,   -39,  -27,  -94,  40,   -112, -48,  127,
>+    58,   14,   38,   -75,  -64,  73,   117,  100,  -119, -11,  6,    32,
>+    -126, -14,  35,   121,  -10,  54,   -60,  89,   -3,   69,   -25,  -20,
>+    43,   -86,  -34,  24,   27,   7,    -81,  -99,  -23,  -16,  -26,  13,
>+    35,   -97,  80,   -29,  -13,  -121, -12,  -65,  -94,  70,   -89,  -126,
>+    -95,  88,   33,   96,   29,   -90,  69,   114,  -78,  65,   90,   -47,
>+    -47,  89,   1,    -12,  3,    8,    30,   5,    2,    -30,  -1,   6,
>+    -7,   10,   -4,   46,   -27,  -40,  22,   -6,   -17,  45,   24,   -9,
>+    23,   -14,  -63,  -26,  -12,  -57,  27,   25,   55,   -76,  -47,  21,
>+    34,   33,   26,   17,   14,   6,    9,    26,   25,   -25,  -25,  -18};
>+
>+const int8_t kInputDenseBias[kInputLayerOutputSize] = {
>+    38,  -6,  127,  127, 127,  -43, -127, 78,  127, 5,   127, 123,
>+    127, 127, -128, -76, -126, 28,  127,  125, -30, 127, -89, -20};
>+
>+const int8_t kHiddenGruWeights[kHiddenLayerWeights] = {
>+    -124, 23,   -123, -33,  -95,  -4,   8,    -84,  4,    101,  -119, 116,
>+    -4,   123,  103,  -51,  29,   -124, -114, -49,  31,   9,    75,   -128,
>+    0,    -49,  37,   -50,  46,   -21,  -63,  -104, 54,   82,   33,   21,
>+    70,   127,  -9,   -79,  -39,  -23,  -127, 107,  122,  -96,  -46,  -18,
>+    -39,  13,   -28,  -48,  14,   56,   -52,  49,   -1,   -121, 25,   -18,
>+    -36,  -52,  -57,  -30,  54,   -124, -26,  -47,  10,   39,   12,   2,
>+    9,    -127, -128, 102,  21,   11,   -64,  -71,  89,   -113, -111, 54,
>+    31,   94,   121,  -40,  30,   40,   -109, 73,   -9,   108,  -92,  2,
>+    -127, 116,  127,  127,  -122, 95,   127,  -37,  -127, 28,   89,   10,
>+    24,   -104, -62,  -67,  -14,  38,   14,   -71,  22,   -41,  20,   -50,
>+    39,   63,   86,   127,  -18,  79,   4,    -51,  2,    33,   117,  -113,
>+    -78,  56,   -91,  37,   34,   -45,  -44,  -22,  21,   -16,  56,   30,
>+    -84,  -79,  38,   -74,  127,  9,    -25,  2,    82,   61,   25,   -26,
>+    26,   11,   117,  -65,  12,   -58,  42,   -62,  -93,  11,   11,   124,
>+    -123, 80,   -125, 11,   -90,  42,   94,   4,    -109, -1,   85,   -52,
>+    45,   -26,  -27,  77,   -5,   30,   90,   0,    95,   -7,   53,   29,
>+    -82,  22,   -9,   74,   2,    -12,  -73,  114,  97,   -64,  122,  -77,
>+    43,   91,   86,   126,  106,  72,   90,   -43,  46,   96,   -51,  21,
>+    22,   68,   22,   41,   79,   75,   -46,  -105, 23,   -116, 127,  -123,
>+    102,  57,   85,   10,   -29,  34,   125,  126,  124,  81,   -15,  54,
>+    96,   -128, 39,   -124, 103,  74,   126,  127,  -50,  -71,  -122, -64,
>+    93,   -75,  71,   105,  122,  123,  126,  122,  -127, 33,   -63,  -74,
>+    124,  -71,  33,   41,   -56,  19,   6,    65,   41,   90,   -116, -3,
>+    -46,  75,   -13,  98,   -74,  -42,  74,   -95,  -96,  81,   24,   32,
>+    -19,  -123, 74,   55,   109,  115,  0,    32,   33,   12,   -20,  9,
>+    127,  127,  -61,  79,   -48,  -54,  -49,  101,  -9,   27,   -106, 74,
>+    119,  77,   87,   -126, -24,  127,  124,  31,   34,   127,  40,   3,
>+    -90,  127,  23,   57,   -53,  127,  -69,  -88,  -33,  127,  19,   -46,
>+    -9,   -125, 13,   -126, -113, 127,  -41,  46,   106,  -62,  3,    -10,
>+    111,  49,   -34,  -24,  -20,  -112, 11,   101,  -50,  -34,  50,   65,
>+    -64,  -106, 70,   -48,  60,   9,    -122, -45,  15,   -112, -26,  -4,
>+    1,    39,   23,   58,   -45,  -80,  127,  82,   58,   30,   -94,  -119,
>+    51,   -89,  95,   -107, 30,   127,  125,  58,   -52,  -42,  -38,  -20,
>+    -122, 115,  39,   -26,  5,    73,   13,   -39,  43,   -23,  -20,  -125,
>+    23,   35,   53,   -61,  -66,  72,   -20,  33,   8,    35,   4,    7,
>+    18,   19,   16,   -45,  -50,  -71,  31,   -29,  -41,  -27,  10,   14,
>+    27,   9,    -23,  98,   6,    -94,  92,   127,  -114, 59,   -26,  -100,
>+    -62,  -127, -17,  -85,  -60,  126,  -42,  -6,   33,   -120, -26,  -126,
>+    -127, -35,  -114, -31,  25,   -126, -100, -126, -64,  -46,  -31,  30,
>+    25,   -74,  -111, -97,  -81,  -104, -114, -19,  -9,   -116, -69,  22,
>+    30,   59,   8,    -51,  16,   -97,  18,   -4,   -89,  80,   -50,  3,
>+    36,   -67,  56,   69,   -26,  107,  -10,  58,   -28,  -4,   -57,  -72,
>+    -111, 0,    -75,  -119, 14,   -75,  -49,  -66,  -49,  8,    -121, 22,
>+    -54,  121,  30,   54,   -26,  -126, -123, 56,   5,    48,   21,   -127,
>+    -11,  23,   25,   -82,  6,    -25,  119,  78,   4,    -104, 27,   61,
>+    -48,  37,   -13,  -52,  50,   -50,  44,   -1,   -22,  -43,  -59,  -78,
>+    -67,  -32,  -26,  9,    -3,   40,   16,   19,   3,    -9,   20,   -6,
>+    -37,  28,   39,   17,   -19,  -10,  1,    6,    -59,  74,   47,   3,
>+    -119, 0,    -128, -107, -25,  -22,  -69,  -23,  -111, -42,  -93,  -120,
>+    90,   -85,  -54,  -118, 76,   -79,  124,  101,  -77,  -75,  -17,  -71,
>+    -114, 68,   55,   79,   -1,   -123, -20,  127,  -65,  -123, -128, -87,
>+    123,  9,    -115, -14,  7,    -4,   127,  -79,  -115, 125,  -28,  89,
>+    -83,  49,   89,   119,  -69,  -5,   12,   -49,  60,   57,   -24,  -99,
>+    -110, 76,   -83,  125,  73,   81,   11,   8,    -45,  1,    83,   13,
>+    -70,  -2,   97,   112,  -97,  53,   -9,   -94,  124,  44,   -49,  -24,
>+    52,   76,   -110, -70,  -114, -12,  72,   -4,   -114, 43,   -43,  81,
>+    102,  -84,  -27,  62,   -40,  52,   58,   124,  -35,  -51,  -123, -43,
>+    56,   -75,  -34,  -35,  -106, 93,   -43,  14,   -16,  46,   62,   -97,
>+    21,   30,   -53,  21,   -11,  -33,  -20,  -95,  4,    -126, 12,   45,
>+    20,   108,  85,   11,   20,   -40,  99,   4,    -25,  -18,  -23,  -12,
>+    -126, -55,  -20,  -44,  -51,  91,   -127, 127,  -44,  7,    127,  78,
>+    38,   125,  -6,   -94,  -103, 73,   126,  -126, 18,   59,   -46,  106,
>+    76,   116,  -31,  75,   -4,   92,   102,  32,   -31,  73,   42,   -21,
>+    -28,  57,   127,  -8,   -107, 115,  124,  -94,  -4,   -128, 29,   -57,
>+    70,   -82,  50,   -13,  -44,  38,   67,   -93,  6,    -39,  -46,  56,
>+    68,   27,   61,   26,   18,   -72,  127,  22,   18,   -31,  127,  61,
>+    -65,  -38,  1,    -67,  -1,   8,    -73,  46,   -116, -94,  58,   -49,
>+    71,   -40,  -63,  -82,  -20,  -60,  93,   76,   69,   -106, 34,   -31,
>+    4,    -25,  107,  -18,  45,   4,    -61,  126,  54,   -126, -125, 41,
>+    19,   44,   32,   -98,  125,  -24,  125,  -96,  -125, 15,   87,   -4,
>+    -90,  18,   -40,  28,   -69,  67,   22,   41,   39,   7,    -48,  -44,
>+    12,   69,   -13,  2,    44,   -38,  111,  -7,   -126, -22,  -9,   74,
>+    -128, -36,  -7,   -123, -15,  -79,  -91,  -37,  -127, -122, 104,  30,
>+    7,    98,   -37,  111,  -116, -47,  127,  -45,  118,  -111, -123, -120,
>+    -77,  -64,  -125, 124,  77,   111,  77,   18,   -113, 117,  -9,   67,
>+    -77,  126,  49,   -20,  -124, 39,   41,   -124, -34,  114,  -87,  -126,
>+    98,   -20,  59,   -17,  -24,  125,  107,  54,   35,   33,   -44,  12,
>+    -29,  125,  -71,  -28,  -63,  -114, 28,   -17,  121,  -36,  127,  89,
>+    -122, -49,  -18,  -48,  17,   24,   19,   -64,  -128, 13,   86,   45,
>+    13,   -49,  55,   84,   48,   80,   -39,  99,   -127, 70,   -33,  30,
>+    50,   126,  -65,  -117, -13,  -20,  -24,  127,  115,  -72,  -104, 63,
>+    126,  -42,  57,   17,   46,   21,   119,  110,  -100, -60,  -112, 62,
>+    -33,  28,   26,   -22,  -60,  -33,  -54,  78,   25,   32,   -114, 86,
>+    44,   26,   43,   76,   121,  19,   97,   -2,   -3,   -73,  -68,  6,
>+    -116, 6,    -43,  -97,  46,   -128, -120, -31,  -119, -29,  16,   16,
>+    -126, -128, -126, -46,  -9,   -3,   92,   -31,  -76,  -126, -3,   -107,
>+    -12,  -23,  -69,  5,    51,   27,   -42,  23,   -70,  -128, -29,  22,
>+    29,   -126, -55,  50,   -71,  -3,   127,  44,   -27,  -70,  -63,  -66,
>+    -70,  104,  86,   115,  29,   -92,  41,   -90,  44,   -11,  -28,  20,
>+    -11,  -63,  -16,  43,   31,   17,   -73,  -31,  -1,   -17,  -11,  -39,
>+    56,   18,   124,  72,   -14,  28,   69,   -121, -125, 34,   127,  63,
>+    86,   -80,  -126, -125, -124, -47,  124,  77,   124,  -19,  23,   -7,
>+    -50,  96,   -128, -93,  102,  -53,  -36,  -87,  119,  -125, 92,   -126,
>+    118,  102,  72,   -2,   125,  10,   97,   124,  -125, 125,  71,   -20,
>+    -47,  -116, -121, -4,   -9,   -32,  79,   -124, -36,  33,   -128, -74,
>+    125,  23,   127,  -29,  -115, -32,  124,  -89,  32,   -107, 43,   -17,
>+    24,   24,   18,   29,   -13,  -15,  -36,  62,   -91,  4,    -41,  95,
>+    28,   -23,  6,    46,   84,   66,   77,   68,   -70,  -1,   -23,  -6,
>+    65,   70,   -21,  9,    77,   -12,  2,    -118, 4,    9,    -108, 84,
>+    52,   2,    52,   13,   -10,  58,   -110, 18,   66,   -95,  -23,  70,
>+    31,   -3,   56,   56,   -3,   -7,   1,    -27,  -48,  -61,  41,   -4,
>+    10,   -62,  32,   -7,   -24,  9,    -48,  -60,  -4,   79,   -20,  -38,
>+    -76,  68,   -49,  -97,  0,    -15,  5,    -100, -49,  -95,  -99,  -115,
>+    -9,   -40,  10,   104,  13,   56,   127,  -27,  -109, -94,  -118, -102,
>+    -44,  -85,  52,   127,  -4,   14,   62,   121,  -122, -26,  -79,  -42,
>+    -34,  1,    25,   -38,  -79,  -58,  -31,  -31,  -90,  -30,  -123, 32,
>+    -56,  125,  66,   124,  -1,   3,    91,   -103, -7,   23,   78,   -18,
>+    9,    69,   -69,  76,   -38,  -33,  -2,   -98,  18,   106,  84,   55,
>+    87,   -47,  35,   -124, 64,   41,   -14,  46,   25,   -2,   120,  -21,
>+    82,   19,   -79,  -37,  -3,   -8,   -16,  21,   19,   -5,   -28,  -112,
>+    39,   -6,   -30,  53,   -69,  53,   46,   127,  123,  78,   20,   28,
>+    -7,   73,   72,   17,   -40,  41,   111,  57,   32,   -95,  29,   28,
>+    -39,  -65,  54,   -20,  -63,  29,   -67,  3,    44,   -57,  -47,  11,
>+    61,   -22,  -44,  61,   48,   -100, 20,   125,  96,   -24,  -16,  3,
>+    -69,  -126, 74,   -125, 9,    45,   -67,  -123, -59,  -72,  118,  69,
>+    45,   50,   -57,  67,   13,   -66,  -106, 47,   62,   22,   -1,   -22,
>+    -25,  -40,  -125, 3,    125,  32,   102,  -56,  -25,  -75,  -30,  122,
>+    60,   -13,  36,   -73,  7,    -84,  124,  40,   -118, 17,   -87,  -118,
>+    -8,   3,    -27,  111,  -40,  40,   -51,  127,  125,  -45,  -30,  -54,
>+    46,   80,   -1,   -30,  101,  -17,  18,   26,   54,   7,    -12,  1,
>+    -127, 123,  -122, -27,  -75,  64,   10,   25,   -15,  -44,  127,  -127,
>+    5,    -84,  -81,  -7,   19,   -26,  126,  15,   116,  -126, 14,   -76,
>+    44,   62,   -110, -124, 125,  -29,  -87,  -3,   -69,  82,   90,   57,
>+    -123, 123,  100,  -19,  -51,  -32,  69,   37,   -57,  -128, -124, -72,
>+    -13,  51,   -7,   -45,  -73,  5,    99,   -26,  -117, -96,  -109, 4,
>+    -31,  -12,  0,    31,   -42,  -27,  12,   -81,  118,  39,   83,   14,
>+    41,   -126, 107,  -82,  94,   -116, -122, -47,  -109, -84,  -128, -35,
>+    -56,  66,   8,    -65,  19,   42,   -46,  -72,  -109, 41,   43,   -127,
>+    -113, 58,   127,  42,   -75,  -1,   65,   117,  -55,  -113, -123, 124,
>+    43,   -96,  -115, -19,  68,   15,   94,   3,    75,   0,    34,   9,
>+    42,   110,  -48,  92,   -76,  99,   -17,  27,   32,   13,   125,  50,
>+    -17,  56,   4,    53,   34,   -8,   99,   80,   -126, -21,  -65,  -11,
>+    -46,  44,   -81,  -3,   -121, 123,  66,   -81,  -84,  119,  127,  84,
>+    105,  45,   -66,  -42,  -23,  32,   -25,  12,   111,  127,  88,   125,
>+    30,   24,   -127, -9,   -54,  127,  -116, -119, 88,   70,   94,   -120,
>+    35,   -93,  15,   22,   -21,  25,   -110, -123, -45,  8,    -109, 125,
>+    -122, -86,  -126, 8,    -14,  -120, -45,  -45,  69,   -125, -122, 6,
>+    81,   86,   125,  95,   54,   77,   54,   -123, 126,  -85,  -117, 56,
>+    11,   0,    -61,  -91,  -12,  -2,   -113, -3,   -15,  -122, -63,  -91,
>+    10,   84,   -111, 125,  93,   21,   62,   -78,  -116, 13,   -57,  28,
>+    -124, 126,  110,  12,   15,   95,   15,   -19,  -125, -97,  52,   -7,
>+    101,  9,    20,   -125, -26,  -56,  72,   77,   12,   -126, 22,   -29,
>+    47,   62,   95,   112,  69,   32,   97,   -83,  -8,   -5,   67,   -63,
>+    -123, 79,   59,   0,    -6,   -17,  4,    -111, -52,  27,   65,   0};
>+
>+const int8_t kHiddenGruRecurrentWeights[kHiddenLayerWeights] = {
>+    65,   83,   35,   56,   24,   -34,  -28,  -2,   125,  19,   42,   -9,
>+    124,  -53,  24,   -87,  11,   35,   -81,  -35,  -125, -31,  123,  -21,
>+    33,   -91,  113,  -93,  45,   -6,   53,   38,   -92,  8,    -27,  87,
>+    4,    43,   43,   10,   -128, -128, -46,  127,  -38,  -45,  25,   -87,
>+    19,   5,    52,   -96,  -23,  -29,  121,  -126, -24,  -20,  -2,   69,
>+    -50,  6,    71,   -81,  -125, 90,   -94,  1,    -38,  36,   89,   17,
>+    -60,  71,   -48,  18,   -15,  44,   -18,  59,   11,   114,  -51,  32,
>+    110,  1,    4,    109,  -24,  127,  27,   60,   88,   24,   45,   -59,
>+    75,   -36,  8,    57,   -32,  -25,  13,   126,  -89,  -61,  -76,  127,
>+    18,   -62,  -68,  23,   -113, 5,    126,  43,   -88,  26,   -78,  18,
>+    75,   21,   9,    -74,  20,   41,   126,  -118, -15,  9,    116,  126,
>+    -127, 34,   -6,   126,  -128, -53,  -54,  -55,  -121, 70,   127,  -12,
>+    -68,  82,   -25,  104,  -126, 126,  -21,  -26,  124,  -75,  -127, -120,
>+    13,   61,   -64,  -108, -63,  -65,  -44,  -35,  -61,  -39,  109,  -74,
>+    113,  -3,   108,  -30,  125,  120,  39,   125,  -128, -95,  -99,  111,
>+    9,    25,   114,  -75,  -92,  -54,  -12,  -32,  -38,  10,   31,   10,
>+    63,   51,   40,   -99,  74,   4,    50,   -128, -36,  -35,  -11,  -28,
>+    -126, -7,   66,   -58,  -126, -22,  -83,  -61,  -127, 49,   126,  -8,
>+    7,    62,   36,   -11,  -32,  -44,  63,   116,  41,   65,   -127, 126,
>+    63,   -30,  -96,  74,   -92,  127,  38,   -18,  -128, 68,   -5,   101,
>+    -4,   85,   58,   79,   0,    -58,  8,    119,  -70,  -1,   -79,  -68,
>+    114,  -28,  -90,  -6,   -112, 2,    127,  -8,   10,   55,   -59,  -126,
>+    127,  125,  80,   72,   35,   -54,  95,   -124, -124, 79,   23,   -46,
>+    -61,  -127, -100, 99,   -77,  8,    -87,  5,    -2,   49,   85,   7,
>+    -71,  82,   53,   -41,  22,   -22,  -93,  -103, 6,    52,   -56,  14,
>+    -8,   -111, 85,   16,   54,   32,   -118, -24,  61,   -53,  96,   -70,
>+    -5,   -17,  -67,  -84,  -7,   -82,  -107, -96,  21,   -83,  -58,  50,
>+    12,   -126, -1,   -28,  34,   -126, 115,  17,   91,   1,    -127, 72,
>+    11,   126,  -81,  6,    96,   -8,   77,   15,   -6,   63,   -27,  20,
>+    -123, -109, 85,   -79,  -17,  126,  -92,  2,    -61,  20,   14,   17,
>+    121,  123,  30,   57,   120,  127,  57,   42,   117,  98,   67,   39,
>+    -20,  -70,  100,  7,    125,  122,  40,   16,   -79,  125,  83,   41,
>+    -106, -57,  24,   55,   27,   -66,  -111, -44,  -7,   -43,  -66,  121,
>+    42,   -128, -45,  35,   15,   -127, 34,   -35,  -34,  -40,  -18,  -6,
>+    63,   111,  31,   116,  127,  19,   24,   -71,  -39,  34,   11,   19,
>+    -40,  27,   12,   106,  -10,  56,   -82,  -106, -2,   -50,  -52,  114,
>+    -126, -34,  -43,  -68,  10,   76,   57,   -118, -128, 37,   -104, 76,
>+    125,  3,    -76,  127,  -29,  84,   -94,  -15,  55,   125,  79,   127,
>+    -57,  -125, 104,  -68,  126,  126,  -77,  51,   45,   33,   -109, 115,
>+    -11,  1,    95,   -121, -5,   -9,   -126, -114, 39,   68,   -126, -107,
>+    -51,  -42,  24,   -8,   51,   -27,  -43,  66,   -45,  62,   -98,  -109,
>+    69,   67,   0,    -125, -128, 49,   31,   126,  -122, 2,    -55,  -67,
>+    -126, -70,  -128, -125, -77,  25,   16,   -8,   -102, 11,   -75,  82,
>+    38,   -5,   5,    19,   34,   47,   -127, -93,  21,   24,   -97,  -18,
>+    31,   39,   34,   -20,  22,   123,  7,    -77,  -81,  -46,  -9,   1,
>+    23,   39,   -127, -43,  -8,   -50,  10,   -21,  59,   -9,   -4,   -13,
>+    -27,  44,   127,  52,   -47,  70,   -43,  52,   101,  -49,  27,   45,
>+    49,   33,   -125, 55,   114,  20,   -1,   76,   -24,  -96,  105,  24,
>+    126,  75,   -21,  -105, 13,   -42,  40,   126,  -30,  -39,  -95,  125,
>+    -63,  11,   6,    125,  125,  -14,  5,    42,   -61,  -4,   49,   88,
>+    6,    -107, -28,  19,   -29,  47,   126,  6,    -46,  -89,  -18,  91,
>+    -20,  -6,   118,  -21,  -22,  39,   115,  11,   -42,  54,   73,   -55,
>+    -77,  62,   -27,  -59,  -99,  -12,  -127, -40,  56,   -3,   -124, -91,
>+    71,   -111, 6,    -19,  82,   -24,  -35,  102,  -42,  7,    -126, -126,
>+    -125, 18,   98,   -52,  127,  105,  -52,  40,   -83,  126,  -122, 109,
>+    5,    127,  48,   6,    5,    -125, 100,  -16,  29,   85,   -89,  8,
>+    4,    41,   62,   -127, 62,   122,  85,   122,  -107, 8,    -125, 93,
>+    -127, 127,  102,  19,   19,   -66,  41,   -42,  114,  127,  -48,  -117,
>+    -29,  -6,   -73,  -102, -3,   -19,  0,    88,   42,   87,   -117, -20,
>+    2,    122,  28,   63,   71,   66,   120,  93,   124,  -43,  49,   103,
>+    31,   90,   -91,  -22,  -126, 26,   -24,  -21,  51,   -126, 87,   -103,
>+    -69,  -10,  -66,  -23,  20,   97,   36,   25,   -127, 30,   -20,  -63,
>+    30,   51,   -116, 23,   40,   -39,  36,   -83,  -77,  -25,  -50,  110,
>+    14,   13,   -109, 125,  -65,  -55,  -87,  124,  -126, -32,  -72,  -108,
>+    127,  127,  -125, -124, 61,   121,  102,  -128, -127, 16,   100,  127,
>+    -124, -68,  72,   -93,  -128, 43,   -93,  -19,  -125, -97,  -113, -33,
>+    83,   127,  -44,  127,  -75,  127,  16,   44,   50,   -122, 23,   118,
>+    46,   19,   26,   -128, 10,   4,    99,   -14,  -82,  -13,  30,   125,
>+    57,   65,   60,   -71,  35,   98,   28,   7,    1,    43,   89,   70,
>+    75,   121,  -59,  82,   -126, -53,  -16,  -116, -65,  52,   -52,  0,
>+    80,   35,   45,   -61,  46,   8,    107,  27,   -26,  -118, 90,   57,
>+    -10,  7,    -15,  0,    -39,  -4,   12,   29,   -1,   116,  84,   79,
>+    119,  125,  -59,  28,   -6,   -25,  -43,  2,    90,   79,   67,   103,
>+    -82,  2,    -6,   125,  19,   73,   0,    -105, 112,  -17,  104,  107,
>+    124,  106,  19,   56,   -44,  55,   -112, 6,    -39,  -83,  126,  -93,
>+    -98,  57,   -120, -23,  -38,  2,    -31,  -48,  106,  127,  127,  69,
>+    16,   110,  71,   104,  62,   -12,  -22,  42,   -37,  -94,  34,   -1,
>+    -32,  -12,  -124, -47,  -13,  60,   -75,  -66,  58,   -127, -2,   64,
>+    76,   -106, 73,   -49,  -31,  127,  126,  31,   16,   127,  -110, 107,
>+    -16,  -53,  20,   69,   -14,  -125, 59,   -44,  15,   120,  125,  125,
>+    43,   6,    19,   -58,  127,  127,  43,   16,   82,   97,   -127, 127,
>+    -93,  -41,  88,   0,    77,   -15,  116,  16,   -124, -31,  -3,   95,
>+    -40,  -126, -54,  -126, -83,  -8,   -59,  6,    67,   -29,  4,    124,
>+    -10,  112,  -28,  -8,   85,   -21,  45,   84,   6,    -8,   11,   72,
>+    32,   84,   -62,  77,   2,    -36,  75,   31,   -50,  116,  126,  119,
>+    -88,  -55,  -14,  -37,  126,  40,   -108, -6,   -6,   57,   64,   -28,
>+    -76,  30,   -117, -93,  31,   -92,  -44,  -64,  94,   58,   65,   114,
>+    41,   47,   71,   42,   -26,  99,   -126, 57,   -5,   74,   -19,  -113,
>+    -1,   67,   -21,  126,  1,    -3,   33,   60,   -82,  37,   -48,  89,
>+    114,  -38,  127,  -114, 35,   58,   -5,   21,   -46,  121,  -123, -43,
>+    127,  115,  123,  122,  -101, 126,  127,  81,   52,   89,   -127, 102,
>+    42,   117,  -9,   -2,   125,  127,  110,  96,   120,  66,   70,   124,
>+    55,   84,   -38,  -58,  119,  -127, -16,  -79,  123,  18,   -127, -50,
>+    -38,  120,  -85,  1,    7,    -56,  108,  -77,  -2,   21,   37,   1,
>+    13,   -105, -69,  28,   -87,  33,   -104, -51,  126,  41,   3,    -121,
>+    28,   71,   58,   86,   -8,   127,  94,   -55,  125,  40,   -19,  127,
>+    -33,  -87,  -23,  7,    -111, -68,  9,    84,   -119, 55,   -82,  78,
>+    -37,  -20,  -9,   -23,  53,   -13,  15,   -46,  116,  126,  -127, 56,
>+    -126, 125,  -7,   -1,   45,   26,   125,  121,  29,   47,   -86,  30,
>+    10,   76,   -125, -7,   23,   92,   -12,  -39,  -18,  92,   -97,  -8,
>+    -85,  -41,  49,   -50,  123,  -37,  -126, -30,  14,   79,   -49,  -65,
>+    9,    -36,  -38,  -96,  85,   -24,  -13,  37,   -25,  -5,   -64,  -127,
>+    55,   -60,  -18,  -61,  -63,  127,  56,   67,   15,   124,  72,   120,
>+    127,  40,   -10,  114,  24,   -23,  46,   78,   -53,  125,  86,   124,
>+    86,   0,    38,   93,   21,   127,  123,  75,   -72,  13,   48,   33,
>+    83,   -51,  15,   -32,  -49,  -33,  120,  64,   7,    9,    65,   60,
>+    21,   -21,  -61,  -53,  -113, 84,   -97,  101,  37,   -114, -27,  41,
>+    73,   126,  -10,  59,   61,   -15,  70,   -13,  82,   -4,   69,   56,
>+    94,   -91,  -50,  92,   -74,  -48,  53,   -7,   -107, 127,  28,   30,
>+    -26,  -21,  -61,  77,   82,   64,   -91,  -125, 122,  -104, 127,  123,
>+    122,  123,  76,   -126, 127,  -6,   -80,  7,    40,   -66,  -65,  54,
>+    -2,   23,   96,   -64,  74,   2,    -53,  -12,  -123, 39,   60,   -20,
>+    16,   -17,  -97,  23,   -4,   -53,  -122, 32,   -16,  -54,  -95,  43,
>+    71,   -1,   -67,  -33,  41,   18,   72,   28,   -83,  31,   -100, -91,
>+    -27,  10,   -128, -106, 2,    76,   -13,  42,   34,   112,  -19,  44,
>+    40,   -9,   -11,  65,   92,   -43,  -125, 2,    47,   -32,  25,   122,
>+    -29,  12,   101,  -8,   -126, -23,  43,   7,    125,  -20,  -124, 82,
>+    -2,   13,   -73,  -106, 115,  31,   116,  -23,  -44,  -71,  84,   3,
>+    47,   91,   127,  127,  -15,  95,   7,    93,   5,    113,  -50,  54,
>+    11,   13,   -127, 17,   72,   43,   -23,  5,    -70,  20,   15,   -27,
>+    99,   69,   -109, -122, -94,  16,   127,  0,    116,  104,  45,   108,
>+    -34,  87,   72,   -14,  118,  46,   42,   109,  -26,  95,   93,   127,
>+    60,   127,  -93,  -54,  -122, 34,   -105, 56,   55,   103,  125,  -71,
>+    -50,  95,   -72,  127,  107,  21,   73,   126,  61,   127,  127,  24,
>+    -62,  90,   73,   90,   -46,  -78,  -124, 72,   123,  -42,  50,   -107,
>+    17,   -32,  -62,  -89,  124,  1,    80,   -2,   117,  119,  -65,  -127,
>+    -95,  -121, -52,  103,  66,   75,   -3,   -62,  -127, 127,  -74,  124,
>+    79,   49,   40,   105,  -67,  -71,  -70,  43,   127,  119,  -4,   66,
>+    43,   23,   91,   -126, 15,   63,   -119, 112,  103,  15,   -99,  31,
>+    -127, 69,   116,  -46,  -67,  2,    -126, -29,  30,   30,   -69,  -98,
>+    -47,  -87,  -70,  -127, 23,   -73,  30,   -7,   94,   -52,  -65,  98,
>+    -45,  97,   53,   23,   -9,   -22,  -52,  -47,  6,    -1,   -85,  -15,
>+    -61,  -14,  68,   110,  -10,  -121, -25,  -35,  -15,  -94,  -123, 27,
>+    75,   48,   -66,  -56,  -44,  93,   109,  67,   -36,  24,   70,   -126,
>+    8,    -127, 126,  52,   11,   -32,  120,  -13,  -26,  -28,  -125, 127,
>+    106,  -50,  124,  36,   -126, -12,  0,    -23,  76,   -71,  -126, -12,
>+    -17,  -82,  12,   124,  57,   33,   4,    77,   -46,  71,   -34,  72,
>+    125,  -128, 124,  -24,  -128, 75,   -120, 69,   -45,  55,   33,   127,
>+    -33,  4,    -105, -41,  -59,  -91,  123,  44,   -127, 127,  -67,  52,
>+    25,   -125, -65,  100,  -25,  123,  6,    11,   -123, -92,  -33,  126,
>+    -17,  -4,   29,   33,   127,  96,   3,    87,   -48,  -18,  -70,  123,
>+    58,   -127, -3,   -52,  -1,   -36,  -41,  127,  51,   -52,  -27,  46,
>+    -83,  57,   9,    126,  127,  94,   79,   -37,  -127, -40,  67,   52,
>+    82,   -66,  122,  -13,  -73,  127,  -8,   -80,  46,   -48,  4,    -54};
>+
>+const int8_t kHiddenGruBias[kHiddenLayerBiases] = {
>+    124, 125, -57, -126, 53,  123, 127,  -75, 68,  102, -2, 116,
>+    124, 127, 124, 125,  126, 123, -16,  48,  125, 126, 78, 85,
>+    11,  126, -30, -30,  -64, -3,  -105, -29, -17, 69,  63, 2,
>+    -32, -10, -62, 113,  -52, 112, -109, 112, 7,   -40, 73, 53,
>+    62,  6,   -2,  0,    0,   100, -16,  26,  -24, 56,  26, -10,
>+    -33, 41,  70,  109,  -29, 127, 34,   -66, 49,  53,  27, 62};
>+
>+const int8_t kOutputDenseWeights[kOutputLayerWeights] = {
>+    127,  127,  127, 127,  127,  20,  127,  -126, -126, -54, 14,  125,
>+    -126, -126, 127, -125, -126, 127, -127, -127, -57,  -30, 127, 80};
>+
>+const int8_t kOutputDenseBias[kOutputLayerOutputSize] = {-50};
>+
>+}  // namespace rnnoise
>diff --git a/Source/ThirdParty/libwebrtc/Source/third_party/rnnoise/src/rnn_vad_weights.h b/Source/ThirdParty/libwebrtc/Source/third_party/rnnoise/src/rnn_vad_weights.h
>new file mode 100644
>index 00000000000..d55e33ddad7
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/third_party/rnnoise/src/rnn_vad_weights.h
>@@ -0,0 +1,37 @@
>+#ifndef THIRD_PARTY_RNNOISE_SRC_RNN_VAD_WEIGHTS_H_
>+#define THIRD_PARTY_RNNOISE_SRC_RNN_VAD_WEIGHTS_H_
>+
>+#include <cstdint>
>+#include <cstring>
>+
>+namespace rnnoise {
>+
>+// Weights scaling factor.
>+const float kWeightsScale = 1.f / 256.f;
>+
>+// Input layer (dense).
>+const size_t kInputLayerInputSize = 42;
>+const size_t kInputLayerOutputSize = 24;
>+const size_t kInputLayerWeights = kInputLayerInputSize * kInputLayerOutputSize;
>+extern const int8_t kInputDenseWeights[kInputLayerWeights];
>+extern const int8_t kInputDenseBias[kInputLayerOutputSize];
>+
>+// Hidden layer (GRU).
>+const size_t kHiddenLayerOutputSize = 24;
>+const size_t kHiddenLayerWeights =
>+    3 * kInputLayerOutputSize * kHiddenLayerOutputSize;
>+const size_t kHiddenLayerBiases = 3 * kHiddenLayerOutputSize;
>+extern const int8_t kHiddenGruWeights[kHiddenLayerWeights];
>+extern const int8_t kHiddenGruRecurrentWeights[kHiddenLayerWeights];
>+extern const int8_t kHiddenGruBias[kHiddenLayerBiases];
>+
>+// Output layer (dense).
>+const size_t kOutputLayerOutputSize = 1;
>+const size_t kOutputLayerWeights =
>+    kHiddenLayerOutputSize * kOutputLayerOutputSize;
>+extern const int8_t kOutputDenseWeights[kOutputLayerWeights];
>+extern const int8_t kOutputDenseBias[kOutputLayerOutputSize];
>+
>+}  // namespace rnnoise
>+
>+#endif  // THIRD_PARTY_RNNOISE_SRC_RNN_VAD_WEIGHTS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/.git-blame-ignore-revs b/Source/ThirdParty/libwebrtc/Source/webrtc/.git-blame-ignore-revs
>index dfa1db349d7..8f312425bb1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/.git-blame-ignore-revs
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/.git-blame-ignore-revs
>@@ -18,3 +18,5 @@
> 
> # Format all Java in WebRTC.
> b6760f9e4442410f2bcb6090b3b89bf709e2fce2
>+# Format all C++ in WebRTC.
>+665174fdbb4e0540eccb27cf7412348f1b65534c
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/.gitignore b/Source/ThirdParty/libwebrtc/Source/webrtc/.gitignore
>index bb2e7420f0f..bf937369ae8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/.gitignore
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/.gitignore
>@@ -37,4 +37,34 @@
> .pydevproject
> .settings
> .sw?
>+/Makefile
>+/base
>+/build
>+/buildtools
>+/ios
>+/mojo
>+/out
>+/testing
>+/third_party
>+/tools
>+/tools_webrtc/android/profiling/flamegraph
>+/tools_webrtc/android/profiling/simpleperf
>+/tools_webrtc/audio_quality/linux/pesq
>+/tools_webrtc/audio_quality/linux/PolqaOem64
>+/tools_webrtc/audio_quality/mac/pesq
>+/tools_webrtc/audio_quality/win/*.exe
>+/tools_webrtc/audio_quality/win/*.dll
>+/tools_webrtc/video_quality_toolchain/linux/ffmpeg
>+/tools_webrtc/video_quality_toolchain/linux/frame_analyzer
>+/tools_webrtc/video_quality_toolchain/linux/zxing
>+/tools_webrtc/video_quality_toolchain/mac/ffmpeg
>+/tools_webrtc/video_quality_toolchain/mac/zxing
>+/tools_webrtc/video_quality_toolchain/win/*.dll
>+/tools_webrtc/video_quality_toolchain/win/*.exe
>+/rtc_tools/testing/*.zip
>+/rtc_tools/testing/*.gz
>+/rtc_tools/testing/golang/*/*.gz
>+/rtc_tools/testing/golang/*/*.zip
>+/x86-generic_out/
>+/xcodebuild
> !webrtc/*
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/.gn
>index 5e3c6ca9aff..9110455155b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/.gn
>@@ -42,7 +42,6 @@ check_targets = [
>   "//system_wrappers/*",
>   "//test/*",
>   "//video/*",
>-  "//voice_engine/*",
>   "//third_party/libyuv/*",
> ]
> 
>@@ -67,6 +66,10 @@ default_args = {
>   # http://bugs.webrtc.org/8570
>   ios_deployment_target = "9.0"
> 
>+  # The SDK API level, in contrast, is set by build/android/AndroidManifest.xml.
>+  android32_ndk_api_level = 16
>+  android64_ndk_api_level = 21
>+
>   # WebRTC does not want to switch to C++14 yet.
>   use_cxx11 = true
> 
>@@ -76,4 +79,6 @@ default_args = {
>   # WebRTC does not provide the gflags dependency. Because libyuv uses it only
>   # for unittests, it can be disabled (see third_party/libyuv/BUILD.gn)
>   libyuv_use_gflags = false
>+
>+  gtest_enable_absl_printers = true
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/.vpython b/Source/ThirdParty/libwebrtc/Source/webrtc/.vpython
>index cf74427542e..05bbe14fde4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/.vpython
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/.vpython
>@@ -30,3 +30,33 @@ wheel: <
>   name: "infra/python/wheels/psutil/${platform}_${py_python}_${py_abi}"
>   version: "version:5.2.2"
> >
>+
>+# Used by:
>+#   build/toolchain/win
>+wheel: <
>+  name: "infra/python/wheels/pypiwin32/${vpython_platform}"
>+  version: "version:219"
>+  match_tag: <
>+    platform: "win32"
>+  >
>+  match_tag: <
>+    platform: "win_amd64"
>+  >
>+>
>+
>+wheel: <
>+  name: "infra/python/wheels/six-py2_py3"
>+  version: "version:1.10.0"
>+>
>+wheel: <
>+  name: "infra/python/wheels/pbr-py2_py3"
>+  version: "version:3.0.0"
>+>
>+wheel: <
>+  name: "infra/python/wheels/funcsigs-py2_py3"
>+  version: "version:1.0.2"
>+>
>+wheel: <
>+  name: "infra/python/wheels/mock-py2_py3"
>+  version: "version:2.0.0"
>+>
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/AUTHORS b/Source/ThirdParty/libwebrtc/Source/webrtc/AUTHORS
>index 6fe77482297..41833d090bf 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/AUTHORS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/AUTHORS
>@@ -14,6 +14,7 @@ Chris Tserng <tserng@amazon.com>
> Christophe Dumez <ch.dumez@samsung.com>
> Cody Barnes <conceptgenesis@gmail.com>
> Colin Plumb
>+David Porter <david@porter.me>
> Dax Booysen <dax@younow.com>
> Dmitry Lizin <sdkdimon@gmail.com>
> Eric Rescorla, RTFM Inc. <ekr@rtfm.com>
>@@ -24,20 +25,24 @@ Gustavo Garcia <gustavogb@gmail.com>
> Hugues Ekra <hekra01@gmail.com>
> Jake Hilton <jakehilton@gmail.com>
> James H. Brown <jbrown@burgoyne.com>
>+Jan Kalab <pitlicek@gmail.com>
> Jens Nielsen <jens.nielsen@berotec.se>
> Jiawei Ou <jiawei.ou@gmail.com>
> Jie Mao <maojie0924@gmail.com>
> Luke Weber <luke.weber@gmail.com>
>+Maksim Khobat <maksimkhobat@gmail.com>
> Mallikarjuna Rao V <vm.arjun@samsung.com>
> Manish Jethani <manish.jethani@gmail.com>
> Martin Storsjo <martin@martin.st>
> Matthias Liebig <matthias.gcode@gmail.com>
> Maxim Potapov <vopatop.skam@gmail.com>
>+Michael Iedema <michael@kapsulate.com>
> Mike Gilbert <floppymaster@gmail.com>
> Mo Zanaty <mzanaty@cisco.com>
> Pali Rohar
> Paul Kapustin <pkapustin@gmail.com>
> Philipp Hancke <philipp.hancke@googlemail.com>
>+Peng Yu <yupeng323@gmail.com>
> Rafael Lopez Diez <rafalopezdiez@gmail.com>
> Ralph Giles <giles@ghostscript.com>
> Riku Voipio <riku.voipio@linaro.org>
>@@ -59,6 +64,10 @@ Yura Yaroshevich <yura.yaroshevich@gmail.com>
> Hans Knoechel <hans@hans-knoechel.de>
> Korniltsev Anatoly <korniltsev.anatoly@gmail.com>
> Todd Wong <todd.wong.ndq@gmail.com>
>+Sergio Garcia Murillo <sergio.garcia.murillo@gmail.com>
>+Maxim Pavlov <pavllovmax@gmail.com>
>+Yusuke Suzuki <utatane.tea@gmail.com>
>+Piasy Xu <xz4215@gmail.com>
> 
> &yet LLC <*@andyet.com>
> Agora IO <*@agora.io>
>@@ -66,6 +75,7 @@ ARM Holdings <*@arm.com>
> BroadSoft Inc. <*@broadsoft.com>
> Facebook Inc. <*@fb.com>
> Google Inc. <*@google.com>
>+HyperConnect Inc. <*@hpcnt.com>
> Life On Air Inc. <*@lifeonair.com>
> Intel Corporation <*@intel.com>
> MIPS Technologies <*@mips.com>
>@@ -78,6 +88,10 @@ Telenor Digital AS <*@telenor.com>
> Temasys Communications <*@temasys.io>
> The Chromium Authors <*@chromium.org>
> The WebRTC Authors <*@webrtc.org>
>+Videxio AS <*@videxio.com>
> Vonage Holdings Corp. <*@vonage.com>
> Wire Swiss GmbH <*@wire.com>
> Miguel Paris <mparisdiaz@gmail.com>
>+Vewd Software AS <*@vewd.com>
>+Highfive, Inc. <*@highfive.com>
>+CoSMo Software Consulting, Pte Ltd <*@cosmosoftware.io>
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/BUILD.gn
>index 2f48b6735a6..3ae4500f393 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/BUILD.gn
>@@ -43,6 +43,7 @@ if (!build_with_chromium) {
>         ":video_engine_tests",
>         ":webrtc_nonparallel_tests",
>         ":webrtc_perf_tests",
>+        "call:fake_network_unittests",
>         "common_audio:common_audio_unittests",
>         "common_video:common_video_unittests",
>         "media:rtc_media_unittests",
>@@ -63,12 +64,11 @@ if (!build_with_chromium) {
>         "video:screenshare_loopback",
>         "video:sv_loopback",
>         "video:video_loopback",
>-        "voice_engine:voice_engine_unittests",
>       ]
>       if (is_android) {
>         deps += [
>           ":android_junit_tests",
>-          "sdk/android:libjingle_peerconnection_android_unittest",
>+          "sdk/android:android_instrumentation_test_apk",
>         ]
>       } else {
>         deps += [ "modules/video_capture:video_capture_tests" ]
>@@ -93,6 +93,10 @@ config("common_inherited_config") {
>     defines += [ "WEBRTC_MOZILLA_BUILD" ]
>   }
> 
>+  if (!rtc_builtin_ssl_root_certificates) {
>+    defines += [ "WEBRTC_EXCLUDE_BUILT_IN_SSL_ROOT_CERTS" ]
>+  }
>+
>   # Some tests need to declare their own trace event handlers. If this define is
>   # not set, the first time TRACE_EVENT_* is called it will store the return
>   # value for the current handler in an static variable, so that subsequent
>@@ -105,13 +109,7 @@ config("common_inherited_config") {
>     defines += [ "WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS=0" ]
>   }
>   if (build_with_chromium) {
>-    defines += [
>-      # TODO(kjellander): Cleanup unused ones and move defines closer to
>-      # the source when webrtc:4256 is completed.
>-      "FEATURE_ENABLE_VOICEMAIL",
>-      "GTEST_RELATIVE_PATH",
>-      "WEBRTC_CHROMIUM_BUILD",
>-    ]
>+    defines += [ "WEBRTC_CHROMIUM_BUILD" ]
>     include_dirs = [
>       # The overrides must be included first as that is the mechanism for
>       # selecting the override headers in Chromium.
>@@ -122,7 +120,7 @@ config("common_inherited_config") {
>       ".",
>     ]
>   }
>-  if (is_posix) {
>+  if (is_posix || is_fuchsia) {
>     defines += [ "WEBRTC_POSIX" ]
>   }
>   if (is_ios) {
>@@ -141,10 +139,7 @@ config("common_inherited_config") {
>     defines += [ "WEBRTC_FUCHSIA" ]
>   }
>   if (is_win) {
>-    defines += [
>-      "WEBRTC_WIN",
>-      "_CRT_SECURE_NO_WARNINGS",  # Suppress warnings about _vsnprinf
>-    ]
>+    defines += [ "WEBRTC_WIN" ]
>   }
>   if (is_android) {
>     defines += [
>@@ -169,22 +164,13 @@ config("common_inherited_config") {
>   if (is_ubsan) {
>     cflags += [ "-fsanitize=float-cast-overflow" ]
>   }
>-
>-  # TODO(GYP): Support these in GN.
>-  # if (is_bsd) {
>-  #   defines += [ "BSD" ]
>-  # }
>-  # if (is_openbsd) {
>-  #   defines += [ "OPENBSD" ]
>-  # }
>-  # if (is_freebsd) {
>-  #   defines += [ "FREEBSD" ]
>-  # }
> }
> 
> config("common_config") {
>   cflags = []
>+  cflags_c = []
>   cflags_cc = []
>+  cflags_objc = []
>   defines = []
> 
>   if (rtc_enable_protobuf) {
>@@ -193,10 +179,6 @@ config("common_config") {
>     defines += [ "WEBRTC_ENABLE_PROTOBUF=0" ]
>   }
> 
>-  if (rtc_restrict_logging) {
>-    defines += [ "WEBRTC_RESTRICT_LOGGING" ]
>-  }
>-
>   if (rtc_include_internal_audio_device) {
>     defines += [ "WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE" ]
>   }
>@@ -213,6 +195,10 @@ config("common_config") {
>     defines += [ "ENABLE_EXTERNAL_AUTH" ]
>   }
> 
>+  if (rtc_use_builtin_sw_codecs) {
>+    defines += [ "USE_BUILTIN_SW_CODECS" ]
>+  }
>+
>   if (build_with_chromium) {
>     defines += [
>       # NOTICE: Since common_inherited_config is used in public_configs for our
>@@ -222,10 +208,9 @@ config("common_config") {
>       "HAVE_WEBRTC_VIDEO",
>       "HAVE_WEBRTC_VOICE",
>       "LOGGING_INSIDE_WEBRTC",
>-      "USE_WEBRTC_DEV_BRANCH",
>     ]
>   } else {
>-    if (is_posix) {
>+    if (is_posix || is_fuchsia) {
>       # Enable more warnings: -Wextra is currently disabled in Chromium.
>       cflags = [
>         "-Wextra",
>@@ -233,8 +218,21 @@ config("common_config") {
>         # Repeat some flags that get overridden by -Wextra.
>         "-Wno-unused-parameter",
>         "-Wno-missing-field-initializers",
>-        "-Wno-strict-overflow",
>       ]
>+      cflags_c += [
>+        # TODO(bugs.webrtc.org/9029): enable commented compiler flags.
>+        # Some of these flags should also be added to cflags_objc.
>+
>+        # "-Wextra",  (used when building C++ but not when building C)
>+        # "-Wmissing-prototypes",  (C/Obj-C only)
>+        # "-Wmissing-declarations",  (ensure this is always used C/C++, etc..)
>+        "-Wstrict-prototypes",
>+
>+        # "-Wpointer-arith",  (ensure this is always used C/C++, etc..)
>+        # "-Wbad-function-cast",  (C/Obj-C only)
>+        # "-Wnested-externs",  (C/Obj-C only)
>+      ]
>+      cflags_objc += [ "-Wstrict-prototypes" ]
>       cflags_cc = [
>         "-Wnon-virtual-dtor",
> 
>@@ -261,6 +259,13 @@ config("common_config") {
>         cflags += [ "-Wunused-lambda-capture" ]
>       }
>     }
>+
>+    if (is_win && !is_clang) {
>+      # MSVC warning suppressions (needed to use Abseil).
>+      # TODO(bugs.webrtc.org/9274): Remove these warnings as soon as MSVC allows
>+      # external headers warning suppression (or fix them upstream).
>+      cflags += [ "/wd4702" ]  # unreachable code
>+    }
>   }
> 
>   if (current_cpu == "arm64") {
>@@ -307,7 +312,7 @@ config("common_config") {
>     ]
>   }
> 
>-  if (use_libfuzzer || use_drfuzz || use_afl) {
>+  if (use_fuzzing_engine && optimize_for_fuzzing) {
>     # Used in Chromium's overrides to disable logging
>     defines += [ "WEBRTC_UNSAFE_FUZZER_MODE" ]
>   }
>@@ -325,6 +330,7 @@ if (!build_with_chromium) {
> 
>     sources = []
>     complete_static_lib = true
>+    suppressed_configs += [ "//build/config/compiler:thin_archive" ]
>     defines = []
> 
>     deps = [
>@@ -342,12 +348,22 @@ if (!build_with_chromium) {
>       "sdk",
>       "system_wrappers:system_wrappers_default",
>       "video",
>-      "voice_engine",
>+    ]
>+
>+    # Additional factory functions to be exposed.
>+    # Rational: These factories are small enough (89 KiB / 32+ MiB)
>+    #           to be unconditionaly included for user convenience.
>+    # That begin said:
>+    # TODO(yvesg) Consider making all non-core APIs optional, so that users
>+    #             can build a customized library tailored to their needs.
>+    deps += [
>+      "api/audio_codecs:builtin_audio_decoder_factory",
>+      "api/audio_codecs:builtin_audio_encoder_factory",
>     ]
> 
>     if (build_with_mozilla) {
>       deps += [
>-        "api:video_frame_api",
>+        "api/video:video_frame",
>         "system_wrappers:field_trial_default",
>         "system_wrappers:metrics_default",
>       ]
>@@ -368,21 +384,13 @@ if (!build_with_chromium) {
>   }
> }
> 
>-rtc_source_set("typedefs") {
>-  sources = [
>-    "typedefs.h",
>-  ]
>-}
>-
>-rtc_static_library("webrtc_common") {
>+rtc_source_set("webrtc_common") {
>   sources = [
>-    "common_types.cc",
>     "common_types.h",
>   ]
>   deps = [
>-    ":typedefs",
>     "api:array_view",
>-    "api:optional",
>+    "api/video:video_bitrate_allocation",
>     "rtc_base:checks",
>     "rtc_base:deprecation",
>     "rtc_base:stringutils",
>@@ -406,25 +414,16 @@ if (use_libfuzzer || use_drfuzz || use_afl) {
> }
> 
> if (rtc_include_tests) {
>-  config("rtc_unittests_config") {
>-    # GN orders flags on a target before flags from configs. The default config
>-    # adds -Wall, and this flag have to be after -Wall -- so they need to
>-    # come from a config and can"t be on the target directly.
>-    if (is_clang) {
>-      cflags = [
>-        "-Wno-sign-compare",
>-        "-Wno-unused-const-variable",
>-      ]
>-    }
>-  }
>-
>   rtc_test("rtc_unittests") {
>     testonly = true
> 
>     deps = [
>       ":webrtc_common",
>       "api:rtc_api_unittests",
>+      "api/audio/test:audio_api_unittests",
>       "api/audio_codecs/test:audio_codecs_api_unittests",
>+      "api/video/test:rtc_api_video_unittests",
>+      "api/video_codecs/test:video_codecs_api_unittests",
>       "p2p:libstunprober_unittests",
>       "p2p:rtc_p2p_unittests",
>       "rtc_base:rtc_base_approved_unittests",
>@@ -434,7 +433,9 @@ if (rtc_include_tests) {
>       "rtc_base:rtc_numerics_unittests",
>       "rtc_base:rtc_task_queue_unittests",
>       "rtc_base:sequenced_task_checker_unittests",
>+      "rtc_base:sigslot_unittest",
>       "rtc_base:weak_ptr_unittests",
>+      "rtc_base/experiments:experiments_unittests",
>       "system_wrappers:metrics_default",
>       "system_wrappers:runtime_enabled_features_default",
>     ]
>@@ -444,12 +445,19 @@ if (rtc_include_tests) {
>     }
> 
>     if (is_android) {
>-      deps += [ "//testing/android/native_test:native_test_support" ]
>+      # Do not use Chromium's launcher. native_unittests defines its own JNI_OnLoad.
>+      use_default_launcher = false
>+
>+      deps += [
>+        "sdk/android:native_unittests",
>+        "sdk/android:native_unittests_java",
>+        "//testing/android/native_test:native_test_support",
>+      ]
>       shard_timeout = 900
>     }
> 
>     if (is_ios || is_mac) {
>-      deps += [ "sdk:sdk_unittests_objc" ]
>+      deps += [ "sdk:rtc_unittests_objc" ]
>     }
>   }
> 
>@@ -526,14 +534,13 @@ if (rtc_include_tests) {
> 
>   rtc_test("webrtc_perf_tests") {
>     testonly = true
>-    configs += [ ":rtc_unittests_config" ]
>-
>     deps = [
>       "audio:audio_perf_tests",
>       "call:call_perf_tests",
>       "modules/audio_coding:audio_coding_perf_tests",
>       "modules/audio_processing:audio_processing_perf_tests",
>       "modules/remote_bitrate_estimator:remote_bitrate_estimator_perf_tests",
>+      "pc:peerconnection_perf_tests",
>       "test:test_main",
>       "video:video_full_stack_tests",
>     ]
>@@ -566,13 +573,35 @@ if (rtc_include_tests) {
>         "examples/androidjunit/src/org/appspot/apprtc/DirectRTCClientTest.java",
>         "examples/androidjunit/src/org/appspot/apprtc/TCPChannelClientTest.java",
>         "sdk/android/tests/src/org/webrtc/CameraEnumerationTest.java",
>+        "sdk/android/tests/src/org/webrtc/CodecTestHelper.java",
>+        "sdk/android/tests/src/org/webrtc/FakeMediaCodecWrapper.java",
>+        "sdk/android/tests/src/org/webrtc/GlGenericDrawerTest.java",
>+        "sdk/android/tests/src/org/webrtc/HardwareVideoEncoderTest.java",
>+        "sdk/android/tests/src/org/webrtc/HardwareVideoDecoderTest.java",
>+        "sdk/android/tests/src/org/webrtc/ScalingSettingsTest.java",
>       ]
> 
>       deps = [
>         "examples:AppRTCMobile_javalib",
>         "sdk/android:libjingle_peerconnection_java",
>         "//base:base_java_test_support",
>+        "//third_party/google-truth:google_truth_java",
>       ]
>     }
>   }
> }
>+
>+# ---- Poisons ----
>+#
>+# Here is one empty dummy target for each poison type (needed because
>+# "being poisonous with poison type foo" is implemented as "depends on
>+# //:poison_foo").
>+#
>+# The set of poison_* targets needs to be kept in sync with the
>+# `all_poison_types` list in webrtc.gni.
>+#
>+group("poison_audio_codecs") {
>+}
>+
>+group("poison_software_video_codecs") {
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/DEPS b/Source/ThirdParty/libwebrtc/Source/webrtc/DEPS
>index 7a50227de5e..2512bc9fa35 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/DEPS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/DEPS
>@@ -7,20 +7,16 @@ vars = {
>   'checkout_configuration': 'default',
>   'checkout_instrumented_libraries': 'checkout_linux and checkout_configuration == "default"',
>   'webrtc_git': 'https://webrtc.googlesource.com',
>-  'chromium_revision': '97bf3ad18eb9c822964de681b94f7f880711c932',
>+  'chromium_revision': '012c9b03138d1b8398b756565826b93d95cf8277',
>   'boringssl_git': 'https://boringssl.googlesource.com',
>   # Three lines of non-changing comments so that
>   # the commit queue can handle CLs rolling swarming_client
>   # and whatever else without interference from each other.
>-  'swarming_revision': '88229872dd17e71658fe96763feaa77915d8cbd6',
>-  # Three lines of non-changing comments so that
>-  # the commit queue can handle CLs rolling openmax_dl
>-  # and whatever else without interference from each other.
>-  'openmax_dl_revision': 'b611996df3b8f6b151339d22c12c21f167009cb6',
>+  'swarming_revision': '486c9b53c4d54dd4b95bb6ce0e31160e600dfc11',
>   # Three lines of non-changing comments so that
>   # the commit queue can handle CLs rolling BoringSSL
>   # and whatever else without interference from each other.
>-  'boringssl_revision': '94cd196a80252c98e329e979870f2a462cc4f402',
>+  'boringssl_revision': '6b0d82229b2c4ce85b1c3f6d55bda661e07bead2',
>   # Three lines of non-changing comments so that
>   # the commit queue can handle CLs rolling lss
>   # and whatever else without interference from each other.
>@@ -28,39 +24,53 @@ vars = {
>   # Three lines of non-changing comments so that
>   # the commit queue can handle CLs rolling catapult
>   # and whatever else without interference from each other.
>-  'catapult_revision': '785486272fdd040ffc53214deaa8d30a293df6f2',
>+  'catapult_revision': 'ed63b1319414a36b099cad8443d497bda8f085a2',
>   # Three lines of non-changing comments so that
>   # the commit queue can handle CLs rolling libFuzzer
>   # and whatever else without interference from each other.
>-  'libfuzzer_revision': 'ba2c1cd6f87accb32b5dbce297387c56a2e53a2f',
>+  'libfuzzer_revision': '658ff786a213703ff0df6ba4a288e9a1e218c074',
>+  # Three lines of non-changing comments so that
>+  # the commit queue can handle CLs rolling freetype
>+  # and whatever else without interference from each other.
>+  'freetype_revision': '96b5e500909cfce39ff78feabefd8063a229b951',
>+  # Three lines of non-changing comments so that
>+  # the commit queue can handle CLs rolling HarfBuzz
>+  # and whatever else without interference from each other.
>+  'harfbuzz_revision': '22defe0965adddaa09eebc13df7fa6c64e2abba3',
> }
> deps = {
>   # TODO(kjellander): Move this to be Android-only once the libevent dependency
>   # in base/third_party/libevent is solved.
>   'src/base':
>-    Var('chromium_git') + '/chromium/src/base' + '@' + 'dc7e66bd14356929c632bd3ebf304092ae9a07fa',
>+    Var('chromium_git') + '/chromium/src/base' + '@' + '33838037fc6db947c6bb42d50c0b9d4da18c04c1',
>   'src/build':
>-    Var('chromium_git') + '/chromium/src/build' + '@' + '2c4f0d08ec19c5be2c3bcb2e35c5bcebd6a61fc3',
>+    Var('chromium_git') + '/chromium/src/build' + '@' + '48f1034a8c3bbc18ce994eaf7dbbc158a4789cf9',
>   'src/buildtools':
>-    Var('chromium_git') + '/chromium/buildtools.git' + '@' + '6fe4a3251488f7af86d64fc25cf442e817cf6133',
>+    Var('chromium_git') + '/chromium/buildtools.git' + '@' + '2dff9c9c74e9d732e6fe57c84ef7fd044cc45d96',
>+  # Gradle 4.3-rc4. Used for testing Android Studio project generation for WebRTC.
>+  'src/examples/androidtests/third_party/gradle': {
>+    'url': Var('chromium_git') + '/external/github.com/gradle/gradle.git' + '@' +
>+      '89af43c4d0506f69980f00dde78c97b2f81437f8',
>+    'condition': 'checkout_android',
>+  },
>   'src/ios': {
>-    'url': Var('chromium_git') + '/chromium/src/ios' + '@' + '406d7b7ac94709f7b8ee20dd6feebe8c722a78af',
>+    'url': Var('chromium_git') + '/chromium/src/ios' + '@' + 'a2b3d838cc8e71f50108611600a57061ba90283f',
>     'condition': 'checkout_ios',
>   },
>   'src/testing':
>-    Var('chromium_git') + '/chromium/src/testing' + '@' + 'e3baccadac00d406bb2c6011c3c80531204af782',
>+    Var('chromium_git') + '/chromium/src/testing' + '@' + 'bc0a21b479692b7bf7a96ce5bb84cce79d655e97',
>   'src/third_party':
>-    Var('chromium_git') + '/chromium/src/third_party' + '@' + 'f01e063dc62a220110fa5a0250fd60b0ed18ab95',
>+    Var('chromium_git') + '/chromium/src/third_party' + '@' + '71a1eaaf86eddb7d6a327fc6b8905a0f0be361bf',
>   'src/third_party/android_ndk': {
>-      'url': Var('chromium_git') + '/android_ndk.git' + '@' + 'e951c37287c7d8cd915bf8d4149fd4a06d808b55',
>+      'url': Var('chromium_git') + '/android_ndk.git' + '@' + '4e2cea441bfd43f0863d14f57b1e1844260b9884',
>       'condition': 'checkout_android',
>   },
>   'src/third_party/android_tools': {
>-    'url': Var('chromium_git') + '/android_tools.git' + '@' + 'c78b25872734e0038ae2a333edc645cd96bc232d',
>+    'url': Var('chromium_git') + '/android_tools.git' + '@' + '130499e25286f4d56acafa252fee09f3cc595c49',
>     'condition': 'checkout_android',
>   },
>   'src/third_party/auto/src': {
>-    'url': Var('chromium_git') + '/external/github.com/google/auto.git' + '@' + '71802f2ae74dae2744abd999f8434e13055c4ee3',
>+    'url': Var('chromium_git') + '/external/github.com/google/auto.git' + '@' + '8a81a858ae7b78a1aef71ac3905fade0bbd64e82',
>     'condition': 'checkout_android',
>   },
>   'src/third_party/boringssl/src':
>@@ -74,24 +84,28 @@ deps = {
>   'src/third_party/colorama/src':
>     Var('chromium_git') + '/external/colorama.git' + '@' + '799604a1041e9b3bc5d2789ecbd7e8db2e18e6b8',
>   'src/third_party/depot_tools':
>-    Var('chromium_git') + '/chromium/tools/depot_tools.git' + '@' + '5d6b00fac64a829cca5637bb76abe618291b8b60',
>+    Var('chromium_git') + '/chromium/tools/depot_tools.git' + '@' + 'efb38bb3d729f9d5b27f5202c766d43186c0103a',
>   'src/third_party/errorprone/lib': {
>-      'url': Var('chromium_git') + '/chromium/third_party/errorprone.git' + '@' + 'ecc57c2b00627667874744b9ad8efe10734d97a8',
>+      'url': Var('chromium_git') + '/chromium/third_party/errorprone.git' + '@' + '980d49e839aa4984015efed34b0134d4b2c9b6d7',
>       'condition': 'checkout_android',
>   },
>   'src/third_party/ffmpeg':
>-    Var('chromium_git') + '/chromium/third_party/ffmpeg.git' + '@' + 'b64dedac9d1d70148a50791d127eede2e7eb93ba',
>+    Var('chromium_git') + '/chromium/third_party/ffmpeg.git' + '@' + '90210b5e10d3917567a3025e4853704bfefd8384',
>   'src/third_party/findbugs': {
>     'url': Var('chromium_git') + '/chromium/deps/findbugs.git' + '@' + '4275d9ac8610db6b1bc9a5e887f97e41b33fac67',
>     'condition': 'checkout_android',
>   },
>+  'src/third_party/freetype/src':
>+    Var('chromium_git') + '/chromium/src/third_party/freetype2.git' + '@' + Var('freetype_revision'),
>+  'src/third_party/harfbuzz-ng/src':
>+    Var('chromium_git') + '/external/github.com/harfbuzz/harfbuzz.git' + '@' + Var('harfbuzz_revision'),
>   # WebRTC-only dependency (not present in Chromium).
>   'src/third_party/gtest-parallel':
>-    Var('chromium_git') + '/external/github.com/google/gtest-parallel' + '@' + '180c2f52341c065fd4d8418a238375ab6c91b336',
>+    Var('chromium_git') + '/external/github.com/google/gtest-parallel' + '@' + 'fe7f791f14769390d0b124ef8231cde4d575eb12',
>   'src/third_party/googletest/src':
>-    Var('chromium_git') + '/external/github.com/google/googletest.git' + '@' + '0062e4869f07a3ef235703ddf63af604b712446c',
>+    Var('chromium_git') + '/external/github.com/google/googletest.git' + '@' + 'd5266326752f0a1dadbd310932d8f4fd8c3c5e7d',
>   'src/third_party/icu': {
>-    'url': Var('chromium_git') + '/chromium/deps/icu.git' + '@' + 'f3d25bcc2e542a10248d5a272219ed7695fe3252',
>+    'url': Var('chromium_git') + '/chromium/deps/icu.git' + '@' + '297a4dd02b9d36c92ab9b4f121e433c9c3bc14f8',
>   },
>   'src/third_party/jsr-305/src': {
>     'url': Var('chromium_git') + '/external/jsr-305.git' + '@' + '642c508235471f7220af6d5df2d3210e3bfc0919',
>@@ -109,23 +123,31 @@ deps = {
>   'src/third_party/libjpeg_turbo':
>     Var('chromium_git') + '/chromium/deps/libjpeg_turbo.git' + '@' + 'a1750dbc79a8792dde3d3f7d7d8ac28ba01ac9dd',
>   'src/third_party/libsrtp':
>-    Var('chromium_git') + '/chromium/deps/libsrtp.git' + '@' + '1d45b8e599dc2db6ea3ae22dbc94a8c504652423',
>+    Var('chromium_git') + '/chromium/deps/libsrtp.git' + '@' + '368abd6bb3091df2b354250818714f72f0692ca5',
>   'src/third_party/libvpx/source/libvpx':
>-    Var('chromium_git') + '/webm/libvpx.git' + '@' +  'bed28a55f593efd3a71a3a9d05cf8bb25d15fa44',
>+    Var('chromium_git') + '/webm/libvpx.git' + '@' +  'b8642738c9851232c9bb1e1a22474953d3d367cf',
>   'src/third_party/libyuv':
>-    Var('chromium_git') + '/libyuv/libyuv.git' + '@' + '50f9e618fa4bcb0561622fd709bee5da922b0fd4',
>+    Var('chromium_git') + '/libyuv/libyuv.git' + '@' + 'd694f0a82b4da9d8ea37e6c453b7a34947eb5790',
>   'src/third_party/lss': {
>     'url': Var('chromium_git') + '/linux-syscall-support.git' + '@' + Var('lss_revision'),
>     'condition': 'checkout_android or checkout_linux',
>   },
>   'src/third_party/mockito/src': {
>-    'url': Var('chromium_git') + '/external/mockito/mockito.git' + '@' + 'de83ad4598ad4cf5ea53c69a8a8053780b04b850',
>+    'url': Var('chromium_git') + '/external/mockito/mockito.git' + '@' + '04a2a289a4222f80ad20717c25144981210d2eac',
>     'condition': 'checkout_android',
>   },
>   'src/third_party/openh264/src':
>-    Var('chromium_git') + '/external/github.com/cisco/openh264' + '@' + '5a5c4f14f471b9f434a55c39e942153453f25661',
>-  'src/third_party/openmax_dl':
>-    Var('webrtc_git') + '/deps/third_party/openmax.git' + '@' +  Var('openmax_dl_revision'),
>+    Var('chromium_git') + '/external/github.com/cisco/openh264' + '@' + '3b51f16a4a41df729f8d647f03e48c5f272911ff',
>+  'src/third_party/r8': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/r8',
>+              'version': 'version:1.2.28-cr0',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>   'src/third_party/requests/src': {
>     'url': Var('chromium_git') + '/external/github.com/kennethreitz/requests.git' + '@' + 'f172b30356d821d180fa4ecfa3e71c7274a32de4',
>     'condition': 'checkout_android',
>@@ -139,10 +161,10 @@ deps = {
>     'condition': 'checkout_android',
>   },
>   'src/third_party/usrsctp/usrsctplib':
>-    Var('chromium_git') + '/external/github.com/sctplab/usrsctp' + '@' + '0e076261b832121cf120ddc04aaff87ac3a34d30',
>+    Var('chromium_git') + '/external/github.com/sctplab/usrsctp' + '@' + '7a8bc9a90ca96634aa56ee712856d97f27d903f8',
>   # WebRTC-only dependency (not present in Chromium).
>   'src/third_party/winsdk_samples': {
>-    'url': Var('webrtc_git') + '/deps/third_party/winsdk_samples_v71' + '@' + '2d31a1cbecc86359e6ec041fb9ff6c082babd073',
>+    'url': Var('webrtc_git') + '/deps/third_party/winsdk_samples_v71' + '@' + 'a59391ef795986633735a1695caa97622a9bfd56',
>     'condition': 'checkout_win',
>   },
>   # Dependency used by libjpeg-turbo.
>@@ -151,13 +173,532 @@ deps = {
>     'condition': 'checkout_win',
>   },
>   'src/third_party/yasm/source/patched-yasm':
>-    Var('chromium_git') + '/chromium/deps/yasm/patched-yasm.git' + '@' + 'b98114e18d8b9b84586b10d24353ab8616d4c5fc',
>+    Var('chromium_git') + '/chromium/deps/yasm/patched-yasm.git' + '@' + '720b70524a4424b15fc57e82263568c8ba0496ad',
>   'src/tools':
>-    Var('chromium_git') + '/chromium/src/tools' + '@' + 'c2699036532f9ce005eb549703b2d64518cabd7a',
>-  'src/tools/gyp':
>-    Var('chromium_git') + '/external/gyp.git' + '@' + 'd61a9397e668fa9843c4aa7da9e79460fe590bfb',
>+    Var('chromium_git') + '/chromium/src/tools' + '@' + 'd6f9a30225a3b6fd4f5509d5727e991f69ea1627',
>   'src/tools/swarming_client':
>     Var('chromium_git') + '/infra/luci/client-py.git' + '@' +  Var('swarming_revision'),
>+
>+  'src/third_party/accessibility_test_framework': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/accessibility-test-framework',
>+              'version': 'version:2.1-cr0',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>+
>+  'src/third_party/android_support_test_runner': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/android_support_test_runner',
>+              'version': 'version:0.5-cr0',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>+
>+  'src/third_party/apk-patch-size-estimator': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/apk-patch-size-estimator',
>+              'version': 'version:0.2-cr0',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>+
>+  'src/third_party/bazel': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/bazel',
>+              'version': 'version:0.10.0',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>+
>+  'src/third_party/bouncycastle': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/bouncycastle',
>+              'version': 'version:1.46-cr0',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>+
>+  'src/third_party/byte_buddy': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/byte_buddy',
>+              'version': 'version:1.8.8-cr0',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>+
>+  'src/third_party/espresso': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/espresso',
>+              'version': 'version:2.2.1-cr0',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>+
>+  'src/third_party/gson': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/gson',
>+              'version': 'version:2.8.0-cr0',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>+
>+  'src/third_party/guava': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/guava',
>+              'version': 'version:23.0-cr0',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>+
>+  'src/third_party/hamcrest': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/hamcrest',
>+              'version': 'version:1.3-cr0',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>+
>+  'src/third_party/icu4j': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/icu4j',
>+              'version': 'version:53.1-cr0',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>+
>+  'src/third_party/intellij': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/intellij',
>+              'version': 'version:12.0-cr0',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>+
>+  'src/third_party/javax_inject': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/javax_inject',
>+              'version': 'version:1-cr0',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>+
>+  'src/third_party/objenesis': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/objenesis',
>+              'version': 'version:2.4-cr0',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>+
>+  'src/third_party/ow2_asm': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/ow2_asm',
>+              'version': 'version:5.0.1-cr0',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>+
>+  'src/third_party/robolectric': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/robolectric',
>+              'version': 'version:3.5.1',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>+
>+  'src/third_party/sqlite4java': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/sqlite4java',
>+              'version': 'version:0.282-cr0',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>+
>+  'src/third_party/xstream': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/xstream',
>+              'version': 'version:1.4.8-cr0',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>+
>+  # === ANDROID_DEPS Start ===
>+  'src/third_party/android_deps/libs/android_arch_core_common': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/android_deps/libs/android_arch_core_common',
>+              'version': 'version:1.0.0-cr0',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>+  'src/third_party/android_deps/libs/android_arch_lifecycle_common': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/android_deps/libs/android_arch_lifecycle_common',
>+              'version': 'version:1.0.0-cr0',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>+  'src/third_party/android_deps/libs/android_arch_lifecycle_runtime': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/android_deps/libs/android_arch_lifecycle_runtime',
>+              'version': 'version:1.0.0-cr0',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>+  'src/third_party/android_deps/libs/com_android_support_animated_vector_drawable': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/android_deps/libs/com_android_support_animated_vector_drawable',
>+              'version': 'version:27.0.0-cr0',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>+  'src/third_party/android_deps/libs/com_android_support_appcompat_v7': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/android_deps/libs/com_android_support_appcompat_v7',
>+              'version': 'version:27.0.0-cr0',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>+  'src/third_party/android_deps/libs/com_android_support_cardview_v7': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/android_deps/libs/com_android_support_cardview_v7',
>+              'version': 'version:27.0.0-cr0',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>+  'src/third_party/android_deps/libs/com_android_support_design': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/android_deps/libs/com_android_support_design',
>+              'version': 'version:27.0.0-cr0',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>+  'src/third_party/android_deps/libs/com_android_support_gridlayout_v7': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/android_deps/libs/com_android_support_gridlayout_v7',
>+              'version': 'version:27.0.0-cr0',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>+  'src/third_party/android_deps/libs/com_android_support_leanback_v17': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/android_deps/libs/com_android_support_leanback_v17',
>+              'version': 'version:27.0.0-cr0',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>+  'src/third_party/android_deps/libs/com_android_support_mediarouter_v7': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/android_deps/libs/com_android_support_mediarouter_v7',
>+              'version': 'version:27.0.0-cr0',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>+  'src/third_party/android_deps/libs/com_android_support_multidex': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/android_deps/libs/com_android_support_multidex',
>+              'version': 'version:1.0.0-cr0',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>+  'src/third_party/android_deps/libs/com_android_support_palette_v7': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/android_deps/libs/com_android_support_palette_v7',
>+              'version': 'version:27.0.0-cr0',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>+  'src/third_party/android_deps/libs/com_android_support_preference_leanback_v17': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/android_deps/libs/com_android_support_preference_leanback_v17',
>+              'version': 'version:27.0.0-cr0',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>+  'src/third_party/android_deps/libs/com_android_support_preference_v14': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/android_deps/libs/com_android_support_preference_v14',
>+              'version': 'version:27.0.0-cr0',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>+  'src/third_party/android_deps/libs/com_android_support_preference_v7': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/android_deps/libs/com_android_support_preference_v7',
>+              'version': 'version:27.0.0-cr0',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>+  'src/third_party/android_deps/libs/com_android_support_recyclerview_v7': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/android_deps/libs/com_android_support_recyclerview_v7',
>+              'version': 'version:27.0.0-cr0',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>+  'src/third_party/android_deps/libs/com_android_support_support_annotations': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/android_deps/libs/com_android_support_support_annotations',
>+              'version': 'version:27.0.0-cr0',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>+  'src/third_party/android_deps/libs/com_android_support_support_compat': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/android_deps/libs/com_android_support_support_compat',
>+              'version': 'version:27.0.0-cr0',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>+  'src/third_party/android_deps/libs/com_android_support_support_core_ui': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/android_deps/libs/com_android_support_support_core_ui',
>+              'version': 'version:27.0.0-cr0',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>+  'src/third_party/android_deps/libs/com_android_support_support_core_utils': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/android_deps/libs/com_android_support_support_core_utils',
>+              'version': 'version:27.0.0-cr0',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>+  'src/third_party/android_deps/libs/com_android_support_support_fragment': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/android_deps/libs/com_android_support_support_fragment',
>+              'version': 'version:27.0.0-cr0',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>+  'src/third_party/android_deps/libs/com_android_support_support_media_compat': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/android_deps/libs/com_android_support_support_media_compat',
>+              'version': 'version:27.0.0-cr0',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>+  'src/third_party/android_deps/libs/com_android_support_support_v13': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/android_deps/libs/com_android_support_support_v13',
>+              'version': 'version:27.0.0-cr0',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>+  'src/third_party/android_deps/libs/com_android_support_support_v4': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/android_deps/libs/com_android_support_support_v4',
>+              'version': 'version:27.0.0-cr0',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>+  'src/third_party/android_deps/libs/com_android_support_support_vector_drawable': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/android_deps/libs/com_android_support_support_vector_drawable',
>+              'version': 'version:27.0.0-cr0',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>+  'src/third_party/android_deps/libs/com_android_support_transition': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/android_deps/libs/com_android_support_transition',
>+              'version': 'version:27.0.0-cr0',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>+  'src/third_party/android_deps/libs/com_google_android_gms_play_services_base': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/android_deps/libs/com_google_android_gms_play_services_base',
>+              'version': 'version:12.0.1-cr0',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>+  'src/third_party/android_deps/libs/com_google_android_gms_play_services_auth_base': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/android_deps/libs/com_google_android_gms_play_services_auth_base',
>+              'version': 'version:12.0.1-cr0',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>+  'src/third_party/android_deps/libs/com_google_android_gms_play_services_base': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/android_deps/libs/com_google_android_gms_play_services_base',
>+              'version': 'version:12.0.1-cr0',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>+  'src/third_party/android_deps/libs/com_google_android_gms_play_services_basement': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/android_deps/libs/com_google_android_gms_play_services_basement',
>+              'version': 'version:12.0.1-cr0',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>+  'src/third_party/android_deps/libs/com_google_android_gms_play_services_tasks': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/android_deps/libs/com_google_android_gms_play_services_tasks',
>+              'version': 'version:12.0.1-cr0',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>+  'src/third_party/google-truth': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/google-truth',
>+              'version': 'version:0.40',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>+  # === ANDROID_DEPS End ===
> }
> 
> hooks = [
>@@ -188,23 +729,62 @@ hooks = [
>     ],
>   },
>   {
>-    # Downloads the current stable linux sysroot to build/linux/ if needed.
>-    # This sysroot updates at about the same rate that the chrome build deps
>-    # change. This script is a no-op except for linux users who are doing
>-    # official chrome builds or cross compiling.
>-    'name': 'sysroot',
>+    'name': 'sysroot_arm',
>+    'pattern': '.',
>+    'condition': 'checkout_linux and checkout_arm',
>+    'action': ['python', 'src/build/linux/sysroot_scripts/install-sysroot.py',
>+               '--arch=arm'],
>+  },
>+  {
>+    'name': 'sysroot_arm64',
>     'pattern': '.',
>-    'condition': 'checkout_linux',
>+    'condition': 'checkout_linux and checkout_arm64',
>     'action': ['python', 'src/build/linux/sysroot_scripts/install-sysroot.py',
>-               '--running-as-hook'],
>+               '--arch=arm64'],
>+  },
>+  {
>+    'name': 'sysroot_x86',
>+    'pattern': '.',
>+    'condition': 'checkout_linux and (checkout_x86 or checkout_x64)',
>+    # TODO(mbonadei): change to --arch=x86.
>+    'action': ['python', 'src/build/linux/sysroot_scripts/install-sysroot.py',
>+               '--arch=i386'],
>+  },
>+  {
>+    'name': 'sysroot_mips',
>+    'pattern': '.',
>+    'condition': 'checkout_linux and checkout_mips',
>+    # TODO(mbonadei): change to --arch=mips.
>+    'action': ['python', 'src/build/linux/sysroot_scripts/install-sysroot.py',
>+               '--arch=mipsel'],
>+  },
>+  {
>+    'name': 'sysroot_x64',
>+    'pattern': '.',
>+    'condition': 'checkout_linux and checkout_x64',
>+    # TODO(mbonadei): change to --arch=x64.
>+    'action': ['python', 'src/build/linux/sysroot_scripts/install-sysroot.py',
>+               '--arch=amd64'],
>+  },
>+  {
>+    # Case-insensitivity for the Win SDK. Must run before win_toolchain below.
>+    'name': 'ciopfs_linux',
>+    'pattern': '.',
>+    'condition': 'checkout_win and host_os == "linux"',
>+    'action': [ 'python',
>+                'src/third_party/depot_tools/download_from_google_storage.py',
>+                '--no_resume',
>+                '--no_auth',
>+                '--bucket', 'chromium-browser-clang/ciopfs',
>+                '-s', 'src/build/ciopfs.sha1',
>+    ]
>   },
>   {
>     # Update the Windows toolchain if necessary. Must run before 'clang' below.
>     'name': 'win_toolchain',
>     'pattern': '.',
>-    # TODO(thakis): Put some condition here. Not just host_os == 'win', because
>-    # we also need this for (mac|linux) -> win cross builds.
>-    'action': ['python', 'src/build/vs_toolchain.py', 'update'],
>+    'condition': 'checkout_win',
>+    'action': ['python', 'src/build/vs_toolchain.py', 'update', '--force'],
>   },
>   {
>     # Update the Mac toolchain if necessary.
>@@ -386,18 +966,6 @@ hooks = [
>                 '-d', 'src/tools/luci-go/linux64',
>     ],
>   },
>-  # Pull the Syzygy binaries, used for optimization and instrumentation.
>-  {
>-    'name': 'syzygy-binaries',
>-    'pattern': '.',
>-    'condition': 'host_os == "win"',
>-    'action': ['python',
>-               'src/build/get_syzygy_binaries.py',
>-               '--output-dir=src/third_party/syzygy/binaries',
>-               '--revision=a8456d9248a126881dcfb8707ca7dcdae56e1ac7',
>-               '--overwrite',
>-    ],
>-  },
>   {
>     'name': 'msan_chained_origins',
>     'pattern': '.',
>@@ -434,15 +1002,6 @@ hooks = [
>                '--bucket', 'chromium-webrtc-resources',
>                'src/resources'],
>   },
>-  {
>-    'name': 'Android CIPD Ensure',
>-    'pattern': '.',
>-    'condition': 'checkout_android',
>-    'action': ['src/build/cipd/cipd_wrapper.py',
>-               '--chromium-root', 'src',
>-               '--ensure-file', 'src/build/cipd/android/android.ensure',
>-    ],
>-  },
>   {
>     # This downloads SDK extras and puts them in the
>     # third_party/android_tools/sdk/extras directory.
>@@ -473,11 +1032,9 @@ include_rules = [
>   "-chromium",
>   "+external/webrtc/webrtc",  # Android platform build.
>   "+libyuv",
>-  # Individual headers that will be moved out of here, see webrtc:4243.
>-  "+call/rtp_config.h",
>+
>+  # These should eventually move out of here.
>   "+common_types.h",
>-  "+transport.h",
>-  "+typedefs.h",
> 
>   "+WebRTC",
>   "+api",
>@@ -485,14 +1042,10 @@ include_rules = [
>   "+rtc_base",
>   "+test",
>   "+rtc_tools",
>-]
> 
>-# The below rules will be removed when webrtc:4243 is fixed.
>-specific_include_rules = {
>-  "video_receive_stream\.h": [
>-    "+call/video_receive_stream.h",
>-  ],
>-  "video_send_stream\.h": [
>-    "+call/video_send_stream.h",
>-  ],
>-}
>+  # Abseil whitelist.
>+  "+absl/container/inlined_vector.h",
>+  "+absl/memory/memory.h",
>+  "+absl/types/optional.h",
>+  "+absl/types/variant.h",
>+]
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/LICENSE_THIRD_PARTY b/Source/ThirdParty/libwebrtc/Source/webrtc/LICENSE_THIRD_PARTY
>deleted file mode 100644
>index af1aa50d37e..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/LICENSE_THIRD_PARTY
>+++ /dev/null
>@@ -1,458 +0,0 @@
>-This source tree contains third party source code which is governed by third
>-party licenses. Paths to the files and associated licenses are collected here.
>-
>-Files governed by third party licenses:
>-base/base64.cc
>-base/base64.h
>-base/md5.cc
>-base/md5.h
>-base/sha1.cc
>-base/sha1.h
>-base/sigslot.cc
>-base/sigslot.h
>-common_audio/fft4g.c
>-common_audio/signal_processing/spl_sqrt_floor.c
>-common_audio/signal_processing/spl_sqrt_floor_arm.S
>-modules/audio_coding/codecs/g711/main/source/g711.c
>-modules/audio_coding/codecs/g711/main/source/g711.h
>-modules/audio_coding/codecs/g722/main/source/g722_decode.c
>-modules/audio_coding/codecs/g722/main/source/g722_enc_dec.h
>-modules/audio_coding/codecs/g722/main/source/g722_encode.c
>-modules/audio_coding/codecs/isac/main/source/fft.c
>-modules/audio_device/mac/portaudio/pa_memorybarrier.h
>-modules/audio_device/mac/portaudio/pa_ringbuffer.c
>-modules/audio_device/mac/portaudio/pa_ringbuffer.h
>-modules/audio_processing/aec/aec_rdft.c
>-system_wrappers/source/condition_variable_event_win.cc
>-system_wrappers/source/set_thread_name_win.h
>-
>-Individual licenses for each file:
>--------------------------------------------------------------------------------
>-Files:
>-base/base64.cc
>-base/base64.h
>-
>-License:
>-//*********************************************************************
>-//* Base64 - a simple base64 encoder and decoder.
>-//*
>-//*     Copyright (c) 1999, Bob Withers - bwit@pobox.com
>-//*
>-//* This code may be freely used for any purpose, either personal
>-//* or commercial, provided the authors copyright notice remains
>-//* intact.
>-//*
>-//* Enhancements by Stanley Yamane:
>-//*     o reverse lookup table for the decode function
>-//*     o reserve string buffer space in advance
>-//*
>-//*********************************************************************
>--------------------------------------------------------------------------------
>-Files:
>-base/md5.cc
>-base/md5.h
>-
>-License:
>-/*
>- * This code implements the MD5 message-digest algorithm.
>- * The algorithm is due to Ron Rivest.  This code was
>- * written by Colin Plumb in 1993, no copyright is claimed.
>- * This code is in the public domain; do with it what you wish.
>- *
>--------------------------------------------------------------------------------
>-Files:
>-base/sha1.cc
>-base/sha1.h
>-
>-License:
>-/*
>- * SHA-1 in C
>- * By Steve Reid <sreid@sea-to-sky.net>
>- * 100% Public Domain
>- *
>- * -----------------
>- * Modified 7/98
>- * By James H. Brown <jbrown@burgoyne.com>
>- * Still 100% Public Domain
>- *
>--------------------------------------------------------------------------------
>-Files:
>-base/sigslot.cc
>-base/sigslot.h
>-
>-License:
>-// sigslot.h: Signal/Slot classes
>-//
>-// Written by Sarah Thompson (sarah@telergy.com) 2002.
>-//
>-// License: Public domain. You are free to use this code however you like, with
>-// the proviso that the author takes on no responsibility or liability for any
>-// use.
>--------------------------------------------------------------------------------
>-Files:
>-common_audio/signal_processing/spl_sqrt_floor.c
>-common_audio/signal_processing/spl_sqrt_floor_arm.S
>-
>-License:
>-/*
>- * Written by Wilco Dijkstra, 1996. The following email exchange establishes the
>- * license.
>- *
>- * From: Wilco Dijkstra <Wilco.Dijkstra@ntlworld.com>
>- * Date: Fri, Jun 24, 2011 at 3:20 AM
>- * Subject: Re: sqrt routine
>- * To: Kevin Ma <kma@google.com>
>- * Hi Kevin,
>- * Thanks for asking. Those routines are public domain (originally posted to
>- * comp.sys.arm a long time ago), so you can use them freely for any purpose.
>- * Cheers,
>- * Wilco
>- *
>- * ----- Original Message -----
>- * From: "Kevin Ma" <kma@google.com>
>- * To: <Wilco.Dijkstra@ntlworld.com>
>- * Sent: Thursday, June 23, 2011 11:44 PM
>- * Subject: Fwd: sqrt routine
>- * Hi Wilco,
>- * I saw your sqrt routine from several web sites, including
>- * http://www.finesse.demon.co.uk/steven/sqrt.html.
>- * Just wonder if there's any copyright information with your Successive
>- * approximation routines, or if I can freely use it for any purpose.
>- * Thanks.
>- * Kevin
>- */
>--------------------------------------------------------------------------------
>-Files:
>-modules/audio_coding/codecs/g711/main/source/g711.c
>-modules/audio_coding/codecs/g711/main/source/g711.h
>-
>-License:
>-/*
>- * SpanDSP - a series of DSP components for telephony
>- *
>- * g711.h - In line A-law and u-law conversion routines
>- *
>- * Written by Steve Underwood <steveu@coppice.org>
>- *
>- * Copyright (C) 2001 Steve Underwood
>- *
>- *  Despite my general liking of the GPL, I place this code in the
>- *  public domain for the benefit of all mankind - even the slimy
>- *  ones who might try to proprietize my work and use it to my
>- *  detriment.
>- */
>--------------------------------------------------------------------------------
>-Files:
>-modules/audio_coding/codecs/g722/main/source/g722_decode.c
>-modules/audio_coding/codecs/g722/main/source/g722_enc_dec.h
>-modules/audio_coding/codecs/g722/main/source/g722_encode.c
>-
>-License:
>-/*
>- * SpanDSP - a series of DSP components for telephony
>- *
>- * g722_decode.c - The ITU G.722 codec, decode part.
>- *
>- * Written by Steve Underwood <steveu@coppice.org>
>- *
>- * Copyright (C) 2005 Steve Underwood
>- *
>- *  Despite my general liking of the GPL, I place my own contributions
>- *  to this code in the public domain for the benefit of all mankind -
>- *  even the slimy ones who might try to proprietize my work and use it
>- *  to my detriment.
>- *
>- * Based in part on a single channel G.722 codec which is:
>- *
>- * Copyright (c) CMU 1993
>- * Computer Science, Speech Group
>- * Chengxiang Lu and Alex Hauptmann
>- */
>--------------------------------------------------------------------------------
>-Files:
>-modules/audio_coding/codecs/isac/main/source/fft.c
>-
>-License:
>-/*
>- * Copyright(c)1995,97 Mark Olesen <olesen@me.QueensU.CA>
>- *    Queen's Univ at Kingston (Canada)
>- *
>- * Permission to use, copy, modify, and distribute this software for
>- * any purpose without fee is hereby granted, provided that this
>- * entire notice is included in all copies of any software which is
>- * or includes a copy or modification of this software and in all
>- * copies of the supporting documentation for such software.
>- *
>- * THIS SOFTWARE IS BEING PROVIDED "AS IS", WITHOUT ANY EXPRESS OR
>- * IMPLIED WARRANTY.  IN PARTICULAR, NEITHER THE AUTHOR NOR QUEEN'S
>- * UNIVERSITY AT KINGSTON MAKES ANY REPRESENTATION OR WARRANTY OF ANY
>- * KIND CONCERNING THE MERCHANTABILITY OF THIS SOFTWARE OR ITS
>- * FITNESS FOR ANY PARTICULAR PURPOSE.
>- *
>- * All of which is to say that you can do what you like with this
>- * source code provided you don't try to sell it as your own and you
>- * include an unaltered copy of this message (including the
>- * copyright).
>- *
>- * It is also implicitly understood that bug fixes and improvements
>- * should make their way back to the general Internet community so
>- * that everyone benefits.
>- */
>--------------------------------------------------------------------------------
>-Files:
>-modules/audio_device/mac/portaudio/pa_memorybarrier.h
>-modules/audio_device/mac/portaudio/pa_ringbuffer.c
>-modules/audio_device/mac/portaudio/pa_ringbuffer.h
>-
>-License:
>-/*
>- * $Id: pa_memorybarrier.h 1240 2007-07-17 13:05:07Z bjornroche $
>- * Portable Audio I/O Library
>- * Memory barrier utilities
>- *
>- * Author: Bjorn Roche, XO Audio, LLC
>- *
>- * This program uses the PortAudio Portable Audio Library.
>- * For more information see: http://www.portaudio.com
>- * Copyright (c) 1999-2000 Ross Bencina and Phil Burk
>- *
>- * Permission is hereby granted, free of charge, to any person obtaining
>- * a copy of this software and associated documentation files
>- * (the "Software"), to deal in the Software without restriction,
>- * including without limitation the rights to use, copy, modify, merge,
>- * publish, distribute, sublicense, and/or sell copies of the Software,
>- * and to permit persons to whom the Software is furnished to do so,
>- * subject to the following conditions:
>- *
>- * The above copyright notice and this permission notice shall be
>- * included in all copies or substantial portions of the Software.
>- *
>- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
>- * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
>- * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
>- * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
>- * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
>- * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
>- * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
>- */
>-
>-/*
>- * The text above constitutes the entire PortAudio license; however,
>- * the PortAudio community also makes the following non-binding requests:
>- *
>- * Any person wishing to distribute modifications to the Software is
>- * requested to send the modifications to the original developer so that
>- * they can be incorporated into the canonical version. It is also
>- * requested that these non-binding requests be included along with the
>- * license above.
>- */
>-
>-/*
>- * $Id: pa_ringbuffer.c 1421 2009-11-18 16:09:05Z bjornroche $
>- * Portable Audio I/O Library
>- * Ring Buffer utility.
>- *
>- * Author: Phil Burk, http://www.softsynth.com
>- * modified for SMP safety on Mac OS X by Bjorn Roche
>- * modified for SMP safety on Linux by Leland Lucius
>- * also, allowed for const where possible
>- * modified for multiple-byte-sized data elements by Sven Fischer
>- *
>- * Note that this is safe only for a single-thread reader and a
>- * single-thread writer.
>- *
>- * This program uses the PortAudio Portable Audio Library.
>- * For more information see: http://www.portaudio.com
>- * Copyright (c) 1999-2000 Ross Bencina and Phil Burk
>- *
>- * Permission is hereby granted, free of charge, to any person obtaining
>- * a copy of this software and associated documentation files
>- * (the "Software"), to deal in the Software without restriction,
>- * including without limitation the rights to use, copy, modify, merge,
>- * publish, distribute, sublicense, and/or sell copies of the Software,
>- * and to permit persons to whom the Software is furnished to do so,
>- * subject to the following conditions:
>- *
>- * The above copyright notice and this permission notice shall be
>- * included in all copies or substantial portions of the Software.
>- *
>- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
>- * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
>- * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
>- * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
>- * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
>- * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
>- * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
>- */
>-
>-/*
>- * The text above constitutes the entire PortAudio license; however,
>- * the PortAudio community also makes the following non-binding requests:
>- *
>- * Any person wishing to distribute modifications to the Software is
>- * requested to send the modifications to the original developer so that
>- * they can be incorporated into the canonical version. It is also
>- * requested that these non-binding requests be included along with the
>- * license above.
>- */
>--------------------------------------------------------------------------------
>-Files:
>-common_audio/fft4g.c
>-modules/audio_processing/aec/aec_rdft.c
>-
>-License:
>-/*
>- * http://www.kurims.kyoto-u.ac.jp/~ooura/fft.html
>- * Copyright Takuya OOURA, 1996-2001
>- *
>- * You may use, copy, modify and distribute this code for any purpose (include
>- * commercial use) and without fee. Please refer to this package when you modify
>- * this code.
>- */
>--------------------------------------------------------------------------------
>-Files:
>-system_wrappers/source/condition_variable_event_win.cc
>-
>-Source:
>-http://www1.cse.wustl.edu/~schmidt/ACE-copying.html
>-
>-License:
>-Copyright and Licensing Information for ACE(TM), TAO(TM), CIAO(TM), DAnCE(TM),
>-and CoSMIC(TM)
>-
>-ACE(TM), TAO(TM), CIAO(TM), DAnCE>(TM), and CoSMIC(TM) (henceforth referred to
>-as "DOC software") are copyrighted by Douglas C. Schmidt and his research
>-group at Washington University, University of California, Irvine, and
>-Vanderbilt University, Copyright (c) 1993-2009, all rights reserved. Since DOC
>-software is open-source, freely available software, you are free to use,
>-modify, copy, and distribute--perpetually and irrevocably--the DOC software
>-source code and object code produced from the source, as well as copy and
>-distribute modified versions of this software. You must, however, include this
>-copyright statement along with any code built using DOC software that you
>-release. No copyright statement needs to be provided if you just ship binary
>-executables of your software products.
>-You can use DOC software in commercial and/or binary software releases and are
>-under no obligation to redistribute any of your source code that is built
>-using DOC software. Note, however, that you may not misappropriate the DOC
>-software code, such as copyrighting it yourself or claiming authorship of the
>-DOC software code, in a way that will prevent DOC software from being
>-distributed freely using an open-source development model. You needn't inform
>-anyone that you're using DOC software in your software, though we encourage
>-you to let us know so we can promote your project in the DOC software success
>-stories.
>-
>-The ACE, TAO, CIAO, DAnCE, and CoSMIC web sites are maintained by the DOC
>-Group at the Institute for Software Integrated Systems (ISIS) and the Center
>-for Distributed Object Computing of Washington University, St. Louis for the
>-development of open-source software as part of the open-source software
>-community. Submissions are provided by the submitter ``as is'' with no
>-warranties whatsoever, including any warranty of merchantability,
>-noninfringement of third party intellectual property, or fitness for any
>-particular purpose. In no event shall the submitter be liable for any direct,
>-indirect, special, exemplary, punitive, or consequential damages, including
>-without limitation, lost profits, even if advised of the possibility of such
>-damages. Likewise, DOC software is provided as is with no warranties of any
>-kind, including the warranties of design, merchantability, and fitness for a
>-particular purpose, noninfringement, or arising from a course of dealing,
>-usage or trade practice. Washington University, UC Irvine, Vanderbilt
>-University, their employees, and students shall have no liability with respect
>-to the infringement of copyrights, trade secrets or any patents by DOC
>-software or any part thereof. Moreover, in no event will Washington
>-University, UC Irvine, or Vanderbilt University, their employees, or students
>-be liable for any lost revenue or profits or other special, indirect and
>-consequential damages.
>-
>-DOC software is provided with no support and without any obligation on the
>-part of Washington University, UC Irvine, Vanderbilt University, their
>-employees, or students to assist in its use, correction, modification, or
>-enhancement. A number of companies around the world provide commercial support
>-for DOC software, however. DOC software is Y2K-compliant, as long as the
>-underlying OS platform is Y2K-compliant. Likewise, DOC software is compliant
>-with the new US daylight savings rule passed by Congress as "The Energy Policy
>-Act of 2005," which established new daylight savings times (DST) rules for the
>-United States that expand DST as of March 2007. Since DOC software obtains
>-time/date and calendaring information from operating systems users will not be
>-affected by the new DST rules as long as they upgrade their operating systems
>-accordingly.
>-
>-The names ACE(TM), TAO(TM), CIAO(TM), DAnCE(TM), CoSMIC(TM), Washington
>-University, UC Irvine, and Vanderbilt University, may not be used to endorse
>-or promote products or services derived from this source without express
>-written permission from Washington University, UC Irvine, or Vanderbilt
>-University. This license grants no permission to call products or services
>-derived from this source ACE(TM), TAO(TM), CIAO(TM), DAnCE(TM), or CoSMIC(TM),
>-nor does it grant permission for the name Washington University, UC Irvine, or
>-Vanderbilt University to appear in their names.
>--------------------------------------------------------------------------------
>-Files:
>-system_wrappers/source/set_thread_name_win.h
>-
>-Source:
>-http://msdn.microsoft.com/en-us/cc300389.aspx#P
>-
>-License:
>-This license governs use of code marked as âsampleâ or âexampleâ available on
>-this web site without a license agreement, as provided under the section above
>-titled âNOTICE SPECIFIC TO SOFTWARE AVAILABLE ON THIS WEB SITE.â If you use
>-such code (the âsoftwareâ), you accept this license. If you do not accept the
>-license, do not use the software.
>-
>-1. Definitions
>-
>-The terms âreproduce,â âreproduction,â âderivative works,â and âdistributionâ
>-have the same meaning here as under U.S. copyright law.
>-
>-A âcontributionâ is the original software, or any additions or changes to the
>-software.
>-
>-A âcontributorâ is any person that distributes its contribution under this
>-license.
>-
>-âLicensed patentsâ are a contributorâs patent claims that read directly on its
>-contribution.
>-
>-2. Grant of Rights
>-
>-(A) Copyright Grant - Subject to the terms of this license, including the
>-license conditions and limitations in section 3, each contributor grants you a
>-non-exclusive, worldwide, royalty-free copyright license to reproduce its
>-contribution, prepare derivative works of its contribution, and distribute its
>-contribution or any derivative works that you create.
>-
>-(B) Patent Grant - Subject to the terms of this license, including the license
>-conditions and limitations in section 3, each contributor grants you a
>-non-exclusive, worldwide, royalty-free license under its licensed patents to
>-make, have made, use, sell, offer for sale, import, and/or otherwise dispose
>-of its contribution in the software or derivative works of the contribution in
>-the software.
>-
>-3. Conditions and Limitations
>-
>-(A) No Trademark License- This license does not grant you rights to use any
>-contributorsâ name, logo, or trademarks.
>-
>-(B) If you bring a patent claim against any contributor over patents that you
>-claim are infringed by the software, your patent license from such contributor
>-to the software ends automatically.
>-
>-(C) If you distribute any portion of the software, you must retain all
>-copyright, patent, trademark, and attribution notices that are present in the
>-software.
>-
>-(D) If you distribute any portion of the software in source code form, you may
>-do so only under this license by including a complete copy of this license
>-with your distribution. If you distribute any portion of the software in
>-compiled or object code form, you may only do so under a license that complies
>-with this license.
>-
>-(E) The software is licensed âas-is.â You bear the risk of using it. The
>-contributors give no express warranties, guarantees or conditions. You may
>-have additional consumer rights under your local laws which this license
>-cannot change. To the extent permitted under your local laws, the contributors
>-exclude the implied warranties of merchantability, fitness for a particular
>-purpose and non-infringement.
>-
>-(F) Platform Limitation - The licenses granted in sections 2(A) and 2(B)
>-extend only to the software or derivative works that you create that run on a
>-Microsoft Windows operating system product.
>-
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/OWNERS
>index f57794a473f..587f727e4d6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/OWNERS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/OWNERS
>@@ -2,16 +2,22 @@ henrika@webrtc.org
> kwiberg@webrtc.org
> mflodman@webrtc.org
> niklas.enbom@webrtc.org
>+phoglund@webrtc.org
> stefan@webrtc.org
> tina.legrand@webrtc.org
> tommi@webrtc.org
> per-file .gitignore=*
>-per-file .gn=phoglund@webrtc.org
>-per-file *.gn=phoglund@webrtc.org
>-per-file *.gni=phoglund@webrtc.org
>+per-file .gn=mbonadei@webrtc.org
>+per-file *.gn=mbonadei@webrtc.org
>+per-file *.gni=mbonadei@webrtc.org
> per-file *.py=phoglund@webrtc.org
> per-file AUTHORS=*
> per-file BUILD.gn=phoglund@webrtc.org
> per-file DEPS=*
> per-file pylintrc=phoglund@webrtc.org
>+per-file THIRD_PARTY_DEPS=phoglund@webrtc.org
>+per-file THIRD_PARTY_DEPS=titovartem@webrtc.org
> per-file WATCHLISTS=*
>+per-file style-guide.md=danilchap@webrtc.org
>+per-file style-guide.md=kwiberg@webrtc.org
>+per-file native-api.md=kwiberg@webrtc.org
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/PRESUBMIT.py b/Source/ThirdParty/libwebrtc/Source/webrtc/PRESUBMIT.py
>old mode 100644
>new mode 100755
>index a621f3b058a..1b435a77e87
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/PRESUBMIT.py
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/PRESUBMIT.py
>@@ -13,7 +13,6 @@ import sys
> from collections import defaultdict
> from contextlib import contextmanager
> 
>-
> # Files and directories that are *skipped* by cpplint in the presubmit script.
> CPPLINT_BLACKLIST = [
>   'api/video_codecs/video_decoder.h',
>@@ -29,11 +28,8 @@ CPPLINT_BLACKLIST = [
>   'modules/audio_processing',
>   'modules/desktop_capture',
>   'modules/include/module_common_types.h',
>-  'modules/media_file',
>   'modules/utility',
>   'modules/video_capture',
>-  'p2p/base/session.cc',
>-  'p2p/base/session.h',
>   'p2p/base/pseudotcp.cc',
>   'p2p/base/pseudotcp.h',
>   'rtc_base',
>@@ -142,6 +138,7 @@ def VerifyNativeApiHeadersListIsValid(input_api, output_api):
>         non_existing_paths)]
>   return []
> 
>+
> API_CHANGE_MSG = """
> You seem to be changing native API header files. Please make sure that you:
>   1. Make compatible changes that don't break existing clients. Usually
>@@ -161,6 +158,7 @@ You seem to be changing native API header files. Please make sure that you:
> Related files:
> """
> 
>+
> def CheckNativeApiHeaderChanges(input_api, output_api):
>   """Checks to remind proper changing of native APIs."""
>   files = []
>@@ -183,12 +181,15 @@ def CheckNativeApiHeaderChanges(input_api, output_api):
>   return []
> 
> 
>-def CheckNoIOStreamInHeaders(input_api, output_api):
>+def CheckNoIOStreamInHeaders(input_api, output_api,
>+                             source_file_filter):
>   """Checks to make sure no .h files include <iostream>."""
>   files = []
>   pattern = input_api.re.compile(r'^#include\s*<iostream>',
>                                  input_api.re.MULTILINE)
>-  for f in input_api.AffectedSourceFiles(input_api.FilterSourceFile):
>+  file_filter = lambda x: (input_api.FilterSourceFile(x)
>+                           and source_file_filter(x))
>+  for f in input_api.AffectedSourceFiles(file_filter):
>     if not f.LocalPath().endswith('.h'):
>       continue
>     contents = input_api.ReadFile(f)
>@@ -204,12 +205,15 @@ def CheckNoIOStreamInHeaders(input_api, output_api):
>   return []
> 
> 
>-def CheckNoPragmaOnce(input_api, output_api):
>+def CheckNoPragmaOnce(input_api, output_api,
>+                      source_file_filter):
>   """Make sure that banned functions are not used."""
>   files = []
>   pattern = input_api.re.compile(r'^#pragma\s+once',
>                                  input_api.re.MULTILINE)
>-  for f in input_api.AffectedSourceFiles(input_api.FilterSourceFile):
>+  file_filter = lambda x: (input_api.FilterSourceFile(x)
>+                           and source_file_filter(x))
>+  for f in input_api.AffectedSourceFiles(file_filter):
>     if not f.LocalPath().endswith('.h'):
>       continue
>     contents = input_api.ReadFile(f)
>@@ -224,13 +228,15 @@ def CheckNoPragmaOnce(input_api, output_api):
>   return []
> 
> 
>-def CheckNoFRIEND_TEST(input_api, output_api):  # pylint: disable=invalid-name
>+def CheckNoFRIEND_TEST(input_api, output_api,  # pylint: disable=invalid-name
>+                       source_file_filter):
>   """Make sure that gtest's FRIEND_TEST() macro is not used, the
>   FRIEND_TEST_ALL_PREFIXES() macro from testsupport/gtest_prod_util.h should be
>   used instead since that allows for FLAKY_, FAILS_ and DISABLED_ prefixes."""
>   problems = []
> 
>-  file_filter = lambda f: f.LocalPath().endswith(('.cc', '.h'))
>+  file_filter = lambda f: (f.LocalPath().endswith(('.cc', '.h'))
>+                           and source_file_filter(f))
>   for f in input_api.AffectedFiles(file_filter=file_filter):
>     for line_num, line in f.ChangedContents():
>       if 'FRIEND_TEST(' in line:
>@@ -252,7 +258,7 @@ def IsLintBlacklisted(blacklist_paths, file_path):
> 
> 
> def CheckApprovedFilesLintClean(input_api, output_api,
>-                                 source_file_filter=None):
>+                                source_file_filter=None):
>   """Checks that all new or non-blacklisted .cc and .h files pass cpplint.py.
>   This check is based on CheckChangeLintsClean in
>   depot_tools/presubmit_canned_checks.py but has less filters and only checks
>@@ -282,7 +288,7 @@ def CheckApprovedFilesLintClean(input_api, output_api,
>   for f in input_api.AffectedSourceFiles(source_file_filter):
>     # Note that moved/renamed files also count as added.
>     if f.Action() == 'A' or not IsLintBlacklisted(blacklist_paths,
>-                                                   f.LocalPath()):
>+                                                  f.LocalPath()):
>       files.append(f.AbsoluteLocalPath())
> 
>   for file_name in files:
>@@ -297,6 +303,7 @@ def CheckApprovedFilesLintClean(input_api, output_api,
> 
>   return result
> 
>+
> def CheckNoSourcesAbove(input_api, gn_files, output_api):
>   # Disallow referencing source files with paths above the GN file location.
>   source_pattern = input_api.re.compile(r' +sources \+?= \[(.*?)\]',
>@@ -325,11 +332,13 @@ def CheckNoSourcesAbove(input_api, gn_files, output_api):
>         items=violating_gn_files)]
>   return []
> 
>+
> def CheckNoMixingSources(input_api, gn_files, output_api):
>   """Disallow mixing C, C++ and Obj-C/Obj-C++ in the same target.
> 
>   See bugs.webrtc.org/7743 for more context.
>   """
>+
>   def _MoreThanOneSourceUsed(*sources_lists):
>     sources_used = 0
>     for source_list in sources_lists:
>@@ -391,6 +400,7 @@ def CheckNoMixingSources(input_api, gn_files, output_api):
>                                        '\n'.join(errors.keys())))]
>   return []
> 
>+
> def CheckNoPackageBoundaryViolations(input_api, gn_files, output_api):
>   cwd = input_api.PresubmitLocalPath()
>   with _AddToPath(input_api.os_path.join(
>@@ -404,19 +414,98 @@ def CheckNoPackageBoundaryViolations(input_api, gn_files, output_api):
>         long_text='\n\n'.join(str(err) for err in errors))]
>   return []
> 
>-def CheckPublicDepsIsNotUsed(gn_files, output_api):
>+
>+def _ReportFileAndLine(filename, line_num):
>+  """Default error formatter for _FindNewViolationsOfRule."""
>+  return '%s (line %s)' % (filename, line_num)
>+
>+
>+def CheckNoWarningSuppressionFlagsAreAdded(gn_files, input_api, output_api,
>+                                           error_formatter=_ReportFileAndLine):
>+  """Make sure that warning suppression flags are not added wihtout a reason."""
>+  msg = ('Usage of //build/config/clang:extra_warnings is discouraged '
>+         'in WebRTC.\n'
>+         'If you are not adding this code (e.g. you are just moving '
>+         'existing code) or you want to add an exception,\n'
>+         'you can add a comment on the line that causes the problem:\n\n'
>+         '"-Wno-odr"  # no-presubmit-check TODO(bugs.webrtc.org/BUG_ID)\n'
>+         '\n'
>+         'Affected files:\n')
>+  errors = []  # 2-element tuples with (file, line number)
>+  clang_warn_re = input_api.re.compile(r'//build/config/clang:extra_warnings')
>+  no_presubmit_re = input_api.re.compile(
>+      r'# no-presubmit-check TODO\(bugs\.webrtc\.org/\d+\)')
>+  for f in gn_files:
>+    for line_num, line in f.ChangedContents():
>+      if clang_warn_re.search(line) and not no_presubmit_re.search(line):
>+        errors.append(error_formatter(f.LocalPath(), line_num))
>+  if errors:
>+    return [output_api.PresubmitError(msg, errors)]
>+  return []
>+
>+def CheckNoStreamUsageIsAdded(input_api, output_api,
>+                              source_file_filter,
>+                              error_formatter=_ReportFileAndLine):
>+  """Make sure that no more dependencies on stringstream are added."""
>+  error_msg = ('Usage of <sstream>, <istream> and <ostream> in WebRTC is '
>+               'deprecated.\n'
>+               'This includes the following types:\n'
>+               'std::istringstream, std::ostringstream, std::wistringstream, '
>+               'std::wostringstream,\n'
>+               'std::wstringstream, std::ostream, std::wostream, std::istream,'
>+               'std::wistream,\n'
>+               'std::iostream, std::wiostream.\n'
>+               'If you are not adding this code (e.g. you are just moving '
>+               'existing code),\n'
>+               'you can add a comment on the line that causes the problem:\n\n'
>+               '#include <sstream>  // no-presubmit-check TODO(webrtc:8982)\n'
>+               'std::ostream& F() {  // no-presubmit-check TODO(webrtc:8982)\n'
>+               '\n'
>+               'If you are adding new code, consider using '
>+               'rtc::SimpleStringBuilder\n'
>+               '(in rtc_base/strings/string_builder.h).\n'
>+               'Affected files:\n')
>+  errors = []  # 2-element tuples with (file, line number)
>+  include_re = input_api.re.compile(r'#include <(i|o|s)stream>')
>+  usage_re = input_api.re.compile(r'std::(w|i|o|io|wi|wo|wio)(string)*stream')
>+  no_presubmit_re = input_api.re.compile(
>+      r'// no-presubmit-check TODO\(webrtc:8982\)')
>+  file_filter = lambda x: (input_api.FilterSourceFile(x)
>+                           and source_file_filter(x))
>+  for f in input_api.AffectedSourceFiles(file_filter):
>+    if f.LocalPath() == 'PRESUBMIT.py':
>+      continue
>+    for line_num, line in f.ChangedContents():
>+      if ((include_re.search(line) or usage_re.search(line))
>+          and not no_presubmit_re.search(line)):
>+        errors.append(error_formatter(f.LocalPath(), line_num))
>+  if errors:
>+    return [output_api.PresubmitError(error_msg, errors)]
>+  return []
>+
>+
>+def CheckPublicDepsIsNotUsed(gn_files, input_api, output_api):
>+  """Checks that public_deps is not used without a good reason."""
>   result = []
>-  error_msg = ('public_deps is not allowed in WebRTC BUILD.gn files because '
>-               'it doesn\'t map well to downstream build systems.\n'
>-               'Used in: %s (line %d).')
>+  no_presubmit_check_re = input_api.re.compile(
>+      r'# no-presubmit-check TODO\(webrtc:8603\)')
>+  error_msg = ('public_deps is not recommended in WebRTC BUILD.gn files '
>+               'because it doesn\'t map well to downstream build systems.\n'
>+               'Used in: %s (line %d).\n'
>+               'If you are not adding this code (e.g. you are just moving '
>+               'existing code) or you have a good reason, you can add a '
>+               'comment on the line that causes the problem:\n\n'
>+               'public_deps = [  # no-presubmit-check TODO(webrtc:8603)\n')
>   for affected_file in gn_files:
>     for (line_number, affected_line) in affected_file.ChangedContents():
>-      if 'public_deps' in affected_line:
>+      if ('public_deps' in affected_line
>+          and not no_presubmit_check_re.search(affected_line)):
>         result.append(
>             output_api.PresubmitError(error_msg % (affected_file.LocalPath(),
>                                                    line_number)))
>   return result
> 
>+
> def CheckCheckIncludesIsNotUsed(gn_files, output_api):
>   result = []
>   error_msg = ('check_includes overrides are not allowed since it can cause '
>@@ -434,13 +523,14 @@ def CheckCheckIncludesIsNotUsed(gn_files, output_api):
>                                                    line_number)))
>   return result
> 
>+
> def CheckGnChanges(input_api, output_api):
>-  source_file_filter = lambda x: input_api.FilterSourceFile(
>+  file_filter = lambda x: (input_api.FilterSourceFile(
>       x, white_list=(r'.+\.(gn|gni)$',),
>-      black_list=(r'.*/presubmit_checks_lib/testdata/.*',))
>+      black_list=(r'.*/presubmit_checks_lib/testdata/.*',)))
> 
>   gn_files = []
>-  for f in input_api.AffectedSourceFiles(source_file_filter):
>+  for f in input_api.AffectedSourceFiles(file_filter):
>     gn_files.append(f)
> 
>   result = []
>@@ -449,10 +539,13 @@ def CheckGnChanges(input_api, output_api):
>     result.extend(CheckNoMixingSources(input_api, gn_files, output_api))
>     result.extend(CheckNoPackageBoundaryViolations(input_api, gn_files,
>                                                    output_api))
>-    result.extend(CheckPublicDepsIsNotUsed(gn_files, output_api))
>+    result.extend(CheckPublicDepsIsNotUsed(gn_files, input_api, output_api))
>     result.extend(CheckCheckIncludesIsNotUsed(gn_files, output_api))
>+    result.extend(CheckNoWarningSuppressionFlagsAreAdded(gn_files, input_api,
>+                                                          output_api))
>   return result
> 
>+
> def CheckGnGen(input_api, output_api):
>   """Runs `gn gen --check` with default args to detect mismatches between
>   #includes and dependencies in the BUILD.gn files, as well as general build
>@@ -469,7 +562,8 @@ def CheckGnGen(input_api, output_api):
>         long_text='\n\n'.join(errors))]
>   return []
> 
>-def CheckUnwantedDependencies(input_api, output_api):
>+
>+def CheckUnwantedDependencies(input_api, output_api, source_file_filter):
>   """Runs checkdeps on #include statements added in this
>   change. Breaking - rules is an error, breaking ! rules is a
>   warning.
>@@ -491,7 +585,7 @@ def CheckUnwantedDependencies(input_api, output_api):
>     from rules import Rule
> 
>   added_includes = []
>-  for f in input_api.AffectedFiles():
>+  for f in input_api.AffectedFiles(file_filter=source_file_filter):
>     if not CppChecker.IsCppFile(f.LocalPath()):
>       continue
> 
>@@ -528,6 +622,7 @@ def CheckUnwantedDependencies(input_api, output_api):
>         warning_descriptions))
>   return results
> 
>+
> def CheckCommitMessageBugEntry(input_api, output_api):
>   """Check that bug entries are well-formed in commit message."""
>   bogus_bug_msg = (
>@@ -553,6 +648,7 @@ def CheckCommitMessageBugEntry(input_api, output_api):
>       results.append(bogus_bug_msg % bug)
>   return [output_api.PresubmitError(r) for r in results]
> 
>+
> def CheckChangeHasBugField(input_api, output_api):
>   """Requires that the changelist is associated with a bug.
> 
>@@ -572,11 +668,13 @@ def CheckChangeHasBugField(input_api, output_api):
>         ' * https://bugs.webrtc.org - reference it using Bug: webrtc:XXXX\n'
>         ' * https://crbug.com - reference it using Bug: chromium:XXXXXX')]
> 
>-def CheckJSONParseErrors(input_api, output_api):
>+
>+def CheckJSONParseErrors(input_api, output_api, source_file_filter):
>   """Check that JSON files do not contain syntax errors."""
> 
>   def FilterFile(affected_file):
>-    return input_api.os_path.splitext(affected_file.LocalPath())[1] == '.json'
>+    return (input_api.os_path.splitext(affected_file.LocalPath())[1] == '.json'
>+            and source_file_filter(affected_file))
> 
>   def GetJSONParseError(input_api, filename):
>     try:
>@@ -593,7 +691,8 @@ def CheckJSONParseErrors(input_api, output_api):
>                                     affected_file.AbsoluteLocalPath())
>     if parse_error:
>       results.append(output_api.PresubmitError('%s could not be parsed: %s' %
>-          (affected_file.LocalPath(), parse_error)))
>+                                               (affected_file.LocalPath(),
>+                                                parse_error)))
>   return results
> 
> 
>@@ -622,12 +721,15 @@ def RunPythonTests(input_api, output_api):
>   return input_api.RunTests(tests, parallel=True)
> 
> 
>-def CheckUsageOfGoogleProtobufNamespace(input_api, output_api):
>+def CheckUsageOfGoogleProtobufNamespace(input_api, output_api,
>+                                        source_file_filter):
>   """Checks that the namespace google::protobuf has not been used."""
>   files = []
>   pattern = input_api.re.compile(r'google::protobuf')
>   proto_utils_path = os.path.join('rtc_base', 'protobuf_utils.h')
>-  for f in input_api.AffectedSourceFiles(input_api.FilterSourceFile):
>+  file_filter = lambda x: (input_api.FilterSourceFile(x)
>+                           and source_file_filter(x))
>+  for f in input_api.AffectedSourceFiles(file_filter):
>     if f.LocalPath() in [proto_utils_path, 'PRESUBMIT.py']:
>       continue
>     contents = input_api.ReadFile(f)
>@@ -690,7 +792,6 @@ def CommonChecks(input_api, output_api):
>                   r'^tools[\\\/].*\.py$',
>                   # TODO(phoglund): should arguably be checked.
>                   r'^tools_webrtc[\\\/]mb[\\\/].*\.py$',
>-                  r'^tools_webrtc[\\\/]valgrind[\\\/].*\.py$',
>                   r'^xcodebuild.*[\\\/].*\.py$',),
>       pylintrc='pylintrc'))
> 
>@@ -704,35 +805,52 @@ def CommonChecks(input_api, output_api):
>   objc_filter_list = (r'.+\.m$', r'.+\.mm$', r'.+objc\/.+\.h$')
>   # Skip long-lines check for DEPS and GN files.
>   build_file_filter_list = (r'.+\.gn$', r'.+\.gni$', 'DEPS')
>+  # Also we will skip most checks for third_party directory.
>+  third_party_filter_list = (r'^third_party[\\\/].+',)
>   eighty_char_sources = lambda x: input_api.FilterSourceFile(x,
>-      black_list=build_file_filter_list + objc_filter_list)
>+      black_list=build_file_filter_list + objc_filter_list +
>+                 third_party_filter_list)
>   hundred_char_sources = lambda x: input_api.FilterSourceFile(x,
>       white_list=objc_filter_list)
>+  non_third_party_sources = lambda x: input_api.FilterSourceFile(x,
>+      black_list=third_party_filter_list)
>+
>   results.extend(input_api.canned_checks.CheckLongLines(
>       input_api, output_api, maxlen=80, source_file_filter=eighty_char_sources))
>   results.extend(input_api.canned_checks.CheckLongLines(
>       input_api, output_api, maxlen=100,
>       source_file_filter=hundred_char_sources))
>-
>   results.extend(input_api.canned_checks.CheckChangeHasNoTabs(
>-      input_api, output_api))
>+      input_api, output_api, source_file_filter=non_third_party_sources))
>   results.extend(input_api.canned_checks.CheckChangeHasNoStrayWhitespace(
>-      input_api, output_api))
>+      input_api, output_api, source_file_filter=non_third_party_sources))
>   results.extend(input_api.canned_checks.CheckAuthorizedAuthor(
>       input_api, output_api))
>   results.extend(input_api.canned_checks.CheckChangeTodoHasOwner(
>+      input_api, output_api, source_file_filter=non_third_party_sources))
>+  results.extend(input_api.canned_checks.CheckPatchFormatted(
>       input_api, output_api))
>   results.extend(CheckNativeApiHeaderChanges(input_api, output_api))
>-  results.extend(CheckNoIOStreamInHeaders(input_api, output_api))
>-  results.extend(CheckNoPragmaOnce(input_api, output_api))
>-  results.extend(CheckNoFRIEND_TEST(input_api, output_api))
>+  results.extend(CheckNoIOStreamInHeaders(
>+      input_api, output_api, source_file_filter=non_third_party_sources))
>+  results.extend(CheckNoPragmaOnce(
>+      input_api, output_api, source_file_filter=non_third_party_sources))
>+  results.extend(CheckNoFRIEND_TEST(
>+      input_api, output_api, source_file_filter=non_third_party_sources))
>   results.extend(CheckGnChanges(input_api, output_api))
>-  results.extend(CheckUnwantedDependencies(input_api, output_api))
>-  results.extend(CheckJSONParseErrors(input_api, output_api))
>+  results.extend(CheckUnwantedDependencies(
>+      input_api, output_api, source_file_filter=non_third_party_sources))
>+  results.extend(CheckJSONParseErrors(
>+      input_api, output_api, source_file_filter=non_third_party_sources))
>   results.extend(RunPythonTests(input_api, output_api))
>-  results.extend(CheckUsageOfGoogleProtobufNamespace(input_api, output_api))
>-  results.extend(CheckOrphanHeaders(input_api, output_api))
>-  results.extend(CheckNewlineAtTheEndOfProtoFiles(input_api, output_api))
>+  results.extend(CheckUsageOfGoogleProtobufNamespace(
>+      input_api, output_api, source_file_filter=non_third_party_sources))
>+  results.extend(CheckOrphanHeaders(
>+      input_api, output_api, source_file_filter=non_third_party_sources))
>+  results.extend(CheckNewlineAtTheEndOfProtoFiles(
>+      input_api, output_api, source_file_filter=non_third_party_sources))
>+  results.extend(CheckNoStreamUsageIsAdded(
>+      input_api, output_api, non_third_party_sources))
>   return results
> 
> 
>@@ -762,7 +880,7 @@ def CheckChangeOnCommit(input_api, output_api):
>   return results
> 
> 
>-def CheckOrphanHeaders(input_api, output_api):
>+def CheckOrphanHeaders(input_api, output_api, source_file_filter):
>   # We need to wait until we have an input_api object and use this
>   # roundabout construct to import prebubmit_checks_lib because this file is
>   # eval-ed and thus doesn't have __file__.
>@@ -776,9 +894,9 @@ def CheckOrphanHeaders(input_api, output_api):
>     from check_orphan_headers import GetBuildGnPathFromFilePath
>     from check_orphan_headers import IsHeaderInBuildGn
> 
>-  source_file_filter = lambda x: input_api.FilterSourceFile(
>-      x, black_list=orphan_blacklist)
>-  for f in input_api.AffectedSourceFiles(source_file_filter):
>+  file_filter = lambda x: input_api.FilterSourceFile(
>+      x, black_list=orphan_blacklist) and source_file_filter(x)
>+  for f in input_api.AffectedSourceFiles(file_filter):
>     if f.LocalPath().endswith('.h'):
>       file_path = os.path.abspath(f.LocalPath())
>       root_dir = os.getcwd()
>@@ -791,13 +909,13 @@ def CheckOrphanHeaders(input_api, output_api):
>   return results
> 
> 
>-def CheckNewlineAtTheEndOfProtoFiles(input_api, output_api):
>+def CheckNewlineAtTheEndOfProtoFiles(input_api, output_api, source_file_filter):
>   """Checks that all .proto files are terminated with a newline."""
>   error_msg = 'File {} must end with exactly one newline.'
>   results = []
>-  source_file_filter = lambda x: input_api.FilterSourceFile(
>-      x, white_list=(r'.+\.proto$',))
>-  for f in input_api.AffectedSourceFiles(source_file_filter):
>+  file_filter = lambda x: input_api.FilterSourceFile(
>+      x, white_list=(r'.+\.proto$',)) and source_file_filter(x)
>+  for f in input_api.AffectedSourceFiles(file_filter):
>     file_path = f.LocalPath()
>     with open(file_path) as f:
>       lines = f.readlines()
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/WATCHLISTS b/Source/ThirdParty/libwebrtc/Source/webrtc/WATCHLISTS
>index ed0a614009b..845ff3c1298 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/WATCHLISTS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/WATCHLISTS
>@@ -16,90 +16,80 @@
>     },
>     'all_webrtc': {
>       # NOTE: if you like this you might like webrtc-reviews@webrtc.org!
>-      'filepath': '^webrtc/.*',
>+      'filepath': '^.*',
>     },
>     'root_files': {
>-      # webrtc/build/ and non-recursive contents of ./ and webrtc/
>-      'filepath': '^[^/]*$|^webrtc/[^/]*$|^webrtc/build/.*',
>-    },
>-    'documented_interfaces': {
>-      'filepath': '^webrtc/[^/]*\.h$|'\
>-                  'webrtc/voice_engine/include/.*',
>+      # Non-recursive contents of ./
>+      'filepath': '^[^/]*$',
>     },
>     'build_files': {
>-      'filepath': '\.gyp$|\.gypi$|Android\.mk$',
>+      'filepath': '\.gni$|\.gn$',
>     },
>     'java_files': {
>       'filepath': '\.java$|\.xml$',
>     },
>     'api': {
>-      'filepath': 'webrtc/api/.*',
>+      'filepath': 'api/.*',
>     },
>     'audio': {
>-      'filepath': 'webrtc/audio/.*',
>+      'filepath': 'audio/.*',
>     },
>     'base': {
>-      'filepath': 'webrtc/base/.*',
>+      'filepath': 'rtc_base/.*',
>     },
>     'call': {
>-      'filepath': 'webrtc/call/.*',
>+      'filepath': 'call/.*',
>     },
>     'media_engine': {
>-      'filepath': 'webrtc/media/engine/.*',
>+      'filepath': 'media/engine/.*',
>     },
>     'video': {
>-      'filepath': 'webrtc/video/.*',
>-    },
>-    'voice_engine': {
>-      'filepath': 'webrtc/voice_engine/.*',
>+      'filepath': 'video/.*',
>     },
>     'common_audio': {
>-      'filepath': 'webrtc/common_audio/.*',
>+      'filepath': 'common_audio/.*',
>     },
>     'common_video': {
>-      'filepath': 'webrtc/common_video/.*',
>+      'filepath': 'common_video/.*',
>     },
>     'video_capture': {
>-      'filepath': 'webrtc/modules/video_capture/.*',
>-    },
>-    'video_render': {
>-      'filepath': 'webrtc/modules/video_render/.*',
>+      'filepath': 'modules/video_capture/.*',
>     },
>     'audio_device': {
>-      'filepath': 'webrtc/modules/audio_device/.*',
>+      'filepath': 'modules/audio_device/.*',
>     },
>     'audio_coding': {
>-      'filepath': 'webrtc/modules/audio_coding/.*',
>+      'filepath': 'modules/audio_coding/.*',
>     },
>     'neteq': {
>-      'filepath': 'webrtc/modules/audio_coding/neteq/.*',
>+      'filepath': 'modules/audio_coding/neteq/.*',
>     },
>     'audio_mixer': {
>-      'filepath': 'webrtc/modules/audio_mixer/.*',
>+      'filepath': 'modules/audio_mixer/.*',
>     },
>     'audio_processing': {
>-      'filepath': 'webrtc/modules/audio_processing/.*',
>+      'filepath': 'modules/audio_processing/.*',
>     },
>     'video_coding': {
>-      'filepath': 'webrtc/modules/video_coding/.*',
>+      'filepath': 'modules/video_coding/.*',
>     },
>     'video_processing': {
>-      'filepath': 'webrtc/modules/video_processing/.*',
>+      'filepath': 'modules/video_processing/.*',
>     },
>     'bitrate_controller': {
>-      'filepath': 'webrtc/modules/bitrate_controller/.*'
>+      'filepath': 'modules/bitrate_controller/.*'
>     },
>     'remote_bitrate_estimator': {
>-      'filepath': 'webrtc/modules/remote_bitrate_estimator/.*'
>+      'filepath': 'modules/remote_bitrate_estimator/.*'
>     },
>     'pacing': {
>-      'filepath': 'webrtc/modules/pacing/.*'
>+      'filepath': 'modules/pacing/.*'
>     },
>     'rtp_rtcp': {
>-      'filepath': 'webrtc/modules/rtp_rtcp/.*'
>+      'filepath': 'modules/rtp_rtcp/.*'
>     },
>     'system_wrappers': {
>-      'filepath': 'webrtc/system_wrappers/.*',
>+      'filepath': 'system_wrappers/.*',
>     },
>   },
> 
>@@ -110,7 +100,7 @@
>                    'peah@webrtc.org',
>                    'qiang.lu@intel.com',
>                    'yujie.mao@webrtc.org'],
>-    'documented_interfaces': ['interface-changes@webrtc.org'],
>+    'build_files': ['mbonadei@webrtc.org'],
>     'common_audio': ['alessiob@webrtc.org',
>                      'aluebs@webrtc.org',
>                      'andrew@webrtc.org',
>@@ -126,28 +116,15 @@
>     'call': ['mflodman@webrtc.org',
>              'solenberg@webrtc.org',
>              'stefan@webrtc.org'],
>-    'media_engine': ['solenberg@webrtc.org'],
>     'video': ['mflodman@webrtc.org',
>               'stefan@webrtc.org',
>               'video-team@agora.io',
>               'yujie.mao@webrtc.org',
>               'zhengzhonghou@agora.io'],
>-    'voice_engine': ['alessiob@webrtc.org',
>-                     'andrew@webrtc.org',
>-                     'audio-team@agora.io',
>-                     'henrika@webrtc.org',
>-                     'henrik.lundin@webrtc.org',
>-                     'minyue@webrtc.org',
>-                     'peah@webrtc.org',
>-                     'solenberg@webrtc.org'],
>     'video_capture': ['mflodman@webrtc.org',
>                       'perkj@webrtc.org',
>                       'sdk-team@agora.io',
>                       'zhengzhonghou@agora.io'],
>-    'video_render': ['mflodman@webrtc.org',
>-                     'perkj@webrtc.org',
>-                     'sdk-team@agora.io',
>-                     'zhengzhonghou@agora.io'],
>     'audio_device': ['audio-team@agora.io',
>                      'henrika@webrtc.org',
>                      'peah@webrtc.org',
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/api/BUILD.gn
>index e3af7bbf2f1..7271ece848a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/BUILD.gn
>@@ -29,25 +29,36 @@ rtc_source_set("call_api") {
> 
>   deps = [
>     # TODO(kjellander): Add remaining dependencies when webrtc:4243 is done.
>-    ":audio_mixer_api",
>     ":transport_api",
>     "..:webrtc_common",
>     "../rtc_base:rtc_base_approved",
>+    "audio:audio_mixer_api",
>     "audio_codecs:audio_codecs_api",
>   ]
> }
> 
>+rtc_source_set("callfactory_api") {
>+  visibility = [ "*" ]
>+  sources = [
>+    "call/callfactoryinterface.h",
>+  ]
>+}
>+
> rtc_static_library("libjingle_peerconnection_api") {
>   visibility = [ "*" ]
>   cflags = []
>   sources = [
>+    "asyncresolverfactory.h",
>+    "bitrate_constraints.h",
>     "candidate.cc",
>     "candidate.h",
>     "cryptoparams.h",
>+    "datachannelinterface.cc",
>     "datachannelinterface.h",
>     "dtmfsenderinterface.h",
>     "jsep.cc",
>     "jsep.h",
>+    "jsepicecandidate.cc",
>     "jsepicecandidate.h",
>     "jsepsessiondescription.h",
>     "mediaconstraintsinterface.cc",
>@@ -60,6 +71,8 @@ rtc_static_library("libjingle_peerconnection_api") {
>     "mediatypes.h",
>     "notifier.h",
>     "peerconnectionfactoryproxy.h",
>+    "peerconnectioninterface.cc",
>+    "peerconnectioninterface.h",
>     "peerconnectionproxy.h",
>     "proxy.cc",
>     "proxy.h",
>@@ -69,39 +82,39 @@ rtc_static_library("libjingle_peerconnection_api") {
>     "rtp_headers.h",
>     "rtpparameters.cc",
>     "rtpparameters.h",
>+    "rtpreceiverinterface.cc",
>     "rtpreceiverinterface.h",
>     "rtpsenderinterface.h",
>+    "rtptransceiverinterface.cc",
>     "rtptransceiverinterface.h",
>     "setremotedescriptionobserverinterface.h",
>     "statstypes.cc",
>     "statstypes.h",
>     "turncustomizer.h",
>-    "umametrics.cc",
>     "umametrics.h",
>-    "videosinkinterface.h",
>-    "videosourceinterface.cc",
>-    "videosourceinterface.h",
>     "videosourceproxy.h",
>   ]
> 
>-  if (!build_with_chromium && is_clang) {
>-    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>-    suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-  }
>-
>   deps = [
>     ":array_view",
>-    ":optional",
>-    ":peerconnection_and_implicit_call_api",
>+    ":audio_options_api",
>+    ":callfactory_api",
>+    ":fec_controller_api",
>+    ":libjingle_logging_api",
>     ":rtc_stats_api",
>-    ":video_frame_api",
>+    "audio:audio_mixer_api",
>     "audio_codecs:audio_codecs_api",
>+    "transport:bitrate_settings",
>+    "transport:network_control",
>+    "video:video_frame",
>+    "//third_party/abseil-cpp/absl/types:optional",
> 
>     # Basically, don't add stuff here. You might break sensitive downstream
>     # targets like pnacl. API should not depend on anything outside of this
>     # file, really. All these should arguably go away in time.
>-    "..:typedefs",
>     "..:webrtc_common",
>+    "../logging:rtc_event_log_api",
>+    "../media:rtc_media_config",
>     "../modules/audio_processing:audio_processing_statistics",
>     "../rtc_base:checks",
>     "../rtc_base:deprecation",
>@@ -109,24 +122,54 @@ rtc_static_library("libjingle_peerconnection_api") {
>     "../rtc_base:rtc_base_approved",
>     "../rtc_base:stringutils",
>   ]
>+
>   if (is_nacl) {
>     # This is needed by .h files included from rtc_base.
>     deps += [ "//native_client_sdk/src/libraries/nacl_io" ]
>   }
> }
> 
>-rtc_source_set("peerconnection_and_implicit_call_api") {
>+rtc_source_set("video_quality_test_fixture_api") {
>   visibility = [ "*" ]
>-
>-  # The peerconnectioninterface.h file pulls in call/callfactoryinterface.h
>-  # and the entire call module with it. We need to either get rid of this
>-  # dependency or pull most of call/ into the API. For now, silence the warnings
>-  # this creates since it creates a circular dependency (call very much depends
>-  # on API). See bugs.webrtc.org/8667.
>-  check_includes = false
>+  testonly = true
>   sources = [
>-    "peerconnectioninterface.h",
>+    "test/video_quality_test_fixture.h",
>   ]
>+  deps = [
>+    ":libjingle_peerconnection_api",
>+    ":simulated_network_api",
>+    "../call:fake_network",
>+    "../call:rtp_interfaces",
>+    "../test:test_common",
>+    "../test:video_test_common",
>+    "video_codecs:video_codecs_api",
>+  ]
>+  if (!build_with_chromium && is_clang) {
>+    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>+    suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>+  }
>+}
>+
>+if (rtc_include_tests) {
>+  rtc_source_set("create_video_quality_test_fixture_api") {
>+    visibility = [ "*" ]
>+    testonly = true
>+    sources = [
>+      "test/create_video_quality_test_fixture.cc",
>+      "test/create_video_quality_test_fixture.h",
>+    ]
>+    deps = [
>+      ":fec_controller_api",
>+      ":video_quality_test_fixture_api",
>+      "../rtc_base:ptr_util",
>+      "../video:video_quality_test",
>+      "//third_party/abseil-cpp/absl/memory",
>+    ]
>+    if (!build_with_chromium && is_clang) {
>+      # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>+      suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>+    }
>+  }
> }
> 
> rtc_source_set("libjingle_logging_api") {
>@@ -159,9 +202,9 @@ rtc_source_set("ortc_api") {
>   # libjingle_peerconnection_api.
>   deps = [
>     ":libjingle_peerconnection_api",
>-    ":optional",
>     "..:webrtc_common",
>     "../rtc_base:rtc_base",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
>   if (!build_with_chromium && is_clang) {
>     # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>@@ -185,57 +228,48 @@ rtc_source_set("rtc_stats_api") {
>   ]
> }
> 
>-rtc_source_set("audio_mixer_api") {
>+rtc_source_set("audio_options_api") {
>   visibility = [ "*" ]
>   sources = [
>-    "audio/audio_mixer.h",
>+    "audio_options.cc",
>+    "audio_options.h",
>   ]
> 
>   deps = [
>-    "../modules:module_api",
>     "../rtc_base:rtc_base_approved",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
> 
> rtc_source_set("transport_api") {
>   visibility = [ "*" ]
>   sources = [
>+    "call/transport.cc",
>     "call/transport.h",
>   ]
> }
> 
>-rtc_source_set("video_frame_api") {
>+rtc_source_set("simulated_network_api") {
>   visibility = [ "*" ]
>   sources = [
>-    "video/video_content_type.cc",
>-    "video/video_content_type.h",
>-    "video/video_frame.cc",
>-    "video/video_frame.h",
>-    "video/video_frame_buffer.cc",
>-    "video/video_frame_buffer.h",
>-    "video/video_rotation.h",
>-    "video/video_timing.cc",
>-    "video/video_timing.h",
>+    "test/simulated_network.h",
>   ]
>-
>   deps = [
>-    "../rtc_base:checks",
>-    "../rtc_base:rtc_base_approved",
>+    "../rtc_base:criticalsection",
>+    "../rtc_base:rtc_base",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
> 
>-rtc_source_set("video_frame_api_i420") {
>+rtc_source_set("fec_controller_api") {
>   visibility = [ "*" ]
>   sources = [
>-    "video/i420_buffer.cc",
>-    "video/i420_buffer.h",
>+    "fec_controller.h",
>   ]
>+
>   deps = [
>-    ":video_frame_api",
>-    "../rtc_base:checks",
>-    "../rtc_base:rtc_base",
>-    "../system_wrappers",
>-    "//third_party/libyuv",
>+    "..:webrtc_common",
>+    "../modules:module_fec_api",
>   ]
> }
> 
>@@ -250,19 +284,6 @@ rtc_source_set("array_view") {
>   ]
> }
> 
>-rtc_source_set("optional") {
>-  visibility = [ "*" ]
>-  sources = [
>-    "optional.cc",
>-    "optional.h",
>-  ]
>-  deps = [
>-    ":array_view",
>-    "../rtc_base:checks",
>-    "../rtc_base:sanitizer",
>-  ]
>-}
>-
> rtc_source_set("refcountedbase") {
>   visibility = [ "*" ]
>   sources = [
>@@ -287,6 +308,86 @@ rtc_source_set("libjingle_peerconnection_test_api") {
> }
> 
> if (rtc_include_tests) {
>+  if (rtc_enable_protobuf) {
>+    rtc_source_set("audioproc_f_api") {
>+      visibility = [ "*" ]
>+      testonly = true
>+      sources = [
>+        "test/audioproc_float.cc",
>+        "test/audioproc_float.h",
>+      ]
>+
>+      deps = [
>+        "../modules/audio_processing:audio_processing",
>+        "../modules/audio_processing:audioproc_f_impl",
>+      ]
>+    }
>+  }
>+
>+  rtc_source_set("simulcast_test_fixture_api") {
>+    visibility = [ "*" ]
>+    testonly = true
>+    sources = [
>+      "test/simulcast_test_fixture.h",
>+    ]
>+  }
>+
>+  rtc_source_set("create_simulcast_test_fixture_api") {
>+    visibility = [ "*" ]
>+    testonly = true
>+    sources = [
>+      "test/create_simulcast_test_fixture.cc",
>+      "test/create_simulcast_test_fixture.h",
>+    ]
>+    deps = [
>+      ":simulcast_test_fixture_api",
>+      "../modules/video_coding:simulcast_test_fixture_impl",
>+      "../rtc_base:rtc_base_approved",
>+      "video_codecs:video_codecs_api",
>+      "//third_party/abseil-cpp/absl/memory",
>+    ]
>+    if (!build_with_chromium && is_clang) {
>+      # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>+      suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>+    }
>+  }
>+
>+  rtc_source_set("videocodec_test_fixture_api") {
>+    visibility = [ "*" ]
>+    testonly = true
>+    sources = [
>+      "test/videocodec_test_fixture.h",
>+      "test/videocodec_test_stats.cc",
>+      "test/videocodec_test_stats.h",
>+    ]
>+    deps = [
>+      "..:webrtc_common",
>+      "../modules/video_coding:video_codec_interface",
>+      "video_codecs:video_codecs_api",
>+    ]
>+  }
>+
>+  rtc_source_set("create_videocodec_test_fixture_api") {
>+    visibility = [ "*" ]
>+    testonly = true
>+    sources = [
>+      "test/create_videocodec_test_fixture.cc",
>+      "test/create_videocodec_test_fixture.h",
>+    ]
>+    deps = [
>+      ":videocodec_test_fixture_api",
>+      "../modules/video_coding:video_codecs_test_framework",
>+      "../modules/video_coding:videocodec_test_impl",
>+      "../rtc_base:rtc_base_approved",
>+      "video_codecs:video_codecs_api",
>+      "//third_party/abseil-cpp/absl/memory",
>+    ]
>+    if (!build_with_chromium && is_clang) {
>+      # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>+      suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>+    }
>+  }
>+
>   rtc_source_set("mock_audio_mixer") {
>     testonly = true
>     sources = [
>@@ -294,9 +395,20 @@ if (rtc_include_tests) {
>     ]
> 
>     deps = [
>-      ":audio_mixer_api",
>       "../test:test_support",
>-      "//testing/gmock",
>+      "audio:audio_mixer_api",
>+    ]
>+  }
>+
>+  rtc_source_set("mock_peerconnectioninterface") {
>+    testonly = true
>+    sources = [
>+      "test/mock_peerconnectioninterface.h",
>+    ]
>+
>+    deps = [
>+      ":libjingle_peerconnection_api",
>+      "../test:test_support",
>     ]
>   }
> 
>@@ -310,43 +422,32 @@ if (rtc_include_tests) {
>     deps = [
>       ":libjingle_peerconnection_api",
>       "../test:test_support",
>-      "//testing/gmock",
>     ]
>   }
> 
>-  rtc_source_set("mock_video_codec_factory") {
>+  rtc_source_set("mock_video_bitrate_allocator") {
>     testonly = true
>     sources = [
>-      "test/mock_video_decoder_factory.h",
>-      "test/mock_video_encoder_factory.h",
>+      "test/mock_video_bitrate_allocator.h",
>     ]
> 
>     deps = [
>-      "../api/video_codecs:video_codecs_api",
>+      "../api/video:video_bitrate_allocator",
>       "../test:test_support",
>-      "//testing/gmock",
>     ]
>   }
> 
>-  rtc_source_set("fakemetricsobserver") {
>+  rtc_source_set("mock_video_codec_factory") {
>     testonly = true
>     sources = [
>-      "fakemetricsobserver.cc",
>-      "fakemetricsobserver.h",
>+      "test/mock_video_decoder_factory.h",
>+      "test/mock_video_encoder_factory.h",
>     ]
>+
>     deps = [
>-      "../api:peerconnection_and_implicit_call_api",
>-      "../media:rtc_media_base",
>-      "../rtc_base:checks",
>-      "../rtc_base:rtc_base_approved",
>+      "../api/video_codecs:video_codecs_api",
>+      "../test:test_support",
>     ]
>-    if (!build_with_chromium && is_clang) {
>-      # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>-      suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-    }
>-    if (!build_with_mozilla) {
>-      deps += [ ":libjingle_peerconnection_api" ]
>-    }
>   }
> 
>   rtc_source_set("rtc_api_unittests") {
>@@ -354,7 +455,6 @@ if (rtc_include_tests) {
> 
>     sources = [
>       "array_view_unittest.cc",
>-      "optional_unittest.cc",
>       "ortc/mediadescription_unittest.cc",
>       "ortc/sessiondescription_unittest.cc",
>       "rtcerror_unittest.cc",
>@@ -370,12 +470,12 @@ if (rtc_include_tests) {
>       ":array_view",
>       ":libjingle_peerconnection_api",
>       ":libjingle_peerconnection_test_api",
>-      ":optional",
>       ":ortc_api",
>       "../rtc_base:checks",
>       "../rtc_base:rtc_base_approved",
>       "../rtc_base:rtc_base_tests_utils",
>       "../test:test_support",
>+      "units:units_unittests",
>     ]
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/DEPS b/Source/ThirdParty/libwebrtc/Source/webrtc/api/DEPS
>index a537633981d..847cce259f9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/DEPS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/DEPS
>@@ -4,22 +4,12 @@ include_rules = [
>   "+media",
>   "+p2p",
>   "+pc",
>+  "+logging/rtc_event_log/rtc_event_log_factory_interface.h",
>+  "+modules/audio_processing/include",
>+  "+system_wrappers/include",
> ]
> 
> specific_include_rules = {
>-  "peerconnection_jni\.cc": [
>-    "+voice_engine",
>-  ],
>-
>-  # TODO(ossu): Remove this exception when {builtin_,}audio_encoder_factory.h
>-  # has moved to api/.
>-  "peerconnectioninterface\.h": [
>-    "+call/callfactoryinterface.h",
>-    "+logging/rtc_event_log/rtc_event_log_factory_interface.h",
>-    "+modules/audio_coding/codecs/audio_encoder_factory.h",
>-    "+modules/audio_coding/codecs/builtin_audio_encoder_factory.h",
>-  ],
>-
>   # Needed because AudioEncoderOpus is in the wrong place for
>   # backwards compatibilty reasons. See
>   # https://bugs.chromium.org/p/webrtc/issues/detail?id=7847
>@@ -33,14 +23,7 @@ specific_include_rules = {
>   # since no one #includes them.
>   ".*\.cc": [
>     "+modules/audio_coding",
>-  ],
>-
>-  ".*i420_buffer\.h": [
>-    "+system_wrappers/include/aligned_malloc.h",
>-  ],
>-
>-  # Needed to use the APM statistics.
>-  "mediastreaminterface.h": [
>-    "+modules/audio_processing/include/audio_processing_statistics.h",
>+    "+modules/audio_processing",
>+    "+modules/video_coding",
>   ],
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/api/OWNERS
>index 1c30e33a68a..593de230d37 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/OWNERS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/OWNERS
>@@ -5,8 +5,9 @@ perkj@webrtc.org
> solenberg@webrtc.org
> tkchin@webrtc.org
> tommi@webrtc.org
>-deadbeef@webrtc.org
> kwiberg@webrtc.org
>+steveanton@webrtc.org
>+shampson@webrtc.org
> 
> per-file peerconnection*=hbos@webrtc.org
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/array_view.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/array_view.h
>index b27bd29a8e2..efc642d6fc6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/array_view.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/array_view.h
>@@ -12,6 +12,7 @@
> #define API_ARRAY_VIEW_H_
> 
> #include <algorithm>
>+#include <array>
> #include <type_traits>
> 
> #include "rtc_base/checks.h"
>@@ -94,7 +95,7 @@ class ArrayViewBase {
>   static_assert(Size > 0, "ArrayView size must be variable or non-negative");
> 
>  public:
>-  ArrayViewBase(T* data, size_t) : data_(data) {}
>+  ArrayViewBase(T* data, size_t size) : data_(data) {}
> 
>   static constexpr size_t size() { return Size; }
>   static constexpr bool empty() { return false; }
>@@ -111,7 +112,7 @@ class ArrayViewBase {
> template <typename T>
> class ArrayViewBase<T, 0> {
>  public:
>-  explicit ArrayViewBase(T*, size_t) {}
>+  explicit ArrayViewBase(T* data, size_t size) {}
> 
>   static constexpr size_t size() { return 0; }
>   static constexpr bool empty() { return true; }
>@@ -169,7 +170,7 @@ class ArrayView final : public impl::ArrayViewBase<T, Size> {
>     RTC_DCHECK_EQ(0, size);
>   }
> 
>-  // Construct an ArrayView from an array.
>+  // Construct an ArrayView from a C-style array.
>   template <typename U, size_t N>
>   ArrayView(U (&array)[N])  // NOLINT
>       : ArrayView(array, N) {
>@@ -177,6 +178,26 @@ class ArrayView final : public impl::ArrayViewBase<T, Size> {
>                   "Array size must match ArrayView size");
>   }
> 
>+  // (Only if size is fixed.) Construct a fixed size ArrayView<T, N> from a
>+  // non-const std::array instance. For an ArrayView with variable size, the
>+  // used ctor is ArrayView(U& u) instead.
>+  template <typename U,
>+            size_t N,
>+            typename std::enable_if<
>+                Size == static_cast<std::ptrdiff_t>(N)>::type* = nullptr>
>+  ArrayView(std::array<U, N>& u)  // NOLINT
>+      : ArrayView(u.data(), u.size()) {}
>+
>+  // (Only if size is fixed.) Construct a fixed size ArrayView<T, N> where T is
>+  // const from a const(expr) std::array instance. For an ArrayView with
>+  // variable size, the used ctor is ArrayView(U& u) instead.
>+  template <typename U,
>+            size_t N,
>+            typename std::enable_if<
>+                Size == static_cast<std::ptrdiff_t>(N)>::type* = nullptr>
>+  ArrayView(const std::array<U, N>& u)  // NOLINT
>+      : ArrayView(u.data(), u.size()) {}
>+
>   // (Only if size is fixed.) Construct an ArrayView from any type U that has a
>   // static constexpr size() method whose return value is equal to Size, and a
>   // data() method whose return value converts implicitly to T*. In particular,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/array_view_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/array_view_unittest.cc
>index 48dff2c266e..694ed0b0085 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/array_view_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/array_view_unittest.cc
>@@ -9,6 +9,7 @@
>  */
> 
> #include <algorithm>
>+#include <array>
> #include <string>
> #include <utility>
> #include <vector>
>@@ -180,6 +181,38 @@ TEST(ArrayViewTest, TestCopyAssignmentFixed) {
>   // v = z;  // Compile error, because can't drop const.
> }
> 
>+TEST(ArrayViewTest, TestStdArray) {
>+  constexpr size_t size = 5;
>+  std::array<float, size> arr{};
>+  // Fixed size view.
>+  rtc::ArrayView<float, size> arr_view_fixed(arr);
>+  EXPECT_EQ(arr.data(), arr_view_fixed.data());
>+  static_assert(size == arr_view_fixed.size(), "");
>+  // Variable size view.
>+  rtc::ArrayView<float> arr_view(arr);
>+  EXPECT_EQ(arr.data(), arr_view.data());
>+  EXPECT_EQ(size, arr_view.size());
>+}
>+
>+TEST(ArrayViewTest, TestConstStdArray) {
>+  constexpr size_t size = 5;
>+
>+  constexpr std::array<float, size> constexpr_arr{};
>+  rtc::ArrayView<const float, size> constexpr_arr_view(constexpr_arr);
>+  EXPECT_EQ(constexpr_arr.data(), constexpr_arr_view.data());
>+  static_assert(constexpr_arr.size() == constexpr_arr_view.size(), "");
>+
>+  const std::array<float, size> const_arr{};
>+  rtc::ArrayView<const float, size> const_arr_view(const_arr);
>+  EXPECT_EQ(const_arr.data(), const_arr_view.data());
>+  static_assert(const_arr.size() == const_arr_view.size(), "");
>+
>+  std::array<float, size> non_const_arr{};
>+  rtc::ArrayView<const float, size> non_const_arr_view(non_const_arr);
>+  EXPECT_EQ(non_const_arr.data(), non_const_arr_view.data());
>+  static_assert(non_const_arr.size() == non_const_arr_view.size(), "");
>+}
>+
> TEST(ArrayViewTest, TestStdVector) {
>   std::vector<int> v;
>   v.push_back(3);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/asyncresolverfactory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/asyncresolverfactory.h
>new file mode 100644
>index 00000000000..3c3bb1e7030
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/asyncresolverfactory.h
>@@ -0,0 +1,33 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_ASYNCRESOLVERFACTORY_H_
>+#define API_ASYNCRESOLVERFACTORY_H_
>+
>+#include "rtc_base/asyncresolverinterface.h"
>+
>+namespace webrtc {
>+
>+// An abstract factory for creating AsyncResolverInterfaces. This allows
>+// client applications to provide WebRTC with their own mechanism for
>+// performing DNS resolution.
>+class AsyncResolverFactory {
>+ public:
>+  AsyncResolverFactory() = default;
>+  virtual ~AsyncResolverFactory() = default;
>+
>+  // The returned object is responsible for deleting itself after address
>+  // resolution has completed.
>+  virtual rtc::AsyncResolverInterface* Create() = 0;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // API_ASYNCRESOLVERFACTORY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/BUILD.gn
>new file mode 100644
>index 00000000000..286a5a6f1c6
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/BUILD.gn
>@@ -0,0 +1,66 @@
>+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+#
>+# Use of this source code is governed by a BSD-style license
>+# that can be found in the LICENSE file in the root of the source
>+# tree. An additional intellectual property rights grant can be found
>+# in the file PATENTS.  All contributing project authors may
>+# be found in the AUTHORS file in the root of the source tree.
>+
>+import("../../webrtc.gni")
>+
>+rtc_source_set("audio_frame_api") {
>+  visibility = [ "*" ]
>+  sources = [
>+    "audio_frame.cc",
>+    "audio_frame.h",
>+  ]
>+
>+  deps = [
>+    "../../rtc_base:checks",
>+    "../../rtc_base:rtc_base_approved",
>+  ]
>+}
>+
>+rtc_source_set("audio_mixer_api") {
>+  visibility = [ "*" ]
>+  sources = [
>+    "audio_mixer.h",
>+  ]
>+
>+  deps = [
>+    ":audio_frame_api",
>+    "../../rtc_base:rtc_base_approved",
>+  ]
>+}
>+
>+rtc_source_set("aec3_config") {
>+  visibility = [ "*" ]
>+  sources = [
>+    "echo_canceller3_config.cc",
>+    "echo_canceller3_config.h",
>+  ]
>+}
>+
>+rtc_source_set("aec3_factory") {
>+  visibility = [ "*" ]
>+  configs += [ "../../modules/audio_processing:apm_debug_dump" ]
>+  sources = [
>+    "echo_canceller3_factory.cc",
>+    "echo_canceller3_factory.h",
>+  ]
>+
>+  deps = [
>+    ":aec3_config",
>+    ":echo_control",
>+    "../../modules/audio_processing/aec3",
>+    "../../rtc_base:rtc_base_approved",
>+    "//third_party/abseil-cpp/absl/memory",
>+  ]
>+}
>+
>+rtc_source_set("echo_control") {
>+  visibility = [ "*" ]
>+  sources = [
>+    "echo_control.h",
>+  ]
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/OWNERS
>new file mode 100644
>index 00000000000..bb499b450fc
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/OWNERS
>@@ -0,0 +1,2 @@
>+gustaf@webrtc.org
>+peah@webrtc.org
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/audio_frame.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/audio_frame.cc
>new file mode 100644
>index 00000000000..75d30b0ba27
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/audio_frame.cc
>@@ -0,0 +1,130 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/audio/audio_frame.h"
>+
>+#include <string.h>
>+
>+#include "rtc_base/checks.h"
>+#include "rtc_base/timeutils.h"
>+
>+namespace webrtc {
>+
>+AudioFrame::AudioFrame() {
>+  // Visual Studio doesn't like this in the class definition.
>+  static_assert(sizeof(data_) == kMaxDataSizeBytes, "kMaxDataSizeBytes");
>+}
>+
>+void AudioFrame::Reset() {
>+  ResetWithoutMuting();
>+  muted_ = true;
>+}
>+
>+void AudioFrame::ResetWithoutMuting() {
>+  // TODO(wu): Zero is a valid value for |timestamp_|. We should initialize
>+  // to an invalid value, or add a new member to indicate invalidity.
>+  timestamp_ = 0;
>+  elapsed_time_ms_ = -1;
>+  ntp_time_ms_ = -1;
>+  samples_per_channel_ = 0;
>+  sample_rate_hz_ = 0;
>+  num_channels_ = 0;
>+  speech_type_ = kUndefined;
>+  vad_activity_ = kVadUnknown;
>+  profile_timestamp_ms_ = 0;
>+}
>+
>+void AudioFrame::UpdateFrame(uint32_t timestamp,
>+                             const int16_t* data,
>+                             size_t samples_per_channel,
>+                             int sample_rate_hz,
>+                             SpeechType speech_type,
>+                             VADActivity vad_activity,
>+                             size_t num_channels) {
>+  timestamp_ = timestamp;
>+  samples_per_channel_ = samples_per_channel;
>+  sample_rate_hz_ = sample_rate_hz;
>+  speech_type_ = speech_type;
>+  vad_activity_ = vad_activity;
>+  num_channels_ = num_channels;
>+
>+  const size_t length = samples_per_channel * num_channels;
>+  RTC_CHECK_LE(length, kMaxDataSizeSamples);
>+  if (data != nullptr) {
>+    memcpy(data_, data, sizeof(int16_t) * length);
>+    muted_ = false;
>+  } else {
>+    muted_ = true;
>+  }
>+}
>+
>+void AudioFrame::CopyFrom(const AudioFrame& src) {
>+  if (this == &src)
>+    return;
>+
>+  timestamp_ = src.timestamp_;
>+  elapsed_time_ms_ = src.elapsed_time_ms_;
>+  ntp_time_ms_ = src.ntp_time_ms_;
>+  muted_ = src.muted();
>+  samples_per_channel_ = src.samples_per_channel_;
>+  sample_rate_hz_ = src.sample_rate_hz_;
>+  speech_type_ = src.speech_type_;
>+  vad_activity_ = src.vad_activity_;
>+  num_channels_ = src.num_channels_;
>+
>+  const size_t length = samples_per_channel_ * num_channels_;
>+  RTC_CHECK_LE(length, kMaxDataSizeSamples);
>+  if (!src.muted()) {
>+    memcpy(data_, src.data(), sizeof(int16_t) * length);
>+    muted_ = false;
>+  }
>+}
>+
>+void AudioFrame::UpdateProfileTimeStamp() {
>+  profile_timestamp_ms_ = rtc::TimeMillis();
>+}
>+
>+int64_t AudioFrame::ElapsedProfileTimeMs() const {
>+  if (profile_timestamp_ms_ == 0) {
>+    // Profiling has not been activated.
>+    return -1;
>+  }
>+  return rtc::TimeSince(profile_timestamp_ms_);
>+}
>+
>+const int16_t* AudioFrame::data() const {
>+  return muted_ ? empty_data() : data_;
>+}
>+
>+// TODO(henrik.lundin) Can we skip zeroing the buffer?
>+// See https://bugs.chromium.org/p/webrtc/issues/detail?id=5647.
>+int16_t* AudioFrame::mutable_data() {
>+  if (muted_) {
>+    memset(data_, 0, kMaxDataSizeBytes);
>+    muted_ = false;
>+  }
>+  return data_;
>+}
>+
>+void AudioFrame::Mute() {
>+  muted_ = true;
>+}
>+
>+bool AudioFrame::muted() const {
>+  return muted_;
>+}
>+
>+// static
>+const int16_t* AudioFrame::empty_data() {
>+  static int16_t* null_data = new int16_t[kMaxDataSizeSamples]();
>+  return &null_data[0];
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/audio_frame.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/audio_frame.h
>new file mode 100644
>index 00000000000..cc078771fae
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/audio_frame.h
>@@ -0,0 +1,131 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_AUDIO_AUDIO_FRAME_H_
>+#define API_AUDIO_AUDIO_FRAME_H_
>+
>+#include <stddef.h>
>+#include <stdint.h>
>+
>+#include "rtc_base/constructormagic.h"
>+
>+namespace webrtc {
>+
>+/* This class holds up to 60 ms of super-wideband (32 kHz) stereo audio. It
>+ * allows for adding and subtracting frames while keeping track of the resulting
>+ * states.
>+ *
>+ * Notes
>+ * - This is a de-facto api, not designed for external use. The AudioFrame class
>+ *   is in need of overhaul or even replacement, and anyone depending on it
>+ *   should be prepared for that.
>+ * - The total number of samples is samples_per_channel_ * num_channels_.
>+ * - Stereo data is interleaved starting with the left channel.
>+ */
>+class AudioFrame {
>+ public:
>+  // Using constexpr here causes linker errors unless the variable also has an
>+  // out-of-class definition, which is impractical in this header-only class.
>+  // (This makes no sense because it compiles as an enum value, which we most
>+  // certainly cannot take the address of, just fine.) C++17 introduces inline
>+  // variables which should allow us to switch to constexpr and keep this a
>+  // header-only class.
>+  enum : size_t {
>+    // Stereo, 32 kHz, 60 ms (2 * 32 * 60)
>+    kMaxDataSizeSamples = 3840,
>+    kMaxDataSizeBytes = kMaxDataSizeSamples * sizeof(int16_t),
>+  };
>+
>+  enum VADActivity { kVadActive = 0, kVadPassive = 1, kVadUnknown = 2 };
>+  enum SpeechType {
>+    kNormalSpeech = 0,
>+    kPLC = 1,
>+    kCNG = 2,
>+    kPLCCNG = 3,
>+    kUndefined = 4
>+  };
>+
>+  AudioFrame();
>+
>+  // Resets all members to their default state.
>+  void Reset();
>+  // Same as Reset(), but leaves mute state unchanged. Muting a frame requires
>+  // the buffer to be zeroed on the next call to mutable_data(). Callers
>+  // intending to write to the buffer immediately after Reset() can instead use
>+  // ResetWithoutMuting() to skip this wasteful zeroing.
>+  void ResetWithoutMuting();
>+
>+  void UpdateFrame(uint32_t timestamp,
>+                   const int16_t* data,
>+                   size_t samples_per_channel,
>+                   int sample_rate_hz,
>+                   SpeechType speech_type,
>+                   VADActivity vad_activity,
>+                   size_t num_channels = 1);
>+
>+  void CopyFrom(const AudioFrame& src);
>+
>+  // Sets a wall-time clock timestamp in milliseconds to be used for profiling
>+  // of time between two points in the audio chain.
>+  // Example:
>+  //   t0: UpdateProfileTimeStamp()
>+  //   t1: ElapsedProfileTimeMs() => t1 - t0 [msec]
>+  void UpdateProfileTimeStamp();
>+  // Returns the time difference between now and when UpdateProfileTimeStamp()
>+  // was last called. Returns -1 if UpdateProfileTimeStamp() has not yet been
>+  // called.
>+  int64_t ElapsedProfileTimeMs() const;
>+
>+  // data() returns a zeroed static buffer if the frame is muted.
>+  // mutable_frame() always returns a non-static buffer; the first call to
>+  // mutable_frame() zeros the non-static buffer and marks the frame unmuted.
>+  const int16_t* data() const;
>+  int16_t* mutable_data();
>+
>+  // Prefer to mute frames using AudioFrameOperations::Mute.
>+  void Mute();
>+  // Frame is muted by default.
>+  bool muted() const;
>+
>+  // RTP timestamp of the first sample in the AudioFrame.
>+  uint32_t timestamp_ = 0;
>+  // Time since the first frame in milliseconds.
>+  // -1 represents an uninitialized value.
>+  int64_t elapsed_time_ms_ = -1;
>+  // NTP time of the estimated capture time in local timebase in milliseconds.
>+  // -1 represents an uninitialized value.
>+  int64_t ntp_time_ms_ = -1;
>+  size_t samples_per_channel_ = 0;
>+  int sample_rate_hz_ = 0;
>+  size_t num_channels_ = 0;
>+  SpeechType speech_type_ = kUndefined;
>+  VADActivity vad_activity_ = kVadUnknown;
>+  // Monotonically increasing timestamp intended for profiling of audio frames.
>+  // Typically used for measuring elapsed time between two different points in
>+  // the audio path. No lock is used to save resources and we are thread safe
>+  // by design. Also, absl::optional is not used since it will cause a "complex
>+  // class/struct needs an explicit out-of-line destructor" build error.
>+  int64_t profile_timestamp_ms_ = 0;
>+
>+ private:
>+  // A permamently zeroed out buffer to represent muted frames. This is a
>+  // header-only class, so the only way to avoid creating a separate empty
>+  // buffer per translation unit is to wrap a static in an inline function.
>+  static const int16_t* empty_data();
>+
>+  int16_t data_[kMaxDataSizeSamples];
>+  bool muted_ = true;
>+
>+  RTC_DISALLOW_COPY_AND_ASSIGN(AudioFrame);
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // API_AUDIO_AUDIO_FRAME_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/audio_mixer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/audio_mixer.h
>index 63b8b8ff9bb..14eefc173be 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/audio_mixer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/audio_mixer.h
>@@ -13,7 +13,7 @@
> 
> #include <memory>
> 
>-#include "modules/include/module_common_types.h"
>+#include "api/audio/audio_frame.h"
> #include "rtc_base/refcount.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/echo_canceller3_config.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/echo_canceller3_config.cc
>new file mode 100644
>index 00000000000..9af7f11c578
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/echo_canceller3_config.cc
>@@ -0,0 +1,24 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#include "api/audio/echo_canceller3_config.h"
>+
>+namespace webrtc {
>+
>+EchoCanceller3Config::EchoCanceller3Config() = default;
>+EchoCanceller3Config::EchoCanceller3Config(const EchoCanceller3Config& e) =
>+    default;
>+EchoCanceller3Config::Mask::Mask() = default;
>+EchoCanceller3Config::Mask::Mask(const EchoCanceller3Config::Mask& m) = default;
>+
>+EchoCanceller3Config::EchoModel::EchoModel() = default;
>+EchoCanceller3Config::EchoModel::EchoModel(
>+    const EchoCanceller3Config::EchoModel& e) = default;
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/echo_canceller3_config.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/echo_canceller3_config.h
>new file mode 100644
>index 00000000000..d633d047b0f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/echo_canceller3_config.h
>@@ -0,0 +1,173 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_AUDIO_ECHO_CANCELLER3_CONFIG_H_
>+#define API_AUDIO_ECHO_CANCELLER3_CONFIG_H_
>+
>+#include <stddef.h>  // size_t
>+
>+namespace webrtc {
>+
>+// Configuration struct for EchoCanceller3
>+struct EchoCanceller3Config {
>+  EchoCanceller3Config();
>+  EchoCanceller3Config(const EchoCanceller3Config& e);
>+  struct Delay {
>+    size_t default_delay = 5;
>+    size_t down_sampling_factor = 4;
>+    size_t num_filters = 6;
>+    size_t api_call_jitter_blocks = 26;
>+    size_t min_echo_path_delay_blocks = 0;
>+    size_t delay_headroom_blocks = 2;
>+    size_t hysteresis_limit_1_blocks = 1;
>+    size_t hysteresis_limit_2_blocks = 1;
>+    size_t skew_hysteresis_blocks = 3;
>+  } delay;
>+
>+  struct Filter {
>+    struct MainConfiguration {
>+      size_t length_blocks;
>+      float leakage_converged;
>+      float leakage_diverged;
>+      float error_floor;
>+      float noise_gate;
>+    };
>+
>+    struct ShadowConfiguration {
>+      size_t length_blocks;
>+      float rate;
>+      float noise_gate;
>+    };
>+
>+    MainConfiguration main = {13, 0.00005f, 0.01f, 0.1f, 20075344.f};
>+    ShadowConfiguration shadow = {13, 0.7f, 20075344.f};
>+
>+    MainConfiguration main_initial = {12, 0.005f, 0.5f, 0.001f, 20075344.f};
>+    ShadowConfiguration shadow_initial = {12, 0.9f, 20075344.f};
>+
>+    size_t config_change_duration_blocks = 250;
>+  } filter;
>+
>+  struct Erle {
>+    float min = 1.f;
>+    float max_l = 4.f;
>+    float max_h = 1.5f;
>+  } erle;
>+
>+  struct EpStrength {
>+    float lf = 1.f;
>+    float mf = 1.f;
>+    float hf = 1.f;
>+    float default_len = 0.88f;
>+    bool reverb_based_on_render = true;
>+    bool echo_can_saturate = true;
>+    bool bounded_erl = false;
>+  } ep_strength;
>+
>+  struct Mask {
>+    Mask();
>+    Mask(const Mask& m);
>+    float m0 = 0.1f;
>+    float m1 = 0.01f;
>+    float m2 = 0.0001f;
>+    float m3 = 0.01f;
>+    float m5 = 0.01f;
>+    float m6 = 0.0001f;
>+    float m7 = 0.01f;
>+    float m8 = 0.0001f;
>+    float m9 = 0.1f;
>+
>+    float gain_curve_offset = 1.45f;
>+    float gain_curve_slope = 5.f;
>+    float temporal_masking_lf = 0.9f;
>+    float temporal_masking_hf = 0.6f;
>+    size_t temporal_masking_lf_bands = 3;
>+  } gain_mask;
>+
>+  struct EchoAudibility {
>+    float low_render_limit = 4 * 64.f;
>+    float normal_render_limit = 64.f;
>+    float floor_power = 2 * 64.f;
>+    float audibility_threshold_lf = 10;
>+    float audibility_threshold_mf = 10;
>+    float audibility_threshold_hf = 10;
>+    bool use_stationary_properties = true;
>+  } echo_audibility;
>+
>+  struct RenderLevels {
>+    float active_render_limit = 100.f;
>+    float poor_excitation_render_limit = 150.f;
>+    float poor_excitation_render_limit_ds8 = 20.f;
>+  } render_levels;
>+
>+  struct GainUpdates {
>+    struct GainChanges {
>+      float max_inc;
>+      float max_dec;
>+      float rate_inc;
>+      float rate_dec;
>+      float min_inc;
>+      float min_dec;
>+    };
>+
>+    GainChanges low_noise = {2.f, 2.f, 1.4f, 1.4f, 1.1f, 1.1f};
>+    GainChanges initial = {2.f, 2.f, 1.5f, 1.5f, 1.2f, 1.2f};
>+    GainChanges normal = {2.f, 2.f, 1.5f, 1.5f, 1.2f, 1.2f};
>+    GainChanges saturation = {1.2f, 1.2f, 1.5f, 1.5f, 1.f, 1.f};
>+    GainChanges nonlinear = {1.5f, 1.5f, 1.2f, 1.2f, 1.1f, 1.1f};
>+
>+    float max_inc_factor = 2.0f;
>+    float max_dec_factor_lf = 0.25f;
>+    float floor_first_increase = 0.00001f;
>+  } gain_updates;
>+
>+  struct EchoRemovalControl {
>+    struct GainRampup {
>+      float initial_gain = 0.0f;
>+      float first_non_zero_gain = 0.001f;
>+      int non_zero_gain_blocks = 187;
>+      int full_gain_blocks = 312;
>+    } gain_rampup;
>+    bool has_clock_drift = false;
>+    bool linear_and_stable_echo_path = false;
>+  } echo_removal_control;
>+
>+  struct EchoModel {
>+    EchoModel();
>+    EchoModel(const EchoModel& e);
>+    size_t noise_floor_hold = 50;
>+    float min_noise_floor_power = 1638400.f;
>+    float stationary_gate_slope = 10.f;
>+    float noise_gate_power = 27509.42f;
>+    float noise_gate_slope = 0.3f;
>+    size_t render_pre_window_size = 1;
>+    size_t render_post_window_size = 1;
>+    size_t render_pre_window_size_init = 10;
>+    size_t render_post_window_size_init = 10;
>+    float nonlinear_hold = 1;
>+    float nonlinear_release = 0.001f;
>+  } echo_model;
>+
>+  struct Suppressor {
>+    size_t bands_with_reliable_coherence = 5;
>+    size_t nearend_average_blocks = 4;
>+
>+    struct MaskingThresholds {
>+      float enr_transparent;
>+      float enr_suppress;
>+      float emr_transparent;
>+    };
>+    MaskingThresholds mask_lf = {.2f, .3f, .3f};
>+    MaskingThresholds mask_hf = {.07f, .1f, .3f};
>+  } suppressor;
>+};
>+}  // namespace webrtc
>+
>+#endif  // API_AUDIO_ECHO_CANCELLER3_CONFIG_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/echo_canceller3_factory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/echo_canceller3_factory.cc
>new file mode 100644
>index 00000000000..07f295f2ea2
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/echo_canceller3_factory.cc
>@@ -0,0 +1,27 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#include "api/audio/echo_canceller3_factory.h"
>+
>+#include <memory>
>+
>+#include "absl/memory/memory.h"
>+#include "modules/audio_processing/aec3/echo_canceller3.h"
>+
>+namespace webrtc {
>+
>+EchoCanceller3Factory::EchoCanceller3Factory() {}
>+
>+EchoCanceller3Factory::EchoCanceller3Factory(const EchoCanceller3Config& config)
>+    : config_(config) {}
>+
>+std::unique_ptr<EchoControl> EchoCanceller3Factory::Create(int sample_rate_hz) {
>+  return absl::make_unique<EchoCanceller3>(config_, sample_rate_hz, true);
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/echo_canceller3_factory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/echo_canceller3_factory.h
>new file mode 100644
>index 00000000000..f6db1168503
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/echo_canceller3_factory.h
>@@ -0,0 +1,38 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_AUDIO_ECHO_CANCELLER3_FACTORY_H_
>+#define API_AUDIO_ECHO_CANCELLER3_FACTORY_H_
>+
>+#include <memory>
>+
>+#include "api/audio/echo_canceller3_config.h"
>+#include "api/audio/echo_control.h"
>+
>+namespace webrtc {
>+
>+class EchoCanceller3Factory : public EchoControlFactory {
>+ public:
>+  // Factory producing EchoCanceller3 instances with the default configuration.
>+  EchoCanceller3Factory();
>+
>+  // Factory producing EchoCanceller3 instances with the specified
>+  // configuration.
>+  explicit EchoCanceller3Factory(const EchoCanceller3Config& config);
>+
>+  // Creates an EchoCanceller3 running at the specified sampling rate.
>+  std::unique_ptr<EchoControl> Create(int sample_rate_hz) override;
>+
>+ private:
>+  const EchoCanceller3Config config_;
>+};
>+}  // namespace webrtc
>+
>+#endif  // API_AUDIO_ECHO_CANCELLER3_FACTORY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/echo_control.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/echo_control.h
>new file mode 100644
>index 00000000000..f549f40fbb0
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/echo_control.h
>@@ -0,0 +1,55 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_AUDIO_ECHO_CONTROL_H_
>+#define API_AUDIO_ECHO_CONTROL_H_
>+
>+#include <memory>
>+
>+namespace webrtc {
>+
>+class AudioBuffer;
>+
>+// Interface for an acoustic echo cancellation (AEC) submodule.
>+class EchoControl {
>+ public:
>+  // Analysis (not changing) of the render signal.
>+  virtual void AnalyzeRender(AudioBuffer* render) = 0;
>+
>+  // Analysis (not changing) of the capture signal.
>+  virtual void AnalyzeCapture(AudioBuffer* capture) = 0;
>+
>+  // Processes the capture signal in order to remove the echo.
>+  virtual void ProcessCapture(AudioBuffer* capture, bool echo_path_change) = 0;
>+
>+  struct Metrics {
>+    double echo_return_loss;
>+    double echo_return_loss_enhancement;
>+    int delay_ms;
>+  };
>+
>+  // Collect current metrics from the echo controller.
>+  virtual Metrics GetMetrics() const = 0;
>+
>+  // Provides an optional external estimate of the audio buffer delay.
>+  virtual void SetAudioBufferDelay(size_t delay_ms) = 0;
>+
>+  virtual ~EchoControl() {}
>+};
>+
>+// Interface for a factory that creates EchoControllers.
>+class EchoControlFactory {
>+ public:
>+  virtual std::unique_ptr<EchoControl> Create(int sample_rate_hz) = 0;
>+  virtual ~EchoControlFactory() = default;
>+};
>+}  // namespace webrtc
>+
>+#endif  // API_AUDIO_ECHO_CONTROL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/test/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/test/BUILD.gn
>new file mode 100644
>index 00000000000..9a797788f68
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/test/BUILD.gn
>@@ -0,0 +1,27 @@
>+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+#
>+# Use of this source code is governed by a BSD-style license
>+# that can be found in the LICENSE file in the root of the source
>+# tree. An additional intellectual property rights grant can be found
>+# in the file PATENTS.  All contributing project authors may
>+# be found in the AUTHORS file in the root of the source tree.
>+
>+import("../../../webrtc.gni")
>+if (is_android) {
>+  import("//build/config/android/config.gni")
>+  import("//build/config/android/rules.gni")
>+}
>+
>+if (rtc_include_tests) {
>+  rtc_source_set("audio_api_unittests") {
>+    testonly = true
>+    sources = [
>+      "audio_frame_unittest.cc",
>+    ]
>+    deps = [
>+      "..:audio_frame_api",
>+      "../../../rtc_base:rtc_base_approved",
>+      "../../../test:test_support",
>+    ]
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/test/audio_frame_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/test/audio_frame_unittest.cc
>new file mode 100644
>index 00000000000..b969d90a4c1
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio/test/audio_frame_unittest.cc
>@@ -0,0 +1,113 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <string.h>  // memcmp
>+
>+#include "api/audio/audio_frame.h"
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+
>+namespace {
>+
>+bool AllSamplesAre(int16_t sample, const AudioFrame& frame) {
>+  const int16_t* frame_data = frame.data();
>+  for (size_t i = 0; i < AudioFrame::kMaxDataSizeSamples; i++) {
>+    if (frame_data[i] != sample) {
>+      return false;
>+    }
>+  }
>+  return true;
>+}
>+
>+constexpr uint32_t kTimestamp = 27;
>+constexpr int kSampleRateHz = 16000;
>+constexpr size_t kNumChannels = 1;
>+constexpr size_t kSamplesPerChannel = kSampleRateHz / 100;
>+
>+}  // namespace
>+
>+TEST(AudioFrameTest, FrameStartsMuted) {
>+  AudioFrame frame;
>+  EXPECT_TRUE(frame.muted());
>+  EXPECT_TRUE(AllSamplesAre(0, frame));
>+}
>+
>+TEST(AudioFrameTest, UnmutedFrameIsInitiallyZeroed) {
>+  AudioFrame frame;
>+  frame.mutable_data();
>+  EXPECT_FALSE(frame.muted());
>+  EXPECT_TRUE(AllSamplesAre(0, frame));
>+}
>+
>+TEST(AudioFrameTest, MutedFrameBufferIsZeroed) {
>+  AudioFrame frame;
>+  int16_t* frame_data = frame.mutable_data();
>+  for (size_t i = 0; i < AudioFrame::kMaxDataSizeSamples; i++) {
>+    frame_data[i] = 17;
>+  }
>+  ASSERT_TRUE(AllSamplesAre(17, frame));
>+  frame.Mute();
>+  EXPECT_TRUE(frame.muted());
>+  EXPECT_TRUE(AllSamplesAre(0, frame));
>+}
>+
>+TEST(AudioFrameTest, UpdateFrame) {
>+  AudioFrame frame;
>+  int16_t samples[kNumChannels * kSamplesPerChannel] = {17};
>+  frame.UpdateFrame(kTimestamp, samples, kSamplesPerChannel, kSampleRateHz,
>+                    AudioFrame::kPLC, AudioFrame::kVadActive, kNumChannels);
>+
>+  EXPECT_EQ(kTimestamp, frame.timestamp_);
>+  EXPECT_EQ(kSamplesPerChannel, frame.samples_per_channel_);
>+  EXPECT_EQ(kSampleRateHz, frame.sample_rate_hz_);
>+  EXPECT_EQ(AudioFrame::kPLC, frame.speech_type_);
>+  EXPECT_EQ(AudioFrame::kVadActive, frame.vad_activity_);
>+  EXPECT_EQ(kNumChannels, frame.num_channels_);
>+
>+  EXPECT_FALSE(frame.muted());
>+  EXPECT_EQ(0, memcmp(samples, frame.data(), sizeof(samples)));
>+
>+  frame.UpdateFrame(kTimestamp, nullptr /* data*/, kSamplesPerChannel,
>+                    kSampleRateHz, AudioFrame::kPLC, AudioFrame::kVadActive,
>+                    kNumChannels);
>+  EXPECT_TRUE(frame.muted());
>+  EXPECT_TRUE(AllSamplesAre(0, frame));
>+}
>+
>+TEST(AudioFrameTest, CopyFrom) {
>+  AudioFrame frame1;
>+  AudioFrame frame2;
>+
>+  int16_t samples[kNumChannels * kSamplesPerChannel] = {17};
>+  frame2.UpdateFrame(kTimestamp, samples, kSamplesPerChannel, kSampleRateHz,
>+                     AudioFrame::kPLC, AudioFrame::kVadActive, kNumChannels);
>+  frame1.CopyFrom(frame2);
>+
>+  EXPECT_EQ(frame2.timestamp_, frame1.timestamp_);
>+  EXPECT_EQ(frame2.samples_per_channel_, frame1.samples_per_channel_);
>+  EXPECT_EQ(frame2.sample_rate_hz_, frame1.sample_rate_hz_);
>+  EXPECT_EQ(frame2.speech_type_, frame1.speech_type_);
>+  EXPECT_EQ(frame2.vad_activity_, frame1.vad_activity_);
>+  EXPECT_EQ(frame2.num_channels_, frame1.num_channels_);
>+
>+  EXPECT_EQ(frame2.muted(), frame1.muted());
>+  EXPECT_EQ(0, memcmp(frame2.data(), frame1.data(), sizeof(samples)));
>+
>+  frame2.UpdateFrame(kTimestamp, nullptr /* data */, kSamplesPerChannel,
>+                     kSampleRateHz, AudioFrame::kPLC, AudioFrame::kVadActive,
>+                     kNumChannels);
>+  frame1.CopyFrom(frame2);
>+
>+  EXPECT_EQ(frame2.muted(), frame1.muted());
>+  EXPECT_EQ(0, memcmp(frame2.data(), frame1.data(), sizeof(samples)));
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/BUILD.gn
>index f7302600154..8e6a0f35225 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/BUILD.gn
>@@ -15,6 +15,8 @@ if (is_android) {
> rtc_source_set("audio_codecs_api") {
>   visibility = [ "*" ]
>   sources = [
>+    "audio_codec_pair_id.cc",
>+    "audio_codec_pair_id.h",
>     "audio_decoder.cc",
>     "audio_decoder.h",
>     "audio_decoder_factory.h",
>@@ -28,18 +30,18 @@ rtc_source_set("audio_codecs_api") {
>   ]
>   deps = [
>     "..:array_view",
>-    "..:optional",
>     "../..:webrtc_common",
>-    "../../:typedefs",
>     "../../rtc_base:checks",
>     "../../rtc_base:deprecation",
>     "../../rtc_base:rtc_base_approved",
>     "../../rtc_base:sanitizer",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
> 
> rtc_static_library("builtin_audio_decoder_factory") {
>   visibility = [ "*" ]
>+  allow_poison = [ "audio_codecs" ]
>   sources = [
>     "builtin_audio_decoder_factory.cc",
>     "builtin_audio_decoder_factory.h",
>@@ -69,6 +71,7 @@ rtc_static_library("builtin_audio_decoder_factory") {
> 
> rtc_static_library("builtin_audio_encoder_factory") {
>   visibility = [ "*" ]
>+  allow_poison = [ "audio_codecs" ]
>   sources = [
>     "builtin_audio_encoder_factory.cc",
>     "builtin_audio_encoder_factory.h",
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/L16/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/L16/BUILD.gn
>index f047bf132de..97248050012 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/L16/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/L16/BUILD.gn
>@@ -14,30 +14,35 @@ if (is_android) {
> 
> rtc_static_library("audio_encoder_L16") {
>   visibility = [ "*" ]
>+  poisonous = [ "audio_codecs" ]
>   sources = [
>     "audio_encoder_L16.cc",
>     "audio_encoder_L16.h",
>   ]
>   deps = [
>     "..:audio_codecs_api",
>-    "../..:optional",
>     "../../..:webrtc_common",
>     "../../../modules/audio_coding:pcm16b",
>     "../../../rtc_base:rtc_base_approved",
>+    "../../../rtc_base:safe_minmax",
>+    "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
> 
> rtc_static_library("audio_decoder_L16") {
>   visibility = [ "*" ]
>+  poisonous = [ "audio_codecs" ]
>   sources = [
>     "audio_decoder_L16.cc",
>     "audio_decoder_L16.h",
>   ]
>   deps = [
>     "..:audio_codecs_api",
>-    "../..:optional",
>     "../../..:webrtc_common",
>     "../../../modules/audio_coding:pcm16b",
>     "../../../rtc_base:rtc_base_approved",
>+    "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/L16/audio_decoder_L16.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/L16/audio_decoder_L16.cc
>index dd14e601f4e..a71e3087007 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/L16/audio_decoder_L16.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/L16/audio_decoder_L16.cc
>@@ -10,22 +10,22 @@
> 
> #include "api/audio_codecs/L16/audio_decoder_L16.h"
> 
>+#include "absl/memory/memory.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/audio_coding/codecs/pcm16b/audio_decoder_pcm16b.h"
> #include "modules/audio_coding/codecs/pcm16b/pcm16b_common.h"
> #include "rtc_base/numerics/safe_conversions.h"
>-#include "rtc_base/ptr_util.h"
> 
> namespace webrtc {
> 
>-rtc::Optional<AudioDecoderL16::Config> AudioDecoderL16::SdpToConfig(
>+absl::optional<AudioDecoderL16::Config> AudioDecoderL16::SdpToConfig(
>     const SdpAudioFormat& format) {
>   Config config;
>   config.sample_rate_hz = format.clockrate_hz;
>   config.num_channels = rtc::checked_cast<int>(format.num_channels);
>   return STR_CASE_CMP(format.name.c_str(), "L16") == 0 && config.IsOk()
>-             ? rtc::Optional<Config>(config)
>-             : rtc::nullopt;
>+             ? absl::optional<Config>(config)
>+             : absl::nullopt;
> }
> 
> void AudioDecoderL16::AppendSupportedDecoders(
>@@ -34,8 +34,9 @@ void AudioDecoderL16::AppendSupportedDecoders(
> }
> 
> std::unique_ptr<AudioDecoder> AudioDecoderL16::MakeAudioDecoder(
>-    const Config& config) {
>-  return config.IsOk() ? rtc::MakeUnique<AudioDecoderPcm16B>(
>+    const Config& config,
>+    absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
>+  return config.IsOk() ? absl::make_unique<AudioDecoderPcm16B>(
>                              config.sample_rate_hz, config.num_channels)
>                        : nullptr;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/L16/audio_decoder_L16.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/L16/audio_decoder_L16.h
>index db863b37de8..184ec24ed92 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/L16/audio_decoder_L16.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/L16/audio_decoder_L16.h
>@@ -14,9 +14,10 @@
> #include <memory>
> #include <vector>
> 
>+#include "absl/types/optional.h"
>+#include "api/audio_codecs/audio_codec_pair_id.h"
> #include "api/audio_codecs/audio_decoder.h"
> #include "api/audio_codecs/audio_format.h"
>-#include "api/optional.h"
> 
> namespace webrtc {
> 
>@@ -34,9 +35,11 @@ struct AudioDecoderL16 {
>     int sample_rate_hz = 8000;
>     int num_channels = 1;
>   };
>-  static rtc::Optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
>+  static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
>   static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
>-  static std::unique_ptr<AudioDecoder> MakeAudioDecoder(const Config& config);
>+  static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
>+      const Config& config,
>+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt);
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/L16/audio_encoder_L16.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/L16/audio_encoder_L16.cc
>index d0d9f6f6448..b516f62863d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/L16/audio_encoder_L16.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/L16/audio_encoder_L16.cc
>@@ -10,25 +10,34 @@
> 
> #include "api/audio_codecs/L16/audio_encoder_L16.h"
> 
>+#include "absl/memory/memory.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
> #include "modules/audio_coding/codecs/pcm16b/pcm16b_common.h"
> #include "rtc_base/numerics/safe_conversions.h"
>-#include "rtc_base/ptr_util.h"
>+#include "rtc_base/numerics/safe_minmax.h"
>+#include "rtc_base/string_to_number.h"
> 
> namespace webrtc {
> 
>-rtc::Optional<AudioEncoderL16::Config> AudioEncoderL16::SdpToConfig(
>+absl::optional<AudioEncoderL16::Config> AudioEncoderL16::SdpToConfig(
>     const SdpAudioFormat& format) {
>   if (!rtc::IsValueInRangeForNumericType<int>(format.num_channels)) {
>-    return rtc::nullopt;
>+    return absl::nullopt;
>   }
>   Config config;
>   config.sample_rate_hz = format.clockrate_hz;
>   config.num_channels = rtc::dchecked_cast<int>(format.num_channels);
>+  auto ptime_iter = format.parameters.find("ptime");
>+  if (ptime_iter != format.parameters.end()) {
>+    const auto ptime = rtc::StringToNumber<int>(ptime_iter->second);
>+    if (ptime && *ptime > 0) {
>+      config.frame_size_ms = rtc::SafeClamp(10 * (*ptime / 10), 10, 60);
>+    }
>+  }
>   return STR_CASE_CMP(format.name.c_str(), "L16") == 0 && config.IsOk()
>-             ? rtc::Optional<Config>(config)
>-             : rtc::nullopt;
>+             ? absl::optional<Config>(config)
>+             : absl::nullopt;
> }
> 
> void AudioEncoderL16::AppendSupportedEncoders(
>@@ -46,14 +55,15 @@ AudioCodecInfo AudioEncoderL16::QueryAudioEncoder(
> 
> std::unique_ptr<AudioEncoder> AudioEncoderL16::MakeAudioEncoder(
>     const AudioEncoderL16::Config& config,
>-    int payload_type) {
>+    int payload_type,
>+    absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
>   RTC_DCHECK(config.IsOk());
>   AudioEncoderPcm16B::Config c;
>   c.sample_rate_hz = config.sample_rate_hz;
>   c.num_channels = config.num_channels;
>   c.frame_size_ms = config.frame_size_ms;
>   c.payload_type = payload_type;
>-  return rtc::MakeUnique<AudioEncoderPcm16B>(c);
>+  return absl::make_unique<AudioEncoderPcm16B>(c);
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/L16/audio_encoder_L16.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/L16/audio_encoder_L16.h
>index e099bd5747d..340e3af47fb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/L16/audio_encoder_L16.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/L16/audio_encoder_L16.h
>@@ -14,9 +14,10 @@
> #include <memory>
> #include <vector>
> 
>+#include "absl/types/optional.h"
>+#include "api/audio_codecs/audio_codec_pair_id.h"
> #include "api/audio_codecs/audio_encoder.h"
> #include "api/audio_codecs/audio_format.h"
>-#include "api/optional.h"
> 
> namespace webrtc {
> 
>@@ -36,11 +37,13 @@ struct AudioEncoderL16 {
>     int num_channels = 1;
>     int frame_size_ms = 10;
>   };
>-  static rtc::Optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
>+  static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
>   static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
>   static AudioCodecInfo QueryAudioEncoder(const Config& config);
>-  static std::unique_ptr<AudioEncoder> MakeAudioEncoder(const Config& config,
>-                                                        int payload_type);
>+  static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
>+      const Config& config,
>+      int payload_type,
>+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt);
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_codec_pair_id.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_codec_pair_id.cc
>new file mode 100644
>index 00000000000..ac84107eda1
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_codec_pair_id.cc
>@@ -0,0 +1,90 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/audio_codecs/audio_codec_pair_id.h"
>+
>+#include <atomic>
>+
>+#include "rtc_base/checks.h"
>+
>+namespace webrtc {
>+
>+namespace {
>+
>+// Returns a new value that it has never returned before. You may call it at
>+// most 2^63 times in the lifetime of the program. Note: The returned values
>+// may be easily predictable.
>+uint64_t GetNextId() {
>+  static std::atomic<uint64_t> next_id(0);
>+
>+  // Atomically increment `next_id`, and return the previous value. Relaxed
>+  // memory order is sufficient, since all we care about is that different
>+  // callers return different values.
>+  const uint64_t new_id = next_id.fetch_add(1, std::memory_order_relaxed);
>+
>+  // This check isn't atomic with the increment, so if we start 2^63 + 1
>+  // invocations of GetNextId() in parallel, the last one to do the atomic
>+  // increment could return the ID 0 before any of the others had time to
>+  // trigger this DCHECK. We blithely assume that this won't happen.
>+  RTC_DCHECK_LT(new_id, uint64_t{1} << 63) << "Used up all ID values";
>+
>+  return new_id;
>+}
>+
>+// Make an integer ID more unpredictable. This is a 1:1 mapping, so you can
>+// feed it any value, but the idea is that you can feed it a sequence such as
>+// 0, 1, 2, ... and get a new sequence that isn't as trivially predictable, so
>+// that users won't rely on it being consecutive or increasing or anything like
>+// that.
>+constexpr uint64_t ObfuscateId(uint64_t id) {
>+  // Any nonzero coefficient that's relatively prime to 2^64 (that is, any odd
>+  // number) and any constant will give a 1:1 mapping. These high-entropy
>+  // values will prevent the sequence from being trivially predictable.
>+  //
>+  // Both the multiplication and the addition going to overflow almost always,
>+  // but that's fine---we *want* arithmetic mod 2^64.
>+  return uint64_t{0x85fdb20e1294309a} + uint64_t{0xc516ef5c37462469} * id;
>+}
>+
>+// The first ten values. Verified against the Python function
>+//
>+//   def f(n):
>+//     return (0x85fdb20e1294309a + 0xc516ef5c37462469 * n) % 2**64
>+//
>+// Callers should obviously not depend on these exact values...
>+//
>+// (On Visual C++, we have to disable warning C4307 (integral constant
>+// overflow), even though unsigned integers have perfectly well-defined
>+// overflow behavior.)
>+#ifdef _MSC_VER
>+#pragma warning(push)
>+#pragma warning(disable : 4307)
>+#endif
>+static_assert(ObfuscateId(0) == uint64_t{0x85fdb20e1294309a}, "");
>+static_assert(ObfuscateId(1) == uint64_t{0x4b14a16a49da5503}, "");
>+static_assert(ObfuscateId(2) == uint64_t{0x102b90c68120796c}, "");
>+static_assert(ObfuscateId(3) == uint64_t{0xd5428022b8669dd5}, "");
>+static_assert(ObfuscateId(4) == uint64_t{0x9a596f7eefacc23e}, "");
>+static_assert(ObfuscateId(5) == uint64_t{0x5f705edb26f2e6a7}, "");
>+static_assert(ObfuscateId(6) == uint64_t{0x24874e375e390b10}, "");
>+static_assert(ObfuscateId(7) == uint64_t{0xe99e3d93957f2f79}, "");
>+static_assert(ObfuscateId(8) == uint64_t{0xaeb52cefccc553e2}, "");
>+static_assert(ObfuscateId(9) == uint64_t{0x73cc1c4c040b784b}, "");
>+#ifdef _MSC_VER
>+#pragma warning(pop)
>+#endif
>+
>+}  // namespace
>+
>+AudioCodecPairId AudioCodecPairId::Create() {
>+  return AudioCodecPairId(ObfuscateId(GetNextId()));
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_codec_pair_id.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_codec_pair_id.h
>new file mode 100644
>index 00000000000..b10f14ea66d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_codec_pair_id.h
>@@ -0,0 +1,74 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_AUDIO_CODECS_AUDIO_CODEC_PAIR_ID_H_
>+#define API_AUDIO_CODECS_AUDIO_CODEC_PAIR_ID_H_
>+
>+#include <stdint.h>
>+
>+#include <utility>
>+
>+namespace webrtc {
>+
>+class AudioCodecPairId final {
>+ public:
>+  // Copyable, but not default constructible.
>+  AudioCodecPairId() = delete;
>+  AudioCodecPairId(const AudioCodecPairId&) = default;
>+  AudioCodecPairId(AudioCodecPairId&&) = default;
>+  AudioCodecPairId& operator=(const AudioCodecPairId&) = default;
>+  AudioCodecPairId& operator=(AudioCodecPairId&&) = default;
>+
>+  friend void swap(AudioCodecPairId& a, AudioCodecPairId& b) {
>+    using std::swap;
>+    swap(a.id_, b.id_);
>+  }
>+
>+  // Creates a new ID, unequal to any previously created ID.
>+  static AudioCodecPairId Create();
>+
>+  // IDs can be tested for equality.
>+  friend bool operator==(AudioCodecPairId a, AudioCodecPairId b) {
>+    return a.id_ == b.id_;
>+  }
>+  friend bool operator!=(AudioCodecPairId a, AudioCodecPairId b) {
>+    return a.id_ != b.id_;
>+  }
>+
>+  // Comparisons. The ordering of ID values is completely arbitrary, but
>+  // stable, so it's useful e.g. if you want to use IDs as keys in an ordered
>+  // map.
>+  friend bool operator<(AudioCodecPairId a, AudioCodecPairId b) {
>+    return a.id_ < b.id_;
>+  }
>+  friend bool operator<=(AudioCodecPairId a, AudioCodecPairId b) {
>+    return a.id_ <= b.id_;
>+  }
>+  friend bool operator>=(AudioCodecPairId a, AudioCodecPairId b) {
>+    return a.id_ >= b.id_;
>+  }
>+  friend bool operator>(AudioCodecPairId a, AudioCodecPairId b) {
>+    return a.id_ > b.id_;
>+  }
>+
>+  // Returns a numeric representation of the ID. The numeric values are
>+  // completely arbitrary, but stable, collision-free, and reasonably evenly
>+  // distributed, so they are e.g. useful as hash values in unordered maps.
>+  uint64_t NumericRepresentation() const { return id_; }
>+
>+ private:
>+  explicit AudioCodecPairId(uint64_t id) : id_(id) {}
>+
>+  uint64_t id_;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // API_AUDIO_CODECS_AUDIO_CODEC_PAIR_ID_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_decoder.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_decoder.cc
>index ddb06d27eeb..00e45d960a4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_decoder.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_decoder.cc
>@@ -33,14 +33,14 @@ class OldStyleEncodedFrame final : public AudioDecoder::EncodedAudioFrame {
>     return ret < 0 ? 0 : static_cast<size_t>(ret);
>   }
> 
>-  rtc::Optional<DecodeResult> Decode(
>+  absl::optional<DecodeResult> Decode(
>       rtc::ArrayView<int16_t> decoded) const override {
>     auto speech_type = AudioDecoder::kSpeech;
>     const int ret = decoder_->Decode(
>         payload_.data(), payload_.size(), decoder_->SampleRateHz(),
>         decoded.size() * sizeof(int16_t), decoded.data(), &speech_type);
>-    return ret < 0 ? rtc::nullopt
>-                   : rtc::Optional<DecodeResult>(
>+    return ret < 0 ? absl::nullopt
>+                   : absl::optional<DecodeResult>(
>                          {static_cast<size_t>(ret), speech_type});
>   }
> 
>@@ -51,6 +51,10 @@ class OldStyleEncodedFrame final : public AudioDecoder::EncodedAudioFrame {
> 
> }  // namespace
> 
>+bool AudioDecoder::EncodedAudioFrame::IsDtxPacket() const {
>+  return false;
>+}
>+
> AudioDecoder::ParseResult::ParseResult() = default;
> AudioDecoder::ParseResult::ParseResult(ParseResult&& b) = default;
> AudioDecoder::ParseResult::ParseResult(uint32_t timestamp,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_decoder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_decoder.h
>index 545bdf52cc0..9a955a6ecc0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_decoder.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_decoder.h
>@@ -14,11 +14,10 @@
> #include <memory>
> #include <vector>
> 
>+#include "absl/types/optional.h"
> #include "api/array_view.h"
>-#include "api/optional.h"
> #include "rtc_base/buffer.h"
> #include "rtc_base/constructormagic.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -48,13 +47,16 @@ class AudioDecoder {
>     // If no duration can be ascertained, returns zero.
>     virtual size_t Duration() const = 0;
> 
>+    // Returns true if this packet contains DTX.
>+    virtual bool IsDtxPacket() const;
>+
>     // Decodes this frame of audio and writes the result in |decoded|.
>     // |decoded| must be large enough to store as many samples as indicated by a
>-    // call to Duration() . On success, returns an rtc::Optional containing the
>+    // call to Duration() . On success, returns an absl::optional containing the
>     // total number of samples across all channels, as well as whether the
>     // decoder produced comfort noise or speech. On failure, returns an empty
>-    // rtc::Optional. Decode may be called at most once per frame object.
>-    virtual rtc::Optional<DecodeResult> Decode(
>+    // absl::optional. Decode may be called at most once per frame object.
>+    virtual absl::optional<DecodeResult> Decode(
>         rtc::ArrayView<int16_t> decoded) const = 0;
>   };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_decoder_factory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_decoder_factory.h
>index ac0f4519d8e..90f93f0cd5f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_decoder_factory.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_decoder_factory.h
>@@ -14,6 +14,8 @@
> #include <memory>
> #include <vector>
> 
>+#include "absl/types/optional.h"
>+#include "api/audio_codecs/audio_codec_pair_id.h"
> #include "api/audio_codecs/audio_decoder.h"
> #include "api/audio_codecs/audio_format.h"
> #include "rtc_base/refcount.h"
>@@ -28,8 +30,18 @@ class AudioDecoderFactory : public rtc::RefCountInterface {
> 
>   virtual bool IsSupportedDecoder(const SdpAudioFormat& format) = 0;
> 
>+  // Create a new decoder instance. The `codec_pair_id` argument is used to
>+  // link encoders and decoders that talk to the same remote entity; if a
>+  // MakeAudioEncoder() and a MakeAudioDecoder() call receive non-null IDs that
>+  // compare equal, the factory implementations may assume that the encoder and
>+  // decoder form a pair.
>+  //
>+  // Note: Implementations need to be robust against combinations other than
>+  // one encoder, one decoder getting the same ID; such decoders must still
>+  // work.
>   virtual std::unique_ptr<AudioDecoder> MakeAudioDecoder(
>-      const SdpAudioFormat& format) = 0;
>+      const SdpAudioFormat& format,
>+      absl::optional<AudioCodecPairId> codec_pair_id) = 0;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_decoder_factory_template.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_decoder_factory_template.h
>index f40c271ec8a..cdbe8bde7a8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_decoder_factory_template.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_decoder_factory_template.h
>@@ -31,7 +31,8 @@ struct Helper<> {
>   static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs) {}
>   static bool IsSupportedDecoder(const SdpAudioFormat& format) { return false; }
>   static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
>-      const SdpAudioFormat& format) {
>+      const SdpAudioFormat& format,
>+      absl::optional<AudioCodecPairId> codec_pair_id) {
>     return nullptr;
>   }
> };
>@@ -47,16 +48,17 @@ struct Helper<T, Ts...> {
>   static bool IsSupportedDecoder(const SdpAudioFormat& format) {
>     auto opt_config = T::SdpToConfig(format);
>     static_assert(std::is_same<decltype(opt_config),
>-                               rtc::Optional<typename T::Config>>::value,
>+                               absl::optional<typename T::Config>>::value,
>                   "T::SdpToConfig() must return a value of type "
>-                  "rtc::Optional<T::Config>");
>+                  "absl::optional<T::Config>");
>     return opt_config ? true : Helper<Ts...>::IsSupportedDecoder(format);
>   }
>   static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
>-      const SdpAudioFormat& format) {
>+      const SdpAudioFormat& format,
>+      absl::optional<AudioCodecPairId> codec_pair_id) {
>     auto opt_config = T::SdpToConfig(format);
>-    return opt_config ? T::MakeAudioDecoder(*opt_config)
>-                      : Helper<Ts...>::MakeAudioDecoder(format);
>+    return opt_config ? T::MakeAudioDecoder(*opt_config, codec_pair_id)
>+                      : Helper<Ts...>::MakeAudioDecoder(format, codec_pair_id);
>   }
> };
> 
>@@ -74,8 +76,9 @@ class AudioDecoderFactoryT : public AudioDecoderFactory {
>   }
> 
>   std::unique_ptr<AudioDecoder> MakeAudioDecoder(
>-      const SdpAudioFormat& format) override {
>-    return Helper<Ts...>::MakeAudioDecoder(format);
>+      const SdpAudioFormat& format,
>+      absl::optional<AudioCodecPairId> codec_pair_id) override {
>+    return Helper<Ts...>::MakeAudioDecoder(format, codec_pair_id);
>   }
> };
> 
>@@ -89,7 +92,7 @@ class AudioDecoderFactoryT : public AudioDecoderFactory {
> //   // Converts |audio_format| to a ConfigType instance. Returns an empty
> //   // optional if |audio_format| doesn't correctly specify an decoder of our
> //   // type.
>-//   rtc::Optional<ConfigType> SdpToConfig(const SdpAudioFormat& audio_format);
>+//   absl::optional<ConfigType> SdpToConfig(const SdpAudioFormat& audio_format);
> //
> //   // Appends zero or more AudioCodecSpecs to the list that will be returned
> //   // by AudioDecoderFactory::GetSupportedDecoders().
>@@ -97,7 +100,9 @@ class AudioDecoderFactoryT : public AudioDecoderFactory {
> //
> //   // Creates an AudioDecoder for the specified format. Used to implement
> //   // AudioDecoderFactory::MakeAudioDecoder().
>-//   std::unique_ptr<AudioDecoder> MakeAudioDecoder(const ConfigType& config);
>+//   std::unique_ptr<AudioDecoder> MakeAudioDecoder(
>+//       const ConfigType& config,
>+//       absl::optional<AudioCodecPairId> codec_pair_id);
> //
> // ConfigType should be a type that encapsulates all the settings needed to
> // create an AudioDecoder. T::Config (where T is the decoder struct) should
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_encoder.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_encoder.cc
>index 4f9b9f0bb28..595c11131e7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_encoder.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_encoder.cc
>@@ -85,12 +85,12 @@ void AudioEncoder::OnReceivedUplinkRecoverablePacketLossFraction(
>     float uplink_recoverable_packet_loss_fraction) {}
> 
> void AudioEncoder::OnReceivedTargetAudioBitrate(int target_audio_bitrate_bps) {
>-  OnReceivedUplinkBandwidth(target_audio_bitrate_bps, rtc::nullopt);
>+  OnReceivedUplinkBandwidth(target_audio_bitrate_bps, absl::nullopt);
> }
> 
> void AudioEncoder::OnReceivedUplinkBandwidth(
>     int target_audio_bitrate_bps,
>-    rtc::Optional<int64_t> bwe_period_ms) {}
>+    absl::optional<int64_t> bwe_period_ms) {}
> 
> void AudioEncoder::OnReceivedRtt(int rtt_ms) {}
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_encoder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_encoder.h
>index 7ad9ba4d095..6a1f6534b21 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_encoder.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_encoder.h
>@@ -16,11 +16,10 @@
> #include <string>
> #include <vector>
> 
>+#include "absl/types/optional.h"
> #include "api/array_view.h"
>-#include "api/optional.h"
> #include "rtc_base/buffer.h"
> #include "rtc_base/deprecation.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -34,30 +33,30 @@ struct ANAStats {
>   // Number of actions taken by the ANA bitrate controller since the start of
>   // the call. If this value is not set, it indicates that the bitrate
>   // controller is disabled.
>-  rtc::Optional<uint32_t> bitrate_action_counter;
>+  absl::optional<uint32_t> bitrate_action_counter;
>   // Number of actions taken by the ANA channel controller since the start of
>   // the call. If this value is not set, it indicates that the channel
>   // controller is disabled.
>-  rtc::Optional<uint32_t> channel_action_counter;
>+  absl::optional<uint32_t> channel_action_counter;
>   // Number of actions taken by the ANA DTX controller since the start of the
>   // call. If this value is not set, it indicates that the DTX controller is
>   // disabled.
>-  rtc::Optional<uint32_t> dtx_action_counter;
>+  absl::optional<uint32_t> dtx_action_counter;
>   // Number of actions taken by the ANA FEC controller since the start of the
>   // call. If this value is not set, it indicates that the FEC controller is
>   // disabled.
>-  rtc::Optional<uint32_t> fec_action_counter;
>+  absl::optional<uint32_t> fec_action_counter;
>   // Number of times the ANA frame length controller decided to increase the
>   // frame length since the start of the call. If this value is not set, it
>   // indicates that the frame length controller is disabled.
>-  rtc::Optional<uint32_t> frame_length_increase_counter;
>+  absl::optional<uint32_t> frame_length_increase_counter;
>   // Number of times the ANA frame length controller decided to decrease the
>   // frame length since the start of the call. If this value is not set, it
>   // indicates that the frame length controller is disabled.
>-  rtc::Optional<uint32_t> frame_length_decrease_counter;
>+  absl::optional<uint32_t> frame_length_decrease_counter;
>   // The uplink packet loss fractions as set by the ANA FEC controller. If this
>   // value is not set, it indicates that the ANA FEC controller is not active.
>-  rtc::Optional<float> uplink_packet_loss_fraction;
>+  absl::optional<float> uplink_packet_loss_fraction;
> };
> 
> // This is the interface class for encoders in AudioCoding module. Each codec
>@@ -220,9 +219,8 @@ class AudioEncoder {
> 
>   // Provides target audio bitrate and corresponding probing interval of
>   // the bandwidth estimator to this encoder to allow it to adapt.
>-  virtual void OnReceivedUplinkBandwidth(
>-      int target_audio_bitrate_bps,
>-      rtc::Optional<int64_t> bwe_period_ms);
>+  virtual void OnReceivedUplinkBandwidth(int target_audio_bitrate_bps,
>+                                         absl::optional<int64_t> bwe_period_ms);
> 
>   // Provides RTT to this encoder to allow it to adapt.
>   virtual void OnReceivedRtt(int rtt_ms);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_encoder_factory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_encoder_factory.h
>index 43461f6b9a6..fb4e23ffe78 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_encoder_factory.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_encoder_factory.h
>@@ -14,6 +14,8 @@
> #include <memory>
> #include <vector>
> 
>+#include "absl/types/optional.h"
>+#include "api/audio_codecs/audio_codec_pair_id.h"
> #include "api/audio_codecs/audio_encoder.h"
> #include "api/audio_codecs/audio_format.h"
> #include "rtc_base/refcount.h"
>@@ -30,15 +32,25 @@ class AudioEncoderFactory : public rtc::RefCountInterface {
>   // Returns information about how this format would be encoded, provided it's
>   // supported. More format and format variations may be supported than those
>   // returned by GetSupportedEncoders().
>-  virtual rtc::Optional<AudioCodecInfo> QueryAudioEncoder(
>+  virtual absl::optional<AudioCodecInfo> QueryAudioEncoder(
>       const SdpAudioFormat& format) = 0;
> 
>-  // Creates an AudioEncoder for the specified format. The encoder will tags its
>-  // payloads with the specified payload type.
>+  // Creates an AudioEncoder for the specified format. The encoder will tags
>+  // its payloads with the specified payload type. The `codec_pair_id` argument
>+  // is used to link encoders and decoders that talk to the same remote entity;
>+  // if a MakeAudioEncoder() and a MakeAudioDecoder() call receive non-null IDs
>+  // that compare equal, the factory implementations may assume that the
>+  // encoder and decoder form a pair.
>+  //
>+  // Note: Implementations need to be robust against combinations other than
>+  // one encoder, one decoder getting the same ID; such encoders must still
>+  // work.
>+  //
>   // TODO(ossu): Try to avoid audio encoders having to know their payload type.
>   virtual std::unique_ptr<AudioEncoder> MakeAudioEncoder(
>       int payload_type,
>-      const SdpAudioFormat& format) = 0;
>+      const SdpAudioFormat& format,
>+      absl::optional<AudioCodecPairId> codec_pair_id) = 0;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_encoder_factory_template.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_encoder_factory_template.h
>index 060ba8cb045..376b39e4c6f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_encoder_factory_template.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_encoder_factory_template.h
>@@ -29,13 +29,14 @@ struct Helper;
> template <>
> struct Helper<> {
>   static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs) {}
>-  static rtc::Optional<AudioCodecInfo> QueryAudioEncoder(
>+  static absl::optional<AudioCodecInfo> QueryAudioEncoder(
>       const SdpAudioFormat& format) {
>-    return rtc::nullopt;
>+    return absl::nullopt;
>   }
>   static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
>       int payload_type,
>-      const SdpAudioFormat& format) {
>+      const SdpAudioFormat& format,
>+      absl::optional<AudioCodecPairId> codec_pair_id) {
>     return nullptr;
>   }
> };
>@@ -48,25 +49,27 @@ struct Helper<T, Ts...> {
>     T::AppendSupportedEncoders(specs);
>     Helper<Ts...>::AppendSupportedEncoders(specs);
>   }
>-  static rtc::Optional<AudioCodecInfo> QueryAudioEncoder(
>+  static absl::optional<AudioCodecInfo> QueryAudioEncoder(
>       const SdpAudioFormat& format) {
>     auto opt_config = T::SdpToConfig(format);
>     static_assert(std::is_same<decltype(opt_config),
>-                               rtc::Optional<typename T::Config>>::value,
>+                               absl::optional<typename T::Config>>::value,
>                   "T::SdpToConfig() must return a value of type "
>-                  "rtc::Optional<T::Config>");
>-    return opt_config ? rtc::Optional<AudioCodecInfo>(
>+                  "absl::optional<T::Config>");
>+    return opt_config ? absl::optional<AudioCodecInfo>(
>                             T::QueryAudioEncoder(*opt_config))
>                       : Helper<Ts...>::QueryAudioEncoder(format);
>   }
>   static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
>       int payload_type,
>-      const SdpAudioFormat& format) {
>+      const SdpAudioFormat& format,
>+      absl::optional<AudioCodecPairId> codec_pair_id) {
>     auto opt_config = T::SdpToConfig(format);
>     if (opt_config) {
>-      return T::MakeAudioEncoder(*opt_config, payload_type);
>+      return T::MakeAudioEncoder(*opt_config, payload_type, codec_pair_id);
>     } else {
>-      return Helper<Ts...>::MakeAudioEncoder(payload_type, format);
>+      return Helper<Ts...>::MakeAudioEncoder(payload_type, format,
>+                                             codec_pair_id);
>     }
>   }
> };
>@@ -80,15 +83,16 @@ class AudioEncoderFactoryT : public AudioEncoderFactory {
>     return specs;
>   }
> 
>-  rtc::Optional<AudioCodecInfo> QueryAudioEncoder(
>+  absl::optional<AudioCodecInfo> QueryAudioEncoder(
>       const SdpAudioFormat& format) override {
>     return Helper<Ts...>::QueryAudioEncoder(format);
>   }
> 
>   std::unique_ptr<AudioEncoder> MakeAudioEncoder(
>       int payload_type,
>-      const SdpAudioFormat& format) override {
>-    return Helper<Ts...>::MakeAudioEncoder(payload_type, format);
>+      const SdpAudioFormat& format,
>+      absl::optional<AudioCodecPairId> codec_pair_id) override {
>+    return Helper<Ts...>::MakeAudioEncoder(payload_type, format, codec_pair_id);
>   }
> };
> 
>@@ -102,7 +106,7 @@ class AudioEncoderFactoryT : public AudioEncoderFactory {
> //   // Converts |audio_format| to a ConfigType instance. Returns an empty
> //   // optional if |audio_format| doesn't correctly specify an encoder of our
> //   // type.
>-//   rtc::Optional<ConfigType> SdpToConfig(const SdpAudioFormat& audio_format);
>+//   absl::optional<ConfigType> SdpToConfig(const SdpAudioFormat& audio_format);
> //
> //   // Appends zero or more AudioCodecSpecs to the list that will be returned
> //   // by AudioEncoderFactory::GetSupportedEncoders().
>@@ -114,8 +118,10 @@ class AudioEncoderFactoryT : public AudioEncoderFactory {
> //
> //   // Creates an AudioEncoder for the specified format. Used to implement
> //   // AudioEncoderFactory::MakeAudioEncoder().
>-//   std::unique_ptr<AudioEncoder> MakeAudioEncoder(const ConfigType& config,
>-//                                                  int payload_type);
>+//   std::unique_ptr<AudioDecoder> MakeAudioEncoder(
>+//       const ConfigType& config,
>+//       int payload_type,
>+//       absl::optional<AudioCodecPairId> codec_pair_id);
> //
> // ConfigType should be a type that encapsulates all the settings needed to
> // create an AudioEncoder. T::Config (where T is the encoder struct) should
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_format.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_format.cc
>index 82c166f5c00..9db5ce01dc8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_format.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_format.cc
>@@ -68,20 +68,6 @@ void swap(SdpAudioFormat& a, SdpAudioFormat& b) {
>   swap(a.parameters, b.parameters);
> }
> 
>-std::ostream& operator<<(std::ostream& os, const SdpAudioFormat& saf) {
>-  os << "{name: " << saf.name;
>-  os << ", clockrate_hz: " << saf.clockrate_hz;
>-  os << ", num_channels: " << saf.num_channels;
>-  os << ", parameters: {";
>-  const char* sep = "";
>-  for (const auto& kv : saf.parameters) {
>-    os << sep << kv.first << ": " << kv.second;
>-    sep = ", ";
>-  }
>-  os << "}}";
>-  return os;
>-}
>-
> AudioCodecInfo::AudioCodecInfo(int sample_rate_hz,
>                                size_t num_channels,
>                                int bitrate_bps)
>@@ -108,23 +94,4 @@ AudioCodecInfo::AudioCodecInfo(int sample_rate_hz,
>   RTC_DCHECK_GE(max_bitrate_bps, default_bitrate_bps);
> }
> 
>-std::ostream& operator<<(std::ostream& os, const AudioCodecInfo& aci) {
>-  os << "{sample_rate_hz: " << aci.sample_rate_hz;
>-  os << ", num_channels: " << aci.num_channels;
>-  os << ", default_bitrate_bps: " << aci.default_bitrate_bps;
>-  os << ", min_bitrate_bps: " << aci.min_bitrate_bps;
>-  os << ", max_bitrate_bps: " << aci.max_bitrate_bps;
>-  os << ", allow_comfort_noise: " << aci.allow_comfort_noise;
>-  os << ", supports_network_adaption: " << aci.supports_network_adaption;
>-  os << "}";
>-  return os;
>-}
>-
>-std::ostream& operator<<(std::ostream& os, const AudioCodecSpec& acs) {
>-  os << "{format: " << acs.format;
>-  os << ", info: " << acs.info;
>-  os << "}";
>-  return os;
>-}
>-
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_format.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_format.h
>index 12e9552e933..d132067d095 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_format.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/audio_format.h
>@@ -12,11 +12,11 @@
> #define API_AUDIO_CODECS_AUDIO_FORMAT_H_
> 
> #include <map>
>-#include <ostream>
> #include <string>
> #include <utility>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
>+#include "rtc_base/checks.h"
> 
> namespace webrtc {
> 
>@@ -61,7 +61,6 @@ struct SdpAudioFormat {
> };
> 
> void swap(SdpAudioFormat& a, SdpAudioFormat& b);
>-std::ostream& operator<<(std::ostream& os, const SdpAudioFormat& saf);
> 
> // Information about how an audio format is treated by the codec implementation.
> // Contains basic information, such as sample rate and number of channels, which
>@@ -120,8 +119,6 @@ struct AudioCodecInfo {
>                                            // network conditions.
> };
> 
>-std::ostream& operator<<(std::ostream& os, const AudioCodecInfo& aci);
>-
> // AudioCodecSpec ties an audio format to specific information about the codec
> // and its implementation.
> struct AudioCodecSpec {
>@@ -135,8 +132,6 @@ struct AudioCodecSpec {
>   AudioCodecInfo info;
> };
> 
>-std::ostream& operator<<(std::ostream& os, const AudioCodecSpec& acs);
>-
> }  // namespace webrtc
> 
> #endif  // API_AUDIO_CODECS_AUDIO_FORMAT_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/builtin_audio_decoder_factory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/builtin_audio_decoder_factory.cc
>index 9520d2a9e73..e3ca1b0e06d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/builtin_audio_decoder_factory.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/builtin_audio_decoder_factory.cc
>@@ -33,14 +33,17 @@ namespace {
> template <typename T>
> struct NotAdvertised {
>   using Config = typename T::Config;
>-  static rtc::Optional<Config> SdpToConfig(const SdpAudioFormat& audio_format) {
>+  static absl::optional<Config> SdpToConfig(
>+      const SdpAudioFormat& audio_format) {
>     return T::SdpToConfig(audio_format);
>   }
>   static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs) {
>     // Don't advertise support for anything.
>   }
>-  static std::unique_ptr<AudioDecoder> MakeAudioDecoder(const Config& config) {
>-    return T::MakeAudioDecoder(config);
>+  static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
>+      const Config& config,
>+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt) {
>+    return T::MakeAudioDecoder(config, codec_pair_id);
>   }
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/builtin_audio_encoder_factory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/builtin_audio_encoder_factory.cc
>index 877f85026f3..c0caff42e28 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/builtin_audio_encoder_factory.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/builtin_audio_encoder_factory.cc
>@@ -33,7 +33,8 @@ namespace {
> template <typename T>
> struct NotAdvertised {
>   using Config = typename T::Config;
>-  static rtc::Optional<Config> SdpToConfig(const SdpAudioFormat& audio_format) {
>+  static absl::optional<Config> SdpToConfig(
>+      const SdpAudioFormat& audio_format) {
>     return T::SdpToConfig(audio_format);
>   }
>   static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs) {
>@@ -42,9 +43,11 @@ struct NotAdvertised {
>   static AudioCodecInfo QueryAudioEncoder(const Config& config) {
>     return T::QueryAudioEncoder(config);
>   }
>-  static std::unique_ptr<AudioEncoder> MakeAudioEncoder(const Config& config,
>-                                                        int payload_type) {
>-    return T::MakeAudioEncoder(config, payload_type);
>+  static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
>+      const Config& config,
>+      int payload_type,
>+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt) {
>+    return T::MakeAudioEncoder(config, payload_type, codec_pair_id);
>   }
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g711/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g711/BUILD.gn
>index b62c6790769..169172a6682 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g711/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g711/BUILD.gn
>@@ -14,30 +14,35 @@ if (is_android) {
> 
> rtc_static_library("audio_encoder_g711") {
>   visibility = [ "*" ]
>+  poisonous = [ "audio_codecs" ]
>   sources = [
>     "audio_encoder_g711.cc",
>     "audio_encoder_g711.h",
>   ]
>   deps = [
>     "..:audio_codecs_api",
>-    "../..:optional",
>     "../../..:webrtc_common",
>     "../../../modules/audio_coding:g711",
>     "../../../rtc_base:rtc_base_approved",
>+    "../../../rtc_base:safe_minmax",
>+    "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
> 
> rtc_static_library("audio_decoder_g711") {
>   visibility = [ "*" ]
>+  poisonous = [ "audio_codecs" ]
>   sources = [
>     "audio_decoder_g711.cc",
>     "audio_decoder_g711.h",
>   ]
>   deps = [
>     "..:audio_codecs_api",
>-    "../..:optional",
>     "../../..:webrtc_common",
>     "../../../modules/audio_coding:g711",
>     "../../../rtc_base:rtc_base_approved",
>+    "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g711/audio_decoder_g711.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g711/audio_decoder_g711.cc
>index 71d363be735..cb16584cdee 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g711/audio_decoder_g711.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g711/audio_decoder_g711.cc
>@@ -13,14 +13,14 @@
> #include <memory>
> #include <vector>
> 
>+#include "absl/memory/memory.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
> #include "rtc_base/numerics/safe_conversions.h"
>-#include "rtc_base/ptr_util.h"
> 
> namespace webrtc {
> 
>-rtc::Optional<AudioDecoderG711::Config> AudioDecoderG711::SdpToConfig(
>+absl::optional<AudioDecoderG711::Config> AudioDecoderG711::SdpToConfig(
>     const SdpAudioFormat& format) {
>   const bool is_pcmu = STR_CASE_CMP(format.name.c_str(), "PCMU") == 0;
>   const bool is_pcma = STR_CASE_CMP(format.name.c_str(), "PCMA") == 0;
>@@ -32,7 +32,7 @@ rtc::Optional<AudioDecoderG711::Config> AudioDecoderG711::SdpToConfig(
>     RTC_DCHECK(config.IsOk());
>     return config;
>   } else {
>-    return rtc::nullopt;
>+    return absl::nullopt;
>   }
> }
> 
>@@ -44,13 +44,14 @@ void AudioDecoderG711::AppendSupportedDecoders(
> }
> 
> std::unique_ptr<AudioDecoder> AudioDecoderG711::MakeAudioDecoder(
>-    const Config& config) {
>+    const Config& config,
>+    absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
>   RTC_DCHECK(config.IsOk());
>   switch (config.type) {
>     case Config::Type::kPcmU:
>-      return rtc::MakeUnique<AudioDecoderPcmU>(config.num_channels);
>+      return absl::make_unique<AudioDecoderPcmU>(config.num_channels);
>     case Config::Type::kPcmA:
>-      return rtc::MakeUnique<AudioDecoderPcmA>(config.num_channels);
>+      return absl::make_unique<AudioDecoderPcmA>(config.num_channels);
>     default:
>       return nullptr;
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g711/audio_decoder_g711.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g711/audio_decoder_g711.h
>index 652e23ebcfd..8275a8c7a14 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g711/audio_decoder_g711.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g711/audio_decoder_g711.h
>@@ -14,9 +14,10 @@
> #include <memory>
> #include <vector>
> 
>+#include "absl/types/optional.h"
>+#include "api/audio_codecs/audio_codec_pair_id.h"
> #include "api/audio_codecs/audio_decoder.h"
> #include "api/audio_codecs/audio_format.h"
>-#include "api/optional.h"
> 
> namespace webrtc {
> 
>@@ -33,9 +34,11 @@ struct AudioDecoderG711 {
>     Type type;
>     int num_channels;
>   };
>-  static rtc::Optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
>+  static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
>   static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
>-  static std::unique_ptr<AudioDecoder> MakeAudioDecoder(const Config& config);
>+  static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
>+      const Config& config,
>+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt);
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g711/audio_encoder_g711.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g711/audio_encoder_g711.cc
>index 7029caeaad1..1d5e541276b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g711/audio_encoder_g711.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g711/audio_encoder_g711.cc
>@@ -13,16 +13,16 @@
> #include <memory>
> #include <vector>
> 
>+#include "absl/memory/memory.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
> #include "rtc_base/numerics/safe_conversions.h"
> #include "rtc_base/numerics/safe_minmax.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/string_to_number.h"
> 
> namespace webrtc {
> 
>-rtc::Optional<AudioEncoderG711::Config> AudioEncoderG711::SdpToConfig(
>+absl::optional<AudioEncoderG711::Config> AudioEncoderG711::SdpToConfig(
>     const SdpAudioFormat& format) {
>   const bool is_pcmu = STR_CASE_CMP(format.name.c_str(), "PCMU") == 0;
>   const bool is_pcma = STR_CASE_CMP(format.name.c_str(), "PCMA") == 0;
>@@ -42,7 +42,7 @@ rtc::Optional<AudioEncoderG711::Config> AudioEncoderG711::SdpToConfig(
>     RTC_DCHECK(config.IsOk());
>     return config;
>   } else {
>-    return rtc::nullopt;
>+    return absl::nullopt;
>   }
> }
> 
>@@ -61,7 +61,8 @@ AudioCodecInfo AudioEncoderG711::QueryAudioEncoder(const Config& config) {
> 
> std::unique_ptr<AudioEncoder> AudioEncoderG711::MakeAudioEncoder(
>     const Config& config,
>-    int payload_type) {
>+    int payload_type,
>+    absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
>   RTC_DCHECK(config.IsOk());
>   switch (config.type) {
>     case Config::Type::kPcmU: {
>@@ -69,14 +70,14 @@ std::unique_ptr<AudioEncoder> AudioEncoderG711::MakeAudioEncoder(
>       impl_config.num_channels = config.num_channels;
>       impl_config.frame_size_ms = config.frame_size_ms;
>       impl_config.payload_type = payload_type;
>-      return rtc::MakeUnique<AudioEncoderPcmU>(impl_config);
>+      return absl::make_unique<AudioEncoderPcmU>(impl_config);
>     }
>     case Config::Type::kPcmA: {
>       AudioEncoderPcmA::Config impl_config;
>       impl_config.num_channels = config.num_channels;
>       impl_config.frame_size_ms = config.frame_size_ms;
>       impl_config.payload_type = payload_type;
>-      return rtc::MakeUnique<AudioEncoderPcmA>(impl_config);
>+      return absl::make_unique<AudioEncoderPcmA>(impl_config);
>     }
>     default: { return nullptr; }
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g711/audio_encoder_g711.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g711/audio_encoder_g711.h
>index ecdb9a39017..6b6eb5fce0c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g711/audio_encoder_g711.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g711/audio_encoder_g711.h
>@@ -14,9 +14,10 @@
> #include <memory>
> #include <vector>
> 
>+#include "absl/types/optional.h"
>+#include "api/audio_codecs/audio_codec_pair_id.h"
> #include "api/audio_codecs/audio_encoder.h"
> #include "api/audio_codecs/audio_format.h"
>-#include "api/optional.h"
> 
> namespace webrtc {
> 
>@@ -35,12 +36,14 @@ struct AudioEncoderG711 {
>     int num_channels = 1;
>     int frame_size_ms = 20;
>   };
>-  static rtc::Optional<AudioEncoderG711::Config> SdpToConfig(
>+  static absl::optional<AudioEncoderG711::Config> SdpToConfig(
>       const SdpAudioFormat& audio_format);
>   static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
>   static AudioCodecInfo QueryAudioEncoder(const Config& config);
>-  static std::unique_ptr<AudioEncoder> MakeAudioEncoder(const Config& config,
>-                                                        int payload_type);
>+  static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
>+      const Config& config,
>+      int payload_type,
>+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt);
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g722/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g722/BUILD.gn
>index 0d12195b692..50b1396d291 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g722/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g722/BUILD.gn
>@@ -21,6 +21,7 @@ rtc_source_set("audio_encoder_g722_config") {
> 
> rtc_static_library("audio_encoder_g722") {
>   visibility = [ "*" ]
>+  poisonous = [ "audio_codecs" ]
>   sources = [
>     "audio_encoder_g722.cc",
>     "audio_encoder_g722.h",
>@@ -28,24 +29,28 @@ rtc_static_library("audio_encoder_g722") {
>   deps = [
>     ":audio_encoder_g722_config",
>     "..:audio_codecs_api",
>-    "../..:optional",
>     "../../..:webrtc_common",
>     "../../../modules/audio_coding:g722",
>     "../../../rtc_base:rtc_base_approved",
>+    "../../../rtc_base:safe_minmax",
>+    "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
> 
> rtc_static_library("audio_decoder_g722") {
>   visibility = [ "*" ]
>+  poisonous = [ "audio_codecs" ]
>   sources = [
>     "audio_decoder_g722.cc",
>     "audio_decoder_g722.h",
>   ]
>   deps = [
>     "..:audio_codecs_api",
>-    "../..:optional",
>     "../../..:webrtc_common",
>     "../../../modules/audio_coding:g722",
>     "../../../rtc_base:rtc_base_approved",
>+    "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g722/audio_decoder_g722.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g722/audio_decoder_g722.cc
>index 961b1267fe1..f1e2afbab50 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g722/audio_decoder_g722.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g722/audio_decoder_g722.cc
>@@ -13,21 +13,21 @@
> #include <memory>
> #include <vector>
> 
>+#include "absl/memory/memory.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/audio_coding/codecs/g722/audio_decoder_g722.h"
> #include "rtc_base/numerics/safe_conversions.h"
>-#include "rtc_base/ptr_util.h"
> 
> namespace webrtc {
> 
>-rtc::Optional<AudioDecoderG722::Config> AudioDecoderG722::SdpToConfig(
>+absl::optional<AudioDecoderG722::Config> AudioDecoderG722::SdpToConfig(
>     const SdpAudioFormat& format) {
>   return STR_CASE_CMP(format.name.c_str(), "G722") == 0 &&
>                  format.clockrate_hz == 8000 &&
>                  (format.num_channels == 1 || format.num_channels == 2)
>-             ? rtc::Optional<Config>(
>+             ? absl::optional<Config>(
>                    Config{rtc::dchecked_cast<int>(format.num_channels)})
>-             : rtc::nullopt;
>+             : absl::nullopt;
> }
> 
> void AudioDecoderG722::AppendSupportedDecoders(
>@@ -36,12 +36,13 @@ void AudioDecoderG722::AppendSupportedDecoders(
> }
> 
> std::unique_ptr<AudioDecoder> AudioDecoderG722::MakeAudioDecoder(
>-    Config config) {
>+    Config config,
>+    absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
>   switch (config.num_channels) {
>     case 1:
>-      return rtc::MakeUnique<AudioDecoderG722Impl>();
>+      return absl::make_unique<AudioDecoderG722Impl>();
>     case 2:
>-      return rtc::MakeUnique<AudioDecoderG722StereoImpl>();
>+      return absl::make_unique<AudioDecoderG722StereoImpl>();
>     default:
>       return nullptr;
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g722/audio_decoder_g722.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g722/audio_decoder_g722.h
>index fddb89aaf84..b7bb0893f33 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g722/audio_decoder_g722.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g722/audio_decoder_g722.h
>@@ -14,9 +14,10 @@
> #include <memory>
> #include <vector>
> 
>+#include "absl/types/optional.h"
>+#include "api/audio_codecs/audio_codec_pair_id.h"
> #include "api/audio_codecs/audio_decoder.h"
> #include "api/audio_codecs/audio_format.h"
>-#include "api/optional.h"
> 
> namespace webrtc {
> 
>@@ -29,9 +30,11 @@ struct AudioDecoderG722 {
>     bool IsOk() const { return num_channels == 1 || num_channels == 2; }
>     int num_channels;
>   };
>-  static rtc::Optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
>+  static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
>   static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
>-  static std::unique_ptr<AudioDecoder> MakeAudioDecoder(Config config);
>+  static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
>+      Config config,
>+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt);
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g722/audio_encoder_g722.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g722/audio_encoder_g722.cc
>index f8aa6162d23..0cf71639bf3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g722/audio_encoder_g722.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g722/audio_encoder_g722.cc
>@@ -13,20 +13,20 @@
> #include <memory>
> #include <vector>
> 
>+#include "absl/memory/memory.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/audio_coding/codecs/g722/audio_encoder_g722.h"
> #include "rtc_base/numerics/safe_conversions.h"
> #include "rtc_base/numerics/safe_minmax.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/string_to_number.h"
> 
> namespace webrtc {
> 
>-rtc::Optional<AudioEncoderG722Config> AudioEncoderG722::SdpToConfig(
>+absl::optional<AudioEncoderG722Config> AudioEncoderG722::SdpToConfig(
>     const SdpAudioFormat& format) {
>   if (STR_CASE_CMP(format.name.c_str(), "g722") != 0 ||
>       format.clockrate_hz != 8000) {
>-    return rtc::nullopt;
>+    return absl::nullopt;
>   }
> 
>   AudioEncoderG722Config config;
>@@ -39,8 +39,8 @@ rtc::Optional<AudioEncoderG722Config> AudioEncoderG722::SdpToConfig(
>       config.frame_size_ms = rtc::SafeClamp<int>(whole_packets * 10, 10, 60);
>     }
>   }
>-  return config.IsOk() ? rtc::Optional<AudioEncoderG722Config>(config)
>-                       : rtc::nullopt;
>+  return config.IsOk() ? absl::optional<AudioEncoderG722Config>(config)
>+                       : absl::nullopt;
> }
> 
> void AudioEncoderG722::AppendSupportedEncoders(
>@@ -59,9 +59,10 @@ AudioCodecInfo AudioEncoderG722::QueryAudioEncoder(
> 
> std::unique_ptr<AudioEncoder> AudioEncoderG722::MakeAudioEncoder(
>     const AudioEncoderG722Config& config,
>-    int payload_type) {
>+    int payload_type,
>+    absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
>   RTC_DCHECK(config.IsOk());
>-  return rtc::MakeUnique<AudioEncoderG722Impl>(config, payload_type);
>+  return absl::make_unique<AudioEncoderG722Impl>(config, payload_type);
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g722/audio_encoder_g722.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g722/audio_encoder_g722.h
>index 0763615cf7c..b97fe1b1478 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g722/audio_encoder_g722.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/g722/audio_encoder_g722.h
>@@ -14,10 +14,11 @@
> #include <memory>
> #include <vector>
> 
>+#include "absl/types/optional.h"
>+#include "api/audio_codecs/audio_codec_pair_id.h"
> #include "api/audio_codecs/audio_encoder.h"
> #include "api/audio_codecs/audio_format.h"
> #include "api/audio_codecs/g722/audio_encoder_g722_config.h"
>-#include "api/optional.h"
> 
> namespace webrtc {
> 
>@@ -27,13 +28,14 @@ namespace webrtc {
> // NOTE: This struct is still under development and may change without notice.
> struct AudioEncoderG722 {
>   using Config = AudioEncoderG722Config;
>-  static rtc::Optional<AudioEncoderG722Config> SdpToConfig(
>+  static absl::optional<AudioEncoderG722Config> SdpToConfig(
>       const SdpAudioFormat& audio_format);
>   static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
>   static AudioCodecInfo QueryAudioEncoder(const AudioEncoderG722Config& config);
>   static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
>       const AudioEncoderG722Config& config,
>-      int payload_type);
>+      int payload_type,
>+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt);
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/BUILD.gn
>index 944d4eb8745..d4f504fd4f9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/BUILD.gn
>@@ -21,6 +21,7 @@ rtc_source_set("audio_encoder_ilbc_config") {
> 
> rtc_static_library("audio_encoder_ilbc") {
>   visibility = [ "*" ]
>+  poisonous = [ "audio_codecs" ]
>   sources = [
>     "audio_encoder_ilbc.cc",
>     "audio_encoder_ilbc.h",
>@@ -28,24 +29,28 @@ rtc_static_library("audio_encoder_ilbc") {
>   deps = [
>     ":audio_encoder_ilbc_config",
>     "..:audio_codecs_api",
>-    "../..:optional",
>     "../../..:webrtc_common",
>     "../../../modules/audio_coding:ilbc",
>     "../../../rtc_base:rtc_base_approved",
>+    "../../../rtc_base:safe_minmax",
>+    "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
> 
> rtc_static_library("audio_decoder_ilbc") {
>   visibility = [ "*" ]
>+  poisonous = [ "audio_codecs" ]
>   sources = [
>     "audio_decoder_ilbc.cc",
>     "audio_decoder_ilbc.h",
>   ]
>   deps = [
>     "..:audio_codecs_api",
>-    "../..:optional",
>     "../../..:webrtc_common",
>     "../../../modules/audio_coding:ilbc",
>     "../../../rtc_base:rtc_base_approved",
>+    "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/audio_decoder_ilbc.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/audio_decoder_ilbc.cc
>index 2ffae480ca8..1f4c475c406 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/audio_decoder_ilbc.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/audio_decoder_ilbc.cc
>@@ -13,18 +13,18 @@
> #include <memory>
> #include <vector>
> 
>+#include "absl/memory/memory.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.h"
>-#include "rtc_base/ptr_util.h"
> 
> namespace webrtc {
> 
>-rtc::Optional<AudioDecoderIlbc::Config> AudioDecoderIlbc::SdpToConfig(
>+absl::optional<AudioDecoderIlbc::Config> AudioDecoderIlbc::SdpToConfig(
>     const SdpAudioFormat& format) {
>   return STR_CASE_CMP(format.name.c_str(), "ILBC") == 0 &&
>                  format.clockrate_hz == 8000 && format.num_channels == 1
>-             ? rtc::Optional<Config>(Config())
>-             : rtc::nullopt;
>+             ? absl::optional<Config>(Config())
>+             : absl::nullopt;
> }
> 
> void AudioDecoderIlbc::AppendSupportedDecoders(
>@@ -33,8 +33,9 @@ void AudioDecoderIlbc::AppendSupportedDecoders(
> }
> 
> std::unique_ptr<AudioDecoder> AudioDecoderIlbc::MakeAudioDecoder(
>-    Config config) {
>-  return rtc::MakeUnique<AudioDecoderIlbcImpl>();
>+    Config config,
>+    absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
>+  return absl::make_unique<AudioDecoderIlbcImpl>();
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/audio_decoder_ilbc.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/audio_decoder_ilbc.h
>index f7292d60c63..20f6ffd287f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/audio_decoder_ilbc.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/audio_decoder_ilbc.h
>@@ -14,9 +14,10 @@
> #include <memory>
> #include <vector>
> 
>+#include "absl/types/optional.h"
>+#include "api/audio_codecs/audio_codec_pair_id.h"
> #include "api/audio_codecs/audio_decoder.h"
> #include "api/audio_codecs/audio_format.h"
>-#include "api/optional.h"
> 
> namespace webrtc {
> 
>@@ -26,9 +27,11 @@ namespace webrtc {
> // NOTE: This struct is still under development and may change without notice.
> struct AudioDecoderIlbc {
>   struct Config {};  // Empty---no config values needed!
>-  static rtc::Optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
>+  static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
>   static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
>-  static std::unique_ptr<AudioDecoder> MakeAudioDecoder(Config config);
>+  static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
>+      Config config,
>+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt);
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.cc
>index 6db88402e39..2ae75474ccd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.cc
>@@ -13,11 +13,11 @@
> #include <memory>
> #include <vector>
> 
>+#include "absl/memory/memory.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
> #include "rtc_base/numerics/safe_conversions.h"
> #include "rtc_base/numerics/safe_minmax.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/string_to_number.h"
> 
> namespace webrtc {
>@@ -38,11 +38,11 @@ int GetIlbcBitrate(int ptime) {
> }
> }  // namespace
> 
>-rtc::Optional<AudioEncoderIlbcConfig> AudioEncoderIlbc::SdpToConfig(
>+absl::optional<AudioEncoderIlbcConfig> AudioEncoderIlbc::SdpToConfig(
>     const SdpAudioFormat& format) {
>   if (STR_CASE_CMP(format.name.c_str(), "ILBC") != 0 ||
>       format.clockrate_hz != 8000 || format.num_channels != 1) {
>-    return rtc::nullopt;
>+    return absl::nullopt;
>   }
> 
>   AudioEncoderIlbcConfig config;
>@@ -54,8 +54,8 @@ rtc::Optional<AudioEncoderIlbcConfig> AudioEncoderIlbc::SdpToConfig(
>       config.frame_size_ms = rtc::SafeClamp<int>(whole_packets * 10, 20, 60);
>     }
>   }
>-  return config.IsOk() ? rtc::Optional<AudioEncoderIlbcConfig>(config)
>-                       : rtc::nullopt;
>+  return config.IsOk() ? absl::optional<AudioEncoderIlbcConfig>(config)
>+                       : absl::nullopt;
> }
> 
> void AudioEncoderIlbc::AppendSupportedEncoders(
>@@ -73,9 +73,10 @@ AudioCodecInfo AudioEncoderIlbc::QueryAudioEncoder(
> 
> std::unique_ptr<AudioEncoder> AudioEncoderIlbc::MakeAudioEncoder(
>     const AudioEncoderIlbcConfig& config,
>-    int payload_type) {
>+    int payload_type,
>+    absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
>   RTC_DCHECK(config.IsOk());
>-  return rtc::MakeUnique<AudioEncoderIlbcImpl>(config, payload_type);
>+  return absl::make_unique<AudioEncoderIlbcImpl>(config, payload_type);
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.h
>index dd653757742..0a86b162e81 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.h
>@@ -14,10 +14,11 @@
> #include <memory>
> #include <vector>
> 
>+#include "absl/types/optional.h"
>+#include "api/audio_codecs/audio_codec_pair_id.h"
> #include "api/audio_codecs/audio_encoder.h"
> #include "api/audio_codecs/audio_format.h"
> #include "api/audio_codecs/ilbc/audio_encoder_ilbc_config.h"
>-#include "api/optional.h"
> 
> namespace webrtc {
> 
>@@ -27,13 +28,14 @@ namespace webrtc {
> // NOTE: This struct is still under development and may change without notice.
> struct AudioEncoderIlbc {
>   using Config = AudioEncoderIlbcConfig;
>-  static rtc::Optional<AudioEncoderIlbcConfig> SdpToConfig(
>+  static absl::optional<AudioEncoderIlbcConfig> SdpToConfig(
>       const SdpAudioFormat& audio_format);
>   static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
>   static AudioCodecInfo QueryAudioEncoder(const AudioEncoderIlbcConfig& config);
>   static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
>       const AudioEncoderIlbcConfig& config,
>-      int payload_type);
>+      int payload_type,
>+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt);
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/BUILD.gn
>index f63435c93aa..ed9d9622577 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/BUILD.gn
>@@ -19,6 +19,7 @@ if (is_android) {
> 
> rtc_source_set("audio_encoder_isac") {
>   visibility = [ "*" ]
>+  poisonous = [ "audio_codecs" ]
>   public = [
>     "audio_encoder_isac.h",
>   ]
>@@ -36,6 +37,7 @@ rtc_source_set("audio_encoder_isac") {
> 
> rtc_source_set("audio_decoder_isac") {
>   visibility = [ "*" ]
>+  poisonous = [ "audio_codecs" ]
>   public = [
>     "audio_decoder_isac.h",
>   ]
>@@ -68,60 +70,68 @@ config("isac_config") {
> 
> rtc_static_library("audio_encoder_isac_fix") {
>   visibility = [ "*" ]
>+  poisonous = [ "audio_codecs" ]
>   sources = [
>     "audio_encoder_isac_fix.cc",
>     "audio_encoder_isac_fix.h",
>   ]
>   deps = [
>     "..:audio_codecs_api",
>-    "../..:optional",
>     "../../..:webrtc_common",
>     "../../../modules/audio_coding:isac_fix",
>     "../../../rtc_base:rtc_base_approved",
>+    "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
> 
> rtc_static_library("audio_decoder_isac_fix") {
>   visibility = [ "*" ]
>+  poisonous = [ "audio_codecs" ]
>   sources = [
>     "audio_decoder_isac_fix.cc",
>     "audio_decoder_isac_fix.h",
>   ]
>   deps = [
>     "..:audio_codecs_api",
>-    "../..:optional",
>     "../../..:webrtc_common",
>     "../../../modules/audio_coding:isac_fix",
>     "../../../rtc_base:rtc_base_approved",
>+    "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
> 
> rtc_static_library("audio_encoder_isac_float") {
>   visibility = [ "*" ]
>+  poisonous = [ "audio_codecs" ]
>   sources = [
>     "audio_encoder_isac_float.cc",
>     "audio_encoder_isac_float.h",
>   ]
>   deps = [
>     "..:audio_codecs_api",
>-    "../..:optional",
>     "../../..:webrtc_common",
>     "../../../modules/audio_coding:isac",
>     "../../../rtc_base:rtc_base_approved",
>+    "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
> 
> rtc_static_library("audio_decoder_isac_float") {
>   visibility = [ "*" ]
>+  poisonous = [ "audio_codecs" ]
>   sources = [
>     "audio_decoder_isac_float.cc",
>     "audio_decoder_isac_float.h",
>   ]
>   deps = [
>     "..:audio_codecs_api",
>-    "../..:optional",
>     "../../..:webrtc_common",
>     "../../../modules/audio_coding:isac",
>     "../../../rtc_base:rtc_base_approved",
>+    "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_decoder_isac_fix.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_decoder_isac_fix.cc
>index 011be4f24a7..446640ff70c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_decoder_isac_fix.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_decoder_isac_fix.cc
>@@ -10,18 +10,18 @@
> 
> #include "api/audio_codecs/isac/audio_decoder_isac_fix.h"
> 
>+#include "absl/memory/memory.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/audio_coding/codecs/isac/fix/include/audio_decoder_isacfix.h"
>-#include "rtc_base/ptr_util.h"
> 
> namespace webrtc {
> 
>-rtc::Optional<AudioDecoderIsacFix::Config> AudioDecoderIsacFix::SdpToConfig(
>+absl::optional<AudioDecoderIsacFix::Config> AudioDecoderIsacFix::SdpToConfig(
>     const SdpAudioFormat& format) {
>   return STR_CASE_CMP(format.name.c_str(), "ISAC") == 0 &&
>                  format.clockrate_hz == 16000 && format.num_channels == 1
>-             ? rtc::Optional<Config>(Config())
>-             : rtc::nullopt;
>+             ? absl::optional<Config>(Config())
>+             : absl::nullopt;
> }
> 
> void AudioDecoderIsacFix::AppendSupportedDecoders(
>@@ -30,8 +30,9 @@ void AudioDecoderIsacFix::AppendSupportedDecoders(
> }
> 
> std::unique_ptr<AudioDecoder> AudioDecoderIsacFix::MakeAudioDecoder(
>-    Config config) {
>-  return rtc::MakeUnique<AudioDecoderIsacFixImpl>(16000);
>+    Config config,
>+    absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
>+  return absl::make_unique<AudioDecoderIsacFixImpl>(16000);
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_decoder_isac_fix.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_decoder_isac_fix.h
>index f3d210eb84b..a4ce6858385 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_decoder_isac_fix.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_decoder_isac_fix.h
>@@ -14,9 +14,10 @@
> #include <memory>
> #include <vector>
> 
>+#include "absl/types/optional.h"
>+#include "api/audio_codecs/audio_codec_pair_id.h"
> #include "api/audio_codecs/audio_decoder.h"
> #include "api/audio_codecs/audio_format.h"
>-#include "api/optional.h"
> 
> namespace webrtc {
> 
>@@ -26,9 +27,11 @@ namespace webrtc {
> // NOTE: This struct is still under development and may change without notice.
> struct AudioDecoderIsacFix {
>   struct Config {};  // Empty---no config values needed!
>-  static rtc::Optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
>+  static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
>   static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
>-  static std::unique_ptr<AudioDecoder> MakeAudioDecoder(Config config);
>+  static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
>+      Config config,
>+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt);
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_decoder_isac_float.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_decoder_isac_float.cc
>index 65ad6ac15d1..1c1926f1a1e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_decoder_isac_float.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_decoder_isac_float.cc
>@@ -10,14 +10,14 @@
> 
> #include "api/audio_codecs/isac/audio_decoder_isac_float.h"
> 
>+#include "absl/memory/memory.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/audio_coding/codecs/isac/main/include/audio_decoder_isac.h"
>-#include "rtc_base/ptr_util.h"
> 
> namespace webrtc {
> 
>-rtc::Optional<AudioDecoderIsacFloat::Config> AudioDecoderIsacFloat::SdpToConfig(
>-    const SdpAudioFormat& format) {
>+absl::optional<AudioDecoderIsacFloat::Config>
>+AudioDecoderIsacFloat::SdpToConfig(const SdpAudioFormat& format) {
>   if (STR_CASE_CMP(format.name.c_str(), "ISAC") == 0 &&
>       (format.clockrate_hz == 16000 || format.clockrate_hz == 32000) &&
>       format.num_channels == 1) {
>@@ -25,7 +25,7 @@ rtc::Optional<AudioDecoderIsacFloat::Config> AudioDecoderIsacFloat::SdpToConfig(
>     config.sample_rate_hz = format.clockrate_hz;
>     return config;
>   } else {
>-    return rtc::nullopt;
>+    return absl::nullopt;
>   }
> }
> 
>@@ -36,9 +36,10 @@ void AudioDecoderIsacFloat::AppendSupportedDecoders(
> }
> 
> std::unique_ptr<AudioDecoder> AudioDecoderIsacFloat::MakeAudioDecoder(
>-    Config config) {
>+    Config config,
>+    absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
>   RTC_DCHECK(config.IsOk());
>-  return rtc::MakeUnique<AudioDecoderIsacFloatImpl>(config.sample_rate_hz);
>+  return absl::make_unique<AudioDecoderIsacFloatImpl>(config.sample_rate_hz);
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_decoder_isac_float.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_decoder_isac_float.h
>index 1decd5af3dc..cc139634196 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_decoder_isac_float.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_decoder_isac_float.h
>@@ -14,9 +14,10 @@
> #include <memory>
> #include <vector>
> 
>+#include "absl/types/optional.h"
>+#include "api/audio_codecs/audio_codec_pair_id.h"
> #include "api/audio_codecs/audio_decoder.h"
> #include "api/audio_codecs/audio_format.h"
>-#include "api/optional.h"
> 
> namespace webrtc {
> 
>@@ -31,9 +32,11 @@ struct AudioDecoderIsacFloat {
>     }
>     int sample_rate_hz = 16000;
>   };
>-  static rtc::Optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
>+  static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
>   static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
>-  static std::unique_ptr<AudioDecoder> MakeAudioDecoder(Config config);
>+  static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
>+      Config config,
>+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt);
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_encoder_isac_fix.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_encoder_isac_fix.cc
>index 17c1a761b81..cd8975398b4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_encoder_isac_fix.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_encoder_isac_fix.cc
>@@ -10,14 +10,14 @@
> 
> #include "api/audio_codecs/isac/audio_encoder_isac_fix.h"
> 
>+#include "absl/memory/memory.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/audio_coding/codecs/isac/fix/include/audio_encoder_isacfix.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/string_to_number.h"
> 
> namespace webrtc {
> 
>-rtc::Optional<AudioEncoderIsacFix::Config> AudioEncoderIsacFix::SdpToConfig(
>+absl::optional<AudioEncoderIsacFix::Config> AudioEncoderIsacFix::SdpToConfig(
>     const SdpAudioFormat& format) {
>   if (STR_CASE_CMP(format.name.c_str(), "ISAC") == 0 &&
>       format.clockrate_hz == 16000 && format.num_channels == 1) {
>@@ -31,7 +31,7 @@ rtc::Optional<AudioEncoderIsacFix::Config> AudioEncoderIsacFix::SdpToConfig(
>     }
>     return config;
>   } else {
>-    return rtc::nullopt;
>+    return absl::nullopt;
>   }
> }
> 
>@@ -50,12 +50,13 @@ AudioCodecInfo AudioEncoderIsacFix::QueryAudioEncoder(
> 
> std::unique_ptr<AudioEncoder> AudioEncoderIsacFix::MakeAudioEncoder(
>     AudioEncoderIsacFix::Config config,
>-    int payload_type) {
>+    int payload_type,
>+    absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
>   RTC_DCHECK(config.IsOk());
>   AudioEncoderIsacFixImpl::Config c;
>   c.frame_size_ms = config.frame_size_ms;
>   c.payload_type = payload_type;
>-  return rtc::MakeUnique<AudioEncoderIsacFixImpl>(c);
>+  return absl::make_unique<AudioEncoderIsacFixImpl>(c);
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_encoder_isac_fix.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_encoder_isac_fix.h
>index 5970c025349..731e48d0c27 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_encoder_isac_fix.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_encoder_isac_fix.h
>@@ -14,9 +14,10 @@
> #include <memory>
> #include <vector>
> 
>+#include "absl/types/optional.h"
>+#include "api/audio_codecs/audio_codec_pair_id.h"
> #include "api/audio_codecs/audio_encoder.h"
> #include "api/audio_codecs/audio_format.h"
>-#include "api/optional.h"
> 
> namespace webrtc {
> 
>@@ -29,11 +30,13 @@ struct AudioEncoderIsacFix {
>     bool IsOk() const { return frame_size_ms == 30 || frame_size_ms == 60; }
>     int frame_size_ms = 30;
>   };
>-  static rtc::Optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
>+  static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
>   static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
>   static AudioCodecInfo QueryAudioEncoder(Config config);
>-  static std::unique_ptr<AudioEncoder> MakeAudioEncoder(Config config,
>-                                                        int payload_type);
>+  static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
>+      Config config,
>+      int payload_type,
>+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt);
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_encoder_isac_float.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_encoder_isac_float.cc
>index bdfbcfbebc3..b70a82e2852 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_encoder_isac_float.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_encoder_isac_float.cc
>@@ -10,15 +10,15 @@
> 
> #include "api/audio_codecs/isac/audio_encoder_isac_float.h"
> 
>+#include "absl/memory/memory.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/audio_coding/codecs/isac/main/include/audio_encoder_isac.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/string_to_number.h"
> 
> namespace webrtc {
> 
>-rtc::Optional<AudioEncoderIsacFloat::Config> AudioEncoderIsacFloat::SdpToConfig(
>-    const SdpAudioFormat& format) {
>+absl::optional<AudioEncoderIsacFloat::Config>
>+AudioEncoderIsacFloat::SdpToConfig(const SdpAudioFormat& format) {
>   if (STR_CASE_CMP(format.name.c_str(), "ISAC") == 0 &&
>       (format.clockrate_hz == 16000 || format.clockrate_hz == 32000) &&
>       format.num_channels == 1) {
>@@ -37,7 +37,7 @@ rtc::Optional<AudioEncoderIsacFloat::Config> AudioEncoderIsacFloat::SdpToConfig(
>     }
>     return config;
>   } else {
>-    return rtc::nullopt;
>+    return absl::nullopt;
>   }
> }
> 
>@@ -61,13 +61,14 @@ AudioCodecInfo AudioEncoderIsacFloat::QueryAudioEncoder(
> 
> std::unique_ptr<AudioEncoder> AudioEncoderIsacFloat::MakeAudioEncoder(
>     const AudioEncoderIsacFloat::Config& config,
>-    int payload_type) {
>+    int payload_type,
>+    absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
>   RTC_DCHECK(config.IsOk());
>   AudioEncoderIsacFloatImpl::Config c;
>   c.sample_rate_hz = config.sample_rate_hz;
>   c.frame_size_ms = config.frame_size_ms;
>   c.payload_type = payload_type;
>-  return rtc::MakeUnique<AudioEncoderIsacFloatImpl>(c);
>+  return absl::make_unique<AudioEncoderIsacFloatImpl>(c);
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_encoder_isac_float.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_encoder_isac_float.h
>index f14c2a25f50..6d98bf99685 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_encoder_isac_float.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/isac/audio_encoder_isac_float.h
>@@ -14,9 +14,10 @@
> #include <memory>
> #include <vector>
> 
>+#include "absl/types/optional.h"
>+#include "api/audio_codecs/audio_codec_pair_id.h"
> #include "api/audio_codecs/audio_encoder.h"
> #include "api/audio_codecs/audio_format.h"
>-#include "api/optional.h"
> 
> namespace webrtc {
> 
>@@ -34,11 +35,13 @@ struct AudioEncoderIsacFloat {
>     int sample_rate_hz = 16000;
>     int frame_size_ms = 30;
>   };
>-  static rtc::Optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
>+  static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
>   static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
>   static AudioCodecInfo QueryAudioEncoder(const Config& config);
>-  static std::unique_ptr<AudioEncoder> MakeAudioEncoder(const Config& config,
>-                                                        int payload_type);
>+  static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
>+      const Config& config,
>+      int payload_type,
>+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt);
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/opus/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/opus/BUILD.gn
>index 13a4b5d3ee1..800abbe279b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/opus/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/opus/BUILD.gn
>@@ -19,8 +19,8 @@ rtc_static_library("audio_encoder_opus_config") {
>     "audio_encoder_opus_config.h",
>   ]
>   deps = [
>-    "../..:optional",
>     "../../../rtc_base:rtc_base_approved",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
>   defines = []
>   if (rtc_opus_variable_complexity) {
>@@ -32,6 +32,7 @@ rtc_static_library("audio_encoder_opus_config") {
> 
> rtc_source_set("audio_encoder_opus") {
>   visibility = [ "*" ]
>+  poisonous = [ "audio_codecs" ]
>   public = [
>     "audio_encoder_opus.h",
>   ]
>@@ -41,27 +42,25 @@ rtc_source_set("audio_encoder_opus") {
>   deps = [
>     ":audio_encoder_opus_config",
>     "..:audio_codecs_api",
>-    "../..:optional",
>     "../../../modules/audio_coding:webrtc_opus",
>     "../../../rtc_base:rtc_base_approved",
>-  ]
>-  public_deps = [
>-    # TODO(kwiberg): Remove this public_dep when bug 7847 has been fixed.
>-    "../../../rtc_base:protobuf_utils",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
> 
> rtc_static_library("audio_decoder_opus") {
>   visibility = [ "*" ]
>+  poisonous = [ "audio_codecs" ]
>   sources = [
>     "audio_decoder_opus.cc",
>     "audio_decoder_opus.h",
>   ]
>   deps = [
>     "..:audio_codecs_api",
>-    "../..:optional",
>     "../../..:webrtc_common",
>     "../../../modules/audio_coding:webrtc_opus",
>     "../../../rtc_base:rtc_base_approved",
>+    "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/opus/audio_decoder_opus.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/opus/audio_decoder_opus.cc
>index 472a0794358..41397f04d20 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/opus/audio_decoder_opus.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/opus/audio_decoder_opus.cc
>@@ -14,15 +14,15 @@
> #include <utility>
> #include <vector>
> 
>+#include "absl/memory/memory.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/audio_coding/codecs/opus/audio_decoder_opus.h"
>-#include "rtc_base/ptr_util.h"
> 
> namespace webrtc {
> 
>-rtc::Optional<AudioDecoderOpus::Config> AudioDecoderOpus::SdpToConfig(
>+absl::optional<AudioDecoderOpus::Config> AudioDecoderOpus::SdpToConfig(
>     const SdpAudioFormat& format) {
>-  const auto num_channels = [&]() -> rtc::Optional<int> {
>+  const auto num_channels = [&]() -> absl::optional<int> {
>     auto stereo = format.parameters.find("stereo");
>     if (stereo != format.parameters.end()) {
>       if (stereo->second == "0") {
>@@ -30,7 +30,7 @@ rtc::Optional<AudioDecoderOpus::Config> AudioDecoderOpus::SdpToConfig(
>       } else if (stereo->second == "1") {
>         return 2;
>       } else {
>-        return rtc::nullopt;  // Bad stereo parameter.
>+        return absl::nullopt;  // Bad stereo parameter.
>       }
>     }
>     return 1;  // Default to mono.
>@@ -40,7 +40,7 @@ rtc::Optional<AudioDecoderOpus::Config> AudioDecoderOpus::SdpToConfig(
>       num_channels) {
>     return Config{*num_channels};
>   } else {
>-    return rtc::nullopt;
>+    return absl::nullopt;
>   }
> }
> 
>@@ -55,8 +55,9 @@ void AudioDecoderOpus::AppendSupportedDecoders(
> }
> 
> std::unique_ptr<AudioDecoder> AudioDecoderOpus::MakeAudioDecoder(
>-    Config config) {
>-  return rtc::MakeUnique<AudioDecoderOpusImpl>(config.num_channels);
>+    Config config,
>+    absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
>+  return absl::make_unique<AudioDecoderOpusImpl>(config.num_channels);
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/opus/audio_decoder_opus.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/opus/audio_decoder_opus.h
>index 0cd917f04b9..de26026b783 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/opus/audio_decoder_opus.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/opus/audio_decoder_opus.h
>@@ -14,9 +14,10 @@
> #include <memory>
> #include <vector>
> 
>+#include "absl/types/optional.h"
>+#include "api/audio_codecs/audio_codec_pair_id.h"
> #include "api/audio_codecs/audio_decoder.h"
> #include "api/audio_codecs/audio_format.h"
>-#include "api/optional.h"
> 
> namespace webrtc {
> 
>@@ -28,9 +29,11 @@ struct AudioDecoderOpus {
>   struct Config {
>     int num_channels;
>   };
>-  static rtc::Optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
>+  static absl::optional<Config> SdpToConfig(const SdpAudioFormat& audio_format);
>   static void AppendSupportedDecoders(std::vector<AudioCodecSpec>* specs);
>-  static std::unique_ptr<AudioDecoder> MakeAudioDecoder(Config config);
>+  static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
>+      Config config,
>+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt);
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/opus/audio_encoder_opus.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/opus/audio_encoder_opus.cc
>index 603da6f223b..36d82b3eff9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/opus/audio_encoder_opus.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/opus/audio_encoder_opus.cc
>@@ -14,7 +14,7 @@
> 
> namespace webrtc {
> 
>-rtc::Optional<AudioEncoderOpusConfig> AudioEncoderOpus::SdpToConfig(
>+absl::optional<AudioEncoderOpusConfig> AudioEncoderOpus::SdpToConfig(
>     const SdpAudioFormat& format) {
>   return AudioEncoderOpusImpl::SdpToConfig(format);
> }
>@@ -31,7 +31,8 @@ AudioCodecInfo AudioEncoderOpus::QueryAudioEncoder(
> 
> std::unique_ptr<AudioEncoder> AudioEncoderOpus::MakeAudioEncoder(
>     const AudioEncoderOpusConfig& config,
>-    int payload_type) {
>+    int payload_type,
>+    absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
>   return AudioEncoderOpusImpl::MakeAudioEncoder(config, payload_type);
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/opus/audio_encoder_opus.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/opus/audio_encoder_opus.h
>index d348a178977..20aaaf726f9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/opus/audio_encoder_opus.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/opus/audio_encoder_opus.h
>@@ -14,10 +14,11 @@
> #include <memory>
> #include <vector>
> 
>+#include "absl/types/optional.h"
>+#include "api/audio_codecs/audio_codec_pair_id.h"
> #include "api/audio_codecs/audio_encoder.h"
> #include "api/audio_codecs/audio_format.h"
> #include "api/audio_codecs/opus/audio_encoder_opus_config.h"
>-#include "api/optional.h"
> 
> namespace webrtc {
> 
>@@ -27,13 +28,14 @@ namespace webrtc {
> // NOTE: This struct is still under development and may change without notice.
> struct AudioEncoderOpus {
>   using Config = AudioEncoderOpusConfig;
>-  static rtc::Optional<AudioEncoderOpusConfig> SdpToConfig(
>+  static absl::optional<AudioEncoderOpusConfig> SdpToConfig(
>       const SdpAudioFormat& audio_format);
>   static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
>   static AudioCodecInfo QueryAudioEncoder(const AudioEncoderOpusConfig& config);
>   static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
>-      const AudioEncoderOpusConfig&,
>-      int payload_type);
>+      const AudioEncoderOpusConfig& config,
>+      int payload_type,
>+      absl::optional<AudioCodecPairId> codec_pair_id = absl::nullopt);
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/opus/audio_encoder_opus_config.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/opus/audio_encoder_opus_config.h
>index d586592ab09..c7067bb1c32 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/opus/audio_encoder_opus_config.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/opus/audio_encoder_opus_config.h
>@@ -15,7 +15,7 @@
> 
> #include <vector>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> 
> namespace webrtc {
> 
>@@ -42,7 +42,7 @@ struct AudioEncoderOpusConfig {
> 
>   // NOTE: This member must always be set.
>   // TODO(kwiberg): Turn it into just an int.
>-  rtc::Optional<int> bitrate_bps;
>+  absl::optional<int> bitrate_bps;
> 
>   bool fec_enabled;
>   bool cbr_enabled;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/test/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/test/BUILD.gn
>index 0f742f57cc2..cc7a4d9cd06 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/test/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/test/BUILD.gn
>@@ -38,7 +38,7 @@ if (rtc_include_tests) {
>       "../isac:audio_encoder_isac_float",
>       "../opus:audio_decoder_opus",
>       "../opus:audio_encoder_opus",
>-      "//testing/gmock",
>+      "//third_party/abseil-cpp/absl/memory",
>     ]
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/test/audio_decoder_factory_template_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/test/audio_decoder_factory_template_unittest.cc
>index 53199034c94..d8fd9e02c12 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/test/audio_decoder_factory_template_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/test/audio_decoder_factory_template_unittest.cc
>@@ -9,6 +9,7 @@
>  */
> 
> #include "api/audio_codecs/audio_decoder_factory_template.h"
>+#include "absl/memory/memory.h"
> #include "api/audio_codecs/L16/audio_decoder_L16.h"
> #include "api/audio_codecs/g711/audio_decoder_g711.h"
> #include "api/audio_codecs/g722/audio_decoder_g722.h"
>@@ -16,7 +17,6 @@
> #include "api/audio_codecs/isac/audio_decoder_isac_fix.h"
> #include "api/audio_codecs/isac/audio_decoder_isac_float.h"
> #include "api/audio_codecs/opus/audio_decoder_opus.h"
>-#include "rtc_base/ptr_util.h"
> #include "test/gmock.h"
> #include "test/gtest.h"
> #include "test/mock_audio_decoder.h"
>@@ -43,12 +43,13 @@ struct AudioDecoderFakeApi {
>     SdpAudioFormat audio_format;
>   };
> 
>-  static rtc::Optional<Config> SdpToConfig(const SdpAudioFormat& audio_format) {
>+  static absl::optional<Config> SdpToConfig(
>+      const SdpAudioFormat& audio_format) {
>     if (Params::AudioFormat() == audio_format) {
>       Config config = {audio_format};
>       return config;
>     } else {
>-      return rtc::nullopt;
>+      return absl::nullopt;
>     }
>   }
> 
>@@ -60,8 +61,10 @@ struct AudioDecoderFakeApi {
>     return Params::CodecInfo();
>   }
> 
>-  static std::unique_ptr<AudioDecoder> MakeAudioDecoder(const Config&) {
>-    auto dec = rtc::MakeUnique<testing::StrictMock<MockAudioDecoder>>();
>+  static std::unique_ptr<AudioDecoder> MakeAudioDecoder(
>+      const Config&,
>+      absl::optional<AudioCodecPairId> /*codec_pair_id*/ = absl::nullopt) {
>+    auto dec = absl::make_unique<testing::StrictMock<MockAudioDecoder>>();
>     EXPECT_CALL(*dec, SampleRateHz())
>         .WillOnce(testing::Return(Params::CodecInfo().sample_rate_hz));
>     EXPECT_CALL(*dec, Die());
>@@ -77,7 +80,8 @@ TEST(AudioDecoderFactoryTemplateTest, NoDecoderTypes) {
>           audio_decoder_factory_template_impl::AudioDecoderFactoryT<>>());
>   EXPECT_THAT(factory->GetSupportedDecoders(), testing::IsEmpty());
>   EXPECT_FALSE(factory->IsSupportedDecoder({"foo", 8000, 1}));
>-  EXPECT_EQ(nullptr, factory->MakeAudioDecoder({"bar", 16000, 1}));
>+  EXPECT_EQ(nullptr,
>+            factory->MakeAudioDecoder({"bar", 16000, 1}, absl::nullopt));
> }
> 
> TEST(AudioDecoderFactoryTemplateTest, OneDecoderType) {
>@@ -87,8 +91,9 @@ TEST(AudioDecoderFactoryTemplateTest, OneDecoderType) {
>                   AudioCodecSpec{{"bogus", 8000, 1}, {8000, 1, 12345}}));
>   EXPECT_FALSE(factory->IsSupportedDecoder({"foo", 8000, 1}));
>   EXPECT_TRUE(factory->IsSupportedDecoder({"bogus", 8000, 1}));
>-  EXPECT_EQ(nullptr, factory->MakeAudioDecoder({"bar", 16000, 1}));
>-  auto dec = factory->MakeAudioDecoder({"bogus", 8000, 1});
>+  EXPECT_EQ(nullptr,
>+            factory->MakeAudioDecoder({"bar", 16000, 1}, absl::nullopt));
>+  auto dec = factory->MakeAudioDecoder({"bogus", 8000, 1}, absl::nullopt);
>   ASSERT_NE(nullptr, dec);
>   EXPECT_EQ(8000, dec->SampleRateHz());
> }
>@@ -105,13 +110,15 @@ TEST(AudioDecoderFactoryTemplateTest, TwoDecoderTypes) {
>   EXPECT_TRUE(factory->IsSupportedDecoder({"bogus", 8000, 1}));
>   EXPECT_TRUE(
>       factory->IsSupportedDecoder({"sham", 16000, 2, {{"param", "value"}}}));
>-  EXPECT_EQ(nullptr, factory->MakeAudioDecoder({"bar", 16000, 1}));
>-  auto dec1 = factory->MakeAudioDecoder({"bogus", 8000, 1});
>+  EXPECT_EQ(nullptr,
>+            factory->MakeAudioDecoder({"bar", 16000, 1}, absl::nullopt));
>+  auto dec1 = factory->MakeAudioDecoder({"bogus", 8000, 1}, absl::nullopt);
>   ASSERT_NE(nullptr, dec1);
>   EXPECT_EQ(8000, dec1->SampleRateHz());
>-  EXPECT_EQ(nullptr, factory->MakeAudioDecoder({"sham", 16000, 2}));
>-  auto dec2 =
>-      factory->MakeAudioDecoder({"sham", 16000, 2, {{"param", "value"}}});
>+  EXPECT_EQ(nullptr,
>+            factory->MakeAudioDecoder({"sham", 16000, 2}, absl::nullopt));
>+  auto dec2 = factory->MakeAudioDecoder(
>+      {"sham", 16000, 2, {{"param", "value"}}}, absl::nullopt);
>   ASSERT_NE(nullptr, dec2);
>   EXPECT_EQ(16000, dec2->SampleRateHz());
> }
>@@ -125,11 +132,12 @@ TEST(AudioDecoderFactoryTemplateTest, G711) {
>   EXPECT_FALSE(factory->IsSupportedDecoder({"G711", 8000, 1}));
>   EXPECT_TRUE(factory->IsSupportedDecoder({"PCMU", 8000, 1}));
>   EXPECT_TRUE(factory->IsSupportedDecoder({"pcma", 8000, 1}));
>-  EXPECT_EQ(nullptr, factory->MakeAudioDecoder({"pcmu", 16000, 1}));
>-  auto dec1 = factory->MakeAudioDecoder({"pcmu", 8000, 1});
>+  EXPECT_EQ(nullptr,
>+            factory->MakeAudioDecoder({"pcmu", 16000, 1}, absl::nullopt));
>+  auto dec1 = factory->MakeAudioDecoder({"pcmu", 8000, 1}, absl::nullopt);
>   ASSERT_NE(nullptr, dec1);
>   EXPECT_EQ(8000, dec1->SampleRateHz());
>-  auto dec2 = factory->MakeAudioDecoder({"PCMA", 8000, 1});
>+  auto dec2 = factory->MakeAudioDecoder({"PCMA", 8000, 1}, absl::nullopt);
>   ASSERT_NE(nullptr, dec2);
>   EXPECT_EQ(8000, dec2->SampleRateHz());
> }
>@@ -141,16 +149,17 @@ TEST(AudioDecoderFactoryTemplateTest, G722) {
>                   AudioCodecSpec{{"G722", 8000, 1}, {16000, 1, 64000}}));
>   EXPECT_FALSE(factory->IsSupportedDecoder({"foo", 8000, 1}));
>   EXPECT_TRUE(factory->IsSupportedDecoder({"G722", 8000, 1}));
>-  EXPECT_EQ(nullptr, factory->MakeAudioDecoder({"bar", 16000, 1}));
>-  auto dec1 = factory->MakeAudioDecoder({"G722", 8000, 1});
>+  EXPECT_EQ(nullptr,
>+            factory->MakeAudioDecoder({"bar", 16000, 1}, absl::nullopt));
>+  auto dec1 = factory->MakeAudioDecoder({"G722", 8000, 1}, absl::nullopt);
>   ASSERT_NE(nullptr, dec1);
>   EXPECT_EQ(16000, dec1->SampleRateHz());
>   EXPECT_EQ(1u, dec1->Channels());
>-  auto dec2 = factory->MakeAudioDecoder({"G722", 8000, 2});
>+  auto dec2 = factory->MakeAudioDecoder({"G722", 8000, 2}, absl::nullopt);
>   ASSERT_NE(nullptr, dec2);
>   EXPECT_EQ(16000, dec2->SampleRateHz());
>   EXPECT_EQ(2u, dec2->Channels());
>-  auto dec3 = factory->MakeAudioDecoder({"G722", 8000, 3});
>+  auto dec3 = factory->MakeAudioDecoder({"G722", 8000, 3}, absl::nullopt);
>   ASSERT_EQ(nullptr, dec3);
> }
> 
>@@ -161,8 +170,9 @@ TEST(AudioDecoderFactoryTemplateTest, Ilbc) {
>                   AudioCodecSpec{{"ILBC", 8000, 1}, {8000, 1, 13300}}));
>   EXPECT_FALSE(factory->IsSupportedDecoder({"foo", 8000, 1}));
>   EXPECT_TRUE(factory->IsSupportedDecoder({"ilbc", 8000, 1}));
>-  EXPECT_EQ(nullptr, factory->MakeAudioDecoder({"bar", 8000, 1}));
>-  auto dec = factory->MakeAudioDecoder({"ilbc", 8000, 1});
>+  EXPECT_EQ(nullptr,
>+            factory->MakeAudioDecoder({"bar", 8000, 1}, absl::nullopt));
>+  auto dec = factory->MakeAudioDecoder({"ilbc", 8000, 1}, absl::nullopt);
>   ASSERT_NE(nullptr, dec);
>   EXPECT_EQ(8000, dec->SampleRateHz());
> }
>@@ -175,8 +185,9 @@ TEST(AudioDecoderFactoryTemplateTest, IsacFix) {
>   EXPECT_FALSE(factory->IsSupportedDecoder({"isac", 16000, 2}));
>   EXPECT_TRUE(factory->IsSupportedDecoder({"isac", 16000, 1}));
>   EXPECT_FALSE(factory->IsSupportedDecoder({"isac", 32000, 1}));
>-  EXPECT_EQ(nullptr, factory->MakeAudioDecoder({"isac", 8000, 1}));
>-  auto dec = factory->MakeAudioDecoder({"isac", 16000, 1});
>+  EXPECT_EQ(nullptr,
>+            factory->MakeAudioDecoder({"isac", 8000, 1}, absl::nullopt));
>+  auto dec = factory->MakeAudioDecoder({"isac", 16000, 1}, absl::nullopt);
>   ASSERT_NE(nullptr, dec);
>   EXPECT_EQ(16000, dec->SampleRateHz());
> }
>@@ -191,11 +202,12 @@ TEST(AudioDecoderFactoryTemplateTest, IsacFloat) {
>   EXPECT_FALSE(factory->IsSupportedDecoder({"isac", 16000, 2}));
>   EXPECT_TRUE(factory->IsSupportedDecoder({"isac", 16000, 1}));
>   EXPECT_TRUE(factory->IsSupportedDecoder({"isac", 32000, 1}));
>-  EXPECT_EQ(nullptr, factory->MakeAudioDecoder({"isac", 8000, 1}));
>-  auto dec1 = factory->MakeAudioDecoder({"isac", 16000, 1});
>+  EXPECT_EQ(nullptr,
>+            factory->MakeAudioDecoder({"isac", 8000, 1}, absl::nullopt));
>+  auto dec1 = factory->MakeAudioDecoder({"isac", 16000, 1}, absl::nullopt);
>   ASSERT_NE(nullptr, dec1);
>   EXPECT_EQ(16000, dec1->SampleRateHz());
>-  auto dec2 = factory->MakeAudioDecoder({"isac", 32000, 1});
>+  auto dec2 = factory->MakeAudioDecoder({"isac", 32000, 1}, absl::nullopt);
>   ASSERT_NE(nullptr, dec2);
>   EXPECT_EQ(32000, dec2->SampleRateHz());
> }
>@@ -214,8 +226,9 @@ TEST(AudioDecoderFactoryTemplateTest, L16) {
>   EXPECT_FALSE(factory->IsSupportedDecoder({"foo", 8000, 1}));
>   EXPECT_TRUE(factory->IsSupportedDecoder({"L16", 48000, 1}));
>   EXPECT_FALSE(factory->IsSupportedDecoder({"L16", 96000, 1}));
>-  EXPECT_EQ(nullptr, factory->MakeAudioDecoder({"L16", 8000, 0}));
>-  auto dec = factory->MakeAudioDecoder({"L16", 48000, 2});
>+  EXPECT_EQ(nullptr,
>+            factory->MakeAudioDecoder({"L16", 8000, 0}, absl::nullopt));
>+  auto dec = factory->MakeAudioDecoder({"L16", 48000, 2}, absl::nullopt);
>   ASSERT_NE(nullptr, dec);
>   EXPECT_EQ(48000, dec->SampleRateHz());
> }
>@@ -231,8 +244,9 @@ TEST(AudioDecoderFactoryTemplateTest, Opus) {
>               testing::ElementsAre(AudioCodecSpec{opus_format, opus_info}));
>   EXPECT_FALSE(factory->IsSupportedDecoder({"opus", 48000, 1}));
>   EXPECT_TRUE(factory->IsSupportedDecoder({"opus", 48000, 2}));
>-  EXPECT_EQ(nullptr, factory->MakeAudioDecoder({"bar", 16000, 1}));
>-  auto dec = factory->MakeAudioDecoder({"opus", 48000, 2});
>+  EXPECT_EQ(nullptr,
>+            factory->MakeAudioDecoder({"bar", 16000, 1}, absl::nullopt));
>+  auto dec = factory->MakeAudioDecoder({"opus", 48000, 2}, absl::nullopt);
>   ASSERT_NE(nullptr, dec);
>   EXPECT_EQ(48000, dec->SampleRateHz());
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/test/audio_encoder_factory_template_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/test/audio_encoder_factory_template_unittest.cc
>index 4f16ff452fd..ca024b9326e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/test/audio_encoder_factory_template_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/test/audio_encoder_factory_template_unittest.cc
>@@ -9,6 +9,7 @@
>  */
> 
> #include "api/audio_codecs/audio_encoder_factory_template.h"
>+#include "absl/memory/memory.h"
> #include "api/audio_codecs/L16/audio_encoder_L16.h"
> #include "api/audio_codecs/g711/audio_encoder_g711.h"
> #include "api/audio_codecs/g722/audio_encoder_g722.h"
>@@ -16,7 +17,6 @@
> #include "api/audio_codecs/isac/audio_encoder_isac_fix.h"
> #include "api/audio_codecs/isac/audio_encoder_isac_float.h"
> #include "api/audio_codecs/opus/audio_encoder_opus.h"
>-#include "rtc_base/ptr_util.h"
> #include "test/gmock.h"
> #include "test/gtest.h"
> #include "test/mock_audio_encoder.h"
>@@ -43,12 +43,13 @@ struct AudioEncoderFakeApi {
>     SdpAudioFormat audio_format;
>   };
> 
>-  static rtc::Optional<Config> SdpToConfig(const SdpAudioFormat& audio_format) {
>+  static absl::optional<Config> SdpToConfig(
>+      const SdpAudioFormat& audio_format) {
>     if (Params::AudioFormat() == audio_format) {
>       Config config = {audio_format};
>       return config;
>     } else {
>-      return rtc::nullopt;
>+      return absl::nullopt;
>     }
>   }
> 
>@@ -60,12 +61,13 @@ struct AudioEncoderFakeApi {
>     return Params::CodecInfo();
>   }
> 
>-  static std::unique_ptr<AudioEncoder> MakeAudioEncoder(const Config&,
>-                                                        int payload_type) {
>-    auto enc = rtc::MakeUnique<testing::StrictMock<MockAudioEncoder>>();
>+  static std::unique_ptr<AudioEncoder> MakeAudioEncoder(
>+      const Config&,
>+      int payload_type,
>+      absl::optional<AudioCodecPairId> /*codec_pair_id*/ = absl::nullopt) {
>+    auto enc = absl::make_unique<testing::StrictMock<MockAudioEncoder>>();
>     EXPECT_CALL(*enc, SampleRateHz())
>         .WillOnce(testing::Return(Params::CodecInfo().sample_rate_hz));
>-    EXPECT_CALL(*enc, Die());
>     return std::move(enc);
>   }
> };
>@@ -77,8 +79,9 @@ TEST(AudioEncoderFactoryTemplateTest, NoEncoderTypes) {
>       new rtc::RefCountedObject<
>           audio_encoder_factory_template_impl::AudioEncoderFactoryT<>>());
>   EXPECT_THAT(factory->GetSupportedEncoders(), testing::IsEmpty());
>-  EXPECT_EQ(rtc::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1}));
>-  EXPECT_EQ(nullptr, factory->MakeAudioEncoder(17, {"bar", 16000, 1}));
>+  EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1}));
>+  EXPECT_EQ(nullptr,
>+            factory->MakeAudioEncoder(17, {"bar", 16000, 1}, absl::nullopt));
> }
> 
> TEST(AudioEncoderFactoryTemplateTest, OneEncoderType) {
>@@ -86,11 +89,12 @@ TEST(AudioEncoderFactoryTemplateTest, OneEncoderType) {
>   EXPECT_THAT(factory->GetSupportedEncoders(),
>               testing::ElementsAre(
>                   AudioCodecSpec{{"bogus", 8000, 1}, {8000, 1, 12345}}));
>-  EXPECT_EQ(rtc::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1}));
>+  EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1}));
>   EXPECT_EQ(AudioCodecInfo(8000, 1, 12345),
>             factory->QueryAudioEncoder({"bogus", 8000, 1}));
>-  EXPECT_EQ(nullptr, factory->MakeAudioEncoder(17, {"bar", 16000, 1}));
>-  auto enc = factory->MakeAudioEncoder(17, {"bogus", 8000, 1});
>+  EXPECT_EQ(nullptr,
>+            factory->MakeAudioEncoder(17, {"bar", 16000, 1}, absl::nullopt));
>+  auto enc = factory->MakeAudioEncoder(17, {"bogus", 8000, 1}, absl::nullopt);
>   ASSERT_NE(nullptr, enc);
>   EXPECT_EQ(8000, enc->SampleRateHz());
> }
>@@ -103,19 +107,21 @@ TEST(AudioEncoderFactoryTemplateTest, TwoEncoderTypes) {
>                   AudioCodecSpec{{"bogus", 8000, 1}, {8000, 1, 12345}},
>                   AudioCodecSpec{{"sham", 16000, 2, {{"param", "value"}}},
>                                  {16000, 2, 23456}}));
>-  EXPECT_EQ(rtc::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1}));
>+  EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1}));
>   EXPECT_EQ(AudioCodecInfo(8000, 1, 12345),
>             factory->QueryAudioEncoder({"bogus", 8000, 1}));
>   EXPECT_EQ(
>       AudioCodecInfo(16000, 2, 23456),
>       factory->QueryAudioEncoder({"sham", 16000, 2, {{"param", "value"}}}));
>-  EXPECT_EQ(nullptr, factory->MakeAudioEncoder(17, {"bar", 16000, 1}));
>-  auto enc1 = factory->MakeAudioEncoder(17, {"bogus", 8000, 1});
>+  EXPECT_EQ(nullptr,
>+            factory->MakeAudioEncoder(17, {"bar", 16000, 1}, absl::nullopt));
>+  auto enc1 = factory->MakeAudioEncoder(17, {"bogus", 8000, 1}, absl::nullopt);
>   ASSERT_NE(nullptr, enc1);
>   EXPECT_EQ(8000, enc1->SampleRateHz());
>-  EXPECT_EQ(nullptr, factory->MakeAudioEncoder(17, {"sham", 16000, 2}));
>-  auto enc2 =
>-      factory->MakeAudioEncoder(17, {"sham", 16000, 2, {{"param", "value"}}});
>+  EXPECT_EQ(nullptr,
>+            factory->MakeAudioEncoder(17, {"sham", 16000, 2}, absl::nullopt));
>+  auto enc2 = factory->MakeAudioEncoder(
>+      17, {"sham", 16000, 2, {{"param", "value"}}}, absl::nullopt);
>   ASSERT_NE(nullptr, enc2);
>   EXPECT_EQ(16000, enc2->SampleRateHz());
> }
>@@ -126,14 +132,15 @@ TEST(AudioEncoderFactoryTemplateTest, G711) {
>               testing::ElementsAre(
>                   AudioCodecSpec{{"PCMU", 8000, 1}, {8000, 1, 64000}},
>                   AudioCodecSpec{{"PCMA", 8000, 1}, {8000, 1, 64000}}));
>-  EXPECT_EQ(rtc::nullopt, factory->QueryAudioEncoder({"PCMA", 16000, 1}));
>+  EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"PCMA", 16000, 1}));
>   EXPECT_EQ(AudioCodecInfo(8000, 1, 64000),
>             factory->QueryAudioEncoder({"PCMA", 8000, 1}));
>-  EXPECT_EQ(nullptr, factory->MakeAudioEncoder(17, {"PCMU", 16000, 1}));
>-  auto enc1 = factory->MakeAudioEncoder(17, {"PCMU", 8000, 1});
>+  EXPECT_EQ(nullptr,
>+            factory->MakeAudioEncoder(17, {"PCMU", 16000, 1}, absl::nullopt));
>+  auto enc1 = factory->MakeAudioEncoder(17, {"PCMU", 8000, 1}, absl::nullopt);
>   ASSERT_NE(nullptr, enc1);
>   EXPECT_EQ(8000, enc1->SampleRateHz());
>-  auto enc2 = factory->MakeAudioEncoder(17, {"PCMA", 8000, 1});
>+  auto enc2 = factory->MakeAudioEncoder(17, {"PCMA", 8000, 1}, absl::nullopt);
>   ASSERT_NE(nullptr, enc2);
>   EXPECT_EQ(8000, enc2->SampleRateHz());
> }
>@@ -143,11 +150,12 @@ TEST(AudioEncoderFactoryTemplateTest, G722) {
>   EXPECT_THAT(factory->GetSupportedEncoders(),
>               testing::ElementsAre(
>                   AudioCodecSpec{{"G722", 8000, 1}, {16000, 1, 64000}}));
>-  EXPECT_EQ(rtc::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1}));
>+  EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1}));
>   EXPECT_EQ(AudioCodecInfo(16000, 1, 64000),
>             factory->QueryAudioEncoder({"G722", 8000, 1}));
>-  EXPECT_EQ(nullptr, factory->MakeAudioEncoder(17, {"bar", 16000, 1}));
>-  auto enc = factory->MakeAudioEncoder(17, {"G722", 8000, 1});
>+  EXPECT_EQ(nullptr,
>+            factory->MakeAudioEncoder(17, {"bar", 16000, 1}, absl::nullopt));
>+  auto enc = factory->MakeAudioEncoder(17, {"G722", 8000, 1}, absl::nullopt);
>   ASSERT_NE(nullptr, enc);
>   EXPECT_EQ(16000, enc->SampleRateHz());
> }
>@@ -157,11 +165,12 @@ TEST(AudioEncoderFactoryTemplateTest, Ilbc) {
>   EXPECT_THAT(factory->GetSupportedEncoders(),
>               testing::ElementsAre(
>                   AudioCodecSpec{{"ILBC", 8000, 1}, {8000, 1, 13333}}));
>-  EXPECT_EQ(rtc::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1}));
>+  EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1}));
>   EXPECT_EQ(AudioCodecInfo(8000, 1, 13333),
>             factory->QueryAudioEncoder({"ilbc", 8000, 1}));
>-  EXPECT_EQ(nullptr, factory->MakeAudioEncoder(17, {"bar", 8000, 1}));
>-  auto enc = factory->MakeAudioEncoder(17, {"ilbc", 8000, 1});
>+  EXPECT_EQ(nullptr,
>+            factory->MakeAudioEncoder(17, {"bar", 8000, 1}, absl::nullopt));
>+  auto enc = factory->MakeAudioEncoder(17, {"ilbc", 8000, 1}, absl::nullopt);
>   ASSERT_NE(nullptr, enc);
>   EXPECT_EQ(8000, enc->SampleRateHz());
> }
>@@ -171,17 +180,18 @@ TEST(AudioEncoderFactoryTemplateTest, IsacFix) {
>   EXPECT_THAT(factory->GetSupportedEncoders(),
>               testing::ElementsAre(AudioCodecSpec{
>                   {"ISAC", 16000, 1}, {16000, 1, 32000, 10000, 32000}}));
>-  EXPECT_EQ(rtc::nullopt, factory->QueryAudioEncoder({"isac", 16000, 2}));
>+  EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"isac", 16000, 2}));
>   EXPECT_EQ(AudioCodecInfo(16000, 1, 32000, 10000, 32000),
>             factory->QueryAudioEncoder({"isac", 16000, 1}));
>-  EXPECT_EQ(rtc::nullopt, factory->QueryAudioEncoder({"isac", 32000, 1}));
>-  EXPECT_EQ(nullptr, factory->MakeAudioEncoder(17, {"isac", 8000, 1}));
>-  auto enc1 = factory->MakeAudioEncoder(17, {"isac", 16000, 1});
>+  EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"isac", 32000, 1}));
>+  EXPECT_EQ(nullptr,
>+            factory->MakeAudioEncoder(17, {"isac", 8000, 1}, absl::nullopt));
>+  auto enc1 = factory->MakeAudioEncoder(17, {"isac", 16000, 1}, absl::nullopt);
>   ASSERT_NE(nullptr, enc1);
>   EXPECT_EQ(16000, enc1->SampleRateHz());
>   EXPECT_EQ(3u, enc1->Num10MsFramesInNextPacket());
>-  auto enc2 =
>-      factory->MakeAudioEncoder(17, {"isac", 16000, 1, {{"ptime", "60"}}});
>+  auto enc2 = factory->MakeAudioEncoder(
>+      17, {"isac", 16000, 1, {{"ptime", "60"}}}, absl::nullopt);
>   ASSERT_NE(nullptr, enc2);
>   EXPECT_EQ(6u, enc2->Num10MsFramesInNextPacket());
> }
>@@ -193,16 +203,17 @@ TEST(AudioEncoderFactoryTemplateTest, IsacFloat) {
>       testing::ElementsAre(
>           AudioCodecSpec{{"ISAC", 16000, 1}, {16000, 1, 32000, 10000, 32000}},
>           AudioCodecSpec{{"ISAC", 32000, 1}, {32000, 1, 56000, 10000, 56000}}));
>-  EXPECT_EQ(rtc::nullopt, factory->QueryAudioEncoder({"isac", 16000, 2}));
>+  EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"isac", 16000, 2}));
>   EXPECT_EQ(AudioCodecInfo(16000, 1, 32000, 10000, 32000),
>             factory->QueryAudioEncoder({"isac", 16000, 1}));
>   EXPECT_EQ(AudioCodecInfo(32000, 1, 56000, 10000, 56000),
>             factory->QueryAudioEncoder({"isac", 32000, 1}));
>-  EXPECT_EQ(nullptr, factory->MakeAudioEncoder(17, {"isac", 8000, 1}));
>-  auto enc1 = factory->MakeAudioEncoder(17, {"isac", 16000, 1});
>+  EXPECT_EQ(nullptr,
>+            factory->MakeAudioEncoder(17, {"isac", 8000, 1}, absl::nullopt));
>+  auto enc1 = factory->MakeAudioEncoder(17, {"isac", 16000, 1}, absl::nullopt);
>   ASSERT_NE(nullptr, enc1);
>   EXPECT_EQ(16000, enc1->SampleRateHz());
>-  auto enc2 = factory->MakeAudioEncoder(17, {"isac", 32000, 1});
>+  auto enc2 = factory->MakeAudioEncoder(17, {"isac", 32000, 1}, absl::nullopt);
>   ASSERT_NE(nullptr, enc2);
>   EXPECT_EQ(32000, enc2->SampleRateHz());
> }
>@@ -218,11 +229,12 @@ TEST(AudioEncoderFactoryTemplateTest, L16) {
>           AudioCodecSpec{{"L16", 8000, 2}, {8000, 2, 8000 * 16 * 2}},
>           AudioCodecSpec{{"L16", 16000, 2}, {16000, 2, 16000 * 16 * 2}},
>           AudioCodecSpec{{"L16", 32000, 2}, {32000, 2, 32000 * 16 * 2}}));
>-  EXPECT_EQ(rtc::nullopt, factory->QueryAudioEncoder({"L16", 8000, 0}));
>+  EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"L16", 8000, 0}));
>   EXPECT_EQ(AudioCodecInfo(48000, 1, 48000 * 16),
>             factory->QueryAudioEncoder({"L16", 48000, 1}));
>-  EXPECT_EQ(nullptr, factory->MakeAudioEncoder(17, {"L16", 8000, 0}));
>-  auto enc = factory->MakeAudioEncoder(17, {"L16", 48000, 2});
>+  EXPECT_EQ(nullptr,
>+            factory->MakeAudioEncoder(17, {"L16", 8000, 0}, absl::nullopt));
>+  auto enc = factory->MakeAudioEncoder(17, {"L16", 48000, 2}, absl::nullopt);
>   ASSERT_NE(nullptr, enc);
>   EXPECT_EQ(48000, enc->SampleRateHz());
> }
>@@ -237,13 +249,14 @@ TEST(AudioEncoderFactoryTemplateTest, Opus) {
>       testing::ElementsAre(AudioCodecSpec{
>           {"opus", 48000, 2, {{"minptime", "10"}, {"useinbandfec", "1"}}},
>           info}));
>-  EXPECT_EQ(rtc::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1}));
>+  EXPECT_EQ(absl::nullopt, factory->QueryAudioEncoder({"foo", 8000, 1}));
>   EXPECT_EQ(
>       info,
>       factory->QueryAudioEncoder(
>           {"opus", 48000, 2, {{"minptime", "10"}, {"useinbandfec", "1"}}}));
>-  EXPECT_EQ(nullptr, factory->MakeAudioEncoder(17, {"bar", 16000, 1}));
>-  auto enc = factory->MakeAudioEncoder(17, {"opus", 48000, 2});
>+  EXPECT_EQ(nullptr,
>+            factory->MakeAudioEncoder(17, {"bar", 16000, 1}, absl::nullopt));
>+  auto enc = factory->MakeAudioEncoder(17, {"opus", 48000, 2}, absl::nullopt);
>   ASSERT_NE(nullptr, enc);
>   EXPECT_EQ(48000, enc->SampleRateHz());
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_options.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_options.cc
>new file mode 100644
>index 00000000000..c196d7d9df0
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_options.cc
>@@ -0,0 +1,18 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/audio_options.h"
>+
>+namespace cricket {
>+
>+AudioOptions::AudioOptions() = default;
>+AudioOptions::~AudioOptions() = default;
>+
>+}  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_options.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_options.h
>new file mode 100644
>index 00000000000..df66d360c56
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_options.h
>@@ -0,0 +1,196 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_AUDIO_OPTIONS_H_
>+#define API_AUDIO_OPTIONS_H_
>+
>+#include <string>
>+
>+#include "absl/types/optional.h"
>+#include "rtc_base/stringencode.h"
>+
>+namespace cricket {
>+
>+// Options that can be applied to a VoiceMediaChannel or a VoiceMediaEngine.
>+// Used to be flags, but that makes it hard to selectively apply options.
>+// We are moving all of the setting of options to structs like this,
>+// but some things currently still use flags.
>+struct AudioOptions {
>+  AudioOptions();
>+  ~AudioOptions();
>+  void SetAll(const AudioOptions& change) {
>+    SetFrom(&echo_cancellation, change.echo_cancellation);
>+#if defined(WEBRTC_IOS)
>+    SetFrom(&ios_force_software_aec_HACK, change.ios_force_software_aec_HACK);
>+#endif
>+    SetFrom(&auto_gain_control, change.auto_gain_control);
>+    SetFrom(&noise_suppression, change.noise_suppression);
>+    SetFrom(&highpass_filter, change.highpass_filter);
>+    SetFrom(&stereo_swapping, change.stereo_swapping);
>+    SetFrom(&audio_jitter_buffer_max_packets,
>+            change.audio_jitter_buffer_max_packets);
>+    SetFrom(&audio_jitter_buffer_fast_accelerate,
>+            change.audio_jitter_buffer_fast_accelerate);
>+    SetFrom(&typing_detection, change.typing_detection);
>+    SetFrom(&aecm_generate_comfort_noise, change.aecm_generate_comfort_noise);
>+    SetFrom(&experimental_agc, change.experimental_agc);
>+    SetFrom(&extended_filter_aec, change.extended_filter_aec);
>+    SetFrom(&delay_agnostic_aec, change.delay_agnostic_aec);
>+    SetFrom(&experimental_ns, change.experimental_ns);
>+    SetFrom(&intelligibility_enhancer, change.intelligibility_enhancer);
>+    SetFrom(&residual_echo_detector, change.residual_echo_detector);
>+    SetFrom(&tx_agc_target_dbov, change.tx_agc_target_dbov);
>+    SetFrom(&tx_agc_digital_compression_gain,
>+            change.tx_agc_digital_compression_gain);
>+    SetFrom(&tx_agc_limiter, change.tx_agc_limiter);
>+    SetFrom(&combined_audio_video_bwe, change.combined_audio_video_bwe);
>+    SetFrom(&audio_network_adaptor, change.audio_network_adaptor);
>+    SetFrom(&audio_network_adaptor_config, change.audio_network_adaptor_config);
>+  }
>+
>+  bool operator==(const AudioOptions& o) const {
>+    return echo_cancellation == o.echo_cancellation &&
>+#if defined(WEBRTC_IOS)
>+           ios_force_software_aec_HACK == o.ios_force_software_aec_HACK &&
>+#endif
>+           auto_gain_control == o.auto_gain_control &&
>+           noise_suppression == o.noise_suppression &&
>+           highpass_filter == o.highpass_filter &&
>+           stereo_swapping == o.stereo_swapping &&
>+           audio_jitter_buffer_max_packets ==
>+               o.audio_jitter_buffer_max_packets &&
>+           audio_jitter_buffer_fast_accelerate ==
>+               o.audio_jitter_buffer_fast_accelerate &&
>+           typing_detection == o.typing_detection &&
>+           aecm_generate_comfort_noise == o.aecm_generate_comfort_noise &&
>+           experimental_agc == o.experimental_agc &&
>+           extended_filter_aec == o.extended_filter_aec &&
>+           delay_agnostic_aec == o.delay_agnostic_aec &&
>+           experimental_ns == o.experimental_ns &&
>+           intelligibility_enhancer == o.intelligibility_enhancer &&
>+           residual_echo_detector == o.residual_echo_detector &&
>+           tx_agc_target_dbov == o.tx_agc_target_dbov &&
>+           tx_agc_digital_compression_gain ==
>+               o.tx_agc_digital_compression_gain &&
>+           tx_agc_limiter == o.tx_agc_limiter &&
>+           combined_audio_video_bwe == o.combined_audio_video_bwe &&
>+           audio_network_adaptor == o.audio_network_adaptor &&
>+           audio_network_adaptor_config == o.audio_network_adaptor_config;
>+  }
>+  bool operator!=(const AudioOptions& o) const { return !(*this == o); }
>+
>+  std::string ToString() const {
>+    std::ostringstream ost;
>+    ost << "AudioOptions {";
>+    ost << ToStringIfSet("aec", echo_cancellation);
>+#if defined(WEBRTC_IOS)
>+    ost << ToStringIfSet("ios_force_software_aec_HACK",
>+                         ios_force_software_aec_HACK);
>+#endif
>+    ost << ToStringIfSet("agc", auto_gain_control);
>+    ost << ToStringIfSet("ns", noise_suppression);
>+    ost << ToStringIfSet("hf", highpass_filter);
>+    ost << ToStringIfSet("swap", stereo_swapping);
>+    ost << ToStringIfSet("audio_jitter_buffer_max_packets",
>+                         audio_jitter_buffer_max_packets);
>+    ost << ToStringIfSet("audio_jitter_buffer_fast_accelerate",
>+                         audio_jitter_buffer_fast_accelerate);
>+    ost << ToStringIfSet("typing", typing_detection);
>+    ost << ToStringIfSet("comfort_noise", aecm_generate_comfort_noise);
>+    ost << ToStringIfSet("experimental_agc", experimental_agc);
>+    ost << ToStringIfSet("extended_filter_aec", extended_filter_aec);
>+    ost << ToStringIfSet("delay_agnostic_aec", delay_agnostic_aec);
>+    ost << ToStringIfSet("experimental_ns", experimental_ns);
>+    ost << ToStringIfSet("intelligibility_enhancer", intelligibility_enhancer);
>+    ost << ToStringIfSet("residual_echo_detector", residual_echo_detector);
>+    ost << ToStringIfSet("tx_agc_target_dbov", tx_agc_target_dbov);
>+    ost << ToStringIfSet("tx_agc_digital_compression_gain",
>+                         tx_agc_digital_compression_gain);
>+    ost << ToStringIfSet("tx_agc_limiter", tx_agc_limiter);
>+    ost << ToStringIfSet("combined_audio_video_bwe", combined_audio_video_bwe);
>+    ost << ToStringIfSet("audio_network_adaptor", audio_network_adaptor);
>+    // The adaptor config is a serialized proto buffer and therefore not human
>+    // readable. So we comment out the following line.
>+    // ost << ToStringIfSet("audio_network_adaptor_config",
>+    //     audio_network_adaptor_config);
>+    ost << "}";
>+    return ost.str();
>+  }
>+
>+  // Audio processing that attempts to filter away the output signal from
>+  // later inbound pickup.
>+  absl::optional<bool> echo_cancellation;
>+#if defined(WEBRTC_IOS)
>+  // Forces software echo cancellation on iOS. This is a temporary workaround
>+  // (until Apple fixes the bug) for a device with non-functioning AEC. May
>+  // improve performance on that particular device, but will cause unpredictable
>+  // behavior in all other cases. See http://bugs.webrtc.org/8682.
>+  absl::optional<bool> ios_force_software_aec_HACK;
>+#endif
>+  // Audio processing to adjust the sensitivity of the local mic dynamically.
>+  absl::optional<bool> auto_gain_control;
>+  // Audio processing to filter out background noise.
>+  absl::optional<bool> noise_suppression;
>+  // Audio processing to remove background noise of lower frequencies.
>+  absl::optional<bool> highpass_filter;
>+  // Audio processing to swap the left and right channels.
>+  absl::optional<bool> stereo_swapping;
>+  // Audio receiver jitter buffer (NetEq) max capacity in number of packets.
>+  absl::optional<int> audio_jitter_buffer_max_packets;
>+  // Audio receiver jitter buffer (NetEq) fast accelerate mode.
>+  absl::optional<bool> audio_jitter_buffer_fast_accelerate;
>+  // Audio processing to detect typing.
>+  absl::optional<bool> typing_detection;
>+  absl::optional<bool> aecm_generate_comfort_noise;
>+  absl::optional<bool> experimental_agc;
>+  absl::optional<bool> extended_filter_aec;
>+  absl::optional<bool> delay_agnostic_aec;
>+  absl::optional<bool> experimental_ns;
>+  absl::optional<bool> intelligibility_enhancer;
>+  // Note that tx_agc_* only applies to non-experimental AGC.
>+  absl::optional<bool> residual_echo_detector;
>+  absl::optional<uint16_t> tx_agc_target_dbov;
>+  absl::optional<uint16_t> tx_agc_digital_compression_gain;
>+  absl::optional<bool> tx_agc_limiter;
>+  // Enable combined audio+bandwidth BWE.
>+  // TODO(pthatcher): This flag is set from the
>+  // "googCombinedAudioVideoBwe", but not used anywhere. So delete it,
>+  // and check if any other AudioOptions members are unused.
>+  absl::optional<bool> combined_audio_video_bwe;
>+  // Enable audio network adaptor.
>+  absl::optional<bool> audio_network_adaptor;
>+  // Config string for audio network adaptor.
>+  absl::optional<std::string> audio_network_adaptor_config;
>+
>+ private:
>+  template <class T>
>+  static std::string ToStringIfSet(const char* key,
>+                                   const absl::optional<T>& val) {
>+    std::string str;
>+    if (val) {
>+      str = key;
>+      str += ": ";
>+      str += val ? rtc::ToString(*val) : "";
>+      str += ", ";
>+    }
>+    return str;
>+  }
>+
>+  template <typename T>
>+  static void SetFrom(absl::optional<T>* s, const absl::optional<T>& o) {
>+    if (o) {
>+      *s = o;
>+    }
>+  }
>+};
>+
>+}  // namespace cricket
>+
>+#endif  // API_AUDIO_OPTIONS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/bitrate_constraints.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/bitrate_constraints.h
>new file mode 100644
>index 00000000000..98e89c08585
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/bitrate_constraints.h
>@@ -0,0 +1,41 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_BITRATE_CONSTRAINTS_H_
>+#define API_BITRATE_CONSTRAINTS_H_
>+
>+#include <algorithm>
>+
>+namespace webrtc {
>+// TODO(srte): BitrateConstraints and BitrateSettings should be merged.
>+// Both represent the same kind data, but are using different default
>+// initializer and representation of unset values.
>+struct BitrateConstraints {
>+  int min_bitrate_bps = 0;
>+  int start_bitrate_bps = kDefaultStartBitrateBps;
>+  int max_bitrate_bps = -1;
>+
>+ private:
>+  static constexpr int kDefaultStartBitrateBps = 300000;
>+};
>+
>+// Like std::min, but considers non-positive values to be unset.
>+template <typename T>
>+static T MinPositive(T a, T b) {
>+  if (a <= 0) {
>+    return b;
>+  }
>+  if (b <= 0) {
>+    return a;
>+  }
>+  return std::min(a, b);
>+}
>+}  // namespace webrtc
>+#endif  // API_BITRATE_CONSTRAINTS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/callfactoryinterface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/call/callfactoryinterface.h
>similarity index 74%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/call/callfactoryinterface.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/api/call/callfactoryinterface.h
>index a3cf6ebc46c..a7f32453c63 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/callfactoryinterface.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/call/callfactoryinterface.h
>@@ -8,15 +8,17 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#ifndef CALL_CALLFACTORYINTERFACE_H_
>-#define CALL_CALLFACTORYINTERFACE_H_
>+#ifndef API_CALL_CALLFACTORYINTERFACE_H_
>+#define API_CALL_CALLFACTORYINTERFACE_H_
> 
> #include <memory>
> 
>-#include "call/call.h"
>-
> namespace webrtc {
> 
>+// These classes are not part of the API, and are treated as opaque pointers.
>+class Call;
>+struct CallConfig;
>+
> // This interface exists to allow webrtc to be optionally built without media
> // support (i.e., if only being used for data channels). PeerConnectionFactory
> // is constructed with a CallFactoryInterface, which may or may not be null.
>@@ -24,11 +26,11 @@ class CallFactoryInterface {
>  public:
>   virtual ~CallFactoryInterface() {}
> 
>-  virtual Call* CreateCall(const Call::Config& config) = 0;
>+  virtual Call* CreateCall(const CallConfig& config) = 0;
> };
> 
> std::unique_ptr<CallFactoryInterface> CreateCallFactory();
> 
> }  // namespace webrtc
> 
>-#endif  // CALL_CALLFACTORYINTERFACE_H_
>+#endif  // API_CALL_CALLFACTORYINTERFACE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/call/transport.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/call/transport.cc
>new file mode 100644
>index 00000000000..0a9dd5bcc7e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/call/transport.cc
>@@ -0,0 +1,21 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/call/transport.h"
>+
>+namespace webrtc {
>+
>+PacketOptions::PacketOptions() = default;
>+
>+PacketOptions::PacketOptions(const PacketOptions&) = default;
>+
>+PacketOptions::~PacketOptions() = default;
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/call/transport.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/call/transport.h
>index 1cdb0d366d8..18d22705cd6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/call/transport.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/call/transport.h
>@@ -13,15 +13,25 @@
> 
> #include <stddef.h>
> #include <stdint.h>
>+#include <vector>
> 
> namespace webrtc {
> 
> // TODO(holmer): Look into unifying this with the PacketOptions in
> // asyncpacketsocket.h.
> struct PacketOptions {
>+  PacketOptions();
>+  PacketOptions(const PacketOptions&);
>+  ~PacketOptions();
>+
>   // A 16 bits positive id. Negative ids are invalid and should be interpreted
>   // as packet_id not being set.
>   int packet_id = -1;
>+  // Additional data bound to the RTP packet for use in application code,
>+  // outside of WebRTC.
>+  std::vector<uint8_t> application_data;
>+  // Whether this is a retransmission of an earlier packet.
>+  bool is_retransmit = false;
> };
> 
> class Transport {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/candidate.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/candidate.cc
>index 62cd1bd1352..d51fb843712 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/candidate.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/candidate.cc
>@@ -73,8 +73,9 @@ std::string Candidate::ToStringInternal(bool sensitive) const {
>       sensitive ? address_.ToSensitiveString() : address_.ToString();
>   ost << "Cand[" << transport_name_ << ":" << foundation_ << ":" << component_
>       << ":" << protocol_ << ":" << priority_ << ":" << address << ":" << type_
>-      << ":" << related_address_ << ":" << username_ << ":" << password_ << ":"
>-      << network_id_ << ":" << network_cost_ << ":" << generation_ << "]";
>+      << ":" << related_address_.ToString() << ":" << username_ << ":"
>+      << password_ << ":" << network_id_ << ":" << network_cost_ << ":"
>+      << generation_ << "]";
>   return ost.str();
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/candidate.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/candidate.h
>index a1f45c215e1..6e0547b5e2f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/candidate.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/candidate.h
>@@ -46,14 +46,14 @@ class Candidate {
>   Candidate(const Candidate&);
>   ~Candidate();
> 
>-  const std::string & id() const { return id_; }
>-  void set_id(const std::string & id) { id_ = id; }
>+  const std::string& id() const { return id_; }
>+  void set_id(const std::string& id) { id_ = id; }
> 
>   int component() const { return component_; }
>   void set_component(int component) { component_ = component; }
> 
>-  const std::string & protocol() const { return protocol_; }
>-  void set_protocol(const std::string & protocol) { protocol_ = protocol; }
>+  const std::string& protocol() const { return protocol_; }
>+  void set_protocol(const std::string& protocol) { protocol_ = protocol; }
> 
>   // The protocol used to talk to relay.
>   const std::string& relay_protocol() const { return relay_protocol_; }
>@@ -61,10 +61,8 @@ class Candidate {
>     relay_protocol_ = protocol;
>   }
> 
>-  const rtc::SocketAddress & address() const { return address_; }
>-  void set_address(const rtc::SocketAddress & address) {
>-    address_ = address;
>-  }
>+  const rtc::SocketAddress& address() const { return address_; }
>+  void set_address(const rtc::SocketAddress& address) { address_ = address; }
> 
>   uint32_t priority() const { return priority_; }
>   void set_priority(const uint32_t priority) { priority_ = priority; }
>@@ -91,17 +89,17 @@ class Candidate {
>   }
> 
>   // TODO(honghaiz): Change to usernameFragment or ufrag.
>-  const std::string & username() const { return username_; }
>-  void set_username(const std::string & username) { username_ = username; }
>+  const std::string& username() const { return username_; }
>+  void set_username(const std::string& username) { username_ = username; }
> 
>-  const std::string & password() const { return password_; }
>-  void set_password(const std::string & password) { password_ = password; }
>+  const std::string& password() const { return password_; }
>+  void set_password(const std::string& password) { password_ = password; }
> 
>-  const std::string & type() const { return type_; }
>-  void set_type(const std::string & type) { type_ = type; }
>+  const std::string& type() const { return type_; }
>+  void set_type(const std::string& type) { type_ = type; }
> 
>-  const std::string & network_name() const { return network_name_; }
>-  void set_network_name(const std::string & network_name) {
>+  const std::string& network_name() const { return network_name_; }
>+  void set_network_name(const std::string& network_name) {
>     network_name_ = network_name;
>   }
> 
>@@ -127,24 +125,17 @@ class Candidate {
>   uint16_t network_id() const { return network_id_; }
>   void set_network_id(uint16_t network_id) { network_id_ = network_id; }
> 
>-  const std::string& foundation() const {
>-    return foundation_;
>-  }
>+  const std::string& foundation() const { return foundation_; }
>   void set_foundation(const std::string& foundation) {
>     foundation_ = foundation;
>   }
> 
>-  const rtc::SocketAddress & related_address() const {
>-    return related_address_;
>-  }
>-  void set_related_address(
>-      const rtc::SocketAddress & related_address) {
>+  const rtc::SocketAddress& related_address() const { return related_address_; }
>+  void set_related_address(const rtc::SocketAddress& related_address) {
>     related_address_ = related_address;
>   }
>   const std::string& tcptype() const { return tcptype_; }
>-  void set_tcptype(const std::string& tcptype) {
>-    tcptype_ = tcptype;
>-  }
>+  void set_tcptype(const std::string& tcptype) { tcptype_ = tcptype; }
> 
>   // The name of the transport channel of this candidate.
>   // TODO(phoglund): remove.
>@@ -164,13 +155,9 @@ class Candidate {
>   // given one when looking for a matching candidate to remove.
>   bool MatchesForRemoval(const Candidate& c) const;
> 
>-  std::string ToString() const {
>-    return ToStringInternal(false);
>-  }
>+  std::string ToString() const { return ToStringInternal(false); }
> 
>-  std::string ToSensitiveString() const {
>-    return ToStringInternal(true);
>-  }
>+  std::string ToSensitiveString() const { return ToStringInternal(true); }
> 
>   uint32_t GetPriority(uint32_t type_preference,
>                        int network_adapter_preference,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/datachannelinterface.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/datachannelinterface.cc
>new file mode 100644
>index 00000000000..141462d9075
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/datachannelinterface.cc
>@@ -0,0 +1,35 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/datachannelinterface.h"
>+
>+namespace webrtc {
>+
>+bool DataChannelInterface::ordered() const {
>+  return false;
>+}
>+
>+uint16_t DataChannelInterface::maxRetransmitTime() const {
>+  return 0;
>+}
>+
>+uint16_t DataChannelInterface::maxRetransmits() const {
>+  return 0;
>+}
>+
>+std::string DataChannelInterface::protocol() const {
>+  return std::string();
>+}
>+
>+bool DataChannelInterface::negotiated() const {
>+  return false;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/datachannelinterface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/datachannelinterface.h
>index 4b5ce9a7333..a0d2b3b51a5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/datachannelinterface.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/datachannelinterface.h
>@@ -16,7 +16,6 @@
> 
> #include <string>
> 
>-#include "rtc_base/basictypes.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/copyonwritebuffer.h"
> #include "rtc_base/refcount.h"
>@@ -24,7 +23,7 @@
> namespace webrtc {
> 
> // C++ version of: https://www.w3.org/TR/webrtc/#idl-def-rtcdatachannelinit
>-// TODO(deadbeef): Use rtc::Optional for the "-1 if unset" things.
>+// TODO(deadbeef): Use absl::optional for the "-1 if unset" things.
> struct DataChannelInit {
>   // Deprecated. Reliability is assumed, and channel will be unreliable if
>   // maxRetransmitTime or MaxRetransmits is set.
>@@ -62,14 +61,10 @@ struct DataChannelInit {
> // as binary or text.
> struct DataBuffer {
>   DataBuffer(const rtc::CopyOnWriteBuffer& data, bool binary)
>-      : data(data),
>-        binary(binary) {
>-  }
>+      : data(data), binary(binary) {}
>   // For convenience for unit tests.
>   explicit DataBuffer(const std::string& text)
>-      : data(text.data(), text.length()),
>-        binary(false) {
>-  }
>+      : data(text.data(), text.length()), binary(false) {}
>   size_t size() const { return data.size(); }
> 
>   rtc::CopyOnWriteBuffer data;
>@@ -90,10 +85,10 @@ class DataChannelObserver {
>   //  A data buffer was successfully received.
>   virtual void OnMessage(const DataBuffer& buffer) = 0;
>   // The data channel's buffered_amount has changed.
>-  virtual void OnBufferedAmountChange(uint64_t) {}
>+  virtual void OnBufferedAmountChange(uint64_t previous_amount) {}
> 
>  protected:
>-  virtual ~DataChannelObserver() {}
>+  virtual ~DataChannelObserver() = default;
> };
> 
> class DataChannelInterface : public rtc::RefCountInterface {
>@@ -139,11 +134,11 @@ class DataChannelInterface : public rtc::RefCountInterface {
>   // TODO(deadbeef): Remove these dummy implementations when all classes have
>   // implemented these APIs. They should all just return the values the
>   // DataChannel was created with.
>-  virtual bool ordered() const { return false; }
>-  virtual uint16_t maxRetransmitTime() const { return 0; }
>-  virtual uint16_t maxRetransmits() const { return 0; }
>-  virtual std::string protocol() const { return std::string(); }
>-  virtual bool negotiated() const { return false; }
>+  virtual bool ordered() const;
>+  virtual uint16_t maxRetransmitTime() const;
>+  virtual uint16_t maxRetransmits() const;
>+  virtual std::string protocol() const;
>+  virtual bool negotiated() const;
> 
>   // Returns the ID from the DataChannelInit, if it was negotiated out-of-band.
>   // If negotiated in-band, this ID will be populated once the DTLS role is
>@@ -175,7 +170,7 @@ class DataChannelInterface : public rtc::RefCountInterface {
>   virtual bool Send(const DataBuffer& buffer) = 0;
> 
>  protected:
>-  virtual ~DataChannelInterface() {}
>+  ~DataChannelInterface() override = default;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/dtmfsenderinterface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/dtmfsenderinterface.h
>index 8f0ab71206c..b79bb318047 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/dtmfsenderinterface.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/dtmfsenderinterface.h
>@@ -29,7 +29,7 @@ class DtmfSenderObserverInterface {
>   virtual void OnToneChange(const std::string& tone) = 0;
> 
>  protected:
>-  virtual ~DtmfSenderObserverInterface() {}
>+  virtual ~DtmfSenderObserverInterface() = default;
> };
> 
> // The interface of native implementation of the RTCDTMFSender defined by the
>@@ -67,14 +67,10 @@ class DtmfSenderInterface : public rtc::RefCountInterface {
>   // If InsertDtmf is called on the same object while an existing task for this
>   // object to generate DTMF is still running, the previous task is canceled.
>   // Returns true on success and false on failure.
>-  virtual bool InsertDtmf(const std::string& tones, int duration,
>+  virtual bool InsertDtmf(const std::string& tones,
>+                          int duration,
>                           int inter_tone_gap) = 0;
> 
>-  // Returns the track given as argument to the constructor. Only exists for
>-  // backwards compatibilty; now that DtmfSenders are tied to RtpSenders, it's
>-  // no longer relevant.
>-  virtual const AudioTrackInterface* track() const = 0;
>-
>   // Returns the tones remaining to be played out.
>   virtual std::string tones() const = 0;
> 
>@@ -89,7 +85,7 @@ class DtmfSenderInterface : public rtc::RefCountInterface {
>   virtual int inter_tone_gap() const = 0;
> 
>  protected:
>-  virtual ~DtmfSenderInterface() {}
>+  ~DtmfSenderInterface() override = default;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/fakemetricsobserver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/fakemetricsobserver.cc
>deleted file mode 100644
>index d09bf171851..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/fakemetricsobserver.cc
>+++ /dev/null
>@@ -1,64 +0,0 @@
>-/*
>- *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "api/fakemetricsobserver.h"
>-#include "rtc_base/checks.h"
>-
>-namespace webrtc {
>-
>-FakeMetricsObserver::FakeMetricsObserver() {
>-  Reset();
>-}
>-
>-void FakeMetricsObserver::Reset() {
>-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>-  counters_.clear();
>-  memset(histogram_samples_, 0, sizeof(histogram_samples_));
>-}
>-
>-void FakeMetricsObserver::IncrementEnumCounter(
>-    PeerConnectionEnumCounterType type,
>-    int counter,
>-    int counter_max) {
>-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>-  if (counters_.size() <= static_cast<size_t>(type)) {
>-    counters_.resize(type + 1);
>-  }
>-  auto& counters = counters_[type];
>-  ++counters[counter];
>-}
>-
>-void FakeMetricsObserver::AddHistogramSample(PeerConnectionMetricsName type,
>-    int value) {
>-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>-  RTC_DCHECK_EQ(histogram_samples_[type], 0);
>-  histogram_samples_[type] = value;
>-}
>-
>-int FakeMetricsObserver::GetEnumCounter(PeerConnectionEnumCounterType type,
>-                                        int counter) const {
>-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>-  if (counters_.size() <= static_cast<size_t>(type)) {
>-    return 0;
>-  }
>-  const auto& it = counters_[type].find(counter);
>-  if (it == counters_[type].end()) {
>-    return 0;
>-  }
>-  return it->second;
>-}
>-
>-int FakeMetricsObserver::GetHistogramSample(
>-    PeerConnectionMetricsName type) const {
>-  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>-  return histogram_samples_[type];
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/fakemetricsobserver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/fakemetricsobserver.h
>deleted file mode 100644
>index afd019391a6..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/fakemetricsobserver.h
>+++ /dev/null
>@@ -1,52 +0,0 @@
>-/*
>- *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef API_FAKEMETRICSOBSERVER_H_
>-#define API_FAKEMETRICSOBSERVER_H_
>-
>-#include <map>
>-#include <string>
>-#include <vector>
>-
>-#include "api/peerconnectioninterface.h"
>-#include "rtc_base/thread_checker.h"
>-
>-namespace webrtc {
>-
>-class FakeMetricsObserver : public MetricsObserverInterface {
>- public:
>-  FakeMetricsObserver();
>-  void Reset();
>-
>-  void IncrementEnumCounter(PeerConnectionEnumCounterType,
>-                            int counter,
>-                            int counter_max) override;
>-  void AddHistogramSample(PeerConnectionMetricsName type,
>-                          int value) override;
>-
>-  // Accessors to be used by the tests.
>-  int GetEnumCounter(PeerConnectionEnumCounterType type, int counter) const;
>-  int GetHistogramSample(PeerConnectionMetricsName type) const;
>-
>- protected:
>-  ~FakeMetricsObserver() {}
>-
>- private:
>-  rtc::ThreadChecker thread_checker_;
>-  // The vector contains maps for each counter type. In the map, it's a mapping
>-  // from individual counter to its count, such that it's memory efficient when
>-  // comes to sparse enum types, like the SSL ciphers in the IANA registry.
>-  std::vector<std::map<int, int>> counters_;
>-  int histogram_samples_[kPeerConnectionMetricsName_Max];
>-};
>-
>-}  // namespace webrtc
>-
>-#endif  // API_FAKEMETRICSOBSERVER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/fec_controller.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/fec_controller.h
>new file mode 100644
>index 00000000000..59e86ccedbd
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/fec_controller.h
>@@ -0,0 +1,91 @@
>+/*
>+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_FEC_CONTROLLER_H_
>+#define API_FEC_CONTROLLER_H_
>+
>+#include <memory>
>+#include <vector>
>+
>+#include "common_types.h"  // NOLINT(build/include)
>+#include "modules/include/module_fec_types.h"
>+
>+namespace webrtc {
>+// TODO(yinwa): work in progress. API in class FecController should not be
>+// used by other users until this comment is removed.
>+
>+// Callback class used for telling the user about how to configure the FEC,
>+// and the rates sent the last second is returned to the VCM.
>+class VCMProtectionCallback {
>+ public:
>+  virtual int ProtectionRequest(const FecProtectionParams* delta_params,
>+                                const FecProtectionParams* key_params,
>+                                uint32_t* sent_video_rate_bps,
>+                                uint32_t* sent_nack_rate_bps,
>+                                uint32_t* sent_fec_rate_bps) = 0;
>+
>+ protected:
>+  virtual ~VCMProtectionCallback() {}
>+};
>+
>+// FecController calculates how much of the allocated network
>+// capacity that can be used by an encoder and how much that
>+// is needed for redundant packets such as FEC and NACK. It uses an
>+// implementation of |VCMProtectionCallback| to set new FEC parameters and get
>+// the bitrate currently used for FEC and NACK.
>+// Usage:
>+// Setup by calling SetProtectionMethod and SetEncodingData.
>+// For each encoded image, call UpdateWithEncodedData.
>+// Each time the bandwidth estimate change, call UpdateFecRates. UpdateFecRates
>+// will return the bitrate that can be used by an encoder.
>+// A lock is used to protect internal states, so methods can be called on an
>+// arbitrary thread.
>+class FecController {
>+ public:
>+  virtual ~FecController() {}
>+
>+  virtual void SetProtectionCallback(
>+      VCMProtectionCallback* protection_callback) = 0;
>+  virtual void SetProtectionMethod(bool enable_fec, bool enable_nack) = 0;
>+
>+  // Informs loss protectoin logic of initial encoding state.
>+  virtual void SetEncodingData(size_t width,
>+                               size_t height,
>+                               size_t num_temporal_layers,
>+                               size_t max_payload_size) = 0;
>+
>+  // Returns target rate for the encoder given the channel parameters.
>+  // Inputs:  estimated_bitrate_bps - the estimated network bitrate in bits/s.
>+  //          actual_framerate - encoder frame rate.
>+  //          fraction_lost - packet loss rate in % in the network.
>+  //          loss_mask_vector - packet loss mask since last time this method
>+  //          was called. round_trip_time_ms - round trip time in milliseconds.
>+  virtual uint32_t UpdateFecRates(uint32_t estimated_bitrate_bps,
>+                                  int actual_framerate,
>+                                  uint8_t fraction_lost,
>+                                  std::vector<bool> loss_mask_vector,
>+                                  int64_t round_trip_time_ms) = 0;
>+
>+  // Informs of encoded output.
>+  virtual void UpdateWithEncodedData(size_t encoded_image_length,
>+                                     FrameType encoded_image_frametype) = 0;
>+
>+  // Returns whether this FEC Controller needs Loss Vector Mask as input.
>+  virtual bool UseLossVectorMask() = 0;
>+};
>+
>+class FecControllerFactoryInterface {
>+ public:
>+  virtual std::unique_ptr<FecController> CreateFecController() = 0;
>+  virtual ~FecControllerFactoryInterface() = default;
>+};
>+
>+}  // namespace webrtc
>+#endif  // API_FEC_CONTROLLER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/jsep.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/jsep.cc
>index 1f4afbaca1c..52a60f95e32 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/jsep.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/jsep.cc
>@@ -21,4 +21,21 @@ size_t SessionDescriptionInterface::RemoveCandidates(
>   return 0;
> }
> 
>+void CreateSessionDescriptionObserver::OnFailure(RTCError error) {
>+  OnFailure(error.message());
>+}
>+
>+void CreateSessionDescriptionObserver::OnFailure(const std::string& error) {
>+  OnFailure(RTCError(RTCErrorType::INTERNAL_ERROR, std::string(error)));
>+}
>+
>+void SetSessionDescriptionObserver::OnFailure(RTCError error) {
>+  std::string message(error.message());
>+  OnFailure(message);
>+}
>+
>+void SetSessionDescriptionObserver::OnFailure(const std::string& error) {
>+  OnFailure(RTCError(RTCErrorType::INTERNAL_ERROR, std::string(error)));
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/jsep.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/jsep.h
>index 581f689d1c6..4d4bcc0bfb6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/jsep.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/jsep.h
>@@ -26,7 +26,8 @@
> #include <string>
> #include <vector>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
>+#include "api/rtcerror.h"
> #include "rtc_base/refcount.h"
> 
> namespace cricket {
>@@ -77,6 +78,12 @@ IceCandidateInterface* CreateIceCandidate(const std::string& sdp_mid,
>                                           const std::string& sdp,
>                                           SdpParseError* error);
> 
>+// Creates an IceCandidateInterface based on a parsed candidate structure.
>+std::unique_ptr<IceCandidateInterface> CreateIceCandidate(
>+    const std::string& sdp_mid,
>+    int sdp_mline_index,
>+    const cricket::Candidate& candidate);
>+
> // This class represents a collection of candidates for a specific m= section.
> // Used in SessionDescriptionInterface.
> class IceCandidateCollection {
>@@ -106,7 +113,7 @@ const char* SdpTypeToString(SdpType type);
> // Returns the SdpType from its string form. The string form can be one of the
> // constants defined in SessionDescriptionInterface. Passing in any other string
> // results in nullopt.
>-rtc::Optional<SdpType> SdpTypeFromString(const std::string& type_str);
>+absl::optional<SdpType> SdpTypeFromString(const std::string& type_str);
> 
> // Class representation of an SDP session description.
> //
>@@ -190,6 +197,14 @@ std::unique_ptr<SessionDescriptionInterface> CreateSessionDescription(
>     const std::string& sdp,
>     SdpParseError* error_out);
> 
>+// Creates a SessionDescriptionInterface based on a parsed SDP structure and the
>+// given type, ID and version.
>+std::unique_ptr<SessionDescriptionInterface> CreateSessionDescription(
>+    SdpType type,
>+    const std::string& session_id,
>+    const std::string& session_version,
>+    std::unique_ptr<cricket::SessionDescription> description);
>+
> // CreateOffer and CreateAnswer callback interface.
> class CreateSessionDescriptionObserver : public rtc::RefCountInterface {
>  public:
>@@ -197,7 +212,15 @@ class CreateSessionDescriptionObserver : public rtc::RefCountInterface {
>   // TODO(deadbeef): Make this take an std::unique_ptr<> to avoid confusion
>   // around ownership.
>   virtual void OnSuccess(SessionDescriptionInterface* desc) = 0;
>-  virtual void OnFailure(const std::string& error) = 0;
>+  // The OnFailure callback takes an RTCError, which consists of an
>+  // error code and a string.
>+  // RTCError is non-copyable, so it must be passed using std::move.
>+  // Earlier versions of the API used a string argument. This version
>+  // is deprecated; in order to let clients remove the old version, it has a
>+  // default implementation. If both versions are unimplemented, the
>+  // result will be a runtime error (stack overflow). This is intentional.
>+  virtual void OnFailure(RTCError error);
>+  virtual void OnFailure(const std::string& error);
> 
>  protected:
>   ~CreateSessionDescriptionObserver() override = default;
>@@ -207,7 +230,10 @@ class CreateSessionDescriptionObserver : public rtc::RefCountInterface {
> class SetSessionDescriptionObserver : public rtc::RefCountInterface {
>  public:
>   virtual void OnSuccess() = 0;
>-  virtual void OnFailure(const std::string& error) = 0;
>+  // See description in CreateSessionDescriptionObserver for OnFailure.
>+  virtual void OnFailure(RTCError error);
>+
>+  virtual void OnFailure(const std::string& error);
> 
>  protected:
>   ~SetSessionDescriptionObserver() override = default;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/jsepicecandidate.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/jsepicecandidate.cc
>new file mode 100644
>index 00000000000..b9ba2fea29a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/jsepicecandidate.cc
>@@ -0,0 +1,83 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/jsepicecandidate.h"
>+
>+namespace webrtc {
>+
>+std::string JsepIceCandidate::sdp_mid() const {
>+  return sdp_mid_;
>+}
>+
>+int JsepIceCandidate::sdp_mline_index() const {
>+  return sdp_mline_index_;
>+}
>+
>+const cricket::Candidate& JsepIceCandidate::candidate() const {
>+  return candidate_;
>+}
>+
>+std::string JsepIceCandidate::server_url() const {
>+  return candidate_.url();
>+}
>+
>+JsepCandidateCollection::JsepCandidateCollection() = default;
>+
>+JsepCandidateCollection::JsepCandidateCollection(JsepCandidateCollection&& o)
>+    : candidates_(std::move(o.candidates_)) {}
>+
>+size_t JsepCandidateCollection::count() const {
>+  return candidates_.size();
>+}
>+
>+void JsepCandidateCollection::add(JsepIceCandidate* candidate) {
>+  candidates_.push_back(candidate);
>+}
>+
>+const IceCandidateInterface* JsepCandidateCollection::at(size_t index) const {
>+  return candidates_[index];
>+}
>+
>+JsepCandidateCollection::~JsepCandidateCollection() {
>+  for (std::vector<JsepIceCandidate*>::iterator it = candidates_.begin();
>+       it != candidates_.end(); ++it) {
>+    delete *it;
>+  }
>+}
>+
>+bool JsepCandidateCollection::HasCandidate(
>+    const IceCandidateInterface* candidate) const {
>+  bool ret = false;
>+  for (std::vector<JsepIceCandidate*>::const_iterator it = candidates_.begin();
>+       it != candidates_.end(); ++it) {
>+    if ((*it)->sdp_mid() == candidate->sdp_mid() &&
>+        (*it)->sdp_mline_index() == candidate->sdp_mline_index() &&
>+        (*it)->candidate().IsEquivalent(candidate->candidate())) {
>+      ret = true;
>+      break;
>+    }
>+  }
>+  return ret;
>+}
>+
>+size_t JsepCandidateCollection::remove(const cricket::Candidate& candidate) {
>+  auto iter = std::find_if(candidates_.begin(), candidates_.end(),
>+                           [candidate](JsepIceCandidate* c) {
>+                             return candidate.MatchesForRemoval(c->candidate());
>+                           });
>+  if (iter != candidates_.end()) {
>+    delete *iter;
>+    candidates_.erase(iter);
>+    return 1;
>+  }
>+  return 0;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/jsepicecandidate.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/jsepicecandidate.h
>index dae6121eade..50520fe7279 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/jsepicecandidate.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/jsepicecandidate.h
>@@ -28,24 +28,23 @@ namespace webrtc {
> class JsepIceCandidate : public IceCandidateInterface {
>  public:
>   JsepIceCandidate(const std::string& sdp_mid, int sdp_mline_index);
>-  JsepIceCandidate(const std::string& sdp_mid, int sdp_mline_index,
>+  JsepIceCandidate(const std::string& sdp_mid,
>+                   int sdp_mline_index,
>                    const cricket::Candidate& candidate);
>-  ~JsepIceCandidate();
>+  ~JsepIceCandidate() override;
>   // |err| may be null.
>   bool Initialize(const std::string& sdp, SdpParseError* err);
>   void SetCandidate(const cricket::Candidate& candidate) {
>     candidate_ = candidate;
>   }
> 
>-  virtual std::string sdp_mid() const { return sdp_mid_; }
>-  virtual int sdp_mline_index() const { return sdp_mline_index_; }
>-  virtual const cricket::Candidate& candidate() const {
>-    return candidate_;
>-  }
>+  std::string sdp_mid() const override;
>+  int sdp_mline_index() const override;
>+  const cricket::Candidate& candidate() const override;
> 
>-  virtual std::string server_url() const { return candidate_.url(); }
>+  std::string server_url() const override;
> 
>-  virtual bool ToString(std::string* out) const;
>+  bool ToString(std::string* out) const override;
> 
>  private:
>   std::string sdp_mid_;
>@@ -58,25 +57,18 @@ class JsepIceCandidate : public IceCandidateInterface {
> // Implementation of IceCandidateCollection which stores JsepIceCandidates.
> class JsepCandidateCollection : public IceCandidateCollection {
>  public:
>-  JsepCandidateCollection() {}
>+  JsepCandidateCollection();
>   // Move constructor is defined so that a vector of JsepCandidateCollections
>   // can be resized.
>-  JsepCandidateCollection(JsepCandidateCollection&& o)
>-      : candidates_(std::move(o.candidates_)) {}
>-  ~JsepCandidateCollection();
>-  virtual size_t count() const {
>-    return candidates_.size();
>-  }
>-  virtual bool HasCandidate(const IceCandidateInterface* candidate) const;
>+  JsepCandidateCollection(JsepCandidateCollection&& o);
>+  ~JsepCandidateCollection() override;
>+  size_t count() const override;
>+  bool HasCandidate(const IceCandidateInterface* candidate) const override;
>   // Adds and takes ownership of the JsepIceCandidate.
>   // TODO(deadbeef): Make this use an std::unique_ptr<>, so ownership logic is
>   // more clear.
>-  virtual void add(JsepIceCandidate* candidate) {
>-    candidates_.push_back(candidate);
>-  }
>-  virtual const IceCandidateInterface* at(size_t index) const {
>-    return candidates_[index];
>-  }
>+  virtual void add(JsepIceCandidate* candidate);
>+  const IceCandidateInterface* at(size_t index) const override;
>   // Removes the candidate that has a matching address and protocol.
>   //
>   // Returns the number of candidates that were removed.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/jsepsessiondescription.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/jsepsessiondescription.h
>index 70bb27710a2..d70829e0fbe 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/jsepsessiondescription.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/jsepsessiondescription.h
>@@ -41,8 +41,8 @@ class JsepSessionDescription : public SessionDescriptionInterface {
>   // TODO(deadbeef): Make this use an std::unique_ptr<>, so ownership logic is
>   // more clear.
>   bool Initialize(cricket::SessionDescription* description,
>-      const std::string& session_id,
>-      const std::string& session_version);
>+                  const std::string& session_id,
>+                  const std::string& session_version);
> 
>   virtual cricket::SessionDescription* description() {
>     return description_.get();
>@@ -50,12 +50,8 @@ class JsepSessionDescription : public SessionDescriptionInterface {
>   virtual const cricket::SessionDescription* description() const {
>     return description_.get();
>   }
>-  virtual std::string session_id() const {
>-    return session_id_;
>-  }
>-  virtual std::string session_version() const {
>-    return session_version_;
>-  }
>+  virtual std::string session_id() const { return session_id_; }
>+  virtual std::string session_version() const { return session_version_; }
>   virtual SdpType GetType() const { return type_; }
>   virtual std::string type() const { return SdpTypeToString(type_); }
>   // Allows changing the type. Used for testing.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediaconstraintsinterface.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediaconstraintsinterface.cc
>index 90a957cfa48..fb4481f35b5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediaconstraintsinterface.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediaconstraintsinterface.cc
>@@ -59,11 +59,11 @@ bool FindConstraint(const webrtc::MediaConstraintsInterface* constraints,
> }
> 
> // Converts a constraint (mandatory takes precedence over optional) to an
>-// rtc::Optional.
>+// absl::optional.
> template <typename T>
> void ConstraintToOptional(const webrtc::MediaConstraintsInterface* constraints,
>                           const std::string& key,
>-                          rtc::Optional<T>* value_out) {
>+                          absl::optional<T>* value_out) {
>   T value;
>   bool present = FindConstraint<T>(constraints, key, &value, nullptr);
>   if (present) {
>@@ -89,8 +89,7 @@ const char MediaConstraintsInterface::kMaxFrameRate[] = "maxFrameRate";
> const char MediaConstraintsInterface::kMinFrameRate[] = "minFrameRate";
> 
> // Audio constraints.
>-const char MediaConstraintsInterface::kEchoCancellation[] =
>-    "echoCancellation";
>+const char MediaConstraintsInterface::kEchoCancellation[] = "echoCancellation";
> const char MediaConstraintsInterface::kGoogEchoCancellation[] =
>     "googEchoCancellation";
> const char MediaConstraintsInterface::kExtendedFilterEchoCancellation[] =
>@@ -107,11 +106,7 @@ const char MediaConstraintsInterface::kExperimentalNoiseSuppression[] =
>     "googNoiseSuppression2";
> const char MediaConstraintsInterface::kIntelligibilityEnhancer[] =
>     "intelligibilityEnhancer";
>-const char MediaConstraintsInterface::kLevelControl[] = "levelControl";
>-const char MediaConstraintsInterface::kLevelControlInitialPeakLevelDBFS[] =
>-    "levelControlInitialPeakLevelDBFS";
>-const char MediaConstraintsInterface::kHighpassFilter[] =
>-    "googHighpassFilter";
>+const char MediaConstraintsInterface::kHighpassFilter[] = "googHighpassFilter";
> const char MediaConstraintsInterface::kTypingNoiseDetection[] =
>     "googTypingNoiseDetection";
> const char MediaConstraintsInterface::kAudioMirroring[] = "googAudioMirroring";
>@@ -128,11 +123,9 @@ const char MediaConstraintsInterface::kOfferToReceiveVideo[] =
>     "OfferToReceiveVideo";
> const char MediaConstraintsInterface::kVoiceActivityDetection[] =
>     "VoiceActivityDetection";
>-const char MediaConstraintsInterface::kIceRestart[] =
>-    "IceRestart";
>+const char MediaConstraintsInterface::kIceRestart[] = "IceRestart";
> // Google specific constraint for BUNDLE enable/disable.
>-const char MediaConstraintsInterface::kUseRtpMux[] =
>-    "googUseRtpMUX";
>+const char MediaConstraintsInterface::kUseRtpMux[] = "googUseRtpMUX";
> 
> // Below constraints should be used during PeerConnection construction.
> const char MediaConstraintsInterface::kEnableDtlsSrtp[] =
>@@ -153,11 +146,11 @@ const char MediaConstraintsInterface::kCpuOveruseDetection[] =
>     "googCpuOveruseDetection";
> const char MediaConstraintsInterface::kPayloadPadding[] = "googPayloadPadding";
> 
>-
> // Set |value| to the value associated with the first appearance of |key|, or
> // return false if |key| is not found.
> bool MediaConstraintsInterface::Constraints::FindFirst(
>-    const std::string& key, std::string* value) const {
>+    const std::string& key,
>+    std::string* value) const {
>   for (Constraints::const_iterator iter = begin(); iter != end(); ++iter) {
>     if (iter->key == key) {
>       *value = iter->value;
>@@ -168,7 +161,8 @@ bool MediaConstraintsInterface::Constraints::FindFirst(
> }
> 
> bool FindConstraint(const MediaConstraintsInterface* constraints,
>-                    const std::string& key, bool* value,
>+                    const std::string& key,
>+                    bool* value,
>                     size_t* mandatory_constraints) {
>   return ::FindConstraint<bool>(constraints, key, value, mandatory_constraints);
> }
>@@ -195,9 +189,9 @@ void CopyConstraintsIntoRtcConfiguration(
>   }
>   FindConstraint(constraints, MediaConstraintsInterface::kEnableDscp,
>                  &configuration->media_config.enable_dscp, nullptr);
>-  FindConstraint(
>-      constraints, MediaConstraintsInterface::kCpuOveruseDetection,
>-      &configuration->media_config.video.enable_cpu_overuse_detection, nullptr);
>+  FindConstraint(constraints, MediaConstraintsInterface::kCpuOveruseDetection,
>+                 &configuration->media_config.video.enable_cpu_adaptation,
>+                 nullptr);
>   FindConstraint(constraints, MediaConstraintsInterface::kEnableRtpDataChannels,
>                  &configuration->enable_rtp_data_channel, nullptr);
>   // Find Suspend Below Min Bitrate constraint.
>@@ -247,9 +241,6 @@ void CopyConstraintsIntoAudioOptions(
>   ConstraintToOptional<bool>(
>       constraints, MediaConstraintsInterface::kIntelligibilityEnhancer,
>       &options->intelligibility_enhancer);
>-  ConstraintToOptional<bool>(constraints,
>-                             MediaConstraintsInterface::kLevelControl,
>-                             &options->level_control);
>   ConstraintToOptional<bool>(constraints,
>                              MediaConstraintsInterface::kHighpassFilter,
>                              &options->highpass_filter);
>@@ -259,9 +250,6 @@ void CopyConstraintsIntoAudioOptions(
>   ConstraintToOptional<bool>(constraints,
>                              MediaConstraintsInterface::kAudioMirroring,
>                              &options->stereo_swapping);
>-  ConstraintToOptional<float>(
>-      constraints, MediaConstraintsInterface::kLevelControlInitialPeakLevelDBFS,
>-      &options->level_control_initial_peak_level_dbfs);
>   ConstraintToOptional<std::string>(
>       constraints, MediaConstraintsInterface::kAudioNetworkAdaptorConfig,
>       &options->audio_network_adaptor_config);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediaconstraintsinterface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediaconstraintsinterface.h
>index 73e4619bca9..54ab7067660 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediaconstraintsinterface.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediaconstraintsinterface.h
>@@ -23,7 +23,7 @@
> #include <string>
> #include <vector>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> #include "api/peerconnectioninterface.h"
> 
> namespace webrtc {
>@@ -39,8 +39,7 @@ class MediaConstraintsInterface {
>   struct Constraint {
>     Constraint() {}
>     Constraint(const std::string& key, const std::string value)
>-        : key(key), value(value) {
>-    }
>+        : key(key), value(value) {}
>     std::string key;
>     std::string value;
>   };
>@@ -54,12 +53,12 @@ class MediaConstraintsInterface {
>   // Specified by draft-alvestrand-constraints-resolution-00b
>   static const char kMinAspectRatio[];  // minAspectRatio
>   static const char kMaxAspectRatio[];  // maxAspectRatio
>-  static const char kMaxWidth[];  // maxWidth
>-  static const char kMinWidth[];  // minWidth
>-  static const char kMaxHeight[];  // maxHeight
>-  static const char kMinHeight[];  // minHeight
>-  static const char kMaxFrameRate[];  // maxFrameRate
>-  static const char kMinFrameRate[];  // minFrameRate
>+  static const char kMaxWidth[];        // maxWidth
>+  static const char kMinWidth[];        // minWidth
>+  static const char kMaxHeight[];       // maxHeight
>+  static const char kMinHeight[];       // minHeight
>+  static const char kMaxFrameRate[];    // maxFrameRate
>+  static const char kMinFrameRate[];    // minFrameRate
> 
>   // Constraint keys used by a local audio source.
>   static const char kEchoCancellation[];  // echoCancellation
>@@ -68,18 +67,15 @@ class MediaConstraintsInterface {
>   static const char kGoogEchoCancellation[];  // googEchoCancellation
> 
>   static const char kExtendedFilterEchoCancellation[];  // googEchoCancellation2
>-  static const char kDAEchoCancellation[];  // googDAEchoCancellation
>-  static const char kAutoGainControl[];  // googAutoGainControl
>-  static const char kExperimentalAutoGainControl[];  // googAutoGainControl2
>-  static const char kNoiseSuppression[];  // googNoiseSuppression
>+  static const char kDAEchoCancellation[];            // googDAEchoCancellation
>+  static const char kAutoGainControl[];               // googAutoGainControl
>+  static const char kExperimentalAutoGainControl[];   // googAutoGainControl2
>+  static const char kNoiseSuppression[];              // googNoiseSuppression
>   static const char kExperimentalNoiseSuppression[];  // googNoiseSuppression2
>-  static const char kIntelligibilityEnhancer[];  // intelligibilityEnhancer
>-  static const char kLevelControl[];             // levelControl
>-  static const char
>-      kLevelControlInitialPeakLevelDBFS[];  // levelControlInitialPeakLevelDBFS
>-  static const char kHighpassFilter[];  // googHighpassFilter
>+  static const char kIntelligibilityEnhancer[];       // intelligibilityEnhancer
>+  static const char kHighpassFilter[];                // googHighpassFilter
>   static const char kTypingNoiseDetection[];  // googTypingNoiseDetection
>-  static const char kAudioMirroring[];  // googAudioMirroring
>+  static const char kAudioMirroring[];        // googAudioMirroring
>   static const char
>       kAudioNetworkAdaptorConfig[];  // goodAudioNetworkAdaptorConfig
> 
>@@ -88,15 +84,15 @@ class MediaConstraintsInterface {
> 
>   // Constraint keys for CreateOffer / CreateAnswer
>   // Specified by the W3C PeerConnection spec
>-  static const char kOfferToReceiveVideo[];  // OfferToReceiveVideo
>-  static const char kOfferToReceiveAudio[];  // OfferToReceiveAudio
>+  static const char kOfferToReceiveVideo[];     // OfferToReceiveVideo
>+  static const char kOfferToReceiveAudio[];     // OfferToReceiveAudio
>   static const char kVoiceActivityDetection[];  // VoiceActivityDetection
>-  static const char kIceRestart[];  // IceRestart
>+  static const char kIceRestart[];              // IceRestart
>   // These keys are google specific.
>   static const char kUseRtpMux[];  // googUseRtpMUX
> 
>   // Constraints values.
>-  static const char kValueTrue[];  // true
>+  static const char kValueTrue[];   // true
>   static const char kValueFalse[];  // false
> 
>   // PeerConnection constraint keys.
>@@ -111,12 +107,12 @@ class MediaConstraintsInterface {
>   static const char kEnableIPv6[];  // googIPv6
>   // Temporary constraint to enable suspend below min bitrate feature.
>   static const char kEnableVideoSuspendBelowMinBitrate[];
>-      // googSuspendBelowMinBitrate
>+  // googSuspendBelowMinBitrate
>   // Constraint to enable combined audio+video bandwidth estimation.
>   static const char kCombinedAudioVideoBwe[];  // googCombinedAudioVideoBwe
>-  static const char kScreencastMinBitrate[];  // googScreencastMinBitrate
>-  static const char kCpuOveruseDetection[];  // googCpuOveruseDetection
>-  static const char kPayloadPadding[];  // googPayloadPadding
>+  static const char kScreencastMinBitrate[];   // googScreencastMinBitrate
>+  static const char kCpuOveruseDetection[];    // googCpuOveruseDetection
>+  static const char kPayloadPadding[];         // googPayloadPadding
> 
>   // The prefix of internal-only constraints whose JS set values should be
>   // stripped by Chrome before passed down to Libjingle.
>@@ -129,7 +125,8 @@ class MediaConstraintsInterface {
> };
> 
> bool FindConstraint(const MediaConstraintsInterface* constraints,
>-                    const std::string& key, bool* value,
>+                    const std::string& key,
>+                    bool* value,
>                     size_t* mandatory_constraints);
> 
> bool FindConstraint(const MediaConstraintsInterface* constraints,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediastreaminterface.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediastreaminterface.cc
>index 0bc5d61e800..6f08a0caf1c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediastreaminterface.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediastreaminterface.cc
>@@ -45,4 +45,17 @@ AudioProcessorInterface::GetStats(bool /*has_remote_tracks*/) {
>   return new_stats;
> }
> 
>+VideoTrackInterface::ContentHint VideoTrackInterface::content_hint() const {
>+  return ContentHint::kNone;
>+}
>+
>+bool AudioTrackInterface::GetSignalLevel(int* level) {
>+  return false;
>+}
>+
>+rtc::scoped_refptr<AudioProcessorInterface>
>+AudioTrackInterface::GetAudioProcessor() {
>+  return nullptr;
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediastreaminterface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediastreaminterface.h
>index 2b7b56a2b12..b661351e9fc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediastreaminterface.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediastreaminterface.h
>@@ -22,13 +22,13 @@
> #include <string>
> #include <vector>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> #include "api/video/video_frame.h"
> // TODO(zhihuang): Remove unrelated headers once downstream applications stop
> // relying on them; they were previously transitively included by
> // mediachannel.h, which is no longer a dependency of this file.
>-#include "api/videosinkinterface.h"
>-#include "api/videosourceinterface.h"
>+#include "api/video/video_sink_interface.h"
>+#include "api/video/video_source_interface.h"
> #include "modules/audio_processing/include/audio_processing_statistics.h"
> #include "rtc_base/ratetracker.h"
> #include "rtc_base/refcount.h"
>@@ -60,19 +60,14 @@ class NotifierInterface {
> class MediaSourceInterface : public rtc::RefCountInterface,
>                              public NotifierInterface {
>  public:
>-  enum SourceState {
>-    kInitializing,
>-    kLive,
>-    kEnded,
>-    kMuted
>-  };
>+  enum SourceState { kInitializing, kLive, kEnded, kMuted };
> 
>   virtual SourceState state() const = 0;
> 
>   virtual bool remote() const = 0;
> 
>  protected:
>-  virtual ~MediaSourceInterface() {}
>+  ~MediaSourceInterface() override = default;
> };
> 
> // C++ version of MediaStreamTrack.
>@@ -106,7 +101,7 @@ class MediaStreamTrackInterface : public rtc::RefCountInterface,
>   virtual TrackState state() const = 0;
> 
>  protected:
>-  virtual ~MediaStreamTrackInterface() {}
>+  ~MediaStreamTrackInterface() override = default;
> };
> 
> // VideoTrackSourceInterface is a reference counted source used for
>@@ -116,9 +111,8 @@ class MediaStreamTrackInterface : public rtc::RefCountInterface,
> // on the worker thread via a VideoTrack. A custom implementation of a source
> // can inherit AdaptedVideoTrackSource instead of directly implementing this
> // interface.
>-class VideoTrackSourceInterface
>-    : public MediaSourceInterface,
>-      public rtc::VideoSourceInterface<VideoFrame> {
>+class VideoTrackSourceInterface : public MediaSourceInterface,
>+                                  public rtc::VideoSourceInterface<VideoFrame> {
>  public:
>   struct Stats {
>     // Original size of captured frame, before video adaptation.
>@@ -138,7 +132,7 @@ class VideoTrackSourceInterface
>   // depending on video codec.
>   // TODO(perkj): Remove this once denoising is done by the source, and not by
>   // the encoder.
>-  virtual rtc::Optional<bool> needs_denoising() const = 0;
>+  virtual absl::optional<bool> needs_denoising() const = 0;
> 
>   // Returns false if no stats are available, e.g, for a remote source, or a
>   // source which has not seen its first frame yet.
>@@ -147,7 +141,7 @@ class VideoTrackSourceInterface
>   virtual bool GetStats(Stats* stats) = 0;
> 
>  protected:
>-  virtual ~VideoTrackSourceInterface() {}
>+  ~VideoTrackSourceInterface() override = default;
> };
> 
> // VideoTrackInterface is designed to be invoked on the signaling thread except
>@@ -156,28 +150,27 @@ class VideoTrackSourceInterface
> // PeerConnectionFactory::CreateVideoTrack can be used for creating a VideoTrack
> // that ensures thread safety and that all methods are called on the right
> // thread.
>-class VideoTrackInterface
>-    : public MediaStreamTrackInterface,
>-      public rtc::VideoSourceInterface<VideoFrame> {
>+class VideoTrackInterface : public MediaStreamTrackInterface,
>+                            public rtc::VideoSourceInterface<VideoFrame> {
>  public:
>   // Video track content hint, used to override the source is_screencast
>   // property.
>-  // See https://crbug.com/653531 and https://github.com/WICG/mst-content-hint.
>-  enum class ContentHint { kNone, kFluid, kDetailed };
>+  // See https://crbug.com/653531 and https://w3c.github.io/mst-content-hint.
>+  enum class ContentHint { kNone, kFluid, kDetailed, kText };
> 
>   // Register a video sink for this track. Used to connect the track to the
>   // underlying video engine.
>-  void AddOrUpdateSink(rtc::VideoSinkInterface<VideoFrame>*,
>-                       const rtc::VideoSinkWants&) override {}
>-  void RemoveSink(rtc::VideoSinkInterface<VideoFrame>*) override {}
>+  void AddOrUpdateSink(rtc::VideoSinkInterface<VideoFrame>* sink,
>+                       const rtc::VideoSinkWants& wants) override {}
>+  void RemoveSink(rtc::VideoSinkInterface<VideoFrame>* sink) override {}
> 
>   virtual VideoTrackSourceInterface* GetSource() const = 0;
> 
>-  virtual ContentHint content_hint() const { return ContentHint::kNone; }
>-  virtual void set_content_hint(ContentHint) {}
>+  virtual ContentHint content_hint() const;
>+  virtual void set_content_hint(ContentHint hint) {}
> 
>  protected:
>-  virtual ~VideoTrackInterface() {}
>+  ~VideoTrackInterface() override = default;
> };
> 
> // Interface for receiving audio data from a AudioTrack.
>@@ -211,15 +204,15 @@ class AudioSourceInterface : public MediaSourceInterface {
>   // Sets the volume of the source. |volume| is in  the range of [0, 10].
>   // TODO(tommi): This method should be on the track and ideally volume should
>   // be applied in the track in a way that does not affect clones of the track.
>-  virtual void SetVolume(double) {}
>+  virtual void SetVolume(double volume) {}
> 
>   // Registers/unregisters observers to the audio source.
>-  virtual void RegisterAudioObserver(AudioObserver*) {}
>-  virtual void UnregisterAudioObserver(AudioObserver*) {}
>+  virtual void RegisterAudioObserver(AudioObserver* observer) {}
>+  virtual void UnregisterAudioObserver(AudioObserver* observer) {}
> 
>   // TODO(tommi): Make pure virtual.
>-  virtual void AddSink(AudioTrackSinkInterface*) {}
>-  virtual void RemoveSink(AudioTrackSinkInterface*) {}
>+  virtual void AddSink(AudioTrackSinkInterface* sink) {}
>+  virtual void RemoveSink(AudioTrackSinkInterface* sink) {}
> };
> 
> // Interface of the audio processor used by the audio track to collect
>@@ -269,14 +262,14 @@ class AudioProcessorInterface : public rtc::RefCountInterface {
>   virtual AudioProcessorStatistics GetStats(bool has_remote_tracks);
> 
>  protected:
>-  virtual ~AudioProcessorInterface() {}
>+  ~AudioProcessorInterface() override = default;
> };
> 
> class AudioTrackInterface : public MediaStreamTrackInterface {
>  public:
>   // TODO(deadbeef): Figure out if the following interface should be const or
>   // not.
>-  virtual AudioSourceInterface* GetSource() const =  0;
>+  virtual AudioSourceInterface* GetSource() const = 0;
> 
>   // Add/Remove a sink that will receive the audio data from the track.
>   virtual void AddSink(AudioTrackSinkInterface* sink) = 0;
>@@ -286,23 +279,19 @@ class AudioTrackInterface : public MediaStreamTrackInterface {
>   // Return true on success, otherwise false.
>   // TODO(deadbeef): Change the interface to int GetSignalLevel() and pure
>   // virtual after it's implemented in chromium.
>-  virtual bool GetSignalLevel(int*) { return false; }
>+  virtual bool GetSignalLevel(int* level);
> 
>   // Get the audio processor used by the audio track. Return null if the track
>   // does not have any processor.
>   // TODO(deadbeef): Make the interface pure virtual.
>-  virtual rtc::scoped_refptr<AudioProcessorInterface> GetAudioProcessor() {
>-    return nullptr;
>-  }
>+  virtual rtc::scoped_refptr<AudioProcessorInterface> GetAudioProcessor();
> 
>  protected:
>-  virtual ~AudioTrackInterface() {}
>+  ~AudioTrackInterface() override = default;
> };
> 
>-typedef std::vector<rtc::scoped_refptr<AudioTrackInterface> >
>-    AudioTrackVector;
>-typedef std::vector<rtc::scoped_refptr<VideoTrackInterface> >
>-    VideoTrackVector;
>+typedef std::vector<rtc::scoped_refptr<AudioTrackInterface> > AudioTrackVector;
>+typedef std::vector<rtc::scoped_refptr<VideoTrackInterface> > VideoTrackVector;
> 
> // C++ version of https://www.w3.org/TR/mediacapture-streams/#mediastream.
> //
>@@ -315,15 +304,14 @@ typedef std::vector<rtc::scoped_refptr<VideoTrackInterface> >
> class MediaStreamInterface : public rtc::RefCountInterface,
>                              public NotifierInterface {
>  public:
>-  // TODO(steveanton): This could be renamed to id() to match the spec.
>-  virtual std::string label() const = 0;
>+  virtual std::string id() const = 0;
> 
>   virtual AudioTrackVector GetAudioTracks() = 0;
>   virtual VideoTrackVector GetVideoTracks() = 0;
>-  virtual rtc::scoped_refptr<AudioTrackInterface>
>-      FindAudioTrack(const std::string& track_id) = 0;
>-  virtual rtc::scoped_refptr<VideoTrackInterface>
>-      FindVideoTrack(const std::string& track_id) = 0;
>+  virtual rtc::scoped_refptr<AudioTrackInterface> FindAudioTrack(
>+      const std::string& track_id) = 0;
>+  virtual rtc::scoped_refptr<VideoTrackInterface> FindVideoTrack(
>+      const std::string& track_id) = 0;
> 
>   virtual bool AddTrack(AudioTrackInterface* track) = 0;
>   virtual bool AddTrack(VideoTrackInterface* track) = 0;
>@@ -331,7 +319,7 @@ class MediaStreamInterface : public rtc::RefCountInterface,
>   virtual bool RemoveTrack(VideoTrackInterface* track) = 0;
> 
>  protected:
>-  virtual ~MediaStreamInterface() {}
>+  ~MediaStreamInterface() override = default;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediastreamproxy.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediastreamproxy.h
>index e32d52b16b7..4c544595aa1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediastreamproxy.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediastreamproxy.h
>@@ -21,22 +21,22 @@ namespace webrtc {
> // TODO(deadbeef): Move this to .cc file and out of api/. What threads methods
> // are called on is an implementation detail.
> BEGIN_SIGNALING_PROXY_MAP(MediaStream)
>-  PROXY_SIGNALING_THREAD_DESTRUCTOR()
>-  PROXY_CONSTMETHOD0(std::string, label)
>-  PROXY_METHOD0(AudioTrackVector, GetAudioTracks)
>-  PROXY_METHOD0(VideoTrackVector, GetVideoTracks)
>-  PROXY_METHOD1(rtc::scoped_refptr<AudioTrackInterface>,
>-                FindAudioTrack,
>-                const std::string&)
>-  PROXY_METHOD1(rtc::scoped_refptr<VideoTrackInterface>,
>-                FindVideoTrack,
>-                const std::string&)
>-  PROXY_METHOD1(bool, AddTrack, AudioTrackInterface*)
>-  PROXY_METHOD1(bool, AddTrack, VideoTrackInterface*)
>-  PROXY_METHOD1(bool, RemoveTrack, AudioTrackInterface*)
>-  PROXY_METHOD1(bool, RemoveTrack, VideoTrackInterface*)
>-  PROXY_METHOD1(void, RegisterObserver, ObserverInterface*)
>-  PROXY_METHOD1(void, UnregisterObserver, ObserverInterface*)
>+PROXY_SIGNALING_THREAD_DESTRUCTOR()
>+PROXY_CONSTMETHOD0(std::string, id)
>+PROXY_METHOD0(AudioTrackVector, GetAudioTracks)
>+PROXY_METHOD0(VideoTrackVector, GetVideoTracks)
>+PROXY_METHOD1(rtc::scoped_refptr<AudioTrackInterface>,
>+              FindAudioTrack,
>+              const std::string&)
>+PROXY_METHOD1(rtc::scoped_refptr<VideoTrackInterface>,
>+              FindVideoTrack,
>+              const std::string&)
>+PROXY_METHOD1(bool, AddTrack, AudioTrackInterface*)
>+PROXY_METHOD1(bool, AddTrack, VideoTrackInterface*)
>+PROXY_METHOD1(bool, RemoveTrack, AudioTrackInterface*)
>+PROXY_METHOD1(bool, RemoveTrack, VideoTrackInterface*)
>+PROXY_METHOD1(void, RegisterObserver, ObserverInterface*)
>+PROXY_METHOD1(void, UnregisterObserver, ObserverInterface*)
> END_PROXY_MAP()
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediastreamtrackproxy.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediastreamtrackproxy.h
>index 57a76955998..77b7bad9b4b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediastreamtrackproxy.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediastreamtrackproxy.h
>@@ -25,39 +25,39 @@ namespace webrtc {
> // are called on is an implementation detail.
> 
> BEGIN_SIGNALING_PROXY_MAP(AudioTrack)
>-  PROXY_SIGNALING_THREAD_DESTRUCTOR()
>-  PROXY_CONSTMETHOD0(std::string, kind)
>-  PROXY_CONSTMETHOD0(std::string, id)
>-  PROXY_CONSTMETHOD0(TrackState, state)
>-  PROXY_CONSTMETHOD0(bool, enabled)
>-  PROXY_CONSTMETHOD0(AudioSourceInterface*, GetSource)
>-  PROXY_METHOD1(void, AddSink, AudioTrackSinkInterface*)
>-  PROXY_METHOD1(void, RemoveSink, AudioTrackSinkInterface*)
>-  PROXY_METHOD1(bool, GetSignalLevel, int*)
>-  PROXY_METHOD0(rtc::scoped_refptr<AudioProcessorInterface>, GetAudioProcessor)
>-  PROXY_METHOD1(bool, set_enabled, bool)
>-  PROXY_METHOD1(void, RegisterObserver, ObserverInterface*)
>-  PROXY_METHOD1(void, UnregisterObserver, ObserverInterface*)
>+PROXY_SIGNALING_THREAD_DESTRUCTOR()
>+PROXY_CONSTMETHOD0(std::string, kind)
>+PROXY_CONSTMETHOD0(std::string, id)
>+PROXY_CONSTMETHOD0(TrackState, state)
>+PROXY_CONSTMETHOD0(bool, enabled)
>+PROXY_CONSTMETHOD0(AudioSourceInterface*, GetSource)
>+PROXY_METHOD1(void, AddSink, AudioTrackSinkInterface*)
>+PROXY_METHOD1(void, RemoveSink, AudioTrackSinkInterface*)
>+PROXY_METHOD1(bool, GetSignalLevel, int*)
>+PROXY_METHOD0(rtc::scoped_refptr<AudioProcessorInterface>, GetAudioProcessor)
>+PROXY_METHOD1(bool, set_enabled, bool)
>+PROXY_METHOD1(void, RegisterObserver, ObserverInterface*)
>+PROXY_METHOD1(void, UnregisterObserver, ObserverInterface*)
> END_PROXY_MAP()
> 
> BEGIN_PROXY_MAP(VideoTrack)
>-  PROXY_SIGNALING_THREAD_DESTRUCTOR()
>-  PROXY_CONSTMETHOD0(std::string, kind)
>-  PROXY_CONSTMETHOD0(std::string, id)
>-  PROXY_CONSTMETHOD0(TrackState, state)
>-  PROXY_CONSTMETHOD0(bool, enabled)
>-  PROXY_METHOD1(bool, set_enabled, bool)
>-  PROXY_CONSTMETHOD0(ContentHint, content_hint)
>-  PROXY_METHOD1(void, set_content_hint, ContentHint)
>-  PROXY_WORKER_METHOD2(void,
>-                       AddOrUpdateSink,
>-                       rtc::VideoSinkInterface<VideoFrame>*,
>-                       const rtc::VideoSinkWants&)
>-  PROXY_WORKER_METHOD1(void, RemoveSink, rtc::VideoSinkInterface<VideoFrame>*)
>-  PROXY_CONSTMETHOD0(VideoTrackSourceInterface*, GetSource)
>+PROXY_SIGNALING_THREAD_DESTRUCTOR()
>+PROXY_CONSTMETHOD0(std::string, kind)
>+PROXY_CONSTMETHOD0(std::string, id)
>+PROXY_CONSTMETHOD0(TrackState, state)
>+PROXY_CONSTMETHOD0(bool, enabled)
>+PROXY_METHOD1(bool, set_enabled, bool)
>+PROXY_CONSTMETHOD0(ContentHint, content_hint)
>+PROXY_METHOD1(void, set_content_hint, ContentHint)
>+PROXY_WORKER_METHOD2(void,
>+                     AddOrUpdateSink,
>+                     rtc::VideoSinkInterface<VideoFrame>*,
>+                     const rtc::VideoSinkWants&)
>+PROXY_WORKER_METHOD1(void, RemoveSink, rtc::VideoSinkInterface<VideoFrame>*)
>+PROXY_CONSTMETHOD0(VideoTrackSourceInterface*, GetSource)
> 
>-  PROXY_METHOD1(void, RegisterObserver, ObserverInterface*)
>-  PROXY_METHOD1(void, UnregisterObserver, ObserverInterface*)
>+PROXY_METHOD1(void, RegisterObserver, ObserverInterface*)
>+PROXY_METHOD1(void, UnregisterObserver, ObserverInterface*)
> END_PROXY_MAP()
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediatypes.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediatypes.h
>index 93ce1a231dd..f2812760255 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediatypes.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediatypes.h
>@@ -13,13 +13,12 @@
> 
> #include <string>
> 
>+// The cricket and webrtc have separate definitions for what a media type is.
>+// They're not compatible. Watch out for this.
>+
> namespace cricket {
> 
>-enum MediaType {
>-  MEDIA_TYPE_AUDIO,
>-  MEDIA_TYPE_VIDEO,
>-  MEDIA_TYPE_DATA
>-};
>+enum MediaType { MEDIA_TYPE_AUDIO, MEDIA_TYPE_VIDEO, MEDIA_TYPE_DATA };
> 
> std::string MediaTypeToString(MediaType type);
> // Aborts on invalid string. Only expected to be used on strings that are
>@@ -28,4 +27,10 @@ MediaType MediaTypeFromString(const std::string& type_str);
> 
> }  // namespace cricket
> 
>+namespace webrtc {
>+
>+enum class MediaType { ANY, AUDIO, VIDEO, DATA };
>+
>+}  // namespace webrtc
>+
> #endif  // API_MEDIATYPES_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/notifier.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/notifier.h
>index ceeda4de557..e5c61c96674 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/notifier.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/notifier.h
>@@ -23,8 +23,7 @@ namespace webrtc {
> template <class T>
> class Notifier : public T {
>  public:
>-  Notifier() {
>-  }
>+  Notifier() {}
> 
>   virtual void RegisterObserver(ObserverInterface* observer) {
>     RTC_DCHECK(observer != nullptr);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/optional.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/optional.cc
>deleted file mode 100644
>index 94126170a5a..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/optional.cc
>+++ /dev/null
>@@ -1,34 +0,0 @@
>-/*
>- *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "api/optional.h"
>-
>-namespace rtc {
>-namespace optional_internal {
>-
>-#if RTC_HAS_ASAN
>-
>-void* FunctionThatDoesNothingImpl(void* x) {
>-  return x;
>-}
>-
>-#endif
>-
>-struct NulloptArg {
>-  constexpr NulloptArg() {}
>-};
>-
>-static NulloptArg nullopt_arg;
>-
>-}  // namespace optional_internal
>-
>-const nullopt_t nullopt(rtc::optional_internal::nullopt_arg);
>-
>-}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/optional.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/optional.h
>deleted file mode 100644
>index 7a623357337..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/optional.h
>+++ /dev/null
>@@ -1,449 +0,0 @@
>-/*
>- *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef API_OPTIONAL_H_
>-#define API_OPTIONAL_H_
>-
>-#include <algorithm>
>-#include <memory>
>-#include <utility>
>-
>-#ifdef UNIT_TEST
>-#include <iomanip>
>-#include <ostream>
>-#endif  // UNIT_TEST
>-
>-#include "api/array_view.h"
>-#include "rtc_base/checks.h"
>-#include "rtc_base/sanitizer.h"
>-
>-namespace rtc {
>-
>-namespace optional_internal {
>-
>-#if RTC_HAS_ASAN
>-
>-// This is a non-inlined function. The optimizer can't see inside it.  It
>-// prevents the compiler from generating optimized code that reads value_ even
>-// if it is unset. Although safe, this causes memory sanitizers to complain.
>-void* FunctionThatDoesNothingImpl(void*);
>-
>-template <typename T>
>-inline T* FunctionThatDoesNothing(T* x) {
>-  return reinterpret_cast<T*>(
>-      FunctionThatDoesNothingImpl(reinterpret_cast<void*>(x)));
>-}
>-
>-#else
>-
>-template <typename T>
>-inline T* FunctionThatDoesNothing(T* x) {
>-  return x;
>-}
>-
>-#endif
>-
>-struct NulloptArg;
>-
>-}  // namespace optional_internal
>-
>-// nullopt_t must be a non-aggregate literal type with a constexpr constructor
>-// that takes some implementation-defined literal type. It mustn't have a
>-// default constructor nor an initializer-list constructor.
>-// See:
>-// http://en.cppreference.com/w/cpp/utility/optional/nullopt_t
>-// That page uses int, though this seems to confuse older versions of GCC.
>-struct nullopt_t {
>-  constexpr explicit nullopt_t(rtc::optional_internal::NulloptArg&) {}
>-};
>-
>-// Specification:
>-// http://en.cppreference.com/w/cpp/utility/optional/nullopt
>-extern const nullopt_t nullopt;
>-
>-// Simple std::optional-wannabe. It either contains a T or not.
>-//
>-// A moved-from Optional<T> may only be destroyed, and assigned to if T allows
>-// being assigned to after having been moved from. Specifically, you may not
>-// assume that it just doesn't contain a value anymore.
>-//
>-// Examples of good places to use Optional:
>-//
>-// - As a class or struct member, when the member doesn't always have a value:
>-//     struct Prisoner {
>-//       std::string name;
>-//       Optional<int> cell_number;  // Empty if not currently incarcerated.
>-//     };
>-//
>-// - As a return value for functions that may fail to return a value on all
>-//   allowed inputs. For example, a function that searches an array might
>-//   return an Optional<size_t> (the index where it found the element, or
>-//   nothing if it didn't find it); and a function that parses numbers might
>-//   return Optional<double> (the parsed number, or nothing if parsing failed).
>-//
>-// Examples of bad places to use Optional:
>-//
>-// - As a return value for functions that may fail because of disallowed
>-//   inputs. For example, a string length function should not return
>-//   Optional<size_t> so that it can return nothing in case the caller passed
>-//   it a null pointer; the function should probably use RTC_[D]CHECK instead,
>-//   and return plain size_t.
>-//
>-// - As a return value for functions that may fail to return a value on all
>-//   allowed inputs, but need to tell the caller what went wrong. Returning
>-//   Optional<double> when parsing a single number as in the example above
>-//   might make sense, but any larger parse job is probably going to need to
>-//   tell the caller what the problem was, not just that there was one.
>-//
>-// - As a non-mutable function argument. When you want to pass a value of a
>-//   type T that can fail to be there, const T* is almost always both fastest
>-//   and cleanest. (If you're *sure* that the the caller will always already
>-//   have an Optional<T>, const Optional<T>& is slightly faster than const T*,
>-//   but this is a micro-optimization. In general, stick to const T*.)
>-//
>-// TODO(kwiberg): Get rid of this class when the standard library has
>-// std::optional (and we're allowed to use it).
>-template <typename T>
>-class Optional final {
>- public:
>-  // Construct an empty Optional.
>-  Optional() : has_value_(false), empty_('\0') { PoisonValue(); }
>-
>-  Optional(rtc::nullopt_t)  // NOLINT(runtime/explicit)
>-      : Optional() {}
>-
>-  // Construct an Optional that contains a value.
>-  Optional(const T& value)  // NOLINT(runtime/explicit)
>-      : has_value_(true) {
>-    new (&value_) T(value);
>-  }
>-  Optional(T&& value)  // NOLINT(runtime/explicit)
>-      : has_value_(true) {
>-    new (&value_) T(std::move(value));
>-  }
>-
>-  // Copy constructor: copies the value from m if it has one.
>-  Optional(const Optional& m) : has_value_(m.has_value_) {
>-    if (has_value_)
>-      new (&value_) T(m.value_);
>-    else
>-      PoisonValue();
>-  }
>-
>-  // Move constructor: if m has a value, moves the value from m, leaving m
>-  // still in a state where it has a value, but a moved-from one (the
>-  // properties of which depends on T; the only general guarantee is that we
>-  // can destroy m).
>-  Optional(Optional&& m) : has_value_(m.has_value_) {
>-    if (has_value_)
>-      new (&value_) T(std::move(m.value_));
>-    else
>-      PoisonValue();
>-  }
>-
>-  ~Optional() {
>-    if (has_value_)
>-      value_.~T();
>-    else
>-      UnpoisonValue();
>-  }
>-
>-  Optional& operator=(rtc::nullopt_t) {
>-    reset();
>-    return *this;
>-  }
>-
>-  // Copy assignment. Uses T's copy assignment if both sides have a value, T's
>-  // copy constructor if only the right-hand side has a value.
>-  Optional& operator=(const Optional& m) {
>-    if (m.has_value_) {
>-      if (has_value_) {
>-        value_ = m.value_;  // T's copy assignment.
>-      } else {
>-        UnpoisonValue();
>-        new (&value_) T(m.value_);  // T's copy constructor.
>-        has_value_ = true;
>-      }
>-    } else {
>-      reset();
>-    }
>-    return *this;
>-  }
>-
>-  // Move assignment. Uses T's move assignment if both sides have a value, T's
>-  // move constructor if only the right-hand side has a value. The state of m
>-  // after it's been moved from is as for the move constructor.
>-  Optional& operator=(Optional&& m) {
>-    if (m.has_value_) {
>-      if (has_value_) {
>-        value_ = std::move(m.value_);  // T's move assignment.
>-      } else {
>-        UnpoisonValue();
>-        new (&value_) T(std::move(m.value_));  // T's move constructor.
>-        has_value_ = true;
>-      }
>-    } else {
>-      reset();
>-    }
>-    return *this;
>-  }
>-
>-  // Swap the values if both m1 and m2 have values; move the value if only one
>-  // of them has one.
>-  friend void swap(Optional& m1, Optional& m2) {
>-    if (m1.has_value_) {
>-      if (m2.has_value_) {
>-        // Both have values: swap.
>-        using std::swap;
>-        swap(m1.value_, m2.value_);
>-      } else {
>-        // Only m1 has a value: move it to m2.
>-        m2.UnpoisonValue();
>-        new (&m2.value_) T(std::move(m1.value_));
>-        m1.value_.~T();  // Destroy the moved-from value.
>-        m1.has_value_ = false;
>-        m2.has_value_ = true;
>-        m1.PoisonValue();
>-      }
>-    } else if (m2.has_value_) {
>-      // Only m2 has a value: move it to m1.
>-      m1.UnpoisonValue();
>-      new (&m1.value_) T(std::move(m2.value_));
>-      m2.value_.~T();  // Destroy the moved-from value.
>-      m1.has_value_ = true;
>-      m2.has_value_ = false;
>-      m2.PoisonValue();
>-    }
>-  }
>-
>-  // Destroy any contained value. Has no effect if we have no value.
>-  void reset() {
>-    if (!has_value_)
>-      return;
>-    value_.~T();
>-    has_value_ = false;
>-    PoisonValue();
>-  }
>-
>-  template <class... Args>
>-  void emplace(Args&&... args) {
>-    if (has_value_)
>-      value_.~T();
>-    else
>-      UnpoisonValue();
>-    new (&value_) T(std::forward<Args>(args)...);
>-    has_value_ = true;
>-  }
>-
>-  // Conversion to bool to test if we have a value.
>-  explicit operator bool() const { return has_value_; }
>-  bool has_value() const { return has_value_; }
>-
>-  // Dereferencing. Only allowed if we have a value.
>-  const T* operator->() const {
>-    RTC_DCHECK(has_value_);
>-    return &value_;
>-  }
>-  T* operator->() {
>-    RTC_DCHECK(has_value_);
>-    return &value_;
>-  }
>-  const T& operator*() const {
>-    RTC_DCHECK(has_value_);
>-    return value_;
>-  }
>-  T& operator*() {
>-    RTC_DCHECK(has_value_);
>-    return value_;
>-  }
>-  const T& value() const {
>-    RTC_DCHECK(has_value_);
>-    return value_;
>-  }
>-  T& value() {
>-    RTC_DCHECK(has_value_);
>-    return value_;
>-  }
>-
>-  // Dereference with a default value in case we don't have a value.
>-  const T& value_or(const T& default_val) const {
>-    // The no-op call prevents the compiler from generating optimized code that
>-    // reads value_ even if !has_value_, but only if FunctionThatDoesNothing is
>-    // not completely inlined; see its declaration.).
>-    return has_value_ ? *optional_internal::FunctionThatDoesNothing(&value_)
>-                      : default_val;
>-  }
>-
>-  // Dereference and move value.
>-  T MoveValue() {
>-    RTC_DCHECK(has_value_);
>-    return std::move(value_);
>-  }
>-
>-  // Equality tests. Two Optionals are equal if they contain equivalent values,
>-  // or if they're both empty.
>-  friend bool operator==(const Optional& m1, const Optional& m2) {
>-    return m1.has_value_ && m2.has_value_ ? m1.value_ == m2.value_
>-                                          : m1.has_value_ == m2.has_value_;
>-  }
>-  friend bool operator==(const Optional& opt, const T& value) {
>-    return opt.has_value_ && opt.value_ == value;
>-  }
>-  friend bool operator==(const T& value, const Optional& opt) {
>-    return opt.has_value_ && value == opt.value_;
>-  }
>-
>-  friend bool operator==(const Optional& opt, rtc::nullopt_t) {
>-    return !opt.has_value_;
>-  }
>-
>-  friend bool operator==(rtc::nullopt_t, const Optional& opt) {
>-    return !opt.has_value_;
>-  }
>-
>-  friend bool operator!=(const Optional& m1, const Optional& m2) {
>-    return m1.has_value_ && m2.has_value_ ? m1.value_ != m2.value_
>-                                          : m1.has_value_ != m2.has_value_;
>-  }
>-  friend bool operator!=(const Optional& opt, const T& value) {
>-    return !opt.has_value_ || opt.value_ != value;
>-  }
>-  friend bool operator!=(const T& value, const Optional& opt) {
>-    return !opt.has_value_ || value != opt.value_;
>-  }
>-
>-  friend bool operator!=(const Optional& opt, rtc::nullopt_t) {
>-    return opt.has_value_;
>-  }
>-
>-  friend bool operator!=(rtc::nullopt_t, const Optional& opt) {
>-    return opt.has_value_;
>-  }
>-
>- private:
>-  // Tell sanitizers that value_ shouldn't be touched.
>-  void PoisonValue() {
>-    rtc::AsanPoison(rtc::MakeArrayView(&value_, 1));
>-    rtc::MsanMarkUninitialized(rtc::MakeArrayView(&value_, 1));
>-  }
>-
>-  // Tell sanitizers that value_ is OK to touch again.
>-  void UnpoisonValue() { rtc::AsanUnpoison(rtc::MakeArrayView(&value_, 1)); }
>-
>-  bool has_value_;  // True iff value_ contains a live value.
>-  union {
>-    // empty_ exists only to make it possible to initialize the union, even when
>-    // it doesn't contain any data. If the union goes uninitialized, it may
>-    // trigger compiler warnings.
>-    char empty_;
>-    // By placing value_ in a union, we get to manage its construction and
>-    // destruction manually: the Optional constructors won't automatically
>-    // construct it, and the Optional destructor won't automatically destroy
>-    // it. Basically, this just allocates a properly sized and aligned block of
>-    // memory in which we can manually put a T with placement new.
>-    T value_;
>-  };
>-};
>-
>-#ifdef UNIT_TEST
>-namespace optional_internal {
>-
>-// Checks if there's a valid PrintTo(const T&, std::ostream*) call for T.
>-template <typename T>
>-struct HasPrintTo {
>- private:
>-  struct No {};
>-
>-  template <typename T2>
>-  static auto Test(const T2& obj)
>-      -> decltype(PrintTo(obj, std::declval<std::ostream*>()));
>-
>-  template <typename>
>-  static No Test(...);
>-
>- public:
>-  static constexpr bool value =
>-      !std::is_same<decltype(Test<T>(std::declval<const T&>())), No>::value;
>-};
>-
>-// Checks if there's a valid operator<<(std::ostream&, const T&) call for T.
>-template <typename T>
>-struct HasOstreamOperator {
>- private:
>-  struct No {};
>-
>-  template <typename T2>
>-  static auto Test(const T2& obj)
>-      -> decltype(std::declval<std::ostream&>() << obj);
>-
>-  template <typename>
>-  static No Test(...);
>-
>- public:
>-  static constexpr bool value =
>-      !std::is_same<decltype(Test<T>(std::declval<const T&>())), No>::value;
>-};
>-
>-// Prefer using PrintTo to print the object.
>-template <typename T>
>-typename std::enable_if<HasPrintTo<T>::value, void>::type OptionalPrintToHelper(
>-    const T& value,
>-    std::ostream* os) {
>-  PrintTo(value, os);
>-}
>-
>-// Fall back to operator<<(std::ostream&, ...) if it exists.
>-template <typename T>
>-typename std::enable_if<HasOstreamOperator<T>::value && !HasPrintTo<T>::value,
>-                        void>::type
>-OptionalPrintToHelper(const T& value, std::ostream* os) {
>-  *os << value;
>-}
>-
>-inline void OptionalPrintObjectBytes(const unsigned char* bytes,
>-                                     size_t size,
>-                                     std::ostream* os) {
>-  *os << "<optional with " << size << "-byte object [";
>-  for (size_t i = 0; i != size; ++i) {
>-    *os << (i == 0 ? "" : ((i & 1) ? "-" : " "));
>-    *os << std::hex << std::setw(2) << std::setfill('0')
>-        << static_cast<int>(bytes[i]);
>-  }
>-  *os << "]>";
>-}
>-
>-// As a final back-up, just print the contents of the objcets byte-wise.
>-template <typename T>
>-typename std::enable_if<!HasOstreamOperator<T>::value && !HasPrintTo<T>::value,
>-                        void>::type
>-OptionalPrintToHelper(const T& value, std::ostream* os) {
>-  OptionalPrintObjectBytes(reinterpret_cast<const unsigned char*>(&value),
>-                           sizeof(value), os);
>-}
>-
>-}  // namespace optional_internal
>-
>-// PrintTo is used by gtest to print out the results of tests. We want to ensure
>-// the object contained in an Optional can be printed out if it's set, while
>-// avoiding touching the object's storage if it is undefined.
>-template <typename T>
>-void PrintTo(const rtc::Optional<T>& opt, std::ostream* os) {
>-  if (opt) {
>-    optional_internal::OptionalPrintToHelper(*opt, os);
>-  } else {
>-    *os << "<empty optional>";
>-  }
>-}
>-
>-#endif  // UNIT_TEST
>-
>-}  // namespace rtc
>-
>-#endif  // API_OPTIONAL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/optional_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/optional_unittest.cc
>deleted file mode 100644
>index 2149033f404..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/optional_unittest.cc
>+++ /dev/null
>@@ -1,902 +0,0 @@
>-/*
>- *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include <memory>
>-#include <sstream>
>-#include <string>
>-#include <utility>
>-#include <vector>
>-
>-#include "api/optional.h"
>-#include "rtc_base/gunit.h"
>-
>-namespace rtc {
>-
>-namespace {
>-
>-struct MyUnprintableType {
>-  int value;
>-};
>-
>-struct MyPrintableType {
>-  int value;
>-};
>-
>-struct MyOstreamPrintableType {
>-  int value;
>-};
>-
>-void PrintTo(const MyPrintableType& mpt, std::ostream* os) {
>-  *os << "The value is " << mpt.value;
>-}
>-
>-std::ostream& operator<<(std::ostream& os, const MyPrintableType& mpt) {
>-  os << mpt.value;
>-  return os;
>-}
>-
>-std::ostream& operator<<(std::ostream& os, const MyOstreamPrintableType& mpt) {
>-  os << mpt.value;
>-  return os;
>-}
>-
>-// Class whose instances logs various method calls (constructor, destructor,
>-// etc.). Each instance has a unique ID (a simple global sequence number) and
>-// an origin ID. When a copy is made, the new object gets a fresh ID but copies
>-// the origin ID from the original. When a new Logger is created from scratch,
>-// it gets a fresh ID, and the origin ID is the same as the ID (default
>-// constructor) or given as an argument (explicit constructor).
>-class Logger {
>- public:
>-  Logger() : id_(g_next_id++), origin_(id_) { Log("default constructor"); }
>-  explicit Logger(int origin) : id_(g_next_id++), origin_(origin) {
>-    Log("explicit constructor");
>-  }
>-  Logger(int origin, const Logger& pass_by_ref, Logger pass_by_value)
>-      : id_(g_next_id++), origin_(origin) {
>-    Log("multi parameter constructor");
>-  }
>-  Logger(const Logger& other) : id_(g_next_id++), origin_(other.origin_) {
>-    LogFrom("copy constructor", other);
>-  }
>-  Logger(Logger&& other) : id_(g_next_id++), origin_(other.origin_) {
>-    LogFrom("move constructor", other);
>-  }
>-  ~Logger() { Log("destructor"); }
>-  Logger& operator=(const Logger& other) {
>-    origin_ = other.origin_;
>-    LogFrom("operator= copy", other);
>-    return *this;
>-  }
>-  Logger& operator=(Logger&& other) {
>-    origin_ = other.origin_;
>-    LogFrom("operator= move", other);
>-    return *this;
>-  }
>-  friend void swap(Logger& a, Logger& b) {
>-    using std::swap;
>-    swap(a.origin_, b.origin_);
>-    Log2("swap", a, b);
>-  }
>-  friend bool operator==(const Logger& a, const Logger& b) {
>-    Log2("operator==", a, b);
>-    return a.origin_ == b.origin_;
>-  }
>-  friend bool operator!=(const Logger& a, const Logger& b) {
>-    Log2("operator!=", a, b);
>-    return a.origin_ != b.origin_;
>-  }
>-  void Foo() { Log("Foo()"); }
>-  void Foo() const { Log("Foo() const"); }
>-  static std::unique_ptr<std::vector<std::string>> Setup() {
>-    std::unique_ptr<std::vector<std::string>> s(new std::vector<std::string>);
>-    g_log = s.get();
>-    g_next_id = 0;
>-    return s;
>-  }
>-
>- private:
>-  int id_;
>-  int origin_;
>-  static std::vector<std::string>* g_log;
>-  static int g_next_id;
>-  void Log(const char* msg) const {
>-    std::ostringstream oss;
>-    oss << id_ << ':' << origin_ << ". " << msg;
>-    g_log->push_back(oss.str());
>-  }
>-  void LogFrom(const char* msg, const Logger& other) const {
>-    std::ostringstream oss;
>-    oss << id_ << ':' << origin_ << ". " << msg << " (from " << other.id_ << ':'
>-        << other.origin_ << ")";
>-    g_log->push_back(oss.str());
>-  }
>-  static void Log2(const char* msg, const Logger& a, const Logger& b) {
>-    std::ostringstream oss;
>-    oss << msg << ' ' << a.id_ << ':' << a.origin_ << ", " << b.id_ << ':'
>-        << b.origin_;
>-    g_log->push_back(oss.str());
>-  }
>-};
>-
>-std::vector<std::string>* Logger::g_log = nullptr;
>-int Logger::g_next_id = 0;
>-
>-// Append all the other args to the vector pointed to by the first arg.
>-template <typename T>
>-void VectorAppend(std::vector<T>* v) {}
>-template <typename T, typename... Ts>
>-void VectorAppend(std::vector<T>* v, const T& e, Ts... es) {
>-  v->push_back(e);
>-  VectorAppend(v, es...);
>-}
>-
>-// Create a vector of strings. Because we're not allowed to use
>-// std::initializer_list.
>-template <typename... Ts>
>-std::vector<std::string> V(Ts... es) {
>-  std::vector<std::string> strings;
>-  VectorAppend(&strings, static_cast<std::string>(es)...);
>-  return strings;
>-}
>-
>-}  // namespace
>-
>-TEST(OptionalTest, TestConstructDefault) {
>-  auto log = Logger::Setup();
>-  {
>-    Optional<Logger> x;
>-    EXPECT_FALSE(x);
>-    EXPECT_FALSE(x.has_value());
>-  }
>-  EXPECT_EQ(V(), *log);
>-}
>-
>-TEST(OptionalTest, TestConstructNullopt) {
>-  auto log = Logger::Setup();
>-  {
>-    Optional<Logger> x(nullopt);
>-    EXPECT_FALSE(x);
>-    EXPECT_FALSE(x.has_value());
>-  }
>-  EXPECT_EQ(V(), *log);
>-}
>-
>-TEST(OptionalTest, TestConstructCopyEmpty) {
>-  auto log = Logger::Setup();
>-  {
>-    Optional<Logger> x;
>-    EXPECT_FALSE(x);
>-    EXPECT_FALSE(x.has_value());
>-    auto y = x;
>-    EXPECT_FALSE(y);
>-    EXPECT_FALSE(y.has_value());
>-  }
>-  EXPECT_EQ(V(), *log);
>-}
>-
>-TEST(OptionalTest, TestConstructCopyFull) {
>-  auto log = Logger::Setup();
>-  {
>-    Logger a;
>-    Optional<Logger> x(a);
>-    EXPECT_TRUE(x);
>-    EXPECT_TRUE(x.has_value());
>-    log->push_back("---");
>-    auto y = x;
>-    EXPECT_TRUE(y);
>-    EXPECT_TRUE(y.has_value());
>-    log->push_back("---");
>-  }
>-  EXPECT_EQ(V("0:0. default constructor", "1:0. copy constructor (from 0:0)",
>-              "---", "2:0. copy constructor (from 1:0)", "---",
>-              "2:0. destructor", "1:0. destructor", "0:0. destructor"),
>-            *log);
>-}
>-
>-TEST(OptionalTest, TestConstructMoveEmpty) {
>-  auto log = Logger::Setup();
>-  {
>-    Optional<Logger> x;
>-    EXPECT_FALSE(x);
>-    EXPECT_FALSE(x.has_value());
>-    auto y = std::move(x);
>-    EXPECT_FALSE(y);
>-    EXPECT_FALSE(y.has_value());
>-  }
>-  EXPECT_EQ(V(), *log);
>-}
>-
>-TEST(OptionalTest, TestConstructMoveFull) {
>-  auto log = Logger::Setup();
>-  {
>-    Optional<Logger> x(Logger(17));
>-    EXPECT_TRUE(x);
>-    EXPECT_TRUE(x.has_value());
>-    log->push_back("---");
>-    auto y = std::move(x);
>-    EXPECT_TRUE(x);
>-    EXPECT_TRUE(x.has_value());
>-    EXPECT_TRUE(y);
>-    EXPECT_TRUE(y.has_value());
>-    log->push_back("---");
>-  }
>-  EXPECT_EQ(
>-      V("0:17. explicit constructor", "1:17. move constructor (from 0:17)",
>-        "0:17. destructor", "---", "2:17. move constructor (from 1:17)", "---",
>-        "2:17. destructor", "1:17. destructor"),
>-      *log);
>-}
>-
>-TEST(OptionalTest, TestCopyAssignToEmptyFromEmpty) {
>-  auto log = Logger::Setup();
>-  {
>-    Optional<Logger> x, y;
>-    x = y;
>-  }
>-  EXPECT_EQ(V(), *log);
>-}
>-
>-TEST(OptionalTest, TestCopyAssignToFullFromEmpty) {
>-  auto log = Logger::Setup();
>-  {
>-    Optional<Logger> x(Logger(17));
>-    Optional<Logger> y;
>-    log->push_back("---");
>-    x = y;
>-    log->push_back("---");
>-  }
>-  EXPECT_EQ(
>-      V("0:17. explicit constructor", "1:17. move constructor (from 0:17)",
>-        "0:17. destructor", "---", "1:17. destructor", "---"),
>-      *log);
>-}
>-
>-TEST(OptionalTest, TestCopyAssignToFullFromNullopt) {
>-  auto log = Logger::Setup();
>-  {
>-    Optional<Logger> x(Logger(17));
>-    log->push_back("---");
>-    x = nullopt;
>-    log->push_back("---");
>-    EXPECT_FALSE(x);
>-  }
>-  EXPECT_EQ(
>-      V("0:17. explicit constructor", "1:17. move constructor (from 0:17)",
>-        "0:17. destructor", "---", "1:17. destructor", "---"),
>-      *log);
>-}
>-
>-TEST(OptionalTest, TestCopyAssignToFullFromEmptyBraces) {
>-  auto log = Logger::Setup();
>-  {
>-    Optional<Logger> x(Logger(17));
>-    log->push_back("---");
>-    x = {};
>-    log->push_back("---");
>-    EXPECT_FALSE(x);
>-  }
>-  EXPECT_EQ(
>-      V("0:17. explicit constructor", "1:17. move constructor (from 0:17)",
>-        "0:17. destructor", "---", "1:17. destructor", "---"),
>-      *log);
>-}
>-
>-TEST(OptionalTest, TestCopyAssignToEmptyFromFull) {
>-  auto log = Logger::Setup();
>-  {
>-    Optional<Logger> x;
>-    Optional<Logger> y(Logger(17));
>-    log->push_back("---");
>-    x = y;
>-    log->push_back("---");
>-  }
>-  EXPECT_EQ(
>-      V("0:17. explicit constructor", "1:17. move constructor (from 0:17)",
>-        "0:17. destructor", "---", "2:17. copy constructor (from 1:17)", "---",
>-        "1:17. destructor", "2:17. destructor"),
>-      *log);
>-}
>-
>-TEST(OptionalTest, TestCopyAssignToFullFromFull) {
>-  auto log = Logger::Setup();
>-  {
>-    Optional<Logger> x(Logger(17));
>-    Optional<Logger> y(Logger(42));
>-    log->push_back("---");
>-    x = y;
>-    log->push_back("---");
>-  }
>-  EXPECT_EQ(
>-      V("0:17. explicit constructor", "1:17. move constructor (from 0:17)",
>-        "0:17. destructor", "2:42. explicit constructor",
>-        "3:42. move constructor (from 2:42)", "2:42. destructor", "---",
>-        "1:42. operator= copy (from 3:42)", "---", "3:42. destructor",
>-        "1:42. destructor"),
>-      *log);
>-}
>-
>-TEST(OptionalTest, TestCopyAssignToEmptyFromT) {
>-  auto log = Logger::Setup();
>-  {
>-    Optional<Logger> x;
>-    Logger y(17);
>-    log->push_back("---");
>-    x = Optional<Logger>(y);
>-    log->push_back("---");
>-  }
>-  EXPECT_EQ(V("0:17. explicit constructor", "---",
>-              "1:17. copy constructor (from 0:17)",
>-              "2:17. move constructor (from 1:17)", "1:17. destructor", "---",
>-              "0:17. destructor", "2:17. destructor"),
>-            *log);
>-}
>-
>-TEST(OptionalTest, TestCopyAssignToFullFromT) {
>-  auto log = Logger::Setup();
>-  {
>-    Optional<Logger> x(Logger(17));
>-    Logger y(42);
>-    log->push_back("---");
>-    x = Optional<Logger>(y);
>-    log->push_back("---");
>-  }
>-  EXPECT_EQ(
>-      V("0:17. explicit constructor", "1:17. move constructor (from 0:17)",
>-        "0:17. destructor", "2:42. explicit constructor", "---",
>-        "3:42. copy constructor (from 2:42)",
>-        "1:42. operator= move (from 3:42)", "3:42. destructor", "---",
>-        "2:42. destructor", "1:42. destructor"),
>-      *log);
>-}
>-
>-TEST(OptionalTest, TestMoveAssignToEmptyFromEmpty) {
>-  auto log = Logger::Setup();
>-  {
>-    Optional<Logger> x, y;
>-    x = std::move(y);
>-  }
>-  EXPECT_EQ(V(), *log);
>-}
>-
>-TEST(OptionalTest, TestMoveAssignToFullFromEmpty) {
>-  auto log = Logger::Setup();
>-  {
>-    Optional<Logger> x(Logger(17));
>-    Optional<Logger> y;
>-    log->push_back("---");
>-    x = std::move(y);
>-    log->push_back("---");
>-  }
>-  EXPECT_EQ(
>-      V("0:17. explicit constructor", "1:17. move constructor (from 0:17)",
>-        "0:17. destructor", "---", "1:17. destructor", "---"),
>-      *log);
>-}
>-
>-TEST(OptionalTest, TestMoveAssignToEmptyFromFull) {
>-  auto log = Logger::Setup();
>-  {
>-    Optional<Logger> x;
>-    Optional<Logger> y(Logger(17));
>-    log->push_back("---");
>-    x = std::move(y);
>-    log->push_back("---");
>-  }
>-  EXPECT_EQ(
>-      V("0:17. explicit constructor", "1:17. move constructor (from 0:17)",
>-        "0:17. destructor", "---", "2:17. move constructor (from 1:17)", "---",
>-        "1:17. destructor", "2:17. destructor"),
>-      *log);
>-}
>-
>-TEST(OptionalTest, TestMoveAssignToFullFromFull) {
>-  auto log = Logger::Setup();
>-  {
>-    Optional<Logger> x(Logger(17));
>-    Optional<Logger> y(Logger(42));
>-    log->push_back("---");
>-    x = std::move(y);
>-    log->push_back("---");
>-  }
>-  EXPECT_EQ(
>-      V("0:17. explicit constructor", "1:17. move constructor (from 0:17)",
>-        "0:17. destructor", "2:42. explicit constructor",
>-        "3:42. move constructor (from 2:42)", "2:42. destructor", "---",
>-        "1:42. operator= move (from 3:42)", "---", "3:42. destructor",
>-        "1:42. destructor"),
>-      *log);
>-}
>-
>-TEST(OptionalTest, TestMoveAssignToEmptyFromT) {
>-  auto log = Logger::Setup();
>-  {
>-    Optional<Logger> x;
>-    Logger y(17);
>-    log->push_back("---");
>-    x = Optional<Logger>(std::move(y));
>-    log->push_back("---");
>-  }
>-  EXPECT_EQ(V("0:17. explicit constructor", "---",
>-              "1:17. move constructor (from 0:17)",
>-              "2:17. move constructor (from 1:17)", "1:17. destructor", "---",
>-              "0:17. destructor", "2:17. destructor"),
>-            *log);
>-}
>-
>-TEST(OptionalTest, TestMoveAssignToFullFromT) {
>-  auto log = Logger::Setup();
>-  {
>-    Optional<Logger> x(Logger(17));
>-    Logger y(42);
>-    log->push_back("---");
>-    x = Optional<Logger>(std::move(y));
>-    log->push_back("---");
>-  }
>-  EXPECT_EQ(
>-      V("0:17. explicit constructor", "1:17. move constructor (from 0:17)",
>-        "0:17. destructor", "2:42. explicit constructor", "---",
>-        "3:42. move constructor (from 2:42)",
>-        "1:42. operator= move (from 3:42)", "3:42. destructor", "---",
>-        "2:42. destructor", "1:42. destructor"),
>-      *log);
>-}
>-
>-TEST(OptionalTest, TestResetEmpty) {
>-  auto log = Logger::Setup();
>-  {
>-    Optional<Logger> x;
>-    x.reset();
>-  }
>-  EXPECT_EQ(V(), *log);
>-}
>-
>-TEST(OptionalTest, TestResetFull) {
>-  auto log = Logger::Setup();
>-  {
>-    Optional<Logger> x(Logger(17));
>-    log->push_back("---");
>-    x.reset();
>-    log->push_back("---");
>-  }
>-  EXPECT_EQ(
>-      V("0:17. explicit constructor", "1:17. move constructor (from 0:17)",
>-        "0:17. destructor", "---", "1:17. destructor", "---"),
>-      *log);
>-}
>-
>-TEST(OptionalTest, TestEmplaceEmptyWithExplicit) {
>-  auto log = Logger::Setup();
>-  {
>-    Optional<Logger> x;
>-    log->push_back("---");
>-    x.emplace(42);
>-    log->push_back("---");
>-  }
>-  // clang-format off
>-  EXPECT_EQ(V("---",
>-              "0:42. explicit constructor",
>-              "---",
>-              "0:42. destructor"),
>-            *log);
>-  // clang-format on
>-}
>-
>-TEST(OptionalTest, TestEmplaceEmptyWithMultipleParameters) {
>-  auto log = Logger::Setup();
>-  {
>-    Optional<Logger> x;
>-    Logger ref(21);
>-    Logger value(35);
>-    log->push_back("---");
>-    x.emplace(42, ref, std::move(value));
>-    log->push_back("---");
>-  }
>-  // clang-format off
>-  EXPECT_EQ(V("0:21. explicit constructor",
>-              "1:35. explicit constructor",
>-              "---",
>-              "2:35. move constructor (from 1:35)",
>-              "3:42. multi parameter constructor",
>-              "2:35. destructor",
>-              "---",
>-              "1:35. destructor",
>-              "0:21. destructor",
>-              "3:42. destructor"),
>-            *log);
>-  // clang-format on
>-}
>-
>-TEST(OptionalTest, TestEmplaceEmptyWithCopy) {
>-  auto log = Logger::Setup();
>-  {
>-    Optional<Logger> x;
>-    Logger y(42);
>-    log->push_back("---");
>-    x.emplace(y);
>-    log->push_back("---");
>-  }
>-  // clang-format off
>-  EXPECT_EQ(V("0:42. explicit constructor",
>-              "---",
>-              "1:42. copy constructor (from 0:42)",
>-              "---",
>-              "0:42. destructor",
>-              "1:42. destructor"),
>-            *log);
>-  // clang-format on
>-}
>-
>-TEST(OptionalTest, TestEmplaceEmptyWithMove) {
>-  auto log = Logger::Setup();
>-  {
>-    Optional<Logger> x;
>-    Logger y(42);
>-    log->push_back("---");
>-    x.emplace(std::move(y));
>-    log->push_back("---");
>-  }
>-  // clang-format off
>-  EXPECT_EQ(V("0:42. explicit constructor",
>-              "---",
>-              "1:42. move constructor (from 0:42)",
>-              "---",
>-              "0:42. destructor",
>-              "1:42. destructor"),
>-            *log);
>-  // clang-format on
>-}
>-
>-TEST(OptionalTest, TestEmplaceFullWithExplicit) {
>-  auto log = Logger::Setup();
>-  {
>-    Optional<Logger> x(Logger(17));
>-    log->push_back("---");
>-    x.emplace(42);
>-    log->push_back("---");
>-  }
>-  // clang-format off
>-  EXPECT_EQ(
>-      V("0:17. explicit constructor",
>-        "1:17. move constructor (from 0:17)",
>-        "0:17. destructor",
>-        "---",
>-        "1:17. destructor",
>-        "2:42. explicit constructor",
>-        "---",
>-        "2:42. destructor"),
>-      *log);
>-  // clang-format on
>-}
>-
>-TEST(OptionalTest, TestEmplaceFullWithMultipleParameters) {
>-  auto log = Logger::Setup();
>-  {
>-    Optional<Logger> x(Logger(17));
>-    Logger ref(21);
>-    Logger value(35);
>-    log->push_back("---");
>-    x.emplace(42, ref, std::move(value));
>-    log->push_back("---");
>-  }
>-  // clang-format off
>-  EXPECT_EQ(V("0:17. explicit constructor",
>-              "1:17. move constructor (from 0:17)",
>-              "0:17. destructor",
>-              "2:21. explicit constructor",
>-              "3:35. explicit constructor",
>-              "---",
>-              "1:17. destructor",
>-              "4:35. move constructor (from 3:35)",
>-              "5:42. multi parameter constructor",
>-              "4:35. destructor",
>-              "---",
>-              "3:35. destructor",
>-              "2:21. destructor",
>-              "5:42. destructor"),
>-            *log);
>-  // clang-format on
>-}
>-
>-TEST(OptionalTest, TestEmplaceFullWithCopy) {
>-  auto log = Logger::Setup();
>-  {
>-    Optional<Logger> x(Logger(17));
>-    Logger y(42);
>-    log->push_back("---");
>-    x.emplace(y);
>-    log->push_back("---");
>-  }
>-  // clang-format off
>-  EXPECT_EQ(V("0:17. explicit constructor",
>-              "1:17. move constructor (from 0:17)",
>-              "0:17. destructor",
>-              "2:42. explicit constructor",
>-              "---",
>-              "1:17. destructor",
>-              "3:42. copy constructor (from 2:42)",
>-              "---",
>-              "2:42. destructor",
>-              "3:42. destructor"),
>-           *log);
>-  // clang-format on
>-}
>-
>-TEST(OptionalTest, TestEmplaceFullWithMove) {
>-  auto log = Logger::Setup();
>-  {
>-    Optional<Logger> x(Logger(17));
>-    Logger y(42);
>-    log->push_back("---");
>-    x.emplace(std::move(y));
>-    log->push_back("---");
>-  }
>-  // clang-format off
>-  EXPECT_EQ(V("0:17. explicit constructor",
>-              "1:17. move constructor (from 0:17)",
>-              "0:17. destructor",
>-              "2:42. explicit constructor",
>-              "---",
>-              "1:17. destructor",
>-              "3:42. move constructor (from 2:42)",
>-              "---",
>-              "2:42. destructor",
>-              "3:42. destructor"),
>-            *log);
>-  // clang-format on
>-}
>-
>-TEST(OptionalTest, TestDereference) {
>-  auto log = Logger::Setup();
>-  {
>-    Optional<Logger> x(Logger(42));
>-    const auto& y = x;
>-    log->push_back("---");
>-    x->Foo();
>-    y->Foo();
>-    std::move(x)->Foo();
>-    std::move(y)->Foo();
>-    log->push_back("---");
>-    (*x).Foo();
>-    (*y).Foo();
>-    (*std::move(x)).Foo();
>-    (*std::move(y)).Foo();
>-    log->push_back("---");
>-    x.value().Foo();
>-    y.value().Foo();
>-    std::move(x).value().Foo();
>-    std::move(y).value().Foo();
>-    log->push_back("---");
>-  }
>-  // clang-format off
>-  EXPECT_EQ(V("0:42. explicit constructor",
>-              "1:42. move constructor (from 0:42)",
>-              "0:42. destructor",
>-              "---",
>-              "1:42. Foo()",
>-              "1:42. Foo() const",
>-              "1:42. Foo()",
>-              "1:42. Foo() const",
>-              "---",
>-              "1:42. Foo()",
>-              "1:42. Foo() const",
>-              "1:42. Foo()",
>-              "1:42. Foo() const",
>-              "---",
>-              "1:42. Foo()",
>-              "1:42. Foo() const",
>-              "1:42. Foo()",
>-              "1:42. Foo() const",
>-              "---",
>-              "1:42. destructor"),
>-            *log);
>-  // clang-format on
>-}
>-
>-TEST(OptionalTest, TestDereferenceWithDefault) {
>-  auto log = Logger::Setup();
>-  {
>-    const Logger a(17), b(42);
>-    Optional<Logger> x(a);
>-    Optional<Logger> y;
>-    log->push_back("-1-");
>-    EXPECT_EQ(a, x.value_or(Logger(42)));
>-    log->push_back("-2-");
>-    EXPECT_EQ(b, y.value_or(Logger(42)));
>-    log->push_back("-3-");
>-    EXPECT_EQ(a, Optional<Logger>(Logger(17)).value_or(b));
>-    log->push_back("-4-");
>-    EXPECT_EQ(b, Optional<Logger>().value_or(b));
>-    log->push_back("-5-");
>-  }
>-  EXPECT_EQ(
>-      V("0:17. explicit constructor", "1:42. explicit constructor",
>-        "2:17. copy constructor (from 0:17)", "-1-",
>-        "3:42. explicit constructor", "operator== 0:17, 2:17",
>-        "3:42. destructor", "-2-", "4:42. explicit constructor",
>-        "operator== 1:42, 4:42", "4:42. destructor", "-3-",
>-        "5:17. explicit constructor", "6:17. move constructor (from 5:17)",
>-        "operator== 0:17, 6:17", "6:17. destructor", "5:17. destructor", "-4-",
>-        "operator== 1:42, 1:42", "-5-", "2:17. destructor", "1:42. destructor",
>-        "0:17. destructor"),
>-      *log);
>-}
>-
>-TEST(OptionalTest, TestEquality) {
>-  auto log = Logger::Setup();
>-  {
>-    Logger a(17), b(42);
>-    Optional<Logger> ma1(a), ma2(a), mb(b), me1, me2;
>-    log->push_back("---");
>-    EXPECT_EQ(ma1, ma1);
>-    EXPECT_EQ(ma1, ma2);
>-    EXPECT_NE(ma1, mb);
>-    EXPECT_NE(ma1, me1);
>-    EXPECT_EQ(me1, me1);
>-    EXPECT_EQ(me1, me2);
>-    log->push_back("---");
>-  }
>-  EXPECT_EQ(
>-      V("0:17. explicit constructor", "1:42. explicit constructor",
>-        "2:17. copy constructor (from 0:17)",
>-        "3:17. copy constructor (from 0:17)",
>-        "4:42. copy constructor (from 1:42)", "---", "operator== 2:17, 2:17",
>-        "operator== 2:17, 3:17", "operator!= 2:17, 4:42", "---",
>-        "4:42. destructor", "3:17. destructor", "2:17. destructor",
>-        "1:42. destructor", "0:17. destructor"),
>-      *log);
>-}
>-
>-TEST(OptionalTest, TestEqualityWithNullopt) {
>-  auto log = Logger::Setup();
>-  {
>-    Logger a(17);
>-    Optional<Logger> ma(a), me;
>-    // Using operator== and operator!= explicitly instead of EXPECT_EQ/EXPECT_NE
>-    // macros because those operators are under test.
>-    log->push_back("---");
>-
>-    EXPECT_FALSE(ma == nullopt);
>-    EXPECT_FALSE(nullopt == ma);
>-    EXPECT_TRUE(me == nullopt);
>-    EXPECT_TRUE(nullopt == me);
>-
>-    EXPECT_TRUE(ma != nullopt);
>-    EXPECT_TRUE(nullopt != ma);
>-    EXPECT_FALSE(me != nullopt);
>-    EXPECT_FALSE(nullopt != me);
>-
>-    log->push_back("---");
>-  }
>-  // clang-format off
>-  EXPECT_EQ(V("0:17. explicit constructor",
>-              "1:17. copy constructor (from 0:17)",
>-              "---",
>-              // No operators should be called when comparing to empty.
>-              "---",
>-              "1:17. destructor",
>-              "0:17. destructor"),
>-            *log);
>-  // clang-format on
>-}
>-
>-TEST(OptionalTest, TestEqualityWithObject) {
>-  auto log = Logger::Setup();
>-  {
>-    Logger a(17), b(42);
>-    Optional<Logger> ma(a), me;
>-    // Using operator== and operator!= explicitly instead of EXPECT_EQ/EXPECT_NE
>-    // macros because those operators are under test.
>-    log->push_back("---");
>-
>-    EXPECT_TRUE(ma == a);
>-    EXPECT_TRUE(a == ma);
>-    EXPECT_FALSE(ma == b);
>-    EXPECT_FALSE(b == ma);
>-    EXPECT_FALSE(me == a);
>-    EXPECT_FALSE(a == me);
>-
>-    EXPECT_FALSE(ma != a);
>-    EXPECT_FALSE(a != ma);
>-    EXPECT_TRUE(ma != b);
>-    EXPECT_TRUE(b != ma);
>-    EXPECT_TRUE(me != a);
>-    EXPECT_TRUE(a != me);
>-
>-    log->push_back("---");
>-  }
>-  // clang-format off
>-  EXPECT_EQ(V("0:17. explicit constructor",
>-              "1:42. explicit constructor",
>-              "2:17. copy constructor (from 0:17)",
>-              "---",
>-              "operator== 2:17, 0:17",
>-              "operator== 0:17, 2:17",
>-              "operator== 2:17, 1:42",
>-              "operator== 1:42, 2:17",
>-              // No operator should be called when comparing to empty.
>-              "operator!= 2:17, 0:17",
>-              "operator!= 0:17, 2:17",
>-              "operator!= 2:17, 1:42",
>-              "operator!= 1:42, 2:17",
>-              // No operator should be called when comparing to empty.
>-              "---",
>-              "2:17. destructor",
>-              "1:42. destructor",
>-              "0:17. destructor"),
>-            *log);
>-  // clang-format on
>-}
>-
>-TEST(OptionalTest, TestSwap) {
>-  auto log = Logger::Setup();
>-  {
>-    Logger a(17), b(42);
>-    Optional<Logger> x1(a), x2(b), y1(a), y2, z1, z2;
>-    log->push_back("---");
>-    swap(x1, x2);  // Swap full <-> full.
>-    swap(y1, y2);  // Swap full <-> empty.
>-    swap(z1, z2);  // Swap empty <-> empty.
>-    log->push_back("---");
>-  }
>-  EXPECT_EQ(V("0:17. explicit constructor", "1:42. explicit constructor",
>-              "2:17. copy constructor (from 0:17)",
>-              "3:42. copy constructor (from 1:42)",
>-              "4:17. copy constructor (from 0:17)", "---", "swap 2:42, 3:17",
>-              "5:17. move constructor (from 4:17)", "4:17. destructor", "---",
>-              "5:17. destructor", "3:17. destructor", "2:42. destructor",
>-              "1:42. destructor", "0:17. destructor"),
>-            *log);
>-}
>-
>-TEST(OptionalTest, TestMoveValue) {
>-  auto log = Logger::Setup();
>-  {
>-    Optional<Logger> x(Logger(42));
>-    log->push_back("---");
>-    Logger moved = x.MoveValue();
>-    log->push_back("---");
>-  }
>-  EXPECT_EQ(
>-      V("0:42. explicit constructor", "1:42. move constructor (from 0:42)",
>-        "0:42. destructor", "---", "2:42. move constructor (from 1:42)", "---",
>-        "2:42. destructor", "1:42. destructor"),
>-      *log);
>-}
>-
>-TEST(OptionalTest, TestPrintTo) {
>-  constexpr char kEmptyOptionalMessage[] = "<empty optional>";
>-  const Optional<MyUnprintableType> empty_unprintable;
>-  const Optional<MyPrintableType> empty_printable;
>-  const Optional<MyOstreamPrintableType> empty_ostream_printable;
>-  EXPECT_EQ(kEmptyOptionalMessage, ::testing::PrintToString(empty_unprintable));
>-  EXPECT_EQ(kEmptyOptionalMessage, ::testing::PrintToString(empty_printable));
>-  EXPECT_EQ(kEmptyOptionalMessage,
>-            ::testing::PrintToString(empty_ostream_printable));
>-  EXPECT_NE("1", ::testing::PrintToString(Optional<MyUnprintableType>({1})));
>-  EXPECT_NE("1", ::testing::PrintToString(Optional<MyPrintableType>({1})));
>-  EXPECT_EQ("The value is 1",
>-            ::testing::PrintToString(Optional<MyPrintableType>({1})));
>-  EXPECT_EQ("1",
>-            ::testing::PrintToString(Optional<MyOstreamPrintableType>({1})));
>-}
>-
>-void UnusedFunctionWorkaround() {
>-  // These are here to ensure we don't get warnings about ostream and PrintTo
>-  // for MyPrintableType never getting called.
>-  const MyPrintableType dont_warn{17};
>-  const MyOstreamPrintableType dont_warn2{18};
>-  std::stringstream sstr;
>-  sstr << dont_warn;
>-  PrintTo(dont_warn, &sstr);
>-  sstr << dont_warn2;
>-}
>-
>-}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/mediadescription.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/mediadescription.h
>index 1a6d0e9037c..5cf1d1a67be 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/mediadescription.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/mediadescription.h
>@@ -15,8 +15,8 @@
> #include <utility>
> #include <vector>
> 
>+#include "absl/types/optional.h"
> #include "api/cryptoparams.h"
>-#include "api/optional.h"
> 
> namespace webrtc {
> 
>@@ -31,7 +31,7 @@ class MediaDescription {
>   // The mid(media stream identification) is used for identifying media streams
>   // within a session description.
>   // https://tools.ietf.org/html/rfc5888#section-6
>-  rtc::Optional<std::string> mid() const { return mid_; }
>+  absl::optional<std::string> mid() const { return mid_; }
>   void set_mid(std::string mid) { mid_.emplace(std::move(mid)); }
> 
>   // Security keys and parameters for this media stream. Can be used to
>@@ -43,7 +43,7 @@ class MediaDescription {
>   }
> 
>  private:
>-  rtc::Optional<std::string> mid_;
>+  absl::optional<std::string> mid_;
> 
>   std::vector<cricket::CryptoParams> sdes_params_;
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/ortcfactoryinterface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/ortcfactoryinterface.h
>index d99fcd44651..ea25c8a3cbc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/ortcfactoryinterface.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/ortcfactoryinterface.h
>@@ -215,20 +215,6 @@ class OrtcFactoryInterface {
>     return CreateAudioSource(cricket::AudioOptions());
>   }
> 
>-  // Creates a video source object wrapping and taking ownership of |capturer|.
>-  //
>-  // |constraints| can be used for selection of resolution and frame rate, and
>-  // may be null if no constraints are desired.
>-  virtual rtc::scoped_refptr<VideoTrackSourceInterface> CreateVideoSource(
>-      std::unique_ptr<cricket::VideoCapturer> capturer,
>-      const MediaConstraintsInterface* constraints) = 0;
>-
>-  // Version of the above method that omits |constraints|.
>-  rtc::scoped_refptr<VideoTrackSourceInterface> CreateVideoSource(
>-      std::unique_ptr<cricket::VideoCapturer> capturer) {
>-    return CreateVideoSource(std::move(capturer), nullptr);
>-  }
>-
>   // Creates a new local video track wrapping |source|. The same |source| can
>   // be used in several tracks.
>   virtual rtc::scoped_refptr<VideoTrackInterface> CreateVideoTrack(
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/packettransportinterface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/packettransportinterface.h
>index 9d53ad311bf..f357f8af63a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/packettransportinterface.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/packettransportinterface.h
>@@ -31,6 +31,7 @@ class PacketTransportInterface {
> 
>   // Classes that can use this internal interface.
>   friend class RtpTransportControllerAdapter;
>+  friend class RtpTransportAdapter;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/rtptransportinterface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/rtptransportinterface.h
>index 716a297c54e..b0d30e8560e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/rtptransportinterface.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/rtptransportinterface.h
>@@ -13,46 +13,17 @@
> 
> #include <string>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> #include "api/ortc/packettransportinterface.h"
> #include "api/rtcerror.h"
> #include "api/rtp_headers.h"
>+#include "api/rtpparameters.h"
> #include "common_types.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
> class RtpTransportAdapter;
> 
>-struct RtcpParameters final {
>-  // The SSRC to be used in the "SSRC of packet sender" field. If not set, one
>-  // will be chosen by the implementation.
>-  // TODO(deadbeef): Not implemented.
>-  rtc::Optional<uint32_t> ssrc;
>-
>-  // The Canonical Name (CNAME) used by RTCP (e.g. in SDES messages).
>-  //
>-  // If empty in the construction of the RtpTransport, one will be generated by
>-  // the implementation, and returned in GetRtcpParameters. Multiple
>-  // RtpTransports created by the same OrtcFactory will use the same generated
>-  // CNAME.
>-  //
>-  // If empty when passed into SetParameters, the CNAME simply won't be
>-  // modified.
>-  std::string cname;
>-
>-  // Send reduced-size RTCP?
>-  bool reduced_size = false;
>-
>-  // Send RTCP multiplexed on the RTP transport?
>-  bool mux = true;
>-
>-  bool operator==(const RtcpParameters& o) const {
>-    return ssrc == o.ssrc && cname == o.cname &&
>-           reduced_size == o.reduced_size && mux == o.mux;
>-  }
>-  bool operator!=(const RtcpParameters& o) const { return !(*this == o); }
>-};
>-
> struct RtpTransportParameters final {
>   RtcpParameters rtcp;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/sessiondescription_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/sessiondescription_unittest.cc
>index fd6f43de437..e4611c6c6b2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/sessiondescription_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/sessiondescription_unittest.cc
>@@ -20,4 +20,4 @@ TEST_F(SessionDescriptionTest, CreateSessionDescription) {
>   EXPECT_EQ(-1, s.session_id());
>   EXPECT_EQ("0", s.session_version());
> }
>-}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/srtptransportinterface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/srtptransportinterface.h
>index 41c8ccc9c14..4b757e400cc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/srtptransportinterface.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/ortc/srtptransportinterface.h
>@@ -11,9 +11,9 @@
> #ifndef API_ORTC_SRTPTRANSPORTINTERFACE_H_
> #define API_ORTC_SRTPTRANSPORTINTERFACE_H_
> 
>+#include "api/cryptoparams.h"
> #include "api/ortc/rtptransportinterface.h"
> #include "api/rtcerror.h"
>-#include "api/cryptoparams.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/peerconnectionfactoryproxy.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/peerconnectionfactoryproxy.h
>index 7601ed144ec..db52083567b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/peerconnectionfactoryproxy.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/peerconnectionfactoryproxy.h
>@@ -24,48 +24,59 @@ namespace webrtc {
> // TODO(deadbeef): Move this to .cc file and out of api/. What threads methods
> // are called on is an implementation detail.
> BEGIN_SIGNALING_PROXY_MAP(PeerConnectionFactory)
>-  PROXY_SIGNALING_THREAD_DESTRUCTOR()
>-  // Use the overloads of CreateVideoSource that take raw VideoCapturer
>-  // pointers from PeerConnectionFactoryInterface.
>-  // TODO(deadbeef): Remove this using statement once those overloads are
>-  // removed.
>-  using PeerConnectionFactoryInterface::CreateVideoSource;
>-  PROXY_METHOD1(void, SetOptions, const Options&)
>-  PROXY_METHOD5(rtc::scoped_refptr<PeerConnectionInterface>,
>-                CreatePeerConnection,
>-                const PeerConnectionInterface::RTCConfiguration&,
>-                const MediaConstraintsInterface*,
>-                std::unique_ptr<cricket::PortAllocator>,
>-                std::unique_ptr<rtc::RTCCertificateGeneratorInterface>,
>-                PeerConnectionObserver*);
>-  PROXY_METHOD4(rtc::scoped_refptr<PeerConnectionInterface>,
>-                CreatePeerConnection,
>-                const PeerConnectionInterface::RTCConfiguration&,
>-                std::unique_ptr<cricket::PortAllocator>,
>-                std::unique_ptr<rtc::RTCCertificateGeneratorInterface>,
>-                PeerConnectionObserver*);
>-  PROXY_METHOD1(rtc::scoped_refptr<MediaStreamInterface>,
>-                CreateLocalMediaStream, const std::string&)
>-  PROXY_METHOD1(rtc::scoped_refptr<AudioSourceInterface>,
>-                CreateAudioSource, const MediaConstraintsInterface*)
>-  PROXY_METHOD1(rtc::scoped_refptr<AudioSourceInterface>,
>-                CreateAudioSource,
>-                const cricket::AudioOptions&)
>-  PROXY_METHOD2(rtc::scoped_refptr<VideoTrackSourceInterface>,
>-                CreateVideoSource,
>-                std::unique_ptr<cricket::VideoCapturer>,
>-                const MediaConstraintsInterface*)
>-  PROXY_METHOD1(rtc::scoped_refptr<VideoTrackSourceInterface>,
>-                CreateVideoSource,
>-                std::unique_ptr<cricket::VideoCapturer>)
>-  PROXY_METHOD2(rtc::scoped_refptr<VideoTrackInterface>,
>-                CreateVideoTrack,
>-                const std::string&,
>-                VideoTrackSourceInterface*)
>-  PROXY_METHOD2(rtc::scoped_refptr<AudioTrackInterface>,
>-                CreateAudioTrack, const std::string&,  AudioSourceInterface*)
>-  PROXY_METHOD2(bool, StartAecDump, rtc::PlatformFile, int64_t)
>-  PROXY_METHOD0(void, StopAecDump)
>+PROXY_SIGNALING_THREAD_DESTRUCTOR()
>+// Use the overloads of CreateVideoSource that take raw VideoCapturer
>+// pointers from PeerConnectionFactoryInterface.
>+// TODO(deadbeef): Remove this using statement once those overloads are
>+// removed.
>+using PeerConnectionFactoryInterface::CreateVideoSource;
>+PROXY_METHOD1(void, SetOptions, const Options&)
>+PROXY_METHOD5(rtc::scoped_refptr<PeerConnectionInterface>,
>+              CreatePeerConnection,
>+              const PeerConnectionInterface::RTCConfiguration&,
>+              const MediaConstraintsInterface*,
>+              std::unique_ptr<cricket::PortAllocator>,
>+              std::unique_ptr<rtc::RTCCertificateGeneratorInterface>,
>+              PeerConnectionObserver*);
>+PROXY_METHOD4(rtc::scoped_refptr<PeerConnectionInterface>,
>+              CreatePeerConnection,
>+              const PeerConnectionInterface::RTCConfiguration&,
>+              std::unique_ptr<cricket::PortAllocator>,
>+              std::unique_ptr<rtc::RTCCertificateGeneratorInterface>,
>+              PeerConnectionObserver*);
>+PROXY_METHOD2(rtc::scoped_refptr<PeerConnectionInterface>,
>+              CreatePeerConnection,
>+              const PeerConnectionInterface::RTCConfiguration&,
>+              PeerConnectionDependencies);
>+PROXY_CONSTMETHOD1(webrtc::RtpCapabilities,
>+                   GetRtpSenderCapabilities,
>+                   cricket::MediaType);
>+PROXY_CONSTMETHOD1(webrtc::RtpCapabilities,
>+                   GetRtpReceiverCapabilities,
>+                   cricket::MediaType);
>+PROXY_METHOD1(rtc::scoped_refptr<MediaStreamInterface>,
>+              CreateLocalMediaStream,
>+              const std::string&)
>+PROXY_METHOD1(rtc::scoped_refptr<AudioSourceInterface>,
>+              CreateAudioSource,
>+              const cricket::AudioOptions&)
>+PROXY_METHOD2(rtc::scoped_refptr<VideoTrackSourceInterface>,
>+              CreateVideoSource,
>+              std::unique_ptr<cricket::VideoCapturer>,
>+              const MediaConstraintsInterface*)
>+PROXY_METHOD1(rtc::scoped_refptr<VideoTrackSourceInterface>,
>+              CreateVideoSource,
>+              std::unique_ptr<cricket::VideoCapturer>)
>+PROXY_METHOD2(rtc::scoped_refptr<VideoTrackInterface>,
>+              CreateVideoTrack,
>+              const std::string&,
>+              VideoTrackSourceInterface*)
>+PROXY_METHOD2(rtc::scoped_refptr<AudioTrackInterface>,
>+              CreateAudioTrack,
>+              const std::string&,
>+              AudioSourceInterface*)
>+PROXY_METHOD2(bool, StartAecDump, rtc::PlatformFile, int64_t)
>+PROXY_METHOD0(void, StopAecDump)
> END_PROXY_MAP()
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/peerconnectioninterface.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/peerconnectioninterface.cc
>new file mode 100644
>index 00000000000..ddaeb362c1a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/peerconnectioninterface.cc
>@@ -0,0 +1,258 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/peerconnectioninterface.h"
>+
>+namespace webrtc {
>+
>+PeerConnectionInterface::IceServer::IceServer() = default;
>+PeerConnectionInterface::IceServer::IceServer(const IceServer& rhs) = default;
>+PeerConnectionInterface::IceServer::~IceServer() = default;
>+
>+PeerConnectionInterface::RTCConfiguration::RTCConfiguration() = default;
>+
>+PeerConnectionInterface::RTCConfiguration::RTCConfiguration(
>+    const RTCConfiguration& rhs) = default;
>+
>+PeerConnectionInterface::RTCConfiguration::RTCConfiguration(
>+    RTCConfigurationType type) {
>+  if (type == RTCConfigurationType::kAggressive) {
>+    // These parameters are also defined in Java and IOS configurations,
>+    // so their values may be overwritten by the Java or IOS configuration.
>+    bundle_policy = kBundlePolicyMaxBundle;
>+    rtcp_mux_policy = kRtcpMuxPolicyRequire;
>+    ice_connection_receiving_timeout = kAggressiveIceConnectionReceivingTimeout;
>+
>+    // These parameters are not defined in Java or IOS configuration,
>+    // so their values will not be overwritten.
>+    enable_ice_renomination = true;
>+    redetermine_role_on_ice_restart = false;
>+  }
>+}
>+
>+PeerConnectionInterface::RTCConfiguration::~RTCConfiguration() = default;
>+
>+RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>>
>+PeerConnectionInterface::AddTrack(
>+    rtc::scoped_refptr<MediaStreamTrackInterface> track,
>+    const std::vector<std::string>& stream_ids) {
>+  return RTCError(RTCErrorType::UNSUPPORTED_OPERATION, "Not implemented");
>+}
>+
>+bool PeerConnectionInterface::RemoveTrack(RtpSenderInterface* sender) {
>+  return RemoveTrackNew(sender).ok();
>+}
>+
>+RTCError PeerConnectionInterface::RemoveTrackNew(
>+    rtc::scoped_refptr<RtpSenderInterface> sender) {
>+  return RTCError(RemoveTrack(sender) ? RTCErrorType::NONE
>+                                      : RTCErrorType::INTERNAL_ERROR);
>+}
>+
>+RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
>+PeerConnectionInterface::AddTransceiver(
>+    rtc::scoped_refptr<MediaStreamTrackInterface> track) {
>+  return RTCError(RTCErrorType::INTERNAL_ERROR, "not implemented");
>+}
>+
>+RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
>+PeerConnectionInterface::AddTransceiver(
>+    rtc::scoped_refptr<MediaStreamTrackInterface> track,
>+    const RtpTransceiverInit& init) {
>+  return RTCError(RTCErrorType::INTERNAL_ERROR, "not implemented");
>+}
>+
>+RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
>+PeerConnectionInterface::AddTransceiver(cricket::MediaType media_type) {
>+  return RTCError(RTCErrorType::INTERNAL_ERROR, "not implemented");
>+}
>+
>+RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
>+PeerConnectionInterface::AddTransceiver(cricket::MediaType media_type,
>+                                        const RtpTransceiverInit& init) {
>+  return RTCError(RTCErrorType::INTERNAL_ERROR, "not implemented");
>+}
>+
>+rtc::scoped_refptr<RtpSenderInterface> PeerConnectionInterface::CreateSender(
>+    const std::string& kind,
>+    const std::string& stream_id) {
>+  return rtc::scoped_refptr<RtpSenderInterface>();
>+}
>+
>+std::vector<rtc::scoped_refptr<RtpSenderInterface>>
>+PeerConnectionInterface::GetSenders() const {
>+  return std::vector<rtc::scoped_refptr<RtpSenderInterface>>();
>+}
>+
>+std::vector<rtc::scoped_refptr<RtpReceiverInterface>>
>+PeerConnectionInterface::GetReceivers() const {
>+  return std::vector<rtc::scoped_refptr<RtpReceiverInterface>>();
>+}
>+
>+std::vector<rtc::scoped_refptr<RtpTransceiverInterface>>
>+PeerConnectionInterface::GetTransceivers() const {
>+  return std::vector<rtc::scoped_refptr<RtpTransceiverInterface>>();
>+}
>+
>+const SessionDescriptionInterface*
>+PeerConnectionInterface::current_local_description() const {
>+  return nullptr;
>+}
>+
>+const SessionDescriptionInterface*
>+PeerConnectionInterface::current_remote_description() const {
>+  return nullptr;
>+}
>+
>+const SessionDescriptionInterface*
>+PeerConnectionInterface::pending_local_description() const {
>+  return nullptr;
>+}
>+
>+const SessionDescriptionInterface*
>+PeerConnectionInterface::pending_remote_description() const {
>+  return nullptr;
>+}
>+
>+PeerConnectionInterface::RTCConfiguration
>+PeerConnectionInterface::GetConfiguration() {
>+  return PeerConnectionInterface::RTCConfiguration();
>+}
>+
>+bool PeerConnectionInterface::SetConfiguration(
>+    const PeerConnectionInterface::RTCConfiguration& config,
>+    RTCError* error) {
>+  return false;
>+}
>+
>+bool PeerConnectionInterface::SetConfiguration(
>+    const PeerConnectionInterface::RTCConfiguration& config) {
>+  return false;
>+}
>+
>+bool PeerConnectionInterface::RemoveIceCandidates(
>+    const std::vector<cricket::Candidate>& candidates) {
>+  return false;
>+}
>+
>+RTCError PeerConnectionInterface::SetBitrate(const BitrateSettings& bitrate) {
>+  BitrateParameters bitrate_parameters;
>+  bitrate_parameters.min_bitrate_bps = bitrate.min_bitrate_bps;
>+  bitrate_parameters.current_bitrate_bps = bitrate.start_bitrate_bps;
>+  bitrate_parameters.max_bitrate_bps = bitrate.max_bitrate_bps;
>+  return SetBitrate(bitrate_parameters);
>+}
>+
>+RTCError PeerConnectionInterface::SetBitrate(
>+    const BitrateParameters& bitrate_parameters) {
>+  BitrateSettings bitrate;
>+  bitrate.min_bitrate_bps = bitrate_parameters.min_bitrate_bps;
>+  bitrate.start_bitrate_bps = bitrate_parameters.current_bitrate_bps;
>+  bitrate.max_bitrate_bps = bitrate_parameters.max_bitrate_bps;
>+  return SetBitrate(bitrate);
>+}
>+
>+bool PeerConnectionInterface::StartRtcEventLog(rtc::PlatformFile file,
>+                                               int64_t max_size_bytes) {
>+  return false;
>+}
>+
>+bool PeerConnectionInterface::StartRtcEventLog(
>+    std::unique_ptr<RtcEventLogOutput> output,
>+    int64_t output_period_ms) {
>+  return false;
>+}
>+
>+PeerConnectionInterface::BitrateParameters::BitrateParameters() = default;
>+
>+PeerConnectionInterface::BitrateParameters::~BitrateParameters() = default;
>+
>+PeerConnectionDependencies::PeerConnectionDependencies(
>+    PeerConnectionObserver* observer_in)
>+    : observer(observer_in) {}
>+
>+PeerConnectionDependencies::PeerConnectionDependencies(
>+    PeerConnectionDependencies&&) = default;
>+
>+PeerConnectionDependencies::~PeerConnectionDependencies() = default;
>+
>+PeerConnectionFactoryDependencies::PeerConnectionFactoryDependencies() =
>+    default;
>+
>+PeerConnectionFactoryDependencies::PeerConnectionFactoryDependencies(
>+    PeerConnectionFactoryDependencies&&) = default;
>+
>+PeerConnectionFactoryDependencies::~PeerConnectionFactoryDependencies() =
>+    default;
>+
>+rtc::scoped_refptr<PeerConnectionInterface>
>+PeerConnectionFactoryInterface::CreatePeerConnection(
>+    const PeerConnectionInterface::RTCConfiguration& configuration,
>+    const MediaConstraintsInterface* constraints,
>+    std::unique_ptr<cricket::PortAllocator> allocator,
>+    std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
>+    PeerConnectionObserver* observer) {
>+  return nullptr;
>+}
>+
>+rtc::scoped_refptr<PeerConnectionInterface>
>+PeerConnectionFactoryInterface::CreatePeerConnection(
>+    const PeerConnectionInterface::RTCConfiguration& configuration,
>+    std::unique_ptr<cricket::PortAllocator> allocator,
>+    std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
>+    PeerConnectionObserver* observer) {
>+  return nullptr;
>+}
>+
>+rtc::scoped_refptr<PeerConnectionInterface>
>+PeerConnectionFactoryInterface::CreatePeerConnection(
>+    const PeerConnectionInterface::RTCConfiguration& configuration,
>+    PeerConnectionDependencies dependencies) {
>+  return nullptr;
>+}
>+
>+RtpCapabilities PeerConnectionFactoryInterface::GetRtpSenderCapabilities(
>+    cricket::MediaType kind) const {
>+  return {};
>+}
>+
>+RtpCapabilities PeerConnectionFactoryInterface::GetRtpReceiverCapabilities(
>+    cricket::MediaType kind) const {
>+  return {};
>+}
>+
>+rtc::scoped_refptr<VideoTrackSourceInterface>
>+PeerConnectionFactoryInterface::CreateVideoSource(
>+    std::unique_ptr<cricket::VideoCapturer> capturer) {
>+  return nullptr;
>+}
>+
>+rtc::scoped_refptr<VideoTrackSourceInterface>
>+PeerConnectionFactoryInterface::CreateVideoSource(
>+    std::unique_ptr<cricket::VideoCapturer> capturer,
>+    const MediaConstraintsInterface* constraints) {
>+  return nullptr;
>+}
>+
>+rtc::scoped_refptr<VideoTrackSourceInterface>
>+PeerConnectionFactoryInterface::CreateVideoSource(
>+    cricket::VideoCapturer* capturer) {
>+  return CreateVideoSource(std::unique_ptr<cricket::VideoCapturer>(capturer));
>+}
>+
>+rtc::scoped_refptr<VideoTrackSourceInterface>
>+PeerConnectionFactoryInterface::CreateVideoSource(
>+    cricket::VideoCapturer* capturer,
>+    const MediaConstraintsInterface* constraints) {
>+  return CreateVideoSource(std::unique_ptr<cricket::VideoCapturer>(capturer),
>+                           constraints);
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/peerconnectioninterface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/peerconnectioninterface.h
>index a4d57eb7dfd..0fc2a2b5b8a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/peerconnectioninterface.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/peerconnectioninterface.h
>@@ -9,7 +9,7 @@
>  */
> 
> // This file contains the PeerConnection interface as defined in
>-// http://dev.w3.org/2011/webrtc/editor/webrtc.html#peer-to-peer-connections.
>+// https://w3c.github.io/webrtc-pc/#peer-to-peer-connections
> //
> // The PeerConnectionFactory class provides factory methods to create
> // PeerConnection, MediaStream and MediaStreamTrack objects.
>@@ -67,19 +67,20 @@
> #ifndef API_PEERCONNECTIONINTERFACE_H_
> #define API_PEERCONNECTIONINTERFACE_H_
> 
>-// TODO(sakal): Remove this define after migration to virtual PeerConnection
>-// observer is complete.
>-#define VIRTUAL_PEERCONNECTION_OBSERVER_DESTRUCTOR
>-
> #include <memory>
> #include <string>
> #include <utility>
> #include <vector>
> 
>+#include "api/asyncresolverfactory.h"
>+#include "api/audio/audio_mixer.h"
> #include "api/audio_codecs/audio_decoder_factory.h"
> #include "api/audio_codecs/audio_encoder_factory.h"
>+#include "api/audio_options.h"
>+#include "api/call/callfactoryinterface.h"
> #include "api/datachannelinterface.h"
> #include "api/dtmfsenderinterface.h"
>+#include "api/fec_controller.h"
> #include "api/jsep.h"
> #include "api/mediastreaminterface.h"
> #include "api/rtcerror.h"
>@@ -90,34 +91,44 @@
> #include "api/setremotedescriptionobserverinterface.h"
> #include "api/stats/rtcstatscollectorcallback.h"
> #include "api/statstypes.h"
>+#include "api/transport/bitrate_settings.h"
>+#include "api/transport/network_control.h"
> #include "api/turncustomizer.h"
> #include "api/umametrics.h"
>-#include "call/callfactoryinterface.h"
> #include "logging/rtc_event_log/rtc_event_log_factory_interface.h"
>-#include "media/base/mediachannel.h"
>-#include "media/base/videocapturer.h"
>-#include "p2p/base/portallocator.h"
>+#include "media/base/mediaconfig.h"
>+// TODO(bugs.webrtc.org/6353): cricket::VideoCapturer is deprecated and should
>+// be deleted from the PeerConnection api.
>+#include "media/base/videocapturer.h"  // nogncheck
>+// TODO(bugs.webrtc.org/7447): We plan to provide a way to let applications
>+// inject a PacketSocketFactory and/or NetworkManager, and not expose
>+// PortAllocator in the PeerConnection api.
>+#include "media/base/mediaengine.h"  // nogncheck
>+#include "p2p/base/portallocator.h"  // nogncheck
>+// TODO(nisse): The interface for bitrate allocation strategy belongs in api/.
>+#include "rtc_base/bitrateallocationstrategy.h"
> #include "rtc_base/network.h"
>+#include "rtc_base/platform_file.h"
> #include "rtc_base/rtccertificate.h"
> #include "rtc_base/rtccertificategenerator.h"
> #include "rtc_base/socketaddress.h"
>+#include "rtc_base/sslcertificate.h"
> #include "rtc_base/sslstreamadapter.h"
> 
> namespace rtc {
> class SSLIdentity;
> class Thread;
>-}
>+}  // namespace rtc
> 
> namespace cricket {
>-class MediaEngineInterface;
> class WebRtcVideoDecoderFactory;
> class WebRtcVideoEncoderFactory;
>-}
>+}  // namespace cricket
> 
> namespace webrtc {
> class AudioDeviceModule;
> class AudioMixer;
>-class CallFactoryInterface;
>+class AudioProcessing;
> class MediaConstraintsInterface;
> class VideoDecoderFactory;
> class VideoEncoderFactory;
>@@ -129,14 +140,12 @@ class StreamCollectionInterface : public rtc::RefCountInterface {
>   virtual size_t count() = 0;
>   virtual MediaStreamInterface* at(size_t index) = 0;
>   virtual MediaStreamInterface* find(const std::string& label) = 0;
>-  virtual MediaStreamTrackInterface* FindAudioTrack(
>-      const std::string& id) = 0;
>-  virtual MediaStreamTrackInterface* FindVideoTrack(
>-      const std::string& id) = 0;
>+  virtual MediaStreamTrackInterface* FindAudioTrack(const std::string& id) = 0;
>+  virtual MediaStreamTrackInterface* FindVideoTrack(const std::string& id) = 0;
> 
>  protected:
>   // Dtor protected as objects shouldn't be deleted via this interface.
>-  ~StreamCollectionInterface() {}
>+  ~StreamCollectionInterface() override = default;
> };
> 
> class StatsObserver : public rtc::RefCountInterface {
>@@ -144,16 +153,14 @@ class StatsObserver : public rtc::RefCountInterface {
>   virtual void OnComplete(const StatsReports& reports) = 0;
> 
>  protected:
>-  virtual ~StatsObserver() {}
>+  ~StatsObserver() override = default;
> };
> 
>-// For now, kDefault is interpreted as kPlanB.
>-// TODO(bugs.webrtc.org/8530): Switch default to kUnifiedPlan.
>-enum class SdpSemantics { kDefault, kPlanB, kUnifiedPlan };
>+enum class SdpSemantics { kPlanB, kUnifiedPlan };
> 
> class PeerConnectionInterface : public rtc::RefCountInterface {
>  public:
>-  // See http://dev.w3.org/2011/webrtc/editor/webrtc.html#state-definitions .
>+  // See https://w3c.github.io/webrtc-pc/#state-definitions
>   enum SignalingState {
>     kStable,
>     kHaveLocalOffer,
>@@ -192,6 +199,10 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>   };
> 
>   struct IceServer {
>+    IceServer();
>+    IceServer(const IceServer&);
>+    ~IceServer();
>+
>     // TODO(jbauch): Remove uri when all code using it has switched to urls.
>     // List of URIs associated with this server. Valid formats are described
>     // in RFC7064 and RFC7065, and more may be added in the future. The "host"
>@@ -231,14 +242,14 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>     kAll
>   };
> 
>-  // https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-08#section-4.1.1
>+  // https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24#section-4.1.1
>   enum BundlePolicy {
>     kBundlePolicyBalanced,
>     kBundlePolicyMaxBundle,
>     kBundlePolicyMaxCompat
>   };
> 
>-  // https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-09#section-4.1.1
>+  // https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24#section-4.1.1
>   enum RtcpMuxPolicy {
>     kRtcpMuxPolicyNegotiate,
>     kRtcpMuxPolicyRequire,
>@@ -254,10 +265,7 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>     kCandidateNetworkPolicyLowCost
>   };
> 
>-  enum ContinualGatheringPolicy {
>-    GATHER_ONCE,
>-    GATHER_CONTINUALLY
>-  };
>+  enum ContinualGatheringPolicy { GATHER_ONCE, GATHER_CONTINUALLY };
> 
>   enum class RTCConfigurationType {
>     // A configuration that is safer to use, despite not having the best
>@@ -281,53 +289,43 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>     // methods for all settings which are of interest to applications,
>     // Chrome in particular.
> 
>-    RTCConfiguration() = default;
>-    explicit RTCConfiguration(RTCConfigurationType type) {
>-      if (type == RTCConfigurationType::kAggressive) {
>-        // These parameters are also defined in Java and IOS configurations,
>-        // so their values may be overwritten by the Java or IOS configuration.
>-        bundle_policy = kBundlePolicyMaxBundle;
>-        rtcp_mux_policy = kRtcpMuxPolicyRequire;
>-        ice_connection_receiving_timeout =
>-            kAggressiveIceConnectionReceivingTimeout;
>-
>-        // These parameters are not defined in Java or IOS configuration,
>-        // so their values will not be overwritten.
>-        enable_ice_renomination = true;
>-        redetermine_role_on_ice_restart = false;
>-      }
>-    }
>+    RTCConfiguration();
>+    RTCConfiguration(const RTCConfiguration&);
>+    explicit RTCConfiguration(RTCConfigurationType type);
>+    ~RTCConfiguration();
> 
>     bool operator==(const RTCConfiguration& o) const;
>     bool operator!=(const RTCConfiguration& o) const;
> 
>-    bool dscp() { return media_config.enable_dscp; }
>+    bool dscp() const { return media_config.enable_dscp; }
>     void set_dscp(bool enable) { media_config.enable_dscp = enable; }
> 
>-    // TODO(nisse): The corresponding flag in MediaConfig and
>-    // elsewhere should be renamed enable_cpu_adaptation.
>-    bool cpu_adaptation() {
>-      return media_config.video.enable_cpu_overuse_detection;
>+    bool cpu_adaptation() const {
>+      return media_config.video.enable_cpu_adaptation;
>     }
>     void set_cpu_adaptation(bool enable) {
>-      media_config.video.enable_cpu_overuse_detection = enable;
>+      media_config.video.enable_cpu_adaptation = enable;
>     }
> 
>-    bool suspend_below_min_bitrate() {
>+    bool suspend_below_min_bitrate() const {
>       return media_config.video.suspend_below_min_bitrate;
>     }
>     void set_suspend_below_min_bitrate(bool enable) {
>       media_config.video.suspend_below_min_bitrate = enable;
>     }
> 
>-    // TODO(nisse): The negation in the corresponding MediaConfig
>-    // attribute is inconsistent, and it should be renamed at some
>-    // point.
>-    bool prerenderer_smoothing() {
>-      return !media_config.video.disable_prerenderer_smoothing;
>+    bool prerenderer_smoothing() const {
>+      return media_config.video.enable_prerenderer_smoothing;
>     }
>     void set_prerenderer_smoothing(bool enable) {
>-      media_config.video.disable_prerenderer_smoothing = !enable;
>+      media_config.video.enable_prerenderer_smoothing = enable;
>+    }
>+
>+    bool experiment_cpu_load_estimator() const {
>+      return media_config.video.experiment_cpu_load_estimator;
>+    }
>+    void set_experiment_cpu_load_estimator(bool enable) {
>+      media_config.video.experiment_cpu_load_estimator = enable;
>     }
> 
>     static const int kUndefined = -1;
>@@ -338,7 +336,7 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
> 
>     ////////////////////////////////////////////////////////////////////////
>     // The below few fields mirror the standard RTCConfiguration dictionary:
>-    // https://www.w3.org/TR/webrtc/#rtcconfiguration-dictionary
>+    // https://w3c.github.io/webrtc-pc/#rtcconfiguration-dictionary
>     ////////////////////////////////////////////////////////////////////////
> 
>     // TODO(pthatcher): Rename this ice_servers, but update Chromium
>@@ -356,7 +354,7 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>     // The below fields correspond to constraints from the deprecated
>     // constraints interface for constructing a PeerConnection.
>     //
>-    // rtc::Optional fields can be "missing", in which case the implementation
>+    // absl::optional fields can be "missing", in which case the implementation
>     // default will be used.
>     //////////////////////////////////////////////////////////////////////////
> 
>@@ -378,6 +376,10 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>     // Can be set to INT_MAX to effectively disable the limit.
>     int max_ipv6_networks = cricket::kDefaultMaxIPv6Networks;
> 
>+    // Exclude link-local network interfaces
>+    // from considertaion for gathering ICE candidates.
>+    bool disable_link_local_networks = false;
>+
>     // If set to true, use RTP data channels instead of SCTP.
>     // TODO(deadbeef): Remove this. We no longer commit to supporting RTP data
>     // channels, though some applications are still working on moving off of
>@@ -387,15 +389,15 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>     // Minimum bitrate at which screencast video tracks will be encoded at.
>     // This means adding padding bits up to this bitrate, which can help
>     // when switching from a static scene to one with motion.
>-    rtc::Optional<int> screencast_min_bitrate;
>+    absl::optional<int> screencast_min_bitrate;
> 
>     // Use new combined audio/video bandwidth estimation?
>-    rtc::Optional<bool> combined_audio_video_bwe;
>+    absl::optional<bool> combined_audio_video_bwe;
> 
>     // Can be used to disable DTLS-SRTP. This should never be done, but can be
>     // useful for testing purposes, for example in setting up a loopback call
>     // with a single PeerConnection.
>-    rtc::Optional<bool> enable_dtls_srtp;
>+    absl::optional<bool> enable_dtls_srtp;
> 
>     /////////////////////////////////////////////////
>     // The below fields are not part of the standard.
>@@ -444,6 +446,9 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>     // standard priority order.
>     bool prioritize_most_likely_ice_candidate_pairs = false;
> 
>+    // Implementation defined settings. A public member only for the benefit of
>+    // the implementation. Applications must not access it directly, and should
>+    // instead use provided accessor methods, e.g., set_cpu_adaptation.
>     struct cricket::MediaConfig media_config;
> 
>     // If set to true, only one preferred TURN allocation will be used per
>@@ -470,15 +475,51 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>     // re-determining was removed in ICEbis (ICE v2).
>     bool redetermine_role_on_ice_restart = true;
> 
>-    // If set, the min interval (max rate) at which we will send ICE checks
>-    // (STUN pings), in milliseconds.
>-    rtc::Optional<int> ice_check_min_interval;
>+    // The following fields define intervals in milliseconds at which ICE
>+    // connectivity checks are sent.
>+    //
>+    // We consider ICE is "strongly connected" for an agent when there is at
>+    // least one candidate pair that currently succeeds in connectivity check
>+    // from its direction i.e. sending a STUN ping and receives a STUN ping
>+    // response, AND all candidate pairs have sent a minimum number of pings for
>+    // connectivity (this number is implementation-specific). Otherwise, ICE is
>+    // considered in "weak connectivity".
>+    //
>+    // Note that the above notion of strong and weak connectivity is not defined
>+    // in RFC 5245, and they apply to our current ICE implementation only.
>+    //
>+    // 1) ice_check_interval_strong_connectivity defines the interval applied to
>+    // ALL candidate pairs when ICE is strongly connected, and it overrides the
>+    // default value of this interval in the ICE implementation;
>+    // 2) ice_check_interval_weak_connectivity defines the counterpart for ALL
>+    // pairs when ICE is weakly connected, and it overrides the default value of
>+    // this interval in the ICE implementation;
>+    // 3) ice_check_min_interval defines the minimal interval (equivalently the
>+    // maximum rate) that overrides the above two intervals when either of them
>+    // is less.
>+    absl::optional<int> ice_check_interval_strong_connectivity;
>+    absl::optional<int> ice_check_interval_weak_connectivity;
>+    absl::optional<int> ice_check_min_interval;
>+
>+    // The min time period for which a candidate pair must wait for response to
>+    // connectivity checks before it becomes unwritable. This parameter
>+    // overrides the default value in the ICE implementation if set.
>+    absl::optional<int> ice_unwritable_timeout;
>+
>+    // The min number of connectivity checks that a candidate pair must sent
>+    // without receiving response before it becomes unwritable. This parameter
>+    // overrides the default value in the ICE implementation if set.
>+    absl::optional<int> ice_unwritable_min_checks;
>+
>+    // The interval in milliseconds at which STUN candidates will resend STUN
>+    // binding requests to keep NAT bindings open.
>+    absl::optional<int> stun_candidate_keepalive_interval;
> 
>     // ICE Periodic Regathering
>     // If set, WebRTC will periodically create and propose candidates without
>     // starting a new ICE generation. The regathering happens continuously with
>     // interval specified in milliseconds by the uniform distribution [a, b].
>-    rtc::Optional<rtc::IntervalRange> ice_regather_interval_range;
>+    absl::optional<rtc::IntervalRange> ice_regather_interval_range;
> 
>     // Optional TurnCustomizer.
>     // With this class one can modify outgoing TURN messages.
>@@ -486,6 +527,12 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>     // called.
>     webrtc::TurnCustomizer* turn_customizer = nullptr;
> 
>+    // Preferred network interface.
>+    // A candidate pair on a preferred network has a higher precedence in ICE
>+    // than one on an un-preferred network, regardless of priority or network
>+    // cost.
>+    absl::optional<rtc::AdapterType> network_preference;
>+
>     // Configure the SDP semantics used by this PeerConnection. Note that the
>     // WebRTC 1.0 specification requires kUnifiedPlan semantics. The
>     // RtpTransceiver API is only available with kUnifiedPlan semantics.
>@@ -493,26 +540,28 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>     // kPlanB will cause PeerConnection to create offers and answers with at
>     // most one audio and one video m= section with multiple RtpSenders and
>     // RtpReceivers specified as multiple a=ssrc lines within the section. This
>-    // will also cause PeerConnection to reject offers/answers with multiple m=
>-    // sections of the same media type.
>+    // will also cause PeerConnection to ignore all but the first m= section of
>+    // the same media type.
>     //
>     // kUnifiedPlan will cause PeerConnection to create offers and answers with
>     // multiple m= sections where each m= section maps to one RtpSender and one
>-    // RtpReceiver (an RtpTransceiver), either both audio or both video. Plan B
>-    // style offers or answers will be rejected in calls to SetLocalDescription
>-    // or SetRemoteDescription.
>-    //
>-    // For users who only send at most one audio and one video track, this
>-    // choice does not matter and should be left as kDefault.
>+    // RtpReceiver (an RtpTransceiver), either both audio or both video. This
>+    // will also cause PeerConnection to ignore all but the first a=ssrc lines
>+    // that form a Plan B stream.
>     //
>     // For users who wish to send multiple audio/video streams and need to stay
>-    // interoperable with legacy WebRTC implementations, specify kPlanB.
>+    // interoperable with legacy WebRTC implementations or use legacy APIs,
>+    // specify kPlanB.
>     //
>-    // For users who wish to send multiple audio/video streams and/or wish to
>-    // use the new RtpTransceiver API, specify kUnifiedPlan.
>-    //
>-    // TODO(steveanton): Implement support for kUnifiedPlan.
>-    SdpSemantics sdp_semantics = SdpSemantics::kDefault;
>+    // For all other users, specify kUnifiedPlan.
>+    SdpSemantics sdp_semantics = SdpSemantics::kPlanB;
>+
>+    // Actively reset the SRTP parameters whenever the DTLS transports
>+    // underneath are reset for every offer/answer negotiation.
>+    // This is only intended to be a workaround for crbug.com/835958
>+    // WARNING: This would cause RTP/RTCP packets decryption failure if not used
>+    // correctly. This flag will be deprecated soon. Do not rely on it.
>+    bool active_reset_srtp_params = false;
> 
>     //
>     // Don't forget to update operator== if adding something.
>@@ -527,8 +576,9 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>     // The default value for constraint offerToReceiveX:true.
>     static const int kOfferToReceiveMediaTrue = 1;
> 
>-    // These have been removed from the standard in favor of the "transceiver"
>-    // API, but given that we don't support that API, we still have them here.
>+    // These options are left as backwards compatibility for clients who need
>+    // "Plan B" semantics. Clients who have switched to "Unified Plan" semantics
>+    // should use the RtpTransceiver API (AddTransceiver) instead.
>     //
>     // offer_to_receive_X set to 1 will cause a media description to be
>     // generated in the offer, even if no tracks of that type have been added.
>@@ -570,12 +620,14 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>   };
> 
>   // Accessor methods to active local streams.
>-  virtual rtc::scoped_refptr<StreamCollectionInterface>
>-      local_streams() = 0;
>+  // This method is not supported with kUnifiedPlan semantics. Please use
>+  // GetSenders() instead.
>+  virtual rtc::scoped_refptr<StreamCollectionInterface> local_streams() = 0;
> 
>   // Accessor methods to remote streams.
>-  virtual rtc::scoped_refptr<StreamCollectionInterface>
>-      remote_streams() = 0;
>+  // This method is not supported with kUnifiedPlan semantics. Please use
>+  // GetReceivers() instead.
>+  virtual rtc::scoped_refptr<StreamCollectionInterface> remote_streams() = 0;
> 
>   // Add a new MediaStream to be sent on this PeerConnection.
>   // Note that a SessionDescription negotiation is needed before the
>@@ -586,39 +638,48 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>   // stream, with the one difference that if "stream->AddTrack(...)" is called
>   // later, the PeerConnection will automatically pick up the new track. Though
>   // this functionality will be deprecated in the future.
>+  //
>+  // This method is not supported with kUnifiedPlan semantics. Please use
>+  // AddTrack instead.
>   virtual bool AddStream(MediaStreamInterface* stream) = 0;
> 
>   // Remove a MediaStream from this PeerConnection.
>   // Note that a SessionDescription negotiation is needed before the
>   // remote peer is notified.
>+  //
>+  // This method is not supported with kUnifiedPlan semantics. Please use
>+  // RemoveTrack instead.
>   virtual void RemoveStream(MediaStreamInterface* stream) = 0;
> 
>   // Add a new MediaStreamTrack to be sent on this PeerConnection, and return
>   // the newly created RtpSender. The RtpSender will be associated with the
>-  // streams specified in the |stream_labels| list.
>+  // streams specified in the |stream_ids| list.
>   //
>   // Errors:
>   // - INVALID_PARAMETER: |track| is null, has a kind other than audio or video,
>   //       or a sender already exists for the track.
>   // - INVALID_STATE: The PeerConnection is closed.
>-  // TODO(steveanton): Remove default implementation once downstream
>-  // implementations have been updated.
>   virtual RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>> AddTrack(
>-      rtc::scoped_refptr<MediaStreamTrackInterface>,
>-      const std::vector<std::string>&) {
>-    return RTCError(RTCErrorType::UNSUPPORTED_OPERATION, "Not implemented");
>-  }
>-  // |streams| indicates which stream labels the track should be associated
>-  // with.
>-  // TODO(steveanton): Remove this overload once callers have moved to the
>-  // signature with stream labels.
>-  virtual rtc::scoped_refptr<RtpSenderInterface> AddTrack(
>-      MediaStreamTrackInterface* track,
>-      std::vector<MediaStreamInterface*> streams) = 0;
>+      rtc::scoped_refptr<MediaStreamTrackInterface> track,
>+      const std::vector<std::string>& stream_ids);
> 
>   // Remove an RtpSender from this PeerConnection.
>   // Returns true on success.
>-  virtual bool RemoveTrack(RtpSenderInterface* sender) = 0;
>+  // TODO(steveanton): Replace with signature that returns RTCError.
>+  virtual bool RemoveTrack(RtpSenderInterface* sender);
>+
>+  // Plan B semantics: Removes the RtpSender from this PeerConnection.
>+  // Unified Plan semantics: Stop sending on the RtpSender and mark the
>+  // corresponding RtpTransceiver direction as no longer sending.
>+  //
>+  // Errors:
>+  // - INVALID_PARAMETER: |sender| is null or (Plan B only) the sender is not
>+  //       associated with this PeerConnection.
>+  // - INVALID_STATE: PeerConnection is closed.
>+  // TODO(bugs.webrtc.org/9534): Rename to RemoveTrack once the other signature
>+  // is removed.
>+  virtual RTCError RemoveTrackNew(
>+      rtc::scoped_refptr<RtpSenderInterface> sender);
> 
>   // AddTransceiver creates a new RtpTransceiver and adds it to the set of
>   // transceivers. Adding a transceiver will cause future calls to CreateOffer
>@@ -647,14 +708,10 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>   // Errors:
>   // - INVALID_PARAMETER: |track| is null.
>   virtual RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
>-  AddTransceiver(rtc::scoped_refptr<MediaStreamTrackInterface>) {
>-    return RTCError(RTCErrorType::INTERNAL_ERROR, "not implemented");
>-  }
>+  AddTransceiver(rtc::scoped_refptr<MediaStreamTrackInterface> track);
>   virtual RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
>-  AddTransceiver(rtc::scoped_refptr<MediaStreamTrackInterface>,
>-                 const RtpTransceiverInit&) {
>-    return RTCError(RTCErrorType::INTERNAL_ERROR, "not implemented");
>-  }
>+  AddTransceiver(rtc::scoped_refptr<MediaStreamTrackInterface> track,
>+                 const RtpTransceiverInit& init);
> 
>   // Adds a transceiver with the given kind. Can either be MEDIA_TYPE_AUDIO or
>   // MEDIA_TYPE_VIDEO.
>@@ -662,22 +719,9 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>   // - INVALID_PARAMETER: |media_type| is not MEDIA_TYPE_AUDIO or
>   //                      MEDIA_TYPE_VIDEO.
>   virtual RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
>-  AddTransceiver(cricket::MediaType) {
>-    return RTCError(RTCErrorType::INTERNAL_ERROR, "not implemented");
>-  }
>+  AddTransceiver(cricket::MediaType media_type);
>   virtual RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
>-  AddTransceiver(cricket::MediaType,
>-                 const RtpTransceiverInit&) {
>-    return RTCError(RTCErrorType::INTERNAL_ERROR, "not implemented");
>-  }
>-
>-  // Returns pointer to a DtmfSender on success. Otherwise returns null.
>-  //
>-  // This API is no longer part of the standard; instead DtmfSenders are
>-  // obtained from RtpSenders. Which is what the implementation does; it finds
>-  // an RtpSender for |track| and just returns its DtmfSender.
>-  virtual rtc::scoped_refptr<DtmfSenderInterface> CreateDtmfSender(
>-      AudioTrackInterface* track) = 0;
>+  AddTransceiver(cricket::MediaType media_type, const RtpTransceiverInit& init);
> 
>   // TODO(deadbeef): Make these pure virtual once all subclasses implement them.
> 
>@@ -692,53 +736,82 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>   //
>   // |stream_id| is used to populate the msid attribute; if empty, one will
>   // be generated automatically.
>+  //
>+  // This method is not supported with kUnifiedPlan semantics. Please use
>+  // AddTransceiver instead.
>   virtual rtc::scoped_refptr<RtpSenderInterface> CreateSender(
>-      const std::string& /* kind */,
>-      const std::string& /* stream_id */) {
>-    return rtc::scoped_refptr<RtpSenderInterface>();
>-  }
>+      const std::string& kind,
>+      const std::string& stream_id);
> 
>-  // Get all RtpSenders, created either through AddStream, AddTrack, or
>-  // CreateSender. Note that these are "Plan B SDP" RtpSenders, not "Unified
>-  // Plan SDP" RtpSenders, which means that all senders of a specific media
>-  // type share the same media description.
>+  // If Plan B semantics are specified, gets all RtpSenders, created either
>+  // through AddStream, AddTrack, or CreateSender. All senders of a specific
>+  // media type share the same media description.
>+  //
>+  // If Unified Plan semantics are specified, gets the RtpSender for each
>+  // RtpTransceiver.
>   virtual std::vector<rtc::scoped_refptr<RtpSenderInterface>> GetSenders()
>-      const {
>-    return std::vector<rtc::scoped_refptr<RtpSenderInterface>>();
>-  }
>+      const;
> 
>-  // Get all RtpReceivers, created when a remote description is applied.
>-  // Note that these are "Plan B SDP" RtpReceivers, not "Unified Plan SDP"
>-  // RtpReceivers, which means that all receivers of a specific media type
>-  // share the same media description.
>+  // If Plan B semantics are specified, gets all RtpReceivers created when a
>+  // remote description is applied. All receivers of a specific media type share
>+  // the same media description. It is also possible to have a media description
>+  // with no associated RtpReceivers, if the directional attribute does not
>+  // indicate that the remote peer is sending any media.
>   //
>-  // It is also possible to have a media description with no associated
>-  // RtpReceivers, if the directional attribute does not indicate that the
>-  // remote peer is sending any media.
>+  // If Unified Plan semantics are specified, gets the RtpReceiver for each
>+  // RtpTransceiver.
>   virtual std::vector<rtc::scoped_refptr<RtpReceiverInterface>> GetReceivers()
>-      const {
>-    return std::vector<rtc::scoped_refptr<RtpReceiverInterface>>();
>-  }
>+      const;
> 
>   // Get all RtpTransceivers, created either through AddTransceiver, AddTrack or
>   // by a remote description applied with SetRemoteDescription.
>+  //
>   // Note: This method is only available when Unified Plan is enabled (see
>   // RTCConfiguration).
>   virtual std::vector<rtc::scoped_refptr<RtpTransceiverInterface>>
>-  GetTransceivers() const {
>-    return {};
>-  }
>+  GetTransceivers() const;
> 
>+  // The legacy non-compliant GetStats() API. This correspond to the
>+  // callback-based version of getStats() in JavaScript. The returned metrics
>+  // are UNDOCUMENTED and many of them rely on implementation-specific details.
>+  // The goal is to DELETE THIS VERSION but we can't today because it is heavily
>+  // relied upon by third parties. See https://crbug.com/822696.
>+  //
>+  // This version is wired up into Chrome. Any stats implemented are
>+  // automatically exposed to the Web Platform. This has BYPASSED the Chrome
>+  // release processes for years and lead to cross-browser incompatibility
>+  // issues and web application reliance on Chrome-only behavior.
>+  //
>+  // This API is in "maintenance mode", serious regressions should be fixed but
>+  // adding new stats is highly discouraged.
>+  //
>+  // TODO(hbos): Deprecate and remove this when third parties have migrated to
>+  // the spec-compliant GetStats() API. https://crbug.com/822696
>   virtual bool GetStats(StatsObserver* observer,
>-                        MediaStreamTrackInterface* track,
>+                        MediaStreamTrackInterface* track,  // Optional
>                         StatsOutputLevel level) = 0;
>-  // Gets stats using the new stats collection API, see webrtc/api/stats/. These
>-  // will replace old stats collection API when the new API has matured enough.
>-  // TODO(hbos): Default implementation that does nothing only exists as to not
>-  // break third party projects. As soon as they have been updated this should
>-  // be changed to "= 0;".
>-  virtual void GetStats(RTCStatsCollectorCallback*) {}
>-  // Clear cached stats in the rtcstatscollector.
>+  // The spec-compliant GetStats() API. This correspond to the promise-based
>+  // version of getStats() in JavaScript. Implementation status is described in
>+  // api/stats/rtcstats_objects.h. For more details on stats, see spec:
>+  // https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnection-getstats
>+  // TODO(hbos): Takes shared ownership, use rtc::scoped_refptr<> instead. This
>+  // requires stop overriding the current version in third party or making third
>+  // party calls explicit to avoid ambiguity during switch. Make the future
>+  // version abstract as soon as third party projects implement it.
>+  virtual void GetStats(RTCStatsCollectorCallback* callback) {}
>+  // Spec-compliant getStats() performing the stats selection algorithm with the
>+  // sender. https://w3c.github.io/webrtc-pc/#dom-rtcrtpsender-getstats
>+  // TODO(hbos): Make abstract as soon as third party projects implement it.
>+  virtual void GetStats(
>+      rtc::scoped_refptr<RtpSenderInterface> selector,
>+      rtc::scoped_refptr<RTCStatsCollectorCallback> callback) {}
>+  // Spec-compliant getStats() performing the stats selection algorithm with the
>+  // receiver. https://w3c.github.io/webrtc-pc/#dom-rtcrtpreceiver-getstats
>+  // TODO(hbos): Make abstract as soon as third party projects implement it.
>+  virtual void GetStats(
>+      rtc::scoped_refptr<RtpReceiverInterface> selector,
>+      rtc::scoped_refptr<RTCStatsCollectorCallback> callback) {}
>+  // Clear cached stats in the RTCStatsCollector.
>   // Exposed for testing while waiting for automatic cache clear to work.
>   // https://bugs.webrtc.org/8693
>   virtual void ClearStatsCache() {}
>@@ -761,44 +834,34 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
> 
>   // A "current" description the one currently negotiated from a complete
>   // offer/answer exchange.
>-  virtual const SessionDescriptionInterface* current_local_description() const {
>-    return nullptr;
>-  }
>-  virtual const SessionDescriptionInterface* current_remote_description()
>-      const {
>-    return nullptr;
>-  }
>+  virtual const SessionDescriptionInterface* current_local_description() const;
>+  virtual const SessionDescriptionInterface* current_remote_description() const;
> 
>   // A "pending" description is one that's part of an incomplete offer/answer
>   // exchange (thus, either an offer or a pranswer). Once the offer/answer
>   // exchange is finished, the "pending" description will become "current".
>-  virtual const SessionDescriptionInterface* pending_local_description() const {
>-    return nullptr;
>-  }
>-  virtual const SessionDescriptionInterface* pending_remote_description()
>-      const {
>-    return nullptr;
>-  }
>+  virtual const SessionDescriptionInterface* pending_local_description() const;
>+  virtual const SessionDescriptionInterface* pending_remote_description() const;
> 
>   // Create a new offer.
>   // The CreateSessionDescriptionObserver callback will be called when done.
>-  virtual void CreateOffer(CreateSessionDescriptionObserver*,
>-                           const MediaConstraintsInterface*) {}
>+  virtual void CreateOffer(CreateSessionDescriptionObserver* observer,
>+                           const MediaConstraintsInterface* constraints) {}
> 
>   // TODO(jiayl): remove the default impl and the old interface when chromium
>   // code is updated.
>-  virtual void CreateOffer(CreateSessionDescriptionObserver*,
>-                           const RTCOfferAnswerOptions&) {}
>+  virtual void CreateOffer(CreateSessionDescriptionObserver* observer,
>+                           const RTCOfferAnswerOptions& options) {}
> 
>   // Create an answer to an offer.
>   // The CreateSessionDescriptionObserver callback will be called when done.
>-  virtual void CreateAnswer(CreateSessionDescriptionObserver*,
>-                            const RTCOfferAnswerOptions&) {}
>+  virtual void CreateAnswer(CreateSessionDescriptionObserver* observer,
>+                            const RTCOfferAnswerOptions& options) {}
>   // Deprecated - use version above.
>   // TODO(hta): Remove and remove default implementations when all callers
>   // are updated.
>-  virtual void CreateAnswer(CreateSessionDescriptionObserver*,
>-                            const MediaConstraintsInterface*) {}
>+  virtual void CreateAnswer(CreateSessionDescriptionObserver* observer,
>+                            const MediaConstraintsInterface* constraints) {}
> 
>   // Sets the local session description.
>   // The PeerConnection takes the ownership of |desc| even if it fails.
>@@ -811,25 +874,16 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>   // The PeerConnection takes the ownership of |desc| even if it fails.
>   // The |observer| callback will be called when done.
>   // TODO(hbos): Remove when Chrome implements the new signature.
>-  virtual void SetRemoteDescription(SetSessionDescriptionObserver*,
>-                                    SessionDescriptionInterface*) {}
>+  virtual void SetRemoteDescription(SetSessionDescriptionObserver* observer,
>+                                    SessionDescriptionInterface* desc) {}
>   // TODO(hbos): Make pure virtual when Chrome has updated its signature.
>   virtual void SetRemoteDescription(
>-      std::unique_ptr<SessionDescriptionInterface>,
>-      rtc::scoped_refptr<SetRemoteDescriptionObserverInterface>) {}
>-  // Deprecated; Replaced by SetConfiguration.
>-  // TODO(deadbeef): Remove once Chrome is moved over to SetConfiguration.
>-  virtual bool UpdateIce(const IceServers&,
>-                         const MediaConstraintsInterface*) {
>-    return false;
>-  }
>-  virtual bool UpdateIce(const IceServers&) { return false; }
>+      std::unique_ptr<SessionDescriptionInterface> desc,
>+      rtc::scoped_refptr<SetRemoteDescriptionObserverInterface> observer) {}
> 
>   // TODO(deadbeef): Make this pure virtual once all Chrome subclasses of
>   // PeerConnectionInterface implement it.
>-  virtual PeerConnectionInterface::RTCConfiguration GetConfiguration() {
>-    return PeerConnectionInterface::RTCConfiguration();
>-  }
>+  virtual PeerConnectionInterface::RTCConfiguration GetConfiguration();
> 
>   // Sets the PeerConnection's global configuration to |config|.
>   //
>@@ -855,16 +909,13 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>   // TODO(deadbeef): Make this pure virtual once all Chrome subclasses of
>   // PeerConnectionInterface implement it.
>   virtual bool SetConfiguration(
>-      const PeerConnectionInterface::RTCConfiguration&,
>-      RTCError*) {
>-    return false;
>-  }
>+      const PeerConnectionInterface::RTCConfiguration& config,
>+      RTCError* error);
>+
>   // Version without error output param for backwards compatibility.
>   // TODO(deadbeef): Remove once chromium is updated.
>   virtual bool SetConfiguration(
>-      const PeerConnectionInterface::RTCConfiguration&) {
>-    return false;
>-  }
>+      const PeerConnectionInterface::RTCConfiguration& config);
> 
>   // Provides a remote candidate to the ICE Agent.
>   // A copy of the |candidate| will be created and added to the remote
>@@ -876,21 +927,16 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>   // continual gathering, to avoid an ever-growing list of candidates as
>   // networks come and go.
>   virtual bool RemoveIceCandidates(
>-      const std::vector<cricket::Candidate>&) {
>-    return false;
>-  }
>-
>-  // Register a metric observer (used by chromium). It's reference counted, and
>-  // this method takes a reference. RegisterUMAObserver(nullptr) will release
>-  // the reference.
>-  // TODO(deadbeef): Take argument as scoped_refptr?
>-  virtual void RegisterUMAObserver(UMAObserver* observer) = 0;
>+      const std::vector<cricket::Candidate>& candidates);
> 
>   // 0 <= min <= current <= max should hold for set parameters.
>   struct BitrateParameters {
>-    rtc::Optional<int> min_bitrate_bps;
>-    rtc::Optional<int> current_bitrate_bps;
>-    rtc::Optional<int> max_bitrate_bps;
>+    BitrateParameters();
>+    ~BitrateParameters();
>+
>+    absl::optional<int> min_bitrate_bps;
>+    absl::optional<int> current_bitrate_bps;
>+    absl::optional<int> max_bitrate_bps;
>   };
> 
>   // SetBitrate limits the bandwidth allocated for all RTP streams sent by
>@@ -899,14 +945,20 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>   //
>   // Setting |current_bitrate_bps| will reset the current bitrate estimate
>   // to the provided value.
>-  virtual RTCError SetBitrate(const BitrateParameters& bitrate) = 0;
>+  virtual RTCError SetBitrate(const BitrateSettings& bitrate);
>+
>+  // TODO(nisse): Deprecated - use version above. These two default
>+  // implementations require subclasses to implement one or the other
>+  // of the methods.
>+  virtual RTCError SetBitrate(const BitrateParameters& bitrate_parameters);
> 
>   // Sets current strategy. If not set default WebRTC allocator will be used.
>   // May be changed during an active session. The strategy
>   // ownership is passed with std::unique_ptr
>   // TODO(alexnarest): Make this pure virtual when tests will be updated
>   virtual void SetBitrateAllocationStrategy(
>-      std::unique_ptr<rtc::BitrateAllocationStrategy>) {}
>+      std::unique_ptr<rtc::BitrateAllocationStrategy>
>+          bitrate_allocation_strategy) {}
> 
>   // Enable/disable playout of received audio streams. Enabled by default. Note
>   // that even if playout is enabled, streams will only be played out if the
>@@ -915,13 +967,13 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>   // for audio data every 10ms to ensure that audio processing happens and the
>   // audio statistics are updated.
>   // TODO(henrika): deprecate and remove this.
>-  virtual void SetAudioPlayout(bool /* playout */) {}
>+  virtual void SetAudioPlayout(bool playout) {}
> 
>   // Enable/disable recording of transmitted audio streams. Enabled by default.
>   // Note that even if recording is enabled, streams will only be recorded if
>   // the appropriate SDP is also applied.
>   // TODO(henrika): deprecate and remove this.
>-  virtual void SetAudioRecording(bool /* recording */) {}
>+  virtual void SetAudioRecording(bool recording) {}
> 
>   // Returns the current SignalingState.
>   virtual SignalingState signaling_state() = 0;
>@@ -940,19 +992,14 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>   // automatically after 10 minutes have passed, or when the StopRtcEventLog
>   // function is called.
>   // TODO(eladalon): Deprecate and remove this.
>-  virtual bool StartRtcEventLog(rtc::PlatformFile,
>-                                int64_t /* max_size_bytes */) {
>-    return false;
>-  }
>+  virtual bool StartRtcEventLog(rtc::PlatformFile file, int64_t max_size_bytes);
> 
>   // Start RtcEventLog using an existing output-sink. Takes ownership of
>   // |output| and passes it on to Call, which will take the ownership. If the
>   // operation fails the output will be closed and deallocated. The event log
>   // will send serialized events to the output object every |output_period_ms|.
>-  virtual bool StartRtcEventLog(std::unique_ptr<RtcEventLogOutput>,
>-                                int64_t /* output_period_ms */) {
>-    return false;
>-  }
>+  virtual bool StartRtcEventLog(std::unique_ptr<RtcEventLogOutput> output,
>+                                int64_t output_period_ms);
> 
>   // Stops logging the RtcEventLog.
>   // TODO(ivoc): Make this pure virtual when Chrome is updated.
>@@ -968,34 +1015,25 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
> 
>  protected:
>   // Dtor protected as objects shouldn't be deleted via this interface.
>-  ~PeerConnectionInterface() {}
>+  ~PeerConnectionInterface() override = default;
> };
> 
> // PeerConnection callback interface, used for RTCPeerConnection events.
> // Application should implement these methods.
> class PeerConnectionObserver {
>  public:
>-  enum StateType {
>-    kSignalingState,
>-    kIceState,
>-  };
>-
>   virtual ~PeerConnectionObserver() = default;
> 
>   // Triggered when the SignalingState changed.
>   virtual void OnSignalingChange(
>       PeerConnectionInterface::SignalingState new_state) = 0;
> 
>-  // TODO(deadbeef): Once all subclasses override the scoped_refptr versions
>-  // of the below three methods, make them pure virtual and remove the raw
>-  // pointer version.
>-
>   // Triggered when media is received on a new stream from remote peer.
>-  virtual void OnAddStream(rtc::scoped_refptr<MediaStreamInterface> stream) = 0;
>+  virtual void OnAddStream(rtc::scoped_refptr<MediaStreamInterface> stream) {}
> 
>-  // Triggered when a remote peer close a stream.
>-  virtual void OnRemoveStream(
>-      rtc::scoped_refptr<MediaStreamInterface> stream) = 0;
>+  // Triggered when a remote peer closes a stream.
>+  virtual void OnRemoveStream(rtc::scoped_refptr<MediaStreamInterface> stream) {
>+  }
> 
>   // Triggered when a remote peer opens a data channel.
>   virtual void OnDataChannel(
>@@ -1025,35 +1063,106 @@ class PeerConnectionObserver {
>   // TODO(honghaiz): Make this a pure virtual method when all its subclasses
>   // implement it.
>   virtual void OnIceCandidatesRemoved(
>-      const std::vector<cricket::Candidate>&) {}
>+      const std::vector<cricket::Candidate>& candidates) {}
> 
>   // Called when the ICE connection receiving status changes.
>-  virtual void OnIceConnectionReceivingChange(bool) {}
>+  virtual void OnIceConnectionReceivingChange(bool receiving) {}
> 
>-  // This is called when a receiver and its track is created.
>+  // This is called when a receiver and its track are created.
>   // TODO(zhihuang): Make this pure virtual when all subclasses implement it.
>+  // Note: This is called with both Plan B and Unified Plan semantics. Unified
>+  // Plan users should prefer OnTrack, OnAddTrack is only called as backwards
>+  // compatibility (and is called in the exact same situations as OnTrack).
>   virtual void OnAddTrack(
>-      rtc::scoped_refptr<RtpReceiverInterface>,
>-      const std::vector<rtc::scoped_refptr<MediaStreamInterface>>&) {}
>-
>-  // TODO(hbos,deadbeef): Add |OnAssociatedStreamsUpdated| with |receiver| and
>-  // |streams| as arguments. This should be called when an existing receiver its
>-  // associated streams updated. https://crbug.com/webrtc/8315
>-  // This may be blocked on supporting multiple streams per sender or else
>-  // this may count as the removal and addition of a track?
>-  // https://crbug.com/webrtc/7932
>-
>-  // Called when a receiver is completely removed. This is current (Plan B SDP)
>-  // behavior that occurs when processing the removal of a remote track, and is
>-  // called when the receiver is removed and the track is muted. When Unified
>-  // Plan SDP is supported, transceivers can change direction (and receivers
>-  // stopped) but receivers are never removed.
>+      rtc::scoped_refptr<RtpReceiverInterface> receiver,
>+      const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams) {}
>+
>+  // This is called when signaling indicates a transceiver will be receiving
>+  // media from the remote endpoint. This is fired during a call to
>+  // SetRemoteDescription. The receiving track can be accessed by:
>+  // |transceiver->receiver()->track()| and its associated streams by
>+  // |transceiver->receiver()->streams()|.
>+  // Note: This will only be called if Unified Plan semantics are specified.
>+  // This behavior is specified in section 2.2.8.2.5 of the "Set the
>+  // RTCSessionDescription" algorithm:
>+  // https://w3c.github.io/webrtc-pc/#set-description
>+  virtual void OnTrack(
>+      rtc::scoped_refptr<RtpTransceiverInterface> transceiver) {}
>+
>+  // Called when signaling indicates that media will no longer be received on a
>+  // track.
>+  // With Plan B semantics, the given receiver will have been removed from the
>+  // PeerConnection and the track muted.
>+  // With Unified Plan semantics, the receiver will remain but the transceiver
>+  // will have changed direction to either sendonly or inactive.
>   // https://w3c.github.io/webrtc-pc/#process-remote-track-removal
>-  // TODO(hbos,deadbeef): When Unified Plan SDP is supported and receivers are
>-  // no longer removed, deprecate and remove this callback.
>   // TODO(hbos,deadbeef): Make pure virtual when all subclasses implement it.
>   virtual void OnRemoveTrack(
>-      rtc::scoped_refptr<RtpReceiverInterface>) {}
>+      rtc::scoped_refptr<RtpReceiverInterface> receiver) {}
>+
>+  // Called when an interesting usage is detected by WebRTC.
>+  // An appropriate action is to add information about the context of the
>+  // PeerConnection and write the event to some kind of "interesting events"
>+  // log function.
>+  // The heuristics for defining what constitutes "interesting" are
>+  // implementation-defined.
>+  virtual void OnInterestingUsage(int usage_pattern) {}
>+};
>+
>+// PeerConnectionDependencies holds all of PeerConnections dependencies.
>+// A dependency is distinct from a configuration as it defines significant
>+// executable code that can be provided by a user of the API.
>+//
>+// All new dependencies should be added as a unique_ptr to allow the
>+// PeerConnection object to be the definitive owner of the dependencies
>+// lifetime making injection safer.
>+struct PeerConnectionDependencies final {
>+  explicit PeerConnectionDependencies(PeerConnectionObserver* observer_in);
>+  // This object is not copyable or assignable.
>+  PeerConnectionDependencies(const PeerConnectionDependencies&) = delete;
>+  PeerConnectionDependencies& operator=(const PeerConnectionDependencies&) =
>+      delete;
>+  // This object is only moveable.
>+  PeerConnectionDependencies(PeerConnectionDependencies&&);
>+  PeerConnectionDependencies& operator=(PeerConnectionDependencies&&) = default;
>+  ~PeerConnectionDependencies();
>+  // Mandatory dependencies
>+  PeerConnectionObserver* observer = nullptr;
>+  // Optional dependencies
>+  std::unique_ptr<cricket::PortAllocator> allocator;
>+  std::unique_ptr<webrtc::AsyncResolverFactory> async_resolver_factory;
>+  std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator;
>+  std::unique_ptr<rtc::SSLCertificateVerifier> tls_cert_verifier;
>+};
>+
>+// PeerConnectionFactoryDependencies holds all of the PeerConnectionFactory
>+// dependencies. All new dependencies should be added here instead of
>+// overloading the function. This simplifies dependency injection and makes it
>+// clear which are mandatory and optional. If possible please allow the peer
>+// connection factory to take ownership of the dependency by adding a unique_ptr
>+// to this structure.
>+struct PeerConnectionFactoryDependencies final {
>+  PeerConnectionFactoryDependencies();
>+  // This object is not copyable or assignable.
>+  PeerConnectionFactoryDependencies(const PeerConnectionFactoryDependencies&) =
>+      delete;
>+  PeerConnectionFactoryDependencies& operator=(
>+      const PeerConnectionFactoryDependencies&) = delete;
>+  // This object is only moveable.
>+  PeerConnectionFactoryDependencies(PeerConnectionFactoryDependencies&&);
>+  PeerConnectionFactoryDependencies& operator=(
>+      PeerConnectionFactoryDependencies&&) = default;
>+  ~PeerConnectionFactoryDependencies();
>+
>+  // Optional dependencies
>+  rtc::Thread* network_thread = nullptr;
>+  rtc::Thread* worker_thread = nullptr;
>+  rtc::Thread* signaling_thread = nullptr;
>+  std::unique_ptr<cricket::MediaEngineInterface> media_engine;
>+  std::unique_ptr<CallFactoryInterface> call_factory;
>+  std::unique_ptr<RtcEventLogFactoryInterface> event_log_factory;
>+  std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory;
>+  std::unique_ptr<NetworkControllerFactoryInterface> network_controller_factory;
> };
> 
> // PeerConnectionFactoryInterface is the factory interface used for creating
>@@ -1108,8 +1217,16 @@ class PeerConnectionFactoryInterface : public rtc::RefCountInterface {
>   // Set the options to be used for subsequently created PeerConnections.
>   virtual void SetOptions(const Options& options) = 0;
> 
>-  // |allocator| and |cert_generator| may be null, in which case default
>-  // implementations will be used.
>+  // The preferred way to create a new peer connection. Simply provide the
>+  // configuration and a PeerConnectionDependencies structure.
>+  // TODO(benwright): Make pure virtual once downstream mock PC factory classes
>+  // are updated.
>+  virtual rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection(
>+      const PeerConnectionInterface::RTCConfiguration& configuration,
>+      PeerConnectionDependencies dependencies);
>+
>+  // Deprecated; |allocator| and |cert_generator| may be null, in which case
>+  // default implementations will be used.
>   //
>   // |observer| must not be null.
>   //
>@@ -1121,7 +1238,7 @@ class PeerConnectionFactoryInterface : public rtc::RefCountInterface {
>       const PeerConnectionInterface::RTCConfiguration& configuration,
>       std::unique_ptr<cricket::PortAllocator> allocator,
>       std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
>-      PeerConnectionObserver* observer) = 0;
>+      PeerConnectionObserver* observer);
> 
>   // Deprecated; should use RTCConfiguration for everything that previously
>   // used constraints.
>@@ -1130,19 +1247,27 @@ class PeerConnectionFactoryInterface : public rtc::RefCountInterface {
>       const MediaConstraintsInterface* constraints,
>       std::unique_ptr<cricket::PortAllocator> allocator,
>       std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
>-      PeerConnectionObserver* observer) = 0;
>+      PeerConnectionObserver* observer);
>+
>+  // Returns the capabilities of an RTP sender of type |kind|.
>+  // If for some reason you pass in MEDIA_TYPE_DATA, returns an empty structure.
>+  // TODO(orphis): Make pure virtual when all subclasses implement it.
>+  virtual RtpCapabilities GetRtpSenderCapabilities(
>+      cricket::MediaType kind) const;
>+
>+  // Returns the capabilities of an RTP receiver of type |kind|.
>+  // If for some reason you pass in MEDIA_TYPE_DATA, returns an empty structure.
>+  // TODO(orphis): Make pure virtual when all subclasses implement it.
>+  virtual RtpCapabilities GetRtpReceiverCapabilities(
>+      cricket::MediaType kind) const;
> 
>-  virtual rtc::scoped_refptr<MediaStreamInterface>
>-      CreateLocalMediaStream(const std::string& label) = 0;
>+  virtual rtc::scoped_refptr<MediaStreamInterface> CreateLocalMediaStream(
>+      const std::string& stream_id) = 0;
> 
>   // Creates an AudioSourceInterface.
>   // |options| decides audio processing settings.
>   virtual rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource(
>       const cricket::AudioOptions& options) = 0;
>-  // Deprecated - use version above.
>-  // Can use CopyConstraintsIntoAudioOptions to bridge the gap.
>-  virtual rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource(
>-      const MediaConstraintsInterface* constraints) = 0;
> 
>   // Creates a VideoTrackSourceInterface from |capturer|.
>   // TODO(deadbeef): We should aim to remove cricket::VideoCapturer from the
>@@ -1152,9 +1277,7 @@ class PeerConnectionFactoryInterface : public rtc::RefCountInterface {
>   // TODO(deadbeef): Make pure virtual once downstream mock PC factory classes
>   // are updated.
>   virtual rtc::scoped_refptr<VideoTrackSourceInterface> CreateVideoSource(
>-      std::unique_ptr<cricket::VideoCapturer>) {
>-    return nullptr;
>-  }
>+      std::unique_ptr<cricket::VideoCapturer> capturer);
> 
>   // A video source creator that allows selection of resolution and frame rate.
>   // |constraints| decides video resolution and frame rate but can be null.
>@@ -1163,23 +1286,16 @@ class PeerConnectionFactoryInterface : public rtc::RefCountInterface {
>   // |constraints| is only used for the invocation of this method, and can
>   // safely be destroyed afterwards.
>   virtual rtc::scoped_refptr<VideoTrackSourceInterface> CreateVideoSource(
>-      std::unique_ptr<cricket::VideoCapturer>,
>-      const MediaConstraintsInterface*) {
>-    return nullptr;
>-  }
>+      std::unique_ptr<cricket::VideoCapturer> capturer,
>+      const MediaConstraintsInterface* constraints);
> 
>   // Deprecated; please use the versions that take unique_ptrs above.
>   // TODO(deadbeef): Remove these once safe to do so.
>   virtual rtc::scoped_refptr<VideoTrackSourceInterface> CreateVideoSource(
>-      cricket::VideoCapturer* capturer) {
>-    return CreateVideoSource(std::unique_ptr<cricket::VideoCapturer>(capturer));
>-  }
>+      cricket::VideoCapturer* capturer);
>   virtual rtc::scoped_refptr<VideoTrackSourceInterface> CreateVideoSource(
>       cricket::VideoCapturer* capturer,
>-      const MediaConstraintsInterface* constraints) {
>-    return CreateVideoSource(std::unique_ptr<cricket::VideoCapturer>(capturer),
>-                             constraints);
>-  }
>+      const MediaConstraintsInterface* constraints);
> 
>   // Creates a new local VideoTrack. The same |source| can be used in several
>   // tracks.
>@@ -1188,9 +1304,9 @@ class PeerConnectionFactoryInterface : public rtc::RefCountInterface {
>       VideoTrackSourceInterface* source) = 0;
> 
>   // Creates an new AudioTrack. At the moment |source| can be null.
>-  virtual rtc::scoped_refptr<AudioTrackInterface>
>-      CreateAudioTrack(const std::string& label,
>-                       AudioSourceInterface* source) = 0;
>+  virtual rtc::scoped_refptr<AudioTrackInterface> CreateAudioTrack(
>+      const std::string& label,
>+      AudioSourceInterface* source) = 0;
> 
>   // Starts AEC dump using existing file. Takes ownership of |file| and passes
>   // it on to VoiceEngine (via other objects) immediately, which will take
>@@ -1208,7 +1324,7 @@ class PeerConnectionFactoryInterface : public rtc::RefCountInterface {
>   // Dtor and ctor protected as objects shouldn't be created or deleted via
>   // this interface.
>   PeerConnectionFactoryInterface() {}
>-  ~PeerConnectionFactoryInterface() {} // NOLINT
>+  ~PeerConnectionFactoryInterface() override = default;
> };
> 
> // Create a new instance of PeerConnectionFactoryInterface.
>@@ -1263,9 +1379,36 @@ rtc::scoped_refptr<PeerConnectionFactoryInterface> CreatePeerConnectionFactory(
>     rtc::scoped_refptr<AudioMixer> audio_mixer,
>     rtc::scoped_refptr<AudioProcessing> audio_processing);
> 
>+// Create a new instance of PeerConnectionFactoryInterface with optional
>+// external audio mixer, audio processing, and fec controller modules.
>+//
>+// If |audio_mixer| is null, an internal audio mixer will be created and used.
>+// If |audio_processing| is null, an internal audio processing module will be
>+// created and used.
>+// If |fec_controller_factory| is null, an internal fec controller module will
>+// be created and used.
>+// If |network_controller_factory| is provided, it will be used if enabled via
>+// field trial.
>+rtc::scoped_refptr<PeerConnectionFactoryInterface> CreatePeerConnectionFactory(
>+    rtc::Thread* network_thread,
>+    rtc::Thread* worker_thread,
>+    rtc::Thread* signaling_thread,
>+    AudioDeviceModule* default_adm,
>+    rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
>+    rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory,
>+    cricket::WebRtcVideoEncoderFactory* video_encoder_factory,
>+    cricket::WebRtcVideoDecoderFactory* video_decoder_factory,
>+    rtc::scoped_refptr<AudioMixer> audio_mixer,
>+    rtc::scoped_refptr<AudioProcessing> audio_processing,
>+    std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory,
>+    std::unique_ptr<NetworkControllerFactoryInterface>
>+        network_controller_factory = nullptr);
>+
> // Create a new instance of PeerConnectionFactoryInterface with optional video
> // codec factories. These video factories represents all video codecs, i.e. no
> // extra internal video codecs will be added.
>+// When building WebRTC with rtc_use_builtin_sw_codecs = false, this is the
>+// only available CreatePeerConnectionFactory overload.
> rtc::scoped_refptr<PeerConnectionFactoryInterface> CreatePeerConnectionFactory(
>     rtc::Thread* network_thread,
>     rtc::Thread* worker_thread,
>@@ -1352,6 +1495,22 @@ CreateModularPeerConnectionFactory(
>     std::unique_ptr<CallFactoryInterface> call_factory,
>     std::unique_ptr<RtcEventLogFactoryInterface> event_log_factory);
> 
>+rtc::scoped_refptr<PeerConnectionFactoryInterface>
>+CreateModularPeerConnectionFactory(
>+    rtc::Thread* network_thread,
>+    rtc::Thread* worker_thread,
>+    rtc::Thread* signaling_thread,
>+    std::unique_ptr<cricket::MediaEngineInterface> media_engine,
>+    std::unique_ptr<CallFactoryInterface> call_factory,
>+    std::unique_ptr<RtcEventLogFactoryInterface> event_log_factory,
>+    std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory,
>+    std::unique_ptr<NetworkControllerFactoryInterface>
>+        network_controller_factory = nullptr);
>+
>+rtc::scoped_refptr<PeerConnectionFactoryInterface>
>+CreateModularPeerConnectionFactory(
>+    PeerConnectionFactoryDependencies dependencies);
>+
> }  // namespace webrtc
> 
> #endif  // API_PEERCONNECTIONINTERFACE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/peerconnectionproxy.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/peerconnectionproxy.h
>index 7235f5b4b2f..7fc5e1716aa 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/peerconnectionproxy.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/peerconnectionproxy.h
>@@ -23,124 +23,124 @@ namespace webrtc {
> // TODO(deadbeef): Move this to .cc file and out of api/. What threads methods
> // are called on is an implementation detail.
> BEGIN_SIGNALING_PROXY_MAP(PeerConnection)
>-  PROXY_SIGNALING_THREAD_DESTRUCTOR()
>-  PROXY_METHOD0(rtc::scoped_refptr<StreamCollectionInterface>, local_streams)
>-  PROXY_METHOD0(rtc::scoped_refptr<StreamCollectionInterface>, remote_streams)
>-  PROXY_METHOD1(bool, AddStream, MediaStreamInterface*)
>-  PROXY_METHOD1(void, RemoveStream, MediaStreamInterface*)
>-  PROXY_METHOD2(RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>>,
>-                AddTrack,
>-                rtc::scoped_refptr<MediaStreamTrackInterface>,
>-                const std::vector<std::string>&);
>-  PROXY_METHOD2(rtc::scoped_refptr<RtpSenderInterface>,
>-                AddTrack,
>-                MediaStreamTrackInterface*,
>-                std::vector<MediaStreamInterface*>)
>-  PROXY_METHOD1(bool, RemoveTrack, RtpSenderInterface*)
>-  PROXY_METHOD1(RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>,
>-                AddTransceiver,
>-                rtc::scoped_refptr<MediaStreamTrackInterface>)
>-  PROXY_METHOD2(RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>,
>-                AddTransceiver,
>-                rtc::scoped_refptr<MediaStreamTrackInterface>,
>-                const RtpTransceiverInit&)
>-  PROXY_METHOD1(RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>,
>-                AddTransceiver,
>-                cricket::MediaType)
>-  PROXY_METHOD2(RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>,
>-                AddTransceiver,
>-                cricket::MediaType,
>-                const RtpTransceiverInit&)
>-  PROXY_METHOD1(rtc::scoped_refptr<DtmfSenderInterface>,
>-                CreateDtmfSender,
>-                AudioTrackInterface*)
>-  PROXY_METHOD2(rtc::scoped_refptr<RtpSenderInterface>,
>-                CreateSender,
>-                const std::string&,
>-                const std::string&)
>-  PROXY_CONSTMETHOD0(std::vector<rtc::scoped_refptr<RtpSenderInterface>>,
>-                     GetSenders)
>-  PROXY_CONSTMETHOD0(std::vector<rtc::scoped_refptr<RtpReceiverInterface>>,
>-                     GetReceivers)
>-  PROXY_CONSTMETHOD0(std::vector<rtc::scoped_refptr<RtpTransceiverInterface>>,
>-                     GetTransceivers)
>-  PROXY_METHOD3(bool,
>-                GetStats,
>-                StatsObserver*,
>-                MediaStreamTrackInterface*,
>-                StatsOutputLevel)
>-  PROXY_METHOD1(void, GetStats, RTCStatsCollectorCallback*)
>-  PROXY_METHOD2(rtc::scoped_refptr<DataChannelInterface>,
>-                CreateDataChannel,
>-                const std::string&,
>-                const DataChannelInit*)
>-  PROXY_CONSTMETHOD0(const SessionDescriptionInterface*, local_description)
>-  PROXY_CONSTMETHOD0(const SessionDescriptionInterface*, remote_description)
>-  PROXY_CONSTMETHOD0(const SessionDescriptionInterface*,
>-                     pending_local_description)
>-  PROXY_CONSTMETHOD0(const SessionDescriptionInterface*,
>-                     pending_remote_description)
>-  PROXY_CONSTMETHOD0(const SessionDescriptionInterface*,
>-                     current_local_description)
>-  PROXY_CONSTMETHOD0(const SessionDescriptionInterface*,
>-                     current_remote_description)
>-  PROXY_METHOD2(void,
>-                CreateOffer,
>-                CreateSessionDescriptionObserver*,
>-                const MediaConstraintsInterface*)
>-  PROXY_METHOD2(void,
>-                CreateAnswer,
>-                CreateSessionDescriptionObserver*,
>-                const MediaConstraintsInterface*)
>-  PROXY_METHOD2(void,
>-                CreateOffer,
>-                CreateSessionDescriptionObserver*,
>-                const RTCOfferAnswerOptions&)
>-  PROXY_METHOD2(void,
>-                CreateAnswer,
>-                CreateSessionDescriptionObserver*,
>-                const RTCOfferAnswerOptions&)
>-  PROXY_METHOD2(void,
>-                SetLocalDescription,
>-                SetSessionDescriptionObserver*,
>-                SessionDescriptionInterface*)
>-  PROXY_METHOD2(void,
>-                SetRemoteDescription,
>-                SetSessionDescriptionObserver*,
>-                SessionDescriptionInterface*)
>-  PROXY_METHOD2(void,
>-                SetRemoteDescription,
>-                std::unique_ptr<SessionDescriptionInterface>,
>-                rtc::scoped_refptr<SetRemoteDescriptionObserverInterface>);
>-  PROXY_METHOD0(PeerConnectionInterface::RTCConfiguration, GetConfiguration);
>-  PROXY_METHOD2(bool,
>-                SetConfiguration,
>-                const PeerConnectionInterface::RTCConfiguration&,
>-                RTCError*);
>-  PROXY_METHOD1(bool,
>-                SetConfiguration,
>-                const PeerConnectionInterface::RTCConfiguration&);
>-  PROXY_METHOD1(bool, AddIceCandidate, const IceCandidateInterface*)
>-  PROXY_METHOD1(bool,
>-                RemoveIceCandidates,
>-                const std::vector<cricket::Candidate>&);
>-  PROXY_METHOD1(void, SetAudioPlayout, bool)
>-  PROXY_METHOD1(void, SetAudioRecording, bool)
>-  PROXY_METHOD1(void, RegisterUMAObserver, UMAObserver*)
>-  PROXY_METHOD1(RTCError, SetBitrate, const BitrateParameters&);
>-  PROXY_METHOD1(void,
>-                SetBitrateAllocationStrategy,
>-                std::unique_ptr<rtc::BitrateAllocationStrategy>);
>-  PROXY_METHOD0(SignalingState, signaling_state)
>-  PROXY_METHOD0(IceConnectionState, ice_connection_state)
>-  PROXY_METHOD0(IceGatheringState, ice_gathering_state)
>-  PROXY_METHOD2(bool, StartRtcEventLog, rtc::PlatformFile, int64_t)
>-  PROXY_METHOD2(bool,
>-                StartRtcEventLog,
>-                std::unique_ptr<RtcEventLogOutput>,
>-                int64_t);
>-  PROXY_METHOD0(void, StopRtcEventLog)
>-  PROXY_METHOD0(void, Close)
>+PROXY_SIGNALING_THREAD_DESTRUCTOR()
>+PROXY_METHOD0(rtc::scoped_refptr<StreamCollectionInterface>, local_streams)
>+PROXY_METHOD0(rtc::scoped_refptr<StreamCollectionInterface>, remote_streams)
>+PROXY_METHOD1(bool, AddStream, MediaStreamInterface*)
>+PROXY_METHOD1(void, RemoveStream, MediaStreamInterface*)
>+PROXY_METHOD2(RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>>,
>+              AddTrack,
>+              rtc::scoped_refptr<MediaStreamTrackInterface>,
>+              const std::vector<std::string>&);
>+PROXY_METHOD1(bool, RemoveTrack, RtpSenderInterface*)
>+PROXY_METHOD1(RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>,
>+              AddTransceiver,
>+              rtc::scoped_refptr<MediaStreamTrackInterface>)
>+PROXY_METHOD2(RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>,
>+              AddTransceiver,
>+              rtc::scoped_refptr<MediaStreamTrackInterface>,
>+              const RtpTransceiverInit&)
>+PROXY_METHOD1(RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>,
>+              AddTransceiver,
>+              cricket::MediaType)
>+PROXY_METHOD2(RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>,
>+              AddTransceiver,
>+              cricket::MediaType,
>+              const RtpTransceiverInit&)
>+PROXY_METHOD2(rtc::scoped_refptr<RtpSenderInterface>,
>+              CreateSender,
>+              const std::string&,
>+              const std::string&)
>+PROXY_CONSTMETHOD0(std::vector<rtc::scoped_refptr<RtpSenderInterface>>,
>+                   GetSenders)
>+PROXY_CONSTMETHOD0(std::vector<rtc::scoped_refptr<RtpReceiverInterface>>,
>+                   GetReceivers)
>+PROXY_CONSTMETHOD0(std::vector<rtc::scoped_refptr<RtpTransceiverInterface>>,
>+                   GetTransceivers)
>+PROXY_METHOD3(bool,
>+              GetStats,
>+              StatsObserver*,
>+              MediaStreamTrackInterface*,
>+              StatsOutputLevel)
>+PROXY_METHOD1(void, GetStats, RTCStatsCollectorCallback*)
>+PROXY_METHOD2(void,
>+              GetStats,
>+              rtc::scoped_refptr<RtpSenderInterface>,
>+              rtc::scoped_refptr<RTCStatsCollectorCallback>);
>+PROXY_METHOD2(void,
>+              GetStats,
>+              rtc::scoped_refptr<RtpReceiverInterface>,
>+              rtc::scoped_refptr<RTCStatsCollectorCallback>);
>+PROXY_METHOD2(rtc::scoped_refptr<DataChannelInterface>,
>+              CreateDataChannel,
>+              const std::string&,
>+              const DataChannelInit*)
>+PROXY_CONSTMETHOD0(const SessionDescriptionInterface*, local_description)
>+PROXY_CONSTMETHOD0(const SessionDescriptionInterface*, remote_description)
>+PROXY_CONSTMETHOD0(const SessionDescriptionInterface*,
>+                   pending_local_description)
>+PROXY_CONSTMETHOD0(const SessionDescriptionInterface*,
>+                   pending_remote_description)
>+PROXY_CONSTMETHOD0(const SessionDescriptionInterface*,
>+                   current_local_description)
>+PROXY_CONSTMETHOD0(const SessionDescriptionInterface*,
>+                   current_remote_description)
>+PROXY_METHOD2(void,
>+              CreateOffer,
>+              CreateSessionDescriptionObserver*,
>+              const MediaConstraintsInterface*)
>+PROXY_METHOD2(void,
>+              CreateAnswer,
>+              CreateSessionDescriptionObserver*,
>+              const MediaConstraintsInterface*)
>+PROXY_METHOD2(void,
>+              CreateOffer,
>+              CreateSessionDescriptionObserver*,
>+              const RTCOfferAnswerOptions&)
>+PROXY_METHOD2(void,
>+              CreateAnswer,
>+              CreateSessionDescriptionObserver*,
>+              const RTCOfferAnswerOptions&)
>+PROXY_METHOD2(void,
>+              SetLocalDescription,
>+              SetSessionDescriptionObserver*,
>+              SessionDescriptionInterface*)
>+PROXY_METHOD2(void,
>+              SetRemoteDescription,
>+              SetSessionDescriptionObserver*,
>+              SessionDescriptionInterface*)
>+PROXY_METHOD2(void,
>+              SetRemoteDescription,
>+              std::unique_ptr<SessionDescriptionInterface>,
>+              rtc::scoped_refptr<SetRemoteDescriptionObserverInterface>);
>+PROXY_METHOD0(PeerConnectionInterface::RTCConfiguration, GetConfiguration);
>+PROXY_METHOD2(bool,
>+              SetConfiguration,
>+              const PeerConnectionInterface::RTCConfiguration&,
>+              RTCError*);
>+PROXY_METHOD1(bool,
>+              SetConfiguration,
>+              const PeerConnectionInterface::RTCConfiguration&);
>+PROXY_METHOD1(bool, AddIceCandidate, const IceCandidateInterface*)
>+PROXY_METHOD1(bool,
>+              RemoveIceCandidates,
>+              const std::vector<cricket::Candidate>&);
>+PROXY_METHOD1(void, SetAudioPlayout, bool)
>+PROXY_METHOD1(void, SetAudioRecording, bool)
>+PROXY_METHOD1(RTCError, SetBitrate, const BitrateSettings&);
>+PROXY_METHOD1(void,
>+              SetBitrateAllocationStrategy,
>+              std::unique_ptr<rtc::BitrateAllocationStrategy>);
>+PROXY_METHOD0(SignalingState, signaling_state)
>+PROXY_METHOD0(IceConnectionState, ice_connection_state)
>+PROXY_METHOD0(IceGatheringState, ice_gathering_state)
>+PROXY_METHOD2(bool, StartRtcEventLog, rtc::PlatformFile, int64_t)
>+PROXY_METHOD2(bool,
>+              StartRtcEventLog,
>+              std::unique_ptr<RtcEventLogOutput>,
>+              int64_t);
>+PROXY_METHOD0(void, StopRtcEventLog)
>+PROXY_METHOD0(void, Close)
> END_PROXY_MAP()
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/proxy.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/proxy.h
>index dd7182e5546..c8962efed7c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/proxy.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/proxy.h
>@@ -64,8 +64,10 @@ namespace webrtc {
> template <typename R>
> class ReturnType {
>  public:
>-  template<typename C, typename M>
>-  void Invoke(C* c, M m) { r_ = (c->*m)(); }
>+  template <typename C, typename M>
>+  void Invoke(C* c, M m) {
>+    r_ = (c->*m)();
>+  }
>   template <typename C, typename M, typename T1>
>   void Invoke(C* c, M m, T1 a1) {
>     r_ = (c->*m)(std::move(a1));
>@@ -78,13 +80,22 @@ class ReturnType {
>   void Invoke(C* c, M m, T1 a1, T2 a2, T3 a3) {
>     r_ = (c->*m)(std::move(a1), std::move(a2), std::move(a3));
>   }
>-  template<typename C, typename M, typename T1, typename T2, typename T3,
>-      typename T4>
>+  template <typename C,
>+            typename M,
>+            typename T1,
>+            typename T2,
>+            typename T3,
>+            typename T4>
>   void Invoke(C* c, M m, T1 a1, T2 a2, T3 a3, T4 a4) {
>     r_ = (c->*m)(std::move(a1), std::move(a2), std::move(a3), std::move(a4));
>   }
>-  template<typename C, typename M, typename T1, typename T2, typename T3,
>-     typename T4, typename T5>
>+  template <typename C,
>+            typename M,
>+            typename T1,
>+            typename T2,
>+            typename T3,
>+            typename T4,
>+            typename T5>
>   void Invoke(C* c, M m, T1 a1, T2 a2, T3 a3, T4 a4, T5 a5) {
>     r_ = (c->*m)(std::move(a1), std::move(a2), std::move(a3), std::move(a4),
>                  std::move(a5));
>@@ -99,8 +110,10 @@ class ReturnType {
> template <>
> class ReturnType<void> {
>  public:
>-  template<typename C, typename M>
>-  void Invoke(C* c, M m) { (c->*m)(); }
>+  template <typename C, typename M>
>+  void Invoke(C* c, M m) {
>+    (c->*m)();
>+  }
>   template <typename C, typename M, typename T1>
>   void Invoke(C* c, M m, T1 a1) {
>     (c->*m)(std::move(a1));
>@@ -119,9 +132,8 @@ class ReturnType<void> {
> 
> namespace internal {
> 
>-class SynchronousMethodCall
>-    : public rtc::MessageData,
>-      public rtc::MessageHandler {
>+class SynchronousMethodCall : public rtc::MessageData,
>+                              public rtc::MessageHandler {
>  public:
>   explicit SynchronousMethodCall(rtc::MessageHandler* proxy);
>   ~SynchronousMethodCall() override;
>@@ -138,8 +150,7 @@ class SynchronousMethodCall
> }  // namespace internal
> 
> template <typename C, typename R>
>-class MethodCall0 : public rtc::Message,
>-                    public rtc::MessageHandler {
>+class MethodCall0 : public rtc::Message, public rtc::MessageHandler {
>  public:
>   typedef R (C::*Method)();
>   MethodCall0(C* c, Method m) : c_(c), m_(m) {}
>@@ -150,7 +161,7 @@ class MethodCall0 : public rtc::Message,
>   }
> 
>  private:
>-  void OnMessage(rtc::Message*) {  r_.Invoke(c_, m_); }
>+  void OnMessage(rtc::Message*) { r_.Invoke(c_, m_); }
> 
>   C* c_;
>   Method m_;
>@@ -158,8 +169,7 @@ class MethodCall0 : public rtc::Message,
> };
> 
> template <typename C, typename R>
>-class ConstMethodCall0 : public rtc::Message,
>-                         public rtc::MessageHandler {
>+class ConstMethodCall0 : public rtc::Message, public rtc::MessageHandler {
>  public:
>   typedef R (C::*Method)() const;
>   ConstMethodCall0(C* c, Method m) : c_(c), m_(m) {}
>@@ -177,9 +187,8 @@ class ConstMethodCall0 : public rtc::Message,
>   ReturnType<R> r_;
> };
> 
>-template <typename C, typename R,  typename T1>
>-class MethodCall1 : public rtc::Message,
>-                    public rtc::MessageHandler {
>+template <typename C, typename R, typename T1>
>+class MethodCall1 : public rtc::Message, public rtc::MessageHandler {
>  public:
>   typedef R (C::*Method)(T1 a1);
>   MethodCall1(C* c, Method m, T1 a1) : c_(c), m_(m), a1_(std::move(a1)) {}
>@@ -198,9 +207,8 @@ class MethodCall1 : public rtc::Message,
>   T1 a1_;
> };
> 
>-template <typename C, typename R,  typename T1>
>-class ConstMethodCall1 : public rtc::Message,
>-                         public rtc::MessageHandler {
>+template <typename C, typename R, typename T1>
>+class ConstMethodCall1 : public rtc::Message, public rtc::MessageHandler {
>  public:
>   typedef R (C::*Method)(T1 a1) const;
>   ConstMethodCall1(C* c, Method m, T1 a1) : c_(c), m_(m), a1_(std::move(a1)) {}
>@@ -220,8 +228,7 @@ class ConstMethodCall1 : public rtc::Message,
> };
> 
> template <typename C, typename R, typename T1, typename T2>
>-class MethodCall2 : public rtc::Message,
>-                    public rtc::MessageHandler {
>+class MethodCall2 : public rtc::Message, public rtc::MessageHandler {
>  public:
>   typedef R (C::*Method)(T1 a1, T2 a2);
>   MethodCall2(C* c, Method m, T1 a1, T2 a2)
>@@ -245,8 +252,7 @@ class MethodCall2 : public rtc::Message,
> };
> 
> template <typename C, typename R, typename T1, typename T2, typename T3>
>-class MethodCall3 : public rtc::Message,
>-                    public rtc::MessageHandler {
>+class MethodCall3 : public rtc::Message, public rtc::MessageHandler {
>  public:
>   typedef R (C::*Method)(T1 a1, T2 a2, T3 a3);
>   MethodCall3(C* c, Method m, T1 a1, T2 a2, T3 a3)
>@@ -274,10 +280,13 @@ class MethodCall3 : public rtc::Message,
>   T3 a3_;
> };
> 
>-template <typename C, typename R, typename T1, typename T2, typename T3,
>-    typename T4>
>-class MethodCall4 : public rtc::Message,
>-                    public rtc::MessageHandler {
>+template <typename C,
>+          typename R,
>+          typename T1,
>+          typename T2,
>+          typename T3,
>+          typename T4>
>+class MethodCall4 : public rtc::Message, public rtc::MessageHandler {
>  public:
>   typedef R (C::*Method)(T1 a1, T2 a2, T3 a3, T4 a4);
>   MethodCall4(C* c, Method m, T1 a1, T2 a2, T3 a3, T4 a4)
>@@ -308,10 +317,14 @@ class MethodCall4 : public rtc::Message,
>   T4 a4_;
> };
> 
>-template <typename C, typename R, typename T1, typename T2, typename T3,
>-    typename T4, typename T5>
>-class MethodCall5 : public rtc::Message,
>-                    public rtc::MessageHandler {
>+template <typename C,
>+          typename R,
>+          typename T1,
>+          typename T2,
>+          typename T3,
>+          typename T4,
>+          typename T5>
>+class MethodCall5 : public rtc::Message, public rtc::MessageHandler {
>  public:
>   typedef R (C::*Method)(T1 a1, T2 a2, T3 a3, T4 a4, T5 a5);
>   MethodCall5(C* c, Method m, T1 a1, T2 a2, T3 a3, T4 a4, T5 a5)
>@@ -344,7 +357,6 @@ class MethodCall5 : public rtc::Message,
>   T5 a5_;
> };
> 
>-
> // Helper macros to reduce code duplication.
> #define PROXY_MAP_BOILERPLATE(c)                          \
>   template <class INTERNAL_CLASS>                         \
>@@ -359,8 +371,12 @@ class MethodCall5 : public rtc::Message,
>     const INTERNAL_CLASS* internal() const { return c_; } \
>     INTERNAL_CLASS* internal() { return c_; }
> 
>+// clang-format off
>+// clang-format would put the semicolon alone,
>+// leading to a presubmit error (cpplint.py)
> #define END_PROXY_MAP() \
>   };
>+// clang-format on
> 
> #define SIGNALING_PROXY_MAP_BOILERPLATE(c)                               \
>  protected:                                                              \
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtcerror.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtcerror.cc
>index f9a31d08934..55ac15e196b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtcerror.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtcerror.cc
>@@ -93,9 +93,10 @@ void RTCError::set_message(std::string&& message) {
>   }
> }
> 
>-std::ostream& operator<<(std::ostream& stream, RTCErrorType error) {
>+// TODO(jonasolsson): Change to use absl::string_view when it's available.
>+std::string ToString(RTCErrorType error) {
>   int index = static_cast<int>(error);
>-  return stream << kRTCErrorTypeNames[index];
>+  return std::string(kRTCErrorTypeNames[index]);
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtcerror.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtcerror.h
>index 962f46dd818..c87ce916501 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtcerror.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtcerror.h
>@@ -11,7 +11,9 @@
> #ifndef API_RTCERROR_H_
> #define API_RTCERROR_H_
> 
>+#ifdef UNIT_TEST
> #include <ostream>
>+#endif  // UNIT_TEST
> #include <string>
> #include <utility>  // For std::move.
> 
>@@ -143,16 +145,24 @@ class RTCError {
> // error type.
> //
> // Only intended to be used for logging/disagnostics.
>-std::ostream& operator<<(std::ostream& stream, RTCErrorType error);
>+std::string ToString(RTCErrorType error);
>+
>+#ifdef UNIT_TEST
>+inline std::ostream& operator<<(  // no-presubmit-check TODO(webrtc:8982)
>+    std::ostream& stream,         // no-presubmit-check TODO(webrtc:8982)
>+    RTCErrorType error) {
>+  return stream << ToString(error);
>+}
>+#endif  // UNIT_TEST
> 
> // Helper macro that can be used by implementations to create an error with a
> // message and log it. |message| should be a string literal or movable
> // std::string.
>-#define LOG_AND_RETURN_ERROR_EX(type, message, severity) \
>-  {                                                      \
>-    RTC_DCHECK(type != RTCErrorType::NONE);              \
>-    RTC_LOG(severity) << message << " (" << type << ")"; \
>-    return webrtc::RTCError(type, message);              \
>+#define LOG_AND_RETURN_ERROR_EX(type, message, severity)           \
>+  {                                                                \
>+    RTC_DCHECK(type != RTCErrorType::NONE);                        \
>+    RTC_LOG(severity) << message << " (" << ToString(type) << ")"; \
>+    return webrtc::RTCError(type, message);                        \
>   }
> 
> #define LOG_AND_RETURN_ERROR(type, message) \
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtcerror_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtcerror_unittest.cc
>index d8f7ca6d49d..90593cf5246 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtcerror_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtcerror_unittest.cc
>@@ -58,15 +58,6 @@ struct MoveOnlyInt2 {
> 
> namespace webrtc {
> 
>-// Simple test for ostream operator for RTCErrorType.
>-TEST(RTCErrorTypeTest, OstreamOperator) {
>-  std::ostringstream oss;
>-  oss << webrtc::RTCErrorType::NONE << ' '
>-      << webrtc::RTCErrorType::INVALID_PARAMETER << ' '
>-      << webrtc::RTCErrorType::INTERNAL_ERROR;
>-  EXPECT_EQ("NONE INVALID_PARAMETER INTERNAL_ERROR", oss.str());
>-}
>-
> // Test that the default constructor creates a "no error" error.
> TEST(RTCErrorTest, DefaultConstructor) {
>   RTCError e;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtp_headers.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtp_headers.h
>index 2624a33e5f5..c762534ab3a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtp_headers.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtp_headers.h
>@@ -13,20 +13,18 @@
> 
> #include <stddef.h>
> #include <string.h>
>-#include <ostream>
> #include <string>
> #include <vector>
> 
>+#include "absl/types/optional.h"
> #include "api/array_view.h"
>-#include "api/optional.h"
> #include "api/video/video_content_type.h"
> #include "api/video/video_rotation.h"
> #include "api/video/video_timing.h"
> 
>+#include "common_types.h"  // NOLINT(build/include)
> #include "rtc_base/checks.h"
> #include "rtc_base/deprecation.h"
>-#include "common_types.h"  // NOLINT(build/include)
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -41,7 +39,14 @@ class StringRtpHeaderExtension {
>   // maximum length that can be encoded with one-byte header extensions.
>   static constexpr size_t kMaxSize = 16;
> 
>-  static bool IsLegalName(rtc::ArrayView<const char> name);
>+  static bool IsLegalMidName(rtc::ArrayView<const char> name);
>+  static bool IsLegalRsidName(rtc::ArrayView<const char> name);
>+
>+  // TODO(bugs.webrtc.org/9537): Deprecate and remove when third parties have
>+  // migrated to "IsLegalRsidName".
>+  static bool IsLegalName(rtc::ArrayView<const char> name) {
>+    return IsLegalRsidName(name);
>+  }
> 
>   StringRtpHeaderExtension() { value_[0] = 0; }
>   explicit StringRtpHeaderExtension(rtc::ArrayView<const char> value) {
>@@ -103,7 +108,7 @@ struct RTPHeaderExtension {
>   bool hasVideoRotation;
>   VideoRotation videoRotation;
> 
>-  // TODO(ilnik): Refactor this and one above to be rtc::Optional() and remove
>+  // TODO(ilnik): Refactor this and one above to be absl::optional() and remove
>   // a corresponding bool flag.
>   bool hasVideoContentType;
>   VideoContentType videoContentType;
>@@ -168,11 +173,6 @@ struct RtpKeepAliveConfig final {
>   bool operator!=(const RtpKeepAliveConfig& o) const { return !(*this == o); }
> };
> 
>-// Currently only VP8/VP9 specific.
>-struct RtpPayloadState {
>-  int16_t picture_id = -1;
>-};
>-
> }  // namespace webrtc
> 
> #endif  // API_RTP_HEADERS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtpparameters.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtpparameters.cc
>index 79fd3a93f93..62ca3fc0254 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtpparameters.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtpparameters.cc
>@@ -10,26 +10,27 @@
> #include "api/rtpparameters.h"
> 
> #include <algorithm>
>-#include <sstream>
> #include <string>
> 
> #include "rtc_base/checks.h"
>+#include "rtc_base/strings/string_builder.h"
> 
> namespace webrtc {
> 
> const double kDefaultBitratePriority = 1.0;
> 
>-RtcpFeedback::RtcpFeedback() {}
>+RtcpFeedback::RtcpFeedback() = default;
> RtcpFeedback::RtcpFeedback(RtcpFeedbackType type) : type(type) {}
> RtcpFeedback::RtcpFeedback(RtcpFeedbackType type,
>                            RtcpFeedbackMessageType message_type)
>     : type(type), message_type(message_type) {}
>-RtcpFeedback::~RtcpFeedback() {}
>+RtcpFeedback::RtcpFeedback(const RtcpFeedback& rhs) = default;
>+RtcpFeedback::~RtcpFeedback() = default;
> 
>-RtpCodecCapability::RtpCodecCapability() {}
>-RtpCodecCapability::~RtpCodecCapability() {}
>+RtpCodecCapability::RtpCodecCapability() = default;
>+RtpCodecCapability::~RtpCodecCapability() = default;
> 
>-RtpHeaderExtensionCapability::RtpHeaderExtensionCapability() {}
>+RtpHeaderExtensionCapability::RtpHeaderExtensionCapability() = default;
> RtpHeaderExtensionCapability::RtpHeaderExtensionCapability(
>     const std::string& uri)
>     : uri(uri) {}
>@@ -37,46 +38,57 @@ RtpHeaderExtensionCapability::RtpHeaderExtensionCapability(
>     const std::string& uri,
>     int preferred_id)
>     : uri(uri), preferred_id(preferred_id) {}
>-RtpHeaderExtensionCapability::~RtpHeaderExtensionCapability() {}
>+RtpHeaderExtensionCapability::~RtpHeaderExtensionCapability() = default;
> 
>-RtpExtension::RtpExtension() {}
>+RtpExtension::RtpExtension() = default;
> RtpExtension::RtpExtension(const std::string& uri, int id) : uri(uri), id(id) {}
> RtpExtension::RtpExtension(const std::string& uri, int id, bool encrypt)
>     : uri(uri), id(id), encrypt(encrypt) {}
>-RtpExtension::~RtpExtension() {}
>+RtpExtension::~RtpExtension() = default;
> 
>-RtpFecParameters::RtpFecParameters() {}
>+RtpFecParameters::RtpFecParameters() = default;
> RtpFecParameters::RtpFecParameters(FecMechanism mechanism)
>     : mechanism(mechanism) {}
> RtpFecParameters::RtpFecParameters(FecMechanism mechanism, uint32_t ssrc)
>     : ssrc(ssrc), mechanism(mechanism) {}
>-RtpFecParameters::~RtpFecParameters() {}
>+RtpFecParameters::RtpFecParameters(const RtpFecParameters& rhs) = default;
>+RtpFecParameters::~RtpFecParameters() = default;
> 
>-RtpRtxParameters::RtpRtxParameters() {}
>+RtpRtxParameters::RtpRtxParameters() = default;
> RtpRtxParameters::RtpRtxParameters(uint32_t ssrc) : ssrc(ssrc) {}
>-RtpRtxParameters::~RtpRtxParameters() {}
>+RtpRtxParameters::RtpRtxParameters(const RtpRtxParameters& rhs) = default;
>+RtpRtxParameters::~RtpRtxParameters() = default;
> 
>-RtpEncodingParameters::RtpEncodingParameters() {}
>-RtpEncodingParameters::~RtpEncodingParameters() {}
>+RtpEncodingParameters::RtpEncodingParameters() = default;
>+RtpEncodingParameters::RtpEncodingParameters(const RtpEncodingParameters& rhs) =
>+    default;
>+RtpEncodingParameters::~RtpEncodingParameters() = default;
> 
>-RtpCodecParameters::RtpCodecParameters() {}
>-RtpCodecParameters::~RtpCodecParameters() {}
>+RtpCodecParameters::RtpCodecParameters() = default;
>+RtpCodecParameters::RtpCodecParameters(const RtpCodecParameters& rhs) = default;
>+RtpCodecParameters::~RtpCodecParameters() = default;
> 
>-RtpCapabilities::RtpCapabilities() {}
>-RtpCapabilities::~RtpCapabilities() {}
>+RtpCapabilities::RtpCapabilities() = default;
>+RtpCapabilities::~RtpCapabilities() = default;
> 
>-RtpParameters::RtpParameters() {}
>-RtpParameters::~RtpParameters() {}
>+RtcpParameters::RtcpParameters() = default;
>+RtcpParameters::RtcpParameters(const RtcpParameters& rhs) = default;
>+RtcpParameters::~RtcpParameters() = default;
>+
>+RtpParameters::RtpParameters() = default;
>+RtpParameters::RtpParameters(const RtpParameters& rhs) = default;
>+RtpParameters::~RtpParameters() = default;
> 
> std::string RtpExtension::ToString() const {
>-  std::stringstream ss;
>-  ss << "{uri: " << uri;
>-  ss << ", id: " << id;
>+  char buf[256];
>+  rtc::SimpleStringBuilder sb(buf);
>+  sb << "{uri: " << uri;
>+  sb << ", id: " << id;
>   if (encrypt) {
>-    ss << ", encrypt";
>+    sb << ", encrypt";
>   }
>-  ss << '}';
>-  return ss.str();
>+  sb << '}';
>+  return sb.str();
> }
> 
> const char RtpExtension::kAudioLevelUri[] =
>@@ -114,6 +126,9 @@ const char RtpExtension::kVideoTimingUri[] =
>     "http://www.webrtc.org/experiments/rtp-hdrext/video-timing";
> const int RtpExtension::kVideoTimingDefaultId = 8;
> 
>+const char RtpExtension::kMidUri[] = "urn:ietf:params:rtp-hdrext:sdes:mid";
>+const int RtpExtension::kMidDefaultId = 9;
>+
> const char RtpExtension::kEncryptHeaderExtensionsUri[] =
>     "urn:ietf:params:rtp-hdrext:encrypt";
> 
>@@ -122,7 +137,8 @@ const int RtpExtension::kMaxId = 14;
> 
> bool RtpExtension::IsSupportedForAudio(const std::string& uri) {
>   return uri == webrtc::RtpExtension::kAudioLevelUri ||
>-         uri == webrtc::RtpExtension::kTransportSequenceNumberUri;
>+         uri == webrtc::RtpExtension::kTransportSequenceNumberUri ||
>+         uri == webrtc::RtpExtension::kMidUri;
> }
> 
> bool RtpExtension::IsSupportedForVideo(const std::string& uri) {
>@@ -132,7 +148,8 @@ bool RtpExtension::IsSupportedForVideo(const std::string& uri) {
>          uri == webrtc::RtpExtension::kTransportSequenceNumberUri ||
>          uri == webrtc::RtpExtension::kPlayoutDelayUri ||
>          uri == webrtc::RtpExtension::kVideoContentTypeUri ||
>-         uri == webrtc::RtpExtension::kVideoTimingUri;
>+         uri == webrtc::RtpExtension::kVideoTimingUri ||
>+         uri == webrtc::RtpExtension::kMidUri;
> }
> 
> bool RtpExtension::IsEncryptionSupported(const std::string& uri) {
>@@ -149,7 +166,8 @@ bool RtpExtension::IsEncryptionSupported(const std::string& uri) {
>          uri == webrtc::RtpExtension::kVideoRotationUri ||
>          uri == webrtc::RtpExtension::kTransportSequenceNumberUri ||
>          uri == webrtc::RtpExtension::kPlayoutDelayUri ||
>-         uri == webrtc::RtpExtension::kVideoContentTypeUri;
>+         uri == webrtc::RtpExtension::kVideoContentTypeUri ||
>+         uri == webrtc::RtpExtension::kMidUri;
> }
> 
> const RtpExtension* RtpExtension::FindHeaderExtensionByUri(
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtpparameters.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtpparameters.h
>index e3408258605..9a29c08d1c9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtpparameters.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtpparameters.h
>@@ -15,8 +15,8 @@
> #include <unordered_map>
> #include <vector>
> 
>+#include "absl/types/optional.h"
> #include "api/mediatypes.h"
>-#include "api/optional.h"
> 
> namespace webrtc {
> 
>@@ -66,9 +66,22 @@ enum class DtxStatus {
>   ENABLED,
> };
> 
>+// Based on the spec in
>+// https://w3c.github.io/webrtc-pc/#idl-def-rtcdegradationpreference.
>+// These options are enforced on a best-effort basis. For instance, all of
>+// these options may suffer some frame drops in order to avoid queuing.
>+// TODO(sprang): Look into possibility of more strictly enforcing the
>+// maintain-framerate option.
>+// TODO(deadbeef): Default to "balanced", as the spec indicates?
> enum class DegradationPreference {
>+  // Don't take any actions based on over-utilization signals. Not part of the
>+  // web API.
>+  DISABLED,
>+  // On over-use, request lower frame rate, possibly causing frame drops.
>   MAINTAIN_FRAMERATE,
>+  // On over-use, request lower resolution, possibly causing down-scaling.
>   MAINTAIN_RESOLUTION,
>+  // Try to strike a "pleasing" balance between frame rate or resolution.
>   BALANCED,
> };
> 
>@@ -81,12 +94,13 @@ struct RtcpFeedback {
>   // 1. It's an enum instead of a string.
>   // 2. Generic NACK feedback is represented by a GENERIC_NACK message type,
>   //    rather than an unset "parameter" value.
>-  rtc::Optional<RtcpFeedbackMessageType> message_type;
>+  absl::optional<RtcpFeedbackMessageType> message_type;
> 
>   // Constructors for convenience.
>   RtcpFeedback();
>   explicit RtcpFeedback(RtcpFeedbackType type);
>   RtcpFeedback(RtcpFeedbackType type, RtcpFeedbackMessageType message_type);
>+  RtcpFeedback(const RtcpFeedback&);
>   ~RtcpFeedback();
> 
>   bool operator==(const RtcpFeedback& o) const {
>@@ -112,23 +126,23 @@ struct RtpCodecCapability {
>   cricket::MediaType kind = cricket::MEDIA_TYPE_AUDIO;
> 
>   // Clock rate in Hertz. If unset, the codec is applicable to any clock rate.
>-  rtc::Optional<int> clock_rate;
>+  absl::optional<int> clock_rate;
> 
>   // Default payload type for this codec. Mainly needed for codecs that use
>   // that have statically assigned payload types.
>-  rtc::Optional<int> preferred_payload_type;
>+  absl::optional<int> preferred_payload_type;
> 
>   // Maximum packetization time supported by an RtpReceiver for this codec.
>   // TODO(deadbeef): Not implemented.
>-  rtc::Optional<int> max_ptime;
>+  absl::optional<int> max_ptime;
> 
>   // Preferred packetization time for an RtpReceiver or RtpSender of this
>   // codec.
>   // TODO(deadbeef): Not implemented.
>-  rtc::Optional<int> ptime;
>+  absl::optional<int> ptime;
> 
>   // The number of audio channels supported. Unused for video codecs.
>-  rtc::Optional<int> num_channels;
>+  absl::optional<int> num_channels;
> 
>   // Feedback mechanisms supported for this codec.
>   std::vector<RtcpFeedback> rtcp_feedback;
>@@ -191,7 +205,7 @@ struct RtpHeaderExtensionCapability {
>   std::string uri;
> 
>   // Preferred value of ID that goes in the packet.
>-  rtc::Optional<int> preferred_id;
>+  absl::optional<int> preferred_id;
> 
>   // If true, it's preferred that the value in the header is encrypted.
>   // TODO(deadbeef): Not implemented.
>@@ -276,6 +290,11 @@ struct RtpExtension {
>   static const char kPlayoutDelayUri[];
>   static const int kPlayoutDelayDefaultId;
> 
>+  // Header extension for identifying media section within a transport.
>+  // https://tools.ietf.org/html/draft-ietf-mmusic-sdp-bundle-negotiation-49#section-15
>+  static const char kMidUri[];
>+  static const int kMidDefaultId;
>+
>   // Encryption of Header Extensions, see RFC 6904 for details:
>   // https://tools.ietf.org/html/rfc6904
>   static const char kEncryptHeaderExtensionsUri[];
>@@ -295,7 +314,7 @@ typedef RtpExtension RtpHeaderExtensionParameters;
> struct RtpFecParameters {
>   // If unset, a value is chosen by the implementation.
>   // Works just like RtpEncodingParameters::ssrc.
>-  rtc::Optional<uint32_t> ssrc;
>+  absl::optional<uint32_t> ssrc;
> 
>   FecMechanism mechanism = FecMechanism::RED;
> 
>@@ -303,6 +322,7 @@ struct RtpFecParameters {
>   RtpFecParameters();
>   explicit RtpFecParameters(FecMechanism mechanism);
>   RtpFecParameters(FecMechanism mechanism, uint32_t ssrc);
>+  RtpFecParameters(const RtpFecParameters&);
>   ~RtpFecParameters();
> 
>   bool operator==(const RtpFecParameters& o) const {
>@@ -314,11 +334,12 @@ struct RtpFecParameters {
> struct RtpRtxParameters {
>   // If unset, a value is chosen by the implementation.
>   // Works just like RtpEncodingParameters::ssrc.
>-  rtc::Optional<uint32_t> ssrc;
>+  absl::optional<uint32_t> ssrc;
> 
>   // Constructors for convenience.
>   RtpRtxParameters();
>   explicit RtpRtxParameters(uint32_t ssrc);
>+  RtpRtxParameters(const RtpRtxParameters&);
>   ~RtpRtxParameters();
> 
>   bool operator==(const RtpRtxParameters& o) const { return ssrc == o.ssrc; }
>@@ -327,6 +348,7 @@ struct RtpRtxParameters {
> 
> struct RtpEncodingParameters {
>   RtpEncodingParameters();
>+  RtpEncodingParameters(const RtpEncodingParameters&);
>   ~RtpEncodingParameters();
> 
>   // If unset, a value is chosen by the implementation.
>@@ -335,7 +357,7 @@ struct RtpEncodingParameters {
>   // may change due to an SSRC conflict, in which case the conflict is handled
>   // internally without any event. Another way of looking at this is that an
>   // unset SSRC acts as a "wildcard" SSRC.
>-  rtc::Optional<uint32_t> ssrc;
>+  absl::optional<uint32_t> ssrc;
> 
>   // Can be used to reference a codec in the |codecs| member of the
>   // RtpParameters that contains this RtpEncodingParameters. If unset, the
>@@ -343,32 +365,45 @@ struct RtpEncodingParameters {
>   // prepare to receive any codec (for a receiver).
>   // TODO(deadbeef): Not implemented. Implementation of RtpSender will always
>   // choose the first codec from the list.
>-  rtc::Optional<int> codec_payload_type;
>+  absl::optional<int> codec_payload_type;
> 
>   // Specifies the FEC mechanism, if set.
>   // TODO(deadbeef): Not implemented. Current implementation will use whatever
>   // FEC codecs are available, including red+ulpfec.
>-  rtc::Optional<RtpFecParameters> fec;
>+  absl::optional<RtpFecParameters> fec;
> 
>   // Specifies the RTX parameters, if set.
>   // TODO(deadbeef): Not implemented with PeerConnection senders/receivers.
>-  rtc::Optional<RtpRtxParameters> rtx;
>+  absl::optional<RtpRtxParameters> rtx;
> 
>   // Only used for audio. If set, determines whether or not discontinuous
>   // transmission will be used, if an available codec supports it. If not
>   // set, the implementation default setting will be used.
>   // TODO(deadbeef): Not implemented. Current implementation will use a CN
>   // codec as long as it's present.
>-  rtc::Optional<DtxStatus> dtx;
>+  absl::optional<DtxStatus> dtx;
> 
>   // The relative bitrate priority of this encoding. Currently this is
>-  // implemented on the sender level (using the first RtpEncodingParameters
>-  // of the rtp parameters).
>+  // implemented for the entire rtp sender by using the value of the first
>+  // encoding parameter.
>+  // TODO(webrtc.bugs.org/8630): Implement this per encoding parameter.
>+  // Currently there is logic for how bitrate is distributed per simulcast layer
>+  // in the VideoBitrateAllocator. This must be updated to incorporate relative
>+  // bitrate priority.
>   double bitrate_priority = kDefaultBitratePriority;
> 
>+  // Indicates the preferred duration of media represented by a packet in
>+  // milliseconds for this encoding. If set, this will take precedence over the
>+  // ptime set in the RtpCodecParameters. This could happen if SDP negotiation
>+  // creates a ptime for a specific codec, which is later changed in the
>+  // RtpEncodingParameters by the application.
>+  // TODO(bugs.webrtc.org/8819): Not implemented.
>+  absl::optional<int> ptime;
>+
>   // If set, this represents the Transport Independent Application Specific
>   // maximum bandwidth defined in RFC3890. If unset, there is no maximum
>-  // bitrate.
>+  // bitrate. Currently this is implemented for the entire rtp sender by using
>+  // the value of the first encoding parameter.
>   //
>   // Just called "maxBitrate" in ORTC spec.
>   //
>@@ -376,23 +411,29 @@ struct RtpEncodingParameters {
>   // bandwidth for the entire bandwidth estimator (audio and video). This is
>   // just always how "b=AS" was handled, but it's not correct and should be
>   // fixed.
>-  rtc::Optional<int> max_bitrate_bps;
>+  absl::optional<int> max_bitrate_bps;
>+
>+  // Specifies the minimum bitrate in bps for video.
>+  // TODO(asapersson): Not implemented for ORTC API.
>+  absl::optional<int> min_bitrate_bps;
> 
>   // TODO(deadbeef): Not implemented.
>-  rtc::Optional<int> max_framerate;
>+  absl::optional<int> max_framerate;
> 
>   // For video, scale the resolution down by this factor.
>   // TODO(deadbeef): Not implemented.
>-  double scale_resolution_down_by = 1.0;
>+  absl::optional<double> scale_resolution_down_by;
> 
>   // Scale the framerate down by this factor.
>   // TODO(deadbeef): Not implemented.
>-  double scale_framerate_down_by = 1.0;
>-
>-  // For an RtpSender, set to true to cause this encoding to be sent, and false
>-  // for it not to be sent. For an RtpReceiver, set to true to cause the
>-  // encoding to be decoded, and false for it to be ignored.
>-  // TODO(deadbeef): Not implemented for PeerConnection RtpReceivers.
>+  absl::optional<double> scale_framerate_down_by;
>+
>+  // For an RtpSender, set to true to cause this encoding to be encoded and
>+  // sent, and false for it not to be encoded and sent. This allows control
>+  // across multiple encodings of a sender for turning simulcast layers on and
>+  // off.
>+  // TODO(webrtc.bugs.org/8807): Updating this parameter will trigger an encoder
>+  // reset, but this isn't necessarily required.
>   bool active = true;
> 
>   // Value to use for RID RTP header extension.
>@@ -408,7 +449,7 @@ struct RtpEncodingParameters {
>   bool operator==(const RtpEncodingParameters& o) const {
>     return ssrc == o.ssrc && codec_payload_type == o.codec_payload_type &&
>            fec == o.fec && rtx == o.rtx && dtx == o.dtx &&
>-           bitrate_priority == o.bitrate_priority &&
>+           bitrate_priority == o.bitrate_priority && ptime == o.ptime &&
>            max_bitrate_bps == o.max_bitrate_bps &&
>            max_framerate == o.max_framerate &&
>            scale_resolution_down_by == o.scale_resolution_down_by &&
>@@ -423,6 +464,7 @@ struct RtpEncodingParameters {
> 
> struct RtpCodecParameters {
>   RtpCodecParameters();
>+  RtpCodecParameters(const RtpCodecParameters&);
>   ~RtpCodecParameters();
> 
>   // Build MIME "type/subtype" string from |name| and |kind|.
>@@ -440,24 +482,24 @@ struct RtpCodecParameters {
>   int payload_type = 0;
> 
>   // If unset, the implementation default is used.
>-  rtc::Optional<int> clock_rate;
>+  absl::optional<int> clock_rate;
> 
>   // The number of audio channels used. Unset for video codecs. If unset for
>   // audio, the implementation default is used.
>   // TODO(deadbeef): The "implementation default" part isn't fully implemented.
>   // Only defaults to 1, even though some codecs (such as opus) should really
>   // default to 2.
>-  rtc::Optional<int> num_channels;
>+  absl::optional<int> num_channels;
> 
>   // The maximum packetization time to be used by an RtpSender.
>   // If |ptime| is also set, this will be ignored.
>   // TODO(deadbeef): Not implemented.
>-  rtc::Optional<int> max_ptime;
>+  absl::optional<int> max_ptime;
> 
>   // The packetization time to be used by an RtpSender.
>   // If unset, will use any time up to max_ptime.
>   // TODO(deadbeef): Not implemented.
>-  rtc::Optional<int> ptime;
>+  absl::optional<int> ptime;
> 
>   // Feedback mechanisms to be used for this codec.
>   // TODO(deadbeef): Not implemented with PeerConnection senders/receivers.
>@@ -470,8 +512,6 @@ struct RtpCodecParameters {
>   // Contrary to ORTC, these parameters are named using all lowercase strings.
>   // This helps make the mapping to SDP simpler, if an application is using
>   // SDP. Boolean values are represented by the string "1".
>-  //
>-  // TODO(deadbeef): Not implemented with PeerConnection senders/receivers.
>   std::unordered_map<std::string, std::string> parameters;
> 
>   bool operator==(const RtpCodecParameters& o) const {
>@@ -508,17 +548,49 @@ struct RtpCapabilities {
>   bool operator!=(const RtpCapabilities& o) const { return !(*this == o); }
> };
> 
>-// Note that unlike in ORTC, an RtcpParameters structure is not included in
>-// RtpParameters, because our API includes an additional "RtpTransport"
>-// abstraction on which RTCP parameters are set.
>+struct RtcpParameters final {
>+  RtcpParameters();
>+  RtcpParameters(const RtcpParameters&);
>+  ~RtcpParameters();
>+
>+  // The SSRC to be used in the "SSRC of packet sender" field. If not set, one
>+  // will be chosen by the implementation.
>+  // TODO(deadbeef): Not implemented.
>+  absl::optional<uint32_t> ssrc;
>+
>+  // The Canonical Name (CNAME) used by RTCP (e.g. in SDES messages).
>+  //
>+  // If empty in the construction of the RtpTransport, one will be generated by
>+  // the implementation, and returned in GetRtcpParameters. Multiple
>+  // RtpTransports created by the same OrtcFactory will use the same generated
>+  // CNAME.
>+  //
>+  // If empty when passed into SetParameters, the CNAME simply won't be
>+  // modified.
>+  std::string cname;
>+
>+  // Send reduced-size RTCP?
>+  bool reduced_size = false;
>+
>+  // Send RTCP multiplexed on the RTP transport?
>+  // Not used with PeerConnection senders/receivers
>+  bool mux = true;
>+
>+  bool operator==(const RtcpParameters& o) const {
>+    return ssrc == o.ssrc && cname == o.cname &&
>+           reduced_size == o.reduced_size && mux == o.mux;
>+  }
>+  bool operator!=(const RtcpParameters& o) const { return !(*this == o); }
>+};
>+
> struct RtpParameters {
>   RtpParameters();
>+  RtpParameters(const RtpParameters&);
>   ~RtpParameters();
> 
>   // Used when calling getParameters/setParameters with a PeerConnection
>   // RtpSender, to ensure that outdated parameters are not unintentionally
>   // applied successfully.
>-  // TODO(deadbeef): Not implemented.
>   std::string transaction_id;
> 
>   // Value to use for MID RTP header extension.
>@@ -528,19 +600,25 @@ struct RtpParameters {
> 
>   std::vector<RtpCodecParameters> codecs;
> 
>-  // TODO(deadbeef): Not implemented with PeerConnection senders/receivers.
>   std::vector<RtpHeaderExtensionParameters> header_extensions;
> 
>   std::vector<RtpEncodingParameters> encodings;
> 
>-  // TODO(deadbeef): Not implemented.
>+  // Only available with a Peerconnection RtpSender.
>+  // In ORTC, our API includes an additional "RtpTransport"
>+  // abstraction on which RTCP parameters are set.
>+  RtcpParameters rtcp;
>+
>+  // When bandwidth is constrained and the RtpSender needs to choose between
>+  // degrading resolution or degrading framerate, degradationPreference
>+  // indicates which is preferred. Only for video tracks.
>   DegradationPreference degradation_preference =
>       DegradationPreference::BALANCED;
> 
>   bool operator==(const RtpParameters& o) const {
>     return mid == o.mid && codecs == o.codecs &&
>            header_extensions == o.header_extensions &&
>-           encodings == o.encodings &&
>+           encodings == o.encodings && rtcp == o.rtcp &&
>            degradation_preference == o.degradation_preference;
>   }
>   bool operator!=(const RtpParameters& o) const { return !(*this == o); }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtpreceiverinterface.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtpreceiverinterface.cc
>new file mode 100644
>index 00000000000..fcdca698927
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtpreceiverinterface.cc
>@@ -0,0 +1,48 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/rtpreceiverinterface.h"
>+
>+namespace webrtc {
>+
>+RtpSource::RtpSource(int64_t timestamp_ms,
>+                     uint32_t source_id,
>+                     RtpSourceType source_type)
>+    : timestamp_ms_(timestamp_ms),
>+      source_id_(source_id),
>+      source_type_(source_type) {}
>+
>+RtpSource::RtpSource(int64_t timestamp_ms,
>+                     uint32_t source_id,
>+                     RtpSourceType source_type,
>+                     uint8_t audio_level)
>+    : timestamp_ms_(timestamp_ms),
>+      source_id_(source_id),
>+      source_type_(source_type),
>+      audio_level_(audio_level) {}
>+
>+RtpSource::RtpSource(const RtpSource&) = default;
>+RtpSource& RtpSource::operator=(const RtpSource&) = default;
>+RtpSource::~RtpSource() = default;
>+
>+std::vector<std::string> RtpReceiverInterface::stream_ids() const {
>+  return {};
>+}
>+
>+std::vector<rtc::scoped_refptr<MediaStreamInterface>>
>+RtpReceiverInterface::streams() const {
>+  return {};
>+}
>+
>+std::vector<RtpSource> RtpReceiverInterface::GetSources() const {
>+  return {};
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtpreceiverinterface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtpreceiverinterface.h
>index ac2e090dad8..b2499b41699 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtpreceiverinterface.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtpreceiverinterface.h
>@@ -34,19 +34,16 @@ enum class RtpSourceType {
> class RtpSource {
>  public:
>   RtpSource() = delete;
>-  RtpSource(int64_t timestamp_ms, uint32_t source_id, RtpSourceType source_type)
>-      : timestamp_ms_(timestamp_ms),
>-        source_id_(source_id),
>-        source_type_(source_type) {}
>-
>+  RtpSource(int64_t timestamp_ms,
>+            uint32_t source_id,
>+            RtpSourceType source_type);
>   RtpSource(int64_t timestamp_ms,
>             uint32_t source_id,
>             RtpSourceType source_type,
>-            uint8_t audio_level)
>-      : timestamp_ms_(timestamp_ms),
>-        source_id_(source_id),
>-        source_type_(source_type),
>-        audio_level_(audio_level) {}
>+            uint8_t audio_level);
>+  RtpSource(const RtpSource&);
>+  RtpSource& operator=(const RtpSource&);
>+  ~RtpSource();
> 
>   int64_t timestamp_ms() const { return timestamp_ms_; }
>   void update_timestamp_ms(int64_t timestamp_ms) {
>@@ -60,8 +57,8 @@ class RtpSource {
>   // The source can be either a contributing source or a synchronization source.
>   RtpSourceType source_type() const { return source_type_; }
> 
>-  rtc::Optional<uint8_t> audio_level() const { return audio_level_; }
>-  void set_audio_level(const rtc::Optional<uint8_t>& level) {
>+  absl::optional<uint8_t> audio_level() const { return audio_level_; }
>+  void set_audio_level(const absl::optional<uint8_t>& level) {
>     audio_level_ = level;
>   }
> 
>@@ -74,7 +71,7 @@ class RtpSource {
>   int64_t timestamp_ms_;
>   uint32_t source_id_;
>   RtpSourceType source_type_;
>-  rtc::Optional<uint8_t> audio_level_;
>+  absl::optional<uint8_t> audio_level_;
> };
> 
> class RtpReceiverObserverInterface {
>@@ -96,12 +93,13 @@ class RtpReceiverInterface : public rtc::RefCountInterface {
>   virtual rtc::scoped_refptr<MediaStreamTrackInterface> track() const = 0;
>   // The list of streams that |track| is associated with. This is the same as
>   // the [[AssociatedRemoteMediaStreams]] internal slot in the spec.
>-  // https://w3c.github.io/webrtc-pc/#dfn-x%5B%5Bassociatedremotemediastreams%5D%5D
>+  // https://w3c.github.io/webrtc-pc/#dfn-associatedremotemediastreams
>   // TODO(hbos): Make pure virtual as soon as Chromium's mock implements this.
>-  virtual std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams()
>-      const {
>-    return std::vector<rtc::scoped_refptr<MediaStreamInterface>>();
>-  }
>+  // TODO(https://crbug.com/webrtc/9480): Remove streams() in favor of
>+  // stream_ids() as soon as downstream projects are no longer dependent on
>+  // stream objects.
>+  virtual std::vector<std::string> stream_ids() const;
>+  virtual std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams() const;
> 
>   // Audio or video receiver?
>   virtual cricket::MediaType media_type() const = 0;
>@@ -124,37 +122,27 @@ class RtpReceiverInterface : public rtc::RefCountInterface {
>   // TODO(zhihuang): Remove the default implementation once the subclasses
>   // implement this. Currently, the only relevant subclass is the
>   // content::FakeRtpReceiver in Chromium.
>-  virtual std::vector<RtpSource> GetSources() const {
>-    return std::vector<RtpSource>();
>-  }
>-  // TODO(hta): Remove default implementation or move function to
>-  // an internal interface. content::FakeRtpReceiver in Chromium needs this.
>-
>-  // Returns an ID that changes if the attached track changes, but
>-  // otherwise remains constant. Used to generate IDs for stats.
>-  // The special value zero means that no track is attached.
>-  virtual int AttachmentId() const { return 0; }
>+  virtual std::vector<RtpSource> GetSources() const;
> 
>  protected:
>-  virtual ~RtpReceiverInterface() {}
>+  ~RtpReceiverInterface() override = default;
> };
> 
> // Define proxy for RtpReceiverInterface.
> // TODO(deadbeef): Move this to .cc file and out of api/. What threads methods
> // are called on is an implementation detail.
> BEGIN_SIGNALING_PROXY_MAP(RtpReceiver)
>-  PROXY_SIGNALING_THREAD_DESTRUCTOR()
>-  PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track)
>-  PROXY_CONSTMETHOD0(std::vector<rtc::scoped_refptr<MediaStreamInterface>>,
>-                     streams)
>-  PROXY_CONSTMETHOD0(cricket::MediaType, media_type)
>-  PROXY_CONSTMETHOD0(std::string, id)
>-  PROXY_CONSTMETHOD0(RtpParameters, GetParameters);
>-  PROXY_METHOD1(bool, SetParameters, const RtpParameters&)
>-  PROXY_METHOD1(void, SetObserver, RtpReceiverObserverInterface*);
>-  PROXY_CONSTMETHOD0(std::vector<RtpSource>, GetSources);
>-  PROXY_CONSTMETHOD0(int, AttachmentId);
>-  END_PROXY_MAP()
>+PROXY_SIGNALING_THREAD_DESTRUCTOR()
>+PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track)
>+PROXY_CONSTMETHOD0(std::vector<rtc::scoped_refptr<MediaStreamInterface>>,
>+                   streams)
>+PROXY_CONSTMETHOD0(cricket::MediaType, media_type)
>+PROXY_CONSTMETHOD0(std::string, id)
>+PROXY_CONSTMETHOD0(RtpParameters, GetParameters);
>+PROXY_METHOD1(bool, SetParameters, const RtpParameters&)
>+PROXY_METHOD1(void, SetObserver, RtpReceiverObserverInterface*);
>+PROXY_CONSTMETHOD0(std::vector<RtpSource>, GetSources);
>+END_PROXY_MAP()
> 
> }  // namespace webrtc
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtpsenderinterface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtpsenderinterface.h
>index a7d7edc4b0b..a7eec4972e5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtpsenderinterface.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtpsenderinterface.h
>@@ -21,7 +21,9 @@
> #include "api/mediastreaminterface.h"
> #include "api/mediatypes.h"
> #include "api/proxy.h"
>+#include "api/rtcerror.h"
> #include "api/rtpparameters.h"
>+#include "rtc_base/deprecation.h"
> #include "rtc_base/refcount.h"
> #include "rtc_base/scoped_ref_ptr.h"
> 
>@@ -36,7 +38,7 @@ class RtpSenderInterface : public rtc::RefCountInterface {
> 
>   // Returns primary SSRC used by this sender for sending media.
>   // Returns 0 if not yet determined.
>-  // TODO(deadbeef): Change to rtc::Optional.
>+  // TODO(deadbeef): Change to absl::optional.
>   // TODO(deadbeef): Remove? With GetParameters this should be redundant.
>   virtual uint32_t ssrc() const = 0;
> 
>@@ -47,45 +49,39 @@ class RtpSenderInterface : public rtc::RefCountInterface {
>   // to uniquely identify a receiver until we implement Unified Plan SDP.
>   virtual std::string id() const = 0;
> 
>-  // Returns a list of streams associated with this sender's track. Although we
>-  // only support one track per stream, in theory the API allows for multiple.
>+  // Returns a list of media stream ids associated with this sender's track.
>+  // These are signalled in the SDP so that the remote side can associate
>+  // tracks.
>   virtual std::vector<std::string> stream_ids() const = 0;
> 
>-  virtual RtpParameters GetParameters() const = 0;
>+  virtual RtpParameters GetParameters() = 0;
>   // Note that only a subset of the parameters can currently be changed. See
>   // rtpparameters.h
>-  virtual bool SetParameters(const RtpParameters& parameters) = 0;
>+  // The encodings are in increasing quality order for simulcast.
>+  virtual RTCError SetParameters(const RtpParameters& parameters) = 0;
> 
>   // Returns null for a video sender.
>   virtual rtc::scoped_refptr<DtmfSenderInterface> GetDtmfSender() const = 0;
> 
>-  // Returns an ID that changes every time SetTrack() is called, but
>-  // otherwise remains constant. Used to generate IDs for stats.
>-  // The special value zero means that no track is attached.
>-  // TODO(hta): Remove default implementation when callers have updated,
>-  // or move function to an internal interface.
>-  virtual int AttachmentId() const { return 0; }
>-
>  protected:
>-  virtual ~RtpSenderInterface() {}
>+  ~RtpSenderInterface() override = default;
> };
> 
> // Define proxy for RtpSenderInterface.
> // TODO(deadbeef): Move this to .cc file and out of api/. What threads methods
> // are called on is an implementation detail.
> BEGIN_SIGNALING_PROXY_MAP(RtpSender)
>-  PROXY_SIGNALING_THREAD_DESTRUCTOR()
>-  PROXY_METHOD1(bool, SetTrack, MediaStreamTrackInterface*)
>-  PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track)
>-  PROXY_CONSTMETHOD0(uint32_t, ssrc)
>-  PROXY_CONSTMETHOD0(cricket::MediaType, media_type)
>-  PROXY_CONSTMETHOD0(std::string, id)
>-  PROXY_CONSTMETHOD0(std::vector<std::string>, stream_ids)
>-  PROXY_CONSTMETHOD0(RtpParameters, GetParameters);
>-  PROXY_METHOD1(bool, SetParameters, const RtpParameters&)
>-  PROXY_CONSTMETHOD0(rtc::scoped_refptr<DtmfSenderInterface>, GetDtmfSender);
>-  PROXY_CONSTMETHOD0(int, AttachmentId);
>-  END_PROXY_MAP()
>+PROXY_SIGNALING_THREAD_DESTRUCTOR()
>+PROXY_METHOD1(bool, SetTrack, MediaStreamTrackInterface*)
>+PROXY_CONSTMETHOD0(rtc::scoped_refptr<MediaStreamTrackInterface>, track)
>+PROXY_CONSTMETHOD0(uint32_t, ssrc)
>+PROXY_CONSTMETHOD0(cricket::MediaType, media_type)
>+PROXY_CONSTMETHOD0(std::string, id)
>+PROXY_CONSTMETHOD0(std::vector<std::string>, stream_ids)
>+PROXY_METHOD0(RtpParameters, GetParameters);
>+PROXY_METHOD1(RTCError, SetParameters, const RtpParameters&)
>+PROXY_CONSTMETHOD0(rtc::scoped_refptr<DtmfSenderInterface>, GetDtmfSender);
>+END_PROXY_MAP()
> 
> }  // namespace webrtc
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtptransceiverinterface.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtptransceiverinterface.cc
>new file mode 100644
>index 00000000000..065ac049778
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtptransceiverinterface.cc
>@@ -0,0 +1,26 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/rtptransceiverinterface.h"
>+
>+namespace webrtc {
>+
>+RtpTransceiverInit::RtpTransceiverInit() = default;
>+
>+RtpTransceiverInit::RtpTransceiverInit(const RtpTransceiverInit& rhs) = default;
>+
>+RtpTransceiverInit::~RtpTransceiverInit() = default;
>+
>+absl::optional<RtpTransceiverDirection>
>+RtpTransceiverInterface::fired_direction() const {
>+  return absl::nullopt;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtptransceiverinterface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtptransceiverinterface.h
>index 5d74ae1023c..4c22957b790 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtptransceiverinterface.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/rtptransceiverinterface.h
>@@ -14,7 +14,8 @@
> #include <string>
> #include <vector>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
>+#include "api/array_view.h"
> #include "api/rtpreceiverinterface.h"
> #include "api/rtpsenderinterface.h"
> #include "rtc_base/refcount.h"
>@@ -29,19 +30,18 @@ enum class RtpTransceiverDirection {
>   kInactive
> };
> 
>-// This is provided as a debugging aid. The format of the output is unspecified.
>-std::ostream& operator<<(std::ostream& os, RtpTransceiverDirection direction);
>-
> // Structure for initializing an RtpTransceiver in a call to
> // PeerConnectionInterface::AddTransceiver.
> // https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiverinit
> struct RtpTransceiverInit final {
>+  RtpTransceiverInit();
>+  RtpTransceiverInit(const RtpTransceiverInit&);
>+  ~RtpTransceiverInit();
>   // Direction of the RtpTransceiver. See RtpTransceiverInterface::direction().
>   RtpTransceiverDirection direction = RtpTransceiverDirection::kSendRecv;
> 
>   // The added RtpTransceiver will be added to these streams.
>-  // TODO(bugs.webrtc.org/7600): Not implemented.
>-  std::vector<std::string> stream_labels;
>+  std::vector<std::string> stream_ids;
> 
>   // TODO(bugs.webrtc.org/7600): Not implemented.
>   std::vector<RtpEncodingParameters> send_encodings;
>@@ -63,11 +63,15 @@ struct RtpTransceiverInit final {
> // https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver
> class RtpTransceiverInterface : public rtc::RefCountInterface {
>  public:
>+  // Media type of the transceiver. Any sender(s)/receiver(s) will have this
>+  // type as well.
>+  virtual cricket::MediaType media_type() const = 0;
>+
>   // The mid attribute is the mid negotiated and present in the local and
>   // remote descriptions. Before negotiation is complete, the mid value may be
>   // null. After rollbacks, the value may change from a non-null value to null.
>   // https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-mid
>-  virtual rtc::Optional<std::string> mid() const = 0;
>+  virtual absl::optional<std::string> mid() const = 0;
> 
>   // The sender attribute exposes the RtpSender corresponding to the RTP media
>   // that may be sent with the transceiver's mid. The sender is always present,
>@@ -104,7 +108,14 @@ class RtpTransceiverInterface : public rtc::RefCountInterface {
>   // for this transceiver. If this transceiver has never been represented in an
>   // offer/answer exchange, or if the transceiver is stopped, the value is null.
>   // https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-currentdirection
>-  virtual rtc::Optional<RtpTransceiverDirection> current_direction() const = 0;
>+  virtual absl::optional<RtpTransceiverDirection> current_direction() const = 0;
>+
>+  // An internal slot designating for which direction the relevant
>+  // PeerConnection events have been fired. This is to ensure that events like
>+  // OnAddTrack only get fired once even if the same session description is
>+  // applied again.
>+  // Exposed in the public interface for use by Chromium.
>+  virtual absl::optional<RtpTransceiverDirection> fired_direction() const;
> 
>   // The Stop method irreversibly stops the RtpTransceiver. The sender of this
>   // transceiver will no longer send, the receiver will no longer receive.
>@@ -119,7 +130,7 @@ class RtpTransceiverInterface : public rtc::RefCountInterface {
>       rtc::ArrayView<RtpCodecCapability> codecs) = 0;
> 
>  protected:
>-  virtual ~RtpTransceiverInterface() = default;
>+  ~RtpTransceiverInterface() override = default;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/stats/rtcstats.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/stats/rtcstats.h
>index 887d602042c..1705f6ab58e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/stats/rtcstats.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/stats/rtcstats.h
>@@ -78,7 +78,7 @@ class RTCStats {
> 
>   // Downcasts the stats object to an |RTCStats| subclass |T|. DCHECKs that the
>   // object is of type |T|.
>-  template<typename T>
>+  template <typename T>
>   const T& cast_to() const {
>     RTC_DCHECK_EQ(type(), T::kType);
>     return static_cast<const T&>(*this);
>@@ -90,8 +90,7 @@ class RTCStats {
>   // shall be reserved in the vector (so that subclasses can allocate a vector
>   // with room for both parent and child members without it having to resize).
>   virtual std::vector<const RTCStatsMemberInterface*>
>-  MembersOfThisObjectAndAncestors(
>-      size_t additional_capacity) const;
>+  MembersOfThisObjectAndAncestors(size_t additional_capacity) const;
> 
>   std::string const id_;
>   int64_t timestamp_us_;
>@@ -138,18 +137,18 @@ class RTCStats {
> //         bar("bar") {
> //   }
> //
>-#define WEBRTC_RTCSTATS_DECL()                                                 \
>- public:                                                                       \
>-  static const char kType[];                                                   \
>-                                                                               \
>-  std::unique_ptr<webrtc::RTCStats> copy() const override;                     \
>-  const char* type() const override;                                           \
>-                                                                               \
>- protected:                                                                    \
>-  std::vector<const webrtc::RTCStatsMemberInterface*>                          \
>-  MembersOfThisObjectAndAncestors(                                             \
>-      size_t local_var_additional_capacity) const override;                    \
>-                                                                               \
>+#define WEBRTC_RTCSTATS_DECL()                                          \
>+ public:                                                                \
>+  static const char kType[];                                            \
>+                                                                        \
>+  std::unique_ptr<webrtc::RTCStats> copy() const override;              \
>+  const char* type() const override;                                    \
>+                                                                        \
>+ protected:                                                             \
>+  std::vector<const webrtc::RTCStatsMemberInterface*>                   \
>+  MembersOfThisObjectAndAncestors(size_t local_var_additional_capacity) \
>+      const override;                                                   \
>+                                                                        \
>  public:
> 
> #define WEBRTC_RTCSTATS_IMPL(this_class, parent_class, type_str, ...)          \
>@@ -159,20 +158,17 @@ class RTCStats {
>     return std::unique_ptr<webrtc::RTCStats>(new this_class(*this));           \
>   }                                                                            \
>                                                                                \
>-  const char* this_class::type() const {                                       \
>-    return this_class::kType;                                                  \
>-  }                                                                            \
>+  const char* this_class::type() const { return this_class::kType; }           \
>                                                                                \
>   std::vector<const webrtc::RTCStatsMemberInterface*>                          \
>   this_class::MembersOfThisObjectAndAncestors(                                 \
>       size_t local_var_additional_capacity) const {                            \
>     const webrtc::RTCStatsMemberInterface* local_var_members[] = {             \
>-      __VA_ARGS__                                                              \
>-    };                                                                         \
>+        __VA_ARGS__};                                                          \
>     size_t local_var_members_count =                                           \
>         sizeof(local_var_members) / sizeof(local_var_members[0]);              \
>-    std::vector<const webrtc::RTCStatsMemberInterface*> local_var_members_vec =\
>-        parent_class::MembersOfThisObjectAndAncestors(                         \
>+    std::vector<const webrtc::RTCStatsMemberInterface*>                        \
>+        local_var_members_vec = parent_class::MembersOfThisObjectAndAncestors( \
>             local_var_members_count + local_var_additional_capacity);          \
>     RTC_DCHECK_GE(                                                             \
>         local_var_members_vec.capacity() - local_var_members_vec.size(),       \
>@@ -191,21 +187,21 @@ class RTCStatsMemberInterface {
>  public:
>   // Member value types.
>   enum Type {
>-    kBool,                  // bool
>-    kInt32,                 // int32_t
>-    kUint32,                // uint32_t
>-    kInt64,                 // int64_t
>-    kUint64,                // uint64_t
>-    kDouble,                // double
>-    kString,                // std::string
>+    kBool,    // bool
>+    kInt32,   // int32_t
>+    kUint32,  // uint32_t
>+    kInt64,   // int64_t
>+    kUint64,  // uint64_t
>+    kDouble,  // double
>+    kString,  // std::string
> 
>-    kSequenceBool,          // std::vector<bool>
>-    kSequenceInt32,         // std::vector<int32_t>
>-    kSequenceUint32,        // std::vector<uint32_t>
>-    kSequenceInt64,         // std::vector<int64_t>
>-    kSequenceUint64,        // std::vector<uint64_t>
>-    kSequenceDouble,        // std::vector<double>
>-    kSequenceString,        // std::vector<std::string>
>+    kSequenceBool,    // std::vector<bool>
>+    kSequenceInt32,   // std::vector<int32_t>
>+    kSequenceUint32,  // std::vector<uint32_t>
>+    kSequenceInt64,   // std::vector<int64_t>
>+    kSequenceUint64,  // std::vector<uint64_t>
>+    kSequenceDouble,  // std::vector<double>
>+    kSequenceString,  // std::vector<std::string>
>   };
> 
>   virtual ~RTCStatsMemberInterface() {}
>@@ -215,6 +211,9 @@ class RTCStatsMemberInterface {
>   virtual bool is_sequence() const = 0;
>   virtual bool is_string() const = 0;
>   bool is_defined() const { return is_defined_; }
>+  // Is this part of the stats spec? Used so that chromium can easily filter
>+  // out anything unstandardized.
>+  virtual bool is_standardized() const = 0;
>   // Type and value comparator. The names are not compared. These operators are
>   // exposed for testing.
>   virtual bool operator==(const RTCStatsMemberInterface& other) const = 0;
>@@ -229,7 +228,7 @@ class RTCStatsMemberInterface {
>   // instead.
>   virtual std::string ValueToJson() const = 0;
> 
>-  template<typename T>
>+  template <typename T>
>   const T& cast_to() const {
>     RTC_DCHECK_EQ(type(), T::kType);
>     return static_cast<const T&>(*this);
>@@ -247,19 +246,17 @@ class RTCStatsMemberInterface {
> // specialized in rtcstats.cc, using a different |T| results in a linker error
> // (undefined reference to |kType|). The supported types are the ones described
> // by |RTCStatsMemberInterface::Type|.
>-template<typename T>
>+template <typename T>
> class RTCStatsMember : public RTCStatsMemberInterface {
>  public:
>   static const Type kType;
> 
>   explicit RTCStatsMember(const char* name)
>-      : RTCStatsMemberInterface(name, false),
>-        value_() {}
>+      : RTCStatsMemberInterface(name, /*is_defined=*/false), value_() {}
>   RTCStatsMember(const char* name, const T& value)
>-      : RTCStatsMemberInterface(name, true),
>-        value_(value) {}
>+      : RTCStatsMemberInterface(name, /*is_defined=*/true), value_(value) {}
>   RTCStatsMember(const char* name, T&& value)
>-      : RTCStatsMemberInterface(name, true),
>+      : RTCStatsMemberInterface(name, /*is_defined=*/true),
>         value_(std::move(value)) {}
>   explicit RTCStatsMember(const RTCStatsMember<T>& other)
>       : RTCStatsMemberInterface(other.name_, other.is_defined_),
>@@ -271,8 +268,9 @@ class RTCStatsMember : public RTCStatsMemberInterface {
>   Type type() const override { return kType; }
>   bool is_sequence() const override;
>   bool is_string() const override;
>+  bool is_standardized() const override { return true; }
>   bool operator==(const RTCStatsMemberInterface& other) const override {
>-    if (type() != other.type())
>+    if (type() != other.type() || is_standardized() != other.is_standardized())
>       return false;
>     const RTCStatsMember<T>& other_t =
>         static_cast<const RTCStatsMember<T>&>(other);
>@@ -298,7 +296,10 @@ class RTCStatsMember : public RTCStatsMemberInterface {
>   }
>   T& operator=(const RTCStatsMember<T>& other) {
>     RTC_DCHECK(other.is_defined_);
>-    value_ = other.is_defined_;
>+    // Shouldn't be attempting to assign an RTCNonStandardStatsMember to an
>+    // RTCStatsMember or vice versa.
>+    RTC_DCHECK(is_standardized() == other.is_standardized());
>+    value_ = other.value_;
>     is_defined_ = true;
>     return value_;
>   }
>@@ -327,6 +328,26 @@ class RTCStatsMember : public RTCStatsMemberInterface {
>   T value_;
> };
> 
>+// Same as above, but "is_standardized" returns false.
>+//
>+// Using inheritance just so that it's obvious from the member's declaration
>+// whether it's standardized or not.
>+template <typename T>
>+class RTCNonStandardStatsMember : public RTCStatsMember<T> {
>+ public:
>+  explicit RTCNonStandardStatsMember(const char* name)
>+      : RTCStatsMember<T>(name) {}
>+  RTCNonStandardStatsMember(const char* name, const T& value)
>+      : RTCStatsMember<T>(name, value) {}
>+  RTCNonStandardStatsMember(const char* name, T&& value)
>+      : RTCStatsMember<T>(name, std::move(value)) {}
>+  explicit RTCNonStandardStatsMember(const RTCNonStandardStatsMember<T>& other)
>+      : RTCStatsMember<T>(other) {}
>+  explicit RTCNonStandardStatsMember(RTCNonStandardStatsMember<T>&& other)
>+      : RTCStatsMember<T>(std::move(other)) {}
>+
>+  bool is_standardized() const override { return false; }
>+};
> }  // namespace webrtc
> 
> #endif  // API_STATS_RTCSTATS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/stats/rtcstats_objects.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/stats/rtcstats_objects.h
>index a052c173834..7e6bcd3228b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/stats/rtcstats_objects.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/stats/rtcstats_objects.h
>@@ -11,6 +11,7 @@
> #ifndef API_STATS_RTCSTATS_OBJECTS_H_
> #define API_STATS_RTCSTATS_OBJECTS_H_
> 
>+#include <memory>
> #include <string>
> #include <vector>
> 
>@@ -183,6 +184,8 @@ class RTCIceCandidatePairStats final : public RTCStats {
> // TODO(hbos): |RTCStatsCollector| only collects candidates that are part of
> // ice candidate pairs, but there could be candidates not paired with anything.
> // crbug.com/632723
>+// TODO(qingsi): Add the stats of STUN binding requests (keepalives) and collect
>+// them in the new PeerConnection::GetStats.
> class RTCIceCandidateStats : public RTCStats {
>  public:
>   WEBRTC_RTCSTATS_DECL();
>@@ -206,8 +209,9 @@ class RTCIceCandidateStats : public RTCStats {
>   RTCStatsMember<bool> deleted;  // = false
> 
>  protected:
>-  RTCIceCandidateStats(
>-      const std::string& id, int64_t timestamp_us, bool is_remote);
>+  RTCIceCandidateStats(const std::string& id,
>+                       int64_t timestamp_us,
>+                       bool is_remote);
>   RTCIceCandidateStats(std::string&& id, int64_t timestamp_us, bool is_remote);
> };
> 
>@@ -215,11 +219,13 @@ class RTCIceCandidateStats : public RTCStats {
> // But here we define them as subclasses of |RTCIceCandidateStats| because the
> // |kType| need to be different ("RTCStatsType type") in the local/remote case.
> // https://w3c.github.io/webrtc-stats/#rtcstatstype-str*
>+// This forces us to have to override copy() and type().
> class RTCLocalIceCandidateStats final : public RTCIceCandidateStats {
>  public:
>   static const char kType[];
>   RTCLocalIceCandidateStats(const std::string& id, int64_t timestamp_us);
>   RTCLocalIceCandidateStats(std::string&& id, int64_t timestamp_us);
>+  std::unique_ptr<RTCStats> copy() const override;
>   const char* type() const override;
> };
> 
>@@ -228,6 +234,7 @@ class RTCRemoteIceCandidateStats final : public RTCIceCandidateStats {
>   static const char kType[];
>   RTCRemoteIceCandidateStats(const std::string& id, int64_t timestamp_us);
>   RTCRemoteIceCandidateStats(std::string&& id, int64_t timestamp_us);
>+  std::unique_ptr<RTCStats> copy() const override;
>   const char* type() const override;
> };
> 
>@@ -252,9 +259,11 @@ class RTCMediaStreamTrackStats final : public RTCStats {
>  public:
>   WEBRTC_RTCSTATS_DECL();
> 
>-  RTCMediaStreamTrackStats(const std::string& id, int64_t timestamp_us,
>+  RTCMediaStreamTrackStats(const std::string& id,
>+                           int64_t timestamp_us,
>                            const char* kind);
>-  RTCMediaStreamTrackStats(std::string&& id, int64_t timestamp_us,
>+  RTCMediaStreamTrackStats(std::string&& id,
>+                           int64_t timestamp_us,
>                            const char* kind);
>   RTCMediaStreamTrackStats(const RTCMediaStreamTrackStats& other);
>   ~RTCMediaStreamTrackStats() override;
>@@ -277,6 +286,7 @@ class RTCMediaStreamTrackStats final : public RTCStats {
>   // TODO(hbos): Not collected by |RTCStatsCollector|. crbug.com/659137
>   RTCStatsMember<double> frames_per_second;
>   RTCStatsMember<uint32_t> frames_sent;
>+  RTCStatsMember<uint32_t> huge_frames_sent;
>   RTCStatsMember<uint32_t> frames_received;
>   RTCStatsMember<uint32_t> frames_decoded;
>   RTCStatsMember<uint32_t> frames_dropped;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/stats/rtcstatscollectorcallback.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/stats/rtcstatscollectorcallback.h
>index 2c67bb81bbc..92422176593 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/stats/rtcstatscollectorcallback.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/stats/rtcstatscollectorcallback.h
>@@ -19,7 +19,7 @@ namespace webrtc {
> 
> class RTCStatsCollectorCallback : public virtual rtc::RefCountInterface {
>  public:
>-  virtual ~RTCStatsCollectorCallback() {}
>+  ~RTCStatsCollectorCallback() override = default;
> 
>   virtual void OnStatsDelivered(
>       const rtc::scoped_refptr<const RTCStatsReport>& report) = 0;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/stats/rtcstatsreport.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/stats/rtcstatsreport.h
>index ee56b35d062..f7410b39f79 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/stats/rtcstatsreport.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/stats/rtcstatsreport.h
>@@ -57,12 +57,16 @@ class RTCStatsReport : public rtc::RefCountInterface {
> 
>   explicit RTCStatsReport(int64_t timestamp_us);
>   RTCStatsReport(const RTCStatsReport& other) = delete;
>+  rtc::scoped_refptr<RTCStatsReport> Copy() const;
> 
>   int64_t timestamp_us() const { return timestamp_us_; }
>   void AddStats(std::unique_ptr<const RTCStats> stats);
>   const RTCStats* Get(const std::string& id) const;
>   size_t size() const { return stats_.size(); }
> 
>+  // Removes the stats object from the report, returning ownership of it or null
>+  // if there is no object with |id|.
>+  std::unique_ptr<const RTCStats> Take(const std::string& id);
>   // Takes ownership of all the stats in |victim|, leaving it empty.
>   void TakeMembersFrom(rtc::scoped_refptr<RTCStatsReport> victim);
> 
>@@ -72,7 +76,7 @@ class RTCStatsReport : public rtc::RefCountInterface {
> 
>   // Gets the subset of stats that are of type |T|, where |T| is any class
>   // descending from |RTCStats|.
>-  template<typename T>
>+  template <typename T>
>   std::vector<const T*> GetStatsOfType() const {
>     std::vector<const T*> stats_of_type;
>     for (const RTCStats& stats : *this) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/statstypes.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/statstypes.cc
>index d1637e3ffd1..ad68894f481 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/statstypes.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/statstypes.cc
>@@ -98,9 +98,8 @@ class TypedIntId : public StatsReport::IdBase {
>   }
> 
>   std::string ToString() const override {
>-    return std::string(InternalTypeToString(type_)) +
>-           kSeparator +
>-           rtc::ToString<int>(id_);
>+    return std::string(InternalTypeToString(type_)) + kSeparator +
>+           rtc::ToString(id_);
>   }
> 
>  protected:
>@@ -109,7 +108,8 @@ class TypedIntId : public StatsReport::IdBase {
> 
> class IdWithDirection : public TypedId {
>  public:
>-  IdWithDirection(StatsReport::StatsType type, const std::string& id,
>+  IdWithDirection(StatsReport::StatsType type,
>+                  const std::string& id,
>                   StatsReport::Direction direction)
>       : TypedId(type, id), direction_(direction) {}
> 
>@@ -132,45 +132,40 @@ class IdWithDirection : public TypedId {
> class CandidateId : public TypedId {
>  public:
>   CandidateId(bool local, const std::string& id)
>-      : TypedId(local ?
>-                    StatsReport::kStatsReportTypeIceLocalCandidate :
>-                    StatsReport::kStatsReportTypeIceRemoteCandidate,
>-                id) {
>-  }
>+      : TypedId(local ? StatsReport::kStatsReportTypeIceLocalCandidate
>+                      : StatsReport::kStatsReportTypeIceRemoteCandidate,
>+                id) {}
> 
>-  std::string ToString() const override {
>-    return "Cand-" + id_;
>-  }
>+  std::string ToString() const override { return "Cand-" + id_; }
> };
> 
> class ComponentId : public StatsReport::IdBase {
>  public:
>   ComponentId(const std::string& content_name, int component)
>-      : ComponentId(StatsReport::kStatsReportTypeComponent, content_name,
>-            component) {}
>+      : ComponentId(StatsReport::kStatsReportTypeComponent,
>+                    content_name,
>+                    component) {}
> 
>   bool Equals(const IdBase& other) const override {
>     return IdBase::Equals(other) &&
>-        static_cast<const ComponentId&>(other).component_ == component_ &&
>-        static_cast<const ComponentId&>(other).content_name_ == content_name_;
>+           static_cast<const ComponentId&>(other).component_ == component_ &&
>+           static_cast<const ComponentId&>(other).content_name_ ==
>+               content_name_;
>   }
> 
>-  std::string ToString() const override {
>-    return ToString("Channel-");
>-  }
>+  std::string ToString() const override { return ToString("Channel-"); }
> 
>  protected:
>-  ComponentId(StatsReport::StatsType type, const std::string& content_name,
>+  ComponentId(StatsReport::StatsType type,
>+              const std::string& content_name,
>               int component)
>-      : IdBase(type),
>-        content_name_(content_name),
>-        component_(component) {}
>+      : IdBase(type), content_name_(content_name), component_(component) {}
> 
>   std::string ToString(const char* prefix) const {
>     std::string ret(prefix);
>     ret += content_name_;
>     ret += '-';
>-    ret += rtc::ToString<>(component_);
>+    ret += rtc::ToString(component_);
>     return ret;
>   }
> 
>@@ -182,19 +177,20 @@ class ComponentId : public StatsReport::IdBase {
> class CandidatePairId : public ComponentId {
>  public:
>   CandidatePairId(const std::string& content_name, int component, int index)
>-      : ComponentId(StatsReport::kStatsReportTypeCandidatePair, content_name,
>-            component),
>+      : ComponentId(StatsReport::kStatsReportTypeCandidatePair,
>+                    content_name,
>+                    component),
>         index_(index) {}
> 
>   bool Equals(const IdBase& other) const override {
>     return ComponentId::Equals(other) &&
>-        static_cast<const CandidatePairId&>(other).index_ == index_;
>+           static_cast<const CandidatePairId&>(other).index_ == index_;
>   }
> 
>   std::string ToString() const override {
>     std::string ret(ComponentId::ToString("Conn-"));
>     ret += '-';
>-    ret += rtc::ToString<>(index_);
>+    ret += rtc::ToString(index_);
>     return ret;
>   }
> 
>@@ -207,7 +203,9 @@ class CandidatePairId : public ComponentId {
> StatsReport::IdBase::IdBase(StatsType type) : type_(type) {}
> StatsReport::IdBase::~IdBase() {}
> 
>-StatsReport::StatsType StatsReport::IdBase::type() const { return type_; }
>+StatsReport::StatsType StatsReport::IdBase::type() const {
>+  return type_;
>+}
> 
> bool StatsReport::IdBase::Equals(const IdBase& other) const {
>   return other.type_ == type_;
>@@ -316,8 +314,8 @@ bool StatsReport::Value::operator==(const char* value) const {
> }
> 
> bool StatsReport::Value::operator==(int64_t value) const {
>-  return type_ == kInt ? value_.int_ == static_cast<int>(value) :
>-      (type_ == kInt64 ? value_.int64_ == value : false);
>+  return type_ == kInt ? value_.int_ == static_cast<int>(value)
>+                       : (type_ == kInt64 ? value_.int64_ == value : false);
> }
> 
> bool StatsReport::Value::operator==(bool value) const {
>@@ -430,7 +428,9 @@ const char* StatsReport::Value::display_name() const {
>     case kStatsValueNameBandwidthLimitedResolution:
>       return "googBandwidthLimitedResolution";
>     // STUN ping related attributes.
>+    //
>     // TODO(zhihuang) Rename these stats to follow the standards.
>+    // Connectivity checks.
>     case kStatsValueNameSentPingRequestsTotal:
>       return "requestsSent";
>     case kStatsValueNameSentPingRequestsBeforeFirstResponse:
>@@ -441,6 +441,15 @@ const char* StatsReport::Value::display_name() const {
>       return "requestsReceived";
>     case kStatsValueNameRecvPingResponses:
>       return "responsesReceived";
>+    // STUN Keepalive pings.
>+    case kStatsValueNameSentStunKeepaliveRequests:
>+      return "stunKeepaliveRequestsSent";
>+    case kStatsValueNameRecvStunKeepaliveResponses:
>+      return "stunKeepaliveResponsesReceived";
>+    case kStatsValueNameStunKeepaliveRttTotal:
>+      return "stunKeepaliveRttTotal";
>+    case kStatsValueNameStunKeepaliveRttSquaredTotal:
>+      return "stunKeepaliveRttSquaredTotal";
> 
>     // Candidate related attributes. Values are taken from
>     // http://w3c.github.io/webrtc-stats/#rtcstatstype-enum*.
>@@ -547,6 +556,8 @@ const char* StatsReport::Value::display_name() const {
>       return "googFrameWidthSent";
>     case kStatsValueNameHasEnteredLowResolution:
>       return "googHasEnteredLowResolution";
>+    case kStatsValueNameHugeFramesSent:
>+      return "hugeFramesSent";
>     case kStatsValueNameInitiator:
>       return "googInitiator";
>     case kStatsValueNameInterframeDelayMaxMs:
>@@ -686,7 +697,9 @@ StatsReport::Id StatsReport::NewTypedIntId(StatsType type, int id) {
> 
> // static
> StatsReport::Id StatsReport::NewIdWithDirection(
>-    StatsType type, const std::string& id, StatsReport::Direction direction) {
>+    StatsType type,
>+    const std::string& id,
>+    StatsReport::Direction direction) {
>   return Id(new RefCountedObject<IdWithDirection>(type, id, direction));
> }
> 
>@@ -696,16 +709,17 @@ StatsReport::Id StatsReport::NewCandidateId(bool local, const std::string& id) {
> }
> 
> // static
>-StatsReport::Id StatsReport::NewComponentId(
>-    const std::string& content_name, int component) {
>+StatsReport::Id StatsReport::NewComponentId(const std::string& content_name,
>+                                            int component) {
>   return Id(new RefCountedObject<ComponentId>(content_name, component));
> }
> 
> // static
>-StatsReport::Id StatsReport::NewCandidatePairId(
>-    const std::string& content_name, int component, int index) {
>-  return Id(new RefCountedObject<CandidatePairId>(
>-      content_name, component, index));
>+StatsReport::Id StatsReport::NewCandidatePairId(const std::string& content_name,
>+                                                int component,
>+                                                int index) {
>+  return Id(
>+      new RefCountedObject<CandidatePairId>(content_name, component, index));
> }
> 
> const char* StatsReport::TypeToString() const {
>@@ -750,8 +764,7 @@ void StatsReport::AddBoolean(StatsReport::StatsValueName name, bool value) {
>     values_[name] = ValuePtr(new Value(name, value));
> }
> 
>-void StatsReport::AddId(StatsReport::StatsValueName name,
>-                        const Id& value) {
>+void StatsReport::AddId(StatsReport::StatsValueName name, const Id& value) {
>   const Value* found = FindValue(name);
>   if (!found || !(*found == value))
>     values_[name] = ValuePtr(new Value(name, value));
>@@ -762,8 +775,7 @@ const StatsReport::Value* StatsReport::FindValue(StatsValueName name) const {
>   return it == values_.end() ? nullptr : it->second.get();
> }
> 
>-StatsCollection::StatsCollection() {
>-}
>+StatsCollection::StatsCollection() {}
> 
> StatsCollection::~StatsCollection() {
>   RTC_DCHECK(thread_checker_.CalledOnValidThread());
>@@ -803,8 +815,9 @@ StatsReport* StatsCollection::FindOrAddNew(const StatsReport::Id& id) {
> StatsReport* StatsCollection::ReplaceOrAddNew(const StatsReport::Id& id) {
>   RTC_DCHECK(thread_checker_.CalledOnValidThread());
>   RTC_DCHECK(id.get());
>-  Container::iterator it = std::find_if(list_.begin(), list_.end(),
>-      [&id](const StatsReport* r)->bool { return r->id()->Equals(id); });
>+  Container::iterator it = std::find_if(
>+      list_.begin(), list_.end(),
>+      [&id](const StatsReport* r) -> bool { return r->id()->Equals(id); });
>   if (it != end()) {
>     StatsReport* report = new StatsReport((*it)->id());
>     delete *it;
>@@ -818,8 +831,9 @@ StatsReport* StatsCollection::ReplaceOrAddNew(const StatsReport::Id& id) {
> // will be returned.
> StatsReport* StatsCollection::Find(const StatsReport::Id& id) {
>   RTC_DCHECK(thread_checker_.CalledOnValidThread());
>-  Container::iterator it = std::find_if(list_.begin(), list_.end(),
>-      [&id](const StatsReport* r)->bool { return r->id()->Equals(id); });
>+  Container::iterator it = std::find_if(
>+      list_.begin(), list_.end(),
>+      [&id](const StatsReport* r) -> bool { return r->id()->Equals(id); });
>   return it == list_.end() ? nullptr : *it;
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/statstypes.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/statstypes.h
>index a034017458e..857b1afaa7b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/statstypes.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/statstypes.h
>@@ -20,7 +20,6 @@
> #include <string>
> #include <vector>
> 
>-#include "rtc_base/basictypes.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/refcount.h"
> #include "rtc_base/scoped_ref_ptr.h"
>@@ -129,6 +128,10 @@ class StatsReport {
>     kStatsValueNameSentPingResponses,
>     kStatsValueNameRecvPingRequests,
>     kStatsValueNameRecvPingResponses,
>+    kStatsValueNameSentStunKeepaliveRequests,
>+    kStatsValueNameRecvStunKeepaliveResponses,
>+    kStatsValueNameStunKeepaliveRttTotal,
>+    kStatsValueNameStunKeepaliveRttSquaredTotal,
> 
>     // Internal StatsValue names.
>     kStatsValueNameAccelerateRate,
>@@ -185,6 +188,7 @@ class StatsReport {
>     kStatsValueNameFrameWidthReceived,
>     kStatsValueNameFrameWidthSent,
>     kStatsValueNameHasEnteredLowResolution,
>+    kStatsValueNameHugeFramesSent,
>     kStatsValueNameInitiator,
>     kStatsValueNameInterframeDelayMaxMs,  // Max over last 10 seconds.
>     kStatsValueNameIssuerId,
>@@ -337,7 +341,7 @@ class StatsReport {
> 
>    private:
>     rtc::ThreadChecker thread_checker_;
>-    mutable int ref_count_ RTC_ACCESS_ON(thread_checker_) = 0;
>+    mutable int ref_count_ RTC_GUARDED_BY(thread_checker_) = 0;
> 
>     const Type type_;
>     // TODO(tommi): Use C++ 11 union and make value_ const.
>@@ -365,13 +369,14 @@ class StatsReport {
>   static Id NewBandwidthEstimationId();
>   static Id NewTypedId(StatsType type, const std::string& id);
>   static Id NewTypedIntId(StatsType type, int id);
>-  static Id NewIdWithDirection(
>-      StatsType type, const std::string& id, Direction direction);
>+  static Id NewIdWithDirection(StatsType type,
>+                               const std::string& id,
>+                               Direction direction);
>   static Id NewCandidateId(bool local, const std::string& id);
>-  static Id NewComponentId(
>-      const std::string& content_name, int component);
>-  static Id NewCandidatePairId(
>-      const std::string& content_name, int component, int index);
>+  static Id NewComponentId(const std::string& content_name, int component);
>+  static Id NewCandidatePairId(const std::string& content_name,
>+                               int component,
>+                               int index);
> 
>   const Id& id() const { return id_; }
>   StatsType type() const { return id_->type(); }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/DEPS b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/DEPS
>new file mode 100644
>index 00000000000..98b1ad391fc
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/DEPS
>@@ -0,0 +1,8 @@
>+specific_include_rules = {
>+  ".*": [
>+    "+modules/video_coding",
>+  ],
>+  ".*": [
>+    "+video"
>+  ],
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/audioproc_float.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/audioproc_float.cc
>new file mode 100644
>index 00000000000..9d3ad7e9d88
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/audioproc_float.cc
>@@ -0,0 +1,27 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/test/audioproc_float.h"
>+
>+#include <utility>
>+
>+#include "modules/audio_processing/test/audioproc_float_impl.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+int AudioprocFloat(std::unique_ptr<AudioProcessingBuilder> ap_builder,
>+                   int argc,
>+                   char* argv[]) {
>+  return AudioprocFloatImpl(std::move(ap_builder), argc, argv);
>+}
>+
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/audioproc_float.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/audioproc_float.h
>new file mode 100644
>index 00000000000..25e4dd5b1cd
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/audioproc_float.h
>@@ -0,0 +1,42 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_TEST_AUDIOPROC_FLOAT_H_
>+#define API_TEST_AUDIOPROC_FLOAT_H_
>+
>+#include <memory>
>+
>+#include "modules/audio_processing/include/audio_processing.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+// This is an interface for the audio processing simulation utility. This
>+// utility can be used to simulate the audioprocessing module using a recording
>+// (either an AEC dump or wav files), and generate the output as a wav file.
>+// The |ap_builder| object will be used to create the AudioProcessing instance
>+// that is used during the simulation. The |ap_builder| supports setting of
>+// injectable components, which will be passed on to the created AudioProcessing
>+// instance. It is needed to pass the command line flags as |argc| and |argv|,
>+// so these can be interpreted properly by the utility.
>+// To get a fully-working audioproc_f utility, all that is needed is to write a
>+// main function, create an AudioProcessingBuilder, optionally set custom
>+// processing components on it, and pass the builder together with the command
>+// line arguments into this function.
>+// To see a list of all supported command line flags, run the executable with
>+// the '--help' flag.
>+int AudioprocFloat(std::unique_ptr<AudioProcessingBuilder> ap_builder,
>+                   int argc,
>+                   char* argv[]);
>+
>+}  // namespace test
>+}  // namespace webrtc
>+
>+#endif  // API_TEST_AUDIOPROC_FLOAT_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/create_simulcast_test_fixture.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/create_simulcast_test_fixture.cc
>new file mode 100644
>index 00000000000..897746d00d2
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/create_simulcast_test_fixture.cc
>@@ -0,0 +1,32 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/test/create_simulcast_test_fixture.h"
>+
>+#include <memory>
>+#include <utility>
>+
>+#include "absl/memory/memory.h"
>+#include "api/test/simulcast_test_fixture.h"
>+#include "modules/video_coding/utility/simulcast_test_fixture_impl.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+std::unique_ptr<SimulcastTestFixture> CreateSimulcastTestFixture(
>+    std::unique_ptr<VideoEncoderFactory> encoder_factory,
>+    std::unique_ptr<VideoDecoderFactory> decoder_factory,
>+    SdpVideoFormat video_format) {
>+  return absl::make_unique<SimulcastTestFixtureImpl>(
>+      std::move(encoder_factory), std::move(decoder_factory), video_format);
>+}
>+
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/create_simulcast_test_fixture.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/create_simulcast_test_fixture.h
>new file mode 100644
>index 00000000000..87f229c0092
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/create_simulcast_test_fixture.h
>@@ -0,0 +1,32 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_TEST_CREATE_SIMULCAST_TEST_FIXTURE_H_
>+#define API_TEST_CREATE_SIMULCAST_TEST_FIXTURE_H_
>+
>+#include <memory>
>+
>+#include "api/test/simulcast_test_fixture.h"
>+#include "api/video_codecs/sdp_video_format.h"
>+#include "api/video_codecs/video_decoder_factory.h"
>+#include "api/video_codecs/video_encoder_factory.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+std::unique_ptr<SimulcastTestFixture> CreateSimulcastTestFixture(
>+    std::unique_ptr<VideoEncoderFactory> encoder_factory,
>+    std::unique_ptr<VideoDecoderFactory> decoder_factory,
>+    SdpVideoFormat video_format);
>+
>+}  // namespace test
>+}  // namespace webrtc
>+
>+#endif  // API_TEST_CREATE_SIMULCAST_TEST_FIXTURE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/create_video_quality_test_fixture.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/create_video_quality_test_fixture.cc
>new file mode 100644
>index 00000000000..994401b65bc
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/create_video_quality_test_fixture.cc
>@@ -0,0 +1,34 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <memory>
>+#include <utility>
>+
>+#include "absl/memory/memory.h"
>+#include "api/test/create_video_quality_test_fixture.h"
>+#include "video/video_quality_test.h"
>+
>+namespace webrtc {
>+
>+std::unique_ptr<VideoQualityTestFixtureInterface>
>+CreateVideoQualityTestFixture() {
>+  // By default, we don't override the FEC module, so pass an empty factory.
>+  return absl::make_unique<VideoQualityTest>(nullptr);
>+}
>+
>+std::unique_ptr<VideoQualityTestFixtureInterface>
>+CreateVideoQualityTestFixture(
>+    std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory) {
>+  return absl::make_unique<VideoQualityTest>(std::move(fec_controller_factory));
>+}
>+
>+}  // namespace webrtc
>+
>+
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/create_video_quality_test_fixture.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/create_video_quality_test_fixture.h
>new file mode 100644
>index 00000000000..07c222f16ae
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/create_video_quality_test_fixture.h
>@@ -0,0 +1,29 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#ifndef API_TEST_CREATE_VIDEO_QUALITY_TEST_FIXTURE_H_
>+#define API_TEST_CREATE_VIDEO_QUALITY_TEST_FIXTURE_H_
>+
>+#include <memory>
>+
>+#include "api/fec_controller.h"
>+#include "api/test/video_quality_test_fixture.h"
>+
>+namespace webrtc {
>+
>+std::unique_ptr<VideoQualityTestFixtureInterface>
>+CreateVideoQualityTestFixture();
>+
>+std::unique_ptr<VideoQualityTestFixtureInterface>
>+CreateVideoQualityTestFixture(
>+    std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory);
>+
>+}
>+
>+#endif  // API_TEST_CREATE_VIDEO_QUALITY_TEST_FIXTURE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/create_videocodec_test_fixture.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/create_videocodec_test_fixture.cc
>new file mode 100644
>index 00000000000..df8ad8b35fc
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/create_videocodec_test_fixture.cc
>@@ -0,0 +1,39 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/test/create_videocodec_test_fixture.h"
>+
>+#include <memory>
>+#include <utility>
>+
>+#include "absl/memory/memory.h"
>+#include "api/test/videocodec_test_fixture.h"
>+#include "modules/video_coding/codecs/test/videocodec_test_fixture_impl.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+using Config = VideoCodecTestFixture::Config;
>+
>+std::unique_ptr<VideoCodecTestFixture> CreateVideoCodecTestFixture(
>+    const Config& config) {
>+  return absl::make_unique<VideoCodecTestFixtureImpl>(config);
>+}
>+
>+std::unique_ptr<VideoCodecTestFixture> CreateVideoCodecTestFixture(
>+    const Config& config,
>+    std::unique_ptr<VideoDecoderFactory> decoder_factory,
>+    std::unique_ptr<VideoEncoderFactory> encoder_factory) {
>+  return absl::make_unique<VideoCodecTestFixtureImpl>(
>+      config, std::move(decoder_factory), std::move(encoder_factory));
>+}
>+
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/create_videocodec_test_fixture.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/create_videocodec_test_fixture.h
>new file mode 100644
>index 00000000000..7a44f6b0580
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/create_videocodec_test_fixture.h
>@@ -0,0 +1,34 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_TEST_CREATE_VIDEOCODEC_TEST_FIXTURE_H_
>+#define API_TEST_CREATE_VIDEOCODEC_TEST_FIXTURE_H_
>+
>+#include <memory>
>+
>+#include "api/test/videocodec_test_fixture.h"
>+#include "api/video_codecs/video_decoder_factory.h"
>+#include "api/video_codecs/video_encoder_factory.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+std::unique_ptr<VideoCodecTestFixture> CreateVideoCodecTestFixture(
>+    const VideoCodecTestFixture::Config& config);
>+
>+std::unique_ptr<VideoCodecTestFixture> CreateVideoCodecTestFixture(
>+    const VideoCodecTestFixture::Config& config,
>+    std::unique_ptr<VideoDecoderFactory> decoder_factory,
>+    std::unique_ptr<VideoEncoderFactory> encoder_factory);
>+
>+}  // namespace test
>+}  // namespace webrtc
>+
>+#endif  // API_TEST_CREATE_VIDEOCODEC_TEST_FIXTURE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/fakeconstraints.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/fakeconstraints.h
>index 2010400aa4e..d9c767de335 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/fakeconstraints.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/fakeconstraints.h
>@@ -21,20 +21,16 @@ namespace webrtc {
> 
> class FakeConstraints : public webrtc::MediaConstraintsInterface {
>  public:
>-  FakeConstraints() { }
>-  virtual ~FakeConstraints() { }
>+  FakeConstraints() {}
>+  virtual ~FakeConstraints() {}
> 
>-  virtual const Constraints& GetMandatory() const {
>-    return mandatory_;
>-  }
>+  virtual const Constraints& GetMandatory() const { return mandatory_; }
> 
>-  virtual const Constraints& GetOptional() const {
>-    return optional_;
>-  }
>+  virtual const Constraints& GetOptional() const { return optional_; }
> 
>   template <class T>
>   void AddMandatory(const std::string& key, const T& value) {
>-    mandatory_.push_back(Constraint(key, rtc::ToString<T>(value)));
>+    mandatory_.push_back(Constraint(key, rtc::ToString(value)));
>   }
> 
>   template <class T>
>@@ -49,12 +45,12 @@ class FakeConstraints : public webrtc::MediaConstraintsInterface {
>         }
>       }
>     }
>-    mandatory_.push_back(Constraint(key, rtc::ToString<T>(value)));
>+    mandatory_.push_back(Constraint(key, rtc::ToString(value)));
>   }
> 
>   template <class T>
>   void AddOptional(const std::string& key, const T& value) {
>-    optional_.push_back(Constraint(key, rtc::ToString<T>(value)));
>+    optional_.push_back(Constraint(key, rtc::ToString(value)));
>   }
> 
>   void SetMandatoryMinAspectRatio(double ratio) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/mock_peerconnectioninterface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/mock_peerconnectioninterface.h
>new file mode 100644
>index 00000000000..58ca741033e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/mock_peerconnectioninterface.h
>@@ -0,0 +1,139 @@
>+/*
>+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_TEST_MOCK_PEERCONNECTIONINTERFACE_H_
>+#define API_TEST_MOCK_PEERCONNECTIONINTERFACE_H_
>+
>+#include <memory>
>+#include <string>
>+#include <utility>
>+#include <vector>
>+
>+#include "api/peerconnectioninterface.h"
>+#include "test/gmock.h"
>+
>+namespace webrtc {
>+
>+class MockPeerConnectionInterface
>+    : public rtc::RefCountedObject<webrtc::PeerConnectionInterface> {
>+ public:
>+  // PeerConnectionInterface
>+  MOCK_METHOD0(local_streams, rtc::scoped_refptr<StreamCollectionInterface>());
>+  MOCK_METHOD0(remote_streams, rtc::scoped_refptr<StreamCollectionInterface>());
>+  MOCK_METHOD1(AddStream, bool(MediaStreamInterface*));
>+  MOCK_METHOD1(RemoveStream, void(MediaStreamInterface*));
>+  MOCK_METHOD2(AddTrack,
>+               RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>>(
>+                   rtc::scoped_refptr<MediaStreamTrackInterface>,
>+                   const std::vector<std::string>&));
>+  MOCK_METHOD2(AddTrack,
>+               rtc::scoped_refptr<RtpSenderInterface>(
>+                   MediaStreamTrackInterface*,
>+                   std::vector<MediaStreamInterface*>));
>+  MOCK_METHOD1(RemoveTrack, bool(RtpSenderInterface*));
>+  MOCK_METHOD1(AddTransceiver,
>+               RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>(
>+                   rtc::scoped_refptr<MediaStreamTrackInterface>));
>+  MOCK_METHOD2(AddTransceiver,
>+               RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>(
>+                   rtc::scoped_refptr<MediaStreamTrackInterface>,
>+                   const RtpTransceiverInit&));
>+  MOCK_METHOD1(AddTransceiver,
>+               RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>(
>+                   cricket::MediaType));
>+  MOCK_METHOD2(AddTransceiver,
>+               RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>(
>+                   cricket::MediaType,
>+                   const RtpTransceiverInit&));
>+  MOCK_METHOD2(CreateSender,
>+               rtc::scoped_refptr<RtpSenderInterface>(const std::string&,
>+                                                      const std::string&));
>+  MOCK_CONST_METHOD0(GetSenders,
>+                     std::vector<rtc::scoped_refptr<RtpSenderInterface>>());
>+  MOCK_CONST_METHOD0(GetReceivers,
>+                     std::vector<rtc::scoped_refptr<RtpReceiverInterface>>());
>+  MOCK_CONST_METHOD0(
>+      GetTransceivers,
>+      std::vector<rtc::scoped_refptr<RtpTransceiverInterface>>());
>+  MOCK_METHOD3(GetStats,
>+               bool(StatsObserver*,
>+                    MediaStreamTrackInterface*,
>+                    StatsOutputLevel));
>+  MOCK_METHOD1(GetStats, void(RTCStatsCollectorCallback*));
>+  MOCK_METHOD2(GetStats,
>+               void(rtc::scoped_refptr<RtpSenderInterface>,
>+                    rtc::scoped_refptr<RTCStatsCollectorCallback>));
>+  MOCK_METHOD2(GetStats,
>+               void(rtc::scoped_refptr<RtpReceiverInterface>,
>+                    rtc::scoped_refptr<RTCStatsCollectorCallback>));
>+  MOCK_METHOD0(ClearStatsCache, void());
>+  MOCK_METHOD2(
>+      CreateDataChannel,
>+      rtc::scoped_refptr<DataChannelInterface>(const std::string&,
>+                                               const DataChannelInit*));
>+  MOCK_CONST_METHOD0(local_description, const SessionDescriptionInterface*());
>+  MOCK_CONST_METHOD0(remote_description, const SessionDescriptionInterface*());
>+  MOCK_CONST_METHOD0(current_local_description,
>+                     const SessionDescriptionInterface*());
>+  MOCK_CONST_METHOD0(current_remote_description,
>+                     const SessionDescriptionInterface*());
>+  MOCK_CONST_METHOD0(pending_local_description,
>+                     const SessionDescriptionInterface*());
>+  MOCK_CONST_METHOD0(pending_remote_description,
>+                     const SessionDescriptionInterface*());
>+  MOCK_METHOD2(CreateOffer,
>+               void(CreateSessionDescriptionObserver*,
>+                    const MediaConstraintsInterface*));
>+  MOCK_METHOD2(CreateOffer,
>+               void(CreateSessionDescriptionObserver*,
>+                    const RTCOfferAnswerOptions&));
>+  MOCK_METHOD2(CreateAnswer,
>+               void(CreateSessionDescriptionObserver*,
>+                    const RTCOfferAnswerOptions&));
>+  MOCK_METHOD2(CreateAnswer,
>+               void(CreateSessionDescriptionObserver*,
>+                    const MediaConstraintsInterface*));
>+  MOCK_METHOD2(SetLocalDescription,
>+               void(SetSessionDescriptionObserver*,
>+                    SessionDescriptionInterface*));
>+  MOCK_METHOD2(SetRemoteDescription,
>+               void(SetSessionDescriptionObserver*,
>+                    SessionDescriptionInterface*));
>+  MOCK_METHOD2(SetRemoteDescription,
>+               void(std::unique_ptr<SessionDescriptionInterface>,
>+                    rtc::scoped_refptr<SetRemoteDescriptionObserverInterface>));
>+  MOCK_METHOD0(GetConfiguration, PeerConnectionInterface::RTCConfiguration());
>+  MOCK_METHOD2(SetConfiguration,
>+               bool(const PeerConnectionInterface::RTCConfiguration&,
>+                    RTCError*));
>+  MOCK_METHOD1(SetConfiguration,
>+               bool(const PeerConnectionInterface::RTCConfiguration&));
>+  MOCK_METHOD1(AddIceCandidate, bool(const IceCandidateInterface*));
>+  MOCK_METHOD1(RemoveIceCandidates,
>+               bool(const std::vector<cricket::Candidate>&));
>+  MOCK_METHOD1(SetBitrate, RTCError(const BitrateSettings&));
>+  MOCK_METHOD1(SetBitrate, RTCError(const BitrateParameters&));
>+  MOCK_METHOD1(SetBitrateAllocationStrategy,
>+               void(std::unique_ptr<rtc::BitrateAllocationStrategy>));
>+  MOCK_METHOD1(SetAudioPlayout, void(bool));
>+  MOCK_METHOD1(SetAudioRecording, void(bool));
>+  MOCK_METHOD0(signaling_state, SignalingState());
>+  MOCK_METHOD0(ice_connection_state, IceConnectionState());
>+  MOCK_METHOD0(ice_gathering_state, IceGatheringState());
>+  MOCK_METHOD2(StartRtcEventLog, bool(rtc::PlatformFile, int64_t));
>+  MOCK_METHOD2(StartRtcEventLog,
>+               bool(std::unique_ptr<RtcEventLogOutput>, int64_t));
>+  MOCK_METHOD0(StopRtcEventLog, void());
>+  MOCK_METHOD0(Close, void());
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // API_TEST_MOCK_PEERCONNECTIONINTERFACE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/mock_rtpreceiver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/mock_rtpreceiver.h
>index 7a612004ba2..de69ceaccdb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/mock_rtpreceiver.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/mock_rtpreceiver.h
>@@ -31,7 +31,6 @@ class MockRtpReceiver : public rtc::RefCountedObject<RtpReceiverInterface> {
>   MOCK_METHOD1(SetParameters, bool(const RtpParameters&));
>   MOCK_METHOD1(SetObserver, void(RtpReceiverObserverInterface*));
>   MOCK_CONST_METHOD0(GetSources, std::vector<RtpSource>());
>-  MOCK_CONST_METHOD0(AttachmentId, int());
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/mock_rtpsender.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/mock_rtpsender.h
>index 35d048c18ac..22f391b86c2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/mock_rtpsender.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/mock_rtpsender.h
>@@ -27,10 +27,9 @@ class MockRtpSender : public rtc::RefCountedObject<RtpSenderInterface> {
>   MOCK_CONST_METHOD0(media_type, cricket::MediaType());
>   MOCK_CONST_METHOD0(id, std::string());
>   MOCK_CONST_METHOD0(stream_ids, std::vector<std::string>());
>-  MOCK_CONST_METHOD0(GetParameters, RtpParameters());
>-  MOCK_METHOD1(SetParameters, bool(const RtpParameters&));
>+  MOCK_METHOD0(GetParameters, RtpParameters());
>+  MOCK_METHOD1(SetParameters, RTCError(const RtpParameters&));
>   MOCK_CONST_METHOD0(GetDtmfSender, rtc::scoped_refptr<DtmfSenderInterface>());
>-  MOCK_CONST_METHOD0(AttachmentId, int());
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/mock_video_bitrate_allocator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/mock_video_bitrate_allocator.h
>new file mode 100644
>index 00000000000..d25537a2b93
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/mock_video_bitrate_allocator.h
>@@ -0,0 +1,28 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_TEST_MOCK_VIDEO_BITRATE_ALLOCATOR_H_
>+#define API_TEST_MOCK_VIDEO_BITRATE_ALLOCATOR_H_
>+
>+#include "api/video/video_bitrate_allocator.h"
>+#include "test/gmock.h"
>+
>+namespace webrtc {
>+
>+class MockVideoBitrateAllocator : public webrtc::VideoBitrateAllocator {
>+  MOCK_METHOD2(GetAllocation,
>+               VideoBitrateAllocation(uint32_t total_bitrate,
>+                                      uint32_t framerate));
>+  MOCK_METHOD1(GetPreferredBitrateBps, uint32_t(uint32_t framerate));
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // API_TEST_MOCK_VIDEO_BITRATE_ALLOCATOR_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/simulated_network.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/simulated_network.h
>new file mode 100644
>index 00000000000..fcb83cd8a8b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/simulated_network.h
>@@ -0,0 +1,87 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_TEST_SIMULATED_NETWORK_H_
>+#define API_TEST_SIMULATED_NETWORK_H_
>+
>+#include <stddef.h>
>+#include <stdint.h>
>+#include <deque>
>+#include <queue>
>+#include <vector>
>+
>+#include "absl/types/optional.h"
>+#include "rtc_base/criticalsection.h"
>+#include "rtc_base/random.h"
>+#include "rtc_base/thread_annotations.h"
>+
>+namespace webrtc {
>+
>+struct PacketInFlightInfo {
>+  PacketInFlightInfo(size_t size, int64_t send_time_us, uint64_t packet_id)
>+      : size(size), send_time_us(send_time_us), packet_id(packet_id) {}
>+
>+  size_t size;
>+  int64_t send_time_us;
>+  // Unique identifier for the packet in relation to other packets in flight.
>+  uint64_t packet_id;
>+};
>+
>+struct PacketDeliveryInfo {
>+  static constexpr int kNotReceived = -1;
>+  PacketDeliveryInfo(PacketInFlightInfo source, int64_t receive_time_us)
>+      : receive_time_us(receive_time_us), packet_id(source.packet_id) {}
>+  int64_t receive_time_us;
>+  uint64_t packet_id;
>+};
>+
>+// DefaultNetworkSimulationConfig is a default network simulation configuration
>+// for default network simulation that will be used by WebRTC if no custom
>+// NetworkSimulationInterface is provided.
>+struct DefaultNetworkSimulationConfig {
>+  DefaultNetworkSimulationConfig() {}
>+  // Queue length in number of packets.
>+  size_t queue_length_packets = 0;
>+  // Delay in addition to capacity induced delay.
>+  int queue_delay_ms = 0;
>+  // Standard deviation of the extra delay.
>+  int delay_standard_deviation_ms = 0;
>+  // Link capacity in kbps.
>+  int link_capacity_kbps = 0;
>+  // Random packet loss.
>+  int loss_percent = 0;
>+  // If packets are allowed to be reordered.
>+  bool allow_reordering = false;
>+  // The average length of a burst of lost packets.
>+  int avg_burst_loss_length = -1;
>+};
>+
>+class NetworkSimulationInterface {
>+ public:
>+  // DO NOT USE. Use DefaultNetworkSimulationConfig directly. This reference
>+  // should be removed when all users will be switched on direct usage.
>+  using SimulatedNetworkConfig = DefaultNetworkSimulationConfig;
>+
>+  // DO NOT USE. Method added temporary for further refactoring and will be
>+  // removed soon.
>+  // Sets a new configuration. This won't affect packets already in the pipe.
>+  virtual void SetConfig(const SimulatedNetworkConfig& config) = 0;
>+
>+  virtual bool EnqueuePacket(PacketInFlightInfo packet_info) = 0;
>+  // Retrieves all packets that should be delivered by the given receive time.
>+  virtual std::vector<PacketDeliveryInfo> DequeueDeliverablePackets(
>+      int64_t receive_time_us) = 0;
>+  virtual absl::optional<int64_t> NextDeliveryTimeUs() const = 0;
>+  virtual ~NetworkSimulationInterface() = default;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // API_TEST_SIMULATED_NETWORK_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/simulcast_test_fixture.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/simulcast_test_fixture.h
>new file mode 100644
>index 00000000000..e7eab241348
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/simulcast_test_fixture.h
>@@ -0,0 +1,41 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_TEST_SIMULCAST_TEST_FIXTURE_H_
>+#define API_TEST_SIMULCAST_TEST_FIXTURE_H_
>+
>+namespace webrtc {
>+namespace test {
>+
>+class SimulcastTestFixture {
>+ public:
>+  virtual ~SimulcastTestFixture() = default;
>+
>+  virtual void TestKeyFrameRequestsOnAllStreams() = 0;
>+  virtual void TestPaddingAllStreams() = 0;
>+  virtual void TestPaddingTwoStreams() = 0;
>+  virtual void TestPaddingTwoStreamsOneMaxedOut() = 0;
>+  virtual void TestPaddingOneStream() = 0;
>+  virtual void TestPaddingOneStreamTwoMaxedOut() = 0;
>+  virtual void TestSendAllStreams() = 0;
>+  virtual void TestDisablingStreams() = 0;
>+  virtual void TestActiveStreams() = 0;
>+  virtual void TestSwitchingToOneStream() = 0;
>+  virtual void TestSwitchingToOneOddStream() = 0;
>+  virtual void TestSwitchingToOneSmallStream() = 0;
>+  virtual void TestSpatioTemporalLayers333PatternEncoder() = 0;
>+  virtual void TestSpatioTemporalLayers321PatternEncoder() = 0;
>+  virtual void TestStrideEncodeDecode() = 0;
>+};
>+
>+}  // namespace test
>+}  // namespace webrtc
>+
>+#endif  // API_TEST_SIMULCAST_TEST_FIXTURE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/video_quality_test_fixture.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/video_quality_test_fixture.h
>new file mode 100644
>index 00000000000..1d8fc346504
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/video_quality_test_fixture.h
>@@ -0,0 +1,111 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_TEST_VIDEO_QUALITY_TEST_FIXTURE_H_
>+#define API_TEST_VIDEO_QUALITY_TEST_FIXTURE_H_
>+
>+#include <map>
>+#include <memory>
>+#include <string>
>+#include <vector>
>+
>+#include "api/bitrate_constraints.h"
>+#include "api/mediatypes.h"
>+#include "api/test/simulated_network.h"
>+#include "api/video_codecs/video_encoder_config.h"
>+
>+namespace webrtc {
>+
>+class VideoQualityTestFixtureInterface {
>+ public:
>+  // Parameters are grouped into smaller structs to make it easier to set
>+  // the desired elements and skip unused, using aggregate initialization.
>+  // Unfortunately, C++11 (as opposed to C11) doesn't support unnamed structs,
>+  // which makes the implementation of VideoQualityTest a bit uglier.
>+  struct Params {
>+    Params();
>+    ~Params();
>+    struct CallConfig {
>+      bool send_side_bwe;
>+      BitrateConstraints call_bitrate_config;
>+      int num_thumbnails;
>+      // Indicates if secondary_(video|ss|screenshare) structures are used.
>+      bool dual_video;
>+    } call;
>+    struct Video {
>+      bool enabled;
>+      size_t width;
>+      size_t height;
>+      int32_t fps;
>+      int min_bitrate_bps;
>+      int target_bitrate_bps;
>+      int max_bitrate_bps;
>+      bool suspend_below_min_bitrate;
>+      std::string codec;
>+      int num_temporal_layers;
>+      int selected_tl;
>+      int min_transmit_bps;
>+      bool ulpfec;
>+      bool flexfec;
>+      bool automatic_scaling;
>+      std::string clip_name;  // "Generator" to generate frames instead.
>+      size_t capture_device_index;
>+      SdpVideoFormat::Parameters sdp_params;
>+    } video[2];
>+    struct Audio {
>+      bool enabled;
>+      bool sync_video;
>+      bool dtx;
>+    } audio;
>+    struct Screenshare {
>+      bool enabled;
>+      bool generate_slides;
>+      int32_t slide_change_interval;
>+      int32_t scroll_duration;
>+      std::vector<std::string> slides;
>+    } screenshare[2];
>+    struct Analyzer {
>+      std::string test_label;
>+      double avg_psnr_threshold;  // (*)
>+      double avg_ssim_threshold;  // (*)
>+      int test_durations_secs;
>+      std::string graph_data_output_filename;
>+      std::string graph_title;
>+    } analyzer;
>+    DefaultNetworkSimulationConfig pipe;
>+    struct SS {                          // Spatial scalability.
>+      std::vector<VideoStream> streams;  // If empty, one stream is assumed.
>+      size_t selected_stream;
>+      int num_spatial_layers;
>+      int selected_sl;
>+      InterLayerPredMode inter_layer_pred;
>+      // If empty, bitrates are generated in VP9Impl automatically.
>+      std::vector<SpatialLayer> spatial_layers;
>+      // If set, default parameters will be used instead of |streams|.
>+      bool infer_streams;
>+    } ss[2];
>+    struct Logging {
>+      std::string rtc_event_log_name;
>+      std::string rtp_dump_name;
>+      std::string encoded_frame_base_path;
>+    } logging;
>+  };
>+
>+  virtual ~VideoQualityTestFixtureInterface() = default;
>+
>+  virtual void RunWithAnalyzer(const Params& params) = 0;
>+  virtual void RunWithRenderers(const Params& params) = 0;
>+
>+  virtual const std::map<uint8_t, webrtc::MediaType>& payload_type_map() = 0;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // API_TEST_VIDEO_QUALITY_TEST_FIXTURE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/videocodec_test_fixture.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/videocodec_test_fixture.h
>new file mode 100644
>index 00000000000..ec69e50ae19
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/videocodec_test_fixture.h
>@@ -0,0 +1,153 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_TEST_VIDEOCODEC_TEST_FIXTURE_H_
>+#define API_TEST_VIDEOCODEC_TEST_FIXTURE_H_
>+
>+#include <string>
>+#include <vector>
>+
>+#include "api/test/videocodec_test_stats.h"
>+#include "api/video_codecs/video_decoder_factory.h"
>+#include "api/video_codecs/video_encoder_factory.h"
>+#include "modules/video_coding/include/video_codec_interface.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+// Rates for the encoder and the frame number when to change profile.
>+struct RateProfile {
>+  size_t target_kbps;
>+  size_t input_fps;
>+  size_t frame_index_rate_update;
>+};
>+
>+struct RateControlThresholds {
>+  double max_avg_bitrate_mismatch_percent;
>+  double max_time_to_reach_target_bitrate_sec;
>+  // TODO(ssilkin): Use absolute threshold for framerate.
>+  double max_avg_framerate_mismatch_percent;
>+  double max_avg_buffer_level_sec;
>+  double max_max_key_frame_delay_sec;
>+  double max_max_delta_frame_delay_sec;
>+  size_t max_num_spatial_resizes;
>+  size_t max_num_key_frames;
>+};
>+
>+struct QualityThresholds {
>+  double min_avg_psnr;
>+  double min_min_psnr;
>+  double min_avg_ssim;
>+  double min_min_ssim;
>+};
>+
>+struct BitstreamThresholds {
>+  size_t max_max_nalu_size_bytes;
>+};
>+
>+// NOTE: This class is still under development and may change without notice.
>+class VideoCodecTestFixture {
>+ public:
>+  class EncodedFrameChecker {
>+   public:
>+    virtual ~EncodedFrameChecker() = default;
>+    virtual void CheckEncodedFrame(webrtc::VideoCodecType codec,
>+                                   const EncodedImage& encoded_frame) const = 0;
>+  };
>+
>+  struct Config {
>+    Config();
>+    void SetCodecSettings(std::string codec_name,
>+                          size_t num_simulcast_streams,
>+                          size_t num_spatial_layers,
>+                          size_t num_temporal_layers,
>+                          bool denoising_on,
>+                          bool frame_dropper_on,
>+                          bool spatial_resize_on,
>+                          size_t width,
>+                          size_t height);
>+
>+    size_t NumberOfCores() const;
>+    size_t NumberOfTemporalLayers() const;
>+    size_t NumberOfSpatialLayers() const;
>+    size_t NumberOfSimulcastStreams() const;
>+
>+    std::string ToString() const;
>+    std::string CodecName() const;
>+    bool IsAsyncCodec() const;
>+
>+    // Plain name of YUV file to process without file extension.
>+    std::string filename;
>+
>+    // File to process. This must be a video file in the YUV format.
>+    std::string filepath;
>+
>+    // Number of frames to process.
>+    size_t num_frames = 0;
>+
>+    // Bitstream constraints.
>+    size_t max_payload_size_bytes = 1440;
>+
>+    // Should we decode the encoded frames?
>+    bool decode = true;
>+
>+    // Force the encoder and decoder to use a single core for processing.
>+    bool use_single_core = false;
>+
>+    // Should cpu usage be measured?
>+    // If set to true, the encoding will run in real-time.
>+    bool measure_cpu = false;
>+
>+    // If > 0: forces the encoder to create a keyframe every Nth frame.
>+    size_t keyframe_interval = 0;
>+
>+    // Codec settings to use.
>+    webrtc::VideoCodec codec_settings;
>+
>+    // Name of the codec being tested.
>+    std::string codec_name;
>+
>+    // H.264 specific settings.
>+    struct H264CodecSettings {
>+      H264::Profile profile = H264::kProfileConstrainedBaseline;
>+      H264PacketizationMode packetization_mode =
>+          webrtc::H264PacketizationMode::NonInterleaved;
>+    } h264_codec_settings;
>+
>+    // Should hardware accelerated codecs be used?
>+    bool hw_encoder = false;
>+    bool hw_decoder = false;
>+
>+    // Custom checker that will be called for each frame.
>+    const EncodedFrameChecker* encoded_frame_checker = nullptr;
>+
>+    // Print out frame level stats.
>+    bool print_frame_level_stats = false;
>+
>+    // Should video be saved persistently to disk for post-run visualization?
>+    struct VisualizationParams {
>+      bool save_encoded_ivf = false;
>+      bool save_decoded_y4m = false;
>+    } visualization_params;
>+  };
>+
>+  virtual ~VideoCodecTestFixture() = default;
>+
>+  virtual void RunTest(const std::vector<RateProfile>& rate_profiles,
>+                       const std::vector<RateControlThresholds>* rc_thresholds,
>+                       const std::vector<QualityThresholds>* quality_thresholds,
>+                       const BitstreamThresholds* bs_thresholds) = 0;
>+  virtual VideoCodecTestStats& GetStats() = 0;
>+};
>+
>+}  // namespace test
>+}  // namespace webrtc
>+
>+#endif  // API_TEST_VIDEOCODEC_TEST_FIXTURE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/videocodec_test_stats.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/videocodec_test_stats.cc
>new file mode 100644
>index 00000000000..72522785cf0
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/videocodec_test_stats.cc
>@@ -0,0 +1,95 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/test/videocodec_test_stats.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+std::string VideoCodecTestStats::FrameStatistics::ToString() const {
>+  std::stringstream ss;
>+  ss << "frame_number " << frame_number;
>+  ss << " decoded_width " << decoded_width;
>+  ss << " decoded_height " << decoded_height;
>+  ss << " spatial_idx " << spatial_idx;
>+  ss << " temporal_idx " << temporal_idx;
>+  ss << " inter_layer_predicted " << inter_layer_predicted;
>+  ss << " non_ref_for_inter_layer_pred " << non_ref_for_inter_layer_pred;
>+  ss << " frame_type " << frame_type;
>+  ss << " length_bytes " << length_bytes;
>+  ss << " qp " << qp;
>+  ss << " psnr " << psnr;
>+  ss << " psnr_y " << psnr_y;
>+  ss << " psnr_u " << psnr_u;
>+  ss << " psnr_v " << psnr_v;
>+  ss << " ssim " << ssim;
>+  ss << " encode_time_us " << encode_time_us;
>+  ss << " decode_time_us " << decode_time_us;
>+  ss << " rtp_timestamp " << rtp_timestamp;
>+  ss << " target_bitrate_kbps " << target_bitrate_kbps;
>+  return ss.str();
>+}
>+
>+VideoCodecTestStats::VideoStatistics::VideoStatistics() = default;
>+VideoCodecTestStats::VideoStatistics::VideoStatistics(const VideoStatistics&) =
>+    default;
>+
>+std::string VideoCodecTestStats::VideoStatistics::ToString(
>+    std::string prefix) const {
>+  std::stringstream ss;
>+  ss << prefix << "target_bitrate_kbps: " << target_bitrate_kbps;
>+  ss << "\n" << prefix << "input_framerate_fps: " << input_framerate_fps;
>+  ss << "\n" << prefix << "spatial_idx: " << spatial_idx;
>+  ss << "\n" << prefix << "temporal_idx: " << temporal_idx;
>+  ss << "\n" << prefix << "width: " << width;
>+  ss << "\n" << prefix << "height: " << height;
>+  ss << "\n" << prefix << "length_bytes: " << length_bytes;
>+  ss << "\n" << prefix << "bitrate_kbps: " << bitrate_kbps;
>+  ss << "\n" << prefix << "framerate_fps: " << framerate_fps;
>+  ss << "\n" << prefix << "enc_speed_fps: " << enc_speed_fps;
>+  ss << "\n" << prefix << "dec_speed_fps: " << dec_speed_fps;
>+  ss << "\n" << prefix << "avg_delay_sec: " << avg_delay_sec;
>+  ss << "\n"
>+     << prefix << "max_key_frame_delay_sec: " << max_key_frame_delay_sec;
>+  ss << "\n"
>+     << prefix << "max_delta_frame_delay_sec: " << max_delta_frame_delay_sec;
>+  ss << "\n"
>+     << prefix << "time_to_reach_target_bitrate_sec: "
>+     << time_to_reach_target_bitrate_sec;
>+  ss << "\n"
>+     << prefix << "avg_key_frame_size_bytes: " << avg_key_frame_size_bytes;
>+  ss << "\n"
>+     << prefix << "avg_delta_frame_size_bytes: " << avg_delta_frame_size_bytes;
>+  ss << "\n" << prefix << "avg_qp: " << avg_qp;
>+  ss << "\n" << prefix << "avg_psnr: " << avg_psnr;
>+  ss << "\n" << prefix << "min_psnr: " << min_psnr;
>+  ss << "\n" << prefix << "avg_ssim: " << avg_ssim;
>+  ss << "\n" << prefix << "min_ssim: " << min_ssim;
>+  ss << "\n" << prefix << "num_input_frames: " << num_input_frames;
>+  ss << "\n" << prefix << "num_encoded_frames: " << num_encoded_frames;
>+  ss << "\n" << prefix << "num_decoded_frames: " << num_decoded_frames;
>+  ss << "\n"
>+     << prefix
>+     << "num_dropped_frames: " << num_input_frames - num_encoded_frames;
>+  ss << "\n" << prefix << "num_key_frames: " << num_key_frames;
>+  ss << "\n" << prefix << "num_spatial_resizes: " << num_spatial_resizes;
>+  ss << "\n" << prefix << "max_nalu_size_bytes: " << max_nalu_size_bytes;
>+  return ss.str();
>+}
>+
>+VideoCodecTestStats::FrameStatistics::FrameStatistics(size_t frame_number,
>+                                                      size_t rtp_timestamp)
>+    : frame_number(frame_number), rtp_timestamp(rtp_timestamp) {}
>+
>+VideoCodecTestStats::FrameStatistics::FrameStatistics(
>+    const FrameStatistics& rhs) = default;
>+
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/videocodec_test_stats.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/videocodec_test_stats.h
>new file mode 100644
>index 00000000000..de6d3500d43
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/test/videocodec_test_stats.h
>@@ -0,0 +1,149 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_TEST_VIDEOCODEC_TEST_STATS_H_
>+#define API_TEST_VIDEOCODEC_TEST_STATS_H_
>+
>+#include <map>
>+#include <string>
>+#include <vector>
>+
>+#include "common_types.h"  // NOLINT(build/include)
>+
>+namespace webrtc {
>+namespace test {
>+
>+// Statistics for a sequence of processed frames. This class is not thread safe.
>+class VideoCodecTestStats {
>+ public:
>+  // Statistics for one processed frame.
>+  struct FrameStatistics {
>+    FrameStatistics(size_t frame_number, size_t rtp_timestamp);
>+    FrameStatistics(const FrameStatistics& rhs);
>+
>+    std::string ToString() const;
>+
>+    size_t frame_number = 0;
>+    size_t rtp_timestamp = 0;
>+
>+    // Encoding.
>+    int64_t encode_start_ns = 0;
>+    int encode_return_code = 0;
>+    bool encoding_successful = false;
>+    size_t encode_time_us = 0;
>+    size_t target_bitrate_kbps = 0;
>+    size_t length_bytes = 0;
>+    webrtc::FrameType frame_type = kVideoFrameDelta;
>+
>+    // Layering.
>+    size_t spatial_idx = 0;
>+    size_t temporal_idx = 0;
>+    bool inter_layer_predicted = false;
>+    bool non_ref_for_inter_layer_pred = true;
>+
>+    // H264 specific.
>+    size_t max_nalu_size_bytes = 0;
>+
>+    // Decoding.
>+    int64_t decode_start_ns = 0;
>+    int decode_return_code = 0;
>+    bool decoding_successful = false;
>+    size_t decode_time_us = 0;
>+    size_t decoded_width = 0;
>+    size_t decoded_height = 0;
>+
>+    // Quantization.
>+    int qp = -1;
>+
>+    // Quality.
>+    float psnr_y = 0.0f;
>+    float psnr_u = 0.0f;
>+    float psnr_v = 0.0f;
>+    float psnr = 0.0f;  // 10 * log10(255^2 / (mse_y + mse_u + mse_v)).
>+    float ssim = 0.0f;  // 0.8 * ssim_y + 0.1 * (ssim_u + ssim_v).
>+  };
>+
>+  struct VideoStatistics {
>+    VideoStatistics();
>+    VideoStatistics(const VideoStatistics&);
>+
>+    std::string ToString(std::string prefix) const;
>+
>+    size_t target_bitrate_kbps = 0;
>+    float input_framerate_fps = 0.0f;
>+
>+    size_t spatial_idx = 0;
>+    size_t temporal_idx = 0;
>+
>+    size_t width = 0;
>+    size_t height = 0;
>+
>+    size_t length_bytes = 0;
>+    size_t bitrate_kbps = 0;
>+    float framerate_fps = 0;
>+
>+    float enc_speed_fps = 0.0f;
>+    float dec_speed_fps = 0.0f;
>+
>+    float avg_delay_sec = 0.0f;
>+    float max_key_frame_delay_sec = 0.0f;
>+    float max_delta_frame_delay_sec = 0.0f;
>+    float time_to_reach_target_bitrate_sec = 0.0f;
>+
>+    float avg_key_frame_size_bytes = 0.0f;
>+    float avg_delta_frame_size_bytes = 0.0f;
>+    float avg_qp = 0.0f;
>+
>+    float avg_psnr_y = 0.0f;
>+    float avg_psnr_u = 0.0f;
>+    float avg_psnr_v = 0.0f;
>+    float avg_psnr = 0.0f;
>+    float min_psnr = 0.0f;
>+    float avg_ssim = 0.0f;
>+    float min_ssim = 0.0f;
>+
>+    size_t num_input_frames = 0;
>+    size_t num_encoded_frames = 0;
>+    size_t num_decoded_frames = 0;
>+    size_t num_key_frames = 0;
>+    size_t num_spatial_resizes = 0;
>+    size_t max_nalu_size_bytes = 0;
>+  };
>+
>+  virtual ~VideoCodecTestStats() = default;
>+
>+  // Creates a FrameStatistics for the next frame to be processed.
>+  virtual FrameStatistics* AddFrame(size_t timestamp, size_t spatial_idx) = 0;
>+
>+  // Returns the FrameStatistics corresponding to |frame_number| or |timestamp|.
>+  virtual FrameStatistics* GetFrame(size_t frame_number,
>+                                    size_t spatial_idx) = 0;
>+  virtual FrameStatistics* GetFrameWithTimestamp(size_t timestamp,
>+                                                 size_t spatial_idx) = 0;
>+
>+  virtual std::vector<VideoStatistics> SliceAndCalcLayerVideoStatistic(
>+      size_t first_frame_num,
>+      size_t last_frame_num) = 0;
>+
>+  virtual VideoStatistics SliceAndCalcAggregatedVideoStatistic(
>+      size_t first_frame_num,
>+      size_t last_frame_num) = 0;
>+
>+  virtual void PrintFrameStatistics() = 0;
>+
>+  virtual size_t Size(size_t spatial_idx) = 0;
>+
>+  virtual void Clear() = 0;
>+};
>+
>+}  // namespace test
>+}  // namespace webrtc
>+
>+#endif  // API_TEST_VIDEOCODEC_TEST_STATS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/BUILD.gn
>new file mode 100644
>index 00000000000..f473179dd14
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/BUILD.gn
>@@ -0,0 +1,54 @@
>+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+#
>+# Use of this source code is governed by a BSD-style license
>+# that can be found in the LICENSE file in the root of the source
>+# tree. An additional intellectual property rights grant can be found
>+# in the file PATENTS.  All contributing project authors may
>+# be found in the AUTHORS file in the root of the source tree.
>+
>+import("../../webrtc.gni")
>+
>+rtc_source_set("bitrate_settings") {
>+  visibility = [ "*" ]
>+  sources = [
>+    "bitrate_settings.cc",
>+    "bitrate_settings.h",
>+  ]
>+  deps = [
>+    "//third_party/abseil-cpp/absl/types:optional",
>+  ]
>+}
>+
>+rtc_static_library("network_control") {
>+  sources = [
>+    "network_control.h",
>+    "network_types.cc",
>+    "network_types.h",
>+  ]
>+
>+  deps = [
>+    "../units:data_rate",
>+    "../units:data_size",
>+    "../units:time_delta",
>+    "../units:timestamp",
>+    "//third_party/abseil-cpp/absl/types:optional",
>+  ]
>+}
>+
>+if (rtc_include_tests) {
>+  rtc_source_set("network_control_test") {
>+    testonly = true
>+    sources = [
>+      "test/mock_network_control.h",
>+      "test/network_control_tester.cc",
>+      "test/network_control_tester.h",
>+    ]
>+    deps = [
>+      ":network_control",
>+      "../../rtc_base:checks",
>+      "../../rtc_base:rtc_base_approved",
>+      "../../test:test_support",
>+      "//third_party/abseil-cpp/absl/types:optional",
>+    ]
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/bitrate_settings.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/bitrate_settings.cc
>new file mode 100644
>index 00000000000..c72bd827911
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/bitrate_settings.cc
>@@ -0,0 +1,19 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/transport/bitrate_settings.h"
>+
>+namespace webrtc {
>+
>+BitrateSettings::BitrateSettings() = default;
>+BitrateSettings::~BitrateSettings() = default;
>+BitrateSettings::BitrateSettings(const BitrateSettings&) = default;
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/bitrate_settings.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/bitrate_settings.h
>new file mode 100644
>index 00000000000..77654bc4016
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/bitrate_settings.h
>@@ -0,0 +1,35 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_TRANSPORT_BITRATE_SETTINGS_H_
>+#define API_TRANSPORT_BITRATE_SETTINGS_H_
>+
>+#include "absl/types/optional.h"
>+
>+namespace webrtc {
>+
>+// Configuration of send bitrate. The |start_bitrate_bps| value is
>+// used for multiple purposes, both as a prior in the bandwidth
>+// estimator, and for initial configuration of the encoder. We may
>+// want to create separate apis for those, and use a smaller struct
>+// with only the min and max constraints.
>+struct BitrateSettings {
>+  BitrateSettings();
>+  ~BitrateSettings();
>+  BitrateSettings(const BitrateSettings&);
>+  // 0 <= min <= start <= max should hold for set parameters.
>+  absl::optional<int> min_bitrate_bps;
>+  absl::optional<int> start_bitrate_bps;
>+  absl::optional<int> max_bitrate_bps;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // API_TRANSPORT_BITRATE_SETTINGS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/network_control.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/network_control.h
>new file mode 100644
>index 00000000000..abd945d8970
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/network_control.h
>@@ -0,0 +1,93 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_TRANSPORT_NETWORK_CONTROL_H_
>+#define API_TRANSPORT_NETWORK_CONTROL_H_
>+#include <stdint.h>
>+#include <memory>
>+
>+#include "api/transport/network_types.h"
>+
>+namespace webrtc {
>+
>+class TargetTransferRateObserver {
>+ public:
>+  virtual ~TargetTransferRateObserver() = default;
>+  // Called to indicate target transfer rate as well as giving information about
>+  // the current estimate of network parameters.
>+  virtual void OnTargetTransferRate(TargetTransferRate) = 0;
>+};
>+
>+// Configuration sent to factory create function. The parameters here are
>+// optional to use for a network controller implementation.
>+struct NetworkControllerConfig {
>+  // The initial constraints to start with, these can be changed at any later
>+  // time by calls to OnTargetRateConstraints.
>+  TargetRateConstraints constraints;
>+  // Initial stream specific configuration, these are changed at any later time
>+  // by calls to OnStreamsConfig.
>+  StreamsConfig stream_based_config;
>+  // The initial bandwidth estimate to base target rate on. This should be used
>+  // as the basis for initial OnTargetTransferRate and OnPacerConfig callbacks.
>+  // Note that starting rate is only provided on construction.
>+  DataRate starting_bandwidth = DataRate::Infinity();
>+};
>+
>+// NetworkControllerInterface is implemented by network controllers. A network
>+// controller is a class that uses information about network state and traffic
>+// to estimate network parameters such as round trip time and bandwidth. Network
>+// controllers does not guarantee thread safety, the interface must be used in a
>+// non-concurrent fashion.
>+class NetworkControllerInterface {
>+ public:
>+  virtual ~NetworkControllerInterface() = default;
>+
>+  // Called when network availabilty changes.
>+  virtual NetworkControlUpdate OnNetworkAvailability(NetworkAvailability) = 0;
>+  // Called when the receiving or sending endpoint changes address.
>+  virtual NetworkControlUpdate OnNetworkRouteChange(NetworkRouteChange) = 0;
>+  // Called periodically with a periodicy as specified by
>+  // NetworkControllerFactoryInterface::GetProcessInterval.
>+  virtual NetworkControlUpdate OnProcessInterval(ProcessInterval) = 0;
>+  // Called when remotely calculated bitrate is received.
>+  virtual NetworkControlUpdate OnRemoteBitrateReport(RemoteBitrateReport) = 0;
>+  // Called round trip time has been calculated by protocol specific mechanisms.
>+  virtual NetworkControlUpdate OnRoundTripTimeUpdate(RoundTripTimeUpdate) = 0;
>+  // Called when a packet is sent on the network.
>+  virtual NetworkControlUpdate OnSentPacket(SentPacket) = 0;
>+  // Called when the stream specific configuration has been updated.
>+  virtual NetworkControlUpdate OnStreamsConfig(StreamsConfig) = 0;
>+  // Called when target transfer rate constraints has been changed.
>+  virtual NetworkControlUpdate OnTargetRateConstraints(
>+      TargetRateConstraints) = 0;
>+  // Called when a protocol specific calculation of packet loss has been made.
>+  virtual NetworkControlUpdate OnTransportLossReport(TransportLossReport) = 0;
>+  // Called with per packet feedback regarding receive time.
>+  virtual NetworkControlUpdate OnTransportPacketsFeedback(
>+      TransportPacketsFeedback) = 0;
>+};
>+
>+// NetworkControllerFactoryInterface is an interface for creating a network
>+// controller.
>+class NetworkControllerFactoryInterface {
>+ public:
>+  virtual ~NetworkControllerFactoryInterface() = default;
>+
>+  // Used to create a new network controller, requires an observer to be
>+  // provided to handle callbacks.
>+  virtual std::unique_ptr<NetworkControllerInterface> Create(
>+      NetworkControllerConfig config) = 0;
>+  // Returns the interval by which the network controller expects
>+  // OnProcessInterval calls.
>+  virtual TimeDelta GetProcessInterval() const = 0;
>+};
>+}  // namespace webrtc
>+
>+#endif  // API_TRANSPORT_NETWORK_CONTROL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/network_types.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/network_types.cc
>new file mode 100644
>index 00000000000..b4c09d47c0c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/network_types.cc
>@@ -0,0 +1,84 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/transport/network_types.h"
>+
>+namespace webrtc {
>+
>+StreamsConfig::StreamsConfig() = default;
>+StreamsConfig::StreamsConfig(const StreamsConfig&) = default;
>+StreamsConfig::~StreamsConfig() = default;
>+
>+TargetRateConstraints::TargetRateConstraints() = default;
>+TargetRateConstraints::TargetRateConstraints(const TargetRateConstraints&) =
>+    default;
>+TargetRateConstraints::~TargetRateConstraints() = default;
>+
>+NetworkRouteChange::NetworkRouteChange() = default;
>+NetworkRouteChange::NetworkRouteChange(const NetworkRouteChange&) = default;
>+NetworkRouteChange::~NetworkRouteChange() = default;
>+
>+PacketResult::PacketResult() = default;
>+PacketResult::PacketResult(const PacketResult& other) = default;
>+PacketResult::~PacketResult() = default;
>+
>+TransportPacketsFeedback::TransportPacketsFeedback() = default;
>+TransportPacketsFeedback::TransportPacketsFeedback(
>+    const TransportPacketsFeedback& other) = default;
>+TransportPacketsFeedback::~TransportPacketsFeedback() = default;
>+
>+std::vector<PacketResult> TransportPacketsFeedback::ReceivedWithSendInfo()
>+    const {
>+  std::vector<PacketResult> res;
>+  for (const PacketResult& fb : packet_feedbacks) {
>+    if (fb.receive_time.IsFinite() && fb.sent_packet.has_value()) {
>+      res.push_back(fb);
>+    }
>+  }
>+  return res;
>+}
>+
>+std::vector<PacketResult> TransportPacketsFeedback::LostWithSendInfo() const {
>+  std::vector<PacketResult> res;
>+  for (const PacketResult& fb : packet_feedbacks) {
>+    if (fb.receive_time.IsInfinite() && fb.sent_packet.has_value()) {
>+      res.push_back(fb);
>+    }
>+  }
>+  return res;
>+}
>+
>+std::vector<PacketResult> TransportPacketsFeedback::PacketsWithFeedback()
>+    const {
>+  return packet_feedbacks;
>+}
>+
>+NetworkControlUpdate::NetworkControlUpdate() = default;
>+NetworkControlUpdate::NetworkControlUpdate(const NetworkControlUpdate&) =
>+    default;
>+NetworkControlUpdate::~NetworkControlUpdate() = default;
>+
>+PacedPacketInfo::PacedPacketInfo() = default;
>+
>+PacedPacketInfo::PacedPacketInfo(int probe_cluster_id,
>+                                 int probe_cluster_min_probes,
>+                                 int probe_cluster_min_bytes)
>+    : probe_cluster_id(probe_cluster_id),
>+      probe_cluster_min_probes(probe_cluster_min_probes),
>+      probe_cluster_min_bytes(probe_cluster_min_bytes) {}
>+
>+bool PacedPacketInfo::operator==(const PacedPacketInfo& rhs) const {
>+  return send_bitrate_bps == rhs.send_bitrate_bps &&
>+         probe_cluster_id == rhs.probe_cluster_id &&
>+         probe_cluster_min_probes == rhs.probe_cluster_min_probes &&
>+         probe_cluster_min_bytes == rhs.probe_cluster_min_bytes;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/network_types.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/network_types.h
>new file mode 100644
>index 00000000000..a3897168825
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/network_types.h
>@@ -0,0 +1,200 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_TRANSPORT_NETWORK_TYPES_H_
>+#define API_TRANSPORT_NETWORK_TYPES_H_
>+#include <stdint.h>
>+#include <vector>
>+
>+#include "absl/types/optional.h"
>+#include "api/units/data_rate.h"
>+#include "api/units/data_size.h"
>+#include "api/units/time_delta.h"
>+#include "api/units/timestamp.h"
>+
>+namespace webrtc {
>+
>+// Configuration
>+
>+// Use StreamsConfig for information about streams that is required for specific
>+// adjustments to the algorithms in network controllers. Especially useful
>+// for experiments.
>+struct StreamsConfig {
>+  StreamsConfig();
>+  StreamsConfig(const StreamsConfig&);
>+  ~StreamsConfig();
>+  Timestamp at_time = Timestamp::Infinity();
>+  bool requests_alr_probing = false;
>+  absl::optional<double> pacing_factor;
>+  absl::optional<DataRate> min_pacing_rate;
>+  absl::optional<DataRate> max_padding_rate;
>+  absl::optional<DataRate> max_total_allocated_bitrate;
>+};
>+
>+struct TargetRateConstraints {
>+  TargetRateConstraints();
>+  TargetRateConstraints(const TargetRateConstraints&);
>+  ~TargetRateConstraints();
>+  Timestamp at_time = Timestamp::Infinity();
>+  absl::optional<DataRate> min_data_rate;
>+  absl::optional<DataRate> max_data_rate;
>+};
>+
>+// Send side information
>+
>+struct NetworkAvailability {
>+  Timestamp at_time = Timestamp::Infinity();
>+  bool network_available = false;
>+};
>+
>+struct NetworkRouteChange {
>+  NetworkRouteChange();
>+  NetworkRouteChange(const NetworkRouteChange&);
>+  ~NetworkRouteChange();
>+  Timestamp at_time = Timestamp::Infinity();
>+  // The TargetRateConstraints are set here so they can be changed synchronously
>+  // when network route changes.
>+  TargetRateConstraints constraints;
>+  absl::optional<DataRate> starting_rate;
>+};
>+
>+struct PacedPacketInfo {
>+  PacedPacketInfo();
>+  PacedPacketInfo(int probe_cluster_id,
>+                  int probe_cluster_min_probes,
>+                  int probe_cluster_min_bytes);
>+
>+  bool operator==(const PacedPacketInfo& rhs) const;
>+
>+  // TODO(srte): Move probing info to a separate, optional struct.
>+  static constexpr int kNotAProbe = -1;
>+  int send_bitrate_bps = -1;
>+  int probe_cluster_id = kNotAProbe;
>+  int probe_cluster_min_probes = -1;
>+  int probe_cluster_min_bytes = -1;
>+};
>+
>+struct SentPacket {
>+  Timestamp send_time = Timestamp::Infinity();
>+  DataSize size = DataSize::Zero();
>+  PacedPacketInfo pacing_info;
>+  // Transport independent sequence number, any tracked packet should have a
>+  // sequence number that is unique over the whole call and increasing by 1 for
>+  // each packet.
>+  int64_t sequence_number;
>+  // Data in flight when the packet was sent, including the packet.
>+  DataSize data_in_flight = DataSize::Zero();
>+};
>+
>+// Transport level feedback
>+
>+struct RemoteBitrateReport {
>+  Timestamp receive_time = Timestamp::Infinity();
>+  DataRate bandwidth = DataRate::Infinity();
>+};
>+
>+struct RoundTripTimeUpdate {
>+  Timestamp receive_time = Timestamp::Infinity();
>+  TimeDelta round_trip_time = TimeDelta::PlusInfinity();
>+  bool smoothed = false;
>+};
>+
>+struct TransportLossReport {
>+  Timestamp receive_time = Timestamp::Infinity();
>+  Timestamp start_time = Timestamp::Infinity();
>+  Timestamp end_time = Timestamp::Infinity();
>+  uint64_t packets_lost_delta = 0;
>+  uint64_t packets_received_delta = 0;
>+};
>+
>+// Packet level feedback
>+
>+struct PacketResult {
>+  PacketResult();
>+  PacketResult(const PacketResult&);
>+  ~PacketResult();
>+
>+  absl::optional<SentPacket> sent_packet;
>+  Timestamp receive_time = Timestamp::Infinity();
>+};
>+
>+struct TransportPacketsFeedback {
>+  TransportPacketsFeedback();
>+  TransportPacketsFeedback(const TransportPacketsFeedback& other);
>+  ~TransportPacketsFeedback();
>+
>+  Timestamp feedback_time = Timestamp::Infinity();
>+  DataSize data_in_flight = DataSize::Zero();
>+  DataSize prior_in_flight = DataSize::Zero();
>+  std::vector<PacketResult> packet_feedbacks;
>+
>+  std::vector<PacketResult> ReceivedWithSendInfo() const;
>+  std::vector<PacketResult> LostWithSendInfo() const;
>+  std::vector<PacketResult> PacketsWithFeedback() const;
>+};
>+
>+// Network estimation
>+
>+struct NetworkEstimate {
>+  Timestamp at_time = Timestamp::Infinity();
>+  DataRate bandwidth = DataRate::Infinity();
>+  TimeDelta round_trip_time = TimeDelta::PlusInfinity();
>+  TimeDelta bwe_period = TimeDelta::PlusInfinity();
>+
>+  float loss_rate_ratio = 0;
>+};
>+
>+// Network control
>+
>+struct PacerConfig {
>+  Timestamp at_time = Timestamp::Infinity();
>+  // Pacer should send at most data_window data over time_window duration.
>+  DataSize data_window = DataSize::Infinity();
>+  TimeDelta time_window = TimeDelta::PlusInfinity();
>+  // Pacer should send at least pad_window data over time_window duration.
>+  DataSize pad_window = DataSize::Zero();
>+  DataRate data_rate() const { return data_window / time_window; }
>+  DataRate pad_rate() const { return pad_window / time_window; }
>+};
>+
>+struct ProbeClusterConfig {
>+  Timestamp at_time = Timestamp::Infinity();
>+  DataRate target_data_rate = DataRate::Zero();
>+  TimeDelta target_duration = TimeDelta::Zero();
>+  int32_t target_probe_count = 0;
>+};
>+
>+struct TargetTransferRate {
>+  Timestamp at_time = Timestamp::Infinity();
>+  // The estimate on which the target rate is based on.
>+  NetworkEstimate network_estimate;
>+  DataRate target_rate = DataRate::Zero();
>+};
>+
>+// Contains updates of network controller comand state. Using optionals to
>+// indicate whether a member has been updated. The array of probe clusters
>+// should be used to send out probes if not empty.
>+struct NetworkControlUpdate {
>+  NetworkControlUpdate();
>+  NetworkControlUpdate(const NetworkControlUpdate&);
>+  ~NetworkControlUpdate();
>+  absl::optional<DataSize> congestion_window;
>+  absl::optional<PacerConfig> pacer_config;
>+  std::vector<ProbeClusterConfig> probe_cluster_configs;
>+  absl::optional<TargetTransferRate> target_rate;
>+};
>+
>+// Process control
>+struct ProcessInterval {
>+  Timestamp at_time = Timestamp::Infinity();
>+};
>+}  // namespace webrtc
>+
>+#endif  // API_TRANSPORT_NETWORK_TYPES_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/test/mock_network_control.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/test/mock_network_control.h
>new file mode 100644
>index 00000000000..df83791e18c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/test/mock_network_control.h
>@@ -0,0 +1,26 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_TRANSPORT_TEST_MOCK_NETWORK_CONTROL_H_
>+#define API_TRANSPORT_TEST_MOCK_NETWORK_CONTROL_H_
>+
>+#include "api/transport/include/network_control.h"
>+#include "test/gmock.h"
>+
>+namespace webrtc {
>+namespace test {
>+class MockTargetTransferRateObserver : public TargetTransferRateObserver {
>+ public:
>+  MOCK_METHOD1(OnTargetTransferRate, void(TargetTransferRate));
>+};
>+}  // namespace test
>+}  // namespace webrtc
>+
>+#endif  // API_TRANSPORT_TEST_MOCK_NETWORK_CONTROL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/test/network_control_tester.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/test/network_control_tester.cc
>new file mode 100644
>index 00000000000..2edabd0717e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/test/network_control_tester.cc
>@@ -0,0 +1,147 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/transport/test/network_control_tester.h"
>+
>+#include <algorithm>
>+
>+#include "api/transport/network_control.h"
>+#include "rtc_base/logging.h"
>+
>+namespace webrtc {
>+namespace test {
>+namespace {
>+void Update(NetworkControlUpdate* target, const NetworkControlUpdate& update) {
>+  if (update.congestion_window) {
>+    RTC_LOG(LS_INFO) << "Received window="
>+                     << ToString(*update.congestion_window);
>+    target->congestion_window = update.congestion_window;
>+  }
>+  if (update.pacer_config) {
>+    RTC_LOG(LS_INFO) << "Received pacing at:"
>+                     << ToString(update.pacer_config->at_time)
>+                     << ": rate=" << ToString(update.pacer_config->data_rate())
>+                     << ", pad=" << ToString(update.pacer_config->pad_rate());
>+    target->pacer_config = update.pacer_config;
>+  }
>+  if (update.target_rate) {
>+    RTC_LOG(LS_INFO)
>+        << "Received target at:" << ToString(update.target_rate->at_time)
>+        << ": rate=" << ToString(update.target_rate->target_rate) << ", rtt="
>+        << ToString(update.target_rate->network_estimate.round_trip_time);
>+    target->target_rate = update.target_rate;
>+  }
>+  for (const auto& probe : update.probe_cluster_configs) {
>+    target->probe_cluster_configs.push_back(probe);
>+    RTC_LOG(LS_INFO) << "Received probe at:" << ToString(probe.at_time)
>+                     << ": target=" << ToString(probe.target_data_rate);
>+  }
>+}
>+}  // namespace
>+
>+SentPacket SimpleTargetRateProducer::ProduceNext(
>+    const NetworkControlUpdate& cache,
>+    Timestamp current_time,
>+    TimeDelta time_delta) {
>+  DataRate actual_send_rate =
>+      std::max(cache.target_rate->target_rate, cache.pacer_config->pad_rate());
>+  SentPacket packet;
>+  packet.send_time = current_time;
>+  packet.size = time_delta * actual_send_rate;
>+  return packet;
>+}
>+
>+NetworkControllerTester::NetworkControllerTester(
>+    NetworkControllerFactoryInterface* factory,
>+    NetworkControllerConfig initial_config)
>+    : current_time_(Timestamp::seconds(100000)),
>+      packet_sequence_number_(1),
>+      accumulated_buffer_(TimeDelta::Zero()) {
>+  initial_config.constraints.at_time = current_time_;
>+  controller_ = factory->Create(initial_config);
>+  process_interval_ = factory->GetProcessInterval();
>+  ProcessInterval interval_msg;
>+  interval_msg.at_time = current_time_;
>+  Update(&state_, controller_->OnProcessInterval(interval_msg));
>+}
>+
>+NetworkControllerTester::~NetworkControllerTester() = default;
>+
>+void NetworkControllerTester::RunSimulation(TimeDelta duration,
>+                                            TimeDelta packet_interval,
>+                                            DataRate actual_bandwidth,
>+                                            TimeDelta propagation_delay,
>+                                            PacketProducer next_packet) {
>+  RTC_CHECK(actual_bandwidth.bps() > 0);
>+  Timestamp start_time = current_time_;
>+  Timestamp last_process_time = current_time_;
>+  while (current_time_ - start_time < duration) {
>+    bool send_packet = true;
>+    if (state_.congestion_window && state_.congestion_window->IsFinite()) {
>+      DataSize data_in_flight = DataSize::Zero();
>+      for (PacketResult& packet : outstanding_packets_)
>+        data_in_flight += packet.sent_packet->size;
>+      if (data_in_flight > *state_.congestion_window)
>+        send_packet = false;
>+    }
>+
>+    if (send_packet) {
>+      SentPacket sent_packet;
>+      sent_packet = next_packet(state_, current_time_, packet_interval);
>+      sent_packet.sequence_number = packet_sequence_number_++;
>+      sent_packet.data_in_flight = sent_packet.size;
>+      for (PacketResult& packet : outstanding_packets_)
>+        sent_packet.data_in_flight += packet.sent_packet->size;
>+      Update(&state_, controller_->OnSentPacket(sent_packet));
>+
>+      TimeDelta time_in_flight = sent_packet.size / actual_bandwidth;
>+      accumulated_buffer_ += time_in_flight;
>+      TimeDelta total_delay = propagation_delay + accumulated_buffer_;
>+      PacketResult result;
>+      result.sent_packet = sent_packet;
>+      result.receive_time = sent_packet.send_time + total_delay;
>+
>+      outstanding_packets_.push_back(result);
>+    }
>+
>+    TimeDelta buffer_consumed = std::min(accumulated_buffer_, packet_interval);
>+    accumulated_buffer_ -= buffer_consumed;
>+
>+    if (outstanding_packets_.size() >= 2 &&
>+        current_time_ >=
>+            outstanding_packets_[1].receive_time + propagation_delay) {
>+      TransportPacketsFeedback feedback;
>+      feedback.prior_in_flight = DataSize::Zero();
>+      for (PacketResult& packet : outstanding_packets_)
>+        feedback.prior_in_flight += packet.sent_packet->size;
>+      while (!outstanding_packets_.empty() &&
>+             current_time_ >= outstanding_packets_.front().receive_time +
>+                                  propagation_delay) {
>+        feedback.packet_feedbacks.push_back(outstanding_packets_.front());
>+        outstanding_packets_.pop_front();
>+      }
>+      feedback.feedback_time =
>+          feedback.packet_feedbacks.back().receive_time + propagation_delay;
>+      feedback.data_in_flight = DataSize::Zero();
>+      for (PacketResult& packet : outstanding_packets_)
>+        feedback.data_in_flight += packet.sent_packet->size;
>+      Update(&state_, controller_->OnTransportPacketsFeedback(feedback));
>+    }
>+    current_time_ += packet_interval;
>+    if (current_time_ - last_process_time > process_interval_) {
>+      ProcessInterval interval_msg;
>+      interval_msg.at_time = current_time_;
>+      Update(&state_, controller_->OnProcessInterval(interval_msg));
>+    }
>+  }
>+}
>+
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/test/network_control_tester.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/test/network_control_tester.h
>new file mode 100644
>index 00000000000..4dfcc1498ad
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/transport/test/network_control_tester.h
>@@ -0,0 +1,73 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_TRANSPORT_TEST_NETWORK_CONTROL_TESTER_H_
>+#define API_TRANSPORT_TEST_NETWORK_CONTROL_TESTER_H_
>+
>+#include <deque>
>+#include <functional>
>+#include <memory>
>+
>+#include "absl/types/optional.h"
>+#include "api/transport/network_control.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+// Produces one packet per time delta
>+class SimpleTargetRateProducer {
>+ public:
>+  static SentPacket ProduceNext(const NetworkControlUpdate& state,
>+                                Timestamp current_time,
>+                                TimeDelta time_delta);
>+};
>+
>+class NetworkControllerTester {
>+ public:
>+  // A PacketProducer is a function that takes a network control state, a
>+  // timestamp representing the expected send time and a time delta of the send
>+  // times (This allows the PacketProducer to be stateless). It returns a
>+  // SentPacket struct with actual send time and packet size.
>+  using PacketProducer = std::function<
>+      SentPacket(const NetworkControlUpdate&, Timestamp, TimeDelta)>;
>+  NetworkControllerTester(NetworkControllerFactoryInterface* factory,
>+                          NetworkControllerConfig initial_config);
>+  ~NetworkControllerTester();
>+
>+  // Runs the simulations for the given duration, the PacketProducer will be
>+  // called repeatedly based on the given packet interval and the network will
>+  // be simulated using given bandwidth and propagation delay. The simulation
>+  // will call the controller under test with OnSentPacket and
>+  // OnTransportPacketsFeedback.
>+
>+  // Note that OnTransportPacketsFeedback will only be called for
>+  // packets with resulting feedback time within the simulated duration. Packets
>+  // with later feedback time are saved and used in the next call to
>+  // RunSimulation where enough simulated time has passed.
>+  void RunSimulation(TimeDelta duration,
>+                     TimeDelta packet_interval,
>+                     DataRate actual_bandwidth,
>+                     TimeDelta propagation_delay,
>+                     PacketProducer next_packet);
>+  NetworkControlUpdate GetState() { return state_; }
>+
>+ private:
>+  std::unique_ptr<NetworkControllerInterface> controller_;
>+  TimeDelta process_interval_ = TimeDelta::PlusInfinity();
>+  Timestamp current_time_;
>+  int64_t packet_sequence_number_;
>+  TimeDelta accumulated_buffer_;
>+  std::deque<PacketResult> outstanding_packets_;
>+  NetworkControlUpdate state_;
>+};
>+}  // namespace test
>+}  // namespace webrtc
>+
>+#endif  // API_TRANSPORT_TEST_NETWORK_CONTROL_TESTER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/turncustomizer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/turncustomizer.h
>index 517abcc8f0d..85c4e77c9c0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/turncustomizer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/turncustomizer.h
>@@ -18,7 +18,6 @@ class PortInterface;
> class StunMessage;
> }  // namespace cricket
> 
>-
> namespace webrtc {
> 
> class TurnCustomizer {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/umametrics.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/umametrics.h
>index 0d66a5e51fe..88ab08c1bef 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/umametrics.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/umametrics.h
>@@ -17,29 +17,6 @@
> 
> namespace webrtc {
> 
>-// Used to specify which enum counter type we're incrementing in
>-// MetricsObserverInterface::IncrementEnumCounter.
>-enum PeerConnectionEnumCounterType {
>-  kEnumCounterAddressFamily,
>-  // For the next 2 counters, we track them separately based on the "first hop"
>-  // protocol used by the local candidate. "First hop" means the local candidate
>-  // type in the case of non-TURN candidates, and the protocol used to connect
>-  // to the TURN server in the case of TURN candidates.
>-  kEnumCounterIceCandidatePairTypeUdp,
>-  kEnumCounterIceCandidatePairTypeTcp,
>-
>-  kEnumCounterAudioSrtpCipher,
>-  kEnumCounterAudioSslCipher,
>-  kEnumCounterVideoSrtpCipher,
>-  kEnumCounterVideoSslCipher,
>-  kEnumCounterDataSrtpCipher,
>-  kEnumCounterDataSslCipher,
>-  kEnumCounterDtlsHandshakeError,
>-  kEnumCounterIceRegathering,
>-  kEnumCounterIceRestart,
>-  kPeerConnectionEnumCounterMax
>-};
>-
> // Currently this contains information related to WebRTC network/transport
> // information.
> 
>@@ -111,26 +88,67 @@ enum IceCandidatePairType {
>   kIceCandidatePairMax
> };
> 
>-class MetricsObserverInterface : public rtc::RefCountInterface {
>- public:
>-  // |type| is the type of the enum counter to be incremented. |counter|
>-  // is the particular counter in that type. |counter_max| is the next sequence
>-  // number after the highest counter.
>-  virtual void IncrementEnumCounter(PeerConnectionEnumCounterType,
>-                                    int,
>-                                    int) {}
>-
>-  // This is used to handle sparse counters like SSL cipher suites.
>-  // TODO(guoweis): Remove the implementation once the dependency's interface
>-  // definition is updated.
>-  virtual void IncrementSparseEnumCounter(PeerConnectionEnumCounterType type,
>-                                          int counter);
>-
>-  virtual void AddHistogramSample(PeerConnectionMetricsName type,
>-                                  int value) = 0;
>+enum KeyExchangeProtocolType {
>+  kEnumCounterKeyProtocolDtls,
>+  kEnumCounterKeyProtocolSdes,
>+  kEnumCounterKeyProtocolMax
>+};
>+
>+enum KeyExchangeProtocolMedia {
>+  kEnumCounterKeyProtocolMediaTypeDtlsAudio,
>+  kEnumCounterKeyProtocolMediaTypeDtlsVideo,
>+  kEnumCounterKeyProtocolMediaTypeDtlsData,
>+  kEnumCounterKeyProtocolMediaTypeSdesAudio,
>+  kEnumCounterKeyProtocolMediaTypeSdesVideo,
>+  kEnumCounterKeyProtocolMediaTypeSdesData,
>+  kEnumCounterKeyProtocolMediaTypeMax
>+};
>+
>+enum SdpSemanticRequested {
>+  kSdpSemanticRequestDefault,
>+  kSdpSemanticRequestPlanB,
>+  kSdpSemanticRequestUnifiedPlan,
>+  kSdpSemanticRequestMax
> };
> 
>-typedef MetricsObserverInterface UMAObserver;
>+enum SdpSemanticNegotiated {
>+  kSdpSemanticNegotiatedNone,
>+  kSdpSemanticNegotiatedPlanB,
>+  kSdpSemanticNegotiatedUnifiedPlan,
>+  kSdpSemanticNegotiatedMixed,
>+  kSdpSemanticNegotiatedMax
>+};
>+
>+// Metric which records the format of the received SDP for tracking how much the
>+// difference between Plan B and Unified Plan affect users.
>+enum SdpFormatReceived {
>+  // No audio or video tracks. This is worth special casing since it seems to be
>+  // the most common scenario (data-channel only).
>+  kSdpFormatReceivedNoTracks,
>+  // No more than one audio and one video track. Should be compatible with both
>+  // Plan B and Unified Plan endpoints.
>+  kSdpFormatReceivedSimple,
>+  // More than one audio track or more than one video track in the Plan B format
>+  // (e.g., one audio media section with multiple streams).
>+  kSdpFormatReceivedComplexPlanB,
>+  // More than one audio track or more than one video track in the Unified Plan
>+  // format (e.g., two audio media sections).
>+  kSdpFormatReceivedComplexUnifiedPlan,
>+  kSdpFormatReceivedMax
>+};
>+
>+// Metric for counting the outcome of adding an ICE candidate
>+enum AddIceCandidateResult {
>+  kAddIceCandidateSuccess,
>+  kAddIceCandidateFailClosed,
>+  kAddIceCandidateFailNoRemoteDescription,
>+  kAddIceCandidateFailNullCandidate,
>+  kAddIceCandidateFailNotValid,
>+  kAddIceCandidateFailNotReady,
>+  kAddIceCandidateFailInAddition,
>+  kAddIceCandidateFailNotUsable,
>+  kAddIceCandidateMax
>+};
> 
> }  // namespace webrtc
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/BUILD.gn
>new file mode 100644
>index 00000000000..7dbddf4e622
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/BUILD.gn
>@@ -0,0 +1,78 @@
>+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+#
>+# Use of this source code is governed by a BSD-style license
>+# that can be found in the LICENSE file in the root of the source
>+# tree. An additional intellectual property rights grant can be found
>+# in the file PATENTS.  All contributing project authors may
>+# be found in the AUTHORS file in the root of the source tree.
>+
>+import("../../webrtc.gni")
>+
>+rtc_source_set("data_rate") {
>+  sources = [
>+    "data_rate.cc",
>+    "data_rate.h",
>+  ]
>+
>+  deps = [
>+    ":data_size",
>+    ":time_delta",
>+    "../../rtc_base:checks",
>+    "../../rtc_base:rtc_base_approved",
>+  ]
>+}
>+
>+rtc_source_set("data_size") {
>+  sources = [
>+    "data_size.cc",
>+    "data_size.h",
>+  ]
>+
>+  deps = [
>+    "../../rtc_base:checks",
>+    "../../rtc_base:rtc_base_approved",
>+  ]
>+}
>+rtc_source_set("time_delta") {
>+  sources = [
>+    "time_delta.cc",
>+    "time_delta.h",
>+  ]
>+
>+  deps = [
>+    "../../rtc_base:checks",
>+    "../../rtc_base:rtc_base_approved",
>+  ]
>+}
>+
>+rtc_source_set("timestamp") {
>+  sources = [
>+    "timestamp.cc",
>+    "timestamp.h",
>+  ]
>+
>+  deps = [
>+    ":time_delta",
>+    "../../rtc_base:checks",
>+    "../../rtc_base:rtc_base_approved",
>+  ]
>+}
>+
>+if (rtc_include_tests) {
>+  rtc_source_set("units_unittests") {
>+    testonly = true
>+    sources = [
>+      "data_rate_unittest.cc",
>+      "data_size_unittest.cc",
>+      "time_delta_unittest.cc",
>+      "timestamp_unittest.cc",
>+    ]
>+    deps = [
>+      ":data_rate",
>+      ":data_size",
>+      ":time_delta",
>+      ":timestamp",
>+      "../../test:test_support",
>+    ]
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/data_rate.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/data_rate.cc
>new file mode 100644
>index 00000000000..4e31d51f46e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/data_rate.cc
>@@ -0,0 +1,27 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/units/data_rate.h"
>+
>+#include "rtc_base/strings/string_builder.h"
>+
>+namespace webrtc {
>+
>+std::string ToString(const DataRate& value) {
>+  char buf[64];
>+  rtc::SimpleStringBuilder sb(buf);
>+  if (value.IsInfinite()) {
>+    sb << "inf bps";
>+  } else {
>+    sb << value.bps() << " bps";
>+  }
>+  return sb.str();
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/data_rate.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/data_rate.h
>new file mode 100644
>index 00000000000..47a143c8897
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/data_rate.h
>@@ -0,0 +1,202 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_UNITS_DATA_RATE_H_
>+#define API_UNITS_DATA_RATE_H_
>+#include <stdint.h>
>+#include <cmath>
>+#include <limits>
>+#include <string>
>+
>+#include "rtc_base/checks.h"
>+#include "rtc_base/numerics/safe_conversions.h"
>+
>+#include "api/units/data_size.h"
>+#include "api/units/time_delta.h"
>+
>+namespace webrtc {
>+namespace data_rate_impl {
>+constexpr int64_t kPlusInfinityVal = std::numeric_limits<int64_t>::max();
>+
>+inline int64_t Microbits(const DataSize& size) {
>+  constexpr int64_t kMaxBeforeConversion =
>+      std::numeric_limits<int64_t>::max() / 8000000;
>+  RTC_DCHECK_LE(size.bytes(), kMaxBeforeConversion)
>+      << "size is too large to be expressed in microbytes";
>+  return size.bytes() * 8000000;
>+}
>+}  // namespace data_rate_impl
>+
>+// DataRate is a class that represents a given data rate. This can be used to
>+// represent bandwidth, encoding bitrate, etc. The internal storage is bits per
>+// second (bps).
>+class DataRate {
>+ public:
>+  DataRate() = delete;
>+  static constexpr DataRate Zero() { return DataRate(0); }
>+  static constexpr DataRate Infinity() {
>+    return DataRate(data_rate_impl::kPlusInfinityVal);
>+  }
>+  template <int64_t bps>
>+  static constexpr DataRate BitsPerSec() {
>+    static_assert(bps >= 0, "");
>+    static_assert(bps < data_rate_impl::kPlusInfinityVal, "");
>+    return DataRate(bps);
>+  }
>+  template <int64_t kbps>
>+  static constexpr DataRate KilobitsPerSec() {
>+    static_assert(kbps >= 0, "");
>+    static_assert(kbps < data_rate_impl::kPlusInfinityVal / 1000, "");
>+    return DataRate(kbps * 1000);
>+  }
>+
>+  template <
>+      typename T,
>+      typename std::enable_if<std::is_integral<T>::value>::type* = nullptr>
>+  static DataRate bps(T bits_per_second) {
>+    RTC_DCHECK_GE(bits_per_second, 0);
>+    RTC_DCHECK_LT(bits_per_second, data_rate_impl::kPlusInfinityVal);
>+    return DataRate(rtc::dchecked_cast<int64_t>(bits_per_second));
>+  }
>+  template <
>+      typename T,
>+      typename std::enable_if<std::is_integral<T>::value>::type* = nullptr>
>+  static DataRate kbps(T kilobits_per_sec) {
>+    RTC_DCHECK_GE(kilobits_per_sec, 0);
>+    RTC_DCHECK_LT(kilobits_per_sec, data_rate_impl::kPlusInfinityVal / 1000);
>+    return DataRate::bps(rtc::dchecked_cast<int64_t>(kilobits_per_sec) * 1000);
>+  }
>+
>+  template <typename T,
>+            typename std::enable_if<std::is_floating_point<T>::value>::type* =
>+                nullptr>
>+  static DataRate bps(T bits_per_second) {
>+    if (bits_per_second == std::numeric_limits<T>::infinity()) {
>+      return Infinity();
>+    } else {
>+      RTC_DCHECK(!std::isnan(bits_per_second));
>+      RTC_DCHECK_GE(bits_per_second, 0);
>+      RTC_DCHECK_LT(bits_per_second, data_rate_impl::kPlusInfinityVal);
>+      return DataRate(rtc::dchecked_cast<int64_t>(bits_per_second));
>+    }
>+  }
>+  template <typename T,
>+            typename std::enable_if<std::is_floating_point<T>::value>::type* =
>+                nullptr>
>+  static DataRate kbps(T kilobits_per_sec) {
>+    return DataRate::bps(kilobits_per_sec * 1e3);
>+  }
>+
>+  template <typename T = int64_t>
>+  typename std::enable_if<std::is_integral<T>::value, T>::type bps() const {
>+    RTC_DCHECK(IsFinite());
>+    return rtc::dchecked_cast<T>(bits_per_sec_);
>+  }
>+  template <typename T = int64_t>
>+  typename std::enable_if<std::is_integral<T>::value, T>::type kbps() const {
>+    RTC_DCHECK(IsFinite());
>+    return rtc::dchecked_cast<T>(UnsafeKilobitsPerSec());
>+  }
>+
>+  template <typename T>
>+  typename std::enable_if<std::is_floating_point<T>::value,
>+                          T>::type constexpr bps() const {
>+    return IsInfinite() ? std::numeric_limits<T>::infinity() : bits_per_sec_;
>+  }
>+  template <typename T>
>+  typename std::enable_if<std::is_floating_point<T>::value,
>+                          T>::type constexpr kbps() const {
>+    return bps<T>() * 1e-3;
>+  }
>+
>+  constexpr int64_t bps_or(int64_t fallback_value) const {
>+    return IsFinite() ? bits_per_sec_ : fallback_value;
>+  }
>+  constexpr int64_t kbps_or(int64_t fallback_value) const {
>+    return IsFinite() ? UnsafeKilobitsPerSec() : fallback_value;
>+  }
>+
>+  constexpr bool IsZero() const { return bits_per_sec_ == 0; }
>+  constexpr bool IsInfinite() const {
>+    return bits_per_sec_ == data_rate_impl::kPlusInfinityVal;
>+  }
>+  constexpr bool IsFinite() const { return !IsInfinite(); }
>+
>+  constexpr double operator/(const DataRate& other) const {
>+    return bps<double>() / other.bps<double>();
>+  }
>+  constexpr bool operator==(const DataRate& other) const {
>+    return bits_per_sec_ == other.bits_per_sec_;
>+  }
>+  constexpr bool operator!=(const DataRate& other) const {
>+    return bits_per_sec_ != other.bits_per_sec_;
>+  }
>+  constexpr bool operator<=(const DataRate& other) const {
>+    return bits_per_sec_ <= other.bits_per_sec_;
>+  }
>+  constexpr bool operator>=(const DataRate& other) const {
>+    return bits_per_sec_ >= other.bits_per_sec_;
>+  }
>+  constexpr bool operator>(const DataRate& other) const {
>+    return bits_per_sec_ > other.bits_per_sec_;
>+  }
>+  constexpr bool operator<(const DataRate& other) const {
>+    return bits_per_sec_ < other.bits_per_sec_;
>+  }
>+
>+ private:
>+  // Bits per second used internally to simplify debugging by making the value
>+  // more recognizable.
>+  explicit constexpr DataRate(int64_t bits_per_second)
>+      : bits_per_sec_(bits_per_second) {}
>+  constexpr int64_t UnsafeKilobitsPerSec() const {
>+    return (bits_per_sec_ + 500) / 1000;
>+  }
>+  int64_t bits_per_sec_;
>+};
>+
>+inline DataRate operator*(const DataRate& rate, const double& scalar) {
>+  return DataRate::bps(std::round(rate.bps() * scalar));
>+}
>+inline DataRate operator*(const double& scalar, const DataRate& rate) {
>+  return rate * scalar;
>+}
>+inline DataRate operator*(const DataRate& rate, const int64_t& scalar) {
>+  return DataRate::bps(rate.bps() * scalar);
>+}
>+inline DataRate operator*(const int64_t& scalar, const DataRate& rate) {
>+  return rate * scalar;
>+}
>+inline DataRate operator*(const DataRate& rate, const int32_t& scalar) {
>+  return DataRate::bps(rate.bps() * scalar);
>+}
>+inline DataRate operator*(const int32_t& scalar, const DataRate& rate) {
>+  return rate * scalar;
>+}
>+
>+inline DataRate operator/(const DataSize& size, const TimeDelta& duration) {
>+  return DataRate::bps(data_rate_impl::Microbits(size) / duration.us());
>+}
>+inline TimeDelta operator/(const DataSize& size, const DataRate& rate) {
>+  return TimeDelta::us(data_rate_impl::Microbits(size) / rate.bps());
>+}
>+inline DataSize operator*(const DataRate& rate, const TimeDelta& duration) {
>+  int64_t microbits = rate.bps() * duration.us();
>+  return DataSize::bytes((microbits + 4000000) / 8000000);
>+}
>+inline DataSize operator*(const TimeDelta& duration, const DataRate& rate) {
>+  return rate * duration;
>+}
>+
>+std::string ToString(const DataRate& value);
>+
>+}  // namespace webrtc
>+
>+#endif  // API_UNITS_DATA_RATE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/data_rate_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/data_rate_unittest.cc
>new file mode 100644
>index 00000000000..9c91fd62470
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/data_rate_unittest.cc
>@@ -0,0 +1,142 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/units/data_rate.h"
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+TEST(DataRateTest, ConstExpr) {
>+  constexpr int64_t kValue = 12345;
>+  constexpr DataRate kDataRateZero = DataRate::Zero();
>+  constexpr DataRate kDataRateInf = DataRate::Infinity();
>+  static_assert(kDataRateZero.IsZero(), "");
>+  static_assert(kDataRateInf.IsInfinite(), "");
>+  static_assert(kDataRateInf.bps_or(-1) == -1, "");
>+  static_assert(kDataRateInf > kDataRateZero, "");
>+
>+  constexpr DataRate kDataRateBps = DataRate::BitsPerSec<kValue>();
>+  constexpr DataRate kDataRateKbps = DataRate::KilobitsPerSec<kValue>();
>+  static_assert(kDataRateBps.bps<double>() == kValue, "");
>+  static_assert(kDataRateBps.bps_or(0) == kValue, "");
>+  static_assert(kDataRateKbps.kbps_or(0) == kValue, "");
>+}
>+
>+TEST(DataRateTest, GetBackSameValues) {
>+  const int64_t kValue = 123 * 8;
>+  EXPECT_EQ(DataRate::bps(kValue).bps(), kValue);
>+  EXPECT_EQ(DataRate::kbps(kValue).kbps(), kValue);
>+}
>+
>+TEST(DataRateTest, GetDifferentPrefix) {
>+  const int64_t kValue = 123 * 8000;
>+  EXPECT_EQ(DataRate::bps(kValue).kbps(), kValue / 1000);
>+}
>+
>+TEST(DataRateTest, IdentityChecks) {
>+  const int64_t kValue = 3000;
>+  EXPECT_TRUE(DataRate::Zero().IsZero());
>+  EXPECT_FALSE(DataRate::bps(kValue).IsZero());
>+
>+  EXPECT_TRUE(DataRate::Infinity().IsInfinite());
>+  EXPECT_FALSE(DataRate::Zero().IsInfinite());
>+  EXPECT_FALSE(DataRate::bps(kValue).IsInfinite());
>+
>+  EXPECT_FALSE(DataRate::Infinity().IsFinite());
>+  EXPECT_TRUE(DataRate::bps(kValue).IsFinite());
>+  EXPECT_TRUE(DataRate::Zero().IsFinite());
>+}
>+
>+TEST(DataRateTest, ComparisonOperators) {
>+  const int64_t kSmall = 450;
>+  const int64_t kLarge = 451;
>+  const DataRate small = DataRate::bps(kSmall);
>+  const DataRate large = DataRate::bps(kLarge);
>+
>+  EXPECT_EQ(DataRate::Zero(), DataRate::bps(0));
>+  EXPECT_EQ(DataRate::Infinity(), DataRate::Infinity());
>+  EXPECT_EQ(small, small);
>+  EXPECT_LE(small, small);
>+  EXPECT_GE(small, small);
>+  EXPECT_NE(small, large);
>+  EXPECT_LE(small, large);
>+  EXPECT_LT(small, large);
>+  EXPECT_GE(large, small);
>+  EXPECT_GT(large, small);
>+  EXPECT_LT(DataRate::Zero(), small);
>+  EXPECT_GT(DataRate::Infinity(), large);
>+}
>+
>+TEST(DataRateTest, ConvertsToAndFromDouble) {
>+  const int64_t kValue = 128;
>+  const double kDoubleValue = static_cast<double>(kValue);
>+  const double kDoubleKbps = kValue * 1e-3;
>+  const double kFloatKbps = static_cast<float>(kDoubleKbps);
>+
>+  EXPECT_EQ(DataRate::bps(kValue).bps<double>(), kDoubleValue);
>+  EXPECT_EQ(DataRate::bps(kValue).kbps<double>(), kDoubleKbps);
>+  EXPECT_EQ(DataRate::bps(kValue).kbps<float>(), kFloatKbps);
>+  EXPECT_EQ(DataRate::bps(kDoubleValue).bps(), kValue);
>+  EXPECT_EQ(DataRate::kbps(kDoubleKbps).bps(), kValue);
>+
>+  const double kInfinity = std::numeric_limits<double>::infinity();
>+  EXPECT_EQ(DataRate::Infinity().bps<double>(), kInfinity);
>+  EXPECT_TRUE(DataRate::bps(kInfinity).IsInfinite());
>+  EXPECT_TRUE(DataRate::kbps(kInfinity).IsInfinite());
>+}
>+
>+TEST(DataRateTest, MathOperations) {
>+  const int64_t kValueA = 450;
>+  const int64_t kValueB = 267;
>+  const DataRate rate_a = DataRate::bps(kValueA);
>+  const DataRate rate_b = DataRate::bps(kValueB);
>+  const int32_t kInt32Value = 123;
>+  const double kFloatValue = 123.0;
>+  EXPECT_EQ((rate_a * kValueB).bps(), kValueA * kValueB);
>+  EXPECT_EQ((rate_a * kInt32Value).bps(), kValueA * kInt32Value);
>+  EXPECT_EQ((rate_a * kFloatValue).bps(), kValueA * kFloatValue);
>+
>+  EXPECT_EQ(rate_a / rate_b, static_cast<double>(kValueA) / kValueB);
>+}
>+
>+TEST(UnitConversionTest, DataRateAndDataSizeAndTimeDelta) {
>+  const int64_t kSeconds = 5;
>+  const int64_t kBitsPerSecond = 440;
>+  const int64_t kBytes = 44000;
>+  const TimeDelta delta_a = TimeDelta::seconds(kSeconds);
>+  const DataRate rate_b = DataRate::bps(kBitsPerSecond);
>+  const DataSize size_c = DataSize::bytes(kBytes);
>+  EXPECT_EQ((delta_a * rate_b).bytes(), kSeconds * kBitsPerSecond / 8);
>+  EXPECT_EQ((rate_b * delta_a).bytes(), kSeconds * kBitsPerSecond / 8);
>+  EXPECT_EQ((size_c / delta_a).bps(), kBytes * 8 / kSeconds);
>+  EXPECT_EQ((size_c / rate_b).seconds(), kBytes * 8 / kBitsPerSecond);
>+}
>+
>+TEST(UnitConversionTest, DivisionFailsOnLargeSize) {
>+  // Note that the failure is expected since the current implementation  is
>+  // implementated in a way that does not support division of large sizes. If
>+  // the implementation is changed, this test can safely be removed.
>+  const int64_t kJustSmallEnoughForDivision =
>+      std::numeric_limits<int64_t>::max() / 8000000;
>+  const DataSize large_size = DataSize::bytes(kJustSmallEnoughForDivision);
>+  const DataRate data_rate = DataRate::kbps(100);
>+  const TimeDelta time_delta = TimeDelta::ms(100);
>+  EXPECT_TRUE((large_size / data_rate).IsFinite());
>+  EXPECT_TRUE((large_size / time_delta).IsFinite());
>+#if GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID) && RTC_DCHECK_IS_ON
>+  const int64_t kToolargeForDivision = kJustSmallEnoughForDivision + 1;
>+  const DataSize too_large_size = DataSize::bytes(kToolargeForDivision);
>+  EXPECT_DEATH(too_large_size / data_rate, "");
>+  EXPECT_DEATH(too_large_size / time_delta, "");
>+#endif  // GTEST_HAS_DEATH_TEST && !!defined(WEBRTC_ANDROID) && RTC_DCHECK_IS_ON
>+}
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/data_size.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/data_size.cc
>new file mode 100644
>index 00000000000..4440f89d03c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/data_size.cc
>@@ -0,0 +1,27 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/units/data_size.h"
>+
>+#include "rtc_base/strings/string_builder.h"
>+
>+namespace webrtc {
>+
>+std::string ToString(const DataSize& value) {
>+  char buf[64];
>+  rtc::SimpleStringBuilder sb(buf);
>+  if (value.IsInfinite()) {
>+    sb << "inf bytes";
>+  } else {
>+    sb << value.bytes() << " bytes";
>+  }
>+  return sb.str();
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/data_size.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/data_size.h
>new file mode 100644
>index 00000000000..00ab2eccf73
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/data_size.h
>@@ -0,0 +1,154 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_UNITS_DATA_SIZE_H_
>+#define API_UNITS_DATA_SIZE_H_
>+
>+#include <stdint.h>
>+#include <cmath>
>+#include <limits>
>+#include <string>
>+#include <type_traits>
>+
>+#include "rtc_base/checks.h"
>+#include "rtc_base/numerics/safe_conversions.h"
>+
>+namespace webrtc {
>+namespace data_size_impl {
>+constexpr int64_t kPlusInfinityVal = std::numeric_limits<int64_t>::max();
>+}  // namespace data_size_impl
>+
>+// DataSize is a class represeting a count of bytes.
>+class DataSize {
>+ public:
>+  DataSize() = delete;
>+  static constexpr DataSize Zero() { return DataSize(0); }
>+  static constexpr DataSize Infinity() {
>+    return DataSize(data_size_impl::kPlusInfinityVal);
>+  }
>+  template <int64_t bytes>
>+  static constexpr DataSize Bytes() {
>+    static_assert(bytes >= 0, "");
>+    static_assert(bytes < data_size_impl::kPlusInfinityVal, "");
>+    return DataSize(bytes);
>+  }
>+
>+  template <
>+      typename T,
>+      typename std::enable_if<std::is_integral<T>::value>::type* = nullptr>
>+  static DataSize bytes(T bytes) {
>+    RTC_DCHECK_GE(bytes, 0);
>+    RTC_DCHECK_LT(bytes, data_size_impl::kPlusInfinityVal);
>+    return DataSize(rtc::dchecked_cast<int64_t>(bytes));
>+  }
>+
>+  template <typename T,
>+            typename std::enable_if<std::is_floating_point<T>::value>::type* =
>+                nullptr>
>+  static DataSize bytes(T bytes) {
>+    if (bytes == std::numeric_limits<T>::infinity()) {
>+      return Infinity();
>+    } else {
>+      RTC_DCHECK(!std::isnan(bytes));
>+      RTC_DCHECK_GE(bytes, 0);
>+      RTC_DCHECK_LT(bytes, data_size_impl::kPlusInfinityVal);
>+      return DataSize(rtc::dchecked_cast<int64_t>(bytes));
>+    }
>+  }
>+
>+  template <typename T = int64_t>
>+  typename std::enable_if<std::is_integral<T>::value, T>::type bytes() const {
>+    RTC_DCHECK(IsFinite());
>+    return rtc::dchecked_cast<T>(bytes_);
>+  }
>+
>+  template <typename T>
>+  constexpr typename std::enable_if<std::is_floating_point<T>::value, T>::type
>+  bytes() const {
>+    return IsInfinite() ? std::numeric_limits<T>::infinity() : bytes_;
>+  }
>+
>+  constexpr int64_t bytes_or(int64_t fallback_value) const {
>+    return IsFinite() ? bytes_ : fallback_value;
>+  }
>+
>+  constexpr bool IsZero() const { return bytes_ == 0; }
>+  constexpr bool IsInfinite() const {
>+    return bytes_ == data_size_impl::kPlusInfinityVal;
>+  }
>+  constexpr bool IsFinite() const { return !IsInfinite(); }
>+  DataSize operator-(const DataSize& other) const {
>+    return DataSize::bytes(bytes() - other.bytes());
>+  }
>+  DataSize operator+(const DataSize& other) const {
>+    return DataSize::bytes(bytes() + other.bytes());
>+  }
>+  DataSize& operator-=(const DataSize& other) {
>+    bytes_ -= other.bytes();
>+    return *this;
>+  }
>+  DataSize& operator+=(const DataSize& other) {
>+    bytes_ += other.bytes();
>+    return *this;
>+  }
>+  constexpr double operator/(const DataSize& other) const {
>+    return bytes<double>() / other.bytes<double>();
>+  }
>+  constexpr bool operator==(const DataSize& other) const {
>+    return bytes_ == other.bytes_;
>+  }
>+  constexpr bool operator!=(const DataSize& other) const {
>+    return bytes_ != other.bytes_;
>+  }
>+  constexpr bool operator<=(const DataSize& other) const {
>+    return bytes_ <= other.bytes_;
>+  }
>+  constexpr bool operator>=(const DataSize& other) const {
>+    return bytes_ >= other.bytes_;
>+  }
>+  constexpr bool operator>(const DataSize& other) const {
>+    return bytes_ > other.bytes_;
>+  }
>+  constexpr bool operator<(const DataSize& other) const {
>+    return bytes_ < other.bytes_;
>+  }
>+
>+ private:
>+  explicit constexpr DataSize(int64_t bytes) : bytes_(bytes) {}
>+  int64_t bytes_;
>+};
>+
>+inline DataSize operator*(const DataSize& size, const double& scalar) {
>+  return DataSize::bytes(std::round(size.bytes() * scalar));
>+}
>+inline DataSize operator*(const double& scalar, const DataSize& size) {
>+  return size * scalar;
>+}
>+inline DataSize operator*(const DataSize& size, const int64_t& scalar) {
>+  return DataSize::bytes(size.bytes() * scalar);
>+}
>+inline DataSize operator*(const int64_t& scalar, const DataSize& size) {
>+  return size * scalar;
>+}
>+inline DataSize operator*(const DataSize& size, const int32_t& scalar) {
>+  return DataSize::bytes(size.bytes() * scalar);
>+}
>+inline DataSize operator*(const int32_t& scalar, const DataSize& size) {
>+  return size * scalar;
>+}
>+inline DataSize operator/(const DataSize& size, const int64_t& scalar) {
>+  return DataSize::bytes(size.bytes() / scalar);
>+}
>+
>+std::string ToString(const DataSize& value);
>+
>+}  // namespace webrtc
>+
>+#endif  // API_UNITS_DATA_SIZE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/data_size_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/data_size_unittest.cc
>new file mode 100644
>index 00000000000..fe7f591dc0d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/data_size_unittest.cc
>@@ -0,0 +1,107 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/units/data_size.h"
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+TEST(DataSizeTest, ConstExpr) {
>+  constexpr int64_t kValue = 12345;
>+  constexpr DataSize kDataSizeZero = DataSize::Zero();
>+  constexpr DataSize kDataSizeInf = DataSize::Infinity();
>+  static_assert(kDataSizeZero.IsZero(), "");
>+  static_assert(kDataSizeInf.IsInfinite(), "");
>+  static_assert(kDataSizeInf.bytes_or(-1) == -1, "");
>+  static_assert(kDataSizeInf > kDataSizeZero, "");
>+
>+  constexpr DataSize kDataSize = DataSize::Bytes<kValue>();
>+  static_assert(kDataSize.bytes_or(-1) == kValue, "");
>+
>+  EXPECT_EQ(kDataSize.bytes(), kValue);
>+}
>+
>+TEST(DataSizeTest, GetBackSameValues) {
>+  const int64_t kValue = 123 * 8;
>+  EXPECT_EQ(DataSize::bytes(kValue).bytes(), kValue);
>+}
>+
>+TEST(DataSizeTest, IdentityChecks) {
>+  const int64_t kValue = 3000;
>+  EXPECT_TRUE(DataSize::Zero().IsZero());
>+  EXPECT_FALSE(DataSize::bytes(kValue).IsZero());
>+
>+  EXPECT_TRUE(DataSize::Infinity().IsInfinite());
>+  EXPECT_FALSE(DataSize::Zero().IsInfinite());
>+  EXPECT_FALSE(DataSize::bytes(kValue).IsInfinite());
>+
>+  EXPECT_FALSE(DataSize::Infinity().IsFinite());
>+  EXPECT_TRUE(DataSize::bytes(kValue).IsFinite());
>+  EXPECT_TRUE(DataSize::Zero().IsFinite());
>+}
>+
>+TEST(DataSizeTest, ComparisonOperators) {
>+  const int64_t kSmall = 450;
>+  const int64_t kLarge = 451;
>+  const DataSize small = DataSize::bytes(kSmall);
>+  const DataSize large = DataSize::bytes(kLarge);
>+
>+  EXPECT_EQ(DataSize::Zero(), DataSize::bytes(0));
>+  EXPECT_EQ(DataSize::Infinity(), DataSize::Infinity());
>+  EXPECT_EQ(small, small);
>+  EXPECT_LE(small, small);
>+  EXPECT_GE(small, small);
>+  EXPECT_NE(small, large);
>+  EXPECT_LE(small, large);
>+  EXPECT_LT(small, large);
>+  EXPECT_GE(large, small);
>+  EXPECT_GT(large, small);
>+  EXPECT_LT(DataSize::Zero(), small);
>+  EXPECT_GT(DataSize::Infinity(), large);
>+}
>+
>+TEST(DataSizeTest, ConvertsToAndFromDouble) {
>+  const int64_t kValue = 128;
>+  const double kDoubleValue = static_cast<double>(kValue);
>+
>+  EXPECT_EQ(DataSize::bytes(kValue).bytes<double>(), kDoubleValue);
>+  EXPECT_EQ(DataSize::bytes(kDoubleValue).bytes(), kValue);
>+
>+  const double kInfinity = std::numeric_limits<double>::infinity();
>+  EXPECT_EQ(DataSize::Infinity().bytes<double>(), kInfinity);
>+  EXPECT_TRUE(DataSize::bytes(kInfinity).IsInfinite());
>+}
>+
>+TEST(DataSizeTest, MathOperations) {
>+  const int64_t kValueA = 450;
>+  const int64_t kValueB = 267;
>+  const DataSize size_a = DataSize::bytes(kValueA);
>+  const DataSize size_b = DataSize::bytes(kValueB);
>+  EXPECT_EQ((size_a + size_b).bytes(), kValueA + kValueB);
>+  EXPECT_EQ((size_a - size_b).bytes(), kValueA - kValueB);
>+
>+  const int32_t kInt32Value = 123;
>+  const double kFloatValue = 123.0;
>+  EXPECT_EQ((size_a * kValueB).bytes(), kValueA * kValueB);
>+  EXPECT_EQ((size_a * kInt32Value).bytes(), kValueA * kInt32Value);
>+  EXPECT_EQ((size_a * kFloatValue).bytes(), kValueA * kFloatValue);
>+
>+  EXPECT_EQ((size_a / 10).bytes(), kValueA / 10);
>+  EXPECT_EQ(size_a / size_b, static_cast<double>(kValueA) / kValueB);
>+
>+  DataSize mutable_size = DataSize::bytes(kValueA);
>+  mutable_size += size_b;
>+  EXPECT_EQ(mutable_size.bytes(), kValueA + kValueB);
>+  mutable_size -= size_a;
>+  EXPECT_EQ(mutable_size.bytes(), kValueB);
>+}
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/time_delta.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/time_delta.cc
>new file mode 100644
>index 00000000000..398df776598
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/time_delta.cc
>@@ -0,0 +1,29 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/units/time_delta.h"
>+
>+#include "rtc_base/strings/string_builder.h"
>+
>+namespace webrtc {
>+
>+std::string ToString(const TimeDelta& value) {
>+  char buf[64];
>+  rtc::SimpleStringBuilder sb(buf);
>+  if (value.IsPlusInfinity()) {
>+    sb << "+inf ms";
>+  } else if (value.IsMinusInfinity()) {
>+    sb << "-inf ms";
>+  } else {
>+    sb << value.ms() << " ms";
>+  }
>+  return sb.str();
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/time_delta.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/time_delta.h
>new file mode 100644
>index 00000000000..1155eb526be
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/time_delta.h
>@@ -0,0 +1,258 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_UNITS_TIME_DELTA_H_
>+#define API_UNITS_TIME_DELTA_H_
>+
>+#include <stdint.h>
>+#include <cmath>
>+#include <limits>
>+#include <string>
>+
>+#include "rtc_base/checks.h"
>+#include "rtc_base/numerics/safe_conversions.h"
>+
>+namespace webrtc {
>+namespace timedelta_impl {
>+constexpr int64_t kPlusInfinityVal = std::numeric_limits<int64_t>::max();
>+constexpr int64_t kMinusInfinityVal = std::numeric_limits<int64_t>::min();
>+}  // namespace timedelta_impl
>+
>+// TimeDelta represents the difference between two timestamps. Commonly this can
>+// be a duration. However since two Timestamps are not guaranteed to have the
>+// same epoch (they might come from different computers, making exact
>+// synchronisation infeasible), the duration covered by a TimeDelta can be
>+// undefined. To simplify usage, it can be constructed and converted to
>+// different units, specifically seconds (s), milliseconds (ms) and
>+// microseconds (us).
>+class TimeDelta {
>+ public:
>+  TimeDelta() = delete;
>+  static constexpr TimeDelta Zero() { return TimeDelta(0); }
>+  static constexpr TimeDelta PlusInfinity() {
>+    return TimeDelta(timedelta_impl::kPlusInfinityVal);
>+  }
>+  static constexpr TimeDelta MinusInfinity() {
>+    return TimeDelta(timedelta_impl::kMinusInfinityVal);
>+  }
>+  template <int64_t seconds>
>+  static constexpr TimeDelta Seconds() {
>+    static_assert(seconds > timedelta_impl::kMinusInfinityVal / 1000000, "");
>+    static_assert(seconds < timedelta_impl::kPlusInfinityVal / 1000000, "");
>+    return TimeDelta(seconds * 1000000);
>+  }
>+  template <int64_t ms>
>+  static constexpr TimeDelta Millis() {
>+    static_assert(ms > timedelta_impl::kMinusInfinityVal / 1000, "");
>+    static_assert(ms < timedelta_impl::kPlusInfinityVal / 1000, "");
>+    return TimeDelta(ms * 1000);
>+  }
>+  template <int64_t us>
>+  static constexpr TimeDelta Micros() {
>+    static_assert(us > timedelta_impl::kMinusInfinityVal, "");
>+    static_assert(us < timedelta_impl::kPlusInfinityVal, "");
>+    return TimeDelta(us);
>+  }
>+
>+  template <
>+      typename T,
>+      typename std::enable_if<std::is_integral<T>::value>::type* = nullptr>
>+  static TimeDelta seconds(T seconds) {
>+    RTC_DCHECK_GT(seconds, timedelta_impl::kMinusInfinityVal / 1000000);
>+    RTC_DCHECK_LT(seconds, timedelta_impl::kPlusInfinityVal / 1000000);
>+    return TimeDelta(rtc::dchecked_cast<int64_t>(seconds) * 1000000);
>+  }
>+  template <
>+      typename T,
>+      typename std::enable_if<std::is_integral<T>::value>::type* = nullptr>
>+  static TimeDelta ms(T milliseconds) {
>+    RTC_DCHECK_GT(milliseconds, timedelta_impl::kMinusInfinityVal / 1000);
>+    RTC_DCHECK_LT(milliseconds, timedelta_impl::kPlusInfinityVal / 1000);
>+    return TimeDelta(rtc::dchecked_cast<int64_t>(milliseconds) * 1000);
>+  }
>+  template <
>+      typename T,
>+      typename std::enable_if<std::is_integral<T>::value>::type* = nullptr>
>+  static TimeDelta us(T microseconds) {
>+    RTC_DCHECK_GT(microseconds, timedelta_impl::kMinusInfinityVal);
>+    RTC_DCHECK_LT(microseconds, timedelta_impl::kPlusInfinityVal);
>+    return TimeDelta(rtc::dchecked_cast<int64_t>(microseconds));
>+  }
>+
>+  template <typename T,
>+            typename std::enable_if<std::is_floating_point<T>::value>::type* =
>+                nullptr>
>+  static TimeDelta seconds(T seconds) {
>+    return TimeDelta::us(seconds * 1e6);
>+  }
>+  template <typename T,
>+            typename std::enable_if<std::is_floating_point<T>::value>::type* =
>+                nullptr>
>+  static TimeDelta ms(T milliseconds) {
>+    return TimeDelta::us(milliseconds * 1e3);
>+  }
>+  template <typename T,
>+            typename std::enable_if<std::is_floating_point<T>::value>::type* =
>+                nullptr>
>+  static TimeDelta us(T microseconds) {
>+    if (microseconds == std::numeric_limits<T>::infinity()) {
>+      return PlusInfinity();
>+    } else if (microseconds == -std::numeric_limits<T>::infinity()) {
>+      return MinusInfinity();
>+    } else {
>+      RTC_DCHECK(!std::isnan(microseconds));
>+      RTC_DCHECK_GT(microseconds, timedelta_impl::kMinusInfinityVal);
>+      RTC_DCHECK_LT(microseconds, timedelta_impl::kPlusInfinityVal);
>+      return TimeDelta(rtc::dchecked_cast<int64_t>(microseconds));
>+    }
>+  }
>+
>+  template <typename T = int64_t>
>+  typename std::enable_if<std::is_integral<T>::value, T>::type seconds() const {
>+    RTC_DCHECK(IsFinite());
>+    return rtc::dchecked_cast<T>(UnsafeSeconds());
>+  }
>+  template <typename T = int64_t>
>+  typename std::enable_if<std::is_integral<T>::value, T>::type ms() const {
>+    RTC_DCHECK(IsFinite());
>+    return rtc::dchecked_cast<T>(UnsafeMillis());
>+  }
>+  template <typename T = int64_t>
>+  typename std::enable_if<std::is_integral<T>::value, T>::type us() const {
>+    RTC_DCHECK(IsFinite());
>+    return rtc::dchecked_cast<T>(microseconds_);
>+  }
>+  template <typename T = int64_t>
>+  typename std::enable_if<std::is_integral<T>::value, T>::type ns() const {
>+    RTC_DCHECK_GE(us(), std::numeric_limits<T>::min() / 1000);
>+    RTC_DCHECK_LE(us(), std::numeric_limits<T>::max() / 1000);
>+    return rtc::dchecked_cast<T>(us() * 1000);
>+  }
>+
>+  template <typename T>
>+  constexpr typename std::enable_if<std::is_floating_point<T>::value, T>::type
>+  seconds() const {
>+    return us<T>() * 1e-6;
>+  }
>+  template <typename T>
>+  constexpr typename std::enable_if<std::is_floating_point<T>::value, T>::type
>+  ms() const {
>+    return us<T>() * 1e-3;
>+  }
>+  template <typename T>
>+  constexpr typename std::enable_if<std::is_floating_point<T>::value, T>::type
>+  us() const {
>+    return IsPlusInfinity()
>+               ? std::numeric_limits<T>::infinity()
>+               : IsMinusInfinity() ? -std::numeric_limits<T>::infinity()
>+                                   : microseconds_;
>+  }
>+  template <typename T>
>+  constexpr typename std::enable_if<std::is_floating_point<T>::value, T>::type
>+  ns() const {
>+    return us<T>() * 1e3;
>+  }
>+
>+  constexpr int64_t seconds_or(int64_t fallback_value) const {
>+    return IsFinite() ? UnsafeSeconds() : fallback_value;
>+  }
>+  constexpr int64_t ms_or(int64_t fallback_value) const {
>+    return IsFinite() ? UnsafeMillis() : fallback_value;
>+  }
>+  constexpr int64_t us_or(int64_t fallback_value) const {
>+    return IsFinite() ? microseconds_ : fallback_value;
>+  }
>+
>+  TimeDelta Abs() const { return TimeDelta::us(std::abs(us())); }
>+  constexpr bool IsZero() const { return microseconds_ == 0; }
>+  constexpr bool IsFinite() const { return !IsInfinite(); }
>+  constexpr bool IsInfinite() const {
>+    return microseconds_ == timedelta_impl::kPlusInfinityVal ||
>+           microseconds_ == timedelta_impl::kMinusInfinityVal;
>+  }
>+  constexpr bool IsPlusInfinity() const {
>+    return microseconds_ == timedelta_impl::kPlusInfinityVal;
>+  }
>+  constexpr bool IsMinusInfinity() const {
>+    return microseconds_ == timedelta_impl::kMinusInfinityVal;
>+  }
>+  TimeDelta operator+(const TimeDelta& other) const {
>+    return TimeDelta::us(us() + other.us());
>+  }
>+  TimeDelta operator-(const TimeDelta& other) const {
>+    return TimeDelta::us(us() - other.us());
>+  }
>+  TimeDelta& operator-=(const TimeDelta& other) {
>+    microseconds_ -= other.us();
>+    return *this;
>+  }
>+  TimeDelta& operator+=(const TimeDelta& other) {
>+    microseconds_ += other.us();
>+    return *this;
>+  }
>+  constexpr double operator/(const TimeDelta& other) const {
>+    return us<double>() / other.us<double>();
>+  }
>+  constexpr bool operator==(const TimeDelta& other) const {
>+    return microseconds_ == other.microseconds_;
>+  }
>+  constexpr bool operator!=(const TimeDelta& other) const {
>+    return microseconds_ != other.microseconds_;
>+  }
>+  constexpr bool operator<=(const TimeDelta& other) const {
>+    return microseconds_ <= other.microseconds_;
>+  }
>+  constexpr bool operator>=(const TimeDelta& other) const {
>+    return microseconds_ >= other.microseconds_;
>+  }
>+  constexpr bool operator>(const TimeDelta& other) const {
>+    return microseconds_ > other.microseconds_;
>+  }
>+  constexpr bool operator<(const TimeDelta& other) const {
>+    return microseconds_ < other.microseconds_;
>+  }
>+
>+ private:
>+  explicit constexpr TimeDelta(int64_t us) : microseconds_(us) {}
>+  constexpr int64_t UnsafeSeconds() const {
>+    return (microseconds_ + (microseconds_ >= 0 ? 500000 : -500000)) / 1000000;
>+  }
>+  constexpr int64_t UnsafeMillis() const {
>+    return (microseconds_ + (microseconds_ >= 0 ? 500 : -500)) / 1000;
>+  }
>+  int64_t microseconds_;
>+};
>+
>+inline TimeDelta operator*(const TimeDelta& delta, const double& scalar) {
>+  return TimeDelta::us(std::round(delta.us() * scalar));
>+}
>+inline TimeDelta operator*(const double& scalar, const TimeDelta& delta) {
>+  return delta * scalar;
>+}
>+inline TimeDelta operator*(const TimeDelta& delta, const int64_t& scalar) {
>+  return TimeDelta::us(delta.us() * scalar);
>+}
>+inline TimeDelta operator*(const int64_t& scalar, const TimeDelta& delta) {
>+  return delta * scalar;
>+}
>+inline TimeDelta operator*(const TimeDelta& delta, const int32_t& scalar) {
>+  return TimeDelta::us(delta.us() * scalar);
>+}
>+inline TimeDelta operator*(const int32_t& scalar, const TimeDelta& delta) {
>+  return delta * scalar;
>+}
>+
>+inline TimeDelta operator/(const TimeDelta& delta, const int64_t& scalar) {
>+  return TimeDelta::us(delta.us() / scalar);
>+}
>+std::string ToString(const TimeDelta& value);
>+}  // namespace webrtc
>+
>+#endif  // API_UNITS_TIME_DELTA_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/time_delta_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/time_delta_unittest.cc
>new file mode 100644
>index 00000000000..9eddee73adf
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/time_delta_unittest.cc
>@@ -0,0 +1,169 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/units/time_delta.h"
>+
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+namespace test {
>+TEST(TimeDeltaTest, ConstExpr) {
>+  constexpr int64_t kValue = -12345;
>+  constexpr TimeDelta kTimeDeltaZero = TimeDelta::Zero();
>+  constexpr TimeDelta kTimeDeltaPlusInf = TimeDelta::PlusInfinity();
>+  constexpr TimeDelta kTimeDeltaMinusInf = TimeDelta::MinusInfinity();
>+  static_assert(kTimeDeltaZero.IsZero(), "");
>+  static_assert(kTimeDeltaPlusInf.IsPlusInfinity(), "");
>+  static_assert(kTimeDeltaMinusInf.IsMinusInfinity(), "");
>+  static_assert(kTimeDeltaPlusInf.ms_or(-1) == -1, "");
>+
>+  static_assert(kTimeDeltaPlusInf > kTimeDeltaZero, "");
>+
>+  constexpr TimeDelta kTimeDeltaSeconds = TimeDelta::Seconds<kValue>();
>+  constexpr TimeDelta kTimeDeltaMs = TimeDelta::Millis<kValue>();
>+  constexpr TimeDelta kTimeDeltaUs = TimeDelta::Micros<kValue>();
>+
>+  static_assert(kTimeDeltaSeconds.seconds_or(0) == kValue, "");
>+  static_assert(kTimeDeltaMs.ms_or(0) == kValue, "");
>+  static_assert(kTimeDeltaUs.us_or(0) == kValue, "");
>+}
>+
>+TEST(TimeDeltaTest, GetBackSameValues) {
>+  const int64_t kValue = 499;
>+  for (int sign = -1; sign <= 1; ++sign) {
>+    int64_t value = kValue * sign;
>+    EXPECT_EQ(TimeDelta::ms(value).ms(), value);
>+    EXPECT_EQ(TimeDelta::us(value).us(), value);
>+    EXPECT_EQ(TimeDelta::seconds(value).seconds(), value);
>+    EXPECT_EQ(TimeDelta::seconds(value).seconds(), value);
>+  }
>+  EXPECT_EQ(TimeDelta::Zero().us(), 0);
>+}
>+
>+TEST(TimeDeltaTest, GetDifferentPrefix) {
>+  const int64_t kValue = 3000000;
>+  EXPECT_EQ(TimeDelta::us(kValue).seconds(), kValue / 1000000);
>+  EXPECT_EQ(TimeDelta::ms(kValue).seconds(), kValue / 1000);
>+  EXPECT_EQ(TimeDelta::us(kValue).ms(), kValue / 1000);
>+
>+  EXPECT_EQ(TimeDelta::ms(kValue).us(), kValue * 1000);
>+  EXPECT_EQ(TimeDelta::seconds(kValue).ms(), kValue * 1000);
>+  EXPECT_EQ(TimeDelta::seconds(kValue).us(), kValue * 1000000);
>+}
>+
>+TEST(TimeDeltaTest, IdentityChecks) {
>+  const int64_t kValue = 3000;
>+  EXPECT_TRUE(TimeDelta::Zero().IsZero());
>+  EXPECT_FALSE(TimeDelta::ms(kValue).IsZero());
>+
>+  EXPECT_TRUE(TimeDelta::PlusInfinity().IsInfinite());
>+  EXPECT_TRUE(TimeDelta::MinusInfinity().IsInfinite());
>+  EXPECT_FALSE(TimeDelta::Zero().IsInfinite());
>+  EXPECT_FALSE(TimeDelta::ms(-kValue).IsInfinite());
>+  EXPECT_FALSE(TimeDelta::ms(kValue).IsInfinite());
>+
>+  EXPECT_FALSE(TimeDelta::PlusInfinity().IsFinite());
>+  EXPECT_FALSE(TimeDelta::MinusInfinity().IsFinite());
>+  EXPECT_TRUE(TimeDelta::ms(-kValue).IsFinite());
>+  EXPECT_TRUE(TimeDelta::ms(kValue).IsFinite());
>+  EXPECT_TRUE(TimeDelta::Zero().IsFinite());
>+}
>+
>+TEST(TimeDeltaTest, ComparisonOperators) {
>+  const int64_t kSmall = 450;
>+  const int64_t kLarge = 451;
>+  const TimeDelta small = TimeDelta::ms(kSmall);
>+  const TimeDelta large = TimeDelta::ms(kLarge);
>+
>+  EXPECT_EQ(TimeDelta::Zero(), TimeDelta::ms(0));
>+  EXPECT_EQ(TimeDelta::PlusInfinity(), TimeDelta::PlusInfinity());
>+  EXPECT_EQ(small, TimeDelta::ms(kSmall));
>+  EXPECT_LE(small, TimeDelta::ms(kSmall));
>+  EXPECT_GE(small, TimeDelta::ms(kSmall));
>+  EXPECT_NE(small, TimeDelta::ms(kLarge));
>+  EXPECT_LE(small, TimeDelta::ms(kLarge));
>+  EXPECT_LT(small, TimeDelta::ms(kLarge));
>+  EXPECT_GE(large, TimeDelta::ms(kSmall));
>+  EXPECT_GT(large, TimeDelta::ms(kSmall));
>+  EXPECT_LT(TimeDelta::Zero(), small);
>+  EXPECT_GT(TimeDelta::Zero(), TimeDelta::ms(-kSmall));
>+  EXPECT_GT(TimeDelta::Zero(), TimeDelta::ms(-kSmall));
>+
>+  EXPECT_GT(TimeDelta::PlusInfinity(), large);
>+  EXPECT_LT(TimeDelta::MinusInfinity(), TimeDelta::Zero());
>+}
>+
>+TEST(TimeDeltaTest, CanBeInititializedFromLargeInt) {
>+  const int kMaxInt = std::numeric_limits<int>::max();
>+  EXPECT_EQ(TimeDelta::seconds(kMaxInt).us(),
>+            static_cast<int64_t>(kMaxInt) * 1000000);
>+  EXPECT_EQ(TimeDelta::ms(kMaxInt).us(), static_cast<int64_t>(kMaxInt) * 1000);
>+}
>+
>+TEST(TimeDeltaTest, ConvertsToAndFromDouble) {
>+  const int64_t kMicros = 17017;
>+  const double kNanosDouble = kMicros * 1e3;
>+  const double kMicrosDouble = kMicros;
>+  const double kMillisDouble = kMicros * 1e-3;
>+  const double kSecondsDouble = kMillisDouble * 1e-3;
>+
>+  EXPECT_EQ(TimeDelta::us(kMicros).seconds<double>(), kSecondsDouble);
>+  EXPECT_EQ(TimeDelta::seconds(kSecondsDouble).us(), kMicros);
>+
>+  EXPECT_EQ(TimeDelta::us(kMicros).ms<double>(), kMillisDouble);
>+  EXPECT_EQ(TimeDelta::ms(kMillisDouble).us(), kMicros);
>+
>+  EXPECT_EQ(TimeDelta::us(kMicros).us<double>(), kMicrosDouble);
>+  EXPECT_EQ(TimeDelta::us(kMicrosDouble).us(), kMicros);
>+
>+  EXPECT_NEAR(TimeDelta::us(kMicros).ns<double>(), kNanosDouble, 1);
>+
>+  const double kPlusInfinity = std::numeric_limits<double>::infinity();
>+  const double kMinusInfinity = -kPlusInfinity;
>+
>+  EXPECT_EQ(TimeDelta::PlusInfinity().seconds<double>(), kPlusInfinity);
>+  EXPECT_EQ(TimeDelta::MinusInfinity().seconds<double>(), kMinusInfinity);
>+  EXPECT_EQ(TimeDelta::PlusInfinity().ms<double>(), kPlusInfinity);
>+  EXPECT_EQ(TimeDelta::MinusInfinity().ms<double>(), kMinusInfinity);
>+  EXPECT_EQ(TimeDelta::PlusInfinity().us<double>(), kPlusInfinity);
>+  EXPECT_EQ(TimeDelta::MinusInfinity().us<double>(), kMinusInfinity);
>+  EXPECT_EQ(TimeDelta::PlusInfinity().ns<double>(), kPlusInfinity);
>+  EXPECT_EQ(TimeDelta::MinusInfinity().ns<double>(), kMinusInfinity);
>+
>+  EXPECT_TRUE(TimeDelta::seconds(kPlusInfinity).IsPlusInfinity());
>+  EXPECT_TRUE(TimeDelta::seconds(kMinusInfinity).IsMinusInfinity());
>+  EXPECT_TRUE(TimeDelta::ms(kPlusInfinity).IsPlusInfinity());
>+  EXPECT_TRUE(TimeDelta::ms(kMinusInfinity).IsMinusInfinity());
>+  EXPECT_TRUE(TimeDelta::us(kPlusInfinity).IsPlusInfinity());
>+  EXPECT_TRUE(TimeDelta::us(kMinusInfinity).IsMinusInfinity());
>+}
>+
>+TEST(TimeDeltaTest, MathOperations) {
>+  const int64_t kValueA = 267;
>+  const int64_t kValueB = 450;
>+  const TimeDelta delta_a = TimeDelta::ms(kValueA);
>+  const TimeDelta delta_b = TimeDelta::ms(kValueB);
>+  EXPECT_EQ((delta_a + delta_b).ms(), kValueA + kValueB);
>+  EXPECT_EQ((delta_a - delta_b).ms(), kValueA - kValueB);
>+
>+  const int32_t kInt32Value = 123;
>+  const double kFloatValue = 123.0;
>+  EXPECT_EQ((TimeDelta::us(kValueA) * kValueB).us(), kValueA * kValueB);
>+  EXPECT_EQ((TimeDelta::us(kValueA) * kInt32Value).us(), kValueA * kInt32Value);
>+  EXPECT_EQ((TimeDelta::us(kValueA) * kFloatValue).us(), kValueA * kFloatValue);
>+
>+  EXPECT_EQ((delta_b / 10).ms(), kValueB / 10);
>+  EXPECT_EQ(delta_b / delta_a, static_cast<double>(kValueB) / kValueA);
>+
>+  EXPECT_EQ(TimeDelta::us(-kValueA).Abs().us(), kValueA);
>+  EXPECT_EQ(TimeDelta::us(kValueA).Abs().us(), kValueA);
>+}
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/timestamp.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/timestamp.cc
>new file mode 100644
>index 00000000000..4b2c44b9a73
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/timestamp.cc
>@@ -0,0 +1,26 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/units/timestamp.h"
>+
>+#include "rtc_base/strings/string_builder.h"
>+
>+namespace webrtc {
>+std::string ToString(const Timestamp& value) {
>+  char buf[64];
>+  rtc::SimpleStringBuilder sb(buf);
>+  if (value.IsInfinite()) {
>+    sb << "inf ms";
>+  } else {
>+    sb << value.ms() << " ms";
>+  }
>+  return sb.str();
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/timestamp.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/timestamp.h
>new file mode 100644
>index 00000000000..1b5e84ff601
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/timestamp.h
>@@ -0,0 +1,208 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_UNITS_TIMESTAMP_H_
>+#define API_UNITS_TIMESTAMP_H_
>+
>+#include <stdint.h>
>+#include <limits>
>+#include <string>
>+
>+#include "api/units/time_delta.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/numerics/safe_conversions.h"
>+
>+namespace webrtc {
>+namespace timestamp_impl {
>+constexpr int64_t kPlusInfinityVal = std::numeric_limits<int64_t>::max();
>+constexpr int64_t kMinusInfinityVal = std::numeric_limits<int64_t>::min();
>+}  // namespace timestamp_impl
>+
>+// Timestamp represents the time that has passed since some unspecified epoch.
>+// The epoch is assumed to be before any represented timestamps, this means that
>+// negative values are not valid. The most notable feature is that the
>+// difference of two Timestamps results in a TimeDelta.
>+class Timestamp {
>+ public:
>+  Timestamp() = delete;
>+  static constexpr Timestamp Infinity() {
>+    return Timestamp(timestamp_impl::kPlusInfinityVal);
>+  }
>+  template <int64_t seconds>
>+  static constexpr Timestamp Seconds() {
>+    static_assert(seconds >= 0, "");
>+    static_assert(seconds < timestamp_impl::kPlusInfinityVal / 1000000, "");
>+    return Timestamp(seconds * 1000000);
>+  }
>+  template <int64_t ms>
>+  static constexpr Timestamp Millis() {
>+    static_assert(ms >= 0, "");
>+    static_assert(ms < timestamp_impl::kPlusInfinityVal / 1000, "");
>+    return Timestamp(ms * 1000);
>+  }
>+  template <int64_t us>
>+  static constexpr Timestamp Micros() {
>+    static_assert(us >= 0, "");
>+    static_assert(us < timestamp_impl::kPlusInfinityVal, "");
>+    return Timestamp(us);
>+  }
>+
>+  template <
>+      typename T,
>+      typename std::enable_if<std::is_integral<T>::value>::type* = nullptr>
>+  static Timestamp seconds(T seconds) {
>+    RTC_DCHECK_GE(seconds, 0);
>+    RTC_DCHECK_LT(seconds, timestamp_impl::kPlusInfinityVal / 1000000);
>+    return Timestamp(rtc::dchecked_cast<int64_t>(seconds) * 1000000);
>+  }
>+
>+  template <
>+      typename T,
>+      typename std::enable_if<std::is_integral<T>::value>::type* = nullptr>
>+  static Timestamp ms(T milliseconds) {
>+    RTC_DCHECK_GE(milliseconds, 0);
>+    RTC_DCHECK_LT(milliseconds, timestamp_impl::kPlusInfinityVal / 1000);
>+    return Timestamp(rtc::dchecked_cast<int64_t>(milliseconds) * 1000);
>+  }
>+
>+  template <
>+      typename T,
>+      typename std::enable_if<std::is_integral<T>::value>::type* = nullptr>
>+  static Timestamp us(T microseconds) {
>+    RTC_DCHECK_GE(microseconds, 0);
>+    RTC_DCHECK_LT(microseconds, timestamp_impl::kPlusInfinityVal);
>+    return Timestamp(rtc::dchecked_cast<int64_t>(microseconds));
>+  }
>+
>+  template <typename T,
>+            typename std::enable_if<std::is_floating_point<T>::value>::type* =
>+                nullptr>
>+  static Timestamp seconds(T seconds) {
>+    return Timestamp::us(seconds * 1e6);
>+  }
>+
>+  template <typename T,
>+            typename std::enable_if<std::is_floating_point<T>::value>::type* =
>+                nullptr>
>+  static Timestamp ms(T milliseconds) {
>+    return Timestamp::us(milliseconds * 1e3);
>+  }
>+  template <typename T,
>+            typename std::enable_if<std::is_floating_point<T>::value>::type* =
>+                nullptr>
>+  static Timestamp us(T microseconds) {
>+    if (microseconds == std::numeric_limits<double>::infinity()) {
>+      return Infinity();
>+    } else {
>+      RTC_DCHECK(!std::isnan(microseconds));
>+      RTC_DCHECK_GE(microseconds, 0);
>+      RTC_DCHECK_LT(microseconds, timestamp_impl::kPlusInfinityVal);
>+      return Timestamp(rtc::dchecked_cast<int64_t>(microseconds));
>+    }
>+  }
>+
>+  template <typename T = int64_t>
>+  typename std::enable_if<std::is_integral<T>::value, T>::type seconds() const {
>+    RTC_DCHECK(IsFinite());
>+    return rtc::dchecked_cast<T>(UnsafeSeconds());
>+  }
>+  template <typename T = int64_t>
>+  typename std::enable_if<std::is_integral<T>::value, T>::type ms() const {
>+    RTC_DCHECK(IsFinite());
>+    return rtc::dchecked_cast<T>(UnsafeMillis());
>+  }
>+  template <typename T = int64_t>
>+  typename std::enable_if<std::is_integral<T>::value, T>::type us() const {
>+    RTC_DCHECK(IsFinite());
>+    return rtc::dchecked_cast<T>(microseconds_);
>+  }
>+
>+  template <typename T>
>+  constexpr typename std::enable_if<std::is_floating_point<T>::value, T>::type
>+  seconds() const {
>+    return us<T>() * 1e-6;
>+  }
>+  template <typename T>
>+  constexpr typename std::enable_if<std::is_floating_point<T>::value, T>::type
>+  ms() const {
>+    return us<T>() * 1e-3;
>+  }
>+  template <typename T>
>+  constexpr typename std::enable_if<std::is_floating_point<T>::value, T>::type
>+  us() const {
>+    return IsInfinite() ? std::numeric_limits<T>::infinity() : microseconds_;
>+  }
>+
>+  constexpr int64_t seconds_or(int64_t fallback_value) const {
>+    return IsFinite() ? UnsafeSeconds() : fallback_value;
>+  }
>+  constexpr int64_t ms_or(int64_t fallback_value) const {
>+    return IsFinite() ? UnsafeMillis() : fallback_value;
>+  }
>+  constexpr int64_t us_or(int64_t fallback_value) const {
>+    return IsFinite() ? microseconds_ : fallback_value;
>+  }
>+
>+  constexpr bool IsInfinite() const {
>+    return microseconds_ == timestamp_impl::kPlusInfinityVal;
>+  }
>+  constexpr bool IsFinite() const { return !IsInfinite(); }
>+  TimeDelta operator-(const Timestamp& other) const {
>+    return TimeDelta::us(us() - other.us());
>+  }
>+  Timestamp operator-(const TimeDelta& delta) const {
>+    return Timestamp::us(us() - delta.us());
>+  }
>+  Timestamp operator+(const TimeDelta& delta) const {
>+    return Timestamp::us(us() + delta.us());
>+  }
>+  Timestamp& operator-=(const TimeDelta& other) {
>+    microseconds_ -= other.us();
>+    return *this;
>+  }
>+  Timestamp& operator+=(const TimeDelta& other) {
>+    microseconds_ += other.us();
>+    return *this;
>+  }
>+  constexpr bool operator==(const Timestamp& other) const {
>+    return microseconds_ == other.microseconds_;
>+  }
>+  constexpr bool operator!=(const Timestamp& other) const {
>+    return microseconds_ != other.microseconds_;
>+  }
>+  constexpr bool operator<=(const Timestamp& other) const {
>+    return microseconds_ <= other.microseconds_;
>+  }
>+  constexpr bool operator>=(const Timestamp& other) const {
>+    return microseconds_ >= other.microseconds_;
>+  }
>+  constexpr bool operator>(const Timestamp& other) const {
>+    return microseconds_ > other.microseconds_;
>+  }
>+  constexpr bool operator<(const Timestamp& other) const {
>+    return microseconds_ < other.microseconds_;
>+  }
>+
>+ private:
>+  explicit constexpr Timestamp(int64_t us) : microseconds_(us) {}
>+  constexpr int64_t UnsafeSeconds() const {
>+    return (microseconds_ + 500000) / 1000000;
>+  }
>+  constexpr int64_t UnsafeMillis() const {
>+    return (microseconds_ + 500) / 1000;
>+  }
>+  int64_t microseconds_;
>+};
>+
>+std::string ToString(const Timestamp& value);
>+
>+}  // namespace webrtc
>+
>+#endif  // API_UNITS_TIMESTAMP_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/timestamp_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/timestamp_unittest.cc
>new file mode 100644
>index 00000000000..db894cca447
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/units/timestamp_unittest.cc
>@@ -0,0 +1,127 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/units/timestamp.h"
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+namespace test {
>+TEST(TimestampTest, ConstExpr) {
>+  constexpr int64_t kValue = 12345;
>+  constexpr Timestamp kTimestampInf = Timestamp::Infinity();
>+  static_assert(kTimestampInf.IsInfinite(), "");
>+  static_assert(kTimestampInf.ms_or(-1) == -1, "");
>+
>+  constexpr Timestamp kTimestampSeconds = Timestamp::Seconds<kValue>();
>+  constexpr Timestamp kTimestampMs = Timestamp::Millis<kValue>();
>+  constexpr Timestamp kTimestampUs = Timestamp::Micros<kValue>();
>+
>+  static_assert(kTimestampSeconds.seconds_or(0) == kValue, "");
>+  static_assert(kTimestampMs.ms_or(0) == kValue, "");
>+  static_assert(kTimestampUs.us_or(0) == kValue, "");
>+
>+  static_assert(kTimestampMs > kTimestampUs, "");
>+
>+  EXPECT_EQ(kTimestampSeconds.seconds(), kValue);
>+  EXPECT_EQ(kTimestampMs.ms(), kValue);
>+  EXPECT_EQ(kTimestampUs.us(), kValue);
>+}
>+
>+TEST(TimestampTest, GetBackSameValues) {
>+  const int64_t kValue = 499;
>+  EXPECT_EQ(Timestamp::ms(kValue).ms(), kValue);
>+  EXPECT_EQ(Timestamp::us(kValue).us(), kValue);
>+  EXPECT_EQ(Timestamp::seconds(kValue).seconds(), kValue);
>+}
>+
>+TEST(TimestampTest, GetDifferentPrefix) {
>+  const int64_t kValue = 3000000;
>+  EXPECT_EQ(Timestamp::us(kValue).seconds(), kValue / 1000000);
>+  EXPECT_EQ(Timestamp::ms(kValue).seconds(), kValue / 1000);
>+  EXPECT_EQ(Timestamp::us(kValue).ms(), kValue / 1000);
>+
>+  EXPECT_EQ(Timestamp::ms(kValue).us(), kValue * 1000);
>+  EXPECT_EQ(Timestamp::seconds(kValue).ms(), kValue * 1000);
>+  EXPECT_EQ(Timestamp::seconds(kValue).us(), kValue * 1000000);
>+}
>+
>+TEST(TimestampTest, IdentityChecks) {
>+  const int64_t kValue = 3000;
>+
>+  EXPECT_TRUE(Timestamp::Infinity().IsInfinite());
>+  EXPECT_FALSE(Timestamp::ms(kValue).IsInfinite());
>+
>+  EXPECT_FALSE(Timestamp::Infinity().IsFinite());
>+  EXPECT_TRUE(Timestamp::ms(kValue).IsFinite());
>+}
>+
>+TEST(TimestampTest, ComparisonOperators) {
>+  const int64_t kSmall = 450;
>+  const int64_t kLarge = 451;
>+
>+  EXPECT_EQ(Timestamp::Infinity(), Timestamp::Infinity());
>+  EXPECT_GE(Timestamp::Infinity(), Timestamp::Infinity());
>+  EXPECT_GT(Timestamp::Infinity(), Timestamp::ms(kLarge));
>+  EXPECT_EQ(Timestamp::ms(kSmall), Timestamp::ms(kSmall));
>+  EXPECT_LE(Timestamp::ms(kSmall), Timestamp::ms(kSmall));
>+  EXPECT_GE(Timestamp::ms(kSmall), Timestamp::ms(kSmall));
>+  EXPECT_NE(Timestamp::ms(kSmall), Timestamp::ms(kLarge));
>+  EXPECT_LE(Timestamp::ms(kSmall), Timestamp::ms(kLarge));
>+  EXPECT_LT(Timestamp::ms(kSmall), Timestamp::ms(kLarge));
>+  EXPECT_GE(Timestamp::ms(kLarge), Timestamp::ms(kSmall));
>+  EXPECT_GT(Timestamp::ms(kLarge), Timestamp::ms(kSmall));
>+}
>+
>+TEST(TimestampTest, CanBeInititializedFromLargeInt) {
>+  const int kMaxInt = std::numeric_limits<int>::max();
>+  EXPECT_EQ(Timestamp::seconds(kMaxInt).us(),
>+            static_cast<int64_t>(kMaxInt) * 1000000);
>+  EXPECT_EQ(Timestamp::ms(kMaxInt).us(), static_cast<int64_t>(kMaxInt) * 1000);
>+}
>+
>+TEST(TimestampTest, ConvertsToAndFromDouble) {
>+  const int64_t kMicros = 17017;
>+  const double kMicrosDouble = kMicros;
>+  const double kMillisDouble = kMicros * 1e-3;
>+  const double kSecondsDouble = kMillisDouble * 1e-3;
>+
>+  EXPECT_EQ(Timestamp::us(kMicros).seconds<double>(), kSecondsDouble);
>+  EXPECT_EQ(Timestamp::seconds(kSecondsDouble).us(), kMicros);
>+
>+  EXPECT_EQ(Timestamp::us(kMicros).ms<double>(), kMillisDouble);
>+  EXPECT_EQ(Timestamp::ms(kMillisDouble).us(), kMicros);
>+
>+  EXPECT_EQ(Timestamp::us(kMicros).us<double>(), kMicrosDouble);
>+  EXPECT_EQ(Timestamp::us(kMicrosDouble).us(), kMicros);
>+
>+  const double kPlusInfinity = std::numeric_limits<double>::infinity();
>+
>+  EXPECT_EQ(Timestamp::Infinity().seconds<double>(), kPlusInfinity);
>+  EXPECT_EQ(Timestamp::Infinity().ms<double>(), kPlusInfinity);
>+  EXPECT_EQ(Timestamp::Infinity().us<double>(), kPlusInfinity);
>+
>+  EXPECT_TRUE(Timestamp::seconds(kPlusInfinity).IsInfinite());
>+  EXPECT_TRUE(Timestamp::ms(kPlusInfinity).IsInfinite());
>+  EXPECT_TRUE(Timestamp::us(kPlusInfinity).IsInfinite());
>+}
>+
>+TEST(UnitConversionTest, TimestampAndTimeDeltaMath) {
>+  const int64_t kValueA = 267;
>+  const int64_t kValueB = 450;
>+  const Timestamp time_a = Timestamp::ms(kValueA);
>+  const Timestamp time_b = Timestamp::ms(kValueB);
>+  const TimeDelta delta_a = TimeDelta::ms(kValueA);
>+
>+  EXPECT_EQ((time_a - time_b), TimeDelta::ms(kValueA - kValueB));
>+  EXPECT_EQ((time_b - delta_a), Timestamp::ms(kValueB - kValueA));
>+  EXPECT_EQ((time_b + delta_a), Timestamp::ms(kValueB + kValueA));
>+}
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/BUILD.gn
>new file mode 100644
>index 00000000000..d5480dd250e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/BUILD.gn
>@@ -0,0 +1,182 @@
>+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+#
>+# Use of this source code is governed by a BSD-style license
>+# that can be found in the LICENSE file in the root of the source
>+# tree. An additional intellectual property rights grant can be found
>+# in the file PATENTS.  All contributing project authors may
>+# be found in the AUTHORS file in the root of the source tree.
>+
>+import("../../webrtc.gni")
>+
>+rtc_source_set("video_frame") {
>+  visibility = [ "*" ]
>+  sources = [
>+    "color_space.cc",
>+    "color_space.h",
>+    "video_content_type.cc",
>+    "video_content_type.h",
>+    "video_frame.cc",
>+    "video_frame.h",
>+    "video_frame_buffer.cc",
>+    "video_frame_buffer.h",
>+    "video_rotation.h",
>+    "video_sink_interface.h",
>+    "video_source_interface.cc",
>+    "video_source_interface.h",
>+    "video_timing.cc",
>+    "video_timing.h",
>+  ]
>+
>+  deps = [
>+    "../../rtc_base:checks",
>+    "../../rtc_base:rtc_base_approved",
>+    "//third_party/abseil-cpp/absl/types:optional",
>+  ]
>+}
>+
>+rtc_source_set("video_frame_i420") {
>+  visibility = [ "*" ]
>+  sources = [
>+    "i420_buffer.cc",
>+    "i420_buffer.h",
>+  ]
>+  deps = [
>+    ":video_frame",
>+    "../../rtc_base:checks",
>+    "../../rtc_base:rtc_base",
>+    "../../rtc_base/memory:aligned_malloc",
>+    "//third_party/libyuv",
>+  ]
>+}
>+
>+rtc_source_set("video_frame_i010") {
>+  visibility = [ "*" ]
>+  sources = [
>+    "i010_buffer.cc",
>+    "i010_buffer.h",
>+  ]
>+  deps = [
>+    ":video_frame",
>+    ":video_frame_i420",
>+    "../../rtc_base:checks",
>+    "../../rtc_base:rtc_base",
>+    "../../rtc_base/memory:aligned_malloc",
>+    "//third_party/libyuv",
>+  ]
>+}
>+
>+rtc_source_set("encoded_frame") {
>+  visibility = [ "*" ]
>+  sources = [
>+    "encoded_frame.cc",
>+    "encoded_frame.h",
>+  ]
>+
>+  deps = [
>+    "../../modules/video_coding:encoded_frame",
>+  ]
>+}
>+
>+rtc_source_set("video_bitrate_allocation") {
>+  visibility = [ "*" ]
>+  sources = [
>+    "video_bitrate_allocation.cc",
>+    "video_bitrate_allocation.h",
>+  ]
>+  deps = [
>+    "../../rtc_base:checks",
>+    "../../rtc_base:safe_conversions",
>+    "../../rtc_base:stringutils",
>+    "//third_party/abseil-cpp/absl/types:optional",
>+  ]
>+}
>+
>+rtc_source_set("video_bitrate_allocator") {
>+  visibility = [ "*" ]
>+  sources = [
>+    "video_bitrate_allocator.h",
>+  ]
>+  deps = [
>+    ":video_bitrate_allocation",
>+  ]
>+}
>+
>+rtc_source_set("video_bitrate_allocator_factory") {
>+  visibility = [ "*" ]
>+  sources = [
>+    "video_bitrate_allocator_factory.h",
>+  ]
>+  deps = [
>+    ":video_bitrate_allocator",
>+    "../../media:rtc_media_base",
>+  ]
>+}
>+
>+rtc_source_set("video_stream_decoder") {
>+  visibility = [ "*" ]
>+  sources = [
>+    "video_stream_decoder.h",
>+  ]
>+
>+  deps = [
>+    ":encoded_frame",
>+    ":video_frame",
>+    "../video_codecs:video_codecs_api",
>+    "//third_party/abseil-cpp/absl/types:optional",
>+  ]
>+}
>+
>+rtc_source_set("video_stream_decoder_create") {
>+  visibility = [ "*" ]
>+  sources = [
>+    "video_stream_decoder_create.cc",
>+    "video_stream_decoder_create.h",
>+  ]
>+
>+  deps = [
>+    ":video_stream_decoder",
>+    "../../rtc_base:rtc_base_approved",
>+    "../../video:video_stream_decoder_impl",
>+    "//third_party/abseil-cpp/absl/memory",
>+  ]
>+}
>+
>+rtc_source_set("video_stream_encoder") {
>+  visibility = [ "*" ]
>+  sources = [
>+    "video_stream_encoder_interface.h",
>+    "video_stream_encoder_observer.h",
>+    "video_stream_encoder_settings.h",
>+  ]
>+
>+  deps = [
>+    ":video_frame",
>+
>+    # For rtpparameters.h
>+    "..:libjingle_peerconnection_api",
>+    "../video_codecs:video_codecs_api",
>+    "//third_party/abseil-cpp/absl/types:optional",
>+  ]
>+}
>+
>+rtc_source_set("video_stream_encoder_create") {
>+  visibility = [ "*" ]
>+  allow_poison = [ "software_video_codecs" ]  # TODO(bugs.webrtc.org/7925): Remove.
>+  sources = [
>+    "video_stream_encoder_create.cc",
>+    "video_stream_encoder_create.h",
>+  ]
>+
>+  if (!build_with_chromium && is_clang) {
>+    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>+    suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>+  }
>+
>+  deps = [
>+    ":video_stream_encoder",
>+    "../../rtc_base:ptr_util",
>+    "../../video:video_stream_encoder_impl",
>+    "../video_codecs:video_codecs_api",
>+    "//third_party/abseil-cpp/absl/memory",
>+  ]
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/DEPS b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/DEPS
>new file mode 100644
>index 00000000000..aec927baaa9
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/DEPS
>@@ -0,0 +1,14 @@
>+specific_include_rules = {
>+  # Until the new VideoStreamDecoder is implemented the current decoding
>+  # pipeline will be used, and therefore EncodedFrame needs to inherit
>+  # VCMEncodedFrame.
>+  "encoded_frame.h": [
>+    "+modules/video_coding/encoded_frame.h",
>+  ],
>+  "video_stream_decoder_create.cc": [
>+    "+video/video_stream_decoder_impl.h",
>+  ],
>+  "video_stream_encoder_create.cc": [
>+    "+video/video_stream_encoder.h",
>+  ],
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/OWNERS
>index 8327124e236..315f85e7d00 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/OWNERS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/OWNERS
>@@ -1,3 +1,4 @@
> magjed@webrtc.org
>+nisse@webrtc.org
> 
> per-file video_timing.h=ilnik@webrtc.org
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/color_space.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/color_space.cc
>new file mode 100644
>index 00000000000..a8be5cd8280
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/color_space.cc
>@@ -0,0 +1,42 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/video/color_space.h"
>+
>+namespace webrtc {
>+
>+ColorSpace::ColorSpace() = default;
>+
>+ColorSpace::ColorSpace(PrimaryID primaries,
>+                       TransferID transfer,
>+                       MatrixID matrix,
>+                       RangeID range)
>+    : primaries_(primaries),
>+      transfer_(transfer),
>+      matrix_(matrix),
>+      range_(range) {}
>+
>+ColorSpace::PrimaryID ColorSpace::primaries() const {
>+  return primaries_;
>+}
>+
>+ColorSpace::TransferID ColorSpace::transfer() const {
>+  return transfer_;
>+}
>+
>+ColorSpace::MatrixID ColorSpace::matrix() const {
>+  return matrix_;
>+}
>+
>+ColorSpace::RangeID ColorSpace::range() const {
>+  return range_;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/color_space.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/color_space.h
>new file mode 100644
>index 00000000000..8102647b6ff
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/color_space.h
>@@ -0,0 +1,107 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_VIDEO_COLOR_SPACE_H_
>+#define API_VIDEO_COLOR_SPACE_H_
>+
>+namespace webrtc {
>+
>+// Used to represent a color space for the purpose of color conversion. This
>+// class only represents color information that can be transferred through the
>+// bitstream of WebRTC's internal supported codecs:
>+// - VP9 supports color profiles, see VP9 Bitstream & Decoding Process
>+// Specification Version 0.6 Section 7.2.2 "Color config semantics" available
>+// from https://www.webmproject.org.
>+// TODO(emircan): Extract these values from decode and add to the existing ones.
>+// - VP8 only supports BT.601, see
>+// https://tools.ietf.org/html/rfc6386#section-9.2
>+// - H264 supports different color primaries, transfer characteristics, matrix
>+// coefficients and range. See T-REC-H.264 E.2.1, "VUI parameters semantics",
>+// available from https://www.itu.int/rec/T-REC-H.264.
>+class ColorSpace {
>+ public:
>+  enum class PrimaryID {
>+    kInvalid,
>+    kBT709,
>+    kBT470M,
>+    kBT470BG,
>+    kSMPTE170M,  // Identical to BT601
>+    kSMPTE240M,
>+    kFILM,
>+    kBT2020,
>+    kSMPTEST428,
>+    kSMPTEST431,
>+    kSMPTEST432,
>+    kJEDECP22,
>+  };
>+
>+  enum class TransferID {
>+    kInvalid,
>+    kBT709,
>+    kGAMMA22,
>+    kGAMMA28,
>+    kSMPTE170M,
>+    kSMPTE240M,
>+    kLINEAR,
>+    kLOG,
>+    kLOG_SQRT,
>+    kIEC61966_2_4,
>+    kBT1361_ECG,
>+    kIEC61966_2_1,
>+    kBT2020_10,
>+    kBT2020_12,
>+    kSMPTEST2084,
>+    kSMPTEST428,
>+    kARIB_STD_B67,
>+  };
>+
>+  enum class MatrixID {
>+    kInvalid,
>+    kRGB,
>+    kBT709,
>+    kFCC,
>+    kBT470BG,
>+    kSMPTE170M,
>+    kSMPTE240M,
>+    kYCOCG,
>+    kBT2020_NCL,
>+    kBT2020_CL,
>+    kSMPTE2085,
>+  };
>+
>+  enum class RangeID {
>+    kInvalid,
>+    // Limited Rec. 709 color range with RGB values ranging from 16 to 235.
>+    kLimited,
>+    // Full RGB color range with RGB valees from 0 to 255.
>+    kFull,
>+  };
>+
>+  ColorSpace();
>+  ColorSpace(PrimaryID primaries,
>+             TransferID transfer,
>+             MatrixID matrix,
>+             RangeID full_range);
>+
>+  PrimaryID primaries() const;
>+  TransferID transfer() const;
>+  MatrixID matrix() const;
>+  RangeID range() const;
>+
>+ private:
>+  PrimaryID primaries_ = PrimaryID::kInvalid;
>+  TransferID transfer_ = TransferID::kInvalid;
>+  MatrixID matrix_ = MatrixID::kInvalid;
>+  RangeID range_ = RangeID::kInvalid;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // API_VIDEO_COLOR_SPACE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/encoded_frame.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/encoded_frame.cc
>new file mode 100644
>index 00000000000..f9152b23f12
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/encoded_frame.cc
>@@ -0,0 +1,29 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/video/encoded_frame.h"
>+
>+namespace webrtc {
>+namespace video_coding {
>+
>+bool EncodedFrame::delayed_by_retransmission() const {
>+  return 0;
>+}
>+
>+uint32_t EncodedFrame::Timestamp() const {
>+  return timestamp;
>+}
>+
>+void EncodedFrame::SetTimestamp(uint32_t rtp_timestamp) {
>+  timestamp = rtp_timestamp;
>+}
>+
>+}  // namespace video_coding
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/encoded_frame.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/encoded_frame.h
>new file mode 100644
>index 00000000000..1b0a26a5fc2
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/encoded_frame.h
>@@ -0,0 +1,93 @@
>+/*
>+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_VIDEO_ENCODED_FRAME_H_
>+#define API_VIDEO_ENCODED_FRAME_H_
>+
>+#include "modules/video_coding/encoded_frame.h"
>+
>+namespace webrtc {
>+namespace video_coding {
>+
>+// NOTE: This class is still under development and may change without notice.
>+struct VideoLayerFrameId {
>+  // TODO(philipel): The default ctor is currently used internaly, but have a
>+  //                 look if we can remove it.
>+  VideoLayerFrameId() : picture_id(-1), spatial_layer(0) {}
>+  VideoLayerFrameId(int64_t picture_id, uint8_t spatial_layer)
>+      : picture_id(picture_id), spatial_layer(spatial_layer) {}
>+
>+  bool operator==(const VideoLayerFrameId& rhs) const {
>+    return picture_id == rhs.picture_id && spatial_layer == rhs.spatial_layer;
>+  }
>+
>+  bool operator!=(const VideoLayerFrameId& rhs) const {
>+    return !(*this == rhs);
>+  }
>+
>+  bool operator<(const VideoLayerFrameId& rhs) const {
>+    if (picture_id == rhs.picture_id)
>+      return spatial_layer < rhs.spatial_layer;
>+    return picture_id < rhs.picture_id;
>+  }
>+
>+  bool operator<=(const VideoLayerFrameId& rhs) const { return !(rhs < *this); }
>+  bool operator>(const VideoLayerFrameId& rhs) const { return rhs < *this; }
>+  bool operator>=(const VideoLayerFrameId& rhs) const { return rhs <= *this; }
>+
>+  int64_t picture_id;
>+  uint8_t spatial_layer;
>+};
>+
>+// TODO(philipel): Remove webrtc::VCMEncodedFrame inheritance.
>+// TODO(philipel): Move transport specific info out of EncodedFrame.
>+// NOTE: This class is still under development and may change without notice.
>+class EncodedFrame : public webrtc::VCMEncodedFrame {
>+ public:
>+  static const uint8_t kMaxFrameReferences = 5;
>+
>+  EncodedFrame() = default;
>+  virtual ~EncodedFrame() {}
>+
>+  virtual bool GetBitstream(uint8_t* destination) const = 0;
>+
>+  // The capture timestamp of this frame, using the 90 kHz RTP clock.
>+  virtual uint32_t Timestamp() const;
>+  virtual void SetTimestamp(uint32_t rtp_timestamp);
>+
>+  // When this frame was received.
>+  virtual int64_t ReceivedTime() const = 0;
>+
>+  // When this frame should be rendered.
>+  virtual int64_t RenderTime() const = 0;
>+
>+  // This information is currently needed by the timing calculation class.
>+  // TODO(philipel): Remove this function when a new timing class has
>+  //                 been implemented.
>+  virtual bool delayed_by_retransmission() const;
>+
>+  size_t size() const { return _length; }
>+
>+  bool is_keyframe() const { return num_references == 0; }
>+
>+  VideoLayerFrameId id;
>+  uint32_t timestamp = 0;
>+
>+  // TODO(philipel): Add simple modify/access functions to prevent adding too
>+  // many |references|.
>+  size_t num_references = 0;
>+  int64_t references[kMaxFrameReferences];
>+  bool inter_layer_predicted = false;
>+};
>+
>+}  // namespace video_coding
>+}  // namespace webrtc
>+
>+#endif  // API_VIDEO_ENCODED_FRAME_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/i010_buffer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/i010_buffer.cc
>new file mode 100644
>index 00000000000..adb5a5e95ae
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/i010_buffer.cc
>@@ -0,0 +1,237 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#include "api/video/i010_buffer.h"
>+
>+#include <algorithm>
>+#include <utility>
>+
>+#include "api/video/i420_buffer.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/refcountedobject.h"
>+#include "third_party/libyuv/include/libyuv/convert.h"
>+#include "third_party/libyuv/include/libyuv/planar_functions.h"
>+#include "third_party/libyuv/include/libyuv/scale.h"
>+
>+// Aligning pointer to 64 bytes for improved performance, e.g. use SIMD.
>+static const int kBufferAlignment = 64;
>+static const int kBytesPerPixel = 2;
>+
>+namespace webrtc {
>+
>+namespace {
>+
>+int I010DataSize(int height, int stride_y, int stride_u, int stride_v) {
>+  return kBytesPerPixel *
>+         (stride_y * height + (stride_u + stride_v) * ((height + 1) / 2));
>+}
>+
>+}  // namespace
>+
>+I010Buffer::I010Buffer(int width,
>+                       int height,
>+                       int stride_y,
>+                       int stride_u,
>+                       int stride_v)
>+    : width_(width),
>+      height_(height),
>+      stride_y_(stride_y),
>+      stride_u_(stride_u),
>+      stride_v_(stride_v),
>+      data_(static_cast<uint16_t*>(
>+          AlignedMalloc(I010DataSize(height, stride_y, stride_u, stride_v),
>+                        kBufferAlignment))) {
>+  RTC_DCHECK_GT(width, 0);
>+  RTC_DCHECK_GT(height, 0);
>+  RTC_DCHECK_GE(stride_y, width);
>+  RTC_DCHECK_GE(stride_u, (width + 1) / 2);
>+  RTC_DCHECK_GE(stride_v, (width + 1) / 2);
>+}
>+
>+I010Buffer::~I010Buffer() {}
>+
>+// static
>+rtc::scoped_refptr<I010Buffer> I010Buffer::Create(int width, int height) {
>+  return new rtc::RefCountedObject<I010Buffer>(
>+      width, height, width, (width + 1) / 2, (width + 1) / 2);
>+}
>+
>+// static
>+rtc::scoped_refptr<I010Buffer> I010Buffer::Copy(
>+    const I010BufferInterface& source) {
>+  const int width = source.width();
>+  const int height = source.height();
>+  rtc::scoped_refptr<I010Buffer> buffer = Create(width, height);
>+  RTC_CHECK_EQ(
>+      0, libyuv::I010Copy(
>+             source.DataY(), source.StrideY(), source.DataU(), source.StrideU(),
>+             source.DataV(), source.StrideV(), buffer->MutableDataY(),
>+             buffer->StrideY(), buffer->MutableDataU(), buffer->StrideU(),
>+             buffer->MutableDataV(), buffer->StrideV(), width, height));
>+  return buffer;
>+}
>+
>+// static
>+rtc::scoped_refptr<I010Buffer> I010Buffer::Copy(
>+    const I420BufferInterface& source) {
>+  const int width = source.width();
>+  const int height = source.height();
>+  rtc::scoped_refptr<I010Buffer> buffer = Create(width, height);
>+  RTC_CHECK_EQ(
>+      0, libyuv::I420ToI010(
>+             source.DataY(), source.StrideY(), source.DataU(), source.StrideU(),
>+             source.DataV(), source.StrideV(), buffer->MutableDataY(),
>+             buffer->StrideY(), buffer->MutableDataU(), buffer->StrideU(),
>+             buffer->MutableDataV(), buffer->StrideV(), width, height));
>+  return buffer;
>+}
>+
>+// static
>+rtc::scoped_refptr<I010Buffer> I010Buffer::Rotate(
>+    const I010BufferInterface& src,
>+    VideoRotation rotation) {
>+  if (rotation == webrtc::kVideoRotation_0)
>+    return Copy(src);
>+
>+  RTC_CHECK(src.DataY());
>+  RTC_CHECK(src.DataU());
>+  RTC_CHECK(src.DataV());
>+  int rotated_width = src.width();
>+  int rotated_height = src.height();
>+  if (rotation == webrtc::kVideoRotation_90 ||
>+      rotation == webrtc::kVideoRotation_270) {
>+    std::swap(rotated_width, rotated_height);
>+  }
>+
>+  rtc::scoped_refptr<webrtc::I010Buffer> buffer =
>+      Create(rotated_width, rotated_height);
>+  // TODO(emircan): Remove this when there is libyuv::I010Rotate().
>+  for (int x = 0; x < src.width(); x++) {
>+    for (int y = 0; y < src.height(); y++) {
>+      int dest_x = x;
>+      int dest_y = y;
>+      switch (rotation) {
>+        // This case is covered by the early return.
>+        case webrtc::kVideoRotation_0:
>+          RTC_NOTREACHED();
>+          break;
>+        case webrtc::kVideoRotation_90:
>+          dest_x = src.height() - y - 1;
>+          dest_y = x;
>+          break;
>+        case webrtc::kVideoRotation_180:
>+          dest_x = src.width() - x - 1;
>+          dest_y = src.height() - y - 1;
>+          break;
>+        case webrtc::kVideoRotation_270:
>+          dest_x = y;
>+          dest_y = src.width() - x - 1;
>+          break;
>+      }
>+      buffer->MutableDataY()[dest_x + buffer->StrideY() * dest_y] =
>+          src.DataY()[x + src.StrideY() * y];
>+      dest_x /= 2;
>+      dest_y /= 2;
>+      int src_x = x / 2;
>+      int src_y = y / 2;
>+      buffer->MutableDataU()[dest_x + buffer->StrideU() * dest_y] =
>+          src.DataU()[src_x + src.StrideU() * src_y];
>+      buffer->MutableDataV()[dest_x + buffer->StrideV() * dest_y] =
>+          src.DataV()[src_x + src.StrideV() * src_y];
>+    }
>+  }
>+  return buffer;
>+}
>+
>+rtc::scoped_refptr<I420BufferInterface> I010Buffer::ToI420() {
>+  rtc::scoped_refptr<I420Buffer> i420_buffer =
>+      I420Buffer::Create(width(), height());
>+  libyuv::I010ToI420(DataY(), StrideY(), DataU(), StrideU(), DataV(), StrideV(),
>+                     i420_buffer->MutableDataY(), i420_buffer->StrideY(),
>+                     i420_buffer->MutableDataU(), i420_buffer->StrideU(),
>+                     i420_buffer->MutableDataV(), i420_buffer->StrideV(),
>+                     width(), height());
>+  return i420_buffer;
>+}
>+
>+int I010Buffer::width() const {
>+  return width_;
>+}
>+
>+int I010Buffer::height() const {
>+  return height_;
>+}
>+
>+const uint16_t* I010Buffer::DataY() const {
>+  return data_.get();
>+}
>+const uint16_t* I010Buffer::DataU() const {
>+  return data_.get() + stride_y_ * height_;
>+}
>+const uint16_t* I010Buffer::DataV() const {
>+  return data_.get() + stride_y_ * height_ + stride_u_ * ((height_ + 1) / 2);
>+}
>+
>+int I010Buffer::StrideY() const {
>+  return stride_y_;
>+}
>+int I010Buffer::StrideU() const {
>+  return stride_u_;
>+}
>+int I010Buffer::StrideV() const {
>+  return stride_v_;
>+}
>+
>+uint16_t* I010Buffer::MutableDataY() {
>+  return const_cast<uint16_t*>(DataY());
>+}
>+uint16_t* I010Buffer::MutableDataU() {
>+  return const_cast<uint16_t*>(DataU());
>+}
>+uint16_t* I010Buffer::MutableDataV() {
>+  return const_cast<uint16_t*>(DataV());
>+}
>+
>+void I010Buffer::CropAndScaleFrom(const I010BufferInterface& src,
>+                                  int offset_x,
>+                                  int offset_y,
>+                                  int crop_width,
>+                                  int crop_height) {
>+  RTC_CHECK_LE(crop_width, src.width());
>+  RTC_CHECK_LE(crop_height, src.height());
>+  RTC_CHECK_LE(crop_width + offset_x, src.width());
>+  RTC_CHECK_LE(crop_height + offset_y, src.height());
>+  RTC_CHECK_GE(offset_x, 0);
>+  RTC_CHECK_GE(offset_y, 0);
>+
>+  // Make sure offset is even so that u/v plane becomes aligned.
>+  const int uv_offset_x = offset_x / 2;
>+  const int uv_offset_y = offset_y / 2;
>+  offset_x = uv_offset_x * 2;
>+  offset_y = uv_offset_y * 2;
>+
>+  const uint16_t* y_plane = src.DataY() + src.StrideY() * offset_y + offset_x;
>+  const uint16_t* u_plane =
>+      src.DataU() + src.StrideU() * uv_offset_y + uv_offset_x;
>+  const uint16_t* v_plane =
>+      src.DataV() + src.StrideV() * uv_offset_y + uv_offset_x;
>+  int res = libyuv::I420Scale_16(
>+      y_plane, src.StrideY(), u_plane, src.StrideU(), v_plane, src.StrideV(),
>+      crop_width, crop_height, MutableDataY(), StrideY(), MutableDataU(),
>+      StrideU(), MutableDataV(), StrideV(), width(), height(),
>+      libyuv::kFilterBox);
>+
>+  RTC_DCHECK_EQ(res, 0);
>+}
>+
>+void I010Buffer::ScaleFrom(const I010BufferInterface& src) {
>+  CropAndScaleFrom(src, 0, 0, src.width(), src.height());
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/i010_buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/i010_buffer.h
>new file mode 100644
>index 00000000000..1208b318525
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/i010_buffer.h
>@@ -0,0 +1,81 @@
>+/*
>+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_VIDEO_I010_BUFFER_H_
>+#define API_VIDEO_I010_BUFFER_H_
>+
>+#include <memory>
>+
>+#include "api/video/video_frame_buffer.h"
>+#include "api/video/video_rotation.h"
>+#include "rtc_base/memory/aligned_malloc.h"
>+
>+namespace webrtc {
>+
>+// Plain I010 buffer in standard memory.
>+class I010Buffer : public I010BufferInterface {
>+ public:
>+  // Create a new buffer.
>+  static rtc::scoped_refptr<I010Buffer> Create(int width, int height);
>+
>+  // Create a new buffer and copy the pixel data.
>+  static rtc::scoped_refptr<I010Buffer> Copy(const I010BufferInterface& buffer);
>+
>+  // Convert and put I420 buffer into a new buffer.
>+  static rtc::scoped_refptr<I010Buffer> Copy(const I420BufferInterface& buffer);
>+
>+  // Return a rotated copy of |src|.
>+  static rtc::scoped_refptr<I010Buffer> Rotate(const I010BufferInterface& src,
>+                                               VideoRotation rotation);
>+
>+  // VideoFrameBuffer implementation.
>+  rtc::scoped_refptr<I420BufferInterface> ToI420() override;
>+
>+  // PlanarYuv16BBuffer implementation.
>+  int width() const override;
>+  int height() const override;
>+  const uint16_t* DataY() const override;
>+  const uint16_t* DataU() const override;
>+  const uint16_t* DataV() const override;
>+  int StrideY() const override;
>+  int StrideU() const override;
>+  int StrideV() const override;
>+
>+  uint16_t* MutableDataY();
>+  uint16_t* MutableDataU();
>+  uint16_t* MutableDataV();
>+
>+  // Scale the cropped area of |src| to the size of |this| buffer, and
>+  // write the result into |this|.
>+  void CropAndScaleFrom(const I010BufferInterface& src,
>+                        int offset_x,
>+                        int offset_y,
>+                        int crop_width,
>+                        int crop_height);
>+
>+  // Scale all of |src| to the size of |this| buffer, with no cropping.
>+  void ScaleFrom(const I010BufferInterface& src);
>+
>+ protected:
>+  I010Buffer(int width, int height, int stride_y, int stride_u, int stride_v);
>+  ~I010Buffer() override;
>+
>+ private:
>+  const int width_;
>+  const int height_;
>+  const int stride_y_;
>+  const int stride_u_;
>+  const int stride_v_;
>+  const std::unique_ptr<uint16_t, AlignedFreeDeleter> data_;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // API_VIDEO_I010_BUFFER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/i420_buffer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/i420_buffer.cc
>index 66071e1ec0d..8d239f320dc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/i420_buffer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/i420_buffer.cc
>@@ -34,8 +34,7 @@ int I420DataSize(int height, int stride_y, int stride_u, int stride_v) {
> }  // namespace
> 
> I420Buffer::I420Buffer(int width, int height)
>-    : I420Buffer(width, height, width, (width + 1) / 2, (width + 1) / 2) {
>-}
>+    : I420Buffer(width, height, width, (width + 1) / 2, (width + 1) / 2) {}
> 
> I420Buffer::I420Buffer(int width,
>                        int height,
>@@ -47,9 +46,9 @@ I420Buffer::I420Buffer(int width,
>       stride_y_(stride_y),
>       stride_u_(stride_u),
>       stride_v_(stride_v),
>-      data_(static_cast<uint8_t*>(AlignedMalloc(
>-          I420DataSize(height, stride_y, stride_u, stride_v),
>-          kBufferAlignment))) {
>+      data_(static_cast<uint8_t*>(
>+          AlignedMalloc(I420DataSize(height, stride_y, stride_u, stride_v),
>+                        kBufferAlignment))) {
>   RTC_DCHECK_GT(width, 0);
>   RTC_DCHECK_GT(height, 0);
>   RTC_DCHECK_GE(stride_y, width);
>@@ -57,8 +56,7 @@ I420Buffer::I420Buffer(int width,
>   RTC_DCHECK_GE(stride_v, (width + 1) / 2);
> }
> 
>-I420Buffer::~I420Buffer() {
>-}
>+I420Buffer::~I420Buffer() {}
> 
> // static
> rtc::scoped_refptr<I420Buffer> I420Buffer::Create(int width, int height) {
>@@ -71,34 +69,34 @@ rtc::scoped_refptr<I420Buffer> I420Buffer::Create(int width,
>                                                   int stride_y,
>                                                   int stride_u,
>                                                   int stride_v) {
>-  return new rtc::RefCountedObject<I420Buffer>(
>-      width, height, stride_y, stride_u, stride_v);
>+  return new rtc::RefCountedObject<I420Buffer>(width, height, stride_y,
>+                                               stride_u, stride_v);
> }
> 
> // static
> rtc::scoped_refptr<I420Buffer> I420Buffer::Copy(
>     const I420BufferInterface& source) {
>-  return Copy(source.width(), source.height(),
>-              source.DataY(), source.StrideY(),
>-              source.DataU(), source.StrideU(),
>-              source.DataV(), source.StrideV());
>+  return Copy(source.width(), source.height(), source.DataY(), source.StrideY(),
>+              source.DataU(), source.StrideU(), source.DataV(),
>+              source.StrideV());
> }
> 
> // static
>-rtc::scoped_refptr<I420Buffer> I420Buffer::Copy(
>-      int width, int height,
>-      const uint8_t* data_y, int stride_y,
>-      const uint8_t* data_u, int stride_u,
>-      const uint8_t* data_v, int stride_v) {
>+rtc::scoped_refptr<I420Buffer> I420Buffer::Copy(int width,
>+                                                int height,
>+                                                const uint8_t* data_y,
>+                                                int stride_y,
>+                                                const uint8_t* data_u,
>+                                                int stride_u,
>+                                                const uint8_t* data_v,
>+                                                int stride_v) {
>   // Note: May use different strides than the input data.
>   rtc::scoped_refptr<I420Buffer> buffer = Create(width, height);
>-  RTC_CHECK_EQ(0, libyuv::I420Copy(data_y, stride_y,
>-                                   data_u, stride_u,
>-                                   data_v, stride_v,
>-                                   buffer->MutableDataY(), buffer->StrideY(),
>-                                   buffer->MutableDataU(), buffer->StrideU(),
>-                                   buffer->MutableDataV(), buffer->StrideV(),
>-                                   width, height));
>+  RTC_CHECK_EQ(0, libyuv::I420Copy(data_y, stride_y, data_u, stride_u, data_v,
>+                                   stride_v, buffer->MutableDataY(),
>+                                   buffer->StrideY(), buffer->MutableDataU(),
>+                                   buffer->StrideU(), buffer->MutableDataV(),
>+                                   buffer->StrideV(), width, height));
>   return buffer;
> }
> 
>@@ -120,14 +118,13 @@ rtc::scoped_refptr<I420Buffer> I420Buffer::Rotate(
>   rtc::scoped_refptr<webrtc::I420Buffer> buffer =
>       I420Buffer::Create(rotated_width, rotated_height);
> 
>-  RTC_CHECK_EQ(0, libyuv::I420Rotate(
>-      src.DataY(), src.StrideY(),
>-      src.DataU(), src.StrideU(),
>-      src.DataV(), src.StrideV(),
>-      buffer->MutableDataY(), buffer->StrideY(), buffer->MutableDataU(),
>-      buffer->StrideU(), buffer->MutableDataV(), buffer->StrideV(),
>-      src.width(), src.height(),
>-      static_cast<libyuv::RotationMode>(rotation)));
>+  RTC_CHECK_EQ(0,
>+               libyuv::I420Rotate(
>+                   src.DataY(), src.StrideY(), src.DataU(), src.StrideU(),
>+                   src.DataV(), src.StrideV(), buffer->MutableDataY(),
>+                   buffer->StrideY(), buffer->MutableDataU(), buffer->StrideU(),
>+                   buffer->MutableDataV(), buffer->StrideV(), src.width(),
>+                   src.height(), static_cast<libyuv::RotationMode>(rotation)));
> 
>   return buffer;
> }
>@@ -179,9 +176,9 @@ uint8_t* I420Buffer::MutableDataV() {
> void I420Buffer::SetBlack(I420Buffer* buffer) {
>   RTC_CHECK(libyuv::I420Rect(buffer->MutableDataY(), buffer->StrideY(),
>                              buffer->MutableDataU(), buffer->StrideU(),
>-                             buffer->MutableDataV(), buffer->StrideV(),
>-                             0, 0, buffer->width(), buffer->height(),
>-                             0, 128, 128) == 0);
>+                             buffer->MutableDataV(), buffer->StrideV(), 0, 0,
>+                             buffer->width(), buffer->height(), 0, 128,
>+                             128) == 0);
> }
> 
> void I420Buffer::CropAndScaleFrom(const I420BufferInterface& src,
>@@ -202,20 +199,16 @@ void I420Buffer::CropAndScaleFrom(const I420BufferInterface& src,
>   offset_x = uv_offset_x * 2;
>   offset_y = uv_offset_y * 2;
> 
>-  const uint8_t* y_plane =
>-      src.DataY() + src.StrideY() * offset_y + offset_x;
>+  const uint8_t* y_plane = src.DataY() + src.StrideY() * offset_y + offset_x;
>   const uint8_t* u_plane =
>       src.DataU() + src.StrideU() * uv_offset_y + uv_offset_x;
>   const uint8_t* v_plane =
>       src.DataV() + src.StrideV() * uv_offset_y + uv_offset_x;
>-  int res = libyuv::I420Scale(y_plane, src.StrideY(),
>-                              u_plane, src.StrideU(),
>-                              v_plane, src.StrideV(),
>-                              crop_width, crop_height,
>-                              MutableDataY(), StrideY(),
>-                              MutableDataU(), StrideU(),
>-                              MutableDataV(), StrideV(),
>-                              width(), height(), libyuv::kFilterBox);
>+  int res =
>+      libyuv::I420Scale(y_plane, src.StrideY(), u_plane, src.StrideU(), v_plane,
>+                        src.StrideV(), crop_width, crop_height, MutableDataY(),
>+                        StrideY(), MutableDataU(), StrideU(), MutableDataV(),
>+                        StrideV(), width(), height(), libyuv::kFilterBox);
> 
>   RTC_DCHECK_EQ(res, 0);
> }
>@@ -226,10 +219,8 @@ void I420Buffer::CropAndScaleFrom(const I420BufferInterface& src) {
>   const int crop_height =
>       std::min(src.height(), height() * src.width() / width());
> 
>-  CropAndScaleFrom(
>-      src,
>-      (src.width() - crop_width) / 2, (src.height() - crop_height) / 2,
>-      crop_width, crop_height);
>+  CropAndScaleFrom(src, (src.width() - crop_width) / 2,
>+                   (src.height() - crop_height) / 2, crop_width, crop_height);
> }
> 
> void I420Buffer::ScaleFrom(const I420BufferInterface& src) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/i420_buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/i420_buffer.h
>index bdac80bbc10..282b2426ca3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/i420_buffer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/i420_buffer.h
>@@ -13,9 +13,9 @@
> 
> #include <memory>
> 
>-#include "api/video/video_rotation.h"
> #include "api/video/video_frame_buffer.h"
>-#include "system_wrappers/include/aligned_malloc.h"
>+#include "api/video/video_rotation.h"
>+#include "rtc_base/memory/aligned_malloc.h"
> 
> namespace webrtc {
> 
>@@ -36,11 +36,14 @@ class I420Buffer : public I420BufferInterface {
>     return Copy(*buffer.GetI420());
>   }
> 
>-  static rtc::scoped_refptr<I420Buffer> Copy(
>-      int width, int height,
>-      const uint8_t* data_y, int stride_y,
>-      const uint8_t* data_u, int stride_u,
>-      const uint8_t* data_v, int stride_v);
>+  static rtc::scoped_refptr<I420Buffer> Copy(int width,
>+                                             int height,
>+                                             const uint8_t* data_y,
>+                                             int stride_y,
>+                                             const uint8_t* data_u,
>+                                             int stride_u,
>+                                             const uint8_t* data_v,
>+                                             int stride_v);
> 
>   // Returns a rotated copy of |src|.
>   static rtc::scoped_refptr<I420Buffer> Rotate(const I420BufferInterface& src,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/memcheck/suppressions_mac.txt b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/test/BUILD.gn
>similarity index 51%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/memcheck/suppressions_mac.txt
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/api/video/test/BUILD.gn
>index ec20df46d8b..cba42458d22 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/memcheck/suppressions_mac.txt
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/test/BUILD.gn
>@@ -1,4 +1,4 @@
>-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
> #
> # Use of this source code is governed by a BSD-style license
> # that can be found in the LICENSE file in the root of the source
>@@ -6,6 +6,15 @@
> # in the file PATENTS.  All contributing project authors may
> # be found in the AUTHORS file in the root of the source tree.
> 
>-# This file is used in addition to the one already maintained in Chrome.
>-# It acts as a place holder for future additions for WebRTC.
>-# It must exist for the Python wrapper script to work properly.
>+import("../../../webrtc.gni")
>+
>+rtc_source_set("rtc_api_video_unittests") {
>+  testonly = true
>+  sources = [
>+    "video_bitrate_allocation_unittest.cc",
>+  ]
>+  deps = [
>+    "..:video_bitrate_allocation",
>+    "../../../test:test_support",
>+  ]
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/test/video_bitrate_allocation_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/test/video_bitrate_allocation_unittest.cc
>new file mode 100644
>index 00000000000..f22957b562e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/test/video_bitrate_allocation_unittest.cc
>@@ -0,0 +1,63 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <memory>
>+#include <string>
>+
>+#include "api/video/video_bitrate_allocation.h"
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+TEST(VideoBitrateAllocation, SimulcastTargetBitrate) {
>+  VideoBitrateAllocation bitrate;
>+  bitrate.SetBitrate(0, 0, 10000);
>+  bitrate.SetBitrate(0, 1, 20000);
>+  bitrate.SetBitrate(1, 0, 40000);
>+  bitrate.SetBitrate(1, 1, 80000);
>+
>+  VideoBitrateAllocation layer0_bitrate;
>+  layer0_bitrate.SetBitrate(0, 0, 10000);
>+  layer0_bitrate.SetBitrate(0, 1, 20000);
>+
>+  VideoBitrateAllocation layer1_bitrate;
>+  layer1_bitrate.SetBitrate(0, 0, 40000);
>+  layer1_bitrate.SetBitrate(0, 1, 80000);
>+
>+  std::vector<absl::optional<VideoBitrateAllocation>> layer_allocations =
>+      bitrate.GetSimulcastAllocations();
>+
>+  EXPECT_EQ(layer0_bitrate, layer_allocations[0]);
>+  EXPECT_EQ(layer1_bitrate, layer_allocations[1]);
>+}
>+
>+TEST(VideoBitrateAllocation, SimulcastTargetBitrateWithInactiveStream) {
>+  // Create bitrate allocation with bitrate only for the first and third stream.
>+  VideoBitrateAllocation bitrate;
>+  bitrate.SetBitrate(0, 0, 10000);
>+  bitrate.SetBitrate(0, 1, 20000);
>+  bitrate.SetBitrate(2, 0, 40000);
>+  bitrate.SetBitrate(2, 1, 80000);
>+
>+  VideoBitrateAllocation layer0_bitrate;
>+  layer0_bitrate.SetBitrate(0, 0, 10000);
>+  layer0_bitrate.SetBitrate(0, 1, 20000);
>+
>+  VideoBitrateAllocation layer2_bitrate;
>+  layer2_bitrate.SetBitrate(0, 0, 40000);
>+  layer2_bitrate.SetBitrate(0, 1, 80000);
>+
>+  std::vector<absl::optional<VideoBitrateAllocation>> layer_allocations =
>+      bitrate.GetSimulcastAllocations();
>+
>+  EXPECT_EQ(layer0_bitrate, layer_allocations[0]);
>+  EXPECT_FALSE(layer_allocations[1]);
>+  EXPECT_EQ(layer2_bitrate, layer_allocations[2]);
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_bitrate_allocation.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_bitrate_allocation.cc
>new file mode 100644
>index 00000000000..6c5ad1eae5f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_bitrate_allocation.cc
>@@ -0,0 +1,185 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/video/video_bitrate_allocation.h"
>+
>+#include <limits>
>+
>+#include "rtc_base/checks.h"
>+#include "rtc_base/numerics/safe_conversions.h"
>+#include "rtc_base/strings/string_builder.h"
>+#include "rtc_base/stringutils.h"
>+
>+namespace webrtc {
>+
>+VideoBitrateAllocation::VideoBitrateAllocation() : sum_(0) {}
>+
>+bool VideoBitrateAllocation::SetBitrate(size_t spatial_index,
>+                                        size_t temporal_index,
>+                                        uint32_t bitrate_bps) {
>+  RTC_CHECK_LT(spatial_index, kMaxSpatialLayers);
>+  RTC_CHECK_LT(temporal_index, kMaxTemporalStreams);
>+  int64_t new_bitrate_sum_bps = sum_;
>+  absl::optional<uint32_t>& layer_bitrate =
>+      bitrates_[spatial_index][temporal_index];
>+  if (layer_bitrate) {
>+    RTC_DCHECK_LE(*layer_bitrate, sum_);
>+    new_bitrate_sum_bps -= *layer_bitrate;
>+  }
>+  new_bitrate_sum_bps += bitrate_bps;
>+  if (new_bitrate_sum_bps > kMaxBitrateBps)
>+    return false;
>+
>+  layer_bitrate = bitrate_bps;
>+  sum_ = rtc::dchecked_cast<uint32_t>(new_bitrate_sum_bps);
>+  return true;
>+}
>+
>+bool VideoBitrateAllocation::HasBitrate(size_t spatial_index,
>+                                        size_t temporal_index) const {
>+  RTC_CHECK_LT(spatial_index, kMaxSpatialLayers);
>+  RTC_CHECK_LT(temporal_index, kMaxTemporalStreams);
>+  return bitrates_[spatial_index][temporal_index].has_value();
>+}
>+
>+uint32_t VideoBitrateAllocation::GetBitrate(size_t spatial_index,
>+                                            size_t temporal_index) const {
>+  RTC_CHECK_LT(spatial_index, kMaxSpatialLayers);
>+  RTC_CHECK_LT(temporal_index, kMaxTemporalStreams);
>+  return bitrates_[spatial_index][temporal_index].value_or(0);
>+}
>+
>+// Whether the specific spatial layers has the bitrate set in any of its
>+// temporal layers.
>+bool VideoBitrateAllocation::IsSpatialLayerUsed(size_t spatial_index) const {
>+  RTC_CHECK_LT(spatial_index, kMaxSpatialLayers);
>+  for (size_t i = 0; i < kMaxTemporalStreams; ++i) {
>+    if (bitrates_[spatial_index][i].has_value())
>+      return true;
>+  }
>+  return false;
>+}
>+
>+// Get the sum of all the temporal layer for a specific spatial layer.
>+uint32_t VideoBitrateAllocation::GetSpatialLayerSum(
>+    size_t spatial_index) const {
>+  RTC_CHECK_LT(spatial_index, kMaxSpatialLayers);
>+  return GetTemporalLayerSum(spatial_index, kMaxTemporalStreams - 1);
>+}
>+
>+uint32_t VideoBitrateAllocation::GetTemporalLayerSum(
>+    size_t spatial_index,
>+    size_t temporal_index) const {
>+  RTC_CHECK_LT(spatial_index, kMaxSpatialLayers);
>+  RTC_CHECK_LT(temporal_index, kMaxTemporalStreams);
>+  uint32_t sum = 0;
>+  for (size_t i = 0; i <= temporal_index; ++i) {
>+    sum += bitrates_[spatial_index][i].value_or(0);
>+  }
>+  return sum;
>+}
>+
>+std::vector<uint32_t> VideoBitrateAllocation::GetTemporalLayerAllocation(
>+    size_t spatial_index) const {
>+  RTC_CHECK_LT(spatial_index, kMaxSpatialLayers);
>+  std::vector<uint32_t> temporal_rates;
>+
>+  // Find the highest temporal layer with a defined bitrate in order to
>+  // determine the size of the temporal layer allocation.
>+  for (size_t i = kMaxTemporalStreams; i > 0; --i) {
>+    if (bitrates_[spatial_index][i - 1].has_value()) {
>+      temporal_rates.resize(i);
>+      break;
>+    }
>+  }
>+
>+  for (size_t i = 0; i < temporal_rates.size(); ++i) {
>+    temporal_rates[i] = bitrates_[spatial_index][i].value_or(0);
>+  }
>+
>+  return temporal_rates;
>+}
>+
>+std::vector<absl::optional<VideoBitrateAllocation>>
>+VideoBitrateAllocation::GetSimulcastAllocations() const {
>+  std::vector<absl::optional<VideoBitrateAllocation>> bitrates;
>+  for (size_t si = 0; si < kMaxSpatialLayers; ++si) {
>+    absl::optional<VideoBitrateAllocation> layer_bitrate;
>+    if (IsSpatialLayerUsed(si)) {
>+      layer_bitrate = VideoBitrateAllocation();
>+      for (int tl = 0; tl < kMaxTemporalStreams; ++tl) {
>+        if (HasBitrate(si, tl))
>+          layer_bitrate->SetBitrate(0, tl, GetBitrate(si, tl));
>+      }
>+    }
>+    bitrates.push_back(layer_bitrate);
>+  }
>+  return bitrates;
>+}
>+
>+bool VideoBitrateAllocation::operator==(
>+    const VideoBitrateAllocation& other) const {
>+  for (size_t si = 0; si < kMaxSpatialLayers; ++si) {
>+    for (size_t ti = 0; ti < kMaxTemporalStreams; ++ti) {
>+      if (bitrates_[si][ti] != other.bitrates_[si][ti])
>+        return false;
>+    }
>+  }
>+  return true;
>+}
>+
>+std::string VideoBitrateAllocation::ToString() const {
>+  if (sum_ == 0)
>+    return "VideoBitrateAllocation [ [] ]";
>+
>+  // Max string length in practice is 260, but let's have some overhead and
>+  // round up to nearest power of two.
>+  char string_buf[512];
>+  rtc::SimpleStringBuilder ssb(string_buf);
>+
>+  ssb << "VideoBitrateAllocation [";
>+  uint32_t spatial_cumulator = 0;
>+  for (size_t si = 0; si < kMaxSpatialLayers; ++si) {
>+    RTC_DCHECK_LE(spatial_cumulator, sum_);
>+    if (spatial_cumulator == sum_)
>+      break;
>+
>+    const uint32_t layer_sum = GetSpatialLayerSum(si);
>+    if (layer_sum == sum_) {
>+      ssb << " [";
>+    } else {
>+      if (si > 0)
>+        ssb << ",";
>+      ssb << '\n' << "  [";
>+    }
>+    spatial_cumulator += layer_sum;
>+
>+    uint32_t temporal_cumulator = 0;
>+    for (size_t ti = 0; ti < kMaxTemporalStreams; ++ti) {
>+      RTC_DCHECK_LE(temporal_cumulator, layer_sum);
>+      if (temporal_cumulator == layer_sum)
>+        break;
>+
>+      if (ti > 0)
>+        ssb << ", ";
>+
>+      uint32_t bitrate = bitrates_[si][ti].value_or(0);
>+      ssb << bitrate;
>+      temporal_cumulator += bitrate;
>+    }
>+    ssb << "]";
>+  }
>+
>+  RTC_DCHECK_EQ(spatial_cumulator, sum_);
>+  ssb << " ]";
>+  return ssb.str();
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_bitrate_allocation.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_bitrate_allocation.h
>new file mode 100644
>index 00000000000..9e2501d85a0
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_bitrate_allocation.h
>@@ -0,0 +1,90 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_VIDEO_VIDEO_BITRATE_ALLOCATION_H_
>+#define API_VIDEO_VIDEO_BITRATE_ALLOCATION_H_
>+
>+#include <limits>
>+#include <string>
>+#include <vector>
>+
>+#include "absl/types/optional.h"
>+
>+namespace webrtc {
>+
>+// TODO(sprang): Move back to common_types when include of this is removed.
>+enum : int { kMaxSimulcastStreams = 4 };
>+enum : int { kMaxSpatialLayers = 5 };
>+enum : int { kMaxTemporalStreams = 4 };
>+
>+// Class that describes how video bitrate, in bps, is allocated across temporal
>+// and spatial layers. Not that bitrates are NOT cumulative. Depending on if
>+// layers are dependent or not, it is up to the user to aggregate.
>+// For each index, the bitrate can also both set and unset. This is used with a
>+// set bps = 0 to signal an explicit "turn off" signal.
>+class VideoBitrateAllocation {
>+ public:
>+  static constexpr uint32_t kMaxBitrateBps =
>+      std::numeric_limits<uint32_t>::max();
>+  VideoBitrateAllocation();
>+
>+  bool SetBitrate(size_t spatial_index,
>+                  size_t temporal_index,
>+                  uint32_t bitrate_bps);
>+
>+  bool HasBitrate(size_t spatial_index, size_t temporal_index) const;
>+
>+  uint32_t GetBitrate(size_t spatial_index, size_t temporal_index) const;
>+
>+  // Whether the specific spatial layers has the bitrate set in any of its
>+  // temporal layers.
>+  bool IsSpatialLayerUsed(size_t spatial_index) const;
>+
>+  // Get the sum of all the temporal layer for a specific spatial layer.
>+  uint32_t GetSpatialLayerSum(size_t spatial_index) const;
>+
>+  // Sum of bitrates of temporal layers, from layer 0 to |temporal_index|
>+  // inclusive, of specified spatial layer |spatial_index|. Bitrates of lower
>+  // spatial layers are not included.
>+  uint32_t GetTemporalLayerSum(size_t spatial_index,
>+                               size_t temporal_index) const;
>+
>+  // Returns a vector of the temporal layer bitrates for the specific spatial
>+  // layer. Length of the returned vector is cropped to the highest temporal
>+  // layer with a defined bitrate.
>+  std::vector<uint32_t> GetTemporalLayerAllocation(size_t spatial_index) const;
>+
>+  // Returns one VideoBitrateAllocation for each spatial layer. This is used to
>+  // configure simulcast streams. Note that the length of the returned vector is
>+  // always kMaxSpatialLayers, the optional is unset for unused layers.
>+  std::vector<absl::optional<VideoBitrateAllocation>> GetSimulcastAllocations()
>+      const;
>+
>+  uint32_t get_sum_bps() const { return sum_; }  // Sum of all bitrates.
>+  uint32_t get_sum_kbps() const {
>+    // Round down to not exceed the allocated bitrate.
>+    return sum_ / 1000;
>+  }
>+
>+  bool operator==(const VideoBitrateAllocation& other) const;
>+  inline bool operator!=(const VideoBitrateAllocation& other) const {
>+    return !(*this == other);
>+  }
>+
>+  std::string ToString() const;
>+
>+ private:
>+  uint32_t sum_;
>+  absl::optional<uint32_t> bitrates_[kMaxSpatialLayers][kMaxTemporalStreams];
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // API_VIDEO_VIDEO_BITRATE_ALLOCATION_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_bitrate_allocator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_bitrate_allocator.h
>new file mode 100644
>index 00000000000..f85c633d0ff
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_bitrate_allocator.h
>@@ -0,0 +1,38 @@
>+/*
>+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_VIDEO_VIDEO_BITRATE_ALLOCATOR_H_
>+#define API_VIDEO_VIDEO_BITRATE_ALLOCATOR_H_
>+
>+#include "api/video/video_bitrate_allocation.h"
>+
>+namespace webrtc {
>+
>+class VideoBitrateAllocator {
>+ public:
>+  VideoBitrateAllocator() {}
>+  virtual ~VideoBitrateAllocator() {}
>+
>+  virtual VideoBitrateAllocation GetAllocation(uint32_t total_bitrate_bps,
>+                                               uint32_t framerate) = 0;
>+};
>+
>+class VideoBitrateAllocationObserver {
>+ public:
>+  VideoBitrateAllocationObserver() {}
>+  virtual ~VideoBitrateAllocationObserver() {}
>+
>+  virtual void OnBitrateAllocationUpdated(
>+      const VideoBitrateAllocation& allocation) = 0;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // API_VIDEO_VIDEO_BITRATE_ALLOCATOR_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_bitrate_allocator_factory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_bitrate_allocator_factory.h
>new file mode 100644
>index 00000000000..bb1d8b60a14
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_bitrate_allocator_factory.h
>@@ -0,0 +1,32 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_VIDEO_VIDEO_BITRATE_ALLOCATOR_FACTORY_H_
>+#define API_VIDEO_VIDEO_BITRATE_ALLOCATOR_FACTORY_H_
>+
>+#include <memory>
>+#include "api/video/video_bitrate_allocator.h"
>+#include "media/base/codec.h"
>+
>+namespace webrtc {
>+
>+// A factory that creates VideoBitrateAllocator.
>+// NOTE: This class is still under development and may change without notice.
>+class VideoBitrateAllocatorFactory {
>+ public:
>+  virtual ~VideoBitrateAllocatorFactory() = default;
>+  // Creates a VideoBitrateAllocator for a specific video codec.
>+  virtual std::unique_ptr<VideoBitrateAllocator> CreateVideoBitrateAllocator(
>+      const VideoCodec& codec) = 0;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // API_VIDEO_VIDEO_BITRATE_ALLOCATOR_FACTORY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_content_type.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_content_type.cc
>index 149b4f9926b..9ba3ece79b6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_content_type.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_content_type.cc
>@@ -68,19 +68,16 @@ bool SetSimulcastId(VideoContentType* content_type, uint8_t simulcast_id) {
>   return true;
> }
> 
>-uint8_t GetExperimentId(
>-    const VideoContentType& content_type) {
>+uint8_t GetExperimentId(const VideoContentType& content_type) {
>   return (static_cast<uint8_t>(content_type) & kExperimentBitsMask) >>
>          kExperimentShift;
> }
>-uint8_t GetSimulcastId(
>-    const VideoContentType& content_type) {
>+uint8_t GetSimulcastId(const VideoContentType& content_type) {
>   return (static_cast<uint8_t>(content_type) & kSimulcastBitsMask) >>
>          kSimulcastShift;
> }
> 
>-bool IsScreenshare(
>-    const VideoContentType& content_type) {
>+bool IsScreenshare(const VideoContentType& content_type) {
>   return (static_cast<uint8_t>(content_type) & kScreenshareBitsMask) > 0;
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_frame.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_frame.cc
>index 93b3c9c6b97..5521ad8a587 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_frame.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_frame.cc
>@@ -15,6 +15,55 @@
> 
> namespace webrtc {
> 
>+VideoFrame::Builder::Builder() = default;
>+
>+VideoFrame::Builder::~Builder() = default;
>+
>+VideoFrame VideoFrame::Builder::build() {
>+  return VideoFrame(video_frame_buffer_, timestamp_us_, timestamp_rtp_,
>+                    ntp_time_ms_, rotation_, color_space_);
>+}
>+
>+VideoFrame::Builder& VideoFrame::Builder::set_video_frame_buffer(
>+    const rtc::scoped_refptr<VideoFrameBuffer>& buffer) {
>+  video_frame_buffer_ = buffer;
>+  return *this;
>+}
>+
>+VideoFrame::Builder& VideoFrame::Builder::set_timestamp_ms(
>+    int64_t timestamp_ms) {
>+  timestamp_us_ = timestamp_ms * rtc::kNumMicrosecsPerMillisec;
>+  return *this;
>+}
>+
>+VideoFrame::Builder& VideoFrame::Builder::set_timestamp_us(
>+    int64_t timestamp_us) {
>+  timestamp_us_ = timestamp_us;
>+  return *this;
>+}
>+
>+VideoFrame::Builder& VideoFrame::Builder::set_timestamp_rtp(
>+    uint32_t timestamp_rtp) {
>+  timestamp_rtp_ = timestamp_rtp;
>+  return *this;
>+}
>+
>+VideoFrame::Builder& VideoFrame::Builder::set_ntp_time_ms(int64_t ntp_time_ms) {
>+  ntp_time_ms_ = ntp_time_ms;
>+  return *this;
>+}
>+
>+VideoFrame::Builder& VideoFrame::Builder::set_rotation(VideoRotation rotation) {
>+  rotation_ = rotation;
>+  return *this;
>+}
>+
>+VideoFrame::Builder& VideoFrame::Builder::set_color_space(
>+    const ColorSpace& color_space) {
>+  color_space_ = color_space;
>+  return *this;
>+}
>+
> VideoFrame::VideoFrame(const rtc::scoped_refptr<VideoFrameBuffer>& buffer,
>                        webrtc::VideoRotation rotation,
>                        int64_t timestamp_us)
>@@ -25,17 +74,30 @@ VideoFrame::VideoFrame(const rtc::scoped_refptr<VideoFrameBuffer>& buffer,
>       rotation_(rotation) {}
> 
> VideoFrame::VideoFrame(const rtc::scoped_refptr<VideoFrameBuffer>& buffer,
>-                       uint32_t timestamp,
>+                       uint32_t timestamp_rtp,
>                        int64_t render_time_ms,
>                        VideoRotation rotation)
>     : video_frame_buffer_(buffer),
>-      timestamp_rtp_(timestamp),
>+      timestamp_rtp_(timestamp_rtp),
>       ntp_time_ms_(0),
>       timestamp_us_(render_time_ms * rtc::kNumMicrosecsPerMillisec),
>       rotation_(rotation) {
>   RTC_DCHECK(buffer);
> }
> 
>+VideoFrame::VideoFrame(const rtc::scoped_refptr<VideoFrameBuffer>& buffer,
>+                       int64_t timestamp_us,
>+                       uint32_t timestamp_rtp,
>+                       int64_t ntp_time_ms,
>+                       VideoRotation rotation,
>+                       const absl::optional<ColorSpace>& color_space)
>+    : video_frame_buffer_(buffer),
>+      timestamp_rtp_(timestamp_rtp),
>+      ntp_time_ms_(ntp_time_ms),
>+      timestamp_us_(timestamp_us),
>+      rotation_(rotation),
>+      color_space_(color_space) {}
>+
> VideoFrame::~VideoFrame() = default;
> 
> VideoFrame::VideoFrame(const VideoFrame&) = default;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_frame.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_frame.h
>index a72bef1d32e..dcb533e29f0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_frame.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_frame.h
>@@ -13,23 +13,46 @@
> 
> #include <stdint.h>
> 
>-#include "api/video/video_rotation.h"
>+#include "absl/types/optional.h"
>+#include "api/video/color_space.h"
> #include "api/video/video_frame_buffer.h"
>+#include "api/video/video_rotation.h"
> 
> namespace webrtc {
> 
> class VideoFrame {
>  public:
>-  // TODO(nisse): This constructor is consistent with the now deleted
>-  // cricket::WebRtcVideoFrame. We should consider whether or not we
>-  // want to stick to this style and deprecate the other constructor.
>+  // Preferred way of building VideoFrame objects.
>+  class Builder {
>+   public:
>+    Builder();
>+    ~Builder();
>+
>+    VideoFrame build();
>+    Builder& set_video_frame_buffer(
>+        const rtc::scoped_refptr<VideoFrameBuffer>& buffer);
>+    Builder& set_timestamp_ms(int64_t timestamp_ms);
>+    Builder& set_timestamp_us(int64_t timestamp_us);
>+    Builder& set_timestamp_rtp(uint32_t timestamp_rtp);
>+    Builder& set_ntp_time_ms(int64_t ntp_time_ms);
>+    Builder& set_rotation(VideoRotation rotation);
>+    Builder& set_color_space(const ColorSpace& color_space);
>+
>+   private:
>+    rtc::scoped_refptr<webrtc::VideoFrameBuffer> video_frame_buffer_;
>+    int64_t timestamp_us_ = 0;
>+    uint32_t timestamp_rtp_ = 0;
>+    int64_t ntp_time_ms_ = 0;
>+    VideoRotation rotation_ = kVideoRotation_0;
>+    absl::optional<ColorSpace> color_space_;
>+  };
>+
>+  // To be deprecated. Migrate all use to Builder.
>   VideoFrame(const rtc::scoped_refptr<VideoFrameBuffer>& buffer,
>              webrtc::VideoRotation rotation,
>              int64_t timestamp_us);
>-
>-  // Preferred constructor.
>   VideoFrame(const rtc::scoped_refptr<VideoFrameBuffer>& buffer,
>-             uint32_t timestamp,
>+             uint32_t timestamp_rtp,
>              int64_t render_time_ms,
>              VideoRotation rotation);
> 
>@@ -87,6 +110,9 @@ class VideoFrame {
>   VideoRotation rotation() const { return rotation_; }
>   void set_rotation(VideoRotation rotation) { rotation_ = rotation; }
> 
>+  // Set Color space when available.
>+  absl::optional<ColorSpace> color_space() const { return color_space_; }
>+
>   // Get render time in milliseconds.
>   // TODO(nisse): Deprecated. Migrate all users to timestamp_us().
>   int64_t render_time_ms() const;
>@@ -102,12 +128,20 @@ class VideoFrame {
>   }
> 
>  private:
>+  VideoFrame(const rtc::scoped_refptr<VideoFrameBuffer>& buffer,
>+             int64_t timestamp_us,
>+             uint32_t timestamp_rtp,
>+             int64_t ntp_time_ms,
>+             VideoRotation rotation,
>+             const absl::optional<ColorSpace>& color_space);
>+
>   // An opaque reference counted handle that stores the pixel data.
>   rtc::scoped_refptr<webrtc::VideoFrameBuffer> video_frame_buffer_;
>   uint32_t timestamp_rtp_;
>   int64_t ntp_time_ms_;
>   int64_t timestamp_us_;
>   VideoRotation rotation_;
>+  absl::optional<ColorSpace> color_space_;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_frame_buffer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_frame_buffer.cc
>index 867f249fe63..41276bec270 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_frame_buffer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_frame_buffer.cc
>@@ -45,6 +45,16 @@ const I444BufferInterface* VideoFrameBuffer::GetI444() const {
>   return static_cast<const I444BufferInterface*>(this);
> }
> 
>+I010BufferInterface* VideoFrameBuffer::GetI010() {
>+  RTC_CHECK(type() == Type::kI010);
>+  return static_cast<I010BufferInterface*>(this);
>+}
>+
>+const I010BufferInterface* VideoFrameBuffer::GetI010() const {
>+  RTC_CHECK(type() == Type::kI010);
>+  return static_cast<const I010BufferInterface*>(this);
>+}
>+
> VideoFrameBuffer::Type I420BufferInterface::type() const {
>   return Type::kI420;
> }
>@@ -77,4 +87,16 @@ int I444BufferInterface::ChromaHeight() const {
>   return height();
> }
> 
>+VideoFrameBuffer::Type I010BufferInterface::type() const {
>+  return Type::kI010;
>+}
>+
>+int I010BufferInterface::ChromaWidth() const {
>+  return (width() + 1) / 2;
>+}
>+
>+int I010BufferInterface::ChromaHeight() const {
>+  return (height() + 1) / 2;
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_frame_buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_frame_buffer.h
>index 2be7e0bb9f6..1e8169ac57c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_frame_buffer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_frame_buffer.h
>@@ -21,6 +21,7 @@ namespace webrtc {
> class I420BufferInterface;
> class I420ABufferInterface;
> class I444BufferInterface;
>+class I010BufferInterface;
> 
> // Base class for frame buffers of different types of pixel format and storage.
> // The tag in type() indicates how the data is represented, and each type is
>@@ -47,6 +48,7 @@ class VideoFrameBuffer : public rtc::RefCountInterface {
>     kI420,
>     kI420A,
>     kI444,
>+    kI010,
>   };
> 
>   // This function specifies in what pixel format the data is stored in.
>@@ -73,33 +75,44 @@ class VideoFrameBuffer : public rtc::RefCountInterface {
>   const I420ABufferInterface* GetI420A() const;
>   I444BufferInterface* GetI444();
>   const I444BufferInterface* GetI444() const;
>+  I010BufferInterface* GetI010();
>+  const I010BufferInterface* GetI010() const;
> 
>  protected:
>   ~VideoFrameBuffer() override {}
> };
> 
>-// This interface represents Type::kI420 and Type::kI444.
>+// This interface represents planar formats.
> class PlanarYuvBuffer : public VideoFrameBuffer {
>  public:
>   virtual int ChromaWidth() const = 0;
>   virtual int ChromaHeight() const = 0;
> 
>+  // Returns the number of steps(in terms of Data*() return type) between
>+  // successive rows for a given plane.
>+  virtual int StrideY() const = 0;
>+  virtual int StrideU() const = 0;
>+  virtual int StrideV() const = 0;
>+
>+ protected:
>+  ~PlanarYuvBuffer() override {}
>+};
>+
>+// This interface represents 8-bit color depth formats: Type::kI420,
>+// Type::kI420A and Type::kI444.
>+class PlanarYuv8Buffer : public PlanarYuvBuffer {
>+ public:
>   // Returns pointer to the pixel data for a given plane. The memory is owned by
>   // the VideoFrameBuffer object and must not be freed by the caller.
>   virtual const uint8_t* DataY() const = 0;
>   virtual const uint8_t* DataU() const = 0;
>   virtual const uint8_t* DataV() const = 0;
> 
>-  // Returns the number of bytes between successive rows for a given plane.
>-  virtual int StrideY() const = 0;
>-  virtual int StrideU() const = 0;
>-  virtual int StrideV() const = 0;
>-
>  protected:
>-  ~PlanarYuvBuffer() override {}
>+  ~PlanarYuv8Buffer() override {}
> };
> 
>-class I420BufferInterface : public PlanarYuvBuffer {
>+class I420BufferInterface : public PlanarYuv8Buffer {
>  public:
>   Type type() const override;
> 
>@@ -122,7 +135,7 @@ class I420ABufferInterface : public I420BufferInterface {
>   ~I420ABufferInterface() override {}
> };
> 
>-class I444BufferInterface : public PlanarYuvBuffer {
>+class I444BufferInterface : public PlanarYuv8Buffer {
>  public:
>   Type type() const final;
> 
>@@ -133,6 +146,32 @@ class I444BufferInterface : public PlanarYuvBuffer {
>   ~I444BufferInterface() override {}
> };
> 
>+// This interface represents 8-bit to 16-bit color depth formats: Type::kI010.
>+class PlanarYuv16BBuffer : public PlanarYuvBuffer {
>+ public:
>+  // Returns pointer to the pixel data for a given plane. The memory is owned by
>+  // the VideoFrameBuffer object and must not be freed by the caller.
>+  virtual const uint16_t* DataY() const = 0;
>+  virtual const uint16_t* DataU() const = 0;
>+  virtual const uint16_t* DataV() const = 0;
>+
>+ protected:
>+  ~PlanarYuv16BBuffer() override {}
>+};
>+
>+// Represents Type::kI010, allocates 16 bits per pixel and fills 10 least
>+// significant bits with color information.
>+class I010BufferInterface : public PlanarYuv16BBuffer {
>+ public:
>+  Type type() const override;
>+
>+  int ChromaWidth() const final;
>+  int ChromaHeight() const final;
>+
>+ protected:
>+  ~I010BufferInterface() override {}
>+};
>+
> }  // namespace webrtc
> 
> #endif  // API_VIDEO_VIDEO_FRAME_BUFFER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/videosinkinterface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_sink_interface.h
>similarity index 81%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/api/videosinkinterface.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_sink_interface.h
>index 23993205417..aac8b4ac38a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/videosinkinterface.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_sink_interface.h
>@@ -8,8 +8,8 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#ifndef API_VIDEOSINKINTERFACE_H_
>-#define API_VIDEOSINKINTERFACE_H_
>+#ifndef API_VIDEO_VIDEO_SINK_INTERFACE_H_
>+#define API_VIDEO_VIDEO_SINK_INTERFACE_H_
> 
> #include <rtc_base/checks.h>
> 
>@@ -18,7 +18,7 @@ namespace rtc {
> template <typename VideoFrameT>
> class VideoSinkInterface {
>  public:
>-  virtual ~VideoSinkInterface() {}
>+  virtual ~VideoSinkInterface() = default;
> 
>   virtual void OnFrame(const VideoFrameT& frame) = 0;
> 
>@@ -29,4 +29,4 @@ class VideoSinkInterface {
> 
> }  // namespace rtc
> 
>-#endif  // API_VIDEOSINKINTERFACE_H_
>+#endif  // API_VIDEO_VIDEO_SINK_INTERFACE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/videosourceinterface.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_source_interface.cc
>similarity index 83%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/api/videosourceinterface.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_source_interface.cc
>index 5eda369df08..70a86c3d645 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/videosourceinterface.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_source_interface.cc
>@@ -8,11 +8,12 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "api/videosourceinterface.h"
>+#include "api/video/video_source_interface.h"
> 
> namespace rtc {
> 
> VideoSinkWants::VideoSinkWants() = default;
>+VideoSinkWants::VideoSinkWants(const VideoSinkWants&) = default;
> VideoSinkWants::~VideoSinkWants() = default;
> 
> }  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/videosourceinterface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_source_interface.h
>similarity index 84%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/api/videosourceinterface.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_source_interface.h
>index ffb017ad51d..4ee47194552 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/videosourceinterface.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_source_interface.h
>@@ -8,13 +8,13 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#ifndef API_VIDEOSOURCEINTERFACE_H_
>-#define API_VIDEOSOURCEINTERFACE_H_
>+#ifndef API_VIDEO_VIDEO_SOURCE_INTERFACE_H_
>+#define API_VIDEO_VIDEO_SOURCE_INTERFACE_H_
> 
> #include <limits>
> 
>-#include "api/optional.h"
>-#include "api/videosinkinterface.h"
>+#include "absl/types/optional.h"
>+#include "api/video/video_sink_interface.h"
> 
> namespace rtc {
> 
>@@ -22,6 +22,7 @@ namespace rtc {
> // should have when it is delivered to a certain sink.
> struct VideoSinkWants {
>   VideoSinkWants();
>+  VideoSinkWants(const VideoSinkWants&);
>   ~VideoSinkWants();
>   // Tells the source whether the sink wants frames with rotation applied.
>   // By default, any rotation must be applied by the sink.
>@@ -37,7 +38,7 @@ struct VideoSinkWants {
>   // have improved after an earlier downgrade. The source should select the
>   // closest resolution to this pixel count, but if max_pixel_count is set, it
>   // still sets the absolute upper bound.
>-  rtc::Optional<int> target_pixel_count;
>+  absl::optional<int> target_pixel_count;
>   // Tells the source the maximum framerate the sink wants.
>   int max_framerate_fps = std::numeric_limits<int>::max();
> };
>@@ -52,8 +53,9 @@ class VideoSourceInterface {
>   virtual void RemoveSink(VideoSinkInterface<VideoFrameT>* sink) = 0;
> 
>  protected:
>+  // Non-public, since one shouldn't own sources via this interface.
>   virtual ~VideoSourceInterface() {}
> };
> 
> }  // namespace rtc
>-#endif  // API_VIDEOSOURCEINTERFACE_H_
>+#endif  // API_VIDEO_VIDEO_SOURCE_INTERFACE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_stream_decoder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_stream_decoder.h
>new file mode 100644
>index 00000000000..dff60d87e63
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_stream_decoder.h
>@@ -0,0 +1,51 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_VIDEO_VIDEO_STREAM_DECODER_H_
>+#define API_VIDEO_VIDEO_STREAM_DECODER_H_
>+
>+#include <map>
>+#include <memory>
>+#include <utility>
>+
>+#include "api/video/encoded_frame.h"
>+#include "api/video/video_frame.h"
>+#include "api/video_codecs/sdp_video_format.h"
>+#include "api/video_codecs/video_decoder_factory.h"
>+
>+namespace webrtc {
>+// NOTE: This class is still under development and may change without notice.
>+class VideoStreamDecoder {
>+ public:
>+  class Callbacks {
>+   public:
>+    virtual ~Callbacks() = default;
>+
>+    // Called when the VideoStreamDecoder enters a non-decodable state.
>+    virtual void OnNonDecodableState() = 0;
>+
>+    // Called with the last continuous frame.
>+    virtual void OnContinuousUntil(
>+        const video_coding::VideoLayerFrameId& key) = 0;
>+
>+    // Called with the decoded frame.
>+    virtual void OnDecodedFrame(VideoFrame decodedImage,
>+                                absl::optional<int> decode_time_ms,
>+                                absl::optional<int> qp) = 0;
>+  };
>+
>+  virtual ~VideoStreamDecoder() = default;
>+
>+  virtual void OnFrame(std::unique_ptr<video_coding::EncodedFrame> frame) = 0;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // API_VIDEO_VIDEO_STREAM_DECODER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_stream_decoder_create.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_stream_decoder_create.cc
>new file mode 100644
>index 00000000000..d57925524bc
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_stream_decoder_create.cc
>@@ -0,0 +1,24 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/video/video_stream_decoder_create.h"
>+
>+#include "absl/memory/memory.h"
>+#include "video/video_stream_decoder_impl.h"
>+
>+namespace webrtc {
>+std::unique_ptr<VideoStreamDecoder> CreateVideoStreamDecoder(
>+    VideoStreamDecoder::Callbacks* callbacks,
>+    VideoDecoderFactory* decoder_factory,
>+    std::map<int, std::pair<SdpVideoFormat, int>> decoder_settings) {
>+  return absl::make_unique<VideoStreamDecoderImpl>(callbacks, decoder_factory,
>+                                                   std::move(decoder_settings));
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_stream_decoder_create.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_stream_decoder_create.h
>new file mode 100644
>index 00000000000..04682901da1
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_stream_decoder_create.h
>@@ -0,0 +1,32 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_VIDEO_VIDEO_STREAM_DECODER_CREATE_H_
>+#define API_VIDEO_VIDEO_STREAM_DECODER_CREATE_H_
>+
>+#include <map>
>+#include <memory>
>+#include <utility>
>+
>+#include "api/video/video_stream_decoder.h"
>+
>+namespace webrtc {
>+// The |decoder_settings| parameter is a map between:
>+// <payload type> -->  <<video format>, <number of cores>>.
>+// The video format is used when instantiating a decoder, and
>+// the number of cores is used when initializing the decoder.
>+std::unique_ptr<VideoStreamDecoder> CreateVideoStreamDecoder(
>+    VideoStreamDecoder::Callbacks* callbacks,
>+    VideoDecoderFactory* decoder_factory,
>+    std::map<int, std::pair<SdpVideoFormat, int>> decoder_settings);
>+
>+}  // namespace webrtc
>+
>+#endif  // API_VIDEO_VIDEO_STREAM_DECODER_CREATE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_stream_encoder_create.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_stream_encoder_create.cc
>new file mode 100644
>index 00000000000..3147d34eea7
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_stream_encoder_create.cc
>@@ -0,0 +1,27 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/video/video_stream_encoder_create.h"
>+
>+#include "absl/memory/memory.h"
>+#include "video/video_stream_encoder.h"
>+
>+namespace webrtc {
>+std::unique_ptr<VideoStreamEncoderInterface> CreateVideoStreamEncoder(
>+    uint32_t number_of_cores,
>+    VideoStreamEncoderObserver* encoder_stats_observer,
>+    const VideoStreamEncoderSettings& settings,
>+    // Deprecated, used for tests only.
>+    rtc::VideoSinkInterface<VideoFrame>* pre_encode_callback) {
>+  return absl::make_unique<VideoStreamEncoder>(
>+      number_of_cores, encoder_stats_observer, settings, pre_encode_callback,
>+      absl::make_unique<OveruseFrameDetector>(encoder_stats_observer));
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_stream_encoder_create.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_stream_encoder_create.h
>new file mode 100644
>index 00000000000..911727327cf
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_stream_encoder_create.h
>@@ -0,0 +1,41 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_VIDEO_VIDEO_STREAM_ENCODER_CREATE_H_
>+#define API_VIDEO_VIDEO_STREAM_ENCODER_CREATE_H_
>+
>+#include <map>
>+#include <memory>
>+#include <utility>
>+
>+#include "api/video/video_stream_encoder_interface.h"
>+#include "api/video/video_stream_encoder_observer.h"
>+#include "api/video/video_stream_encoder_settings.h"
>+
>+namespace webrtc {
>+
>+std::unique_ptr<VideoStreamEncoderInterface> CreateVideoStreamEncoder(
>+    uint32_t number_of_cores,
>+    VideoStreamEncoderObserver* encoder_stats_observer,
>+    const VideoStreamEncoderSettings& settings,
>+    // Deprecated, used for tests only.
>+    rtc::VideoSinkInterface<VideoFrame>* pre_encode_callback);
>+
>+inline std::unique_ptr<VideoStreamEncoderInterface> CreateVideoStreamEncoder(
>+    uint32_t number_of_cores,
>+    VideoStreamEncoderObserver* encoder_stats_observer,
>+    const VideoStreamEncoderSettings& settings) {
>+  return CreateVideoStreamEncoder(number_of_cores, encoder_stats_observer,
>+                                  settings, nullptr);
>+}
>+
>+}  // namespace webrtc
>+
>+#endif  // API_VIDEO_VIDEO_STREAM_ENCODER_CREATE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_stream_encoder_interface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_stream_encoder_interface.h
>new file mode 100644
>index 00000000000..2f95e5851e1
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_stream_encoder_interface.h
>@@ -0,0 +1,105 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_VIDEO_VIDEO_STREAM_ENCODER_INTERFACE_H_
>+#define API_VIDEO_VIDEO_STREAM_ENCODER_INTERFACE_H_
>+
>+#include <vector>
>+
>+#include "api/rtpparameters.h"  // For DegradationPreference.
>+#include "api/video/video_sink_interface.h"
>+#include "api/video/video_source_interface.h"
>+#include "api/video_codecs/video_encoder.h"
>+#include "api/video_codecs/video_encoder_config.h"
>+
>+namespace webrtc {
>+
>+// TODO(nisse): Move full declaration to api/.
>+class VideoBitrateAllocationObserver;
>+
>+// This interface represents a class responsible for creating and driving the
>+// encoder(s) for a single video stream. It is also responsible for adaptation
>+// decisions related to video quality, requesting reduced frame rate or
>+// resolution from the VideoSource when needed.
>+// TODO(bugs.webrtc.org/8830): This interface is under development. Changes
>+// under consideration include:
>+//
>+// 1. Taking out responsibility for adaptation decisions, instead only reporting
>+//    per-frame measurements to the decision maker.
>+//
>+// 2. Moving responsibility for simulcast and for software fallback into this
>+//    class.
>+class VideoStreamEncoderInterface : public rtc::VideoSinkInterface<VideoFrame> {
>+ public:
>+  // Interface for receiving encoded video frames and notifications about
>+  // configuration changes.
>+  class EncoderSink : public EncodedImageCallback {
>+   public:
>+    virtual void OnEncoderConfigurationChanged(
>+        std::vector<VideoStream> streams,
>+        int min_transmit_bitrate_bps) = 0;
>+  };
>+
>+  // Sets the source that will provide video frames to the VideoStreamEncoder's
>+  // OnFrame method. |degradation_preference| control whether or not resolution
>+  // or frame rate may be reduced. The VideoStreamEncoder registers itself with
>+  // |source|, and signals adaptation decisions to the source in the form of
>+  // VideoSinkWants.
>+  // TODO(nisse): When adaptation logic is extracted from this class,
>+  // it no longer needs to know the source.
>+  virtual void SetSource(
>+      rtc::VideoSourceInterface<VideoFrame>* source,
>+      const DegradationPreference& degradation_preference) = 0;
>+
>+  // Sets the |sink| that gets the encoded frames. |rotation_applied| means
>+  // that the source must support rotation. Only set |rotation_applied| if the
>+  // remote side does not support the rotation extension.
>+  virtual void SetSink(EncoderSink* sink, bool rotation_applied) = 0;
>+
>+  // Sets an initial bitrate, later overriden by OnBitrateUpdated. Mainly
>+  // affects the resolution of the initial key frame: If incoming frames are
>+  // larger than reasonable for the start bitrate, and scaling is enabled,
>+  // VideoStreamEncoder asks the source to scale down and drops a few initial
>+  // frames.
>+  // TODO(nisse): This is a poor interface, and mixes bandwidth estimation and
>+  // codec configuration in an undesired way. For the actual send bandwidth, we
>+  // should always be somewhat conservative, but we may nevertheless want to let
>+  // the application configure a more optimistic quality for the initial
>+  // resolution. Should be replaced by a construction time setting.
>+  virtual void SetStartBitrate(int start_bitrate_bps) = 0;
>+
>+  // Request a key frame. Used for signalling from the remote receiver.
>+  virtual void SendKeyFrame() = 0;
>+
>+  // Set the currently estimated network properties. A |bitrate_bps|
>+  // of zero pauses the encoder.
>+  virtual void OnBitrateUpdated(uint32_t bitrate_bps,
>+                                uint8_t fraction_lost,
>+                                int64_t round_trip_time_ms) = 0;
>+
>+  // Register observer for the bitrate allocation between the temporal
>+  // and spatial layers.
>+  virtual void SetBitrateAllocationObserver(
>+      VideoBitrateAllocationObserver* bitrate_observer) = 0;
>+
>+  // Creates and configures an encoder with the given |config|. The
>+  // |max_data_payload_length| is used to support single NAL unit
>+  // packetization for H.264.
>+  virtual void ConfigureEncoder(VideoEncoderConfig config,
>+                                size_t max_data_payload_length) = 0;
>+
>+  // Permanently stop encoding. After this method has returned, it is
>+  // guaranteed that no encoded frames will be delivered to the sink.
>+  virtual void Stop() = 0;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // API_VIDEO_VIDEO_STREAM_ENCODER_INTERFACE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_stream_encoder_observer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_stream_encoder_observer.h
>new file mode 100644
>index 00000000000..ac5c8a1f5be
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_stream_encoder_observer.h
>@@ -0,0 +1,94 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_VIDEO_VIDEO_STREAM_ENCODER_OBSERVER_H_
>+#define API_VIDEO_VIDEO_STREAM_ENCODER_OBSERVER_H_
>+
>+#include <vector>
>+
>+#include "absl/types/optional.h"
>+#include "api/video_codecs/video_encoder.h"
>+#include "api/video_codecs/video_encoder_config.h"
>+
>+namespace webrtc {
>+
>+// TODO(nisse): Used for the OnSendEncodedImage callback below. The callback
>+// wants metadata such as size, encode timing, qp, but doesn't need actual
>+// encoded data. So use some other type to represent that.
>+class EncodedImage;
>+
>+// Broken out into a base class, with public inheritance below, only to ease
>+// unit testing of the internal class OveruseFrameDetector.
>+class CpuOveruseMetricsObserver {
>+ public:
>+  virtual ~CpuOveruseMetricsObserver() = default;
>+  virtual void OnEncodedFrameTimeMeasured(int encode_duration_ms,
>+                                          int encode_usage_percent) = 0;
>+};
>+
>+class VideoStreamEncoderObserver : public CpuOveruseMetricsObserver {
>+ public:
>+  // Number of resolution and framerate reductions (unset if disabled).
>+  struct AdaptationSteps {
>+    absl::optional<int> num_resolution_reductions = 0;
>+    absl::optional<int> num_framerate_reductions = 0;
>+  };
>+
>+  // TODO(nisse): There are too many enums to represent this. Besides
>+  // this one, see AdaptationObserverInterface::AdaptReason and
>+  // WebRtcVideoChannel::AdaptReason.
>+  enum class AdaptationReason {
>+    kNone,  // Used for reset of counters.
>+    kCpu,
>+    kQuality,
>+  };
>+
>+  // TODO(nisse): Duplicates enum EncodedImageCallback::DropReason.
>+  enum class DropReason {
>+    kSource,
>+    kEncoderQueue,
>+    kEncoder,
>+    kMediaOptimization
>+  };
>+
>+  virtual ~VideoStreamEncoderObserver() = default;
>+
>+  virtual void OnIncomingFrame(int width, int height) = 0;
>+
>+  // TODO(nisse): Merge into one callback per encoded frame.
>+  using CpuOveruseMetricsObserver::OnEncodedFrameTimeMeasured;
>+  virtual void OnSendEncodedImage(const EncodedImage& encoded_image,
>+                                  const CodecSpecificInfo* codec_info) = 0;
>+
>+  virtual void OnFrameDropped(DropReason reason) = 0;
>+
>+  // Used to indicate change in content type, which may require a change in
>+  // how stats are collected and set the configured preferred media bitrate.
>+  virtual void OnEncoderReconfigured(
>+      const VideoEncoderConfig& encoder_config,
>+      const std::vector<VideoStream>& streams) = 0;
>+
>+  virtual void OnAdaptationChanged(AdaptationReason reason,
>+                                   const AdaptationSteps& cpu_steps,
>+                                   const AdaptationSteps& quality_steps) = 0;
>+  virtual void OnMinPixelLimitReached() = 0;
>+  virtual void OnInitialQualityResolutionAdaptDown() = 0;
>+
>+  virtual void OnSuspendChange(bool is_suspended) = 0;
>+
>+  // TODO(nisse): VideoStreamEncoder wants to query the stats, which makes this
>+  // not a pure observer. GetInputFrameRate is needed for the cpu adaptation, so
>+  // can be deleted if that responsibility is moved out to a VideoStreamAdaptor
>+  // class.
>+  virtual int GetInputFrameRate() const = 0;
>+};
>+
>+}  // namespace webrtc
>+#endif  // API_VIDEO_VIDEO_STREAM_ENCODER_OBSERVER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_stream_encoder_settings.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_stream_encoder_settings.h
>new file mode 100644
>index 00000000000..b67f33c982d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_stream_encoder_settings.h
>@@ -0,0 +1,31 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_VIDEO_VIDEO_STREAM_ENCODER_SETTINGS_H_
>+#define API_VIDEO_VIDEO_STREAM_ENCODER_SETTINGS_H_
>+
>+#include "api/video_codecs/video_encoder_factory.h"
>+
>+namespace webrtc {
>+
>+struct VideoStreamEncoderSettings {
>+  VideoStreamEncoderSettings() = default;
>+
>+  // Enables the new method to estimate the cpu load from encoding, used for
>+  // cpu adaptation.
>+  bool experiment_cpu_load_estimator = false;
>+
>+  // Ownership stays with WebrtcVideoEngine (delegated from PeerConnection).
>+  VideoEncoderFactory* encoder_factory = nullptr;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // API_VIDEO_VIDEO_STREAM_ENCODER_SETTINGS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_timing.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_timing.cc
>index 3ccbe4eae5b..40de011f0cc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_timing.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_timing.cc
>@@ -10,7 +10,7 @@
> 
> #include "api/video/video_timing.h"
> 
>-#include <sstream>
>+#include "rtc_base/strings/string_builder.h"
> 
> namespace webrtc {
> 
>@@ -28,7 +28,7 @@ TimingFrameInfo::TimingFrameInfo()
>       decode_start_ms(-1),
>       decode_finish_ms(-1),
>       render_time_ms(-1),
>-      flags(TimingFrameFlags::kDefault) {}
>+      flags(VideoSendTiming::kNotTriggered) {}
> 
> int64_t TimingFrameInfo::EndToEndDelay() const {
>   return capture_time_ms >= 0 ? decode_finish_ms - capture_time_ms : -1;
>@@ -48,31 +48,34 @@ bool TimingFrameInfo::operator<=(const TimingFrameInfo& other) const {
> }
> 
> bool TimingFrameInfo::IsOutlier() const {
>-  return !IsInvalid() && (flags & TimingFrameFlags::kTriggeredBySize);
>+  return !IsInvalid() && (flags & VideoSendTiming::kTriggeredBySize);
> }
> 
> bool TimingFrameInfo::IsTimerTriggered() const {
>-  return !IsInvalid() && (flags & TimingFrameFlags::kTriggeredByTimer);
>+  return !IsInvalid() && (flags & VideoSendTiming::kTriggeredByTimer);
> }
> 
> bool TimingFrameInfo::IsInvalid() const {
>-  return flags == TimingFrameFlags::kInvalid;
>+  return flags == VideoSendTiming::kInvalid;
> }
> 
> std::string TimingFrameInfo::ToString() const {
>-  std::stringstream out;
>   if (IsInvalid()) {
>-    out << "";
>-  } else {
>-    out << rtp_timestamp << ',' << capture_time_ms << ',' << encode_start_ms
>-        << ',' << encode_finish_ms << ',' << packetization_finish_ms << ','
>-        << pacer_exit_ms << ',' << network_timestamp_ms << ','
>-        << network2_timestamp_ms << ',' << receive_start_ms << ','
>-        << receive_finish_ms << ',' << decode_start_ms << ','
>-        << decode_finish_ms << ',' << render_time_ms << ','
>-        << IsOutlier() << ',' << IsTimerTriggered();
>+    return "";
>   }
>-  return out.str();
>+
>+  char buf[1024];
>+  rtc::SimpleStringBuilder sb(buf);
>+
>+  sb << rtp_timestamp << ',' << capture_time_ms << ',' << encode_start_ms << ','
>+     << encode_finish_ms << ',' << packetization_finish_ms << ','
>+     << pacer_exit_ms << ',' << network_timestamp_ms << ','
>+     << network2_timestamp_ms << ',' << receive_start_ms << ','
>+     << receive_finish_ms << ',' << decode_start_ms << ',' << decode_finish_ms
>+     << ',' << render_time_ms << ',' << IsOutlier() << ','
>+     << IsTimerTriggered();
>+
>+  return sb.str();
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_timing.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_timing.h
>index ab8cd99136f..e787a4541f4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_timing.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video/video_timing.h
>@@ -21,20 +21,17 @@
> 
> namespace webrtc {
> 
>-enum TimingFrameFlags : uint8_t {
>-  kNotTriggered = 0,           // Timing info valid, but not to be transmitted.
>-                               // Used on send-side only.
>-  // TODO(ilnik): Delete compatibility alias.
>-  // Used to be sent over the wire, for the old protocol.
>-  kDefault = 0,                // Old name, for API compatibility.
>-  kTriggeredByTimer = 1 << 0,  // Frame marked for tracing by periodic timer.
>-  kTriggeredBySize = 1 << 1,   // Frame marked for tracing due to size.
>-  kInvalid = std::numeric_limits<uint8_t>::max()  // Invalid, ignore!
>-};
>-
> // Video timing timestamps in ms counted from capture_time_ms of a frame.
> // This structure represents data sent in video-timing RTP header extension.
> struct VideoSendTiming {
>+  enum TimingFrameFlags : uint8_t {
>+    kNotTriggered = 0,  // Timing info valid, but not to be transmitted.
>+                        // Used on send-side only.
>+    kTriggeredByTimer = 1 << 0,  // Frame marked for tracing by periodic timer.
>+    kTriggeredBySize = 1 << 1,   // Frame marked for tracing due to size.
>+    kInvalid = std::numeric_limits<uint8_t>::max()  // Invalid, ignore!
>+  };
>+
>   // Offsets of the fields in the RTP header extension, counting from the first
>   // byte after the one-byte header.
>   static constexpr uint8_t kFlagsOffset = 0;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/BUILD.gn
>index 5da89807561..e523191082b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/BUILD.gn
>@@ -15,21 +15,95 @@ if (is_android) {
> rtc_source_set("video_codecs_api") {
>   visibility = [ "*" ]
>   sources = [
>+    "sdp_video_format.cc",
>     "sdp_video_format.h",
>+    "video_codec.cc",
>+    "video_codec.h",
>+    "video_decoder.cc",
>     "video_decoder.h",
>     "video_decoder_factory.h",
>     "video_encoder.cc",
>     "video_encoder.h",
>+    "video_encoder_config.cc",
>+    "video_encoder_config.h",
>     "video_encoder_factory.h",
>   ]
> 
>   deps = [
>-    "..:optional",
>-    "..:video_frame_api",
>     "../..:webrtc_common",
>-    "../../:typedefs",
>     "../../common_video",
>     "../../rtc_base:checks",
>     "../../rtc_base:rtc_base_approved",
>+    "../video:video_bitrate_allocation",
>+    "../video:video_frame",
>+    "//third_party/abseil-cpp/absl/types:optional",
>+  ]
>+}
>+
>+rtc_static_library("builtin_video_decoder_factory") {
>+  visibility = [ "*" ]
>+  allow_poison = [
>+    "audio_codecs",  # TODO(bugs.webrtc.org/8396): Remove.
>+    "software_video_codecs",
>+  ]
>+  sources = [
>+    "builtin_video_decoder_factory.cc",
>+    "builtin_video_decoder_factory.h",
>+  ]
>+
>+  deps = [
>+    ":video_codecs_api",
>+    "../../media:rtc_internal_video_codecs",
>+    "../../rtc_base:ptr_util",
>+    "//third_party/abseil-cpp/absl/memory",
>+  ]
>+}
>+
>+rtc_static_library("builtin_video_encoder_factory") {
>+  visibility = [ "*" ]
>+  allow_poison = [
>+    "audio_codecs",  # TODO(bugs.webrtc.org/8396): Remove.
>+    "software_video_codecs",
>+  ]
>+  sources = [
>+    "builtin_video_encoder_factory.cc",
>+    "builtin_video_encoder_factory.h",
>+  ]
>+
>+  deps = [
>+    ":video_codecs_api",
>+    "../../media:rtc_internal_video_codecs",
>+    "../../media:rtc_media_base",
>+    "../../rtc_base:ptr_util",
>+    "//third_party/abseil-cpp/absl/memory",
>+  ]
>+}
>+
>+rtc_static_library("rtc_software_fallback_wrappers") {
>+  visibility = [ "*" ]
>+
>+  sources = [
>+    "video_decoder_software_fallback_wrapper.cc",
>+    "video_decoder_software_fallback_wrapper.h",
>+    "video_encoder_software_fallback_wrapper.cc",
>+    "video_encoder_software_fallback_wrapper.h",
>+  ]
>+
>+  if (!build_with_chromium && is_clang) {
>+    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>+    suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>+  }
>+
>+  deps = [
>+    ":video_codecs_api",
>+    "../../media:rtc_h264_profile_id",
>+    "../../media:rtc_media_base",
>+    "../../modules/video_coding:video_codec_interface",
>+    "../../rtc_base:checks",
>+    "../../rtc_base:rtc_base_approved",
>+    "../../rtc_base/system:fallthrough",
>+    "../../system_wrappers:field_trial_api",
>+    "../video:video_bitrate_allocation",
>+    "//third_party/abseil-cpp/absl/memory",
>   ]
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/builtin_video_decoder_factory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/builtin_video_decoder_factory.cc
>new file mode 100644
>index 00000000000..883c1ebd2e4
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/builtin_video_decoder_factory.cc
>@@ -0,0 +1,22 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/video_codecs/builtin_video_decoder_factory.h"
>+
>+#include "absl/memory/memory.h"
>+#include "media/engine/internaldecoderfactory.h"
>+
>+namespace webrtc {
>+
>+std::unique_ptr<VideoDecoderFactory> CreateBuiltinVideoDecoderFactory() {
>+  return absl::make_unique<InternalDecoderFactory>();
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/builtin_video_decoder_factory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/builtin_video_decoder_factory.h
>new file mode 100644
>index 00000000000..1f8e75c9ae7
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/builtin_video_decoder_factory.h
>@@ -0,0 +1,25 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_VIDEO_CODECS_BUILTIN_VIDEO_DECODER_FACTORY_H_
>+#define API_VIDEO_CODECS_BUILTIN_VIDEO_DECODER_FACTORY_H_
>+
>+#include <memory>
>+
>+#include "api/video_codecs/video_decoder_factory.h"
>+
>+namespace webrtc {
>+
>+// Creates a new factory that can create the built-in types of video decoders.
>+std::unique_ptr<VideoDecoderFactory> CreateBuiltinVideoDecoderFactory();
>+
>+}  // namespace webrtc
>+
>+#endif  // API_VIDEO_CODECS_BUILTIN_VIDEO_DECODER_FACTORY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/builtin_video_encoder_factory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/builtin_video_encoder_factory.cc
>new file mode 100644
>index 00000000000..ca389b99e3b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/builtin_video_encoder_factory.cc
>@@ -0,0 +1,85 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/video_codecs/builtin_video_encoder_factory.h"
>+
>+#include <vector>
>+
>+#include "absl/memory/memory.h"
>+#include "api/video_codecs/sdp_video_format.h"
>+#include "media/base/codec.h"
>+#include "media/base/mediaconstants.h"
>+#include "media/engine/internalencoderfactory.h"
>+#include "media/engine/vp8_encoder_simulcast_proxy.h"
>+
>+namespace webrtc {
>+
>+namespace {
>+
>+bool IsFormatSupported(const std::vector<SdpVideoFormat>& supported_formats,
>+                       const SdpVideoFormat& format) {
>+  for (const SdpVideoFormat& supported_format : supported_formats) {
>+    if (cricket::IsSameCodec(format.name, format.parameters,
>+                             supported_format.name,
>+                             supported_format.parameters)) {
>+      return true;
>+    }
>+  }
>+  return false;
>+}
>+
>+// This class wraps the internal factory and adds simulcast.
>+class BuiltinVideoEncoderFactory : public VideoEncoderFactory {
>+ public:
>+  BuiltinVideoEncoderFactory()
>+      : internal_encoder_factory_(new InternalEncoderFactory()) {}
>+
>+  VideoEncoderFactory::CodecInfo QueryVideoEncoder(
>+      const SdpVideoFormat& format) const override {
>+    // Format must be one of the internal formats.
>+    RTC_DCHECK(IsFormatSupported(
>+        internal_encoder_factory_->GetSupportedFormats(), format));
>+    VideoEncoderFactory::CodecInfo info;
>+    info.has_internal_source = false;
>+    info.is_hardware_accelerated = false;
>+    return info;
>+  }
>+
>+  std::unique_ptr<VideoEncoder> CreateVideoEncoder(
>+      const SdpVideoFormat& format) override {
>+    // Try creating internal encoder.
>+    std::unique_ptr<VideoEncoder> internal_encoder;
>+    if (IsFormatSupported(internal_encoder_factory_->GetSupportedFormats(),
>+                          format)) {
>+      internal_encoder =
>+          cricket::CodecNamesEq(format.name.c_str(), cricket::kVp8CodecName)
>+              ? absl::make_unique<VP8EncoderSimulcastProxy>(
>+                    internal_encoder_factory_.get(), format)
>+              : internal_encoder_factory_->CreateVideoEncoder(format);
>+    }
>+
>+    return internal_encoder;
>+  }
>+
>+  std::vector<SdpVideoFormat> GetSupportedFormats() const override {
>+    return internal_encoder_factory_->GetSupportedFormats();
>+  }
>+
>+ private:
>+  const std::unique_ptr<VideoEncoderFactory> internal_encoder_factory_;
>+};
>+
>+}  // namespace
>+
>+std::unique_ptr<VideoEncoderFactory> CreateBuiltinVideoEncoderFactory() {
>+  return absl::make_unique<BuiltinVideoEncoderFactory>();
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/builtin_video_encoder_factory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/builtin_video_encoder_factory.h
>new file mode 100644
>index 00000000000..6a6618fb327
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/builtin_video_encoder_factory.h
>@@ -0,0 +1,26 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_VIDEO_CODECS_BUILTIN_VIDEO_ENCODER_FACTORY_H_
>+#define API_VIDEO_CODECS_BUILTIN_VIDEO_ENCODER_FACTORY_H_
>+
>+#include <memory>
>+
>+#include "api/video_codecs/video_encoder_factory.h"
>+
>+namespace webrtc {
>+
>+// Creates a new factory that can create the built-in types of video encoders.
>+// The factory has simulcast support for VP8.
>+std::unique_ptr<VideoEncoderFactory> CreateBuiltinVideoEncoderFactory();
>+
>+}  // namespace webrtc
>+
>+#endif  // API_VIDEO_CODECS_BUILTIN_VIDEO_ENCODER_FACTORY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/sdp_video_format.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/sdp_video_format.cc
>new file mode 100644
>index 00000000000..909904c19c0
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/sdp_video_format.cc
>@@ -0,0 +1,32 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/video_codecs/sdp_video_format.h"
>+
>+namespace webrtc {
>+
>+SdpVideoFormat::SdpVideoFormat(const std::string& name) : name(name) {}
>+
>+SdpVideoFormat::SdpVideoFormat(const std::string& name,
>+                               const Parameters& parameters)
>+    : name(name), parameters(parameters) {}
>+
>+SdpVideoFormat::SdpVideoFormat(const SdpVideoFormat&) = default;
>+SdpVideoFormat::SdpVideoFormat(SdpVideoFormat&&) = default;
>+SdpVideoFormat& SdpVideoFormat::operator=(const SdpVideoFormat&) = default;
>+SdpVideoFormat& SdpVideoFormat::operator=(SdpVideoFormat&&) = default;
>+
>+SdpVideoFormat::~SdpVideoFormat() = default;
>+
>+bool operator==(const SdpVideoFormat& a, const SdpVideoFormat& b) {
>+  return a.name == b.name && a.parameters == b.parameters;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/sdp_video_format.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/sdp_video_format.h
>index 542353aea7e..c25b857fde7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/sdp_video_format.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/sdp_video_format.h
>@@ -21,14 +21,16 @@ namespace webrtc {
> struct SdpVideoFormat {
>   using Parameters = std::map<std::string, std::string>;
> 
>-  explicit SdpVideoFormat(const std::string& name) : name(name) {}
>-  SdpVideoFormat(const std::string& name, const Parameters& parameters)
>-      : name(name), parameters(parameters) {}
>+  explicit SdpVideoFormat(const std::string& name);
>+  SdpVideoFormat(const std::string& name, const Parameters& parameters);
>+  SdpVideoFormat(const SdpVideoFormat&);
>+  SdpVideoFormat(SdpVideoFormat&&);
>+  SdpVideoFormat& operator=(const SdpVideoFormat&);
>+  SdpVideoFormat& operator=(SdpVideoFormat&&);
> 
>-  friend bool operator==(const SdpVideoFormat& a, const SdpVideoFormat& b) {
>-    return a.name == b.name && a.parameters == b.parameters;
>-  }
>+  ~SdpVideoFormat();
> 
>+  friend bool operator==(const SdpVideoFormat& a, const SdpVideoFormat& b);
>   friend bool operator!=(const SdpVideoFormat& a, const SdpVideoFormat& b) {
>     return !(a == b);
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/test/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/test/BUILD.gn
>new file mode 100644
>index 00000000000..18943095f56
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/test/BUILD.gn
>@@ -0,0 +1,37 @@
>+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+#
>+# Use of this source code is governed by a BSD-style license
>+# that can be found in the LICENSE file in the root of the source
>+# tree. An additional intellectual property rights grant can be found
>+# in the file PATENTS.  All contributing project authors may
>+# be found in the AUTHORS file in the root of the source tree.
>+
>+import("../../../webrtc.gni")
>+
>+if (rtc_include_tests) {
>+  rtc_source_set("video_codecs_api_unittests") {
>+    testonly = true
>+    sources = [
>+      "builtin_video_encoder_factory_unittest.cc",
>+      "video_decoder_software_fallback_wrapper_unittest.cc",
>+      "video_encoder_software_fallback_wrapper_unittest.cc",
>+    ]
>+
>+    deps = [
>+      "..:builtin_video_encoder_factory",
>+      "..:rtc_software_fallback_wrappers",
>+      "..:video_codecs_api",
>+      "../../../modules/video_coding:video_codec_interface",
>+      "../../../modules/video_coding:video_coding_utility",
>+      "../../../modules/video_coding:webrtc_vp8",
>+      "../../../rtc_base:checks",
>+      "../../../rtc_base:rtc_base_tests_utils",
>+      "../../../system_wrappers:metrics_default",
>+      "../../../test:field_trial",
>+      "../../../test:test_support",
>+      "../../video:video_bitrate_allocation",
>+      "../../video:video_frame_i420",
>+      "//testing/gtest",
>+    ]
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/test/builtin_video_encoder_factory_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/test/builtin_video_encoder_factory_unittest.cc
>new file mode 100644
>index 00000000000..73957c999f0
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/test/builtin_video_encoder_factory_unittest.cc
>@@ -0,0 +1,37 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/video_codecs/builtin_video_encoder_factory.h"
>+
>+#include <memory>
>+
>+#include "api/video_codecs/sdp_video_format.h"
>+#include "test/gmock.h"
>+
>+namespace webrtc {
>+
>+TEST(BuiltinVideoEncoderFactoryTest, AnnouncesVp9AccordingToBuildFlags) {
>+  std::unique_ptr<VideoEncoderFactory> factory =
>+      CreateBuiltinVideoEncoderFactory();
>+  bool claims_vp9_support = false;
>+  for (const SdpVideoFormat& format : factory->GetSupportedFormats()) {
>+    if (format.name == "VP9") {
>+      claims_vp9_support = true;
>+      break;
>+    }
>+  }
>+#if defined(RTC_DISABLE_VP9)
>+  EXPECT_FALSE(claims_vp9_support);
>+#else
>+  EXPECT_TRUE(claims_vp9_support);
>+#endif  // defined(RTC_DISABLE_VP9)
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/videodecodersoftwarefallbackwrapper_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/test/video_decoder_software_fallback_wrapper_unittest.cc
>similarity index 62%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/videodecodersoftwarefallbackwrapper_unittest.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/test/video_decoder_software_fallback_wrapper_unittest.cc
>index 367527eff79..d2d0e2b62b2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/videodecodersoftwarefallbackwrapper_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/test/video_decoder_software_fallback_wrapper_unittest.cc
>@@ -8,7 +8,7 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "media/engine/videodecodersoftwarefallbackwrapper.h"
>+#include "api/video_codecs/video_decoder_software_fallback_wrapper.h"
> #include "api/video_codecs/video_decoder.h"
> #include "modules/video_coding/codecs/vp8/include/vp8.h"
> #include "modules/video_coding/include/video_error_codes.h"
>@@ -21,8 +21,9 @@ class VideoDecoderSoftwareFallbackWrapperTest : public ::testing::Test {
>  protected:
>   VideoDecoderSoftwareFallbackWrapperTest()
>       : fake_decoder_(new CountingFakeDecoder()),
>-        fallback_wrapper_(std::unique_ptr<VideoDecoder>(VP8Decoder::Create()),
>-                          std::unique_ptr<VideoDecoder>(fake_decoder_)) {}
>+        fallback_wrapper_(CreateVideoDecoderSoftwareFallbackWrapper(
>+            std::unique_ptr<VideoDecoder>(VP8Decoder::Create()),
>+            std::unique_ptr<VideoDecoder>(fake_decoder_))) {}
> 
>   class CountingFakeDecoder : public VideoDecoder {
>    public:
>@@ -34,7 +35,6 @@ class VideoDecoderSoftwareFallbackWrapperTest : public ::testing::Test {
> 
>     int32_t Decode(const EncodedImage& input_image,
>                    bool missing_frames,
>-                   const RTPFragmentationHeader* fragmentation,
>                    const CodecSpecificInfo* codec_specific_info,
>                    int64_t render_time_ms) override {
>       ++decode_count_;
>@@ -52,9 +52,7 @@ class VideoDecoderSoftwareFallbackWrapperTest : public ::testing::Test {
>       return WEBRTC_VIDEO_CODEC_OK;
>     }
> 
>-    const char* ImplementationName() const override {
>-      return "fake-decoder";
>-    }
>+    const char* ImplementationName() const override { return "fake-decoder"; }
> 
>     int init_decode_count_ = 0;
>     int decode_count_ = 0;
>@@ -66,17 +64,17 @@ class VideoDecoderSoftwareFallbackWrapperTest : public ::testing::Test {
>   };
>   // |fake_decoder_| is owned and released by |fallback_wrapper_|.
>   CountingFakeDecoder* fake_decoder_;
>-  VideoDecoderSoftwareFallbackWrapper fallback_wrapper_;
>+  std::unique_ptr<VideoDecoder> fallback_wrapper_;
> };
> 
> TEST_F(VideoDecoderSoftwareFallbackWrapperTest, InitializesDecoder) {
>   VideoCodec codec = {};
>-  fallback_wrapper_.InitDecode(&codec, 2);
>+  fallback_wrapper_->InitDecode(&codec, 2);
>   EXPECT_EQ(1, fake_decoder_->init_decode_count_);
> 
>   EncodedImage encoded_image;
>   encoded_image._frameType = kVideoFrameKey;
>-  fallback_wrapper_.Decode(encoded_image, false, nullptr, nullptr, -1);
>+  fallback_wrapper_->Decode(encoded_image, false, nullptr, -1);
>   EXPECT_EQ(1, fake_decoder_->init_decode_count_)
>       << "Initialized decoder should not be reinitialized.";
>   EXPECT_EQ(1, fake_decoder_->decode_count_);
>@@ -85,16 +83,15 @@ TEST_F(VideoDecoderSoftwareFallbackWrapperTest, InitializesDecoder) {
> TEST_F(VideoDecoderSoftwareFallbackWrapperTest,
>        UsesFallbackDecoderAfterAnyInitDecodeFailure) {
>   VideoCodec codec = {};
>-  fake_decoder_->init_decode_return_code_ =
>-      WEBRTC_VIDEO_CODEC_UNINITIALIZED;
>-  fallback_wrapper_.InitDecode(&codec, 2);
>+  fake_decoder_->init_decode_return_code_ = WEBRTC_VIDEO_CODEC_UNINITIALIZED;
>+  fallback_wrapper_->InitDecode(&codec, 2);
>   EXPECT_EQ(1, fake_decoder_->init_decode_count_);
> 
>   EncodedImage encoded_image;
>   encoded_image._frameType = kVideoFrameKey;
>-  fallback_wrapper_.Decode(encoded_image, false, nullptr, nullptr, -1);
>-  EXPECT_EQ(2, fake_decoder_->init_decode_count_)
>-      << "Should have attempted reinitializing the fallback decoder on "
>+  fallback_wrapper_->Decode(encoded_image, false, nullptr, -1);
>+  EXPECT_EQ(1, fake_decoder_->init_decode_count_)
>+      << "Should not have attempted reinitializing the fallback decoder on "
>          "keyframe.";
>   // Unfortunately faking a VP8 frame is hard. Rely on no Decode -> using SW
>   // decoder.
>@@ -102,62 +99,70 @@ TEST_F(VideoDecoderSoftwareFallbackWrapperTest,
>       << "Decoder used even though no InitDecode had succeeded.";
> }
> 
>-TEST_F(VideoDecoderSoftwareFallbackWrapperTest,
>-       CanRecoverFromSoftwareFallback) {
>+TEST_F(VideoDecoderSoftwareFallbackWrapperTest, IsSoftwareFallbackSticky) {
>   VideoCodec codec = {};
>-  fallback_wrapper_.InitDecode(&codec, 2);
>-  // Unfortunately faking a VP8 frame is hard. Rely on no Decode -> using SW
>-  // decoder.
>+  fallback_wrapper_->InitDecode(&codec, 2);
>+
>   fake_decoder_->decode_return_code_ = WEBRTC_VIDEO_CODEC_FALLBACK_SOFTWARE;
>   EncodedImage encoded_image;
>-  fallback_wrapper_.Decode(encoded_image, false, nullptr, nullptr, -1);
>+  fallback_wrapper_->Decode(encoded_image, false, nullptr, -1);
>   EXPECT_EQ(1, fake_decoder_->decode_count_);
> 
>-  // Fail -> fake_decoder shouldn't be used anymore.
>-  fallback_wrapper_.Decode(encoded_image, false, nullptr, nullptr, -1);
>-  EXPECT_EQ(1, fake_decoder_->decode_count_)
>-      << "Decoder used even though fallback should be active.";
>-
>-  // Should be able to recover on a keyframe.
>+  // Software fallback should be sticky, fake_decoder_ shouldn't be used.
>   encoded_image._frameType = kVideoFrameKey;
>-  fake_decoder_->decode_return_code_ = WEBRTC_VIDEO_CODEC_OK;
>-  fallback_wrapper_.Decode(encoded_image, false, nullptr, nullptr, -1);
>-  EXPECT_EQ(2, fake_decoder_->decode_count_)
>-      << "Wrapper did not try to decode a keyframe using registered decoder.";
>+  fallback_wrapper_->Decode(encoded_image, false, nullptr, -1);
>+  EXPECT_EQ(1, fake_decoder_->decode_count_)
>+      << "Decoder shouldn't be used after failure.";
> 
>-  encoded_image._frameType = kVideoFrameDelta;
>-  fallback_wrapper_.Decode(encoded_image, false, nullptr, nullptr, -1);
>-  EXPECT_EQ(3, fake_decoder_->decode_count_)
>-      << "Decoder not used on future delta frames.";
>+  // fake_decoder_ should have only been initialized once during the test.
>+  EXPECT_EQ(1, fake_decoder_->init_decode_count_);
> }
> 
> TEST_F(VideoDecoderSoftwareFallbackWrapperTest, DoesNotFallbackOnEveryError) {
>   VideoCodec codec = {};
>-  fallback_wrapper_.InitDecode(&codec, 2);
>+  fallback_wrapper_->InitDecode(&codec, 2);
>   fake_decoder_->decode_return_code_ = WEBRTC_VIDEO_CODEC_ERROR;
>   EncodedImage encoded_image;
>-  EXPECT_EQ(
>-      fake_decoder_->decode_return_code_,
>-      fallback_wrapper_.Decode(encoded_image, false, nullptr, nullptr, -1));
>+  EXPECT_EQ(fake_decoder_->decode_return_code_,
>+            fallback_wrapper_->Decode(encoded_image, false, nullptr, -1));
>   EXPECT_EQ(1, fake_decoder_->decode_count_);
> 
>-  fallback_wrapper_.Decode(encoded_image, false, nullptr, nullptr, -1);
>+  fallback_wrapper_->Decode(encoded_image, false, nullptr, -1);
>   EXPECT_EQ(2, fake_decoder_->decode_count_)
>       << "Decoder should be active even though previous decode failed.";
> }
> 
>+TEST_F(VideoDecoderSoftwareFallbackWrapperTest, UsesHwDecoderAfterReinit) {
>+  VideoCodec codec = {};
>+  fallback_wrapper_->InitDecode(&codec, 2);
>+
>+  fake_decoder_->decode_return_code_ = WEBRTC_VIDEO_CODEC_FALLBACK_SOFTWARE;
>+  EncodedImage encoded_image;
>+  fallback_wrapper_->Decode(encoded_image, false, nullptr, -1);
>+  EXPECT_EQ(1, fake_decoder_->decode_count_);
>+
>+  fallback_wrapper_->Release();
>+  fallback_wrapper_->InitDecode(&codec, 2);
>+
>+  fake_decoder_->decode_return_code_ = WEBRTC_VIDEO_CODEC_OK;
>+  fallback_wrapper_->Decode(encoded_image, false, nullptr, -1);
>+  EXPECT_EQ(2, fake_decoder_->decode_count_)
>+      << "Should not be using fallback after reinit.";
>+}
>+
> TEST_F(VideoDecoderSoftwareFallbackWrapperTest, ForwardsReleaseCall) {
>   VideoCodec codec = {};
>-  fallback_wrapper_.InitDecode(&codec, 2);
>-  fallback_wrapper_.Release();
>+  fallback_wrapper_->InitDecode(&codec, 2);
>+  fallback_wrapper_->Release();
>   EXPECT_EQ(1, fake_decoder_->release_count_);
> 
>+  fallback_wrapper_->InitDecode(&codec, 2);
>   fake_decoder_->decode_return_code_ = WEBRTC_VIDEO_CODEC_FALLBACK_SOFTWARE;
>   EncodedImage encoded_image;
>-  fallback_wrapper_.Decode(encoded_image, false, nullptr, nullptr, -1);
>-  EXPECT_EQ(1, fake_decoder_->release_count_)
>-      << "Decoder should not be released during fallback.";
>-  fallback_wrapper_.Release();
>+  fallback_wrapper_->Decode(encoded_image, false, nullptr, -1);
>+  EXPECT_EQ(2, fake_decoder_->release_count_)
>+      << "Decoder should be released during fallback.";
>+  fallback_wrapper_->Release();
>   EXPECT_EQ(2, fake_decoder_->release_count_);
> }
> 
>@@ -167,43 +172,37 @@ TEST_F(VideoDecoderSoftwareFallbackWrapperTest,
>        ForwardsRegisterDecodeCompleteCallback) {
>   class FakeDecodedImageCallback : public DecodedImageCallback {
>     int32_t Decoded(VideoFrame& decodedImage) override { return 0; }
>-    int32_t Decoded(
>-        webrtc::VideoFrame& decodedImage, int64_t decode_time_ms) override {
>+    int32_t Decoded(webrtc::VideoFrame& decodedImage,
>+                    int64_t decode_time_ms) override {
>       RTC_NOTREACHED();
>       return -1;
>     }
>     void Decoded(webrtc::VideoFrame& decodedImage,
>-                 rtc::Optional<int32_t> decode_time_ms,
>-                 rtc::Optional<uint8_t> qp) override {
>+                 absl::optional<int32_t> decode_time_ms,
>+                 absl::optional<uint8_t> qp) override {
>       RTC_NOTREACHED();
>     }
>-  } callback, callback2;
>+  } callback;
> 
>   VideoCodec codec = {};
>-  fallback_wrapper_.InitDecode(&codec, 2);
>-  fallback_wrapper_.RegisterDecodeCompleteCallback(&callback);
>+  fallback_wrapper_->InitDecode(&codec, 2);
>+  fallback_wrapper_->RegisterDecodeCompleteCallback(&callback);
>   EXPECT_EQ(&callback, fake_decoder_->decode_complete_callback_);
>-
>-  fake_decoder_->decode_return_code_ = WEBRTC_VIDEO_CODEC_FALLBACK_SOFTWARE;
>-  EncodedImage encoded_image;
>-  fallback_wrapper_.Decode(encoded_image, false, nullptr, nullptr, -1);
>-  fallback_wrapper_.RegisterDecodeCompleteCallback(&callback2);
>-  EXPECT_EQ(&callback2, fake_decoder_->decode_complete_callback_);
> }
> 
> TEST_F(VideoDecoderSoftwareFallbackWrapperTest,
>        ReportsFallbackImplementationName) {
>   VideoCodec codec = {};
>-  fallback_wrapper_.InitDecode(&codec, 2);
>+  fallback_wrapper_->InitDecode(&codec, 2);
> 
>   fake_decoder_->decode_return_code_ = WEBRTC_VIDEO_CODEC_FALLBACK_SOFTWARE;
>   EncodedImage encoded_image;
>-  fallback_wrapper_.Decode(encoded_image, false, nullptr, nullptr, -1);
>+  fallback_wrapper_->Decode(encoded_image, false, nullptr, -1);
>   // Hard coded expected value since libvpx is the software implementation name
>   // for VP8. Change accordingly if the underlying implementation does.
>   EXPECT_STREQ("libvpx (fallback from: fake-decoder)",
>-               fallback_wrapper_.ImplementationName());
>-  fallback_wrapper_.Release();
>+               fallback_wrapper_->ImplementationName());
>+  fallback_wrapper_->Release();
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/videoencodersoftwarefallbackwrapper_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/test/video_encoder_software_fallback_wrapper_unittest.cc
>similarity index 82%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/videoencodersoftwarefallbackwrapper_unittest.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/test/video_encoder_software_fallback_wrapper_unittest.cc
>index e39b02f71ce..a1a43b42027 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/videoencodersoftwarefallbackwrapper_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/test/video_encoder_software_fallback_wrapper_unittest.cc
>@@ -8,16 +8,17 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "media/engine/videoencodersoftwarefallbackwrapper.h"
>+#include "api/video_codecs/video_encoder_software_fallback_wrapper.h"
> 
> #include <utility>
> 
> #include "api/video/i420_buffer.h"
>+#include "api/video/video_bitrate_allocation.h"
> #include "modules/video_coding/codecs/vp8/include/vp8.h"
>-#include "modules/video_coding/codecs/vp8/simulcast_rate_allocator.h"
> #include "modules/video_coding/codecs/vp8/temporal_layers.h"
> #include "modules/video_coding/include/video_codec_interface.h"
> #include "modules/video_coding/include/video_error_codes.h"
>+#include "modules/video_coding/utility/simulcast_rate_allocator.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/fakeclock.h"
> #include "test/field_trial.h"
>@@ -43,8 +44,9 @@ class VideoEncoderSoftwareFallbackWrapperTest : public ::testing::Test {
>       const std::string& field_trials)
>       : override_field_trials_(field_trials),
>         fake_encoder_(new CountingFakeEncoder()),
>-        fallback_wrapper_(std::unique_ptr<VideoEncoder>(VP8Encoder::Create()),
>-                          std::unique_ptr<VideoEncoder>(fake_encoder_)) {}
>+        fallback_wrapper_(CreateVideoEncoderSoftwareFallbackWrapper(
>+            std::unique_ptr<VideoEncoder>(VP8Encoder::Create()),
>+            std::unique_ptr<VideoEncoder>(fake_encoder_))) {}
> 
>   class CountingFakeEncoder : public VideoEncoder {
>    public:
>@@ -84,7 +86,7 @@ class VideoEncoderSoftwareFallbackWrapperTest : public ::testing::Test {
>       return WEBRTC_VIDEO_CODEC_OK;
>     }
> 
>-    int32_t SetRateAllocation(const BitrateAllocation& bitrate_allocation,
>+    int32_t SetRateAllocation(const VideoBitrateAllocation& bitrate_allocation,
>                               uint32_t framerate) override {
>       ++set_rates_count_;
>       return WEBRTC_VIDEO_CODEC_OK;
>@@ -95,12 +97,10 @@ class VideoEncoderSoftwareFallbackWrapperTest : public ::testing::Test {
>       return supports_native_handle_;
>     }
> 
>-    const char* ImplementationName() const override {
>-      return "fake-encoder";
>-    }
>+    const char* ImplementationName() const override { return "fake-encoder"; }
> 
>     VideoEncoder::ScalingSettings GetScalingSettings() const override {
>-      return VideoEncoder::ScalingSettings(true, kLowThreshold, kHighThreshold);
>+      return VideoEncoder::ScalingSettings(kLowThreshold, kHighThreshold);
>     }
> 
>     int init_encode_count_ = 0;
>@@ -141,7 +141,7 @@ class VideoEncoderSoftwareFallbackWrapperTest : public ::testing::Test {
>   FakeEncodedImageCallback callback_;
>   // |fake_encoder_| is owned and released by |fallback_wrapper_|.
>   CountingFakeEncoder* fake_encoder_;
>-  VideoEncoderSoftwareFallbackWrapper fallback_wrapper_;
>+  std::unique_ptr<VideoEncoder> fallback_wrapper_;
>   VideoCodec codec_ = {};
>   std::unique_ptr<VideoFrame> frame_;
>   std::unique_ptr<SimulcastRateAllocator> rate_allocator_;
>@@ -157,12 +157,13 @@ void VideoEncoderSoftwareFallbackWrapperTest::EncodeFrame(int expected_ret) {
>   I420Buffer::SetBlack(buffer);
>   std::vector<FrameType> types(1, kVideoFrameKey);
> 
>-  frame_.reset(new VideoFrame(buffer, 0, 0, webrtc::kVideoRotation_0));
>-  EXPECT_EQ(expected_ret, fallback_wrapper_.Encode(*frame_, nullptr, &types));
>+  frame_.reset(
>+      new VideoFrame(buffer, webrtc::kVideoRotation_0, 0 /* timestamp_us */));
>+  EXPECT_EQ(expected_ret, fallback_wrapper_->Encode(*frame_, nullptr, &types));
> }
> 
> void VideoEncoderSoftwareFallbackWrapperTest::UtilizeFallbackEncoder() {
>-  fallback_wrapper_.RegisterEncodeCompleteCallback(&callback_);
>+  fallback_wrapper_->RegisterEncodeCompleteCallback(&callback_);
>   EXPECT_EQ(&callback_, fake_encoder_->encode_complete_callback_);
> 
>   // Register with failing fake encoder. Should succeed with VP8 fallback.
>@@ -171,18 +172,14 @@ void VideoEncoderSoftwareFallbackWrapperTest::UtilizeFallbackEncoder() {
>   codec_.width = kWidth;
>   codec_.height = kHeight;
>   codec_.VP8()->numberOfTemporalLayers = 1;
>-  std::unique_ptr<TemporalLayersFactory> tl_factory(
>-      new TemporalLayersFactory());
>-  codec_.VP8()->tl_factory = tl_factory.get();
>-  rate_allocator_.reset(
>-      new SimulcastRateAllocator(codec_, std::move(tl_factory)));
>+  rate_allocator_.reset(new SimulcastRateAllocator(codec_));
> 
>   fake_encoder_->init_encode_return_code_ = WEBRTC_VIDEO_CODEC_ERROR;
>   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>-            fallback_wrapper_.InitEncode(&codec_, kNumCores, kMaxPayloadSize));
>+            fallback_wrapper_->InitEncode(&codec_, kNumCores, kMaxPayloadSize));
>   EXPECT_EQ(
>       WEBRTC_VIDEO_CODEC_OK,
>-      fallback_wrapper_.SetRateAllocation(
>+      fallback_wrapper_->SetRateAllocation(
>           rate_allocator_->GetAllocation(300000, kFramerate), kFramerate));
> 
>   int callback_count = callback_.callback_count_;
>@@ -193,21 +190,17 @@ void VideoEncoderSoftwareFallbackWrapperTest::UtilizeFallbackEncoder() {
> }
> 
> void VideoEncoderSoftwareFallbackWrapperTest::FallbackFromEncodeRequest() {
>-  fallback_wrapper_.RegisterEncodeCompleteCallback(&callback_);
>+  fallback_wrapper_->RegisterEncodeCompleteCallback(&callback_);
>   codec_.codecType = kVideoCodecVP8;
>   codec_.maxFramerate = kFramerate;
>   codec_.width = kWidth;
>   codec_.height = kHeight;
>   codec_.VP8()->numberOfTemporalLayers = 1;
>-  std::unique_ptr<TemporalLayersFactory> tl_factory(
>-      new TemporalLayersFactory());
>-  codec_.VP8()->tl_factory = tl_factory.get();
>-  rate_allocator_.reset(
>-      new SimulcastRateAllocator(codec_, std::move(tl_factory)));
>-  fallback_wrapper_.InitEncode(&codec_, 2, kMaxPayloadSize);
>+  rate_allocator_.reset(new SimulcastRateAllocator(codec_));
>+  fallback_wrapper_->InitEncode(&codec_, 2, kMaxPayloadSize);
>   EXPECT_EQ(
>       WEBRTC_VIDEO_CODEC_OK,
>-      fallback_wrapper_.SetRateAllocation(
>+      fallback_wrapper_->SetRateAllocation(
>           rate_allocator_->GetAllocation(300000, kFramerate), kFramerate));
>   EXPECT_EQ(1, fake_encoder_->init_encode_count_);
> 
>@@ -223,7 +216,7 @@ void VideoEncoderSoftwareFallbackWrapperTest::FallbackFromEncodeRequest() {
> 
> TEST_F(VideoEncoderSoftwareFallbackWrapperTest, InitializesEncoder) {
>   VideoCodec codec = {};
>-  fallback_wrapper_.InitEncode(&codec, 2, kMaxPayloadSize);
>+  fallback_wrapper_->InitEncode(&codec, 2, kMaxPayloadSize);
>   EXPECT_EQ(1, fake_encoder_->init_encode_count_);
> }
> 
>@@ -237,7 +230,7 @@ TEST_F(VideoEncoderSoftwareFallbackWrapperTest, EncodeRequestsFallback) {
> 
> TEST_F(VideoEncoderSoftwareFallbackWrapperTest, CanUtilizeFallbackEncoder) {
>   UtilizeFallbackEncoder();
>-  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, fallback_wrapper_.Release());
>+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, fallback_wrapper_->Release());
> }
> 
> TEST_F(VideoEncoderSoftwareFallbackWrapperTest,
>@@ -245,7 +238,7 @@ TEST_F(VideoEncoderSoftwareFallbackWrapperTest,
>   EXPECT_EQ(0, fake_encoder_->release_count_);
>   UtilizeFallbackEncoder();
>   EXPECT_EQ(1, fake_encoder_->release_count_);
>-  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, fallback_wrapper_.Release());
>+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, fallback_wrapper_->Release());
>   // No extra release when the fallback is released.
>   EXPECT_EQ(1, fake_encoder_->release_count_);
> }
>@@ -257,7 +250,7 @@ TEST_F(VideoEncoderSoftwareFallbackWrapperTest,
>   EncodeFrame();
>   EXPECT_EQ(encode_count, fake_encoder_->encode_count_);
> 
>-  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, fallback_wrapper_.Release());
>+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, fallback_wrapper_->Release());
> }
> 
> TEST_F(VideoEncoderSoftwareFallbackWrapperTest,
>@@ -266,55 +259,55 @@ TEST_F(VideoEncoderSoftwareFallbackWrapperTest,
>   // Registering an encode-complete callback should still work when fallback
>   // encoder is being used.
>   FakeEncodedImageCallback callback2;
>-  fallback_wrapper_.RegisterEncodeCompleteCallback(&callback2);
>+  fallback_wrapper_->RegisterEncodeCompleteCallback(&callback2);
>   EXPECT_EQ(&callback2, fake_encoder_->encode_complete_callback_);
> 
>   // Encoding a frame using the fallback should arrive at the new callback.
>   std::vector<FrameType> types(1, kVideoFrameKey);
>   frame_->set_timestamp(frame_->timestamp() + 1000);
>   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>-            fallback_wrapper_.Encode(*frame_, nullptr, &types));
>+            fallback_wrapper_->Encode(*frame_, nullptr, &types));
> 
>-  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, fallback_wrapper_.Release());
>+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, fallback_wrapper_->Release());
> }
> 
> TEST_F(VideoEncoderSoftwareFallbackWrapperTest,
>        SetChannelParametersForwardedDuringFallback) {
>   UtilizeFallbackEncoder();
>   EXPECT_EQ(0, fake_encoder_->set_channel_parameters_count_);
>-  fallback_wrapper_.SetChannelParameters(1, 1);
>+  fallback_wrapper_->SetChannelParameters(1, 1);
>   EXPECT_EQ(1, fake_encoder_->set_channel_parameters_count_);
>-  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, fallback_wrapper_.Release());
>+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, fallback_wrapper_->Release());
> }
> 
> TEST_F(VideoEncoderSoftwareFallbackWrapperTest,
>        SetRatesForwardedDuringFallback) {
>   UtilizeFallbackEncoder();
>   EXPECT_EQ(1, fake_encoder_->set_rates_count_);
>-  fallback_wrapper_.SetRateAllocation(BitrateAllocation(), 1);
>+  fallback_wrapper_->SetRateAllocation(VideoBitrateAllocation(), 1);
>   EXPECT_EQ(2, fake_encoder_->set_rates_count_);
>-  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, fallback_wrapper_.Release());
>+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, fallback_wrapper_->Release());
> }
> 
> TEST_F(VideoEncoderSoftwareFallbackWrapperTest,
>        SupportsNativeHandleForwardedWithoutFallback) {
>-  fallback_wrapper_.SupportsNativeHandle();
>+  fallback_wrapper_->SupportsNativeHandle();
>   EXPECT_EQ(1, fake_encoder_->supports_native_handle_count_);
> }
> 
> TEST_F(VideoEncoderSoftwareFallbackWrapperTest,
>        SupportsNativeHandleNotForwardedDuringFallback) {
>   UtilizeFallbackEncoder();
>-  fallback_wrapper_.SupportsNativeHandle();
>+  fallback_wrapper_->SupportsNativeHandle();
>   EXPECT_EQ(0, fake_encoder_->supports_native_handle_count_);
>-  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, fallback_wrapper_.Release());
>+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, fallback_wrapper_->Release());
> }
> 
> TEST_F(VideoEncoderSoftwareFallbackWrapperTest, ReportsImplementationName) {
>   codec_.width = kWidth;
>   codec_.height = kHeight;
>-  fallback_wrapper_.RegisterEncodeCompleteCallback(&callback_);
>-  fallback_wrapper_.InitEncode(&codec_, kNumCores, kMaxPayloadSize);
>+  fallback_wrapper_->RegisterEncodeCompleteCallback(&callback_);
>+  fallback_wrapper_->InitEncode(&codec_, kNumCores, kMaxPayloadSize);
>   EncodeFrame();
>   CheckLastEncoderName("fake-encoder");
> }
>@@ -347,13 +340,11 @@ class ForcedFallbackTest : public VideoEncoderSoftwareFallbackWrapperTest {
>   }
> 
>   void TearDown() override {
>-    EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, fallback_wrapper_.Release());
>+    EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, fallback_wrapper_->Release());
>   }
> 
>   void ConfigureVp8Codec() {
>-    fallback_wrapper_.RegisterEncodeCompleteCallback(&callback_);
>-    std::unique_ptr<TemporalLayersFactory> tl_factory(
>-        new TemporalLayersFactory());
>+    fallback_wrapper_->RegisterEncodeCompleteCallback(&callback_);
>     codec_.codecType = kVideoCodecVP8;
>     codec_.maxFramerate = kFramerate;
>     codec_.width = kWidth;
>@@ -361,21 +352,19 @@ class ForcedFallbackTest : public VideoEncoderSoftwareFallbackWrapperTest {
>     codec_.VP8()->numberOfTemporalLayers = 1;
>     codec_.VP8()->automaticResizeOn = true;
>     codec_.VP8()->frameDroppingOn = true;
>-    codec_.VP8()->tl_factory = tl_factory.get();
>-    rate_allocator_.reset(
>-        new SimulcastRateAllocator(codec_, std::move(tl_factory)));
>+    rate_allocator_.reset(new SimulcastRateAllocator(codec_));
>   }
> 
>   void InitEncode(int width, int height) {
>     codec_.width = width;
>     codec_.height = height;
>-    EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, fallback_wrapper_.InitEncode(
>+    EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, fallback_wrapper_->InitEncode(
>                                          &codec_, kNumCores, kMaxPayloadSize));
>     SetRateAllocation(kBitrateKbps);
>   }
> 
>   void SetRateAllocation(uint32_t bitrate_kbps) {
>-    EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, fallback_wrapper_.SetRateAllocation(
>+    EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, fallback_wrapper_->SetRateAllocation(
>                                          rate_allocator_->GetAllocation(
>                                              bitrate_kbps * 1000, kFramerate),
>                                          kFramerate));
>@@ -500,8 +489,8 @@ TEST_F(ForcedFallbackTestDisabled, GetScaleSettings) {
>   EncodeFrameAndVerifyLastName("fake-encoder");
> 
>   // Default min pixels per frame should be used.
>-  const auto settings = fallback_wrapper_.GetScalingSettings();
>-  EXPECT_TRUE(settings.enabled);
>+  const auto settings = fallback_wrapper_->GetScalingSettings();
>+  EXPECT_TRUE(settings.thresholds.has_value());
>   EXPECT_EQ(kDefaultMinPixelsPerFrame, settings.min_pixels_per_frame);
> }
> 
>@@ -511,8 +500,7 @@ TEST_F(ForcedFallbackTestEnabled, GetScaleSettingsWithNoFallback) {
>   EncodeFrameAndVerifyLastName("fake-encoder");
> 
>   // Configured min pixels per frame should be used.
>-  const auto settings = fallback_wrapper_.GetScalingSettings();
>-  EXPECT_TRUE(settings.enabled);
>+  const auto settings = fallback_wrapper_->GetScalingSettings();
>   EXPECT_EQ(kMinPixelsPerFrame, settings.min_pixels_per_frame);
>   ASSERT_TRUE(settings.thresholds);
>   EXPECT_EQ(kLowThreshold, settings.thresholds->low);
>@@ -525,8 +513,8 @@ TEST_F(ForcedFallbackTestEnabled, GetScaleSettingsWithFallback) {
>   EncodeFrameAndVerifyLastName("libvpx");
> 
>   // Configured min pixels per frame should be used.
>-  const auto settings = fallback_wrapper_.GetScalingSettings();
>-  EXPECT_TRUE(settings.enabled);
>+  const auto settings = fallback_wrapper_->GetScalingSettings();
>+  EXPECT_TRUE(settings.thresholds.has_value());
>   EXPECT_EQ(kMinPixelsPerFrame, settings.min_pixels_per_frame);
> }
> 
>@@ -537,8 +525,8 @@ TEST_F(ForcedFallbackTestEnabled, ScalingDisabledIfResizeOff) {
>   EncodeFrameAndVerifyLastName("libvpx");
> 
>   // Should be disabled for automatic resize off.
>-  const auto settings = fallback_wrapper_.GetScalingSettings();
>-  EXPECT_FALSE(settings.enabled);
>+  const auto settings = fallback_wrapper_->GetScalingSettings();
>+  EXPECT_FALSE(settings.thresholds.has_value());
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_codec.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_codec.cc
>new file mode 100644
>index 00000000000..d36535b3100
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_codec.cc
>@@ -0,0 +1,155 @@
>+/*
>+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/video_codecs/video_codec.h"
>+
>+#include <string.h>
>+#include <algorithm>
>+#include <limits>
>+#include <string>
>+#include <type_traits>
>+
>+#include "rtc_base/checks.h"
>+#include "rtc_base/strings/string_builder.h"
>+#include "rtc_base/stringutils.h"
>+
>+namespace webrtc {
>+
>+bool VideoCodecVP8::operator==(const VideoCodecVP8& other) const {
>+  return (complexity == other.complexity &&
>+          numberOfTemporalLayers == other.numberOfTemporalLayers &&
>+          denoisingOn == other.denoisingOn &&
>+          automaticResizeOn == other.automaticResizeOn &&
>+          frameDroppingOn == other.frameDroppingOn &&
>+          keyFrameInterval == other.keyFrameInterval);
>+}
>+
>+bool VideoCodecVP9::operator==(const VideoCodecVP9& other) const {
>+  return (complexity == other.complexity &&
>+          numberOfTemporalLayers == other.numberOfTemporalLayers &&
>+          denoisingOn == other.denoisingOn &&
>+          frameDroppingOn == other.frameDroppingOn &&
>+          keyFrameInterval == other.keyFrameInterval &&
>+          adaptiveQpMode == other.adaptiveQpMode &&
>+          automaticResizeOn == other.automaticResizeOn &&
>+          numberOfSpatialLayers == other.numberOfSpatialLayers &&
>+          flexibleMode == other.flexibleMode);
>+}
>+
>+bool VideoCodecH264::operator==(const VideoCodecH264& other) const {
>+  return (frameDroppingOn == other.frameDroppingOn &&
>+          keyFrameInterval == other.keyFrameInterval &&
>+          spsLen == other.spsLen && ppsLen == other.ppsLen &&
>+          profile == other.profile &&
>+          (spsLen == 0 || memcmp(spsData, other.spsData, spsLen) == 0) &&
>+          (ppsLen == 0 || memcmp(ppsData, other.ppsData, ppsLen) == 0));
>+}
>+
>+bool SpatialLayer::operator==(const SpatialLayer& other) const {
>+  return (width == other.width && height == other.height &&
>+          numberOfTemporalLayers == other.numberOfTemporalLayers &&
>+          maxBitrate == other.maxBitrate &&
>+          targetBitrate == other.targetBitrate &&
>+          minBitrate == other.minBitrate && qpMax == other.qpMax &&
>+          active == other.active);
>+}
>+
>+VideoCodec::VideoCodec()
>+    : codecType(kVideoCodecGeneric),
>+      plType(0),
>+      width(0),
>+      height(0),
>+      startBitrate(0),
>+      maxBitrate(0),
>+      minBitrate(0),
>+      targetBitrate(0),
>+      maxFramerate(0),
>+      active(true),
>+      qpMax(0),
>+      numberOfSimulcastStreams(0),
>+      simulcastStream(),
>+      spatialLayers(),
>+      mode(VideoCodecMode::kRealtimeVideo),
>+      expect_encode_from_texture(false),
>+      timing_frame_thresholds({0, 0}),
>+      codec_specific_() {}
>+
>+VideoCodecVP8* VideoCodec::VP8() {
>+  RTC_DCHECK_EQ(codecType, kVideoCodecVP8);
>+  return &codec_specific_.VP8;
>+}
>+
>+const VideoCodecVP8& VideoCodec::VP8() const {
>+  RTC_DCHECK_EQ(codecType, kVideoCodecVP8);
>+  return codec_specific_.VP8;
>+}
>+
>+VideoCodecVP9* VideoCodec::VP9() {
>+  RTC_DCHECK_EQ(codecType, kVideoCodecVP9);
>+  return &codec_specific_.VP9;
>+}
>+
>+const VideoCodecVP9& VideoCodec::VP9() const {
>+  RTC_DCHECK_EQ(codecType, kVideoCodecVP9);
>+  return codec_specific_.VP9;
>+}
>+
>+VideoCodecH264* VideoCodec::H264() {
>+  RTC_DCHECK_EQ(codecType, kVideoCodecH264);
>+  return &codec_specific_.H264;
>+}
>+
>+const VideoCodecH264& VideoCodec::H264() const {
>+  RTC_DCHECK_EQ(codecType, kVideoCodecH264);
>+  return codec_specific_.H264;
>+}
>+
>+static const char* kPayloadNameVp8 = "VP8";
>+static const char* kPayloadNameVp9 = "VP9";
>+static const char* kPayloadNameH264 = "H264";
>+static const char* kPayloadNameI420 = "I420";
>+static const char* kPayloadNameGeneric = "Generic";
>+static const char* kPayloadNameMultiplex = "Multiplex";
>+
>+static bool CodecNamesEq(const char* name1, const char* name2) {
>+  return _stricmp(name1, name2) == 0;
>+}
>+
>+const char* CodecTypeToPayloadString(VideoCodecType type) {
>+  switch (type) {
>+    case kVideoCodecVP8:
>+      return kPayloadNameVp8;
>+    case kVideoCodecVP9:
>+      return kPayloadNameVp9;
>+    case kVideoCodecH264:
>+      return kPayloadNameH264;
>+    case kVideoCodecI420:
>+      return kPayloadNameI420;
>+    // Other codecs default to generic.
>+    default:
>+      return kPayloadNameGeneric;
>+  }
>+}
>+
>+VideoCodecType PayloadStringToCodecType(const std::string& name) {
>+  if (CodecNamesEq(name.c_str(), kPayloadNameVp8))
>+    return kVideoCodecVP8;
>+  if (CodecNamesEq(name.c_str(), kPayloadNameVp9))
>+    return kVideoCodecVP9;
>+  if (CodecNamesEq(name.c_str(), kPayloadNameH264))
>+    return kVideoCodecH264;
>+  if (CodecNamesEq(name.c_str(), kPayloadNameI420))
>+    return kVideoCodecI420;
>+  if (CodecNamesEq(name.c_str(), kPayloadNameMultiplex))
>+    return kVideoCodecMultiplex;
>+  return kVideoCodecGeneric;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_codec.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_codec.h
>new file mode 100644
>index 00000000000..5b4ebb25c07
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_codec.h
>@@ -0,0 +1,166 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_VIDEO_CODECS_VIDEO_CODEC_H_
>+#define API_VIDEO_CODECS_VIDEO_CODEC_H_
>+
>+#include <string>
>+
>+#include "common_types.h"  // NOLINT(build/include)
>+
>+namespace webrtc {
>+
>+// The VideoCodec class represents an old defacto-apis, which we're migrating
>+// away from slowly.
>+
>+// Video codec
>+enum class VideoCodecComplexity {
>+  kComplexityNormal = 0,
>+  kComplexityHigh = 1,
>+  kComplexityHigher = 2,
>+  kComplexityMax = 3
>+};
>+
>+// VP8 specific
>+struct VideoCodecVP8 {
>+  bool operator==(const VideoCodecVP8& other) const;
>+  bool operator!=(const VideoCodecVP8& other) const {
>+    return !(*this == other);
>+  }
>+  VideoCodecComplexity complexity;
>+  unsigned char numberOfTemporalLayers;
>+  bool denoisingOn;
>+  bool automaticResizeOn;
>+  bool frameDroppingOn;
>+  int keyFrameInterval;
>+};
>+
>+enum class InterLayerPredMode {
>+  kOn,       // Allow inter-layer prediction for all frames.
>+             // Frame of low spatial layer can be used for
>+             // prediction of next spatial layer frame.
>+  kOff,      // Encoder produces independent spatial layers.
>+  kOnKeyPic  // Allow inter-layer prediction only for frames
>+             // within key picture.
>+};
>+
>+// VP9 specific.
>+struct VideoCodecVP9 {
>+  bool operator==(const VideoCodecVP9& other) const;
>+  bool operator!=(const VideoCodecVP9& other) const {
>+    return !(*this == other);
>+  }
>+  VideoCodecComplexity complexity;
>+  unsigned char numberOfTemporalLayers;
>+  bool denoisingOn;
>+  bool frameDroppingOn;
>+  int keyFrameInterval;
>+  bool adaptiveQpMode;
>+  bool automaticResizeOn;
>+  unsigned char numberOfSpatialLayers;
>+  bool flexibleMode;
>+  InterLayerPredMode interLayerPred;
>+};
>+
>+// H264 specific.
>+struct VideoCodecH264 {
>+  bool operator==(const VideoCodecH264& other) const;
>+  bool operator!=(const VideoCodecH264& other) const {
>+    return !(*this == other);
>+  }
>+  bool frameDroppingOn;
>+  int keyFrameInterval;
>+  // These are NULL/0 if not externally negotiated.
>+  const uint8_t* spsData;
>+  size_t spsLen;
>+  const uint8_t* ppsData;
>+  size_t ppsLen;
>+  H264::Profile profile;
>+};
>+
>+// Translates from name of codec to codec type and vice versa.
>+const char* CodecTypeToPayloadString(VideoCodecType type);
>+VideoCodecType PayloadStringToCodecType(const std::string& name);
>+
>+union VideoCodecUnion {
>+  VideoCodecVP8 VP8;
>+  VideoCodecVP9 VP9;
>+  VideoCodecH264 H264;
>+};
>+
>+enum class VideoCodecMode { kRealtimeVideo, kScreensharing };
>+
>+// Common video codec properties
>+class VideoCodec {
>+ public:
>+  VideoCodec();
>+
>+  // Public variables. TODO(hta): Make them private with accessors.
>+  VideoCodecType codecType;
>+  unsigned char plType;
>+
>+  // TODO(nisse): Change to int, for consistency.
>+  uint16_t width;
>+  uint16_t height;
>+
>+  unsigned int startBitrate;   // kilobits/sec.
>+  unsigned int maxBitrate;     // kilobits/sec.
>+  unsigned int minBitrate;     // kilobits/sec.
>+  unsigned int targetBitrate;  // kilobits/sec.
>+
>+  uint32_t maxFramerate;
>+
>+  // This enables/disables encoding and sending when there aren't multiple
>+  // simulcast streams,by allocating 0 bitrate if inactive.
>+  bool active;
>+
>+  unsigned int qpMax;
>+  unsigned char numberOfSimulcastStreams;
>+  SimulcastStream simulcastStream[kMaxSimulcastStreams];
>+  SpatialLayer spatialLayers[kMaxSpatialLayers];
>+
>+  VideoCodecMode mode;
>+  bool expect_encode_from_texture;
>+
>+  // Timing frames configuration. There is delay of delay_ms between two
>+  // consequent timing frames, excluding outliers. Frame is always made a
>+  // timing frame if it's at least outlier_ratio in percent of "ideal" average
>+  // frame given bitrate and framerate, i.e. if it's bigger than
>+  // |outlier_ratio / 100.0 * bitrate_bps / fps| in bits. This way, timing
>+  // frames will not be sent too often usually. Yet large frames will always
>+  // have timing information for debug purposes because they are more likely to
>+  // cause extra delays.
>+  struct TimingFrameTriggerThresholds {
>+    int64_t delay_ms;
>+    uint16_t outlier_ratio_percent;
>+  } timing_frame_thresholds;
>+
>+  bool operator==(const VideoCodec& other) const = delete;
>+  bool operator!=(const VideoCodec& other) const = delete;
>+
>+  // Accessors for codec specific information.
>+  // There is a const version of each that returns a reference,
>+  // and a non-const version that returns a pointer, in order
>+  // to allow modification of the parameters.
>+  VideoCodecVP8* VP8();
>+  const VideoCodecVP8& VP8() const;
>+  VideoCodecVP9* VP9();
>+  const VideoCodecVP9& VP9() const;
>+  VideoCodecH264* H264();
>+  const VideoCodecH264& H264() const;
>+
>+ private:
>+  // TODO(hta): Consider replacing the union with a pointer type.
>+  // This will allow removing the VideoCodec* types from this file.
>+  VideoCodecUnion codec_specific_;
>+};
>+
>+}  // namespace webrtc
>+#endif  // API_VIDEO_CODECS_VIDEO_CODEC_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_decoder.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_decoder.cc
>new file mode 100644
>index 00000000000..b5fff32e5fe
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_decoder.cc
>@@ -0,0 +1,44 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/video_codecs/video_decoder.h"
>+
>+namespace webrtc {
>+
>+int32_t DecodedImageCallback::Decoded(VideoFrame& decodedImage,
>+                                      int64_t decode_time_ms) {
>+  // The default implementation ignores custom decode time value.
>+  return Decoded(decodedImage);
>+}
>+
>+void DecodedImageCallback::Decoded(VideoFrame& decodedImage,
>+                                   absl::optional<int32_t> decode_time_ms,
>+                                   absl::optional<uint8_t> qp) {
>+  Decoded(decodedImage, decode_time_ms.value_or(-1));
>+}
>+
>+int32_t DecodedImageCallback::ReceivedDecodedReferenceFrame(
>+    const uint64_t pictureId) {
>+  return -1;
>+}
>+
>+int32_t DecodedImageCallback::ReceivedDecodedFrame(const uint64_t pictureId) {
>+  return -1;
>+}
>+
>+bool VideoDecoder::PrefersLateDecoding() const {
>+  return true;
>+}
>+
>+const char* VideoDecoder::ImplementationName() const {
>+  return "unknown";
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_decoder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_decoder.h
>index 7d3a7f59781..7995fccb056 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_decoder.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_decoder.h
>@@ -16,13 +16,11 @@
> #include <vector>
> 
> #include "api/video/video_frame.h"
>-#include "common_types.h"  // NOLINT(build/include)
>+#include "api/video_codecs/video_codec.h"
> #include "common_video/include/video_frame.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>-class RTPFragmentationHeader;
> // TODO(pbos): Expose these through a public (root) header or change these APIs.
> struct CodecSpecificInfo;
> class VideoCodec;
>@@ -36,24 +34,17 @@ class DecodedImageCallback {
>   // decode time excluding waiting time for any previous pending frame to
>   // return. This is necessary for breaking positive feedback in the delay
>   // estimation when the decoder has a single output buffer.
>-  virtual int32_t Decoded(VideoFrame& decodedImage, int64_t /* decode_time_ms */) {
>-    // The default implementation ignores custom decode time value.
>-    return Decoded(decodedImage);
>-  }
>+  virtual int32_t Decoded(VideoFrame& decodedImage, int64_t decode_time_ms);
>+
>   // TODO(sakal): Remove other implementations when upstream projects have been
>   // updated.
>   virtual void Decoded(VideoFrame& decodedImage,
>-                       rtc::Optional<int32_t> decode_time_ms,
>-                       rtc::Optional<uint8_t> /* qp */) {
>-    Decoded(decodedImage,
>-            decode_time_ms ? static_cast<int32_t>(*decode_time_ms) : -1);
>-  }
>+                       absl::optional<int32_t> decode_time_ms,
>+                       absl::optional<uint8_t> qp);
> 
>-  virtual int32_t ReceivedDecodedReferenceFrame(const uint64_t /* pictureId */) {
>-    return -1;
>-  }
>+  virtual int32_t ReceivedDecodedReferenceFrame(const uint64_t pictureId);
> 
>-  virtual int32_t ReceivedDecodedFrame(const uint64_t /* pictureId */) { return -1; }
>+  virtual int32_t ReceivedDecodedFrame(const uint64_t pictureId);
> };
> 
> class VideoDecoder {
>@@ -65,9 +56,8 @@ class VideoDecoder {
> 
>   virtual int32_t Decode(const EncodedImage& input_image,
>                          bool missing_frames,
>-                         const RTPFragmentationHeader* fragmentation,
>-                         const CodecSpecificInfo* codec_specific_info = NULL,
>-                         int64_t render_time_ms = -1) = 0;
>+                         const CodecSpecificInfo* codec_specific_info,
>+                         int64_t render_time_ms) = 0;
> 
>   virtual int32_t RegisterDecodeCompleteCallback(
>       DecodedImageCallback* callback) = 0;
>@@ -77,9 +67,9 @@ class VideoDecoder {
>   // Returns true if the decoder prefer to decode frames late.
>   // That is, it can not decode infinite number of frames before the decoded
>   // frame is consumed.
>-  virtual bool PrefersLateDecoding() const { return true; }
>+  virtual bool PrefersLateDecoding() const;
> 
>-  virtual const char* ImplementationName() const { return "unknown"; }
>+  virtual const char* ImplementationName() const;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_decoder_software_fallback_wrapper.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_decoder_software_fallback_wrapper.cc
>new file mode 100644
>index 00000000000..1666bc32773
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_decoder_software_fallback_wrapper.cc
>@@ -0,0 +1,224 @@
>+/*
>+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/video_codecs/video_decoder_software_fallback_wrapper.h"
>+
>+#include <string>
>+#include <utility>
>+
>+#include "absl/memory/memory.h"
>+#include "modules/video_coding/include/video_error_codes.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/logging.h"
>+#include "rtc_base/system/fallthrough.h"
>+#include "rtc_base/trace_event.h"
>+
>+namespace webrtc {
>+
>+namespace {
>+
>+class VideoDecoderSoftwareFallbackWrapper final : public VideoDecoder {
>+ public:
>+  VideoDecoderSoftwareFallbackWrapper(
>+      std::unique_ptr<VideoDecoder> sw_fallback_decoder,
>+      std::unique_ptr<VideoDecoder> hw_decoder);
>+  ~VideoDecoderSoftwareFallbackWrapper() override;
>+
>+  int32_t InitDecode(const VideoCodec* codec_settings,
>+                     int32_t number_of_cores) override;
>+
>+  int32_t Decode(const EncodedImage& input_image,
>+                 bool missing_frames,
>+                 const CodecSpecificInfo* codec_specific_info,
>+                 int64_t render_time_ms) override;
>+
>+  int32_t RegisterDecodeCompleteCallback(
>+      DecodedImageCallback* callback) override;
>+
>+  int32_t Release() override;
>+  bool PrefersLateDecoding() const override;
>+
>+  const char* ImplementationName() const override;
>+
>+ private:
>+  bool InitFallbackDecoder();
>+  int32_t InitHwDecoder();
>+
>+  VideoDecoder& active_decoder() const;
>+
>+  // Determines if we are trying to use the HW or SW decoder.
>+  enum class DecoderType {
>+    kNone,
>+    kHardware,
>+    kFallback,
>+  } decoder_type_;
>+  std::unique_ptr<VideoDecoder> hw_decoder_;
>+
>+  VideoCodec codec_settings_;
>+  int32_t number_of_cores_;
>+  const std::unique_ptr<VideoDecoder> fallback_decoder_;
>+  const std::string fallback_implementation_name_;
>+  DecodedImageCallback* callback_;
>+};
>+
>+VideoDecoderSoftwareFallbackWrapper::VideoDecoderSoftwareFallbackWrapper(
>+    std::unique_ptr<VideoDecoder> sw_fallback_decoder,
>+    std::unique_ptr<VideoDecoder> hw_decoder)
>+    : decoder_type_(DecoderType::kNone),
>+      hw_decoder_(std::move(hw_decoder)),
>+      fallback_decoder_(std::move(sw_fallback_decoder)),
>+      fallback_implementation_name_(
>+          std::string(fallback_decoder_->ImplementationName()) +
>+          " (fallback from: " + hw_decoder_->ImplementationName() + ")"),
>+      callback_(nullptr) {}
>+VideoDecoderSoftwareFallbackWrapper::~VideoDecoderSoftwareFallbackWrapper() =
>+    default;
>+
>+int32_t VideoDecoderSoftwareFallbackWrapper::InitDecode(
>+    const VideoCodec* codec_settings,
>+    int32_t number_of_cores) {
>+  codec_settings_ = *codec_settings;
>+  number_of_cores_ = number_of_cores;
>+
>+  int32_t status = InitHwDecoder();
>+  if (status == WEBRTC_VIDEO_CODEC_OK) {
>+    return WEBRTC_VIDEO_CODEC_OK;
>+  }
>+
>+  RTC_DCHECK(decoder_type_ == DecoderType::kNone);
>+  if (InitFallbackDecoder()) {
>+    return WEBRTC_VIDEO_CODEC_OK;
>+  }
>+
>+  return status;
>+}
>+
>+int32_t VideoDecoderSoftwareFallbackWrapper::InitHwDecoder() {
>+  RTC_DCHECK(decoder_type_ == DecoderType::kNone);
>+  int32_t status = hw_decoder_->InitDecode(&codec_settings_, number_of_cores_);
>+  if (status != WEBRTC_VIDEO_CODEC_OK) {
>+    return status;
>+  }
>+
>+  decoder_type_ = DecoderType::kHardware;
>+  if (callback_)
>+    hw_decoder_->RegisterDecodeCompleteCallback(callback_);
>+  return status;
>+}
>+
>+bool VideoDecoderSoftwareFallbackWrapper::InitFallbackDecoder() {
>+  RTC_DCHECK(decoder_type_ == DecoderType::kNone ||
>+             decoder_type_ == DecoderType::kHardware);
>+  RTC_LOG(LS_WARNING) << "Decoder falling back to software decoding.";
>+  int32_t status =
>+      fallback_decoder_->InitDecode(&codec_settings_, number_of_cores_);
>+  if (status != WEBRTC_VIDEO_CODEC_OK) {
>+    RTC_LOG(LS_ERROR) << "Failed to initialize software-decoder fallback.";
>+    return false;
>+  }
>+
>+  if (decoder_type_ == DecoderType::kHardware) {
>+    hw_decoder_->Release();
>+  }
>+  decoder_type_ = DecoderType::kFallback;
>+
>+  if (callback_)
>+    fallback_decoder_->RegisterDecodeCompleteCallback(callback_);
>+  return true;
>+}
>+
>+int32_t VideoDecoderSoftwareFallbackWrapper::Decode(
>+    const EncodedImage& input_image,
>+    bool missing_frames,
>+    const CodecSpecificInfo* codec_specific_info,
>+    int64_t render_time_ms) {
>+  TRACE_EVENT0("webrtc", "VideoDecoderSoftwareFallbackWrapper::Decode");
>+  switch (decoder_type_) {
>+    case DecoderType::kNone:
>+      return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
>+    case DecoderType::kHardware: {
>+      int32_t ret = WEBRTC_VIDEO_CODEC_FALLBACK_SOFTWARE;
>+      ret = hw_decoder_->Decode(input_image, missing_frames,
>+                                codec_specific_info, render_time_ms);
>+      if (ret != WEBRTC_VIDEO_CODEC_FALLBACK_SOFTWARE) {
>+        return ret;
>+      }
>+
>+      // HW decoder returned WEBRTC_VIDEO_CODEC_FALLBACK_SOFTWARE or
>+      // initialization failed, fallback to software.
>+      if (!InitFallbackDecoder()) {
>+        return ret;
>+      }
>+
>+      // Fallback decoder initialized, fall-through.
>+      RTC_FALLTHROUGH();
>+    }
>+    case DecoderType::kFallback:
>+      return fallback_decoder_->Decode(input_image, missing_frames,
>+                                       codec_specific_info, render_time_ms);
>+    default:
>+      RTC_NOTREACHED();
>+      return WEBRTC_VIDEO_CODEC_ERROR;
>+  }
>+}
>+
>+int32_t VideoDecoderSoftwareFallbackWrapper::RegisterDecodeCompleteCallback(
>+    DecodedImageCallback* callback) {
>+  callback_ = callback;
>+  return active_decoder().RegisterDecodeCompleteCallback(callback);
>+}
>+
>+int32_t VideoDecoderSoftwareFallbackWrapper::Release() {
>+  int32_t status;
>+  switch (decoder_type_) {
>+    case DecoderType::kHardware:
>+      status = hw_decoder_->Release();
>+      break;
>+    case DecoderType::kFallback:
>+      RTC_LOG(LS_INFO) << "Releasing software fallback decoder.";
>+      status = fallback_decoder_->Release();
>+      break;
>+    case DecoderType::kNone:
>+      status = WEBRTC_VIDEO_CODEC_OK;
>+      break;
>+    default:
>+      RTC_NOTREACHED();
>+      status = WEBRTC_VIDEO_CODEC_ERROR;
>+  }
>+
>+  decoder_type_ = DecoderType::kNone;
>+  return status;
>+}
>+
>+bool VideoDecoderSoftwareFallbackWrapper::PrefersLateDecoding() const {
>+  return active_decoder().PrefersLateDecoding();
>+}
>+
>+const char* VideoDecoderSoftwareFallbackWrapper::ImplementationName() const {
>+  return decoder_type_ == DecoderType::kFallback
>+             ? fallback_implementation_name_.c_str()
>+             : hw_decoder_->ImplementationName();
>+}
>+
>+VideoDecoder& VideoDecoderSoftwareFallbackWrapper::active_decoder() const {
>+  return decoder_type_ == DecoderType::kFallback ? *fallback_decoder_
>+                                                 : *hw_decoder_;
>+}
>+
>+}  // namespace
>+
>+std::unique_ptr<VideoDecoder> CreateVideoDecoderSoftwareFallbackWrapper(
>+    std::unique_ptr<VideoDecoder> sw_fallback_decoder,
>+    std::unique_ptr<VideoDecoder> hw_decoder) {
>+  return absl::make_unique<VideoDecoderSoftwareFallbackWrapper>(
>+      std::move(sw_fallback_decoder), std::move(hw_decoder));
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_decoder_software_fallback_wrapper.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_decoder_software_fallback_wrapper.h
>new file mode 100644
>index 00000000000..0608715182f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_decoder_software_fallback_wrapper.h
>@@ -0,0 +1,29 @@
>+/*
>+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_VIDEO_CODECS_VIDEO_DECODER_SOFTWARE_FALLBACK_WRAPPER_H_
>+#define API_VIDEO_CODECS_VIDEO_DECODER_SOFTWARE_FALLBACK_WRAPPER_H_
>+
>+#include <memory>
>+
>+#include "api/video_codecs/video_decoder.h"
>+
>+namespace webrtc {
>+
>+// Used to wrap external VideoDecoders to provide a fallback option on
>+// software decoding when a hardware decoder fails to decode a stream due to
>+// hardware restrictions, such as max resolution.
>+std::unique_ptr<VideoDecoder> CreateVideoDecoderSoftwareFallbackWrapper(
>+    std::unique_ptr<VideoDecoder> sw_fallback_decoder,
>+    std::unique_ptr<VideoDecoder> hw_decoder);
>+
>+}  // namespace webrtc
>+
>+#endif  // API_VIDEO_CODECS_VIDEO_DECODER_SOFTWARE_FALLBACK_WRAPPER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_encoder.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_encoder.cc
>index e1cc0cfac7c..008780e38c4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_encoder.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_encoder.cc
>@@ -17,10 +17,8 @@ VideoCodecVP8 VideoEncoder::GetDefaultVp8Settings() {
>   VideoCodecVP8 vp8_settings;
>   memset(&vp8_settings, 0, sizeof(vp8_settings));
> 
>-  vp8_settings.resilience = kResilientStream;
>   vp8_settings.numberOfTemporalLayers = 1;
>   vp8_settings.denoisingOn = true;
>-  vp8_settings.errorConcealmentOn = false;
>   vp8_settings.automaticResizeOn = false;
>   vp8_settings.frameDroppingOn = true;
>   vp8_settings.keyFrameInterval = 3000;
>@@ -32,7 +30,6 @@ VideoCodecVP9 VideoEncoder::GetDefaultVp9Settings() {
>   VideoCodecVP9 vp9_settings;
>   memset(&vp9_settings, 0, sizeof(vp9_settings));
> 
>-  vp9_settings.resilienceOn = true;
>   vp9_settings.numberOfTemporalLayers = 1;
>   vp9_settings.denoisingOn = true;
>   vp9_settings.frameDroppingOn = true;
>@@ -41,6 +38,7 @@ VideoCodecVP9 VideoEncoder::GetDefaultVp9Settings() {
>   vp9_settings.automaticResizeOn = true;
>   vp9_settings.numberOfSpatialLayers = 1;
>   vp9_settings.flexibleMode = false;
>+  vp9_settings.interLayerPred = InterLayerPredMode::kOn;
> 
>   return vp9_settings;
> }
>@@ -60,24 +58,26 @@ VideoCodecH264 VideoEncoder::GetDefaultH264Settings() {
>   return h264_settings;
> }
> 
>-VideoEncoder::ScalingSettings::ScalingSettings(bool on, int low, int high)
>-    : enabled(on), thresholds(QpThresholds(low, high)) {}
>+VideoEncoder::ScalingSettings::ScalingSettings() = default;
> 
>-VideoEncoder::ScalingSettings::ScalingSettings(bool on,
>-                                               int low,
>+VideoEncoder::ScalingSettings::ScalingSettings(KOff) : ScalingSettings() {}
>+
>+VideoEncoder::ScalingSettings::ScalingSettings(int low, int high)
>+    : thresholds(QpThresholds(low, high)) {}
>+
>+VideoEncoder::ScalingSettings::ScalingSettings(int low,
>                                                int high,
>                                                int min_pixels)
>-    : enabled(on),
>-      thresholds(QpThresholds(low, high)),
>-      min_pixels_per_frame(min_pixels) {}
>-
>-VideoEncoder::ScalingSettings::ScalingSettings(bool on, int min_pixels)
>-    : enabled(on), min_pixels_per_frame(min_pixels) {}
>+    : thresholds(QpThresholds(low, high)), min_pixels_per_frame(min_pixels) {}
> 
>-VideoEncoder::ScalingSettings::ScalingSettings(bool on) : enabled(on) {}
>+VideoEncoder::ScalingSettings::ScalingSettings(const ScalingSettings&) =
>+    default;
> 
> VideoEncoder::ScalingSettings::~ScalingSettings() {}
> 
>+// static
>+constexpr VideoEncoder::ScalingSettings::KOff
>+    VideoEncoder::ScalingSettings::kOff;
> 
> int32_t VideoEncoder::SetRates(uint32_t bitrate, uint32_t framerate) {
>   RTC_NOTREACHED() << "SetRate(uint32_t, uint32_t) is deprecated.";
>@@ -85,17 +85,13 @@ int32_t VideoEncoder::SetRates(uint32_t bitrate, uint32_t framerate) {
> }
> 
> int32_t VideoEncoder::SetRateAllocation(
>-    const BitrateAllocation& allocation,
>+    const VideoBitrateAllocation& allocation,
>     uint32_t framerate) {
>   return SetRates(allocation.get_sum_kbps(), framerate);
> }
> 
> VideoEncoder::ScalingSettings VideoEncoder::GetScalingSettings() const {
>-  return ScalingSettings(false);
>-}
>-
>-int32_t VideoEncoder::SetPeriodicKeyFrames(bool enable) {
>-  return -1;
>+  return ScalingSettings::kOff;
> }
> 
> bool VideoEncoder::SupportsNativeHandle() const {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_encoder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_encoder.h
>index 64e33069099..68d9b445580 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_encoder.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_encoder.h
>@@ -15,19 +15,18 @@
> #include <string>
> #include <vector>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
>+#include "api/video/video_bitrate_allocation.h"
> #include "api/video/video_frame.h"
>-#include "common_types.h"  // NOLINT(build/include)
>+#include "api/video_codecs/video_codec.h"
> #include "common_video/include/video_frame.h"
> #include "rtc_base/checks.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
> class RTPFragmentationHeader;
> // TODO(pbos): Expose these through a public (root) header or change these APIs.
> struct CodecSpecificInfo;
>-class VideoCodec;
> 
> class EncodedImageCallback {
>  public:
>@@ -71,10 +70,7 @@ class EncodedImageCallback {
>       const CodecSpecificInfo* codec_specific_info,
>       const RTPFragmentationHeader* fragmentation) = 0;
> 
>-  // Deprecated. TODO(ilnik): Remove this in few weeks.
>-  virtual void OnDroppedFrame() {}
>-
>-  virtual void OnDroppedFrame(DropReason) {}
>+  virtual void OnDroppedFrame(DropReason reason) {}
> };
> 
> class VideoEncoder {
>@@ -85,22 +81,39 @@ class VideoEncoder {
>     int low;
>     int high;
>   };
>+  // Quality scaling is enabled if thresholds are provided.
>   struct ScalingSettings {
>-    ScalingSettings(bool on, int low, int high);
>-    ScalingSettings(bool on, int low, int high, int min_pixels);
>-    ScalingSettings(bool on, int min_pixels);
>-    explicit ScalingSettings(bool on);
>+   private:
>+    // Private magic type for kOff, implicitly convertible to
>+    // ScalingSettings.
>+    struct KOff {};
>+
>+   public:
>+    // TODO(nisse): Would be nicer if kOff were a constant ScalingSettings
>+    // rather than a magic value. However, absl::optional is not trivially copy
>+    // constructible, and hence a constant ScalingSettings needs a static
>+    // initializer, which is strongly discouraged in Chrome. We can hopefully
>+    // fix this when we switch to absl::optional or std::optional.
>+    static constexpr KOff kOff = {};
>+
>+    ScalingSettings(int low, int high);
>+    ScalingSettings(int low, int high, int min_pixels);
>     ScalingSettings(const ScalingSettings&);
>+    ScalingSettings(KOff);  // NOLINT(runtime/explicit)
>     ~ScalingSettings();
> 
>-    const bool enabled;
>-    const rtc::Optional<QpThresholds> thresholds;
>+    const absl::optional<QpThresholds> thresholds;
> 
>     // We will never ask for a resolution lower than this.
>     // TODO(kthelgason): Lower this limit when better testing
>     // on MediaCodec and fallback implementations are in place.
>     // See https://bugs.chromium.org/p/webrtc/issues/detail?id=7206
>     const int min_pixels_per_frame = 320 * 180;
>+
>+   private:
>+    // Private constructor; to get an object without thresholds, use
>+    // the magic constant ScalingSettings::kOff.
>+    ScalingSettings();
>   };
> 
>   static VideoCodecVP8 GetDefaultVp8Settings();
>@@ -180,14 +193,13 @@ class VideoEncoder {
> 
>   // Default fallback: Just use the sum of bitrates as the single target rate.
>   // TODO(sprang): Remove this default implementation when we remove SetRates().
>-  virtual int32_t SetRateAllocation(const BitrateAllocation& allocation,
>+  virtual int32_t SetRateAllocation(const VideoBitrateAllocation& allocation,
>                                     uint32_t framerate);
> 
>   // Any encoder implementation wishing to use the WebRTC provided
>   // quality scaler must implement this method.
>   virtual ScalingSettings GetScalingSettings() const;
> 
>-  virtual int32_t SetPeriodicKeyFrames(bool enable);
>   virtual bool SupportsNativeHandle() const;
>   virtual const char* ImplementationName() const;
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/video_config.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_encoder_config.cc
>similarity index 87%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/call/video_config.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_encoder_config.cc
>index 85140515f79..74977eaea3f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/video_config.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_encoder_config.cc
>@@ -7,13 +7,13 @@
>  *  in the file PATENTS.  All contributing project authors may
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
>-#include "call/video_config.h"
>+#include "api/video_codecs/video_encoder_config.h"
> 
> #include <algorithm>
>-#include <sstream>
> #include <string>
> 
> #include "rtc_base/checks.h"
>+#include "rtc_base/strings/string_builder.h"
> 
> namespace webrtc {
> VideoStream::VideoStream()
>@@ -30,7 +30,8 @@ VideoStream::VideoStream(const VideoStream& other) = default;
> VideoStream::~VideoStream() = default;
> 
> std::string VideoStream::ToString() const {
>-  std::stringstream ss;
>+  char buf[1024];
>+  rtc::SimpleStringBuilder ss(buf);
>   ss << "{width: " << width;
>   ss << ", height: " << height;
>   ss << ", max_framerate: " << max_framerate;
>@@ -38,22 +39,17 @@ std::string VideoStream::ToString() const {
>   ss << ", target_bitrate_bps:" << target_bitrate_bps;
>   ss << ", max_bitrate_bps:" << max_bitrate_bps;
>   ss << ", max_qp: " << max_qp;
>+  ss << ", num_temporal_layers: " << num_temporal_layers.value_or(0);
>+  ss << ", bitrate_priority: " << bitrate_priority.value_or(0);
>   ss << ", active: " << active;
> 
>-  ss << ", temporal_layer_thresholds_bps: [";
>-  for (size_t i = 0; i < temporal_layer_thresholds_bps.size(); ++i) {
>-    ss << temporal_layer_thresholds_bps[i];
>-    if (i != temporal_layer_thresholds_bps.size() - 1)
>-      ss << ", ";
>-  }
>-  ss << ']';
>-
>-  ss << '}';
>   return ss.str();
> }
> 
> VideoEncoderConfig::VideoEncoderConfig()
>-    : content_type(ContentType::kRealtimeVideo),
>+    : codec_type(kVideoCodecGeneric),
>+      video_format("Unset"),
>+      content_type(ContentType::kRealtimeVideo),
>       encoder_specific_settings(nullptr),
>       min_transmit_bitrate_bps(0),
>       max_bitrate_bps(0),
>@@ -65,8 +61,11 @@ VideoEncoderConfig::VideoEncoderConfig(VideoEncoderConfig&&) = default;
> VideoEncoderConfig::~VideoEncoderConfig() = default;
> 
> std::string VideoEncoderConfig::ToString() const {
>-  std::stringstream ss;
>-  ss << "{content_type: ";
>+  char buf[1024];
>+  rtc::SimpleStringBuilder ss(buf);
>+  ss << "{codec_type: ";
>+  ss << CodecTypeToPayloadString(codec_type);
>+  ss << ", content_type: ";
>   switch (content_type) {
>     case ContentType::kRealtimeVideo:
>       ss << "kRealtimeVideo";
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/video_config.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_encoder_config.h
>similarity index 81%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/call/video_config.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_encoder_config.h
>index 038cdb2b5b0..e10f08190c6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/video_config.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_encoder_config.h
>@@ -8,18 +8,17 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#ifndef CALL_VIDEO_CONFIG_H_
>-#define CALL_VIDEO_CONFIG_H_
>+#ifndef API_VIDEO_CODECS_VIDEO_ENCODER_CONFIG_H_
>+#define API_VIDEO_CODECS_VIDEO_ENCODER_CONFIG_H_
> 
> #include <string>
> #include <vector>
> 
>-#include "api/optional.h"
>-#include "common_types.h"  // NOLINT(build/include)
>-#include "rtc_base/basictypes.h"
>+#include "absl/types/optional.h"
>+#include "api/video_codecs/sdp_video_format.h"
>+#include "api/video_codecs/video_codec.h"
> #include "rtc_base/refcount.h"
> #include "rtc_base/scoped_ref_ptr.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -36,24 +35,14 @@ struct VideoStream {
>   int min_bitrate_bps;
>   int target_bitrate_bps;
>   int max_bitrate_bps;
>-  rtc::Optional<double> bitrate_priority;
>-
>   int max_qp;
> 
>+  absl::optional<size_t> num_temporal_layers;
>+
>+  absl::optional<double> bitrate_priority;
>+
>   // TODO(bugs.webrtc.org/8653): Support active per-simulcast layer.
>   bool active;
>-
>-  // Bitrate thresholds for enabling additional temporal layers. Since these are
>-  // thresholds in between layers, we have one additional layer. One threshold
>-  // gives two temporal layers, one below the threshold and one above, two give
>-  // three, and so on.
>-  // The VideoEncoder may redistribute bitrates over the temporal layers so a
>-  // bitrate threshold of 100k and an estimate of 105k does not imply that we
>-  // get 100k in one temporal layer and 5k in the other, just that the bitrate
>-  // in the first temporal layer should not exceed 100k.
>-  // TODO(kthelgason): Apart from a special case for two-layer screencast these
>-  // thresholds are not propagated to the VideoEncoder. To be implemented.
>-  std::vector<int> temporal_layer_thresholds_bps;
> };
> 
> class VideoEncoderConfig {
>@@ -61,7 +50,7 @@ class VideoEncoderConfig {
>   // These are reference counted to permit copying VideoEncoderConfig and be
>   // kept alive until all encoder_specific_settings go out of scope.
>   // TODO(kthelgason): Consider removing the need for copying VideoEncoderConfig
>-  // and use rtc::Optional for encoder_specific_settings instead.
>+  // and use absl::optional for encoder_specific_settings instead.
>   class EncoderSpecificSettings : public rtc::RefCountInterface {
>    public:
>     // TODO(pbos): Remove FillEncoderSpecificSettings as soon as VideoCodec is
>@@ -136,6 +125,10 @@ class VideoEncoderConfig {
>   ~VideoEncoderConfig();
>   std::string ToString() const;
> 
>+  // TODO(nisse): Consolidate on one of these.
>+  VideoCodecType codec_type;
>+  SdpVideoFormat video_format;
>+
>   rtc::scoped_refptr<VideoStreamFactoryInterface> video_stream_factory;
>   std::vector<SpatialLayer> spatial_layers;
>   ContentType content_type;
>@@ -153,6 +146,8 @@ class VideoEncoderConfig {
>   // The simulcast layer's configurations set by the application for this video
>   // sender. These are modified by the video_stream_factory before being passed
>   // down to lower layers for the video encoding.
>+  // |simulcast_layers| is also used for configuring non-simulcast (when there
>+  // is a single VideoStream).
>   std::vector<VideoStream> simulcast_layers;
> 
>   // Max number of encoded VideoStreams to produce.
>@@ -166,4 +161,4 @@ class VideoEncoderConfig {
> 
> }  // namespace webrtc
> 
>-#endif  // CALL_VIDEO_CONFIG_H_
>+#endif  // API_VIDEO_CODECS_VIDEO_ENCODER_CONFIG_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/videoencodersoftwarefallbackwrapper.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_encoder_software_fallback_wrapper.cc
>similarity index 72%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/videoencodersoftwarefallbackwrapper.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_encoder_software_fallback_wrapper.cc
>index 69254f5fb50..99f29707e44 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/videoencodersoftwarefallbackwrapper.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_encoder_software_fallback_wrapper.cc
>@@ -8,12 +8,16 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "media/engine/videoencodersoftwarefallbackwrapper.h"
>+#include "api/video_codecs/video_encoder_software_fallback_wrapper.h"
> 
>+#include <cstdio>
>+#include <string>
> #include <utility>
>+#include <vector>
> 
>+#include "absl/memory/memory.h"
>+#include "media/base/codec.h"
> #include "media/base/h264_profile_level_id.h"
>-#include "media/engine/internalencoderfactory.h"
> #include "modules/video_coding/include/video_error_codes.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>@@ -21,7 +25,9 @@
> #include "system_wrappers/include/field_trial.h"
> 
> namespace webrtc {
>+
> namespace {
>+
> const char kVp8ForceFallbackEncoderFieldTrial[] =
>     "WebRTC-VP8-Forced-Fallback-Encoder-v2";
> 
>@@ -62,7 +68,81 @@ void GetForcedFallbackParamsFromFieldTrialGroup(int* param_min_pixels,
>   *param_min_pixels = min_pixels;
>   *param_max_pixels = max_pixels;
> }
>-}  // namespace
>+
>+class VideoEncoderSoftwareFallbackWrapper final : public VideoEncoder {
>+ public:
>+  VideoEncoderSoftwareFallbackWrapper(
>+      std::unique_ptr<webrtc::VideoEncoder> sw_encoder,
>+      std::unique_ptr<webrtc::VideoEncoder> hw_encoder);
>+  ~VideoEncoderSoftwareFallbackWrapper() override;
>+
>+  int32_t InitEncode(const VideoCodec* codec_settings,
>+                     int32_t number_of_cores,
>+                     size_t max_payload_size) override;
>+
>+  int32_t RegisterEncodeCompleteCallback(
>+      EncodedImageCallback* callback) override;
>+
>+  int32_t Release() override;
>+  int32_t Encode(const VideoFrame& frame,
>+                 const CodecSpecificInfo* codec_specific_info,
>+                 const std::vector<FrameType>* frame_types) override;
>+  int32_t SetChannelParameters(uint32_t packet_loss, int64_t rtt) override;
>+  int32_t SetRateAllocation(const VideoBitrateAllocation& bitrate_allocation,
>+                            uint32_t framerate) override;
>+  bool SupportsNativeHandle() const override;
>+  ScalingSettings GetScalingSettings() const override;
>+  const char* ImplementationName() const override;
>+
>+ private:
>+  bool InitFallbackEncoder();
>+
>+  // If |forced_fallback_possible_| is true:
>+  // The forced fallback is requested if the resolution is less than or equal to
>+  // |max_pixels_|. The resolution is allowed to be scaled down to
>+  // |min_pixels_|.
>+  class ForcedFallbackParams {
>+   public:
>+    bool IsValid(const VideoCodec& codec) const {
>+      return codec.width * codec.height <= max_pixels_;
>+    }
>+
>+    bool active_ = false;
>+    int min_pixels_ = 320 * 180;
>+    int max_pixels_ = 320 * 240;
>+  };
>+
>+  bool TryInitForcedFallbackEncoder();
>+  bool TryReInitForcedFallbackEncoder();
>+  void ValidateSettingsForForcedFallback();
>+  bool IsForcedFallbackActive() const;
>+  void MaybeModifyCodecForFallback();
>+
>+  // Settings used in the last InitEncode call and used if a dynamic fallback to
>+  // software is required.
>+  VideoCodec codec_settings_;
>+  int32_t number_of_cores_;
>+  size_t max_payload_size_;
>+
>+  // The last bitrate/framerate set, and a flag for noting they are set.
>+  bool rates_set_;
>+  VideoBitrateAllocation bitrate_allocation_;
>+  uint32_t framerate_;
>+
>+  // The last channel parameters set, and a flag for noting they are set.
>+  bool channel_parameters_set_;
>+  uint32_t packet_loss_;
>+  int64_t rtt_;
>+
>+  bool use_fallback_encoder_;
>+  const std::unique_ptr<webrtc::VideoEncoder> encoder_;
>+
>+  const std::unique_ptr<webrtc::VideoEncoder> fallback_encoder_;
>+  EncodedImageCallback* callback_;
>+
>+  bool forced_fallback_possible_;
>+  ForcedFallbackParams forced_fallback_;
>+};
> 
> VideoEncoderSoftwareFallbackWrapper::VideoEncoderSoftwareFallbackWrapper(
>     std::unique_ptr<webrtc::VideoEncoder> sw_encoder,
>@@ -86,6 +166,8 @@ VideoEncoderSoftwareFallbackWrapper::VideoEncoderSoftwareFallbackWrapper(
>             1);  // No HW below.
>   }
> }
>+VideoEncoderSoftwareFallbackWrapper::~VideoEncoderSoftwareFallbackWrapper() =
>+    default;
> 
> bool VideoEncoderSoftwareFallbackWrapper::InitFallbackEncoder() {
>   RTC_LOG(LS_WARNING) << "Encoder falling back to software encoding.";
>@@ -182,13 +264,6 @@ int32_t VideoEncoderSoftwareFallbackWrapper::Encode(
>   // If requested, try a software fallback.
>   bool fallback_requested = (ret == WEBRTC_VIDEO_CODEC_FALLBACK_SOFTWARE);
>   if (fallback_requested && InitFallbackEncoder()) {
>-    if (frame.video_frame_buffer()->type() == VideoFrameBuffer::Type::kNative &&
>-        !fallback_encoder_->SupportsNativeHandle()) {
>-      RTC_LOG(LS_WARNING) << "Fallback encoder doesn't support native frames, "
>-                          << "dropping one frame.";
>-      return WEBRTC_VIDEO_CODEC_ERROR;
>-    }
>-
>     // Start using the fallback with this frame.
>     return fallback_encoder_->Encode(frame, codec_specific_info, frame_types);
>   }
>@@ -208,7 +283,7 @@ int32_t VideoEncoderSoftwareFallbackWrapper::SetChannelParameters(
> }
> 
> int32_t VideoEncoderSoftwareFallbackWrapper::SetRateAllocation(
>-    const BitrateAllocation& bitrate_allocation,
>+    const VideoBitrateAllocation& bitrate_allocation,
>     uint32_t framerate) {
>   rates_set_ = true;
>   bitrate_allocation_ = bitrate_allocation;
>@@ -227,19 +302,14 @@ bool VideoEncoderSoftwareFallbackWrapper::SupportsNativeHandle() const {
> VideoEncoder::ScalingSettings
> VideoEncoderSoftwareFallbackWrapper::GetScalingSettings() const {
>   if (forced_fallback_possible_) {
>-    if (forced_fallback_.active_) {
>-      return VideoEncoder::ScalingSettings(
>-          codec_settings_.VP8().automaticResizeOn,
>-          forced_fallback_.min_pixels_);
>-    }
>-    const auto settings = encoder_->GetScalingSettings();
>-    if (settings.thresholds) {
>-      return VideoEncoder::ScalingSettings(
>-          settings.enabled, settings.thresholds->low, settings.thresholds->high,
>-          forced_fallback_.min_pixels_);
>-    }
>-    return VideoEncoder::ScalingSettings(settings.enabled,
>-                                         forced_fallback_.min_pixels_);
>+    const auto settings = forced_fallback_.active_
>+                              ? fallback_encoder_->GetScalingSettings()
>+                              : encoder_->GetScalingSettings();
>+    return settings.thresholds
>+               ? VideoEncoder::ScalingSettings(settings.thresholds->low,
>+                                               settings.thresholds->high,
>+                                               forced_fallback_.min_pixels_)
>+               : VideoEncoder::ScalingSettings::kOff;
>   }
>   return encoder_->GetScalingSettings();
> }
>@@ -305,4 +375,13 @@ void VideoEncoderSoftwareFallbackWrapper::ValidateSettingsForForcedFallback() {
>   }
> }
> 
>+}  // namespace
>+
>+std::unique_ptr<VideoEncoder> CreateVideoEncoderSoftwareFallbackWrapper(
>+    std::unique_ptr<VideoEncoder> sw_fallback_encoder,
>+    std::unique_ptr<VideoEncoder> hw_encoder) {
>+  return absl::make_unique<VideoEncoderSoftwareFallbackWrapper>(
>+      std::move(sw_fallback_encoder), std::move(hw_encoder));
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_encoder_software_fallback_wrapper.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_encoder_software_fallback_wrapper.h
>new file mode 100644
>index 00000000000..0f46ad41bad
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_encoder_software_fallback_wrapper.h
>@@ -0,0 +1,29 @@
>+/*
>+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef API_VIDEO_CODECS_VIDEO_ENCODER_SOFTWARE_FALLBACK_WRAPPER_H_
>+#define API_VIDEO_CODECS_VIDEO_ENCODER_SOFTWARE_FALLBACK_WRAPPER_H_
>+
>+#include <memory>
>+
>+#include "api/video_codecs/video_encoder.h"
>+
>+namespace webrtc {
>+
>+// Used to wrap external VideoEncoders to provide a fallback option on
>+// software encoding when a hardware encoder fails to encode a stream due to
>+// hardware restrictions, such as max resolution.
>+std::unique_ptr<VideoEncoder> CreateVideoEncoderSoftwareFallbackWrapper(
>+    std::unique_ptr<VideoEncoder> sw_fallback_encoder,
>+    std::unique_ptr<VideoEncoder> hw_encoder);
>+
>+}  // namespace webrtc
>+
>+#endif  // API_VIDEO_CODECS_VIDEO_ENCODER_SOFTWARE_FALLBACK_WRAPPER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/videosourceproxy.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/videosourceproxy.h
>index f2d8be06813..dbd9045be87 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/videosourceproxy.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/videosourceproxy.h
>@@ -11,8 +11,8 @@
> #ifndef API_VIDEOSOURCEPROXY_H_
> #define API_VIDEOSOURCEPROXY_H_
> 
>-#include "api/proxy.h"
> #include "api/mediastreaminterface.h"
>+#include "api/proxy.h"
> 
> namespace webrtc {
> 
>@@ -21,19 +21,19 @@ namespace webrtc {
> // TODO(deadbeef): Move this to .cc file and out of api/. What threads methods
> // are called on is an implementation detail.
> BEGIN_PROXY_MAP(VideoTrackSource)
>-  PROXY_SIGNALING_THREAD_DESTRUCTOR()
>-  PROXY_CONSTMETHOD0(SourceState, state)
>-  PROXY_CONSTMETHOD0(bool, remote)
>-  PROXY_CONSTMETHOD0(bool, is_screencast)
>-  PROXY_CONSTMETHOD0(rtc::Optional<bool>, needs_denoising)
>-  PROXY_METHOD1(bool, GetStats, Stats*)
>-  PROXY_WORKER_METHOD2(void,
>-                       AddOrUpdateSink,
>-                       rtc::VideoSinkInterface<VideoFrame>*,
>-                       const rtc::VideoSinkWants&)
>-  PROXY_WORKER_METHOD1(void, RemoveSink, rtc::VideoSinkInterface<VideoFrame>*)
>-  PROXY_METHOD1(void, RegisterObserver, ObserverInterface*)
>-  PROXY_METHOD1(void, UnregisterObserver, ObserverInterface*)
>+PROXY_SIGNALING_THREAD_DESTRUCTOR()
>+PROXY_CONSTMETHOD0(SourceState, state)
>+PROXY_CONSTMETHOD0(bool, remote)
>+PROXY_CONSTMETHOD0(bool, is_screencast)
>+PROXY_CONSTMETHOD0(absl::optional<bool>, needs_denoising)
>+PROXY_METHOD1(bool, GetStats, Stats*)
>+PROXY_WORKER_METHOD2(void,
>+                     AddOrUpdateSink,
>+                     rtc::VideoSinkInterface<VideoFrame>*,
>+                     const rtc::VideoSinkWants&)
>+PROXY_WORKER_METHOD1(void, RemoveSink, rtc::VideoSinkInterface<VideoFrame>*)
>+PROXY_METHOD1(void, RegisterObserver, ObserverInterface*)
>+PROXY_METHOD1(void, UnregisterObserver, ObserverInterface*)
> END_PROXY_MAP()
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/BUILD.gn
>index d3af836947e..2aaacc75a09 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/BUILD.gn
>@@ -14,6 +14,8 @@ if (is_android) {
> 
> rtc_static_library("audio") {
>   sources = [
>+    "audio_level.cc",
>+    "audio_level.h",
>     "audio_receive_stream.cc",
>     "audio_receive_stream.h",
>     "audio_send_stream.cc",
>@@ -22,11 +24,19 @@ rtc_static_library("audio") {
>     "audio_state.h",
>     "audio_transport_impl.cc",
>     "audio_transport_impl.h",
>+    "channel.cc",
>+    "channel.h",
>+    "channel_proxy.cc",
>+    "channel_proxy.h",
>     "conversion.h",
>     "null_audio_poller.cc",
>     "null_audio_poller.h",
>+    "remix_resample.cc",
>+    "remix_resample.h",
>     "time_interval.cc",
>     "time_interval.h",
>+    "transport_feedback_packet_loss_tracker.cc",
>+    "transport_feedback_packet_loss_tracker.h",
>   ]
> 
>   if (!build_with_chromium && is_clang) {
>@@ -36,32 +46,47 @@ rtc_static_library("audio") {
> 
>   deps = [
>     "..:webrtc_common",
>-    "../api:audio_mixer_api",
>+    "../api:array_view",
>     "../api:call_api",
>-    "../api:optional",
>+    "../api:libjingle_peerconnection_api",
>+    "../api:transport_api",
>+    "../api/audio:aec3_factory",
>+    "../api/audio:audio_frame_api",
>+    "../api/audio:audio_mixer_api",
>     "../api/audio_codecs:audio_codecs_api",
>-    "../api/audio_codecs:builtin_audio_encoder_factory",
>     "../call:bitrate_allocator",
>     "../call:call_interfaces",
>     "../call:rtp_interfaces",
>     "../common_audio",
>+    "../common_audio:common_audio_c",
>+    "../logging:rtc_event_audio",
>+    "../logging:rtc_event_log_api",
>+    "../modules/audio_coding",
>+    "../modules/audio_coding:audio_format_conversion",
>+    "../modules/audio_coding:audio_network_adaptor_config",
>     "../modules/audio_coding:cng",
>     "../modules/audio_device",
>     "../modules/audio_processing",
>     "../modules/bitrate_controller:bitrate_controller",
>-    "../modules/congestion_controller:congestion_controller",
>     "../modules/pacing:pacing",
>     "../modules/remote_bitrate_estimator:remote_bitrate_estimator",
>     "../modules/rtp_rtcp",
>+    "../modules/rtp_rtcp:rtp_rtcp_format",
>+    "../modules/utility",
>+    "../rtc_base:audio_format_to_string",
>     "../rtc_base:checks",
>+    "../rtc_base:rate_limiter",
>     "../rtc_base:rtc_base",
>     "../rtc_base:rtc_base_approved",
>     "../rtc_base:rtc_task_queue",
>+    "../rtc_base:safe_minmax",
>+    "../rtc_base:stringutils",
>     "../system_wrappers",
>     "../system_wrappers:field_trial_api",
>-    "../voice_engine",
>-    "../voice_engine:audio_level",
>+    "../system_wrappers:metrics_api",
>     "utility:audio_frame_operations",
>+    "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
> if (rtc_include_tests) {
>@@ -75,7 +100,6 @@ if (rtc_include_tests) {
>     deps = [
>       ":audio",
>       "../system_wrappers:system_wrappers",
>-      "../test:fake_audio_device",
>       "../test:test_common",
>       "../test:test_support",
>     ]
>@@ -94,28 +118,33 @@ if (rtc_include_tests) {
>       "audio_send_stream_tests.cc",
>       "audio_send_stream_unittest.cc",
>       "audio_state_unittest.cc",
>+      "mock_voe_channel_proxy.h",
>+      "remix_resample_unittest.cc",
>+      "test/audio_stats_test.cc",
>       "time_interval_unittest.cc",
>+      "transport_feedback_packet_loss_tracker_unittest.cc",
>     ]
>     deps = [
>       ":audio",
>       ":audio_end_to_end_test",
>       "../api:mock_audio_mixer",
>+      "../api/audio:audio_frame_api",
>+      "../api/units:time_delta",
>       "../call:mock_call_interfaces",
>       "../call:mock_rtp_interfaces",
>       "../call:rtp_interfaces",
>       "../call:rtp_receiver",
>+      "../common_audio",
>       "../logging:mocks",
>       "../modules/audio_device:mock_audio_device",
>       "../modules/audio_mixer:audio_mixer_impl",
>       "../modules/audio_processing:audio_processing_statistics",
>       "../modules/audio_processing:mocks",
>       "../modules/bitrate_controller:mocks",
>-      "../modules/congestion_controller:congestion_controller",
>-      "../modules/congestion_controller:mock_congestion_controller",
>-      "../modules/pacing:mock_paced_sender",
>       "../modules/pacing:pacing",
>       "../modules/rtp_rtcp:mock_rtp_rtcp",
>       "../modules/rtp_rtcp:rtp_rtcp_format",
>+      "../rtc_base:checks",
>       "../rtc_base:rtc_base_approved",
>       "../rtc_base:rtc_base_tests_utils",
>       "../rtc_base:rtc_task_queue",
>@@ -125,17 +154,11 @@ if (rtc_include_tests) {
>       "../test:rtp_test_utils",
>       "../test:test_common",
>       "../test:test_support",
>-      "../voice_engine",
>       "utility:utility_tests",
>-      "//testing/gmock",
>       "//testing/gtest",
>+      "//third_party/abseil-cpp/absl/memory",
>     ]
> 
>-    if (!rtc_use_memcheck) {
>-      # This test is timing dependent, which rules out running on memcheck bots.
>-      sources += [ "test/audio_stats_test.cc" ]
>-    }
>-
>     if (!build_with_chromium && is_clang) {
>       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>@@ -155,10 +178,9 @@ if (rtc_include_tests) {
>         "../common_audio",
>         "../rtc_base:rtc_base_approved",
>         "../system_wrappers",
>-        "../test:fake_audio_device",
>+        "../test:fileutils",
>         "../test:test_common",
>         "../test:test_main",
>-        "//testing/gmock",
>         "//testing/gtest",
>       ]
>       if (is_android) {
>@@ -166,11 +188,34 @@ if (rtc_include_tests) {
>       }
> 
>       data = [
>-        "../resources/voice_engine/audio_dtx16.wav",
>         "../resources/voice_engine/audio_tiny16.wav",
>         "../resources/voice_engine/audio_tiny48.wav",
>+      ]
>+
>+      if (!build_with_chromium && is_clang) {
>+        # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163)
>+        suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>+      }
>+    }
>+
>+    group("low_bandwidth_audio_perf_test") {
>+      testonly = true
>+
>+      deps = [
>+        ":low_bandwidth_audio_test",
>+      ]
>+
>+      data = [
>         "test/low_bandwidth_audio_test.py",
>+        "../resources/voice_engine/audio_tiny16.wav",
>+        "../resources/voice_engine/audio_tiny48.wav",
>       ]
>+      if (is_win) {
>+        data += [ "${root_out_dir}/low_bandwidth_audio_test.exe" ]
>+      } else {
>+        data += [ "${root_out_dir}/low_bandwidth_audio_test" ]
>+      }
>+
>       if (is_linux || is_android) {
>         data += [
>           "../tools_webrtc/audio_quality/linux/PolqaOem64",
>@@ -189,10 +234,7 @@ if (rtc_include_tests) {
>         data += [ "../tools_webrtc/audio_quality/mac/pesq" ]
>       }
> 
>-      if (!build_with_chromium && is_clang) {
>-        # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163)
>-        suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-      }
>+      write_runtime_deps = "${root_out_dir}/${target_name}.runtime_deps"
>     }
>   }
> 
>@@ -207,13 +249,13 @@ if (rtc_include_tests) {
>       "../common_audio",
>       "../rtc_base:rtc_base_approved",
>       "../system_wrappers",
>-      "../test:fake_audio_device",
>       "../test:field_trial",
>+      "../test:fileutils",
>       "../test:single_threaded_task_queue",
>       "../test:test_common",
>       "../test:test_main",
>-      "//testing/gmock",
>       "//testing/gtest",
>+      "//third_party/abseil-cpp/absl/memory",
>     ]
> 
>     data = [
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/DEPS b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/DEPS
>index 70e33469df9..8bb1f808052 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/DEPS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/DEPS
>@@ -5,14 +5,15 @@ include_rules = [
>   "+modules/audio_coding",
>   "+modules/audio_device",
>   "+modules/audio_mixer",
>+  "+modules/audio_processing",
>   "+modules/audio_processing/include",
>   "+modules/bitrate_controller",
>   "+modules/congestion_controller",
>   "+modules/pacing",
>   "+modules/remote_bitrate_estimator",
>   "+modules/rtp_rtcp",
>+  "+modules/utility",
>   "+system_wrappers",
>-  "+voice_engine",
> ]
> 
> specific_include_rules = {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/voice_engine/audio_level.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_level.cc
>similarity index 65%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/voice_engine/audio_level.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_level.cc
>index 57b485546be..3b8df9778eb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/voice_engine/audio_level.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_level.cc
>@@ -8,33 +8,21 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "voice_engine/audio_level.h"
>+#include "audio/audio_level.h"
> 
>+#include "api/audio/audio_frame.h"
> #include "common_audio/signal_processing/include/signal_processing_library.h"
>-#include "modules/include/module_common_types.h"
> 
> namespace webrtc {
> namespace voe {
> 
>-// Number of bars on the indicator.
>-// Note that the number of elements is specified because we are indexing it
>-// in the range of 0-32
>-constexpr int8_t kPermutation[33] = {0, 1, 2, 3, 4, 4, 5, 5, 5, 5, 6,
>-                                     6, 6, 6, 6, 7, 7, 7, 7, 8, 8, 8,
>-                                     9, 9, 9, 9, 9, 9, 9, 9, 9, 9, 9};
>-
> AudioLevel::AudioLevel()
>-    : abs_max_(0), count_(0), current_level_(0), current_level_full_range_(0) {
>+    : abs_max_(0), count_(0), current_level_full_range_(0) {
>   WebRtcSpl_Init();
> }
> 
> AudioLevel::~AudioLevel() {}
> 
>-int8_t AudioLevel::Level() const {
>-  rtc::CritScope cs(&crit_sect_);
>-  return current_level_;
>-}
>-
> int16_t AudioLevel::LevelFullRange() const {
>   rtc::CritScope cs(&crit_sect_);
>   return current_level_full_range_;
>@@ -44,7 +32,6 @@ void AudioLevel::Clear() {
>   rtc::CritScope cs(&crit_sect_);
>   abs_max_ = 0;
>   count_ = 0;
>-  current_level_ = 0;
>   current_level_full_range_ = 0;
> }
> 
>@@ -60,10 +47,12 @@ double AudioLevel::TotalDuration() const {
> 
> void AudioLevel::ComputeLevel(const AudioFrame& audioFrame, double duration) {
>   // Check speech level (works for 2 channels as well)
>-  int16_t abs_value = audioFrame.muted() ? 0 :
>-      WebRtcSpl_MaxAbsValueW16(
>-          audioFrame.data(),
>-          audioFrame.samples_per_channel_ * audioFrame.num_channels_);
>+  int16_t abs_value =
>+      audioFrame.muted()
>+          ? 0
>+          : WebRtcSpl_MaxAbsValueW16(
>+                audioFrame.data(),
>+                audioFrame.samples_per_channel_ * audioFrame.num_channels_);
> 
>   // Protect member access using a lock since this method is called on a
>   // dedicated audio thread in the RecordedDataIsAvailable() callback.
>@@ -78,18 +67,6 @@ void AudioLevel::ComputeLevel(const AudioFrame& audioFrame, double duration) {
> 
>     count_ = 0;
> 
>-    // Highest value for a int16_t is 0x7fff = 32767
>-    // Divide with 1000 to get in the range of 0-32 which is the range of the
>-    // permutation vector
>-    int32_t position = abs_max_ / 1000;
>-
>-    // Make it less likely that the bar stays at position 0. I.e. only if it's
>-    // in the range 0-250 (instead of 0-1000)
>-    if ((position == 0) && (abs_max_ > 250)) {
>-      position = 1;
>-    }
>-    current_level_ = kPermutation[position];
>-
>     // Decay the absolute maximum (divide by 4)
>     abs_max_ >>= 2;
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/voice_engine/audio_level.h b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_level.h
>similarity index 84%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/voice_engine/audio_level.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_level.h
>index a1951edba90..3bbe5fdd201 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/voice_engine/audio_level.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_level.h
>@@ -8,12 +8,11 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#ifndef VOICE_ENGINE_AUDIO_LEVEL_H_
>-#define VOICE_ENGINE_AUDIO_LEVEL_H_
>+#ifndef AUDIO_AUDIO_LEVEL_H_
>+#define AUDIO_AUDIO_LEVEL_H_
> 
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/thread_annotations.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -26,8 +25,7 @@ class AudioLevel {
>   ~AudioLevel();
> 
>   // Called on "API thread(s)" from APIs like VoEBase::CreateChannel(),
>-  // VoEBase::StopSend(), VoEVolumeControl::GetSpeechOutputLevel().
>-  int8_t Level() const;
>+  // VoEBase::StopSend()
>   int16_t LevelFullRange() const;
>   void Clear();
>   // See the description for "totalAudioEnergy" in the WebRTC stats spec
>@@ -47,7 +45,6 @@ class AudioLevel {
> 
>   int16_t abs_max_ RTC_GUARDED_BY(crit_sect_);
>   int16_t count_ RTC_GUARDED_BY(crit_sect_);
>-  int8_t current_level_ RTC_GUARDED_BY(crit_sect_);
>   int16_t current_level_full_range_ RTC_GUARDED_BY(crit_sect_);
> 
>   double total_energy_ RTC_GUARDED_BY(crit_sect_) = 0.0;
>@@ -57,4 +54,4 @@ class AudioLevel {
> }  // namespace voe
> }  // namespace webrtc
> 
>-#endif  // VOICE_ENGINE_AUDIO_LEVEL_H_
>+#endif  // AUDIO_AUDIO_LEVEL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_receive_stream.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_receive_stream.cc
>index 294dad0094b..b507afc1676 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_receive_stream.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_receive_stream.cc
>@@ -16,20 +16,21 @@
> #include "api/call/audio_sink.h"
> #include "audio/audio_send_stream.h"
> #include "audio/audio_state.h"
>+#include "audio/channel_proxy.h"
> #include "audio/conversion.h"
> #include "call/rtp_stream_receiver_controller_interface.h"
> #include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
>-#include "modules/rtp_rtcp/include/rtp_receiver.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>+#include "rtc_base/strings/string_builder.h"
> #include "rtc_base/timeutils.h"
>-#include "voice_engine/channel_proxy.h"
> 
> namespace webrtc {
> 
> std::string AudioReceiveStream::Config::Rtp::ToString() const {
>-  std::stringstream ss;
>+  char ss_buf[1024];
>+  rtc::SimpleStringBuilder ss(ss_buf);
>   ss << "{remote_ssrc: " << remote_ssrc;
>   ss << ", local_ssrc: " << local_ssrc;
>   ss << ", transport_cc: " << (transport_cc ? "on" : "off");
>@@ -47,7 +48,8 @@ std::string AudioReceiveStream::Config::Rtp::ToString() const {
> }
> 
> std::string AudioReceiveStream::Config::ToString() const {
>-  std::stringstream ss;
>+  char ss_buf[1024];
>+  rtc::SimpleStringBuilder ss(ss_buf);
>   ss << "{rtp: " << rtp.ToString();
>   ss << ", rtcp_send_transport: "
>      << (rtcp_send_transport ? "(Transport)" : "null");
>@@ -63,17 +65,16 @@ namespace {
> std::unique_ptr<voe::ChannelProxy> CreateChannelAndProxy(
>     webrtc::AudioState* audio_state,
>     ProcessThread* module_process_thread,
>-    const webrtc::AudioReceiveStream::Config& config) {
>+    const webrtc::AudioReceiveStream::Config& config,
>+    RtcEventLog* event_log) {
>   RTC_DCHECK(audio_state);
>   internal::AudioState* internal_audio_state =
>       static_cast<internal::AudioState*>(audio_state);
>-  return std::unique_ptr<voe::ChannelProxy>(new voe::ChannelProxy(
>-      std::unique_ptr<voe::Channel>(new voe::Channel(
>-              module_process_thread,
>-              internal_audio_state->audio_device_module(),
>-              config.jitter_buffer_max_packets,
>-              config.jitter_buffer_fast_accelerate,
>-              config.decoder_factory))));
>+  return absl::make_unique<voe::ChannelProxy>(absl::make_unique<voe::Channel>(
>+      module_process_thread, internal_audio_state->audio_device_module(),
>+      nullptr /* RtcpRttStats */, event_log, config.rtp.remote_ssrc,
>+      config.jitter_buffer_max_packets, config.jitter_buffer_fast_accelerate,
>+      config.decoder_factory, config.codec_pair_id));
> }
> }  // namespace
> 
>@@ -91,7 +92,8 @@ AudioReceiveStream::AudioReceiveStream(
>                          event_log,
>                          CreateChannelAndProxy(audio_state.get(),
>                                                module_process_thread,
>-                                               config)) {}
>+                                               config,
>+                                               event_log)) {}
> 
> AudioReceiveStream::AudioReceiveStream(
>     RtpStreamReceiverControllerInterface* receiver_controller,
>@@ -100,9 +102,8 @@ AudioReceiveStream::AudioReceiveStream(
>     const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
>     webrtc::RtcEventLog* event_log,
>     std::unique_ptr<voe::ChannelProxy> channel_proxy)
>-    : audio_state_(audio_state),
>-      channel_proxy_(std::move(channel_proxy)) {
>-  RTC_LOG(LS_INFO) << "AudioReceiveStream: " << config.ToString();
>+    : audio_state_(audio_state), channel_proxy_(std::move(channel_proxy)) {
>+  RTC_LOG(LS_INFO) << "AudioReceiveStream: " << config.rtp.remote_ssrc;
>   RTC_DCHECK(receiver_controller);
>   RTC_DCHECK(packet_router);
>   RTC_DCHECK(config.decoder_factory);
>@@ -111,34 +112,30 @@ AudioReceiveStream::AudioReceiveStream(
> 
>   module_process_thread_checker_.DetachFromThread();
> 
>-  channel_proxy_->SetRtcEventLog(event_log);
>   channel_proxy_->RegisterTransport(config.rtcp_send_transport);
> 
>   // Configure bandwidth estimation.
>   channel_proxy_->RegisterReceiverCongestionControlObjects(packet_router);
> 
>   // Register with transport.
>-  rtp_stream_receiver_ =
>-      receiver_controller->CreateReceiver(config.rtp.remote_ssrc,
>-                                          channel_proxy_.get());
>+  rtp_stream_receiver_ = receiver_controller->CreateReceiver(
>+      config.rtp.remote_ssrc, channel_proxy_.get());
> 
>   ConfigureStream(this, config, true);
> }
> 
> AudioReceiveStream::~AudioReceiveStream() {
>   RTC_DCHECK_RUN_ON(&worker_thread_checker_);
>-  RTC_LOG(LS_INFO) << "~AudioReceiveStream: " << config_.ToString();
>+  RTC_LOG(LS_INFO) << "~AudioReceiveStream: " << config_.rtp.remote_ssrc;
>   Stop();
>   channel_proxy_->DisassociateSendChannel();
>   channel_proxy_->RegisterTransport(nullptr);
>   channel_proxy_->ResetReceiverCongestionControlObjects();
>-  channel_proxy_->SetRtcEventLog(nullptr);
> }
> 
> void AudioReceiveStream::Reconfigure(
>     const webrtc::AudioReceiveStream::Config& config) {
>   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  RTC_LOG(LS_INFO) << "AudioReceiveStream::Reconfigure: " << config_.ToString();
>   ConfigureStream(this, config, false);
> }
> 
>@@ -222,11 +219,6 @@ webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const {
>   return stats;
> }
> 
>-int AudioReceiveStream::GetOutputLevel() const {
>-  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
>-  return channel_proxy_->GetSpeechOutputLevel();
>-}
>-
> void AudioReceiveStream::SetSink(AudioSinkInterface* sink) {
>   RTC_DCHECK_RUN_ON(&worker_thread_checker_);
>   channel_proxy_->SetSink(sink);
>@@ -261,30 +253,14 @@ int AudioReceiveStream::id() const {
>   return config_.rtp.remote_ssrc;
> }
> 
>-rtc::Optional<Syncable::Info> AudioReceiveStream::GetInfo() const {
>+absl::optional<Syncable::Info> AudioReceiveStream::GetInfo() const {
>   RTC_DCHECK_RUN_ON(&module_process_thread_checker_);
>-  Syncable::Info info;
>-
>-  RtpRtcp* rtp_rtcp = nullptr;
>-  RtpReceiver* rtp_receiver = nullptr;
>-  channel_proxy_->GetRtpRtcp(&rtp_rtcp, &rtp_receiver);
>-  RTC_DCHECK(rtp_rtcp);
>-  RTC_DCHECK(rtp_receiver);
>-
>-  if (!rtp_receiver->GetLatestTimestamps(
>-          &info.latest_received_capture_timestamp,
>-          &info.latest_receive_time_ms)) {
>-    return rtc::nullopt;
>-  }
>-  if (rtp_rtcp->RemoteNTP(&info.capture_time_ntp_secs,
>-                          &info.capture_time_ntp_frac,
>-                          nullptr,
>-                          nullptr,
>-                          &info.capture_time_source_clock) != 0) {
>-    return rtc::nullopt;
>-  }
>+  absl::optional<Syncable::Info> info = channel_proxy_->GetSyncInfo();
>+
>+  if (!info)
>+    return absl::nullopt;
> 
>-  info.current_delay_ms = channel_proxy_->GetDelayEstimate();
>+  info->current_delay_ms = channel_proxy_->GetDelayEstimate();
>   return info;
> }
> 
>@@ -333,8 +309,8 @@ const webrtc::AudioReceiveStream::Config& AudioReceiveStream::config() const {
>   return config_;
> }
> 
>-const AudioSendStream*
>-    AudioReceiveStream::GetAssociatedSendStreamForTesting() const {
>+const AudioSendStream* AudioReceiveStream::GetAssociatedSendStreamForTesting()
>+    const {
>   RTC_DCHECK_RUN_ON(&worker_thread_checker_);
>   return associated_send_stream_;
> }
>@@ -348,6 +324,8 @@ internal::AudioState* AudioReceiveStream::audio_state() const {
> void AudioReceiveStream::ConfigureStream(AudioReceiveStream* stream,
>                                          const Config& new_config,
>                                          bool first_time) {
>+  RTC_LOG(LS_INFO) << "AudioReceiveStream::ConfigureStream: "
>+                   << new_config.ToString();
>   RTC_DCHECK(stream);
>   const auto& channel_proxy = stream->channel_proxy_;
>   const auto& old_config = stream->config_;
>@@ -365,6 +343,12 @@ void AudioReceiveStream::ConfigureStream(AudioReceiveStream* stream,
>   if (first_time || old_config.rtp.local_ssrc != new_config.rtp.local_ssrc) {
>     channel_proxy->SetLocalSSRC(new_config.rtp.local_ssrc);
>   }
>+
>+  if (!first_time) {
>+    // Remote ssrc can't be changed mid-stream.
>+    RTC_DCHECK_EQ(old_config.rtp.remote_ssrc, new_config.rtp.remote_ssrc);
>+  }
>+
>   // TODO(solenberg): Config NACK history window (which is a packet count),
>   // using the actual packet size for the configured codec.
>   if (first_time || old_config.rtp.nack.rtp_history_ms !=
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_receive_stream.h b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_receive_stream.h
>index 7d0eed9cb21..64552e35c0d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_receive_stream.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_receive_stream.h
>@@ -15,6 +15,7 @@
> #include <vector>
> 
> #include "api/audio/audio_mixer.h"
>+#include "api/rtp_headers.h"
> #include "audio/audio_state.h"
> #include "call/audio_receive_stream.h"
> #include "call/rtp_packet_sink_interface.h"
>@@ -61,7 +62,6 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream,
>   void Start() override;
>   void Stop() override;
>   webrtc::AudioReceiveStream::Stats GetStats() const override;
>-  int GetOutputLevel() const override;
>   void SetSink(AudioSinkInterface* sink) override;
>   void SetGain(float gain) override;
>   std::vector<webrtc::RtpSource> GetSources() const override;
>@@ -80,7 +80,7 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream,
> 
>   // Syncable
>   int id() const override;
>-  rtc::Optional<Syncable::Info> GetInfo() const override;
>+  absl::optional<Syncable::Info> GetInfo() const override;
>   uint32_t GetPlayoutTimestamp() const override;
>   void SetMinimumPlayoutDelay(int delay_ms) override;
> 
>@@ -104,7 +104,7 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream,
>   std::unique_ptr<voe::ChannelProxy> channel_proxy_;
>   AudioSendStream* associated_send_stream_ = nullptr;
> 
>-  bool playing_ RTC_ACCESS_ON(worker_thread_checker_) = false;
>+  bool playing_ RTC_GUARDED_BY(worker_thread_checker_) = false;
> 
>   std::unique_ptr<RtpStreamReceiverInterface> rtp_stream_receiver_;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_receive_stream_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_receive_stream_unittest.cc
>index c8318aa967c..eb93bd0aa5d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_receive_stream_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_receive_stream_unittest.cc
>@@ -15,6 +15,7 @@
> #include "api/test/mock_audio_mixer.h"
> #include "audio/audio_receive_stream.h"
> #include "audio/conversion.h"
>+#include "audio/mock_voe_channel_proxy.h"
> #include "call/rtp_stream_receiver_controller.h"
> #include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
> #include "modules/audio_device/include/mock_audio_device.h"
>@@ -24,7 +25,6 @@
> #include "modules/rtp_rtcp/source/byte_io.h"
> #include "test/gtest.h"
> #include "test/mock_audio_decoder_factory.h"
>-#include "test/mock_voe_channel_proxy.h"
> 
> namespace webrtc {
> namespace test {
>@@ -59,18 +59,16 @@ const unsigned int kSpeechOutputLevel = 99;
> const double kTotalOutputEnergy = 0.25;
> const double kTotalOutputDuration = 0.5;
> 
>-const CallStatistics kCallStats = {
>-    345,  678,  901, 234, -12, 3456, 7890, 567, 890, 123};
>-const CodecInst kCodecInst = {
>-    123, "codec_name_recv", 96000, -187, 0, -103};
>+const CallStatistics kCallStats = {345,  678,  901, 234, -12,
>+                                   3456, 7890, 567, 890, 123};
>+const CodecInst kCodecInst = {123, "codec_name_recv", 96000, -187, 0, -103};
> const NetworkStatistics kNetworkStats = {
>     123, 456, false, 789012, 3456, 123, 456, 0,  {}, 789, 12,
>     345, 678, 901,   0,      -1,   -1,  -1,  -1, -1, 0};
> const AudioDecodingCallStats kAudioDecodeStats = MakeAudioDecodeStatsForTest();
> 
> struct ConfigHelper {
>-  ConfigHelper()
>-      : ConfigHelper(new rtc::RefCountedObject<MockAudioMixer>()) {}
>+  ConfigHelper() : ConfigHelper(new rtc::RefCountedObject<MockAudioMixer>()) {}
> 
>   explicit ConfigHelper(rtc::scoped_refptr<MockAudioMixer> audio_mixer)
>       : audio_mixer_(audio_mixer) {
>@@ -87,23 +85,16 @@ struct ConfigHelper {
>     EXPECT_CALL(*channel_proxy_, SetLocalSSRC(kLocalSsrc)).Times(1);
>     EXPECT_CALL(*channel_proxy_, SetNACKStatus(true, 15)).Times(1);
>     EXPECT_CALL(*channel_proxy_,
>-        RegisterReceiverCongestionControlObjects(&packet_router_))
>-            .Times(1);
>+                RegisterReceiverCongestionControlObjects(&packet_router_))
>+        .Times(1);
>     EXPECT_CALL(*channel_proxy_, ResetReceiverCongestionControlObjects())
>         .Times(1);
>     EXPECT_CALL(*channel_proxy_, RegisterTransport(nullptr)).Times(2);
>-    testing::Expectation expect_set =
>-        EXPECT_CALL(*channel_proxy_, SetRtcEventLog(&event_log_))
>-            .Times(1);
>-    EXPECT_CALL(*channel_proxy_, SetRtcEventLog(testing::IsNull()))
>-        .Times(1)
>-        .After(expect_set);
>     EXPECT_CALL(*channel_proxy_, DisassociateSendChannel()).Times(1);
>     EXPECT_CALL(*channel_proxy_, SetReceiveCodecs(_))
>-        .WillRepeatedly(
>-            Invoke([](const std::map<int, SdpAudioFormat>& codecs) {
>-              EXPECT_THAT(codecs, testing::IsEmpty());
>-            }));
>+        .WillRepeatedly(Invoke([](const std::map<int, SdpAudioFormat>& codecs) {
>+          EXPECT_THAT(codecs, testing::IsEmpty());
>+        }));
> 
>     stream_config_.rtp.local_ssrc = kLocalSsrc;
>     stream_config_.rtp.remote_ssrc = kRemoteSsrc;
>@@ -119,11 +110,8 @@ struct ConfigHelper {
>   std::unique_ptr<internal::AudioReceiveStream> CreateAudioReceiveStream() {
>     return std::unique_ptr<internal::AudioReceiveStream>(
>         new internal::AudioReceiveStream(
>-            &rtp_stream_receiver_controller_,
>-            &packet_router_,
>-            stream_config_,
>-            audio_state_,
>-            &event_log_,
>+            &rtp_stream_receiver_controller_, &packet_router_, stream_config_,
>+            audio_state_, &event_log_,
>             std::unique_ptr<voe::ChannelProxy>(channel_proxy_)));
>   }
> 
>@@ -242,11 +230,11 @@ TEST(AudioReceiveStreamTest, ReceiveRtpPacket) {
>   const int kTransportSequenceNumberValue = 1234;
>   std::vector<uint8_t> rtp_packet = CreateRtpHeaderWithOneByteExtension(
>       kTransportSequenceNumberId, kTransportSequenceNumberValue, 2);
>-  PacketTime packet_time(5678000, 0);
>+  constexpr int64_t packet_time_us = 5678000;
> 
>   RtpPacketReceived parsed_packet;
>   ASSERT_TRUE(parsed_packet.Parse(&rtp_packet[0], rtp_packet.size()));
>-  parsed_packet.set_arrival_time_ms((packet_time.timestamp + 500) / 1000);
>+  parsed_packet.set_arrival_time_ms((packet_time_us + 500) / 1000);
> 
>   EXPECT_CALL(*helper.channel_proxy(),
>               OnRtpPacket(testing::Ref(parsed_packet)));
>@@ -322,7 +310,7 @@ TEST(AudioReceiveStreamTest, SetGain) {
>   ConfigHelper helper;
>   auto recv_stream = helper.CreateAudioReceiveStream();
>   EXPECT_CALL(*helper.channel_proxy(),
>-      SetChannelOutputVolumeScaling(FloatEq(0.765f)));
>+              SetChannelOutputVolumeScaling(FloatEq(0.765f)));
>   recv_stream->SetGain(0.765f);
> }
> 
>@@ -370,10 +358,10 @@ TEST(AudioReceiveStreamTest, ReconfigureWithUpdatedConfig) {
>   new_config.rtp.nack.rtp_history_ms = 300 + 20;
>   new_config.rtp.extensions.clear();
>   new_config.rtp.extensions.push_back(
>-        RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId + 1));
>-  new_config.rtp.extensions.push_back(RtpExtension(
>-        RtpExtension::kTransportSequenceNumberUri,
>-        kTransportSequenceNumberId + 1));
>+      RtpExtension(RtpExtension::kAudioLevelUri, kAudioLevelId + 1));
>+  new_config.rtp.extensions.push_back(
>+      RtpExtension(RtpExtension::kTransportSequenceNumberUri,
>+                   kTransportSequenceNumberId + 1));
>   new_config.decoder_map.emplace(1, SdpAudioFormat("foo", 8000, 1));
> 
>   MockVoEChannelProxy& channel_proxy = *helper.channel_proxy();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_send_stream.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_send_stream.cc
>index 053302e22b8..0f725c4da17 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_send_stream.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_send_stream.cc
>@@ -14,21 +14,21 @@
> #include <utility>
> #include <vector>
> 
>+#include "absl/memory/memory.h"
>+
> #include "audio/audio_state.h"
>+#include "audio/channel_proxy.h"
> #include "audio/conversion.h"
> #include "call/rtp_transport_controller_send_interface.h"
> #include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
>-#include "modules/bitrate_controller/include/bitrate_controller.h"
>-#include "modules/congestion_controller/include/send_side_congestion_controller.h"
>-#include "modules/pacing/paced_sender.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/event.h"
> #include "rtc_base/function_view.h"
> #include "rtc_base/logging.h"
>+#include "rtc_base/strings/audio_format_to_string.h"
> #include "rtc_base/task_queue.h"
> #include "rtc_base/timeutils.h"
> #include "system_wrappers/include/field_trial.h"
>-#include "voice_engine/channel_proxy.h"
> 
> namespace webrtc {
> namespace internal {
>@@ -49,15 +49,15 @@ void CallEncoder(const std::unique_ptr<voe::ChannelProxy>& channel_proxy,
> std::unique_ptr<voe::ChannelProxy> CreateChannelAndProxy(
>     webrtc::AudioState* audio_state,
>     rtc::TaskQueue* worker_queue,
>-    ProcessThread* module_process_thread) {
>+    ProcessThread* module_process_thread,
>+    RtcpRttStats* rtcp_rtt_stats,
>+    RtcEventLog* event_log) {
>   RTC_DCHECK(audio_state);
>   internal::AudioState* internal_audio_state =
>       static_cast<internal::AudioState*>(audio_state);
>-  return std::unique_ptr<voe::ChannelProxy>(new voe::ChannelProxy(
>-      std::unique_ptr<voe::Channel>(new voe::Channel(
>-          worker_queue,
>-          module_process_thread,
>-          internal_audio_state->audio_device_module()))));
>+  return absl::make_unique<voe::ChannelProxy>(absl::make_unique<voe::Channel>(
>+      worker_queue, module_process_thread,
>+      internal_audio_state->audio_device_module(), rtcp_rtt_stats, event_log));
> }
> }  // namespace
> 
>@@ -93,7 +93,7 @@ AudioSendStream::AudioSendStream(
>     BitrateAllocator* bitrate_allocator,
>     RtcEventLog* event_log,
>     RtcpRttStats* rtcp_rtt_stats,
>-    const rtc::Optional<RtpState>& suspended_rtp_state,
>+    const absl::optional<RtpState>& suspended_rtp_state,
>     TimeInterval* overall_call_lifetime)
>     : AudioSendStream(config,
>                       audio_state,
>@@ -106,7 +106,9 @@ AudioSendStream::AudioSendStream(
>                       overall_call_lifetime,
>                       CreateChannelAndProxy(audio_state.get(),
>                                             worker_queue,
>-                                            module_process_thread)) {}
>+                                            module_process_thread,
>+                                            rtcp_rtt_stats,
>+                                            event_log)) {}
> 
> AudioSendStream::AudioSendStream(
>     const webrtc::AudioSendStream::Config& config,
>@@ -116,7 +118,7 @@ AudioSendStream::AudioSendStream(
>     BitrateAllocator* bitrate_allocator,
>     RtcEventLog* event_log,
>     RtcpRttStats* rtcp_rtt_stats,
>-    const rtc::Optional<RtpState>& suspended_rtp_state,
>+    const absl::optional<RtpState>& suspended_rtp_state,
>     TimeInterval* overall_call_lifetime,
>     std::unique_ptr<voe::ChannelProxy> channel_proxy)
>     : worker_queue_(worker_queue),
>@@ -132,38 +134,32 @@ AudioSendStream::AudioSendStream(
>       rtp_rtcp_module_(nullptr),
>       suspended_rtp_state_(suspended_rtp_state),
>       overall_call_lifetime_(overall_call_lifetime) {
>-  RTC_LOG(LS_INFO) << "AudioSendStream: " << config.ToString();
>+  RTC_LOG(LS_INFO) << "AudioSendStream: " << config.rtp.ssrc;
>   RTC_DCHECK(worker_queue_);
>   RTC_DCHECK(audio_state_);
>   RTC_DCHECK(channel_proxy_);
>   RTC_DCHECK(bitrate_allocator_);
>   RTC_DCHECK(transport);
>-  RTC_DCHECK(transport->send_side_cc());
>   RTC_DCHECK(overall_call_lifetime_);
> 
>-  channel_proxy_->SetRtcEventLog(event_log_);
>-  channel_proxy_->SetRtcpRttStats(rtcp_rtt_stats);
>   channel_proxy_->SetRTCPStatus(true);
>-  RtpReceiver* rtpReceiver = nullptr;  // Unused, but required for call.
>-  channel_proxy_->GetRtpRtcp(&rtp_rtcp_module_, &rtpReceiver);
>+  rtp_rtcp_module_ = channel_proxy_->GetRtpRtcp();
>   RTC_DCHECK(rtp_rtcp_module_);
> 
>   ConfigureStream(this, config, true);
> 
>   pacer_thread_checker_.DetachFromThread();
>   // Signal congestion controller this object is ready for OnPacket* callbacks.
>-  transport_->send_side_cc()->RegisterPacketFeedbackObserver(this);
>+  transport_->RegisterPacketFeedbackObserver(this);
> }
> 
> AudioSendStream::~AudioSendStream() {
>   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  RTC_LOG(LS_INFO) << "~AudioSendStream: " << config_.ToString();
>+  RTC_LOG(LS_INFO) << "~AudioSendStream: " << config_.rtp.ssrc;
>   RTC_DCHECK(!sending_);
>-  transport_->send_side_cc()->DeRegisterPacketFeedbackObserver(this);
>+  transport_->DeRegisterPacketFeedbackObserver(this);
>   channel_proxy_->RegisterTransport(nullptr);
>   channel_proxy_->ResetSenderCongestionControlObjects();
>-  channel_proxy_->SetRtcEventLog(nullptr);
>-  channel_proxy_->SetRtcpRttStats(nullptr);
>   // Lifetime can only be updated after deregistering
>   // |timed_send_transport_adapter_| in the underlying channel object to avoid
>   // data races in |active_lifetime_|.
>@@ -189,6 +185,8 @@ AudioSendStream::ExtensionIds AudioSendStream::FindExtensionIds(
>       ids.audio_level = extension.id;
>     } else if (extension.uri == RtpExtension::kTransportSequenceNumberUri) {
>       ids.transport_sequence_number = extension.id;
>+    } else if (extension.uri == RtpExtension::kMidUri) {
>+      ids.mid = extension.id;
>     }
>   }
>   return ids;
>@@ -198,7 +196,8 @@ void AudioSendStream::ConfigureStream(
>     webrtc::internal::AudioSendStream* stream,
>     const webrtc::AudioSendStream::Config& new_config,
>     bool first_time) {
>-  RTC_LOG(LS_INFO) << "AudioSendStream::Configuring: " << new_config.ToString();
>+  RTC_LOG(LS_INFO) << "AudioSendStream::ConfigureStream: "
>+                   << new_config.ToString();
>   const auto& channel_proxy = stream->channel_proxy_;
>   const auto& old_config = stream->config_;
> 
>@@ -219,8 +218,7 @@ void AudioSendStream::ConfigureStream(
>                                  new_config.rtp.nack.rtp_history_ms / 20);
>   }
> 
>-  if (first_time ||
>-      new_config.send_transport != old_config.send_transport) {
>+  if (first_time || new_config.send_transport != old_config.send_transport) {
>     if (old_config.send_transport) {
>       channel_proxy->RegisterTransport(nullptr);
>     }
>@@ -243,28 +241,38 @@ void AudioSendStream::ConfigureStream(
>   }
>   bool transport_seq_num_id_changed =
>       new_ids.transport_sequence_number != old_ids.transport_sequence_number;
>-  if (first_time || transport_seq_num_id_changed) {
>+  if (first_time ||
>+      (transport_seq_num_id_changed &&
>+       !webrtc::field_trial::IsEnabled("WebRTC-Audio-ForceNoTWCC"))) {
>     if (!first_time) {
>       channel_proxy->ResetSenderCongestionControlObjects();
>     }
> 
>     RtcpBandwidthObserver* bandwidth_observer = nullptr;
>-    bool has_transport_sequence_number = new_ids.transport_sequence_number != 0;
>+    bool has_transport_sequence_number =
>+        new_ids.transport_sequence_number != 0 &&
>+        !webrtc::field_trial::IsEnabled("WebRTC-Audio-ForceNoTWCC");
>     if (has_transport_sequence_number) {
>       channel_proxy->EnableSendTransportSequenceNumber(
>           new_ids.transport_sequence_number);
>       // Probing in application limited region is only used in combination with
>       // send side congestion control, wich depends on feedback packets which
>       // requires transport sequence numbers to be enabled.
>-      stream->transport_->send_side_cc()->EnablePeriodicAlrProbing(true);
>-      bandwidth_observer =
>-          stream->transport_->send_side_cc()->GetBandwidthObserver();
>+      stream->transport_->EnablePeriodicAlrProbing(true);
>+      bandwidth_observer = stream->transport_->GetBandwidthObserver();
>     }
> 
>     channel_proxy->RegisterSenderCongestionControlObjects(stream->transport_,
>                                                           bandwidth_observer);
>   }
> 
>+  // MID RTP header extension.
>+  if ((first_time || new_ids.mid != old_ids.mid ||
>+       new_config.rtp.mid != old_config.rtp.mid) &&
>+      new_ids.mid != 0 && !new_config.rtp.mid.empty()) {
>+    channel_proxy->SetMid(new_config.rtp.mid, new_ids.mid);
>+  }
>+
>   if (!ReconfigureSendCodec(stream, new_config)) {
>     RTC_LOG(LS_ERROR) << "Failed to set up send codec state.";
>   }
>@@ -281,14 +289,18 @@ void AudioSendStream::Start() {
>     return;
>   }
> 
>+  bool has_transport_sequence_number =
>+      FindExtensionIds(config_.rtp.extensions).transport_sequence_number != 0 &&
>+      !webrtc::field_trial::IsEnabled("WebRTC-Audio-ForceNoTWCC");
>   if (config_.min_bitrate_bps != -1 && config_.max_bitrate_bps != -1 &&
>-      (FindExtensionIds(config_.rtp.extensions).transport_sequence_number !=
>-           0 ||
>-       !webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe"))) {
>+      (has_transport_sequence_number ||
>+       !webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe") ||
>+       webrtc::field_trial::IsEnabled("WebRTC-Audio-ABWENoTWCC"))) {
>     // Audio BWE is enabled.
>     transport_->packet_sender()->SetAccountForAudioPackets(true);
>     ConfigureBitrateObserver(config_.min_bitrate_bps, config_.max_bitrate_bps,
>-                             config_.bitrate_priority);
>+                             config_.bitrate_priority,
>+                             has_transport_sequence_number);
>   }
>   channel_proxy_->StartSend();
>   sending_ = true;
>@@ -314,7 +326,8 @@ void AudioSendStream::SendAudioData(std::unique_ptr<AudioFrame> audio_frame) {
> }
> 
> bool AudioSendStream::SendTelephoneEvent(int payload_type,
>-                                         int payload_frequency, int event,
>+                                         int payload_frequency,
>+                                         int event,
>                                          int duration_ms) {
>   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>   return channel_proxy_->SetSendTelephoneEventPayloadType(payload_type,
>@@ -397,14 +410,19 @@ uint32_t AudioSendStream::OnBitrateUpdated(uint32_t bitrate_bps,
>                                            uint8_t fraction_loss,
>                                            int64_t rtt,
>                                            int64_t bwe_period_ms) {
>+  // Audio transport feedback will not be reported in this mode, instead update
>+  // acknowledged bitrate estimator with the bitrate allocated for audio.
>+  if (webrtc::field_trial::IsEnabled("WebRTC-Audio-ABWENoTWCC")) {
>+    transport_->SetAllocatedBitrateWithoutFeedback(bitrate_bps);
>+  }
>+
>   // A send stream may be allocated a bitrate of zero if the allocator decides
>   // to disable it. For now we ignore this decision and keep sending on min
>   // bitrate.
>   if (bitrate_bps == 0) {
>     bitrate_bps = config_.min_bitrate_bps;
>   }
>-  RTC_DCHECK_GE(bitrate_bps,
>-                static_cast<uint32_t>(config_.min_bitrate_bps));
>+  RTC_DCHECK_GE(bitrate_bps, static_cast<uint32_t>(config_.min_bitrate_bps));
>   // The bitrate allocator might allocate an higher than max configured bitrate
>   // if there is room, to allow for, as example, extra FEC. Ignore that for now.
>   const uint32_t max_bitrate_bps = config_.max_bitrate_bps;
>@@ -433,8 +451,8 @@ void AudioSendStream::OnPacketAdded(uint32_t ssrc, uint16_t seq_num) {
> void AudioSendStream::OnPacketFeedbackVector(
>     const std::vector<PacketFeedback>& packet_feedback_vector) {
>   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  rtc::Optional<float> plr;
>-  rtc::Optional<float> rplr;
>+  absl::optional<float> plr;
>+  absl::optional<float> rplr;
>   {
>     rtc::CritScope lock(&packet_loss_tracker_cs_);
>     packet_loss_tracker_.OnPacketFeedbackVector(packet_feedback_vector);
>@@ -454,8 +472,6 @@ void AudioSendStream::OnPacketFeedbackVector(
> 
> void AudioSendStream::SetTransportOverhead(int transport_overhead_per_packet) {
>   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  transport_->send_side_cc()->SetTransportOverhead(
>-      transport_overhead_per_packet);
>   channel_proxy_->SetTransportOverhead(transport_overhead_per_packet);
> }
> 
>@@ -501,16 +517,25 @@ bool AudioSendStream::SetupSendCodec(AudioSendStream* stream,
> 
>   RTC_DCHECK(new_config.encoder_factory);
>   std::unique_ptr<AudioEncoder> encoder =
>-      new_config.encoder_factory->MakeAudioEncoder(spec.payload_type,
>-                                                   spec.format);
>+      new_config.encoder_factory->MakeAudioEncoder(
>+          spec.payload_type, spec.format, new_config.codec_pair_id);
> 
>   if (!encoder) {
>-    RTC_LOG(LS_ERROR) << "Unable to create encoder for " << spec.format;
>+    RTC_DLOG(LS_ERROR) << "Unable to create encoder for "
>+                       << rtc::ToString(spec.format);
>     return false;
>   }
>+
>+  // If other side does not support audio TWCC and WebRTC-Audio-ABWENoTWCC is
>+  // not enabled, do not update target audio bitrate if we are in
>+  // WebRTC-Audio-SendSideBwe-For-Video experiment
>+  const bool do_not_update_target_bitrate =
>+      !webrtc::field_trial::IsEnabled("WebRTC-Audio-ABWENoTWCC") &&
>+      webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe-For-Video") &&
>+      !FindExtensionIds(new_config.rtp.extensions).transport_sequence_number;
>   // If a bitrate has been specified for the codec, use it over the
>   // codec's default.
>-  if (spec.target_bitrate_bps) {
>+  if (!do_not_update_target_bitrate && spec.target_bitrate_bps) {
>     encoder->OnReceivedTargetAudioBitrate(*spec.target_bitrate_bps);
>   }
> 
>@@ -518,8 +543,8 @@ bool AudioSendStream::SetupSendCodec(AudioSendStream* stream,
>   if (new_config.audio_network_adaptor_config) {
>     if (encoder->EnableAudioNetworkAdaptor(
>             *new_config.audio_network_adaptor_config, stream->event_log_)) {
>-      RTC_LOG(LS_INFO) << "Audio network adaptor enabled on SSRC "
>-                       << new_config.rtp.ssrc;
>+      RTC_DLOG(LS_INFO) << "Audio network adaptor enabled on SSRC "
>+                        << new_config.rtp.ssrc;
>     } else {
>       RTC_NOTREACHED();
>     }
>@@ -573,11 +598,19 @@ bool AudioSendStream::ReconfigureSendCodec(AudioSendStream* stream,
>     return SetupSendCodec(stream, new_config);
>   }
> 
>-  const rtc::Optional<int>& new_target_bitrate_bps =
>+  // If other side does not support audio TWCC and WebRTC-Audio-ABWENoTWCC is
>+  // not enabled, do not update target audio bitrate if we are in
>+  // WebRTC-Audio-SendSideBwe-For-Video experiment
>+  const bool do_not_update_target_bitrate =
>+      !webrtc::field_trial::IsEnabled("WebRTC-Audio-ABWENoTWCC") &&
>+      webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe-For-Video") &&
>+      !FindExtensionIds(new_config.rtp.extensions).transport_sequence_number;
>+
>+  const absl::optional<int>& new_target_bitrate_bps =
>       new_config.send_codec_spec->target_bitrate_bps;
>   // If a bitrate has been specified for the codec, use it over the
>   // codec's default.
>-  if (new_target_bitrate_bps &&
>+  if (!do_not_update_target_bitrate && new_target_bitrate_bps &&
>       new_target_bitrate_bps !=
>           old_config.send_codec_spec->target_bitrate_bps) {
>     CallEncoder(stream->channel_proxy_, [&](AudioEncoder* encoder) {
>@@ -601,8 +634,8 @@ void AudioSendStream::ReconfigureANA(AudioSendStream* stream,
>     CallEncoder(stream->channel_proxy_, [&](AudioEncoder* encoder) {
>       if (encoder->EnableAudioNetworkAdaptor(
>               *new_config.audio_network_adaptor_config, stream->event_log_)) {
>-        RTC_LOG(LS_INFO) << "Audio network adaptor enabled on SSRC "
>-                         << new_config.rtp.ssrc;
>+        RTC_DLOG(LS_INFO) << "Audio network adaptor enabled on SSRC "
>+                          << new_config.rtp.ssrc;
>       } else {
>         RTC_NOTREACHED();
>       }
>@@ -611,8 +644,8 @@ void AudioSendStream::ReconfigureANA(AudioSendStream* stream,
>     CallEncoder(stream->channel_proxy_, [&](AudioEncoder* encoder) {
>       encoder->DisableAudioNetworkAdaptor();
>     });
>-    RTC_LOG(LS_INFO) << "Audio network adaptor disabled on SSRC "
>-                     << new_config.rtp.ssrc;
>+    RTC_DLOG(LS_INFO) << "Audio network adaptor disabled on SSRC "
>+                      << new_config.rtp.ssrc;
>   }
> }
> 
>@@ -676,12 +709,13 @@ void AudioSendStream::ReconfigureBitrateObserver(
>     return;
>   }
> 
>+  bool has_transport_sequence_number = new_transport_seq_num_id != 0;
>   if (new_config.min_bitrate_bps != -1 && new_config.max_bitrate_bps != -1 &&
>-      (new_transport_seq_num_id != 0 ||
>+      (has_transport_sequence_number ||
>        !webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe"))) {
>-    stream->ConfigureBitrateObserver(new_config.min_bitrate_bps,
>-                                     new_config.max_bitrate_bps,
>-                                     new_config.bitrate_priority);
>+    stream->ConfigureBitrateObserver(
>+        new_config.min_bitrate_bps, new_config.max_bitrate_bps,
>+        new_config.bitrate_priority, has_transport_sequence_number);
>   } else {
>     stream->RemoveBitrateObserver();
>   }
>@@ -689,7 +723,8 @@ void AudioSendStream::ReconfigureBitrateObserver(
> 
> void AudioSendStream::ConfigureBitrateObserver(int min_bitrate_bps,
>                                                int max_bitrate_bps,
>-                                               double bitrate_priority) {
>+                                               double bitrate_priority,
>+                                               bool has_packet_feedback) {
>   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>   RTC_DCHECK_GE(max_bitrate_bps, min_bitrate_bps);
>   rtc::Event thread_sync_event(false /* manual_reset */, false);
>@@ -700,8 +735,11 @@ void AudioSendStream::ConfigureBitrateObserver(int min_bitrate_bps,
>     config_.max_bitrate_bps = max_bitrate_bps;
>     config_.bitrate_priority = bitrate_priority;
>     // This either updates the current observer or adds a new observer.
>-    bitrate_allocator_->AddObserver(this, min_bitrate_bps, max_bitrate_bps, 0,
>-                                    true, config_.track_id, bitrate_priority);
>+    bitrate_allocator_->AddObserver(
>+        this, MediaStreamAllocationConfig{
>+                  static_cast<uint32_t>(min_bitrate_bps),
>+                  static_cast<uint32_t>(max_bitrate_bps), 0, true,
>+                  config_.track_id, bitrate_priority, has_packet_feedback});
>     thread_sync_event.Set();
>   });
>   thread_sync_event.Wait(rtc::Event::kForever);
>@@ -723,8 +761,8 @@ void AudioSendStream::RegisterCngPayloadType(int payload_type,
>   if (rtp_rtcp_module_->RegisterSendPayload(codec) != 0) {
>     rtp_rtcp_module_->DeRegisterSendPayload(codec.pltype);
>     if (rtp_rtcp_module_->RegisterSendPayload(codec) != 0) {
>-      RTC_LOG(LS_ERROR) << "RegisterCngPayloadType() failed to register CN to "
>-                           "RTP/RTCP module";
>+      RTC_DLOG(LS_ERROR) << "RegisterCngPayloadType() failed to register CN to "
>+                            "RTP/RTCP module";
>     }
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_send_stream.h b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_send_stream.h
>index b2bcac54537..ba87202bbe3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_send_stream.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_send_stream.h
>@@ -15,6 +15,7 @@
> #include <vector>
> 
> #include "audio/time_interval.h"
>+#include "audio/transport_feedback_packet_loss_tracker.h"
> #include "call/audio_send_stream.h"
> #include "call/audio_state.h"
> #include "call/bitrate_allocator.h"
>@@ -22,7 +23,6 @@
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/race_checker.h"
> #include "rtc_base/thread_checker.h"
>-#include "voice_engine/transport_feedback_packet_loss_tracker.h"
> 
> namespace webrtc {
> class RtcEventLog;
>@@ -49,7 +49,7 @@ class AudioSendStream final : public webrtc::AudioSendStream,
>                   BitrateAllocator* bitrate_allocator,
>                   RtcEventLog* event_log,
>                   RtcpRttStats* rtcp_rtt_stats,
>-                  const rtc::Optional<RtpState>& suspended_rtp_state,
>+                  const absl::optional<RtpState>& suspended_rtp_state,
>                   TimeInterval* overall_call_lifetime);
>   // For unit tests, which need to supply a mock channel proxy.
>   AudioSendStream(const webrtc::AudioSendStream::Config& config,
>@@ -59,7 +59,7 @@ class AudioSendStream final : public webrtc::AudioSendStream,
>                   BitrateAllocator* bitrate_allocator,
>                   RtcEventLog* event_log,
>                   RtcpRttStats* rtcp_rtt_stats,
>-                  const rtc::Optional<RtpState>& suspended_rtp_state,
>+                  const absl::optional<RtpState>& suspended_rtp_state,
>                   TimeInterval* overall_call_lifetime,
>                   std::unique_ptr<voe::ChannelProxy> channel_proxy);
>   ~AudioSendStream() override;
>@@ -70,7 +70,9 @@ class AudioSendStream final : public webrtc::AudioSendStream,
>   void Start() override;
>   void Stop() override;
>   void SendAudioData(std::unique_ptr<AudioFrame> audio_frame) override;
>-  bool SendTelephoneEvent(int payload_type, int payload_frequency, int event,
>+  bool SendTelephoneEvent(int payload_type,
>+                          int payload_frequency,
>+                          int event,
>                           int duration_ms) override;
>   void SetMuted(bool muted) override;
>   webrtc::AudioSendStream::Stats GetStats() const override;
>@@ -119,7 +121,8 @@ class AudioSendStream final : public webrtc::AudioSendStream,
> 
>   void ConfigureBitrateObserver(int min_bitrate_bps,
>                                 int max_bitrate_bps,
>-                                double bitrate_priority);
>+                                double bitrate_priority,
>+                                bool has_packet_feedback);
>   void RemoveBitrateObserver();
> 
>   void RegisterCngPayloadType(int payload_type, int clockrate_hz);
>@@ -145,7 +148,7 @@ class AudioSendStream final : public webrtc::AudioSendStream,
>       RTC_GUARDED_BY(&packet_loss_tracker_cs_);
> 
>   RtpRtcp* rtp_rtcp_module_;
>-  rtc::Optional<RtpState> const suspended_rtp_state_;
>+  absl::optional<RtpState> const suspended_rtp_state_;
> 
>   std::unique_ptr<TimedTransport> timed_send_transport_adapter_;
>   TimeInterval active_lifetime_;
>@@ -157,6 +160,7 @@ class AudioSendStream final : public webrtc::AudioSendStream,
>   struct ExtensionIds {
>     int audio_level = 0;
>     int transport_sequence_number = 0;
>+    int mid = 0;
>   };
>   static ExtensionIds FindExtensionIds(
>       const std::vector<RtpExtension>& extensions);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_send_stream_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_send_stream_tests.cc
>index 3f96c3350ff..7deeff3b4d1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_send_stream_tests.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_send_stream_tests.cc
>@@ -20,15 +20,9 @@ class AudioSendTest : public SendTest {
>  public:
>   AudioSendTest() : SendTest(CallTest::kDefaultTimeoutMs) {}
> 
>-  size_t GetNumVideoStreams() const override {
>-    return 0;
>-  }
>-  size_t GetNumAudioStreams() const override {
>-    return 1;
>-  }
>-  size_t GetNumFlexfecStreams() const override {
>-    return 0;
>-  }
>+  size_t GetNumVideoStreams() const override { return 0; }
>+  size_t GetNumAudioStreams() const override { return 1; }
>+  size_t GetNumFlexfecStreams() const override { return 0; }
> };
> }  // namespace
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_send_stream_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_send_stream_unittest.cc
>index 109f15d1b2a..f5d6796af5e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_send_stream_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_send_stream_unittest.cc
>@@ -12,30 +12,27 @@
> #include <utility>
> #include <vector>
> 
>+#include "absl/memory/memory.h"
>+#include "api/units/time_delta.h"
> #include "audio/audio_send_stream.h"
> #include "audio/audio_state.h"
> #include "audio/conversion.h"
>-#include "call/fake_rtp_transport_controller_send.h"
>-#include "call/rtp_transport_controller_send_interface.h"
>+#include "audio/mock_voe_channel_proxy.h"
>+#include "call/test/mock_rtp_transport_controller_send.h"
> #include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
> #include "modules/audio_device/include/mock_audio_device.h"
> #include "modules/audio_mixer/audio_mixer_impl.h"
> #include "modules/audio_processing/include/audio_processing_statistics.h"
> #include "modules/audio_processing/include/mock_audio_processing.h"
>-#include "modules/congestion_controller/include/mock/mock_congestion_observer.h"
>-#include "modules/congestion_controller/include/send_side_congestion_controller.h"
>-#include "modules/pacing/mock/mock_paced_sender.h"
>+#include "modules/rtp_rtcp/mocks/mock_rtcp_bandwidth_observer.h"
> #include "modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h"
> #include "modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
> #include "rtc_base/fakeclock.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/task_queue.h"
>-#include "rtc_base/timedelta.h"
> #include "test/gtest.h"
> #include "test/mock_audio_encoder.h"
> #include "test/mock_audio_encoder_factory.h"
> #include "test/mock_transport.h"
>-#include "test/mock_voe_channel_proxy.h"
> 
> namespace webrtc {
> namespace test {
>@@ -59,8 +56,8 @@ const double kEchoReturnLoss = -65;
> const double kEchoReturnLossEnhancement = 101;
> const double kResidualEchoLikelihood = -1.0f;
> const double kResidualEchoLikelihoodMax = 23.0f;
>-const CallStatistics kCallStats = {
>-    1345,  1678,  1901, 1234,  112, 13456, 17890, 1567, -1890, -1123};
>+const CallStatistics kCallStats = {1345,  1678,  1901, 1234,  112,
>+                                   13456, 17890, 1567, -1890, -1123};
> const ReportBlock kReportBlock = {456, 780, 123, 567, 890, 132, 143, 13354};
> const int kTelephoneEventPayloadType = 123;
> const int kTelephoneEventPayloadFrequency = 65432;
>@@ -78,9 +75,11 @@ const AudioCodecSpec kCodecSpecs[] = {
> 
> class MockLimitObserver : public BitrateAllocator::LimitObserver {
>  public:
>-  MOCK_METHOD2(OnAllocationLimitsChanged,
>+  MOCK_METHOD4(OnAllocationLimitsChanged,
>                void(uint32_t min_send_bitrate_bps,
>-                    uint32_t max_padding_bitrate_bps));
>+                    uint32_t max_padding_bitrate_bps,
>+                    uint32_t total_bitrate_bps,
>+                    bool has_packet_feedback));
> };
> 
> std::unique_ptr<MockAudioEncoder> SetupAudioEncoderMock(
>@@ -88,7 +87,8 @@ std::unique_ptr<MockAudioEncoder> SetupAudioEncoderMock(
>     const SdpAudioFormat& format) {
>   for (const auto& spec : kCodecSpecs) {
>     if (format == spec.format) {
>-      std::unique_ptr<MockAudioEncoder> encoder(new MockAudioEncoder);
>+      std::unique_ptr<MockAudioEncoder> encoder(
>+          new testing::NiceMock<MockAudioEncoder>());
>       ON_CALL(*encoder.get(), SampleRateHz())
>           .WillByDefault(Return(spec.info.sample_rate_hz));
>       ON_CALL(*encoder.get(), NumChannels())
>@@ -109,16 +109,17 @@ rtc::scoped_refptr<MockAudioEncoderFactory> SetupEncoderFactoryMock() {
>           std::begin(kCodecSpecs), std::end(kCodecSpecs))));
>   ON_CALL(*factory.get(), QueryAudioEncoder(_))
>       .WillByDefault(Invoke(
>-          [](const SdpAudioFormat& format) -> rtc::Optional<AudioCodecInfo> {
>+          [](const SdpAudioFormat& format) -> absl::optional<AudioCodecInfo> {
>             for (const auto& spec : kCodecSpecs) {
>               if (format == spec.format) {
>                 return spec.info;
>               }
>             }
>-            return rtc::nullopt;
>+            return absl::nullopt;
>           }));
>-  ON_CALL(*factory.get(), MakeAudioEncoderMock(_, _, _))
>+  ON_CALL(*factory.get(), MakeAudioEncoderMock(_, _, _, _))
>       .WillByDefault(Invoke([](int payload_type, const SdpAudioFormat& format,
>+                               absl::optional<AudioCodecPairId> codec_pair_id,
>                                std::unique_ptr<AudioEncoder>* return_value) {
>         *return_value = SetupAudioEncoderMock(payload_type, format);
>       }));
>@@ -129,13 +130,6 @@ struct ConfigHelper {
>   ConfigHelper(bool audio_bwe_enabled, bool expect_set_encoder_call)
>       : stream_config_(nullptr),
>         audio_processing_(new rtc::RefCountedObject<MockAudioProcessing>()),
>-        simulated_clock_(123456),
>-        send_side_cc_(rtc::MakeUnique<SendSideCongestionController>(
>-            &simulated_clock_,
>-            nullptr /* observer */,
>-            &event_log_,
>-            &pacer_)),
>-        fake_transport_(&packet_router_, &pacer_, send_side_cc_.get()),
>         bitrate_allocator_(&limit_observer_),
>         worker_queue_("ConfigHelper_worker_queue"),
>         audio_encoder_(nullptr) {
>@@ -171,8 +165,8 @@ struct ConfigHelper {
>   std::unique_ptr<internal::AudioSendStream> CreateAudioSendStream() {
>     return std::unique_ptr<internal::AudioSendStream>(
>         new internal::AudioSendStream(
>-            stream_config_, audio_state_, &worker_queue_, &fake_transport_,
>-            &bitrate_allocator_, &event_log_, &rtcp_rtt_stats_, rtc::nullopt,
>+            stream_config_, audio_state_, &worker_queue_, &rtp_transport_,
>+            &bitrate_allocator_, &event_log_, &rtcp_rtt_stats_, absl::nullopt,
>             &active_lifetime_,
>             std::unique_ptr<voe::ChannelProxy>(channel_proxy_)));
>   }
>@@ -183,24 +177,21 @@ struct ConfigHelper {
>         stream_config_.encoder_factory.get());
>   }
>   MockVoEChannelProxy* channel_proxy() { return channel_proxy_; }
>-  RtpTransportControllerSendInterface* transport() { return &fake_transport_; }
>+  RtpTransportControllerSendInterface* transport() { return &rtp_transport_; }
>   TimeInterval* active_lifetime() { return &active_lifetime_; }
> 
>   static void AddBweToConfig(AudioSendStream::Config* config) {
>-    config->rtp.extensions.push_back(
>-        RtpExtension(RtpExtension::kTransportSequenceNumberUri,
>-                     kTransportSequenceNumberId));
>+    config->rtp.extensions.push_back(RtpExtension(
>+        RtpExtension::kTransportSequenceNumberUri, kTransportSequenceNumberId));
>     config->send_codec_spec->transport_cc_enabled = true;
>   }
> 
>   void SetupDefaultChannelProxy(bool audio_bwe_enabled) {
>+    EXPECT_TRUE(channel_proxy_ == nullptr);
>     channel_proxy_ = new testing::StrictMock<MockVoEChannelProxy>();
>-    EXPECT_CALL(*channel_proxy_, GetRtpRtcp(_, _))
>-        .WillRepeatedly(Invoke(
>-            [this](RtpRtcp** rtp_rtcp_module, RtpReceiver** rtp_receiver) {
>-              *rtp_rtcp_module = &this->rtp_rtcp_;
>-              *rtp_receiver = nullptr;  // Not deemed necessary for tests yet.
>-            }));
>+    EXPECT_CALL(*channel_proxy_, GetRtpRtcp()).WillRepeatedly(Invoke([this]() {
>+      return &this->rtp_rtcp_;
>+    }));
>     EXPECT_CALL(*channel_proxy_, SetRTCPStatus(true)).Times(1);
>     EXPECT_CALL(*channel_proxy_, SetLocalSSRC(kSsrc)).Times(1);
>     EXPECT_CALL(*channel_proxy_, SetRTCP_CNAME(StrEq(kCName))).Times(1);
>@@ -208,16 +199,19 @@ struct ConfigHelper {
>     EXPECT_CALL(*channel_proxy_,
>                 SetSendAudioLevelIndicationStatus(true, kAudioLevelId))
>         .Times(1);
>+    EXPECT_CALL(rtp_transport_, GetBandwidthObserver())
>+        .WillRepeatedly(Return(&bandwidth_observer_));
>     if (audio_bwe_enabled) {
>       EXPECT_CALL(*channel_proxy_,
>                   EnableSendTransportSequenceNumber(kTransportSequenceNumberId))
>           .Times(1);
>-      EXPECT_CALL(*channel_proxy_, RegisterSenderCongestionControlObjects(
>-                                       &fake_transport_, Ne(nullptr)))
>+      EXPECT_CALL(*channel_proxy_,
>+                  RegisterSenderCongestionControlObjects(
>+                      &rtp_transport_, Eq(&bandwidth_observer_)))
>           .Times(1);
>     } else {
>       EXPECT_CALL(*channel_proxy_, RegisterSenderCongestionControlObjects(
>-                                       &fake_transport_, Eq(nullptr)))
>+                                       &rtp_transport_, Eq(nullptr)))
>           .Times(1);
>     }
>     EXPECT_CALL(*channel_proxy_, ResetSenderCongestionControlObjects())
>@@ -227,12 +221,6 @@ struct ConfigHelper {
>       EXPECT_CALL(*channel_proxy_, RegisterTransport(_)).Times(1);
>       EXPECT_CALL(*channel_proxy_, RegisterTransport(nullptr)).Times(1);
>     }
>-    EXPECT_CALL(*channel_proxy_, SetRtcEventLog(testing::NotNull())).Times(1);
>-    EXPECT_CALL(*channel_proxy_, SetRtcEventLog(testing::IsNull()))
>-        .Times(1);  // Destructor resets the event log
>-    EXPECT_CALL(*channel_proxy_, SetRtcpRttStats(&rtcp_rtt_stats_)).Times(1);
>-    EXPECT_CALL(*channel_proxy_, SetRtcpRttStats(testing::IsNull()))
>-        .Times(1);  // Destructor resets the rtt stats.
>   }
> 
>   void SetupMockForSetupSendCodec(bool expect_set_encoder_call) {
>@@ -259,13 +247,14 @@ struct ConfigHelper {
> 
>   void SetupMockForSendTelephoneEvent() {
>     EXPECT_TRUE(channel_proxy_);
>-    EXPECT_CALL(*channel_proxy_,
>-        SetSendTelephoneEventPayloadType(kTelephoneEventPayloadType,
>-                                         kTelephoneEventPayloadFrequency))
>-            .WillOnce(Return(true));
>-    EXPECT_CALL(*channel_proxy_,
>+    EXPECT_CALL(*channel_proxy_, SetSendTelephoneEventPayloadType(
>+                                     kTelephoneEventPayloadType,
>+                                     kTelephoneEventPayloadFrequency))
>+        .WillOnce(Return(true));
>+    EXPECT_CALL(
>+        *channel_proxy_,
>         SendTelephoneEventOutband(kTelephoneEventCode, kTelephoneEventDuration))
>-            .WillOnce(Return(true));
>+        .WillOnce(Return(true));
>   }
> 
>   void SetupMockForGetStats() {
>@@ -310,14 +299,11 @@ struct ConfigHelper {
>   testing::StrictMock<MockVoEChannelProxy>* channel_proxy_ = nullptr;
>   rtc::scoped_refptr<MockAudioProcessing> audio_processing_;
>   AudioProcessingStats audio_processing_stats_;
>-  SimulatedClock simulated_clock_;
>   TimeInterval active_lifetime_;
>-  PacketRouter packet_router_;
>-  testing::NiceMock<MockPacedSender> pacer_;
>-  std::unique_ptr<SendSideCongestionController> send_side_cc_;
>-  FakeRtpTransportControllerSend fake_transport_;
>-  MockRtcEventLog event_log_;
>-  MockRtpRtcp rtp_rtcp_;
>+  testing::StrictMock<MockRtcpBandwidthObserver> bandwidth_observer_;
>+  testing::NiceMock<MockRtcEventLog> event_log_;
>+  testing::NiceMock<MockRtpTransportControllerSend> rtp_transport_;
>+  testing::NiceMock<MockRtpRtcp> rtp_rtcp_;
>   MockRtcpRttStats rtcp_rtt_stats_;
>   testing::NiceMock<MockLimitObserver> limit_observer_;
>   BitrateAllocator bitrate_allocator_;
>@@ -363,9 +349,9 @@ TEST(AudioSendStreamTest, SendTelephoneEvent) {
>   ConfigHelper helper(false, true);
>   auto send_stream = helper.CreateAudioSendStream();
>   helper.SetupMockForSendTelephoneEvent();
>-  EXPECT_TRUE(send_stream->SendTelephoneEvent(kTelephoneEventPayloadType,
>-      kTelephoneEventPayloadFrequency, kTelephoneEventCode,
>-      kTelephoneEventDuration));
>+  EXPECT_TRUE(send_stream->SendTelephoneEvent(
>+      kTelephoneEventPayloadType, kTelephoneEventPayloadFrequency,
>+      kTelephoneEventCode, kTelephoneEventDuration));
> }
> 
> TEST(AudioSendStreamTest, SetMuted) {
>@@ -393,8 +379,7 @@ TEST(AudioSendStreamTest, GetStats) {
>   EXPECT_EQ(kSsrc, stats.local_ssrc);
>   EXPECT_EQ(static_cast<int64_t>(kCallStats.bytesSent), stats.bytes_sent);
>   EXPECT_EQ(kCallStats.packetsSent, stats.packets_sent);
>-  EXPECT_EQ(static_cast<int32_t>(kReportBlock.cumulative_num_packets_lost),
>-            stats.packets_lost);
>+  EXPECT_EQ(kReportBlock.cumulative_num_packets_lost, stats.packets_lost);
>   EXPECT_EQ(Q8ToFloat(kReportBlock.fraction_lost), stats.fraction_lost);
>   EXPECT_EQ(std::string(kIsacCodec.plname), stats.codec_name);
>   EXPECT_EQ(static_cast<int32_t>(kReportBlock.extended_highest_sequence_number),
>@@ -429,9 +414,10 @@ TEST(AudioSendStreamTest, SendCodecAppliesAudioNetworkAdaptor) {
> 
>   helper.config().audio_network_adaptor_config = kAnaConfigString;
> 
>-  EXPECT_CALL(helper.mock_encoder_factory(), MakeAudioEncoderMock(_, _, _))
>+  EXPECT_CALL(helper.mock_encoder_factory(), MakeAudioEncoderMock(_, _, _, _))
>       .WillOnce(Invoke([&kAnaConfigString, &kAnaReconfigString](
>                            int payload_type, const SdpAudioFormat& format,
>+                           absl::optional<AudioCodecPairId> codec_pair_id,
>                            std::unique_ptr<AudioEncoder>* return_value) {
>         auto mock_encoder = SetupAudioEncoderMock(payload_type, format);
>         EXPECT_CALL(*mock_encoder,
>@@ -525,7 +511,7 @@ TEST(AudioSendStreamTest, ReconfigureTransportCcResetsFirst) {
>     EXPECT_CALL(*helper.channel_proxy(), ResetSenderCongestionControlObjects())
>         .Times(1);
>     EXPECT_CALL(*helper.channel_proxy(), RegisterSenderCongestionControlObjects(
>-        helper.transport(), Ne(nullptr)))
>+                                             helper.transport(), Ne(nullptr)))
>         .Times(1);
>   }
>   send_stream->Reconfigure(new_config);
>@@ -552,8 +538,7 @@ TEST(AudioSendStreamTest, UpdateLifetime) {
>     EXPECT_CALL(mock_transport, SendRtp(_, _, _)).Times(2);
>     const PacketOptions options;
>     registered_transport->SendRtp(nullptr, 0, options);
>-    fake_clock.AdvanceTime(
>-        rtc::TimeDelta::FromMilliseconds(kTimeBetweenSendRtpCallsMs));
>+    fake_clock.AdvanceTime(TimeDelta::ms(kTimeBetweenSendRtpCallsMs));
>     registered_transport->SendRtp(nullptr, 0, options);
>   }
>   EXPECT_TRUE(!helper.active_lifetime()->Empty());
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_state.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_state.cc
>index 24a7ce801ed..35ea03d3840 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_state.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_state.cc
>@@ -14,12 +14,12 @@
> #include <utility>
> #include <vector>
> 
>+#include "absl/memory/memory.h"
> #include "audio/audio_receive_stream.h"
> #include "modules/audio_device/include/audio_device.h"
> #include "rtc_base/atomicops.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/thread.h"
> 
> namespace webrtc {
>@@ -27,9 +27,7 @@ namespace internal {
> 
> AudioState::AudioState(const AudioState::Config& config)
>     : config_(config),
>-      audio_transport_(config_.audio_mixer,
>-                       config_.audio_processing.get(),
>-                       config_.audio_device_module.get()) {
>+      audio_transport_(config_.audio_mixer, config_.audio_processing.get()) {
>   process_thread_checker_.DetachFromThread();
>   RTC_DCHECK(config_.audio_mixer);
>   RTC_DCHECK(config_.audio_device_module);
>@@ -51,8 +49,8 @@ void AudioState::AddReceivingStream(webrtc::AudioReceiveStream* stream) {
>   RTC_DCHECK_EQ(0, receiving_streams_.count(stream));
>   receiving_streams_.insert(stream);
>   if (!config_.audio_mixer->AddSource(
>-      static_cast<internal::AudioReceiveStream*>(stream))) {
>-    RTC_LOG(LS_ERROR) << "Failed to add source to mixer.";
>+          static_cast<internal::AudioReceiveStream*>(stream))) {
>+    RTC_DLOG(LS_ERROR) << "Failed to add source to mixer.";
>   }
> 
>   // Make sure playback is initialized; start playing if enabled.
>@@ -80,7 +78,8 @@ void AudioState::RemoveReceivingStream(webrtc::AudioReceiveStream* stream) {
> }
> 
> void AudioState::AddSendingStream(webrtc::AudioSendStream* stream,
>-                                  int sample_rate_hz, size_t num_channels) {
>+                                  int sample_rate_hz,
>+                                  size_t num_channels) {
>   RTC_DCHECK(thread_checker_.CalledOnValidThread());
>   auto& properties = sending_streams_[stream];
>   properties.sample_rate_hz = sample_rate_hz;
>@@ -123,7 +122,7 @@ void AudioState::SetPlayout(bool enabled) {
>     } else {
>       config_.audio_device_module->StopPlayout();
>       null_audio_poller_ =
>-          rtc::MakeUnique<NullAudioPoller>(&audio_transport_);
>+          absl::make_unique<NullAudioPoller>(&audio_transport_);
>     }
>   }
> }
>@@ -150,9 +149,6 @@ AudioState::Stats AudioState::GetAudioInputStats() const {
>   result.audio_level = audio_level.LevelFullRange();
>   RTC_DCHECK_LE(0, result.audio_level);
>   RTC_DCHECK_GE(32767, result.audio_level);
>-  result.quantized_audio_level = audio_level.Level();
>-  RTC_DCHECK_LE(0, result.quantized_audio_level);
>-  RTC_DCHECK_GE(9, result.quantized_audio_level);
>   result.total_energy = audio_level.TotalEnergy();
>   result.total_duration = audio_level.TotalDuration();
>   return result;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_state.h b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_state.h
>index d4e4e3fd3ce..689534b0eec 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_state.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_state.h
>@@ -39,9 +39,7 @@ class AudioState final : public webrtc::AudioState {
>     RTC_DCHECK(config_.audio_processing);
>     return config_.audio_processing.get();
>   }
>-  AudioTransport* audio_transport() override {
>-    return &audio_transport_;
>-  }
>+  AudioTransport* audio_transport() override { return &audio_transport_; }
> 
>   void SetPlayout(bool enabled) override;
>   void SetRecording(bool enabled) override;
>@@ -60,7 +58,8 @@ class AudioState final : public webrtc::AudioState {
>   void RemoveReceivingStream(webrtc::AudioReceiveStream* stream);
> 
>   void AddSendingStream(webrtc::AudioSendStream* stream,
>-                        int sample_rate_hz, size_t num_channels);
>+                        int sample_rate_hz,
>+                        size_t num_channels);
>   void RemoveSendingStream(webrtc::AudioSendStream* stream);
> 
>  private:
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_state_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_state_unittest.cc
>index b4452add9a1..dc622df00de 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_state_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_state_unittest.cc
>@@ -71,8 +71,7 @@ std::vector<int16_t> Create10msTestData(int sample_rate_hz,
>   const float inc = (2 * 3.14159265f * 1000) / sample_rate_hz;
>   float w = 0.f;
>   for (int i = 0; i < samples_per_channel; ++i) {
>-    audio_data[i * num_channels] =
>-        static_cast<int16_t>(32767.f * std::sin(w));
>+    audio_data[i * num_channels] = static_cast<int16_t>(32767.f * std::sin(w));
>     w += inc;
>   }
>   return audio_data;
>@@ -111,16 +110,18 @@ TEST(AudioStateTest, RecordedAudioArrivesAtSingleStream) {
>   MockAudioSendStream stream;
>   audio_state->AddSendingStream(&stream, 8000, 2);
> 
>-  EXPECT_CALL(stream, SendAudioDataForMock(testing::AllOf(
>-      testing::Field(&AudioFrame::sample_rate_hz_, testing::Eq(8000)),
>-      testing::Field(&AudioFrame::num_channels_, testing::Eq(2u)))))
>-          .WillOnce(
>-              // Verify that channels are not swapped by default.
>-              testing::Invoke([](AudioFrame* audio_frame) {
>-                auto levels = ComputeChannelLevels(audio_frame);
>-                EXPECT_LT(0u, levels[0]);
>-                EXPECT_EQ(0u, levels[1]);
>-              }));
>+  EXPECT_CALL(
>+      stream,
>+      SendAudioDataForMock(testing::AllOf(
>+          testing::Field(&AudioFrame::sample_rate_hz_, testing::Eq(8000)),
>+          testing::Field(&AudioFrame::num_channels_, testing::Eq(2u)))))
>+      .WillOnce(
>+          // Verify that channels are not swapped by default.
>+          testing::Invoke([](AudioFrame* audio_frame) {
>+            auto levels = ComputeChannelLevels(audio_frame);
>+            EXPECT_LT(0u, levels[0]);
>+            EXPECT_EQ(0u, levels[1]);
>+          }));
>   MockAudioProcessing* ap =
>       static_cast<MockAudioProcessing*>(audio_state->audio_processing());
>   EXPECT_CALL(*ap, set_stream_delay_ms(0));
>@@ -132,8 +133,8 @@ TEST(AudioStateTest, RecordedAudioArrivesAtSingleStream) {
>   auto audio_data = Create10msTestData(kSampleRate, kNumChannels);
>   uint32_t new_mic_level = 667;
>   audio_state->audio_transport()->RecordedDataIsAvailable(
>-      &audio_data[0], kSampleRate / 100, kNumChannels * 2,
>-      kNumChannels, kSampleRate, 0, 0, 0, false, new_mic_level);
>+      &audio_data[0], kSampleRate / 100, kNumChannels * 2, kNumChannels,
>+      kSampleRate, 0, 0, 0, false, new_mic_level);
>   EXPECT_EQ(667u, new_mic_level);
> 
>   audio_state->RemoveSendingStream(&stream);
>@@ -149,24 +150,28 @@ TEST(AudioStateTest, RecordedAudioArrivesAtMultipleStreams) {
>   audio_state->AddSendingStream(&stream_1, 8001, 2);
>   audio_state->AddSendingStream(&stream_2, 32000, 1);
> 
>-  EXPECT_CALL(stream_1, SendAudioDataForMock(testing::AllOf(
>-      testing::Field(&AudioFrame::sample_rate_hz_, testing::Eq(16000)),
>-      testing::Field(&AudioFrame::num_channels_, testing::Eq(1u)))))
>-          .WillOnce(
>-              // Verify that there is output signal.
>-              testing::Invoke([](AudioFrame* audio_frame) {
>-                auto levels = ComputeChannelLevels(audio_frame);
>-                EXPECT_LT(0u, levels[0]);
>-              }));
>-  EXPECT_CALL(stream_2, SendAudioDataForMock(testing::AllOf(
>-      testing::Field(&AudioFrame::sample_rate_hz_, testing::Eq(16000)),
>-      testing::Field(&AudioFrame::num_channels_, testing::Eq(1u)))))
>-          .WillOnce(
>-              // Verify that there is output signal.
>-              testing::Invoke([](AudioFrame* audio_frame) {
>-                auto levels = ComputeChannelLevels(audio_frame);
>-                EXPECT_LT(0u, levels[0]);
>-              }));
>+  EXPECT_CALL(
>+      stream_1,
>+      SendAudioDataForMock(testing::AllOf(
>+          testing::Field(&AudioFrame::sample_rate_hz_, testing::Eq(16000)),
>+          testing::Field(&AudioFrame::num_channels_, testing::Eq(1u)))))
>+      .WillOnce(
>+          // Verify that there is output signal.
>+          testing::Invoke([](AudioFrame* audio_frame) {
>+            auto levels = ComputeChannelLevels(audio_frame);
>+            EXPECT_LT(0u, levels[0]);
>+          }));
>+  EXPECT_CALL(
>+      stream_2,
>+      SendAudioDataForMock(testing::AllOf(
>+          testing::Field(&AudioFrame::sample_rate_hz_, testing::Eq(16000)),
>+          testing::Field(&AudioFrame::num_channels_, testing::Eq(1u)))))
>+      .WillOnce(
>+          // Verify that there is output signal.
>+          testing::Invoke([](AudioFrame* audio_frame) {
>+            auto levels = ComputeChannelLevels(audio_frame);
>+            EXPECT_LT(0u, levels[0]);
>+          }));
>   MockAudioProcessing* ap =
>       static_cast<MockAudioProcessing*>(audio_state->audio_processing());
>   EXPECT_CALL(*ap, set_stream_delay_ms(5));
>@@ -178,8 +183,8 @@ TEST(AudioStateTest, RecordedAudioArrivesAtMultipleStreams) {
>   auto audio_data = Create10msTestData(kSampleRate, kNumChannels);
>   uint32_t new_mic_level = 667;
>   audio_state->audio_transport()->RecordedDataIsAvailable(
>-      &audio_data[0], kSampleRate / 100, kNumChannels * 2,
>-      kNumChannels, kSampleRate, 5, 0, 0, true, new_mic_level);
>+      &audio_data[0], kSampleRate / 100, kNumChannels * 2, kNumChannels,
>+      kSampleRate, 5, 0, 0, true, new_mic_level);
>   EXPECT_EQ(667u, new_mic_level);
> 
>   audio_state->RemoveSendingStream(&stream_1);
>@@ -210,8 +215,8 @@ TEST(AudioStateTest, EnableChannelSwap) {
>   auto audio_data = Create10msTestData(kSampleRate, kNumChannels);
>   uint32_t new_mic_level = 667;
>   audio_state->audio_transport()->RecordedDataIsAvailable(
>-      &audio_data[0], kSampleRate / 100, kNumChannels * 2,
>-      kNumChannels, kSampleRate, 0, 0, 0, false, new_mic_level);
>+      &audio_data[0], kSampleRate / 100, kNumChannels * 2, kNumChannels,
>+      kSampleRate, 0, 0, 0, false, new_mic_level);
>   EXPECT_EQ(667u, new_mic_level);
> 
>   audio_state->RemoveSendingStream(&stream);
>@@ -230,11 +235,10 @@ TEST(AudioStateTest, InputLevelStats) {
>     auto audio_data = Create10msSilentTestData(kSampleRate, kNumChannels);
>     uint32_t new_mic_level = 667;
>     audio_state->audio_transport()->RecordedDataIsAvailable(
>-        &audio_data[0], kSampleRate / 100, kNumChannels * 2,
>-        kNumChannels, kSampleRate, 0, 0, 0, false, new_mic_level);
>+        &audio_data[0], kSampleRate / 100, kNumChannels * 2, kNumChannels,
>+        kSampleRate, 0, 0, 0, false, new_mic_level);
>     auto stats = audio_state->GetAudioInputStats();
>     EXPECT_EQ(0, stats.audio_level);
>-    EXPECT_EQ(0, stats.quantized_audio_level);
>     EXPECT_THAT(stats.total_energy, testing::DoubleEq(0.0));
>     EXPECT_THAT(stats.total_duration, testing::DoubleEq(0.01));
>   }
>@@ -245,12 +249,11 @@ TEST(AudioStateTest, InputLevelStats) {
>     uint32_t new_mic_level = 667;
>     for (int i = 0; i < 10; ++i) {
>       audio_state->audio_transport()->RecordedDataIsAvailable(
>-          &audio_data[0], kSampleRate / 100, kNumChannels * 2,
>-          kNumChannels, kSampleRate, 0, 0, 0, false, new_mic_level);
>+          &audio_data[0], kSampleRate / 100, kNumChannels * 2, kNumChannels,
>+          kSampleRate, 0, 0, 0, false, new_mic_level);
>     }
>     auto stats = audio_state->GetAudioInputStats();
>     EXPECT_EQ(32767, stats.audio_level);
>-    EXPECT_EQ(9, stats.quantized_audio_level);
>     EXPECT_THAT(stats.total_energy, testing::DoubleEq(0.01));
>     EXPECT_THAT(stats.total_duration, testing::DoubleEq(0.11));
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_transport_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_transport_impl.cc
>index 586fc4660b4..44e95aa909a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_transport_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_transport_impl.cc
>@@ -14,10 +14,10 @@
> #include <memory>
> #include <utility>
> 
>+#include "audio/remix_resample.h"
> #include "audio/utility/audio_frame_operations.h"
> #include "call/audio_send_stream.h"
> #include "rtc_base/logging.h"
>-#include "voice_engine/utility.h"
> 
> namespace webrtc {
> 
>@@ -43,8 +43,7 @@ void InitializeCaptureFrame(int input_sample_rate,
>   audio_frame->num_channels_ = std::min(input_num_channels, send_num_channels);
> }
> 
>-void ProcessCaptureFrame(int analog_level,
>-                         uint32_t delay_ms,
>+void ProcessCaptureFrame(uint32_t delay_ms,
>                          bool key_pressed,
>                          bool swap_stereo_channels,
>                          AudioProcessing* audio_processing,
>@@ -53,13 +52,9 @@ void ProcessCaptureFrame(int analog_level,
>   RTC_DCHECK(audio_frame);
>   RTC_DCHECK(
>       !audio_processing->echo_cancellation()->is_drift_compensation_enabled());
>-  GainControl* agc = audio_processing->gain_control();
>-  int error = agc->set_stream_analog_level(analog_level);
>-  RTC_DCHECK_EQ(0, error) <<
>-      "set_stream_analog_level failed: analog_level = " << analog_level;
>   audio_processing->set_stream_delay_ms(delay_ms);
>   audio_processing->set_stream_key_pressed(key_pressed);
>-  error = audio_processing->ProcessStream(audio_frame);
>+  int error = audio_processing->ProcessStream(audio_frame);
>   RTC_DCHECK_EQ(0, error) << "ProcessStream() error: " << error;
>   if (swap_stereo_channels) {
>     AudioFrameOperations::SwapStereoChannels(audio_frame);
>@@ -87,14 +82,10 @@ int Resample(const AudioFrame& frame,
> }  // namespace
> 
> AudioTransportImpl::AudioTransportImpl(AudioMixer* mixer,
>-                                       AudioProcessing* audio_processing,
>-                                       AudioDeviceModule* audio_device_module)
>-    : audio_processing_(audio_processing),
>-      audio_device_module_(audio_device_module),
>-      mixer_(mixer) {
>+                                       AudioProcessing* audio_processing)
>+    : audio_processing_(audio_processing), mixer_(mixer) {
>   RTC_DCHECK(mixer);
>   RTC_DCHECK(audio_processing);
>-  RTC_DCHECK(audio_device_module);
> }
> 
> AudioTransportImpl::~AudioTransportImpl() {}
>@@ -109,7 +100,7 @@ int32_t AudioTransportImpl::RecordedDataIsAvailable(
>     const uint32_t sample_rate,
>     const uint32_t audio_delay_milliseconds,
>     const int32_t /*clock_drift*/,
>-    const uint32_t volume,
>+    const uint32_t /*volume*/,
>     const bool key_pressed,
>     uint32_t& /*new_mic_volume*/) {  // NOLINT: to avoid changing APIs
>   RTC_DCHECK(audio_data);
>@@ -122,35 +113,6 @@ int32_t AudioTransportImpl::RecordedDataIsAvailable(
>   RTC_DCHECK_LE(bytes_per_sample * number_of_frames * number_of_channels,
>                 AudioFrame::kMaxDataSizeBytes);
> 
>-  // TODO(solenberg): Remove volume handling since it is now always 0.
>-  uint16_t voe_mic_level = 0;
>-  {
>-    constexpr uint32_t kMaxVolumeLevel = 255;
>-    uint32_t max_volume = 0;
>-
>-    // Check for zero to skip this calculation; the consumer may use this to
>-    // indicate no volume is available.
>-    if (volume != 0) {
>-      // Scale from ADM to VoE level range
>-      if (audio_device_module_->MaxMicrophoneVolume(&max_volume) == 0) {
>-        if (max_volume != 0) {
>-          voe_mic_level = static_cast<uint16_t>(
>-              (volume * kMaxVolumeLevel + static_cast<int>(max_volume / 2)) /
>-              max_volume);
>-        }
>-      }
>-      // We learned that on certain systems (e.g Linux) the voe_mic_level
>-      // can be greater than the maxVolumeLevel therefore
>-      // we are going to cap the voe_mic_level to the maxVolumeLevel
>-      // and change the maxVolume to volume if it turns out that
>-      // the voe_mic_level is indeed greater than the maxVolumeLevel.
>-      if (voe_mic_level > kMaxVolumeLevel) {
>-        voe_mic_level = kMaxVolumeLevel;
>-        max_volume = volume;
>-      }
>-    }
>-  }
>-
>   int send_sample_rate_hz = 0;
>   size_t send_num_channels = 0;
>   bool swap_stereo_channels = false;
>@@ -162,13 +124,12 @@ int32_t AudioTransportImpl::RecordedDataIsAvailable(
>   }
> 
>   std::unique_ptr<AudioFrame> audio_frame(new AudioFrame());
>-  InitializeCaptureFrame(sample_rate, send_sample_rate_hz,
>-                         number_of_channels, send_num_channels,
>-                         audio_frame.get());
>+  InitializeCaptureFrame(sample_rate, send_sample_rate_hz, number_of_channels,
>+                         send_num_channels, audio_frame.get());
>   voe::RemixAndResample(static_cast<const int16_t*>(audio_data),
>                         number_of_frames, number_of_channels, sample_rate,
>                         &capture_resampler_, audio_frame.get());
>-  ProcessCaptureFrame(voe_mic_level, audio_delay_milliseconds, key_pressed,
>+  ProcessCaptureFrame(audio_delay_milliseconds, key_pressed,
>                       swap_stereo_channels, audio_processing_,
>                       audio_frame.get());
> 
>@@ -212,13 +173,13 @@ int32_t AudioTransportImpl::RecordedDataIsAvailable(
> // Mix all received streams, feed the result to the AudioProcessing module, then
> // resample the result to the requested output rate.
> int32_t AudioTransportImpl::NeedMorePlayData(const size_t nSamples,
>-                                              const size_t nBytesPerSample,
>-                                              const size_t nChannels,
>-                                              const uint32_t samplesPerSec,
>-                                              void* audioSamples,
>-                                              size_t& nSamplesOut,
>-                                              int64_t* elapsed_time_ms,
>-                                              int64_t* ntp_time_ms) {
>+                                             const size_t nBytesPerSample,
>+                                             const size_t nChannels,
>+                                             const uint32_t samplesPerSec,
>+                                             void* audioSamples,
>+                                             size_t& nSamplesOut,
>+                                             int64_t* elapsed_time_ms,
>+                                             int64_t* ntp_time_ms) {
>   RTC_DCHECK_EQ(sizeof(int16_t) * nChannels, nBytesPerSample);
>   RTC_DCHECK_GE(nChannels, 1);
>   RTC_DCHECK_LE(nChannels, 2);
>@@ -247,12 +208,12 @@ int32_t AudioTransportImpl::NeedMorePlayData(const size_t nSamples,
> // Used by Chromium - same as NeedMorePlayData() but because Chrome has its
> // own APM instance, does not call audio_processing_->ProcessReverseStream().
> void AudioTransportImpl::PullRenderData(int bits_per_sample,
>-                                         int sample_rate,
>-                                         size_t number_of_channels,
>-                                         size_t number_of_frames,
>-                                         void* audio_data,
>-                                         int64_t* elapsed_time_ms,
>-                                         int64_t* ntp_time_ms) {
>+                                        int sample_rate,
>+                                        size_t number_of_channels,
>+                                        size_t number_of_frames,
>+                                        void* audio_data,
>+                                        int64_t* elapsed_time_ms,
>+                                        int64_t* ntp_time_ms) {
>   RTC_DCHECK_EQ(bits_per_sample, 16);
>   RTC_DCHECK_GE(number_of_channels, 1);
>   RTC_DCHECK_LE(number_of_channels, 2);
>@@ -274,7 +235,8 @@ void AudioTransportImpl::PullRenderData(int bits_per_sample,
> }
> 
> void AudioTransportImpl::UpdateSendingStreams(
>-    std::vector<AudioSendStream*> streams, int send_sample_rate_hz,
>+    std::vector<AudioSendStream*> streams,
>+    int send_sample_rate_hz,
>     size_t send_num_channels) {
>   rtc::CritScope lock(&capture_lock_);
>   sending_streams_ = std::move(streams);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_transport_impl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_transport_impl.h
>index e7de7e9f485..3a3155cdbd4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_transport_impl.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/audio_transport_impl.h
>@@ -14,6 +14,7 @@
> #include <vector>
> 
> #include "api/audio/audio_mixer.h"
>+#include "audio/audio_level.h"
> #include "common_audio/resampler/include/push_resampler.h"
> #include "modules/audio_device/include/audio_device.h"
> #include "modules/audio_processing/include/audio_processing.h"
>@@ -22,7 +23,6 @@
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/scoped_ref_ptr.h"
> #include "rtc_base/thread_annotations.h"
>-#include "voice_engine/audio_level.h"
> 
> namespace webrtc {
> 
>@@ -30,9 +30,7 @@ class AudioSendStream;
> 
> class AudioTransportImpl : public AudioTransport {
>  public:
>-  AudioTransportImpl(AudioMixer* mixer,
>-                     AudioProcessing* audio_processing,
>-                     AudioDeviceModule* audio_device_module);
>+  AudioTransportImpl(AudioMixer* mixer, AudioProcessing* audio_processing);
>   ~AudioTransportImpl() override;
> 
>   int32_t RecordedDataIsAvailable(const void* audioSamples,
>@@ -64,12 +62,11 @@ class AudioTransportImpl : public AudioTransport {
>                       int64_t* ntp_time_ms) override;
> 
>   void UpdateSendingStreams(std::vector<AudioSendStream*> streams,
>-                            int send_sample_rate_hz, size_t send_num_channels);
>+                            int send_sample_rate_hz,
>+                            size_t send_num_channels);
>   void SetStereoChannelSwapping(bool enable);
>   bool typing_noise_detected() const;
>-  const voe::AudioLevel& audio_level() const {
>-    return audio_level_;
>-  }
>+  const voe::AudioLevel& audio_level() const { return audio_level_; }
> 
>  private:
>   // Shared.
>@@ -82,7 +79,6 @@ class AudioTransportImpl : public AudioTransport {
>   size_t send_num_channels_ RTC_GUARDED_BY(capture_lock_) = 1;
>   bool typing_noise_detected_ RTC_GUARDED_BY(capture_lock_) = false;
>   bool swap_stereo_channels_ RTC_GUARDED_BY(capture_lock_) = false;
>-  AudioDeviceModule* audio_device_module_ = nullptr;
>   PushResampler<int16_t> capture_resampler_;
>   voe::AudioLevel audio_level_;
>   TypingDetection typing_detection_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/voice_engine/channel.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/channel.cc
>similarity index 84%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/voice_engine/channel.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/audio/channel.cc
>index efc76bdcab4..90443645aed 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/voice_engine/channel.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/channel.cc
>@@ -8,7 +8,7 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "voice_engine/channel.h"
>+#include "audio/channel.h"
> 
> #include <algorithm>
> #include <map>
>@@ -17,20 +17,18 @@
> #include <utility>
> #include <vector>
> 
>+#include "absl/memory/memory.h"
> #include "api/array_view.h"
> #include "audio/utility/audio_frame_operations.h"
> #include "call/rtp_transport_controller_send_interface.h"
>-#include "logging/rtc_event_log/rtc_event_log.h"
> #include "logging/rtc_event_log/events/rtc_event_audio_playout.h"
>+#include "logging/rtc_event_log/rtc_event_log.h"
> #include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
> #include "modules/audio_coding/codecs/audio_format_conversion.h"
> #include "modules/audio_device/include/audio_device.h"
> #include "modules/audio_processing/include/audio_processing.h"
>-#include "modules/include/module_common_types.h"
> #include "modules/pacing/packet_router.h"
> #include "modules/rtp_rtcp/include/receive_statistics.h"
>-#include "modules/rtp_rtcp/include/rtp_payload_registry.h"
>-#include "modules/rtp_rtcp/include/rtp_receiver.h"
> #include "modules/rtp_rtcp/source/rtp_packet_received.h"
> #include "modules/rtp_rtcp/source/rtp_receiver_strategy.h"
> #include "modules/utility/include/process_thread.h"
>@@ -39,7 +37,6 @@
> #include "rtc_base/format_macros.h"
> #include "rtc_base/location.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/rate_limiter.h"
> #include "rtc_base/task_queue.h"
> #include "rtc_base/thread_checker.h"
>@@ -64,64 +61,6 @@ constexpr int kVoiceEngineMaxMinPlayoutDelayMs = 10000;
> 
> const int kTelephoneEventAttenuationdB = 10;
> 
>-class RtcEventLogProxy final : public webrtc::RtcEventLog {
>- public:
>-  RtcEventLogProxy() : event_log_(nullptr) {}
>-
>-  bool StartLogging(std::unique_ptr<RtcEventLogOutput> output,
>-                    int64_t output_period_ms) override {
>-    RTC_NOTREACHED();
>-    return false;
>-  }
>-
>-  void StopLogging() override { RTC_NOTREACHED(); }
>-
>-  void Log(std::unique_ptr<RtcEvent> event) override {
>-    rtc::CritScope lock(&crit_);
>-    if (event_log_) {
>-      event_log_->Log(std::move(event));
>-    }
>-  }
>-
>-  void SetEventLog(RtcEventLog* event_log) {
>-    rtc::CritScope lock(&crit_);
>-    event_log_ = event_log;
>-  }
>-
>- private:
>-  rtc::CriticalSection crit_;
>-  RtcEventLog* event_log_ RTC_GUARDED_BY(crit_);
>-  RTC_DISALLOW_COPY_AND_ASSIGN(RtcEventLogProxy);
>-};
>-
>-class RtcpRttStatsProxy final : public RtcpRttStats {
>- public:
>-  RtcpRttStatsProxy() : rtcp_rtt_stats_(nullptr) {}
>-
>-  void OnRttUpdate(int64_t rtt) override {
>-    rtc::CritScope lock(&crit_);
>-    if (rtcp_rtt_stats_)
>-      rtcp_rtt_stats_->OnRttUpdate(rtt);
>-  }
>-
>-  int64_t LastProcessedRtt() const override {
>-    rtc::CritScope lock(&crit_);
>-    if (!rtcp_rtt_stats_)
>-      return 0;
>-    return rtcp_rtt_stats_->LastProcessedRtt();
>-  }
>-
>-  void SetRtcpRttStats(RtcpRttStats* rtcp_rtt_stats) {
>-    rtc::CritScope lock(&crit_);
>-    rtcp_rtt_stats_ = rtcp_rtt_stats;
>-  }
>-
>- private:
>-  rtc::CriticalSection crit_;
>-  RtcpRttStats* rtcp_rtt_stats_ RTC_GUARDED_BY(crit_);
>-  RTC_DISALLOW_COPY_AND_ASSIGN(RtcpRttStatsProxy);
>-};
>-
> class TransportFeedbackProxy : public TransportFeedbackObserver {
>  public:
>   TransportFeedbackProxy() : feedback_observer_(nullptr) {
>@@ -153,10 +92,6 @@ class TransportFeedbackProxy : public TransportFeedbackObserver {
>     if (feedback_observer_)
>       feedback_observer_->OnTransportFeedback(feedback);
>   }
>-  std::vector<PacketFeedback> GetTransportFeedbackVector() const override {
>-    RTC_NOTREACHED();
>-    return std::vector<PacketFeedback>();
>-  }
> 
>  private:
>   rtc::CriticalSection crit_;
>@@ -345,7 +280,7 @@ int32_t Channel::SendData(FrameType frameType,
>           // received from the capture device as
>           // undefined for voice for now.
>           -1, payloadData, payloadSize, fragmentation, nullptr, nullptr)) {
>-    RTC_LOG(LS_ERROR)
>+    RTC_DLOG(LS_ERROR)
>         << "Channel::SendData() failed to send data to RTP/RTCP module";
>     return -1;
>   }
>@@ -359,17 +294,14 @@ bool Channel::SendRtp(const uint8_t* data,
>   rtc::CritScope cs(&_callbackCritSect);
> 
>   if (_transportPtr == NULL) {
>-    RTC_LOG(LS_ERROR)
>+    RTC_DLOG(LS_ERROR)
>         << "Channel::SendPacket() failed to send RTP packet due to"
>         << " invalid transport object";
>     return false;
>   }
> 
>-  uint8_t* bufferToSendPtr = (uint8_t*)data;
>-  size_t bufferLength = len;
>-
>-  if (!_transportPtr->SendRtp(bufferToSendPtr, bufferLength, options)) {
>-    RTC_LOG(LS_ERROR) << "Channel::SendPacket() RTP transmission failed";
>+  if (!_transportPtr->SendRtp(data, len, options)) {
>+    RTC_DLOG(LS_ERROR) << "Channel::SendPacket() RTP transmission failed";
>     return false;
>   }
>   return true;
>@@ -378,44 +310,20 @@ bool Channel::SendRtp(const uint8_t* data,
> bool Channel::SendRtcp(const uint8_t* data, size_t len) {
>   rtc::CritScope cs(&_callbackCritSect);
>   if (_transportPtr == NULL) {
>-    RTC_LOG(LS_ERROR) << "Channel::SendRtcp() failed to send RTCP packet due to"
>-                      << " invalid transport object";
>+    RTC_DLOG(LS_ERROR)
>+        << "Channel::SendRtcp() failed to send RTCP packet due to"
>+        << " invalid transport object";
>     return false;
>   }
> 
>-  uint8_t* bufferToSendPtr = (uint8_t*)data;
>-  size_t bufferLength = len;
>-
>-  int n = _transportPtr->SendRtcp(bufferToSendPtr, bufferLength);
>+  int n = _transportPtr->SendRtcp(data, len);
>   if (n < 0) {
>-    RTC_LOG(LS_ERROR) << "Channel::SendRtcp() transmission failed";
>+    RTC_DLOG(LS_ERROR) << "Channel::SendRtcp() transmission failed";
>     return false;
>   }
>   return true;
> }
> 
>-void Channel::OnIncomingSSRCChanged(uint32_t ssrc) {
>-  // Update ssrc so that NTP for AV sync can be updated.
>-  _rtpRtcpModule->SetRemoteSSRC(ssrc);
>-}
>-
>-void Channel::OnIncomingCSRCChanged(uint32_t CSRC, bool added) {
>-  // TODO(saza): remove.
>-}
>-
>-int32_t Channel::OnInitializeDecoder(int payload_type,
>-                                     const SdpAudioFormat& audio_format,
>-                                     uint32_t rate) {
>-  if (!audio_coding_->RegisterReceiveCodec(payload_type, audio_format)) {
>-    RTC_LOG(LS_WARNING) << "Channel::OnInitializeDecoder() invalid codec (pt="
>-                        << payload_type << ", " << audio_format
>-                        << ") received -1";
>-    return -1;
>-  }
>-
>-  return 0;
>-}
>-
> int32_t Channel::OnReceivedPayloadData(const uint8_t* payloadData,
>                                        size_t payloadSize,
>                                        const WebRtcRTPHeader* rtpHeader) {
>@@ -428,14 +336,13 @@ int32_t Channel::OnReceivedPayloadData(const uint8_t* payloadData,
>   // Push the incoming payload (parsed and ready for decoding) into the ACM
>   if (audio_coding_->IncomingPacket(payloadData, payloadSize, *rtpHeader) !=
>       0) {
>-    RTC_LOG(LS_ERROR)
>+    RTC_DLOG(LS_ERROR)
>         << "Channel::OnReceivedPayloadData() unable to push data to the ACM";
>     return -1;
>   }
> 
>   int64_t round_trip_time = 0;
>-  _rtpRtcpModule->RTT(rtp_receiver_->SSRC(), &round_trip_time, NULL, NULL,
>-                      NULL);
>+  _rtpRtcpModule->RTT(remote_ssrc_, &round_trip_time, NULL, NULL, NULL);
> 
>   std::vector<uint16_t> nack_list = audio_coding_->GetNackList(round_trip_time);
>   if (!nack_list.empty()) {
>@@ -453,12 +360,12 @@ AudioMixer::Source::AudioFrameInfo Channel::GetAudioFrameWithInfo(
> 
>   unsigned int ssrc;
>   RTC_CHECK_EQ(GetRemoteSSRC(ssrc), 0);
>-  event_log_proxy_->Log(rtc::MakeUnique<RtcEventAudioPlayout>(ssrc));
>+  event_log_->Log(absl::make_unique<RtcEventAudioPlayout>(ssrc));
>   // Get 10ms raw PCM data from the ACM (mixer limits output frequency)
>   bool muted;
>   if (audio_coding_->PlayoutData10Ms(audio_frame->sample_rate_hz_, audio_frame,
>                                      &muted) == -1) {
>-    RTC_LOG(LS_ERROR) << "Channel::GetAudioFrame() PlayoutData10Ms() failed!";
>+    RTC_DLOG(LS_ERROR) << "Channel::GetAudioFrame() PlayoutData10Ms() failed!";
>     // In all likelihood, the audio in this frame is garbage. We return an
>     // error so that the audio mixer module doesn't add it to the mix. As
>     // a result, it won't be played out and the actions skipped here are
>@@ -561,32 +468,35 @@ int Channel::PreferredSampleRate() const {
> 
> Channel::Channel(rtc::TaskQueue* encoder_queue,
>                  ProcessThread* module_process_thread,
>-                 AudioDeviceModule* audio_device_module)
>+                 AudioDeviceModule* audio_device_module,
>+                 RtcpRttStats* rtcp_rtt_stats,
>+                 RtcEventLog* rtc_event_log)
>     : Channel(module_process_thread,
>               audio_device_module,
>+              rtcp_rtt_stats,
>+              rtc_event_log,
>+              0,
>               0,
>               false,
>-              rtc::scoped_refptr<AudioDecoderFactory>()) {
>+              rtc::scoped_refptr<AudioDecoderFactory>(),
>+              absl::nullopt) {
>   RTC_DCHECK(encoder_queue);
>   encoder_queue_ = encoder_queue;
> }
> 
> Channel::Channel(ProcessThread* module_process_thread,
>                  AudioDeviceModule* audio_device_module,
>+                 RtcpRttStats* rtcp_rtt_stats,
>+                 RtcEventLog* rtc_event_log,
>+                 uint32_t remote_ssrc,
>                  size_t jitter_buffer_max_packets,
>                  bool jitter_buffer_fast_playout,
>-                 rtc::scoped_refptr<AudioDecoderFactory> decoder_factory)
>-    : event_log_proxy_(new RtcEventLogProxy()),
>-      rtcp_rtt_stats_proxy_(new RtcpRttStatsProxy()),
>-      rtp_payload_registry_(new RTPPayloadRegistry()),
>+                 rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
>+                 absl::optional<AudioCodecPairId> codec_pair_id)
>+    : event_log_(rtc_event_log),
>       rtp_receive_statistics_(
>           ReceiveStatistics::Create(Clock::GetRealTimeClock())),
>-      rtp_receiver_(
>-          RtpReceiver::CreateAudioReceiver(Clock::GetRealTimeClock(),
>-                                           this,
>-                                           this,
>-                                           rtp_payload_registry_.get())),
>-      telephone_event_handler_(rtp_receiver_->GetTelephoneEventHandler()),
>+      remote_ssrc_(remote_ssrc),
>       _outputAudioLevel(),
>       _timeStamp(0),  // This is just an offset, RTP module will add it's own
>                       // random offset
>@@ -619,6 +529,7 @@ Channel::Channel(ProcessThread* module_process_thread,
>   RTC_DCHECK(audio_device_module);
>   AudioCodingModule::Config acm_config;
>   acm_config.decoder_factory = decoder_factory;
>+  acm_config.neteq_config.codec_pair_id = codec_pair_id;
>   acm_config.neteq_config.max_packets_in_buffer = jitter_buffer_max_packets;
>   acm_config.neteq_config.enable_fast_accelerate = jitter_buffer_fast_playout;
>   acm_config.neteq_config.enable_muted_state = true;
>@@ -638,14 +549,14 @@ Channel::Channel(ProcessThread* module_process_thread,
>         seq_num_allocator_proxy_.get();
>     configuration.transport_feedback_callback = feedback_observer_proxy_.get();
>   }
>-  configuration.event_log = &(*event_log_proxy_);
>-  configuration.rtt_stats = &(*rtcp_rtt_stats_proxy_);
>+  configuration.event_log = event_log_;
>+  configuration.rtt_stats = rtcp_rtt_stats;
>   configuration.retransmission_rate_limiter =
>       retransmission_rate_limiter_.get();
> 
>   _rtpRtcpModule.reset(RtpRtcp::CreateRtpRtcp(configuration));
>   _rtpRtcpModule->SetSendingMediaStatus(false);
>-
>+  _rtpRtcpModule->SetRemoteSSRC(remote_ssrc_);
>   Init();
> }
> 
>@@ -672,7 +583,6 @@ void Channel::Init() {
>   // disabled by the user.
>   // After StopListen (when no sockets exists), RTCP packets will no longer
>   // be transmitted since the Transport object will then be invalid.
>-  telephone_event_handler_->SetTelephoneEventForwardToDecoder(true);
>   // RTCP is enabled by default.
>   _rtpRtcpModule->SetRTCPStatus(RtcpMode::kCompound);
> 
>@@ -742,7 +652,7 @@ int32_t Channel::StartSend() {
>   }
>   _rtpRtcpModule->SetSendingMediaStatus(true);
>   if (_rtpRtcpModule->SetSendingStatus(true) != 0) {
>-    RTC_LOG(LS_ERROR) << "StartSend() RTP/RTCP failed to start sending";
>+    RTC_DLOG(LS_ERROR) << "StartSend() RTP/RTCP failed to start sending";
>     _rtpRtcpModule->SetSendingMediaStatus(false);
>     rtc::CritScope cs(&_callbackCritSect);
>     channel_state_.SetSending(false);
>@@ -791,7 +701,7 @@ void Channel::StopSend() {
>   // Reset sending SSRC and sequence number and triggers direct transmission
>   // of RTCP BYE
>   if (_rtpRtcpModule->SetSendingStatus(false) == -1) {
>-    RTC_LOG(LS_ERROR) << "StartSend() RTP/RTCP failed to stop sending";
>+    RTC_DLOG(LS_ERROR) << "StartSend() RTP/RTCP failed to stop sending";
>   }
>   _rtpRtcpModule->SetSendingMediaStatus(false);
> }
>@@ -822,7 +732,7 @@ bool Channel::SetEncoder(int payload_type,
>   if (_rtpRtcpModule->RegisterSendPayload(rtp_codec) != 0) {
>     _rtpRtcpModule->DeRegisterSendPayload(payload_type);
>     if (_rtpRtcpModule->RegisterSendPayload(rtp_codec) != 0) {
>-      RTC_LOG(LS_ERROR)
>+      RTC_DLOG(LS_ERROR)
>           << "SetEncoder() failed to register codec to RTP/RTCP module";
>       return false;
>     }
>@@ -870,6 +780,22 @@ void Channel::OnRecoverableUplinkPacketLossRate(
>   });
> }
> 
>+std::vector<webrtc::RtpSource> Channel::GetSources() const {
>+  int64_t now_ms = rtc::TimeMillis();
>+  std::vector<RtpSource> sources;
>+  {
>+    rtc::CritScope cs(&rtp_sources_lock_);
>+    sources = contributing_sources_.GetSources(now_ms);
>+    if (last_received_rtp_system_time_ms_ >=
>+        now_ms - ContributingSources::kHistoryMs) {
>+      sources.emplace_back(*last_received_rtp_system_time_ms_, remote_ssrc_,
>+                           RtpSourceType::SSRC);
>+      sources.back().set_audio_level(last_received_rtp_audio_level_);
>+    }
>+  }
>+  return sources;
>+}
>+
> void Channel::OnUplinkPacketLossRate(float packet_loss_rate) {
>   if (use_twcc_plr_for_ana_)
>     return;
>@@ -881,7 +807,10 @@ void Channel::OnUplinkPacketLossRate(float packet_loss_rate) {
> }
> 
> void Channel::SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs) {
>-  rtp_payload_registry_->SetAudioReceivePayloads(codecs);
>+  for (const auto& kv : codecs) {
>+    RTC_DCHECK_GE(kv.second.clockrate_hz, 1000);
>+    payload_type_frequencies_[kv.first] = kv.second.clockrate_hz;
>+  }
>   audio_coding_->SetReceiveCodecs(codecs);
> }
> 
>@@ -889,8 +818,8 @@ bool Channel::EnableAudioNetworkAdaptor(const std::string& config_string) {
>   bool success = false;
>   audio_coding_->ModifyEncoder([&](std::unique_ptr<AudioEncoder>* encoder) {
>     if (*encoder) {
>-      success = (*encoder)->EnableAudioNetworkAdaptor(config_string,
>-                                                      event_log_proxy_.get());
>+      success =
>+          (*encoder)->EnableAudioNetworkAdaptor(config_string, event_log_);
>     }
>   });
>   return success;
>@@ -918,23 +847,39 @@ void Channel::RegisterTransport(Transport* transport) {
>   _transportPtr = transport;
> }
> 
>+// TODO(nisse): Move receive logic up to AudioReceiveStream.
> void Channel::OnRtpPacket(const RtpPacketReceived& packet) {
>+  int64_t now_ms = rtc::TimeMillis();
>+  uint8_t audio_level;
>+  bool voice_activity;
>+  bool has_audio_level =
>+      packet.GetExtension<::webrtc::AudioLevel>(&voice_activity, &audio_level);
>+
>+  {
>+    rtc::CritScope cs(&rtp_sources_lock_);
>+    last_received_rtp_timestamp_ = packet.Timestamp();
>+    last_received_rtp_system_time_ms_ = now_ms;
>+    if (has_audio_level)
>+      last_received_rtp_audio_level_ = audio_level;
>+    std::vector<uint32_t> csrcs = packet.Csrcs();
>+    contributing_sources_.Update(now_ms, csrcs);
>+  }
>+
>   RTPHeader header;
>   packet.GetHeader(&header);
> 
>   // Store playout timestamp for the received RTP packet
>   UpdatePlayoutTimestamp(false);
> 
>-  header.payload_type_frequency =
>-      rtp_payload_registry_->GetPayloadTypeFrequency(header.payloadType);
>-  if (header.payload_type_frequency >= 0) {
>-    bool in_order = IsPacketInOrder(header);
>-    rtp_receive_statistics_->IncomingPacket(
>-        header, packet.size(), IsPacketRetransmitted(header, in_order));
>-    rtp_payload_registry_->SetIncomingPayloadType(header);
>+  const auto& it = payload_type_frequencies_.find(header.payloadType);
>+  if (it == payload_type_frequencies_.end())
>+    return;
>+  header.payload_type_frequency = it->second;
> 
>-    ReceivePacket(packet.data(), packet.size(), header);
>-  }
>+  rtp_receive_statistics_->IncomingPacket(header, packet.size(),
>+                                          IsPacketRetransmitted(header));
>+
>+  ReceivePacket(packet.data(), packet.size(), header);
> }
> 
> bool Channel::ReceivePacket(const uint8_t* packet,
>@@ -943,33 +888,25 @@ bool Channel::ReceivePacket(const uint8_t* packet,
>   const uint8_t* payload = packet + header.headerLength;
>   assert(packet_length >= header.headerLength);
>   size_t payload_length = packet_length - header.headerLength;
>-  const auto pl =
>-      rtp_payload_registry_->PayloadTypeToPayload(header.payloadType);
>-  if (!pl) {
>-    return false;
>-  }
>-  return rtp_receiver_->IncomingRtpPacket(header, payload, payload_length,
>-                                          pl->typeSpecific);
>-}
>+  WebRtcRTPHeader webrtc_rtp_header = {};
>+  webrtc_rtp_header.header = header;
> 
>-bool Channel::IsPacketInOrder(const RTPHeader& header) const {
>-  StreamStatistician* statistician =
>-      rtp_receive_statistics_->GetStatistician(header.ssrc);
>-  if (!statistician)
>-    return false;
>-  return statistician->IsPacketInOrder(header.sequenceNumber);
>+  const size_t payload_data_length = payload_length - header.paddingLength;
>+  if (payload_data_length == 0) {
>+    webrtc_rtp_header.frameType = kEmptyFrame;
>+    return OnReceivedPayloadData(nullptr, 0, &webrtc_rtp_header);
>+  }
>+  return OnReceivedPayloadData(payload, payload_data_length,
>+                               &webrtc_rtp_header);
> }
> 
>-bool Channel::IsPacketRetransmitted(const RTPHeader& header,
>-                                    bool in_order) const {
>+bool Channel::IsPacketRetransmitted(const RTPHeader& header) const {
>   StreamStatistician* statistician =
>       rtp_receive_statistics_->GetStatistician(header.ssrc);
>   if (!statistician)
>     return false;
>   // Check if this is a retransmission.
>-  int64_t min_rtt = 0;
>-  _rtpRtcpModule->RTT(rtp_receiver_->SSRC(), NULL, NULL, &min_rtt, NULL);
>-  return !in_order && statistician->IsRetransmitOfOldPacket(header, min_rtt);
>+  return statistician->IsRetransmitOfOldPacket(header);
> }
> 
> int32_t Channel::ReceivedRTCPPacket(const uint8_t* data, size_t length) {
>@@ -1002,9 +939,8 @@ int32_t Channel::ReceivedRTCPPacket(const uint8_t* data, size_t length) {
>   uint32_t ntp_secs = 0;
>   uint32_t ntp_frac = 0;
>   uint32_t rtp_timestamp = 0;
>-  if (0 !=
>-      _rtpRtcpModule->RemoteNTP(&ntp_secs, &ntp_frac, NULL, NULL,
>-                                &rtp_timestamp)) {
>+  if (0 != _rtpRtcpModule->RemoteNTP(&ntp_secs, &ntp_frac, NULL, NULL,
>+                                     &rtp_timestamp)) {
>     // Waiting for RTCP.
>     return 0;
>   }
>@@ -1016,10 +952,6 @@ int32_t Channel::ReceivedRTCPPacket(const uint8_t* data, size_t length) {
>   return 0;
> }
> 
>-int Channel::GetSpeechOutputLevel() const {
>-  return _outputAudioLevel.Level();
>-}
>-
> int Channel::GetSpeechOutputLevelFullRange() const {
>   return _outputAudioLevel.LevelFullRange();
> }
>@@ -1056,8 +988,8 @@ int Channel::SendTelephoneEventOutband(int event, int duration_ms) {
>     return -1;
>   }
>   if (_rtpRtcpModule->SendTelephoneEventOutband(
>-      event, duration_ms, kTelephoneEventAttenuationdB) != 0) {
>-    RTC_LOG(LS_ERROR) << "SendTelephoneEventOutband() failed to send event";
>+          event, duration_ms, kTelephoneEventAttenuationdB) != 0) {
>+    RTC_DLOG(LS_ERROR) << "SendTelephoneEventOutband() failed to send event";
>     return -1;
>   }
>   return 0;
>@@ -1074,7 +1006,7 @@ int Channel::SetSendTelephoneEventPayloadType(int payload_type,
>   if (_rtpRtcpModule->RegisterSendPayload(codec) != 0) {
>     _rtpRtcpModule->DeRegisterSendPayload(codec.pltype);
>     if (_rtpRtcpModule->RegisterSendPayload(codec) != 0) {
>-      RTC_LOG(LS_ERROR)
>+      RTC_DLOG(LS_ERROR)
>           << "SetSendTelephoneEventPayloadType() failed to register "
>              "send payload type";
>       return -1;
>@@ -1085,15 +1017,22 @@ int Channel::SetSendTelephoneEventPayloadType(int payload_type,
> 
> int Channel::SetLocalSSRC(unsigned int ssrc) {
>   if (channel_state_.Get().sending) {
>-    RTC_LOG(LS_ERROR) << "SetLocalSSRC() already sending";
>+    RTC_DLOG(LS_ERROR) << "SetLocalSSRC() already sending";
>     return -1;
>   }
>   _rtpRtcpModule->SetSSRC(ssrc);
>   return 0;
> }
> 
>+void Channel::SetMid(const std::string& mid, int extension_id) {
>+  int ret = SetSendRtpHeaderExtension(true, kRtpExtensionMid, extension_id);
>+  RTC_DCHECK_EQ(0, ret);
>+  _rtpRtcpModule->SetMid(mid);
>+}
>+
>+// TODO(nisse): Pass ssrc in return value instead.
> int Channel::GetRemoteSSRC(unsigned int& ssrc) {
>-  ssrc = rtp_receiver_->SSRC();
>+  ssrc = remote_ssrc_;
>   return 0;
> }
> 
>@@ -1163,7 +1102,7 @@ void Channel::SetRTCPStatus(bool enable) {
> 
> int Channel::SetRTCP_CNAME(const char cName[256]) {
>   if (_rtpRtcpModule->SetCNAME(cName) != 0) {
>-    RTC_LOG(LS_ERROR) << "SetRTCP_CNAME() failed to set RTCP CNAME";
>+    RTC_DLOG(LS_ERROR) << "SetRTCP_CNAME() failed to set RTCP CNAME";
>     return -1;
>   }
>   return 0;
>@@ -1172,7 +1111,7 @@ int Channel::SetRTCP_CNAME(const char cName[256]) {
> int Channel::GetRemoteRTCPReportBlocks(
>     std::vector<ReportBlock>* report_blocks) {
>   if (report_blocks == NULL) {
>-    RTC_LOG(LS_ERROR) << "GetRemoteRTCPReportBlock()s invalid report_blocks.";
>+    RTC_DLOG(LS_ERROR) << "GetRemoteRTCPReportBlock()s invalid report_blocks.";
>     return -1;
>   }
> 
>@@ -1207,14 +1146,16 @@ int Channel::GetRemoteRTCPReportBlocks(
> int Channel::GetRTPStatistics(CallStatistics& stats) {
>   // --- RtcpStatistics
> 
>-  // The jitter statistics is updated for each received RTP packet and is
>-  // based on received packets.
>+  // Jitter, cumulative loss, and extended max sequence number is updated for
>+  // each received RTP packet.
>   RtcpStatistics statistics;
>   StreamStatistician* statistician =
>-      rtp_receive_statistics_->GetStatistician(rtp_receiver_->SSRC());
>+      rtp_receive_statistics_->GetStatistician(remote_ssrc_);
>   if (statistician) {
>-    statistician->GetStatistics(&statistics,
>-                                _rtpRtcpModule->RTCP() == RtcpMode::kOff);
>+    // Recompute |fraction_lost| only if RTCP is off. If it's on, then
>+    // |fraction_lost| should only be recomputed when an RTCP SR or RR is sent.
>+    bool update_fraction_lost = _rtpRtcpModule->RTCP() == RtcpMode::kOff;
>+    statistician->GetStatistics(&statistics, update_fraction_lost);
>   }
> 
>   stats.fractionLost = statistics.fraction_lost;
>@@ -1237,7 +1178,7 @@ int Channel::GetRTPStatistics(CallStatistics& stats) {
>   }
> 
>   if (_rtpRtcpModule->DataCountersRTP(&bytesSent, &packetsSent) != 0) {
>-    RTC_LOG(LS_WARNING)
>+    RTC_DLOG(LS_WARNING)
>         << "GetRTPStatistics() failed to retrieve RTP datacounters"
>         << " => output will not be complete";
>   }
>@@ -1320,7 +1261,7 @@ void Channel::ProcessAndEncodeAudioOnTaskQueue(AudioFrame* audio_input) {
>   // is done and payload is ready for packetization and transmission.
>   // Otherwise, it will return without invoking the callback.
>   if (audio_coding_->Add10MsData(*audio_input) < 0) {
>-    RTC_LOG(LS_ERROR) << "ACM::Add10MsData() failed.";
>+    RTC_DLOG(LS_ERROR) << "ACM::Add10MsData() failed.";
>     return;
>   }
> 
>@@ -1333,14 +1274,6 @@ void Channel::SetAssociatedSendChannel(Channel* channel) {
>   associated_send_channel_ = channel;
> }
> 
>-void Channel::SetRtcEventLog(RtcEventLog* event_log) {
>-  event_log_proxy_->SetEventLog(event_log);
>-}
>-
>-void Channel::SetRtcpRttStats(RtcpRttStats* rtcp_rtt_stats) {
>-  rtcp_rtt_stats_proxy_->SetRtcpRttStats(rtcp_rtt_stats);
>-}
>-
> void Channel::UpdateOverheadForEncoder() {
>   size_t overhead_per_packet =
>       transport_overhead_per_packet_ + rtp_overhead_per_packet_;
>@@ -1384,11 +1317,11 @@ uint32_t Channel::GetDelayEstimate() const {
> int Channel::SetMinimumPlayoutDelay(int delayMs) {
>   if ((delayMs < kVoiceEngineMinMinPlayoutDelayMs) ||
>       (delayMs > kVoiceEngineMaxMinPlayoutDelayMs)) {
>-    RTC_LOG(LS_ERROR) << "SetMinimumPlayoutDelay() invalid min delay";
>+    RTC_DLOG(LS_ERROR) << "SetMinimumPlayoutDelay() invalid min delay";
>     return -1;
>   }
>   if (audio_coding_->SetMinimumPlayoutDelay(delayMs) != 0) {
>-    RTC_LOG(LS_ERROR)
>+    RTC_DLOG(LS_ERROR)
>         << "SetMinimumPlayoutDelay() failed to set min playout delay";
>     return -1;
>   }
>@@ -1402,18 +1335,33 @@ int Channel::GetPlayoutTimestamp(unsigned int& timestamp) {
>     playout_timestamp_rtp = playout_timestamp_rtp_;
>   }
>   if (playout_timestamp_rtp == 0) {
>-    RTC_LOG(LS_ERROR) << "GetPlayoutTimestamp() failed to retrieve timestamp";
>+    RTC_DLOG(LS_ERROR) << "GetPlayoutTimestamp() failed to retrieve timestamp";
>     return -1;
>   }
>   timestamp = playout_timestamp_rtp;
>   return 0;
> }
> 
>-int Channel::GetRtpRtcp(RtpRtcp** rtpRtcpModule,
>-                        RtpReceiver** rtp_receiver) const {
>-  *rtpRtcpModule = _rtpRtcpModule.get();
>-  *rtp_receiver = rtp_receiver_.get();
>-  return 0;
>+RtpRtcp* Channel::GetRtpRtcp() const {
>+  return _rtpRtcpModule.get();
>+}
>+
>+absl::optional<Syncable::Info> Channel::GetSyncInfo() const {
>+  Syncable::Info info;
>+  if (_rtpRtcpModule->RemoteNTP(&info.capture_time_ntp_secs,
>+                                &info.capture_time_ntp_frac, nullptr, nullptr,
>+                                &info.capture_time_source_clock) != 0) {
>+    return absl::nullopt;
>+  }
>+  {
>+    rtc::CritScope cs(&rtp_sources_lock_);
>+    if (!last_received_rtp_timestamp_ || !last_received_rtp_system_time_ms_) {
>+      return absl::nullopt;
>+    }
>+    info.latest_received_capture_timestamp = *last_received_rtp_timestamp_;
>+    info.latest_receive_time_ms = *last_received_rtp_system_time_ms_;
>+  }
>+  return info;
> }
> 
> void Channel::UpdatePlayoutTimestamp(bool rtcp) {
>@@ -1427,8 +1375,8 @@ void Channel::UpdatePlayoutTimestamp(bool rtcp) {
> 
>   uint16_t delay_ms = 0;
>   if (_audioDeviceModulePtr->PlayoutDelay(&delay_ms) == -1) {
>-    RTC_LOG(LS_WARNING) << "Channel::UpdatePlayoutTimestamp() failed to read"
>-                        << " playout delay from the ADM";
>+    RTC_DLOG(LS_WARNING) << "Channel::UpdatePlayoutTimestamp() failed to read"
>+                         << " playout delay from the ADM";
>     return;
>   }
> 
>@@ -1493,25 +1441,23 @@ int64_t Channel::GetRTT(bool allow_associate_channel) const {
>     return rtt;
>   }
> 
>-  uint32_t remoteSSRC = rtp_receiver_->SSRC();
>   std::vector<RTCPReportBlock>::const_iterator it = report_blocks.begin();
>   for (; it != report_blocks.end(); ++it) {
>-    if (it->sender_ssrc == remoteSSRC)
>+    if (it->sender_ssrc == remote_ssrc_)
>       break;
>   }
>-  if (it == report_blocks.end()) {
>-    // We have not received packets with SSRC matching the report blocks.
>-    // To calculate RTT we try with the SSRC of the first report block.
>-    // This is very important for send-only channels where we don't know
>-    // the SSRC of the other end.
>-    remoteSSRC = report_blocks[0].sender_ssrc;
>-  }
>+
>+  // If we have not received packets with SSRC matching the report blocks, use
>+  // the SSRC of the first report block for calculating the RTT. This is very
>+  // important for send-only channels where we don't know the SSRC of the other
>+  // end.
>+  uint32_t ssrc =
>+      (it == report_blocks.end()) ? report_blocks[0].sender_ssrc : remote_ssrc_;
> 
>   int64_t avg_rtt = 0;
>   int64_t max_rtt = 0;
>   int64_t min_rtt = 0;
>-  if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) !=
>-      0) {
>+  if (_rtpRtcpModule->RTT(ssrc, &rtt, &avg_rtt, &min_rtt, &max_rtt) != 0) {
>     return 0;
>   }
>   return rtt;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/voice_engine/channel.h b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/channel.h
>similarity index 83%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/voice_engine/channel.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/audio/channel.h
>index 3d6dd8f226c..98f26134925 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/voice_engine/channel.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/channel.h
>@@ -8,28 +8,37 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#ifndef VOICE_ENGINE_CHANNEL_H_
>-#define VOICE_ENGINE_CHANNEL_H_
>+#ifndef AUDIO_CHANNEL_H_
>+#define AUDIO_CHANNEL_H_
> 
>+#include <map>
> #include <memory>
>+#include <string>
>+#include <vector>
> 
>+#include "absl/types/optional.h"
> #include "api/audio/audio_mixer.h"
> #include "api/audio_codecs/audio_encoder.h"
> #include "api/call/audio_sink.h"
> #include "api/call/transport.h"
>-#include "api/optional.h"
>+#include "api/rtpreceiverinterface.h"
>+#include "audio/audio_level.h"
>+#include "call/syncable.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/audio_coding/include/audio_coding_module.h"
> #include "modules/audio_processing/rms_level.h"
> #include "modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
> #include "modules/rtp_rtcp/include/rtp_header_parser.h"
>-#include "modules/rtp_rtcp/include/rtp_receiver.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp.h"
>+#include "modules/rtp_rtcp/source/contributing_sources.h"
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/event.h"
> #include "rtc_base/task_queue.h"
> #include "rtc_base/thread_checker.h"
>-#include "voice_engine/audio_level.h"
>+
>+// TODO(solenberg, nisse): This file contains a few NOLINT marks, to silence
>+// warnings about use of unsigned short, and non-const reference arguments.
>+// These need cleanup, in a separate cl.
> 
> namespace rtc {
> class TimestampWrapAroundHandler;
>@@ -44,17 +53,14 @@ class RateLimiter;
> class ReceiveStatistics;
> class RemoteNtpTimeEstimator;
> class RtcEventLog;
>-class RTPPayloadRegistry;
>-class RTPReceiverAudio;
> class RtpPacketReceived;
> class RtpRtcp;
> class RtpTransportControllerSendInterface;
>-class TelephoneEventHandler;
> 
> struct SenderInfo;
> 
> struct CallStatistics {
>-  unsigned short fractionLost;
>+  unsigned short fractionLost;  // NOLINT
>   unsigned int cumulativeLost;
>   unsigned int extendedMax;
>   unsigned int jitterSamples;
>@@ -73,7 +79,7 @@ struct ReportBlock {
>   uint32_t sender_SSRC;  // SSRC of sender
>   uint32_t source_SSRC;
>   uint8_t fraction_lost;
>-  uint32_t cumulative_num_packets_lost;
>+  int32_t cumulative_num_packets_lost;
>   uint32_t extended_highest_sequence_number;
>   uint32_t interarrival_jitter;
>   uint32_t last_SR_timestamp;
>@@ -82,8 +88,6 @@ struct ReportBlock {
> 
> namespace voe {
> 
>-class RtcEventLogProxy;
>-class RtcpRttStatsProxy;
> class RtpPacketSenderProxy;
> class TransportFeedbackProxy;
> class TransportSequenceNumberProxy;
>@@ -131,7 +135,6 @@ class ChannelState {
> 
> class Channel
>     : public RtpData,
>-      public RtpFeedback,
>       public Transport,
>       public AudioPacketizationCallback,  // receive encoded packets from the
>                                           // ACM
>@@ -144,13 +147,19 @@ class Channel
>   // Used for send streams.
>   Channel(rtc::TaskQueue* encoder_queue,
>           ProcessThread* module_process_thread,
>-          AudioDeviceModule* audio_device_module);
>+          AudioDeviceModule* audio_device_module,
>+          RtcpRttStats* rtcp_rtt_stats,
>+          RtcEventLog* rtc_event_log);
>   // Used for receive streams.
>   Channel(ProcessThread* module_process_thread,
>           AudioDeviceModule* audio_device_module,
>+          RtcpRttStats* rtcp_rtt_stats,
>+          RtcEventLog* rtc_event_log,
>+          uint32_t remote_ssrc,
>           size_t jitter_buffer_max_packets,
>           bool jitter_buffer_fast_playout,
>-          rtc::scoped_refptr<AudioDecoderFactory> decoder_factory);
>+          rtc::scoped_refptr<AudioDecoderFactory> decoder_factory,
>+          absl::optional<AudioCodecPairId> codec_pair_id);
>   virtual ~Channel();
> 
>   void SetSink(AudioSinkInterface* sink);
>@@ -171,7 +180,7 @@ class Channel
>   void StopSend();
> 
>   // Codecs
>-  int32_t GetRecCodec(CodecInst& codec);
>+  int32_t GetRecCodec(CodecInst& codec);  // NOLINT
>   void SetBitRate(int bitrate_bps, int64_t probing_interval_ms);
>   bool EnableAudioNetworkAdaptor(const std::string& config_string);
>   void DisableAudioNetworkAdaptor();
>@@ -187,7 +196,6 @@ class Channel
>   // Muting, Volume and Level.
>   void SetInputMute(bool enable);
>   void SetChannelOutputVolumeScaling(float scaling);
>-  int GetSpeechOutputLevel() const;
>   int GetSpeechOutputLevelFullRange() const;
>   // See description of "totalAudioEnergy" in the WebRTC stats spec:
>   // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy
>@@ -195,22 +203,28 @@ class Channel
>   double GetTotalOutputDuration() const;
> 
>   // Stats.
>-  int GetNetworkStatistics(NetworkStatistics& stats);
>+  int GetNetworkStatistics(NetworkStatistics& stats);  // NOLINT
>   void GetDecodingCallStatistics(AudioDecodingCallStats* stats) const;
>   ANAStats GetANAStatistics() const;
> 
>   // Audio+Video Sync.
>   uint32_t GetDelayEstimate() const;
>   int SetMinimumPlayoutDelay(int delayMs);
>-  int GetPlayoutTimestamp(unsigned int& timestamp);
>-  int GetRtpRtcp(RtpRtcp** rtpRtcpModule, RtpReceiver** rtp_receiver) const;
>+  int GetPlayoutTimestamp(unsigned int& timestamp);  // NOLINT
>+
>+  // Used by AudioSendStream.
>+  RtpRtcp* GetRtpRtcp() const;
> 
>+  // Produces the transport-related timestamps; current_delay_ms is left unset.
>+  absl::optional<Syncable::Info> GetSyncInfo() const;
>   // DTMF.
>   int SendTelephoneEventOutband(int event, int duration_ms);
>   int SetSendTelephoneEventPayloadType(int payload_type, int payload_frequency);
> 
>   // RTP+RTCP
>   int SetLocalSSRC(unsigned int ssrc);
>+
>+  void SetMid(const std::string& mid, int extension_id);
>   int SetSendAudioLevelIndicationStatus(bool enable, unsigned char id);
>   void EnableSendTransportSequenceNumber(int id);
> 
>@@ -223,7 +237,7 @@ class Channel
>   void SetRTCPStatus(bool enable);
>   int SetRTCP_CNAME(const char cName[256]);
>   int GetRemoteRTCPReportBlocks(std::vector<ReportBlock>* report_blocks);
>-  int GetRTPStatistics(CallStatistics& stats);
>+  int GetRTPStatistics(CallStatistics& stats);  // NOLINT
>   void SetNACKStatus(bool enable, int maxNumberOfPackets);
> 
>   // From AudioPacketizationCallback in the ACM
>@@ -239,13 +253,6 @@ class Channel
>                                 size_t payloadSize,
>                                 const WebRtcRTPHeader* rtpHeader) override;
> 
>-  // From RtpFeedback in the RTP/RTCP module
>-  int32_t OnInitializeDecoder(int payload_type,
>-                              const SdpAudioFormat& audio_format,
>-                              uint32_t rate) override;
>-  void OnIncomingSSRCChanged(uint32_t ssrc) override;
>-  void OnIncomingCSRCChanged(uint32_t CSRC, bool added) override;
>-
>   // From Transport (called by the RTP/RTCP module)
>   bool SendRtp(const uint8_t* data,
>                size_t len,
>@@ -262,7 +269,6 @@ class Channel
>   bool Playing() const { return channel_state_.Get().playing; }
>   bool Sending() const { return channel_state_.Get().sending; }
>   RtpRtcp* RtpRtcpModulePtr() const { return _rtpRtcpModule.get(); }
>-  int8_t OutputEnergyLevel() const { return _outputAudioLevel.Level(); }
> 
>   // ProcessAndEncodeAudio() posts a task on the shared encoder task queue,
>   // which in turn calls (on the queue) ProcessAndEncodeAudioOnTaskQueue() where
>@@ -279,10 +285,6 @@ class Channel
>   // Used for obtaining RTT for a receive-only channel.
>   void SetAssociatedSendChannel(Channel* channel);
> 
>-  // Set a RtcEventLog logging object.
>-  void SetRtcEventLog(RtcEventLog* event_log);
>-
>-  void SetRtcpRttStats(RtcpRttStats* rtcp_rtt_stats);
>   void SetTransportOverhead(size_t transport_overhead_per_packet);
> 
>   // From OverheadObserver in the RTP/RTCP module
>@@ -296,9 +298,7 @@ class Channel
> 
>   void OnRecoverableUplinkPacketLossRate(float recoverable_packet_loss_rate);
> 
>-  std::vector<RtpSource> GetSources() const {
>-    return rtp_receiver_->GetSources();
>-  }
>+  std::vector<RtpSource> GetSources() const;
> 
>  private:
>   class ProcessAndEncodeAudioTask;
>@@ -306,15 +306,14 @@ class Channel
>   void Init();
>   void Terminate();
> 
>-  int GetRemoteSSRC(unsigned int& ssrc);
>+  int GetRemoteSSRC(unsigned int& ssrc);  // NOLINT
>   void OnUplinkPacketLossRate(float packet_loss_rate);
>   bool InputMute() const;
> 
>   bool ReceivePacket(const uint8_t* packet,
>                      size_t packet_length,
>                      const RTPHeader& header);
>-  bool IsPacketInOrder(const RTPHeader& header) const;
>-  bool IsPacketRetransmitted(const RTPHeader& header, bool in_order) const;
>+  bool IsPacketRetransmitted(const RTPHeader& header) const;
>   int ResendPackets(const uint16_t* sequence_numbers, int length);
>   void UpdatePlayoutTimestamp(bool rtcp);
> 
>@@ -337,23 +336,35 @@ class Channel
> 
>   ChannelState channel_state_;
> 
>-  std::unique_ptr<voe::RtcEventLogProxy> event_log_proxy_;
>-  std::unique_ptr<voe::RtcpRttStatsProxy> rtcp_rtt_stats_proxy_;
>+  RtcEventLog* const event_log_;
>+
>+  // Indexed by payload type.
>+  std::map<uint8_t, int> payload_type_frequencies_;
> 
>-  std::unique_ptr<RTPPayloadRegistry> rtp_payload_registry_;
>   std::unique_ptr<ReceiveStatistics> rtp_receive_statistics_;
>-  std::unique_ptr<RtpReceiver> rtp_receiver_;
>-  TelephoneEventHandler* telephone_event_handler_;
>   std::unique_ptr<RtpRtcp> _rtpRtcpModule;
>+  const uint32_t remote_ssrc_;
>+
>+  // Info for GetSources and GetSyncInfo is updated on network or worker thread,
>+  // queried on the worker thread.
>+  rtc::CriticalSection rtp_sources_lock_;
>+  ContributingSources contributing_sources_ RTC_GUARDED_BY(&rtp_sources_lock_);
>+  absl::optional<uint32_t> last_received_rtp_timestamp_
>+      RTC_GUARDED_BY(&rtp_sources_lock_);
>+  absl::optional<int64_t> last_received_rtp_system_time_ms_
>+      RTC_GUARDED_BY(&rtp_sources_lock_);
>+  absl::optional<uint8_t> last_received_rtp_audio_level_
>+      RTC_GUARDED_BY(&rtp_sources_lock_);
>+
>   std::unique_ptr<AudioCodingModule> audio_coding_;
>   AudioSinkInterface* audio_sink_ = nullptr;
>   AudioLevel _outputAudioLevel;
>-  uint32_t _timeStamp RTC_ACCESS_ON(encoder_queue_);
>+  uint32_t _timeStamp RTC_GUARDED_BY(encoder_queue_);
> 
>   RemoteNtpTimeEstimator ntp_estimator_ RTC_GUARDED_BY(ts_stats_lock_);
> 
>   // Timestamp of the audio pulled from NetEq.
>-  rtc::Optional<uint32_t> jitter_buffer_playout_timestamp_;
>+  absl::optional<uint32_t> jitter_buffer_playout_timestamp_;
> 
>   rtc::CriticalSection video_sync_lock_;
>   uint32_t playout_timestamp_rtp_ RTC_GUARDED_BY(video_sync_lock_);
>@@ -373,9 +384,9 @@ class Channel
>   ProcessThread* _moduleProcessThreadPtr;
>   AudioDeviceModule* _audioDeviceModulePtr;
>   Transport* _transportPtr;  // WebRtc socket or external transport
>-  RmsLevel rms_level_ RTC_ACCESS_ON(encoder_queue_);
>+  RmsLevel rms_level_ RTC_GUARDED_BY(encoder_queue_);
>   bool input_mute_ RTC_GUARDED_BY(volume_settings_critsect_);
>-  bool previous_frame_muted_ RTC_ACCESS_ON(encoder_queue_);
>+  bool previous_frame_muted_ RTC_GUARDED_BY(encoder_queue_);
>   float _outputGain RTC_GUARDED_BY(volume_settings_critsect_);
>   // VoeRTP_RTCP
>   // TODO(henrika): can today be accessed on the main thread and on the
>@@ -410,4 +421,4 @@ class Channel
> }  // namespace voe
> }  // namespace webrtc
> 
>-#endif  // VOICE_ENGINE_CHANNEL_H_
>+#endif  // AUDIO_CHANNEL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/voice_engine/channel_proxy.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/channel_proxy.cc
>similarity index 91%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/voice_engine/channel_proxy.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/audio/channel_proxy.cc
>index 70564d4c911..5e8181a7031 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/voice_engine/channel_proxy.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/channel_proxy.cc
>@@ -8,7 +8,7 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "voice_engine/channel_proxy.h"
>+#include "audio/channel_proxy.h"
> 
> #include <utility>
> 
>@@ -22,8 +22,8 @@ namespace webrtc {
> namespace voe {
> ChannelProxy::ChannelProxy() {}
> 
>-ChannelProxy::ChannelProxy(std::unique_ptr<Channel> channel) :
>-    channel_(std::move(channel)) {
>+ChannelProxy::ChannelProxy(std::unique_ptr<Channel> channel)
>+    : channel_(std::move(channel)) {
>   RTC_DCHECK(channel_);
>   module_process_thread_checker_.DetachFromThread();
> }
>@@ -53,6 +53,11 @@ void ChannelProxy::SetLocalSSRC(uint32_t ssrc) {
>   RTC_DCHECK_EQ(0, error);
> }
> 
>+void ChannelProxy::SetMid(const std::string& mid, int extension_id) {
>+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>+  channel_->SetMid(mid, extension_id);
>+}
>+
> void ChannelProxy::SetRTCP_CNAME(const std::string& c_name) {
>   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>   // Note: VoERTP_RTCP::SetRTCP_CNAME() accepts a char[256] array.
>@@ -82,7 +87,7 @@ void ChannelProxy::RegisterSenderCongestionControlObjects(
>     RtcpBandwidthObserver* bandwidth_observer) {
>   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>   channel_->RegisterSenderCongestionControlObjects(transport,
>-                                                    bandwidth_observer);
>+                                                   bandwidth_observer);
> }
> 
> void ChannelProxy::RegisterReceiverCongestionControlObjects(
>@@ -137,11 +142,6 @@ ANAStats ChannelProxy::GetANAStatistics() const {
>   return channel_->GetANAStatistics();
> }
> 
>-int ChannelProxy::GetSpeechOutputLevel() const {
>-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  return channel_->GetSpeechOutputLevel();
>-}
>-
> int ChannelProxy::GetSpeechOutputLevelFullRange() const {
>   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>   return channel_->GetSpeechOutputLevelFullRange();
>@@ -167,7 +167,7 @@ bool ChannelProxy::SetSendTelephoneEventPayloadType(int payload_type,
>                                                     int payload_frequency) {
>   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>   return channel_->SetSendTelephoneEventPayloadType(payload_type,
>-                                                     payload_frequency) == 0;
>+                                                    payload_frequency) == 0;
> }
> 
> bool ChannelProxy::SendTelephoneEventOutband(int event, int duration_ms) {
>@@ -221,11 +221,6 @@ void ChannelProxy::SetChannelOutputVolumeScaling(float scaling) {
>   channel_->SetChannelOutputVolumeScaling(scaling);
> }
> 
>-void ChannelProxy::SetRtcEventLog(RtcEventLog* event_log) {
>-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  channel_->SetRtcEventLog(event_log);
>-}
>-
> AudioMixer::Source::AudioFrameInfo ChannelProxy::GetAudioFrameWithInfo(
>     int sample_rate_hz,
>     AudioFrame* audio_frame) {
>@@ -260,13 +255,14 @@ void ChannelProxy::DisassociateSendChannel() {
>   channel_->SetAssociatedSendChannel(nullptr);
> }
> 
>-void ChannelProxy::GetRtpRtcp(RtpRtcp** rtp_rtcp,
>-                              RtpReceiver** rtp_receiver) const {
>+RtpRtcp* ChannelProxy::GetRtpRtcp() const {
>   RTC_DCHECK(module_process_thread_checker_.CalledOnValidThread());
>-  RTC_DCHECK(rtp_rtcp);
>-  RTC_DCHECK(rtp_receiver);
>-  int error = channel_->GetRtpRtcp(rtp_rtcp, rtp_receiver);
>-  RTC_DCHECK_EQ(0, error);
>+  return channel_->GetRtpRtcp();
>+}
>+
>+absl::optional<Syncable::Info> ChannelProxy::GetSyncInfo() const {
>+  RTC_DCHECK(module_process_thread_checker_.CalledOnValidThread());
>+  return channel_->GetSyncInfo();
> }
> 
> uint32_t ChannelProxy::GetPlayoutTimestamp() const {
>@@ -288,11 +284,6 @@ void ChannelProxy::SetMinimumPlayoutDelay(int delay_ms) {
>   }
> }
> 
>-void ChannelProxy::SetRtcpRttStats(RtcpRttStats* rtcp_rtt_stats) {
>-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  channel_->SetRtcpRttStats(rtcp_rtt_stats);
>-}
>-
> bool ChannelProxy::GetRecCodec(CodecInst* codec_inst) const {
>   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>   return channel_->GetRecCodec(*codec_inst) == 0;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/voice_engine/channel_proxy.h b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/channel_proxy.h
>similarity index 92%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/voice_engine/channel_proxy.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/audio/channel_proxy.h
>index ef81174c0ea..f82c1fd2811 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/voice_engine/channel_proxy.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/channel_proxy.h
>@@ -8,9 +8,10 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#ifndef VOICE_ENGINE_CHANNEL_PROXY_H_
>-#define VOICE_ENGINE_CHANNEL_PROXY_H_
>+#ifndef AUDIO_CHANNEL_PROXY_H_
>+#define AUDIO_CHANNEL_PROXY_H_
> 
>+#include <map>
> #include <memory>
> #include <string>
> #include <vector>
>@@ -18,11 +19,11 @@
> #include "api/audio/audio_mixer.h"
> #include "api/audio_codecs/audio_encoder.h"
> #include "api/rtpreceiverinterface.h"
>+#include "audio/channel.h"
> #include "call/rtp_packet_sink_interface.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/race_checker.h"
> #include "rtc_base/thread_checker.h"
>-#include "voice_engine/channel.h"
> 
> namespace webrtc {
> 
>@@ -33,7 +34,6 @@ class RtcpBandwidthObserver;
> class RtcpRttStats;
> class RtpPacketSender;
> class RtpPacketReceived;
>-class RtpReceiver;
> class RtpRtcp;
> class RtpTransportControllerSendInterface;
> class Transport;
>@@ -61,6 +61,7 @@ class ChannelProxy : public RtpPacketSinkInterface {
> 
>   virtual void SetRTCPStatus(bool enable);
>   virtual void SetLocalSSRC(uint32_t ssrc);
>+  virtual void SetMid(const std::string& mid, int extension_id);
>   virtual void SetRTCP_CNAME(const std::string& c_name);
>   virtual void SetNACKStatus(bool enable, int max_packets);
>   virtual void SetSendAudioLevelIndicationStatus(bool enable, int id);
>@@ -77,7 +78,6 @@ class ChannelProxy : public RtpPacketSinkInterface {
>   virtual NetworkStatistics GetNetworkStatistics() const;
>   virtual AudioDecodingCallStats GetDecodingCallStatistics() const;
>   virtual ANAStats GetANAStatistics() const;
>-  virtual int GetSpeechOutputLevel() const;
>   virtual int GetSpeechOutputLevelFullRange() const;
>   // See description of "totalAudioEnergy" in the WebRTC stats spec:
>   // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy
>@@ -97,7 +97,6 @@ class ChannelProxy : public RtpPacketSinkInterface {
>   void OnRtpPacket(const RtpPacketReceived& packet) override;
>   virtual bool ReceivedRTCPPacket(const uint8_t* packet, size_t length);
>   virtual void SetChannelOutputVolumeScaling(float scaling);
>-  virtual void SetRtcEventLog(RtcEventLog* event_log);
>   virtual AudioMixer::Source::AudioFrameInfo GetAudioFrameWithInfo(
>       int sample_rate_hz,
>       AudioFrame* audio_frame);
>@@ -106,11 +105,12 @@ class ChannelProxy : public RtpPacketSinkInterface {
>   virtual void SetTransportOverhead(int transport_overhead_per_packet);
>   virtual void AssociateSendChannel(const ChannelProxy& send_channel_proxy);
>   virtual void DisassociateSendChannel();
>-  virtual void GetRtpRtcp(RtpRtcp** rtp_rtcp,
>-                          RtpReceiver** rtp_receiver) const;
>+  virtual RtpRtcp* GetRtpRtcp() const;
>+
>+  // Produces the transport-related timestamps; current_delay_ms is left unset.
>+  absl::optional<Syncable::Info> GetSyncInfo() const;
>   virtual uint32_t GetPlayoutTimestamp() const;
>   virtual void SetMinimumPlayoutDelay(int delay_ms);
>-  virtual void SetRtcpRttStats(RtcpRttStats* rtcp_rtt_stats);
>   virtual bool GetRecCodec(CodecInst* codec_inst) const;
>   virtual void OnTwccBasedUplinkPacketLossRate(float packet_loss_rate);
>   virtual void OnRecoverableUplinkPacketLossRate(
>@@ -141,4 +141,4 @@ class ChannelProxy : public RtpPacketSinkInterface {
> }  // namespace voe
> }  // namespace webrtc
> 
>-#endif  // VOICE_ENGINE_CHANNEL_PROXY_H_
>+#endif  // AUDIO_CHANNEL_PROXY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/mock_voe_channel_proxy.h b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/mock_voe_channel_proxy.h
>similarity index 71%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/test/mock_voe_channel_proxy.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/audio/mock_voe_channel_proxy.h
>index 9bc48b10a09..f6a2637c420 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/mock_voe_channel_proxy.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/mock_voe_channel_proxy.h
>@@ -8,28 +8,30 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#ifndef TEST_MOCK_VOE_CHANNEL_PROXY_H_
>-#define TEST_MOCK_VOE_CHANNEL_PROXY_H_
>+#ifndef AUDIO_MOCK_VOE_CHANNEL_PROXY_H_
>+#define AUDIO_MOCK_VOE_CHANNEL_PROXY_H_
> 
>+#include <map>
>+#include <memory>
> #include <string>
>+#include <vector>
> 
>+#include "audio/channel_proxy.h"
> #include "modules/rtp_rtcp/source/rtp_packet_received.h"
> #include "test/gmock.h"
>-#include "voice_engine/channel_proxy.h"
> 
> namespace webrtc {
> namespace test {
> 
> class MockVoEChannelProxy : public voe::ChannelProxy {
>  public:
>-  // GTest doesn't like move-only types, like std::unique_ptr
>-  bool SetEncoder(int payload_type,
>-                  std::unique_ptr<AudioEncoder> encoder) {
>+  // GMock doesn't like move-only types, like std::unique_ptr.
>+  virtual bool SetEncoder(int payload_type,
>+                          std::unique_ptr<AudioEncoder> encoder) {
>     return SetEncoderForMock(payload_type, &encoder);
>   }
>   MOCK_METHOD2(SetEncoderForMock,
>-               bool(int payload_type,
>-                    std::unique_ptr<AudioEncoder>* encoder));
>+               bool(int payload_type, std::unique_ptr<AudioEncoder>* encoder));
>   MOCK_METHOD1(
>       ModifyEncoder,
>       void(rtc::FunctionView<void(std::unique_ptr<AudioEncoder>*)> modifier));
>@@ -38,9 +40,7 @@ class MockVoEChannelProxy : public voe::ChannelProxy {
>   MOCK_METHOD1(SetRTCP_CNAME, void(const std::string& c_name));
>   MOCK_METHOD2(SetNACKStatus, void(bool enable, int max_packets));
>   MOCK_METHOD2(SetSendAudioLevelIndicationStatus, void(bool enable, int id));
>-  MOCK_METHOD2(SetReceiveAudioLevelIndicationStatus, void(bool enable, int id));
>   MOCK_METHOD1(EnableSendTransportSequenceNumber, void(int id));
>-  MOCK_METHOD1(EnableReceiveTransportSequenceNumber, void(int id));
>   MOCK_METHOD2(RegisterSenderCongestionControlObjects,
>                void(RtpTransportControllerSendInterface* transport,
>                     RtcpBandwidthObserver* bandwidth_observer));
>@@ -53,36 +53,35 @@ class MockVoEChannelProxy : public voe::ChannelProxy {
>   MOCK_CONST_METHOD0(GetNetworkStatistics, NetworkStatistics());
>   MOCK_CONST_METHOD0(GetDecodingCallStatistics, AudioDecodingCallStats());
>   MOCK_CONST_METHOD0(GetANAStatistics, ANAStats());
>-  MOCK_CONST_METHOD0(GetSpeechOutputLevel, int());
>   MOCK_CONST_METHOD0(GetSpeechOutputLevelFullRange, int());
>   MOCK_CONST_METHOD0(GetTotalOutputEnergy, double());
>   MOCK_CONST_METHOD0(GetTotalOutputDuration, double());
>   MOCK_CONST_METHOD0(GetDelayEstimate, uint32_t());
>-  MOCK_METHOD2(SetSendTelephoneEventPayloadType, bool(int payload_type,
>-                                                      int payload_frequency));
>+  MOCK_METHOD2(SetSendTelephoneEventPayloadType,
>+               bool(int payload_type, int payload_frequency));
>   MOCK_METHOD2(SendTelephoneEventOutband, bool(int event, int duration_ms));
>   MOCK_METHOD2(SetBitrate, void(int bitrate_bps, int64_t probing_interval_ms));
>-  // TODO(solenberg): Talk the compiler into accepting this mock method:
>-  // MOCK_METHOD1(SetSink, void(std::unique_ptr<AudioSinkInterface> sink));
>+  MOCK_METHOD1(SetSink, void(AudioSinkInterface* sink));
>   MOCK_METHOD1(SetInputMute, void(bool muted));
>   MOCK_METHOD1(RegisterTransport, void(Transport* transport));
>   MOCK_METHOD1(OnRtpPacket, void(const RtpPacketReceived& packet));
>   MOCK_METHOD2(ReceivedRTCPPacket, bool(const uint8_t* packet, size_t length));
>-  MOCK_CONST_METHOD0(GetAudioDecoderFactory,
>-                     const rtc::scoped_refptr<AudioDecoderFactory>&());
>   MOCK_METHOD1(SetChannelOutputVolumeScaling, void(float scaling));
>-  MOCK_METHOD1(SetRtcEventLog, void(RtcEventLog* event_log));
>-  MOCK_METHOD1(SetRtcpRttStats, void(RtcpRttStats* rtcp_rtt_stats));
>   MOCK_METHOD2(GetAudioFrameWithInfo,
>-      AudioMixer::Source::AudioFrameInfo(int sample_rate_hz,
>-                                         AudioFrame* audio_frame));
>+               AudioMixer::Source::AudioFrameInfo(int sample_rate_hz,
>+                                                  AudioFrame* audio_frame));
>   MOCK_CONST_METHOD0(PreferredSampleRate, int());
>+  // GMock doesn't like move-only types, like std::unique_ptr.
>+  virtual void ProcessAndEncodeAudio(std::unique_ptr<AudioFrame> audio_frame) {
>+    ProcessAndEncodeAudioForMock(&audio_frame);
>+  }
>+  MOCK_METHOD1(ProcessAndEncodeAudioForMock,
>+               void(std::unique_ptr<AudioFrame>* audio_frame));
>   MOCK_METHOD1(SetTransportOverhead, void(int transport_overhead_per_packet));
>   MOCK_METHOD1(AssociateSendChannel,
>                void(const ChannelProxy& send_channel_proxy));
>   MOCK_METHOD0(DisassociateSendChannel, void());
>-  MOCK_CONST_METHOD2(GetRtpRtcp, void(RtpRtcp** rtp_rtcp,
>-                                      RtpReceiver** rtp_receiver));
>+  MOCK_CONST_METHOD0(GetRtpRtcp, RtpRtcp*());
>   MOCK_CONST_METHOD0(GetPlayoutTimestamp, uint32_t());
>   MOCK_METHOD1(SetMinimumPlayoutDelay, void(int delay_ms));
>   MOCK_CONST_METHOD1(GetRecCodec, bool(CodecInst* codec_inst));
>@@ -92,8 +91,12 @@ class MockVoEChannelProxy : public voe::ChannelProxy {
>   MOCK_METHOD1(OnRecoverableUplinkPacketLossRate,
>                void(float recoverable_packet_loss_rate));
>   MOCK_CONST_METHOD0(GetSources, std::vector<RtpSource>());
>+  MOCK_METHOD0(StartSend, void());
>+  MOCK_METHOD0(StopSend, void());
>+  MOCK_METHOD0(StartPlayout, void());
>+  MOCK_METHOD0(StopPlayout, void());
> };
> }  // namespace test
> }  // namespace webrtc
> 
>-#endif  // TEST_MOCK_VOE_CHANNEL_PROXY_H_
>+#endif  // AUDIO_MOCK_VOE_CHANNEL_PROXY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/voice_engine/utility.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/remix_resample.cc
>similarity index 92%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/voice_engine/utility.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/audio/remix_resample.cc
>index 5caa73ae451..97222e947a4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/voice_engine/utility.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/remix_resample.cc
>@@ -8,13 +8,13 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "voice_engine/utility.h"
>+#include "audio/remix_resample.h"
> 
>+#include "api/audio/audio_frame.h"
> #include "audio/utility/audio_frame_operations.h"
> #include "common_audio/resampler/include/push_resampler.h"
> #include "common_audio/signal_processing/include/signal_processing_library.h"
> #include "common_types.h"  // NOLINT(build/include)
>-#include "modules/include/module_common_types.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> 
>@@ -68,9 +68,9 @@ void RemixAndResample(const int16_t* src_data,
>   // how much to zero here; or 2) make resampler accept a hint that the input is
>   // zeroed.
>   const size_t src_length = samples_per_channel * audio_ptr_num_channels;
>-  int out_length = resampler->Resample(audio_ptr, src_length,
>-                                       dst_frame->mutable_data(),
>-                                       AudioFrame::kMaxDataSizeSamples);
>+  int out_length =
>+      resampler->Resample(audio_ptr, src_length, dst_frame->mutable_data(),
>+                          AudioFrame::kMaxDataSizeSamples);
>   if (out_length == -1) {
>     RTC_FATAL() << "Resample failed: audio_ptr = " << audio_ptr
>             << ", src_length = " << src_length
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/voice_engine/utility.h b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/remix_resample.h
>similarity index 86%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/voice_engine/utility.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/audio/remix_resample.h
>index dc23e1667dc..a45270b39a6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/voice_engine/utility.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/remix_resample.h
>@@ -8,20 +8,13 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-/*
>- *  Contains functions often used by different parts of VoiceEngine.
>- */
>-
>-#ifndef VOICE_ENGINE_UTILITY_H_
>-#define VOICE_ENGINE_UTILITY_H_
>+#ifndef AUDIO_REMIX_RESAMPLE_H_
>+#define AUDIO_REMIX_RESAMPLE_H_
> 
>+#include "api/audio/audio_frame.h"
> #include "common_audio/resampler/include/push_resampler.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
>-
>-class AudioFrame;
>-
> namespace voe {
> 
> // Upmix or downmix and resample the audio to |dst_frame|. Expects |dst_frame|
>@@ -48,4 +41,4 @@ void RemixAndResample(const int16_t* src_data,
> }  // namespace voe
> }  // namespace webrtc
> 
>-#endif  // VOICE_ENGINE_UTILITY_H_
>+#endif  // AUDIO_REMIX_RESAMPLE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/voice_engine/utility_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/remix_resample_unittest.cc
>similarity index 94%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/voice_engine/utility_unittest.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/audio/remix_resample_unittest.cc
>index c798582d4b0..f1fb5f7dd05 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/voice_engine/utility_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/remix_resample_unittest.cc
>@@ -10,12 +10,12 @@
> 
> #include <math.h>
> 
>+#include "audio/remix_resample.h"
> #include "common_audio/resampler/include/push_resampler.h"
>-#include "modules/include/module_common_types.h"
> #include "rtc_base/arraysize.h"
>+#include "rtc_base/checks.h"
> #include "rtc_base/format_macros.h"
> #include "test/gtest.h"
>-#include "voice_engine/utility.h"
> 
> namespace webrtc {
> namespace voe {
>@@ -113,7 +113,8 @@ void VerifyParams(const AudioFrame& ref_frame, const AudioFrame& test_frame) {
> // Computes the best SNR based on the error between |ref_frame| and
> // |test_frame|. It allows for up to a |max_delay| in samples between the
> // signals to compensate for the resampling delay.
>-float ComputeSNR(const AudioFrame& ref_frame, const AudioFrame& test_frame,
>+float ComputeSNR(const AudioFrame& ref_frame,
>+                 const AudioFrame& test_frame,
>                  size_t max_delay) {
>   VerifyParams(ref_frame, test_frame);
>   float best_snr = 0;
>@@ -123,8 +124,9 @@ float ComputeSNR(const AudioFrame& ref_frame, const AudioFrame& test_frame,
>     float variance = 0;
>     const int16_t* ref_frame_data = ref_frame.data();
>     const int16_t* test_frame_data = test_frame.data();
>-    for (size_t i = 0; i < ref_frame.samples_per_channel_ *
>-        ref_frame.num_channels_ - delay; i++) {
>+    for (size_t i = 0;
>+         i < ref_frame.samples_per_channel_ * ref_frame.num_channels_ - delay;
>+         i++) {
>       int error = ref_frame_data[i] - test_frame_data[i + delay];
>       mse += error * error;
>       variance += ref_frame_data[i] * ref_frame_data[i];
>@@ -145,7 +147,7 @@ void VerifyFramesAreEqual(const AudioFrame& ref_frame,
>                           const AudioFrame& test_frame) {
>   VerifyParams(ref_frame, test_frame);
>   const int16_t* ref_frame_data = ref_frame.data();
>-  const int16_t* test_frame_data  = test_frame.data();
>+  const int16_t* test_frame_data = test_frame.data();
>   for (size_t i = 0;
>        i < ref_frame.samples_per_channel_ * ref_frame.num_channels_; i++) {
>     EXPECT_EQ(ref_frame_data[i], test_frame_data[i]);
>@@ -161,8 +163,8 @@ void UtilityTest::RunResampleTest(int src_channels,
>   const int16_t kSrcCh2 = 15;
>   const int16_t kSrcCh3 = 22;
>   const int16_t kSrcCh4 = 8;
>-  const float resampling_factor = (1.0 * src_sample_rate_hz) /
>-      dst_sample_rate_hz;
>+  const float resampling_factor =
>+      (1.0 * src_sample_rate_hz) / dst_sample_rate_hz;
>   const float dst_ch1 = resampling_factor * kSrcCh1;
>   const float dst_ch2 = resampling_factor * kSrcCh2;
>   const float dst_ch3 = resampling_factor * kSrcCh3;
>@@ -206,7 +208,7 @@ void UtilityTest::RunResampleTest(int src_channels,
>       static_cast<double>(dst_sample_rate_hz) / src_sample_rate_hz *
>       kInputKernelDelaySamples * dst_channels * 2);
>   printf("(%d, %d Hz) -> (%d, %d Hz) ",  // SNR reported on the same line later.
>-      src_channels, src_sample_rate_hz, dst_channels, dst_sample_rate_hz);
>+         src_channels, src_sample_rate_hz, dst_channels, dst_sample_rate_hz);
>   RemixAndResample(src_frame_, &resampler, &dst_frame_);
> 
>   if (src_sample_rate_hz == 96000 && dst_sample_rate_hz == 8000) {
>@@ -258,8 +260,7 @@ TEST_F(UtilityTest, RemixAndResampleSucceeds) {
> 
>   for (int src_rate = 0; src_rate < kSampleRatesSize; src_rate++) {
>     for (int dst_rate = 0; dst_rate < kSampleRatesSize; dst_rate++) {
>-      for (int src_channel = 0; src_channel < kSrcChannelsSize;
>-           src_channel++) {
>+      for (int src_channel = 0; src_channel < kSrcChannelsSize; src_channel++) {
>         for (int dst_channel = 0; dst_channel < kDstChannelsSize;
>              dst_channel++) {
>           RunResampleTest(kSrcChannels[src_channel], kSampleRates[src_rate],
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/audio_bwe_integration_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/audio_bwe_integration_test.cc
>index f89ced97f4d..4d8ac804c87 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/audio_bwe_integration_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/audio_bwe_integration_test.cc
>@@ -10,8 +10,8 @@
> 
> #include "audio/test/audio_bwe_integration_test.h"
> 
>+#include "absl/memory/memory.h"
> #include "common_audio/wav_file.h"
>-#include "rtc_base/ptr_util.h"
> #include "system_wrappers/include/sleep.h"
> #include "test/field_trial.h"
> #include "test/gtest.h"
>@@ -37,14 +37,14 @@ size_t AudioBweTest::GetNumFlexfecStreams() const {
>   return 0;
> }
> 
>-std::unique_ptr<test::FakeAudioDevice::Capturer>
>+std::unique_ptr<TestAudioDeviceModule::Capturer>
> AudioBweTest::CreateCapturer() {
>-  return test::FakeAudioDevice::CreateWavFileReader(AudioInputFile());
>+  return TestAudioDeviceModule::CreateWavFileReader(AudioInputFile());
> }
> 
> void AudioBweTest::OnFakeAudioDevicesCreated(
>-    test::FakeAudioDevice* send_audio_device,
>-    test::FakeAudioDevice* recv_audio_device) {
>+    TestAudioDeviceModule* send_audio_device,
>+    TestAudioDeviceModule* recv_audio_device) {
>   send_audio_device_ = send_audio_device;
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/audio_bwe_integration_test.h b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/audio_bwe_integration_test.h
>index 0b0cb6c73bd..a88b047993c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/audio_bwe_integration_test.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/audio_bwe_integration_test.h
>@@ -14,7 +14,6 @@
> #include <string>
> 
> #include "test/call_test.h"
>-#include "test/fake_audio_device.h"
> #include "test/single_threaded_task_queue.h"
> 
> namespace webrtc {
>@@ -33,11 +32,11 @@ class AudioBweTest : public test::EndToEndTest {
>   size_t GetNumAudioStreams() const override;
>   size_t GetNumFlexfecStreams() const override;
> 
>-  std::unique_ptr<test::FakeAudioDevice::Capturer> CreateCapturer() override;
>+  std::unique_ptr<TestAudioDeviceModule::Capturer> CreateCapturer() override;
> 
>   void OnFakeAudioDevicesCreated(
>-      test::FakeAudioDevice* send_audio_device,
>-      test::FakeAudioDevice* recv_audio_device) override;
>+      TestAudioDeviceModule* send_audio_device,
>+      TestAudioDeviceModule* recv_audio_device) override;
> 
>   test::PacketTransport* CreateSendTransport(
>       SingleThreadedTaskQueueForTesting* task_queue,
>@@ -48,7 +47,7 @@ class AudioBweTest : public test::EndToEndTest {
>   void PerformTest() override;
> 
>  private:
>-  test::FakeAudioDevice* send_audio_device_;
>+  TestAudioDeviceModule* send_audio_device_;
> };
> 
> }  // namespace test
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/audio_end_to_end_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/audio_end_to_end_test.cc
>index 44bf3f775ac..820b46444e7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/audio_end_to_end_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/audio_end_to_end_test.cc
>@@ -12,7 +12,6 @@
> 
> #include "audio/test/audio_end_to_end_test.h"
> #include "system_wrappers/include/sleep.h"
>-#include "test/fake_audio_device.h"
> #include "test/gtest.h"
> 
> namespace webrtc {
>@@ -43,19 +42,19 @@ size_t AudioEndToEndTest::GetNumFlexfecStreams() const {
>   return 0;
> }
> 
>-std::unique_ptr<test::FakeAudioDevice::Capturer>
>-    AudioEndToEndTest::CreateCapturer() {
>-  return test::FakeAudioDevice::CreatePulsedNoiseCapturer(32000, kSampleRate);
>+std::unique_ptr<TestAudioDeviceModule::Capturer>
>+AudioEndToEndTest::CreateCapturer() {
>+  return TestAudioDeviceModule::CreatePulsedNoiseCapturer(32000, kSampleRate);
> }
> 
>-std::unique_ptr<test::FakeAudioDevice::Renderer>
>-    AudioEndToEndTest::CreateRenderer() {
>-  return test::FakeAudioDevice::CreateDiscardRenderer(kSampleRate);
>+std::unique_ptr<TestAudioDeviceModule::Renderer>
>+AudioEndToEndTest::CreateRenderer() {
>+  return TestAudioDeviceModule::CreateDiscardRenderer(kSampleRate);
> }
> 
> void AudioEndToEndTest::OnFakeAudioDevicesCreated(
>-    test::FakeAudioDevice* send_audio_device,
>-    test::FakeAudioDevice* recv_audio_device) {
>+    TestAudioDeviceModule* send_audio_device,
>+    TestAudioDeviceModule* recv_audio_device) {
>   send_audio_device_ = send_audio_device;
> }
> 
>@@ -68,15 +67,15 @@ test::PacketTransport* AudioEndToEndTest::CreateSendTransport(
> }
> 
> test::PacketTransport* AudioEndToEndTest::CreateReceiveTransport(
>-      SingleThreadedTaskQueueForTesting* task_queue) {
>+    SingleThreadedTaskQueueForTesting* task_queue) {
>   return new test::PacketTransport(
>       task_queue, nullptr, this, test::PacketTransport::kReceiver,
>       test::CallTest::payload_type_map_, GetNetworkPipeConfig());
> }
> 
> void AudioEndToEndTest::ModifyAudioConfigs(
>-  AudioSendStream::Config* send_config,
>-  std::vector<AudioReceiveStream::Config>* receive_configs) {
>+    AudioSendStream::Config* send_config,
>+    std::vector<AudioReceiveStream::Config>* receive_configs) {
>   // Large bitrate by default.
>   const webrtc::SdpAudioFormat kDefaultFormat("opus", 48000, 2,
>                                               {{"stereo", "1"}});
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/audio_end_to_end_test.h b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/audio_end_to_end_test.h
>index 921de844833..cbe25f5b3a9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/audio_end_to_end_test.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/audio_end_to_end_test.h
>@@ -24,7 +24,7 @@ class AudioEndToEndTest : public test::EndToEndTest {
>   AudioEndToEndTest();
> 
>  protected:
>-  test::FakeAudioDevice* send_audio_device() { return send_audio_device_; }
>+  TestAudioDeviceModule* send_audio_device() { return send_audio_device_; }
>   const AudioSendStream* send_stream() const { return send_stream_; }
>   const AudioReceiveStream* receive_stream() const { return receive_stream_; }
> 
>@@ -34,12 +34,12 @@ class AudioEndToEndTest : public test::EndToEndTest {
>   size_t GetNumAudioStreams() const override;
>   size_t GetNumFlexfecStreams() const override;
> 
>-  std::unique_ptr<test::FakeAudioDevice::Capturer> CreateCapturer() override;
>-  std::unique_ptr<test::FakeAudioDevice::Renderer> CreateRenderer() override;
>+  std::unique_ptr<TestAudioDeviceModule::Capturer> CreateCapturer() override;
>+  std::unique_ptr<TestAudioDeviceModule::Renderer> CreateRenderer() override;
> 
>   void OnFakeAudioDevicesCreated(
>-      test::FakeAudioDevice* send_audio_device,
>-      test::FakeAudioDevice* recv_audio_device) override;
>+      TestAudioDeviceModule* send_audio_device,
>+      TestAudioDeviceModule* recv_audio_device) override;
> 
>   test::PacketTransport* CreateSendTransport(
>       SingleThreadedTaskQueueForTesting* task_queue,
>@@ -57,7 +57,7 @@ class AudioEndToEndTest : public test::EndToEndTest {
>   void PerformTest() override;
> 
>  private:
>-  test::FakeAudioDevice* send_audio_device_ = nullptr;
>+  TestAudioDeviceModule* send_audio_device_ = nullptr;
>   AudioSendStream* send_stream_ = nullptr;
>   AudioReceiveStream* receive_stream_ = nullptr;
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/low_bandwidth_audio_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/low_bandwidth_audio_test.cc
>index cf84e19924b..169f4bf592c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/low_bandwidth_audio_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/low_bandwidth_audio_test.cc
>@@ -13,10 +13,12 @@
> #include "system_wrappers/include/sleep.h"
> #include "test/testsupport/fileutils.h"
> 
>-DEFINE_int(sample_rate_hz, 16000,
>+DEFINE_int(sample_rate_hz,
>+           16000,
>            "Sample rate (Hz) of the produced audio files.");
> 
>-DEFINE_bool(quick, false,
>+DEFINE_bool(quick,
>+            false,
>             "Don't do the full audio recording. "
>             "Used to quickly check that the test runs without crashing.");
> 
>@@ -42,15 +44,15 @@ class AudioQualityTest : public AudioEndToEndTest {
>     const ::testing::TestInfo* const test_info =
>         ::testing::UnitTest::GetInstance()->current_test_info();
>     return webrtc::test::OutputPath() + "LowBandwidth_" + test_info->name() +
>-        "_" + FileSampleRateSuffix() + ".wav";
>+           "_" + FileSampleRateSuffix() + ".wav";
>   }
> 
>-  std::unique_ptr<test::FakeAudioDevice::Capturer> CreateCapturer() override {
>-    return test::FakeAudioDevice::CreateWavFileReader(AudioInputFile());
>+  std::unique_ptr<TestAudioDeviceModule::Capturer> CreateCapturer() override {
>+    return TestAudioDeviceModule::CreateWavFileReader(AudioInputFile());
>   }
> 
>-  std::unique_ptr<test::FakeAudioDevice::Renderer> CreateRenderer() override {
>-    return test::FakeAudioDevice::CreateBoundedWavFileWriter(
>+  std::unique_ptr<TestAudioDeviceModule::Renderer> CreateRenderer() override {
>+    return TestAudioDeviceModule::CreateBoundedWavFileWriter(
>         AudioOutputFile(), FLAG_sample_rate_hz);
>   }
> 
>@@ -69,22 +71,21 @@ class AudioQualityTest : public AudioEndToEndTest {
> 
>     // Output information about the input and output audio files so that further
>     // processing can be done by an external process.
>-    printf("TEST %s %s %s\n", test_info->name(),
>-           AudioInputFile().c_str(), AudioOutputFile().c_str());
>+    printf("TEST %s %s %s\n", test_info->name(), AudioInputFile().c_str(),
>+           AudioOutputFile().c_str());
>   }
> };
> 
> class Mobile2GNetworkTest : public AudioQualityTest {
>-  void ModifyAudioConfigs(AudioSendStream::Config* send_config,
>+  void ModifyAudioConfigs(
>+      AudioSendStream::Config* send_config,
>       std::vector<AudioReceiveStream::Config>* receive_configs) override {
>     send_config->send_codec_spec = AudioSendStream::Config::SendCodecSpec(
>         test::CallTest::kAudioSendPayloadType,
>         {"OPUS",
>          48000,
>          2,
>-         {{"maxaveragebitrate", "6000"},
>-           {"ptime", "60"},
>-           {"stereo", "1"}}});
>+         {{"maxaveragebitrate", "6000"}, {"ptime", "60"}, {"stereo", "1"}}});
>   }
> 
>   FakeNetworkPipe::Config GetNetworkPipeConfig() const override {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/low_bandwidth_audio_test.py b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/low_bandwidth_audio_test.py
>index 9e2597741b9..b81a0476faf 100755
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/low_bandwidth_audio_test.py
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/test/low_bandwidth_audio_test.py
>@@ -55,13 +55,15 @@ def _ParseArgs():
>   parser.add_argument('--android', action='store_true',
>       help='Perform the test on a connected Android device instead.')
>   parser.add_argument('--adb-path', help='Path to adb binary.', default='adb')
>-  parser.add_argument('--chartjson-result-file',
>+  parser.add_argument('--num-retries', default='0',
>+                      help='Number of times to retry the test on Android.')
>+  parser.add_argument('--isolated-script-test-perf-output',
>       help='Where to store perf results in chartjson format.', default=None)
> 
>   # Ignore Chromium-specific flags
>   parser.add_argument('--isolated-script-test-output',
>                       type=str, default=None)
>-  parser.add_argument('--isolated-script-test-perf-output',
>+  parser.add_argument('--test-launcher-summary-output',
>                       type=str, default=None)
>   args = parser.parse_args()
> 
>@@ -221,7 +223,8 @@ def main():
>   out_dir = os.path.join(args.build_dir, '..')
>   if args.android:
>     test_command = [os.path.join(args.build_dir, 'bin',
>-                                 'run_low_bandwidth_audio_test'), '-v']
>+                                 'run_low_bandwidth_audio_test'),
>+                    '-v', '--num-retries', args.num_retries]
>   else:
>     test_command = [os.path.join(args.build_dir, 'low_bandwidth_audio_test')]
> 
>@@ -267,8 +270,8 @@ def main():
>     finally:
>       test_process.terminate()
> 
>-  if args.chartjson_result_file:
>-    with open(args.chartjson_result_file, 'w') as f:
>+  if args.isolated_script_test_perf_output:
>+    with open(args.isolated_script_test_perf_output, 'w') as f:
>       json.dump({"format_version": "1.0", "charts": charts}, f)
> 
>   return test_process.wait()
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/time_interval.h b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/time_interval.h
>index 88b2f7dd4f5..79fe29d9d56 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/time_interval.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/time_interval.h
>@@ -13,7 +13,7 @@
> 
> #include <stdint.h>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> 
> namespace webrtc {
> 
>@@ -57,7 +57,7 @@ class TimeInterval {
> 
>     int64_t first, last;
>   };
>-  rtc::Optional<Interval> interval_;
>+  absl::optional<Interval> interval_;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/time_interval_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/time_interval_unittest.cc
>index 7f8b44ecec5..deff6e363da 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/time_interval_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/time_interval_unittest.cc
>@@ -9,8 +9,8 @@
>  */
> 
> #include "audio/time_interval.h"
>+#include "api/units/time_delta.h"
> #include "rtc_base/fakeclock.h"
>-#include "rtc_base/timedelta.h"
> #include "test/gtest.h"
> 
> namespace webrtc {
>@@ -19,7 +19,7 @@ TEST(TimeIntervalTest, TimeInMs) {
>   rtc::ScopedFakeClock fake_clock;
>   TimeInterval interval;
>   interval.Extend();
>-  fake_clock.AdvanceTime(rtc::TimeDelta::FromMilliseconds(100));
>+  fake_clock.AdvanceTime(TimeDelta::ms(100));
>   interval.Extend();
>   EXPECT_EQ(interval.Length(), 100);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/voice_engine/transport_feedback_packet_loss_tracker.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/transport_feedback_packet_loss_tracker.cc
>similarity index 96%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/voice_engine/transport_feedback_packet_loss_tracker.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/audio/transport_feedback_packet_loss_tracker.cc
>index 774faf5219b..7e0c5c542f8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/voice_engine/transport_feedback_packet_loss_tracker.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/transport_feedback_packet_loss_tracker.cc
>@@ -8,7 +8,7 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "voice_engine/transport_feedback_packet_loss_tracker.h"
>+#include "audio/transport_feedback_packet_loss_tracker.h"
> 
> #include <limits>
> #include <utility>
>@@ -116,12 +116,12 @@ void TransportFeedbackPacketLossTracker::OnPacketFeedbackVector(
>   }
> }
> 
>-rtc::Optional<float>
>-TransportFeedbackPacketLossTracker::GetPacketLossRate() const {
>+absl::optional<float> TransportFeedbackPacketLossTracker::GetPacketLossRate()
>+    const {
>   return plr_state_.GetMetric();
> }
> 
>-rtc::Optional<float>
>+absl::optional<float>
> TransportFeedbackPacketLossTracker::GetRecoverablePacketLossRate() const {
>   return rplr_state_.GetMetric();
> }
>@@ -344,20 +344,20 @@ void TransportFeedbackPacketLossTracker::Validate() const {  // Testing only!
>   RTC_CHECK_EQ(rplr_state_.num_recoverable_losses_, recoverable_losses);
> }
> 
>-rtc::Optional<float>
>-TransportFeedbackPacketLossTracker::PlrState::GetMetric() const {
>+absl::optional<float> TransportFeedbackPacketLossTracker::PlrState::GetMetric()
>+    const {
>   const size_t total = num_lost_packets_ + num_received_packets_;
>   if (total < min_num_acked_packets_) {
>-    return rtc::nullopt;
>+    return absl::nullopt;
>   } else {
>     return static_cast<float>(num_lost_packets_) / total;
>   }
> }
> 
>-rtc::Optional<float>
>-TransportFeedbackPacketLossTracker::RplrState::GetMetric() const {
>+absl::optional<float> TransportFeedbackPacketLossTracker::RplrState::GetMetric()
>+    const {
>   if (num_acked_pairs_ < min_num_acked_pairs_) {
>-    return rtc::nullopt;
>+    return absl::nullopt;
>   } else {
>     return static_cast<float>(num_recoverable_losses_) / num_acked_pairs_;
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/voice_engine/transport_feedback_packet_loss_tracker.h b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/transport_feedback_packet_loss_tracker.h
>similarity index 91%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/voice_engine/transport_feedback_packet_loss_tracker.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/audio/transport_feedback_packet_loss_tracker.h
>index d7420785f7b..7d58d6c4bd3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/voice_engine/transport_feedback_packet_loss_tracker.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/transport_feedback_packet_loss_tracker.h
>@@ -8,13 +8,13 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#ifndef VOICE_ENGINE_TRANSPORT_FEEDBACK_PACKET_LOSS_TRACKER_H_
>-#define VOICE_ENGINE_TRANSPORT_FEEDBACK_PACKET_LOSS_TRACKER_H_
>+#ifndef AUDIO_TRANSPORT_FEEDBACK_PACKET_LOSS_TRACKER_H_
>+#define AUDIO_TRANSPORT_FEEDBACK_PACKET_LOSS_TRACKER_H_
> 
> #include <map>
>+#include <vector>
> 
>-#include "api/optional.h"
>-#include "modules/include/module_common_types.h"
>+#include "absl/types/optional.h"
> 
> namespace webrtc {
> 
>@@ -43,11 +43,11 @@ class TransportFeedbackPacketLossTracker final {
> 
>   // Returns the packet loss rate, if the window has enough packet statuses to
>   // reliably compute it. Otherwise, returns empty.
>-  rtc::Optional<float> GetPacketLossRate() const;
>+  absl::optional<float> GetPacketLossRate() const;
> 
>   // Returns the first-order-FEC recoverable packet loss rate, if the window has
>   // enough status pairs to reliably compute it. Otherwise, returns empty.
>-  rtc::Optional<float> GetRecoverablePacketLossRate() const;
>+  absl::optional<float> GetRecoverablePacketLossRate() const;
> 
>   // Verifies that the internal states are correct. Only used for tests.
>   void Validate() const;
>@@ -108,7 +108,7 @@ class TransportFeedbackPacketLossTracker final {
>       num_received_packets_ = 0;
>       num_lost_packets_ = 0;
>     }
>-    rtc::Optional<float> GetMetric() const;
>+    absl::optional<float> GetMetric() const;
>     const size_t min_num_acked_packets_;
>     size_t num_received_packets_;
>     size_t num_lost_packets_;
>@@ -124,7 +124,7 @@ class TransportFeedbackPacketLossTracker final {
>       num_acked_pairs_ = 0;
>       num_recoverable_losses_ = 0;
>     }
>-    rtc::Optional<float> GetMetric() const;
>+    absl::optional<float> GetMetric() const;
>     // Recoverable packets are those which were lost, but immediately followed
>     // by a properly received packet. If that second packet carried FEC,
>     // the data from the former (lost) packet could be recovered.
>@@ -138,4 +138,4 @@ class TransportFeedbackPacketLossTracker final {
> 
> }  // namespace webrtc
> 
>-#endif  // VOICE_ENGINE_TRANSPORT_FEEDBACK_PACKET_LOSS_TRACKER_H_
>+#endif  // AUDIO_TRANSPORT_FEEDBACK_PACKET_LOSS_TRACKER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/voice_engine/transport_feedback_packet_loss_tracker_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/transport_feedback_packet_loss_tracker_unittest.cc
>similarity index 93%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/voice_engine/transport_feedback_packet_loss_tracker_unittest.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/audio/transport_feedback_packet_loss_tracker_unittest.cc
>index 55626bedd0f..2f9bf68444f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/voice_engine/transport_feedback_packet_loss_tracker_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/transport_feedback_packet_loss_tracker_unittest.cc
>@@ -13,12 +13,12 @@
> #include <numeric>
> #include <vector>
> 
>+#include "audio/transport_feedback_packet_loss_tracker.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> #include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
> #include "rtc_base/checks.h"
> #include "test/gmock.h"
> #include "test/gtest.h"
>-#include "voice_engine/transport_feedback_packet_loss_tracker.h"
> 
> namespace webrtc {
> 
>@@ -94,18 +94,18 @@ class TransportFeedbackPacketLossTrackerTest
>   // value is as expected.
>   void ValidatePacketLossStatistics(
>       const TransportFeedbackPacketLossTracker& tracker,
>-      rtc::Optional<float> expected_plr,
>-      rtc::Optional<float> expected_rplr) {
>-    // TODO(eladalon): Comparing the rtc::Optional<float> directly would have
>+      absl::optional<float> expected_plr,
>+      absl::optional<float> expected_rplr) {
>+    // TODO(eladalon): Comparing the absl::optional<float> directly would have
>     // given concise code, but less readable error messages. If we modify
>-    // the way rtc::Optional is printed, we can get rid of this.
>-    rtc::Optional<float> plr = tracker.GetPacketLossRate();
>+    // the way absl::optional is printed, we can get rid of this.
>+    absl::optional<float> plr = tracker.GetPacketLossRate();
>     EXPECT_EQ(static_cast<bool>(expected_plr), static_cast<bool>(plr));
>     if (expected_plr && plr) {
>       EXPECT_EQ(*expected_plr, *plr);
>     }
> 
>-    rtc::Optional<float> rplr = tracker.GetRecoverablePacketLossRate();
>+    absl::optional<float> rplr = tracker.GetRecoverablePacketLossRate();
>     EXPECT_EQ(static_cast<bool>(expected_rplr), static_cast<bool>(rplr));
>     if (expected_rplr && rplr) {
>       EXPECT_EQ(*expected_rplr, *rplr);
>@@ -127,7 +127,7 @@ TEST_P(TransportFeedbackPacketLossTrackerTest, EmptyWindow) {
>   TransportFeedbackPacketLossTracker tracker(kDefaultMaxWindowSizeMs, 5, 5);
> 
>   // PLR and RPLR reported as unknown before reception of first feedback.
>-  ValidatePacketLossStatistics(tracker, rtc::nullopt, rtc::nullopt);
>+  ValidatePacketLossStatistics(tracker, absl::nullopt, absl::nullopt);
> }
> 
> // A feedback received for an empty window has no effect.
>@@ -136,7 +136,7 @@ TEST_P(TransportFeedbackPacketLossTrackerTest, EmptyWindowFeedback) {
> 
>   // Feedback doesn't correspond to any packets - ignored.
>   AddTransportFeedbackAndValidate(&tracker, base_, {true, false, true});
>-  ValidatePacketLossStatistics(tracker, rtc::nullopt, rtc::nullopt);
>+  ValidatePacketLossStatistics(tracker, absl::nullopt, absl::nullopt);
> 
>   // After the packets are transmitted, acking them would have an effect.
>   SendPackets(&tracker, base_, 3, kDefaultSendIntervalMs);
>@@ -153,7 +153,7 @@ TEST_P(TransportFeedbackPacketLossTrackerTest, PartiallyFilledWindow) {
>   // Expected window contents: [] -> [1001].
>   SendPackets(&tracker, base_, 3, kDefaultSendIntervalMs);
>   AddTransportFeedbackAndValidate(&tracker, base_, {true, false, false, true});
>-  ValidatePacketLossStatistics(tracker, rtc::nullopt, rtc::nullopt);
>+  ValidatePacketLossStatistics(tracker, absl::nullopt, absl::nullopt);
> }
> 
> // Sanity check on minimum filled window - PLR known, RPLR unknown.
>@@ -166,7 +166,7 @@ TEST_P(TransportFeedbackPacketLossTrackerTest, PlrMinimumFilledWindow) {
>   SendPackets(&tracker, base_, 5, kDefaultSendIntervalMs);
>   AddTransportFeedbackAndValidate(&tracker, base_,
>                                   {true, false, false, true, true});
>-  ValidatePacketLossStatistics(tracker, 2.0f / 5.0f, rtc::nullopt);
>+  ValidatePacketLossStatistics(tracker, 2.0f / 5.0f, absl::nullopt);
> }
> 
> // Sanity check on minimum filled window - PLR unknown, RPLR known.
>@@ -179,7 +179,7 @@ TEST_P(TransportFeedbackPacketLossTrackerTest, RplrMinimumFilledWindow) {
>   SendPackets(&tracker, base_, 5, kDefaultSendIntervalMs);
>   AddTransportFeedbackAndValidate(&tracker, base_,
>                                   {true, false, false, true, true});
>-  ValidatePacketLossStatistics(tracker, rtc::nullopt, 1.0f / 4.0f);
>+  ValidatePacketLossStatistics(tracker, absl::nullopt, 1.0f / 4.0f);
> }
> 
> // If packets are sent close enough together that the clock reading for both
>@@ -203,7 +203,7 @@ TEST_P(TransportFeedbackPacketLossTrackerTest, ExtendWindow) {
>   // Expected window contents: [] -> [10011].
>   AddTransportFeedbackAndValidate(&tracker, base_,
>                                   {true, false, false, true, true});
>-  ValidatePacketLossStatistics(tracker, 2.0f / 5.0f, rtc::nullopt);
>+  ValidatePacketLossStatistics(tracker, 2.0f / 5.0f, absl::nullopt);
> 
>   // Expected window contents: [10011] -> [1001110101].
>   AddTransportFeedbackAndValidate(&tracker, base_ + 5,
>@@ -412,34 +412,30 @@ TEST_P(TransportFeedbackPacketLossTrackerTest, InsertionCompletesTwoPairs) {
> TEST_P(TransportFeedbackPacketLossTrackerTest, SanityGapsInSequenceNumbers) {
>   TransportFeedbackPacketLossTracker tracker(50 * kDefaultSendIntervalMs, 5, 1);
> 
>-  SendPackets(&tracker,
>-              {static_cast<uint16_t>(base_),
>-               static_cast<uint16_t>(base_ + 2),
>-               static_cast<uint16_t>(base_ + 4),
>-               static_cast<uint16_t>(base_ + 6),
>-               static_cast<uint16_t>(base_ + 8)},
>-              kDefaultSendIntervalMs);
>+  SendPackets(
>+      &tracker,
>+      {static_cast<uint16_t>(base_), static_cast<uint16_t>(base_ + 2),
>+       static_cast<uint16_t>(base_ + 4), static_cast<uint16_t>(base_ + 6),
>+       static_cast<uint16_t>(base_ + 8)},
>+      kDefaultSendIntervalMs);
> 
>   // Gaps in sequence numbers not considered as gaps in window, because  only
>   // those sequence numbers which were associated with the stream count.
>   // Expected window contents: [] -> [11011].
>   AddTransportFeedbackAndValidate(
>       // Note: Left packets belong to this stream, right ones ignored.
>-      &tracker, base_, {true, false,
>-                        true, false,
>-                        false, false,
>-                        true, false,
>-                        true, true});
>+      &tracker, base_,
>+      {true, false, true, false, false, false, true, false, true, true});
>   ValidatePacketLossStatistics(tracker, 1.0f / 5.0f, 1.0f / 4.0f);
> 
>   // Create gap by sending [base + 10] but not acking it.
>   // Note: Acks for [base + 11] and [base + 13] ignored (other stream).
>   // Expected window contents: [11011] -> [11011-GAP-01].
>-  SendPackets(&tracker,
>-              {static_cast<uint16_t>(base_ + 10),
>-               static_cast<uint16_t>(base_ + 12),
>-               static_cast<uint16_t>(base_ + 14)},
>-              kDefaultSendIntervalMs);
>+  SendPackets(
>+      &tracker,
>+      {static_cast<uint16_t>(base_ + 10), static_cast<uint16_t>(base_ + 12),
>+       static_cast<uint16_t>(base_ + 14)},
>+      kDefaultSendIntervalMs);
>   AddTransportFeedbackAndValidate(&tracker, base_ + 11,
>                                   {false, false, false, true, true});
>   ValidatePacketLossStatistics(tracker, 2.0f / 7.0f, 2.0f / 5.0f);
>@@ -520,7 +516,7 @@ TEST_P(TransportFeedbackPacketLossTrackerTest, RepeatedSeqNumResetsWindow) {
>   // A reset occurs.
>   SendPackets(&tracker, {static_cast<uint16_t>(base_ + 2)},
>               kDefaultSendIntervalMs);
>-  ValidatePacketLossStatistics(tracker, rtc::nullopt, rtc::nullopt);
>+  ValidatePacketLossStatistics(tracker, absl::nullopt, absl::nullopt);
> }
> 
> // The window is reset by the sending of a packet which is 0x8000 or more
>@@ -539,7 +535,7 @@ TEST_P(TransportFeedbackPacketLossTrackerTest,
>   // A reset occurs.
>   SendPackets(&tracker, {static_cast<uint16_t>(base_ + 5 + 0x8000)},
>               kDefaultSendIntervalMs);
>-  ValidatePacketLossStatistics(tracker, rtc::nullopt, rtc::nullopt);
>+  ValidatePacketLossStatistics(tracker, absl::nullopt, absl::nullopt);
> }
> 
> #if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/utility/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/utility/BUILD.gn
>index aa8445c90b3..76c09a51a38 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/utility/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/utility/BUILD.gn
>@@ -22,9 +22,7 @@ rtc_static_library("audio_frame_operations") {
> 
>   deps = [
>     "../..:webrtc_common",
>-    "../../:typedefs",
>-    "../../modules:module_api",
>-    "../../modules/audio_coding:audio_format_conversion",
>+    "../../api/audio:audio_frame_api",
>     "../../rtc_base:checks",
>     "../../rtc_base:rtc_base_approved",
>   ]
>@@ -38,7 +36,6 @@ if (rtc_include_tests) {
>     ]
>     deps = [
>       ":audio_frame_operations",
>-      "../../modules:module_api",
>       "../../rtc_base:checks",
>       "../../rtc_base:rtc_base_approved",
>       "../../test:test_support",
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/utility/audio_frame_operations.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/utility/audio_frame_operations.cc
>index a7c77821f65..fb1f3b07711 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/utility/audio_frame_operations.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/utility/audio_frame_operations.cc
>@@ -10,9 +10,9 @@
> 
> #include "audio/utility/audio_frame_operations.h"
> 
>+#include <string.h>
> #include <algorithm>
> 
>-#include "modules/include/module_common_types.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/numerics/safe_conversions.h"
> 
>@@ -159,7 +159,8 @@ void AudioFrameOperations::QuadToMono(const int16_t* src_audio,
>   for (size_t i = 0; i < samples_per_channel; i++) {
>     dst_audio[i] =
>         (static_cast<int32_t>(src_audio[4 * i]) + src_audio[4 * i + 1] +
>-         src_audio[4 * i + 2] + src_audio[4 * i + 3]) >> 2;
>+         src_audio[4 * i + 2] + src_audio[4 * i + 3]) >>
>+        2;
>   }
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/utility/audio_frame_operations.h b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/utility/audio_frame_operations.h
>index cd55f19fc12..599352356f6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/utility/audio_frame_operations.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/utility/audio_frame_operations.h
>@@ -13,12 +13,10 @@
> 
> #include <stddef.h>
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include "api/audio/audio_frame.h"
> 
> namespace webrtc {
> 
>-class AudioFrame;
>-
> // TODO(andrew): consolidate this with utility.h and audio_frame_manipulator.h.
> // Change reference parameters to pointers. Consider using a namespace rather
> // than a class.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/utility/audio_frame_operations_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/utility/audio_frame_operations_unittest.cc
>index 6d23731a05d..76f1dcdb52f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/audio/utility/audio_frame_operations_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/audio/utility/audio_frame_operations_unittest.cc
>@@ -9,7 +9,6 @@
>  */
> 
> #include "audio/utility/audio_frame_operations.h"
>-#include "modules/include/module_common_types.h"
> #include "rtc_base/checks.h"
> #include "test/gtest.h"
> 
>@@ -51,27 +50,29 @@ void SetFrameData(int16_t left, int16_t right, AudioFrame* frame) {
> 
> void SetFrameData(int16_t data, AudioFrame* frame) {
>   int16_t* frame_data = frame->mutable_data();
>-  for (size_t i = 0;
>-       i < frame->samples_per_channel_ * frame->num_channels_; i++) {
>+  for (size_t i = 0; i < frame->samples_per_channel_ * frame->num_channels_;
>+       i++) {
>     frame_data[i] = data;
>   }
> }
> 
> void VerifyFramesAreEqual(const AudioFrame& frame1, const AudioFrame& frame2) {
>   EXPECT_EQ(frame1.num_channels_, frame2.num_channels_);
>-  EXPECT_EQ(frame1.samples_per_channel_,
>-            frame2.samples_per_channel_);
>+  EXPECT_EQ(frame1.samples_per_channel_, frame2.samples_per_channel_);
>   const int16_t* frame1_data = frame1.data();
>   const int16_t* frame2_data = frame2.data();
>   for (size_t i = 0; i < frame1.samples_per_channel_ * frame1.num_channels_;
>-      i++) {
>+       i++) {
>     EXPECT_EQ(frame1_data[i], frame2_data[i]);
>   }
>   EXPECT_EQ(frame1.muted(), frame2.muted());
> }
> 
>-void InitFrame(AudioFrame* frame, size_t channels, size_t samples_per_channel,
>-               int16_t left_data, int16_t right_data) {
>+void InitFrame(AudioFrame* frame,
>+               size_t channels,
>+               size_t samples_per_channel,
>+               int16_t left_data,
>+               int16_t right_data) {
>   RTC_DCHECK(frame);
>   RTC_DCHECK_GE(2, channels);
>   RTC_DCHECK_GE(AudioFrame::kMaxDataSizeSamples,
>@@ -91,7 +92,9 @@ int16_t GetChannelData(const AudioFrame& frame, size_t channel, size_t index) {
>   return frame.data()[index * frame.num_channels_ + channel];
> }
> 
>-void VerifyFrameDataBounds(const AudioFrame& frame, size_t channel, int16_t max,
>+void VerifyFrameDataBounds(const AudioFrame& frame,
>+                           size_t channel,
>+                           int16_t max,
>                            int16_t min) {
>   for (size_t i = 0; i < frame.samples_per_channel_; ++i) {
>     int16_t s = GetChannelData(frame, channel, i);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/call/BUILD.gn
>index f61d57c2ee1..838728ad78b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/BUILD.gn
>@@ -10,12 +10,18 @@ import("../webrtc.gni")
> 
> rtc_source_set("call_interfaces") {
>   sources = [
>+    "audio_receive_stream.cc",
>     "audio_receive_stream.h",
>     "audio_send_stream.h",
>+    "audio_state.cc",
>     "audio_state.h",
>     "call.h",
>-    "callfactoryinterface.h",
>+    "call_config.cc",
>+    "call_config.h",
>+    "flexfec_receive_stream.cc",
>     "flexfec_receive_stream.h",
>+    "packet_receiver.cc",
>+    "packet_receiver.h",
>     "syncable.cc",
>     "syncable.h",
>   ]
>@@ -26,15 +32,19 @@ rtc_source_set("call_interfaces") {
>     ":rtp_interfaces",
>     ":video_stream_api",
>     "..:webrtc_common",
>-    "../:typedefs",
>-    "../api:audio_mixer_api",
>+    "../api:fec_controller_api",
>     "../api:libjingle_peerconnection_api",
>-    "../api:optional",
>     "../api:transport_api",
>+    "../api/audio:audio_mixer_api",
>     "../api/audio_codecs:audio_codecs_api",
>+    "../api/transport:network_control",
>+    "../modules/audio_device:audio_device",
>+    "../modules/audio_processing:audio_processing",
>     "../modules/audio_processing:audio_processing_statistics",
>+    "../rtc_base:audio_format_to_string",
>     "../rtc_base:rtc_base",
>     "../rtc_base:rtc_base_approved",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
> 
>@@ -51,7 +61,12 @@ rtc_source_set("rtp_interfaces") {
>   ]
>   deps = [
>     "../api:array_view",
>+    "../api:libjingle_peerconnection_api",
>+    "../api/transport:bitrate_settings",
>+    "../logging:rtc_event_log_api",
>+    "../modules/rtp_rtcp:rtp_rtcp_format",
>     "../rtc_base:rtc_base_approved",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
> 
>@@ -75,24 +90,61 @@ rtc_source_set("rtp_receiver") {
>     "..:webrtc_common",
>     "../api:array_view",
>     "../api:libjingle_peerconnection_api",
>-    "../api:optional",
>     "../modules/rtp_rtcp",
>     "../modules/rtp_rtcp:rtp_rtcp_format",
>     "../rtc_base:checks",
>     "../rtc_base:rtc_base_approved",
>+    "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
> 
> rtc_source_set("rtp_sender") {
>   sources = [
>+    "rtp_payload_params.cc",
>+    "rtp_payload_params.h",
>     "rtp_transport_controller_send.cc",
>     "rtp_transport_controller_send.h",
>+    "rtp_video_sender.cc",
>+    "rtp_video_sender.h",
>+    "rtp_video_sender_interface.h",
>   ]
>   deps = [
>+    ":bitrate_configurator",
>     ":rtp_interfaces",
>     "..:webrtc_common",
>+    "../api:transport_api",
>+    "../api/transport:network_control",
>+    "../api/video_codecs:video_codecs_api",
>+    "../logging:rtc_event_log_api",
>     "../modules/congestion_controller",
>+    "../modules/congestion_controller/rtp:congestion_controller",
>     "../modules/pacing",
>+    "../modules/rtp_rtcp:rtp_rtcp",
>+    "../modules/rtp_rtcp:rtp_rtcp_format",
>+    "../modules/rtp_rtcp:rtp_video_header",
>+    "../modules/utility",
>+    "../modules/video_coding:video_codec_interface",
>+    "../rtc_base:checks",
>+    "../rtc_base:rate_limiter",
>+    "../rtc_base:rtc_base",
>+    "../rtc_base:rtc_base_approved",
>+    "../rtc_base:rtc_task_queue",
>+    "../system_wrappers:field_trial_api",
>+    "//third_party/abseil-cpp/absl/memory",
>+  ]
>+}
>+
>+rtc_source_set("bitrate_configurator") {
>+  sources = [
>+    "rtp_bitrate_configurator.cc",
>+    "rtp_bitrate_configurator.h",
>+  ]
>+  deps = [
>+    ":rtp_interfaces",
>+    "../api:libjingle_peerconnection_api",
>+    "../api/transport:bitrate_settings",
>+    "../rtc_base:checks",
>     "../rtc_base:rtc_base_approved",
>   ]
> }
>@@ -108,12 +160,9 @@ rtc_source_set("bitrate_allocator") {
>     "../rtc_base:rtc_base_approved",
>     "../rtc_base:sequenced_task_checker",
>     "../system_wrappers",
>+    "../system_wrappers:field_trial_api",
>     "../system_wrappers:metrics_api",
>   ]
>-  if (!build_with_chromium && is_clang) {
>-    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>-    suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-  }
> }
> 
> rtc_static_library("call") {
>@@ -121,8 +170,12 @@ rtc_static_library("call") {
>     "call.cc",
>     "callfactory.cc",
>     "callfactory.h",
>+    "degraded_call.cc",
>+    "degraded_call.h",
>     "flexfec_receive_stream_impl.cc",
>     "flexfec_receive_stream_impl.h",
>+    "receive_time_calculator.cc",
>+    "receive_time_calculator.h",
>   ]
> 
>   if (!build_with_chromium && is_clang) {
>@@ -133,36 +186,46 @@ rtc_static_library("call") {
>   deps = [
>     ":bitrate_allocator",
>     ":call_interfaces",
>+    ":fake_network",
>     ":rtp_interfaces",
>     ":rtp_receiver",
>     ":rtp_sender",
>     ":video_stream_api",
>     "..:webrtc_common",
>-    "../api:optional",
>+    "../api:callfactory_api",
>     "../api:transport_api",
>+    "../api/transport:network_control",
>     "../audio",
>+    "../logging:rtc_event_audio",
>     "../logging:rtc_event_log_api",
>-    "../logging:rtc_event_log_impl",
>+    "../logging:rtc_event_rtp_rtcp",
>+    "../logging:rtc_event_video",
>+    "../logging:rtc_stream_config",
>     "../modules/bitrate_controller",
>     "../modules/congestion_controller",
>     "../modules/pacing",
>     "../modules/rtp_rtcp",
>     "../modules/rtp_rtcp:rtp_rtcp_format",
>     "../modules/utility",
>+    "../modules/video_coding:video_coding",
>     "../rtc_base:checks",
>+    "../rtc_base:rate_limiter",
>     "../rtc_base:rtc_base_approved",
>     "../rtc_base:rtc_task_queue",
>+    "../rtc_base:safe_minmax",
>     "../rtc_base:sequenced_task_checker",
>+    "../rtc_base/synchronization:rw_lock_wrapper",
>     "../system_wrappers",
>+    "../system_wrappers:field_trial_api",
>     "../system_wrappers:metrics_api",
>     "../video",
>+    "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
> 
> rtc_source_set("video_stream_api") {
>   sources = [
>-    "video_config.cc",
>-    "video_config.h",
>     "video_receive_stream.cc",
>     "video_receive_stream.h",
>     "video_send_stream.cc",
>@@ -170,16 +233,49 @@ rtc_source_set("video_stream_api") {
>   ]
>   deps = [
>     ":rtp_interfaces",
>-    "../:typedefs",
>     "../:webrtc_common",
>     "../api:libjingle_peerconnection_api",
>-    "../api:optional",
>     "../api:transport_api",
>-    "../api:video_frame_api",
>+    "../api/video:video_frame",
>+    "../api/video:video_stream_encoder",
>+    "../api/video_codecs:video_codecs_api",
>     "../common_video:common_video",
>     "../modules/rtp_rtcp:rtp_rtcp_format",
>     "../rtc_base:checks",
>     "../rtc_base:rtc_base_approved",
>+    "//third_party/abseil-cpp/absl/types:optional",
>+  ]
>+}
>+
>+rtc_source_set("simulated_network") {
>+  sources = [
>+    "simulated_network.cc",
>+    "simulated_network.h",
>+  ]
>+  deps = [
>+    "../api:simulated_network_api",
>+    "../rtc_base:rtc_base_approved",
>+    "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/types:optional",
>+  ]
>+}
>+
>+rtc_source_set("fake_network") {
>+  sources = [
>+    "fake_network_pipe.cc",
>+    "fake_network_pipe.h",
>+  ]
>+  deps = [
>+    ":call_interfaces",
>+    ":simulated_network",
>+    "..:webrtc_common",
>+    "../api:simulated_network_api",
>+    "../api:transport_api",
>+    "../modules:module_api",
>+    "../rtc_base:rtc_base_approved",
>+    "../rtc_base:sequenced_task_checker",
>+    "../system_wrappers",
>+    "//third_party/abseil-cpp/absl/memory",
>   ]
> }
> 
>@@ -192,13 +288,18 @@ if (rtc_include_tests) {
>       "bitrate_estimator_tests.cc",
>       "call_unittest.cc",
>       "flexfec_receive_stream_unittest.cc",
>+      "receive_time_calculator_unittest.cc",
>       "rtcp_demuxer_unittest.cc",
>+      "rtp_bitrate_configurator_unittest.cc",
>       "rtp_demuxer_unittest.cc",
>+      "rtp_payload_params_unittest.cc",
>       "rtp_rtcp_demuxer_helper_unittest.cc",
>+      "rtp_video_sender_unittest.cc",
>       "rtx_receive_stream_unittest.cc",
>     ]
>     deps = [
>       ":bitrate_allocator",
>+      ":bitrate_configurator",
>       ":call",
>       ":call_interfaces",
>       ":mock_rtp_interfaces",
>@@ -212,29 +313,33 @@ if (rtc_include_tests) {
>       "../api/audio_codecs:builtin_audio_decoder_factory",
>       "../audio:audio",
>       "../logging:rtc_event_log_api",
>+      "../logging:rtc_event_log_impl_base",
>       "../modules/audio_device:mock_audio_device",
>       "../modules/audio_mixer",
>       "../modules/audio_mixer:audio_mixer_impl",
>       "../modules/audio_processing:mocks",
>       "../modules/bitrate_controller",
>       "../modules/congestion_controller",
>-      "../modules/congestion_controller:mock_congestion_controller",
>       "../modules/pacing",
>       "../modules/pacing:mock_paced_sender",
>       "../modules/rtp_rtcp",
>       "../modules/rtp_rtcp:mock_rtp_rtcp",
>       "../modules/rtp_rtcp:rtp_rtcp_format",
>       "../modules/utility:mock_process_thread",
>+      "../modules/video_coding:video_codec_interface",
>       "../rtc_base:checks",
>+      "../rtc_base:rate_limiter",
>       "../rtc_base:rtc_base_approved",
>       "../system_wrappers",
>       "../test:audio_codec_mocks",
>       "../test:direct_transport",
>+      "../test:field_trial",
>       "../test:test_common",
>       "../test:test_support",
>       "../test:video_test_common",
>-      "//testing/gmock",
>+      "../video:video",
>       "//testing/gtest",
>+      "//third_party/abseil-cpp/absl/memory",
>     ]
>     if (!build_with_chromium && is_clang) {
>       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>@@ -255,8 +360,12 @@ if (rtc_include_tests) {
>       ":video_stream_api",
>       "..:webrtc_common",
>       "../api/audio_codecs:builtin_audio_encoder_factory",
>+      "../api/video:video_bitrate_allocation",
>+      "../api/video_codecs:video_codecs_api",
>       "../logging:rtc_event_log_api",
>       "../modules/audio_coding",
>+      "../modules/audio_device",
>+      "../modules/audio_device:audio_device_impl",
>       "../modules/audio_mixer:audio_mixer_impl",
>       "../modules/rtp_rtcp",
>       "../rtc_base:checks",
>@@ -265,15 +374,15 @@ if (rtc_include_tests) {
>       "../system_wrappers:metrics_default",
>       "../system_wrappers:runtime_enabled_features_default",
>       "../test:direct_transport",
>-      "../test:fake_audio_device",
>       "../test:field_trial",
>+      "../test:fileutils",
>       "../test:perf_test",
>       "../test:test_common",
>       "../test:test_support",
>       "../test:video_test_common",
>       "../video",
>-      "../voice_engine",
>       "//testing/gtest",
>+      "//third_party/abseil-cpp/absl/memory",
>     ]
>     if (!build_with_chromium && is_clang) {
>       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>@@ -286,19 +395,30 @@ if (rtc_include_tests) {
>     testonly = true
> 
>     sources = [
>-      "fake_rtp_transport_controller_send.h",
>       "test/mock_rtp_packet_sink_interface.h",
>+      "test/mock_rtp_transport_controller_send.h",
>     ]
>     deps = [
>       ":rtp_interfaces",
>-      "..:webrtc_common",
>-      "../modules/congestion_controller:congestion_controller",
>-      "../modules/pacing:pacing",
>+      "../api:libjingle_peerconnection_api",
>+      "../modules/congestion_controller",
>+      "../modules/pacing",
>+      "../rtc_base:rate_limiter",
>+      "../rtc_base:rtc_base",
>       "../test:test_support",
>-      "//testing/gmock",
>     ]
>   }
>+  rtc_source_set("mock_bitrate_allocator") {
>+    testonly = true
> 
>+    sources = [
>+      "test/mock_bitrate_allocator.h",
>+    ]
>+    deps = [
>+      ":bitrate_allocator",
>+      "//test:test_support",
>+    ]
>+  }
>   rtc_source_set("mock_call_interfaces") {
>     testonly = true
> 
>@@ -310,4 +430,21 @@ if (rtc_include_tests) {
>       "//test:test_support",
>     ]
>   }
>+
>+  rtc_test("fake_network_unittests") {
>+    deps = [
>+      ":call_interfaces",
>+      ":fake_network",
>+      ":simulated_network",
>+      "../modules/rtp_rtcp",
>+      "../rtc_base:rtc_base_approved",
>+      "../system_wrappers",
>+      "../test:test_common",
>+      "../test:test_main",
>+      "//testing/gtest",
>+    ]
>+    sources = [
>+      "test/fake_network_pipe_unittest.cc",
>+    ]
>+  }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/DEPS b/Source/ThirdParty/libwebrtc/Source/webrtc/call/DEPS
>index 7622e241168..f823a7b9c37 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/DEPS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/DEPS
>@@ -7,11 +7,11 @@ include_rules = [
>   "+modules/audio_processing",
>   "+modules/bitrate_controller",
>   "+modules/congestion_controller",
>+  "+modules/video_coding",
>   "+modules/pacing",
>   "+modules/rtp_rtcp",
>   "+modules/utility",
>   "+system_wrappers",
>-  "+voice_engine",
>   "+video",
> ]
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/call/OWNERS
>index fc0487dc2ad..af7f98ae3f8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/OWNERS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/OWNERS
>@@ -1,6 +1,8 @@
> mflodman@webrtc.org
>+nisse@webrtc.org
> solenberg@webrtc.org
> stefan@webrtc.org
>+srte@webrtc.org
> 
> # These are for the common case of adding or renaming files. If you're doing
> # structural changes, please get a review from a reviewer in this file.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/audio_receive_stream.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/audio_receive_stream.cc
>new file mode 100644
>index 00000000000..c3c2ac77d06
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/audio_receive_stream.cc
>@@ -0,0 +1,24 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "call/audio_receive_stream.h"
>+
>+namespace webrtc {
>+
>+AudioReceiveStream::Stats::Stats() = default;
>+AudioReceiveStream::Stats::~Stats() = default;
>+
>+AudioReceiveStream::Config::Config() = default;
>+AudioReceiveStream::Config::~Config() = default;
>+
>+AudioReceiveStream::Config::Rtp::Rtp() = default;
>+AudioReceiveStream::Config::Rtp::~Rtp() = default;
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/audio_receive_stream.h b/Source/ThirdParty/libwebrtc/Source/webrtc/call/audio_receive_stream.h
>index f8d787cb504..9595a293dac 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/audio_receive_stream.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/audio_receive_stream.h
>@@ -16,15 +16,14 @@
> #include <string>
> #include <vector>
> 
>+#include "absl/types/optional.h"
> #include "api/audio_codecs/audio_decoder_factory.h"
> #include "api/call/transport.h"
>-#include "api/optional.h"
> #include "api/rtpparameters.h"
> #include "api/rtpreceiverinterface.h"
> #include "call/rtp_config.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "rtc_base/scoped_ref_ptr.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> class AudioSinkInterface;
>@@ -32,13 +31,15 @@ class AudioSinkInterface;
> class AudioReceiveStream {
>  public:
>   struct Stats {
>+    Stats();
>+    ~Stats();
>     uint32_t remote_ssrc = 0;
>     int64_t bytes_rcvd = 0;
>     uint32_t packets_rcvd = 0;
>     uint32_t packets_lost = 0;
>     float fraction_lost = 0.0f;
>     std::string codec_name;
>-    rtc::Optional<int> codec_payload_type;
>+    absl::optional<int> codec_payload_type;
>     uint32_t ext_seqnum = 0;
>     uint32_t jitter_ms = 0;
>     uint32_t jitter_buffer_ms = 0;
>@@ -71,10 +72,16 @@ class AudioReceiveStream {
>   };
> 
>   struct Config {
>+    Config();
>+    ~Config();
>+
>     std::string ToString() const;
> 
>     // Receive-stream specific RTP settings.
>     struct Rtp {
>+      Rtp();
>+      ~Rtp();
>+
>       std::string ToString() const;
> 
>       // Synchronization source (stream identifier) to be received.
>@@ -98,9 +105,6 @@ class AudioReceiveStream {
> 
>     Transport* rtcp_send_transport = nullptr;
> 
>-    // TODO(solenberg): Remove once clients don't use it anymore.
>-    int voe_channel_id = -1;
>-
>     // NetEq settings.
>     size_t jitter_buffer_max_packets = 50;
>     bool jitter_buffer_fast_accelerate = false;
>@@ -114,6 +118,8 @@ class AudioReceiveStream {
>     std::map<int, SdpAudioFormat> decoder_map;
> 
>     rtc::scoped_refptr<AudioDecoderFactory> decoder_factory;
>+
>+    absl::optional<AudioCodecPairId> codec_pair_id;
>   };
> 
>   // Reconfigure the stream according to the Configuration.
>@@ -127,8 +133,6 @@ class AudioReceiveStream {
>   virtual void Stop() = 0;
> 
>   virtual Stats GetStats() const = 0;
>-  // TODO(solenberg): Remove, once AudioMonitor is gone.
>-  virtual int GetOutputLevel() const = 0;
> 
>   // Sets an audio sink that receives unmixed audio from the receive stream.
>   // Ownership of the sink is managed by the caller.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/audio_send_stream.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/audio_send_stream.cc
>index 274a6ebdb4d..e2cf20cf844 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/audio_send_stream.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/audio_send_stream.cc
>@@ -9,8 +9,9 @@
>  */
> 
> #include "call/audio_send_stream.h"
>-
>-#include <string>
>+#include "rtc_base/stringencode.h"
>+#include "rtc_base/strings/audio_format_to_string.h"
>+#include "rtc_base/strings/string_builder.h"
> 
> namespace webrtc {
> 
>@@ -23,7 +24,8 @@ AudioSendStream::Config::Config(Transport* send_transport)
> AudioSendStream::Config::~Config() = default;
> 
> std::string AudioSendStream::Config::ToString() const {
>-  std::stringstream ss;
>+  char buf[1024];
>+  rtc::SimpleStringBuilder ss(buf);
>   ss << "{rtp: " << rtp.ToString();
>   ss << ", send_transport: " << (send_transport ? "(Transport)" : "null");
>   ss << ", min_bitrate_bps: " << min_bitrate_bps;
>@@ -39,7 +41,8 @@ AudioSendStream::Config::Rtp::Rtp() = default;
> AudioSendStream::Config::Rtp::~Rtp() = default;
> 
> std::string AudioSendStream::Config::Rtp::ToString() const {
>-  std::stringstream ss;
>+  char buf[1024];
>+  rtc::SimpleStringBuilder ss(buf);
>   ss << "{ssrc: " << ssrc;
>   ss << ", extensions: [";
>   for (size_t i = 0; i < extensions.size(); ++i) {
>@@ -62,13 +65,14 @@ AudioSendStream::Config::SendCodecSpec::SendCodecSpec(
> AudioSendStream::Config::SendCodecSpec::~SendCodecSpec() = default;
> 
> std::string AudioSendStream::Config::SendCodecSpec::ToString() const {
>-  std::stringstream ss;
>+  char buf[1024];
>+  rtc::SimpleStringBuilder ss(buf);
>   ss << "{nack_enabled: " << (nack_enabled ? "true" : "false");
>   ss << ", transport_cc_enabled: " << (transport_cc_enabled ? "true" : "false");
>   ss << ", cng_payload_type: "
>-     << (cng_payload_type ? std::to_string(*cng_payload_type) : "<unset>");
>+     << (cng_payload_type ? rtc::ToString(*cng_payload_type) : "<unset>");
>   ss << ", payload_type: " << payload_type;
>-  ss << ", format: " << format;
>+  ss << ", format: " << rtc::ToString(format);
>   ss << '}';
>   return ss.str();
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/audio_send_stream.h b/Source/ThirdParty/libwebrtc/Source/webrtc/call/audio_send_stream.h
>index 03f32b79e62..60909ae128b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/audio_send_stream.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/audio_send_stream.h
>@@ -15,16 +15,16 @@
> #include <string>
> #include <vector>
> 
>+#include "absl/types/optional.h"
>+#include "api/audio_codecs/audio_codec_pair_id.h"
> #include "api/audio_codecs/audio_encoder.h"
> #include "api/audio_codecs/audio_encoder_factory.h"
> #include "api/audio_codecs/audio_format.h"
> #include "api/call/transport.h"
>-#include "api/optional.h"
> #include "api/rtpparameters.h"
> #include "call/rtp_config.h"
> #include "modules/audio_processing/include/audio_processing_statistics.h"
> #include "rtc_base/scoped_ref_ptr.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -43,7 +43,7 @@ class AudioSendStream {
>     int32_t packets_lost = -1;
>     float fraction_lost = -1.0f;
>     std::string codec_name;
>-    rtc::Optional<int> codec_payload_type;
>+    absl::optional<int> codec_payload_type;
>     int32_t ext_seqnum = -1;
>     int32_t jitter_ms = -1;
>     int64_t rtt_ms = -1;
>@@ -73,6 +73,10 @@ class AudioSendStream {
>       // Sender SSRC.
>       uint32_t ssrc = 0;
> 
>+      // The value to send in the MID RTP header extension if the extension is
>+      // included in the list of extensions.
>+      std::string mid;
>+
>       // RTP header extensions used for the sent stream.
>       std::vector<RtpExtension> extensions;
> 
>@@ -87,9 +91,6 @@ class AudioSendStream {
>     // the entire life of the AudioSendStream and is owned by the API client.
>     Transport* send_transport = nullptr;
> 
>-    // TODO(solenberg): Remove once clients don't use it anymore.
>-    int voe_channel_id = -1;
>-
>     // Bitrate limits used for variable audio bitrate streams. Set both to -1 to
>     // disable audio bitrate adaptation.
>     // Note: This is still an experimental feature and not ready for real usage.
>@@ -100,7 +101,7 @@ class AudioSendStream {
> 
>     // Defines whether to turn on audio network adaptor, and defines its config
>     // string.
>-    rtc::Optional<std::string> audio_network_adaptor_config;
>+    absl::optional<std::string> audio_network_adaptor_config;
> 
>     struct SendCodecSpec {
>       SendCodecSpec(int payload_type, const SdpAudioFormat& format);
>@@ -116,13 +117,14 @@ class AudioSendStream {
>       SdpAudioFormat format;
>       bool nack_enabled = false;
>       bool transport_cc_enabled = false;
>-      rtc::Optional<int> cng_payload_type;
>+      absl::optional<int> cng_payload_type;
>       // If unset, use the encoder's default target bitrate.
>-      rtc::Optional<int> target_bitrate_bps;
>+      absl::optional<int> target_bitrate_bps;
>     };
> 
>-    rtc::Optional<SendCodecSpec> send_codec_spec;
>+    absl::optional<SendCodecSpec> send_codec_spec;
>     rtc::scoped_refptr<AudioEncoderFactory> encoder_factory;
>+    absl::optional<AudioCodecPairId> codec_pair_id;
> 
>     // Track ID as specified during track creation.
>     std::string track_id;
>@@ -147,8 +149,10 @@ class AudioSendStream {
>       std::unique_ptr<webrtc::AudioFrame> audio_frame) = 0;
> 
>   // TODO(solenberg): Make payload_type a config property instead.
>-  virtual bool SendTelephoneEvent(int payload_type, int payload_frequency,
>-                                  int event, int duration_ms) = 0;
>+  virtual bool SendTelephoneEvent(int payload_type,
>+                                  int payload_frequency,
>+                                  int event,
>+                                  int duration_ms) = 0;
> 
>   virtual void SetMuted(bool muted) = 0;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/audio_state.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/audio_state.cc
>new file mode 100644
>index 00000000000..725d27f423d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/audio_state.cc
>@@ -0,0 +1,18 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "call/audio_state.h"
>+
>+namespace webrtc {
>+
>+AudioState::Config::Config() = default;
>+AudioState::Config::~Config() = default;
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/audio_state.h b/Source/ThirdParty/libwebrtc/Source/webrtc/call/audio_state.h
>index a8e57f0f3f5..104f493fb6e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/audio_state.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/audio_state.h
>@@ -11,23 +11,22 @@
> #define CALL_AUDIO_STATE_H_
> 
> #include "api/audio/audio_mixer.h"
>+#include "modules/audio_device/include/audio_device.h"
>+#include "modules/audio_processing/include/audio_processing.h"
> #include "rtc_base/refcount.h"
> #include "rtc_base/scoped_ref_ptr.h"
> 
> namespace webrtc {
> 
>-class AudioDeviceModule;
>-class AudioProcessing;
> class AudioTransport;
>-class VoiceEngine;
> 
> // AudioState holds the state which must be shared between multiple instances of
> // webrtc::Call for audio processing purposes.
> class AudioState : public rtc::RefCountInterface {
>  public:
>   struct Config {
>-    // TODO(solenberg): Remove once clients don't use it anymore.
>-    VoiceEngine* voice_engine = nullptr;
>+    Config();
>+    ~Config();
> 
>     // The audio mixer connected to active receive streams. One per
>     // AudioState.
>@@ -43,9 +42,8 @@ class AudioState : public rtc::RefCountInterface {
>   struct Stats {
>     // Audio peak level (max(abs())), linearly on the interval [0,32767].
>     int32_t audio_level = -1;
>-    // Audio peak level (max(abs())), logarithmically on the interval [0,9].
>-    int8_t quantized_audio_level = -1;
>-    // See: https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy
>+    // See:
>+    // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudioenergy
>     double total_energy = 0.0f;
>     double total_duration = 0.0f;
>   };
>@@ -71,7 +69,7 @@ class AudioState : public rtc::RefCountInterface {
>   static rtc::scoped_refptr<AudioState> Create(
>       const AudioState::Config& config);
> 
>-  virtual ~AudioState() {}
>+  ~AudioState() override {}
> };
> }  // namespace webrtc
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/bitrate_allocator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/bitrate_allocator.cc
>index 0b8ce39c4b5..c7049ba2111 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/bitrate_allocator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/bitrate_allocator.cc
>@@ -20,16 +20,14 @@
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> #include "system_wrappers/include/clock.h"
>+#include "system_wrappers/include/field_trial.h"
> #include "system_wrappers/include/metrics.h"
> 
> namespace webrtc {
> 
> // Allow packets to be transmitted in up to 2 times max video bitrate if the
> // bandwidth estimate allows it.
>-// TODO(bugs.webrtc.org/8541): May be worth to refactor to keep this logic in
>-// video send stream. Similar logic is implemented in
>-// AudioPriorityBitrateAllocationStrategy.
>-const int kTransmissionMaxBitrateMultiplier = 2;
>+const uint8_t kTransmissionMaxBitrateMultiplier = 2;
> const int kDefaultBitrateBps = 300000;
> 
> // Require a bitrate increase of max(10%, 20kbps) to resume paused streams.
>@@ -61,7 +59,11 @@ BitrateAllocator::BitrateAllocator(LimitObserver* limit_observer)
>       last_bwe_log_time_(0),
>       total_requested_padding_bitrate_(0),
>       total_requested_min_bitrate_(0),
>-      bitrate_allocation_strategy_(nullptr) {
>+      total_requested_max_bitrate_(0),
>+      has_packet_feedback_(false),
>+      bitrate_allocation_strategy_(nullptr),
>+      transmission_max_bitrate_multiplier_(
>+          GetTransmissionMaxBitrateMultiplier()) {
>   sequenced_checker_.Detach();
> }
> 
>@@ -70,6 +72,20 @@ BitrateAllocator::~BitrateAllocator() {
>                            num_pause_events_);
> }
> 
>+// static
>+uint8_t BitrateAllocator::GetTransmissionMaxBitrateMultiplier() {
>+  uint64_t multiplier = strtoul(webrtc::field_trial::FindFullName(
>+                                    "WebRTC-TransmissionMaxBitrateMultiplier")
>+                                    .c_str(),
>+                                nullptr, 10);
>+  if (multiplier > 0 && multiplier <= kTransmissionMaxBitrateMultiplier) {
>+    RTC_LOG(LS_INFO) << "TransmissionMaxBitrateMultiplier is set to "
>+                     << multiplier;
>+    return static_cast<uint8_t>(multiplier);
>+  }
>+  return kTransmissionMaxBitrateMultiplier;
>+}
>+
> void BitrateAllocator::OnNetworkChanged(uint32_t target_bitrate_bps,
>                                         uint8_t fraction_loss,
>                                         int64_t rtt,
>@@ -94,8 +110,7 @@ void BitrateAllocator::OnNetworkChanged(uint32_t target_bitrate_bps,
>   for (auto& config : bitrate_observer_configs_) {
>     uint32_t allocated_bitrate = allocation[config.observer];
>     uint32_t protection_bitrate = config.observer->OnBitrateUpdated(
>-        allocated_bitrate, last_fraction_loss_, last_rtt_,
>-        last_bwe_period_ms_);
>+        allocated_bitrate, last_fraction_loss_, last_rtt_, last_bwe_period_ms_);
> 
>     if (allocated_bitrate == 0 && config.allocated_bitrate_bps > 0) {
>       if (target_bitrate_bps > 0)
>@@ -127,28 +142,24 @@ void BitrateAllocator::OnNetworkChanged(uint32_t target_bitrate_bps,
> }
> 
> void BitrateAllocator::AddObserver(BitrateAllocatorObserver* observer,
>-                                   uint32_t min_bitrate_bps,
>-                                   uint32_t max_bitrate_bps,
>-                                   uint32_t pad_up_bitrate_bps,
>-                                   bool enforce_min_bitrate,
>-                                   std::string track_id,
>-                                   double bitrate_priority) {
>+                                   MediaStreamAllocationConfig config) {
>   RTC_DCHECK_CALLED_SEQUENTIALLY(&sequenced_checker_);
>-  RTC_DCHECK_GT(bitrate_priority, 0);
>-  RTC_DCHECK(std::isnormal(bitrate_priority));
>+  RTC_DCHECK_GT(config.bitrate_priority, 0);
>+  RTC_DCHECK(std::isnormal(config.bitrate_priority));
>   auto it = FindObserverConfig(observer);
> 
>   // Update settings if the observer already exists, create a new one otherwise.
>   if (it != bitrate_observer_configs_.end()) {
>-    it->min_bitrate_bps = min_bitrate_bps;
>-    it->max_bitrate_bps = max_bitrate_bps;
>-    it->pad_up_bitrate_bps = pad_up_bitrate_bps;
>-    it->enforce_min_bitrate = enforce_min_bitrate;
>-    it->bitrate_priority = bitrate_priority;
>+    it->min_bitrate_bps = config.min_bitrate_bps;
>+    it->max_bitrate_bps = config.max_bitrate_bps;
>+    it->pad_up_bitrate_bps = config.pad_up_bitrate_bps;
>+    it->enforce_min_bitrate = config.enforce_min_bitrate;
>+    it->bitrate_priority = config.bitrate_priority;
>   } else {
>     bitrate_observer_configs_.push_back(ObserverConfig(
>-        observer, min_bitrate_bps, max_bitrate_bps, pad_up_bitrate_bps,
>-        enforce_min_bitrate, track_id, bitrate_priority));
>+        observer, config.min_bitrate_bps, config.max_bitrate_bps,
>+        config.pad_up_bitrate_bps, config.enforce_min_bitrate, config.track_id,
>+        config.bitrate_priority, config.has_packet_feedback));
>   }
> 
>   ObserverAllocation allocation;
>@@ -176,10 +187,10 @@ void BitrateAllocator::AddObserver(BitrateAllocatorObserver* observer,
> }
> 
> void BitrateAllocator::UpdateAllocationLimits() {
>-  RTC_DCHECK_CALLED_SEQUENTIALLY(&sequenced_checker_);
>   uint32_t total_requested_padding_bitrate = 0;
>   uint32_t total_requested_min_bitrate = 0;
>-
>+  uint32_t total_requested_max_bitrate = 0;
>+  bool has_packet_feedback = false;
>   for (const auto& config : bitrate_observer_configs_) {
>     uint32_t stream_padding = config.pad_up_bitrate_bps;
>     if (config.enforce_min_bitrate) {
>@@ -189,22 +200,32 @@ void BitrateAllocator::UpdateAllocationLimits() {
>           std::max(config.MinBitrateWithHysteresis(), stream_padding);
>     }
>     total_requested_padding_bitrate += stream_padding;
>+    total_requested_max_bitrate += config.max_bitrate_bps;
>+    if (config.allocated_bitrate_bps > 0 && config.has_packet_feedback)
>+      has_packet_feedback = true;
>   }
> 
>   if (total_requested_padding_bitrate == total_requested_padding_bitrate_ &&
>-      total_requested_min_bitrate == total_requested_min_bitrate_) {
>+      total_requested_min_bitrate == total_requested_min_bitrate_ &&
>+      total_requested_max_bitrate == total_requested_max_bitrate_ &&
>+      has_packet_feedback == has_packet_feedback_) {
>     return;
>   }
> 
>   total_requested_min_bitrate_ = total_requested_min_bitrate;
>   total_requested_padding_bitrate_ = total_requested_padding_bitrate;
>+  total_requested_max_bitrate_ = total_requested_max_bitrate;
>+  has_packet_feedback_ = has_packet_feedback;
> 
>   RTC_LOG(LS_INFO) << "UpdateAllocationLimits : total_requested_min_bitrate: "
>                    << total_requested_min_bitrate
>                    << "bps, total_requested_padding_bitrate: "
>-                   << total_requested_padding_bitrate << "bps";
>-  limit_observer_->OnAllocationLimitsChanged(total_requested_min_bitrate,
>-                                             total_requested_padding_bitrate);
>+                   << total_requested_padding_bitrate
>+                   << "bps, total_requested_max_bitrate: "
>+                   << total_requested_max_bitrate << "bps";
>+  limit_observer_->OnAllocationLimitsChanged(
>+      total_requested_min_bitrate, total_requested_padding_bitrate,
>+      total_requested_max_bitrate, has_packet_feedback);
> }
> 
> void BitrateAllocator::RemoveObserver(BitrateAllocatorObserver* observer) {
>@@ -244,7 +265,6 @@ void BitrateAllocator::SetBitrateAllocationStrategy(
> 
> BitrateAllocator::ObserverConfigs::iterator
> BitrateAllocator::FindObserverConfig(const BitrateAllocatorObserver* observer) {
>-  RTC_DCHECK_CALLED_SEQUENTIALLY(&sequenced_checker_);
>   for (auto it = bitrate_observer_configs_.begin();
>        it != bitrate_observer_configs_.end(); ++it) {
>     if (it->observer == observer)
>@@ -255,7 +275,6 @@ BitrateAllocator::FindObserverConfig(const BitrateAllocatorObserver* observer) {
> 
> BitrateAllocator::ObserverAllocation BitrateAllocator::AllocateBitrates(
>     uint32_t bitrate) {
>-  RTC_DCHECK_CALLED_SEQUENTIALLY(&sequenced_checker_);
>   if (bitrate_observer_configs_.empty())
>     return ObserverAllocation();
> 
>@@ -299,12 +318,11 @@ BitrateAllocator::ObserverAllocation BitrateAllocator::AllocateBitrates(
>   if (bitrate <= sum_max_bitrates)
>     return NormalRateAllocation(bitrate, sum_min_bitrates);
> 
>-  // All observers will get up to kTransmissionMaxBitrateMultiplier x max.
>+  // All observers will get up to transmission_max_bitrate_multiplier_ x max.
>   return MaxRateAllocation(bitrate, sum_max_bitrates);
> }
> 
> BitrateAllocator::ObserverAllocation BitrateAllocator::ZeroRateAllocation() {
>-  RTC_DCHECK_CALLED_SEQUENTIALLY(&sequenced_checker_);
>   ObserverAllocation allocation;
>   for (const auto& observer_config : bitrate_observer_configs_)
>     allocation[observer_config.observer] = 0;
>@@ -313,7 +331,6 @@ BitrateAllocator::ObserverAllocation BitrateAllocator::ZeroRateAllocation() {
> 
> BitrateAllocator::ObserverAllocation BitrateAllocator::LowRateAllocation(
>     uint32_t bitrate) {
>-  RTC_DCHECK_CALLED_SEQUENTIALLY(&sequenced_checker_);
>   ObserverAllocation allocation;
>   // Start by allocating bitrate to observers enforcing a min bitrate, hence
>   // remaining_bitrate might turn negative.
>@@ -375,7 +392,6 @@ BitrateAllocator::ObserverAllocation BitrateAllocator::LowRateAllocation(
> BitrateAllocator::ObserverAllocation BitrateAllocator::NormalRateAllocation(
>     uint32_t bitrate,
>     uint32_t sum_min_bitrates) {
>-  RTC_DCHECK_CALLED_SEQUENTIALLY(&sequenced_checker_);
>   ObserverAllocation allocation;
>   ObserverAllocation observers_capacities;
>   for (const auto& observer_config : bitrate_observer_configs_) {
>@@ -396,14 +412,13 @@ BitrateAllocator::ObserverAllocation BitrateAllocator::NormalRateAllocation(
> BitrateAllocator::ObserverAllocation BitrateAllocator::MaxRateAllocation(
>     uint32_t bitrate,
>     uint32_t sum_max_bitrates) {
>-  RTC_DCHECK_CALLED_SEQUENTIALLY(&sequenced_checker_);
>   ObserverAllocation allocation;
> 
>   for (const auto& observer_config : bitrate_observer_configs_) {
>     allocation[observer_config.observer] = observer_config.max_bitrate_bps;
>     bitrate -= observer_config.max_bitrate_bps;
>   }
>-  DistributeBitrateEvenly(bitrate, true, kTransmissionMaxBitrateMultiplier,
>+  DistributeBitrateEvenly(bitrate, true, transmission_max_bitrate_multiplier_,
>                           &allocation);
>   return allocation;
> }
>@@ -437,7 +452,6 @@ void BitrateAllocator::DistributeBitrateEvenly(uint32_t bitrate,
>                                                bool include_zero_allocations,
>                                                int max_multiplier,
>                                                ObserverAllocation* allocation) {
>-  RTC_DCHECK_CALLED_SEQUENTIALLY(&sequenced_checker_);
>   RTC_DCHECK_EQ(allocation->size(), bitrate_observer_configs_.size());
> 
>   ObserverSortingMap list_max_bitrates;
>@@ -470,7 +484,6 @@ void BitrateAllocator::DistributeBitrateEvenly(uint32_t bitrate,
> 
> bool BitrateAllocator::EnoughBitrateForAllObservers(uint32_t bitrate,
>                                                     uint32_t sum_min_bitrates) {
>-  RTC_DCHECK_CALLED_SEQUENTIALLY(&sequenced_checker_);
>   if (bitrate < sum_min_bitrates)
>     return false;
> 
>@@ -490,7 +503,6 @@ void BitrateAllocator::DistributeBitrateRelatively(
>     uint32_t remaining_bitrate,
>     const ObserverAllocation& observers_capacities,
>     ObserverAllocation* allocation) {
>-  RTC_DCHECK_CALLED_SEQUENTIALLY(&sequenced_checker_);
>   RTC_DCHECK_EQ(allocation->size(), bitrate_observer_configs_.size());
>   RTC_DCHECK_EQ(observers_capacities.size(), bitrate_observer_configs_.size());
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/bitrate_allocator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/call/bitrate_allocator.h
>index b1afc8c424e..c29ea5e0899 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/bitrate_allocator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/bitrate_allocator.h
>@@ -43,25 +43,62 @@ class BitrateAllocatorObserver {
>   virtual ~BitrateAllocatorObserver() {}
> };
> 
>+// Struct describing parameters for how a media stream should get bitrate
>+// allocated to it. |min_bitrate_bps| = 0 equals no min bitrate.
>+// |max_bitrate_bps| = 0 equals no max bitrate.
>+// |enforce_min_bitrate| = 'true' will allocate at least |min_bitrate_bps| for
>+//    this observer, even if the BWE is too low, 'false' will allocate 0 to
>+//    the observer if BWE doesn't allow |min_bitrate_bps|.
>+// |has_packet_feedback| indicates whether the data produced by the
>+// corresponding media stream will receive per packet feedback. This is
>+// tracked here to communicate to limit observers whether packet feedback can
>+// be expected, which is true if any of the active observers has packet
>+// feedback enabled. Note that |observer|->OnBitrateUpdated() will be called
>+// within the scope of this method with the current rtt, fraction_loss and
>+// available bitrate and that the bitrate in OnBitrateUpdated will be zero if
>+// the |observer| is currently not allowed to send data.
>+struct MediaStreamAllocationConfig {
>+  uint32_t min_bitrate_bps;
>+  uint32_t max_bitrate_bps;
>+  uint32_t pad_up_bitrate_bps;
>+  bool enforce_min_bitrate;
>+  std::string track_id;
>+  double bitrate_priority;
>+  bool has_packet_feedback;
>+};
>+
>+// Interface used for mocking
>+class BitrateAllocatorInterface {
>+ public:
>+  virtual void AddObserver(BitrateAllocatorObserver* observer,
>+                           MediaStreamAllocationConfig config) = 0;
>+  virtual void RemoveObserver(BitrateAllocatorObserver* observer) = 0;
>+  virtual int GetStartBitrate(BitrateAllocatorObserver* observer) = 0;
>+
>+ protected:
>+  virtual ~BitrateAllocatorInterface() = default;
>+};
>+
> // Usage: this class will register multiple RtcpBitrateObserver's one at each
> // RTCP module. It will aggregate the results and run one bandwidth estimation
> // and push the result to the encoders via BitrateAllocatorObserver(s).
>-class BitrateAllocator {
>+class BitrateAllocator : public BitrateAllocatorInterface {
>  public:
>   // Used to get notified when send stream limits such as the minimum send
>   // bitrate and max padding bitrate is changed.
>   class LimitObserver {
>    public:
>-    virtual void OnAllocationLimitsChanged(
>-        uint32_t min_send_bitrate_bps,
>-        uint32_t max_padding_bitrate_bps) = 0;
>+    virtual void OnAllocationLimitsChanged(uint32_t min_send_bitrate_bps,
>+                                           uint32_t max_padding_bitrate_bps,
>+                                           uint32_t total_bitrate_bps,
>+                                           bool has_packet_feedback) = 0;
> 
>    protected:
>-    virtual ~LimitObserver() {}
>+    virtual ~LimitObserver() = default;
>   };
> 
>   explicit BitrateAllocator(LimitObserver* limit_observer);
>-  ~BitrateAllocator();
>+  ~BitrateAllocator() override;
> 
>   // Allocate target_bitrate across the registered BitrateAllocatorObservers.
>   void OnNetworkChanged(uint32_t target_bitrate_bps,
>@@ -69,33 +106,19 @@ class BitrateAllocator {
>                         int64_t rtt,
>                         int64_t bwe_period_ms);
> 
>-  // Set the start and max send bitrate used by the bandwidth management.
>-  //
>+  // Set the configuration used by the bandwidth management.
>   // |observer| updates bitrates if already in use.
>-  // |min_bitrate_bps| = 0 equals no min bitrate.
>-  // |max_bitrate_bps| = 0 equals no max bitrate.
>-  // |enforce_min_bitrate| = 'true' will allocate at least |min_bitrate_bps| for
>-  //    this observer, even if the BWE is too low, 'false' will allocate 0 to
>-  //    the observer if BWE doesn't allow |min_bitrate_bps|.
>-  // Note that |observer|->OnBitrateUpdated() will be called within the scope of
>-  // this method with the current rtt, fraction_loss and available bitrate and
>-  // that the bitrate in OnBitrateUpdated will be zero if the |observer| is
>-  // currently not allowed to send data.
>+  // |config| is the configuration to use for allocation.
>   void AddObserver(BitrateAllocatorObserver* observer,
>-                   uint32_t min_bitrate_bps,
>-                   uint32_t max_bitrate_bps,
>-                   uint32_t pad_up_bitrate_bps,
>-                   bool enforce_min_bitrate,
>-                   std::string track_id,
>-                   double bitrate_priority);
>+                   MediaStreamAllocationConfig config) override;
> 
>   // Removes a previously added observer, but will not trigger a new bitrate
>   // allocation.
>-  void RemoveObserver(BitrateAllocatorObserver* observer);
>+  void RemoveObserver(BitrateAllocatorObserver* observer) override;
> 
>   // Returns initial bitrate allocated for |observer|. If |observer| is not in
>   // the list of added observers, a best guess is returned.
>-  int GetStartBitrate(BitrateAllocatorObserver* observer);
>+  int GetStartBitrate(BitrateAllocatorObserver* observer) override;
> 
>   // Sets external allocation strategy. If strategy is not set default WebRTC
>   // allocation mechanism will be used. The strategy may be changed during call.
>@@ -112,7 +135,8 @@ class BitrateAllocator {
>                    uint32_t pad_up_bitrate_bps,
>                    bool enforce_min_bitrate,
>                    std::string track_id,
>-                   double bitrate_priority)
>+                   double bitrate_priority,
>+                   bool has_packet_feedback)
>         : TrackConfig(min_bitrate_bps,
>                       max_bitrate_bps,
>                       enforce_min_bitrate,
>@@ -121,7 +145,8 @@ class BitrateAllocator {
>           pad_up_bitrate_bps(pad_up_bitrate_bps),
>           allocated_bitrate_bps(-1),
>           media_ratio(1.0),
>-          bitrate_priority(bitrate_priority) {}
>+          bitrate_priority(bitrate_priority),
>+          has_packet_feedback(has_packet_feedback) {}
> 
>     BitrateAllocatorObserver* observer;
>     uint32_t pad_up_bitrate_bps;
>@@ -131,6 +156,7 @@ class BitrateAllocator {
>     // observers. If an observer has twice the bitrate_priority of other
>     // observers, it should be allocated twice the bitrate above its min.
>     double bitrate_priority;
>+    bool has_packet_feedback;
> 
>     uint32_t LastAllocatedBitrate() const;
>     // The minimum bitrate required by this observer, including
>@@ -140,31 +166,35 @@ class BitrateAllocator {
> 
>   // Calculates the minimum requested send bitrate and max padding bitrate and
>   // calls LimitObserver::OnAllocationLimitsChanged.
>-  void UpdateAllocationLimits();
>+  void UpdateAllocationLimits() RTC_RUN_ON(&sequenced_checker_);
> 
>   typedef std::vector<ObserverConfig> ObserverConfigs;
>   ObserverConfigs::iterator FindObserverConfig(
>-      const BitrateAllocatorObserver* observer);
>+      const BitrateAllocatorObserver* observer) RTC_RUN_ON(&sequenced_checker_);
> 
>   typedef std::multimap<uint32_t, const ObserverConfig*> ObserverSortingMap;
>   typedef std::map<BitrateAllocatorObserver*, int> ObserverAllocation;
> 
>-  ObserverAllocation AllocateBitrates(uint32_t bitrate);
>+  ObserverAllocation AllocateBitrates(uint32_t bitrate)
>+      RTC_RUN_ON(&sequenced_checker_);
> 
>   // Allocates zero bitrate to all observers.
>-  ObserverAllocation ZeroRateAllocation();
>+  ObserverAllocation ZeroRateAllocation() RTC_RUN_ON(&sequenced_checker_);
>   // Allocates bitrate to observers when there isn't enough to allocate the
>   // minimum to all observers.
>-  ObserverAllocation LowRateAllocation(uint32_t bitrate);
>+  ObserverAllocation LowRateAllocation(uint32_t bitrate)
>+      RTC_RUN_ON(&sequenced_checker_);
>   // Allocates bitrate to all observers when the available bandwidth is enough
>   // to allocate the minimum to all observers but not enough to allocate the
>   // max bitrate of each observer.
>   ObserverAllocation NormalRateAllocation(uint32_t bitrate,
>-                                          uint32_t sum_min_bitrates);
>+                                          uint32_t sum_min_bitrates)
>+      RTC_RUN_ON(&sequenced_checker_);
>   // Allocates bitrate to observers when there is enough available bandwidth
>   // for all observers to be allocated their max bitrate.
>   ObserverAllocation MaxRateAllocation(uint32_t bitrate,
>-                                       uint32_t sum_max_bitrates);
>+                                       uint32_t sum_max_bitrates)
>+      RTC_RUN_ON(&sequenced_checker_);
> 
>   // Splits |bitrate| evenly to observers already in |allocation|.
>   // |include_zero_allocations| decides if zero allocations should be part of
>@@ -173,9 +203,10 @@ class BitrateAllocator {
>   void DistributeBitrateEvenly(uint32_t bitrate,
>                                bool include_zero_allocations,
>                                int max_multiplier,
>-                               ObserverAllocation* allocation);
>-  bool EnoughBitrateForAllObservers(uint32_t bitrate,
>-                                    uint32_t sum_min_bitrates);
>+                               ObserverAllocation* allocation)
>+      RTC_RUN_ON(&sequenced_checker_);
>+  bool EnoughBitrateForAllObservers(uint32_t bitrate, uint32_t sum_min_bitrates)
>+      RTC_RUN_ON(&sequenced_checker_);
> 
>   // From the available |bitrate|, each observer will be allocated a
>   // proportional amount based upon its bitrate priority. If that amount is
>@@ -186,7 +217,14 @@ class BitrateAllocator {
>   void DistributeBitrateRelatively(
>       uint32_t bitrate,
>       const ObserverAllocation& observers_capacities,
>-      ObserverAllocation* allocation);
>+      ObserverAllocation* allocation) RTC_RUN_ON(&sequenced_checker_);
>+
>+  // Allow packets to be transmitted in up to 2 times max video bitrate if the
>+  // bandwidth estimate allows it.
>+  // TODO(bugs.webrtc.org/8541): May be worth to refactor to keep this logic in
>+  // video send stream. Similar logic is implemented in
>+  // AudioPriorityBitrateAllocationStrategy.
>+  static uint8_t GetTransmissionMaxBitrateMultiplier();
> 
>   rtc::SequencedTaskChecker sequenced_checker_;
>   LimitObserver* const limit_observer_ RTC_GUARDED_BY(&sequenced_checker_);
>@@ -203,8 +241,11 @@ class BitrateAllocator {
>   int64_t last_bwe_log_time_ RTC_GUARDED_BY(&sequenced_checker_);
>   uint32_t total_requested_padding_bitrate_ RTC_GUARDED_BY(&sequenced_checker_);
>   uint32_t total_requested_min_bitrate_ RTC_GUARDED_BY(&sequenced_checker_);
>+  uint32_t total_requested_max_bitrate_ RTC_GUARDED_BY(&sequenced_checker_);
>+  bool has_packet_feedback_ RTC_GUARDED_BY(&sequenced_checker_);
>   std::unique_ptr<rtc::BitrateAllocationStrategy> bitrate_allocation_strategy_
>       RTC_GUARDED_BY(&sequenced_checker_);
>+  const uint8_t transmission_max_bitrate_multiplier_;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/bitrate_allocator_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/bitrate_allocator_unittest.cc
>index 154d14e3d76..7814655358c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/bitrate_allocator_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/bitrate_allocator_unittest.cc
>@@ -21,12 +21,28 @@ using ::testing::NiceMock;
> using ::testing::_;
> 
> namespace webrtc {
>+// Emulating old interface for test suite compatibility.
>+// TODO(srte): Update tests to reflect new interface.
>+class LimitObserverWrapper : public BitrateAllocator::LimitObserver {
>+ public:
>+  void OnAllocationLimitsChanged(uint32_t min_send_bitrate_bps,
>+                                 uint32_t max_padding_bitrate_bps,
>+                                 uint32_t total_bitrate_bps,
>+                                 bool has_packet_feedback) override {
>+    OnAllocationLimitsChanged(min_send_bitrate_bps, max_padding_bitrate_bps,
>+                              total_bitrate_bps);
>+  }
>+  virtual void OnAllocationLimitsChanged(uint32_t min_send_bitrate_bps,
>+                                         uint32_t max_padding_bitrate_bps,
>+                                         uint32_t total_bitrate_bps) = 0;
>+};
> 
>-class MockLimitObserver : public BitrateAllocator::LimitObserver {
>+class MockLimitObserver : public LimitObserverWrapper {
>  public:
>-  MOCK_METHOD2(OnAllocationLimitsChanged,
>+  MOCK_METHOD3(OnAllocationLimitsChanged,
>                void(uint32_t min_send_bitrate_bps,
>-                    uint32_t max_padding_bitrate_bps));
>+                    uint32_t max_padding_bitrate_bps,
>+                    uint32_t total_bitrate_bps));
> };
> 
> class TestBitrateObserver : public BitrateAllocatorObserver {
>@@ -62,7 +78,7 @@ class TestBitrateObserver : public BitrateAllocatorObserver {
> namespace {
> constexpr int64_t kDefaultProbingIntervalMs = 3000;
> const double kDefaultBitratePriority = 1.0;
>-}
>+}  // namespace
> 
> class BitrateAllocatorTest : public ::testing::Test {
>  protected:
>@@ -70,6 +86,17 @@ class BitrateAllocatorTest : public ::testing::Test {
>     allocator_->OnNetworkChanged(300000u, 0, 0, kDefaultProbingIntervalMs);
>   }
>   ~BitrateAllocatorTest() {}
>+  void AddObserver(BitrateAllocatorObserver* observer,
>+                   uint32_t min_bitrate_bps,
>+                   uint32_t max_bitrate_bps,
>+                   uint32_t pad_up_bitrate_bps,
>+                   bool enforce_min_bitrate,
>+                   std::string track_id,
>+                   double bitrate_priority) {
>+    allocator_->AddObserver(
>+        observer, {min_bitrate_bps, max_bitrate_bps, pad_up_bitrate_bps,
>+                   enforce_min_bitrate, track_id, bitrate_priority, false});
>+  }
> 
>   NiceMock<MockLimitObserver> limit_observer_;
>   std::unique_ptr<BitrateAllocator> allocator_;
>@@ -79,12 +106,13 @@ TEST_F(BitrateAllocatorTest, UpdatingBitrateObserver) {
>   TestBitrateObserver bitrate_observer;
>   const uint32_t kMinSendBitrateBps = 100000;
>   const uint32_t kPadUpToBitrateBps = 50000;
>+  const uint32_t kMaxBitrateBps = 1500000;
> 
>-  EXPECT_CALL(limit_observer_, OnAllocationLimitsChanged(kMinSendBitrateBps,
>-                                                         kPadUpToBitrateBps));
>-  allocator_->AddObserver(&bitrate_observer, kMinSendBitrateBps, 1500000,
>-                          kPadUpToBitrateBps, true, "",
>-                          kDefaultBitratePriority);
>+  EXPECT_CALL(limit_observer_,
>+              OnAllocationLimitsChanged(kMinSendBitrateBps, kPadUpToBitrateBps,
>+                                        kMaxBitrateBps));
>+  AddObserver(&bitrate_observer, kMinSendBitrateBps, kMaxBitrateBps,
>+              kPadUpToBitrateBps, true, "", kDefaultBitratePriority);
>   EXPECT_EQ(300000, allocator_->GetStartBitrate(&bitrate_observer));
>   allocator_->OnNetworkChanged(200000, 0, 0, kDefaultProbingIntervalMs);
>   EXPECT_EQ(200000, allocator_->GetStartBitrate(&bitrate_observer));
>@@ -96,36 +124,49 @@ TEST_F(BitrateAllocatorTest, UpdatingBitrateObserver) {
> 
>   // Expect |max_padding_bitrate_bps| to change to 0 if the observer is updated.
>   EXPECT_CALL(limit_observer_,
>-              OnAllocationLimitsChanged(kMinSendBitrateBps, 0));
>-  allocator_->AddObserver(&bitrate_observer, kMinSendBitrateBps, 4000000, 0,
>-                          true, "", kDefaultBitratePriority);
>+              OnAllocationLimitsChanged(kMinSendBitrateBps, 0, _));
>+  AddObserver(&bitrate_observer, kMinSendBitrateBps, 4000000, 0, true, "",
>+              kDefaultBitratePriority);
>+  EXPECT_CALL(limit_observer_,
>+              OnAllocationLimitsChanged(kMinSendBitrateBps, 0, _));
>   EXPECT_EQ(4000000, allocator_->GetStartBitrate(&bitrate_observer));
> 
>-  allocator_->AddObserver(&bitrate_observer, kMinSendBitrateBps, 1500000, 0,
>-                          true, "", kDefaultBitratePriority);
>+  AddObserver(&bitrate_observer, kMinSendBitrateBps, kMaxBitrateBps, 0, true,
>+              "", kDefaultBitratePriority);
>   EXPECT_EQ(3000000, allocator_->GetStartBitrate(&bitrate_observer));
>   EXPECT_EQ(3000000u, bitrate_observer.last_bitrate_bps_);
>-  allocator_->OnNetworkChanged(1500000, 0, 0, kDefaultProbingIntervalMs);
>+  allocator_->OnNetworkChanged(kMaxBitrateBps, 0, 0, kDefaultProbingIntervalMs);
>   EXPECT_EQ(1500000u, bitrate_observer.last_bitrate_bps_);
> }
> 
> TEST_F(BitrateAllocatorTest, TwoBitrateObserversOneRtcpObserver) {
>   TestBitrateObserver bitrate_observer_1;
>   TestBitrateObserver bitrate_observer_2;
>-  EXPECT_CALL(limit_observer_, OnAllocationLimitsChanged(100000, 0));
>-  allocator_->AddObserver(&bitrate_observer_1, 100000, 300000, 0, true, "",
>-                          kDefaultBitratePriority);
>-  EXPECT_EQ(300000, allocator_->GetStartBitrate(&bitrate_observer_1));
>-  EXPECT_CALL(limit_observer_,
>-              OnAllocationLimitsChanged(100000 + 200000, 0));
>-  allocator_->AddObserver(&bitrate_observer_2, 200000, 300000, 0, true, "",
>-                          kDefaultBitratePriority);
>-  EXPECT_EQ(200000, allocator_->GetStartBitrate(&bitrate_observer_2));
>+  const uint32_t kObs1StartBitrateBps = 100000;
>+  const uint32_t kObs2StartBitrateBps = 200000;
>+  const uint32_t kObs1MaxBitrateBps = 300000;
>+  const uint32_t kObs2MaxBitrateBps = 300000;
>+
>+  EXPECT_CALL(
>+      limit_observer_,
>+      OnAllocationLimitsChanged(kObs1StartBitrateBps, 0, kObs1MaxBitrateBps));
>+  AddObserver(&bitrate_observer_1, kObs1StartBitrateBps, kObs1MaxBitrateBps, 0,
>+              true, "", kDefaultBitratePriority);
>+  EXPECT_EQ(static_cast<int>(kObs1MaxBitrateBps),
>+            allocator_->GetStartBitrate(&bitrate_observer_1));
>+  EXPECT_CALL(limit_observer_, OnAllocationLimitsChanged(
>+                                   kObs1StartBitrateBps + kObs2StartBitrateBps,
>+                                   0, kObs1MaxBitrateBps + kObs2MaxBitrateBps));
>+  AddObserver(&bitrate_observer_2, kObs2StartBitrateBps, kObs2MaxBitrateBps, 0,
>+              true, "", kDefaultBitratePriority);
>+  EXPECT_EQ(static_cast<int>(kObs2StartBitrateBps),
>+            allocator_->GetStartBitrate(&bitrate_observer_2));
> 
>   // Test too low start bitrate, hence lower than sum of min. Min bitrates
>   // will
>   // be allocated to all observers.
>-  allocator_->OnNetworkChanged(200000, 0, 50, kDefaultProbingIntervalMs);
>+  allocator_->OnNetworkChanged(kObs2StartBitrateBps, 0, 50,
>+                               kDefaultProbingIntervalMs);
>   EXPECT_EQ(100000u, bitrate_observer_1.last_bitrate_bps_);
>   EXPECT_EQ(0, bitrate_observer_1.last_fraction_loss_);
>   EXPECT_EQ(50, bitrate_observer_1.last_rtt_ms_);
>@@ -135,7 +176,8 @@ TEST_F(BitrateAllocatorTest, TwoBitrateObserversOneRtcpObserver) {
> 
>   // Test a bitrate which should be distributed equally.
>   allocator_->OnNetworkChanged(500000, 0, 50, kDefaultProbingIntervalMs);
>-  const uint32_t kBitrateToShare = 500000 - 200000 - 100000;
>+  const uint32_t kBitrateToShare =
>+      500000 - kObs2StartBitrateBps - kObs1StartBitrateBps;
>   EXPECT_EQ(100000u + kBitrateToShare / 2,
>             bitrate_observer_1.last_bitrate_bps_);
>   EXPECT_EQ(200000u + kBitrateToShare / 2,
>@@ -159,13 +201,14 @@ TEST_F(BitrateAllocatorTest, RemoveObserverTriggersLimitObserver) {
>   TestBitrateObserver bitrate_observer;
>   const uint32_t kMinSendBitrateBps = 100000;
>   const uint32_t kPadUpToBitrateBps = 50000;
>+  const uint32_t kMaxBitrateBps = 1500000;
> 
>-  EXPECT_CALL(limit_observer_, OnAllocationLimitsChanged(kMinSendBitrateBps,
>-                                                         kPadUpToBitrateBps));
>-  allocator_->AddObserver(&bitrate_observer, kMinSendBitrateBps, 1500000,
>-                          kPadUpToBitrateBps, true, "",
>-                          kDefaultBitratePriority);
>-  EXPECT_CALL(limit_observer_, OnAllocationLimitsChanged(0, 0));
>+  EXPECT_CALL(limit_observer_,
>+              OnAllocationLimitsChanged(kMinSendBitrateBps, kPadUpToBitrateBps,
>+                                        kMaxBitrateBps));
>+  AddObserver(&bitrate_observer, kMinSendBitrateBps, kMaxBitrateBps,
>+              kPadUpToBitrateBps, true, "", kDefaultBitratePriority);
>+  EXPECT_CALL(limit_observer_, OnAllocationLimitsChanged(0, 0, _));
>   allocator_->RemoveObserver(&bitrate_observer);
> }
> 
>@@ -176,7 +219,17 @@ class BitrateAllocatorTestNoEnforceMin : public ::testing::Test {
>     allocator_->OnNetworkChanged(300000u, 0, 0, kDefaultProbingIntervalMs);
>   }
>   ~BitrateAllocatorTestNoEnforceMin() {}
>-
>+  void AddObserver(BitrateAllocatorObserver* observer,
>+                   uint32_t min_bitrate_bps,
>+                   uint32_t max_bitrate_bps,
>+                   uint32_t pad_up_bitrate_bps,
>+                   bool enforce_min_bitrate,
>+                   std::string track_id,
>+                   double bitrate_priority) {
>+    allocator_->AddObserver(
>+        observer, {min_bitrate_bps, max_bitrate_bps, pad_up_bitrate_bps,
>+                   enforce_min_bitrate, track_id, bitrate_priority, false});
>+  }
>   NiceMock<MockLimitObserver> limit_observer_;
>   std::unique_ptr<BitrateAllocator> allocator_;
> };
>@@ -187,9 +240,10 @@ TEST_F(BitrateAllocatorTestNoEnforceMin, OneBitrateObserver) {
>   TestBitrateObserver bitrate_observer_1;
>   // Expect OnAllocationLimitsChanged with |min_send_bitrate_bps| = 0 since
>   // AddObserver is called with |enforce_min_bitrate| = false.
>-  EXPECT_CALL(limit_observer_, OnAllocationLimitsChanged(0, 120000));
>-  allocator_->AddObserver(&bitrate_observer_1, 100000, 400000, 0, false, "",
>-                          kDefaultBitratePriority);
>+  EXPECT_CALL(limit_observer_, OnAllocationLimitsChanged(0, 0, _));
>+  EXPECT_CALL(limit_observer_, OnAllocationLimitsChanged(0, 120000, _));
>+  AddObserver(&bitrate_observer_1, 100000, 400000, 0, false, "",
>+              kDefaultBitratePriority);
>   EXPECT_EQ(300000, allocator_->GetStartBitrate(&bitrate_observer_1));
> 
>   // High BWE.
>@@ -200,7 +254,7 @@ TEST_F(BitrateAllocatorTestNoEnforceMin, OneBitrateObserver) {
>   allocator_->OnNetworkChanged(10000, 0, 0, kDefaultProbingIntervalMs);
>   EXPECT_EQ(0u, bitrate_observer_1.last_bitrate_bps_);
> 
>-  EXPECT_CALL(limit_observer_, OnAllocationLimitsChanged(0, 0));
>+  EXPECT_CALL(limit_observer_, OnAllocationLimitsChanged(0, 0, _));
>   allocator_->RemoveObserver(&bitrate_observer_1);
> }
> 
>@@ -209,17 +263,17 @@ TEST_F(BitrateAllocatorTestNoEnforceMin, ThreeBitrateObservers) {
>   TestBitrateObserver bitrate_observer_2;
>   TestBitrateObserver bitrate_observer_3;
>   // Set up the observers with min bitrates at 100000, 200000, and 300000.
>-  allocator_->AddObserver(&bitrate_observer_1, 100000, 400000, 0, false, "",
>-                          kDefaultBitratePriority);
>+  AddObserver(&bitrate_observer_1, 100000, 400000, 0, false, "",
>+              kDefaultBitratePriority);
>   EXPECT_EQ(300000, allocator_->GetStartBitrate(&bitrate_observer_1));
> 
>-  allocator_->AddObserver(&bitrate_observer_2, 200000, 400000, 0, false, "",
>-                          kDefaultBitratePriority);
>+  AddObserver(&bitrate_observer_2, 200000, 400000, 0, false, "",
>+              kDefaultBitratePriority);
>   EXPECT_EQ(200000, allocator_->GetStartBitrate(&bitrate_observer_2));
>   EXPECT_EQ(100000u, bitrate_observer_1.last_bitrate_bps_);
> 
>-  allocator_->AddObserver(&bitrate_observer_3, 300000, 400000, 0, false, "",
>-                          kDefaultBitratePriority);
>+  AddObserver(&bitrate_observer_3, 300000, 400000, 0, false, "",
>+              kDefaultBitratePriority);
>   EXPECT_EQ(0, allocator_->GetStartBitrate(&bitrate_observer_3));
>   EXPECT_EQ(100000u, bitrate_observer_1.last_bitrate_bps_);
>   EXPECT_EQ(200000u, bitrate_observer_2.last_bitrate_bps_);
>@@ -241,14 +295,14 @@ TEST_F(BitrateAllocatorTestNoEnforceMin, ThreeBitrateObservers) {
>   allocator_->OnNetworkChanged(300000, 0, 0, kDefaultProbingIntervalMs);
>   EXPECT_EQ(100000u, bitrate_observer_1.last_bitrate_bps_);  // Min bitrate.
>   EXPECT_EQ(200000u, bitrate_observer_2.last_bitrate_bps_);  // Min bitrate.
>-  EXPECT_EQ(0u, bitrate_observer_3.last_bitrate_bps_);  // Nothing.
>+  EXPECT_EQ(0u, bitrate_observer_3.last_bitrate_bps_);       // Nothing.
> 
>   // Increased BWE, but still below the sum of configured min bitrates for all
>   // observers and too little for observer 3. 1 and 2 will share the rest.
>   allocator_->OnNetworkChanged(500000, 0, 0, kDefaultProbingIntervalMs);
>   EXPECT_EQ(200000u, bitrate_observer_1.last_bitrate_bps_);  // Min + split.
>   EXPECT_EQ(300000u, bitrate_observer_2.last_bitrate_bps_);  // Min + split.
>-  EXPECT_EQ(0u, bitrate_observer_3.last_bitrate_bps_);  // Nothing.
>+  EXPECT_EQ(0u, bitrate_observer_3.last_bitrate_bps_);       // Nothing.
> 
>   // Below min for all.
>   allocator_->OnNetworkChanged(10000, 0, 0, kDefaultProbingIntervalMs);
>@@ -272,9 +326,10 @@ TEST_F(BitrateAllocatorTestNoEnforceMin, OneBitrateObserverWithPacketLoss) {
>   TestBitrateObserver bitrate_observer;
>   // Expect OnAllocationLimitsChanged with |min_send_bitrate_bps| = 0 since
>   // AddObserver is called with |enforce_min_bitrate| = false.
>-  EXPECT_CALL(limit_observer_, OnAllocationLimitsChanged(0, 168000));
>-  allocator_->AddObserver(&bitrate_observer, 100000, 400000, 0, false, "",
>-                          kDefaultBitratePriority);
>+  EXPECT_CALL(limit_observer_, OnAllocationLimitsChanged(0, 0, _));
>+  EXPECT_CALL(limit_observer_, OnAllocationLimitsChanged(0, 168000, _));
>+  AddObserver(&bitrate_observer, 100000, 400000, 0, false, "",
>+              kDefaultBitratePriority);
>   EXPECT_EQ(300000, allocator_->GetStartBitrate(&bitrate_observer));
> 
>   // High BWE.
>@@ -300,7 +355,7 @@ TEST_F(BitrateAllocatorTestNoEnforceMin, OneBitrateObserverWithPacketLoss) {
>   EXPECT_EQ(0u, bitrate_observer.last_bitrate_bps_);
> 
>   // Just enough to enable video again.
>-  EXPECT_CALL(limit_observer_, OnAllocationLimitsChanged(0, 0));
>+  EXPECT_CALL(limit_observer_, OnAllocationLimitsChanged(0, 0, _));
>   allocator_->OnNetworkChanged(168000, 0, fraction_loss,
>                                kDefaultProbingIntervalMs);
>   EXPECT_EQ(168000u, bitrate_observer.last_bitrate_bps_);
>@@ -313,6 +368,7 @@ TEST_F(BitrateAllocatorTestNoEnforceMin, OneBitrateObserverWithPacketLoss) {
>   allocator_->OnNetworkChanged(139000, 0, 0, kDefaultProbingIntervalMs);
>   EXPECT_EQ(139000u, bitrate_observer.last_bitrate_bps_);
> 
>+  EXPECT_CALL(limit_observer_, OnAllocationLimitsChanged(0, 0, _));
>   allocator_->RemoveObserver(&bitrate_observer);
> }
> 
>@@ -320,11 +376,11 @@ TEST_F(BitrateAllocatorTestNoEnforceMin, TwoBitrateObserverWithPacketLoss) {
>   TestBitrateObserver bitrate_observer_1;
>   TestBitrateObserver bitrate_observer_2;
> 
>-  allocator_->AddObserver(&bitrate_observer_1, 100000, 400000, 0, false, "",
>-                          kDefaultBitratePriority);
>+  AddObserver(&bitrate_observer_1, 100000, 400000, 0, false, "",
>+              kDefaultBitratePriority);
>   EXPECT_EQ(300000, allocator_->GetStartBitrate(&bitrate_observer_1));
>-  allocator_->AddObserver(&bitrate_observer_2, 200000, 400000, 0, false, "",
>-                          kDefaultBitratePriority);
>+  AddObserver(&bitrate_observer_2, 200000, 400000, 0, false, "",
>+              kDefaultBitratePriority);
>   EXPECT_EQ(200000, allocator_->GetStartBitrate(&bitrate_observer_2));
>   EXPECT_EQ(100000u, bitrate_observer_1.last_bitrate_bps_);
> 
>@@ -373,17 +429,17 @@ TEST_F(BitrateAllocatorTest, ThreeBitrateObserversLowBweEnforceMin) {
>   TestBitrateObserver bitrate_observer_2;
>   TestBitrateObserver bitrate_observer_3;
> 
>-  allocator_->AddObserver(&bitrate_observer_1, 100000, 400000, 0, true, "",
>-                          kDefaultBitratePriority);
>+  AddObserver(&bitrate_observer_1, 100000, 400000, 0, true, "",
>+              kDefaultBitratePriority);
>   EXPECT_EQ(300000, allocator_->GetStartBitrate(&bitrate_observer_1));
> 
>-  allocator_->AddObserver(&bitrate_observer_2, 200000, 400000, 0, true, "",
>-                          kDefaultBitratePriority);
>+  AddObserver(&bitrate_observer_2, 200000, 400000, 0, true, "",
>+              kDefaultBitratePriority);
>   EXPECT_EQ(200000, allocator_->GetStartBitrate(&bitrate_observer_2));
>   EXPECT_EQ(100000u, bitrate_observer_1.last_bitrate_bps_);
> 
>-  allocator_->AddObserver(&bitrate_observer_3, 300000, 400000, 0, true, "",
>-                          kDefaultBitratePriority);
>+  AddObserver(&bitrate_observer_3, 300000, 400000, 0, true, "",
>+              kDefaultBitratePriority);
>   EXPECT_EQ(300000, allocator_->GetStartBitrate(&bitrate_observer_3));
>   EXPECT_EQ(100000, static_cast<int>(bitrate_observer_1.last_bitrate_bps_));
>   EXPECT_EQ(200000, static_cast<int>(bitrate_observer_2.last_bitrate_bps_));
>@@ -402,10 +458,10 @@ TEST_F(BitrateAllocatorTest, ThreeBitrateObserversLowBweEnforceMin) {
> 
> TEST_F(BitrateAllocatorTest, AddObserverWhileNetworkDown) {
>   TestBitrateObserver bitrate_observer_1;
>-  EXPECT_CALL(limit_observer_, OnAllocationLimitsChanged(50000, 0));
>+  EXPECT_CALL(limit_observer_, OnAllocationLimitsChanged(50000, 0, _));
> 
>-  allocator_->AddObserver(&bitrate_observer_1, 50000, 400000, 0, true, "",
>-                          kDefaultBitratePriority);
>+  AddObserver(&bitrate_observer_1, 50000, 400000, 0, true, "",
>+              kDefaultBitratePriority);
>   EXPECT_EQ(300000, allocator_->GetStartBitrate(&bitrate_observer_1));
> 
>   // Set network down, ie, no available bitrate.
>@@ -415,9 +471,9 @@ TEST_F(BitrateAllocatorTest, AddObserverWhileNetworkDown) {
> 
>   TestBitrateObserver bitrate_observer_2;
>   // Adding an observer while the network is down should not affect the limits.
>-  EXPECT_CALL(limit_observer_, OnAllocationLimitsChanged(50000 + 50000, 0));
>-  allocator_->AddObserver(&bitrate_observer_2, 50000, 400000, 0, true, "",
>-                          kDefaultBitratePriority);
>+  EXPECT_CALL(limit_observer_, OnAllocationLimitsChanged(50000 + 50000, 0, _));
>+  AddObserver(&bitrate_observer_2, 50000, 400000, 0, true, "",
>+              kDefaultBitratePriority);
> 
>   // Expect the start_bitrate to be set as if the network was still up but that
>   // the new observer have been notified that the network is down.
>@@ -433,13 +489,13 @@ TEST_F(BitrateAllocatorTest, AddObserverWhileNetworkDown) {
> 
> TEST_F(BitrateAllocatorTest, MixedEnforecedConfigs) {
>   TestBitrateObserver enforced_observer;
>-  allocator_->AddObserver(&enforced_observer, 6000, 30000, 0, true, "",
>-                          kDefaultBitratePriority);
>+  AddObserver(&enforced_observer, 6000, 30000, 0, true, "",
>+              kDefaultBitratePriority);
>   EXPECT_EQ(60000, allocator_->GetStartBitrate(&enforced_observer));
> 
>   TestBitrateObserver not_enforced_observer;
>-  allocator_->AddObserver(&not_enforced_observer, 30000, 2500000, 0, false, "",
>-                          kDefaultBitratePriority);
>+  AddObserver(&not_enforced_observer, 30000, 2500000, 0, false, "",
>+              kDefaultBitratePriority);
>   EXPECT_EQ(270000, allocator_->GetStartBitrate(&not_enforced_observer));
>   EXPECT_EQ(30000u, enforced_observer.last_bitrate_bps_);
> 
>@@ -477,8 +533,7 @@ TEST_F(BitrateAllocatorTest, MixedEnforecedConfigs) {
> 
> TEST_F(BitrateAllocatorTest, AvoidToggleAbsolute) {
>   TestBitrateObserver observer;
>-  allocator_->AddObserver(&observer, 30000, 300000, 0, false, "",
>-                          kDefaultBitratePriority);
>+  AddObserver(&observer, 30000, 300000, 0, false, "", kDefaultBitratePriority);
>   EXPECT_EQ(300000, allocator_->GetStartBitrate(&observer));
> 
>   allocator_->OnNetworkChanged(30000, 0, 50, kDefaultProbingIntervalMs);
>@@ -504,8 +559,7 @@ TEST_F(BitrateAllocatorTest, AvoidToggleAbsolute) {
> 
> TEST_F(BitrateAllocatorTest, AvoidTogglePercent) {
>   TestBitrateObserver observer;
>-  allocator_->AddObserver(&observer, 300000, 600000, 0, false, "",
>-                          kDefaultBitratePriority);
>+  AddObserver(&observer, 300000, 600000, 0, false, "", kDefaultBitratePriority);
>   EXPECT_EQ(300000, allocator_->GetStartBitrate(&observer));
> 
>   allocator_->OnNetworkChanged(300000, 0, 50, kDefaultProbingIntervalMs);
>@@ -531,8 +585,7 @@ TEST_F(BitrateAllocatorTest, AvoidTogglePercent) {
> 
> TEST_F(BitrateAllocatorTest, PassProbingInterval) {
>   TestBitrateObserver observer;
>-  allocator_->AddObserver(&observer, 300000, 600000, 0, false, "",
>-                          kDefaultBitratePriority);
>+  AddObserver(&observer, 300000, 600000, 0, false, "", kDefaultBitratePriority);
>   EXPECT_EQ(300000, allocator_->GetStartBitrate(&observer));
> 
>   allocator_->OnNetworkChanged(300000, 0, 50, 5000);
>@@ -547,8 +600,8 @@ TEST_F(BitrateAllocatorTest, PriorityRateOneObserverBasic) {
>   const uint32_t kMaxSendBitrateBps = 60;
>   const uint32_t kNetworkBandwidthBps = 30;
> 
>-  allocator_->AddObserver(&observer, kMinSendBitrateBps, kMaxSendBitrateBps, 0,
>-                          true, "", 2.0);
>+  AddObserver(&observer, kMinSendBitrateBps, kMaxSendBitrateBps, 0, true, "",
>+              2.0);
>   allocator_->OnNetworkChanged(kNetworkBandwidthBps, 0, 0,
>                                kDefaultProbingIntervalMs);
> 
>@@ -565,10 +618,10 @@ TEST_F(BitrateAllocatorTest, PriorityRateTwoObserversBasic) {
>   const uint32_t kMinSendBitrateBps = 10;
>   const uint32_t kMaxSendBitrateBps = 60;
>   const uint32_t kNetworkBandwidthBps = 60;
>-  allocator_->AddObserver(&observer_low_1, kMinSendBitrateBps,
>-                          kMaxSendBitrateBps, 0, false, "low1", 2.0);
>-  allocator_->AddObserver(&observer_low_2, kMinSendBitrateBps,
>-                          kMaxSendBitrateBps, 0, false, "low2", 2.0);
>+  AddObserver(&observer_low_1, kMinSendBitrateBps, kMaxSendBitrateBps, 0, false,
>+              "low1", 2.0);
>+  AddObserver(&observer_low_2, kMinSendBitrateBps, kMaxSendBitrateBps, 0, false,
>+              "low2", 2.0);
>   allocator_->OnNetworkChanged(kNetworkBandwidthBps, 0, 0,
>                                kDefaultProbingIntervalMs);
> 
>@@ -587,10 +640,10 @@ TEST_F(BitrateAllocatorTest, PriorityRateTwoObserversBasicMinEnforced) {
>   const uint32_t kMinSendBitrateBps = 0;
>   const uint32_t kMaxSendBitrateBps = 60;
>   const uint32_t kNetworkBandwidthBps = 60;
>-  allocator_->AddObserver(&observer_low_1, kMinSendBitrateBps,
>-                          kMaxSendBitrateBps, 0, true, "low1", 2.0);
>-  allocator_->AddObserver(&observer_low_2, kMinSendBitrateBps,
>-                          kMaxSendBitrateBps, 0, true, "low2", 2.0);
>+  AddObserver(&observer_low_1, kMinSendBitrateBps, kMaxSendBitrateBps, 0, true,
>+              "low1", 2.0);
>+  AddObserver(&observer_low_2, kMinSendBitrateBps, kMaxSendBitrateBps, 0, true,
>+              "low2", 2.0);
>   allocator_->OnNetworkChanged(kNetworkBandwidthBps, 0, 0,
>                                kDefaultProbingIntervalMs);
> 
>@@ -609,10 +662,10 @@ TEST_F(BitrateAllocatorTest, PriorityRateTwoObserversBothAllocatedMax) {
>   const uint32_t kMinSendBitrateBps = 0;
>   const uint32_t kMaxSendBitrateBps = 60;
>   const uint32_t kNetworkBandwidthBps = kMaxSendBitrateBps * 2;
>-  allocator_->AddObserver(&observer_low, kMinSendBitrateBps, kMaxSendBitrateBps,
>-                          0, true, "low", 2.0);
>-  allocator_->AddObserver(&observer_mid, kMinSendBitrateBps, kMaxSendBitrateBps,
>-                          0, true, "mid", 4.0);
>+  AddObserver(&observer_low, kMinSendBitrateBps, kMaxSendBitrateBps, 0, true,
>+              "low", 2.0);
>+  AddObserver(&observer_mid, kMinSendBitrateBps, kMaxSendBitrateBps, 0, true,
>+              "mid", 4.0);
>   allocator_->OnNetworkChanged(kNetworkBandwidthBps, 0, 0,
>                                kDefaultProbingIntervalMs);
> 
>@@ -629,8 +682,8 @@ TEST_F(BitrateAllocatorTest, PriorityRateTwoObserversBothAllocatedMax) {
> TEST_F(BitrateAllocatorTest, PriorityRateTwoObserversOneAllocatedToMax) {
>   TestBitrateObserver observer_low;
>   TestBitrateObserver observer_mid;
>-  allocator_->AddObserver(&observer_low, 10, 50, 0, false, "low", 2.0);
>-  allocator_->AddObserver(&observer_mid, 10, 50, 0, false, "mid", 4.0);
>+  AddObserver(&observer_low, 10, 50, 0, false, "low", 2.0);
>+  AddObserver(&observer_mid, 10, 50, 0, false, "mid", 4.0);
>   allocator_->OnNetworkChanged(90, 0, 0, kDefaultProbingIntervalMs);
> 
>   EXPECT_EQ(40u, observer_low.last_bitrate_bps_);
>@@ -655,12 +708,12 @@ TEST_F(BitrateAllocatorTest,
>   const double kHighBitratePriority = 8.0;
>   const double kTotalBitratePriority =
>       kLowBitratePriority + kMidBitratePriority + kHighBitratePriority;
>-  allocator_->AddObserver(&observer_low, 0, kMaxBitrate, 0, false, "low",
>-                          kLowBitratePriority);
>-  allocator_->AddObserver(&observer_mid, 0, kMaxBitrate, 0, false, "mid",
>-                          kMidBitratePriority);
>-  allocator_->AddObserver(&observer_high, 0, kMaxBitrate, 0, false, "high",
>-                          kHighBitratePriority);
>+  AddObserver(&observer_low, 0, kMaxBitrate, 0, false, "low",
>+              kLowBitratePriority);
>+  AddObserver(&observer_mid, 0, kMaxBitrate, 0, false, "mid",
>+              kMidBitratePriority);
>+  AddObserver(&observer_high, 0, kMaxBitrate, 0, false, "high",
>+              kHighBitratePriority);
>   allocator_->OnNetworkChanged(kNetworkBandwidthBps, 0, 0,
>                                kDefaultProbingIntervalMs);
> 
>@@ -701,12 +754,12 @@ TEST_F(BitrateAllocatorTest, PriorityRateThreeObserversHighAllocatedToMax) {
>   const uint32_t kRemainingBitrate =
>       kAvailableBitrate - kMaxBitrate - (2 * kMinBitrate);
> 
>-  allocator_->AddObserver(&observer_low, kMinBitrate, kMaxBitrate, 0, false,
>-                          "low", kLowBitratePriority);
>-  allocator_->AddObserver(&observer_mid, kMinBitrate, kMaxBitrate, 0, false,
>-                          "mid", kMidBitratePriority);
>-  allocator_->AddObserver(&observer_high, kMinBitrate, kMaxBitrate, 0, false,
>-                          "high", kHighBitratePriority);
>+  AddObserver(&observer_low, kMinBitrate, kMaxBitrate, 0, false, "low",
>+              kLowBitratePriority);
>+  AddObserver(&observer_mid, kMinBitrate, kMaxBitrate, 0, false, "mid",
>+              kMidBitratePriority);
>+  AddObserver(&observer_high, kMinBitrate, kMaxBitrate, 0, false, "high",
>+              kHighBitratePriority);
>   allocator_->OnNetworkChanged(kAvailableBitrate, 0, 0,
>                                kDefaultProbingIntervalMs);
> 
>@@ -745,12 +798,12 @@ TEST_F(BitrateAllocatorTest, PriorityRateThreeObserversLowAllocatedToMax) {
>   // available bitrate, so 70 bps would be sufficient network bandwidth.
>   const uint32_t kRemainingBitrate = kAvailableBitrate - kLowMaxBitrate;
> 
>-  allocator_->AddObserver(&observer_low, kMinBitrate, kLowMaxBitrate, 0, false,
>-                          "low", kLowBitratePriority);
>-  allocator_->AddObserver(&observer_mid, kMinBitrate, kMaxBitrate, 0, false,
>-                          "mid", kMidBitratePriority);
>-  allocator_->AddObserver(&observer_high, kMinBitrate, kMaxBitrate, 0, false,
>-                          "high", kHighBitratePriority);
>+  AddObserver(&observer_low, kMinBitrate, kLowMaxBitrate, 0, false, "low",
>+              kLowBitratePriority);
>+  AddObserver(&observer_mid, kMinBitrate, kMaxBitrate, 0, false, "mid",
>+              kMidBitratePriority);
>+  AddObserver(&observer_high, kMinBitrate, kMaxBitrate, 0, false, "high",
>+              kHighBitratePriority);
>   allocator_->OnNetworkChanged(kAvailableBitrate, 0, 0,
>                                kDefaultProbingIntervalMs);
> 
>@@ -779,12 +832,12 @@ TEST_F(BitrateAllocatorTest, PriorityRateThreeObserversTwoAllocatedToMax) {
>   TestBitrateObserver observer_low;
>   TestBitrateObserver observer_mid;
>   TestBitrateObserver observer_high;
>-  allocator_->AddObserver(&observer_low, 10, 40, 0, false, "low", 2.0);
>+  AddObserver(&observer_low, 10, 40, 0, false, "low", 2.0);
>   // Scaled allocation above the min allocation is the same for these two,
>   // meaning they will get allocated  their max at the same time.
>   // Scaled (target allocation) = (max - min) / bitrate priority
>-  allocator_->AddObserver(&observer_mid, 10, 30, 0, false, "mid", 4.0);
>-  allocator_->AddObserver(&observer_high, 10, 50, 0, false, "high", 8.0);
>+  AddObserver(&observer_mid, 10, 30, 0, false, "mid", 4.0);
>+  AddObserver(&observer_high, 10, 50, 0, false, "high", 8.0);
>   allocator_->OnNetworkChanged(110, 0, 0, kDefaultProbingIntervalMs);
> 
>   EXPECT_EQ(30u, observer_low.last_bitrate_bps_);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/bitrate_estimator_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/bitrate_estimator_tests.cc
>index 4c397875f97..b3f7856ca35 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/bitrate_estimator_tests.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/bitrate_estimator_tests.cc
>@@ -103,9 +103,7 @@ class BitrateEstimatorTest : public test::CallTest {
> 
>   virtual void SetUp() {
>     task_queue_.SendTask([this]() {
>-      Call::Config config(event_log_.get());
>-      receiver_call_.reset(Call::Create(config));
>-      sender_call_.reset(Call::Create(config));
>+      CreateCalls();
> 
>       send_transport_.reset(new test::DirectTransport(
>           &task_queue_, sender_call_.get(), payload_type_map_));
>@@ -114,18 +112,20 @@ class BitrateEstimatorTest : public test::CallTest {
>           &task_queue_, receiver_call_.get(), payload_type_map_));
>       receive_transport_->SetReceiver(sender_call_->Receiver());
> 
>-      video_send_config_ = VideoSendStream::Config(send_transport_.get());
>-      video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[0]);
>-      // Encoders will be set separately per stream.
>-      video_send_config_.encoder_settings.encoder = nullptr;
>-      video_send_config_.encoder_settings.payload_name = "FAKE";
>-      video_send_config_.encoder_settings.payload_type =
>-          kFakeVideoSendPayloadType;
>-      test::FillEncoderConfiguration(1, &video_encoder_config_);
>+      VideoSendStream::Config video_send_config(send_transport_.get());
>+      video_send_config.rtp.ssrcs.push_back(kVideoSendSsrcs[0]);
>+      video_send_config.encoder_settings.encoder_factory =
>+          &fake_encoder_factory_;
>+      video_send_config.rtp.payload_name = "FAKE";
>+      video_send_config.rtp.payload_type = kFakeVideoSendPayloadType;
>+      SetVideoSendConfig(video_send_config);
>+      VideoEncoderConfig video_encoder_config;
>+      test::FillEncoderConfiguration(kVideoCodecVP8, 1, &video_encoder_config);
>+      SetVideoEncoderConfig(video_encoder_config);
> 
>       receive_config_ = VideoReceiveStream::Config(receive_transport_.get());
>       // receive_config_.decoders will be set by every stream separately.
>-      receive_config_.rtp.remote_ssrc = video_send_config_.rtp.ssrcs[0];
>+      receive_config_.rtp.remote_ssrc = GetVideoSendConfig()->rtp.ssrcs[0];
>       receive_config_.rtp.local_ssrc = kReceiverLocalVideoSsrc;
>       receive_config_.rtp.remb = true;
>       receive_config_.rtp.extensions.push_back(
>@@ -137,19 +137,16 @@ class BitrateEstimatorTest : public test::CallTest {
> 
>   virtual void TearDown() {
>     task_queue_.SendTask([this]() {
>-      std::for_each(streams_.begin(), streams_.end(),
>-                    std::mem_fun(&Stream::StopSending));
>-
>-      while (!streams_.empty()) {
>-        delete streams_.back();
>-        streams_.pop_back();
>+      for (auto* stream : streams_) {
>+        stream->StopSending();
>+        delete stream;
>       }
>+      streams_.clear();
> 
>       send_transport_.reset();
>       receive_transport_.reset();
> 
>-      receiver_call_.reset();
>-      sender_call_.reset();
>+      DestroyCalls();
>     });
>   }
> 
>@@ -163,33 +160,28 @@ class BitrateEstimatorTest : public test::CallTest {
>           is_sending_receiving_(false),
>           send_stream_(nullptr),
>           frame_generator_capturer_(),
>-          fake_encoder_(Clock::GetRealTimeClock()),
>           fake_decoder_() {
>-      test_->video_send_config_.rtp.ssrcs[0]++;
>-      test_->video_send_config_.encoder_settings.encoder = &fake_encoder_;
>+      test_->GetVideoSendConfig()->rtp.ssrcs[0]++;
>       send_stream_ = test_->sender_call_->CreateVideoSendStream(
>-          test_->video_send_config_.Copy(),
>-          test_->video_encoder_config_.Copy());
>-      RTC_DCHECK_EQ(1, test_->video_encoder_config_.number_of_streams);
>+          test_->GetVideoSendConfig()->Copy(),
>+          test_->GetVideoEncoderConfig()->Copy());
>+      RTC_DCHECK_EQ(1, test_->GetVideoEncoderConfig()->number_of_streams);
>       frame_generator_capturer_.reset(test::FrameGeneratorCapturer::Create(
>-          kDefaultWidth, kDefaultHeight, kDefaultFramerate,
>-          Clock::GetRealTimeClock()));
>-      send_stream_->SetSource(
>-          frame_generator_capturer_.get(),
>-          VideoSendStream::DegradationPreference::kMaintainFramerate);
>+          kDefaultWidth, kDefaultHeight, absl::nullopt, absl::nullopt,
>+          kDefaultFramerate, Clock::GetRealTimeClock()));
>+      send_stream_->SetSource(frame_generator_capturer_.get(),
>+                              DegradationPreference::MAINTAIN_FRAMERATE);
>       send_stream_->Start();
>       frame_generator_capturer_->Start();
> 
>       VideoReceiveStream::Decoder decoder;
>       decoder.decoder = &fake_decoder_;
>-      decoder.payload_type =
>-          test_->video_send_config_.encoder_settings.payload_type;
>-      decoder.payload_name =
>-          test_->video_send_config_.encoder_settings.payload_name;
>+      decoder.payload_type = test_->GetVideoSendConfig()->rtp.payload_type;
>+      decoder.payload_name = test_->GetVideoSendConfig()->rtp.payload_name;
>       test_->receive_config_.decoders.clear();
>       test_->receive_config_.decoders.push_back(decoder);
>       test_->receive_config_.rtp.remote_ssrc =
>-          test_->video_send_config_.rtp.ssrcs[0];
>+          test_->GetVideoSendConfig()->rtp.ssrcs[0];
>       test_->receive_config_.rtp.local_ssrc++;
>       test_->receive_config_.renderer = &test->fake_renderer_;
>       video_receive_stream_ = test_->receiver_call_->CreateVideoReceiveStream(
>@@ -226,15 +218,12 @@ class BitrateEstimatorTest : public test::CallTest {
>     VideoSendStream* send_stream_;
>     VideoReceiveStream* video_receive_stream_;
>     std::unique_ptr<test::FrameGeneratorCapturer> frame_generator_capturer_;
>-    test::FakeEncoder fake_encoder_;
>     test::FakeDecoder fake_decoder_;
>   };
> 
>   LogObserver receiver_log_;
>   std::unique_ptr<test::DirectTransport> send_transport_;
>   std::unique_ptr<test::DirectTransport> receive_transport_;
>-  std::unique_ptr<Call> sender_call_;
>-  std::unique_ptr<Call> receiver_call_;
>   VideoReceiveStream::Config receive_config_;
>   std::vector<Stream*> streams_;
> };
>@@ -246,7 +235,7 @@ static const char* kSingleStreamLog =
> 
> TEST_F(BitrateEstimatorTest, InstantiatesTOFPerDefaultForVideo) {
>   task_queue_.SendTask([this]() {
>-    video_send_config_.rtp.extensions.push_back(
>+    GetVideoSendConfig()->rtp.extensions.push_back(
>         RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId));
>     receiver_log_.PushExpectedLogLine(kSingleStreamLog);
>     receiver_log_.PushExpectedLogLine(kSingleStreamLog);
>@@ -257,7 +246,7 @@ TEST_F(BitrateEstimatorTest, InstantiatesTOFPerDefaultForVideo) {
> 
> TEST_F(BitrateEstimatorTest, ImmediatelySwitchToASTForVideo) {
>   task_queue_.SendTask([this]() {
>-    video_send_config_.rtp.extensions.push_back(
>+    GetVideoSendConfig()->rtp.extensions.push_back(
>         RtpExtension(RtpExtension::kAbsSendTimeUri, kASTExtensionId));
>     receiver_log_.PushExpectedLogLine(kSingleStreamLog);
>     receiver_log_.PushExpectedLogLine(kSingleStreamLog);
>@@ -270,7 +259,7 @@ TEST_F(BitrateEstimatorTest, ImmediatelySwitchToASTForVideo) {
> 
> TEST_F(BitrateEstimatorTest, SwitchesToASTForVideo) {
>   task_queue_.SendTask([this]() {
>-    video_send_config_.rtp.extensions.push_back(
>+    GetVideoSendConfig()->rtp.extensions.push_back(
>         RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId));
>     receiver_log_.PushExpectedLogLine(kSingleStreamLog);
>     receiver_log_.PushExpectedLogLine(kSingleStreamLog);
>@@ -279,7 +268,7 @@ TEST_F(BitrateEstimatorTest, SwitchesToASTForVideo) {
>   EXPECT_TRUE(receiver_log_.Wait());
> 
>   task_queue_.SendTask([this]() {
>-    video_send_config_.rtp.extensions[0] =
>+    GetVideoSendConfig()->rtp.extensions[0] =
>         RtpExtension(RtpExtension::kAbsSendTimeUri, kASTExtensionId);
>     receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
>     receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
>@@ -291,7 +280,7 @@ TEST_F(BitrateEstimatorTest, SwitchesToASTForVideo) {
> // This test is flaky. See webrtc:5790.
> TEST_F(BitrateEstimatorTest, DISABLED_SwitchesToASTThenBackToTOFForVideo) {
>   task_queue_.SendTask([this]() {
>-    video_send_config_.rtp.extensions.push_back(
>+    GetVideoSendConfig()->rtp.extensions.push_back(
>         RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId));
>     receiver_log_.PushExpectedLogLine(kSingleStreamLog);
>     receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
>@@ -301,7 +290,7 @@ TEST_F(BitrateEstimatorTest, DISABLED_SwitchesToASTThenBackToTOFForVideo) {
>   EXPECT_TRUE(receiver_log_.Wait());
> 
>   task_queue_.SendTask([this]() {
>-    video_send_config_.rtp.extensions[0] =
>+    GetVideoSendConfig()->rtp.extensions[0] =
>         RtpExtension(RtpExtension::kAbsSendTimeUri, kASTExtensionId);
>     receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
>     receiver_log_.PushExpectedLogLine("Switching to absolute send time RBE.");
>@@ -310,7 +299,7 @@ TEST_F(BitrateEstimatorTest, DISABLED_SwitchesToASTThenBackToTOFForVideo) {
>   EXPECT_TRUE(receiver_log_.Wait());
> 
>   task_queue_.SendTask([this]() {
>-    video_send_config_.rtp.extensions[0] =
>+    GetVideoSendConfig()->rtp.extensions[0] =
>         RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId);
>     receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
>     receiver_log_.PushExpectedLogLine(
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/call.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/call.cc
>index 08e1d937fa6..113109a823c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/call.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/call.cc
>@@ -16,7 +16,9 @@
> #include <utility>
> #include <vector>
> 
>-#include "api/optional.h"
>+#include "absl/memory/memory.h"
>+#include "absl/types/optional.h"
>+#include "api/transport/network_control.h"
> #include "audio/audio_receive_stream.h"
> #include "audio/audio_send_stream.h"
> #include "audio/audio_state.h"
>@@ -24,6 +26,7 @@
> #include "call/bitrate_allocator.h"
> #include "call/call.h"
> #include "call/flexfec_receive_stream_impl.h"
>+#include "call/receive_time_calculator.h"
> #include "call/rtp_stream_receiver_controller.h"
> #include "call/rtp_transport_controller_send.h"
> #include "logging/rtc_event_log/events/rtc_event_audio_receive_stream_config.h"
>@@ -42,20 +45,21 @@
> #include "modules/rtp_rtcp/source/byte_io.h"
> #include "modules/rtp_rtcp/source/rtp_packet_received.h"
> #include "modules/utility/include/process_thread.h"
>-#include "rtc_base/basictypes.h"
>+#include "modules/video_coding/fec_controller_default.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/location.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/ptr_util.h"
>+#include "rtc_base/numerics/safe_minmax.h"
> #include "rtc_base/sequenced_task_checker.h"
>+#include "rtc_base/strings/string_builder.h"
>+#include "rtc_base/synchronization/rw_lock_wrapper.h"
> #include "rtc_base/task_queue.h"
> #include "rtc_base/thread_annotations.h"
> #include "rtc_base/trace_event.h"
> #include "system_wrappers/include/clock.h"
> #include "system_wrappers/include/cpu_info.h"
> #include "system_wrappers/include/metrics.h"
>-#include "system_wrappers/include/rw_lock_wrapper.h"
> #include "video/call_stats.h"
> #include "video/send_delay_stats.h"
> #include "video/stats_counter.h"
>@@ -65,7 +69,6 @@
> namespace webrtc {
> 
> namespace {
>-
> // TODO(nisse): This really begs for a shared context struct.
> bool UseSendSideBwe(const std::vector<RtpExtension>& extensions,
>                     bool transport_cc) {
>@@ -100,7 +103,7 @@ const int* FindKeyByValue(const std::map<int, int>& m, int v) {
> 
> std::unique_ptr<rtclog::StreamConfig> CreateRtcLogStreamConfig(
>     const VideoReceiveStream::Config& config) {
>-  auto rtclog_config = rtc::MakeUnique<rtclog::StreamConfig>();
>+  auto rtclog_config = absl::make_unique<rtclog::StreamConfig>();
>   rtclog_config->remote_ssrc = config.rtp.remote_ssrc;
>   rtclog_config->local_ssrc = config.rtp.local_ssrc;
>   rtclog_config->rtx_ssrc = config.rtp.rtx_ssrc;
>@@ -112,7 +115,7 @@ std::unique_ptr<rtclog::StreamConfig> CreateRtcLogStreamConfig(
>     const int* search =
>         FindKeyByValue(config.rtp.rtx_associated_payload_types, d.payload_type);
>     rtclog_config->codecs.emplace_back(d.payload_name, d.payload_type,
>-                                      search ? *search : 0);
>+                                       search ? *search : 0);
>   }
>   return rtclog_config;
> }
>@@ -120,7 +123,7 @@ std::unique_ptr<rtclog::StreamConfig> CreateRtcLogStreamConfig(
> std::unique_ptr<rtclog::StreamConfig> CreateRtcLogStreamConfig(
>     const VideoSendStream::Config& config,
>     size_t ssrc_index) {
>-  auto rtclog_config = rtc::MakeUnique<rtclog::StreamConfig>();
>+  auto rtclog_config = absl::make_unique<rtclog::StreamConfig>();
>   rtclog_config->local_ssrc = config.rtp.ssrcs[ssrc_index];
>   if (ssrc_index < config.rtp.rtx.ssrcs.size()) {
>     rtclog_config->rtx_ssrc = config.rtp.rtx.ssrcs[ssrc_index];
>@@ -128,15 +131,15 @@ std::unique_ptr<rtclog::StreamConfig> CreateRtcLogStreamConfig(
>   rtclog_config->rtcp_mode = config.rtp.rtcp_mode;
>   rtclog_config->rtp_extensions = config.rtp.extensions;
> 
>-  rtclog_config->codecs.emplace_back(config.encoder_settings.payload_name,
>-                                     config.encoder_settings.payload_type,
>+  rtclog_config->codecs.emplace_back(config.rtp.payload_name,
>+                                     config.rtp.payload_type,
>                                      config.rtp.rtx.payload_type);
>   return rtclog_config;
> }
> 
> std::unique_ptr<rtclog::StreamConfig> CreateRtcLogStreamConfig(
>     const AudioReceiveStream::Config& config) {
>-  auto rtclog_config = rtc::MakeUnique<rtclog::StreamConfig>();
>+  auto rtclog_config = absl::make_unique<rtclog::StreamConfig>();
>   rtclog_config->remote_ssrc = config.rtp.remote_ssrc;
>   rtclog_config->local_ssrc = config.rtp.local_ssrc;
>   rtclog_config->rtp_extensions = config.rtp.extensions;
>@@ -145,7 +148,7 @@ std::unique_ptr<rtclog::StreamConfig> CreateRtcLogStreamConfig(
> 
> std::unique_ptr<rtclog::StreamConfig> CreateRtcLogStreamConfig(
>     const AudioSendStream::Config& config) {
>-  auto rtclog_config = rtc::MakeUnique<rtclog::StreamConfig>();
>+  auto rtclog_config = absl::make_unique<rtclog::StreamConfig>();
>   rtclog_config->local_ssrc = config.rtp.ssrc;
>   rtclog_config->rtp_extensions = config.rtp.extensions;
>   if (config.send_codec_spec) {
>@@ -159,11 +162,11 @@ std::unique_ptr<rtclog::StreamConfig> CreateRtcLogStreamConfig(
> 
> namespace internal {
> 
>-class Call : public webrtc::Call,
>-             public PacketReceiver,
>-             public RecoveredPacketReceiver,
>-             public SendSideCongestionController::Observer,
>-             public BitrateAllocator::LimitObserver {
>+class Call final : public webrtc::Call,
>+                   public PacketReceiver,
>+                   public RecoveredPacketReceiver,
>+                   public TargetTransferRateObserver,
>+                   public BitrateAllocator::LimitObserver {
>  public:
>   Call(const Call::Config& config,
>        std::unique_ptr<RtpTransportControllerSendInterface> transport_send);
>@@ -184,6 +187,10 @@ class Call : public webrtc::Call,
>   webrtc::VideoSendStream* CreateVideoSendStream(
>       webrtc::VideoSendStream::Config config,
>       VideoEncoderConfig encoder_config) override;
>+  webrtc::VideoSendStream* CreateVideoSendStream(
>+      webrtc::VideoSendStream::Config config,
>+      VideoEncoderConfig encoder_config,
>+      std::unique_ptr<FecController> fec_controller) override;
>   void DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) override;
> 
>   webrtc::VideoReceiveStream* CreateVideoReceiveStream(
>@@ -196,22 +203,18 @@ class Call : public webrtc::Call,
>   void DestroyFlexfecReceiveStream(
>       FlexfecReceiveStream* receive_stream) override;
> 
>+  RtpTransportControllerSendInterface* GetTransportControllerSend() override;
>+
>   Stats GetStats() const override;
> 
>   // Implements PacketReceiver.
>   DeliveryStatus DeliverPacket(MediaType media_type,
>                                rtc::CopyOnWriteBuffer packet,
>-                               const PacketTime& packet_time) override;
>+                               int64_t packet_time_us) override;
> 
>   // Implements RecoveredPacketReceiver.
>   void OnRecoveredPacket(const uint8_t* packet, size_t length) override;
> 
>-  void SetBitrateConfig(
>-      const webrtc::Call::Config::BitrateConfig& bitrate_config) override;
>-
>-  void SetBitrateConfigMask(
>-      const webrtc::Call::Config::BitrateConfigMask& bitrate_config) override;
>-
>   void SetBitrateAllocationStrategy(
>       std::unique_ptr<rtc::BitrateAllocationStrategy>
>           bitrate_allocation_strategy) override;
>@@ -221,27 +224,24 @@ class Call : public webrtc::Call,
>   void OnTransportOverheadChanged(MediaType media,
>                                   int transport_overhead_per_packet) override;
> 
>-  void OnNetworkRouteChanged(const std::string& transport_name,
>-                             const rtc::NetworkRoute& network_route) override;
>-
>   void OnSentPacket(const rtc::SentPacket& sent_packet) override;
> 
>-  // Implements BitrateObserver.
>-  void OnNetworkChanged(uint32_t bitrate_bps,
>-                        uint8_t fraction_loss,
>-                        int64_t rtt_ms,
>-                        int64_t probing_interval_ms) override;
>+  // Implements TargetTransferRateObserver,
>+  void OnTargetTransferRate(TargetTransferRate msg) override;
> 
>   // Implements BitrateAllocator::LimitObserver.
>   void OnAllocationLimitsChanged(uint32_t min_send_bitrate_bps,
>-                                 uint32_t max_padding_bitrate_bps) override;
>+                                 uint32_t max_padding_bitrate_bps,
>+                                 uint32_t total_bitrate_bps,
>+                                 bool has_packet_feedback) override;
> 
>  private:
>-  DeliveryStatus DeliverRtcp(MediaType media_type, const uint8_t* packet,
>+  DeliveryStatus DeliverRtcp(MediaType media_type,
>+                             const uint8_t* packet,
>                              size_t length);
>   DeliveryStatus DeliverRtp(MediaType media_type,
>                             rtc::CopyOnWriteBuffer packet,
>-                            const PacketTime& packet_time);
>+                            int64_t packet_time_us);
>   void ConfigureSync(const std::string& sync_group)
>       RTC_EXCLUSIVE_LOCKS_REQUIRED(receive_crit_);
> 
>@@ -255,15 +255,10 @@ class Call : public webrtc::Call,
>   void UpdateHistograms();
>   void UpdateAggregateNetworkState();
> 
>-  // Applies update to the BitrateConfig cached in |config_|, restarting
>-  // bandwidth estimation from |new_start| if set.
>-  void UpdateCurrentBitrateConfig(const rtc::Optional<int>& new_start);
>-
>   Clock* const clock_;
> 
>   const int num_cpu_cores_;
>   const std::unique_ptr<ProcessThread> module_process_thread_;
>-  const std::unique_ptr<ProcessThread> pacer_thread_;
>   const std::unique_ptr<CallStats> call_stats_;
>   const std::unique_ptr<BitrateAllocator> bitrate_allocator_;
>   Call::Config config_;
>@@ -271,6 +266,8 @@ class Call : public webrtc::Call,
> 
>   NetworkState audio_network_state_;
>   NetworkState video_network_state_;
>+  rtc::CriticalSection aggregate_network_up_crit_;
>+  bool aggregate_network_up_ RTC_GUARDED_BY(aggregate_network_up_crit_);
> 
>   std::unique_ptr<RWLockWrapper> receive_crit_;
>   // Audio, Video, and FlexFEC receive streams are owned by the client that
>@@ -295,19 +292,24 @@ class Call : public webrtc::Call,
>   // single mapping from ssrc to a more abstract receive stream, with
>   // accessor methods for all configuration we need at this level.
>   struct ReceiveRtpConfig {
>-    ReceiveRtpConfig() = default;  // Needed by std::map
>-    ReceiveRtpConfig(const std::vector<RtpExtension>& extensions,
>-                     bool use_send_side_bwe)
>-        : extensions(extensions), use_send_side_bwe(use_send_side_bwe) {}
>+    explicit ReceiveRtpConfig(const webrtc::AudioReceiveStream::Config& config)
>+        : extensions(config.rtp.extensions),
>+          use_send_side_bwe(UseSendSideBwe(config)) {}
>+    explicit ReceiveRtpConfig(const webrtc::VideoReceiveStream::Config& config)
>+        : extensions(config.rtp.extensions),
>+          use_send_side_bwe(UseSendSideBwe(config)) {}
>+    explicit ReceiveRtpConfig(const FlexfecReceiveStream::Config& config)
>+        : extensions(config.rtp_header_extensions),
>+          use_send_side_bwe(UseSendSideBwe(config)) {}
> 
>     // Registered RTP header extensions for each stream. Note that RTP header
>     // extensions are negotiated per track ("m= line") in the SDP, but we have
>     // no notion of tracks at the Call level. We therefore store the RTP header
>     // extensions per SSRC instead, which leads to some storage overhead.
>-    RtpHeaderExtensionMap extensions;
>+    const RtpHeaderExtensionMap extensions;
>     // Set if both RTP extension the RTCP feedback message needed for
>     // send side BWE are negotiated.
>-    bool use_send_side_bwe = false;
>+    const bool use_send_side_bwe;
>   };
>   std::map<uint32_t, ReceiveRtpConfig> receive_rtp_config_
>       RTC_GUARDED_BY(receive_crit_);
>@@ -339,12 +341,14 @@ class Call : public webrtc::Call,
>   RateCounter received_audio_bytes_per_second_counter_;
>   RateCounter received_video_bytes_per_second_counter_;
>   RateCounter received_rtcp_bytes_per_second_counter_;
>-  rtc::Optional<int64_t> first_received_rtp_audio_ms_;
>-  rtc::Optional<int64_t> last_received_rtp_audio_ms_;
>-  rtc::Optional<int64_t> first_received_rtp_video_ms_;
>-  rtc::Optional<int64_t> last_received_rtp_video_ms_;
>+  absl::optional<int64_t> first_received_rtp_audio_ms_;
>+  absl::optional<int64_t> last_received_rtp_audio_ms_;
>+  absl::optional<int64_t> first_received_rtp_video_ms_;
>+  absl::optional<int64_t> last_received_rtp_video_ms_;
>   TimeInterval sent_rtp_audio_timer_ms_;
> 
>+  rtc::CriticalSection last_bandwidth_bps_crit_;
>+  uint32_t last_bandwidth_bps_ RTC_GUARDED_BY(&last_bandwidth_bps_crit_);
>   // TODO(holmer): Remove this lock once BitrateController no longer calls
>   // OnNetworkChanged from multiple threads.
>   rtc::CriticalSection bitrate_crit_;
>@@ -354,32 +358,28 @@ class Call : public webrtc::Call,
>       RTC_GUARDED_BY(&bitrate_crit_);
>   AvgCounter pacer_bitrate_kbps_counter_ RTC_GUARDED_BY(&bitrate_crit_);
> 
>-  std::map<std::string, rtc::NetworkRoute> network_routes_;
>-
>-  std::unique_ptr<RtpTransportControllerSendInterface> transport_send_;
>   ReceiveSideCongestionController receive_side_cc_;
>-  const std::unique_ptr<SendDelayStats> video_send_delay_stats_;
>-  const int64_t start_ms_;
>-  // TODO(perkj): |worker_queue_| is supposed to replace
>-  // |module_process_thread_|.
>-  // |worker_queue| is defined last to ensure all pending tasks are cancelled
>-  // and deleted before any other members.
>-  rtc::TaskQueue worker_queue_;
> 
>-  // The config mask set by SetBitrateConfigMask.
>-  // 0 <= min <= start <= max
>-  Config::BitrateConfigMask bitrate_config_mask_;
>+  const std::unique_ptr<ReceiveTimeCalculator> receive_time_calculator_;
> 
>-  // The config set by SetBitrateConfig.
>-  // min >= 0, start != 0, max == -1 || max > 0
>-  Config::BitrateConfig base_bitrate_config_;
>+  const std::unique_ptr<SendDelayStats> video_send_delay_stats_;
>+  const int64_t start_ms_;
> 
>+  // Caches transport_send_.get(), to avoid racing with destructor.
>+  // Note that this is declared before transport_send_ to ensure that it is not
>+  // invalidated until no more tasks can be running on the transport_send_ task
>+  // queue.
>+  RtpTransportControllerSendInterface* transport_send_ptr_;
>+  // Declared last since it will issue callbacks from a task queue. Declaring it
>+  // last ensures that it is destroyed first and any running tasks are finished.
>+  std::unique_ptr<RtpTransportControllerSendInterface> transport_send_;
>   RTC_DISALLOW_COPY_AND_ASSIGN(Call);
> };
> }  // namespace internal
> 
> std::string Call::Stats::ToString(int64_t time_ms) const {
>-  std::stringstream ss;
>+  char buf[1024];
>+  rtc::SimpleStringBuilder ss(buf);
>   ss << "Call stats: " << time_ms << ", {";
>   ss << "send_bw_bps: " << send_bandwidth_bps << ", ";
>   ss << "recv_bw_bps: " << recv_bandwidth_bps << ", ";
>@@ -391,9 +391,10 @@ std::string Call::Stats::ToString(int64_t time_ms) const {
> }
> 
> Call* Call::Create(const Call::Config& config) {
>-  return new internal::Call(config,
>-                            rtc::MakeUnique<RtpTransportControllerSend>(
>-                                Clock::GetRealTimeClock(), config.event_log));
>+  return new internal::Call(
>+      config, absl::make_unique<RtpTransportControllerSend>(
>+                  Clock::GetRealTimeClock(), config.event_log,
>+                  config.network_controller_factory, config.bitrate_config));
> }
> 
> Call* Call::Create(
>@@ -402,6 +403,16 @@ Call* Call::Create(
>   return new internal::Call(config, std::move(transport_send));
> }
> 
>+// This method here to avoid subclasses has to implement this method.
>+// Call perf test will use Internal::Call::CreateVideoSendStream() to inject
>+// FecController.
>+VideoSendStream* Call::CreateVideoSendStream(
>+    VideoSendStream::Config config,
>+    VideoEncoderConfig encoder_config,
>+    std::unique_ptr<FecController> fec_controller) {
>+  return nullptr;
>+}
>+
> namespace internal {
> 
> Call::Call(const Call::Config& config,
>@@ -409,12 +420,12 @@ Call::Call(const Call::Config& config,
>     : clock_(Clock::GetRealTimeClock()),
>       num_cpu_cores_(CpuInfo::DetectNumberOfCores()),
>       module_process_thread_(ProcessThread::Create("ModuleProcessThread")),
>-      pacer_thread_(ProcessThread::Create("PacerThread")),
>-      call_stats_(new CallStats(clock_)),
>+      call_stats_(new CallStats(clock_, module_process_thread_.get())),
>       bitrate_allocator_(new BitrateAllocator(this)),
>       config_(config),
>       audio_network_state_(kNetworkDown),
>       video_network_state_(kNetworkDown),
>+      aggregate_network_up_(false),
>       receive_crit_(RWLockWrapper::CreateRWLock()),
>       send_crit_(RWLockWrapper::CreateRWLock()),
>       event_log_(config.event_log),
>@@ -422,45 +433,27 @@ Call::Call(const Call::Config& config,
>       received_audio_bytes_per_second_counter_(clock_, nullptr, true),
>       received_video_bytes_per_second_counter_(clock_, nullptr, true),
>       received_rtcp_bytes_per_second_counter_(clock_, nullptr, true),
>+      last_bandwidth_bps_(0),
>       min_allocated_send_bitrate_bps_(0),
>       configured_max_padding_bitrate_bps_(0),
>       estimated_send_bitrate_kbps_counter_(clock_, nullptr, true),
>       pacer_bitrate_kbps_counter_(clock_, nullptr, true),
>       receive_side_cc_(clock_, transport_send->packet_router()),
>+      receive_time_calculator_(ReceiveTimeCalculator::CreateFromFieldTrial()),
>       video_send_delay_stats_(new SendDelayStats(clock_)),
>-      start_ms_(clock_->TimeInMilliseconds()),
>-      worker_queue_("call_worker_queue"),
>-      base_bitrate_config_(config.bitrate_config) {
>+      start_ms_(clock_->TimeInMilliseconds()) {
>   RTC_DCHECK(config.event_log != nullptr);
>-  RTC_DCHECK_GE(config.bitrate_config.min_bitrate_bps, 0);
>-  RTC_DCHECK_GE(config.bitrate_config.start_bitrate_bps,
>-                config.bitrate_config.min_bitrate_bps);
>-  if (config.bitrate_config.max_bitrate_bps != -1) {
>-    RTC_DCHECK_GE(config.bitrate_config.max_bitrate_bps,
>-                  config.bitrate_config.start_bitrate_bps);
>-  }
>-  transport_send->send_side_cc()->RegisterNetworkObserver(this);
>+  transport_send->RegisterTargetTransferRateObserver(this);
>   transport_send_ = std::move(transport_send);
>-  transport_send_->send_side_cc()->SignalNetworkState(kNetworkDown);
>-  transport_send_->send_side_cc()->SetBweBitrates(
>-      config_.bitrate_config.min_bitrate_bps,
>-      config_.bitrate_config.start_bitrate_bps,
>-      config_.bitrate_config.max_bitrate_bps);
>+  transport_send_ptr_ = transport_send_.get();
>+
>   call_stats_->RegisterStatsObserver(&receive_side_cc_);
>-  call_stats_->RegisterStatsObserver(transport_send_->send_side_cc());
>+  call_stats_->RegisterStatsObserver(transport_send_->GetCallStatsObserver());
> 
>-  // We have to attach the pacer to the pacer thread before starting the
>-  // module process thread to avoid a race accessing the process thread
>-  // both from the process thread and the pacer thread.
>-  pacer_thread_->RegisterModule(transport_send_->pacer(), RTC_FROM_HERE);
>-  pacer_thread_->RegisterModule(
>+  module_process_thread_->RegisterModule(
>       receive_side_cc_.GetRemoteBitrateEstimator(true), RTC_FROM_HERE);
>-  pacer_thread_->Start();
>-
>   module_process_thread_->RegisterModule(call_stats_.get(), RTC_FROM_HERE);
>   module_process_thread_->RegisterModule(&receive_side_cc_, RTC_FROM_HERE);
>-  module_process_thread_->RegisterModule(transport_send_->send_side_cc(),
>-                                         RTC_FROM_HERE);
>   module_process_thread_->Start();
> }
> 
>@@ -473,23 +466,15 @@ Call::~Call() {
>   RTC_CHECK(audio_receive_streams_.empty());
>   RTC_CHECK(video_receive_streams_.empty());
> 
>-  // The send-side congestion controller must be de-registered prior to
>-  // the pacer thread being stopped to avoid a race when accessing the
>-  // pacer thread object on the module process thread at the same time as
>-  // the pacer thread is stopped.
>-  module_process_thread_->DeRegisterModule(transport_send_->send_side_cc());
>-  pacer_thread_->Stop();
>-  pacer_thread_->DeRegisterModule(transport_send_->pacer());
>-  pacer_thread_->DeRegisterModule(
>+  module_process_thread_->DeRegisterModule(
>       receive_side_cc_.GetRemoteBitrateEstimator(true));
>   module_process_thread_->DeRegisterModule(&receive_side_cc_);
>   module_process_thread_->DeRegisterModule(call_stats_.get());
>   module_process_thread_->Stop();
>   call_stats_->DeregisterStatsObserver(&receive_side_cc_);
>-  call_stats_->DeregisterStatsObserver(transport_send_->send_side_cc());
>+  call_stats_->DeregisterStatsObserver(transport_send_->GetCallStatsObserver());
> 
>-  int64_t first_sent_packet_ms =
>-      transport_send_->send_side_cc()->GetFirstPacketTimeMs();
>+  int64_t first_sent_packet_ms = transport_send_->GetFirstPacketTimeMs();
>   // Only update histograms after process threads have been shut down, so that
>   // they won't try to concurrently update stats.
>   {
>@@ -592,10 +577,10 @@ webrtc::AudioSendStream* Call::CreateAudioSendStream(
>     const webrtc::AudioSendStream::Config& config) {
>   TRACE_EVENT0("webrtc", "Call::CreateAudioSendStream");
>   RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
>-  event_log_->Log(rtc::MakeUnique<RtcEventAudioSendStreamConfig>(
>+  event_log_->Log(absl::make_unique<RtcEventAudioSendStreamConfig>(
>       CreateRtcLogStreamConfig(config)));
> 
>-  rtc::Optional<RtpState> suspended_rtp_state;
>+  absl::optional<RtpState> suspended_rtp_state;
>   {
>     const auto& iter = suspended_audio_send_ssrcs_.find(config.rtp.ssrc);
>     if (iter != suspended_audio_send_ssrcs_.end()) {
>@@ -603,11 +588,14 @@ webrtc::AudioSendStream* Call::CreateAudioSendStream(
>     }
>   }
> 
>+  // TODO(srte): AudioSendStream should call GetWorkerQueue directly rather than
>+  // having it injected.
>+
>   AudioSendStream* send_stream = new AudioSendStream(
>-      config, config_.audio_state, &worker_queue_, module_process_thread_.get(),
>-      transport_send_.get(), bitrate_allocator_.get(), event_log_,
>-      call_stats_->rtcp_rtt_stats(), suspended_rtp_state,
>-      &sent_rtp_audio_timer_ms_);
>+      config, config_.audio_state, transport_send_ptr_->GetWorkerQueue(),
>+      module_process_thread_.get(), transport_send_ptr_,
>+      bitrate_allocator_.get(), event_log_, call_stats_.get(),
>+      suspended_rtp_state, &sent_rtp_audio_timer_ms_);
>   {
>     WriteLockScoped write_lock(*send_crit_);
>     RTC_DCHECK(audio_send_ssrcs_.find(config.rtp.ssrc) ==
>@@ -659,15 +647,15 @@ webrtc::AudioReceiveStream* Call::CreateAudioReceiveStream(
>     const webrtc::AudioReceiveStream::Config& config) {
>   TRACE_EVENT0("webrtc", "Call::CreateAudioReceiveStream");
>   RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
>-  event_log_->Log(rtc::MakeUnique<RtcEventAudioReceiveStreamConfig>(
>+  event_log_->Log(absl::make_unique<RtcEventAudioReceiveStreamConfig>(
>       CreateRtcLogStreamConfig(config)));
>   AudioReceiveStream* receive_stream = new AudioReceiveStream(
>-      &audio_receiver_controller_, transport_send_->packet_router(),
>+      &audio_receiver_controller_, transport_send_ptr_->packet_router(),
>       module_process_thread_.get(), config, config_.audio_state, event_log_);
>   {
>     WriteLockScoped write_lock(*receive_crit_);
>-    receive_rtp_config_[config.rtp.remote_ssrc] =
>-        ReceiveRtpConfig(config.rtp.extensions, UseSendSideBwe(config));
>+    receive_rtp_config_.emplace(config.rtp.remote_ssrc,
>+                                ReceiveRtpConfig(config));
>     audio_receive_streams_.insert(receive_stream);
> 
>     ConfigureSync(config.sync_group);
>@@ -711,16 +699,18 @@ void Call::DestroyAudioReceiveStream(
>   delete audio_receive_stream;
> }
> 
>+// This method can be used for Call tests with external fec controller factory.
> webrtc::VideoSendStream* Call::CreateVideoSendStream(
>     webrtc::VideoSendStream::Config config,
>-    VideoEncoderConfig encoder_config) {
>+    VideoEncoderConfig encoder_config,
>+    std::unique_ptr<FecController> fec_controller) {
>   TRACE_EVENT0("webrtc", "Call::CreateVideoSendStream");
>   RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
> 
>   video_send_delay_stats_->AddSsrcs(config);
>   for (size_t ssrc_index = 0; ssrc_index < config.rtp.ssrcs.size();
>        ++ssrc_index) {
>-    event_log_->Log(rtc::MakeUnique<RtcEventVideoSendStreamConfig>(
>+    event_log_->Log(absl::make_unique<RtcEventVideoSendStreamConfig>(
>         CreateRtcLogStreamConfig(config, ssrc_index)));
>   }
> 
>@@ -728,12 +718,16 @@ webrtc::VideoSendStream* Call::CreateVideoSendStream(
>   // the call has already started.
>   // Copy ssrcs from |config| since |config| is moved.
>   std::vector<uint32_t> ssrcs = config.rtp.ssrcs;
>+
>+  // TODO(srte): VideoSendStream should call GetWorkerQueue directly rather than
>+  // having it injected.
>   VideoSendStream* send_stream = new VideoSendStream(
>-      num_cpu_cores_, module_process_thread_.get(), &worker_queue_,
>-      call_stats_.get(), transport_send_.get(), bitrate_allocator_.get(),
>+      num_cpu_cores_, module_process_thread_.get(),
>+      transport_send_ptr_->GetWorkerQueue(), call_stats_.get(),
>+      transport_send_ptr_, bitrate_allocator_.get(),
>       video_send_delay_stats_.get(), event_log_, std::move(config),
>       std::move(encoder_config), suspended_video_send_ssrcs_,
>-      suspended_video_payload_states_);
>+      suspended_video_payload_states_, std::move(fec_controller));
> 
>   {
>     WriteLockScoped write_lock(*send_crit_);
>@@ -743,12 +737,25 @@ webrtc::VideoSendStream* Call::CreateVideoSendStream(
>     }
>     video_send_streams_.insert(send_stream);
>   }
>-  send_stream->SignalNetworkState(video_network_state_);
>   UpdateAggregateNetworkState();
> 
>   return send_stream;
> }
> 
>+webrtc::VideoSendStream* Call::CreateVideoSendStream(
>+    webrtc::VideoSendStream::Config config,
>+    VideoEncoderConfig encoder_config) {
>+  if (config_.fec_controller_factory) {
>+    RTC_LOG(LS_INFO) << "External FEC Controller will be used.";
>+  }
>+  std::unique_ptr<FecController> fec_controller =
>+      config_.fec_controller_factory
>+          ? config_.fec_controller_factory->CreateFecController()
>+          : absl::make_unique<FecControllerDefault>(Clock::GetRealTimeClock());
>+  return CreateVideoSendStream(std::move(config), std::move(encoder_config),
>+                               std::move(fec_controller));
>+}
>+
> void Call::DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) {
>   TRACE_EVENT0("webrtc", "Call::DestroyVideoSendStream");
>   RTC_DCHECK(send_stream != nullptr);
>@@ -794,12 +801,10 @@ webrtc::VideoReceiveStream* Call::CreateVideoReceiveStream(
> 
>   VideoReceiveStream* receive_stream = new VideoReceiveStream(
>       &video_receiver_controller_, num_cpu_cores_,
>-      transport_send_->packet_router(), std::move(configuration),
>+      transport_send_ptr_->packet_router(), std::move(configuration),
>       module_process_thread_.get(), call_stats_.get());
> 
>   const webrtc::VideoReceiveStream::Config& config = receive_stream->config();
>-  ReceiveRtpConfig receive_config(config.rtp.extensions,
>-                                  UseSendSideBwe(config));
>   {
>     WriteLockScoped write_lock(*receive_crit_);
>     if (config.rtp.rtx_ssrc) {
>@@ -807,15 +812,17 @@ webrtc::VideoReceiveStream* Call::CreateVideoReceiveStream(
>       // stream. Since the transport_send_cc negotiation is per payload
>       // type, we may get an incorrect value for the rtx stream, but
>       // that is unlikely to matter in practice.
>-      receive_rtp_config_[config.rtp.rtx_ssrc] = receive_config;
>+      receive_rtp_config_.emplace(config.rtp.rtx_ssrc,
>+                                  ReceiveRtpConfig(config));
>     }
>-    receive_rtp_config_[config.rtp.remote_ssrc] = receive_config;
>+    receive_rtp_config_.emplace(config.rtp.remote_ssrc,
>+                                ReceiveRtpConfig(config));
>     video_receive_streams_.insert(receive_stream);
>     ConfigureSync(config.sync_group);
>   }
>   receive_stream->SignalNetworkState(video_network_state_);
>   UpdateAggregateNetworkState();
>-  event_log_->Log(rtc::MakeUnique<RtcEventVideoReceiveStreamConfig>(
>+  event_log_->Log(absl::make_unique<RtcEventVideoReceiveStreamConfig>(
>       CreateRtcLogStreamConfig(config)));
>   return receive_stream;
> }
>@@ -868,12 +875,11 @@ FlexfecReceiveStream* Call::CreateFlexfecReceiveStream(
>     // this locked scope.
>     receive_stream = new FlexfecReceiveStreamImpl(
>         &video_receiver_controller_, config, recovered_packet_receiver,
>-        call_stats_->rtcp_rtt_stats(), module_process_thread_.get());
>+        call_stats_.get(), module_process_thread_.get());
> 
>     RTC_DCHECK(receive_rtp_config_.find(config.remote_ssrc) ==
>                receive_rtp_config_.end());
>-    receive_rtp_config_[config.remote_ssrc] =
>-        ReceiveRtpConfig(config.rtp_header_extensions, UseSendSideBwe(config));
>+    receive_rtp_config_.emplace(config.remote_ssrc, ReceiveRtpConfig(config));
>   }
> 
>   // TODO(brandtr): Store config in RtcEventLog here.
>@@ -902,23 +908,36 @@ void Call::DestroyFlexfecReceiveStream(FlexfecReceiveStream* receive_stream) {
>   delete receive_stream;
> }
> 
>+RtpTransportControllerSendInterface* Call::GetTransportControllerSend() {
>+  return transport_send_ptr_;
>+}
>+
> Call::Stats Call::GetStats() const {
>   // TODO(solenberg): Some test cases in EndToEndTest use this from a different
>   // thread. Re-enable once that is fixed.
>   // RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
>   Stats stats;
>   // Fetch available send/receive bitrates.
>-  uint32_t send_bandwidth = 0;
>-  transport_send_->send_side_cc()->AvailableBandwidth(&send_bandwidth);
>   std::vector<unsigned int> ssrcs;
>   uint32_t recv_bandwidth = 0;
>   receive_side_cc_.GetRemoteBitrateEstimator(false)->LatestEstimate(
>       &ssrcs, &recv_bandwidth);
>-  stats.send_bandwidth_bps = send_bandwidth;
>+
>+  {
>+    rtc::CritScope cs(&last_bandwidth_bps_crit_);
>+    stats.send_bandwidth_bps = last_bandwidth_bps_;
>+  }
>   stats.recv_bandwidth_bps = recv_bandwidth;
>-  stats.pacer_delay_ms =
>-      transport_send_->send_side_cc()->GetPacerQueuingDelayMs();
>-  stats.rtt_ms = call_stats_->rtcp_rtt_stats()->LastProcessedRtt();
>+  // TODO(srte): It is unclear if we only want to report queues if network is
>+  // available.
>+  {
>+    rtc::CritScope cs(&aggregate_network_up_crit_);
>+    stats.pacer_delay_ms = aggregate_network_up_
>+                               ? transport_send_ptr_->GetPacerQueuingDelayMs()
>+                               : 0;
>+  }
>+
>+  stats.rtt_ms = call_stats_->LastProcessedRtt();
>   {
>     rtc::CritScope cs(&bitrate_crit_);
>     stats.max_padding_bitrate_bps = configured_max_padding_bitrate_bps_;
>@@ -926,103 +945,22 @@ Call::Stats Call::GetStats() const {
>   return stats;
> }
> 
>-void Call::SetBitrateConfig(
>-    const webrtc::Call::Config::BitrateConfig& bitrate_config) {
>-  TRACE_EVENT0("webrtc", "Call::SetBitrateConfig");
>-  RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
>-  RTC_DCHECK_GE(bitrate_config.min_bitrate_bps, 0);
>-  RTC_DCHECK_NE(bitrate_config.start_bitrate_bps, 0);
>-  if (bitrate_config.max_bitrate_bps != -1) {
>-    RTC_DCHECK_GT(bitrate_config.max_bitrate_bps, 0);
>-  }
>-
>-  rtc::Optional<int> new_start;
>-  // Only update the "start" bitrate if it's set, and different from the old
>-  // value. In practice, this value comes from the x-google-start-bitrate codec
>-  // parameter in SDP, and setting the same remote description twice shouldn't
>-  // restart bandwidth estimation.
>-  if (bitrate_config.start_bitrate_bps != -1 &&
>-      bitrate_config.start_bitrate_bps !=
>-          base_bitrate_config_.start_bitrate_bps) {
>-    new_start.emplace(bitrate_config.start_bitrate_bps);
>-  }
>-  base_bitrate_config_ = bitrate_config;
>-  UpdateCurrentBitrateConfig(new_start);
>-}
>-
>-void Call::SetBitrateConfigMask(
>-    const webrtc::Call::Config::BitrateConfigMask& mask) {
>-  TRACE_EVENT0("webrtc", "Call::SetBitrateConfigMask");
>-  RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
>-
>-  bitrate_config_mask_ = mask;
>-  UpdateCurrentBitrateConfig(mask.start_bitrate_bps);
>-}
>-
>-void Call::UpdateCurrentBitrateConfig(const rtc::Optional<int>& new_start) {
>-  Config::BitrateConfig updated;
>-  updated.min_bitrate_bps =
>-      std::max(bitrate_config_mask_.min_bitrate_bps.value_or(0),
>-               base_bitrate_config_.min_bitrate_bps);
>-
>-  updated.max_bitrate_bps =
>-      MinPositive(bitrate_config_mask_.max_bitrate_bps.value_or(-1),
>-                  base_bitrate_config_.max_bitrate_bps);
>-
>-  // If the combined min ends up greater than the combined max, the max takes
>-  // priority.
>-  if (updated.max_bitrate_bps != -1 &&
>-      updated.min_bitrate_bps > updated.max_bitrate_bps) {
>-    updated.min_bitrate_bps = updated.max_bitrate_bps;
>-  }
>-
>-  // If there is nothing to update (min/max unchanged, no new bandwidth
>-  // estimation start value), return early.
>-  if (updated.min_bitrate_bps == config_.bitrate_config.min_bitrate_bps &&
>-      updated.max_bitrate_bps == config_.bitrate_config.max_bitrate_bps &&
>-      !new_start) {
>-    RTC_LOG(LS_VERBOSE) << "WebRTC.Call.UpdateCurrentBitrateConfig: "
>-                        << "nothing to update";
>-    return;
>-  }
>-
>-  if (new_start) {
>-    // Clamp start by min and max.
>-    updated.start_bitrate_bps = MinPositive(
>-        std::max(*new_start, updated.min_bitrate_bps), updated.max_bitrate_bps);
>-  } else {
>-    updated.start_bitrate_bps = -1;
>-  }
>-
>-  RTC_LOG(INFO) << "WebRTC.Call.UpdateCurrentBitrateConfig: "
>-                << "calling SetBweBitrates with args ("
>-                << updated.min_bitrate_bps << ", " << updated.start_bitrate_bps
>-                << ", " << updated.max_bitrate_bps << ")";
>-  transport_send_->send_side_cc()->SetBweBitrates(updated.min_bitrate_bps,
>-                                                  updated.start_bitrate_bps,
>-                                                  updated.max_bitrate_bps);
>-  if (!new_start) {
>-    updated.start_bitrate_bps = config_.bitrate_config.start_bitrate_bps;
>-  }
>-  config_.bitrate_config = updated;
>-}
>-
> void Call::SetBitrateAllocationStrategy(
>     std::unique_ptr<rtc::BitrateAllocationStrategy>
>         bitrate_allocation_strategy) {
>-  if (!worker_queue_.IsCurrent()) {
>-    rtc::BitrateAllocationStrategy* strategy_raw =
>-        bitrate_allocation_strategy.release();
>-    auto functor = [this, strategy_raw]() {
>-      SetBitrateAllocationStrategy(
>-          rtc::WrapUnique<rtc::BitrateAllocationStrategy>(strategy_raw));
>-    };
>-    worker_queue_.PostTask([functor] { functor(); });
>-    return;
>-  }
>-  RTC_DCHECK_RUN_ON(&worker_queue_);
>-  bitrate_allocator_->SetBitrateAllocationStrategy(
>-      std::move(bitrate_allocation_strategy));
>+  // TODO(srte): This function should be moved to RtpTransportControllerSend
>+  // when BitrateAllocator is moved there.
>+  struct Functor {
>+    void operator()() {
>+      bitrate_allocator_->SetBitrateAllocationStrategy(
>+          std::move(bitrate_allocation_strategy_));
>+    }
>+    BitrateAllocator* bitrate_allocator_;
>+    std::unique_ptr<rtc::BitrateAllocationStrategy>
>+        bitrate_allocation_strategy_;
>+  };
>+  transport_send_ptr_->GetWorkerQueue()->PostTask(Functor{
>+      bitrate_allocator_.get(), std::move(bitrate_allocation_strategy)});
> }
> 
> void Call::SignalChannelNetworkState(MediaType media, NetworkState state) {
>@@ -1046,9 +984,6 @@ void Call::SignalChannelNetworkState(MediaType media, NetworkState state) {
>     for (auto& kv : audio_send_ssrcs_) {
>       kv.second->SignalNetworkState(audio_network_state_);
>     }
>-    for (auto& kv : video_send_ssrcs_) {
>-      kv.second->SignalNetworkState(video_network_state_);
>-    }
>   }
>   {
>     ReadLockScoped read_lock(*receive_crit_);
>@@ -1085,45 +1020,6 @@ void Call::OnTransportOverheadChanged(MediaType media,
>   }
> }
> 
>-// TODO(honghaiz): Add tests for this method.
>-void Call::OnNetworkRouteChanged(const std::string& transport_name,
>-                                 const rtc::NetworkRoute& network_route) {
>-  RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
>-  // Check if the network route is connected.
>-  if (!network_route.connected) {
>-    RTC_LOG(LS_INFO) << "Transport " << transport_name << " is disconnected";
>-    // TODO(honghaiz): Perhaps handle this in SignalChannelNetworkState and
>-    // consider merging these two methods.
>-    return;
>-  }
>-
>-  // Check whether the network route has changed on each transport.
>-  auto result =
>-      network_routes_.insert(std::make_pair(transport_name, network_route));
>-  auto kv = result.first;
>-  bool inserted = result.second;
>-  if (inserted) {
>-    // No need to reset BWE if this is the first time the network connects.
>-    return;
>-  }
>-  if (kv->second != network_route) {
>-    kv->second = network_route;
>-    RTC_LOG(LS_INFO)
>-        << "Network route changed on transport " << transport_name
>-        << ": new local network id " << network_route.local_network_id
>-        << " new remote network id " << network_route.remote_network_id
>-        << " Reset bitrates to min: " << config_.bitrate_config.min_bitrate_bps
>-        << " bps, start: " << config_.bitrate_config.start_bitrate_bps
>-        << " bps,  max: " << config_.bitrate_config.start_bitrate_bps
>-        << " bps.";
>-    RTC_DCHECK_GT(config_.bitrate_config.start_bitrate_bps, 0);
>-    transport_send_->send_side_cc()->OnNetworkRouteChanged(
>-        network_route, config_.bitrate_config.start_bitrate_bps,
>-        config_.bitrate_config.min_bitrate_bps,
>-        config_.bitrate_config.max_bitrate_bps);
>-  }
>-}
>-
> void Call::UpdateAggregateNetworkState() {
>   RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
> 
>@@ -1144,39 +1040,37 @@ void Call::UpdateAggregateNetworkState() {
>       have_video = true;
>   }
> 
>-  NetworkState aggregate_state = kNetworkDown;
>-  if ((have_video && video_network_state_ == kNetworkUp) ||
>-      (have_audio && audio_network_state_ == kNetworkUp)) {
>-    aggregate_state = kNetworkUp;
>-  }
>+  bool aggregate_network_up =
>+      ((have_video && video_network_state_ == kNetworkUp) ||
>+       (have_audio && audio_network_state_ == kNetworkUp));
> 
>   RTC_LOG(LS_INFO) << "UpdateAggregateNetworkState: aggregate_state="
>-                   << (aggregate_state == kNetworkUp ? "up" : "down");
>-
>-  transport_send_->send_side_cc()->SignalNetworkState(aggregate_state);
>+                   << (aggregate_network_up ? "up" : "down");
>+  {
>+    rtc::CritScope cs(&aggregate_network_up_crit_);
>+    aggregate_network_up_ = aggregate_network_up;
>+  }
>+  transport_send_ptr_->OnNetworkAvailability(aggregate_network_up);
> }
> 
> void Call::OnSentPacket(const rtc::SentPacket& sent_packet) {
>   video_send_delay_stats_->OnSentPacket(sent_packet.packet_id,
>                                         clock_->TimeInMilliseconds());
>-  transport_send_->send_side_cc()->OnSentPacket(sent_packet);
>+  transport_send_ptr_->OnSentPacket(sent_packet);
> }
> 
>-void Call::OnNetworkChanged(uint32_t target_bitrate_bps,
>-                            uint8_t fraction_loss,
>-                            int64_t rtt_ms,
>-                            int64_t probing_interval_ms) {
>-  // TODO(perkj): Consider making sure CongestionController operates on
>-  // |worker_queue_|.
>-  if (!worker_queue_.IsCurrent()) {
>-    worker_queue_.PostTask(
>-        [this, target_bitrate_bps, fraction_loss, rtt_ms, probing_interval_ms] {
>-          OnNetworkChanged(target_bitrate_bps, fraction_loss, rtt_ms,
>-                           probing_interval_ms);
>-        });
>-    return;
>+void Call::OnTargetTransferRate(TargetTransferRate msg) {
>+  uint32_t target_bitrate_bps = msg.target_rate.bps();
>+  int loss_ratio_255 = msg.network_estimate.loss_rate_ratio * 255;
>+  uint8_t fraction_loss =
>+      rtc::dchecked_cast<uint8_t>(rtc::SafeClamp(loss_ratio_255, 0, 255));
>+  int64_t rtt_ms = msg.network_estimate.round_trip_time.ms();
>+  int64_t probing_interval_ms = msg.network_estimate.bwe_period.ms();
>+  uint32_t bandwidth_bps = msg.network_estimate.bandwidth.bps();
>+  {
>+    rtc::CritScope cs(&last_bandwidth_bps_crit_);
>+    last_bandwidth_bps_ = bandwidth_bps;
>   }
>-  RTC_DCHECK_RUN_ON(&worker_queue_);
>   // For controlling the rate of feedback messages.
>   receive_side_cc_.OnBitrateChanged(target_bitrate_bps);
>   bitrate_allocator_->OnNetworkChanged(target_bitrate_bps, fraction_loss,
>@@ -1211,9 +1105,12 @@ void Call::OnNetworkChanged(uint32_t target_bitrate_bps,
> }
> 
> void Call::OnAllocationLimitsChanged(uint32_t min_send_bitrate_bps,
>-                                     uint32_t max_padding_bitrate_bps) {
>-  transport_send_->SetAllocatedSendBitrateLimits(min_send_bitrate_bps,
>-                                                 max_padding_bitrate_bps);
>+                                     uint32_t max_padding_bitrate_bps,
>+                                     uint32_t total_bitrate_bps,
>+                                     bool has_packet_feedback) {
>+  transport_send_ptr_->SetAllocatedSendBitrateLimits(
>+      min_send_bitrate_bps, max_padding_bitrate_bps, total_bitrate_bps);
>+  transport_send_ptr_->SetPerPacketFeedbackAvailable(has_packet_feedback);
>   rtc::CritScope lock(&bitrate_crit_);
>   min_allocated_send_bitrate_bps_ = min_send_bitrate_bps;
>   configured_max_padding_bitrate_bps_ = max_padding_bitrate_bps;
>@@ -1312,7 +1209,7 @@ PacketReceiver::DeliveryStatus Call::DeliverRtcp(MediaType media_type,
>   }
> 
>   if (rtcp_delivered) {
>-    event_log_->Log(rtc::MakeUnique<RtcEventRtcpPacketIncoming>(
>+    event_log_->Log(absl::make_unique<RtcEventRtcpPacketIncoming>(
>         rtc::MakeArrayView(packet, length)));
>   }
> 
>@@ -1321,15 +1218,19 @@ PacketReceiver::DeliveryStatus Call::DeliverRtcp(MediaType media_type,
> 
> PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
>                                                 rtc::CopyOnWriteBuffer packet,
>-                                                const PacketTime& packet_time) {
>+                                                int64_t packet_time_us) {
>   TRACE_EVENT0("webrtc", "Call::DeliverRtp");
> 
>   RtpPacketReceived parsed_packet;
>   if (!parsed_packet.Parse(std::move(packet)))
>     return DELIVERY_PACKET_ERROR;
> 
>-  if (packet_time.timestamp != -1) {
>-    parsed_packet.set_arrival_time_ms((packet_time.timestamp + 500) / 1000);
>+  if (packet_time_us != -1) {
>+    if (receive_time_calculator_) {
>+      packet_time_us = receive_time_calculator_->ReconcileReceiveTimes(
>+          packet_time_us, clock_->TimeInMicroseconds());
>+    }
>+    parsed_packet.set_arrival_time_ms((packet_time_us + 500) / 1000);
>   } else {
>     parsed_packet.set_arrival_time_ms(clock_->TimeInMilliseconds());
>   }
>@@ -1367,7 +1268,7 @@ PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
>       received_bytes_per_second_counter_.Add(length);
>       received_audio_bytes_per_second_counter_.Add(length);
>       event_log_->Log(
>-          rtc::MakeUnique<RtcEventRtpPacketIncoming>(parsed_packet));
>+          absl::make_unique<RtcEventRtpPacketIncoming>(parsed_packet));
>       const int64_t arrival_time_ms = parsed_packet.arrival_time_ms();
>       if (!first_received_rtp_audio_ms_) {
>         first_received_rtp_audio_ms_.emplace(arrival_time_ms);
>@@ -1376,11 +1277,12 @@ PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
>       return DELIVERY_OK;
>     }
>   } else if (media_type == MediaType::VIDEO) {
>+    parsed_packet.set_payload_type_frequency(kVideoPayloadTypeFrequency);
>     if (video_receiver_controller_.OnRtpPacket(parsed_packet)) {
>       received_bytes_per_second_counter_.Add(length);
>       received_video_bytes_per_second_counter_.Add(length);
>       event_log_->Log(
>-          rtc::MakeUnique<RtcEventRtpPacketIncoming>(parsed_packet));
>+          absl::make_unique<RtcEventRtpPacketIncoming>(parsed_packet));
>       const int64_t arrival_time_ms = parsed_packet.arrival_time_ms();
>       if (!first_received_rtp_video_ms_) {
>         first_received_rtp_video_ms_.emplace(arrival_time_ms);
>@@ -1395,12 +1297,12 @@ PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
> PacketReceiver::DeliveryStatus Call::DeliverPacket(
>     MediaType media_type,
>     rtc::CopyOnWriteBuffer packet,
>-    const PacketTime& packet_time) {
>+    int64_t packet_time_us) {
>   RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
>   if (RtpHeaderParser::IsRtcp(packet.cdata(), packet.size()))
>     return DeliverRtcp(media_type, packet.cdata(), packet.size());
> 
>-  return DeliverRtp(media_type, std::move(packet), packet_time);
>+  return DeliverRtp(media_type, std::move(packet), packet_time_us);
> }
> 
> void Call::OnRecoveredPacket(const uint8_t* packet, size_t length) {
>@@ -1420,12 +1322,13 @@ void Call::OnRecoveredPacket(const uint8_t* packet, size_t length) {
>     // deregistering in the |receive_rtp_config_| map is protected by that lock.
>     // So by not passing the packet on to demuxing in this case, we prevent
>     // incoming packets to be passed on via the demuxer to a receive stream
>-    // which is being torned down.
>+    // which is being torn down.
>     return;
>   }
>   parsed_packet.IdentifyExtensions(it->second.extensions);
> 
>   // TODO(brandtr): Update here when we support protecting audio packets too.
>+  parsed_packet.set_payload_type_frequency(kVideoPayloadTypeFrequency);
>   video_receiver_controller_.OnRtpPacket(parsed_packet);
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/call.h b/Source/ThirdParty/libwebrtc/Source/webrtc/call/call.h
>index b6e0aeabbfb..d8b1b5777db 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/call.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/call.h
>@@ -15,11 +15,12 @@
> #include <string>
> #include <vector>
> 
>-#include "api/rtcerror.h"
>+#include "api/mediatypes.h"
> #include "call/audio_receive_stream.h"
> #include "call/audio_send_stream.h"
>-#include "call/audio_state.h"
>+#include "call/call_config.h"
> #include "call/flexfec_receive_stream.h"
>+#include "call/packet_receiver.h"
> #include "call/rtp_transport_controller_send_interface.h"
> #include "call/video_receive_stream.h"
> #include "call/video_send_stream.h"
>@@ -27,93 +28,16 @@
> #include "rtc_base/bitrateallocationstrategy.h"
> #include "rtc_base/copyonwritebuffer.h"
> #include "rtc_base/networkroute.h"
>-#include "rtc_base/platform_file.h"
> #include "rtc_base/socket.h"
> 
> namespace webrtc {
> 
>-class AudioProcessing;
>-class RtcEventLog;
>-
>-enum class MediaType {
>-  ANY,
>-  AUDIO,
>-  VIDEO,
>-  DATA
>-};
>-
>-// Like std::min, but considers non-positive values to be unset.
>-// TODO(zstein): Remove once all callers use rtc::Optional.
>-template <typename T>
>-static T MinPositive(T a, T b) {
>-  if (a <= 0) {
>-    return b;
>-  }
>-  if (b <= 0) {
>-    return a;
>-  }
>-  return std::min(a, b);
>-}
>-
>-class PacketReceiver {
>- public:
>-  enum DeliveryStatus {
>-    DELIVERY_OK,
>-    DELIVERY_UNKNOWN_SSRC,
>-    DELIVERY_PACKET_ERROR,
>-  };
>-
>-  virtual DeliveryStatus DeliverPacket(MediaType media_type,
>-                                       rtc::CopyOnWriteBuffer packet,
>-                                       const PacketTime& packet_time) = 0;
>-
>- protected:
>-  virtual ~PacketReceiver() {}
>-};
>-
> // A Call instance can contain several send and/or receive streams. All streams
> // are assumed to have the same remote endpoint and will share bitrate estimates
> // etc.
> class Call {
>  public:
>-  struct Config {
>-    explicit Config(RtcEventLog* event_log) : event_log(event_log) {
>-      RTC_DCHECK(event_log);
>-    }
>-
>-    static constexpr int kDefaultStartBitrateBps = 300000;
>-
>-    // Bitrate config used until valid bitrate estimates are calculated. Also
>-    // used to cap total bitrate used. This comes from the remote connection.
>-    struct BitrateConfig {
>-      int min_bitrate_bps = 0;
>-      int start_bitrate_bps = kDefaultStartBitrateBps;
>-      int max_bitrate_bps = -1;
>-    } bitrate_config;
>-
>-    // The local client's bitrate preferences. The actual configuration used
>-    // is a combination of this and |bitrate_config|. The combination is
>-    // currently more complicated than a simple mask operation (see
>-    // SetBitrateConfig and SetBitrateConfigMask). Assumes that 0 <= min <=
>-    // start <= max holds for set parameters.
>-    struct BitrateConfigMask {
>-      rtc::Optional<int> min_bitrate_bps;
>-      rtc::Optional<int> start_bitrate_bps;
>-      rtc::Optional<int> max_bitrate_bps;
>-    };
>-
>-    // AudioState which is possibly shared between multiple calls.
>-    // TODO(solenberg): Change this to a shared_ptr once we can use C++11.
>-    rtc::scoped_refptr<AudioState> audio_state;
>-
>-    // Audio Processing Module to be used in this call.
>-    // TODO(solenberg): Change this to a shared_ptr once we can use C++11.
>-    AudioProcessing* audio_processing = nullptr;
>-
>-    // RtcEventLog to use for this call. Required.
>-    // Use webrtc::RtcEventLog::CreateNull() for a null implementation.
>-    RtcEventLog* event_log = nullptr;
>-  };
>+  using Config = CallConfig;
> 
>   struct Stats {
>     std::string ToString(int64_t time_ms) const;
>@@ -144,6 +68,10 @@ class Call {
>   virtual VideoSendStream* CreateVideoSendStream(
>       VideoSendStream::Config config,
>       VideoEncoderConfig encoder_config) = 0;
>+  virtual VideoSendStream* CreateVideoSendStream(
>+      VideoSendStream::Config config,
>+      VideoEncoderConfig encoder_config,
>+      std::unique_ptr<FecController> fec_controller);
>   virtual void DestroyVideoSendStream(VideoSendStream* send_stream) = 0;
> 
>   virtual VideoReceiveStream* CreateVideoReceiveStream(
>@@ -164,26 +92,17 @@ class Call {
>   // Call instance exists.
>   virtual PacketReceiver* Receiver() = 0;
> 
>+  // This is used to access the transport controller send instance owned by
>+  // Call. The send transport controller is currently owned by Call for legacy
>+  // reasons. (for instance  variants of call tests are built on this assumtion)
>+  // TODO(srte): Move ownership of transport controller send out of Call and
>+  // remove this method interface.
>+  virtual RtpTransportControllerSendInterface* GetTransportControllerSend() = 0;
>+
>   // Returns the call statistics, such as estimated send and receive bandwidth,
>   // pacing delay, etc.
>   virtual Stats GetStats() const = 0;
> 
>-  // The greater min and smaller max set by this and SetBitrateConfigMask will
>-  // be used. The latest non-negative start value from either call will be used.
>-  // Specifying a start bitrate (>0) will reset the current bitrate estimate.
>-  // This is due to how the 'x-google-start-bitrate' flag is currently
>-  // implemented. Passing -1 leaves the start bitrate unchanged. Behavior is not
>-  // guaranteed for other negative values or 0.
>-  virtual void SetBitrateConfig(
>-      const Config::BitrateConfig& bitrate_config) = 0;
>-
>-  // The greater min and smaller max set by this and SetBitrateConfig will be
>-  // used. The latest non-negative start value form either call will be used.
>-  // Specifying a start bitrate will reset the current bitrate estimate.
>-  // Assumes 0 <= min <= start <= max holds for set parameters.
>-  virtual void SetBitrateConfigMask(
>-      const Config::BitrateConfigMask& bitrate_mask) = 0;
>-
>   virtual void SetBitrateAllocationStrategy(
>       std::unique_ptr<rtc::BitrateAllocationStrategy>
>           bitrate_allocation_strategy) = 0;
>@@ -198,10 +117,6 @@ class Call {
>       MediaType media,
>       int transport_overhead_per_packet) = 0;
> 
>-  virtual void OnNetworkRouteChanged(
>-      const std::string& transport_name,
>-      const rtc::NetworkRoute& network_route) = 0;
>-
>   virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0;
> 
>   virtual ~Call() {}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/call_config.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/call_config.cc
>new file mode 100644
>index 00000000000..ca5fb60b341
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/call_config.cc
>@@ -0,0 +1,20 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "call/call_config.h"
>+
>+namespace webrtc {
>+
>+CallConfig::CallConfig(RtcEventLog* event_log) : event_log(event_log) {
>+  RTC_DCHECK(event_log);
>+}
>+CallConfig::~CallConfig() = default;
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/call_config.h b/Source/ThirdParty/libwebrtc/Source/webrtc/call/call_config.h
>new file mode 100644
>index 00000000000..438929f4d38
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/call_config.h
>@@ -0,0 +1,56 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#ifndef CALL_CALL_CONFIG_H_
>+#define CALL_CALL_CONFIG_H_
>+
>+#include "api/bitrate_constraints.h"
>+#include "api/fec_controller.h"
>+#include "api/rtcerror.h"
>+#include "api/transport/network_control.h"
>+#include "call/audio_state.h"
>+#include "rtc_base/platform_file.h"
>+
>+namespace webrtc {
>+
>+class AudioProcessing;
>+class RtcEventLog;
>+
>+struct CallConfig {
>+  explicit CallConfig(RtcEventLog* event_log);
>+  ~CallConfig();
>+
>+  RTC_DEPRECATED static constexpr int kDefaultStartBitrateBps = 300000;
>+
>+  // Bitrate config used until valid bitrate estimates are calculated. Also
>+  // used to cap total bitrate used. This comes from the remote connection.
>+  BitrateConstraints bitrate_config;
>+
>+  // AudioState which is possibly shared between multiple calls.
>+  // TODO(solenberg): Change this to a shared_ptr once we can use C++11.
>+  rtc::scoped_refptr<AudioState> audio_state;
>+
>+  // Audio Processing Module to be used in this call.
>+  // TODO(solenberg): Change this to a shared_ptr once we can use C++11.
>+  AudioProcessing* audio_processing = nullptr;
>+
>+  // RtcEventLog to use for this call. Required.
>+  // Use webrtc::RtcEventLog::CreateNull() for a null implementation.
>+  RtcEventLog* event_log = nullptr;
>+
>+  // FecController to use for this call.
>+  FecControllerFactoryInterface* fec_controller_factory = nullptr;
>+
>+  // Network controller factory to use for this call.
>+  NetworkControllerFactoryInterface* network_controller_factory = nullptr;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // CALL_CALL_CONFIG_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/call_perf_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/call_perf_tests.cc
>index e9747a4fe6c..54e01b12324 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/call_perf_tests.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/call_perf_tests.cc
>@@ -13,23 +13,25 @@
> #include <memory>
> #include <string>
> 
>+#include "absl/memory/memory.h"
> #include "api/audio_codecs/builtin_audio_encoder_factory.h"
>+#include "api/video/video_bitrate_allocation.h"
>+#include "api/video_codecs/video_encoder_config.h"
> #include "call/call.h"
>-#include "call/video_config.h"
> #include "logging/rtc_event_log/rtc_event_log.h"
> #include "modules/audio_coding/include/audio_coding_module.h"
>+#include "modules/audio_device/include/test_audio_device.h"
> #include "modules/audio_mixer/audio_mixer_impl.h"
> #include "modules/rtp_rtcp/include/rtp_header_parser.h"
> #include "rtc_base/bitrateallocationstrategy.h"
> #include "rtc_base/checks.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/thread_annotations.h"
> #include "system_wrappers/include/metrics_default.h"
> #include "test/call_test.h"
> #include "test/direct_transport.h"
> #include "test/drifting_clock.h"
>+#include "test/encoder_proxy_factory.h"
> #include "test/encoder_settings.h"
>-#include "test/fake_audio_device.h"
> #include "test/fake_encoder.h"
> #include "test/field_trial.h"
> #include "test/frame_generator.h"
>@@ -42,18 +44,13 @@
> #include "video/transport_adapter.h"
> 
> using webrtc::test::DriftingClock;
>-using webrtc::test::FakeAudioDevice;
> 
> namespace webrtc {
> 
> class CallPerfTest : public test::CallTest {
>  protected:
>-  enum class FecMode {
>-    kOn, kOff
>-  };
>-  enum class CreateOrder {
>-    kAudioFirst, kVideoFirst
>-  };
>+  enum class FecMode { kOn, kOff };
>+  enum class CreateOrder { kAudioFirst, kVideoFirst };
>   void TestAudioVideoSync(FecMode fec,
>                           CreateOrder create_first,
>                           float video_ntp_speed,
>@@ -170,10 +167,11 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec,
> 
>   task_queue_.SendTask([&]() {
>     metrics::Reset();
>-    rtc::scoped_refptr<FakeAudioDevice> fake_audio_device =
>-        new rtc::RefCountedObject<FakeAudioDevice>(
>-            FakeAudioDevice::CreatePulsedNoiseCapturer(256, 48000),
>-            FakeAudioDevice::CreateDiscardRenderer(48000), audio_rtp_speed);
>+    rtc::scoped_refptr<TestAudioDeviceModule> fake_audio_device =
>+        TestAudioDeviceModule::CreateTestAudioDeviceModule(
>+            TestAudioDeviceModule::CreatePulsedNoiseCapturer(256, 48000),
>+            TestAudioDeviceModule::CreateDiscardRenderer(48000),
>+            audio_rtp_speed);
>     EXPECT_EQ(0, fake_audio_device->Init());
> 
>     AudioState::Config send_audio_state_config;
>@@ -181,12 +179,12 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec,
>     send_audio_state_config.audio_processing =
>         AudioProcessingBuilder().Create();
>     send_audio_state_config.audio_device_module = fake_audio_device;
>-    Call::Config sender_config(event_log_.get());
>+    Call::Config sender_config(send_event_log_.get());
> 
>     auto audio_state = AudioState::Create(send_audio_state_config);
>     fake_audio_device->RegisterAudioCallback(audio_state->audio_transport());
>     sender_config.audio_state = audio_state;
>-    Call::Config receiver_config(event_log_.get());
>+    Call::Config receiver_config(recv_event_log_.get());
>     receiver_config.audio_state = audio_state;
>     CreateCalls(sender_config, receiver_config);
> 
>@@ -201,18 +199,18 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec,
>                    return pair.second == MediaType::VIDEO;
>                  });
> 
>-    audio_send_transport = rtc::MakeUnique<test::PacketTransport>(
>+    audio_send_transport = absl::make_unique<test::PacketTransport>(
>         &task_queue_, sender_call_.get(), &observer,
>         test::PacketTransport::kSender, audio_pt_map, audio_net_config);
>     audio_send_transport->SetReceiver(receiver_call_->Receiver());
> 
>-    video_send_transport = rtc::MakeUnique<test::PacketTransport>(
>+    video_send_transport = absl::make_unique<test::PacketTransport>(
>         &task_queue_, sender_call_.get(), &observer,
>         test::PacketTransport::kSender, video_pt_map,
>         FakeNetworkPipe::Config());
>     video_send_transport->SetReceiver(receiver_call_->Receiver());
> 
>-    receive_transport = rtc::MakeUnique<test::PacketTransport>(
>+    receive_transport = absl::make_unique<test::PacketTransport>(
>         &task_queue_, receiver_call_.get(), &observer,
>         test::PacketTransport::kReceiver, payload_type_map_,
>         FakeNetworkPipe::Config());
>@@ -228,10 +226,10 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec,
>     audio_send_config.encoder_factory = CreateBuiltinAudioEncoderFactory();
>     audio_send_stream = sender_call_->CreateAudioSendStream(audio_send_config);
> 
>-    video_send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
>+    GetVideoSendConfig()->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
>     if (fec == FecMode::kOn) {
>-      video_send_config_.rtp.ulpfec.red_payload_type = kRedPayloadType;
>-      video_send_config_.rtp.ulpfec.ulpfec_payload_type = kUlpfecPayloadType;
>+      GetVideoSendConfig()->rtp.ulpfec.red_payload_type = kRedPayloadType;
>+      GetVideoSendConfig()->rtp.ulpfec.ulpfec_payload_type = kUlpfecPayloadType;
>       video_receive_configs_[0].rtp.red_payload_type = kRedPayloadType;
>       video_receive_configs_[0].rtp.ulpfec_payload_type = kUlpfecPayloadType;
>     }
>@@ -243,7 +241,7 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec,
>     audio_recv_config.rtp.remote_ssrc = kAudioSendSsrc;
>     audio_recv_config.rtp.local_ssrc = kAudioRecvSsrc;
>     audio_recv_config.sync_group = kSyncGroup;
>-    audio_recv_config.decoder_factory = decoder_factory_;
>+    audio_recv_config.decoder_factory = audio_decoder_factory_;
>     audio_recv_config.decoder_map = {
>         {kAudioSendPayloadType, {"ISAC", 16000, 1}}};
> 
>@@ -258,7 +256,7 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec,
>     }
>     EXPECT_EQ(1u, video_receive_streams_.size());
>     observer.set_receive_stream(video_receive_streams_[0]);
>-    drifting_clock = rtc::MakeUnique<DriftingClock>(clock_, video_ntp_speed);
>+    drifting_clock = absl::make_unique<DriftingClock>(clock_, video_ntp_speed);
>     CreateFrameGeneratorCapturerWithDrift(drifting_clock.get(), video_rtp_speed,
>                                           kDefaultFramerate, kDefaultWidth,
>                                           kDefaultHeight);
>@@ -298,6 +296,12 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec,
>   }
> }
> 
>+TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithoutClockDrift) {
>+  TestAudioVideoSync(FecMode::kOff, CreateOrder::kAudioFirst,
>+                     DriftingClock::kNoDrift, DriftingClock::kNoDrift,
>+                     DriftingClock::kNoDrift, "_video_no_drift");
>+}
>+
> TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithVideoNtpDrift) {
>   TestAudioVideoSync(FecMode::kOff, CreateOrder::kAudioFirst,
>                      DriftingClock::PercentsFaster(10.0f),
>@@ -543,8 +547,7 @@ TEST_F(CallPerfTest, ReceivesCpuOveruseAndUnderuse) {
>     void ModifyVideoConfigs(
>         VideoSendStream::Config* send_config,
>         std::vector<VideoReceiveStream::Config>* receive_configs,
>-        VideoEncoderConfig* encoder_config) override {
>-    }
>+        VideoEncoderConfig* encoder_config) override {}
> 
>     void PerformTest() override {
>       EXPECT_TRUE(Wait()) << "Timed out before receiving an overuse callback.";
>@@ -637,20 +640,29 @@ void CallPerfTest::TestMinTransmitBitrate(bool pad_to_min_bitrate) {
>     std::vector<double> bitrate_kbps_list_;
>   } test(pad_to_min_bitrate);
> 
>-  fake_encoder_.SetMaxBitrate(kMaxEncodeBitrateKbps);
>+  fake_encoder_max_bitrate_ = kMaxEncodeBitrateKbps;
>   RunBaseTest(&test);
> }
> 
>-TEST_F(CallPerfTest, PadsToMinTransmitBitrate) { TestMinTransmitBitrate(true); }
>+TEST_F(CallPerfTest, PadsToMinTransmitBitrate) {
>+  TestMinTransmitBitrate(true);
>+}
> 
> TEST_F(CallPerfTest, NoPadWithoutMinTransmitBitrate) {
>   TestMinTransmitBitrate(false);
> }
> 
>-TEST_F(CallPerfTest, KeepsHighBitrateWhenReconfiguringSender) {
>+// TODO(bugs.webrtc.org/8878)
>+#if defined(WEBRTC_MAC)
>+#define MAYBE_KeepsHighBitrateWhenReconfiguringSender \
>+  DISABLED_KeepsHighBitrateWhenReconfiguringSender
>+#else
>+#define MAYBE_KeepsHighBitrateWhenReconfiguringSender \
>+  KeepsHighBitrateWhenReconfiguringSender
>+#endif
>+TEST_F(CallPerfTest, MAYBE_KeepsHighBitrateWhenReconfiguringSender) {
>   static const uint32_t kInitialBitrateKbps = 400;
>   static const uint32_t kReconfigureThresholdKbps = 600;
>-  static const uint32_t kPermittedReconfiguredBitrateDiffKbps = 100;
> 
>   class VideoStreamFactory
>       : public VideoEncoderConfig::VideoStreamFactoryInterface {
>@@ -679,7 +691,8 @@ TEST_F(CallPerfTest, KeepsHighBitrateWhenReconfiguringSender) {
>           encoder_inits_(0),
>           last_set_bitrate_kbps_(0),
>           send_stream_(nullptr),
>-          frame_generator_(nullptr) {}
>+          frame_generator_(nullptr),
>+          encoder_factory_(this) {}
> 
>     int32_t InitEncode(const VideoCodec* config,
>                        int32_t number_of_cores,
>@@ -700,16 +713,14 @@ TEST_F(CallPerfTest, KeepsHighBitrateWhenReconfiguringSender) {
>         EXPECT_EQ(2 * kDefaultWidth, config->width);
>         EXPECT_EQ(2 * kDefaultHeight, config->height);
>         EXPECT_GE(last_set_bitrate_kbps_, kReconfigureThresholdKbps);
>-        EXPECT_GT(
>-            config->startBitrate,
>-            last_set_bitrate_kbps_ - kPermittedReconfiguredBitrateDiffKbps)
>+        EXPECT_GT(config->startBitrate, kReconfigureThresholdKbps)
>             << "Encoder reconfigured with bitrate too far away from last set.";
>         observation_complete_.Set();
>       }
>       return FakeEncoder::InitEncode(config, number_of_cores, max_payload_size);
>     }
> 
>-    int32_t SetRateAllocation(const BitrateAllocation& rate_allocation,
>+    int32_t SetRateAllocation(const VideoBitrateAllocation& rate_allocation,
>                               uint32_t framerate) override {
>       last_set_bitrate_kbps_ = rate_allocation.get_sum_kbps();
>       if (encoder_inits_ == 1 &&
>@@ -719,18 +730,15 @@ TEST_F(CallPerfTest, KeepsHighBitrateWhenReconfiguringSender) {
>       return FakeEncoder::SetRateAllocation(rate_allocation, framerate);
>     }
> 
>-    Call::Config GetSenderCallConfig() override {
>-      Call::Config config = EndToEndTest::GetSenderCallConfig();
>-      config.event_log = event_log_.get();
>-      config.bitrate_config.start_bitrate_bps = kInitialBitrateKbps * 1000;
>-      return config;
>+    void ModifySenderCallConfig(Call::Config* config) override {
>+      config->bitrate_config.start_bitrate_bps = kInitialBitrateKbps * 1000;
>     }
> 
>     void ModifyVideoConfigs(
>         VideoSendStream::Config* send_config,
>         std::vector<VideoReceiveStream::Config>* receive_configs,
>         VideoEncoderConfig* encoder_config) override {
>-      send_config->encoder_settings.encoder = this;
>+      send_config->encoder_settings.encoder_factory = &encoder_factory_;
>       encoder_config->max_bitrate_bps = 2 * kReconfigureThresholdKbps * 1000;
>       encoder_config->video_stream_factory =
>           new rtc::RefCountedObject<VideoStreamFactory>();
>@@ -765,6 +773,7 @@ TEST_F(CallPerfTest, KeepsHighBitrateWhenReconfiguringSender) {
>     uint32_t last_set_bitrate_kbps_;
>     VideoSendStream* send_stream_;
>     test::FrameGeneratorCapturer* frame_generator_;
>+    test::EncoderProxyFactory encoder_factory_;
>     VideoEncoderConfig encoder_config_;
>   } test;
> 
>@@ -880,11 +889,12 @@ void CallPerfTest::TestMinAudioVideoBitrate(
> 
>     void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
>       sender_call_ = sender_call;
>-      Call::Config::BitrateConfig bitrate_config;
>+      BitrateConstraints bitrate_config;
>       bitrate_config.min_bitrate_bps = min_bwe_;
>       bitrate_config.start_bitrate_bps = start_bwe_;
>       bitrate_config.max_bitrate_bps = max_bwe_;
>-      sender_call->SetBitrateConfig(bitrate_config);
>+      sender_call->GetTransportControllerSend()->SetSdpBitrateParameters(
>+          bitrate_config);
>       if (use_bitrate_allocation_strategy_) {
>         sender_call->SetBitrateAllocationStrategy(
>             std::move(allocation_strategy_));
>@@ -904,7 +914,7 @@ void CallPerfTest::TestMinAudioVideoBitrate(
>         send_config->max_bitrate_bps = kOpusBitrateFbBps;
>       } else {
>         send_config->send_codec_spec->target_bitrate_bps =
>-            rtc::Optional<int>(kOpusBitrateFbBps);
>+            absl::optional<int>(kOpusBitrateFbBps);
>       }
>     }
> 
>@@ -926,7 +936,13 @@ void CallPerfTest::TestMinAudioVideoBitrate(
>   RunBaseTest(&test);
> }
> 
>-TEST_F(CallPerfTest, MinVideoAndAudioBitrate) {
>+// TODO(bugs.webrtc.org/8878)
>+#if defined(WEBRTC_MAC)
>+#define MAYBE_MinVideoAndAudioBitrate DISABLED_MinVideoAndAudioBitrate
>+#else
>+#define MAYBE_MinVideoAndAudioBitrate MinVideoAndAudioBitrate
>+#endif
>+TEST_F(CallPerfTest, MAYBE_MinVideoAndAudioBitrate) {
>   TestMinAudioVideoBitrate(false, 110, 40, -10, 10000, 70000, 200000);
> }
> TEST_F(CallPerfTest, MinVideoAndAudioBitrateWStrategy) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/call_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/call_unittest.cc
>index dc36ce20720..1084bcfb46e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/call_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/call_unittest.cc
>@@ -13,20 +13,18 @@
> #include <memory>
> #include <utility>
> 
>+#include "absl/memory/memory.h"
> #include "api/audio_codecs/builtin_audio_decoder_factory.h"
> #include "api/test/mock_audio_mixer.h"
>-#include "audio/audio_send_stream.h"
> #include "audio/audio_receive_stream.h"
>+#include "audio/audio_send_stream.h"
> #include "call/audio_state.h"
> #include "call/call.h"
>-#include "call/fake_rtp_transport_controller_send.h"
> #include "logging/rtc_event_log/rtc_event_log.h"
> #include "modules/audio_device/include/mock_audio_device.h"
> #include "modules/audio_processing/include/mock_audio_processing.h"
>-#include "modules/congestion_controller/include/mock/mock_send_side_congestion_controller.h"
> #include "modules/pacing/mock/mock_paced_sender.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp.h"
>-#include "rtc_base/ptr_util.h"
> #include "test/fake_encoder.h"
> #include "test/gtest.h"
> #include "test/mock_audio_decoder_factory.h"
>@@ -252,115 +250,6 @@ TEST(CallTest, MultipleFlexfecReceiveStreamsProtectingSingleVideoStream) {
>   }
> }
> 
>-namespace {
>-struct CallBitrateHelper {
>-  CallBitrateHelper() : CallBitrateHelper(Call::Config::BitrateConfig()) {}
>-
>-  explicit CallBitrateHelper(const Call::Config::BitrateConfig& bitrate_config)
>-      : mock_cc_(Clock::GetRealTimeClock(), &event_log_, &pacer_) {
>-    Call::Config config(&event_log_);
>-    config.bitrate_config = bitrate_config;
>-    call_.reset(
>-        Call::Create(config, rtc::MakeUnique<FakeRtpTransportControllerSend>(
>-                                 &packet_router_, &pacer_, &mock_cc_)));
>-  }
>-
>-  webrtc::Call* operator->() { return call_.get(); }
>-  testing::NiceMock<test::MockSendSideCongestionController>& mock_cc() {
>-    return mock_cc_;
>-  }
>-
>- private:
>-  webrtc::RtcEventLogNullImpl event_log_;
>-  PacketRouter packet_router_;
>-  testing::NiceMock<MockPacedSender> pacer_;
>-  testing::NiceMock<test::MockSendSideCongestionController> mock_cc_;
>-  std::unique_ptr<Call> call_;
>-};
>-}  // namespace
>-
>-TEST(CallBitrateTest, SetBitrateConfigWithValidConfigCallsSetBweBitrates) {
>-  CallBitrateHelper call;
>-
>-  Call::Config::BitrateConfig bitrate_config;
>-  bitrate_config.min_bitrate_bps = 1;
>-  bitrate_config.start_bitrate_bps = 2;
>-  bitrate_config.max_bitrate_bps = 3;
>-
>-  EXPECT_CALL(call.mock_cc(), SetBweBitrates(1, 2, 3));
>-  call->SetBitrateConfig(bitrate_config);
>-}
>-
>-TEST(CallBitrateTest, SetBitrateConfigWithDifferentMinCallsSetBweBitrates) {
>-  CallBitrateHelper call;
>-
>-  Call::Config::BitrateConfig bitrate_config;
>-  bitrate_config.min_bitrate_bps = 10;
>-  bitrate_config.start_bitrate_bps = 20;
>-  bitrate_config.max_bitrate_bps = 30;
>-  call->SetBitrateConfig(bitrate_config);
>-
>-  bitrate_config.min_bitrate_bps = 11;
>-  EXPECT_CALL(call.mock_cc(), SetBweBitrates(11, -1, 30));
>-  call->SetBitrateConfig(bitrate_config);
>-}
>-
>-TEST(CallBitrateTest, SetBitrateConfigWithDifferentStartCallsSetBweBitrates) {
>-  CallBitrateHelper call;
>-
>-  Call::Config::BitrateConfig bitrate_config;
>-  bitrate_config.min_bitrate_bps = 10;
>-  bitrate_config.start_bitrate_bps = 20;
>-  bitrate_config.max_bitrate_bps = 30;
>-  call->SetBitrateConfig(bitrate_config);
>-
>-  bitrate_config.start_bitrate_bps = 21;
>-  EXPECT_CALL(call.mock_cc(), SetBweBitrates(10, 21, 30));
>-  call->SetBitrateConfig(bitrate_config);
>-}
>-
>-TEST(CallBitrateTest, SetBitrateConfigWithDifferentMaxCallsSetBweBitrates) {
>-  CallBitrateHelper call;
>-
>-  Call::Config::BitrateConfig bitrate_config;
>-  bitrate_config.min_bitrate_bps = 10;
>-  bitrate_config.start_bitrate_bps = 20;
>-  bitrate_config.max_bitrate_bps = 30;
>-  call->SetBitrateConfig(bitrate_config);
>-
>-  bitrate_config.max_bitrate_bps = 31;
>-  EXPECT_CALL(call.mock_cc(), SetBweBitrates(10, -1, 31));
>-  call->SetBitrateConfig(bitrate_config);
>-}
>-
>-TEST(CallBitrateTest, SetBitrateConfigWithSameConfigElidesSecondCall) {
>-  CallBitrateHelper call;
>-  Call::Config::BitrateConfig bitrate_config;
>-  bitrate_config.min_bitrate_bps = 1;
>-  bitrate_config.start_bitrate_bps = 2;
>-  bitrate_config.max_bitrate_bps = 3;
>-
>-  EXPECT_CALL(call.mock_cc(), SetBweBitrates(1, 2, 3)).Times(1);
>-  call->SetBitrateConfig(bitrate_config);
>-  call->SetBitrateConfig(bitrate_config);
>-}
>-
>-TEST(CallBitrateTest,
>-     SetBitrateConfigWithSameMinMaxAndNegativeStartElidesSecondCall) {
>-  CallBitrateHelper call;
>-
>-  Call::Config::BitrateConfig bitrate_config;
>-  bitrate_config.min_bitrate_bps = 1;
>-  bitrate_config.start_bitrate_bps = 2;
>-  bitrate_config.max_bitrate_bps = 3;
>-
>-  EXPECT_CALL(call.mock_cc(), SetBweBitrates(1, 2, 3)).Times(1);
>-  call->SetBitrateConfig(bitrate_config);
>-
>-  bitrate_config.start_bitrate_bps = -1;
>-  call->SetBitrateConfig(bitrate_config);
>-}
>-
> TEST(CallTest, RecreatingAudioStreamWithSameSsrcReusesRtpState) {
>   constexpr uint32_t kSSRC = 12345;
>   CallHelper call;
>@@ -387,222 +276,4 @@ TEST(CallTest, RecreatingAudioStreamWithSameSsrcReusesRtpState) {
>   EXPECT_EQ(rtp_state1.media_has_been_sent, rtp_state2.media_has_been_sent);
> }
> 
>-TEST(CallBitrateTest, BiggerMaskMinUsed) {
>-  CallBitrateHelper call;
>-  Call::Config::BitrateConfigMask mask;
>-  mask.min_bitrate_bps = 1234;
>-
>-  EXPECT_CALL(call.mock_cc(),
>-              SetBweBitrates(*mask.min_bitrate_bps, testing::_, testing::_));
>-  call->SetBitrateConfigMask(mask);
>-}
>-
>-TEST(CallBitrateTest, BiggerConfigMinUsed) {
>-  CallBitrateHelper call;
>-  Call::Config::BitrateConfigMask mask;
>-  mask.min_bitrate_bps = 1000;
>-  EXPECT_CALL(call.mock_cc(), SetBweBitrates(1000, testing::_, testing::_));
>-  call->SetBitrateConfigMask(mask);
>-
>-  Call::Config::BitrateConfig config;
>-  config.min_bitrate_bps = 1234;
>-
>-  EXPECT_CALL(call.mock_cc(), SetBweBitrates(1234, testing::_, testing::_));
>-  call->SetBitrateConfig(config);
>-}
>-
>-// The last call to set start should be used.
>-TEST(CallBitrateTest, LatestStartMaskPreferred) {
>-  CallBitrateHelper call;
>-  Call::Config::BitrateConfigMask mask;
>-  mask.start_bitrate_bps = 1300;
>-
>-  EXPECT_CALL(call.mock_cc(),
>-              SetBweBitrates(testing::_, *mask.start_bitrate_bps, testing::_));
>-  call->SetBitrateConfigMask(mask);
>-
>-  Call::Config::BitrateConfig bitrate_config;
>-  bitrate_config.start_bitrate_bps = 1200;
>-
>-  EXPECT_CALL(
>-      call.mock_cc(),
>-      SetBweBitrates(testing::_, bitrate_config.start_bitrate_bps, testing::_));
>-  call->SetBitrateConfig(bitrate_config);
>-}
>-
>-TEST(CallBitrateTest, SmallerMaskMaxUsed) {
>-  Call::Config::BitrateConfig bitrate_config;
>-  bitrate_config.max_bitrate_bps = bitrate_config.start_bitrate_bps + 2000;
>-  CallBitrateHelper call(bitrate_config);
>-
>-  Call::Config::BitrateConfigMask mask;
>-  mask.max_bitrate_bps = bitrate_config.start_bitrate_bps + 1000;
>-
>-  EXPECT_CALL(call.mock_cc(),
>-              SetBweBitrates(testing::_, testing::_, *mask.max_bitrate_bps));
>-  call->SetBitrateConfigMask(mask);
>-}
>-
>-TEST(CallBitrateTest, SmallerConfigMaxUsed) {
>-  Call::Config::BitrateConfig bitrate_config;
>-  bitrate_config.max_bitrate_bps = bitrate_config.start_bitrate_bps + 1000;
>-  CallBitrateHelper call(bitrate_config);
>-
>-  Call::Config::BitrateConfigMask mask;
>-  mask.max_bitrate_bps = bitrate_config.start_bitrate_bps + 2000;
>-
>-  // Expect no calls because nothing changes
>-  EXPECT_CALL(call.mock_cc(),
>-              SetBweBitrates(testing::_, testing::_, testing::_))
>-      .Times(0);
>-  call->SetBitrateConfigMask(mask);
>-}
>-
>-TEST(CallBitrateTest, MaskStartLessThanConfigMinClamped) {
>-  Call::Config::BitrateConfig bitrate_config;
>-  bitrate_config.min_bitrate_bps = 2000;
>-  CallBitrateHelper call(bitrate_config);
>-
>-  Call::Config::BitrateConfigMask mask;
>-  mask.start_bitrate_bps = 1000;
>-
>-  EXPECT_CALL(call.mock_cc(), SetBweBitrates(2000, 2000, testing::_));
>-  call->SetBitrateConfigMask(mask);
>-}
>-
>-TEST(CallBitrateTest, MaskStartGreaterThanConfigMaxClamped) {
>-  Call::Config::BitrateConfig bitrate_config;
>-  bitrate_config.start_bitrate_bps = 2000;
>-  CallBitrateHelper call(bitrate_config);
>-
>-  Call::Config::BitrateConfigMask mask;
>-  mask.max_bitrate_bps = 1000;
>-
>-  EXPECT_CALL(call.mock_cc(), SetBweBitrates(testing::_, -1, 1000));
>-  call->SetBitrateConfigMask(mask);
>-}
>-
>-TEST(CallBitrateTest, MaskMinGreaterThanConfigMaxClamped) {
>-  Call::Config::BitrateConfig bitrate_config;
>-  bitrate_config.min_bitrate_bps = 2000;
>-  CallBitrateHelper call(bitrate_config);
>-
>-  Call::Config::BitrateConfigMask mask;
>-  mask.max_bitrate_bps = 1000;
>-
>-  EXPECT_CALL(call.mock_cc(), SetBweBitrates(1000, testing::_, 1000));
>-  call->SetBitrateConfigMask(mask);
>-}
>-
>-TEST(CallBitrateTest, SettingMaskStartForcesUpdate) {
>-  CallBitrateHelper call;
>-
>-  Call::Config::BitrateConfigMask mask;
>-  mask.start_bitrate_bps = 1000;
>-
>-  // SetBweBitrates should be called twice with the same params since
>-  // start_bitrate_bps is set.
>-  EXPECT_CALL(call.mock_cc(), SetBweBitrates(testing::_, 1000, testing::_))
>-      .Times(2);
>-  call->SetBitrateConfigMask(mask);
>-  call->SetBitrateConfigMask(mask);
>-}
>-
>-TEST(CallBitrateTest, SetBitrateConfigWithNoChangesDoesNotCallSetBweBitrates) {
>-  CallBitrateHelper call;
>-
>-  Call::Config::BitrateConfig config1;
>-  config1.min_bitrate_bps = 0;
>-  config1.start_bitrate_bps = 1000;
>-  config1.max_bitrate_bps = -1;
>-
>-  Call::Config::BitrateConfig config2;
>-  config2.min_bitrate_bps = 0;
>-  config2.start_bitrate_bps = -1;
>-  config2.max_bitrate_bps = -1;
>-
>-  // The second call should not call SetBweBitrates because it doesn't
>-  // change any values.
>-  EXPECT_CALL(call.mock_cc(), SetBweBitrates(0, 1000, -1));
>-  call->SetBitrateConfig(config1);
>-  call->SetBitrateConfig(config2);
>-}
>-
>-// If SetBitrateConfig changes the max, but not the effective max,
>-// SetBweBitrates shouldn't be called, to avoid unnecessary encoder
>-// reconfigurations.
>-TEST(CallBitrateTest, SetBweBitratesNotCalledWhenEffectiveMaxUnchanged) {
>-  CallBitrateHelper call;
>-
>-  Call::Config::BitrateConfig config;
>-  config.min_bitrate_bps = 0;
>-  config.start_bitrate_bps = -1;
>-  config.max_bitrate_bps = 2000;
>-  EXPECT_CALL(call.mock_cc(), SetBweBitrates(testing::_, testing::_, 2000));
>-  call->SetBitrateConfig(config);
>-
>-  // Reduce effective max to 1000 with the mask.
>-  Call::Config::BitrateConfigMask mask;
>-  mask.max_bitrate_bps = 1000;
>-  EXPECT_CALL(call.mock_cc(), SetBweBitrates(testing::_, testing::_, 1000));
>-  call->SetBitrateConfigMask(mask);
>-
>-  // This leaves the effective max unchanged, so SetBweBitrates shouldn't be
>-  // called again.
>-  config.max_bitrate_bps = 1000;
>-  call->SetBitrateConfig(config);
>-}
>-
>-// When the "start bitrate" mask is removed, SetBweBitrates shouldn't be called
>-// again, since nothing's changing.
>-TEST(CallBitrateTest, SetBweBitratesNotCalledWhenStartMaskRemoved) {
>-  CallBitrateHelper call;
>-
>-  Call::Config::BitrateConfigMask mask;
>-  mask.start_bitrate_bps = 1000;
>-  EXPECT_CALL(call.mock_cc(), SetBweBitrates(0, 1000, -1));
>-  call->SetBitrateConfigMask(mask);
>-
>-  mask.start_bitrate_bps.reset();
>-  call->SetBitrateConfigMask(mask);
>-}
>-
>-// Test that if SetBitrateConfig is called after SetBitrateConfigMask applies a
>-// "start" value, the SetBitrateConfig call won't apply that start value a
>-// second time.
>-TEST(CallBitrateTest, SetBitrateConfigAfterSetBitrateConfigMaskWithStart) {
>-  CallBitrateHelper call;
>-
>-  Call::Config::BitrateConfigMask mask;
>-  mask.start_bitrate_bps = 1000;
>-  EXPECT_CALL(call.mock_cc(), SetBweBitrates(0, 1000, -1));
>-  call->SetBitrateConfigMask(mask);
>-
>-  Call::Config::BitrateConfig config;
>-  config.min_bitrate_bps = 0;
>-  config.start_bitrate_bps = -1;
>-  config.max_bitrate_bps = 5000;
>-  // The start value isn't changing, so SetBweBitrates should be called with
>-  // -1.
>-  EXPECT_CALL(call.mock_cc(), SetBweBitrates(0, -1, 5000));
>-  call->SetBitrateConfig(config);
>-}
>-
>-TEST(CallBitrateTest, SetBweBitratesNotCalledWhenClampedMinUnchanged) {
>-  Call::Config::BitrateConfig bitrate_config;
>-  bitrate_config.start_bitrate_bps = 500;
>-  bitrate_config.max_bitrate_bps = 1000;
>-  CallBitrateHelper call(bitrate_config);
>-
>-  // Set min to 2000; it is clamped to the max (1000).
>-  Call::Config::BitrateConfigMask mask;
>-  mask.min_bitrate_bps = 2000;
>-  EXPECT_CALL(call.mock_cc(), SetBweBitrates(1000, -1, 1000));
>-  call->SetBitrateConfigMask(mask);
>-
>-  // Set min to 3000; the clamped value stays the same so nothing happens.
>-  mask.min_bitrate_bps = 3000;
>-  call->SetBitrateConfigMask(mask);
>-}
>-
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/callfactory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/callfactory.cc
>index 82acb6533c0..b5c28d2e456 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/callfactory.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/callfactory.cc
>@@ -11,10 +11,74 @@
> #include "call/callfactory.h"
> 
> #include <memory>
>+#include <string>
>+#include <utility>
>+
>+#include "absl/types/optional.h"
>+#include "call/call.h"
>+#include "call/degraded_call.h"
>+#include "call/fake_network_pipe.h"
>+#include "system_wrappers/include/field_trial.h"
> 
> namespace webrtc {
>+namespace {
>+bool ParseConfigParam(std::string exp_name, int* field) {
>+  std::string group = field_trial::FindFullName(exp_name);
>+  if (group == "")
>+    return false;
>+
>+  return (sscanf(group.c_str(), "%d", field) == 1);
>+}
>+
>+absl::optional<webrtc::FakeNetworkPipe::Config> ParseDegradationConfig(
>+    bool send) {
>+  std::string exp_prefix = "WebRTCFakeNetwork";
>+  if (send) {
>+    exp_prefix += "Send";
>+  } else {
>+    exp_prefix += "Receive";
>+  }
>+
>+  webrtc::FakeNetworkPipe::Config config;
>+  bool configured = false;
>+  configured |=
>+      ParseConfigParam(exp_prefix + "DelayMs", &config.queue_delay_ms);
>+  configured |= ParseConfigParam(exp_prefix + "DelayStdDevMs",
>+                                 &config.delay_standard_deviation_ms);
>+  int queue_length = 0;
>+  if (ParseConfigParam(exp_prefix + "QueueLength", &queue_length)) {
>+    RTC_CHECK_GE(queue_length, 0);
>+    config.queue_length_packets = queue_length;
>+    configured = true;
>+  }
>+  configured |=
>+      ParseConfigParam(exp_prefix + "CapacityKbps", &config.link_capacity_kbps);
>+  configured |=
>+      ParseConfigParam(exp_prefix + "LossPercent", &config.loss_percent);
>+  int allow_reordering = 0;
>+  if (ParseConfigParam(exp_prefix + "AllowReordering", &allow_reordering)) {
>+    config.allow_reordering = true;
>+    configured = true;
>+  }
>+  configured |= ParseConfigParam(exp_prefix + "AvgBurstLossLength",
>+                                 &config.avg_burst_loss_length);
>+  return configured ? absl::optional<webrtc::FakeNetworkPipe::Config>(config)
>+                    : absl::nullopt;
>+}
>+}  // namespace
> 
> Call* CallFactory::CreateCall(const Call::Config& config) {
>+  absl::optional<webrtc::FakeNetworkPipe::Config> send_degradation_config =
>+      ParseDegradationConfig(true);
>+  absl::optional<webrtc::FakeNetworkPipe::Config> receive_degradation_config =
>+      ParseDegradationConfig(false);
>+
>+  if (send_degradation_config || receive_degradation_config) {
>+    return new DegradedCall(std::unique_ptr<Call>(Call::Create(config)),
>+                            send_degradation_config,
>+                            receive_degradation_config);
>+  }
>+
>   return Call::Create(config);
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/callfactory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/call/callfactory.h
>index 167b82ad541..1c57bd435f8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/callfactory.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/callfactory.h
>@@ -11,14 +11,14 @@
> #ifndef CALL_CALLFACTORY_H_
> #define CALL_CALLFACTORY_H_
> 
>-#include "call/callfactoryinterface.h"
>+#include "api/call/callfactoryinterface.h"
> 
> namespace webrtc {
> 
> class CallFactory : public CallFactoryInterface {
>   ~CallFactory() override {}
> 
>-  Call* CreateCall(const Call::Config& config) override;
>+  Call* CreateCall(const CallConfig& config) override;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/degraded_call.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/degraded_call.cc
>new file mode 100644
>index 00000000000..f24ef594144
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/degraded_call.cc
>@@ -0,0 +1,209 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <utility>
>+
>+#include "call/degraded_call.h"
>+
>+#include "absl/memory/memory.h"
>+
>+namespace webrtc {
>+DegradedCall::DegradedCall(
>+    std::unique_ptr<Call> call,
>+    absl::optional<FakeNetworkPipe::Config> send_config,
>+    absl::optional<FakeNetworkPipe::Config> receive_config)
>+    : clock_(Clock::GetRealTimeClock()),
>+      call_(std::move(call)),
>+      send_config_(send_config),
>+      send_process_thread_(
>+          send_config_ ? ProcessThread::Create("DegradedSendThread") : nullptr),
>+      num_send_streams_(0),
>+      receive_config_(receive_config) {
>+  if (receive_config_) {
>+    receive_pipe_ =
>+        absl::make_unique<webrtc::FakeNetworkPipe>(clock_, *receive_config_);
>+    receive_pipe_->SetReceiver(call_->Receiver());
>+  }
>+  if (send_process_thread_) {
>+    send_process_thread_->Start();
>+  }
>+}
>+
>+DegradedCall::~DegradedCall() {
>+  if (send_pipe_) {
>+    send_process_thread_->DeRegisterModule(send_pipe_.get());
>+  }
>+  if (send_process_thread_) {
>+    send_process_thread_->Stop();
>+  }
>+}
>+
>+AudioSendStream* DegradedCall::CreateAudioSendStream(
>+    const AudioSendStream::Config& config) {
>+  return call_->CreateAudioSendStream(config);
>+}
>+
>+void DegradedCall::DestroyAudioSendStream(AudioSendStream* send_stream) {
>+  call_->DestroyAudioSendStream(send_stream);
>+}
>+
>+AudioReceiveStream* DegradedCall::CreateAudioReceiveStream(
>+    const AudioReceiveStream::Config& config) {
>+  return call_->CreateAudioReceiveStream(config);
>+}
>+
>+void DegradedCall::DestroyAudioReceiveStream(
>+    AudioReceiveStream* receive_stream) {
>+  call_->DestroyAudioReceiveStream(receive_stream);
>+}
>+
>+VideoSendStream* DegradedCall::CreateVideoSendStream(
>+    VideoSendStream::Config config,
>+    VideoEncoderConfig encoder_config) {
>+  if (send_config_ && !send_pipe_) {
>+    send_pipe_ = absl::make_unique<FakeNetworkPipe>(clock_, *send_config_,
>+                                                    config.send_transport);
>+    config.send_transport = this;
>+    send_process_thread_->RegisterModule(send_pipe_.get(), RTC_FROM_HERE);
>+  }
>+  ++num_send_streams_;
>+  return call_->CreateVideoSendStream(std::move(config),
>+                                      std::move(encoder_config));
>+}
>+
>+VideoSendStream* DegradedCall::CreateVideoSendStream(
>+    VideoSendStream::Config config,
>+    VideoEncoderConfig encoder_config,
>+    std::unique_ptr<FecController> fec_controller) {
>+  if (send_config_ && !send_pipe_) {
>+    send_pipe_ = absl::make_unique<FakeNetworkPipe>(clock_, *send_config_,
>+                                                    config.send_transport);
>+    config.send_transport = this;
>+    send_process_thread_->RegisterModule(send_pipe_.get(), RTC_FROM_HERE);
>+  }
>+  ++num_send_streams_;
>+  return call_->CreateVideoSendStream(
>+      std::move(config), std::move(encoder_config), std::move(fec_controller));
>+}
>+
>+void DegradedCall::DestroyVideoSendStream(VideoSendStream* send_stream) {
>+  call_->DestroyVideoSendStream(send_stream);
>+  if (send_pipe_ && num_send_streams_ > 0) {
>+    --num_send_streams_;
>+    if (num_send_streams_ == 0) {
>+      send_process_thread_->DeRegisterModule(send_pipe_.get());
>+      send_pipe_.reset();
>+    }
>+  }
>+}
>+
>+VideoReceiveStream* DegradedCall::CreateVideoReceiveStream(
>+    VideoReceiveStream::Config configuration) {
>+  return call_->CreateVideoReceiveStream(std::move(configuration));
>+}
>+
>+void DegradedCall::DestroyVideoReceiveStream(
>+    VideoReceiveStream* receive_stream) {
>+  call_->DestroyVideoReceiveStream(receive_stream);
>+}
>+
>+FlexfecReceiveStream* DegradedCall::CreateFlexfecReceiveStream(
>+    const FlexfecReceiveStream::Config& config) {
>+  return call_->CreateFlexfecReceiveStream(config);
>+}
>+
>+void DegradedCall::DestroyFlexfecReceiveStream(
>+    FlexfecReceiveStream* receive_stream) {
>+  call_->DestroyFlexfecReceiveStream(receive_stream);
>+}
>+
>+PacketReceiver* DegradedCall::Receiver() {
>+  if (receive_config_) {
>+    return this;
>+  }
>+  return call_->Receiver();
>+}
>+
>+RtpTransportControllerSendInterface*
>+DegradedCall::GetTransportControllerSend() {
>+  return call_->GetTransportControllerSend();
>+}
>+
>+Call::Stats DegradedCall::GetStats() const {
>+  return call_->GetStats();
>+}
>+
>+void DegradedCall::SetBitrateAllocationStrategy(
>+    std::unique_ptr<rtc::BitrateAllocationStrategy>
>+        bitrate_allocation_strategy) {
>+  call_->SetBitrateAllocationStrategy(std::move(bitrate_allocation_strategy));
>+}
>+
>+void DegradedCall::SignalChannelNetworkState(MediaType media,
>+                                             NetworkState state) {
>+  call_->SignalChannelNetworkState(media, state);
>+}
>+
>+void DegradedCall::OnTransportOverheadChanged(
>+    MediaType media,
>+    int transport_overhead_per_packet) {
>+  call_->OnTransportOverheadChanged(media, transport_overhead_per_packet);
>+}
>+
>+void DegradedCall::OnSentPacket(const rtc::SentPacket& sent_packet) {
>+  if (send_config_) {
>+    // If we have a degraded send-transport, we have already notified call
>+    // about the supposed network send time. Discard the actual network send
>+    // time in order to properly fool the BWE.
>+    return;
>+  }
>+  call_->OnSentPacket(sent_packet);
>+}
>+
>+bool DegradedCall::SendRtp(const uint8_t* packet,
>+                           size_t length,
>+                           const PacketOptions& options) {
>+  // A call here comes from the RTP stack (probably pacer). We intercept it and
>+  // put it in the fake network pipe instead, but report to Call that is has
>+  // been sent, so that the bandwidth estimator sees the delay we add.
>+  send_pipe_->SendRtp(packet, length, options);
>+  if (options.packet_id != -1) {
>+    rtc::SentPacket packet_info;
>+    packet_info.packet_id = options.packet_id;
>+    packet_info.send_time_ms = clock_->TimeInMilliseconds();
>+    call_->OnSentPacket(packet_info);
>+  }
>+  return true;
>+}
>+
>+bool DegradedCall::SendRtcp(const uint8_t* packet, size_t length) {
>+  send_pipe_->SendRtcp(packet, length);
>+  return true;
>+}
>+
>+PacketReceiver::DeliveryStatus DegradedCall::DeliverPacket(
>+    MediaType media_type,
>+    rtc::CopyOnWriteBuffer packet,
>+    int64_t packet_time_us) {
>+  PacketReceiver::DeliveryStatus status = receive_pipe_->DeliverPacket(
>+      media_type, std::move(packet), packet_time_us);
>+  // This is not optimal, but there are many places where there are thread
>+  // checks that fail if we're not using the worker thread call into this
>+  // method. If we want to fix this we probably need a task queue to do handover
>+  // of all overriden methods, which feels like overikill for the current use
>+  // case.
>+  // By just having this thread call out via the Process() method we work around
>+  // that, with the tradeoff that a non-zero delay may become a little larger
>+  // than anticipated at very low packet rates.
>+  receive_pipe_->Process();
>+  return status;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/degraded_call.h b/Source/ThirdParty/libwebrtc/Source/webrtc/call/degraded_call.h
>new file mode 100644
>index 00000000000..b1e22495874
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/degraded_call.h
>@@ -0,0 +1,104 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef CALL_DEGRADED_CALL_H_
>+#define CALL_DEGRADED_CALL_H_
>+
>+#include <memory>
>+
>+#include "absl/types/optional.h"
>+#include "api/call/transport.h"
>+#include "call/call.h"
>+#include "call/fake_network_pipe.h"
>+#include "modules/utility/include/process_thread.h"
>+#include "system_wrappers/include/clock.h"
>+
>+namespace webrtc {
>+
>+class DegradedCall : public Call, private Transport, private PacketReceiver {
>+ public:
>+  explicit DegradedCall(std::unique_ptr<Call> call,
>+                        absl::optional<FakeNetworkPipe::Config> send_config,
>+                        absl::optional<FakeNetworkPipe::Config> receive_config);
>+  ~DegradedCall() override;
>+
>+  // Implements Call.
>+  AudioSendStream* CreateAudioSendStream(
>+      const AudioSendStream::Config& config) override;
>+  void DestroyAudioSendStream(AudioSendStream* send_stream) override;
>+
>+  AudioReceiveStream* CreateAudioReceiveStream(
>+      const AudioReceiveStream::Config& config) override;
>+  void DestroyAudioReceiveStream(AudioReceiveStream* receive_stream) override;
>+
>+  VideoSendStream* CreateVideoSendStream(
>+      VideoSendStream::Config config,
>+      VideoEncoderConfig encoder_config) override;
>+  VideoSendStream* CreateVideoSendStream(
>+      VideoSendStream::Config config,
>+      VideoEncoderConfig encoder_config,
>+      std::unique_ptr<FecController> fec_controller) override;
>+  void DestroyVideoSendStream(VideoSendStream* send_stream) override;
>+
>+  VideoReceiveStream* CreateVideoReceiveStream(
>+      VideoReceiveStream::Config configuration) override;
>+  void DestroyVideoReceiveStream(VideoReceiveStream* receive_stream) override;
>+
>+  FlexfecReceiveStream* CreateFlexfecReceiveStream(
>+      const FlexfecReceiveStream::Config& config) override;
>+  void DestroyFlexfecReceiveStream(
>+      FlexfecReceiveStream* receive_stream) override;
>+
>+  PacketReceiver* Receiver() override;
>+
>+  RtpTransportControllerSendInterface* GetTransportControllerSend() override;
>+
>+  Stats GetStats() const override;
>+
>+  void SetBitrateAllocationStrategy(
>+      std::unique_ptr<rtc::BitrateAllocationStrategy>
>+          bitrate_allocation_strategy) override;
>+
>+  void SignalChannelNetworkState(MediaType media, NetworkState state) override;
>+
>+  void OnTransportOverheadChanged(MediaType media,
>+                                  int transport_overhead_per_packet) override;
>+
>+  void OnSentPacket(const rtc::SentPacket& sent_packet) override;
>+
>+ protected:
>+  // Implements Transport.
>+  bool SendRtp(const uint8_t* packet,
>+               size_t length,
>+               const PacketOptions& options) override;
>+
>+  bool SendRtcp(const uint8_t* packet, size_t length) override;
>+
>+  // Implements PacketReceiver.
>+  DeliveryStatus DeliverPacket(MediaType media_type,
>+                               rtc::CopyOnWriteBuffer packet,
>+                               int64_t packet_time_us) override;
>+
>+ private:
>+  Clock* const clock_;
>+  const std::unique_ptr<Call> call_;
>+
>+  const absl::optional<FakeNetworkPipe::Config> send_config_;
>+  const std::unique_ptr<ProcessThread> send_process_thread_;
>+  std::unique_ptr<FakeNetworkPipe> send_pipe_;
>+  size_t num_send_streams_;
>+
>+  const absl::optional<FakeNetworkPipe::Config> receive_config_;
>+  std::unique_ptr<FakeNetworkPipe> receive_pipe_;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // CALL_DEGRADED_CALL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/fake_network_pipe.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/fake_network_pipe.cc
>new file mode 100644
>index 00000000000..55af767cb2f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/fake_network_pipe.cc
>@@ -0,0 +1,422 @@
>+/*
>+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <assert.h>
>+#include <math.h>
>+#include <string.h>
>+
>+#include <algorithm>
>+#include <utility>
>+
>+#include "absl/memory/memory.h"
>+#include "call/call.h"
>+#include "call/fake_network_pipe.h"
>+#include "call/simulated_network.h"
>+#include "rtc_base/logging.h"
>+#include "system_wrappers/include/clock.h"
>+
>+namespace webrtc {
>+
>+namespace {
>+constexpr int64_t kDefaultProcessIntervalMs = 5;
>+constexpr int64_t kLogIntervalMs = 5000;
>+}  // namespace
>+
>+NetworkPacket::NetworkPacket(rtc::CopyOnWriteBuffer packet,
>+                             int64_t send_time,
>+                             int64_t arrival_time,
>+                             absl::optional<PacketOptions> packet_options,
>+                             bool is_rtcp,
>+                             MediaType media_type,
>+                             absl::optional<int64_t> packet_time_us)
>+    : packet_(std::move(packet)),
>+      send_time_(send_time),
>+      arrival_time_(arrival_time),
>+      packet_options_(packet_options),
>+      is_rtcp_(is_rtcp),
>+      media_type_(media_type),
>+      packet_time_us_(packet_time_us) {}
>+
>+NetworkPacket::NetworkPacket(rtc::CopyOnWriteBuffer packet,
>+                             int64_t send_time,
>+                             int64_t arrival_time,
>+                             absl::optional<PacketOptions> packet_options,
>+                             bool is_rtcp,
>+                             MediaType media_type,
>+                             absl::optional<PacketTime> packet_time)
>+    : NetworkPacket(packet,
>+                    send_time,
>+                    arrival_time,
>+                    packet_options,
>+                    is_rtcp,
>+                    media_type,
>+                    packet_time
>+                        ? absl::optional<int64_t>(packet_time->timestamp)
>+                        : absl::nullopt) {}
>+
>+NetworkPacket::NetworkPacket(NetworkPacket&& o)
>+    : packet_(std::move(o.packet_)),
>+      send_time_(o.send_time_),
>+      arrival_time_(o.arrival_time_),
>+      packet_options_(o.packet_options_),
>+      is_rtcp_(o.is_rtcp_),
>+      media_type_(o.media_type_),
>+      packet_time_us_(o.packet_time_us_) {}
>+
>+NetworkPacket::~NetworkPacket() = default;
>+
>+NetworkPacket& NetworkPacket::operator=(NetworkPacket&& o) {
>+  packet_ = std::move(o.packet_);
>+  send_time_ = o.send_time_;
>+  arrival_time_ = o.arrival_time_;
>+  packet_options_ = o.packet_options_;
>+  is_rtcp_ = o.is_rtcp_;
>+  media_type_ = o.media_type_;
>+  packet_time_us_ = o.packet_time_us_;
>+
>+  return *this;
>+}
>+
>+FakeNetworkPipe::FakeNetworkPipe(Clock* clock,
>+                                 const FakeNetworkPipe::Config& config)
>+    : FakeNetworkPipe(clock, config, nullptr, 1) {}
>+
>+FakeNetworkPipe::FakeNetworkPipe(Clock* clock,
>+                                 const FakeNetworkPipe::Config& config,
>+                                 PacketReceiver* receiver)
>+    : FakeNetworkPipe(clock, config, receiver, 1) {}
>+
>+FakeNetworkPipe::FakeNetworkPipe(Clock* clock,
>+                                 const FakeNetworkPipe::Config& config,
>+                                 PacketReceiver* receiver,
>+                                 uint64_t seed)
>+    : clock_(clock),
>+      network_simulation_(absl::make_unique<SimulatedNetwork>(config, seed)),
>+      receiver_(receiver),
>+      transport_(nullptr),
>+      clock_offset_ms_(0),
>+      dropped_packets_(0),
>+      sent_packets_(0),
>+      total_packet_delay_us_(0),
>+      next_process_time_us_(clock_->TimeInMicroseconds()),
>+      last_log_time_us_(clock_->TimeInMicroseconds()) {}
>+
>+FakeNetworkPipe::FakeNetworkPipe(Clock* clock,
>+                                 const FakeNetworkPipe::Config& config,
>+                                 Transport* transport)
>+    : clock_(clock),
>+      network_simulation_(absl::make_unique<SimulatedNetwork>(config, 1)),
>+      receiver_(nullptr),
>+      transport_(transport),
>+      clock_offset_ms_(0),
>+      dropped_packets_(0),
>+      sent_packets_(0),
>+      total_packet_delay_us_(0),
>+      next_process_time_us_(clock_->TimeInMicroseconds()),
>+      last_log_time_us_(clock_->TimeInMicroseconds()) {}
>+
>+FakeNetworkPipe::FakeNetworkPipe(
>+    Clock* clock,
>+    std::unique_ptr<NetworkSimulationInterface> network_simulation)
>+    : FakeNetworkPipe(clock, std::move(network_simulation), nullptr, 1) {}
>+
>+FakeNetworkPipe::FakeNetworkPipe(
>+    Clock* clock,
>+    std::unique_ptr<NetworkSimulationInterface> network_simulation,
>+    PacketReceiver* receiver)
>+    : FakeNetworkPipe(clock, std::move(network_simulation), receiver, 1) {}
>+
>+FakeNetworkPipe::FakeNetworkPipe(
>+    Clock* clock,
>+    std::unique_ptr<NetworkSimulationInterface> network_simulation,
>+    PacketReceiver* receiver,
>+    uint64_t seed)
>+    : clock_(clock),
>+      network_simulation_(std::move(network_simulation)),
>+      receiver_(receiver),
>+      transport_(nullptr),
>+      clock_offset_ms_(0),
>+      dropped_packets_(0),
>+      sent_packets_(0),
>+      total_packet_delay_us_(0),
>+      next_process_time_us_(clock_->TimeInMicroseconds()),
>+      last_log_time_us_(clock_->TimeInMicroseconds()) {}
>+
>+FakeNetworkPipe::FakeNetworkPipe(
>+    Clock* clock,
>+    std::unique_ptr<NetworkSimulationInterface> network_simulation,
>+    Transport* transport)
>+    : clock_(clock),
>+      network_simulation_(std::move(network_simulation)),
>+      receiver_(nullptr),
>+      transport_(transport),
>+      clock_offset_ms_(0),
>+      dropped_packets_(0),
>+      sent_packets_(0),
>+      total_packet_delay_us_(0),
>+      next_process_time_us_(clock_->TimeInMicroseconds()),
>+      last_log_time_us_(clock_->TimeInMicroseconds()) {}
>+
>+FakeNetworkPipe::~FakeNetworkPipe() = default;
>+
>+void FakeNetworkPipe::SetReceiver(PacketReceiver* receiver) {
>+  rtc::CritScope crit(&config_lock_);
>+  receiver_ = receiver;
>+}
>+
>+bool FakeNetworkPipe::SendRtp(const uint8_t* packet,
>+                              size_t length,
>+                              const PacketOptions& options) {
>+  RTC_DCHECK(HasTransport());
>+  EnqueuePacket(rtc::CopyOnWriteBuffer(packet, length), options, false,
>+                MediaType::ANY);
>+  return true;
>+}
>+
>+bool FakeNetworkPipe::SendRtcp(const uint8_t* packet, size_t length) {
>+  RTC_DCHECK(HasTransport());
>+  EnqueuePacket(rtc::CopyOnWriteBuffer(packet, length), absl::nullopt, true,
>+                MediaType::ANY);
>+  return true;
>+}
>+
>+PacketReceiver::DeliveryStatus FakeNetworkPipe::DeliverPacket(
>+    MediaType media_type,
>+    rtc::CopyOnWriteBuffer packet,
>+    int64_t packet_time_us) {
>+  return EnqueuePacket(std::move(packet), absl::nullopt, false, media_type,
>+                       packet_time_us)
>+             ? PacketReceiver::DELIVERY_OK
>+             : PacketReceiver::DELIVERY_PACKET_ERROR;
>+}
>+
>+void FakeNetworkPipe::SetClockOffset(int64_t offset_ms) {
>+  rtc::CritScope crit(&config_lock_);
>+  clock_offset_ms_ = offset_ms;
>+}
>+
>+void FakeNetworkPipe::SetConfig(const FakeNetworkPipe::Config& config) {
>+  network_simulation_->SetConfig(config);
>+}
>+
>+
>+FakeNetworkPipe::StoredPacket::StoredPacket(NetworkPacket&& packet)
>+    : packet(std::move(packet)) {}
>+
>+bool FakeNetworkPipe::EnqueuePacket(rtc::CopyOnWriteBuffer packet,
>+                                    absl::optional<PacketOptions> options,
>+                                    bool is_rtcp,
>+                                    MediaType media_type,
>+                                    absl::optional<int64_t> packet_time_us) {
>+  absl::optional<PacketTime> packet_time;
>+  if (packet_time_us) {
>+    packet_time = PacketTime(*packet_time_us, -1);
>+  }
>+  return EnqueuePacket(packet, options, is_rtcp, media_type, packet_time);
>+}
>+
>+bool FakeNetworkPipe::EnqueuePacket(rtc::CopyOnWriteBuffer packet,
>+                                    absl::optional<PacketOptions> options,
>+                                    bool is_rtcp,
>+                                    MediaType media_type,
>+                                    absl::optional<PacketTime> packet_time) {
>+  int64_t time_now_us = clock_->TimeInMicroseconds();
>+  rtc::CritScope crit(&process_lock_);
>+  size_t packet_size = packet.size();
>+  NetworkPacket net_packet(
>+      std::move(packet), time_now_us, time_now_us, options, is_rtcp, media_type,
>+      packet_time ? absl::optional<int64_t>(packet_time->timestamp)
>+                  : absl::nullopt);
>+
>+  packets_in_flight_.emplace_back(StoredPacket(std::move(net_packet)));
>+  int64_t packet_id = reinterpret_cast<uint64_t>(&packets_in_flight_.back());
>+  bool sent = network_simulation_->EnqueuePacket(
>+      PacketInFlightInfo(packet_size, time_now_us, packet_id));
>+
>+  if (!sent) {
>+    packets_in_flight_.pop_back();
>+    ++dropped_packets_;
>+  }
>+  return sent;
>+}
>+
>+float FakeNetworkPipe::PercentageLoss() {
>+  rtc::CritScope crit(&process_lock_);
>+  if (sent_packets_ == 0)
>+    return 0;
>+
>+  return static_cast<float>(dropped_packets_) /
>+         (sent_packets_ + dropped_packets_);
>+}
>+
>+int FakeNetworkPipe::AverageDelay() {
>+  rtc::CritScope crit(&process_lock_);
>+  if (sent_packets_ == 0)
>+    return 0;
>+
>+  return static_cast<int>(total_packet_delay_us_ /
>+                          (1000 * static_cast<int64_t>(sent_packets_)));
>+}
>+
>+size_t FakeNetworkPipe::DroppedPackets() {
>+  rtc::CritScope crit(&process_lock_);
>+  return dropped_packets_;
>+}
>+
>+size_t FakeNetworkPipe::SentPackets() {
>+  rtc::CritScope crit(&process_lock_);
>+  return sent_packets_;
>+}
>+
>+void FakeNetworkPipe::Process() {
>+  int64_t time_now_us = clock_->TimeInMicroseconds();
>+  std::queue<NetworkPacket> packets_to_deliver;
>+  {
>+    rtc::CritScope crit(&process_lock_);
>+    if (time_now_us - last_log_time_us_ > kLogIntervalMs * 1000) {
>+      int64_t queueing_delay_us = 0;
>+      if (!packets_in_flight_.empty())
>+        queueing_delay_us =
>+            time_now_us - packets_in_flight_.front().packet.send_time();
>+
>+      RTC_LOG(LS_INFO) << "Network queue: " << queueing_delay_us / 1000
>+                       << " ms.";
>+      last_log_time_us_ = time_now_us;
>+    }
>+
>+    std::vector<PacketDeliveryInfo> delivery_infos =
>+        network_simulation_->DequeueDeliverablePackets(time_now_us);
>+    for (auto& delivery_info : delivery_infos) {
>+      // In the common case where no reordering happens, find will return early
>+      // as the first packet will be a match.
>+      auto packet_it =
>+          std::find_if(packets_in_flight_.begin(), packets_in_flight_.end(),
>+                       [&delivery_info](StoredPacket& packet_ref) {
>+                         return reinterpret_cast<uint64_t>(&packet_ref) ==
>+                                delivery_info.packet_id;
>+                       });
>+      // Check that the packet is in the deque of packets in flight.
>+      RTC_CHECK(packet_it != packets_in_flight_.end());
>+      // Check that the packet is not already removed.
>+      RTC_DCHECK(!packet_it->removed);
>+
>+      NetworkPacket packet = std::move(packet_it->packet);
>+      packet_it->removed = true;
>+
>+      // Cleanup of removed packets at the beginning of the deque.
>+      while (!packets_in_flight_.empty() &&
>+             packets_in_flight_.front().removed) {
>+        packets_in_flight_.pop_front();
>+      }
>+
>+      if (delivery_info.receive_time_us != PacketDeliveryInfo::kNotReceived) {
>+        int64_t added_delay_us =
>+            delivery_info.receive_time_us - packet.send_time();
>+        packet.IncrementArrivalTime(added_delay_us);
>+        packets_to_deliver.emplace(std::move(packet));
>+        // |time_now_us| might be later than when the packet should have
>+        // arrived, due to NetworkProcess being called too late. For stats, use
>+        // the time it should have been on the link.
>+        total_packet_delay_us_ += added_delay_us;
>+      }
>+    }
>+    sent_packets_ += packets_to_deliver.size();
>+  }
>+
>+  rtc::CritScope crit(&config_lock_);
>+  while (!packets_to_deliver.empty()) {
>+    NetworkPacket packet = std::move(packets_to_deliver.front());
>+    packets_to_deliver.pop();
>+    DeliverNetworkPacket(&packet);
>+  }
>+  absl::optional<int64_t> delivery_us =
>+      network_simulation_->NextDeliveryTimeUs();
>+  next_process_time_us_ = delivery_us
>+                              ? *delivery_us
>+                              : time_now_us + kDefaultProcessIntervalMs * 1000;
>+}
>+
>+void FakeNetworkPipe::DeliverNetworkPacket(NetworkPacket* packet) {
>+  if (transport_) {
>+    RTC_DCHECK(!receiver_);
>+    if (packet->is_rtcp()) {
>+      transport_->SendRtcp(packet->data(), packet->data_length());
>+    } else {
>+      transport_->SendRtp(packet->data(), packet->data_length(),
>+                          packet->packet_options());
>+    }
>+  } else if (receiver_) {
>+    int64_t packet_time_us = packet->packet_time_us().value_or(-1);
>+    if (packet_time_us != -1) {
>+      int64_t queue_time_us = packet->arrival_time() - packet->send_time();
>+      RTC_CHECK(queue_time_us >= 0);
>+      packet_time_us += queue_time_us;
>+      packet_time_us += (clock_offset_ms_ * 1000);
>+    }
>+    receiver_->DeliverPacket(packet->media_type(),
>+                             std::move(*packet->raw_packet()), packet_time_us);
>+  }
>+}
>+
>+int64_t FakeNetworkPipe::TimeUntilNextProcess() {
>+  rtc::CritScope crit(&process_lock_);
>+  int64_t delay_us = next_process_time_us_ - clock_->TimeInMicroseconds();
>+  return std::max<int64_t>((delay_us + 500) / 1000, 0);
>+}
>+
>+bool FakeNetworkPipe::HasTransport() const {
>+  rtc::CritScope crit(&config_lock_);
>+  return transport_ != nullptr;
>+}
>+bool FakeNetworkPipe::HasReceiver() const {
>+  rtc::CritScope crit(&config_lock_);
>+  return receiver_ != nullptr;
>+}
>+
>+void FakeNetworkPipe::DeliverPacketWithLock(NetworkPacket* packet) {
>+  rtc::CritScope crit(&config_lock_);
>+  DeliverNetworkPacket(packet);
>+}
>+
>+void FakeNetworkPipe::ResetStats() {
>+  rtc::CritScope crit(&process_lock_);
>+  dropped_packets_ = 0;
>+  sent_packets_ = 0;
>+  total_packet_delay_us_ = 0;
>+}
>+
>+void FakeNetworkPipe::AddToPacketDropCount() {
>+  rtc::CritScope crit(&process_lock_);
>+  ++dropped_packets_;
>+}
>+
>+void FakeNetworkPipe::AddToPacketSentCount(int count) {
>+  rtc::CritScope crit(&process_lock_);
>+  sent_packets_ += count;
>+}
>+
>+void FakeNetworkPipe::AddToTotalDelay(int delay_us) {
>+  rtc::CritScope crit(&process_lock_);
>+  total_packet_delay_us_ += delay_us;
>+}
>+
>+int64_t FakeNetworkPipe::GetTimeInMicroseconds() const {
>+  return clock_->TimeInMicroseconds();
>+}
>+
>+bool FakeNetworkPipe::ShouldProcess(int64_t time_now_us) const {
>+  return time_now_us >= next_process_time_us_;
>+}
>+
>+void FakeNetworkPipe::SetTimeToNextProcess(int64_t skip_us) {
>+  next_process_time_us_ += skip_us;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/fake_network_pipe.h b/Source/ThirdParty/libwebrtc/Source/webrtc/call/fake_network_pipe.h
>new file mode 100644
>index 00000000000..1f660b80636
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/fake_network_pipe.h
>@@ -0,0 +1,274 @@
>+/*
>+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef CALL_FAKE_NETWORK_PIPE_H_
>+#define CALL_FAKE_NETWORK_PIPE_H_
>+
>+#include <deque>
>+#include <map>
>+#include <memory>
>+#include <queue>
>+#include <set>
>+#include <string>
>+#include <vector>
>+
>+#include "api/call/transport.h"
>+#include "api/test/simulated_network.h"
>+#include "call/call.h"
>+#include "common_types.h"  // NOLINT(build/include)
>+#include "modules/include/module.h"
>+#include "rtc_base/constructormagic.h"
>+#include "rtc_base/criticalsection.h"
>+#include "rtc_base/thread_annotations.h"
>+
>+namespace webrtc {
>+
>+class Clock;
>+class PacketReceiver;
>+enum class MediaType;
>+
>+class NetworkPacket {
>+ public:
>+  NetworkPacket(rtc::CopyOnWriteBuffer packet,
>+                int64_t send_time,
>+                int64_t arrival_time,
>+                absl::optional<PacketOptions> packet_options,
>+                bool is_rtcp,
>+                MediaType media_type,
>+                absl::optional<int64_t> packet_time_us);
>+  // TODO(nisse): Deprecated.
>+  NetworkPacket(rtc::CopyOnWriteBuffer packet,
>+                int64_t send_time,
>+                int64_t arrival_time,
>+                absl::optional<PacketOptions> packet_options,
>+                bool is_rtcp,
>+                MediaType media_type,
>+                absl::optional<PacketTime> packet_time);
>+
>+  // Disallow copy constructor and copy assignment (no deep copies of |data_|).
>+  NetworkPacket(const NetworkPacket&) = delete;
>+  ~NetworkPacket();
>+  NetworkPacket& operator=(const NetworkPacket&) = delete;
>+  // Allow move constructor/assignment, so that we can use in stl containers.
>+  NetworkPacket(NetworkPacket&&);
>+  NetworkPacket& operator=(NetworkPacket&&);
>+
>+  const uint8_t* data() const { return packet_.data(); }
>+  size_t data_length() const { return packet_.size(); }
>+  rtc::CopyOnWriteBuffer* raw_packet() { return &packet_; }
>+  int64_t send_time() const { return send_time_; }
>+  int64_t arrival_time() const { return arrival_time_; }
>+  void IncrementArrivalTime(int64_t extra_delay) {
>+    arrival_time_ += extra_delay;
>+  }
>+  PacketOptions packet_options() const {
>+    return packet_options_.value_or(PacketOptions());
>+  }
>+  bool is_rtcp() const { return is_rtcp_; }
>+  MediaType media_type() const { return media_type_; }
>+  absl::optional<int64_t> packet_time_us() const { return packet_time_us_; }
>+  // TODO(nisse): Deprecated.
>+  PacketTime packet_time() const {
>+    return PacketTime(packet_time_us_.value_or(-1), -1);
>+  }
>+
>+ private:
>+  rtc::CopyOnWriteBuffer packet_;
>+  // The time the packet was sent out on the network.
>+  int64_t send_time_;
>+  // The time the packet should arrive at the receiver.
>+  int64_t arrival_time_;
>+  // If using a Transport for outgoing degradation, populate with
>+  // PacketOptions (transport-wide sequence number) for RTP.
>+  absl::optional<PacketOptions> packet_options_;
>+  bool is_rtcp_;
>+  // If using a PacketReceiver for incoming degradation, populate with
>+  // appropriate MediaType and PacketTime. This type/timing will be kept and
>+  // forwarded. The PacketTime might be altered to reflect time spent in fake
>+  // network pipe.
>+  MediaType media_type_;
>+  absl::optional<int64_t> packet_time_us_;
>+};
>+
>+// Class faking a network link, internally is uses an implementation of a
>+// SimulatedNetworkInterface to simulate network behavior.
>+class FakeNetworkPipe : public Transport, public PacketReceiver, public Module {
>+ public:
>+  using Config = NetworkSimulationInterface::SimulatedNetworkConfig;
>+
>+  // Deprecated. DO NOT USE. To be removed. Use corresponding version with
>+  // NetworkSimulationInterface instance instead.
>+  // Use these constructors if you plan to insert packets using DeliverPacket().
>+  FakeNetworkPipe(Clock* clock, const FakeNetworkPipe::Config& config);
>+  // Will keep |network_simulation| alive while pipe is alive itself.
>+  // Use these constructors if you plan to insert packets using DeliverPacket().
>+  FakeNetworkPipe(
>+      Clock* clock,
>+      std::unique_ptr<NetworkSimulationInterface> network_simulation);
>+  // Deprecated. DO NOT USE. To be removed. Use corresponding version with
>+  // NetworkSimulationInterface instance instead.
>+  FakeNetworkPipe(Clock* clock,
>+                  const FakeNetworkPipe::Config& config,
>+                  PacketReceiver* receiver);
>+  FakeNetworkPipe(
>+      Clock* clock,
>+      std::unique_ptr<NetworkSimulationInterface> network_simulation,
>+      PacketReceiver* receiver);
>+  // Deprecated. DO NOT USE. To be removed. Use corresponding version with
>+  // NetworkSimulationInterface instance instead.
>+  FakeNetworkPipe(Clock* clock,
>+                  const FakeNetworkPipe::Config& config,
>+                  PacketReceiver* receiver,
>+                  uint64_t seed);
>+  FakeNetworkPipe(
>+      Clock* clock,
>+      std::unique_ptr<NetworkSimulationInterface> network_simulation,
>+      PacketReceiver* receiver,
>+      uint64_t seed);
>+
>+  // Deprecated. DO NOT USE. To be removed. Use corresponding version with
>+  // NetworkSimulationInterface instance instead.
>+  // Use this constructor if you plan to insert packets using SendRt[c?]p().
>+  FakeNetworkPipe(Clock* clock,
>+                  const FakeNetworkPipe::Config& config,
>+                  Transport* transport);
>+  // Use this constructor if you plan to insert packets using SendRt[c?]p().
>+  FakeNetworkPipe(
>+      Clock* clock,
>+      std::unique_ptr<NetworkSimulationInterface> network_simulation,
>+      Transport* transport);
>+
>+  ~FakeNetworkPipe() override;
>+
>+  void SetClockOffset(int64_t offset_ms);
>+
>+  // Deprecated. DO NOT USE. Hold direct reference on NetworkSimulationInterface
>+  // instead and call SetConfig on that object directly. Will be removed soon.
>+  // Sets a new configuration. This won't affect packets already in the pipe.
>+  void SetConfig(const FakeNetworkPipe::Config& config);
>+
>+  // Must not be called in parallel with DeliverPacket or Process.
>+  void SetReceiver(PacketReceiver* receiver);
>+
>+  // Implements Transport interface. When/if packets are delivered, they will
>+  // be passed to the transport instance given in SetReceiverTransport(). These
>+  // methods should only be called if a Transport instance was provided in the
>+  // constructor.
>+  bool SendRtp(const uint8_t* packet,
>+               size_t length,
>+               const PacketOptions& options) override;
>+  bool SendRtcp(const uint8_t* packet, size_t length) override;
>+
>+  // Implements the PacketReceiver interface. When/if packets are delivered,
>+  // they will be passed directly to the receiver instance given in
>+  // SetReceiver(), without passing through a Demuxer. The receive time in
>+  // PacketTime will be increased by the amount of time the packet spent in the
>+  // fake network pipe.
>+  PacketReceiver::DeliveryStatus DeliverPacket(MediaType media_type,
>+                                               rtc::CopyOnWriteBuffer packet,
>+                                               int64_t packet_time_us) override;
>+
>+  // TODO(bugs.webrtc.org/9584): Needed to inherit the alternative signature for
>+  // this method.
>+  using PacketReceiver::DeliverPacket;
>+
>+  // Processes the network queues and trigger PacketReceiver::IncomingPacket for
>+  // packets ready to be delivered.
>+  void Process() override;
>+  int64_t TimeUntilNextProcess() override;
>+
>+  // Get statistics.
>+  float PercentageLoss();
>+  int AverageDelay();
>+  size_t DroppedPackets();
>+  size_t SentPackets();
>+  void ResetStats();
>+
>+ protected:
>+  void DeliverPacketWithLock(NetworkPacket* packet);
>+  void AddToPacketDropCount();
>+  void AddToPacketSentCount(int count);
>+  void AddToTotalDelay(int delay_us);
>+  int64_t GetTimeInMicroseconds() const;
>+  bool ShouldProcess(int64_t time_now_us) const;
>+  void SetTimeToNextProcess(int64_t skip_us);
>+
>+ private:
>+  struct StoredPacket {
>+    NetworkPacket packet;
>+    bool removed = false;
>+    explicit StoredPacket(NetworkPacket&& packet);
>+    StoredPacket(StoredPacket&&) = default;
>+    StoredPacket(const StoredPacket&) = delete;
>+    StoredPacket& operator=(const StoredPacket&) = delete;
>+    StoredPacket() = delete;
>+  };
>+
>+  // Returns true if enqueued, or false if packet was dropped.
>+  virtual bool EnqueuePacket(rtc::CopyOnWriteBuffer packet,
>+                             absl::optional<PacketOptions> options,
>+                             bool is_rtcp,
>+                             MediaType media_type,
>+                             absl::optional<int64_t> packet_time_us);
>+
>+  // TODO(nisse): Deprecated. Delete as soon as overrides in downstream code are
>+  // updated.
>+  virtual bool EnqueuePacket(rtc::CopyOnWriteBuffer packet,
>+                             absl::optional<PacketOptions> options,
>+                             bool is_rtcp,
>+                             MediaType media_type,
>+                             absl::optional<PacketTime> packet_time);
>+  bool EnqueuePacket(rtc::CopyOnWriteBuffer packet,
>+                     absl::optional<PacketOptions> options,
>+                     bool is_rtcp,
>+                     MediaType media_type) {
>+    return EnqueuePacket(packet, options, is_rtcp, media_type,
>+                         absl::optional<PacketTime>());
>+  }
>+  void DeliverNetworkPacket(NetworkPacket* packet)
>+      RTC_EXCLUSIVE_LOCKS_REQUIRED(config_lock_);
>+  bool HasTransport() const;
>+  bool HasReceiver() const;
>+
>+  Clock* const clock_;
>+  // |config_lock| guards the mostly constant things like the callbacks.
>+  rtc::CriticalSection config_lock_;
>+  const std::unique_ptr<NetworkSimulationInterface> network_simulation_;
>+  PacketReceiver* receiver_ RTC_GUARDED_BY(config_lock_);
>+  Transport* const transport_ RTC_GUARDED_BY(config_lock_);
>+
>+  // |process_lock| guards the data structures involved in delay and loss
>+  // processes, such as the packet queues.
>+  rtc::CriticalSection process_lock_;
>+
>+  // Packets  are added at the back of the deque, this makes the deque ordered
>+  // by increasing send time. The common case when removing packets from the
>+  // deque is removing early packets, which will be close to the front of the
>+  // deque. This makes finding the packets in the deque efficient in the common
>+  // case.
>+  std::deque<StoredPacket> packets_in_flight_ RTC_GUARDED_BY(process_lock_);
>+
>+  int64_t clock_offset_ms_ RTC_GUARDED_BY(config_lock_);
>+
>+  // Statistics.
>+  size_t dropped_packets_ RTC_GUARDED_BY(process_lock_);
>+  size_t sent_packets_ RTC_GUARDED_BY(process_lock_);
>+  int64_t total_packet_delay_us_ RTC_GUARDED_BY(process_lock_);
>+
>+  int64_t next_process_time_us_;
>+
>+  int64_t last_log_time_us_;
>+
>+  RTC_DISALLOW_COPY_AND_ASSIGN(FakeNetworkPipe);
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // CALL_FAKE_NETWORK_PIPE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/fake_rtp_transport_controller_send.h b/Source/ThirdParty/libwebrtc/Source/webrtc/call/fake_rtp_transport_controller_send.h
>deleted file mode 100644
>index dda2e5f645c..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/fake_rtp_transport_controller_send.h
>+++ /dev/null
>@@ -1,68 +0,0 @@
>-/*
>- *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef CALL_FAKE_RTP_TRANSPORT_CONTROLLER_SEND_H_
>-#define CALL_FAKE_RTP_TRANSPORT_CONTROLLER_SEND_H_
>-
>-#include "call/rtp_transport_controller_send_interface.h"
>-#include "common_types.h"  // NOLINT(build/include)
>-#include "modules/congestion_controller/include/send_side_congestion_controller.h"
>-#include "modules/pacing/packet_router.h"
>-
>-namespace webrtc {
>-
>-class FakeRtpTransportControllerSend
>-    : public RtpTransportControllerSendInterface {
>- public:
>-  explicit FakeRtpTransportControllerSend(
>-      PacketRouter* packet_router,
>-      PacedSender* paced_sender,
>-      SendSideCongestionController* send_side_cc)
>-      : packet_router_(packet_router),
>-        paced_sender_(paced_sender),
>-        send_side_cc_(send_side_cc) {
>-    RTC_DCHECK(send_side_cc);
>-  }
>-
>-  PacketRouter* packet_router() override { return packet_router_; }
>-
>-  SendSideCongestionController* send_side_cc() override {
>-    return send_side_cc_;
>-  }
>-
>-  TransportFeedbackObserver* transport_feedback_observer() override {
>-    return send_side_cc_;
>-  }
>-
>-  PacedSender* pacer() override { return paced_sender_; }
>-
>-  RtpPacketSender* packet_sender() override { return paced_sender_; }
>-
>-  const RtpKeepAliveConfig& keepalive_config() const override {
>-    return keepalive_;
>-  }
>-
>-  void SetAllocatedSendBitrateLimits(int min_send_bitrate_bps,
>-                                     int max_padding_bitrate_bps) override {}
>-
>-  void set_keepalive_config(const RtpKeepAliveConfig& keepalive_config) {
>-    keepalive_ = keepalive_config;
>-  }
>-
>- private:
>-  PacketRouter* packet_router_;
>-  PacedSender* paced_sender_;
>-  SendSideCongestionController* send_side_cc_;
>-  RtpKeepAliveConfig keepalive_;
>-};
>-
>-}  // namespace webrtc
>-
>-#endif  // CALL_FAKE_RTP_TRANSPORT_CONTROLLER_SEND_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/flexfec_receive_stream.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/flexfec_receive_stream.cc
>new file mode 100644
>index 00000000000..86c000623d3
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/flexfec_receive_stream.cc
>@@ -0,0 +1,21 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "call/flexfec_receive_stream.h"
>+
>+namespace webrtc {
>+
>+FlexfecReceiveStream::Config::Config(Transport* rtcp_send_transport)
>+    : rtcp_send_transport(rtcp_send_transport) {
>+  RTC_DCHECK(rtcp_send_transport);
>+}
>+FlexfecReceiveStream::Config::~Config() = default;
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/flexfec_receive_stream.h b/Source/ThirdParty/libwebrtc/Source/webrtc/call/flexfec_receive_stream.h
>index 98ce3510959..d64cb26f707 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/flexfec_receive_stream.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/flexfec_receive_stream.h
>@@ -16,8 +16,8 @@
> #include <string>
> #include <vector>
> 
>-#include "api/rtp_headers.h"
> #include "api/call/transport.h"
>+#include "api/rtp_headers.h"
> #include "api/rtpparameters.h"
> #include "call/rtp_packet_sink_interface.h"
> #include "common_types.h"  // NOLINT(build/include)
>@@ -36,10 +36,8 @@ class FlexfecReceiveStream : public RtpPacketSinkInterface {
>   };
> 
>   struct Config {
>-    explicit Config(Transport* rtcp_send_transport)
>-        : rtcp_send_transport(rtcp_send_transport) {
>-      RTC_DCHECK(rtcp_send_transport);
>-    }
>+    explicit Config(Transport* rtcp_send_transport);
>+    ~Config();
> 
>     std::string ToString() const;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/flexfec_receive_stream_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/flexfec_receive_stream_impl.cc
>index 038c1f6a1b7..973df661643 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/flexfec_receive_stream_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/flexfec_receive_stream_impl.cc
>@@ -21,19 +21,22 @@
> #include "rtc_base/checks.h"
> #include "rtc_base/location.h"
> #include "rtc_base/logging.h"
>+#include "rtc_base/strings/string_builder.h"
> #include "system_wrappers/include/clock.h"
> 
> namespace webrtc {
> 
> std::string FlexfecReceiveStream::Stats::ToString(int64_t time_ms) const {
>-  std::stringstream ss;
>+  char buf[1024];
>+  rtc::SimpleStringBuilder ss(buf);
>   ss << "FlexfecReceiveStream stats: " << time_ms
>      << ", {flexfec_bitrate_bps: " << flexfec_bitrate_bps << "}";
>   return ss.str();
> }
> 
> std::string FlexfecReceiveStream::Config::ToString() const {
>-  std::stringstream ss;
>+  char buf[1024];
>+  rtc::SimpleStringBuilder ss(buf);
>   ss << "{payload_type: " << payload_type;
>   ss << ", remote_ssrc: " << remote_ssrc;
>   ss << ", local_ssrc: " << local_ssrc;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/flexfec_receive_stream_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/flexfec_receive_stream_unittest.cc
>index 21dbeb7609d..da87c685d1c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/flexfec_receive_stream_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/flexfec_receive_stream_unittest.cc
>@@ -13,6 +13,7 @@
> #include <stdint.h>
> #include <memory>
> 
>+#include "absl/memory/memory.h"
> #include "api/array_view.h"
> #include "call/rtp_stream_receiver_controller.h"
> #include "modules/pacing/packet_router.h"
>@@ -22,7 +23,6 @@
> #include "modules/rtp_rtcp/source/byte_io.h"
> #include "modules/rtp_rtcp/source/rtp_header_extensions.h"
> #include "modules/utility/include/mock/mock_process_thread.h"
>-#include "rtc_base/ptr_util.h"
> #include "test/gmock.h"
> #include "test/gtest.h"
> #include "test/mock_transport.h"
>@@ -84,7 +84,7 @@ class FlexfecReceiveStreamTest : public ::testing::Test {
>   FlexfecReceiveStreamTest()
>       : config_(CreateDefaultConfig(&rtcp_send_transport_)) {
>     EXPECT_CALL(process_thread_, RegisterModule(_, _)).Times(1);
>-    receive_stream_ = rtc::MakeUnique<FlexfecReceiveStreamImpl>(
>+    receive_stream_ = absl::make_unique<FlexfecReceiveStreamImpl>(
>         &rtp_stream_receiver_controller_, config_, &recovered_packet_receiver_,
>         &rtt_stats_, &process_thread_);
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/packet_receiver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/packet_receiver.cc
>new file mode 100644
>index 00000000000..d6786d29a7d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/packet_receiver.cc
>@@ -0,0 +1,31 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "call/packet_receiver.h"
>+
>+namespace webrtc {
>+
>+PacketReceiver::DeliveryStatus PacketReceiver::DeliverPacket(
>+    MediaType media_type,
>+    rtc::CopyOnWriteBuffer packet,
>+    int64_t packet_time_us) {
>+  return DeliverPacket(media_type, packet, PacketTime(packet_time_us, -1));
>+}
>+
>+// TODO(bugs.webrtc.org/9584): Deprecated. Over the transition, default
>+// implementations are used, and subclasses must override one or the other.
>+PacketReceiver::DeliveryStatus PacketReceiver::DeliverPacket(
>+    MediaType media_type,
>+    rtc::CopyOnWriteBuffer packet,
>+    const PacketTime& packet_time) {
>+  return DeliverPacket(media_type, packet, packet_time.timestamp);
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/packet_receiver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/call/packet_receiver.h
>new file mode 100644
>index 00000000000..2a0e3ef7b3b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/packet_receiver.h
>@@ -0,0 +1,48 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#ifndef CALL_PACKET_RECEIVER_H_
>+#define CALL_PACKET_RECEIVER_H_
>+
>+#include <algorithm>
>+#include <memory>
>+#include <string>
>+#include <vector>
>+
>+#include "api/mediatypes.h"
>+#include "common_types.h"  // NOLINT(build/include)
>+#include "rtc_base/copyonwritebuffer.h"
>+
>+namespace webrtc {
>+
>+class PacketReceiver {
>+ public:
>+  enum DeliveryStatus {
>+    DELIVERY_OK,
>+    DELIVERY_UNKNOWN_SSRC,
>+    DELIVERY_PACKET_ERROR,
>+  };
>+
>+  virtual DeliveryStatus DeliverPacket(MediaType media_type,
>+                                       rtc::CopyOnWriteBuffer packet,
>+                                       int64_t packet_time_us);
>+
>+  // TODO(bugs.webrtc.org/9584): Deprecated. Over the transition, default
>+  // implementations are used, and subclasses must override one or the other.
>+  virtual DeliveryStatus DeliverPacket(MediaType media_type,
>+                                       rtc::CopyOnWriteBuffer packet,
>+                                       const PacketTime& packet_time);
>+
>+ protected:
>+  virtual ~PacketReceiver() {}
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // CALL_PACKET_RECEIVER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rampup_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rampup_tests.cc
>index d0354168cfe..fcf32ad4793 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rampup_tests.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rampup_tests.cc
>@@ -13,6 +13,7 @@
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/platform_thread.h"
>+#include "rtc_base/stringencode.h"
> #include "test/encoder_settings.h"
> #include "test/gtest.h"
> #include "test/testsupport/perf_test.h"
>@@ -24,7 +25,9 @@ static const int64_t kPollIntervalMs = 20;
> static const int kExpectedHighVideoBitrateBps = 80000;
> static const int kExpectedHighAudioBitrateBps = 30000;
> static const int kLowBandwidthLimitBps = 20000;
>-static const int kExpectedLowBitrateBps = 20000;
>+// Set target detected bitrate to slightly larger than the target bitrate to
>+// avoid flakiness.
>+static const int kLowBitrateMarginBps = 2000;
> 
> std::vector<uint32_t> GenerateSsrcs(size_t num_streams, uint32_t ssrc_offset) {
>   std::vector<uint32_t> ssrcs;
>@@ -72,16 +75,13 @@ RampUpTester::RampUpTester(size_t num_video_streams,
>   EXPECT_LE(num_audio_streams_, 1u);
> }
> 
>-RampUpTester::~RampUpTester() {
>-}
>+RampUpTester::~RampUpTester() {}
> 
>-Call::Config RampUpTester::GetSenderCallConfig() {
>-  Call::Config call_config(&event_log_);
>+void RampUpTester::ModifySenderCallConfig(Call::Config* config) {
>   if (start_bitrate_bps_ != 0) {
>-    call_config.bitrate_config.start_bitrate_bps = start_bitrate_bps_;
>+    config->bitrate_config.start_bitrate_bps = start_bitrate_bps_;
>   }
>-  call_config.bitrate_config.min_bitrate_bps = 10000;
>-  return call_config;
>+  config->bitrate_config.min_bitrate_bps = 10000;
> }
> 
> void RampUpTester::OnVideoStreamsCreated(
>@@ -217,7 +217,7 @@ void RampUpTester::ModifyVideoConfigs(
>       recv_config.rtp.rtx_ssrc = video_rtx_ssrcs_[i];
>       recv_config.rtp
>           .rtx_associated_payload_types[send_config->rtp.rtx.payload_type] =
>-          send_config->encoder_settings.payload_type;
>+          send_config->rtp.payload_type;
>     }
>     ++i;
>   }
>@@ -436,26 +436,20 @@ void RampUpDownUpTester::PollStats() {
>   } while (!stop_event_.Wait(kPollIntervalMs));
> }
> 
>-Call::Config RampUpDownUpTester::GetReceiverCallConfig() {
>-  Call::Config config(&event_log_);
>-  config.bitrate_config.min_bitrate_bps = 10000;
>-  return config;
>+void RampUpDownUpTester::ModifyReceiverCallConfig(Call::Config* config) {
>+  config->bitrate_config.min_bitrate_bps = 10000;
> }
> 
> std::string RampUpDownUpTester::GetModifierString() const {
>   std::string str("_");
>   if (num_video_streams_ > 0) {
>-    std::ostringstream s;
>-    s << num_video_streams_;
>-    str += s.str();
>+    str += rtc::ToString(num_video_streams_);
>     str += "stream";
>     str += (num_video_streams_ > 1 ? "s" : "");
>     str += "_";
>   }
>   if (num_audio_streams_ > 0) {
>-    std::ostringstream s;
>-    s << num_audio_streams_;
>-    str += s.str();
>+    str += rtc::ToString(num_audio_streams_);
>     str += "stream";
>     str += (num_audio_streams_ > 1 ? "s" : "");
>     str += "_";
>@@ -510,7 +504,7 @@ void RampUpDownUpTester::EvolveTestState(int bitrate_bps, bool suspended) {
>     case kLowRate: {
>       // Audio streams are never suspended.
>       bool check_suspend_state = num_video_streams_ > 0;
>-      if (bitrate_bps < kExpectedLowBitrateBps &&
>+      if (bitrate_bps < kLowBandwidthLimitBps + kLowBitrateMarginBps &&
>           suspended == check_suspend_state) {
>         if (report_perf_stats_) {
>           webrtc::test::PrintResult("ramp_up_down_up", GetModifierString(),
>@@ -561,11 +555,6 @@ void RampUpDownUpTester::EvolveTestState(int bitrate_bps, bool suspended) {
> class RampUpTest : public test::CallTest {
>  public:
>   RampUpTest() {}
>-
>-  virtual ~RampUpTest() {
>-    EXPECT_EQ(nullptr, video_send_stream_);
>-    EXPECT_TRUE(video_receive_streams_.empty());
>-  }
> };
> 
> static const uint32_t kStartBitrateBps = 60000;
>@@ -578,7 +567,15 @@ TEST_F(RampUpTest, UpDownUpAbsSendTimeSimulcastRedRtx) {
>   RunBaseTest(&test);
> }
> 
>-TEST_F(RampUpTest, UpDownUpTransportSequenceNumberRtx) {
>+// TODO(bugs.webrtc.org/8878)
>+#if defined(WEBRTC_MAC)
>+#define MAYBE_UpDownUpTransportSequenceNumberRtx \
>+  DISABLED_UpDownUpTransportSequenceNumberRtx
>+#else
>+#define MAYBE_UpDownUpTransportSequenceNumberRtx \
>+  UpDownUpTransportSequenceNumberRtx
>+#endif
>+TEST_F(RampUpTest, MAYBE_UpDownUpTransportSequenceNumberRtx) {
>   std::vector<int> loss_rates = {0, 0, 0, 0};
>   RampUpDownUpTester test(3, 0, 0, kStartBitrateBps,
>                           RtpExtension::kTransportSequenceNumberUri, true,
>@@ -598,7 +595,15 @@ TEST_F(RampUpTest, DISABLED_UpDownUpTransportSequenceNumberPacketLoss) {
>   RunBaseTest(&test);
> }
> 
>-TEST_F(RampUpTest, UpDownUpAudioVideoTransportSequenceNumberRtx) {
>+// TODO(bugs.webrtc.org/8878)
>+#if defined(WEBRTC_MAC)
>+#define MAYBE_UpDownUpAudioVideoTransportSequenceNumberRtx \
>+  DISABLED_UpDownUpAudioVideoTransportSequenceNumberRtx
>+#else
>+#define MAYBE_UpDownUpAudioVideoTransportSequenceNumberRtx \
>+  UpDownUpAudioVideoTransportSequenceNumberRtx
>+#endif
>+TEST_F(RampUpTest, MAYBE_UpDownUpAudioVideoTransportSequenceNumberRtx) {
>   std::vector<int> loss_rates = {0, 0, 0, 0};
>   RampUpDownUpTester test(3, 1, 0, kStartBitrateBps,
>                           RtpExtension::kTransportSequenceNumberUri, true,
>@@ -650,7 +655,13 @@ TEST_F(RampUpTest, TransportSequenceNumberSimulcastRedRtx) {
>   RunBaseTest(&test);
> }
> 
>-TEST_F(RampUpTest, AudioTransportSequenceNumber) {
>+// TODO(bugs.webrtc.org/8878)
>+#if defined(WEBRTC_MAC)
>+#define MAYBE_AudioTransportSequenceNumber DISABLED_AudioTransportSequenceNumber
>+#else
>+#define MAYBE_AudioTransportSequenceNumber AudioTransportSequenceNumber
>+#endif
>+TEST_F(RampUpTest, MAYBE_AudioTransportSequenceNumber) {
>   RampUpTester test(0, 1, 0, 300000, 10000,
>                     RtpExtension::kTransportSequenceNumberUri, false, false,
>                     false);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rampup_tests.h b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rampup_tests.h
>index 1339f1f2dc7..a22dfc9934c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rampup_tests.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rampup_tests.h
>@@ -62,7 +62,6 @@ class RampUpTester : public test::EndToEndTest {
>                     const std::string& units) const;
>   void TriggerTestDone();
> 
>-  webrtc::RtcEventLogNullImpl event_log_;
>   rtc::Event stop_event_;
>   Clock* const clock_;
>   FakeNetworkPipe::Config forward_transport_config_;
>@@ -80,7 +79,7 @@ class RampUpTester : public test::EndToEndTest {
>   typedef std::map<uint32_t, uint32_t> SsrcMap;
>   class VideoStreamFactory;
> 
>-  Call::Config GetSenderCallConfig() override;
>+  void ModifySenderCallConfig(Call::Config* config) override;
>   void OnVideoStreamsCreated(
>       VideoSendStream* send_stream,
>       const std::vector<VideoReceiveStream*>& receive_streams) override;
>@@ -139,7 +138,7 @@ class RampUpDownUpTester : public RampUpTester {
>     kTransitionToNextState,
>   };
> 
>-  Call::Config GetReceiverCallConfig() override;
>+  void ModifyReceiverCallConfig(Call::Config* config);
> 
>   std::string GetModifierString() const;
>   int GetExpectedHighBitrate() const;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/receive_time_calculator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/receive_time_calculator.cc
>new file mode 100644
>index 00000000000..16c6a43cc3f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/receive_time_calculator.cc
>@@ -0,0 +1,67 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "call/receive_time_calculator.h"
>+#include "absl/memory/memory.h"
>+#include "rtc_base/logging.h"
>+#include "system_wrappers/include/field_trial.h"
>+
>+namespace webrtc {
>+namespace {
>+using ::webrtc::field_trial::FindFullName;
>+using ::webrtc::field_trial::IsEnabled;
>+
>+const char kBweReceiveTimeCorrection[] = "WebRTC-BweReceiveTimeCorrection";
>+}  // namespace
>+
>+ReceiveTimeCalculator::ReceiveTimeCalculator(int64_t min_delta_ms,
>+                                             int64_t max_delta_diff_ms)
>+    : min_delta_us_(min_delta_ms * 1000),
>+      max_delta_diff_us_(max_delta_diff_ms * 1000) {}
>+
>+std::unique_ptr<ReceiveTimeCalculator>
>+ReceiveTimeCalculator::CreateFromFieldTrial() {
>+  if (!IsEnabled(kBweReceiveTimeCorrection))
>+    return nullptr;
>+  int min, max;
>+  if (sscanf(FindFullName(kBweReceiveTimeCorrection).c_str(), "Enabled,%d,%d",
>+             &min, &max) != 2) {
>+    RTC_LOG(LS_WARNING) << "Invalid number of parameters provided.";
>+    return nullptr;
>+  }
>+  return absl::make_unique<ReceiveTimeCalculator>(min, max);
>+}
>+
>+int64_t ReceiveTimeCalculator::ReconcileReceiveTimes(int64_t packet_time_us_,
>+                                                     int64_t safe_time_us_) {
>+  if (!receive_time_offset_us_) {
>+    receive_time_offset_us_ = safe_time_us_ - packet_time_us_;
>+  } else {
>+    int64_t safe_delta_us = safe_time_us_ - last_safe_time_us_;
>+    int64_t packet_delta_us_ = packet_time_us_ - last_packet_time_us_;
>+    int64_t delta_diff = packet_delta_us_ - safe_delta_us;
>+    // Packet time should not decrease significantly, a large decrease indicates
>+    // a reset of the packet time clock and we should reset the offest
>+    // parameter. The safe reference time can increase in large jumps if the
>+    // thread measuring it is backgrounded for longer periods. But if the packet
>+    // time increases significantly more than the safe time, it indicates a
>+    // clock reset and we should reset the offset.
>+
>+    if (packet_delta_us_ < min_delta_us_ || delta_diff > max_delta_diff_us_) {
>+      RTC_LOG(LS_WARNING) << "Received a clock jump of " << delta_diff
>+                          << " resetting offset";
>+      receive_time_offset_us_ = safe_time_us_ - packet_time_us_;
>+    }
>+  }
>+  last_packet_time_us_ = packet_time_us_;
>+  last_safe_time_us_ = safe_time_us_;
>+  return packet_time_us_ + *receive_time_offset_us_;
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/receive_time_calculator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/call/receive_time_calculator.h
>new file mode 100644
>index 00000000000..a8217cd1012
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/receive_time_calculator.h
>@@ -0,0 +1,47 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#ifndef CALL_RECEIVE_TIME_CALCULATOR_H_
>+#define CALL_RECEIVE_TIME_CALCULATOR_H_
>+
>+#include <stdint.h>
>+#include <memory>
>+
>+#include "absl/types/optional.h"
>+
>+namespace webrtc {
>+
>+// The receive time calculator serves the purpose of combining packet time
>+// stamps with a safely incremental clock. This assumes that the packet time
>+// stamps are based on lower layer timestamps that have more accurate time
>+// increments since they are based on the exact receive time. They might
>+// however, have large jumps due to clock resets in the system. To compensate
>+// this they are combined with a safe clock source that is guaranteed to be
>+// consistent, but it will not be able to measure the exact time when a packet
>+// is received.
>+class ReceiveTimeCalculator {
>+ public:
>+  static std::unique_ptr<ReceiveTimeCalculator> CreateFromFieldTrial();
>+  // The min delta is used to correct for jumps backwards in time, to allow some
>+  // packet reordering a small negative value is appropriate to use. The max
>+  // delta difference is used as margin when detecting when packet time
>+  // increases more than the safe clock. This should be larger than the largest
>+  // expected sysmtem induced delay in the safe clock timestamp.
>+  ReceiveTimeCalculator(int64_t min_delta_ms, int64_t max_delta_diff_ms);
>+  int64_t ReconcileReceiveTimes(int64_t packet_time_us_, int64_t safe_time_us_);
>+
>+ private:
>+  const int64_t min_delta_us_;
>+  const int64_t max_delta_diff_us_;
>+  absl::optional<int64_t> receive_time_offset_us_;
>+  int64_t last_packet_time_us_ = 0;
>+  int64_t last_safe_time_us_ = 0;
>+};
>+}  // namespace webrtc
>+#endif  // CALL_RECEIVE_TIME_CALCULATOR_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/receive_time_calculator_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/receive_time_calculator_unittest.cc
>new file mode 100644
>index 00000000000..92d5a279eae
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/receive_time_calculator_unittest.cc
>@@ -0,0 +1,74 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "call/receive_time_calculator.h"
>+
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+namespace test {
>+namespace {
>+
>+int64_t ReconcileMs(ReceiveTimeCalculator* calc,
>+                    int64_t packet_time_ms,
>+                    int64_t safe_time_ms) {
>+  return calc->ReconcileReceiveTimes(packet_time_ms * 1000,
>+                                     safe_time_ms * 1000) /
>+         1000;
>+}
>+}  // namespace
>+
>+TEST(ReceiveTimeCalculatorTest, UsesSmallerIncrements) {
>+  int64_t kMinDeltaMs = -20;
>+  int64_t kMaxDeltaDiffMs = 100;
>+  ReceiveTimeCalculator calc(kMinDeltaMs, kMaxDeltaDiffMs);
>+  // Initialize offset.
>+  ReconcileMs(&calc, 10000, 20000);
>+
>+  EXPECT_EQ(ReconcileMs(&calc, 10010, 20100), 20010);
>+  EXPECT_EQ(ReconcileMs(&calc, 10020, 20100), 20020);
>+  EXPECT_EQ(ReconcileMs(&calc, 10030, 20100), 20030);
>+
>+  EXPECT_EQ(ReconcileMs(&calc, 10110, 20200), 20110);
>+  EXPECT_EQ(ReconcileMs(&calc, 10120, 20200), 20120);
>+  EXPECT_EQ(ReconcileMs(&calc, 10130, 20200), 20130);
>+
>+  // Small jumps backwards are let trough, they might be due to reordering.
>+  EXPECT_EQ(ReconcileMs(&calc, 10120, 20200), 20120);
>+  // The safe clock might be smaller than the packet clock.
>+  EXPECT_EQ(ReconcileMs(&calc, 10210, 20200), 20210);
>+  EXPECT_EQ(ReconcileMs(&calc, 10240, 20200), 20240);
>+}
>+
>+TEST(ReceiveTimeCalculatorTest, CorrectsJumps) {
>+  int64_t kMinDeltaMs = -20;
>+  int64_t kMaxDeltaDiffMs = 100;
>+  ReceiveTimeCalculator calc(kMinDeltaMs, kMaxDeltaDiffMs);
>+  // Initialize offset.
>+  ReconcileMs(&calc, 10000, 20000);
>+
>+  EXPECT_EQ(ReconcileMs(&calc, 10010, 20100), 20010);
>+  EXPECT_EQ(ReconcileMs(&calc, 10020, 20100), 20020);
>+  EXPECT_EQ(ReconcileMs(&calc, 10030, 20100), 20030);
>+
>+  // Jump forward in time.
>+  EXPECT_EQ(ReconcileMs(&calc, 10240, 20200), 20200);
>+  EXPECT_EQ(ReconcileMs(&calc, 10250, 20200), 20210);
>+  EXPECT_EQ(ReconcileMs(&calc, 10260, 20200), 20220);
>+
>+  // Jump backward in time.
>+  EXPECT_EQ(ReconcileMs(&calc, 10230, 20300), 20300);
>+  EXPECT_EQ(ReconcileMs(&calc, 10240, 20300), 20310);
>+  EXPECT_EQ(ReconcileMs(&calc, 10250, 20300), 20320);
>+}
>+
>+}  // namespace test
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtcp_demuxer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtcp_demuxer.cc
>index 0e78ddea186..d441c05b3db 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtcp_demuxer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtcp_demuxer.cc
>@@ -35,7 +35,7 @@ void RtcpDemuxer::AddSink(uint32_t sender_ssrc, RtcpPacketSinkInterface* sink) {
> 
> void RtcpDemuxer::AddSink(const std::string& rsid,
>                           RtcpPacketSinkInterface* sink) {
>-  RTC_DCHECK(StreamId::IsLegalName(rsid));
>+  RTC_DCHECK(StreamId::IsLegalRsidName(rsid));
>   RTC_DCHECK(sink);
>   RTC_DCHECK(!ContainerHasKey(broadcast_sinks_, sink));
>   RTC_DCHECK(!MultimapAssociationExists(rsid_sinks_, rsid, sink));
>@@ -66,7 +66,7 @@ void RtcpDemuxer::RemoveBroadcastSink(const RtcpPacketSinkInterface* sink) {
> 
> void RtcpDemuxer::OnRtcpPacket(rtc::ArrayView<const uint8_t> packet) {
>   // Perform sender-SSRC-based demuxing for packets with a sender-SSRC.
>-  rtc::Optional<uint32_t> sender_ssrc = ParseRtcpPacketSenderSsrc(packet);
>+  absl::optional<uint32_t> sender_ssrc = ParseRtcpPacketSenderSsrc(packet);
>   if (sender_ssrc) {
>     auto it_range = ssrc_sinks_.equal_range(*sender_ssrc);
>     for (auto it = it_range.first; it != it_range.second; ++it) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtcp_demuxer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtcp_demuxer.h
>index 87b5816282a..494e0cea4b5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtcp_demuxer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtcp_demuxer.h
>@@ -17,7 +17,6 @@
> 
> #include "api/array_view.h"
> #include "call/ssrc_binding_observer.h"
>-#include "rtc_base/basictypes.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtcp_demuxer_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtcp_demuxer_unittest.cc
>index 0e1c95b9f5c..808364acc5e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtcp_demuxer_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtcp_demuxer_unittest.cc
>@@ -13,14 +13,13 @@
> #include <memory>
> #include <set>
> 
>+#include "absl/memory/memory.h"
> #include "api/rtp_headers.h"
> #include "call/rtcp_packet_sink_interface.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/rtp_rtcp/source/rtcp_packet/bye.h"
> #include "rtc_base/arraysize.h"
>-#include "rtc_base/basictypes.h"
> #include "rtc_base/checks.h"
>-#include "rtc_base/ptr_util.h"
> #include "test/gmock.h"
> #include "test/gtest.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_bitrate_configurator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_bitrate_configurator.cc
>new file mode 100644
>index 00000000000..fafd8ec9972
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_bitrate_configurator.cc
>@@ -0,0 +1,107 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "call/rtp_bitrate_configurator.h"
>+
>+#include <algorithm>
>+
>+#include "rtc_base/checks.h"
>+
>+namespace webrtc {
>+RtpBitrateConfigurator::RtpBitrateConfigurator(
>+    const BitrateConstraints& bitrate_config)
>+    : bitrate_config_(bitrate_config), base_bitrate_config_(bitrate_config) {
>+  RTC_DCHECK_GE(bitrate_config.min_bitrate_bps, 0);
>+  RTC_DCHECK_GE(bitrate_config.start_bitrate_bps,
>+                bitrate_config.min_bitrate_bps);
>+  if (bitrate_config.max_bitrate_bps != -1) {
>+    RTC_DCHECK_GE(bitrate_config.max_bitrate_bps,
>+                  bitrate_config.start_bitrate_bps);
>+  }
>+}
>+
>+RtpBitrateConfigurator::~RtpBitrateConfigurator() = default;
>+
>+BitrateConstraints RtpBitrateConfigurator::GetConfig() const {
>+  return bitrate_config_;
>+}
>+
>+absl::optional<BitrateConstraints>
>+RtpBitrateConfigurator::UpdateWithSdpParameters(
>+    const BitrateConstraints& bitrate_config) {
>+  RTC_DCHECK_GE(bitrate_config.min_bitrate_bps, 0);
>+  RTC_DCHECK_NE(bitrate_config.start_bitrate_bps, 0);
>+  if (bitrate_config.max_bitrate_bps != -1) {
>+    RTC_DCHECK_GT(bitrate_config.max_bitrate_bps, 0);
>+  }
>+
>+  absl::optional<int> new_start;
>+  // Only update the "start" bitrate if it's set, and different from the old
>+  // value. In practice, this value comes from the x-google-start-bitrate codec
>+  // parameter in SDP, and setting the same remote description twice shouldn't
>+  // restart bandwidth estimation.
>+  if (bitrate_config.start_bitrate_bps != -1 &&
>+      bitrate_config.start_bitrate_bps !=
>+          base_bitrate_config_.start_bitrate_bps) {
>+    new_start.emplace(bitrate_config.start_bitrate_bps);
>+  }
>+  base_bitrate_config_ = bitrate_config;
>+  return UpdateConstraints(new_start);
>+}
>+
>+absl::optional<BitrateConstraints>
>+RtpBitrateConfigurator::UpdateWithClientPreferences(
>+    const BitrateSettings& bitrate_mask) {
>+  bitrate_config_mask_ = bitrate_mask;
>+  return UpdateConstraints(bitrate_mask.start_bitrate_bps);
>+}
>+
>+absl::optional<BitrateConstraints> RtpBitrateConfigurator::UpdateConstraints(
>+    const absl::optional<int>& new_start) {
>+  BitrateConstraints updated;
>+  updated.min_bitrate_bps =
>+      std::max(bitrate_config_mask_.min_bitrate_bps.value_or(0),
>+               base_bitrate_config_.min_bitrate_bps);
>+
>+  updated.max_bitrate_bps =
>+      MinPositive(bitrate_config_mask_.max_bitrate_bps.value_or(-1),
>+                  base_bitrate_config_.max_bitrate_bps);
>+
>+  // If the combined min ends up greater than the combined max, the max takes
>+  // priority.
>+  if (updated.max_bitrate_bps != -1 &&
>+      updated.min_bitrate_bps > updated.max_bitrate_bps) {
>+    updated.min_bitrate_bps = updated.max_bitrate_bps;
>+  }
>+
>+  // If there is nothing to update (min/max unchanged, no new bandwidth
>+  // estimation start value), return early.
>+  if (updated.min_bitrate_bps == bitrate_config_.min_bitrate_bps &&
>+      updated.max_bitrate_bps == bitrate_config_.max_bitrate_bps &&
>+      !new_start) {
>+    return absl::nullopt;
>+  }
>+
>+  if (new_start) {
>+    // Clamp start by min and max.
>+    updated.start_bitrate_bps = MinPositive(
>+        std::max(*new_start, updated.min_bitrate_bps), updated.max_bitrate_bps);
>+  } else {
>+    updated.start_bitrate_bps = -1;
>+  }
>+  BitrateConstraints config_to_return = updated;
>+  if (!new_start) {
>+    updated.start_bitrate_bps = bitrate_config_.start_bitrate_bps;
>+  }
>+  bitrate_config_ = updated;
>+  return config_to_return;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_bitrate_configurator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_bitrate_configurator.h
>new file mode 100644
>index 00000000000..07fe8d41a65
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_bitrate_configurator.h
>@@ -0,0 +1,69 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef CALL_RTP_BITRATE_CONFIGURATOR_H_
>+#define CALL_RTP_BITRATE_CONFIGURATOR_H_
>+
>+#include "api/bitrate_constraints.h"
>+#include "api/transport/bitrate_settings.h"
>+#include "rtc_base/constructormagic.h"
>+
>+namespace webrtc {
>+
>+// RtpBitrateConfigurator calculates the bitrate configuration based on received
>+// remote configuration combined with local overrides.
>+class RtpBitrateConfigurator {
>+ public:
>+  explicit RtpBitrateConfigurator(const BitrateConstraints& bitrate_config);
>+  ~RtpBitrateConfigurator();
>+  BitrateConstraints GetConfig() const;
>+
>+  // The greater min and smaller max set by this and SetClientBitratePreferences
>+  // will be used. The latest non-negative start value from either call will be
>+  // used. Specifying a start bitrate (>0) will reset the current bitrate
>+  // estimate. This is due to how the 'x-google-start-bitrate' flag is currently
>+  // implemented. Passing -1 leaves the start bitrate unchanged. Behavior is not
>+  // guaranteed for other negative values or 0.
>+  // The optional return value is set with new configuration if it was updated.
>+  absl::optional<BitrateConstraints> UpdateWithSdpParameters(
>+      const BitrateConstraints& bitrate_config_);
>+
>+  // The greater min and smaller max set by this and SetSdpBitrateParameters
>+  // will be used. The latest non-negative start value form either call will be
>+  // used. Specifying a start bitrate will reset the current bitrate estimate.
>+  // Assumes 0 <= min <= start <= max holds for set parameters.
>+  // Update the bitrate configuration
>+  // The optional return value is set with new configuration if it was updated.
>+  absl::optional<BitrateConstraints> UpdateWithClientPreferences(
>+      const BitrateSettings& bitrate_mask);
>+
>+ private:
>+  // Applies update to the BitrateConstraints cached in |config_|, resetting
>+  // with |new_start| if set.
>+  absl::optional<BitrateConstraints> UpdateConstraints(
>+      const absl::optional<int>& new_start);
>+
>+  // Bitrate config used until valid bitrate estimates are calculated. Also
>+  // used to cap total bitrate used. This comes from the remote connection.
>+  BitrateConstraints bitrate_config_;
>+
>+  // The config mask set by SetClientBitratePreferences.
>+  // 0 <= min <= start <= max
>+  BitrateSettings bitrate_config_mask_;
>+
>+  // The config set by SetSdpBitrateParameters.
>+  // min >= 0, start != 0, max == -1 || max > 0
>+  BitrateConstraints base_bitrate_config_;
>+
>+  RTC_DISALLOW_COPY_AND_ASSIGN(RtpBitrateConfigurator);
>+};
>+}  // namespace webrtc
>+
>+#endif  // CALL_RTP_BITRATE_CONFIGURATOR_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_bitrate_configurator_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_bitrate_configurator_unittest.cc
>new file mode 100644
>index 00000000000..b177db7a8f6
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_bitrate_configurator_unittest.cc
>@@ -0,0 +1,299 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#include <memory>
>+
>+#include "call/rtp_bitrate_configurator.h"
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+using absl::nullopt;
>+
>+class RtpBitrateConfiguratorTest : public testing::Test {
>+ public:
>+  RtpBitrateConfiguratorTest()
>+      : configurator_(new RtpBitrateConfigurator(BitrateConstraints())) {}
>+  std::unique_ptr<RtpBitrateConfigurator> configurator_;
>+  void UpdateConfigMatches(BitrateConstraints bitrate_config,
>+                           absl::optional<int> min_bitrate_bps,
>+                           absl::optional<int> start_bitrate_bps,
>+                           absl::optional<int> max_bitrate_bps) {
>+    absl::optional<BitrateConstraints> result =
>+        configurator_->UpdateWithSdpParameters(bitrate_config);
>+    EXPECT_TRUE(result.has_value());
>+    if (start_bitrate_bps.has_value())
>+      EXPECT_EQ(result->start_bitrate_bps, start_bitrate_bps);
>+    if (min_bitrate_bps.has_value())
>+      EXPECT_EQ(result->min_bitrate_bps, min_bitrate_bps);
>+    if (max_bitrate_bps.has_value())
>+      EXPECT_EQ(result->max_bitrate_bps, max_bitrate_bps);
>+  }
>+
>+  void UpdateMaskMatches(BitrateSettings bitrate_mask,
>+                         absl::optional<int> min_bitrate_bps,
>+                         absl::optional<int> start_bitrate_bps,
>+                         absl::optional<int> max_bitrate_bps) {
>+    absl::optional<BitrateConstraints> result =
>+        configurator_->UpdateWithClientPreferences(bitrate_mask);
>+    EXPECT_TRUE(result.has_value());
>+    if (start_bitrate_bps.has_value())
>+      EXPECT_EQ(result->start_bitrate_bps, start_bitrate_bps);
>+    if (min_bitrate_bps.has_value())
>+      EXPECT_EQ(result->min_bitrate_bps, min_bitrate_bps);
>+    if (max_bitrate_bps.has_value())
>+      EXPECT_EQ(result->max_bitrate_bps, max_bitrate_bps);
>+  }
>+};
>+
>+TEST_F(RtpBitrateConfiguratorTest, NewConfigWithValidConfigReturnsNewConfig) {
>+  BitrateConstraints bitrate_config;
>+  bitrate_config.min_bitrate_bps = 1;
>+  bitrate_config.start_bitrate_bps = 2;
>+  bitrate_config.max_bitrate_bps = 3;
>+
>+  UpdateConfigMatches(bitrate_config, 1, 2, 3);
>+}
>+
>+TEST_F(RtpBitrateConfiguratorTest, NewConfigWithDifferentMinReturnsNewConfig) {
>+  BitrateConstraints bitrate_config;
>+  bitrate_config.min_bitrate_bps = 10;
>+  bitrate_config.start_bitrate_bps = 20;
>+  bitrate_config.max_bitrate_bps = 30;
>+  configurator_.reset(new RtpBitrateConfigurator(bitrate_config));
>+
>+  bitrate_config.min_bitrate_bps = 11;
>+  UpdateConfigMatches(bitrate_config, 11, -1, 30);
>+}
>+
>+TEST_F(RtpBitrateConfiguratorTest,
>+       NewConfigWithDifferentStartReturnsNewConfig) {
>+  BitrateConstraints bitrate_config;
>+  bitrate_config.min_bitrate_bps = 10;
>+  bitrate_config.start_bitrate_bps = 20;
>+  bitrate_config.max_bitrate_bps = 30;
>+  configurator_.reset(new RtpBitrateConfigurator(bitrate_config));
>+
>+  bitrate_config.start_bitrate_bps = 21;
>+  UpdateConfigMatches(bitrate_config, 10, 21, 30);
>+}
>+
>+TEST_F(RtpBitrateConfiguratorTest, NewConfigWithDifferentMaxReturnsNewConfig) {
>+  BitrateConstraints bitrate_config;
>+  bitrate_config.min_bitrate_bps = 10;
>+  bitrate_config.start_bitrate_bps = 20;
>+  bitrate_config.max_bitrate_bps = 30;
>+  configurator_.reset(new RtpBitrateConfigurator(bitrate_config));
>+
>+  bitrate_config.max_bitrate_bps = 31;
>+  UpdateConfigMatches(bitrate_config, 10, -1, 31);
>+}
>+
>+TEST_F(RtpBitrateConfiguratorTest, NewConfigWithSameConfigElidesSecondCall) {
>+  BitrateConstraints bitrate_config;
>+  bitrate_config.min_bitrate_bps = 1;
>+  bitrate_config.start_bitrate_bps = 2;
>+  bitrate_config.max_bitrate_bps = 3;
>+
>+  UpdateConfigMatches(bitrate_config, 1, 2, 3);
>+  EXPECT_FALSE(
>+      configurator_->UpdateWithSdpParameters(bitrate_config).has_value());
>+}
>+
>+TEST_F(RtpBitrateConfiguratorTest,
>+       NewConfigWithSameMinMaxAndNegativeStartElidesSecondCall) {
>+  BitrateConstraints bitrate_config;
>+  bitrate_config.min_bitrate_bps = 1;
>+  bitrate_config.start_bitrate_bps = 2;
>+  bitrate_config.max_bitrate_bps = 3;
>+
>+  UpdateConfigMatches(bitrate_config, 1, 2, 3);
>+
>+  bitrate_config.start_bitrate_bps = -1;
>+  EXPECT_FALSE(
>+      configurator_->UpdateWithSdpParameters(bitrate_config).has_value());
>+}
>+
>+TEST_F(RtpBitrateConfiguratorTest, BiggerMaskMinUsed) {
>+  BitrateSettings mask;
>+  mask.min_bitrate_bps = 1234;
>+  UpdateMaskMatches(mask, *mask.min_bitrate_bps, nullopt, nullopt);
>+}
>+
>+TEST_F(RtpBitrateConfiguratorTest, BiggerConfigMinUsed) {
>+  BitrateSettings mask;
>+  mask.min_bitrate_bps = 1000;
>+  UpdateMaskMatches(mask, 1000, nullopt, nullopt);
>+
>+  BitrateConstraints config;
>+  config.min_bitrate_bps = 1234;
>+  UpdateConfigMatches(config, 1234, nullopt, nullopt);
>+}
>+
>+// The last call to set start should be used.
>+TEST_F(RtpBitrateConfiguratorTest, LatestStartMaskPreferred) {
>+  BitrateSettings mask;
>+  mask.start_bitrate_bps = 1300;
>+  UpdateMaskMatches(mask, nullopt, *mask.start_bitrate_bps, nullopt);
>+
>+  BitrateConstraints bitrate_config;
>+  bitrate_config.start_bitrate_bps = 1200;
>+
>+  UpdateConfigMatches(bitrate_config, nullopt, bitrate_config.start_bitrate_bps,
>+                      nullopt);
>+}
>+
>+TEST_F(RtpBitrateConfiguratorTest, SmallerMaskMaxUsed) {
>+  BitrateConstraints bitrate_config;
>+  bitrate_config.max_bitrate_bps = bitrate_config.start_bitrate_bps + 2000;
>+  configurator_.reset(new RtpBitrateConfigurator(bitrate_config));
>+
>+  BitrateSettings mask;
>+  mask.max_bitrate_bps = bitrate_config.start_bitrate_bps + 1000;
>+
>+  UpdateMaskMatches(mask, nullopt, nullopt, *mask.max_bitrate_bps);
>+}
>+
>+TEST_F(RtpBitrateConfiguratorTest, SmallerConfigMaxUsed) {
>+  BitrateConstraints bitrate_config;
>+  bitrate_config.max_bitrate_bps = bitrate_config.start_bitrate_bps + 1000;
>+  configurator_.reset(new RtpBitrateConfigurator(bitrate_config));
>+
>+  BitrateSettings mask;
>+  mask.max_bitrate_bps = bitrate_config.start_bitrate_bps + 2000;
>+
>+  // Expect no return because nothing changes
>+  EXPECT_FALSE(configurator_->UpdateWithClientPreferences(mask).has_value());
>+}
>+
>+TEST_F(RtpBitrateConfiguratorTest, MaskStartLessThanConfigMinClamped) {
>+  BitrateConstraints bitrate_config;
>+  bitrate_config.min_bitrate_bps = 2000;
>+  configurator_.reset(new RtpBitrateConfigurator(bitrate_config));
>+
>+  BitrateSettings mask;
>+  mask.start_bitrate_bps = 1000;
>+  UpdateMaskMatches(mask, 2000, 2000, nullopt);
>+}
>+
>+TEST_F(RtpBitrateConfiguratorTest, MaskStartGreaterThanConfigMaxClamped) {
>+  BitrateConstraints bitrate_config;
>+  bitrate_config.start_bitrate_bps = 2000;
>+  configurator_.reset(new RtpBitrateConfigurator(bitrate_config));
>+
>+  BitrateSettings mask;
>+  mask.max_bitrate_bps = 1000;
>+
>+  UpdateMaskMatches(mask, nullopt, -1, 1000);
>+}
>+
>+TEST_F(RtpBitrateConfiguratorTest, MaskMinGreaterThanConfigMaxClamped) {
>+  BitrateConstraints bitrate_config;
>+  bitrate_config.min_bitrate_bps = 2000;
>+  configurator_.reset(new RtpBitrateConfigurator(bitrate_config));
>+
>+  BitrateSettings mask;
>+  mask.max_bitrate_bps = 1000;
>+
>+  UpdateMaskMatches(mask, 1000, nullopt, 1000);
>+}
>+
>+TEST_F(RtpBitrateConfiguratorTest, SettingMaskStartForcesUpdate) {
>+  BitrateSettings mask;
>+  mask.start_bitrate_bps = 1000;
>+
>+  // Config should be returned twice with the same params since
>+  // start_bitrate_bps is set.
>+  UpdateMaskMatches(mask, nullopt, 1000, nullopt);
>+  UpdateMaskMatches(mask, nullopt, 1000, nullopt);
>+}
>+
>+TEST_F(RtpBitrateConfiguratorTest, NewConfigWithNoChangesDoesNotCallNewConfig) {
>+  BitrateConstraints config1;
>+  config1.min_bitrate_bps = 0;
>+  config1.start_bitrate_bps = 1000;
>+  config1.max_bitrate_bps = -1;
>+
>+  BitrateConstraints config2;
>+  config2.min_bitrate_bps = 0;
>+  config2.start_bitrate_bps = -1;
>+  config2.max_bitrate_bps = -1;
>+
>+  // The second call should not return anything because it doesn't
>+  // change any values.
>+  UpdateConfigMatches(config1, 0, 1000, -1);
>+  EXPECT_FALSE(configurator_->UpdateWithSdpParameters(config2).has_value());
>+}
>+
>+// If config changes the max, but not the effective max,
>+// new config shouldn't be returned, to avoid unnecessary encoder
>+// reconfigurations.
>+TEST_F(RtpBitrateConfiguratorTest,
>+       NewConfigNotReturnedWhenEffectiveMaxUnchanged) {
>+  BitrateConstraints config;
>+  config.min_bitrate_bps = 0;
>+  config.start_bitrate_bps = -1;
>+  config.max_bitrate_bps = 2000;
>+  UpdateConfigMatches(config, nullopt, nullopt, 2000);
>+
>+  // Reduce effective max to 1000 with the mask.
>+  BitrateSettings mask;
>+  mask.max_bitrate_bps = 1000;
>+  UpdateMaskMatches(mask, nullopt, nullopt, 1000);
>+
>+  // This leaves the effective max unchanged, so new config shouldn't be
>+  // returned again.
>+  config.max_bitrate_bps = 1000;
>+  EXPECT_FALSE(configurator_->UpdateWithSdpParameters(config).has_value());
>+}
>+
>+// When the "start bitrate" mask is removed, new config shouldn't be returned
>+// again, since nothing's changing.
>+TEST_F(RtpBitrateConfiguratorTest, NewConfigNotReturnedWhenStartMaskRemoved) {
>+  BitrateSettings mask;
>+  mask.start_bitrate_bps = 1000;
>+  UpdateMaskMatches(mask, 0, 1000, -1);
>+
>+  mask.start_bitrate_bps.reset();
>+  EXPECT_FALSE(configurator_->UpdateWithClientPreferences(mask).has_value());
>+}
>+
>+// Test that if a new config is returned after BitrateSettings applies a
>+// "start" value, the new config won't return that start value a
>+// second time.
>+TEST_F(RtpBitrateConfiguratorTest, NewConfigAfterBitrateConfigMaskWithStart) {
>+  BitrateSettings mask;
>+  mask.start_bitrate_bps = 1000;
>+  UpdateMaskMatches(mask, 0, 1000, -1);
>+
>+  BitrateConstraints config;
>+  config.min_bitrate_bps = 0;
>+  config.start_bitrate_bps = -1;
>+  config.max_bitrate_bps = 5000;
>+  // The start value isn't changing, so new config should be returned with
>+  // -1.
>+  UpdateConfigMatches(config, 0, -1, 5000);
>+}
>+
>+TEST_F(RtpBitrateConfiguratorTest,
>+       NewConfigNotReturnedWhenClampedMinUnchanged) {
>+  BitrateConstraints bitrate_config;
>+  bitrate_config.start_bitrate_bps = 500;
>+  bitrate_config.max_bitrate_bps = 1000;
>+  configurator_.reset(new RtpBitrateConfigurator(bitrate_config));
>+
>+  // Set min to 2000; it is clamped to the max (1000).
>+  BitrateSettings mask;
>+  mask.min_bitrate_bps = 2000;
>+  UpdateMaskMatches(mask, 1000, -1, 1000);
>+
>+  // Set min to 3000; the clamped value stays the same so nothing happens.
>+  mask.min_bitrate_bps = 3000;
>+  EXPECT_FALSE(configurator_->UpdateWithClientPreferences(mask).has_value());
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_config.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_config.cc
>index 3621f728903..1445c2552e4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_config.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_config.cc
>@@ -10,19 +10,21 @@
> 
> #include "call/rtp_config.h"
> 
>-#include <sstream>
>+#include "rtc_base/strings/string_builder.h"
> 
> namespace webrtc {
> 
> std::string NackConfig::ToString() const {
>-  std::stringstream ss;
>+  char buf[1024];
>+  rtc::SimpleStringBuilder ss(buf);
>   ss << "{rtp_history_ms: " << rtp_history_ms;
>   ss << '}';
>   return ss.str();
> }
> 
> std::string UlpfecConfig::ToString() const {
>-  std::stringstream ss;
>+  char buf[1024];
>+  rtc::SimpleStringBuilder ss(buf);
>   ss << "{ulpfec_payload_type: " << ulpfec_payload_type;
>   ss << ", red_payload_type: " << red_payload_type;
>   ss << ", red_rtx_payload_type: " << red_rtx_payload_type;
>@@ -35,4 +37,89 @@ bool UlpfecConfig::operator==(const UlpfecConfig& other) const {
>          red_payload_type == other.red_payload_type &&
>          red_rtx_payload_type == other.red_rtx_payload_type;
> }
>+
>+RtpConfig::RtpConfig() = default;
>+RtpConfig::RtpConfig(const RtpConfig&) = default;
>+RtpConfig::~RtpConfig() = default;
>+
>+RtpConfig::Flexfec::Flexfec() = default;
>+RtpConfig::Flexfec::Flexfec(const Flexfec&) = default;
>+RtpConfig::Flexfec::~Flexfec() = default;
>+
>+std::string RtpConfig::ToString() const {
>+  char buf[2 * 1024];
>+  rtc::SimpleStringBuilder ss(buf);
>+  ss << "{ssrcs: [";
>+  for (size_t i = 0; i < ssrcs.size(); ++i) {
>+    ss << ssrcs[i];
>+    if (i != ssrcs.size() - 1)
>+      ss << ", ";
>+  }
>+  ss << ']';
>+  ss << ", rtcp_mode: "
>+     << (rtcp_mode == RtcpMode::kCompound ? "RtcpMode::kCompound"
>+                                          : "RtcpMode::kReducedSize");
>+  ss << ", max_packet_size: " << max_packet_size;
>+  ss << ", extensions: [";
>+  for (size_t i = 0; i < extensions.size(); ++i) {
>+    ss << extensions[i].ToString();
>+    if (i != extensions.size() - 1)
>+      ss << ", ";
>+  }
>+  ss << ']';
>+
>+  ss << ", nack: {rtp_history_ms: " << nack.rtp_history_ms << '}';
>+  ss << ", ulpfec: " << ulpfec.ToString();
>+  ss << ", payload_name: " << payload_name;
>+  ss << ", payload_type: " << payload_type;
>+
>+  ss << ", flexfec: {payload_type: " << flexfec.payload_type;
>+  ss << ", ssrc: " << flexfec.ssrc;
>+  ss << ", protected_media_ssrcs: [";
>+  for (size_t i = 0; i < flexfec.protected_media_ssrcs.size(); ++i) {
>+    ss << flexfec.protected_media_ssrcs[i];
>+    if (i != flexfec.protected_media_ssrcs.size() - 1)
>+      ss << ", ";
>+  }
>+  ss << "]}";
>+
>+  ss << ", rtx: " << rtx.ToString();
>+  ss << ", c_name: " << c_name;
>+  ss << '}';
>+  return ss.str();
>+}
>+
>+RtpConfig::Rtx::Rtx() = default;
>+RtpConfig::Rtx::Rtx(const Rtx&) = default;
>+RtpConfig::Rtx::~Rtx() = default;
>+
>+std::string RtpConfig::Rtx::ToString() const {
>+  char buf[1024];
>+  rtc::SimpleStringBuilder ss(buf);
>+  ss << "{ssrcs: [";
>+  for (size_t i = 0; i < ssrcs.size(); ++i) {
>+    ss << ssrcs[i];
>+    if (i != ssrcs.size() - 1)
>+      ss << ", ";
>+  }
>+  ss << ']';
>+
>+  ss << ", payload_type: " << payload_type;
>+  ss << '}';
>+  return ss.str();
>+}
>+
>+RtcpConfig::RtcpConfig() = default;
>+RtcpConfig::RtcpConfig(const RtcpConfig&) = default;
>+RtcpConfig::~RtcpConfig() = default;
>+
>+std::string RtcpConfig::ToString() const {
>+  char buf[1024];
>+  rtc::SimpleStringBuilder ss(buf);
>+  ss << "{video_report_interval_ms: " << video_report_interval_ms;
>+  ss << ", audio_report_interval_ms: " << audio_report_interval_ms;
>+  ss << '}';
>+  return ss.str();
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_config.h b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_config.h
>index 86d32ac104d..8f8d03a3a82 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_config.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_config.h
>@@ -12,8 +12,18 @@
> #define CALL_RTP_CONFIG_H_
> 
> #include <string>
>+#include <vector>
>+
>+#include "api/rtp_headers.h"
>+#include "api/rtpparameters.h"
> 
> namespace webrtc {
>+// Currently only VP8/VP9 specific.
>+struct RtpPayloadState {
>+  int16_t picture_id = -1;
>+  uint8_t tl0_pic_idx = 0;
>+  int64_t shared_frame_id = 0;
>+};
> // Settings for NACK, see RFC 4585 for details.
> struct NackConfig {
>   NackConfig() : rtp_history_ms(0) {}
>@@ -44,5 +54,92 @@ struct UlpfecConfig {
>   // RTX payload type for RED payload.
>   int red_rtx_payload_type;
> };
>+
>+static const size_t kDefaultMaxPacketSize = 1500 - 40;  // TCP over IPv4.
>+struct RtpConfig {
>+  RtpConfig();
>+  RtpConfig(const RtpConfig&);
>+  ~RtpConfig();
>+  std::string ToString() const;
>+
>+  std::vector<uint32_t> ssrcs;
>+
>+  // The value to send in the MID RTP header extension if the extension is
>+  // included in the list of extensions.
>+  std::string mid;
>+
>+  // See RtcpMode for description.
>+  RtcpMode rtcp_mode = RtcpMode::kCompound;
>+
>+  // Max RTP packet size delivered to send transport from VideoEngine.
>+  size_t max_packet_size = kDefaultMaxPacketSize;
>+
>+  // RTP header extensions to use for this send stream.
>+  std::vector<RtpExtension> extensions;
>+
>+  // TODO(nisse): For now, these are fixed, but we'd like to support
>+  // changing codec without recreating the VideoSendStream. Then these
>+  // fields must be removed, and association between payload type and codec
>+  // must move above the per-stream level. Ownership could be with
>+  // RtpTransportControllerSend, with a reference from PayloadRouter, where
>+  // the latter would be responsible for mapping the codec type of encoded
>+  // images to the right payload type.
>+  std::string payload_name;
>+  int payload_type = -1;
>+
>+  // See NackConfig for description.
>+  NackConfig nack;
>+
>+  // See UlpfecConfig for description.
>+  UlpfecConfig ulpfec;
>+
>+  struct Flexfec {
>+    Flexfec();
>+    Flexfec(const Flexfec&);
>+    ~Flexfec();
>+    // Payload type of FlexFEC. Set to -1 to disable sending FlexFEC.
>+    int payload_type = -1;
>+
>+    // SSRC of FlexFEC stream.
>+    uint32_t ssrc = 0;
>+
>+    // Vector containing a single element, corresponding to the SSRC of the
>+    // media stream being protected by this FlexFEC stream.
>+    // The vector MUST have size 1.
>+    //
>+    // TODO(brandtr): Update comment above when we support
>+    // multistream protection.
>+    std::vector<uint32_t> protected_media_ssrcs;
>+  } flexfec;
>+
>+  // Settings for RTP retransmission payload format, see RFC 4588 for
>+  // details.
>+  struct Rtx {
>+    Rtx();
>+    Rtx(const Rtx&);
>+    ~Rtx();
>+    std::string ToString() const;
>+    // SSRCs to use for the RTX streams.
>+    std::vector<uint32_t> ssrcs;
>+
>+    // Payload type to use for the RTX stream.
>+    int payload_type = -1;
>+  } rtx;
>+
>+  // RTCP CNAME, see RFC 3550.
>+  std::string c_name;
>+};
>+
>+struct RtcpConfig {
>+  RtcpConfig();
>+  RtcpConfig(const RtcpConfig&);
>+  ~RtcpConfig();
>+  std::string ToString() const;
>+
>+  // Time interval between RTCP report for video
>+  int64_t video_report_interval_ms = 1000;
>+  // Time interval between RTCP report for audio
>+  int64_t audio_report_interval_ms = 5000;
>+};
> }  // namespace webrtc
> #endif  // CALL_RTP_CONFIG_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_demuxer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_demuxer.cc
>index 6a9cae8f77a..8f0e2e529d2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_demuxer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_demuxer.cc
>@@ -38,8 +38,8 @@ bool RtpDemuxer::AddSink(const RtpDemuxerCriteria& criteria,
>                          RtpPacketSinkInterface* sink) {
>   RTC_DCHECK(!criteria.payload_types.empty() || !criteria.ssrcs.empty() ||
>              !criteria.mid.empty() || !criteria.rsid.empty());
>-  RTC_DCHECK(criteria.mid.empty() || Mid::IsLegalName(criteria.mid));
>-  RTC_DCHECK(criteria.rsid.empty() || StreamId::IsLegalName(criteria.rsid));
>+  RTC_DCHECK(criteria.mid.empty() || Mid::IsLegalMidName(criteria.mid));
>+  RTC_DCHECK(criteria.rsid.empty() || StreamId::IsLegalRsidName(criteria.rsid));
>   RTC_DCHECK(sink);
> 
>   // We return false instead of DCHECKing for logical conflicts with the new
>@@ -168,7 +168,7 @@ RtpPacketSinkInterface* RtpDemuxer::ResolveSink(
>   // RSID and RRID are routed to the same sinks. If an RSID is specified on a
>   // repair packet, it should be ignored and the RRID should be used.
>   std::string packet_mid, packet_rsid;
>-  bool has_mid = packet.GetExtension<RtpMid>(&packet_mid);
>+  bool has_mid = use_mid_ && packet.GetExtension<RtpMid>(&packet_mid);
>   bool has_rsid = packet.GetExtension<RepairedRtpStreamId>(&packet_rsid);
>   if (!has_rsid) {
>     has_rsid = packet.GetExtension<RtpStreamId>(&packet_rsid);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_demuxer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_demuxer.h
>index 971c1516480..0a8acc27bb6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_demuxer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_demuxer.h
>@@ -137,6 +137,10 @@ class RtpDemuxer {
>   // Deprecated: Use the above method.
>   void DeregisterRsidResolutionObserver(const SsrcBindingObserver* observer);
> 
>+  // Configure whether to look at the MID header extension when demuxing
>+  // incoming RTP packets. By default this is enabled.
>+  void set_use_mid(bool use_mid) { use_mid_ = use_mid; }
>+
>  private:
>   // Returns true if adding a sink with the given criteria would cause conflicts
>   // with the existing criteria and should be rejected.
>@@ -197,6 +201,8 @@ class RtpDemuxer {
>   // Observers which will be notified when an RSID association to an SSRC is
>   // resolved by this object.
>   std::vector<SsrcBindingObserver*> ssrc_binding_observers_;
>+
>+  bool use_mid_ = true;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_demuxer_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_demuxer_unittest.cc
>index ef092ee98f4..b5ede3cd03a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_demuxer_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_demuxer_unittest.cc
>@@ -14,6 +14,7 @@
> #include <set>
> #include <string>
> 
>+#include "absl/memory/memory.h"
> #include "call/ssrc_binding_observer.h"
> #include "call/test/mock_rtp_packet_sink_interface.h"
> #include "common_types.h"  // NOLINT(build/include)
>@@ -21,10 +22,8 @@
> #include "modules/rtp_rtcp/source/rtp_header_extensions.h"
> #include "modules/rtp_rtcp/source/rtp_packet_received.h"
> #include "rtc_base/arraysize.h"
>-#include "rtc_base/basictypes.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/numerics/safe_conversions.h"
>-#include "rtc_base/ptr_util.h"
> #include "test/gmock.h"
> #include "test/gtest.h"
> 
>@@ -136,7 +135,7 @@ class RtpDemuxerTest : public testing::Test {
>   std::unique_ptr<RtpPacketReceived> CreatePacket(
>       uint32_t ssrc,
>       RtpPacketReceived::ExtensionManager* extension_manager) {
>-    auto packet = rtc::MakeUnique<RtpPacketReceived>(extension_manager);
>+    auto packet = absl::make_unique<RtpPacketReceived>(extension_manager);
>     packet->SetSsrc(ssrc);
>     packet->SetSequenceNumber(next_sequence_number_++);
>     return packet;
>@@ -347,8 +346,8 @@ TEST_F(RtpDemuxerTest, OnRtpPacketCalledOnCorrectSinkByRsid) {
>   }
> 
>   for (size_t i = 0; i < arraysize(rsids); i++) {
>-    auto packet = CreatePacketWithSsrcRsid(rtc::checked_cast<uint32_t>(i),
>-                                           rsids[i]);
>+    auto packet =
>+        CreatePacketWithSsrcRsid(rtc::checked_cast<uint32_t>(i), rsids[i]);
>     EXPECT_CALL(sinks[i], OnRtpPacket(SamePacketAs(*packet))).Times(1);
>     EXPECT_TRUE(demuxer_.OnRtpPacket(*packet));
>   }
>@@ -362,8 +361,8 @@ TEST_F(RtpDemuxerTest, OnRtpPacketCalledOnCorrectSinkByMid) {
>   }
> 
>   for (size_t i = 0; i < arraysize(mids); i++) {
>-    auto packet = CreatePacketWithSsrcMid(rtc::checked_cast<uint32_t>(i),
>-                                          mids[i]);
>+    auto packet =
>+        CreatePacketWithSsrcMid(rtc::checked_cast<uint32_t>(i), mids[i]);
>     EXPECT_CALL(sinks[i], OnRtpPacket(SamePacketAs(*packet))).Times(1);
>     EXPECT_TRUE(demuxer_.OnRtpPacket(*packet));
>   }
>@@ -1492,9 +1491,9 @@ TEST_F(RtpDemuxerTest, RsidMustBeAlphaNumeric) {
>   EXPECT_DEATH(AddSinkOnlyRsid("a_3", &sink), "");
> }
> 
>-TEST_F(RtpDemuxerTest, MidMustBeAlphaNumeric) {
>+TEST_F(RtpDemuxerTest, MidMustBeToken) {
>   MockRtpPacketSink sink;
>-  EXPECT_DEATH(AddSinkOnlyMid("a_3", &sink), "");
>+  EXPECT_DEATH(AddSinkOnlyMid("a(3)", &sink), "");
> }
> 
> TEST_F(RtpDemuxerTest, RsidMustNotExceedMaximumLength) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_payload_params.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_payload_params.cc
>new file mode 100644
>index 00000000000..d61d40fc8bb
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_payload_params.cc
>@@ -0,0 +1,183 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "call/rtp_payload_params.h"
>+
>+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>+#include "modules/video_coding/include/video_codec_interface.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/random.h"
>+#include "rtc_base/timeutils.h"
>+
>+namespace webrtc {
>+
>+namespace {
>+void PopulateRtpWithCodecSpecifics(const CodecSpecificInfo& info,
>+                                   RTPVideoHeader* rtp) {
>+  rtp->codec = info.codecType;
>+  switch (info.codecType) {
>+    case kVideoCodecVP8: {
>+      rtp->vp8().InitRTPVideoHeaderVP8();
>+      rtp->vp8().nonReference = info.codecSpecific.VP8.nonReference;
>+      rtp->vp8().temporalIdx = info.codecSpecific.VP8.temporalIdx;
>+      rtp->vp8().layerSync = info.codecSpecific.VP8.layerSync;
>+      rtp->vp8().keyIdx = info.codecSpecific.VP8.keyIdx;
>+      rtp->simulcastIdx = info.codecSpecific.VP8.simulcastIdx;
>+      return;
>+    }
>+    case kVideoCodecVP9: {
>+      auto& vp9_header = rtp->video_type_header.emplace<RTPVideoHeaderVP9>();
>+      vp9_header.InitRTPVideoHeaderVP9();
>+      vp9_header.inter_pic_predicted =
>+          info.codecSpecific.VP9.inter_pic_predicted;
>+      vp9_header.flexible_mode = info.codecSpecific.VP9.flexible_mode;
>+      vp9_header.ss_data_available = info.codecSpecific.VP9.ss_data_available;
>+      vp9_header.non_ref_for_inter_layer_pred =
>+          info.codecSpecific.VP9.non_ref_for_inter_layer_pred;
>+      vp9_header.temporal_idx = info.codecSpecific.VP9.temporal_idx;
>+      vp9_header.spatial_idx = info.codecSpecific.VP9.spatial_idx;
>+      vp9_header.temporal_up_switch = info.codecSpecific.VP9.temporal_up_switch;
>+      vp9_header.inter_layer_predicted =
>+          info.codecSpecific.VP9.inter_layer_predicted;
>+      vp9_header.gof_idx = info.codecSpecific.VP9.gof_idx;
>+      vp9_header.num_spatial_layers = info.codecSpecific.VP9.num_spatial_layers;
>+
>+      if (info.codecSpecific.VP9.ss_data_available) {
>+        vp9_header.spatial_layer_resolution_present =
>+            info.codecSpecific.VP9.spatial_layer_resolution_present;
>+        if (info.codecSpecific.VP9.spatial_layer_resolution_present) {
>+          for (size_t i = 0; i < info.codecSpecific.VP9.num_spatial_layers;
>+               ++i) {
>+            vp9_header.width[i] = info.codecSpecific.VP9.width[i];
>+            vp9_header.height[i] = info.codecSpecific.VP9.height[i];
>+          }
>+        }
>+        vp9_header.gof.CopyGofInfoVP9(info.codecSpecific.VP9.gof);
>+      }
>+
>+      vp9_header.num_ref_pics = info.codecSpecific.VP9.num_ref_pics;
>+      for (int i = 0; i < info.codecSpecific.VP9.num_ref_pics; ++i) {
>+        vp9_header.pid_diff[i] = info.codecSpecific.VP9.p_diff[i];
>+      }
>+      vp9_header.end_of_picture = info.codecSpecific.VP9.end_of_picture;
>+      return;
>+    }
>+    case kVideoCodecH264: {
>+      auto& h264_header = rtp->video_type_header.emplace<RTPVideoHeaderH264>();
>+      h264_header.packetization_mode =
>+          info.codecSpecific.H264.packetization_mode;
>+      rtp->simulcastIdx = info.codecSpecific.H264.simulcast_idx;
>+      return;
>+    }
>+    case kVideoCodecMultiplex:
>+    case kVideoCodecGeneric:
>+      rtp->codec = kVideoCodecGeneric;
>+      rtp->simulcastIdx = info.codecSpecific.generic.simulcast_idx;
>+      return;
>+    default:
>+      return;
>+  }
>+}
>+
>+void SetVideoTiming(const EncodedImage& image, VideoSendTiming* timing) {
>+  if (image.timing_.flags == VideoSendTiming::TimingFrameFlags::kInvalid ||
>+      image.timing_.flags == VideoSendTiming::TimingFrameFlags::kNotTriggered) {
>+    timing->flags = VideoSendTiming::TimingFrameFlags::kInvalid;
>+    return;
>+  }
>+
>+  timing->encode_start_delta_ms = VideoSendTiming::GetDeltaCappedMs(
>+      image.capture_time_ms_, image.timing_.encode_start_ms);
>+  timing->encode_finish_delta_ms = VideoSendTiming::GetDeltaCappedMs(
>+      image.capture_time_ms_, image.timing_.encode_finish_ms);
>+  timing->packetization_finish_delta_ms = 0;
>+  timing->pacer_exit_delta_ms = 0;
>+  timing->network_timestamp_delta_ms = 0;
>+  timing->network2_timestamp_delta_ms = 0;
>+  timing->flags = image.timing_.flags;
>+}
>+}  // namespace
>+
>+RtpPayloadParams::RtpPayloadParams(const uint32_t ssrc,
>+                                   const RtpPayloadState* state)
>+    : ssrc_(ssrc) {
>+  Random random(rtc::TimeMicros());
>+  state_.picture_id =
>+      state ? state->picture_id : (random.Rand<int16_t>() & 0x7FFF);
>+  state_.tl0_pic_idx = state ? state->tl0_pic_idx : (random.Rand<uint8_t>());
>+}
>+RtpPayloadParams::~RtpPayloadParams() {}
>+
>+RTPVideoHeader RtpPayloadParams::GetRtpVideoHeader(
>+    const EncodedImage& image,
>+    const CodecSpecificInfo* codec_specific_info) {
>+  RTPVideoHeader rtp_video_header;
>+  if (codec_specific_info) {
>+    PopulateRtpWithCodecSpecifics(*codec_specific_info, &rtp_video_header);
>+  }
>+  rtp_video_header.rotation = image.rotation_;
>+  rtp_video_header.content_type = image.content_type_;
>+  rtp_video_header.playout_delay = image.playout_delay_;
>+
>+  SetVideoTiming(image, &rtp_video_header.video_timing);
>+
>+  // Sets picture id and tl0 pic idx.
>+  const bool first_frame_in_picture =
>+      (codec_specific_info && codec_specific_info->codecType == kVideoCodecVP9)
>+          ? codec_specific_info->codecSpecific.VP9.first_frame_in_picture
>+          : true;
>+  Set(&rtp_video_header, first_frame_in_picture);
>+  return rtp_video_header;
>+}
>+
>+uint32_t RtpPayloadParams::ssrc() const {
>+  return ssrc_;
>+}
>+
>+RtpPayloadState RtpPayloadParams::state() const {
>+  return state_;
>+}
>+
>+void RtpPayloadParams::Set(RTPVideoHeader* rtp_video_header,
>+                           bool first_frame_in_picture) {
>+  // Always set picture id. Set tl0_pic_idx iff temporal index is set.
>+  if (first_frame_in_picture) {
>+    state_.picture_id = (static_cast<uint16_t>(state_.picture_id) + 1) & 0x7FFF;
>+  }
>+  if (rtp_video_header->codec == kVideoCodecVP8) {
>+    rtp_video_header->vp8().pictureId = state_.picture_id;
>+
>+    if (rtp_video_header->vp8().temporalIdx != kNoTemporalIdx) {
>+      if (rtp_video_header->vp8().temporalIdx == 0) {
>+        ++state_.tl0_pic_idx;
>+      }
>+      rtp_video_header->vp8().tl0PicIdx = state_.tl0_pic_idx;
>+    }
>+  }
>+  if (rtp_video_header->codec == kVideoCodecVP9) {
>+    auto& vp9_header =
>+        absl::get<RTPVideoHeaderVP9>(rtp_video_header->video_type_header);
>+    vp9_header.picture_id = state_.picture_id;
>+
>+    // Note that in the case that we have no temporal layers but we do have
>+    // spatial layers, packets will carry layering info with a temporal_idx of
>+    // zero, and we then have to set and increment tl0_pic_idx.
>+    if (vp9_header.temporal_idx != kNoTemporalIdx ||
>+        vp9_header.spatial_idx != kNoSpatialIdx) {
>+      if (first_frame_in_picture &&
>+          (vp9_header.temporal_idx == 0 ||
>+           vp9_header.temporal_idx == kNoTemporalIdx)) {
>+        ++state_.tl0_pic_idx;
>+      }
>+      vp9_header.tl0_pic_idx = state_.tl0_pic_idx;
>+    }
>+  }
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_payload_params.h b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_payload_params.h
>new file mode 100644
>index 00000000000..0c71a7b5f55
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_payload_params.h
>@@ -0,0 +1,49 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef CALL_RTP_PAYLOAD_PARAMS_H_
>+#define CALL_RTP_PAYLOAD_PARAMS_H_
>+
>+#include <map>
>+#include <vector>
>+
>+#include "api/video_codecs/video_encoder.h"
>+#include "call/rtp_config.h"
>+#include "common_types.h"  // NOLINT(build/include)
>+#include "modules/rtp_rtcp/source/rtp_video_header.h"
>+
>+namespace webrtc {
>+
>+class RTPFragmentationHeader;
>+class RtpRtcp;
>+
>+// State for setting picture id and tl0 pic idx, for VP8 and VP9
>+// TODO(nisse): Make these properties not codec specific.
>+class RtpPayloadParams final {
>+ public:
>+  RtpPayloadParams(const uint32_t ssrc, const RtpPayloadState* state);
>+  ~RtpPayloadParams();
>+
>+  RTPVideoHeader GetRtpVideoHeader(
>+      const EncodedImage& image,
>+      const CodecSpecificInfo* codec_specific_info);
>+
>+  uint32_t ssrc() const;
>+
>+  RtpPayloadState state() const;
>+
>+ private:
>+  void Set(RTPVideoHeader* rtp_video_header, bool first_frame_in_picture);
>+
>+  const uint32_t ssrc_;
>+  RtpPayloadState state_;
>+};
>+}  // namespace webrtc
>+#endif  // CALL_RTP_PAYLOAD_PARAMS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_payload_params_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_payload_params_unittest.cc
>new file mode 100644
>index 00000000000..b2339cd3b52
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_payload_params_unittest.cc
>@@ -0,0 +1,256 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <memory>
>+
>+#include "call/rtp_payload_params.h"
>+#include "modules/video_coding/include/video_codec_interface.h"
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+namespace {
>+const uint32_t kSsrc1 = 12345;
>+const uint32_t kSsrc2 = 23456;
>+const int16_t kPictureId = 123;
>+const int16_t kTl0PicIdx = 20;
>+const uint8_t kTemporalIdx = 1;
>+const int16_t kInitialPictureId1 = 222;
>+const int16_t kInitialTl0PicIdx1 = 99;
>+}  // namespace
>+
>+TEST(RtpPayloadParamsTest, InfoMappedToRtpVideoHeader_Vp8) {
>+  RtpPayloadState state2;
>+  state2.picture_id = kPictureId;
>+  state2.tl0_pic_idx = kTl0PicIdx;
>+  std::map<uint32_t, RtpPayloadState> states = {{kSsrc2, state2}};
>+
>+  RtpPayloadParams params(kSsrc2, &state2);
>+  EncodedImage encoded_image;
>+  encoded_image.rotation_ = kVideoRotation_90;
>+  encoded_image.content_type_ = VideoContentType::SCREENSHARE;
>+
>+  CodecSpecificInfo codec_info;
>+  memset(&codec_info, 0, sizeof(CodecSpecificInfo));
>+  codec_info.codecType = kVideoCodecVP8;
>+  codec_info.codecSpecific.VP8.simulcastIdx = 1;
>+  codec_info.codecSpecific.VP8.temporalIdx = kTemporalIdx;
>+  codec_info.codecSpecific.VP8.keyIdx = kNoKeyIdx;
>+  codec_info.codecSpecific.VP8.layerSync = true;
>+  codec_info.codecSpecific.VP8.nonReference = true;
>+
>+  RTPVideoHeader header = params.GetRtpVideoHeader(encoded_image, &codec_info);
>+
>+  EXPECT_EQ(kVideoRotation_90, header.rotation);
>+  EXPECT_EQ(VideoContentType::SCREENSHARE, header.content_type);
>+  EXPECT_EQ(1, header.simulcastIdx);
>+  EXPECT_EQ(kVideoCodecVP8, header.codec);
>+  EXPECT_EQ(kPictureId + 1, header.vp8().pictureId);
>+  EXPECT_EQ(kTemporalIdx, header.vp8().temporalIdx);
>+  EXPECT_EQ(kTl0PicIdx, header.vp8().tl0PicIdx);
>+  EXPECT_EQ(kNoKeyIdx, header.vp8().keyIdx);
>+  EXPECT_TRUE(header.vp8().layerSync);
>+  EXPECT_TRUE(header.vp8().nonReference);
>+}
>+
>+TEST(RtpPayloadParamsTest, InfoMappedToRtpVideoHeader_Vp9) {
>+  RtpPayloadState state;
>+  state.picture_id = kPictureId;
>+  state.tl0_pic_idx = kTl0PicIdx;
>+  RtpPayloadParams params(kSsrc1, &state);
>+
>+  EncodedImage encoded_image;
>+  encoded_image.rotation_ = kVideoRotation_90;
>+  encoded_image.content_type_ = VideoContentType::SCREENSHARE;
>+
>+  CodecSpecificInfo codec_info;
>+  memset(&codec_info, 0, sizeof(CodecSpecificInfo));
>+  codec_info.codecType = kVideoCodecVP9;
>+  codec_info.codecSpecific.VP9.num_spatial_layers = 3;
>+  codec_info.codecSpecific.VP9.first_frame_in_picture = true;
>+  codec_info.codecSpecific.VP9.spatial_idx = 0;
>+  codec_info.codecSpecific.VP9.temporal_idx = 2;
>+  codec_info.codecSpecific.VP9.end_of_picture = false;
>+
>+  RTPVideoHeader header = params.GetRtpVideoHeader(encoded_image, &codec_info);
>+
>+  EXPECT_EQ(kVideoRotation_90, header.rotation);
>+  EXPECT_EQ(VideoContentType::SCREENSHARE, header.content_type);
>+  EXPECT_EQ(kVideoCodecVP9, header.codec);
>+  const auto& vp9_header =
>+      absl::get<RTPVideoHeaderVP9>(header.video_type_header);
>+  EXPECT_EQ(kPictureId + 1, vp9_header.picture_id);
>+  EXPECT_EQ(kTl0PicIdx, vp9_header.tl0_pic_idx);
>+  EXPECT_EQ(vp9_header.temporal_idx, codec_info.codecSpecific.VP9.temporal_idx);
>+  EXPECT_EQ(vp9_header.spatial_idx, codec_info.codecSpecific.VP9.spatial_idx);
>+  EXPECT_EQ(vp9_header.num_spatial_layers,
>+            codec_info.codecSpecific.VP9.num_spatial_layers);
>+  EXPECT_EQ(vp9_header.end_of_picture,
>+            codec_info.codecSpecific.VP9.end_of_picture);
>+
>+  // Next spatial layer.
>+  codec_info.codecSpecific.VP9.first_frame_in_picture = false;
>+  codec_info.codecSpecific.VP9.spatial_idx += 1;
>+  codec_info.codecSpecific.VP9.end_of_picture = true;
>+
>+  header = params.GetRtpVideoHeader(encoded_image, &codec_info);
>+
>+  EXPECT_EQ(kVideoRotation_90, header.rotation);
>+  EXPECT_EQ(VideoContentType::SCREENSHARE, header.content_type);
>+  EXPECT_EQ(kVideoCodecVP9, header.codec);
>+  EXPECT_EQ(kPictureId + 1, vp9_header.picture_id);
>+  EXPECT_EQ(kTl0PicIdx, vp9_header.tl0_pic_idx);
>+  EXPECT_EQ(vp9_header.temporal_idx, codec_info.codecSpecific.VP9.temporal_idx);
>+  EXPECT_EQ(vp9_header.spatial_idx, codec_info.codecSpecific.VP9.spatial_idx);
>+  EXPECT_EQ(vp9_header.num_spatial_layers,
>+            codec_info.codecSpecific.VP9.num_spatial_layers);
>+  EXPECT_EQ(vp9_header.end_of_picture,
>+            codec_info.codecSpecific.VP9.end_of_picture);
>+}
>+
>+TEST(RtpPayloadParamsTest, InfoMappedToRtpVideoHeader_H264) {
>+  RtpPayloadParams params(kSsrc1, {});
>+
>+  EncodedImage encoded_image;
>+  CodecSpecificInfo codec_info;
>+  memset(&codec_info, 0, sizeof(CodecSpecificInfo));
>+  codec_info.codecType = kVideoCodecH264;
>+  codec_info.codecSpecific.H264.packetization_mode =
>+      H264PacketizationMode::SingleNalUnit;
>+
>+  RTPVideoHeader header = params.GetRtpVideoHeader(encoded_image, &codec_info);
>+
>+  EXPECT_EQ(0, header.simulcastIdx);
>+  EXPECT_EQ(kVideoCodecH264, header.codec);
>+  const auto& h264 = absl::get<RTPVideoHeaderH264>(header.video_type_header);
>+  EXPECT_EQ(H264PacketizationMode::SingleNalUnit, h264.packetization_mode);
>+}
>+
>+TEST(RtpPayloadParamsTest, PictureIdIsSetForVp8) {
>+  RtpPayloadState state;
>+  state.picture_id = kInitialPictureId1;
>+  state.tl0_pic_idx = kInitialTl0PicIdx1;
>+
>+  EncodedImage encoded_image;
>+  CodecSpecificInfo codec_info;
>+  memset(&codec_info, 0, sizeof(CodecSpecificInfo));
>+  codec_info.codecType = kVideoCodecVP8;
>+  codec_info.codecSpecific.VP8.simulcastIdx = 0;
>+
>+  RtpPayloadParams params(kSsrc1, &state);
>+  RTPVideoHeader header = params.GetRtpVideoHeader(encoded_image, &codec_info);
>+  EXPECT_EQ(kVideoCodecVP8, header.codec);
>+  EXPECT_EQ(kInitialPictureId1 + 1, header.vp8().pictureId);
>+
>+  // State should hold latest used picture id and tl0_pic_idx.
>+  state = params.state();
>+  EXPECT_EQ(kInitialPictureId1 + 1, state.picture_id);
>+  EXPECT_EQ(kInitialTl0PicIdx1 + 1, state.tl0_pic_idx);
>+}
>+
>+TEST(RtpPayloadParamsTest, PictureIdWraps) {
>+  RtpPayloadState state;
>+  state.picture_id = kMaxTwoBytePictureId;
>+  state.tl0_pic_idx = kInitialTl0PicIdx1;
>+
>+  EncodedImage encoded_image;
>+  CodecSpecificInfo codec_info;
>+  memset(&codec_info, 0, sizeof(CodecSpecificInfo));
>+  codec_info.codecType = kVideoCodecVP8;
>+  codec_info.codecSpecific.VP8.temporalIdx = kNoTemporalIdx;
>+
>+  RtpPayloadParams params(kSsrc1, &state);
>+  RTPVideoHeader header = params.GetRtpVideoHeader(encoded_image, &codec_info);
>+  EXPECT_EQ(kVideoCodecVP8, header.codec);
>+  EXPECT_EQ(0, header.vp8().pictureId);
>+
>+  // State should hold latest used picture id and tl0_pic_idx.
>+  EXPECT_EQ(0, params.state().picture_id);  // Wrapped.
>+  EXPECT_EQ(kInitialTl0PicIdx1, params.state().tl0_pic_idx);
>+}
>+
>+TEST(RtpPayloadParamsTest, Tl0PicIdxUpdatedForVp8) {
>+  RtpPayloadState state;
>+  state.picture_id = kInitialPictureId1;
>+  state.tl0_pic_idx = kInitialTl0PicIdx1;
>+
>+  EncodedImage encoded_image;
>+  // Modules are sending for this test.
>+  // OnEncodedImage, temporalIdx: 1.
>+  CodecSpecificInfo codec_info;
>+  memset(&codec_info, 0, sizeof(CodecSpecificInfo));
>+  codec_info.codecType = kVideoCodecVP8;
>+  codec_info.codecSpecific.VP8.temporalIdx = 1;
>+
>+  RtpPayloadParams params(kSsrc1, &state);
>+  RTPVideoHeader header = params.GetRtpVideoHeader(encoded_image, &codec_info);
>+
>+  EXPECT_EQ(kVideoCodecVP8, header.codec);
>+  EXPECT_EQ(kInitialPictureId1 + 1, header.vp8().pictureId);
>+  EXPECT_EQ(kInitialTl0PicIdx1, header.vp8().tl0PicIdx);
>+
>+  // OnEncodedImage, temporalIdx: 0.
>+  codec_info.codecSpecific.VP8.temporalIdx = 0;
>+
>+  header = params.GetRtpVideoHeader(encoded_image, &codec_info);
>+  EXPECT_EQ(kVideoCodecVP8, header.codec);
>+  EXPECT_EQ(kInitialPictureId1 + 2, header.vp8().pictureId);
>+  EXPECT_EQ(kInitialTl0PicIdx1 + 1, header.vp8().tl0PicIdx);
>+
>+  // State should hold latest used picture id and tl0_pic_idx.
>+  EXPECT_EQ(kInitialPictureId1 + 2, params.state().picture_id);
>+  EXPECT_EQ(kInitialTl0PicIdx1 + 1, params.state().tl0_pic_idx);
>+}
>+
>+TEST(RtpPayloadParamsTest, Tl0PicIdxUpdatedForVp9) {
>+  RtpPayloadState state;
>+  state.picture_id = kInitialPictureId1;
>+  state.tl0_pic_idx = kInitialTl0PicIdx1;
>+
>+  EncodedImage encoded_image;
>+  // Modules are sending for this test.
>+  // OnEncodedImage, temporalIdx: 1.
>+  CodecSpecificInfo codec_info;
>+  memset(&codec_info, 0, sizeof(CodecSpecificInfo));
>+  codec_info.codecType = kVideoCodecVP9;
>+  codec_info.codecSpecific.VP9.temporal_idx = 1;
>+  codec_info.codecSpecific.VP9.first_frame_in_picture = true;
>+
>+  RtpPayloadParams params(kSsrc1, &state);
>+  RTPVideoHeader header = params.GetRtpVideoHeader(encoded_image, &codec_info);
>+
>+  EXPECT_EQ(kVideoCodecVP9, header.codec);
>+  const auto& vp9_header =
>+      absl::get<RTPVideoHeaderVP9>(header.video_type_header);
>+  EXPECT_EQ(kInitialPictureId1 + 1, vp9_header.picture_id);
>+  EXPECT_EQ(kInitialTl0PicIdx1, vp9_header.tl0_pic_idx);
>+
>+  // OnEncodedImage, temporalIdx: 0.
>+  codec_info.codecSpecific.VP9.temporal_idx = 0;
>+
>+  header = params.GetRtpVideoHeader(encoded_image, &codec_info);
>+
>+  EXPECT_EQ(kVideoCodecVP9, header.codec);
>+  EXPECT_EQ(kInitialPictureId1 + 2, vp9_header.picture_id);
>+  EXPECT_EQ(kInitialTl0PicIdx1 + 1, vp9_header.tl0_pic_idx);
>+
>+  // OnEncodedImage, first_frame_in_picture = false
>+  codec_info.codecSpecific.VP9.first_frame_in_picture = false;
>+
>+  header = params.GetRtpVideoHeader(encoded_image, &codec_info);
>+
>+  EXPECT_EQ(kVideoCodecVP9, header.codec);
>+  EXPECT_EQ(kInitialPictureId1 + 2, vp9_header.picture_id);
>+  EXPECT_EQ(kInitialTl0PicIdx1 + 1, vp9_header.tl0_pic_idx);
>+
>+  // State should hold latest used picture id and tl0_pic_idx.
>+  EXPECT_EQ(kInitialPictureId1 + 2, params.state().picture_id);
>+  EXPECT_EQ(kInitialTl0PicIdx1 + 1, params.state().tl0_pic_idx);
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_rtcp_demuxer_helper.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_rtcp_demuxer_helper.cc
>index deabe738424..125169b0772 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_rtcp_demuxer_helper.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_rtcp_demuxer_helper.cc
>@@ -21,13 +21,13 @@
> 
> namespace webrtc {
> 
>-rtc::Optional<uint32_t> ParseRtcpPacketSenderSsrc(
>+absl::optional<uint32_t> ParseRtcpPacketSenderSsrc(
>     rtc::ArrayView<const uint8_t> packet) {
>   rtcp::CommonHeader header;
>   for (const uint8_t* next_packet = packet.begin(); next_packet < packet.end();
>        next_packet = header.NextPacket()) {
>     if (!header.Parse(next_packet, packet.end() - next_packet)) {
>-      return rtc::nullopt;
>+      return absl::nullopt;
>     }
> 
>     switch (header.type()) {
>@@ -43,13 +43,13 @@ rtc::Optional<uint32_t> ParseRtcpPacketSenderSsrc(
>               ByteReader<uint32_t>::ReadBigEndian(header.payload());
>           return ssrc_sender;
>         } else {
>-          return rtc::nullopt;
>+          return absl::nullopt;
>         }
>       }
>     }
>   }
> 
>-  return rtc::nullopt;
>+  return absl::nullopt;
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_rtcp_demuxer_helper.h b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_rtcp_demuxer_helper.h
>index 32408e8a7a0..4b66713538e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_rtcp_demuxer_helper.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_rtcp_demuxer_helper.h
>@@ -15,9 +15,8 @@
> #include <map>
> #include <utility>
> 
>+#include "absl/types/optional.h"
> #include "api/array_view.h"
>-#include "api/optional.h"
>-#include "rtc_base/basictypes.h"
> 
> namespace webrtc {
> 
>@@ -90,7 +89,7 @@ bool MultimapHasKey(const Container& c,
>   return it_range.first != it_range.second;
> }
> 
>-rtc::Optional<uint32_t> ParseRtcpPacketSenderSsrc(
>+absl::optional<uint32_t> ParseRtcpPacketSenderSsrc(
>     rtc::ArrayView<const uint8_t> packet);
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_rtcp_demuxer_helper_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_rtcp_demuxer_helper_unittest.cc
>index cb8a092616a..9f378864a13 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_rtcp_demuxer_helper_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_rtcp_demuxer_helper_unittest.cc
>@@ -20,7 +20,6 @@
> #include "modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
> #include "modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
> #include "rtc_base/arraysize.h"
>-#include "rtc_base/basictypes.h"
> #include "rtc_base/buffer.h"
> #include "test/gtest.h"
> 
>@@ -35,7 +34,7 @@ TEST(RtpRtcpDemuxerHelperTest, ParseRtcpPacketSenderSsrc_ByePacket) {
>   rtcp_packet.SetSenderSsrc(kSsrc);
>   rtc::Buffer raw_packet = rtcp_packet.Build();
> 
>-  rtc::Optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(raw_packet);
>+  absl::optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(raw_packet);
>   EXPECT_EQ(ssrc, kSsrc);
> }
> 
>@@ -45,7 +44,7 @@ TEST(RtpRtcpDemuxerHelperTest,
>   rtcp_packet.SetSenderSsrc(kSsrc);
>   rtc::Buffer raw_packet = rtcp_packet.Build();
> 
>-  rtc::Optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(raw_packet);
>+  absl::optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(raw_packet);
>   EXPECT_EQ(ssrc, kSsrc);
> }
> 
>@@ -54,7 +53,7 @@ TEST(RtpRtcpDemuxerHelperTest, ParseRtcpPacketSenderSsrc_PsfbPacket) {
>   rtcp_packet.SetSenderSsrc(kSsrc);
>   rtc::Buffer raw_packet = rtcp_packet.Build();
> 
>-  rtc::Optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(raw_packet);
>+  absl::optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(raw_packet);
>   EXPECT_EQ(ssrc, kSsrc);
> }
> 
>@@ -63,7 +62,7 @@ TEST(RtpRtcpDemuxerHelperTest, ParseRtcpPacketSenderSsrc_ReceiverReportPacket) {
>   rtcp_packet.SetSenderSsrc(kSsrc);
>   rtc::Buffer raw_packet = rtcp_packet.Build();
> 
>-  rtc::Optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(raw_packet);
>+  absl::optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(raw_packet);
>   EXPECT_EQ(ssrc, kSsrc);
> }
> 
>@@ -73,7 +72,7 @@ TEST(RtpRtcpDemuxerHelperTest, ParseRtcpPacketSenderSsrc_RtpfbPacket) {
>   rtcp_packet.SetSenderSsrc(kSsrc);
>   rtc::Buffer raw_packet = rtcp_packet.Build();
> 
>-  rtc::Optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(raw_packet);
>+  absl::optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(raw_packet);
>   EXPECT_EQ(ssrc, kSsrc);
> }
> 
>@@ -82,7 +81,7 @@ TEST(RtpRtcpDemuxerHelperTest, ParseRtcpPacketSenderSsrc_SenderReportPacket) {
>   rtcp_packet.SetSenderSsrc(kSsrc);
>   rtc::Buffer raw_packet = rtcp_packet.Build();
> 
>-  rtc::Optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(raw_packet);
>+  absl::optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(raw_packet);
>   EXPECT_EQ(ssrc, kSsrc);
> }
> 
>@@ -90,7 +89,7 @@ TEST(RtpRtcpDemuxerHelperTest, ParseRtcpPacketSenderSsrc_MalformedRtcpPacket) {
>   uint8_t garbage[100];
>   memset(&garbage[0], 0, arraysize(garbage));
> 
>-  rtc::Optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(garbage);
>+  absl::optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(garbage);
>   EXPECT_FALSE(ssrc);
> }
> 
>@@ -99,7 +98,7 @@ TEST(RtpRtcpDemuxerHelperTest,
>   webrtc::rtcp::ExtendedJitterReport rtcp_packet;  // Has no sender SSRC.
>   rtc::Buffer raw_packet = rtcp_packet.Build();
> 
>-  rtc::Optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(raw_packet);
>+  absl::optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(raw_packet);
>   EXPECT_FALSE(ssrc);
> }
> 
>@@ -111,7 +110,7 @@ TEST(RtpRtcpDemuxerHelperTest, ParseRtcpPacketSenderSsrc_TruncatedRtcpMessage) {
>   constexpr size_t rtcp_length_bytes = 8;
>   ASSERT_EQ(rtcp_length_bytes, raw_packet.size());
> 
>-  rtc::Optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(
>+  absl::optional<uint32_t> ssrc = ParseRtcpPacketSenderSsrc(
>       rtc::ArrayView<const uint8_t>(raw_packet.data(), rtcp_length_bytes - 1));
>   EXPECT_FALSE(ssrc);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_stream_receiver_controller.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_stream_receiver_controller.cc
>index a5d73f5784f..3fae021188b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_stream_receiver_controller.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_stream_receiver_controller.cc
>@@ -10,8 +10,8 @@
> 
> #include "call/rtp_stream_receiver_controller.h"
> 
>+#include "absl/memory/memory.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/ptr_util.h"
> 
> namespace webrtc {
> 
>@@ -35,14 +35,18 @@ RtpStreamReceiverController::Receiver::~Receiver() {
>   controller_->RemoveSink(sink_);
> }
> 
>-RtpStreamReceiverController::RtpStreamReceiverController() = default;
>+RtpStreamReceiverController::RtpStreamReceiverController() {
>+  // At this level the demuxer is only configured to demux by SSRC, so don't
>+  // worry about MIDs (MIDs are handled by upper layers).
>+  demuxer_.set_use_mid(false);
>+}
>+
> RtpStreamReceiverController::~RtpStreamReceiverController() = default;
> 
> std::unique_ptr<RtpStreamReceiverInterface>
>-RtpStreamReceiverController::CreateReceiver(
>-    uint32_t ssrc,
>-    RtpPacketSinkInterface* sink) {
>-  return rtc::MakeUnique<Receiver>(this, ssrc, sink);
>+RtpStreamReceiverController::CreateReceiver(uint32_t ssrc,
>+                                            RtpPacketSinkInterface* sink) {
>+  return absl::make_unique<Receiver>(this, ssrc, sink);
> }
> 
> bool RtpStreamReceiverController::OnRtpPacket(const RtpPacketReceived& packet) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_transport_controller_send.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_transport_controller_send.cc
>index 090c261e4e5..b0dd8d8e295 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_transport_controller_send.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_transport_controller_send.cc
>@@ -7,32 +7,159 @@
>  *  in the file PATENTS.  All contributing project authors may
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
>+#include <utility>
>+#include <vector>
> 
>+#include "absl/memory/memory.h"
> #include "call/rtp_transport_controller_send.h"
>+#include "modules/congestion_controller/include/send_side_congestion_controller.h"
>+#include "modules/congestion_controller/rtp/include/send_side_congestion_controller.h"
>+#include "rtc_base/location.h"
>+#include "rtc_base/logging.h"
>+#include "rtc_base/rate_limiter.h"
>+#include "system_wrappers/include/field_trial.h"
> 
> namespace webrtc {
>+namespace {
>+static const int64_t kRetransmitWindowSizeMs = 500;
>+const char kTaskQueueExperiment[] = "WebRTC-TaskQueueCongestionControl";
>+using TaskQueueController = webrtc::webrtc_cc::SendSideCongestionController;
>+
>+bool TaskQueueExperimentEnabled() {
>+  std::string trial = webrtc::field_trial::FindFullName(kTaskQueueExperiment);
>+  return trial.find("Enable") == 0;
>+}
>+
>+std::unique_ptr<SendSideCongestionControllerInterface> CreateController(
>+    Clock* clock,
>+    rtc::TaskQueue* task_queue,
>+    webrtc::RtcEventLog* event_log,
>+    PacedSender* pacer,
>+    const BitrateConstraints& bitrate_config,
>+    bool task_queue_controller,
>+    NetworkControllerFactoryInterface* controller_factory) {
>+  if (task_queue_controller) {
>+    RTC_LOG(LS_INFO) << "Using TaskQueue based SSCC";
>+    return absl::make_unique<webrtc::webrtc_cc::SendSideCongestionController>(
>+        clock, task_queue, event_log, pacer, bitrate_config.start_bitrate_bps,
>+        bitrate_config.min_bitrate_bps, bitrate_config.max_bitrate_bps,
>+        controller_factory);
>+  }
>+  RTC_LOG(LS_INFO) << "Using Legacy SSCC";
>+  auto cc = absl::make_unique<webrtc::SendSideCongestionController>(
>+      clock, nullptr /* observer */, event_log, pacer);
>+  cc->SignalNetworkState(kNetworkDown);
>+  cc->SetBweBitrates(bitrate_config.min_bitrate_bps,
>+                     bitrate_config.start_bitrate_bps,
>+                     bitrate_config.max_bitrate_bps);
>+  return std::move(cc);
>+}
>+}  // namespace
> 
> RtpTransportControllerSend::RtpTransportControllerSend(
>     Clock* clock,
>-    webrtc::RtcEventLog* event_log)
>-    : pacer_(clock, &packet_router_, event_log),
>-      send_side_cc_(clock, nullptr /* observer */, event_log, &pacer_) {}
>+    webrtc::RtcEventLog* event_log,
>+    NetworkControllerFactoryInterface* controller_factory,
>+    const BitrateConstraints& bitrate_config)
>+    : clock_(clock),
>+      pacer_(clock, &packet_router_, event_log),
>+      bitrate_configurator_(bitrate_config),
>+      process_thread_(ProcessThread::Create("SendControllerThread")),
>+      observer_(nullptr),
>+      retransmission_rate_limiter_(clock, kRetransmitWindowSizeMs),
>+      task_queue_("rtp_send_controller") {
>+  // Created after task_queue to be able to post to the task queue internally.
>+  send_side_cc_ =
>+      CreateController(clock, &task_queue_, event_log, &pacer_, bitrate_config,
>+                       TaskQueueExperimentEnabled(), controller_factory);
> 
>-PacketRouter* RtpTransportControllerSend::packet_router() {
>-  return &packet_router_;
>+  process_thread_->RegisterModule(&pacer_, RTC_FROM_HERE);
>+  process_thread_->RegisterModule(send_side_cc_.get(), RTC_FROM_HERE);
>+  process_thread_->Start();
> }
> 
>-PacedSender* RtpTransportControllerSend::pacer() {
>-  return &pacer_;
>+RtpTransportControllerSend::~RtpTransportControllerSend() {
>+  process_thread_->Stop();
>+  process_thread_->DeRegisterModule(send_side_cc_.get());
>+  process_thread_->DeRegisterModule(&pacer_);
> }
> 
>-SendSideCongestionController* RtpTransportControllerSend::send_side_cc() {
>-  return &send_side_cc_;
>+RtpVideoSenderInterface* RtpTransportControllerSend::CreateRtpVideoSender(
>+    const std::vector<uint32_t>& ssrcs,
>+    std::map<uint32_t, RtpState> suspended_ssrcs,
>+    const std::map<uint32_t, RtpPayloadState>& states,
>+    const RtpConfig& rtp_config,
>+    const RtcpConfig& rtcp_config,
>+    Transport* send_transport,
>+    const RtpSenderObservers& observers,
>+    RtcEventLog* event_log) {
>+  video_rtp_senders_.push_back(absl::make_unique<RtpVideoSender>(
>+      ssrcs, suspended_ssrcs, states, rtp_config, rtcp_config, send_transport,
>+      observers,
>+      // TODO(holmer): Remove this circular dependency by injecting
>+      // the parts of RtpTransportControllerSendInterface that are really used.
>+      this, event_log, &retransmission_rate_limiter_));
>+  return video_rtp_senders_.back().get();
>+}
>+
>+void RtpTransportControllerSend::DestroyRtpVideoSender(
>+    RtpVideoSenderInterface* rtp_video_sender) {
>+  std::vector<std::unique_ptr<RtpVideoSenderInterface>>::iterator it =
>+      video_rtp_senders_.end();
>+  for (it = video_rtp_senders_.begin(); it != video_rtp_senders_.end(); ++it) {
>+    if (it->get() == rtp_video_sender) {
>+      break;
>+    }
>+  }
>+  RTC_DCHECK(it != video_rtp_senders_.end());
>+  video_rtp_senders_.erase(it);
>+}
>+
>+void RtpTransportControllerSend::OnNetworkChanged(uint32_t bitrate_bps,
>+                                                  uint8_t fraction_loss,
>+                                                  int64_t rtt_ms,
>+                                                  int64_t probing_interval_ms) {
>+  // TODO(srte): Skip this step when old SendSideCongestionController is
>+  // deprecated.
>+  TargetTransferRate msg;
>+  msg.at_time = Timestamp::ms(clock_->TimeInMilliseconds());
>+  msg.target_rate = DataRate::bps(bitrate_bps);
>+  msg.network_estimate.at_time = msg.at_time;
>+  msg.network_estimate.bwe_period = TimeDelta::ms(probing_interval_ms);
>+  uint32_t bandwidth_bps;
>+  if (send_side_cc_->AvailableBandwidth(&bandwidth_bps))
>+    msg.network_estimate.bandwidth = DataRate::bps(bandwidth_bps);
>+  msg.network_estimate.loss_rate_ratio = fraction_loss / 255.0;
>+  msg.network_estimate.round_trip_time = TimeDelta::ms(rtt_ms);
>+
>+  retransmission_rate_limiter_.SetMaxRate(bandwidth_bps);
>+
>+  if (!task_queue_.IsCurrent()) {
>+    task_queue_.PostTask([this, msg] {
>+      rtc::CritScope cs(&observer_crit_);
>+      // We won't register as observer until we have an observers.
>+      RTC_DCHECK(observer_ != nullptr);
>+      observer_->OnTargetTransferRate(msg);
>+    });
>+  } else {
>+    rtc::CritScope cs(&observer_crit_);
>+    // We won't register as observer until we have an observers.
>+    RTC_DCHECK(observer_ != nullptr);
>+    observer_->OnTargetTransferRate(msg);
>+  }
>+}
>+
>+rtc::TaskQueue* RtpTransportControllerSend::GetWorkerQueue() {
>+  return &task_queue_;
>+}
>+
>+PacketRouter* RtpTransportControllerSend::packet_router() {
>+  return &packet_router_;
> }
> 
> TransportFeedbackObserver*
> RtpTransportControllerSend::transport_feedback_observer() {
>-  return &send_side_cc_;
>+  return send_side_cc_.get();
> }
> 
> RtpPacketSender* RtpTransportControllerSend::packet_sender() {
>@@ -45,13 +172,141 @@ const RtpKeepAliveConfig& RtpTransportControllerSend::keepalive_config() const {
> 
> void RtpTransportControllerSend::SetAllocatedSendBitrateLimits(
>     int min_send_bitrate_bps,
>-    int max_padding_bitrate_bps) {
>-  pacer_.SetSendBitrateLimits(min_send_bitrate_bps, max_padding_bitrate_bps);
>+    int max_padding_bitrate_bps,
>+    int max_total_bitrate_bps) {
>+  send_side_cc_->SetAllocatedSendBitrateLimits(
>+      min_send_bitrate_bps, max_padding_bitrate_bps, max_total_bitrate_bps);
> }
> 
> void RtpTransportControllerSend::SetKeepAliveConfig(
>     const RtpKeepAliveConfig& config) {
>   keepalive_ = config;
> }
>+void RtpTransportControllerSend::SetPacingFactor(float pacing_factor) {
>+  send_side_cc_->SetPacingFactor(pacing_factor);
>+}
>+void RtpTransportControllerSend::SetQueueTimeLimit(int limit_ms) {
>+  pacer_.SetQueueTimeLimit(limit_ms);
>+}
>+CallStatsObserver* RtpTransportControllerSend::GetCallStatsObserver() {
>+  return send_side_cc_.get();
>+}
>+void RtpTransportControllerSend::RegisterPacketFeedbackObserver(
>+    PacketFeedbackObserver* observer) {
>+  send_side_cc_->RegisterPacketFeedbackObserver(observer);
>+}
>+void RtpTransportControllerSend::DeRegisterPacketFeedbackObserver(
>+    PacketFeedbackObserver* observer) {
>+  send_side_cc_->DeRegisterPacketFeedbackObserver(observer);
>+}
>+
>+void RtpTransportControllerSend::RegisterTargetTransferRateObserver(
>+    TargetTransferRateObserver* observer) {
>+  {
>+    rtc::CritScope cs(&observer_crit_);
>+    RTC_DCHECK(observer_ == nullptr);
>+    observer_ = observer;
>+  }
>+  send_side_cc_->RegisterNetworkObserver(this);
>+}
>+void RtpTransportControllerSend::OnNetworkRouteChanged(
>+    const std::string& transport_name,
>+    const rtc::NetworkRoute& network_route) {
>+  // Check if the network route is connected.
>+  if (!network_route.connected) {
>+    RTC_LOG(LS_INFO) << "Transport " << transport_name << " is disconnected";
>+    // TODO(honghaiz): Perhaps handle this in SignalChannelNetworkState and
>+    // consider merging these two methods.
>+    return;
>+  }
>+
>+  // Check whether the network route has changed on each transport.
>+  auto result =
>+      network_routes_.insert(std::make_pair(transport_name, network_route));
>+  auto kv = result.first;
>+  bool inserted = result.second;
>+  if (inserted) {
>+    // No need to reset BWE if this is the first time the network connects.
>+    return;
>+  }
>+  if (kv->second != network_route) {
>+    kv->second = network_route;
>+    BitrateConstraints bitrate_config = bitrate_configurator_.GetConfig();
>+    RTC_LOG(LS_INFO) << "Network route changed on transport " << transport_name
>+                     << ": new local network id "
>+                     << network_route.local_network_id
>+                     << " new remote network id "
>+                     << network_route.remote_network_id
>+                     << " Reset bitrates to min: "
>+                     << bitrate_config.min_bitrate_bps
>+                     << " bps, start: " << bitrate_config.start_bitrate_bps
>+                     << " bps,  max: " << bitrate_config.max_bitrate_bps
>+                     << " bps.";
>+    RTC_DCHECK_GT(bitrate_config.start_bitrate_bps, 0);
>+    send_side_cc_->OnNetworkRouteChanged(
>+        network_route, bitrate_config.start_bitrate_bps,
>+        bitrate_config.min_bitrate_bps, bitrate_config.max_bitrate_bps);
>+  }
>+}
>+void RtpTransportControllerSend::OnNetworkAvailability(bool network_available) {
>+  send_side_cc_->SignalNetworkState(network_available ? kNetworkUp
>+                                                      : kNetworkDown);
>+  for (auto& rtp_sender : video_rtp_senders_) {
>+    rtp_sender->OnNetworkAvailability(network_available);
>+  }
>+}
>+RtcpBandwidthObserver* RtpTransportControllerSend::GetBandwidthObserver() {
>+  return send_side_cc_->GetBandwidthObserver();
>+}
>+int64_t RtpTransportControllerSend::GetPacerQueuingDelayMs() const {
>+  return pacer_.QueueInMs();
>+}
>+int64_t RtpTransportControllerSend::GetFirstPacketTimeMs() const {
>+  return pacer_.FirstSentPacketTimeMs();
>+}
>+void RtpTransportControllerSend::SetPerPacketFeedbackAvailable(bool available) {
>+  send_side_cc_->SetPerPacketFeedbackAvailable(available);
>+}
>+void RtpTransportControllerSend::EnablePeriodicAlrProbing(bool enable) {
>+  send_side_cc_->EnablePeriodicAlrProbing(enable);
>+}
>+void RtpTransportControllerSend::OnSentPacket(
>+    const rtc::SentPacket& sent_packet) {
>+  send_side_cc_->OnSentPacket(sent_packet);
>+}
> 
>+void RtpTransportControllerSend::SetSdpBitrateParameters(
>+    const BitrateConstraints& constraints) {
>+  absl::optional<BitrateConstraints> updated =
>+      bitrate_configurator_.UpdateWithSdpParameters(constraints);
>+  if (updated.has_value()) {
>+    send_side_cc_->SetBweBitrates(updated->min_bitrate_bps,
>+                                  updated->start_bitrate_bps,
>+                                  updated->max_bitrate_bps);
>+  } else {
>+    RTC_LOG(LS_VERBOSE)
>+        << "WebRTC.RtpTransportControllerSend.SetSdpBitrateParameters: "
>+        << "nothing to update";
>+  }
>+}
>+
>+void RtpTransportControllerSend::SetClientBitratePreferences(
>+    const BitrateSettings& preferences) {
>+  absl::optional<BitrateConstraints> updated =
>+      bitrate_configurator_.UpdateWithClientPreferences(preferences);
>+  if (updated.has_value()) {
>+    send_side_cc_->SetBweBitrates(updated->min_bitrate_bps,
>+                                  updated->start_bitrate_bps,
>+                                  updated->max_bitrate_bps);
>+  } else {
>+    RTC_LOG(LS_VERBOSE)
>+        << "WebRTC.RtpTransportControllerSend.SetClientBitratePreferences: "
>+        << "nothing to update";
>+  }
>+}
>+
>+void RtpTransportControllerSend::SetAllocatedBitrateWithoutFeedback(
>+    uint32_t bitrate_bps) {
>+  send_side_cc_->SetAllocatedBitrateWithoutFeedback(bitrate_bps);
>+}
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_transport_controller_send.h b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_transport_controller_send.h
>index 45d6db0feee..496a00ede5f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_transport_controller_send.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_transport_controller_send.h
>@@ -11,11 +11,22 @@
> #ifndef CALL_RTP_TRANSPORT_CONTROLLER_SEND_H_
> #define CALL_RTP_TRANSPORT_CONTROLLER_SEND_H_
> 
>+#include <map>
>+#include <memory>
>+#include <string>
>+#include <vector>
>+
>+#include "api/transport/network_control.h"
>+#include "call/rtp_bitrate_configurator.h"
> #include "call/rtp_transport_controller_send_interface.h"
>+#include "call/rtp_video_sender.h"
> #include "common_types.h"  // NOLINT(build/include)
>-#include "modules/congestion_controller/include/send_side_congestion_controller.h"
>+#include "modules/congestion_controller/include/send_side_congestion_controller_interface.h"
> #include "modules/pacing/packet_router.h"
>+#include "modules/utility/include/process_thread.h"
> #include "rtc_base/constructormagic.h"
>+#include "rtc_base/networkroute.h"
>+#include "rtc_base/task_queue.h"
> 
> namespace webrtc {
> class Clock;
>@@ -24,33 +35,91 @@ class RtcEventLog;
> // TODO(nisse): When we get the underlying transports here, we should
> // have one object implementing RtpTransportControllerSendInterface
> // per transport, sharing the same congestion controller.
>-class RtpTransportControllerSend : public RtpTransportControllerSendInterface {
>+class RtpTransportControllerSend final
>+    : public RtpTransportControllerSendInterface,
>+      public NetworkChangedObserver {
>  public:
>-  RtpTransportControllerSend(Clock* clock, webrtc::RtcEventLog* event_log);
>+  RtpTransportControllerSend(
>+      Clock* clock,
>+      RtcEventLog* event_log,
>+      NetworkControllerFactoryInterface* controller_factory,
>+      const BitrateConstraints& bitrate_config);
>+  ~RtpTransportControllerSend() override;
>+
>+  RtpVideoSenderInterface* CreateRtpVideoSender(
>+      const std::vector<uint32_t>& ssrcs,
>+      std::map<uint32_t, RtpState> suspended_ssrcs,
>+      const std::map<uint32_t, RtpPayloadState>&
>+          states,  // move states into RtpTransportControllerSend
>+      const RtpConfig& rtp_config,
>+      const RtcpConfig& rtcp_config,
>+      Transport* send_transport,
>+      const RtpSenderObservers& observers,
>+      RtcEventLog* event_log) override;
>+  void DestroyRtpVideoSender(
>+      RtpVideoSenderInterface* rtp_video_sender) override;
>+
>+  // Implements NetworkChangedObserver interface.
>+  void OnNetworkChanged(uint32_t bitrate_bps,
>+                        uint8_t fraction_loss,
>+                        int64_t rtt_ms,
>+                        int64_t probing_interval_ms) override;
> 
>   // Implements RtpTransportControllerSendInterface
>+  rtc::TaskQueue* GetWorkerQueue() override;
>   PacketRouter* packet_router() override;
>-  // TODO(holmer): Temporarily exposed, should be removed and the
>-  // appropriate methods should be added to this class instead.
>-  // In addition the PacedSender should be driven by this class, either
>-  // by owning the process thread, or later by using a task queue.
>-  PacedSender* pacer() override;
>-  SendSideCongestionController* send_side_cc() override;
>+
>   TransportFeedbackObserver* transport_feedback_observer() override;
>   RtpPacketSender* packet_sender() override;
>   const RtpKeepAliveConfig& keepalive_config() const override;
> 
>   void SetAllocatedSendBitrateLimits(int min_send_bitrate_bps,
>-                                     int max_padding_bitrate_bps) override;
>+                                     int max_padding_bitrate_bps,
>+                                     int max_total_bitrate_bps) override;
> 
>   void SetKeepAliveConfig(const RtpKeepAliveConfig& config);
>+  void SetPacingFactor(float pacing_factor) override;
>+  void SetQueueTimeLimit(int limit_ms) override;
>+  CallStatsObserver* GetCallStatsObserver() override;
>+  void RegisterPacketFeedbackObserver(
>+      PacketFeedbackObserver* observer) override;
>+  void DeRegisterPacketFeedbackObserver(
>+      PacketFeedbackObserver* observer) override;
>+  void RegisterTargetTransferRateObserver(
>+      TargetTransferRateObserver* observer) override;
>+  void OnNetworkRouteChanged(const std::string& transport_name,
>+                             const rtc::NetworkRoute& network_route) override;
>+  void OnNetworkAvailability(bool network_available) override;
>+  RtcpBandwidthObserver* GetBandwidthObserver() override;
>+  int64_t GetPacerQueuingDelayMs() const override;
>+  int64_t GetFirstPacketTimeMs() const override;
>+  void SetPerPacketFeedbackAvailable(bool available) override;
>+  void EnablePeriodicAlrProbing(bool enable) override;
>+  void OnSentPacket(const rtc::SentPacket& sent_packet) override;
>+
>+  void SetSdpBitrateParameters(const BitrateConstraints& constraints) override;
>+  void SetClientBitratePreferences(const BitrateSettings& preferences) override;
>+
>+  void SetAllocatedBitrateWithoutFeedback(uint32_t bitrate_bps) override;
> 
>  private:
>+  const Clock* const clock_;
>   PacketRouter packet_router_;
>+  std::vector<std::unique_ptr<RtpVideoSenderInterface>> video_rtp_senders_;
>   PacedSender pacer_;
>-  SendSideCongestionController send_side_cc_;
>   RtpKeepAliveConfig keepalive_;
>+  RtpBitrateConfigurator bitrate_configurator_;
>+  std::map<std::string, rtc::NetworkRoute> network_routes_;
>+  const std::unique_ptr<ProcessThread> process_thread_;
>+  rtc::CriticalSection observer_crit_;
>+  TargetTransferRateObserver* observer_ RTC_GUARDED_BY(observer_crit_);
>+  std::unique_ptr<SendSideCongestionControllerInterface> send_side_cc_;
>+  RateLimiter retransmission_rate_limiter_;
> 
>+  // TODO(perkj): |task_queue_| is supposed to replace |process_thread_|.
>+  // |task_queue_| is defined last to ensure all pending tasks are cancelled
>+  // and deleted before any other members.
>+  rtc::TaskQueue task_queue_;
>   RTC_DISALLOW_COPY_AND_ASSIGN(RtpTransportControllerSend);
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_transport_controller_send_interface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_transport_controller_send_interface.h
>index 7b0dbefd14f..828fec40cf0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_transport_controller_send_interface.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_transport_controller_send_interface.h
>@@ -10,16 +10,57 @@
> 
> #ifndef CALL_RTP_TRANSPORT_CONTROLLER_SEND_INTERFACE_H_
> #define CALL_RTP_TRANSPORT_CONTROLLER_SEND_INTERFACE_H_
>+#include <stddef.h>
>+#include <stdint.h>
> 
>+#include <map>
>+#include <string>
>+#include <vector>
>+
>+#include "absl/types/optional.h"
>+#include "api/bitrate_constraints.h"
>+#include "api/transport/bitrate_settings.h"
>+#include "call/rtp_config.h"
>+#include "logging/rtc_event_log/rtc_event_log.h"
>+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>+
>+namespace rtc {
>+struct SentPacket;
>+struct NetworkRoute;
>+class TaskQueue;
>+}  // namespace rtc
> namespace webrtc {
> 
>+class CallStats;
>+class CallStatsObserver;
>+class TargetTransferRateObserver;
>+class Transport;
>+class Module;
> class PacedSender;
>+class PacketFeedbackObserver;
> class PacketRouter;
>+class RtpVideoSenderInterface;
>+class RateLimiter;
>+class RtcpBandwidthObserver;
> class RtpPacketSender;
> struct RtpKeepAliveConfig;
>-class SendSideCongestionController;
>+class SendDelayStats;
>+class SendStatisticsProxy;
> class TransportFeedbackObserver;
> 
>+struct RtpSenderObservers {
>+  RtcpRttStats* rtcp_rtt_stats;
>+  RtcpIntraFrameObserver* intra_frame_callback;
>+  RtcpStatisticsCallback* rtcp_stats;
>+  StreamDataCountersCallback* rtp_stats;
>+  BitrateStatisticsObserver* bitrate_observer;
>+  FrameCountObserver* frame_count_observer;
>+  RtcpPacketTypeCounterObserver* rtcp_type_observer;
>+  SendSideDelayObserver* send_delay_observer;
>+  SendPacketObserver* send_packet_observer;
>+  OverheadObserver* overhead_observer;
>+};
>+
> // An RtpTransportController should own everything related to the RTP
> // transport to/from a remote endpoint. We should have separate
> // interfaces for send and receive side, even if they are implemented
>@@ -46,10 +87,22 @@ class TransportFeedbackObserver;
> class RtpTransportControllerSendInterface {
>  public:
>   virtual ~RtpTransportControllerSendInterface() {}
>+  virtual rtc::TaskQueue* GetWorkerQueue() = 0;
>   virtual PacketRouter* packet_router() = 0;
>-  virtual PacedSender* pacer() = 0;
>-  // Currently returning the same pointer, but with different types.
>-  virtual SendSideCongestionController* send_side_cc() = 0;
>+
>+  virtual RtpVideoSenderInterface* CreateRtpVideoSender(
>+      const std::vector<uint32_t>& ssrcs,
>+      std::map<uint32_t, RtpState> suspended_ssrcs,
>+      // TODO(holmer): Move states into RtpTransportControllerSend.
>+      const std::map<uint32_t, RtpPayloadState>& states,
>+      const RtpConfig& rtp_config,
>+      const RtcpConfig& rtcp_config,
>+      Transport* send_transport,
>+      const RtpSenderObservers& observers,
>+      RtcEventLog* event_log) = 0;
>+  virtual void DestroyRtpVideoSender(
>+      RtpVideoSenderInterface* rtp_video_sender) = 0;
>+
>   virtual TransportFeedbackObserver* transport_feedback_observer() = 0;
> 
>   virtual RtpPacketSender* packet_sender() = 0;
>@@ -65,7 +118,37 @@ class RtpTransportControllerSendInterface {
>   // current network estimate and tells the PacedSender how much it should max
>   // pad unless there is real packets to send.
>   virtual void SetAllocatedSendBitrateLimits(int min_send_bitrate_bps,
>-                                             int max_padding_bitrate_bps) = 0;
>+                                             int max_padding_bitrate_bps,
>+                                             int total_bitrate_bps) = 0;
>+
>+  virtual void SetPacingFactor(float pacing_factor) = 0;
>+  virtual void SetQueueTimeLimit(int limit_ms) = 0;
>+
>+  virtual CallStatsObserver* GetCallStatsObserver() = 0;
>+
>+  virtual void RegisterPacketFeedbackObserver(
>+      PacketFeedbackObserver* observer) = 0;
>+  virtual void DeRegisterPacketFeedbackObserver(
>+      PacketFeedbackObserver* observer) = 0;
>+  virtual void RegisterTargetTransferRateObserver(
>+      TargetTransferRateObserver* observer) = 0;
>+  virtual void OnNetworkRouteChanged(
>+      const std::string& transport_name,
>+      const rtc::NetworkRoute& network_route) = 0;
>+  virtual void OnNetworkAvailability(bool network_available) = 0;
>+  virtual RtcpBandwidthObserver* GetBandwidthObserver() = 0;
>+  virtual int64_t GetPacerQueuingDelayMs() const = 0;
>+  virtual int64_t GetFirstPacketTimeMs() const = 0;
>+  virtual void EnablePeriodicAlrProbing(bool enable) = 0;
>+  virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0;
>+  virtual void SetPerPacketFeedbackAvailable(bool available) = 0;
>+
>+  virtual void SetSdpBitrateParameters(
>+      const BitrateConstraints& constraints) = 0;
>+  virtual void SetClientBitratePreferences(
>+      const BitrateSettings& preferences) = 0;
>+
>+  virtual void SetAllocatedBitrateWithoutFeedback(uint32_t bitrate_bps) = 0;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_video_sender.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_video_sender.cc
>new file mode 100644
>index 00000000000..d9dcb87eb8c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_video_sender.cc
>@@ -0,0 +1,553 @@
>+/*
>+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "call/rtp_video_sender.h"
>+
>+#include <algorithm>
>+#include <memory>
>+#include <string>
>+#include <utility>
>+
>+#include "call/rtp_transport_controller_send_interface.h"
>+#include "modules/pacing/packet_router.h"
>+#include "modules/rtp_rtcp/include/rtp_rtcp.h"
>+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>+#include "modules/rtp_rtcp/source/rtp_sender.h"
>+#include "modules/utility/include/process_thread.h"
>+#include "modules/video_coding/include/video_codec_interface.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/location.h"
>+#include "rtc_base/logging.h"
>+#include "system_wrappers/include/field_trial.h"
>+
>+namespace webrtc {
>+
>+namespace {
>+static const int kMinSendSidePacketHistorySize = 600;
>+
>+std::vector<std::unique_ptr<RtpRtcp>> CreateRtpRtcpModules(
>+    const std::vector<uint32_t>& ssrcs,
>+    const std::vector<uint32_t>& protected_media_ssrcs,
>+    const RtcpConfig& rtcp_config,
>+    Transport* send_transport,
>+    RtcpIntraFrameObserver* intra_frame_callback,
>+    RtcpBandwidthObserver* bandwidth_callback,
>+    RtpTransportControllerSendInterface* transport,
>+    RtcpRttStats* rtt_stats,
>+    FlexfecSender* flexfec_sender,
>+    BitrateStatisticsObserver* bitrate_observer,
>+    FrameCountObserver* frame_count_observer,
>+    RtcpPacketTypeCounterObserver* rtcp_type_observer,
>+    SendSideDelayObserver* send_delay_observer,
>+    SendPacketObserver* send_packet_observer,
>+    RtcEventLog* event_log,
>+    RateLimiter* retransmission_rate_limiter,
>+    OverheadObserver* overhead_observer,
>+    RtpKeepAliveConfig keepalive_config) {
>+  RTC_DCHECK_GT(ssrcs.size(), 0);
>+  RtpRtcp::Configuration configuration;
>+  configuration.audio = false;
>+  configuration.receiver_only = false;
>+  configuration.outgoing_transport = send_transport;
>+  configuration.intra_frame_callback = intra_frame_callback;
>+  configuration.bandwidth_callback = bandwidth_callback;
>+  configuration.transport_feedback_callback =
>+      transport->transport_feedback_observer();
>+  configuration.rtt_stats = rtt_stats;
>+  configuration.rtcp_packet_type_counter_observer = rtcp_type_observer;
>+  configuration.paced_sender = transport->packet_sender();
>+  configuration.transport_sequence_number_allocator =
>+      transport->packet_router();
>+  configuration.send_bitrate_observer = bitrate_observer;
>+  configuration.send_frame_count_observer = frame_count_observer;
>+  configuration.send_side_delay_observer = send_delay_observer;
>+  configuration.send_packet_observer = send_packet_observer;
>+  configuration.event_log = event_log;
>+  configuration.retransmission_rate_limiter = retransmission_rate_limiter;
>+  configuration.overhead_observer = overhead_observer;
>+  configuration.keepalive_config = keepalive_config;
>+  configuration.rtcp_interval_config.video_interval_ms =
>+      rtcp_config.video_report_interval_ms;
>+  configuration.rtcp_interval_config.audio_interval_ms =
>+      rtcp_config.audio_report_interval_ms;
>+  std::vector<std::unique_ptr<RtpRtcp>> modules;
>+  const std::vector<uint32_t>& flexfec_protected_ssrcs = protected_media_ssrcs;
>+  for (uint32_t ssrc : ssrcs) {
>+    bool enable_flexfec = flexfec_sender != nullptr &&
>+                          std::find(flexfec_protected_ssrcs.begin(),
>+                                    flexfec_protected_ssrcs.end(),
>+                                    ssrc) != flexfec_protected_ssrcs.end();
>+    configuration.flexfec_sender = enable_flexfec ? flexfec_sender : nullptr;
>+    std::unique_ptr<RtpRtcp> rtp_rtcp =
>+        std::unique_ptr<RtpRtcp>(RtpRtcp::CreateRtpRtcp(configuration));
>+    rtp_rtcp->SetSendingStatus(false);
>+    rtp_rtcp->SetSendingMediaStatus(false);
>+    rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound);
>+    modules.push_back(std::move(rtp_rtcp));
>+  }
>+  return modules;
>+}
>+
>+absl::optional<size_t> GetSimulcastIdx(const CodecSpecificInfo* info) {
>+  if (!info)
>+    return absl::nullopt;
>+  switch (info->codecType) {
>+    case kVideoCodecVP8:
>+      return absl::optional<size_t>(info->codecSpecific.VP8.simulcastIdx);
>+    case kVideoCodecH264:
>+      return absl::optional<size_t>(info->codecSpecific.H264.simulcast_idx);
>+    case kVideoCodecMultiplex:
>+    case kVideoCodecGeneric:
>+      return absl::optional<size_t>(info->codecSpecific.generic.simulcast_idx);
>+    default:
>+      return absl::nullopt;
>+  }
>+}
>+bool PayloadTypeSupportsSkippingFecPackets(const std::string& payload_name) {
>+  const VideoCodecType codecType = PayloadStringToCodecType(payload_name);
>+  if (codecType == kVideoCodecVP8 || codecType == kVideoCodecVP9) {
>+    return true;
>+  }
>+  return false;
>+}
>+
>+// TODO(brandtr): Update this function when we support multistream protection.
>+std::unique_ptr<FlexfecSender> MaybeCreateFlexfecSender(
>+    const RtpConfig& rtp,
>+    const std::map<uint32_t, RtpState>& suspended_ssrcs) {
>+  if (rtp.flexfec.payload_type < 0) {
>+    return nullptr;
>+  }
>+  RTC_DCHECK_GE(rtp.flexfec.payload_type, 0);
>+  RTC_DCHECK_LE(rtp.flexfec.payload_type, 127);
>+  if (rtp.flexfec.ssrc == 0) {
>+    RTC_LOG(LS_WARNING) << "FlexFEC is enabled, but no FlexFEC SSRC given. "
>+                           "Therefore disabling FlexFEC.";
>+    return nullptr;
>+  }
>+  if (rtp.flexfec.protected_media_ssrcs.empty()) {
>+    RTC_LOG(LS_WARNING)
>+        << "FlexFEC is enabled, but no protected media SSRC given. "
>+           "Therefore disabling FlexFEC.";
>+    return nullptr;
>+  }
>+
>+  if (rtp.flexfec.protected_media_ssrcs.size() > 1) {
>+    RTC_LOG(LS_WARNING)
>+        << "The supplied FlexfecConfig contained multiple protected "
>+           "media streams, but our implementation currently only "
>+           "supports protecting a single media stream. "
>+           "To avoid confusion, disabling FlexFEC completely.";
>+    return nullptr;
>+  }
>+
>+  const RtpState* rtp_state = nullptr;
>+  auto it = suspended_ssrcs.find(rtp.flexfec.ssrc);
>+  if (it != suspended_ssrcs.end()) {
>+    rtp_state = &it->second;
>+  }
>+
>+  RTC_DCHECK_EQ(1U, rtp.flexfec.protected_media_ssrcs.size());
>+  return absl::make_unique<FlexfecSender>(
>+      rtp.flexfec.payload_type, rtp.flexfec.ssrc,
>+      rtp.flexfec.protected_media_ssrcs[0], rtp.mid, rtp.extensions,
>+      RTPSender::FecExtensionSizes(), rtp_state, Clock::GetRealTimeClock());
>+}
>+}  // namespace
>+
>+RtpVideoSender::RtpVideoSender(
>+    const std::vector<uint32_t>& ssrcs,
>+    std::map<uint32_t, RtpState> suspended_ssrcs,
>+    const std::map<uint32_t, RtpPayloadState>& states,
>+    const RtpConfig& rtp_config,
>+    const RtcpConfig& rtcp_config,
>+    Transport* send_transport,
>+    const RtpSenderObservers& observers,
>+    RtpTransportControllerSendInterface* transport,
>+    RtcEventLog* event_log,
>+    RateLimiter* retransmission_limiter)
>+    : active_(false),
>+      module_process_thread_(nullptr),
>+      suspended_ssrcs_(std::move(suspended_ssrcs)),
>+      flexfec_sender_(MaybeCreateFlexfecSender(rtp_config, suspended_ssrcs_)),
>+      rtp_modules_(
>+          CreateRtpRtcpModules(ssrcs,
>+                               rtp_config.flexfec.protected_media_ssrcs,
>+                               rtcp_config,
>+                               send_transport,
>+                               observers.intra_frame_callback,
>+                               transport->GetBandwidthObserver(),
>+                               transport,
>+                               observers.rtcp_rtt_stats,
>+                               flexfec_sender_.get(),
>+                               observers.bitrate_observer,
>+                               observers.frame_count_observer,
>+                               observers.rtcp_type_observer,
>+                               observers.send_delay_observer,
>+                               observers.send_packet_observer,
>+                               event_log,
>+                               retransmission_limiter,
>+                               observers.overhead_observer,
>+                               transport->keepalive_config())),
>+      rtp_config_(rtp_config),
>+      transport_(transport) {
>+  RTC_DCHECK_EQ(ssrcs.size(), rtp_modules_.size());
>+  module_process_thread_checker_.DetachFromThread();
>+  // SSRCs are assumed to be sorted in the same order as |rtp_modules|.
>+  for (uint32_t ssrc : ssrcs) {
>+    // Restore state if it previously existed.
>+    const RtpPayloadState* state = nullptr;
>+    auto it = states.find(ssrc);
>+    if (it != states.end()) {
>+      state = &it->second;
>+      shared_frame_id_ = std::max(shared_frame_id_, state->shared_frame_id);
>+    }
>+    params_.push_back(RtpPayloadParams(ssrc, state));
>+  }
>+
>+  // RTP/RTCP initialization.
>+
>+  // We add the highest spatial layer first to ensure it'll be prioritized
>+  // when sending padding, with the hope that the packet rate will be smaller,
>+  // and that it's more important to protect than the lower layers.
>+
>+  // TODO(nisse): Consider moving registration with PacketRouter last, after the
>+  // modules are fully configured.
>+  for (auto& rtp_rtcp : rtp_modules_) {
>+    constexpr bool remb_candidate = true;
>+    transport->packet_router()->AddSendRtpModule(rtp_rtcp.get(),
>+                                                 remb_candidate);
>+  }
>+
>+  for (size_t i = 0; i < rtp_config_.extensions.size(); ++i) {
>+    const std::string& extension = rtp_config_.extensions[i].uri;
>+    int id = rtp_config_.extensions[i].id;
>+    // One-byte-extension local identifiers are in the range 1-14 inclusive.
>+    RTC_DCHECK_GE(id, 1);
>+    RTC_DCHECK_LE(id, 14);
>+    RTC_DCHECK(RtpExtension::IsSupportedForVideo(extension));
>+    for (auto& rtp_rtcp : rtp_modules_) {
>+      RTC_CHECK_EQ(0, rtp_rtcp->RegisterSendRtpHeaderExtension(
>+                          StringToRtpExtensionType(extension), id));
>+    }
>+  }
>+
>+  ConfigureProtection(rtp_config);
>+  ConfigureSsrcs(rtp_config);
>+
>+  if (!rtp_config.mid.empty()) {
>+    for (auto& rtp_rtcp : rtp_modules_) {
>+      rtp_rtcp->SetMid(rtp_config.mid);
>+    }
>+  }
>+
>+  // TODO(pbos): Should we set CNAME on all RTP modules?
>+  rtp_modules_.front()->SetCNAME(rtp_config.c_name.c_str());
>+
>+  for (auto& rtp_rtcp : rtp_modules_) {
>+    rtp_rtcp->RegisterRtcpStatisticsCallback(observers.rtcp_stats);
>+    rtp_rtcp->RegisterSendChannelRtpStatisticsCallback(observers.rtp_stats);
>+    rtp_rtcp->SetMaxRtpPacketSize(rtp_config.max_packet_size);
>+    rtp_rtcp->RegisterVideoSendPayload(rtp_config.payload_type,
>+                                       rtp_config.payload_name.c_str());
>+  }
>+}
>+
>+RtpVideoSender::~RtpVideoSender() {
>+  for (auto& rtp_rtcp : rtp_modules_) {
>+    transport_->packet_router()->RemoveSendRtpModule(rtp_rtcp.get());
>+  }
>+}
>+
>+void RtpVideoSender::RegisterProcessThread(
>+    ProcessThread* module_process_thread) {
>+  RTC_DCHECK_RUN_ON(&module_process_thread_checker_);
>+  RTC_DCHECK(!module_process_thread_);
>+  module_process_thread_ = module_process_thread;
>+
>+  for (auto& rtp_rtcp : rtp_modules_)
>+    module_process_thread_->RegisterModule(rtp_rtcp.get(), RTC_FROM_HERE);
>+}
>+
>+void RtpVideoSender::DeRegisterProcessThread() {
>+  RTC_DCHECK_RUN_ON(&module_process_thread_checker_);
>+  for (auto& rtp_rtcp : rtp_modules_)
>+    module_process_thread_->DeRegisterModule(rtp_rtcp.get());
>+}
>+
>+void RtpVideoSender::SetActive(bool active) {
>+  rtc::CritScope lock(&crit_);
>+  if (active_ == active)
>+    return;
>+  const std::vector<bool> active_modules(rtp_modules_.size(), active);
>+  SetActiveModules(active_modules);
>+}
>+
>+void RtpVideoSender::SetActiveModules(const std::vector<bool> active_modules) {
>+  rtc::CritScope lock(&crit_);
>+  RTC_DCHECK_EQ(rtp_modules_.size(), active_modules.size());
>+  active_ = false;
>+  for (size_t i = 0; i < active_modules.size(); ++i) {
>+    if (active_modules[i]) {
>+      active_ = true;
>+    }
>+    // Sends a kRtcpByeCode when going from true to false.
>+    rtp_modules_[i]->SetSendingStatus(active_modules[i]);
>+    // If set to false this module won't send media.
>+    rtp_modules_[i]->SetSendingMediaStatus(active_modules[i]);
>+  }
>+}
>+
>+bool RtpVideoSender::IsActive() {
>+  rtc::CritScope lock(&crit_);
>+  return active_ && !rtp_modules_.empty();
>+}
>+
>+EncodedImageCallback::Result RtpVideoSender::OnEncodedImage(
>+    const EncodedImage& encoded_image,
>+    const CodecSpecificInfo* codec_specific_info,
>+    const RTPFragmentationHeader* fragmentation) {
>+  rtc::CritScope lock(&crit_);
>+  RTC_DCHECK(!rtp_modules_.empty());
>+  if (!active_)
>+    return Result(Result::ERROR_SEND_FAILED);
>+
>+  size_t stream_index = GetSimulcastIdx(codec_specific_info).value_or(0);
>+  RTC_DCHECK_LT(stream_index, rtp_modules_.size());
>+  RTPVideoHeader rtp_video_header = params_[stream_index].GetRtpVideoHeader(
>+      encoded_image, codec_specific_info);
>+
>+  uint32_t frame_id;
>+  if (!rtp_modules_[stream_index]->Sending()) {
>+    // The payload router could be active but this module isn't sending.
>+    return Result(Result::ERROR_SEND_FAILED);
>+  }
>+  bool send_result = rtp_modules_[stream_index]->SendOutgoingData(
>+      encoded_image._frameType, rtp_config_.payload_type,
>+      encoded_image._timeStamp, encoded_image.capture_time_ms_,
>+      encoded_image._buffer, encoded_image._length, fragmentation,
>+      &rtp_video_header, &frame_id);
>+  if (!send_result)
>+    return Result(Result::ERROR_SEND_FAILED);
>+
>+  return Result(Result::OK, frame_id);
>+}
>+
>+void RtpVideoSender::OnBitrateAllocationUpdated(
>+    const VideoBitrateAllocation& bitrate) {
>+  rtc::CritScope lock(&crit_);
>+  if (IsActive()) {
>+    if (rtp_modules_.size() == 1) {
>+      // If spatial scalability is enabled, it is covered by a single stream.
>+      rtp_modules_[0]->SetVideoBitrateAllocation(bitrate);
>+    } else {
>+      std::vector<absl::optional<VideoBitrateAllocation>> layer_bitrates =
>+          bitrate.GetSimulcastAllocations();
>+      // Simulcast is in use, split the VideoBitrateAllocation into one struct
>+      // per rtp stream, moving over the temporal layer allocation.
>+      for (size_t i = 0; i < rtp_modules_.size(); ++i) {
>+        // The next spatial layer could be used if the current one is
>+        // inactive.
>+        if (layer_bitrates[i]) {
>+          rtp_modules_[i]->SetVideoBitrateAllocation(*layer_bitrates[i]);
>+        }
>+      }
>+    }
>+  }
>+}
>+
>+void RtpVideoSender::ConfigureProtection(const RtpConfig& rtp_config) {
>+  // Consistency of FlexFEC parameters is checked in MaybeCreateFlexfecSender.
>+  const bool flexfec_enabled = (flexfec_sender_ != nullptr);
>+
>+  // Consistency of NACK and RED+ULPFEC parameters is checked in this function.
>+  const bool nack_enabled = rtp_config.nack.rtp_history_ms > 0;
>+  int red_payload_type = rtp_config.ulpfec.red_payload_type;
>+  int ulpfec_payload_type = rtp_config.ulpfec.ulpfec_payload_type;
>+
>+  // Shorthands.
>+  auto IsRedEnabled = [&]() { return red_payload_type >= 0; };
>+  auto IsUlpfecEnabled = [&]() { return ulpfec_payload_type >= 0; };
>+  auto DisableRedAndUlpfec = [&]() {
>+    red_payload_type = -1;
>+    ulpfec_payload_type = -1;
>+  };
>+
>+  if (webrtc::field_trial::IsEnabled("WebRTC-DisableUlpFecExperiment")) {
>+    RTC_LOG(LS_INFO) << "Experiment to disable sending ULPFEC is enabled.";
>+    DisableRedAndUlpfec();
>+  }
>+
>+  // If enabled, FlexFEC takes priority over RED+ULPFEC.
>+  if (flexfec_enabled) {
>+    if (IsUlpfecEnabled()) {
>+      RTC_LOG(LS_INFO)
>+          << "Both FlexFEC and ULPFEC are configured. Disabling ULPFEC.";
>+    }
>+    DisableRedAndUlpfec();
>+  }
>+
>+  // Payload types without picture ID cannot determine that a stream is complete
>+  // without retransmitting FEC, so using ULPFEC + NACK for H.264 (for instance)
>+  // is a waste of bandwidth since FEC packets still have to be transmitted.
>+  // Note that this is not the case with FlexFEC.
>+  if (nack_enabled && IsUlpfecEnabled() &&
>+      !PayloadTypeSupportsSkippingFecPackets(rtp_config.payload_name)) {
>+    RTC_LOG(LS_WARNING)
>+        << "Transmitting payload type without picture ID using "
>+           "NACK+ULPFEC is a waste of bandwidth since ULPFEC packets "
>+           "also have to be retransmitted. Disabling ULPFEC.";
>+    DisableRedAndUlpfec();
>+  }
>+
>+  // Verify payload types.
>+  if (IsUlpfecEnabled() ^ IsRedEnabled()) {
>+    RTC_LOG(LS_WARNING)
>+        << "Only RED or only ULPFEC enabled, but not both. Disabling both.";
>+    DisableRedAndUlpfec();
>+  }
>+
>+  for (auto& rtp_rtcp : rtp_modules_) {
>+    // Set NACK.
>+    rtp_rtcp->SetStorePacketsStatus(true, kMinSendSidePacketHistorySize);
>+    // Set RED/ULPFEC information.
>+    rtp_rtcp->SetUlpfecConfig(red_payload_type, ulpfec_payload_type);
>+  }
>+}
>+
>+bool RtpVideoSender::FecEnabled() const {
>+  const bool flexfec_enabled = (flexfec_sender_ != nullptr);
>+  int ulpfec_payload_type = rtp_config_.ulpfec.ulpfec_payload_type;
>+  return flexfec_enabled || ulpfec_payload_type >= 0;
>+}
>+
>+bool RtpVideoSender::NackEnabled() const {
>+  const bool nack_enabled = rtp_config_.nack.rtp_history_ms > 0;
>+  return nack_enabled;
>+}
>+
>+void RtpVideoSender::DeliverRtcp(const uint8_t* packet, size_t length) {
>+  // Runs on a network thread.
>+  for (auto& rtp_rtcp : rtp_modules_)
>+    rtp_rtcp->IncomingRtcpPacket(packet, length);
>+}
>+
>+void RtpVideoSender::ProtectionRequest(const FecProtectionParams* delta_params,
>+                                       const FecProtectionParams* key_params,
>+                                       uint32_t* sent_video_rate_bps,
>+                                       uint32_t* sent_nack_rate_bps,
>+                                       uint32_t* sent_fec_rate_bps) {
>+  *sent_video_rate_bps = 0;
>+  *sent_nack_rate_bps = 0;
>+  *sent_fec_rate_bps = 0;
>+  for (auto& rtp_rtcp : rtp_modules_) {
>+    uint32_t not_used = 0;
>+    uint32_t module_video_rate = 0;
>+    uint32_t module_fec_rate = 0;
>+    uint32_t module_nack_rate = 0;
>+    rtp_rtcp->SetFecParameters(*delta_params, *key_params);
>+    rtp_rtcp->BitrateSent(&not_used, &module_video_rate, &module_fec_rate,
>+                          &module_nack_rate);
>+    *sent_video_rate_bps += module_video_rate;
>+    *sent_nack_rate_bps += module_nack_rate;
>+    *sent_fec_rate_bps += module_fec_rate;
>+  }
>+}
>+
>+void RtpVideoSender::SetMaxRtpPacketSize(size_t max_rtp_packet_size) {
>+  for (auto& rtp_rtcp : rtp_modules_) {
>+    rtp_rtcp->SetMaxRtpPacketSize(max_rtp_packet_size);
>+  }
>+}
>+
>+void RtpVideoSender::ConfigureSsrcs(const RtpConfig& rtp_config) {
>+  // Configure regular SSRCs.
>+  for (size_t i = 0; i < rtp_config.ssrcs.size(); ++i) {
>+    uint32_t ssrc = rtp_config.ssrcs[i];
>+    RtpRtcp* const rtp_rtcp = rtp_modules_[i].get();
>+    rtp_rtcp->SetSSRC(ssrc);
>+
>+    // Restore RTP state if previous existed.
>+    auto it = suspended_ssrcs_.find(ssrc);
>+    if (it != suspended_ssrcs_.end())
>+      rtp_rtcp->SetRtpState(it->second);
>+  }
>+
>+  // Set up RTX if available.
>+  if (rtp_config.rtx.ssrcs.empty())
>+    return;
>+
>+  // Configure RTX SSRCs.
>+  RTC_DCHECK_EQ(rtp_config.rtx.ssrcs.size(), rtp_config.ssrcs.size());
>+  for (size_t i = 0; i < rtp_config.rtx.ssrcs.size(); ++i) {
>+    uint32_t ssrc = rtp_config.rtx.ssrcs[i];
>+    RtpRtcp* const rtp_rtcp = rtp_modules_[i].get();
>+    rtp_rtcp->SetRtxSsrc(ssrc);
>+    auto it = suspended_ssrcs_.find(ssrc);
>+    if (it != suspended_ssrcs_.end())
>+      rtp_rtcp->SetRtxState(it->second);
>+  }
>+
>+  // Configure RTX payload types.
>+  RTC_DCHECK_GE(rtp_config.rtx.payload_type, 0);
>+  for (auto& rtp_rtcp : rtp_modules_) {
>+    rtp_rtcp->SetRtxSendPayloadType(rtp_config.rtx.payload_type,
>+                                    rtp_config.payload_type);
>+    rtp_rtcp->SetRtxSendStatus(kRtxRetransmitted | kRtxRedundantPayloads);
>+  }
>+  if (rtp_config.ulpfec.red_payload_type != -1 &&
>+      rtp_config.ulpfec.red_rtx_payload_type != -1) {
>+    for (auto& rtp_rtcp : rtp_modules_) {
>+      rtp_rtcp->SetRtxSendPayloadType(rtp_config.ulpfec.red_rtx_payload_type,
>+                                      rtp_config.ulpfec.red_payload_type);
>+    }
>+  }
>+}
>+
>+void RtpVideoSender::OnNetworkAvailability(bool network_available) {
>+  for (auto& rtp_rtcp : rtp_modules_) {
>+    rtp_rtcp->SetRTCPStatus(network_available ? rtp_config_.rtcp_mode
>+                                              : RtcpMode::kOff);
>+  }
>+}
>+
>+std::map<uint32_t, RtpState> RtpVideoSender::GetRtpStates() const {
>+  std::map<uint32_t, RtpState> rtp_states;
>+
>+  for (size_t i = 0; i < rtp_config_.ssrcs.size(); ++i) {
>+    uint32_t ssrc = rtp_config_.ssrcs[i];
>+    RTC_DCHECK_EQ(ssrc, rtp_modules_[i]->SSRC());
>+    rtp_states[ssrc] = rtp_modules_[i]->GetRtpState();
>+  }
>+
>+  for (size_t i = 0; i < rtp_config_.rtx.ssrcs.size(); ++i) {
>+    uint32_t ssrc = rtp_config_.rtx.ssrcs[i];
>+    rtp_states[ssrc] = rtp_modules_[i]->GetRtxState();
>+  }
>+
>+  if (flexfec_sender_) {
>+    uint32_t ssrc = rtp_config_.flexfec.ssrc;
>+    rtp_states[ssrc] = flexfec_sender_->GetRtpState();
>+  }
>+
>+  return rtp_states;
>+}
>+
>+std::map<uint32_t, RtpPayloadState> RtpVideoSender::GetRtpPayloadStates()
>+    const {
>+  rtc::CritScope lock(&crit_);
>+  std::map<uint32_t, RtpPayloadState> payload_states;
>+  for (const auto& param : params_) {
>+    payload_states[param.ssrc()] = param.state();
>+    payload_states[param.ssrc()].shared_frame_id = shared_frame_id_;
>+  }
>+  return payload_states;
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_video_sender.h b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_video_sender.h
>new file mode 100644
>index 00000000000..1735dc751ae
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_video_sender.h
>@@ -0,0 +1,133 @@
>+/*
>+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef CALL_RTP_VIDEO_SENDER_H_
>+#define CALL_RTP_VIDEO_SENDER_H_
>+
>+#include <map>
>+#include <memory>
>+#include <vector>
>+
>+#include "api/call/transport.h"
>+#include "api/video_codecs/video_encoder.h"
>+#include "call/rtp_config.h"
>+#include "call/rtp_payload_params.h"
>+#include "call/rtp_transport_controller_send_interface.h"
>+#include "call/rtp_video_sender_interface.h"
>+#include "common_types.h"  // NOLINT(build/include)
>+#include "logging/rtc_event_log/rtc_event_log.h"
>+#include "modules/rtp_rtcp/include/flexfec_sender.h"
>+#include "modules/rtp_rtcp/source/rtp_video_header.h"
>+#include "modules/utility/include/process_thread.h"
>+#include "rtc_base/constructormagic.h"
>+#include "rtc_base/criticalsection.h"
>+#include "rtc_base/rate_limiter.h"
>+#include "rtc_base/thread_annotations.h"
>+#include "rtc_base/thread_checker.h"
>+
>+namespace webrtc {
>+
>+class RTPFragmentationHeader;
>+class RtpRtcp;
>+class RtpTransportControllerSendInterface;
>+
>+// RtpVideoSender routes outgoing data to the correct sending RTP module, based
>+// on the simulcast layer in RTPVideoHeader.
>+class RtpVideoSender : public RtpVideoSenderInterface {
>+ public:
>+  // Rtp modules are assumed to be sorted in simulcast index order.
>+  RtpVideoSender(
>+      const std::vector<uint32_t>& ssrcs,
>+      std::map<uint32_t, RtpState> suspended_ssrcs,
>+      const std::map<uint32_t, RtpPayloadState>& states,
>+      const RtpConfig& rtp_config,
>+      const RtcpConfig& rtcp_config,
>+      Transport* send_transport,
>+      const RtpSenderObservers& observers,
>+      RtpTransportControllerSendInterface* transport,
>+      RtcEventLog* event_log,
>+      RateLimiter* retransmission_limiter);  // move inside RtpTransport
>+  ~RtpVideoSender() override;
>+
>+  // RegisterProcessThread register |module_process_thread| with those objects
>+  // that use it. Registration has to happen on the thread were
>+  // |module_process_thread| was created (libjingle's worker thread).
>+  // TODO(perkj): Replace the use of |module_process_thread| with a TaskQueue,
>+  // maybe |worker_queue|.
>+  void RegisterProcessThread(ProcessThread* module_process_thread) override;
>+  void DeRegisterProcessThread() override;
>+
>+  // RtpVideoSender will only route packets if being active, all packets will be
>+  // dropped otherwise.
>+  void SetActive(bool active) override;
>+  // Sets the sending status of the rtp modules and appropriately sets the
>+  // payload router to active if any rtp modules are active.
>+  void SetActiveModules(const std::vector<bool> active_modules) override;
>+  bool IsActive() override;
>+
>+  void OnNetworkAvailability(bool network_available) override;
>+  std::map<uint32_t, RtpState> GetRtpStates() const override;
>+  std::map<uint32_t, RtpPayloadState> GetRtpPayloadStates() const override;
>+
>+  bool FecEnabled() const override;
>+
>+  bool NackEnabled() const override;
>+
>+  void DeliverRtcp(const uint8_t* packet, size_t length) override;
>+
>+  void ProtectionRequest(const FecProtectionParams* delta_params,
>+                         const FecProtectionParams* key_params,
>+                         uint32_t* sent_video_rate_bps,
>+                         uint32_t* sent_nack_rate_bps,
>+                         uint32_t* sent_fec_rate_bps) override;
>+
>+  void SetMaxRtpPacketSize(size_t max_rtp_packet_size) override;
>+
>+  // Implements EncodedImageCallback.
>+  // Returns 0 if the packet was routed / sent, -1 otherwise.
>+  EncodedImageCallback::Result OnEncodedImage(
>+      const EncodedImage& encoded_image,
>+      const CodecSpecificInfo* codec_specific_info,
>+      const RTPFragmentationHeader* fragmentation) override;
>+
>+  void OnBitrateAllocationUpdated(
>+      const VideoBitrateAllocation& bitrate) override;
>+
>+ private:
>+  void UpdateModuleSendingState() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
>+  void ConfigureProtection(const RtpConfig& rtp_config);
>+  void ConfigureSsrcs(const RtpConfig& rtp_config);
>+
>+  rtc::CriticalSection crit_;
>+  bool active_ RTC_GUARDED_BY(crit_);
>+
>+  ProcessThread* module_process_thread_;
>+  rtc::ThreadChecker module_process_thread_checker_;
>+  std::map<uint32_t, RtpState> suspended_ssrcs_;
>+
>+  std::unique_ptr<FlexfecSender> flexfec_sender_;
>+  // Rtp modules are assumed to be sorted in simulcast index order.
>+  const std::vector<std::unique_ptr<RtpRtcp>> rtp_modules_;
>+  const RtpConfig rtp_config_;
>+  RtpTransportControllerSendInterface* const transport_;
>+
>+  // When using the generic descriptor we want all simulcast streams to share
>+  // one frame id space (so that the SFU can switch stream without having to
>+  // rewrite the frame id), therefore |shared_frame_id| has to live in a place
>+  // where we are aware of all the different streams.
>+  int64_t shared_frame_id_ = 0;
>+  std::vector<RtpPayloadParams> params_ RTC_GUARDED_BY(crit_);
>+
>+  RTC_DISALLOW_COPY_AND_ASSIGN(RtpVideoSender);
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // CALL_RTP_VIDEO_SENDER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_video_sender_interface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_video_sender_interface.h
>new file mode 100644
>index 00000000000..c69f1bab02a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_video_sender_interface.h
>@@ -0,0 +1,60 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef CALL_RTP_VIDEO_SENDER_INTERFACE_H_
>+#define CALL_RTP_VIDEO_SENDER_INTERFACE_H_
>+
>+#include <map>
>+#include <vector>
>+
>+#include "call/rtp_config.h"
>+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>+#include "modules/utility/include/process_thread.h"
>+#include "modules/video_coding/include/video_codec_interface.h"
>+
>+namespace webrtc {
>+class VideoBitrateAllocation;
>+struct FecProtectionParams;
>+
>+class RtpVideoSenderInterface : public EncodedImageCallback {
>+ public:
>+  virtual void RegisterProcessThread(ProcessThread* module_process_thread) = 0;
>+  virtual void DeRegisterProcessThread() = 0;
>+
>+  // RtpVideoSender will only route packets if being active, all
>+  // packets will be dropped otherwise.
>+  virtual void SetActive(bool active) = 0;
>+  // Sets the sending status of the rtp modules and appropriately sets the
>+  // RtpVideoSender to active if any rtp modules are active.
>+  virtual void SetActiveModules(const std::vector<bool> active_modules) = 0;
>+  virtual bool IsActive() = 0;
>+
>+  virtual void OnNetworkAvailability(bool network_available) = 0;
>+  virtual std::map<uint32_t, RtpState> GetRtpStates() const = 0;
>+  virtual std::map<uint32_t, RtpPayloadState> GetRtpPayloadStates() const = 0;
>+
>+  virtual bool FecEnabled() const = 0;
>+
>+  virtual bool NackEnabled() const = 0;
>+
>+  virtual void DeliverRtcp(const uint8_t* packet, size_t length) = 0;
>+
>+  virtual void ProtectionRequest(const FecProtectionParams* delta_params,
>+                                 const FecProtectionParams* key_params,
>+                                 uint32_t* sent_video_rate_bps,
>+                                 uint32_t* sent_nack_rate_bps,
>+                                 uint32_t* sent_fec_rate_bps) = 0;
>+
>+  virtual void SetMaxRtpPacketSize(size_t max_rtp_packet_size) = 0;
>+  virtual void OnBitrateAllocationUpdated(
>+      const VideoBitrateAllocation& bitrate) = 0;
>+};
>+}  // namespace webrtc
>+#endif  // CALL_RTP_VIDEO_SENDER_INTERFACE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_video_sender_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_video_sender_unittest.cc
>new file mode 100644
>index 00000000000..b819e696d93
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtp_video_sender_unittest.cc
>@@ -0,0 +1,299 @@
>+/*
>+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <memory>
>+#include <string>
>+
>+#include "call/rtp_transport_controller_send.h"
>+#include "call/rtp_video_sender.h"
>+#include "modules/video_coding/include/video_codec_interface.h"
>+#include "rtc_base/rate_limiter.h"
>+#include "test/field_trial.h"
>+#include "test/gmock.h"
>+#include "test/gtest.h"
>+#include "test/mock_transport.h"
>+#include "video/call_stats.h"
>+#include "video/send_delay_stats.h"
>+#include "video/send_statistics_proxy.h"
>+
>+using ::testing::_;
>+using ::testing::AnyNumber;
>+using ::testing::Invoke;
>+using ::testing::NiceMock;
>+using ::testing::Return;
>+using ::testing::Unused;
>+
>+namespace webrtc {
>+namespace {
>+const int8_t kPayloadType = 96;
>+const uint32_t kSsrc1 = 12345;
>+const uint32_t kSsrc2 = 23456;
>+const int16_t kInitialPictureId1 = 222;
>+const int16_t kInitialPictureId2 = 44;
>+const int16_t kInitialTl0PicIdx1 = 99;
>+const int16_t kInitialTl0PicIdx2 = 199;
>+const int64_t kRetransmitWindowSizeMs = 500;
>+
>+class MockRtcpIntraFrameObserver : public RtcpIntraFrameObserver {
>+ public:
>+  MOCK_METHOD1(OnReceivedIntraFrameRequest, void(uint32_t));
>+};
>+
>+class MockOverheadObserver : public OverheadObserver {
>+ public:
>+  MOCK_METHOD1(OnOverheadChanged, void(size_t overhead_bytes_per_packet));
>+};
>+
>+class MockCongestionObserver : public NetworkChangedObserver {
>+ public:
>+  MOCK_METHOD4(OnNetworkChanged,
>+               void(uint32_t bitrate_bps,
>+                    uint8_t fraction_loss,
>+                    int64_t rtt_ms,
>+                    int64_t probing_interval_ms));
>+};
>+
>+RtpSenderObservers CreateObservers(
>+    RtcpRttStats* rtcp_rtt_stats,
>+    RtcpIntraFrameObserver* intra_frame_callback,
>+    RtcpStatisticsCallback* rtcp_stats,
>+    StreamDataCountersCallback* rtp_stats,
>+    BitrateStatisticsObserver* bitrate_observer,
>+    FrameCountObserver* frame_count_observer,
>+    RtcpPacketTypeCounterObserver* rtcp_type_observer,
>+    SendSideDelayObserver* send_delay_observer,
>+    SendPacketObserver* send_packet_observer,
>+    OverheadObserver* overhead_observer) {
>+  RtpSenderObservers observers;
>+  observers.rtcp_rtt_stats = rtcp_rtt_stats;
>+  observers.intra_frame_callback = intra_frame_callback;
>+  observers.rtcp_stats = rtcp_stats;
>+  observers.rtp_stats = rtp_stats;
>+  observers.bitrate_observer = bitrate_observer;
>+  observers.frame_count_observer = frame_count_observer;
>+  observers.rtcp_type_observer = rtcp_type_observer;
>+  observers.send_delay_observer = send_delay_observer;
>+  observers.send_packet_observer = send_packet_observer;
>+  observers.overhead_observer = overhead_observer;
>+  return observers;
>+}
>+
>+class RtpVideoSenderTestFixture {
>+ public:
>+  RtpVideoSenderTestFixture(
>+      const std::vector<uint32_t>& ssrcs,
>+      int payload_type,
>+      const std::map<uint32_t, RtpPayloadState>& suspended_payload_states)
>+      : clock_(0),
>+        config_(&transport_),
>+        send_delay_stats_(&clock_),
>+        transport_controller_(&clock_, &event_log_, nullptr, bitrate_config_),
>+        process_thread_(ProcessThread::Create("test_thread")),
>+        call_stats_(&clock_, process_thread_.get()),
>+        stats_proxy_(&clock_,
>+                     config_,
>+                     VideoEncoderConfig::ContentType::kRealtimeVideo),
>+        retransmission_rate_limiter_(&clock_, kRetransmitWindowSizeMs) {
>+    for (uint32_t ssrc : ssrcs) {
>+      config_.rtp.ssrcs.push_back(ssrc);
>+    }
>+    config_.rtp.payload_type = payload_type;
>+    std::map<uint32_t, RtpState> suspended_ssrcs;
>+    router_ = absl::make_unique<RtpVideoSender>(
>+        config_.rtp.ssrcs, suspended_ssrcs, suspended_payload_states,
>+        config_.rtp, config_.rtcp, &transport_,
>+        CreateObservers(&call_stats_, &encoder_feedback_, &stats_proxy_,
>+                        &stats_proxy_, &stats_proxy_, &stats_proxy_,
>+                        &stats_proxy_, &stats_proxy_, &send_delay_stats_,
>+                        &overhead_observer_),
>+        &transport_controller_, &event_log_, &retransmission_rate_limiter_);
>+  }
>+
>+  RtpVideoSender* router() { return router_.get(); }
>+
>+ private:
>+  NiceMock<MockTransport> transport_;
>+  NiceMock<MockCongestionObserver> congestion_observer_;
>+  NiceMock<MockOverheadObserver> overhead_observer_;
>+  NiceMock<MockRtcpIntraFrameObserver> encoder_feedback_;
>+  SimulatedClock clock_;
>+  RtcEventLogNullImpl event_log_;
>+  VideoSendStream::Config config_;
>+  SendDelayStats send_delay_stats_;
>+  BitrateConstraints bitrate_config_;
>+  RtpTransportControllerSend transport_controller_;
>+  std::unique_ptr<ProcessThread> process_thread_;
>+  CallStats call_stats_;
>+  SendStatisticsProxy stats_proxy_;
>+  RateLimiter retransmission_rate_limiter_;
>+  std::unique_ptr<RtpVideoSender> router_;
>+};
>+}  // namespace
>+
>+TEST(RtpVideoSenderTest, SendOnOneModule) {
>+  uint8_t payload = 'a';
>+  EncodedImage encoded_image;
>+  encoded_image._timeStamp = 1;
>+  encoded_image.capture_time_ms_ = 2;
>+  encoded_image._frameType = kVideoFrameKey;
>+  encoded_image._buffer = &payload;
>+  encoded_image._length = 1;
>+
>+  RtpVideoSenderTestFixture test({kSsrc1}, kPayloadType, {});
>+  EXPECT_NE(
>+      EncodedImageCallback::Result::OK,
>+      test.router()->OnEncodedImage(encoded_image, nullptr, nullptr).error);
>+
>+  test.router()->SetActive(true);
>+  EXPECT_EQ(
>+      EncodedImageCallback::Result::OK,
>+      test.router()->OnEncodedImage(encoded_image, nullptr, nullptr).error);
>+
>+  test.router()->SetActive(false);
>+  EXPECT_NE(
>+      EncodedImageCallback::Result::OK,
>+      test.router()->OnEncodedImage(encoded_image, nullptr, nullptr).error);
>+
>+  test.router()->SetActive(true);
>+  EXPECT_EQ(
>+      EncodedImageCallback::Result::OK,
>+      test.router()->OnEncodedImage(encoded_image, nullptr, nullptr).error);
>+}
>+
>+TEST(RtpVideoSenderTest, SendSimulcastSetActive) {
>+  uint8_t payload = 'a';
>+  EncodedImage encoded_image;
>+  encoded_image._timeStamp = 1;
>+  encoded_image.capture_time_ms_ = 2;
>+  encoded_image._frameType = kVideoFrameKey;
>+  encoded_image._buffer = &payload;
>+  encoded_image._length = 1;
>+
>+  RtpVideoSenderTestFixture test({kSsrc1, kSsrc2}, kPayloadType, {});
>+
>+  CodecSpecificInfo codec_info_1;
>+  memset(&codec_info_1, 0, sizeof(CodecSpecificInfo));
>+  codec_info_1.codecType = kVideoCodecVP8;
>+  codec_info_1.codecSpecific.VP8.simulcastIdx = 0;
>+
>+  test.router()->SetActive(true);
>+  EXPECT_EQ(EncodedImageCallback::Result::OK,
>+            test.router()
>+                ->OnEncodedImage(encoded_image, &codec_info_1, nullptr)
>+                .error);
>+
>+  CodecSpecificInfo codec_info_2;
>+  memset(&codec_info_2, 0, sizeof(CodecSpecificInfo));
>+  codec_info_2.codecType = kVideoCodecVP8;
>+  codec_info_2.codecSpecific.VP8.simulcastIdx = 1;
>+  EXPECT_EQ(EncodedImageCallback::Result::OK,
>+            test.router()
>+                ->OnEncodedImage(encoded_image, &codec_info_2, nullptr)
>+                .error);
>+
>+  // Inactive.
>+  test.router()->SetActive(false);
>+  EXPECT_NE(EncodedImageCallback::Result::OK,
>+            test.router()
>+                ->OnEncodedImage(encoded_image, &codec_info_1, nullptr)
>+                .error);
>+  EXPECT_NE(EncodedImageCallback::Result::OK,
>+            test.router()
>+                ->OnEncodedImage(encoded_image, &codec_info_2, nullptr)
>+                .error);
>+}
>+
>+// Tests how setting individual rtp modules to active affects the overall
>+// behavior of the payload router. First sets one module to active and checks
>+// that outgoing data can be sent on this module, and checks that no data can
>+// be sent if both modules are inactive.
>+TEST(RtpVideoSenderTest, SendSimulcastSetActiveModules) {
>+  uint8_t payload = 'a';
>+  EncodedImage encoded_image;
>+  encoded_image._timeStamp = 1;
>+  encoded_image.capture_time_ms_ = 2;
>+  encoded_image._frameType = kVideoFrameKey;
>+  encoded_image._buffer = &payload;
>+  encoded_image._length = 1;
>+
>+  RtpVideoSenderTestFixture test({kSsrc1, kSsrc2}, kPayloadType, {});
>+  CodecSpecificInfo codec_info_1;
>+  memset(&codec_info_1, 0, sizeof(CodecSpecificInfo));
>+  codec_info_1.codecType = kVideoCodecVP8;
>+  codec_info_1.codecSpecific.VP8.simulcastIdx = 0;
>+  CodecSpecificInfo codec_info_2;
>+  memset(&codec_info_2, 0, sizeof(CodecSpecificInfo));
>+  codec_info_2.codecType = kVideoCodecVP8;
>+  codec_info_2.codecSpecific.VP8.simulcastIdx = 1;
>+
>+  // Only setting one stream to active will still set the payload router to
>+  // active and allow sending data on the active stream.
>+  std::vector<bool> active_modules({true, false});
>+  test.router()->SetActiveModules(active_modules);
>+  EXPECT_EQ(EncodedImageCallback::Result::OK,
>+            test.router()
>+                ->OnEncodedImage(encoded_image, &codec_info_1, nullptr)
>+                .error);
>+
>+  // Setting both streams to inactive will turn the payload router to
>+  // inactive.
>+  active_modules = {false, false};
>+  test.router()->SetActiveModules(active_modules);
>+  // An incoming encoded image will not ask the module to send outgoing data
>+  // because the payload router is inactive.
>+  EXPECT_NE(EncodedImageCallback::Result::OK,
>+            test.router()
>+                ->OnEncodedImage(encoded_image, &codec_info_1, nullptr)
>+                .error);
>+  EXPECT_NE(EncodedImageCallback::Result::OK,
>+            test.router()
>+                ->OnEncodedImage(encoded_image, &codec_info_2, nullptr)
>+                .error);
>+}
>+
>+TEST(RtpVideoSenderTest, CreateWithNoPreviousStates) {
>+  RtpVideoSenderTestFixture test({kSsrc1, kSsrc2}, kPayloadType, {});
>+  test.router()->SetActive(true);
>+
>+  std::map<uint32_t, RtpPayloadState> initial_states =
>+      test.router()->GetRtpPayloadStates();
>+  EXPECT_EQ(2u, initial_states.size());
>+  EXPECT_NE(initial_states.find(kSsrc1), initial_states.end());
>+  EXPECT_NE(initial_states.find(kSsrc2), initial_states.end());
>+}
>+
>+TEST(RtpVideoSenderTest, CreateWithPreviousStates) {
>+  const int64_t kState1SharedFrameId = 123;
>+  const int64_t kState2SharedFrameId = 234;
>+  RtpPayloadState state1;
>+  state1.picture_id = kInitialPictureId1;
>+  state1.tl0_pic_idx = kInitialTl0PicIdx1;
>+  state1.shared_frame_id = kState1SharedFrameId;
>+  RtpPayloadState state2;
>+  state2.picture_id = kInitialPictureId2;
>+  state2.tl0_pic_idx = kInitialTl0PicIdx2;
>+  state2.shared_frame_id = kState2SharedFrameId;
>+  std::map<uint32_t, RtpPayloadState> states = {{kSsrc1, state1},
>+                                                {kSsrc2, state2}};
>+
>+  RtpVideoSenderTestFixture test({kSsrc1, kSsrc2}, kPayloadType, states);
>+  test.router()->SetActive(true);
>+
>+  std::map<uint32_t, RtpPayloadState> initial_states =
>+      test.router()->GetRtpPayloadStates();
>+  EXPECT_EQ(2u, initial_states.size());
>+  EXPECT_EQ(kInitialPictureId1, initial_states[kSsrc1].picture_id);
>+  EXPECT_EQ(kInitialTl0PicIdx1, initial_states[kSsrc1].tl0_pic_idx);
>+  EXPECT_EQ(kInitialPictureId2, initial_states[kSsrc2].picture_id);
>+  EXPECT_EQ(kInitialTl0PicIdx2, initial_states[kSsrc2].tl0_pic_idx);
>+  EXPECT_EQ(kState2SharedFrameId, initial_states[kSsrc1].shared_frame_id);
>+  EXPECT_EQ(kState2SharedFrameId, initial_states[kSsrc2].shared_frame_id);
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtx_receive_stream.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtx_receive_stream.cc
>index 09200791f88..5d5fec19bda 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtx_receive_stream.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtx_receive_stream.cc
>@@ -63,8 +63,7 @@ void RtxReceiveStream::OnRtpPacket(const RtpPacketReceived& rtx_packet) {
>   media_packet.set_recovered(true);
> 
>   // Skip the RTX header.
>-  rtc::ArrayView<const uint8_t> rtx_payload =
>-      payload.subview(kRtxHeaderSize);
>+  rtc::ArrayView<const uint8_t> rtx_payload = payload.subview(kRtxHeaderSize);
> 
>   uint8_t* media_payload = media_packet.AllocatePayload(rtx_payload.size());
>   RTC_DCHECK(media_payload != nullptr);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtx_receive_stream_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtx_receive_stream_unittest.cc
>index 65ab82b0dd7..a9cfd53b3c9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtx_receive_stream_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/rtx_receive_stream_unittest.cc
>@@ -36,7 +36,7 @@ constexpr uint8_t kRtxPacket[] = {
>     0x11, 0x11, 0x11, 0x11,  // Timestamp.
>     0x22, 0x22, 0x22, 0x22,  // SSRC.
>     // RTX header.
>-    0x56, 0x57,              // Orig seqno.
>+    0x56, 0x57,  // Orig seqno.
>     // Payload.
>     0xee,
> };
>@@ -50,7 +50,7 @@ constexpr uint8_t kRtxPacketWithCVO[] = {
>     0xbe, 0xde, 0x00, 0x01,  // Extension header.
>     0x30, 0x01, 0x00, 0x00,  // 90 degree rotation.
>     // RTX header.
>-    0x56, 0x57,              // Orig seqno.
>+    0x56, 0x57,  // Orig seqno.
>     // Payload.
>     0xee,
> };
>@@ -73,8 +73,8 @@ TEST(RtxReceiveStreamTest, RestoresPacketPayload) {
>   RtpPacketReceived rtx_packet;
>   EXPECT_TRUE(rtx_packet.Parse(rtc::ArrayView<const uint8_t>(kRtxPacket)));
> 
>-  EXPECT_CALL(media_sink, OnRtpPacket(_)).WillOnce(testing::Invoke(
>-      [](const RtpPacketReceived& packet) {
>+  EXPECT_CALL(media_sink, OnRtpPacket(_))
>+      .WillOnce(testing::Invoke([](const RtpPacketReceived& packet) {
>         EXPECT_EQ(packet.SequenceNumber(), kMediaSeqno);
>         EXPECT_EQ(packet.Ssrc(), kMediaSSRC);
>         EXPECT_EQ(packet.PayloadType(), kMediaPayloadType);
>@@ -124,15 +124,15 @@ TEST(RtxReceiveStreamTest, CopiesRtpHeaderExtensions) {
>   RtpHeaderExtensionMap extension_map;
>   extension_map.RegisterByType(3, kRtpExtensionVideoRotation);
>   RtpPacketReceived rtx_packet(&extension_map);
>-  EXPECT_TRUE(rtx_packet.Parse(
>-      rtc::ArrayView<const uint8_t>(kRtxPacketWithCVO)));
>+  EXPECT_TRUE(
>+      rtx_packet.Parse(rtc::ArrayView<const uint8_t>(kRtxPacketWithCVO)));
> 
>   VideoRotation rotation = kVideoRotation_0;
>   EXPECT_TRUE(rtx_packet.GetExtension<VideoOrientation>(&rotation));
>   EXPECT_EQ(kVideoRotation_90, rotation);
> 
>-  EXPECT_CALL(media_sink, OnRtpPacket(_)).WillOnce(testing::Invoke(
>-      [](const RtpPacketReceived& packet) {
>+  EXPECT_CALL(media_sink, OnRtpPacket(_))
>+      .WillOnce(testing::Invoke([](const RtpPacketReceived& packet) {
>         EXPECT_EQ(packet.SequenceNumber(), kMediaSeqno);
>         EXPECT_EQ(packet.Ssrc(), kMediaSSRC);
>         EXPECT_EQ(packet.PayloadType(), kMediaPayloadType);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/simulated_network.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/simulated_network.cc
>new file mode 100644
>index 00000000000..ee58dce7182
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/simulated_network.cc
>@@ -0,0 +1,186 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "call/simulated_network.h"
>+
>+#include <algorithm>
>+#include <cmath>
>+#include <utility>
>+
>+namespace webrtc {
>+
>+SimulatedNetwork::SimulatedNetwork(SimulatedNetwork::Config config,
>+                                   uint64_t random_seed)
>+    : random_(random_seed), bursting_(false) {
>+  SetConfig(config);
>+}
>+
>+SimulatedNetwork::~SimulatedNetwork() = default;
>+
>+void SimulatedNetwork::SetConfig(const SimulatedNetwork::Config& config) {
>+  rtc::CritScope crit(&config_lock_);
>+  config_ = config;  // Shallow copy of the struct.
>+  double prob_loss = config.loss_percent / 100.0;
>+  if (config_.avg_burst_loss_length == -1) {
>+    // Uniform loss
>+    prob_loss_bursting_ = prob_loss;
>+    prob_start_bursting_ = prob_loss;
>+  } else {
>+    // Lose packets according to a gilbert-elliot model.
>+    int avg_burst_loss_length = config.avg_burst_loss_length;
>+    int min_avg_burst_loss_length = std::ceil(prob_loss / (1 - prob_loss));
>+
>+    RTC_CHECK_GT(avg_burst_loss_length, min_avg_burst_loss_length)
>+        << "For a total packet loss of " << config.loss_percent << "%% then"
>+        << " avg_burst_loss_length must be " << min_avg_burst_loss_length + 1
>+        << " or higher.";
>+
>+    prob_loss_bursting_ = (1.0 - 1.0 / avg_burst_loss_length);
>+    prob_start_bursting_ = prob_loss / (1 - prob_loss) / avg_burst_loss_length;
>+  }
>+}
>+
>+void SimulatedNetwork::PauseTransmissionUntil(int64_t until_us) {
>+  rtc::CritScope crit(&config_lock_);
>+  pause_transmission_until_us_ = until_us;
>+}
>+
>+bool SimulatedNetwork::EnqueuePacket(PacketInFlightInfo packet) {
>+  Config config;
>+  {
>+    rtc::CritScope crit(&config_lock_);
>+    config = config_;
>+  }
>+  rtc::CritScope crit(&process_lock_);
>+  if (config.queue_length_packets > 0 &&
>+      capacity_link_.size() >= config.queue_length_packets) {
>+    // Too many packet on the link, drop this one.
>+    return false;
>+  }
>+
>+  // Delay introduced by the link capacity.
>+  int64_t capacity_delay_ms = 0;
>+  if (config.link_capacity_kbps > 0) {
>+    // Using bytes per millisecond to avoid losing precision.
>+    const int64_t bytes_per_millisecond = config.link_capacity_kbps / 8;
>+    // To round to the closest millisecond we add half a milliseconds worth of
>+    // bytes to the delay calculation.
>+    capacity_delay_ms = (packet.size + capacity_delay_error_bytes_ +
>+                         bytes_per_millisecond / 2) /
>+                        bytes_per_millisecond;
>+    capacity_delay_error_bytes_ +=
>+        packet.size - capacity_delay_ms * bytes_per_millisecond;
>+  }
>+  int64_t network_start_time_us = packet.send_time_us;
>+
>+  {
>+    rtc::CritScope crit(&config_lock_);
>+    if (pause_transmission_until_us_) {
>+      network_start_time_us =
>+          std::max(network_start_time_us, *pause_transmission_until_us_);
>+      pause_transmission_until_us_.reset();
>+    }
>+  }
>+  // Check if there already are packets on the link and change network start
>+  // time forward if there is.
>+  if (!capacity_link_.empty() &&
>+      network_start_time_us < capacity_link_.back().arrival_time_us)
>+    network_start_time_us = capacity_link_.back().arrival_time_us;
>+
>+  int64_t arrival_time_us = network_start_time_us + capacity_delay_ms * 1000;
>+  capacity_link_.push({packet, arrival_time_us});
>+  return true;
>+}
>+
>+absl::optional<int64_t> SimulatedNetwork::NextDeliveryTimeUs() const {
>+  if (!delay_link_.empty())
>+    return delay_link_.begin()->arrival_time_us;
>+  return absl::nullopt;
>+}
>+std::vector<PacketDeliveryInfo> SimulatedNetwork::DequeueDeliverablePackets(
>+    int64_t receive_time_us) {
>+  int64_t time_now_us = receive_time_us;
>+  Config config;
>+  double prob_loss_bursting;
>+  double prob_start_bursting;
>+  {
>+    rtc::CritScope crit(&config_lock_);
>+    config = config_;
>+    prob_loss_bursting = prob_loss_bursting_;
>+    prob_start_bursting = prob_start_bursting_;
>+  }
>+  {
>+    rtc::CritScope crit(&process_lock_);
>+    // Check the capacity link first.
>+    if (!capacity_link_.empty()) {
>+      int64_t last_arrival_time_us =
>+          delay_link_.empty() ? -1 : delay_link_.back().arrival_time_us;
>+      bool needs_sort = false;
>+      while (!capacity_link_.empty() &&
>+             time_now_us >= capacity_link_.front().arrival_time_us) {
>+        // Time to get this packet.
>+        PacketInfo packet = std::move(capacity_link_.front());
>+        capacity_link_.pop();
>+
>+        // Drop packets at an average rate of |config_.loss_percent| with
>+        // and average loss burst length of |config_.avg_burst_loss_length|.
>+        if ((bursting_ && random_.Rand<double>() < prob_loss_bursting) ||
>+            (!bursting_ && random_.Rand<double>() < prob_start_bursting)) {
>+          bursting_ = true;
>+          continue;
>+        } else {
>+          bursting_ = false;
>+        }
>+
>+        int64_t arrival_time_jitter_us = std::max(
>+            random_.Gaussian(config.queue_delay_ms * 1000,
>+                             config.delay_standard_deviation_ms * 1000),
>+            0.0);
>+
>+        // If reordering is not allowed then adjust arrival_time_jitter
>+        // to make sure all packets are sent in order.
>+        if (!config.allow_reordering && !delay_link_.empty() &&
>+            packet.arrival_time_us + arrival_time_jitter_us <
>+                last_arrival_time_us) {
>+          arrival_time_jitter_us =
>+              last_arrival_time_us - packet.arrival_time_us;
>+        }
>+        packet.arrival_time_us += arrival_time_jitter_us;
>+        if (packet.arrival_time_us >= last_arrival_time_us) {
>+          last_arrival_time_us = packet.arrival_time_us;
>+        } else {
>+          needs_sort = true;
>+        }
>+        delay_link_.emplace_back(std::move(packet));
>+      }
>+
>+      if (needs_sort) {
>+        // Packet(s) arrived out of order, make sure list is sorted.
>+        std::sort(delay_link_.begin(), delay_link_.end(),
>+                  [](const PacketInfo& p1, const PacketInfo& p2) {
>+                    return p1.arrival_time_us < p2.arrival_time_us;
>+                  });
>+      }
>+    }
>+
>+    std::vector<PacketDeliveryInfo> packets_to_deliver;
>+    // Check the extra delay queue.
>+    while (!delay_link_.empty() &&
>+           time_now_us >= delay_link_.front().arrival_time_us) {
>+      PacketInfo packet_info = delay_link_.front();
>+      packets_to_deliver.emplace_back(
>+          PacketDeliveryInfo(packet_info.packet, packet_info.arrival_time_us));
>+      delay_link_.pop_front();
>+    }
>+    return packets_to_deliver;
>+  }
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/simulated_network.h b/Source/ThirdParty/libwebrtc/Source/webrtc/call/simulated_network.h
>new file mode 100644
>index 00000000000..0ddd1f85fc5
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/simulated_network.h
>@@ -0,0 +1,80 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#ifndef CALL_SIMULATED_NETWORK_H_
>+#define CALL_SIMULATED_NETWORK_H_
>+
>+#include <deque>
>+#include <queue>
>+#include <vector>
>+
>+#include "absl/memory/memory.h"
>+#include "absl/types/optional.h"
>+#include "api/test/simulated_network.h"
>+#include "rtc_base/criticalsection.h"
>+#include "rtc_base/random.h"
>+#include "rtc_base/thread_annotations.h"
>+
>+namespace webrtc {
>+
>+// Class simulating a network link. This is a simple and naive solution just
>+// faking capacity and adding an extra transport delay in addition to the
>+// capacity introduced delay.
>+class SimulatedNetwork : public NetworkSimulationInterface {
>+ public:
>+  using Config = NetworkSimulationInterface::SimulatedNetworkConfig;
>+  explicit SimulatedNetwork(Config config, uint64_t random_seed = 1);
>+  ~SimulatedNetwork() override;
>+
>+  // Sets a new configuration. This won't affect packets already in the pipe.
>+  void SetConfig(const Config& config) override;
>+  void PauseTransmissionUntil(int64_t until_us);
>+
>+  // NetworkSimulationInterface
>+  bool EnqueuePacket(PacketInFlightInfo packet) override;
>+  std::vector<PacketDeliveryInfo> DequeueDeliverablePackets(
>+      int64_t receive_time_us) override;
>+
>+  absl::optional<int64_t> NextDeliveryTimeUs() const override;
>+
>+ private:
>+  struct PacketInfo {
>+    PacketInFlightInfo packet;
>+    int64_t arrival_time_us;
>+  };
>+  rtc::CriticalSection config_lock_;
>+
>+  // |process_lock| guards the data structures involved in delay and loss
>+  // processes, such as the packet queues.
>+  rtc::CriticalSection process_lock_;
>+  std::queue<PacketInfo> capacity_link_ RTC_GUARDED_BY(process_lock_);
>+  Random random_;
>+
>+  std::deque<PacketInfo> delay_link_;
>+
>+  // Link configuration.
>+  Config config_ RTC_GUARDED_BY(config_lock_);
>+  absl::optional<int64_t> pause_transmission_until_us_
>+      RTC_GUARDED_BY(config_lock_);
>+
>+  // Are we currently dropping a burst of packets?
>+  bool bursting_;
>+
>+  // The probability to drop the packet if we are currently dropping a
>+  // burst of packet
>+  double prob_loss_bursting_ RTC_GUARDED_BY(config_lock_);
>+
>+  // The probability to drop a burst of packets.
>+  double prob_start_bursting_ RTC_GUARDED_BY(config_lock_);
>+  int64_t capacity_delay_error_bytes_ = 0;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // CALL_SIMULATED_NETWORK_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/ssrc_binding_observer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/call/ssrc_binding_observer.h
>index 73a8e8e5864..ada505610fa 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/ssrc_binding_observer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/ssrc_binding_observer.h
>@@ -12,8 +12,6 @@
> 
> #include <string>
> 
>-#include "rtc_base/basictypes.h"
>-
> namespace webrtc {
> 
> // With newer versions of SDP, SSRC is often not explicitly signaled and must
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/syncable.h b/Source/ThirdParty/libwebrtc/Source/webrtc/call/syncable.h
>index a97990b2be0..a914793d78a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/syncable.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/syncable.h
>@@ -16,7 +16,7 @@
> 
> #include <stdint.h>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> 
> namespace webrtc {
> 
>@@ -34,7 +34,7 @@ class Syncable {
>   virtual ~Syncable();
> 
>   virtual int id() const = 0;
>-  virtual rtc::Optional<Info> GetInfo() const = 0;
>+  virtual absl::optional<Info> GetInfo() const = 0;
>   virtual uint32_t GetPlayoutTimestamp() const = 0;
>   virtual void SetMinimumPlayoutDelay(int delay_ms) = 0;
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_network_pipe_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/test/fake_network_pipe_unittest.cc
>similarity index 62%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_network_pipe_unittest.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/call/test/fake_network_pipe_unittest.cc
>index 759003b294a..0578f33191e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_network_pipe_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/test/fake_network_pipe_unittest.cc
>@@ -8,12 +8,14 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>+#include "call/fake_network_pipe.h"
>+
> #include <memory>
> 
> #include "call/call.h"
>+#include "call/simulated_network.h"
> #include "modules/rtp_rtcp/include/rtp_header_parser.h"
> #include "system_wrappers/include/clock.h"
>-#include "test/fake_network_pipe.h"
> #include "test/gmock.h"
> #include "test/gtest.h"
> 
>@@ -24,37 +26,27 @@ using ::testing::Invoke;
> 
> namespace webrtc {
> 
>-class TestDemuxer : public Demuxer {
>+class MockReceiver : public PacketReceiver {
>  public:
>-  void IncomingPacket(NetworkPacket* packet) {
>-    DeliverPacket(packet, PacketTime());
>-  }
>-
>-  MOCK_METHOD1(SetReceiver, void(PacketReceiver* receiver));
>-  MOCK_METHOD2(DeliverPacket,
>-               void(const NetworkPacket* packet,
>-                    const PacketTime& packet_time));
>+  MOCK_METHOD3(DeliverPacket,
>+               DeliveryStatus(MediaType, rtc::CopyOnWriteBuffer, int64_t));
>+  virtual ~MockReceiver() = default;
> };
> 
>-class ReorderTestDemuxer : public TestDemuxer {
>+class ReorderTestReceiver : public MockReceiver {
>  public:
>-  void DeliverPacket(const NetworkPacket* packet,
>-                     const PacketTime& packet_time) override {
>-    RTC_DCHECK_GE(packet->data_length(), sizeof(int));
>+  DeliveryStatus DeliverPacket(MediaType media_type,
>+                               rtc::CopyOnWriteBuffer packet,
>+                               int64_t /* packet_time_us */) override {
>+    RTC_DCHECK_GE(packet.size(), sizeof(int));
>     int seq_num;
>-    memcpy(&seq_num, packet->data(), sizeof(int));
>+    memcpy(&seq_num, packet.data<uint8_t>(), sizeof(int));
>     delivered_sequence_numbers_.push_back(seq_num);
>+    return DeliveryStatus::DELIVERY_OK;
>   }
>   std::vector<int> delivered_sequence_numbers_;
> };
> 
>-class MockReceiver : public PacketReceiver {
>- public:
>-  MOCK_METHOD4(
>-      DeliverPacket,
>-      DeliveryStatus(MediaType, const uint8_t*, size_t, const PacketTime&));
>-};
>-
> class FakeNetworkPipeTest : public ::testing::Test {
>  public:
>   FakeNetworkPipeTest() : fake_clock_(12345) {}
>@@ -67,7 +59,8 @@ class FakeNetworkPipeTest : public ::testing::Test {
>       // Set a sequence number for the packets by
>       // using the first bytes in the packet.
>       memcpy(packet.get(), &i, sizeof(int));
>-      pipe->SendPacket(packet.get(), packet_size);
>+      rtc::CopyOnWriteBuffer buffer(packet.get(), packet_size);
>+      pipe->DeliverPacket(MediaType::ANY, buffer, /* packet_time_us */ -1);
>     }
>   }
> 
>@@ -83,9 +76,9 @@ TEST_F(FakeNetworkPipeTest, CapacityTest) {
>   FakeNetworkPipe::Config config;
>   config.queue_length_packets = 20;
>   config.link_capacity_kbps = 80;
>-  TestDemuxer* demuxer = new TestDemuxer();
>-  std::unique_ptr<FakeNetworkPipe> pipe(new FakeNetworkPipe(
>-      &fake_clock_, config, std::unique_ptr<Demuxer>(demuxer)));
>+  MockReceiver receiver;
>+  std::unique_ptr<FakeNetworkPipe> pipe(
>+      new FakeNetworkPipe(&fake_clock_, config, &receiver));
> 
>   // Add 10 packets of 1000 bytes, = 80 kb, and verify it takes one second to
>   // get through the pipe.
>@@ -94,26 +87,26 @@ TEST_F(FakeNetworkPipeTest, CapacityTest) {
>   SendPackets(pipe.get(), kNumPackets, kPacketSize);
> 
>   // Time to get one packet through the link.
>-  const int kPacketTimeMs = PacketTimeMs(config.link_capacity_kbps,
>-                                         kPacketSize);
>+  const int kPacketTimeMs =
>+      PacketTimeMs(config.link_capacity_kbps, kPacketSize);
> 
>   // Time haven't increased yet, so we souldn't get any packets.
>-  EXPECT_CALL(*demuxer, DeliverPacket(_, _)).Times(0);
>+  EXPECT_CALL(receiver, DeliverPacket(_, _, _)).Times(0);
>   pipe->Process();
> 
>   // Advance enough time to release one packet.
>   fake_clock_.AdvanceTimeMilliseconds(kPacketTimeMs);
>-  EXPECT_CALL(*demuxer, DeliverPacket(_, _)).Times(1);
>+  EXPECT_CALL(receiver, DeliverPacket(_, _, _)).Times(1);
>   pipe->Process();
> 
>   // Release all but one packet
>   fake_clock_.AdvanceTimeMilliseconds(9 * kPacketTimeMs - 1);
>-  EXPECT_CALL(*demuxer, DeliverPacket(_, _)).Times(8);
>+  EXPECT_CALL(receiver, DeliverPacket(_, _, _)).Times(8);
>   pipe->Process();
> 
>   // And the last one.
>   fake_clock_.AdvanceTimeMilliseconds(1);
>-  EXPECT_CALL(*demuxer, DeliverPacket(_, _)).Times(1);
>+  EXPECT_CALL(receiver, DeliverPacket(_, _, _)).Times(1);
>   pipe->Process();
> }
> 
>@@ -123,31 +116,31 @@ TEST_F(FakeNetworkPipeTest, ExtraDelayTest) {
>   config.queue_length_packets = 20;
>   config.queue_delay_ms = 100;
>   config.link_capacity_kbps = 80;
>-  TestDemuxer* demuxer = new TestDemuxer();
>-  std::unique_ptr<FakeNetworkPipe> pipe(new FakeNetworkPipe(
>-      &fake_clock_, config, std::unique_ptr<Demuxer>(demuxer)));
>+  MockReceiver receiver;
>+  std::unique_ptr<FakeNetworkPipe> pipe(
>+      new FakeNetworkPipe(&fake_clock_, config, &receiver));
> 
>   const int kNumPackets = 2;
>   const int kPacketSize = 1000;
>   SendPackets(pipe.get(), kNumPackets, kPacketSize);
> 
>   // Time to get one packet through the link.
>-  const int kPacketTimeMs = PacketTimeMs(config.link_capacity_kbps,
>-                                         kPacketSize);
>+  const int kPacketTimeMs =
>+      PacketTimeMs(config.link_capacity_kbps, kPacketSize);
> 
>   // Increase more than kPacketTimeMs, but not more than the extra delay.
>   fake_clock_.AdvanceTimeMilliseconds(kPacketTimeMs);
>-  EXPECT_CALL(*demuxer, DeliverPacket(_, _)).Times(0);
>+  EXPECT_CALL(receiver, DeliverPacket(_, _, _)).Times(0);
>   pipe->Process();
> 
>   // Advance the network delay to get the first packet.
>   fake_clock_.AdvanceTimeMilliseconds(config.queue_delay_ms);
>-  EXPECT_CALL(*demuxer, DeliverPacket(_, _)).Times(1);
>+  EXPECT_CALL(receiver, DeliverPacket(_, _, _)).Times(1);
>   pipe->Process();
> 
>   // Advance one more kPacketTimeMs to get the last packet.
>   fake_clock_.AdvanceTimeMilliseconds(kPacketTimeMs);
>-  EXPECT_CALL(*demuxer, DeliverPacket(_, _)).Times(1);
>+  EXPECT_CALL(receiver, DeliverPacket(_, _, _)).Times(1);
>   pipe->Process();
> }
> 
>@@ -157,13 +150,13 @@ TEST_F(FakeNetworkPipeTest, QueueLengthTest) {
>   FakeNetworkPipe::Config config;
>   config.queue_length_packets = 2;
>   config.link_capacity_kbps = 80;
>-  TestDemuxer* demuxer = new TestDemuxer();
>-  std::unique_ptr<FakeNetworkPipe> pipe(new FakeNetworkPipe(
>-      &fake_clock_, config, std::unique_ptr<Demuxer>(demuxer)));
>+  MockReceiver receiver;
>+  std::unique_ptr<FakeNetworkPipe> pipe(
>+      new FakeNetworkPipe(&fake_clock_, config, &receiver));
> 
>   const int kPacketSize = 1000;
>-  const int kPacketTimeMs = PacketTimeMs(config.link_capacity_kbps,
>-                                         kPacketSize);
>+  const int kPacketTimeMs =
>+      PacketTimeMs(config.link_capacity_kbps, kPacketSize);
> 
>   // Send three packets and verify only 2 are delivered.
>   SendPackets(pipe.get(), 3, kPacketSize);
>@@ -171,7 +164,7 @@ TEST_F(FakeNetworkPipeTest, QueueLengthTest) {
>   // Increase time enough to deliver all three packets, verify only two are
>   // delivered.
>   fake_clock_.AdvanceTimeMilliseconds(3 * kPacketTimeMs);
>-  EXPECT_CALL(*demuxer, DeliverPacket(_, _)).Times(2);
>+  EXPECT_CALL(receiver, DeliverPacket(_, _, _)).Times(2);
>   pipe->Process();
> }
> 
>@@ -181,28 +174,28 @@ TEST_F(FakeNetworkPipeTest, StatisticsTest) {
>   config.queue_length_packets = 2;
>   config.queue_delay_ms = 20;
>   config.link_capacity_kbps = 80;
>-  TestDemuxer* demuxer = new TestDemuxer();
>-  std::unique_ptr<FakeNetworkPipe> pipe(new FakeNetworkPipe(
>-      &fake_clock_, config, std::unique_ptr<Demuxer>(demuxer)));
>+  MockReceiver receiver;
>+  std::unique_ptr<FakeNetworkPipe> pipe(
>+      new FakeNetworkPipe(&fake_clock_, config, &receiver));
> 
>   const int kPacketSize = 1000;
>-  const int kPacketTimeMs = PacketTimeMs(config.link_capacity_kbps,
>-                                         kPacketSize);
>+  const int kPacketTimeMs =
>+      PacketTimeMs(config.link_capacity_kbps, kPacketSize);
> 
>   // Send three packets and verify only 2 are delivered.
>   SendPackets(pipe.get(), 3, kPacketSize);
>   fake_clock_.AdvanceTimeMilliseconds(3 * kPacketTimeMs +
>                                       config.queue_delay_ms);
> 
>-  EXPECT_CALL(*demuxer, DeliverPacket(_, _)).Times(2);
>+  EXPECT_CALL(receiver, DeliverPacket(_, _, _)).Times(2);
>   pipe->Process();
> 
>   // Packet 1: kPacketTimeMs + config.queue_delay_ms,
>   // packet 2: 2 * kPacketTimeMs + config.queue_delay_ms => 170 ms average.
>   EXPECT_EQ(pipe->AverageDelay(), 170);
>-  EXPECT_EQ(pipe->sent_packets(), 2u);
>-  EXPECT_EQ(pipe->dropped_packets(), 1u);
>-  EXPECT_EQ(pipe->PercentageLoss(), 1/3.f);
>+  EXPECT_EQ(pipe->SentPackets(), 2u);
>+  EXPECT_EQ(pipe->DroppedPackets(), 1u);
>+  EXPECT_EQ(pipe->PercentageLoss(), 1 / 3.f);
> }
> 
> // Change the link capacity half-way through the test and verify that the
>@@ -211,9 +204,11 @@ TEST_F(FakeNetworkPipeTest, ChangingCapacityWithEmptyPipeTest) {
>   FakeNetworkPipe::Config config;
>   config.queue_length_packets = 20;
>   config.link_capacity_kbps = 80;
>-  TestDemuxer* demuxer = new TestDemuxer();
>-  std::unique_ptr<FakeNetworkPipe> pipe(new FakeNetworkPipe(
>-      &fake_clock_, config, std::unique_ptr<Demuxer>(demuxer)));
>+  MockReceiver receiver;
>+  std::unique_ptr<SimulatedNetwork> network(new SimulatedNetwork(config));
>+  SimulatedNetwork* simulated_network = network.get();
>+  std::unique_ptr<FakeNetworkPipe> pipe(
>+      new FakeNetworkPipe(&fake_clock_, std::move(network), &receiver));
> 
>   // Add 10 packets of 1000 bytes, = 80 kb, and verify it takes one second to
>   // get through the pipe.
>@@ -225,19 +220,19 @@ TEST_F(FakeNetworkPipeTest, ChangingCapacityWithEmptyPipeTest) {
>   int packet_time_ms = PacketTimeMs(config.link_capacity_kbps, kPacketSize);
> 
>   // Time hasn't increased yet, so we souldn't get any packets.
>-  EXPECT_CALL(*demuxer, DeliverPacket(_, _)).Times(0);
>+  EXPECT_CALL(receiver, DeliverPacket(_, _, _)).Times(0);
>   pipe->Process();
> 
>   // Advance time in steps to release one packet at a time.
>   for (int i = 0; i < kNumPackets; ++i) {
>     fake_clock_.AdvanceTimeMilliseconds(packet_time_ms);
>-    EXPECT_CALL(*demuxer, DeliverPacket(_, _)).Times(1);
>+    EXPECT_CALL(receiver, DeliverPacket(_, _, _)).Times(1);
>     pipe->Process();
>   }
> 
>   // Change the capacity.
>   config.link_capacity_kbps /= 2;  // Reduce to 50%.
>-  pipe->SetConfig(config);
>+  simulated_network->SetConfig(config);
> 
>   // Add another 10 packets of 1000 bytes, = 80 kb, and verify it takes two
>   // seconds to get them through the pipe.
>@@ -247,20 +242,20 @@ TEST_F(FakeNetworkPipeTest, ChangingCapacityWithEmptyPipeTest) {
>   packet_time_ms = PacketTimeMs(config.link_capacity_kbps, kPacketSize);
> 
>   // Time hasn't increased yet, so we souldn't get any packets.
>-  EXPECT_CALL(*demuxer, DeliverPacket(_, _)).Times(0);
>+  EXPECT_CALL(receiver, DeliverPacket(_, _, _)).Times(0);
>   pipe->Process();
> 
>   // Advance time in steps to release one packet at a time.
>   for (int i = 0; i < kNumPackets; ++i) {
>     fake_clock_.AdvanceTimeMilliseconds(packet_time_ms);
>-    EXPECT_CALL(*demuxer, DeliverPacket(_, _)).Times(1);
>+    EXPECT_CALL(receiver, DeliverPacket(_, _, _)).Times(1);
>     pipe->Process();
>   }
> 
>   // Check that all the packets were sent.
>-  EXPECT_EQ(static_cast<size_t>(2 * kNumPackets), pipe->sent_packets());
>+  EXPECT_EQ(static_cast<size_t>(2 * kNumPackets), pipe->SentPackets());
>   fake_clock_.AdvanceTimeMilliseconds(pipe->TimeUntilNextProcess());
>-  EXPECT_CALL(*demuxer, DeliverPacket(_, _)).Times(0);
>+  EXPECT_CALL(receiver, DeliverPacket(_, _, _)).Times(0);
>   pipe->Process();
> }
> 
>@@ -270,9 +265,11 @@ TEST_F(FakeNetworkPipeTest, ChangingCapacityWithPacketsInPipeTest) {
>   FakeNetworkPipe::Config config;
>   config.queue_length_packets = 20;
>   config.link_capacity_kbps = 80;
>-  TestDemuxer* demuxer = new TestDemuxer();
>-  std::unique_ptr<FakeNetworkPipe> pipe(new FakeNetworkPipe(
>-      &fake_clock_, config, std::unique_ptr<Demuxer>(demuxer)));
>+  MockReceiver receiver;
>+  std::unique_ptr<SimulatedNetwork> network(new SimulatedNetwork(config));
>+  SimulatedNetwork* simulated_network = network.get();
>+  std::unique_ptr<FakeNetworkPipe> pipe(
>+      new FakeNetworkPipe(&fake_clock_, std::move(network), &receiver));
> 
>   // Add 10 packets of 1000 bytes, = 80 kb.
>   const int kNumPackets = 10;
>@@ -284,7 +281,7 @@ TEST_F(FakeNetworkPipeTest, ChangingCapacityWithPacketsInPipeTest) {
> 
>   // Change the capacity.
>   config.link_capacity_kbps *= 2;  // Double the capacity.
>-  pipe->SetConfig(config);
>+  simulated_network->SetConfig(config);
> 
>   // Add another 10 packets of 1000 bytes, = 80 kb, and verify it takes two
>   // seconds to get them through the pipe.
>@@ -294,27 +291,27 @@ TEST_F(FakeNetworkPipeTest, ChangingCapacityWithPacketsInPipeTest) {
>   int packet_time_2_ms = PacketTimeMs(config.link_capacity_kbps, kPacketSize);
> 
>   // Time hasn't increased yet, so we souldn't get any packets.
>-  EXPECT_CALL(*demuxer, DeliverPacket(_, _)).Times(0);
>+  EXPECT_CALL(receiver, DeliverPacket(_, _, _)).Times(0);
>   pipe->Process();
> 
>   // Advance time in steps to release one packet at a time.
>   for (int i = 0; i < kNumPackets; ++i) {
>     fake_clock_.AdvanceTimeMilliseconds(packet_time_1_ms);
>-    EXPECT_CALL(*demuxer, DeliverPacket(_, _)).Times(1);
>+    EXPECT_CALL(receiver, DeliverPacket(_, _, _)).Times(1);
>     pipe->Process();
>   }
> 
>   // Advance time in steps to release one packet at a time.
>   for (int i = 0; i < kNumPackets; ++i) {
>     fake_clock_.AdvanceTimeMilliseconds(packet_time_2_ms);
>-    EXPECT_CALL(*demuxer, DeliverPacket(_, _)).Times(1);
>+    EXPECT_CALL(receiver, DeliverPacket(_, _, _)).Times(1);
>     pipe->Process();
>   }
> 
>   // Check that all the packets were sent.
>-  EXPECT_EQ(static_cast<size_t>(2 * kNumPackets), pipe->sent_packets());
>+  EXPECT_EQ(static_cast<size_t>(2 * kNumPackets), pipe->SentPackets());
>   fake_clock_.AdvanceTimeMilliseconds(pipe->TimeUntilNextProcess());
>-  EXPECT_CALL(*demuxer, DeliverPacket(_, _)).Times(0);
>+  EXPECT_CALL(receiver, DeliverPacket(_, _, _)).Times(0);
>   pipe->Process();
> }
> 
>@@ -325,9 +322,11 @@ TEST_F(FakeNetworkPipeTest, DisallowReorderingThenAllowReordering) {
>   config.link_capacity_kbps = 800;
>   config.queue_delay_ms = 100;
>   config.delay_standard_deviation_ms = 10;
>-  ReorderTestDemuxer* demuxer = new ReorderTestDemuxer();
>-  std::unique_ptr<FakeNetworkPipe> pipe(new FakeNetworkPipe(
>-      &fake_clock_, config, std::unique_ptr<Demuxer>(demuxer)));
>+  ReorderTestReceiver receiver;
>+  std::unique_ptr<SimulatedNetwork> network(new SimulatedNetwork(config));
>+  SimulatedNetwork* simulated_network = network.get();
>+  std::unique_ptr<FakeNetworkPipe> pipe(
>+      new FakeNetworkPipe(&fake_clock_, std::move(network), &receiver));
> 
>   const uint32_t kNumPackets = 100;
>   const int kPacketSize = 10;
>@@ -336,26 +335,26 @@ TEST_F(FakeNetworkPipeTest, DisallowReorderingThenAllowReordering) {
>   pipe->Process();
> 
>   // Confirm that all packets have been delivered in order.
>-  EXPECT_EQ(kNumPackets, demuxer->delivered_sequence_numbers_.size());
>+  EXPECT_EQ(kNumPackets, receiver.delivered_sequence_numbers_.size());
>   int last_seq_num = -1;
>-  for (int seq_num : demuxer->delivered_sequence_numbers_) {
>+  for (int seq_num : receiver.delivered_sequence_numbers_) {
>     EXPECT_GT(seq_num, last_seq_num);
>     last_seq_num = seq_num;
>   }
> 
>   config.allow_reordering = true;
>-  pipe->SetConfig(config);
>+  simulated_network->SetConfig(config);
>   SendPackets(pipe.get(), kNumPackets, kPacketSize);
>   fake_clock_.AdvanceTimeMilliseconds(1000);
>-  demuxer->delivered_sequence_numbers_.clear();
>+  receiver.delivered_sequence_numbers_.clear();
>   pipe->Process();
> 
>   // Confirm that all packets have been delivered
>   // and that reordering has occured.
>-  EXPECT_EQ(kNumPackets, demuxer->delivered_sequence_numbers_.size());
>+  EXPECT_EQ(kNumPackets, receiver.delivered_sequence_numbers_.size());
>   bool reordering_has_occured = false;
>   last_seq_num = -1;
>-  for (int seq_num : demuxer->delivered_sequence_numbers_) {
>+  for (int seq_num : receiver.delivered_sequence_numbers_) {
>     if (last_seq_num > seq_num) {
>       reordering_has_occured = true;
>       break;
>@@ -375,26 +374,26 @@ TEST_F(FakeNetworkPipeTest, BurstLoss) {
>   config.queue_length_packets = kNumPackets;
>   config.loss_percent = kLossPercent;
>   config.avg_burst_loss_length = kAvgBurstLength;
>-  ReorderTestDemuxer* demuxer = new ReorderTestDemuxer();
>-  std::unique_ptr<FakeNetworkPipe> pipe(new FakeNetworkPipe(
>-      &fake_clock_, config, std::unique_ptr<Demuxer>(demuxer)));
>+  ReorderTestReceiver receiver;
>+  std::unique_ptr<FakeNetworkPipe> pipe(
>+      new FakeNetworkPipe(&fake_clock_, config, &receiver));
> 
>   SendPackets(pipe.get(), kNumPackets, kPacketSize);
>   fake_clock_.AdvanceTimeMilliseconds(1000);
>   pipe->Process();
> 
>   // Check that the average loss is |kLossPercent| percent.
>-  int lost_packets = kNumPackets - demuxer->delivered_sequence_numbers_.size();
>+  int lost_packets = kNumPackets - receiver.delivered_sequence_numbers_.size();
>   double loss_fraction = lost_packets / static_cast<double>(kNumPackets);
> 
>   EXPECT_NEAR(kLossPercent / 100.0, loss_fraction, 0.05);
> 
>   // Find the number of bursts that has occurred.
>-  size_t received_packets = demuxer->delivered_sequence_numbers_.size();
>+  size_t received_packets = receiver.delivered_sequence_numbers_.size();
>   int num_bursts = 0;
>   for (size_t i = 0; i < received_packets - 1; ++i) {
>-    int diff = demuxer->delivered_sequence_numbers_[i + 1] -
>-               demuxer->delivered_sequence_numbers_[i];
>+    int diff = receiver.delivered_sequence_numbers_[i + 1] -
>+               receiver.delivered_sequence_numbers_[i];
>     if (diff > 1)
>       ++num_bursts;
>   }
>@@ -406,40 +405,27 @@ TEST_F(FakeNetworkPipeTest, BurstLoss) {
> 
> TEST_F(FakeNetworkPipeTest, SetReceiver) {
>   FakeNetworkPipe::Config config;
>-  TestDemuxer* demuxer = new TestDemuxer();
>-  std::unique_ptr<FakeNetworkPipe> pipe(new FakeNetworkPipe(
>-      &fake_clock_, config, std::unique_ptr<Demuxer>(demuxer)));
>-  MockReceiver packet_receiver;
>-  EXPECT_CALL(*demuxer, SetReceiver(&packet_receiver)).Times(1);
>-  pipe->SetReceiver(&packet_receiver);
>-}
>+  config.link_capacity_kbps = 800;
>+  MockReceiver receiver;
>+  std::unique_ptr<FakeNetworkPipe> pipe(
>+      new FakeNetworkPipe(&fake_clock_, config, &receiver));
>+
>+  const int kPacketSize = 1000;
>+  const int kPacketTimeMs =
>+      PacketTimeMs(config.link_capacity_kbps, kPacketSize);
>+  SendPackets(pipe.get(), 1, kPacketSize);
>+  fake_clock_.AdvanceTimeMilliseconds(kPacketTimeMs);
>+  EXPECT_CALL(receiver, DeliverPacket(_, _, _)).Times(1);
>+  pipe->Process();
>+
>+  MockReceiver new_receiver;
>+  pipe->SetReceiver(&new_receiver);
> 
>-TEST(DemuxerImplTest, Demuxing) {
>-  constexpr uint8_t kVideoPayloadType = 100;
>-  constexpr uint8_t kAudioPayloadType = 101;
>-  constexpr int64_t kTimeNow = 12345;
>-  constexpr int64_t kArrivalTime = kTimeNow - 1;
>-  constexpr size_t kPacketSize = 10;
>-  DemuxerImpl demuxer({{kVideoPayloadType, MediaType::VIDEO},
>-                       {kAudioPayloadType, MediaType::AUDIO}});
>-
>-  MockReceiver mock_receiver;
>-  demuxer.SetReceiver(&mock_receiver);
>-
>-  std::vector<uint8_t> data(kPacketSize);
>-  data[1] = kVideoPayloadType;
>-  std::unique_ptr<NetworkPacket> packet(
>-      new NetworkPacket(&data[0], kPacketSize, kTimeNow, kArrivalTime));
>-  EXPECT_CALL(mock_receiver, DeliverPacket(MediaType::VIDEO, _, _, _))
>-      .WillOnce(Return(PacketReceiver::DELIVERY_OK));
>-  demuxer.DeliverPacket(packet.get(), PacketTime());
>-
>-  data[1] = kAudioPayloadType;
>-  packet.reset(
>-      new NetworkPacket(&data[0], kPacketSize, kTimeNow, kArrivalTime));
>-  EXPECT_CALL(mock_receiver, DeliverPacket(MediaType::AUDIO, _, _, _))
>-      .WillOnce(Return(PacketReceiver::DELIVERY_OK));
>-  demuxer.DeliverPacket(packet.get(), PacketTime());
>+  SendPackets(pipe.get(), 1, kPacketSize);
>+  fake_clock_.AdvanceTimeMilliseconds(kPacketTimeMs);
>+  EXPECT_CALL(receiver, DeliverPacket(_, _, _)).Times(0);
>+  EXPECT_CALL(new_receiver, DeliverPacket(_, _, _)).Times(1);
>+  pipe->Process();
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/test/mock_audio_send_stream.h b/Source/ThirdParty/libwebrtc/Source/webrtc/call/test/mock_audio_send_stream.h
>index 4eb11663166..489e826d0eb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/test/mock_audio_send_stream.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/test/mock_audio_send_stream.h
>@@ -26,14 +26,14 @@ class MockAudioSendStream : public AudioSendStream {
>   MOCK_METHOD0(Start, void());
>   MOCK_METHOD0(Stop, void());
>   // GMock doesn't like move-only types, such as std::unique_ptr.
>-  virtual void SendAudioData(
>-      std::unique_ptr<webrtc::AudioFrame> audio_frame) {
>+  virtual void SendAudioData(std::unique_ptr<webrtc::AudioFrame> audio_frame) {
>     SendAudioDataForMock(audio_frame.get());
>   }
>-  MOCK_METHOD1(SendAudioDataForMock,
>-               void(webrtc::AudioFrame* audio_frame));
>+  MOCK_METHOD1(SendAudioDataForMock, void(webrtc::AudioFrame* audio_frame));
>   MOCK_METHOD4(SendTelephoneEvent,
>-               bool(int payload_type, int payload_frequency, int event,
>+               bool(int payload_type,
>+                    int payload_frequency,
>+                    int event,
>                     int duration_ms));
>   MOCK_METHOD1(SetMuted, void(bool muted));
>   MOCK_CONST_METHOD0(GetStats, Stats());
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/test/mock_bitrate_allocator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/call/test/mock_bitrate_allocator.h
>new file mode 100644
>index 00000000000..714c02541f1
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/test/mock_bitrate_allocator.h
>@@ -0,0 +1,27 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#ifndef CALL_TEST_MOCK_BITRATE_ALLOCATOR_H_
>+#define CALL_TEST_MOCK_BITRATE_ALLOCATOR_H_
>+
>+#include <string>
>+
>+#include "call/bitrate_allocator.h"
>+#include "test/gmock.h"
>+
>+namespace webrtc {
>+class MockBitrateAllocator : public BitrateAllocatorInterface {
>+ public:
>+  MOCK_METHOD2(AddObserver,
>+               void(BitrateAllocatorObserver*, MediaStreamAllocationConfig));
>+  MOCK_METHOD1(RemoveObserver, void(BitrateAllocatorObserver*));
>+  MOCK_METHOD1(GetStartBitrate, int(BitrateAllocatorObserver*));
>+};
>+}  // namespace webrtc
>+#endif  // CALL_TEST_MOCK_BITRATE_ALLOCATOR_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/test/mock_rtp_transport_controller_send.h b/Source/ThirdParty/libwebrtc/Source/webrtc/call/test/mock_rtp_transport_controller_send.h
>new file mode 100644
>index 00000000000..828b030378b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/test/mock_rtp_transport_controller_send.h
>@@ -0,0 +1,70 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef CALL_TEST_MOCK_RTP_TRANSPORT_CONTROLLER_SEND_H_
>+#define CALL_TEST_MOCK_RTP_TRANSPORT_CONTROLLER_SEND_H_
>+
>+#include <map>
>+#include <string>
>+#include <vector>
>+
>+#include "api/bitrate_constraints.h"
>+#include "call/rtp_transport_controller_send_interface.h"
>+#include "modules/congestion_controller/include/network_changed_observer.h"
>+#include "modules/pacing/packet_router.h"
>+#include "rtc_base/networkroute.h"
>+#include "rtc_base/rate_limiter.h"
>+#include "rtc_base/socket.h"
>+#include "test/gmock.h"
>+
>+namespace webrtc {
>+
>+class MockRtpTransportControllerSend
>+    : public RtpTransportControllerSendInterface {
>+ public:
>+  MOCK_METHOD8(
>+      CreateRtpVideoSender,
>+      RtpVideoSenderInterface*(const std::vector<uint32_t>&,
>+                               std::map<uint32_t, RtpState>,
>+                               const std::map<uint32_t, RtpPayloadState>&,
>+                               const RtpConfig&,
>+                               const RtcpConfig&,
>+                               Transport*,
>+                               const RtpSenderObservers&,
>+                               RtcEventLog*));
>+  MOCK_METHOD1(DestroyRtpVideoSender, void(RtpVideoSenderInterface*));
>+  MOCK_METHOD0(GetWorkerQueue, rtc::TaskQueue*());
>+  MOCK_METHOD0(packet_router, PacketRouter*());
>+  MOCK_METHOD0(transport_feedback_observer, TransportFeedbackObserver*());
>+  MOCK_METHOD0(packet_sender, RtpPacketSender*());
>+  MOCK_CONST_METHOD0(keepalive_config, RtpKeepAliveConfig&());
>+  MOCK_METHOD3(SetAllocatedSendBitrateLimits, void(int, int, int));
>+  MOCK_METHOD1(SetPacingFactor, void(float));
>+  MOCK_METHOD1(SetQueueTimeLimit, void(int));
>+  MOCK_METHOD0(GetCallStatsObserver, CallStatsObserver*());
>+  MOCK_METHOD1(RegisterPacketFeedbackObserver, void(PacketFeedbackObserver*));
>+  MOCK_METHOD1(DeRegisterPacketFeedbackObserver, void(PacketFeedbackObserver*));
>+  MOCK_METHOD1(RegisterTargetTransferRateObserver,
>+               void(TargetTransferRateObserver*));
>+  MOCK_METHOD2(OnNetworkRouteChanged,
>+               void(const std::string&, const rtc::NetworkRoute&));
>+  MOCK_METHOD1(OnNetworkAvailability, void(bool));
>+  MOCK_METHOD0(GetBandwidthObserver, RtcpBandwidthObserver*());
>+  MOCK_CONST_METHOD0(GetPacerQueuingDelayMs, int64_t());
>+  MOCK_CONST_METHOD0(GetFirstPacketTimeMs, int64_t());
>+  MOCK_METHOD1(SetPerPacketFeedbackAvailable, void(bool));
>+  MOCK_METHOD1(EnablePeriodicAlrProbing, void(bool));
>+  MOCK_METHOD1(OnSentPacket, void(const rtc::SentPacket&));
>+  MOCK_METHOD1(SetSdpBitrateParameters, void(const BitrateConstraints&));
>+  MOCK_METHOD1(SetClientBitratePreferences, void(const BitrateSettings&));
>+  MOCK_METHOD1(SetAllocatedBitrateWithoutFeedback, void(uint32_t));
>+};
>+}  // namespace webrtc
>+#endif  // CALL_TEST_MOCK_RTP_TRANSPORT_CONTROLLER_SEND_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/video_receive_stream.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/video_receive_stream.cc
>index f338805746d..7662afff72d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/video_receive_stream.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/video_receive_stream.cc
>@@ -9,6 +9,7 @@
>  */
> 
> #include "call/video_receive_stream.h"
>+#include "rtc_base/strings/string_builder.h"
> 
> namespace webrtc {
> 
>@@ -17,7 +18,8 @@ VideoReceiveStream::Decoder::Decoder(const Decoder&) = default;
> VideoReceiveStream::Decoder::~Decoder() = default;
> 
> std::string VideoReceiveStream::Decoder::ToString() const {
>-  std::stringstream ss;
>+  char buf[1024];
>+  rtc::SimpleStringBuilder ss(buf);
>   ss << "{decoder: " << (decoder ? "(VideoDecoder)" : "nullptr");
>   ss << ", payload_type: " << payload_type;
>   ss << ", payload_name: " << payload_name;
>@@ -34,7 +36,8 @@ VideoReceiveStream::Stats::Stats() = default;
> VideoReceiveStream::Stats::~Stats() = default;
> 
> std::string VideoReceiveStream::Stats::ToString(int64_t time_ms) const {
>-  std::stringstream ss;
>+  char buf[1024];
>+  rtc::SimpleStringBuilder ss(buf);
>   ss << "VideoReceiveStream stats: " << time_ms << ", {ssrc: " << ssrc << ", ";
>   ss << "total_bps: " << total_bitrate_bps << ", ";
>   ss << "width: " << width << ", ";
>@@ -71,7 +74,8 @@ VideoReceiveStream::Config& VideoReceiveStream::Config::operator=(Config&&) =
> VideoReceiveStream::Config::Config::~Config() = default;
> 
> std::string VideoReceiveStream::Config::ToString() const {
>-  std::stringstream ss;
>+  char buf[4 * 1024];
>+  rtc::SimpleStringBuilder ss(buf);
>   ss << "{decoders: [";
>   for (size_t i = 0; i < decoders.size(); ++i) {
>     ss << decoders[i].ToString();
>@@ -84,8 +88,6 @@ std::string VideoReceiveStream::Config::ToString() const {
>   ss << ", render_delay_ms: " << render_delay_ms;
>   if (!sync_group.empty())
>     ss << ", sync_group: " << sync_group;
>-  ss << ", pre_decode_callback: "
>-     << (pre_decode_callback ? "(EncodedFrameObserver)" : "nullptr");
>   ss << ", target_delay_ms: " << target_delay_ms;
>   ss << '}';
> 
>@@ -97,7 +99,8 @@ VideoReceiveStream::Config::Rtp::Rtp(const Rtp&) = default;
> VideoReceiveStream::Config::Rtp::~Rtp() = default;
> 
> std::string VideoReceiveStream::Config::Rtp::ToString() const {
>-  std::stringstream ss;
>+  char buf[2 * 1024];
>+  rtc::SimpleStringBuilder ss(buf);
>   ss << "{remote_ssrc: " << remote_ssrc;
>   ss << ", local_ssrc: " << local_ssrc;
>   ss << ", rtcp_mode: "
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/video_receive_stream.h b/Source/ThirdParty/libwebrtc/Source/webrtc/call/video_receive_stream.h
>index 1adc696461a..5009ddb43f4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/video_receive_stream.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/video_receive_stream.h
>@@ -16,12 +16,12 @@
> #include <string>
> #include <vector>
> 
>-#include "api/rtp_headers.h"
> #include "api/call/transport.h"
>+#include "api/rtp_headers.h"
> #include "api/rtpparameters.h"
> #include "api/video/video_content_type.h"
>+#include "api/video/video_sink_interface.h"
> #include "api/video/video_timing.h"
>-#include "api/videosinkinterface.h"
> #include "call/rtp_config.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "common_video/include/frame_callback.h"
>@@ -82,7 +82,7 @@ class VideoReceiveStream {
>     int render_delay_ms = 10;
>     int64_t interframe_delay_max_ms = -1;
>     uint32_t frames_decoded = 0;
>-    rtc::Optional<uint64_t> qp_sum;
>+    absl::optional<uint64_t> qp_sum;
> 
>     int current_payload_type = -1;
> 
>@@ -104,7 +104,7 @@ class VideoReceiveStream {
> 
>     // Timing frame info: all important timestamps for a full lifetime of a
>     // single 'timing frame'.
>-    rtc::Optional<webrtc::TimingFrameInfo> timing_frame_info;
>+    absl::optional<webrtc::TimingFrameInfo> timing_frame_info;
>   };
> 
>   struct Config {
>@@ -213,11 +213,6 @@ class VideoReceiveStream {
>     // to one of the audio streams.
>     std::string sync_group;
> 
>-    // Called for each incoming video frame, i.e. in encoded state. E.g. used
>-    // when
>-    // saving the stream to a file. 'nullptr' disables the callback.
>-    EncodedFrameObserver* pre_decode_callback = nullptr;
>-
>     // Target delay in milliseconds. A positive value indicates this stream is
>     // used for streaming instead of a real-time call.
>     int target_delay_ms = 0;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/video_send_stream.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/call/video_send_stream.cc
>index 4f8c059ee42..6b93fe28238 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/video_send_stream.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/video_send_stream.cc
>@@ -9,6 +9,7 @@
>  */
> 
> #include "call/video_send_stream.h"
>+#include "rtc_base/strings/string_builder.h"
> 
> namespace webrtc {
> 
>@@ -16,7 +17,8 @@ VideoSendStream::StreamStats::StreamStats() = default;
> VideoSendStream::StreamStats::~StreamStats() = default;
> 
> std::string VideoSendStream::StreamStats::ToString() const {
>-  std::stringstream ss;
>+  char buf[1024];
>+  rtc::SimpleStringBuilder ss(buf);
>   ss << "width: " << width << ", ";
>   ss << "height: " << height << ", ";
>   ss << "key: " << frame_counts.key_frames << ", ";
>@@ -37,7 +39,8 @@ VideoSendStream::Stats::Stats() = default;
> VideoSendStream::Stats::~Stats() = default;
> 
> std::string VideoSendStream::Stats::ToString(int64_t time_ms) const {
>-  std::stringstream ss;
>+  char buf[1024];
>+  rtc::SimpleStringBuilder ss(buf);
>   ss << "VideoSendStream stats: " << time_ms << ", {";
>   ss << "input_fps: " << input_frame_rate << ", ";
>   ss << "encode_fps: " << encode_frame_rate << ", ";
>@@ -45,7 +48,6 @@ std::string VideoSendStream::Stats::ToString(int64_t time_ms) const {
>   ss << "encode_usage_perc: " << encode_usage_percent << ", ";
>   ss << "target_bps: " << target_media_bitrate_bps << ", ";
>   ss << "media_bps: " << media_bitrate_bps << ", ";
>-  ss << "preferred_media_bitrate_bps: " << preferred_media_bitrate_bps << ", ";
>   ss << "suspended: " << (suspended ? "true" : "false") << ", ";
>   ss << "bw_adapted: " << (bw_limited_resolution ? "true" : "false");
>   ss << '}';
>@@ -68,9 +70,12 @@ VideoSendStream::Config& VideoSendStream::Config::operator=(Config&&) = default;
> VideoSendStream::Config::Config::~Config() = default;
> 
> std::string VideoSendStream::Config::ToString() const {
>-  std::stringstream ss;
>-  ss << "{encoder_settings: " << encoder_settings.ToString();
>+  char buf[2 * 1024];
>+  rtc::SimpleStringBuilder ss(buf);
>+  ss << "{encoder_settings: { experiment_cpu_load_estimator: "
>+     << (encoder_settings.experiment_cpu_load_estimator ? "on" : "off") << "}}";
>   ss << ", rtp: " << rtp.ToString();
>+  ss << ", rtcp: " << rtcp.ToString();
>   ss << ", pre_encode_callback: "
>      << (pre_encode_callback ? "(VideoSinkInterface)" : "nullptr");
>   ss << ", post_encode_callback: "
>@@ -83,80 +88,4 @@ std::string VideoSendStream::Config::ToString() const {
>   return ss.str();
> }
> 
>-std::string VideoSendStream::Config::EncoderSettings::ToString() const {
>-  std::stringstream ss;
>-  ss << "{payload_name: " << payload_name;
>-  ss << ", payload_type: " << payload_type;
>-  ss << ", encoder: " << (encoder ? "(VideoEncoder)" : "nullptr");
>-  ss << '}';
>-  return ss.str();
>-}
>-
>-VideoSendStream::Config::Rtp::Rtp() = default;
>-VideoSendStream::Config::Rtp::Rtp(const Rtp&) = default;
>-VideoSendStream::Config::Rtp::~Rtp() = default;
>-
>-VideoSendStream::Config::Rtp::Flexfec::Flexfec() = default;
>-VideoSendStream::Config::Rtp::Flexfec::Flexfec(const Flexfec&) = default;
>-VideoSendStream::Config::Rtp::Flexfec::~Flexfec() = default;
>-
>-std::string VideoSendStream::Config::Rtp::ToString() const {
>-  std::stringstream ss;
>-  ss << "{ssrcs: [";
>-  for (size_t i = 0; i < ssrcs.size(); ++i) {
>-    ss << ssrcs[i];
>-    if (i != ssrcs.size() - 1)
>-      ss << ", ";
>-  }
>-  ss << ']';
>-  ss << ", rtcp_mode: "
>-     << (rtcp_mode == RtcpMode::kCompound ? "RtcpMode::kCompound"
>-                                          : "RtcpMode::kReducedSize");
>-  ss << ", max_packet_size: " << max_packet_size;
>-  ss << ", extensions: [";
>-  for (size_t i = 0; i < extensions.size(); ++i) {
>-    ss << extensions[i].ToString();
>-    if (i != extensions.size() - 1)
>-      ss << ", ";
>-  }
>-  ss << ']';
>-
>-  ss << ", nack: {rtp_history_ms: " << nack.rtp_history_ms << '}';
>-  ss << ", ulpfec: " << ulpfec.ToString();
>-
>-  ss << ", flexfec: {payload_type: " << flexfec.payload_type;
>-  ss << ", ssrc: " << flexfec.ssrc;
>-  ss << ", protected_media_ssrcs: [";
>-  for (size_t i = 0; i < flexfec.protected_media_ssrcs.size(); ++i) {
>-    ss << flexfec.protected_media_ssrcs[i];
>-    if (i != flexfec.protected_media_ssrcs.size() - 1)
>-      ss << ", ";
>-  }
>-  ss << "]}";
>-
>-  ss << ", rtx: " << rtx.ToString();
>-  ss << ", c_name: " << c_name;
>-  ss << '}';
>-  return ss.str();
>-}
>-
>-VideoSendStream::Config::Rtp::Rtx::Rtx() = default;
>-VideoSendStream::Config::Rtp::Rtx::Rtx(const Rtx&) = default;
>-VideoSendStream::Config::Rtp::Rtx::~Rtx() = default;
>-
>-std::string VideoSendStream::Config::Rtp::Rtx::ToString() const {
>-  std::stringstream ss;
>-  ss << "{ssrcs: [";
>-  for (size_t i = 0; i < ssrcs.size(); ++i) {
>-    ss << ssrcs[i];
>-    if (i != ssrcs.size() - 1)
>-      ss << ", ";
>-  }
>-  ss << ']';
>-
>-  ss << ", payload_type: " << payload_type;
>-  ss << '}';
>-  return ss.str();
>-}
>-
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/call/video_send_stream.h b/Source/ThirdParty/libwebrtc/Source/webrtc/call/video_send_stream.h
>index a3c6f4f13aa..ef98ae4094e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/call/video_send_stream.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/call/video_send_stream.h
>@@ -17,12 +17,12 @@
> #include <vector>
> 
> #include "api/call/transport.h"
>-#include "api/rtpparameters.h"
>-#include "api/rtp_headers.h"
>-#include "api/videosinkinterface.h"
>-#include "api/videosourceinterface.h"
>+#include "api/video/video_sink_interface.h"
>+#include "api/video/video_source_interface.h"
>+#include "api/video/video_stream_encoder_settings.h"
>+#include "api/video_codecs/video_encoder_config.h"
>+#include "api/video_codecs/video_encoder_factory.h"
> #include "call/rtp_config.h"
>-#include "call/video_config.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "common_video/include/frame_callback.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>@@ -30,8 +30,6 @@
> 
> namespace webrtc {
> 
>-class VideoEncoder;
>-
> class VideoSendStream {
>  public:
>   struct StreamStats {
>@@ -69,15 +67,12 @@ class VideoSendStream {
>     uint32_t frames_dropped_by_encoder_queue = 0;
>     uint32_t frames_dropped_by_rate_limiter = 0;
>     uint32_t frames_dropped_by_encoder = 0;
>-    rtc::Optional<uint64_t> qp_sum;
>+    absl::optional<uint64_t> qp_sum;
>     // Bitrate the encoder is currently configured to use due to bandwidth
>     // limitations.
>     int target_media_bitrate_bps = 0;
>     // Bitrate the encoder is actually producing.
>     int media_bitrate_bps = 0;
>-    // Media bitrate this VideoSendStream is configured to prefer if there are
>-    // no bandwidth limitations.
>-    int preferred_media_bitrate_bps = 0;
>     bool suspended = false;
>     bool bw_limited_resolution = false;
>     bool cpu_limited_resolution = false;
>@@ -91,6 +86,7 @@ class VideoSendStream {
>     std::map<uint32_t, StreamStats> substreams;
>     webrtc::VideoContentType content_type =
>         webrtc::VideoContentType::UNSPECIFIED;
>+    uint32_t huge_frames_sent = 0;
>   };
> 
>   struct Config {
>@@ -109,93 +105,11 @@ class VideoSendStream {
> 
>     std::string ToString() const;
> 
>-    struct EncoderSettings {
>-      EncoderSettings() = default;
>-      EncoderSettings(std::string payload_name,
>-                      int payload_type,
>-                      VideoEncoder* encoder)
>-          : payload_name(std::move(payload_name)),
>-            payload_type(payload_type),
>-            encoder(encoder) {}
>-      std::string ToString() const;
>-
>-      std::string payload_name;
>-      int payload_type = -1;
>-
>-      // TODO(sophiechang): Delete this field when no one is using internal
>-      // sources anymore.
>-      bool internal_source = false;
>-
>-      // Allow 100% encoder utilization. Used for HW encoders where CPU isn't
>-      // expected to be the limiting factor, but a chip could be running at
>-      // 30fps (for example) exactly.
>-      bool full_overuse_time = false;
>-
>-      // Uninitialized VideoEncoder instance to be used for encoding. Will be
>-      // initialized from inside the VideoSendStream.
>-      VideoEncoder* encoder = nullptr;
>-    } encoder_settings;
>-
>-    static const size_t kDefaultMaxPacketSize = 1500 - 40;  // TCP over IPv4.
>-    struct Rtp {
>-      Rtp();
>-      Rtp(const Rtp&);
>-      ~Rtp();
>-      std::string ToString() const;
>-
>-      std::vector<uint32_t> ssrcs;
>-
>-      // See RtcpMode for description.
>-      RtcpMode rtcp_mode = RtcpMode::kCompound;
>-
>-      // Max RTP packet size delivered to send transport from VideoEngine.
>-      size_t max_packet_size = kDefaultMaxPacketSize;
>-
>-      // RTP header extensions to use for this send stream.
>-      std::vector<RtpExtension> extensions;
>-
>-      // See NackConfig for description.
>-      NackConfig nack;
>-
>-      // See UlpfecConfig for description.
>-      UlpfecConfig ulpfec;
>-
>-      struct Flexfec {
>-        Flexfec();
>-        Flexfec(const Flexfec&);
>-        ~Flexfec();
>-        // Payload type of FlexFEC. Set to -1 to disable sending FlexFEC.
>-        int payload_type = -1;
>-
>-        // SSRC of FlexFEC stream.
>-        uint32_t ssrc = 0;
>-
>-        // Vector containing a single element, corresponding to the SSRC of the
>-        // media stream being protected by this FlexFEC stream.
>-        // The vector MUST have size 1.
>-        //
>-        // TODO(brandtr): Update comment above when we support
>-        // multistream protection.
>-        std::vector<uint32_t> protected_media_ssrcs;
>-      } flexfec;
>-
>-      // Settings for RTP retransmission payload format, see RFC 4588 for
>-      // details.
>-      struct Rtx {
>-        Rtx();
>-        Rtx(const Rtx&);
>-        ~Rtx();
>-        std::string ToString() const;
>-        // SSRCs to use for the RTX streams.
>-        std::vector<uint32_t> ssrcs;
>-
>-        // Payload type to use for the RTX stream.
>-        int payload_type = -1;
>-      } rtx;
>-
>-      // RTCP CNAME, see RFC 3550.
>-      std::string c_name;
>-    } rtp;
>+    VideoStreamEncoderSettings encoder_settings;
>+
>+    RtpConfig rtp;
>+
>+    RtcpConfig rtcp;
> 
>     // Transport for outgoing packets.
>     Transport* send_transport = nullptr;
>@@ -236,6 +150,16 @@ class VideoSendStream {
>     Config(const Config&);
>   };
> 
>+  // Updates the sending state for all simulcast layers that the video send
>+  // stream owns. This can mean updating the activity one or for multiple
>+  // layers. The ordering of active layers is the order in which the
>+  // rtp modules are stored in the VideoSendStream.
>+  // Note: This starts stream activity if it is inactive and one of the layers
>+  // is active. This stops stream activity if it is active and all layers are
>+  // inactive.
>+  virtual void UpdateActiveSimulcastLayers(
>+      const std::vector<bool> active_layers) = 0;
>+
>   // Starts stream activity.
>   // When a stream is active, it can receive, process and deliver packets.
>   virtual void Start() = 0;
>@@ -243,23 +167,6 @@ class VideoSendStream {
>   // When a stream is stopped, it can't receive, process or deliver packets.
>   virtual void Stop() = 0;
> 
>-  // Based on the spec in
>-  // https://w3c.github.io/webrtc-pc/#idl-def-rtcdegradationpreference.
>-  // These options are enforced on a best-effort basis. For instance, all of
>-  // these options may suffer some frame drops in order to avoid queuing.
>-  // TODO(sprang): Look into possibility of more strictly enforcing the
>-  // maintain-framerate option.
>-  enum class DegradationPreference {
>-    // Don't take any actions based on over-utilization signals.
>-    kDegradationDisabled,
>-    // On over-use, request lower frame rate, possibly causing frame drops.
>-    kMaintainResolution,
>-    // On over-use, request lower resolution, possibly causing down-scaling.
>-    kMaintainFramerate,
>-    // Try to strike a "pleasing" balance between frame rate or resolution.
>-    kBalanced,
>-  };
>-
>   virtual void SetSource(
>       rtc::VideoSourceInterface<webrtc::VideoFrame>* source,
>       const DegradationPreference& degradation_preference) = 0;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/BUILD.gn
>index 34cdddc8036..50bd0f4022f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/BUILD.gn
>@@ -6,19 +6,10 @@
> # in the file PATENTS.  All contributing project authors may
> # be found in the AUTHORS file in the root of the source tree.
> 
>-import("//build/config/arm.gni")
> import("../webrtc.gni")
> 
> visibility = [ ":*" ]
> 
>-config("common_audio_config") {
>-  include_dirs = [
>-    "resampler/include",
>-    "signal_processing/include",
>-    "vad/include",
>-  ]
>-}
>-
> rtc_static_library("common_audio") {
>   visibility += [ "*" ]
>   sources = [
>@@ -60,47 +51,27 @@ rtc_static_library("common_audio") {
>   ]
> 
>   deps = [
>+    ":common_audio_c",
>     ":sinc_resampler",
>     "..:webrtc_common",
>-    "../:typedefs",
>-    "../api:optional",
>     "../rtc_base:checks",
>     "../rtc_base:gtest_prod",
>     "../rtc_base:rtc_base_approved",
>+    "../rtc_base/memory:aligned_array",
>+    "../rtc_base/memory:aligned_malloc",
>+    "../rtc_base/system:arch",
>     "../system_wrappers",
>     "../system_wrappers:cpu_features_api",
>-  ]
>-  public_deps = [
>-    ":common_audio_c",
>+    "third_party/fft4g:fft4g",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> 
>   defines = []
>-  if (rtc_use_openmax_dl) {
>-    sources += [
>-      "real_fourier_openmax.cc",
>-      "real_fourier_openmax.h",
>-    ]
>-    defines += [ "RTC_USE_OPENMAX_DL" ]
>-    if (rtc_build_openmax_dl) {
>-      deps += [ "//third_party/openmax_dl/dl" ]
>-    }
>-  }
> 
>   if (rtc_build_with_neon) {
>     deps += [ ":common_audio_neon" ]
>   }
> 
>-  if (is_win) {
>-    cflags = [ "/wd4334" ]  # Ignore warning on shift operator promotion.
>-  }
>-
>-  public_configs = [ ":common_audio_config" ]
>-
>-  if (!build_with_chromium && is_clang) {
>-    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>-    suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-  }
>-
>   if (current_cpu == "x86" || current_cpu == "x64") {
>     deps += [ ":common_audio_sse2" ]
>   }
>@@ -121,24 +92,22 @@ rtc_source_set("mock_common_audio") {
> 
> rtc_source_set("common_audio_c_arm_asm") {
>   sources = []
>+  deps = []
>   if (current_cpu == "arm") {
>-    sources += [
>-      "signal_processing/complex_bit_reverse_arm.S",
>-      "signal_processing/spl_sqrt_floor_arm.S",
>-    ]
>+    sources += [ "signal_processing/complex_bit_reverse_arm.S" ]
> 
>     if (arm_version >= 7) {
>       sources += [ "signal_processing/filter_ar_fast_q12_armv7.S" ]
>     } else {
>       sources += [ "signal_processing/filter_ar_fast_q12.c" ]
>     }
>+    deps += [ "../rtc_base/system:asm_defines" ]
>   }
> }
> 
> rtc_source_set("common_audio_c") {
>+  visibility += webrtc_default_visibility
>   sources = [
>-    "fft4g.c",
>-    "fft4g.h",
>     "ring_buffer.c",
>     "ring_buffer.h",
>     "signal_processing/auto_corr_to_refl_coef.c",
>@@ -198,7 +167,6 @@ rtc_source_set("common_audio_c") {
>       "signal_processing/include/spl_inl_mips.h",
>       "signal_processing/min_max_operations_mips.c",
>       "signal_processing/resample_by_2_mips.c",
>-      "signal_processing/spl_sqrt_floor_mips.c",
>     ]
>     if (mips_dsp_rev > 0) {
>       sources += [ "signal_processing/vector_scaling_operations_mips.c" ]
>@@ -211,26 +179,22 @@ rtc_source_set("common_audio_c") {
>     sources += [
>       "signal_processing/complex_bit_reverse.c",
>       "signal_processing/filter_ar_fast_q12.c",
>-      "signal_processing/spl_sqrt_floor.c",
>     ]
>   }
> 
>-  if (is_win) {
>-    cflags = [ "/wd4334" ]  # Ignore warning on shift operator promotion.
>-  }
>-
>-  public_configs = [ ":common_audio_config" ]
>   deps = [
>     ":common_audio_c_arm_asm",
>     ":common_audio_cc",
>     "..:webrtc_common",
>-    "../:typedefs",
>     "../rtc_base:checks",
>     "../rtc_base:compile_assert_c",
>     "../rtc_base:rtc_base_approved",
>     "../rtc_base:sanitizer",
>+    "../rtc_base/system:arch",
>     "../system_wrappers",
>     "../system_wrappers:cpu_features_api",
>+    "third_party/fft4g:fft4g",
>+    "third_party/spl_sqrt_floor",
>   ]
> }
> 
>@@ -240,10 +204,8 @@ rtc_source_set("common_audio_cc") {
>     "signal_processing/dot_product_with_scale.h",
>   ]
> 
>-  public_configs = [ ":common_audio_config" ]
>   deps = [
>     "..:webrtc_common",
>-    "../:typedefs",
>     "../rtc_base:rtc_base_approved",
>     "../system_wrappers",
>   ]
>@@ -255,9 +217,10 @@ rtc_source_set("sinc_resampler") {
>   ]
>   deps = [
>     "..:webrtc_common",
>-    "../:typedefs",
>     "../rtc_base:gtest_prod",
>     "../rtc_base:rtc_base_approved",
>+    "../rtc_base/memory:aligned_malloc",
>+    "../rtc_base/system:arch",
>     "../system_wrappers",
>   ]
> }
>@@ -281,6 +244,7 @@ rtc_source_set("fir_filter_factory") {
>     ":fir_filter",
>     "../rtc_base:checks",
>     "../rtc_base:rtc_base_approved",
>+    "../rtc_base/system:arch",
>     "../system_wrappers:cpu_features_api",
>   ]
>   if (current_cpu == "x86" || current_cpu == "x64") {
>@@ -299,20 +263,16 @@ if (current_cpu == "x86" || current_cpu == "x64") {
>       "resampler/sinc_resampler_sse.cc",
>     ]
> 
>-    if (is_posix) {
>+    if (is_posix || is_fuchsia) {
>       cflags = [ "-msse2" ]
>     }
> 
>-    if (!build_with_chromium && is_clang) {
>-      # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>-      suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-    }
>     deps = [
>       ":fir_filter",
>       ":sinc_resampler",
>       "../rtc_base:checks",
>       "../rtc_base:rtc_base_approved",
>-      "../system_wrappers",
>+      "../rtc_base/memory:aligned_malloc",
>     ]
>   }
> }
>@@ -326,9 +286,7 @@ if (rtc_build_with_neon) {
>     ]
> 
>     if (current_cpu != "arm64") {
>-      # Enable compilation for the NEON instruction set. This is needed
>-      # since //build/config/arm.gni only enables NEON for iOS, not Android.
>-      # This provides the same functionality as webrtc/build/arm_neon.gypi.
>+      # Enable compilation for the NEON instruction set.
>       suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
>       cflags = [ "-mfpu=neon" ]
>     }
>@@ -342,25 +300,18 @@ if (rtc_build_with_neon) {
>       ]
>     }
> 
>-    if (!build_with_chromium && is_clang) {
>-      # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>-      suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-    }
>-
>     deps = [
>+      ":common_audio_neon_c",
>       ":fir_filter",
>       ":sinc_resampler",
>       "../rtc_base:checks",
>       "../rtc_base:rtc_base_approved",
>-      "../system_wrappers",
>-    ]
>-
>-    public_deps = [
>-      ":common_audio_neon_c",
>+      "../rtc_base/memory:aligned_malloc",
>     ]
>   }
> 
>   rtc_source_set("common_audio_neon_c") {
>+    visibility += webrtc_default_visibility
>     sources = [
>       "signal_processing/cross_correlation_neon.c",
>       "signal_processing/downsample_fast_neon.c",
>@@ -368,9 +319,7 @@ if (rtc_build_with_neon) {
>     ]
> 
>     if (current_cpu != "arm64") {
>-      # Enable compilation for the NEON instruction set. This is needed
>-      # since //build/config/arm.gni only enables NEON for iOS, not Android.
>-      # This provides the same functionality as webrtc/build/arm_neon.gypi.
>+      # Enable compilation for the NEON instruction set.
>       suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
>       cflags = [ "-mfpu=neon" ]
>     }
>@@ -384,14 +333,11 @@ if (rtc_build_with_neon) {
>       ]
>     }
> 
>-    if (!build_with_chromium && is_clang) {
>-      # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>-      suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-    }
>     deps = [
>       ":common_audio_c",
>       "../rtc_base:checks",
>       "../rtc_base:rtc_base_approved",
>+      "../rtc_base/system:arch",
>     ]
>   }
> }
>@@ -436,28 +382,20 @@ if (rtc_include_tests) {
>       sources += [ "resampler/sinc_resampler_unittest.cc" ]
>     }
> 
>-    if (rtc_use_openmax_dl) {
>-      defines = [ "RTC_USE_OPENMAX_DL" ]
>-    }
>-
>-    if (!build_with_chromium && is_clang) {
>-      # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>-      suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-    }
>-
>     deps = [
>       ":common_audio",
>+      ":common_audio_c",
>       ":fir_filter",
>       ":fir_filter_factory",
>       ":sinc_resampler",
>       "..:webrtc_common",
>-      "../:typedefs",
>       "../rtc_base:checks",
>       "../rtc_base:rtc_base_approved",
>       "../rtc_base:rtc_base_tests_utils",
>+      "../rtc_base/system:arch",
>       "../system_wrappers:cpu_features_api",
>+      "../test:fileutils",
>       "../test:test_main",
>-      "//testing/gmock",
>       "//testing/gtest",
>     ]
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/DEPS b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/DEPS
>index 47ce4c32b6c..8a9adf19f97 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/DEPS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/DEPS
>@@ -1,4 +1,3 @@
> include_rules = [
>-  "+dl/sp/api",  # For openmax_dl.
>   "+system_wrappers",
> ]
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/audio_converter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/audio_converter.cc
>index 47d2be280ea..0f97abb86a3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/audio_converter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/audio_converter.cc
>@@ -26,12 +26,16 @@ namespace webrtc {
> 
> class CopyConverter : public AudioConverter {
>  public:
>-  CopyConverter(size_t src_channels, size_t src_frames, size_t dst_channels,
>+  CopyConverter(size_t src_channels,
>+                size_t src_frames,
>+                size_t dst_channels,
>                 size_t dst_frames)
>       : AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {}
>-  ~CopyConverter() override {};
>+  ~CopyConverter() override{};
> 
>-  void Convert(const float* const* src, size_t src_size, float* const* dst,
>+  void Convert(const float* const* src,
>+               size_t src_size,
>+               float* const* dst,
>                size_t dst_capacity) override {
>     CheckSizes(src_size, dst_capacity);
>     if (src != dst) {
>@@ -43,12 +47,16 @@ class CopyConverter : public AudioConverter {
> 
> class UpmixConverter : public AudioConverter {
>  public:
>-  UpmixConverter(size_t src_channels, size_t src_frames, size_t dst_channels,
>+  UpmixConverter(size_t src_channels,
>+                 size_t src_frames,
>+                 size_t dst_channels,
>                  size_t dst_frames)
>       : AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {}
>-  ~UpmixConverter() override {};
>+  ~UpmixConverter() override{};
> 
>-  void Convert(const float* const* src, size_t src_size, float* const* dst,
>+  void Convert(const float* const* src,
>+               size_t src_size,
>+               float* const* dst,
>                size_t dst_capacity) override {
>     CheckSizes(src_size, dst_capacity);
>     for (size_t i = 0; i < dst_frames(); ++i) {
>@@ -61,13 +69,16 @@ class UpmixConverter : public AudioConverter {
> 
> class DownmixConverter : public AudioConverter {
>  public:
>-  DownmixConverter(size_t src_channels, size_t src_frames, size_t dst_channels,
>+  DownmixConverter(size_t src_channels,
>+                   size_t src_frames,
>+                   size_t dst_channels,
>                    size_t dst_frames)
>-      : AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {
>-  }
>-  ~DownmixConverter() override {};
>+      : AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {}
>+  ~DownmixConverter() override{};
> 
>-  void Convert(const float* const* src, size_t src_size, float* const* dst,
>+  void Convert(const float* const* src,
>+               size_t src_size,
>+               float* const* dst,
>                size_t dst_capacity) override {
>     CheckSizes(src_size, dst_capacity);
>     float* dst_mono = dst[0];
>@@ -82,7 +93,9 @@ class DownmixConverter : public AudioConverter {
> 
> class ResampleConverter : public AudioConverter {
>  public:
>-  ResampleConverter(size_t src_channels, size_t src_frames, size_t dst_channels,
>+  ResampleConverter(size_t src_channels,
>+                    size_t src_frames,
>+                    size_t dst_channels,
>                     size_t dst_frames)
>       : AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {
>     resamplers_.reserve(src_channels);
>@@ -90,9 +103,11 @@ class ResampleConverter : public AudioConverter {
>       resamplers_.push_back(std::unique_ptr<PushSincResampler>(
>           new PushSincResampler(src_frames, dst_frames)));
>   }
>-  ~ResampleConverter() override {};
>+  ~ResampleConverter() override{};
> 
>-  void Convert(const float* const* src, size_t src_size, float* const* dst,
>+  void Convert(const float* const* src,
>+               size_t src_size,
>+               float* const* dst,
>                size_t dst_capacity) override {
>     CheckSizes(src_size, dst_capacity);
>     for (size_t i = 0; i < resamplers_.size(); ++i)
>@@ -108,7 +123,7 @@ class ResampleConverter : public AudioConverter {
> class CompositionConverter : public AudioConverter {
>  public:
>   explicit CompositionConverter(
>-    std::vector<std::unique_ptr<AudioConverter>> converters)
>+      std::vector<std::unique_ptr<AudioConverter>> converters)
>       : converters_(std::move(converters)) {
>     RTC_CHECK_GE(converters_.size(), 2);
>     // We need an intermediate buffer after every converter.
>@@ -117,19 +132,19 @@ class CompositionConverter : public AudioConverter {
>           std::unique_ptr<ChannelBuffer<float>>(new ChannelBuffer<float>(
>               (*it)->dst_frames(), (*it)->dst_channels())));
>   }
>-  ~CompositionConverter() override {};
>+  ~CompositionConverter() override{};
> 
>-  void Convert(const float* const* src, size_t src_size, float* const* dst,
>+  void Convert(const float* const* src,
>+               size_t src_size,
>+               float* const* dst,
>                size_t dst_capacity) override {
>     converters_.front()->Convert(src, src_size, buffers_.front()->channels(),
>                                  buffers_.front()->size());
>     for (size_t i = 2; i < converters_.size(); ++i) {
>       auto& src_buffer = buffers_[i - 2];
>       auto& dst_buffer = buffers_[i - 1];
>-      converters_[i]->Convert(src_buffer->channels(),
>-                              src_buffer->size(),
>-                              dst_buffer->channels(),
>-                              dst_buffer->size());
>+      converters_[i]->Convert(src_buffer->channels(), src_buffer->size(),
>+                              dst_buffer->channels(), dst_buffer->size());
>     }
>     converters_.back()->Convert(buffers_.back()->channels(),
>                                 buffers_.back()->size(), dst, dst_capacity);
>@@ -175,8 +190,8 @@ std::unique_ptr<AudioConverter> AudioConverter::Create(size_t src_channels,
>     sp.reset(new ResampleConverter(src_channels, src_frames, dst_channels,
>                                    dst_frames));
>   } else {
>-    sp.reset(new CopyConverter(src_channels, src_frames, dst_channels,
>-                               dst_frames));
>+    sp.reset(
>+        new CopyConverter(src_channels, src_frames, dst_channels, dst_frames));
>   }
> 
>   return sp;
>@@ -184,13 +199,12 @@ std::unique_ptr<AudioConverter> AudioConverter::Create(size_t src_channels,
> 
> // For CompositionConverter.
> AudioConverter::AudioConverter()
>-    : src_channels_(0),
>-      src_frames_(0),
>-      dst_channels_(0),
>-      dst_frames_(0) {}
>+    : src_channels_(0), src_frames_(0), dst_channels_(0), dst_frames_(0) {}
> 
>-AudioConverter::AudioConverter(size_t src_channels, size_t src_frames,
>-                               size_t dst_channels, size_t dst_frames)
>+AudioConverter::AudioConverter(size_t src_channels,
>+                               size_t src_frames,
>+                               size_t dst_channels,
>+                               size_t dst_frames)
>     : src_channels_(src_channels),
>       src_frames_(src_frames),
>       dst_channels_(dst_channels),
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/audio_converter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/audio_converter.h
>index 3f7b9a852cb..769d724e2fb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/audio_converter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/audio_converter.h
>@@ -37,8 +37,10 @@ class AudioConverter {
>   // capacity of |dst_capacity|. Both point to a series of buffers containing
>   // the samples for each channel. The sizes must correspond to the format
>   // passed to Create().
>-  virtual void Convert(const float* const* src, size_t src_size,
>-                       float* const* dst, size_t dst_capacity) = 0;
>+  virtual void Convert(const float* const* src,
>+                       size_t src_size,
>+                       float* const* dst,
>+                       size_t dst_capacity) = 0;
> 
>   size_t src_channels() const { return src_channels_; }
>   size_t src_frames() const { return src_frames_; }
>@@ -47,7 +49,9 @@ class AudioConverter {
> 
>  protected:
>   AudioConverter();
>-  AudioConverter(size_t src_channels, size_t src_frames, size_t dst_channels,
>+  AudioConverter(size_t src_channels,
>+                 size_t src_frames,
>+                 size_t dst_channels,
>                  size_t dst_frames);
> 
>   // Helper to RTC_CHECK that inputs are correctly sized.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/audio_converter_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/audio_converter_unittest.cc
>index e9937fd8cca..b99d825c2b5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/audio_converter_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/audio_converter_unittest.cc
>@@ -8,8 +8,8 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include <cmath>
> #include <algorithm>
>+#include <cmath>
> #include <memory>
> #include <vector>
> 
>@@ -52,8 +52,7 @@ float ComputeSNR(const ChannelBuffer<float>& ref,
> 
>   // Search within one sample of the expected delay.
>   for (size_t delay = std::max(expected_delay, static_cast<size_t>(1)) - 1;
>-       delay <= std::min(expected_delay + 1, ref.num_frames());
>-       ++delay) {
>+       delay <= std::min(expected_delay + 1, ref.num_frames()); ++delay) {
>     float mse = 0;
>     float variance = 0;
>     float mean = 0;
>@@ -92,8 +91,8 @@ void RunAudioConverterTest(size_t src_channels,
>                            int dst_sample_rate_hz) {
>   const float kSrcLeft = 0.0002f;
>   const float kSrcRight = 0.0001f;
>-  const float resampling_factor = (1.f * src_sample_rate_hz) /
>-      dst_sample_rate_hz;
>+  const float resampling_factor =
>+      (1.f * src_sample_rate_hz) / dst_sample_rate_hz;
>   const float dst_left = resampling_factor * kSrcLeft;
>   const float dst_right = resampling_factor * kSrcRight;
>   const float dst_mono = (dst_left + dst_right) / 2;
>@@ -124,13 +123,15 @@ void RunAudioConverterTest(size_t src_channels,
>   ScopedBuffer ref_buffer = CreateBuffer(ref_data, dst_frames);
> 
>   // The sinc resampler has a known delay, which we compute here.
>-  const size_t delay_frames = src_sample_rate_hz == dst_sample_rate_hz ? 0 :
>-      static_cast<size_t>(
>-          PushSincResampler::AlgorithmicDelaySeconds(src_sample_rate_hz) *
>-          dst_sample_rate_hz);
>+  const size_t delay_frames =
>+      src_sample_rate_hz == dst_sample_rate_hz
>+          ? 0
>+          : static_cast<size_t>(
>+                PushSincResampler::AlgorithmicDelaySeconds(src_sample_rate_hz) *
>+                dst_sample_rate_hz);
>   // SNR reported on the same line later.
>-  printf("(%" PRIuS ", %d Hz) -> (%" PRIuS ", %d Hz) ",
>-         src_channels, src_sample_rate_hz, dst_channels, dst_sample_rate_hz);
>+  printf("(%" PRIuS ", %d Hz) -> (%" PRIuS ", %d Hz) ", src_channels,
>+         src_sample_rate_hz, dst_channels, dst_sample_rate_hz);
> 
>   std::unique_ptr<AudioConverter> converter = AudioConverter::Create(
>       src_channels, src_frames, dst_channels, dst_frames);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/audio_ring_buffer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/audio_ring_buffer.cc
>index e7b5d81ac6a..b3bdc252f67 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/audio_ring_buffer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/audio_ring_buffer.cc
>@@ -24,11 +24,12 @@ AudioRingBuffer::AudioRingBuffer(size_t channels, size_t max_frames) {
> }
> 
> AudioRingBuffer::~AudioRingBuffer() {
>-  for (auto buf : buffers_)
>+  for (auto* buf : buffers_)
>     WebRtc_FreeBuffer(buf);
> }
> 
>-void AudioRingBuffer::Write(const float* const* data, size_t channels,
>+void AudioRingBuffer::Write(const float* const* data,
>+                            size_t channels,
>                             size_t frames) {
>   RTC_DCHECK_EQ(buffers_.size(), channels);
>   for (size_t i = 0; i < channels; ++i) {
>@@ -57,7 +58,7 @@ size_t AudioRingBuffer::WriteFramesAvailable() const {
> }
> 
> void AudioRingBuffer::MoveReadPositionForward(size_t frames) {
>-  for (auto buf : buffers_) {
>+  for (auto* buf : buffers_) {
>     const size_t moved =
>         static_cast<size_t>(WebRtc_MoveReadPtr(buf, static_cast<int>(frames)));
>     RTC_CHECK_EQ(moved, frames);
>@@ -65,7 +66,7 @@ void AudioRingBuffer::MoveReadPositionForward(size_t frames) {
> }
> 
> void AudioRingBuffer::MoveReadPositionBackward(size_t frames) {
>-  for (auto buf : buffers_) {
>+  for (auto* buf : buffers_) {
>     const size_t moved = static_cast<size_t>(
>         -WebRtc_MoveReadPtr(buf, -static_cast<int>(frames)));
>     RTC_CHECK_EQ(moved, frames);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/audio_ring_buffer_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/audio_ring_buffer_unittest.cc
>index 2fcf80035f5..d411195aa73 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/audio_ring_buffer_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/audio_ring_buffer_unittest.cc
>@@ -17,8 +17,8 @@
> 
> namespace webrtc {
> 
>-class AudioRingBufferTest :
>-    public ::testing::TestWithParam< ::testing::tuple<int, int, int, int> > {
>+class AudioRingBufferTest
>+    : public ::testing::TestWithParam< ::testing::tuple<int, int, int, int> > {
> };
> 
> void ReadAndWriteTest(const ChannelBuffer<float>& input,
>@@ -72,10 +72,8 @@ TEST_P(AudioRingBufferTest, ReadDataMatchesWrittenData) {
>       input.channels()[i][j] = (i + 1) * (j + 1);
> 
>   ChannelBuffer<float> output(kFrames, static_cast<int>(num_channels));
>-  ReadAndWriteTest(input,
>-                   ::testing::get<0>(GetParam()),
>-                   ::testing::get<1>(GetParam()),
>-                   ::testing::get<2>(GetParam()),
>+  ReadAndWriteTest(input, ::testing::get<0>(GetParam()),
>+                   ::testing::get<1>(GetParam()), ::testing::get<2>(GetParam()),
>                    &output);
> 
>   // Verify the read data matches the input.
>@@ -85,7 +83,8 @@ TEST_P(AudioRingBufferTest, ReadDataMatchesWrittenData) {
> }
> 
> INSTANTIATE_TEST_CASE_P(
>-    AudioRingBufferTest, AudioRingBufferTest,
>+    AudioRingBufferTest,
>+    AudioRingBufferTest,
>     ::testing::Combine(::testing::Values(10, 20, 42),  // num_write_chunk_frames
>                        ::testing::Values(1, 10, 17),   // num_read_chunk_frames
>                        ::testing::Values(100, 256),    // buffer_frames
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/audio_util.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/audio_util.cc
>index b442a146f7a..735ba5f1886 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/audio_util.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/audio_util.cc
>@@ -10,8 +10,6 @@
> 
> #include "common_audio/include/audio_util.h"
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>-
> namespace webrtc {
> 
> void FloatToS16(const float* src, size_t size, int16_t* dest) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/audio_util_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/audio_util_unittest.cc
>index 7af3c369602..cf85a2d46cb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/audio_util_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/audio_util_unittest.cc
>@@ -9,9 +9,10 @@
>  */
> 
> #include "common_audio/include/audio_util.h"
>+
>+#include "rtc_base/arraysize.h"
> #include "test/gmock.h"
> #include "test/gtest.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> namespace {
>@@ -26,84 +27,121 @@ void ExpectArraysEq(const int16_t* ref, const int16_t* test, size_t length) {
> 
> void ExpectArraysEq(const float* ref, const float* test, size_t length) {
>   for (size_t i = 0; i < length; ++i) {
>-    EXPECT_FLOAT_EQ(ref[i], test[i]);
>+    EXPECT_NEAR(ref[i], test[i], 0.01f);
>   }
> }
> 
> TEST(AudioUtilTest, FloatToS16) {
>-  const size_t kSize = 9;
>-  const float kInput[kSize] = {0.f,
>-                               0.4f / 32767.f,
>-                               0.6f / 32767.f,
>-                               -0.4f / 32768.f,
>-                               -0.6f / 32768.f,
>-                               1.f,
>-                               -1.f,
>-                               1.1f,
>-                               -1.1f};
>-  const int16_t kReference[kSize] = {0,     0,      1,     0,     -1,
>-                                     32767, -32768, 32767, -32768};
>+  static constexpr float kInput[] = {0.f,
>+                                     0.4f / 32767.f,
>+                                     0.6f / 32767.f,
>+                                     -0.4f / 32768.f,
>+                                     -0.6f / 32768.f,
>+                                     1.f,
>+                                     -1.f,
>+                                     1.1f,
>+                                     -1.1f};
>+  static constexpr int16_t kReference[] = {0,     0,      1,     0,     -1,
>+                                           32767, -32768, 32767, -32768};
>+  static constexpr size_t kSize = arraysize(kInput);
>+  static_assert(arraysize(kReference) == kSize, "");
>   int16_t output[kSize];
>   FloatToS16(kInput, kSize, output);
>   ExpectArraysEq(kReference, output, kSize);
> }
> 
> TEST(AudioUtilTest, S16ToFloat) {
>-  const size_t kSize = 7;
>-  const int16_t kInput[kSize] = {0, 1, -1, 16384, -16384, 32767, -32768};
>-  const float kReference[kSize] = {
>+  static constexpr int16_t kInput[] = {0, 1, -1, 16384, -16384, 32767, -32768};
>+  static constexpr float kReference[] = {
>       0.f, 1.f / 32767.f, -1.f / 32768.f, 16384.f / 32767.f, -0.5f, 1.f, -1.f};
>+  static constexpr size_t kSize = arraysize(kInput);
>+  static_assert(arraysize(kReference) == kSize, "");
>   float output[kSize];
>   S16ToFloat(kInput, kSize, output);
>   ExpectArraysEq(kReference, output, kSize);
> }
> 
> TEST(AudioUtilTest, FloatS16ToS16) {
>-  const size_t kSize = 7;
>-  const float kInput[kSize] = {0.f,   0.4f,    0.5f,    -0.4f,
>-                               -0.5f, 32768.f, -32769.f};
>-  const int16_t kReference[kSize] = {0, 0, 1, 0, -1, 32767, -32768};
>+  static constexpr float kInput[] = {0.f,   0.4f,    0.5f,    -0.4f,
>+                                     -0.5f, 32768.f, -32769.f};
>+  static constexpr int16_t kReference[] = {0, 0, 1, 0, -1, 32767, -32768};
>+  static constexpr size_t kSize = arraysize(kInput);
>+  static_assert(arraysize(kReference) == kSize, "");
>   int16_t output[kSize];
>   FloatS16ToS16(kInput, kSize, output);
>   ExpectArraysEq(kReference, output, kSize);
> }
> 
> TEST(AudioUtilTest, FloatToFloatS16) {
>-  const size_t kSize = 9;
>-  const float kInput[kSize] = {0.f,
>-                               0.4f / 32767.f,
>-                               0.6f / 32767.f,
>-                               -0.4f / 32768.f,
>-                               -0.6f / 32768.f,
>-                               1.f,
>-                               -1.f,
>-                               1.1f,
>-                               -1.1f};
>-  const float kReference[kSize] = {0.f,     0.4f,     0.6f,     -0.4f,    -0.6f,
>-                                   32767.f, -32768.f, 36043.7f, -36044.8f};
>+  static constexpr float kInput[] = {0.f,
>+                                     0.4f / 32767.f,
>+                                     0.6f / 32767.f,
>+                                     -0.4f / 32768.f,
>+                                     -0.6f / 32768.f,
>+                                     1.f,
>+                                     -1.f,
>+                                     1.1f,
>+                                     -1.1f};
>+  static constexpr float kReference[] = {
>+      0.f, 0.4f, 0.6f, -0.4f, -0.6f, 32767.f, -32768.f, 36043.7f, -36044.8f};
>+  static constexpr size_t kSize = arraysize(kInput);
>+  static_assert(arraysize(kReference) == kSize, "");
>   float output[kSize];
>   FloatToFloatS16(kInput, kSize, output);
>   ExpectArraysEq(kReference, output, kSize);
> }
> 
> TEST(AudioUtilTest, FloatS16ToFloat) {
>-  const size_t kSize = 9;
>-  const float kInput[kSize] = {0.f,     0.4f,     0.6f,     -0.4f,    -0.6f,
>-                               32767.f, -32768.f, 36043.7f, -36044.8f};
>-  const float kReference[kSize] = {0.f,
>-                                   0.4f / 32767.f,
>-                                   0.6f / 32767.f,
>-                                   -0.4f / 32768.f,
>-                                   -0.6f / 32768.f,
>-                                   1.f,
>-                                   -1.f,
>-                                   1.1f,
>-                                   -1.1f};
>+  static constexpr float kInput[] = {
>+      0.f, 0.4f, 0.6f, -0.4f, -0.6f, 32767.f, -32768.f, 36043.7f, -36044.8f};
>+  static constexpr float kReference[] = {0.f,
>+                                         0.4f / 32767.f,
>+                                         0.6f / 32767.f,
>+                                         -0.4f / 32768.f,
>+                                         -0.6f / 32768.f,
>+                                         1.f,
>+                                         -1.f,
>+                                         1.1f,
>+                                         -1.1f};
>+  static constexpr size_t kSize = arraysize(kInput);
>+  static_assert(arraysize(kReference) == kSize, "");
>   float output[kSize];
>   FloatS16ToFloat(kInput, kSize, output);
>   ExpectArraysEq(kReference, output, kSize);
> }
> 
>+TEST(AudioUtilTest, DbfsToFloatS16) {
>+  static constexpr float kInput[] = {-90.f, -70.f, -30.f, -20.f, -10.f,
>+                                     -5.f,  -1.f,  0.f,   1.f};
>+  static constexpr float kReference[] = {
>+      1.036215186f, 10.36215115f, 1036.215088f, 3276.800049f, 10362.15137f,
>+      18426.80078f, 29204.51172f, 32768.f,      36766.30078f};
>+  static constexpr size_t kSize = arraysize(kInput);
>+  static_assert(arraysize(kReference) == kSize, "");
>+  float output[kSize];
>+  for (size_t i = 0; i < kSize; ++i) {
>+    output[i] = DbfsToFloatS16(kInput[i]);
>+  }
>+  ExpectArraysEq(kReference, output, kSize);
>+}
>+
>+TEST(AudioUtilTest, FloatS16ToDbfs) {
>+  static constexpr float kInput[] = {1.036215143f, 10.36215143f,  1036.215143f,
>+                                     3276.8f,      10362.151436f, 18426.800543f,
>+                                     29204.51074f, 32768.0f,      36766.30071f};
>+
>+  static constexpr float kReference[] = {
>+      -90.f, -70.f, -30.f, -20.f, -10.f, -5.f, -1.f, 0.f, 0.9999923706f};
>+  static constexpr size_t kSize = arraysize(kInput);
>+  static_assert(arraysize(kReference) == kSize, "");
>+
>+  float output[kSize];
>+  for (size_t i = 0; i < kSize; ++i) {
>+    output[i] = FloatS16ToDbfs(kInput[i]);
>+  }
>+  ExpectArraysEq(kReference, output, kSize);
>+}
>+
> TEST(AudioUtilTest, InterleavingStereo) {
>   const int16_t kInterleaved[] = {2, 3, 4, 9, 8, 27, 16, 81};
>   const size_t kSamplesPerChannel = 4;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/blocker.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/blocker.cc
>index 7d09d21ea84..3dc8ed8040f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/blocker.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/blocker.cc
>@@ -41,8 +41,7 @@ void CopyFrames(const float* const* src,
>                 float* const* dst,
>                 size_t dst_start_index) {
>   for (size_t i = 0; i < num_channels; ++i) {
>-    memcpy(&dst[i][dst_start_index],
>-           &src[i][src_start_index],
>+    memcpy(&dst[i][dst_start_index], &src[i][src_start_index],
>            num_frames * sizeof(dst[i][dst_start_index]));
>   }
> }
>@@ -55,8 +54,7 @@ void MoveFrames(const float* const* src,
>                 float* const* dst,
>                 size_t dst_start_index) {
>   for (size_t i = 0; i < num_channels; ++i) {
>-    memmove(&dst[i][dst_start_index],
>-            &src[i][src_start_index],
>+    memmove(&dst[i][dst_start_index], &src[i][src_start_index],
>             num_frames * sizeof(dst[i][dst_start_index]));
>   }
> }
>@@ -87,9 +85,9 @@ void ApplyWindow(const float* window,
> size_t gcd(size_t a, size_t b) {
>   size_t tmp;
>   while (b) {
>-     tmp = a;
>-     a = b;
>-     b = tmp % b;
>+    tmp = a;
>+    a = b;
>+    b = tmp % b;
>   }
>   return a;
> }
>@@ -184,51 +182,30 @@ void Blocker::ProcessChunk(const float* const* input,
>                        block_size_);
>     input_buffer_.MoveReadPositionBackward(block_size_ - shift_amount_);
> 
>-    ApplyWindow(window_.get(),
>-                block_size_,
>-                num_input_channels_,
>+    ApplyWindow(window_.get(), block_size_, num_input_channels_,
>                 input_block_.channels());
>-    callback_->ProcessBlock(input_block_.channels(),
>-                            block_size_,
>-                            num_input_channels_,
>-                            num_output_channels_,
>+    callback_->ProcessBlock(input_block_.channels(), block_size_,
>+                            num_input_channels_, num_output_channels_,
>                             output_block_.channels());
>-    ApplyWindow(window_.get(),
>-                block_size_,
>-                num_output_channels_,
>+    ApplyWindow(window_.get(), block_size_, num_output_channels_,
>                 output_block_.channels());
> 
>-    AddFrames(output_buffer_.channels(),
>-              first_frame_in_block,
>-              output_block_.channels(),
>-              0,
>-              block_size_,
>-              num_output_channels_,
>-              output_buffer_.channels(),
>-              first_frame_in_block);
>+    AddFrames(output_buffer_.channels(), first_frame_in_block,
>+              output_block_.channels(), 0, block_size_, num_output_channels_,
>+              output_buffer_.channels(), first_frame_in_block);
> 
>     first_frame_in_block += shift_amount_;
>   }
> 
>   // Copy output buffer to output
>-  CopyFrames(output_buffer_.channels(),
>-             0,
>-             chunk_size_,
>-             num_output_channels_,
>-             output,
>-             0);
>+  CopyFrames(output_buffer_.channels(), 0, chunk_size_, num_output_channels_,
>+             output, 0);
> 
>   // Copy output buffer [chunk_size_, chunk_size_ + initial_delay]
>   // to output buffer [0, initial_delay], zero the rest.
>-  MoveFrames(output_buffer_.channels(),
>-             chunk_size,
>-             initial_delay_,
>-             num_output_channels_,
>-             output_buffer_.channels(),
>-             0);
>-  ZeroOut(output_buffer_.channels(),
>-          initial_delay_,
>-          chunk_size_,
>+  MoveFrames(output_buffer_.channels(), chunk_size, initial_delay_,
>+             num_output_channels_, output_buffer_.channels(), 0);
>+  ZeroOut(output_buffer_.channels(), initial_delay_, chunk_size_,
>           num_output_channels_);
> 
>   // Calculate new starting frames.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/blocker_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/blocker_unittest.cc
>index 296efab15b8..85a24f68478 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/blocker_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/blocker_unittest.cc
>@@ -71,11 +71,8 @@ class BlockerTest : public ::testing::Test {
>     size_t end = chunk_size - 1;
>     while (end < num_frames) {
>       CopyTo(input_chunk, 0, start, num_input_channels, chunk_size, input);
>-      blocker->ProcessChunk(input_chunk,
>-                            chunk_size,
>-                            num_input_channels,
>-                            num_output_channels,
>-                            output_chunk);
>+      blocker->ProcessChunk(input_chunk, chunk_size, num_input_channels,
>+                            num_output_channels, output_chunk);
>       CopyTo(output, start, 0, num_output_channels, chunk_size, output_chunk);
> 
>       start += chunk_size;
>@@ -116,8 +113,7 @@ class BlockerTest : public ::testing::Test {
>                      size_t num_frames,
>                      const float* const* src) {
>     for (size_t i = 0; i < num_channels; ++i) {
>-      memcpy(&dst[i][start_index_dst],
>-             &src[i][start_index_src],
>+      memcpy(&dst[i][start_index_dst], &src[i][start_index_src],
>              num_frames * sizeof(float));
>     }
>   }
>@@ -152,27 +148,15 @@ TEST_F(BlockerTest, TestBlockerMutuallyPrimeChunkandBlockSize) {
>   ChannelBuffer<float> output_chunk_cb(kChunkSize, kNumOutputChannels);
> 
>   PlusThreeBlockerCallback callback;
>-  Blocker blocker(kChunkSize,
>-                  kBlockSize,
>-                  kNumInputChannels,
>-                  kNumOutputChannels,
>-                  kWindow,
>-                  kShiftAmount,
>-                  &callback);
>-
>-  RunTest(&blocker,
>-          kChunkSize,
>-          kNumFrames,
>-          input_cb.channels(),
>-          input_chunk_cb.channels(),
>-          actual_output_cb.channels(),
>-          output_chunk_cb.channels(),
>-          kNumInputChannels,
>-          kNumOutputChannels);
>+  Blocker blocker(kChunkSize, kBlockSize, kNumInputChannels, kNumOutputChannels,
>+                  kWindow, kShiftAmount, &callback);
>+
>+  RunTest(&blocker, kChunkSize, kNumFrames, input_cb.channels(),
>+          input_chunk_cb.channels(), actual_output_cb.channels(),
>+          output_chunk_cb.channels(), kNumInputChannels, kNumOutputChannels);
> 
>   ValidateSignalEquality(expected_output_cb.channels(),
>-                         actual_output_cb.channels(),
>-                         kNumOutputChannels,
>+                         actual_output_cb.channels(), kNumOutputChannels,
>                          kNumFrames);
> }
> 
>@@ -205,27 +189,15 @@ TEST_F(BlockerTest, TestBlockerMutuallyPrimeShiftAndBlockSize) {
>   ChannelBuffer<float> output_chunk_cb(kChunkSize, kNumOutputChannels);
> 
>   PlusThreeBlockerCallback callback;
>-  Blocker blocker(kChunkSize,
>-                  kBlockSize,
>-                  kNumInputChannels,
>-                  kNumOutputChannels,
>-                  kWindow,
>-                  kShiftAmount,
>-                  &callback);
>-
>-  RunTest(&blocker,
>-          kChunkSize,
>-          kNumFrames,
>-          input_cb.channels(),
>-          input_chunk_cb.channels(),
>-          actual_output_cb.channels(),
>-          output_chunk_cb.channels(),
>-          kNumInputChannels,
>-          kNumOutputChannels);
>+  Blocker blocker(kChunkSize, kBlockSize, kNumInputChannels, kNumOutputChannels,
>+                  kWindow, kShiftAmount, &callback);
>+
>+  RunTest(&blocker, kChunkSize, kNumFrames, input_cb.channels(),
>+          input_chunk_cb.channels(), actual_output_cb.channels(),
>+          output_chunk_cb.channels(), kNumInputChannels, kNumOutputChannels);
> 
>   ValidateSignalEquality(expected_output_cb.channels(),
>-                         actual_output_cb.channels(),
>-                         kNumOutputChannels,
>+                         actual_output_cb.channels(), kNumOutputChannels,
>                          kNumFrames);
> }
> 
>@@ -258,27 +230,15 @@ TEST_F(BlockerTest, TestBlockerNoOverlap) {
>   ChannelBuffer<float> output_chunk_cb(kChunkSize, kNumOutputChannels);
> 
>   PlusThreeBlockerCallback callback;
>-  Blocker blocker(kChunkSize,
>-                  kBlockSize,
>-                  kNumInputChannels,
>-                  kNumOutputChannels,
>-                  kWindow,
>-                  kShiftAmount,
>-                  &callback);
>-
>-  RunTest(&blocker,
>-          kChunkSize,
>-          kNumFrames,
>-          input_cb.channels(),
>-          input_chunk_cb.channels(),
>-          actual_output_cb.channels(),
>-          output_chunk_cb.channels(),
>-          kNumInputChannels,
>-          kNumOutputChannels);
>+  Blocker blocker(kChunkSize, kBlockSize, kNumInputChannels, kNumOutputChannels,
>+                  kWindow, kShiftAmount, &callback);
>+
>+  RunTest(&blocker, kChunkSize, kNumFrames, input_cb.channels(),
>+          input_chunk_cb.channels(), actual_output_cb.channels(),
>+          output_chunk_cb.channels(), kNumInputChannels, kNumOutputChannels);
> 
>   ValidateSignalEquality(expected_output_cb.channels(),
>-                         actual_output_cb.channels(),
>-                         kNumOutputChannels,
>+                         actual_output_cb.channels(), kNumOutputChannels,
>                          kNumFrames);
> }
> 
>@@ -286,14 +246,14 @@ TEST_F(BlockerTest, InitialDelaysAreMinimum) {
>   const size_t kNumInputChannels = 3;
>   const size_t kNumOutputChannels = 2;
>   const size_t kNumFrames = 1280;
>-  const size_t kChunkSize[] =
>-      {80, 80, 80, 80, 80, 80, 160, 160, 160, 160, 160, 160};
>-  const size_t kBlockSize[] =
>-      {64, 64, 64, 128, 128, 128, 128, 128, 128, 256, 256, 256};
>-  const size_t kShiftAmount[] =
>-      {16, 32, 64, 32, 64, 128, 32, 64, 128, 64, 128, 256};
>-  const size_t kInitialDelay[] =
>-      {48, 48, 48, 112, 112, 112, 96, 96, 96, 224, 224, 224};
>+  const size_t kChunkSize[] = {80,  80,  80,  80,  80,  80,
>+                               160, 160, 160, 160, 160, 160};
>+  const size_t kBlockSize[] = {64,  64,  64,  128, 128, 128,
>+                               128, 128, 128, 256, 256, 256};
>+  const size_t kShiftAmount[] = {16, 32, 64,  32, 64,  128,
>+                                 32, 64, 128, 64, 128, 256};
>+  const size_t kInitialDelay[] = {48, 48, 48, 112, 112, 112,
>+                                  96, 96, 96, 224, 224, 224};
> 
>   float input[kNumInputChannels][kNumFrames];
>   for (size_t i = 0; i < kNumInputChannels; ++i) {
>@@ -317,27 +277,15 @@ TEST_F(BlockerTest, InitialDelaysAreMinimum) {
>     ChannelBuffer<float> input_chunk_cb(kChunkSize[i], kNumInputChannels);
>     ChannelBuffer<float> output_chunk_cb(kChunkSize[i], kNumOutputChannels);
> 
>-    Blocker blocker(kChunkSize[i],
>-                    kBlockSize[i],
>-                    kNumInputChannels,
>-                    kNumOutputChannels,
>-                    window.get(),
>-                    kShiftAmount[i],
>+    Blocker blocker(kChunkSize[i], kBlockSize[i], kNumInputChannels,
>+                    kNumOutputChannels, window.get(), kShiftAmount[i],
>                     &callback);
> 
>-    RunTest(&blocker,
>-            kChunkSize[i],
>-            kNumFrames,
>-            input_cb.channels(),
>-            input_chunk_cb.channels(),
>-            output_cb.channels(),
>-            output_chunk_cb.channels(),
>-            kNumInputChannels,
>-            kNumOutputChannels);
>-
>-    ValidateInitialDelay(output_cb.channels(),
>-                         kNumOutputChannels,
>-                         kNumFrames,
>+    RunTest(&blocker, kChunkSize[i], kNumFrames, input_cb.channels(),
>+            input_chunk_cb.channels(), output_cb.channels(),
>+            output_chunk_cb.channels(), kNumInputChannels, kNumOutputChannels);
>+
>+    ValidateInitialDelay(output_cb.channels(), kNumOutputChannels, kNumFrames,
>                          kInitialDelay[i]);
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/channel_buffer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/channel_buffer.cc
>index df45f6d0f6b..38d231e09d0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/channel_buffer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/channel_buffer.cc
>@@ -68,9 +68,7 @@ void IFChannelBuffer::RefreshI() const {
>     ibuf_.set_num_channels(fbuf_.num_channels());
>     const float* const* float_channels = fbuf_.channels();
>     for (size_t i = 0; i < fbuf_.num_channels(); ++i) {
>-      FloatS16ToS16(float_channels[i],
>-                    ibuf_.num_frames(),
>-                    int_channels[i]);
>+      FloatS16ToS16(float_channels[i], ibuf_.num_frames(), int_channels[i]);
>     }
>     ivalid_ = true;
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/channel_buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/channel_buffer.h
>index 024868c46cd..3f9ba9c0e16 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/channel_buffer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/channel_buffer.h
>@@ -40,9 +40,7 @@ namespace webrtc {
> template <typename T>
> class ChannelBuffer {
>  public:
>-  ChannelBuffer(size_t num_frames,
>-                size_t num_channels,
>-                size_t num_bands = 1)
>+  ChannelBuffer(size_t num_frames, size_t num_channels, size_t num_bands = 1)
>       : data_(new T[num_frames * num_channels]()),
>         channels_(new T*[num_channels * num_bands]),
>         bands_(new T*[num_channels * num_bands]),
>@@ -119,7 +117,7 @@ class ChannelBuffer {
>   size_t num_frames_per_band() const { return num_frames_per_band_; }
>   size_t num_channels() const { return num_channels_; }
>   size_t num_bands() const { return num_bands_; }
>-  size_t size() const {return num_frames_ * num_allocated_channels_; }
>+  size_t size() const { return num_frames_ * num_allocated_channels_; }
> 
>   void set_num_channels(size_t num_channels) {
>     RTC_DCHECK_LE(num_channels, num_allocated_channels_);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/fir_filter_c.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/fir_filter_c.cc
>index 6fe247011bf..3418434672a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/fir_filter_c.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/fir_filter_c.cc
>@@ -20,8 +20,7 @@
> 
> namespace webrtc {
> 
>-FIRFilterC::~FIRFilterC() {
>-}
>+FIRFilterC::~FIRFilterC() {}
> 
> FIRFilterC::FIRFilterC(const float* coefficients, size_t coefficients_length)
>     : coefficients_length_(coefficients_length),
>@@ -52,11 +51,10 @@ void FIRFilterC::Filter(const float* in, size_t length, float* out) {
> 
>   // Update current state.
>   if (length >= state_length_) {
>-    memcpy(
>-        state_.get(), &in[length - state_length_], state_length_ * sizeof(*in));
>+    memcpy(state_.get(), &in[length - state_length_],
>+           state_length_ * sizeof(*in));
>   } else {
>-    memmove(state_.get(),
>-            &state_[length],
>+    memmove(state_.get(), &state_[length],
>             (state_length_ - length) * sizeof(state_[0]));
>     memcpy(&state_[state_length_ - length], in, length * sizeof(*in));
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/fir_filter_c.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/fir_filter_c.h
>index ffce838a67a..d263e1ba606 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/fir_filter_c.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/fir_filter_c.h
>@@ -20,8 +20,7 @@ namespace webrtc {
> 
> class FIRFilterC : public FIRFilter {
>  public:
>-  FIRFilterC(const float* coefficients,
>-             size_t coefficients_length);
>+  FIRFilterC(const float* coefficients, size_t coefficients_length);
>   ~FIRFilterC() override;
> 
>   void Filter(const float* in, size_t length, float* out) override;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/fir_filter_factory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/fir_filter_factory.cc
>index c15c2e0c26c..3243b27ef9a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/fir_filter_factory.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/fir_filter_factory.cc
>@@ -12,6 +12,7 @@
> 
> #include "common_audio/fir_filter_c.h"
> #include "rtc_base/checks.h"
>+#include "rtc_base/system/arch.h"
> #include "system_wrappers/include/cpu_features_wrapper.h"
> 
> #if defined(WEBRTC_HAS_NEON)
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/fir_filter_factory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/fir_filter_factory.h
>index 2e7ca9b89c9..a952541035e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/fir_filter_factory.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/fir_filter_factory.h
>@@ -1,4 +1,4 @@
>-  /*
>+/*
>  *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
>  *
>  *  Use of this source code is governed by a BSD-style license
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/fir_filter_neon.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/fir_filter_neon.cc
>index d9f91b76002..f668841be63 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/fir_filter_neon.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/fir_filter_neon.cc
>@@ -14,12 +14,11 @@
> #include <string.h>
> 
> #include "rtc_base/checks.h"
>-#include "system_wrappers/include/aligned_malloc.h"
>+#include "rtc_base/memory/aligned_malloc.h"
> 
> namespace webrtc {
> 
>-FIRFilterNEON::~FIRFilterNEON() {
>-}
>+FIRFilterNEON::~FIRFilterNEON() {}
> 
> FIRFilterNEON::FIRFilterNEON(const float* coefficients,
>                              size_t coefficients_length,
>@@ -40,8 +39,7 @@ FIRFilterNEON::FIRFilterNEON(const float* coefficients,
>   for (size_t i = 0; i < coefficients_length; ++i) {
>     coefficients_[i + padding] = coefficients[coefficients_length - i - 1];
>   }
>-  memset(state_.get(),
>-         0.f,
>+  memset(state_.get(), 0.f,
>          (max_input_length + state_length_) * sizeof(state_[0]));
> }
> 
>@@ -60,8 +58,8 @@ void FIRFilterNEON::Filter(const float* in, size_t length, float* out) {
>     float32x4_t m_in;
> 
>     for (size_t j = 0; j < coefficients_length_; j += 4) {
>-       m_in = vld1q_f32(in_ptr + j);
>-       m_sum = vmlaq_f32(m_sum, m_in, vld1q_f32(coef_ptr + j));
>+      m_in = vld1q_f32(in_ptr + j);
>+      m_sum = vmlaq_f32(m_sum, m_in, vld1q_f32(coef_ptr + j));
>     }
> 
>     float32x2_t m_half = vadd_f32(vget_high_f32(m_sum), vget_low_f32(m_sum));
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/fir_filter_neon.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/fir_filter_neon.h
>index 5696df8f0e0..1ffefd80dc5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/fir_filter_neon.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/fir_filter_neon.h
>@@ -14,7 +14,7 @@
> #include <memory>
> 
> #include "common_audio/fir_filter.h"
>-#include "system_wrappers/include/aligned_malloc.h"
>+#include "rtc_base/memory/aligned_malloc.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/fir_filter_sse.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/fir_filter_sse.cc
>index 3302d564917..ee75fb38ad4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/fir_filter_sse.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/fir_filter_sse.cc
>@@ -15,12 +15,11 @@
> #include <xmmintrin.h>
> 
> #include "rtc_base/checks.h"
>-#include "system_wrappers/include/aligned_malloc.h"
>+#include "rtc_base/memory/aligned_malloc.h"
> 
> namespace webrtc {
> 
>-FIRFilterSSE2::~FIRFilterSSE2() {
>-}
>+FIRFilterSSE2::~FIRFilterSSE2() {}
> 
> FIRFilterSSE2::FIRFilterSSE2(const float* coefficients,
>                              size_t coefficients_length,
>@@ -41,8 +40,7 @@ FIRFilterSSE2::FIRFilterSSE2(const float* coefficients,
>   for (size_t i = 0; i < coefficients_length; ++i) {
>     coefficients_[i + padding] = coefficients[coefficients_length - i - 1];
>   }
>-  memset(state_.get(),
>-         0,
>+  memset(state_.get(), 0,
>          (max_input_length + state_length_) * sizeof(state_[0]));
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/fir_filter_sse.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/fir_filter_sse.h
>index 6506024aff5..7707f933d9e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/fir_filter_sse.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/fir_filter_sse.h
>@@ -14,7 +14,7 @@
> #include <memory>
> 
> #include "common_audio/fir_filter.h"
>-#include "system_wrappers/include/aligned_malloc.h"
>+#include "rtc_base/memory/aligned_malloc.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/fir_filter_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/fir_filter_unittest.cc
>index 46966217d77..07abf2094fe 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/fir_filter_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/fir_filter_unittest.cc
>@@ -21,20 +21,18 @@ namespace webrtc {
> namespace {
> 
> static const float kCoefficients[] = {0.2f, 0.3f, 0.5f, 0.7f, 0.11f};
>-static const size_t kCoefficientsLength = sizeof(kCoefficients) /
>-                                       sizeof(kCoefficients[0]);
>+static const size_t kCoefficientsLength =
>+    sizeof(kCoefficients) / sizeof(kCoefficients[0]);
> 
>-static const float kInput[] = {1.f, 2.f, 3.f, 4.f, 5.f, 6.f, 7.f,
>-                                      8.f, 9.f, 10.f};
>-static const size_t kInputLength = sizeof(kInput) /
>-                                      sizeof(kInput[0]);
>+static const float kInput[] = {1.f, 2.f, 3.f, 4.f, 5.f,
>+                               6.f, 7.f, 8.f, 9.f, 10.f};
>+static const size_t kInputLength = sizeof(kInput) / sizeof(kInput[0]);
> 
> void VerifyOutput(const float* expected_output,
>                   const float* output,
>                   size_t length) {
>-  EXPECT_EQ(0, memcmp(expected_output,
>-                      output,
>-                      length * sizeof(expected_output[0])));
>+  EXPECT_EQ(
>+      0, memcmp(expected_output, output, length * sizeof(expected_output[0])));
> }
> 
> }  // namespace
>@@ -97,8 +95,8 @@ TEST(FIRFilterTest, FilterInLengthLesserOrEqualToCoefficientsLength) {
> 
>   EXPECT_FLOAT_EQ(0.2f, output[0]);
>   EXPECT_FLOAT_EQ(0.7f, output[1]);
>-  filter.reset(CreateFirFilter(
>-      kCoefficients, kCoefficientsLength, kCoefficientsLength));
>+  filter.reset(
>+      CreateFirFilter(kCoefficients, kCoefficientsLength, kCoefficientsLength));
>   filter->Filter(kInput, kCoefficientsLength, output);
> 
>   EXPECT_FLOAT_EQ(0.2f, output[0]);
>@@ -149,19 +147,17 @@ TEST(FIRFilterTest, VerifySampleBasedVsBlockBasedFiltering) {
>     filter->Filter(&kInput[i], 1, &output_sample_based[i]);
>   }
> 
>-  EXPECT_EQ(0, memcmp(output_sample_based,
>-                      output_block_based,
>-                      kInputLength));
>+  EXPECT_EQ(0, memcmp(output_sample_based, output_block_based, kInputLength));
> }
> 
> TEST(FIRFilterTest, SimplestHighPassFilter) {
>   const float kCoefficients[] = {1.f, -1.f};
>-  const size_t kCoefficientsLength = sizeof(kCoefficients) /
>-                                  sizeof(kCoefficients[0]);
>+  const size_t kCoefficientsLength =
>+      sizeof(kCoefficients) / sizeof(kCoefficients[0]);
> 
>   float kConstantInput[] = {1.f, 1.f, 1.f, 1.f, 1.f, 1.f, 1.f, 1.f};
>-  const size_t kConstantInputLength = sizeof(kConstantInput) /
>-      sizeof(kConstantInput[0]);
>+  const size_t kConstantInputLength =
>+      sizeof(kConstantInput) / sizeof(kConstantInput[0]);
> 
>   float output[kConstantInputLength];
>   std::unique_ptr<FIRFilter> filter(CreateFirFilter(
>@@ -175,12 +171,12 @@ TEST(FIRFilterTest, SimplestHighPassFilter) {
> 
> TEST(FIRFilterTest, SimplestLowPassFilter) {
>   const float kCoefficients[] = {1.f, 1.f};
>-  const size_t kCoefficientsLength = sizeof(kCoefficients) /
>-                                  sizeof(kCoefficients[0]);
>+  const size_t kCoefficientsLength =
>+      sizeof(kCoefficients) / sizeof(kCoefficients[0]);
> 
>   float kHighFrequencyInput[] = {-1.f, 1.f, -1.f, 1.f, -1.f, 1.f, -1.f, 1.f};
>-  const size_t kHighFrequencyInputLength = sizeof(kHighFrequencyInput) /
>-                                        sizeof(kHighFrequencyInput[0]);
>+  const size_t kHighFrequencyInputLength =
>+      sizeof(kHighFrequencyInput) / sizeof(kHighFrequencyInput[0]);
> 
>   float output[kHighFrequencyInputLength];
>   std::unique_ptr<FIRFilter> filter(CreateFirFilter(
>@@ -195,16 +191,16 @@ TEST(FIRFilterTest, SimplestLowPassFilter) {
> TEST(FIRFilterTest, SameOutputWhenSwapedCoefficientsAndInput) {
>   float output[kCoefficientsLength];
>   float output_swaped[kCoefficientsLength];
>-  std::unique_ptr<FIRFilter> filter(CreateFirFilter(
>-      kCoefficients, kCoefficientsLength, kCoefficientsLength));
>+  std::unique_ptr<FIRFilter> filter(
>+      CreateFirFilter(kCoefficients, kCoefficientsLength, kCoefficientsLength));
>   // Use kCoefficientsLength for in_length to get same-length outputs.
>   filter->Filter(kInput, kCoefficientsLength, output);
> 
>-  filter.reset(CreateFirFilter(
>-      kInput, kCoefficientsLength, kCoefficientsLength));
>+  filter.reset(
>+      CreateFirFilter(kInput, kCoefficientsLength, kCoefficientsLength));
>   filter->Filter(kCoefficients, kCoefficientsLength, output_swaped);
> 
>-  for (size_t i = 0 ; i < kCoefficientsLength; ++i) {
>+  for (size_t i = 0; i < kCoefficientsLength; ++i) {
>     EXPECT_FLOAT_EQ(output[i], output_swaped[i]);
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/include/audio_util.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/include/audio_util.h
>index b9e1b26b0db..de242a49be8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/include/audio_util.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/include/audio_util.h
>@@ -12,11 +12,11 @@
> #define COMMON_AUDIO_INCLUDE_AUDIO_UTIL_H_
> 
> #include <algorithm>
>-#include <limits>
>+#include <cmath>
> #include <cstring>
>+#include <limits>
> 
> #include "rtc_base/checks.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -26,6 +26,10 @@ typedef std::numeric_limits<int16_t> limits_int16;
> // S16:      int16_t [-32768, 32767]
> // Float:    float   [-1.0, 1.0]
> // FloatS16: float   [-32768.0, 32767.0]
>+// Dbfs: float [-20.0*log(10, 32768), 0] = [-90.3, 0]
>+// The ratio conversion functions use this naming convention:
>+// Ratio: float (0, +inf)
>+// Db: float (-inf, +inf)
> static inline int16_t FloatToS16(float v) {
>   if (v > 0)
>     return v >= 1 ? limits_int16::max()
>@@ -65,6 +69,27 @@ void FloatS16ToS16(const float* src, size_t size, int16_t* dest);
> void FloatToFloatS16(const float* src, size_t size, float* dest);
> void FloatS16ToFloat(const float* src, size_t size, float* dest);
> 
>+inline float DbToRatio(float v) {
>+  return std::pow(10.0f, v / 20.0f);
>+}
>+
>+inline float DbfsToFloatS16(float v) {
>+  static constexpr float kMaximumAbsFloatS16 = -limits_int16::min();
>+  return DbToRatio(v) * kMaximumAbsFloatS16;
>+}
>+
>+inline float FloatS16ToDbfs(float v) {
>+  RTC_DCHECK_GE(v, 0);
>+
>+  // kMinDbfs is equal to -20.0 * log10(-limits_int16::min())
>+  static constexpr float kMinDbfs = -90.30899869919436f;
>+  if (v <= 1.0f) {
>+    return kMinDbfs;
>+  }
>+  // Equal to 20 * log10(v / (-limits_int16::min()))
>+  return 20.0f * std::log10(v) + kMinDbfs;
>+}
>+
> // Copy audio from |src| channels to |dest| channels unless |src| and |dest|
> // point to the same address. |src| and |dest| must have the same number of
> // channels, and there must be sufficient space allocated in |dest|.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/lapped_transform.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/lapped_transform.cc
>index 517709f4622..72c2ad79f6f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/lapped_transform.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/lapped_transform.cc
>@@ -29,20 +29,17 @@ void LappedTransform::BlockThunk::ProcessBlock(const float* const* input,
>   RTC_CHECK_EQ(parent_->block_length_, num_frames);
> 
>   for (size_t i = 0; i < num_input_channels; ++i) {
>-    memcpy(parent_->real_buf_.Row(i), input[i],
>-           num_frames * sizeof(*input[0]));
>+    memcpy(parent_->real_buf_.Row(i), input[i], num_frames * sizeof(*input[0]));
>     parent_->fft_->Forward(parent_->real_buf_.Row(i),
>                            parent_->cplx_pre_.Row(i));
>   }
> 
>-  size_t block_length = RealFourier::ComplexLength(
>-      RealFourier::FftOrder(num_frames));
>+  size_t block_length =
>+      RealFourier::ComplexLength(RealFourier::FftOrder(num_frames));
>   RTC_CHECK_EQ(parent_->cplx_length_, block_length);
>-  parent_->block_processor_->ProcessAudioBlock(parent_->cplx_pre_.Array(),
>-                                               num_input_channels,
>-                                               parent_->cplx_length_,
>-                                               num_output_channels,
>-                                               parent_->cplx_post_.Array());
>+  parent_->block_processor_->ProcessAudioBlock(
>+      parent_->cplx_pre_.Array(), num_input_channels, parent_->cplx_length_,
>+      num_output_channels, parent_->cplx_post_.Array());
> 
>   for (size_t i = 0; i < num_output_channels; ++i) {
>     parent_->fft_->Inverse(parent_->cplx_post_.Row(i),
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/lapped_transform.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/lapped_transform.h
>index fe3a8cd8c88..1ab2a9fee75 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/lapped_transform.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/lapped_transform.h
>@@ -16,7 +16,7 @@
> 
> #include "common_audio/blocker.h"
> #include "common_audio/real_fourier.h"
>-#include "system_wrappers/include/aligned_array.h"
>+#include "rtc_base/memory/aligned_array.h"
> 
> namespace webrtc {
> 
>@@ -35,7 +35,8 @@ class LappedTransform {
>     virtual ~Callback() {}
> 
>     virtual void ProcessAudioBlock(const std::complex<float>* const* in_block,
>-                                   size_t num_in_channels, size_t frames,
>+                                   size_t num_in_channels,
>+                                   size_t frames,
>                                    size_t num_out_channels,
>                                    std::complex<float>* const* out_block) = 0;
>   };
>@@ -128,4 +129,3 @@ class LappedTransform {
> }  // namespace webrtc
> 
> #endif  // COMMON_AUDIO_LAPPED_TRANSFORM_H_
>-
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/lapped_transform_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/lapped_transform_unittest.cc
>index d6a312db517..687df8986a1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/lapped_transform_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/lapped_transform_unittest.cc
>@@ -24,11 +24,11 @@ class NoopCallback : public webrtc::LappedTransform::Callback {
>  public:
>   NoopCallback() : block_num_(0) {}
> 
>-  virtual void ProcessAudioBlock(const complex<float>* const* in_block,
>-                                 size_t in_channels,
>-                                 size_t frames,
>-                                 size_t out_channels,
>-                                 complex<float>* const* out_block) {
>+  void ProcessAudioBlock(const complex<float>* const* in_block,
>+                         size_t in_channels,
>+                         size_t frames,
>+                         size_t out_channels,
>+                         complex<float>* const* out_block) override {
>     RTC_CHECK_EQ(in_channels, out_channels);
>     for (size_t i = 0; i < out_channels; ++i) {
>       memcpy(out_block[i], in_block[i], sizeof(**in_block) * frames);
>@@ -36,9 +36,7 @@ class NoopCallback : public webrtc::LappedTransform::Callback {
>     ++block_num_;
>   }
> 
>-  size_t block_num() {
>-    return block_num_;
>-  }
>+  size_t block_num() { return block_num_; }
> 
>  private:
>   size_t block_num_;
>@@ -48,11 +46,11 @@ class FftCheckerCallback : public webrtc::LappedTransform::Callback {
>  public:
>   FftCheckerCallback() : block_num_(0) {}
> 
>-  virtual void ProcessAudioBlock(const complex<float>* const* in_block,
>-                                 size_t in_channels,
>-                                 size_t frames,
>-                                 size_t out_channels,
>-                                 complex<float>* const* out_block) {
>+  void ProcessAudioBlock(const complex<float>* const* in_block,
>+                         size_t in_channels,
>+                         size_t frames,
>+                         size_t out_channels,
>+                         complex<float>* const* out_block) override {
>     RTC_CHECK_EQ(in_channels, out_channels);
> 
>     size_t full_length = (frames - 1) * 2;
>@@ -69,9 +67,7 @@ class FftCheckerCallback : public webrtc::LappedTransform::Callback {
>     }
>   }
> 
>-  size_t block_num() {
>-    return block_num_;
>-  }
>+  size_t block_num() { return block_num_; }
> 
>  private:
>   size_t block_num_;
>@@ -150,8 +146,7 @@ TEST(LappedTransformTest, IdentityProcessor) {
>   trans.ProcessChunk(&in_chunk, &out_chunk);
> 
>   for (size_t i = 0; i < kChunkLength; ++i) {
>-    ASSERT_NEAR(out_chunk[i],
>-                (i < kBlockLength - kShiftAmount) ? 0.0f : 2.0f,
>+    ASSERT_NEAR(out_chunk[i], (i < kBlockLength - kShiftAmount) ? 0.0f : 2.0f,
>                 1e-5f);
>   }
> 
>@@ -167,8 +162,8 @@ TEST(LappedTransformTest, Callbacks) {
>   float window[kBlockLength];
>   std::fill(window, &window[kBlockLength], 1.0f);
> 
>-  LappedTransform trans(1, 1, kChunkLength, window, kBlockLength,
>-                        kBlockLength, &call);
>+  LappedTransform trans(1, 1, kChunkLength, window, kBlockLength, kBlockLength,
>+                        &call);
>   float in_buffer[kChunkLength];
>   float* in_chunk = in_buffer;
>   float out_buffer[kChunkLength];
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/mocks/mock_smoothing_filter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/mocks/mock_smoothing_filter.h
>index dec6ea5f470..712049fa6a3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/mocks/mock_smoothing_filter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/mocks/mock_smoothing_filter.h
>@@ -19,7 +19,7 @@ namespace webrtc {
> class MockSmoothingFilter : public SmoothingFilter {
>  public:
>   MOCK_METHOD1(AddSample, void(float));
>-  MOCK_METHOD0(GetAverage, rtc::Optional<float>());
>+  MOCK_METHOD0(GetAverage, absl::optional<float>());
>   MOCK_METHOD1(SetTimeConstantMs, bool(int));
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/real_fourier.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/real_fourier.cc
>index f759778a35a..7365844e8d0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/real_fourier.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/real_fourier.cc
>@@ -11,7 +11,6 @@
> #include "common_audio/real_fourier.h"
> 
> #include "common_audio/real_fourier_ooura.h"
>-#include "common_audio/real_fourier_openmax.h"
> #include "common_audio/signal_processing/include/signal_processing_library.h"
> #include "rtc_base/checks.h"
> 
>@@ -22,11 +21,7 @@ using std::complex;
> const size_t RealFourier::kFftBufferAlignment = 32;
> 
> std::unique_ptr<RealFourier> RealFourier::Create(int fft_order) {
>-#if defined(RTC_USE_OPENMAX_DL)
>-  return std::unique_ptr<RealFourier>(new RealFourierOpenmax(fft_order));
>-#else
>   return std::unique_ptr<RealFourier>(new RealFourierOoura(fft_order));
>-#endif
> }
> 
> int RealFourier::FftOrder(size_t length) {
>@@ -36,7 +31,7 @@ int RealFourier::FftOrder(size_t length) {
> 
> size_t RealFourier::FftLength(int order) {
>   RTC_CHECK_GE(order, 0);
>-  return static_cast<size_t>(1 << order);
>+  return size_t{1} << order;
> }
> 
> size_t RealFourier::ComplexLength(int order) {
>@@ -54,4 +49,3 @@ RealFourier::fft_cplx_scoper RealFourier::AllocCplxBuffer(int count) {
> }
> 
> }  // namespace webrtc
>-
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/real_fourier.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/real_fourier.h
>index 4c69c3c7685..a3e4dd16a82 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/real_fourier.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/real_fourier.h
>@@ -14,7 +14,7 @@
> #include <complex>
> #include <memory>
> 
>-#include "system_wrappers/include/aligned_malloc.h"
>+#include "rtc_base/memory/aligned_malloc.h"
> 
> // Uniform interface class for the real DFT and its inverse, for power-of-2
> // input lengths. Also contains helper functions for buffer allocation, taking
>@@ -72,4 +72,3 @@ class RealFourier {
> }  // namespace webrtc
> 
> #endif  // COMMON_AUDIO_REAL_FOURIER_H_
>-
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/real_fourier_ooura.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/real_fourier_ooura.cc
>index 5d75717bc75..89694c16678 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/real_fourier_ooura.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/real_fourier_ooura.cc
>@@ -10,10 +10,10 @@
> 
> #include "common_audio/real_fourier_ooura.h"
> 
>-#include <cmath>
> #include <algorithm>
>+#include <cmath>
> 
>-#include "common_audio/fft4g.h"
>+#include "common_audio/third_party/fft4g/fft4g.h"
> #include "rtc_base/checks.h"
> 
> namespace webrtc {
>@@ -28,8 +28,8 @@ void Conjugate(complex<float>* array, size_t complex_length) {
> }
> 
> size_t ComputeWorkIpSize(size_t fft_length) {
>-  return static_cast<size_t>(2 + std::ceil(std::sqrt(
>-      static_cast<float>(fft_length))));
>+  return static_cast<size_t>(
>+      2 + std::ceil(std::sqrt(static_cast<float>(fft_length))));
> }
> 
> }  // namespace
>@@ -45,11 +45,13 @@ RealFourierOoura::RealFourierOoura(int fft_order)
>   RTC_CHECK_GE(fft_order, 1);
> }
> 
>+RealFourierOoura::~RealFourierOoura() = default;
>+
> void RealFourierOoura::Forward(const float* src, complex<float>* dest) const {
>   {
>     // This cast is well-defined since C++11. See "Non-static data members" at:
>     // http://en.cppreference.com/w/cpp/numeric/complex
>-    auto dest_float = reinterpret_cast<float*>(dest);
>+    auto* dest_float = reinterpret_cast<float*>(dest);
>     std::copy(src, src + length_, dest_float);
>     WebRtc_rdft(length_, 1, dest_float, work_ip_.get(), work_w_.get());
>   }
>@@ -63,7 +65,7 @@ void RealFourierOoura::Forward(const float* src, complex<float>* dest) const {
> 
> void RealFourierOoura::Inverse(const complex<float>* src, float* dest) const {
>   {
>-    auto dest_complex = reinterpret_cast<complex<float>*>(dest);
>+    auto* dest_complex = reinterpret_cast<complex<float>*>(dest);
>     // The real output array is shorter than the input complex array by one
>     // complex element.
>     const size_t dest_complex_length = complex_length_ - 1;
>@@ -71,8 +73,8 @@ void RealFourierOoura::Inverse(const complex<float>* src, float* dest) const {
>     // Restore Ooura's conjugate definition.
>     Conjugate(dest_complex, dest_complex_length);
>     // Restore real[n/2] to imag[0].
>-    dest_complex[0] = complex<float>(dest_complex[0].real(),
>-                                     src[complex_length_ - 1].real());
>+    dest_complex[0] =
>+        complex<float>(dest_complex[0].real(), src[complex_length_ - 1].real());
>   }
> 
>   WebRtc_rdft(length_, -1, dest, work_ip_.get(), work_w_.get());
>@@ -82,4 +84,8 @@ void RealFourierOoura::Inverse(const complex<float>* src, float* dest) const {
>   std::for_each(dest, dest + length_, [scale](float& v) { v *= scale; });
> }
> 
>+int RealFourierOoura::order() const {
>+  return order_;
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/real_fourier_ooura.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/real_fourier_ooura.h
>index f885a34f58c..bb8eef96dff 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/real_fourier_ooura.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/real_fourier_ooura.h
>@@ -21,13 +21,12 @@ namespace webrtc {
> class RealFourierOoura : public RealFourier {
>  public:
>   explicit RealFourierOoura(int fft_order);
>+  ~RealFourierOoura() override;
> 
>   void Forward(const float* src, std::complex<float>* dest) const override;
>   void Inverse(const std::complex<float>* src, float* dest) const override;
> 
>-  int order() const override {
>-    return order_;
>-  }
>+  int order() const override;
> 
>  private:
>   const int order_;
>@@ -42,4 +41,3 @@ class RealFourierOoura : public RealFourier {
> }  // namespace webrtc
> 
> #endif  // COMMON_AUDIO_REAL_FOURIER_OOURA_H_
>-
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/real_fourier_openmax.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/real_fourier_openmax.cc
>deleted file mode 100644
>index 6c5c9cee928..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/real_fourier_openmax.cc
>+++ /dev/null
>@@ -1,69 +0,0 @@
>-/*
>- *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "common_audio/real_fourier_openmax.h"
>-
>-#include <cstdlib>
>-
>-#include "dl/sp/api/omxSP.h"
>-#include "rtc_base/checks.h"
>-
>-namespace webrtc {
>-
>-using std::complex;
>-
>-namespace {
>-
>-// Creates and initializes the Openmax state. Transfers ownership to caller.
>-OMXFFTSpec_R_F32* CreateOpenmaxState(int order) {
>-  RTC_CHECK_GE(order, 1);
>-  // The omx implementation uses this macro to check order validity.
>-  RTC_CHECK_LE(order, TWIDDLE_TABLE_ORDER);
>-
>-  OMX_INT buffer_size;
>-  OMXResult r = omxSP_FFTGetBufSize_R_F32(order, &buffer_size);
>-  RTC_CHECK_EQ(r, OMX_Sts_NoErr);
>-
>-  OMXFFTSpec_R_F32* omx_spec = malloc(buffer_size);
>-  RTC_DCHECK(omx_spec);
>-
>-  r = omxSP_FFTInit_R_F32(omx_spec, order);
>-  RTC_CHECK_EQ(r, OMX_Sts_NoErr);
>-  return omx_spec;
>-}
>-
>-}  // namespace
>-
>-RealFourierOpenmax::RealFourierOpenmax(int fft_order)
>-    : order_(fft_order),
>-      omx_spec_(CreateOpenmaxState(order_)) {
>-}
>-
>-RealFourierOpenmax::~RealFourierOpenmax() {
>-  free(omx_spec_);
>-}
>-
>-void RealFourierOpenmax::Forward(const float* src, complex<float>* dest) const {
>-  // This cast is well-defined since C++11. See "Non-static data members" at:
>-  // http://en.cppreference.com/w/cpp/numeric/complex
>-  OMXResult r =
>-      omxSP_FFTFwd_RToCCS_F32(src, reinterpret_cast<OMX_F32*>(dest), omx_spec_);
>-  RTC_CHECK_EQ(r, OMX_Sts_NoErr);
>-}
>-
>-void RealFourierOpenmax::Inverse(const complex<float>* src, float* dest) const {
>-  OMXResult r =
>-      omxSP_FFTInv_CCSToR_F32(reinterpret_cast<const OMX_F32*>(src), dest,
>-                              omx_spec_);
>-  RTC_CHECK_EQ(r, OMX_Sts_NoErr);
>-}
>-
>-}  // namespace webrtc
>-
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/real_fourier_openmax.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/real_fourier_openmax.h
>deleted file mode 100644
>index 7b4299aa3c9..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/real_fourier_openmax.h
>+++ /dev/null
>@@ -1,44 +0,0 @@
>-/*
>- *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef COMMON_AUDIO_REAL_FOURIER_OPENMAX_H_
>-#define COMMON_AUDIO_REAL_FOURIER_OPENMAX_H_
>-
>-#include <complex>
>-
>-#include "common_audio/real_fourier.h"
>-
>-namespace webrtc {
>-
>-class RealFourierOpenmax : public RealFourier {
>- public:
>-  explicit RealFourierOpenmax(int fft_order);
>-  ~RealFourierOpenmax() override;
>-
>-  void Forward(const float* src, std::complex<float>* dest) const override;
>-  void Inverse(const std::complex<float>* src, float* dest) const override;
>-
>-  int order() const override {
>-    return order_;
>-  }
>-
>- private:
>-  // Basically a forward declare of OMXFFTSpec_R_F32. To get rid of the
>-  // dependency on openmax.
>-  typedef void OMXFFTSpec_R_F32_;
>-  const int order_;
>-
>-  OMXFFTSpec_R_F32_* const omx_spec_;
>-};
>-
>-}  // namespace webrtc
>-
>-#endif  // COMMON_AUDIO_REAL_FOURIER_OPENMAX_H_
>-
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/real_fourier_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/real_fourier_unittest.cc
>index 97849b965b1..132488747ad 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/real_fourier_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/real_fourier_unittest.cc
>@@ -13,7 +13,6 @@
> #include <stdlib.h>
> 
> #include "common_audio/real_fourier_ooura.h"
>-#include "common_audio/real_fourier_openmax.h"
> #include "test/gtest.h"
> 
> namespace webrtc {
>@@ -61,19 +60,14 @@ class RealFourierTest : public ::testing::Test {
>         real_buffer_(RealFourier::AllocRealBuffer(4)),
>         cplx_buffer_(RealFourier::AllocCplxBuffer(3)) {}
> 
>-  ~RealFourierTest() {
>-  }
>+  ~RealFourierTest() {}
> 
>   T rf_;
>   const RealFourier::fft_real_scoper real_buffer_;
>   const RealFourier::fft_cplx_scoper cplx_buffer_;
> };
> 
>-using FftTypes = ::testing::Types<
>-#if defined(RTC_USE_OPENMAX_DL)
>-    RealFourierOpenmax,
>-#endif
>-    RealFourierOoura>;
>+using FftTypes = ::testing::Types<RealFourierOoura>;
> TYPED_TEST_CASE(RealFourierTest, FftTypes);
> 
> TYPED_TEST(RealFourierTest, SimpleForwardTransform) {
>@@ -106,4 +100,3 @@ TYPED_TEST(RealFourierTest, SimpleBackwardTransform) {
> }
> 
> }  // namespace webrtc
>-
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/include/push_resampler.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/include/push_resampler.h
>index 046415b87e8..082cdc62840 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/include/push_resampler.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/include/push_resampler.h
>@@ -13,8 +13,6 @@
> 
> #include <memory>
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>-
> namespace webrtc {
> 
> class PushSincResampler;
>@@ -29,7 +27,8 @@ class PushResampler {
> 
>   // Must be called whenever the parameters change. Free to be called at any
>   // time as it is a no-op if parameters have not changed since the last call.
>-  int InitializeIfNeeded(int src_sample_rate_hz, int dst_sample_rate_hz,
>+  int InitializeIfNeeded(int src_sample_rate_hz,
>+                         int dst_sample_rate_hz,
>                          size_t num_channels);
> 
>   // Returns the total number of samples provided in destination (e.g. 32 kHz,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/include/resampler.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/include/resampler.h
>index fec2c1a7bf2..04c487b331e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/include/resampler.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/include/resampler.h
>@@ -8,7 +8,6 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-
> /*
>  * A wrapper for resampling a numerous amount of sampling combinations.
>  */
>@@ -17,8 +16,7 @@
> #define COMMON_AUDIO_RESAMPLER_INCLUDE_RESAMPLER_H_
> 
> #include <stddef.h>
>-
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include <stdint.h>
> 
> namespace webrtc {
> 
>@@ -36,8 +34,11 @@ class Resampler {
>   int ResetIfNeeded(int inFreq, int outFreq, size_t num_channels);
> 
>   // Resample samplesIn to samplesOut.
>-  int Push(const int16_t* samplesIn, size_t lengthIn, int16_t* samplesOut,
>-           size_t maxLen, size_t& outLen);  // NOLINT: to avoid changing APIs
>+  int Push(const int16_t* samplesIn,
>+           size_t lengthIn,
>+           int16_t* samplesOut,
>+           size_t maxLen,
>+           size_t& outLen);  // NOLINT: to avoid changing APIs
> 
>  private:
>   enum ResamplerMode {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/push_resampler.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/push_resampler.cc
>index 3930624b2ad..cc24c4b3f65 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/push_resampler.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/push_resampler.cc
>@@ -25,7 +25,8 @@ namespace {
> // caused the compiler to generate code that threw off the linker.
> // TODO(tommi): Re-enable when we've figured out what the problem is.
> // http://crbug.com/615050
>-void CheckValidInitParams(int src_sample_rate_hz, int dst_sample_rate_hz,
>+void CheckValidInitParams(int src_sample_rate_hz,
>+                          int dst_sample_rate_hz,
>                           size_t num_channels) {
> // The below checks are temporarily disabled on WEBRTC_WIN due to problems
> // with clang debug builds.
>@@ -57,14 +58,10 @@ void CheckExpectedBufferSizes(size_t src_length,
> 
> template <typename T>
> PushResampler<T>::PushResampler()
>-    : src_sample_rate_hz_(0),
>-      dst_sample_rate_hz_(0),
>-      num_channels_(0) {
>-}
>+    : src_sample_rate_hz_(0), dst_sample_rate_hz_(0), num_channels_(0) {}
> 
> template <typename T>
>-PushResampler<T>::~PushResampler() {
>-}
>+PushResampler<T>::~PushResampler() {}
> 
> template <typename T>
> int PushResampler<T>::InitializeIfNeeded(int src_sample_rate_hz,
>@@ -92,22 +89,24 @@ int PushResampler<T>::InitializeIfNeeded(int src_sample_rate_hz,
>       static_cast<size_t>(src_sample_rate_hz / 100);
>   const size_t dst_size_10ms_mono =
>       static_cast<size_t>(dst_sample_rate_hz / 100);
>-  sinc_resampler_.reset(new PushSincResampler(src_size_10ms_mono,
>-                                              dst_size_10ms_mono));
>+  sinc_resampler_.reset(
>+      new PushSincResampler(src_size_10ms_mono, dst_size_10ms_mono));
>   if (num_channels_ == 2) {
>     src_left_.reset(new T[src_size_10ms_mono]);
>     src_right_.reset(new T[src_size_10ms_mono]);
>     dst_left_.reset(new T[dst_size_10ms_mono]);
>     dst_right_.reset(new T[dst_size_10ms_mono]);
>-    sinc_resampler_right_.reset(new PushSincResampler(src_size_10ms_mono,
>-                                                      dst_size_10ms_mono));
>+    sinc_resampler_right_.reset(
>+        new PushSincResampler(src_size_10ms_mono, dst_size_10ms_mono));
>   }
> 
>   return 0;
> }
> 
> template <typename T>
>-int PushResampler<T>::Resample(const T* src, size_t src_length, T* dst,
>+int PushResampler<T>::Resample(const T* src,
>+                               size_t src_length,
>+                               T* dst,
>                                size_t dst_capacity) {
>   CheckExpectedBufferSizes(src_length, dst_capacity, num_channels_,
>                            src_sample_rate_hz_, dst_sample_rate_hz_);
>@@ -124,9 +123,8 @@ int PushResampler<T>::Resample(const T* src, size_t src_length, T* dst,
>     T* deinterleaved[] = {src_left_.get(), src_right_.get()};
>     Deinterleave(src, src_length_mono, num_channels_, deinterleaved);
> 
>-    size_t dst_length_mono =
>-        sinc_resampler_->Resample(src_left_.get(), src_length_mono,
>-                                  dst_left_.get(), dst_capacity_mono);
>+    size_t dst_length_mono = sinc_resampler_->Resample(
>+        src_left_.get(), src_length_mono, dst_left_.get(), dst_capacity_mono);
>     sinc_resampler_right_->Resample(src_right_.get(), src_length_mono,
>                                     dst_right_.get(), dst_capacity_mono);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/push_sinc_resampler.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/push_sinc_resampler.cc
>index 14ab330cdfd..3bfead26e75 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/push_sinc_resampler.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/push_sinc_resampler.cc
>@@ -28,8 +28,7 @@ PushSincResampler::PushSincResampler(size_t source_frames,
>       first_pass_(true),
>       source_available_(0) {}
> 
>-PushSincResampler::~PushSincResampler() {
>-}
>+PushSincResampler::~PushSincResampler() {}
> 
> size_t PushSincResampler::Resample(const int16_t* source,
>                                    size_t source_length,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/push_sinc_resampler.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/push_sinc_resampler.h
>index bdc03a2c00f..1ffe73f79ba 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/push_sinc_resampler.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/push_sinc_resampler.h
>@@ -15,7 +15,6 @@
> 
> #include "common_audio/resampler/sinc_resampler.h"
> #include "rtc_base/constructormagic.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -36,8 +35,10 @@ class PushSincResampler : public SincResamplerCallback {
>   // at least as large as |destination_frames|. Returns the number of samples
>   // provided in destination (for convenience, since this will always be equal
>   // to |destination_frames|).
>-  size_t Resample(const int16_t* source, size_t source_frames,
>-                  int16_t* destination, size_t destination_capacity);
>+  size_t Resample(const int16_t* source,
>+                  size_t source_frames,
>+                  int16_t* destination,
>+                  size_t destination_capacity);
>   size_t Resample(const float* source,
>                   size_t source_frames,
>                   float* destination,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/push_sinc_resampler_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/push_sinc_resampler_unittest.cc
>index 3be7c0a3651..2f53a745ae2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/push_sinc_resampler_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/push_sinc_resampler_unittest.cc
>@@ -19,7 +19,6 @@
> #include "rtc_base/timeutils.h"
> #include "test/gmock.h"
> #include "test/gtest.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> namespace {
>@@ -36,14 +35,13 @@ T DBFS(T x) {
> }  // namespace
> 
> class PushSincResamplerTest : public ::testing::TestWithParam<
>-    ::testing::tuple<int, int, double, double>> {
>+                                  ::testing::tuple<int, int, double, double>> {
>  public:
>   PushSincResamplerTest()
>       : input_rate_(::testing::get<0>(GetParam())),
>         output_rate_(::testing::get<1>(GetParam())),
>         rms_error_(::testing::get<2>(GetParam())),
>-        low_freq_error_(::testing::get<3>(GetParam())) {
>-  }
>+        low_freq_error_(::testing::get<3>(GetParam())) {}
> 
>   ~PushSincResamplerTest() override {}
> 
>@@ -59,7 +57,7 @@ class PushSincResamplerTest : public ::testing::TestWithParam<
> 
> class ZeroSource : public SincResamplerCallback {
>  public:
>-  void Run(size_t frames, float* destination) {
>+  void Run(size_t frames, float* destination) override {
>     std::memset(destination, 0, sizeof(float) * frames);
>   }
> };
>@@ -82,8 +80,8 @@ void PushSincResamplerTest::ResampleBenchmarkTest(bool int_format) {
>     source_int[i] = static_cast<int16_t>(floor(32767 * source[i] + 0.5));
>   }
> 
>-  printf("Benchmarking %d iterations of %d Hz -> %d Hz:\n",
>-         kResampleIterations, input_rate_, output_rate_);
>+  printf("Benchmarking %d iterations of %d Hz -> %d Hz:\n", kResampleIterations,
>+         input_rate_, output_rate_);
>   const double io_ratio = input_rate_ / static_cast<double>(output_rate_);
>   SincResampler sinc_resampler(io_ratio, SincResampler::kDefaultRequestSize,
>                                &resampler_source);
>@@ -101,25 +99,23 @@ void PushSincResamplerTest::ResampleBenchmarkTest(bool int_format) {
>   if (int_format) {
>     for (int i = 0; i < kResampleIterations; ++i) {
>       EXPECT_EQ(output_samples,
>-                resampler.Resample(source_int.get(),
>-                                   input_samples,
>-                                   destination_int.get(),
>-                                   output_samples));
>+                resampler.Resample(source_int.get(), input_samples,
>+                                   destination_int.get(), output_samples));
>     }
>   } else {
>     for (int i = 0; i < kResampleIterations; ++i) {
>-      EXPECT_EQ(output_samples,
>-                resampler.Resample(source.get(),
>-                                   input_samples,
>-                                   resampled_destination.get(),
>-                                   output_samples));
>+      EXPECT_EQ(output_samples, resampler.Resample(source.get(), input_samples,
>+                                                   resampled_destination.get(),
>+                                                   output_samples));
>     }
>   }
>   double total_time_us =
>       (rtc::TimeNanos() - start) / rtc::kNumNanosecsPerMicrosec;
>-  printf("PushSincResampler took %.2f us per frame; which is a %.1f%% overhead "
>-         "on SincResampler.\n\n", total_time_us / kResampleIterations,
>-         (total_time_us - total_time_sinc_us) / total_time_sinc_us * 100);
>+  printf(
>+      "PushSincResampler took %.2f us per frame; which is a %.1f%% overhead "
>+      "on SincResampler.\n\n",
>+      total_time_us / kResampleIterations,
>+      (total_time_us - total_time_sinc_us) / total_time_sinc_us * 100);
> }
> 
> // Disabled because it takes too long to run routinely. Use for performance
>@@ -149,8 +145,8 @@ void PushSincResamplerTest::ResampleTest(bool int_format) {
>   const double input_nyquist_freq = 0.5 * input_rate_;
> 
>   // Source for data to be resampled.
>-  SinusoidalLinearChirpSource resampler_source(
>-      input_rate_, input_samples, input_nyquist_freq, 0);
>+  SinusoidalLinearChirpSource resampler_source(input_rate_, input_samples,
>+                                               input_nyquist_freq, 0);
> 
>   PushSincResampler resampler(input_block_size, output_block_size);
> 
>@@ -168,8 +164,8 @@ void PushSincResamplerTest::ResampleTest(bool int_format) {
>   // deal with it in the test by delaying the "pure" source to match. It must be
>   // checked before the first call to Resample(), because ChunkSize() will
>   // change afterwards.
>-  const size_t output_delay_samples = output_block_size -
>-      resampler.get_resampler_for_testing()->ChunkSize();
>+  const size_t output_delay_samples =
>+      output_block_size - resampler.get_resampler_for_testing()->ChunkSize();
> 
>   // Generate resampled signal.
>   // With the PushSincResampler, we produce the signal block-by-10ms-block
>@@ -178,21 +174,18 @@ void PushSincResamplerTest::ResampleTest(bool int_format) {
>   if (int_format) {
>     for (size_t i = 0; i < kNumBlocks; ++i) {
>       FloatToS16(&source[i * input_block_size], input_block_size,
>-               source_int.get());
>+                 source_int.get());
>       EXPECT_EQ(output_block_size,
>-                resampler.Resample(source_int.get(),
>-                                   input_block_size,
>-                                   destination_int.get(),
>-                                   output_block_size));
>+                resampler.Resample(source_int.get(), input_block_size,
>+                                   destination_int.get(), output_block_size));
>       S16ToFloat(destination_int.get(), output_block_size,
>-               &resampled_destination[i * output_block_size]);
>+                 &resampled_destination[i * output_block_size]);
>     }
>   } else {
>     for (size_t i = 0; i < kNumBlocks; ++i) {
>       EXPECT_EQ(
>           output_block_size,
>-          resampler.Resample(&source[i * input_block_size],
>-                             input_block_size,
>+          resampler.Resample(&source[i * input_block_size], input_block_size,
>                              &resampled_destination[i * output_block_size],
>                              output_block_size));
>     }
>@@ -252,9 +245,13 @@ void PushSincResamplerTest::ResampleTest(bool int_format) {
>   EXPECT_LE(high_freq_max_error, kHighFrequencyMaxError);
> }
> 
>-TEST_P(PushSincResamplerTest, ResampleInt) { ResampleTest(true); }
>+TEST_P(PushSincResamplerTest, ResampleInt) {
>+  ResampleTest(true);
>+}
> 
>-TEST_P(PushSincResamplerTest, ResampleFloat) { ResampleTest(false); }
>+TEST_P(PushSincResamplerTest, ResampleFloat) {
>+  ResampleTest(false);
>+}
> 
> // Thresholds chosen arbitrarily based on what each resampling reported during
> // testing.  All thresholds are in dbFS, http://en.wikipedia.org/wiki/DBFS.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/resampler.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/resampler.cc
>index ea85d827e0d..aa3a4bac2aa 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/resampler.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/resampler.cc
>@@ -8,7 +8,6 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-
> /*
>  * A wrapper for resampling a numerous amount of sampling combinations.
>  */
>@@ -37,8 +36,7 @@ Resampler::Resampler()
>       my_mode_(kResamplerMode1To1),
>       num_channels_(0),
>       slave_left_(nullptr),
>-      slave_right_(nullptr) {
>-}
>+      slave_right_(nullptr) {}
> 
> Resampler::Resampler(int inFreq, int outFreq, size_t num_channels)
>     : Resampler() {
>@@ -73,9 +71,9 @@ int Resampler::ResetIfNeeded(int inFreq, int outFreq, size_t num_channels) {
>   int tmpInFreq_kHz = inFreq / 1000;
>   int tmpOutFreq_kHz = outFreq / 1000;
> 
>-  if ((tmpInFreq_kHz != my_in_frequency_khz_)
>-      || (tmpOutFreq_kHz != my_out_frequency_khz_)
>-      || (num_channels != num_channels_)) {
>+  if ((tmpInFreq_kHz != my_in_frequency_khz_) ||
>+      (tmpOutFreq_kHz != my_out_frequency_khz_) ||
>+      (num_channels != num_channels_)) {
>     return Reset(inFreq, outFreq, num_channels);
>   } else {
>     return 0;
>@@ -191,7 +189,7 @@ int Resampler::Reset(int inFreq, int outFreq, size_t num_channels) {
>       // 2:6
>       state1_ = malloc(sizeof(WebRtcSpl_State16khzTo48khz));
>       WebRtcSpl_ResetResample16khzTo48khz(
>-        static_cast<WebRtcSpl_State16khzTo48khz*>(state1_));
>+          static_cast<WebRtcSpl_State16khzTo48khz*>(state1_));
>       // 6:3
>       state2_ = malloc(8 * sizeof(int32_t));
>       memset(state2_, 0, 8 * sizeof(int32_t));
>@@ -395,8 +393,11 @@ int Resampler::ComputeResamplerMode(int in_freq_hz,
> }
> 
> // Synchronous resampling, all output samples are written to samplesOut
>-int Resampler::Push(const int16_t * samplesIn, size_t lengthIn,
>-                    int16_t* samplesOut, size_t maxLen, size_t& outLen) {
>+int Resampler::Push(const int16_t* samplesIn,
>+                    size_t lengthIn,
>+                    int16_t* samplesOut,
>+                    size_t maxLen,
>+                    size_t& outLen) {
>   if (num_channels_ == 2) {
>     // Split up the signal and call the slave object for each channel
>     int16_t* left =
>@@ -575,7 +576,7 @@ int Resampler::Push(const int16_t * samplesIn, size_t lengthIn,
>       if ((lengthIn % 160) != 0) {
>         return -1;
>       }
>-      tmp = static_cast<int16_t*> (malloc(sizeof(int16_t) * lengthIn * 3));
>+      tmp = static_cast<int16_t*>(malloc(sizeof(int16_t) * lengthIn * 3));
>       tmp_mem = static_cast<int32_t*>(malloc(336 * sizeof(int32_t)));
>       for (size_t i = 0; i < lengthIn; i += 160) {
>         WebRtcSpl_Resample16khzTo48khz(
>@@ -827,7 +828,7 @@ int Resampler::Push(const int16_t * samplesIn, size_t lengthIn,
>         return -1;
>       }
>       // 3:6
>-      tmp = static_cast<int16_t*> (malloc(sizeof(int16_t) * lengthIn * 2));
>+      tmp = static_cast<int16_t*>(malloc(sizeof(int16_t) * lengthIn * 2));
>       WebRtcSpl_UpsampleBy2(samplesIn, lengthIn, tmp,
>                             static_cast<int32_t*>(state1_));
>       lengthIn *= 2;
>@@ -858,8 +859,8 @@ int Resampler::Push(const int16_t * samplesIn, size_t lengthIn,
>         return -1;
>       }
>       tmp_mem = static_cast<int32_t*>(malloc(126 * sizeof(int32_t)));
>-      tmp = static_cast<int16_t*>(
>-          malloc((lengthIn * 4) / 11 * sizeof(int16_t)));
>+      tmp =
>+          static_cast<int16_t*>(malloc((lengthIn * 4) / 11 * sizeof(int16_t)));
> 
>       for (size_t i = 0; i < lengthIn; i += 220) {
>         WebRtcSpl_Resample22khzTo8khz(
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/resampler_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/resampler_unittest.cc
>index 03007196374..fd636e29980 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/resampler_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/resampler_unittest.cc
>@@ -23,17 +23,10 @@ const size_t kNumChannelsSize = sizeof(kNumChannels) / sizeof(*kNumChannels);
> 
> // Rates we must support.
> const int kMaxRate = 96000;
>-const int kRates[] = {
>-  8000,
>-  16000,
>-  32000,
>-  44000,
>-  48000,
>-  kMaxRate
>-};
>+const int kRates[] = {8000, 16000, 32000, 44000, 48000, kMaxRate};
> const size_t kRatesSize = sizeof(kRates) / sizeof(*kRates);
> const int kMaxChannels = 2;
>-const size_t kDataSize = static_cast<size_t> (kMaxChannels * kMaxRate / 100);
>+const size_t kDataSize = static_cast<size_t>(kMaxChannels * kMaxRate / 100);
> 
> // TODO(andrew): should we be supporting these combinations?
> bool ValidRates(int in_rate, int out_rate) {
>@@ -49,8 +42,8 @@ bool ValidRates(int in_rate, int out_rate) {
> class ResamplerTest : public testing::Test {
>  protected:
>   ResamplerTest();
>-  virtual void SetUp();
>-  virtual void TearDown();
>+  void SetUp() override;
>+  void TearDown() override;
> 
>   void ResetIfNeededAndPush(int in_rate, int out_rate, int num_channels);
> 
>@@ -99,7 +92,7 @@ TEST_F(ResamplerTest, Reset) {
>       for (size_t k = 0; k < kNumChannelsSize; ++k) {
>         std::ostringstream ss;
>         ss << "Input rate: " << kRates[i] << ", output rate: " << kRates[j]
>-            << ", channels: " << kNumChannels[k];
>+           << ", channels: " << kNumChannels[k];
>         SCOPED_TRACE(ss.str());
>         if (ValidRates(kRates[i], kRates[j]))
>           EXPECT_EQ(0, rs_.Reset(kRates[i], kRates[j], kNumChannels[k]));
>@@ -124,8 +117,8 @@ TEST_F(ResamplerTest, Mono) {
>         size_t in_length = static_cast<size_t>(kRates[i] / 100);
>         size_t out_length = 0;
>         EXPECT_EQ(0, rs_.Reset(kRates[i], kRates[j], kChannels));
>-        EXPECT_EQ(0, rs_.Push(data_in_, in_length, data_out_, kDataSize,
>-                              out_length));
>+        EXPECT_EQ(
>+            0, rs_.Push(data_in_, in_length, data_out_, kDataSize, out_length));
>         EXPECT_EQ(static_cast<size_t>(kRates[j] / 100), out_length);
>       } else {
>         EXPECT_EQ(-1, rs_.Reset(kRates[i], kRates[j], kChannels));
>@@ -145,14 +138,12 @@ TEST_F(ResamplerTest, Stereo) {
>       if (ValidRates(kRates[i], kRates[j])) {
>         size_t in_length = static_cast<size_t>(kChannels * kRates[i] / 100);
>         size_t out_length = 0;
>-        EXPECT_EQ(0, rs_.Reset(kRates[i], kRates[j],
>-                               kChannels));
>-        EXPECT_EQ(0, rs_.Push(data_in_, in_length, data_out_, kDataSize,
>-                              out_length));
>+        EXPECT_EQ(0, rs_.Reset(kRates[i], kRates[j], kChannels));
>+        EXPECT_EQ(
>+            0, rs_.Push(data_in_, in_length, data_out_, kDataSize, out_length));
>         EXPECT_EQ(static_cast<size_t>(kChannels * kRates[j] / 100), out_length);
>       } else {
>-        EXPECT_EQ(-1, rs_.Reset(kRates[i], kRates[j],
>-                                kChannels));
>+        EXPECT_EQ(-1, rs_.Reset(kRates[i], kRates[j], kChannels));
>       }
>     }
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/sinc_resampler.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/sinc_resampler.cc
>index c8577550c95..5aa2061ab75 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/sinc_resampler.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/sinc_resampler.cc
>@@ -93,8 +93,8 @@
> #include <limits>
> 
> #include "rtc_base/checks.h"
>+#include "rtc_base/system/arch.h"
> #include "system_wrappers/include/cpu_features_wrapper.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -160,12 +160,12 @@ SincResampler::SincResampler(double io_sample_rate_ratio,
>           AlignedMalloc(sizeof(float) * kKernelStorageSize, 16))),
>       input_buffer_(static_cast<float*>(
>           AlignedMalloc(sizeof(float) * input_buffer_size_, 16))),
>-#if defined(WEBRTC_CPU_DETECTION)
>+#if defined(WEBRTC_ARCH_X86_FAMILY) && !defined(__SSE2__)
>       convolve_proc_(nullptr),
> #endif
>       r1_(input_buffer_.get()),
>       r2_(input_buffer_.get() + kKernelSize / 2) {
>-#if defined(WEBRTC_CPU_DETECTION)
>+#if defined(WEBRTC_ARCH_X86_FAMILY) && !defined(__SSE2__)
>   InitializeCPUSpecificFeatures();
>   RTC_DCHECK(convolve_proc_);
> #endif
>@@ -217,23 +217,23 @@ void SincResampler::InitializeKernel() {
> 
>     for (size_t i = 0; i < kKernelSize; ++i) {
>       const size_t idx = i + offset_idx * kKernelSize;
>-      const float pre_sinc = static_cast<float>(M_PI *
>-          (static_cast<int>(i) - static_cast<int>(kKernelSize / 2) -
>-           subsample_offset));
>+      const float pre_sinc = static_cast<float>(
>+          M_PI * (static_cast<int>(i) - static_cast<int>(kKernelSize / 2) -
>+                  subsample_offset));
>       kernel_pre_sinc_storage_[idx] = pre_sinc;
> 
>       // Compute Blackman window, matching the offset of the sinc().
>       const float x = (i - subsample_offset) / kKernelSize;
>       const float window = static_cast<float>(kA0 - kA1 * cos(2.0 * M_PI * x) +
>-          kA2 * cos(4.0 * M_PI * x));
>+                                              kA2 * cos(4.0 * M_PI * x));
>       kernel_window_storage_[idx] = window;
> 
>       // Compute the sinc with offset, then window the sinc() function and store
>       // at the correct offset.
>-      kernel_storage_[idx] = static_cast<float>(window *
>-          ((pre_sinc == 0) ?
>-              sinc_scale_factor :
>-              (sin(sinc_scale_factor * pre_sinc) / pre_sinc)));
>+      kernel_storage_[idx] = static_cast<float>(
>+          window * ((pre_sinc == 0)
>+                        ? sinc_scale_factor
>+                        : (sin(sinc_scale_factor * pre_sinc) / pre_sinc)));
>     }
>   }
> }
>@@ -255,10 +255,10 @@ void SincResampler::SetRatio(double io_sample_rate_ratio) {
>       const float window = kernel_window_storage_[idx];
>       const float pre_sinc = kernel_pre_sinc_storage_[idx];
> 
>-      kernel_storage_[idx] = static_cast<float>(window *
>-          ((pre_sinc == 0) ?
>-              sinc_scale_factor :
>-              (sin(sinc_scale_factor * pre_sinc) / pre_sinc)));
>+      kernel_storage_[idx] = static_cast<float>(
>+          window * ((pre_sinc == 0)
>+                        ? sinc_scale_factor
>+                        : (sin(sinc_scale_factor * pre_sinc) / pre_sinc)));
>     }
>   }
> }
>@@ -312,8 +312,8 @@ void SincResampler::Resample(size_t frames, float* destination) {
>       // Figure out how much to weight each kernel's "convolution".
>       const double kernel_interpolation_factor =
>           virtual_offset_idx - offset_idx;
>-      *destination++ = CONVOLVE_FUNC(
>-          input_ptr, k1, k2, kernel_interpolation_factor);
>+      *destination++ =
>+          CONVOLVE_FUNC(input_ptr, k1, k2, kernel_interpolation_factor);
> 
>       // Advance the virtual index.
>       virtual_source_idx_ += current_io_ratio;
>@@ -352,7 +352,8 @@ void SincResampler::Flush() {
>   UpdateRegions(false);
> }
> 
>-float SincResampler::Convolve_C(const float* input_ptr, const float* k1,
>+float SincResampler::Convolve_C(const float* input_ptr,
>+                                const float* k1,
>                                 const float* k2,
>                                 double kernel_interpolation_factor) {
>   float sum1 = 0;
>@@ -368,7 +369,7 @@ float SincResampler::Convolve_C(const float* input_ptr, const float* k1,
> 
>   // Linearly interpolate the two "convolutions".
>   return static_cast<float>((1.0 - kernel_interpolation_factor) * sum1 +
>-      kernel_interpolation_factor * sum2);
>+                            kernel_interpolation_factor * sum2);
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/sinc_resampler.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/sinc_resampler.h
>index d0112240554..8a833ce88c9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/sinc_resampler.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/sinc_resampler.h
>@@ -18,8 +18,8 @@
> 
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/gtest_prod_util.h"
>-#include "system_wrappers/include/aligned_malloc.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include "rtc_base/memory/aligned_malloc.h"
>+#include "rtc_base/system/arch.h"
> 
> namespace webrtc {
> 
>@@ -101,14 +101,18 @@ class SincResampler {
>   // Compute convolution of |k1| and |k2| over |input_ptr|, resultant sums are
>   // linearly interpolated using |kernel_interpolation_factor|.  On x86 and ARM
>   // the underlying implementation is chosen at run time.
>-  static float Convolve_C(const float* input_ptr, const float* k1,
>-                          const float* k2, double kernel_interpolation_factor);
>+  static float Convolve_C(const float* input_ptr,
>+                          const float* k1,
>+                          const float* k2,
>+                          double kernel_interpolation_factor);
> #if defined(WEBRTC_ARCH_X86_FAMILY)
>-  static float Convolve_SSE(const float* input_ptr, const float* k1,
>+  static float Convolve_SSE(const float* input_ptr,
>+                            const float* k1,
>                             const float* k2,
>                             double kernel_interpolation_factor);
> #elif defined(WEBRTC_HAS_NEON)
>-  static float Convolve_NEON(const float* input_ptr, const float* k1,
>+  static float Convolve_NEON(const float* input_ptr,
>+                             const float* k1,
>                              const float* k2,
>                              double kernel_interpolation_factor);
> #endif
>@@ -145,12 +149,14 @@ class SincResampler {
>   // Data from the source is copied into this buffer for each processing pass.
>   std::unique_ptr<float[], AlignedFreeDeleter> input_buffer_;
> 
>-  // Stores the runtime selection of which Convolve function to use.
>-  // TODO(ajm): Move to using a global static which must only be initialized
>-  // once by the user. We're not doing this initially, because we don't have
>-  // e.g. a LazyInstance helper in webrtc.
>-#if defined(WEBRTC_CPU_DETECTION)
>-  typedef float (*ConvolveProc)(const float*, const float*, const float*,
>+// Stores the runtime selection of which Convolve function to use.
>+// TODO(ajm): Move to using a global static which must only be initialized
>+// once by the user. We're not doing this initially, because we don't have
>+// e.g. a LazyInstance helper in webrtc.
>+#if defined(WEBRTC_ARCH_X86_FAMILY) && !defined(__SSE2__)
>+  typedef float (*ConvolveProc)(const float*,
>+                                const float*,
>+                                const float*,
>                                 double);
>   ConvolveProc convolve_proc_;
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/sinc_resampler_neon.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/sinc_resampler_neon.cc
>index 9d77f0d12c8..3649324d126 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/sinc_resampler_neon.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/sinc_resampler_neon.cc
>@@ -17,7 +17,8 @@
> 
> namespace webrtc {
> 
>-float SincResampler::Convolve_NEON(const float* input_ptr, const float* k1,
>+float SincResampler::Convolve_NEON(const float* input_ptr,
>+                                   const float* k1,
>                                    const float* k2,
>                                    double kernel_interpolation_factor) {
>   float32x4_t m_input;
>@@ -25,7 +26,7 @@ float SincResampler::Convolve_NEON(const float* input_ptr, const float* k1,
>   float32x4_t m_sums2 = vmovq_n_f32(0);
> 
>   const float* upper = input_ptr + kKernelSize;
>-  for (; input_ptr < upper; ) {
>+  for (; input_ptr < upper;) {
>     m_input = vld1q_f32(input_ptr);
>     input_ptr += 4;
>     m_sums1 = vmlaq_f32(m_sums1, m_input, vld1q_f32(k1));
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/sinc_resampler_sse.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/sinc_resampler_sse.cc
>index 711110858f6..3906a797528 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/sinc_resampler_sse.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/sinc_resampler_sse.cc
>@@ -17,7 +17,8 @@
> 
> namespace webrtc {
> 
>-float SincResampler::Convolve_SSE(const float* input_ptr, const float* k1,
>+float SincResampler::Convolve_SSE(const float* input_ptr,
>+                                  const float* k1,
>                                   const float* k2,
>                                   double kernel_interpolation_factor) {
>   __m128 m_input;
>@@ -41,17 +42,18 @@ float SincResampler::Convolve_SSE(const float* input_ptr, const float* k1,
>   }
> 
>   // Linearly interpolate the two "convolutions".
>-  m_sums1 = _mm_mul_ps(m_sums1, _mm_set_ps1(
>-      static_cast<float>(1.0 - kernel_interpolation_factor)));
>-  m_sums2 = _mm_mul_ps(m_sums2, _mm_set_ps1(
>-      static_cast<float>(kernel_interpolation_factor)));
>+  m_sums1 = _mm_mul_ps(
>+      m_sums1,
>+      _mm_set_ps1(static_cast<float>(1.0 - kernel_interpolation_factor)));
>+  m_sums2 = _mm_mul_ps(
>+      m_sums2, _mm_set_ps1(static_cast<float>(kernel_interpolation_factor)));
>   m_sums1 = _mm_add_ps(m_sums1, m_sums2);
> 
>   // Sum components together.
>   float result;
>   m_sums2 = _mm_add_ps(_mm_movehl_ps(m_sums1, m_sums1), m_sums1);
>-  _mm_store_ss(&result, _mm_add_ss(m_sums2, _mm_shuffle_ps(
>-      m_sums2, m_sums2, 1)));
>+  _mm_store_ss(&result,
>+               _mm_add_ss(m_sums2, _mm_shuffle_ps(m_sums2, m_sums2, 1)));
> 
>   return result;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/sinc_resampler_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/sinc_resampler_unittest.cc
>index 87e991d3b6a..eb5424d54e2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/sinc_resampler_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/sinc_resampler_unittest.cc
>@@ -23,6 +23,7 @@
> #include "common_audio/resampler/sinc_resampler.h"
> #include "common_audio/resampler/sinusoidal_linear_chirp_source.h"
> #include "rtc_base/stringize_macros.h"
>+#include "rtc_base/system/arch.h"
> #include "rtc_base/timeutils.h"
> #include "system_wrappers/include/cpu_features_wrapper.h"
> #include "test/gmock.h"
>@@ -67,14 +68,14 @@ TEST(SincResamplerTest, ChunkedResample) {
>   std::unique_ptr<float[]> resampled_destination(new float[max_chunk_size]);
> 
>   // Verify requesting ChunkSize() frames causes a single callback.
>-  EXPECT_CALL(mock_source, Run(_, _))
>-      .Times(1).WillOnce(ClearBuffer());
>+  EXPECT_CALL(mock_source, Run(_, _)).Times(1).WillOnce(ClearBuffer());
>   resampler.Resample(resampler.ChunkSize(), resampled_destination.get());
> 
>   // Verify requesting kChunks * ChunkSize() frames causes kChunks callbacks.
>   testing::Mock::VerifyAndClear(&mock_source);
>   EXPECT_CALL(mock_source, Run(_, _))
>-      .Times(kChunks).WillRepeatedly(ClearBuffer());
>+      .Times(kChunks)
>+      .WillRepeatedly(ClearBuffer());
>   resampler.Resample(max_chunk_size, resampled_destination.get());
> }
> 
>@@ -87,16 +88,14 @@ TEST(SincResamplerTest, Flush) {
>       new float[resampler.ChunkSize()]);
> 
>   // Fill the resampler with junk data.
>-  EXPECT_CALL(mock_source, Run(_, _))
>-      .Times(1).WillOnce(FillBuffer());
>+  EXPECT_CALL(mock_source, Run(_, _)).Times(1).WillOnce(FillBuffer());
>   resampler.Resample(resampler.ChunkSize() / 2, resampled_destination.get());
>   ASSERT_NE(resampled_destination[0], 0);
> 
>   // Flush and request more data, which should all be zeros now.
>   resampler.Flush();
>   testing::Mock::VerifyAndClear(&mock_source);
>-  EXPECT_CALL(mock_source, Run(_, _))
>-      .Times(1).WillOnce(ClearBuffer());
>+  EXPECT_CALL(mock_source, Run(_, _)).Times(1).WillOnce(ClearBuffer());
>   resampler.Resample(resampler.ChunkSize() / 2, resampled_destination.get());
>   for (size_t i = 0; i < resampler.ChunkSize() / 2; ++i)
>     ASSERT_FLOAT_EQ(resampled_destination[i], 0);
>@@ -116,7 +115,6 @@ TEST(SincResamplerTest, DISABLED_SetRatioBench) {
>   printf("SetRatio() took %.2fms.\n", total_time_c_us / 1000);
> }
> 
>-
> // Define platform independent function name for Convolve* tests.
> #if defined(WEBRTC_ARCH_X86_FAMILY)
> #define CONVOLVE_FUNC Convolve_SSE
>@@ -232,8 +230,7 @@ TEST(SincResamplerTest, ConvolveBenchmark) {
> #undef CONVOLVE_FUNC
> 
> typedef std::tuple<int, int, double, double> SincResamplerTestData;
>-class SincResamplerTest
>-    : public testing::TestWithParam<SincResamplerTestData> {
>+class SincResamplerTest : public testing::TestWithParam<SincResamplerTestData> {
>  public:
>   SincResamplerTest()
>       : input_rate_(std::get<0>(GetParam())),
>@@ -263,8 +260,8 @@ TEST_P(SincResamplerTest, Resample) {
>   const double input_nyquist_freq = 0.5 * input_rate_;
> 
>   // Source for data to be resampled.
>-  SinusoidalLinearChirpSource resampler_source(
>-      input_rate_, input_samples, input_nyquist_freq, 0);
>+  SinusoidalLinearChirpSource resampler_source(input_rate_, input_samples,
>+                                               input_nyquist_freq, 0);
> 
>   const double io_ratio = input_rate_ / static_cast<double>(output_rate_);
>   SincResampler resampler(io_ratio, SincResampler::kDefaultRequestSize,
>@@ -291,8 +288,8 @@ TEST_P(SincResamplerTest, Resample) {
>   resampler.Resample(output_samples, resampled_destination.get());
> 
>   // Generate pure signal.
>-  SinusoidalLinearChirpSource pure_source(
>-      output_rate_, output_samples, input_nyquist_freq, 0);
>+  SinusoidalLinearChirpSource pure_source(output_rate_, output_samples,
>+                                          input_nyquist_freq, 0);
>   pure_source.Run(output_samples, pure_destination.get());
> 
>   // Range of the Nyquist frequency (0.5 * min(input rate, output_rate)) which
>@@ -324,8 +321,8 @@ TEST_P(SincResamplerTest, Resample) {
> 
>   double rms_error = sqrt(sum_of_squares / output_samples);
> 
>-  // Convert each error to dbFS.
>-  #define DBFS(x) 20 * log10(x)
>+// Convert each error to dbFS.
>+#define DBFS(x) 20 * log10(x)
>   rms_error = DBFS(rms_error);
>   low_freq_max_error = DBFS(low_freq_max_error);
>   high_freq_max_error = DBFS(high_freq_max_error);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/sinusoidal_linear_chirp_source.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/sinusoidal_linear_chirp_source.cc
>index 134044e7786..2afdd1be476 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/sinusoidal_linear_chirp_source.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/sinusoidal_linear_chirp_source.cc
>@@ -43,8 +43,7 @@ void SinusoidalLinearChirpSource::Run(size_t frames, float* destination) {
>       } else {
>         // Sinusoidal linear chirp.
>         double t = (current_index_ - delay_samples_) / sample_rate_;
>-        destination[i] =
>-            sin(2 * M_PI * (kMinFrequency * t + (k_ / 2) * t * t));
>+        destination[i] = sin(2 * M_PI * (kMinFrequency * t + (k_ / 2) * t * t));
>       }
>     }
>   }
>@@ -52,7 +51,7 @@ void SinusoidalLinearChirpSource::Run(size_t frames, float* destination) {
> 
> double SinusoidalLinearChirpSource::Frequency(size_t position) {
>   return kMinFrequency + (position - delay_samples_) *
>-      (max_frequency_ - kMinFrequency) / total_samples_;
>+                             (max_frequency_ - kMinFrequency) / total_samples_;
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/sinusoidal_linear_chirp_source.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/sinusoidal_linear_chirp_source.h
>index 7fcbaa08214..71dcff5eb96 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/sinusoidal_linear_chirp_source.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/resampler/sinusoidal_linear_chirp_source.h
>@@ -26,19 +26,19 @@ class SinusoidalLinearChirpSource : public SincResamplerCallback {
>  public:
>   // |delay_samples| can be used to insert a fractional sample delay into the
>   // source.  It will produce zeros until non-negative time is reached.
>-  SinusoidalLinearChirpSource(int sample_rate, size_t samples,
>-                              double max_frequency, double delay_samples);
>+  SinusoidalLinearChirpSource(int sample_rate,
>+                              size_t samples,
>+                              double max_frequency,
>+                              double delay_samples);
> 
>-  virtual ~SinusoidalLinearChirpSource() {}
>+  ~SinusoidalLinearChirpSource() override {}
> 
>   void Run(size_t frames, float* destination) override;
> 
>   double Frequency(size_t position);
> 
>  private:
>-  enum {
>-    kMinFrequency = 5
>-  };
>+  enum { kMinFrequency = 5 };
> 
>   int sample_rate_;
>   size_t total_samples_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/ring_buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/ring_buffer.h
>index aa2ac27375b..0bbe879980d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/ring_buffer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/ring_buffer.h
>@@ -53,7 +53,8 @@ size_t WebRtc_ReadBuffer(RingBuffer* handle,
>                          size_t element_count);
> 
> // Writes |data| to buffer and returns the number of elements written.
>-size_t WebRtc_WriteBuffer(RingBuffer* handle, const void* data,
>+size_t WebRtc_WriteBuffer(RingBuffer* handle,
>+                          const void* data,
>                           size_t element_count);
> 
> // Moves the buffer read position and returns the number of elements moved.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/ring_buffer_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/ring_buffer_unittest.cc
>index 4bb1497b17b..130e1246733 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/ring_buffer_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/ring_buffer_unittest.cc
>@@ -21,9 +21,7 @@
> namespace webrtc {
> 
> struct FreeBufferDeleter {
>-  inline void operator()(void* ptr) const {
>-    WebRtc_FreeBuffer(ptr);
>-  }
>+  inline void operator()(void* ptr) const { WebRtc_FreeBuffer(ptr); }
> };
> typedef std::unique_ptr<RingBuffer, FreeBufferDeleter> scoped_ring_buffer;
> 
>@@ -31,7 +29,8 @@ static void AssertElementEq(int expected, int actual) {
>   ASSERT_EQ(expected, actual);
> }
> 
>-static int SetIncrementingData(int* data, int num_elements,
>+static int SetIncrementingData(int* data,
>+                               int num_elements,
>                                int starting_value) {
>   for (int i = 0; i < num_elements; i++) {
>     data[i] = starting_value++;
>@@ -39,7 +38,8 @@ static int SetIncrementingData(int* data, int num_elements,
>   return starting_value;
> }
> 
>-static int CheckIncrementingData(int* data, int num_elements,
>+static int CheckIncrementingData(int* data,
>+                                 int num_elements,
>                                  int starting_value) {
>   for (int i = 0; i < num_elements; i++) {
>     AssertElementEq(starting_value++, data[i]);
>@@ -70,35 +70,33 @@ static void RandomStressTest(int** data_ptr) {
>     int read_element = 0;
>     for (int j = 0; j < kNumOps; j++) {
>       const bool write = rand() % 2 == 0 ? true : false;  // NOLINT
>-      const int num_elements = rand() % buffer_size;  // NOLINT
>+      const int num_elements = rand() % buffer_size;      // NOLINT
>       if (write) {
>         const int buffer_available = buffer_size - buffer_consumed;
>         ASSERT_EQ(static_cast<size_t>(buffer_available),
>                   WebRtc_available_write(buffer.get()));
>         const int expected_elements = std::min(num_elements, buffer_available);
>         write_element = SetIncrementingData(write_data.get(), expected_elements,
>-                                     write_element);
>-        ASSERT_EQ(static_cast<size_t>(expected_elements),
>-                  WebRtc_WriteBuffer(buffer.get(), write_data.get(),
>-                                     num_elements));
>-        buffer_consumed = std::min(buffer_consumed + expected_elements,
>-                                   buffer_size);
>+                                            write_element);
>+        ASSERT_EQ(
>+            static_cast<size_t>(expected_elements),
>+            WebRtc_WriteBuffer(buffer.get(), write_data.get(), num_elements));
>+        buffer_consumed =
>+            std::min(buffer_consumed + expected_elements, buffer_size);
>       } else {
>-        const int expected_elements = std::min(num_elements,
>-                                               buffer_consumed);
>+        const int expected_elements = std::min(num_elements, buffer_consumed);
>         ASSERT_EQ(static_cast<size_t>(buffer_consumed),
>                   WebRtc_available_read(buffer.get()));
>-        ASSERT_EQ(static_cast<size_t>(expected_elements),
>-                  WebRtc_ReadBuffer(buffer.get(),
>-                                    reinterpret_cast<void**>(data_ptr),
>-                                    read_data.get(),
>-                                    num_elements));
>+        ASSERT_EQ(
>+            static_cast<size_t>(expected_elements),
>+            WebRtc_ReadBuffer(buffer.get(), reinterpret_cast<void**>(data_ptr),
>+                              read_data.get(), num_elements));
>         int* check_ptr = read_data.get();
>         if (data_ptr) {
>           check_ptr = *data_ptr;
>         }
>-        read_element = CheckIncrementingData(check_ptr, expected_elements,
>-                                             read_element);
>+        read_element =
>+            CheckIncrementingData(check_ptr, expected_elements, read_element);
>         buffer_consumed = std::max(buffer_consumed - expected_elements, 0);
>       }
>     }
>@@ -127,8 +125,9 @@ TEST(RingBufferTest, PassingNulltoReadBufferForcesMemcpy) {
>   SetIncrementingData(write_data, kDataSize, 0);
>   EXPECT_EQ(kDataSize, WebRtc_WriteBuffer(buffer.get(), write_data, kDataSize));
>   SetIncrementingData(read_data, kDataSize, kDataSize);
>-  EXPECT_EQ(kDataSize, WebRtc_ReadBuffer(buffer.get(),
>-      reinterpret_cast<void**>(&data_ptr), read_data, kDataSize));
>+  EXPECT_EQ(kDataSize,
>+            WebRtc_ReadBuffer(buffer.get(), reinterpret_cast<void**>(&data_ptr),
>+                              read_data, kDataSize));
>   // Copying was not necessary, so |read_data| has not been updated.
>   CheckIncrementingData(data_ptr, kDataSize, 0);
>   CheckIncrementingData(read_data, kDataSize, kDataSize);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/complex_bit_reverse_arm.S b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/complex_bit_reverse_arm.S
>index c70349aaffa..be8e181aa7b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/complex_bit_reverse_arm.S
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/complex_bit_reverse_arm.S
>@@ -12,7 +12,7 @@
> @ for ARMv5 platforms.
> @ Reference C code is in file complex_bit_reverse.c. Bit-exact.
> 
>-#include "system_wrappers/include/asm_defines.h"
>+#include "rtc_base/system/asm_defines.h"
> 
> GLOBAL_FUNCTION WebRtcSpl_ComplexBitReverse
> .align  2
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/complex_fft.c b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/complex_fft.c
>index 36689b3c462..ddc9a97b599 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/complex_fft.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/complex_fft.c
>@@ -17,6 +17,7 @@
> 
> #include "common_audio/signal_processing/complex_fft_tables.h"
> #include "common_audio/signal_processing/include/signal_processing_library.h"
>+#include "rtc_base/system/arch.h"
> 
> #define CFFTSFT 14
> #define CFFTRND 1
>@@ -166,7 +167,7 @@ int WebRtcSpl_ComplexIFFT(int16_t frfi[], int stages, int mode)
>     /* The 1024-value is a constant given from the size of kSinTable1024[],
>      * and should not be changed depending on the input parameter 'stages'
>      */
>-    n = 1 << stages;
>+    n = ((size_t)1) << stages;
>     if (n > 1024)
>         return -1;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/complex_fft_tables.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/complex_fft_tables.h
>index 6c3fcfdbe05..90fac072d27 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/complex_fft_tables.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/complex_fft_tables.h
>@@ -8,141 +8,125 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-
> #ifndef COMMON_AUDIO_SIGNAL_PROCESSING_COMPLEX_FFT_TABLES_H_
> #define COMMON_AUDIO_SIGNAL_PROCESSING_COMPLEX_FFT_TABLES_H_
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include <stdint.h>
> 
> static const int16_t kSinTable1024[] = {
>-       0,    201,    402,    603,    804,   1005,   1206,   1406,
>-    1607,   1808,   2009,   2209,   2410,   2610,   2811,   3011,
>-    3211,   3411,   3611,   3811,   4011,   4210,   4409,   4608,
>-    4807,   5006,   5205,   5403,   5601,   5799,   5997,   6195,
>-    6392,   6589,   6786,   6982,   7179,   7375,   7571,   7766,
>-    7961,   8156,   8351,   8545,   8739,   8932,   9126,   9319,
>-    9511,   9703,   9895,  10087,  10278,  10469,  10659,  10849,
>-   11038,  11227,  11416,  11604,  11792,  11980,  12166,  12353,
>-   12539,  12724,  12909,  13094,  13278,  13462,  13645,  13827,
>-   14009,  14191,  14372,  14552,  14732,  14911,  15090,  15268,
>-   15446,  15623,  15799,  15975,  16150,  16325,  16499,  16672,
>-   16845,  17017,  17189,  17360,  17530,  17699,  17868,  18036,
>-   18204,  18371,  18537,  18702,  18867,  19031,  19194,  19357,
>-   19519,  19680,  19840,  20000,  20159,  20317,  20474,  20631,
>-   20787,  20942,  21096,  21249,  21402,  21554,  21705,  21855,
>-   22004,  22153,  22301,  22448,  22594,  22739,  22883,  23027,
>-   23169,  23311,  23452,  23592,  23731,  23869,  24006,  24143,
>-   24278,  24413,  24546,  24679,  24811,  24942,  25072,  25201,
>-   25329,  25456,  25582,  25707,  25831,  25954,  26077,  26198,
>-   26318,  26437,  26556,  26673,  26789,  26905,  27019,  27132,
>-   27244,  27355,  27466,  27575,  27683,  27790,  27896,  28001,
>-   28105,  28208,  28309,  28410,  28510,  28608,  28706,  28802,
>-   28897,  28992,  29085,  29177,  29268,  29358,  29446,  29534,
>-   29621,  29706,  29790,  29873,  29955,  30036,  30116,  30195,
>-   30272,  30349,  30424,  30498,  30571,  30643,  30713,  30783,
>-   30851,  30918,  30984,  31049,  31113,  31175,  31236,  31297,
>-   31356,  31413,  31470,  31525,  31580,  31633,  31684,  31735,
>-   31785,  31833,  31880,  31926,  31970,  32014,  32056,  32097,
>-   32137,  32176,  32213,  32249,  32284,  32318,  32350,  32382,
>-   32412,  32441,  32468,  32495,  32520,  32544,  32567,  32588,
>-   32609,  32628,  32646,  32662,  32678,  32692,  32705,  32717,
>-   32727,  32736,  32744,  32751,  32757,  32761,  32764,  32766,
>-   32767,  32766,  32764,  32761,  32757,  32751,  32744,  32736,
>-   32727,  32717,  32705,  32692,  32678,  32662,  32646,  32628,
>-   32609,  32588,  32567,  32544,  32520,  32495,  32468,  32441,
>-   32412,  32382,  32350,  32318,  32284,  32249,  32213,  32176,
>-   32137,  32097,  32056,  32014,  31970,  31926,  31880,  31833,
>-   31785,  31735,  31684,  31633,  31580,  31525,  31470,  31413,
>-   31356,  31297,  31236,  31175,  31113,  31049,  30984,  30918,
>-   30851,  30783,  30713,  30643,  30571,  30498,  30424,  30349,
>-   30272,  30195,  30116,  30036,  29955,  29873,  29790,  29706,
>-   29621,  29534,  29446,  29358,  29268,  29177,  29085,  28992,
>-   28897,  28802,  28706,  28608,  28510,  28410,  28309,  28208,
>-   28105,  28001,  27896,  27790,  27683,  27575,  27466,  27355,
>-   27244,  27132,  27019,  26905,  26789,  26673,  26556,  26437,
>-   26318,  26198,  26077,  25954,  25831,  25707,  25582,  25456,
>-   25329,  25201,  25072,  24942,  24811,  24679,  24546,  24413,
>-   24278,  24143,  24006,  23869,  23731,  23592,  23452,  23311,
>-   23169,  23027,  22883,  22739,  22594,  22448,  22301,  22153,
>-   22004,  21855,  21705,  21554,  21402,  21249,  21096,  20942,
>-   20787,  20631,  20474,  20317,  20159,  20000,  19840,  19680,
>-   19519,  19357,  19194,  19031,  18867,  18702,  18537,  18371,
>-   18204,  18036,  17868,  17699,  17530,  17360,  17189,  17017,
>-   16845,  16672,  16499,  16325,  16150,  15975,  15799,  15623,
>-   15446,  15268,  15090,  14911,  14732,  14552,  14372,  14191,
>-   14009,  13827,  13645,  13462,  13278,  13094,  12909,  12724,
>-   12539,  12353,  12166,  11980,  11792,  11604,  11416,  11227,
>-   11038,  10849,  10659,  10469,  10278,  10087,   9895,   9703,
>-    9511,   9319,   9126,   8932,   8739,   8545,   8351,   8156,
>-    7961,   7766,   7571,   7375,   7179,   6982,   6786,   6589,
>-    6392,   6195,   5997,   5799,   5601,   5403,   5205,   5006,
>-    4807,   4608,   4409,   4210,   4011,   3811,   3611,   3411,
>-    3211,   3011,   2811,   2610,   2410,   2209,   2009,   1808,
>-    1607,   1406,   1206,   1005,    804,    603,    402,    201,
>-       0,   -201,   -402,   -603,   -804,  -1005,  -1206,  -1406,
>-   -1607,  -1808,  -2009,  -2209,  -2410,  -2610,  -2811,  -3011,
>-   -3211,  -3411,  -3611,  -3811,  -4011,  -4210,  -4409,  -4608,
>-   -4807,  -5006,  -5205,  -5403,  -5601,  -5799,  -5997,  -6195,
>-   -6392,  -6589,  -6786,  -6982,  -7179,  -7375,  -7571,  -7766,
>-   -7961,  -8156,  -8351,  -8545,  -8739,  -8932,  -9126,  -9319,
>-   -9511,  -9703,  -9895, -10087, -10278, -10469, -10659, -10849,
>-  -11038, -11227, -11416, -11604, -11792, -11980, -12166, -12353,
>-  -12539, -12724, -12909, -13094, -13278, -13462, -13645, -13827,
>-  -14009, -14191, -14372, -14552, -14732, -14911, -15090, -15268,
>-  -15446, -15623, -15799, -15975, -16150, -16325, -16499, -16672,
>-  -16845, -17017, -17189, -17360, -17530, -17699, -17868, -18036,
>-  -18204, -18371, -18537, -18702, -18867, -19031, -19194, -19357,
>-  -19519, -19680, -19840, -20000, -20159, -20317, -20474, -20631,
>-  -20787, -20942, -21096, -21249, -21402, -21554, -21705, -21855,
>-  -22004, -22153, -22301, -22448, -22594, -22739, -22883, -23027,
>-  -23169, -23311, -23452, -23592, -23731, -23869, -24006, -24143,
>-  -24278, -24413, -24546, -24679, -24811, -24942, -25072, -25201,
>-  -25329, -25456, -25582, -25707, -25831, -25954, -26077, -26198,
>-  -26318, -26437, -26556, -26673, -26789, -26905, -27019, -27132,
>-  -27244, -27355, -27466, -27575, -27683, -27790, -27896, -28001,
>-  -28105, -28208, -28309, -28410, -28510, -28608, -28706, -28802,
>-  -28897, -28992, -29085, -29177, -29268, -29358, -29446, -29534,
>-  -29621, -29706, -29790, -29873, -29955, -30036, -30116, -30195,
>-  -30272, -30349, -30424, -30498, -30571, -30643, -30713, -30783,
>-  -30851, -30918, -30984, -31049, -31113, -31175, -31236, -31297,
>-  -31356, -31413, -31470, -31525, -31580, -31633, -31684, -31735,
>-  -31785, -31833, -31880, -31926, -31970, -32014, -32056, -32097,
>-  -32137, -32176, -32213, -32249, -32284, -32318, -32350, -32382,
>-  -32412, -32441, -32468, -32495, -32520, -32544, -32567, -32588,
>-  -32609, -32628, -32646, -32662, -32678, -32692, -32705, -32717,
>-  -32727, -32736, -32744, -32751, -32757, -32761, -32764, -32766,
>-  -32767, -32766, -32764, -32761, -32757, -32751, -32744, -32736,
>-  -32727, -32717, -32705, -32692, -32678, -32662, -32646, -32628,
>-  -32609, -32588, -32567, -32544, -32520, -32495, -32468, -32441,
>-  -32412, -32382, -32350, -32318, -32284, -32249, -32213, -32176,
>-  -32137, -32097, -32056, -32014, -31970, -31926, -31880, -31833,
>-  -31785, -31735, -31684, -31633, -31580, -31525, -31470, -31413,
>-  -31356, -31297, -31236, -31175, -31113, -31049, -30984, -30918,
>-  -30851, -30783, -30713, -30643, -30571, -30498, -30424, -30349,
>-  -30272, -30195, -30116, -30036, -29955, -29873, -29790, -29706,
>-  -29621, -29534, -29446, -29358, -29268, -29177, -29085, -28992,
>-  -28897, -28802, -28706, -28608, -28510, -28410, -28309, -28208,
>-  -28105, -28001, -27896, -27790, -27683, -27575, -27466, -27355,
>-  -27244, -27132, -27019, -26905, -26789, -26673, -26556, -26437,
>-  -26318, -26198, -26077, -25954, -25831, -25707, -25582, -25456,
>-  -25329, -25201, -25072, -24942, -24811, -24679, -24546, -24413,
>-  -24278, -24143, -24006, -23869, -23731, -23592, -23452, -23311,
>-  -23169, -23027, -22883, -22739, -22594, -22448, -22301, -22153,
>-  -22004, -21855, -21705, -21554, -21402, -21249, -21096, -20942,
>-  -20787, -20631, -20474, -20317, -20159, -20000, -19840, -19680,
>-  -19519, -19357, -19194, -19031, -18867, -18702, -18537, -18371,
>-  -18204, -18036, -17868, -17699, -17530, -17360, -17189, -17017,
>-  -16845, -16672, -16499, -16325, -16150, -15975, -15799, -15623,
>-  -15446, -15268, -15090, -14911, -14732, -14552, -14372, -14191,
>-  -14009, -13827, -13645, -13462, -13278, -13094, -12909, -12724,
>-  -12539, -12353, -12166, -11980, -11792, -11604, -11416, -11227,
>-  -11038, -10849, -10659, -10469, -10278, -10087,  -9895,  -9703,
>-   -9511,  -9319,  -9126,  -8932,  -8739,  -8545,  -8351,  -8156,
>-   -7961,  -7766,  -7571,  -7375,  -7179,  -6982,  -6786,  -6589,
>-   -6392,  -6195,  -5997,  -5799,  -5601,  -5403,  -5205,  -5006,
>-   -4807,  -4608,  -4409,  -4210,  -4011,  -3811,  -3611,  -3411,
>-   -3211,  -3011,  -2811,  -2610,  -2410,  -2209,  -2009,  -1808,
>-   -1607,  -1406,  -1206,  -1005,   -804,   -603,   -402,   -201
>-};
>+    0,      201,    402,    603,    804,    1005,   1206,   1406,   1607,
>+    1808,   2009,   2209,   2410,   2610,   2811,   3011,   3211,   3411,
>+    3611,   3811,   4011,   4210,   4409,   4608,   4807,   5006,   5205,
>+    5403,   5601,   5799,   5997,   6195,   6392,   6589,   6786,   6982,
>+    7179,   7375,   7571,   7766,   7961,   8156,   8351,   8545,   8739,
>+    8932,   9126,   9319,   9511,   9703,   9895,   10087,  10278,  10469,
>+    10659,  10849,  11038,  11227,  11416,  11604,  11792,  11980,  12166,
>+    12353,  12539,  12724,  12909,  13094,  13278,  13462,  13645,  13827,
>+    14009,  14191,  14372,  14552,  14732,  14911,  15090,  15268,  15446,
>+    15623,  15799,  15975,  16150,  16325,  16499,  16672,  16845,  17017,
>+    17189,  17360,  17530,  17699,  17868,  18036,  18204,  18371,  18537,
>+    18702,  18867,  19031,  19194,  19357,  19519,  19680,  19840,  20000,
>+    20159,  20317,  20474,  20631,  20787,  20942,  21096,  21249,  21402,
>+    21554,  21705,  21855,  22004,  22153,  22301,  22448,  22594,  22739,
>+    22883,  23027,  23169,  23311,  23452,  23592,  23731,  23869,  24006,
>+    24143,  24278,  24413,  24546,  24679,  24811,  24942,  25072,  25201,
>+    25329,  25456,  25582,  25707,  25831,  25954,  26077,  26198,  26318,
>+    26437,  26556,  26673,  26789,  26905,  27019,  27132,  27244,  27355,
>+    27466,  27575,  27683,  27790,  27896,  28001,  28105,  28208,  28309,
>+    28410,  28510,  28608,  28706,  28802,  28897,  28992,  29085,  29177,
>+    29268,  29358,  29446,  29534,  29621,  29706,  29790,  29873,  29955,
>+    30036,  30116,  30195,  30272,  30349,  30424,  30498,  30571,  30643,
>+    30713,  30783,  30851,  30918,  30984,  31049,  31113,  31175,  31236,
>+    31297,  31356,  31413,  31470,  31525,  31580,  31633,  31684,  31735,
>+    31785,  31833,  31880,  31926,  31970,  32014,  32056,  32097,  32137,
>+    32176,  32213,  32249,  32284,  32318,  32350,  32382,  32412,  32441,
>+    32468,  32495,  32520,  32544,  32567,  32588,  32609,  32628,  32646,
>+    32662,  32678,  32692,  32705,  32717,  32727,  32736,  32744,  32751,
>+    32757,  32761,  32764,  32766,  32767,  32766,  32764,  32761,  32757,
>+    32751,  32744,  32736,  32727,  32717,  32705,  32692,  32678,  32662,
>+    32646,  32628,  32609,  32588,  32567,  32544,  32520,  32495,  32468,
>+    32441,  32412,  32382,  32350,  32318,  32284,  32249,  32213,  32176,
>+    32137,  32097,  32056,  32014,  31970,  31926,  31880,  31833,  31785,
>+    31735,  31684,  31633,  31580,  31525,  31470,  31413,  31356,  31297,
>+    31236,  31175,  31113,  31049,  30984,  30918,  30851,  30783,  30713,
>+    30643,  30571,  30498,  30424,  30349,  30272,  30195,  30116,  30036,
>+    29955,  29873,  29790,  29706,  29621,  29534,  29446,  29358,  29268,
>+    29177,  29085,  28992,  28897,  28802,  28706,  28608,  28510,  28410,
>+    28309,  28208,  28105,  28001,  27896,  27790,  27683,  27575,  27466,
>+    27355,  27244,  27132,  27019,  26905,  26789,  26673,  26556,  26437,
>+    26318,  26198,  26077,  25954,  25831,  25707,  25582,  25456,  25329,
>+    25201,  25072,  24942,  24811,  24679,  24546,  24413,  24278,  24143,
>+    24006,  23869,  23731,  23592,  23452,  23311,  23169,  23027,  22883,
>+    22739,  22594,  22448,  22301,  22153,  22004,  21855,  21705,  21554,
>+    21402,  21249,  21096,  20942,  20787,  20631,  20474,  20317,  20159,
>+    20000,  19840,  19680,  19519,  19357,  19194,  19031,  18867,  18702,
>+    18537,  18371,  18204,  18036,  17868,  17699,  17530,  17360,  17189,
>+    17017,  16845,  16672,  16499,  16325,  16150,  15975,  15799,  15623,
>+    15446,  15268,  15090,  14911,  14732,  14552,  14372,  14191,  14009,
>+    13827,  13645,  13462,  13278,  13094,  12909,  12724,  12539,  12353,
>+    12166,  11980,  11792,  11604,  11416,  11227,  11038,  10849,  10659,
>+    10469,  10278,  10087,  9895,   9703,   9511,   9319,   9126,   8932,
>+    8739,   8545,   8351,   8156,   7961,   7766,   7571,   7375,   7179,
>+    6982,   6786,   6589,   6392,   6195,   5997,   5799,   5601,   5403,
>+    5205,   5006,   4807,   4608,   4409,   4210,   4011,   3811,   3611,
>+    3411,   3211,   3011,   2811,   2610,   2410,   2209,   2009,   1808,
>+    1607,   1406,   1206,   1005,   804,    603,    402,    201,    0,
>+    -201,   -402,   -603,   -804,   -1005,  -1206,  -1406,  -1607,  -1808,
>+    -2009,  -2209,  -2410,  -2610,  -2811,  -3011,  -3211,  -3411,  -3611,
>+    -3811,  -4011,  -4210,  -4409,  -4608,  -4807,  -5006,  -5205,  -5403,
>+    -5601,  -5799,  -5997,  -6195,  -6392,  -6589,  -6786,  -6982,  -7179,
>+    -7375,  -7571,  -7766,  -7961,  -8156,  -8351,  -8545,  -8739,  -8932,
>+    -9126,  -9319,  -9511,  -9703,  -9895,  -10087, -10278, -10469, -10659,
>+    -10849, -11038, -11227, -11416, -11604, -11792, -11980, -12166, -12353,
>+    -12539, -12724, -12909, -13094, -13278, -13462, -13645, -13827, -14009,
>+    -14191, -14372, -14552, -14732, -14911, -15090, -15268, -15446, -15623,
>+    -15799, -15975, -16150, -16325, -16499, -16672, -16845, -17017, -17189,
>+    -17360, -17530, -17699, -17868, -18036, -18204, -18371, -18537, -18702,
>+    -18867, -19031, -19194, -19357, -19519, -19680, -19840, -20000, -20159,
>+    -20317, -20474, -20631, -20787, -20942, -21096, -21249, -21402, -21554,
>+    -21705, -21855, -22004, -22153, -22301, -22448, -22594, -22739, -22883,
>+    -23027, -23169, -23311, -23452, -23592, -23731, -23869, -24006, -24143,
>+    -24278, -24413, -24546, -24679, -24811, -24942, -25072, -25201, -25329,
>+    -25456, -25582, -25707, -25831, -25954, -26077, -26198, -26318, -26437,
>+    -26556, -26673, -26789, -26905, -27019, -27132, -27244, -27355, -27466,
>+    -27575, -27683, -27790, -27896, -28001, -28105, -28208, -28309, -28410,
>+    -28510, -28608, -28706, -28802, -28897, -28992, -29085, -29177, -29268,
>+    -29358, -29446, -29534, -29621, -29706, -29790, -29873, -29955, -30036,
>+    -30116, -30195, -30272, -30349, -30424, -30498, -30571, -30643, -30713,
>+    -30783, -30851, -30918, -30984, -31049, -31113, -31175, -31236, -31297,
>+    -31356, -31413, -31470, -31525, -31580, -31633, -31684, -31735, -31785,
>+    -31833, -31880, -31926, -31970, -32014, -32056, -32097, -32137, -32176,
>+    -32213, -32249, -32284, -32318, -32350, -32382, -32412, -32441, -32468,
>+    -32495, -32520, -32544, -32567, -32588, -32609, -32628, -32646, -32662,
>+    -32678, -32692, -32705, -32717, -32727, -32736, -32744, -32751, -32757,
>+    -32761, -32764, -32766, -32767, -32766, -32764, -32761, -32757, -32751,
>+    -32744, -32736, -32727, -32717, -32705, -32692, -32678, -32662, -32646,
>+    -32628, -32609, -32588, -32567, -32544, -32520, -32495, -32468, -32441,
>+    -32412, -32382, -32350, -32318, -32284, -32249, -32213, -32176, -32137,
>+    -32097, -32056, -32014, -31970, -31926, -31880, -31833, -31785, -31735,
>+    -31684, -31633, -31580, -31525, -31470, -31413, -31356, -31297, -31236,
>+    -31175, -31113, -31049, -30984, -30918, -30851, -30783, -30713, -30643,
>+    -30571, -30498, -30424, -30349, -30272, -30195, -30116, -30036, -29955,
>+    -29873, -29790, -29706, -29621, -29534, -29446, -29358, -29268, -29177,
>+    -29085, -28992, -28897, -28802, -28706, -28608, -28510, -28410, -28309,
>+    -28208, -28105, -28001, -27896, -27790, -27683, -27575, -27466, -27355,
>+    -27244, -27132, -27019, -26905, -26789, -26673, -26556, -26437, -26318,
>+    -26198, -26077, -25954, -25831, -25707, -25582, -25456, -25329, -25201,
>+    -25072, -24942, -24811, -24679, -24546, -24413, -24278, -24143, -24006,
>+    -23869, -23731, -23592, -23452, -23311, -23169, -23027, -22883, -22739,
>+    -22594, -22448, -22301, -22153, -22004, -21855, -21705, -21554, -21402,
>+    -21249, -21096, -20942, -20787, -20631, -20474, -20317, -20159, -20000,
>+    -19840, -19680, -19519, -19357, -19194, -19031, -18867, -18702, -18537,
>+    -18371, -18204, -18036, -17868, -17699, -17530, -17360, -17189, -17017,
>+    -16845, -16672, -16499, -16325, -16150, -15975, -15799, -15623, -15446,
>+    -15268, -15090, -14911, -14732, -14552, -14372, -14191, -14009, -13827,
>+    -13645, -13462, -13278, -13094, -12909, -12724, -12539, -12353, -12166,
>+    -11980, -11792, -11604, -11416, -11227, -11038, -10849, -10659, -10469,
>+    -10278, -10087, -9895,  -9703,  -9511,  -9319,  -9126,  -8932,  -8739,
>+    -8545,  -8351,  -8156,  -7961,  -7766,  -7571,  -7375,  -7179,  -6982,
>+    -6786,  -6589,  -6392,  -6195,  -5997,  -5799,  -5601,  -5403,  -5205,
>+    -5006,  -4807,  -4608,  -4409,  -4210,  -4011,  -3811,  -3611,  -3411,
>+    -3211,  -3011,  -2811,  -2610,  -2410,  -2209,  -2009,  -1808,  -1607,
>+    -1406,  -1206,  -1005,  -804,   -603,   -402,   -201};
> 
> #endif  // COMMON_AUDIO_SIGNAL_PROCESSING_COMPLEX_FFT_TABLES_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/cross_correlation_neon.c b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/cross_correlation_neon.c
>index fdd03f1976d..f2afbdf9f53 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/cross_correlation_neon.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/cross_correlation_neon.c
>@@ -9,6 +9,7 @@
>  */
> 
> #include "common_audio/signal_processing/include/signal_processing_library.h"
>+#include "rtc_base/system/arch.h"
> 
> #include <arm_neon.h>
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/dot_product_with_scale.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/dot_product_with_scale.h
>index ff3c525066a..bb892d40cb3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/dot_product_with_scale.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/dot_product_with_scale.h
>@@ -11,10 +11,9 @@
> #ifndef COMMON_AUDIO_SIGNAL_PROCESSING_DOT_PRODUCT_WITH_SCALE_H_
> #define COMMON_AUDIO_SIGNAL_PROCESSING_DOT_PRODUCT_WITH_SCALE_H_
> 
>+#include <stdint.h>
> #include <string.h>
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>-
> #ifdef __cplusplus
> extern "C" {
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/downsample_fast.c b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/downsample_fast.c
>index 9a2ea053da2..80fdc58a499 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/downsample_fast.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/downsample_fast.c
>@@ -42,8 +42,13 @@ int WebRtcSpl_DownsampleFastC(const int16_t* data_in,
>     out_s32 = 2048;  // Round value, 0.5 in Q12.
> 
>     for (j = 0; j < coefficients_length; j++) {
>-      rtc_MsanCheckInitialized(&data_in[i - j], sizeof(data_in[0]), 1);
>-      out_s32 += coefficients[j] * data_in[i - j];  // Q12.
>+      // Negative overflow is permitted here, because this is
>+      // auto-regressive filters, and the state for each batch run is
>+      // stored in the "negative" positions of the output vector.
>+      rtc_MsanCheckInitialized(&data_in[(ptrdiff_t) i - (ptrdiff_t) j],
>+          sizeof(data_in[0]), 1);
>+      // out_s32 is in Q12 domain.
>+      out_s32 += coefficients[j] * data_in[(ptrdiff_t) i - (ptrdiff_t) j];
>     }
> 
>     out_s32 >>= 12;  // Q0.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/filter_ar.c b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/filter_ar.c
>index 49d5d61960f..2471cd11178 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/filter_ar.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/filter_ar.c
>@@ -17,6 +17,8 @@
> 
> #include "common_audio/signal_processing/include/signal_processing_library.h"
> 
>+#include "rtc_base/checks.h"
>+
> size_t WebRtcSpl_FilterAR(const int16_t* a,
>                           size_t a_length,
>                           const int16_t* x,
>@@ -40,8 +42,10 @@ size_t WebRtcSpl_FilterAR(const int16_t* a,
>     {
>         // Calculate filtered[i] and filtered_low[i]
>         const int16_t* a_ptr = &a[1];
>-        int16_t* filtered_ptr = &filtered[i - 1];
>-        int16_t* filtered_low_ptr = &filtered_low[i - 1];
>+        // The index can become negative, but the arrays will never be indexed
>+        // with it when negative. Nevertheless, the index cannot be a size_t
>+        // because of this.
>+        int filtered_ix = (int)i - 1;
>         int16_t* state_ptr = &state[state_length - 1];
>         int16_t* state_low_ptr = &state_low[state_length - 1];
> 
>@@ -51,8 +55,10 @@ size_t WebRtcSpl_FilterAR(const int16_t* a,
>         stop = (i < a_length) ? i + 1 : a_length;
>         for (j = 1; j < stop; j++)
>         {
>-          o -= *a_ptr * *filtered_ptr--;
>-          oLOW -= *a_ptr++ * *filtered_low_ptr--;
>+          RTC_DCHECK_GE(filtered_ix, 0);
>+          o -= *a_ptr * filtered[filtered_ix];
>+          oLOW -= *a_ptr++ * filtered_low[filtered_ix];
>+          --filtered_ix;
>         }
>         for (j = i + 1; j < a_length; j++)
>         {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/filter_ar_fast_q12.c b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/filter_ar_fast_q12.c
>index df9e51889ab..8b8bdb1af5e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/filter_ar_fast_q12.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/filter_ar_fast_q12.c
>@@ -8,6 +8,8 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>+#include "stddef.h"
>+
> #include "rtc_base/checks.h"
> #include "common_audio/signal_processing/include/signal_processing_library.h"
> 
>@@ -29,7 +31,10 @@ void WebRtcSpl_FilterARFastQ12(const int16_t* data_in,
>     int64_t sum = 0;
> 
>     for (j = coefficients_length - 1; j > 0; j--) {
>-      sum += coefficients[j] * data_out[i - j];
>+      // Negative overflow is permitted here, because this is
>+      // auto-regressive filters, and the state for each batch run is
>+      // stored in the "negative" positions of the output vector.
>+      sum += coefficients[j] * data_out[(ptrdiff_t) i - (ptrdiff_t) j];
>     }
> 
>     output = coefficients[0] * data_in[i];
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/filter_ar_fast_q12_armv7.S b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/filter_ar_fast_q12_armv7.S
>index c6397c2055c..60319d29ffd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/filter_ar_fast_q12_armv7.S
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/filter_ar_fast_q12_armv7.S
>@@ -35,7 +35,7 @@
> @ r11: Scratch
> @ r12: &coefficients[j]
> 
>-#include "system_wrappers/include/asm_defines.h"
>+#include "rtc_base/system/asm_defines.h"
> 
> GLOBAL_FUNCTION WebRtcSpl_FilterARFastQ12
> .align  2
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/filter_ma_fast_q12.c b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/filter_ma_fast_q12.c
>index 9596ef19222..329d47e14f9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/filter_ma_fast_q12.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/filter_ma_fast_q12.c
>@@ -37,7 +37,10 @@ void WebRtcSpl_FilterMAFastQ12(const int16_t* in_ptr,
> 
>         for (j = 0; j < B_length; j++)
>         {
>-          o += B[j] * in_ptr[i - j];
>+          // Negative overflow is permitted here, because this is
>+          // auto-regressive filters, and the state for each batch run is
>+          // stored in the "negative" positions of the output vector.
>+          o += B[j] * in_ptr[(ptrdiff_t) i - (ptrdiff_t) j];
>         }
> 
>         // If output is higher than 32768, saturate it. Same with negative side
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/include/real_fft.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/include/real_fft.h
>index 7d21072bc71..84450667d1a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/include/real_fft.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/include/real_fft.h
>@@ -11,12 +11,11 @@
> #ifndef COMMON_AUDIO_SIGNAL_PROCESSING_INCLUDE_REAL_FFT_H_
> #define COMMON_AUDIO_SIGNAL_PROCESSING_INCLUDE_REAL_FFT_H_
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include <stdint.h>
> 
> // For ComplexFFT(), the maximum fft order is 10;
>-// for OpenMax FFT in ARM, it is 12;
> // WebRTC APM uses orders of only 7 and 8.
>-enum {kMaxFFTOrder = 10};
>+enum { kMaxFFTOrder = 10 };
> 
> struct RealFFT;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/include/signal_processing_library.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/include/signal_processing_library.h
>index 27fea6f8377..33469d429b2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/include/signal_processing_library.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/include/signal_processing_library.h
>@@ -8,11 +8,10 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-
> /*
>- * This header file includes all of the fix point signal processing library (SPL) function
>- * descriptions and declarations.
>- * For specific function calls, see bottom of file.
>+ * This header file includes all of the fix point signal processing library
>+ * (SPL) function descriptions and declarations. For specific function calls,
>+ * see bottom of file.
>  */
> 
> #ifndef COMMON_AUDIO_SIGNAL_PROCESSING_INCLUDE_SIGNAL_PROCESSING_LIBRARY_H_
>@@ -20,66 +19,60 @@
> 
> #include <string.h>
> #include "common_audio/signal_processing/dot_product_with_scale.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> // Macros specific for the fixed point implementation
>-#define WEBRTC_SPL_WORD16_MAX       32767
>-#define WEBRTC_SPL_WORD16_MIN       -32768
>-#define WEBRTC_SPL_WORD32_MAX       (int32_t)0x7fffffff
>-#define WEBRTC_SPL_WORD32_MIN       (int32_t)0x80000000
>-#define WEBRTC_SPL_MAX_LPC_ORDER    14
>-#define WEBRTC_SPL_MIN(A, B)        (A < B ? A : B)  // Get min value
>-#define WEBRTC_SPL_MAX(A, B)        (A > B ? A : B)  // Get max value
>+#define WEBRTC_SPL_WORD16_MAX 32767
>+#define WEBRTC_SPL_WORD16_MIN -32768
>+#define WEBRTC_SPL_WORD32_MAX (int32_t)0x7fffffff
>+#define WEBRTC_SPL_WORD32_MIN (int32_t)0x80000000
>+#define WEBRTC_SPL_MAX_LPC_ORDER 14
>+#define WEBRTC_SPL_MIN(A, B) (A < B ? A : B)  // Get min value
>+#define WEBRTC_SPL_MAX(A, B) (A > B ? A : B)  // Get max value
> // TODO(kma/bjorn): For the next two macros, investigate how to correct the code
> // for inputs of a = WEBRTC_SPL_WORD16_MIN or WEBRTC_SPL_WORD32_MIN.
>-#define WEBRTC_SPL_ABS_W16(a) \
>-    (((int16_t)a >= 0) ? ((int16_t)a) : -((int16_t)a))
>-#define WEBRTC_SPL_ABS_W32(a) \
>-    (((int32_t)a >= 0) ? ((int32_t)a) : -((int32_t)a))
>-
>-#define WEBRTC_SPL_MUL(a, b) \
>-    ((int32_t) ((int32_t)(a) * (int32_t)(b)))
>-#define WEBRTC_SPL_UMUL(a, b) \
>-    ((uint32_t) ((uint32_t)(a) * (uint32_t)(b)))
>-#define WEBRTC_SPL_UMUL_32_16(a, b) \
>-    ((uint32_t) ((uint32_t)(a) * (uint16_t)(b)))
>-#define WEBRTC_SPL_MUL_16_U16(a, b) \
>-    ((int32_t)(int16_t)(a) * (uint16_t)(b))
>+#define WEBRTC_SPL_ABS_W16(a) (((int16_t)a >= 0) ? ((int16_t)a) : -((int16_t)a))
>+#define WEBRTC_SPL_ABS_W32(a) (((int32_t)a >= 0) ? ((int32_t)a) : -((int32_t)a))
>+
>+#define WEBRTC_SPL_MUL(a, b) ((int32_t)((int32_t)(a) * (int32_t)(b)))
>+#define WEBRTC_SPL_UMUL(a, b) ((uint32_t)((uint32_t)(a) * (uint32_t)(b)))
>+#define WEBRTC_SPL_UMUL_32_16(a, b) ((uint32_t)((uint32_t)(a) * (uint16_t)(b)))
>+#define WEBRTC_SPL_MUL_16_U16(a, b) ((int32_t)(int16_t)(a) * (uint16_t)(b))
> 
>+// clang-format off
>+// clang-format would choose some identation
>+// leading to presubmit error (cpplint.py)
> #ifndef WEBRTC_ARCH_ARM_V7
> // For ARMv7 platforms, these are inline functions in spl_inl_armv7.h
> #ifndef MIPS32_LE
> // For MIPS platforms, these are inline functions in spl_inl_mips.h
>-#define WEBRTC_SPL_MUL_16_16(a, b) \
>-    ((int32_t) (((int16_t)(a)) * ((int16_t)(b))))
>+#define WEBRTC_SPL_MUL_16_16(a, b) ((int32_t)(((int16_t)(a)) * ((int16_t)(b))))
> #define WEBRTC_SPL_MUL_16_32_RSFT16(a, b) \
>-    (WEBRTC_SPL_MUL_16_16(a, b >> 16) \
>-     + ((WEBRTC_SPL_MUL_16_16(a, (b & 0xffff) >> 1) + 0x4000) >> 15))
>+        (WEBRTC_SPL_MUL_16_16(a, b >> 16) +     \
>+        ((WEBRTC_SPL_MUL_16_16(a, (b & 0xffff) >> 1) + 0x4000) >> 15))
> #endif
> #endif
> 
> #define WEBRTC_SPL_MUL_16_32_RSFT11(a, b)          \
>-  (WEBRTC_SPL_MUL_16_16(a, (b) >> 16) * (1 << 5) + \
>-    (((WEBRTC_SPL_MUL_16_U16(a, (uint16_t)(b)) >> 1) + 0x0200) >> 10))
>+        (WEBRTC_SPL_MUL_16_16(a, (b) >> 16) * (1 << 5) + \
>+        (((WEBRTC_SPL_MUL_16_U16(a, (uint16_t)(b)) >> 1) + 0x0200) >> 10))
> #define WEBRTC_SPL_MUL_16_32_RSFT14(a, b)          \
>-  (WEBRTC_SPL_MUL_16_16(a, (b) >> 16) * (1 << 2) + \
>-    (((WEBRTC_SPL_MUL_16_U16(a, (uint16_t)(b)) >> 1) + 0x1000) >> 13))
>+        (WEBRTC_SPL_MUL_16_16(a, (b) >> 16) * (1 << 2) + \
>+        (((WEBRTC_SPL_MUL_16_U16(a, (uint16_t)(b)) >> 1) + 0x1000) >> 13))
> #define WEBRTC_SPL_MUL_16_32_RSFT15(a, b)            \
>-  ((WEBRTC_SPL_MUL_16_16(a, (b) >> 16) * (1 << 1)) + \
>-    (((WEBRTC_SPL_MUL_16_U16(a, (uint16_t)(b)) >> 1) + 0x2000) >> 14))
>+        ((WEBRTC_SPL_MUL_16_16(a, (b) >> 16) * (1 << 1)) + \
>+        (((WEBRTC_SPL_MUL_16_U16(a, (uint16_t)(b)) >> 1) + 0x2000) >> 14))
>+// clang-format on
> 
>-#define WEBRTC_SPL_MUL_16_16_RSFT(a, b, c) \
>-    (WEBRTC_SPL_MUL_16_16(a, b) >> (c))
>+#define WEBRTC_SPL_MUL_16_16_RSFT(a, b, c) (WEBRTC_SPL_MUL_16_16(a, b) >> (c))
> 
> #define WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(a, b, c) \
>-    ((WEBRTC_SPL_MUL_16_16(a, b) + ((int32_t) \
>-                                  (((int32_t)1) << ((c) - 1)))) >> (c))
>+  ((WEBRTC_SPL_MUL_16_16(a, b) + ((int32_t)(((int32_t)1) << ((c)-1)))) >> (c))
> 
> // C + the 32 most significant bits of A * B
> #define WEBRTC_SPL_SCALEDIFF32(A, B, C) \
>-    (C + (B >> 16) * A + (((uint32_t)(B & 0x0000FFFF) * A) >> 16))
>+  (C + (B >> 16) * A + (((uint32_t)(B & 0x0000FFFF) * A) >> 16))
> 
>-#define WEBRTC_SPL_SAT(a, b, c)         (b > a ? a : b < c ? c : b)
>+#define WEBRTC_SPL_SAT(a, b, c) (b > a ? a : b < c ? c : b)
> 
> // Shifting with negative numbers allowed
> // Positive means left shift
>@@ -87,12 +80,11 @@
> 
> // Shifting with negative numbers not allowed
> // We cannot do casting here due to signed/unsigned problem
>-#define WEBRTC_SPL_LSHIFT_W32(x, c)     ((x) << (c))
>+#define WEBRTC_SPL_LSHIFT_W32(x, c) ((x) << (c))
> 
>-#define WEBRTC_SPL_RSHIFT_U32(x, c)     ((uint32_t)(x) >> (c))
>+#define WEBRTC_SPL_RSHIFT_U32(x, c) ((uint32_t)(x) >> (c))
> 
>-#define WEBRTC_SPL_RAND(a) \
>-    ((int16_t)((((int16_t)a * 18816) >> 7) & 0x00007fff))
>+#define WEBRTC_SPL_RAND(a) ((int16_t)((((int16_t)a * 18816) >> 7) & 0x00007fff))
> 
> #ifdef __cplusplus
> extern "C" {
>@@ -104,6 +96,9 @@ extern "C" {
> // inline functions:
> #include "common_audio/signal_processing/include/spl_inl.h"
> 
>+// third party math functions
>+#include "common_audio/third_party/spl_sqrt_floor/spl_sqrt_floor.h"
>+
> // Initialize SPL. Currently it contains only function pointer initialization.
> // If the underlying platform is known to be ARM-Neon (WEBRTC_HAS_NEON defined),
> // the pointers will be assigned to code optimized for Neon; otherwise, generic
>@@ -131,13 +126,10 @@ void WebRtcSpl_CopyFromEndW16(const int16_t* in_vector,
>                               size_t in_vector_length,
>                               size_t samples,
>                               int16_t* out_vector);
>-void WebRtcSpl_ZerosArrayW16(int16_t* vector,
>-                             size_t vector_length);
>-void WebRtcSpl_ZerosArrayW32(int32_t* vector,
>-                             size_t vector_length);
>+void WebRtcSpl_ZerosArrayW16(int16_t* vector, size_t vector_length);
>+void WebRtcSpl_ZerosArrayW32(int32_t* vector, size_t vector_length);
> // End: Copy and set operations.
> 
>-
> // Minimum and maximum operation functions and their pointers.
> // Implementation in min_max_operations.c.
> 
>@@ -297,7 +289,6 @@ size_t WebRtcSpl_MinIndexW32(const int32_t* vector, size_t length);
> 
> // End: Minimum and maximum operations.
> 
>-
> // Vector scaling operations. Implementation in vector_scaling_operations.c.
> // Description at bottom of file.
> void WebRtcSpl_VectorBitShiftW16(int16_t* out_vector,
>@@ -323,9 +314,11 @@ void WebRtcSpl_ScaleVectorWithSat(const int16_t* in_vector,
>                                   size_t vector_length,
>                                   int16_t right_shifts);
> void WebRtcSpl_ScaleAndAddVectors(const int16_t* in_vector1,
>-                                  int16_t gain1, int right_shifts1,
>+                                  int16_t gain1,
>+                                  int right_shifts1,
>                                   const int16_t* in_vector2,
>-                                  int16_t gain2, int right_shifts2,
>+                                  int16_t gain2,
>+                                  int right_shifts2,
>                                   int16_t* out_vector,
>                                   size_t vector_length);
> 
>@@ -583,7 +576,6 @@ int16_t WebRtcSpl_RandUArray(int16_t* vector,
> 
> // Math functions
> int32_t WebRtcSpl_Sqrt(int32_t value);
>-int32_t WebRtcSpl_SqrtFloor(int32_t value);
> 
> // Divisions. Implementations collected in division_operations.c and
> // descriptions at bottom of this file.
>@@ -777,7 +769,8 @@ typedef struct {
>   int32_t S_16_8[8];
> } WebRtcSpl_State22khzTo8khz;
> 
>-void WebRtcSpl_Resample22khzTo8khz(const int16_t* in, int16_t* out,
>+void WebRtcSpl_Resample22khzTo8khz(const int16_t* in,
>+                                   int16_t* out,
>                                    WebRtcSpl_State22khzTo8khz* state,
>                                    int32_t* tmpmem);
> 
>@@ -790,7 +783,8 @@ typedef struct {
>   int32_t S_11_22[8];
> } WebRtcSpl_State8khzTo22khz;
> 
>-void WebRtcSpl_Resample8khzTo22khz(const int16_t* in, int16_t* out,
>+void WebRtcSpl_Resample8khzTo22khz(const int16_t* in,
>+                                   int16_t* out,
>                                    WebRtcSpl_State8khzTo22khz* state,
>                                    int32_t* tmpmem);
> 
>@@ -830,7 +824,8 @@ typedef struct {
>   int32_t S_32_16[8];
> } WebRtcSpl_State48khzTo16khz;
> 
>-void WebRtcSpl_Resample48khzTo16khz(const int16_t* in, int16_t* out,
>+void WebRtcSpl_Resample48khzTo16khz(const int16_t* in,
>+                                    int16_t* out,
>                                     WebRtcSpl_State48khzTo16khz* state,
>                                     int32_t* tmpmem);
> 
>@@ -842,7 +837,8 @@ typedef struct {
>   int32_t S_24_48[8];
> } WebRtcSpl_State16khzTo48khz;
> 
>-void WebRtcSpl_Resample16khzTo48khz(const int16_t* in, int16_t* out,
>+void WebRtcSpl_Resample16khzTo48khz(const int16_t* in,
>+                                    int16_t* out,
>                                     WebRtcSpl_State16khzTo48khz* state,
>                                     int32_t* tmpmem);
> 
>@@ -855,7 +851,8 @@ typedef struct {
>   int32_t S_16_8[8];
> } WebRtcSpl_State48khzTo8khz;
> 
>-void WebRtcSpl_Resample48khzTo8khz(const int16_t* in, int16_t* out,
>+void WebRtcSpl_Resample48khzTo8khz(const int16_t* in,
>+                                   int16_t* out,
>                                    WebRtcSpl_State48khzTo8khz* state,
>                                    int32_t* tmpmem);
> 
>@@ -868,7 +865,8 @@ typedef struct {
>   int32_t S_24_48[8];
> } WebRtcSpl_State8khzTo48khz;
> 
>-void WebRtcSpl_Resample8khzTo48khz(const int16_t* in, int16_t* out,
>+void WebRtcSpl_Resample8khzTo48khz(const int16_t* in,
>+                                   int16_t* out,
>                                    WebRtcSpl_State8khzTo48khz* state,
>                                    int32_t* tmpmem);
> 
>@@ -881,11 +879,15 @@ void WebRtcSpl_ResetResample8khzTo48khz(WebRtcSpl_State8khzTo48khz* state);
>  *
>  ******************************************************************/
> 
>-void WebRtcSpl_DownsampleBy2(const int16_t* in, size_t len,
>-                             int16_t* out, int32_t* filtState);
>+void WebRtcSpl_DownsampleBy2(const int16_t* in,
>+                             size_t len,
>+                             int16_t* out,
>+                             int32_t* filtState);
> 
>-void WebRtcSpl_UpsampleBy2(const int16_t* in, size_t len,
>-                           int16_t* out, int32_t* filtState);
>+void WebRtcSpl_UpsampleBy2(const int16_t* in,
>+                           size_t len,
>+                           int16_t* out,
>+                           int32_t* filtState);
> 
> /************************************************************
>  * END OF RESAMPLING FUNCTIONS
>@@ -1334,23 +1336,6 @@ void WebRtcSpl_SynthesisQMF(const int16_t* low_band,
> // Return value     : Result of the sqrt calculation
> //
> 
>-//
>-// WebRtcSpl_SqrtFloor(...)
>-//
>-// Returns the square root of the input value |value|. The precision of this
>-// function is rounding down integer precision, i.e., sqrt(8) gives 2 as answer.
>-// If |value| is a negative number then 0 is returned.
>-//
>-// Algorithm:
>-//
>-// An iterative 4 cylce/bit routine
>-//
>-// Input:
>-//      - value     : Value to calculate sqrt of
>-//
>-// Return value     : Result of the sqrt calculation
>-//
>-
> //
> // WebRtcSpl_DivU32U16(...)
> //
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/include/spl_inl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/include/spl_inl.h
>index ba3a11376b9..656a3125bbe 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/include/spl_inl.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/include/spl_inl.h
>@@ -8,7 +8,6 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-
> // This header file includes the inline functions in
> // the fix point signal processing library.
> 
>@@ -73,7 +72,7 @@ static __inline int WebRtcSpl_CountLeadingZeros64(uint64_t n) {
> 
> #if !defined(MIPS_DSP_R1_LE)
> static __inline int16_t WebRtcSpl_SatW32ToW16(int32_t value32) {
>-  int16_t out16 = (int16_t) value32;
>+  int16_t out16 = (int16_t)value32;
> 
>   if (value32 > 32767)
>     out16 = 32767;
>@@ -112,11 +111,11 @@ static __inline int32_t WebRtcSpl_SubSatW32(int32_t a, int32_t b) {
> }
> 
> static __inline int16_t WebRtcSpl_AddSatW16(int16_t a, int16_t b) {
>-  return WebRtcSpl_SatW32ToW16((int32_t) a + (int32_t) b);
>+  return WebRtcSpl_SatW32ToW16((int32_t)a + (int32_t)b);
> }
> 
> static __inline int16_t WebRtcSpl_SubSatW16(int16_t var1, int16_t var2) {
>-  return WebRtcSpl_SatW32ToW16((int32_t) var1 - (int32_t) var2);
>+  return WebRtcSpl_SatW32ToW16((int32_t)var1 - (int32_t)var2);
> }
> #endif  // #if !defined(MIPS_DSP_R1_LE)
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/include/spl_inl_armv7.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/include/spl_inl_armv7.h
>index 97179f92f2a..930e91e2b35 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/include/spl_inl_armv7.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/include/spl_inl_armv7.h
>@@ -8,7 +8,6 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-
> /* This header file includes the inline functions for ARM processors in
>  * the fix point signal processing library.
>  */
>@@ -26,35 +25,37 @@
>  */
> static __inline int32_t WEBRTC_SPL_MUL_16_32_RSFT16(int16_t a, int32_t b) {
>   int32_t tmp = 0;
>-  __asm __volatile ("smulwb %0, %1, %2":"=r"(tmp):"r"(b), "r"(a));
>+  __asm __volatile("smulwb %0, %1, %2" : "=r"(tmp) : "r"(b), "r"(a));
>   return tmp;
> }
> 
> static __inline int32_t WEBRTC_SPL_MUL_16_16(int16_t a, int16_t b) {
>   int32_t tmp = 0;
>-  __asm __volatile ("smulbb %0, %1, %2":"=r"(tmp):"r"(a), "r"(b));
>+  __asm __volatile("smulbb %0, %1, %2" : "=r"(tmp) : "r"(a), "r"(b));
>   return tmp;
> }
> 
> // TODO(kma): add unit test.
> static __inline int32_t WebRtc_MulAccumW16(int16_t a, int16_t b, int32_t c) {
>   int32_t tmp = 0;
>-  __asm __volatile ("smlabb %0, %1, %2, %3":"=r"(tmp):"r"(a), "r"(b), "r"(c));
>+  __asm __volatile("smlabb %0, %1, %2, %3"
>+                   : "=r"(tmp)
>+                   : "r"(a), "r"(b), "r"(c));
>   return tmp;
> }
> 
> static __inline int16_t WebRtcSpl_AddSatW16(int16_t a, int16_t b) {
>   int32_t s_sum = 0;
> 
>-  __asm __volatile ("qadd16 %0, %1, %2":"=r"(s_sum):"r"(a), "r"(b));
>+  __asm __volatile("qadd16 %0, %1, %2" : "=r"(s_sum) : "r"(a), "r"(b));
> 
>-  return (int16_t) s_sum;
>+  return (int16_t)s_sum;
> }
> 
> static __inline int32_t WebRtcSpl_AddSatW32(int32_t l_var1, int32_t l_var2) {
>   int32_t l_sum = 0;
> 
>-  __asm __volatile ("qadd %0, %1, %2":"=r"(l_sum):"r"(l_var1), "r"(l_var2));
>+  __asm __volatile("qadd %0, %1, %2" : "=r"(l_sum) : "r"(l_var1), "r"(l_var2));
> 
>   return l_sum;
> }
>@@ -62,7 +63,7 @@ static __inline int32_t WebRtcSpl_AddSatW32(int32_t l_var1, int32_t l_var2) {
> static __inline int32_t WebRtcSpl_SubSatW32(int32_t l_var1, int32_t l_var2) {
>   int32_t l_sub = 0;
> 
>-  __asm __volatile ("qsub %0, %1, %2":"=r"(l_sub):"r"(l_var1), "r"(l_var2));
>+  __asm __volatile("qsub %0, %1, %2" : "=r"(l_sub) : "r"(l_var1), "r"(l_var2));
> 
>   return l_sub;
> }
>@@ -70,7 +71,7 @@ static __inline int32_t WebRtcSpl_SubSatW32(int32_t l_var1, int32_t l_var2) {
> static __inline int16_t WebRtcSpl_SubSatW16(int16_t var1, int16_t var2) {
>   int32_t s_sub = 0;
> 
>-  __asm __volatile ("qsub16 %0, %1, %2":"=r"(s_sub):"r"(var1), "r"(var2));
>+  __asm __volatile("qsub16 %0, %1, %2" : "=r"(s_sub) : "r"(var1), "r"(var2));
> 
>   return (int16_t)s_sub;
> }
>@@ -78,7 +79,7 @@ static __inline int16_t WebRtcSpl_SubSatW16(int16_t var1, int16_t var2) {
> static __inline int16_t WebRtcSpl_GetSizeInBits(uint32_t n) {
>   int32_t tmp = 0;
> 
>-  __asm __volatile ("clz %0, %1":"=r"(tmp):"r"(n));
>+  __asm __volatile("clz %0, %1" : "=r"(tmp) : "r"(n));
> 
>   return (int16_t)(32 - tmp);
> }
>@@ -92,7 +93,7 @@ static __inline int16_t WebRtcSpl_NormW32(int32_t a) {
>     a ^= 0xFFFFFFFF;
>   }
> 
>-  __asm __volatile ("clz %0, %1":"=r"(tmp):"r"(a));
>+  __asm __volatile("clz %0, %1" : "=r"(tmp) : "r"(a));
> 
>   return (int16_t)(tmp - 1);
> }
>@@ -100,9 +101,10 @@ static __inline int16_t WebRtcSpl_NormW32(int32_t a) {
> static __inline int16_t WebRtcSpl_NormU32(uint32_t a) {
>   int tmp = 0;
> 
>-  if (a == 0) return 0;
>+  if (a == 0)
>+    return 0;
> 
>-  __asm __volatile ("clz %0, %1":"=r"(tmp):"r"(a));
>+  __asm __volatile("clz %0, %1" : "=r"(tmp) : "r"(a));
> 
>   return (int16_t)tmp;
> }
>@@ -117,7 +119,7 @@ static __inline int16_t WebRtcSpl_NormW16(int16_t a) {
>     a_32 ^= 0xFFFFFFFF;
>   }
> 
>-  __asm __volatile ("clz %0, %1":"=r"(tmp):"r"(a_32));
>+  __asm __volatile("clz %0, %1" : "=r"(tmp) : "r"(a_32));
> 
>   return (int16_t)(tmp - 17);
> }
>@@ -126,7 +128,7 @@ static __inline int16_t WebRtcSpl_NormW16(int16_t a) {
> static __inline int16_t WebRtcSpl_SatW32ToW16(int32_t value32) {
>   int32_t out = 0;
> 
>-  __asm __volatile ("ssat %0, #16, %1" : "=r"(out) : "r"(value32));
>+  __asm __volatile("ssat %0, #16, %1" : "=r"(out) : "r"(value32));
> 
>   return (int16_t)out;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/include/spl_inl_mips.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/include/spl_inl_mips.h
>index 5819d0f86bd..1db95e8254d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/include/spl_inl_mips.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/include/spl_inl_mips.h
>@@ -8,69 +8,65 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-
> // This header file includes the inline functions in
> // the fix point signal processing library.
> 
> #ifndef COMMON_AUDIO_SIGNAL_PROCESSING_INCLUDE_SPL_INL_MIPS_H_
> #define COMMON_AUDIO_SIGNAL_PROCESSING_INCLUDE_SPL_INL_MIPS_H_
> 
>-static __inline int32_t WEBRTC_SPL_MUL_16_16(int32_t a,
>-                                             int32_t b) {
>+static __inline int32_t WEBRTC_SPL_MUL_16_16(int32_t a, int32_t b) {
>   int32_t value32 = 0;
>   int32_t a1 = 0, b1 = 0;
> 
>   __asm __volatile(
> #if defined(MIPS32_R2_LE)
>-    "seh    %[a1],          %[a]                \n\t"
>-    "seh    %[b1],          %[b]                \n\t"
>+      "seh    %[a1],          %[a]                \n\t"
>+      "seh    %[b1],          %[b]                \n\t"
> #else
>-    "sll    %[a1],          %[a],         16    \n\t"
>-    "sll    %[b1],          %[b],         16    \n\t"
>-    "sra    %[a1],          %[a1],        16    \n\t"
>-    "sra    %[b1],          %[b1],        16    \n\t"
>+      "sll    %[a1],          %[a],         16    \n\t"
>+      "sll    %[b1],          %[b],         16    \n\t"
>+      "sra    %[a1],          %[a1],        16    \n\t"
>+      "sra    %[b1],          %[b1],        16    \n\t"
> #endif
>-    "mul    %[value32],     %[a1],  %[b1]       \n\t"
>-    : [value32] "=r" (value32), [a1] "=&r" (a1), [b1] "=&r" (b1)
>-    : [a] "r" (a), [b] "r" (b)
>-    : "hi", "lo");
>+      "mul    %[value32],     %[a1],  %[b1]       \n\t"
>+      : [value32] "=r"(value32), [a1] "=&r"(a1), [b1] "=&r"(b1)
>+      : [a] "r"(a), [b] "r"(b)
>+      : "hi", "lo");
>   return value32;
> }
> 
>-static __inline int32_t WEBRTC_SPL_MUL_16_32_RSFT16(int16_t a,
>-                                                    int32_t b) {
>+static __inline int32_t WEBRTC_SPL_MUL_16_32_RSFT16(int16_t a, int32_t b) {
>   int32_t value32 = 0, b1 = 0, b2 = 0;
>   int32_t a1 = 0;
> 
>   __asm __volatile(
> #if defined(MIPS32_R2_LE)
>-    "seh    %[a1],          %[a]                        \n\t"
>+      "seh    %[a1],          %[a]                        \n\t"
> #else
>-    "sll    %[a1],          %[a],           16          \n\t"
>-    "sra    %[a1],          %[a1],          16          \n\t"
>+      "sll    %[a1],          %[a],           16          \n\t"
>+      "sra    %[a1],          %[a1],          16          \n\t"
> #endif
>-    "andi   %[b2],          %[b],           0xFFFF      \n\t"
>-    "sra    %[b1],          %[b],           16          \n\t"
>-    "sra    %[b2],          %[b2],          1           \n\t"
>-    "mul    %[value32],     %[a1],          %[b1]       \n\t"
>-    "mul    %[b2],          %[a1],          %[b2]       \n\t"
>-    "addiu  %[b2],          %[b2],          0x4000      \n\t"
>-    "sra    %[b2],          %[b2],          15          \n\t"
>-    "addu   %[value32],     %[value32],     %[b2]       \n\t"
>-    : [value32] "=&r" (value32), [b1] "=&r" (b1), [b2] "=&r" (b2),
>-      [a1] "=&r" (a1)
>-    : [a] "r" (a), [b] "r" (b)
>-    : "hi", "lo");
>+      "andi   %[b2],          %[b],           0xFFFF      \n\t"
>+      "sra    %[b1],          %[b],           16          \n\t"
>+      "sra    %[b2],          %[b2],          1           \n\t"
>+      "mul    %[value32],     %[a1],          %[b1]       \n\t"
>+      "mul    %[b2],          %[a1],          %[b2]       \n\t"
>+      "addiu  %[b2],          %[b2],          0x4000      \n\t"
>+      "sra    %[b2],          %[b2],          15          \n\t"
>+      "addu   %[value32],     %[value32],     %[b2]       \n\t"
>+      : [value32] "=&r"(value32), [b1] "=&r"(b1), [b2] "=&r"(b2), [a1] "=&r"(a1)
>+      : [a] "r"(a), [b] "r"(b)
>+      : "hi", "lo");
>   return value32;
> }
> 
> #if defined(MIPS_DSP_R1_LE)
> static __inline int16_t WebRtcSpl_SatW32ToW16(int32_t value32) {
>   __asm __volatile(
>-    "shll_s.w   %[value32], %[value32], 16      \n\t"
>-    "sra        %[value32], %[value32], 16      \n\t"
>-    : [value32] "+r" (value32)
>-    :);
>+      "shll_s.w   %[value32], %[value32], 16      \n\t"
>+      "sra        %[value32], %[value32], 16      \n\t"
>+      : [value32] "+r"(value32)
>+      :);
>   int16_t out16 = (int16_t)value32;
>   return out16;
> }
>@@ -78,10 +74,9 @@ static __inline int16_t WebRtcSpl_SatW32ToW16(int32_t value32) {
> static __inline int16_t WebRtcSpl_AddSatW16(int16_t a, int16_t b) {
>   int32_t value32 = 0;
> 
>-  __asm __volatile(
>-    "addq_s.ph      %[value32],     %[a],   %[b]    \n\t"
>-    : [value32] "=r" (value32)
>-    : [a] "r" (a), [b] "r" (b) );
>+  __asm __volatile("addq_s.ph      %[value32],     %[a],   %[b]    \n\t"
>+                   : [value32] "=r"(value32)
>+                   : [a] "r"(a), [b] "r"(b));
>   return (int16_t)value32;
> }
> 
>@@ -89,9 +84,9 @@ static __inline int32_t WebRtcSpl_AddSatW32(int32_t l_var1, int32_t l_var2) {
>   int32_t l_sum;
> 
>   __asm __volatile(
>-    "addq_s.w   %[l_sum],       %[l_var1],      %[l_var2]    \n\t"
>-    : [l_sum] "=r" (l_sum)
>-    : [l_var1] "r" (l_var1), [l_var2] "r" (l_var2) );
>+      "addq_s.w   %[l_sum],       %[l_var1],      %[l_var2]    \n\t"
>+      : [l_sum] "=r"(l_sum)
>+      : [l_var1] "r"(l_var1), [l_var2] "r"(l_var2));
> 
>   return l_sum;
> }
>@@ -99,10 +94,9 @@ static __inline int32_t WebRtcSpl_AddSatW32(int32_t l_var1, int32_t l_var2) {
> static __inline int16_t WebRtcSpl_SubSatW16(int16_t var1, int16_t var2) {
>   int32_t value32;
> 
>-  __asm __volatile(
>-    "subq_s.ph  %[value32], %[var1],    %[var2]     \n\t"
>-    : [value32] "=r" (value32)
>-    : [var1] "r" (var1), [var2] "r" (var2) );
>+  __asm __volatile("subq_s.ph  %[value32], %[var1],    %[var2]     \n\t"
>+                   : [value32] "=r"(value32)
>+                   : [var1] "r"(var1), [var2] "r"(var2));
> 
>   return (int16_t)value32;
> }
>@@ -111,9 +105,9 @@ static __inline int32_t WebRtcSpl_SubSatW32(int32_t l_var1, int32_t l_var2) {
>   int32_t l_diff;
> 
>   __asm __volatile(
>-    "subq_s.w   %[l_diff],      %[l_var1],      %[l_var2]    \n\t"
>-    : [l_diff] "=r" (l_diff)
>-    : [l_var1] "r" (l_var1), [l_var2] "r" (l_var2) );
>+      "subq_s.w   %[l_diff],      %[l_var1],      %[l_var2]    \n\t"
>+      : [l_diff] "=r"(l_diff)
>+      : [l_var1] "r"(l_var1), [l_var2] "r"(l_var2));
> 
>   return l_diff;
> }
>@@ -124,10 +118,10 @@ static __inline int16_t WebRtcSpl_GetSizeInBits(uint32_t n) {
>   int i32 = 32;
> 
>   __asm __volatile(
>-    "clz    %[bits],    %[n]                    \n\t"
>-    "subu   %[bits],    %[i32],     %[bits]     \n\t"
>-    : [bits] "=&r" (bits)
>-    : [n] "r" (n), [i32] "r" (i32) );
>+      "clz    %[bits],    %[n]                    \n\t"
>+      "subu   %[bits],    %[i32],     %[bits]     \n\t"
>+      : [bits] "=&r"(bits)
>+      : [n] "r"(n), [i32] "r"(i32));
> 
>   return (int16_t)bits;
> }
>@@ -136,20 +130,20 @@ static __inline int16_t WebRtcSpl_NormW32(int32_t a) {
>   int zeros = 0;
> 
>   __asm __volatile(
>-    ".set       push                                \n\t"
>-    ".set       noreorder                           \n\t"
>-    "bnez       %[a],       1f                      \n\t"
>-    " sra       %[zeros],   %[a],       31          \n\t"
>-    "b          2f                                  \n\t"
>-    " move      %[zeros],   $zero                   \n\t"
>-   "1:                                              \n\t"
>-    "xor        %[zeros],   %[a],       %[zeros]    \n\t"
>-    "clz        %[zeros],   %[zeros]                \n\t"
>-    "addiu      %[zeros],   %[zeros],   -1          \n\t"
>-   "2:                                              \n\t"
>-    ".set       pop                                 \n\t"
>-    : [zeros]"=&r"(zeros)
>-    : [a] "r" (a) );
>+      ".set       push                                \n\t"
>+      ".set       noreorder                           \n\t"
>+      "bnez       %[a],       1f                      \n\t"
>+      " sra       %[zeros],   %[a],       31          \n\t"
>+      "b          2f                                  \n\t"
>+      " move      %[zeros],   $zero                   \n\t"
>+      "1:                                              \n\t"
>+      "xor        %[zeros],   %[a],       %[zeros]    \n\t"
>+      "clz        %[zeros],   %[zeros]                \n\t"
>+      "addiu      %[zeros],   %[zeros],   -1          \n\t"
>+      "2:                                              \n\t"
>+      ".set       pop                                 \n\t"
>+      : [zeros] "=&r"(zeros)
>+      : [a] "r"(a));
> 
>   return (int16_t)zeros;
> }
>@@ -157,10 +151,9 @@ static __inline int16_t WebRtcSpl_NormW32(int32_t a) {
> static __inline int16_t WebRtcSpl_NormU32(uint32_t a) {
>   int zeros = 0;
> 
>-  __asm __volatile(
>-    "clz    %[zeros],   %[a]    \n\t"
>-    : [zeros] "=r" (zeros)
>-    : [a] "r" (a) );
>+  __asm __volatile("clz    %[zeros],   %[a]    \n\t"
>+                   : [zeros] "=r"(zeros)
>+                   : [a] "r"(a));
> 
>   return (int16_t)(zeros & 0x1f);
> }
>@@ -170,43 +163,41 @@ static __inline int16_t WebRtcSpl_NormW16(int16_t a) {
>   int a0 = a << 16;
> 
>   __asm __volatile(
>-    ".set       push                                \n\t"
>-    ".set       noreorder                           \n\t"
>-    "bnez       %[a0],      1f                      \n\t"
>-    " sra       %[zeros],   %[a0],      31          \n\t"
>-    "b          2f                                  \n\t"
>-    " move      %[zeros],   $zero                   \n\t"
>-   "1:                                              \n\t"
>-    "xor        %[zeros],   %[a0],      %[zeros]    \n\t"
>-    "clz        %[zeros],   %[zeros]                \n\t"
>-    "addiu      %[zeros],   %[zeros],   -1          \n\t"
>-   "2:                                              \n\t"
>-    ".set       pop                                 \n\t"
>-    : [zeros]"=&r"(zeros)
>-    : [a0] "r" (a0) );
>+      ".set       push                                \n\t"
>+      ".set       noreorder                           \n\t"
>+      "bnez       %[a0],      1f                      \n\t"
>+      " sra       %[zeros],   %[a0],      31          \n\t"
>+      "b          2f                                  \n\t"
>+      " move      %[zeros],   $zero                   \n\t"
>+      "1:                                              \n\t"
>+      "xor        %[zeros],   %[a0],      %[zeros]    \n\t"
>+      "clz        %[zeros],   %[zeros]                \n\t"
>+      "addiu      %[zeros],   %[zeros],   -1          \n\t"
>+      "2:                                              \n\t"
>+      ".set       pop                                 \n\t"
>+      : [zeros] "=&r"(zeros)
>+      : [a0] "r"(a0));
> 
>   return (int16_t)zeros;
> }
> 
>-static __inline int32_t WebRtc_MulAccumW16(int16_t a,
>-                                           int16_t b,
>-                                           int32_t c) {
>+static __inline int32_t WebRtc_MulAccumW16(int16_t a, int16_t b, int32_t c) {
>   int32_t res = 0, c1 = 0;
>   __asm __volatile(
> #if defined(MIPS32_R2_LE)
>-    "seh    %[a],       %[a]            \n\t"
>-    "seh    %[b],       %[b]            \n\t"
>+      "seh    %[a],       %[a]            \n\t"
>+      "seh    %[b],       %[b]            \n\t"
> #else
>-    "sll    %[a],       %[a],   16      \n\t"
>-    "sll    %[b],       %[b],   16      \n\t"
>-    "sra    %[a],       %[a],   16      \n\t"
>-    "sra    %[b],       %[b],   16      \n\t"
>+      "sll    %[a],       %[a],   16      \n\t"
>+      "sll    %[b],       %[b],   16      \n\t"
>+      "sra    %[a],       %[a],   16      \n\t"
>+      "sra    %[b],       %[b],   16      \n\t"
> #endif
>-    "mul    %[res],     %[a],   %[b]    \n\t"
>-    "addu   %[c1],      %[c],   %[res]  \n\t"
>-    : [c1] "=r" (c1), [res] "=&r" (res)
>-    : [a] "r" (a), [b] "r" (b), [c] "r" (c)
>-    : "hi", "lo");
>+      "mul    %[res],     %[a],   %[b]    \n\t"
>+      "addu   %[c1],      %[c],   %[res]  \n\t"
>+      : [c1] "=r"(c1), [res] "=&r"(res)
>+      : [a] "r"(a), [b] "r"(b), [c] "r"(c)
>+      : "hi", "lo");
>   return (c1);
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/real_fft_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/real_fft_unittest.cc
>index 4f5b5c72b2c..3b4312381ae 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/real_fft_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/real_fft_unittest.cc
>@@ -11,7 +11,6 @@
> #include "common_audio/signal_processing/include/real_fft.h"
> #include "common_audio/signal_processing/include/signal_processing_library.h"
> #include "test/gtest.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> namespace {
>@@ -27,17 +26,14 @@ const int kFreqDataLength = (1 << kOrder) + 2;
> const int kComplexFftDataLength = 2 << kOrder;
> // Reference data for time signal.
> const int16_t kRefData[kTimeDataLength] = {
>-  11739, 6848, -8688, 31980, -30295, 25242, 27085, 19410,
>-  -26299, 15607, -10791, 11778, -23819, 14498, -25772, 10076,
>-  1173, 6848, -8688, 31980, -30295, 2522, 27085, 19410,
>-  -2629, 5607, -3, 1178, -23819, 1498, -25772, 10076
>-};
>+    11739,  6848,  -8688,  31980, -30295, 25242, 27085,  19410,
>+    -26299, 15607, -10791, 11778, -23819, 14498, -25772, 10076,
>+    1173,   6848,  -8688,  31980, -30295, 2522,  27085,  19410,
>+    -2629,  5607,  -3,     1178,  -23819, 1498,  -25772, 10076};
> 
> class RealFFTTest : public ::testing::Test {
>  protected:
>-  RealFFTTest() {
>-    WebRtcSpl_Init();
>-  }
>+  RealFFTTest() { WebRtcSpl_Init(); }
> };
> 
> TEST_F(RealFFTTest, CreateFailsOnBadInput) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/resample_by_2_internal.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/resample_by_2_internal.h
>index b0d19691c41..145395a4cbd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/resample_by_2_internal.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/resample_by_2_internal.h
>@@ -8,7 +8,6 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-
> /*
>  * This header file contains some internal resampling functions.
>  *
>@@ -17,31 +16,45 @@
> #ifndef COMMON_AUDIO_SIGNAL_PROCESSING_RESAMPLE_BY_2_INTERNAL_H_
> #define COMMON_AUDIO_SIGNAL_PROCESSING_RESAMPLE_BY_2_INTERNAL_H_
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include <stdint.h>
> 
> /*******************************************************************
>  * resample_by_2_fast.c
>  * Functions for internal use in the other resample functions
>  ******************************************************************/
>-void WebRtcSpl_DownBy2IntToShort(int32_t *in, int32_t len, int16_t *out,
>-                                 int32_t *state);
>-
>-void WebRtcSpl_DownBy2ShortToInt(const int16_t *in, int32_t len,
>-                                 int32_t *out, int32_t *state);
>-
>-void WebRtcSpl_UpBy2ShortToInt(const int16_t *in, int32_t len,
>-                               int32_t *out, int32_t *state);
>-
>-void WebRtcSpl_UpBy2IntToInt(const int32_t *in, int32_t len, int32_t *out,
>-                             int32_t *state);
>+void WebRtcSpl_DownBy2IntToShort(int32_t* in,
>+                                 int32_t len,
>+                                 int16_t* out,
>+                                 int32_t* state);
>+
>+void WebRtcSpl_DownBy2ShortToInt(const int16_t* in,
>+                                 int32_t len,
>+                                 int32_t* out,
>+                                 int32_t* state);
>+
>+void WebRtcSpl_UpBy2ShortToInt(const int16_t* in,
>+                               int32_t len,
>+                               int32_t* out,
>+                               int32_t* state);
>+
>+void WebRtcSpl_UpBy2IntToInt(const int32_t* in,
>+                             int32_t len,
>+                             int32_t* out,
>+                             int32_t* state);
> 
>-void WebRtcSpl_UpBy2IntToShort(const int32_t *in, int32_t len,
>-                               int16_t *out, int32_t *state);
>+void WebRtcSpl_UpBy2IntToShort(const int32_t* in,
>+                               int32_t len,
>+                               int16_t* out,
>+                               int32_t* state);
> 
>-void WebRtcSpl_LPBy2ShortToInt(const int16_t* in, int32_t len,
>-                               int32_t* out, int32_t* state);
>+void WebRtcSpl_LPBy2ShortToInt(const int16_t* in,
>+                               int32_t len,
>+                               int32_t* out,
>+                               int32_t* state);
> 
>-void WebRtcSpl_LPBy2IntToInt(const int32_t* in, int32_t len, int32_t* out,
>+void WebRtcSpl_LPBy2IntToInt(const int32_t* in,
>+                             int32_t len,
>+                             int32_t* out,
>                              int32_t* state);
> 
> #endif  // COMMON_AUDIO_SIGNAL_PROCESSING_RESAMPLE_BY_2_INTERNAL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/signal_processing_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/signal_processing_unittest.cc
>index b9efe01fe7e..8e65ebd2bca 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/signal_processing_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/signal_processing_unittest.cc
>@@ -15,111 +15,113 @@
> #include "test/gtest.h"
> 
> static const size_t kVector16Size = 9;
>-static const int16_t vector16[kVector16Size] = {1, -15511, 4323, 1963,
>-  WEBRTC_SPL_WORD16_MAX, 0, WEBRTC_SPL_WORD16_MIN + 5, -3333, 345};
>+static const int16_t vector16[kVector16Size] = {1,
>+                                                -15511,
>+                                                4323,
>+                                                1963,
>+                                                WEBRTC_SPL_WORD16_MAX,
>+                                                0,
>+                                                WEBRTC_SPL_WORD16_MIN + 5,
>+                                                -3333,
>+                                                345};
> 
> class SplTest : public testing::Test {
>  protected:
>-  SplTest() {
>-    WebRtcSpl_Init();
>-  }
>-  virtual ~SplTest() {
>-  }
>+  SplTest() { WebRtcSpl_Init(); }
>+  ~SplTest() override {}
> };
> 
> TEST_F(SplTest, MacroTest) {
>-    // Macros with inputs.
>-    int A = 10;
>-    int B = 21;
>-    int a = -3;
>-    int b = WEBRTC_SPL_WORD32_MAX;
>-
>-    EXPECT_EQ(10, WEBRTC_SPL_MIN(A, B));
>-    EXPECT_EQ(21, WEBRTC_SPL_MAX(A, B));
>-
>-    EXPECT_EQ(3, WEBRTC_SPL_ABS_W16(a));
>-    EXPECT_EQ(3, WEBRTC_SPL_ABS_W32(a));
>-
>-    EXPECT_EQ(-63, WEBRTC_SPL_MUL(a, B));
>-    EXPECT_EQ(2147483651u, WEBRTC_SPL_UMUL(a, b));
>-    b = WEBRTC_SPL_WORD16_MAX >> 1;
>-    EXPECT_EQ(4294918147u, WEBRTC_SPL_UMUL_32_16(a, b));
>-    EXPECT_EQ(-49149, WEBRTC_SPL_MUL_16_U16(a, b));
>-
>-    a = b;
>-    b = -3;
>-
>-    EXPECT_EQ(-1, WEBRTC_SPL_MUL_16_32_RSFT16(a, b));
>-    EXPECT_EQ(-1, WEBRTC_SPL_MUL_16_32_RSFT15(a, b));
>-    EXPECT_EQ(-3, WEBRTC_SPL_MUL_16_32_RSFT14(a, b));
>-    EXPECT_EQ(-24, WEBRTC_SPL_MUL_16_32_RSFT11(a, b));
>-
>-    EXPECT_EQ(-12288, WEBRTC_SPL_MUL_16_16_RSFT(a, b, 2));
>-    EXPECT_EQ(-12287, WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(a, b, 2));
>-
>-    EXPECT_EQ(21, WEBRTC_SPL_SAT(a, A, B));
>-    EXPECT_EQ(21, WEBRTC_SPL_SAT(a, B, A));
>-
>-    // Shifting with negative numbers allowed
>-    int shift_amount = 1;  // Workaround compiler warning using variable here.
>-    // Positive means left shift
>-    EXPECT_EQ(32766, WEBRTC_SPL_SHIFT_W32(a, shift_amount));
>-
>-    // Shifting with negative numbers not allowed
>-    // We cannot do casting here due to signed/unsigned problem
>-    EXPECT_EQ(32766, WEBRTC_SPL_LSHIFT_W32(a, 1));
>-
>-    EXPECT_EQ(8191u, WEBRTC_SPL_RSHIFT_U32(a, 1));
>-
>-    EXPECT_EQ(1470, WEBRTC_SPL_RAND(A));
>-
>-    EXPECT_EQ(-49149, WEBRTC_SPL_MUL_16_16(a, b));
>-    EXPECT_EQ(1073676289, WEBRTC_SPL_MUL_16_16(WEBRTC_SPL_WORD16_MAX,
>-                                               WEBRTC_SPL_WORD16_MAX));
>-    EXPECT_EQ(1073709055, WEBRTC_SPL_MUL_16_32_RSFT16(WEBRTC_SPL_WORD16_MAX,
>-                                                      WEBRTC_SPL_WORD32_MAX));
>-    EXPECT_EQ(1073741824, WEBRTC_SPL_MUL_16_32_RSFT16(WEBRTC_SPL_WORD16_MIN,
>-                                                      WEBRTC_SPL_WORD32_MIN));
>+  // Macros with inputs.
>+  int A = 10;
>+  int B = 21;
>+  int a = -3;
>+  int b = WEBRTC_SPL_WORD32_MAX;
>+
>+  EXPECT_EQ(10, WEBRTC_SPL_MIN(A, B));
>+  EXPECT_EQ(21, WEBRTC_SPL_MAX(A, B));
>+
>+  EXPECT_EQ(3, WEBRTC_SPL_ABS_W16(a));
>+  EXPECT_EQ(3, WEBRTC_SPL_ABS_W32(a));
>+
>+  EXPECT_EQ(-63, WEBRTC_SPL_MUL(a, B));
>+  EXPECT_EQ(2147483651u, WEBRTC_SPL_UMUL(a, b));
>+  b = WEBRTC_SPL_WORD16_MAX >> 1;
>+  EXPECT_EQ(4294918147u, WEBRTC_SPL_UMUL_32_16(a, b));
>+  EXPECT_EQ(-49149, WEBRTC_SPL_MUL_16_U16(a, b));
>+
>+  a = b;
>+  b = -3;
>+
>+  EXPECT_EQ(-1, WEBRTC_SPL_MUL_16_32_RSFT16(a, b));
>+  EXPECT_EQ(-1, WEBRTC_SPL_MUL_16_32_RSFT15(a, b));
>+  EXPECT_EQ(-3, WEBRTC_SPL_MUL_16_32_RSFT14(a, b));
>+  EXPECT_EQ(-24, WEBRTC_SPL_MUL_16_32_RSFT11(a, b));
>+
>+  EXPECT_EQ(-12288, WEBRTC_SPL_MUL_16_16_RSFT(a, b, 2));
>+  EXPECT_EQ(-12287, WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(a, b, 2));
>+
>+  EXPECT_EQ(21, WEBRTC_SPL_SAT(a, A, B));
>+  EXPECT_EQ(21, WEBRTC_SPL_SAT(a, B, A));
>+
>+  // Shifting with negative numbers allowed
>+  int shift_amount = 1;  // Workaround compiler warning using variable here.
>+  // Positive means left shift
>+  EXPECT_EQ(32766, WEBRTC_SPL_SHIFT_W32(a, shift_amount));
>+
>+  // Shifting with negative numbers not allowed
>+  // We cannot do casting here due to signed/unsigned problem
>+  EXPECT_EQ(32766, WEBRTC_SPL_LSHIFT_W32(a, 1));
>+
>+  EXPECT_EQ(8191u, WEBRTC_SPL_RSHIFT_U32(a, 1));
>+
>+  EXPECT_EQ(1470, WEBRTC_SPL_RAND(A));
>+
>+  EXPECT_EQ(-49149, WEBRTC_SPL_MUL_16_16(a, b));
>+  EXPECT_EQ(1073676289,
>+            WEBRTC_SPL_MUL_16_16(WEBRTC_SPL_WORD16_MAX, WEBRTC_SPL_WORD16_MAX));
>+  EXPECT_EQ(1073709055, WEBRTC_SPL_MUL_16_32_RSFT16(WEBRTC_SPL_WORD16_MAX,
>+                                                    WEBRTC_SPL_WORD32_MAX));
>+  EXPECT_EQ(1073741824, WEBRTC_SPL_MUL_16_32_RSFT16(WEBRTC_SPL_WORD16_MIN,
>+                                                    WEBRTC_SPL_WORD32_MIN));
> #ifdef WEBRTC_ARCH_ARM_V7
>-    EXPECT_EQ(-1073741824,
>-              WEBRTC_SPL_MUL_16_32_RSFT16(WEBRTC_SPL_WORD16_MIN,
>-                                          WEBRTC_SPL_WORD32_MAX));
>+  EXPECT_EQ(-1073741824, WEBRTC_SPL_MUL_16_32_RSFT16(WEBRTC_SPL_WORD16_MIN,
>+                                                     WEBRTC_SPL_WORD32_MAX));
> #else
>-    EXPECT_EQ(-1073741823,
>-              WEBRTC_SPL_MUL_16_32_RSFT16(WEBRTC_SPL_WORD16_MIN,
>-                                          WEBRTC_SPL_WORD32_MAX));
>+  EXPECT_EQ(-1073741823, WEBRTC_SPL_MUL_16_32_RSFT16(WEBRTC_SPL_WORD16_MIN,
>+                                                     WEBRTC_SPL_WORD32_MAX));
> #endif
> }
> 
> TEST_F(SplTest, InlineTest) {
>-    int16_t a16 = 121;
>-    int16_t b16 = -17;
>-    int32_t a32 = 111121;
>-    int32_t b32 = -1711;
>-
>-    EXPECT_EQ(17, WebRtcSpl_GetSizeInBits(a32));
>-
>-    EXPECT_EQ(0, WebRtcSpl_NormW32(0));
>-    EXPECT_EQ(31, WebRtcSpl_NormW32(-1));
>-    EXPECT_EQ(0, WebRtcSpl_NormW32(WEBRTC_SPL_WORD32_MIN));
>-    EXPECT_EQ(14, WebRtcSpl_NormW32(a32));
>-
>-    EXPECT_EQ(0, WebRtcSpl_NormW16(0));
>-    EXPECT_EQ(15, WebRtcSpl_NormW16(-1));
>-    EXPECT_EQ(0, WebRtcSpl_NormW16(WEBRTC_SPL_WORD16_MIN));
>-    EXPECT_EQ(4, WebRtcSpl_NormW16(b32));
>-    for (int ii = 0; ii < 15; ++ii) {
>-      int16_t value = 1 << ii;
>-      EXPECT_EQ(14 - ii, WebRtcSpl_NormW16(value));
>-      EXPECT_EQ(15 - ii, WebRtcSpl_NormW16(-value));
>-    }
>+  int16_t a16 = 121;
>+  int16_t b16 = -17;
>+  int32_t a32 = 111121;
>+  int32_t b32 = -1711;
>+
>+  EXPECT_EQ(17, WebRtcSpl_GetSizeInBits(a32));
>+
>+  EXPECT_EQ(0, WebRtcSpl_NormW32(0));
>+  EXPECT_EQ(31, WebRtcSpl_NormW32(-1));
>+  EXPECT_EQ(0, WebRtcSpl_NormW32(WEBRTC_SPL_WORD32_MIN));
>+  EXPECT_EQ(14, WebRtcSpl_NormW32(a32));
>+
>+  EXPECT_EQ(0, WebRtcSpl_NormW16(0));
>+  EXPECT_EQ(15, WebRtcSpl_NormW16(-1));
>+  EXPECT_EQ(0, WebRtcSpl_NormW16(WEBRTC_SPL_WORD16_MIN));
>+  EXPECT_EQ(4, WebRtcSpl_NormW16(b32));
>+  for (int ii = 0; ii < 15; ++ii) {
>+    int16_t value = 1 << ii;
>+    EXPECT_EQ(14 - ii, WebRtcSpl_NormW16(value));
>+    EXPECT_EQ(15 - ii, WebRtcSpl_NormW16(-value));
>+  }
> 
>-    EXPECT_EQ(0, WebRtcSpl_NormU32(0u));
>-    EXPECT_EQ(0, WebRtcSpl_NormU32(0xffffffff));
>-    EXPECT_EQ(15, WebRtcSpl_NormU32(static_cast<uint32_t>(a32)));
>+  EXPECT_EQ(0, WebRtcSpl_NormU32(0u));
>+  EXPECT_EQ(0, WebRtcSpl_NormU32(0xffffffff));
>+  EXPECT_EQ(15, WebRtcSpl_NormU32(static_cast<uint32_t>(a32)));
> 
>-    EXPECT_EQ(104, WebRtcSpl_AddSatW16(a16, b16));
>-    EXPECT_EQ(138, WebRtcSpl_SubSatW16(a16, b16));
>+  EXPECT_EQ(104, WebRtcSpl_AddSatW16(a16, b16));
>+  EXPECT_EQ(138, WebRtcSpl_SubSatW16(a16, b16));
> }
> 
> TEST_F(SplTest, AddSubSatW32) {
>@@ -168,84 +170,83 @@ TEST_F(SplTest, CountLeadingZeros64) {
> }
> 
> TEST_F(SplTest, MathOperationsTest) {
>-    int A = 1134567892;
>-    int32_t num = 117;
>-    int32_t den = -5;
>-    uint16_t denU = 5;
>-    EXPECT_EQ(33700, WebRtcSpl_Sqrt(A));
>-    EXPECT_EQ(33683, WebRtcSpl_SqrtFloor(A));
>-
>-
>-    EXPECT_EQ(-91772805, WebRtcSpl_DivResultInQ31(den, num));
>-    EXPECT_EQ(-23, WebRtcSpl_DivW32W16ResW16(num, (int16_t)den));
>-    EXPECT_EQ(-23, WebRtcSpl_DivW32W16(num, (int16_t)den));
>-    EXPECT_EQ(23u, WebRtcSpl_DivU32U16(num, denU));
>-    EXPECT_EQ(0, WebRtcSpl_DivW32HiLow(128, 0, 256));
>+  int A = 1134567892;
>+  int32_t num = 117;
>+  int32_t den = -5;
>+  uint16_t denU = 5;
>+  EXPECT_EQ(33700, WebRtcSpl_Sqrt(A));
>+  EXPECT_EQ(33683, WebRtcSpl_SqrtFloor(A));
>+
>+  EXPECT_EQ(-91772805, WebRtcSpl_DivResultInQ31(den, num));
>+  EXPECT_EQ(-23, WebRtcSpl_DivW32W16ResW16(num, (int16_t)den));
>+  EXPECT_EQ(-23, WebRtcSpl_DivW32W16(num, (int16_t)den));
>+  EXPECT_EQ(23u, WebRtcSpl_DivU32U16(num, denU));
>+  EXPECT_EQ(0, WebRtcSpl_DivW32HiLow(128, 0, 256));
> }
> 
> TEST_F(SplTest, BasicArrayOperationsTest) {
>-    const size_t kVectorSize = 4;
>-    int B[] = {4, 12, 133, 1100};
>-    int16_t b16[kVectorSize];
>-    int32_t b32[kVectorSize];
>+  const size_t kVectorSize = 4;
>+  int B[] = {4, 12, 133, 1100};
>+  int16_t b16[kVectorSize];
>+  int32_t b32[kVectorSize];
> 
>-    int16_t bTmp16[kVectorSize];
>-    int32_t bTmp32[kVectorSize];
>+  int16_t bTmp16[kVectorSize];
>+  int32_t bTmp32[kVectorSize];
> 
>-    WebRtcSpl_MemSetW16(b16, 3, kVectorSize);
>-    for (size_t kk = 0; kk < kVectorSize; ++kk) {
>-        EXPECT_EQ(3, b16[kk]);
>-    }
>-    WebRtcSpl_ZerosArrayW16(b16, kVectorSize);
>-    for (size_t kk = 0; kk < kVectorSize; ++kk) {
>-        EXPECT_EQ(0, b16[kk]);
>-    }
>-    WebRtcSpl_MemSetW32(b32, 3, kVectorSize);
>-    for (size_t kk = 0; kk < kVectorSize; ++kk) {
>-        EXPECT_EQ(3, b32[kk]);
>-    }
>-    WebRtcSpl_ZerosArrayW32(b32, kVectorSize);
>-    for (size_t kk = 0; kk < kVectorSize; ++kk) {
>-        EXPECT_EQ(0, b32[kk]);
>-    }
>-    for (size_t kk = 0; kk < kVectorSize; ++kk) {
>-        bTmp16[kk] = (int16_t)kk;
>-        bTmp32[kk] = (int32_t)kk;
>-    }
>-    WEBRTC_SPL_MEMCPY_W16(b16, bTmp16, kVectorSize);
>-    for (size_t kk = 0; kk < kVectorSize; ++kk) {
>-        EXPECT_EQ(b16[kk], bTmp16[kk]);
>-    }
>-//    WEBRTC_SPL_MEMCPY_W32(b32, bTmp32, kVectorSize);
>-//    for (int kk = 0; kk < kVectorSize; ++kk) {
>-//        EXPECT_EQ(b32[kk], bTmp32[kk]);
>-//    }
>-    WebRtcSpl_CopyFromEndW16(b16, kVectorSize, 2, bTmp16);
>-    for (size_t kk = 0; kk < 2; ++kk) {
>-        EXPECT_EQ(static_cast<int16_t>(kk+2), bTmp16[kk]);
>-    }
>+  WebRtcSpl_MemSetW16(b16, 3, kVectorSize);
>+  for (size_t kk = 0; kk < kVectorSize; ++kk) {
>+    EXPECT_EQ(3, b16[kk]);
>+  }
>+  WebRtcSpl_ZerosArrayW16(b16, kVectorSize);
>+  for (size_t kk = 0; kk < kVectorSize; ++kk) {
>+    EXPECT_EQ(0, b16[kk]);
>+  }
>+  WebRtcSpl_MemSetW32(b32, 3, kVectorSize);
>+  for (size_t kk = 0; kk < kVectorSize; ++kk) {
>+    EXPECT_EQ(3, b32[kk]);
>+  }
>+  WebRtcSpl_ZerosArrayW32(b32, kVectorSize);
>+  for (size_t kk = 0; kk < kVectorSize; ++kk) {
>+    EXPECT_EQ(0, b32[kk]);
>+  }
>+  for (size_t kk = 0; kk < kVectorSize; ++kk) {
>+    bTmp16[kk] = (int16_t)kk;
>+    bTmp32[kk] = (int32_t)kk;
>+  }
>+  WEBRTC_SPL_MEMCPY_W16(b16, bTmp16, kVectorSize);
>+  for (size_t kk = 0; kk < kVectorSize; ++kk) {
>+    EXPECT_EQ(b16[kk], bTmp16[kk]);
>+  }
>+  //    WEBRTC_SPL_MEMCPY_W32(b32, bTmp32, kVectorSize);
>+  //    for (int kk = 0; kk < kVectorSize; ++kk) {
>+  //        EXPECT_EQ(b32[kk], bTmp32[kk]);
>+  //    }
>+  WebRtcSpl_CopyFromEndW16(b16, kVectorSize, 2, bTmp16);
>+  for (size_t kk = 0; kk < 2; ++kk) {
>+    EXPECT_EQ(static_cast<int16_t>(kk + 2), bTmp16[kk]);
>+  }
> 
>-    for (size_t kk = 0; kk < kVectorSize; ++kk) {
>-        b32[kk] = B[kk];
>-        b16[kk] = (int16_t)B[kk];
>-    }
>-    WebRtcSpl_VectorBitShiftW32ToW16(bTmp16, kVectorSize, b32, 1);
>-    for (size_t kk = 0; kk < kVectorSize; ++kk) {
>-        EXPECT_EQ((B[kk]>>1), bTmp16[kk]);
>-    }
>-    WebRtcSpl_VectorBitShiftW16(bTmp16, kVectorSize, b16, 1);
>-    for (size_t kk = 0; kk < kVectorSize; ++kk) {
>-        EXPECT_EQ((B[kk]>>1), bTmp16[kk]);
>-    }
>-    WebRtcSpl_VectorBitShiftW32(bTmp32, kVectorSize, b32, 1);
>-    for (size_t kk = 0; kk < kVectorSize; ++kk) {
>-        EXPECT_EQ((B[kk]>>1), bTmp32[kk]);
>-    }
>+  for (size_t kk = 0; kk < kVectorSize; ++kk) {
>+    b32[kk] = B[kk];
>+    b16[kk] = (int16_t)B[kk];
>+  }
>+  WebRtcSpl_VectorBitShiftW32ToW16(bTmp16, kVectorSize, b32, 1);
>+  for (size_t kk = 0; kk < kVectorSize; ++kk) {
>+    EXPECT_EQ((B[kk] >> 1), bTmp16[kk]);
>+  }
>+  WebRtcSpl_VectorBitShiftW16(bTmp16, kVectorSize, b16, 1);
>+  for (size_t kk = 0; kk < kVectorSize; ++kk) {
>+    EXPECT_EQ((B[kk] >> 1), bTmp16[kk]);
>+  }
>+  WebRtcSpl_VectorBitShiftW32(bTmp32, kVectorSize, b32, 1);
>+  for (size_t kk = 0; kk < kVectorSize; ++kk) {
>+    EXPECT_EQ((B[kk] >> 1), bTmp32[kk]);
>+  }
> 
>-    WebRtcSpl_MemCpyReversedOrder(&bTmp16[3], b16, kVectorSize);
>-    for (size_t kk = 0; kk < kVectorSize; ++kk) {
>-        EXPECT_EQ(b16[3-kk], bTmp16[kk]);
>-    }
>+  WebRtcSpl_MemCpyReversedOrder(&bTmp16[3], b16, kVectorSize);
>+  for (size_t kk = 0; kk < kVectorSize; ++kk) {
>+    EXPECT_EQ(b16[3 - kk], bTmp16[kk]);
>+  }
> }
> 
> TEST_F(SplTest, MinMaxOperationsTest) {
>@@ -253,12 +254,40 @@ TEST_F(SplTest, MinMaxOperationsTest) {
> 
>   // Vectors to test the cases where minimum values have to be caught
>   // outside of the unrolled loops in ARM-Neon.
>-  int16_t vector16[kVectorSize] = {-1, 7485, 0, 3333,
>-      -18283, 0, 12334, -29871, 988, -3333,
>-      345, -456, 222, 999,  888, 8774, WEBRTC_SPL_WORD16_MIN};
>-  int32_t vector32[kVectorSize] = {-1, 0, 283211, 3333,
>-      8712345, 0, -3333, 89345, -374585456, 222, 999, 122345334,
>-      -12389756, -987329871, 888, -2, WEBRTC_SPL_WORD32_MIN};
>+  int16_t vector16[kVectorSize] = {-1,
>+                                   7485,
>+                                   0,
>+                                   3333,
>+                                   -18283,
>+                                   0,
>+                                   12334,
>+                                   -29871,
>+                                   988,
>+                                   -3333,
>+                                   345,
>+                                   -456,
>+                                   222,
>+                                   999,
>+                                   888,
>+                                   8774,
>+                                   WEBRTC_SPL_WORD16_MIN};
>+  int32_t vector32[kVectorSize] = {-1,
>+                                   0,
>+                                   283211,
>+                                   3333,
>+                                   8712345,
>+                                   0,
>+                                   -3333,
>+                                   89345,
>+                                   -374585456,
>+                                   222,
>+                                   999,
>+                                   122345334,
>+                                   -12389756,
>+                                   -987329871,
>+                                   888,
>+                                   -2,
>+                                   WEBRTC_SPL_WORD32_MIN};
> 
>   EXPECT_EQ(WEBRTC_SPL_WORD16_MIN,
>             WebRtcSpl_MinValueW16(vector16, kVectorSize));
>@@ -312,76 +341,75 @@ TEST_F(SplTest, MinMaxOperationsTest) {
> }
> 
> TEST_F(SplTest, VectorOperationsTest) {
>-    const size_t kVectorSize = 4;
>-    int B[] = {4, 12, 133, 1100};
>-    int16_t a16[kVectorSize];
>-    int16_t b16[kVectorSize];
>-    int16_t bTmp16[kVectorSize];
>-
>-    for (size_t kk = 0; kk < kVectorSize; ++kk) {
>-        a16[kk] = B[kk];
>-        b16[kk] = B[kk];
>-    }
>+  const size_t kVectorSize = 4;
>+  int B[] = {4, 12, 133, 1100};
>+  int16_t a16[kVectorSize];
>+  int16_t b16[kVectorSize];
>+  int16_t bTmp16[kVectorSize];
>+
>+  for (size_t kk = 0; kk < kVectorSize; ++kk) {
>+    a16[kk] = B[kk];
>+    b16[kk] = B[kk];
>+  }
> 
>-    WebRtcSpl_AffineTransformVector(bTmp16, b16, 3, 7, 2, kVectorSize);
>-    for (size_t kk = 0; kk < kVectorSize; ++kk) {
>-        EXPECT_EQ((B[kk]*3+7)>>2, bTmp16[kk]);
>-    }
>-    WebRtcSpl_ScaleAndAddVectorsWithRound(b16, 3, b16, 2, 2, bTmp16,
>-                                          kVectorSize);
>-    for (size_t kk = 0; kk < kVectorSize; ++kk) {
>-        EXPECT_EQ((B[kk]*3+B[kk]*2+2)>>2, bTmp16[kk]);
>-    }
>+  WebRtcSpl_AffineTransformVector(bTmp16, b16, 3, 7, 2, kVectorSize);
>+  for (size_t kk = 0; kk < kVectorSize; ++kk) {
>+    EXPECT_EQ((B[kk] * 3 + 7) >> 2, bTmp16[kk]);
>+  }
>+  WebRtcSpl_ScaleAndAddVectorsWithRound(b16, 3, b16, 2, 2, bTmp16, kVectorSize);
>+  for (size_t kk = 0; kk < kVectorSize; ++kk) {
>+    EXPECT_EQ((B[kk] * 3 + B[kk] * 2 + 2) >> 2, bTmp16[kk]);
>+  }
> 
>-    WebRtcSpl_AddAffineVectorToVector(bTmp16, b16, 3, 7, 2, kVectorSize);
>-    for (size_t kk = 0; kk < kVectorSize; ++kk) {
>-        EXPECT_EQ(((B[kk]*3+B[kk]*2+2)>>2)+((b16[kk]*3+7)>>2), bTmp16[kk]);
>-    }
>+  WebRtcSpl_AddAffineVectorToVector(bTmp16, b16, 3, 7, 2, kVectorSize);
>+  for (size_t kk = 0; kk < kVectorSize; ++kk) {
>+    EXPECT_EQ(((B[kk] * 3 + B[kk] * 2 + 2) >> 2) + ((b16[kk] * 3 + 7) >> 2),
>+              bTmp16[kk]);
>+  }
> 
>-    WebRtcSpl_ScaleVector(b16, bTmp16, 13, kVectorSize, 2);
>-    for (size_t kk = 0; kk < kVectorSize; ++kk) {
>-        EXPECT_EQ((b16[kk]*13)>>2, bTmp16[kk]);
>-    }
>-    WebRtcSpl_ScaleVectorWithSat(b16, bTmp16, 13, kVectorSize, 2);
>-    for (size_t kk = 0; kk < kVectorSize; ++kk) {
>-        EXPECT_EQ((b16[kk]*13)>>2, bTmp16[kk]);
>-    }
>-    WebRtcSpl_ScaleAndAddVectors(a16, 13, 2, b16, 7, 2, bTmp16, kVectorSize);
>-    for (size_t kk = 0; kk < kVectorSize; ++kk) {
>-        EXPECT_EQ(((a16[kk]*13)>>2)+((b16[kk]*7)>>2), bTmp16[kk]);
>-    }
>+  WebRtcSpl_ScaleVector(b16, bTmp16, 13, kVectorSize, 2);
>+  for (size_t kk = 0; kk < kVectorSize; ++kk) {
>+    EXPECT_EQ((b16[kk] * 13) >> 2, bTmp16[kk]);
>+  }
>+  WebRtcSpl_ScaleVectorWithSat(b16, bTmp16, 13, kVectorSize, 2);
>+  for (size_t kk = 0; kk < kVectorSize; ++kk) {
>+    EXPECT_EQ((b16[kk] * 13) >> 2, bTmp16[kk]);
>+  }
>+  WebRtcSpl_ScaleAndAddVectors(a16, 13, 2, b16, 7, 2, bTmp16, kVectorSize);
>+  for (size_t kk = 0; kk < kVectorSize; ++kk) {
>+    EXPECT_EQ(((a16[kk] * 13) >> 2) + ((b16[kk] * 7) >> 2), bTmp16[kk]);
>+  }
> 
>-    WebRtcSpl_AddVectorsAndShift(bTmp16, a16, b16, kVectorSize, 2);
>-    for (size_t kk = 0; kk < kVectorSize; ++kk) {
>-        EXPECT_EQ(B[kk] >> 1, bTmp16[kk]);
>-    }
>-    WebRtcSpl_ReverseOrderMultArrayElements(bTmp16, a16, &b16[3],
>-                                            kVectorSize, 2);
>-    for (size_t kk = 0; kk < kVectorSize; ++kk) {
>-        EXPECT_EQ((a16[kk]*b16[3-kk])>>2, bTmp16[kk]);
>-    }
>-    WebRtcSpl_ElementwiseVectorMult(bTmp16, a16, b16, kVectorSize, 6);
>-    for (size_t kk = 0; kk < kVectorSize; ++kk) {
>-        EXPECT_EQ((a16[kk]*b16[kk])>>6, bTmp16[kk]);
>-    }
>+  WebRtcSpl_AddVectorsAndShift(bTmp16, a16, b16, kVectorSize, 2);
>+  for (size_t kk = 0; kk < kVectorSize; ++kk) {
>+    EXPECT_EQ(B[kk] >> 1, bTmp16[kk]);
>+  }
>+  WebRtcSpl_ReverseOrderMultArrayElements(bTmp16, a16, &b16[3], kVectorSize, 2);
>+  for (size_t kk = 0; kk < kVectorSize; ++kk) {
>+    EXPECT_EQ((a16[kk] * b16[3 - kk]) >> 2, bTmp16[kk]);
>+  }
>+  WebRtcSpl_ElementwiseVectorMult(bTmp16, a16, b16, kVectorSize, 6);
>+  for (size_t kk = 0; kk < kVectorSize; ++kk) {
>+    EXPECT_EQ((a16[kk] * b16[kk]) >> 6, bTmp16[kk]);
>+  }
> 
>-    WebRtcSpl_SqrtOfOneMinusXSquared(b16, kVectorSize, bTmp16);
>-    for (size_t kk = 0; kk < kVectorSize - 1; ++kk) {
>-        EXPECT_EQ(32767, bTmp16[kk]);
>-    }
>-    EXPECT_EQ(32749, bTmp16[kVectorSize - 1]);
>+  WebRtcSpl_SqrtOfOneMinusXSquared(b16, kVectorSize, bTmp16);
>+  for (size_t kk = 0; kk < kVectorSize - 1; ++kk) {
>+    EXPECT_EQ(32767, bTmp16[kk]);
>+  }
>+  EXPECT_EQ(32749, bTmp16[kVectorSize - 1]);
> 
>-    EXPECT_EQ(0, WebRtcSpl_GetScalingSquare(b16, kVectorSize, 1));
>+  EXPECT_EQ(0, WebRtcSpl_GetScalingSquare(b16, kVectorSize, 1));
> }
> 
> TEST_F(SplTest, EstimatorsTest) {
>   const size_t kOrder = 2;
>-  const int32_t unstable_filter[] = { 4, 12, 133, 1100 };
>-  const int32_t stable_filter[] = { 1100, 133, 12, 4 };
>-  int16_t lpc[kOrder + 2] = { 0 };
>-  int16_t refl[kOrder + 2] = { 0 };
>-  int16_t lpc_result[] = { 4096, -497, 15, 0 };
>-  int16_t refl_result[] = { -3962, 123, 0, 0 };
>+  const int32_t unstable_filter[] = {4, 12, 133, 1100};
>+  const int32_t stable_filter[] = {1100, 133, 12, 4};
>+  int16_t lpc[kOrder + 2] = {0};
>+  int16_t refl[kOrder + 2] = {0};
>+  int16_t lpc_result[] = {4096, -497, 15, 0};
>+  int16_t refl_result[] = {-3962, 123, 0, 0};
> 
>   EXPECT_EQ(0, WebRtcSpl_LevinsonDurbin(unstable_filter, lpc, refl, kOrder));
>   EXPECT_EQ(1, WebRtcSpl_LevinsonDurbin(stable_filter, lpc, refl, kOrder));
>@@ -392,69 +420,61 @@ TEST_F(SplTest, EstimatorsTest) {
> }
> 
> TEST_F(SplTest, FilterTest) {
>-    const size_t kVectorSize = 4;
>-    const size_t kFilterOrder = 3;
>-    int16_t A[] = {1, 2, 33, 100};
>-    int16_t A5[] = {1, 2, 33, 100, -5};
>-    int16_t B[] = {4, 12, 133, 110};
>-    int16_t data_in[kVectorSize];
>-    int16_t data_out[kVectorSize];
>-    int16_t bTmp16Low[kVectorSize];
>-    int16_t bState[kVectorSize];
>-    int16_t bStateLow[kVectorSize];
>-
>-    WebRtcSpl_ZerosArrayW16(bState, kVectorSize);
>-    WebRtcSpl_ZerosArrayW16(bStateLow, kVectorSize);
>-
>-    for (size_t kk = 0; kk < kVectorSize; ++kk) {
>-        data_in[kk] = A[kk];
>-        data_out[kk] = 0;
>-    }
>+  const size_t kVectorSize = 4;
>+  const size_t kFilterOrder = 3;
>+  int16_t A[] = {1, 2, 33, 100};
>+  int16_t A5[] = {1, 2, 33, 100, -5};
>+  int16_t B[] = {4, 12, 133, 110};
>+  int16_t data_in[kVectorSize];
>+  int16_t data_out[kVectorSize];
>+  int16_t bTmp16Low[kVectorSize];
>+  int16_t bState[kVectorSize];
>+  int16_t bStateLow[kVectorSize];
>+
>+  WebRtcSpl_ZerosArrayW16(bState, kVectorSize);
>+  WebRtcSpl_ZerosArrayW16(bStateLow, kVectorSize);
>+
>+  for (size_t kk = 0; kk < kVectorSize; ++kk) {
>+    data_in[kk] = A[kk];
>+    data_out[kk] = 0;
>+  }
> 
>-    // MA filters.
>-    // Note that the input data has |kFilterOrder| states before the actual
>-    // data (one sample).
>-    WebRtcSpl_FilterMAFastQ12(&data_in[kFilterOrder], data_out, B,
>-                              kFilterOrder + 1, 1);
>-    EXPECT_EQ(0, data_out[0]);
>-    // AR filters.
>-    // Note that the output data has |kFilterOrder| states before the actual
>-    // data (one sample).
>-    WebRtcSpl_FilterARFastQ12(data_in, &data_out[kFilterOrder], A,
>-                              kFilterOrder + 1, 1);
>-    EXPECT_EQ(0, data_out[kFilterOrder]);
>-
>-    EXPECT_EQ(kVectorSize, WebRtcSpl_FilterAR(A5,
>-                                              5,
>-                                              data_in,
>-                                              kVectorSize,
>-                                              bState,
>-                                              kVectorSize,
>-                                              bStateLow,
>-                                              kVectorSize,
>-                                              data_out,
>-                                              bTmp16Low,
>-                                              kVectorSize));
>+  // MA filters.
>+  // Note that the input data has |kFilterOrder| states before the actual
>+  // data (one sample).
>+  WebRtcSpl_FilterMAFastQ12(&data_in[kFilterOrder], data_out, B,
>+                            kFilterOrder + 1, 1);
>+  EXPECT_EQ(0, data_out[0]);
>+  // AR filters.
>+  // Note that the output data has |kFilterOrder| states before the actual
>+  // data (one sample).
>+  WebRtcSpl_FilterARFastQ12(data_in, &data_out[kFilterOrder], A,
>+                            kFilterOrder + 1, 1);
>+  EXPECT_EQ(0, data_out[kFilterOrder]);
>+
>+  EXPECT_EQ(kVectorSize, WebRtcSpl_FilterAR(A5, 5, data_in, kVectorSize, bState,
>+                                            kVectorSize, bStateLow, kVectorSize,
>+                                            data_out, bTmp16Low, kVectorSize));
> }
> 
> TEST_F(SplTest, RandTest) {
>-    const int kVectorSize = 4;
>-    int16_t BU[] = {3653, 12446, 8525, 30691};
>-    int16_t b16[kVectorSize];
>-    uint32_t bSeed = 100000;
>-
>-    EXPECT_EQ(7086, WebRtcSpl_RandU(&bSeed));
>-    EXPECT_EQ(31565, WebRtcSpl_RandU(&bSeed));
>-    EXPECT_EQ(-9786, WebRtcSpl_RandN(&bSeed));
>-    EXPECT_EQ(kVectorSize, WebRtcSpl_RandUArray(b16, kVectorSize, &bSeed));
>-    for (int kk = 0; kk < kVectorSize; ++kk) {
>-        EXPECT_EQ(BU[kk], b16[kk]);
>-    }
>+  const int kVectorSize = 4;
>+  int16_t BU[] = {3653, 12446, 8525, 30691};
>+  int16_t b16[kVectorSize];
>+  uint32_t bSeed = 100000;
>+
>+  EXPECT_EQ(7086, WebRtcSpl_RandU(&bSeed));
>+  EXPECT_EQ(31565, WebRtcSpl_RandU(&bSeed));
>+  EXPECT_EQ(-9786, WebRtcSpl_RandN(&bSeed));
>+  EXPECT_EQ(kVectorSize, WebRtcSpl_RandUArray(b16, kVectorSize, &bSeed));
>+  for (int kk = 0; kk < kVectorSize; ++kk) {
>+    EXPECT_EQ(BU[kk], b16[kk]);
>+  }
> }
> 
> TEST_F(SplTest, DotProductWithScaleTest) {
>-  EXPECT_EQ(605362796, WebRtcSpl_DotProductWithScale(vector16,
>-      vector16, kVector16Size, 2));
>+  EXPECT_EQ(605362796, WebRtcSpl_DotProductWithScale(vector16, vector16,
>+                                                     kVector16Size, 2));
> }
> 
> TEST_F(SplTest, CrossCorrelationTest) {
>@@ -464,8 +484,9 @@ TEST_F(SplTest, CrossCorrelationTest) {
>   const int kStep = 1;
>   const size_t kSeqDimension = 6;
> 
>-  const int16_t kVector16[kVector16Size] = {1, 4323, 1963,
>-    WEBRTC_SPL_WORD16_MAX, WEBRTC_SPL_WORD16_MIN + 5, -3333, -876, 8483, 142};
>+  const int16_t kVector16[kVector16Size] = {
>+      1,    4323, 1963, WEBRTC_SPL_WORD16_MAX, WEBRTC_SPL_WORD16_MIN + 5, -3333,
>+      -876, 8483, 142};
>   int32_t vector32[kCrossCorrelationDimension] = {0};
> 
>   WebRtcSpl_CrossCorrelation(vector32, vector16, kVector16, kSeqDimension,
>@@ -473,12 +494,12 @@ TEST_F(SplTest, CrossCorrelationTest) {
> 
>   // WebRtcSpl_CrossCorrelationC() and WebRtcSpl_CrossCorrelationNeon()
>   // are not bit-exact.
>-  const int32_t kExpected[kCrossCorrelationDimension] =
>-      {-266947903, -15579555, -171282001};
>+  const int32_t kExpected[kCrossCorrelationDimension] = {-266947903, -15579555,
>+                                                         -171282001};
>   const int32_t* expected = kExpected;
> #if !defined(MIPS32_LE)
>-  const int32_t kExpectedNeon[kCrossCorrelationDimension] =
>-      {-266947901, -15579553, -171281999};
>+  const int32_t kExpectedNeon[kCrossCorrelationDimension] = {
>+      -266947901, -15579553, -171281999};
>   if (WebRtcSpl_CrossCorrelation != WebRtcSpl_CrossCorrelationC) {
>     expected = kExpectedNeon;
>   }
>@@ -491,8 +512,9 @@ TEST_F(SplTest, CrossCorrelationTest) {
> TEST_F(SplTest, AutoCorrelationTest) {
>   int scale = 0;
>   int32_t vector32[kVector16Size];
>-  const int32_t expected[kVector16Size] = {302681398, 14223410, -121705063,
>-    -85221647, -17104971, 61806945, 6644603, -669329, 43};
>+  const int32_t expected[kVector16Size] = {302681398, 14223410,  -121705063,
>+                                           -85221647, -17104971, 61806945,
>+                                           6644603,   -669329,   43};
> 
>   EXPECT_EQ(kVector16Size,
>             WebRtcSpl_AutoCorrelation(vector16, kVector16Size,
>@@ -504,63 +526,60 @@ TEST_F(SplTest, AutoCorrelationTest) {
> }
> 
> TEST_F(SplTest, SignalProcessingTest) {
>-    const size_t kVectorSize = 4;
>-    int A[] = {1, 2, 33, 100};
>-    const int16_t kHanning[4] = { 2399, 8192, 13985, 16384 };
>-    int16_t b16[kVectorSize];
>+  const size_t kVectorSize = 4;
>+  int A[] = {1, 2, 33, 100};
>+  const int16_t kHanning[4] = {2399, 8192, 13985, 16384};
>+  int16_t b16[kVectorSize];
> 
>-    int16_t bTmp16[kVectorSize];
>+  int16_t bTmp16[kVectorSize];
> 
>-    int bScale = 0;
>+  int bScale = 0;
> 
>-    for (size_t kk = 0; kk < kVectorSize; ++kk) {
>-        b16[kk] = A[kk];
>-    }
>+  for (size_t kk = 0; kk < kVectorSize; ++kk) {
>+    b16[kk] = A[kk];
>+  }
> 
>-    // TODO(bjornv): Activate the Reflection Coefficient tests when refactoring.
>-//    WebRtcSpl_ReflCoefToLpc(b16, kVectorSize, bTmp16);
>-////    for (int kk = 0; kk < kVectorSize; ++kk) {
>-////        EXPECT_EQ(aTmp16[kk], bTmp16[kk]);
>-////    }
>-//    WebRtcSpl_LpcToReflCoef(bTmp16, kVectorSize, b16);
>-////    for (int kk = 0; kk < kVectorSize; ++kk) {
>-////        EXPECT_EQ(a16[kk], b16[kk]);
>-////    }
>-//    WebRtcSpl_AutoCorrToReflCoef(b32, kVectorSize, bTmp16);
>-////    for (int kk = 0; kk < kVectorSize; ++kk) {
>-////        EXPECT_EQ(aTmp16[kk], bTmp16[kk]);
>-////    }
>-
>-    WebRtcSpl_GetHanningWindow(bTmp16, kVectorSize);
>-    for (size_t kk = 0; kk < kVectorSize; ++kk) {
>-        EXPECT_EQ(kHanning[kk], bTmp16[kk]);
>-    }
>+  // TODO(bjornv): Activate the Reflection Coefficient tests when refactoring.
>+  //    WebRtcSpl_ReflCoefToLpc(b16, kVectorSize, bTmp16);
>+  ////    for (int kk = 0; kk < kVectorSize; ++kk) {
>+  ////        EXPECT_EQ(aTmp16[kk], bTmp16[kk]);
>+  ////    }
>+  //    WebRtcSpl_LpcToReflCoef(bTmp16, kVectorSize, b16);
>+  ////    for (int kk = 0; kk < kVectorSize; ++kk) {
>+  ////        EXPECT_EQ(a16[kk], b16[kk]);
>+  ////    }
>+  //    WebRtcSpl_AutoCorrToReflCoef(b32, kVectorSize, bTmp16);
>+  ////    for (int kk = 0; kk < kVectorSize; ++kk) {
>+  ////        EXPECT_EQ(aTmp16[kk], bTmp16[kk]);
>+  ////    }
>+
>+  WebRtcSpl_GetHanningWindow(bTmp16, kVectorSize);
>+  for (size_t kk = 0; kk < kVectorSize; ++kk) {
>+    EXPECT_EQ(kHanning[kk], bTmp16[kk]);
>+  }
> 
>-    for (size_t kk = 0; kk < kVectorSize; ++kk) {
>-        b16[kk] = A[kk];
>-    }
>-    EXPECT_EQ(11094 , WebRtcSpl_Energy(b16, kVectorSize, &bScale));
>-    EXPECT_EQ(0, bScale);
>+  for (size_t kk = 0; kk < kVectorSize; ++kk) {
>+    b16[kk] = A[kk];
>+  }
>+  EXPECT_EQ(11094, WebRtcSpl_Energy(b16, kVectorSize, &bScale));
>+  EXPECT_EQ(0, bScale);
> }
> 
> TEST_F(SplTest, FFTTest) {
>-    int16_t B[] = {1, 2, 33, 100,
>-            2, 3, 34, 101,
>-            3, 4, 35, 102,
>-            4, 5, 36, 103};
>-
>-    EXPECT_EQ(0, WebRtcSpl_ComplexFFT(B, 3, 1));
>-//    for (int kk = 0; kk < 16; ++kk) {
>-//        EXPECT_EQ(A[kk], B[kk]);
>-//    }
>-    EXPECT_EQ(0, WebRtcSpl_ComplexIFFT(B, 3, 1));
>-//    for (int kk = 0; kk < 16; ++kk) {
>-//        EXPECT_EQ(A[kk], B[kk]);
>-//    }
>-    WebRtcSpl_ComplexBitReverse(B, 3);
>-    for (int kk = 0; kk < 16; ++kk) {
>-//      EXPECT_EQ(A[kk], B[kk]);
>-    }
>+  int16_t B[] = {1, 2, 33, 100, 2, 3, 34, 101, 3, 4, 35, 102, 4, 5, 36, 103};
>+
>+  EXPECT_EQ(0, WebRtcSpl_ComplexFFT(B, 3, 1));
>+  //    for (int kk = 0; kk < 16; ++kk) {
>+  //        EXPECT_EQ(A[kk], B[kk]);
>+  //    }
>+  EXPECT_EQ(0, WebRtcSpl_ComplexIFFT(B, 3, 1));
>+  //    for (int kk = 0; kk < 16; ++kk) {
>+  //        EXPECT_EQ(A[kk], B[kk]);
>+  //    }
>+  WebRtcSpl_ComplexBitReverse(B, 3);
>+  for (int kk = 0; kk < 16; ++kk) {
>+    //      EXPECT_EQ(A[kk], B[kk]);
>+  }
> }
> 
> TEST_F(SplTest, Resample48WithSaturationTest) {
>@@ -570,14 +589,13 @@ TEST_F(SplTest, Resample48WithSaturationTest) {
> 
>   // Saturated input vector of 48 samples.
>   const int32_t kVectorSaturated[3 * kBlockSize + 7] = {
>-     -32768, -32768, -32768, -32768, -32768, -32768, -32768, -32768,
>-     -32768, -32768, -32768, -32768, -32768, -32768, -32768, -32768,
>-     -32768, -32768, -32768, -32768, -32768, -32768, -32768, -32768,
>-     32767, 32767, 32767, 32767, 32767, 32767, 32767, 32767,
>-     32767, 32767, 32767, 32767, 32767, 32767, 32767, 32767,
>-     32767, 32767, 32767, 32767, 32767, 32767, 32767, 32767,
>-     32767, 32767, 32767, 32767, 32767, 32767, 32767
>-  };
>+      -32768, -32768, -32768, -32768, -32768, -32768, -32768, -32768,
>+      -32768, -32768, -32768, -32768, -32768, -32768, -32768, -32768,
>+      -32768, -32768, -32768, -32768, -32768, -32768, -32768, -32768,
>+      32767,  32767,  32767,  32767,  32767,  32767,  32767,  32767,
>+      32767,  32767,  32767,  32767,  32767,  32767,  32767,  32767,
>+      32767,  32767,  32767,  32767,  32767,  32767,  32767,  32767,
>+      32767,  32767,  32767,  32767,  32767,  32767,  32767};
> 
>   // All values in |out_vector| should be |kRefValue32kHz|.
>   const int32_t kRefValue32kHz1 = -1077493760;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/spl_init.c b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/spl_init.c
>index 0f41bc13211..80e91973477 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/spl_init.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/spl_init.c
>@@ -30,7 +30,7 @@ ScaleAndAddVectorsWithRound WebRtcSpl_ScaleAndAddVectorsWithRound;
> 
> #if (!defined(WEBRTC_HAS_NEON)) && !defined(MIPS32_LE)
> /* Initialize function pointers to the generic C version. */
>-static void InitPointersToC() {
>+static void InitPointersToC(void) {
>   WebRtcSpl_MaxAbsValueW16 = WebRtcSpl_MaxAbsValueW16C;
>   WebRtcSpl_MaxAbsValueW32 = WebRtcSpl_MaxAbsValueW32C;
>   WebRtcSpl_MaxValueW16 = WebRtcSpl_MaxValueW16C;
>@@ -46,7 +46,7 @@ static void InitPointersToC() {
> 
> #if defined(WEBRTC_HAS_NEON)
> /* Initialize function pointers to the Neon version. */
>-static void InitPointersToNeon() {
>+static void InitPointersToNeon(void) {
>   WebRtcSpl_MaxAbsValueW16 = WebRtcSpl_MaxAbsValueW16Neon;
>   WebRtcSpl_MaxAbsValueW32 = WebRtcSpl_MaxAbsValueW32Neon;
>   WebRtcSpl_MaxValueW16 = WebRtcSpl_MaxValueW16Neon;
>@@ -62,7 +62,7 @@ static void InitPointersToNeon() {
> 
> #if defined(MIPS32_LE)
> /* Initialize function pointers to the MIPS version. */
>-static void InitPointersToMIPS() {
>+static void InitPointersToMIPS(void) {
>   WebRtcSpl_MaxAbsValueW16 = WebRtcSpl_MaxAbsValueW16_mips;
>   WebRtcSpl_MaxValueW16 = WebRtcSpl_MaxValueW16_mips;
>   WebRtcSpl_MaxValueW32 = WebRtcSpl_MaxValueW32_mips;
>@@ -128,6 +128,6 @@ static void once(void (*func)(void)) {
>  */
> #endif  /* WEBRTC_POSIX */
> 
>-void WebRtcSpl_Init() {
>+void WebRtcSpl_Init(void) {
>   once(InitFunctionPointers);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/smoothing_filter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/smoothing_filter.cc
>index ecfb5c252b0..d426bda2501 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/smoothing_filter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/smoothing_filter.cc
>@@ -52,10 +52,10 @@ void SmoothingFilterImpl::AddSample(float sample) {
>   last_sample_ = sample;
> }
> 
>-rtc::Optional<float> SmoothingFilterImpl::GetAverage() {
>+absl::optional<float> SmoothingFilterImpl::GetAverage() {
>   if (!init_end_time_ms_) {
>     // |init_end_time_ms_| undefined since we have not received any sample.
>-    return rtc::nullopt;
>+    return absl::nullopt;
>   }
>   ExtrapolateLastSample(rtc::TimeMillis());
>   return state_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/smoothing_filter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/smoothing_filter.h
>index b8ab4e5919c..cff746953aa 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/smoothing_filter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/smoothing_filter.h
>@@ -11,7 +11,7 @@
> #ifndef COMMON_AUDIO_SMOOTHING_FILTER_H_
> #define COMMON_AUDIO_SMOOTHING_FILTER_H_
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> #include "rtc_base/constructormagic.h"
> #include "system_wrappers/include/clock.h"
> 
>@@ -21,7 +21,7 @@ class SmoothingFilter {
>  public:
>   virtual ~SmoothingFilter() = default;
>   virtual void AddSample(float sample) = 0;
>-  virtual rtc::Optional<float> GetAverage() = 0;
>+  virtual absl::optional<float> GetAverage() = 0;
>   virtual bool SetTimeConstantMs(int time_constant_ms) = 0;
> };
> 
>@@ -44,7 +44,7 @@ class SmoothingFilterImpl final : public SmoothingFilter {
>   ~SmoothingFilterImpl() override;
> 
>   void AddSample(float sample) override;
>-  rtc::Optional<float> GetAverage() override;
>+  absl::optional<float> GetAverage() override;
>   bool SetTimeConstantMs(int time_constant_ms) override;
> 
>   // Methods used for unittests.
>@@ -58,7 +58,7 @@ class SmoothingFilterImpl final : public SmoothingFilter {
>   const float init_factor_;
>   const float init_const_;
> 
>-  rtc::Optional<int64_t> init_end_time_ms_;
>+  absl::optional<int64_t> init_end_time_ms_;
>   float last_sample_;
>   float alpha_;
>   float state_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/smoothing_filter_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/smoothing_filter_unittest.cc
>index d173c9ab850..abe4272a825 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/smoothing_filter_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/smoothing_filter_unittest.cc
>@@ -25,7 +25,7 @@ constexpr int64_t kClockInitialTime = 123456;
> struct SmoothingFilterStates {
>   explicit SmoothingFilterStates(int init_time_ms)
>       : smoothing_filter(init_time_ms) {
>-    fake_clock.AdvanceTime(rtc::TimeDelta::FromMilliseconds(kClockInitialTime));
>+    fake_clock.AdvanceTime(TimeDelta::ms(kClockInitialTime));
>   }
>   rtc::ScopedFakeClock fake_clock;
>   SmoothingFilterImpl smoothing_filter;
>@@ -41,8 +41,7 @@ void CheckOutput(SmoothingFilterStates* states,
>                  int advance_time_ms,
>                  float expected_ouput) {
>   states->smoothing_filter.AddSample(sample);
>-  states->fake_clock.AdvanceTime(
>-      rtc::TimeDelta::FromMilliseconds(advance_time_ms));
>+  states->fake_clock.AdvanceTime(TimeDelta::ms(advance_time_ms));
>   auto output = states->smoothing_filter.GetAverage();
>   EXPECT_TRUE(output);
>   EXPECT_NEAR(expected_ouput, *output, kMaxAbsError);
>@@ -141,14 +140,13 @@ TEST(SmoothingFilterTest, CannotChangeTimeConstantDuringInitialization) {
>   states.smoothing_filter.AddSample(0.0);
> 
>   // During initialization, |SetTimeConstantMs| does not take effect.
>-  states.fake_clock.AdvanceTime(
>-      rtc::TimeDelta::FromMilliseconds(kInitTimeMs - 1));
>+  states.fake_clock.AdvanceTime(TimeDelta::ms(kInitTimeMs - 1));
>   states.smoothing_filter.AddSample(0.0);
> 
>   EXPECT_FALSE(states.smoothing_filter.SetTimeConstantMs(kInitTimeMs * 2));
>   EXPECT_NE(exp(-1.0f / (kInitTimeMs * 2)), states.smoothing_filter.alpha());
> 
>-  states.fake_clock.AdvanceTime(rtc::TimeDelta::FromMilliseconds(1));
>+  states.fake_clock.AdvanceTime(TimeDelta::ms(1));
>   states.smoothing_filter.AddSample(0.0);
>   // When initialization finishes, the time constant should be come
>   // |kInitTimeConstantMs|.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/sparse_fir_filter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/sparse_fir_filter.cc
>index ed2d79bbe16..0fc032722e8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/sparse_fir_filter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/sparse_fir_filter.cc
>@@ -34,8 +34,8 @@ void SparseFIRFilter::Filter(const float* in, size_t length, float* out) {
>   for (size_t i = 0; i < length; ++i) {
>     out[i] = 0.f;
>     size_t j;
>-    for (j = 0; i >= j * sparsity_ + offset_ &&
>-                j < nonzero_coeffs_.size(); ++j) {
>+    for (j = 0; i >= j * sparsity_ + offset_ && j < nonzero_coeffs_.size();
>+         ++j) {
>       out[i] += in[i - j * sparsity_ - offset_] * nonzero_coeffs_[j];
>     }
>     for (; j < nonzero_coeffs_.size(); ++j) {
>@@ -47,12 +47,10 @@ void SparseFIRFilter::Filter(const float* in, size_t length, float* out) {
>   // Update current state.
>   if (state_.size() > 0u) {
>     if (length >= state_.size()) {
>-      std::memcpy(&state_[0],
>-                  &in[length - state_.size()],
>+      std::memcpy(&state_[0], &in[length - state_.size()],
>                   state_.size() * sizeof(*in));
>     } else {
>-      std::memmove(&state_[0],
>-                   &state_[length],
>+      std::memmove(&state_[0], &state_[length],
>                    (state_.size() - length) * sizeof(state_[0]));
>       std::memcpy(&state_[state_.size() - length], in, length * sizeof(*in));
>     }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/sparse_fir_filter_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/sparse_fir_filter_unittest.cc
>index 434daaa6d96..b6cd6f9d9b6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/sparse_fir_filter_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/sparse_fir_filter_unittest.cc
>@@ -21,8 +21,8 @@ namespace webrtc {
> namespace {
> 
> static const float kCoeffs[] = {0.2f, 0.3f, 0.5f, 0.7f, 0.11f};
>-static const float kInput[] =
>-    {1.f, 2.f, 3.f, 4.f, 5.f, 6.f, 7.f, 8.f, 9.f, 10.f};
>+static const float kInput[] = {1.f, 2.f, 3.f, 4.f, 5.f,
>+                               6.f, 7.f, 8.f, 9.f, 10.f};
> 
> template <size_t N>
> void VerifyOutput(const float (&expected_output)[N], const float (&output)[N]) {
>@@ -50,13 +50,9 @@ TEST(SparseFIRFilterTest, SameOutputForScalarCoefficientAndDifferentSparsity) {
>   const size_t kOffset = 0;
>   float low_sparsity_output[arraysize(kInput)];
>   float high_sparsity_output[arraysize(kInput)];
>-  SparseFIRFilter low_sparsity_filter(&kCoeff,
>-                                      kNumCoeff,
>-                                      kLowSparsity,
>+  SparseFIRFilter low_sparsity_filter(&kCoeff, kNumCoeff, kLowSparsity,
>                                       kOffset);
>-  SparseFIRFilter high_sparsity_filter(&kCoeff,
>-                                       kNumCoeff,
>-                                       kHighSparsity,
>+  SparseFIRFilter high_sparsity_filter(&kCoeff, kNumCoeff, kHighSparsity,
>                                        kOffset);
>   low_sparsity_filter.Filter(kInput, arraysize(kInput), low_sparsity_output);
>   high_sparsity_filter.Filter(kInput, arraysize(kInput), high_sparsity_output);
>@@ -146,15 +142,10 @@ TEST(SparseFIRFilterTest, VerifySampleBasedVsBlockBasedFiltering) {
>   const size_t kSparsity = 3;
>   const size_t kOffset = 1;
>   float output_block_based[arraysize(kInput)];
>-  SparseFIRFilter filter_block(kCoeffs,
>-                               arraysize(kCoeffs),
>-                               kSparsity,
>-                               kOffset);
>+  SparseFIRFilter filter_block(kCoeffs, arraysize(kCoeffs), kSparsity, kOffset);
>   filter_block.Filter(kInput, arraysize(kInput), output_block_based);
>   float output_sample_based[arraysize(kInput)];
>-  SparseFIRFilter filter_sample(kCoeffs,
>-                                arraysize(kCoeffs),
>-                                kSparsity,
>+  SparseFIRFilter filter_sample(kCoeffs, arraysize(kCoeffs), kSparsity,
>                                 kOffset);
>   for (size_t i = 0; i < arraysize(kInput); ++i)
>     filter_sample.Filter(&kInput[i], 1, &output_sample_based[i]);
>@@ -165,8 +156,8 @@ TEST(SparseFIRFilterTest, SimpleHighPassFilter) {
>   const size_t kSparsity = 2;
>   const size_t kOffset = 2;
>   const float kHPCoeffs[] = {1.f, -1.f};
>-  const float kConstantInput[] =
>-      {1.f, 1.f, 1.f, 1.f, 1.f, 1.f, 1.f, 1.f, 1.f, 1.f};
>+  const float kConstantInput[] = {1.f, 1.f, 1.f, 1.f, 1.f,
>+                                  1.f, 1.f, 1.f, 1.f, 1.f};
>   float output[arraysize(kConstantInput)];
>   SparseFIRFilter filter(kHPCoeffs, arraysize(kHPCoeffs), kSparsity, kOffset);
>   filter.Filter(kConstantInput, arraysize(kConstantInput), output);
>@@ -182,8 +173,8 @@ TEST(SparseFIRFilterTest, SimpleLowPassFilter) {
>   const size_t kSparsity = 2;
>   const size_t kOffset = 2;
>   const float kLPCoeffs[] = {1.f, 1.f};
>-  const float kHighFrequencyInput[] =
>-      {1.f, 1.f, -1.f, -1.f, 1.f, 1.f, -1.f, -1.f, 1.f, 1.f};
>+  const float kHighFrequencyInput[] = {1.f, 1.f,  -1.f, -1.f, 1.f,
>+                                       1.f, -1.f, -1.f, 1.f,  1.f};
>   float output[arraysize(kHighFrequencyInput)];
>   SparseFIRFilter filter(kLPCoeffs, arraysize(kLPCoeffs), kSparsity, kOffset);
>   filter.Filter(kHighFrequencyInput, arraysize(kHighFrequencyInput), output);
>@@ -203,9 +194,7 @@ TEST(SparseFIRFilterTest, SameOutputWhenSwappedCoefficientsAndInput) {
>   SparseFIRFilter filter(kCoeffs, arraysize(kCoeffs), kSparsity, kOffset);
>   // Use arraysize(kCoeffs) for in_length to get same-length outputs.
>   filter.Filter(kInput, arraysize(kCoeffs), output);
>-  SparseFIRFilter filter_swapped(kInput,
>-                                 arraysize(kCoeffs),
>-                                 kSparsity,
>+  SparseFIRFilter filter_swapped(kInput, arraysize(kCoeffs), kSparsity,
>                                  kOffset);
>   filter_swapped.Filter(kCoeffs, arraysize(kCoeffs), output_swapped);
>   VerifyOutput(output, output_swapped);
>@@ -218,9 +207,7 @@ TEST(SparseFIRFilterTest, SameOutputAsFIRFilterWhenSparsityOneAndOffsetZero) {
>   float sparse_output[arraysize(kInput)];
>   std::unique_ptr<FIRFilter> filter(
>       CreateFirFilter(kCoeffs, arraysize(kCoeffs), arraysize(kInput)));
>-  SparseFIRFilter sparse_filter(kCoeffs,
>-                                arraysize(kCoeffs),
>-                                kSparsity,
>+  SparseFIRFilter sparse_filter(kCoeffs, arraysize(kCoeffs), kSparsity,
>                                 kOffset);
>   filter->Filter(kInput, arraysize(kInput), output);
>   sparse_filter.Filter(kInput, arraysize(kInput), sparse_output);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/third_party/fft4g/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/third_party/fft4g/BUILD.gn
>new file mode 100644
>index 00000000000..ae0d5f6c00c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/third_party/fft4g/BUILD.gn
>@@ -0,0 +1,16 @@
>+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+#
>+# Use of this source code is governed by a BSD-style license
>+# that can be found in the ../../../LICENSE file in the root of the source
>+# tree. An additional intellectual property rights grant can be found
>+# in the file PATENTS.  All contributing project authors may
>+# be found in the AUTHORS file in the root of the source tree.
>+
>+import("../../../webrtc.gni")
>+
>+rtc_source_set("fft4g") {
>+  sources = [
>+    "fft4g.c",
>+    "fft4g.h",
>+  ]
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/third_party/fft4g/LICENSE b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/third_party/fft4g/LICENSE
>new file mode 100644
>index 00000000000..3bf870aa3cc
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/third_party/fft4g/LICENSE
>@@ -0,0 +1,8 @@
>+/*
>+ * http://www.kurims.kyoto-u.ac.jp/~ooura/fft.html
>+ * Copyright Takuya OOURA, 1996-2001
>+ *
>+ * You may use, copy, modify and distribute this code for any purpose (include
>+ * commercial use) and without fee. Please refer to this package when you modify
>+ * this code.
>+ */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/third_party/fft4g/README.chromium b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/third_party/fft4g/README.chromium
>new file mode 100644
>index 00000000000..9df2ddb5e28
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/third_party/fft4g/README.chromium
>@@ -0,0 +1,13 @@
>+Name: General Purpose FFT (Fast Fourier/Cosine/Sine Transform) Package
>+Short Name: fft4g
>+URL: http://www.kurims.kyoto-u.ac.jp/~ooura/fft.html
>+Version: 0
>+Date: 2018-06-19
>+License: Custome license
>+License File: LICENSE
>+Security Critical: yes
>+
>+Description:
>+This is a package to calculate Discrete Fourier/Cosine/Sine Transforms of
>+1-dimensional sequences of length 2^N. This package contains C and Fortran
>+FFT codes.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/fft4g.c b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/third_party/fft4g/fft4g.c
>similarity index 100%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/fft4g.c
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/third_party/fft4g/fft4g.c
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/fft4g.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/third_party/fft4g/fft4g.h
>similarity index 52%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/fft4g.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/third_party/fft4g/fft4g.h
>index 1f0e29dd5c8..f1f31a182d9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/fft4g.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/third_party/fft4g/fft4g.h
>@@ -1,25 +1,25 @@
> /*
>- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>  *
>  *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>+ *  that can be found in the ../../../LICENSE file in the root of the source
>  *  tree. An additional intellectual property rights grant can be found
>  *  in the file PATENTS.  All contributing project authors may
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#ifndef COMMON_AUDIO_FFT4G_H_
>-#define COMMON_AUDIO_FFT4G_H_
>+#ifndef COMMON_AUDIO_THIRD_PARTY_FFT4G_FFT4G_H_
>+#define COMMON_AUDIO_THIRD_PARTY_FFT4G_FFT4G_H_
> 
> #if defined(__cplusplus)
> extern "C" {
> #endif
> 
> // Refer to fft4g.c for documentation.
>-void WebRtc_rdft(size_t n, int isgn, float *a, size_t *ip, float *w);
>+void WebRtc_rdft(size_t n, int isgn, float* a, size_t* ip, float* w);
> 
> #if defined(__cplusplus)
> }
> #endif
> 
>-#endif  // COMMON_AUDIO_FFT4G_H_
>+#endif  /* COMMON_AUDIO_THIRD_PARTY_FFT4G_FFT4G_H_ */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/third_party/spl_sqrt_floor/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/third_party/spl_sqrt_floor/BUILD.gn
>new file mode 100644
>index 00000000000..194899ebda3
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/third_party/spl_sqrt_floor/BUILD.gn
>@@ -0,0 +1,26 @@
>+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+#
>+# Use of this source code is governed by a BSD-style license
>+# that can be found in the ../../../LICENSE file in the root of the source
>+# tree. An additional intellectual property rights grant can be found
>+# in the file PATENTS.  All contributing project authors may
>+# be found in the AUTHORS file in the root of the source tree.
>+
>+import("../../../webrtc.gni")
>+
>+rtc_source_set("spl_sqrt_floor") {
>+  visibility = [ "../..:common_audio_c" ]
>+  sources = [
>+    "spl_sqrt_floor.h",
>+  ]
>+  deps = []
>+  if (current_cpu == "arm") {
>+    sources += [ "spl_sqrt_floor_arm.S" ]
>+
>+    deps += [ "../../../rtc_base/system:asm_defines" ]
>+  } else if (current_cpu == "mipsel") {
>+    sources += [ "spl_sqrt_floor_mips.c" ]
>+  } else {
>+    sources += [ "spl_sqrt_floor.c" ]
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/third_party/spl_sqrt_floor/LICENSE b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/third_party/spl_sqrt_floor/LICENSE
>new file mode 100644
>index 00000000000..fdf17a2041e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/third_party/spl_sqrt_floor/LICENSE
>@@ -0,0 +1,27 @@
>+/*
>+ * Written by Wilco Dijkstra, 1996. The following email exchange establishes the
>+ * license.
>+ *
>+ * From: Wilco Dijkstra <Wilco.Dijkstra@ntlworld.com>
>+ * Date: Fri, Jun 24, 2011 at 3:20 AM
>+ * Subject: Re: sqrt routine
>+ * To: Kevin Ma <kma@google.com>
>+ * Hi Kevin,
>+ * Thanks for asking. Those routines are public domain (originally posted to
>+ * comp.sys.arm a long time ago), so you can use them freely for any purpose.
>+ * Cheers,
>+ * Wilco
>+ *
>+ * ----- Original Message -----
>+ * From: "Kevin Ma" <kma@google.com>
>+ * To: <Wilco.Dijkstra@ntlworld.com>
>+ * Sent: Thursday, June 23, 2011 11:44 PM
>+ * Subject: Fwd: sqrt routine
>+ * Hi Wilco,
>+ * I saw your sqrt routine from several web sites, including
>+ * http://www.finesse.demon.co.uk/steven/sqrt.html.
>+ * Just wonder if there's any copyright information with your Successive
>+ * approximation routines, or if I can freely use it for any purpose.
>+ * Thanks.
>+ * Kevin
>+ */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/third_party/spl_sqrt_floor/README.chromium b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/third_party/spl_sqrt_floor/README.chromium
>new file mode 100644
>index 00000000000..b226490e85f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/third_party/spl_sqrt_floor/README.chromium
>@@ -0,0 +1,12 @@
>+Name: sql sqrt floor
>+Short Name: sql_sqrt_floor
>+URL: http://www.pertinentdetail.org/sqrt
>+Version: 0
>+Date: 2018-03-22
>+License: Custom license
>+License File: LICENSE
>+Security Critical: yes
>+
>+Description:
>+Sqrt routine, originally was posted to the USENET group comp.sys.arm on
>+20 Jun 1996.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/spl_sqrt_floor.c b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/third_party/spl_sqrt_floor/spl_sqrt_floor.c
>similarity index 96%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/spl_sqrt_floor.c
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/third_party/spl_sqrt_floor/spl_sqrt_floor.c
>index 714138696bc..b478a41b969 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/spl_sqrt_floor.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/third_party/spl_sqrt_floor/spl_sqrt_floor.c
>@@ -28,7 +28,7 @@
> 
> // Minor modifications in code style for WebRTC, 2012.
> 
>-#include "common_audio/signal_processing/include/signal_processing_library.h"
>+#include "common_audio/third_party/spl_sqrt_floor/spl_sqrt_floor.h"
> 
> /*
>  * Algorithm:
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/third_party/spl_sqrt_floor/spl_sqrt_floor.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/third_party/spl_sqrt_floor/spl_sqrt_floor.h
>new file mode 100644
>index 00000000000..eaa58e30a4b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/third_party/spl_sqrt_floor/spl_sqrt_floor.h
>@@ -0,0 +1,29 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <stdint.h>
>+
>+//
>+// WebRtcSpl_SqrtFloor(...)
>+//
>+// Returns the square root of the input value |value|. The precision of this
>+// function is rounding down integer precision, i.e., sqrt(8) gives 2 as answer.
>+// If |value| is a negative number then 0 is returned.
>+//
>+// Algorithm:
>+//
>+// An iterative 4 cylce/bit routine
>+//
>+// Input:
>+//      - value     : Value to calculate sqrt of
>+//
>+// Return value     : Result of the sqrt calculation
>+//
>+int32_t WebRtcSpl_SqrtFloor(int32_t value);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/spl_sqrt_floor_arm.S b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/third_party/spl_sqrt_floor/spl_sqrt_floor_arm.S
>similarity index 98%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/spl_sqrt_floor_arm.S
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/third_party/spl_sqrt_floor/spl_sqrt_floor_arm.S
>index 29e6d4d4552..228e68e6ca6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/spl_sqrt_floor_arm.S
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/third_party/spl_sqrt_floor/spl_sqrt_floor_arm.S
>@@ -32,7 +32,7 @@
> @ Output:             r0 = INT (SQRT (r0)), precision is 16 bits
> @ Registers touched:  r1, r2
> 
>-#include "system_wrappers/include/asm_defines.h"
>+#include "rtc_base/system/asm_defines.h"
> 
> GLOBAL_FUNCTION WebRtcSpl_SqrtFloor
> .align  2
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/spl_sqrt_floor_mips.c b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/third_party/spl_sqrt_floor/spl_sqrt_floor_mips.c
>similarity index 99%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/spl_sqrt_floor_mips.c
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/third_party/spl_sqrt_floor/spl_sqrt_floor_mips.c
>index 7128fbd381a..04033c14dee 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/signal_processing/spl_sqrt_floor_mips.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/third_party/spl_sqrt_floor/spl_sqrt_floor_mips.c
>@@ -29,7 +29,7 @@
> // Minor modifications in code style for WebRTC, 2012.
> // Code optimizations for MIPS, 2013.
> 
>-#include "common_audio/signal_processing/include/signal_processing_library.h"
>+#include "common_audio/third_party/spl_sqrt_floor/spl_sqrt_floor.h"
> 
> /*
>  * Algorithm:
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/include/vad.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/include/vad.h
>index bd10756c99d..b15275b1668 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/include/vad.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/include/vad.h
>@@ -15,7 +15,6 @@
> 
> #include "common_audio/vad/include/webrtc_vad.h"
> #include "rtc_base/checks.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/include/webrtc_vad.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/include/webrtc_vad.h
>index 353dbf0861d..f5bbadf5b78 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/include/webrtc_vad.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/include/webrtc_vad.h
>@@ -8,17 +8,16 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-
> /*
>- * This header file includes the VAD API calls. Specific function calls are given below.
>+ * This header file includes the VAD API calls. Specific function calls are
>+ * given below.
>  */
> 
> #ifndef COMMON_AUDIO_VAD_INCLUDE_WEBRTC_VAD_H_  // NOLINT
> #define COMMON_AUDIO_VAD_INCLUDE_WEBRTC_VAD_H_
> 
> #include <stddef.h>
>-
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include <stdint.h>
> 
> typedef struct WebRtcVadInst VadInst;
> 
>@@ -67,7 +66,9 @@ int WebRtcVad_set_mode(VadInst* handle, int mode);
> // returns              : 1 - (Active Voice),
> //                        0 - (Non-active Voice),
> //                       -1 - (Error)
>-int WebRtcVad_Process(VadInst* handle, int fs, const int16_t* audio_frame,
>+int WebRtcVad_Process(VadInst* handle,
>+                      int fs,
>+                      const int16_t* audio_frame,
>                       size_t frame_length);
> 
> // Checks for valid combinations of |rate| and |frame_length|. We support 10,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_core.c b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_core.c
>index 7316b45ce8b..55927cea40f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_core.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_core.c
>@@ -15,7 +15,6 @@
> #include "common_audio/vad/vad_filterbank.h"
> #include "common_audio/vad/vad_gmm.h"
> #include "common_audio/vad/vad_sp.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> // Spectrum Weighting
> static const int16_t kSpectrumWeight[kNumChannels] = { 6, 8, 10, 12, 14, 16 };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_core.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_core.h
>index 6541819f53e..e79696cef0b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_core.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_core.h
>@@ -8,7 +8,6 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-
> /*
>  * This header file includes the descriptions of the core VAD calls.
>  */
>@@ -17,39 +16,38 @@
> #define COMMON_AUDIO_VAD_VAD_CORE_H_
> 
> #include "common_audio/signal_processing/include/signal_processing_library.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
>-enum { kNumChannels = 6 };  // Number of frequency bands (named channels).
>+enum { kNumChannels = 6 };   // Number of frequency bands (named channels).
> enum { kNumGaussians = 2 };  // Number of Gaussians per channel in the GMM.
> enum { kTableSize = kNumChannels * kNumGaussians };
> enum { kMinEnergy = 10 };  // Minimum energy required to trigger audio signal.
> 
> typedef struct VadInstT_ {
>-    int vad;
>-    int32_t downsampling_filter_states[4];
>-    WebRtcSpl_State48khzTo8khz state_48_to_8;
>-    int16_t noise_means[kTableSize];
>-    int16_t speech_means[kTableSize];
>-    int16_t noise_stds[kTableSize];
>-    int16_t speech_stds[kTableSize];
>-    // TODO(bjornv): Change to |frame_count|.
>-    int32_t frame_counter;
>-    int16_t over_hang;  // Over Hang
>-    int16_t num_of_speech;
>-    // TODO(bjornv): Change to |age_vector|.
>-    int16_t index_vector[16 * kNumChannels];
>-    int16_t low_value_vector[16 * kNumChannels];
>-    // TODO(bjornv): Change to |median|.
>-    int16_t mean_value[kNumChannels];
>-    int16_t upper_state[5];
>-    int16_t lower_state[5];
>-    int16_t hp_filter_state[4];
>-    int16_t over_hang_max_1[3];
>-    int16_t over_hang_max_2[3];
>-    int16_t individual[3];
>-    int16_t total[3];
>+  int vad;
>+  int32_t downsampling_filter_states[4];
>+  WebRtcSpl_State48khzTo8khz state_48_to_8;
>+  int16_t noise_means[kTableSize];
>+  int16_t speech_means[kTableSize];
>+  int16_t noise_stds[kTableSize];
>+  int16_t speech_stds[kTableSize];
>+  // TODO(bjornv): Change to |frame_count|.
>+  int32_t frame_counter;
>+  int16_t over_hang;  // Over Hang
>+  int16_t num_of_speech;
>+  // TODO(bjornv): Change to |age_vector|.
>+  int16_t index_vector[16 * kNumChannels];
>+  int16_t low_value_vector[16 * kNumChannels];
>+  // TODO(bjornv): Change to |median|.
>+  int16_t mean_value[kNumChannels];
>+  int16_t upper_state[5];
>+  int16_t lower_state[5];
>+  int16_t hp_filter_state[4];
>+  int16_t over_hang_max_1[3];
>+  int16_t over_hang_max_2[3];
>+  int16_t individual[3];
>+  int16_t total[3];
> 
>-    int init_flag;
>+  int init_flag;
> } VadInstT;
> 
> // Initializes the core VAD component. The default aggressiveness mode is
>@@ -100,13 +98,17 @@ int WebRtcVad_set_mode_core(VadInstT* self, int mode);
>  *                        0 - No active speech
>  *                        1-6 - Active speech
>  */
>-int WebRtcVad_CalcVad48khz(VadInstT* inst, const int16_t* speech_frame,
>+int WebRtcVad_CalcVad48khz(VadInstT* inst,
>+                           const int16_t* speech_frame,
>                            size_t frame_length);
>-int WebRtcVad_CalcVad32khz(VadInstT* inst, const int16_t* speech_frame,
>+int WebRtcVad_CalcVad32khz(VadInstT* inst,
>+                           const int16_t* speech_frame,
>                            size_t frame_length);
>-int WebRtcVad_CalcVad16khz(VadInstT* inst, const int16_t* speech_frame,
>+int WebRtcVad_CalcVad16khz(VadInstT* inst,
>+                           const int16_t* speech_frame,
>                            size_t frame_length);
>-int WebRtcVad_CalcVad8khz(VadInstT* inst, const int16_t* speech_frame,
>+int WebRtcVad_CalcVad8khz(VadInstT* inst,
>+                          const int16_t* speech_frame,
>                           size_t frame_length);
> 
> #endif  // COMMON_AUDIO_VAD_VAD_CORE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_core_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_core_unittest.cc
>index 0587878105a..3131a86ae32 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_core_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_core_unittest.cc
>@@ -12,7 +12,6 @@
> 
> #include "common_audio/vad/vad_unittest.h"
> #include "test/gtest.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> extern "C" {
> #include "common_audio/vad/vad_core.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_filterbank.c b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_filterbank.c
>index 82cff25c36e..1513153c48b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_filterbank.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_filterbank.c
>@@ -12,7 +12,6 @@
> 
> #include "rtc_base/checks.h"
> #include "common_audio/signal_processing/include/signal_processing_library.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> // Constants used in LogOfEnergy().
> static const int16_t kLogConst = 24660;  // 160*log10(2) in Q9.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_filterbank.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_filterbank.h
>index 620f96a6991..53bbbe10fdc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_filterbank.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_filterbank.h
>@@ -16,7 +16,6 @@
> #define COMMON_AUDIO_VAD_VAD_FILTERBANK_H_
> 
> #include "common_audio/vad/vad_core.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> // Takes |data_length| samples of |data_in| and calculates the logarithm of the
> // energy of each of the |kNumChannels| = 6 frequency bands used by the VAD:
>@@ -38,7 +37,9 @@
> // - features     [o]   : 10 * log10(energy in each frequency band), Q4.
> // - returns            : Total energy of the signal (NOTE! This value is not
> //                        exact. It is only used in a comparison.)
>-int16_t WebRtcVad_CalculateFeatures(VadInstT* self, const int16_t* data_in,
>-                                    size_t data_length, int16_t* features);
>+int16_t WebRtcVad_CalculateFeatures(VadInstT* self,
>+                                    const int16_t* data_in,
>+                                    size_t data_length,
>+                                    int16_t* features);
> 
> #endif  // COMMON_AUDIO_VAD_VAD_FILTERBANK_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_filterbank_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_filterbank_unittest.cc
>index 55b12793dad..51d8d0fefdf 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_filterbank_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_filterbank_unittest.cc
>@@ -12,7 +12,6 @@
> 
> #include "common_audio/vad/vad_unittest.h"
> #include "test/gtest.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> extern "C" {
> #include "common_audio/vad/vad_core.h"
>@@ -26,14 +25,12 @@ const int kNumValidFrameLengths = 3;
> 
> TEST_F(VadTest, vad_filterbank) {
>   VadInstT* self = reinterpret_cast<VadInstT*>(malloc(sizeof(VadInstT)));
>-  static const int16_t kReference[kNumValidFrameLengths] = { 48, 11, 11 };
>+  static const int16_t kReference[kNumValidFrameLengths] = {48, 11, 11};
>   static const int16_t kFeatures[kNumValidFrameLengths * kNumChannels] = {
>-      1213, 759, 587, 462, 434, 272,
>-      1479, 1385, 1291, 1200, 1103, 1099,
>-      1732, 1692, 1681, 1629, 1436, 1436
>-  };
>-  static const int16_t kOffsetVector[kNumChannels] = {
>-      368, 368, 272, 176, 176, 176 };
>+      1213, 759,  587,  462,  434,  272,  1479, 1385, 1291,
>+      1200, 1103, 1099, 1732, 1692, 1681, 1629, 1436, 1436};
>+  static const int16_t kOffsetVector[kNumChannels] = {368, 368, 272,
>+                                                      176, 176, 176};
>   int16_t features[kNumChannels];
> 
>   // Construct a speech signal that will trigger the VAD in all modes. It is
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_gmm.c b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_gmm.c
>index b746fd53642..ddc87b6edd3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_gmm.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_gmm.c
>@@ -11,7 +11,6 @@
> #include "common_audio/vad/vad_gmm.h"
> 
> #include "common_audio/signal_processing/include/signal_processing_library.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> static const int32_t kCompVar = 22005;
> static const int16_t kLog2Exp = 5909;  // log2(exp(1)) in Q12.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_gmm.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_gmm.h
>index 79f15c89574..6b2d11ba336 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_gmm.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_gmm.h
>@@ -13,7 +13,7 @@
> #ifndef COMMON_AUDIO_VAD_VAD_GMM_H_
> #define COMMON_AUDIO_VAD_VAD_GMM_H_
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include <stdint.h>
> 
> // Calculates the probability for |input|, given that |input| comes from a
> // normal distribution with mean and standard deviation (|mean|, |std|).
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_gmm_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_gmm_unittest.cc
>index e77603d6ec0..be61f7f9715 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_gmm_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_gmm_unittest.cc
>@@ -10,7 +10,6 @@
> 
> #include "common_audio/vad/vad_unittest.h"
> #include "test/gtest.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> extern "C" {
> #include "common_audio/vad/vad_gmm.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_sp.c b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_sp.c
>index 915fb375206..d367c8b0af1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_sp.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_sp.c
>@@ -13,7 +13,6 @@
> #include "rtc_base/checks.h"
> #include "common_audio/signal_processing/include/signal_processing_library.h"
> #include "common_audio/vad/vad_core.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> // Allpass filter coefficients, upper and lower, in Q13.
> // Upper: 0.64, Lower: 0.17.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_sp.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_sp.h
>index 21fed11b1af..ff367603c58 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_sp.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_sp.h
>@@ -8,14 +8,12 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-
> // This file includes specific signal processing tools used in vad_core.c.
> 
> #ifndef COMMON_AUDIO_VAD_VAD_SP_H_
> #define COMMON_AUDIO_VAD_VAD_SP_H_
> 
> #include "common_audio/vad/vad_core.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> // Downsamples the signal by a factor 2, eg. 32->16 or 16->8.
> //
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_sp_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_sp_unittest.cc
>index 7eb6794d217..cdde56d1ca2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_sp_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_sp_unittest.cc
>@@ -12,7 +12,6 @@
> 
> #include "common_audio/vad/vad_unittest.h"
> #include "test/gtest.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> extern "C" {
> #include "common_audio/vad/vad_core.h"
>@@ -25,19 +24,17 @@ namespace test {
> TEST_F(VadTest, vad_sp) {
>   VadInstT* self = reinterpret_cast<VadInstT*>(malloc(sizeof(VadInstT)));
>   const size_t kMaxFrameLenSp = 960;  // Maximum frame length in this unittest.
>-  int16_t zeros[kMaxFrameLenSp] = { 0 };
>-  int32_t state[2] = { 0 };
>+  int16_t zeros[kMaxFrameLenSp] = {0};
>+  int32_t state[2] = {0};
>   int16_t data_in[kMaxFrameLenSp];
>   int16_t data_out[kMaxFrameLenSp];
> 
>   // We expect the first value to be 1600 as long as |frame_counter| is zero,
>   // which is true for the first iteration.
>   static const int16_t kReferenceMin[32] = {
>-      1600, 720, 509, 512, 532, 552, 570, 588,
>-       606, 624, 642, 659, 675, 691, 707, 723,
>-      1600, 544, 502, 522, 542, 561, 579, 597,
>-       615, 633, 651, 667, 683, 699, 715, 731
>-  };
>+      1600, 720, 509, 512, 532, 552,  570, 588, 606, 624, 642,
>+      659,  675, 691, 707, 723, 1600, 544, 502, 522, 542, 561,
>+      579,  597, 615, 633, 651, 667,  683, 699, 715, 731};
> 
>   // Construct a speech signal that will trigger the VAD in all modes. It is
>   // known that (i * i) will wrap around, but that doesn't matter in this case.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_unittest.cc
>index f79678b8a00..c54014efce8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_unittest.cc
>@@ -17,7 +17,6 @@
> #include "rtc_base/arraysize.h"
> #include "rtc_base/checks.h"
> #include "test/gtest.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> VadTest::VadTest() {}
> 
>@@ -60,7 +59,7 @@ TEST_F(VadTest, ApiTest) {
>   // combinations.
> 
>   VadInst* handle = WebRtcVad_Create();
>-  int16_t zeros[kMaxFrameLength] = { 0 };
>+  int16_t zeros[kMaxFrameLength] = {0};
> 
>   // Construct a speech signal that will trigger the VAD in all modes. It is
>   // known that (i * i) will wrap around, but that doesn't matter in this case.
>@@ -87,12 +86,10 @@ TEST_F(VadTest, ApiTest) {
> 
>   // WebRtcVad_set_mode() invalid modes tests. Tries smallest supported value
>   // minus one and largest supported value plus one.
>-  EXPECT_EQ(-1, WebRtcVad_set_mode(handle,
>-                                   WebRtcSpl_MinValueW32(kModes,
>-                                                         kModesSize) - 1));
>-  EXPECT_EQ(-1, WebRtcVad_set_mode(handle,
>-                                   WebRtcSpl_MaxValueW32(kModes,
>-                                                         kModesSize) + 1));
>+  EXPECT_EQ(-1, WebRtcVad_set_mode(
>+                    handle, WebRtcSpl_MinValueW32(kModes, kModesSize) - 1));
>+  EXPECT_EQ(-1, WebRtcVad_set_mode(
>+                    handle, WebRtcSpl_MaxValueW32(kModes, kModesSize) + 1));
> 
>   // WebRtcVad_Process() tests
>   // nullptr as speech pointer
>@@ -109,14 +106,10 @@ TEST_F(VadTest, ApiTest) {
>     for (size_t i = 0; i < kRatesSize; i++) {
>       for (size_t j = 0; j < kFrameLengthsSize; j++) {
>         if (ValidRatesAndFrameLengths(kRates[i], kFrameLengths[j])) {
>-          EXPECT_EQ(1, WebRtcVad_Process(handle,
>-                                         kRates[i],
>-                                         speech,
>+          EXPECT_EQ(1, WebRtcVad_Process(handle, kRates[i], speech,
>                                          kFrameLengths[j]));
>         } else {
>-          EXPECT_EQ(-1, WebRtcVad_Process(handle,
>-                                          kRates[i],
>-                                          speech,
>+          EXPECT_EQ(-1, WebRtcVad_Process(handle, kRates[i], speech,
>                                           kFrameLengths[j]));
>         }
>       }
>@@ -130,22 +123,20 @@ TEST_F(VadTest, ValidRatesFrameLengths) {
>   // This test verifies valid and invalid rate/frame_length combinations. We
>   // loop through some sampling rates and frame lengths from negative values to
>   // values larger than possible.
>-  const int kRates[] = {
>-    -8000, -4000, 0, 4000, 8000, 8001, 15999, 16000, 32000, 48000, 48001, 96000
>-  };
>+  const int kRates[] = {-8000, -4000, 0,     4000,  8000,  8001,
>+                        15999, 16000, 32000, 48000, 48001, 96000};
> 
>-  const size_t kFrameLengths[] = {
>-    0, 80, 81, 159, 160, 240, 320, 480, 640, 960, 1440, 2000
>-  };
>+  const size_t kFrameLengths[] = {0,   80,  81,  159, 160,  240,
>+                                  320, 480, 640, 960, 1440, 2000};
> 
>   for (size_t i = 0; i < arraysize(kRates); i++) {
>     for (size_t j = 0; j < arraysize(kFrameLengths); j++) {
>       if (ValidRatesAndFrameLengths(kRates[i], kFrameLengths[j])) {
>-        EXPECT_EQ(0, WebRtcVad_ValidRateAndFrameLength(kRates[i],
>-                                                       kFrameLengths[j]));
>+        EXPECT_EQ(
>+            0, WebRtcVad_ValidRateAndFrameLength(kRates[i], kFrameLengths[j]));
>       } else {
>-        EXPECT_EQ(-1, WebRtcVad_ValidRateAndFrameLength(kRates[i],
>-                                                        kFrameLengths[j]));
>+        EXPECT_EQ(
>+            -1, WebRtcVad_ValidRateAndFrameLength(kRates[i], kFrameLengths[j]));
>       }
>     }
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_unittest.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_unittest.h
>index f982f52eb5a..ee642063af5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_unittest.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/vad_unittest.h
>@@ -14,23 +14,22 @@
> #include <stddef.h>  // size_t
> 
> #include "test/gtest.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> namespace test {
> 
> // Modes we support
>-const int kModes[] = { 0, 1, 2, 3 };
>+const int kModes[] = {0, 1, 2, 3};
> const size_t kModesSize = sizeof(kModes) / sizeof(*kModes);
> 
> // Rates we support.
>-const int kRates[] = { 8000, 12000, 16000, 24000, 32000, 48000 };
>+const int kRates[] = {8000, 12000, 16000, 24000, 32000, 48000};
> const size_t kRatesSize = sizeof(kRates) / sizeof(*kRates);
> 
> // Frame lengths we support.
> const size_t kMaxFrameLength = 1440;
>-const size_t kFrameLengths[] = { 80, 120, 160, 240, 320, 480, 640, 960,
>-    kMaxFrameLength };
>+const size_t kFrameLengths[] = {
>+    80, 120, 160, 240, 320, 480, 640, 960, kMaxFrameLength};
> const size_t kFrameLengthsSize = sizeof(kFrameLengths) / sizeof(*kFrameLengths);
> 
> }  // namespace test
>@@ -39,8 +38,8 @@ const size_t kFrameLengthsSize = sizeof(kFrameLengths) / sizeof(*kFrameLengths);
> class VadTest : public ::testing::Test {
>  protected:
>   VadTest();
>-  virtual void SetUp();
>-  virtual void TearDown();
>+  void SetUp() override;
>+  void TearDown() override;
> 
>   // Returns true if the rate and frame length combination is valid.
>   bool ValidRatesAndFrameLengths(int rate, size_t frame_length);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/webrtc_vad.c b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/webrtc_vad.c
>index 7fc4d65ed27..d8457eeebe8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/webrtc_vad.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/vad/webrtc_vad.c
>@@ -15,7 +15,6 @@
> 
> #include "common_audio/signal_processing/include/signal_processing_library.h"
> #include "common_audio/vad/vad_core.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> static const int kInitCheck = 42;
> static const int kValidRates[] = { 8000, 16000, 32000, 48000 };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/wav_file.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/wav_file.cc
>index 37f249e7fa8..0209f52d2c7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/wav_file.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/wav_file.cc
>@@ -13,12 +13,13 @@
> #include <algorithm>
> #include <cstdio>
> #include <limits>
>-#include <sstream>
> 
> #include "common_audio/include/audio_util.h"
> #include "common_audio/wav_header.h"
> #include "rtc_base/checks.h"
>+#include "rtc_base/logging.h"
> #include "rtc_base/numerics/safe_conversions.h"
>+#include "rtc_base/system/arch.h"
> 
> namespace webrtc {
> 
>@@ -30,7 +31,7 @@ static const size_t kBytesPerSample = 2;
> class ReadableWavFile : public ReadableWav {
>  public:
>   explicit ReadableWavFile(FILE* file) : file_(file) {}
>-  virtual size_t Read(void* buf, size_t num_bytes) {
>+  size_t Read(void* buf, size_t num_bytes) override {
>     return fread(buf, 1, num_bytes, file_);
>   }
> 
>@@ -38,17 +39,22 @@ class ReadableWavFile : public ReadableWav {
>   FILE* file_;
> };
> 
>-std::string WavFile::FormatAsString() const {
>-  std::ostringstream s;
>-  s << "Sample rate: " << sample_rate() << " Hz, Channels: " << num_channels()
>-    << ", Duration: "
>-    << (1.f * num_samples()) / (num_channels() * sample_rate()) << " s";
>-  return s.str();
>-}
>-
> WavReader::WavReader(const std::string& filename)
>-    : file_handle_(fopen(filename.c_str(), "rb")) {
>-  RTC_CHECK(file_handle_) << "Could not open wav file for reading.";
>+    : WavReader(rtc::OpenPlatformFileReadOnly(filename)) {}
>+
>+WavReader::WavReader(rtc::PlatformFile file) {
>+  RTC_CHECK_NE(file, rtc::kInvalidPlatformFileValue)
>+      << "Invalid file. Could not create file handle for wav file.";
>+  file_handle_ = rtc::FdopenPlatformFile(file, "rb");
>+  if (!file_handle_) {
>+    RTC_LOG(LS_ERROR) << "Could not open wav file for reading: " << errno;
>+    // Even though we failed to open a FILE*, the file is still open
>+    // and needs to be closed.
>+    if (!rtc::ClosePlatformFile(file)) {
>+      RTC_LOG(LS_ERROR) << "Can't close file.";
>+    }
>+    RTC_FATAL() << "Could not open wav file for reading.";
>+  }
> 
>   ReadableWavFile readable(file_handle_);
>   WavFormat format;
>@@ -110,13 +116,31 @@ void WavReader::Close() {
>   file_handle_ = nullptr;
> }
> 
>-WavWriter::WavWriter(const std::string& filename, int sample_rate,
>+WavWriter::WavWriter(const std::string& filename,
>+                     int sample_rate,
>                      size_t num_channels)
>-    : sample_rate_(sample_rate),
>-      num_channels_(num_channels),
>-      num_samples_(0),
>-      file_handle_(fopen(filename.c_str(), "wb")) {
>-  RTC_CHECK(file_handle_) << "Could not open wav file for writing.";
>+    // Unlike plain fopen, CreatePlatformFile takes care of filename utf8 ->
>+    // wchar conversion on windows.
>+    : WavWriter(rtc::CreatePlatformFile(filename), sample_rate, num_channels) {}
>+
>+WavWriter::WavWriter(rtc::PlatformFile file,
>+                     int sample_rate,
>+                     size_t num_channels)
>+    : sample_rate_(sample_rate), num_channels_(num_channels), num_samples_(0) {
>+  // Handle errors from the CreatePlatformFile call in above constructor.
>+  RTC_CHECK_NE(file, rtc::kInvalidPlatformFileValue)
>+      << "Invalid file. Could not create wav file.";
>+  file_handle_ = rtc::FdopenPlatformFile(file, "wb");
>+  if (!file_handle_) {
>+    RTC_LOG(LS_ERROR) << "Could not open wav file for writing.";
>+    // Even though we failed to open a FILE*, the file is still open
>+    // and needs to be closed.
>+    if (!rtc::ClosePlatformFile(file)) {
>+      RTC_LOG(LS_ERROR) << "Can't close file.";
>+    }
>+    RTC_FATAL() << "Could not open wav file for writing.";
>+  }
>+
>   RTC_CHECK(CheckWavParameters(num_channels_, sample_rate_, kWavFormat,
>                                kBytesPerSample, num_samples_));
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/wav_file.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/wav_file.h
>index f7afe9239c6..cbce59d364a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/wav_file.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/wav_file.h
>@@ -18,6 +18,7 @@
> #include <string>
> 
> #include "rtc_base/constructormagic.h"
>+#include "rtc_base/platform_file.h"
> 
> namespace webrtc {
> 
>@@ -29,9 +30,6 @@ class WavFile {
>   virtual int sample_rate() const = 0;
>   virtual size_t num_channels() const = 0;
>   virtual size_t num_samples() const = 0;
>-
>-  // Returns a human-readable string containing the audio format.
>-  std::string FormatAsString() const;
> };
> 
> // Simple C++ class for writing 16-bit PCM WAV files. All error handling is
>@@ -41,6 +39,9 @@ class WavWriter final : public WavFile {
>   // Open a new WAV file for writing.
>   WavWriter(const std::string& filename, int sample_rate, size_t num_channels);
> 
>+  // Open a new WAV file for writing.
>+  WavWriter(rtc::PlatformFile file, int sample_rate, size_t num_channels);
>+
>   // Close the WAV file, after writing its header.
>   ~WavWriter() override;
> 
>@@ -59,7 +60,7 @@ class WavWriter final : public WavFile {
>   const int sample_rate_;
>   const size_t num_channels_;
>   size_t num_samples_;  // Total number of samples written to file.
>-  FILE* file_handle_;  // Output file, owned by this class
>+  FILE* file_handle_;   // Output file, owned by this class
> 
>   RTC_DISALLOW_COPY_AND_ASSIGN(WavWriter);
> };
>@@ -70,6 +71,9 @@ class WavReader final : public WavFile {
>   // Opens an existing WAV file for reading.
>   explicit WavReader(const std::string& filename);
> 
>+  // Opens an existing WAV file for reading.
>+  explicit WavReader(rtc::PlatformFile file);
>+
>   // Close the WAV file.
>   ~WavReader() override;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/wav_file_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/wav_file_unittest.cc
>index 7113b47c75c..d40229a1ebd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/wav_file_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/wav_file_unittest.cc
>@@ -19,12 +19,22 @@
> #include "test/gtest.h"
> #include "test/testsupport/fileutils.h"
> 
>+// WavWriterTest.CPPFileDescriptor and WavWriterTest.CPP flaky on Mac.
>+// See webrtc:9247.
>+#if defined(WEBRTC_MAC)
>+#define MAYBE_CPP DISABLED_CPP
>+#define MAYBE_CPPFileDescriptor DISABLED_CPPFileDescriptor
>+#else
>+#define MAYBE_CPP CPP
>+#define MAYBE_CPPFileDescriptor CPPFileDescriptor
>+#endif
>+
> namespace webrtc {
> 
> static const float kSamples[] = {0.0, 10.0, 4e4, -1e9};
> 
> // Write a tiny WAV file with the C++ interface and verify the result.
>-TEST(WavWriterTest, CPP) {
>+TEST(WavWriterTest, MAYBE_CPP) {
>   const std::string outfile = test::OutputPath() + "wavtest1.wav";
>   static const size_t kNumSamples = 3;
>   {
>@@ -46,6 +56,8 @@ TEST(WavWriterTest, CPP) {
>     fclose(f);
>   }
>   static const uint8_t kExpectedContents[] = {
>+      // clang-format off
>+      // clang formatting doesn't respect inline comments.
>     'R', 'I', 'F', 'F',
>     42, 0, 0, 0,  // size of whole file - 8: 6 + 44 - 8
>     'W', 'A', 'V', 'E',
>@@ -63,6 +75,7 @@ TEST(WavWriterTest, CPP) {
>     10, 0,  // second sample: 10.0
>     0xff, 0x7f,  // third sample: 4e4 (saturated)
>     kMetadata[0], kMetadata[1],
>+      // clang-format on
>   };
>   static const size_t kContentSize =
>       kWavHeaderSize + kNumSamples * sizeof(int16_t) + sizeof(kMetadata);
>@@ -102,6 +115,8 @@ TEST(WavWriterTest, C) {
>   EXPECT_EQ(kNumSamples, rtc_WavNumSamples(w));
>   rtc_WavClose(w);
>   static const uint8_t kExpectedContents[] = {
>+      // clang-format off
>+      // clang formatting doesn't respect inline comments.
>     'R', 'I', 'F', 'F',
>     44, 0, 0, 0,  // size of whole file - 8: 8 + 44 - 8
>     'W', 'A', 'V', 'E',
>@@ -119,6 +134,7 @@ TEST(WavWriterTest, C) {
>     10, 0,  // second sample: 10.0
>     0xff, 0x7f,  // third sample: 4e4 (saturated)
>     0, 0x80,  // fourth sample: -1e9 (saturated)
>+      // clang-format on
>   };
>   static const size_t kContentSize =
>       kWavHeaderSize + kNumSamples * sizeof(int16_t);
>@@ -174,4 +190,73 @@ TEST(WavWriterTest, LargeFile) {
>   }
> }
> 
>+// Write a tiny WAV file with the the std::FILE interface and verify the
>+// result.
>+TEST(WavWriterTest, MAYBE_CPPFileDescriptor) {
>+  const std::string outfile = test::OutputPath() + "wavtest1.wav";
>+  static constexpr size_t kNumSamples = 3;
>+  {
>+    WavWriter w(rtc::CreatePlatformFile(outfile), 14099, 1);
>+    EXPECT_EQ(14099, w.sample_rate());
>+    EXPECT_EQ(1u, w.num_channels());
>+    EXPECT_EQ(0u, w.num_samples());
>+    w.WriteSamples(kSamples, kNumSamples);
>+    EXPECT_EQ(kNumSamples, w.num_samples());
>+  }
>+  // Write some extra "metadata" to the file that should be silently ignored
>+  // by WavReader. We don't use WavWriter directly for this because it doesn't
>+  // support metadata.
>+  static constexpr uint8_t kMetadata[] = {101, 202};
>+  {
>+    FILE* f = fopen(outfile.c_str(), "ab");
>+    ASSERT_TRUE(f);
>+    ASSERT_EQ(1u, fwrite(kMetadata, sizeof(kMetadata), 1, f));
>+    fclose(f);
>+  }
>+  static const uint8_t kExpectedContents[] = {
>+      // clang-format off
>+      // clang formatting doesn't respect inline comments.
>+    'R', 'I', 'F', 'F',
>+    42, 0, 0, 0,       // size of whole file - 8: 6 + 44 - 8
>+    'W', 'A', 'V', 'E',
>+    'f', 'm', 't', ' ',
>+    16, 0, 0, 0,       // size of fmt block - 8: 24 - 8
>+    1, 0,              // format: PCM (1)
>+    1, 0,              // channels: 1
>+    0x13, 0x37, 0, 0,  // sample rate: 14099
>+    0x26, 0x6e, 0, 0,  // byte rate: 2 * 14099
>+    2, 0,              // block align: NumChannels * BytesPerSample
>+    16, 0,             // bits per sample: 2 * 8
>+    'd', 'a', 't', 'a',
>+    6, 0, 0, 0,        // size of payload: 6
>+    0, 0,              // first sample: 0.0
>+    10, 0,             // second sample: 10.0
>+    0xff, 0x7f,        // third sample: 4e4 (saturated)
>+    kMetadata[0], kMetadata[1],
>+      // clang-format on
>+  };
>+  static constexpr size_t kContentSize =
>+      kWavHeaderSize + kNumSamples * sizeof(int16_t) + sizeof(kMetadata);
>+  static_assert(sizeof(kExpectedContents) == kContentSize, "");
>+  EXPECT_EQ(kContentSize, test::GetFileSize(outfile));
>+  FILE* f = fopen(outfile.c_str(), "rb");
>+  ASSERT_TRUE(f);
>+  uint8_t contents[kContentSize];
>+  ASSERT_EQ(1u, fread(contents, kContentSize, 1, f));
>+  EXPECT_EQ(0, fclose(f));
>+  EXPECT_EQ(0, memcmp(kExpectedContents, contents, kContentSize));
>+
>+  {
>+    WavReader r(rtc::OpenPlatformFileReadOnly(outfile));
>+    EXPECT_EQ(14099, r.sample_rate());
>+    EXPECT_EQ(1u, r.num_channels());
>+    EXPECT_EQ(kNumSamples, r.num_samples());
>+    static constexpr float kTruncatedSamples[] = {0.0, 10.0, 32767.0};
>+    float samples[kNumSamples];
>+    EXPECT_EQ(kNumSamples, r.ReadSamples(kNumSamples, samples));
>+    EXPECT_EQ(0, memcmp(kTruncatedSamples, samples, sizeof(samples)));
>+    EXPECT_EQ(0u, r.ReadSamples(kNumSamples, samples));
>+  }
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/wav_header.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/wav_header.cc
>index a57e917249c..8fc5fef5085 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/wav_header.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/wav_header.cc
>@@ -21,6 +21,7 @@
> 
> #include "common_audio/include/audio_util.h"
> #include "rtc_base/checks.h"
>+#include "rtc_base/system/arch.h"
> 
> namespace webrtc {
> namespace {
>@@ -112,17 +113,23 @@ bool CheckWavParameters(size_t num_channels,
> }
> 
> #ifdef WEBRTC_ARCH_LITTLE_ENDIAN
>-static inline void WriteLE16(uint16_t* f, uint16_t x) { *f = x; }
>-static inline void WriteLE32(uint32_t* f, uint32_t x) { *f = x; }
>+static inline void WriteLE16(uint16_t* f, uint16_t x) {
>+  *f = x;
>+}
>+static inline void WriteLE32(uint32_t* f, uint32_t x) {
>+  *f = x;
>+}
> static inline void WriteFourCC(uint32_t* f, char a, char b, char c, char d) {
>-  *f = static_cast<uint32_t>(a)
>-      | static_cast<uint32_t>(b) << 8
>-      | static_cast<uint32_t>(c) << 16
>-      | static_cast<uint32_t>(d) << 24;
>+  *f = static_cast<uint32_t>(a) | static_cast<uint32_t>(b) << 8 |
>+       static_cast<uint32_t>(c) << 16 | static_cast<uint32_t>(d) << 24;
> }
> 
>-static inline uint16_t ReadLE16(uint16_t x) { return x; }
>-static inline uint32_t ReadLE32(uint32_t x) { return x; }
>+static inline uint16_t ReadLE16(uint16_t x) {
>+  return x;
>+}
>+static inline uint32_t ReadLE32(uint32_t x) {
>+  return x;
>+}
> static inline std::string ReadFourCC(uint32_t x) {
>   return std::string(reinterpret_cast<char*>(&x), 4);
> }
>@@ -131,11 +138,12 @@ static inline std::string ReadFourCC(uint32_t x) {
> #endif
> 
> static inline uint32_t RiffChunkSize(size_t bytes_in_payload) {
>-  return static_cast<uint32_t>(
>-      bytes_in_payload + kWavHeaderSize - sizeof(ChunkHeader));
>+  return static_cast<uint32_t>(bytes_in_payload + kWavHeaderSize -
>+                               sizeof(ChunkHeader));
> }
> 
>-static inline uint32_t ByteRate(size_t num_channels, int sample_rate,
>+static inline uint32_t ByteRate(size_t num_channels,
>+                                int sample_rate,
>                                 size_t bytes_per_sample) {
>   return static_cast<uint32_t>(num_channels * sample_rate * bytes_per_sample);
> }
>@@ -166,8 +174,8 @@ void WriteWavHeader(uint8_t* buf,
>   WriteLE16(&header.fmt.AudioFormat, format);
>   WriteLE16(&header.fmt.NumChannels, static_cast<uint16_t>(num_channels));
>   WriteLE32(&header.fmt.SampleRate, sample_rate);
>-  WriteLE32(&header.fmt.ByteRate, ByteRate(num_channels, sample_rate,
>-                                           bytes_per_sample));
>+  WriteLE32(&header.fmt.ByteRate,
>+            ByteRate(num_channels, sample_rate, bytes_per_sample));
>   WriteLE16(&header.fmt.BlockAlign, BlockAlign(num_channels, bytes_per_sample));
>   WriteLE16(&header.fmt.BitsPerSample,
>             static_cast<uint16_t>(8 * bytes_per_sample));
>@@ -239,5 +247,4 @@ bool ReadWavHeader(ReadableWav* readable,
>                             *bytes_per_sample, *num_samples);
> }
> 
>-
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/wav_header.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/wav_header.h
>index 2295fbe6adc..872d3abce5c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/wav_header.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/wav_header.h
>@@ -26,8 +26,8 @@ class ReadableWav {
> };
> 
> enum WavFormat {
>-  kWavFormatPcm   = 1,  // PCM, each sample of size bytes_per_sample
>-  kWavFormatALaw  = 6,  // 8-bit ITU-T G.711 A-law
>+  kWavFormatPcm = 1,    // PCM, each sample of size bytes_per_sample
>+  kWavFormatALaw = 6,   // 8-bit ITU-T G.711 A-law
>   kWavFormatMuLaw = 7,  // 8-bit ITU-T G.711 mu-law
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/wav_header_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/wav_header_unittest.cc
>index c6f605f7298..b7169b53c8f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/wav_header_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/wav_header_unittest.cc
>@@ -8,6 +8,7 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>+#include <string.h>
> #include <limits>
> 
> #include "common_audio/wav_header.h"
>@@ -31,13 +32,13 @@ class ReadableWavBuffer : public ReadableWav {
>         buf_exhausted_(false),
>         check_read_size_(check_read_size) {}
> 
>-  virtual ~ReadableWavBuffer() {
>+  ~ReadableWavBuffer() override {
>     // Verify the entire buffer has been read.
>     if (check_read_size_)
>       EXPECT_EQ(size_, pos_);
>   }
> 
>-  virtual size_t Read(void* buf, size_t num_bytes) {
>+  size_t Read(void* buf, size_t num_bytes) override {
>     // Verify we don't try to read outside of a properly sized header.
>     if (size_ >= kWavHeaderSize)
>       EXPECT_GE(size_, pos_ + num_bytes);
>@@ -83,8 +84,8 @@ TEST(WavHeaderTest, CheckWavParameters) {
> 
>   // Too large values.
>   EXPECT_FALSE(CheckWavParameters(1 << 20, 1 << 20, kWavFormatPcm, 1, 0));
>-  EXPECT_FALSE(CheckWavParameters(
>-      1, 8000, kWavFormatPcm, 1, std::numeric_limits<uint32_t>::max()));
>+  EXPECT_FALSE(CheckWavParameters(1, 8000, kWavFormatPcm, 1,
>+                                  std::numeric_limits<uint32_t>::max()));
> 
>   // Not the same number of samples for each channel.
>   EXPECT_FALSE(CheckWavParameters(3, 8000, kWavFormatPcm, 1, 5));
>@@ -103,6 +104,8 @@ TEST(WavHeaderTest, ReadWavHeaderWithErrors) {
>   // *BAD*.
>   {
>     static const uint8_t kBadRiffID[] = {
>+        // clang-format off
>+        // clang formatting doesn't respect inline comments.
>       'R', 'i', 'f', 'f',  // *BAD*
>       0xbd, 0xd0, 0x5b, 0x07,  // size of whole file - 8: 123457689 + 44 - 8
>       'W', 'A', 'V', 'E',
>@@ -116,14 +119,16 @@ TEST(WavHeaderTest, ReadWavHeaderWithErrors) {
>       8, 0,  // bits per sample: 1 * 8
>       'd', 'a', 't', 'a',
>       0x99, 0xd0, 0x5b, 0x07,  // size of payload: 123457689
>+        // clang-format on
>     };
>     ReadableWavBuffer r(kBadRiffID, sizeof(kBadRiffID));
>-    EXPECT_FALSE(
>-        ReadWavHeader(&r, &num_channels, &sample_rate, &format,
>-                      &bytes_per_sample, &num_samples));
>+    EXPECT_FALSE(ReadWavHeader(&r, &num_channels, &sample_rate, &format,
>+                               &bytes_per_sample, &num_samples));
>   }
>   {
>     static const uint8_t kBadBitsPerSample[] = {
>+        // clang-format off
>+        // clang formatting doesn't respect inline comments.
>       'R', 'I', 'F', 'F',
>       0xbd, 0xd0, 0x5b, 0x07,  // size of whole file - 8: 123457689 + 44 - 8
>       'W', 'A', 'V', 'E',
>@@ -137,14 +142,16 @@ TEST(WavHeaderTest, ReadWavHeaderWithErrors) {
>       1, 0,  // bits per sample: *BAD*
>       'd', 'a', 't', 'a',
>       0x99, 0xd0, 0x5b, 0x07,  // size of payload: 123457689
>+        // clang-format on
>     };
>     ReadableWavBuffer r(kBadBitsPerSample, sizeof(kBadBitsPerSample));
>-    EXPECT_FALSE(
>-        ReadWavHeader(&r, &num_channels, &sample_rate, &format,
>-                      &bytes_per_sample, &num_samples));
>+    EXPECT_FALSE(ReadWavHeader(&r, &num_channels, &sample_rate, &format,
>+                               &bytes_per_sample, &num_samples));
>   }
>   {
>     static const uint8_t kBadByteRate[] = {
>+        // clang-format off
>+        // clang formatting doesn't respect inline comments.
>       'R', 'I', 'F', 'F',
>       0xbd, 0xd0, 0x5b, 0x07,  // size of whole file - 8: 123457689 + 44 - 8
>       'W', 'A', 'V', 'E',
>@@ -158,14 +165,16 @@ TEST(WavHeaderTest, ReadWavHeaderWithErrors) {
>       8, 0,  // bits per sample: 1 * 8
>       'd', 'a', 't', 'a',
>       0x99, 0xd0, 0x5b, 0x07,  // size of payload: 123457689
>+        // clang-format on
>     };
>     ReadableWavBuffer r(kBadByteRate, sizeof(kBadByteRate));
>-    EXPECT_FALSE(
>-        ReadWavHeader(&r, &num_channels, &sample_rate, &format,
>-                      &bytes_per_sample, &num_samples));
>+    EXPECT_FALSE(ReadWavHeader(&r, &num_channels, &sample_rate, &format,
>+                               &bytes_per_sample, &num_samples));
>   }
>   {
>     static const uint8_t kBadFmtHeaderSize[] = {
>+        // clang-format off
>+        // clang formatting doesn't respect inline comments.
>       'R', 'I', 'F', 'F',
>       0xbd, 0xd0, 0x5b, 0x07,  // size of whole file - 8: 123457689 + 44 - 8
>       'W', 'A', 'V', 'E',
>@@ -180,14 +189,16 @@ TEST(WavHeaderTest, ReadWavHeaderWithErrors) {
>       0,  // extra (though invalid) header byte
>       'd', 'a', 't', 'a',
>       0x99, 0xd0, 0x5b, 0x07,  // size of payload: 123457689
>+        // clang-format on
>     };
>     ReadableWavBuffer r(kBadFmtHeaderSize, sizeof(kBadFmtHeaderSize), false);
>-    EXPECT_FALSE(
>-        ReadWavHeader(&r, &num_channels, &sample_rate, &format,
>-                      &bytes_per_sample, &num_samples));
>+    EXPECT_FALSE(ReadWavHeader(&r, &num_channels, &sample_rate, &format,
>+                               &bytes_per_sample, &num_samples));
>   }
>   {
>     static const uint8_t kNonZeroExtensionField[] = {
>+        // clang-format off
>+        // clang formatting doesn't respect inline comments.
>       'R', 'I', 'F', 'F',
>       0xbd, 0xd0, 0x5b, 0x07,  // size of whole file - 8: 123457689 + 44 - 8
>       'W', 'A', 'V', 'E',
>@@ -202,15 +213,17 @@ TEST(WavHeaderTest, ReadWavHeaderWithErrors) {
>       1, 0,  // non-zero extension field *BAD*
>       'd', 'a', 't', 'a',
>       0x99, 0xd0, 0x5b, 0x07,  // size of payload: 123457689
>+        // clang-format on
>     };
>     ReadableWavBuffer r(kNonZeroExtensionField, sizeof(kNonZeroExtensionField),
>                         false);
>-    EXPECT_FALSE(
>-        ReadWavHeader(&r, &num_channels, &sample_rate, &format,
>-                      &bytes_per_sample, &num_samples));
>+    EXPECT_FALSE(ReadWavHeader(&r, &num_channels, &sample_rate, &format,
>+                               &bytes_per_sample, &num_samples));
>   }
>   {
>     static const uint8_t kMissingDataChunk[] = {
>+        // clang-format off
>+        // clang formatting doesn't respect inline comments.
>       'R', 'I', 'F', 'F',
>       0xbd, 0xd0, 0x5b, 0x07,  // size of whole file - 8: 123457689 + 44 - 8
>       'W', 'A', 'V', 'E',
>@@ -222,23 +235,25 @@ TEST(WavHeaderTest, ReadWavHeaderWithErrors) {
>       0xc9, 0x33, 0x03, 0,  // byte rate: 1 * 17 * 12345
>       17, 0,  // block align: NumChannels * BytesPerSample
>       8, 0,  // bits per sample: 1 * 8
>+        // clang-format on
>     };
>     ReadableWavBuffer r(kMissingDataChunk, sizeof(kMissingDataChunk));
>-    EXPECT_FALSE(
>-        ReadWavHeader(&r, &num_channels, &sample_rate, &format,
>-                      &bytes_per_sample, &num_samples));
>+    EXPECT_FALSE(ReadWavHeader(&r, &num_channels, &sample_rate, &format,
>+                               &bytes_per_sample, &num_samples));
>   }
>   {
>     static const uint8_t kMissingFmtAndDataChunks[] = {
>+        // clang-format off
>+        // clang formatting doesn't respect inline comments.
>       'R', 'I', 'F', 'F',
>       0xbd, 0xd0, 0x5b, 0x07,  // size of whole file - 8: 123457689 + 44 - 8
>       'W', 'A', 'V', 'E',
>+        // clang-format on
>     };
>     ReadableWavBuffer r(kMissingFmtAndDataChunks,
>                         sizeof(kMissingFmtAndDataChunks));
>-    EXPECT_FALSE(
>-        ReadWavHeader(&r, &num_channels, &sample_rate, &format,
>-                      &bytes_per_sample, &num_samples));
>+    EXPECT_FALSE(ReadWavHeader(&r, &num_channels, &sample_rate, &format,
>+                               &bytes_per_sample, &num_samples));
>   }
> }
> 
>@@ -249,6 +264,8 @@ TEST(WavHeaderTest, WriteAndReadWavHeader) {
>   memset(buf, 0xa4, sizeof(buf));
>   WriteWavHeader(buf + 4, 17, 12345, kWavFormatALaw, 1, 123457689);
>   static const uint8_t kExpectedBuf[] = {
>+      // clang-format off
>+      // clang formatting doesn't respect inline comments.
>     0xa4, 0xa4, 0xa4, 0xa4,  // untouched bytes before header
>     'R', 'I', 'F', 'F',
>     0xbd, 0xd0, 0x5b, 0x07,  // size of whole file - 8: 123457689 + 44 - 8
>@@ -264,6 +281,7 @@ TEST(WavHeaderTest, WriteAndReadWavHeader) {
>     'd', 'a', 't', 'a',
>     0x99, 0xd0, 0x5b, 0x07,  // size of payload: 123457689
>     0xa4, 0xa4, 0xa4, 0xa4,  // untouched bytes after header
>+      // clang-format on
>   };
>   static_assert(sizeof(kExpectedBuf) == kSize, "buffer size");
>   EXPECT_EQ(0, memcmp(kExpectedBuf, buf, kSize));
>@@ -274,9 +292,8 @@ TEST(WavHeaderTest, WriteAndReadWavHeader) {
>   size_t bytes_per_sample = 0;
>   size_t num_samples = 0;
>   ReadableWavBuffer r(buf + 4, sizeof(buf) - 8);
>-  EXPECT_TRUE(
>-      ReadWavHeader(&r, &num_channels, &sample_rate, &format,
>-                    &bytes_per_sample, &num_samples));
>+  EXPECT_TRUE(ReadWavHeader(&r, &num_channels, &sample_rate, &format,
>+                            &bytes_per_sample, &num_samples));
>   EXPECT_EQ(17u, num_channels);
>   EXPECT_EQ(12345, sample_rate);
>   EXPECT_EQ(kWavFormatALaw, format);
>@@ -287,6 +304,8 @@ TEST(WavHeaderTest, WriteAndReadWavHeader) {
> // Try reading an atypical but valid WAV header and make sure it's parsed OK.
> TEST(WavHeaderTest, ReadAtypicalWavHeader) {
>   static const uint8_t kBuf[] = {
>+      // clang-format off
>+      // clang formatting doesn't respect inline comments.
>     'R', 'I', 'F', 'F',
>     0x3d, 0xd1, 0x5b, 0x07,  // size of whole file - 8 + an extra 128 bytes of
>                              // "metadata": 123457689 + 44 - 8 + 128. (atypical)
>@@ -302,6 +321,7 @@ TEST(WavHeaderTest, ReadAtypicalWavHeader) {
>     0, 0,  // zero extension size field (atypical)
>     'd', 'a', 't', 'a',
>     0x99, 0xd0, 0x5b, 0x07,  // size of payload: 123457689
>+      // clang-format on
>   };
> 
>   size_t num_channels = 0;
>@@ -310,9 +330,8 @@ TEST(WavHeaderTest, ReadAtypicalWavHeader) {
>   size_t bytes_per_sample = 0;
>   size_t num_samples = 0;
>   ReadableWavBuffer r(kBuf, sizeof(kBuf));
>-  EXPECT_TRUE(
>-      ReadWavHeader(&r, &num_channels, &sample_rate, &format,
>-                    &bytes_per_sample, &num_samples));
>+  EXPECT_TRUE(ReadWavHeader(&r, &num_channels, &sample_rate, &format,
>+                            &bytes_per_sample, &num_samples));
>   EXPECT_EQ(17u, num_channels);
>   EXPECT_EQ(12345, sample_rate);
>   EXPECT_EQ(kWavFormatALaw, format);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/window_generator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/window_generator.cc
>index 823d2b7387e..da5603d9e7a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/window_generator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/window_generator.cc
>@@ -25,12 +25,11 @@ namespace {
> complex<float> I0(complex<float> x) {
>   complex<float> y = x / 3.75f;
>   y *= y;
>-  return 1.0f + y * (
>-    3.5156229f + y * (
>-      3.0899424f + y * (
>-        1.2067492f + y * (
>-          0.2659732f + y * (
>-            0.360768e-1f + y * 0.45813e-2f)))));
>+  return 1.0f + y * (3.5156229f +
>+                     y * (3.0899424f +
>+                          y * (1.2067492f +
>+                               y * (0.2659732f +
>+                                    y * (0.360768e-1f + y * 0.45813e-2f)))));
> }
> 
> }  // namespace
>@@ -41,12 +40,13 @@ void WindowGenerator::Hanning(int length, float* window) {
>   RTC_CHECK_GT(length, 1);
>   RTC_CHECK(window != nullptr);
>   for (int i = 0; i < length; ++i) {
>-    window[i] = 0.5f * (1 - cosf(2 * static_cast<float>(M_PI) * i /
>-                                 (length - 1)));
>+    window[i] =
>+        0.5f * (1 - cosf(2 * static_cast<float>(M_PI) * i / (length - 1)));
>   }
> }
> 
>-void WindowGenerator::KaiserBesselDerived(float alpha, size_t length,
>+void WindowGenerator::KaiserBesselDerived(float alpha,
>+                                          size_t length,
>                                           float* window) {
>   RTC_CHECK_GT(length, 1U);
>   RTC_CHECK(window != nullptr);
>@@ -69,4 +69,3 @@ void WindowGenerator::KaiserBesselDerived(float alpha, size_t length,
> }
> 
> }  // namespace webrtc
>-
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/window_generator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/window_generator.h
>index 5fc738eaf07..ad3b4456b9e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/window_generator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/window_generator.h
>@@ -30,4 +30,3 @@ class WindowGenerator {
> }  // namespace webrtc
> 
> #endif  // COMMON_AUDIO_WINDOW_GENERATOR_H_
>-
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/window_generator_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/window_generator_unittest.cc
>index b2089d471aa..cf339327c6a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/window_generator_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_audio/window_generator_unittest.cc
>@@ -89,4 +89,3 @@ TEST(WindowGeneratorTest, Hanning) {
> }
> 
> }  // namespace webrtc
>-
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_types.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_types.cc
>deleted file mode 100644
>index 07f77b77778..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_types.cc
>+++ /dev/null
>@@ -1,234 +0,0 @@
>-/*
>- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "common_types.h"  // NOLINT(build/include)
>-
>-#include <string.h>
>-#include <algorithm>
>-#include <limits>
>-#include <type_traits>
>-
>-#include "rtc_base/checks.h"
>-#include "rtc_base/stringutils.h"
>-
>-namespace webrtc {
>-
>-VideoCodec::VideoCodec()
>-    : codecType(kVideoCodecUnknown),
>-      plName(),
>-      plType(0),
>-      width(0),
>-      height(0),
>-      startBitrate(0),
>-      maxBitrate(0),
>-      minBitrate(0),
>-      targetBitrate(0),
>-      maxFramerate(0),
>-      qpMax(0),
>-      numberOfSimulcastStreams(0),
>-      simulcastStream(),
>-      spatialLayers(),
>-      mode(kRealtimeVideo),
>-      expect_encode_from_texture(false),
>-      timing_frame_thresholds({0, 0}),
>-      codec_specific_() {}
>-
>-VideoCodecVP8* VideoCodec::VP8() {
>-  RTC_DCHECK_EQ(codecType, kVideoCodecVP8);
>-  return &codec_specific_.VP8;
>-}
>-
>-const VideoCodecVP8& VideoCodec::VP8() const {
>-  RTC_DCHECK_EQ(codecType, kVideoCodecVP8);
>-  return codec_specific_.VP8;
>-}
>-
>-VideoCodecVP9* VideoCodec::VP9() {
>-  RTC_DCHECK_EQ(codecType, kVideoCodecVP9);
>-  return &codec_specific_.VP9;
>-}
>-
>-const VideoCodecVP9& VideoCodec::VP9() const {
>-  RTC_DCHECK_EQ(codecType, kVideoCodecVP9);
>-  return codec_specific_.VP9;
>-}
>-
>-VideoCodecH264* VideoCodec::H264() {
>-  RTC_DCHECK_EQ(codecType, kVideoCodecH264);
>-  return &codec_specific_.H264;
>-}
>-
>-const VideoCodecH264& VideoCodec::H264() const {
>-  RTC_DCHECK_EQ(codecType, kVideoCodecH264);
>-  return codec_specific_.H264;
>-}
>-
>-static const char* kPayloadNameVp8 = "VP8";
>-static const char* kPayloadNameVp9 = "VP9";
>-static const char* kPayloadNameH264 = "H264";
>-static const char* kPayloadNameI420 = "I420";
>-static const char* kPayloadNameRED = "RED";
>-static const char* kPayloadNameULPFEC = "ULPFEC";
>-static const char* kPayloadNameGeneric = "Generic";
>-static const char* kPayloadNameStereo = "Stereo";
>-
>-static bool CodecNamesEq(const char* name1, const char* name2) {
>-  return _stricmp(name1, name2) == 0;
>-}
>-
>-const char* CodecTypeToPayloadString(VideoCodecType type) {
>-  switch (type) {
>-    case kVideoCodecVP8:
>-      return kPayloadNameVp8;
>-    case kVideoCodecVP9:
>-      return kPayloadNameVp9;
>-    case kVideoCodecH264:
>-      return kPayloadNameH264;
>-    case kVideoCodecI420:
>-      return kPayloadNameI420;
>-    case kVideoCodecRED:
>-      return kPayloadNameRED;
>-    case kVideoCodecULPFEC:
>-      return kPayloadNameULPFEC;
>-    // Other codecs default to generic.
>-    case kVideoCodecStereo:
>-    case kVideoCodecFlexfec:
>-    case kVideoCodecGeneric:
>-    case kVideoCodecUnknown:
>-      return kPayloadNameGeneric;
>-  }
>-  return kPayloadNameGeneric;
>-}
>-
>-VideoCodecType PayloadStringToCodecType(const std::string& name) {
>-  if (CodecNamesEq(name.c_str(), kPayloadNameVp8))
>-    return kVideoCodecVP8;
>-  if (CodecNamesEq(name.c_str(), kPayloadNameVp9))
>-    return kVideoCodecVP9;
>-  if (CodecNamesEq(name.c_str(), kPayloadNameH264))
>-    return kVideoCodecH264;
>-  if (CodecNamesEq(name.c_str(), kPayloadNameI420))
>-    return kVideoCodecI420;
>-  if (CodecNamesEq(name.c_str(), kPayloadNameRED))
>-    return kVideoCodecRED;
>-  if (CodecNamesEq(name.c_str(), kPayloadNameULPFEC))
>-    return kVideoCodecULPFEC;
>-  if (CodecNamesEq(name.c_str(), kPayloadNameStereo))
>-    return kVideoCodecStereo;
>-  return kVideoCodecGeneric;
>-}
>-
>-const uint32_t BitrateAllocation::kMaxBitrateBps =
>-    std::numeric_limits<uint32_t>::max();
>-
>-BitrateAllocation::BitrateAllocation() : sum_(0), bitrates_{}, has_bitrate_{} {}
>-
>-bool BitrateAllocation::SetBitrate(size_t spatial_index,
>-                                   size_t temporal_index,
>-                                   uint32_t bitrate_bps) {
>-  RTC_CHECK_LT(spatial_index, kMaxSpatialLayers);
>-  RTC_CHECK_LT(temporal_index, kMaxTemporalStreams);
>-  RTC_CHECK_LE(bitrates_[spatial_index][temporal_index], sum_);
>-  uint64_t new_bitrate_sum_bps = sum_;
>-  new_bitrate_sum_bps -= bitrates_[spatial_index][temporal_index];
>-  new_bitrate_sum_bps += bitrate_bps;
>-  if (new_bitrate_sum_bps > kMaxBitrateBps)
>-    return false;
>-
>-  bitrates_[spatial_index][temporal_index] = bitrate_bps;
>-  has_bitrate_[spatial_index][temporal_index] = true;
>-  sum_ = static_cast<uint32_t>(new_bitrate_sum_bps);
>-  return true;
>-}
>-
>-bool BitrateAllocation::HasBitrate(size_t spatial_index,
>-                                   size_t temporal_index) const {
>-  RTC_CHECK_LT(spatial_index, kMaxSpatialLayers);
>-  RTC_CHECK_LT(temporal_index, kMaxTemporalStreams);
>-  return has_bitrate_[spatial_index][temporal_index];
>-}
>-
>-uint32_t BitrateAllocation::GetBitrate(size_t spatial_index,
>-                                       size_t temporal_index) const {
>-  RTC_CHECK_LT(spatial_index, kMaxSpatialLayers);
>-  RTC_CHECK_LT(temporal_index, kMaxTemporalStreams);
>-  return bitrates_[spatial_index][temporal_index];
>-}
>-
>-// Whether the specific spatial layers has the bitrate set in any of its
>-// temporal layers.
>-bool BitrateAllocation::IsSpatialLayerUsed(size_t spatial_index) const {
>-  RTC_CHECK_LT(spatial_index, kMaxSpatialLayers);
>-  for (int i = 0; i < kMaxTemporalStreams; ++i) {
>-    if (has_bitrate_[spatial_index][i])
>-      return true;
>-  }
>-  return false;
>-}
>-
>-// Get the sum of all the temporal layer for a specific spatial layer.
>-uint32_t BitrateAllocation::GetSpatialLayerSum(size_t spatial_index) const {
>-  RTC_CHECK_LT(spatial_index, kMaxSpatialLayers);
>-  uint32_t sum = 0;
>-  for (int i = 0; i < kMaxTemporalStreams; ++i)
>-    sum += bitrates_[spatial_index][i];
>-  return sum;
>-}
>-
>-std::string BitrateAllocation::ToString() const {
>-  if (sum_ == 0)
>-    return "BitrateAllocation [ [] ]";
>-
>-  // TODO(sprang): Replace this stringstream with something cheaper.
>-  std::ostringstream oss;
>-  oss << "BitrateAllocation [";
>-  uint32_t spatial_cumulator = 0;
>-  for (int si = 0; si < kMaxSpatialLayers; ++si) {
>-    RTC_DCHECK_LE(spatial_cumulator, sum_);
>-    if (spatial_cumulator == sum_)
>-      break;
>-
>-    const uint32_t layer_sum = GetSpatialLayerSum(si);
>-    if (layer_sum == sum_) {
>-      oss << " [";
>-    } else {
>-      if (si > 0)
>-        oss << ",";
>-      oss << std::endl << "  [";
>-    }
>-    spatial_cumulator += layer_sum;
>-
>-    uint32_t temporal_cumulator = 0;
>-    for (int ti = 0; ti < kMaxTemporalStreams; ++ti) {
>-      RTC_DCHECK_LE(temporal_cumulator, layer_sum);
>-      if (temporal_cumulator == layer_sum)
>-        break;
>-
>-      if (ti > 0)
>-        oss << ", ";
>-
>-      uint32_t bitrate = bitrates_[si][ti];
>-      oss << bitrate;
>-      temporal_cumulator += bitrate;
>-    }
>-    oss << "]";
>-  }
>-
>-  RTC_DCHECK_EQ(spatial_cumulator, sum_);
>-  oss << " ]";
>-  return oss.str();
>-}
>-
>-std::ostream& BitrateAllocation::operator<<(std::ostream& os) const {
>-  os << ToString();
>-  return os;
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_types.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_types.h
>index 9f1ad8f9f79..c9eeee434e6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_types.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_types.h
>@@ -13,15 +13,14 @@
> 
> #include <stddef.h>
> #include <string.h>
>-#include <ostream>
> #include <string>
> #include <vector>
> 
> #include "api/array_view.h"
>-#include "api/optional.h"
>+// TODO(sprang): Remove this include when all usage includes it directly.
>+#include "api/video/video_bitrate_allocation.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/deprecation.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> #if defined(_MSC_VER)
> // Disable "new behavior: elements of array will be default initialized"
>@@ -29,18 +28,6 @@
> #pragma warning(disable : 4351)
> #endif
> 
>-#if defined(WEBRTC_EXPORT)
>-#define WEBRTC_DLLEXPORT _declspec(dllexport)
>-#elif defined(WEBRTC_DLL)
>-#define WEBRTC_DLLEXPORT _declspec(dllimport)
>-#else
>-#define WEBRTC_DLLEXPORT
>-#endif
>-
>-#ifndef NULL
>-#define NULL 0
>-#endif
>-
> #define RTP_PAYLOAD_NAME_SIZE 32u
> 
> #if defined(WEBRTC_WIN) || defined(WIN32)
>@@ -55,34 +42,6 @@
> 
> namespace webrtc {
> 
>-class RewindableStream {
>- public:
>-  virtual ~RewindableStream() {}
>-  virtual int Rewind() = 0;
>-};
>-
>-class InStream : public RewindableStream {
>- public:
>-  // Reads |len| bytes from file to |buf|. Returns the number of bytes read
>-  // or -1 on error.
>-  virtual int Read(void* buf, size_t len) = 0;
>-};
>-
>-class OutStream : public RewindableStream {
>- public:
>-  // Writes |len| bytes from |buf| to file. The actual writing may happen
>-  // some time later. Call Flush() to force a write.
>-  virtual bool Write(const void* buf, size_t len) = 0;
>-};
>-
>-// For the deprecated MediaFile module.
>-enum FileFormats {
>-  kFileFormatWavFile = 1,
>-  kFileFormatPcm16kHzFile = 7,
>-  kFileFormatPcm8kHzFile = 8,
>-  kFileFormatPcm32kHzFile = 9,
>-};
>-
> enum FrameType {
>   kEmptyFrame = 0,
>   kAudioFrameSpeech = 1,
>@@ -100,14 +59,8 @@ struct RtcpStatistics {
>         jitter(0) {}
> 
>   uint8_t fraction_lost;
>-  union {
>-    int32_t packets_lost;  // Defined as a 24 bit signed integer in RTCP
>-    RTC_DEPRECATED uint32_t cumulative_lost;
>-  };
>-  union {
>-    uint32_t extended_highest_sequence_number;
>-    RTC_DEPRECATED uint32_t extended_max_sequence_number;
>-  };
>+  int32_t packets_lost;  // Defined as a 24 bit signed integer in RTCP
>+  uint32_t extended_highest_sequence_number;
>   uint32_t jitter;
> };
> 
>@@ -186,14 +139,6 @@ class RtcpPacketTypeCounterObserver {
>       const RtcpPacketTypeCounter& packet_counter) = 0;
> };
> 
>-// Rate statistics for a stream.
>-struct BitrateStatistics {
>-  BitrateStatistics() : bitrate_bps(0), packet_rate(0) {}
>-
>-  uint32_t bitrate_bps;  // Bitrate in bits per second.
>-  uint32_t packet_rate;  // Packet rate in packets per second.
>-};
>-
> // Callback, used to notify an observer whenever new rates have been estimated.
> class BitrateStatisticsObserver {
>  public:
>@@ -268,35 +213,11 @@ struct CodecInst {
>   }
> 
>   bool operator!=(const CodecInst& other) const { return !(*this == other); }
>-
>-  friend std::ostream& operator<<(std::ostream& os, const CodecInst& ci) {
>-    os << "{pltype: " << ci.pltype;
>-    os << ", plname: " << ci.plname;
>-    os << ", plfreq: " << ci.plfreq;
>-    os << ", pacsize: " << ci.pacsize;
>-    os << ", channels: " << ci.channels;
>-    os << ", rate: " << ci.rate << "}";
>-    return os;
>-  }
> };
> 
> // RTP
> enum { kRtpCsrcSize = 15 };  // RFC 3550 page 13
> 
>-enum PayloadFrequencies {
>-  kFreq8000Hz = 8000,
>-  kFreq16000Hz = 16000,
>-  kFreq32000Hz = 32000
>-};
>-
>-// Degree of bandwidth reduction.
>-enum VadModes {
>-  kVadConventional = 0,  // lowest reduction
>-  kVadAggressiveLow,
>-  kVadAggressiveMid,
>-  kVadAggressiveHigh  // highest reduction
>-};
>-
> // NETEQ statistics.
> struct NetworkStatistics {
>   // current jitter buffer size in ms
>@@ -367,7 +288,7 @@ struct AudioDecodingCallStats {
>   int decoded_normal;  // Number of calls where audio RTP packet decoded.
>   int decoded_plc;     // Number of calls resulted in PLC.
>   int decoded_cng;  // Number of calls where comfort noise generated due to DTX.
>-  int decoded_plc_cng;  // Number of calls resulted where PLC faded to CNG.
>+  int decoded_plc_cng;       // Number of calls resulted where PLC faded to CNG.
>   int decoded_muted_output;  // Number of calls returning a muted state output.
> };
> 
>@@ -396,60 +317,6 @@ enum class VideoType {
>   kBGRA,
> };
> 
>-// Video codec
>-enum { kPayloadNameSize = 32 };
>-enum { kMaxSimulcastStreams = 4 };
>-enum { kMaxSpatialLayers = 5 };
>-enum { kMaxTemporalStreams = 4 };
>-
>-enum VideoCodecComplexity {
>-  kComplexityNormal = 0,
>-  kComplexityHigh = 1,
>-  kComplexityHigher = 2,
>-  kComplexityMax = 3
>-};
>-
>-enum VP8ResilienceMode {
>-  kResilienceOff,    // The stream produced by the encoder requires a
>-                     // recovery frame (typically a key frame) to be
>-                     // decodable after a packet loss.
>-  kResilientStream,  // A stream produced by the encoder is resilient to
>-                     // packet losses, but packets within a frame subsequent
>-                     // to a loss can't be decoded.
>-  kResilientFrames   // Same as kResilientStream but with added resilience
>-                     // within a frame.
>-};
>-
>-class TemporalLayersFactory;
>-// VP8 specific
>-struct VideoCodecVP8 {
>-  // TODO(nisse): Unused, delete?
>-  bool pictureLossIndicationOn;
>-  VideoCodecComplexity complexity;
>-  VP8ResilienceMode resilience;
>-  unsigned char numberOfTemporalLayers;
>-  bool denoisingOn;
>-  bool errorConcealmentOn;
>-  bool automaticResizeOn;
>-  bool frameDroppingOn;
>-  int keyFrameInterval;
>-  TemporalLayersFactory* tl_factory;
>-};
>-
>-// VP9 specific.
>-struct VideoCodecVP9 {
>-  VideoCodecComplexity complexity;
>-  bool resilienceOn;
>-  unsigned char numberOfTemporalLayers;
>-  bool denoisingOn;
>-  bool frameDroppingOn;
>-  int keyFrameInterval;
>-  bool adaptiveQpMode;
>-  bool automaticResizeOn;
>-  unsigned char numberOfSpatialLayers;
>-  bool flexibleMode;
>-};
>-
> // TODO(magjed): Move this and other H264 related classes out to their own file.
> namespace H264 {
> 
>@@ -463,45 +330,35 @@ enum Profile {
> 
> }  // namespace H264
> 
>-// H264 specific.
>-struct VideoCodecH264 {
>-  bool frameDroppingOn;
>-  int keyFrameInterval;
>-  // These are NULL/0 if not externally negotiated.
>-  const uint8_t* spsData;
>-  size_t spsLen;
>-  const uint8_t* ppsData;
>-  size_t ppsLen;
>-  H264::Profile profile;
>-};
>-
> // Video codec types
> enum VideoCodecType {
>+  // There are various memset(..., 0, ...) calls in the code that rely on
>+  // kVideoCodecGeneric being zero.
>+  kVideoCodecGeneric = 0,
>   kVideoCodecVP8,
>   kVideoCodecVP9,
>   kVideoCodecH264,
>   kVideoCodecI420,
>-  kVideoCodecRED,
>-  kVideoCodecULPFEC,
>-  kVideoCodecFlexfec,
>-  kVideoCodecGeneric,
>-  kVideoCodecStereo,
>-  kVideoCodecUnknown
>+  kVideoCodecMultiplex,
>+  // DEPRECATED. Do not use.
>+  kVideoCodecUnknown,
>+
>+  // TODO(nisse): Deprecated aliases, for code expecting RtpVideoCodecTypes.
>+  kRtpVideoNone = kVideoCodecGeneric,
>+  kRtpVideoGeneric = kVideoCodecGeneric,
>+  kRtpVideoVp8 = kVideoCodecVP8,
>+  kRtpVideoVp9 = kVideoCodecVP9,
>+  kRtpVideoH264 = kVideoCodecH264,
> };
> 
> // Translates from name of codec to codec type and vice versa.
> const char* CodecTypeToPayloadString(VideoCodecType type);
> VideoCodecType PayloadStringToCodecType(const std::string& name);
> 
>-union VideoCodecUnion {
>-  VideoCodecVP8 VP8;
>-  VideoCodecVP9 VP9;
>-  VideoCodecH264 H264;
>-};
>+struct SpatialLayer {
>+  bool operator==(const SpatialLayer& other) const;
>+  bool operator!=(const SpatialLayer& other) const { return !(*this == other); }
> 
>-// Simulcast is when the same stream is encoded multiple times with different
>-// settings such as resolution.
>-struct SimulcastStream {
>   unsigned short width;
>   unsigned short height;
>   unsigned char numberOfTemporalLayers;
>@@ -509,117 +366,15 @@ struct SimulcastStream {
>   unsigned int targetBitrate;  // kilobits/sec.
>   unsigned int minBitrate;     // kilobits/sec.
>   unsigned int qpMax;          // minimum quality
>+  bool active;                 // encoded and sent.
> };
> 
>-struct SpatialLayer {
>-  int scaling_factor_num;
>-  int scaling_factor_den;
>-  int target_bitrate_bps;
>-  // TODO(ivica): Add max_quantizer and min_quantizer?
>-};
>-
>-enum VideoCodecMode { kRealtimeVideo, kScreensharing };
>-
>-// Common video codec properties
>-class VideoCodec {
>- public:
>-  VideoCodec();
>-
>-  // Public variables. TODO(hta): Make them private with accessors.
>-  VideoCodecType codecType;
>-  char plName[kPayloadNameSize];
>-  unsigned char plType;
>-
>-  unsigned short width;
>-  unsigned short height;
>-
>-  unsigned int startBitrate;   // kilobits/sec.
>-  unsigned int maxBitrate;     // kilobits/sec.
>-  unsigned int minBitrate;     // kilobits/sec.
>-  unsigned int targetBitrate;  // kilobits/sec.
>-
>-  uint32_t maxFramerate;
>-
>-  unsigned int qpMax;
>-  unsigned char numberOfSimulcastStreams;
>-  SimulcastStream simulcastStream[kMaxSimulcastStreams];
>-  SpatialLayer spatialLayers[kMaxSpatialLayers];
>-
>-  VideoCodecMode mode;
>-  bool expect_encode_from_texture;
>-
>-  // Timing frames configuration. There is delay of delay_ms between two
>-  // consequent timing frames, excluding outliers. Frame is always made a
>-  // timing frame if it's at least outlier_ratio in percent of "ideal" average
>-  // frame given bitrate and framerate, i.e. if it's bigger than
>-  // |outlier_ratio / 100.0 * bitrate_bps / fps| in bits. This way, timing
>-  // frames will not be sent too often usually. Yet large frames will always
>-  // have timing information for debug purposes because they are more likely to
>-  // cause extra delays.
>-  struct TimingFrameTriggerThresholds {
>-    int64_t delay_ms;
>-    uint16_t outlier_ratio_percent;
>-  } timing_frame_thresholds;
>-
>-  bool operator==(const VideoCodec& other) const = delete;
>-  bool operator!=(const VideoCodec& other) const = delete;
>-
>-  // Accessors for codec specific information.
>-  // There is a const version of each that returns a reference,
>-  // and a non-const version that returns a pointer, in order
>-  // to allow modification of the parameters.
>-  VideoCodecVP8* VP8();
>-  const VideoCodecVP8& VP8() const;
>-  VideoCodecVP9* VP9();
>-  const VideoCodecVP9& VP9() const;
>-  VideoCodecH264* H264();
>-  const VideoCodecH264& H264() const;
>-
>- private:
>-  // TODO(hta): Consider replacing the union with a pointer type.
>-  // This will allow removing the VideoCodec* types from this file.
>-  VideoCodecUnion codec_specific_;
>-};
>-
>-class BitrateAllocation {
>- public:
>-  static const uint32_t kMaxBitrateBps;
>-  BitrateAllocation();
>-
>-  bool SetBitrate(size_t spatial_index,
>-                  size_t temporal_index,
>-                  uint32_t bitrate_bps);
>-
>-  bool HasBitrate(size_t spatial_index, size_t temporal_index) const;
>-
>-  uint32_t GetBitrate(size_t spatial_index, size_t temporal_index) const;
>-
>-  // Whether the specific spatial layers has the bitrate set in any of its
>-  // temporal layers.
>-  bool IsSpatialLayerUsed(size_t spatial_index) const;
>-
>-  // Get the sum of all the temporal layer for a specific spatial layer.
>-  uint32_t GetSpatialLayerSum(size_t spatial_index) const;
>-
>-  uint32_t get_sum_bps() const { return sum_; }  // Sum of all bitrates.
>-  uint32_t get_sum_kbps() const { return (sum_ + 500) / 1000; }
>-
>-  inline bool operator==(const BitrateAllocation& other) const {
>-    return memcmp(bitrates_, other.bitrates_, sizeof(bitrates_)) == 0;
>-  }
>-  inline bool operator!=(const BitrateAllocation& other) const {
>-    return !(*this == other);
>-  }
>-
>-  // Expensive, please use only in tests.
>-  std::string ToString() const;
>-  std::ostream& operator<<(std::ostream& os) const;
>+// Simulcast is when the same stream is encoded multiple times with different
>+// settings such as resolution.
>+typedef SpatialLayer SimulcastStream;
> 
>- private:
>-  uint32_t sum_;
>-  uint32_t bitrates_[kMaxSpatialLayers][kMaxTemporalStreams];
>-  bool has_bitrate_[kMaxSpatialLayers][kMaxTemporalStreams];
>-};
>+// TODO(sprang): Remove this when downstream projects have been updated.
>+using BitrateAllocation = VideoBitrateAllocation;
> 
> // Bandwidth over-use detector options.  These are used to drive
> // experimentation with bandwidth estimation parameters.
>@@ -649,6 +404,9 @@ struct OverUseDetectorOptions {
>   double initial_var_noise;
> };
> 
>+// TODO(nisse): This struct is phased out, delete as soon as down stream code is
>+// updated.
>+
> // This structure will have the information about when packet is actually
> // received by socket.
> struct PacketTime {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/BUILD.gn
>index 1c298fdba09..5bc4eb4ad38 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/BUILD.gn
>@@ -8,10 +8,6 @@
> 
> import("../webrtc.gni")
> 
>-config("common_video_config") {
>-  include_dirs = [ "include" ]
>-}
>-
> rtc_static_library("common_video") {
>   visibility = [ "*" ]
> 
>@@ -45,28 +41,19 @@ rtc_static_library("common_video") {
>     "video_render_frames.h",
>   ]
> 
>-  include_dirs = [ "../modules/interface" ]
>-
>-  public_configs = [ ":common_video_config" ]
>-
>-  if (!build_with_chromium && is_clang) {
>-    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>-    suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-  }
>-
>   deps = [
>     "..:webrtc_common",
>-    "../:typedefs",
>-    "../api:libjingle_peerconnection_api",
>-    "../api:optional",
>-    "../api:video_frame_api",
>-    "../api:video_frame_api_i420",
>+    "../api/video:video_bitrate_allocation",
>+    "../api/video:video_bitrate_allocator",
>+    "../api/video:video_frame",
>+    "../api/video:video_frame_i420",
>     "../media:rtc_h264_profile_id",
>     "../modules:module_api",
>     "../rtc_base:checks",
>     "../rtc_base:rtc_base",
>     "../rtc_base:rtc_task_queue",
>-    "../system_wrappers",
>+    "../rtc_base:safe_minmax",
>+    "//third_party/abseil-cpp/absl/types:optional",
>     "//third_party/libyuv",
>   ]
> }
>@@ -95,29 +82,22 @@ if (rtc_include_tests) {
>       "h264/sps_parser_unittest.cc",
>       "h264/sps_vui_rewriter_unittest.cc",
>       "i420_buffer_pool_unittest.cc",
>-      "i420_video_frame_unittest.cc",
>       "libyuv/libyuv_unittest.cc",
>+      "video_frame_unittest.cc",
>     ]
> 
>-    # TODO(jschuh): Bug 1348: fix this warning.
>-    configs += [ "//build/config/compiler:no_size_t_to_int_warning" ]
>-
>-    if (!build_with_chromium && is_clang) {
>-      # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>-      suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-    }
>-
>     deps = [
>       ":common_video",
>-      "../api:video_frame_api",
>-      "../api:video_frame_api_i420",
>+      "../api/video:video_frame",
>+      "../api/video:video_frame_i010",
>+      "../api/video:video_frame_i420",
>       "../modules/video_capture:video_capture",
>       "../rtc_base:rtc_base",
>       "../rtc_base:rtc_base_approved",
>-      "../system_wrappers:system_wrappers",
>+      "../rtc_base:rtc_base_tests_utils",
>+      "../test:fileutils",
>       "../test:test_main",
>       "../test:video_test_common",
>-      "//testing/gmock",
>       "//testing/gtest",
>       "//third_party/libyuv",
>     ]
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/bitrate_adjuster.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/bitrate_adjuster.cc
>index 0d3458c6c84..163c4b19814 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/bitrate_adjuster.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/bitrate_adjuster.cc
>@@ -15,7 +15,7 @@
> 
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>-#include "system_wrappers/include/clock.h"
>+#include "rtc_base/timeutils.h"
> 
> namespace webrtc {
> 
>@@ -30,11 +30,9 @@ const float BitrateAdjuster::kBitrateTolerancePct = .1f;
> 
> const float BitrateAdjuster::kBytesPerMsToBitsPerSecond = 8 * 1000;
> 
>-BitrateAdjuster::BitrateAdjuster(Clock* clock,
>-                                 float min_adjusted_bitrate_pct,
>+BitrateAdjuster::BitrateAdjuster(float min_adjusted_bitrate_pct,
>                                  float max_adjusted_bitrate_pct)
>-    : clock_(clock),
>-      min_adjusted_bitrate_pct_(min_adjusted_bitrate_pct),
>+    : min_adjusted_bitrate_pct_(min_adjusted_bitrate_pct),
>       max_adjusted_bitrate_pct_(max_adjusted_bitrate_pct),
>       bitrate_tracker_(1.5 * kBitrateUpdateIntervalMs,
>                        kBytesPerMsToBitsPerSecond) {
>@@ -70,14 +68,14 @@ uint32_t BitrateAdjuster::GetAdjustedBitrateBps() const {
>   return adjusted_bitrate_bps_;
> }
> 
>-rtc::Optional<uint32_t> BitrateAdjuster::GetEstimatedBitrateBps() {
>+absl::optional<uint32_t> BitrateAdjuster::GetEstimatedBitrateBps() {
>   rtc::CritScope cs(&crit_);
>-  return bitrate_tracker_.Rate(clock_->TimeInMilliseconds());
>+  return bitrate_tracker_.Rate(rtc::TimeMillis());
> }
> 
> void BitrateAdjuster::Update(size_t frame_size) {
>   rtc::CritScope cs(&crit_);
>-  uint32_t current_time_ms = clock_->TimeInMilliseconds();
>+  uint32_t current_time_ms = rtc::TimeMillis();
>   bitrate_tracker_.Update(frame_size, current_time_ms);
>   UpdateBitrate(current_time_ms);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/bitrate_adjuster_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/bitrate_adjuster_unittest.cc
>index b75f184cf2d..6e8d3fb856d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/bitrate_adjuster_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/bitrate_adjuster_unittest.cc
>@@ -9,7 +9,7 @@
>  */
> 
> #include "common_video/include/bitrate_adjuster.h"
>-#include "system_wrappers/include/clock.h"
>+#include "rtc_base/fakeclock.h"
> #include "test/gtest.h"
> 
> namespace webrtc {
>@@ -17,8 +17,7 @@ namespace webrtc {
> class BitrateAdjusterTest : public ::testing::Test {
>  public:
>   BitrateAdjusterTest()
>-      : clock_(0),
>-        adjuster_(&clock_, kMinAdjustedBitratePct, kMaxAdjustedBitratePct) {}
>+      : adjuster_(kMinAdjustedBitratePct, kMaxAdjustedBitratePct) {}
> 
>   // Simulate an output bitrate for one update cycle of BitrateAdjuster.
>   void SimulateBitrateBps(uint32_t bitrate_bps) {
>@@ -33,7 +32,7 @@ class BitrateAdjusterTest : public ::testing::Test {
>     const size_t frame_size_bytes =
>         (bitrate_bps * frame_interval_ms) / (8 * 1000);
>     for (size_t i = 0; i < update_frame_interval; ++i) {
>-      clock_.AdvanceTimeMilliseconds(frame_interval_ms);
>+      clock_.AdvanceTime(webrtc::TimeDelta::ms(frame_interval_ms));
>       adjuster_.Update(frame_size_bytes);
>     }
>   }
>@@ -63,7 +62,7 @@ class BitrateAdjusterTest : public ::testing::Test {
>  protected:
>   static const float kMinAdjustedBitratePct;
>   static const float kMaxAdjustedBitratePct;
>-  SimulatedClock clock_;
>+  rtc::ScopedFakeClock clock_;
>   BitrateAdjuster adjuster_;
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/h264_bitstream_parser.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/h264_bitstream_parser.cc
>index 439beecec29..d8f8a62378c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/h264_bitstream_parser.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/h264_bitstream_parser.cc
>@@ -22,7 +22,7 @@ namespace {
> const int kMaxAbsQpDeltaValue = 51;
> const int kMinQpValue = 0;
> const int kMaxQpValue = 51;
>-}
>+}  // namespace
> 
> namespace webrtc {
> 
>@@ -44,7 +44,7 @@ H264BitstreamParser::Result H264BitstreamParser::ParseNonParameterSetNalu(
>   if (!sps_ || !pps_)
>     return kInvalidStream;
> 
>-  last_slice_qp_delta_ = rtc::nullopt;
>+  last_slice_qp_delta_ = absl::nullopt;
>   const std::vector<uint8_t> slice_rbsp =
>       H264::ParseRbsp(source, source_length);
>   if (slice_rbsp.size() < H264::kNaluTypeSize)
>@@ -247,8 +247,8 @@ H264BitstreamParser::Result H264BitstreamParser::ParseNonParameterSetNalu(
>       }
>     }
>   }
>-  if (pps_->entropy_coding_mode_flag &&
>-      slice_type != H264::SliceType::kI && slice_type != H264::SliceType::kSi) {
>+  if (pps_->entropy_coding_mode_flag && slice_type != H264::SliceType::kI &&
>+      slice_type != H264::SliceType::kSi) {
>     // cabac_init_idc: ue(v)
>     RETURN_INV_ON_FAIL(slice_reader.ReadExponentialGolomb(&golomb_tmp));
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/h264_bitstream_parser.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/h264_bitstream_parser.h
>index 4ef6b40b440..b3fac7bb215 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/h264_bitstream_parser.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/h264_bitstream_parser.h
>@@ -13,7 +13,7 @@
> #include <stddef.h>
> #include <stdint.h>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> #include "common_video/h264/pps_parser.h"
> #include "common_video/h264/sps_parser.h"
> 
>@@ -53,11 +53,11 @@ class H264BitstreamParser {
>                                   uint8_t nalu_type);
> 
>   // SPS/PPS state, updated when parsing new SPS/PPS, used to parse slices.
>-  rtc::Optional<SpsParser::SpsState> sps_;
>-  rtc::Optional<PpsParser::PpsState> pps_;
>+  absl::optional<SpsParser::SpsState> sps_;
>+  absl::optional<PpsParser::PpsState> pps_;
> 
>   // Last parsed slice QP.
>-  rtc::Optional<int32_t> last_slice_qp_delta_;
>+  absl::optional<int32_t> last_slice_qp_delta_;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/h264_bitstream_parser_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/h264_bitstream_parser_unittest.cc
>index 3bfbdb27042..1509d677535 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/h264_bitstream_parser_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/h264_bitstream_parser_unittest.cc
>@@ -28,9 +28,9 @@ uint8_t kH264BitstreamChunk[] = {
> };
> 
> uint8_t kH264BitstreamChunkCabac[] = {
>-  0x00, 0x00, 0x00, 0x01, 0x27, 0x64, 0x00, 0x0d, 0xac, 0x52, 0x30, 0x50,
>-  0x7e, 0xc0, 0x5a, 0x81, 0x01, 0x01, 0x18, 0x56, 0xbd, 0xef, 0x80, 0x80,
>-  0x00, 0x00, 0x00, 0x01, 0x28, 0xfe, 0x09, 0x8b,
>+    0x00, 0x00, 0x00, 0x01, 0x27, 0x64, 0x00, 0x0d, 0xac, 0x52, 0x30,
>+    0x50, 0x7e, 0xc0, 0x5a, 0x81, 0x01, 0x01, 0x18, 0x56, 0xbd, 0xef,
>+    0x80, 0x80, 0x00, 0x00, 0x00, 0x01, 0x28, 0xfe, 0x09, 0x8b,
> };
> 
> // Contains enough of the image slice to contain slice QP.
>@@ -40,8 +40,8 @@ uint8_t kH264BitstreamNextImageSliceChunk[] = {
> 
> // Contains enough of the image slice to contain slice QP.
> uint8_t kH264BitstreamNextImageSliceChunkCabac[] = {
>-  0x00, 0x00, 0x00, 0x01, 0x21, 0xe1, 0x05, 0x11, 0x3f, 0x9a, 0xae, 0x46,
>-  0x70, 0xbf, 0xc1, 0x4a, 0x16, 0x8f, 0x51, 0xf4, 0xca, 0xfb, 0xa3, 0x65,
>+    0x00, 0x00, 0x00, 0x01, 0x21, 0xe1, 0x05, 0x11, 0x3f, 0x9a, 0xae, 0x46,
>+    0x70, 0xbf, 0xc1, 0x4a, 0x16, 0x8f, 0x51, 0xf4, 0xca, 0xfb, 0xa3, 0x65,
> };
> 
> TEST(H264BitstreamParserTest, ReportsNoQpWithoutParsedSlices) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/pps_parser.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/pps_parser.cc
>index 407b15c7227..5bc29f35925 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/pps_parser.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/pps_parser.cc
>@@ -19,13 +19,13 @@
> 
> #define RETURN_EMPTY_ON_FAIL(x) \
>   if (!(x)) {                   \
>-    return rtc::nullopt;        \
>+    return absl::nullopt;       \
>   }
> 
> namespace {
> const int kMaxPicInitQpDeltaValue = 25;
> const int kMinPicInitQpDeltaValue = -26;
>-}
>+}  // namespace
> 
> namespace webrtc {
> 
>@@ -33,8 +33,8 @@ namespace webrtc {
> // You can find it on this page:
> // http://www.itu.int/rec/T-REC-H.264
> 
>-rtc::Optional<PpsParser::PpsState> PpsParser::ParsePps(const uint8_t* data,
>-                                                       size_t length) {
>+absl::optional<PpsParser::PpsState> PpsParser::ParsePps(const uint8_t* data,
>+                                                        size_t length) {
>   // First, parse out rbsp, which is basically the source buffer minus emulation
>   // bytes (the last byte of a 0x00 0x00 0x03 sequence). RBSP is defined in
>   // section 7.3.1 of the H.264 standard.
>@@ -57,26 +57,26 @@ bool PpsParser::ParsePpsIds(const uint8_t* data,
>   return ParsePpsIdsInternal(&bit_buffer, pps_id, sps_id);
> }
> 
>-rtc::Optional<uint32_t> PpsParser::ParsePpsIdFromSlice(const uint8_t* data,
>-                                                       size_t length) {
>+absl::optional<uint32_t> PpsParser::ParsePpsIdFromSlice(const uint8_t* data,
>+                                                        size_t length) {
>   std::vector<uint8_t> unpacked_buffer = H264::ParseRbsp(data, length);
>   rtc::BitBuffer slice_reader(unpacked_buffer.data(), unpacked_buffer.size());
> 
>   uint32_t golomb_tmp;
>   // first_mb_in_slice: ue(v)
>   if (!slice_reader.ReadExponentialGolomb(&golomb_tmp))
>-    return rtc::nullopt;
>+    return absl::nullopt;
>   // slice_type: ue(v)
>   if (!slice_reader.ReadExponentialGolomb(&golomb_tmp))
>-    return rtc::nullopt;
>+    return absl::nullopt;
>   // pic_parameter_set_id: ue(v)
>   uint32_t slice_pps_id;
>   if (!slice_reader.ReadExponentialGolomb(&slice_pps_id))
>-    return rtc::nullopt;
>+    return absl::nullopt;
>   return slice_pps_id;
> }
> 
>-rtc::Optional<PpsParser::PpsState> PpsParser::ParseInternal(
>+absl::optional<PpsParser::PpsState> PpsParser::ParseInternal(
>     rtc::BitBuffer* bit_buffer) {
>   PpsState pps;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/pps_parser.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/pps_parser.h
>index 571af9773a0..d6c31b06887 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/pps_parser.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/pps_parser.h
>@@ -11,7 +11,7 @@
> #ifndef COMMON_VIDEO_H264_PPS_PARSER_H_
> #define COMMON_VIDEO_H264_PPS_PARSER_H_
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> 
> namespace rtc {
> class BitBuffer;
>@@ -38,20 +38,20 @@ class PpsParser {
>   };
> 
>   // Unpack RBSP and parse PPS state from the supplied buffer.
>-  static rtc::Optional<PpsState> ParsePps(const uint8_t* data, size_t length);
>+  static absl::optional<PpsState> ParsePps(const uint8_t* data, size_t length);
> 
>   static bool ParsePpsIds(const uint8_t* data,
>                           size_t length,
>                           uint32_t* pps_id,
>                           uint32_t* sps_id);
> 
>-  static rtc::Optional<uint32_t> ParsePpsIdFromSlice(const uint8_t* data,
>-                                                     size_t length);
>+  static absl::optional<uint32_t> ParsePpsIdFromSlice(const uint8_t* data,
>+                                                      size_t length);
> 
>  protected:
>   // Parse the PPS state, for a bit buffer where RBSP decoding has already been
>   // performed.
>-  static rtc::Optional<PpsState> ParseInternal(rtc::BitBuffer* bit_buffer);
>+  static absl::optional<PpsState> ParseInternal(rtc::BitBuffer* bit_buffer);
>   static bool ParsePpsIdsInternal(rtc::BitBuffer* bit_buffer,
>                                   uint32_t* pps_id,
>                                   uint32_t* sps_id);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/pps_parser_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/pps_parser_unittest.cc
>index 14a425f6c4e..bfafecfc654 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/pps_parser_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/pps_parser_unittest.cc
>@@ -140,7 +140,7 @@ void WritePps(const PpsParser::PpsState& pps,
> class PpsParserTest : public ::testing::Test {
>  public:
>   PpsParserTest() {}
>-  virtual ~PpsParserTest() {}
>+  ~PpsParserTest() override {}
> 
>   void RunTest() {
>     VerifyParsing(generated_pps_, 0, 1, 0);
>@@ -192,7 +192,7 @@ class PpsParserTest : public ::testing::Test {
> 
>   PpsParser::PpsState generated_pps_;
>   rtc::Buffer buffer_;
>-  rtc::Optional<PpsParser::PpsState> parsed_pps_;
>+  absl::optional<PpsParser::PpsState> parsed_pps_;
> };
> 
> TEST_F(PpsParserTest, ZeroPps) {
>@@ -215,7 +215,7 @@ TEST_F(PpsParserTest, MaxPps) {
> }
> 
> TEST_F(PpsParserTest, PpsIdFromSlice) {
>-  rtc::Optional<uint32_t> pps_id = PpsParser::ParsePpsIdFromSlice(
>+  absl::optional<uint32_t> pps_id = PpsParser::ParsePpsIdFromSlice(
>       kH264BitstreamChunk, sizeof(kH264BitstreamChunk));
>   ASSERT_TRUE(pps_id);
>   EXPECT_EQ(2u, *pps_id);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/profile_level_id_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/profile_level_id_unittest.cc
>index d7b7cfb3518..66ad300131e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/profile_level_id_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/profile_level_id_unittest.cc
>@@ -125,7 +125,7 @@ TEST(H264ProfileLevelId, TestToStringInvalid) {
> }
> 
> TEST(H264ProfileLevelId, TestParseSdpProfileLevelIdEmpty) {
>-  const rtc::Optional<ProfileLevelId> profile_level_id =
>+  const absl::optional<ProfileLevelId> profile_level_id =
>       ParseSdpProfileLevelId(CodecParameterMap());
>   EXPECT_TRUE(profile_level_id);
>   EXPECT_EQ(kProfileConstrainedBaseline, profile_level_id->profile);
>@@ -135,7 +135,7 @@ TEST(H264ProfileLevelId, TestParseSdpProfileLevelIdEmpty) {
> TEST(H264ProfileLevelId, TestParseSdpProfileLevelIdConstrainedHigh) {
>   CodecParameterMap params;
>   params["profile-level-id"] = "640c2a";
>-  const rtc::Optional<ProfileLevelId> profile_level_id =
>+  const absl::optional<ProfileLevelId> profile_level_id =
>       ParseSdpProfileLevelId(params);
>   EXPECT_TRUE(profile_level_id);
>   EXPECT_EQ(kProfileConstrainedHigh, profile_level_id->profile);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/sps_parser.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/sps_parser.cc
>index 2ee7391f8c4..ce12834fe1e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/sps_parser.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/sps_parser.cc
>@@ -18,7 +18,7 @@
> #include "rtc_base/logging.h"
> 
> namespace {
>-typedef rtc::Optional<webrtc::SpsParser::SpsState> OptionalSps;
>+typedef absl::optional<webrtc::SpsParser::SpsState> OptionalSps;
> 
> #define RETURN_EMPTY_ON_FAIL(x) \
>   if (!(x)) {                   \
>@@ -31,19 +31,23 @@ constexpr int kScaldingDeltaMax = 127;
> 
> namespace webrtc {
> 
>+SpsParser::SpsState::SpsState() = default;
>+SpsParser::SpsState::SpsState(const SpsState&) = default;
>+SpsParser::SpsState::~SpsState() = default;
>+
> // General note: this is based off the 02/2014 version of the H.264 standard.
> // You can find it on this page:
> // http://www.itu.int/rec/T-REC-H.264
> 
> // Unpack RBSP and parse SPS state from the supplied buffer.
>-rtc::Optional<SpsParser::SpsState> SpsParser::ParseSps(const uint8_t* data,
>-                                                       size_t length) {
>+absl::optional<SpsParser::SpsState> SpsParser::ParseSps(const uint8_t* data,
>+                                                        size_t length) {
>   std::vector<uint8_t> unpacked_buffer = H264::ParseRbsp(data, length);
>   rtc::BitBuffer bit_buffer(unpacked_buffer.data(), unpacked_buffer.size());
>   return ParseSpsUpToVui(&bit_buffer);
> }
> 
>-rtc::Optional<SpsParser::SpsState> SpsParser::ParseSpsUpToVui(
>+absl::optional<SpsParser::SpsState> SpsParser::ParseSpsUpToVui(
>     rtc::BitBuffer* buffer) {
>   // Now, we need to use a bit buffer to parse through the actual AVC SPS
>   // format. See Section 7.3.2.1.1 ("Sequence parameter set data syntax") of the
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/sps_parser.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/sps_parser.h
>index 0ccc481356d..386d4689369 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/sps_parser.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/sps_parser.h
>@@ -11,7 +11,7 @@
> #ifndef COMMON_VIDEO_H264_SPS_PARSER_H_
> #define COMMON_VIDEO_H264_SPS_PARSER_H_
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> 
> namespace rtc {
> class BitBuffer;
>@@ -25,7 +25,9 @@ class SpsParser {
>   // The parsed state of the SPS. Only some select values are stored.
>   // Add more as they are actually needed.
>   struct SpsState {
>-    SpsState() = default;
>+    SpsState();
>+    SpsState(const SpsState&);
>+    ~SpsState();
> 
>     uint32_t width = 0;
>     uint32_t height = 0;
>@@ -41,12 +43,12 @@ class SpsParser {
>   };
> 
>   // Unpack RBSP and parse SPS state from the supplied buffer.
>-  static rtc::Optional<SpsState> ParseSps(const uint8_t* data, size_t length);
>+  static absl::optional<SpsState> ParseSps(const uint8_t* data, size_t length);
> 
>  protected:
>   // Parse the SPS state, up till the VUI part, for a bit buffer where RBSP
>   // decoding has already been performed.
>-  static rtc::Optional<SpsState> ParseSpsUpToVui(rtc::BitBuffer* buffer);
>+  static absl::optional<SpsState> ParseSpsUpToVui(rtc::BitBuffer* buffer);
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/sps_parser_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/sps_parser_unittest.cc
>index 6856c1bbf2a..e88d0c31191 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/sps_parser_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/sps_parser_unittest.cc
>@@ -110,9 +110,9 @@ void GenerateFakeSps(uint16_t width,
> class H264SpsParserTest : public ::testing::Test {
>  public:
>   H264SpsParserTest() {}
>-  virtual ~H264SpsParserTest() {}
>+  ~H264SpsParserTest() override {}
> 
>-  rtc::Optional<SpsParser::SpsState> sps_;
>+  absl::optional<SpsParser::SpsState> sps_;
> };
> 
> TEST_F(H264SpsParserTest, TestSampleSPSHdLandscape) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/sps_vui_rewriter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/sps_vui_rewriter.cc
>index c346865090e..749b62e6d21 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/sps_vui_rewriter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/sps_vui_rewriter.cc
>@@ -72,13 +72,13 @@ bool CopyRemainingBits(rtc::BitBuffer* source,
> SpsVuiRewriter::ParseResult SpsVuiRewriter::ParseAndRewriteSps(
>     const uint8_t* buffer,
>     size_t length,
>-    rtc::Optional<SpsParser::SpsState>* sps,
>+    absl::optional<SpsParser::SpsState>* sps,
>     rtc::Buffer* destination) {
>   // Create temporary RBSP decoded buffer of the payload (exlcuding the
>   // leading nalu type header byte (the SpsParser uses only the payload).
>   std::vector<uint8_t> rbsp_buffer = H264::ParseRbsp(buffer, length);
>   rtc::BitBuffer source_buffer(rbsp_buffer.data(), rbsp_buffer.size());
>-  rtc::Optional<SpsParser::SpsState> sps_state =
>+  absl::optional<SpsParser::SpsState> sps_state =
>       SpsParser::ParseSpsUpToVui(&source_buffer);
>   if (!sps_state)
>     return ParseResult::kFailure;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/sps_vui_rewriter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/sps_vui_rewriter.h
>index f639c0d8931..233051d2d61 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/sps_vui_rewriter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/sps_vui_rewriter.h
>@@ -12,7 +12,7 @@
> #ifndef COMMON_VIDEO_H264_SPS_VUI_REWRITER_H_
> #define COMMON_VIDEO_H264_SPS_VUI_REWRITER_H_
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> #include "common_video/h264/sps_parser.h"
> #include "rtc_base/buffer.h"
> 
>@@ -43,10 +43,11 @@ class SpsVuiRewriter : private SpsParser {
>   // SPS state. This function assumes that any previous headers
>   // (NALU start, type, Stap-A, etc) have already been parsed and that RBSP
>   // decoding has been performed.
>-  static ParseResult ParseAndRewriteSps(const uint8_t* buffer,
>-                                        size_t length,
>-                                        rtc::Optional<SpsParser::SpsState>* sps,
>-                                        rtc::Buffer* destination);
>+  static ParseResult ParseAndRewriteSps(
>+      const uint8_t* buffer,
>+      size_t length,
>+      absl::optional<SpsParser::SpsState>* sps,
>+      rtc::Buffer* destination);
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/sps_vui_rewriter_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/sps_vui_rewriter_unittest.cc
>index 0de5d3372dc..9464de80771 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/sps_vui_rewriter_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/h264/sps_vui_rewriter_unittest.cc
>@@ -159,7 +159,7 @@ void TestSps(SpsMode mode, SpsVuiRewriter::ParseResult expected_parse_result) {
>   index.payload_start_offset += H264::kNaluTypeSize;
>   index.payload_size -= H264::kNaluTypeSize;
> 
>-  rtc::Optional<SpsParser::SpsState> sps;
>+  absl::optional<SpsParser::SpsState> sps;
>   rtc::Buffer out_buffer;
>   SpsVuiRewriter::ParseResult result =
>       SpsVuiRewriter::ParseAndRewriteSps(&buffer[index.payload_start_offset],
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/i420_buffer_pool.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/i420_buffer_pool.cc
>index 48d6512971d..c403eeb4fcb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/i420_buffer_pool.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/i420_buffer_pool.cc
>@@ -14,10 +14,14 @@
> 
> namespace webrtc {
> 
>+I420BufferPool::I420BufferPool() : I420BufferPool(false) {}
>+I420BufferPool::I420BufferPool(bool zero_initialize)
>+    : I420BufferPool(zero_initialize, std::numeric_limits<size_t>::max()) {}
> I420BufferPool::I420BufferPool(bool zero_initialize,
>                                size_t max_number_of_buffers)
>     : zero_initialize_(zero_initialize),
>       max_number_of_buffers_(max_number_of_buffers) {}
>+I420BufferPool::~I420BufferPool() = default;
> 
> void I420BufferPool::Release() {
>   buffers_.clear();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/i420_video_frame_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/i420_video_frame_unittest.cc
>deleted file mode 100644
>index 7004d9dab14..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/i420_video_frame_unittest.cc
>+++ /dev/null
>@@ -1,298 +0,0 @@
>-/*
>- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include <math.h>
>-#include <string.h>
>-
>-#include "api/video/i420_buffer.h"
>-#include "api/video/video_frame.h"
>-#include "rtc_base/bind.h"
>-#include "rtc_base/timeutils.h"
>-#include "test/fake_texture_frame.h"
>-#include "test/frame_utils.h"
>-#include "test/gtest.h"
>-
>-namespace webrtc {
>-
>-namespace {
>-
>-rtc::scoped_refptr<I420Buffer> CreateGradient(int width, int height) {
>-  rtc::scoped_refptr<I420Buffer> buffer(
>-      I420Buffer::Create(width, height));
>-  // Initialize with gradient, Y = 128(x/w + y/h), U = 256 x/w, V = 256 y/h
>-  for (int x = 0; x < width; x++) {
>-    for (int y = 0; y < height; y++) {
>-      buffer->MutableDataY()[x + y * width] =
>-          128 * (x * height + y * width) / (width * height);
>-    }
>-  }
>-  int chroma_width = buffer->ChromaWidth();
>-  int chroma_height = buffer->ChromaHeight();
>-  for (int x = 0; x < chroma_width; x++) {
>-    for (int y = 0; y < chroma_height; y++) {
>-      buffer->MutableDataU()[x + y * chroma_width] =
>-          255 * x / (chroma_width - 1);
>-      buffer->MutableDataV()[x + y * chroma_width] =
>-          255 * y / (chroma_height - 1);
>-    }
>-  }
>-  return buffer;
>-}
>-
>-// The offsets and sizes describe the rectangle extracted from the
>-// original (gradient) frame, in relative coordinates where the
>-// original frame correspond to the unit square, 0.0 <= x, y < 1.0.
>-void CheckCrop(const webrtc::I420BufferInterface& frame,
>-               double offset_x,
>-               double offset_y,
>-               double rel_width,
>-               double rel_height) {
>-  int width = frame.width();
>-  int height = frame.height();
>-  // Check that pixel values in the corners match the gradient used
>-  // for initialization.
>-  for (int i = 0; i < 2; i++) {
>-    for (int j = 0; j < 2; j++) {
>-      // Pixel coordinates of the corner.
>-      int x = i * (width - 1);
>-      int y = j * (height - 1);
>-      // Relative coordinates, range 0.0 - 1.0 correspond to the
>-      // size of the uncropped input frame.
>-      double orig_x = offset_x + i * rel_width;
>-      double orig_y = offset_y + j * rel_height;
>-
>-      EXPECT_NEAR(frame.DataY()[x + y * frame.StrideY()] / 256.0,
>-                  (orig_x + orig_y) / 2, 0.02);
>-      EXPECT_NEAR(frame.DataU()[x / 2 + (y / 2) * frame.StrideU()] / 256.0,
>-                  orig_x, 0.02);
>-      EXPECT_NEAR(frame.DataV()[x / 2 + (y / 2) * frame.StrideV()] / 256.0,
>-                  orig_y, 0.02);
>-    }
>-  }
>-}
>-
>-void CheckRotate(int width,
>-                 int height,
>-                 webrtc::VideoRotation rotation,
>-                 const webrtc::I420BufferInterface& rotated) {
>-  int rotated_width = width;
>-  int rotated_height = height;
>-
>-  if (rotation == kVideoRotation_90 || rotation == kVideoRotation_270) {
>-    std::swap(rotated_width, rotated_height);
>-  }
>-  EXPECT_EQ(rotated_width, rotated.width());
>-  EXPECT_EQ(rotated_height, rotated.height());
>-
>-  // Clock-wise order (with 0,0 at top-left)
>-  const struct { int x; int y; } corners[] = {
>-    { 0, 0 }, { 1, 0 }, { 1, 1 }, { 0, 1 }
>-  };
>-  // Corresponding corner colors of the frame produced by CreateGradient.
>-  const struct { int y; int u; int v; } colors[] = {
>-    {0, 0, 0}, { 127, 255, 0}, { 255, 255, 255 }, {127, 0, 255}
>-  };
>-  int corner_offset = static_cast<int>(rotation) / 90;
>-
>-  for (int i = 0; i < 4; i++) {
>-    int j = (i + corner_offset) % 4;
>-    int x = corners[j].x * (rotated_width - 1);
>-    int y = corners[j].y * (rotated_height - 1);
>-    EXPECT_EQ(colors[i].y, rotated.DataY()[x + y * rotated.StrideY()]);
>-    EXPECT_EQ(colors[i].u,
>-              rotated.DataU()[(x / 2) + (y / 2) * rotated.StrideU()]);
>-    EXPECT_EQ(colors[i].v,
>-              rotated.DataV()[(x / 2) + (y / 2) * rotated.StrideV()]);
>-  }
>-}
>-
>-}  // namespace
>-
>-TEST(TestVideoFrame, WidthHeightValues) {
>-  VideoFrame frame(I420Buffer::Create(10, 10, 10, 14, 90),
>-                   webrtc::kVideoRotation_0,
>-                   789 * rtc::kNumMicrosecsPerMillisec);
>-  const int valid_value = 10;
>-  EXPECT_EQ(valid_value, frame.width());
>-  EXPECT_EQ(valid_value, frame.height());
>-  frame.set_timestamp(123u);
>-  EXPECT_EQ(123u, frame.timestamp());
>-  frame.set_ntp_time_ms(456);
>-  EXPECT_EQ(456, frame.ntp_time_ms());
>-  EXPECT_EQ(789, frame.render_time_ms());
>-}
>-
>-TEST(TestVideoFrame, ShallowCopy) {
>-  uint32_t timestamp = 1;
>-  int64_t ntp_time_ms = 2;
>-  int64_t timestamp_us = 3;
>-  int stride_y = 15;
>-  int stride_u = 10;
>-  int stride_v = 10;
>-  int width = 15;
>-  int height = 15;
>-
>-  const int kSizeY = 400;
>-  const int kSizeU = 100;
>-  const int kSizeV = 100;
>-  const VideoRotation kRotation = kVideoRotation_270;
>-  uint8_t buffer_y[kSizeY];
>-  uint8_t buffer_u[kSizeU];
>-  uint8_t buffer_v[kSizeV];
>-  memset(buffer_y, 16, kSizeY);
>-  memset(buffer_u, 8, kSizeU);
>-  memset(buffer_v, 4, kSizeV);
>-
>-  VideoFrame frame1(
>-      I420Buffer::Copy(width, height,
>-                       buffer_y, stride_y,
>-                       buffer_u, stride_u,
>-                       buffer_v, stride_v),
>-      kRotation, 0);
>-  frame1.set_timestamp(timestamp);
>-  frame1.set_ntp_time_ms(ntp_time_ms);
>-  frame1.set_timestamp_us(timestamp_us);
>-  VideoFrame frame2(frame1);
>-
>-  EXPECT_EQ(frame1.video_frame_buffer(), frame2.video_frame_buffer());
>-  rtc::scoped_refptr<I420BufferInterface> yuv1 =
>-      frame1.video_frame_buffer()->GetI420();
>-  rtc::scoped_refptr<I420BufferInterface> yuv2 =
>-      frame2.video_frame_buffer()->GetI420();
>-  EXPECT_EQ(yuv1->DataY(), yuv2->DataY());
>-  EXPECT_EQ(yuv1->DataU(), yuv2->DataU());
>-  EXPECT_EQ(yuv1->DataV(), yuv2->DataV());
>-
>-  EXPECT_EQ(frame2.timestamp(), frame1.timestamp());
>-  EXPECT_EQ(frame2.ntp_time_ms(), frame1.ntp_time_ms());
>-  EXPECT_EQ(frame2.timestamp_us(), frame1.timestamp_us());
>-  EXPECT_EQ(frame2.rotation(), frame1.rotation());
>-
>-  frame2.set_timestamp(timestamp + 1);
>-  frame2.set_ntp_time_ms(ntp_time_ms + 1);
>-  frame2.set_timestamp_us(timestamp_us + 1);
>-  frame2.set_rotation(kVideoRotation_90);
>-
>-  EXPECT_NE(frame2.timestamp(), frame1.timestamp());
>-  EXPECT_NE(frame2.ntp_time_ms(), frame1.ntp_time_ms());
>-  EXPECT_NE(frame2.timestamp_us(), frame1.timestamp_us());
>-  EXPECT_NE(frame2.rotation(), frame1.rotation());
>-}
>-
>-TEST(TestVideoFrame, TextureInitialValues) {
>-  VideoFrame frame = test::FakeNativeBuffer::CreateFrame(
>-      640, 480, 100, 10, webrtc::kVideoRotation_0);
>-  EXPECT_EQ(640, frame.width());
>-  EXPECT_EQ(480, frame.height());
>-  EXPECT_EQ(100u, frame.timestamp());
>-  EXPECT_EQ(10, frame.render_time_ms());
>-  ASSERT_TRUE(frame.video_frame_buffer() != nullptr);
>-  EXPECT_TRUE(frame.video_frame_buffer()->type() ==
>-              VideoFrameBuffer::Type::kNative);
>-
>-  frame.set_timestamp(200);
>-  EXPECT_EQ(200u, frame.timestamp());
>-  frame.set_timestamp_us(20);
>-  EXPECT_EQ(20, frame.timestamp_us());
>-}
>-
>-TEST(TestI420FrameBuffer, Copy) {
>-  rtc::scoped_refptr<I420Buffer> buf1(
>-      I420Buffer::Create(20, 10));
>-  memset(buf1->MutableDataY(), 1, 200);
>-  memset(buf1->MutableDataU(), 2, 50);
>-  memset(buf1->MutableDataV(), 3, 50);
>-  rtc::scoped_refptr<I420Buffer> buf2 = I420Buffer::Copy(*buf1);
>-  EXPECT_TRUE(test::FrameBufsEqual(buf1, buf2));
>-}
>-
>-TEST(TestI420FrameBuffer, Scale) {
>-  rtc::scoped_refptr<I420Buffer> buf = CreateGradient(200, 100);
>-
>-  // Pure scaling, no cropping.
>-  rtc::scoped_refptr<I420Buffer> scaled_buffer(
>-      I420Buffer::Create(150, 75));
>-
>-  scaled_buffer->ScaleFrom(*buf);
>-  CheckCrop(*scaled_buffer, 0.0, 0.0, 1.0, 1.0);
>-}
>-
>-TEST(TestI420FrameBuffer, CropXCenter) {
>-  rtc::scoped_refptr<I420Buffer> buf = CreateGradient(200, 100);
>-
>-  // Pure center cropping, no scaling.
>-  rtc::scoped_refptr<I420Buffer> scaled_buffer(
>-      I420Buffer::Create(100, 100));
>-
>-  scaled_buffer->CropAndScaleFrom(*buf, 50, 0, 100, 100);
>-  CheckCrop(*scaled_buffer, 0.25, 0.0, 0.5, 1.0);
>-}
>-
>-TEST(TestI420FrameBuffer, CropXNotCenter) {
>-  rtc::scoped_refptr<I420Buffer> buf = CreateGradient(200, 100);
>-
>-  // Non-center cropping, no scaling.
>-  rtc::scoped_refptr<I420Buffer> scaled_buffer(
>-      I420Buffer::Create(100, 100));
>-
>-  scaled_buffer->CropAndScaleFrom(*buf, 25, 0, 100, 100);
>-  CheckCrop(*scaled_buffer, 0.125, 0.0, 0.5, 1.0);
>-}
>-
>-TEST(TestI420FrameBuffer, CropYCenter) {
>-  rtc::scoped_refptr<I420Buffer> buf = CreateGradient(100, 200);
>-
>-  // Pure center cropping, no scaling.
>-  rtc::scoped_refptr<I420Buffer> scaled_buffer(
>-      I420Buffer::Create(100, 100));
>-
>-  scaled_buffer->CropAndScaleFrom(*buf, 0, 50, 100, 100);
>-  CheckCrop(*scaled_buffer, 0.0, 0.25, 1.0, 0.5);
>-}
>-
>-TEST(TestI420FrameBuffer, CropYNotCenter) {
>-  rtc::scoped_refptr<I420Buffer> buf = CreateGradient(100, 200);
>-
>-  // Non-center cropping, no scaling.
>-  rtc::scoped_refptr<I420Buffer> scaled_buffer(
>-      I420Buffer::Create(100, 100));
>-
>-  scaled_buffer->CropAndScaleFrom(*buf, 0, 25, 100, 100);
>-  CheckCrop(*scaled_buffer, 0.0, 0.125, 1.0, 0.5);
>-}
>-
>-TEST(TestI420FrameBuffer, CropAndScale16x9) {
>-  rtc::scoped_refptr<I420Buffer> buf = CreateGradient(640, 480);
>-
>-  // Center crop to 640 x 360 (16/9 aspect), then scale down by 2.
>-  rtc::scoped_refptr<I420Buffer> scaled_buffer(
>-      I420Buffer::Create(320, 180));
>-
>-  scaled_buffer->CropAndScaleFrom(*buf);
>-  CheckCrop(*scaled_buffer, 0.0, 0.125, 1.0, 0.75);
>-}
>-
>-class TestI420BufferRotate
>-    : public ::testing::TestWithParam<webrtc::VideoRotation> {};
>-
>-TEST_P(TestI420BufferRotate, Rotates) {
>-  rtc::scoped_refptr<I420BufferInterface> buffer = CreateGradient(640, 480);
>-  rtc::scoped_refptr<I420BufferInterface> rotated_buffer =
>-      I420Buffer::Rotate(*buffer, GetParam());
>-  CheckRotate(640, 480, GetParam(), *rotated_buffer);
>-}
>-
>-INSTANTIATE_TEST_CASE_P(Rotate, TestI420BufferRotate,
>-                        ::testing::Values(kVideoRotation_0,
>-                                          kVideoRotation_90,
>-                                          kVideoRotation_180,
>-                                          kVideoRotation_270));
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/bitrate_adjuster.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/bitrate_adjuster.h
>index fd83555469c..ee312e4f9a4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/bitrate_adjuster.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/bitrate_adjuster.h
>@@ -18,8 +18,6 @@
> 
> namespace webrtc {
> 
>-class Clock;
>-
> // Certain hardware encoders tend to consistently overshoot the bitrate that
> // they are configured to encode at. This class estimates an adjusted bitrate
> // that when set on the encoder will produce the desired bitrate.
>@@ -28,8 +26,7 @@ class BitrateAdjuster {
>   // min_adjusted_bitrate_pct and max_adjusted_bitrate_pct are the lower and
>   // upper bound outputted adjusted bitrates as a percentage of the target
>   // bitrate.
>-  BitrateAdjuster(Clock* clock,
>-                  float min_adjusted_bitrate_pct,
>+  BitrateAdjuster(float min_adjusted_bitrate_pct,
>                   float max_adjusted_bitrate_pct);
>   virtual ~BitrateAdjuster() {}
> 
>@@ -47,7 +44,7 @@ class BitrateAdjuster {
>   uint32_t GetAdjustedBitrateBps() const;
> 
>   // Returns what we think the current bitrate is.
>-  rtc::Optional<uint32_t> GetEstimatedBitrateBps();
>+  absl::optional<uint32_t> GetEstimatedBitrateBps();
> 
>   // This should be called after each frame is encoded. The timestamp at which
>   // it is called is used to estimate the output bitrate of the encoder.
>@@ -68,7 +65,6 @@ class BitrateAdjuster {
>       RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
> 
>   rtc::CriticalSection crit_;
>-  Clock* const clock_;
>   const float min_adjusted_bitrate_pct_;
>   const float max_adjusted_bitrate_pct_;
>   // The bitrate we want.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/i420_buffer_pool.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/i420_buffer_pool.h
>index 863eb101804..79f7eec23af 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/i420_buffer_pool.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/i420_buffer_pool.h
>@@ -11,8 +11,8 @@
> #ifndef COMMON_VIDEO_INCLUDE_I420_BUFFER_POOL_H_
> #define COMMON_VIDEO_INCLUDE_I420_BUFFER_POOL_H_
> 
>-#include <list>
> #include <limits>
>+#include <list>
> 
> #include "api/video/i420_buffer.h"
> #include "rtc_base/race_checker.h"
>@@ -29,11 +29,10 @@ namespace webrtc {
> // are created. This is to prevent memory leaks where frames are not returned.
> class I420BufferPool {
>  public:
>-  I420BufferPool()
>-      : I420BufferPool(false) {}
>-  explicit I420BufferPool(bool zero_initialize)
>-      : I420BufferPool(zero_initialize, std::numeric_limits<size_t>::max()) {}
>+  I420BufferPool();
>+  explicit I420BufferPool(bool zero_initialize);
>   I420BufferPool(bool zero_initialze, size_t max_number_of_buffers);
>+  ~I420BufferPool();
> 
>   // Returns a buffer from the pool. If no suitable buffer exist in the pool
>   // and there are less than |max_number_of_buffers| pending, a buffer is
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/incoming_video_stream.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/incoming_video_stream.h
>index 405416c2d81..8063061d313 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/incoming_video_stream.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/incoming_video_stream.h
>@@ -11,7 +11,7 @@
> #ifndef COMMON_VIDEO_INCLUDE_INCOMING_VIDEO_STREAM_H_
> #define COMMON_VIDEO_INCLUDE_INCOMING_VIDEO_STREAM_H_
> 
>-#include "api/videosinkinterface.h"
>+#include "api/video/video_sink_interface.h"
> #include "common_video/video_render_frames.h"
> #include "rtc_base/race_checker.h"
> #include "rtc_base/task_queue.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/video_bitrate_allocator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/video_bitrate_allocator.h
>index 597b6259a4a..77a0fef1dd0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/video_bitrate_allocator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/video_bitrate_allocator.h
>@@ -11,29 +11,6 @@
> #ifndef COMMON_VIDEO_INCLUDE_VIDEO_BITRATE_ALLOCATOR_H_
> #define COMMON_VIDEO_INCLUDE_VIDEO_BITRATE_ALLOCATOR_H_
> 
>-#include "common_types.h"  // NOLINT(build/include)
>-
>-namespace webrtc {
>-
>-class VideoBitrateAllocator {
>- public:
>-  VideoBitrateAllocator() {}
>-  virtual ~VideoBitrateAllocator() {}
>-
>-  virtual BitrateAllocation GetAllocation(uint32_t total_bitrate,
>-                                          uint32_t framerate) = 0;
>-  virtual uint32_t GetPreferredBitrateBps(uint32_t framerate) = 0;
>-};
>-
>-class VideoBitrateAllocationObserver {
>- public:
>-  VideoBitrateAllocationObserver() {}
>-  virtual ~VideoBitrateAllocationObserver() {}
>-
>-  virtual void OnBitrateAllocationUpdated(
>-      const BitrateAllocation& allocation) = 0;
>-};
>-
>-}  // namespace webrtc
>+#include "api/video/video_bitrate_allocator.h"
> 
> #endif  // COMMON_VIDEO_INCLUDE_VIDEO_BITRATE_ALLOCATOR_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/video_frame.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/video_frame.h
>index 572a4e47033..ccb709f7bc5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/video_frame.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/video_frame.h
>@@ -16,9 +16,11 @@
> // to refactor and clean up related interfaces, at which point it
> // should be moved to somewhere under api/.
> 
>-#include "api/rtp_headers.h"
>+#include "absl/types/optional.h"
>+#include "api/video/video_content_type.h"
>+#include "api/video/video_rotation.h"
>+#include "api/video/video_timing.h"
> #include "common_types.h"  // NOLINT(build/include)
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -32,10 +34,22 @@ class EncodedImage {
>   static size_t GetBufferPaddingBytes(VideoCodecType codec_type);
> 
>   EncodedImage();
>+  EncodedImage(const EncodedImage&);
>   EncodedImage(uint8_t* buffer, size_t length, size_t size);
> 
>   void SetEncodeTime(int64_t encode_start_ms, int64_t encode_finish_ms);
> 
>+  absl::optional<int> SpatialIndex() const {
>+    if (spatial_index_ < 0)
>+      return absl::nullopt;
>+    return spatial_index_;
>+  }
>+  void SetSpatialIndex(absl::optional<int> spatial_index) {
>+    RTC_DCHECK_GE(spatial_index.value_or(0), 0);
>+    RTC_DCHECK_LT(spatial_index.value_or(0), kMaxSpatialLayers);
>+    spatial_index_ = spatial_index.value_or(-1);
>+  }
>+
>   uint32_t _encodedWidth = 0;
>   uint32_t _encodedHeight = 0;
>   uint32_t _timeStamp = 0;
>@@ -57,7 +71,7 @@ class EncodedImage {
>   PlayoutDelay playout_delay_ = {-1, -1};
> 
>   struct Timing {
>-    uint8_t flags = TimingFrameFlags::kInvalid;
>+    uint8_t flags = VideoSendTiming::kInvalid;
>     int64_t encode_start_ms = 0;
>     int64_t encode_finish_ms = 0;
>     int64_t packetization_finish_ms = 0;
>@@ -67,6 +81,11 @@ class EncodedImage {
>     int64_t receive_start_ms = 0;
>     int64_t receive_finish_ms = 0;
>   } timing_;
>+
>+ private:
>+  // -1 means not set. Use a plain int rather than optional, to keep this class
>+  // copyable with memcpy.
>+  int spatial_index_ = -1;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/video_frame_buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/video_frame_buffer.h
>index 312979b05d8..11bb8125316 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/video_frame_buffer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/video_frame_buffer.h
>@@ -103,6 +103,17 @@ rtc::scoped_refptr<PlanarYuvBuffer> WrapYuvBuffer(
>     int v_stride,
>     const rtc::Callback0<void>& no_longer_used);
> 
>+rtc::scoped_refptr<I010BufferInterface> WrapI010Buffer(
>+    int width,
>+    int height,
>+    const uint16_t* y_plane,
>+    int y_stride,
>+    const uint16_t* u_plane,
>+    int u_stride,
>+    const uint16_t* v_plane,
>+    int v_stride,
>+    const rtc::Callback0<void>& no_longer_used);
>+
> }  // namespace webrtc
> 
> #endif  // COMMON_VIDEO_INCLUDE_VIDEO_FRAME_BUFFER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/incoming_video_stream.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/incoming_video_stream.cc
>index 8e715407ea6..efca5145566 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/incoming_video_stream.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/incoming_video_stream.cc
>@@ -15,7 +15,6 @@
> #include "common_video/video_render_frames.h"
> #include "rtc_base/timeutils.h"
> #include "rtc_base/trace_event.h"
>-#include "system_wrappers/include/event_wrapper.h"
> 
> namespace webrtc {
> namespace {
>@@ -70,7 +69,7 @@ void IncomingVideoStream::OnFrame(const VideoFrame& video_frame) {
> void IncomingVideoStream::Dequeue() {
>   TRACE_EVENT0("webrtc", "IncomingVideoStream::Dequeue");
>   RTC_DCHECK(incoming_render_queue_.IsCurrent());
>-  rtc::Optional<VideoFrame> frame_to_render = render_buffers_.FrameToRender();
>+  absl::optional<VideoFrame> frame_to_render = render_buffers_.FrameToRender();
>   if (frame_to_render)
>     callback_->OnFrame(*frame_to_render);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/libyuv/include/webrtc_libyuv.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/libyuv/include/webrtc_libyuv.h
>index 5b2a3af8359..4cbfec20187 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/libyuv/include/webrtc_libyuv.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/libyuv/include/webrtc_libyuv.h
>@@ -20,11 +20,9 @@
> 
> #include "api/video/video_frame.h"
> #include "common_types.h"  // NOLINT(build/include)  // VideoTypes.
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>-
> // This is the max PSNR value our algorithms can return.
> const double kPerfectPSNR = 48.0f;
> 
>@@ -93,12 +91,18 @@ double I420SSIM(const I420BufferInterface& ref_buffer,
> // |tmp_buffer| should be:
> //   (src_width/2) * (src_height/2) * 2 + (dst_width/2) * (dst_height/2) * 2
> void NV12Scale(uint8_t* tmp_buffer,
>-               const uint8_t* src_y, int src_stride_y,
>-               const uint8_t* src_uv, int src_stride_uv,
>-               int src_width, int src_height,
>-               uint8_t* dst_y, int dst_stride_y,
>-               uint8_t* dst_uv, int dst_stride_uv,
>-               int dst_width, int dst_height);
>+               const uint8_t* src_y,
>+               int src_stride_y,
>+               const uint8_t* src_uv,
>+               int src_stride_uv,
>+               int src_width,
>+               int src_height,
>+               uint8_t* dst_y,
>+               int dst_stride_y,
>+               uint8_t* dst_uv,
>+               int dst_stride_uv,
>+               int dst_width,
>+               int dst_height);
> 
> // Helper class for directly converting and scaling NV12 to I420. The Y-plane
> // will be scaled directly to the I420 destination, which makes this faster
>@@ -107,13 +111,21 @@ class NV12ToI420Scaler {
>  public:
>   NV12ToI420Scaler();
>   ~NV12ToI420Scaler();
>-  void NV12ToI420Scale(const uint8_t* src_y, int src_stride_y,
>-                       const uint8_t* src_uv, int src_stride_uv,
>-                       int src_width, int src_height,
>-                       uint8_t* dst_y, int dst_stride_y,
>-                       uint8_t* dst_u, int dst_stride_u,
>-                       uint8_t* dst_v, int dst_stride_v,
>-                       int dst_width, int dst_height);
>+  void NV12ToI420Scale(const uint8_t* src_y,
>+                       int src_stride_y,
>+                       const uint8_t* src_uv,
>+                       int src_stride_uv,
>+                       int src_width,
>+                       int src_height,
>+                       uint8_t* dst_y,
>+                       int dst_stride_y,
>+                       uint8_t* dst_u,
>+                       int dst_stride_u,
>+                       uint8_t* dst_v,
>+                       int dst_stride_v,
>+                       int dst_width,
>+                       int dst_height);
>+
>  private:
>   std::vector<uint8_t> tmp_uv_planes_;
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/libyuv/libyuv_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/libyuv/libyuv_unittest.cc
>index ba65544bf01..79e7cb61b98 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/libyuv/libyuv_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/libyuv/libyuv_unittest.cc
>@@ -35,8 +35,8 @@ void Calc16ByteAlignedStride(int width, int* stride_y, int* stride_uv) {
> class TestLibYuv : public ::testing::Test {
>  protected:
>   TestLibYuv();
>-  virtual void SetUp();
>-  virtual void TearDown();
>+  void SetUp() override;
>+  void TearDown() override;
> 
>   FILE* source_file_;
>   std::unique_ptr<VideoFrame> orig_frame_;
>@@ -57,11 +57,11 @@ TestLibYuv::TestLibYuv()
>       frame_length_(CalcBufferSize(VideoType::kI420, 352, 288)) {}
> 
> void TestLibYuv::SetUp() {
>-  const std::string input_file_name = webrtc::test::ResourcePath("foreman_cif",
>-                                                                 "yuv");
>-  source_file_  = fopen(input_file_name.c_str(), "rb");
>-  ASSERT_TRUE(source_file_ != NULL) << "Cannot read file: "<<
>-                                       input_file_name << "\n";
>+  const std::string input_file_name =
>+      webrtc::test::ResourcePath("foreman_cif", "yuv");
>+  source_file_ = fopen(input_file_name.c_str(), "rb");
>+  ASSERT_TRUE(source_file_ != NULL)
>+      << "Cannot read file: " << input_file_name << "\n";
> 
>   rtc::scoped_refptr<I420BufferInterface> buffer(
>       test::ReadI420Buffer(width_, height_, source_file_));
>@@ -83,9 +83,9 @@ TEST_F(TestLibYuv, ConvertSanityTest) {
> TEST_F(TestLibYuv, ConvertTest) {
>   // Reading YUV frame - testing on the first frame of the foreman sequence
>   int j = 0;
>-  std::string output_file_name = webrtc::test::OutputPath() +
>-                                 "LibYuvTest_conversion.yuv";
>-  FILE*  output_file = fopen(output_file_name.c_str(), "wb");
>+  std::string output_file_name =
>+      webrtc::test::OutputPath() + "LibYuvTest_conversion.yuv";
>+  FILE* output_file = fopen(output_file_name.c_str(), "wb");
>   ASSERT_TRUE(output_file != NULL);
> 
>   double psnr = 0.0;
>@@ -249,9 +249,9 @@ TEST_F(TestLibYuv, ConvertTest) {
> 
> TEST_F(TestLibYuv, ConvertAlignedFrame) {
>   // Reading YUV frame - testing on the first frame of the foreman sequence
>-  std::string output_file_name = webrtc::test::OutputPath() +
>-                                 "LibYuvTest_conversion.yuv";
>-  FILE*  output_file = fopen(output_file_name.c_str(), "wb");
>+  std::string output_file_name =
>+      webrtc::test::OutputPath() + "LibYuvTest_conversion.yuv";
>+  FILE* output_file = fopen(output_file_name.c_str(), "wb");
>   ASSERT_TRUE(output_file != NULL);
> 
>   double psnr = 0.0;
>@@ -290,33 +290,24 @@ static uint8_t Average(int a, int b, int c, int d) {
> }
> 
> TEST_F(TestLibYuv, NV12Scale2x2to2x2) {
>-  const std::vector<uint8_t> src_y = {0, 1,
>-                                      2, 3};
>+  const std::vector<uint8_t> src_y = {0, 1, 2, 3};
>   const std::vector<uint8_t> src_uv = {0, 1};
>   std::vector<uint8_t> dst_y(4);
>   std::vector<uint8_t> dst_uv(2);
> 
>   uint8_t* tmp_buffer = nullptr;
> 
>-  NV12Scale(tmp_buffer,
>-            src_y.data(), 2,
>-            src_uv.data(), 2,
>-            2, 2,
>-            dst_y.data(), 2,
>-            dst_uv.data(), 2,
>-            2, 2);
>+  NV12Scale(tmp_buffer, src_y.data(), 2, src_uv.data(), 2, 2, 2, dst_y.data(),
>+            2, dst_uv.data(), 2, 2, 2);
> 
>   EXPECT_THAT(dst_y, ::testing::ContainerEq(src_y));
>   EXPECT_THAT(dst_uv, ::testing::ContainerEq(src_uv));
> }
> 
> TEST_F(TestLibYuv, NV12Scale4x4to2x2) {
>-  const uint8_t src_y[] = { 0,  1,  2,  3,
>-                            4,  5,  6,  7,
>-                            8,  9, 10, 11,
>-                           12, 13, 14, 15};
>-  const uint8_t src_uv[] = {0, 1, 2, 3,
>-                            4, 5, 6, 7};
>+  const uint8_t src_y[] = {0, 1, 2,  3,  4,  5,  6,  7,
>+                           8, 9, 10, 11, 12, 13, 14, 15};
>+  const uint8_t src_uv[] = {0, 1, 2, 3, 4, 5, 6, 7};
>   std::vector<uint8_t> dst_y(4);
>   std::vector<uint8_t> dst_uv(2);
> 
>@@ -329,13 +320,8 @@ TEST_F(TestLibYuv, NV12Scale4x4to2x2) {
>                     dst_chroma_width * dst_chroma_height * 2);
>   tmp_buffer.shrink_to_fit();
> 
>-  NV12Scale(tmp_buffer.data(),
>-            src_y, 4,
>-            src_uv, 4,
>-            4, 4,
>-            dst_y.data(), 2,
>-            dst_uv.data(), 2,
>-            2, 2);
>+  NV12Scale(tmp_buffer.data(), src_y, 4, src_uv, 4, 4, 4, dst_y.data(), 2,
>+            dst_uv.data(), 2, 2, 2);
> 
>   EXPECT_THAT(dst_y, ::testing::ElementsAre(
>                          Average(0, 1, 4, 5), Average(2, 3, 6, 7),
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/libyuv/webrtc_libyuv.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/libyuv/webrtc_libyuv.cc
>index 676dad3d0c3..6bdfabd5e9c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/libyuv/webrtc_libyuv.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/libyuv/webrtc_libyuv.cc
>@@ -13,6 +13,8 @@
> #include <string.h>
> 
> #include "api/video/i420_buffer.h"
>+#include "common_video/include/video_frame_buffer.h"
>+#include "rtc_base/bind.h"
> #include "rtc_base/checks.h"
> #include "third_party/libyuv/include/libyuv.h"
> 
>@@ -73,18 +75,15 @@ int PrintVideoFrame(const I420BufferInterface& frame, FILE* file) {
>   int chroma_width = frame.ChromaWidth();
>   int chroma_height = frame.ChromaHeight();
> 
>-  if (PrintPlane(frame.DataY(), width, height,
>-                 frame.StrideY(), file) < 0) {
>+  if (PrintPlane(frame.DataY(), width, height, frame.StrideY(), file) < 0) {
>     return -1;
>   }
>-  if (PrintPlane(frame.DataU(),
>-                 chroma_width, chroma_height,
>-                 frame.StrideU(), file) < 0) {
>+  if (PrintPlane(frame.DataU(), chroma_width, chroma_height, frame.StrideU(),
>+                 file) < 0) {
>     return -1;
>   }
>-  if (PrintPlane(frame.DataV(),
>-                 chroma_width, chroma_height,
>-                 frame.StrideV(), file) < 0) {
>+  if (PrintPlane(frame.DataV(), chroma_width, chroma_height, frame.StrideV(),
>+                 file) < 0) {
>     return -1;
>   }
>   return 0;
>@@ -104,23 +103,18 @@ int ExtractBuffer(const rtc::scoped_refptr<I420BufferInterface>& input_frame,
>   int height = input_frame->height();
>   size_t length = CalcBufferSize(VideoType::kI420, width, height);
>   if (size < length) {
>-     return -1;
>+    return -1;
>   }
> 
>   int chroma_width = input_frame->ChromaWidth();
>   int chroma_height = input_frame->ChromaHeight();
> 
>-  libyuv::I420Copy(input_frame->DataY(),
>-                   input_frame->StrideY(),
>-                   input_frame->DataU(),
>-                   input_frame->StrideU(),
>-                   input_frame->DataV(),
>-                   input_frame->StrideV(),
>-                   buffer, width,
>-                   buffer + width*height, chroma_width,
>-                   buffer + width*height + chroma_width*chroma_height,
>-                   chroma_width,
>-                   width, height);
>+  libyuv::I420Copy(input_frame->DataY(), input_frame->StrideY(),
>+                   input_frame->DataU(), input_frame->StrideU(),
>+                   input_frame->DataV(), input_frame->StrideV(), buffer, width,
>+                   buffer + width * height, chroma_width,
>+                   buffer + width * height + chroma_width * chroma_height,
>+                   chroma_width, width, height);
> 
>   return static_cast<int>(length);
> }
>@@ -132,24 +126,25 @@ int ExtractBuffer(const VideoFrame& input_frame, size_t size, uint8_t* buffer) {
> 
> int ConvertNV12ToRGB565(const uint8_t* src_frame,
>                         uint8_t* dst_frame,
>-                        int width, int height) {
>+                        int width,
>+                        int height) {
>   int abs_height = (height < 0) ? -height : height;
>   const uint8_t* yplane = src_frame;
>   const uint8_t* uvInterlaced = src_frame + (width * abs_height);
> 
>-  return libyuv::NV12ToRGB565(yplane, width,
>-                              uvInterlaced, (width + 1) >> 1,
>-                              dst_frame, width,
>-                              width, height);
>+  return libyuv::NV12ToRGB565(yplane, width, uvInterlaced, (width + 1) >> 1,
>+                              dst_frame, width, width, height);
> }
> 
>-int ConvertRGB24ToARGB(const uint8_t* src_frame, uint8_t* dst_frame,
>-                       int width, int height, int dst_stride) {
>+int ConvertRGB24ToARGB(const uint8_t* src_frame,
>+                       uint8_t* dst_frame,
>+                       int width,
>+                       int height,
>+                       int dst_stride) {
>   if (dst_stride == 0)
>     dst_stride = width;
>-  return libyuv::RGB24ToARGB(src_frame, width,
>-                             dst_frame, dst_stride,
>-                             width, height);
>+  return libyuv::RGB24ToARGB(src_frame, width, dst_frame, dst_stride, width,
>+                             height);
> }
> 
> int ConvertVideoType(VideoType video_type) {
>@@ -203,6 +198,78 @@ int ConvertFromI420(const VideoFrame& src_frame,
>       ConvertVideoType(dst_video_type));
> }
> 
>+// Helper functions for keeping references alive.
>+void KeepBufferRefs(rtc::scoped_refptr<webrtc::VideoFrameBuffer>,
>+                    rtc::scoped_refptr<webrtc::VideoFrameBuffer>) {}
>+
>+rtc::scoped_refptr<I420ABufferInterface> ScaleI420ABuffer(
>+    const I420ABufferInterface& buffer,
>+    int target_width,
>+    int target_height) {
>+  rtc::scoped_refptr<I420Buffer> yuv_buffer =
>+      I420Buffer::Create(target_width, target_height);
>+  yuv_buffer->ScaleFrom(buffer);
>+  rtc::scoped_refptr<I420Buffer> axx_buffer =
>+      I420Buffer::Create(target_width, target_height);
>+  libyuv::ScalePlane(buffer.DataA(), buffer.StrideA(), buffer.width(),
>+                     buffer.height(), axx_buffer->MutableDataY(),
>+                     axx_buffer->StrideY(), target_width, target_height,
>+                     libyuv::kFilterBox);
>+  rtc::scoped_refptr<I420ABufferInterface> merged_buffer = WrapI420ABuffer(
>+      yuv_buffer->width(), yuv_buffer->height(), yuv_buffer->DataY(),
>+      yuv_buffer->StrideY(), yuv_buffer->DataU(), yuv_buffer->StrideU(),
>+      yuv_buffer->DataV(), yuv_buffer->StrideV(), axx_buffer->DataY(),
>+      axx_buffer->StrideY(),
>+      rtc::Bind(&KeepBufferRefs, yuv_buffer, axx_buffer));
>+  return merged_buffer;
>+}
>+
>+// Compute PSNR for an I420A frame (all planes). Can upscale test frame.
>+double I420APSNR(const I420ABufferInterface& ref_buffer,
>+                 const I420ABufferInterface& test_buffer) {
>+  RTC_DCHECK_GE(ref_buffer.width(), test_buffer.width());
>+  RTC_DCHECK_GE(ref_buffer.height(), test_buffer.height());
>+  if ((ref_buffer.width() != test_buffer.width()) ||
>+      (ref_buffer.height() != test_buffer.height())) {
>+    rtc::scoped_refptr<I420ABufferInterface> scaled_buffer =
>+        ScaleI420ABuffer(test_buffer, ref_buffer.width(), ref_buffer.height());
>+    return I420APSNR(ref_buffer, *scaled_buffer);
>+  }
>+  const int width = test_buffer.width();
>+  const int height = test_buffer.height();
>+  const uint64_t sse_y = libyuv::ComputeSumSquareErrorPlane(
>+      ref_buffer.DataY(), ref_buffer.StrideY(), test_buffer.DataY(),
>+      test_buffer.StrideY(), width, height);
>+  const int width_uv = (width + 1) >> 1;
>+  const int height_uv = (height + 1) >> 1;
>+  const uint64_t sse_u = libyuv::ComputeSumSquareErrorPlane(
>+      ref_buffer.DataU(), ref_buffer.StrideU(), test_buffer.DataU(),
>+      test_buffer.StrideU(), width_uv, height_uv);
>+  const uint64_t sse_v = libyuv::ComputeSumSquareErrorPlane(
>+      ref_buffer.DataV(), ref_buffer.StrideV(), test_buffer.DataV(),
>+      test_buffer.StrideV(), width_uv, height_uv);
>+  const uint64_t sse_a = libyuv::ComputeSumSquareErrorPlane(
>+      ref_buffer.DataA(), ref_buffer.StrideA(), test_buffer.DataA(),
>+      test_buffer.StrideA(), width, height);
>+  const uint64_t samples = 2 * (uint64_t)width * (uint64_t)height +
>+                           2 * ((uint64_t)width_uv * (uint64_t)height_uv);
>+  const uint64_t sse = sse_y + sse_u + sse_v + sse_a;
>+  const double psnr = libyuv::SumSquareErrorToPsnr(sse, samples);
>+  return (psnr > kPerfectPSNR) ? kPerfectPSNR : psnr;
>+}
>+
>+// Compute PSNR for an I420A frame (all planes)
>+double I420APSNR(const VideoFrame* ref_frame, const VideoFrame* test_frame) {
>+  if (!ref_frame || !test_frame)
>+    return -1;
>+  RTC_DCHECK(ref_frame->video_frame_buffer()->type() ==
>+             VideoFrameBuffer::Type::kI420A);
>+  RTC_DCHECK(test_frame->video_frame_buffer()->type() ==
>+             VideoFrameBuffer::Type::kI420A);
>+  return I420APSNR(*ref_frame->video_frame_buffer()->GetI420A(),
>+                   *test_frame->video_frame_buffer()->GetI420A());
>+}
>+
> // Compute PSNR for an I420 frame (all planes). Can upscale test frame.
> double I420PSNR(const I420BufferInterface& ref_buffer,
>                 const I420BufferInterface& test_buffer) {
>@@ -234,6 +301,41 @@ double I420PSNR(const VideoFrame* ref_frame, const VideoFrame* test_frame) {
>                   *test_frame->video_frame_buffer()->ToI420());
> }
> 
>+// Compute SSIM for an I420A frame (all planes). Can upscale test frame.
>+double I420ASSIM(const I420ABufferInterface& ref_buffer,
>+                 const I420ABufferInterface& test_buffer) {
>+  RTC_DCHECK_GE(ref_buffer.width(), test_buffer.width());
>+  RTC_DCHECK_GE(ref_buffer.height(), test_buffer.height());
>+  if ((ref_buffer.width() != test_buffer.width()) ||
>+      (ref_buffer.height() != test_buffer.height())) {
>+    rtc::scoped_refptr<I420ABufferInterface> scaled_buffer =
>+        ScaleI420ABuffer(test_buffer, ref_buffer.width(), ref_buffer.height());
>+    return I420ASSIM(ref_buffer, *scaled_buffer);
>+  }
>+  const double yuv_ssim = libyuv::I420Ssim(
>+      ref_buffer.DataY(), ref_buffer.StrideY(), ref_buffer.DataU(),
>+      ref_buffer.StrideU(), ref_buffer.DataV(), ref_buffer.StrideV(),
>+      test_buffer.DataY(), test_buffer.StrideY(), test_buffer.DataU(),
>+      test_buffer.StrideU(), test_buffer.DataV(), test_buffer.StrideV(),
>+      test_buffer.width(), test_buffer.height());
>+  const double a_ssim = libyuv::CalcFrameSsim(
>+      ref_buffer.DataA(), ref_buffer.StrideA(), test_buffer.DataA(),
>+      test_buffer.StrideA(), test_buffer.width(), test_buffer.height());
>+  return (yuv_ssim + (a_ssim * 0.8)) / 1.8;
>+}
>+
>+// Compute SSIM for an I420A frame (all planes)
>+double I420ASSIM(const VideoFrame* ref_frame, const VideoFrame* test_frame) {
>+  if (!ref_frame || !test_frame)
>+    return -1;
>+  RTC_DCHECK(ref_frame->video_frame_buffer()->type() ==
>+             VideoFrameBuffer::Type::kI420A);
>+  RTC_DCHECK(test_frame->video_frame_buffer()->type() ==
>+             VideoFrameBuffer::Type::kI420A);
>+  return I420ASSIM(*ref_frame->video_frame_buffer()->GetI420A(),
>+                   *test_frame->video_frame_buffer()->GetI420A());
>+}
>+
> // Compute SSIM for an I420 frame (all planes). Can upscale test_buffer.
> double I420SSIM(const I420BufferInterface& ref_buffer,
>                 const I420BufferInterface& test_buffer) {
>@@ -253,6 +355,7 @@ double I420SSIM(const I420BufferInterface& ref_buffer,
>       test_buffer.StrideU(), test_buffer.DataV(), test_buffer.StrideV(),
>       test_buffer.width(), test_buffer.height());
> }
>+
> double I420SSIM(const VideoFrame* ref_frame, const VideoFrame* test_frame) {
>   if (!ref_frame || !test_frame)
>     return -1;
>@@ -261,12 +364,18 @@ double I420SSIM(const VideoFrame* ref_frame, const VideoFrame* test_frame) {
> }
> 
> void NV12Scale(uint8_t* tmp_buffer,
>-               const uint8_t* src_y, int src_stride_y,
>-               const uint8_t* src_uv, int src_stride_uv,
>-               int src_width, int src_height,
>-               uint8_t* dst_y, int dst_stride_y,
>-               uint8_t* dst_uv, int dst_stride_uv,
>-               int dst_width, int dst_height) {
>+               const uint8_t* src_y,
>+               int src_stride_y,
>+               const uint8_t* src_uv,
>+               int src_stride_uv,
>+               int src_width,
>+               int src_height,
>+               uint8_t* dst_y,
>+               int dst_stride_y,
>+               uint8_t* dst_uv,
>+               int dst_stride_uv,
>+               int dst_width,
>+               int dst_height) {
>   const int src_chroma_width = (src_width + 1) / 2;
>   const int src_chroma_height = (src_height + 1) / 2;
> 
>@@ -290,51 +399,44 @@ void NV12Scale(uint8_t* tmp_buffer,
>   uint8_t* const dst_v = dst_u + dst_chroma_width * dst_chroma_height;
> 
>   // Split source UV plane into separate U and V plane using the temporary data.
>-  libyuv::SplitUVPlane(src_uv, src_stride_uv,
>-                       src_u, src_chroma_width,
>-                       src_v, src_chroma_width,
>-                       src_chroma_width, src_chroma_height);
>+  libyuv::SplitUVPlane(src_uv, src_stride_uv, src_u, src_chroma_width, src_v,
>+                       src_chroma_width, src_chroma_width, src_chroma_height);
> 
>   // Scale the planes.
>-  libyuv::I420Scale(src_y, src_stride_y,
>-                    src_u, src_chroma_width,
>-                    src_v, src_chroma_width,
>-                    src_width, src_height,
>-                    dst_y, dst_stride_y,
>-                    dst_u, dst_chroma_width,
>-                    dst_v, dst_chroma_width,
>-                    dst_width, dst_height,
>-                    libyuv::kFilterBox);
>+  libyuv::I420Scale(
>+      src_y, src_stride_y, src_u, src_chroma_width, src_v, src_chroma_width,
>+      src_width, src_height, dst_y, dst_stride_y, dst_u, dst_chroma_width,
>+      dst_v, dst_chroma_width, dst_width, dst_height, libyuv::kFilterBox);
> 
>   // Merge the UV planes into the destination.
>-  libyuv::MergeUVPlane(dst_u, dst_chroma_width,
>-                       dst_v, dst_chroma_width,
>-                       dst_uv, dst_stride_uv,
>-                       dst_chroma_width, dst_chroma_height);
>+  libyuv::MergeUVPlane(dst_u, dst_chroma_width, dst_v, dst_chroma_width, dst_uv,
>+                       dst_stride_uv, dst_chroma_width, dst_chroma_height);
> }
> 
> NV12ToI420Scaler::NV12ToI420Scaler() = default;
> NV12ToI420Scaler::~NV12ToI420Scaler() = default;
> 
>-void NV12ToI420Scaler::NV12ToI420Scale(
>-    const uint8_t* src_y, int src_stride_y,
>-    const uint8_t* src_uv, int src_stride_uv,
>-    int src_width, int src_height,
>-    uint8_t* dst_y, int dst_stride_y,
>-    uint8_t* dst_u, int dst_stride_u,
>-    uint8_t* dst_v, int dst_stride_v,
>-    int dst_width, int dst_height) {
>+void NV12ToI420Scaler::NV12ToI420Scale(const uint8_t* src_y,
>+                                       int src_stride_y,
>+                                       const uint8_t* src_uv,
>+                                       int src_stride_uv,
>+                                       int src_width,
>+                                       int src_height,
>+                                       uint8_t* dst_y,
>+                                       int dst_stride_y,
>+                                       uint8_t* dst_u,
>+                                       int dst_stride_u,
>+                                       uint8_t* dst_v,
>+                                       int dst_stride_v,
>+                                       int dst_width,
>+                                       int dst_height) {
>   if (src_width == dst_width && src_height == dst_height) {
>     // No scaling.
>     tmp_uv_planes_.clear();
>     tmp_uv_planes_.shrink_to_fit();
>-    libyuv::NV12ToI420(
>-        src_y, src_stride_y,
>-        src_uv, src_stride_uv,
>-        dst_y, dst_stride_y,
>-        dst_u, dst_stride_u,
>-        dst_v, dst_stride_v,
>-        src_width, src_height);
>+    libyuv::NV12ToI420(src_y, src_stride_y, src_uv, src_stride_uv, dst_y,
>+                       dst_stride_y, dst_u, dst_stride_u, dst_v, dst_stride_v,
>+                       src_width, src_height);
>     return;
>   }
> 
>@@ -348,21 +450,14 @@ void NV12ToI420Scaler::NV12ToI420Scale(
>   // Split source UV plane into separate U and V plane using the temporary data.
>   uint8_t* const src_u = tmp_uv_planes_.data();
>   uint8_t* const src_v = tmp_uv_planes_.data() + src_uv_width * src_uv_height;
>-  libyuv::SplitUVPlane(src_uv, src_stride_uv,
>-                       src_u, src_uv_width,
>-                       src_v, src_uv_width,
>-                       src_uv_width, src_uv_height);
>+  libyuv::SplitUVPlane(src_uv, src_stride_uv, src_u, src_uv_width, src_v,
>+                       src_uv_width, src_uv_width, src_uv_height);
> 
>   // Scale the planes into the destination.
>-  libyuv::I420Scale(src_y, src_stride_y,
>-                    src_u, src_uv_width,
>-                    src_v, src_uv_width,
>-                    src_width, src_height,
>-                    dst_y, dst_stride_y,
>-                    dst_u, dst_stride_u,
>-                    dst_v, dst_stride_v,
>-                    dst_width, dst_height,
>-                    libyuv::kFilterBox);
>+  libyuv::I420Scale(src_y, src_stride_y, src_u, src_uv_width, src_v,
>+                    src_uv_width, src_width, src_height, dst_y, dst_stride_y,
>+                    dst_u, dst_stride_u, dst_v, dst_stride_v, dst_width,
>+                    dst_height, libyuv::kFilterBox);
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/video_frame.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/video_frame.cc
>index 58342a91831..f4618146711 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/video_frame.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/video_frame.cc
>@@ -25,28 +25,19 @@ const size_t EncodedImage::kBufferPaddingBytesH264 = 8;
> 
> size_t EncodedImage::GetBufferPaddingBytes(VideoCodecType codec_type) {
>   switch (codec_type) {
>-    case kVideoCodecVP8:
>-    case kVideoCodecVP9:
>-      return 0;
>     case kVideoCodecH264:
>       return kBufferPaddingBytesH264;
>-    case kVideoCodecI420:
>-    case kVideoCodecRED:
>-    case kVideoCodecULPFEC:
>-    case kVideoCodecFlexfec:
>-    case kVideoCodecGeneric:
>-    case kVideoCodecStereo:
>-    case kVideoCodecUnknown:
>+    default:
>       return 0;
>   }
>-  RTC_NOTREACHED();
>-  return 0;
> }
> 
> EncodedImage::EncodedImage() : EncodedImage(nullptr, 0, 0) {}
> 
>+EncodedImage::EncodedImage(const EncodedImage&) = default;
>+
> EncodedImage::EncodedImage(uint8_t* buffer, size_t length, size_t size)
>-      : _buffer(buffer), _length(length), _size(size) {}
>+    : _buffer(buffer), _length(length), _size(size) {}
> 
> void EncodedImage::SetEncodeTime(int64_t encode_start_ms,
>                                  int64_t encode_finish_ms) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/video_frame_buffer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/video_frame_buffer.cc
>index eebae2ad94b..be0b4927bbe 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/video_frame_buffer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/video_frame_buffer.cc
>@@ -130,6 +130,77 @@ rtc::scoped_refptr<I420BufferInterface> I444BufferBase::ToI420() {
>   return i420_buffer;
> }
> 
>+// Template to implement a wrapped buffer for a PlanarYuv16BBuffer.
>+template <typename Base>
>+class WrappedYuv16BBuffer : public Base {
>+ public:
>+  WrappedYuv16BBuffer(int width,
>+                      int height,
>+                      const uint16_t* y_plane,
>+                      int y_stride,
>+                      const uint16_t* u_plane,
>+                      int u_stride,
>+                      const uint16_t* v_plane,
>+                      int v_stride,
>+                      const rtc::Callback0<void>& no_longer_used)
>+      : width_(width),
>+        height_(height),
>+        y_plane_(y_plane),
>+        u_plane_(u_plane),
>+        v_plane_(v_plane),
>+        y_stride_(y_stride),
>+        u_stride_(u_stride),
>+        v_stride_(v_stride),
>+        no_longer_used_cb_(no_longer_used) {}
>+
>+  ~WrappedYuv16BBuffer() override { no_longer_used_cb_(); }
>+
>+  int width() const override { return width_; }
>+
>+  int height() const override { return height_; }
>+
>+  const uint16_t* DataY() const override { return y_plane_; }
>+
>+  const uint16_t* DataU() const override { return u_plane_; }
>+
>+  const uint16_t* DataV() const override { return v_plane_; }
>+
>+  int StrideY() const override { return y_stride_; }
>+
>+  int StrideU() const override { return u_stride_; }
>+
>+  int StrideV() const override { return v_stride_; }
>+
>+ private:
>+  friend class rtc::RefCountedObject<WrappedYuv16BBuffer>;
>+
>+  const int width_;
>+  const int height_;
>+  const uint16_t* const y_plane_;
>+  const uint16_t* const u_plane_;
>+  const uint16_t* const v_plane_;
>+  const int y_stride_;
>+  const int u_stride_;
>+  const int v_stride_;
>+  rtc::Callback0<void> no_longer_used_cb_;
>+};
>+
>+class I010BufferBase : public I010BufferInterface {
>+ public:
>+  rtc::scoped_refptr<I420BufferInterface> ToI420() final;
>+};
>+
>+rtc::scoped_refptr<I420BufferInterface> I010BufferBase::ToI420() {
>+  rtc::scoped_refptr<I420Buffer> i420_buffer =
>+      I420Buffer::Create(width(), height());
>+  libyuv::I010ToI420(DataY(), StrideY(), DataU(), StrideU(), DataV(), StrideV(),
>+                     i420_buffer->MutableDataY(), i420_buffer->StrideY(),
>+                     i420_buffer->MutableDataU(), i420_buffer->StrideU(),
>+                     i420_buffer->MutableDataV(), i420_buffer->StrideV(),
>+                     width(), height());
>+  return i420_buffer;
>+}
>+
> }  // namespace
> 
> WrappedI420Buffer::WrappedI420Buffer(int width,
>@@ -149,8 +220,7 @@ WrappedI420Buffer::WrappedI420Buffer(int width,
>       y_stride_(y_stride),
>       u_stride_(u_stride),
>       v_stride_(v_stride),
>-      no_longer_used_cb_(no_longer_used) {
>-}
>+      no_longer_used_cb_(no_longer_used) {}
> 
> WrappedI420Buffer::~WrappedI420Buffer() {
>   no_longer_used_cb_();
>@@ -258,4 +328,20 @@ rtc::scoped_refptr<PlanarYuvBuffer> WrapYuvBuffer(
>   }
> }
> 
>+rtc::scoped_refptr<I010BufferInterface> WrapI010Buffer(
>+    int width,
>+    int height,
>+    const uint16_t* y_plane,
>+    int y_stride,
>+    const uint16_t* u_plane,
>+    int u_stride,
>+    const uint16_t* v_plane,
>+    int v_stride,
>+    const rtc::Callback0<void>& no_longer_used) {
>+  return rtc::scoped_refptr<I010BufferInterface>(
>+      new rtc::RefCountedObject<WrappedYuv16BBuffer<I010BufferBase>>(
>+          width, height, y_plane, y_stride, u_plane, u_stride, v_plane,
>+          v_stride, no_longer_used));
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/video_frame_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/video_frame_unittest.cc
>new file mode 100644
>index 00000000000..a4b110b6b93
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/video_frame_unittest.cc
>@@ -0,0 +1,435 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <math.h>
>+#include <string.h>
>+
>+#include "api/video/i010_buffer.h"
>+#include "api/video/i420_buffer.h"
>+#include "api/video/video_frame.h"
>+#include "rtc_base/bind.h"
>+#include "rtc_base/timeutils.h"
>+#include "test/fake_texture_frame.h"
>+#include "test/frame_utils.h"
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+
>+namespace {
>+
>+// Helper class to delegate calls to appropriate container.
>+class PlanarYuvBufferFactory {
>+ public:
>+  static rtc::scoped_refptr<PlanarYuvBuffer> Create(VideoFrameBuffer::Type type,
>+                                                    int width,
>+                                                    int height) {
>+    switch (type) {
>+      case VideoFrameBuffer::Type::kI420:
>+        return I420Buffer::Create(width, height);
>+      case VideoFrameBuffer::Type::kI010:
>+        return I010Buffer::Create(width, height);
>+      default:
>+        RTC_NOTREACHED();
>+    }
>+    return nullptr;
>+  }
>+
>+  static rtc::scoped_refptr<PlanarYuvBuffer> Copy(const VideoFrameBuffer& src) {
>+    switch (src.type()) {
>+      case VideoFrameBuffer::Type::kI420:
>+        return I420Buffer::Copy(src);
>+      case VideoFrameBuffer::Type::kI010:
>+        return I010Buffer::Copy(*src.GetI010());
>+      default:
>+        RTC_NOTREACHED();
>+    }
>+    return nullptr;
>+  }
>+
>+  static rtc::scoped_refptr<PlanarYuvBuffer> Rotate(const VideoFrameBuffer& src,
>+                                                    VideoRotation rotation) {
>+    switch (src.type()) {
>+      case VideoFrameBuffer::Type::kI420:
>+        return I420Buffer::Rotate(src, rotation);
>+      case VideoFrameBuffer::Type::kI010:
>+        return I010Buffer::Rotate(*src.GetI010(), rotation);
>+      default:
>+        RTC_NOTREACHED();
>+    }
>+    return nullptr;
>+  }
>+
>+  static rtc::scoped_refptr<PlanarYuvBuffer> CropAndScaleFrom(
>+      const VideoFrameBuffer& src,
>+      int offset_x,
>+      int offset_y,
>+      int crop_width,
>+      int crop_height) {
>+    switch (src.type()) {
>+      case VideoFrameBuffer::Type::kI420: {
>+        rtc::scoped_refptr<I420Buffer> buffer =
>+            I420Buffer::Create(crop_width, crop_height);
>+        buffer->CropAndScaleFrom(*src.GetI420(), offset_x, offset_y, crop_width,
>+                                 crop_height);
>+        return buffer;
>+      }
>+      case VideoFrameBuffer::Type::kI010: {
>+        rtc::scoped_refptr<I010Buffer> buffer =
>+            I010Buffer::Create(crop_width, crop_height);
>+        buffer->CropAndScaleFrom(*src.GetI010(), offset_x, offset_y, crop_width,
>+                                 crop_height);
>+        return buffer;
>+      }
>+      default:
>+        RTC_NOTREACHED();
>+    }
>+    return nullptr;
>+  }
>+
>+  static rtc::scoped_refptr<PlanarYuvBuffer> CropAndScaleFrom(
>+      const VideoFrameBuffer& src,
>+      int crop_width,
>+      int crop_height) {
>+    const int out_width =
>+        std::min(src.width(), crop_width * src.height() / crop_height);
>+    const int out_height =
>+        std::min(src.height(), crop_height * src.width() / crop_width);
>+    return CropAndScaleFrom(src, (src.width() - out_width) / 2,
>+                            (src.height() - out_height) / 2, out_width,
>+                            out_height);
>+  }
>+
>+  static rtc::scoped_refptr<PlanarYuvBuffer>
>+  ScaleFrom(const VideoFrameBuffer& src, int crop_width, int crop_height) {
>+    switch (src.type()) {
>+      case VideoFrameBuffer::Type::kI420: {
>+        rtc::scoped_refptr<I420Buffer> buffer =
>+            I420Buffer::Create(crop_width, crop_height);
>+        buffer->ScaleFrom(*src.GetI420());
>+        return buffer;
>+      }
>+      case VideoFrameBuffer::Type::kI010: {
>+        rtc::scoped_refptr<I010Buffer> buffer =
>+            I010Buffer::Create(crop_width, crop_height);
>+        buffer->ScaleFrom(*src.GetI010());
>+        return buffer;
>+      }
>+      default:
>+        RTC_NOTREACHED();
>+    }
>+    return nullptr;
>+  }
>+};
>+
>+rtc::scoped_refptr<PlanarYuvBuffer> CreateGradient(VideoFrameBuffer::Type type,
>+                                                   int width,
>+                                                   int height) {
>+  rtc::scoped_refptr<I420Buffer> buffer(I420Buffer::Create(width, height));
>+  // Initialize with gradient, Y = 128(x/w + y/h), U = 256 x/w, V = 256 y/h
>+  for (int x = 0; x < width; x++) {
>+    for (int y = 0; y < height; y++) {
>+      buffer->MutableDataY()[x + y * width] =
>+          128 * (x * height + y * width) / (width * height);
>+    }
>+  }
>+  int chroma_width = buffer->ChromaWidth();
>+  int chroma_height = buffer->ChromaHeight();
>+  for (int x = 0; x < chroma_width; x++) {
>+    for (int y = 0; y < chroma_height; y++) {
>+      buffer->MutableDataU()[x + y * chroma_width] =
>+          255 * x / (chroma_width - 1);
>+      buffer->MutableDataV()[x + y * chroma_width] =
>+          255 * y / (chroma_height - 1);
>+    }
>+  }
>+  if (type == VideoFrameBuffer::Type::kI420)
>+    return buffer;
>+
>+  RTC_DCHECK(type == VideoFrameBuffer::Type::kI010);
>+  return I010Buffer::Copy(*buffer);
>+}
>+
>+// The offsets and sizes describe the rectangle extracted from the
>+// original (gradient) frame, in relative coordinates where the
>+// original frame correspond to the unit square, 0.0 <= x, y < 1.0.
>+void CheckCrop(const webrtc::I420BufferInterface& frame,
>+               double offset_x,
>+               double offset_y,
>+               double rel_width,
>+               double rel_height) {
>+  int width = frame.width();
>+  int height = frame.height();
>+  // Check that pixel values in the corners match the gradient used
>+  // for initialization.
>+  for (int i = 0; i < 2; i++) {
>+    for (int j = 0; j < 2; j++) {
>+      // Pixel coordinates of the corner.
>+      int x = i * (width - 1);
>+      int y = j * (height - 1);
>+      // Relative coordinates, range 0.0 - 1.0 correspond to the
>+      // size of the uncropped input frame.
>+      double orig_x = offset_x + i * rel_width;
>+      double orig_y = offset_y + j * rel_height;
>+
>+      EXPECT_NEAR(frame.DataY()[x + y * frame.StrideY()] / 256.0,
>+                  (orig_x + orig_y) / 2, 0.02);
>+      EXPECT_NEAR(frame.DataU()[x / 2 + (y / 2) * frame.StrideU()] / 256.0,
>+                  orig_x, 0.02);
>+      EXPECT_NEAR(frame.DataV()[x / 2 + (y / 2) * frame.StrideV()] / 256.0,
>+                  orig_y, 0.02);
>+    }
>+  }
>+}
>+
>+void CheckRotate(int width,
>+                 int height,
>+                 webrtc::VideoRotation rotation,
>+                 const webrtc::I420BufferInterface& rotated) {
>+  int rotated_width = width;
>+  int rotated_height = height;
>+
>+  if (rotation == kVideoRotation_90 || rotation == kVideoRotation_270) {
>+    std::swap(rotated_width, rotated_height);
>+  }
>+  EXPECT_EQ(rotated_width, rotated.width());
>+  EXPECT_EQ(rotated_height, rotated.height());
>+
>+  // Clock-wise order (with 0,0 at top-left)
>+  const struct {
>+    int x;
>+    int y;
>+  } corners[] = {{0, 0}, {1, 0}, {1, 1}, {0, 1}};
>+  // Corresponding corner colors of the frame produced by CreateGradient.
>+  const struct {
>+    int y;
>+    int u;
>+    int v;
>+  } colors[] = {{0, 0, 0}, {127, 255, 0}, {255, 255, 255}, {127, 0, 255}};
>+  int corner_offset = static_cast<int>(rotation) / 90;
>+
>+  for (int i = 0; i < 4; i++) {
>+    int j = (i + corner_offset) % 4;
>+    int x = corners[j].x * (rotated_width - 1);
>+    int y = corners[j].y * (rotated_height - 1);
>+    EXPECT_EQ(colors[i].y, rotated.DataY()[x + y * rotated.StrideY()]);
>+    EXPECT_EQ(colors[i].u,
>+              rotated.DataU()[(x / 2) + (y / 2) * rotated.StrideU()]);
>+    EXPECT_EQ(colors[i].v,
>+              rotated.DataV()[(x / 2) + (y / 2) * rotated.StrideV()]);
>+  }
>+}
>+
>+}  // namespace
>+
>+TEST(TestVideoFrame, WidthHeightValues) {
>+  VideoFrame frame(I420Buffer::Create(10, 10, 10, 14, 90),
>+                   webrtc::kVideoRotation_0,
>+                   789 * rtc::kNumMicrosecsPerMillisec);
>+  const int valid_value = 10;
>+  EXPECT_EQ(valid_value, frame.width());
>+  EXPECT_EQ(valid_value, frame.height());
>+  frame.set_timestamp(123u);
>+  EXPECT_EQ(123u, frame.timestamp());
>+  frame.set_ntp_time_ms(456);
>+  EXPECT_EQ(456, frame.ntp_time_ms());
>+  EXPECT_EQ(789, frame.render_time_ms());
>+}
>+
>+TEST(TestVideoFrame, ShallowCopy) {
>+  uint32_t timestamp = 1;
>+  int64_t ntp_time_ms = 2;
>+  int64_t timestamp_us = 3;
>+  int stride_y = 15;
>+  int stride_u = 10;
>+  int stride_v = 10;
>+  int width = 15;
>+  int height = 15;
>+
>+  const int kSizeY = 400;
>+  const int kSizeU = 100;
>+  const int kSizeV = 100;
>+  const VideoRotation kRotation = kVideoRotation_270;
>+  uint8_t buffer_y[kSizeY];
>+  uint8_t buffer_u[kSizeU];
>+  uint8_t buffer_v[kSizeV];
>+  memset(buffer_y, 16, kSizeY);
>+  memset(buffer_u, 8, kSizeU);
>+  memset(buffer_v, 4, kSizeV);
>+
>+  VideoFrame frame1(I420Buffer::Copy(width, height, buffer_y, stride_y,
>+                                     buffer_u, stride_u, buffer_v, stride_v),
>+                    kRotation, 0);
>+  frame1.set_timestamp(timestamp);
>+  frame1.set_ntp_time_ms(ntp_time_ms);
>+  frame1.set_timestamp_us(timestamp_us);
>+  VideoFrame frame2(frame1);
>+
>+  EXPECT_EQ(frame1.video_frame_buffer(), frame2.video_frame_buffer());
>+  rtc::scoped_refptr<I420BufferInterface> yuv1 =
>+      frame1.video_frame_buffer()->GetI420();
>+  rtc::scoped_refptr<I420BufferInterface> yuv2 =
>+      frame2.video_frame_buffer()->GetI420();
>+  EXPECT_EQ(yuv1->DataY(), yuv2->DataY());
>+  EXPECT_EQ(yuv1->DataU(), yuv2->DataU());
>+  EXPECT_EQ(yuv1->DataV(), yuv2->DataV());
>+
>+  EXPECT_EQ(frame2.timestamp(), frame1.timestamp());
>+  EXPECT_EQ(frame2.ntp_time_ms(), frame1.ntp_time_ms());
>+  EXPECT_EQ(frame2.timestamp_us(), frame1.timestamp_us());
>+  EXPECT_EQ(frame2.rotation(), frame1.rotation());
>+
>+  frame2.set_timestamp(timestamp + 1);
>+  frame2.set_ntp_time_ms(ntp_time_ms + 1);
>+  frame2.set_timestamp_us(timestamp_us + 1);
>+  frame2.set_rotation(kVideoRotation_90);
>+
>+  EXPECT_NE(frame2.timestamp(), frame1.timestamp());
>+  EXPECT_NE(frame2.ntp_time_ms(), frame1.ntp_time_ms());
>+  EXPECT_NE(frame2.timestamp_us(), frame1.timestamp_us());
>+  EXPECT_NE(frame2.rotation(), frame1.rotation());
>+}
>+
>+TEST(TestVideoFrame, TextureInitialValues) {
>+  VideoFrame frame = test::FakeNativeBuffer::CreateFrame(
>+      640, 480, 100, 10, webrtc::kVideoRotation_0);
>+  EXPECT_EQ(640, frame.width());
>+  EXPECT_EQ(480, frame.height());
>+  EXPECT_EQ(100u, frame.timestamp());
>+  EXPECT_EQ(10, frame.render_time_ms());
>+  ASSERT_TRUE(frame.video_frame_buffer() != nullptr);
>+  EXPECT_TRUE(frame.video_frame_buffer()->type() ==
>+              VideoFrameBuffer::Type::kNative);
>+
>+  frame.set_timestamp(200);
>+  EXPECT_EQ(200u, frame.timestamp());
>+  frame.set_timestamp_us(20);
>+  EXPECT_EQ(20, frame.timestamp_us());
>+}
>+
>+class TestPlanarYuvBuffer
>+    : public ::testing::TestWithParam<VideoFrameBuffer::Type> {};
>+
>+rtc::scoped_refptr<I420Buffer> CreateAndFillBuffer() {
>+  auto buf = I420Buffer::Create(20, 10);
>+  memset(buf->MutableDataY(), 1, 200);
>+  memset(buf->MutableDataU(), 2, 50);
>+  memset(buf->MutableDataV(), 3, 50);
>+  return buf;
>+}
>+
>+TEST_P(TestPlanarYuvBuffer, Copy) {
>+  rtc::scoped_refptr<PlanarYuvBuffer> buf1;
>+  switch (GetParam()) {
>+    case VideoFrameBuffer::Type::kI420: {
>+      buf1 = CreateAndFillBuffer();
>+      break;
>+    }
>+    case VideoFrameBuffer::Type::kI010: {
>+      buf1 = I010Buffer::Copy(*CreateAndFillBuffer());
>+      break;
>+    }
>+    default:
>+      RTC_NOTREACHED();
>+  }
>+
>+  rtc::scoped_refptr<PlanarYuvBuffer> buf2 =
>+      PlanarYuvBufferFactory::Copy(*buf1);
>+  EXPECT_TRUE(test::FrameBufsEqual(buf1->ToI420(), buf2->ToI420()));
>+}
>+
>+TEST_P(TestPlanarYuvBuffer, Scale) {
>+  rtc::scoped_refptr<PlanarYuvBuffer> buf =
>+      CreateGradient(GetParam(), 200, 100);
>+
>+  // Pure scaling, no cropping.
>+  rtc::scoped_refptr<PlanarYuvBuffer> scaled_buffer =
>+      PlanarYuvBufferFactory::ScaleFrom(*buf, 150, 75);
>+  CheckCrop(*scaled_buffer->ToI420(), 0.0, 0.0, 1.0, 1.0);
>+}
>+
>+TEST_P(TestPlanarYuvBuffer, CropXCenter) {
>+  rtc::scoped_refptr<PlanarYuvBuffer> buf =
>+      CreateGradient(GetParam(), 200, 100);
>+
>+  // Pure center cropping, no scaling.
>+  rtc::scoped_refptr<PlanarYuvBuffer> scaled_buffer =
>+      PlanarYuvBufferFactory::CropAndScaleFrom(*buf, 50, 0, 100, 100);
>+  CheckCrop(*scaled_buffer->ToI420(), 0.25, 0.0, 0.5, 1.0);
>+}
>+
>+TEST_P(TestPlanarYuvBuffer, CropXNotCenter) {
>+  rtc::scoped_refptr<PlanarYuvBuffer> buf =
>+      CreateGradient(GetParam(), 200, 100);
>+
>+  // Non-center cropping, no scaling.
>+  rtc::scoped_refptr<PlanarYuvBuffer> scaled_buffer =
>+      PlanarYuvBufferFactory::CropAndScaleFrom(*buf, 25, 0, 100, 100);
>+  CheckCrop(*scaled_buffer->ToI420(), 0.125, 0.0, 0.5, 1.0);
>+}
>+
>+TEST_P(TestPlanarYuvBuffer, CropYCenter) {
>+  rtc::scoped_refptr<PlanarYuvBuffer> buf =
>+      CreateGradient(GetParam(), 100, 200);
>+
>+  // Pure center cropping, no scaling.
>+  rtc::scoped_refptr<PlanarYuvBuffer> scaled_buffer =
>+      PlanarYuvBufferFactory::CropAndScaleFrom(*buf, 0, 50, 100, 100);
>+  CheckCrop(*scaled_buffer->ToI420(), 0.0, 0.25, 1.0, 0.5);
>+}
>+
>+TEST_P(TestPlanarYuvBuffer, CropYNotCenter) {
>+  rtc::scoped_refptr<PlanarYuvBuffer> buf =
>+      CreateGradient(GetParam(), 100, 200);
>+
>+  // Pure center cropping, no scaling.
>+  rtc::scoped_refptr<PlanarYuvBuffer> scaled_buffer =
>+      PlanarYuvBufferFactory::CropAndScaleFrom(*buf, 0, 25, 100, 100);
>+  CheckCrop(*scaled_buffer->ToI420(), 0.0, 0.125, 1.0, 0.5);
>+}
>+
>+TEST_P(TestPlanarYuvBuffer, CropAndScale16x9) {
>+  rtc::scoped_refptr<PlanarYuvBuffer> buf =
>+      CreateGradient(GetParam(), 640, 480);
>+
>+  // Pure center cropping, no scaling.
>+  rtc::scoped_refptr<PlanarYuvBuffer> scaled_buffer =
>+      PlanarYuvBufferFactory::CropAndScaleFrom(*buf, 320, 180);
>+  CheckCrop(*scaled_buffer->ToI420(), 0.0, 0.125, 1.0, 0.75);
>+}
>+
>+INSTANTIATE_TEST_CASE_P(,
>+                        TestPlanarYuvBuffer,
>+                        ::testing::Values(VideoFrameBuffer::Type::kI420,
>+                                          VideoFrameBuffer::Type::kI010));
>+
>+class TestPlanarYuvBufferRotate
>+    : public ::testing::TestWithParam<
>+          std::tuple<webrtc::VideoRotation, VideoFrameBuffer::Type>> {};
>+
>+TEST_P(TestPlanarYuvBufferRotate, Rotates) {
>+  const webrtc::VideoRotation rotation = std::get<0>(GetParam());
>+  const VideoFrameBuffer::Type type = std::get<1>(GetParam());
>+  rtc::scoped_refptr<PlanarYuvBuffer> buffer = CreateGradient(type, 640, 480);
>+  rtc::scoped_refptr<PlanarYuvBuffer> rotated_buffer =
>+      PlanarYuvBufferFactory::Rotate(*buffer, rotation);
>+  CheckRotate(640, 480, rotation, *rotated_buffer->ToI420());
>+}
>+
>+INSTANTIATE_TEST_CASE_P(
>+    Rotate,
>+    TestPlanarYuvBufferRotate,
>+    ::testing::Combine(::testing::Values(kVideoRotation_0,
>+                                         kVideoRotation_90,
>+                                         kVideoRotation_180,
>+                                         kVideoRotation_270),
>+                       ::testing::Values(VideoFrameBuffer::Type::kI420,
>+                                         VideoFrameBuffer::Type::kI010)));
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/video_render_frames.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/video_render_frames.cc
>index 5c732df2cf8..02a105ac07a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/video_render_frames.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/video_render_frames.cc
>@@ -38,6 +38,8 @@ uint32_t EnsureValidRenderDelay(uint32_t render_delay) {
> VideoRenderFrames::VideoRenderFrames(uint32_t render_delay_ms)
>     : render_delay_ms_(EnsureValidRenderDelay(render_delay_ms)) {}
> 
>+VideoRenderFrames::~VideoRenderFrames() = default;
>+
> int32_t VideoRenderFrames::AddFrame(VideoFrame&& new_frame) {
>   const int64_t time_now = rtc::TimeMillis();
> 
>@@ -73,8 +75,8 @@ int32_t VideoRenderFrames::AddFrame(VideoFrame&& new_frame) {
>   return static_cast<int32_t>(incoming_frames_.size());
> }
> 
>-rtc::Optional<VideoFrame> VideoRenderFrames::FrameToRender() {
>-  rtc::Optional<VideoFrame> render_frame;
>+absl::optional<VideoFrame> VideoRenderFrames::FrameToRender() {
>+  absl::optional<VideoFrame> render_frame;
>   // Get the newest frame that can be released for rendering.
>   while (!incoming_frames_.empty() && TimeToNextFrameRelease() <= 0) {
>     render_frame = std::move(incoming_frames_.front());
>@@ -88,8 +90,7 @@ uint32_t VideoRenderFrames::TimeToNextFrameRelease() {
>     return kEventMaxWaitTimeMs;
>   }
>   const int64_t time_to_release = incoming_frames_.front().render_time_ms() -
>-                                  render_delay_ms_ -
>-                                  rtc::TimeMillis();
>+                                  render_delay_ms_ - rtc::TimeMillis();
>   return time_to_release < 0 ? 0u : static_cast<uint32_t>(time_to_release);
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/video_render_frames.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/video_render_frames.h
>index af254f2c63b..0b47e0a0586 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/video_render_frames.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/video_render_frames.h
>@@ -15,7 +15,7 @@
> 
> #include <list>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> #include "api/video/video_frame.h"
> 
> namespace webrtc {
>@@ -25,12 +25,13 @@ class VideoRenderFrames {
>  public:
>   explicit VideoRenderFrames(uint32_t render_delay_ms);
>   VideoRenderFrames(const VideoRenderFrames&) = delete;
>+  ~VideoRenderFrames();
> 
>   // Add a frame to the render queue
>   int32_t AddFrame(VideoFrame&& new_frame);
> 
>   // Get a frame for rendering, or false if it's not time to render.
>-  rtc::Optional<VideoFrame> FrameToRender();
>+  absl::optional<VideoFrame> FrameToRender();
> 
>   // Returns the number of ms to next frame to render
>   uint32_t TimeToNextFrameRelease();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/data/audio_processing/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/data/audio_processing/OWNERS
>new file mode 100644
>index 00000000000..59d718ff995
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/data/audio_processing/OWNERS
>@@ -0,0 +1,3 @@
>+peah@webrtc.org
>+henrik.lundin@webrtc.org
>+
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/data/audio_processing/android/output_data_fixed.pb b/Source/ThirdParty/libwebrtc/Source/webrtc/data/audio_processing/android/output_data_fixed.pb
>new file mode 100644
>index 00000000000..2f45fd367eb
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/data/audio_processing/android/output_data_fixed.pb differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/data/audio_processing/android/output_data_float.pb b/Source/ThirdParty/libwebrtc/Source/webrtc/data/audio_processing/android/output_data_float.pb
>new file mode 100644
>index 00000000000..1bf18c2a2a6
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/data/audio_processing/android/output_data_float.pb differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/data/common_video/jpeg/webrtc_logo.jpg b/Source/ThirdParty/libwebrtc/Source/webrtc/data/common_video/jpeg/webrtc_logo.jpg
>new file mode 100644
>index 00000000000..ddb6192bc7b
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/data/common_video/jpeg/webrtc_logo.jpg differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/data/rtp_rtcp/H263Foreman_CIF_Iframe.bin b/Source/ThirdParty/libwebrtc/Source/webrtc/data/rtp_rtcp/H263Foreman_CIF_Iframe.bin
>new file mode 100644
>index 00000000000..00e3f8058ee
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/data/rtp_rtcp/H263Foreman_CIF_Iframe.bin differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/data/rtp_rtcp/H263Foreman_CIF_Pframe.bin b/Source/ThirdParty/libwebrtc/Source/webrtc/data/rtp_rtcp/H263Foreman_CIF_Pframe.bin
>new file mode 100644
>index 00000000000..57f94c3a525
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/data/rtp_rtcp/H263Foreman_CIF_Pframe.bin differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/data/rtp_rtcp/H263_CIF_IFRAME.bin b/Source/ThirdParty/libwebrtc/Source/webrtc/data/rtp_rtcp/H263_CIF_IFRAME.bin
>new file mode 100644
>index 00000000000..00e3f8058ee
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/data/rtp_rtcp/H263_CIF_IFRAME.bin differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/data/rtp_rtcp/H263_CIF_PFRAME.bin b/Source/ThirdParty/libwebrtc/Source/webrtc/data/rtp_rtcp/H263_CIF_PFRAME.bin
>new file mode 100644
>index 00000000000..248f3a19834
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/data/rtp_rtcp/H263_CIF_PFRAME.bin differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/data/rtp_rtcp/H263_QCIF_IFRAME.bin b/Source/ThirdParty/libwebrtc/Source/webrtc/data/rtp_rtcp/H263_QCIF_IFRAME.bin
>new file mode 100644
>index 00000000000..0fa144c9c17
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/data/rtp_rtcp/H263_QCIF_IFRAME.bin differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/data/rtp_rtcp/RTCPPacketTMMBR0.bin b/Source/ThirdParty/libwebrtc/Source/webrtc/data/rtp_rtcp/RTCPPacketTMMBR0.bin
>new file mode 100644
>index 00000000000..19df13c4d74
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/data/rtp_rtcp/RTCPPacketTMMBR0.bin differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/data/rtp_rtcp/RTCPPacketTMMBR1.bin b/Source/ThirdParty/libwebrtc/Source/webrtc/data/rtp_rtcp/RTCPPacketTMMBR1.bin
>new file mode 100644
>index 00000000000..b7b7c941ed6
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/data/rtp_rtcp/RTCPPacketTMMBR1.bin differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/data/rtp_rtcp/RTCPPacketTMMBR2.bin b/Source/ThirdParty/libwebrtc/Source/webrtc/data/rtp_rtcp/RTCPPacketTMMBR2.bin
>new file mode 100644
>index 00000000000..257835c616d
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/data/rtp_rtcp/RTCPPacketTMMBR2.bin differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/data/rtp_rtcp/RTCPPacketTMMBR3.bin b/Source/ThirdParty/libwebrtc/Source/webrtc/data/rtp_rtcp/RTCPPacketTMMBR3.bin
>new file mode 100644
>index 00000000000..4a8e375ee3f
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/data/rtp_rtcp/RTCPPacketTMMBR3.bin differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/data/rtp_rtcp/RTCPPacketTMMBR4.bin b/Source/ThirdParty/libwebrtc/Source/webrtc/data/rtp_rtcp/RTCPPacketTMMBR4.bin
>new file mode 100644
>index 00000000000..28cd99ce9f1
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/data/rtp_rtcp/RTCPPacketTMMBR4.bin differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/data/rtp_rtcp/RTCPPacketTMMBR4_1.bin b/Source/ThirdParty/libwebrtc/Source/webrtc/data/rtp_rtcp/RTCPPacketTMMBR4_1.bin
>new file mode 100644
>index 00000000000..5080b88ab9f
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/data/rtp_rtcp/RTCPPacketTMMBR4_1.bin differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/data/rtp_rtcp/RTCPPacketTMMBR4_2.bin b/Source/ThirdParty/libwebrtc/Source/webrtc/data/rtp_rtcp/RTCPPacketTMMBR4_2.bin
>new file mode 100644
>index 00000000000..2c2f288fcb1
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/data/rtp_rtcp/RTCPPacketTMMBR4_2.bin differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/data/rtp_rtcp/RTCPPacketTMMBR5.bin b/Source/ThirdParty/libwebrtc/Source/webrtc/data/rtp_rtcp/RTCPPacketTMMBR5.bin
>new file mode 100644
>index 00000000000..da7235a56eb
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/data/rtp_rtcp/RTCPPacketTMMBR5.bin differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/audio_long16.wav b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/audio_long16.wav
>new file mode 100644
>index 00000000000..ebe91c44377
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/audio_long16.wav differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/audio_long16big_endian.pcm b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/audio_long16big_endian.pcm
>new file mode 100644
>index 00000000000..563e4e9932a
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/audio_long16big_endian.pcm differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/audio_long16noise.pcm b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/audio_long16noise.pcm
>new file mode 100644
>index 00000000000..a7be53779c2
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/audio_long16noise.pcm differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/audio_long8.pcm b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/audio_long8.pcm
>new file mode 100644
>index 00000000000..85d17e5dd91
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/audio_long8.pcm differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/audio_long8mulaw.wav b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/audio_long8mulaw.wav
>new file mode 100644
>index 00000000000..2d3d8b3ff44
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/audio_long8mulaw.wav differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/audio_short16.pcm b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/audio_short16.pcm
>new file mode 100644
>index 00000000000..15a0f1811c7
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/audio_short16.pcm differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/audio_tiny11.wav b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/audio_tiny11.wav
>new file mode 100644
>index 00000000000..6db80d536ee
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/audio_tiny11.wav differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/audio_tiny16.wav b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/audio_tiny16.wav
>new file mode 100644
>index 00000000000..baab0acce59
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/audio_tiny16.wav differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/audio_tiny22.wav b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/audio_tiny22.wav
>new file mode 100644
>index 00000000000..b4218679735
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/audio_tiny22.wav differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/audio_tiny32.wav b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/audio_tiny32.wav
>new file mode 100644
>index 00000000000..773ac23eb2e
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/audio_tiny32.wav differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/audio_tiny44.wav b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/audio_tiny44.wav
>new file mode 100644
>index 00000000000..c9faa45f504
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/audio_tiny44.wav differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/audio_tiny8.wav b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/audio_tiny8.wav
>new file mode 100644
>index 00000000000..d71c65e89e4
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/audio_tiny8.wav differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/stereo_rtp_files/HRTF_pcm16wb.rtp b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/stereo_rtp_files/HRTF_pcm16wb.rtp
>new file mode 100644
>index 00000000000..02abbc24f7f
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/stereo_rtp_files/HRTF_pcm16wb.rtp differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/stereo_rtp_files/HRTF_pcm16wb_jitter.rtp b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/stereo_rtp_files/HRTF_pcm16wb_jitter.rtp
>new file mode 100644
>index 00000000000..4ed110b5fa8
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/stereo_rtp_files/HRTF_pcm16wb_jitter.rtp differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/stereo_rtp_files/README.txt b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/stereo_rtp_files/README.txt
>new file mode 100644
>index 00000000000..976ac56c88b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/stereo_rtp_files/README.txt
>@@ -0,0 +1,4 @@
>+Use RTP Play tool with command 'rtpplay.exe -v -T -f <path>\<file.rtp> 127.0.0.1/1236' 
>+Example: rtpplay.exe -v -T -f hrtf_g722_1C_48.rtp 127.0.0.1/1236.  
>+This sends the stereo rtp file to port 1236.  
>+You can hear the voice getting panned from left, right and center.   
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/stereo_rtp_files/hrtf_g722_1C_48.rtp b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/stereo_rtp_files/hrtf_g722_1C_48.rtp
>new file mode 100644
>index 00000000000..b96d59b89fc
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/stereo_rtp_files/hrtf_g722_1C_48.rtp differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/stereo_rtp_files/hrtf_g722_1C_48_jitterT2.rtp b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/stereo_rtp_files/hrtf_g722_1C_48_jitterT2.rtp
>new file mode 100644
>index 00000000000..527a50a5c73
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/stereo_rtp_files/hrtf_g722_1C_48_jitterT2.rtp differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/stereo_rtp_files/rtpplay.exe b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/stereo_rtp_files/rtpplay.exe
>new file mode 100755
>index 00000000000..6f938c8fb22
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/stereo_rtp_files/rtpplay.exe differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/stereo_rtp_files/stereo_g729.rtp b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/stereo_rtp_files/stereo_g729.rtp
>new file mode 100644
>index 00000000000..3c36e30786e
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/stereo_rtp_files/stereo_g729.rtp differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/stereo_rtp_files/stereo_g729_jitter.rtp b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/stereo_rtp_files/stereo_g729_jitter.rtp
>new file mode 100644
>index 00000000000..913226cc919
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/stereo_rtp_files/stereo_g729_jitter.rtp differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/stereo_rtp_files/stereo_pcm16wb.rtp b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/stereo_rtp_files/stereo_pcm16wb.rtp
>new file mode 100644
>index 00000000000..729b565178b
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/stereo_rtp_files/stereo_pcm16wb.rtp differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/stereo_rtp_files/stereo_pcm16wb_jitter.rtp b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/stereo_rtp_files/stereo_pcm16wb_jitter.rtp
>new file mode 100644
>index 00000000000..efa2800c132
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/stereo_rtp_files/stereo_pcm16wb_jitter.rtp differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/stereo_rtp_files/stereo_pcmu.rtp b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/stereo_rtp_files/stereo_pcmu.rtp
>new file mode 100644
>index 00000000000..bb2d93c1c47
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/stereo_rtp_files/stereo_pcmu.rtp differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/stereo_rtp_files/stereo_pcmu_jitter.rtp b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/stereo_rtp_files/stereo_pcmu_jitter.rtp
>new file mode 100644
>index 00000000000..fb7937863d0
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/stereo_rtp_files/stereo_pcmu_jitter.rtp differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/stereo_rtp_files/stereo_pcmu_vad.rtp b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/stereo_rtp_files/stereo_pcmu_vad.rtp
>new file mode 100644
>index 00000000000..eebcf349854
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/stereo_rtp_files/stereo_pcmu_vad.rtp differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/stereo_rtp_files/stereo_pcmu_vad_jitter.rtp b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/stereo_rtp_files/stereo_pcmu_vad_jitter.rtp
>new file mode 100644
>index 00000000000..5c368b4bcf9
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/stereo_rtp_files/stereo_pcmu_vad_jitter.rtp differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/stereo_rtp_files/toggling_stereo_g729_pt18_pt125.rtp b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/stereo_rtp_files/toggling_stereo_g729_pt18_pt125.rtp
>new file mode 100644
>index 00000000000..1f713f6e813
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/data/voice_engine/stereo_rtp_files/toggling_stereo_g729_pt18_pt125.rtp differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/BUILD.gn
>index eef20d4bb89..5a19a372ec0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/BUILD.gn
>@@ -7,6 +7,7 @@
> # be found in the AUTHORS file in the root of the source tree.
> 
> import("../webrtc.gni")
>+
> if (is_android) {
>   import("//build/config/android/config.gni")
>   import("//build/config/android/rules.gni")
>@@ -19,26 +20,33 @@ if (is_android) {
> group("examples") {
>   # This target shall build all targets in examples.
>   testonly = true
>-  public_deps = []
>+  deps = []
> 
>   if (is_android) {
>-    public_deps += [
>+    deps += [
>       ":AppRTCMobile",
>-      ":AppRTCMobileTest",
>-      ":AppRTCMobileTestStubbedVideoIO",
>+      ":AppRTCMobile_stubbed_video_io_test_apk",
>+      ":AppRTCMobile_test_apk",
>+      ":libwebrtc_unity",
>     ]
>+
>+    # TODO(sakal): We include some code from the tests. Remove this dependency
>+    # and remove this if-clause.
>+    if (rtc_include_tests) {
>+      deps += [ "androidnativeapi:androidnativeapi" ]
>+    }
>   }
> 
>   if (!build_with_chromium) {
>-    public_deps += [ ":stun_prober" ]
>+    deps += [ ":stun_prober" ]
>   }
> 
>   if (is_ios || (is_mac && target_cpu != "x86")) {
>-    public_deps += [ ":AppRTCMobile" ]
>+    deps += [ ":AppRTCMobile" ]
>   }
> 
>   if (is_linux || is_win) {
>-    public_deps += [
>+    deps += [
>       ":peerconnection_client",
>       ":peerconnection_server",
>       ":relayserver",
>@@ -46,6 +54,10 @@ group("examples") {
>       ":turnserver",
>     ]
>   }
>+
>+  if (is_android || is_win) {
>+    deps += [ ":webrtc_unity_plugin" ]
>+  }
> }
> 
> if (is_android) {
>@@ -58,7 +70,6 @@ if (is_android) {
>       ":AppRTCMobile_javalib",
>       ":AppRTCMobile_resources",
>       "../rtc_base:base_java",
>-      "//base:base_java",
>     ]
> 
>     shared_libraries = [ "../sdk/android:libjingle_peerconnection_so" ]
>@@ -66,7 +77,7 @@ if (is_android) {
> 
>   rtc_android_library("AppRTCMobile_javalib") {
>     testonly = true
>-    android_manifest = "androidapp/AndroidManifest.xml"
>+    android_manifest_for_lint = "androidapp/AndroidManifest.xml"
> 
>     java_files = [
>       "androidapp/src/org/appspot/apprtc/AppRTCAudioManager.java",
>@@ -82,6 +93,8 @@ if (is_android) {
>       "androidapp/src/org/appspot/apprtc/HudFragment.java",
>       "androidapp/src/org/appspot/apprtc/PeerConnectionClient.java",
>       "androidapp/src/org/appspot/apprtc/RoomParametersFetcher.java",
>+      "androidapp/src/org/appspot/apprtc/RtcEventLog.java",
>+      "androidapp/src/org/appspot/apprtc/RecordedAudioToFileController.java",
>       "androidapp/src/org/appspot/apprtc/SettingsActivity.java",
>       "androidapp/src/org/appspot/apprtc/SettingsFragment.java",
>       "androidapp/src/org/appspot/apprtc/TCPChannelClient.java",
>@@ -96,9 +109,23 @@ if (is_android) {
>       ":AppRTCMobile_resources",
>       "../modules/audio_device:audio_device_java",
>       "../rtc_base:base_java",
>+      "../sdk/android:audio_api_java",
>+      "../sdk/android:base_java",
>+      "../sdk/android:camera_java",
>+      "../sdk/android:default_video_codec_factory_java",
>+      "../sdk/android:filevideo_java",
>+      "../sdk/android:hwcodecs_java",
>+      "../sdk/android:java_audio_device_module_java",
>       "../sdk/android:libjingle_peerconnection_java",
>       "../sdk/android:libjingle_peerconnection_metrics_default_java",
>+      "../sdk/android:peerconnection_java",
>+      "../sdk/android:screencapturer_java",
>+      "../sdk/android:surfaceviewrenderer_java",
>+      "../sdk/android:swcodecs_java",
>+      "../sdk/android:video_api_java",
>+      "../sdk/android:video_java",
>       "androidapp/third_party/autobanh:autobanh_java",
>+      "//third_party/jsr-305:jsr_305_javalib",
>     ]
>   }
> 
>@@ -108,23 +135,26 @@ if (is_android) {
>     custom_package = "org.appspot.apprtc"
>   }
> 
>-  rtc_instrumentation_test_apk("AppRTCMobileTest") {
>+  rtc_instrumentation_test_apk("AppRTCMobile_test_apk") {
>     apk_name = "AppRTCMobileTest"
>     android_manifest = "androidtests/AndroidManifest.xml"
> 
>-    java_files = [ "androidtests/src/org/appspot/apprtc/test/PeerConnectionClientTest.java" ]
>+    java_files = [
>+      "androidtests/src/org/appspot/apprtc/test/PeerConnectionClientTest.java",
>+    ]
> 
>     apk_under_test = ":AppRTCMobile"
> 
>     deps = [
>       ":AppRTCMobile_javalib",
>       "../sdk/android:libjingle_peerconnection_java",
>+      "../sdk/android:video_java",
>       "//third_party/android_support_test_runner:runner_java",
>       "//third_party/junit",
>     ]
>   }
> 
>-  rtc_instrumentation_test_apk("AppRTCMobileTestStubbedVideoIO") {
>+  rtc_instrumentation_test_apk("AppRTCMobile_stubbed_video_io_test_apk") {
>     apk_name = "AppRTCMobileTestStubbedVideoIO"
>     android_manifest = "androidtests/AndroidManifest.xml"
> 
>@@ -143,7 +173,19 @@ if (is_android) {
>     ]
> 
>     data = [
>+      "../build/android/adb_reverse_forwarder.py",
>+      "../examples/androidtests/video_quality_loopback_test.py",
>       "../resources/reference_video_640x360_30fps.y4m",
>+      "../rtc_tools/barcode_tools/barcode_decoder.py",
>+      "../rtc_tools/barcode_tools/helper_functions.py",
>+      "../rtc_tools/compare_videos.py",
>+      "../rtc_tools/testing/prebuilt_apprtc.zip",
>+      "../rtc_tools/testing/golang/linux/go.tar.gz",
>+      "../rtc_tools/testing/build_apprtc.py",
>+      "../rtc_tools/testing/utils.py",
>+      "../tools_webrtc/video_quality_toolchain/linux/ffmpeg",
>+      "../tools_webrtc/video_quality_toolchain/linux/frame_analyzer",
>+      "../tools_webrtc/video_quality_toolchain/linux/zxing",
>     ]
>   }
> }
>@@ -162,28 +204,27 @@ if (is_ios || (is_mac && target_cpu != "x86")) {
>     public_configs = [ ":apprtc_common_config" ]
> 
>     if (is_ios) {
>+      # iOS must use WebRTC.framework which is dynamically linked.
>+      # 'gn check' is disabled in order to avoid confusion and
>+      # errors caused by multiple implementations.
>+      check_includes = false
>       deps = [
>-        ":AppRTCMobile_ios_frameworks",
>+        "../sdk:framework_objc",
>+        "../system_wrappers:field_trial_default",
>+        "../system_wrappers:runtime_enabled_features_default",
>       ]
>     } else {
>       deps = [
>         "../sdk:common_objc",
>         "../system_wrappers:field_trial_default",
>         "../system_wrappers:metrics_default",
>+        "../system_wrappers:runtime_enabled_features_default",
>       ]
>     }
>   }
> 
>   config("apprtc_signaling_config") {
>     include_dirs = [ "objc/AppRTCMobile" ]
>-
>-    # GN orders flags on a target before flags from configs. The default config
>-    # adds these flags so to cancel them out they need to come from a config and
>-    # cannot be on the target directly.
>-    cflags = [
>-      "-Wno-sign-compare",
>-      "-Wno-unused-variable",
>-    ]
>   }
> 
>   rtc_static_library("apprtc_signaling") {
>@@ -198,6 +239,8 @@ if (is_ios || (is_mac && target_cpu != "x86")) {
>       "objc/AppRTCMobile/ARDBitrateTracker.m",
>       "objc/AppRTCMobile/ARDCaptureController.h",
>       "objc/AppRTCMobile/ARDCaptureController.m",
>+      "objc/AppRTCMobile/ARDExternalSampleCapturer.h",
>+      "objc/AppRTCMobile/ARDExternalSampleCapturer.m",
>       "objc/AppRTCMobile/ARDJoinResponse+Internal.h",
>       "objc/AppRTCMobile/ARDJoinResponse.h",
>       "objc/AppRTCMobile/ARDJoinResponse.m",
>@@ -235,10 +278,23 @@ if (is_ios || (is_mac && target_cpu != "x86")) {
>       ":socketrocket",
>     ]
>     if (is_ios) {
>-      deps += [ ":AppRTCMobile_ios_frameworks" ]
>+      # iOS must use WebRTC.framework which is dynamically linked.
>+      # 'gn check' is disabled in order to avoid confusion and
>+      # errors caused by multiple implementations.
>+      check_includes = false
>+      deps += [
>+        ":AppRTCMobile_ios_frameworks",
>+        "../sdk:framework_objc",
>+      ]
>     } else {
>-      public_deps = [
>-        "../sdk:peerconnection_objc",
>+      deps += [
>+        "../sdk:common_objc",
>+        "../sdk:mediaconstraints_objc",
>+        "../sdk:peerconnectionfactory_base_objc",
>+        "../sdk:videocapture_objc",
>+        "../sdk:videocodec_objc",
>+        "../sdk:videoframebuffer_objc",
>+        "../sdk:videosource_objc",
>       ]
>     }
>     libs = [ "QuartzCore.framework" ]
>@@ -246,8 +302,13 @@ if (is_ios || (is_mac && target_cpu != "x86")) {
> 
>   if (is_ios) {
>     rtc_static_library("AppRTCMobile_lib") {
>+      # iOS must use WebRTC.framework which is dynamically linked.
>+      # 'gn check' is disabled in order to avoid confusion and
>+      # errors caused by multiple implementations.
>+      check_includes = false
>       testonly = true
>       sources = [
>+        "objc/AppRTCMobile/ios/ARDAppDelegate.h",
>         "objc/AppRTCMobile/ios/ARDAppDelegate.m",
>         "objc/AppRTCMobile/ios/ARDFileCaptureController.h",
>         "objc/AppRTCMobile/ios/ARDFileCaptureController.m",
>@@ -273,6 +334,7 @@ if (is_ios || (is_mac && target_cpu != "x86")) {
>         ":AppRTCMobile_ios_frameworks",
>         ":apprtc_common",
>         ":apprtc_signaling",
>+        "../sdk:framework_objc",
>       ]
>     }
> 
>@@ -291,15 +353,96 @@ if (is_ios || (is_mac && target_cpu != "x86")) {
>         ":AppRTCMobile_ios_bundle_data",
>         ":AppRTCMobile_ios_frameworks",
>         ":AppRTCMobile_lib",
>+        "../sdk:framework_objc",
>       ]
> 
>+      if (rtc_apprtcmobile_broadcast_extension) {
>+        deps += [
>+          ":AppRTCMobileBroadcastSetupUI_extension_bundle",
>+          ":AppRTCMobileBroadcastUpload_extension_bundle",
>+        ]
>+      }
>+
>       if (target_cpu == "x86") {
>         deps += [ "//testing/iossim:iossim" ]
>       }
>     }
> 
>+    if (rtc_apprtcmobile_broadcast_extension) {
>+      bundle_data("AppRTCMobileBroadcastUpload_extension_bundle") {
>+        testonly = true
>+        public_deps = [
>+          ":AppRTCMobileBroadcastUpload",
>+        ]
>+        sources = [
>+          "$root_out_dir/AppRTCMobileBroadcastUpload.appex",
>+        ]
>+        outputs = [
>+          "{{bundle_plugins_dir}}/{{source_file_part}}",
>+        ]
>+      }
>+
>+      bundle_data("AppRTCMobileBroadcastSetupUI_extension_bundle") {
>+        testonly = true
>+        public_deps = [
>+          ":AppRTCMobileBroadcastSetupUI",
>+        ]
>+        sources = [
>+          "$root_out_dir/AppRTCMobileBroadcastSetupUI.appex",
>+        ]
>+        outputs = [
>+          "{{bundle_plugins_dir}}/{{source_file_part}}",
>+        ]
>+      }
>+
>+      rtc_static_library("AppRTCMobileBroadcastUpload_lib") {
>+        check_includes = false
>+        testonly = true
>+        sources = [
>+          "objc/AppRTCMobile/ios/broadcast_extension/ARDBroadcastSampleHandler.h",
>+          "objc/AppRTCMobile/ios/broadcast_extension/ARDBroadcastSampleHandler.m",
>+        ]
>+
>+        deps = [
>+          ":AppRTCMobile_ios_frameworks",
>+          ":apprtc_signaling",
>+          "../sdk:framework_objc",
>+        ]
>+
>+        libs = [ "ReplayKit.framework" ]
>+      }
>+
>+      ios_appex_bundle("AppRTCMobileBroadcastUpload") {
>+        testonly = true
>+        configs += [ "..:common_config" ]
>+        public_configs = [ "..:common_inherited_config" ]
>+
>+        info_plist = "objc/AppRTCMobile/ios/broadcast_extension/BroadcastUploadInfo.plist"
>+
>+        deps = [
>+          ":AppRTCMobileBroadcastUpload_lib",
>+          "../sdk:framework_objc",
>+        ]
>+      }
>+
>+      ios_appex_bundle("AppRTCMobileBroadcastSetupUI") {
>+        sources = [
>+          "objc/AppRTCMobile/ios/broadcast_extension/ARDBroadcastSetupViewController.h",
>+          "objc/AppRTCMobile/ios/broadcast_extension/ARDBroadcastSetupViewController.m",
>+        ]
>+
>+        info_plist = "objc/AppRTCMobile/ios/broadcast_extension/BroadcastSetupUIInfo.plist"
>+
>+        libs = [ "ReplayKit.framework" ]
>+
>+        deps = [
>+          ":AppRTCMobile_ios_bundle_data",
>+        ]
>+      }
>+    }
>+
>     bundle_data("AppRTCMobile_ios_frameworks") {
>-      public_deps = [
>+      deps = [
>         "../sdk:framework_objc+link",
>       ]
>       sources = [
>@@ -338,6 +481,67 @@ if (is_ios || (is_mac && target_cpu != "x86")) {
>         "{{bundle_resources_dir}}/{{source_file_part}}",
>       ]
>     }
>+
>+    rtc_static_library("ObjCNativeAPIDemo_lib") {
>+      testonly = true
>+      sources = [
>+        "objcnativeapi/objc/NADAppDelegate.h",
>+        "objcnativeapi/objc/NADAppDelegate.m",
>+        "objcnativeapi/objc/NADViewController.h",
>+        "objcnativeapi/objc/NADViewController.mm",
>+        "objcnativeapi/objc/objccallclient.h",
>+        "objcnativeapi/objc/objccallclient.mm",
>+      ]
>+
>+      if (!build_with_chromium && is_clang) {
>+        # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>+        suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>+      }
>+
>+      deps = [
>+        "../api:libjingle_peerconnection_api",
>+        "../api/audio_codecs:builtin_audio_decoder_factory",
>+        "../api/audio_codecs:builtin_audio_encoder_factory",
>+        "../logging:rtc_event_log_impl_base",
>+        "../media:rtc_audio_video",
>+        "../modules/audio_processing:audio_processing",
>+        "../pc:libjingle_peerconnection",
>+        "../rtc_base:rtc_base",
>+        "../sdk:default_codec_factory_objc",
>+        "../sdk:framework_objc",
>+        "../sdk:native_api",
>+        "../sdk:ui_objc",
>+        "../sdk:videotoolbox_objc",
>+        "../system_wrappers:field_trial_default",
>+        "../system_wrappers:metrics_default",
>+        "../system_wrappers:runtime_enabled_features_default",
>+        "//third_party/abseil-cpp/absl/memory",
>+      ]
>+
>+      if (current_cpu == "arm64") {
>+        deps += [ "../sdk:metal_objc" ]
>+      }
>+    }
>+
>+    ios_app_bundle("ObjCNativeAPIDemo") {
>+      testonly = true
>+      sources = [
>+        "objcnativeapi/objc/main.m",
>+      ]
>+
>+      info_plist = "objcnativeapi/Info.plist"
>+
>+      configs += [ "..:common_config" ]
>+      public_configs = [ "..:common_inherited_config" ]
>+
>+      deps = [
>+        ":ObjCNativeAPIDemo_lib",
>+      ]
>+
>+      if (target_cpu == "x86") {
>+        deps += [ "//testing/iossim:iossim" ]
>+      }
>+    }
>   }
> 
>   if (is_mac) {
>@@ -353,7 +557,14 @@ if (is_ios || (is_mac && target_cpu != "x86")) {
>       deps = [
>         ":apprtc_common",
>         ":apprtc_signaling",
>+        "../sdk:default_codec_factory_objc",
>+        "../sdk:metal_objc",
>+        "../sdk:peerconnectionfactory_base_objc",
>         "../sdk:ui_objc",
>+        "../sdk:videocapture_objc",
>+        "../sdk:videocodec_objc",
>+        "../sdk:videocodec_objc",
>+        "../sdk:videotoolbox_objc",
>       ]
>     }
> 
>@@ -417,6 +628,10 @@ if (is_ios || (is_mac && target_cpu != "x86")) {
>     # TODO(kthelgason): compile xctests on mac when chromium supports it.
>     if (is_ios) {
>       rtc_source_set("apprtcmobile_test_sources") {
>+        # iOS must use WebRTC.framework which is dynamically linked.
>+        # 'gn check' is disabled in order to avoid confusion and
>+        # errors caused by multiple implementations.
>+        check_includes = false
>         testonly = true
>         include_dirs = [
>           "objc/AppRTCMobile",
>@@ -429,11 +644,11 @@ if (is_ios || (is_mac && target_cpu != "x86")) {
>           "objc/AppRTCMobile/tests/ARDSettingsModel_xctest.mm",
>         ]
>         deps = [
>-          "../rtc_base:rtc_base",
>-        ]
>-        public_deps = [
>           ":AppRTCMobile_ios_frameworks",
>           ":AppRTCMobile_lib",
>+          ":apprtc_signaling",
>+          "../rtc_base:rtc_base",
>+          "../sdk:framework_objc",
>           "//build/config/ios:xctest",
>           "//third_party/ocmock",
>         ]
>@@ -445,7 +660,9 @@ if (is_ios || (is_mac && target_cpu != "x86")) {
>           "objc/AppRTCMobile/ios/main.m",
>         ]
>         deps = [
>+          ":AppRTCMobile_lib",
>           ":apprtcmobile_test_sources",
>+          "../sdk:framework_objc",
>         ]
>         ldflags = [ "-all_load" ]
>       }
>@@ -454,35 +671,6 @@ if (is_ios || (is_mac && target_cpu != "x86")) {
> }
> 
> if (is_linux || is_win) {
>-  config("peerconnection_client_warnings_config") {
>-    cflags = []
>-    if (is_win && is_clang) {
>-      cflags += [
>-        # Disable warnings failing when compiling with Clang on Windows.
>-        # https://bugs.chromium.org/p/webrtc/issues/detail?id=5366
>-        "-Wno-format",
>-
>-        # See https://bugs.chromium.org/p/webrtc/issues/detail?id=6271
>-        # for -Wno-reorder and -Wno-sign-compare
>-        "-Wno-reorder",
>-        "-Wno-sign-compare",
>-      ]
>-    }
>-    if (is_linux && target_cpu == "x86") {
>-      cflags += [
>-        # Needed to compile on Linux 32-bit.
>-        "-Wno-sentinel",
>-      ]
>-    }
>-
>-    if (is_clang) {
>-      # TODO(ehmaldonado): Make peerconnection_client compile with the standard
>-      # set of warnings.
>-      # See https://bugs.chromium.org/p/webrtc/issues/detail?id=6306
>-      cflags += [ "-Wno-inconsistent-missing-override" ]
>-    }
>-  }
>-
>   rtc_executable("peerconnection_client") {
>     testonly = true
>     sources = [
>@@ -499,7 +687,11 @@ if (is_linux || is_win) {
>       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>     }
>     deps = [
>-      "../api:video_frame_api_i420",
>+      "../api:libjingle_peerconnection_api",
>+      "../api/video:video_frame_i420",
>+      "../rtc_base:checks",
>+      "../rtc_base:stringutils",
>+      "../rtc_base/third_party/sigslot",
>     ]
>     if (is_win) {
>       sources += [
>@@ -508,7 +700,6 @@ if (is_linux || is_win) {
>         "peerconnection/client/main_wnd.cc",
>         "peerconnection/client/main_wnd.h",
>       ]
>-      cflags = [ "/wd4245" ]
>       configs += [ "//build/config/win:windowed" ]
>       deps += [ "../media:rtc_media_base" ]
>     }
>@@ -527,14 +718,18 @@ if (is_linux || is_win) {
>       ]
>       deps += [ "//build/config/linux/gtk" ]
>     }
>-    configs += [ ":peerconnection_client_warnings_config" ]
> 
>     deps += [
>+      "../api:libjingle_peerconnection_api",
>       "../api:libjingle_peerconnection_test_api",
>-      "../api:video_frame_api",
>       "../api/audio_codecs:builtin_audio_decoder_factory",
>       "../api/audio_codecs:builtin_audio_encoder_factory",
>-      "../media:rtc_media",
>+      "../api/video:video_frame",
>+      "../api/video_codecs:builtin_video_decoder_factory",
>+      "../api/video_codecs:builtin_video_encoder_factory",
>+      "../media:rtc_audio_video",
>+      "../modules/audio_device:audio_device",
>+      "../modules/audio_processing:audio_processing",
>       "../modules/video_capture:video_capture_module",
>       "../pc:libjingle_peerconnection",
>       "../rtc_base:rtc_base",
>@@ -542,6 +737,7 @@ if (is_linux || is_win) {
>       "../rtc_base:rtc_json",
>       "../system_wrappers:field_trial_default",
>       "../system_wrappers:metrics_default",
>+      "../system_wrappers:runtime_enabled_features_default",
>       "//third_party/libyuv",
>     ]
>   }
>@@ -560,6 +756,7 @@ if (is_linux || is_win) {
>     deps = [
>       "..:webrtc_common",
>       "../rtc_base:rtc_base_approved",
>+      "../rtc_base:stringutils",
>       "../rtc_tools:command_line_parser",
>     ]
>     if (!build_with_chromium && is_clang) {
>@@ -579,6 +776,7 @@ if (is_linux || is_win) {
>       "../rtc_base:rtc_base_approved",
>       "../system_wrappers:field_trial_default",
>       "../system_wrappers:metrics_default",
>+      "../system_wrappers:runtime_enabled_features_default",
>     ]
>     if (!build_with_chromium && is_clang) {
>       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>@@ -597,6 +795,7 @@ if (is_linux || is_win) {
>       "../rtc_base:rtc_base_approved",
>       "../system_wrappers:field_trial_default",
>       "../system_wrappers:metrics_default",
>+      "../system_wrappers:runtime_enabled_features_default",
>     ]
>     if (!build_with_chromium && is_clang) {
>       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>@@ -615,6 +814,7 @@ if (is_linux || is_win) {
>       "../rtc_base:rtc_base_approved",
>       "../system_wrappers:field_trial_default",
>       "../system_wrappers:metrics_default",
>+      "../system_wrappers:runtime_enabled_features_default",
>     ]
>     if (!build_with_chromium && is_clang) {
>       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>@@ -641,6 +841,7 @@ if (is_win || is_android) {
>         "unityplugin/classreferenceholder.h",
>         "unityplugin/jni_onload.cc",
>       ]
>+      suppressed_configs += [ "//build/config/android:hide_all_but_jni_onload" ]
>     }
> 
>     if (!build_with_chromium && is_clang) {
>@@ -648,25 +849,33 @@ if (is_win || is_android) {
>       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>     }
>     if (is_win) {
>-      cflags = [ "/wd4245" ]
>-      configs += [
>-        "//build/config/win:windowed",
>-        ":peerconnection_client_warnings_config",
>-      ]
>+      configs += [ "//build/config/win:windowed" ]
>     }
>     deps = [
>+      "../api:libjingle_peerconnection_api",
>       "../api:libjingle_peerconnection_test_api",
>-      "../api:video_frame_api",
>+      "../api/audio_codecs:builtin_audio_decoder_factory",
>+      "../api/audio_codecs:builtin_audio_encoder_factory",
>+      "../api/video:video_frame",
>+      "../media:rtc_audio_video",
>+      "../media:rtc_internal_video_codecs",
>       "../media:rtc_media",
>       "../media:rtc_media_base",
>+      "../modules/audio_device:audio_device",
>+      "../modules/audio_processing:audio_processing",
>       "../modules/video_capture:video_capture_module",
>       "../pc:libjingle_peerconnection",
>       "../rtc_base:rtc_base",
>       "../system_wrappers:field_trial_default",
>       "../system_wrappers:metrics_default",
>+      "../system_wrappers:runtime_enabled_features_default",
>+      "//third_party/abseil-cpp/absl/memory",
>     ]
>     if (is_android) {
>-      deps += [ "../sdk/android:libjingle_peerconnection_jni" ]
>+      deps += [
>+        "../modules/utility:utility",
>+        "../sdk/android:libjingle_peerconnection_jni",
>+      ]
>     }
>   }
> }
>@@ -683,8 +892,9 @@ if (is_android) {
>   dist_jar("libwebrtc_unity") {
>     _target_dir_name = get_label_info(":$target_name", "dir")
>     output = "${root_out_dir}/lib.java${_target_dir_name}/${target_name}.jar"
>-    direct_deps_only = true
>+    direct_deps_only = false
>     use_interface_jars = false
>+    use_unprocessed_jars = false
>     requires_android = true
>     deps = [
>       ":webrtc_unity_java",
>@@ -713,8 +923,10 @@ if (!build_with_chromium) {
>     deps = [
>       "../p2p:libstunprober",
>       "../p2p:rtc_p2p",
>+      "../rtc_base:checks",
>       "../rtc_base:rtc_base",
>       "../rtc_base:rtc_base_approved",
>+      "../rtc_base:stringutils",
>       "../system_wrappers:field_trial_default",
>     ]
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/DEPS b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/DEPS
>index 4ade6af1054..15702200975 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/DEPS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/DEPS
>@@ -1,9 +1,13 @@
> include_rules = [
>   "+WebRTC",
>   "+api",
>+  "+common_video",
>   "+media",
>   "+modules/audio_device",
>   "+modules/video_capture",
>+  "+modules/audio_processing",
>   "+p2p",
>   "+pc",
>+  "+sdk/objc/Framework/Native/api",
>+  "+third_party/libyuv",
> ]
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/aarproject/.gitignore.rej b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/aarproject/.gitignore.rej
>deleted file mode 100644
>index bb40f75ab3e..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/aarproject/.gitignore.rej
>+++ /dev/null
>@@ -1,18 +0,0 @@
>-diff a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/aarproject/.gitignore b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/aarproject/.gitignore	(rejected hunks)
>-@@ -0,0 +1,16 @@
>-+# Default ignores by Android Studio
>-+*.iml
>-+.gradle
>-+# We want to specify our own SDK.
>-+# /local.properties
>-+/.idea/workspace.xml
>-+/.idea/libraries
>-+.DS_Store
>-+/build
>-+/captures
>-+.externalNativeBuild
>-+
>-+# Additional ignores
>-+/gradlew
>-+/gradlew.bat
>-+/gradle
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/aarproject/.idea/encodings.xml.rej b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/aarproject/.idea/encodings.xml.rej
>deleted file mode 100644
>index d1a368af363..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/aarproject/.idea/encodings.xml.rej
>+++ /dev/null
>@@ -1,9 +0,0 @@
>-diff a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/aarproject/.idea/encodings.xml b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/aarproject/.idea/encodings.xml	(rejected hunks)
>-@@ -0,0 +1,6 @@
>-+<?xml version="1.0" encoding="UTF-8"?>
>-+<project version="4">
>-+  <component name="Encoding">
>-+    <file url="PROJECT" charset="UTF-8" />
>-+  </component>
>-+</project>
>-\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/aarproject/.idea/gradle.xml.rej b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/aarproject/.idea/gradle.xml.rej
>deleted file mode 100644
>index c8ae0e7b7b2..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/aarproject/.idea/gradle.xml.rej
>+++ /dev/null
>@@ -1,21 +0,0 @@
>-diff a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/aarproject/.idea/gradle.xml b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/aarproject/.idea/gradle.xml	(rejected hunks)
>-@@ -0,0 +1,18 @@
>-+<?xml version="1.0" encoding="UTF-8"?>
>-+<project version="4">
>-+  <component name="GradleSettings">
>-+    <option name="linkedExternalProjectsSettings">
>-+      <GradleProjectSettings>
>-+        <option name="distributionType" value="DEFAULT_WRAPPED" />
>-+        <option name="externalProjectPath" value="$PROJECT_DIR$" />
>-+        <option name="modules">
>-+          <set>
>-+            <option value="$PROJECT_DIR$" />
>-+            <option value="$PROJECT_DIR$/app" />
>-+          </set>
>-+        </option>
>-+        <option name="resolveModulePerSourceSet" value="false" />
>-+      </GradleProjectSettings>
>-+    </option>
>-+  </component>
>-+</project>
>-\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/aarproject/.idea/misc.xml.rej b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/aarproject/.idea/misc.xml.rej
>deleted file mode 100644
>index 78ad0ae47ac..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/aarproject/.idea/misc.xml.rej
>+++ /dev/null
>@@ -1,36 +0,0 @@
>-diff a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/aarproject/.idea/misc.xml b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/aarproject/.idea/misc.xml	(rejected hunks)
>-@@ -0,0 +1,33 @@
>-+<?xml version="1.0" encoding="UTF-8"?>
>-+<project version="4">
>-+  <component name="NullableNotNullManager">
>-+    <option name="myDefaultNullable" value="android.support.annotation.Nullable" />
>-+    <option name="myDefaultNotNull" value="android.support.annotation.NonNull" />
>-+    <option name="myNullables">
>-+      <value>
>-+        <list size="4">
>-+          <item index="0" class="java.lang.String" itemvalue="org.jetbrains.annotations.Nullable" />
>-+          <item index="1" class="java.lang.String" itemvalue="javax.annotation.Nullable" />
>-+          <item index="2" class="java.lang.String" itemvalue="edu.umd.cs.findbugs.annotations.Nullable" />
>-+          <item index="3" class="java.lang.String" itemvalue="android.support.annotation.Nullable" />
>-+        </list>
>-+      </value>
>-+    </option>
>-+    <option name="myNotNulls">
>-+      <value>
>-+        <list size="4">
>-+          <item index="0" class="java.lang.String" itemvalue="org.jetbrains.annotations.NotNull" />
>-+          <item index="1" class="java.lang.String" itemvalue="javax.annotation.Nonnull" />
>-+          <item index="2" class="java.lang.String" itemvalue="edu.umd.cs.findbugs.annotations.NonNull" />
>-+          <item index="3" class="java.lang.String" itemvalue="android.support.annotation.NonNull" />
>-+        </list>
>-+      </value>
>-+    </option>
>-+  </component>
>-+  <component name="ProjectRootManager" version="2" languageLevel="JDK_1_8" default="true" project-jdk-name="1.8" project-jdk-type="JavaSDK">
>-+    <output url="file://$PROJECT_DIR$/build/classes" />
>-+  </component>
>-+  <component name="ProjectType">
>-+    <option name="id" value="Android" />
>-+  </component>
>-+</project>
>-\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/aarproject/.idea/modules.xml.rej b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/aarproject/.idea/modules.xml.rej
>deleted file mode 100644
>index 3ca43f07ea7..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/aarproject/.idea/modules.xml.rej
>+++ /dev/null
>@@ -1,12 +0,0 @@
>-diff a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/aarproject/.idea/modules.xml b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/aarproject/.idea/modules.xml	(rejected hunks)
>-@@ -0,0 +1,9 @@
>-+<?xml version="1.0" encoding="UTF-8"?>
>-+<project version="4">
>-+  <component name="ProjectModuleManager">
>-+    <modules>
>-+      <module fileurl="file://$PROJECT_DIR$/aarproject.iml" filepath="$PROJECT_DIR$/aarproject.iml" />
>-+      <module fileurl="file://$PROJECT_DIR$/app/app.iml" filepath="$PROJECT_DIR$/app/app.iml" />
>-+    </modules>
>-+  </component>
>-+</project>
>-\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/aarproject/.idea/runConfigurations.xml.rej b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/aarproject/.idea/runConfigurations.xml.rej
>deleted file mode 100644
>index 29c687fe1b6..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/aarproject/.idea/runConfigurations.xml.rej
>+++ /dev/null
>@@ -1,15 +0,0 @@
>-diff a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/aarproject/.idea/runConfigurations.xml b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/aarproject/.idea/runConfigurations.xml	(rejected hunks)
>-@@ -0,0 +1,12 @@
>-+<?xml version="1.0" encoding="UTF-8"?>
>-+<project version="4">
>-+  <component name="RunConfigurationProducerService">
>-+    <option name="ignoredProducers">
>-+      <set>
>-+        <option value="org.jetbrains.plugins.gradle.execution.test.runner.AllInPackageGradleConfigurationProducer" />
>-+        <option value="org.jetbrains.plugins.gradle.execution.test.runner.TestClassGradleConfigurationProducer" />
>-+        <option value="org.jetbrains.plugins.gradle.execution.test.runner.TestMethodGradleConfigurationProducer" />
>-+      </set>
>-+    </option>
>-+  </component>
>-+</project>
>-\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/aarproject/app/.gitignore.rej b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/aarproject/app/.gitignore.rej
>deleted file mode 100644
>index 9071603f0da..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/aarproject/app/.gitignore.rej
>+++ /dev/null
>@@ -1,3 +0,0 @@
>-diff a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/aarproject/app/.gitignore b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/aarproject/app/.gitignore	(rejected hunks)
>-@@ -0,0 +1 @@
>-+/build
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/aarproject/app/build.gradle b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/aarproject/app/build.gradle
>index 1dd15112d6a..ca9793fc3a0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/aarproject/app/build.gradle
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/aarproject/app/build.gradle
>@@ -1,8 +1,8 @@
> apply plugin: 'com.android.application'
> 
> android {
>-    compileSdkVersion 26
>-    buildToolsVersion "26.0.0"
>+    compileSdkVersion 27
>+    buildToolsVersion "27.0.1"
>     defaultConfig {
>         applicationId "org.appspot.apprtc"
>         minSdkVersion 16
>@@ -46,6 +46,7 @@ dependencies {
>     implementation fileTree(dir: 'libs', include: ['*.jar'])
>     implementation fileTree(dir: '../../androidapp/third_party/autobanh/lib', include: ['autobanh.jar'])
>     implementation 'com.android.support:appcompat-v7:26.1.0'
>+    implementation 'com.google.code.findbugs:jsr305:3.0.2'
>     implementation 'org.webrtc:google-webrtc:1.0.+'
>     testImplementation 'junit:junit:4.12'
>     androidTestImplementation 'com.android.support.test:runner:1.0.1'
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/res/values/strings.xml b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/res/values/strings.xml
>index 231475f587f..9006dc1cc7b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/res/values/strings.xml
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/res/values/strings.xml
>@@ -125,6 +125,11 @@
>     <string name="pref_aecdump_dlg">Enable diagnostic audio recordings.</string>
>     <string name="pref_aecdump_default">false</string>
> 
>+    <string name="pref_enable_save_input_audio_to_file_key">enable_key</string>
>+    <string name="pref_enable_save_input_audio_to_file_title">Save input audio to file.</string>
>+    <string name="pref_enable_save_input_audio_to_file_dlg">Save input audio to file.</string>
>+    <string name="pref_enable_save_input_audio_to_file_default">false</string>
>+
>     <string name="pref_opensles_key">opensles_preference</string>
>     <string name="pref_opensles_title">Use OpenSL ES for audio playback.</string>
>     <string name="pref_opensles_dlg">Use OpenSL ES for audio playback.</string>
>@@ -148,10 +153,6 @@
>     <string name="pref_disable_built_in_ns_default">false</string>
>     <string name="pref_built_in_ns_not_available">Hardware NS is not available</string>
> 
>-    <string name="pref_enable_level_control_key">enable_level_control_preference</string>
>-    <string name="pref_enable_level_control_title">Enable level control.</string>
>-    <string name="pref_enable_level_control_default">false</string>
>-
>     <string name="pref_disable_webrtc_agc_and_hpf_key">disable_webrtc_agc_and_hpf_preference</string>
>     <string name="pref_disable_webrtc_agc_and_hpf_title">Disable WebRTC AGC and HPF.</string>
>     <string name="pref_disable_webrtc_agc_default">false</string>
>@@ -213,4 +214,12 @@
>     <string name="pref_tracing_title">Debug performance tracing.</string>
>     <string name="pref_tracing_dlg">Debug performance tracing.</string>
>     <string name="pref_tracing_default" translatable="false">false</string>
>+
>+    <string name="pref_enable_rtceventlog_key">enable_rtceventlog_key</string>
>+    <string name="pref_enable_rtceventlog_title">Enable RtcEventLog.</string>
>+    <string name="pref_enable_rtceventlog_default">false</string>
>+
>+    <string name="pref_use_legacy_audio_device_key">use_legacy_audio_device_key</string>
>+    <string name="pref_use_legacy_audio_device_title">Use legacy audio device.</string>
>+    <string name="pref_use_legacy_audio_device_default">false</string>
> </resources>
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/res/xml/preferences.xml b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/res/xml/preferences.xml
>index c9fe750d9e2..ffe2daa8f57 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/res/xml/preferences.xml
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/res/xml/preferences.xml
>@@ -123,6 +123,12 @@
>             android:dialogTitle="@string/pref_aecdump_dlg"
>             android:defaultValue="@string/pref_aecdump_default" />
> 
>+        <CheckBoxPreference
>+            android:key="@string/pref_enable_save_input_audio_to_file_key"
>+            android:title="@string/pref_enable_save_input_audio_to_file_title"
>+            android:dialogTitle="@string/pref_enable_save_input_audio_to_file_dlg"
>+            android:defaultValue="@string/pref_enable_save_input_audio_to_file_default" />
>+
>         <CheckBoxPreference
>             android:key="@string/pref_opensles_key"
>             android:title="@string/pref_opensles_title"
>@@ -147,11 +153,6 @@
>             android:dialogTitle="@string/pref_disable_built_in_ns_dlg"
>             android:defaultValue="@string/pref_disable_built_in_ns_default" />
> 
>-        <CheckBoxPreference
>-            android:key="@string/pref_enable_level_control_key"
>-            android:title="@string/pref_enable_level_control_title"
>-            android:defaultValue="@string/pref_enable_level_control_default" />
>-
>         <CheckBoxPreference
>             android:key="@string/pref_disable_webrtc_agc_and_hpf_key"
>             android:title="@string/pref_disable_webrtc_agc_and_hpf_title"
>@@ -164,6 +165,11 @@
>             android:dialogTitle="@string/pref_speakerphone_dlg"
>             android:entries="@array/speakerphone"
>             android:entryValues="@array/speakerphoneValues" />
>+
>+        <CheckBoxPreference
>+            android:key="@string/pref_use_legacy_audio_device_key"
>+            android:title="@string/pref_use_legacy_audio_device_title"
>+            android:defaultValue="@string/pref_use_legacy_audio_device_default" />
>     </PreferenceCategory>
> 
>     <PreferenceCategory
>@@ -229,13 +235,18 @@
>             android:key="@string/pref_displayhud_key"
>             android:title="@string/pref_displayhud_title"
>             android:dialogTitle="@string/pref_displayhud_dlg"
>-           android:defaultValue="@string/pref_displayhud_default" />
>+            android:defaultValue="@string/pref_displayhud_default" />
> 
>         <CheckBoxPreference
>             android:key="@string/pref_tracing_key"
>             android:title="@string/pref_tracing_title"
>             android:dialogTitle="@string/pref_tracing_dlg"
>-           android:defaultValue="@string/pref_tracing_default" />
>+            android:defaultValue="@string/pref_tracing_default" />
>+
>+        <CheckBoxPreference
>+            android:key="@string/pref_enable_rtceventlog_key"
>+            android:title="@string/pref_enable_rtceventlog_title"
>+            android:defaultValue="@string/pref_enable_rtceventlog_default"/>
>     </PreferenceCategory>
> 
> </PreferenceScreen>
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/AppRTCAudioManager.java b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/AppRTCAudioManager.java
>index 5418ca0c311..aff198e88cd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/AppRTCAudioManager.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/AppRTCAudioManager.java
>@@ -24,6 +24,7 @@ import android.util.Log;
> import java.util.Collections;
> import java.util.HashSet;
> import java.util.Set;
>+import javax.annotation.Nullable;
> import org.appspot.apprtc.util.AppRTCUtils;
> import org.webrtc.ThreadUtils;
> 
>@@ -57,8 +58,10 @@ public class AppRTCAudioManager {
>   }
> 
>   private final Context apprtcContext;
>+  @Nullable
>   private AudioManager audioManager;
> 
>+  @Nullable
>   private AudioManagerEvents audioManagerEvents;
>   private AudioManagerState amState;
>   private int savedAudioMode = AudioManager.MODE_INVALID;
>@@ -90,6 +93,7 @@ public class AppRTCAudioManager {
>   // relative to the view screen of a device and can therefore be used to
>   // assist device switching (close to ear <=> use headset earpiece if
>   // available, far from ear <=> use speaker phone).
>+  @Nullable
>   private AppRTCProximitySensor proximitySensor = null;
> 
>   // Handles all tasks related to Bluetooth headset devices.
>@@ -103,6 +107,7 @@ public class AppRTCAudioManager {
>   private BroadcastReceiver wiredHeadsetReceiver;
> 
>   // Callback method for changes in audio focus.
>+  @Nullable
>   private AudioManager.OnAudioFocusChangeListener audioFocusChangeListener;
> 
>   /**
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/AppRTCBluetoothManager.java b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/AppRTCBluetoothManager.java
>index 00e6b04e418..6720e927c1c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/AppRTCBluetoothManager.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/AppRTCBluetoothManager.java
>@@ -24,6 +24,7 @@ import android.media.AudioManager;
> import android.os.Handler;
> import android.os.Looper;
> import android.os.Process;
>+import javax.annotation.Nullable;
> import android.util.Log;
> import java.util.List;
> import java.util.Set;
>@@ -64,14 +65,18 @@ public class AppRTCBluetoothManager {
> 
>   private final Context apprtcContext;
>   private final AppRTCAudioManager apprtcAudioManager;
>+  @Nullable
>   private final AudioManager audioManager;
>   private final Handler handler;
> 
>   int scoConnectionAttempts;
>   private State bluetoothState;
>   private final BluetoothProfile.ServiceListener bluetoothServiceListener;
>+  @Nullable
>   private BluetoothAdapter bluetoothAdapter;
>+  @Nullable
>   private BluetoothHeadset bluetoothHeadset;
>+  @Nullable
>   private BluetoothDevice bluetoothDevice;
>   private final BroadcastReceiver bluetoothHeadsetReceiver;
> 
>@@ -390,6 +395,7 @@ public class AppRTCBluetoothManager {
>   /**
>    * Stubs for test mocks.
>    */
>+  @Nullable
>   protected AudioManager getAudioManager(Context context) {
>     return (AudioManager) context.getSystemService(Context.AUDIO_SERVICE);
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/AppRTCProximitySensor.java b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/AppRTCProximitySensor.java
>index b63e8b867f5..ac03a8025ef 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/AppRTCProximitySensor.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/AppRTCProximitySensor.java
>@@ -16,6 +16,7 @@ import android.hardware.SensorEvent;
> import android.hardware.SensorEventListener;
> import android.hardware.SensorManager;
> import android.os.Build;
>+import javax.annotation.Nullable;
> import android.util.Log;
> import org.appspot.apprtc.util.AppRTCUtils;
> import org.webrtc.ThreadUtils;
>@@ -39,6 +40,7 @@ public class AppRTCProximitySensor implements SensorEventListener {
> 
>   private final Runnable onSensorStateListener;
>   private final SensorManager sensorManager;
>+  @Nullable
>   private Sensor proximitySensor = null;
>   private boolean lastStateReportIsNear = false;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/CallActivity.java b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/CallActivity.java
>index dabff97de5b..b66b91c39c6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/CallActivity.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/CallActivity.java
>@@ -36,6 +36,7 @@ import java.lang.RuntimeException;
> import java.util.ArrayList;
> import java.util.List;
> import java.util.Set;
>+import javax.annotation.Nullable;
> import org.appspot.apprtc.AppRTCAudioManager.AudioDevice;
> import org.appspot.apprtc.AppRTCAudioManager.AudioManagerEvents;
> import org.appspot.apprtc.AppRTCClient.RoomConnectionParameters;
>@@ -45,6 +46,7 @@ import org.appspot.apprtc.PeerConnectionClient.PeerConnectionParameters;
> import org.webrtc.Camera1Enumerator;
> import org.webrtc.Camera2Enumerator;
> import org.webrtc.CameraEnumerator;
>+import org.webrtc.EglBase;
> import org.webrtc.FileVideoCapturer;
> import org.webrtc.IceCandidate;
> import org.webrtc.Logging;
>@@ -57,7 +59,6 @@ import org.webrtc.SurfaceViewRenderer;
> import org.webrtc.VideoCapturer;
> import org.webrtc.VideoFileRenderer;
> import org.webrtc.VideoFrame;
>-import org.webrtc.VideoRenderer;
> import org.webrtc.VideoSink;
> 
> /**
>@@ -90,11 +91,12 @@ public class CallActivity extends Activity implements AppRTCClient.SignalingEven
>   public static final String EXTRA_NOAUDIOPROCESSING_ENABLED =
>       "org.appspot.apprtc.NOAUDIOPROCESSING";
>   public static final String EXTRA_AECDUMP_ENABLED = "org.appspot.apprtc.AECDUMP";
>+  public static final String EXTRA_SAVE_INPUT_AUDIO_TO_FILE_ENABLED =
>+      "org.appspot.apprtc.SAVE_INPUT_AUDIO_TO_FILE";
>   public static final String EXTRA_OPENSLES_ENABLED = "org.appspot.apprtc.OPENSLES";
>   public static final String EXTRA_DISABLE_BUILT_IN_AEC = "org.appspot.apprtc.DISABLE_BUILT_IN_AEC";
>   public static final String EXTRA_DISABLE_BUILT_IN_AGC = "org.appspot.apprtc.DISABLE_BUILT_IN_AGC";
>   public static final String EXTRA_DISABLE_BUILT_IN_NS = "org.appspot.apprtc.DISABLE_BUILT_IN_NS";
>-  public static final String EXTRA_ENABLE_LEVEL_CONTROL = "org.appspot.apprtc.ENABLE_LEVEL_CONTROL";
>   public static final String EXTRA_DISABLE_WEBRTC_AGC_AND_HPF =
>       "org.appspot.apprtc.DISABLE_WEBRTC_GAIN_CONTROL";
>   public static final String EXTRA_DISPLAY_HUD = "org.appspot.apprtc.DISPLAY_HUD";
>@@ -117,6 +119,9 @@ public class CallActivity extends Activity implements AppRTCClient.SignalingEven
>   public static final String EXTRA_PROTOCOL = "org.appspot.apprtc.PROTOCOL";
>   public static final String EXTRA_NEGOTIATED = "org.appspot.apprtc.NEGOTIATED";
>   public static final String EXTRA_ID = "org.appspot.apprtc.ID";
>+  public static final String EXTRA_ENABLE_RTCEVENTLOG = "org.appspot.apprtc.ENABLE_RTCEVENTLOG";
>+  public static final String EXTRA_USE_LEGACY_AUDIO_DEVICE =
>+      "org.appspot.apprtc.USE_LEGACY_AUDIO_DEVICE";
> 
>   private static final int CAPTURE_PERMISSION_REQUEST_CODE = 1;
> 
>@@ -127,25 +132,6 @@ public class CallActivity extends Activity implements AppRTCClient.SignalingEven
>   // Peer connection statistics callback period in ms.
>   private static final int STAT_CALLBACK_PERIOD = 1000;
> 
>-  private static class ProxyRenderer implements VideoRenderer.Callbacks {
>-    private VideoRenderer.Callbacks target;
>-
>-    @Override
>-    synchronized public void renderFrame(VideoRenderer.I420Frame frame) {
>-      if (target == null) {
>-        Logging.d(TAG, "Dropping frame in proxy because target is null.");
>-        VideoRenderer.renderFrameDone(frame);
>-        return;
>-      }
>-
>-      target.renderFrame(frame);
>-    }
>-
>-    synchronized public void setTarget(VideoRenderer.Callbacks target) {
>-      this.target = target;
>-    }
>-  }
>-
>   private static class ProxyVideoSink implements VideoSink {
>     private VideoSink target;
> 
>@@ -164,20 +150,28 @@ public class CallActivity extends Activity implements AppRTCClient.SignalingEven
>     }
>   }
> 
>-  private final ProxyRenderer remoteProxyRenderer = new ProxyRenderer();
>+  private final ProxyVideoSink remoteProxyRenderer = new ProxyVideoSink();
>   private final ProxyVideoSink localProxyVideoSink = new ProxyVideoSink();
>+  @Nullable
>   private PeerConnectionClient peerConnectionClient = null;
>+  @Nullable
>   private AppRTCClient appRtcClient;
>+  @Nullable
>   private SignalingParameters signalingParameters;
>+  @Nullable
>   private AppRTCAudioManager audioManager = null;
>+  @Nullable
>   private SurfaceViewRenderer pipRenderer;
>+  @Nullable
>   private SurfaceViewRenderer fullscreenRenderer;
>+  @Nullable
>   private VideoFileRenderer videoFileRenderer;
>-  private final List<VideoRenderer.Callbacks> remoteRenderers = new ArrayList<>();
>+  private final List<VideoSink> remoteSinks = new ArrayList<>();
>   private Toast logToast;
>   private boolean commandLineRun;
>   private boolean activityRunning;
>   private RoomConnectionParameters roomConnectionParameters;
>+  @Nullable
>   private PeerConnectionParameters peerConnectionParameters;
>   private boolean iceConnected;
>   private boolean isError;
>@@ -196,6 +190,9 @@ public class CallActivity extends Activity implements AppRTCClient.SignalingEven
>   private CpuMonitor cpuMonitor;
> 
>   @Override
>+  // TODO(bugs.webrtc.org/8580): LayoutParams.FLAG_TURN_SCREEN_ON and
>+  // LayoutParams.FLAG_SHOW_WHEN_LOCKED are deprecated.
>+  @SuppressWarnings("deprecation")
>   public void onCreate(Bundle savedInstanceState) {
>     super.onCreate(savedInstanceState);
>     Thread.setDefaultUncaughtExceptionHandler(new UnhandledExceptionHandler(this));
>@@ -234,15 +231,13 @@ public class CallActivity extends Activity implements AppRTCClient.SignalingEven
>     });
> 
>     fullscreenRenderer.setOnClickListener(listener);
>-    remoteRenderers.add(remoteProxyRenderer);
>+    remoteSinks.add(remoteProxyRenderer);
> 
>     final Intent intent = getIntent();
>-
>-    // Create peer connection client.
>-    peerConnectionClient = new PeerConnectionClient();
>+    final EglBase eglBase = EglBase.create();
> 
>     // Create video renderers.
>-    pipRenderer.init(peerConnectionClient.getRenderContext(), null);
>+    pipRenderer.init(eglBase.getEglBaseContext(), null);
>     pipRenderer.setScalingType(ScalingType.SCALE_ASPECT_FIT);
>     String saveRemoteVideoToFile = intent.getStringExtra(EXTRA_SAVE_REMOTE_VIDEO_TO_FILE);
> 
>@@ -251,20 +246,20 @@ public class CallActivity extends Activity implements AppRTCClient.SignalingEven
>       int videoOutWidth = intent.getIntExtra(EXTRA_SAVE_REMOTE_VIDEO_TO_FILE_WIDTH, 0);
>       int videoOutHeight = intent.getIntExtra(EXTRA_SAVE_REMOTE_VIDEO_TO_FILE_HEIGHT, 0);
>       try {
>-        videoFileRenderer = new VideoFileRenderer(saveRemoteVideoToFile, videoOutWidth,
>-            videoOutHeight, peerConnectionClient.getRenderContext());
>-        remoteRenderers.add(videoFileRenderer);
>+        videoFileRenderer = new VideoFileRenderer(
>+            saveRemoteVideoToFile, videoOutWidth, videoOutHeight, eglBase.getEglBaseContext());
>+        remoteSinks.add(videoFileRenderer);
>       } catch (IOException e) {
>         throw new RuntimeException(
>             "Failed to open video file for output: " + saveRemoteVideoToFile, e);
>       }
>     }
>-    fullscreenRenderer.init(peerConnectionClient.getRenderContext(), null);
>+    fullscreenRenderer.init(eglBase.getEglBaseContext(), null);
>     fullscreenRenderer.setScalingType(ScalingType.SCALE_ASPECT_FILL);
> 
>     pipRenderer.setZOrderMediaOverlay(true);
>     pipRenderer.setEnableHardwareScaler(true /* enabled */);
>-    fullscreenRenderer.setEnableHardwareScaler(true /* enabled */);
>+    fullscreenRenderer.setEnableHardwareScaler(false /* enabled */);
>     // Start with local feed in fullscreen and swap it to the pip when the call is connected.
>     setSwappedFeeds(true /* isSwappedFeeds */);
> 
>@@ -327,12 +322,14 @@ public class CallActivity extends Activity implements AppRTCClient.SignalingEven
>             intent.getIntExtra(EXTRA_AUDIO_BITRATE, 0), intent.getStringExtra(EXTRA_AUDIOCODEC),
>             intent.getBooleanExtra(EXTRA_NOAUDIOPROCESSING_ENABLED, false),
>             intent.getBooleanExtra(EXTRA_AECDUMP_ENABLED, false),
>+            intent.getBooleanExtra(EXTRA_SAVE_INPUT_AUDIO_TO_FILE_ENABLED, false),
>             intent.getBooleanExtra(EXTRA_OPENSLES_ENABLED, false),
>             intent.getBooleanExtra(EXTRA_DISABLE_BUILT_IN_AEC, false),
>             intent.getBooleanExtra(EXTRA_DISABLE_BUILT_IN_AGC, false),
>             intent.getBooleanExtra(EXTRA_DISABLE_BUILT_IN_NS, false),
>-            intent.getBooleanExtra(EXTRA_ENABLE_LEVEL_CONTROL, false),
>-            intent.getBooleanExtra(EXTRA_DISABLE_WEBRTC_AGC_AND_HPF, false), dataChannelParameters);
>+            intent.getBooleanExtra(EXTRA_DISABLE_WEBRTC_AGC_AND_HPF, false),
>+            intent.getBooleanExtra(EXTRA_ENABLE_RTCEVENTLOG, false),
>+            intent.getBooleanExtra(EXTRA_USE_LEGACY_AUDIO_DEVICE, false), dataChannelParameters);
>     commandLineRun = intent.getBooleanExtra(EXTRA_CMDLINE, false);
>     int runTimeMs = intent.getIntExtra(EXTRA_RUNTIME, 0);
> 
>@@ -352,7 +349,7 @@ public class CallActivity extends Activity implements AppRTCClient.SignalingEven
>         new RoomConnectionParameters(roomUri.toString(), roomId, loopback, urlParameters);
> 
>     // Create CPU monitor
>-    if (cpuMonitor.isSupported()) {
>+    if (CpuMonitor.isSupported()) {
>       cpuMonitor = new CpuMonitor(this);
>       hudFragment.setCpuMonitor(cpuMonitor);
>     }
>@@ -376,13 +373,14 @@ public class CallActivity extends Activity implements AppRTCClient.SignalingEven
>       }, runTimeMs);
>     }
> 
>+    // Create peer connection client.
>+    peerConnectionClient = new PeerConnectionClient(
>+        getApplicationContext(), eglBase, peerConnectionParameters, CallActivity.this);
>+    PeerConnectionFactory.Options options = new PeerConnectionFactory.Options();
>     if (loopback) {
>-      PeerConnectionFactory.Options options = new PeerConnectionFactory.Options();
>       options.networkIgnoreMask = 0;
>-      peerConnectionClient.setPeerConnectionFactoryOptions(options);
>     }
>-    peerConnectionClient.createPeerConnectionFactory(
>-        getApplicationContext(), peerConnectionParameters, CallActivity.this);
>+    peerConnectionClient.createPeerConnectionFactory(options);
> 
>     if (screencaptureEnabled) {
>       startScreenCapture();
>@@ -435,7 +433,7 @@ public class CallActivity extends Activity implements AppRTCClient.SignalingEven
>     return getIntent().getBooleanExtra(EXTRA_CAPTURETOTEXTURE_ENABLED, false);
>   }
> 
>-  private VideoCapturer createCameraCapturer(CameraEnumerator enumerator) {
>+  private @Nullable VideoCapturer createCameraCapturer(CameraEnumerator enumerator) {
>     final String[] deviceNames = enumerator.getDeviceNames();
> 
>     // First, try to find front facing camera
>@@ -468,7 +466,7 @@ public class CallActivity extends Activity implements AppRTCClient.SignalingEven
>   }
> 
>   @TargetApi(21)
>-  private VideoCapturer createScreenCapturer() {
>+  private @Nullable VideoCapturer createScreenCapturer() {
>     if (mediaProjectionPermissionResultCode != Activity.RESULT_OK) {
>       reportError("User didn't give permission to capture the screen.");
>       return null;
>@@ -705,7 +703,7 @@ public class CallActivity extends Activity implements AppRTCClient.SignalingEven
>     });
>   }
> 
>-  private VideoCapturer createVideoCapturer() {
>+  private @Nullable VideoCapturer createVideoCapturer() {
>     final VideoCapturer videoCapturer;
>     String videoFileAsCamera = getIntent().getStringExtra(EXTRA_VIDEO_FILE_AS_CAMERA);
>     if (videoFileAsCamera != null) {
>@@ -758,7 +756,7 @@ public class CallActivity extends Activity implements AppRTCClient.SignalingEven
>       videoCapturer = createVideoCapturer();
>     }
>     peerConnectionClient.createPeerConnection(
>-        localProxyVideoSink, remoteRenderers, videoCapturer, signalingParameters);
>+        localProxyVideoSink, remoteSinks, videoCapturer, signalingParameters);
> 
>     if (signalingParameters.initiator) {
>       logAndToast("Creating OFFER...");
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/ConnectActivity.java b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/ConnectActivity.java
>index d6f2ec3e99b..54cd0d7400d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/ConnectActivity.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/ConnectActivity.java
>@@ -235,7 +235,7 @@ public class ConnectActivity extends Activity {
>    */
>   private boolean sharedPrefGetBoolean(
>       int attributeId, String intentName, int defaultId, boolean useFromIntent) {
>-    boolean defaultValue = Boolean.valueOf(getString(defaultId));
>+    boolean defaultValue = Boolean.parseBoolean(getString(defaultId));
>     if (useFromIntent) {
>       return getIntent().getBooleanExtra(intentName, defaultValue);
>     } else {
>@@ -266,6 +266,7 @@ public class ConnectActivity extends Activity {
>     }
>   }
> 
>+  @SuppressWarnings("StringSplitter")
>   private void connectToRoom(String roomId, boolean commandLineRun, boolean loopback,
>       boolean useValuesFromIntent, int runTimeMs) {
>     ConnectActivity.commandLineRun = commandLineRun;
>@@ -314,10 +315,14 @@ public class ConnectActivity extends Activity {
>         CallActivity.EXTRA_NOAUDIOPROCESSING_ENABLED, R.string.pref_noaudioprocessing_default,
>         useValuesFromIntent);
> 
>-    // Check Disable Audio Processing flag.
>     boolean aecDump = sharedPrefGetBoolean(R.string.pref_aecdump_key,
>         CallActivity.EXTRA_AECDUMP_ENABLED, R.string.pref_aecdump_default, useValuesFromIntent);
> 
>+    boolean saveInputAudioToFile =
>+        sharedPrefGetBoolean(R.string.pref_enable_save_input_audio_to_file_key,
>+            CallActivity.EXTRA_SAVE_INPUT_AUDIO_TO_FILE_ENABLED,
>+            R.string.pref_enable_save_input_audio_to_file_default, useValuesFromIntent);
>+
>     // Check OpenSL ES enabled flag.
>     boolean useOpenSLES = sharedPrefGetBoolean(R.string.pref_opensles_key,
>         CallActivity.EXTRA_OPENSLES_ENABLED, R.string.pref_opensles_default, useValuesFromIntent);
>@@ -337,11 +342,6 @@ public class ConnectActivity extends Activity {
>         CallActivity.EXTRA_DISABLE_BUILT_IN_NS, R.string.pref_disable_built_in_ns_default,
>         useValuesFromIntent);
> 
>-    // Check Enable level control.
>-    boolean enableLevelControl = sharedPrefGetBoolean(R.string.pref_enable_level_control_key,
>-        CallActivity.EXTRA_ENABLE_LEVEL_CONTROL, R.string.pref_enable_level_control_key,
>-        useValuesFromIntent);
>-
>     // Check Disable gain control
>     boolean disableWebRtcAGCAndHPF = sharedPrefGetBoolean(
>         R.string.pref_disable_webrtc_agc_and_hpf_key, CallActivity.EXTRA_DISABLE_WEBRTC_AGC_AND_HPF,
>@@ -429,6 +429,15 @@ public class ConnectActivity extends Activity {
>     boolean tracing = sharedPrefGetBoolean(R.string.pref_tracing_key, CallActivity.EXTRA_TRACING,
>         R.string.pref_tracing_default, useValuesFromIntent);
> 
>+    // Check Enable RtcEventLog.
>+    boolean rtcEventLogEnabled = sharedPrefGetBoolean(R.string.pref_enable_rtceventlog_key,
>+        CallActivity.EXTRA_ENABLE_RTCEVENTLOG, R.string.pref_enable_rtceventlog_default,
>+        useValuesFromIntent);
>+
>+    boolean useLegacyAudioDevice = sharedPrefGetBoolean(R.string.pref_use_legacy_audio_device_key,
>+        CallActivity.EXTRA_USE_LEGACY_AUDIO_DEVICE, R.string.pref_use_legacy_audio_device_default,
>+        useValuesFromIntent);
>+
>     // Get datachannel options
>     boolean dataChannelEnabled = sharedPrefGetBoolean(R.string.pref_enable_datachannel_key,
>         CallActivity.EXTRA_DATA_CHANNEL_ENABLED, R.string.pref_enable_datachannel_default,
>@@ -470,18 +479,20 @@ public class ConnectActivity extends Activity {
>       intent.putExtra(CallActivity.EXTRA_FLEXFEC_ENABLED, flexfecEnabled);
>       intent.putExtra(CallActivity.EXTRA_NOAUDIOPROCESSING_ENABLED, noAudioProcessing);
>       intent.putExtra(CallActivity.EXTRA_AECDUMP_ENABLED, aecDump);
>+      intent.putExtra(CallActivity.EXTRA_SAVE_INPUT_AUDIO_TO_FILE_ENABLED, saveInputAudioToFile);
>       intent.putExtra(CallActivity.EXTRA_OPENSLES_ENABLED, useOpenSLES);
>       intent.putExtra(CallActivity.EXTRA_DISABLE_BUILT_IN_AEC, disableBuiltInAEC);
>       intent.putExtra(CallActivity.EXTRA_DISABLE_BUILT_IN_AGC, disableBuiltInAGC);
>       intent.putExtra(CallActivity.EXTRA_DISABLE_BUILT_IN_NS, disableBuiltInNS);
>-      intent.putExtra(CallActivity.EXTRA_ENABLE_LEVEL_CONTROL, enableLevelControl);
>       intent.putExtra(CallActivity.EXTRA_DISABLE_WEBRTC_AGC_AND_HPF, disableWebRtcAGCAndHPF);
>       intent.putExtra(CallActivity.EXTRA_AUDIO_BITRATE, audioStartBitrate);
>       intent.putExtra(CallActivity.EXTRA_AUDIOCODEC, audioCodec);
>       intent.putExtra(CallActivity.EXTRA_DISPLAY_HUD, displayHud);
>       intent.putExtra(CallActivity.EXTRA_TRACING, tracing);
>+      intent.putExtra(CallActivity.EXTRA_ENABLE_RTCEVENTLOG, rtcEventLogEnabled);
>       intent.putExtra(CallActivity.EXTRA_CMDLINE, commandLineRun);
>       intent.putExtra(CallActivity.EXTRA_RUNTIME, runTimeMs);
>+      intent.putExtra(CallActivity.EXTRA_USE_LEGACY_AUDIO_DEVICE, useLegacyAudioDevice);
> 
>       intent.putExtra(CallActivity.EXTRA_DATA_CHANNEL_ENABLED, dataChannelEnabled);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/CpuMonitor.java b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/CpuMonitor.java
>index 4a7ab9ecae6..8e2881c2d7d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/CpuMonitor.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/CpuMonitor.java
>@@ -31,6 +31,7 @@ import java.util.concurrent.Executors;
> import java.util.concurrent.Future;
> import java.util.concurrent.ScheduledExecutorService;
> import java.util.concurrent.TimeUnit;
>+import javax.annotation.Nullable;
> 
> /**
>  * Simple CPU monitor.  The caller creates a CpuMonitor object which can then
>@@ -91,6 +92,7 @@ class CpuMonitor {
>   // CPU frequency in percentage from maximum.
>   private final MovingAverage frequencyScale;
> 
>+  @Nullable
>   private ScheduledExecutorService executor;
>   private long lastStatLogTimeMs;
>   private long[] cpuFreqMax;
>@@ -101,6 +103,7 @@ class CpuMonitor {
>   private String[] maxPath;
>   private String[] curPath;
>   private double[] curFreqScales;
>+  @Nullable
>   private ProcStat lastProcStat;
> 
>   private static class ProcStat {
>@@ -484,7 +487,8 @@ class CpuMonitor {
>    * Read the current utilization of all CPUs using the cumulative first line
>    * of /proc/stat.
>    */
>-  private ProcStat readProcStat() {
>+  @SuppressWarnings("StringSplitter")
>+  private @Nullable ProcStat readProcStat() {
>     long userTime = 0;
>     long systemTime = 0;
>     long idleTime = 0;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/DirectRTCClient.java b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/DirectRTCClient.java
>index ae3a5228d63..9762e79aa03 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/DirectRTCClient.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/DirectRTCClient.java
>@@ -10,6 +10,7 @@
> 
> package org.appspot.apprtc;
> 
>+import javax.annotation.Nullable;
> import android.util.Log;
> 
> import org.json.JSONArray;
>@@ -53,6 +54,7 @@ public class DirectRTCClient implements AppRTCClient, TCPChannelClient.TCPChanne
> 
>   private final ExecutorService executor;
>   private final SignalingEvents events;
>+  @Nullable
>   private TCPChannelClient tcpClient;
>   private RoomConnectionParameters connectionParameters;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/PeerConnectionClient.java b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/PeerConnectionClient.java
>index fb11e4ae0bb..ae1f77b05ab 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/PeerConnectionClient.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/PeerConnectionClient.java
>@@ -13,23 +13,30 @@ package org.appspot.apprtc;
> import android.content.Context;
> import android.os.Environment;
> import android.os.ParcelFileDescriptor;
>+import android.preference.PreferenceManager;
> import android.util.Log;
> import java.io.File;
> import java.io.IOException;
> import java.nio.ByteBuffer;
> import java.nio.charset.Charset;
>+import java.text.DateFormat;
>+import java.text.SimpleDateFormat;
> import java.util.ArrayList;
> import java.util.Arrays;
> import java.util.Collections;
>+import java.util.Date;
> import java.util.Iterator;
> import java.util.List;
>+import java.util.Locale;
> import java.util.Timer;
> import java.util.TimerTask;
> import java.util.concurrent.ExecutorService;
> import java.util.concurrent.Executors;
> import java.util.regex.Matcher;
> import java.util.regex.Pattern;
>+import javax.annotation.Nullable;
> import org.appspot.apprtc.AppRTCClient.SignalingParameters;
>+import org.appspot.apprtc.RecordedAudioToFileController;
> import org.webrtc.AudioSource;
> import org.webrtc.AudioTrack;
> import org.webrtc.CameraVideoCapturer;
>@@ -41,25 +48,33 @@ import org.webrtc.IceCandidate;
> import org.webrtc.Logging;
> import org.webrtc.MediaConstraints;
> import org.webrtc.MediaStream;
>+import org.webrtc.MediaStreamTrack;
>+import org.webrtc.MediaStreamTrack.MediaType;
> import org.webrtc.PeerConnection;
> import org.webrtc.PeerConnection.IceConnectionState;
> import org.webrtc.PeerConnectionFactory;
> import org.webrtc.RtpParameters;
> import org.webrtc.RtpReceiver;
> import org.webrtc.RtpSender;
>+import org.webrtc.RtpTransceiver;
> import org.webrtc.SdpObserver;
> import org.webrtc.SessionDescription;
> import org.webrtc.SoftwareVideoDecoderFactory;
> import org.webrtc.SoftwareVideoEncoderFactory;
> import org.webrtc.StatsObserver;
> import org.webrtc.StatsReport;
>+import org.webrtc.SurfaceTextureHelper;
> import org.webrtc.VideoCapturer;
> import org.webrtc.VideoDecoderFactory;
> import org.webrtc.VideoEncoderFactory;
>-import org.webrtc.VideoRenderer;
> import org.webrtc.VideoSink;
> import org.webrtc.VideoSource;
> import org.webrtc.VideoTrack;
>+import org.webrtc.audio.AudioDeviceModule;
>+import org.webrtc.audio.JavaAudioDeviceModule;
>+import org.webrtc.audio.JavaAudioDeviceModule.AudioRecordErrorCallback;
>+import org.webrtc.audio.JavaAudioDeviceModule.AudioTrackErrorCallback;
>+import org.webrtc.audio.LegacyAudioDeviceModule;
> import org.webrtc.voiceengine.WebRtcAudioManager;
> import org.webrtc.voiceengine.WebRtcAudioRecord;
> import org.webrtc.voiceengine.WebRtcAudioRecord.AudioRecordStartErrorCode;
>@@ -91,23 +106,18 @@ public class PeerConnectionClient {
>   private static final String VIDEO_FLEXFEC_FIELDTRIAL =
>       "WebRTC-FlexFEC-03-Advertised/Enabled/WebRTC-FlexFEC-03/Enabled/";
>   private static final String VIDEO_VP8_INTEL_HW_ENCODER_FIELDTRIAL = "WebRTC-IntelVP8/Enabled/";
>-  private static final String VIDEO_H264_HIGH_PROFILE_FIELDTRIAL =
>-      "WebRTC-H264HighProfile/Enabled/";
>   private static final String DISABLE_WEBRTC_AGC_FIELDTRIAL =
>       "WebRTC-Audio-MinimizeResamplingOnMobile/Enabled/";
>-  private static final String VIDEO_FRAME_EMIT_FIELDTRIAL =
>-      PeerConnectionFactory.VIDEO_FRAME_EMIT_TRIAL + "/" + PeerConnectionFactory.TRIAL_ENABLED
>-      + "/";
>   private static final String AUDIO_CODEC_PARAM_BITRATE = "maxaveragebitrate";
>   private static final String AUDIO_ECHO_CANCELLATION_CONSTRAINT = "googEchoCancellation";
>   private static final String AUDIO_AUTO_GAIN_CONTROL_CONSTRAINT = "googAutoGainControl";
>   private static final String AUDIO_HIGH_PASS_FILTER_CONSTRAINT = "googHighpassFilter";
>   private static final String AUDIO_NOISE_SUPPRESSION_CONSTRAINT = "googNoiseSuppression";
>-  private static final String AUDIO_LEVEL_CONTROL_CONSTRAINT = "levelControl";
>   private static final String DTLS_SRTP_KEY_AGREEMENT_CONSTRAINT = "DtlsSrtpKeyAgreement";
>   private static final int HD_VIDEO_WIDTH = 1280;
>   private static final int HD_VIDEO_HEIGHT = 720;
>   private static final int BPS_IN_KBPS = 1000;
>+  private static final String RTCEVENTLOG_OUTPUT_DIR_NAME = "rtc_event_log";
> 
>   // Executor thread is started once in private ctor and is used for all
>   // peer connection API calls to ensure new peer connection factory is
>@@ -116,48 +126,64 @@ public class PeerConnectionClient {
> 
>   private final PCObserver pcObserver = new PCObserver();
>   private final SDPObserver sdpObserver = new SDPObserver();
>-
>+  private final Timer statsTimer = new Timer();
>   private final EglBase rootEglBase;
>+  private final Context appContext;
>+  private final PeerConnectionParameters peerConnectionParameters;
>+  private final PeerConnectionEvents events;
>+
>+  @Nullable
>   private PeerConnectionFactory factory;
>+  @Nullable
>   private PeerConnection peerConnection;
>-  PeerConnectionFactory.Options options = null;
>+  @Nullable
>   private AudioSource audioSource;
>-  private VideoSource videoSource;
>-  private boolean videoCallEnabled;
>+  @Nullable private SurfaceTextureHelper surfaceTextureHelper;
>+  @Nullable private VideoSource videoSource;
>   private boolean preferIsac;
>-  private String preferredVideoCodec;
>   private boolean videoCapturerStopped;
>   private boolean isError;
>-  private Timer statsTimer;
>+  @Nullable
>   private VideoSink localRender;
>-  private List<VideoRenderer.Callbacks> remoteRenders;
>+  @Nullable private List<VideoSink> remoteSinks;
>   private SignalingParameters signalingParameters;
>-  private MediaConstraints pcConstraints;
>   private int videoWidth;
>   private int videoHeight;
>   private int videoFps;
>   private MediaConstraints audioConstraints;
>   private MediaConstraints sdpMediaConstraints;
>-  private PeerConnectionParameters peerConnectionParameters;
>   // Queued remote ICE candidates are consumed only after both local and
>   // remote descriptions are set. Similarly local ICE candidates are sent to
>   // remote peer after both local and remote description are set.
>+  @Nullable
>   private List<IceCandidate> queuedRemoteCandidates;
>-  private PeerConnectionEvents events;
>   private boolean isInitiator;
>+  @Nullable
>   private SessionDescription localSdp; // either offer or answer SDP
>-  private MediaStream mediaStream;
>+  @Nullable
>   private VideoCapturer videoCapturer;
>   // enableVideo is set to true if video should be rendered and sent.
>-  private boolean renderVideo;
>+  private boolean renderVideo = true;
>+  @Nullable
>   private VideoTrack localVideoTrack;
>+  @Nullable
>   private VideoTrack remoteVideoTrack;
>+  @Nullable
>   private RtpSender localVideoSender;
>   // enableAudio is set to true if audio should be sent.
>-  private boolean enableAudio;
>+  private boolean enableAudio = true;
>+  @Nullable
>   private AudioTrack localAudioTrack;
>+  @Nullable
>   private DataChannel dataChannel;
>-  private boolean dataChannelEnabled;
>+  private final boolean dataChannelEnabled;
>+  // Enable RtcEventLog.
>+  @Nullable
>+  private RtcEventLog rtcEventLog;
>+  // Implements the WebRtcAudioRecordSamplesReadyCallback interface and writes
>+  // recorded audio samples to an output file.
>+  @Nullable
>+  private RecordedAudioToFileController saveRecordedAudioToFile = null;
> 
>   /**
>    * Peer connection parameters.
>@@ -199,33 +225,23 @@ public class PeerConnectionClient {
>     public final String audioCodec;
>     public final boolean noAudioProcessing;
>     public final boolean aecDump;
>+    public final boolean saveInputAudioToFile;
>     public final boolean useOpenSLES;
>     public final boolean disableBuiltInAEC;
>     public final boolean disableBuiltInAGC;
>     public final boolean disableBuiltInNS;
>-    public final boolean enableLevelControl;
>     public final boolean disableWebRtcAGCAndHPF;
>+    public final boolean enableRtcEventLog;
>+    public final boolean useLegacyAudioDevice;
>     private final DataChannelParameters dataChannelParameters;
> 
>     public PeerConnectionParameters(boolean videoCallEnabled, boolean loopback, boolean tracing,
>         int videoWidth, int videoHeight, int videoFps, int videoMaxBitrate, String videoCodec,
>         boolean videoCodecHwAcceleration, boolean videoFlexfecEnabled, int audioStartBitrate,
>-        String audioCodec, boolean noAudioProcessing, boolean aecDump, boolean useOpenSLES,
>-        boolean disableBuiltInAEC, boolean disableBuiltInAGC, boolean disableBuiltInNS,
>-        boolean enableLevelControl, boolean disableWebRtcAGCAndHPF) {
>-      this(videoCallEnabled, loopback, tracing, videoWidth, videoHeight, videoFps, videoMaxBitrate,
>-          videoCodec, videoCodecHwAcceleration, videoFlexfecEnabled, audioStartBitrate, audioCodec,
>-          noAudioProcessing, aecDump, useOpenSLES, disableBuiltInAEC, disableBuiltInAGC,
>-          disableBuiltInNS, enableLevelControl, disableWebRtcAGCAndHPF, null);
>-    }
>-
>-    public PeerConnectionParameters(boolean videoCallEnabled, boolean loopback, boolean tracing,
>-        int videoWidth, int videoHeight, int videoFps, int videoMaxBitrate, String videoCodec,
>-        boolean videoCodecHwAcceleration, boolean videoFlexfecEnabled, int audioStartBitrate,
>-        String audioCodec, boolean noAudioProcessing, boolean aecDump, boolean useOpenSLES,
>-        boolean disableBuiltInAEC, boolean disableBuiltInAGC, boolean disableBuiltInNS,
>-        boolean enableLevelControl, boolean disableWebRtcAGCAndHPF,
>-        DataChannelParameters dataChannelParameters) {
>+        String audioCodec, boolean noAudioProcessing, boolean aecDump, boolean saveInputAudioToFile,
>+        boolean useOpenSLES, boolean disableBuiltInAEC, boolean disableBuiltInAGC,
>+        boolean disableBuiltInNS, boolean disableWebRtcAGCAndHPF, boolean enableRtcEventLog,
>+        boolean useLegacyAudioDevice, DataChannelParameters dataChannelParameters) {
>       this.videoCallEnabled = videoCallEnabled;
>       this.loopback = loopback;
>       this.tracing = tracing;
>@@ -240,12 +256,14 @@ public class PeerConnectionClient {
>       this.audioCodec = audioCodec;
>       this.noAudioProcessing = noAudioProcessing;
>       this.aecDump = aecDump;
>+      this.saveInputAudioToFile = saveInputAudioToFile;
>       this.useOpenSLES = useOpenSLES;
>       this.disableBuiltInAEC = disableBuiltInAEC;
>       this.disableBuiltInAGC = disableBuiltInAGC;
>       this.disableBuiltInNS = disableBuiltInNS;
>-      this.enableLevelControl = enableLevelControl;
>       this.disableWebRtcAGCAndHPF = disableWebRtcAGCAndHPF;
>+      this.enableRtcEventLog = enableRtcEventLog;
>+      this.useLegacyAudioDevice = useLegacyAudioDevice;
>       this.dataChannelParameters = dataChannelParameters;
>     }
>   }
>@@ -297,141 +315,83 @@ public class PeerConnectionClient {
>     void onPeerConnectionError(final String description);
>   }
> 
>-  public PeerConnectionClient() {
>-    rootEglBase = EglBase.create();
>-  }
>-
>-  public void setPeerConnectionFactoryOptions(PeerConnectionFactory.Options options) {
>-    this.options = options;
>-  }
>-
>-  public void createPeerConnectionFactory(final Context context,
>-      final PeerConnectionParameters peerConnectionParameters, final PeerConnectionEvents events) {
>-    this.peerConnectionParameters = peerConnectionParameters;
>+  /**
>+   * Create a PeerConnectionClient with the specified parameters. PeerConnectionClient takes
>+   * ownership of |eglBase|.
>+   */
>+  public PeerConnectionClient(Context appContext, EglBase eglBase,
>+      PeerConnectionParameters peerConnectionParameters, PeerConnectionEvents events) {
>+    this.rootEglBase = eglBase;
>+    this.appContext = appContext;
>     this.events = events;
>-    videoCallEnabled = peerConnectionParameters.videoCallEnabled;
>-    dataChannelEnabled = peerConnectionParameters.dataChannelParameters != null;
>-    // Reset variables to initial states.
>-    factory = null;
>-    peerConnection = null;
>-    preferIsac = false;
>-    videoCapturerStopped = false;
>-    isError = false;
>-    queuedRemoteCandidates = null;
>-    localSdp = null; // either offer or answer SDP
>-    mediaStream = null;
>-    videoCapturer = null;
>-    renderVideo = true;
>-    localVideoTrack = null;
>-    remoteVideoTrack = null;
>-    localVideoSender = null;
>-    enableAudio = true;
>-    localAudioTrack = null;
>-    statsTimer = new Timer();
>-
>-    executor.execute(new Runnable() {
>-      @Override
>-      public void run() {
>-        createPeerConnectionFactoryInternal(context);
>-      }
>+    this.peerConnectionParameters = peerConnectionParameters;
>+    this.dataChannelEnabled = peerConnectionParameters.dataChannelParameters != null;
>+
>+    Log.d(TAG, "Preferred video codec: " + getSdpVideoCodecName(peerConnectionParameters));
>+
>+    final String fieldTrials = getFieldTrials(peerConnectionParameters);
>+    executor.execute(() -> {
>+      Log.d(TAG, "Initialize WebRTC. Field trials: " + fieldTrials);
>+      PeerConnectionFactory.initialize(
>+          PeerConnectionFactory.InitializationOptions.builder(appContext)
>+              .setFieldTrials(fieldTrials)
>+              .setEnableInternalTracer(true)
>+              .createInitializationOptions());
>     });
>   }
> 
>-  public void createPeerConnection(final VideoSink localRender,
>-      final VideoRenderer.Callbacks remoteRender, final VideoCapturer videoCapturer,
>-      final SignalingParameters signalingParameters) {
>+  /**
>+   * This function should only be called once.
>+   */
>+  public void createPeerConnectionFactory(PeerConnectionFactory.Options options) {
>+    if (factory != null) {
>+      throw new IllegalStateException("PeerConnectionFactory has already been constructed");
>+    }
>+    executor.execute(() -> createPeerConnectionFactoryInternal(options));
>+  }
>+
>+  public void createPeerConnection(final VideoSink localRender, final VideoSink remoteSink,
>+      final VideoCapturer videoCapturer, final SignalingParameters signalingParameters) {
>+    if (peerConnectionParameters.videoCallEnabled && videoCapturer == null) {
>+      Log.w(TAG, "Video call enabled but no video capturer provided.");
>+    }
>     createPeerConnection(
>-        localRender, Collections.singletonList(remoteRender), videoCapturer, signalingParameters);
>+        localRender, Collections.singletonList(remoteSink), videoCapturer, signalingParameters);
>   }
> 
>-  public void createPeerConnection(final VideoSink localRender,
>-      final List<VideoRenderer.Callbacks> remoteRenders, final VideoCapturer videoCapturer,
>-      final SignalingParameters signalingParameters) {
>+  public void createPeerConnection(final VideoSink localRender, final List<VideoSink> remoteSinks,
>+      final VideoCapturer videoCapturer, final SignalingParameters signalingParameters) {
>     if (peerConnectionParameters == null) {
>       Log.e(TAG, "Creating peer connection without initializing factory.");
>       return;
>     }
>     this.localRender = localRender;
>-    this.remoteRenders = remoteRenders;
>+    this.remoteSinks = remoteSinks;
>     this.videoCapturer = videoCapturer;
>     this.signalingParameters = signalingParameters;
>-    executor.execute(new Runnable() {
>-      @Override
>-      public void run() {
>-        try {
>-          createMediaConstraintsInternal();
>-          createPeerConnectionInternal();
>-        } catch (Exception e) {
>-          reportError("Failed to create peer connection: " + e.getMessage());
>-          throw e;
>-        }
>+    executor.execute(() -> {
>+      try {
>+        createMediaConstraintsInternal();
>+        createPeerConnectionInternal();
>+        maybeCreateAndStartRtcEventLog();
>+      } catch (Exception e) {
>+        reportError("Failed to create peer connection: " + e.getMessage());
>+        throw e;
>       }
>     });
>   }
> 
>   public void close() {
>-    executor.execute(new Runnable() {
>-      @Override
>-      public void run() {
>-        closeInternal();
>-      }
>-    });
>+    executor.execute(this ::closeInternal);
>   }
> 
>-  public boolean isVideoCallEnabled() {
>-    return videoCallEnabled;
>+  private boolean isVideoCallEnabled() {
>+    return peerConnectionParameters.videoCallEnabled && videoCapturer != null;
>   }
> 
>-  private void createPeerConnectionFactoryInternal(Context context) {
>+  private void createPeerConnectionFactoryInternal(PeerConnectionFactory.Options options) {
>     isError = false;
> 
>-    // Initialize field trials.
>-    String fieldTrials = "";
>-    if (peerConnectionParameters.videoFlexfecEnabled) {
>-      fieldTrials += VIDEO_FLEXFEC_FIELDTRIAL;
>-      Log.d(TAG, "Enable FlexFEC field trial.");
>-    }
>-    fieldTrials += VIDEO_VP8_INTEL_HW_ENCODER_FIELDTRIAL;
>-    if (peerConnectionParameters.disableWebRtcAGCAndHPF) {
>-      fieldTrials += DISABLE_WEBRTC_AGC_FIELDTRIAL;
>-      Log.d(TAG, "Disable WebRTC AGC field trial.");
>-    }
>-    fieldTrials += VIDEO_FRAME_EMIT_FIELDTRIAL;
>-
>-    // Check preferred video codec.
>-    preferredVideoCodec = VIDEO_CODEC_VP8;
>-    if (videoCallEnabled && peerConnectionParameters.videoCodec != null) {
>-      switch (peerConnectionParameters.videoCodec) {
>-        case VIDEO_CODEC_VP8:
>-          preferredVideoCodec = VIDEO_CODEC_VP8;
>-          break;
>-        case VIDEO_CODEC_VP9:
>-          preferredVideoCodec = VIDEO_CODEC_VP9;
>-          break;
>-        case VIDEO_CODEC_H264_BASELINE:
>-          preferredVideoCodec = VIDEO_CODEC_H264;
>-          break;
>-        case VIDEO_CODEC_H264_HIGH:
>-          // TODO(magjed): Strip High from SDP when selecting Baseline instead of using field trial.
>-          fieldTrials += VIDEO_H264_HIGH_PROFILE_FIELDTRIAL;
>-          preferredVideoCodec = VIDEO_CODEC_H264;
>-          break;
>-        default:
>-          preferredVideoCodec = VIDEO_CODEC_VP8;
>-      }
>-    }
>-    Log.d(TAG, "Preferred video codec: " + preferredVideoCodec);
>-
>-    // Initialize WebRTC
>-    Log.d(TAG,
>-        "Initialize WebRTC. Field trials: " + fieldTrials + " Enable video HW acceleration: "
>-            + peerConnectionParameters.videoCodecHwAcceleration);
>-    PeerConnectionFactory.initialize(
>-        PeerConnectionFactory.InitializationOptions.builder(context)
>-            .setFieldTrials(fieldTrials)
>-            .setEnableVideoHwAcceleration(peerConnectionParameters.videoCodecHwAcceleration)
>-            .setEnableInternalTracer(true)
>-            .createInitializationOptions());
>     if (peerConnectionParameters.tracing) {
>       PeerConnectionFactory.startInternalTracingCapture(
>           Environment.getExternalStorageDirectory().getAbsolutePath() + File.separator
>@@ -442,6 +402,55 @@ public class PeerConnectionClient {
>     preferIsac = peerConnectionParameters.audioCodec != null
>         && peerConnectionParameters.audioCodec.equals(AUDIO_CODEC_ISAC);
> 
>+    // It is possible to save a copy in raw PCM format on a file by checking
>+    // the "Save input audio to file" checkbox in the Settings UI. A callback
>+    // interface is set when this flag is enabled. As a result, a copy of recorded
>+    // audio samples are provided to this client directly from the native audio
>+    // layer in Java.
>+    if (peerConnectionParameters.saveInputAudioToFile) {
>+      if (!peerConnectionParameters.useOpenSLES) {
>+        Log.d(TAG, "Enable recording of microphone input audio to file");
>+        saveRecordedAudioToFile = new RecordedAudioToFileController(executor);
>+      } else {
>+        // TODO(henrika): ensure that the UI reflects that if OpenSL ES is selected,
>+        // then the "Save inut audio to file" option shall be grayed out.
>+        Log.e(TAG, "Recording of input audio is not supported for OpenSL ES");
>+      }
>+    }
>+
>+    final AudioDeviceModule adm = peerConnectionParameters.useLegacyAudioDevice
>+        ? createLegacyAudioDevice()
>+        : createJavaAudioDevice();
>+
>+    // Create peer connection factory.
>+    if (options != null) {
>+      Log.d(TAG, "Factory networkIgnoreMask option: " + options.networkIgnoreMask);
>+    }
>+    final boolean enableH264HighProfile =
>+        VIDEO_CODEC_H264_HIGH.equals(peerConnectionParameters.videoCodec);
>+    final VideoEncoderFactory encoderFactory;
>+    final VideoDecoderFactory decoderFactory;
>+
>+    if (peerConnectionParameters.videoCodecHwAcceleration) {
>+      encoderFactory = new DefaultVideoEncoderFactory(
>+          rootEglBase.getEglBaseContext(), true /* enableIntelVp8Encoder */, enableH264HighProfile);
>+      decoderFactory = new DefaultVideoDecoderFactory(rootEglBase.getEglBaseContext());
>+    } else {
>+      encoderFactory = new SoftwareVideoEncoderFactory();
>+      decoderFactory = new SoftwareVideoDecoderFactory();
>+    }
>+
>+    factory = PeerConnectionFactory.builder()
>+                  .setOptions(options)
>+                  .setAudioDeviceModule(adm)
>+                  .setVideoEncoderFactory(encoderFactory)
>+                  .setVideoDecoderFactory(decoderFactory)
>+                  .createPeerConnectionFactory();
>+    Log.d(TAG, "Peer connection factory created.");
>+    adm.release();
>+  }
>+
>+  AudioDeviceModule createLegacyAudioDevice() {
>     // Enable/disable OpenSL ES playback.
>     if (!peerConnectionParameters.useOpenSLES) {
>       Log.d(TAG, "Disable OpenSL ES audio even if device supports it");
>@@ -459,14 +468,6 @@ public class PeerConnectionClient {
>       WebRtcAudioUtils.setWebRtcBasedAcousticEchoCanceler(false);
>     }
> 
>-    if (peerConnectionParameters.disableBuiltInAGC) {
>-      Log.d(TAG, "Disable built-in AGC even if device supports it");
>-      WebRtcAudioUtils.setWebRtcBasedAutomaticGainControl(true);
>-    } else {
>-      Log.d(TAG, "Enable built-in AGC if device supports it");
>-      WebRtcAudioUtils.setWebRtcBasedAutomaticGainControl(false);
>-    }
>-
>     if (peerConnectionParameters.disableBuiltInNS) {
>       Log.d(TAG, "Disable built-in NS even if device supports it");
>       WebRtcAudioUtils.setWebRtcBasedNoiseSuppressor(true);
>@@ -475,6 +476,8 @@ public class PeerConnectionClient {
>       WebRtcAudioUtils.setWebRtcBasedNoiseSuppressor(false);
>     }
> 
>+    WebRtcAudioRecord.setOnAudioSamplesReady(saveRecordedAudioToFile);
>+
>     // Set audio record error callbacks.
>     WebRtcAudioRecord.setErrorCallback(new WebRtcAudioRecordErrorCallback() {
>       @Override
>@@ -518,47 +521,71 @@ public class PeerConnectionClient {
>       }
>     });
> 
>-    // Create peer connection factory.
>-    if (options != null) {
>-      Log.d(TAG, "Factory networkIgnoreMask option: " + options.networkIgnoreMask);
>-    }
>-    final boolean enableH264HighProfile =
>-        VIDEO_CODEC_H264_HIGH.equals(peerConnectionParameters.videoCodec);
>-    final VideoEncoderFactory encoderFactory;
>-    final VideoDecoderFactory decoderFactory;
>+    return new LegacyAudioDeviceModule();
>+  }
> 
>-    if (peerConnectionParameters.videoCodecHwAcceleration) {
>-      encoderFactory = new DefaultVideoEncoderFactory(
>-          rootEglBase.getEglBaseContext(), true /* enableIntelVp8Encoder */, enableH264HighProfile);
>-      decoderFactory = new DefaultVideoDecoderFactory(rootEglBase.getEglBaseContext());
>-    } else {
>-      encoderFactory = new SoftwareVideoEncoderFactory();
>-      decoderFactory = new SoftwareVideoDecoderFactory();
>+  AudioDeviceModule createJavaAudioDevice() {
>+    // Enable/disable OpenSL ES playback.
>+    if (!peerConnectionParameters.useOpenSLES) {
>+      Log.w(TAG, "External OpenSLES ADM not implemented yet.");
>+      // TODO(magjed): Add support for external OpenSLES ADM.
>     }
> 
>-    factory = new PeerConnectionFactory(options, encoderFactory, decoderFactory);
>-    Log.d(TAG, "Peer connection factory created.");
>+    // Set audio record error callbacks.
>+    AudioRecordErrorCallback audioRecordErrorCallback = new AudioRecordErrorCallback() {
>+      @Override
>+      public void onWebRtcAudioRecordInitError(String errorMessage) {
>+        Log.e(TAG, "onWebRtcAudioRecordInitError: " + errorMessage);
>+        reportError(errorMessage);
>+      }
>+
>+      @Override
>+      public void onWebRtcAudioRecordStartError(
>+          JavaAudioDeviceModule.AudioRecordStartErrorCode errorCode, String errorMessage) {
>+        Log.e(TAG, "onWebRtcAudioRecordStartError: " + errorCode + ". " + errorMessage);
>+        reportError(errorMessage);
>+      }
>+
>+      @Override
>+      public void onWebRtcAudioRecordError(String errorMessage) {
>+        Log.e(TAG, "onWebRtcAudioRecordError: " + errorMessage);
>+        reportError(errorMessage);
>+      }
>+    };
>+
>+    AudioTrackErrorCallback audioTrackErrorCallback = new AudioTrackErrorCallback() {
>+      @Override
>+      public void onWebRtcAudioTrackInitError(String errorMessage) {
>+        Log.e(TAG, "onWebRtcAudioTrackInitError: " + errorMessage);
>+        reportError(errorMessage);
>+      }
>+
>+      @Override
>+      public void onWebRtcAudioTrackStartError(
>+          JavaAudioDeviceModule.AudioTrackStartErrorCode errorCode, String errorMessage) {
>+        Log.e(TAG, "onWebRtcAudioTrackStartError: " + errorCode + ". " + errorMessage);
>+        reportError(errorMessage);
>+      }
>+
>+      @Override
>+      public void onWebRtcAudioTrackError(String errorMessage) {
>+        Log.e(TAG, "onWebRtcAudioTrackError: " + errorMessage);
>+        reportError(errorMessage);
>+      }
>+    };
>+
>+    return JavaAudioDeviceModule.builder(appContext)
>+        .setSamplesReadyCallback(saveRecordedAudioToFile)
>+        .setUseHardwareAcousticEchoCanceler(!peerConnectionParameters.disableBuiltInAEC)
>+        .setUseHardwareNoiseSuppressor(!peerConnectionParameters.disableBuiltInNS)
>+        .setAudioRecordErrorCallback(audioRecordErrorCallback)
>+        .setAudioTrackErrorCallback(audioTrackErrorCallback)
>+        .createAudioDeviceModule();
>   }
> 
>   private void createMediaConstraintsInternal() {
>-    // Create peer connection constraints.
>-    pcConstraints = new MediaConstraints();
>-    // Enable DTLS for normal calls and disable for loopback calls.
>-    if (peerConnectionParameters.loopback) {
>-      pcConstraints.optional.add(
>-          new MediaConstraints.KeyValuePair(DTLS_SRTP_KEY_AGREEMENT_CONSTRAINT, "false"));
>-    } else {
>-      pcConstraints.optional.add(
>-          new MediaConstraints.KeyValuePair(DTLS_SRTP_KEY_AGREEMENT_CONSTRAINT, "true"));
>-    }
>-
>-    // Check if there is a camera on device and disable video call if not.
>-    if (videoCapturer == null) {
>-      Log.w(TAG, "No camera on device. Switch to audio only call.");
>-      videoCallEnabled = false;
>-    }
>     // Create video constraints if video call is enabled.
>-    if (videoCallEnabled) {
>+    if (isVideoCallEnabled()) {
>       videoWidth = peerConnectionParameters.videoWidth;
>       videoHeight = peerConnectionParameters.videoHeight;
>       videoFps = peerConnectionParameters.videoFps;
>@@ -590,22 +617,12 @@ public class PeerConnectionClient {
>       audioConstraints.mandatory.add(
>           new MediaConstraints.KeyValuePair(AUDIO_NOISE_SUPPRESSION_CONSTRAINT, "false"));
>     }
>-    if (peerConnectionParameters.enableLevelControl) {
>-      Log.d(TAG, "Enabling level control.");
>-      audioConstraints.mandatory.add(
>-          new MediaConstraints.KeyValuePair(AUDIO_LEVEL_CONTROL_CONSTRAINT, "true"));
>-    }
>     // Create SDP constraints.
>     sdpMediaConstraints = new MediaConstraints();
>     sdpMediaConstraints.mandatory.add(
>         new MediaConstraints.KeyValuePair("OfferToReceiveAudio", "true"));
>-    if (videoCallEnabled || peerConnectionParameters.loopback) {
>-      sdpMediaConstraints.mandatory.add(
>-          new MediaConstraints.KeyValuePair("OfferToReceiveVideo", "true"));
>-    } else {
>-      sdpMediaConstraints.mandatory.add(
>-          new MediaConstraints.KeyValuePair("OfferToReceiveVideo", "false"));
>-    }
>+    sdpMediaConstraints.mandatory.add(new MediaConstraints.KeyValuePair(
>+        "OfferToReceiveVideo", Boolean.toString(isVideoCallEnabled())));
>   }
> 
>   private void createPeerConnectionInternal() {
>@@ -615,14 +632,8 @@ public class PeerConnectionClient {
>     }
>     Log.d(TAG, "Create peer connection.");
> 
>-    Log.d(TAG, "PCConstraints: " + pcConstraints.toString());
>     queuedRemoteCandidates = new ArrayList<>();
> 
>-    if (videoCallEnabled) {
>-      factory.setVideoHwAccelerationOptions(
>-          rootEglBase.getEglBaseContext(), rootEglBase.getEglBaseContext());
>-    }
>-
>     PeerConnection.RTCConfiguration rtcConfig =
>         new PeerConnection.RTCConfiguration(signalingParameters.iceServers);
>     // TCP candidates are only useful when connecting to a server that supports
>@@ -633,8 +644,11 @@ public class PeerConnectionClient {
>     rtcConfig.continualGatheringPolicy = PeerConnection.ContinualGatheringPolicy.GATHER_CONTINUALLY;
>     // Use ECDSA encryption.
>     rtcConfig.keyType = PeerConnection.KeyType.ECDSA;
>+    // Enable DTLS for normal calls and disable for loopback calls.
>+    rtcConfig.enableDtlsSrtp = !peerConnectionParameters.loopback;
>+    rtcConfig.sdpSemantics = PeerConnection.SdpSemantics.UNIFIED_PLAN;
> 
>-    peerConnection = factory.createPeerConnection(rtcConfig, pcConstraints, pcObserver);
>+    peerConnection = factory.createPeerConnection(rtcConfig, pcObserver);
> 
>     if (dataChannelEnabled) {
>       DataChannel.Init init = new DataChannel.Init();
>@@ -652,14 +666,19 @@ public class PeerConnectionClient {
>     // NOTE: this _must_ happen while |factory| is alive!
>     Logging.enableLogToDebugOutput(Logging.Severity.LS_INFO);
> 
>-    mediaStream = factory.createLocalMediaStream("ARDAMS");
>-    if (videoCallEnabled) {
>-      mediaStream.addTrack(createVideoTrack(videoCapturer));
>+    List<String> mediaStreamLabels = Collections.singletonList("ARDAMS");
>+    if (isVideoCallEnabled()) {
>+      peerConnection.addTrack(createVideoTrack(videoCapturer), mediaStreamLabels);
>+      // We can add the renderers right away because we don't need to wait for an
>+      // answer to get the remote track.
>+      remoteVideoTrack = getRemoteVideoTrack();
>+      remoteVideoTrack.setEnabled(renderVideo);
>+      for (VideoSink remoteSink : remoteSinks) {
>+        remoteVideoTrack.addSink(remoteSink);
>+      }
>     }
>-
>-    mediaStream.addTrack(createAudioTrack());
>-    peerConnection.addStream(mediaStream);
>-    if (videoCallEnabled) {
>+    peerConnection.addTrack(createAudioTrack(), mediaStreamLabels);
>+    if (isVideoCallEnabled()) {
>       findVideoSender();
>     }
> 
>@@ -670,15 +689,40 @@ public class PeerConnectionClient {
>                                           + File.separator + "Download/audio.aecdump"),
>                 ParcelFileDescriptor.MODE_READ_WRITE | ParcelFileDescriptor.MODE_CREATE
>                     | ParcelFileDescriptor.MODE_TRUNCATE);
>-        factory.startAecDump(aecDumpFileDescriptor.getFd(), -1);
>+        factory.startAecDump(aecDumpFileDescriptor.detachFd(), -1);
>       } catch (IOException e) {
>         Log.e(TAG, "Can not open aecdump file", e);
>       }
>     }
> 
>+    if (saveRecordedAudioToFile != null) {
>+      if (saveRecordedAudioToFile.start()) {
>+        Log.d(TAG, "Recording input audio to file is activated");
>+      }
>+    }
>     Log.d(TAG, "Peer connection created.");
>   }
> 
>+  private File createRtcEventLogOutputFile() {
>+    DateFormat dateFormat = new SimpleDateFormat("yyyyMMdd_hhmm_ss", Locale.getDefault());
>+    Date date = new Date();
>+    final String outputFileName = "event_log_" + dateFormat.format(date) + ".log";
>+    return new File(
>+        appContext.getDir(RTCEVENTLOG_OUTPUT_DIR_NAME, Context.MODE_PRIVATE), outputFileName);
>+  }
>+
>+  private void maybeCreateAndStartRtcEventLog() {
>+    if (appContext == null || peerConnection == null) {
>+      return;
>+    }
>+    if (!peerConnectionParameters.enableRtcEventLog) {
>+      Log.d(TAG, "RtcEventLog is disabled.");
>+      return;
>+    }
>+    rtcEventLog = new RtcEventLog(peerConnection);
>+    rtcEventLog.start(createRtcEventLogOutputFile());
>+  }
>+
>   private void closeInternal() {
>     if (factory != null && peerConnectionParameters.aecDump) {
>       factory.stopAecDump();
>@@ -689,6 +733,11 @@ public class PeerConnectionClient {
>       dataChannel.dispose();
>       dataChannel = null;
>     }
>+    if (rtcEventLog != null) {
>+      // RtcEventLog should stop before the peer connection is disposed.
>+      rtcEventLog.stop();
>+      rtcEventLog = null;
>+    }
>     if (peerConnection != null) {
>       peerConnection.dispose();
>       peerConnection = null;
>@@ -714,28 +763,31 @@ public class PeerConnectionClient {
>       videoSource.dispose();
>       videoSource = null;
>     }
>+    if (surfaceTextureHelper != null) {
>+      surfaceTextureHelper.dispose();
>+      surfaceTextureHelper = null;
>+    }
>+    if (saveRecordedAudioToFile != null) {
>+      Log.d(TAG, "Closing audio file for recorded input audio.");
>+      saveRecordedAudioToFile.stop();
>+      saveRecordedAudioToFile = null;
>+    }
>     localRender = null;
>-    remoteRenders = null;
>+    remoteSinks = null;
>     Log.d(TAG, "Closing peer connection factory.");
>     if (factory != null) {
>       factory.dispose();
>       factory = null;
>     }
>-    options = null;
>     rootEglBase.release();
>     Log.d(TAG, "Closing peer connection done.");
>     events.onPeerConnectionClosed();
>     PeerConnectionFactory.stopInternalTracingCapture();
>     PeerConnectionFactory.shutdownInternalTracer();
>-    events = null;
>   }
> 
>   public boolean isHDVideo() {
>-    return videoCallEnabled && videoWidth * videoHeight >= 1280 * 720;
>-  }
>-
>-  public EglBase.Context getRenderContext() {
>-    return rootEglBase.getEglBaseContext();
>+    return isVideoCallEnabled() && videoWidth * videoHeight >= 1280 * 720;
>   }
> 
>   @SuppressWarnings("deprecation") // TODO(sakal): getStats is deprecated.
>@@ -760,12 +812,7 @@ public class PeerConnectionClient {
>         statsTimer.schedule(new TimerTask() {
>           @Override
>           public void run() {
>-            executor.execute(new Runnable() {
>-              @Override
>-              public void run() {
>-                getStats();
>-              }
>-            });
>+            executor.execute(() -> getStats());
>           }
>         }, 0, periodMs);
>       } catch (Exception e) {
>@@ -777,186 +824,155 @@ public class PeerConnectionClient {
>   }
> 
>   public void setAudioEnabled(final boolean enable) {
>-    executor.execute(new Runnable() {
>-      @Override
>-      public void run() {
>-        enableAudio = enable;
>-        if (localAudioTrack != null) {
>-          localAudioTrack.setEnabled(enableAudio);
>-        }
>+    executor.execute(() -> {
>+      enableAudio = enable;
>+      if (localAudioTrack != null) {
>+        localAudioTrack.setEnabled(enableAudio);
>       }
>     });
>   }
> 
>   public void setVideoEnabled(final boolean enable) {
>-    executor.execute(new Runnable() {
>-      @Override
>-      public void run() {
>-        renderVideo = enable;
>-        if (localVideoTrack != null) {
>-          localVideoTrack.setEnabled(renderVideo);
>-        }
>-        if (remoteVideoTrack != null) {
>-          remoteVideoTrack.setEnabled(renderVideo);
>-        }
>+    executor.execute(() -> {
>+      renderVideo = enable;
>+      if (localVideoTrack != null) {
>+        localVideoTrack.setEnabled(renderVideo);
>+      }
>+      if (remoteVideoTrack != null) {
>+        remoteVideoTrack.setEnabled(renderVideo);
>       }
>     });
>   }
> 
>   public void createOffer() {
>-    executor.execute(new Runnable() {
>-      @Override
>-      public void run() {
>-        if (peerConnection != null && !isError) {
>-          Log.d(TAG, "PC Create OFFER");
>-          isInitiator = true;
>-          peerConnection.createOffer(sdpObserver, sdpMediaConstraints);
>-        }
>+    executor.execute(() -> {
>+      if (peerConnection != null && !isError) {
>+        Log.d(TAG, "PC Create OFFER");
>+        isInitiator = true;
>+        peerConnection.createOffer(sdpObserver, sdpMediaConstraints);
>       }
>     });
>   }
> 
>   public void createAnswer() {
>-    executor.execute(new Runnable() {
>-      @Override
>-      public void run() {
>-        if (peerConnection != null && !isError) {
>-          Log.d(TAG, "PC create ANSWER");
>-          isInitiator = false;
>-          peerConnection.createAnswer(sdpObserver, sdpMediaConstraints);
>-        }
>+    executor.execute(() -> {
>+      if (peerConnection != null && !isError) {
>+        Log.d(TAG, "PC create ANSWER");
>+        isInitiator = false;
>+        peerConnection.createAnswer(sdpObserver, sdpMediaConstraints);
>       }
>     });
>   }
> 
>   public void addRemoteIceCandidate(final IceCandidate candidate) {
>-    executor.execute(new Runnable() {
>-      @Override
>-      public void run() {
>-        if (peerConnection != null && !isError) {
>-          if (queuedRemoteCandidates != null) {
>-            queuedRemoteCandidates.add(candidate);
>-          } else {
>-            peerConnection.addIceCandidate(candidate);
>-          }
>+    executor.execute(() -> {
>+      if (peerConnection != null && !isError) {
>+        if (queuedRemoteCandidates != null) {
>+          queuedRemoteCandidates.add(candidate);
>+        } else {
>+          peerConnection.addIceCandidate(candidate);
>         }
>       }
>     });
>   }
> 
>   public void removeRemoteIceCandidates(final IceCandidate[] candidates) {
>-    executor.execute(new Runnable() {
>-      @Override
>-      public void run() {
>-        if (peerConnection == null || isError) {
>-          return;
>-        }
>-        // Drain the queued remote candidates if there is any so that
>-        // they are processed in the proper order.
>-        drainCandidates();
>-        peerConnection.removeIceCandidates(candidates);
>+    executor.execute(() -> {
>+      if (peerConnection == null || isError) {
>+        return;
>       }
>+      // Drain the queued remote candidates if there is any so that
>+      // they are processed in the proper order.
>+      drainCandidates();
>+      peerConnection.removeIceCandidates(candidates);
>     });
>   }
> 
>   public void setRemoteDescription(final SessionDescription sdp) {
>-    executor.execute(new Runnable() {
>-      @Override
>-      public void run() {
>-        if (peerConnection == null || isError) {
>-          return;
>-        }
>-        String sdpDescription = sdp.description;
>-        if (preferIsac) {
>-          sdpDescription = preferCodec(sdpDescription, AUDIO_CODEC_ISAC, true);
>-        }
>-        if (videoCallEnabled) {
>-          sdpDescription = preferCodec(sdpDescription, preferredVideoCodec, false);
>-        }
>-        if (peerConnectionParameters.audioStartBitrate > 0) {
>-          sdpDescription = setStartBitrate(
>-              AUDIO_CODEC_OPUS, false, sdpDescription, peerConnectionParameters.audioStartBitrate);
>-        }
>-        Log.d(TAG, "Set remote SDP.");
>-        SessionDescription sdpRemote = new SessionDescription(sdp.type, sdpDescription);
>-        peerConnection.setRemoteDescription(sdpObserver, sdpRemote);
>+    executor.execute(() -> {
>+      if (peerConnection == null || isError) {
>+        return;
>+      }
>+      String sdpDescription = sdp.description;
>+      if (preferIsac) {
>+        sdpDescription = preferCodec(sdpDescription, AUDIO_CODEC_ISAC, true);
>+      }
>+      if (isVideoCallEnabled()) {
>+        sdpDescription =
>+            preferCodec(sdpDescription, getSdpVideoCodecName(peerConnectionParameters), false);
>       }
>+      if (peerConnectionParameters.audioStartBitrate > 0) {
>+        sdpDescription = setStartBitrate(
>+            AUDIO_CODEC_OPUS, false, sdpDescription, peerConnectionParameters.audioStartBitrate);
>+      }
>+      Log.d(TAG, "Set remote SDP.");
>+      SessionDescription sdpRemote = new SessionDescription(sdp.type, sdpDescription);
>+      peerConnection.setRemoteDescription(sdpObserver, sdpRemote);
>     });
>   }
> 
>   public void stopVideoSource() {
>-    executor.execute(new Runnable() {
>-      @Override
>-      public void run() {
>-        if (videoCapturer != null && !videoCapturerStopped) {
>-          Log.d(TAG, "Stop video source.");
>-          try {
>-            videoCapturer.stopCapture();
>-          } catch (InterruptedException e) {
>-          }
>-          videoCapturerStopped = true;
>+    executor.execute(() -> {
>+      if (videoCapturer != null && !videoCapturerStopped) {
>+        Log.d(TAG, "Stop video source.");
>+        try {
>+          videoCapturer.stopCapture();
>+        } catch (InterruptedException e) {
>         }
>+        videoCapturerStopped = true;
>       }
>     });
>   }
> 
>   public void startVideoSource() {
>-    executor.execute(new Runnable() {
>-      @Override
>-      public void run() {
>-        if (videoCapturer != null && videoCapturerStopped) {
>-          Log.d(TAG, "Restart video source.");
>-          videoCapturer.startCapture(videoWidth, videoHeight, videoFps);
>-          videoCapturerStopped = false;
>-        }
>+    executor.execute(() -> {
>+      if (videoCapturer != null && videoCapturerStopped) {
>+        Log.d(TAG, "Restart video source.");
>+        videoCapturer.startCapture(videoWidth, videoHeight, videoFps);
>+        videoCapturerStopped = false;
>       }
>     });
>   }
> 
>-  public void setVideoMaxBitrate(final Integer maxBitrateKbps) {
>-    executor.execute(new Runnable() {
>-      @Override
>-      public void run() {
>-        if (peerConnection == null || localVideoSender == null || isError) {
>-          return;
>-        }
>-        Log.d(TAG, "Requested max video bitrate: " + maxBitrateKbps);
>-        if (localVideoSender == null) {
>-          Log.w(TAG, "Sender is not ready.");
>-          return;
>-        }
>+  public void setVideoMaxBitrate(@Nullable final Integer maxBitrateKbps) {
>+    executor.execute(() -> {
>+      if (peerConnection == null || localVideoSender == null || isError) {
>+        return;
>+      }
>+      Log.d(TAG, "Requested max video bitrate: " + maxBitrateKbps);
>+      if (localVideoSender == null) {
>+        Log.w(TAG, "Sender is not ready.");
>+        return;
>+      }
> 
>-        RtpParameters parameters = localVideoSender.getParameters();
>-        if (parameters.encodings.size() == 0) {
>-          Log.w(TAG, "RtpParameters are not ready.");
>-          return;
>-        }
>+      RtpParameters parameters = localVideoSender.getParameters();
>+      if (parameters.encodings.size() == 0) {
>+        Log.w(TAG, "RtpParameters are not ready.");
>+        return;
>+      }
> 
>-        for (RtpParameters.Encoding encoding : parameters.encodings) {
>-          // Null value means no limit.
>-          encoding.maxBitrateBps = maxBitrateKbps == null ? null : maxBitrateKbps * BPS_IN_KBPS;
>-        }
>-        if (!localVideoSender.setParameters(parameters)) {
>-          Log.e(TAG, "RtpSender.setParameters failed.");
>-        }
>-        Log.d(TAG, "Configured max video bitrate to: " + maxBitrateKbps);
>+      for (RtpParameters.Encoding encoding : parameters.encodings) {
>+        // Null value means no limit.
>+        encoding.maxBitrateBps = maxBitrateKbps == null ? null : maxBitrateKbps * BPS_IN_KBPS;
>       }
>+      if (!localVideoSender.setParameters(parameters)) {
>+        Log.e(TAG, "RtpSender.setParameters failed.");
>+      }
>+      Log.d(TAG, "Configured max video bitrate to: " + maxBitrateKbps);
>     });
>   }
> 
>   private void reportError(final String errorMessage) {
>     Log.e(TAG, "Peerconnection error: " + errorMessage);
>-    executor.execute(new Runnable() {
>-      @Override
>-      public void run() {
>-        if (!isError) {
>-          events.onPeerConnectionError(errorMessage);
>-          isError = true;
>-        }
>+    executor.execute(() -> {
>+      if (!isError) {
>+        events.onPeerConnectionError(errorMessage);
>+        isError = true;
>       }
>     });
>   }
> 
>+  @Nullable
>   private AudioTrack createAudioTrack() {
>     audioSource = factory.createAudioSource(audioConstraints);
>     localAudioTrack = factory.createAudioTrack(AUDIO_TRACK_ID, audioSource);
>@@ -964,8 +980,12 @@ public class PeerConnectionClient {
>     return localAudioTrack;
>   }
> 
>+  @Nullable
>   private VideoTrack createVideoTrack(VideoCapturer capturer) {
>-    videoSource = factory.createVideoSource(capturer);
>+    surfaceTextureHelper =
>+        SurfaceTextureHelper.create("CaptureThread", rootEglBase.getEglBaseContext());
>+    videoSource = factory.createVideoSource(capturer.isScreencast());
>+    capturer.initialize(surfaceTextureHelper, appContext, videoSource.getCapturerObserver());
>     capturer.startCapture(videoWidth, videoHeight, videoFps);
> 
>     localVideoTrack = factory.createVideoTrack(VIDEO_TRACK_ID, videoSource);
>@@ -986,6 +1006,46 @@ public class PeerConnectionClient {
>     }
>   }
> 
>+  // Returns the remote VideoTrack, assuming there is only one.
>+  private @Nullable VideoTrack getRemoteVideoTrack() {
>+    for (RtpTransceiver transceiver : peerConnection.getTransceivers()) {
>+      MediaStreamTrack track = transceiver.getReceiver().track();
>+      if (track instanceof VideoTrack) {
>+        return (VideoTrack) track;
>+      }
>+    }
>+    return null;
>+  }
>+
>+  private static String getSdpVideoCodecName(PeerConnectionParameters parameters) {
>+    switch (parameters.videoCodec) {
>+      case VIDEO_CODEC_VP8:
>+        return VIDEO_CODEC_VP8;
>+      case VIDEO_CODEC_VP9:
>+        return VIDEO_CODEC_VP9;
>+      case VIDEO_CODEC_H264_HIGH:
>+      case VIDEO_CODEC_H264_BASELINE:
>+        return VIDEO_CODEC_H264;
>+      default:
>+        return VIDEO_CODEC_VP8;
>+    }
>+  }
>+
>+  private static String getFieldTrials(PeerConnectionParameters peerConnectionParameters) {
>+    String fieldTrials = "";
>+    if (peerConnectionParameters.videoFlexfecEnabled) {
>+      fieldTrials += VIDEO_FLEXFEC_FIELDTRIAL;
>+      Log.d(TAG, "Enable FlexFEC field trial.");
>+    }
>+    fieldTrials += VIDEO_VP8_INTEL_HW_ENCODER_FIELDTRIAL;
>+    if (peerConnectionParameters.disableWebRtcAGCAndHPF) {
>+      fieldTrials += DISABLE_WEBRTC_AGC_FIELDTRIAL;
>+      Log.d(TAG, "Disable WebRTC AGC field trial.");
>+    }
>+    return fieldTrials;
>+  }
>+
>+  @SuppressWarnings("StringSplitter")
>   private static String setStartBitrate(
>       String codec, boolean isVideoCodec, String sdpDescription, int bitrateKbps) {
>     String[] lines = sdpDescription.split("\r\n");
>@@ -1076,7 +1136,8 @@ public class PeerConnectionClient {
>     return buffer.toString();
>   }
> 
>-  private static String movePayloadTypesToFront(List<String> preferredPayloadTypes, String mLine) {
>+  private static @Nullable String movePayloadTypesToFront(
>+      List<String> preferredPayloadTypes, String mLine) {
>     // The format of the media description line should be: m=<media> <port> <proto> <fmt> ...
>     final List<String> origLineParts = Arrays.asList(mLine.split(" "));
>     if (origLineParts.size() <= 3) {
>@@ -1140,8 +1201,9 @@ public class PeerConnectionClient {
> 
>   private void switchCameraInternal() {
>     if (videoCapturer instanceof CameraVideoCapturer) {
>-      if (!videoCallEnabled || isError) {
>-        Log.e(TAG, "Failed to switch camera. Video: " + videoCallEnabled + ". Error : " + isError);
>+      if (!isVideoCallEnabled() || isError) {
>+        Log.e(TAG,
>+            "Failed to switch camera. Video: " + isVideoCallEnabled() + ". Error : " + isError);
>         return; // No video is sent or only one camera is available or error happened.
>       }
>       Log.d(TAG, "Switch camera");
>@@ -1153,27 +1215,18 @@ public class PeerConnectionClient {
>   }
> 
>   public void switchCamera() {
>-    executor.execute(new Runnable() {
>-      @Override
>-      public void run() {
>-        switchCameraInternal();
>-      }
>-    });
>+    executor.execute(this ::switchCameraInternal);
>   }
> 
>   public void changeCaptureFormat(final int width, final int height, final int framerate) {
>-    executor.execute(new Runnable() {
>-      @Override
>-      public void run() {
>-        changeCaptureFormatInternal(width, height, framerate);
>-      }
>-    });
>+    executor.execute(() -> changeCaptureFormatInternal(width, height, framerate));
>   }
> 
>   private void changeCaptureFormatInternal(int width, int height, int framerate) {
>-    if (!videoCallEnabled || isError || videoCapturer == null) {
>+    if (!isVideoCallEnabled() || isError || videoCapturer == null) {
>       Log.e(TAG,
>-          "Failed to change capture format. Video: " + videoCallEnabled + ". Error : " + isError);
>+          "Failed to change capture format. Video: " + isVideoCallEnabled()
>+              + ". Error : " + isError);
>       return;
>     }
>     Log.d(TAG, "changeCaptureFormat: " + width + "x" + height + "@" + framerate);
>@@ -1184,22 +1237,12 @@ public class PeerConnectionClient {
>   private class PCObserver implements PeerConnection.Observer {
>     @Override
>     public void onIceCandidate(final IceCandidate candidate) {
>-      executor.execute(new Runnable() {
>-        @Override
>-        public void run() {
>-          events.onIceCandidate(candidate);
>-        }
>-      });
>+      executor.execute(() -> events.onIceCandidate(candidate));
>     }
> 
>     @Override
>     public void onIceCandidatesRemoved(final IceCandidate[] candidates) {
>-      executor.execute(new Runnable() {
>-        @Override
>-        public void run() {
>-          events.onIceCandidatesRemoved(candidates);
>-        }
>-      });
>+      executor.execute(() -> events.onIceCandidatesRemoved(candidates));
>     }
> 
>     @Override
>@@ -1209,17 +1252,14 @@ public class PeerConnectionClient {
> 
>     @Override
>     public void onIceConnectionChange(final PeerConnection.IceConnectionState newState) {
>-      executor.execute(new Runnable() {
>-        @Override
>-        public void run() {
>-          Log.d(TAG, "IceConnectionState: " + newState);
>-          if (newState == IceConnectionState.CONNECTED) {
>-            events.onIceConnected();
>-          } else if (newState == IceConnectionState.DISCONNECTED) {
>-            events.onIceDisconnected();
>-          } else if (newState == IceConnectionState.FAILED) {
>-            reportError("ICE connection failed.");
>-          }
>+      executor.execute(() -> {
>+        Log.d(TAG, "IceConnectionState: " + newState);
>+        if (newState == IceConnectionState.CONNECTED) {
>+          events.onIceConnected();
>+        } else if (newState == IceConnectionState.DISCONNECTED) {
>+          events.onIceDisconnected();
>+        } else if (newState == IceConnectionState.FAILED) {
>+          reportError("ICE connection failed.");
>         }
>       });
>     }
>@@ -1235,37 +1275,10 @@ public class PeerConnectionClient {
>     }
> 
>     @Override
>-    public void onAddStream(final MediaStream stream) {
>-      executor.execute(new Runnable() {
>-        @Override
>-        public void run() {
>-          if (peerConnection == null || isError) {
>-            return;
>-          }
>-          if (stream.audioTracks.size() > 1 || stream.videoTracks.size() > 1) {
>-            reportError("Weird-looking stream: " + stream);
>-            return;
>-          }
>-          if (stream.videoTracks.size() == 1) {
>-            remoteVideoTrack = stream.videoTracks.get(0);
>-            remoteVideoTrack.setEnabled(renderVideo);
>-            for (VideoRenderer.Callbacks remoteRender : remoteRenders) {
>-              remoteVideoTrack.addRenderer(new VideoRenderer(remoteRender));
>-            }
>-          }
>-        }
>-      });
>-    }
>+    public void onAddStream(final MediaStream stream) {}
> 
>     @Override
>-    public void onRemoveStream(final MediaStream stream) {
>-      executor.execute(new Runnable() {
>-        @Override
>-        public void run() {
>-          remoteVideoTrack = null;
>-        }
>-      });
>-    }
>+    public void onRemoveStream(final MediaStream stream) {}
> 
>     @Override
>     public void onDataChannel(final DataChannel dc) {
>@@ -1323,57 +1336,52 @@ public class PeerConnectionClient {
>       if (preferIsac) {
>         sdpDescription = preferCodec(sdpDescription, AUDIO_CODEC_ISAC, true);
>       }
>-      if (videoCallEnabled) {
>-        sdpDescription = preferCodec(sdpDescription, preferredVideoCodec, false);
>+      if (isVideoCallEnabled()) {
>+        sdpDescription =
>+            preferCodec(sdpDescription, getSdpVideoCodecName(peerConnectionParameters), false);
>       }
>       final SessionDescription sdp = new SessionDescription(origSdp.type, sdpDescription);
>       localSdp = sdp;
>-      executor.execute(new Runnable() {
>-        @Override
>-        public void run() {
>-          if (peerConnection != null && !isError) {
>-            Log.d(TAG, "Set local SDP from " + sdp.type);
>-            peerConnection.setLocalDescription(sdpObserver, sdp);
>-          }
>+      executor.execute(() -> {
>+        if (peerConnection != null && !isError) {
>+          Log.d(TAG, "Set local SDP from " + sdp.type);
>+          peerConnection.setLocalDescription(sdpObserver, sdp);
>         }
>       });
>     }
> 
>     @Override
>     public void onSetSuccess() {
>-      executor.execute(new Runnable() {
>-        @Override
>-        public void run() {
>-          if (peerConnection == null || isError) {
>-            return;
>+      executor.execute(() -> {
>+        if (peerConnection == null || isError) {
>+          return;
>+        }
>+        if (isInitiator) {
>+          // For offering peer connection we first create offer and set
>+          // local SDP, then after receiving answer set remote SDP.
>+          if (peerConnection.getRemoteDescription() == null) {
>+            // We've just set our local SDP so time to send it.
>+            Log.d(TAG, "Local SDP set succesfully");
>+            events.onLocalDescription(localSdp);
>+          } else {
>+            // We've just set remote description, so drain remote
>+            // and send local ICE candidates.
>+            Log.d(TAG, "Remote SDP set succesfully");
>+            drainCandidates();
>           }
>-          if (isInitiator) {
>-            // For offering peer connection we first create offer and set
>-            // local SDP, then after receiving answer set remote SDP.
>-            if (peerConnection.getRemoteDescription() == null) {
>-              // We've just set our local SDP so time to send it.
>-              Log.d(TAG, "Local SDP set succesfully");
>-              events.onLocalDescription(localSdp);
>-            } else {
>-              // We've just set remote description, so drain remote
>-              // and send local ICE candidates.
>-              Log.d(TAG, "Remote SDP set succesfully");
>-              drainCandidates();
>-            }
>+        } else {
>+          // For answering peer connection we set remote SDP and then
>+          // create answer and set local SDP.
>+          if (peerConnection.getLocalDescription() != null) {
>+            // We've just set our local SDP so time to send it, drain
>+            // remote and send local ICE candidates.
>+            Log.d(TAG, "Local SDP set succesfully");
>+            events.onLocalDescription(localSdp);
>+            drainCandidates();
>           } else {
>-            // For answering peer connection we set remote SDP and then
>-            // create answer and set local SDP.
>-            if (peerConnection.getLocalDescription() != null) {
>-              // We've just set our local SDP so time to send it, drain
>-              // remote and send local ICE candidates.
>-              Log.d(TAG, "Local SDP set succesfully");
>-              events.onLocalDescription(localSdp);
>-              drainCandidates();
>-            } else {
>-              // We've just set remote SDP - do nothing for now -
>-              // answer will be created soon.
>-              Log.d(TAG, "Remote SDP set succesfully");
>-            }
>+            // We've just set remote SDP - do nothing for now -
>+            // answer will be created soon.
>+            Log.d(TAG, "Remote SDP set succesfully");
>           }
>         }
>       });
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/RecordedAudioToFileController.java b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/RecordedAudioToFileController.java
>new file mode 100644
>index 00000000000..ede98194330
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/RecordedAudioToFileController.java
>@@ -0,0 +1,154 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.appspot.apprtc;
>+
>+import android.media.AudioFormat;
>+import android.os.Environment;
>+import javax.annotation.Nullable;
>+import android.util.Log;
>+import java.io.File;
>+import java.io.FileOutputStream;
>+import java.io.FileNotFoundException;
>+import java.io.IOException;
>+import java.io.OutputStream;
>+import java.util.concurrent.ExecutorService;
>+import org.webrtc.audio.JavaAudioDeviceModule;
>+import org.webrtc.audio.JavaAudioDeviceModule.SamplesReadyCallback;
>+import org.webrtc.voiceengine.WebRtcAudioRecord;
>+import org.webrtc.voiceengine.WebRtcAudioRecord.WebRtcAudioRecordSamplesReadyCallback;
>+
>+/**
>+ * Implements the AudioRecordSamplesReadyCallback interface and writes
>+ * recorded raw audio samples to an output file.
>+ */
>+public class RecordedAudioToFileController
>+    implements SamplesReadyCallback, WebRtcAudioRecordSamplesReadyCallback {
>+  private static final String TAG = "RecordedAudioToFile";
>+  private static final long MAX_FILE_SIZE_IN_BYTES = 58348800L;
>+
>+  private final Object lock = new Object();
>+  private final ExecutorService executor;
>+  @Nullable
>+  private OutputStream rawAudioFileOutputStream = null;
>+  private boolean isRunning;
>+  private long fileSizeInBytes = 0;
>+
>+  public RecordedAudioToFileController(ExecutorService executor) {
>+    Log.d(TAG, "ctor");
>+    this.executor = executor;
>+  }
>+
>+  /**
>+   * Should be called on the same executor thread as the one provided at
>+   * construction.
>+   */
>+  public boolean start() {
>+    Log.d(TAG, "start");
>+    if (!isExternalStorageWritable()) {
>+      Log.e(TAG, "Writing to external media is not possible");
>+      return false;
>+    }
>+    synchronized (lock) {
>+      isRunning = true;
>+    }
>+    return true;
>+  }
>+
>+  /**
>+   * Should be called on the same executor thread as the one provided at
>+   * construction.
>+   */
>+  public void stop() {
>+    Log.d(TAG, "stop");
>+    synchronized (lock) {
>+      isRunning = false;
>+      if (rawAudioFileOutputStream != null) {
>+        try {
>+          rawAudioFileOutputStream.close();
>+        } catch (IOException e) {
>+          Log.e(TAG, "Failed to close file with saved input audio: " + e);
>+        }
>+        rawAudioFileOutputStream = null;
>+      }
>+      fileSizeInBytes = 0;
>+    }
>+  }
>+
>+  // Checks if external storage is available for read and write.
>+  private boolean isExternalStorageWritable() {
>+    String state = Environment.getExternalStorageState();
>+    if (Environment.MEDIA_MOUNTED.equals(state)) {
>+      return true;
>+    }
>+    return false;
>+  }
>+
>+  // Utilizes audio parameters to create a file name which contains sufficient
>+  // information so that the file can be played using an external file player.
>+  // Example: /sdcard/recorded_audio_16bits_48000Hz_mono.pcm.
>+  private void openRawAudioOutputFile(int sampleRate, int channelCount) {
>+    final String fileName = Environment.getExternalStorageDirectory().getPath() + File.separator
>+        + "recorded_audio_16bits_" + String.valueOf(sampleRate) + "Hz"
>+        + ((channelCount == 1) ? "_mono" : "_stereo") + ".pcm";
>+    final File outputFile = new File(fileName);
>+    try {
>+      rawAudioFileOutputStream = new FileOutputStream(outputFile);
>+    } catch (FileNotFoundException e) {
>+      Log.e(TAG, "Failed to open audio output file: " + e.getMessage());
>+    }
>+    Log.d(TAG, "Opened file for recording: " + fileName);
>+  }
>+
>+  // Called when new audio samples are ready.
>+  @Override
>+  public void onWebRtcAudioRecordSamplesReady(WebRtcAudioRecord.AudioSamples samples) {
>+    onWebRtcAudioRecordSamplesReady(new JavaAudioDeviceModule.AudioSamples(samples.getAudioFormat(),
>+        samples.getChannelCount(), samples.getSampleRate(), samples.getData()));
>+  }
>+
>+  // Called when new audio samples are ready.
>+  @Override
>+  public void onWebRtcAudioRecordSamplesReady(JavaAudioDeviceModule.AudioSamples samples) {
>+    // The native audio layer on Android should use 16-bit PCM format.
>+    if (samples.getAudioFormat() != AudioFormat.ENCODING_PCM_16BIT) {
>+      Log.e(TAG, "Invalid audio format");
>+      return;
>+    }
>+    synchronized (lock) {
>+      // Abort early if stop() has been called.
>+      if (!isRunning) {
>+        return;
>+      }
>+      // Open a new file for the first callback only since it allows us to add audio parameters to
>+      // the file name.
>+      if (rawAudioFileOutputStream == null) {
>+        openRawAudioOutputFile(samples.getSampleRate(), samples.getChannelCount());
>+        fileSizeInBytes = 0;
>+      }
>+    }
>+    // Append the recorded 16-bit audio samples to the open output file.
>+    executor.execute(() -> {
>+      if (rawAudioFileOutputStream != null) {
>+        try {
>+          // Set a limit on max file size. 58348800 bytes corresponds to
>+          // approximately 10 minutes of recording in mono at 48kHz.
>+          if (fileSizeInBytes < MAX_FILE_SIZE_IN_BYTES) {
>+            // Writes samples.getData().length bytes to output stream.
>+            rawAudioFileOutputStream.write(samples.getData());
>+            fileSizeInBytes += samples.getData().length;
>+          }
>+        } catch (IOException e) {
>+          Log.e(TAG, "Failed to write audio to file: " + e.getMessage());
>+        }
>+      }
>+    });
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/RtcEventLog.java b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/RtcEventLog.java
>new file mode 100644
>index 00000000000..bbbd06b1eec
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/RtcEventLog.java
>@@ -0,0 +1,74 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.appspot.apprtc;
>+
>+import android.content.Context;
>+import android.os.ParcelFileDescriptor;
>+import android.util.Log;
>+import java.io.File;
>+import java.io.IOException;
>+import org.webrtc.PeerConnection;
>+
>+public class RtcEventLog {
>+  private static final String TAG = "RtcEventLog";
>+  private static final int OUTPUT_FILE_MAX_BYTES = 10_000_000;
>+  private final PeerConnection peerConnection;
>+  private RtcEventLogState state = RtcEventLogState.INACTIVE;
>+
>+  enum RtcEventLogState {
>+    INACTIVE,
>+    STARTED,
>+    STOPPED,
>+  }
>+
>+  public RtcEventLog(PeerConnection peerConnection) {
>+    if (peerConnection == null) {
>+      throw new NullPointerException("The peer connection is null.");
>+    }
>+    this.peerConnection = peerConnection;
>+  }
>+
>+  public void start(final File outputFile) {
>+    if (state == RtcEventLogState.STARTED) {
>+      Log.e(TAG, "RtcEventLog has already started.");
>+      return;
>+    }
>+    final ParcelFileDescriptor fileDescriptor;
>+    try {
>+      fileDescriptor = ParcelFileDescriptor.open(outputFile,
>+          ParcelFileDescriptor.MODE_READ_WRITE | ParcelFileDescriptor.MODE_CREATE
>+              | ParcelFileDescriptor.MODE_TRUNCATE);
>+    } catch (IOException e) {
>+      Log.e(TAG, "Failed to create a new file", e);
>+      return;
>+    }
>+
>+    // Passes ownership of the file to WebRTC.
>+    boolean success =
>+        peerConnection.startRtcEventLog(fileDescriptor.detachFd(), OUTPUT_FILE_MAX_BYTES);
>+    if (!success) {
>+      Log.e(TAG, "Failed to start RTC event log.");
>+      return;
>+    }
>+    state = RtcEventLogState.STARTED;
>+    Log.d(TAG, "RtcEventLog started.");
>+  }
>+
>+  public void stop() {
>+    if (state != RtcEventLogState.STARTED) {
>+      Log.e(TAG, "RtcEventLog was not started.");
>+      return;
>+    }
>+    peerConnection.stopRtcEventLog();
>+    state = RtcEventLogState.STOPPED;
>+    Log.d(TAG, "RtcEventLog stopped.");
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/SettingsActivity.java b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/SettingsActivity.java
>index 43b8a0a662e..8a556c335f6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/SettingsActivity.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/SettingsActivity.java
>@@ -17,7 +17,7 @@ import android.os.Bundle;
> import android.preference.ListPreference;
> import android.preference.Preference;
> import org.webrtc.Camera2Enumerator;
>-import org.webrtc.voiceengine.WebRtcAudioUtils;
>+import org.webrtc.audio.JavaAudioDeviceModule;
> 
> /**
>  * Settings activity for AppRTC.
>@@ -42,17 +42,18 @@ public class SettingsActivity extends Activity implements OnSharedPreferenceChan
>   private String keyPrefAudioCodec;
>   private String keyprefNoAudioProcessing;
>   private String keyprefAecDump;
>+  private String keyprefEnableSaveInputAudioToFile;
>   private String keyprefOpenSLES;
>   private String keyprefDisableBuiltInAEC;
>   private String keyprefDisableBuiltInAGC;
>   private String keyprefDisableBuiltInNS;
>-  private String keyprefEnableLevelControl;
>   private String keyprefDisableWebRtcAGCAndHPF;
>   private String keyprefSpeakerphone;
> 
>   private String keyPrefRoomServerUrl;
>   private String keyPrefDisplayHud;
>   private String keyPrefTracing;
>+  private String keyprefEnabledRtcEventLog;
> 
>   private String keyprefEnableDataChannel;
>   private String keyprefOrdered;
>@@ -61,6 +62,7 @@ public class SettingsActivity extends Activity implements OnSharedPreferenceChan
>   private String keyprefDataProtocol;
>   private String keyprefNegotiated;
>   private String keyprefDataId;
>+  private String keyprefUseLegacyAudioDevice;
> 
>   @Override
>   protected void onCreate(Bundle savedInstanceState) {
>@@ -83,11 +85,12 @@ public class SettingsActivity extends Activity implements OnSharedPreferenceChan
>     keyPrefAudioCodec = getString(R.string.pref_audiocodec_key);
>     keyprefNoAudioProcessing = getString(R.string.pref_noaudioprocessing_key);
>     keyprefAecDump = getString(R.string.pref_aecdump_key);
>+    keyprefEnableSaveInputAudioToFile =
>+        getString(R.string.pref_enable_save_input_audio_to_file_key);
>     keyprefOpenSLES = getString(R.string.pref_opensles_key);
>     keyprefDisableBuiltInAEC = getString(R.string.pref_disable_built_in_aec_key);
>     keyprefDisableBuiltInAGC = getString(R.string.pref_disable_built_in_agc_key);
>     keyprefDisableBuiltInNS = getString(R.string.pref_disable_built_in_ns_key);
>-    keyprefEnableLevelControl = getString(R.string.pref_enable_level_control_key);
>     keyprefDisableWebRtcAGCAndHPF = getString(R.string.pref_disable_webrtc_agc_and_hpf_key);
>     keyprefSpeakerphone = getString(R.string.pref_speakerphone_key);
> 
>@@ -102,6 +105,8 @@ public class SettingsActivity extends Activity implements OnSharedPreferenceChan
>     keyPrefRoomServerUrl = getString(R.string.pref_room_server_url_key);
>     keyPrefDisplayHud = getString(R.string.pref_displayhud_key);
>     keyPrefTracing = getString(R.string.pref_tracing_key);
>+    keyprefEnabledRtcEventLog = getString(R.string.pref_enable_rtceventlog_key);
>+    keyprefUseLegacyAudioDevice = getString(R.string.pref_use_legacy_audio_device_key);
> 
>     // Display the fragment as the main content.
>     settingsFragment = new SettingsFragment();
>@@ -138,11 +143,11 @@ public class SettingsActivity extends Activity implements OnSharedPreferenceChan
>     updateSummary(sharedPreferences, keyPrefAudioCodec);
>     updateSummaryB(sharedPreferences, keyprefNoAudioProcessing);
>     updateSummaryB(sharedPreferences, keyprefAecDump);
>+    updateSummaryB(sharedPreferences, keyprefEnableSaveInputAudioToFile);
>     updateSummaryB(sharedPreferences, keyprefOpenSLES);
>     updateSummaryB(sharedPreferences, keyprefDisableBuiltInAEC);
>     updateSummaryB(sharedPreferences, keyprefDisableBuiltInAGC);
>     updateSummaryB(sharedPreferences, keyprefDisableBuiltInNS);
>-    updateSummaryB(sharedPreferences, keyprefEnableLevelControl);
>     updateSummaryB(sharedPreferences, keyprefDisableWebRtcAGCAndHPF);
>     updateSummaryList(sharedPreferences, keyprefSpeakerphone);
> 
>@@ -158,6 +163,8 @@ public class SettingsActivity extends Activity implements OnSharedPreferenceChan
>     updateSummary(sharedPreferences, keyPrefRoomServerUrl);
>     updateSummaryB(sharedPreferences, keyPrefDisplayHud);
>     updateSummaryB(sharedPreferences, keyPrefTracing);
>+    updateSummaryB(sharedPreferences, keyprefEnabledRtcEventLog);
>+    updateSummaryB(sharedPreferences, keyprefUseLegacyAudioDevice);
> 
>     if (!Camera2Enumerator.isSupported(this)) {
>       Preference camera2Preference = settingsFragment.findPreference(keyprefCamera2);
>@@ -166,10 +173,7 @@ public class SettingsActivity extends Activity implements OnSharedPreferenceChan
>       camera2Preference.setEnabled(false);
>     }
> 
>-    // Disable forcing WebRTC based AEC so it won't affect our value.
>-    // Otherwise, if it was enabled, isAcousticEchoCancelerSupported would always return false.
>-    WebRtcAudioUtils.setWebRtcBasedAcousticEchoCanceler(false);
>-    if (!WebRtcAudioUtils.isAcousticEchoCancelerSupported()) {
>+    if (!JavaAudioDeviceModule.isBuiltInAcousticEchoCancelerSupported()) {
>       Preference disableBuiltInAECPreference =
>           settingsFragment.findPreference(keyprefDisableBuiltInAEC);
> 
>@@ -177,17 +181,13 @@ public class SettingsActivity extends Activity implements OnSharedPreferenceChan
>       disableBuiltInAECPreference.setEnabled(false);
>     }
> 
>-    WebRtcAudioUtils.setWebRtcBasedAutomaticGainControl(false);
>-    if (!WebRtcAudioUtils.isAutomaticGainControlSupported()) {
>-      Preference disableBuiltInAGCPreference =
>-          settingsFragment.findPreference(keyprefDisableBuiltInAGC);
>+    Preference disableBuiltInAGCPreference =
>+        settingsFragment.findPreference(keyprefDisableBuiltInAGC);
> 
>-      disableBuiltInAGCPreference.setSummary(getString(R.string.pref_built_in_agc_not_available));
>-      disableBuiltInAGCPreference.setEnabled(false);
>-    }
>+    disableBuiltInAGCPreference.setSummary(getString(R.string.pref_built_in_agc_not_available));
>+    disableBuiltInAGCPreference.setEnabled(false);
> 
>-    WebRtcAudioUtils.setWebRtcBasedNoiseSuppressor(false);
>-    if (!WebRtcAudioUtils.isNoiseSuppressorSupported()) {
>+    if (!JavaAudioDeviceModule.isBuiltInNoiseSuppressorSupported()) {
>       Preference disableBuiltInNSPreference =
>           settingsFragment.findPreference(keyprefDisableBuiltInNS);
> 
>@@ -232,16 +232,18 @@ public class SettingsActivity extends Activity implements OnSharedPreferenceChan
>         || key.equals(keyprefFlexfec)
>         || key.equals(keyprefNoAudioProcessing)
>         || key.equals(keyprefAecDump)
>+        || key.equals(keyprefEnableSaveInputAudioToFile)
>         || key.equals(keyprefOpenSLES)
>         || key.equals(keyprefDisableBuiltInAEC)
>         || key.equals(keyprefDisableBuiltInAGC)
>         || key.equals(keyprefDisableBuiltInNS)
>-        || key.equals(keyprefEnableLevelControl)
>         || key.equals(keyprefDisableWebRtcAGCAndHPF)
>         || key.equals(keyPrefDisplayHud)
>         || key.equals(keyprefEnableDataChannel)
>         || key.equals(keyprefOrdered)
>-        || key.equals(keyprefNegotiated)) {
>+        || key.equals(keyprefNegotiated)
>+        || key.equals(keyprefEnabledRtcEventLog)
>+        || key.equals(keyprefUseLegacyAudioDevice)) {
>       updateSummaryB(sharedPreferences, key);
>     } else if (key.equals(keyprefSpeakerphone)) {
>       updateSummaryList(sharedPreferences, key);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/TCPChannelClient.java b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/TCPChannelClient.java
>index 39f41f86174..e9fd7be4059 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/TCPChannelClient.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/TCPChannelClient.java
>@@ -10,6 +10,7 @@
> 
> package org.appspot.apprtc;
> 
>+import javax.annotation.Nullable;
> import android.util.Log;
> import java.io.BufferedReader;
> import java.io.IOException;
>@@ -123,7 +124,9 @@ public class TCPChannelClient {
>   private abstract class TCPSocket extends Thread {
>     // Lock for editing out and rawSocket
>     protected final Object rawSocketLock;
>+    @Nullable
>     private PrintWriter out;
>+    @Nullable
>     private Socket rawSocket;
> 
>     /**
>@@ -131,6 +134,7 @@ public class TCPChannelClient {
>      *
>      * @return Socket connection, null if connection failed.
>      */
>+    @Nullable
>     public abstract Socket connect();
> 
>     /** Returns true if sockets is a server rawSocket. */
>@@ -263,6 +267,7 @@ public class TCPChannelClient {
> 
>   private class TCPSocketServer extends TCPSocket {
>     // Server socket is also guarded by rawSocketLock.
>+    @Nullable
>     private ServerSocket serverSocket;
> 
>     final private InetAddress address;
>@@ -274,6 +279,7 @@ public class TCPChannelClient {
>     }
> 
>     /** Opens a listening socket and waits for a connection. */
>+    @Nullable
>     @Override
>     public Socket connect() {
>       Log.d(TAG, "Listening on [" + address.getHostAddress() + "]:" + Integer.toString(port));
>@@ -335,6 +341,7 @@ public class TCPChannelClient {
>     }
> 
>     /** Connects to the peer. */
>+    @Nullable
>     @Override
>     public Socket connect() {
>       Log.d(TAG, "Connecting to [" + address.getHostAddress() + "]:" + Integer.toString(port));
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/WebSocketChannelClient.java b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/WebSocketChannelClient.java
>index d3e5c8751b0..07b78472f41 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/WebSocketChannelClient.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/WebSocketChannelClient.java
>@@ -11,6 +11,7 @@
> package org.appspot.apprtc;
> 
> import android.os.Handler;
>+import javax.annotation.Nullable;
> import android.util.Log;
> import de.tavendo.autobahn.WebSocket.WebSocketConnectionObserver;
> import de.tavendo.autobahn.WebSocketConnection;
>@@ -39,7 +40,9 @@ public class WebSocketChannelClient {
>   private WebSocketConnection ws;
>   private String wsServerUrl;
>   private String postServerUrl;
>+  @Nullable
>   private String roomID;
>+  @Nullable
>   private String clientID;
>   private WebSocketConnectionState state;
>   // Do not remove this member variable. If this is removed, the observer gets garbage collected and
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/WebSocketRTCClient.java b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/WebSocketRTCClient.java
>index 32150519346..dbe34bed1b1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/WebSocketRTCClient.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/src/org/appspot/apprtc/WebSocketRTCClient.java
>@@ -10,6 +10,7 @@
> 
> package org.appspot.apprtc;
> 
>+import javax.annotation.Nullable;
> import org.appspot.apprtc.RoomParametersFetcher.RoomParametersFetcherEvents;
> import org.appspot.apprtc.WebSocketChannelClient.WebSocketChannelEvents;
> import org.appspot.apprtc.WebSocketChannelClient.WebSocketConnectionState;
>@@ -380,7 +381,7 @@ public class WebSocketRTCClient implements AppRTCClient, WebSocketChannelEvents
> 
>   // Send SDP or ICE candidate to a room server.
>   private void sendPostMessage(
>-      final MessageType messageType, final String url, final String message) {
>+      final MessageType messageType, final String url, @Nullable final String message) {
>     String logInfo = url;
>     if (message != null) {
>       logInfo += ". Message: " + message;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/third_party/autobanh/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/third_party/autobanh/BUILD.gn
>index d5f6b0f2d6a..b671239bae7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/third_party/autobanh/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidapp/third_party/autobanh/BUILD.gn
>@@ -6,8 +6,10 @@
> # in the file PATENTS.  All contributing project authors may
> # be found in the AUTHORS file in the root of the source tree.
> 
>-import("//build/config/android/rules.gni")
>+if (is_android) {
>+  import("//build/config/android/rules.gni")
> 
>-android_java_prebuilt("autobanh_java") {
>-  jar_path = "lib/autobanh.jar"
>+  android_java_prebuilt("autobanh_java") {
>+    jar_path = "lib/autobanh.jar"
>+  }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidjunit/src/org/appspot/apprtc/BluetoothManagerTest.java b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidjunit/src/org/appspot/apprtc/BluetoothManagerTest.java
>index 0724ec17c0c..b97f1f0bf6f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidjunit/src/org/appspot/apprtc/BluetoothManagerTest.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidjunit/src/org/appspot/apprtc/BluetoothManagerTest.java
>@@ -11,11 +11,8 @@
> package org.appspot.apprtc;
> 
> import static org.junit.Assert.assertEquals;
>-import static org.junit.Assert.assertFalse;
> import static org.junit.Assert.assertNotNull;
> import static org.junit.Assert.assertNull;
>-import static org.junit.Assert.assertTrue;
>-import static org.mockito.Mockito.doCallRealMethod;
> import static org.mockito.Mockito.mock;
> import static org.mockito.Mockito.never;
> import static org.mockito.Mockito.times;
>@@ -39,8 +36,8 @@ import org.chromium.testing.local.LocalRobolectricTestRunner;
> import org.junit.Before;
> import org.junit.Test;
> import org.junit.runner.RunWith;
>+import org.robolectric.RuntimeEnvironment;
> import org.robolectric.annotation.Config;
>-import org.robolectric.shadows.ShadowApplication;
> import org.robolectric.shadows.ShadowLog;
> 
> /**
>@@ -68,7 +65,7 @@ public class BluetoothManagerTest {
>   @Before
>   public void setUp() {
>     ShadowLog.stream = System.out;
>-    context = ShadowApplication.getInstance().getApplicationContext();
>+    context = RuntimeEnvironment.application;
>     mockedAppRtcAudioManager = mock(AppRTCAudioManager.class);
>     mockedAudioManager = mock(AudioManager.class);
>     mockedBluetoothHeadset = mock(BluetoothHeadset.class);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidjunit/src/org/appspot/apprtc/TCPChannelClientTest.java b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidjunit/src/org/appspot/apprtc/TCPChannelClientTest.java
>index 0f4916aa48c..8c5f38ccb31 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidjunit/src/org/appspot/apprtc/TCPChannelClientTest.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidjunit/src/org/appspot/apprtc/TCPChannelClientTest.java
>@@ -35,10 +35,10 @@ import java.util.concurrent.TimeUnit;
> public class TCPChannelClientTest {
>   private static final int PORT = 8888;
>   /**
>-   * How long we wait before trying to connect to the server. Chosen quite arbitrarily and
>-   * could be made smaller if need be.
>+   * How long we wait before trying to connect to the server. Note: was
>+   * previously only 10, which was too short (tests were flaky).
>    */
>-  private static final int SERVER_WAIT = 10;
>+  private static final int SERVER_WAIT = 100;
>   private static final int CONNECT_TIMEOUT = 100;
>   private static final int SEND_TIMEOUT = 100;
>   private static final int DISCONNECT_TIMEOUT = 100;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidnativeapi/AndroidManifest.xml b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidnativeapi/AndroidManifest.xml
>new file mode 100644
>index 00000000000..f10f55a1b68
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidnativeapi/AndroidManifest.xml
>@@ -0,0 +1,23 @@
>+<?xml version="1.0" encoding="utf-8"?>
>+<manifest xmlns:android="http://schemas.android.com/apk/res/android"
>+  package="org.webrtc.examples.androidnativeapi">
>+
>+  <uses-sdk android:minSdkVersion="19" android:targetSdkVersion="27" />
>+
>+  <uses-permission android:name="android.permission.INTERNET" />
>+  <uses-permission android:name="android.permission.CAMERA" />
>+
>+  <application
>+    android:allowBackup="true"
>+    android:label="@string/app_name"
>+    android:supportsRtl="true">
>+    <activity android:name=".MainActivity">
>+      <intent-filter>
>+        <action android:name="android.intent.action.MAIN" />
>+
>+        <category android:name="android.intent.category.LAUNCHER" />
>+      </intent-filter>
>+    </activity>
>+  </application>
>+
>+</manifest>
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidnativeapi/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidnativeapi/BUILD.gn
>new file mode 100644
>index 00000000000..f96fab2e8e4
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidnativeapi/BUILD.gn
>@@ -0,0 +1,75 @@
>+import("//webrtc.gni")
>+
>+if (is_android) {
>+  rtc_android_apk("androidnativeapi") {
>+    testonly = true
>+    apk_name = "androidnativeapi"
>+    android_manifest = "AndroidManifest.xml"
>+
>+    java_files = [
>+      "java/org/webrtc/examples/androidnativeapi/MainActivity.java",
>+      "java/org/webrtc/examples/androidnativeapi/CallClient.java",
>+    ]
>+
>+    deps = [
>+      ":resources",
>+      "//modules/audio_device:audio_device_java",
>+      "//sdk/android:camera_java",
>+      "//sdk/android:surfaceviewrenderer_java",
>+      "//sdk/android:video_api_java",
>+      "//sdk/android:video_java",
>+    ]
>+
>+    shared_libraries = [ ":examples_androidnativeapi_jni" ]
>+  }
>+
>+  generate_jni("generated_jni") {
>+    testonly = true
>+    sources = [
>+      "java/org/webrtc/examples/androidnativeapi/CallClient.java",
>+    ]
>+    jni_package = ""
>+    namespace = "webrtc_examples"
>+    jni_generator_include = "//sdk/android/src/jni/jni_generator_helper.h"
>+  }
>+
>+  rtc_shared_library("examples_androidnativeapi_jni") {
>+    testonly = true
>+    sources = [
>+      "jni/androidcallclient.cc",
>+      "jni/androidcallclient.h",
>+      "jni/onload.cc",
>+    ]
>+
>+    suppressed_configs += [ "//build/config/android:hide_all_but_jni_onload" ]
>+    configs += [ "//build/config/android:hide_all_but_jni" ]
>+
>+    deps = [
>+      ":generated_jni",
>+      "//api:libjingle_peerconnection_api",
>+      "//api/audio_codecs:builtin_audio_decoder_factory",
>+      "//api/audio_codecs:builtin_audio_encoder_factory",
>+      "//logging:rtc_event_log_impl_base",
>+      "//media:rtc_audio_video",
>+      "//media:rtc_internal_video_codecs",
>+      "//modules/audio_processing",
>+      "//modules/utility:utility",
>+      "//pc:libjingle_peerconnection",
>+      "//rtc_base:rtc_base",
>+      "//rtc_base:rtc_base_approved",
>+      "//sdk/android:native_api_base",
>+      "//sdk/android:native_api_jni",
>+      "//sdk/android:native_api_video",
>+      "//system_wrappers:field_trial_default",
>+      "//system_wrappers:metrics_default",
>+      "//system_wrappers:runtime_enabled_features_default",
>+      "//third_party/abseil-cpp/absl/memory",
>+    ]
>+  }
>+
>+  android_resources("resources") {
>+    testonly = true
>+    resource_dirs = [ "res" ]
>+    custom_package = "org.webrtc.examples.androidnativeapi"
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidnativeapi/DEPS b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidnativeapi/DEPS
>new file mode 100644
>index 00000000000..2d4c0d8bfbd
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidnativeapi/DEPS
>@@ -0,0 +1,4 @@
>+include_rules = [
>+  "+modules/utility/include",
>+  "+sdk/android/native_api",
>+]
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidnativeapi/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidnativeapi/OWNERS
>new file mode 100644
>index 00000000000..3c4e54174e6
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidnativeapi/OWNERS
>@@ -0,0 +1 @@
>+sakal@webrtc.org
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidnativeapi/java/org/webrtc/examples/androidnativeapi/CallClient.java b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidnativeapi/java/org/webrtc/examples/androidnativeapi/CallClient.java
>new file mode 100644
>index 00000000000..7369a1286d0
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidnativeapi/java/org/webrtc/examples/androidnativeapi/CallClient.java
>@@ -0,0 +1,72 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc.examples.androidnativeapi;
>+
>+import android.content.Context;
>+import android.os.Handler;
>+import android.os.HandlerThread;
>+import org.webrtc.CapturerObserver;
>+import org.webrtc.SurfaceTextureHelper;
>+import org.webrtc.VideoCapturer;
>+import org.webrtc.VideoSink;
>+
>+public class CallClient {
>+  private static final String TAG = "CallClient";
>+  private static final int CAPTURE_WIDTH = 640;
>+  private static final int CAPTURE_HEIGHT = 480;
>+  private static final int CAPTURE_FPS = 30;
>+
>+  private final Context applicationContext;
>+  private final HandlerThread thread;
>+  private final Handler handler;
>+
>+  private long nativeClient;
>+  private SurfaceTextureHelper surfaceTextureHelper;
>+  private VideoCapturer videoCapturer;
>+
>+  public CallClient(Context applicationContext) {
>+    this.applicationContext = applicationContext;
>+    thread = new HandlerThread(TAG + "Thread");
>+    thread.start();
>+    handler = new Handler(thread.getLooper());
>+    handler.post(() -> { nativeClient = nativeCreateClient(); });
>+  }
>+
>+  public void call(VideoSink localSink, VideoSink remoteSink, VideoCapturer videoCapturer,
>+      SurfaceTextureHelper videoCapturerSurfaceTextureHelper) {
>+    handler.post(() -> {
>+      nativeCall(nativeClient, localSink, remoteSink);
>+      videoCapturer.initialize(videoCapturerSurfaceTextureHelper, applicationContext,
>+          nativeGetJavaVideoCapturerObserver(nativeClient));
>+      videoCapturer.startCapture(CAPTURE_WIDTH, CAPTURE_HEIGHT, CAPTURE_FPS);
>+    });
>+  }
>+
>+  public void hangup() {
>+    handler.post(() -> { nativeHangup(nativeClient); });
>+  }
>+
>+  public void close() {
>+    handler.post(() -> {
>+      nativeDelete(nativeClient);
>+      nativeClient = 0;
>+    });
>+    thread.quitSafely();
>+  }
>+
>+  private static native long nativeCreateClient();
>+  private static native void nativeCall(
>+      long nativeAndroidCallClient, VideoSink localSink, VideoSink remoteSink);
>+  private static native void nativeHangup(long nativeAndroidCallClient);
>+  private static native void nativeDelete(long nativeAndroidCallClient);
>+  private static native CapturerObserver nativeGetJavaVideoCapturerObserver(
>+      long nativeAndroidCallClient);
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidnativeapi/java/org/webrtc/examples/androidnativeapi/MainActivity.java b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidnativeapi/java/org/webrtc/examples/androidnativeapi/MainActivity.java
>new file mode 100644
>index 00000000000..801e238782d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidnativeapi/java/org/webrtc/examples/androidnativeapi/MainActivity.java
>@@ -0,0 +1,120 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc.examples.androidnativeapi;
>+
>+import android.app.Activity;
>+import android.content.Context;
>+import android.os.Bundle;
>+import android.widget.Button;
>+import javax.annotation.Nullable;
>+import org.webrtc.Camera1Enumerator;
>+import org.webrtc.Camera2Enumerator;
>+import org.webrtc.CameraEnumerator;
>+import org.webrtc.ContextUtils;
>+import org.webrtc.EglBase;
>+import org.webrtc.GlRectDrawer;
>+import org.webrtc.SurfaceTextureHelper;
>+import org.webrtc.SurfaceViewRenderer;
>+import org.webrtc.VideoCapturer;
>+
>+public class MainActivity extends Activity {
>+  private @Nullable CallClient callClient;
>+  private @Nullable EglBase eglBase;
>+  private @Nullable SurfaceViewRenderer localRenderer;
>+  private @Nullable SurfaceViewRenderer remoteRenderer;
>+  private @Nullable SurfaceTextureHelper videoCapturerSurfaceTextureHelper;
>+  private @Nullable VideoCapturer videoCapturer;
>+
>+  @Override
>+  protected void onCreate(Bundle savedInstance) {
>+    ContextUtils.initialize(getApplicationContext());
>+
>+    super.onCreate(savedInstance);
>+    setContentView(R.layout.activity_main);
>+
>+    System.loadLibrary("examples_androidnativeapi_jni");
>+    callClient = new CallClient(getApplicationContext());
>+
>+    Button callButton = (Button) findViewById(R.id.call_button);
>+    callButton.setOnClickListener((view) -> {
>+      if (videoCapturer == null) {
>+        videoCapturer = createVideoCapturer(getApplicationContext());
>+      }
>+      callClient.call(
>+          localRenderer, remoteRenderer, videoCapturer, videoCapturerSurfaceTextureHelper);
>+    });
>+
>+    Button hangupButton = (Button) findViewById(R.id.hangup_button);
>+    hangupButton.setOnClickListener((view) -> { hangup(); });
>+  }
>+
>+  @Override
>+  protected void onStart() {
>+    super.onStart();
>+
>+    eglBase = EglBase.create(null /* sharedContext */, EglBase.CONFIG_PLAIN);
>+    localRenderer = (SurfaceViewRenderer) findViewById(R.id.local_renderer);
>+    remoteRenderer = (SurfaceViewRenderer) findViewById(R.id.remote_renderer);
>+
>+    localRenderer.init(eglBase.getEglBaseContext(), null /* rendererEvents */, EglBase.CONFIG_PLAIN,
>+        new GlRectDrawer());
>+    remoteRenderer.init(eglBase.getEglBaseContext(), null /* rendererEvents */,
>+        EglBase.CONFIG_PLAIN, new GlRectDrawer());
>+
>+    videoCapturerSurfaceTextureHelper =
>+        SurfaceTextureHelper.create("VideoCapturerThread", eglBase.getEglBaseContext());
>+  }
>+
>+  @Override
>+  protected void onStop() {
>+    hangup();
>+
>+    localRenderer.release();
>+    remoteRenderer.release();
>+    videoCapturerSurfaceTextureHelper.dispose();
>+    eglBase.release();
>+
>+    localRenderer = null;
>+    remoteRenderer = null;
>+    videoCapturerSurfaceTextureHelper = null;
>+    eglBase = null;
>+
>+    super.onStop();
>+  }
>+
>+  @Override
>+  protected void onDestroy() {
>+    callClient.close();
>+    callClient = null;
>+
>+    super.onDestroy();
>+  }
>+
>+  private void hangup() {
>+    if (videoCapturer != null) {
>+      try {
>+        videoCapturer.stopCapture();
>+      } catch (InterruptedException e) {
>+        throw new RuntimeException(e);
>+      }
>+      videoCapturer.dispose();
>+      videoCapturer = null;
>+    }
>+    callClient.hangup();
>+  }
>+
>+  private static VideoCapturer createVideoCapturer(Context context) {
>+    CameraEnumerator enumerator = Camera2Enumerator.isSupported(context)
>+        ? new Camera2Enumerator(context)
>+        : new Camera1Enumerator();
>+    return enumerator.createCapturer(enumerator.getDeviceNames()[0], null /* eventsHandler */);
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidnativeapi/jni/androidcallclient.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidnativeapi/jni/androidcallclient.cc
>new file mode 100644
>index 00000000000..005f369a6d1
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidnativeapi/jni/androidcallclient.cc
>@@ -0,0 +1,290 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "examples/androidnativeapi/jni/androidcallclient.h"
>+
>+#include <utility>
>+
>+#include "absl/memory/memory.h"
>+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
>+#include "api/audio_codecs/builtin_audio_encoder_factory.h"
>+#include "api/peerconnectioninterface.h"
>+#include "examples/androidnativeapi/generated_jni/jni/CallClient_jni.h"
>+#include "media/engine/internaldecoderfactory.h"
>+#include "media/engine/internalencoderfactory.h"
>+#include "media/engine/webrtcmediaengine.h"
>+#include "modules/audio_processing/include/audio_processing.h"
>+#include "sdk/android/native_api/jni/java_types.h"
>+#include "sdk/android/native_api/video/wrapper.h"
>+
>+namespace webrtc_examples {
>+
>+class AndroidCallClient::PCObserver : public webrtc::PeerConnectionObserver {
>+ public:
>+  explicit PCObserver(AndroidCallClient* client);
>+
>+  void OnSignalingChange(
>+      webrtc::PeerConnectionInterface::SignalingState new_state) override;
>+  void OnDataChannel(
>+      rtc::scoped_refptr<webrtc::DataChannelInterface> data_channel) override;
>+  void OnRenegotiationNeeded() override;
>+  void OnIceConnectionChange(
>+      webrtc::PeerConnectionInterface::IceConnectionState new_state) override;
>+  void OnIceGatheringChange(
>+      webrtc::PeerConnectionInterface::IceGatheringState new_state) override;
>+  void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override;
>+
>+ private:
>+  const AndroidCallClient* client_;
>+};
>+
>+namespace {
>+
>+class CreateOfferObserver : public webrtc::CreateSessionDescriptionObserver {
>+ public:
>+  explicit CreateOfferObserver(
>+      rtc::scoped_refptr<webrtc::PeerConnectionInterface> pc);
>+
>+  void OnSuccess(webrtc::SessionDescriptionInterface* desc) override;
>+  void OnFailure(webrtc::RTCError error) override;
>+
>+ private:
>+  const rtc::scoped_refptr<webrtc::PeerConnectionInterface> pc_;
>+};
>+
>+class SetRemoteSessionDescriptionObserver
>+    : public webrtc::SetRemoteDescriptionObserverInterface {
>+ public:
>+  void OnSetRemoteDescriptionComplete(webrtc::RTCError error) override;
>+};
>+
>+class SetLocalSessionDescriptionObserver
>+    : public webrtc::SetSessionDescriptionObserver {
>+ public:
>+  void OnSuccess() override;
>+  void OnFailure(webrtc::RTCError error) override;
>+};
>+
>+}  // namespace
>+
>+AndroidCallClient::AndroidCallClient()
>+    : call_started_(false), pc_observer_(absl::make_unique<PCObserver>(this)) {
>+  thread_checker_.DetachFromThread();
>+  CreatePeerConnectionFactory();
>+}
>+
>+AndroidCallClient::~AndroidCallClient() = default;
>+
>+void AndroidCallClient::Call(JNIEnv* env,
>+                             const webrtc::JavaRef<jobject>& cls,
>+                             const webrtc::JavaRef<jobject>& local_sink,
>+                             const webrtc::JavaRef<jobject>& remote_sink) {
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+
>+  rtc::CritScope lock(&pc_mutex_);
>+  if (call_started_) {
>+    RTC_LOG(LS_WARNING) << "Call already started.";
>+    return;
>+  }
>+  call_started_ = true;
>+
>+  local_sink_ = webrtc::JavaToNativeVideoSink(env, local_sink.obj());
>+  remote_sink_ = webrtc::JavaToNativeVideoSink(env, remote_sink.obj());
>+
>+  video_source_ = webrtc::CreateJavaVideoSource(env, signaling_thread_.get(),
>+                                                false /* is_screencast */);
>+
>+  CreatePeerConnection();
>+  Connect();
>+}
>+
>+void AndroidCallClient::Hangup(JNIEnv* env,
>+                               const webrtc::JavaRef<jobject>& cls) {
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+
>+  call_started_ = false;
>+
>+  {
>+    rtc::CritScope lock(&pc_mutex_);
>+    if (pc_ != nullptr) {
>+      pc_->Close();
>+      pc_ = nullptr;
>+    }
>+  }
>+
>+  local_sink_ = nullptr;
>+  remote_sink_ = nullptr;
>+  video_source_ = nullptr;
>+}
>+
>+void AndroidCallClient::Delete(JNIEnv* env,
>+                               const webrtc::JavaRef<jobject>& cls) {
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+
>+  delete this;
>+}
>+
>+webrtc::ScopedJavaLocalRef<jobject>
>+AndroidCallClient::GetJavaVideoCapturerObserver(
>+    JNIEnv* env,
>+    const webrtc::JavaRef<jobject>& cls) {
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+
>+  return video_source_->GetJavaVideoCapturerObserver(env);
>+}
>+
>+void AndroidCallClient::CreatePeerConnectionFactory() {
>+  network_thread_ = rtc::Thread::CreateWithSocketServer();
>+  network_thread_->SetName("network_thread", nullptr);
>+  RTC_CHECK(network_thread_->Start()) << "Failed to start thread";
>+
>+  worker_thread_ = rtc::Thread::Create();
>+  worker_thread_->SetName("worker_thread", nullptr);
>+  RTC_CHECK(worker_thread_->Start()) << "Failed to start thread";
>+
>+  signaling_thread_ = rtc::Thread::Create();
>+  signaling_thread_->SetName("signaling_thread", nullptr);
>+  RTC_CHECK(signaling_thread_->Start()) << "Failed to start thread";
>+
>+  std::unique_ptr<cricket::MediaEngineInterface> media_engine =
>+      cricket::WebRtcMediaEngineFactory::Create(
>+          nullptr /* adm */, webrtc::CreateBuiltinAudioEncoderFactory(),
>+          webrtc::CreateBuiltinAudioDecoderFactory(),
>+          absl::make_unique<webrtc::InternalEncoderFactory>(),
>+          absl::make_unique<webrtc::InternalDecoderFactory>(),
>+          nullptr /* audio_mixer */, webrtc::AudioProcessingBuilder().Create());
>+  RTC_LOG(LS_INFO) << "Media engine created: " << media_engine.get();
>+
>+  pcf_ = CreateModularPeerConnectionFactory(
>+      network_thread_.get(), worker_thread_.get(), signaling_thread_.get(),
>+      std::move(media_engine), webrtc::CreateCallFactory(),
>+      webrtc::CreateRtcEventLogFactory());
>+  RTC_LOG(LS_INFO) << "PeerConnectionFactory created: " << pcf_;
>+}
>+
>+void AndroidCallClient::CreatePeerConnection() {
>+  rtc::CritScope lock(&pc_mutex_);
>+  webrtc::PeerConnectionInterface::RTCConfiguration config;
>+  config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan;
>+  // DTLS SRTP has to be disabled for loopback to work.
>+  config.enable_dtls_srtp = false;
>+  pc_ = pcf_->CreatePeerConnection(config, nullptr /* port_allocator */,
>+                                   nullptr /* cert_generator */,
>+                                   pc_observer_.get());
>+  RTC_LOG(LS_INFO) << "PeerConnection created: " << pc_;
>+
>+  rtc::scoped_refptr<webrtc::VideoTrackInterface> local_video_track =
>+      pcf_->CreateVideoTrack("video", video_source_);
>+  local_video_track->AddOrUpdateSink(local_sink_.get(), rtc::VideoSinkWants());
>+  pc_->AddTransceiver(local_video_track);
>+  RTC_LOG(LS_INFO) << "Local video sink set up: " << local_video_track;
>+
>+  for (const rtc::scoped_refptr<webrtc::RtpTransceiverInterface>& tranceiver :
>+       pc_->GetTransceivers()) {
>+    rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> track =
>+        tranceiver->receiver()->track();
>+    if (track &&
>+        track->kind() == webrtc::MediaStreamTrackInterface::kVideoKind) {
>+      static_cast<webrtc::VideoTrackInterface*>(track.get())
>+          ->AddOrUpdateSink(remote_sink_.get(), rtc::VideoSinkWants());
>+      RTC_LOG(LS_INFO) << "Remote video sink set up: " << track;
>+      break;
>+    }
>+  }
>+}
>+
>+void AndroidCallClient::Connect() {
>+  rtc::CritScope lock(&pc_mutex_);
>+  pc_->CreateOffer(new rtc::RefCountedObject<CreateOfferObserver>(pc_),
>+                   webrtc::PeerConnectionInterface::RTCOfferAnswerOptions());
>+}
>+
>+AndroidCallClient::PCObserver::PCObserver(AndroidCallClient* client)
>+    : client_(client) {}
>+
>+void AndroidCallClient::PCObserver::OnSignalingChange(
>+    webrtc::PeerConnectionInterface::SignalingState new_state) {
>+  RTC_LOG(LS_INFO) << "OnSignalingChange: " << new_state;
>+}
>+
>+void AndroidCallClient::PCObserver::OnDataChannel(
>+    rtc::scoped_refptr<webrtc::DataChannelInterface> data_channel) {
>+  RTC_LOG(LS_INFO) << "OnDataChannel";
>+}
>+
>+void AndroidCallClient::PCObserver::OnRenegotiationNeeded() {
>+  RTC_LOG(LS_INFO) << "OnRenegotiationNeeded";
>+}
>+
>+void AndroidCallClient::PCObserver::OnIceConnectionChange(
>+    webrtc::PeerConnectionInterface::IceConnectionState new_state) {
>+  RTC_LOG(LS_INFO) << "OnIceConnectionChange: " << new_state;
>+}
>+
>+void AndroidCallClient::PCObserver::OnIceGatheringChange(
>+    webrtc::PeerConnectionInterface::IceGatheringState new_state) {
>+  RTC_LOG(LS_INFO) << "OnIceGatheringChange: " << new_state;
>+}
>+
>+void AndroidCallClient::PCObserver::OnIceCandidate(
>+    const webrtc::IceCandidateInterface* candidate) {
>+  RTC_LOG(LS_INFO) << "OnIceCandidate: " << candidate->server_url();
>+  rtc::CritScope lock(&client_->pc_mutex_);
>+  RTC_DCHECK(client_->pc_ != nullptr);
>+  client_->pc_->AddIceCandidate(candidate);
>+}
>+
>+CreateOfferObserver::CreateOfferObserver(
>+    rtc::scoped_refptr<webrtc::PeerConnectionInterface> pc)
>+    : pc_(pc) {}
>+
>+void CreateOfferObserver::OnSuccess(webrtc::SessionDescriptionInterface* desc) {
>+  std::string sdp;
>+  desc->ToString(&sdp);
>+  RTC_LOG(LS_INFO) << "Created offer: " << sdp;
>+
>+  // Ownership of desc was transferred to us, now we transfer it forward.
>+  pc_->SetLocalDescription(
>+      new rtc::RefCountedObject<SetLocalSessionDescriptionObserver>(), desc);
>+
>+  // Generate a fake answer.
>+  std::unique_ptr<webrtc::SessionDescriptionInterface> answer(
>+      webrtc::CreateSessionDescription(webrtc::SdpType::kAnswer, sdp));
>+  pc_->SetRemoteDescription(
>+      std::move(answer),
>+      new rtc::RefCountedObject<SetRemoteSessionDescriptionObserver>());
>+}
>+
>+void CreateOfferObserver::OnFailure(webrtc::RTCError error) {
>+  RTC_LOG(LS_INFO) << "Failed to create offer: " << ToString(error.type())
>+                   << ": " << error.message();
>+}
>+
>+void SetRemoteSessionDescriptionObserver::OnSetRemoteDescriptionComplete(
>+    webrtc::RTCError error) {
>+  RTC_LOG(LS_INFO) << "Set remote description: " << error.message();
>+}
>+
>+void SetLocalSessionDescriptionObserver::OnSuccess() {
>+  RTC_LOG(LS_INFO) << "Set local description success!";
>+}
>+
>+void SetLocalSessionDescriptionObserver::OnFailure(webrtc::RTCError error) {
>+  RTC_LOG(LS_INFO) << "Set local description failure: "
>+                   << ToString(error.type()) << ": " << error.message();
>+}
>+
>+static jlong JNI_CallClient_CreateClient(
>+    JNIEnv* env,
>+    const webrtc::JavaParamRef<jclass>& cls) {
>+  return webrtc::NativeToJavaPointer(new webrtc_examples::AndroidCallClient());
>+}
>+
>+}  // namespace webrtc_examples
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidnativeapi/jni/androidcallclient.h b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidnativeapi/jni/androidcallclient.h
>new file mode 100644
>index 00000000000..d456e9c10a9
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidnativeapi/jni/androidcallclient.h
>@@ -0,0 +1,79 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef EXAMPLES_ANDROIDNATIVEAPI_JNI_ANDROIDCALLCLIENT_H_
>+#define EXAMPLES_ANDROIDNATIVEAPI_JNI_ANDROIDCALLCLIENT_H_
>+
>+#include <jni.h>
>+
>+#include <memory>
>+#include <string>
>+
>+#include "api/peerconnectioninterface.h"
>+#include "rtc_base/criticalsection.h"
>+#include "rtc_base/scoped_ref_ptr.h"
>+#include "rtc_base/thread_checker.h"
>+#include "sdk/android/native_api/jni/scoped_java_ref.h"
>+#include "sdk/android/native_api/video/videosource.h"
>+
>+namespace webrtc_examples {
>+
>+class AndroidCallClient {
>+ public:
>+  AndroidCallClient();
>+  ~AndroidCallClient();
>+
>+  void Call(JNIEnv* env,
>+            const webrtc::JavaRef<jobject>& cls,
>+            const webrtc::JavaRef<jobject>& local_sink,
>+            const webrtc::JavaRef<jobject>& remote_sink);
>+  void Hangup(JNIEnv* env, const webrtc::JavaRef<jobject>& cls);
>+  // A helper method for Java code to delete this object. Calls delete this.
>+  void Delete(JNIEnv* env, const webrtc::JavaRef<jobject>& cls);
>+
>+  webrtc::ScopedJavaLocalRef<jobject> GetJavaVideoCapturerObserver(
>+      JNIEnv* env,
>+      const webrtc::JavaRef<jobject>& cls);
>+
>+ private:
>+  class PCObserver;
>+
>+  void CreatePeerConnectionFactory() RTC_RUN_ON(thread_checker_);
>+  void CreatePeerConnection() RTC_RUN_ON(thread_checker_);
>+  void Connect() RTC_RUN_ON(thread_checker_);
>+
>+  rtc::ThreadChecker thread_checker_;
>+
>+  bool call_started_ RTC_GUARDED_BY(thread_checker_);
>+
>+  const std::unique_ptr<PCObserver> pc_observer_;
>+
>+  rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> pcf_
>+      RTC_GUARDED_BY(thread_checker_);
>+  std::unique_ptr<rtc::Thread> network_thread_ RTC_GUARDED_BY(thread_checker_);
>+  std::unique_ptr<rtc::Thread> worker_thread_ RTC_GUARDED_BY(thread_checker_);
>+  std::unique_ptr<rtc::Thread> signaling_thread_
>+      RTC_GUARDED_BY(thread_checker_);
>+
>+  std::unique_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>> local_sink_
>+      RTC_GUARDED_BY(thread_checker_);
>+  std::unique_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>> remote_sink_
>+      RTC_GUARDED_BY(thread_checker_);
>+  rtc::scoped_refptr<webrtc::JavaVideoTrackSourceInterface> video_source_
>+      RTC_GUARDED_BY(thread_checker_);
>+
>+  rtc::CriticalSection pc_mutex_;
>+  rtc::scoped_refptr<webrtc::PeerConnectionInterface> pc_
>+      RTC_GUARDED_BY(pc_mutex_);
>+};
>+
>+}  // namespace webrtc_examples
>+
>+#endif  // EXAMPLES_ANDROIDNATIVEAPI_JNI_ANDROIDCALLCLIENT_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidnativeapi/jni/onload.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidnativeapi/jni/onload.cc
>new file mode 100644
>index 00000000000..4b4b5d960a2
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidnativeapi/jni/onload.cc
>@@ -0,0 +1,30 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <jni.h>
>+
>+#include "modules/utility/include/jvm_android.h"
>+#include "rtc_base/ssladapter.h"
>+#include "sdk/android/native_api/base/init.h"
>+
>+namespace webrtc_examples {
>+
>+extern "C" jint JNIEXPORT JNICALL JNI_OnLoad(JavaVM* jvm, void* reserved) {
>+  webrtc::InitAndroid(jvm);
>+  webrtc::JVM::Initialize(jvm);
>+  RTC_CHECK(rtc::InitializeSSL()) << "Failed to InitializeSSL()";
>+  return JNI_VERSION_1_6;
>+}
>+
>+extern "C" void JNIEXPORT JNICALL JNI_OnUnLoad(JavaVM* jvm, void* reserved) {
>+  RTC_CHECK(rtc::CleanupSSL()) << "Failed to CleanupSSL()";
>+}
>+
>+}  // namespace webrtc_examples
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidnativeapi/res/layout/activity_main.xml b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidnativeapi/res/layout/activity_main.xml
>new file mode 100644
>index 00000000000..ac8037320f6
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidnativeapi/res/layout/activity_main.xml
>@@ -0,0 +1,52 @@
>+<?xml version="1.0" encoding="utf-8"?>
>+<LinearLayout xmlns:android="http://schemas.android.com/apk/res/android"
>+  xmlns:app="http://schemas.android.com/apk/res-auto"
>+  xmlns:tools="http://schemas.android.com/tools"
>+  android:orientation="vertical"
>+  android:layout_width="match_parent"
>+  android:layout_height="match_parent"
>+  android:padding="8dp"
>+  tools:context="org.webrtc.examples.androidnativeapi.MainActivity">
>+
>+  <org.webrtc.SurfaceViewRenderer
>+    android:id="@+id/local_renderer"
>+    android:layout_width="match_parent"
>+    android:layout_height="0dp"
>+    android:layout_weight="1"
>+    android:layout_margin="8dp"/>
>+
>+  <org.webrtc.SurfaceViewRenderer
>+    android:id="@+id/remote_renderer"
>+    android:layout_width="match_parent"
>+    android:layout_height="0dp"
>+    android:layout_weight="1"
>+    android:layout_margin="8dp"/>
>+
>+
>+  <LinearLayout
>+    android:orientation="horizontal"
>+    android:layout_width="match_parent"
>+    android:layout_height="48dp"
>+    style="?android:attr/buttonBarStyle">
>+
>+    <Button
>+      android:id="@+id/call_button"
>+      android:text="@string/call_button"
>+      style="?android:attr/buttonBarButtonStyle"
>+      android:layout_width="0dp"
>+      android:layout_height="48dp"
>+      android:layout_weight="1"
>+      android:layout_margin="8dp"/>
>+
>+    <Button
>+      android:id="@+id/hangup_button"
>+      android:text="@string/hangup_button"
>+      style="?android:attr/buttonBarButtonStyle"
>+      android:layout_width="0dp"
>+      android:layout_height="48dp"
>+      android:layout_weight="1"
>+      android:layout_margin="8dp"/>
>+
>+  </LinearLayout>
>+
>+</LinearLayout>
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidnativeapi/res/values/strings.xml b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidnativeapi/res/values/strings.xml
>new file mode 100644
>index 00000000000..a00920c92b6
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidnativeapi/res/values/strings.xml
>@@ -0,0 +1,5 @@
>+<resources>
>+  <string name="app_name">androidnativeapi</string>
>+  <string name="call_button">Call</string>
>+  <string name="hangup_button">Hangup</string>
>+</resources>
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidtests/README b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidtests/README
>index 49ae6c81d69..0701b0e8966 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidtests/README
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidtests/README
>@@ -4,11 +4,11 @@ Example of building & using the app:
> 
> - Build Android AppRTCMobile and AppRTCMobile unit test:
> cd <path/to/webrtc>/src
>-ninja -C out/Debug AppRTCMobileTest
>+ninja -C out/Debug AppRTCMobile_test_apk
> 
> - Install AppRTCMobile and AppRTCMobileTest:
> adb install -r out/Debug/apks/AppRTCMobile.apk
> adb install -r out/Debug/apks/AppRTCMobileTest.apk
> 
> - Run unit tests:
>-adb shell am instrument -w org.appspot.apprtc.test/android.test.InstrumentationTestRunner
>\ No newline at end of file
>+adb shell am instrument -w org.appspot.apprtc.test/android.test.InstrumentationTestRunner
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidtests/src/org/appspot/apprtc/test/PeerConnectionClientTest.java b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidtests/src/org/appspot/apprtc/test/PeerConnectionClientTest.java
>index 56eb3f72843..ec06bb9dde5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidtests/src/org/appspot/apprtc/test/PeerConnectionClientTest.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidtests/src/org/appspot/apprtc/test/PeerConnectionClientTest.java
>@@ -35,15 +35,14 @@ import org.junit.runner.RunWith;
> import org.webrtc.Camera1Enumerator;
> import org.webrtc.Camera2Enumerator;
> import org.webrtc.CameraEnumerator;
>+import org.webrtc.EglBase;
> import org.webrtc.IceCandidate;
>-import org.webrtc.MediaCodecVideoEncoder;
> import org.webrtc.PeerConnection;
> import org.webrtc.PeerConnectionFactory;
> import org.webrtc.SessionDescription;
> import org.webrtc.StatsReport;
> import org.webrtc.VideoCapturer;
> import org.webrtc.VideoFrame;
>-import org.webrtc.VideoRenderer;
> import org.webrtc.VideoSink;
> 
> @RunWith(AndroidJUnit4.class)
>@@ -85,53 +84,6 @@ public class PeerConnectionClientTest implements PeerConnectionEvents {
>   private final Object iceConnectedEvent = new Object();
>   private final Object closeEvent = new Object();
> 
>-  // Mock renderer implementation.
>-  private static class MockRenderer implements VideoRenderer.Callbacks {
>-    // These are protected by 'this' since we gets called from worker threads.
>-    private String rendererName;
>-    private boolean renderFrameCalled = false;
>-
>-    // Thread-safe in itself.
>-    private CountDownLatch doneRendering;
>-
>-    public MockRenderer(int expectedFrames, String rendererName) {
>-      this.rendererName = rendererName;
>-      reset(expectedFrames);
>-    }
>-
>-    // Resets render to wait for new amount of video frames.
>-    // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
>-    @SuppressWarnings("NoSynchronizedMethodCheck")
>-    public synchronized void reset(int expectedFrames) {
>-      renderFrameCalled = false;
>-      doneRendering = new CountDownLatch(expectedFrames);
>-    }
>-
>-    @Override
>-    // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
>-    @SuppressWarnings("NoSynchronizedMethodCheck")
>-    public synchronized void renderFrame(VideoRenderer.I420Frame frame) {
>-      if (!renderFrameCalled) {
>-        if (rendererName != null) {
>-          Log.d(TAG, rendererName + " render frame: " + frame.rotatedWidth() + " x "
>-                  + frame.rotatedHeight());
>-        } else {
>-          Log.d(TAG, "Render frame: " + frame.rotatedWidth() + " x " + frame.rotatedHeight());
>-        }
>-      }
>-      renderFrameCalled = true;
>-      VideoRenderer.renderFrameDone(frame);
>-      doneRendering.countDown();
>-    }
>-
>-    // This method shouldn't hold any locks or touch member variables since it
>-    // blocks.
>-    public boolean waitForFramesRendered(int timeoutMs) throws InterruptedException {
>-      doneRendering.await(timeoutMs, TimeUnit.MILLISECONDS);
>-      return (doneRendering.getCount() <= 0);
>-    }
>-  }
>-
>   // Mock VideoSink implementation.
>   private static class MockSink implements VideoSink {
>     // These are protected by 'this' since we gets called from worker threads.
>@@ -306,22 +258,22 @@ public class PeerConnectionClientTest implements PeerConnectionEvents {
>     }
>   }
> 
>-  PeerConnectionClient createPeerConnectionClient(MockSink localRenderer,
>-      MockRenderer remoteRenderer, PeerConnectionParameters peerConnectionParameters,
>-      VideoCapturer videoCapturer) {
>+  PeerConnectionClient createPeerConnectionClient(MockSink localRenderer, MockSink remoteRenderer,
>+      PeerConnectionParameters peerConnectionParameters, VideoCapturer videoCapturer) {
>     List<PeerConnection.IceServer> iceServers = new ArrayList<>();
>     SignalingParameters signalingParameters =
>         new SignalingParameters(iceServers, true, // iceServers, initiator.
>             null, null, null, // clientId, wssUrl, wssPostUrl.
>             null, null); // offerSdp, iceCandidates.
> 
>-    PeerConnectionClient client = new PeerConnectionClient();
>+    final EglBase eglBase = EglBase.create();
>+    PeerConnectionClient client =
>+        new PeerConnectionClient(InstrumentationRegistry.getTargetContext(), eglBase,
>+            peerConnectionParameters, this /* PeerConnectionEvents */);
>     PeerConnectionFactory.Options options = new PeerConnectionFactory.Options();
>     options.networkIgnoreMask = 0;
>     options.disableNetworkMonitor = true;
>-    client.setPeerConnectionFactoryOptions(options);
>-    client.createPeerConnectionFactory(
>-        InstrumentationRegistry.getTargetContext(), peerConnectionParameters, this);
>+    client.createPeerConnectionFactory(options);
>     client.createPeerConnection(localRenderer, remoteRenderer, videoCapturer, signalingParameters);
>     client.createOffer();
>     return client;
>@@ -344,8 +296,10 @@ public class PeerConnectionClientTest implements PeerConnectionEvents {
>         "OPUS", /* audioCodec */
>         false, /* noAudioProcessing */
>         false, /* aecDump */
>+        false, /* saveInputAudioToFile */
>         false /* useOpenSLES */, false /* disableBuiltInAEC */, false /* disableBuiltInAGC */,
>-        false /* disableBuiltInNS */, false /* enableLevelControl */, false /* disableWebRtcAGC */);
>+        false /* disableBuiltInNS */, false /* disableWebRtcAGC */, false /* enableRtcEventLog */,
>+        false /* useLegacyAudioDevice */, null /* dataChannelParameters */);
>   }
> 
>   private VideoCapturer createCameraCapturer(boolean captureToTexture) {
>@@ -379,8 +333,10 @@ public class PeerConnectionClientTest implements PeerConnectionEvents {
>         "OPUS", /* audioCodec */
>         false, /* noAudioProcessing */
>         false, /* aecDump */
>+        false, /* saveInputAudioToFile */
>         false /* useOpenSLES */, false /* disableBuiltInAEC */, false /* disableBuiltInAGC */,
>-        false /* disableBuiltInNS */, false /* enableLevelControl */, false /* disableWebRtcAGC */);
>+        false /* disableBuiltInNS */, false /* disableWebRtcAGC */, false /* enableRtcEventLog */,
>+        false /* useLegacyAudioDevice */, null /* dataChannelParameters */);
>   }
> 
>   @Before
>@@ -398,7 +354,8 @@ public class PeerConnectionClientTest implements PeerConnectionEvents {
>   public void testSetLocalOfferMakesVideoFlowLocally() throws InterruptedException {
>     Log.d(TAG, "testSetLocalOfferMakesVideoFlowLocally");
>     MockSink localRenderer = new MockSink(EXPECTED_VIDEO_FRAMES, LOCAL_RENDERER_NAME);
>-    pcClient = createPeerConnectionClient(localRenderer, new MockRenderer(0, null),
>+    pcClient = createPeerConnectionClient(localRenderer,
>+        new MockSink(/* expectedFrames= */ 0, /* rendererName= */ null),
>         createParametersForVideoCall(VIDEO_CODEC_VP8),
>         createCameraCapturer(false /* captureToTexture */));
> 
>@@ -420,11 +377,11 @@ public class PeerConnectionClientTest implements PeerConnectionEvents {
>       boolean decodeToTexture) throws InterruptedException {
>     loopback = true;
>     MockSink localRenderer = null;
>-    MockRenderer remoteRenderer = null;
>+    MockSink remoteRenderer = null;
>     if (parameters.videoCallEnabled) {
>       Log.d(TAG, "testLoopback for video " + parameters.videoCodec);
>       localRenderer = new MockSink(EXPECTED_VIDEO_FRAMES, LOCAL_RENDERER_NAME);
>-      remoteRenderer = new MockRenderer(EXPECTED_VIDEO_FRAMES, REMOTE_RENDERER_NAME);
>+      remoteRenderer = new MockSink(EXPECTED_VIDEO_FRAMES, REMOTE_RENDERER_NAME);
>     } else {
>       Log.d(TAG, "testLoopback for audio.");
>     }
>@@ -523,12 +480,6 @@ public class PeerConnectionClientTest implements PeerConnectionEvents {
>       Log.i(TAG, "Encode to textures is not supported. Requires SDK version 19");
>       return;
>     }
>-    // TODO(perkj): If we can always capture to textures, there is no need to check if the
>-    // hardware encoder supports to encode from a texture.
>-    if (!MediaCodecVideoEncoder.isVp8HwSupportedUsingTextures()) {
>-      Log.i(TAG, "VP8 encode to textures is not supported.");
>-      return;
>-    }
>     doLoopbackTest(createParametersForVideoCall(VIDEO_CODEC_VP8),
>         createCameraCapturer(true /* captureToTexture */), true /* decodeToTexture */);
>   }
>@@ -540,12 +491,6 @@ public class PeerConnectionClientTest implements PeerConnectionEvents {
>       Log.i(TAG, "Encode to textures is not supported. Requires KITKAT");
>       return;
>     }
>-    // TODO(perkj): If we can always capture to textures, there is no need to check if the
>-    // hardware encoder supports to encode from a texture.
>-    if (!MediaCodecVideoEncoder.isH264HwSupportedUsingTextures()) {
>-      Log.i(TAG, "H264 encode to textures is not supported.");
>-      return;
>-    }
>     doLoopbackTest(createParametersForVideoCall(VIDEO_CODEC_H264),
>         createCameraCapturer(true /* captureToTexture */), true /* decodeToTexture */);
>   }
>@@ -559,7 +504,7 @@ public class PeerConnectionClientTest implements PeerConnectionEvents {
>     loopback = true;
> 
>     MockSink localRenderer = new MockSink(EXPECTED_VIDEO_FRAMES, LOCAL_RENDERER_NAME);
>-    MockRenderer remoteRenderer = new MockRenderer(EXPECTED_VIDEO_FRAMES, REMOTE_RENDERER_NAME);
>+    MockSink remoteRenderer = new MockSink(EXPECTED_VIDEO_FRAMES, REMOTE_RENDERER_NAME);
> 
>     pcClient = createPeerConnectionClient(localRenderer, remoteRenderer,
>         createParametersForVideoCall(VIDEO_CODEC_VP8),
>@@ -607,7 +552,7 @@ public class PeerConnectionClientTest implements PeerConnectionEvents {
>     loopback = true;
> 
>     MockSink localRenderer = new MockSink(EXPECTED_VIDEO_FRAMES, LOCAL_RENDERER_NAME);
>-    MockRenderer remoteRenderer = new MockRenderer(EXPECTED_VIDEO_FRAMES, REMOTE_RENDERER_NAME);
>+    MockSink remoteRenderer = new MockSink(EXPECTED_VIDEO_FRAMES, REMOTE_RENDERER_NAME);
> 
>     pcClient = createPeerConnectionClient(localRenderer, remoteRenderer,
>         createParametersForVideoCall(VIDEO_CODEC_VP8),
>@@ -656,7 +601,7 @@ public class PeerConnectionClientTest implements PeerConnectionEvents {
>     loopback = true;
> 
>     MockSink localRenderer = new MockSink(EXPECTED_VIDEO_FRAMES, LOCAL_RENDERER_NAME);
>-    MockRenderer remoteRenderer = new MockRenderer(EXPECTED_VIDEO_FRAMES, REMOTE_RENDERER_NAME);
>+    MockSink remoteRenderer = new MockSink(EXPECTED_VIDEO_FRAMES, REMOTE_RENDERER_NAME);
> 
>     pcClient = createPeerConnectionClient(localRenderer, remoteRenderer,
>         createParametersForVideoCall(VIDEO_CODEC_VP8),
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidtests/video_quality_loopback_test.py b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidtests/video_quality_loopback_test.py
>index 6991a0a57b6..225f885273c 100755
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidtests/video_quality_loopback_test.py
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/androidtests/video_quality_loopback_test.py
>@@ -22,7 +22,6 @@ import argparse
> import json
> import logging
> import os
>-import shutil
> import subprocess
> import sys
> import tempfile
>@@ -31,11 +30,15 @@ import time
> 
> SCRIPT_DIR = os.path.dirname(os.path.abspath(__file__))
> SRC_DIR = os.path.normpath(os.path.join(SCRIPT_DIR, os.pardir, os.pardir))
>+RTC_TOOLS_DIR = os.path.join(SRC_DIR, 'rtc_tools', 'testing')
>+TOOLCHAIN_DIR = os.path.join(SRC_DIR, 'tools_webrtc', 'video_quality_toolchain',
>+                             'linux')
> BAD_DEVICES_JSON = os.path.join(SRC_DIR,
>                                 os.environ.get('CHROMIUM_OUT_DIR', 'out'),
>                                 'bad_devices.json')
>-sys.path.append(os.path.join(SRC_DIR, 'build'))
>-import find_depot_tools
>+
>+sys.path.append(RTC_TOOLS_DIR)
>+import utils
> 
> 
> class Error(Exception):
>@@ -46,19 +49,19 @@ class VideoQualityTestError(Error):
>   pass
> 
> 
>-def _RunCommand(argv, cwd=SRC_DIR, **kwargs):
>+def _RunCommand(argv, **kwargs):
>   logging.info('Running %r', argv)
>-  subprocess.check_call(argv, cwd=cwd, **kwargs)
>+  subprocess.check_call(argv, **kwargs)
> 
> 
>-def _RunCommandWithOutput(argv, cwd=SRC_DIR, **kwargs):
>+def _RunCommandWithOutput(argv, **kwargs):
>   logging.info('Running %r', argv)
>-  return subprocess.check_output(argv, cwd=cwd, **kwargs)
>+  return subprocess.check_output(argv, **kwargs)
> 
> 
>-def _RunBackgroundCommand(argv, cwd=SRC_DIR):
>+def _RunBackgroundCommand(argv):
>   logging.info('Running %r', argv)
>-  process = subprocess.Popen(argv, cwd=cwd)
>+  process = subprocess.Popen(argv)
>   time.sleep(0.5)
>   status = process.poll()
>   if status:  # is not None or 0
>@@ -66,6 +69,14 @@ def _RunBackgroundCommand(argv, cwd=SRC_DIR):
>   return process
> 
> 
>+def CreateEmptyDir(suggested_dir):
>+  if not suggested_dir:
>+    return tempfile.mkdtemp()
>+  utils.RemoveDirectory(suggested_dir)
>+  os.makedirs(suggested_dir)
>+  return suggested_dir
>+
>+
> def _ParseArgs():
>   parser = argparse.ArgumentParser(description='Start loopback video analysis.')
>   parser.add_argument('build_dir_android',
>@@ -75,50 +86,27 @@ def _ParseArgs():
>   parser.add_argument('--temp_dir',
>       help='A temporary directory to put the output.')
>   parser.add_argument('--adb-path', help='Path to adb binary.', default='adb')
>+  parser.add_argument('--num-retries', default='0',
>+                      help='Number of times to retry the test on Android.')
>+  parser.add_argument('--isolated-script-test-perf-output',
>+      help='Where to store perf results in chartjson format.', default=None)
> 
>-  args = parser.parse_args()
>-  return args
>-
>-
>-def main():
>-  logging.basicConfig(level=logging.INFO)
>-
>-  args = _ParseArgs()
>-
>-  build_dir_android = args.build_dir_android
>-  build_dir_x86 = args.build_dir_x86
>-  temp_dir = args.temp_dir
>-  adb_path = args.adb_path
>-  if not temp_dir:
>-    temp_dir = tempfile.mkdtemp()
>-  else:
>-    if not os.path.exists(temp_dir):
>-      os.makedirs(temp_dir)
>-
>-  if not build_dir_x86:
>-    build_dir_x86 = os.path.join(temp_dir, 'LocalBuild')
>+  args, unknown_args = parser.parse_known_args()
> 
>-    def DepotToolPath(*args):
>-      return os.path.join(find_depot_tools.DEPOT_TOOLS_PATH, *args)
>+  # Ignore Chromium-specific flags
>+  parser = argparse.ArgumentParser()
>+  parser.add_argument('--isolated-script-test-output',
>+                      type=str, default=None)
>+  parser.add_argument('--test-launcher-summary-output',
>+                      type=str, default=None)
> 
>-    _RunCommand([sys.executable, DepotToolPath('gn.py'), 'gen', build_dir_x86])
>-    _RunCommand([DepotToolPath('ninja'), '-C', build_dir_x86,
>-                 'frame_analyzer'])
>+  parser.parse_args(unknown_args)
> 
>-  tools_dir = os.path.join(SRC_DIR, 'tools_webrtc')
>-  toolchain_dir = os.path.join(tools_dir, 'video_quality_toolchain')
>-
>-  # Download ffmpeg and zxing.
>-  download_tools_script = os.path.join(tools_dir, 'download_tools.py')
>-  _RunCommand([sys.executable, download_tools_script, toolchain_dir])
>-
>-  testing_tools_dir = os.path.join(SRC_DIR, 'rtc_tools', 'testing')
>+  return args
> 
>-  # Download, extract and build AppRTC.
>-  setup_apprtc_script = os.path.join(testing_tools_dir, 'setup_apprtc.py')
>-  _RunCommand([sys.executable, setup_apprtc_script, temp_dir])
> 
>-  # Select an Android device in case multiple are connected
>+def SelectAndroidDevice(adb_path):
>+  # Select an Android device in case multiple are connected.
>   try:
>     with open(BAD_DEVICES_JSON) as bad_devices_file:
>       bad_devices = json.load(bad_devices_file)
>@@ -131,91 +119,124 @@ def main():
>     if line.endswith('\tdevice'):
>       android_device = line.split('\t')[0]
>       if android_device not in bad_devices:
>-        break
>-  else:
>-    raise VideoQualityTestError('Cannot find any connected Android device.')
>+        return android_device
>+  raise VideoQualityTestError('Cannot find any connected Android device.')
>+
>+
>+def SetUpTools(android_device, temp_dir, processes):
>+  # Extract AppRTC.
>+  apprtc_archive = os.path.join(RTC_TOOLS_DIR, 'prebuilt_apprtc.zip')
>+  golang_archive = os.path.join(RTC_TOOLS_DIR, 'golang', 'linux', 'go.tar.gz')
>+
>+  utils.UnpackArchiveTo(apprtc_archive, temp_dir)
>+  utils.UnpackArchiveTo(golang_archive, temp_dir)
>+
>+  # Build AppRTC.
>+  build_apprtc_script = os.path.join(RTC_TOOLS_DIR, 'build_apprtc.py')
>+  apprtc_dir = os.path.join(temp_dir, 'apprtc')
>+  go_dir = os.path.join(temp_dir, 'go')
>+  collider_dir = os.path.join(temp_dir, 'collider')
>+
>+  _RunCommand([sys.executable, build_apprtc_script, apprtc_dir, go_dir,
>+               collider_dir])
>+
>+  # Start AppRTC Server.
>+  dev_appserver = os.path.join(temp_dir, 'apprtc', 'temp', 'google-cloud-sdk',
>+                              'bin', 'dev_appserver.py')
>+  appengine_dir = os.path.join(temp_dir, 'apprtc', 'out', 'app_engine')
>+  processes.append(_RunBackgroundCommand([
>+      sys.executable, dev_appserver, appengine_dir, '--port=9999',
>+      '--admin_port=9998', '--skip_sdk_update_check', '--clear_datastore=yes']))
>+
>+  # Start Collider.
>+  collider_path = os.path.join(temp_dir, 'collider', 'collidermain')
>+  processes.append(_RunBackgroundCommand([
>+      collider_path, '-tls=false', '-port=8089',
>+      '-room-server=http://localhost:9999']))
>+
>+  # Start adb reverse forwarder.
>+  reverseforwarder_path = os.path.join(
>+      SRC_DIR, 'build', 'android', 'adb_reverse_forwarder.py')
>+  processes.append(_RunBackgroundCommand([
>+      reverseforwarder_path, '--device', android_device, '9999', '9999', '8089',
>+      '8089']))
>+
>+
>+def RunTest(android_device, adb_path, build_dir, temp_dir, num_retries,
>+            chartjson_result_file):
>+  ffmpeg_path = os.path.join(TOOLCHAIN_DIR, 'ffmpeg')
>+  def ConvertVideo(input_video, output_video):
>+    _RunCommand([ffmpeg_path, '-y', '-i', input_video, output_video])
>+
>+  # Start loopback call and record video.
>+  test_script = os.path.join(
>+      build_dir, 'bin', 'run_AppRTCMobile_stubbed_video_io_test_apk')
>+  _RunCommand([test_script, '--device', android_device,
>+               '--num-retries', num_retries])
>+
>+  # Pull the recorded video.
>+  test_video = os.path.join(temp_dir, 'test_video.y4m')
>+  _RunCommand([adb_path, '-s', android_device,
>+              'pull', '/sdcard/output.y4m', test_video])
>+
>+  # Convert the recorded and reference videos to YUV.
>+  reference_video = os.path.join(SRC_DIR,
>+      'resources', 'reference_video_640x360_30fps.y4m')
>+
>+  test_video_yuv = os.path.join(temp_dir, 'test_video.yuv')
>+  reference_video_yuv = os.path.join(
>+      temp_dir, 'reference_video_640x360_30fps.yuv')
>+
>+  ConvertVideo(test_video, test_video_yuv)
>+  ConvertVideo(reference_video, reference_video_yuv)
>+
>+  # Run comparison script.
>+  compare_script = os.path.join(SRC_DIR, 'rtc_tools', 'compare_videos.py')
>+  frame_analyzer = os.path.join(TOOLCHAIN_DIR, 'frame_analyzer')
>+  zxing_path = os.path.join(TOOLCHAIN_DIR, 'zxing')
>+  stats_file_ref = os.path.join(temp_dir, 'stats_ref.txt')
>+  stats_file_test = os.path.join(temp_dir, 'stats_test.txt')
>+
>+  args = [
>+      '--ref_video', reference_video_yuv,
>+      '--test_video', test_video_yuv,
>+      '--yuv_frame_width', '640',
>+      '--yuv_frame_height', '360',
>+      '--stats_file_ref', stats_file_ref,
>+      '--stats_file_test', stats_file_test,
>+      '--frame_analyzer', frame_analyzer,
>+      '--ffmpeg_path', ffmpeg_path,
>+      '--zxing_path', zxing_path,
>+  ]
>+  if chartjson_result_file:
>+    args.extend(['--chartjson_result_file', chartjson_result_file])
>+
>+  _RunCommand([sys.executable, compare_script] + args)
>+
>+
>+def main():
>+  logging.basicConfig(level=logging.INFO)
>+
>+  args = _ParseArgs()
>+
>+  temp_dir = args.temp_dir
>+  build_dir = args.build_dir_android
>+  adb_path = args.adb_path
> 
>   processes = []
>+  temp_dir = CreateEmptyDir(temp_dir)
>   try:
>-    # Start AppRTC Server
>-    dev_appserver = os.path.join(temp_dir, 'apprtc', 'temp', 'google-cloud-sdk',
>-                                'bin', 'dev_appserver.py')
>-    appengine_dir = os.path.join(temp_dir, 'apprtc', 'out', 'app_engine')
>-    processes.append(_RunBackgroundCommand([
>-        'python', dev_appserver, appengine_dir,
>-        '--port=9999', '--admin_port=9998',
>-        '--skip_sdk_update_check', '--clear_datastore=yes']))
>-
>-    # Start Collider
>-    collider_path = os.path.join(temp_dir, 'collider', 'collidermain')
>-    processes.append(_RunBackgroundCommand([
>-        collider_path, '-tls=false', '-port=8089',
>-        '-room-server=http://localhost:9999']))
>-
>-    # Start adb reverse forwarder
>-    reverseforwarder_path = os.path.join(
>-        SRC_DIR, 'build', 'android', 'adb_reverse_forwarder.py')
>-    processes.append(_RunBackgroundCommand([
>-        reverseforwarder_path, '--device', android_device,
>-        '9999', '9999', '8089', '8089']))
>-
>-    # Run the Espresso code.
>-    test_script = os.path.join(build_dir_android,
>-        'bin', 'run_AppRTCMobileTestStubbedVideoIO')
>-    _RunCommand([test_script, '--device', android_device])
>-
>-    # Pull the output video.
>-    test_video = os.path.join(temp_dir, 'test_video.y4m')
>-    _RunCommand([adb_path, '-s', android_device,
>-                'pull', '/sdcard/output.y4m', test_video])
>-
>-    test_video_yuv = os.path.join(temp_dir, 'test_video.yuv')
>-
>-    ffmpeg_path = os.path.join(toolchain_dir, 'linux', 'ffmpeg')
>-
>-    def ConvertVideo(input_video, output_video):
>-      _RunCommand([ffmpeg_path, '-y', '-i', input_video, output_video])
>-
>-    ConvertVideo(test_video, test_video_yuv)
>-
>-    reference_video = os.path.join(SRC_DIR,
>-        'resources', 'reference_video_640x360_30fps.y4m')
>-
>-    reference_video_yuv = os.path.join(temp_dir,
>-        'reference_video_640x360_30fps.yuv')
>-
>-    ConvertVideo(reference_video, reference_video_yuv)
>-
>-    # Run compare script.
>-    compare_script = os.path.join(SRC_DIR, 'rtc_tools', 'compare_videos.py')
>-    zxing_path = os.path.join(toolchain_dir, 'linux', 'zxing')
>-
>-    # The frame_analyzer binary should be built for local computer and not for
>-    # Android
>-    frame_analyzer = os.path.join(build_dir_x86, 'frame_analyzer')
>-
>-    frame_width = 640
>-    frame_height = 360
>-
>-    stats_file_ref = os.path.join(temp_dir, 'stats_ref.txt')
>-    stats_file_test = os.path.join(temp_dir, 'stats_test.txt')
>-
>-    _RunCommand([
>-        sys.executable, compare_script, '--ref_video', reference_video_yuv,
>-        '--test_video', test_video_yuv, '--yuv_frame_width', str(frame_width),
>-        '--yuv_frame_height', str(frame_height),
>-        '--stats_file_ref', stats_file_ref,
>-        '--stats_file_test', stats_file_test,
>-        '--frame_analyzer', frame_analyzer,
>-        '--ffmpeg_path', ffmpeg_path, '--zxing_path', zxing_path])
>-
>+    android_device = SelectAndroidDevice(adb_path)
>+    SetUpTools(android_device, temp_dir, processes)
>+    RunTest(android_device, adb_path, build_dir, temp_dir, args.num_retries,
>+            args.isolated_script_test_perf_output)
>   finally:
>     for process in processes:
>       if process:
>         process.terminate()
>         process.wait()
> 
>-    shutil.rmtree(temp_dir)
>+    utils.RemoveDirectory(temp_dir)
> 
> 
> if __name__ == '__main__':
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDAppClient+Internal.h b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDAppClient+Internal.h
>index 3cf40400f59..8e96b169090 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDAppClient+Internal.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDAppClient+Internal.h
>@@ -18,8 +18,7 @@
> 
> @class RTCPeerConnectionFactory;
> 
>-@interface ARDAppClient () <ARDSignalingChannelDelegate,
>-  RTCPeerConnectionDelegate>
>+@interface ARDAppClient () <ARDSignalingChannelDelegate, RTCPeerConnectionDelegate>
> 
> // All properties should only be mutated from the main queue.
> @property(nonatomic, strong) id<ARDRoomServerClient> roomServerClient;
>@@ -43,8 +42,7 @@
> @property(nonatomic, strong) NSURL *webSocketRestURL;
> @property(nonatomic, readonly) BOOL isLoopback;
> 
>-@property(nonatomic, strong)
>-    RTCMediaConstraints *defaultPeerConnectionConstraints;
>+@property(nonatomic, strong) RTCMediaConstraints *defaultPeerConnectionConstraints;
> 
> - (instancetype)initWithRoomServerClient:(id<ARDRoomServerClient>)rsClient
>                         signalingChannel:(id<ARDSignalingChannel>)channel
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDAppClient.h b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDAppClient.h
>index 5054c28676f..07cd53d1675 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDAppClient.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDAppClient.h
>@@ -23,6 +23,7 @@ typedef NS_ENUM(NSInteger, ARDAppClientState) {
> 
> @class ARDAppClient;
> @class ARDSettingsModel;
>+@class ARDExternalSampleCapturer;
> @class RTCMediaConstraints;
> @class RTCCameraVideoCapturer;
> @class RTCFileVideoCapturer;
>@@ -31,30 +32,28 @@ typedef NS_ENUM(NSInteger, ARDAppClientState) {
> // main queue.
> @protocol ARDAppClientDelegate <NSObject>
> 
>-- (void)appClient:(ARDAppClient *)client
>-    didChangeState:(ARDAppClientState)state;
>+- (void)appClient:(ARDAppClient *)client didChangeState:(ARDAppClientState)state;
> 
>-- (void)appClient:(ARDAppClient *)client
>-    didChangeConnectionState:(RTCIceConnectionState)state;
>+- (void)appClient:(ARDAppClient *)client didChangeConnectionState:(RTCIceConnectionState)state;
> 
> - (void)appClient:(ARDAppClient *)client
>     didCreateLocalCapturer:(RTCCameraVideoCapturer *)localCapturer;
> 
>-- (void)appClient:(ARDAppClient *)client
>-    didReceiveLocalVideoTrack:(RTCVideoTrack *)localVideoTrack;
>+- (void)appClient:(ARDAppClient *)client didReceiveLocalVideoTrack:(RTCVideoTrack *)localVideoTrack;
> 
> - (void)appClient:(ARDAppClient *)client
>     didReceiveRemoteVideoTrack:(RTCVideoTrack *)remoteVideoTrack;
> 
>-- (void)appClient:(ARDAppClient *)client
>-         didError:(NSError *)error;
>+- (void)appClient:(ARDAppClient *)client didError:(NSError *)error;
> 
>-- (void)appClient:(ARDAppClient *)client
>-      didGetStats:(NSArray *)stats;
>+- (void)appClient:(ARDAppClient *)client didGetStats:(NSArray *)stats;
> 
> @optional
> - (void)appClient:(ARDAppClient *)client
>-didCreateLocalFileCapturer:(RTCFileVideoCapturer *)fileCapturer;
>+    didCreateLocalFileCapturer:(RTCFileVideoCapturer *)fileCapturer;
>+
>+- (void)appClient:(ARDAppClient *)client
>+    didCreateLocalExternalSampleCapturer:(ARDExternalSampleCapturer *)externalSampleCapturer;
> 
> @end
> 
>@@ -67,6 +66,8 @@ didCreateLocalFileCapturer:(RTCFileVideoCapturer *)fileCapturer;
> @property(nonatomic, assign) BOOL shouldGetStats;
> @property(nonatomic, readonly) ARDAppClientState state;
> @property(nonatomic, weak) id<ARDAppClientDelegate> delegate;
>+@property(nonatomic, assign, getter=isBroadcast) BOOL broadcast;
>+
> // Convenience constructor since all expected use cases will need a delegate
> // in order to receive remote tracks.
> - (instancetype)initWithDelegate:(id<ARDAppClientDelegate>)delegate;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDAppClient.m b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDAppClient.m
>index 10be40a89ea..465ce8c9b80 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDAppClient.m
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDAppClient.m
>@@ -10,7 +10,6 @@
> 
> #import "ARDAppClient+Internal.h"
> 
>-#import "WebRTC/RTCAVFoundationVideoSource.h"
> #import "WebRTC/RTCAudioTrack.h"
> #import "WebRTC/RTCCameraVideoCapturer.h"
> #import "WebRTC/RTCConfiguration.h"
>@@ -22,11 +21,14 @@
> #import "WebRTC/RTCMediaStream.h"
> #import "WebRTC/RTCPeerConnectionFactory.h"
> #import "WebRTC/RTCRtpSender.h"
>+#import "WebRTC/RTCRtpTransceiver.h"
> #import "WebRTC/RTCTracing.h"
> #import "WebRTC/RTCVideoCodecFactory.h"
>+#import "WebRTC/RTCVideoSource.h"
> #import "WebRTC/RTCVideoTrack.h"
> 
> #import "ARDAppEngineClient.h"
>+#import "ARDExternalSampleCapturer.h"
> #import "ARDJoinResponse.h"
> #import "ARDMessageResponse.h"
> #import "ARDSettingsModel.h"
>@@ -52,10 +54,12 @@
> static NSString * const kARDVideoTrackKind = @"video";
> 
> // TODO(tkchin): Add these as UI options.
>+#if defined(WEBRTC_IOS)
> static BOOL const kARDAppClientEnableTracing = NO;
> static BOOL const kARDAppClientEnableRtcEventLog = YES;
> static int64_t const kARDAppClientAecDumpMaxSizeInBytes = 5e6;  // 5 MB.
> static int64_t const kARDAppClientRtcEventLogMaxSizeInBytes = 5e6;  // 5 MB.
>+#endif
> static int const kKbpsMultiplier = 1000;
> 
> // We need a proxy to NSTimer because it causes a strong retain cycle. When
>@@ -127,6 +131,7 @@ @implementation ARDAppClient {
> @synthesize defaultPeerConnectionConstraints =
>     _defaultPeerConnectionConstraints;
> @synthesize isLoopback = _isLoopback;
>+@synthesize broadcast = _broadcast;
> 
> - (instancetype)init {
>   return [self initWithDelegate:nil];
>@@ -236,8 +241,7 @@ - (void)connectToRoomWithId:(NSString *)roomId
>   [_turnClient requestServersWithCompletionHandler:^(NSArray *turnServers,
>                                                      NSError *error) {
>     if (error) {
>-      RTCLogError("Error retrieving TURN servers: %@",
>-                  error.localizedDescription);
>+      RTCLogError(@"Error retrieving TURN servers: %@", error.localizedDescription);
>     }
>     ARDAppClient *strongSelf = weakSelf;
>     [strongSelf.iceServers addObjectsFromArray:turnServers];
>@@ -371,15 +375,15 @@ - (void)peerConnection:(RTCPeerConnection *)peerConnection
> 
> - (void)peerConnection:(RTCPeerConnection *)peerConnection
>           didAddStream:(RTCMediaStream *)stream {
>-  dispatch_async(dispatch_get_main_queue(), ^{
>-    RTCLog(@"Received %lu video tracks and %lu audio tracks",
>-        (unsigned long)stream.videoTracks.count,
>-        (unsigned long)stream.audioTracks.count);
>-    if (stream.videoTracks.count) {
>-      RTCVideoTrack *videoTrack = stream.videoTracks[0];
>-      [_delegate appClient:self didReceiveRemoteVideoTrack:videoTrack];
>-    }
>-  });
>+  RTCLog(@"Stream with %lu video tracks and %lu audio tracks was added.",
>+         (unsigned long)stream.videoTracks.count,
>+         (unsigned long)stream.audioTracks.count);
>+}
>+
>+- (void)peerConnection:(RTCPeerConnection *)peerConnection
>+    didStartReceivingOnTransceiver:(RTCRtpTransceiver *)transceiver {
>+  RTCMediaStreamTrack *track = transceiver.receiver.track;
>+  RTCLog(@"Now receiving %@ on track %@.", track.kind, track.trackId);
> }
> 
> - (void)peerConnection:(RTCPeerConnection *)peerConnection
>@@ -529,7 +533,14 @@ - (void)startSignalingIfReady {
>   // Create peer connection.
>   RTCMediaConstraints *constraints = [self defaultPeerConnectionConstraints];
>   RTCConfiguration *config = [[RTCConfiguration alloc] init];
>+  RTCCertificate *pcert = [RTCCertificate generateCertificateWithParams:@{
>+    @"expires" : @100000,
>+    @"name" : @"RSASSA-PKCS1-v1_5"
>+  }];
>   config.iceServers = _iceServers;
>+  config.sdpSemantics = RTCSdpSemanticsUnifiedPlan;
>+  config.certificate = pcert;
>+
>   _peerConnection = [_factory peerConnectionWithConfiguration:config
>                                                   constraints:constraints
>                                                      delegate:self];
>@@ -676,18 +687,31 @@ - (void)setMaxBitrate:(NSNumber *)maxBitrate forVideoSender:(RTCRtpSender *)send
>   [sender setParameters:parametersToModify];
> }
> 
>+- (RTCRtpTransceiver *)videoTransceiver {
>+  for (RTCRtpTransceiver *transceiver in _peerConnection.transceivers) {
>+    if (transceiver.mediaType == RTCRtpMediaTypeVideo) {
>+      return transceiver;
>+    }
>+  }
>+  return nil;
>+}
>+
> - (void)createMediaSenders {
>   RTCMediaConstraints *constraints = [self defaultMediaAudioConstraints];
>   RTCAudioSource *source = [_factory audioSourceWithConstraints:constraints];
>   RTCAudioTrack *track = [_factory audioTrackWithSource:source
>                                                 trackId:kARDAudioTrackId];
>-  RTCMediaStream *stream = [_factory mediaStreamWithStreamId:kARDMediaStreamId];
>-  [stream addAudioTrack:track];
>+  [_peerConnection addTrack:track streamIds:@[ kARDMediaStreamId ]];
>   _localVideoTrack = [self createLocalVideoTrack];
>   if (_localVideoTrack) {
>-    [stream addVideoTrack:_localVideoTrack];
>+    [_peerConnection addTrack:_localVideoTrack streamIds:@[ kARDMediaStreamId ]];
>+    [_delegate appClient:self didReceiveLocalVideoTrack:_localVideoTrack];
>+    // We can set up rendering for the remote track right away since the transceiver already has an
>+    // RTCRtpReceiver with a track. The track will automatically get unmuted and produce frames
>+    // once RTP is received.
>+    RTCVideoTrack *track = (RTCVideoTrack *)([self videoTransceiver].receiver.track);
>+    [_delegate appClient:self didReceiveRemoteVideoTrack:track];
>   }
>-  [_peerConnection addStream:stream];
> }
> 
> - (RTCVideoTrack *)createLocalVideoTrack {
>@@ -698,9 +722,14 @@ - (RTCVideoTrack *)createLocalVideoTrack {
>   RTCVideoSource *source = [_factory videoSource];
> 
> #if !TARGET_IPHONE_SIMULATOR
>-  RTCCameraVideoCapturer *capturer = [[RTCCameraVideoCapturer alloc] initWithDelegate:source];
>-  [_delegate appClient:self didCreateLocalCapturer:capturer];
>-
>+  if (self.isBroadcast) {
>+    ARDExternalSampleCapturer *capturer =
>+        [[ARDExternalSampleCapturer alloc] initWithDelegate:source];
>+    [_delegate appClient:self didCreateLocalExternalSampleCapturer:capturer];
>+  } else {
>+    RTCCameraVideoCapturer *capturer = [[RTCCameraVideoCapturer alloc] initWithDelegate:source];
>+    [_delegate appClient:self didCreateLocalCapturer:capturer];
>+  }
> #else
> #if defined(__IPHONE_11_0) && (__IPHONE_OS_VERSION_MAX_ALLOWED >= __IPHONE_11_0)
>   if (@available(iOS 10, *)) {
>@@ -740,10 +769,7 @@ - (void)registerWithColliderIfReady {
> #pragma mark - Defaults
> 
>  - (RTCMediaConstraints *)defaultMediaAudioConstraints {
>-   NSString *valueLevelControl = [_settings currentUseLevelControllerSettingFromStore] ?
>-       kRTCMediaConstraintsValueTrue :
>-       kRTCMediaConstraintsValueFalse;
>-   NSDictionary *mandatoryConstraints = @{ kRTCMediaConstraintsLevelControl : valueLevelControl };
>+   NSDictionary *mandatoryConstraints = @{};
>    RTCMediaConstraints *constraints =
>        [[RTCMediaConstraints alloc] initWithMandatoryConstraints:mandatoryConstraints
>                                              optionalConstraints:nil];
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDAppEngineClient.h b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDAppEngineClient.h
>index 7514f3645c4..db4e2efd7f8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDAppEngineClient.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDAppEngineClient.h
>@@ -10,5 +10,5 @@
> 
> #import "ARDRoomServerClient.h"
> 
>-@interface ARDAppEngineClient : NSObject <ARDRoomServerClient>
>+@interface ARDAppEngineClient : NSObject<ARDRoomServerClient>
> @end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDAppEngineClient.m b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDAppEngineClient.m
>index ad7a6c2bf2b..0c22ea4b0f3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDAppEngineClient.m
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDAppEngineClient.m
>@@ -55,10 +55,8 @@ - (void)joinRoomWithRoomId:(NSString *)roomId
>   RTCLog(@"Joining room:%@ on room server.", roomId);
>   NSMutableURLRequest *request = [NSMutableURLRequest requestWithURL:roomURL];
>   request.HTTPMethod = @"POST";
>-  __weak ARDAppEngineClient *weakSelf = self;
>   [NSURLConnection sendAsyncRequest:request
>                   completionHandler:^(NSURLResponse *response, NSData *data, NSError *error) {
>-                    ARDAppEngineClient *strongSelf = weakSelf;
>                     if (error) {
>                       if (completionHandler) {
>                         completionHandler(nil, error);
>@@ -97,12 +95,10 @@ - (void)sendMessage:(ARDSignalingMessage *)message
>   NSMutableURLRequest *request = [NSMutableURLRequest requestWithURL:url];
>   request.HTTPMethod = @"POST";
>   request.HTTPBody = data;
>-  __weak ARDAppEngineClient *weakSelf = self;
>   [NSURLConnection sendAsyncRequest:request
>                   completionHandler:^(NSURLResponse *response,
>                                       NSData *data,
>                                       NSError *error) {
>-    ARDAppEngineClient *strongSelf = weakSelf;
>     if (error) {
>       if (completionHandler) {
>         completionHandler(nil, error);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDCaptureController.m b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDCaptureController.m
>index 6830e88c23e..068e18ec929 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDCaptureController.m
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDCaptureController.m
>@@ -13,6 +13,8 @@
> #import "ARDSettingsModel.h"
> #import "WebRTC/RTCLogging.h"
> 
>+const Float64 kFramerateLimit = 30.0;
>+
> @implementation ARDCaptureController {
>   RTCCameraVideoCapturer *_capturer;
>   ARDSettingsModel *_settings;
>@@ -21,7 +23,7 @@ @implementation ARDCaptureController {
> 
> - (instancetype)initWithCapturer:(RTCCameraVideoCapturer *)capturer
>                         settings:(ARDSettingsModel *)settings {
>-  if ([super init]) {
>+  if (self = [super init]) {
>     _capturer = capturer;
>     _settings = settings;
>     _usingFrontCamera = YES;
>@@ -93,11 +95,11 @@ - (AVCaptureDeviceFormat *)selectFormatForDevice:(AVCaptureDevice *)device {
> }
> 
> - (NSInteger)selectFpsForFormat:(AVCaptureDeviceFormat *)format {
>-  Float64 maxFramerate = 0;
>+  Float64 maxSupportedFramerate = 0;
>   for (AVFrameRateRange *fpsRange in format.videoSupportedFrameRateRanges) {
>-    maxFramerate = fmax(maxFramerate, fpsRange.maxFrameRate);
>+    maxSupportedFramerate = fmax(maxSupportedFramerate, fpsRange.maxFrameRate);
>   }
>-  return maxFramerate;
>+  return fmin(maxSupportedFramerate, kFramerateLimit);
> }
> 
> @end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDExternalSampleCapturer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDExternalSampleCapturer.h
>new file mode 100644
>index 00000000000..98a60fc7eda
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDExternalSampleCapturer.h
>@@ -0,0 +1,18 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#import <WebRTC/RTCVideoCapturer.h>
>+
>+@protocol ARDExternalSampleDelegate <NSObject>
>+- (void)didCaptureSampleBuffer:(CMSampleBufferRef)sampleBuffer;
>+@end
>+
>+@interface ARDExternalSampleCapturer : RTCVideoCapturer <ARDExternalSampleDelegate>
>+@end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDExternalSampleCapturer.m b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDExternalSampleCapturer.m
>new file mode 100644
>index 00000000000..d377033bde2
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDExternalSampleCapturer.m
>@@ -0,0 +1,43 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#import "ARDExternalSampleCapturer.h"
>+
>+#import "WebRTC/RTCVideoFrameBuffer.h"
>+
>+@implementation ARDExternalSampleCapturer
>+
>+- (instancetype)initWithDelegate:(__weak id<RTCVideoCapturerDelegate>)delegate {
>+  return [super initWithDelegate:delegate];
>+}
>+
>+#pragma mark - ARDExternalSampleDelegate
>+
>+- (void)didCaptureSampleBuffer:(CMSampleBufferRef)sampleBuffer {
>+  if (CMSampleBufferGetNumSamples(sampleBuffer) != 1 || !CMSampleBufferIsValid(sampleBuffer) ||
>+      !CMSampleBufferDataIsReady(sampleBuffer)) {
>+    return;
>+  }
>+
>+  CVPixelBufferRef pixelBuffer = CMSampleBufferGetImageBuffer(sampleBuffer);
>+  if (pixelBuffer == nil) {
>+    return;
>+  }
>+
>+  RTCCVPixelBuffer *rtcPixelBuffer = [[RTCCVPixelBuffer alloc] initWithPixelBuffer:pixelBuffer];
>+  int64_t timeStampNs =
>+      CMTimeGetSeconds(CMSampleBufferGetPresentationTimeStamp(sampleBuffer)) * NSEC_PER_SEC;
>+  RTCVideoFrame *videoFrame = [[RTCVideoFrame alloc] initWithBuffer:rtcPixelBuffer
>+                                                           rotation:RTCVideoRotation_0
>+                                                        timeStampNs:timeStampNs];
>+  [self.delegate capturer:self didCaptureVideoFrame:videoFrame];
>+}
>+
>+@end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDJoinResponse+Internal.h b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDJoinResponse+Internal.h
>index b3202999d08..0edf7083c0c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDJoinResponse+Internal.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDJoinResponse+Internal.h
>@@ -14,10 +14,10 @@
> 
> @property(nonatomic, assign) ARDJoinResultType result;
> @property(nonatomic, assign) BOOL isInitiator;
>-@property(nonatomic, strong) NSString *roomId;
>-@property(nonatomic, strong) NSString *clientId;
>-@property(nonatomic, strong) NSArray *messages;
>-@property(nonatomic, strong) NSURL *webSocketURL;
>-@property(nonatomic, strong) NSURL *webSocketRestURL;
>+@property(nonatomic, strong) NSString* roomId;
>+@property(nonatomic, strong) NSString* clientId;
>+@property(nonatomic, strong) NSArray* messages;
>+@property(nonatomic, strong) NSURL* webSocketURL;
>+@property(nonatomic, strong) NSURL* webSocketRestURL;
> 
> @end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDRoomServerClient.h b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDRoomServerClient.h
>index 70694a8c9f1..3a5818d6d6f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDRoomServerClient.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDRoomServerClient.h
>@@ -18,14 +18,12 @@
> 
> - (void)joinRoomWithRoomId:(NSString *)roomId
>                 isLoopback:(BOOL)isLoopback
>-         completionHandler:(void (^)(ARDJoinResponse *response,
>-                                     NSError *error))completionHandler;
>+         completionHandler:(void (^)(ARDJoinResponse *response, NSError *error))completionHandler;
> 
> - (void)sendMessage:(ARDSignalingMessage *)message
>             forRoomId:(NSString *)roomId
>              clientId:(NSString *)clientId
>-    completionHandler:(void (^)(ARDMessageResponse *response,
>-                                NSError *error))completionHandler;
>+    completionHandler:(void (^)(ARDMessageResponse *response, NSError *error))completionHandler;
> 
> - (void)leaveRoomWithRoomId:(NSString *)roomId
>                    clientId:(NSString *)clientId
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDSettingsModel.h b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDSettingsModel.h
>index 625b533a0e3..597ab38fbf5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDSettingsModel.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDSettingsModel.h
>@@ -106,19 +106,6 @@ NS_ASSUME_NONNULL_BEGIN
>  */
> - (void)storeCreateAecDumpSetting:(BOOL)createAecDump;
> 
>-/**
>- * Returns current setting whether to use level controller from store if present or default (NO)
>- * otherwise.
>- */
>-- (BOOL)currentUseLevelControllerSettingFromStore;
>-
>-/**
>- * Stores the provided use level controller setting into the store.
>- *
>- * @param setting the boolean value to be stored.
>- */
>-- (void)storeUseLevelControllerSetting:(BOOL)useLevelController;
>-
> /**
>  * Returns current setting whether to use manual audio config from store if present or default (YES)
>  * otherwise.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDSettingsModel.m b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDSettingsModel.m
>index a4eea731d45..34a47a49894 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDSettingsModel.m
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDSettingsModel.m
>@@ -39,7 +39,11 @@ @implementation ARDSettingsModel
>   NSArray<NSArray<NSNumber *> *> *sortedResolutions =
>       [[resolutions allObjects] sortedArrayUsingComparator:^NSComparisonResult(
>                                     NSArray<NSNumber *> *obj1, NSArray<NSNumber *> *obj2) {
>-        return obj1.firstObject > obj2.firstObject;
>+        NSComparisonResult cmp = [obj1.firstObject compare:obj2.firstObject];
>+        if (cmp != NSOrderedSame) {
>+          return cmp;
>+        }
>+        return [obj1.lastObject compare:obj2.lastObject];
>       }];
> 
>   NSMutableArray<NSString *> *resolutionStrings = [[NSMutableArray<NSString *> alloc] init];
>@@ -109,14 +113,6 @@ - (void)storeCreateAecDumpSetting:(BOOL)createAecDump {
>   [[self settingsStore] setCreateAecDump:createAecDump];
> }
> 
>-- (BOOL)currentUseLevelControllerSettingFromStore {
>-  return [[self settingsStore] useLevelController];
>-}
>-
>-- (void)storeUseLevelControllerSetting:(BOOL)useLevelController {
>-  [[self settingsStore] setUseLevelController:useLevelController];
>-}
>-
> - (BOOL)currentUseManualAudioConfigSettingFromStore {
>   return [[self settingsStore] useManualAudioConfig];
> }
>@@ -168,15 +164,12 @@ - (int)videoResolutionComponentAtIndex:(int)index inString:(NSString *)resolutio
> }
> 
> - (void)registerStoreDefaults {
>-  NSString *defaultVideoResolutionSetting = [self defaultVideoResolutionSetting];
>-
>   NSData *codecData = [NSKeyedArchiver archivedDataWithRootObject:[self defaultVideoCodecSetting]];
>   [ARDSettingsStore setDefaultsForVideoResolution:[self defaultVideoResolutionSetting]
>                                        videoCodec:codecData
>                                           bitrate:nil
>                                         audioOnly:NO
>                                     createAecDump:NO
>-                               useLevelController:NO
>                              useManualAudioConfig:YES];
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDSettingsStore.h b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDSettingsStore.h
>index b68adb737af..bb051dbb26d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDSettingsStore.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDSettingsStore.h
>@@ -28,7 +28,6 @@ NS_ASSUME_NONNULL_BEGIN
>                               bitrate:(nullable NSNumber *)bitrate
>                             audioOnly:(BOOL)audioOnly
>                         createAecDump:(BOOL)createAecDump
>-                   useLevelController:(BOOL)useLevelController
>                  useManualAudioConfig:(BOOL)useManualAudioConfig;
> 
> @property(nonatomic) NSString *videoResolution;
>@@ -47,7 +46,6 @@ NS_ASSUME_NONNULL_BEGIN
> 
> @property(nonatomic) BOOL audioOnly;
> @property(nonatomic) BOOL createAecDump;
>-@property(nonatomic) BOOL useLevelController;
> @property(nonatomic) BOOL useManualAudioConfig;
> 
> @end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDSettingsStore.m b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDSettingsStore.m
>index 86bdaaf54c7..a3713e2f0e6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDSettingsStore.m
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDSettingsStore.m
>@@ -15,7 +15,6 @@
> static NSString *const kBitrateKey = @"rtc_max_bitrate_key";
> static NSString *const kAudioOnlyKey = @"rtc_audio_only_key";
> static NSString *const kCreateAecDumpKey = @"rtc_create_aec_dump_key";
>-static NSString *const kUseLevelControllerKey = @"rtc_use_level_controller_key";
> static NSString *const kUseManualAudioConfigKey = @"rtc_use_manual_audio_config_key";
> 
> NS_ASSUME_NONNULL_BEGIN
>@@ -32,12 +31,10 @@ + (void)setDefaultsForVideoResolution:(NSString *)videoResolution
>                               bitrate:(nullable NSNumber *)bitrate
>                             audioOnly:(BOOL)audioOnly
>                         createAecDump:(BOOL)createAecDump
>-                   useLevelController:(BOOL)useLevelController
>                  useManualAudioConfig:(BOOL)useManualAudioConfig {
>   NSMutableDictionary<NSString *, id> *defaultsDictionary = [@{
>     kAudioOnlyKey : @(audioOnly),
>     kCreateAecDumpKey : @(createAecDump),
>-    kUseLevelControllerKey : @(useLevelController),
>     kUseManualAudioConfigKey : @(useManualAudioConfig)
>   } mutableCopy];
> 
>@@ -105,15 +102,6 @@ - (void)setCreateAecDump:(BOOL)createAecDump {
>   [self.storage synchronize];
> }
> 
>-- (BOOL)useLevelController {
>-  return [self.storage boolForKey:kUseLevelControllerKey];
>-}
>-
>-- (void)setUseLevelController:(BOOL)useLevelController {
>-  [self.storage setBool:useLevelController forKey:kUseLevelControllerKey];
>-  [self.storage synchronize];
>-}
>-
> - (BOOL)useManualAudioConfig {
>   return [self.storage boolForKey:kUseManualAudioConfigKey];
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDSignalingChannel.h b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDSignalingChannel.h
>index 70ba2ff4c95..396b117b179 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDSignalingChannel.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDSignalingChannel.h
>@@ -26,11 +26,9 @@ typedef NS_ENUM(NSInteger, ARDSignalingChannelState) {
> @protocol ARDSignalingChannel;
> @protocol ARDSignalingChannelDelegate <NSObject>
> 
>-- (void)channel:(id<ARDSignalingChannel>)channel
>-    didChangeState:(ARDSignalingChannelState)state;
>+- (void)channel:(id<ARDSignalingChannel>)channel didChangeState:(ARDSignalingChannelState)state;
> 
>-- (void)channel:(id<ARDSignalingChannel>)channel
>-    didReceiveMessage:(ARDSignalingMessage *)message;
>+- (void)channel:(id<ARDSignalingChannel>)channel didReceiveMessage:(ARDSignalingMessage *)message;
> 
> @end
> 
>@@ -42,11 +40,9 @@ typedef NS_ENUM(NSInteger, ARDSignalingChannelState) {
> @property(nonatomic, weak) id<ARDSignalingChannelDelegate> delegate;
> 
> // Registers the channel for the given room and client id.
>-- (void)registerForRoomId:(NSString *)roomId
>-                 clientId:(NSString *)clientId;
>+- (void)registerForRoomId:(NSString *)roomId clientId:(NSString *)clientId;
> 
> // Sends signaling message over the channel.
> - (void)sendMessage:(ARDSignalingMessage *)message;
> 
> @end
>-
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDSignalingMessage.h b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDSignalingMessage.h
>index e60517282b9..93ff4862cf2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDSignalingMessage.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDSignalingMessage.h
>@@ -42,8 +42,7 @@ typedef enum {
> 
> @property(nonatomic, readonly) NSArray<RTCIceCandidate *> *candidates;
> 
>-- (instancetype)initWithRemovedCandidates:
>-    (NSArray<RTCIceCandidate *> *)candidates;
>+- (instancetype)initWithRemovedCandidates:(NSArray<RTCIceCandidate *> *)candidates;
> 
> @end
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDTURNClient.h b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDTURNClient.h
>index 75ccffcb8ca..0cefaf6c199 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDTURNClient.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDTURNClient.h
>@@ -15,8 +15,7 @@
> @protocol ARDTURNClient <NSObject>
> 
> // Returns TURN server urls if successful.
>-- (void)requestServersWithCompletionHandler:
>-    (void (^)(NSArray *turnServers,
>-              NSError *error))completionHandler;
>+- (void)requestServersWithCompletionHandler:(void (^)(NSArray *turnServers,
>+                                                      NSError *error))completionHandler;
> 
> @end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDWebSocketChannel.h b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDWebSocketChannel.h
>index ffb0b724afa..81888e6e839 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDWebSocketChannel.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDWebSocketChannel.h
>@@ -21,8 +21,7 @@
> 
> // Registers with the WebSocket server for the given room and client id once
> // the web socket connection is open.
>-- (void)registerForRoomId:(NSString *)roomId
>-                 clientId:(NSString *)clientId;
>+- (void)registerForRoomId:(NSString *)roomId clientId:(NSString *)clientId;
> 
> // Sends message over the WebSocket connection if registered, otherwise POSTs to
> // the web socket server instead.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/RTCIceCandidate+JSON.h b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/RTCIceCandidate+JSON.h
>index d2e5e33b5bb..18265558618 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/RTCIceCandidate+JSON.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/RTCIceCandidate+JSON.h
>@@ -13,8 +13,7 @@
> @interface RTCIceCandidate (JSON)
> 
> + (RTCIceCandidate *)candidateFromJSONDictionary:(NSDictionary *)dictionary;
>-+ (NSArray<RTCIceCandidate *> *)candidatesFromJSONDictionary:
>-    (NSDictionary *)dictionary;
>++ (NSArray<RTCIceCandidate *> *)candidatesFromJSONDictionary:(NSDictionary *)dictionary;
> + (NSData *)JSONDataForIceCandidates:(NSArray<RTCIceCandidate *> *)candidates
>                             withType:(NSString *)typeValue;
> - (NSData *)JSONData;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/RTCMediaConstraints+JSON.h b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/RTCMediaConstraints+JSON.h
>index 74f89a953fb..42d7ab0f019 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/RTCMediaConstraints+JSON.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/RTCMediaConstraints+JSON.h
>@@ -12,8 +12,6 @@
> 
> @interface RTCMediaConstraints (JSON)
> 
>-+ (RTCMediaConstraints *)constraintsFromJSONDictionary:
>-    (NSDictionary *)dictionary;
>++ (RTCMediaConstraints *)constraintsFromJSONDictionary:(NSDictionary *)dictionary;
> 
> @end
>-
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/RTCSessionDescription+JSON.h b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/RTCSessionDescription+JSON.h
>index cccff9ad74c..f5671213a89 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/RTCSessionDescription+JSON.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/RTCSessionDescription+JSON.h
>@@ -12,8 +12,7 @@
> 
> @interface RTCSessionDescription (JSON)
> 
>-+ (RTCSessionDescription *)descriptionFromJSONDictionary:
>-    (NSDictionary *)dictionary;
>++ (RTCSessionDescription *)descriptionFromJSONDictionary:(NSDictionary *)dictionary;
> - (NSData *)JSONData;
> 
> @end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/common/ARDUtilities.h b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/common/ARDUtilities.h
>index 8a5c1262ab9..5f0d7dbef71 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/common/ARDUtilities.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/common/ARDUtilities.h
>@@ -22,17 +22,14 @@
> 
> // Issues an asynchronous request that calls back on main queue.
> + (void)sendAsyncRequest:(NSURLRequest *)request
>-       completionHandler:(void (^)(NSURLResponse *response,
>-                                   NSData *data,
>-                                   NSError *error))completionHandler;
>+       completionHandler:
>+           (void (^)(NSURLResponse *response, NSData *data, NSError *error))completionHandler;
> 
> // Posts data to the specified URL.
> + (void)sendAsyncPostToURL:(NSURL *)url
>                   withData:(NSData *)data
>-         completionHandler:(void (^)(BOOL succeeded,
>-                                     NSData *data))completionHandler;
>+         completionHandler:(void (^)(BOOL succeeded, NSData *data))completionHandler;
> 
> @end
> 
>-NSInteger ARDGetCpuUsagePercentage();
>-
>+NSInteger ARDGetCpuUsagePercentage(void);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ios/ARDAppDelegate.h b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ios/ARDAppDelegate.h
>index 7eafff8ebcd..623e8590960 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ios/ARDAppDelegate.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ios/ARDAppDelegate.h
>@@ -13,5 +13,5 @@
> // The main application class of the AppRTCMobile iOS app demonstrating
> // interoperability between the Objective C implementation of PeerConnection
> // and the appr.tc demo webapp.
>-@interface ARDAppDelegate : NSObject <UIApplicationDelegate>
>+@interface ARDAppDelegate : NSObject<UIApplicationDelegate>
> @end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ios/ARDAppDelegate.m b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ios/ARDAppDelegate.m
>index 07c83a0826f..4f444e428d1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ios/ARDAppDelegate.m
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ios/ARDAppDelegate.m
>@@ -25,9 +25,7 @@ @implementation ARDAppDelegate {
> 
> - (BOOL)application:(UIApplication *)application
>     didFinishLaunchingWithOptions:(NSDictionary *)launchOptions {
>-  NSDictionary *fieldTrials = @{
>-    kRTCFieldTrialH264HighProfileKey: kRTCFieldTrialEnabledValue,
>-  };
>+  NSDictionary *fieldTrials = @{};
>   RTCInitFieldTrialDictionary(fieldTrials);
>   RTCInitializeSSL();
>   RTCSetupInternalTracer();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ios/ARDMainView.m b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ios/ARDMainView.m
>index fcce3c30159..c3bd24a7a06 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ios/ARDMainView.m
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ios/ARDMainView.m
>@@ -29,7 +29,7 @@ - (instancetype)initWithFrame:(CGRect)frame {
>   if (self = [super initWithFrame:frame]) {
>     _roomText = [[UITextField alloc] initWithFrame:CGRectZero];
>     _roomText.borderStyle = UITextBorderStyleNone;
>-    _roomText.font = [UIFont fontWithName:@"Roboto" size:12];
>+    _roomText.font = [UIFont systemFontOfSize:12];
>     _roomText.placeholder = @"Room name";
>     _roomText.autocorrectionType = UITextAutocorrectionTypeNo;
>     _roomText.autocapitalizationType = UITextAutocapitalizationTypeNone;
>@@ -73,10 +73,8 @@ - (BOOL)textFieldShouldReturn:(UITextField *)textField {
> 
> @implementation ARDMainView {
>   ARDRoomTextField *_roomText;
>-  UILabel *_callOptionsLabel;
>-  UISwitch *_loopbackSwitch;
>-  UILabel *_loopbackLabel;
>-  UIButton *_startCallButton;
>+  UIButton *_startRegularCallButton;
>+  UIButton *_startLoopbackCallButton;
>   UIButton *_audioLoopButton;
> }
> 
>@@ -88,40 +86,38 @@ - (instancetype)initWithFrame:(CGRect)frame {
>     _roomText = [[ARDRoomTextField alloc] initWithFrame:CGRectZero];
>     [self addSubview:_roomText];
> 
>-    UIFont *controlFont = [UIFont fontWithName:@"Roboto" size:20];
>-    UIColor *controlFontColor = [UIColor colorWithWhite:0 alpha:.6];
>-
>-    _callOptionsLabel = [[UILabel alloc] initWithFrame:CGRectZero];
>-    _callOptionsLabel.text = @"Call Options";
>-    _callOptionsLabel.font = controlFont;
>-    _callOptionsLabel.textColor = controlFontColor;
>-    [_callOptionsLabel sizeToFit];
>-    [self addSubview:_callOptionsLabel];
>-
>-    _loopbackSwitch = [[UISwitch alloc] initWithFrame:CGRectZero];
>-    [_loopbackSwitch sizeToFit];
>-    [self addSubview:_loopbackSwitch];
>-
>-    _loopbackLabel = [[UILabel alloc] initWithFrame:CGRectZero];
>-    _loopbackLabel.text = @"Loopback mode";
>-    _loopbackLabel.font = controlFont;
>-    _loopbackLabel.textColor = controlFontColor;
>-    [_loopbackLabel sizeToFit];
>-    [self addSubview:_loopbackLabel];
>-
>-    _startCallButton = [UIButton buttonWithType:UIButtonTypeSystem];
>-    [_startCallButton setTitle:@"Start call"
>-                      forState:UIControlStateNormal];
>-    _startCallButton.titleLabel.font = controlFont;
>-    [_startCallButton sizeToFit];
>-    [_startCallButton addTarget:self
>-                         action:@selector(onStartCall:)
>-               forControlEvents:UIControlEventTouchUpInside];
>-    [self addSubview:_startCallButton];
>+    UIFont *controlFont = [UIFont boldSystemFontOfSize:18.0];
>+    UIColor *controlFontColor = [UIColor whiteColor];
>+
>+    _startRegularCallButton = [UIButton buttonWithType:UIButtonTypeSystem];
>+    _startRegularCallButton.titleLabel.font = controlFont;
>+    [_startRegularCallButton setTitleColor:controlFontColor forState:UIControlStateNormal];
>+    _startRegularCallButton.backgroundColor
>+        = [UIColor colorWithRed:66.0/255.0 green:200.0/255.0 blue:90.0/255.0 alpha:1.0];
>+    [_startRegularCallButton setTitle:@"Call room" forState:UIControlStateNormal];
>+    [_startRegularCallButton addTarget:self
>+                                action:@selector(onStartRegularCall:)
>+                      forControlEvents:UIControlEventTouchUpInside];
>+    [self addSubview:_startRegularCallButton];
>+
>+    _startLoopbackCallButton = [UIButton buttonWithType:UIButtonTypeSystem];
>+    _startLoopbackCallButton.titleLabel.font = controlFont;
>+    [_startLoopbackCallButton setTitleColor:controlFontColor forState:UIControlStateNormal];
>+    _startLoopbackCallButton.backgroundColor =
>+        [UIColor colorWithRed:0.0 green:122.0/255.0 blue:1.0 alpha:1.0];
>+    [_startLoopbackCallButton setTitle:@"Loopback call" forState:UIControlStateNormal];
>+    [_startLoopbackCallButton addTarget:self
>+                                 action:@selector(onStartLoopbackCall:)
>+                       forControlEvents:UIControlEventTouchUpInside];
>+    [self addSubview:_startLoopbackCallButton];
>+
> 
>     // Used to test what happens to sounds when calls are in progress.
>     _audioLoopButton = [UIButton buttonWithType:UIButtonTypeSystem];
>     _audioLoopButton.titleLabel.font = controlFont;
>+    [_audioLoopButton setTitleColor:controlFontColor forState:UIControlStateNormal];
>+    _audioLoopButton.backgroundColor =
>+        [UIColor colorWithRed:1.0 green:149.0/255.0 blue:0.0 alpha:1.0];
>     [self updateAudioLoopButton];
>     [_audioLoopButton addTarget:self
>                          action:@selector(onToggleAudioLoop:)
>@@ -146,51 +142,42 @@ - (void)layoutSubviews {
>   CGFloat roomTextWidth = bounds.size.width - 2 * kRoomTextFieldMargin;
>   CGFloat roomTextHeight = [_roomText sizeThatFits:bounds.size].height;
>   _roomText.frame =
>-      CGRectMake(kRoomTextFieldMargin, kRoomTextFieldMargin, roomTextWidth, roomTextHeight);
>-
>-  CGFloat callOptionsLabelTop =
>-      CGRectGetMaxY(_roomText.frame) + kCallControlMargin * 4;
>-  _callOptionsLabel.frame = CGRectMake(kCallControlMargin,
>-                                       callOptionsLabelTop,
>-                                       _callOptionsLabel.frame.size.width,
>-                                       _callOptionsLabel.frame.size.height);
>-
>-  CGFloat loopbackModeTop = CGRectGetMaxY(_callOptionsLabel.frame) + kCallControlMargin * 2;
>-  CGRect loopbackModeRect = CGRectMake(kCallControlMargin * 3,
>-                                       loopbackModeTop,
>-                                       _loopbackSwitch.frame.size.width,
>-                                       _loopbackSwitch.frame.size.height);
>-  _loopbackSwitch.frame = loopbackModeRect;
>-  CGFloat loopbackModeLabelCenterX = CGRectGetMaxX(loopbackModeRect) +
>-      kCallControlMargin + _loopbackLabel.frame.size.width / 2;
>-  _loopbackLabel.center = CGPointMake(loopbackModeLabelCenterX,
>-                                      CGRectGetMidY(loopbackModeRect));
>-
>-  CGFloat audioLoopTop = CGRectGetMaxY(loopbackModeRect) + kCallControlMargin * 3;
>-  _audioLoopButton.frame = CGRectMake(kCallControlMargin,
>-                                      audioLoopTop,
>-                                      _audioLoopButton.frame.size.width,
>-                                      _audioLoopButton.frame.size.height);
>-
>-  CGFloat startCallTop =
>-      CGRectGetMaxY(_audioLoopButton.frame) + kCallControlMargin * 3;
>-  _startCallButton.frame = CGRectMake(kCallControlMargin,
>-                                      startCallTop,
>-                                      _startCallButton.frame.size.width,
>-                                      _startCallButton.frame.size.height);
>+      CGRectMake(kRoomTextFieldMargin, kRoomTextFieldMargin, roomTextWidth,
>+                 roomTextHeight);
>+
>+  CGFloat buttonHeight =
>+      (CGRectGetMaxY(self.bounds) - CGRectGetMaxY(_roomText.frame) - kCallControlMargin * 4) / 3;
>+
>+  CGFloat regularCallFrameTop = CGRectGetMaxY(_roomText.frame) + kCallControlMargin;
>+  CGRect regularCallFrame = CGRectMake(kCallControlMargin,
>+                                       regularCallFrameTop,
>+                                       bounds.size.width - 2*kCallControlMargin,
>+                                       buttonHeight);
>+
>+  CGFloat loopbackCallFrameTop = CGRectGetMaxY(regularCallFrame) + kCallControlMargin;
>+  CGRect loopbackCallFrame = CGRectMake(kCallControlMargin,
>+                                        loopbackCallFrameTop,
>+                                        bounds.size.width - 2*kCallControlMargin,
>+                                        buttonHeight);
>+
>+  CGFloat audioLoopTop = CGRectGetMaxY(loopbackCallFrame) + kCallControlMargin;
>+  CGRect audioLoopFrame = CGRectMake(kCallControlMargin,
>+                                     audioLoopTop,
>+                                     bounds.size.width - 2*kCallControlMargin,
>+                                     buttonHeight);
>+
>+  _startRegularCallButton.frame = regularCallFrame;
>+  _startLoopbackCallButton.frame = loopbackCallFrame;
>+  _audioLoopButton.frame = audioLoopFrame;
> }
> 
> #pragma mark - Private
> 
> - (void)updateAudioLoopButton {
>   if (_isAudioLoopPlaying) {
>-    [_audioLoopButton setTitle:@"Stop sound"
>-                      forState:UIControlStateNormal];
>-    [_audioLoopButton sizeToFit];
>+    [_audioLoopButton setTitle:@"Stop sound" forState:UIControlStateNormal];
>   } else {
>-    [_audioLoopButton setTitle:@"Play sound"
>-                      forState:UIControlStateNormal];
>-    [_audioLoopButton sizeToFit];
>+    [_audioLoopButton setTitle:@"Play sound" forState:UIControlStateNormal];
>   }
> }
> 
>@@ -198,8 +185,12 @@ - (void)onToggleAudioLoop:(id)sender {
>   [_delegate mainViewDidToggleAudioLoop:self];
> }
> 
>-- (void)onStartCall:(id)sender {
>-  [_delegate mainView:self didInputRoom:_roomText.roomText isLoopback:_loopbackSwitch.isOn];
>+- (void)onStartRegularCall:(id)sender {
>+  [_delegate mainView:self didInputRoom:_roomText.roomText isLoopback:NO];
>+}
>+
>+- (void)onStartLoopbackCall:(id)sender {
>+  [_delegate mainView:self didInputRoom:_roomText.roomText isLoopback:YES];
> }
> 
> @end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ios/ARDSettingsViewController.m b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ios/ARDSettingsViewController.m
>index 27aa757ff74..f2fea15c094 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ios/ARDSettingsViewController.m
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ios/ARDSettingsViewController.m
>@@ -24,7 +24,6 @@ typedef NS_ENUM(int, ARDSettingsSections) {
> typedef NS_ENUM(int, ARDAudioSettingsOptions) {
>   ARDAudioSettingsAudioOnly = 0,
>   ARDAudioSettingsCreateAecDump,
>-  ARDAudioSettingsUseLevelController,
>   ARDAudioSettingsUseManualAudioConfig,
> };
> 
>@@ -92,7 +91,7 @@ - (NSInteger)numberOfSectionsInTableView:(UITableView *)tableView {
> - (NSInteger)tableView:(UITableView *)tableView numberOfRowsInSection:(NSInteger)section {
>   switch (section) {
>     case ARDSettingsSectionAudioSettings:
>-      return 4;
>+      return 3;
>     case ARDSettingsSectionVideoResolution:
>       return self.videoResolutionArray.count;
>     case ARDSettingsSectionVideoCodec:
>@@ -319,8 +318,6 @@ - (NSString *)labelForAudioSettingAtIndexPathRow:(NSInteger)setting {
>       return @"Audio only";
>     case ARDAudioSettingsCreateAecDump:
>       return @"Create AecDump";
>-    case ARDAudioSettingsUseLevelController:
>-      return @"Use level controller";
>     case ARDAudioSettingsUseManualAudioConfig:
>       return @"Use manual audio config";
>     default:
>@@ -334,8 +331,6 @@ - (BOOL)valueForAudioSettingAtIndexPathRow:(NSInteger)setting {
>       return [_settingsModel currentAudioOnlySettingFromStore];
>     case ARDAudioSettingsCreateAecDump:
>       return [_settingsModel currentCreateAecDumpSettingFromStore];
>-    case ARDAudioSettingsUseLevelController:
>-      return [_settingsModel currentUseLevelControllerSettingFromStore];
>     case ARDAudioSettingsUseManualAudioConfig:
>       return [_settingsModel currentUseManualAudioConfigSettingFromStore];
>     default:
>@@ -353,10 +348,6 @@ - (void)audioSettingSwitchChanged:(UISwitch *)sender {
>       [_settingsModel storeCreateAecDumpSetting:sender.isOn];
>       break;
>     }
>-    case ARDAudioSettingsUseLevelController: {
>-      [_settingsModel storeUseLevelControllerSetting:sender.isOn];
>-      break;
>-    }
>     case ARDAudioSettingsUseManualAudioConfig: {
>       [_settingsModel storeUseManualAudioConfigSetting:sender.isOn];
>       break;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ios/ARDVideoCallViewController.m b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ios/ARDVideoCallViewController.m
>index bf349ceea96..dbd78ea279f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ios/ARDVideoCallViewController.m
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ios/ARDVideoCallViewController.m
>@@ -18,7 +18,6 @@
> #import "ARDFileCaptureController.h"
> #import "ARDSettingsModel.h"
> #import "ARDVideoCallView.h"
>-#import "WebRTC/RTCAVFoundationVideoSource.h"
> #import "WebRTC/RTCDispatcher.h"
> #import "WebRTC/RTCLogging.h"
> #import "WebRTC/RTCMediaConstraints.h"
>@@ -125,7 +124,11 @@ - (void)appClient:(ARDAppClient *)client
> - (void)appClient:(ARDAppClient *)client
>     didReceiveRemoteVideoTrack:(RTCVideoTrack *)remoteVideoTrack {
>   self.remoteVideoTrack = remoteVideoTrack;
>-  _videoCallView.statusLabel.hidden = YES;
>+  __weak ARDVideoCallViewController *weakSelf = self;
>+  dispatch_async(dispatch_get_main_queue(), ^{
>+    ARDVideoCallViewController *strongSelf = weakSelf;
>+    strongSelf.videoCallView.statusLabel.hidden = YES;
>+  });
> }
> 
> - (void)appClient:(ARDAppClient *)client
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/ARDBroadcastSampleHandler.h b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/ARDBroadcastSampleHandler.h
>new file mode 100644
>index 00000000000..2218261991f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/ARDBroadcastSampleHandler.h
>@@ -0,0 +1,25 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#import <ReplayKit/ReplayKit.h>
>+
>+#import "WebRTC/RTCLogging.h"
>+
>+#import "ARDAppClient.h"
>+
>+@protocol ARDExternalSampleDelegate;
>+
>+API_AVAILABLE(ios(10.0))
>+@interface ARDBroadcastSampleHandler
>+    : RPBroadcastSampleHandler<ARDAppClientDelegate>
>+
>+@property(nonatomic, strong) id<ARDExternalSampleDelegate> capturer;
>+
>+@end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/ARDBroadcastSampleHandler.m b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/ARDBroadcastSampleHandler.m
>new file mode 100644
>index 00000000000..d54bec7583a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/ARDBroadcastSampleHandler.m
>@@ -0,0 +1,130 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#import "ARDBroadcastSampleHandler.h"
>+
>+#import <os/log.h>
>+
>+#import "ARDExternalSampleCapturer.h"
>+#import "ARDSettingsModel.h"
>+
>+#import "WebRTC/RTCCallbackLogger.h"
>+#import "WebRTC/RTCLogging.h"
>+
>+@implementation ARDBroadcastSampleHandler {
>+  ARDAppClient *_client;
>+  RTCCallbackLogger *_callbackLogger;
>+}
>+
>+@synthesize capturer = _capturer;
>+
>+- (instancetype)init {
>+  if (self = [super init]) {
>+    _callbackLogger = [[RTCCallbackLogger alloc] init];
>+    os_log_t rtc_os_log = os_log_create("com.google.AppRTCMobile", "RTCLog");
>+    [_callbackLogger start:^(NSString *logMessage) {
>+      os_log(rtc_os_log, "%{public}s", [logMessage cStringUsingEncoding:NSUTF8StringEncoding]);
>+    }];
>+  }
>+  return self;
>+}
>+
>+- (void)broadcastStartedWithSetupInfo:(NSDictionary<NSString *, NSObject *> *)setupInfo {
>+  // User has requested to start the broadcast. Setup info from the UI extension can be supplied but
>+  // optional.
>+  ARDSettingsModel *settingsModel = [[ARDSettingsModel alloc] init];
>+
>+  _client = [[ARDAppClient alloc] initWithDelegate:self];
>+  _client.broadcast = YES;
>+
>+  NSString *roomName = nil;
>+  if (setupInfo[@"roomName"]) {
>+    roomName = (NSString *)setupInfo[@"roomName"];
>+  } else {
>+    u_int32_t randomRoomSuffix = arc4random_uniform(1000);
>+    roomName = [NSString stringWithFormat:@"broadcast_%d", randomRoomSuffix];
>+  }
>+  [_client connectToRoomWithId:roomName settings:settingsModel isLoopback:NO];
>+  RTCLog(@"Broadcast started.");
>+}
>+
>+- (void)broadcastPaused {
>+  // User has requested to pause the broadcast. Samples will stop being delivered.
>+}
>+
>+- (void)broadcastResumed {
>+  // User has requested to resume the broadcast. Samples delivery will resume.
>+}
>+
>+- (void)broadcastFinished {
>+  // User has requested to finish the broadcast.
>+  [_client disconnect];
>+}
>+
>+- (void)processSampleBuffer:(CMSampleBufferRef)sampleBuffer
>+                   withType:(RPSampleBufferType)sampleBufferType {
>+  switch (sampleBufferType) {
>+    case RPSampleBufferTypeVideo:
>+      [self.capturer didCaptureSampleBuffer:sampleBuffer];
>+      break;
>+    case RPSampleBufferTypeAudioApp:
>+      break;
>+    case RPSampleBufferTypeAudioMic:
>+      break;
>+    default:
>+      break;
>+  }
>+}
>+
>+#pragma mark - ARDAppClientDelegate
>+
>+- (void)appClient:(ARDAppClient *)client didChangeState:(ARDAppClientState)state {
>+  switch (state) {
>+    case kARDAppClientStateConnected:
>+      RTCLog(@"Client connected.");
>+      break;
>+    case kARDAppClientStateConnecting:
>+      RTCLog("Client connecting.");
>+      break;
>+    case kARDAppClientStateDisconnected:
>+      RTCLog(@"Client disconnected.");
>+      break;
>+  }
>+}
>+
>+- (void)appClient:(ARDAppClient *)client didChangeConnectionState:(RTCIceConnectionState)state {
>+  RTCLog(@"ICE state changed: %ld", (long)state);
>+}
>+
>+- (void)appClient:(ARDAppClient *)client
>+    didCreateLocalCapturer:(RTCCameraVideoCapturer *)localCapturer {
>+}
>+
>+- (void)appClient:(ARDAppClient *)client
>+    didCreateLocalExternalSampleCapturer:(ARDExternalSampleCapturer *)externalSampleCapturer {
>+  self.capturer = externalSampleCapturer;
>+}
>+
>+- (void)appClient:(ARDAppClient *)client
>+    didReceiveLocalVideoTrack:(RTCVideoTrack *)localVideoTrack {
>+}
>+
>+- (void)appClient:(ARDAppClient *)client
>+    didReceiveRemoteVideoTrack:(RTCVideoTrack *)remoteVideoTrack {
>+}
>+
>+- (void)appClient:(ARDAppClient *)client didGetStats:(NSArray *)stats {
>+}
>+
>+- (void)appClient:(ARDAppClient *)client didError:(NSError *)error {
>+  RTCLog(@"Error: %@", error);
>+}
>+
>+@end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/ARDBroadcastSetupViewController.h b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/ARDBroadcastSetupViewController.h
>new file mode 100644
>index 00000000000..e95c5cc226d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/ARDBroadcastSetupViewController.h
>@@ -0,0 +1,18 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#import <ReplayKit/ReplayKit.h>
>+#import <UIKit/UIKit.h>
>+
>+API_AVAILABLE(ios(11.0))
>+@interface ARDBroadcastSetupViewController
>+    : UIViewController<UITextFieldDelegate>
>+
>+@end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/ARDBroadcastSetupViewController.m b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/ARDBroadcastSetupViewController.m
>new file mode 100644
>index 00000000000..55438f17d87
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/ARDBroadcastSetupViewController.m
>@@ -0,0 +1,107 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#import "ARDBroadcastSetupViewController.h"
>+
>+@implementation ARDBroadcastSetupViewController {
>+  UITextField *_roomNameField;
>+}
>+
>+- (void)loadView {
>+  UIView *view = [[UIView alloc] initWithFrame:CGRectZero];
>+  view.backgroundColor = [UIColor colorWithWhite:1.0 alpha:0.7];
>+
>+  UIImageView *imageView = [[UIImageView alloc] initWithImage:[UIImage imageNamed:@"Icon-180"]];
>+  imageView.translatesAutoresizingMaskIntoConstraints = NO;
>+  [view addSubview:imageView];
>+
>+  _roomNameField = [[UITextField alloc] initWithFrame:CGRectZero];
>+  _roomNameField.borderStyle = UITextBorderStyleRoundedRect;
>+  _roomNameField.font = [UIFont systemFontOfSize:14.0];
>+  _roomNameField.translatesAutoresizingMaskIntoConstraints = NO;
>+  _roomNameField.placeholder = @"Room name";
>+  _roomNameField.returnKeyType = UIReturnKeyDone;
>+  _roomNameField.delegate = self;
>+  [view addSubview:_roomNameField];
>+
>+  UIButton *doneButton = [UIButton buttonWithType:UIButtonTypeSystem];
>+  doneButton.translatesAutoresizingMaskIntoConstraints = NO;
>+  doneButton.titleLabel.font = [UIFont systemFontOfSize:20.0];
>+  [doneButton setTitle:@"Done" forState:UIControlStateNormal];
>+  [doneButton addTarget:self
>+                 action:@selector(userDidFinishSetup)
>+       forControlEvents:UIControlEventTouchUpInside];
>+  [view addSubview:doneButton];
>+
>+  UIButton *cancelButton = [UIButton buttonWithType:UIButtonTypeSystem];
>+  cancelButton.translatesAutoresizingMaskIntoConstraints = NO;
>+  cancelButton.titleLabel.font = [UIFont systemFontOfSize:20.0];
>+  [cancelButton setTitle:@"Cancel" forState:UIControlStateNormal];
>+  [cancelButton addTarget:self
>+                   action:@selector(userDidCancelSetup)
>+         forControlEvents:UIControlEventTouchUpInside];
>+  [view addSubview:cancelButton];
>+
>+  UILayoutGuide *margin = view.layoutMarginsGuide;
>+  [imageView.widthAnchor constraintEqualToConstant:60.0].active = YES;
>+  [imageView.heightAnchor constraintEqualToConstant:60.0].active = YES;
>+  [imageView.topAnchor constraintEqualToAnchor:margin.topAnchor constant:20].active = YES;
>+  [imageView.centerXAnchor constraintEqualToAnchor:view.centerXAnchor].active = YES;
>+
>+  [_roomNameField.leadingAnchor constraintEqualToAnchor:margin.leadingAnchor].active = YES;
>+  [_roomNameField.topAnchor constraintEqualToAnchor:imageView.bottomAnchor constant:20].active =
>+      YES;
>+  [_roomNameField.trailingAnchor constraintEqualToAnchor:margin.trailingAnchor].active = YES;
>+
>+  [doneButton.leadingAnchor constraintEqualToAnchor:margin.leadingAnchor].active = YES;
>+  [doneButton.bottomAnchor constraintEqualToAnchor:margin.bottomAnchor constant:-20].active = YES;
>+
>+  [cancelButton.trailingAnchor constraintEqualToAnchor:margin.trailingAnchor].active = YES;
>+  [cancelButton.bottomAnchor constraintEqualToAnchor:margin.bottomAnchor constant:-20].active = YES;
>+
>+  UITapGestureRecognizer *tgr =
>+      [[UITapGestureRecognizer alloc] initWithTarget:self action:@selector(didTap:)];
>+  [view addGestureRecognizer:tgr];
>+
>+  self.view = view;
>+}
>+
>+- (IBAction)didTap:(id)sender {
>+  [self.view endEditing:YES];
>+}
>+
>+- (void)userDidFinishSetup {
>+  // URL of the resource where broadcast can be viewed that will be returned to the application
>+  NSURL *broadcastURL = [NSURL
>+      URLWithString:[NSString stringWithFormat:@"https://appr.tc/r/%@", _roomNameField.text]];
>+
>+  // Dictionary with setup information that will be provided to broadcast extension when broadcast
>+  // is started
>+  NSDictionary *setupInfo = @{@"roomName" : _roomNameField.text};
>+
>+  // Tell ReplayKit that the extension is finished setting up and can begin broadcasting
>+  [self.extensionContext completeRequestWithBroadcastURL:broadcastURL setupInfo:setupInfo];
>+}
>+
>+- (void)userDidCancelSetup {
>+  // Tell ReplayKit that the extension was cancelled by the user
>+  [self.extensionContext cancelRequestWithError:[NSError errorWithDomain:@"com.google.AppRTCMobile"
>+                                                                    code:-1
>+                                                                userInfo:nil]];
>+}
>+
>+#pragma mark - UITextFieldDelegate
>+
>+- (BOOL)textFieldShouldReturn:(UITextField *)textField {
>+  [self userDidFinishSetup];
>+  return YES;
>+}
>+
>+@end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/BroadcastSetupUIInfo.plist b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/BroadcastSetupUIInfo.plist
>new file mode 100644
>index 00000000000..a123c111e55
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/BroadcastSetupUIInfo.plist
>@@ -0,0 +1,39 @@
>+<?xml version="1.0" encoding="UTF-8"?>
>+<!DOCTYPE plist PUBLIC "-//Apple//DTD PLIST 1.0//EN" "http://www.apple.com/DTDs/PropertyList-1.0.dtd">
>+<plist version="1.0">
>+<dict>
>+	<key>CFBundleDevelopmentRegion</key>
>+	<string>en</string>
>+	<key>CFBundleDisplayName</key>
>+	<string>AppRTCMobile</string>
>+	<key>CFBundleExecutable</key>
>+	<string>$(EXECUTABLE_NAME)</string>
>+	<key>CFBundleIdentifier</key>
>+	<string>com.google.AppRTCMobile.BroadcastSetupUI</string>
>+	<key>CFBundleInfoDictionaryVersion</key>
>+	<string>6.0</string>
>+	<key>CFBundleName</key>
>+	<string>$(PRODUCT_NAME)</string>
>+	<key>CFBundlePackageType</key>
>+	<string>XPC!</string>
>+	<key>CFBundleShortVersionString</key>
>+	<string>1.0</string>
>+	<key>CFBundleVersion</key>
>+	<string>1</string>
>+	<key>NSExtension</key>
>+	<dict>
>+		<key>NSExtensionAttributes</key>
>+		<dict>
>+			<key>NSExtensionActivationRule</key>
>+			<dict>
>+				<key>NSExtensionActivationSupportsReplayKitStreaming</key>
>+				<true/>
>+			</dict>
>+		</dict>
>+		<key>NSExtensionPointIdentifier</key>
>+		<string>com.apple.broadcast-services-setupui</string>
>+		<key>NSExtensionPrincipalClass</key>
>+		<string>ARDBroadcastSetupViewController</string>
>+	</dict>
>+</dict>
>+</plist>
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/BroadcastUploadInfo.plist b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/BroadcastUploadInfo.plist
>new file mode 100644
>index 00000000000..2bab60ea8fb
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ios/broadcast_extension/BroadcastUploadInfo.plist
>@@ -0,0 +1,33 @@
>+<?xml version="1.0" encoding="UTF-8"?>
>+<!DOCTYPE plist PUBLIC "-//Apple//DTD PLIST 1.0//EN" "http://www.apple.com/DTDs/PropertyList-1.0.dtd">
>+<plist version="1.0">
>+<dict>
>+	<key>CFBundleDevelopmentRegion</key>
>+	<string>en</string>
>+	<key>CFBundleDisplayName</key>
>+	<string>AppRTCMobile</string>
>+	<key>CFBundleExecutable</key>
>+	<string>$(EXECUTABLE_NAME)</string>
>+	<key>CFBundleIdentifier</key>
>+	<string>com.google.AppRTCMobile.BroadcastUpload</string>
>+	<key>CFBundleInfoDictionaryVersion</key>
>+	<string>6.0</string>
>+	<key>CFBundleName</key>
>+	<string>$(PRODUCT_NAME)</string>
>+	<key>CFBundlePackageType</key>
>+	<string>XPC!</string>
>+	<key>CFBundleShortVersionString</key>
>+	<string>1.0</string>
>+	<key>CFBundleVersion</key>
>+	<string>1</string>
>+	<key>NSExtension</key>
>+	<dict>
>+		<key>NSExtensionPointIdentifier</key>
>+		<string>com.apple.broadcast-services-upload</string>
>+		<key>NSExtensionPrincipalClass</key>
>+		<string>ARDBroadcastSampleHandler</string>
>+		<key>RPBroadcastProcessMode</key>
>+		<string>RPBroadcastProcessModeSampleBuffer</string>
>+	</dict>
>+</dict>
>+</plist>
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ios/resources/foreman.mp4 b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ios/resources/foreman.mp4
>new file mode 100644
>index 00000000000..ccffbf47225
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ios/resources/foreman.mp4 differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/Icon-120.png b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/Icon-120.png
>new file mode 100644
>index 00000000000..938fef477b0
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/Icon-120.png differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/Icon-180.png b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/Icon-180.png
>new file mode 100644
>index 00000000000..a5b76096800
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/Icon-180.png differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objcnativeapi/Info.plist b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objcnativeapi/Info.plist
>new file mode 100644
>index 00000000000..cbc9e5f9f3c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objcnativeapi/Info.plist
>@@ -0,0 +1,45 @@
>+<?xml version="1.0" encoding="UTF-8"?>
>+<!DOCTYPE plist PUBLIC "-//Apple//DTD PLIST 1.0//EN" "http://www.apple.com/DTDs/PropertyList-1.0.dtd">
>+<plist version="1.0">
>+<dict>
>+  <key>CFBundleDevelopmentRegion</key>
>+  <string>en</string>
>+  <key>CFBundleExecutable</key>
>+  <string>$(EXECUTABLE_NAME)</string>
>+  <key>CFBundleIdentifier</key>
>+  <string>com.google.ObjCNativeAPIDemo</string>
>+  <key>CFBundleInfoDictionaryVersion</key>
>+  <string>6.0</string>
>+  <key>CFBundleName</key>
>+  <string>ObjCNativeAPIDemo</string>
>+  <key>CFBundlePackageType</key>
>+  <string>APPL</string>
>+  <key>CFBundleShortVersionString</key>
>+  <string>1.0</string>
>+  <key>CFBundleVersion</key>
>+  <string>1</string>
>+  <key>LSRequiresIPhoneOS</key>
>+  <true/>
>+  <key>UIRequiredDeviceCapabilities</key>
>+  <array>
>+    <string>armv7</string>
>+  </array>
>+  <key>UISupportedInterfaceOrientations</key>
>+  <array>
>+    <string>UIInterfaceOrientationPortrait</string>
>+    <string>UIInterfaceOrientationLandscapeLeft</string>
>+    <string>UIInterfaceOrientationLandscapeRight</string>
>+  </array>
>+  <key>UISupportedInterfaceOrientations~ipad</key>
>+  <array>
>+    <string>UIInterfaceOrientationPortrait</string>
>+    <string>UIInterfaceOrientationPortraitUpsideDown</string>
>+    <string>UIInterfaceOrientationLandscapeLeft</string>
>+    <string>UIInterfaceOrientationLandscapeRight</string>
>+  </array>
>+  <key>NSCameraUsageDescription</key>
>+  <string>Camera access needed for video calling</string>
>+  <key>NSMicrophoneUsageDescription</key>
>+  <string>Microphone access needed for video calling</string>
>+</dict>
>+</plist>
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/SDK/PodTest/PodTest/AppDelegate.h b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objcnativeapi/objc/NADAppDelegate.h
>similarity index 68%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/SDK/PodTest/PodTest/AppDelegate.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/examples/objcnativeapi/objc/NADAppDelegate.h
>index de2cc0465db..d4211208427 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/SDK/PodTest/PodTest/AppDelegate.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objcnativeapi/objc/NADAppDelegate.h
>@@ -1,5 +1,5 @@
> /*
>- *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>  *
>  *  Use of this source code is governed by a BSD-style license
>  *  that can be found in the LICENSE file in the root of the source
>@@ -10,8 +10,8 @@
> 
> #import <UIKit/UIKit.h>
> 
>-@interface AppDelegate : UIResponder <UIApplicationDelegate>
>+@interface NADAppDelegate : UIResponder<UIApplicationDelegate>
> 
>-@property (strong, nonatomic) UIWindow *window;
>+@property(strong, nonatomic) UIWindow* window;
> 
> @end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objcnativeapi/objc/NADAppDelegate.m b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objcnativeapi/objc/NADAppDelegate.m
>new file mode 100644
>index 00000000000..254dd3be76f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objcnativeapi/objc/NADAppDelegate.m
>@@ -0,0 +1,63 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#import "NADAppDelegate.h"
>+
>+#import "NADViewController.h"
>+
>+@interface NADAppDelegate ()
>+@end
>+
>+@implementation NADAppDelegate
>+
>+@synthesize window = _window;
>+
>+- (BOOL)application:(UIApplication *)application
>+    didFinishLaunchingWithOptions:(NSDictionary *)launchOptions {
>+  _window = [[UIWindow alloc] initWithFrame:[[UIScreen mainScreen] bounds]];
>+  [_window makeKeyAndVisible];
>+
>+  NADViewController *viewController = [[NADViewController alloc] init];
>+  _window.rootViewController = viewController;
>+
>+  return YES;
>+}
>+
>+- (void)applicationWillResignActive:(UIApplication *)application {
>+  // Sent when the application is about to move from active to inactive state. This can occur for
>+  // certain types of temporary interruptions (such as an incoming phone call or SMS message) or
>+  // when the user quits the application and it begins the transition to the background state. Use
>+  // this method to pause ongoing tasks, disable timers, and invalidate graphics rendering
>+  // callbacks. Games should use this method to pause the game.
>+}
>+
>+- (void)applicationDidEnterBackground:(UIApplication *)application {
>+  // Use this method to release shared resources, save user data, invalidate timers, and store
>+  // enough application state information to restore your application to its current state in case
>+  // it is terminated later. If your application supports background execution, this method is
>+  // called instead of applicationWillTerminate: when the user quits.
>+}
>+
>+- (void)applicationWillEnterForeground:(UIApplication *)application {
>+  // Called as part of the transition from the background to the active state; here you can undo
>+  // many of the changes made on entering the background.
>+}
>+
>+- (void)applicationDidBecomeActive:(UIApplication *)application {
>+  // Restart any tasks that were paused (or not yet started) while the application was inactive. If
>+  // the application was previously in the background, optionally refresh the user interface.
>+}
>+
>+- (void)applicationWillTerminate:(UIApplication *)application {
>+  // Called when the application is about to terminate. Save data if appropriate. See also
>+  // applicationDidEnterBackground:.
>+}
>+
>+@end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/SDK/PodTest/PodTest/ViewController.h b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objcnativeapi/objc/NADViewController.h
>similarity index 77%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/SDK/PodTest/PodTest/ViewController.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/examples/objcnativeapi/objc/NADViewController.h
>index bdf8f31856d..c43bebb52d4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/SDK/PodTest/PodTest/ViewController.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objcnativeapi/objc/NADViewController.h
>@@ -1,5 +1,5 @@
> /*
>- *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>  *
>  *  Use of this source code is governed by a BSD-style license
>  *  that can be found in the LICENSE file in the root of the source
>@@ -10,6 +10,6 @@
> 
> #import <UIKit/UIKit.h>
> 
>-@interface ViewController : UIViewController
>+@interface NADViewController : UIViewController
> 
> @end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objcnativeapi/objc/NADViewController.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objcnativeapi/objc/NADViewController.mm
>new file mode 100644
>index 00000000000..0d5ac9a3b91
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objcnativeapi/objc/NADViewController.mm
>@@ -0,0 +1,157 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#import "NADViewController.h"
>+
>+#import <WebRTC/RTCCameraPreviewView.h>
>+#import <WebRTC/RTCCameraVideoCapturer.h>
>+#import <WebRTC/RTCEAGLVideoView.h>
>+#import <WebRTC/RTCMTLVideoView.h>
>+#import <WebRTC/RTCVideoRenderer.h>
>+#include <memory>
>+
>+#include "objccallclient.h"
>+
>+@interface NADViewController ()
>+
>+@property(nonatomic) RTCCameraVideoCapturer *capturer;
>+@property(nonatomic) RTCCameraPreviewView *localVideoView;
>+@property(nonatomic) __kindof UIView<RTCVideoRenderer> *remoteVideoView;
>+@property(nonatomic) UIButton *callButton;
>+@property(nonatomic) UIButton *hangUpButton;
>+
>+@end
>+
>+@implementation NADViewController {
>+  std::unique_ptr<webrtc_examples::ObjCCallClient> _call_client;
>+
>+  UIView *_view;
>+}
>+
>+@synthesize capturer = _capturer;
>+@synthesize localVideoView = _localVideoView;
>+@synthesize remoteVideoView = _remoteVideoView;
>+@synthesize callButton = _callButton;
>+@synthesize hangUpButton = _hangUpButton;
>+
>+#pragma mark - View controller lifecycle
>+
>+- (void)loadView {
>+  _view = [[UIView alloc] initWithFrame:CGRectZero];
>+
>+#if defined(RTC_SUPPORTS_METAL)
>+  _remoteVideoView = [[RTCMTLVideoView alloc] initWithFrame:CGRectZero];
>+#else
>+  _remoteVideoView = [[RTCEAGLVideoView alloc] initWithFrame:CGRectZero];
>+#endif
>+  _remoteVideoView.translatesAutoresizingMaskIntoConstraints = NO;
>+  [_view addSubview:_remoteVideoView];
>+
>+  _localVideoView = [[RTCCameraPreviewView alloc] initWithFrame:CGRectZero];
>+  _localVideoView.translatesAutoresizingMaskIntoConstraints = NO;
>+  [_view addSubview:_localVideoView];
>+
>+  _callButton = [UIButton buttonWithType:UIButtonTypeSystem];
>+  _callButton.translatesAutoresizingMaskIntoConstraints = NO;
>+  [_callButton setTitle:@"Call" forState:UIControlStateNormal];
>+  [_callButton addTarget:self action:@selector(call:) forControlEvents:UIControlEventTouchUpInside];
>+  [_view addSubview:_callButton];
>+
>+  _hangUpButton = [UIButton buttonWithType:UIButtonTypeSystem];
>+  _hangUpButton.translatesAutoresizingMaskIntoConstraints = NO;
>+  [_hangUpButton setTitle:@"Hang up" forState:UIControlStateNormal];
>+  [_hangUpButton addTarget:self
>+                    action:@selector(hangUp:)
>+          forControlEvents:UIControlEventTouchUpInside];
>+  [_view addSubview:_hangUpButton];
>+
>+  UILayoutGuide *margin = _view.layoutMarginsGuide;
>+  [_remoteVideoView.leadingAnchor constraintEqualToAnchor:margin.leadingAnchor].active = YES;
>+  [_remoteVideoView.topAnchor constraintEqualToAnchor:margin.topAnchor].active = YES;
>+  [_remoteVideoView.trailingAnchor constraintEqualToAnchor:margin.trailingAnchor].active = YES;
>+  [_remoteVideoView.bottomAnchor constraintEqualToAnchor:margin.bottomAnchor].active = YES;
>+
>+  [_localVideoView.leadingAnchor constraintEqualToAnchor:margin.leadingAnchor constant:8.0].active =
>+      YES;
>+  [_localVideoView.topAnchor constraintEqualToAnchor:margin.topAnchor constant:8.0].active = YES;
>+  [_localVideoView.widthAnchor constraintEqualToConstant:60].active = YES;
>+  [_localVideoView.heightAnchor constraintEqualToConstant:60].active = YES;
>+
>+  [_callButton.leadingAnchor constraintEqualToAnchor:margin.leadingAnchor constant:8.0].active =
>+      YES;
>+  [_callButton.bottomAnchor constraintEqualToAnchor:margin.bottomAnchor constant:8.0].active = YES;
>+  [_callButton.widthAnchor constraintEqualToConstant:100].active = YES;
>+  [_callButton.heightAnchor constraintEqualToConstant:40].active = YES;
>+
>+  [_hangUpButton.trailingAnchor constraintEqualToAnchor:margin.trailingAnchor constant:8.0].active =
>+      YES;
>+  [_hangUpButton.bottomAnchor constraintEqualToAnchor:margin.bottomAnchor constant:8.0].active =
>+      YES;
>+  [_hangUpButton.widthAnchor constraintEqualToConstant:100].active = YES;
>+  [_hangUpButton.heightAnchor constraintEqualToConstant:40].active = YES;
>+
>+  self.view = _view;
>+}
>+
>+- (void)viewDidLoad {
>+  [super viewDidLoad];
>+
>+  self.capturer = [[RTCCameraVideoCapturer alloc] init];
>+  self.localVideoView.captureSession = self.capturer.captureSession;
>+
>+  _call_client.reset(new webrtc_examples::ObjCCallClient());
>+
>+  // Start capturer.
>+  AVCaptureDevice *selectedDevice = nil;
>+  NSArray<AVCaptureDevice *> *captureDevices = [RTCCameraVideoCapturer captureDevices];
>+  for (AVCaptureDevice *device in captureDevices) {
>+    if (device.position == AVCaptureDevicePositionFront) {
>+      selectedDevice = device;
>+      break;
>+    }
>+  }
>+
>+  AVCaptureDeviceFormat *selectedFormat = nil;
>+  int targetWidth = 640;
>+  int targetHeight = 480;
>+  int currentDiff = INT_MAX;
>+  NSArray<AVCaptureDeviceFormat *> *formats =
>+      [RTCCameraVideoCapturer supportedFormatsForDevice:selectedDevice];
>+  for (AVCaptureDeviceFormat *format in formats) {
>+    CMVideoDimensions dimension = CMVideoFormatDescriptionGetDimensions(format.formatDescription);
>+    FourCharCode pixelFormat = CMFormatDescriptionGetMediaSubType(format.formatDescription);
>+    int diff = abs(targetWidth - dimension.width) + abs(targetHeight - dimension.height);
>+    if (diff < currentDiff) {
>+      selectedFormat = format;
>+      currentDiff = diff;
>+    } else if (diff == currentDiff && pixelFormat == [_capturer preferredOutputPixelFormat]) {
>+      selectedFormat = format;
>+    }
>+  }
>+
>+  [self.capturer startCaptureWithDevice:selectedDevice format:selectedFormat fps:30];
>+}
>+
>+- (void)didReceiveMemoryWarning {
>+  [super didReceiveMemoryWarning];
>+  // Dispose of any resources that can be recreated.
>+}
>+
>+#pragma mark - Actions
>+
>+- (IBAction)call:(id)sender {
>+  _call_client->Call(self.capturer, self.remoteVideoView);
>+}
>+
>+- (IBAction)hangUp:(id)sender {
>+  _call_client->Hangup();
>+}
>+
>+@end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/SDK/PodTest/PodTest/main.m b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objcnativeapi/objc/main.m
>similarity index 62%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/SDK/PodTest/PodTest/main.m
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/examples/objcnativeapi/objc/main.m
>index 4169aa9fabd..2c3b5fbbfb1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/SDK/PodTest/PodTest/main.m
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objcnativeapi/objc/main.m
>@@ -1,5 +1,5 @@
> /*
>- *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>  *
>  *  Use of this source code is governed by a BSD-style license
>  *  that can be found in the LICENSE file in the root of the source
>@@ -9,13 +9,10 @@
>  */
> 
> #import <UIKit/UIKit.h>
>-#import "AppDelegate.h"
>+#import "NADAppDelegate.h"
> 
> int main(int argc, char* argv[]) {
>   @autoreleasepool {
>-    return UIApplicationMain(argc,
>-                             argv,
>-                             nil,
>-                             NSStringFromClass([AppDelegate class]));
>+    return UIApplicationMain(argc, argv, nil, NSStringFromClass([NADAppDelegate class]));
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objcnativeapi/objc/objccallclient.h b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objcnativeapi/objc/objccallclient.h
>new file mode 100644
>index 00000000000..e48aae47e1e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objcnativeapi/objc/objccallclient.h
>@@ -0,0 +1,84 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef EXAMPLES_OBJCNATIVEAPI_OBJCCALLCLIENT_H_
>+#define EXAMPLES_OBJCNATIVEAPI_OBJCCALLCLIENT_H_
>+
>+#include <memory>
>+#include <string>
>+
>+#include "api/peerconnectioninterface.h"
>+#include "rtc_base/criticalsection.h"
>+#include "rtc_base/scoped_ref_ptr.h"
>+#include "rtc_base/thread_checker.h"
>+
>+@class RTCVideoCapturer;
>+@protocol RTCVideoRenderer;
>+
>+namespace webrtc_examples {
>+
>+class ObjCCallClient {
>+ public:
>+  ObjCCallClient();
>+
>+  void Call(RTCVideoCapturer* capturer, id<RTCVideoRenderer> remote_renderer);
>+  void Hangup();
>+
>+ private:
>+  class PCObserver : public webrtc::PeerConnectionObserver {
>+   public:
>+    explicit PCObserver(ObjCCallClient* client);
>+
>+    void OnSignalingChange(
>+        webrtc::PeerConnectionInterface::SignalingState new_state) override;
>+    void OnDataChannel(
>+        rtc::scoped_refptr<webrtc::DataChannelInterface> data_channel) override;
>+    void OnRenegotiationNeeded() override;
>+    void OnIceConnectionChange(
>+        webrtc::PeerConnectionInterface::IceConnectionState new_state) override;
>+    void OnIceGatheringChange(
>+        webrtc::PeerConnectionInterface::IceGatheringState new_state) override;
>+    void OnIceCandidate(
>+        const webrtc::IceCandidateInterface* candidate) override;
>+
>+   private:
>+    const ObjCCallClient* client_;
>+  };
>+
>+  void CreatePeerConnectionFactory() RTC_RUN_ON(thread_checker_);
>+  void CreatePeerConnection() RTC_RUN_ON(thread_checker_);
>+  void Connect() RTC_RUN_ON(thread_checker_);
>+
>+  rtc::ThreadChecker thread_checker_;
>+
>+  bool call_started_ RTC_GUARDED_BY(thread_checker_);
>+
>+  const std::unique_ptr<PCObserver> pc_observer_;
>+
>+  rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> pcf_
>+      RTC_GUARDED_BY(thread_checker_);
>+  std::unique_ptr<rtc::Thread> network_thread_ RTC_GUARDED_BY(thread_checker_);
>+  std::unique_ptr<rtc::Thread> worker_thread_ RTC_GUARDED_BY(thread_checker_);
>+  std::unique_ptr<rtc::Thread> signaling_thread_
>+      RTC_GUARDED_BY(thread_checker_);
>+
>+  std::unique_ptr<rtc::VideoSinkInterface<webrtc::VideoFrame>> remote_sink_
>+      RTC_GUARDED_BY(thread_checker_);
>+  rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> video_source_
>+      RTC_GUARDED_BY(thread_checker_);
>+
>+  rtc::CriticalSection pc_mutex_;
>+  rtc::scoped_refptr<webrtc::PeerConnectionInterface> pc_
>+      RTC_GUARDED_BY(pc_mutex_);
>+};
>+
>+}  // namespace webrtc_examples
>+
>+#endif  // EXAMPLES_OBJCNATIVEAPI_OBJCCALLCLIENT_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objcnativeapi/objc/objccallclient.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objcnativeapi/objc/objccallclient.mm
>new file mode 100644
>index 00000000000..0e3b8e05db0
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objcnativeapi/objc/objccallclient.mm
>@@ -0,0 +1,237 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "examples/objcnativeapi/objc/objccallclient.h"
>+
>+#include <utility>
>+
>+#import <WebRTC/RTCCameraPreviewView.h>
>+#import <WebRTC/RTCVideoCodecFactory.h>
>+#import <WebRTC/RTCVideoRenderer.h>
>+
>+#include "absl/memory/memory.h"
>+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
>+#include "api/audio_codecs/builtin_audio_encoder_factory.h"
>+#include "api/peerconnectioninterface.h"
>+#include "media/engine/webrtcmediaengine.h"
>+#include "modules/audio_processing/include/audio_processing.h"
>+#include "sdk/objc/Framework/Native/api/video_capturer.h"
>+#include "sdk/objc/Framework/Native/api/video_decoder_factory.h"
>+#include "sdk/objc/Framework/Native/api/video_encoder_factory.h"
>+#include "sdk/objc/Framework/Native/api/video_renderer.h"
>+
>+namespace webrtc_examples {
>+
>+namespace {
>+
>+class CreateOfferObserver : public webrtc::CreateSessionDescriptionObserver {
>+ public:
>+  explicit CreateOfferObserver(rtc::scoped_refptr<webrtc::PeerConnectionInterface> pc);
>+
>+  void OnSuccess(webrtc::SessionDescriptionInterface* desc) override;
>+  void OnFailure(webrtc::RTCError error) override;
>+
>+ private:
>+  const rtc::scoped_refptr<webrtc::PeerConnectionInterface> pc_;
>+};
>+
>+class SetRemoteSessionDescriptionObserver : public webrtc::SetRemoteDescriptionObserverInterface {
>+ public:
>+  void OnSetRemoteDescriptionComplete(webrtc::RTCError error) override;
>+};
>+
>+class SetLocalSessionDescriptionObserver : public webrtc::SetSessionDescriptionObserver {
>+ public:
>+  void OnSuccess() override;
>+  void OnFailure(webrtc::RTCError error) override;
>+};
>+
>+}  // namespace
>+
>+ObjCCallClient::ObjCCallClient()
>+    : call_started_(false), pc_observer_(absl::make_unique<PCObserver>(this)) {
>+  thread_checker_.DetachFromThread();
>+  CreatePeerConnectionFactory();
>+}
>+
>+void ObjCCallClient::Call(RTCVideoCapturer* capturer, id<RTCVideoRenderer> remote_renderer) {
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+
>+  rtc::CritScope lock(&pc_mutex_);
>+  if (call_started_) {
>+    RTC_LOG(LS_WARNING) << "Call already started.";
>+    return;
>+  }
>+  call_started_ = true;
>+
>+  remote_sink_ = webrtc::ObjCToNativeVideoRenderer(remote_renderer);
>+
>+  video_source_ =
>+      webrtc::ObjCToNativeVideoCapturer(capturer, signaling_thread_.get(), worker_thread_.get());
>+
>+  CreatePeerConnection();
>+  Connect();
>+}
>+
>+void ObjCCallClient::Hangup() {
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+
>+  call_started_ = false;
>+
>+  {
>+    rtc::CritScope lock(&pc_mutex_);
>+    if (pc_ != nullptr) {
>+      pc_->Close();
>+      pc_ = nullptr;
>+    }
>+  }
>+
>+  remote_sink_ = nullptr;
>+  video_source_ = nullptr;
>+}
>+
>+void ObjCCallClient::CreatePeerConnectionFactory() {
>+  network_thread_ = rtc::Thread::CreateWithSocketServer();
>+  network_thread_->SetName("network_thread", nullptr);
>+  RTC_CHECK(network_thread_->Start()) << "Failed to start thread";
>+
>+  worker_thread_ = rtc::Thread::Create();
>+  worker_thread_->SetName("worker_thread", nullptr);
>+  RTC_CHECK(worker_thread_->Start()) << "Failed to start thread";
>+
>+  signaling_thread_ = rtc::Thread::Create();
>+  signaling_thread_->SetName("signaling_thread", nullptr);
>+  RTC_CHECK(signaling_thread_->Start()) << "Failed to start thread";
>+
>+  std::unique_ptr<webrtc::VideoDecoderFactory> videoDecoderFactory =
>+      webrtc::ObjCToNativeVideoDecoderFactory([[RTCDefaultVideoDecoderFactory alloc] init]);
>+  std::unique_ptr<webrtc::VideoEncoderFactory> videoEncoderFactory =
>+      webrtc::ObjCToNativeVideoEncoderFactory([[RTCDefaultVideoEncoderFactory alloc] init]);
>+
>+  std::unique_ptr<cricket::MediaEngineInterface> media_engine =
>+      cricket::WebRtcMediaEngineFactory::Create(nullptr /* adm */,
>+                                                webrtc::CreateBuiltinAudioEncoderFactory(),
>+                                                webrtc::CreateBuiltinAudioDecoderFactory(),
>+                                                std::move(videoEncoderFactory),
>+                                                std::move(videoDecoderFactory),
>+                                                nullptr /* audio_mixer */,
>+                                                webrtc::AudioProcessingBuilder().Create());
>+  RTC_LOG(LS_INFO) << "Media engine created: " << media_engine.get();
>+
>+  pcf_ = webrtc::CreateModularPeerConnectionFactory(network_thread_.get(),
>+                                                    worker_thread_.get(),
>+                                                    signaling_thread_.get(),
>+                                                    std::move(media_engine),
>+                                                    webrtc::CreateCallFactory(),
>+                                                    webrtc::CreateRtcEventLogFactory());
>+  RTC_LOG(LS_INFO) << "PeerConnectionFactory created: " << pcf_;
>+}
>+
>+void ObjCCallClient::CreatePeerConnection() {
>+  rtc::CritScope lock(&pc_mutex_);
>+  webrtc::PeerConnectionInterface::RTCConfiguration config;
>+  config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan;
>+  // DTLS SRTP has to be disabled for loopback to work.
>+  config.enable_dtls_srtp = false;
>+  pc_ = pcf_->CreatePeerConnection(
>+      config, nullptr /* port_allocator */, nullptr /* cert_generator */, pc_observer_.get());
>+  RTC_LOG(LS_INFO) << "PeerConnection created: " << pc_;
>+
>+  rtc::scoped_refptr<webrtc::VideoTrackInterface> local_video_track =
>+      pcf_->CreateVideoTrack("video", video_source_);
>+  pc_->AddTransceiver(local_video_track);
>+  RTC_LOG(LS_INFO) << "Local video sink set up: " << local_video_track;
>+
>+  for (const rtc::scoped_refptr<webrtc::RtpTransceiverInterface>& tranceiver :
>+       pc_->GetTransceivers()) {
>+    rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> track = tranceiver->receiver()->track();
>+    if (track && track->kind() == webrtc::MediaStreamTrackInterface::kVideoKind) {
>+      static_cast<webrtc::VideoTrackInterface*>(track.get())
>+          ->AddOrUpdateSink(remote_sink_.get(), rtc::VideoSinkWants());
>+      RTC_LOG(LS_INFO) << "Remote video sink set up: " << track;
>+      break;
>+    }
>+  }
>+}
>+
>+void ObjCCallClient::Connect() {
>+  rtc::CritScope lock(&pc_mutex_);
>+  pc_->CreateOffer(new rtc::RefCountedObject<CreateOfferObserver>(pc_),
>+                   webrtc::PeerConnectionInterface::RTCOfferAnswerOptions());
>+}
>+
>+ObjCCallClient::PCObserver::PCObserver(ObjCCallClient* client) : client_(client) {}
>+
>+void ObjCCallClient::PCObserver::OnSignalingChange(
>+    webrtc::PeerConnectionInterface::SignalingState new_state) {
>+  RTC_LOG(LS_INFO) << "OnSignalingChange: " << new_state;
>+}
>+
>+void ObjCCallClient::PCObserver::OnDataChannel(
>+    rtc::scoped_refptr<webrtc::DataChannelInterface> data_channel) {
>+  RTC_LOG(LS_INFO) << "OnDataChannel";
>+}
>+
>+void ObjCCallClient::PCObserver::OnRenegotiationNeeded() {
>+  RTC_LOG(LS_INFO) << "OnRenegotiationNeeded";
>+}
>+
>+void ObjCCallClient::PCObserver::OnIceConnectionChange(
>+    webrtc::PeerConnectionInterface::IceConnectionState new_state) {
>+  RTC_LOG(LS_INFO) << "OnIceConnectionChange: " << new_state;
>+}
>+
>+void ObjCCallClient::PCObserver::OnIceGatheringChange(
>+    webrtc::PeerConnectionInterface::IceGatheringState new_state) {
>+  RTC_LOG(LS_INFO) << "OnIceGatheringChange: " << new_state;
>+}
>+
>+void ObjCCallClient::PCObserver::OnIceCandidate(const webrtc::IceCandidateInterface* candidate) {
>+  RTC_LOG(LS_INFO) << "OnIceCandidate: " << candidate->server_url();
>+  rtc::CritScope lock(&client_->pc_mutex_);
>+  RTC_DCHECK(client_->pc_ != nullptr);
>+  client_->pc_->AddIceCandidate(candidate);
>+}
>+
>+CreateOfferObserver::CreateOfferObserver(rtc::scoped_refptr<webrtc::PeerConnectionInterface> pc)
>+    : pc_(pc) {}
>+
>+void CreateOfferObserver::OnSuccess(webrtc::SessionDescriptionInterface* desc) {
>+  std::string sdp;
>+  desc->ToString(&sdp);
>+  RTC_LOG(LS_INFO) << "Created offer: " << sdp;
>+
>+  // Ownership of desc was transferred to us, now we transfer it forward.
>+  pc_->SetLocalDescription(new rtc::RefCountedObject<SetLocalSessionDescriptionObserver>(), desc);
>+
>+  // Generate a fake answer.
>+  std::unique_ptr<webrtc::SessionDescriptionInterface> answer(
>+      webrtc::CreateSessionDescription(webrtc::SdpType::kAnswer, sdp));
>+  pc_->SetRemoteDescription(std::move(answer),
>+                            new rtc::RefCountedObject<SetRemoteSessionDescriptionObserver>());
>+}
>+
>+void CreateOfferObserver::OnFailure(webrtc::RTCError error) {
>+  RTC_LOG(LS_INFO) << "Failed to create offer: " << error.message();
>+}
>+
>+void SetRemoteSessionDescriptionObserver::OnSetRemoteDescriptionComplete(webrtc::RTCError error) {
>+  RTC_LOG(LS_INFO) << "Set remote description: " << error.message();
>+}
>+
>+void SetLocalSessionDescriptionObserver::OnSuccess() {
>+  RTC_LOG(LS_INFO) << "Set local description success!";
>+}
>+
>+void SetLocalSessionDescriptionObserver::OnFailure(webrtc::RTCError error) {
>+  RTC_LOG(LS_INFO) << "Set local description failure: " << error.message();
>+}
>+
>+}  // namespace webrtc_examples
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/conductor.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/conductor.cc
>index 7394137c5d8..89c89846928 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/conductor.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/conductor.cc
>@@ -17,8 +17,12 @@
> #include "api/audio_codecs/builtin_audio_decoder_factory.h"
> #include "api/audio_codecs/builtin_audio_encoder_factory.h"
> #include "api/test/fakeconstraints.h"
>+#include "api/video_codecs/builtin_video_decoder_factory.h"
>+#include "api/video_codecs/builtin_video_encoder_factory.h"
> #include "examples/peerconnection/client/defaults.h"
> #include "media/engine/webrtcvideocapturerfactory.h"
>+#include "modules/audio_device/include/audio_device.h"
>+#include "modules/audio_processing/include/audio_processing.h"
> #include "modules/video_capture/video_capture_factory.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/json.h"
>@@ -33,41 +37,31 @@ const char kCandidateSdpName[] = "candidate";
> const char kSessionDescriptionTypeName[] = "type";
> const char kSessionDescriptionSdpName[] = "sdp";
> 
>-#define DTLS_ON  true
>-#define DTLS_OFF false
>-
> class DummySetSessionDescriptionObserver
>     : public webrtc::SetSessionDescriptionObserver {
>  public:
>   static DummySetSessionDescriptionObserver* Create() {
>-    return
>-        new rtc::RefCountedObject<DummySetSessionDescriptionObserver>();
>+    return new rtc::RefCountedObject<DummySetSessionDescriptionObserver>();
>   }
>   virtual void OnSuccess() { RTC_LOG(INFO) << __FUNCTION__; }
>-  virtual void OnFailure(const std::string& error) {
>-    RTC_LOG(INFO) << __FUNCTION__ << " " << error;
>+  virtual void OnFailure(webrtc::RTCError error) {
>+    RTC_LOG(INFO) << __FUNCTION__ << " " << ToString(error.type()) << ": "
>+                  << error.message();
>   }
>-
>- protected:
>-  DummySetSessionDescriptionObserver() {}
>-  ~DummySetSessionDescriptionObserver() {}
> };
> 
> Conductor::Conductor(PeerConnectionClient* client, MainWindow* main_wnd)
>-  : peer_id_(-1),
>-    loopback_(false),
>-    client_(client),
>-    main_wnd_(main_wnd) {
>+    : peer_id_(-1), loopback_(false), client_(client), main_wnd_(main_wnd) {
>   client_->RegisterObserver(this);
>   main_wnd->RegisterObserver(this);
> }
> 
> Conductor::~Conductor() {
>-  RTC_DCHECK(peer_connection_.get() == NULL);
>+  RTC_DCHECK(!peer_connection_);
> }
> 
> bool Conductor::connection_active() const {
>-  return peer_connection_.get() != NULL;
>+  return peer_connection_ != nullptr;
> }
> 
> void Conductor::Close() {
>@@ -76,77 +70,77 @@ void Conductor::Close() {
> }
> 
> bool Conductor::InitializePeerConnection() {
>-  RTC_DCHECK(peer_connection_factory_.get() == NULL);
>-  RTC_DCHECK(peer_connection_.get() == NULL);
>+  RTC_DCHECK(!peer_connection_factory_);
>+  RTC_DCHECK(!peer_connection_);
> 
>   peer_connection_factory_ = webrtc::CreatePeerConnectionFactory(
>+      nullptr /* network_thread */, nullptr /* worker_thread */,
>+      nullptr /* signaling_thread */, nullptr /* default_adm */,
>       webrtc::CreateBuiltinAudioEncoderFactory(),
>-      webrtc::CreateBuiltinAudioDecoderFactory());
>+      webrtc::CreateBuiltinAudioDecoderFactory(),
>+      webrtc::CreateBuiltinVideoEncoderFactory(),
>+      webrtc::CreateBuiltinVideoDecoderFactory(), nullptr /* audio_mixer */,
>+      nullptr /* audio_processing */);
> 
>-  if (!peer_connection_factory_.get()) {
>-    main_wnd_->MessageBox("Error",
>-        "Failed to initialize PeerConnectionFactory", true);
>+  if (!peer_connection_factory_) {
>+    main_wnd_->MessageBox("Error", "Failed to initialize PeerConnectionFactory",
>+                          true);
>     DeletePeerConnection();
>     return false;
>   }
> 
>-  if (!CreatePeerConnection(DTLS_ON)) {
>-    main_wnd_->MessageBox("Error",
>-        "CreatePeerConnection failed", true);
>+  if (!CreatePeerConnection(/*dtls=*/true)) {
>+    main_wnd_->MessageBox("Error", "CreatePeerConnection failed", true);
>     DeletePeerConnection();
>   }
>-  AddStreams();
>-  return peer_connection_.get() != NULL;
>+
>+  AddTracks();
>+
>+  return peer_connection_ != nullptr;
> }
> 
> bool Conductor::ReinitializePeerConnectionForLoopback() {
>   loopback_ = true;
>-  rtc::scoped_refptr<webrtc::StreamCollectionInterface> streams(
>-      peer_connection_->local_streams());
>-  peer_connection_ = NULL;
>-  if (CreatePeerConnection(DTLS_OFF)) {
>-    for (size_t i = 0; i < streams->count(); ++i)
>-      peer_connection_->AddStream(streams->at(i));
>-    peer_connection_->CreateOffer(this, NULL);
>+  std::vector<rtc::scoped_refptr<webrtc::RtpSenderInterface>> senders =
>+      peer_connection_->GetSenders();
>+  peer_connection_ = nullptr;
>+  if (CreatePeerConnection(/*dtls=*/false)) {
>+    for (const auto& sender : senders) {
>+      peer_connection_->AddTrack(sender->track(), sender->stream_ids());
>+    }
>+    peer_connection_->CreateOffer(
>+        this, webrtc::PeerConnectionInterface::RTCOfferAnswerOptions());
>   }
>-  return peer_connection_.get() != NULL;
>+  return peer_connection_ != nullptr;
> }
> 
> bool Conductor::CreatePeerConnection(bool dtls) {
>-  RTC_DCHECK(peer_connection_factory_.get() != NULL);
>-  RTC_DCHECK(peer_connection_.get() == NULL);
>+  RTC_DCHECK(peer_connection_factory_);
>+  RTC_DCHECK(!peer_connection_);
> 
>   webrtc::PeerConnectionInterface::RTCConfiguration config;
>+  config.sdp_semantics = webrtc::SdpSemantics::kUnifiedPlan;
>+  config.enable_dtls_srtp = dtls;
>   webrtc::PeerConnectionInterface::IceServer server;
>   server.uri = GetPeerConnectionString();
>   config.servers.push_back(server);
> 
>-  webrtc::FakeConstraints constraints;
>-  if (dtls) {
>-    constraints.AddOptional(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
>-                            "true");
>-  } else {
>-    constraints.AddOptional(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
>-                            "false");
>-  }
>-
>   peer_connection_ = peer_connection_factory_->CreatePeerConnection(
>-      config, &constraints, NULL, NULL, this);
>-  return peer_connection_.get() != NULL;
>+      config, nullptr, nullptr, this);
>+  return peer_connection_ != nullptr;
> }
> 
> void Conductor::DeletePeerConnection() {
>-  peer_connection_ = NULL;
>-  active_streams_.clear();
>   main_wnd_->StopLocalRenderer();
>   main_wnd_->StopRemoteRenderer();
>-  peer_connection_factory_ = NULL;
>+  peer_connection_ = nullptr;
>+  peer_connection_factory_ = nullptr;
>   peer_id_ = -1;
>   loopback_ = false;
> }
> 
> void Conductor::EnsureStreamingUI() {
>-  RTC_DCHECK(peer_connection_.get() != NULL);
>+  RTC_DCHECK(peer_connection_);
>   if (main_wnd_->IsWindow()) {
>     if (main_wnd_->current_ui() != MainWindow::STREAMING)
>       main_wnd_->SwitchToStreamingUI();
>@@ -157,17 +151,19 @@ void Conductor::EnsureStreamingUI() {
> // PeerConnectionObserver implementation.
> //
> 
>-// Called when a remote stream is added
>-void Conductor::OnAddStream(
>-    rtc::scoped_refptr<webrtc::MediaStreamInterface> stream) {
>-  RTC_LOG(INFO) << __FUNCTION__ << " " << stream->label();
>-  main_wnd_->QueueUIThreadCallback(NEW_STREAM_ADDED, stream.release());
>+void Conductor::OnAddTrack(
>+    rtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver,
>+    const std::vector<rtc::scoped_refptr<webrtc::MediaStreamInterface>>&
>+        streams) {
>+  RTC_LOG(INFO) << __FUNCTION__ << " " << receiver->id();
>+  main_wnd_->QueueUIThreadCallback(NEW_TRACK_ADDED,
>+                                   receiver->track().release());
> }
> 
>-void Conductor::OnRemoveStream(
>-    rtc::scoped_refptr<webrtc::MediaStreamInterface> stream) {
>-  RTC_LOG(INFO) << __FUNCTION__ << " " << stream->label();
>-  main_wnd_->QueueUIThreadCallback(STREAM_REMOVED, stream.release());
>+void Conductor::OnRemoveTrack(
>+    rtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver) {
>+  RTC_LOG(INFO) << __FUNCTION__ << " " << receiver->id();
>+  main_wnd_->QueueUIThreadCallback(TRACK_REMOVED, receiver->track().release());
> }
> 
> void Conductor::OnIceCandidate(const webrtc::IceCandidateInterface* candidate) {
>@@ -258,12 +254,13 @@ void Conductor::OnMessageFromPeer(int peer_id, const std::string& message) {
>     RTC_LOG(WARNING) << "Received unknown message. " << message;
>     return;
>   }
>-  std::string type;
>+  std::string type_str;
>   std::string json_object;
> 
>-  rtc::GetStringFromJsonObject(jmessage, kSessionDescriptionTypeName, &type);
>-  if (!type.empty()) {
>-    if (type == "offer-loopback") {
>+  rtc::GetStringFromJsonObject(jmessage, kSessionDescriptionTypeName,
>+                               &type_str);
>+  if (!type_str.empty()) {
>+    if (type_str == "offer-loopback") {
>       // This is a loopback call.
>       // Recreate the peerconnection with DTLS disabled.
>       if (!ReinitializePeerConnectionForLoopback()) {
>@@ -273,7 +270,13 @@ void Conductor::OnMessageFromPeer(int peer_id, const std::string& message) {
>       }
>       return;
>     }
>-
>+    absl::optional<webrtc::SdpType> type_maybe =
>+        webrtc::SdpTypeFromString(type_str);
>+    if (!type_maybe) {
>+      RTC_LOG(LS_ERROR) << "Unknown SDP type: " << type_str;
>+      return;
>+    }
>+    webrtc::SdpType type = *type_maybe;
>     std::string sdp;
>     if (!rtc::GetStringFromJsonObject(jmessage, kSessionDescriptionSdpName,
>                                       &sdp)) {
>@@ -281,8 +284,8 @@ void Conductor::OnMessageFromPeer(int peer_id, const std::string& message) {
>       return;
>     }
>     webrtc::SdpParseError error;
>-    webrtc::SessionDescriptionInterface* session_description(
>-        webrtc::CreateSessionDescription(type, sdp, &error));
>+    std::unique_ptr<webrtc::SessionDescriptionInterface> session_description =
>+        webrtc::CreateSessionDescription(type, sdp, &error);
>     if (!session_description) {
>       RTC_LOG(WARNING) << "Can't parse received session description message. "
>                        << "SdpParseError was: " << error.description;
>@@ -290,12 +293,12 @@ void Conductor::OnMessageFromPeer(int peer_id, const std::string& message) {
>     }
>     RTC_LOG(INFO) << " Received session description :" << message;
>     peer_connection_->SetRemoteDescription(
>-        DummySetSessionDescriptionObserver::Create(), session_description);
>-    if (session_description->type() ==
>-        webrtc::SessionDescriptionInterface::kOffer) {
>-      peer_connection_->CreateAnswer(this, NULL);
>+        DummySetSessionDescriptionObserver::Create(),
>+        session_description.release());
>+    if (type == webrtc::SdpType::kOffer) {
>+      peer_connection_->CreateAnswer(
>+          this, webrtc::PeerConnectionInterface::RTCOfferAnswerOptions());
>     }
>-    return;
>   } else {
>     std::string sdp_mid;
>     int sdp_mlineindex = 0;
>@@ -321,7 +324,6 @@ void Conductor::OnMessageFromPeer(int peer_id, const std::string& message) {
>       return;
>     }
>     RTC_LOG(INFO) << " Received candidate :" << message;
>-    return;
>   }
> }
> 
>@@ -331,8 +333,8 @@ void Conductor::OnMessageSent(int err) {
> }
> 
> void Conductor::OnServerConnectionFailure() {
>-    main_wnd_->MessageBox("Error", ("Failed to connect to " + server_).c_str(),
>-                          true);
>+  main_wnd_->MessageBox("Error", ("Failed to connect to " + server_).c_str(),
>+                        true);
> }
> 
> //
>@@ -356,14 +358,15 @@ void Conductor::ConnectToPeer(int peer_id) {
>   RTC_DCHECK(peer_id != -1);
> 
>   if (peer_connection_.get()) {
>-    main_wnd_->MessageBox("Error",
>-        "We only support connecting to one peer at a time", true);
>+    main_wnd_->MessageBox(
>+        "Error", "We only support connecting to one peer at a time", true);
>     return;
>   }
> 
>   if (InitializePeerConnection()) {
>     peer_id_ = peer_id;
>-    peer_connection_->CreateOffer(this, NULL);
>+    peer_connection_->CreateOffer(
>+        this, webrtc::PeerConnectionInterface::RTCOfferAnswerOptions());
>   } else {
>     main_wnd_->MessageBox("Error", "Failed to initialize PeerConnection", true);
>   }
>@@ -399,33 +402,39 @@ std::unique_ptr<cricket::VideoCapturer> Conductor::OpenVideoCaptureDevice() {
>   return capturer;
> }
> 
>-void Conductor::AddStreams() {
>-  if (active_streams_.find(kStreamLabel) != active_streams_.end())
>-    return;  // Already added.
>+void Conductor::AddTracks() {
>+  if (!peer_connection_->GetSenders().empty()) {
>+    return;  // Already added tracks.
>+  }
> 
>   rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
>       peer_connection_factory_->CreateAudioTrack(
>-          kAudioLabel, peer_connection_factory_->CreateAudioSource(NULL)));
>-
>-  rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track(
>-      peer_connection_factory_->CreateVideoTrack(
>-          kVideoLabel,
>-          peer_connection_factory_->CreateVideoSource(OpenVideoCaptureDevice(),
>-                                                      NULL)));
>-  main_wnd_->StartLocalRenderer(video_track);
>-
>-  rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
>-      peer_connection_factory_->CreateLocalMediaStream(kStreamLabel);
>-
>-  stream->AddTrack(audio_track);
>-  stream->AddTrack(video_track);
>-  if (!peer_connection_->AddStream(stream)) {
>-    RTC_LOG(LS_ERROR) << "Adding stream to PeerConnection failed";
>+          kAudioLabel, peer_connection_factory_->CreateAudioSource(
>+                           cricket::AudioOptions())));
>+  auto result_or_error = peer_connection_->AddTrack(audio_track, {kStreamId});
>+  if (!result_or_error.ok()) {
>+    RTC_LOG(LS_ERROR) << "Failed to add audio track to PeerConnection: "
>+                      << result_or_error.error().message();
>   }
>-  typedef std::pair<std::string,
>-                    rtc::scoped_refptr<webrtc::MediaStreamInterface> >
>-      MediaStreamPair;
>-  active_streams_.insert(MediaStreamPair(stream->label(), stream));
>+
>+  std::unique_ptr<cricket::VideoCapturer> video_device =
>+      OpenVideoCaptureDevice();
>+  if (video_device) {
>+    rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track_(
>+        peer_connection_factory_->CreateVideoTrack(
>+            kVideoLabel, peer_connection_factory_->CreateVideoSource(
>+                             std::move(video_device), nullptr)));
>+    main_wnd_->StartLocalRenderer(video_track_);
>+
>+    result_or_error = peer_connection_->AddTrack(video_track_, {kStreamId});
>+    if (!result_or_error.ok()) {
>+      RTC_LOG(LS_ERROR) << "Failed to add video track to PeerConnection: "
>+                        << result_or_error.error().message();
>+    }
>+  } else {
>+    RTC_LOG(LS_ERROR) << "OpenVideoCaptureDevice failed";
>+  }
>+
>   main_wnd_->SwitchToStreamingUI();
> }
> 
>@@ -446,8 +455,6 @@ void Conductor::UIThreadCallback(int msg_id, void* data) {
>       RTC_LOG(INFO) << "PEER_CONNECTION_CLOSED";
>       DeletePeerConnection();
> 
>-      RTC_DCHECK(active_streams_.empty());
>-
>       if (main_wnd_->IsWindow()) {
>         if (client_->is_connected()) {
>           main_wnd_->SwitchToPeerList(client_->peers());
>@@ -486,26 +493,20 @@ void Conductor::UIThreadCallback(int msg_id, void* data) {
>       break;
>     }
> 
>-    case NEW_STREAM_ADDED: {
>-      webrtc::MediaStreamInterface* stream =
>-          reinterpret_cast<webrtc::MediaStreamInterface*>(
>-          data);
>-      webrtc::VideoTrackVector tracks = stream->GetVideoTracks();
>-      // Only render the first track.
>-      if (!tracks.empty()) {
>-        webrtc::VideoTrackInterface* track = tracks[0];
>-        main_wnd_->StartRemoteRenderer(track);
>+    case NEW_TRACK_ADDED: {
>+      auto* track = reinterpret_cast<webrtc::MediaStreamTrackInterface*>(data);
>+      if (track->kind() == webrtc::MediaStreamTrackInterface::kVideoKind) {
>+        auto* video_track = static_cast<webrtc::VideoTrackInterface*>(track);
>+        main_wnd_->StartRemoteRenderer(video_track);
>       }
>-      stream->Release();
>+      track->Release();
>       break;
>     }
> 
>-    case STREAM_REMOVED: {
>-      // Remote peer stopped sending a stream.
>-      webrtc::MediaStreamInterface* stream =
>-          reinterpret_cast<webrtc::MediaStreamInterface*>(
>-          data);
>-      stream->Release();
>+    case TRACK_REMOVED: {
>+      // Remote peer stopped sending a track.
>+      auto* track = reinterpret_cast<webrtc::MediaStreamTrackInterface*>(data);
>+      track->Release();
>       break;
>     }
> 
>@@ -525,22 +526,24 @@ void Conductor::OnSuccess(webrtc::SessionDescriptionInterface* desc) {
>   // For loopback test. To save some connecting delay.
>   if (loopback_) {
>     // Replace message type from "offer" to "answer"
>-    webrtc::SessionDescriptionInterface* session_description(
>-        webrtc::CreateSessionDescription("answer", sdp, nullptr));
>+    std::unique_ptr<webrtc::SessionDescriptionInterface> session_description =
>+        webrtc::CreateSessionDescription(webrtc::SdpType::kAnswer, sdp);
>     peer_connection_->SetRemoteDescription(
>-        DummySetSessionDescriptionObserver::Create(), session_description);
>+        DummySetSessionDescriptionObserver::Create(),
>+        session_description.release());
>     return;
>   }
> 
>   Json::StyledWriter writer;
>   Json::Value jmessage;
>-  jmessage[kSessionDescriptionTypeName] = desc->type();
>+  jmessage[kSessionDescriptionTypeName] =
>+      webrtc::SdpTypeToString(desc->GetType());
>   jmessage[kSessionDescriptionSdpName] = sdp;
>   SendMessage(writer.write(jmessage));
> }
> 
>-void Conductor::OnFailure(const std::string& error) {
>-  RTC_LOG(LERROR) << error;
>+void Conductor::OnFailure(webrtc::RTCError error) {
>+  RTC_LOG(LERROR) << ToString(error.type()) << ": " << error.message();
> }
> 
> void Conductor::SendMessage(const std::string& json_object) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/conductor.h b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/conductor.h
>index ebd2b6baed9..a038743915b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/conductor.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/conductor.h
>@@ -14,8 +14,8 @@
> #include <deque>
> #include <map>
> #include <memory>
>-#include <set>
> #include <string>
>+#include <vector>
> 
> #include "api/mediastreaminterface.h"
> #include "api/peerconnectioninterface.h"
>@@ -30,25 +30,24 @@ namespace cricket {
> class VideoRenderer;
> }  // namespace cricket
> 
>-class Conductor
>-  : public webrtc::PeerConnectionObserver,
>-    public webrtc::CreateSessionDescriptionObserver,
>-    public PeerConnectionClientObserver,
>-    public MainWndCallback {
>+class Conductor : public webrtc::PeerConnectionObserver,
>+                  public webrtc::CreateSessionDescriptionObserver,
>+                  public PeerConnectionClientObserver,
>+                  public MainWndCallback {
>  public:
>   enum CallbackID {
>     MEDIA_CHANNELS_INITIALIZED = 1,
>     PEER_CONNECTION_CLOSED,
>     SEND_MESSAGE_TO_PEER,
>-    NEW_STREAM_ADDED,
>-    STREAM_REMOVED,
>+    NEW_TRACK_ADDED,
>+    TRACK_REMOVED,
>   };
> 
>   Conductor(PeerConnectionClient* client, MainWindow* main_wnd);
> 
>   bool connection_active() const;
> 
>-  virtual void Close();
>+  void Close() override;
> 
>  protected:
>   ~Conductor();
>@@ -57,7 +56,7 @@ class Conductor
>   bool CreatePeerConnection(bool dtls);
>   void DeletePeerConnection();
>   void EnsureStreamingUI();
>-  void AddStreams();
>+  void AddTracks();
>   std::unique_ptr<cricket::VideoCapturer> OpenVideoCaptureDevice();
> 
>   //
>@@ -66,10 +65,12 @@ class Conductor
> 
>   void OnSignalingChange(
>       webrtc::PeerConnectionInterface::SignalingState new_state) override{};
>-  void OnAddStream(
>-      rtc::scoped_refptr<webrtc::MediaStreamInterface> stream) override;
>-  void OnRemoveStream(
>-      rtc::scoped_refptr<webrtc::MediaStreamInterface> stream) override;
>+  void OnAddTrack(
>+      rtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver,
>+      const std::vector<rtc::scoped_refptr<webrtc::MediaStreamInterface>>&
>+          streams) override;
>+  void OnRemoveTrack(
>+      rtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver) override;
>   void OnDataChannel(
>       rtc::scoped_refptr<webrtc::DataChannelInterface> channel) override {}
>   void OnRenegotiationNeeded() override {}
>@@ -114,7 +115,7 @@ class Conductor
> 
>   // CreateSessionDescriptionObserver implementation.
>   void OnSuccess(webrtc::SessionDescriptionInterface* desc) override;
>-  void OnFailure(const std::string& error) override;
>+  void OnFailure(webrtc::RTCError error) override;
> 
>  protected:
>   // Send a message to the remote peer.
>@@ -128,8 +129,6 @@ class Conductor
>   PeerConnectionClient* client_;
>   MainWindow* main_wnd_;
>   std::deque<std::string*> pending_messages_;
>-  std::map<std::string, rtc::scoped_refptr<webrtc::MediaStreamInterface> >
>-      active_streams_;
>   std::string server_;
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/defaults.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/defaults.cc
>index 51732ad4848..aeb5cb488d2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/defaults.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/defaults.cc
>@@ -23,7 +23,7 @@
> 
> const char kAudioLabel[] = "audio_label";
> const char kVideoLabel[] = "video_label";
>-const char kStreamLabel[] = "stream_label";
>+const char kStreamId[] = "stream_id";
> const uint16_t kDefaultServerPort = 8888;
> 
> std::string GetEnvVarOrDefault(const char* env_var_name,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/defaults.h b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/defaults.h
>index c46011e3eab..30936fd9d4a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/defaults.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/defaults.h
>@@ -17,7 +17,7 @@
> 
> extern const char kAudioLabel[];
> extern const char kVideoLabel[];
>-extern const char kStreamLabel[];
>+extern const char kStreamId[];
> extern const uint16_t kDefaultServerPort;
> 
> std::string GetEnvVarOrDefault(const char* env_var_name,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/flagdefs.h b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/flagdefs.h
>index cc66c6f34a6..564e0e95f90 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/flagdefs.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/flagdefs.h
>@@ -20,13 +20,19 @@ extern const uint16_t kDefaultServerPort;  // From defaults.[h|cc]
> // for each platform.
> 
> DEFINE_bool(help, false, "Prints this message");
>-DEFINE_bool(autoconnect, false, "Connect to the server without user "
>-                                "intervention.");
>+DEFINE_bool(autoconnect,
>+            false,
>+            "Connect to the server without user "
>+            "intervention.");
> DEFINE_string(server, "localhost", "The server to connect to.");
>-DEFINE_int(port, kDefaultServerPort,
>+DEFINE_int(port,
>+           kDefaultServerPort,
>            "The port on which the server is listening.");
>-DEFINE_bool(autocall, false, "Call the first available other client on "
>-  "the server without user intervention.  Note: this flag should only be set "
>-  "to true on one of the two clients.");
>+DEFINE_bool(
>+    autocall,
>+    false,
>+    "Call the first available other client on "
>+    "the server without user intervention.  Note: this flag should only be set "
>+    "to true on one of the two clients.");
> 
> #endif  // EXAMPLES_PEERCONNECTION_CLIENT_FLAGDEFS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/linux/main.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/linux/main.cc
>index c074d7c2387..50179c4ba55 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/linux/main.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/linux/main.cc
>@@ -32,21 +32,21 @@ class CustomSocketServer : public rtc::PhysicalSocketServer {
>   void set_conductor(Conductor* conductor) { conductor_ = conductor; }
> 
>   // Override so that we can also pump the GTK message loop.
>-  virtual bool Wait(int cms, bool process_io) {
>+  bool Wait(int cms, bool process_io) override {
>     // Pump GTK events.
>     // TODO(henrike): We really should move either the socket server or UI to a
>     // different thread.  Alternatively we could look at merging the two loops
>     // by implementing a dispatcher for the socket server and/or use
>     // g_main_context_set_poll_func.
>-      while (gtk_events_pending())
>-        gtk_main_iteration();
>+    while (gtk_events_pending())
>+      gtk_main_iteration();
> 
>     if (!wnd_->IsWindow() && !conductor_->connection_active() &&
>         client_ != NULL && !client_->is_connected()) {
>       message_queue_->Quit();
>     }
>-    return rtc::PhysicalSocketServer::Wait(0/*cms == -1 ? 1 : cms*/,
>-                                                 process_io);
>+    return rtc::PhysicalSocketServer::Wait(0 /*cms == -1 ? 1 : cms*/,
>+                                           process_io);
>   }
> 
>  protected:
>@@ -58,15 +58,15 @@ class CustomSocketServer : public rtc::PhysicalSocketServer {
> 
> int main(int argc, char* argv[]) {
>   gtk_init(&argc, &argv);
>-  // g_type_init API is deprecated (and does nothing) since glib 2.35.0, see:
>-  // https://mail.gnome.org/archives/commits-list/2012-November/msg07809.html
>+// g_type_init API is deprecated (and does nothing) since glib 2.35.0, see:
>+// https://mail.gnome.org/archives/commits-list/2012-November/msg07809.html
> #if !GLIB_CHECK_VERSION(2, 35, 0)
>-    g_type_init();
>+  g_type_init();
> #endif
>-  // g_thread_init API is deprecated since glib 2.31.0, see release note:
>-  // http://mail.gnome.org/archives/gnome-announce-list/2011-October/msg00041.html
>+// g_thread_init API is deprecated since glib 2.31.0, see release note:
>+// http://mail.gnome.org/archives/gnome-announce-list/2011-October/msg00041.html
> #if !GLIB_CHECK_VERSION(2, 31, 0)
>-    g_thread_init(NULL);
>+  g_thread_init(NULL);
> #endif
> 
>   rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/linux/main_wnd.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/linux/main_wnd.cc
>index f7863d9461c..52b0d88523c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/linux/main_wnd.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/linux/main_wnd.cc
>@@ -14,12 +14,12 @@
> #include <gtk/gtk.h>
> #include <stddef.h>
> 
>-#include "libyuv/convert_from.h"
> #include "api/video/i420_buffer.h"
> #include "examples/peerconnection/client/defaults.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/stringutils.h"
>+#include "third_party/libyuv/include/libyuv/convert_from.h"
> 
> using rtc::sprintfn;
> 
>@@ -30,7 +30,8 @@ namespace {
> // GtkMainWnd instance.
> //
> 
>-gboolean OnDestroyedCallback(GtkWidget* widget, GdkEvent* event,
>+gboolean OnDestroyedCallback(GtkWidget* widget,
>+                             GdkEvent* event,
>                              gpointer data) {
>   reinterpret_cast<GtkMainWnd*>(data)->OnDestroyed(widget, event);
>   return FALSE;
>@@ -45,14 +46,17 @@ gboolean SimulateButtonClick(gpointer button) {
>   return false;
> }
> 
>-gboolean OnKeyPressCallback(GtkWidget* widget, GdkEventKey* key,
>+gboolean OnKeyPressCallback(GtkWidget* widget,
>+                            GdkEventKey* key,
>                             gpointer data) {
>   reinterpret_cast<GtkMainWnd*>(data)->OnKeyPress(widget, key);
>   return false;
> }
> 
>-void OnRowActivatedCallback(GtkTreeView* tree_view, GtkTreePath* path,
>-                            GtkTreeViewColumn* column, gpointer data) {
>+void OnRowActivatedCallback(GtkTreeView* tree_view,
>+                            GtkTreePath* path,
>+                            GtkTreeViewColumn* column,
>+                            gpointer data) {
>   reinterpret_cast<GtkMainWnd*>(data)->OnRowActivated(tree_view, path, column);
> }
> 
>@@ -90,8 +94,8 @@ void InitializeList(GtkWidget* list) {
> 
> // Adds an entry to a tree view.
> void AddToList(GtkWidget* list, const gchar* str, int value) {
>-  GtkListStore* store = GTK_LIST_STORE(
>-      gtk_tree_view_get_model(GTK_TREE_VIEW(list)));
>+  GtkListStore* store =
>+      GTK_LIST_STORE(gtk_tree_view_get_model(GTK_TREE_VIEW(list)));
> 
>   GtkTreeIter iter;
>   gtk_list_store_append(store, &iter);
>@@ -131,11 +135,20 @@ gboolean Draw(GtkWidget* widget, cairo_t* cr, gpointer data) {
> // GtkMainWnd implementation.
> //
> 
>-GtkMainWnd::GtkMainWnd(const char* server, int port, bool autoconnect,
>+GtkMainWnd::GtkMainWnd(const char* server,
>+                       int port,
>+                       bool autoconnect,
>                        bool autocall)
>-    : window_(NULL), draw_area_(NULL), vbox_(NULL), server_edit_(NULL),
>-      port_edit_(NULL), peer_list_(NULL), callback_(NULL),
>-      server_(server), autoconnect_(autoconnect), autocall_(autocall) {
>+    : window_(NULL),
>+      draw_area_(NULL),
>+      vbox_(NULL),
>+      server_edit_(NULL),
>+      port_edit_(NULL),
>+      peer_list_(NULL),
>+      callback_(NULL),
>+      server_(server),
>+      autoconnect_(autoconnect),
>+      autocall_(autocall) {
>   char buffer[10];
>   sprintfn(buffer, sizeof(buffer), "%i", port);
>   port_ = buffer;
>@@ -153,12 +166,13 @@ bool GtkMainWnd::IsWindow() {
>   return window_ != NULL && GTK_IS_WINDOW(window_);
> }
> 
>-void GtkMainWnd::MessageBox(const char* caption, const char* text,
>+void GtkMainWnd::MessageBox(const char* caption,
>+                            const char* text,
>                             bool is_error) {
>-  GtkWidget* dialog = gtk_message_dialog_new(GTK_WINDOW(window_),
>-      GTK_DIALOG_DESTROY_WITH_PARENT,
>-      is_error ? GTK_MESSAGE_ERROR : GTK_MESSAGE_INFO,
>-      GTK_BUTTONS_CLOSE, "%s", text);
>+  GtkWidget* dialog = gtk_message_dialog_new(
>+      GTK_WINDOW(window_), GTK_DIALOG_DESTROY_WITH_PARENT,
>+      is_error ? GTK_MESSAGE_ERROR : GTK_MESSAGE_INFO, GTK_BUTTONS_CLOSE, "%s",
>+      text);
>   gtk_window_set_title(GTK_WINDOW(dialog), caption);
>   gtk_dialog_run(GTK_DIALOG(dialog));
>   gtk_widget_destroy(dialog);
>@@ -174,7 +188,6 @@ MainWindow::UI GtkMainWnd::current_ui() {
>   return STREAMING;
> }
> 
>-
> void GtkMainWnd::StartLocalRenderer(webrtc::VideoTrackInterface* local_video) {
>   local_renderer_.reset(new VideoRenderer(this, local_video));
> }
>@@ -394,7 +407,8 @@ void GtkMainWnd::OnKeyPress(GtkWidget* widget, GdkEventKey* key) {
>   }
> }
> 
>-void GtkMainWnd::OnRowActivated(GtkTreeView* tree_view, GtkTreePath* path,
>+void GtkMainWnd::OnRowActivated(GtkTreeView* tree_view,
>+                                GtkTreePath* path,
>                                 GtkTreeViewColumn* column) {
>   RTC_DCHECK(peer_list_ != NULL);
>   GtkTreeIter iter;
>@@ -402,12 +416,12 @@ void GtkMainWnd::OnRowActivated(GtkTreeView* tree_view, GtkTreePath* path,
>   GtkTreeSelection* selection =
>       gtk_tree_view_get_selection(GTK_TREE_VIEW(tree_view));
>   if (gtk_tree_selection_get_selected(selection, &model, &iter)) {
>-     char* text;
>-     int id = -1;
>-     gtk_tree_model_get(model, &iter, 0, &text, 1, &id,  -1);
>-     if (id != -1)
>-       callback_->ConnectToPeer(id);
>-     g_free(text);
>+    char* text;
>+    int id = -1;
>+    gtk_tree_model_get(model, &iter, 0, &text, 1, &id, -1);
>+    if (id != -1)
>+      callback_->ConnectToPeer(id);
>+    g_free(text);
>   }
> }
> 
>@@ -519,8 +533,7 @@ void GtkMainWnd::VideoRenderer::SetSize(int width, int height) {
>   gdk_threads_leave();
> }
> 
>-void GtkMainWnd::VideoRenderer::OnFrame(
>-    const webrtc::VideoFrame& video_frame) {
>+void GtkMainWnd::VideoRenderer::OnFrame(const webrtc::VideoFrame& video_frame) {
>   gdk_threads_enter();
> 
>   rtc::scoped_refptr<webrtc::I420BufferInterface> buffer(
>@@ -536,11 +549,10 @@ void GtkMainWnd::VideoRenderer::OnFrame(
>   // This was supposed to be a call to libyuv::I420ToRGBA but it was resulting
>   // in a reddish video output (see https://bugs.webrtc.org/6857) because it
>   // was producing an unexpected byte order (ABGR, byte swapped).
>-  libyuv::I420ToABGR(buffer->DataY(), buffer->StrideY(),
>-                     buffer->DataU(), buffer->StrideU(),
>-                     buffer->DataV(), buffer->StrideV(),
>-                     image_.get(), width_ * 4,
>-                     buffer->width(), buffer->height());
>+  libyuv::I420ToABGR(buffer->DataY(), buffer->StrideY(), buffer->DataU(),
>+                     buffer->StrideU(), buffer->DataV(), buffer->StrideV(),
>+                     image_.get(), width_ * 4, buffer->width(),
>+                     buffer->height());
> 
>   gdk_threads_leave();
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/linux/main_wnd.h b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/linux/main_wnd.h
>index fdeee4ba7b3..a2edb747577 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/linux/main_wnd.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/linux/main_wnd.h
>@@ -39,8 +39,7 @@ class GtkMainWnd : public MainWindow {
>   virtual void SwitchToConnectUI();
>   virtual void SwitchToPeerList(const Peers& peers);
>   virtual void SwitchToStreamingUI();
>-  virtual void MessageBox(const char* caption, const char* text,
>-                          bool is_error);
>+  virtual void MessageBox(const char* caption, const char* text, bool is_error);
>   virtual MainWindow::UI current_ui();
>   virtual void StartLocalRenderer(webrtc::VideoTrackInterface* local_video);
>   virtual void StopLocalRenderer();
>@@ -67,7 +66,8 @@ class GtkMainWnd : public MainWindow {
> 
>   // Callback when the user double clicks a peer in order to initiate a
>   // connection.
>-  void OnRowActivated(GtkTreeView* tree_view, GtkTreePath* path,
>+  void OnRowActivated(GtkTreeView* tree_view,
>+                      GtkTreePath* path,
>                       GtkTreeViewColumn* column);
> 
>   void OnRedraw();
>@@ -86,13 +86,9 @@ class GtkMainWnd : public MainWindow {
> 
>     const uint8_t* image() const { return image_.get(); }
> 
>-    int width() const {
>-      return width_;
>-    }
>+    int width() const { return width_; }
> 
>-    int height() const {
>-      return height_;
>-    }
>+    int height() const { return height_; }
> 
>    protected:
>     void SetSize(int width, int height);
>@@ -104,9 +100,9 @@ class GtkMainWnd : public MainWindow {
>   };
> 
>  protected:
>-  GtkWidget* window_;  // Our main window.
>+  GtkWidget* window_;     // Our main window.
>   GtkWidget* draw_area_;  // The drawing surface for rendering video streams.
>-  GtkWidget* vbox_;  // Container for the Connect UI.
>+  GtkWidget* vbox_;       // Container for the Connect UI.
>   GtkWidget* server_edit_;
>   GtkWidget* port_edit_;
>   GtkWidget* peer_list_;  // The list of peers.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/main.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/main.cc
>index bcf86f185ee..ee70da620f2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/main.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/main.cc
>@@ -17,8 +17,10 @@
> #include "rtc_base/win32socketinit.h"
> #include "rtc_base/win32socketserver.h"
> 
>-int PASCAL wWinMain(HINSTANCE instance, HINSTANCE prev_instance,
>-                    wchar_t* cmd_line, int cmd_show) {
>+int PASCAL wWinMain(HINSTANCE instance,
>+                    HINSTANCE prev_instance,
>+                    wchar_t* cmd_line,
>+                    int cmd_show) {
>   rtc::EnsureWinsockInit();
>   rtc::Win32SocketServer w32_ss;
>   rtc::Win32Thread w32_thread(&w32_ss);
>@@ -26,7 +28,7 @@ int PASCAL wWinMain(HINSTANCE instance, HINSTANCE prev_instance,
> 
>   rtc::WindowsCommandLineArguments win_args;
>   int argc = win_args.argc();
>-  char **argv = win_args.argv();
>+  char** argv = win_args.argv();
> 
>   rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true);
>   if (FLAG_help) {
>@@ -50,7 +52,7 @@ int PASCAL wWinMain(HINSTANCE instance, HINSTANCE prev_instance,
>   rtc::InitializeSSL();
>   PeerConnectionClient client;
>   rtc::scoped_refptr<Conductor> conductor(
>-        new rtc::RefCountedObject<Conductor>(&client, &wnd));
>+      new rtc::RefCountedObject<Conductor>(&client, &wnd));
> 
>   // Main loop.
>   MSG msg;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/main_wnd.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/main_wnd.cc
>index a33391d5ed1..3ad248843a8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/main_wnd.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/main_wnd.cc
>@@ -12,12 +12,13 @@
> 
> #include <math.h>
> 
>-#include "libyuv/convert_argb.h"
> #include "api/video/i420_buffer.h"
> #include "examples/peerconnection/client/defaults.h"
> #include "rtc_base/arraysize.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>+#include "rtc_base/stringutils.h"
>+#include "third_party/libyuv/include/libyuv/convert_argb.h"
> 
> ATOM MainWnd::wnd_class_ = 0;
> const wchar_t MainWnd::kClassName[] = L"WebRTC_MainWnd";
>@@ -30,8 +31,10 @@ const char kConnecting[] = "Connecting... ";
> const char kNoVideoStreams[] = "(no video streams either way)";
> const char kNoIncomingStream[] = "(no incoming video)";
> 
>-void CalculateWindowSizeForText(HWND wnd, const wchar_t* text,
>-                                size_t* width, size_t* height) {
>+void CalculateWindowSizeForText(HWND wnd,
>+                                const wchar_t* text,
>+                                size_t* width,
>+                                size_t* height) {
>   HDC dc = ::GetDC(wnd);
>   RECT text_rc = {0};
>   ::DrawText(dc, text, -1, &text_rc, DT_CALCRECT | DT_SINGLELINE);
>@@ -41,11 +44,9 @@ void CalculateWindowSizeForText(HWND wnd, const wchar_t* text,
>   ::GetWindowRect(wnd, &window);
> 
>   *width = text_rc.right - text_rc.left;
>-  *width += (window.right - window.left) -
>-            (client.right - client.left);
>+  *width += (window.right - window.left) - (client.right - client.left);
>   *height = text_rc.bottom - text_rc.top;
>-  *height += (window.bottom - window.top) -
>-             (client.bottom - client.top);
>+  *height += (window.bottom - window.top) - (client.bottom - client.top);
> }
> 
> HFONT GetDefaultFont() {
>@@ -61,18 +62,30 @@ std::string GetWindowText(HWND wnd) {
> 
> void AddListBoxItem(HWND listbox, const std::string& str, LPARAM item_data) {
>   LRESULT index = ::SendMessageA(listbox, LB_ADDSTRING, 0,
>-      reinterpret_cast<LPARAM>(str.c_str()));
>+                                 reinterpret_cast<LPARAM>(str.c_str()));
>   ::SendMessageA(listbox, LB_SETITEMDATA, index, item_data);
> }
> 
> }  // namespace
> 
>-MainWnd::MainWnd(const char* server, int port, bool auto_connect,
>+MainWnd::MainWnd(const char* server,
>+                 int port,
>+                 bool auto_connect,
>                  bool auto_call)
>-  : ui_(CONNECT_TO_SERVER), wnd_(NULL), edit1_(NULL), edit2_(NULL),
>-    label1_(NULL), label2_(NULL), button_(NULL), listbox_(NULL),
>-    destroyed_(false), callback_(NULL), nested_msg_(NULL),
>-    server_(server), auto_connect_(auto_connect), auto_call_(auto_call) {
>+    : ui_(CONNECT_TO_SERVER),
>+      wnd_(NULL),
>+      edit1_(NULL),
>+      edit2_(NULL),
>+      label1_(NULL),
>+      label2_(NULL),
>+      button_(NULL),
>+      listbox_(NULL),
>+      destroyed_(false),
>+      nested_msg_(NULL),
>+      callback_(NULL),
>+      server_(server),
>+      auto_connect_(auto_connect),
>+      auto_call_(auto_call) {
>   char buffer[10] = {0};
>   sprintfn(buffer, sizeof(buffer), "%i", port);
>   port_ = buffer;
>@@ -88,10 +101,11 @@ bool MainWnd::Create() {
>     return false;
> 
>   ui_thread_id_ = ::GetCurrentThreadId();
>-  wnd_ = ::CreateWindowExW(WS_EX_OVERLAPPEDWINDOW, kClassName, L"WebRTC",
>-      WS_OVERLAPPEDWINDOW | WS_VISIBLE | WS_CLIPCHILDREN,
>-      CW_USEDEFAULT, CW_USEDEFAULT, CW_USEDEFAULT, CW_USEDEFAULT,
>-      NULL, NULL, GetModuleHandle(NULL), this);
>+  wnd_ =
>+      ::CreateWindowExW(WS_EX_OVERLAPPEDWINDOW, kClassName, L"WebRTC",
>+                        WS_OVERLAPPEDWINDOW | WS_VISIBLE | WS_CLIPCHILDREN,
>+                        CW_USEDEFAULT, CW_USEDEFAULT, CW_USEDEFAULT,
>+                        CW_USEDEFAULT, NULL, NULL, GetModuleHandle(NULL), this);
> 
>   ::SendMessage(wnd_, WM_SETFONT, reinterpret_cast<WPARAM>(GetDefaultFont()),
>                 TRUE);
>@@ -175,10 +189,10 @@ void MainWnd::SwitchToPeerList(const Peers& peers) {
>     LRESULT count = ::SendMessage(listbox_, LB_GETCOUNT, 0, 0);
>     if (count != LB_ERR) {
>       // Select the last item in the list
>-      LRESULT selection = ::SendMessage(listbox_, LB_SETCURSEL , count - 1, 0);
>+      LRESULT selection = ::SendMessage(listbox_, LB_SETCURSEL, count - 1, 0);
>       if (selection != LB_ERR)
>-        ::PostMessage(wnd_, WM_COMMAND, MAKEWPARAM(GetDlgCtrlID(listbox_),
>-                                                   LBN_DBLCLK),
>+        ::PostMessage(wnd_, WM_COMMAND,
>+                      MAKEWPARAM(GetDlgCtrlID(listbox_), LBN_DBLCLK),
>                       reinterpret_cast<LPARAM>(listbox_));
>     }
>   }
>@@ -198,7 +212,6 @@ void MainWnd::MessageBox(const char* caption, const char* text, bool is_error) {
>   ::MessageBoxA(handle(), text, caption, flags);
> }
> 
>-
> void MainWnd::StartLocalRenderer(webrtc::VideoTrackInterface* local_video) {
>   local_renderer_.reset(new VideoRenderer(handle(), 1, 1, local_video));
> }
>@@ -217,7 +230,8 @@ void MainWnd::StopRemoteRenderer() {
> 
> void MainWnd::QueueUIThreadCallback(int msg_id, void* data) {
>   ::PostThreadMessage(ui_thread_id_, UI_THREAD_CALLBACK,
>-      static_cast<WPARAM>(msg_id), reinterpret_cast<LPARAM>(data));
>+                      static_cast<WPARAM>(msg_id),
>+                      reinterpret_cast<LPARAM>(data));
> }
> 
> void MainWnd::OnPaint() {
>@@ -243,8 +257,8 @@ void MainWnd::OnPaint() {
>       ::SetStretchBltMode(dc_mem, HALFTONE);
> 
>       // Set the map mode so that the ratio will be maintained for us.
>-      HDC all_dc[] = { ps.hdc, dc_mem };
>-      for (int i = 0; i < arraysize(all_dc); ++i) {
>+      HDC all_dc[] = {ps.hdc, dc_mem};
>+      for (size_t i = 0; i < arraysize(all_dc); ++i) {
>         SetMapMode(all_dc[i], MM_ISOTROPIC);
>         SetWindowExtEx(all_dc[i], width, height, NULL);
>         SetViewportExtEx(all_dc[i], rc.right, rc.bottom, NULL);
>@@ -253,35 +267,34 @@ void MainWnd::OnPaint() {
>       HBITMAP bmp_mem = ::CreateCompatibleBitmap(ps.hdc, rc.right, rc.bottom);
>       HGDIOBJ bmp_old = ::SelectObject(dc_mem, bmp_mem);
> 
>-      POINT logical_area = { rc.right, rc.bottom };
>+      POINT logical_area = {rc.right, rc.bottom};
>       DPtoLP(ps.hdc, &logical_area, 1);
> 
>       HBRUSH brush = ::CreateSolidBrush(RGB(0, 0, 0));
>-      RECT logical_rect = {0, 0, logical_area.x, logical_area.y };
>+      RECT logical_rect = {0, 0, logical_area.x, logical_area.y};
>       ::FillRect(dc_mem, &logical_rect, brush);
>       ::DeleteObject(brush);
> 
>       int x = (logical_area.x / 2) - (width / 2);
>       int y = (logical_area.y / 2) - (height / 2);
> 
>-      StretchDIBits(dc_mem, x, y, width, height,
>-                    0, 0, width, height, image, &bmi, DIB_RGB_COLORS, SRCCOPY);
>+      StretchDIBits(dc_mem, x, y, width, height, 0, 0, width, height, image,
>+                    &bmi, DIB_RGB_COLORS, SRCCOPY);
> 
>       if ((rc.right - rc.left) > 200 && (rc.bottom - rc.top) > 200) {
>         const BITMAPINFO& bmi = local_renderer->bmi();
>         image = local_renderer->image();
>         int thumb_width = bmi.bmiHeader.biWidth / 4;
>         int thumb_height = abs(bmi.bmiHeader.biHeight) / 4;
>-        StretchDIBits(dc_mem,
>-            logical_area.x - thumb_width - 10,
>-            logical_area.y - thumb_height - 10,
>-            thumb_width, thumb_height,
>-            0, 0, bmi.bmiHeader.biWidth, -bmi.bmiHeader.biHeight,
>-            image, &bmi, DIB_RGB_COLORS, SRCCOPY);
>+        StretchDIBits(dc_mem, logical_area.x - thumb_width - 10,
>+                      logical_area.y - thumb_height - 10, thumb_width,
>+                      thumb_height, 0, 0, bmi.bmiHeader.biWidth,
>+                      -bmi.bmiHeader.biHeight, image, &bmi, DIB_RGB_COLORS,
>+                      SRCCOPY);
>       }
> 
>-      BitBlt(ps.hdc, 0, 0, logical_area.x, logical_area.y,
>-             dc_mem, 0, 0, SRCCOPY);
>+      BitBlt(ps.hdc, 0, 0, logical_area.x, logical_area.y, dc_mem, 0, 0,
>+             SRCCOPY);
> 
>       // Cleanup.
>       ::SelectObject(dc_mem, bmp_old);
>@@ -304,7 +317,7 @@ void MainWnd::OnPaint() {
>         text += kNoIncomingStream;
>       }
>       ::DrawTextA(ps.hdc, text.c_str(), -1, &rc,
>-          DT_SINGLELINE | DT_CENTER | DT_VCENTER);
>+                  DT_SINGLELINE | DT_CENTER | DT_VCENTER);
>       ::SelectObject(ps.hdc, old_font);
>     }
>   } else {
>@@ -392,8 +405,8 @@ bool MainWnd::OnMessage(UINT msg, WPARAM wp, LPARAM lp, LRESULT* result) {
> 
> // static
> LRESULT CALLBACK MainWnd::WndProc(HWND hwnd, UINT msg, WPARAM wp, LPARAM lp) {
>-  MainWnd* me = reinterpret_cast<MainWnd*>(
>-      ::GetWindowLongPtr(hwnd, GWLP_USERDATA));
>+  MainWnd* me =
>+      reinterpret_cast<MainWnd*>(::GetWindowLongPtr(hwnd, GWLP_USERDATA));
>   if (!me && WM_CREATE == msg) {
>     CREATESTRUCT* cs = reinterpret_cast<CREATESTRUCT*>(lp);
>     me = reinterpret_cast<MainWnd*>(cs->lpCreateParams);
>@@ -432,7 +445,7 @@ bool MainWnd::RegisterWindowClass() {
>   if (wnd_class_)
>     return true;
> 
>-  WNDCLASSEX wcex = { sizeof(WNDCLASSEX) };
>+  WNDCLASSEX wcex = {sizeof(WNDCLASSEX)};
>   wcex.style = CS_DBLCLKS;
>   wcex.hInstance = GetModuleHandle(NULL);
>   wcex.hbrBackground = reinterpret_cast<HBRUSH>(COLOR_WINDOW + 1);
>@@ -444,17 +457,18 @@ bool MainWnd::RegisterWindowClass() {
>   return wnd_class_ != 0;
> }
> 
>-void MainWnd::CreateChildWindow(HWND* wnd, MainWnd::ChildWindowID id,
>-                                const wchar_t* class_name, DWORD control_style,
>+void MainWnd::CreateChildWindow(HWND* wnd,
>+                                MainWnd::ChildWindowID id,
>+                                const wchar_t* class_name,
>+                                DWORD control_style,
>                                 DWORD ex_style) {
>   if (::IsWindow(*wnd))
>     return;
> 
>   // Child windows are invisible at first, and shown after being resized.
>   DWORD style = WS_CHILD | control_style;
>-  *wnd = ::CreateWindowEx(ex_style, class_name, L"", style,
>-                          100, 100, 100, 100, wnd_,
>-                          reinterpret_cast<HMENU>(id),
>+  *wnd = ::CreateWindowEx(ex_style, class_name, L"", style, 100, 100, 100, 100,
>+                          wnd_, reinterpret_cast<HMENU>(id),
>                           GetModuleHandle(NULL), NULL);
>   RTC_DCHECK(::IsWindow(*wnd) != FALSE);
>   ::SendMessage(*wnd, WM_SETFONT, reinterpret_cast<WPARAM>(GetDefaultFont()),
>@@ -485,11 +499,9 @@ void MainWnd::LayoutConnectUI(bool show) {
>     size_t width;
>     size_t height;
>   } windows[] = {
>-    { label1_, L"Server" },
>-    { edit1_, L"XXXyyyYYYgggXXXyyyYYYggg" },
>-    { label2_, L":" },
>-    { edit2_, L"XyXyX" },
>-    { button_, L"Connect" },
>+      {label1_, L"Server"},  {edit1_, L"XXXyyyYYYgggXXXyyyYYYggg"},
>+      {label2_, L":"},       {edit2_, L"XyXyX"},
>+      {button_, L"Connect"},
>   };
> 
>   if (show) {
>@@ -510,8 +522,7 @@ void MainWnd::LayoutConnectUI(bool show) {
>       size_t top = y - (windows[i].height / 2);
>       ::MoveWindow(windows[i].wnd, static_cast<int>(x), static_cast<int>(top),
>                    static_cast<int>(windows[i].width),
>-                   static_cast<int>(windows[i].height),
>-                   TRUE);
>+                   static_cast<int>(windows[i].height), TRUE);
>       x += kSeparator + windows[i].width;
>       if (windows[i].text[0] != 'X')
>         ::SetWindowText(windows[i].wnd, windows[i].text);
>@@ -565,7 +576,9 @@ void MainWnd::HandleTabbing() {
> //
> 
> MainWnd::VideoRenderer::VideoRenderer(
>-    HWND wnd, int width, int height,
>+    HWND wnd,
>+    int width,
>+    int height,
>     webrtc::VideoTrackInterface* track_to_render)
>     : wnd_(wnd), rendered_track_(track_to_render) {
>   ::InitializeCriticalSection(&buffer_lock_);
>@@ -576,8 +589,8 @@ MainWnd::VideoRenderer::VideoRenderer(
>   bmi_.bmiHeader.biCompression = BI_RGB;
>   bmi_.bmiHeader.biWidth = width;
>   bmi_.bmiHeader.biHeight = -height;
>-  bmi_.bmiHeader.biSizeImage = width * height *
>-                              (bmi_.bmiHeader.biBitCount >> 3);
>+  bmi_.bmiHeader.biSizeImage =
>+      width * height * (bmi_.bmiHeader.biBitCount >> 3);
>   rendered_track_->AddOrUpdateSink(this, rtc::VideoSinkWants());
> }
> 
>@@ -595,14 +608,12 @@ void MainWnd::VideoRenderer::SetSize(int width, int height) {
> 
>   bmi_.bmiHeader.biWidth = width;
>   bmi_.bmiHeader.biHeight = -height;
>-  bmi_.bmiHeader.biSizeImage = width * height *
>-                               (bmi_.bmiHeader.biBitCount >> 3);
>+  bmi_.bmiHeader.biSizeImage =
>+      width * height * (bmi_.bmiHeader.biBitCount >> 3);
>   image_.reset(new uint8_t[bmi_.bmiHeader.biSizeImage]);
> }
> 
>-void MainWnd::VideoRenderer::OnFrame(
>-    const webrtc::VideoFrame& video_frame) {
>-
>+void MainWnd::VideoRenderer::OnFrame(const webrtc::VideoFrame& video_frame) {
>   {
>     AutoLock<VideoRenderer> lock(this);
> 
>@@ -615,12 +626,10 @@ void MainWnd::VideoRenderer::OnFrame(
>     SetSize(buffer->width(), buffer->height());
> 
>     RTC_DCHECK(image_.get() != NULL);
>-    libyuv::I420ToARGB(buffer->DataY(), buffer->StrideY(),
>-                       buffer->DataU(), buffer->StrideU(),
>-                       buffer->DataV(), buffer->StrideV(),
>+    libyuv::I420ToARGB(buffer->DataY(), buffer->StrideY(), buffer->DataU(),
>+                       buffer->StrideU(), buffer->DataV(), buffer->StrideV(),
>                        image_.get(),
>-                       bmi_.bmiHeader.biWidth *
>-                           bmi_.bmiHeader.biBitCount / 8,
>+                       bmi_.bmiHeader.biWidth * bmi_.bmiHeader.biBitCount / 8,
>                        buffer->width(), buffer->height());
>   }
>   InvalidateRect(wnd_, NULL, TRUE);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/main_wnd.h b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/main_wnd.h
>index f3ed2f576cb..57002583ea4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/main_wnd.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/main_wnd.h
>@@ -20,7 +20,9 @@
> #include "examples/peerconnection/client/peer_connection_client.h"
> #include "media/base/mediachannel.h"
> #include "media/base/videocommon.h"
>+#if defined(WEBRTC_WIN)
> #include "rtc_base/win32.h"
>+#endif  // WEBRTC_WIN
> 
> class MainWndCallback {
>  public:
>@@ -30,6 +32,7 @@ class MainWndCallback {
>   virtual void DisconnectFromCurrentPeer() = 0;
>   virtual void UIThreadCallback(int msg_id, void* data) = 0;
>   virtual void Close() = 0;
>+
>  protected:
>   virtual ~MainWndCallback() {}
> };
>@@ -48,7 +51,8 @@ class MainWindow {
>   virtual void RegisterObserver(MainWndCallback* callback) = 0;
> 
>   virtual bool IsWindow() = 0;
>-  virtual void MessageBox(const char* caption, const char* text,
>+  virtual void MessageBox(const char* caption,
>+                          const char* text,
>                           bool is_error) = 0;
> 
>   virtual UI current_ui() = 0;
>@@ -88,8 +92,7 @@ class MainWnd : public MainWindow {
>   virtual void SwitchToConnectUI();
>   virtual void SwitchToPeerList(const Peers& peers);
>   virtual void SwitchToStreamingUI();
>-  virtual void MessageBox(const char* caption, const char* text,
>-                          bool is_error);
>+  virtual void MessageBox(const char* caption, const char* text, bool is_error);
>   virtual UI current_ui() { return ui_; }
> 
>   virtual void StartLocalRenderer(webrtc::VideoTrackInterface* local_video);
>@@ -103,17 +106,15 @@ class MainWnd : public MainWindow {
> 
>   class VideoRenderer : public rtc::VideoSinkInterface<webrtc::VideoFrame> {
>    public:
>-    VideoRenderer(HWND wnd, int width, int height,
>+    VideoRenderer(HWND wnd,
>+                  int width,
>+                  int height,
>                   webrtc::VideoTrackInterface* track_to_render);
>     virtual ~VideoRenderer();
> 
>-    void Lock() {
>-      ::EnterCriticalSection(&buffer_lock_);
>-    }
>+    void Lock() { ::EnterCriticalSection(&buffer_lock_); }
> 
>-    void Unlock() {
>-      ::LeaveCriticalSection(&buffer_lock_);
>-    }
>+    void Unlock() { ::LeaveCriticalSection(&buffer_lock_); }
> 
>     // VideoSinkInterface implementation
>     void OnFrame(const webrtc::VideoFrame& frame) override;
>@@ -143,6 +144,7 @@ class MainWnd : public MainWindow {
>    public:
>     explicit AutoLock(T* obj) : obj_(obj) { obj_->Lock(); }
>     ~AutoLock() { obj_->Unlock(); }
>+
>    protected:
>     T* obj_;
>   };
>@@ -166,8 +168,11 @@ class MainWnd : public MainWindow {
>   static LRESULT CALLBACK WndProc(HWND hwnd, UINT msg, WPARAM wp, LPARAM lp);
>   static bool RegisterWindowClass();
> 
>-  void CreateChildWindow(HWND* wnd, ChildWindowID id, const wchar_t* class_name,
>-                         DWORD control_style, DWORD ex_style);
>+  void CreateChildWindow(HWND* wnd,
>+                         ChildWindowID id,
>+                         const wchar_t* class_name,
>+                         DWORD control_style,
>+                         DWORD ex_style);
>   void CreateChildWindows();
> 
>   void LayoutConnectUI(bool show);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/peer_connection_client.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/peer_connection_client.cc
>index 40dfec55fe1..bb79d167911 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/peer_connection_client.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/peer_connection_client.cc
>@@ -46,30 +46,23 @@ rtc::AsyncSocket* CreateClientSocket(int family) {
> }  // namespace
> 
> PeerConnectionClient::PeerConnectionClient()
>-  : callback_(NULL),
>-    resolver_(NULL),
>-    state_(NOT_CONNECTED),
>-    my_id_(-1) {
>-}
>+    : callback_(NULL), resolver_(NULL), state_(NOT_CONNECTED), my_id_(-1) {}
> 
>-PeerConnectionClient::~PeerConnectionClient() {
>-}
>+PeerConnectionClient::~PeerConnectionClient() {}
> 
> void PeerConnectionClient::InitSocketSignals() {
>   RTC_DCHECK(control_socket_.get() != NULL);
>   RTC_DCHECK(hanging_get_.get() != NULL);
>   control_socket_->SignalCloseEvent.connect(this,
>-      &PeerConnectionClient::OnClose);
>-  hanging_get_->SignalCloseEvent.connect(this,
>-      &PeerConnectionClient::OnClose);
>+                                            &PeerConnectionClient::OnClose);
>+  hanging_get_->SignalCloseEvent.connect(this, &PeerConnectionClient::OnClose);
>   control_socket_->SignalConnectEvent.connect(this,
>-      &PeerConnectionClient::OnConnect);
>-  hanging_get_->SignalConnectEvent.connect(this,
>-      &PeerConnectionClient::OnHangingGetConnect);
>-  control_socket_->SignalReadEvent.connect(this,
>-      &PeerConnectionClient::OnRead);
>-  hanging_get_->SignalReadEvent.connect(this,
>-      &PeerConnectionClient::OnHangingGetRead);
>+                                              &PeerConnectionClient::OnConnect);
>+  hanging_get_->SignalConnectEvent.connect(
>+      this, &PeerConnectionClient::OnHangingGetConnect);
>+  control_socket_->SignalReadEvent.connect(this, &PeerConnectionClient::OnRead);
>+  hanging_get_->SignalReadEvent.connect(
>+      this, &PeerConnectionClient::OnHangingGetRead);
> }
> 
> int PeerConnectionClient::id() const {
>@@ -90,7 +83,8 @@ void PeerConnectionClient::RegisterObserver(
>   callback_ = callback;
> }
> 
>-void PeerConnectionClient::Connect(const std::string& server, int port,
>+void PeerConnectionClient::Connect(const std::string& server,
>+                                   int port,
>                                    const std::string& client_name) {
>   RTC_DCHECK(!server.empty());
>   RTC_DCHECK(!client_name.empty());
>@@ -142,8 +136,8 @@ void PeerConnectionClient::DoConnect() {
>   hanging_get_.reset(CreateClientSocket(server_address_.ipaddr().family()));
>   InitSocketSignals();
>   char buffer[1024];
>-  sprintfn(buffer, sizeof(buffer),
>-           "GET /sign_in?%s HTTP/1.0\r\n\r\n", client_name_.c_str());
>+  sprintfn(buffer, sizeof(buffer), "GET /sign_in?%s HTTP/1.0\r\n\r\n",
>+           client_name_.c_str());
>   onconnect_data_ = buffer;
> 
>   bool ret = ConnectControlSocket();
>@@ -165,11 +159,11 @@ bool PeerConnectionClient::SendToPeer(int peer_id, const std::string& message) {
> 
>   char headers[1024];
>   sprintfn(headers, sizeof(headers),
>-      "POST /message?peer_id=%i&to=%i HTTP/1.0\r\n"
>-      "Content-Length: %i\r\n"
>-      "Content-Type: text/plain\r\n"
>-      "\r\n",
>-      my_id_, peer_id, message.length());
>+           "POST /message?peer_id=%i&to=%i HTTP/1.0\r\n"
>+           "Content-Length: %i\r\n"
>+           "Content-Type: text/plain\r\n"
>+           "\r\n",
>+           my_id_, peer_id, message.length());
>   onconnect_data_ = headers;
>   onconnect_data_ += message;
>   return ConnectControlSocket();
>@@ -197,7 +191,7 @@ bool PeerConnectionClient::SignOut() {
>     if (my_id_ != -1) {
>       char buffer[1024];
>       sprintfn(buffer, sizeof(buffer),
>-          "GET /sign_out?peer_id=%i HTTP/1.0\r\n\r\n", my_id_);
>+               "GET /sign_out?peer_id=%i HTTP/1.0\r\n\r\n", my_id_);
>       onconnect_data_ = buffer;
>       return ConnectControlSocket();
>     } else {
>@@ -243,8 +237,8 @@ void PeerConnectionClient::OnConnect(rtc::AsyncSocket* socket) {
> 
> void PeerConnectionClient::OnHangingGetConnect(rtc::AsyncSocket* socket) {
>   char buffer[1024];
>-  sprintfn(buffer, sizeof(buffer),
>-           "GET /wait?peer_id=%i HTTP/1.0\r\n\r\n", my_id_);
>+  sprintfn(buffer, sizeof(buffer), "GET /wait?peer_id=%i HTTP/1.0\r\n\r\n",
>+           my_id_);
>   int len = static_cast<int>(strlen(buffer));
>   int sent = socket->Send(buffer, len);
>   RTC_DCHECK(sent == len);
>@@ -273,7 +267,8 @@ bool PeerConnectionClient::GetHeaderValue(const std::string& data,
>   return false;
> }
> 
>-bool PeerConnectionClient::GetHeaderValue(const std::string& data, size_t eoh,
>+bool PeerConnectionClient::GetHeaderValue(const std::string& data,
>+                                          size_t eoh,
>                                           const char* header_pattern,
>                                           std::string* value) {
>   RTC_DCHECK(value != NULL);
>@@ -331,8 +326,8 @@ void PeerConnectionClient::OnRead(rtc::AsyncSocket* socket) {
>   size_t content_length = 0;
>   if (ReadIntoBuffer(socket, &control_data_, &content_length)) {
>     size_t peer_id = 0, eoh = 0;
>-    bool ok = ParseServerResponse(control_data_, content_length, &peer_id,
>-                                  &eoh);
>+    bool ok =
>+        ParseServerResponse(control_data_, content_length, &peer_id, &eoh);
>     if (ok) {
>       if (my_id_ == -1) {
>         // First response.  Let's store our server assigned ID.
>@@ -351,7 +346,8 @@ void PeerConnectionClient::OnRead(rtc::AsyncSocket* socket) {
>             std::string name;
>             bool connected;
>             if (ParseEntry(control_data_.substr(pos, eol - pos), &name, &id,
>-                           &connected) && id != my_id_) {
>+                           &connected) &&
>+                id != my_id_) {
>               peers_[id] = name;
>               callback_->OnPeerConnected(id, name);
>             }
>@@ -383,8 +379,8 @@ void PeerConnectionClient::OnHangingGetRead(rtc::AsyncSocket* socket) {
>   size_t content_length = 0;
>   if (ReadIntoBuffer(socket, &notification_data_, &content_length)) {
>     size_t peer_id = 0, eoh = 0;
>-    bool ok = ParseServerResponse(notification_data_, content_length,
>-                                  &peer_id, &eoh);
>+    bool ok =
>+        ParseServerResponse(notification_data_, content_length, &peer_id, &eoh);
> 
>     if (ok) {
>       // Store the position where the body begins.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/peer_connection_client.h b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/peer_connection_client.h
>index f5f87ec381e..8dd28f79933 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/peer_connection_client.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/client/peer_connection_client.h
>@@ -18,7 +18,7 @@
> #include "rtc_base/nethelpers.h"
> #include "rtc_base/physicalsocketserver.h"
> #include "rtc_base/signalthread.h"
>-#include "rtc_base/sigslot.h"
>+#include "rtc_base/third_party/sigslot/sigslot.h"
> 
> typedef std::map<int, std::string> Peers;
> 
>@@ -56,7 +56,8 @@ class PeerConnectionClient : public sigslot::has_slots<>,
> 
>   void RegisterObserver(PeerConnectionClientObserver* callback);
> 
>-  void Connect(const std::string& server, int port,
>+  void Connect(const std::string& server,
>+               int port,
>                const std::string& client_name);
> 
>   bool SendToPeer(int peer_id, const std::string& message);
>@@ -78,14 +79,19 @@ class PeerConnectionClient : public sigslot::has_slots<>,
>   void OnMessageFromPeer(int peer_id, const std::string& message);
> 
>   // Quick and dirty support for parsing HTTP header values.
>-  bool GetHeaderValue(const std::string& data, size_t eoh,
>-                      const char* header_pattern, size_t* value);
>+  bool GetHeaderValue(const std::string& data,
>+                      size_t eoh,
>+                      const char* header_pattern,
>+                      size_t* value);
> 
>-  bool GetHeaderValue(const std::string& data, size_t eoh,
>-                      const char* header_pattern, std::string* value);
>+  bool GetHeaderValue(const std::string& data,
>+                      size_t eoh,
>+                      const char* header_pattern,
>+                      std::string* value);
> 
>   // Returns true if the whole response has been read.
>-  bool ReadIntoBuffer(rtc::AsyncSocket* socket, std::string* data,
>+  bool ReadIntoBuffer(rtc::AsyncSocket* socket,
>+                      std::string* data,
>                       size_t* content_length);
> 
>   void OnRead(rtc::AsyncSocket* socket);
>@@ -93,13 +99,17 @@ class PeerConnectionClient : public sigslot::has_slots<>,
>   void OnHangingGetRead(rtc::AsyncSocket* socket);
> 
>   // Parses a single line entry in the form "<name>,<id>,<connected>"
>-  bool ParseEntry(const std::string& entry, std::string* name, int* id,
>+  bool ParseEntry(const std::string& entry,
>+                  std::string* name,
>+                  int* id,
>                   bool* connected);
> 
>   int GetResponseStatus(const std::string& response);
> 
>-  bool ParseServerResponse(const std::string& response, size_t content_length,
>-                           size_t* peer_id, size_t* eoh);
>+  bool ParseServerResponse(const std::string& response,
>+                           size_t content_length,
>+                           size_t* peer_id,
>+                           size_t* eoh);
> 
>   void OnClose(rtc::AsyncSocket* socket, int err);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/server/data_socket.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/server/data_socket.cc
>index 7a924316aa2..4369117e1f6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/server/data_socket.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/server/data_socket.cc
>@@ -29,7 +29,7 @@ const char DataSocket::kCrossOriginAllowHeaders[] =
>     "Access-Control-Allow-Credentials: true\r\n"
>     "Access-Control-Allow-Methods: POST, GET, OPTIONS\r\n"
>     "Access-Control-Allow-Headers: Content-Type, "
>-        "Content-Length, Connection, Cache-Control\r\n"
>+    "Content-Length, Connection, Cache-Control\r\n"
>     "Access-Control-Expose-Headers: Content-Length, X-Peer-Id\r\n";
> 
> #if defined(WIN32)
>@@ -116,10 +116,11 @@ bool DataSocket::OnDataAvailable(bool* close_socket) {
> 
> bool DataSocket::Send(const std::string& data) const {
>   return send(socket_, data.data(), static_cast<int>(data.length()), 0) !=
>-      SOCKET_ERROR;
>+         SOCKET_ERROR;
> }
> 
>-bool DataSocket::Send(const std::string& status, bool connection_close,
>+bool DataSocket::Send(const std::string& status,
>+                      bool connection_close,
>                       const std::string& content_type,
>                       const std::string& extra_headers,
>                       const std::string& data) const {
>@@ -127,8 +128,9 @@ bool DataSocket::Send(const std::string& status, bool connection_close,
>   assert(!status.empty());
>   std::string buffer("HTTP/1.1 " + status + "\r\n");
> 
>-  buffer += "Server: PeerConnectionTestServer/0.1\r\n"
>-            "Cache-Control: no-cache\r\n";
>+  buffer +=
>+      "Server: PeerConnectionTestServer/0.1\r\n"
>+      "Cache-Control: no-cache\r\n";
> 
>   if (connection_close)
>     buffer += "Connection: close\r\n";
>@@ -136,8 +138,8 @@ bool DataSocket::Send(const std::string& status, bool connection_close,
>   if (!content_type.empty())
>     buffer += "Content-Type: " + content_type + "\r\n";
> 
>-  buffer += "Content-Length: " + int2str(static_cast<int>(data.size())) +
>-            "\r\n";
>+  buffer +=
>+      "Content-Length: " + int2str(static_cast<int>(data.size())) + "\r\n";
> 
>   if (!extra_headers.empty()) {
>     buffer += extra_headers;
>@@ -190,9 +192,7 @@ bool DataSocket::ParseMethodAndPath(const char* begin, size_t len) {
>     size_t method_name_len;
>     RequestMethod id;
>   } supported_methods[] = {
>-    { "GET", 3, GET },
>-    { "POST", 4, POST },
>-    { "OPTIONS", 7, OPTIONS },
>+      {"GET", 3, GET}, {"POST", 4, POST}, {"OPTIONS", 7, OPTIONS},
>   };
> 
>   const char* path = NULL;
>@@ -231,15 +231,15 @@ bool DataSocket::ParseContentLengthAndType(const char* headers, size_t length) {
>       static const char kContentLength[] = "Content-Length:";
>       static const char kContentType[] = "Content-Type:";
>       if ((headers + ARRAYSIZE(kContentLength)) < end &&
>-          strncmp(headers, kContentLength,
>-                  ARRAYSIZE(kContentLength) - 1) == 0) {
>+          strncmp(headers, kContentLength, ARRAYSIZE(kContentLength) - 1) ==
>+              0) {
>         headers += ARRAYSIZE(kContentLength) - 1;
>         while (headers[0] == ' ')
>           ++headers;
>         content_length_ = atoi(headers);
>       } else if ((headers + ARRAYSIZE(kContentType)) < end &&
>-                 strncmp(headers, kContentType,
>-                         ARRAYSIZE(kContentType) - 1) == 0) {
>+                 strncmp(headers, kContentType, ARRAYSIZE(kContentType) - 1) ==
>+                     0) {
>         headers += ARRAYSIZE(kContentType) - 1;
>         while (headers[0] == ' ')
>           ++headers;
>@@ -267,13 +267,13 @@ bool ListeningSocket::Listen(unsigned short port) {
>   assert(valid());
>   int enabled = 1;
>   setsockopt(socket_, SOL_SOCKET, SO_REUSEADDR,
>-      reinterpret_cast<const char*>(&enabled), sizeof(enabled));
>+             reinterpret_cast<const char*>(&enabled), sizeof(enabled));
>   struct sockaddr_in addr = {0};
>   addr.sin_family = AF_INET;
>   addr.sin_addr.s_addr = htonl(INADDR_ANY);
>   addr.sin_port = htons(port);
>-  if (bind(socket_, reinterpret_cast<const sockaddr*>(&addr),
>-           sizeof(addr)) == SOCKET_ERROR) {
>+  if (bind(socket_, reinterpret_cast<const sockaddr*>(&addr), sizeof(addr)) ==
>+      SOCKET_ERROR) {
>     printf("bind failed\n");
>     return false;
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/server/data_socket.h b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/server/data_socket.h
>index 6bafa830a9e..4429bc5a00f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/server/data_socket.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/server/data_socket.h
>@@ -27,7 +27,7 @@ typedef int NativeSocket;
> #endif
> 
> #ifndef INVALID_SOCKET
>-#define INVALID_SOCKET  static_cast<NativeSocket>(-1)
>+#define INVALID_SOCKET static_cast<NativeSocket>(-1)
> #endif
> #endif
> 
>@@ -35,8 +35,8 @@ typedef int NativeSocket;
> 
> class SocketBase {
>  public:
>-  SocketBase() : socket_(INVALID_SOCKET) { }
>-  explicit SocketBase(NativeSocket socket) : socket_(socket) { }
>+  SocketBase() : socket_(INVALID_SOCKET) {}
>+  explicit SocketBase(NativeSocket socket) : socket_(socket) {}
>   ~SocketBase() { Close(); }
> 
>   NativeSocket socket() const { return socket_; }
>@@ -60,13 +60,9 @@ class DataSocket : public SocketBase {
>   };
> 
>   explicit DataSocket(NativeSocket socket)
>-      : SocketBase(socket),
>-        method_(INVALID),
>-        content_length_(0) {
>-  }
>+      : SocketBase(socket), method_(INVALID), content_length_(0) {}
> 
>-  ~DataSocket() {
>-  }
>+  ~DataSocket() {}
> 
>   static const char kCrossOriginAllowHeaders[];
> 
>@@ -110,9 +106,11 @@ class DataSocket : public SocketBase {
>   // header terminates with "\r\n".
>   // |data| is the body of the message.  It's length will be specified via
>   // a "Content-Length" header.
>-  bool Send(const std::string& status, bool connection_close,
>+  bool Send(const std::string& status,
>+            bool connection_close,
>             const std::string& content_type,
>-            const std::string& extra_headers, const std::string& data) const;
>+            const std::string& extra_headers,
>+            const std::string& data) const;
> 
>   // Clears all held state and prepares the socket for receiving a new request.
>   void Clear();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/server/main.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/server/main.cc
>index d26a1da4d7f..eafda606303 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/server/main.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/server/main.cc
>@@ -46,7 +46,7 @@ void HandleBrowserRequest(DataSocket* ds, bool* quit) {
>   }
> }
> 
>-int main(int argc, char** argv) {
>+int main(int argc, char* argv[]) {
>   std::string program_name = argv[0];
>   std::string usage = "Example usage: " + program_name + " --port=8888";
>   webrtc::test::CommandLineParser parser;
>@@ -94,7 +94,7 @@ int main(int argc, char** argv) {
>     for (SocketArray::iterator i = sockets.begin(); i != sockets.end(); ++i)
>       FD_SET((*i)->socket(), &socket_set);
> 
>-    struct timeval timeout = { 10, 0 };
>+    struct timeval timeout = {10, 0};
>     if (select(FD_SETSIZE, &socket_set, NULL, NULL, &timeout) == SOCKET_ERROR) {
>       printf("select failed\n");
>       break;
>@@ -111,8 +111,7 @@ int main(int argc, char** argv) {
>               if (s->PathEquals("/sign_in")) {
>                 clients.AddMember(s);
>               } else {
>-                printf("No member found for: %s\n",
>-                    s->request_path().c_str());
>+                printf("No member found for: %s\n", s->request_path().c_str());
>                 s->Send("500 Error", true, "text/plain", "",
>                         "Peer most likely gone.");
>               }
>@@ -127,7 +126,7 @@ int main(int argc, char** argv) {
>                 s->Send("200 OK", true, "text/plain", "", "");
>               } else {
>                 printf("Couldn't find target for request: %s\n",
>-                    s->request_path().c_str());
>+                       s->request_path().c_str());
>                 s->Send("500 Error", true, "text/plain", "",
>                         "Peer most likely gone.");
>               }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/server/peer_channel.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/server/peer_channel.cc
>index 69a0e104f3e..6ef08d7b21d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/server/peer_channel.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/server/peer_channel.cc
>@@ -37,7 +37,7 @@ using rtc::sprintfn;
> static const char kPeerIdHeader[] = "Pragma: ";
> 
> static const char* kRequestPaths[] = {
>-  "/wait", "/sign_out", "/message",
>+    "/wait", "/sign_out", "/message",
> };
> 
> enum RequestPathIndex {
>@@ -55,8 +55,10 @@ const size_t kMaxNameLength = 512;
> int ChannelMember::s_member_id_ = 0;
> 
> ChannelMember::ChannelMember(DataSocket* socket)
>-  : waiting_socket_(NULL), id_(++s_member_id_),
>-    connected_(true), timestamp_(time(NULL)) {
>+    : waiting_socket_(NULL),
>+      id_(++s_member_id_),
>+      connected_(true),
>+      timestamp_(time(NULL)) {
>   assert(socket);
>   assert(socket->method() == DataSocket::GET);
>   assert(socket->PathEquals("/sign_in"));
>@@ -69,8 +71,7 @@ ChannelMember::ChannelMember(DataSocket* socket)
>   std::replace(name_.begin(), name_.end(), ',', '_');
> }
> 
>-ChannelMember::~ChannelMember() {
>-}
>+ChannelMember::~ChannelMember() {}
> 
> bool ChannelMember::is_wait_request(DataSocket* ds) const {
>   return ds && ds->PathEquals(kRequestPaths[kWait]);
>@@ -87,8 +88,7 @@ std::string ChannelMember::GetPeerIdHeader() const {
> 
> bool ChannelMember::NotifyOfOtherMember(const ChannelMember& other) {
>   assert(&other != this);
>-  QueueResponse("200 OK", "text/plain", GetPeerIdHeader(),
>-                other.GetEntry());
>+  QueueResponse("200 OK", "text/plain", GetPeerIdHeader(), other.GetEntry());
>   return true;
> }
> 
>@@ -110,11 +110,9 @@ void ChannelMember::ForwardRequestToPeer(DataSocket* ds, ChannelMember* peer) {
>   std::string extra_headers(GetPeerIdHeader());
> 
>   if (peer == this) {
>-    ds->Send("200 OK", true, ds->content_type(), extra_headers,
>-             ds->data());
>+    ds->Send("200 OK", true, ds->content_type(), extra_headers, ds->data());
>   } else {
>-    printf("Client %s sending to %s\n",
>-        name_.c_str(), peer->name().c_str());
>+    printf("Client %s sending to %s\n", name_.c_str(), peer->name().c_str());
>     peer->QueueResponse("200 OK", ds->content_type(), extra_headers,
>                         ds->data());
>     ds->Send("200 OK", true, "text/plain", "", "");
>@@ -135,8 +133,8 @@ void ChannelMember::QueueResponse(const std::string& status,
>   if (waiting_socket_) {
>     assert(queue_.size() == 0);
>     assert(waiting_socket_->method() == DataSocket::GET);
>-    bool ok = waiting_socket_->Send(status, true, content_type, extra_headers,
>-                                    data);
>+    bool ok =
>+        waiting_socket_->Send(status, true, content_type, extra_headers, data);
>     if (!ok) {
>       printf("Failed to deliver data to waiting socket\n");
>     }
>@@ -251,7 +249,7 @@ bool PeerChannel::AddMember(DataSocket* ds) {
>   members_.push_back(new_guy);
> 
>   printf("New member added (total=%s): %s\n",
>-      size_t2str(members_.size()).c_str(), new_guy->name().c_str());
>+         size_t2str(members_.size()).c_str(), new_guy->name().c_str());
> 
>   // Let the newly connected peer know about other members of the channel.
>   std::string content_type;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/server/peer_channel.h b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/server/peer_channel.h
>index a9343d9aebb..c3bb1ad10b5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/server/peer_channel.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/server/peer_channel.h
>@@ -44,8 +44,10 @@ class ChannelMember {
> 
>   void OnClosing(DataSocket* ds);
> 
>-  void QueueResponse(const std::string& status, const std::string& content_type,
>-                     const std::string& extra_headers, const std::string& data);
>+  void QueueResponse(const std::string& status,
>+                     const std::string& content_type,
>+                     const std::string& extra_headers,
>+                     const std::string& data);
> 
>   void SetWaitingSocket(DataSocket* ds);
> 
>@@ -68,12 +70,9 @@ class PeerChannel {
>  public:
>   typedef std::vector<ChannelMember*> Members;
> 
>-  PeerChannel() {
>-  }
>+  PeerChannel() {}
> 
>-  ~PeerChannel() {
>-    DeleteAll();
>-  }
>+  ~PeerChannel() { DeleteAll(); }
> 
>   const Members& members() const { return members_; }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/server/utils.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/server/utils.cc
>index 1bcbe6bd916..dcd1ef74db8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/server/utils.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/peerconnection/server/utils.cc
>@@ -17,9 +17,9 @@
> using rtc::ToString;
> 
> std::string int2str(int i) {
>-  return ToString<int>(i);
>+  return ToString(i);
> }
> 
> std::string size_t2str(size_t i) {
>-  return ToString<size_t>(i);
>+  return ToString(i);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/relayserver/relayserver_main.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/relayserver/relayserver_main.cc
>index e3b18f1a92c..b98caba044f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/relayserver/relayserver_main.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/relayserver/relayserver_main.cc
>@@ -14,7 +14,7 @@
> #include "p2p/base/relayserver.h"
> #include "rtc_base/thread.h"
> 
>-int main(int argc, char **argv) {
>+int main(int argc, char** argv) {
>   if (argc != 3) {
>     std::cerr << "usage: relayserver internal-address external-address"
>               << std::endl;
>@@ -33,21 +33,21 @@ int main(int argc, char **argv) {
>     return 1;
>   }
> 
>-  rtc::Thread *pthMain = rtc::Thread::Current();
>+  rtc::Thread* pthMain = rtc::Thread::Current();
> 
>   std::unique_ptr<rtc::AsyncUDPSocket> int_socket(
>       rtc::AsyncUDPSocket::Create(pthMain->socketserver(), int_addr));
>   if (!int_socket) {
>-    std::cerr << "Failed to create a UDP socket bound at"
>-              << int_addr.ToString() << std::endl;
>+    std::cerr << "Failed to create a UDP socket bound at" << int_addr.ToString()
>+              << std::endl;
>     return 1;
>   }
> 
>   std::unique_ptr<rtc::AsyncUDPSocket> ext_socket(
>       rtc::AsyncUDPSocket::Create(pthMain->socketserver(), ext_addr));
>   if (!ext_socket) {
>-    std::cerr << "Failed to create a UDP socket bound at"
>-              << ext_addr.ToString() << std::endl;
>+    std::cerr << "Failed to create a UDP socket bound at" << ext_addr.ToString()
>+              << std::endl;
>     return 1;
>   }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/stunprober/main.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/stunprober/main.cc
>index e0f98773fe4..973db39d778 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/stunprober/main.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/stunprober/main.cc
>@@ -100,7 +100,7 @@ void StopTrial(rtc::Thread* thread, StunProber* prober, int result) {
> 
> }  // namespace
> 
>-int main(int argc, char** argv) {
>+int main(int argc, char* argv[]) {
>   rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true);
>   if (FLAG_help) {
>     rtc::FlagList::Print(nullptr, false);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/stunserver/stunserver_main.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/stunserver/stunserver_main.cc
>index 8129e1226d5..64d8b419a97 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/stunserver/stunserver_main.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/stunserver/stunserver_main.cc
>@@ -31,7 +31,7 @@ int main(int argc, char* argv[]) {
>     return 1;
>   }
> 
>-  rtc::Thread *pthMain = rtc::Thread::Current();
>+  rtc::Thread* pthMain = rtc::Thread::Current();
> 
>   rtc::AsyncUDPSocket* server_socket =
>       rtc::AsyncUDPSocket::Create(pthMain->socketserver(), server_addr);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/turnserver/turnserver_main.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/turnserver/turnserver_main.cc
>index 5433f951c10..edc0b699b4c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/turnserver/turnserver_main.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/turnserver/turnserver_main.cc
>@@ -21,10 +21,9 @@ static const char kSoftware[] = "libjingle TurnServer";
> 
> class TurnFileAuth : public cricket::TurnAuthInterface {
>  public:
>-  explicit TurnFileAuth(const std::string& path) : file_(path) {
>-    file_.Load();
>-  }
>-  virtual bool GetKey(const std::string& username, const std::string& realm,
>+  explicit TurnFileAuth(const std::string& path) : file_(path) { file_.Load(); }
>+  virtual bool GetKey(const std::string& username,
>+                      const std::string& realm,
>                       std::string* key) {
>     // File is stored as lines of <username>=<HA1>.
>     // Generate HA1 via "echo -n "<username>:<realm>:<password>" | md5sum"
>@@ -37,11 +36,12 @@ class TurnFileAuth : public cricket::TurnAuthInterface {
>     }
>     return ret;
>   }
>+
>  private:
>   rtc::OptionsFile file_;
> };
> 
>-int main(int argc, char **argv) {
>+int main(int argc, char* argv[]) {
>   if (argc != 5) {
>     std::cerr << "usage: turnserver int-addr ext-ip realm auth-file"
>               << std::endl;
>@@ -64,8 +64,8 @@ int main(int argc, char **argv) {
>   rtc::AsyncUDPSocket* int_socket =
>       rtc::AsyncUDPSocket::Create(main->socketserver(), int_addr);
>   if (!int_socket) {
>-    std::cerr << "Failed to create a UDP socket bound at"
>-              << int_addr.ToString() << std::endl;
>+    std::cerr << "Failed to create a UDP socket bound at" << int_addr.ToString()
>+              << std::endl;
>     return 1;
>   }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/unityplugin/ANDROID_INSTRUCTION b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/unityplugin/ANDROID_INSTRUCTION
>index 7f69faa8349..f0149eb953e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/unityplugin/ANDROID_INSTRUCTION
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/unityplugin/ANDROID_INSTRUCTION
>@@ -1,29 +1,25 @@
>-Instruction of Running webrtc_unity_plugin on Android Unity
>+Instruction of running webrtc_unity_plugin on Android Unity
> 
> 1. On Linux machine, compile target webrtc_unity_plugin.
>    Checkout WebRTC codebase: fetch --no-hooks webrtc_android
>    If you already have a checkout for linux, add target_os=âandroidâ into .gclient file.
>       Run gclient sync
>-   Modify files src/build/android/android_only_jni_exports.lst and src/build/android/android_only_explicit_jni_exports.lst to expose all functions. Namely, change "global" section to "*", and remove "local" section. Otherwise, Unity C# code will not be able to access the functions defined in the plugin. If you do this step after you create a build folder, you may have to clean and recreate the build folder.
>       Run gn args out/Android, and again set target_os=âandroidâ in the args.gn
>       Run ninja -C out/Android webrtc_unity_plugin
> 
>-2. On Linux machine, compile target libwebrtc_unity under webrtc checkout. This is the java code for webrtc to work on Android.
>+2. On Linux machine, build target libwebrtc_unity under webrtc checkout. This is the java code for webrtc to work on Android.
> 
> 3. Copy libwebrtc_unity.jar and libwebrtc_unity_plugin.so into Unity project folder, under Assets/Plugins/Android folder.
> 
>-4. Rename libwebrtc_unity_plugin.so to libjingle_peerconnection_so.so. Again, this is hacky, and the purpose is to let the java code in libwebrtc.jar to find their JNI implementation. And simultaneously, in your C# wrapper script for the native plugin libjingle_peerconnection_so.so, the dll_path should be set to âjingle_peerconnection_soâ.
>+4. Rename libwebrtc_unity_plugin.so to libjingle_peerconnection_so.so. This is hacky, and the purpose is to let the java code in libwebrtc_unity.jar to find their JNI implementations. Simultaneously, in your C# wrapper script for the native plugin libjingle_peerconnection_so.so, the dll_path should be set to âjingle_peerconnection_soâ.
> 
> 5. In the Unity Main Sceneâs Start method, write the following code to initialize the Java environment for webrtc (otherwise, webrtc will not be able to access audio device or camera from C++ code):
> 
> #if UNITY_ANDROID
>-        AndroidJavaClass playerClass = new AndroidJavaClass("com.unity3d.player.UnityPlayer");
>-        AndroidJavaObject activity = playerClass.GetStatic<AndroidJavaObject>("currentActivity");
>-        AndroidJavaClass webrtcClass = new AndroidJavaClass("org.webrtc.PeerConnectionFactory");
>-        if (webrtcClass != null)
>-        {
>-            webrtcClass.CallStatic("initializeAndroidGlobals", new object[2] { activity, false });
>-        }
>+    AndroidJavaClass playerClass = new AndroidJavaClass("com.unity3d.player.UnityPlayer");
>+    AndroidJavaObject activity = playerClass.GetStatic<AndroidJavaObject>("currentActivity");
>+    AndroidJavaClass utilityClass = new AndroidJavaClass("org.webrtc.UnityUtility");
>+    utilityClass.CallStatic("InitializePeerConncectionFactory", new object[1] { activity });
> #endif
> 
> 6. Compile the unity project into an APK, and decompile the apk using apktool that you can download from https://ibotpeaches.github.io/Apktool/
>@@ -32,6 +28,6 @@ Then copy the AndroidManifest.xml in the decompiled folder to the Assets/Plugins
>   <uses-permission android:name="android.permission.RECORD_AUDIO" />
>   <uses-permission android:name="android.permission.CAMERA" />
> 
>-  The purpose of using apktool is to get a well-written android manifest xml file. If you know how to write manifest file from scratch, you can skip using apktool.
>+The purpose of using apktool is to get a well-written android manifest xml file. If you know how to write manifest file from scratch, you can skip using apktool.
> 
> 7. Compile the unity project into an APK again and deploy it to an android device.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/unityplugin/DEPS b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/unityplugin/DEPS
>index 608c6ade4b7..604005ac734 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/unityplugin/DEPS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/unityplugin/DEPS
>@@ -1,3 +1,4 @@
> include_rules = [
>+  "+modules/utility",
>   "+sdk",
> ]
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/unityplugin/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/unityplugin/OWNERS
>index 61ea9a97f5d..343f8600f10 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/unityplugin/OWNERS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/unityplugin/OWNERS
>@@ -1 +1,2 @@
> gyzhou@chromium.org
>+qiangchen@chromium.org
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/unityplugin/java/src/org/webrtc/UnityUtility.java b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/unityplugin/java/src/org/webrtc/UnityUtility.java
>index b16d20ad403..5118bac5076 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/unityplugin/java/src/org/webrtc/UnityUtility.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/unityplugin/java/src/org/webrtc/UnityUtility.java
>@@ -12,6 +12,7 @@ package org.webrtc;
> 
> import android.content.Context;
> import java.util.List;
>+import javax.annotation.Nullable;
> 
> public class UnityUtility {
>   private static final String VIDEO_CAPTURER_THREAD_NAME = "VideoCapturerThread";
>@@ -26,7 +27,7 @@ public class UnityUtility {
>     return Camera2Enumerator.isSupported(ContextUtils.getApplicationContext());
>   }
> 
>-  private static VideoCapturer createCameraCapturer(CameraEnumerator enumerator) {
>+  private static @Nullable VideoCapturer createCameraCapturer(CameraEnumerator enumerator) {
>     final String[] deviceNames = enumerator.getDeviceNames();
> 
>     for (String deviceName : deviceNames) {
>@@ -47,11 +48,10 @@ public class UnityUtility {
>     VideoCapturer capturer =
>         createCameraCapturer(new Camera2Enumerator(ContextUtils.getApplicationContext()));
> 
>-    VideoCapturer.CapturerObserver capturerObserver =
>-        new AndroidVideoTrackSourceObserver(nativeTrackSource);
>+    VideoSource videoSource = new VideoSource(nativeTrackSource);
> 
>-    capturer.initialize(
>-        surfaceTextureHelper, ContextUtils.getApplicationContext(), capturerObserver);
>+    capturer.initialize(surfaceTextureHelper, ContextUtils.getApplicationContext(),
>+        videoSource.getCapturerObserver());
> 
>     capturer.startCapture(720, 480, 30);
>     return capturer;
>@@ -61,4 +61,9 @@ public class UnityUtility {
>     camera.stopCapture();
>     camera.dispose();
>   }
>+
>+  public static void InitializePeerConncectionFactory(Context context) throws InterruptedException {
>+    PeerConnectionFactory.initialize(
>+        PeerConnectionFactory.InitializationOptions.builder(context).createInitializationOptions());
>+  }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/unityplugin/simple_peer_connection.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/unityplugin/simple_peer_connection.cc
>index 9332a83a27b..df03e3767b6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/unityplugin/simple_peer_connection.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/unityplugin/simple_peer_connection.cc
>@@ -12,21 +12,32 @@
> 
> #include <utility>
> 
>+#include "absl/memory/memory.h"
>+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
>+#include "api/audio_codecs/builtin_audio_encoder_factory.h"
> #include "api/test/fakeconstraints.h"
> #include "api/videosourceproxy.h"
>+#include "media/engine/internaldecoderfactory.h"
>+#include "media/engine/internalencoderfactory.h"
>+#include "media/engine/multiplexcodecfactory.h"
> #include "media/engine/webrtcvideocapturerfactory.h"
>+#include "media/engine/webrtcvideodecoderfactory.h"
>+#include "media/engine/webrtcvideoencoderfactory.h"
>+#include "modules/audio_device/include/audio_device.h"
>+#include "modules/audio_processing/include/audio_processing.h"
> #include "modules/video_capture/video_capture_factory.h"
> 
> #if defined(WEBRTC_ANDROID)
> #include "examples/unityplugin/classreferenceholder.h"
>+#include "modules/utility/include/helpers_android.h"
> #include "sdk/android/src/jni/androidvideotracksource.h"
> #include "sdk/android/src/jni/jni_helpers.h"
> #endif
> 
>-// Names used for media stream labels.
>+// Names used for media stream ids.
> const char kAudioLabel[] = "audio_label";
> const char kVideoLabel[] = "video_label";
>-const char kStreamLabel[] = "stream_label";
>+const char kStreamId[] = "stream_id";
> 
> namespace {
> static int g_peer_count = 0;
>@@ -65,8 +76,9 @@ class DummySetSessionDescriptionObserver
>     return new rtc::RefCountedObject<DummySetSessionDescriptionObserver>();
>   }
>   virtual void OnSuccess() { RTC_LOG(INFO) << __FUNCTION__; }
>-  virtual void OnFailure(const std::string& error) {
>-    RTC_LOG(INFO) << __FUNCTION__ << " " << error;
>+  virtual void OnFailure(webrtc::RTCError error) {
>+    RTC_LOG(INFO) << __FUNCTION__ << " " << ToString(error.type()) << ": "
>+                  << error.message();
>   }
> 
>  protected:
>@@ -91,7 +103,15 @@ bool SimplePeerConnection::InitializePeerConnection(const char** turn_urls,
> 
>     g_peer_connection_factory = webrtc::CreatePeerConnectionFactory(
>         g_worker_thread.get(), g_worker_thread.get(), g_signaling_thread.get(),
>-        nullptr, nullptr, nullptr);
>+        nullptr, webrtc::CreateBuiltinAudioEncoderFactory(),
>+        webrtc::CreateBuiltinAudioDecoderFactory(),
>+        std::unique_ptr<webrtc::VideoEncoderFactory>(
>+            new webrtc::MultiplexEncoderFactory(
>+                absl::make_unique<webrtc::InternalEncoderFactory>())),
>+        std::unique_ptr<webrtc::VideoDecoderFactory>(
>+            new webrtc::MultiplexDecoderFactory(
>+                absl::make_unique<webrtc::InternalDecoderFactory>())),
>+        nullptr, nullptr);
>   }
>   if (!g_peer_connection_factory.get()) {
>     DeletePeerConnection();
>@@ -99,19 +119,19 @@ bool SimplePeerConnection::InitializePeerConnection(const char** turn_urls,
>   }
> 
>   g_peer_count++;
>-  if (!CreatePeerConnection(turn_urls, no_of_urls, username, credential,
>-                            is_receiver)) {
>+  if (!CreatePeerConnection(turn_urls, no_of_urls, username, credential)) {
>     DeletePeerConnection();
>     return false;
>   }
>+
>+  mandatory_receive_ = is_receiver;
>   return peer_connection_.get() != nullptr;
> }
> 
> bool SimplePeerConnection::CreatePeerConnection(const char** turn_urls,
>                                                 const int no_of_urls,
>                                                 const char* username,
>-                                                const char* credential,
>-                                                bool is_receiver) {
>+                                                const char* credential) {
>   RTC_DCHECK(g_peer_connection_factory.get() != nullptr);
>   RTC_DCHECK(peer_connection_.get() == nullptr);
> 
>@@ -148,11 +168,6 @@ bool SimplePeerConnection::CreatePeerConnection(const char** turn_urls,
>   webrtc::FakeConstraints constraints;
>   constraints.SetAllowDtlsSctpDataChannels();
> 
>-  if (is_receiver) {
>-    constraints.SetMandatoryReceiveAudio(true);
>-    constraints.SetMandatoryReceiveVideo(true);
>-  }
>-
>   peer_connection_ = g_peer_connection_factory->CreatePeerConnection(
>       config_, &constraints, nullptr, nullptr, this);
> 
>@@ -167,7 +182,7 @@ void SimplePeerConnection::DeletePeerConnection() {
>     JNIEnv* env = webrtc::jni::GetEnv();
>     jclass pc_factory_class =
>         unity_plugin::FindClass(env, "org/webrtc/UnityUtility");
>-    jmethodID stop_camera_method = webrtc::jni::GetStaticMethodID(
>+    jmethodID stop_camera_method = webrtc::GetStaticMethodID(
>         env, pc_factory_class, "StopCamera", "(Lorg/webrtc/VideoCapturer;)V");
> 
>     env->CallStaticVoidMethod(pc_factory_class, stop_camera_method, g_camera);
>@@ -192,7 +207,12 @@ bool SimplePeerConnection::CreateOffer() {
>   if (!peer_connection_.get())
>     return false;
> 
>-  peer_connection_->CreateOffer(this, nullptr);
>+  webrtc::FakeConstraints constraints;
>+  if (mandatory_receive_) {
>+    constraints.SetMandatoryReceiveAudio(true);
>+    constraints.SetMandatoryReceiveVideo(true);
>+  }
>+  peer_connection_->CreateOffer(this, &constraints);
>   return true;
> }
> 
>@@ -200,7 +220,12 @@ bool SimplePeerConnection::CreateAnswer() {
>   if (!peer_connection_.get())
>     return false;
> 
>-  peer_connection_->CreateAnswer(this, nullptr);
>+  webrtc::FakeConstraints constraints;
>+  if (mandatory_receive_) {
>+    constraints.SetMandatoryReceiveAudio(true);
>+    constraints.SetMandatoryReceiveVideo(true);
>+  }
>+  peer_connection_->CreateAnswer(this, &constraints);
>   return true;
> }
> 
>@@ -216,11 +241,12 @@ void SimplePeerConnection::OnSuccess(
>     OnLocalSdpReady(desc->type().c_str(), sdp.c_str());
> }
> 
>-void SimplePeerConnection::OnFailure(const std::string& error) {
>-  RTC_LOG(LERROR) << error;
>+void SimplePeerConnection::OnFailure(webrtc::RTCError error) {
>+  RTC_LOG(LERROR) << ToString(error.type()) << ": " << error.message();
> 
>+  // TODO(hta): include error.type in the message
>   if (OnFailureMessage)
>-    OnFailureMessage(error.c_str());
>+    OnFailureMessage(error.message());
> }
> 
> void SimplePeerConnection::OnIceCandidate(
>@@ -354,7 +380,7 @@ void SimplePeerConnection::SetAudioControl() {
> 
> void SimplePeerConnection::OnAddStream(
>     rtc::scoped_refptr<webrtc::MediaStreamInterface> stream) {
>-  RTC_LOG(INFO) << __FUNCTION__ << " " << stream->label();
>+  RTC_LOG(INFO) << __FUNCTION__ << " " << stream->id();
>   remote_stream_ = stream;
>   if (remote_video_observer_ && !remote_stream_->GetVideoTracks().empty()) {
>     remote_stream_->GetVideoTracks()[0]->AddOrUpdateSink(
>@@ -395,15 +421,16 @@ SimplePeerConnection::OpenVideoCaptureDevice() {
> }
> 
> void SimplePeerConnection::AddStreams(bool audio_only) {
>-  if (active_streams_.find(kStreamLabel) != active_streams_.end())
>+  if (active_streams_.find(kStreamId) != active_streams_.end())
>     return;  // Already added.
> 
>   rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
>-      g_peer_connection_factory->CreateLocalMediaStream(kStreamLabel);
>+      g_peer_connection_factory->CreateLocalMediaStream(kStreamId);
> 
>   rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
>       g_peer_connection_factory->CreateAudioTrack(
>-          kAudioLabel, g_peer_connection_factory->CreateAudioSource(nullptr)));
>+          kAudioLabel, g_peer_connection_factory->CreateAudioSource(
>+                           cricket::AudioOptions())));
>   std::string id = audio_track->id();
>   stream->AddTrack(audio_track);
> 
>@@ -412,7 +439,7 @@ void SimplePeerConnection::AddStreams(bool audio_only) {
>     JNIEnv* env = webrtc::jni::GetEnv();
>     jclass pc_factory_class =
>         unity_plugin::FindClass(env, "org/webrtc/UnityUtility");
>-    jmethodID load_texture_helper_method = webrtc::jni::GetStaticMethodID(
>+    jmethodID load_texture_helper_method = webrtc::GetStaticMethodID(
>         env, pc_factory_class, "LoadSurfaceTextureHelper",
>         "()Lorg/webrtc/SurfaceTextureHelper;");
>     jobject texture_helper = env->CallStaticObjectMethod(
>@@ -421,15 +448,15 @@ void SimplePeerConnection::AddStreams(bool audio_only) {
>     RTC_DCHECK(texture_helper != nullptr)
>         << "Cannot get the Surface Texture Helper.";
> 
>-    rtc::scoped_refptr<AndroidVideoTrackSource> source(
>-        new rtc::RefCountedObject<AndroidVideoTrackSource>(
>-            g_signaling_thread.get(), env, texture_helper, false));
>+    rtc::scoped_refptr<webrtc::jni::AndroidVideoTrackSource> source(
>+        new rtc::RefCountedObject<webrtc::jni::AndroidVideoTrackSource>(
>+            g_signaling_thread.get(), env, false));
>     rtc::scoped_refptr<webrtc::VideoTrackSourceProxy> proxy_source =
>         webrtc::VideoTrackSourceProxy::Create(g_signaling_thread.get(),
>                                               g_worker_thread.get(), source);
> 
>     // link with VideoCapturer (Camera);
>-    jmethodID link_camera_method = webrtc::jni::GetStaticMethodID(
>+    jmethodID link_camera_method = webrtc::GetStaticMethodID(
>         env, pc_factory_class, "LinkCamera",
>         "(JLorg/webrtc/SurfaceTextureHelper;)Lorg/webrtc/VideoCapturer;");
>     jobject camera_tmp =
>@@ -466,7 +493,7 @@ void SimplePeerConnection::AddStreams(bool audio_only) {
>   typedef std::pair<std::string,
>                     rtc::scoped_refptr<webrtc::MediaStreamInterface>>
>       MediaStreamPair;
>-  active_streams_.insert(MediaStreamPair(stream->label(), stream));
>+  active_streams_.insert(MediaStreamPair(stream->id(), stream));
> }
> 
> bool SimplePeerConnection::CreateDataChannel() {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/unityplugin/simple_peer_connection.h b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/unityplugin/simple_peer_connection.h
>index 0c7a2615d76..5b30778711f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/unityplugin/simple_peer_connection.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/unityplugin/simple_peer_connection.h
>@@ -64,8 +64,7 @@ class SimplePeerConnection : public webrtc::PeerConnectionObserver,
>   bool CreatePeerConnection(const char** turn_urls,
>                             const int no_of_urls,
>                             const char* username,
>-                            const char* credential,
>-                            bool is_receiver);
>+                            const char* credential);
>   void CloseDataChannel();
>   std::unique_ptr<cricket::VideoCapturer> OpenVideoCaptureDevice();
>   void SetAudioControl();
>@@ -89,7 +88,7 @@ class SimplePeerConnection : public webrtc::PeerConnectionObserver,
> 
>   // CreateSessionDescriptionObserver implementation.
>   void OnSuccess(webrtc::SessionDescriptionInterface* desc) override;
>-  void OnFailure(const std::string& error) override;
>+  void OnFailure(webrtc::RTCError error) override;
> 
>   // DataChannelObserver implementation.
>   void OnStateChange() override;
>@@ -127,6 +126,7 @@ class SimplePeerConnection : public webrtc::PeerConnectionObserver,
> 
>   bool is_mute_audio_ = false;
>   bool is_record_audio_ = false;
>+  bool mandatory_receive_ = false;
> 
>   // disallow copy-and-assign
>   SimplePeerConnection(const SimplePeerConnection&) = delete;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/unityplugin/unity_plugin_apis.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/unityplugin/unity_plugin_apis.cc
>index ae98a833f5d..34c28d926a3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/unityplugin/unity_plugin_apis.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/unityplugin/unity_plugin_apis.cc
>@@ -24,12 +24,13 @@ static std::map<int, rtc::scoped_refptr<SimplePeerConnection>>
> int CreatePeerConnection(const char** turn_urls,
>                          const int no_of_urls,
>                          const char* username,
>-                         const char* credential) {
>+                         const char* credential,
>+                         bool mandatory_receive_video) {
>   g_peer_connection_map[g_peer_connection_id] =
>       new rtc::RefCountedObject<SimplePeerConnection>();
> 
>   if (!g_peer_connection_map[g_peer_connection_id]->InitializePeerConnection(
>-          turn_urls, no_of_urls, username, credential, false))
>+          turn_urls, no_of_urls, username, credential, mandatory_receive_video))
>     return -1;
> 
>   return g_peer_connection_id++;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/unityplugin/unity_plugin_apis.h b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/unityplugin/unity_plugin_apis.h
>index 814b9675fb2..b32f9e2caf2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/unityplugin/unity_plugin_apis.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/unityplugin/unity_plugin_apis.h
>@@ -19,9 +19,11 @@
> typedef void (*I420FRAMEREADY_CALLBACK)(const uint8_t* data_y,
>                                         const uint8_t* data_u,
>                                         const uint8_t* data_v,
>+                                        const uint8_t* data_a,
>                                         int stride_y,
>                                         int stride_u,
>                                         int stride_v,
>+                                        int stride_a,
>                                         uint32_t width,
>                                         uint32_t height);
> typedef void (*LOCALDATACHANNELREADY_CALLBACK)();
>@@ -47,7 +49,8 @@ extern "C" {
> WEBRTC_PLUGIN_API int CreatePeerConnection(const char** turn_urls,
>                                            const int no_of_urls,
>                                            const char* username,
>-                                           const char* credential);
>+                                           const char* credential,
>+                                           bool mandatory_receive_video);
> // Close a peerconnection.
> WEBRTC_PLUGIN_API bool ClosePeerConnection(int peer_connection_id);
> // Add a audio stream. If audio_only is true, the stream only has an audio
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/unityplugin/video_observer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/unityplugin/video_observer.cc
>index 821acd66aaa..a78ef57e417 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/unityplugin/video_observer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/unityplugin/video_observer.cc
>@@ -17,11 +17,28 @@ void VideoObserver::SetVideoCallback(I420FRAMEREADY_CALLBACK callback) {
> 
> void VideoObserver::OnFrame(const webrtc::VideoFrame& frame) {
>   std::unique_lock<std::mutex> lock(mutex);
>-  rtc::scoped_refptr<webrtc::PlanarYuvBuffer> buffer(
>-      frame.video_frame_buffer()->ToI420());
>-  if (OnI420FrameReady) {
>-    OnI420FrameReady(buffer->DataY(), buffer->DataU(), buffer->DataV(),
>-                     buffer->StrideY(), buffer->StrideU(), buffer->StrideV(),
>+  if (!OnI420FrameReady)
>+    return;
>+
>+  rtc::scoped_refptr<webrtc::VideoFrameBuffer> buffer(
>+      frame.video_frame_buffer());
>+
>+  if (buffer->type() != webrtc::VideoFrameBuffer::Type::kI420A) {
>+    rtc::scoped_refptr<webrtc::I420BufferInterface> i420_buffer =
>+        buffer->ToI420();
>+    OnI420FrameReady(i420_buffer->DataY(), i420_buffer->DataU(),
>+                     i420_buffer->DataV(), nullptr, i420_buffer->StrideY(),
>+                     i420_buffer->StrideU(), i420_buffer->StrideV(), 0,
>+                     frame.width(), frame.height());
>+
>+  } else {
>+    // The buffer has alpha channel.
>+    webrtc::I420ABufferInterface* i420a_buffer = buffer->GetI420A();
>+
>+    OnI420FrameReady(i420a_buffer->DataY(), i420a_buffer->DataU(),
>+                     i420a_buffer->DataV(), i420a_buffer->DataA(),
>+                     i420a_buffer->StrideY(), i420a_buffer->StrideU(),
>+                     i420a_buffer->StrideV(), i420a_buffer->StrideA(),
>                      frame.width(), frame.height());
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/unityplugin/video_observer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/unityplugin/video_observer.h
>index f2b06b3d495..84c03d329a9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/unityplugin/video_observer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/examples/unityplugin/video_observer.h
>@@ -14,8 +14,8 @@
> #include <mutex>
> 
> #include "api/mediastreaminterface.h"
>+#include "api/video/video_sink_interface.h"
> #include "examples/unityplugin/unity_plugin_apis.h"
>-#include "media/base/videosinkinterface.h"
> 
> class VideoObserver : public rtc::VideoSinkInterface<webrtc::VideoFrame> {
>  public:
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/infra/config/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/infra/config/OWNERS
>index b159542bc40..67dfd2fc52b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/infra/config/OWNERS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/infra/config/OWNERS
>@@ -1,3 +1,4 @@
> set noparent
> sergiyb@chromium.org
> phoglund@webrtc.org
>+oprypin@webrtc.org
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/infra/config/cq.cfg b/Source/ThirdParty/libwebrtc/Source/webrtc/infra/config/cq.cfg
>index ef656d3c2f9..d4f8d963e3a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/infra/config/cq.cfg
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/infra/config/cq.cfg
>@@ -2,7 +2,6 @@
> # at http://luci-config.appspot.com/schemas/projects/refs:cq.cfg.
> 
> version: 1
>-cq_name: "webrtc"
> cq_status_url: "https://chromium-cq-status.appspot.com"
> git_repo_url: "https://webrtc.googlesource.com/src"
> commit_burst_delay: 60
>@@ -22,16 +21,17 @@ verifiers {
> 
>   try_job {
>     buckets {
>-      name: "master.tryserver.webrtc"
>+      name: "luci.webrtc.try"
>       builders { name: "android_dbg" }
>       builders { name: "android_rel" }
>       builders { name: "android_arm64_rel" }
>-      builders { name: "android_compile_mips_dbg" }
>+      builders { name: "android_chromium_compile" }
>+      builders { name: "android_compile_arm64_rel" }
>+      builders { name: "android_compile_rel" }
>       builders { name: "android_compile_x86_dbg" }
>       builders { name: "android_compile_x86_rel" }
>       builders { name: "android_compile_x64_dbg" }
>       builders { name: "android_more_configs" }
>-      builders { name: "android_webrtc_compile_rel" }
>       builders { name: "ios_arm64_dbg" }
>       builders { name: "ios_arm64_rel" }
>       builders { name: "ios_dbg" }
>@@ -48,12 +48,10 @@ verifiers {
>       builders { name: "linux_arm64_dbg" }
>       builders { name: "linux_arm64_rel" }
>       builders { name: "linux_asan" }
>-      builders { name: "linux_baremetal" }
>-      builders { name: "linux_chromium_webrtc_compile_rel_ng" }
>+      builders { name: "linux_chromium_compile" }
>       builders { name: "linux_compile_dbg" }
>+      builders { name: "linux_compile_rel" }
>       builders { name: "linux_libfuzzer_rel" }
>-      # TODO(bugs.webrtc.org/8356): Investigate fixing and re-enabling.
>-      #builders { name: "linux_memcheck" }
>       builders { name: "linux_msan" }
>       builders { name: "linux_rel" }
>       builders { name: "linux_tsan2" }
>@@ -62,14 +60,12 @@ verifiers {
>       builders { name: "linux_gcc_rel" }
>       builders { name: "linux_more_configs" }
>       builders { name: "mac_asan" }
>-      builders { name: "mac_baremetal" }
>-      builders { name: "mac_chromium_webrtc_compile_rel_ng" }
>+      builders { name: "mac_chromium_compile" }
>       builders { name: "mac_compile_dbg" }
>       builders { name: "mac_rel" }
>       builders { name: "presubmit" }
>       builders { name: "win_asan" }
>-      builders { name: "win_baremetal" }
>-      builders { name: "win_chromium_webrtc_compile_rel_ng" }
>+      builders { name: "win_chromium_compile" }
>       builders { name: "win_clang_dbg" }
>       builders { name: "win_clang_rel" }
>       builders { name: "win_dbg" }
>@@ -79,6 +75,15 @@ verifiers {
>       builders { name: "win_x64_dbg" }
>       builders { name: "win_x64_rel" }
>       builders { name: "win_more_configs" }
>+      builders {
>+        name: "noop"
>+        equivalent_to {
>+          bucket: "master.internal.tryserver.corp.webrtc"
>+          builder: "linux_internal_compile_lite"
>+          owner_whitelist_group: "project-webrtc-internal-tryjob-access"
>+          percentage: 100
>+        }
>+      }
>     }
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/BUILD.gn
>index f1afddccd54..843c38dbd93 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/BUILD.gn
>@@ -17,18 +17,62 @@ if (is_android) {
> 
> group("logging") {
>   deps = [
>-    ":rtc_event_log_impl",
>+    ":rtc_event_audio",
>+    ":rtc_event_bwe",
>+    ":rtc_event_log_impl_base",
>+    ":rtc_event_log_impl_encoder",
>+    ":rtc_event_log_impl_output",
>+    ":rtc_event_pacing",
>+    ":rtc_event_rtp_rtcp",
>+    ":rtc_event_video",
>   ]
>-  if (rtc_enable_protobuf) {
>-    deps += [ ":rtc_event_log_parser" ]
>-  }
> }
> 
> rtc_source_set("rtc_event_log_api") {
>   sources = [
>+    "rtc_event_log/encoder/rtc_event_log_encoder.h",
>     "rtc_event_log/events/rtc_event.h",
>+    "rtc_event_log/rtc_event_log.cc",
>+    "rtc_event_log/rtc_event_log.h",
>+    "rtc_event_log/rtc_event_log_factory_interface.h",
>+  ]
>+
>+  deps = [
>+    "../api:libjingle_logging_api",
>+    "../rtc_base:ptr_util",
>+    "../rtc_base:rtc_base_approved",
>+    "../rtc_base:rtc_task_queue",
>+  ]
>+}
>+
>+rtc_source_set("rtc_stream_config") {
>+  sources = [
>+    "rtc_event_log/rtc_stream_config.cc",
>+    "rtc_event_log/rtc_stream_config.h",
>+  ]
>+
>+  deps = [
>+    ":rtc_event_log_api",
>+    "..:webrtc_common",
>+    "../api:libjingle_peerconnection_api",
>+  ]
>+}
>+
>+rtc_source_set("rtc_event_pacing") {
>+  sources = [
>     "rtc_event_log/events/rtc_event_alr_state.cc",
>     "rtc_event_log/events/rtc_event_alr_state.h",
>+  ]
>+
>+  deps = [
>+    ":rtc_event_log_api",
>+    "../rtc_base:ptr_util",
>+    "//third_party/abseil-cpp/absl/memory",
>+  ]
>+}
>+
>+rtc_source_set("rtc_event_audio") {
>+  sources = [
>     "rtc_event_log/events/rtc_event_audio_network_adaptation.cc",
>     "rtc_event_log/events/rtc_event_audio_network_adaptation.h",
>     "rtc_event_log/events/rtc_event_audio_playout.cc",
>@@ -37,6 +81,19 @@ rtc_source_set("rtc_event_log_api") {
>     "rtc_event_log/events/rtc_event_audio_receive_stream_config.h",
>     "rtc_event_log/events/rtc_event_audio_send_stream_config.cc",
>     "rtc_event_log/events/rtc_event_audio_send_stream_config.h",
>+  ]
>+
>+  deps = [
>+    ":rtc_event_log_api",
>+    ":rtc_stream_config",
>+    "../modules/audio_coding:audio_network_adaptor_config",
>+    "../rtc_base:ptr_util",
>+    "//third_party/abseil-cpp/absl/memory",
>+  ]
>+}
>+
>+rtc_source_set("rtc_event_bwe") {
>+  sources = [
>     "rtc_event_log/events/rtc_event_bwe_update_delay_based.cc",
>     "rtc_event_log/events/rtc_event_bwe_update_delay_based.h",
>     "rtc_event_log/events/rtc_event_bwe_update_loss_based.cc",
>@@ -47,6 +104,18 @@ rtc_source_set("rtc_event_log_api") {
>     "rtc_event_log/events/rtc_event_probe_result_failure.h",
>     "rtc_event_log/events/rtc_event_probe_result_success.cc",
>     "rtc_event_log/events/rtc_event_probe_result_success.h",
>+  ]
>+
>+  deps = [
>+    ":rtc_event_log_api",
>+    "../modules/remote_bitrate_estimator:remote_bitrate_estimator",
>+    "../rtc_base:ptr_util",
>+    "//third_party/abseil-cpp/absl/memory",
>+  ]
>+}
>+
>+rtc_source_set("rtc_event_rtp_rtcp") {
>+  sources = [
>     "rtc_event_log/events/rtc_event_rtcp_packet_incoming.cc",
>     "rtc_event_log/events/rtc_event_rtcp_packet_incoming.h",
>     "rtc_event_log/events/rtc_event_rtcp_packet_outgoing.cc",
>@@ -55,77 +124,124 @@ rtc_source_set("rtc_event_log_api") {
>     "rtc_event_log/events/rtc_event_rtp_packet_incoming.h",
>     "rtc_event_log/events/rtc_event_rtp_packet_outgoing.cc",
>     "rtc_event_log/events/rtc_event_rtp_packet_outgoing.h",
>+  ]
>+
>+  deps = [
>+    ":rtc_event_log_api",
>+    "../api:array_view",
>+    "../modules/rtp_rtcp:rtp_rtcp_format",
>+    "../rtc_base:ptr_util",
>+    "../rtc_base:rtc_base_approved",
>+    "//third_party/abseil-cpp/absl/memory",
>+  ]
>+}
>+
>+rtc_source_set("rtc_event_video") {
>+  sources = [
>     "rtc_event_log/events/rtc_event_video_receive_stream_config.cc",
>     "rtc_event_log/events/rtc_event_video_receive_stream_config.h",
>     "rtc_event_log/events/rtc_event_video_send_stream_config.cc",
>     "rtc_event_log/events/rtc_event_video_send_stream_config.h",
>-    "rtc_event_log/output/rtc_event_log_output_file.cc",
>-    "rtc_event_log/output/rtc_event_log_output_file.h",
>-    "rtc_event_log/rtc_event_log.h",
>-    "rtc_event_log/rtc_event_log_factory_interface.h",
>-    "rtc_event_log/rtc_stream_config.cc",
>-    "rtc_event_log/rtc_stream_config.h",
>   ]
> 
>   deps = [
>-    "..:webrtc_common",
>-    "../:typedefs",
>-    "../api:array_view",
>-    "../api:libjingle_logging_api",
>-    "../api:libjingle_peerconnection_api",
>-    "../call:video_stream_api",
>-    "../modules/audio_coding:audio_network_adaptor_config",
>+    ":rtc_event_log_api",
>+    ":rtc_stream_config",
>+    "../rtc_base:ptr_util",
>+    "//third_party/abseil-cpp/absl/memory",
>+  ]
>+}
>+
>+rtc_static_library("rtc_event_log_impl_encoder") {
>+  visibility = [ "*" ]
>+  sources = [
>+    "rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc",
>+    "rtc_event_log/encoder/rtc_event_log_encoder_legacy.h",
>+  ]
>+
>+  defines = []
>+
>+  deps = [
>+    ":ice_log",
>+    ":rtc_event_audio",
>+    ":rtc_event_bwe",
>+    ":rtc_event_log_api",
>+    ":rtc_event_log_impl_output",
>+    ":rtc_event_pacing",
>+    ":rtc_event_rtp_rtcp",
>+    ":rtc_event_video",
>+    ":rtc_stream_config",
>+    "../modules/audio_coding:audio_network_adaptor",
>     "../modules/remote_bitrate_estimator:remote_bitrate_estimator",
>     "../modules/rtp_rtcp:rtp_rtcp_format",
>     "../rtc_base:checks",
>     "../rtc_base:rtc_base_approved",
>-    "../system_wrappers",
>   ]
> 
>-  # TODO(eladalon): Remove this.
>-  if (!build_with_chromium && is_clang) {
>-    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>-    suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>+  if (rtc_enable_protobuf) {
>+    defines += [ "ENABLE_RTC_EVENT_LOG" ]
>+    deps += [ ":rtc_event_log_proto" ]
>   }
> }
> 
>-rtc_static_library("rtc_event_log_impl") {
>+rtc_source_set("rtc_event_log_impl_output") {
>+  sources = [
>+    "rtc_event_log/output/rtc_event_log_output_file.cc",
>+    "rtc_event_log/output/rtc_event_log_output_file.h",
>+  ]
>+
>+  deps = [
>+    ":rtc_event_log_api",
>+    "../api:libjingle_logging_api",
>+    "../rtc_base:checks",
>+    "../rtc_base:rtc_base_approved",
>+  ]
>+}
>+
>+rtc_static_library("rtc_event_log_impl_base") {
>   visibility = [ "*" ]
>   sources = [
>-    "rtc_event_log/encoder/rtc_event_log_encoder.h",
>-    "rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc",
>-    "rtc_event_log/encoder/rtc_event_log_encoder_legacy.h",
>-    "rtc_event_log/rtc_event_log.cc",
>     "rtc_event_log/rtc_event_log_factory.cc",
>     "rtc_event_log/rtc_event_log_factory.h",
>+    "rtc_event_log/rtc_event_log_impl.cc",
>   ]
> 
>   defines = []
> 
>   deps = [
>+    ":ice_log",
>     ":rtc_event_log_api",
>-    "..:webrtc_common",
>-    "../modules/audio_coding:audio_network_adaptor",
>-    "../modules/remote_bitrate_estimator:remote_bitrate_estimator",
>-    "../modules/rtp_rtcp:rtp_rtcp_format",
>+    ":rtc_event_log_impl_encoder",
>+    ":rtc_event_log_impl_output",
>     "../rtc_base:checks",
>-    "../rtc_base:protobuf_utils",
>     "../rtc_base:rtc_base_approved",
>     "../rtc_base:rtc_task_queue",
>+    "../rtc_base:safe_minmax",
>     "../rtc_base:sequenced_task_checker",
>-    "../system_wrappers",
>+    "//third_party/abseil-cpp/absl/memory",
>   ]
> 
>   if (rtc_enable_protobuf) {
>     defines += [ "ENABLE_RTC_EVENT_LOG" ]
>     deps += [ ":rtc_event_log_proto" ]
>   }
>+}
> 
>-  # TODO(eladalon): Remove this.
>-  if (!build_with_chromium && is_clang) {
>-    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>-    suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-  }
>+rtc_source_set("fake_rtc_event_log") {
>+  testonly = true
>+  sources = [
>+    "rtc_event_log/fake_rtc_event_log.cc",
>+    "rtc_event_log/fake_rtc_event_log.h",
>+    "rtc_event_log/fake_rtc_event_log_factory.cc",
>+    "rtc_event_log/fake_rtc_event_log_factory.h",
>+  ]
>+
>+  deps = [
>+    ":ice_log",
>+    ":rtc_event_log_api",
>+    "../rtc_base:checks",
>+    "../rtc_base:rtc_base",
>+  ]
> }
> 
> if (rtc_enable_protobuf) {
>@@ -144,16 +260,23 @@ if (rtc_enable_protobuf) {
>   }
> 
>   rtc_static_library("rtc_event_log_parser") {
>+    visibility = [ "*" ]
>     sources = [
>       "rtc_event_log/rtc_event_log_parser.cc",
>       "rtc_event_log/rtc_event_log_parser.h",
>+      "rtc_event_log/rtc_event_log_parser_new.cc",
>+      "rtc_event_log/rtc_event_log_parser_new.h",
>     ]
> 
>     deps = [
>+      ":ice_log",
>+      ":rtc_event_bwe",
>       ":rtc_event_log2_proto",
>       ":rtc_event_log_api",
>       ":rtc_event_log_proto",
>+      ":rtc_stream_config",
>       "..:webrtc_common",
>+      "../api:libjingle_peerconnection_api",
>       "../call:video_stream_api",
>       "../modules/audio_coding:audio_network_adaptor",
>       "../modules/remote_bitrate_estimator:remote_bitrate_estimator",
>@@ -162,7 +285,7 @@ if (rtc_enable_protobuf) {
>       "../rtc_base:checks",
>       "../rtc_base:protobuf_utils",
>       "../rtc_base:rtc_base_approved",
>-      "../system_wrappers",
>+      "//third_party/abseil-cpp/absl/memory",
>     ]
> 
>     if (!build_with_chromium && is_clang) {
>@@ -176,9 +299,6 @@ if (rtc_enable_protobuf) {
>       testonly = true
>       assert(rtc_enable_protobuf)
>       defines = [ "ENABLE_RTC_EVENT_LOG" ]
>-      if (rtc_use_memcheck) {
>-        defines += [ "WEBRTC_USE_MEMCHECK" ]
>-      }
>       sources = [
>         "rtc_event_log/encoder/rtc_event_log_encoder_unittest.cc",
>         "rtc_event_log/output/rtc_event_log_output_file_unittest.cc",
>@@ -187,10 +307,19 @@ if (rtc_enable_protobuf) {
>         "rtc_event_log/rtc_event_log_unittest_helper.h",
>       ]
>       deps = [
>+        ":ice_log",
>+        ":rtc_event_audio",
>+        ":rtc_event_bwe",
>         ":rtc_event_log_api",
>-        ":rtc_event_log_impl",
>+        ":rtc_event_log_impl_base",
>+        ":rtc_event_log_impl_encoder",
>+        ":rtc_event_log_impl_output",
>         ":rtc_event_log_parser",
>         ":rtc_event_log_proto",
>+        ":rtc_event_pacing",
>+        ":rtc_event_rtp_rtcp",
>+        ":rtc_event_video",
>+        ":rtc_stream_config",
>         "../api:libjingle_peerconnection_api",
>         "../call",
>         "../call:call_interfaces",
>@@ -200,16 +329,17 @@ if (rtc_enable_protobuf) {
>         "../rtc_base:checks",
>         "../rtc_base:rtc_base_approved",
>         "../rtc_base:rtc_base_tests_utils",
>-        "../system_wrappers:metrics_default",
>+        "../test:fileutils",
>         "../test:test_support",
>-        "//testing/gmock",
>         "//testing/gtest",
>+        "//third_party/abseil-cpp/absl/memory",
>       ]
>       if (!build_with_chromium && is_clang) {
>         # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>         suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>       }
>     }
>+
>     rtc_test("rtc_event_log2rtp_dump") {
>       testonly = true
>       sources = [
>@@ -217,7 +347,6 @@ if (rtc_enable_protobuf) {
>       ]
>       deps = [
>         ":rtc_event_log_api",
>-        ":rtc_event_log_impl",
>         ":rtc_event_log_parser",
>         "../modules/rtp_rtcp",
>         "../modules/rtp_rtcp:rtp_rtcp_format",
>@@ -243,7 +372,6 @@ if (rtc_enable_protobuf) {
>       ]
>       deps = [
>         ":rtc_event_log_api",
>-        ":rtc_event_log_impl",
>         ":rtc_event_log_parser",
>         "../:webrtc_common",
>         "../call:video_stream_api",
>@@ -251,6 +379,7 @@ if (rtc_enable_protobuf) {
>         "../rtc_base:checks",
>         "../rtc_base:protobuf_utils",
>         "../rtc_base:rtc_base_approved",
>+        "../rtc_base:stringutils",
> 
>         # TODO(kwiberg): Remove this dependency.
>         "../api/audio_codecs:audio_codecs_api",
>@@ -271,28 +400,42 @@ if (rtc_enable_protobuf) {
>       ]
>       deps = [
>         ":rtc_event_log_api",
>-        ":rtc_event_log_impl",
>         ":rtc_event_log_proto",
>         "../rtc_base:checks",
>         "../rtc_base:rtc_base_approved",
>       ]
>-      if (!build_with_chromium && is_clang) {
>-        # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>-        suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-      }
>     }
>   }
> }
> 
>+rtc_source_set("ice_log") {
>+  sources = [
>+    "rtc_event_log/events/rtc_event_ice_candidate_pair.cc",
>+    "rtc_event_log/events/rtc_event_ice_candidate_pair.h",
>+    "rtc_event_log/events/rtc_event_ice_candidate_pair_config.cc",
>+    "rtc_event_log/events/rtc_event_ice_candidate_pair_config.h",
>+    "rtc_event_log/icelogger.cc",
>+    "rtc_event_log/icelogger.h",
>+  ]
>+
>+  deps = [
>+    ":rtc_event_log_api",
>+    "../api:libjingle_logging_api",
>+    "../rtc_base:rtc_base_approved",
>+    "//third_party/abseil-cpp/absl/memory",
>+  ]
>+}
>+
> if (rtc_include_tests) {
>   rtc_source_set("mocks") {
>     testonly = true
>     sources = [
>+      "rtc_event_log/mock/mock_rtc_event_log.cc",
>       "rtc_event_log/mock/mock_rtc_event_log.h",
>     ]
>     deps = [
>       ":rtc_event_log_api",
>-      "../../test:test_support",
>+      "../test:test_support",
>     ]
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/OWNERS
>index 492fb7ab32c..96b781d87d9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/OWNERS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/OWNERS
>@@ -1,3 +1,2 @@
>-skvlad@webrtc.org
> stefan@webrtc.org
> terelius@webrtc.org
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/DEPS b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/DEPS
>index ec93428e809..d09e30eace6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/DEPS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/DEPS
>@@ -3,5 +3,4 @@ include_rules = [
>   "+modules/audio_coding/audio_network_adaptor",
>   "+modules/remote_bitrate_estimator/include",
>   "+modules/rtp_rtcp",
>-  "+system_wrappers",
> ]
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder.h
>index 8f509ce896e..beb711c9dd8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder.h
>@@ -25,8 +25,6 @@ class RtcEventLogEncoder {
>   virtual std::string EncodeLogStart(int64_t timestamp_us) = 0;
>   virtual std::string EncodeLogEnd(int64_t timestamp_us) = 0;
> 
>-  virtual std::string Encode(const RtcEvent& event) = 0;
>-
>   virtual std::string EncodeBatch(
>       std::deque<std::unique_ptr<RtcEvent>>::const_iterator begin,
>       std::deque<std::unique_ptr<RtcEvent>>::const_iterator end) = 0;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc
>index c05f7717c85..7ef8d677c60 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.cc
>@@ -17,6 +17,8 @@
> #include "logging/rtc_event_log/events/rtc_event_audio_send_stream_config.h"
> #include "logging/rtc_event_log/events/rtc_event_bwe_update_delay_based.h"
> #include "logging/rtc_event_log/events/rtc_event_bwe_update_loss_based.h"
>+#include "logging/rtc_event_log/events/rtc_event_ice_candidate_pair.h"
>+#include "logging/rtc_event_log/events/rtc_event_ice_candidate_pair_config.h"
> #include "logging/rtc_event_log/events/rtc_event_probe_cluster_created.h"
> #include "logging/rtc_event_log/events/rtc_event_probe_result_failure.h"
> #include "logging/rtc_event_log/events/rtc_event_probe_result_success.h"
>@@ -103,8 +105,123 @@ rtclog::VideoReceiveConfig_RtcpMode ConvertRtcpMode(RtcpMode rtcp_mode) {
>   RTC_NOTREACHED();
>   return rtclog::VideoReceiveConfig::RTCP_COMPOUND;
> }
>-}  // namespace
> 
>+rtclog::IceCandidatePairConfig::IceCandidatePairConfigType
>+ConvertIceCandidatePairConfigType(IceCandidatePairConfigType type) {
>+  switch (type) {
>+    case IceCandidatePairConfigType::kAdded:
>+      return rtclog::IceCandidatePairConfig::ADDED;
>+    case IceCandidatePairConfigType::kUpdated:
>+      return rtclog::IceCandidatePairConfig::UPDATED;
>+    case IceCandidatePairConfigType::kDestroyed:
>+      return rtclog::IceCandidatePairConfig::DESTROYED;
>+    case IceCandidatePairConfigType::kSelected:
>+      return rtclog::IceCandidatePairConfig::SELECTED;
>+    case IceCandidatePairConfigType::kNumValues:
>+      RTC_NOTREACHED();
>+  }
>+  RTC_NOTREACHED();
>+  return rtclog::IceCandidatePairConfig::ADDED;
>+}
>+
>+rtclog::IceCandidatePairConfig::IceCandidateType ConvertIceCandidateType(
>+    IceCandidateType type) {
>+  switch (type) {
>+    case IceCandidateType::kUnknown:
>+      return rtclog::IceCandidatePairConfig::UNKNOWN_CANDIDATE_TYPE;
>+    case IceCandidateType::kLocal:
>+      return rtclog::IceCandidatePairConfig::LOCAL;
>+    case IceCandidateType::kStun:
>+      return rtclog::IceCandidatePairConfig::STUN;
>+    case IceCandidateType::kPrflx:
>+      return rtclog::IceCandidatePairConfig::PRFLX;
>+    case IceCandidateType::kRelay:
>+      return rtclog::IceCandidatePairConfig::RELAY;
>+    case IceCandidateType::kNumValues:
>+      RTC_NOTREACHED();
>+  }
>+  RTC_NOTREACHED();
>+  return rtclog::IceCandidatePairConfig::UNKNOWN_CANDIDATE_TYPE;
>+}
>+
>+rtclog::IceCandidatePairConfig::Protocol ConvertIceCandidatePairProtocol(
>+    IceCandidatePairProtocol protocol) {
>+  switch (protocol) {
>+    case IceCandidatePairProtocol::kUnknown:
>+      return rtclog::IceCandidatePairConfig::UNKNOWN_PROTOCOL;
>+    case IceCandidatePairProtocol::kUdp:
>+      return rtclog::IceCandidatePairConfig::UDP;
>+    case IceCandidatePairProtocol::kTcp:
>+      return rtclog::IceCandidatePairConfig::TCP;
>+    case IceCandidatePairProtocol::kSsltcp:
>+      return rtclog::IceCandidatePairConfig::SSLTCP;
>+    case IceCandidatePairProtocol::kTls:
>+      return rtclog::IceCandidatePairConfig::TLS;
>+    case IceCandidatePairProtocol::kNumValues:
>+      RTC_NOTREACHED();
>+  }
>+  RTC_NOTREACHED();
>+  return rtclog::IceCandidatePairConfig::UNKNOWN_PROTOCOL;
>+}
>+
>+rtclog::IceCandidatePairConfig::AddressFamily
>+ConvertIceCandidatePairAddressFamily(
>+    IceCandidatePairAddressFamily address_family) {
>+  switch (address_family) {
>+    case IceCandidatePairAddressFamily::kUnknown:
>+      return rtclog::IceCandidatePairConfig::UNKNOWN_ADDRESS_FAMILY;
>+    case IceCandidatePairAddressFamily::kIpv4:
>+      return rtclog::IceCandidatePairConfig::IPV4;
>+    case IceCandidatePairAddressFamily::kIpv6:
>+      return rtclog::IceCandidatePairConfig::IPV6;
>+    case IceCandidatePairAddressFamily::kNumValues:
>+      RTC_NOTREACHED();
>+  }
>+  RTC_NOTREACHED();
>+  return rtclog::IceCandidatePairConfig::UNKNOWN_ADDRESS_FAMILY;
>+}
>+
>+rtclog::IceCandidatePairConfig::NetworkType ConvertIceCandidateNetworkType(
>+    IceCandidateNetworkType network_type) {
>+  switch (network_type) {
>+    case IceCandidateNetworkType::kUnknown:
>+      return rtclog::IceCandidatePairConfig::UNKNOWN_NETWORK_TYPE;
>+    case IceCandidateNetworkType::kEthernet:
>+      return rtclog::IceCandidatePairConfig::ETHERNET;
>+    case IceCandidateNetworkType::kLoopback:
>+      return rtclog::IceCandidatePairConfig::LOOPBACK;
>+    case IceCandidateNetworkType::kWifi:
>+      return rtclog::IceCandidatePairConfig::WIFI;
>+    case IceCandidateNetworkType::kVpn:
>+      return rtclog::IceCandidatePairConfig::VPN;
>+    case IceCandidateNetworkType::kCellular:
>+      return rtclog::IceCandidatePairConfig::CELLULAR;
>+    case IceCandidateNetworkType::kNumValues:
>+      RTC_NOTREACHED();
>+  }
>+  RTC_NOTREACHED();
>+  return rtclog::IceCandidatePairConfig::UNKNOWN_NETWORK_TYPE;
>+}
>+
>+rtclog::IceCandidatePairEvent::IceCandidatePairEventType
>+ConvertIceCandidatePairEventType(IceCandidatePairEventType type) {
>+  switch (type) {
>+    case IceCandidatePairEventType::kCheckSent:
>+      return rtclog::IceCandidatePairEvent::CHECK_SENT;
>+    case IceCandidatePairEventType::kCheckReceived:
>+      return rtclog::IceCandidatePairEvent::CHECK_RECEIVED;
>+    case IceCandidatePairEventType::kCheckResponseSent:
>+      return rtclog::IceCandidatePairEvent::CHECK_RESPONSE_SENT;
>+    case IceCandidatePairEventType::kCheckResponseReceived:
>+      return rtclog::IceCandidatePairEvent::CHECK_RESPONSE_RECEIVED;
>+    case IceCandidatePairEventType::kNumValues:
>+      RTC_NOTREACHED();
>+  }
>+  RTC_NOTREACHED();
>+  return rtclog::IceCandidatePairEvent::CHECK_SENT;
>+}
>+
>+}  // namespace
> 
> std::string RtcEventLogEncoderLegacy::EncodeLogStart(int64_t timestamp_us) {
>   rtclog::Event rtclog_event;
>@@ -127,6 +244,7 @@ std::string RtcEventLogEncoderLegacy::EncodeBatch(
>   for (auto it = begin; it != end; ++it) {
>     // TODO(terelius): Can we avoid the slight inefficiency of reallocating the
>     // string?
>+    RTC_CHECK(it->get() != nullptr);
>     encoded_output += Encode(**it);
>   }
>   return encoded_output;
>@@ -172,6 +290,17 @@ std::string RtcEventLogEncoderLegacy::Encode(const RtcEvent& event) {
>       return EncodeBweUpdateLossBased(rtc_event);
>     }
> 
>+    case RtcEvent::Type::IceCandidatePairConfig: {
>+      auto& rtc_event =
>+          static_cast<const RtcEventIceCandidatePairConfig&>(event);
>+      return EncodeIceCandidatePairConfig(rtc_event);
>+    }
>+
>+    case RtcEvent::Type::IceCandidatePairEvent: {
>+      auto& rtc_event = static_cast<const RtcEventIceCandidatePair&>(event);
>+      return EncodeIceCandidatePairEvent(rtc_event);
>+    }
>+
>     case RtcEvent::Type::ProbeClusterCreated: {
>       auto& rtc_event = static_cast<const RtcEventProbeClusterCreated&>(event);
>       return EncodeProbeClusterCreated(rtc_event);
>@@ -231,7 +360,7 @@ std::string RtcEventLogEncoderLegacy::EncodeAlrState(
>   rtclog_event.set_timestamp_us(event.timestamp_us_);
>   rtclog_event.set_type(rtclog::Event::ALR_STATE_EVENT);
> 
>-  auto alr_state = rtclog_event.mutable_alr_state();
>+  auto* alr_state = rtclog_event.mutable_alr_state();
>   alr_state->set_in_alr(event.in_alr_);
>   return Serialize(&rtclog_event);
> }
>@@ -242,7 +371,7 @@ std::string RtcEventLogEncoderLegacy::EncodeAudioNetworkAdaptation(
>   rtclog_event.set_timestamp_us(event.timestamp_us_);
>   rtclog_event.set_type(rtclog::Event::AUDIO_NETWORK_ADAPTATION_EVENT);
> 
>-  auto audio_network_adaptation =
>+  auto* audio_network_adaptation =
>       rtclog_event.mutable_audio_network_adaptation();
>   if (event.config_->bitrate_bps)
>     audio_network_adaptation->set_bitrate_bps(*event.config_->bitrate_bps);
>@@ -269,7 +398,7 @@ std::string RtcEventLogEncoderLegacy::EncodeAudioPlayout(
>   rtclog_event.set_timestamp_us(event.timestamp_us_);
>   rtclog_event.set_type(rtclog::Event::AUDIO_PLAYOUT_EVENT);
> 
>-  auto playout_event = rtclog_event.mutable_audio_playout_event();
>+  auto* playout_event = rtclog_event.mutable_audio_playout_event();
>   playout_event->set_local_ssrc(event.ssrc_);
> 
>   return Serialize(&rtclog_event);
>@@ -323,7 +452,7 @@ std::string RtcEventLogEncoderLegacy::EncodeBweUpdateDelayBased(
>   rtclog_event.set_timestamp_us(event.timestamp_us_);
>   rtclog_event.set_type(rtclog::Event::DELAY_BASED_BWE_UPDATE);
> 
>-  auto bwe_event = rtclog_event.mutable_delay_based_bwe_update();
>+  auto* bwe_event = rtclog_event.mutable_delay_based_bwe_update();
>   bwe_event->set_bitrate_bps(event.bitrate_bps_);
>   bwe_event->set_detector_state(ConvertDetectorState(event.detector_state_));
> 
>@@ -336,7 +465,7 @@ std::string RtcEventLogEncoderLegacy::EncodeBweUpdateLossBased(
>   rtclog_event.set_timestamp_us(event.timestamp_us_);
>   rtclog_event.set_type(rtclog::Event::LOSS_BASED_BWE_UPDATE);
> 
>-  auto bwe_event = rtclog_event.mutable_loss_based_bwe_update();
>+  auto* bwe_event = rtclog_event.mutable_loss_based_bwe_update();
>   bwe_event->set_bitrate_bps(event.bitrate_bps_);
>   bwe_event->set_fraction_loss(event.fraction_loss_);
>   bwe_event->set_total_packets(event.total_packets_);
>@@ -344,13 +473,56 @@ std::string RtcEventLogEncoderLegacy::EncodeBweUpdateLossBased(
>   return Serialize(&rtclog_event);
> }
> 
>+std::string RtcEventLogEncoderLegacy::EncodeIceCandidatePairConfig(
>+    const RtcEventIceCandidatePairConfig& event) {
>+  rtclog::Event encoded_rtc_event;
>+  encoded_rtc_event.set_timestamp_us(event.timestamp_us_);
>+  encoded_rtc_event.set_type(rtclog::Event::ICE_CANDIDATE_PAIR_CONFIG);
>+
>+  auto* encoded_ice_event =
>+      encoded_rtc_event.mutable_ice_candidate_pair_config();
>+  encoded_ice_event->set_config_type(
>+      ConvertIceCandidatePairConfigType(event.type_));
>+  encoded_ice_event->set_candidate_pair_id(event.candidate_pair_id_);
>+  const auto& desc = event.candidate_pair_desc_;
>+  encoded_ice_event->set_local_candidate_type(
>+      ConvertIceCandidateType(desc.local_candidate_type));
>+  encoded_ice_event->set_local_relay_protocol(
>+      ConvertIceCandidatePairProtocol(desc.local_relay_protocol));
>+  encoded_ice_event->set_local_network_type(
>+      ConvertIceCandidateNetworkType(desc.local_network_type));
>+  encoded_ice_event->set_local_address_family(
>+      ConvertIceCandidatePairAddressFamily(desc.local_address_family));
>+  encoded_ice_event->set_remote_candidate_type(
>+      ConvertIceCandidateType(desc.remote_candidate_type));
>+  encoded_ice_event->set_remote_address_family(
>+      ConvertIceCandidatePairAddressFamily(desc.remote_address_family));
>+  encoded_ice_event->set_candidate_pair_protocol(
>+      ConvertIceCandidatePairProtocol(desc.candidate_pair_protocol));
>+  return Serialize(&encoded_rtc_event);
>+}
>+
>+std::string RtcEventLogEncoderLegacy::EncodeIceCandidatePairEvent(
>+    const RtcEventIceCandidatePair& event) {
>+  rtclog::Event encoded_rtc_event;
>+  encoded_rtc_event.set_timestamp_us(event.timestamp_us_);
>+  encoded_rtc_event.set_type(rtclog::Event::ICE_CANDIDATE_PAIR_EVENT);
>+
>+  auto* encoded_ice_event =
>+      encoded_rtc_event.mutable_ice_candidate_pair_event();
>+  encoded_ice_event->set_event_type(
>+      ConvertIceCandidatePairEventType(event.type_));
>+  encoded_ice_event->set_candidate_pair_id(event.candidate_pair_id_);
>+  return Serialize(&encoded_rtc_event);
>+}
>+
> std::string RtcEventLogEncoderLegacy::EncodeProbeClusterCreated(
>     const RtcEventProbeClusterCreated& event) {
>   rtclog::Event rtclog_event;
>   rtclog_event.set_timestamp_us(event.timestamp_us_);
>   rtclog_event.set_type(rtclog::Event::BWE_PROBE_CLUSTER_CREATED_EVENT);
> 
>-  auto probe_cluster = rtclog_event.mutable_probe_cluster();
>+  auto* probe_cluster = rtclog_event.mutable_probe_cluster();
>   probe_cluster->set_id(event.id_);
>   probe_cluster->set_bitrate_bps(event.bitrate_bps_);
>   probe_cluster->set_min_packets(event.min_probes_);
>@@ -365,7 +537,7 @@ std::string RtcEventLogEncoderLegacy::EncodeProbeResultFailure(
>   rtclog_event.set_timestamp_us(event.timestamp_us_);
>   rtclog_event.set_type(rtclog::Event::BWE_PROBE_RESULT_EVENT);
> 
>-  auto probe_result = rtclog_event.mutable_probe_result();
>+  auto* probe_result = rtclog_event.mutable_probe_result();
>   probe_result->set_id(event.id_);
>   probe_result->set_result(ConvertProbeResultType(event.failure_reason_));
> 
>@@ -378,7 +550,7 @@ std::string RtcEventLogEncoderLegacy::EncodeProbeResultSuccess(
>   rtclog_event.set_timestamp_us(event.timestamp_us_);
>   rtclog_event.set_type(rtclog::Event::BWE_PROBE_RESULT_EVENT);
> 
>-  auto probe_result = rtclog_event.mutable_probe_result();
>+  auto* probe_result = rtclog_event.mutable_probe_result();
>   probe_result->set_id(event.id_);
>   probe_result->set_result(rtclog::BweProbeResult::SUCCESS);
>   probe_result->set_bitrate_bps(event.bitrate_bps_);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.h
>index a8173135aef..87db039e3af 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.h
>@@ -32,6 +32,8 @@ class RtcEventAudioReceiveStreamConfig;
> class RtcEventAudioSendStreamConfig;
> class RtcEventBweUpdateDelayBased;
> class RtcEventBweUpdateLossBased;
>+class RtcEventIceCandidatePairConfig;
>+class RtcEventIceCandidatePair;
> class RtcEventLoggingStarted;
> class RtcEventLoggingStopped;
> class RtcEventProbeClusterCreated;
>@@ -50,8 +52,6 @@ class RtcEventLogEncoderLegacy final : public RtcEventLogEncoder {
>  public:
>   ~RtcEventLogEncoderLegacy() override = default;
> 
>-  std::string Encode(const RtcEvent& event) override;
>-
>   std::string EncodeLogStart(int64_t timestamp_us) override;
>   std::string EncodeLogEnd(int64_t timestamp_us) override;
> 
>@@ -60,6 +60,7 @@ class RtcEventLogEncoderLegacy final : public RtcEventLogEncoder {
>       std::deque<std::unique_ptr<RtcEvent>>::const_iterator end) override;
> 
>  private:
>+  std::string Encode(const RtcEvent& event);
>   // Encoding entry-point for the various RtcEvent subclasses.
>   std::string EncodeAlrState(const RtcEventAlrState& event);
>   std::string EncodeAudioNetworkAdaptation(
>@@ -72,6 +73,10 @@ class RtcEventLogEncoderLegacy final : public RtcEventLogEncoder {
>   std::string EncodeBweUpdateDelayBased(
>       const RtcEventBweUpdateDelayBased& event);
>   std::string EncodeBweUpdateLossBased(const RtcEventBweUpdateLossBased& event);
>+  std::string EncodeIceCandidatePairConfig(
>+      const RtcEventIceCandidatePairConfig& event);
>+  std::string EncodeIceCandidatePairEvent(
>+      const RtcEventIceCandidatePair& event);
>   std::string EncodeProbeClusterCreated(
>       const RtcEventProbeClusterCreated& event);
>   std::string EncodeProbeResultFailure(const RtcEventProbeResultFailure& event);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_unittest.cc
>index 00dad29d0f1..990fa6080bb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/encoder/rtc_event_log_encoder_unittest.cc
>@@ -8,14 +8,13 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include <cmath>
>+#include <deque>
> #include <limits>
>-#include <memory>
> #include <string>
>-#include <utility>
> 
>-#include "api/rtpparameters.h"  // RtpExtension
>+#include "absl/memory/memory.h"
> #include "logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.h"
>+#include "logging/rtc_event_log/events/rtc_event_alr_state.h"
> #include "logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h"
> #include "logging/rtc_event_log/events/rtc_event_audio_playout.h"
> #include "logging/rtc_event_log/events/rtc_event_audio_receive_stream_config.h"
>@@ -31,51 +30,24 @@
> #include "logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.h"
> #include "logging/rtc_event_log/events/rtc_event_video_receive_stream_config.h"
> #include "logging/rtc_event_log/events/rtc_event_video_send_stream_config.h"
>-#include "logging/rtc_event_log/rtc_event_log_parser.h"
>+#include "logging/rtc_event_log/rtc_event_log_parser_new.h"
>+#include "logging/rtc_event_log/rtc_event_log_unittest_helper.h"
> #include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
> #include "modules/remote_bitrate_estimator/include/bwe_defines.h"
>-#include "modules/rtp_rtcp/source/rtcp_packet/bye.h"  // Arbitrary RTCP message.
> #include "modules/rtp_rtcp/source/rtp_header_extensions.h"
>-#include "modules/rtp_rtcp/source/rtp_packet_received.h"
>-#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
>-#include "rtc_base/arraysize.h"
>-#include "rtc_base/numerics/safe_conversions.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/random.h"
> #include "test/gtest.h"
> 
> namespace webrtc {
> 
>-namespace {
>-struct ExtensionInfo {
>-  RTPExtensionType type;
>-  const char* uri;
>-};
>-
>-template <typename Extension>
>-constexpr ExtensionInfo CreateExtensionInfo() {
>-  return {Extension::kId, Extension::kUri};
>-}
>-
>-constexpr ExtensionInfo kExtensions[] = {
>-    CreateExtensionInfo<TransmissionOffset>(),
>-    CreateExtensionInfo<AudioLevel>(),
>-    CreateExtensionInfo<AbsoluteSendTime>(),
>-    CreateExtensionInfo<VideoOrientation>(),
>-    CreateExtensionInfo<TransportSequenceNumber>(),
>-    CreateExtensionInfo<PlayoutDelayLimits>(),
>-    CreateExtensionInfo<VideoContentTypeExtension>(),
>-    CreateExtensionInfo<VideoTimingExtension>(),
>-    CreateExtensionInfo<RtpStreamId>(),
>-    CreateExtensionInfo<RepairedRtpStreamId>(),
>-    CreateExtensionInfo<RtpMid>(),
>-};
>-}  // namespace
> 
> class RtcEventLogEncoderTest : public testing::TestWithParam<int> {
>  protected:
>   RtcEventLogEncoderTest()
>-      : encoder_(new RtcEventLogEncoderLegacy), prng_(GetParam()) {}
>+      : encoder_(new RtcEventLogEncoderLegacy),
>+        seed_(GetParam()),
>+        prng_(seed_),
>+        gen_(seed_ * 880001UL) {}
>   ~RtcEventLogEncoderTest() override = default;
> 
>   // ANA events have some optional fields, so we want to make sure that we get
>@@ -84,74 +56,57 @@ class RtcEventLogEncoderTest : public testing::TestWithParam<int> {
>   void TestRtcEventAudioNetworkAdaptation(
>       std::unique_ptr<AudioEncoderRuntimeConfig> runtime_config);
> 
>-  // These help prevent code duplication between incoming/outgoing variants.
>-  void TestRtcEventRtcpPacket(PacketDirection direction);
>-  void TestRtcEventRtpPacket(PacketDirection direction);
>-
>-  int RandomInt() {
>-    // Don't run this on a SNES.
>-    static_assert(8 * sizeof(int) >= 32, "Don't run this on a SNES.");
>-    int32_t rand = prng_.Rand(0, std::numeric_limits<int32_t>::max());
>-    return rtc::saturated_cast<int>(rand);
>-  }
>-
>-  int RandomPositiveInt() {
>-    int32_t rand = prng_.Rand(1, std::numeric_limits<int32_t>::max());
>-    return rtc::saturated_cast<int>(rand);
>-  }
>-
>-  uint32_t RandomSsrc() {
>-    return prng_.Rand(std::numeric_limits<uint32_t>::max());
>-  }
>-
>-  std::vector<RtpExtension> RandomRtpExtensions() {
>-    RTC_DCHECK(arraysize(kExtensions) >= 2);
>-    size_t id_1 = prng_.Rand(0u, arraysize(kExtensions) - 1);
>-    size_t id_2 = prng_.Rand(0u, arraysize(kExtensions) - 2);
>-    if (id_2 == id_1)
>-      id_2 = arraysize(kExtensions) - 1;
>-    return std::vector<RtpExtension>{
>-        RtpExtension(kExtensions[id_1].uri, kExtensions[id_1].type),
>-        RtpExtension(kExtensions[id_2].uri, kExtensions[id_2].type)};
>-  }
>-
>-  int RandomBitrate() { return RandomInt(); }
>-
>-  // TODO(eladalon): Once we have more than once possible encoder, parameterize
>+  std::deque<std::unique_ptr<RtcEvent>> history_;
>+  // TODO(eladalon): Once we have more than one possible encoder, parameterize
>   // encoder selection.
>   std::unique_ptr<RtcEventLogEncoder> encoder_;
>-  ParsedRtcEventLog parsed_log_;
>+  ParsedRtcEventLogNew parsed_log_;
>+  const uint64_t seed_;
>   Random prng_;
>+  test::EventGenerator gen_;
> };
> 
>+TEST_P(RtcEventLogEncoderTest, RtcEventAlrState) {
>+  std::unique_ptr<RtcEventAlrState> event = gen_.NewAlrState();
>+  history_.push_back(event->Copy());
>+
>+  std::string encoded = encoder_->EncodeBatch(history_.begin(), history_.end());
>+  ASSERT_TRUE(parsed_log_.ParseString(encoded));
>+  const auto& alr_state_events = parsed_log_.alr_state_events();
>+
>+  ASSERT_EQ(alr_state_events.size(), 1u);
>+  EXPECT_TRUE(test::VerifyLoggedAlrStateEvent(*event, alr_state_events[0]));
>+}
>+
> void RtcEventLogEncoderTest::TestRtcEventAudioNetworkAdaptation(
>     std::unique_ptr<AudioEncoderRuntimeConfig> runtime_config) {
>+  // This function is called repeatedly. Clear state between calls.
>+  history_.clear();
>   auto original_runtime_config = *runtime_config;
>-  auto event = rtc::MakeUnique<RtcEventAudioNetworkAdaptation>(
>+  auto event = absl::make_unique<RtcEventAudioNetworkAdaptation>(
>       std::move(runtime_config));
>   const int64_t timestamp_us = event->timestamp_us_;
>+  history_.push_back(std::move(event));
> 
>-  ASSERT_TRUE(parsed_log_.ParseString(encoder_->Encode(*event)));
>-  ASSERT_EQ(parsed_log_.GetNumberOfEvents(), 1u);
>-  ASSERT_EQ(parsed_log_.GetEventType(0),
>-            ParsedRtcEventLog::AUDIO_NETWORK_ADAPTATION_EVENT);
>-
>-  AudioEncoderRuntimeConfig parsed_runtime_config;
>-  parsed_log_.GetAudioNetworkAdaptation(0, &parsed_runtime_config);
>+  std::string encoded = encoder_->EncodeBatch(history_.begin(), history_.end());
>+  ASSERT_TRUE(parsed_log_.ParseString(encoded));
>+  const auto& ana_configs = parsed_log_.audio_network_adaptation_events();
>+  ASSERT_EQ(ana_configs.size(), 1u);
> 
>-  EXPECT_EQ(parsed_log_.GetTimestamp(0), timestamp_us);
>-  EXPECT_EQ(parsed_runtime_config, original_runtime_config);
>+  EXPECT_EQ(ana_configs[0].timestamp_us, timestamp_us);
>+  EXPECT_EQ(ana_configs[0].config, original_runtime_config);
> }
> 
> TEST_P(RtcEventLogEncoderTest, RtcEventAudioNetworkAdaptationBitrate) {
>-  auto runtime_config = rtc::MakeUnique<AudioEncoderRuntimeConfig>();
>-  const int bitrate_bps = RandomBitrate();
>+  auto runtime_config = absl::make_unique<AudioEncoderRuntimeConfig>();
>+  const int bitrate_bps = rtc::checked_cast<int>(
>+      prng_.Rand(0, std::numeric_limits<int32_t>::max()));
>   runtime_config->bitrate_bps = bitrate_bps;
>   TestRtcEventAudioNetworkAdaptation(std::move(runtime_config));
> }
> 
> TEST_P(RtcEventLogEncoderTest, RtcEventAudioNetworkAdaptationFrameLength) {
>-  auto runtime_config = rtc::MakeUnique<AudioEncoderRuntimeConfig>();
>+  auto runtime_config = absl::make_unique<AudioEncoderRuntimeConfig>();
>   const int frame_length_ms = prng_.Rand(1, 1000);
>   runtime_config->frame_length_ms = frame_length_ms;
>   TestRtcEventAudioNetworkAdaptation(std::move(runtime_config));
>@@ -160,7 +115,7 @@ TEST_P(RtcEventLogEncoderTest, RtcEventAudioNetworkAdaptationFrameLength) {
> TEST_P(RtcEventLogEncoderTest, RtcEventAudioNetworkAdaptationPacketLoss) {
>   // To simplify the test, we just check powers of two.
>   const float plr = std::pow(0.5f, prng_.Rand(1, 8));
>-  auto runtime_config = rtc::MakeUnique<AudioEncoderRuntimeConfig>();
>+  auto runtime_config = absl::make_unique<AudioEncoderRuntimeConfig>();
>   runtime_config->uplink_packet_loss_fraction = plr;
>   TestRtcEventAudioNetworkAdaptation(std::move(runtime_config));
> }
>@@ -169,7 +124,7 @@ TEST_P(RtcEventLogEncoderTest, RtcEventAudioNetworkAdaptationFec) {
>   // The test might be trivially passing for one of the two boolean values, so
>   // for safety's sake, we test both.
>   for (bool fec_enabled : {false, true}) {
>-    auto runtime_config = rtc::MakeUnique<AudioEncoderRuntimeConfig>();
>+    auto runtime_config = absl::make_unique<AudioEncoderRuntimeConfig>();
>     runtime_config->enable_fec = fec_enabled;
>     TestRtcEventAudioNetworkAdaptation(std::move(runtime_config));
>   }
>@@ -179,7 +134,7 @@ TEST_P(RtcEventLogEncoderTest, RtcEventAudioNetworkAdaptationDtx) {
>   // The test might be trivially passing for one of the two boolean values, so
>   // for safety's sake, we test both.
>   for (bool dtx_enabled : {false, true}) {
>-    auto runtime_config = rtc::MakeUnique<AudioEncoderRuntimeConfig>();
>+    auto runtime_config = absl::make_unique<AudioEncoderRuntimeConfig>();
>     runtime_config->enable_dtx = dtx_enabled;
>     TestRtcEventAudioNetworkAdaptation(std::move(runtime_config));
>   }
>@@ -189,20 +144,21 @@ TEST_P(RtcEventLogEncoderTest, RtcEventAudioNetworkAdaptationChannels) {
>   // The test might be trivially passing for one of the two possible values, so
>   // for safety's sake, we test both.
>   for (size_t channels : {1, 2}) {
>-    auto runtime_config = rtc::MakeUnique<AudioEncoderRuntimeConfig>();
>+    auto runtime_config = absl::make_unique<AudioEncoderRuntimeConfig>();
>     runtime_config->num_channels = channels;
>     TestRtcEventAudioNetworkAdaptation(std::move(runtime_config));
>   }
> }
> 
> TEST_P(RtcEventLogEncoderTest, RtcEventAudioNetworkAdaptationAll) {
>-  const int bitrate_bps = RandomBitrate();
>+  const int bitrate_bps = rtc::checked_cast<int>(
>+      prng_.Rand(0, std::numeric_limits<int32_t>::max()));
>   const int frame_length_ms = prng_.Rand(1, 1000);
>   const float plr = std::pow(0.5f, prng_.Rand(1, 8));
>   for (bool fec_enabled : {false, true}) {
>     for (bool dtx_enabled : {false, true}) {
>       for (size_t channels : {1, 2}) {
>-        auto runtime_config = rtc::MakeUnique<AudioEncoderRuntimeConfig>();
>+        auto runtime_config = absl::make_unique<AudioEncoderRuntimeConfig>();
>         runtime_config->bitrate_bps = bitrate_bps;
>         runtime_config->frame_length_ms = frame_length_ms;
>         runtime_config->uplink_packet_loss_fraction = plr;
>@@ -217,350 +173,268 @@ TEST_P(RtcEventLogEncoderTest, RtcEventAudioNetworkAdaptationAll) {
> }
> 
> TEST_P(RtcEventLogEncoderTest, RtcEventAudioPlayout) {
>-  const uint32_t ssrc = RandomSsrc();
>-  auto event = rtc::MakeUnique<RtcEventAudioPlayout>(ssrc);
>-  const int64_t timestamp_us = event->timestamp_us_;
>-
>-  ASSERT_TRUE(parsed_log_.ParseString(encoder_->Encode(*event)));
>-  ASSERT_EQ(parsed_log_.GetNumberOfEvents(), 1u);
>-  ASSERT_EQ(parsed_log_.GetEventType(0),
>-            ParsedRtcEventLog::AUDIO_PLAYOUT_EVENT);
>-
>-  uint32_t parsed_ssrc;
>-  parsed_log_.GetAudioPlayout(0, &parsed_ssrc);
>-
>-  EXPECT_EQ(parsed_log_.GetTimestamp(0), timestamp_us);
>-  EXPECT_EQ(parsed_ssrc, ssrc);
>+  uint32_t ssrc = prng_.Rand<uint32_t>();
>+  std::unique_ptr<RtcEventAudioPlayout> event = gen_.NewAudioPlayout(ssrc);
>+  history_.push_back(event->Copy());
>+
>+  std::string encoded = encoder_->EncodeBatch(history_.begin(), history_.end());
>+  ASSERT_TRUE(parsed_log_.ParseString(encoded));
>+
>+  const auto& playout_events = parsed_log_.audio_playout_events();
>+  ASSERT_EQ(playout_events.size(), 1u);
>+  const auto playout_stream = playout_events.find(ssrc);
>+  ASSERT_TRUE(playout_stream != playout_events.end());
>+  ASSERT_EQ(playout_stream->second.size(), 1u);
>+  LoggedAudioPlayoutEvent playout_event = playout_stream->second[0];
>+  EXPECT_TRUE(test::VerifyLoggedAudioPlayoutEvent(*event, playout_event));
> }
> 
> TEST_P(RtcEventLogEncoderTest, RtcEventAudioReceiveStreamConfig) {
>-  auto stream_config = rtc::MakeUnique<rtclog::StreamConfig>();
>-  stream_config->local_ssrc = RandomSsrc();
>-  stream_config->remote_ssrc = RandomSsrc();
>-  // TODO(eladalon): Verify that the extensions are used correctly when
>-  // parsing RTP packets headers. Here and elsewhere.
>-  std::vector<RtpExtension> extensions = RandomRtpExtensions();
>-  for (const auto& extension : extensions)
>-    stream_config->rtp_extensions.push_back(extension);
>-
>-  auto original_stream_config = *stream_config;
>-
>-  auto event = rtc::MakeUnique<RtcEventAudioReceiveStreamConfig>(
>-      std::move(stream_config));
>-  const int64_t timestamp_us = event->timestamp_us_;
>-
>-  ASSERT_TRUE(parsed_log_.ParseString(encoder_->Encode(*event)));
>-  ASSERT_EQ(parsed_log_.GetNumberOfEvents(), 1u);
>-  ASSERT_EQ(parsed_log_.GetEventType(0),
>-            ParsedRtcEventLog::AUDIO_RECEIVER_CONFIG_EVENT);
>-
>-  auto parsed_event = parsed_log_.GetAudioReceiveConfig(0);
>-  EXPECT_EQ(parsed_log_.GetTimestamp(0), timestamp_us);
>-  EXPECT_EQ(parsed_event, original_stream_config);
>+  uint32_t ssrc = prng_.Rand<uint32_t>();
>+  RtpHeaderExtensionMap extensions = gen_.NewRtpHeaderExtensionMap();
>+  std::unique_ptr<RtcEventAudioReceiveStreamConfig> event =
>+      gen_.NewAudioReceiveStreamConfig(ssrc, extensions);
>+  history_.push_back(event->Copy());
>+
>+  std::string encoded = encoder_->EncodeBatch(history_.begin(), history_.end());
>+  ASSERT_TRUE(parsed_log_.ParseString(encoded));
>+  const auto& audio_recv_configs = parsed_log_.audio_recv_configs();
>+
>+  ASSERT_EQ(audio_recv_configs.size(), 1u);
>+  EXPECT_TRUE(test::VerifyLoggedAudioRecvConfig(*event, audio_recv_configs[0]));
> }
> 
> TEST_P(RtcEventLogEncoderTest, RtcEventAudioSendStreamConfig) {
>-  auto stream_config = rtc::MakeUnique<rtclog::StreamConfig>();
>-  stream_config->local_ssrc = RandomSsrc();
>-  std::vector<RtpExtension> extensions = RandomRtpExtensions();
>-  for (const auto& extension : extensions)
>-    stream_config->rtp_extensions.push_back(extension);
>-
>-  auto original_stream_config = *stream_config;
>+  uint32_t ssrc = prng_.Rand<uint32_t>();
>+  RtpHeaderExtensionMap extensions = gen_.NewRtpHeaderExtensionMap();
>+  std::unique_ptr<RtcEventAudioSendStreamConfig> event =
>+      gen_.NewAudioSendStreamConfig(ssrc, extensions);
>+  history_.push_back(event->Copy());
>+
>+  std::string encoded = encoder_->EncodeBatch(history_.begin(), history_.end());
>+  ASSERT_TRUE(parsed_log_.ParseString(encoded));
>+  const auto& audio_send_configs = parsed_log_.audio_send_configs();
>+
>+  ASSERT_EQ(audio_send_configs.size(), 1u);
>+  EXPECT_TRUE(test::VerifyLoggedAudioSendConfig(*event, audio_send_configs[0]));
>+}
> 
>-  auto event =
>-      rtc::MakeUnique<RtcEventAudioSendStreamConfig>(std::move(stream_config));
>-  const int64_t timestamp_us = event->timestamp_us_;
>+TEST_P(RtcEventLogEncoderTest, RtcEventBweUpdateDelayBased) {
>+  std::unique_ptr<RtcEventBweUpdateDelayBased> event =
>+      gen_.NewBweUpdateDelayBased();
>+  history_.push_back(event->Copy());
> 
>-  ASSERT_TRUE(parsed_log_.ParseString(encoder_->Encode(*event)));
>-  ASSERT_EQ(parsed_log_.GetNumberOfEvents(), 1u);
>-  ASSERT_EQ(parsed_log_.GetEventType(0),
>-            ParsedRtcEventLog::AUDIO_SENDER_CONFIG_EVENT);
>+  std::string encoded = encoder_->EncodeBatch(history_.begin(), history_.end());
>+  ASSERT_TRUE(parsed_log_.ParseString(encoded));
>+  const auto& bwe_delay_updates = parsed_log_.bwe_delay_updates();
> 
>-  auto parsed_event = parsed_log_.GetAudioSendConfig(0);
>-  EXPECT_EQ(parsed_log_.GetTimestamp(0), timestamp_us);
>-  EXPECT_EQ(parsed_event, original_stream_config);
>+  ASSERT_EQ(bwe_delay_updates.size(), 1u);
>+  EXPECT_TRUE(
>+      test::VerifyLoggedBweDelayBasedUpdate(*event, bwe_delay_updates[0]));
> }
> 
>-TEST_P(RtcEventLogEncoderTest, RtcEventBweUpdateDelayBased) {
>-  const int32_t bitrate_bps = RandomBitrate();
>-  const BandwidthUsage detector_state = static_cast<BandwidthUsage>(
>-      prng_.Rand(0, static_cast<int32_t>(BandwidthUsage::kLast) - 1));
>-  auto event =
>-      rtc::MakeUnique<RtcEventBweUpdateDelayBased>(bitrate_bps, detector_state);
>-  const int64_t timestamp_us = event->timestamp_us_;
>+TEST_P(RtcEventLogEncoderTest, RtcEventBweUpdateLossBased) {
>+  std::unique_ptr<RtcEventBweUpdateLossBased> event =
>+      gen_.NewBweUpdateLossBased();
>+  history_.push_back(event->Copy());
> 
>-  ASSERT_TRUE(parsed_log_.ParseString(encoder_->Encode(*event)));
>-  ASSERT_EQ(parsed_log_.GetNumberOfEvents(), 1u);
>-  ASSERT_EQ(parsed_log_.GetEventType(0),
>-            ParsedRtcEventLog::DELAY_BASED_BWE_UPDATE);
>+  std::string encoded = encoder_->EncodeBatch(history_.begin(), history_.end());
>+  ASSERT_TRUE(parsed_log_.ParseString(encoded));
>+  const auto& bwe_loss_updates = parsed_log_.bwe_loss_updates();
>+
>+  ASSERT_EQ(bwe_loss_updates.size(), 1u);
>+  EXPECT_TRUE(
>+      test::VerifyLoggedBweLossBasedUpdate(*event, bwe_loss_updates[0]));
>+}
> 
>-  auto parsed_event = parsed_log_.GetDelayBasedBweUpdate(0);
>+TEST_P(RtcEventLogEncoderTest, RtcEventIceCandidatePairConfig) {
>+  std::unique_ptr<RtcEventIceCandidatePairConfig> event =
>+      gen_.NewIceCandidatePairConfig();
>+  history_.push_back(event->Copy());
> 
>-  EXPECT_EQ(parsed_log_.GetTimestamp(0), timestamp_us);
>-  EXPECT_EQ(parsed_event.bitrate_bps, bitrate_bps);
>-  EXPECT_EQ(parsed_event.detector_state, detector_state);
>+  std::string encoded = encoder_->EncodeBatch(history_.begin(), history_.end());
>+  ASSERT_TRUE(parsed_log_.ParseString(encoded));
>+  const auto& ice_candidate_pair_configs =
>+      parsed_log_.ice_candidate_pair_configs();
>+
>+  ASSERT_EQ(ice_candidate_pair_configs.size(), 1u);
>+  EXPECT_TRUE(test::VerifyLoggedIceCandidatePairConfig(
>+      *event, ice_candidate_pair_configs[0]));
> }
> 
>-TEST_P(RtcEventLogEncoderTest, RtcEventBweUpdateLossBased) {
>-  const int32_t bitrate_bps = RandomBitrate();
>-  const uint8_t fraction_loss = rtc::dchecked_cast<uint8_t>(
>-      prng_.Rand(0, std::numeric_limits<uint8_t>::max()));
>-  const int32_t total_packets = RandomInt();
>+TEST_P(RtcEventLogEncoderTest, RtcEventIceCandidatePair) {
>+  std::unique_ptr<RtcEventIceCandidatePair> event = gen_.NewIceCandidatePair();
>+  history_.push_back(event->Copy());
> 
>-  auto event = rtc::MakeUnique<RtcEventBweUpdateLossBased>(
>-      bitrate_bps, fraction_loss, total_packets);
>-  const int64_t timestamp_us = event->timestamp_us_;
>+  std::string encoded = encoder_->EncodeBatch(history_.begin(), history_.end());
>+  ASSERT_TRUE(parsed_log_.ParseString(encoded));
>+  const auto& ice_candidate_pair_events =
>+      parsed_log_.ice_candidate_pair_events();
> 
>-  ASSERT_TRUE(parsed_log_.ParseString(encoder_->Encode(*event)));
>-  ASSERT_EQ(parsed_log_.GetNumberOfEvents(), 1u);
>-  ASSERT_EQ(parsed_log_.GetEventType(0),
>-            ParsedRtcEventLog::LOSS_BASED_BWE_UPDATE);
>-
>-  int32_t parsed_bitrate_bps;
>-  uint8_t parsed_fraction_loss;
>-  int32_t parsed_total_packets;
>-  parsed_log_.GetLossBasedBweUpdate(
>-      0, &parsed_bitrate_bps, &parsed_fraction_loss, &parsed_total_packets);
>-
>-  EXPECT_EQ(parsed_log_.GetTimestamp(0), timestamp_us);
>-  EXPECT_EQ(parsed_bitrate_bps, bitrate_bps);
>-  EXPECT_EQ(parsed_fraction_loss, fraction_loss);
>-  EXPECT_EQ(parsed_total_packets, total_packets);
>+  ASSERT_EQ(ice_candidate_pair_events.size(), 1u);
>+  EXPECT_TRUE(test::VerifyLoggedIceCandidatePairEvent(
>+      *event, ice_candidate_pair_events[0]));
> }
> 
> TEST_P(RtcEventLogEncoderTest, RtcEventLoggingStarted) {
>   const int64_t timestamp_us = rtc::TimeMicros();
> 
>   ASSERT_TRUE(parsed_log_.ParseString(encoder_->EncodeLogStart(timestamp_us)));
>-  ASSERT_EQ(parsed_log_.GetNumberOfEvents(), 1u);
>-  ASSERT_EQ(parsed_log_.GetEventType(0), ParsedRtcEventLog::LOG_START);
>+  const auto& start_log_events = parsed_log_.start_log_events();
> 
>-  EXPECT_EQ(parsed_log_.GetTimestamp(0), timestamp_us);
>+  ASSERT_EQ(start_log_events.size(), 1u);
>+  EXPECT_EQ(start_log_events[0].timestamp_us, timestamp_us);
> }
> 
> TEST_P(RtcEventLogEncoderTest, RtcEventLoggingStopped) {
>   const int64_t timestamp_us = rtc::TimeMicros();
> 
>   ASSERT_TRUE(parsed_log_.ParseString(encoder_->EncodeLogEnd(timestamp_us)));
>-  ASSERT_EQ(parsed_log_.GetNumberOfEvents(), 1u);
>-  ASSERT_EQ(parsed_log_.GetEventType(0), ParsedRtcEventLog::LOG_END);
>+  const auto& stop_log_events = parsed_log_.stop_log_events();
> 
>-  EXPECT_EQ(parsed_log_.GetTimestamp(0), timestamp_us);
>+  ASSERT_EQ(stop_log_events.size(), 1u);
>+  EXPECT_EQ(stop_log_events[0].timestamp_us, timestamp_us);
> }
> 
> TEST_P(RtcEventLogEncoderTest, RtcEventProbeClusterCreated) {
>-  const int id = RandomPositiveInt();
>-  const int bitrate_bps = RandomBitrate();
>-  const int min_probes = RandomPositiveInt();
>-  const int min_bytes = RandomPositiveInt();
>-
>-  auto event = rtc::MakeUnique<RtcEventProbeClusterCreated>(
>-      id, bitrate_bps, min_probes, min_bytes);
>-  const int64_t timestamp_us = event->timestamp_us_;
>-
>-  ASSERT_TRUE(parsed_log_.ParseString(encoder_->Encode(*event)));
>-  ASSERT_EQ(parsed_log_.GetNumberOfEvents(), 1u);
>-  ASSERT_EQ(parsed_log_.GetEventType(0),
>-            ParsedRtcEventLog::BWE_PROBE_CLUSTER_CREATED_EVENT);
>-
>-  auto parsed_event = parsed_log_.GetBweProbeClusterCreated(0);
>-
>-  EXPECT_EQ(parsed_log_.GetTimestamp(0), timestamp_us);
>-  EXPECT_EQ(rtc::dchecked_cast<int>(parsed_event.id), id);
>-  EXPECT_EQ(rtc::dchecked_cast<int>(parsed_event.bitrate_bps), bitrate_bps);
>-  EXPECT_EQ(rtc::dchecked_cast<int>(parsed_event.min_packets), min_probes);
>-  EXPECT_EQ(rtc::dchecked_cast<int>(parsed_event.min_bytes), min_bytes);
>+  std::unique_ptr<RtcEventProbeClusterCreated> event =
>+      gen_.NewProbeClusterCreated();
>+  history_.push_back(event->Copy());
>+
>+  std::string encoded = encoder_->EncodeBatch(history_.begin(), history_.end());
>+  ASSERT_TRUE(parsed_log_.ParseString(encoded));
>+  const auto& bwe_probe_cluster_created_events =
>+      parsed_log_.bwe_probe_cluster_created_events();
>+
>+  ASSERT_EQ(bwe_probe_cluster_created_events.size(), 1u);
>+  EXPECT_TRUE(test::VerifyLoggedBweProbeClusterCreatedEvent(
>+      *event, bwe_probe_cluster_created_events[0]));
> }
> 
> TEST_P(RtcEventLogEncoderTest, RtcEventProbeResultFailure) {
>-  const int id = RandomPositiveInt();
>-  const ProbeFailureReason failure_reason = static_cast<ProbeFailureReason>(
>-      prng_.Rand(0, static_cast<int32_t>(ProbeFailureReason::kLast) - 1));
>-
>-  auto event = rtc::MakeUnique<RtcEventProbeResultFailure>(id, failure_reason);
>-  const int64_t timestamp_us = event->timestamp_us_;
>-
>-  ASSERT_TRUE(parsed_log_.ParseString(encoder_->Encode(*event)));
>-  ASSERT_EQ(parsed_log_.GetNumberOfEvents(), 1u);
>-  ASSERT_EQ(parsed_log_.GetEventType(0),
>-            ParsedRtcEventLog::BWE_PROBE_RESULT_EVENT);
>+  std::unique_ptr<RtcEventProbeResultFailure> event =
>+      gen_.NewProbeResultFailure();
>+  history_.push_back(event->Copy());
> 
>-  auto parsed_event = parsed_log_.GetBweProbeResult(0);
>+  std::string encoded = encoder_->EncodeBatch(history_.begin(), history_.end());
>+  ASSERT_TRUE(parsed_log_.ParseString(encoded));
>+  const auto& bwe_probe_failure_events = parsed_log_.bwe_probe_failure_events();
> 
>-  EXPECT_EQ(parsed_log_.GetTimestamp(0), timestamp_us);
>-  EXPECT_EQ(rtc::dchecked_cast<int>(parsed_event.id), id);
>-  ASSERT_FALSE(parsed_event.bitrate_bps);
>-  ASSERT_TRUE(parsed_event.failure_reason);
>-  EXPECT_EQ(parsed_event.failure_reason, failure_reason);
>+  ASSERT_EQ(bwe_probe_failure_events.size(), 1u);
>+  EXPECT_TRUE(test::VerifyLoggedBweProbeFailureEvent(
>+      *event, bwe_probe_failure_events[0]));
> }
> 
> TEST_P(RtcEventLogEncoderTest, RtcEventProbeResultSuccess) {
>-  const int id = RandomPositiveInt();
>-  const int bitrate_bps = RandomBitrate();
>-
>-  auto event = rtc::MakeUnique<RtcEventProbeResultSuccess>(id, bitrate_bps);
>-  const int64_t timestamp_us = event->timestamp_us_;
>-
>-  ASSERT_TRUE(parsed_log_.ParseString(encoder_->Encode(*event)));
>-  ASSERT_EQ(parsed_log_.GetNumberOfEvents(), 1u);
>-  ASSERT_EQ(parsed_log_.GetEventType(0),
>-            ParsedRtcEventLog::BWE_PROBE_RESULT_EVENT);
>+  std::unique_ptr<RtcEventProbeResultSuccess> event =
>+      gen_.NewProbeResultSuccess();
>+  history_.push_back(event->Copy());
> 
>-  auto parsed_event = parsed_log_.GetBweProbeResult(0);
>+  std::string encoded = encoder_->EncodeBatch(history_.begin(), history_.end());
>+  ASSERT_TRUE(parsed_log_.ParseString(encoded));
>+  const auto& bwe_probe_success_events = parsed_log_.bwe_probe_success_events();
> 
>-  EXPECT_EQ(parsed_log_.GetTimestamp(0), timestamp_us);
>-  EXPECT_EQ(rtc::dchecked_cast<int>(parsed_event.id), id);
>-  ASSERT_TRUE(parsed_event.bitrate_bps);
>-  EXPECT_EQ(parsed_event.bitrate_bps, bitrate_bps);
>-  ASSERT_FALSE(parsed_event.failure_reason);
>+  ASSERT_EQ(bwe_probe_success_events.size(), 1u);
>+  EXPECT_TRUE(test::VerifyLoggedBweProbeSuccessEvent(
>+      *event, bwe_probe_success_events[0]));
> }
> 
>-void RtcEventLogEncoderTest::TestRtcEventRtcpPacket(PacketDirection direction) {
>-  rtcp::Bye bye_packet;  // Arbitrarily chosen RTCP packet type.
>-  bye_packet.SetReason("a man's reach should exceed his grasp");
>-  auto rtcp_packet = bye_packet.Build();
>-
>-  std::unique_ptr<RtcEvent> event;
>-  if (direction == PacketDirection::kIncomingPacket) {
>-    event = rtc::MakeUnique<RtcEventRtcpPacketIncoming>(rtcp_packet);
>-  } else {
>-    event = rtc::MakeUnique<RtcEventRtcpPacketOutgoing>(rtcp_packet);
>-  }
>-  const int64_t timestamp_us = event->timestamp_us_;
>-
>-  ASSERT_TRUE(parsed_log_.ParseString(encoder_->Encode(*event)));
>-  ASSERT_EQ(parsed_log_.GetNumberOfEvents(), 1u);
>-  ASSERT_EQ(parsed_log_.GetEventType(0), ParsedRtcEventLog::RTCP_EVENT);
>+TEST_P(RtcEventLogEncoderTest, RtcEventRtcpPacketIncoming) {
>+  std::unique_ptr<RtcEventRtcpPacketIncoming> event =
>+      gen_.NewRtcpPacketIncoming();
>+  history_.push_back(event->Copy());
> 
>-  PacketDirection parsed_direction;
>-  uint8_t parsed_packet[IP_PACKET_SIZE];  // "Parsed" = after event-encoding.
>-  size_t parsed_packet_length;
>-  parsed_log_.GetRtcpPacket(0, &parsed_direction, parsed_packet,
>-                            &parsed_packet_length);
>+  std::string encoded = encoder_->EncodeBatch(history_.begin(), history_.end());
>+  ASSERT_TRUE(parsed_log_.ParseString(encoded));
>+  const auto& incoming_rtcp_packets = parsed_log_.incoming_rtcp_packets();
> 
>-  EXPECT_EQ(parsed_direction, direction);
>-  EXPECT_EQ(parsed_log_.GetTimestamp(0), timestamp_us);
>-  ASSERT_EQ(parsed_packet_length, rtcp_packet.size());
>-  ASSERT_EQ(memcmp(parsed_packet, rtcp_packet.data(), parsed_packet_length), 0);
>-}
>-
>-TEST_P(RtcEventLogEncoderTest, RtcEventRtcpPacketIncoming) {
>-  TestRtcEventRtcpPacket(PacketDirection::kIncomingPacket);
>+  ASSERT_EQ(incoming_rtcp_packets.size(), 1u);
>+  EXPECT_TRUE(
>+      test::VerifyLoggedRtcpPacketIncoming(*event, incoming_rtcp_packets[0]));
> }
> 
> TEST_P(RtcEventLogEncoderTest, RtcEventRtcpPacketOutgoing) {
>-  TestRtcEventRtcpPacket(PacketDirection::kOutgoingPacket);
>-}
>+  std::unique_ptr<RtcEventRtcpPacketOutgoing> event =
>+      gen_.NewRtcpPacketOutgoing();
>+  history_.push_back(event->Copy());
> 
>-void RtcEventLogEncoderTest::TestRtcEventRtpPacket(PacketDirection direction) {
>-  const int probe_cluster_id = RandomPositiveInt();
>-
>-  std::unique_ptr<RtpPacketReceived> packet_received;
>-  std::unique_ptr<RtpPacketToSend> packet_to_send;
>-  RtpPacket* packet;
>-  if (direction == PacketDirection::kIncomingPacket) {
>-    packet_received = rtc::MakeUnique<RtpPacketReceived>();
>-    packet = packet_received.get();
>-  } else {
>-    packet_to_send = rtc::MakeUnique<RtpPacketToSend>(nullptr);
>-    packet = packet_to_send.get();
>-  }
>-  packet->SetSsrc(RandomSsrc());
>-  packet->SetSequenceNumber(static_cast<uint16_t>(RandomInt()));
>-  packet->SetPayloadSize(prng_.Rand(0u, 1000u));
>-  // TODO(terelius): Add marker bit, capture timestamp, CSRCs, and header
>-  // extensions.
>-
>-  std::unique_ptr<RtcEvent> event;
>-  if (direction == PacketDirection::kIncomingPacket) {
>-    event = rtc::MakeUnique<RtcEventRtpPacketIncoming>(*packet_received);
>-  } else {
>-    event = rtc::MakeUnique<RtcEventRtpPacketOutgoing>(*packet_to_send,
>-                                                       probe_cluster_id);
>-  }
>-  const int64_t timestamp_us = event->timestamp_us_;
>+  std::string encoded = encoder_->EncodeBatch(history_.begin(), history_.end());
>+  ASSERT_TRUE(parsed_log_.ParseString(encoded));
>+  const auto& outgoing_rtcp_packets = parsed_log_.outgoing_rtcp_packets();
> 
>-  ASSERT_TRUE(parsed_log_.ParseString(encoder_->Encode(*event)));
>-  ASSERT_EQ(parsed_log_.GetNumberOfEvents(), 1u);
>-  ASSERT_EQ(parsed_log_.GetEventType(0), ParsedRtcEventLog::RTP_EVENT);
>-
>-  PacketDirection parsed_direction;
>-  uint8_t parsed_rtp_header[IP_PACKET_SIZE];
>-  size_t parsed_header_length;
>-  size_t parsed_total_length;
>-  int parsed_probe_cluster_id;
>-  parsed_log_.GetRtpHeader(0, &parsed_direction, parsed_rtp_header,
>-                           &parsed_header_length, &parsed_total_length,
>-                           &parsed_probe_cluster_id);
>-
>-  EXPECT_EQ(parsed_log_.GetTimestamp(0), timestamp_us);
>-  EXPECT_EQ(parsed_direction, direction);
>-  if (parsed_direction == PacketDirection::kOutgoingPacket) {
>-    EXPECT_EQ(parsed_probe_cluster_id, probe_cluster_id);
>-  }
>-  EXPECT_EQ(memcmp(parsed_rtp_header, packet->data(), parsed_header_length), 0);
>-  EXPECT_EQ(parsed_header_length, packet->headers_size());
>-  EXPECT_EQ(parsed_total_length, packet->size());
>+  ASSERT_EQ(outgoing_rtcp_packets.size(), 1u);
>+  EXPECT_TRUE(
>+      test::VerifyLoggedRtcpPacketOutgoing(*event, outgoing_rtcp_packets[0]));
> }
> 
> TEST_P(RtcEventLogEncoderTest, RtcEventRtpPacketIncoming) {
>-  TestRtcEventRtpPacket(PacketDirection::kIncomingPacket);
>+  uint32_t ssrc = prng_.Rand<uint32_t>();
>+  RtpHeaderExtensionMap extension_map;  // TODO(terelius): Test extensions too.
>+  std::unique_ptr<RtcEventRtpPacketIncoming> event =
>+      gen_.NewRtpPacketIncoming(ssrc, extension_map);
>+  history_.push_back(event->Copy());
>+
>+  std::string encoded = encoder_->EncodeBatch(history_.begin(), history_.end());
>+  ASSERT_TRUE(parsed_log_.ParseString(encoded));
>+  const auto& incoming_rtp_packets_by_ssrc =
>+      parsed_log_.incoming_rtp_packets_by_ssrc();
>+
>+  ASSERT_EQ(incoming_rtp_packets_by_ssrc.size(), 1u);
>+  const auto& stream = incoming_rtp_packets_by_ssrc[0];
>+  EXPECT_EQ(stream.ssrc, ssrc);
>+  ASSERT_EQ(stream.incoming_packets.size(), 1u);
>+  EXPECT_TRUE(
>+      test::VerifyLoggedRtpPacketIncoming(*event, stream.incoming_packets[0]));
> }
> 
> TEST_P(RtcEventLogEncoderTest, RtcEventRtpPacketOutgoing) {
>-  TestRtcEventRtpPacket(PacketDirection::kOutgoingPacket);
>+  uint32_t ssrc = prng_.Rand<uint32_t>();
>+  RtpHeaderExtensionMap extension_map;  // TODO(terelius): Test extensions too.
>+  std::unique_ptr<RtcEventRtpPacketOutgoing> event =
>+      gen_.NewRtpPacketOutgoing(ssrc, extension_map);
>+  history_.push_back(event->Copy());
>+
>+  std::string encoded = encoder_->EncodeBatch(history_.begin(), history_.end());
>+  ASSERT_TRUE(parsed_log_.ParseString(encoded));
>+  const auto& outgoing_rtp_packets_by_ssrc =
>+      parsed_log_.outgoing_rtp_packets_by_ssrc();
>+
>+  ASSERT_EQ(outgoing_rtp_packets_by_ssrc.size(), 1u);
>+  const auto& stream = outgoing_rtp_packets_by_ssrc[0];
>+  EXPECT_EQ(stream.ssrc, ssrc);
>+  ASSERT_EQ(stream.outgoing_packets.size(), 1u);
>+  EXPECT_TRUE(
>+      test::VerifyLoggedRtpPacketOutgoing(*event, stream.outgoing_packets[0]));
> }
> 
> TEST_P(RtcEventLogEncoderTest, RtcEventVideoReceiveStreamConfig) {
>-  auto stream_config = rtc::MakeUnique<rtclog::StreamConfig>();
>-  stream_config->local_ssrc = RandomSsrc();
>-  stream_config->remote_ssrc = RandomSsrc();
>-  stream_config->rtcp_mode = RtcpMode::kCompound;
>-  stream_config->remb = prng_.Rand<bool>();
>-  std::vector<RtpExtension> extensions = RandomRtpExtensions();
>-  for (const auto& extension : extensions)
>-    stream_config->rtp_extensions.push_back(extension);
>-  stream_config->codecs.emplace_back("CODEC", 122, 7);
>-
>-  auto original_stream_config = *stream_config;
>-
>-  auto event = rtc::MakeUnique<RtcEventVideoReceiveStreamConfig>(
>-      std::move(stream_config));
>-  const int64_t timestamp_us = event->timestamp_us_;
>-
>-  ASSERT_TRUE(parsed_log_.ParseString(encoder_->Encode(*event)));
>-  ASSERT_EQ(parsed_log_.GetNumberOfEvents(), 1u);
>-  ASSERT_EQ(parsed_log_.GetEventType(0),
>-            ParsedRtcEventLog::VIDEO_RECEIVER_CONFIG_EVENT);
>-
>-  auto parsed_event = parsed_log_.GetVideoReceiveConfig(0);
>-  EXPECT_EQ(parsed_log_.GetTimestamp(0), timestamp_us);
>-  EXPECT_EQ(parsed_event, original_stream_config);
>+  uint32_t ssrc = prng_.Rand<uint32_t>();
>+  RtpHeaderExtensionMap extensions = gen_.NewRtpHeaderExtensionMap();
>+  std::unique_ptr<RtcEventVideoReceiveStreamConfig> event =
>+      gen_.NewVideoReceiveStreamConfig(ssrc, extensions);
>+  history_.push_back(event->Copy());
>+
>+  std::string encoded = encoder_->EncodeBatch(history_.begin(), history_.end());
>+  ASSERT_TRUE(parsed_log_.ParseString(encoded));
>+  const auto& video_recv_configs = parsed_log_.video_recv_configs();
>+
>+  ASSERT_EQ(video_recv_configs.size(), 1u);
>+  EXPECT_TRUE(test::VerifyLoggedVideoRecvConfig(*event, video_recv_configs[0]));
> }
> 
> TEST_P(RtcEventLogEncoderTest, RtcEventVideoSendStreamConfig) {
>-  auto stream_config = rtc::MakeUnique<rtclog::StreamConfig>();
>-  stream_config->local_ssrc = RandomSsrc();
>-  std::vector<RtpExtension> extensions = RandomRtpExtensions();
>-  for (const auto& extension : extensions)
>-    stream_config->rtp_extensions.push_back(extension);
>-  stream_config->codecs.emplace_back("CODEC", 120, 3);
>-
>-  auto original_stream_config = *stream_config;
>-
>-  auto event =
>-      rtc::MakeUnique<RtcEventVideoSendStreamConfig>(std::move(stream_config));
>-  const int64_t timestamp_us = event->timestamp_us_;
>-
>-  ASSERT_TRUE(parsed_log_.ParseString(encoder_->Encode(*event)));
>-  ASSERT_EQ(parsed_log_.GetNumberOfEvents(), 1u);
>-  ASSERT_EQ(parsed_log_.GetEventType(0),
>-            ParsedRtcEventLog::VIDEO_SENDER_CONFIG_EVENT);
>-
>-  auto parsed_event = parsed_log_.GetVideoSendConfig(0)[0];
>-  EXPECT_EQ(parsed_log_.GetTimestamp(0), timestamp_us);
>-  EXPECT_EQ(parsed_event, original_stream_config);
>+  uint32_t ssrc = prng_.Rand<uint32_t>();
>+  RtpHeaderExtensionMap extensions = gen_.NewRtpHeaderExtensionMap();
>+  std::unique_ptr<RtcEventVideoSendStreamConfig> event =
>+      gen_.NewVideoSendStreamConfig(ssrc, extensions);
>+  history_.push_back(event->Copy());
>+
>+  std::string encoded = encoder_->EncodeBatch(history_.begin(), history_.end());
>+  ASSERT_TRUE(parsed_log_.ParseString(encoded));
>+  const auto& video_send_configs = parsed_log_.video_send_configs();
>+
>+  ASSERT_EQ(video_send_configs.size(), 1u);
>+  EXPECT_TRUE(test::VerifyLoggedVideoSendConfig(*event, video_send_configs[0]));
> }
> 
> INSTANTIATE_TEST_CASE_P(RandomSeeds,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event.h
>index 6410698673d..21d6e48e0f5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event.h
>@@ -11,7 +11,7 @@
> #ifndef LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_H_
> #define LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_H_
> 
>-#include <typedefs.h>
>+#include <memory>
> 
> #include "rtc_base/timeutils.h"
> 
>@@ -37,6 +37,8 @@ class RtcEvent {
>     AudioSendStreamConfig,
>     BweUpdateDelayBased,
>     BweUpdateLossBased,
>+    IceCandidatePairConfig,
>+    IceCandidatePairEvent,
>     ProbeClusterCreated,
>     ProbeResultFailure,
>     ProbeResultSuccess,
>@@ -55,7 +57,12 @@ class RtcEvent {
> 
>   virtual bool IsConfigEvent() const = 0;
> 
>+  virtual std::unique_ptr<RtcEvent> Copy() const = 0;
>+
>   const int64_t timestamp_us_;
>+
>+ protected:
>+  explicit RtcEvent(int64_t timestamp_us) : timestamp_us_(timestamp_us) {}
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_alr_state.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_alr_state.cc
>index e329079c613..c1f1b395320 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_alr_state.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_alr_state.cc
>@@ -9,11 +9,15 @@
>  */
> 
> #include "logging/rtc_event_log/events/rtc_event_alr_state.h"
>+#include "absl/memory/memory.h"
> 
> namespace webrtc {
> 
> RtcEventAlrState::RtcEventAlrState(bool in_alr) : in_alr_(in_alr) {}
> 
>+RtcEventAlrState::RtcEventAlrState(const RtcEventAlrState& other)
>+    : RtcEvent(other.timestamp_us_), in_alr_(other.in_alr_) {}
>+
> RtcEventAlrState::~RtcEventAlrState() = default;
> 
> RtcEvent::Type RtcEventAlrState::GetType() const {
>@@ -24,4 +28,8 @@ bool RtcEventAlrState::IsConfigEvent() const {
>   return false;
> }
> 
>+std::unique_ptr<RtcEvent> RtcEventAlrState::Copy() const {
>+  return absl::WrapUnique<RtcEvent>(new RtcEventAlrState(*this));
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_alr_state.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_alr_state.h
>index d07f3f2019b..e737c915e1d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_alr_state.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_alr_state.h
>@@ -11,9 +11,9 @@
> #ifndef LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_ALR_STATE_H_
> #define LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_ALR_STATE_H_
> 
>-#include "logging/rtc_event_log/events/rtc_event.h"
>+#include <memory>
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include "logging/rtc_event_log/events/rtc_event.h"
> 
> namespace webrtc {
> 
>@@ -26,7 +26,12 @@ class RtcEventAlrState final : public RtcEvent {
> 
>   bool IsConfigEvent() const override;
> 
>+  std::unique_ptr<RtcEvent> Copy() const override;
>+
>   const bool in_alr_;
>+
>+ private:
>+  RtcEventAlrState(const RtcEventAlrState& other);
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_network_adaptation.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_network_adaptation.cc
>index 5ab7da67965..dcf2d7de592 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_network_adaptation.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_network_adaptation.cc
>@@ -12,6 +12,7 @@
> 
> #include <utility>
> 
>+#include "absl/memory/memory.h"
> #include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
> 
> namespace webrtc {
>@@ -20,6 +21,11 @@ RtcEventAudioNetworkAdaptation::RtcEventAudioNetworkAdaptation(
>     std::unique_ptr<AudioEncoderRuntimeConfig> config)
>     : config_(std::move(config)) {}
> 
>+RtcEventAudioNetworkAdaptation::RtcEventAudioNetworkAdaptation(
>+    const RtcEventAudioNetworkAdaptation& other)
>+    : RtcEvent(other.timestamp_us_),
>+      config_(absl::make_unique<AudioEncoderRuntimeConfig>(*other.config_)) {}
>+
> RtcEventAudioNetworkAdaptation::~RtcEventAudioNetworkAdaptation() = default;
> 
> RtcEvent::Type RtcEventAudioNetworkAdaptation::GetType() const {
>@@ -30,4 +36,8 @@ bool RtcEventAudioNetworkAdaptation::IsConfigEvent() const {
>   return false;
> }
> 
>+std::unique_ptr<RtcEvent> RtcEventAudioNetworkAdaptation::Copy() const {
>+  return absl::WrapUnique(new RtcEventAudioNetworkAdaptation(*this));
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h
>index 9b39d24bfa6..65fbad8621d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h
>@@ -29,7 +29,12 @@ class RtcEventAudioNetworkAdaptation final : public RtcEvent {
> 
>   bool IsConfigEvent() const override;
> 
>+  std::unique_ptr<RtcEvent> Copy() const override;
>+
>   const std::unique_ptr<const AudioEncoderRuntimeConfig> config_;
>+
>+ private:
>+  RtcEventAudioNetworkAdaptation(const RtcEventAudioNetworkAdaptation& other);
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_playout.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_playout.cc
>index 8378993acec..a9643ba868c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_playout.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_playout.cc
>@@ -10,10 +10,15 @@
> 
> #include "logging/rtc_event_log/events/rtc_event_audio_playout.h"
> 
>+#include "absl/memory/memory.h"
>+
> namespace webrtc {
> 
> RtcEventAudioPlayout::RtcEventAudioPlayout(uint32_t ssrc) : ssrc_(ssrc) {}
> 
>+RtcEventAudioPlayout::RtcEventAudioPlayout(const RtcEventAudioPlayout& other)
>+    : RtcEvent(other.timestamp_us_), ssrc_(other.ssrc_) {}
>+
> RtcEvent::Type RtcEventAudioPlayout::GetType() const {
>   return RtcEvent::Type::AudioPlayout;
> }
>@@ -22,4 +27,8 @@ bool RtcEventAudioPlayout::IsConfigEvent() const {
>   return false;
> }
> 
>+std::unique_ptr<RtcEvent> RtcEventAudioPlayout::Copy() const {
>+  return absl::WrapUnique<RtcEvent>(new RtcEventAudioPlayout(*this));
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_playout.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_playout.h
>index c74148109bd..3080f5b9f0e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_playout.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_playout.h
>@@ -11,6 +11,8 @@
> #ifndef LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_AUDIO_PLAYOUT_H_
> #define LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_AUDIO_PLAYOUT_H_
> 
>+#include <memory>
>+
> #include "logging/rtc_event_log/events/rtc_event.h"
> 
> namespace webrtc {
>@@ -24,7 +26,12 @@ class RtcEventAudioPlayout final : public RtcEvent {
> 
>   bool IsConfigEvent() const override;
> 
>+  std::unique_ptr<RtcEvent> Copy() const override;
>+
>   const uint32_t ssrc_;
>+
>+ private:
>+  RtcEventAudioPlayout(const RtcEventAudioPlayout& other);
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_receive_stream_config.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_receive_stream_config.cc
>index 6bc9dc6b306..fee7c3c766c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_receive_stream_config.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_receive_stream_config.cc
>@@ -12,6 +12,7 @@
> 
> #include <utility>
> 
>+#include "absl/memory/memory.h"
> #include "logging/rtc_event_log/rtc_stream_config.h"
> 
> namespace webrtc {
>@@ -20,6 +21,11 @@ RtcEventAudioReceiveStreamConfig::RtcEventAudioReceiveStreamConfig(
>     std::unique_ptr<rtclog::StreamConfig> config)
>     : config_(std::move(config)) {}
> 
>+RtcEventAudioReceiveStreamConfig::RtcEventAudioReceiveStreamConfig(
>+    const RtcEventAudioReceiveStreamConfig& other)
>+    : RtcEvent(other.timestamp_us_),
>+      config_(absl::make_unique<rtclog::StreamConfig>(*other.config_)) {}
>+
> RtcEventAudioReceiveStreamConfig::~RtcEventAudioReceiveStreamConfig() = default;
> 
> RtcEvent::Type RtcEventAudioReceiveStreamConfig::GetType() const {
>@@ -30,4 +36,10 @@ bool RtcEventAudioReceiveStreamConfig::IsConfigEvent() const {
>   return true;
> }
> 
>+std::unique_ptr<RtcEvent> RtcEventAudioReceiveStreamConfig::Copy() const {
>+  auto config_copy = absl::make_unique<rtclog::StreamConfig>(*config_);
>+  return absl::WrapUnique<RtcEvent>(
>+      new RtcEventAudioReceiveStreamConfig(*this));
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_receive_stream_config.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_receive_stream_config.h
>index ae79bbf44a4..27576503143 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_receive_stream_config.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_receive_stream_config.h
>@@ -31,7 +31,13 @@ class RtcEventAudioReceiveStreamConfig final : public RtcEvent {
> 
>   bool IsConfigEvent() const override;
> 
>+  std::unique_ptr<RtcEvent> Copy() const override;
>+
>   const std::unique_ptr<const rtclog::StreamConfig> config_;
>+
>+ private:
>+  RtcEventAudioReceiveStreamConfig(
>+      const RtcEventAudioReceiveStreamConfig& other);
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_send_stream_config.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_send_stream_config.cc
>index de88bcad5c0..47939c7bdc3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_send_stream_config.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_send_stream_config.cc
>@@ -12,6 +12,7 @@
> 
> #include <utility>
> 
>+#include "absl/memory/memory.h"
> #include "logging/rtc_event_log/rtc_stream_config.h"
> 
> namespace webrtc {
>@@ -20,6 +21,11 @@ RtcEventAudioSendStreamConfig::RtcEventAudioSendStreamConfig(
>     std::unique_ptr<rtclog::StreamConfig> config)
>     : config_(std::move(config)) {}
> 
>+RtcEventAudioSendStreamConfig::RtcEventAudioSendStreamConfig(
>+    const RtcEventAudioSendStreamConfig& other)
>+    : RtcEvent(other.timestamp_us_),
>+      config_(absl::make_unique<rtclog::StreamConfig>(*other.config_)) {}
>+
> RtcEventAudioSendStreamConfig::~RtcEventAudioSendStreamConfig() = default;
> 
> RtcEvent::Type RtcEventAudioSendStreamConfig::GetType() const {
>@@ -30,4 +36,9 @@ bool RtcEventAudioSendStreamConfig::IsConfigEvent() const {
>   return true;
> }
> 
>+std::unique_ptr<RtcEvent> RtcEventAudioSendStreamConfig::Copy() const {
>+  auto config_copy = absl::make_unique<rtclog::StreamConfig>(*config_);
>+  return absl::WrapUnique<RtcEvent>(new RtcEventAudioSendStreamConfig(*this));
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_send_stream_config.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_send_stream_config.h
>index 44611f1e4b6..b248a619150 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_send_stream_config.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_audio_send_stream_config.h
>@@ -31,7 +31,12 @@ class RtcEventAudioSendStreamConfig final : public RtcEvent {
> 
>   bool IsConfigEvent() const override;
> 
>+  std::unique_ptr<RtcEvent> Copy() const override;
>+
>   const std::unique_ptr<const rtclog::StreamConfig> config_;
>+
>+ private:
>+  RtcEventAudioSendStreamConfig(const RtcEventAudioSendStreamConfig& other);
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_bwe_update_delay_based.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_bwe_update_delay_based.cc
>index caf0f77c446..8c179712605 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_bwe_update_delay_based.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_bwe_update_delay_based.cc
>@@ -10,6 +10,7 @@
> 
> #include "logging/rtc_event_log/events/rtc_event_bwe_update_delay_based.h"
> 
>+#include "absl/memory/memory.h"
> #include "modules/remote_bitrate_estimator/include/bwe_defines.h"
> 
> namespace webrtc {
>@@ -19,6 +20,12 @@ RtcEventBweUpdateDelayBased::RtcEventBweUpdateDelayBased(
>     BandwidthUsage detector_state)
>     : bitrate_bps_(bitrate_bps), detector_state_(detector_state) {}
> 
>+RtcEventBweUpdateDelayBased::RtcEventBweUpdateDelayBased(
>+    const RtcEventBweUpdateDelayBased& other)
>+    : RtcEvent(other.timestamp_us_),
>+      bitrate_bps_(other.bitrate_bps_),
>+      detector_state_(other.detector_state_) {}
>+
> RtcEventBweUpdateDelayBased::~RtcEventBweUpdateDelayBased() = default;
> 
> RtcEvent::Type RtcEventBweUpdateDelayBased::GetType() const {
>@@ -29,4 +36,8 @@ bool RtcEventBweUpdateDelayBased::IsConfigEvent() const {
>   return false;
> }
> 
>+std::unique_ptr<RtcEvent> RtcEventBweUpdateDelayBased::Copy() const {
>+  return absl::WrapUnique<RtcEvent>(new RtcEventBweUpdateDelayBased(*this));
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_bwe_update_delay_based.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_bwe_update_delay_based.h
>index b59b78cdcd9..3f47dc7f970 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_bwe_update_delay_based.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_bwe_update_delay_based.h
>@@ -11,6 +11,8 @@
> #ifndef LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_BWE_UPDATE_DELAY_BASED_H_
> #define LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_BWE_UPDATE_DELAY_BASED_H_
> 
>+#include <memory>
>+
> #include "logging/rtc_event_log/events/rtc_event.h"
> 
> namespace webrtc {
>@@ -27,8 +29,13 @@ class RtcEventBweUpdateDelayBased final : public RtcEvent {
> 
>   bool IsConfigEvent() const override;
> 
>+  std::unique_ptr<RtcEvent> Copy() const override;
>+
>   const int32_t bitrate_bps_;
>   const BandwidthUsage detector_state_;
>+
>+ private:
>+  RtcEventBweUpdateDelayBased(const RtcEventBweUpdateDelayBased& other);
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_bwe_update_loss_based.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_bwe_update_loss_based.cc
>index 5d11200e736..b0e72ddb0aa 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_bwe_update_loss_based.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_bwe_update_loss_based.cc
>@@ -10,6 +10,8 @@
> 
> #include "logging/rtc_event_log/events/rtc_event_bwe_update_loss_based.h"
> 
>+#include "absl/memory/memory.h"
>+
> namespace webrtc {
> 
> RtcEventBweUpdateLossBased::RtcEventBweUpdateLossBased(int32_t bitrate_bps,
>@@ -19,6 +21,13 @@ RtcEventBweUpdateLossBased::RtcEventBweUpdateLossBased(int32_t bitrate_bps,
>       fraction_loss_(fraction_loss),
>       total_packets_(total_packets) {}
> 
>+RtcEventBweUpdateLossBased::RtcEventBweUpdateLossBased(
>+    const RtcEventBweUpdateLossBased& other)
>+    : RtcEvent(other.timestamp_us_),
>+      bitrate_bps_(other.bitrate_bps_),
>+      fraction_loss_(other.fraction_loss_),
>+      total_packets_(other.total_packets_) {}
>+
> RtcEventBweUpdateLossBased::~RtcEventBweUpdateLossBased() = default;
> 
> RtcEvent::Type RtcEventBweUpdateLossBased::GetType() const {
>@@ -29,4 +38,8 @@ bool RtcEventBweUpdateLossBased::IsConfigEvent() const {
>   return false;
> }
> 
>+std::unique_ptr<RtcEvent> RtcEventBweUpdateLossBased::Copy() const {
>+  return absl::WrapUnique<RtcEvent>(new RtcEventBweUpdateLossBased(*this));
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_bwe_update_loss_based.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_bwe_update_loss_based.h
>index 474b9de0404..2d4ffda1b1e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_bwe_update_loss_based.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_bwe_update_loss_based.h
>@@ -11,6 +11,8 @@
> #ifndef LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_BWE_UPDATE_LOSS_BASED_H_
> #define LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_BWE_UPDATE_LOSS_BASED_H_
> 
>+#include <memory>
>+
> #include "logging/rtc_event_log/events/rtc_event.h"
> 
> namespace webrtc {
>@@ -26,9 +28,14 @@ class RtcEventBweUpdateLossBased final : public RtcEvent {
> 
>   bool IsConfigEvent() const override;
> 
>+  std::unique_ptr<RtcEvent> Copy() const override;
>+
>   const int32_t bitrate_bps_;
>   const uint8_t fraction_loss_;
>   const int32_t total_packets_;
>+
>+ private:
>+  RtcEventBweUpdateLossBased(const RtcEventBweUpdateLossBased& other);
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_ice_candidate_pair.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_ice_candidate_pair.cc
>new file mode 100644
>index 00000000000..fa32de68d5e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_ice_candidate_pair.cc
>@@ -0,0 +1,42 @@
>+/*
>+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "logging/rtc_event_log/events/rtc_event_ice_candidate_pair.h"
>+
>+#include "absl/memory/memory.h"
>+
>+namespace webrtc {
>+
>+RtcEventIceCandidatePair::RtcEventIceCandidatePair(
>+    IceCandidatePairEventType type,
>+    uint32_t candidate_pair_id)
>+    : type_(type), candidate_pair_id_(candidate_pair_id) {}
>+
>+RtcEventIceCandidatePair::RtcEventIceCandidatePair(
>+    const RtcEventIceCandidatePair& other)
>+    : RtcEvent(other.timestamp_us_),
>+      type_(other.type_),
>+      candidate_pair_id_(other.candidate_pair_id_) {}
>+
>+RtcEventIceCandidatePair::~RtcEventIceCandidatePair() = default;
>+
>+RtcEvent::Type RtcEventIceCandidatePair::GetType() const {
>+  return RtcEvent::Type::IceCandidatePairEvent;
>+}
>+
>+bool RtcEventIceCandidatePair::IsConfigEvent() const {
>+  return false;
>+}
>+
>+std::unique_ptr<RtcEvent> RtcEventIceCandidatePair::Copy() const {
>+  return absl::WrapUnique<RtcEvent>(new RtcEventIceCandidatePair(*this));
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_ice_candidate_pair.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_ice_candidate_pair.h
>new file mode 100644
>index 00000000000..f49a0d1792c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_ice_candidate_pair.h
>@@ -0,0 +1,50 @@
>+/*
>+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_ICE_CANDIDATE_PAIR_H_
>+#define LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_ICE_CANDIDATE_PAIR_H_
>+
>+#include <memory>
>+
>+#include "logging/rtc_event_log/events/rtc_event.h"
>+
>+namespace webrtc {
>+
>+enum class IceCandidatePairEventType {
>+  kCheckSent,
>+  kCheckReceived,
>+  kCheckResponseSent,
>+  kCheckResponseReceived,
>+  kNumValues,
>+};
>+
>+class RtcEventIceCandidatePair final : public RtcEvent {
>+ public:
>+  RtcEventIceCandidatePair(IceCandidatePairEventType type,
>+                           uint32_t candidate_pair_id);
>+
>+  ~RtcEventIceCandidatePair() override;
>+
>+  Type GetType() const override;
>+
>+  bool IsConfigEvent() const override;
>+
>+  std::unique_ptr<RtcEvent> Copy() const override;
>+
>+  const IceCandidatePairEventType type_;
>+  const uint32_t candidate_pair_id_;
>+
>+ private:
>+  RtcEventIceCandidatePair(const RtcEventIceCandidatePair& other);
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_ICE_CANDIDATE_PAIR_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_ice_candidate_pair_config.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_ice_candidate_pair_config.cc
>new file mode 100644
>index 00000000000..f89de7740c7
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_ice_candidate_pair_config.cc
>@@ -0,0 +1,71 @@
>+/*
>+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "logging/rtc_event_log/events/rtc_event_ice_candidate_pair_config.h"
>+
>+#include "absl/memory/memory.h"
>+
>+namespace webrtc {
>+
>+IceCandidatePairDescription::IceCandidatePairDescription() {
>+  local_candidate_type = IceCandidateType::kUnknown;
>+  local_relay_protocol = IceCandidatePairProtocol::kUnknown;
>+  local_network_type = IceCandidateNetworkType::kUnknown;
>+  local_address_family = IceCandidatePairAddressFamily::kUnknown;
>+  remote_candidate_type = IceCandidateType::kUnknown;
>+  remote_address_family = IceCandidatePairAddressFamily::kUnknown;
>+  candidate_pair_protocol = IceCandidatePairProtocol::kUnknown;
>+}
>+
>+IceCandidatePairDescription::IceCandidatePairDescription(
>+    const IceCandidatePairDescription& other) {
>+  local_candidate_type = other.local_candidate_type;
>+  local_relay_protocol = other.local_relay_protocol;
>+  local_network_type = other.local_network_type;
>+  local_address_family = other.local_address_family;
>+  remote_candidate_type = other.remote_candidate_type;
>+  remote_address_family = other.remote_address_family;
>+  candidate_pair_protocol = other.candidate_pair_protocol;
>+}
>+
>+IceCandidatePairDescription::~IceCandidatePairDescription() {}
>+
>+RtcEventIceCandidatePairConfig::RtcEventIceCandidatePairConfig(
>+    IceCandidatePairConfigType type,
>+    uint32_t candidate_pair_id,
>+    const IceCandidatePairDescription& candidate_pair_desc)
>+    : type_(type),
>+      candidate_pair_id_(candidate_pair_id),
>+      candidate_pair_desc_(candidate_pair_desc) {}
>+
>+RtcEventIceCandidatePairConfig::RtcEventIceCandidatePairConfig(
>+    const RtcEventIceCandidatePairConfig& other)
>+    : RtcEvent(other.timestamp_us_),
>+      type_(other.type_),
>+      candidate_pair_id_(other.candidate_pair_id_),
>+      candidate_pair_desc_(other.candidate_pair_desc_) {}
>+
>+RtcEventIceCandidatePairConfig::~RtcEventIceCandidatePairConfig() = default;
>+
>+RtcEvent::Type RtcEventIceCandidatePairConfig::GetType() const {
>+  return RtcEvent::Type::IceCandidatePairConfig;
>+}
>+
>+// The ICE candidate pair config event is not equivalent to a RtcEventLog config
>+// event.
>+bool RtcEventIceCandidatePairConfig::IsConfigEvent() const {
>+  return false;
>+}
>+
>+std::unique_ptr<RtcEvent> RtcEventIceCandidatePairConfig::Copy() const {
>+  return absl::WrapUnique<RtcEvent>(new RtcEventIceCandidatePairConfig(*this));
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_ice_candidate_pair_config.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_ice_candidate_pair_config.h
>new file mode 100644
>index 00000000000..1f6eb8ceb0f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_ice_candidate_pair_config.h
>@@ -0,0 +1,107 @@
>+/*
>+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_ICE_CANDIDATE_PAIR_CONFIG_H_
>+#define LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_ICE_CANDIDATE_PAIR_CONFIG_H_
>+
>+#include <memory>
>+
>+#include "logging/rtc_event_log/events/rtc_event.h"
>+
>+namespace webrtc {
>+
>+enum class IceCandidatePairConfigType {
>+  kAdded,
>+  kUpdated,
>+  kDestroyed,
>+  kSelected,
>+  kNumValues,
>+};
>+
>+// TODO(qingsi): Change the names of candidate types to "host", "srflx", "prflx"
>+// and "relay" after the naming is spec-compliant in the signaling part
>+enum class IceCandidateType {
>+  kUnknown,
>+  kLocal,
>+  kStun,
>+  kPrflx,
>+  kRelay,
>+  kNumValues,
>+};
>+
>+enum class IceCandidatePairProtocol {
>+  kUnknown,
>+  kUdp,
>+  kTcp,
>+  kSsltcp,
>+  kTls,
>+  kNumValues,
>+};
>+
>+enum class IceCandidatePairAddressFamily {
>+  kUnknown,
>+  kIpv4,
>+  kIpv6,
>+  kNumValues,
>+};
>+
>+enum class IceCandidateNetworkType {
>+  kUnknown,
>+  kEthernet,
>+  kLoopback,
>+  kWifi,
>+  kVpn,
>+  kCellular,
>+  kNumValues,
>+};
>+
>+class IceCandidatePairDescription {
>+ public:
>+  IceCandidatePairDescription();
>+  explicit IceCandidatePairDescription(
>+      const IceCandidatePairDescription& other);
>+
>+  ~IceCandidatePairDescription();
>+
>+  IceCandidateType local_candidate_type;
>+  IceCandidatePairProtocol local_relay_protocol;
>+  IceCandidateNetworkType local_network_type;
>+  IceCandidatePairAddressFamily local_address_family;
>+  IceCandidateType remote_candidate_type;
>+  IceCandidatePairAddressFamily remote_address_family;
>+  IceCandidatePairProtocol candidate_pair_protocol;
>+};
>+
>+class RtcEventIceCandidatePairConfig final : public RtcEvent {
>+ public:
>+  RtcEventIceCandidatePairConfig(
>+      IceCandidatePairConfigType type,
>+      uint32_t candidate_pair_id,
>+      const IceCandidatePairDescription& candidate_pair_desc);
>+
>+  ~RtcEventIceCandidatePairConfig() override;
>+
>+  Type GetType() const override;
>+
>+  bool IsConfigEvent() const override;
>+
>+  std::unique_ptr<RtcEvent> Copy() const override;
>+
>+  const IceCandidatePairConfigType type_;
>+  const uint32_t candidate_pair_id_;
>+  const IceCandidatePairDescription candidate_pair_desc_;
>+
>+ private:
>+  RtcEventIceCandidatePairConfig(const RtcEventIceCandidatePairConfig& other);
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_ICE_CANDIDATE_PAIR_CONFIG_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_probe_cluster_created.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_probe_cluster_created.cc
>index d5dad741220..a60765dbd0f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_probe_cluster_created.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_probe_cluster_created.cc
>@@ -10,17 +10,27 @@
> 
> #include "logging/rtc_event_log/events/rtc_event_probe_cluster_created.h"
> 
>+#include "absl/memory/memory.h"
>+
> namespace webrtc {
> 
>-RtcEventProbeClusterCreated::RtcEventProbeClusterCreated(int id,
>-                                                         int bitrate_bps,
>-                                                         int min_probes,
>-                                                         int min_bytes)
>+RtcEventProbeClusterCreated::RtcEventProbeClusterCreated(int32_t id,
>+                                                         int32_t bitrate_bps,
>+                                                         uint32_t min_probes,
>+                                                         uint32_t min_bytes)
>     : id_(id),
>       bitrate_bps_(bitrate_bps),
>       min_probes_(min_probes),
>       min_bytes_(min_bytes) {}
> 
>+RtcEventProbeClusterCreated::RtcEventProbeClusterCreated(
>+    const RtcEventProbeClusterCreated& other)
>+    : RtcEvent(other.timestamp_us_),
>+      id_(other.id_),
>+      bitrate_bps_(other.bitrate_bps_),
>+      min_probes_(other.min_probes_),
>+      min_bytes_(other.min_bytes_) {}
>+
> RtcEvent::Type RtcEventProbeClusterCreated::GetType() const {
>   return RtcEvent::Type::ProbeClusterCreated;
> }
>@@ -29,4 +39,8 @@ bool RtcEventProbeClusterCreated::IsConfigEvent() const {
>   return false;
> }
> 
>+std::unique_ptr<RtcEvent> RtcEventProbeClusterCreated::Copy() const {
>+  return absl::WrapUnique<RtcEvent>(new RtcEventProbeClusterCreated(*this));
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_probe_cluster_created.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_probe_cluster_created.h
>index 938ba4f32ef..3e8c21f5c92 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_probe_cluster_created.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_probe_cluster_created.h
>@@ -11,26 +11,33 @@
> #ifndef LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_PROBE_CLUSTER_CREATED_H_
> #define LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_PROBE_CLUSTER_CREATED_H_
> 
>+#include <memory>
>+
> #include "logging/rtc_event_log/events/rtc_event.h"
> 
> namespace webrtc {
> 
> class RtcEventProbeClusterCreated final : public RtcEvent {
>  public:
>-  RtcEventProbeClusterCreated(int id,
>-                              int bitrate_bps,
>-                              int min_probes,
>-                              int min_bytes);
>+  RtcEventProbeClusterCreated(int32_t id,
>+                              int32_t bitrate_bps,
>+                              uint32_t min_probes,
>+                              uint32_t min_bytes);
>   ~RtcEventProbeClusterCreated() override = default;
> 
>   Type GetType() const override;
> 
>   bool IsConfigEvent() const override;
> 
>-  const int id_;
>-  const int bitrate_bps_;
>-  const int min_probes_;
>-  const int min_bytes_;
>+  std::unique_ptr<RtcEvent> Copy() const override;
>+
>+  const int32_t id_;
>+  const int32_t bitrate_bps_;
>+  const uint32_t min_probes_;
>+  const uint32_t min_bytes_;
>+
>+ private:
>+  RtcEventProbeClusterCreated(const RtcEventProbeClusterCreated& other);
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_probe_result_failure.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_probe_result_failure.cc
>index 1242ab825a3..aa6091ca775 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_probe_result_failure.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_probe_result_failure.cc
>@@ -10,13 +10,21 @@
> 
> #include "logging/rtc_event_log/events/rtc_event_probe_result_failure.h"
> 
>+#include "absl/memory/memory.h"
>+
> namespace webrtc {
> 
> RtcEventProbeResultFailure::RtcEventProbeResultFailure(
>-    int id,
>+    int32_t id,
>     ProbeFailureReason failure_reason)
>     : id_(id), failure_reason_(failure_reason) {}
> 
>+RtcEventProbeResultFailure::RtcEventProbeResultFailure(
>+    const RtcEventProbeResultFailure& other)
>+    : RtcEvent(other.timestamp_us_),
>+      id_(other.id_),
>+      failure_reason_(other.failure_reason_) {}
>+
> RtcEvent::Type RtcEventProbeResultFailure::GetType() const {
>   return RtcEvent::Type::ProbeResultFailure;
> }
>@@ -25,4 +33,8 @@ bool RtcEventProbeResultFailure::IsConfigEvent() const {
>   return false;
> }
> 
>+std::unique_ptr<RtcEvent> RtcEventProbeResultFailure::Copy() const {
>+  return absl::WrapUnique<RtcEvent>(new RtcEventProbeResultFailure(*this));
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_probe_result_failure.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_probe_result_failure.h
>index 90ef4dec515..181c6940370 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_probe_result_failure.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_probe_result_failure.h
>@@ -11,6 +11,8 @@
> #ifndef LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_PROBE_RESULT_FAILURE_H_
> #define LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_PROBE_RESULT_FAILURE_H_
> 
>+#include <memory>
>+
> #include "logging/rtc_event_log/events/rtc_event.h"
> 
> namespace webrtc {
>@@ -24,15 +26,20 @@ enum class ProbeFailureReason {
> 
> class RtcEventProbeResultFailure final : public RtcEvent {
>  public:
>-  RtcEventProbeResultFailure(int id, ProbeFailureReason failure_reason);
>+  RtcEventProbeResultFailure(int32_t id, ProbeFailureReason failure_reason);
>   ~RtcEventProbeResultFailure() override = default;
> 
>   Type GetType() const override;
> 
>   bool IsConfigEvent() const override;
> 
>-  const int id_;
>+  std::unique_ptr<RtcEvent> Copy() const override;
>+
>+  const int32_t id_;
>   const ProbeFailureReason failure_reason_;
>+
>+ private:
>+  RtcEventProbeResultFailure(const RtcEventProbeResultFailure& other);
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_probe_result_success.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_probe_result_success.cc
>index f7b06fe28e9..55907295deb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_probe_result_success.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_probe_result_success.cc
>@@ -10,11 +10,20 @@
> 
> #include "logging/rtc_event_log/events/rtc_event_probe_result_success.h"
> 
>+#include "absl/memory/memory.h"
>+
> namespace webrtc {
> 
>-RtcEventProbeResultSuccess::RtcEventProbeResultSuccess(int id, int bitrate_bps)
>+RtcEventProbeResultSuccess::RtcEventProbeResultSuccess(int32_t id,
>+                                                       int32_t bitrate_bps)
>     : id_(id), bitrate_bps_(bitrate_bps) {}
> 
>+RtcEventProbeResultSuccess::RtcEventProbeResultSuccess(
>+    const RtcEventProbeResultSuccess& other)
>+    : RtcEvent(other.timestamp_us_),
>+      id_(other.id_),
>+      bitrate_bps_(other.bitrate_bps_) {}
>+
> RtcEvent::Type RtcEventProbeResultSuccess::GetType() const {
>   return RtcEvent::Type::ProbeResultSuccess;
> }
>@@ -23,4 +32,8 @@ bool RtcEventProbeResultSuccess::IsConfigEvent() const {
>   return false;
> }
> 
>+std::unique_ptr<RtcEvent> RtcEventProbeResultSuccess::Copy() const {
>+  return absl::WrapUnique<RtcEvent>(new RtcEventProbeResultSuccess(*this));
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_probe_result_success.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_probe_result_success.h
>index f8f02164d64..15ff1833bf0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_probe_result_success.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_probe_result_success.h
>@@ -11,21 +11,28 @@
> #ifndef LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_PROBE_RESULT_SUCCESS_H_
> #define LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_PROBE_RESULT_SUCCESS_H_
> 
>+#include <memory>
>+
> #include "logging/rtc_event_log/events/rtc_event.h"
> 
> namespace webrtc {
> 
> class RtcEventProbeResultSuccess final : public RtcEvent {
>  public:
>-  RtcEventProbeResultSuccess(int id, int bitrate_bps);
>+  RtcEventProbeResultSuccess(int32_t id, int32_t bitrate_bps);
>   ~RtcEventProbeResultSuccess() override = default;
> 
>   Type GetType() const override;
> 
>   bool IsConfigEvent() const override;
> 
>-  const int id_;
>-  const int bitrate_bps_;
>+  std::unique_ptr<RtcEvent> Copy() const override;
>+
>+  const int32_t id_;
>+  const int32_t bitrate_bps_;
>+
>+ private:
>+  RtcEventProbeResultSuccess(const RtcEventProbeResultSuccess& other);
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtcp_packet_incoming.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtcp_packet_incoming.cc
>index 1f0765e40cc..119938c7f71 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtcp_packet_incoming.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtcp_packet_incoming.cc
>@@ -10,12 +10,19 @@
> 
> #include "logging/rtc_event_log/events/rtc_event_rtcp_packet_incoming.h"
> 
>+#include "absl/memory/memory.h"
>+
> namespace webrtc {
> 
> RtcEventRtcpPacketIncoming::RtcEventRtcpPacketIncoming(
>     rtc::ArrayView<const uint8_t> packet)
>     : packet_(packet.data(), packet.size()) {}
> 
>+RtcEventRtcpPacketIncoming::RtcEventRtcpPacketIncoming(
>+    const RtcEventRtcpPacketIncoming& other)
>+    : RtcEvent(other.timestamp_us_),
>+      packet_(other.packet_.data(), other.packet_.size()) {}
>+
> RtcEventRtcpPacketIncoming::~RtcEventRtcpPacketIncoming() = default;
> 
> RtcEvent::Type RtcEventRtcpPacketIncoming::GetType() const {
>@@ -26,4 +33,8 @@ bool RtcEventRtcpPacketIncoming::IsConfigEvent() const {
>   return false;
> }
> 
>+std::unique_ptr<RtcEvent> RtcEventRtcpPacketIncoming::Copy() const {
>+  return absl::WrapUnique<RtcEvent>(new RtcEventRtcpPacketIncoming(*this));
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtcp_packet_incoming.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtcp_packet_incoming.h
>index f7af462c101..6d904953a48 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtcp_packet_incoming.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtcp_packet_incoming.h
>@@ -11,6 +11,8 @@
> #ifndef LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_RTCP_PACKET_INCOMING_H_
> #define LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_RTCP_PACKET_INCOMING_H_
> 
>+#include <memory>
>+
> #include "api/array_view.h"
> #include "logging/rtc_event_log/events/rtc_event.h"
> #include "rtc_base/buffer.h"
>@@ -26,7 +28,12 @@ class RtcEventRtcpPacketIncoming final : public RtcEvent {
> 
>   bool IsConfigEvent() const override;
> 
>+  std::unique_ptr<RtcEvent> Copy() const override;
>+
>   rtc::Buffer packet_;
>+
>+ private:
>+  RtcEventRtcpPacketIncoming(const RtcEventRtcpPacketIncoming& other);
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtcp_packet_outgoing.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtcp_packet_outgoing.cc
>index b2dfaf08677..4085ac0f313 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtcp_packet_outgoing.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtcp_packet_outgoing.cc
>@@ -10,12 +10,19 @@
> 
> #include "logging/rtc_event_log/events/rtc_event_rtcp_packet_outgoing.h"
> 
>+#include "absl/memory/memory.h"
>+
> namespace webrtc {
> 
> RtcEventRtcpPacketOutgoing::RtcEventRtcpPacketOutgoing(
>     rtc::ArrayView<const uint8_t> packet)
>     : packet_(packet.data(), packet.size()) {}
> 
>+RtcEventRtcpPacketOutgoing::RtcEventRtcpPacketOutgoing(
>+    const RtcEventRtcpPacketOutgoing& other)
>+    : RtcEvent(other.timestamp_us_),
>+      packet_(other.packet_.data(), other.packet_.size()) {}
>+
> RtcEventRtcpPacketOutgoing::~RtcEventRtcpPacketOutgoing() = default;
> 
> RtcEvent::Type RtcEventRtcpPacketOutgoing::GetType() const {
>@@ -26,4 +33,8 @@ bool RtcEventRtcpPacketOutgoing::IsConfigEvent() const {
>   return false;
> }
> 
>+std::unique_ptr<RtcEvent> RtcEventRtcpPacketOutgoing::Copy() const {
>+  return absl::WrapUnique<RtcEvent>(new RtcEventRtcpPacketOutgoing(*this));
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtcp_packet_outgoing.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtcp_packet_outgoing.h
>index 2c38ae08785..3aa9e7199ad 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtcp_packet_outgoing.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtcp_packet_outgoing.h
>@@ -11,6 +11,8 @@
> #ifndef LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_RTCP_PACKET_OUTGOING_H_
> #define LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_RTCP_PACKET_OUTGOING_H_
> 
>+#include <memory>
>+
> #include "api/array_view.h"
> #include "logging/rtc_event_log/events/rtc_event.h"
> #include "rtc_base/buffer.h"
>@@ -26,7 +28,12 @@ class RtcEventRtcpPacketOutgoing final : public RtcEvent {
> 
>   bool IsConfigEvent() const override;
> 
>+  std::unique_ptr<RtcEvent> Copy() const override;
>+
>   rtc::Buffer packet_;
>+
>+ private:
>+  RtcEventRtcpPacketOutgoing(const RtcEventRtcpPacketOutgoing& other);
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtp_packet_incoming.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtp_packet_incoming.cc
>index e0dcec5e793..8e15d021751 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtp_packet_incoming.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtp_packet_incoming.cc
>@@ -10,6 +10,7 @@
> 
> #include "logging/rtc_event_log/events/rtc_event_rtp_packet_incoming.h"
> 
>+#include "absl/memory/memory.h"
> #include "modules/rtp_rtcp/source/rtp_packet_received.h"
> 
> namespace webrtc {
>@@ -20,6 +21,12 @@ RtcEventRtpPacketIncoming::RtcEventRtpPacketIncoming(
>   header_.CopyHeaderFrom(packet);
> }
> 
>+RtcEventRtpPacketIncoming::RtcEventRtpPacketIncoming(
>+    const RtcEventRtpPacketIncoming& other)
>+    : RtcEvent(other.timestamp_us_), packet_length_(other.packet_length_) {
>+  header_.CopyHeaderFrom(other.header_);
>+}
>+
> RtcEventRtpPacketIncoming::~RtcEventRtpPacketIncoming() = default;
> 
> RtcEvent::Type RtcEventRtpPacketIncoming::GetType() const {
>@@ -30,4 +37,8 @@ bool RtcEventRtpPacketIncoming::IsConfigEvent() const {
>   return false;
> }
> 
>+std::unique_ptr<RtcEvent> RtcEventRtpPacketIncoming::Copy() const {
>+  return absl::WrapUnique<RtcEvent>(new RtcEventRtpPacketIncoming(*this));
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtp_packet_incoming.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtp_packet_incoming.h
>index 4aaf86f289b..6b141b6d576 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtp_packet_incoming.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtp_packet_incoming.h
>@@ -11,6 +11,8 @@
> #ifndef LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_RTP_PACKET_INCOMING_H_
> #define LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_RTP_PACKET_INCOMING_H_
> 
>+#include <memory>
>+
> #include "logging/rtc_event_log/events/rtc_event.h"
> #include "modules/rtp_rtcp/source/rtp_packet.h"
> 
>@@ -27,8 +29,13 @@ class RtcEventRtpPacketIncoming final : public RtcEvent {
> 
>   bool IsConfigEvent() const override;
> 
>+  std::unique_ptr<RtcEvent> Copy() const override;
>+
>   RtpPacket header_;            // Only the packet's header will be stored here.
>   const size_t packet_length_;  // Length before stripping away all but header.
>+
>+ private:
>+  RtcEventRtpPacketIncoming(const RtcEventRtpPacketIncoming& other);
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.cc
>index 207f76a511d..103440ad6e9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.cc
>@@ -10,6 +10,7 @@
> 
> #include "logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.h"
> 
>+#include "absl/memory/memory.h"
> #include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
> 
> namespace webrtc {
>@@ -21,6 +22,14 @@ RtcEventRtpPacketOutgoing::RtcEventRtpPacketOutgoing(
>   header_.CopyHeaderFrom(packet);
> }
> 
>+RtcEventRtpPacketOutgoing::RtcEventRtpPacketOutgoing(
>+    const RtcEventRtpPacketOutgoing& other)
>+    : RtcEvent(other.timestamp_us_),
>+      packet_length_(other.packet_length_),
>+      probe_cluster_id_(other.probe_cluster_id_) {
>+  header_.CopyHeaderFrom(other.header_);
>+}
>+
> RtcEventRtpPacketOutgoing::~RtcEventRtpPacketOutgoing() = default;
> 
> RtcEvent::Type RtcEventRtpPacketOutgoing::GetType() const {
>@@ -31,4 +40,8 @@ bool RtcEventRtpPacketOutgoing::IsConfigEvent() const {
>   return false;
> }
> 
>+std::unique_ptr<RtcEvent> RtcEventRtpPacketOutgoing::Copy() const {
>+  return absl::WrapUnique<RtcEvent>(new RtcEventRtpPacketOutgoing(*this));
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.h
>index 898cdce40ed..171a21c420e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.h
>@@ -11,6 +11,8 @@
> #ifndef LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_RTP_PACKET_OUTGOING_H_
> #define LOGGING_RTC_EVENT_LOG_EVENTS_RTC_EVENT_RTP_PACKET_OUTGOING_H_
> 
>+#include <memory>
>+
> #include "logging/rtc_event_log/events/rtc_event.h"
> #include "modules/rtp_rtcp/source/rtp_packet.h"
> 
>@@ -28,9 +30,14 @@ class RtcEventRtpPacketOutgoing final : public RtcEvent {
> 
>   bool IsConfigEvent() const override;
> 
>+  std::unique_ptr<RtcEvent> Copy() const override;
>+
>   RtpPacket header_;            // Only the packet's header will be stored here.
>   const size_t packet_length_;  // Length before stripping away all but header.
>   const int probe_cluster_id_;
>+
>+ private:
>+  RtcEventRtpPacketOutgoing(const RtcEventRtpPacketOutgoing& other);
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_video_receive_stream_config.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_video_receive_stream_config.cc
>index 514974b2ec5..7f4dd9b7b80 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_video_receive_stream_config.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_video_receive_stream_config.cc
>@@ -12,12 +12,19 @@
> 
> #include <utility>
> 
>+#include "absl/memory/memory.h"
>+
> namespace webrtc {
> 
> RtcEventVideoReceiveStreamConfig::RtcEventVideoReceiveStreamConfig(
>     std::unique_ptr<rtclog::StreamConfig> config)
>     : config_(std::move(config)) {}
> 
>+RtcEventVideoReceiveStreamConfig::RtcEventVideoReceiveStreamConfig(
>+    const RtcEventVideoReceiveStreamConfig& other)
>+    : RtcEvent(other.timestamp_us_),
>+      config_(absl::make_unique<rtclog::StreamConfig>(*other.config_)) {}
>+
> RtcEventVideoReceiveStreamConfig::~RtcEventVideoReceiveStreamConfig() = default;
> 
> RtcEvent::Type RtcEventVideoReceiveStreamConfig::GetType() const {
>@@ -28,4 +35,9 @@ bool RtcEventVideoReceiveStreamConfig::IsConfigEvent() const {
>   return true;
> }
> 
>+std::unique_ptr<RtcEvent> RtcEventVideoReceiveStreamConfig::Copy() const {
>+  return absl::WrapUnique<RtcEvent>(
>+      new RtcEventVideoReceiveStreamConfig(*this));
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_video_receive_stream_config.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_video_receive_stream_config.h
>index 1d470e50ce3..c6263c3fdbe 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_video_receive_stream_config.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_video_receive_stream_config.h
>@@ -28,7 +28,13 @@ class RtcEventVideoReceiveStreamConfig final : public RtcEvent {
> 
>   bool IsConfigEvent() const override;
> 
>+  std::unique_ptr<RtcEvent> Copy() const override;
>+
>   const std::unique_ptr<const rtclog::StreamConfig> config_;
>+
>+ private:
>+  RtcEventVideoReceiveStreamConfig(
>+      const RtcEventVideoReceiveStreamConfig& other);
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_video_send_stream_config.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_video_send_stream_config.cc
>index 9bb2b5e38e8..b4b78a8a52a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_video_send_stream_config.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_video_send_stream_config.cc
>@@ -12,12 +12,19 @@
> 
> #include <utility>
> 
>+#include "absl/memory/memory.h"
>+
> namespace webrtc {
> 
> RtcEventVideoSendStreamConfig::RtcEventVideoSendStreamConfig(
>     std::unique_ptr<rtclog::StreamConfig> config)
>     : config_(std::move(config)) {}
> 
>+RtcEventVideoSendStreamConfig::RtcEventVideoSendStreamConfig(
>+    const RtcEventVideoSendStreamConfig& other)
>+    : RtcEvent(other.timestamp_us_),
>+      config_(absl::make_unique<rtclog::StreamConfig>(*other.config_)) {}
>+
> RtcEventVideoSendStreamConfig::~RtcEventVideoSendStreamConfig() = default;
> 
> RtcEvent::Type RtcEventVideoSendStreamConfig::GetType() const {
>@@ -28,4 +35,8 @@ bool RtcEventVideoSendStreamConfig::IsConfigEvent() const {
>   return true;
> }
> 
>+std::unique_ptr<RtcEvent> RtcEventVideoSendStreamConfig::Copy() const {
>+  return absl::WrapUnique<RtcEvent>(new RtcEventVideoSendStreamConfig(*this));
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_video_send_stream_config.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_video_send_stream_config.h
>index 97bdba9613b..3472f8855be 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_video_send_stream_config.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/events/rtc_event_video_send_stream_config.h
>@@ -28,7 +28,12 @@ class RtcEventVideoSendStreamConfig final : public RtcEvent {
> 
>   bool IsConfigEvent() const override;
> 
>+  std::unique_ptr<RtcEvent> Copy() const override;
>+
>   const std::unique_ptr<const rtclog::StreamConfig> config_;
>+
>+ private:
>+  RtcEventVideoSendStreamConfig(const RtcEventVideoSendStreamConfig& other);
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/fake_rtc_event_log.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/fake_rtc_event_log.cc
>new file mode 100644
>index 00000000000..55f4b582c78
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/fake_rtc_event_log.cc
>@@ -0,0 +1,41 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "logging/rtc_event_log/fake_rtc_event_log.h"
>+
>+#include "logging/rtc_event_log/events/rtc_event_ice_candidate_pair.h"
>+#include "rtc_base/bind.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/logging.h"
>+
>+namespace webrtc {
>+
>+FakeRtcEventLog::FakeRtcEventLog(rtc::Thread* thread) : thread_(thread) {
>+  RTC_DCHECK(thread_);
>+}
>+FakeRtcEventLog::~FakeRtcEventLog() = default;
>+
>+bool FakeRtcEventLog::StartLogging(std::unique_ptr<RtcEventLogOutput> output,
>+                                   int64_t output_period_ms) {
>+  return true;
>+}
>+
>+void FakeRtcEventLog::StopLogging() {
>+  invoker_.Flush(thread_);
>+}
>+
>+void FakeRtcEventLog::Log(std::unique_ptr<RtcEvent> event) {
>+  RtcEvent::Type rtc_event_type = event->GetType();
>+  invoker_.AsyncInvoke<void>(
>+      RTC_FROM_HERE, thread_,
>+      rtc::Bind(&FakeRtcEventLog::IncrementEventCount, this, rtc_event_type));
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/fake_rtc_event_log.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/fake_rtc_event_log.h
>new file mode 100644
>index 00000000000..c5ea08aa690
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/fake_rtc_event_log.h
>@@ -0,0 +1,43 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef LOGGING_RTC_EVENT_LOG_FAKE_RTC_EVENT_LOG_H_
>+#define LOGGING_RTC_EVENT_LOG_FAKE_RTC_EVENT_LOG_H_
>+
>+#include <map>
>+#include <memory>
>+
>+#include "logging/rtc_event_log/events/rtc_event.h"
>+#include "logging/rtc_event_log/rtc_event_log.h"
>+#include "rtc_base/asyncinvoker.h"
>+#include "rtc_base/thread.h"
>+
>+namespace webrtc {
>+
>+class FakeRtcEventLog : public RtcEventLog {
>+ public:
>+  explicit FakeRtcEventLog(rtc::Thread* thread);
>+  ~FakeRtcEventLog() override;
>+  bool StartLogging(std::unique_ptr<RtcEventLogOutput> output,
>+                    int64_t output_period_ms) override;
>+  void StopLogging() override;
>+  void Log(std::unique_ptr<RtcEvent> event) override;
>+  int GetEventCount(RtcEvent::Type event_type) { return count_[event_type]; }
>+
>+ private:
>+  void IncrementEventCount(RtcEvent::Type event_type) { ++count_[event_type]; }
>+  std::map<RtcEvent::Type, int> count_;
>+  rtc::Thread* thread_;
>+  rtc::AsyncInvoker invoker_;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // LOGGING_RTC_EVENT_LOG_FAKE_RTC_EVENT_LOG_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/fake_rtc_event_log_factory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/fake_rtc_event_log_factory.cc
>new file mode 100644
>index 00000000000..b3f638cbb4d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/fake_rtc_event_log_factory.cc
>@@ -0,0 +1,34 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "logging/rtc_event_log/fake_rtc_event_log_factory.h"
>+
>+#include <utility>
>+
>+#include "logging/rtc_event_log/rtc_event_log.h"
>+
>+namespace webrtc {
>+
>+std::unique_ptr<RtcEventLog> FakeRtcEventLogFactory::CreateRtcEventLog(
>+    RtcEventLog::EncodingType encoding_type) {
>+  std::unique_ptr<RtcEventLog> fake_event_log(new FakeRtcEventLog(thread()));
>+  last_log_created_ = fake_event_log.get();
>+  return fake_event_log;
>+}
>+
>+std::unique_ptr<RtcEventLog> FakeRtcEventLogFactory::CreateRtcEventLog(
>+    RtcEventLog::EncodingType encoding_type,
>+    std::unique_ptr<rtc::TaskQueue> task_queue) {
>+  std::unique_ptr<RtcEventLog> fake_event_log(new FakeRtcEventLog(thread()));
>+  last_log_created_ = fake_event_log.get();
>+  return fake_event_log;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/fake_rtc_event_log_factory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/fake_rtc_event_log_factory.h
>new file mode 100644
>index 00000000000..1160e5464c6
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/fake_rtc_event_log_factory.h
>@@ -0,0 +1,44 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef LOGGING_RTC_EVENT_LOG_FAKE_RTC_EVENT_LOG_FACTORY_H_
>+#define LOGGING_RTC_EVENT_LOG_FAKE_RTC_EVENT_LOG_FACTORY_H_
>+
>+#include <memory>
>+
>+#include "logging/rtc_event_log/fake_rtc_event_log.h"
>+#include "logging/rtc_event_log/rtc_event_log_factory_interface.h"
>+#include "rtc_base/thread.h"
>+
>+namespace webrtc {
>+
>+class FakeRtcEventLogFactory : public RtcEventLogFactoryInterface {
>+ public:
>+  explicit FakeRtcEventLogFactory(rtc::Thread* thread) : thread_(thread) {}
>+  ~FakeRtcEventLogFactory() override {}
>+
>+  std::unique_ptr<RtcEventLog> CreateRtcEventLog(
>+      RtcEventLog::EncodingType encoding_type) override;
>+
>+  std::unique_ptr<RtcEventLog> CreateRtcEventLog(
>+      RtcEventLog::EncodingType encoding_type,
>+      std::unique_ptr<rtc::TaskQueue> task_queue) override;
>+
>+  webrtc::RtcEventLog* last_log_created() { return last_log_created_; }
>+  rtc::Thread* thread() { return thread_; }
>+
>+ private:
>+  webrtc::RtcEventLog* last_log_created_;
>+  rtc::Thread* thread_;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // LOGGING_RTC_EVENT_LOG_FAKE_RTC_EVENT_LOG_FACTORY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/icelogger.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/icelogger.cc
>new file mode 100644
>index 00000000000..5f91c5e7bdb
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/icelogger.cc
>@@ -0,0 +1,50 @@
>+/*
>+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "logging/rtc_event_log/icelogger.h"
>+
>+#include "absl/memory/memory.h"
>+#include "logging/rtc_event_log/rtc_event_log.h"
>+
>+namespace webrtc {
>+
>+IceEventLog::IceEventLog() {}
>+IceEventLog::~IceEventLog() {}
>+
>+void IceEventLog::LogCandidatePairConfig(
>+    IceCandidatePairConfigType type,
>+    uint32_t candidate_pair_id,
>+    const IceCandidatePairDescription& candidate_pair_desc) {
>+  if (event_log_ == nullptr) {
>+    return;
>+  }
>+  candidate_pair_desc_by_id_[candidate_pair_id] = candidate_pair_desc;
>+  event_log_->Log(absl::make_unique<RtcEventIceCandidatePairConfig>(
>+      type, candidate_pair_id, candidate_pair_desc));
>+}
>+
>+void IceEventLog::LogCandidatePairEvent(IceCandidatePairEventType type,
>+                                        uint32_t candidate_pair_id) {
>+  if (event_log_ == nullptr) {
>+    return;
>+  }
>+  event_log_->Log(
>+      absl::make_unique<RtcEventIceCandidatePair>(type, candidate_pair_id));
>+}
>+
>+void IceEventLog::DumpCandidatePairDescriptionToMemoryAsConfigEvents() const {
>+  for (const auto& desc_id_pair : candidate_pair_desc_by_id_) {
>+    event_log_->Log(absl::make_unique<RtcEventIceCandidatePairConfig>(
>+        IceCandidatePairConfigType::kUpdated, desc_id_pair.first,
>+        desc_id_pair.second));
>+  }
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/icelogger.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/icelogger.h
>new file mode 100644
>index 00000000000..b590fd7cb32
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/icelogger.h
>@@ -0,0 +1,55 @@
>+/*
>+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef LOGGING_RTC_EVENT_LOG_ICELOGGER_H_
>+#define LOGGING_RTC_EVENT_LOG_ICELOGGER_H_
>+
>+#include <unordered_map>
>+
>+#include "logging/rtc_event_log/events/rtc_event_ice_candidate_pair.h"
>+#include "logging/rtc_event_log/events/rtc_event_ice_candidate_pair_config.h"
>+
>+namespace webrtc {
>+
>+class RtcEventLog;
>+
>+// IceEventLog wraps RtcEventLog and provides structural logging of ICE-specific
>+// events. The logged events are serialized with other RtcEvent's if protobuf is
>+// enabled in the build.
>+class IceEventLog {
>+ public:
>+  IceEventLog();
>+  ~IceEventLog();
>+
>+  void set_event_log(RtcEventLog* event_log) { event_log_ = event_log; }
>+
>+  void LogCandidatePairConfig(
>+      IceCandidatePairConfigType type,
>+      uint32_t candidate_pair_id,
>+      const IceCandidatePairDescription& candidate_pair_desc);
>+
>+  void LogCandidatePairEvent(IceCandidatePairEventType type,
>+                             uint32_t candidate_pair_id);
>+
>+  // This method constructs a config event for each candidate pair with their
>+  // description and logs these config events. It is intended to be called when
>+  // logging starts to ensure that we have at least one config for each
>+  // candidate pair id.
>+  void DumpCandidatePairDescriptionToMemoryAsConfigEvents() const;
>+
>+ private:
>+  RtcEventLog* event_log_ = nullptr;
>+  std::unordered_map<uint32_t, IceCandidatePairDescription>
>+      candidate_pair_desc_by_id_;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // LOGGING_RTC_EVENT_LOG_ICELOGGER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/test/video_capture_main_mac.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.cc
>similarity index 56%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/test/video_capture_main_mac.mm
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.cc
>index 680160fb9fb..240a4f66939 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/test/video_capture_main_mac.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.cc
>@@ -1,5 +1,5 @@
> /*
>- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>  *
>  *  Use of this source code is governed by a BSD-style license
>  *  that can be found in the LICENSE file in the root of the source
>@@ -8,10 +8,11 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "test/gtest.h"
>-#include "test/testsupport/mac/run_threaded_main_mac.h"
>+#include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
> 
>-int ImplementThisToRunYourTest(int argc, char** argv) {
>-  testing::InitGoogleTest(&argc, argv);
>-  return RUN_ALL_TESTS();
>-}
>+namespace webrtc {
>+
>+MockRtcEventLog::MockRtcEventLog() = default;
>+MockRtcEventLog::~MockRtcEventLog() = default;
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h
>index 6f69169eff8..44207ff2baf 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h
>@@ -20,6 +20,9 @@ namespace webrtc {
> 
> class MockRtcEventLog : public RtcEventLog {
>  public:
>+  MockRtcEventLog();
>+  ~MockRtcEventLog();
>+
>   virtual bool StartLogging(std::unique_ptr<RtcEventLogOutput> output,
>                             int64_t output_period_ms) {
>     return StartLoggingProxy(output.get(), output_period_ms);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/output/rtc_event_log_output_file_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/output/rtc_event_log_output_file_unittest.cc
>index a2a02049614..c42a7111488 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/output/rtc_event_log_output_file_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/output/rtc_event_log_output_file_unittest.cc
>@@ -8,12 +8,14 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>+#include "logging/rtc_event_log/output/rtc_event_log_output_file.h"
>+
> #include <fstream>
> #include <memory>
> #include <string>
> 
>-#include "logging/rtc_event_log/output/rtc_event_log_output_file.h"
>-#include "rtc_base/ptr_util.h"
>+#include "absl/memory/memory.h"
>+#include "rtc_base/checks.h"
> #include "test/gtest.h"
> #include "test/testsupport/fileutils.h"
> 
>@@ -49,13 +51,14 @@ class RtcEventLogOutputFileTest : public ::testing::Test {
> };
> 
> TEST_F(RtcEventLogOutputFileTest, NonDefectiveOutputsStartOutActive) {
>-  auto output_file = rtc::MakeUnique<RtcEventLogOutputFile>(output_file_name_);
>+  auto output_file =
>+      absl::make_unique<RtcEventLogOutputFile>(output_file_name_);
>   EXPECT_TRUE(output_file->IsActive());
> }
> 
> TEST_F(RtcEventLogOutputFileTest, DefectiveOutputsStartOutInactive) {
>   const std::string illegal_filename = "/////////";
>-  auto output_file = rtc::MakeUnique<RtcEventLogOutputFile>(illegal_filename);
>+  auto output_file = absl::make_unique<RtcEventLogOutputFile>(illegal_filename);
>   EXPECT_FALSE(output_file->IsActive());
> }
> 
>@@ -63,7 +66,8 @@ TEST_F(RtcEventLogOutputFileTest, DefectiveOutputsStartOutInactive) {
> TEST_F(RtcEventLogOutputFileTest, UnlimitedOutputFile) {
>   const std::string output_str = "one two three";
> 
>-  auto output_file = rtc::MakeUnique<RtcEventLogOutputFile>(output_file_name_);
>+  auto output_file =
>+      absl::make_unique<RtcEventLogOutputFile>(output_file_name_);
>   output_file->Write(output_str);
>   output_file.reset();  // Closing the file flushes the buffer to disk.
> 
>@@ -74,7 +78,7 @@ TEST_F(RtcEventLogOutputFileTest, UnlimitedOutputFile) {
> TEST_F(RtcEventLogOutputFileTest, LimitedOutputFileCappedToCapacity) {
>   // Fit two bytes, then the third should be rejected.
>   auto output_file =
>-      rtc::MakeUnique<RtcEventLogOutputFile>(output_file_name_, 2);
>+      absl::make_unique<RtcEventLogOutputFile>(output_file_name_, 2);
> 
>   output_file->Write("1");
>   output_file->Write("2");
>@@ -94,7 +98,7 @@ TEST_F(RtcEventLogOutputFileTest, DoNotWritePartialLines) {
>   // Set a file size limit just shy of fitting the entire second line.
>   const size_t size_limit = output_str_1.length() + output_str_2.length() - 1;
>   auto output_file =
>-      rtc::MakeUnique<RtcEventLogOutputFile>(output_file_name_, size_limit);
>+      absl::make_unique<RtcEventLogOutputFile>(output_file_name_, size_limit);
> 
>   output_file->Write(output_str_1);
>   output_file->Write(output_str_2);
>@@ -105,20 +109,20 @@ TEST_F(RtcEventLogOutputFileTest, DoNotWritePartialLines) {
> 
> TEST_F(RtcEventLogOutputFileTest, UnsuccessfulWriteReturnsFalse) {
>   auto output_file =
>-      rtc::MakeUnique<RtcEventLogOutputFile>(output_file_name_, 2);
>+      absl::make_unique<RtcEventLogOutputFile>(output_file_name_, 2);
>   EXPECT_FALSE(output_file->Write("abc"));
> }
> 
> TEST_F(RtcEventLogOutputFileTest, SuccessfulWriteReturnsTrue) {
>   auto output_file =
>-      rtc::MakeUnique<RtcEventLogOutputFile>(output_file_name_, 3);
>+      absl::make_unique<RtcEventLogOutputFile>(output_file_name_, 3);
>   EXPECT_TRUE(output_file->Write("abc"));
> }
> 
> // Even if capacity is reached, a successful write leaves the output active.
> TEST_F(RtcEventLogOutputFileTest, FileStillActiveAfterSuccessfulWrite) {
>   auto output_file =
>-      rtc::MakeUnique<RtcEventLogOutputFile>(output_file_name_, 3);
>+      absl::make_unique<RtcEventLogOutputFile>(output_file_name_, 3);
>   ASSERT_TRUE(output_file->Write("abc"));
>   EXPECT_TRUE(output_file->IsActive());
> }
>@@ -127,19 +131,18 @@ TEST_F(RtcEventLogOutputFileTest, FileStillActiveAfterSuccessfulWrite) {
> // not yet been reached.
> TEST_F(RtcEventLogOutputFileTest, FileInactiveAfterUnsuccessfulWrite) {
>   auto output_file =
>-      rtc::MakeUnique<RtcEventLogOutputFile>(output_file_name_, 2);
>+      absl::make_unique<RtcEventLogOutputFile>(output_file_name_, 2);
>   ASSERT_FALSE(output_file->Write("abc"));
>   EXPECT_FALSE(output_file->IsActive());
> }
> 
> TEST_F(RtcEventLogOutputFileTest, AllowReasonableFileSizeLimits) {
>-  auto output_file = rtc::MakeUnique<RtcEventLogOutputFile>(
>+  auto output_file = absl::make_unique<RtcEventLogOutputFile>(
>       output_file_name_, RtcEventLogOutputFile::kMaxReasonableFileSize);
>   EXPECT_TRUE(output_file->IsActive());
> }
> 
> #if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
>-#if !defined(WEBRTC_USE_MEMCHECK)  // Crashing expected to leak memory.
> TEST_F(RtcEventLogOutputFileTest, WritingToInactiveFileForbidden) {
>   RtcEventLogOutputFile output_file(output_file_name_, 2);
>   ASSERT_FALSE(output_file.Write("abc"));
>@@ -154,12 +157,11 @@ TEST_F(RtcEventLogOutputFileTest, DisallowUnreasonableFileSizeLimits) {
>   auto create_output_file = [&] {
>     const size_t unreasonable_size =
>         RtcEventLogOutputFile::kMaxReasonableFileSize + 1;
>-    output_file = rtc::MakeUnique<RtcEventLogOutputFile>(output_file_name_,
>-                                                         unreasonable_size);
>+    output_file = absl::make_unique<RtcEventLogOutputFile>(output_file_name_,
>+                                                           unreasonable_size);
>   };
>   EXPECT_DEATH(create_output_file(), "");
> }
>-#endif  // !WEBRTC_USE_MEMCHECK
> #endif
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log.cc
>index 2173590662d..73a598d6b1e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log.cc
>@@ -1,5 +1,5 @@
> /*
>- *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>  *
>  *  Use of this source code is governed by a BSD-style license
>  *  that can be found in the LICENSE file in the root of the source
>@@ -10,371 +10,12 @@
> 
> #include "logging/rtc_event_log/rtc_event_log.h"
> 
>-#include <atomic>
>-#include <deque>
>-#include <functional>
>-#include <limits>
>-#include <memory>
>-#include <utility>
>-#include <vector>
>-
>-#include "logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.h"
>-#include "logging/rtc_event_log/output/rtc_event_log_output_file.h"
>-#include "rtc_base/checks.h"
>-#include "rtc_base/constructormagic.h"
>-#include "rtc_base/event.h"
>-#include "rtc_base/logging.h"
>-#include "rtc_base/numerics/safe_conversions.h"
>-#include "rtc_base/numerics/safe_minmax.h"
>-#include "rtc_base/ptr_util.h"
>-#include "rtc_base/sequenced_task_checker.h"
>-#include "rtc_base/task_queue.h"
>-#include "rtc_base/thread_annotations.h"
>-
> namespace webrtc {
> 
>-#ifdef ENABLE_RTC_EVENT_LOG
>-
>-namespace {
>-constexpr size_t kMaxEventsInHistory = 10000;
>-// The config-history is supposed to be unbounded, but needs to have some bound
>-// to prevent an attack via unreasonable memory use.
>-constexpr size_t kMaxEventsInConfigHistory = 1000;
>-
>-// Observe a limit on the number of concurrent logs, so as not to run into
>-// OS-imposed limits on open files and/or threads/task-queues.
>-// TODO(eladalon): Known issue - there's a race over |rtc_event_log_count|.
>-std::atomic<int> rtc_event_log_count(0);
>-
>-// TODO(eladalon): This class exists because C++11 doesn't allow transferring a
>-// unique_ptr to a lambda (a copy constructor is required). We should get
>-// rid of this when we move to C++14.
>-template <typename T>
>-class ResourceOwningTask final : public rtc::QueuedTask {
>- public:
>-  ResourceOwningTask(std::unique_ptr<T> resource,
>-                     std::function<void(std::unique_ptr<T>)> handler)
>-      : resource_(std::move(resource)), handler_(handler) {}
>-
>-  bool Run() override {
>-    handler_(std::move(resource_));
>-    return true;
>-  }
>-
>- private:
>-  std::unique_ptr<T> resource_;
>-  std::function<void(std::unique_ptr<T>)> handler_;
>-};
>-
>-std::unique_ptr<RtcEventLogEncoder> CreateEncoder(
>-    RtcEventLog::EncodingType type) {
>-  switch (type) {
>-    case RtcEventLog::EncodingType::Legacy:
>-      return rtc::MakeUnique<RtcEventLogEncoderLegacy>();
>-    default:
>-      RTC_LOG(LS_ERROR) << "Unknown RtcEventLog encoder type (" << int(type)
>-                        << ")";
>-      RTC_NOTREACHED();
>-      return std::unique_ptr<RtcEventLogEncoder>(nullptr);
>-  }
>-}
>-
>-class RtcEventLogImpl final : public RtcEventLog {
>- public:
>-  explicit RtcEventLogImpl(std::unique_ptr<RtcEventLogEncoder> event_encoder);
>-  ~RtcEventLogImpl() override;
>-
>-  // TODO(eladalon): We should change these name to reflect that what we're
>-  // actually starting/stopping is the output of the log, not the log itself.
>-  bool StartLogging(std::unique_ptr<RtcEventLogOutput> output,
>-                    int64_t output_period_ms) override;
>-  void StopLogging() override;
>-
>-  void Log(std::unique_ptr<RtcEvent> event) override;
>-
>- private:
>-  void LogToMemory(std::unique_ptr<RtcEvent> event) RTC_RUN_ON(&task_queue_);
>-  void LogEventsFromMemoryToOutput() RTC_RUN_ON(&task_queue_);
>-
>-  void StopOutput() RTC_RUN_ON(&task_queue_);
>-
>-  void WriteConfigsAndHistoryToOutput(const std::string& encoded_configs,
>-                                      const std::string& encoded_history)
>-      RTC_RUN_ON(&task_queue_);
>-  void WriteToOutput(const std::string& output_string) RTC_RUN_ON(&task_queue_);
>-
>-  void StopLoggingInternal() RTC_RUN_ON(&task_queue_);
>-
>-  void ScheduleOutput() RTC_RUN_ON(&task_queue_);
>-
>-  // Make sure that the event log is "managed" - created/destroyed, as well
>-  // as started/stopped - from the same thread/task-queue.
>-  rtc::SequencedTaskChecker owner_sequence_checker_;
>-
>-  // History containing all past configuration events.
>-  std::deque<std::unique_ptr<RtcEvent>> config_history_
>-      RTC_ACCESS_ON(task_queue_);
>-
>-  // History containing the most recent (non-configuration) events (~10s).
>-  std::deque<std::unique_ptr<RtcEvent>> history_ RTC_ACCESS_ON(task_queue_);
>-
>-  size_t max_size_bytes_ RTC_ACCESS_ON(task_queue_);
>-  size_t written_bytes_ RTC_ACCESS_ON(task_queue_);
>-
>-  std::unique_ptr<RtcEventLogEncoder> event_encoder_ RTC_ACCESS_ON(task_queue_);
>-  std::unique_ptr<RtcEventLogOutput> event_output_ RTC_ACCESS_ON(task_queue_);
>-
>-  size_t num_config_events_written_ RTC_ACCESS_ON(task_queue_);
>-  int64_t output_period_ms_ RTC_ACCESS_ON(task_queue_);
>-  int64_t last_output_ms_ RTC_ACCESS_ON(task_queue_);
>-  bool output_scheduled_ RTC_ACCESS_ON(task_queue_);
>-
>-  // Since we are posting tasks bound to |this|,  it is critical that the event
>-  // log and it's members outlive the |task_queue_|. Keep the "task_queue_|
>-  // last to ensure it destructs first, or else tasks living on the queue might
>-  // access other members after they've been torn down.
>-  rtc::TaskQueue task_queue_;
>-
>-  RTC_DISALLOW_COPY_AND_ASSIGN(RtcEventLogImpl);
>-};
>-
>-RtcEventLogImpl::RtcEventLogImpl(
>-    std::unique_ptr<RtcEventLogEncoder> event_encoder)
>-    : max_size_bytes_(std::numeric_limits<decltype(max_size_bytes_)>::max()),
>-      written_bytes_(0),
>-      event_encoder_(std::move(event_encoder)),
>-      num_config_events_written_(0),
>-      output_period_ms_(kImmediateOutput),
>-      last_output_ms_(rtc::TimeMillis()),
>-      output_scheduled_(false),
>-      task_queue_("rtc_event_log") {}
>-
>-RtcEventLogImpl::~RtcEventLogImpl() {
>-  RTC_DCHECK_CALLED_SEQUENTIALLY(&owner_sequence_checker_);
>-
>-  // If we're logging to the output, this will stop that. Blocking function.
>-  StopLogging();
>-
>-  int count = std::atomic_fetch_sub(&rtc_event_log_count, 1) - 1;
>-  RTC_DCHECK_GE(count, 0);
>-}
>-
>-bool RtcEventLogImpl::StartLogging(std::unique_ptr<RtcEventLogOutput> output,
>-                                   int64_t output_period_ms) {
>-  RTC_DCHECK_CALLED_SEQUENTIALLY(&owner_sequence_checker_);
>-
>-  RTC_DCHECK(output_period_ms == kImmediateOutput || output_period_ms > 0);
>-
>-  if (!output->IsActive()) {
>-    // TODO(eladalon): We may want to remove the IsActive method. Otherwise
>-    // we probably want to be consistent and terminate any existing output.
>-    return false;
>-  }
>-
>-  RTC_LOG(LS_INFO) << "Starting WebRTC event log.";
>-
>-  const int64_t timestamp_us = rtc::TimeMicros();
>-
>-  // Binding to |this| is safe because |this| outlives the |task_queue_|.
>-  auto start = [this, timestamp_us](std::unique_ptr<RtcEventLogOutput> output) {
>-    RTC_DCHECK_RUN_ON(&task_queue_);
>-    RTC_DCHECK(output->IsActive());
>-    event_output_ = std::move(output);
>-    num_config_events_written_ = 0;
>-    WriteToOutput(event_encoder_->EncodeLogStart(timestamp_us));
>-    LogEventsFromMemoryToOutput();
>-  };
>-
>-  task_queue_.PostTask(rtc::MakeUnique<ResourceOwningTask<RtcEventLogOutput>>(
>-      std::move(output), start));
>-
>-  return true;
>-}
>-
>-void RtcEventLogImpl::StopLogging() {
>-  RTC_DCHECK_CALLED_SEQUENTIALLY(&owner_sequence_checker_);
>-
>-  RTC_LOG(LS_INFO) << "Stopping WebRTC event log.";
>-
>-  rtc::Event output_stopped(true, false);
>-
>-  // Binding to |this| is safe because |this| outlives the |task_queue_|.
>-  task_queue_.PostTask([this, &output_stopped]() {
>-    RTC_DCHECK_RUN_ON(&task_queue_);
>-    if (event_output_) {
>-      RTC_DCHECK(event_output_->IsActive());
>-      LogEventsFromMemoryToOutput();
>-    }
>-    StopLoggingInternal();
>-    output_stopped.Set();
>-  });
>-
>-  output_stopped.Wait(rtc::Event::kForever);
>-
>-  RTC_LOG(LS_INFO) << "WebRTC event log successfully stopped.";
>-}
>-
>-void RtcEventLogImpl::Log(std::unique_ptr<RtcEvent> event) {
>-  RTC_DCHECK(event);
>-
>-  // Binding to |this| is safe because |this| outlives the |task_queue_|.
>-  auto event_handler = [this](std::unique_ptr<RtcEvent> unencoded_event) {
>-    RTC_DCHECK_RUN_ON(&task_queue_);
>-    LogToMemory(std::move(unencoded_event));
>-    if (event_output_)
>-      ScheduleOutput();
>-  };
>-
>-  task_queue_.PostTask(rtc::MakeUnique<ResourceOwningTask<RtcEvent>>(
>-      std::move(event), event_handler));
>-}
>-
>-void RtcEventLogImpl::ScheduleOutput() {
>-  RTC_DCHECK(event_output_ && event_output_->IsActive());
>-  if (history_.size() >= kMaxEventsInHistory) {
>-    // We have to emergency drain the buffer. We can't wait for the scheduled
>-    // output task because there might be other event incoming before that.
>-    LogEventsFromMemoryToOutput();
>-    return;
>-  }
>-
>-  if (output_period_ms_ == kImmediateOutput) {
>-    // We are already on the |task_queue_| so there is no reason to post a task
>-    // if we want to output immediately.
>-    LogEventsFromMemoryToOutput();
>-    return;
>-  }
>-
>-  if (!output_scheduled_) {
>-    output_scheduled_ = true;
>-    // Binding to |this| is safe because |this| outlives the |task_queue_|.
>-    auto output_task = [this]() {
>-      RTC_DCHECK_RUN_ON(&task_queue_);
>-      if (event_output_) {
>-        RTC_DCHECK(event_output_->IsActive());
>-        LogEventsFromMemoryToOutput();
>-      }
>-      output_scheduled_ = false;
>-    };
>-    int64_t now_ms = rtc::TimeMillis();
>-    int64_t time_since_output_ms = now_ms - last_output_ms_;
>-    uint32_t delay = rtc::SafeClamp(output_period_ms_ - time_since_output_ms, 0,
>-                                    output_period_ms_);
>-    task_queue_.PostDelayedTask(output_task, delay);
>-  }
>-}
>-
>-void RtcEventLogImpl::LogToMemory(std::unique_ptr<RtcEvent> event) {
>-  std::deque<std::unique_ptr<RtcEvent>>& container =
>-      event->IsConfigEvent() ? config_history_ : history_;
>-  const size_t container_max_size =
>-      event->IsConfigEvent() ? kMaxEventsInConfigHistory : kMaxEventsInHistory;
>-
>-  if (container.size() >= container_max_size) {
>-    RTC_DCHECK(!event_output_);  // Shouldn't lose events if we have an output.
>-    container.pop_front();
>-  }
>-  container.push_back(std::move(event));
>-}
>-
>-void RtcEventLogImpl::LogEventsFromMemoryToOutput() {
>-  RTC_DCHECK(event_output_ && event_output_->IsActive());
>-  last_output_ms_ = rtc::TimeMillis();
>-
>-  // Serialize all stream configurations that haven't already been written to
>-  // this output. |num_config_events_written_| is used to track which configs we
>-  // have already written. (Note that the config may have been written to
>-  // previous outputs; configs are not discarded.)
>-  std::string encoded_configs;
>-  RTC_DCHECK_LE(num_config_events_written_, config_history_.size());
>-  if (num_config_events_written_ < config_history_.size()) {
>-    const auto begin = config_history_.begin() + num_config_events_written_;
>-    const auto end = config_history_.end();
>-    encoded_configs = event_encoder_->EncodeBatch(begin, end);
>-    num_config_events_written_ = config_history_.size();
>-  }
>-
>-  // Serialize the events in the event queue. Note that the write may fail,
>-  // for example if we are writing to a file and have reached the maximum limit.
>-  // We don't get any feedback if this happens, so we still remove the events
>-  // from the event log history. This is normally not a problem, but if another
>-  // log is started immediately after the first one becomes full, then one
>-  // cannot rely on the second log to contain everything that isn't in the first
>-  // log; one batch of events might be missing.
>-  std::string encoded_history =
>-      event_encoder_->EncodeBatch(history_.begin(), history_.end());
>-  history_.clear();
>-
>-  WriteConfigsAndHistoryToOutput(encoded_configs, encoded_history);
>-}
>-
>-void RtcEventLogImpl::WriteConfigsAndHistoryToOutput(
>-    const std::string& encoded_configs,
>-    const std::string& encoded_history) {
>-  // This function is used to merge the strings instead of calling the output
>-  // object twice with small strings. The function also avoids copying any
>-  // strings in the typical case where there are no config events.
>-  if (encoded_configs.size() == 0) {
>-    WriteToOutput(encoded_history);  // Typical case.
>-  } else if (encoded_history.size() == 0) {
>-    WriteToOutput(encoded_configs);  // Very unusual case.
>-  } else {
>-    WriteToOutput(encoded_configs + encoded_history);
>-  }
>-}
>-
>-void RtcEventLogImpl::StopOutput() {
>-  max_size_bytes_ = std::numeric_limits<decltype(max_size_bytes_)>::max();
>-  written_bytes_ = 0;
>-  event_output_.reset();
>-}
>-
>-void RtcEventLogImpl::StopLoggingInternal() {
>-  if (event_output_) {
>-    RTC_DCHECK(event_output_->IsActive());
>-    const int64_t timestamp_us = rtc::TimeMicros();
>-    event_output_->Write(event_encoder_->EncodeLogEnd(timestamp_us));
>-  }
>-  StopOutput();
>-}
>-
>-void RtcEventLogImpl::WriteToOutput(const std::string& output_string) {
>-  RTC_DCHECK(event_output_ && event_output_->IsActive());
>-  if (!event_output_->Write(output_string)) {
>-    RTC_LOG(LS_ERROR) << "Failed to write RTC event to output.";
>-    // The first failure closes the output.
>-    RTC_DCHECK(!event_output_->IsActive());
>-    StopOutput();  // Clean-up.
>-    return;
>-  }
>-  written_bytes_ += output_string.size();
>-}
>-
>-}  // namespace
>-
>-#endif  // ENABLE_RTC_EVENT_LOG
>-
>-// RtcEventLog member functions.
>-std::unique_ptr<RtcEventLog> RtcEventLog::Create(EncodingType encoding_type) {
>-#ifdef ENABLE_RTC_EVENT_LOG
>-  // TODO(eladalon): Known issue - there's a race over |rtc_event_log_count|.
>-  constexpr int kMaxLogCount = 5;
>-  int count = 1 + std::atomic_fetch_add(&rtc_event_log_count, 1);
>-  if (count > kMaxLogCount) {
>-    RTC_LOG(LS_WARNING) << "Denied creation of additional WebRTC event logs. "
>-                        << count - 1 << " logs open already.";
>-    std::atomic_fetch_sub(&rtc_event_log_count, 1);
>-    return CreateNull();
>-  }
>-  auto encoder = CreateEncoder(encoding_type);
>-  return rtc::MakeUnique<RtcEventLogImpl>(std::move(encoder));
>-#else
>-  return CreateNull();
>-#endif  // ENABLE_RTC_EVENT_LOG
>-}
>-
>-std::unique_ptr<RtcEventLog> RtcEventLog::CreateNull() {
>-  return std::unique_ptr<RtcEventLog>(new RtcEventLogNullImpl());
>+bool RtcEventLogNullImpl::StartLogging(
>+    std::unique_ptr<RtcEventLogOutput> output,
>+    int64_t output_period_ms) {
>+  return false;
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log.h
>index 89ace2c9e95..0c714069024 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log.h
>@@ -17,11 +17,11 @@
> 
> #include "api/rtceventlogoutput.h"
> #include "logging/rtc_event_log/events/rtc_event.h"
>+#include "rtc_base/task_queue.h"
> 
> namespace webrtc {
> 
>-class Clock;
>-
>+// TODO(terelius): Move this to the parser.
> enum PacketDirection { kIncomingPacket = 0, kOutgoingPacket };
> 
> class RtcEventLog {
>@@ -38,13 +38,12 @@ class RtcEventLog {
> 
>   // Factory method to create an RtcEventLog object.
>   static std::unique_ptr<RtcEventLog> Create(EncodingType encoding_type);
>-  // TODO(nisse): webrtc::Clock is deprecated. Delete this method and
>-  // above forward declaration of Clock when
>-  // webrtc/system_wrappers/include/clock.h is deleted.
>-  static std::unique_ptr<RtcEventLog> Create(const Clock*,
>-                                             EncodingType encoding_type) {
>-    return Create(encoding_type);
>-  }
>+
>+  // Factory method to create an RtcEventLog object which uses the given
>+  // rtc::TaskQueue for emitting output.
>+  static std::unique_ptr<RtcEventLog> Create(
>+      EncodingType encoding_type,
>+      std::unique_ptr<rtc::TaskQueue> task_queue);
> 
>   // Create an RtcEventLog object that does nothing.
>   static std::unique_ptr<RtcEventLog> CreateNull();
>@@ -65,12 +64,10 @@ class RtcEventLog {
> // No-op implementation is used if flag is not set, or in tests.
> class RtcEventLogNullImpl : public RtcEventLog {
>  public:
>-  bool StartLogging(std::unique_ptr<RtcEventLogOutput>,
>-                    int64_t) override {
>-    return false;
>-  }
>+  bool StartLogging(std::unique_ptr<RtcEventLogOutput> output,
>+                    int64_t output_period_ms) override;
>   void StopLogging() override {}
>-  void Log(std::unique_ptr<RtcEvent>) override {}
>+  void Log(std::unique_ptr<RtcEvent> event) override {}
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log.proto b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log.proto
>index 8d2e22a6f87..94e288e0245 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log.proto
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log.proto
>@@ -41,6 +41,8 @@ message Event {
>     BWE_PROBE_CLUSTER_CREATED_EVENT = 17;
>     BWE_PROBE_RESULT_EVENT = 18;
>     ALR_STATE_EVENT = 19;
>+    ICE_CANDIDATE_PAIR_CONFIG = 20;
>+    ICE_CANDIDATE_PAIR_EVENT = 21;
>   }
> 
>   // required - Indicates the type of this event
>@@ -85,6 +87,12 @@ message Event {
> 
>     // required if type == ALR_STATE_EVENT
>     AlrState alr_state = 19;
>+
>+    // required if type == ICE_CANDIDATE_PAIR_CONFIG
>+    IceCandidatePairConfig ice_candidate_pair_config = 20;
>+
>+    // required if type == ICE_CANDIDATE_PAIR_EVENT
>+    IceCandidatePairEvent ice_candidate_pair_event = 21;
>   }
> }
> 
>@@ -101,7 +109,7 @@ message RtpPacket {
>   optional bytes header = 4;
> 
>   // optional - The probe cluster id.
>-  optional uint32 probe_cluster_id = 5;
>+  optional int32 probe_cluster_id = 5;
> 
>   // Do not add code to log user payload data without a privacy review!
> }
>@@ -289,10 +297,10 @@ message AudioNetworkAdaptation {
> 
> message BweProbeCluster {
>   // required - The id of this probe cluster.
>-  optional uint32 id = 1;
>+  optional int32 id = 1;
> 
>   // required - The bitrate in bps that this probe cluster is meant to probe.
>-  optional uint64 bitrate_bps = 2;
>+  optional int32 bitrate_bps = 2;
> 
>   // required - The minimum number of packets used to probe the given bitrate.
>   optional uint32 min_packets = 3;
>@@ -303,7 +311,7 @@ message BweProbeCluster {
> 
> message BweProbeResult {
>   // required - The id of this probe cluster.
>-  optional uint32 id = 1;
>+  optional int32 id = 1;
> 
>   enum ResultType {
>     SUCCESS = 0;
>@@ -316,10 +324,92 @@ message BweProbeResult {
>   optional ResultType result = 2;
> 
>   // optional - but required if result == SUCCESS. The resulting bitrate in bps.
>-  optional uint64 bitrate_bps = 3;
>+  optional int32 bitrate_bps = 3;
> }
> 
> message AlrState {
>   // required - If we are in ALR or not.
>   optional bool in_alr = 1;
> }
>+
>+message IceCandidatePairConfig {
>+  enum IceCandidatePairConfigType {
>+    ADDED = 0;
>+    UPDATED = 1;
>+    DESTROYED = 2;
>+    SELECTED = 3;
>+  }
>+
>+  enum IceCandidateType {
>+    LOCAL = 0;
>+    STUN = 1;
>+    PRFLX = 2;
>+    RELAY = 3;
>+    UNKNOWN_CANDIDATE_TYPE = 4;
>+  }
>+
>+  enum Protocol {
>+    UDP = 0;
>+    TCP = 1;
>+    SSLTCP = 2;
>+    TLS = 3;
>+    UNKNOWN_PROTOCOL = 4;
>+  }
>+
>+  enum AddressFamily {
>+    IPV4 = 0;
>+    IPV6 = 1;
>+    UNKNOWN_ADDRESS_FAMILY = 2;
>+  }
>+
>+  enum NetworkType {
>+    ETHERNET = 0;
>+    LOOPBACK = 1;
>+    WIFI = 2;
>+    VPN = 3;
>+    CELLULAR = 4;
>+    UNKNOWN_NETWORK_TYPE = 5;
>+  }
>+
>+  // required
>+  optional IceCandidatePairConfigType config_type = 1;
>+
>+  // required
>+  optional uint32 candidate_pair_id = 2;
>+
>+  // required
>+  optional IceCandidateType local_candidate_type = 3;
>+
>+  // required
>+  optional Protocol local_relay_protocol = 4;
>+
>+  // required
>+  optional NetworkType local_network_type = 5;
>+
>+  // required
>+  optional AddressFamily local_address_family = 6;
>+
>+  // required
>+  optional IceCandidateType remote_candidate_type = 7;
>+
>+  // required
>+  optional AddressFamily remote_address_family = 8;
>+
>+  // required
>+  optional Protocol candidate_pair_protocol = 9;
>+}
>+
>+message IceCandidatePairEvent {
>+  enum IceCandidatePairEventType {
>+    CHECK_SENT = 0;
>+    CHECK_RECEIVED = 1;
>+    CHECK_RESPONSE_SENT = 2;
>+    CHECK_RESPONSE_RECEIVED = 3;
>+  }
>+
>+  // required
>+  optional IceCandidatePairEventType event_type = 1;
>+
>+  // required
>+  optional uint32 candidate_pair_id = 2;
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log2rtp_dump.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log2rtp_dump.cc
>index c6fa12997b1..b62759b0d51 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log2rtp_dump.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log2rtp_dump.cc
>@@ -16,7 +16,7 @@
> #include <string>
> 
> #include "logging/rtc_event_log/rtc_event_log.h"
>-#include "logging/rtc_event_log/rtc_event_log_parser.h"
>+#include "logging/rtc_event_log/rtc_event_log_parser_new.h"
> #include "modules/rtp_rtcp/source/byte_io.h"
> #include "modules/rtp_rtcp/source/rtp_utility.h"
> #include "rtc_base/checks.h"
>@@ -25,7 +25,7 @@
> 
> namespace {
> 
>-using MediaType = webrtc::ParsedRtcEventLog::MediaType;
>+using MediaType = webrtc::ParsedRtcEventLogNew::MediaType;
> 
> DEFINE_bool(
>     audio,
>@@ -84,8 +84,8 @@ int main(int argc, char* argv[]) {
>       " --help for usage.\n"
>       "Example usage:\n" +
>       program_name + " input.rel output.rtp\n";
>-  if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) ||
>-      FLAG_help || argc != 3) {
>+  if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) || FLAG_help ||
>+      argc != 3) {
>     std::cout << usage;
>     if (FLAG_help) {
>       rtc::FlagList::Print(nullptr, false);
>@@ -102,7 +102,7 @@ int main(int argc, char* argv[]) {
>     RTC_CHECK(ParseSsrc(FLAG_ssrc, &ssrc_filter))
>         << "Flag verification has failed.";
> 
>-  webrtc::ParsedRtcEventLog parsed_stream;
>+  webrtc::ParsedRtcEventLogNew parsed_stream;
>   if (!parsed_stream.ParseFile(input_file)) {
>     std::cerr << "Error while parsing input file: " << input_file << std::endl;
>     return -1;
>@@ -127,8 +127,8 @@ int main(int argc, char* argv[]) {
>     // some required fields and we attempt to access them. We could consider
>     // a softer failure option, but it does not seem useful to generate
>     // RTP dumps based on broken event logs.
>-    if (FLAG_rtp &&
>-        parsed_stream.GetEventType(i) == webrtc::ParsedRtcEventLog::RTP_EVENT) {
>+    if (FLAG_rtp && parsed_stream.GetEventType(i) ==
>+                        webrtc::ParsedRtcEventLogNew::EventType::RTP_EVENT) {
>       webrtc::test::RtpPacket packet;
>       webrtc::PacketDirection direction;
>       parsed_stream.GetRtpHeader(i, &direction, packet.data, &packet.length,
>@@ -166,7 +166,7 @@ int main(int argc, char* argv[]) {
>       rtp_counter++;
>     }
>     if (FLAG_rtcp && parsed_stream.GetEventType(i) ==
>-                         webrtc::ParsedRtcEventLog::RTCP_EVENT) {
>+                         webrtc::ParsedRtcEventLogNew::EventType::RTCP_EVENT) {
>       webrtc::test::RtpPacket packet;
>       webrtc::PacketDirection direction;
>       parsed_stream.GetRtcpPacket(i, &direction, packet.data, &packet.length);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log2stats.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log2stats.cc
>index f38322d5bd2..928c829f49f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log2stats.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log2stats.cc
>@@ -166,6 +166,10 @@ std::string EventTypeToString(webrtc::rtclog::Event::EventType event_type) {
>       return "BWE_PROBE_RESULT";
>     case webrtc::rtclog::Event::ALR_STATE_EVENT:
>       return "ALR_STATE_EVENT";
>+    case webrtc::rtclog::Event::ICE_CANDIDATE_PAIR_CONFIG:
>+      return "ICE_CANDIDATE_PAIR_CONFIG";
>+    case webrtc::rtclog::Event::ICE_CANDIDATE_PAIR_EVENT:
>+      return "ICE_CANDIDATE_PAIR_EVENT";
>   }
>   RTC_NOTREACHED();
>   return "UNKNOWN_EVENT";
>@@ -185,8 +189,8 @@ int main(int argc, char* argv[]) {
>       " --help for usage.\n"
>       "Example usage:\n" +
>       program_name + " input.rel\n";
>-  if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) ||
>-      FLAG_help || argc != 2) {
>+  if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) || FLAG_help ||
>+      argc != 2) {
>     std::cout << usage;
>     if (FLAG_help) {
>       rtc::FlagList::Print(nullptr, false);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log2text.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log2text.cc
>index 42ff7324303..f08e6ece5ec 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log2text.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log2text.cc
>@@ -13,13 +13,11 @@
> #include <iomanip>  // setfill, setw
> #include <iostream>
> #include <map>
>-#include <sstream>
> #include <string>
> #include <utility>  // pair
> 
>-#include "call/video_config.h"
> #include "common_types.h"  // NOLINT(build/include)
>-#include "logging/rtc_event_log/rtc_event_log_parser.h"
>+#include "logging/rtc_event_log/rtc_event_log_parser_new.h"
> #include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
> #include "modules/rtp_rtcp/source/rtcp_packet/bye.h"
> #include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
>@@ -40,6 +38,7 @@
> #include "rtc_base/checks.h"
> #include "rtc_base/flags.h"
> #include "rtc_base/logging.h"
>+#include "rtc_base/strings/string_builder.h"
> 
> namespace {
> 
>@@ -60,6 +59,7 @@ DEFINE_bool(rtcp, true, "Use --nortcp to exclude RTCP packets.");
> DEFINE_bool(playout, true, "Use --noplayout to exclude audio playout events.");
> DEFINE_bool(ana, true, "Use --noana to exclude ANA events.");
> DEFINE_bool(probe, true, "Use --noprobe to exclude probe events.");
>+DEFINE_bool(ice, true, "Use --noice to exclude ICE events.");
> 
> DEFINE_bool(print_full_packets,
>             false,
>@@ -72,7 +72,7 @@ DEFINE_string(ssrc,
>               "starting with 0x).");
> DEFINE_bool(help, false, "Prints this message.");
> 
>-using MediaType = webrtc::ParsedRtcEventLog::MediaType;
>+using MediaType = webrtc::ParsedRtcEventLogNew::MediaType;
> 
> static uint32_t filtered_ssrc = 0;
> 
>@@ -164,7 +164,7 @@ webrtc::RtpHeaderExtensionMap GetDefaultHeaderExtensionMap() {
>   return default_map;
> }
> 
>-void PrintSenderReport(const webrtc::ParsedRtcEventLog& parsed_stream,
>+void PrintSenderReport(const webrtc::ParsedRtcEventLogNew& parsed_stream,
>                        const webrtc::rtcp::CommonHeader& rtcp_block,
>                        uint64_t log_timestamp,
>                        webrtc::PacketDirection direction) {
>@@ -181,7 +181,7 @@ void PrintSenderReport(const webrtc::ParsedRtcEventLog& parsed_stream,
>             << "\ttimestamp=" << sr.rtp_timestamp() << std::endl;
> }
> 
>-void PrintReceiverReport(const webrtc::ParsedRtcEventLog& parsed_stream,
>+void PrintReceiverReport(const webrtc::ParsedRtcEventLogNew& parsed_stream,
>                          const webrtc::rtcp::CommonHeader& rtcp_block,
>                          uint64_t log_timestamp,
>                          webrtc::PacketDirection direction) {
>@@ -197,7 +197,7 @@ void PrintReceiverReport(const webrtc::ParsedRtcEventLog& parsed_stream,
>             << "\tssrc=" << rr.sender_ssrc() << std::endl;
> }
> 
>-void PrintXr(const webrtc::ParsedRtcEventLog& parsed_stream,
>+void PrintXr(const webrtc::ParsedRtcEventLogNew& parsed_stream,
>              const webrtc::rtcp::CommonHeader& rtcp_block,
>              uint64_t log_timestamp,
>              webrtc::PacketDirection direction) {
>@@ -222,7 +222,7 @@ void PrintSdes(const webrtc::rtcp::CommonHeader& rtcp_block,
>   RTC_NOTREACHED() << "SDES should have been redacted when writing the log";
> }
> 
>-void PrintBye(const webrtc::ParsedRtcEventLog& parsed_stream,
>+void PrintBye(const webrtc::ParsedRtcEventLogNew& parsed_stream,
>               const webrtc::rtcp::CommonHeader& rtcp_block,
>               uint64_t log_timestamp,
>               webrtc::PacketDirection direction) {
>@@ -238,7 +238,7 @@ void PrintBye(const webrtc::ParsedRtcEventLog& parsed_stream,
>             << "\tssrc=" << bye.sender_ssrc() << std::endl;
> }
> 
>-void PrintRtpFeedback(const webrtc::ParsedRtcEventLog& parsed_stream,
>+void PrintRtpFeedback(const webrtc::ParsedRtcEventLogNew& parsed_stream,
>                       const webrtc::rtcp::CommonHeader& rtcp_block,
>                       uint64_t log_timestamp,
>                       webrtc::PacketDirection direction) {
>@@ -318,7 +318,7 @@ void PrintRtpFeedback(const webrtc::ParsedRtcEventLog& parsed_stream,
>   }
> }
> 
>-void PrintPsFeedback(const webrtc::ParsedRtcEventLog& parsed_stream,
>+void PrintPsFeedback(const webrtc::ParsedRtcEventLogNew& parsed_stream,
>                      const webrtc::rtcp::CommonHeader& rtcp_block,
>                      uint64_t log_timestamp,
>                      webrtc::PacketDirection direction) {
>@@ -369,30 +369,53 @@ void PrintPsFeedback(const webrtc::ParsedRtcEventLog& parsed_stream,
>   }
> }
> 
>+enum class InputSource {
>+  STDIN,
>+  FILE,
>+};
>+
>+void PrintUsageGuide(const std::string& program_name) {
>+  std::cout
>+      << "Tool for printing packet information from an RtcEventLog as text.\n"
>+      << "* Run " + program_name + " --help for usage.\n"
>+      << "* Example usage for parsing a file:\n"
>+      << "  " << program_name + " input.rel\n"
>+      << "* Example usage for parsing the stdin:\n"
>+      << "  " << program_name + "\n";
>+}
>+
>+// TODO(eladalon): Return a stream or file descriptor instead.
>+InputSource ParseCommandLineFlags(int argc, char* argv[]) {
>+  if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) != 0) {
>+    PrintUsageGuide(argv[0]);
>+    exit(-1);
>+  }
>+
>+  if (FLAG_help) {
>+    PrintUsageGuide(argv[0]);
>+    std::cout << std::endl;
>+    rtc::FlagList::Print(nullptr, false);
>+    exit(0);
>+  }
>+
>+  switch (argc) {
>+    case 1:
>+      return InputSource::STDIN;
>+    case 2:
>+      return InputSource::FILE;
>+    default:
>+      PrintUsageGuide(argv[0]);
>+      exit(-1);
>+  }
>+}
>+
> }  // namespace
> 
> // This utility will print basic information about each packet to stdout.
> // Note that parser will assert if the protobuf event is missing some required
> // fields and we attempt to access them. We don't handle this at the moment.
> int main(int argc, char* argv[]) {
>-  std::string program_name = argv[0];
>-  std::string usage =
>-      "Tool for printing packet information from an RtcEventLog as text.\n"
>-      "Run " +
>-      program_name +
>-      " --help for usage.\n"
>-      "Example usage:\n" +
>-      program_name + " input.rel\n";
>-  if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) ||
>-      FLAG_help || argc != 2) {
>-    std::cout << usage;
>-    if (FLAG_help) {
>-      rtc::FlagList::Print(nullptr, false);
>-      return 0;
>-    }
>-    return 1;
>-  }
>-  std::string input_file = argv[1];
>+  InputSource input_source = ParseCommandLineFlags(argc, argv);
> 
>   if (strlen(FLAG_ssrc) > 0)
>     RTC_CHECK(ParseSsrc(FLAG_ssrc)) << "Flag verification has failed.";
>@@ -400,16 +423,30 @@ int main(int argc, char* argv[]) {
>   webrtc::RtpHeaderExtensionMap default_map = GetDefaultHeaderExtensionMap();
>   bool default_map_used = false;
> 
>-  webrtc::ParsedRtcEventLog parsed_stream;
>-  if (!parsed_stream.ParseFile(input_file)) {
>-    std::cerr << "Error while parsing input file: " << input_file << std::endl;
>-    return -1;
>+  webrtc::ParsedRtcEventLogNew parsed_stream;
>+
>+  switch (input_source) {
>+    case InputSource::STDIN: {
>+      if (!parsed_stream.ParseStream(std::cin)) {
>+        std::cerr << "Error while parsing input stream." << std::endl;
>+        return -1;
>+      }
>+      break;
>+    }
>+    case InputSource::FILE: {
>+      if (!parsed_stream.ParseFile(argv[1])) {
>+        std::cerr << "Error while parsing input file: " << argv[1] << std::endl;
>+        return -1;
>+      }
>+      break;
>+    }
>+    default: { RTC_NOTREACHED() << "Unsupported input source."; }
>   }
> 
>   for (size_t i = 0; i < parsed_stream.GetNumberOfEvents(); i++) {
>     bool event_recognized = false;
>     switch (parsed_stream.GetEventType(i)) {
>-      case webrtc::ParsedRtcEventLog::UNKNOWN_EVENT: {
>+      case webrtc::ParsedRtcEventLogNew::EventType::UNKNOWN_EVENT: {
>         if (FLAG_unknown) {
>           std::cout << parsed_stream.GetTimestamp(i) << "\tUNKNOWN_EVENT"
>                     << std::endl;
>@@ -418,7 +455,7 @@ int main(int argc, char* argv[]) {
>         break;
>       }
> 
>-      case webrtc::ParsedRtcEventLog::LOG_START: {
>+      case webrtc::ParsedRtcEventLogNew::EventType::LOG_START: {
>         if (FLAG_startstop) {
>           std::cout << parsed_stream.GetTimestamp(i) << "\tLOG_START"
>                     << std::endl;
>@@ -427,7 +464,7 @@ int main(int argc, char* argv[]) {
>         break;
>       }
> 
>-      case webrtc::ParsedRtcEventLog::LOG_END: {
>+      case webrtc::ParsedRtcEventLogNew::EventType::LOG_END: {
>         if (FLAG_startstop) {
>           std::cout << parsed_stream.GetTimestamp(i) << "\tLOG_END"
>                     << std::endl;
>@@ -436,13 +473,13 @@ int main(int argc, char* argv[]) {
>         break;
>       }
> 
>-      case webrtc::ParsedRtcEventLog::RTP_EVENT: {
>+      case webrtc::ParsedRtcEventLogNew::EventType::RTP_EVENT: {
>         if (FLAG_rtp) {
>           size_t header_length;
>           size_t total_length;
>           uint8_t header[IP_PACKET_SIZE];
>           webrtc::PacketDirection direction;
>-          webrtc::RtpHeaderExtensionMap* extension_map =
>+          const webrtc::RtpHeaderExtensionMap* extension_map =
>               parsed_stream.GetRtpHeader(i, &direction, header, &header_length,
>                                          &total_length, nullptr);
> 
>@@ -492,8 +529,8 @@ int main(int argc, char* argv[]) {
>                       << parsed_header.extension.transmissionTimeOffset;
>           }
>           if (parsed_header.extension.hasAudioLevel) {
>-            std::cout << "\tAudioLevel=" <<
>-                static_cast<int>(parsed_header.extension.audioLevel);
>+            std::cout << "\tAudioLevel="
>+                      << static_cast<int>(parsed_header.extension.audioLevel);
>           }
>           std::cout << std::endl;
>           if (FLAG_print_full_packets) {
>@@ -513,7 +550,7 @@ int main(int argc, char* argv[]) {
>         break;
>       }
> 
>-      case webrtc::ParsedRtcEventLog::RTCP_EVENT: {
>+      case webrtc::ParsedRtcEventLogNew::EventType::RTCP_EVENT: {
>         if (FLAG_rtcp) {
>           size_t length;
>           uint8_t packet[IP_PACKET_SIZE];
>@@ -580,37 +617,34 @@ int main(int argc, char* argv[]) {
>         break;
>       }
> 
>-      case webrtc::ParsedRtcEventLog::AUDIO_PLAYOUT_EVENT: {
>+      case webrtc::ParsedRtcEventLogNew::EventType::AUDIO_PLAYOUT_EVENT: {
>         if (FLAG_playout) {
>-          uint32_t ssrc;
>-          parsed_stream.GetAudioPlayout(i, &ssrc);
>-          std::cout << parsed_stream.GetTimestamp(i) << "\tAUDIO_PLAYOUT"
>-                    << "\tssrc=" << ssrc << std::endl;
>+          auto audio_playout = parsed_stream.GetAudioPlayout(i);
>+          std::cout << audio_playout.log_time_us() << "\tAUDIO_PLAYOUT"
>+                    << "\tssrc=" << audio_playout.ssrc << std::endl;
>         }
>         event_recognized = true;
>         break;
>       }
> 
>-      case webrtc::ParsedRtcEventLog::LOSS_BASED_BWE_UPDATE: {
>+      case webrtc::ParsedRtcEventLogNew::EventType::LOSS_BASED_BWE_UPDATE: {
>         if (FLAG_bwe) {
>-          int32_t bitrate_bps;
>-          uint8_t fraction_loss;
>-          int32_t total_packets;
>-          parsed_stream.GetLossBasedBweUpdate(i, &bitrate_bps, &fraction_loss,
>-                                              &total_packets);
>-          std::cout << parsed_stream.GetTimestamp(i) << "\tBWE(LOSS_BASED)"
>-                    << "\tbitrate_bps=" << bitrate_bps << "\tfraction_loss="
>-                    << static_cast<unsigned>(fraction_loss)
>-                    << "\ttotal_packets=" << total_packets << std::endl;
>+          auto bwe_update = parsed_stream.GetLossBasedBweUpdate(i);
>+          std::cout << bwe_update.log_time_us() << "\tBWE(LOSS_BASED)"
>+                    << "\tbitrate_bps=" << bwe_update.bitrate_bps
>+                    << "\tfraction_lost="
>+                    << static_cast<unsigned>(bwe_update.fraction_lost)
>+                    << "\texpected_packets=" << bwe_update.expected_packets
>+                    << std::endl;
>         }
>         event_recognized = true;
>         break;
>       }
> 
>-      case webrtc::ParsedRtcEventLog::DELAY_BASED_BWE_UPDATE: {
>+      case webrtc::ParsedRtcEventLogNew::EventType::DELAY_BASED_BWE_UPDATE: {
>         if (FLAG_bwe) {
>           auto bwe_update = parsed_stream.GetDelayBasedBweUpdate(i);
>-          std::cout << parsed_stream.GetTimestamp(i) << "\tBWE(DELAY_BASED)"
>+          std::cout << bwe_update.log_time_us() << "\tBWE(DELAY_BASED)"
>                     << "\tbitrate_bps=" << bwe_update.bitrate_bps
>                     << "\tdetector_state="
>                     << static_cast<int>(bwe_update.detector_state) << std::endl;
>@@ -619,7 +653,8 @@ int main(int argc, char* argv[]) {
>         break;
>       }
> 
>-      case webrtc::ParsedRtcEventLog::VIDEO_RECEIVER_CONFIG_EVENT: {
>+      case webrtc::ParsedRtcEventLogNew::EventType::
>+          VIDEO_RECEIVER_CONFIG_EVENT: {
>         if (FLAG_config && FLAG_video && FLAG_incoming) {
>           webrtc::rtclog::StreamConfig config =
>               parsed_stream.GetVideoReceiveConfig(i);
>@@ -644,7 +679,7 @@ int main(int argc, char* argv[]) {
>         break;
>       }
> 
>-      case webrtc::ParsedRtcEventLog::VIDEO_SENDER_CONFIG_EVENT: {
>+      case webrtc::ParsedRtcEventLogNew::EventType::VIDEO_SENDER_CONFIG_EVENT: {
>         if (FLAG_config && FLAG_video && FLAG_outgoing) {
>           std::vector<webrtc::rtclog::StreamConfig> configs =
>               parsed_stream.GetVideoSendConfig(i);
>@@ -671,7 +706,8 @@ int main(int argc, char* argv[]) {
>         break;
>       }
> 
>-      case webrtc::ParsedRtcEventLog::AUDIO_RECEIVER_CONFIG_EVENT: {
>+      case webrtc::ParsedRtcEventLogNew::EventType::
>+          AUDIO_RECEIVER_CONFIG_EVENT: {
>         if (FLAG_config && FLAG_audio && FLAG_incoming) {
>           webrtc::rtclog::StreamConfig config =
>               parsed_stream.GetAudioReceiveConfig(i);
>@@ -696,7 +732,7 @@ int main(int argc, char* argv[]) {
>         break;
>       }
> 
>-      case webrtc::ParsedRtcEventLog::AUDIO_SENDER_CONFIG_EVENT: {
>+      case webrtc::ParsedRtcEventLogNew::EventType::AUDIO_SENDER_CONFIG_EVENT: {
>         if (FLAG_config && FLAG_audio && FLAG_outgoing) {
>           webrtc::rtclog::StreamConfig config =
>               parsed_stream.GetAudioSendConfig(i);
>@@ -720,41 +756,43 @@ int main(int argc, char* argv[]) {
>         break;
>       }
> 
>-      case webrtc::ParsedRtcEventLog::AUDIO_NETWORK_ADAPTATION_EVENT: {
>+      case webrtc::ParsedRtcEventLogNew::EventType::
>+          AUDIO_NETWORK_ADAPTATION_EVENT: {
>         if (FLAG_ana) {
>-          webrtc::AudioEncoderRuntimeConfig ana_config;
>-          parsed_stream.GetAudioNetworkAdaptation(i, &ana_config);
>-          std::stringstream ss;
>-          ss << parsed_stream.GetTimestamp(i) << "\tANA_UPDATE";
>-          if (ana_config.bitrate_bps) {
>-            ss << "\tbitrate_bps=" << *ana_config.bitrate_bps;
>+          auto ana_event = parsed_stream.GetAudioNetworkAdaptation(i);
>+          char buffer[300];
>+          rtc::SimpleStringBuilder builder(buffer);
>+          builder << parsed_stream.GetTimestamp(i) << "\tANA_UPDATE";
>+          if (ana_event.config.bitrate_bps) {
>+            builder << "\tbitrate_bps=" << *ana_event.config.bitrate_bps;
>           }
>-          if (ana_config.frame_length_ms) {
>-            ss << "\tframe_length_ms=" << *ana_config.frame_length_ms;
>+          if (ana_event.config.frame_length_ms) {
>+            builder << "\tframe_length_ms="
>+                    << *ana_event.config.frame_length_ms;
>           }
>-          if (ana_config.uplink_packet_loss_fraction) {
>-            ss << "\tuplink_packet_loss_fraction="
>-               << *ana_config.uplink_packet_loss_fraction;
>+          if (ana_event.config.uplink_packet_loss_fraction) {
>+            builder << "\tuplink_packet_loss_fraction="
>+                    << *ana_event.config.uplink_packet_loss_fraction;
>           }
>-          if (ana_config.enable_fec) {
>-            ss << "\tenable_fec=" << *ana_config.enable_fec;
>+          if (ana_event.config.enable_fec) {
>+            builder << "\tenable_fec=" << *ana_event.config.enable_fec;
>           }
>-          if (ana_config.enable_dtx) {
>-            ss << "\tenable_dtx=" << *ana_config.enable_dtx;
>+          if (ana_event.config.enable_dtx) {
>+            builder << "\tenable_dtx=" << *ana_event.config.enable_dtx;
>           }
>-          if (ana_config.num_channels) {
>-            ss << "\tnum_channels=" << *ana_config.num_channels;
>+          if (ana_event.config.num_channels) {
>+            builder << "\tnum_channels=" << *ana_event.config.num_channels;
>           }
>-          std::cout << ss.str() << std::endl;
>+          std::cout << builder.str() << std::endl;
>         }
>         event_recognized = true;
>         break;
>       }
> 
>-      case webrtc::ParsedRtcEventLog::BWE_PROBE_CLUSTER_CREATED_EVENT: {
>+      case webrtc::ParsedRtcEventLogNew::EventType::
>+          BWE_PROBE_CLUSTER_CREATED_EVENT: {
>         if (FLAG_probe) {
>-          webrtc::ParsedRtcEventLog::BweProbeClusterCreatedEvent probe_event =
>-              parsed_stream.GetBweProbeClusterCreated(i);
>+          auto probe_event = parsed_stream.GetBweProbeClusterCreated(i);
>           std::cout << parsed_stream.GetTimestamp(i) << "\tPROBE_CREATED("
>                     << probe_event.id << ")"
>                     << "\tbitrate_bps=" << probe_event.bitrate_bps
>@@ -765,42 +803,76 @@ int main(int argc, char* argv[]) {
>         break;
>       }
> 
>-      case webrtc::ParsedRtcEventLog::BWE_PROBE_RESULT_EVENT: {
>+      case webrtc::ParsedRtcEventLogNew::EventType::BWE_PROBE_FAILURE_EVENT: {
>         if (FLAG_probe) {
>-          webrtc::ParsedRtcEventLog::BweProbeResultEvent probe_result =
>-              parsed_stream.GetBweProbeResult(i);
>-          if (probe_result.failure_reason) {
>-            std::cout << parsed_stream.GetTimestamp(i) << "\tPROBE_SUCCESS("
>-                      << probe_result.id << ")"
>-                      << "\tfailure_reason="
>-                      << static_cast<int>(*probe_result.failure_reason)
>-                      << std::endl;
>-          } else {
>-            std::cout << parsed_stream.GetTimestamp(i) << "\tPROBE_SUCCESS("
>-                      << probe_result.id << ")"
>-                      << "\tbitrate_bps=" << *probe_result.bitrate_bps
>-                      << std::endl;
>-          }
>+          webrtc::LoggedBweProbeFailureEvent probe_result =
>+              parsed_stream.GetBweProbeFailure(i);
>+          std::cout << parsed_stream.GetTimestamp(i) << "\tPROBE_FAILURE("
>+                    << probe_result.id << ")"
>+                    << "\tfailure_reason="
>+                    << static_cast<int>(probe_result.failure_reason)
>+                    << std::endl;
>         }
>         event_recognized = true;
>         break;
>       }
> 
>-      case webrtc::ParsedRtcEventLog::ALR_STATE_EVENT: {
>+      case webrtc::ParsedRtcEventLogNew::EventType::BWE_PROBE_SUCCESS_EVENT: {
>+        if (FLAG_probe) {
>+          webrtc::LoggedBweProbeSuccessEvent probe_result =
>+              parsed_stream.GetBweProbeSuccess(i);
>+          std::cout << parsed_stream.GetTimestamp(i) << "\tPROBE_SUCCESS("
>+                    << probe_result.id << ")"
>+                    << "\tbitrate_bps=" << probe_result.bitrate_bps
>+                    << std::endl;
>+        }
>+        event_recognized = true;
>+        break;
>+      }
>+
>+      case webrtc::ParsedRtcEventLogNew::EventType::ALR_STATE_EVENT: {
>         if (FLAG_bwe) {
>-          webrtc::ParsedRtcEventLog::AlrStateEvent alr_state =
>-              parsed_stream.GetAlrState(i);
>+          webrtc::LoggedAlrStateEvent alr_state = parsed_stream.GetAlrState(i);
>           std::cout << parsed_stream.GetTimestamp(i) << "\tALR_STATE"
>                     << "\tin_alr=" << alr_state.in_alr << std::endl;
>         }
>         event_recognized = true;
>         break;
>       }
>+
>+      case webrtc::ParsedRtcEventLogNew::EventType::ICE_CANDIDATE_PAIR_CONFIG: {
>+        if (FLAG_ice) {
>+          webrtc::LoggedIceCandidatePairConfig ice_cp_config =
>+              parsed_stream.GetIceCandidatePairConfig(i);
>+          // TODO(qingsi): convert the numeric representation of states to text
>+          std::cout << parsed_stream.GetTimestamp(i)
>+                    << "\tICE_CANDIDATE_PAIR_CONFIG"
>+                    << "\ttype=" << static_cast<int>(ice_cp_config.type)
>+                    << std::endl;
>+        }
>+        event_recognized = true;
>+        break;
>+      }
>+
>+      case webrtc::ParsedRtcEventLogNew::EventType::ICE_CANDIDATE_PAIR_EVENT: {
>+        if (FLAG_ice) {
>+          webrtc::LoggedIceCandidatePairEvent ice_cp_event =
>+              parsed_stream.GetIceCandidatePairEvent(i);
>+          // TODO(qingsi): convert the numeric representation of states to text
>+          std::cout << parsed_stream.GetTimestamp(i)
>+                    << "\tICE_CANDIDATE_PAIR_EVENT"
>+                    << "\ttype=" << static_cast<int>(ice_cp_event.type)
>+                    << std::endl;
>+        }
>+        event_recognized = true;
>+        break;
>+      }
>     }
> 
>     if (!event_recognized) {
>-      std::cout << "Unrecognized event (" << parsed_stream.GetEventType(i)
>-                << ")" << std::endl;
>+      std::cout << "Unrecognized event ("
>+                << static_cast<int>(parsed_stream.GetEventType(i)) << ")"
>+                << std::endl;
>     }
>   }
>   return 0;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_factory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_factory.cc
>index 566736c8b70..e9d983d96e4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_factory.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_factory.cc
>@@ -10,6 +10,8 @@
> 
> #include "logging/rtc_event_log/rtc_event_log_factory.h"
> 
>+#include <utility>
>+
> #include "logging/rtc_event_log/rtc_event_log.h"
> 
> namespace webrtc {
>@@ -19,6 +21,12 @@ std::unique_ptr<RtcEventLog> RtcEventLogFactory::CreateRtcEventLog(
>   return RtcEventLog::Create(encoding_type);
> }
> 
>+std::unique_ptr<RtcEventLog> RtcEventLogFactory::CreateRtcEventLog(
>+    RtcEventLog::EncodingType encoding_type,
>+    std::unique_ptr<rtc::TaskQueue> task_queue) {
>+  return RtcEventLog::Create(encoding_type, std::move(task_queue));
>+}
>+
> std::unique_ptr<RtcEventLogFactoryInterface> CreateRtcEventLogFactory() {
>   return std::unique_ptr<RtcEventLogFactoryInterface>(new RtcEventLogFactory());
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_factory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_factory.h
>index 1b75d66c10d..7f61ca92a79 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_factory.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_factory.h
>@@ -14,6 +14,7 @@
> #include <memory>
> 
> #include "logging/rtc_event_log/rtc_event_log_factory_interface.h"
>+#include "rtc_base/task_queue.h"
> 
> namespace webrtc {
> 
>@@ -23,6 +24,10 @@ class RtcEventLogFactory : public RtcEventLogFactoryInterface {
> 
>   std::unique_ptr<RtcEventLog> CreateRtcEventLog(
>       RtcEventLog::EncodingType encoding_type) override;
>+
>+  std::unique_ptr<RtcEventLog> CreateRtcEventLog(
>+      RtcEventLog::EncodingType encoding_type,
>+      std::unique_ptr<rtc::TaskQueue> task_queue) override;
> };
> 
> std::unique_ptr<RtcEventLogFactoryInterface> CreateRtcEventLogFactory();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_factory_interface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_factory_interface.h
>index a28002d7827..395f2dc9a0f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_factory_interface.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_factory_interface.h
>@@ -14,6 +14,7 @@
> #include <memory>
> 
> #include "logging/rtc_event_log/rtc_event_log.h"
>+#include "rtc_base/task_queue.h"
> 
> namespace webrtc {
> 
>@@ -26,6 +27,10 @@ class RtcEventLogFactoryInterface {
> 
>   virtual std::unique_ptr<RtcEventLog> CreateRtcEventLog(
>       RtcEventLog::EncodingType encoding_type) = 0;
>+
>+  virtual std::unique_ptr<RtcEventLog> CreateRtcEventLog(
>+      RtcEventLog::EncodingType encoding_type,
>+      std::unique_ptr<rtc::TaskQueue> task_queue) = 0;
> };
> 
> std::unique_ptr<RtcEventLogFactoryInterface> CreateRtcEventLogFactory();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_impl.cc
>new file mode 100644
>index 00000000000..554a0283fd6
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_impl.cc
>@@ -0,0 +1,379 @@
>+/*
>+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "logging/rtc_event_log/rtc_event_log.h"
>+
>+#include <deque>
>+#include <functional>
>+#include <limits>
>+#include <memory>
>+#include <utility>
>+#include <vector>
>+
>+#include "absl/memory/memory.h"
>+#include "logging/rtc_event_log/encoder/rtc_event_log_encoder_legacy.h"
>+#include "logging/rtc_event_log/output/rtc_event_log_output_file.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/constructormagic.h"
>+#include "rtc_base/event.h"
>+#include "rtc_base/logging.h"
>+#include "rtc_base/numerics/safe_conversions.h"
>+#include "rtc_base/numerics/safe_minmax.h"
>+#include "rtc_base/sequenced_task_checker.h"
>+#include "rtc_base/task_queue.h"
>+#include "rtc_base/thread_annotations.h"
>+
>+namespace webrtc {
>+
>+#ifdef ENABLE_RTC_EVENT_LOG
>+
>+namespace {
>+constexpr size_t kMaxEventsInHistory = 10000;
>+// The config-history is supposed to be unbounded, but needs to have some bound
>+// to prevent an attack via unreasonable memory use.
>+constexpr size_t kMaxEventsInConfigHistory = 1000;
>+
>+// TODO(eladalon): This class exists because C++11 doesn't allow transferring a
>+// unique_ptr to a lambda (a copy constructor is required). We should get
>+// rid of this when we move to C++14.
>+template <typename T>
>+class ResourceOwningTask final : public rtc::QueuedTask {
>+ public:
>+  ResourceOwningTask(std::unique_ptr<T> resource,
>+                     std::function<void(std::unique_ptr<T>)> handler)
>+      : resource_(std::move(resource)), handler_(handler) {}
>+
>+  bool Run() override {
>+    handler_(std::move(resource_));
>+    return true;
>+  }
>+
>+ private:
>+  std::unique_ptr<T> resource_;
>+  std::function<void(std::unique_ptr<T>)> handler_;
>+};
>+
>+std::unique_ptr<RtcEventLogEncoder> CreateEncoder(
>+    RtcEventLog::EncodingType type) {
>+  switch (type) {
>+    case RtcEventLog::EncodingType::Legacy:
>+      return absl::make_unique<RtcEventLogEncoderLegacy>();
>+    default:
>+      RTC_LOG(LS_ERROR) << "Unknown RtcEventLog encoder type (" << int(type)
>+                        << ")";
>+      RTC_NOTREACHED();
>+      return std::unique_ptr<RtcEventLogEncoder>(nullptr);
>+  }
>+}
>+
>+class RtcEventLogImpl final : public RtcEventLog {
>+ public:
>+  RtcEventLogImpl(std::unique_ptr<RtcEventLogEncoder> event_encoder,
>+                  std::unique_ptr<rtc::TaskQueue> task_queue);
>+
>+  ~RtcEventLogImpl() override;
>+
>+  // TODO(eladalon): We should change these name to reflect that what we're
>+  // actually starting/stopping is the output of the log, not the log itself.
>+  bool StartLogging(std::unique_ptr<RtcEventLogOutput> output,
>+                    int64_t output_period_ms) override;
>+  void StopLogging() override;
>+
>+  void Log(std::unique_ptr<RtcEvent> event) override;
>+
>+ private:
>+  void LogToMemory(std::unique_ptr<RtcEvent> event) RTC_RUN_ON(task_queue_);
>+  void LogEventsFromMemoryToOutput() RTC_RUN_ON(task_queue_);
>+
>+  void StopOutput() RTC_RUN_ON(task_queue_);
>+
>+  void WriteConfigsAndHistoryToOutput(const std::string& encoded_configs,
>+                                      const std::string& encoded_history)
>+      RTC_RUN_ON(task_queue_);
>+  void WriteToOutput(const std::string& output_string) RTC_RUN_ON(task_queue_);
>+
>+  void StopLoggingInternal() RTC_RUN_ON(task_queue_);
>+
>+  void ScheduleOutput() RTC_RUN_ON(task_queue_);
>+
>+  // Make sure that the event log is "managed" - created/destroyed, as well
>+  // as started/stopped - from the same thread/task-queue.
>+  rtc::SequencedTaskChecker owner_sequence_checker_;
>+
>+  // History containing all past configuration events.
>+  std::deque<std::unique_ptr<RtcEvent>> config_history_
>+      RTC_GUARDED_BY(*task_queue_);
>+
>+  // History containing the most recent (non-configuration) events (~10s).
>+  std::deque<std::unique_ptr<RtcEvent>> history_ RTC_GUARDED_BY(*task_queue_);
>+
>+  size_t max_size_bytes_ RTC_GUARDED_BY(*task_queue_);
>+  size_t written_bytes_ RTC_GUARDED_BY(*task_queue_);
>+
>+  std::unique_ptr<RtcEventLogEncoder> event_encoder_
>+      RTC_GUARDED_BY(*task_queue_);
>+  std::unique_ptr<RtcEventLogOutput> event_output_ RTC_GUARDED_BY(*task_queue_);
>+
>+  size_t num_config_events_written_ RTC_GUARDED_BY(*task_queue_);
>+  int64_t output_period_ms_ RTC_GUARDED_BY(*task_queue_);
>+  int64_t last_output_ms_ RTC_GUARDED_BY(*task_queue_);
>+  bool output_scheduled_ RTC_GUARDED_BY(*task_queue_);
>+
>+  // Since we are posting tasks bound to |this|,  it is critical that the event
>+  // log and it's members outlive the |task_queue_|. Keep the "task_queue_|
>+  // last to ensure it destructs first, or else tasks living on the queue might
>+  // access other members after they've been torn down.
>+  std::unique_ptr<rtc::TaskQueue> task_queue_;
>+
>+  RTC_DISALLOW_COPY_AND_ASSIGN(RtcEventLogImpl);
>+};
>+
>+RtcEventLogImpl::RtcEventLogImpl(
>+    std::unique_ptr<RtcEventLogEncoder> event_encoder,
>+    std::unique_ptr<rtc::TaskQueue> task_queue)
>+    : max_size_bytes_(std::numeric_limits<decltype(max_size_bytes_)>::max()),
>+      written_bytes_(0),
>+      event_encoder_(std::move(event_encoder)),
>+      num_config_events_written_(0),
>+      output_period_ms_(kImmediateOutput),
>+      last_output_ms_(rtc::TimeMillis()),
>+      output_scheduled_(false),
>+      task_queue_(std::move(task_queue)) {
>+  RTC_DCHECK(task_queue_);
>+}
>+
>+RtcEventLogImpl::~RtcEventLogImpl() {
>+  RTC_DCHECK_CALLED_SEQUENTIALLY(&owner_sequence_checker_);
>+
>+  // If we're logging to the output, this will stop that. Blocking function.
>+  StopLogging();
>+}
>+
>+bool RtcEventLogImpl::StartLogging(std::unique_ptr<RtcEventLogOutput> output,
>+                                   int64_t output_period_ms) {
>+  RTC_DCHECK_CALLED_SEQUENTIALLY(&owner_sequence_checker_);
>+
>+  RTC_DCHECK(output_period_ms == kImmediateOutput || output_period_ms > 0);
>+
>+  if (!output->IsActive()) {
>+    // TODO(eladalon): We may want to remove the IsActive method. Otherwise
>+    // we probably want to be consistent and terminate any existing output.
>+    return false;
>+  }
>+
>+  // TODO(terelius): The mapping between log timestamps and UTC should be stored
>+  // in the event_log START event.
>+  const int64_t timestamp_us = rtc::TimeMicros();
>+  const int64_t utc_time_us = rtc::TimeUTCMicros();
>+  RTC_LOG(LS_INFO) << "Starting WebRTC event log. (Timestamp, UTC) = "
>+                   << "(" << timestamp_us << ", " << utc_time_us << ").";
>+
>+  // Binding to |this| is safe because |this| outlives the |task_queue_|.
>+  auto start = [this, timestamp_us](std::unique_ptr<RtcEventLogOutput> output) {
>+    RTC_DCHECK_RUN_ON(task_queue_.get());
>+    RTC_DCHECK(output->IsActive());
>+    event_output_ = std::move(output);
>+    num_config_events_written_ = 0;
>+    WriteToOutput(event_encoder_->EncodeLogStart(timestamp_us));
>+    LogEventsFromMemoryToOutput();
>+  };
>+
>+  task_queue_->PostTask(
>+      absl::make_unique<ResourceOwningTask<RtcEventLogOutput>>(
>+          std::move(output), start));
>+
>+  return true;
>+}
>+
>+void RtcEventLogImpl::StopLogging() {
>+  RTC_DCHECK_CALLED_SEQUENTIALLY(&owner_sequence_checker_);
>+
>+  RTC_LOG(LS_INFO) << "Stopping WebRTC event log.";
>+
>+  rtc::Event output_stopped(true, false);
>+
>+  // Binding to |this| is safe because |this| outlives the |task_queue_|.
>+  task_queue_->PostTask([this, &output_stopped]() {
>+    RTC_DCHECK_RUN_ON(task_queue_.get());
>+    if (event_output_) {
>+      RTC_DCHECK(event_output_->IsActive());
>+      LogEventsFromMemoryToOutput();
>+    }
>+    StopLoggingInternal();
>+    output_stopped.Set();
>+  });
>+
>+  output_stopped.Wait(rtc::Event::kForever);
>+
>+  RTC_LOG(LS_INFO) << "WebRTC event log successfully stopped.";
>+}
>+
>+void RtcEventLogImpl::Log(std::unique_ptr<RtcEvent> event) {
>+  RTC_CHECK(event);
>+
>+  // Binding to |this| is safe because |this| outlives the |task_queue_|.
>+  auto event_handler = [this](std::unique_ptr<RtcEvent> unencoded_event) {
>+    RTC_DCHECK_RUN_ON(task_queue_.get());
>+    LogToMemory(std::move(unencoded_event));
>+    if (event_output_)
>+      ScheduleOutput();
>+  };
>+
>+  task_queue_->PostTask(absl::make_unique<ResourceOwningTask<RtcEvent>>(
>+      std::move(event), event_handler));
>+}
>+
>+void RtcEventLogImpl::ScheduleOutput() {
>+  RTC_DCHECK(event_output_ && event_output_->IsActive());
>+  if (history_.size() >= kMaxEventsInHistory) {
>+    // We have to emergency drain the buffer. We can't wait for the scheduled
>+    // output task because there might be other event incoming before that.
>+    LogEventsFromMemoryToOutput();
>+    return;
>+  }
>+
>+  if (output_period_ms_ == kImmediateOutput) {
>+    // We are already on the |task_queue_| so there is no reason to post a task
>+    // if we want to output immediately.
>+    LogEventsFromMemoryToOutput();
>+    return;
>+  }
>+
>+  if (!output_scheduled_) {
>+    output_scheduled_ = true;
>+    // Binding to |this| is safe because |this| outlives the |task_queue_|.
>+    auto output_task = [this]() {
>+      RTC_DCHECK_RUN_ON(task_queue_.get());
>+      if (event_output_) {
>+        RTC_DCHECK(event_output_->IsActive());
>+        LogEventsFromMemoryToOutput();
>+      }
>+      output_scheduled_ = false;
>+    };
>+    int64_t now_ms = rtc::TimeMillis();
>+    int64_t time_since_output_ms = now_ms - last_output_ms_;
>+    uint32_t delay = rtc::SafeClamp(output_period_ms_ - time_since_output_ms, 0,
>+                                    output_period_ms_);
>+    task_queue_->PostDelayedTask(output_task, delay);
>+  }
>+}
>+
>+void RtcEventLogImpl::LogToMemory(std::unique_ptr<RtcEvent> event) {
>+  std::deque<std::unique_ptr<RtcEvent>>& container =
>+      event->IsConfigEvent() ? config_history_ : history_;
>+  const size_t container_max_size =
>+      event->IsConfigEvent() ? kMaxEventsInConfigHistory : kMaxEventsInHistory;
>+
>+  if (container.size() >= container_max_size) {
>+    RTC_DCHECK(!event_output_);  // Shouldn't lose events if we have an output.
>+    container.pop_front();
>+  }
>+  container.push_back(std::move(event));
>+}
>+
>+void RtcEventLogImpl::LogEventsFromMemoryToOutput() {
>+  RTC_DCHECK(event_output_ && event_output_->IsActive());
>+  last_output_ms_ = rtc::TimeMillis();
>+
>+  // Serialize all stream configurations that haven't already been written to
>+  // this output. |num_config_events_written_| is used to track which configs we
>+  // have already written. (Note that the config may have been written to
>+  // previous outputs; configs are not discarded.)
>+  std::string encoded_configs;
>+  RTC_DCHECK_LE(num_config_events_written_, config_history_.size());
>+  if (num_config_events_written_ < config_history_.size()) {
>+    const auto begin = config_history_.begin() + num_config_events_written_;
>+    const auto end = config_history_.end();
>+    encoded_configs = event_encoder_->EncodeBatch(begin, end);
>+    num_config_events_written_ = config_history_.size();
>+  }
>+
>+  // Serialize the events in the event queue. Note that the write may fail,
>+  // for example if we are writing to a file and have reached the maximum limit.
>+  // We don't get any feedback if this happens, so we still remove the events
>+  // from the event log history. This is normally not a problem, but if another
>+  // log is started immediately after the first one becomes full, then one
>+  // cannot rely on the second log to contain everything that isn't in the first
>+  // log; one batch of events might be missing.
>+  std::string encoded_history =
>+      event_encoder_->EncodeBatch(history_.begin(), history_.end());
>+  history_.clear();
>+
>+  WriteConfigsAndHistoryToOutput(encoded_configs, encoded_history);
>+}
>+
>+void RtcEventLogImpl::WriteConfigsAndHistoryToOutput(
>+    const std::string& encoded_configs,
>+    const std::string& encoded_history) {
>+  // This function is used to merge the strings instead of calling the output
>+  // object twice with small strings. The function also avoids copying any
>+  // strings in the typical case where there are no config events.
>+  if (encoded_configs.size() == 0) {
>+    WriteToOutput(encoded_history);  // Typical case.
>+  } else if (encoded_history.size() == 0) {
>+    WriteToOutput(encoded_configs);  // Very unusual case.
>+  } else {
>+    WriteToOutput(encoded_configs + encoded_history);
>+  }
>+}
>+
>+void RtcEventLogImpl::StopOutput() {
>+  max_size_bytes_ = std::numeric_limits<decltype(max_size_bytes_)>::max();
>+  written_bytes_ = 0;
>+  event_output_.reset();
>+}
>+
>+void RtcEventLogImpl::StopLoggingInternal() {
>+  if (event_output_) {
>+    RTC_DCHECK(event_output_->IsActive());
>+    const int64_t timestamp_us = rtc::TimeMicros();
>+    event_output_->Write(event_encoder_->EncodeLogEnd(timestamp_us));
>+  }
>+  StopOutput();
>+}
>+
>+void RtcEventLogImpl::WriteToOutput(const std::string& output_string) {
>+  RTC_DCHECK(event_output_ && event_output_->IsActive());
>+  if (!event_output_->Write(output_string)) {
>+    RTC_LOG(LS_ERROR) << "Failed to write RTC event to output.";
>+    // The first failure closes the output.
>+    RTC_DCHECK(!event_output_->IsActive());
>+    StopOutput();  // Clean-up.
>+    return;
>+  }
>+  written_bytes_ += output_string.size();
>+}
>+
>+}  // namespace
>+
>+#endif  // ENABLE_RTC_EVENT_LOG
>+
>+// RtcEventLog member functions.
>+std::unique_ptr<RtcEventLog> RtcEventLog::Create(EncodingType encoding_type) {
>+  return Create(encoding_type,
>+                absl::make_unique<rtc::TaskQueue>("rtc_event_log"));
>+}
>+
>+std::unique_ptr<RtcEventLog> RtcEventLog::Create(
>+    EncodingType encoding_type,
>+    std::unique_ptr<rtc::TaskQueue> task_queue) {
>+#ifdef ENABLE_RTC_EVENT_LOG
>+  return absl::make_unique<RtcEventLogImpl>(CreateEncoder(encoding_type),
>+                                            std::move(task_queue));
>+#else
>+  return CreateNull();
>+#endif  // ENABLE_RTC_EVENT_LOG
>+}
>+
>+std::unique_ptr<RtcEventLog> RtcEventLog::CreateNull() {
>+  return std::unique_ptr<RtcEventLog>(new RtcEventLogNullImpl());
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc
>index 194da57b6d1..27bb2b26d65 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc
>@@ -76,6 +76,10 @@ ParsedRtcEventLog::EventType GetRuntimeEventType(
>       return ParsedRtcEventLog::EventType::BWE_PROBE_RESULT_EVENT;
>     case rtclog::Event::ALR_STATE_EVENT:
>       return ParsedRtcEventLog::EventType::ALR_STATE_EVENT;
>+    case rtclog::Event::ICE_CANDIDATE_PAIR_CONFIG:
>+      return ParsedRtcEventLog::EventType::ICE_CANDIDATE_PAIR_CONFIG;
>+    case rtclog::Event::ICE_CANDIDATE_PAIR_EVENT:
>+      return ParsedRtcEventLog::EventType::ICE_CANDIDATE_PAIR_EVENT;
>   }
>   return ParsedRtcEventLog::EventType::UNKNOWN_EVENT;
> }
>@@ -94,6 +98,108 @@ BandwidthUsage GetRuntimeDetectorState(
>   return BandwidthUsage::kBwNormal;
> }
> 
>+IceCandidatePairConfigType GetRuntimeIceCandidatePairConfigType(
>+    rtclog::IceCandidatePairConfig::IceCandidatePairConfigType type) {
>+  switch (type) {
>+    case rtclog::IceCandidatePairConfig::ADDED:
>+      return IceCandidatePairConfigType::kAdded;
>+    case rtclog::IceCandidatePairConfig::UPDATED:
>+      return IceCandidatePairConfigType::kUpdated;
>+    case rtclog::IceCandidatePairConfig::DESTROYED:
>+      return IceCandidatePairConfigType::kDestroyed;
>+    case rtclog::IceCandidatePairConfig::SELECTED:
>+      return IceCandidatePairConfigType::kSelected;
>+  }
>+  RTC_NOTREACHED();
>+  return IceCandidatePairConfigType::kAdded;
>+}
>+
>+IceCandidateType GetRuntimeIceCandidateType(
>+    rtclog::IceCandidatePairConfig::IceCandidateType type) {
>+  switch (type) {
>+    case rtclog::IceCandidatePairConfig::LOCAL:
>+      return IceCandidateType::kLocal;
>+    case rtclog::IceCandidatePairConfig::STUN:
>+      return IceCandidateType::kStun;
>+    case rtclog::IceCandidatePairConfig::PRFLX:
>+      return IceCandidateType::kPrflx;
>+    case rtclog::IceCandidatePairConfig::RELAY:
>+      return IceCandidateType::kRelay;
>+    case rtclog::IceCandidatePairConfig::UNKNOWN_CANDIDATE_TYPE:
>+      return IceCandidateType::kUnknown;
>+  }
>+  RTC_NOTREACHED();
>+  return IceCandidateType::kUnknown;
>+}
>+
>+IceCandidatePairProtocol GetRuntimeIceCandidatePairProtocol(
>+    rtclog::IceCandidatePairConfig::Protocol protocol) {
>+  switch (protocol) {
>+    case rtclog::IceCandidatePairConfig::UDP:
>+      return IceCandidatePairProtocol::kUdp;
>+    case rtclog::IceCandidatePairConfig::TCP:
>+      return IceCandidatePairProtocol::kTcp;
>+    case rtclog::IceCandidatePairConfig::SSLTCP:
>+      return IceCandidatePairProtocol::kSsltcp;
>+    case rtclog::IceCandidatePairConfig::TLS:
>+      return IceCandidatePairProtocol::kTls;
>+    case rtclog::IceCandidatePairConfig::UNKNOWN_PROTOCOL:
>+      return IceCandidatePairProtocol::kUnknown;
>+  }
>+  RTC_NOTREACHED();
>+  return IceCandidatePairProtocol::kUnknown;
>+}
>+
>+IceCandidatePairAddressFamily GetRuntimeIceCandidatePairAddressFamily(
>+    rtclog::IceCandidatePairConfig::AddressFamily address_family) {
>+  switch (address_family) {
>+    case rtclog::IceCandidatePairConfig::IPV4:
>+      return IceCandidatePairAddressFamily::kIpv4;
>+    case rtclog::IceCandidatePairConfig::IPV6:
>+      return IceCandidatePairAddressFamily::kIpv6;
>+    case rtclog::IceCandidatePairConfig::UNKNOWN_ADDRESS_FAMILY:
>+      return IceCandidatePairAddressFamily::kUnknown;
>+  }
>+  RTC_NOTREACHED();
>+  return IceCandidatePairAddressFamily::kUnknown;
>+}
>+
>+IceCandidateNetworkType GetRuntimeIceCandidateNetworkType(
>+    rtclog::IceCandidatePairConfig::NetworkType network_type) {
>+  switch (network_type) {
>+    case rtclog::IceCandidatePairConfig::ETHERNET:
>+      return IceCandidateNetworkType::kEthernet;
>+    case rtclog::IceCandidatePairConfig::LOOPBACK:
>+      return IceCandidateNetworkType::kLoopback;
>+    case rtclog::IceCandidatePairConfig::WIFI:
>+      return IceCandidateNetworkType::kWifi;
>+    case rtclog::IceCandidatePairConfig::VPN:
>+      return IceCandidateNetworkType::kVpn;
>+    case rtclog::IceCandidatePairConfig::CELLULAR:
>+      return IceCandidateNetworkType::kCellular;
>+    case rtclog::IceCandidatePairConfig::UNKNOWN_NETWORK_TYPE:
>+      return IceCandidateNetworkType::kUnknown;
>+  }
>+  RTC_NOTREACHED();
>+  return IceCandidateNetworkType::kUnknown;
>+}
>+
>+IceCandidatePairEventType GetRuntimeIceCandidatePairEventType(
>+    rtclog::IceCandidatePairEvent::IceCandidatePairEventType type) {
>+  switch (type) {
>+    case rtclog::IceCandidatePairEvent::CHECK_SENT:
>+      return IceCandidatePairEventType::kCheckSent;
>+    case rtclog::IceCandidatePairEvent::CHECK_RECEIVED:
>+      return IceCandidatePairEventType::kCheckReceived;
>+    case rtclog::IceCandidatePairEvent::CHECK_RESPONSE_SENT:
>+      return IceCandidatePairEventType::kCheckResponseSent;
>+    case rtclog::IceCandidatePairEvent::CHECK_RESPONSE_RECEIVED:
>+      return IceCandidatePairEventType::kCheckResponseReceived;
>+  }
>+  RTC_NOTREACHED();
>+  return IceCandidatePairEventType::kCheckSent;
>+}
>+
> std::pair<uint64_t, bool> ParseVarInt(std::istream& stream) {
>   uint64_t varint = 0;
>   for (size_t bytes_read = 0; bytes_read < 10; ++bytes_read) {
>@@ -115,10 +221,9 @@ std::pair<uint64_t, bool> ParseVarInt(std::istream& stream) {
>   return std::make_pair(varint, false);
> }
> 
>-void GetHeaderExtensions(
>-    std::vector<RtpExtension>* header_extensions,
>-    const RepeatedPtrField<rtclog::RtpHeaderExtension>&
>-    proto_header_extensions) {
>+void GetHeaderExtensions(std::vector<RtpExtension>* header_extensions,
>+                         const RepeatedPtrField<rtclog::RtpHeaderExtension>&
>+                             proto_header_extensions) {
>   header_extensions->clear();
>   for (auto& p : proto_header_extensions) {
>     RTC_CHECK(p.has_name());
>@@ -670,6 +775,61 @@ ParsedRtcEventLog::AlrStateEvent ParsedRtcEventLog::GetAlrState(
>   return res;
> }
> 
>+ParsedRtcEventLog::IceCandidatePairConfig
>+ParsedRtcEventLog::GetIceCandidatePairConfig(size_t index) const {
>+  RTC_CHECK_LT(index, GetNumberOfEvents());
>+  const rtclog::Event& rtc_event = events_[index];
>+  RTC_CHECK(rtc_event.has_type());
>+  RTC_CHECK_EQ(rtc_event.type(), rtclog::Event::ICE_CANDIDATE_PAIR_CONFIG);
>+  IceCandidatePairConfig res;
>+  const rtclog::IceCandidatePairConfig& config =
>+      rtc_event.ice_candidate_pair_config();
>+  res.timestamp = GetTimestamp(index);
>+  RTC_CHECK(config.has_config_type());
>+  res.type = GetRuntimeIceCandidatePairConfigType(config.config_type());
>+  RTC_CHECK(config.has_candidate_pair_id());
>+  res.candidate_pair_id = config.candidate_pair_id();
>+  RTC_CHECK(config.has_local_candidate_type());
>+  res.local_candidate_type =
>+      GetRuntimeIceCandidateType(config.local_candidate_type());
>+  RTC_CHECK(config.has_local_relay_protocol());
>+  res.local_relay_protocol =
>+      GetRuntimeIceCandidatePairProtocol(config.local_relay_protocol());
>+  RTC_CHECK(config.has_local_network_type());
>+  res.local_network_type =
>+      GetRuntimeIceCandidateNetworkType(config.local_network_type());
>+  RTC_CHECK(config.has_local_address_family());
>+  res.local_address_family =
>+      GetRuntimeIceCandidatePairAddressFamily(config.local_address_family());
>+  RTC_CHECK(config.has_remote_candidate_type());
>+  res.remote_candidate_type =
>+      GetRuntimeIceCandidateType(config.remote_candidate_type());
>+  RTC_CHECK(config.has_remote_address_family());
>+  res.remote_address_family =
>+      GetRuntimeIceCandidatePairAddressFamily(config.remote_address_family());
>+  RTC_CHECK(config.has_candidate_pair_protocol());
>+  res.candidate_pair_protocol =
>+      GetRuntimeIceCandidatePairProtocol(config.candidate_pair_protocol());
>+  return res;
>+}
>+
>+ParsedRtcEventLog::IceCandidatePairEvent
>+ParsedRtcEventLog::GetIceCandidatePairEvent(size_t index) const {
>+  RTC_CHECK_LT(index, GetNumberOfEvents());
>+  const rtclog::Event& rtc_event = events_[index];
>+  RTC_CHECK(rtc_event.has_type());
>+  RTC_CHECK_EQ(rtc_event.type(), rtclog::Event::ICE_CANDIDATE_PAIR_EVENT);
>+  IceCandidatePairEvent res;
>+  const rtclog::IceCandidatePairEvent& event =
>+      rtc_event.ice_candidate_pair_event();
>+  res.timestamp = GetTimestamp(index);
>+  RTC_CHECK(event.has_event_type());
>+  res.type = GetRuntimeIceCandidatePairEventType(event.event_type());
>+  RTC_CHECK(event.has_candidate_pair_id());
>+  res.candidate_pair_id = event.candidate_pair_id();
>+  return res;
>+}
>+
> // Returns the MediaType for registered SSRCs. Search from the end to use last
> // registered types first.
> ParsedRtcEventLog::MediaType ParsedRtcEventLog::GetMediaType(
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_parser.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_parser.h
>index 64389187748..862a48d42c4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_parser.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_parser.h
>@@ -17,6 +17,8 @@
> 
> #include "call/video_receive_stream.h"
> #include "call/video_send_stream.h"
>+#include "logging/rtc_event_log/events/rtc_event_ice_candidate_pair.h"
>+#include "logging/rtc_event_log/events/rtc_event_ice_candidate_pair_config.h"
> #include "logging/rtc_event_log/events/rtc_event_probe_result_failure.h"
> #include "logging/rtc_event_log/rtc_event_log.h"
> #include "logging/rtc_event_log/rtc_stream_config.h"
>@@ -55,8 +57,8 @@ class ParsedRtcEventLog {
>   struct BweProbeResultEvent {
>     uint64_t timestamp;
>     uint32_t id;
>-    rtc::Optional<uint64_t> bitrate_bps;
>-    rtc::Optional<ProbeFailureReason> failure_reason;
>+    absl::optional<uint64_t> bitrate_bps;
>+    absl::optional<ProbeFailureReason> failure_reason;
>   };
> 
>   struct BweDelayBasedUpdate {
>@@ -70,6 +72,25 @@ class ParsedRtcEventLog {
>     bool in_alr;
>   };
> 
>+  struct IceCandidatePairConfig {
>+    uint64_t timestamp;
>+    IceCandidatePairConfigType type;
>+    uint32_t candidate_pair_id;
>+    IceCandidateType local_candidate_type;
>+    IceCandidatePairProtocol local_relay_protocol;
>+    IceCandidateNetworkType local_network_type;
>+    IceCandidatePairAddressFamily local_address_family;
>+    IceCandidateType remote_candidate_type;
>+    IceCandidatePairAddressFamily remote_address_family;
>+    IceCandidatePairProtocol candidate_pair_protocol;
>+  };
>+
>+  struct IceCandidatePairEvent {
>+    uint64_t timestamp;
>+    IceCandidatePairEventType type;
>+    uint32_t candidate_pair_id;
>+  };
>+
>   enum EventType {
>     UNKNOWN_EVENT = 0,
>     LOG_START = 1,
>@@ -86,7 +107,9 @@ class ParsedRtcEventLog {
>     AUDIO_NETWORK_ADAPTATION_EVENT = 16,
>     BWE_PROBE_CLUSTER_CREATED_EVENT = 17,
>     BWE_PROBE_RESULT_EVENT = 18,
>-    ALR_STATE_EVENT = 19
>+    ALR_STATE_EVENT = 19,
>+    ICE_CANDIDATE_PAIR_CONFIG = 20,
>+    ICE_CANDIDATE_PAIR_EVENT = 21,
>   };
> 
>   enum class MediaType { ANY, AUDIO, VIDEO, DATA };
>@@ -188,6 +211,9 @@ class ParsedRtcEventLog {
> 
>   AlrStateEvent GetAlrState(size_t index) const;
> 
>+  IceCandidatePairConfig GetIceCandidatePairConfig(size_t index) const;
>+  IceCandidatePairEvent GetIceCandidatePairEvent(size_t index) const;
>+
>  private:
>   rtclog::StreamConfig GetVideoReceiveConfig(const rtclog::Event& event) const;
>   std::vector<rtclog::StreamConfig> GetVideoSendConfig(
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_parser_new.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_parser_new.cc
>new file mode 100644
>index 00000000000..b2b32642828
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_parser_new.cc
>@@ -0,0 +1,1308 @@
>+/*
>+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "logging/rtc_event_log/rtc_event_log_parser_new.h"
>+
>+#include <stdint.h>
>+#include <string.h>
>+
>+#include <algorithm>
>+#include <fstream>
>+#include <istream>  // no-presubmit-check TODO(webrtc:8982)
>+#include <limits>
>+#include <map>
>+#include <utility>
>+
>+#include "absl/memory/memory.h"
>+#include "api/rtp_headers.h"
>+#include "api/rtpparameters.h"
>+#include "logging/rtc_event_log/rtc_event_log.h"
>+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
>+#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
>+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>+#include "modules/rtp_rtcp/source/byte_io.h"
>+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
>+#include "modules/rtp_rtcp/source/rtp_utility.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/logging.h"
>+#include "rtc_base/protobuf_utils.h"
>+
>+namespace webrtc {
>+
>+namespace {
>+RtcpMode GetRuntimeRtcpMode(rtclog::VideoReceiveConfig::RtcpMode rtcp_mode) {
>+  switch (rtcp_mode) {
>+    case rtclog::VideoReceiveConfig::RTCP_COMPOUND:
>+      return RtcpMode::kCompound;
>+    case rtclog::VideoReceiveConfig::RTCP_REDUCEDSIZE:
>+      return RtcpMode::kReducedSize;
>+  }
>+  RTC_NOTREACHED();
>+  return RtcpMode::kOff;
>+}
>+
>+ParsedRtcEventLogNew::EventType GetRuntimeEventType(
>+    rtclog::Event::EventType event_type) {
>+  switch (event_type) {
>+    case rtclog::Event::UNKNOWN_EVENT:
>+      return ParsedRtcEventLogNew::EventType::UNKNOWN_EVENT;
>+    case rtclog::Event::LOG_START:
>+      return ParsedRtcEventLogNew::EventType::LOG_START;
>+    case rtclog::Event::LOG_END:
>+      return ParsedRtcEventLogNew::EventType::LOG_END;
>+    case rtclog::Event::RTP_EVENT:
>+      return ParsedRtcEventLogNew::EventType::RTP_EVENT;
>+    case rtclog::Event::RTCP_EVENT:
>+      return ParsedRtcEventLogNew::EventType::RTCP_EVENT;
>+    case rtclog::Event::AUDIO_PLAYOUT_EVENT:
>+      return ParsedRtcEventLogNew::EventType::AUDIO_PLAYOUT_EVENT;
>+    case rtclog::Event::LOSS_BASED_BWE_UPDATE:
>+      return ParsedRtcEventLogNew::EventType::LOSS_BASED_BWE_UPDATE;
>+    case rtclog::Event::DELAY_BASED_BWE_UPDATE:
>+      return ParsedRtcEventLogNew::EventType::DELAY_BASED_BWE_UPDATE;
>+    case rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT:
>+      return ParsedRtcEventLogNew::EventType::VIDEO_RECEIVER_CONFIG_EVENT;
>+    case rtclog::Event::VIDEO_SENDER_CONFIG_EVENT:
>+      return ParsedRtcEventLogNew::EventType::VIDEO_SENDER_CONFIG_EVENT;
>+    case rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT:
>+      return ParsedRtcEventLogNew::EventType::AUDIO_RECEIVER_CONFIG_EVENT;
>+    case rtclog::Event::AUDIO_SENDER_CONFIG_EVENT:
>+      return ParsedRtcEventLogNew::EventType::AUDIO_SENDER_CONFIG_EVENT;
>+    case rtclog::Event::AUDIO_NETWORK_ADAPTATION_EVENT:
>+      return ParsedRtcEventLogNew::EventType::AUDIO_NETWORK_ADAPTATION_EVENT;
>+    case rtclog::Event::BWE_PROBE_CLUSTER_CREATED_EVENT:
>+      return ParsedRtcEventLogNew::EventType::BWE_PROBE_CLUSTER_CREATED_EVENT;
>+    case rtclog::Event::BWE_PROBE_RESULT_EVENT:
>+      // Probe successes and failures are currently stored in the same proto
>+      // message, we are moving towards separate messages. Probe results
>+      // therefore need special treatment in the parser.
>+      return ParsedRtcEventLogNew::EventType::UNKNOWN_EVENT;
>+    case rtclog::Event::ALR_STATE_EVENT:
>+      return ParsedRtcEventLogNew::EventType::ALR_STATE_EVENT;
>+    case rtclog::Event::ICE_CANDIDATE_PAIR_CONFIG:
>+      return ParsedRtcEventLogNew::EventType::ICE_CANDIDATE_PAIR_CONFIG;
>+    case rtclog::Event::ICE_CANDIDATE_PAIR_EVENT:
>+      return ParsedRtcEventLogNew::EventType::ICE_CANDIDATE_PAIR_EVENT;
>+  }
>+  return ParsedRtcEventLogNew::EventType::UNKNOWN_EVENT;
>+}
>+
>+BandwidthUsage GetRuntimeDetectorState(
>+    rtclog::DelayBasedBweUpdate::DetectorState detector_state) {
>+  switch (detector_state) {
>+    case rtclog::DelayBasedBweUpdate::BWE_NORMAL:
>+      return BandwidthUsage::kBwNormal;
>+    case rtclog::DelayBasedBweUpdate::BWE_UNDERUSING:
>+      return BandwidthUsage::kBwUnderusing;
>+    case rtclog::DelayBasedBweUpdate::BWE_OVERUSING:
>+      return BandwidthUsage::kBwOverusing;
>+  }
>+  RTC_NOTREACHED();
>+  return BandwidthUsage::kBwNormal;
>+}
>+
>+IceCandidatePairConfigType GetRuntimeIceCandidatePairConfigType(
>+    rtclog::IceCandidatePairConfig::IceCandidatePairConfigType type) {
>+  switch (type) {
>+    case rtclog::IceCandidatePairConfig::ADDED:
>+      return IceCandidatePairConfigType::kAdded;
>+    case rtclog::IceCandidatePairConfig::UPDATED:
>+      return IceCandidatePairConfigType::kUpdated;
>+    case rtclog::IceCandidatePairConfig::DESTROYED:
>+      return IceCandidatePairConfigType::kDestroyed;
>+    case rtclog::IceCandidatePairConfig::SELECTED:
>+      return IceCandidatePairConfigType::kSelected;
>+  }
>+  RTC_NOTREACHED();
>+  return IceCandidatePairConfigType::kAdded;
>+}
>+
>+IceCandidateType GetRuntimeIceCandidateType(
>+    rtclog::IceCandidatePairConfig::IceCandidateType type) {
>+  switch (type) {
>+    case rtclog::IceCandidatePairConfig::LOCAL:
>+      return IceCandidateType::kLocal;
>+    case rtclog::IceCandidatePairConfig::STUN:
>+      return IceCandidateType::kStun;
>+    case rtclog::IceCandidatePairConfig::PRFLX:
>+      return IceCandidateType::kPrflx;
>+    case rtclog::IceCandidatePairConfig::RELAY:
>+      return IceCandidateType::kRelay;
>+    case rtclog::IceCandidatePairConfig::UNKNOWN_CANDIDATE_TYPE:
>+      return IceCandidateType::kUnknown;
>+  }
>+  RTC_NOTREACHED();
>+  return IceCandidateType::kUnknown;
>+}
>+
>+IceCandidatePairProtocol GetRuntimeIceCandidatePairProtocol(
>+    rtclog::IceCandidatePairConfig::Protocol protocol) {
>+  switch (protocol) {
>+    case rtclog::IceCandidatePairConfig::UDP:
>+      return IceCandidatePairProtocol::kUdp;
>+    case rtclog::IceCandidatePairConfig::TCP:
>+      return IceCandidatePairProtocol::kTcp;
>+    case rtclog::IceCandidatePairConfig::SSLTCP:
>+      return IceCandidatePairProtocol::kSsltcp;
>+    case rtclog::IceCandidatePairConfig::TLS:
>+      return IceCandidatePairProtocol::kTls;
>+    case rtclog::IceCandidatePairConfig::UNKNOWN_PROTOCOL:
>+      return IceCandidatePairProtocol::kUnknown;
>+  }
>+  RTC_NOTREACHED();
>+  return IceCandidatePairProtocol::kUnknown;
>+}
>+
>+IceCandidatePairAddressFamily GetRuntimeIceCandidatePairAddressFamily(
>+    rtclog::IceCandidatePairConfig::AddressFamily address_family) {
>+  switch (address_family) {
>+    case rtclog::IceCandidatePairConfig::IPV4:
>+      return IceCandidatePairAddressFamily::kIpv4;
>+    case rtclog::IceCandidatePairConfig::IPV6:
>+      return IceCandidatePairAddressFamily::kIpv6;
>+    case rtclog::IceCandidatePairConfig::UNKNOWN_ADDRESS_FAMILY:
>+      return IceCandidatePairAddressFamily::kUnknown;
>+  }
>+  RTC_NOTREACHED();
>+  return IceCandidatePairAddressFamily::kUnknown;
>+}
>+
>+IceCandidateNetworkType GetRuntimeIceCandidateNetworkType(
>+    rtclog::IceCandidatePairConfig::NetworkType network_type) {
>+  switch (network_type) {
>+    case rtclog::IceCandidatePairConfig::ETHERNET:
>+      return IceCandidateNetworkType::kEthernet;
>+    case rtclog::IceCandidatePairConfig::LOOPBACK:
>+      return IceCandidateNetworkType::kLoopback;
>+    case rtclog::IceCandidatePairConfig::WIFI:
>+      return IceCandidateNetworkType::kWifi;
>+    case rtclog::IceCandidatePairConfig::VPN:
>+      return IceCandidateNetworkType::kVpn;
>+    case rtclog::IceCandidatePairConfig::CELLULAR:
>+      return IceCandidateNetworkType::kCellular;
>+    case rtclog::IceCandidatePairConfig::UNKNOWN_NETWORK_TYPE:
>+      return IceCandidateNetworkType::kUnknown;
>+  }
>+  RTC_NOTREACHED();
>+  return IceCandidateNetworkType::kUnknown;
>+}
>+
>+IceCandidatePairEventType GetRuntimeIceCandidatePairEventType(
>+    rtclog::IceCandidatePairEvent::IceCandidatePairEventType type) {
>+  switch (type) {
>+    case rtclog::IceCandidatePairEvent::CHECK_SENT:
>+      return IceCandidatePairEventType::kCheckSent;
>+    case rtclog::IceCandidatePairEvent::CHECK_RECEIVED:
>+      return IceCandidatePairEventType::kCheckReceived;
>+    case rtclog::IceCandidatePairEvent::CHECK_RESPONSE_SENT:
>+      return IceCandidatePairEventType::kCheckResponseSent;
>+    case rtclog::IceCandidatePairEvent::CHECK_RESPONSE_RECEIVED:
>+      return IceCandidatePairEventType::kCheckResponseReceived;
>+  }
>+  RTC_NOTREACHED();
>+  return IceCandidatePairEventType::kCheckSent;
>+}
>+
>+// Return default values for header extensions, to use on streams without stored
>+// mapping data. Currently this only applies to audio streams, since the mapping
>+// is not stored in the event log.
>+// TODO(ivoc): Remove this once this mapping is stored in the event log for
>+//             audio streams. Tracking bug: webrtc:6399
>+webrtc::RtpHeaderExtensionMap GetDefaultHeaderExtensionMap() {
>+  webrtc::RtpHeaderExtensionMap default_map;
>+  default_map.Register<AudioLevel>(webrtc::RtpExtension::kAudioLevelDefaultId);
>+  default_map.Register<TransmissionOffset>(
>+      webrtc::RtpExtension::kTimestampOffsetDefaultId);
>+  default_map.Register<AbsoluteSendTime>(
>+      webrtc::RtpExtension::kAbsSendTimeDefaultId);
>+  default_map.Register<VideoOrientation>(
>+      webrtc::RtpExtension::kVideoRotationDefaultId);
>+  default_map.Register<VideoContentTypeExtension>(
>+      webrtc::RtpExtension::kVideoContentTypeDefaultId);
>+  default_map.Register<VideoTimingExtension>(
>+      webrtc::RtpExtension::kVideoTimingDefaultId);
>+  default_map.Register<TransportSequenceNumber>(
>+      webrtc::RtpExtension::kTransportSequenceNumberDefaultId);
>+  default_map.Register<PlayoutDelayLimits>(
>+      webrtc::RtpExtension::kPlayoutDelayDefaultId);
>+  return default_map;
>+}
>+
>+std::pair<uint64_t, bool> ParseVarInt(
>+    std::istream& stream) {  // no-presubmit-check TODO(webrtc:8982)
>+  uint64_t varint = 0;
>+  for (size_t bytes_read = 0; bytes_read < 10; ++bytes_read) {
>+    // The most significant bit of each byte is 0 if it is the last byte in
>+    // the varint and 1 otherwise. Thus, we take the 7 least significant bits
>+    // of each byte and shift them 7 bits for each byte read previously to get
>+    // the (unsigned) integer.
>+    int byte = stream.get();
>+    if (stream.eof()) {
>+      return std::make_pair(varint, false);
>+    }
>+    RTC_DCHECK_GE(byte, 0);
>+    RTC_DCHECK_LE(byte, 255);
>+    varint |= static_cast<uint64_t>(byte & 0x7F) << (7 * bytes_read);
>+    if ((byte & 0x80) == 0) {
>+      return std::make_pair(varint, true);
>+    }
>+  }
>+  return std::make_pair(varint, false);
>+}
>+
>+void GetHeaderExtensions(std::vector<RtpExtension>* header_extensions,
>+                         const RepeatedPtrField<rtclog::RtpHeaderExtension>&
>+                             proto_header_extensions) {
>+  header_extensions->clear();
>+  for (auto& p : proto_header_extensions) {
>+    RTC_CHECK(p.has_name());
>+    RTC_CHECK(p.has_id());
>+    const std::string& name = p.name();
>+    int id = p.id();
>+    header_extensions->push_back(RtpExtension(name, id));
>+  }
>+}
>+
>+}  // namespace
>+
>+ParsedRtcEventLogNew::ParsedRtcEventLogNew(
>+    UnconfiguredHeaderExtensions parse_unconfigured_header_extensions)
>+    : parse_unconfigured_header_extensions_(
>+          parse_unconfigured_header_extensions) {
>+  Clear();
>+}
>+
>+void ParsedRtcEventLogNew::Clear() {
>+  events_.clear();
>+  default_extension_map_ = GetDefaultHeaderExtensionMap();
>+
>+  incoming_rtx_ssrcs_.clear();
>+  incoming_video_ssrcs_.clear();
>+  incoming_audio_ssrcs_.clear();
>+  outgoing_rtx_ssrcs_.clear();
>+  outgoing_video_ssrcs_.clear();
>+  outgoing_audio_ssrcs_.clear();
>+
>+  incoming_rtp_packets_map_.clear();
>+  outgoing_rtp_packets_map_.clear();
>+  incoming_rtp_packets_by_ssrc_.clear();
>+  outgoing_rtp_packets_by_ssrc_.clear();
>+  incoming_rtp_packet_views_by_ssrc_.clear();
>+  outgoing_rtp_packet_views_by_ssrc_.clear();
>+
>+  incoming_rtcp_packets_.clear();
>+  outgoing_rtcp_packets_.clear();
>+
>+  incoming_rr_.clear();
>+  outgoing_rr_.clear();
>+  incoming_sr_.clear();
>+  outgoing_sr_.clear();
>+  incoming_nack_.clear();
>+  outgoing_nack_.clear();
>+  incoming_remb_.clear();
>+  outgoing_remb_.clear();
>+  incoming_transport_feedback_.clear();
>+  outgoing_transport_feedback_.clear();
>+
>+  start_log_events_.clear();
>+  stop_log_events_.clear();
>+  audio_playout_events_.clear();
>+  audio_network_adaptation_events_.clear();
>+  bwe_probe_cluster_created_events_.clear();
>+  bwe_probe_failure_events_.clear();
>+  bwe_probe_success_events_.clear();
>+  bwe_delay_updates_.clear();
>+  bwe_loss_updates_.clear();
>+  alr_state_events_.clear();
>+  ice_candidate_pair_configs_.clear();
>+  ice_candidate_pair_events_.clear();
>+  audio_recv_configs_.clear();
>+  audio_send_configs_.clear();
>+  video_recv_configs_.clear();
>+  video_send_configs_.clear();
>+
>+  memset(last_incoming_rtcp_packet_, 0, IP_PACKET_SIZE);
>+  last_incoming_rtcp_packet_length_ = 0;
>+
>+  first_timestamp_ = std::numeric_limits<int64_t>::max();
>+  last_timestamp_ = std::numeric_limits<int64_t>::min();
>+
>+  incoming_rtp_extensions_maps_.clear();
>+  outgoing_rtp_extensions_maps_.clear();
>+}
>+
>+bool ParsedRtcEventLogNew::ParseFile(const std::string& filename) {
>+  std::ifstream file(  // no-presubmit-check TODO(webrtc:8982)
>+      filename, std::ios_base::in | std::ios_base::binary);
>+  if (!file.good() || !file.is_open()) {
>+    RTC_LOG(LS_WARNING) << "Could not open file for reading.";
>+    return false;
>+  }
>+
>+  return ParseStream(file);
>+}
>+
>+bool ParsedRtcEventLogNew::ParseString(const std::string& s) {
>+  std::istringstream stream(  // no-presubmit-check TODO(webrtc:8982)
>+      s, std::ios_base::in | std::ios_base::binary);
>+  return ParseStream(stream);
>+}
>+
>+bool ParsedRtcEventLogNew::ParseStream(
>+    std::istream& stream) {  // no-presubmit-check TODO(webrtc:8982)
>+  Clear();
>+  bool success = ParseStreamInternal(stream);
>+
>+  // ParseStreamInternal stores the RTP packets in a map indexed by SSRC.
>+  // Since we dont need rapid lookup based on SSRC after parsing, we move the
>+  // packets_streams from map to vector.
>+  incoming_rtp_packets_by_ssrc_.reserve(incoming_rtp_packets_map_.size());
>+  for (const auto& kv : incoming_rtp_packets_map_) {
>+    incoming_rtp_packets_by_ssrc_.emplace_back(LoggedRtpStreamIncoming());
>+    incoming_rtp_packets_by_ssrc_.back().ssrc = kv.first;
>+    incoming_rtp_packets_by_ssrc_.back().incoming_packets =
>+        std::move(kv.second);
>+  }
>+  incoming_rtp_packets_map_.clear();
>+  outgoing_rtp_packets_by_ssrc_.reserve(outgoing_rtp_packets_map_.size());
>+  for (const auto& kv : outgoing_rtp_packets_map_) {
>+    outgoing_rtp_packets_by_ssrc_.emplace_back(LoggedRtpStreamOutgoing());
>+    outgoing_rtp_packets_by_ssrc_.back().ssrc = kv.first;
>+    outgoing_rtp_packets_by_ssrc_.back().outgoing_packets =
>+        std::move(kv.second);
>+  }
>+  outgoing_rtp_packets_map_.clear();
>+
>+  // Build PacketViews for easier iteration over RTP packets
>+  for (const auto& stream : incoming_rtp_packets_by_ssrc_) {
>+    incoming_rtp_packet_views_by_ssrc_.emplace_back(
>+        LoggedRtpStreamView(stream.ssrc, stream.incoming_packets.data(),
>+                            stream.incoming_packets.size()));
>+  }
>+  for (const auto& stream : outgoing_rtp_packets_by_ssrc_) {
>+    outgoing_rtp_packet_views_by_ssrc_.emplace_back(
>+        LoggedRtpStreamView(stream.ssrc, stream.outgoing_packets.data(),
>+                            stream.outgoing_packets.size()));
>+  }
>+
>+  return success;
>+}
>+
>+bool ParsedRtcEventLogNew::ParseStreamInternal(
>+    std::istream& stream) {  // no-presubmit-check TODO(webrtc:8982)
>+  const size_t kMaxEventSize = (1u << 16) - 1;
>+  std::vector<char> tmp_buffer(kMaxEventSize);
>+  uint64_t tag;
>+  uint64_t message_length;
>+  bool success;
>+
>+  RTC_DCHECK(stream.good());
>+
>+  while (1) {
>+    // Check whether we have reached end of file.
>+    stream.peek();
>+    if (stream.eof()) {
>+      break;
>+    }
>+
>+    // Read the next message tag. The tag number is defined as
>+    // (fieldnumber << 3) | wire_type. In our case, the field number is
>+    // supposed to be 1 and the wire type for an
>+    // length-delimited field is 2.
>+    const uint64_t kExpectedTag = (1 << 3) | 2;
>+    std::tie(tag, success) = ParseVarInt(stream);
>+    if (!success) {
>+      RTC_LOG(LS_WARNING)
>+          << "Missing field tag from beginning of protobuf event.";
>+      return false;
>+    } else if (tag != kExpectedTag) {
>+      RTC_LOG(LS_WARNING)
>+          << "Unexpected field tag at beginning of protobuf event.";
>+      return false;
>+    }
>+
>+    // Read the length field.
>+    std::tie(message_length, success) = ParseVarInt(stream);
>+    if (!success) {
>+      RTC_LOG(LS_WARNING) << "Missing message length after protobuf field tag.";
>+      return false;
>+    } else if (message_length > kMaxEventSize) {
>+      RTC_LOG(LS_WARNING) << "Protobuf message length is too large.";
>+      return false;
>+    }
>+
>+    // Read the next protobuf event to a temporary char buffer.
>+    stream.read(tmp_buffer.data(), message_length);
>+    if (stream.gcount() != static_cast<int>(message_length)) {
>+      RTC_LOG(LS_WARNING) << "Failed to read protobuf message from file.";
>+      return false;
>+    }
>+
>+    // Parse the protobuf event from the buffer.
>+    rtclog::Event event;
>+    if (!event.ParseFromArray(tmp_buffer.data(), message_length)) {
>+      RTC_LOG(LS_WARNING) << "Failed to parse protobuf message.";
>+      return false;
>+    }
>+
>+    StoreParsedEvent(event);
>+    events_.push_back(event);
>+  }
>+  return true;
>+}
>+
>+void ParsedRtcEventLogNew::StoreParsedEvent(const rtclog::Event& event) {
>+  if (event.type() != rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT &&
>+      event.type() != rtclog::Event::VIDEO_SENDER_CONFIG_EVENT &&
>+      event.type() != rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT &&
>+      event.type() != rtclog::Event::AUDIO_SENDER_CONFIG_EVENT &&
>+      event.type() != rtclog::Event::LOG_START &&
>+      event.type() != rtclog::Event::LOG_END) {
>+    RTC_CHECK(event.has_timestamp_us());
>+    int64_t timestamp = event.timestamp_us();
>+    first_timestamp_ = std::min(first_timestamp_, timestamp);
>+    last_timestamp_ = std::max(last_timestamp_, timestamp);
>+  }
>+
>+  switch (GetEventType(event)) {
>+    case ParsedRtcEventLogNew::EventType::VIDEO_RECEIVER_CONFIG_EVENT: {
>+      rtclog::StreamConfig config = GetVideoReceiveConfig(event);
>+      video_recv_configs_.emplace_back(GetTimestamp(event), config);
>+      incoming_rtp_extensions_maps_[config.remote_ssrc] =
>+          RtpHeaderExtensionMap(config.rtp_extensions);
>+      // TODO(terelius): I don't understand the reason for configuring header
>+      // extensions for the local SSRC. I think it should be removed, but for
>+      // now I want to preserve the previous functionality.
>+      incoming_rtp_extensions_maps_[config.local_ssrc] =
>+          RtpHeaderExtensionMap(config.rtp_extensions);
>+      incoming_video_ssrcs_.insert(config.remote_ssrc);
>+      incoming_video_ssrcs_.insert(config.rtx_ssrc);
>+      incoming_rtx_ssrcs_.insert(config.rtx_ssrc);
>+      break;
>+    }
>+    case ParsedRtcEventLogNew::EventType::VIDEO_SENDER_CONFIG_EVENT: {
>+      std::vector<rtclog::StreamConfig> configs = GetVideoSendConfig(event);
>+      video_send_configs_.emplace_back(GetTimestamp(event), configs);
>+      for (const auto& config : configs) {
>+        outgoing_rtp_extensions_maps_[config.local_ssrc] =
>+            RtpHeaderExtensionMap(config.rtp_extensions);
>+        outgoing_rtp_extensions_maps_[config.rtx_ssrc] =
>+            RtpHeaderExtensionMap(config.rtp_extensions);
>+        outgoing_video_ssrcs_.insert(config.local_ssrc);
>+        outgoing_video_ssrcs_.insert(config.rtx_ssrc);
>+        outgoing_rtx_ssrcs_.insert(config.rtx_ssrc);
>+      }
>+      break;
>+    }
>+    case ParsedRtcEventLogNew::EventType::AUDIO_RECEIVER_CONFIG_EVENT: {
>+      rtclog::StreamConfig config = GetAudioReceiveConfig(event);
>+      audio_recv_configs_.emplace_back(GetTimestamp(event), config);
>+      incoming_rtp_extensions_maps_[config.remote_ssrc] =
>+          RtpHeaderExtensionMap(config.rtp_extensions);
>+      incoming_rtp_extensions_maps_[config.local_ssrc] =
>+          RtpHeaderExtensionMap(config.rtp_extensions);
>+      incoming_audio_ssrcs_.insert(config.remote_ssrc);
>+      break;
>+    }
>+    case ParsedRtcEventLogNew::EventType::AUDIO_SENDER_CONFIG_EVENT: {
>+      rtclog::StreamConfig config = GetAudioSendConfig(event);
>+      audio_send_configs_.emplace_back(GetTimestamp(event), config);
>+      outgoing_rtp_extensions_maps_[config.local_ssrc] =
>+          RtpHeaderExtensionMap(config.rtp_extensions);
>+      outgoing_audio_ssrcs_.insert(config.local_ssrc);
>+      break;
>+    }
>+    case ParsedRtcEventLogNew::EventType::RTP_EVENT: {
>+      PacketDirection direction;
>+      uint8_t header[IP_PACKET_SIZE];
>+      size_t header_length;
>+      size_t total_length;
>+      const RtpHeaderExtensionMap* extension_map = GetRtpHeader(
>+          event, &direction, header, &header_length, &total_length, nullptr);
>+      RtpUtility::RtpHeaderParser rtp_parser(header, header_length);
>+      RTPHeader parsed_header;
>+      if (extension_map != nullptr) {
>+        rtp_parser.Parse(&parsed_header, extension_map);
>+      } else {
>+        // Use the default extension map.
>+        // TODO(terelius): This should be removed. GetRtpHeader will return the
>+        // default map if the parser is configured for it.
>+        // TODO(ivoc): Once configuration of audio streams is stored in the
>+        //             event log, this can be removed.
>+        //             Tracking bug: webrtc:6399
>+        rtp_parser.Parse(&parsed_header, &default_extension_map_);
>+      }
>+      RTC_CHECK(event.has_timestamp_us());
>+      uint64_t timestamp_us = event.timestamp_us();
>+      if (direction == kIncomingPacket) {
>+        incoming_rtp_packets_map_[parsed_header.ssrc].push_back(
>+            LoggedRtpPacketIncoming(timestamp_us, parsed_header, header_length,
>+                                    total_length));
>+      } else {
>+        outgoing_rtp_packets_map_[parsed_header.ssrc].push_back(
>+            LoggedRtpPacketOutgoing(timestamp_us, parsed_header, header_length,
>+                                    total_length));
>+      }
>+      break;
>+    }
>+    case ParsedRtcEventLogNew::EventType::RTCP_EVENT: {
>+      PacketDirection direction;
>+      uint8_t packet[IP_PACKET_SIZE];
>+      size_t total_length;
>+      GetRtcpPacket(event, &direction, packet, &total_length);
>+      uint64_t timestamp_us = GetTimestamp(event);
>+      RTC_CHECK_LE(total_length, IP_PACKET_SIZE);
>+      if (direction == kIncomingPacket) {
>+        // Currently incoming RTCP packets are logged twice, both for audio and
>+        // video. Only act on one of them. Compare against the previous parsed
>+        // incoming RTCP packet.
>+        if (total_length == last_incoming_rtcp_packet_length_ &&
>+            memcmp(last_incoming_rtcp_packet_, packet, total_length) == 0)
>+          break;
>+        incoming_rtcp_packets_.push_back(
>+            LoggedRtcpPacketIncoming(timestamp_us, packet, total_length));
>+        last_incoming_rtcp_packet_length_ = total_length;
>+        memcpy(last_incoming_rtcp_packet_, packet, total_length);
>+      } else {
>+        outgoing_rtcp_packets_.push_back(
>+            LoggedRtcpPacketOutgoing(timestamp_us, packet, total_length));
>+      }
>+      rtcp::CommonHeader header;
>+      const uint8_t* packet_end = packet + total_length;
>+      for (const uint8_t* block = packet; block < packet_end;
>+           block = header.NextPacket()) {
>+        RTC_CHECK(header.Parse(block, packet_end - block));
>+        if (header.type() == rtcp::TransportFeedback::kPacketType &&
>+            header.fmt() == rtcp::TransportFeedback::kFeedbackMessageType) {
>+          if (direction == kIncomingPacket) {
>+            incoming_transport_feedback_.emplace_back();
>+            LoggedRtcpPacketTransportFeedback& parsed_block =
>+                incoming_transport_feedback_.back();
>+            parsed_block.timestamp_us = GetTimestamp(event);
>+            if (!parsed_block.transport_feedback.Parse(header))
>+              incoming_transport_feedback_.pop_back();
>+          } else {
>+            outgoing_transport_feedback_.emplace_back();
>+            LoggedRtcpPacketTransportFeedback& parsed_block =
>+                outgoing_transport_feedback_.back();
>+            parsed_block.timestamp_us = GetTimestamp(event);
>+            if (!parsed_block.transport_feedback.Parse(header))
>+              outgoing_transport_feedback_.pop_back();
>+          }
>+        } else if (header.type() == rtcp::SenderReport::kPacketType) {
>+          LoggedRtcpPacketSenderReport parsed_block;
>+          parsed_block.timestamp_us = GetTimestamp(event);
>+          if (parsed_block.sr.Parse(header)) {
>+            if (direction == kIncomingPacket)
>+              incoming_sr_.push_back(std::move(parsed_block));
>+            else
>+              outgoing_sr_.push_back(std::move(parsed_block));
>+          }
>+        } else if (header.type() == rtcp::ReceiverReport::kPacketType) {
>+          LoggedRtcpPacketReceiverReport parsed_block;
>+          parsed_block.timestamp_us = GetTimestamp(event);
>+          if (parsed_block.rr.Parse(header)) {
>+            if (direction == kIncomingPacket)
>+              incoming_rr_.push_back(std::move(parsed_block));
>+            else
>+              outgoing_rr_.push_back(std::move(parsed_block));
>+          }
>+        } else if (header.type() == rtcp::Remb::kPacketType &&
>+                   header.fmt() == rtcp::Remb::kFeedbackMessageType) {
>+          LoggedRtcpPacketRemb parsed_block;
>+          parsed_block.timestamp_us = GetTimestamp(event);
>+          if (parsed_block.remb.Parse(header)) {
>+            if (direction == kIncomingPacket)
>+              incoming_remb_.push_back(std::move(parsed_block));
>+            else
>+              outgoing_remb_.push_back(std::move(parsed_block));
>+          }
>+        } else if (header.type() == rtcp::Nack::kPacketType &&
>+                   header.fmt() == rtcp::Nack::kFeedbackMessageType) {
>+          LoggedRtcpPacketNack parsed_block;
>+          parsed_block.timestamp_us = GetTimestamp(event);
>+          if (parsed_block.nack.Parse(header)) {
>+            if (direction == kIncomingPacket)
>+              incoming_nack_.push_back(std::move(parsed_block));
>+            else
>+              outgoing_nack_.push_back(std::move(parsed_block));
>+          }
>+        }
>+      }
>+      break;
>+    }
>+    case ParsedRtcEventLogNew::EventType::LOG_START: {
>+      start_log_events_.push_back(LoggedStartEvent(GetTimestamp(event)));
>+      break;
>+    }
>+    case ParsedRtcEventLogNew::EventType::LOG_END: {
>+      stop_log_events_.push_back(LoggedStopEvent(GetTimestamp(event)));
>+      break;
>+    }
>+    case ParsedRtcEventLogNew::EventType::AUDIO_PLAYOUT_EVENT: {
>+      LoggedAudioPlayoutEvent playout_event = GetAudioPlayout(event);
>+      audio_playout_events_[playout_event.ssrc].push_back(playout_event);
>+      break;
>+    }
>+    case ParsedRtcEventLogNew::EventType::LOSS_BASED_BWE_UPDATE: {
>+      bwe_loss_updates_.push_back(GetLossBasedBweUpdate(event));
>+      break;
>+    }
>+    case ParsedRtcEventLogNew::EventType::DELAY_BASED_BWE_UPDATE: {
>+      bwe_delay_updates_.push_back(GetDelayBasedBweUpdate(event));
>+      break;
>+    }
>+    case ParsedRtcEventLogNew::EventType::AUDIO_NETWORK_ADAPTATION_EVENT: {
>+      LoggedAudioNetworkAdaptationEvent ana_event =
>+          GetAudioNetworkAdaptation(event);
>+      audio_network_adaptation_events_.push_back(ana_event);
>+      break;
>+    }
>+    case ParsedRtcEventLogNew::EventType::BWE_PROBE_CLUSTER_CREATED_EVENT: {
>+      bwe_probe_cluster_created_events_.push_back(
>+          GetBweProbeClusterCreated(event));
>+      break;
>+    }
>+    case ParsedRtcEventLogNew::EventType::BWE_PROBE_FAILURE_EVENT: {
>+      bwe_probe_failure_events_.push_back(GetBweProbeFailure(event));
>+      break;
>+    }
>+    case ParsedRtcEventLogNew::EventType::BWE_PROBE_SUCCESS_EVENT: {
>+      bwe_probe_success_events_.push_back(GetBweProbeSuccess(event));
>+      break;
>+    }
>+    case ParsedRtcEventLogNew::EventType::ALR_STATE_EVENT: {
>+      alr_state_events_.push_back(GetAlrState(event));
>+      break;
>+    }
>+    case ParsedRtcEventLogNew::EventType::ICE_CANDIDATE_PAIR_CONFIG: {
>+      ice_candidate_pair_configs_.push_back(GetIceCandidatePairConfig(event));
>+      break;
>+    }
>+    case ParsedRtcEventLogNew::EventType::ICE_CANDIDATE_PAIR_EVENT: {
>+      ice_candidate_pair_events_.push_back(GetIceCandidatePairEvent(event));
>+      break;
>+    }
>+    case ParsedRtcEventLogNew::EventType::UNKNOWN_EVENT: {
>+      break;
>+    }
>+  }
>+}
>+
>+size_t ParsedRtcEventLogNew::GetNumberOfEvents() const {
>+  return events_.size();
>+}
>+
>+int64_t ParsedRtcEventLogNew::GetTimestamp(size_t index) const {
>+  RTC_CHECK_LT(index, GetNumberOfEvents());
>+  const rtclog::Event& event = events_[index];
>+  return GetTimestamp(event);
>+}
>+
>+int64_t ParsedRtcEventLogNew::GetTimestamp(const rtclog::Event& event) const {
>+  RTC_CHECK(event.has_timestamp_us());
>+  return event.timestamp_us();
>+}
>+
>+ParsedRtcEventLogNew::EventType ParsedRtcEventLogNew::GetEventType(
>+    size_t index) const {
>+  RTC_CHECK_LT(index, GetNumberOfEvents());
>+  const rtclog::Event& event = events_[index];
>+  return GetEventType(event);
>+}
>+
>+ParsedRtcEventLogNew::EventType ParsedRtcEventLogNew::GetEventType(
>+    const rtclog::Event& event) const {
>+  RTC_CHECK(event.has_type());
>+  if (event.type() == rtclog::Event::BWE_PROBE_RESULT_EVENT) {
>+    RTC_CHECK(event.has_probe_result());
>+    RTC_CHECK(event.probe_result().has_result());
>+    if (event.probe_result().result() == rtclog::BweProbeResult::SUCCESS)
>+      return ParsedRtcEventLogNew::EventType::BWE_PROBE_SUCCESS_EVENT;
>+    return ParsedRtcEventLogNew::EventType::BWE_PROBE_FAILURE_EVENT;
>+  }
>+  return GetRuntimeEventType(event.type());
>+}
>+
>+// The header must have space for at least IP_PACKET_SIZE bytes.
>+const webrtc::RtpHeaderExtensionMap* ParsedRtcEventLogNew::GetRtpHeader(
>+    size_t index,
>+    PacketDirection* incoming,
>+    uint8_t* header,
>+    size_t* header_length,
>+    size_t* total_length,
>+    int* probe_cluster_id) const {
>+  RTC_CHECK_LT(index, GetNumberOfEvents());
>+  const rtclog::Event& event = events_[index];
>+  return GetRtpHeader(event, incoming, header, header_length, total_length,
>+                      probe_cluster_id);
>+}
>+
>+const webrtc::RtpHeaderExtensionMap* ParsedRtcEventLogNew::GetRtpHeader(
>+    const rtclog::Event& event,
>+    PacketDirection* incoming,
>+    uint8_t* header,
>+    size_t* header_length,
>+    size_t* total_length,
>+    int* probe_cluster_id) const {
>+  RTC_CHECK(event.has_type());
>+  RTC_CHECK_EQ(event.type(), rtclog::Event::RTP_EVENT);
>+  RTC_CHECK(event.has_rtp_packet());
>+  const rtclog::RtpPacket& rtp_packet = event.rtp_packet();
>+  // Get direction of packet.
>+  RTC_CHECK(rtp_packet.has_incoming());
>+  if (incoming != nullptr) {
>+    *incoming = rtp_packet.incoming() ? kIncomingPacket : kOutgoingPacket;
>+  }
>+  // Get packet length.
>+  RTC_CHECK(rtp_packet.has_packet_length());
>+  if (total_length != nullptr) {
>+    *total_length = rtp_packet.packet_length();
>+  }
>+  // Get header length.
>+  RTC_CHECK(rtp_packet.has_header());
>+  if (header_length != nullptr) {
>+    *header_length = rtp_packet.header().size();
>+  }
>+  if (probe_cluster_id != nullptr) {
>+    if (rtp_packet.has_probe_cluster_id()) {
>+      *probe_cluster_id = rtp_packet.probe_cluster_id();
>+      RTC_CHECK_NE(*probe_cluster_id, PacedPacketInfo::kNotAProbe);
>+    } else {
>+      *probe_cluster_id = PacedPacketInfo::kNotAProbe;
>+    }
>+  }
>+  // Get header contents.
>+  if (header != nullptr) {
>+    const size_t kMinRtpHeaderSize = 12;
>+    RTC_CHECK_GE(rtp_packet.header().size(), kMinRtpHeaderSize);
>+    RTC_CHECK_LE(rtp_packet.header().size(),
>+                 static_cast<size_t>(IP_PACKET_SIZE));
>+    memcpy(header, rtp_packet.header().data(), rtp_packet.header().size());
>+    uint32_t ssrc = ByteReader<uint32_t>::ReadBigEndian(header + 8);
>+    auto& extensions_maps = rtp_packet.incoming()
>+                                ? incoming_rtp_extensions_maps_
>+                                : outgoing_rtp_extensions_maps_;
>+    auto it = extensions_maps.find(ssrc);
>+    if (it != extensions_maps.end()) {
>+      return &(it->second);
>+    }
>+    if (parse_unconfigured_header_extensions_ ==
>+        UnconfiguredHeaderExtensions::kAttemptWebrtcDefaultConfig) {
>+      RTC_LOG(LS_WARNING) << "Using default header extension map for SSRC "
>+                          << ssrc;
>+      extensions_maps.insert(std::make_pair(ssrc, default_extension_map_));
>+      return &default_extension_map_;
>+    }
>+  }
>+  return nullptr;
>+}
>+
>+// The packet must have space for at least IP_PACKET_SIZE bytes.
>+void ParsedRtcEventLogNew::GetRtcpPacket(size_t index,
>+                                         PacketDirection* incoming,
>+                                         uint8_t* packet,
>+                                         size_t* length) const {
>+  RTC_CHECK_LT(index, GetNumberOfEvents());
>+  const rtclog::Event& event = events_[index];
>+  GetRtcpPacket(event, incoming, packet, length);
>+}
>+
>+void ParsedRtcEventLogNew::GetRtcpPacket(const rtclog::Event& event,
>+                                         PacketDirection* incoming,
>+                                         uint8_t* packet,
>+                                         size_t* length) const {
>+  RTC_CHECK(event.has_type());
>+  RTC_CHECK_EQ(event.type(), rtclog::Event::RTCP_EVENT);
>+  RTC_CHECK(event.has_rtcp_packet());
>+  const rtclog::RtcpPacket& rtcp_packet = event.rtcp_packet();
>+  // Get direction of packet.
>+  RTC_CHECK(rtcp_packet.has_incoming());
>+  if (incoming != nullptr) {
>+    *incoming = rtcp_packet.incoming() ? kIncomingPacket : kOutgoingPacket;
>+  }
>+  // Get packet length.
>+  RTC_CHECK(rtcp_packet.has_packet_data());
>+  if (length != nullptr) {
>+    *length = rtcp_packet.packet_data().size();
>+  }
>+  // Get packet contents.
>+  if (packet != nullptr) {
>+    RTC_CHECK_LE(rtcp_packet.packet_data().size(),
>+                 static_cast<unsigned>(IP_PACKET_SIZE));
>+    memcpy(packet, rtcp_packet.packet_data().data(),
>+           rtcp_packet.packet_data().size());
>+  }
>+}
>+
>+rtclog::StreamConfig ParsedRtcEventLogNew::GetVideoReceiveConfig(
>+    size_t index) const {
>+  RTC_CHECK_LT(index, GetNumberOfEvents());
>+  return GetVideoReceiveConfig(events_[index]);
>+}
>+
>+rtclog::StreamConfig ParsedRtcEventLogNew::GetVideoReceiveConfig(
>+    const rtclog::Event& event) const {
>+  rtclog::StreamConfig config;
>+  RTC_CHECK(event.has_type());
>+  RTC_CHECK_EQ(event.type(), rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT);
>+  RTC_CHECK(event.has_video_receiver_config());
>+  const rtclog::VideoReceiveConfig& receiver_config =
>+      event.video_receiver_config();
>+  // Get SSRCs.
>+  RTC_CHECK(receiver_config.has_remote_ssrc());
>+  config.remote_ssrc = receiver_config.remote_ssrc();
>+  RTC_CHECK(receiver_config.has_local_ssrc());
>+  config.local_ssrc = receiver_config.local_ssrc();
>+  config.rtx_ssrc = 0;
>+  // Get RTCP settings.
>+  RTC_CHECK(receiver_config.has_rtcp_mode());
>+  config.rtcp_mode = GetRuntimeRtcpMode(receiver_config.rtcp_mode());
>+  RTC_CHECK(receiver_config.has_remb());
>+  config.remb = receiver_config.remb();
>+
>+  // Get RTX map.
>+  std::map<uint32_t, const rtclog::RtxConfig> rtx_map;
>+  for (int i = 0; i < receiver_config.rtx_map_size(); i++) {
>+    const rtclog::RtxMap& map = receiver_config.rtx_map(i);
>+    RTC_CHECK(map.has_payload_type());
>+    RTC_CHECK(map.has_config());
>+    RTC_CHECK(map.config().has_rtx_ssrc());
>+    RTC_CHECK(map.config().has_rtx_payload_type());
>+    rtx_map.insert(std::make_pair(map.payload_type(), map.config()));
>+  }
>+
>+  // Get header extensions.
>+  GetHeaderExtensions(&config.rtp_extensions,
>+                      receiver_config.header_extensions());
>+  // Get decoders.
>+  config.codecs.clear();
>+  for (int i = 0; i < receiver_config.decoders_size(); i++) {
>+    RTC_CHECK(receiver_config.decoders(i).has_name());
>+    RTC_CHECK(receiver_config.decoders(i).has_payload_type());
>+    int rtx_payload_type = 0;
>+    auto rtx_it = rtx_map.find(receiver_config.decoders(i).payload_type());
>+    if (rtx_it != rtx_map.end()) {
>+      rtx_payload_type = rtx_it->second.rtx_payload_type();
>+      if (config.rtx_ssrc != 0 &&
>+          config.rtx_ssrc != rtx_it->second.rtx_ssrc()) {
>+        RTC_LOG(LS_WARNING)
>+            << "RtcEventLog protobuf contained different SSRCs for "
>+               "different received RTX payload types. Will only use "
>+               "rtx_ssrc = "
>+            << config.rtx_ssrc << ".";
>+      } else {
>+        config.rtx_ssrc = rtx_it->second.rtx_ssrc();
>+      }
>+    }
>+    config.codecs.emplace_back(receiver_config.decoders(i).name(),
>+                               receiver_config.decoders(i).payload_type(),
>+                               rtx_payload_type);
>+  }
>+  return config;
>+}
>+
>+std::vector<rtclog::StreamConfig> ParsedRtcEventLogNew::GetVideoSendConfig(
>+    size_t index) const {
>+  RTC_CHECK_LT(index, GetNumberOfEvents());
>+  return GetVideoSendConfig(events_[index]);
>+}
>+
>+std::vector<rtclog::StreamConfig> ParsedRtcEventLogNew::GetVideoSendConfig(
>+    const rtclog::Event& event) const {
>+  std::vector<rtclog::StreamConfig> configs;
>+  RTC_CHECK(event.has_type());
>+  RTC_CHECK_EQ(event.type(), rtclog::Event::VIDEO_SENDER_CONFIG_EVENT);
>+  RTC_CHECK(event.has_video_sender_config());
>+  const rtclog::VideoSendConfig& sender_config = event.video_sender_config();
>+  if (sender_config.rtx_ssrcs_size() > 0 &&
>+      sender_config.ssrcs_size() != sender_config.rtx_ssrcs_size()) {
>+    RTC_LOG(WARNING)
>+        << "VideoSendConfig is configured for RTX but the number of "
>+           "SSRCs doesn't match the number of RTX SSRCs.";
>+  }
>+  configs.resize(sender_config.ssrcs_size());
>+  for (int i = 0; i < sender_config.ssrcs_size(); i++) {
>+    // Get SSRCs.
>+    configs[i].local_ssrc = sender_config.ssrcs(i);
>+    if (sender_config.rtx_ssrcs_size() > 0 &&
>+        i < sender_config.rtx_ssrcs_size()) {
>+      RTC_CHECK(sender_config.has_rtx_payload_type());
>+      configs[i].rtx_ssrc = sender_config.rtx_ssrcs(i);
>+    }
>+    // Get header extensions.
>+    GetHeaderExtensions(&configs[i].rtp_extensions,
>+                        sender_config.header_extensions());
>+
>+    // Get the codec.
>+    RTC_CHECK(sender_config.has_encoder());
>+    RTC_CHECK(sender_config.encoder().has_name());
>+    RTC_CHECK(sender_config.encoder().has_payload_type());
>+    configs[i].codecs.emplace_back(
>+        sender_config.encoder().name(), sender_config.encoder().payload_type(),
>+        sender_config.has_rtx_payload_type() ? sender_config.rtx_payload_type()
>+                                             : 0);
>+  }
>+  return configs;
>+}
>+
>+rtclog::StreamConfig ParsedRtcEventLogNew::GetAudioReceiveConfig(
>+    size_t index) const {
>+  RTC_CHECK_LT(index, GetNumberOfEvents());
>+  return GetAudioReceiveConfig(events_[index]);
>+}
>+
>+rtclog::StreamConfig ParsedRtcEventLogNew::GetAudioReceiveConfig(
>+    const rtclog::Event& event) const {
>+  rtclog::StreamConfig config;
>+  RTC_CHECK(event.has_type());
>+  RTC_CHECK_EQ(event.type(), rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT);
>+  RTC_CHECK(event.has_audio_receiver_config());
>+  const rtclog::AudioReceiveConfig& receiver_config =
>+      event.audio_receiver_config();
>+  // Get SSRCs.
>+  RTC_CHECK(receiver_config.has_remote_ssrc());
>+  config.remote_ssrc = receiver_config.remote_ssrc();
>+  RTC_CHECK(receiver_config.has_local_ssrc());
>+  config.local_ssrc = receiver_config.local_ssrc();
>+  // Get header extensions.
>+  GetHeaderExtensions(&config.rtp_extensions,
>+                      receiver_config.header_extensions());
>+  return config;
>+}
>+
>+rtclog::StreamConfig ParsedRtcEventLogNew::GetAudioSendConfig(
>+    size_t index) const {
>+  RTC_CHECK_LT(index, GetNumberOfEvents());
>+  return GetAudioSendConfig(events_[index]);
>+}
>+
>+rtclog::StreamConfig ParsedRtcEventLogNew::GetAudioSendConfig(
>+    const rtclog::Event& event) const {
>+  rtclog::StreamConfig config;
>+  RTC_CHECK(event.has_type());
>+  RTC_CHECK_EQ(event.type(), rtclog::Event::AUDIO_SENDER_CONFIG_EVENT);
>+  RTC_CHECK(event.has_audio_sender_config());
>+  const rtclog::AudioSendConfig& sender_config = event.audio_sender_config();
>+  // Get SSRCs.
>+  RTC_CHECK(sender_config.has_ssrc());
>+  config.local_ssrc = sender_config.ssrc();
>+  // Get header extensions.
>+  GetHeaderExtensions(&config.rtp_extensions,
>+                      sender_config.header_extensions());
>+  return config;
>+}
>+
>+LoggedAudioPlayoutEvent ParsedRtcEventLogNew::GetAudioPlayout(
>+    size_t index) const {
>+  RTC_CHECK_LT(index, GetNumberOfEvents());
>+  const rtclog::Event& event = events_[index];
>+  return GetAudioPlayout(event);
>+}
>+
>+LoggedAudioPlayoutEvent ParsedRtcEventLogNew::GetAudioPlayout(
>+    const rtclog::Event& event) const {
>+  RTC_CHECK(event.has_type());
>+  RTC_CHECK_EQ(event.type(), rtclog::Event::AUDIO_PLAYOUT_EVENT);
>+  RTC_CHECK(event.has_audio_playout_event());
>+  const rtclog::AudioPlayoutEvent& playout_event = event.audio_playout_event();
>+  LoggedAudioPlayoutEvent res;
>+  res.timestamp_us = GetTimestamp(event);
>+  RTC_CHECK(playout_event.has_local_ssrc());
>+  res.ssrc = playout_event.local_ssrc();
>+  return res;
>+}
>+
>+LoggedBweLossBasedUpdate ParsedRtcEventLogNew::GetLossBasedBweUpdate(
>+    size_t index) const {
>+  RTC_CHECK_LT(index, GetNumberOfEvents());
>+  const rtclog::Event& event = events_[index];
>+  return GetLossBasedBweUpdate(event);
>+}
>+
>+LoggedBweLossBasedUpdate ParsedRtcEventLogNew::GetLossBasedBweUpdate(
>+    const rtclog::Event& event) const {
>+  RTC_CHECK(event.has_type());
>+  RTC_CHECK_EQ(event.type(), rtclog::Event::LOSS_BASED_BWE_UPDATE);
>+  RTC_CHECK(event.has_loss_based_bwe_update());
>+  const rtclog::LossBasedBweUpdate& loss_event = event.loss_based_bwe_update();
>+
>+  LoggedBweLossBasedUpdate bwe_update;
>+  bwe_update.timestamp_us = GetTimestamp(event);
>+  RTC_CHECK(loss_event.has_bitrate_bps());
>+  bwe_update.bitrate_bps = loss_event.bitrate_bps();
>+  RTC_CHECK(loss_event.has_fraction_loss());
>+  bwe_update.fraction_lost = loss_event.fraction_loss();
>+  RTC_CHECK(loss_event.has_total_packets());
>+  bwe_update.expected_packets = loss_event.total_packets();
>+  return bwe_update;
>+}
>+
>+LoggedBweDelayBasedUpdate ParsedRtcEventLogNew::GetDelayBasedBweUpdate(
>+    size_t index) const {
>+  RTC_CHECK_LT(index, GetNumberOfEvents());
>+  const rtclog::Event& event = events_[index];
>+  return GetDelayBasedBweUpdate(event);
>+}
>+
>+LoggedBweDelayBasedUpdate ParsedRtcEventLogNew::GetDelayBasedBweUpdate(
>+    const rtclog::Event& event) const {
>+  RTC_CHECK(event.has_type());
>+  RTC_CHECK_EQ(event.type(), rtclog::Event::DELAY_BASED_BWE_UPDATE);
>+  RTC_CHECK(event.has_delay_based_bwe_update());
>+  const rtclog::DelayBasedBweUpdate& delay_event =
>+      event.delay_based_bwe_update();
>+
>+  LoggedBweDelayBasedUpdate res;
>+  res.timestamp_us = GetTimestamp(event);
>+  RTC_CHECK(delay_event.has_bitrate_bps());
>+  res.bitrate_bps = delay_event.bitrate_bps();
>+  RTC_CHECK(delay_event.has_detector_state());
>+  res.detector_state = GetRuntimeDetectorState(delay_event.detector_state());
>+  return res;
>+}
>+
>+LoggedAudioNetworkAdaptationEvent
>+ParsedRtcEventLogNew::GetAudioNetworkAdaptation(size_t index) const {
>+  RTC_CHECK_LT(index, GetNumberOfEvents());
>+  const rtclog::Event& event = events_[index];
>+  return GetAudioNetworkAdaptation(event);
>+}
>+
>+LoggedAudioNetworkAdaptationEvent
>+ParsedRtcEventLogNew::GetAudioNetworkAdaptation(
>+    const rtclog::Event& event) const {
>+  RTC_CHECK(event.has_type());
>+  RTC_CHECK_EQ(event.type(), rtclog::Event::AUDIO_NETWORK_ADAPTATION_EVENT);
>+  RTC_CHECK(event.has_audio_network_adaptation());
>+  const rtclog::AudioNetworkAdaptation& ana_event =
>+      event.audio_network_adaptation();
>+
>+  LoggedAudioNetworkAdaptationEvent res;
>+  res.timestamp_us = GetTimestamp(event);
>+  if (ana_event.has_bitrate_bps())
>+    res.config.bitrate_bps = ana_event.bitrate_bps();
>+  if (ana_event.has_enable_fec())
>+    res.config.enable_fec = ana_event.enable_fec();
>+  if (ana_event.has_enable_dtx())
>+    res.config.enable_dtx = ana_event.enable_dtx();
>+  if (ana_event.has_frame_length_ms())
>+    res.config.frame_length_ms = ana_event.frame_length_ms();
>+  if (ana_event.has_num_channels())
>+    res.config.num_channels = ana_event.num_channels();
>+  if (ana_event.has_uplink_packet_loss_fraction())
>+    res.config.uplink_packet_loss_fraction =
>+        ana_event.uplink_packet_loss_fraction();
>+  return res;
>+}
>+
>+LoggedBweProbeClusterCreatedEvent
>+ParsedRtcEventLogNew::GetBweProbeClusterCreated(size_t index) const {
>+  RTC_CHECK_LT(index, GetNumberOfEvents());
>+  const rtclog::Event& event = events_[index];
>+  return GetBweProbeClusterCreated(event);
>+}
>+
>+LoggedBweProbeClusterCreatedEvent
>+ParsedRtcEventLogNew::GetBweProbeClusterCreated(
>+    const rtclog::Event& event) const {
>+  RTC_CHECK(event.has_type());
>+  RTC_CHECK_EQ(event.type(), rtclog::Event::BWE_PROBE_CLUSTER_CREATED_EVENT);
>+  RTC_CHECK(event.has_probe_cluster());
>+  const rtclog::BweProbeCluster& pcc_event = event.probe_cluster();
>+  LoggedBweProbeClusterCreatedEvent res;
>+  res.timestamp_us = GetTimestamp(event);
>+  RTC_CHECK(pcc_event.has_id());
>+  res.id = pcc_event.id();
>+  RTC_CHECK(pcc_event.has_bitrate_bps());
>+  res.bitrate_bps = pcc_event.bitrate_bps();
>+  RTC_CHECK(pcc_event.has_min_packets());
>+  res.min_packets = pcc_event.min_packets();
>+  RTC_CHECK(pcc_event.has_min_bytes());
>+  res.min_bytes = pcc_event.min_bytes();
>+  return res;
>+}
>+
>+LoggedBweProbeFailureEvent ParsedRtcEventLogNew::GetBweProbeFailure(
>+    size_t index) const {
>+  RTC_CHECK_LT(index, GetNumberOfEvents());
>+  const rtclog::Event& event = events_[index];
>+  return GetBweProbeFailure(event);
>+}
>+
>+LoggedBweProbeFailureEvent ParsedRtcEventLogNew::GetBweProbeFailure(
>+    const rtclog::Event& event) const {
>+  RTC_CHECK(event.has_type());
>+  RTC_CHECK_EQ(event.type(), rtclog::Event::BWE_PROBE_RESULT_EVENT);
>+  RTC_CHECK(event.has_probe_result());
>+  const rtclog::BweProbeResult& pr_event = event.probe_result();
>+  RTC_CHECK(pr_event.has_result());
>+  RTC_CHECK_NE(pr_event.result(), rtclog::BweProbeResult::SUCCESS);
>+
>+  LoggedBweProbeFailureEvent res;
>+  res.timestamp_us = GetTimestamp(event);
>+  RTC_CHECK(pr_event.has_id());
>+  res.id = pr_event.id();
>+  RTC_CHECK(pr_event.has_result());
>+  if (pr_event.result() ==
>+      rtclog::BweProbeResult::INVALID_SEND_RECEIVE_INTERVAL) {
>+    res.failure_reason = ProbeFailureReason::kInvalidSendReceiveInterval;
>+  } else if (pr_event.result() ==
>+             rtclog::BweProbeResult::INVALID_SEND_RECEIVE_RATIO) {
>+    res.failure_reason = ProbeFailureReason::kInvalidSendReceiveRatio;
>+  } else if (pr_event.result() == rtclog::BweProbeResult::TIMEOUT) {
>+    res.failure_reason = ProbeFailureReason::kTimeout;
>+  } else {
>+    RTC_NOTREACHED();
>+  }
>+  RTC_CHECK(!pr_event.has_bitrate_bps());
>+
>+  return res;
>+}
>+
>+LoggedBweProbeSuccessEvent ParsedRtcEventLogNew::GetBweProbeSuccess(
>+    size_t index) const {
>+  RTC_CHECK_LT(index, GetNumberOfEvents());
>+  const rtclog::Event& event = events_[index];
>+  return GetBweProbeSuccess(event);
>+}
>+
>+LoggedBweProbeSuccessEvent ParsedRtcEventLogNew::GetBweProbeSuccess(
>+    const rtclog::Event& event) const {
>+  RTC_CHECK(event.has_type());
>+  RTC_CHECK_EQ(event.type(), rtclog::Event::BWE_PROBE_RESULT_EVENT);
>+  RTC_CHECK(event.has_probe_result());
>+  const rtclog::BweProbeResult& pr_event = event.probe_result();
>+  RTC_CHECK(pr_event.has_result());
>+  RTC_CHECK_EQ(pr_event.result(), rtclog::BweProbeResult::SUCCESS);
>+
>+  LoggedBweProbeSuccessEvent res;
>+  res.timestamp_us = GetTimestamp(event);
>+  RTC_CHECK(pr_event.has_id());
>+  res.id = pr_event.id();
>+  RTC_CHECK(pr_event.has_bitrate_bps());
>+  res.bitrate_bps = pr_event.bitrate_bps();
>+
>+  return res;
>+}
>+
>+LoggedAlrStateEvent ParsedRtcEventLogNew::GetAlrState(size_t index) const {
>+  RTC_CHECK_LT(index, GetNumberOfEvents());
>+  const rtclog::Event& event = events_[index];
>+  return GetAlrState(event);
>+}
>+
>+LoggedAlrStateEvent ParsedRtcEventLogNew::GetAlrState(
>+    const rtclog::Event& event) const {
>+  RTC_CHECK(event.has_type());
>+  RTC_CHECK_EQ(event.type(), rtclog::Event::ALR_STATE_EVENT);
>+  RTC_CHECK(event.has_alr_state());
>+  const rtclog::AlrState& alr_event = event.alr_state();
>+  LoggedAlrStateEvent res;
>+  res.timestamp_us = GetTimestamp(event);
>+  RTC_CHECK(alr_event.has_in_alr());
>+  res.in_alr = alr_event.in_alr();
>+
>+  return res;
>+}
>+
>+LoggedIceCandidatePairConfig ParsedRtcEventLogNew::GetIceCandidatePairConfig(
>+    size_t index) const {
>+  RTC_CHECK_LT(index, GetNumberOfEvents());
>+  const rtclog::Event& rtc_event = events_[index];
>+  return GetIceCandidatePairConfig(rtc_event);
>+}
>+
>+LoggedIceCandidatePairConfig ParsedRtcEventLogNew::GetIceCandidatePairConfig(
>+    const rtclog::Event& rtc_event) const {
>+  RTC_CHECK(rtc_event.has_type());
>+  RTC_CHECK_EQ(rtc_event.type(), rtclog::Event::ICE_CANDIDATE_PAIR_CONFIG);
>+  LoggedIceCandidatePairConfig res;
>+  const rtclog::IceCandidatePairConfig& config =
>+      rtc_event.ice_candidate_pair_config();
>+  res.timestamp_us = GetTimestamp(rtc_event);
>+  RTC_CHECK(config.has_config_type());
>+  res.type = GetRuntimeIceCandidatePairConfigType(config.config_type());
>+  RTC_CHECK(config.has_candidate_pair_id());
>+  res.candidate_pair_id = config.candidate_pair_id();
>+  RTC_CHECK(config.has_local_candidate_type());
>+  res.local_candidate_type =
>+      GetRuntimeIceCandidateType(config.local_candidate_type());
>+  RTC_CHECK(config.has_local_relay_protocol());
>+  res.local_relay_protocol =
>+      GetRuntimeIceCandidatePairProtocol(config.local_relay_protocol());
>+  RTC_CHECK(config.has_local_network_type());
>+  res.local_network_type =
>+      GetRuntimeIceCandidateNetworkType(config.local_network_type());
>+  RTC_CHECK(config.has_local_address_family());
>+  res.local_address_family =
>+      GetRuntimeIceCandidatePairAddressFamily(config.local_address_family());
>+  RTC_CHECK(config.has_remote_candidate_type());
>+  res.remote_candidate_type =
>+      GetRuntimeIceCandidateType(config.remote_candidate_type());
>+  RTC_CHECK(config.has_remote_address_family());
>+  res.remote_address_family =
>+      GetRuntimeIceCandidatePairAddressFamily(config.remote_address_family());
>+  RTC_CHECK(config.has_candidate_pair_protocol());
>+  res.candidate_pair_protocol =
>+      GetRuntimeIceCandidatePairProtocol(config.candidate_pair_protocol());
>+  return res;
>+}
>+
>+LoggedIceCandidatePairEvent ParsedRtcEventLogNew::GetIceCandidatePairEvent(
>+    size_t index) const {
>+  RTC_CHECK_LT(index, GetNumberOfEvents());
>+  const rtclog::Event& rtc_event = events_[index];
>+  return GetIceCandidatePairEvent(rtc_event);
>+}
>+
>+LoggedIceCandidatePairEvent ParsedRtcEventLogNew::GetIceCandidatePairEvent(
>+    const rtclog::Event& rtc_event) const {
>+  RTC_CHECK(rtc_event.has_type());
>+  RTC_CHECK_EQ(rtc_event.type(), rtclog::Event::ICE_CANDIDATE_PAIR_EVENT);
>+  LoggedIceCandidatePairEvent res;
>+  const rtclog::IceCandidatePairEvent& event =
>+      rtc_event.ice_candidate_pair_event();
>+  res.timestamp_us = GetTimestamp(rtc_event);
>+  RTC_CHECK(event.has_event_type());
>+  res.type = GetRuntimeIceCandidatePairEventType(event.event_type());
>+  RTC_CHECK(event.has_candidate_pair_id());
>+  res.candidate_pair_id = event.candidate_pair_id();
>+  return res;
>+}
>+
>+// Returns the MediaType for registered SSRCs. Search from the end to use last
>+// registered types first.
>+ParsedRtcEventLogNew::MediaType ParsedRtcEventLogNew::GetMediaType(
>+    uint32_t ssrc,
>+    PacketDirection direction) const {
>+  if (direction == kIncomingPacket) {
>+    if (std::find(incoming_video_ssrcs_.begin(), incoming_video_ssrcs_.end(),
>+                  ssrc) != incoming_video_ssrcs_.end()) {
>+      return MediaType::VIDEO;
>+    }
>+    if (std::find(incoming_audio_ssrcs_.begin(), incoming_audio_ssrcs_.end(),
>+                  ssrc) != incoming_audio_ssrcs_.end()) {
>+      return MediaType::AUDIO;
>+    }
>+  } else {
>+    if (std::find(outgoing_video_ssrcs_.begin(), outgoing_video_ssrcs_.end(),
>+                  ssrc) != outgoing_video_ssrcs_.end()) {
>+      return MediaType::VIDEO;
>+    }
>+    if (std::find(outgoing_audio_ssrcs_.begin(), outgoing_audio_ssrcs_.end(),
>+                  ssrc) != outgoing_audio_ssrcs_.end()) {
>+      return MediaType::AUDIO;
>+    }
>+  }
>+  return MediaType::ANY;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_parser_new.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_parser_new.h
>new file mode 100644
>index 00000000000..a4a130228c9
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_parser_new.h
>@@ -0,0 +1,982 @@
>+/*
>+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#ifndef LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_PARSER_NEW_H_
>+#define LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_PARSER_NEW_H_
>+
>+#include <iterator>
>+#include <map>
>+#include <set>
>+#include <string>
>+#include <utility>  // pair
>+#include <vector>
>+
>+#include "call/video_receive_stream.h"
>+#include "call/video_send_stream.h"
>+#include "logging/rtc_event_log/events/rtc_event_ice_candidate_pair.h"
>+#include "logging/rtc_event_log/events/rtc_event_ice_candidate_pair_config.h"
>+#include "logging/rtc_event_log/events/rtc_event_probe_result_failure.h"
>+#include "logging/rtc_event_log/rtc_event_log.h"
>+#include "logging/rtc_event_log/rtc_stream_config.h"
>+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
>+#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
>+#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
>+#include "modules/rtp_rtcp/source/rtcp_packet/nack.h"
>+#include "modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
>+#include "modules/rtp_rtcp/source/rtcp_packet/remb.h"
>+#include "modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
>+#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
>+#include "rtc_base/ignore_wundef.h"
>+
>+// Files generated at build-time by the protobuf compiler.
>+RTC_PUSH_IGNORING_WUNDEF()
>+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
>+#include "external/webrtc/webrtc/logging/rtc_event_log/rtc_event_log.pb.h"
>+#else
>+#include "logging/rtc_event_log/rtc_event_log.pb.h"
>+#endif
>+RTC_POP_IGNORING_WUNDEF()
>+
>+namespace webrtc {
>+
>+enum class BandwidthUsage;
>+struct AudioEncoderRuntimeConfig;
>+
>+struct LoggedAlrStateEvent {
>+  int64_t timestamp_us;
>+  bool in_alr;
>+  int64_t log_time_us() const { return timestamp_us; }
>+  int64_t log_time_ms() const { return timestamp_us / 1000; }
>+};
>+
>+struct LoggedAudioPlayoutEvent {
>+  int64_t timestamp_us;
>+  uint32_t ssrc;
>+  int64_t log_time_us() const { return timestamp_us; }
>+  int64_t log_time_ms() const { return timestamp_us / 1000; }
>+};
>+
>+struct LoggedAudioNetworkAdaptationEvent {
>+  int64_t timestamp_us;
>+  AudioEncoderRuntimeConfig config;
>+  int64_t log_time_us() const { return timestamp_us; }
>+  int64_t log_time_ms() const { return timestamp_us / 1000; }
>+};
>+
>+struct LoggedBweDelayBasedUpdate {
>+  int64_t timestamp_us;
>+  int32_t bitrate_bps;
>+  BandwidthUsage detector_state;
>+  int64_t log_time_us() const { return timestamp_us; }
>+  int64_t log_time_ms() const { return timestamp_us / 1000; }
>+};
>+
>+struct LoggedBweLossBasedUpdate {
>+  int64_t timestamp_us;
>+  int32_t bitrate_bps;
>+  uint8_t fraction_lost;
>+  int32_t expected_packets;
>+  int64_t log_time_us() const { return timestamp_us; }
>+  int64_t log_time_ms() const { return timestamp_us / 1000; }
>+};
>+
>+struct LoggedBweProbeClusterCreatedEvent {
>+  int64_t timestamp_us;
>+  int32_t id;
>+  int32_t bitrate_bps;
>+  uint32_t min_packets;
>+  uint32_t min_bytes;
>+  int64_t log_time_us() const { return timestamp_us; }
>+  int64_t log_time_ms() const { return timestamp_us / 1000; }
>+};
>+
>+struct LoggedBweProbeSuccessEvent {
>+  int64_t timestamp_us;
>+  int32_t id;
>+  int32_t bitrate_bps;
>+  int64_t log_time_us() const { return timestamp_us; }
>+  int64_t log_time_ms() const { return timestamp_us / 1000; }
>+};
>+
>+struct LoggedBweProbeFailureEvent {
>+  int64_t timestamp_us;
>+  int32_t id;
>+  ProbeFailureReason failure_reason;
>+  int64_t log_time_us() const { return timestamp_us; }
>+  int64_t log_time_ms() const { return timestamp_us / 1000; }
>+};
>+
>+struct LoggedIceCandidatePairConfig {
>+  int64_t timestamp_us;
>+  IceCandidatePairConfigType type;
>+  uint32_t candidate_pair_id;
>+  IceCandidateType local_candidate_type;
>+  IceCandidatePairProtocol local_relay_protocol;
>+  IceCandidateNetworkType local_network_type;
>+  IceCandidatePairAddressFamily local_address_family;
>+  IceCandidateType remote_candidate_type;
>+  IceCandidatePairAddressFamily remote_address_family;
>+  IceCandidatePairProtocol candidate_pair_protocol;
>+  int64_t log_time_us() const { return timestamp_us; }
>+  int64_t log_time_ms() const { return timestamp_us / 1000; }
>+};
>+
>+struct LoggedIceCandidatePairEvent {
>+  int64_t timestamp_us;
>+  IceCandidatePairEventType type;
>+  uint32_t candidate_pair_id;
>+  int64_t log_time_us() const { return timestamp_us; }
>+  int64_t log_time_ms() const { return timestamp_us / 1000; }
>+};
>+
>+struct LoggedRtpPacket {
>+  LoggedRtpPacket(uint64_t timestamp_us,
>+                  RTPHeader header,
>+                  size_t header_length,
>+                  size_t total_length)
>+      : timestamp_us(timestamp_us),
>+        header(header),
>+        header_length(header_length),
>+        total_length(total_length) {}
>+  int64_t timestamp_us;
>+  // TODO(terelius): This allocates space for 15 CSRCs even if none are used.
>+  RTPHeader header;
>+  size_t header_length;
>+  size_t total_length;
>+  int64_t log_time_us() const { return timestamp_us; }
>+  int64_t log_time_ms() const { return timestamp_us / 1000; }
>+};
>+
>+struct LoggedRtpPacketIncoming {
>+  LoggedRtpPacketIncoming(uint64_t timestamp_us,
>+                          RTPHeader header,
>+                          size_t header_length,
>+                          size_t total_length)
>+      : rtp(timestamp_us, header, header_length, total_length) {}
>+  LoggedRtpPacket rtp;
>+  int64_t log_time_us() const { return rtp.timestamp_us; }
>+  int64_t log_time_ms() const { return rtp.timestamp_us / 1000; }
>+};
>+
>+struct LoggedRtpPacketOutgoing {
>+  LoggedRtpPacketOutgoing(uint64_t timestamp_us,
>+                          RTPHeader header,
>+                          size_t header_length,
>+                          size_t total_length)
>+      : rtp(timestamp_us, header, header_length, total_length) {}
>+  LoggedRtpPacket rtp;
>+  int64_t log_time_us() const { return rtp.timestamp_us; }
>+  int64_t log_time_ms() const { return rtp.timestamp_us / 1000; }
>+};
>+
>+struct LoggedRtcpPacket {
>+  LoggedRtcpPacket(uint64_t timestamp_us,
>+                   const uint8_t* packet,
>+                   size_t total_length)
>+      : timestamp_us(timestamp_us), raw_data(packet, packet + total_length) {}
>+  int64_t timestamp_us;
>+  std::vector<uint8_t> raw_data;
>+  int64_t log_time_us() const { return timestamp_us; }
>+  int64_t log_time_ms() const { return timestamp_us / 1000; }
>+};
>+
>+struct LoggedRtcpPacketIncoming {
>+  LoggedRtcpPacketIncoming(uint64_t timestamp_us,
>+                           const uint8_t* packet,
>+                           size_t total_length)
>+      : rtcp(timestamp_us, packet, total_length) {}
>+  LoggedRtcpPacket rtcp;
>+  int64_t log_time_us() const { return rtcp.timestamp_us; }
>+  int64_t log_time_ms() const { return rtcp.timestamp_us / 1000; }
>+};
>+
>+struct LoggedRtcpPacketOutgoing {
>+  LoggedRtcpPacketOutgoing(uint64_t timestamp_us,
>+                           const uint8_t* packet,
>+                           size_t total_length)
>+      : rtcp(timestamp_us, packet, total_length) {}
>+  LoggedRtcpPacket rtcp;
>+  int64_t log_time_us() const { return rtcp.timestamp_us; }
>+  int64_t log_time_ms() const { return rtcp.timestamp_us / 1000; }
>+};
>+
>+struct LoggedRtcpPacketReceiverReport {
>+  int64_t timestamp_us;
>+  rtcp::ReceiverReport rr;
>+  int64_t log_time_us() const { return timestamp_us; }
>+  int64_t log_time_ms() const { return timestamp_us / 1000; }
>+};
>+
>+struct LoggedRtcpPacketSenderReport {
>+  int64_t timestamp_us;
>+  rtcp::SenderReport sr;
>+  int64_t log_time_us() const { return timestamp_us; }
>+  int64_t log_time_ms() const { return timestamp_us / 1000; }
>+};
>+
>+struct LoggedRtcpPacketRemb {
>+  int64_t timestamp_us;
>+  rtcp::Remb remb;
>+  int64_t log_time_us() const { return timestamp_us; }
>+  int64_t log_time_ms() const { return timestamp_us / 1000; }
>+};
>+
>+struct LoggedRtcpPacketNack {
>+  int64_t timestamp_us;
>+  rtcp::Nack nack;
>+  int64_t log_time_us() const { return timestamp_us; }
>+  int64_t log_time_ms() const { return timestamp_us / 1000; }
>+};
>+
>+struct LoggedRtcpPacketTransportFeedback {
>+  int64_t timestamp_us;
>+  rtcp::TransportFeedback transport_feedback;
>+  int64_t log_time_us() const { return timestamp_us; }
>+  int64_t log_time_ms() const { return timestamp_us / 1000; }
>+};
>+
>+struct LoggedStartEvent {
>+  explicit LoggedStartEvent(uint64_t timestamp_us)
>+      : timestamp_us(timestamp_us) {}
>+  int64_t timestamp_us;
>+  int64_t log_time_us() const { return timestamp_us; }
>+  int64_t log_time_ms() const { return timestamp_us / 1000; }
>+};
>+
>+struct LoggedStopEvent {
>+  explicit LoggedStopEvent(uint64_t timestamp_us)
>+      : timestamp_us(timestamp_us) {}
>+  int64_t timestamp_us;
>+  int64_t log_time_us() const { return timestamp_us; }
>+  int64_t log_time_ms() const { return timestamp_us / 1000; }
>+};
>+
>+struct LoggedAudioRecvConfig {
>+  LoggedAudioRecvConfig(int64_t timestamp_us, const rtclog::StreamConfig config)
>+      : timestamp_us(timestamp_us), config(config) {}
>+  int64_t timestamp_us;
>+  rtclog::StreamConfig config;
>+  int64_t log_time_us() const { return timestamp_us; }
>+  int64_t log_time_ms() const { return timestamp_us / 1000; }
>+};
>+
>+struct LoggedAudioSendConfig {
>+  LoggedAudioSendConfig(int64_t timestamp_us, const rtclog::StreamConfig config)
>+      : timestamp_us(timestamp_us), config(config) {}
>+  int64_t timestamp_us;
>+  rtclog::StreamConfig config;
>+  int64_t log_time_us() const { return timestamp_us; }
>+  int64_t log_time_ms() const { return timestamp_us / 1000; }
>+};
>+
>+struct LoggedVideoRecvConfig {
>+  LoggedVideoRecvConfig(int64_t timestamp_us, const rtclog::StreamConfig config)
>+      : timestamp_us(timestamp_us), config(config) {}
>+  int64_t timestamp_us;
>+  rtclog::StreamConfig config;
>+  int64_t log_time_us() const { return timestamp_us; }
>+  int64_t log_time_ms() const { return timestamp_us / 1000; }
>+};
>+
>+struct LoggedVideoSendConfig {
>+  LoggedVideoSendConfig(int64_t timestamp_us,
>+                        const std::vector<rtclog::StreamConfig> configs)
>+      : timestamp_us(timestamp_us), configs(configs) {}
>+  int64_t timestamp_us;
>+  std::vector<rtclog::StreamConfig> configs;
>+  int64_t log_time_us() const { return timestamp_us; }
>+  int64_t log_time_ms() const { return timestamp_us / 1000; }
>+};
>+
>+template <typename T>
>+class PacketView;
>+
>+template <typename T>
>+class PacketIterator {
>+  friend class PacketView<T>;
>+
>+ public:
>+  // Standard iterator traits.
>+  using difference_type = std::ptrdiff_t;
>+  using value_type = T;
>+  using pointer = T*;
>+  using reference = T&;
>+  using iterator_category = std::bidirectional_iterator_tag;
>+
>+  // The default-contructed iterator is meaningless, but is required by the
>+  // ForwardIterator concept.
>+  PacketIterator() : ptr_(nullptr), element_size_(0) {}
>+  PacketIterator(const PacketIterator& other)
>+      : ptr_(other.ptr_), element_size_(other.element_size_) {}
>+  PacketIterator(const PacketIterator&& other)
>+      : ptr_(other.ptr_), element_size_(other.element_size_) {}
>+  ~PacketIterator() = default;
>+
>+  PacketIterator& operator=(const PacketIterator& other) {
>+    ptr_ = other.ptr_;
>+    element_size_ = other.element_size_;
>+    return *this;
>+  }
>+  PacketIterator& operator=(const PacketIterator&& other) {
>+    ptr_ = other.ptr_;
>+    element_size_ = other.element_size_;
>+    return *this;
>+  }
>+
>+  bool operator==(const PacketIterator<T>& other) const {
>+    RTC_DCHECK_EQ(element_size_, other.element_size_);
>+    return ptr_ == other.ptr_;
>+  }
>+  bool operator!=(const PacketIterator<T>& other) const {
>+    RTC_DCHECK_EQ(element_size_, other.element_size_);
>+    return ptr_ != other.ptr_;
>+  }
>+
>+  PacketIterator& operator++() {
>+    ptr_ += element_size_;
>+    return *this;
>+  }
>+  PacketIterator& operator--() {
>+    ptr_ -= element_size_;
>+    return *this;
>+  }
>+  PacketIterator operator++(int) {
>+    PacketIterator iter_copy(ptr_, element_size_);
>+    ptr_ += element_size_;
>+    return iter_copy;
>+  }
>+  PacketIterator operator--(int) {
>+    PacketIterator iter_copy(ptr_, element_size_);
>+    ptr_ -= element_size_;
>+    return iter_copy;
>+  }
>+
>+  T& operator*() { return *reinterpret_cast<T*>(ptr_); }
>+  const T& operator*() const { return *reinterpret_cast<const T*>(ptr_); }
>+
>+ private:
>+  PacketIterator(typename std::conditional<std::is_const<T>::value,
>+                                           const void*,
>+                                           void*>::type p,
>+                 size_t s)
>+      : ptr_(reinterpret_cast<decltype(ptr_)>(p)), element_size_(s) {}
>+
>+  typename std::conditional<std::is_const<T>::value, const char*, char*>::type
>+      ptr_;
>+  size_t element_size_;
>+};
>+
>+// Suppose that we have a struct S where we are only interested in a specific
>+// member M. Given an array of S, PacketView can be used to treat the array
>+// as an array of M, without exposing the type S to surrounding code and without
>+// accessing the member through a virtual function. In this case, we want to
>+// have a common view for incoming and outgoing RtpPackets, hence the PacketView
>+// name.
>+// Note that constructing a PacketView bypasses the typesystem, so the caller
>+// has to take extra care when constructing these objects. The implementation
>+// also requires that the containing struct is standard-layout (e.g. POD).
>+//
>+// Usage example:
>+// struct A {...};
>+// struct B { A a; ...};
>+// struct C { A a; ...};
>+// size_t len = 10;
>+// B* array1 = new B[len];
>+// C* array2 = new C[len];
>+//
>+// PacketView<A> view1 = PacketView<A>::Create<B>(array1, len, offsetof(B, a));
>+// PacketView<A> view2 = PacketView<A>::Create<C>(array2, len, offsetof(C, a));
>+//
>+// The following code works with either view1 or view2.
>+// void f(PacketView<A> view)
>+// for (A& a : view) {
>+//   DoSomething(a);
>+// }
>+template <typename T>
>+class PacketView {
>+ public:
>+  template <typename U>
>+  static PacketView Create(U* ptr, size_t num_elements, size_t offset) {
>+    static_assert(std::is_standard_layout<U>::value,
>+                  "PacketView can only be created for standard layout types.");
>+    static_assert(std::is_standard_layout<T>::value,
>+                  "PacketView can only be created for standard layout types.");
>+    return PacketView(ptr, num_elements, offset, sizeof(U));
>+  }
>+
>+  using iterator = PacketIterator<T>;
>+  using const_iterator = PacketIterator<const T>;
>+  using reverse_iterator = std::reverse_iterator<iterator>;
>+  using const_reverse_iterator = std::reverse_iterator<const_iterator>;
>+
>+  iterator begin() { return iterator(data_, element_size_); }
>+  iterator end() {
>+    auto end_ptr = data_ + num_elements_ * element_size_;
>+    return iterator(end_ptr, element_size_);
>+  }
>+
>+  const_iterator begin() const { return const_iterator(data_, element_size_); }
>+  const_iterator end() const {
>+    auto end_ptr = data_ + num_elements_ * element_size_;
>+    return const_iterator(end_ptr, element_size_);
>+  }
>+
>+  reverse_iterator rbegin() { return reverse_iterator(end()); }
>+  reverse_iterator rend() { return reverse_iterator(begin()); }
>+
>+  const_reverse_iterator rbegin() const {
>+    return const_reverse_iterator(end());
>+  }
>+  const_reverse_iterator rend() const {
>+    return const_reverse_iterator(begin());
>+  }
>+
>+  size_t size() const { return num_elements_; }
>+
>+  T& operator[](size_t i) {
>+    auto elem_ptr = data_ + i * element_size_;
>+    return *reinterpret_cast<T*>(elem_ptr);
>+  }
>+
>+  const T& operator[](size_t i) const {
>+    auto elem_ptr = data_ + i * element_size_;
>+    return *reinterpret_cast<const T*>(elem_ptr);
>+  }
>+
>+ private:
>+  PacketView(typename std::conditional<std::is_const<T>::value,
>+                                       const void*,
>+                                       void*>::type data,
>+             size_t num_elements,
>+             size_t offset,
>+             size_t element_size)
>+      : data_(reinterpret_cast<decltype(data_)>(data) + offset),
>+        num_elements_(num_elements),
>+        element_size_(element_size) {}
>+
>+  typename std::conditional<std::is_const<T>::value, const char*, char*>::type
>+      data_;
>+  size_t num_elements_;
>+  size_t element_size_;
>+};
>+
>+class ParsedRtcEventLogNew {
>+  friend class RtcEventLogTestHelper;
>+
>+ public:
>+  enum class EventType {
>+    UNKNOWN_EVENT = 0,
>+    LOG_START = 1,
>+    LOG_END = 2,
>+    RTP_EVENT = 3,
>+    RTCP_EVENT = 4,
>+    AUDIO_PLAYOUT_EVENT = 5,
>+    LOSS_BASED_BWE_UPDATE = 6,
>+    DELAY_BASED_BWE_UPDATE = 7,
>+    VIDEO_RECEIVER_CONFIG_EVENT = 8,
>+    VIDEO_SENDER_CONFIG_EVENT = 9,
>+    AUDIO_RECEIVER_CONFIG_EVENT = 10,
>+    AUDIO_SENDER_CONFIG_EVENT = 11,
>+    AUDIO_NETWORK_ADAPTATION_EVENT = 16,
>+    BWE_PROBE_CLUSTER_CREATED_EVENT = 17,
>+    BWE_PROBE_FAILURE_EVENT = 18,
>+    BWE_PROBE_SUCCESS_EVENT = 19,
>+    ALR_STATE_EVENT = 20,
>+    ICE_CANDIDATE_PAIR_CONFIG = 21,
>+    ICE_CANDIDATE_PAIR_EVENT = 22,
>+  };
>+
>+  enum class MediaType { ANY, AUDIO, VIDEO, DATA };
>+  enum class UnconfiguredHeaderExtensions {
>+    kDontParse,
>+    kAttemptWebrtcDefaultConfig
>+  };
>+
>+  struct LoggedRtpStreamIncoming {
>+    uint32_t ssrc;
>+    std::vector<LoggedRtpPacketIncoming> incoming_packets;
>+  };
>+
>+  struct LoggedRtpStreamOutgoing {
>+    uint32_t ssrc;
>+    std::vector<LoggedRtpPacketOutgoing> outgoing_packets;
>+  };
>+
>+  struct LoggedRtpStreamView {
>+    LoggedRtpStreamView(uint32_t ssrc,
>+                        const LoggedRtpPacketIncoming* ptr,
>+                        size_t num_elements)
>+        : ssrc(ssrc),
>+          packet_view(PacketView<const LoggedRtpPacket>::Create(
>+              ptr,
>+              num_elements,
>+              offsetof(LoggedRtpPacketIncoming, rtp))) {}
>+    LoggedRtpStreamView(uint32_t ssrc,
>+                        const LoggedRtpPacketOutgoing* ptr,
>+                        size_t num_elements)
>+        : ssrc(ssrc),
>+          packet_view(PacketView<const LoggedRtpPacket>::Create(
>+              ptr,
>+              num_elements,
>+              offsetof(LoggedRtpPacketOutgoing, rtp))) {}
>+    uint32_t ssrc;
>+    PacketView<const LoggedRtpPacket> packet_view;
>+  };
>+
>+  explicit ParsedRtcEventLogNew(
>+      UnconfiguredHeaderExtensions parse_unconfigured_header_extensions =
>+          UnconfiguredHeaderExtensions::kDontParse);
>+
>+  // Clears previously parsed events and resets the ParsedRtcEventLogNew to an
>+  // empty state.
>+  void Clear();
>+
>+  // Reads an RtcEventLog file and returns true if parsing was successful.
>+  bool ParseFile(const std::string& file_name);
>+
>+  // Reads an RtcEventLog from a string and returns true if successful.
>+  bool ParseString(const std::string& s);
>+
>+  // Reads an RtcEventLog from an istream and returns true if successful.
>+  bool ParseStream(
>+      std::istream& stream);  // no-presubmit-check TODO(webrtc:8982)
>+
>+  // Returns the number of events in an EventStream.
>+  size_t GetNumberOfEvents() const;
>+
>+  // Reads the arrival timestamp (in microseconds) from a rtclog::Event.
>+  int64_t GetTimestamp(size_t index) const;
>+  int64_t GetTimestamp(const rtclog::Event& event) const;
>+
>+  // Reads the event type of the rtclog::Event at |index|.
>+  EventType GetEventType(size_t index) const;
>+  EventType GetEventType(const rtclog::Event& event) const;
>+
>+  // Reads the header, direction, header length and packet length from the RTP
>+  // event at |index|, and stores the values in the corresponding output
>+  // parameters. Each output parameter can be set to nullptr if that value
>+  // isn't needed.
>+  // NB: The header must have space for at least IP_PACKET_SIZE bytes.
>+  // Returns: a pointer to a header extensions map acquired from parsing
>+  // corresponding Audio/Video Sender/Receiver config events.
>+  // Warning: if the same SSRC is reused by both video and audio streams during
>+  // call, extensions maps may be incorrect (the last one would be returned).
>+  const webrtc::RtpHeaderExtensionMap* GetRtpHeader(
>+      size_t index,
>+      PacketDirection* incoming,
>+      uint8_t* header,
>+      size_t* header_length,
>+      size_t* total_length,
>+      int* probe_cluster_id) const;
>+  const webrtc::RtpHeaderExtensionMap* GetRtpHeader(
>+      const rtclog::Event& event,
>+      PacketDirection* incoming,
>+      uint8_t* header,
>+      size_t* header_length,
>+      size_t* total_length,
>+      int* probe_cluster_id) const;
>+
>+  // Reads packet, direction and packet length from the RTCP event at |index|,
>+  // and stores the values in the corresponding output parameters.
>+  // Each output parameter can be set to nullptr if that value isn't needed.
>+  // NB: The packet must have space for at least IP_PACKET_SIZE bytes.
>+  void GetRtcpPacket(size_t index,
>+                     PacketDirection* incoming,
>+                     uint8_t* packet,
>+                     size_t* length) const;
>+  void GetRtcpPacket(const rtclog::Event& event,
>+                     PacketDirection* incoming,
>+                     uint8_t* packet,
>+                     size_t* length) const;
>+
>+  // Reads a video receive config event to a StreamConfig struct.
>+  // Only the fields that are stored in the protobuf will be written.
>+  rtclog::StreamConfig GetVideoReceiveConfig(size_t index) const;
>+
>+  // Reads a video send config event to a StreamConfig struct. If the proto
>+  // contains multiple SSRCs and RTX SSRCs (this used to be the case for
>+  // simulcast streams) then we return one StreamConfig per SSRC,RTX_SSRC pair.
>+  // Only the fields that are stored in the protobuf will be written.
>+  std::vector<rtclog::StreamConfig> GetVideoSendConfig(size_t index) const;
>+
>+  // Reads a audio receive config event to a StreamConfig struct.
>+  // Only the fields that are stored in the protobuf will be written.
>+  rtclog::StreamConfig GetAudioReceiveConfig(size_t index) const;
>+
>+  // Reads a config event to a StreamConfig struct.
>+  // Only the fields that are stored in the protobuf will be written.
>+  rtclog::StreamConfig GetAudioSendConfig(size_t index) const;
>+
>+  // Reads the SSRC from the audio playout event at |index|. The SSRC is stored
>+  // in the output parameter ssrc. The output parameter can be set to nullptr
>+  // and in that case the function only asserts that the event is well formed.
>+  LoggedAudioPlayoutEvent GetAudioPlayout(size_t index) const;
>+
>+  // Reads bitrate, fraction loss (as defined in RFC 1889) and total number of
>+  // expected packets from the loss based BWE event at |index| and stores the
>+  // values in
>+  // the corresponding output parameters. Each output parameter can be set to
>+  // nullptr if that
>+  // value isn't needed.
>+  LoggedBweLossBasedUpdate GetLossBasedBweUpdate(size_t index) const;
>+
>+  // Reads bitrate and detector_state from the delay based BWE event at |index|
>+  // and stores the values in the corresponding output parameters. Each output
>+  // parameter can be set to nullptr if that
>+  // value isn't needed.
>+  LoggedBweDelayBasedUpdate GetDelayBasedBweUpdate(size_t index) const;
>+
>+  // Reads a audio network adaptation event to a (non-NULL)
>+  // AudioEncoderRuntimeConfig struct. Only the fields that are
>+  // stored in the protobuf will be written.
>+  LoggedAudioNetworkAdaptationEvent GetAudioNetworkAdaptation(
>+      size_t index) const;
>+
>+  LoggedBweProbeClusterCreatedEvent GetBweProbeClusterCreated(
>+      size_t index) const;
>+
>+  LoggedBweProbeFailureEvent GetBweProbeFailure(size_t index) const;
>+  LoggedBweProbeSuccessEvent GetBweProbeSuccess(size_t index) const;
>+
>+  MediaType GetMediaType(uint32_t ssrc, PacketDirection direction) const;
>+
>+  LoggedAlrStateEvent GetAlrState(size_t index) const;
>+
>+  LoggedIceCandidatePairConfig GetIceCandidatePairConfig(size_t index) const;
>+
>+  LoggedIceCandidatePairEvent GetIceCandidatePairEvent(size_t index) const;
>+
>+  // Configured SSRCs.
>+  const std::set<uint32_t>& incoming_rtx_ssrcs() const {
>+    return incoming_rtx_ssrcs_;
>+  }
>+
>+  const std::set<uint32_t>& incoming_video_ssrcs() const {
>+    return incoming_video_ssrcs_;
>+  }
>+
>+  const std::set<uint32_t>& incoming_audio_ssrcs() const {
>+    return incoming_audio_ssrcs_;
>+  }
>+
>+  const std::set<uint32_t>& outgoing_rtx_ssrcs() const {
>+    return outgoing_rtx_ssrcs_;
>+  }
>+
>+  const std::set<uint32_t>& outgoing_video_ssrcs() const {
>+    return outgoing_video_ssrcs_;
>+  }
>+
>+  const std::set<uint32_t>& outgoing_audio_ssrcs() const {
>+    return outgoing_audio_ssrcs_;
>+  }
>+
>+  // Stream configurations.
>+  const std::vector<LoggedAudioRecvConfig>& audio_recv_configs() const {
>+    return audio_recv_configs_;
>+  }
>+
>+  const std::vector<LoggedAudioSendConfig>& audio_send_configs() const {
>+    return audio_send_configs_;
>+  }
>+
>+  const std::vector<LoggedVideoRecvConfig>& video_recv_configs() const {
>+    return video_recv_configs_;
>+  }
>+
>+  const std::vector<LoggedVideoSendConfig>& video_send_configs() const {
>+    return video_send_configs_;
>+  }
>+
>+  // Beginning and end of log segments.
>+  const std::vector<LoggedStartEvent>& start_log_events() const {
>+    return start_log_events_;
>+  }
>+
>+  const std::vector<LoggedStopEvent>& stop_log_events() const {
>+    return stop_log_events_;
>+  }
>+
>+  // Audio
>+  const std::map<uint32_t, std::vector<LoggedAudioPlayoutEvent>>&
>+  audio_playout_events() const {
>+    return audio_playout_events_;
>+  }
>+
>+  const std::vector<LoggedAudioNetworkAdaptationEvent>&
>+  audio_network_adaptation_events() const {
>+    return audio_network_adaptation_events_;
>+  }
>+
>+  // Bandwidth estimation
>+  const std::vector<LoggedBweProbeClusterCreatedEvent>&
>+  bwe_probe_cluster_created_events() const {
>+    return bwe_probe_cluster_created_events_;
>+  }
>+
>+  const std::vector<LoggedBweProbeFailureEvent>& bwe_probe_failure_events()
>+      const {
>+    return bwe_probe_failure_events_;
>+  }
>+
>+  const std::vector<LoggedBweProbeSuccessEvent>& bwe_probe_success_events()
>+      const {
>+    return bwe_probe_success_events_;
>+  }
>+
>+  const std::vector<LoggedBweDelayBasedUpdate>& bwe_delay_updates() const {
>+    return bwe_delay_updates_;
>+  }
>+
>+  const std::vector<LoggedBweLossBasedUpdate>& bwe_loss_updates() const {
>+    return bwe_loss_updates_;
>+  }
>+
>+  const std::vector<LoggedAlrStateEvent>& alr_state_events() const {
>+    return alr_state_events_;
>+  }
>+
>+  // ICE events
>+  const std::vector<LoggedIceCandidatePairConfig>& ice_candidate_pair_configs()
>+      const {
>+    return ice_candidate_pair_configs_;
>+  }
>+
>+  const std::vector<LoggedIceCandidatePairEvent>& ice_candidate_pair_events()
>+      const {
>+    return ice_candidate_pair_events_;
>+  }
>+
>+  // RTP
>+  const std::vector<LoggedRtpStreamIncoming>& incoming_rtp_packets_by_ssrc()
>+      const {
>+    return incoming_rtp_packets_by_ssrc_;
>+  }
>+
>+  const std::vector<LoggedRtpStreamOutgoing>& outgoing_rtp_packets_by_ssrc()
>+      const {
>+    return outgoing_rtp_packets_by_ssrc_;
>+  }
>+
>+  const std::vector<LoggedRtpStreamView>& rtp_packets_by_ssrc(
>+      PacketDirection direction) const {
>+    if (direction == kIncomingPacket)
>+      return incoming_rtp_packet_views_by_ssrc_;
>+    else
>+      return outgoing_rtp_packet_views_by_ssrc_;
>+  }
>+
>+  // RTCP
>+  const std::vector<LoggedRtcpPacketIncoming>& incoming_rtcp_packets() const {
>+    return incoming_rtcp_packets_;
>+  }
>+
>+  const std::vector<LoggedRtcpPacketOutgoing>& outgoing_rtcp_packets() const {
>+    return outgoing_rtcp_packets_;
>+  }
>+
>+  const std::vector<LoggedRtcpPacketReceiverReport>& receiver_reports(
>+      PacketDirection direction) const {
>+    if (direction == kIncomingPacket) {
>+      return incoming_rr_;
>+    } else {
>+      return outgoing_rr_;
>+    }
>+  }
>+
>+  const std::vector<LoggedRtcpPacketSenderReport>& sender_reports(
>+      PacketDirection direction) const {
>+    if (direction == kIncomingPacket) {
>+      return incoming_sr_;
>+    } else {
>+      return outgoing_sr_;
>+    }
>+  }
>+
>+  const std::vector<LoggedRtcpPacketNack>& nacks(
>+      PacketDirection direction) const {
>+    if (direction == kIncomingPacket) {
>+      return incoming_nack_;
>+    } else {
>+      return outgoing_nack_;
>+    }
>+  }
>+
>+  const std::vector<LoggedRtcpPacketRemb>& rembs(
>+      PacketDirection direction) const {
>+    if (direction == kIncomingPacket) {
>+      return incoming_remb_;
>+    } else {
>+      return outgoing_remb_;
>+    }
>+  }
>+
>+  const std::vector<LoggedRtcpPacketTransportFeedback>& transport_feedbacks(
>+      PacketDirection direction) const {
>+    if (direction == kIncomingPacket) {
>+      return incoming_transport_feedback_;
>+    } else {
>+      return outgoing_transport_feedback_;
>+    }
>+  }
>+
>+  int64_t first_timestamp() const { return first_timestamp_; }
>+  int64_t last_timestamp() const { return last_timestamp_; }
>+
>+ private:
>+  bool ParseStreamInternal(
>+      std::istream& stream);  // no-presubmit-check TODO(webrtc:8982)
>+
>+  void StoreParsedEvent(const rtclog::Event& event);
>+
>+  rtclog::StreamConfig GetVideoReceiveConfig(const rtclog::Event& event) const;
>+  std::vector<rtclog::StreamConfig> GetVideoSendConfig(
>+      const rtclog::Event& event) const;
>+  rtclog::StreamConfig GetAudioReceiveConfig(const rtclog::Event& event) const;
>+  rtclog::StreamConfig GetAudioSendConfig(const rtclog::Event& event) const;
>+
>+  LoggedAudioPlayoutEvent GetAudioPlayout(const rtclog::Event& event) const;
>+
>+  LoggedBweLossBasedUpdate GetLossBasedBweUpdate(
>+      const rtclog::Event& event) const;
>+  LoggedBweDelayBasedUpdate GetDelayBasedBweUpdate(
>+      const rtclog::Event& event) const;
>+
>+  LoggedAudioNetworkAdaptationEvent GetAudioNetworkAdaptation(
>+      const rtclog::Event& event) const;
>+
>+  LoggedBweProbeClusterCreatedEvent GetBweProbeClusterCreated(
>+      const rtclog::Event& event) const;
>+  LoggedBweProbeFailureEvent GetBweProbeFailure(
>+      const rtclog::Event& event) const;
>+  LoggedBweProbeSuccessEvent GetBweProbeSuccess(
>+      const rtclog::Event& event) const;
>+
>+  LoggedAlrStateEvent GetAlrState(const rtclog::Event& event) const;
>+
>+  LoggedIceCandidatePairConfig GetIceCandidatePairConfig(
>+      const rtclog::Event& event) const;
>+  LoggedIceCandidatePairEvent GetIceCandidatePairEvent(
>+      const rtclog::Event& event) const;
>+
>+  std::vector<rtclog::Event> events_;
>+
>+  struct Stream {
>+    Stream(uint32_t ssrc,
>+           MediaType media_type,
>+           PacketDirection direction,
>+           webrtc::RtpHeaderExtensionMap map)
>+        : ssrc(ssrc),
>+          media_type(media_type),
>+          direction(direction),
>+          rtp_extensions_map(map) {}
>+    uint32_t ssrc;
>+    MediaType media_type;
>+    PacketDirection direction;
>+    webrtc::RtpHeaderExtensionMap rtp_extensions_map;
>+  };
>+
>+  const UnconfiguredHeaderExtensions parse_unconfigured_header_extensions_;
>+
>+  // Make a default extension map for streams without configuration information.
>+  // TODO(ivoc): Once configuration of audio streams is stored in the event log,
>+  //             this can be removed. Tracking bug: webrtc:6399
>+  RtpHeaderExtensionMap default_extension_map_;
>+
>+  // Tracks what each stream is configured for. Note that a single SSRC can be
>+  // in several sets. For example, the SSRC used for sending video over RTX
>+  // will appear in both video_ssrcs_ and rtx_ssrcs_. In the unlikely case that
>+  // an SSRC is reconfigured to a different media type mid-call, it will also
>+  // appear in multiple sets.
>+  std::set<uint32_t> incoming_rtx_ssrcs_;
>+  std::set<uint32_t> incoming_video_ssrcs_;
>+  std::set<uint32_t> incoming_audio_ssrcs_;
>+  std::set<uint32_t> outgoing_rtx_ssrcs_;
>+  std::set<uint32_t> outgoing_video_ssrcs_;
>+  std::set<uint32_t> outgoing_audio_ssrcs_;
>+
>+  // Maps an SSRC to the parsed  RTP headers in that stream. Header extensions
>+  // are parsed if the stream has been configured. This is only used for
>+  // grouping the events by SSRC during parsing; the events are moved to
>+  // incoming_rtp_packets_by_ssrc_ once the parsing is done.
>+  std::map<uint32_t, std::vector<LoggedRtpPacketIncoming>>
>+      incoming_rtp_packets_map_;
>+  std::map<uint32_t, std::vector<LoggedRtpPacketOutgoing>>
>+      outgoing_rtp_packets_map_;
>+
>+  // RTP headers.
>+  std::vector<LoggedRtpStreamIncoming> incoming_rtp_packets_by_ssrc_;
>+  std::vector<LoggedRtpStreamOutgoing> outgoing_rtp_packets_by_ssrc_;
>+  std::vector<LoggedRtpStreamView> incoming_rtp_packet_views_by_ssrc_;
>+  std::vector<LoggedRtpStreamView> outgoing_rtp_packet_views_by_ssrc_;
>+
>+  // Raw RTCP packets.
>+  std::vector<LoggedRtcpPacketIncoming> incoming_rtcp_packets_;
>+  std::vector<LoggedRtcpPacketOutgoing> outgoing_rtcp_packets_;
>+
>+  // Parsed RTCP messages. Currently not separated based on SSRC.
>+  std::vector<LoggedRtcpPacketReceiverReport> incoming_rr_;
>+  std::vector<LoggedRtcpPacketReceiverReport> outgoing_rr_;
>+  std::vector<LoggedRtcpPacketSenderReport> incoming_sr_;
>+  std::vector<LoggedRtcpPacketSenderReport> outgoing_sr_;
>+  std::vector<LoggedRtcpPacketNack> incoming_nack_;
>+  std::vector<LoggedRtcpPacketNack> outgoing_nack_;
>+  std::vector<LoggedRtcpPacketRemb> incoming_remb_;
>+  std::vector<LoggedRtcpPacketRemb> outgoing_remb_;
>+  std::vector<LoggedRtcpPacketTransportFeedback> incoming_transport_feedback_;
>+  std::vector<LoggedRtcpPacketTransportFeedback> outgoing_transport_feedback_;
>+
>+  std::vector<LoggedStartEvent> start_log_events_;
>+  std::vector<LoggedStopEvent> stop_log_events_;
>+
>+  std::map<uint32_t, std::vector<LoggedAudioPlayoutEvent>>
>+      audio_playout_events_;
>+
>+  std::vector<LoggedAudioNetworkAdaptationEvent>
>+      audio_network_adaptation_events_;
>+
>+  std::vector<LoggedBweProbeClusterCreatedEvent>
>+      bwe_probe_cluster_created_events_;
>+
>+  std::vector<LoggedBweProbeFailureEvent> bwe_probe_failure_events_;
>+  std::vector<LoggedBweProbeSuccessEvent> bwe_probe_success_events_;
>+
>+  std::vector<LoggedBweDelayBasedUpdate> bwe_delay_updates_;
>+
>+  // A list of all updates from the send-side loss-based bandwidth estimator.
>+  std::vector<LoggedBweLossBasedUpdate> bwe_loss_updates_;
>+
>+  std::vector<LoggedAlrStateEvent> alr_state_events_;
>+
>+  std::vector<LoggedIceCandidatePairConfig> ice_candidate_pair_configs_;
>+
>+  std::vector<LoggedIceCandidatePairEvent> ice_candidate_pair_events_;
>+
>+  std::vector<LoggedAudioRecvConfig> audio_recv_configs_;
>+  std::vector<LoggedAudioSendConfig> audio_send_configs_;
>+  std::vector<LoggedVideoRecvConfig> video_recv_configs_;
>+  std::vector<LoggedVideoSendConfig> video_send_configs_;
>+
>+  uint8_t last_incoming_rtcp_packet_[IP_PACKET_SIZE];
>+  uint8_t last_incoming_rtcp_packet_length_;
>+
>+  int64_t first_timestamp_;
>+  int64_t last_timestamp_;
>+
>+  // The extension maps are mutable to allow us to insert the default
>+  // configuration when parsing an RTP header for an unconfigured stream.
>+  mutable std::map<uint32_t, webrtc::RtpHeaderExtensionMap>
>+      incoming_rtp_extensions_maps_;
>+  mutable std::map<uint32_t, webrtc::RtpHeaderExtensionMap>
>+      outgoing_rtp_extensions_maps_;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_PARSER_NEW_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc
>index 1cea3138ab3..e238f848846 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc
>@@ -8,15 +8,15 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>+#include <algorithm>
> #include <map>
> #include <memory>
>-#include <ostream>
> #include <string>
> #include <tuple>
> #include <utility>
> #include <vector>
> 
>-#include "call/call.h"
>+#include "absl/memory/memory.h"
> #include "logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h"
> #include "logging/rtc_event_log/events/rtc_event_audio_playout.h"
> #include "logging/rtc_event_log/events/rtc_event_audio_receive_stream_config.h"
>@@ -34,21 +34,13 @@
> #include "logging/rtc_event_log/events/rtc_event_video_send_stream_config.h"
> #include "logging/rtc_event_log/output/rtc_event_log_output_file.h"
> #include "logging/rtc_event_log/rtc_event_log.h"
>-#include "logging/rtc_event_log/rtc_event_log_parser.h"
>+#include "logging/rtc_event_log/rtc_event_log_parser_new.h"
> #include "logging/rtc_event_log/rtc_event_log_unittest_helper.h"
> #include "logging/rtc_event_log/rtc_stream_config.h"
>-#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
>-#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
> #include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
>-#include "modules/rtp_rtcp/source/rtcp_packet.h"
>-#include "modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
> #include "modules/rtp_rtcp/source/rtp_header_extensions.h"
>-#include "modules/rtp_rtcp/source/rtp_packet_received.h"
>-#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
>-#include "rtc_base/buffer.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/fakeclock.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/random.h"
> #include "test/gtest.h"
> #include "test/testsupport/fileutils.h"
>@@ -57,662 +49,653 @@ namespace webrtc {
> 
> namespace {
> 
>-const uint8_t kTransmissionTimeOffsetExtensionId = 1;
>-const uint8_t kAbsoluteSendTimeExtensionId = 14;
>-const uint8_t kTransportSequenceNumberExtensionId = 13;
>-const uint8_t kAudioLevelExtensionId = 9;
>-const uint8_t kVideoRotationExtensionId = 5;
>-
>-const uint8_t kExtensionIds[] = {
>-    kTransmissionTimeOffsetExtensionId, kAbsoluteSendTimeExtensionId,
>-    kTransportSequenceNumberExtensionId, kAudioLevelExtensionId,
>-    kVideoRotationExtensionId};
>-const RTPExtensionType kExtensionTypes[] = {
>-    RTPExtensionType::kRtpExtensionTransmissionTimeOffset,
>-    RTPExtensionType::kRtpExtensionAbsoluteSendTime,
>-    RTPExtensionType::kRtpExtensionTransportSequenceNumber,
>-    RTPExtensionType::kRtpExtensionAudioLevel,
>-    RTPExtensionType::kRtpExtensionVideoRotation};
>-const char* kExtensionNames[] = {
>-    RtpExtension::kTimestampOffsetUri, RtpExtension::kAbsSendTimeUri,
>-    RtpExtension::kTransportSequenceNumberUri, RtpExtension::kAudioLevelUri,
>-    RtpExtension::kVideoRotationUri};
>-
>-const size_t kNumExtensions = 5;
>-
>-struct BweLossEvent {
>-  int32_t bitrate_bps;
>-  uint8_t fraction_loss;
>-  int32_t total_packets;
>-};
>-
>-// TODO(terelius): Merge with event type in parser once updated?
>-enum class EventType {
>-  kIncomingRtp,
>-  kOutgoingRtp,
>-  kIncomingRtcp,
>-  kOutgoingRtcp,
>-  kAudioPlayout,
>-  kBweLossUpdate,
>-  kBweDelayUpdate,
>-  kVideoRecvConfig,
>-  kVideoSendConfig,
>-  kAudioRecvConfig,
>-  kAudioSendConfig,
>-  kAudioNetworkAdaptation,
>-  kBweProbeClusterCreated,
>-  kBweProbeResult,
>-};
>-
>-const std::map<EventType, std::string> event_type_to_string(
>-    {{EventType::kIncomingRtp, "RTP(in)"},
>-     {EventType::kOutgoingRtp, "RTP(out)"},
>-     {EventType::kIncomingRtcp, "RTCP(in)"},
>-     {EventType::kOutgoingRtcp, "RTCP(out)"},
>-     {EventType::kAudioPlayout, "PLAYOUT"},
>-     {EventType::kBweLossUpdate, "BWE_LOSS"},
>-     {EventType::kBweDelayUpdate, "BWE_DELAY"},
>-     {EventType::kVideoRecvConfig, "VIDEO_RECV_CONFIG"},
>-     {EventType::kVideoSendConfig, "VIDEO_SEND_CONFIG"},
>-     {EventType::kAudioRecvConfig, "AUDIO_RECV_CONFIG"},
>-     {EventType::kAudioSendConfig, "AUDIO_SEND_CONFIG"},
>-     {EventType::kAudioNetworkAdaptation, "AUDIO_NETWORK_ADAPTATION"},
>-     {EventType::kBweProbeClusterCreated, "BWE_PROBE_CREATED"},
>-     {EventType::kBweProbeResult, "BWE_PROBE_RESULT"}});
>-
>-const std::map<ParsedRtcEventLog::EventType, std::string>
>-    parsed_event_type_to_string(
>-        {{ParsedRtcEventLog::EventType::UNKNOWN_EVENT, "UNKNOWN_EVENT"},
>-         {ParsedRtcEventLog::EventType::LOG_START, "LOG_START"},
>-         {ParsedRtcEventLog::EventType::LOG_END, "LOG_END"},
>-         {ParsedRtcEventLog::EventType::RTP_EVENT, "RTP"},
>-         {ParsedRtcEventLog::EventType::RTCP_EVENT, "RTCP"},
>-         {ParsedRtcEventLog::EventType::AUDIO_PLAYOUT_EVENT, "AUDIO_PLAYOUT"},
>-         {ParsedRtcEventLog::EventType::LOSS_BASED_BWE_UPDATE,
>-          "LOSS_BASED_BWE_UPDATE"},
>-         {ParsedRtcEventLog::EventType::DELAY_BASED_BWE_UPDATE,
>-          "DELAY_BASED_BWE_UPDATE"},
>-         {ParsedRtcEventLog::EventType::VIDEO_RECEIVER_CONFIG_EVENT,
>-          "VIDEO_RECV_CONFIG"},
>-         {ParsedRtcEventLog::EventType::VIDEO_SENDER_CONFIG_EVENT,
>-          "VIDEO_SEND_CONFIG"},
>-         {ParsedRtcEventLog::EventType::AUDIO_RECEIVER_CONFIG_EVENT,
>-          "AUDIO_RECV_CONFIG"},
>-         {ParsedRtcEventLog::EventType::AUDIO_SENDER_CONFIG_EVENT,
>-          "AUDIO_SEND_CONFIG"},
>-         {ParsedRtcEventLog::EventType::AUDIO_NETWORK_ADAPTATION_EVENT,
>-          "AUDIO_NETWORK_ADAPTATION"},
>-         {ParsedRtcEventLog::EventType::BWE_PROBE_CLUSTER_CREATED_EVENT,
>-          "BWE_PROBE_CREATED"},
>-         {ParsedRtcEventLog::EventType::BWE_PROBE_RESULT_EVENT,
>-          "BWE_PROBE_RESULT"}});
>-}  // namespace
>-
>-void PrintActualEvents(const ParsedRtcEventLog& parsed_log,
>-                       std::ostream& stream);
>+RtpHeaderExtensionMap ExtensionMapWithAllSupportedExtensions() {
>+  RtpHeaderExtensionMap all_extensions;
>+  all_extensions.Register<AudioLevel>(RtpExtension::kAudioLevelDefaultId);
>+  all_extensions.Register<TransmissionOffset>(
>+      RtpExtension::kTimestampOffsetDefaultId);
>+  all_extensions.Register<AbsoluteSendTime>(
>+      RtpExtension::kAbsSendTimeDefaultId);
>+  all_extensions.Register<VideoOrientation>(
>+      RtpExtension::kVideoRotationDefaultId);
>+  all_extensions.Register<TransportSequenceNumber>(
>+      RtpExtension::kTransportSequenceNumberDefaultId);
>+  return all_extensions;
>+}
> 
>-RtpPacketToSend GenerateOutgoingRtpPacket(
>-    const RtpHeaderExtensionMap* extensions,
>-    uint32_t csrcs_count,
>-    size_t packet_size,
>-    Random* prng) {
>-  RTC_CHECK_GE(packet_size, 16 + 4 * csrcs_count + 4 * kNumExtensions);
>+struct EventCounts {
>+  size_t audio_send_streams = 0;
>+  size_t audio_recv_streams = 0;
>+  size_t video_send_streams = 0;
>+  size_t video_recv_streams = 0;
>+  size_t alr_states = 0;
>+  size_t audio_playouts = 0;
>+  size_t ana_configs = 0;
>+  size_t bwe_loss_events = 0;
>+  size_t bwe_delay_events = 0;
>+  size_t probe_creations = 0;
>+  size_t probe_successes = 0;
>+  size_t probe_failures = 0;
>+  size_t ice_configs = 0;
>+  size_t ice_events = 0;
>+  size_t incoming_rtp_packets = 0;
>+  size_t outgoing_rtp_packets = 0;
>+  size_t incoming_rtcp_packets = 0;
>+  size_t outgoing_rtcp_packets = 0;
>+
>+  size_t total_nonconfig_events() const {
>+    return alr_states + audio_playouts + ana_configs + bwe_loss_events +
>+           bwe_delay_events + probe_creations + probe_successes +
>+           probe_failures + ice_configs + ice_events + incoming_rtp_packets +
>+           outgoing_rtp_packets + incoming_rtcp_packets + outgoing_rtcp_packets;
>+  }
> 
>-  std::vector<uint32_t> csrcs;
>-  for (unsigned i = 0; i < csrcs_count; i++) {
>-    csrcs.push_back(prng->Rand<uint32_t>());
>+  size_t total_config_events() const {
>+    return audio_send_streams + audio_recv_streams + video_send_streams +
>+           video_recv_streams;
>   }
> 
>-  RtpPacketToSend rtp_packet(extensions, packet_size);
>-  rtp_packet.SetPayloadType(prng->Rand(127));
>-  rtp_packet.SetMarker(prng->Rand<bool>());
>-  rtp_packet.SetSequenceNumber(prng->Rand<uint16_t>());
>-  rtp_packet.SetSsrc(prng->Rand<uint32_t>());
>-  rtp_packet.SetTimestamp(prng->Rand<uint32_t>());
>-  rtp_packet.SetCsrcs(csrcs);
>-
>-  rtp_packet.SetExtension<TransmissionOffset>(prng->Rand(0x00ffffff));
>-  rtp_packet.SetExtension<AudioLevel>(prng->Rand<bool>(), prng->Rand(127));
>-  rtp_packet.SetExtension<AbsoluteSendTime>(prng->Rand(0x00ffffff));
>-  rtp_packet.SetExtension<VideoOrientation>(prng->Rand(2));
>-  rtp_packet.SetExtension<TransportSequenceNumber>(prng->Rand<uint16_t>());
>-
>-  size_t payload_size = packet_size - rtp_packet.headers_size();
>-  uint8_t* payload = rtp_packet.AllocatePayload(payload_size);
>-  for (size_t i = 0; i < payload_size; i++) {
>-    payload[i] = prng->Rand<uint8_t>();
>+  size_t total_events() const {
>+    return total_nonconfig_events() + total_config_events();
>   }
>-  return rtp_packet;
>-}
>+};
> 
>-RtpPacketReceived GenerateIncomingRtpPacket(
>-    const RtpHeaderExtensionMap* extensions,
>-    uint32_t csrcs_count,
>-    size_t packet_size,
>-    Random* prng) {
>-  RtpPacketToSend packet_out =
>-      GenerateOutgoingRtpPacket(extensions, csrcs_count, packet_size, prng);
>-  RtpPacketReceived packet_in(extensions);
>-  packet_in.Parse(packet_out.data(), packet_out.size());
>-  return packet_in;
>-}
>+class RtcEventLogSession
>+    : public ::testing::TestWithParam<std::tuple<uint64_t, int64_t>> {
>+ public:
>+  RtcEventLogSession()
>+      : seed_(std::get<0>(GetParam())),
>+        prng_(seed_),
>+        gen_(seed_ * 880001UL),
>+        output_period_ms_(std::get<1>(GetParam())) {
>+    clock_.SetTimeMicros(prng_.Rand<uint32_t>());
>+    // Find the name of the current test, in order to use it as a temporary
>+    // filename.
>+    // TODO(terelius): Use a general utility function to generate a temp file.
>+    auto test_info = ::testing::UnitTest::GetInstance()->current_test_info();
>+    std::string test_name =
>+        std::string(test_info->test_case_name()) + "_" + test_info->name();
>+    std::replace(test_name.begin(), test_name.end(), '/', '_');
>+    temp_filename_ = test::OutputPath() + test_name;
>+  }
> 
>-rtc::Buffer GenerateRtcpPacket(Random* prng) {
>-  rtcp::ReportBlock report_block;
>-  report_block.SetMediaSsrc(prng->Rand<uint32_t>());  // Remote SSRC.
>-  report_block.SetFractionLost(prng->Rand(50));
>+  // Create and buffer the config events and |num_events_before_log_start|
>+  // randomized non-config events. Then call StartLogging and finally create and
>+  // write the remaining non-config events.
>+  void WriteLog(EventCounts count, size_t num_events_before_log_start);
>+  void ReadAndVerifyLog();
> 
>-  rtcp::SenderReport sender_report;
>-  sender_report.SetSenderSsrc(prng->Rand<uint32_t>());
>-  sender_report.SetNtp(NtpTime(prng->Rand<uint32_t>(), prng->Rand<uint32_t>()));
>-  sender_report.SetPacketCount(prng->Rand<uint32_t>());
>-  sender_report.AddReportBlock(report_block);
>+ private:
>+  void WriteAudioRecvConfigs(size_t audio_recv_streams, RtcEventLog* event_log);
>+  void WriteAudioSendConfigs(size_t audio_send_streams, RtcEventLog* event_log);
>+  void WriteVideoRecvConfigs(size_t video_recv_streams, RtcEventLog* event_log);
>+  void WriteVideoSendConfigs(size_t video_send_streams, RtcEventLog* event_log);
>+
>+  std::vector<std::pair<uint32_t, RtpHeaderExtensionMap>> incoming_extensions_;
>+  std::vector<std::pair<uint32_t, RtpHeaderExtensionMap>> outgoing_extensions_;
>+
>+  // Config events.
>+  std::vector<std::unique_ptr<RtcEventAudioSendStreamConfig>>
>+      audio_send_config_list_;
>+  std::vector<std::unique_ptr<RtcEventAudioReceiveStreamConfig>>
>+      audio_recv_config_list_;
>+  std::vector<std::unique_ptr<RtcEventVideoSendStreamConfig>>
>+      video_send_config_list_;
>+  std::vector<std::unique_ptr<RtcEventVideoReceiveStreamConfig>>
>+      video_recv_config_list_;
>+
>+  // Regular events.
>+  std::vector<std::unique_ptr<RtcEventAlrState>> alr_state_list_;
>+  std::map<uint32_t, std::vector<std::unique_ptr<RtcEventAudioPlayout>>>
>+      audio_playout_map_;  // Groups audio by SSRC.
>+  std::vector<std::unique_ptr<RtcEventAudioNetworkAdaptation>>
>+      ana_configs_list_;
>+  std::vector<std::unique_ptr<RtcEventBweUpdateLossBased>> bwe_loss_list_;
>+  std::vector<std::unique_ptr<RtcEventBweUpdateDelayBased>> bwe_delay_list_;
>+  std::vector<std::unique_ptr<RtcEventProbeClusterCreated>>
>+      probe_creation_list_;
>+  std::vector<std::unique_ptr<RtcEventProbeResultSuccess>> probe_success_list_;
>+  std::vector<std::unique_ptr<RtcEventProbeResultFailure>> probe_failure_list_;
>+  std::vector<std::unique_ptr<RtcEventIceCandidatePairConfig>> ice_config_list_;
>+  std::vector<std::unique_ptr<RtcEventIceCandidatePair>> ice_event_list_;
>+  std::map<uint32_t, std::vector<std::unique_ptr<RtcEventRtpPacketIncoming>>>
>+      incoming_rtp_map_;  // Groups incoming RTP by SSRC.
>+  std::map<uint32_t, std::vector<std::unique_ptr<RtcEventRtpPacketOutgoing>>>
>+      outgoing_rtp_map_;  // Groups outgoing RTP by SSRC.
>+  std::vector<std::unique_ptr<RtcEventRtcpPacketIncoming>> incoming_rtcp_list_;
>+  std::vector<std::unique_ptr<RtcEventRtcpPacketOutgoing>> outgoing_rtcp_list_;
>+
>+  int64_t start_time_us_;
>+  int64_t stop_time_us_;
>+
>+  const uint64_t seed_;
>+  Random prng_;
>+  test::EventGenerator gen_;
>+  int64_t output_period_ms_;
>+  rtc::ScopedFakeClock clock_;
>+  std::string temp_filename_;
>+};
> 
>-  return sender_report.Build();
>+bool SsrcUsed(
>+    uint32_t ssrc,
>+    const std::vector<std::pair<uint32_t, RtpHeaderExtensionMap>>& streams) {
>+  for (const auto& kv : streams) {
>+    if (kv.first == ssrc)
>+      return true;
>+  }
>+  return false;
> }
> 
>-void GenerateVideoReceiveConfig(const RtpHeaderExtensionMap& extensions,
>-                                rtclog::StreamConfig* config,
>-                                Random* prng) {
>-  // Add SSRCs for the stream.
>-  config->remote_ssrc = prng->Rand<uint32_t>();
>-  config->local_ssrc = prng->Rand<uint32_t>();
>-  // Add extensions and settings for RTCP.
>-  config->rtcp_mode =
>-      prng->Rand<bool>() ? RtcpMode::kCompound : RtcpMode::kReducedSize;
>-  config->remb = prng->Rand<bool>();
>-  config->rtx_ssrc = prng->Rand<uint32_t>();
>-  config->codecs.emplace_back(prng->Rand<bool>() ? "VP8" : "H264",
>-                              prng->Rand(1, 127), prng->Rand(1, 127));
>-  // Add header extensions.
>-  for (unsigned i = 0; i < kNumExtensions; i++) {
>-    uint8_t id = extensions.GetId(kExtensionTypes[i]);
>-    if (id != RtpHeaderExtensionMap::kInvalidId) {
>-      config->rtp_extensions.emplace_back(kExtensionNames[i], id);
>-    }
>+void RtcEventLogSession::WriteAudioRecvConfigs(size_t audio_recv_streams,
>+                                               RtcEventLog* event_log) {
>+  RTC_CHECK(event_log != nullptr);
>+  uint32_t ssrc;
>+  for (size_t i = 0; i < audio_recv_streams; i++) {
>+    clock_.AdvanceTimeMicros(prng_.Rand(20) * 1000);
>+    do {
>+      ssrc = prng_.Rand<uint32_t>();
>+    } while (SsrcUsed(ssrc, incoming_extensions_));
>+    RtpHeaderExtensionMap extensions = gen_.NewRtpHeaderExtensionMap();
>+    incoming_extensions_.emplace_back(ssrc, extensions);
>+    auto event = gen_.NewAudioReceiveStreamConfig(ssrc, extensions);
>+    event_log->Log(event->Copy());
>+    audio_recv_config_list_.push_back(std::move(event));
>   }
> }
> 
>-void GenerateVideoSendConfig(const RtpHeaderExtensionMap& extensions,
>-                             rtclog::StreamConfig* config,
>-                             Random* prng) {
>-  config->codecs.emplace_back(prng->Rand<bool>() ? "VP8" : "H264",
>-                              prng->Rand(1, 127), prng->Rand(1, 127));
>-  config->local_ssrc = prng->Rand<uint32_t>();
>-  config->rtx_ssrc = prng->Rand<uint32_t>();
>-  // Add header extensions.
>-  for (unsigned i = 0; i < kNumExtensions; i++) {
>-    uint8_t id = extensions.GetId(kExtensionTypes[i]);
>-    if (id != RtpHeaderExtensionMap::kInvalidId) {
>-      config->rtp_extensions.emplace_back(kExtensionNames[i], id);
>-    }
>+void RtcEventLogSession::WriteAudioSendConfigs(size_t audio_send_streams,
>+                                               RtcEventLog* event_log) {
>+  RTC_CHECK(event_log != nullptr);
>+  uint32_t ssrc;
>+  for (size_t i = 0; i < audio_send_streams; i++) {
>+    clock_.AdvanceTimeMicros(prng_.Rand(20) * 1000);
>+    do {
>+      ssrc = prng_.Rand<uint32_t>();
>+    } while (SsrcUsed(ssrc, outgoing_extensions_));
>+    RtpHeaderExtensionMap extensions = gen_.NewRtpHeaderExtensionMap();
>+    outgoing_extensions_.emplace_back(ssrc, extensions);
>+    auto event = gen_.NewAudioSendStreamConfig(ssrc, extensions);
>+    event_log->Log(event->Copy());
>+    audio_send_config_list_.push_back(std::move(event));
>   }
> }
> 
>-void GenerateAudioReceiveConfig(const RtpHeaderExtensionMap& extensions,
>-                                rtclog::StreamConfig* config,
>-                                Random* prng) {
>-  // Add SSRCs for the stream.
>-  config->remote_ssrc = prng->Rand<uint32_t>();
>-  config->local_ssrc = prng->Rand<uint32_t>();
>-  // Add header extensions.
>-  for (unsigned i = 0; i < kNumExtensions; i++) {
>-    uint8_t id = extensions.GetId(kExtensionTypes[i]);
>-    if (id != RtpHeaderExtensionMap::kInvalidId) {
>-      config->rtp_extensions.emplace_back(kExtensionNames[i], id);
>-    }
>+void RtcEventLogSession::WriteVideoRecvConfigs(size_t video_recv_streams,
>+                                               RtcEventLog* event_log) {
>+  RTC_CHECK(event_log != nullptr);
>+  RTC_CHECK_GE(video_recv_streams, 1);
>+
>+  // Force least one stream to use all header extensions, to ensure
>+  // (statistically) that every extension is tested in packet creation.
>+  RtpHeaderExtensionMap all_extensions =
>+      ExtensionMapWithAllSupportedExtensions();
>+
>+  clock_.AdvanceTimeMicros(prng_.Rand(20) * 1000);
>+  uint32_t ssrc = prng_.Rand<uint32_t>();
>+  incoming_extensions_.emplace_back(prng_.Rand<uint32_t>(), all_extensions);
>+  auto event = gen_.NewVideoReceiveStreamConfig(ssrc, all_extensions);
>+  event_log->Log(event->Copy());
>+  video_recv_config_list_.push_back(std::move(event));
>+  for (size_t i = 1; i < video_recv_streams; i++) {
>+    clock_.AdvanceTimeMicros(prng_.Rand(20) * 1000);
>+    do {
>+      ssrc = prng_.Rand<uint32_t>();
>+    } while (SsrcUsed(ssrc, incoming_extensions_));
>+    RtpHeaderExtensionMap extensions = gen_.NewRtpHeaderExtensionMap();
>+    incoming_extensions_.emplace_back(ssrc, extensions);
>+    auto event = gen_.NewVideoReceiveStreamConfig(ssrc, extensions);
>+    event_log->Log(event->Copy());
>+    video_recv_config_list_.push_back(std::move(event));
>   }
> }
> 
>-void GenerateAudioSendConfig(const RtpHeaderExtensionMap& extensions,
>-                             rtclog::StreamConfig* config,
>-                             Random* prng) {
>-  // Add SSRC to the stream.
>-  config->local_ssrc = prng->Rand<uint32_t>();
>-  // Add header extensions.
>-  for (unsigned i = 0; i < kNumExtensions; i++) {
>-    uint8_t id = extensions.GetId(kExtensionTypes[i]);
>-    if (id != RtpHeaderExtensionMap::kInvalidId) {
>-      config->rtp_extensions.emplace_back(kExtensionNames[i], id);
>-    }
>+void RtcEventLogSession::WriteVideoSendConfigs(size_t video_send_streams,
>+                                               RtcEventLog* event_log) {
>+  RTC_CHECK(event_log != nullptr);
>+  RTC_CHECK_GE(video_send_streams, 1);
>+
>+  // Force least one stream to use all header extensions, to ensure
>+  // (statistically) that every extension is tested in packet creation.
>+  RtpHeaderExtensionMap all_extensions =
>+      ExtensionMapWithAllSupportedExtensions();
>+
>+  clock_.AdvanceTimeMicros(prng_.Rand(20) * 1000);
>+  uint32_t ssrc = prng_.Rand<uint32_t>();
>+  outgoing_extensions_.emplace_back(prng_.Rand<uint32_t>(), all_extensions);
>+  auto event = gen_.NewVideoSendStreamConfig(ssrc, all_extensions);
>+  event_log->Log(event->Copy());
>+  video_send_config_list_.push_back(std::move(event));
>+  for (size_t i = 1; i < video_send_streams; i++) {
>+    clock_.AdvanceTimeMicros(prng_.Rand(20) * 1000);
>+    do {
>+      ssrc = prng_.Rand<uint32_t>();
>+    } while (SsrcUsed(ssrc, outgoing_extensions_));
>+    RtpHeaderExtensionMap extensions = gen_.NewRtpHeaderExtensionMap();
>+    outgoing_extensions_.emplace_back(ssrc, extensions);
>+    auto event = gen_.NewVideoSendStreamConfig(ssrc, extensions);
>+    event_log->Log(event->Copy());
>+    video_send_config_list_.push_back(std::move(event));
>   }
> }
> 
>-BweLossEvent GenerateBweLossEvent(Random* prng) {
>-  BweLossEvent loss_event;
>-  loss_event.bitrate_bps = prng->Rand(6000, 10000000);
>-  loss_event.fraction_loss = prng->Rand<uint8_t>();
>-  loss_event.total_packets = prng->Rand(1, 1000);
>-  return loss_event;
>-}
>+void RtcEventLogSession::WriteLog(EventCounts count,
>+                                  size_t num_events_before_start) {
>+  // TODO(terelius): Allow test to run with either a real or a fake clock_.
>+  // Maybe always use the ScopedFakeClock, but conditionally SleepMs()?
> 
>-void GenerateAudioNetworkAdaptation(const RtpHeaderExtensionMap& extensions,
>-                                    AudioEncoderRuntimeConfig* config,
>-                                    Random* prng) {
>-  config->bitrate_bps = prng->Rand(0, 3000000);
>-  config->enable_fec = prng->Rand<bool>();
>-  config->enable_dtx = prng->Rand<bool>();
>-  config->frame_length_ms = prng->Rand(10, 120);
>-  config->num_channels = prng->Rand(1, 2);
>-  config->uplink_packet_loss_fraction = prng->Rand<float>();
>-}
>+  // The log file will be flushed to disk when the event_log goes out of scope.
>+  std::unique_ptr<RtcEventLog> event_log(
>+      RtcEventLog::Create(RtcEventLog::EncodingType::Legacy));
> 
>-class RtcEventLogSession
>-    : public ::testing::TestWithParam<std::tuple<uint64_t, int64_t>> {
>- public:
>-  RtcEventLogSession()
>-      : prng(std::get<0>(GetParam())),
>-        output_period_ms(std::get<1>(GetParam())) {}
>-  void GenerateSessionDescription(size_t incoming_rtp_count,
>-                                  size_t outgoing_rtp_count,
>-                                  size_t incoming_rtcp_count,
>-                                  size_t outgoing_rtcp_count,
>-                                  size_t playout_count,
>-                                  size_t bwe_loss_count,
>-                                  size_t bwe_delay_count,
>-                                  const RtpHeaderExtensionMap& extensions,
>-                                  uint32_t csrcs_count);
>-  void WriteSession();
>-  void ReadAndVerifySession();
>-  void PrintExpectedEvents(std::ostream& stream);
>+  // We can't send or receive packets without configured streams.
>+  RTC_CHECK_GE(count.video_recv_streams, 1);
>+  RTC_CHECK_GE(count.video_send_streams, 1);
>+
>+  WriteAudioRecvConfigs(count.audio_recv_streams, event_log.get());
>+  WriteAudioSendConfigs(count.audio_send_streams, event_log.get());
>+  WriteVideoRecvConfigs(count.video_recv_streams, event_log.get());
>+  WriteVideoSendConfigs(count.video_send_streams, event_log.get());
>+
>+  size_t remaining_events = count.total_nonconfig_events();
>+  ASSERT_LE(num_events_before_start, remaining_events);
>+  size_t remaining_events_at_start = remaining_events - num_events_before_start;
>+  for (; remaining_events > 0; remaining_events--) {
>+    if (remaining_events == remaining_events_at_start) {
>+      clock_.AdvanceTimeMicros(prng_.Rand(20) * 1000);
>+      event_log->StartLogging(
>+          absl::make_unique<RtcEventLogOutputFile>(temp_filename_, 10000000),
>+          output_period_ms_);
>+      start_time_us_ = rtc::TimeMicros();
>+    }
> 
>- private:
>-  std::vector<RtpPacketReceived> incoming_rtp_packets;
>-  std::vector<RtpPacketToSend> outgoing_rtp_packets;
>-  std::vector<rtc::Buffer> incoming_rtcp_packets;
>-  std::vector<rtc::Buffer> outgoing_rtcp_packets;
>-  std::vector<uint32_t> playout_ssrcs;
>-  std::vector<BweLossEvent> bwe_loss_updates;
>-  std::vector<std::pair<int32_t, BandwidthUsage> > bwe_delay_updates;
>-  std::vector<rtclog::StreamConfig> receiver_configs;
>-  std::vector<rtclog::StreamConfig> sender_configs;
>-  std::vector<EventType> event_types;
>-  Random prng;
>-  int64_t output_period_ms;
>-};
>+    clock_.AdvanceTimeMicros(prng_.Rand(20) * 1000);
>+    size_t selection = prng_.Rand(remaining_events - 1);
> 
>-void RtcEventLogSession::GenerateSessionDescription(
>-    size_t incoming_rtp_count,
>-    size_t outgoing_rtp_count,
>-    size_t incoming_rtcp_count,
>-    size_t outgoing_rtcp_count,
>-    size_t playout_count,
>-    size_t bwe_loss_count,
>-    size_t bwe_delay_count,
>-    const RtpHeaderExtensionMap& extensions,
>-    uint32_t csrcs_count) {
>-  // Create configuration for the video receive stream.
>-  receiver_configs.push_back(rtclog::StreamConfig());
>-  GenerateVideoReceiveConfig(extensions, &receiver_configs.back(), &prng);
>-  event_types.push_back(EventType::kVideoRecvConfig);
>-
>-  // Create configuration for the video send stream.
>-  sender_configs.push_back(rtclog::StreamConfig());
>-  GenerateVideoSendConfig(extensions, &sender_configs.back(), &prng);
>-  event_types.push_back(EventType::kVideoSendConfig);
>-  const size_t config_count = 2;
>-
>-  // Create incoming and outgoing RTP packets containing random data.
>-  for (size_t i = 0; i < incoming_rtp_count; i++) {
>-    size_t packet_size = prng.Rand(1000, 1100);
>-    incoming_rtp_packets.push_back(GenerateIncomingRtpPacket(
>-        &extensions, csrcs_count, packet_size, &prng));
>-    event_types.push_back(EventType::kIncomingRtp);
>+    if (selection < count.alr_states) {
>+      auto event = gen_.NewAlrState();
>+      event_log->Log(event->Copy());
>+      alr_state_list_.push_back(std::move(event));
>+      count.alr_states--;
>+      continue;
>+    }
>+    selection -= count.alr_states;
>+
>+    if (selection < count.audio_playouts) {
>+      size_t stream = prng_.Rand(incoming_extensions_.size() - 1);
>+      // This might be a video SSRC, but the parser does not use the config.
>+      uint32_t ssrc = incoming_extensions_[stream].first;
>+      auto event = gen_.NewAudioPlayout(ssrc);
>+      event_log->Log(event->Copy());
>+      audio_playout_map_[ssrc].push_back(std::move(event));
>+      count.audio_playouts--;
>+      continue;
>+    }
>+    selection -= count.audio_playouts;
>+
>+    if (selection < count.ana_configs) {
>+      auto event = gen_.NewAudioNetworkAdaptation();
>+      event_log->Log(event->Copy());
>+      ana_configs_list_.push_back(std::move(event));
>+      count.ana_configs--;
>+      continue;
>+    }
>+    selection -= count.ana_configs;
>+
>+    if (selection < count.bwe_loss_events) {
>+      auto event = gen_.NewBweUpdateLossBased();
>+      event_log->Log(event->Copy());
>+      bwe_loss_list_.push_back(std::move(event));
>+      count.bwe_loss_events--;
>+      continue;
>+    }
>+    selection -= count.bwe_loss_events;
>+
>+    if (selection < count.bwe_delay_events) {
>+      auto event = gen_.NewBweUpdateDelayBased();
>+      event_log->Log(event->Copy());
>+      bwe_delay_list_.push_back(std::move(event));
>+      count.bwe_delay_events--;
>+      continue;
>+    }
>+    selection -= count.bwe_delay_events;
>+
>+    if (selection < count.probe_creations) {
>+      auto event = gen_.NewProbeClusterCreated();
>+      event_log->Log(event->Copy());
>+      probe_creation_list_.push_back(std::move(event));
>+      count.probe_creations--;
>+      continue;
>+    }
>+    selection -= count.probe_creations;
>+
>+    if (selection < count.probe_successes) {
>+      auto event = gen_.NewProbeResultSuccess();
>+      event_log->Log(event->Copy());
>+      probe_success_list_.push_back(std::move(event));
>+      count.probe_successes--;
>+      continue;
>+    }
>+    selection -= count.probe_successes;
>+
>+    if (selection < count.probe_failures) {
>+      auto event = gen_.NewProbeResultFailure();
>+      event_log->Log(event->Copy());
>+      probe_failure_list_.push_back(std::move(event));
>+      count.probe_failures--;
>+      continue;
>+    }
>+    selection -= count.probe_failures;
>+
>+    if (selection < count.ice_configs) {
>+      auto event = gen_.NewIceCandidatePairConfig();
>+      event_log->Log(event->Copy());
>+      ice_config_list_.push_back(std::move(event));
>+      count.ice_configs--;
>+      continue;
>+    }
>+    selection -= count.ice_configs;
>+
>+    if (selection < count.ice_events) {
>+      auto event = gen_.NewIceCandidatePair();
>+      event_log->Log(event->Copy());
>+      ice_event_list_.push_back(std::move(event));
>+      count.ice_events--;
>+      continue;
>+    }
>+    selection -= count.ice_events;
>+
>+    if (selection < count.incoming_rtp_packets) {
>+      size_t stream = prng_.Rand(incoming_extensions_.size() - 1);
>+      uint32_t ssrc = incoming_extensions_[stream].first;
>+      auto event =
>+          gen_.NewRtpPacketIncoming(ssrc, incoming_extensions_[stream].second);
>+      event_log->Log(event->Copy());
>+      incoming_rtp_map_[ssrc].push_back(std::move(event));
>+      count.incoming_rtp_packets--;
>+      continue;
>+    }
>+    selection -= count.incoming_rtp_packets;
>+
>+    if (selection < count.outgoing_rtp_packets) {
>+      size_t stream = prng_.Rand(outgoing_extensions_.size() - 1);
>+      uint32_t ssrc = outgoing_extensions_[stream].first;
>+      auto event =
>+          gen_.NewRtpPacketOutgoing(ssrc, outgoing_extensions_[stream].second);
>+      event_log->Log(event->Copy());
>+      outgoing_rtp_map_[ssrc].push_back(std::move(event));
>+      count.outgoing_rtp_packets--;
>+      continue;
>+    }
>+    selection -= count.outgoing_rtp_packets;
>+
>+    if (selection < count.incoming_rtcp_packets) {
>+      auto event = gen_.NewRtcpPacketIncoming();
>+      event_log->Log(event->Copy());
>+      incoming_rtcp_list_.push_back(std::move(event));
>+      count.incoming_rtcp_packets--;
>+      continue;
>+    }
>+    selection -= count.incoming_rtcp_packets;
>+
>+    if (selection < count.outgoing_rtcp_packets) {
>+      auto event = gen_.NewRtcpPacketOutgoing();
>+      event_log->Log(event->Copy());
>+      outgoing_rtcp_list_.push_back(std::move(event));
>+      count.outgoing_rtcp_packets--;
>+      continue;
>+    }
>+    selection -= count.outgoing_rtcp_packets;
>+
>+    RTC_NOTREACHED();
>   }
>-  for (size_t i = 0; i < outgoing_rtp_count; i++) {
>-    size_t packet_size = prng.Rand(1000, 1100);
>-    outgoing_rtp_packets.push_back(GenerateOutgoingRtpPacket(
>-        &extensions, csrcs_count, packet_size, &prng));
>-    event_types.push_back(EventType::kOutgoingRtp);
>+
>+  event_log->StopLogging();
>+  stop_time_us_ = rtc::TimeMicros();
>+
>+  ASSERT_EQ(count.total_nonconfig_events(), static_cast<size_t>(0));
>+}
>+
>+// Read the file and verify that what we read back from the event log is the
>+// same as what we wrote down.
>+void RtcEventLogSession::ReadAndVerifyLog() {
>+  // Read the generated file from disk.
>+  ParsedRtcEventLogNew parsed_log;
>+  ASSERT_TRUE(parsed_log.ParseFile(temp_filename_));
>+
>+  // Start and stop events.
>+  auto& parsed_start_log_events = parsed_log.start_log_events();
>+  ASSERT_EQ(parsed_start_log_events.size(), static_cast<size_t>(1));
>+  EXPECT_TRUE(
>+      test::VerifyLoggedStartEvent(start_time_us_, parsed_start_log_events[0]));
>+
>+  auto& parsed_stop_log_events = parsed_log.stop_log_events();
>+  ASSERT_EQ(parsed_stop_log_events.size(), static_cast<size_t>(1));
>+  EXPECT_TRUE(
>+      test::VerifyLoggedStopEvent(stop_time_us_, parsed_stop_log_events[0]));
>+
>+  auto& parsed_alr_state_events = parsed_log.alr_state_events();
>+  ASSERT_EQ(parsed_alr_state_events.size(), alr_state_list_.size());
>+  for (size_t i = 0; i < parsed_alr_state_events.size(); i++) {
>+    EXPECT_TRUE(test::VerifyLoggedAlrStateEvent(*alr_state_list_[i],
>+                                                parsed_alr_state_events[i]));
>   }
>-  // Create incoming and outgoing RTCP packets containing random data.
>-  for (size_t i = 0; i < incoming_rtcp_count; i++) {
>-    incoming_rtcp_packets.push_back(GenerateRtcpPacket(&prng));
>-    event_types.push_back(EventType::kIncomingRtcp);
>+
>+  const auto& parsed_audio_playout_map = parsed_log.audio_playout_events();
>+  ASSERT_EQ(parsed_audio_playout_map.size(), audio_playout_map_.size());
>+  for (const auto& kv : parsed_audio_playout_map) {
>+    uint32_t ssrc = kv.first;
>+    const auto& parsed_audio_playout_stream = kv.second;
>+    const auto& audio_playout_stream = audio_playout_map_[ssrc];
>+    ASSERT_EQ(parsed_audio_playout_stream.size(), audio_playout_stream.size());
>+    for (size_t i = 0; i < parsed_audio_playout_map.size(); i++) {
>+      EXPECT_TRUE(test::VerifyLoggedAudioPlayoutEvent(
>+          *audio_playout_stream[i], parsed_audio_playout_stream[i]));
>+    }
>   }
>-  for (size_t i = 0; i < outgoing_rtcp_count; i++) {
>-    outgoing_rtcp_packets.push_back(GenerateRtcpPacket(&prng));
>-    event_types.push_back(EventType::kOutgoingRtcp);
>+
>+  auto& parsed_audio_network_adaptation_events =
>+      parsed_log.audio_network_adaptation_events();
>+  ASSERT_EQ(parsed_audio_network_adaptation_events.size(),
>+            ana_configs_list_.size());
>+  for (size_t i = 0; i < parsed_audio_network_adaptation_events.size(); i++) {
>+    EXPECT_TRUE(test::VerifyLoggedAudioNetworkAdaptationEvent(
>+        *ana_configs_list_[i], parsed_audio_network_adaptation_events[i]));
>   }
>-  // Create random SSRCs to use when logging AudioPlayout events.
>-  for (size_t i = 0; i < playout_count; i++) {
>-    playout_ssrcs.push_back(prng.Rand<uint32_t>());
>-    event_types.push_back(EventType::kAudioPlayout);
>+
>+  auto& parsed_bwe_delay_updates = parsed_log.bwe_delay_updates();
>+  ASSERT_EQ(parsed_bwe_delay_updates.size(), bwe_delay_list_.size());
>+  for (size_t i = 0; i < parsed_bwe_delay_updates.size(); i++) {
>+    EXPECT_TRUE(test::VerifyLoggedBweDelayBasedUpdate(
>+        *bwe_delay_list_[i], parsed_bwe_delay_updates[i]));
>   }
>-  // Create random bitrate updates for LossBasedBwe.
>-  for (size_t i = 0; i < bwe_loss_count; i++) {
>-    bwe_loss_updates.push_back(GenerateBweLossEvent(&prng));
>-    event_types.push_back(EventType::kBweLossUpdate);
>+
>+  auto& parsed_bwe_loss_updates = parsed_log.bwe_loss_updates();
>+  ASSERT_EQ(parsed_bwe_loss_updates.size(), bwe_loss_list_.size());
>+  for (size_t i = 0; i < parsed_bwe_loss_updates.size(); i++) {
>+    EXPECT_TRUE(test::VerifyLoggedBweLossBasedUpdate(
>+        *bwe_loss_list_[i], parsed_bwe_loss_updates[i]));
>   }
>-  // Create random bitrate updates for DelayBasedBwe.
>-  for (size_t i = 0; i < bwe_delay_count; i++) {
>-    bwe_delay_updates.push_back(std::make_pair(
>-        prng.Rand(6000, 10000000), prng.Rand<bool>()
>-                                       ? BandwidthUsage::kBwOverusing
>-                                       : BandwidthUsage::kBwUnderusing));
>-    event_types.push_back(EventType::kBweDelayUpdate);
>+
>+  auto& parsed_bwe_probe_cluster_created_events =
>+      parsed_log.bwe_probe_cluster_created_events();
>+  ASSERT_EQ(parsed_bwe_probe_cluster_created_events.size(),
>+            probe_creation_list_.size());
>+  for (size_t i = 0; i < parsed_bwe_probe_cluster_created_events.size(); i++) {
>+    EXPECT_TRUE(test::VerifyLoggedBweProbeClusterCreatedEvent(
>+        *probe_creation_list_[i], parsed_bwe_probe_cluster_created_events[i]));
>   }
> 
>-  // Order the events randomly. The configurations are stored in a separate
>-  // buffer, so they might be written before any othe events. Hence, we can't
>-  // mix the config events with other events.
>-  for (size_t i = config_count; i < event_types.size(); i++) {
>-    size_t other = prng.Rand(static_cast<uint32_t>(i),
>-                             static_cast<uint32_t>(event_types.size() - 1));
>-    RTC_CHECK(i <= other && other < event_types.size());
>-    std::swap(event_types[i], event_types[other]);
>+  auto& parsed_bwe_probe_failure_events = parsed_log.bwe_probe_failure_events();
>+  ASSERT_EQ(parsed_bwe_probe_failure_events.size(), probe_failure_list_.size());
>+  for (size_t i = 0; i < parsed_bwe_probe_failure_events.size(); i++) {
>+    EXPECT_TRUE(test::VerifyLoggedBweProbeFailureEvent(
>+        *probe_failure_list_[i], parsed_bwe_probe_failure_events[i]));
>   }
>-}
> 
>-void RtcEventLogSession::WriteSession() {
>-  // Find the name of the current test, in order to use it as a temporary
>-  // filename.
>-  auto test_info = ::testing::UnitTest::GetInstance()->current_test_info();
>-  std::string test_name = test_info->name();
>-  std::replace(test_name.begin(), test_name.end(), '/', '_');
>-  const std::string temp_filename =
>-      test::OutputPath() + "RtcEventLogTest_" + test_name;
>+  auto& parsed_bwe_probe_success_events = parsed_log.bwe_probe_success_events();
>+  ASSERT_EQ(parsed_bwe_probe_success_events.size(), probe_success_list_.size());
>+  for (size_t i = 0; i < parsed_bwe_probe_success_events.size(); i++) {
>+    EXPECT_TRUE(test::VerifyLoggedBweProbeSuccessEvent(
>+        *probe_success_list_[i], parsed_bwe_probe_success_events[i]));
>+  }
> 
>-  rtc::ScopedFakeClock fake_clock;
>-  fake_clock.SetTimeMicros(prng.Rand<uint32_t>());
>+  auto& parsed_ice_candidate_pair_configs =
>+      parsed_log.ice_candidate_pair_configs();
>+  ASSERT_EQ(parsed_ice_candidate_pair_configs.size(), ice_config_list_.size());
>+  for (size_t i = 0; i < parsed_ice_candidate_pair_configs.size(); i++) {
>+    EXPECT_TRUE(test::VerifyLoggedIceCandidatePairConfig(
>+        *ice_config_list_[i], parsed_ice_candidate_pair_configs[i]));
>+  }
> 
>-  // When log_dumper goes out of scope, it causes the log file to be flushed
>-  // to disk.
>-  std::unique_ptr<RtcEventLog> log_dumper(
>-      RtcEventLog::Create(RtcEventLog::EncodingType::Legacy));
>+  auto& parsed_ice_candidate_pair_events =
>+      parsed_log.ice_candidate_pair_events();
>+  ASSERT_EQ(parsed_ice_candidate_pair_events.size(),
>+            parsed_ice_candidate_pair_events.size());
>+  for (size_t i = 0; i < parsed_ice_candidate_pair_events.size(); i++) {
>+    EXPECT_TRUE(test::VerifyLoggedIceCandidatePairEvent(
>+        *ice_event_list_[i], parsed_ice_candidate_pair_events[i]));
>+  }
> 
>-  size_t incoming_rtp_written = 0;
>-  size_t outgoing_rtp_written = 0;
>-  size_t incoming_rtcp_written = 0;
>-  size_t outgoing_rtcp_written = 0;
>-  size_t playouts_written = 0;
>-  size_t bwe_loss_written = 0;
>-  size_t bwe_delay_written = 0;
>-  size_t recv_configs_written = 0;
>-  size_t send_configs_written = 0;
>-
>-  for (size_t i = 0; i < event_types.size(); i++) {
>-    fake_clock.AdvanceTimeMicros(prng.Rand(1, 1000));
>-    if (i == event_types.size() / 2)
>-      log_dumper->StartLogging(
>-          rtc::MakeUnique<RtcEventLogOutputFile>(temp_filename, 10000000),
>-          output_period_ms);
>-    switch (event_types[i]) {
>-      case EventType::kIncomingRtp:
>-        RTC_CHECK(incoming_rtp_written < incoming_rtp_packets.size());
>-        log_dumper->Log(rtc::MakeUnique<RtcEventRtpPacketIncoming>(
>-            incoming_rtp_packets[incoming_rtp_written++]));
>-        break;
>-      case EventType::kOutgoingRtp: {
>-        RTC_CHECK(outgoing_rtp_written < outgoing_rtp_packets.size());
>-        constexpr int kNotAProbe = PacedPacketInfo::kNotAProbe;  // Compiler...
>-        log_dumper->Log(rtc::MakeUnique<RtcEventRtpPacketOutgoing>(
>-            outgoing_rtp_packets[outgoing_rtp_written++], kNotAProbe));
>-        break;
>-      }
>-      case EventType::kIncomingRtcp:
>-        RTC_CHECK(incoming_rtcp_written < incoming_rtcp_packets.size());
>-        log_dumper->Log(rtc::MakeUnique<RtcEventRtcpPacketIncoming>(
>-            incoming_rtcp_packets[incoming_rtcp_written++]));
>-        break;
>-      case EventType::kOutgoingRtcp:
>-        RTC_CHECK(outgoing_rtcp_written < outgoing_rtcp_packets.size());
>-        log_dumper->Log(rtc::MakeUnique<RtcEventRtcpPacketOutgoing>(
>-            outgoing_rtcp_packets[outgoing_rtcp_written++]));
>-        break;
>-      case EventType::kAudioPlayout:
>-        RTC_CHECK(playouts_written < playout_ssrcs.size());
>-        log_dumper->Log(rtc::MakeUnique<RtcEventAudioPlayout>(
>-            playout_ssrcs[playouts_written++]));
>-        break;
>-      case EventType::kBweLossUpdate:
>-        RTC_CHECK(bwe_loss_written < bwe_loss_updates.size());
>-        log_dumper->Log(rtc::MakeUnique<RtcEventBweUpdateLossBased>(
>-            bwe_loss_updates[bwe_loss_written].bitrate_bps,
>-            bwe_loss_updates[bwe_loss_written].fraction_loss,
>-            bwe_loss_updates[bwe_loss_written].total_packets));
>-        bwe_loss_written++;
>-        break;
>-      case EventType::kBweDelayUpdate:
>-        RTC_CHECK(bwe_delay_written < bwe_delay_updates.size());
>-        log_dumper->Log(rtc::MakeUnique<RtcEventBweUpdateDelayBased>(
>-            bwe_delay_updates[bwe_delay_written].first,
>-            bwe_delay_updates[bwe_delay_written].second));
>-        bwe_delay_written++;
>-        break;
>-      case EventType::kVideoRecvConfig:
>-        RTC_CHECK(recv_configs_written < receiver_configs.size());
>-        log_dumper->Log(rtc::MakeUnique<RtcEventVideoReceiveStreamConfig>(
>-            rtc::MakeUnique<rtclog::StreamConfig>(
>-                receiver_configs[recv_configs_written++])));
>-        break;
>-      case EventType::kVideoSendConfig:
>-        RTC_CHECK(send_configs_written < sender_configs.size());
>-        log_dumper->Log(rtc::MakeUnique<RtcEventVideoSendStreamConfig>(
>-            rtc::MakeUnique<rtclog::StreamConfig>(
>-                sender_configs[send_configs_written++])));
>-        break;
>-      case EventType::kAudioRecvConfig:
>-        // Not implemented
>-        RTC_NOTREACHED();
>-        break;
>-      case EventType::kAudioSendConfig:
>-        // Not implemented
>-        RTC_NOTREACHED();
>-        break;
>-      case EventType::kAudioNetworkAdaptation:
>-        // Not implemented
>-        RTC_NOTREACHED();
>-        break;
>-      case EventType::kBweProbeClusterCreated:
>-        // Not implemented
>-        RTC_NOTREACHED();
>-        break;
>-      case EventType::kBweProbeResult:
>-        // Not implemented
>-        RTC_NOTREACHED();
>-        break;
>+  auto& parsed_incoming_rtp_packets_by_ssrc =
>+      parsed_log.incoming_rtp_packets_by_ssrc();
>+  ASSERT_EQ(parsed_incoming_rtp_packets_by_ssrc.size(),
>+            incoming_rtp_map_.size());
>+  for (const auto& kv : parsed_incoming_rtp_packets_by_ssrc) {
>+    uint32_t ssrc = kv.ssrc;
>+    const auto& parsed_rtp_stream = kv.incoming_packets;
>+    const auto& rtp_stream = incoming_rtp_map_[ssrc];
>+    ASSERT_EQ(parsed_rtp_stream.size(), rtp_stream.size());
>+    for (size_t i = 0; i < parsed_rtp_stream.size(); i++) {
>+      EXPECT_TRUE(test::VerifyLoggedRtpPacketIncoming(*rtp_stream[i],
>+                                                      parsed_rtp_stream[i]));
>     }
>   }
> 
>-  log_dumper->StopLogging();
>-}
>-
>-// Read the file and verify that what we read back from the event log is the
>-// same as what we wrote down.
>-void RtcEventLogSession::ReadAndVerifySession() {
>-  // Find the name of the current test, in order to use it as a temporary
>-  // filename.
>-  auto test_info = ::testing::UnitTest::GetInstance()->current_test_info();
>-  std::string test_name = test_info->name();
>-  std::replace(test_name.begin(), test_name.end(), '/', '_');
>-  const std::string temp_filename =
>-      test::OutputPath() + "RtcEventLogTest_" + test_name;
>-
>-  // Read the generated file from disk.
>-  ParsedRtcEventLog parsed_log;
>-  ASSERT_TRUE(parsed_log.ParseFile(temp_filename));
>-  EXPECT_GE(5000u, event_types.size() + 2);  // The events must fit.
>-  EXPECT_EQ(event_types.size() + 2, parsed_log.GetNumberOfEvents());
>-
>-  size_t incoming_rtp_read = 0;
>-  size_t outgoing_rtp_read = 0;
>-  size_t incoming_rtcp_read = 0;
>-  size_t outgoing_rtcp_read = 0;
>-  size_t playouts_read = 0;
>-  size_t bwe_loss_read = 0;
>-  size_t bwe_delay_read = 0;
>-  size_t recv_configs_read = 0;
>-  size_t send_configs_read = 0;
>-
>-  RtcEventLogTestHelper::VerifyLogStartEvent(parsed_log, 0);
>-
>-  for (size_t i = 0; i < event_types.size(); i++) {
>-    switch (event_types[i]) {
>-      case EventType::kIncomingRtp:
>-        RTC_CHECK(incoming_rtp_read < incoming_rtp_packets.size());
>-        RtcEventLogTestHelper::VerifyIncomingRtpEvent(
>-            parsed_log, i + 1, incoming_rtp_packets[incoming_rtp_read++]);
>-        break;
>-      case EventType::kOutgoingRtp:
>-        RTC_CHECK(outgoing_rtp_read < outgoing_rtp_packets.size());
>-        RtcEventLogTestHelper::VerifyOutgoingRtpEvent(
>-            parsed_log, i + 1, outgoing_rtp_packets[outgoing_rtp_read++]);
>-        break;
>-      case EventType::kIncomingRtcp:
>-        RTC_CHECK(incoming_rtcp_read < incoming_rtcp_packets.size());
>-        RtcEventLogTestHelper::VerifyRtcpEvent(
>-            parsed_log, i + 1, kIncomingPacket,
>-            incoming_rtcp_packets[incoming_rtcp_read].data(),
>-            incoming_rtcp_packets[incoming_rtcp_read].size());
>-        incoming_rtcp_read++;
>-        break;
>-      case EventType::kOutgoingRtcp:
>-        RTC_CHECK(outgoing_rtcp_read < outgoing_rtcp_packets.size());
>-        RtcEventLogTestHelper::VerifyRtcpEvent(
>-            parsed_log, i + 1, kOutgoingPacket,
>-            outgoing_rtcp_packets[outgoing_rtcp_read].data(),
>-            outgoing_rtcp_packets[outgoing_rtcp_read].size());
>-        outgoing_rtcp_read++;
>-        break;
>-      case EventType::kAudioPlayout:
>-        RTC_CHECK(playouts_read < playout_ssrcs.size());
>-        RtcEventLogTestHelper::VerifyPlayoutEvent(
>-            parsed_log, i + 1, playout_ssrcs[playouts_read++]);
>-        break;
>-      case EventType::kBweLossUpdate:
>-        RTC_CHECK(bwe_loss_read < bwe_loss_updates.size());
>-        RtcEventLogTestHelper::VerifyBweLossEvent(
>-            parsed_log, i + 1, bwe_loss_updates[bwe_loss_read].bitrate_bps,
>-            bwe_loss_updates[bwe_loss_read].fraction_loss,
>-            bwe_loss_updates[bwe_loss_read].total_packets);
>-        bwe_loss_read++;
>-        break;
>-      case EventType::kBweDelayUpdate:
>-        RTC_CHECK(bwe_delay_read < bwe_delay_updates.size());
>-        RtcEventLogTestHelper::VerifyBweDelayEvent(
>-            parsed_log, i + 1, bwe_delay_updates[bwe_delay_read].first,
>-            bwe_delay_updates[bwe_delay_read].second);
>-        bwe_delay_read++;
>-        break;
>-      case EventType::kVideoRecvConfig:
>-        RTC_CHECK(recv_configs_read < receiver_configs.size());
>-        RtcEventLogTestHelper::VerifyVideoReceiveStreamConfig(
>-            parsed_log, i + 1, receiver_configs[recv_configs_read++]);
>-        break;
>-      case EventType::kVideoSendConfig:
>-        RTC_CHECK(send_configs_read < sender_configs.size());
>-        RtcEventLogTestHelper::VerifyVideoSendStreamConfig(
>-            parsed_log, i + 1, sender_configs[send_configs_read++]);
>-        break;
>-      case EventType::kAudioRecvConfig:
>-        // Not implemented
>-        RTC_NOTREACHED();
>-        break;
>-      case EventType::kAudioSendConfig:
>-        // Not implemented
>-        RTC_NOTREACHED();
>-        break;
>-      case EventType::kAudioNetworkAdaptation:
>-        // Not implemented
>-        RTC_NOTREACHED();
>-        break;
>-      case EventType::kBweProbeClusterCreated:
>-        // Not implemented
>-        RTC_NOTREACHED();
>-        break;
>-      case EventType::kBweProbeResult:
>-        // Not implemented
>-        RTC_NOTREACHED();
>-        break;
>+  auto& parsed_outgoing_rtp_packets_by_ssrc =
>+      parsed_log.outgoing_rtp_packets_by_ssrc();
>+  ASSERT_EQ(parsed_outgoing_rtp_packets_by_ssrc.size(),
>+            outgoing_rtp_map_.size());
>+  for (const auto& kv : parsed_outgoing_rtp_packets_by_ssrc) {
>+    uint32_t ssrc = kv.ssrc;
>+    const auto& parsed_rtp_stream = kv.outgoing_packets;
>+    const auto& rtp_stream = outgoing_rtp_map_[ssrc];
>+    ASSERT_EQ(parsed_rtp_stream.size(), rtp_stream.size());
>+    for (size_t i = 0; i < parsed_rtp_stream.size(); i++) {
>+      EXPECT_TRUE(test::VerifyLoggedRtpPacketOutgoing(*rtp_stream[i],
>+                                                      parsed_rtp_stream[i]));
>     }
>   }
> 
>-  RtcEventLogTestHelper::VerifyLogEndEvent(parsed_log,
>-                                           parsed_log.GetNumberOfEvents() - 1);
>-
>-  // Clean up temporary file - can be pretty slow.
>-  remove(temp_filename.c_str());
>-}
>-
>-void RtcEventLogSession::PrintExpectedEvents(std::ostream& stream) {
>-  for (size_t i = 0; i < event_types.size(); i++) {
>-    auto it = event_type_to_string.find(event_types[i]);
>-    RTC_CHECK(it != event_type_to_string.end());
>-    stream << it->second << " ";
>+  auto& parsed_incoming_rtcp_packets = parsed_log.incoming_rtcp_packets();
>+  ASSERT_EQ(parsed_incoming_rtcp_packets.size(), incoming_rtcp_list_.size());
>+  for (size_t i = 0; i < parsed_incoming_rtcp_packets.size(); i++) {
>+    EXPECT_TRUE(test::VerifyLoggedRtcpPacketIncoming(
>+        *incoming_rtcp_list_[i], parsed_incoming_rtcp_packets[i]));
>   }
>-  stream << std::endl;
>-}
> 
>-void PrintActualEvents(const ParsedRtcEventLog& parsed_log,
>-                       std::ostream& stream) {
>-  for (size_t i = 0; i < parsed_log.GetNumberOfEvents(); i++) {
>-    auto it = parsed_event_type_to_string.find(parsed_log.GetEventType(i));
>-    RTC_CHECK(it != parsed_event_type_to_string.end());
>-    stream << it->second << " ";
>+  auto& parsed_outgoing_rtcp_packets = parsed_log.outgoing_rtcp_packets();
>+  ASSERT_EQ(parsed_outgoing_rtcp_packets.size(), outgoing_rtcp_list_.size());
>+  for (size_t i = 0; i < parsed_outgoing_rtcp_packets.size(); i++) {
>+    EXPECT_TRUE(test::VerifyLoggedRtcpPacketOutgoing(
>+        *outgoing_rtcp_list_[i], parsed_outgoing_rtcp_packets[i]));
>+  }
>+  auto& parsed_audio_recv_configs = parsed_log.audio_recv_configs();
>+  ASSERT_EQ(parsed_audio_recv_configs.size(), audio_recv_config_list_.size());
>+  for (size_t i = 0; i < parsed_audio_recv_configs.size(); i++) {
>+    EXPECT_TRUE(test::VerifyLoggedAudioRecvConfig(
>+        *audio_recv_config_list_[i], parsed_audio_recv_configs[i]));
>+  }
>+  auto& parsed_audio_send_configs = parsed_log.audio_send_configs();
>+  ASSERT_EQ(parsed_audio_send_configs.size(), audio_send_config_list_.size());
>+  for (size_t i = 0; i < parsed_audio_send_configs.size(); i++) {
>+    EXPECT_TRUE(test::VerifyLoggedAudioSendConfig(
>+        *audio_send_config_list_[i], parsed_audio_send_configs[i]));
>+  }
>+  auto& parsed_video_recv_configs = parsed_log.video_recv_configs();
>+  ASSERT_EQ(parsed_video_recv_configs.size(), video_recv_config_list_.size());
>+  for (size_t i = 0; i < parsed_video_recv_configs.size(); i++) {
>+    EXPECT_TRUE(test::VerifyLoggedVideoRecvConfig(
>+        *video_recv_config_list_[i], parsed_video_recv_configs[i]));
>+  }
>+  auto& parsed_video_send_configs = parsed_log.video_send_configs();
>+  ASSERT_EQ(parsed_video_send_configs.size(), video_send_config_list_.size());
>+  for (size_t i = 0; i < parsed_video_send_configs.size(); i++) {
>+    EXPECT_TRUE(test::VerifyLoggedVideoSendConfig(
>+        *video_send_config_list_[i], parsed_video_send_configs[i]));
>   }
>-  stream << std::endl;
>-}
> 
>-TEST_P(RtcEventLogSession, LogSessionAndReadBack) {
>-  RtpHeaderExtensionMap extensions;
>-  GenerateSessionDescription(3,           // Number of incoming RTP packets.
>-                             2,           // Number of outgoing RTP packets.
>-                             1,           // Number of incoming RTCP packets.
>-                             1,           // Number of outgoing RTCP packets.
>-                             0,           // Number of playout events.
>-                             0,           // Number of BWE loss events.
>-                             0,           // Number of BWE delay events.
>-                             extensions,  // No extensions.
>-                             0);          // Number of contributing sources.
>-  WriteSession();
>-  ReadAndVerifySession();
>+  // Clean up temporary file - can be pretty slow.
>+  remove(temp_filename_.c_str());
> }
> 
>-TEST_P(RtcEventLogSession, LogSessionAndReadBackWith2Extensions) {
>-  RtpHeaderExtensionMap extensions;
>-  extensions.Register(kRtpExtensionAbsoluteSendTime,
>-                      kAbsoluteSendTimeExtensionId);
>-  extensions.Register(kRtpExtensionTransportSequenceNumber,
>-                      kTransportSequenceNumberExtensionId);
>-  GenerateSessionDescription(4, 4, 1, 1, 0, 0, 0, extensions, 0);
>-  WriteSession();
>-  ReadAndVerifySession();
>-}
>+}  // namespace
> 
>-TEST_P(RtcEventLogSession, LogSessionAndReadBackWithAllExtensions) {
>-  RtpHeaderExtensionMap extensions;
>-  for (uint32_t i = 0; i < kNumExtensions; i++) {
>-    extensions.Register(kExtensionTypes[i], kExtensionIds[i]);
>-  }
>-  GenerateSessionDescription(5, 4, 1, 1, 3, 2, 2, extensions, 2);
>-  WriteSession();
>-  ReadAndVerifySession();
>+TEST_P(RtcEventLogSession, StartLoggingFromBeginning) {
>+  EventCounts count;
>+  count.audio_send_streams = 2;
>+  count.audio_recv_streams = 2;
>+  count.video_send_streams = 3;
>+  count.video_recv_streams = 4;
>+  count.alr_states = 4;
>+  count.audio_playouts = 100;
>+  count.ana_configs = 3;
>+  count.bwe_loss_events = 20;
>+  count.bwe_delay_events = 20;
>+  count.probe_creations = 4;
>+  count.probe_successes = 2;
>+  count.probe_failures = 2;
>+  count.ice_configs = 3;
>+  count.ice_events = 10;
>+  count.incoming_rtp_packets = 100;
>+  count.outgoing_rtp_packets = 100;
>+  count.incoming_rtcp_packets = 20;
>+  count.outgoing_rtcp_packets = 20;
>+
>+  WriteLog(count, 0);
>+  ReadAndVerifyLog();
> }
> 
>-TEST_P(RtcEventLogSession, LogLongSessionAndReadBack) {
>-  RtpHeaderExtensionMap extensions;
>-  for (uint32_t i = 0; i < kNumExtensions; i++) {
>-    extensions.Register(kExtensionTypes[i], kExtensionIds[i]);
>-  }
>-  GenerateSessionDescription(1000, 1000, 250, 250, 200, 100, 100, extensions,
>-                             1);
>-  WriteSession();
>-  ReadAndVerifySession();
>+TEST_P(RtcEventLogSession, StartLoggingInTheMiddle) {
>+  EventCounts count;
>+  count.audio_send_streams = 3;
>+  count.audio_recv_streams = 4;
>+  count.video_send_streams = 5;
>+  count.video_recv_streams = 6;
>+  count.alr_states = 10;
>+  count.audio_playouts = 500;
>+  count.ana_configs = 10;
>+  count.bwe_loss_events = 50;
>+  count.bwe_delay_events = 50;
>+  count.probe_creations = 10;
>+  count.probe_successes = 5;
>+  count.probe_failures = 5;
>+  count.ice_configs = 10;
>+  count.ice_events = 20;
>+  count.incoming_rtp_packets = 500;
>+  count.outgoing_rtp_packets = 500;
>+  count.incoming_rtcp_packets = 50;
>+  count.outgoing_rtcp_packets = 50;
>+
>+  WriteLog(count, 500);
>+  ReadAndVerifyLog();
> }
> 
> TEST(RtcEventLogTest, CircularBufferKeepsMostRecentEvents) {
>+  // TODO(terelius): Maybe make a separate RtcEventLogImplTest that can access
>+  // the size of the cyclic buffer?
>   constexpr size_t kNumEvents = 20000;
>   constexpr int64_t kStartTime = 1000000;
>+  constexpr int32_t kStartBitrate = 1000000;
> 
>   auto test_info = ::testing::UnitTest::GetInstance()->current_test_info();
>-  std::string test_name = test_info->name();
>+  std::string test_name =
>+      std::string(test_info->test_case_name()) + "_" + test_info->name();
>   std::replace(test_name.begin(), test_name.end(), '/', '_');
>-  const std::string temp_filename =
>-      test::OutputPath() + "RtcEventLogTest_" + test_name;
>+  const std::string temp_filename = test::OutputPath() + test_name;
> 
>-  rtc::ScopedFakeClock fake_clock;
>-  fake_clock.SetTimeMicros(kStartTime);
>+  std::unique_ptr<rtc::ScopedFakeClock> fake_clock =
>+      absl::make_unique<rtc::ScopedFakeClock>();
>+  fake_clock->SetTimeMicros(kStartTime);
> 
>   // When log_dumper goes out of scope, it causes the log file to be flushed
>   // to disk.
>@@ -722,50 +705,62 @@ TEST(RtcEventLogTest, CircularBufferKeepsMostRecentEvents) {
>   for (size_t i = 0; i < kNumEvents; i++) {
>     // The purpose of the test is to verify that the log can handle
>     // more events than what fits in the internal circular buffer. The exact
>-    // type of events does not matter so we chose AudioPlayouts for simplicity.
>-    // We use the index as an ssrc to get a strict relationship between the ssrc
>-    // and the timestamp. We use this for some basic consistency checks when we
>-    // read back.
>-    log_dumper->Log(rtc::MakeUnique<RtcEventAudioPlayout>(i));
>-    fake_clock.AdvanceTimeMicros(10000);
>+    // type of events does not matter so we chose ProbeSuccess events for
>+    // simplicity.
>+    // We base the various values on the index. We use this for some basic
>+    // consistency checks when we read back.
>+    log_dumper->Log(absl::make_unique<RtcEventProbeResultSuccess>(
>+        i, kStartBitrate + i * 1000));
>+    fake_clock->AdvanceTimeMicros(10000);
>   }
>+  int64_t start_time_us = rtc::TimeMicros();
>   log_dumper->StartLogging(
>-      rtc::MakeUnique<RtcEventLogOutputFile>(temp_filename, 10000000),
>+      absl::make_unique<RtcEventLogOutputFile>(temp_filename, 10000000),
>       RtcEventLog::kImmediateOutput);
>+  fake_clock->AdvanceTimeMicros(10000);
>+  int64_t stop_time_us = rtc::TimeMicros();
>   log_dumper->StopLogging();
> 
>   // Read the generated file from disk.
>-  ParsedRtcEventLog parsed_log;
>+  ParsedRtcEventLogNew parsed_log;
>   ASSERT_TRUE(parsed_log.ParseFile(temp_filename));
>+
>+  const auto& start_log_events = parsed_log.start_log_events();
>+  ASSERT_EQ(start_log_events.size(), 1u);
>+  EXPECT_TRUE(test::VerifyLoggedStartEvent(start_time_us, start_log_events[0]));
>+
>+  const auto& stop_log_events = parsed_log.stop_log_events();
>+  ASSERT_EQ(stop_log_events.size(), 1u);
>+  EXPECT_TRUE(test::VerifyLoggedStopEvent(stop_time_us, stop_log_events[0]));
>+
>+  const auto& probe_success_events = parsed_log.bwe_probe_success_events();
>   // If the following fails, it probably means that kNumEvents isn't larger
>   // than the size of the cyclic buffer in the event log. Try increasing
>   // kNumEvents.
>-  EXPECT_LT(parsed_log.GetNumberOfEvents(), kNumEvents);
>-  // We expect a start event, some number of playouts events and a stop event.
>-  EXPECT_GT(parsed_log.GetNumberOfEvents(), 2u);
>-
>-  RtcEventLogTestHelper::VerifyLogStartEvent(parsed_log, 0);
>-  rtc::Optional<int64_t> last_timestamp;
>-  rtc::Optional<uint32_t> last_ssrc;
>-  for (size_t i = 1; i < parsed_log.GetNumberOfEvents() - 1; i++) {
>-    EXPECT_EQ(parsed_log.GetEventType(i),
>-              ParsedRtcEventLog::EventType::AUDIO_PLAYOUT_EVENT);
>-    uint32_t ssrc;
>-    parsed_log.GetAudioPlayout(i, &ssrc);
>-    int64_t timestamp = parsed_log.GetTimestamp(i);
>-    EXPECT_LT(ssrc, kNumEvents);
>-    EXPECT_EQ(static_cast<int64_t>(kStartTime + 10000 * ssrc), timestamp);
>-    if (last_ssrc)
>-      EXPECT_EQ(ssrc, *last_ssrc + 1);
>-    if (last_timestamp)
>-      EXPECT_EQ(timestamp, *last_timestamp + 10000);
>-    last_ssrc = ssrc;
>-    last_timestamp = timestamp;
>+  EXPECT_LT(probe_success_events.size(), kNumEvents);
>+
>+  ASSERT_GT(probe_success_events.size(), 1u);
>+  int64_t first_timestamp_us = probe_success_events[0].timestamp_us;
>+  uint32_t first_id = probe_success_events[0].id;
>+  int32_t first_bitrate_bps = probe_success_events[0].bitrate_bps;
>+  // We want to reset the time to what we used when generating the events, but
>+  // the fake clock implementation DCHECKS if time moves backwards. We therefore
>+  // recreate the clock. However we must ensure that the old fake_clock is
>+  // destroyed before the new one is created, so we have to reset() first.
>+  fake_clock.reset();
>+  fake_clock = absl::make_unique<rtc::ScopedFakeClock>();
>+  fake_clock->SetTimeMicros(first_timestamp_us);
>+  for (size_t i = 1; i < probe_success_events.size(); i++) {
>+    fake_clock->AdvanceTimeMicros(10000);
>+    ASSERT_TRUE(test::VerifyLoggedBweProbeSuccessEvent(
>+        RtcEventProbeResultSuccess(first_id + i, first_bitrate_bps + i * 1000),
>+        probe_success_events[i]));
>   }
>-  RtcEventLogTestHelper::VerifyLogEndEvent(parsed_log,
>-                                           parsed_log.GetNumberOfEvents() - 1);
> }
> 
>+// TODO(terelius): Verify parser behavior if the timestamps are not
>+// monotonically increasing in the log.
>+
> INSTANTIATE_TEST_CASE_P(
>     RtcEventLogTest,
>     RtcEventLogSession,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.cc
>index 23e15b5033b..273539c1aca 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.cc
>@@ -10,610 +10,809 @@
> 
> #include "logging/rtc_event_log/rtc_event_log_unittest_helper.h"
> 
>-#include <string.h>
>+#include <string.h>  // memcmp
> 
>-#include <string>
>+#include <limits>
>+#include <memory>
>+#include <numeric>
>+#include <utility>
> #include <vector>
> 
> #include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
> #include "modules/remote_bitrate_estimator/include/bwe_defines.h"
>+#include "modules/rtp_rtcp/include/rtp_cvo.h"
>+#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
>+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
>+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
> #include "rtc_base/checks.h"
>-#include "test/gmock.h"
>-#include "test/gtest.h"
>-#include "test/testsupport/fileutils.h"
>-
>-// Files generated at build-time by the protobuf compiler.
>-#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
>-#include "external/webrtc/webrtc/logging/rtc_event_log/rtc_event_log.pb.h"
>-#else
>-#include "logging/rtc_event_log/rtc_event_log.pb.h"
>-#endif
> 
> namespace webrtc {
> 
>+namespace test {
>+
> namespace {
> 
>-BandwidthUsage GetRuntimeDetectorState(
>-    rtclog::DelayBasedBweUpdate::DetectorState detector_state) {
>-  switch (detector_state) {
>-    case rtclog::DelayBasedBweUpdate::BWE_NORMAL:
>-      return BandwidthUsage::kBwNormal;
>-    case rtclog::DelayBasedBweUpdate::BWE_UNDERUSING:
>-      return BandwidthUsage::kBwUnderusing;
>-    case rtclog::DelayBasedBweUpdate::BWE_OVERUSING:
>-      return BandwidthUsage::kBwOverusing;
>-  }
>-  RTC_NOTREACHED();
>-  return BandwidthUsage::kBwNormal;
>-}
>-
>-rtclog::BweProbeResult::ResultType GetProbeResultType(
>-    ProbeFailureReason failure_reason) {
>-  switch (failure_reason) {
>-    case ProbeFailureReason::kInvalidSendReceiveInterval:
>-      return rtclog::BweProbeResult::INVALID_SEND_RECEIVE_INTERVAL;
>-    case ProbeFailureReason::kInvalidSendReceiveRatio:
>-      return rtclog::BweProbeResult::INVALID_SEND_RECEIVE_RATIO;
>-    case ProbeFailureReason::kTimeout:
>-      return rtclog::BweProbeResult::TIMEOUT;
>-    case ProbeFailureReason::kLast:
>-      RTC_NOTREACHED();
>+struct ExtensionPair {
>+  RTPExtensionType type;
>+  const char* name;
>+};
>+
>+constexpr int kMaxCsrcs = 3;
>+
>+// Maximum serialized size of a header extension, including 1 byte ID.
>+constexpr int kMaxExtensionSizeBytes = 4;
>+constexpr int kMaxNumExtensions = 5;
>+
>+constexpr ExtensionPair kExtensions[kMaxNumExtensions] = {
>+    {RTPExtensionType::kRtpExtensionTransmissionTimeOffset,
>+     RtpExtension::kTimestampOffsetUri},
>+    {RTPExtensionType::kRtpExtensionAbsoluteSendTime,
>+     RtpExtension::kAbsSendTimeUri},
>+    {RTPExtensionType::kRtpExtensionTransportSequenceNumber,
>+     RtpExtension::kTransportSequenceNumberUri},
>+    {RTPExtensionType::kRtpExtensionAudioLevel, RtpExtension::kAudioLevelUri},
>+    {RTPExtensionType::kRtpExtensionVideoRotation,
>+     RtpExtension::kVideoRotationUri}};
>+
>+template <typename T>
>+void ShuffleInPlace(Random* prng, rtc::ArrayView<T> array) {
>+  RTC_DCHECK_LE(array.size(), std::numeric_limits<uint32_t>::max());
>+  for (uint32_t i = 0; i + 1 < array.size(); i++) {
>+    uint32_t other = prng->Rand(i, static_cast<uint32_t>(array.size() - 1));
>+    std::swap(array[i], array[other]);
>   }
>-  RTC_NOTREACHED();
>-  return rtclog::BweProbeResult::SUCCESS;
> }
> }  // namespace
> 
>-// Checks that the event has a timestamp, a type and exactly the data field
>-// corresponding to the type.
>-::testing::AssertionResult IsValidBasicEvent(const rtclog::Event& event) {
>-  if (!event.has_timestamp_us()) {
>-    return ::testing::AssertionFailure() << "Event has no timestamp";
>-  }
>-  if (!event.has_type()) {
>-    return ::testing::AssertionFailure() << "Event has no event type";
>+std::unique_ptr<RtcEventAlrState> EventGenerator::NewAlrState() {
>+  return absl::make_unique<RtcEventAlrState>(prng_.Rand<bool>());
>+}
>+
>+std::unique_ptr<RtcEventAudioPlayout> EventGenerator::NewAudioPlayout(
>+    uint32_t ssrc) {
>+  return absl::make_unique<RtcEventAudioPlayout>(ssrc);
>+}
>+
>+std::unique_ptr<RtcEventAudioNetworkAdaptation>
>+EventGenerator::NewAudioNetworkAdaptation() {
>+  std::unique_ptr<AudioEncoderRuntimeConfig> config =
>+      absl::make_unique<AudioEncoderRuntimeConfig>();
>+
>+  config->bitrate_bps = prng_.Rand(0, 3000000);
>+  config->enable_fec = prng_.Rand<bool>();
>+  config->enable_dtx = prng_.Rand<bool>();
>+  config->frame_length_ms = prng_.Rand(10, 120);
>+  config->num_channels = prng_.Rand(1, 2);
>+  config->uplink_packet_loss_fraction = prng_.Rand<float>();
>+
>+  return absl::make_unique<RtcEventAudioNetworkAdaptation>(std::move(config));
>+}
>+
>+std::unique_ptr<RtcEventBweUpdateDelayBased>
>+EventGenerator::NewBweUpdateDelayBased() {
>+  constexpr int32_t kMaxBweBps = 20000000;
>+  int32_t bitrate_bps = prng_.Rand(0, kMaxBweBps);
>+  BandwidthUsage state = static_cast<BandwidthUsage>(
>+      prng_.Rand(static_cast<uint32_t>(BandwidthUsage::kLast) - 1));
>+  return absl::make_unique<RtcEventBweUpdateDelayBased>(bitrate_bps, state);
>+}
>+
>+std::unique_ptr<RtcEventBweUpdateLossBased>
>+EventGenerator::NewBweUpdateLossBased() {
>+  constexpr int32_t kMaxBweBps = 20000000;
>+  constexpr int32_t kMaxPackets = 1000;
>+  int32_t bitrate_bps = prng_.Rand(0, kMaxBweBps);
>+  uint8_t fraction_lost = prng_.Rand<uint8_t>();
>+  int32_t total_packets = prng_.Rand(1, kMaxPackets);
>+
>+  return absl::make_unique<RtcEventBweUpdateLossBased>(
>+      bitrate_bps, fraction_lost, total_packets);
>+}
>+
>+std::unique_ptr<RtcEventProbeClusterCreated>
>+EventGenerator::NewProbeClusterCreated() {
>+  constexpr int kMaxBweBps = 20000000;
>+  constexpr int kMaxNumProbes = 10000;
>+  int id = prng_.Rand(1, kMaxNumProbes);
>+  int bitrate_bps = prng_.Rand(0, kMaxBweBps);
>+  int min_probes = prng_.Rand(5, 50);
>+  int min_bytes = prng_.Rand(500, 50000);
>+
>+  return absl::make_unique<RtcEventProbeClusterCreated>(id, bitrate_bps,
>+                                                        min_probes, min_bytes);
>+}
>+
>+std::unique_ptr<RtcEventProbeResultFailure>
>+EventGenerator::NewProbeResultFailure() {
>+  constexpr int kMaxNumProbes = 10000;
>+  int id = prng_.Rand(1, kMaxNumProbes);
>+  ProbeFailureReason reason = static_cast<ProbeFailureReason>(
>+      prng_.Rand(static_cast<uint32_t>(ProbeFailureReason::kLast) - 1));
>+
>+  return absl::make_unique<RtcEventProbeResultFailure>(id, reason);
>+}
>+
>+std::unique_ptr<RtcEventProbeResultSuccess>
>+EventGenerator::NewProbeResultSuccess() {
>+  constexpr int kMaxBweBps = 20000000;
>+  constexpr int kMaxNumProbes = 10000;
>+  int id = prng_.Rand(1, kMaxNumProbes);
>+  int bitrate_bps = prng_.Rand(0, kMaxBweBps);
>+
>+  return absl::make_unique<RtcEventProbeResultSuccess>(id, bitrate_bps);
>+}
>+
>+std::unique_ptr<RtcEventIceCandidatePairConfig>
>+EventGenerator::NewIceCandidatePairConfig() {
>+  IceCandidateType local_candidate_type = static_cast<IceCandidateType>(
>+      prng_.Rand(static_cast<uint32_t>(IceCandidateType::kNumValues) - 1));
>+  IceCandidateNetworkType local_network_type =
>+      static_cast<IceCandidateNetworkType>(prng_.Rand(
>+          static_cast<uint32_t>(IceCandidateNetworkType::kNumValues) - 1));
>+  IceCandidatePairAddressFamily local_address_family =
>+      static_cast<IceCandidatePairAddressFamily>(prng_.Rand(
>+          static_cast<uint32_t>(IceCandidatePairAddressFamily::kNumValues) -
>+          1));
>+  IceCandidateType remote_candidate_type = static_cast<IceCandidateType>(
>+      prng_.Rand(static_cast<uint32_t>(IceCandidateType::kNumValues) - 1));
>+  IceCandidatePairAddressFamily remote_address_family =
>+      static_cast<IceCandidatePairAddressFamily>(prng_.Rand(
>+          static_cast<uint32_t>(IceCandidatePairAddressFamily::kNumValues) -
>+          1));
>+  IceCandidatePairProtocol protocol_type =
>+      static_cast<IceCandidatePairProtocol>(prng_.Rand(
>+          static_cast<uint32_t>(IceCandidatePairProtocol::kNumValues) - 1));
>+
>+  IceCandidatePairDescription desc;
>+  desc.local_candidate_type = local_candidate_type;
>+  desc.local_relay_protocol = protocol_type;
>+  desc.local_network_type = local_network_type;
>+  desc.local_address_family = local_address_family;
>+  desc.remote_candidate_type = remote_candidate_type;
>+  desc.remote_address_family = remote_address_family;
>+  desc.candidate_pair_protocol = protocol_type;
>+
>+  IceCandidatePairConfigType type =
>+      static_cast<IceCandidatePairConfigType>(prng_.Rand(
>+          static_cast<uint32_t>(IceCandidatePairConfigType::kNumValues) - 1));
>+  uint32_t pair_id = prng_.Rand<uint32_t>();
>+  return absl::make_unique<RtcEventIceCandidatePairConfig>(type, pair_id, desc);
>+}
>+
>+std::unique_ptr<RtcEventIceCandidatePair>
>+EventGenerator::NewIceCandidatePair() {
>+  IceCandidatePairEventType type =
>+      static_cast<IceCandidatePairEventType>(prng_.Rand(
>+          static_cast<uint32_t>(IceCandidatePairEventType::kNumValues) - 1));
>+  uint32_t pair_id = prng_.Rand<uint32_t>();
>+
>+  return absl::make_unique<RtcEventIceCandidatePair>(type, pair_id);
>+}
>+
>+rtcp::ReportBlock EventGenerator::NewReportBlock() {
>+  rtcp::ReportBlock report_block;
>+  report_block.SetMediaSsrc(prng_.Rand<uint32_t>());
>+  report_block.SetFractionLost(prng_.Rand<uint8_t>());
>+  // cumulative_lost is a 3-byte signed value.
>+  RTC_DCHECK(report_block.SetCumulativeLost(
>+      prng_.Rand(-(1 << 23) + 1, (1 << 23) - 1)));
>+  report_block.SetExtHighestSeqNum(prng_.Rand<uint32_t>());
>+  report_block.SetJitter(prng_.Rand<uint32_t>());
>+  report_block.SetLastSr(prng_.Rand<uint32_t>());
>+  report_block.SetDelayLastSr(prng_.Rand<uint32_t>());
>+  return report_block;
>+}
>+
>+rtcp::SenderReport EventGenerator::NewSenderReport() {
>+  rtcp::SenderReport sender_report;
>+  sender_report.SetSenderSsrc(prng_.Rand<uint32_t>());
>+  sender_report.SetNtp(NtpTime(prng_.Rand<uint32_t>(), prng_.Rand<uint32_t>()));
>+  sender_report.SetPacketCount(prng_.Rand<uint32_t>());
>+  sender_report.AddReportBlock(NewReportBlock());
>+  return sender_report;
>+}
>+
>+rtcp::ReceiverReport EventGenerator::NewReceiverReport() {
>+  rtcp::ReceiverReport receiver_report;
>+  receiver_report.SetSenderSsrc(prng_.Rand<uint32_t>());
>+  receiver_report.AddReportBlock(NewReportBlock());
>+  return receiver_report;
>+}
>+
>+std::unique_ptr<RtcEventRtcpPacketIncoming>
>+EventGenerator::NewRtcpPacketIncoming() {
>+  // TODO(terelius): Test the other RTCP types too.
>+  switch (prng_.Rand(0, 1)) {
>+    case 0: {
>+      rtcp::SenderReport sender_report = NewSenderReport();
>+      rtc::Buffer buffer = sender_report.Build();
>+      return absl::make_unique<RtcEventRtcpPacketIncoming>(buffer);
>+    }
>+    case 1: {
>+      rtcp::ReceiverReport receiver_report = NewReceiverReport();
>+      rtc::Buffer buffer = receiver_report.Build();
>+      return absl::make_unique<RtcEventRtcpPacketIncoming>(buffer);
>+    }
>+    default:
>+      RTC_NOTREACHED();
>+      rtc::Buffer buffer;
>+      return absl::make_unique<RtcEventRtcpPacketIncoming>(buffer);
>   }
>-  rtclog::Event_EventType type = event.type();
>-  if ((type == rtclog::Event::RTP_EVENT) != event.has_rtp_packet()) {
>-    return ::testing::AssertionFailure()
>-           << "Event of type " << type << " has "
>-           << (event.has_rtp_packet() ? "" : "no ") << "RTP packet";
>+}
>+
>+std::unique_ptr<RtcEventRtcpPacketOutgoing>
>+EventGenerator::NewRtcpPacketOutgoing() {
>+  // TODO(terelius): Test the other RTCP types too.
>+  switch (prng_.Rand(0, 1)) {
>+    case 0: {
>+      rtcp::SenderReport sender_report = NewSenderReport();
>+      rtc::Buffer buffer = sender_report.Build();
>+      return absl::make_unique<RtcEventRtcpPacketOutgoing>(buffer);
>+    }
>+    case 1: {
>+      rtcp::ReceiverReport receiver_report = NewReceiverReport();
>+      rtc::Buffer buffer = receiver_report.Build();
>+      return absl::make_unique<RtcEventRtcpPacketOutgoing>(buffer);
>+    }
>+    default:
>+      RTC_NOTREACHED();
>+      rtc::Buffer buffer;
>+      return absl::make_unique<RtcEventRtcpPacketOutgoing>(buffer);
>   }
>-  if ((type == rtclog::Event::RTCP_EVENT) != event.has_rtcp_packet()) {
>-    return ::testing::AssertionFailure()
>-           << "Event of type " << type << " has "
>-           << (event.has_rtcp_packet() ? "" : "no ") << "RTCP packet";
>+}
>+
>+void EventGenerator::RandomizeRtpPacket(
>+    size_t packet_size,
>+    uint32_t ssrc,
>+    const RtpHeaderExtensionMap& extension_map,
>+    RtpPacket* rtp_packet) {
>+  constexpr int kMaxPayloadType = 127;
>+  rtp_packet->SetPayloadType(prng_.Rand(kMaxPayloadType));
>+  rtp_packet->SetMarker(prng_.Rand<bool>());
>+  rtp_packet->SetSequenceNumber(prng_.Rand<uint16_t>());
>+  rtp_packet->SetSsrc(ssrc);
>+  rtp_packet->SetTimestamp(prng_.Rand<uint32_t>());
>+
>+  uint32_t csrcs_count = prng_.Rand(0, kMaxCsrcs);
>+  std::vector<uint32_t> csrcs;
>+  for (size_t i = 0; i < csrcs_count; i++) {
>+    csrcs.push_back(prng_.Rand<uint32_t>());
>   }
>-  if ((type == rtclog::Event::LOSS_BASED_BWE_UPDATE) !=
>-      event.has_loss_based_bwe_update()) {
>-    return ::testing::AssertionFailure()
>-           << "Event of type " << type << " has "
>-           << (event.has_loss_based_bwe_update() ? "" : "no ") << "loss update";
>+  rtp_packet->SetCsrcs(csrcs);
>+
>+  if (extension_map.IsRegistered(TransmissionOffset::kId))
>+    rtp_packet->SetExtension<TransmissionOffset>(prng_.Rand(0x00ffffff));
>+  if (extension_map.IsRegistered(AudioLevel::kId))
>+    rtp_packet->SetExtension<AudioLevel>(prng_.Rand<bool>(), prng_.Rand(127));
>+  if (extension_map.IsRegistered(AbsoluteSendTime::kId))
>+    rtp_packet->SetExtension<AbsoluteSendTime>(prng_.Rand(0x00ffffff));
>+  if (extension_map.IsRegistered(VideoOrientation::kId))
>+    rtp_packet->SetExtension<VideoOrientation>(prng_.Rand(3));
>+  if (extension_map.IsRegistered(TransportSequenceNumber::kId))
>+    rtp_packet->SetExtension<TransportSequenceNumber>(prng_.Rand<uint16_t>());
>+
>+  RTC_DCHECK_GE(packet_size, rtp_packet->headers_size());
>+  size_t payload_size = packet_size - rtp_packet->headers_size();
>+  RTC_CHECK_LE(rtp_packet->headers_size() + payload_size, IP_PACKET_SIZE);
>+  uint8_t* payload = rtp_packet->AllocatePayload(payload_size);
>+  RTC_DCHECK(payload != nullptr);
>+  for (size_t i = 0; i < payload_size; i++) {
>+    payload[i] = prng_.Rand<uint8_t>();
>   }
>-  if ((type == rtclog::Event::DELAY_BASED_BWE_UPDATE) !=
>-      event.has_delay_based_bwe_update()) {
>-    return ::testing::AssertionFailure()
>-           << "Event of type " << type << " has "
>-           << (event.has_delay_based_bwe_update() ? "" : "no ")
>-           << "delay update";
>+}
>+
>+std::unique_ptr<RtcEventRtpPacketIncoming> EventGenerator::NewRtpPacketIncoming(
>+    uint32_t ssrc,
>+    const RtpHeaderExtensionMap& extension_map) {
>+  // 12 bytes RTP header, 4 bytes for 0xBEDE + alignment, 4 bytes per CSRC.
>+  constexpr size_t kMaxHeaderSize =
>+      16 + 4 * kMaxCsrcs + kMaxExtensionSizeBytes * kMaxNumExtensions;
>+  size_t packet_size =
>+      prng_.Rand(kMaxHeaderSize, static_cast<uint32_t>(IP_PACKET_SIZE - 1));
>+
>+  RtpPacketReceived rtp_packet(&extension_map);
>+  RandomizeRtpPacket(packet_size, ssrc, extension_map, &rtp_packet);
>+
>+  return absl::make_unique<RtcEventRtpPacketIncoming>(rtp_packet);
>+}
>+
>+std::unique_ptr<RtcEventRtpPacketOutgoing> EventGenerator::NewRtpPacketOutgoing(
>+    uint32_t ssrc,
>+    const RtpHeaderExtensionMap& extension_map) {
>+  // 12 bytes RTP header, 4 bytes for 0xBEDE + alignment, 4 bytes per CSRC.
>+  constexpr size_t kMaxHeaderSize =
>+      16 + 4 * kMaxCsrcs + kMaxExtensionSizeBytes * kMaxNumExtensions;
>+  size_t packet_size =
>+      prng_.Rand(kMaxHeaderSize, static_cast<uint32_t>(IP_PACKET_SIZE - 1));
>+
>+  RtpPacketToSend rtp_packet(&extension_map, packet_size);
>+  RandomizeRtpPacket(packet_size, ssrc, extension_map, &rtp_packet);
>+
>+  int probe_cluster_id = prng_.Rand(0, 100000);
>+  return absl::make_unique<RtcEventRtpPacketOutgoing>(rtp_packet,
>+                                                      probe_cluster_id);
>+}
>+
>+RtpHeaderExtensionMap EventGenerator::NewRtpHeaderExtensionMap() {
>+  RtpHeaderExtensionMap extension_map;
>+  std::vector<int> id(RtpExtension::kMaxId - RtpExtension::kMinId + 1);
>+  std::iota(id.begin(), id.end(), RtpExtension::kMinId);
>+  ShuffleInPlace(&prng_, rtc::ArrayView<int>(id));
>+
>+  if (prng_.Rand<bool>()) {
>+    extension_map.Register<AudioLevel>(id[0]);
>   }
>-  if ((type == rtclog::Event::AUDIO_PLAYOUT_EVENT) !=
>-      event.has_audio_playout_event()) {
>-    return ::testing::AssertionFailure()
>-           << "Event of type " << type << " has "
>-           << (event.has_audio_playout_event() ? "" : "no ")
>-           << "audio_playout event";
>+  if (prng_.Rand<bool>()) {
>+    extension_map.Register<TransmissionOffset>(id[1]);
>   }
>-  if ((type == rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT) !=
>-      event.has_video_receiver_config()) {
>-    return ::testing::AssertionFailure()
>-           << "Event of type " << type << " has "
>-           << (event.has_video_receiver_config() ? "" : "no ")
>-           << "receiver config";
>+  if (prng_.Rand<bool>()) {
>+    extension_map.Register<AbsoluteSendTime>(id[2]);
>   }
>-  if ((type == rtclog::Event::VIDEO_SENDER_CONFIG_EVENT) !=
>-      event.has_video_sender_config()) {
>-    return ::testing::AssertionFailure()
>-           << "Event of type " << type << " has "
>-           << (event.has_video_sender_config() ? "" : "no ") << "sender config";
>+  if (prng_.Rand<bool>()) {
>+    extension_map.Register<VideoOrientation>(id[3]);
>   }
>-  if ((type == rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT) !=
>-      event.has_audio_receiver_config()) {
>-    return ::testing::AssertionFailure()
>-           << "Event of type " << type << " has "
>-           << (event.has_audio_receiver_config() ? "" : "no ")
>-           << "audio receiver config";
>+  if (prng_.Rand<bool>()) {
>+    extension_map.Register<TransportSequenceNumber>(id[4]);
>   }
>-  if ((type == rtclog::Event::AUDIO_SENDER_CONFIG_EVENT) !=
>-      event.has_audio_sender_config()) {
>-    return ::testing::AssertionFailure()
>-           << "Event of type " << type << " has "
>-           << (event.has_audio_sender_config() ? "" : "no ")
>-           << "audio sender config";
>+
>+  return extension_map;
>+}
>+
>+std::unique_ptr<RtcEventAudioReceiveStreamConfig>
>+EventGenerator::NewAudioReceiveStreamConfig(
>+    uint32_t ssrc,
>+    const RtpHeaderExtensionMap& extensions) {
>+  auto config = absl::make_unique<rtclog::StreamConfig>();
>+  // Add SSRCs for the stream.
>+  config->remote_ssrc = ssrc;
>+  config->local_ssrc = prng_.Rand<uint32_t>();
>+  // Add header extensions.
>+  for (size_t i = 0; i < kMaxNumExtensions; i++) {
>+    uint8_t id = extensions.GetId(kExtensions[i].type);
>+    if (id != RtpHeaderExtensionMap::kInvalidId) {
>+      config->rtp_extensions.emplace_back(kExtensions[i].name, id);
>+    }
>   }
>-  if ((type == rtclog::Event::AUDIO_NETWORK_ADAPTATION_EVENT) !=
>-      event.has_audio_network_adaptation()) {
>-    return ::testing::AssertionFailure()
>-           << "Event of type " << type << " has "
>-           << (event.has_audio_network_adaptation() ? "" : "no ")
>-           << "audio network adaptation";
>+
>+  return absl::make_unique<RtcEventAudioReceiveStreamConfig>(std::move(config));
>+}
>+
>+std::unique_ptr<RtcEventAudioSendStreamConfig>
>+EventGenerator::NewAudioSendStreamConfig(
>+    uint32_t ssrc,
>+    const RtpHeaderExtensionMap& extensions) {
>+  auto config = absl::make_unique<rtclog::StreamConfig>();
>+  // Add SSRC to the stream.
>+  config->local_ssrc = ssrc;
>+  // Add header extensions.
>+  for (size_t i = 0; i < kMaxNumExtensions; i++) {
>+    uint8_t id = extensions.GetId(kExtensions[i].type);
>+    if (id != RtpHeaderExtensionMap::kInvalidId) {
>+      config->rtp_extensions.emplace_back(kExtensions[i].name, id);
>+    }
>   }
>-  if ((type == rtclog::Event::BWE_PROBE_CLUSTER_CREATED_EVENT) !=
>-      event.has_probe_cluster()) {
>-    return ::testing::AssertionFailure()
>-           << "Event of type " << type << " has "
>-           << (event.has_probe_cluster() ? "" : "no ") << "bwe probe cluster";
>+  return absl::make_unique<RtcEventAudioSendStreamConfig>(std::move(config));
>+}
>+
>+std::unique_ptr<RtcEventVideoReceiveStreamConfig>
>+EventGenerator::NewVideoReceiveStreamConfig(
>+    uint32_t ssrc,
>+    const RtpHeaderExtensionMap& extensions) {
>+  auto config = absl::make_unique<rtclog::StreamConfig>();
>+
>+  // Add SSRCs for the stream.
>+  config->remote_ssrc = ssrc;
>+  config->local_ssrc = prng_.Rand<uint32_t>();
>+  // Add extensions and settings for RTCP.
>+  config->rtcp_mode =
>+      prng_.Rand<bool>() ? RtcpMode::kCompound : RtcpMode::kReducedSize;
>+  config->remb = prng_.Rand<bool>();
>+  config->rtx_ssrc = prng_.Rand<uint32_t>();
>+  config->codecs.emplace_back(prng_.Rand<bool>() ? "VP8" : "H264",
>+                              prng_.Rand(127), prng_.Rand(127));
>+  // Add header extensions.
>+  for (size_t i = 0; i < kMaxNumExtensions; i++) {
>+    uint8_t id = extensions.GetId(kExtensions[i].type);
>+    if (id != RtpHeaderExtensionMap::kInvalidId) {
>+      config->rtp_extensions.emplace_back(kExtensions[i].name, id);
>+    }
>   }
>-  if ((type == rtclog::Event::BWE_PROBE_RESULT_EVENT) !=
>-      event.has_probe_result()) {
>-    return ::testing::AssertionFailure()
>-           << "Event of type " << type << " has "
>-           << (event.has_probe_result() ? "" : "no ") << "bwe probe result";
>+  return absl::make_unique<RtcEventVideoReceiveStreamConfig>(std::move(config));
>+}
>+
>+std::unique_ptr<RtcEventVideoSendStreamConfig>
>+EventGenerator::NewVideoSendStreamConfig(
>+    uint32_t ssrc,
>+    const RtpHeaderExtensionMap& extensions) {
>+  auto config = absl::make_unique<rtclog::StreamConfig>();
>+
>+  config->codecs.emplace_back(prng_.Rand<bool>() ? "VP8" : "H264",
>+                              prng_.Rand(127), prng_.Rand(127));
>+  config->local_ssrc = ssrc;
>+  config->rtx_ssrc = prng_.Rand<uint32_t>();
>+  // Add header extensions.
>+  for (size_t i = 0; i < kMaxNumExtensions; i++) {
>+    uint8_t id = extensions.GetId(kExtensions[i].type);
>+    if (id != RtpHeaderExtensionMap::kInvalidId) {
>+      config->rtp_extensions.emplace_back(kExtensions[i].name, id);
>+    }
>   }
>-  return ::testing::AssertionSuccess();
>+  return absl::make_unique<RtcEventVideoSendStreamConfig>(std::move(config));
>+}
>+
>+bool VerifyLoggedAlrStateEvent(const RtcEventAlrState& original_event,
>+                               const LoggedAlrStateEvent& logged_event) {
>+  if (original_event.timestamp_us_ != logged_event.log_time_us())
>+    return false;
>+  if (original_event.in_alr_ != logged_event.in_alr)
>+    return false;
>+  return true;
> }
> 
>-void VerifyStreamConfigsAreEqual(const rtclog::StreamConfig& config_1,
>-                                 const rtclog::StreamConfig& config_2) {
>-  EXPECT_EQ(config_1.remote_ssrc, config_2.remote_ssrc);
>-  EXPECT_EQ(config_1.local_ssrc, config_2.local_ssrc);
>-  EXPECT_EQ(config_1.rtx_ssrc, config_2.rtx_ssrc);
>-  EXPECT_EQ(config_1.rtcp_mode, config_2.rtcp_mode);
>-  EXPECT_EQ(config_1.remb, config_2.remb);
>+bool VerifyLoggedAudioPlayoutEvent(
>+    const RtcEventAudioPlayout& original_event,
>+    const LoggedAudioPlayoutEvent& logged_event) {
>+  if (original_event.timestamp_us_ != logged_event.log_time_us())
>+    return false;
>+  if (original_event.ssrc_ != logged_event.ssrc)
>+    return false;
>+  return true;
>+}
> 
>-  ASSERT_EQ(config_1.rtp_extensions.size(), config_2.rtp_extensions.size());
>-  for (size_t i = 0; i < config_2.rtp_extensions.size(); i++) {
>-    EXPECT_EQ(config_1.rtp_extensions[i].uri, config_2.rtp_extensions[i].uri);
>-    EXPECT_EQ(config_1.rtp_extensions[i].id, config_2.rtp_extensions[i].id);
>-  }
>-  ASSERT_EQ(config_1.codecs.size(), config_2.codecs.size());
>-  for (size_t i = 0; i < config_2.codecs.size(); i++) {
>-    EXPECT_EQ(config_1.codecs[i].payload_name, config_2.codecs[i].payload_name);
>-    EXPECT_EQ(config_1.codecs[i].payload_type, config_2.codecs[i].payload_type);
>-    EXPECT_EQ(config_1.codecs[i].rtx_payload_type,
>-              config_2.codecs[i].rtx_payload_type);
>-  }
>+bool VerifyLoggedAudioNetworkAdaptationEvent(
>+    const RtcEventAudioNetworkAdaptation& original_event,
>+    const LoggedAudioNetworkAdaptationEvent& logged_event) {
>+  if (original_event.timestamp_us_ != logged_event.log_time_us())
>+    return false;
>+
>+  if (original_event.config_->bitrate_bps != logged_event.config.bitrate_bps)
>+    return false;
>+  if (original_event.config_->enable_dtx != logged_event.config.enable_dtx)
>+    return false;
>+  if (original_event.config_->enable_fec != logged_event.config.enable_fec)
>+    return false;
>+  if (original_event.config_->frame_length_ms !=
>+      logged_event.config.frame_length_ms)
>+    return false;
>+  if (original_event.config_->num_channels != logged_event.config.num_channels)
>+    return false;
>+  if (original_event.config_->uplink_packet_loss_fraction !=
>+      logged_event.config.uplink_packet_loss_fraction)
>+    return false;
>+
>+  return true;
> }
> 
>-void RtcEventLogTestHelper::VerifyVideoReceiveStreamConfig(
>-    const ParsedRtcEventLog& parsed_log,
>-    size_t index,
>-    const rtclog::StreamConfig& config) {
>-  ASSERT_LT(index, parsed_log.events_.size());
>-  const rtclog::Event& event = parsed_log.events_[index];
>-  ASSERT_TRUE(IsValidBasicEvent(event));
>-  ASSERT_EQ(rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT, event.type());
>-  const rtclog::VideoReceiveConfig& receiver_config =
>-      event.video_receiver_config();
>-  // Check SSRCs.
>-  ASSERT_TRUE(receiver_config.has_remote_ssrc());
>-  EXPECT_EQ(config.remote_ssrc, receiver_config.remote_ssrc());
>-  ASSERT_TRUE(receiver_config.has_local_ssrc());
>-  EXPECT_EQ(config.local_ssrc, receiver_config.local_ssrc());
>-  // Check RTCP settings.
>-  ASSERT_TRUE(receiver_config.has_rtcp_mode());
>-  if (config.rtcp_mode == RtcpMode::kCompound) {
>-    EXPECT_EQ(rtclog::VideoReceiveConfig::RTCP_COMPOUND,
>-              receiver_config.rtcp_mode());
>-  } else {
>-    EXPECT_EQ(rtclog::VideoReceiveConfig::RTCP_REDUCEDSIZE,
>-              receiver_config.rtcp_mode());
>-  }
>-  ASSERT_TRUE(receiver_config.has_remb());
>-  EXPECT_EQ(config.remb, receiver_config.remb());
>-  // Check RTX map.
>-  for (const rtclog::RtxMap& rtx_map : receiver_config.rtx_map()) {
>-    ASSERT_TRUE(rtx_map.has_payload_type());
>-    ASSERT_TRUE(rtx_map.has_config());
>-    const rtclog::RtxConfig& rtx_config = rtx_map.config();
>-    ASSERT_TRUE(rtx_config.has_rtx_ssrc());
>-    ASSERT_TRUE(rtx_config.has_rtx_payload_type());
>-
>-    EXPECT_EQ(config.rtx_ssrc, rtx_config.rtx_ssrc());
>-    auto codec_found =
>-        std::find_if(config.codecs.begin(), config.codecs.end(),
>-                     [&rtx_map](const rtclog::StreamConfig::Codec& codec) {
>-                       return rtx_map.payload_type() == codec.payload_type;
>-                     });
>-    ASSERT_TRUE(codec_found != config.codecs.end());
>-    EXPECT_EQ(rtx_config.rtx_payload_type(), codec_found->rtx_payload_type);
>+bool VerifyLoggedBweDelayBasedUpdate(
>+    const RtcEventBweUpdateDelayBased& original_event,
>+    const LoggedBweDelayBasedUpdate& logged_event) {
>+  if (original_event.timestamp_us_ != logged_event.log_time_us())
>+    return false;
>+  if (original_event.bitrate_bps_ != logged_event.bitrate_bps)
>+    return false;
>+  if (original_event.detector_state_ != logged_event.detector_state)
>+    return false;
>+  return true;
>+}
>+
>+bool VerifyLoggedBweLossBasedUpdate(
>+    const RtcEventBweUpdateLossBased& original_event,
>+    const LoggedBweLossBasedUpdate& logged_event) {
>+  if (original_event.timestamp_us_ != logged_event.log_time_us())
>+    return false;
>+  if (original_event.bitrate_bps_ != logged_event.bitrate_bps)
>+    return false;
>+  if (original_event.fraction_loss_ != logged_event.fraction_lost)
>+    return false;
>+  if (original_event.total_packets_ != logged_event.expected_packets)
>+    return false;
>+  return true;
>+}
>+
>+bool VerifyLoggedBweProbeClusterCreatedEvent(
>+    const RtcEventProbeClusterCreated& original_event,
>+    const LoggedBweProbeClusterCreatedEvent& logged_event) {
>+  if (original_event.timestamp_us_ != logged_event.log_time_us())
>+    return false;
>+  if (original_event.id_ != logged_event.id)
>+    return false;
>+  if (original_event.bitrate_bps_ != logged_event.bitrate_bps)
>+    return false;
>+  if (original_event.min_probes_ != logged_event.min_packets)
>+    return false;
>+  if (original_event.min_bytes_ != logged_event.min_bytes)
>+    return false;
>+
>+  return true;
>+}
>+
>+bool VerifyLoggedBweProbeFailureEvent(
>+    const RtcEventProbeResultFailure& original_event,
>+    const LoggedBweProbeFailureEvent& logged_event) {
>+  if (original_event.timestamp_us_ != logged_event.log_time_us())
>+    return false;
>+  if (original_event.id_ != logged_event.id)
>+    return false;
>+  if (original_event.failure_reason_ != logged_event.failure_reason)
>+    return false;
>+  return true;
>+}
>+
>+bool VerifyLoggedBweProbeSuccessEvent(
>+    const RtcEventProbeResultSuccess& original_event,
>+    const LoggedBweProbeSuccessEvent& logged_event) {
>+  if (original_event.timestamp_us_ != logged_event.log_time_us())
>+    return false;
>+  if (original_event.id_ != logged_event.id)
>+    return false;
>+  if (original_event.bitrate_bps_ != logged_event.bitrate_bps)
>+    return false;
>+  return true;
>+}
>+
>+bool VerifyLoggedIceCandidatePairConfig(
>+    const RtcEventIceCandidatePairConfig& original_event,
>+    const LoggedIceCandidatePairConfig& logged_event) {
>+  if (original_event.timestamp_us_ != logged_event.log_time_us())
>+    return false;
>+
>+  if (original_event.type_ != logged_event.type)
>+    return false;
>+  if (original_event.candidate_pair_id_ != logged_event.candidate_pair_id)
>+    return false;
>+  if (original_event.candidate_pair_desc_.local_candidate_type !=
>+      logged_event.local_candidate_type)
>+    return false;
>+  if (original_event.candidate_pair_desc_.local_relay_protocol !=
>+      logged_event.local_relay_protocol)
>+    return false;
>+  if (original_event.candidate_pair_desc_.local_network_type !=
>+      logged_event.local_network_type)
>+    return false;
>+  if (original_event.candidate_pair_desc_.local_address_family !=
>+      logged_event.local_address_family)
>+    return false;
>+  if (original_event.candidate_pair_desc_.remote_candidate_type !=
>+      logged_event.remote_candidate_type)
>+    return false;
>+  if (original_event.candidate_pair_desc_.remote_address_family !=
>+      logged_event.remote_address_family)
>+    return false;
>+  if (original_event.candidate_pair_desc_.candidate_pair_protocol !=
>+      logged_event.candidate_pair_protocol)
>+    return false;
>+
>+  return true;
>+}
>+
>+bool VerifyLoggedIceCandidatePairEvent(
>+    const RtcEventIceCandidatePair& original_event,
>+    const LoggedIceCandidatePairEvent& logged_event) {
>+  if (original_event.timestamp_us_ != logged_event.log_time_us())
>+    return false;
>+
>+  if (original_event.type_ != logged_event.type)
>+    return false;
>+  if (original_event.candidate_pair_id_ != logged_event.candidate_pair_id)
>+    return false;
>+
>+  return true;
>+}
>+
>+bool VerifyLoggedRtpHeader(const RtpPacket& original_header,
>+                           const RTPHeader& logged_header) {
>+  // Standard RTP header.
>+  if (original_header.Marker() != logged_header.markerBit)
>+    return false;
>+  if (original_header.PayloadType() != logged_header.payloadType)
>+    return false;
>+  if (original_header.SequenceNumber() != logged_header.sequenceNumber)
>+    return false;
>+  if (original_header.Timestamp() != logged_header.timestamp)
>+    return false;
>+  if (original_header.Ssrc() != logged_header.ssrc)
>+    return false;
>+  if (original_header.Csrcs().size() != logged_header.numCSRCs)
>+    return false;
>+  for (size_t i = 0; i < logged_header.numCSRCs; i++) {
>+    if (original_header.Csrcs()[i] != logged_header.arrOfCSRCs[i])
>+      return false;
>   }
>-  // Check header extensions.
>-  ASSERT_EQ(static_cast<int>(config.rtp_extensions.size()),
>-            receiver_config.header_extensions_size());
>-  for (int i = 0; i < receiver_config.header_extensions_size(); i++) {
>-    ASSERT_TRUE(receiver_config.header_extensions(i).has_name());
>-    ASSERT_TRUE(receiver_config.header_extensions(i).has_id());
>-    const std::string& name = receiver_config.header_extensions(i).name();
>-    int id = receiver_config.header_extensions(i).id();
>-    EXPECT_EQ(config.rtp_extensions[i].id, id);
>-    EXPECT_EQ(config.rtp_extensions[i].uri, name);
>+
>+  if (original_header.padding_size() != logged_header.paddingLength)
>+    return false;
>+  if (original_header.headers_size() != logged_header.headerLength)
>+    return false;
>+
>+  // TransmissionOffset header extension.
>+  if (original_header.HasExtension<TransmissionOffset>() !=
>+      logged_header.extension.hasTransmissionTimeOffset)
>+    return false;
>+  if (logged_header.extension.hasTransmissionTimeOffset) {
>+    int32_t offset;
>+    original_header.GetExtension<TransmissionOffset>(&offset);
>+    if (offset != logged_header.extension.transmissionTimeOffset)
>+      return false;
>   }
>-  // Check decoders.
>-  ASSERT_EQ(static_cast<int>(config.codecs.size()),
>-            receiver_config.decoders_size());
>-  for (int i = 0; i < receiver_config.decoders_size(); i++) {
>-    ASSERT_TRUE(receiver_config.decoders(i).has_name());
>-    ASSERT_TRUE(receiver_config.decoders(i).has_payload_type());
>-    const std::string& decoder_name = receiver_config.decoders(i).name();
>-    int decoder_type = receiver_config.decoders(i).payload_type();
>-    EXPECT_EQ(config.codecs[i].payload_name, decoder_name);
>-    EXPECT_EQ(config.codecs[i].payload_type, decoder_type);
>+
>+  // AbsoluteSendTime header extension.
>+  if (original_header.HasExtension<AbsoluteSendTime>() !=
>+      logged_header.extension.hasAbsoluteSendTime)
>+    return false;
>+  if (logged_header.extension.hasAbsoluteSendTime) {
>+    uint32_t sendtime;
>+    original_header.GetExtension<AbsoluteSendTime>(&sendtime);
>+    if (sendtime != logged_header.extension.absoluteSendTime)
>+      return false;
>   }
> 
>-  // Check consistency of the parser.
>-  rtclog::StreamConfig parsed_config = parsed_log.GetVideoReceiveConfig(index);
>-  VerifyStreamConfigsAreEqual(config, parsed_config);
>-}
>-
>-void RtcEventLogTestHelper::VerifyVideoSendStreamConfig(
>-    const ParsedRtcEventLog& parsed_log,
>-    size_t index,
>-    const rtclog::StreamConfig& config) {
>-  ASSERT_LT(index, parsed_log.events_.size());
>-  const rtclog::Event& event = parsed_log.events_[index];
>-  ASSERT_TRUE(IsValidBasicEvent(event));
>-  ASSERT_EQ(rtclog::Event::VIDEO_SENDER_CONFIG_EVENT, event.type());
>-  const rtclog::VideoSendConfig& sender_config = event.video_sender_config();
>-
>-  EXPECT_EQ(config.local_ssrc, sender_config.ssrcs(0));
>-  EXPECT_EQ(config.rtx_ssrc, sender_config.rtx_ssrcs(0));
>-
>-  // Check header extensions.
>-  ASSERT_EQ(static_cast<int>(config.rtp_extensions.size()),
>-            sender_config.header_extensions_size());
>-  for (int i = 0; i < sender_config.header_extensions_size(); i++) {
>-    ASSERT_TRUE(sender_config.header_extensions(i).has_name());
>-    ASSERT_TRUE(sender_config.header_extensions(i).has_id());
>-    const std::string& name = sender_config.header_extensions(i).name();
>-    int id = sender_config.header_extensions(i).id();
>-    EXPECT_EQ(config.rtp_extensions[i].id, id);
>-    EXPECT_EQ(config.rtp_extensions[i].uri, name);
>+  // TransportSequenceNumber header extension.
>+  if (original_header.HasExtension<TransportSequenceNumber>() !=
>+      logged_header.extension.hasTransportSequenceNumber)
>+    return false;
>+  if (logged_header.extension.hasTransportSequenceNumber) {
>+    uint16_t seqnum;
>+    original_header.GetExtension<TransportSequenceNumber>(&seqnum);
>+    if (seqnum != logged_header.extension.transportSequenceNumber)
>+      return false;
>   }
>-  // Check encoder.
>-  ASSERT_TRUE(sender_config.has_encoder());
>-  ASSERT_TRUE(sender_config.encoder().has_name());
>-  ASSERT_TRUE(sender_config.encoder().has_payload_type());
>-  EXPECT_EQ(config.codecs[0].payload_name, sender_config.encoder().name());
>-  EXPECT_EQ(config.codecs[0].payload_type,
>-            sender_config.encoder().payload_type());
>-
>-  EXPECT_EQ(config.codecs[0].rtx_payload_type,
>-            sender_config.rtx_payload_type());
>-
>-  // Check consistency of the parser.
>-  std::vector<rtclog::StreamConfig> parsed_configs =
>-      parsed_log.GetVideoSendConfig(index);
>-  ASSERT_EQ(1u, parsed_configs.size());
>-  VerifyStreamConfigsAreEqual(config, parsed_configs[0]);
>-}
>-
>-void RtcEventLogTestHelper::VerifyAudioReceiveStreamConfig(
>-    const ParsedRtcEventLog& parsed_log,
>-    size_t index,
>-    const rtclog::StreamConfig& config) {
>-  ASSERT_LT(index, parsed_log.events_.size());
>-  const rtclog::Event& event = parsed_log.events_[index];
>-  ASSERT_TRUE(IsValidBasicEvent(event));
>-  ASSERT_EQ(rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT, event.type());
>-  const rtclog::AudioReceiveConfig& receiver_config =
>-      event.audio_receiver_config();
>-  // Check SSRCs.
>-  ASSERT_TRUE(receiver_config.has_remote_ssrc());
>-  EXPECT_EQ(config.remote_ssrc, receiver_config.remote_ssrc());
>-  ASSERT_TRUE(receiver_config.has_local_ssrc());
>-  EXPECT_EQ(config.local_ssrc, receiver_config.local_ssrc());
>-  // Check header extensions.
>-  ASSERT_EQ(static_cast<int>(config.rtp_extensions.size()),
>-            receiver_config.header_extensions_size());
>-  for (int i = 0; i < receiver_config.header_extensions_size(); i++) {
>-    ASSERT_TRUE(receiver_config.header_extensions(i).has_name());
>-    ASSERT_TRUE(receiver_config.header_extensions(i).has_id());
>-    const std::string& name = receiver_config.header_extensions(i).name();
>-    int id = receiver_config.header_extensions(i).id();
>-    EXPECT_EQ(config.rtp_extensions[i].id, id);
>-    EXPECT_EQ(config.rtp_extensions[i].uri, name);
>+
>+  // AudioLevel header extension.
>+  if (original_header.HasExtension<AudioLevel>() !=
>+      logged_header.extension.hasAudioLevel)
>+    return false;
>+  if (logged_header.extension.hasAudioLevel) {
>+    bool voice_activity;
>+    uint8_t audio_level;
>+    original_header.GetExtension<AudioLevel>(&voice_activity, &audio_level);
>+    if (voice_activity != logged_header.extension.voiceActivity)
>+      return false;
>+    if (audio_level != logged_header.extension.audioLevel)
>+      return false;
>   }
> 
>-  // Check consistency of the parser.
>-  rtclog::StreamConfig parsed_config = parsed_log.GetAudioReceiveConfig(index);
>-  EXPECT_EQ(config.remote_ssrc, parsed_config.remote_ssrc);
>-  EXPECT_EQ(config.local_ssrc, parsed_config.local_ssrc);
>-  // Check header extensions.
>-  EXPECT_EQ(config.rtp_extensions.size(), parsed_config.rtp_extensions.size());
>-  for (size_t i = 0; i < parsed_config.rtp_extensions.size(); i++) {
>-    EXPECT_EQ(config.rtp_extensions[i].uri,
>-              parsed_config.rtp_extensions[i].uri);
>-    EXPECT_EQ(config.rtp_extensions[i].id, parsed_config.rtp_extensions[i].id);
>+  // VideoOrientation header extension.
>+  if (original_header.HasExtension<VideoOrientation>() !=
>+      logged_header.extension.hasVideoRotation)
>+    return false;
>+  if (logged_header.extension.hasVideoRotation) {
>+    uint8_t rotation;
>+    original_header.GetExtension<VideoOrientation>(&rotation);
>+    if (ConvertCVOByteToVideoRotation(rotation) !=
>+        logged_header.extension.videoRotation)
>+      return false;
>   }
>+
>+  return true;
>+}
>+
>+bool VerifyLoggedRtpPacketIncoming(
>+    const RtcEventRtpPacketIncoming& original_event,
>+    const LoggedRtpPacketIncoming& logged_event) {
>+  if (original_event.timestamp_us_ != logged_event.log_time_us())
>+    return false;
>+
>+  if (original_event.header_.headers_size() != logged_event.rtp.header_length)
>+    return false;
>+
>+  if (original_event.packet_length_ != logged_event.rtp.total_length)
>+    return false;
>+
>+  if (!VerifyLoggedRtpHeader(original_event.header_, logged_event.rtp.header))
>+    return false;
>+
>+  return true;
>+}
>+
>+bool VerifyLoggedRtpPacketOutgoing(
>+    const RtcEventRtpPacketOutgoing& original_event,
>+    const LoggedRtpPacketOutgoing& logged_event) {
>+  if (original_event.timestamp_us_ != logged_event.log_time_us())
>+    return false;
>+
>+  if (original_event.header_.headers_size() != logged_event.rtp.header_length)
>+    return false;
>+
>+  if (original_event.packet_length_ != logged_event.rtp.total_length)
>+    return false;
>+
>+  // TODO(terelius): Probe cluster ID isn't parsed, used or tested. Unless
>+  // someone has a strong reason to keep it, it'll be removed.
>+
>+  if (!VerifyLoggedRtpHeader(original_event.header_, logged_event.rtp.header))
>+    return false;
>+
>+  return true;
> }
> 
>-void RtcEventLogTestHelper::VerifyAudioSendStreamConfig(
>-    const ParsedRtcEventLog& parsed_log,
>-    size_t index,
>-    const rtclog::StreamConfig& config) {
>-  ASSERT_LT(index, parsed_log.events_.size());
>-  const rtclog::Event& event = parsed_log.events_[index];
>-  ASSERT_TRUE(IsValidBasicEvent(event));
>-  ASSERT_EQ(rtclog::Event::AUDIO_SENDER_CONFIG_EVENT, event.type());
>-  const rtclog::AudioSendConfig& sender_config = event.audio_sender_config();
>-  // Check SSRCs.
>-  EXPECT_EQ(config.local_ssrc, sender_config.ssrc());
>-  // Check header extensions.
>-  ASSERT_EQ(static_cast<int>(config.rtp_extensions.size()),
>-            sender_config.header_extensions_size());
>-  for (int i = 0; i < sender_config.header_extensions_size(); i++) {
>-    ASSERT_TRUE(sender_config.header_extensions(i).has_name());
>-    ASSERT_TRUE(sender_config.header_extensions(i).has_id());
>-    const std::string& name = sender_config.header_extensions(i).name();
>-    int id = sender_config.header_extensions(i).id();
>-    EXPECT_EQ(config.rtp_extensions[i].id, id);
>-    EXPECT_EQ(config.rtp_extensions[i].uri, name);
>+bool VerifyLoggedRtcpPacketIncoming(
>+    const RtcEventRtcpPacketIncoming& original_event,
>+    const LoggedRtcpPacketIncoming& logged_event) {
>+  if (original_event.timestamp_us_ != logged_event.log_time_us())
>+    return false;
>+
>+  if (original_event.packet_.size() != logged_event.rtcp.raw_data.size())
>+    return false;
>+  if (memcmp(original_event.packet_.data(), logged_event.rtcp.raw_data.data(),
>+             original_event.packet_.size()) != 0) {
>+    return false;
>   }
>+  return true;
>+}
> 
>-  // Check consistency of the parser.
>-  rtclog::StreamConfig parsed_config = parsed_log.GetAudioSendConfig(index);
>-  VerifyStreamConfigsAreEqual(config, parsed_config);
>-}
>-
>-void RtcEventLogTestHelper::VerifyIncomingRtpEvent(
>-    const ParsedRtcEventLog& parsed_log,
>-    size_t index,
>-    const RtpPacketReceived& expected_packet) {
>-  ASSERT_LT(index, parsed_log.events_.size());
>-  const rtclog::Event& event = parsed_log.events_[index];
>-  ASSERT_TRUE(IsValidBasicEvent(event));
>-  ASSERT_EQ(rtclog::Event::RTP_EVENT, event.type());
>-  const rtclog::RtpPacket& rtp_packet = event.rtp_packet();
>-  ASSERT_TRUE(rtp_packet.has_incoming());
>-  EXPECT_TRUE(rtp_packet.incoming());
>-  ASSERT_TRUE(rtp_packet.has_packet_length());
>-  EXPECT_EQ(expected_packet.size(), rtp_packet.packet_length());
>-  size_t header_size = expected_packet.headers_size();
>-  ASSERT_TRUE(rtp_packet.has_header());
>-  EXPECT_THAT(testing::make_tuple(expected_packet.data(), header_size),
>-              testing::ElementsAreArray(rtp_packet.header().data(),
>-                                        rtp_packet.header().size()));
>-
>-  // Check consistency of the parser.
>-  PacketDirection parsed_direction;
>-  uint8_t parsed_header[1500];
>-  size_t parsed_header_size, parsed_total_size;
>-  parsed_log.GetRtpHeader(index, &parsed_direction, parsed_header,
>-                          &parsed_header_size, &parsed_total_size, nullptr);
>-  EXPECT_EQ(kIncomingPacket, parsed_direction);
>-  EXPECT_THAT(testing::make_tuple(expected_packet.data(), header_size),
>-              testing::ElementsAreArray(parsed_header, parsed_header_size));
>-  EXPECT_EQ(expected_packet.size(), parsed_total_size);
>-}
>-
>-void RtcEventLogTestHelper::VerifyOutgoingRtpEvent(
>-    const ParsedRtcEventLog& parsed_log,
>-    size_t index,
>-    const RtpPacketToSend& expected_packet) {
>-  ASSERT_LT(index, parsed_log.events_.size());
>-  const rtclog::Event& event = parsed_log.events_[index];
>-  ASSERT_TRUE(IsValidBasicEvent(event));
>-  ASSERT_EQ(rtclog::Event::RTP_EVENT, event.type());
>-  const rtclog::RtpPacket& rtp_packet = event.rtp_packet();
>-  ASSERT_TRUE(rtp_packet.has_incoming());
>-  EXPECT_FALSE(rtp_packet.incoming());
>-  ASSERT_TRUE(rtp_packet.has_packet_length());
>-  EXPECT_EQ(expected_packet.size(), rtp_packet.packet_length());
>-  size_t header_size = expected_packet.headers_size();
>-  ASSERT_TRUE(rtp_packet.has_header());
>-  EXPECT_THAT(testing::make_tuple(expected_packet.data(), header_size),
>-              testing::ElementsAreArray(rtp_packet.header().data(),
>-                                        rtp_packet.header().size()));
>-
>-  // Check consistency of the parser.
>-  PacketDirection parsed_direction;
>-  uint8_t parsed_header[1500];
>-  size_t parsed_header_size, parsed_total_size;
>-  parsed_log.GetRtpHeader(index, &parsed_direction, parsed_header,
>-                          &parsed_header_size, &parsed_total_size, nullptr);
>-  EXPECT_EQ(kOutgoingPacket, parsed_direction);
>-  EXPECT_THAT(testing::make_tuple(expected_packet.data(), header_size),
>-              testing::ElementsAreArray(parsed_header, parsed_header_size));
>-  EXPECT_EQ(expected_packet.size(), parsed_total_size);
>-}
>-
>-void RtcEventLogTestHelper::VerifyRtcpEvent(const ParsedRtcEventLog& parsed_log,
>-                                            size_t index,
>-                                            PacketDirection direction,
>-                                            const uint8_t* packet,
>-                                            size_t total_size) {
>-  ASSERT_LT(index, parsed_log.events_.size());
>-  const rtclog::Event& event = parsed_log.events_[index];
>-  ASSERT_TRUE(IsValidBasicEvent(event));
>-  ASSERT_EQ(rtclog::Event::RTCP_EVENT, event.type());
>-  const rtclog::RtcpPacket& rtcp_packet = event.rtcp_packet();
>-  ASSERT_TRUE(rtcp_packet.has_incoming());
>-  EXPECT_EQ(direction == kIncomingPacket, rtcp_packet.incoming());
>-  ASSERT_TRUE(rtcp_packet.has_packet_data());
>-  ASSERT_EQ(total_size, rtcp_packet.packet_data().size());
>-  for (size_t i = 0; i < total_size; i++) {
>-    EXPECT_EQ(packet[i], static_cast<uint8_t>(rtcp_packet.packet_data()[i]));
>+bool VerifyLoggedRtcpPacketOutgoing(
>+    const RtcEventRtcpPacketOutgoing& original_event,
>+    const LoggedRtcpPacketOutgoing& logged_event) {
>+  if (original_event.timestamp_us_ != logged_event.log_time_us())
>+    return false;
>+
>+  if (original_event.packet_.size() != logged_event.rtcp.raw_data.size())
>+    return false;
>+  if (memcmp(original_event.packet_.data(), logged_event.rtcp.raw_data.data(),
>+             original_event.packet_.size()) != 0) {
>+    return false;
>   }
>+  return true;
>+}
> 
>-  // Check consistency of the parser.
>-  PacketDirection parsed_direction;
>-  uint8_t parsed_packet[1500];
>-  size_t parsed_total_size;
>-  parsed_log.GetRtcpPacket(index, &parsed_direction, parsed_packet,
>-                           &parsed_total_size);
>-  EXPECT_EQ(direction, parsed_direction);
>-  ASSERT_EQ(total_size, parsed_total_size);
>-  EXPECT_EQ(0, std::memcmp(packet, parsed_packet, total_size));
>+bool VerifyLoggedStartEvent(int64_t start_time_us,
>+                            const LoggedStartEvent& logged_event) {
>+  if (start_time_us != logged_event.log_time_us())
>+    return false;
>+  return true;
> }
> 
>-void RtcEventLogTestHelper::VerifyPlayoutEvent(
>-    const ParsedRtcEventLog& parsed_log,
>-    size_t index,
>-    uint32_t ssrc) {
>-  ASSERT_LT(index, parsed_log.events_.size());
>-  const rtclog::Event& event = parsed_log.events_[index];
>-  ASSERT_TRUE(IsValidBasicEvent(event));
>-  ASSERT_EQ(rtclog::Event::AUDIO_PLAYOUT_EVENT, event.type());
>-  const rtclog::AudioPlayoutEvent& playout_event = event.audio_playout_event();
>-  ASSERT_TRUE(playout_event.has_local_ssrc());
>-  EXPECT_EQ(ssrc, playout_event.local_ssrc());
>-
>-  // Check consistency of the parser.
>-  uint32_t parsed_ssrc;
>-  parsed_log.GetAudioPlayout(index, &parsed_ssrc);
>-  EXPECT_EQ(ssrc, parsed_ssrc);
>-}
>-
>-void RtcEventLogTestHelper::VerifyBweLossEvent(
>-    const ParsedRtcEventLog& parsed_log,
>-    size_t index,
>-    int32_t bitrate,
>-    uint8_t fraction_loss,
>-    int32_t total_packets) {
>-  ASSERT_LT(index, parsed_log.events_.size());
>-  const rtclog::Event& event = parsed_log.events_[index];
>-  ASSERT_TRUE(IsValidBasicEvent(event));
>-  ASSERT_EQ(rtclog::Event::LOSS_BASED_BWE_UPDATE, event.type());
>-  const rtclog::LossBasedBweUpdate& bwe_event = event.loss_based_bwe_update();
>-  ASSERT_TRUE(bwe_event.has_bitrate_bps());
>-  EXPECT_EQ(bitrate, bwe_event.bitrate_bps());
>-  ASSERT_TRUE(bwe_event.has_fraction_loss());
>-  EXPECT_EQ(fraction_loss, bwe_event.fraction_loss());
>-  ASSERT_TRUE(bwe_event.has_total_packets());
>-  EXPECT_EQ(total_packets, bwe_event.total_packets());
>-
>-  // Check consistency of the parser.
>-  int32_t parsed_bitrate;
>-  uint8_t parsed_fraction_loss;
>-  int32_t parsed_total_packets;
>-  parsed_log.GetLossBasedBweUpdate(
>-      index, &parsed_bitrate, &parsed_fraction_loss, &parsed_total_packets);
>-  EXPECT_EQ(bitrate, parsed_bitrate);
>-  EXPECT_EQ(fraction_loss, parsed_fraction_loss);
>-  EXPECT_EQ(total_packets, parsed_total_packets);
>-}
>-
>-void RtcEventLogTestHelper::VerifyBweDelayEvent(
>-    const ParsedRtcEventLog& parsed_log,
>-    size_t index,
>-    int32_t bitrate,
>-    BandwidthUsage detector_state) {
>-  ASSERT_LT(index, parsed_log.events_.size());
>-  const rtclog::Event& event = parsed_log.events_[index];
>-  ASSERT_TRUE(IsValidBasicEvent(event));
>-  ASSERT_EQ(rtclog::Event::DELAY_BASED_BWE_UPDATE, event.type());
>-  const rtclog::DelayBasedBweUpdate& bwe_event = event.delay_based_bwe_update();
>-  ASSERT_TRUE(bwe_event.has_bitrate_bps());
>-  EXPECT_EQ(bitrate, bwe_event.bitrate_bps());
>-  ASSERT_TRUE(bwe_event.has_detector_state());
>-  EXPECT_EQ(detector_state,
>-            GetRuntimeDetectorState(bwe_event.detector_state()));
>-
>-  // Check consistency of the parser.
>-  ParsedRtcEventLog::BweDelayBasedUpdate res =
>-      parsed_log.GetDelayBasedBweUpdate(index);
>-  EXPECT_EQ(res.bitrate_bps, bitrate);
>-  EXPECT_EQ(res.detector_state, detector_state);
>-}
>-
>-void RtcEventLogTestHelper::VerifyAudioNetworkAdaptation(
>-    const ParsedRtcEventLog& parsed_log,
>-    size_t index,
>-    const AudioEncoderRuntimeConfig& config) {
>-  AudioEncoderRuntimeConfig parsed_config;
>-  parsed_log.GetAudioNetworkAdaptation(index, &parsed_config);
>-  EXPECT_EQ(config.bitrate_bps, parsed_config.bitrate_bps);
>-  EXPECT_EQ(config.enable_dtx, parsed_config.enable_dtx);
>-  EXPECT_EQ(config.enable_fec, parsed_config.enable_fec);
>-  EXPECT_EQ(config.frame_length_ms, parsed_config.frame_length_ms);
>-  EXPECT_EQ(config.num_channels, parsed_config.num_channels);
>-  EXPECT_EQ(config.uplink_packet_loss_fraction,
>-            parsed_config.uplink_packet_loss_fraction);
>-}
>-
>-void RtcEventLogTestHelper::VerifyLogStartEvent(
>-    const ParsedRtcEventLog& parsed_log,
>-    size_t index) {
>-  ASSERT_LT(index, parsed_log.events_.size());
>-  const rtclog::Event& event = parsed_log.events_[index];
>-  ASSERT_TRUE(IsValidBasicEvent(event));
>-  EXPECT_EQ(rtclog::Event::LOG_START, event.type());
>-}
>-
>-void RtcEventLogTestHelper::VerifyLogEndEvent(
>-    const ParsedRtcEventLog& parsed_log,
>-    size_t index) {
>-  ASSERT_LT(index, parsed_log.events_.size());
>-  const rtclog::Event& event = parsed_log.events_[index];
>-  ASSERT_TRUE(IsValidBasicEvent(event));
>-  EXPECT_EQ(rtclog::Event::LOG_END, event.type());
>-}
>-
>-void RtcEventLogTestHelper::VerifyBweProbeCluster(
>-    const ParsedRtcEventLog& parsed_log,
>-    size_t index,
>-    uint32_t id,
>-    uint32_t bitrate_bps,
>-    uint32_t min_probes,
>-    uint32_t min_bytes) {
>-  ASSERT_LT(index, parsed_log.events_.size());
>-  const rtclog::Event& event = parsed_log.events_[index];
>-  ASSERT_TRUE(IsValidBasicEvent(event));
>-  EXPECT_EQ(rtclog::Event::BWE_PROBE_CLUSTER_CREATED_EVENT, event.type());
>-
>-  const rtclog::BweProbeCluster& bwe_event = event.probe_cluster();
>-  ASSERT_TRUE(bwe_event.has_id());
>-  EXPECT_EQ(id, bwe_event.id());
>-  ASSERT_TRUE(bwe_event.has_bitrate_bps());
>-  EXPECT_EQ(bitrate_bps, bwe_event.bitrate_bps());
>-  ASSERT_TRUE(bwe_event.has_min_packets());
>-  EXPECT_EQ(min_probes, bwe_event.min_packets());
>-  ASSERT_TRUE(bwe_event.has_min_bytes());
>-  EXPECT_EQ(min_bytes, bwe_event.min_bytes());
>-
>-  // TODO(philipel): Verify the parser when parsing has been implemented.
>-}
>-
>-void RtcEventLogTestHelper::VerifyProbeResultSuccess(
>-    const ParsedRtcEventLog& parsed_log,
>-    size_t index,
>-    uint32_t id,
>-    uint32_t bitrate_bps) {
>-  ASSERT_LT(index, parsed_log.events_.size());
>-  const rtclog::Event& event = parsed_log.events_[index];
>-  ASSERT_TRUE(IsValidBasicEvent(event));
>-  EXPECT_EQ(rtclog::Event::BWE_PROBE_RESULT_EVENT, event.type());
>-
>-  const rtclog::BweProbeResult& bwe_event = event.probe_result();
>-  ASSERT_TRUE(bwe_event.has_id());
>-  EXPECT_EQ(id, bwe_event.id());
>-  ASSERT_TRUE(bwe_event.has_bitrate_bps());
>-  EXPECT_EQ(bitrate_bps, bwe_event.bitrate_bps());
>-  ASSERT_TRUE(bwe_event.has_result());
>-  EXPECT_EQ(rtclog::BweProbeResult::SUCCESS, bwe_event.result());
>-
>-  // TODO(philipel): Verify the parser when parsing has been implemented.
>-}
>-
>-void RtcEventLogTestHelper::VerifyProbeResultFailure(
>-    const ParsedRtcEventLog& parsed_log,
>-    size_t index,
>-    uint32_t id,
>-    ProbeFailureReason failure_reason) {
>-  ASSERT_LT(index, parsed_log.events_.size());
>-  const rtclog::Event& event = parsed_log.events_[index];
>-  ASSERT_TRUE(IsValidBasicEvent(event));
>-  EXPECT_EQ(rtclog::Event::BWE_PROBE_RESULT_EVENT, event.type());
>-
>-  const rtclog::BweProbeResult& bwe_event = event.probe_result();
>-  ASSERT_TRUE(bwe_event.has_id());
>-  EXPECT_EQ(id, bwe_event.id());
>-  ASSERT_TRUE(bwe_event.has_result());
>-  EXPECT_EQ(GetProbeResultType(failure_reason), bwe_event.result());
>-  ASSERT_FALSE(bwe_event.has_bitrate_bps());
>-
>-  // TODO(philipel): Verify the parser when parsing has been implemented.
>+bool VerifyLoggedStopEvent(int64_t stop_time_us,
>+                           const LoggedStopEvent& logged_event) {
>+  if (stop_time_us != logged_event.log_time_us())
>+    return false;
>+  return true;
>+}
>+
>+bool VerifyLoggedAudioRecvConfig(
>+    const RtcEventAudioReceiveStreamConfig& original_event,
>+    const LoggedAudioRecvConfig& logged_event) {
>+  if (original_event.timestamp_us_ != logged_event.log_time_us())
>+    return false;
>+  if (*original_event.config_ != logged_event.config)
>+    return false;
>+  return true;
>+}
>+
>+bool VerifyLoggedAudioSendConfig(
>+    const RtcEventAudioSendStreamConfig& original_event,
>+    const LoggedAudioSendConfig& logged_event) {
>+  if (original_event.timestamp_us_ != logged_event.log_time_us())
>+    return false;
>+  if (*original_event.config_ != logged_event.config)
>+    return false;
>+  return true;
>+}
>+
>+bool VerifyLoggedVideoRecvConfig(
>+    const RtcEventVideoReceiveStreamConfig& original_event,
>+    const LoggedVideoRecvConfig& logged_event) {
>+  if (original_event.timestamp_us_ != logged_event.log_time_us())
>+    return false;
>+  if (*original_event.config_ != logged_event.config)
>+    return false;
>+  return true;
>+}
>+
>+bool VerifyLoggedVideoSendConfig(
>+    const RtcEventVideoSendStreamConfig& original_event,
>+    const LoggedVideoSendConfig& logged_event) {
>+  if (original_event.timestamp_us_ != logged_event.log_time_us())
>+    return false;
>+  // TODO(terelius): In the past, we allowed storing multiple RtcStreamConfigs
>+  // in the same RtcEventVideoSendStreamConfig. Look into whether we should drop
>+  // backwards compatibility in the parser.
>+  if (logged_event.configs.size() != 1)
>+    return false;
>+  if (*original_event.config_ != logged_event.configs[0])
>+    return false;
>+  return true;
> }
> 
>+}  // namespace test
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.h
>index 630f160a185..d67469c0ffa 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.h
>@@ -11,81 +11,180 @@
> #ifndef LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_UNITTEST_HELPER_H_
> #define LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_UNITTEST_HELPER_H_
> 
>-#include "call/call.h"
>-#include "logging/rtc_event_log/rtc_event_log_parser.h"
>-#include "modules/rtp_rtcp/source/rtp_packet_received.h"
>-#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
>+#include <memory>
>+
>+#include "logging/rtc_event_log/events/rtc_event.h"
>+#include "logging/rtc_event_log/events/rtc_event_alr_state.h"
>+#include "logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h"
>+#include "logging/rtc_event_log/events/rtc_event_audio_playout.h"
>+#include "logging/rtc_event_log/events/rtc_event_audio_receive_stream_config.h"
>+#include "logging/rtc_event_log/events/rtc_event_audio_send_stream_config.h"
>+#include "logging/rtc_event_log/events/rtc_event_bwe_update_delay_based.h"
>+#include "logging/rtc_event_log/events/rtc_event_bwe_update_loss_based.h"
>+#include "logging/rtc_event_log/events/rtc_event_ice_candidate_pair.h"
>+#include "logging/rtc_event_log/events/rtc_event_ice_candidate_pair_config.h"
>+#include "logging/rtc_event_log/events/rtc_event_probe_cluster_created.h"
>+#include "logging/rtc_event_log/events/rtc_event_probe_result_failure.h"
>+#include "logging/rtc_event_log/events/rtc_event_probe_result_success.h"
>+#include "logging/rtc_event_log/events/rtc_event_rtcp_packet_incoming.h"
>+#include "logging/rtc_event_log/events/rtc_event_rtcp_packet_outgoing.h"
>+#include "logging/rtc_event_log/events/rtc_event_rtp_packet_incoming.h"
>+#include "logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.h"
>+#include "logging/rtc_event_log/events/rtc_event_video_receive_stream_config.h"
>+#include "logging/rtc_event_log/events/rtc_event_video_send_stream_config.h"
>+#include "logging/rtc_event_log/rtc_event_log_parser_new.h"
>+#include "modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
>+#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
>+#include "modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
>+#include "rtc_base/random.h"
> 
> namespace webrtc {
> 
>-class RtcEventLogTestHelper {
>+namespace test {
>+
>+class EventGenerator {
>  public:
>-  static void VerifyVideoReceiveStreamConfig(
>-      const ParsedRtcEventLog& parsed_log,
>-      size_t index,
>-      const rtclog::StreamConfig& config);
>-  static void VerifyVideoSendStreamConfig(const ParsedRtcEventLog& parsed_log,
>-                                          size_t index,
>-                                          const rtclog::StreamConfig& config);
>-  static void VerifyAudioReceiveStreamConfig(
>-      const ParsedRtcEventLog& parsed_log,
>-      size_t index,
>-      const rtclog::StreamConfig& config);
>-  static void VerifyAudioSendStreamConfig(const ParsedRtcEventLog& parsed_log,
>-                                          size_t index,
>-                                          const rtclog::StreamConfig& config);
>-  static void VerifyIncomingRtpEvent(const ParsedRtcEventLog& parsed_log,
>-                                     size_t index,
>-                                     const RtpPacketReceived& expected_packet);
>-  static void VerifyOutgoingRtpEvent(const ParsedRtcEventLog& parsed_log,
>-                                     size_t index,
>-                                     const RtpPacketToSend& expected_packet);
>-  static void VerifyRtcpEvent(const ParsedRtcEventLog& parsed_log,
>-                              size_t index,
>-                              PacketDirection direction,
>-                              const uint8_t* packet,
>-                              size_t total_size);
>-  static void VerifyPlayoutEvent(const ParsedRtcEventLog& parsed_log,
>-                                 size_t index,
>-                                 uint32_t ssrc);
>-  static void VerifyBweLossEvent(const ParsedRtcEventLog& parsed_log,
>-                                 size_t index,
>-                                 int32_t bitrate,
>-                                 uint8_t fraction_loss,
>-                                 int32_t total_packets);
>-  static void VerifyBweDelayEvent(const ParsedRtcEventLog& parsed_log,
>-                                  size_t index,
>-                                  int32_t bitrate,
>-                                  BandwidthUsage detector_state);
>-
>-  static void VerifyAudioNetworkAdaptation(
>-      const ParsedRtcEventLog& parsed_log,
>-      size_t index,
>-      const AudioEncoderRuntimeConfig& config);
>-
>-  static void VerifyLogStartEvent(const ParsedRtcEventLog& parsed_log,
>-                                  size_t index);
>-  static void VerifyLogEndEvent(const ParsedRtcEventLog& parsed_log,
>-                                size_t index);
>-
>-  static void VerifyBweProbeCluster(const ParsedRtcEventLog& parsed_log,
>-                                    size_t index,
>-                                    uint32_t id,
>-                                    uint32_t bitrate_bps,
>-                                    uint32_t min_probes,
>-                                    uint32_t min_bytes);
>-
>-  static void VerifyProbeResultSuccess(const ParsedRtcEventLog& parsed_log,
>-                                       size_t index,
>-                                       uint32_t id,
>-                                       uint32_t bitrate_bps);
>-
>-  static void VerifyProbeResultFailure(const ParsedRtcEventLog& parsed_log,
>-                                       size_t index,
>-                                       uint32_t id,
>-                                       ProbeFailureReason failure_reason);
>+  explicit EventGenerator(uint64_t seed) : prng_(seed) {}
>+
>+  std::unique_ptr<RtcEventAlrState> NewAlrState();
>+
>+  std::unique_ptr<RtcEventAudioPlayout> NewAudioPlayout(uint32_t ssrc);
>+
>+  std::unique_ptr<RtcEventAudioNetworkAdaptation> NewAudioNetworkAdaptation();
>+
>+  std::unique_ptr<RtcEventBweUpdateDelayBased> NewBweUpdateDelayBased();
>+
>+  std::unique_ptr<RtcEventBweUpdateLossBased> NewBweUpdateLossBased();
>+
>+  std::unique_ptr<RtcEventProbeClusterCreated> NewProbeClusterCreated();
>+
>+  std::unique_ptr<RtcEventProbeResultFailure> NewProbeResultFailure();
>+
>+  std::unique_ptr<RtcEventProbeResultSuccess> NewProbeResultSuccess();
>+
>+  std::unique_ptr<RtcEventIceCandidatePairConfig> NewIceCandidatePairConfig();
>+
>+  std::unique_ptr<RtcEventIceCandidatePair> NewIceCandidatePair();
>+
>+  std::unique_ptr<RtcEventRtcpPacketIncoming> NewRtcpPacketIncoming();
>+
>+  std::unique_ptr<RtcEventRtcpPacketOutgoing> NewRtcpPacketOutgoing();
>+
>+  void RandomizeRtpPacket(size_t packet_size,
>+                          uint32_t ssrc,
>+                          const RtpHeaderExtensionMap& extension_map,
>+                          RtpPacket* rtp_packet);
>+
>+  std::unique_ptr<RtcEventRtpPacketIncoming> NewRtpPacketIncoming(
>+      uint32_t ssrc,
>+      const RtpHeaderExtensionMap& extension_map);
>+
>+  std::unique_ptr<RtcEventRtpPacketOutgoing> NewRtpPacketOutgoing(
>+      uint32_t ssrc,
>+      const RtpHeaderExtensionMap& extension_map);
>+
>+  RtpHeaderExtensionMap NewRtpHeaderExtensionMap();
>+
>+  std::unique_ptr<RtcEventAudioReceiveStreamConfig> NewAudioReceiveStreamConfig(
>+      uint32_t ssrc,
>+      const RtpHeaderExtensionMap& extensions);
>+
>+  std::unique_ptr<RtcEventAudioSendStreamConfig> NewAudioSendStreamConfig(
>+      uint32_t ssrc,
>+      const RtpHeaderExtensionMap& extensions);
>+
>+  std::unique_ptr<RtcEventVideoReceiveStreamConfig> NewVideoReceiveStreamConfig(
>+      uint32_t ssrc,
>+      const RtpHeaderExtensionMap& extensions);
>+
>+  std::unique_ptr<RtcEventVideoSendStreamConfig> NewVideoSendStreamConfig(
>+      uint32_t ssrc,
>+      const RtpHeaderExtensionMap& extensions);
>+
>+ private:
>+  rtcp::ReportBlock NewReportBlock();
>+  rtcp::SenderReport NewSenderReport();
>+  rtcp::ReceiverReport NewReceiverReport();
>+
>+  Random prng_;
> };
> 
>+bool VerifyLoggedAlrStateEvent(const RtcEventAlrState& original_event,
>+                               const LoggedAlrStateEvent& logged_event);
>+
>+bool VerifyLoggedAudioPlayoutEvent(const RtcEventAudioPlayout& original_event,
>+                                   const LoggedAudioPlayoutEvent& logged_event);
>+
>+bool VerifyLoggedAudioNetworkAdaptationEvent(
>+    const RtcEventAudioNetworkAdaptation& original_event,
>+    const LoggedAudioNetworkAdaptationEvent& logged_event);
>+
>+bool VerifyLoggedBweDelayBasedUpdate(
>+    const RtcEventBweUpdateDelayBased& original_event,
>+    const LoggedBweDelayBasedUpdate& logged_event);
>+
>+bool VerifyLoggedBweLossBasedUpdate(
>+    const RtcEventBweUpdateLossBased& original_event,
>+    const LoggedBweLossBasedUpdate& logged_event);
>+
>+bool VerifyLoggedBweProbeClusterCreatedEvent(
>+    const RtcEventProbeClusterCreated& original_event,
>+    const LoggedBweProbeClusterCreatedEvent& logged_event);
>+
>+bool VerifyLoggedBweProbeFailureEvent(
>+    const RtcEventProbeResultFailure& original_event,
>+    const LoggedBweProbeFailureEvent& logged_event);
>+
>+bool VerifyLoggedBweProbeSuccessEvent(
>+    const RtcEventProbeResultSuccess& original_event,
>+    const LoggedBweProbeSuccessEvent& logged_event);
>+
>+bool VerifyLoggedIceCandidatePairConfig(
>+    const RtcEventIceCandidatePairConfig& original_event,
>+    const LoggedIceCandidatePairConfig& logged_event);
>+
>+bool VerifyLoggedIceCandidatePairEvent(
>+    const RtcEventIceCandidatePair& original_event,
>+    const LoggedIceCandidatePairEvent& logged_event);
>+
>+bool VerifyLoggedRtpPacketIncoming(
>+    const RtcEventRtpPacketIncoming& original_event,
>+    const LoggedRtpPacketIncoming& logged_event);
>+
>+bool VerifyLoggedRtpPacketOutgoing(
>+    const RtcEventRtpPacketOutgoing& original_event,
>+    const LoggedRtpPacketOutgoing& logged_event);
>+
>+bool VerifyLoggedRtcpPacketIncoming(
>+    const RtcEventRtcpPacketIncoming& original_event,
>+    const LoggedRtcpPacketIncoming& logged_event);
>+
>+bool VerifyLoggedRtcpPacketOutgoing(
>+    const RtcEventRtcpPacketOutgoing& original_event,
>+    const LoggedRtcpPacketOutgoing& logged_event);
>+
>+bool VerifyLoggedStartEvent(int64_t start_time_us,
>+                            const LoggedStartEvent& logged_event);
>+bool VerifyLoggedStopEvent(int64_t stop_time_us,
>+                           const LoggedStopEvent& logged_event);
>+
>+bool VerifyLoggedAudioRecvConfig(
>+    const RtcEventAudioReceiveStreamConfig& original_event,
>+    const LoggedAudioRecvConfig& logged_event);
>+
>+bool VerifyLoggedAudioSendConfig(
>+    const RtcEventAudioSendStreamConfig& original_event,
>+    const LoggedAudioSendConfig& logged_event);
>+
>+bool VerifyLoggedVideoRecvConfig(
>+    const RtcEventVideoReceiveStreamConfig& original_event,
>+    const LoggedVideoRecvConfig& logged_event);
>+
>+bool VerifyLoggedVideoSendConfig(
>+    const RtcEventVideoSendStreamConfig& original_event,
>+    const LoggedVideoSendConfig& logged_event);
>+
>+}  // namespace test
> }  // namespace webrtc
> 
> #endif  // LOGGING_RTC_EVENT_LOG_RTC_EVENT_LOG_UNITTEST_HELPER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_stream_config.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_stream_config.cc
>index b43c6dd6f36..d4d30d00bc5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_stream_config.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_stream_config.cc
>@@ -17,6 +17,8 @@ StreamConfig::StreamConfig() {}
> 
> StreamConfig::~StreamConfig() {}
> 
>+StreamConfig::StreamConfig(const StreamConfig& other) = default;
>+
> bool StreamConfig::operator==(const StreamConfig& other) const {
>   return local_ssrc == other.local_ssrc && remote_ssrc == other.remote_ssrc &&
>          rtx_ssrc == other.rtx_ssrc && rsid == other.rsid &&
>@@ -24,12 +26,16 @@ bool StreamConfig::operator==(const StreamConfig& other) const {
>          rtp_extensions == other.rtp_extensions && codecs == other.codecs;
> }
> 
>+bool StreamConfig::operator!=(const StreamConfig& other) const {
>+  return !(*this == other);
>+}
>+
> StreamConfig::Codec::Codec(const std::string& payload_name,
>                            int payload_type,
>                            int rtx_payload_type)
>-        : payload_name(payload_name),
>-          payload_type(payload_type),
>-          rtx_payload_type(rtx_payload_type) {}
>+    : payload_name(payload_name),
>+      payload_type(payload_type),
>+      rtx_payload_type(rtx_payload_type) {}
> 
> bool StreamConfig::Codec::operator==(const Codec& other) const {
>   return payload_name == other.payload_name &&
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_stream_config.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_stream_config.h
>index 771ba66c6a4..d95c8359ece 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_stream_config.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_stream_config.h
>@@ -23,9 +23,11 @@ namespace rtclog {
> 
> struct StreamConfig {
>   StreamConfig();
>+  StreamConfig(const StreamConfig& other);
>   ~StreamConfig();
> 
>   bool operator==(const StreamConfig& other) const;
>+  bool operator!=(const StreamConfig& other) const;
> 
>   uint32_t local_ssrc = 0;
>   uint32_t remote_ssrc = 0;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/media/BUILD.gn
>index 0dc20b4acb6..3042ac2a865 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/BUILD.gn
>@@ -26,15 +26,6 @@ config("rtc_media_defines_config") {
>   ]
> }
> 
>-config("rtc_media_warnings_config") {
>-  # GN orders flags on a target before flags from configs. The default config
>-  # adds these flags so to cancel them out they need to come from a config and
>-  # cannot be on the target directly.
>-  if (!is_win) {
>-    cflags = [ "-Wno-deprecated-declarations" ]
>-  }
>-}
>-
> rtc_source_set("rtc_h264_profile_id") {
>   visibility = [ "*" ]
>   sources = [
>@@ -49,9 +40,30 @@ rtc_source_set("rtc_h264_profile_id") {
> 
>   deps = [
>     "..:webrtc_common",
>-    "../api:optional",
>     "../rtc_base:rtc_base",
>     "../rtc_base:rtc_base_approved",
>+    "//third_party/abseil-cpp/absl/types:optional",
>+  ]
>+}
>+
>+rtc_source_set("rtc_media_config") {
>+  visibility = [ "*" ]
>+  sources = [
>+    "base/mediaconfig.h",
>+  ]
>+}
>+
>+rtc_source_set("rtc_vp9_profile") {
>+  sources = [
>+    "base/vp9_profile.cc",
>+    "base/vp9_profile.h",
>+  ]
>+
>+  deps = [
>+    "..:webrtc_common",
>+    "../api/video_codecs:video_codecs_api",
>+    "../rtc_base:rtc_base_approved",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
> 
>@@ -60,8 +72,12 @@ rtc_static_library("rtc_media_base") {
>   defines = []
>   libs = []
>   deps = [
>+    "../api:audio_options_api",
>     "../rtc_base:checks",
>+    "../rtc_base:rtc_base_approved",
>+    "../rtc_base:rtc_task_queue",
>     "../rtc_base:sanitizer",
>+    "../rtc_base:sequenced_task_checker",
>     "../rtc_base:stringutils",
>   ]
>   sources = [
>@@ -72,6 +88,7 @@ rtc_static_library("rtc_media_base") {
>     "base/codec.h",
>     "base/cryptoparams.h",
>     "base/device.h",
>+    "base/mediachannel.cc",
>     "base/mediachannel.h",
>     "base/mediaconstants.cc",
>     "base/mediaconstants.h",
>@@ -105,20 +122,22 @@ rtc_static_library("rtc_media_base") {
> 
>   deps += [
>     ":rtc_h264_profile_id",
>+    ":rtc_media_config",
>+    ":rtc_vp9_profile",
>     "..:webrtc_common",
>     "../api:libjingle_peerconnection_api",
>-    "../api:optional",
>-    "../api:video_frame_api",
>-    "../api:video_frame_api_i420",
>     "../api/audio_codecs:audio_codecs_api",
>+    "../api/video:video_frame",
>+    "../api/video:video_frame_i420",
>     "../api/video_codecs:video_codecs_api",
>     "../call:call_interfaces",
>-    "../call:video_stream_api",
>     "../common_video",
>     "../modules/audio_processing:audio_processing_statistics",
>     "../rtc_base:rtc_base",
>     "../rtc_base:rtc_base_approved",
>+    "../rtc_base/third_party/sigslot",
>     "../system_wrappers:field_trial_api",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> 
>   if (!build_with_mozilla) {
>@@ -142,11 +161,13 @@ rtc_static_library("rtc_constants") {
> 
> rtc_static_library("rtc_internal_video_codecs") {
>   visibility = [ "*" ]
>+  allow_poison = [ "software_video_codecs" ]
>   defines = []
>   libs = []
>   deps = [
>     ":rtc_h264_profile_id",
>     "../modules/video_coding:video_codec_interface",
>+    "//third_party/abseil-cpp/absl/memory",
>   ]
>   sources = [
>     "engine/convert_legacy_video_factory.cc",
>@@ -155,43 +176,25 @@ rtc_static_library("rtc_internal_video_codecs") {
>     "engine/internaldecoderfactory.h",
>     "engine/internalencoderfactory.cc",
>     "engine/internalencoderfactory.h",
>+    "engine/multiplexcodecfactory.cc",
>+    "engine/multiplexcodecfactory.h",
>     "engine/scopedvideodecoder.cc",
>     "engine/scopedvideodecoder.h",
>     "engine/scopedvideoencoder.cc",
>     "engine/scopedvideoencoder.h",
>-    "engine/simulcast.cc",
>-    "engine/simulcast.h",
>     "engine/simulcast_encoder_adapter.cc",
>     "engine/simulcast_encoder_adapter.h",
>-    "engine/stereocodecfactory.cc",
>-    "engine/stereocodecfactory.h",
>-    "engine/videodecodersoftwarefallbackwrapper.cc",
>-    "engine/videodecodersoftwarefallbackwrapper.h",
>-    "engine/videoencodersoftwarefallbackwrapper.cc",
>-    "engine/videoencodersoftwarefallbackwrapper.h",
>     "engine/vp8_encoder_simulcast_proxy.cc",
>     "engine/vp8_encoder_simulcast_proxy.h",
>-    "engine/webrtcvideodecoderfactory.cc",
>     "engine/webrtcvideodecoderfactory.h",
>-    "engine/webrtcvideoencoderfactory.cc",
>     "engine/webrtcvideoencoderfactory.h",
>   ]
> 
>-  configs += [ ":rtc_media_warnings_config" ]
>-
>   if (!build_with_chromium && is_clang) {
>     # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>     suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>   }
> 
>-  if (is_win) {
>-    cflags = [
>-      "/wd4245",  # conversion from "int" to "size_t", signed/unsigned mismatch.
>-      "/wd4267",  # conversion from "size_t" to "int", possible loss of data.
>-      "/wd4389",  # signed/unsigned mismatch.
>-    ]
>-  }
>-
>   include_dirs = []
> 
>   public_configs = []
>@@ -199,14 +202,16 @@ rtc_static_library("rtc_internal_video_codecs") {
>     ":rtc_constants",
>     ":rtc_media_base",
>     "..:webrtc_common",
>-    "../api:video_frame_api_i420",
>+    "../api/video:video_bitrate_allocation",
>+    "../api/video:video_frame_i420",
>+    "../api/video_codecs:rtc_software_fallback_wrappers",
>     "../api/video_codecs:video_codecs_api",
>     "../call:call_interfaces",
>     "../call:video_stream_api",
>+    "../modules/video_coding:video_coding_utility",
>     "../modules/video_coding:webrtc_h264",
>-    "../modules/video_coding:webrtc_stereo",
>+    "../modules/video_coding:webrtc_multiplex",
>     "../modules/video_coding:webrtc_vp8",
>-    "../modules/video_coding:webrtc_vp8_helpers",
>     "../modules/video_coding:webrtc_vp9",
>     "../rtc_base:checks",
>     "../rtc_base:rtc_base_approved",
>@@ -220,12 +225,20 @@ rtc_static_library("rtc_internal_video_codecs") {
> 
> rtc_static_library("rtc_audio_video") {
>   visibility = [ "*" ]
>+  allow_poison = [
>+    "audio_codecs",  # TODO(bugs.webrtc.org/8396): Remove.
>+    "software_video_codecs",  # TODO(bugs.webrtc.org/7925): Remove.
>+  ]
>   defines = []
>   libs = []
>   deps = [
>+    "../modules/audio_processing/aec_dump:aec_dump",
>     "../modules/video_coding:video_codec_interface",
>     "../modules/video_coding:video_coding",
>+    "../modules/video_coding:video_coding_utility",
>+    "../rtc_base:audio_format_to_string",
>     "../rtc_base:checks",
>+    "../rtc_base/third_party/base64",
>   ]
> 
>   sources = [
>@@ -236,33 +249,29 @@ rtc_static_library("rtc_audio_video") {
>     "engine/nullwebrtcvideoengine.h",
>     "engine/payload_type_mapper.cc",
>     "engine/payload_type_mapper.h",
>+    "engine/simulcast.cc",
>+    "engine/simulcast.h",
>     "engine/webrtcmediaengine.cc",
>     "engine/webrtcmediaengine.h",
>     "engine/webrtcvideocapturer.cc",
>     "engine/webrtcvideocapturer.h",
>     "engine/webrtcvideocapturerfactory.cc",
>     "engine/webrtcvideocapturerfactory.h",
>+    "engine/webrtcvideodecoderfactory.cc",
>+    "engine/webrtcvideodecoderfactory.h",
>+    "engine/webrtcvideoencoderfactory.cc",
>+    "engine/webrtcvideoencoderfactory.h",
>     "engine/webrtcvideoengine.cc",
>     "engine/webrtcvideoengine.h",
>     "engine/webrtcvoiceengine.cc",
>     "engine/webrtcvoiceengine.h",
>   ]
> 
>-  configs += [ ":rtc_media_warnings_config" ]
>-
>   if (!build_with_chromium && is_clang) {
>     # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>     suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>   }
> 
>-  if (is_win) {
>-    cflags = [
>-      "/wd4245",  # conversion from "int" to "size_t", signed/unsigned mismatch.
>-      "/wd4267",  # conversion from "size_t" to "int", possible loss of data.
>-      "/wd4389",  # signed/unsigned mismatch.
>-    ]
>-  }
>-
>   if (rtc_enable_intelligibility_enhancer) {
>     defines += [ "WEBRTC_INTELLIGIBILITY_ENHANCER=1" ]
>   } else {
>@@ -289,24 +298,27 @@ rtc_static_library("rtc_audio_video") {
>   } else {
>     deps += [ "../modules/audio_processing/aec_dump:null_aec_dump_factory" ]
>   }
>+  if (rtc_use_builtin_sw_codecs) {
>+    deps += [ ":rtc_internal_video_codecs" ]
>+  }
>   deps += [
>     ":rtc_constants",
>-    ":rtc_internal_video_codecs",
>     ":rtc_media_base",
>     "..:webrtc_common",
>     "../api:call_api",
>     "../api:libjingle_peerconnection_api",
>-    "../api:optional",
>     "../api:transport_api",
>-    "../api:video_frame_api",
>-    "../api:video_frame_api_i420",
>     "../api/audio_codecs:audio_codecs_api",
>+    "../api/video:video_frame",
>+    "../api/video:video_frame_i420",
>+    "../api/video_codecs:rtc_software_fallback_wrappers",
>     "../api/video_codecs:video_codecs_api",
>     "../call",
>     "../call:call_interfaces",
>     "../call:video_stream_api",
>     "../common_video:common_video",
>     "../modules/audio_device:audio_device",
>+    "../modules/audio_device:audio_device_impl",
>     "../modules/audio_mixer:audio_mixer_impl",
>     "../modules/audio_processing:audio_processing",
>     "../modules/video_capture:video_capture_module",
>@@ -317,7 +329,7 @@ rtc_static_library("rtc_audio_video") {
>     "../system_wrappers",
>     "../system_wrappers:field_trial_api",
>     "../system_wrappers:metrics_api",
>-    "../voice_engine",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
> 
>@@ -338,21 +350,11 @@ rtc_static_library("rtc_data") {
>     ]
>   }
> 
>-  configs += [ ":rtc_media_warnings_config" ]
>-
>   if (!build_with_chromium && is_clang) {
>     # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>     suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>   }
> 
>-  if (is_win) {
>-    cflags = [
>-      "/wd4245",  # conversion from "int" to "size_t", signed/unsigned mismatch.
>-      "/wd4267",  # conversion from "size_t" to "int", possible loss of data.
>-      "/wd4389",  # signed/unsigned mismatch.
>-    ]
>-  }
>-
>   if (rtc_enable_sctp && rtc_build_usrsctp) {
>     include_dirs = [
>       # TODO(jiayl): move this into the public_configs of
>@@ -370,12 +372,17 @@ rtc_static_library("rtc_data") {
>     "../p2p:rtc_p2p",
>     "../rtc_base:rtc_base",
>     "../rtc_base:rtc_base_approved",
>+    "../rtc_base/third_party/sigslot",
>     "../system_wrappers",
>   ]
> }
> 
> rtc_source_set("rtc_media") {
>   visibility = [ "*" ]
>+  allow_poison = [
>+    "audio_codecs",  # TODO(bugs.webrtc.org/8396): Remove.
>+    "software_video_codecs",  # TODO(bugs.webrtc.org/7925): Remove.
>+  ]
>   deps = [
>     ":rtc_audio_video",
>     ":rtc_data",
>@@ -383,23 +390,15 @@ rtc_source_set("rtc_media") {
> }
> 
> if (rtc_include_tests) {
>-  config("rtc_unittest_main_config") {
>-    # GN orders flags on a target before flags from configs. The default config
>-    # adds -Wall, and this flag have to be after -Wall -- so they need to
>-    # come from a config and can"t be on the target directly.
>-    if (is_clang && is_ios) {
>-      cflags = [ "-Wno-unused-variable" ]
>-    }
>-  }
>-
>   rtc_source_set("rtc_media_tests_utils") {
>     testonly = true
> 
>+    defines = []
>     include_dirs = []
>     deps = [
>       ":rtc_audio_video",
>       "../api:libjingle_peerconnection_api",
>-      "../api:video_frame_api_i420",
>+      "../api/video:video_frame_i420",
>       "../call:video_stream_api",
>       "../common_video:common_video",
>       "../modules/audio_coding:rent_a_codec",
>@@ -409,16 +408,21 @@ if (rtc_include_tests) {
>       "../modules/video_coding:video_coding_utility",
>       "../p2p:rtc_p2p",
>       "../rtc_base:checks",
>+      "../rtc_base:rtc_task_queue",
>       "../rtc_base:stringutils",
>+      "//third_party/abseil-cpp/absl/memory",
>     ]
>     sources = [
>+      "base/fakeframesource.cc",
>+      "base/fakeframesource.h",
>       "base/fakemediaengine.h",
>       "base/fakenetworkinterface.h",
>       "base/fakertp.cc",
>       "base/fakertp.h",
>+      "base/fakevideocapturer.cc",
>       "base/fakevideocapturer.h",
>+      "base/fakevideorenderer.cc",
>       "base/fakevideorenderer.h",
>-      "base/test/mock_mediachannel.h",
>       "base/testutils.cc",
>       "base/testutils.h",
>       "engine/fakewebrtccall.cc",
>@@ -426,52 +430,40 @@ if (rtc_include_tests) {
>       "engine/fakewebrtcdeviceinfo.h",
>       "engine/fakewebrtcvcmfactory.h",
>       "engine/fakewebrtcvideocapturemodule.h",
>+      "engine/fakewebrtcvideoengine.cc",
>       "engine/fakewebrtcvideoengine.h",
>     ]
> 
>-    configs += [ ":rtc_unittest_main_config" ]
>-
>     if (!build_with_chromium && is_clang) {
>       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>     }
> 
>+    if (rtc_use_h264) {
>+      defines += [ "WEBRTC_USE_H264" ]
>+    }
>+
>     deps += [
>       ":rtc_internal_video_codecs",
>       ":rtc_media",
>       ":rtc_media_base",
>       "..:webrtc_common",
>       "../api:call_api",
>-      "../api:video_frame_api",
>+      "../api/video:video_bitrate_allocation",
>+      "../api/video:video_frame",
>       "../api/video_codecs:video_codecs_api",
>       "../call:call_interfaces",
>+      "../call:mock_rtp_interfaces",
>       "../rtc_base:rtc_base",
>       "../rtc_base:rtc_base_approved",
>       "../rtc_base:rtc_base_tests_utils",
>+      "../rtc_base:rtc_task_queue_for_test",
>+      "../rtc_base/third_party/sigslot",
>       "../test:test_support",
>-      "//testing/gmock",
>       "//testing/gtest",
>     ]
>   }
> 
>-  config("rtc_media_unittests_config") {
>-    # GN orders flags on a target before flags from configs. The default config
>-    # adds -Wall, and this flag have to be after -Wall -- so they need to
>-    # come from a config and can"t be on the target directly.
>-    # TODO(kjellander): Make the code compile without disabling these flags.
>-    # See https://bugs.webrtc.org/3307.
>-    if (is_clang && is_win) {
>-      cflags = [
>-        # See https://bugs.chromium.org/p/webrtc/issues/detail?id=6266
>-        # for -Wno-sign-compare
>-        "-Wno-sign-compare",
>-      ]
>-    }
>-    if (!is_win) {
>-      cflags = [ "-Wno-sign-compare" ]
>-    }
>-  }
>-
>   rtc_media_unittests_resources = [
>     "../resources/media/captured-320x240-2s-48.frames",
>     "../resources/media/faces.1280x720_P420.yuv",
>@@ -500,14 +492,19 @@ if (rtc_include_tests) {
>       ":rtc_audio_video",
>       ":rtc_constants",
>       ":rtc_data",
>-      "../api:video_frame_api_i420",
>+      "../api/video:video_frame_i420",
>       "../modules/audio_processing:mocks",
>+      "../modules/rtp_rtcp",
>       "../modules/video_coding:video_codec_interface",
>+      "../modules/video_coding:webrtc_vp8",
>       "../pc:rtc_pc",
>       "../pc:rtc_pc_base",
>       "../rtc_base:checks",
>+      "../rtc_base:rtc_task_queue",
>       "../rtc_base:stringutils",
>       "../test:field_trial",
>+      "../test:test_common",
>+      "//third_party/abseil-cpp/absl/memory",
>     ]
>     sources = [
>       "base/codec_unittest.cc",
>@@ -519,16 +516,13 @@ if (rtc_include_tests) {
>       "base/videobroadcaster_unittest.cc",
>       "base/videocapturer_unittest.cc",
>       "base/videocommon_unittest.cc",
>-      "base/videoengine_unittest.h",
>       "engine/apm_helpers_unittest.cc",
>       "engine/internaldecoderfactory_unittest.cc",
>+      "engine/multiplexcodecfactory_unittest.cc",
>       "engine/nullwebrtcvideoengine_unittest.cc",
>       "engine/payload_type_mapper_unittest.cc",
>       "engine/simulcast_encoder_adapter_unittest.cc",
>       "engine/simulcast_unittest.cc",
>-      "engine/stereocodecfactory_unittest.cc",
>-      "engine/videodecodersoftwarefallbackwrapper_unittest.cc",
>-      "engine/videoencodersoftwarefallbackwrapper_unittest.cc",
>       "engine/vp8_encoder_simulcast_proxy_unittest.cc",
>       "engine/webrtcmediaengine_unittest.cc",
>       "engine/webrtcvideocapturer_unittest.cc",
>@@ -546,8 +540,6 @@ if (rtc_include_tests) {
>       sources += [ "sctp/sctptransport_unittest.cc" ]
>     }
> 
>-    configs += [ ":rtc_media_unittests_config" ]
>-
>     if (rtc_use_h264) {
>       defines += [ "WEBRTC_USE_H264" ]
>     }
>@@ -558,18 +550,8 @@ if (rtc_include_tests) {
>       defines += [ "WEBRTC_OPUS_SUPPORT_120MS_PTIME=0" ]
>     }
> 
>-    if (is_win) {
>-      cflags = [
>-        "/wd4245",  # conversion from int to size_t, signed/unsigned mismatch.
>-        "/wd4373",  # virtual function override.
>-        "/wd4389",  # signed/unsigned mismatch.
>-      ]
>-    }
>-
>     if (!build_with_chromium && is_clang) {
>       suppressed_configs += [
>-        "//build/config/clang:extra_warnings",
>-
>         # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>         "//build/config/clang:find_bad_constructs",
>       ]
>@@ -591,22 +573,26 @@ if (rtc_include_tests) {
>       ":rtc_media",
>       ":rtc_media_base",
>       ":rtc_media_tests_utils",
>+      ":rtc_vp9_profile",
>+      "../api:create_simulcast_test_fixture_api",
>       "../api:libjingle_peerconnection_api",
>       "../api:mock_video_codec_factory",
>-      "../api:video_frame_api",
>+      "../api:simulcast_test_fixture_api",
>       "../api/audio_codecs:builtin_audio_decoder_factory",
>       "../api/audio_codecs:builtin_audio_encoder_factory",
>+      "../api/video:video_bitrate_allocation",
>+      "../api/video:video_frame",
>+      "../api/video_codecs:builtin_video_decoder_factory",
>+      "../api/video_codecs:builtin_video_encoder_factory",
>       "../api/video_codecs:video_codecs_api",
>       "../audio",
>       "../call:call_interfaces",
>       "../common_video:common_video",
>       "../logging:rtc_event_log_api",
>+      "../logging:rtc_event_log_impl_base",
>       "../modules/audio_device:mock_audio_device",
>       "../modules/audio_processing:audio_processing",
>-      "../modules/video_coding:simulcast_test_utility",
>-      "../modules/video_coding:video_coding_utility",
>-      "../modules/video_coding:webrtc_vp8",
>-      "../modules/video_coding:webrtc_vp8_helpers",
>+      "../modules/video_coding:simulcast_test_fixture_impl",
>       "../p2p:p2p_test_utils",
>       "../rtc_base:rtc_base",
>       "../rtc_base:rtc_base_approved",
>@@ -617,7 +603,6 @@ if (rtc_include_tests) {
>       "../test:audio_codec_mocks",
>       "../test:test_support",
>       "../test:video_test_common",
>-      "../voice_engine:voice_engine",
>     ]
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/DEPS b/Source/ThirdParty/libwebrtc/Source/webrtc/media/DEPS
>index 7a266a23faf..ab54b440369 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/DEPS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/DEPS
>@@ -10,11 +10,11 @@ include_rules = [
>   "+modules/rtp_rtcp",
>   "+modules/video_capture",
>   "+modules/video_coding",
>+  "+modules/video_coding/utility",
>   "+p2p",
>   "+pc",
>   "+sound",
>   "+system_wrappers",
>-  "+voice_engine",
>   "+usrsctplib",
>   "+third_party/libyuv",
> ]
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/media/OWNERS
>index bf977f493e7..2688b07b00d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/OWNERS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/OWNERS
>@@ -3,7 +3,6 @@ mflodman@webrtc.org
> perkj@webrtc.org
> pthatcher@webrtc.org
> solenberg@webrtc.org
>-deadbeef@webrtc.org
> 
> # These are for the common case of adding or renaming files. If you're doing
> # structural changes, please get a review from a reviewer in this file.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/adaptedvideotracksource.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/adaptedvideotracksource.cc
>index 5a7168bfb86..11952770a46 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/adaptedvideotracksource.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/adaptedvideotracksource.cc
>@@ -22,6 +22,7 @@ AdaptedVideoTrackSource::AdaptedVideoTrackSource(int required_alignment)
>     : video_adapter_(required_alignment) {
>   thread_checker_.DetachFromThread();
> }
>+AdaptedVideoTrackSource::~AdaptedVideoTrackSource() = default;
> 
> bool AdaptedVideoTrackSource::GetStats(Stats* stats) {
>   rtc::CritScope lock(&stats_crit_);
>@@ -103,8 +104,8 @@ bool AdaptedVideoTrackSource::AdaptFrame(int width,
>   }
> 
>   if (!video_adapter_.AdaptFrameResolution(
>-          width, height, time_us * rtc::kNumNanosecsPerMicrosec,
>-          crop_width, crop_height, out_width, out_height)) {
>+          width, height, time_us * rtc::kNumNanosecsPerMicrosec, crop_width,
>+          crop_height, out_width, out_height)) {
>     broadcaster_.OnDiscardedFrame();
>     // VideoAdapter dropped the frame.
>     return false;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/adaptedvideotracksource.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/adaptedvideotracksource.h
>index 0db381f5fbc..7e7ba7eeed1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/adaptedvideotracksource.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/adaptedvideotracksource.h
>@@ -26,6 +26,7 @@ class AdaptedVideoTrackSource
>     : public webrtc::Notifier<webrtc::VideoTrackSourceInterface> {
>  public:
>   AdaptedVideoTrackSource();
>+  ~AdaptedVideoTrackSource() override;
> 
>  protected:
>   // Allows derived classes to initialize |video_adapter_| with a custom
>@@ -74,7 +75,7 @@ class AdaptedVideoTrackSource
>   cricket::VideoAdapter video_adapter_;
> 
>   rtc::CriticalSection stats_crit_;
>-  rtc::Optional<Stats> stats_ RTC_GUARDED_BY(stats_crit_);
>+  absl::optional<Stats> stats_ RTC_GUARDED_BY(stats_crit_);
> 
>   VideoBroadcaster broadcaster_;
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/codec.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/codec.cc
>index 98e52d68489..ee7eda343b3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/codec.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/codec.cc
>@@ -11,21 +11,23 @@
> #include "media/base/codec.h"
> 
> #include <algorithm>
>-#include <sstream>
> 
> #include "media/base/h264_profile_level_id.h"
>+#include "media/base/vp9_profile.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/stringencode.h"
>+#include "rtc_base/strings/string_builder.h"
> #include "rtc_base/stringutils.h"
> 
> namespace cricket {
> 
> FeedbackParams::FeedbackParams() = default;
>+FeedbackParams::~FeedbackParams() = default;
> 
> bool FeedbackParam::operator==(const FeedbackParam& other) const {
>   return _stricmp(other.id().c_str(), id().c_str()) == 0 &&
>-      _stricmp(other.param().c_str(), param().c_str()) == 0;
>+         _stricmp(other.param().c_str(), param().c_str()) == 0;
> }
> 
> bool FeedbackParams::operator==(const FeedbackParams& other) const {
>@@ -117,7 +119,7 @@ void Codec::SetParam(const std::string& name, const std::string& value) {
>   params[name] = value;
> }
> 
>-void Codec::SetParam(const std::string& name, int value)  {
>+void Codec::SetParam(const std::string& name, int value) {
>   params[name] = rtc::ToString(value);
> }
> 
>@@ -142,6 +144,7 @@ webrtc::RtpCodecParameters Codec::ToCodecParameters() const {
>   codec_params.payload_type = id;
>   codec_params.name = name;
>   codec_params.clock_rate = clockrate;
>+  codec_params.parameters.insert(params.begin(), params.end());
>   return codec_params;
> }
> 
>@@ -152,8 +155,7 @@ AudioCodec::AudioCodec(int id,
>                        size_t channels)
>     : Codec(id, name, clockrate), bitrate(bitrate), channels(channels) {}
> 
>-AudioCodec::AudioCodec() : Codec(), bitrate(0), channels(0) {
>-}
>+AudioCodec::AudioCodec() : Codec(), bitrate(0), channels(0) {}
> 
> AudioCodec::AudioCodec(const AudioCodec& c) = default;
> AudioCodec::AudioCodec(AudioCodec&& c) = default;
>@@ -175,17 +177,18 @@ bool AudioCodec::Matches(const AudioCodec& codec) const {
>   // Preference is ignored.
>   // TODO(juberti): Treat a zero clockrate as 8000Hz, the RTP default clockrate.
>   return Codec::Matches(codec) &&
>-      ((codec.clockrate == 0 /*&& clockrate == 8000*/) ||
>+         ((codec.clockrate == 0 /*&& clockrate == 8000*/) ||
>           clockrate == codec.clockrate) &&
>-      (codec.bitrate == 0 || bitrate <= 0 || bitrate == codec.bitrate) &&
>-      ((codec.channels < 2 && channels < 2) || channels == codec.channels);
>+         (codec.bitrate == 0 || bitrate <= 0 || bitrate == codec.bitrate) &&
>+         ((codec.channels < 2 && channels < 2) || channels == codec.channels);
> }
> 
> std::string AudioCodec::ToString() const {
>-  std::ostringstream os;
>-  os << "AudioCodec[" << id << ":" << name << ":" << clockrate << ":" << bitrate
>+  char buf[256];
>+  rtc::SimpleStringBuilder sb(buf);
>+  sb << "AudioCodec[" << id << ":" << name << ":" << clockrate << ":" << bitrate
>      << ":" << channels << "]";
>-  return os.str();
>+  return sb.str();
> }
> 
> webrtc::RtpCodecParameters AudioCodec::ToCodecParameters() const {
>@@ -196,9 +199,10 @@ webrtc::RtpCodecParameters AudioCodec::ToCodecParameters() const {
> }
> 
> std::string VideoCodec::ToString() const {
>-  std::ostringstream os;
>-  os << "VideoCodec[" << id << ":" << name << "]";
>-  return os.str();
>+  char buf[256];
>+  rtc::SimpleStringBuilder sb(buf);
>+  sb << "VideoCodec[" << id << ":" << name << "]";
>+  return sb.str();
> }
> 
> webrtc::RtpCodecParameters VideoCodec::ToCodecParameters() const {
>@@ -244,11 +248,33 @@ bool VideoCodec::operator==(const VideoCodec& c) const {
>   return Codec::operator==(c);
> }
> 
>+static bool IsSameH264PacketizationMode(const CodecParameterMap& ours,
>+                                        const CodecParameterMap& theirs) {
>+  // If packetization-mode is not present, default to "0".
>+  // https://tools.ietf.org/html/rfc6184#section-6.2
>+  std::string our_packetization_mode = "0";
>+  std::string their_packetization_mode = "0";
>+  auto ours_it = ours.find(kH264FmtpPacketizationMode);
>+  if (ours_it != ours.end()) {
>+    our_packetization_mode = ours_it->second;
>+  }
>+  auto theirs_it = theirs.find(kH264FmtpPacketizationMode);
>+  if (theirs_it != theirs.end()) {
>+    their_packetization_mode = theirs_it->second;
>+  }
>+  return our_packetization_mode == their_packetization_mode;
>+}
>+
> bool VideoCodec::Matches(const VideoCodec& other) const {
>   if (!Codec::Matches(other))
>     return false;
>   if (CodecNamesEq(name.c_str(), kH264CodecName))
>-    return webrtc::H264::IsSameH264Profile(params, other.params);
>+    return webrtc::H264::IsSameH264Profile(params, other.params) &&
>+           IsSameH264PacketizationMode(params, other.params);
>+#if !defined(RTC_DISABLE_VP9)
>+  if (CodecNamesEq(name.c_str(), kVp9CodecName))
>+    return webrtc::IsSameVP9Profile(params, other.params);
>+#endif
>   return true;
> }
> 
>@@ -312,9 +338,10 @@ DataCodec& DataCodec::operator=(const DataCodec& c) = default;
> DataCodec& DataCodec::operator=(DataCodec&& c) = default;
> 
> std::string DataCodec::ToString() const {
>-  std::ostringstream os;
>-  os << "DataCodec[" << id << ":" << name << "]";
>-  return os.str();
>+  char buf[256];
>+  rtc::SimpleStringBuilder sb(buf);
>+  sb << "DataCodec[" << id << ":" << name << "]";
>+  return sb.str();
> }
> 
> bool HasNack(const Codec& codec) {
>@@ -327,6 +354,11 @@ bool HasRemb(const Codec& codec) {
>       FeedbackParam(kRtcpFbParamRemb, kParamValueEmpty));
> }
> 
>+bool HasRrtr(const Codec& codec) {
>+  return codec.HasFeedbackParam(
>+      FeedbackParam(kRtcpFbParamRrtr, kParamValueEmpty));
>+}
>+
> bool HasTransportCc(const Codec& codec) {
>   return codec.HasFeedbackParam(
>       FeedbackParam(kRtcpFbParamTransportCc, kParamValueEmpty));
>@@ -359,9 +391,14 @@ bool IsSameCodec(const std::string& name1,
>   // If different names (case insensitive), then not same formats.
>   if (!CodecNamesEq(name1, name2))
>     return false;
>-  // For every format besides H264, comparing names is enough.
>-  return !CodecNamesEq(name1.c_str(), kH264CodecName) ||
>-         webrtc::H264::IsSameH264Profile(params1, params2);
>+  // For every format besides H264 and VP9, comparing names is enough.
>+  if (CodecNamesEq(name1.c_str(), kH264CodecName))
>+    return webrtc::H264::IsSameH264Profile(params1, params2);
>+#if !defined(RTC_DISABLE_VP9)
>+  if (CodecNamesEq(name1.c_str(), kVp9CodecName))
>+    return webrtc::IsSameVP9Profile(params1, params2);
>+#endif
>+  return true;
> }
> 
> }  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/codec.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/codec.h
>index 6a2dcf4529e..db133f6e27f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/codec.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/codec.h
>@@ -29,26 +29,24 @@ class FeedbackParam {
>  public:
>   FeedbackParam() = default;
>   FeedbackParam(const std::string& id, const std::string& param)
>-      : id_(id),
>-        param_(param) {
>-  }
>+      : id_(id), param_(param) {}
>   explicit FeedbackParam(const std::string& id)
>-      : id_(id),
>-        param_(kParamValueEmpty) {
>-  }
>+      : id_(id), param_(kParamValueEmpty) {}
>+
>   bool operator==(const FeedbackParam& other) const;
> 
>   const std::string& id() const { return id_; }
>   const std::string& param() const { return param_; }
> 
>  private:
>-  std::string id_;  // e.g. "nack", "ccm"
>+  std::string id_;     // e.g. "nack", "ccm"
>   std::string param_;  // e.g. "", "rpsi", "fir"
> };
> 
> class FeedbackParams {
>  public:
>   FeedbackParams();
>+  ~FeedbackParams();
>   bool operator==(const FeedbackParams& other) const;
> 
>   bool Has(const FeedbackParam& param) const;
>@@ -57,6 +55,7 @@ class FeedbackParams {
>   void Intersect(const FeedbackParams& from);
> 
>   const std::vector<FeedbackParam>& params() const { return params_; }
>+
>  private:
>   bool HasDuplicateEntries() const;
> 
>@@ -100,9 +99,7 @@ struct Codec {
> 
>   bool operator==(const Codec& c) const;
> 
>-  bool operator!=(const Codec& c) const {
>-    return !(*this == c);
>-  }
>+  bool operator!=(const Codec& c) const { return !(*this == c); }
> 
>  protected:
>   // A Codec can't be created without a subclass.
>@@ -142,33 +139,9 @@ struct AudioCodec : public Codec {
> 
>   bool operator==(const AudioCodec& c) const;
> 
>-  bool operator!=(const AudioCodec& c) const {
>-    return !(*this == c);
>-  }
>+  bool operator!=(const AudioCodec& c) const { return !(*this == c); }
> };
> 
>-inline std::ostream& operator<<(std::ostream& os, const AudioCodec& ac) {
>-  os << "{id: " << ac.id;
>-  os << ", name: " << ac.name;
>-  os << ", clockrate: " << ac.clockrate;
>-  os << ", bitrate: " << ac.bitrate;
>-  os << ", channels: " << ac.channels;
>-  os << ", params: {";
>-  const char* sep = "";
>-  for (const auto& kv : ac.params) {
>-    os << sep << kv.first << ": " << kv.second;
>-    sep = ", ";
>-  }
>-  os << "}, feedback_params: {";
>-  sep = "";
>-  for (const FeedbackParam& fp : ac.feedback_params.params()) {
>-    os << sep << fp.id() << ": " << fp.param();
>-    sep = ", ";
>-  }
>-  os << "}}";
>-  return os;
>-}
>-
> struct VideoCodec : public Codec {
>   // Creates a codec with the given parameters.
>   VideoCodec(int id, const std::string& name);
>@@ -195,9 +168,7 @@ struct VideoCodec : public Codec {
> 
>   bool operator==(const VideoCodec& c) const;
> 
>-  bool operator!=(const VideoCodec& c) const {
>-    return !(*this == c);
>-  }
>+  bool operator!=(const VideoCodec& c) const { return !(*this == c); }
> 
>   static VideoCodec CreateRtxCodec(int rtx_payload_type,
>                                    int associated_payload_type);
>@@ -248,6 +219,7 @@ bool CodecNamesEq(const std::string& name1, const std::string& name2);
> bool CodecNamesEq(const char* name1, const char* name2);
> bool HasNack(const Codec& codec);
> bool HasRemb(const Codec& codec);
>+bool HasRrtr(const Codec& codec);
> bool HasTransportCc(const Codec& codec);
> // Returns the first codec in |supported_codecs| that matches |codec|, or
> // nullptr if no codec matches.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/codec_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/codec_unittest.cc
>index 03d8684c64a..54396a92447 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/codec_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/codec_unittest.cc
>@@ -9,6 +9,8 @@
>  */
> 
> #include "media/base/codec.h"
>+
>+#include "media/base/vp9_profile.h"
> #include "rtc_base/gunit.h"
> 
> using cricket::AudioCodec;
>@@ -186,6 +188,75 @@ TEST(CodecTest, TestVideoCodecMatches) {
>   EXPECT_FALSE(c1.Matches(VideoCodec(95, "V")));
> }
> 
>+// VP9 codecs compare profile information.
>+TEST(CodecTest, TestVP9CodecMatches) {
>+  const char kProfile0[] = "0";
>+  const char kProfile2[] = "2";
>+
>+  VideoCodec c_no_profile(95, cricket::kVp9CodecName);
>+  VideoCodec c_profile0(95, cricket::kVp9CodecName);
>+  c_profile0.params[webrtc::kVP9FmtpProfileId] = kProfile0;
>+
>+  EXPECT_TRUE(c_profile0.Matches(c_no_profile));
>+
>+  {
>+    VideoCodec c_profile0_eq(95, cricket::kVp9CodecName);
>+    c_profile0_eq.params[webrtc::kVP9FmtpProfileId] = kProfile0;
>+    EXPECT_TRUE(c_profile0.Matches(c_profile0_eq));
>+  }
>+
>+  {
>+    VideoCodec c_profile2(95, cricket::kVp9CodecName);
>+    c_profile2.params[webrtc::kVP9FmtpProfileId] = kProfile2;
>+    EXPECT_FALSE(c_profile0.Matches(c_profile2));
>+    EXPECT_FALSE(c_no_profile.Matches(c_profile2));
>+  }
>+
>+  {
>+    VideoCodec c_no_profile_eq(95, cricket::kVp9CodecName);
>+    EXPECT_TRUE(c_no_profile.Matches(c_no_profile_eq));
>+  }
>+}
>+
>+// Matching H264 codecs also need to have matching profile-level-id and
>+// packetization-mode.
>+TEST(CodecTest, TestH264CodecMatches) {
>+  const char kProfileLevelId1[] = "42e01f";
>+  const char kProfileLevelId2[] = "42a01e";
>+
>+  VideoCodec pli_1_pm_0(95, "H264");
>+  pli_1_pm_0.params[cricket::kH264FmtpProfileLevelId] = kProfileLevelId1;
>+  pli_1_pm_0.params[cricket::kH264FmtpPacketizationMode] = "0";
>+
>+  {
>+    VideoCodec pli_1_pm_blank(95, "H264");
>+    pli_1_pm_blank.params[cricket::kH264FmtpProfileLevelId] = kProfileLevelId1;
>+    pli_1_pm_blank.params.erase(
>+        pli_1_pm_blank.params.find(cricket::kH264FmtpPacketizationMode));
>+
>+    // Matches since if packetization-mode is not specified it defaults to "0".
>+    EXPECT_TRUE(pli_1_pm_0.Matches(pli_1_pm_blank));
>+  }
>+
>+  {
>+    VideoCodec pli_1_pm_1(95, "H264");
>+    pli_1_pm_1.params[cricket::kH264FmtpProfileLevelId] = kProfileLevelId1;
>+    pli_1_pm_1.params[cricket::kH264FmtpPacketizationMode] = "1";
>+
>+    // Does not match since packetization-mode is different.
>+    EXPECT_FALSE(pli_1_pm_0.Matches(pli_1_pm_1));
>+  }
>+
>+  {
>+    VideoCodec pli_2_pm_0(95, "H264");
>+    pli_2_pm_0.params[cricket::kH264FmtpProfileLevelId] = kProfileLevelId2;
>+    pli_2_pm_0.params[cricket::kH264FmtpPacketizationMode] = "0";
>+
>+    // Does not match since profile-level-id is different.
>+    EXPECT_FALSE(pli_1_pm_0.Matches(pli_2_pm_0));
>+  }
>+}
>+
> TEST(CodecTest, TestDataCodecMatches) {
>   // Test a codec with a static payload type.
>   DataCodec c0(95, "D");
>@@ -308,19 +379,27 @@ TEST(CodecTest, TestValidateCodecFormat) {
> }
> 
> TEST(CodecTest, TestToCodecParameters) {
>-  const VideoCodec v(96, "V");
>+  VideoCodec v(96, "V");
>+  v.SetParam("p1", "v1");
>   webrtc::RtpCodecParameters codec_params_1 = v.ToCodecParameters();
>   EXPECT_EQ(96, codec_params_1.payload_type);
>   EXPECT_EQ(cricket::MEDIA_TYPE_VIDEO, codec_params_1.kind);
>   EXPECT_EQ("V", codec_params_1.name);
>   EXPECT_EQ(cricket::kVideoCodecClockrate, codec_params_1.clock_rate);
>-  EXPECT_EQ(rtc::nullopt, codec_params_1.num_channels);
>+  EXPECT_EQ(absl::nullopt, codec_params_1.num_channels);
>+  ASSERT_EQ(1u, codec_params_1.parameters.size());
>+  EXPECT_EQ("p1", codec_params_1.parameters.begin()->first);
>+  EXPECT_EQ("v1", codec_params_1.parameters.begin()->second);
> 
>-  const AudioCodec a(97, "A", 44100, 20000, 2);
>+  AudioCodec a(97, "A", 44100, 20000, 2);
>+  a.SetParam("p1", "a1");
>   webrtc::RtpCodecParameters codec_params_2 = a.ToCodecParameters();
>   EXPECT_EQ(97, codec_params_2.payload_type);
>   EXPECT_EQ(cricket::MEDIA_TYPE_AUDIO, codec_params_2.kind);
>   EXPECT_EQ("A", codec_params_2.name);
>   EXPECT_EQ(44100, codec_params_2.clock_rate);
>   EXPECT_EQ(2, codec_params_2.num_channels);
>+  ASSERT_EQ(1u, codec_params_2.parameters.size());
>+  EXPECT_EQ("p1", codec_params_2.parameters.begin()->first);
>+  EXPECT_EQ("a1", codec_params_2.parameters.begin()->second);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/device.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/device.h
>index f47293823f5..7910c974008 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/device.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/device.h
>@@ -20,12 +20,8 @@ namespace cricket {
> // Used to represent an audio or video capture or render device.
> struct Device {
>   Device() {}
>-  Device(const std::string& name, int id)
>-      : name(name),
>-        id(rtc::ToString(id)) {
>-  }
>-  Device(const std::string& name, const std::string& id)
>-      : name(name), id(id) {}
>+  Device(const std::string& name, int id) : name(name), id(rtc::ToString(id)) {}
>+  Device(const std::string& name, const std::string& id) : name(name), id(id) {}
> 
>   std::string name;
>   std::string id;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/fakeframesource.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/fakeframesource.cc
>new file mode 100644
>index 00000000000..ffc3eee773f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/fakeframesource.cc
>@@ -0,0 +1,71 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "media/base/fakeframesource.h"
>+
>+#include "api/video/i420_buffer.h"
>+#include "rtc_base/checks.h"
>+
>+namespace cricket {
>+
>+FakeFrameSource::FakeFrameSource(int width, int height, int interval_us)
>+    : width_(width), height_(height), interval_us_(interval_us) {
>+  RTC_CHECK_GT(width_, 0);
>+  RTC_CHECK_GT(height_, 0);
>+  RTC_CHECK_GT(interval_us_, 0);
>+}
>+
>+webrtc::VideoRotation FakeFrameSource::GetRotation() const {
>+  return rotation_;
>+}
>+
>+void FakeFrameSource::SetRotation(webrtc::VideoRotation rotation) {
>+  rotation_ = rotation;
>+}
>+
>+webrtc::VideoFrame FakeFrameSource::GetFrameRotationApplied() {
>+  switch (rotation_) {
>+    case webrtc::kVideoRotation_0:
>+    case webrtc::kVideoRotation_180:
>+      return GetFrame(width_, height_, webrtc::kVideoRotation_0, interval_us_);
>+    case webrtc::kVideoRotation_90:
>+    case webrtc::kVideoRotation_270:
>+      return GetFrame(height_, width_, webrtc::kVideoRotation_0, interval_us_);
>+  }
>+  RTC_NOTREACHED() << "Invalid rotation value: " << static_cast<int>(rotation_);
>+  // Without this return, the Windows Visual Studio compiler complains
>+  // "not all control paths return a value".
>+  return GetFrame();
>+}
>+
>+webrtc::VideoFrame FakeFrameSource::GetFrame() {
>+  return GetFrame(width_, height_, rotation_, interval_us_);
>+}
>+
>+webrtc::VideoFrame FakeFrameSource::GetFrame(int width,
>+                                             int height,
>+                                             webrtc::VideoRotation rotation,
>+                                             int interval_us) {
>+  RTC_CHECK_GT(width, 0);
>+  RTC_CHECK_GT(height, 0);
>+  RTC_CHECK_GT(interval_us, 0);
>+
>+  rtc::scoped_refptr<webrtc::I420Buffer> buffer(
>+      webrtc::I420Buffer::Create(width, height));
>+
>+  buffer->InitializeData();
>+  webrtc::VideoFrame frame =
>+      webrtc::VideoFrame(buffer, rotation, next_timestamp_us_);
>+
>+  next_timestamp_us_ += interval_us;
>+  return frame;
>+}
>+
>+}  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/fakeframesource.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/fakeframesource.h
>new file mode 100644
>index 00000000000..db440457c03
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/fakeframesource.h
>@@ -0,0 +1,47 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MEDIA_BASE_FAKEFRAMESOURCE_H_
>+#define MEDIA_BASE_FAKEFRAMESOURCE_H_
>+
>+#include "api/video/video_frame.h"
>+
>+#include "rtc_base/timeutils.h"
>+
>+namespace cricket {
>+
>+class FakeFrameSource {
>+ public:
>+  FakeFrameSource(int width, int height, int interval_us);
>+
>+  webrtc::VideoRotation GetRotation() const;
>+  void SetRotation(webrtc::VideoRotation rotation);
>+
>+  webrtc::VideoFrame GetFrame();
>+  webrtc::VideoFrame GetFrameRotationApplied();
>+
>+  // Override configuration.
>+  webrtc::VideoFrame GetFrame(int width,
>+                              int height,
>+                              webrtc::VideoRotation rotation,
>+                              int interval_us);
>+
>+ private:
>+  const int width_;
>+  const int height_;
>+  const int interval_us_;
>+
>+  webrtc::VideoRotation rotation_ = webrtc::kVideoRotation_0;
>+  int64_t next_timestamp_us_ = rtc::kNumMicrosecsPerMillisec;
>+};
>+
>+}  // namespace cricket
>+
>+#endif  // MEDIA_BASE_FAKEFRAMESOURCE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/fakemediaengine.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/fakemediaengine.h
>index 609b62f910a..5d381f7b41a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/fakemediaengine.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/fakemediaengine.h
>@@ -20,6 +20,7 @@
> #include <utility>
> #include <vector>
> 
>+#include "absl/memory/memory.h"
> #include "api/call/audio_sink.h"
> #include "media/base/audiosource.h"
> #include "media/base/mediaengine.h"
>@@ -30,7 +31,6 @@
> #include "rtc_base/checks.h"
> #include "rtc_base/copyonwritebuffer.h"
> #include "rtc_base/networkroute.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/stringutils.h"
> 
> using webrtc::RtpExtension;
>@@ -42,7 +42,8 @@ class FakeVideoEngine;
> class FakeVoiceEngine;
> 
> // A common helper class that handles sending and receiving RTP/RTCP packets.
>-template <class Base> class RtpHelper : public Base {
>+template <class Base>
>+class RtpHelper : public Base {
>  public:
>   RtpHelper()
>       : sending_(false),
>@@ -110,7 +111,7 @@ template <class Base> class RtpHelper : public Base {
>     }
>     send_streams_.push_back(sp);
>     rtp_send_parameters_[sp.first_ssrc()] =
>-        CreateRtpParametersWithOneEncoding();
>+        CreateRtpParametersWithEncodings(sp);
>     return true;
>   }
>   virtual bool RemoveSendStream(uint32_t ssrc) {
>@@ -145,16 +146,17 @@ template <class Base> class RtpHelper : public Base {
>     }
>     return webrtc::RtpParameters();
>   }
>-  virtual bool SetRtpSendParameters(uint32_t ssrc,
>-                                    const webrtc::RtpParameters& parameters) {
>+  virtual webrtc::RTCError SetRtpSendParameters(
>+      uint32_t ssrc,
>+      const webrtc::RtpParameters& parameters) {
>     auto parameters_iterator = rtp_send_parameters_.find(ssrc);
>     if (parameters_iterator != rtp_send_parameters_.end()) {
>       parameters_iterator->second = parameters;
>-      return true;
>+      return webrtc::RTCError::OK();
>     }
>     // Replicate the behavior of the real media channel: return false
>     // when setting parameters for unknown SSRCs.
>-    return false;
>+    return webrtc::RTCError(webrtc::RTCErrorType::INTERNAL_ERROR);
>   }
> 
>   virtual webrtc::RtpParameters GetRtpReceiveParameters(uint32_t ssrc) const {
>@@ -216,9 +218,7 @@ template <class Base> class RtpHelper : public Base {
>   const RtcpParameters& send_rtcp_parameters() { return send_rtcp_parameters_; }
>   const RtcpParameters& recv_rtcp_parameters() { return recv_rtcp_parameters_; }
> 
>-  bool ready_to_send() const {
>-    return ready_to_send_;
>-  }
>+  bool ready_to_send() const { return ready_to_send_; }
> 
>   int transport_overhead_per_packet() const {
>     return transport_overhead_per_packet_;
>@@ -269,9 +269,7 @@ template <class Base> class RtpHelper : public Base {
>                               const rtc::PacketTime& packet_time) {
>     rtcp_packets_.push_back(std::string(packet->data<char>(), packet->size()));
>   }
>-  virtual void OnReadyToSend(bool ready) {
>-    ready_to_send_ = ready;
>-  }
>+  virtual void OnReadyToSend(bool ready) { ready_to_send_ = ready; }
> 
>   virtual void OnNetworkRouteChanged(const std::string& transport_name,
>                                      const rtc::NetworkRoute& network_route) {
>@@ -310,9 +308,7 @@ class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> {
>  public:
>   struct DtmfInfo {
>     DtmfInfo(uint32_t ssrc, int event_code, int duration)
>-        : ssrc(ssrc),
>-          event_code(event_code),
>-          duration(duration) {}
>+        : ssrc(ssrc), event_code(event_code), duration(duration) {}
>     uint32_t ssrc;
>     int event_code;
>     int duration;
>@@ -381,9 +377,6 @@ class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> {
>     return true;
>   }
> 
>-  virtual bool GetActiveStreams(StreamList* streams) { return true; }
>-  virtual int GetOutputLevel() { return 0; }
>-
>   virtual bool CanInsertDtmf() {
>     for (std::vector<AudioCodec>::const_iterator it = send_codecs_.begin();
>          it != send_codecs_.end(); ++it) {
>@@ -394,9 +387,7 @@ class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> {
>     }
>     return false;
>   }
>-  virtual bool InsertDtmf(uint32_t ssrc,
>-                          int event_code,
>-                          int duration) {
>+  virtual bool InsertDtmf(uint32_t ssrc, int event_code, int duration) {
>     dtmf_info_queue_.push_back(DtmfInfo(ssrc, event_code, duration));
>     return true;
>   }
>@@ -488,7 +479,7 @@ class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> {
>         RTC_CHECK(it->second->source() == source);
>       } else {
>         local_sinks_.insert(std::make_pair(
>-            ssrc, rtc::MakeUnique<VoiceChannelAudioSink>(source)));
>+            ssrc, absl::make_unique<VoiceChannelAudioSink>(source)));
>       }
>     } else {
>       if (it != local_sinks_.end()) {
>@@ -579,13 +570,9 @@ class FakeVideoMediaChannel : public RtpHelper<VideoMediaChannel> {
>   bool SetSend(bool send) override { return set_sending(send); }
>   bool SetVideoSend(
>       uint32_t ssrc,
>-      bool enable,
>       const VideoOptions* options,
>       rtc::VideoSourceInterface<webrtc::VideoFrame>* source) override {
>-    if (!RtpHelper<VideoMediaChannel>::MuteStream(ssrc, !enable)) {
>-      return false;
>-    }
>-    if (enable && options) {
>+    if (options) {
>       if (!SetOptions(*options)) {
>         return false;
>       }
>@@ -741,9 +728,7 @@ class FakeDataMediaChannel : public RtpHelper<DataMediaChannel> {
> // and FakeVideoEngine.
> class FakeBaseEngine {
>  public:
>-  FakeBaseEngine()
>-      : options_changed_(false),
>-        fail_create_channel_(false) {}
>+  FakeBaseEngine() : options_changed_(false), fail_create_channel_(false) {}
>   void set_fail_create_channel(bool fail) { fail_create_channel_ = fail; }
> 
>   RtpCapabilities GetCapabilities() const { return capabilities_; }
>@@ -881,8 +866,8 @@ class FakeVideoEngine : public FakeBaseEngine {
>   friend class FakeMediaEngine;
> };
> 
>-class FakeMediaEngine :
>-    public CompositeMediaEngine<FakeVoiceEngine, FakeVideoEngine> {
>+class FakeMediaEngine
>+    : public CompositeMediaEngine<FakeVoiceEngine, FakeVideoEngine> {
>  public:
>   FakeMediaEngine()
>       : CompositeMediaEngine<FakeVoiceEngine, FakeVideoEngine>(std::tuple<>(),
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/fakenetworkinterface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/fakenetworkinterface.h
>index 3d98d1f9177..98d9c469a52 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/fakenetworkinterface.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/fakenetworkinterface.h
>@@ -37,8 +37,7 @@ class FakeNetworkInterface : public MediaChannel::NetworkInterface,
>         conf_(false),
>         sendbuf_size_(-1),
>         recvbuf_size_(-1),
>-        dscp_(rtc::DSCP_NO_CHANGE) {
>-  }
>+        dscp_(rtc::DSCP_NO_CHANGE) {}
> 
>   void SetDestination(MediaChannel* dest) { dest_ = dest; }
> 
>@@ -125,8 +124,7 @@ class FakeNetworkInterface : public MediaChannel::NetworkInterface,
>     rtp_packets_.push_back(*packet);
>     if (conf_) {
>       for (size_t i = 0; i < conf_sent_ssrcs_.size(); ++i) {
>-        if (!SetRtpSsrc(packet->data(), packet->size(),
>-                        conf_sent_ssrcs_[i])) {
>+        if (!SetRtpSsrc(packet->data(), packet->size(), conf_sent_ssrcs_[i])) {
>           return false;
>         }
>         PostMessage(ST_RTP, *packet);
>@@ -148,8 +146,7 @@ class FakeNetworkInterface : public MediaChannel::NetworkInterface,
>     return true;
>   }
> 
>-  virtual int SetOption(SocketType type, rtc::Socket::Option opt,
>-                        int option) {
>+  virtual int SetOption(SocketType type, rtc::Socket::Option opt, int option) {
>     if (opt == rtc::Socket::OPT_SNDBUF) {
>       sendbuf_size_ = option;
>     } else if (opt == rtc::Socket::OPT_RCVBUF) {
>@@ -166,15 +163,12 @@ class FakeNetworkInterface : public MediaChannel::NetworkInterface,
> 
>   virtual void OnMessage(rtc::Message* msg) {
>     rtc::TypedMessageData<rtc::CopyOnWriteBuffer>* msg_data =
>-        static_cast<rtc::TypedMessageData<rtc::CopyOnWriteBuffer>*>(
>-            msg->pdata);
>+        static_cast<rtc::TypedMessageData<rtc::CopyOnWriteBuffer>*>(msg->pdata);
>     if (dest_) {
>       if (msg->message_id == ST_RTP) {
>-        dest_->OnPacketReceived(&msg_data->data(),
>-                                rtc::CreatePacketTime(0));
>+        dest_->OnPacketReceived(&msg_data->data(), rtc::CreatePacketTime(0));
>       } else {
>-        dest_->OnRtcpReceived(&msg_data->data(),
>-                              rtc::CreatePacketTime(0));
>+        dest_->OnRtcpReceived(&msg_data->data(), rtc::CreatePacketTime(0));
>       }
>     }
>     delete msg_data;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/fakertp.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/fakertp.cc
>index be1631b73f7..fc9a058b87c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/fakertp.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/fakertp.cc
>@@ -13,9 +13,12 @@
> #include "media/base/fakertp.h"
> #include "rtc_base/gunit.h"
> 
>-void CompareHeaderExtensions(const char* packet1, size_t packet1_size,
>-    const char* packet2, size_t packet2_size,
>-    const std::vector<int> encrypted_headers, bool expect_equal) {
>+void CompareHeaderExtensions(const char* packet1,
>+                             size_t packet1_size,
>+                             const char* packet2,
>+                             size_t packet2_size,
>+                             const std::vector<int> encrypted_headers,
>+                             bool expect_equal) {
>   // Sanity check: packets must be large enough to contain the RTP header and
>   // extensions header.
>   RTC_CHECK_GE(packet1_size, 12 + 4);
>@@ -36,7 +39,7 @@ void CompareHeaderExtensions(const char* packet1, size_t packet1_size,
>   RTC_CHECK_GE(packet2_size, 12 + 4 + extension_words * 4);
>   while (extension_data1 < extension_end1) {
>     uint8_t id = (*extension_data1 & 0xf0) >> 4;
>-    uint8_t len = (*extension_data1 & 0x0f) +1;
>+    uint8_t len = (*extension_data1 & 0x0f) + 1;
>     extension_data1++;
>     extension_data2++;
>     EXPECT_LE(extension_data1, extension_end1);
>@@ -48,8 +51,8 @@ void CompareHeaderExtensions(const char* packet1, size_t packet1_size,
>     // The header extension doesn't get encrypted if the id is not in the
>     // list of header extensions to encrypt.
>     if (expect_equal ||
>-        std::find(encrypted_headers.begin(), encrypted_headers.end(), id)
>-            == encrypted_headers.end()) {
>+        std::find(encrypted_headers.begin(), encrypted_headers.end(), id) ==
>+            encrypted_headers.end()) {
>       EXPECT_EQ(0, memcmp(extension_data1, extension_data2, len));
>     } else {
>       EXPECT_NE(0, memcmp(extension_data1, extension_data2, len));
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/fakertp.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/fakertp.h
>index 6786bab6d92..946f302b00d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/fakertp.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/fakertp.h
>@@ -13,33 +13,28 @@
> #ifndef MEDIA_BASE_FAKERTP_H_
> #define MEDIA_BASE_FAKERTP_H_
> 
>+#include <cstddef>  // size_t
> #include <vector>
> 
> // A typical PCMU RTP packet.
> // PT=0, SN=1, TS=0, SSRC=1
> // all data FF
> static const unsigned char kPcmuFrame[] = {
>-  0x80, 0x00, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>+    0x80, 0x00, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01,
>+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>+    0xFF, 0xFF, 0xFF, 0xFF,
> };
> 
> static const int kHeaderExtensionIDs[] = {1, 4};
>@@ -48,93 +43,81 @@ static const int kHeaderExtensionIDs[] = {1, 4};
> // PT=0, SN=1, TS=0, SSRC=1
> // all data FF
> static const unsigned char kPcmuFrameWithExtensions[] = {
>-  0x90, 0x00, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01,
>-  // RFC 5285, section 4.2. One-Byte Header.
>-  0xBE, 0xDE,
>-  // Header extension length 6 * 32 bits.
>-  0x00, 0x06,
>-  // 8 bytes header id 1.
>-  0x17, 0x41, 0x42, 0x73, 0xA4, 0x75, 0x26, 0x27, 0x48,
>-  // 3 bytes header id 2.
>-  0x22, 0x00, 0x00, 0xC8,
>-  // 1 byte header id 3.
>-  0x30, 0x8E,
>-  // 7 bytes header id 4.
>-  0x46, 0x55, 0x99, 0x63, 0x86, 0xB3, 0x95, 0xFB,
>-  // 1 byte header padding.
>-  0x00,
>-  // Payload data.
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>+    0x90, 0x00, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01,
>+    // RFC 5285, section 4.2. One-Byte Header.
>+    0xBE, 0xDE,
>+    // Header extension length 6 * 32 bits.
>+    0x00, 0x06,
>+    // 8 bytes header id 1.
>+    0x17, 0x41, 0x42, 0x73, 0xA4, 0x75, 0x26, 0x27, 0x48,
>+    // 3 bytes header id 2.
>+    0x22, 0x00, 0x00, 0xC8,
>+    // 1 byte header id 3.
>+    0x30, 0x8E,
>+    // 7 bytes header id 4.
>+    0x46, 0x55, 0x99, 0x63, 0x86, 0xB3, 0x95, 0xFB,
>+    // 1 byte header padding.
>+    0x00,
>+    // Payload data.
>+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>+    0xFF, 0xFF, 0xFF, 0xFF,
> };
> 
> // A typical Receiver Report RTCP packet.
> // PT=RR, LN=1, SSRC=1
> // send SSRC=2, all other fields 0
> static const unsigned char kRtcpReport[] = {
>-  0x80, 0xc9, 0x00, 0x01, 0x00, 0x00, 0x00, 0x01,
>-  0x00, 0x00, 0x00, 0x02, 0x00, 0x00, 0x00, 0x00,
>-  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
>-  0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00
>-};
>+    0x80, 0xc9, 0x00, 0x01, 0x00, 0x00, 0x00, 0x01, 0x00, 0x00, 0x00,
>+    0x02, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
>+    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00};
> 
> // PT = 97, TS = 0, Seq = 1, SSRC = 2
> // H264 - NRI = 1, Type = 1, bit stream = FF
> 
> static const unsigned char kH264Packet[] = {
>-  0x80, 0x61, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x02,
>-  0x21, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>-  0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>+    0x80, 0x61, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x02,
>+    0x21, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>+    0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF,
>+    0xFF, 0xFF, 0xFF, 0xFF,
> };
> 
> // PT= 101, SN=2, TS=3, SSRC = 4
> static const unsigned char kDataPacket[] = {
>-  0x80, 0x65, 0x00, 0x02, 0x00, 0x00, 0x00, 0x03, 0x00, 0x00, 0x00, 0x04,
>-  0x00, 0x00, 0x00, 0x00,
>-  0x01, 0x02, 0x03, 0x04, 0x05, 0x06, 0x07, 0x08,
>+    0x80, 0x65, 0x00, 0x02, 0x00, 0x00, 0x00, 0x03, 0x00, 0x00, 0x00, 0x04,
>+    0x00, 0x00, 0x00, 0x00, 0x01, 0x02, 0x03, 0x04, 0x05, 0x06, 0x07, 0x08,
> };
> 
> // This expects both packets to be based on kPcmuFrameWithExtensions.
> // Header extensions with an id in "encrypted_headers" are expected to be
> // different in the packets unless "expect_equal" is set to "true".
>-void CompareHeaderExtensions(const char* packet1, size_t packet1_size,
>-    const char* packet2, size_t packet2_size,
>-    const std::vector<int> encrypted_headers, bool expect_equal);
>+void CompareHeaderExtensions(const char* packet1,
>+                             size_t packet1_size,
>+                             const char* packet2,
>+                             size_t packet2_size,
>+                             const std::vector<int> encrypted_headers,
>+                             bool expect_equal);
> 
> #endif  // MEDIA_BASE_FAKERTP_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/fakevideocapturer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/fakevideocapturer.cc
>new file mode 100644
>index 00000000000..94900ba37b6
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/fakevideocapturer.cc
>@@ -0,0 +1,158 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "media/base/fakevideocapturer.h"
>+
>+#include "rtc_base/arraysize.h"
>+
>+namespace cricket {
>+
>+FakeVideoCapturer::FakeVideoCapturer(bool is_screencast)
>+    : running_(false),
>+      is_screencast_(is_screencast),
>+      rotation_(webrtc::kVideoRotation_0) {
>+  // Default supported formats. Use ResetSupportedFormats to over write.
>+  using cricket::VideoFormat;
>+  static const VideoFormat formats[] = {
>+      {1280, 720, VideoFormat::FpsToInterval(30), cricket::FOURCC_I420},
>+      {640, 480, VideoFormat::FpsToInterval(30), cricket::FOURCC_I420},
>+      {320, 240, VideoFormat::FpsToInterval(30), cricket::FOURCC_I420},
>+      {160, 120, VideoFormat::FpsToInterval(30), cricket::FOURCC_I420},
>+      {1280, 720, VideoFormat::FpsToInterval(60), cricket::FOURCC_I420},
>+  };
>+  ResetSupportedFormats({&formats[0], &formats[arraysize(formats)]});
>+}
>+
>+FakeVideoCapturer::FakeVideoCapturer() : FakeVideoCapturer(false) {}
>+
>+FakeVideoCapturer::~FakeVideoCapturer() {
>+  SignalDestroyed(this);
>+}
>+
>+void FakeVideoCapturer::ResetSupportedFormats(
>+    const std::vector<cricket::VideoFormat>& formats) {
>+  SetSupportedFormats(formats);
>+}
>+
>+bool FakeVideoCapturer::CaptureFrame() {
>+  if (!GetCaptureFormat()) {
>+    return false;
>+  }
>+  RTC_CHECK_EQ(GetCaptureFormat()->fourcc, FOURCC_I420);
>+  return CaptureFrame(frame_source_->GetFrame());
>+}
>+
>+bool FakeVideoCapturer::CaptureCustomFrame(int width, int height) {
>+  // Default to 30fps.
>+  // TODO(nisse): Would anything break if we always stick to
>+  // the configure frame interval?
>+  return CaptureFrame(frame_source_->GetFrame(width, height, rotation_,
>+                                              rtc::kNumMicrosecsPerSec / 30));
>+}
>+
>+bool FakeVideoCapturer::CaptureFrame(const webrtc::VideoFrame& frame) {
>+  if (!running_) {
>+    return false;
>+  }
>+
>+  int adapted_width;
>+  int adapted_height;
>+  int crop_width;
>+  int crop_height;
>+  int crop_x;
>+  int crop_y;
>+
>+  // TODO(nisse): It's a bit silly to have this logic in a fake
>+  // class. Child classes of VideoCapturer are expected to call
>+  // AdaptFrame, and the test case
>+  // VideoCapturerTest.SinkWantsMaxPixelAndMaxPixelCountStepUp
>+  // depends on this.
>+  if (AdaptFrame(frame.width(), frame.height(), frame.timestamp_us(),
>+                 frame.timestamp_us(), &adapted_width, &adapted_height,
>+                 &crop_width, &crop_height, &crop_x, &crop_y, nullptr)) {
>+    rtc::scoped_refptr<webrtc::I420Buffer> buffer(
>+        webrtc::I420Buffer::Create(adapted_width, adapted_height));
>+    buffer->InitializeData();
>+
>+    OnFrame(webrtc::VideoFrame(buffer, frame.rotation(), frame.timestamp_us()),
>+            frame.width(), frame.height());
>+  }
>+
>+  return true;
>+}
>+
>+cricket::CaptureState FakeVideoCapturer::Start(
>+    const cricket::VideoFormat& format) {
>+  SetCaptureFormat(&format);
>+  running_ = true;
>+  SetCaptureState(cricket::CS_RUNNING);
>+  frame_source_ = absl::make_unique<FakeFrameSource>(
>+      format.width, format.height,
>+      format.interval / rtc::kNumNanosecsPerMicrosec);
>+  frame_source_->SetRotation(rotation_);
>+  return cricket::CS_RUNNING;
>+}
>+
>+void FakeVideoCapturer::Stop() {
>+  running_ = false;
>+  SetCaptureFormat(NULL);
>+  SetCaptureState(cricket::CS_STOPPED);
>+}
>+
>+bool FakeVideoCapturer::IsRunning() {
>+  return running_;
>+}
>+
>+bool FakeVideoCapturer::IsScreencast() const {
>+  return is_screencast_;
>+}
>+
>+bool FakeVideoCapturer::GetPreferredFourccs(std::vector<uint32_t>* fourccs) {
>+  fourccs->push_back(cricket::FOURCC_I420);
>+  fourccs->push_back(cricket::FOURCC_MJPG);
>+  return true;
>+}
>+
>+void FakeVideoCapturer::SetRotation(webrtc::VideoRotation rotation) {
>+  rotation_ = rotation;
>+  if (frame_source_)
>+    frame_source_->SetRotation(rotation_);
>+}
>+
>+webrtc::VideoRotation FakeVideoCapturer::GetRotation() {
>+  return rotation_;
>+}
>+
>+FakeVideoCapturerWithTaskQueue::FakeVideoCapturerWithTaskQueue(
>+    bool is_screencast)
>+    : FakeVideoCapturer(is_screencast) {}
>+
>+FakeVideoCapturerWithTaskQueue::FakeVideoCapturerWithTaskQueue() {}
>+
>+bool FakeVideoCapturerWithTaskQueue::CaptureFrame() {
>+  if (task_queue_.IsCurrent())
>+    return FakeVideoCapturer::CaptureFrame();
>+  bool ret = false;
>+  task_queue_.SendTask(
>+      [this, &ret]() { ret = FakeVideoCapturer::CaptureFrame(); });
>+  return ret;
>+}
>+
>+bool FakeVideoCapturerWithTaskQueue::CaptureCustomFrame(int width, int height) {
>+  if (task_queue_.IsCurrent())
>+    return FakeVideoCapturer::CaptureCustomFrame(width, height);
>+  bool ret = false;
>+  task_queue_.SendTask([this, &ret, width, height]() {
>+    ret = FakeVideoCapturer::CaptureCustomFrame(width, height);
>+  });
>+  return ret;
>+}
>+
>+}  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/fakevideocapturer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/fakevideocapturer.h
>index 536fe16356e..4a49f4504ad 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/fakevideocapturer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/fakevideocapturer.h
>@@ -18,8 +18,10 @@
> 
> #include "api/video/i420_buffer.h"
> #include "api/video/video_frame.h"
>+#include "media/base/fakeframesource.h"
> #include "media/base/videocapturer.h"
> #include "media/base/videocommon.h"
>+#include "rtc_base/task_queue_for_test.h"
> #include "rtc_base/timeutils.h"
> 
> namespace cricket {
>@@ -27,124 +29,50 @@ namespace cricket {
> // Fake video capturer that allows the test to manually pump in frames.
> class FakeVideoCapturer : public cricket::VideoCapturer {
>  public:
>-  explicit FakeVideoCapturer(bool is_screencast)
>-      : running_(false),
>-        initial_timestamp_(rtc::TimeNanos()),
>-        next_timestamp_(rtc::kNumNanosecsPerMillisec),
>-        is_screencast_(is_screencast),
>-        rotation_(webrtc::kVideoRotation_0) {
>-    // Default supported formats. Use ResetSupportedFormats to over write.
>-    std::vector<cricket::VideoFormat> formats;
>-    formats.push_back(cricket::VideoFormat(1280, 720,
>-        cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>-    formats.push_back(cricket::VideoFormat(640, 480,
>-        cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>-    formats.push_back(cricket::VideoFormat(320, 240,
>-        cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>-    formats.push_back(cricket::VideoFormat(160, 120,
>-        cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>-    formats.push_back(cricket::VideoFormat(1280, 720,
>-        cricket::VideoFormat::FpsToInterval(60), cricket::FOURCC_I420));
>-    ResetSupportedFormats(formats);
>-  }
>-  FakeVideoCapturer() : FakeVideoCapturer(false) {}
>-
>-  ~FakeVideoCapturer() {
>-    SignalDestroyed(this);
>-  }
>-
>-  void ResetSupportedFormats(const std::vector<cricket::VideoFormat>& formats) {
>-    SetSupportedFormats(formats);
>-  }
>-  bool CaptureFrame() {
>-    if (!GetCaptureFormat()) {
>-      return false;
>-    }
>-    return CaptureCustomFrame(GetCaptureFormat()->width,
>-                              GetCaptureFormat()->height,
>-                              GetCaptureFormat()->interval,
>-                              GetCaptureFormat()->fourcc);
>-  }
>-  bool CaptureCustomFrame(int width, int height, uint32_t fourcc) {
>-    // Default to 30fps.
>-    return CaptureCustomFrame(width, height, rtc::kNumNanosecsPerSec / 30,
>-                              fourcc);
>-  }
>-  bool CaptureCustomFrame(int width,
>-                          int height,
>-                          int64_t timestamp_interval,
>-                          uint32_t fourcc) {
>-    if (!running_) {
>-      return false;
>-    }
>-    RTC_CHECK(fourcc == FOURCC_I420);
>-    RTC_CHECK(width > 0);
>-    RTC_CHECK(height > 0);
>-
>-    int adapted_width;
>-    int adapted_height;
>-    int crop_width;
>-    int crop_height;
>-    int crop_x;
>-    int crop_y;
>-
>-    // TODO(nisse): It's a bit silly to have this logic in a fake
>-    // class. Child classes of VideoCapturer are expected to call
>-    // AdaptFrame, and the test case
>-    // VideoCapturerTest.SinkWantsMaxPixelAndMaxPixelCountStepUp
>-    // depends on this.
>-    if (AdaptFrame(width, height,
>-                   next_timestamp_ / rtc::kNumNanosecsPerMicrosec,
>-                   next_timestamp_ / rtc::kNumNanosecsPerMicrosec,
>-                   &adapted_width, &adapted_height, &crop_width, &crop_height,
>-                   &crop_x, &crop_y, nullptr)) {
>-      rtc::scoped_refptr<webrtc::I420Buffer> buffer(
>-          webrtc::I420Buffer::Create(adapted_width, adapted_height));
>-      buffer->InitializeData();
>-
>-      OnFrame(webrtc::VideoFrame(
>-                  buffer, rotation_,
>-                  next_timestamp_ / rtc::kNumNanosecsPerMicrosec),
>-              width, height);
>-    }
>-    next_timestamp_ += timestamp_interval;
>-
>-    return true;
>-  }
>+  explicit FakeVideoCapturer(bool is_screencast);
>+  FakeVideoCapturer();
>+
>+  ~FakeVideoCapturer() override;
>+
>+  void ResetSupportedFormats(const std::vector<cricket::VideoFormat>& formats);
>+  virtual bool CaptureFrame();
>+  virtual bool CaptureCustomFrame(int width, int height);
> 
>   sigslot::signal1<FakeVideoCapturer*> SignalDestroyed;
> 
>-  cricket::CaptureState Start(const cricket::VideoFormat& format) override {
>-    SetCaptureFormat(&format);
>-    running_ = true;
>-    SetCaptureState(cricket::CS_RUNNING);
>-    return cricket::CS_RUNNING;
>-  }
>-  void Stop() override {
>-    running_ = false;
>-    SetCaptureFormat(NULL);
>-    SetCaptureState(cricket::CS_STOPPED);
>-  }
>-  bool IsRunning() override { return running_; }
>-  bool IsScreencast() const override { return is_screencast_; }
>-  bool GetPreferredFourccs(std::vector<uint32_t>* fourccs) override {
>-    fourccs->push_back(cricket::FOURCC_I420);
>-    fourccs->push_back(cricket::FOURCC_MJPG);
>-    return true;
>-  }
>-
>-  void SetRotation(webrtc::VideoRotation rotation) {
>-    rotation_ = rotation;
>-  }
>-
>-  webrtc::VideoRotation GetRotation() { return rotation_; }
>+  cricket::CaptureState Start(const cricket::VideoFormat& format) override;
>+  void Stop() override;
>+  bool IsRunning() override;
>+  bool IsScreencast() const override;
>+  bool GetPreferredFourccs(std::vector<uint32_t>* fourccs) override;
>+
>+  void SetRotation(webrtc::VideoRotation rotation);
>+
>+  webrtc::VideoRotation GetRotation();
> 
>  private:
>+  bool CaptureFrame(const webrtc::VideoFrame& frame);
>+
>   bool running_;
>-  int64_t initial_timestamp_;
>-  int64_t next_timestamp_;
>   const bool is_screencast_;
>+  // Duplicates FakeFrameSource::rotation_, but needed to support
>+  // SetRotation before Start.
>   webrtc::VideoRotation rotation_;
>+  std::unique_ptr<FakeFrameSource> frame_source_;
>+};
>+
>+// Inherits from FakeVideoCapturer but adds a TaskQueue so that frames can be
>+// delivered on a TaskQueue as expected by VideoSinkInterface implementations.
>+class FakeVideoCapturerWithTaskQueue : public FakeVideoCapturer {
>+ public:
>+  explicit FakeVideoCapturerWithTaskQueue(bool is_screencast);
>+  FakeVideoCapturerWithTaskQueue();
>+
>+  bool CaptureFrame() override;
>+  bool CaptureCustomFrame(int width, int height) override;
>+
>+ protected:
>+  rtc::test::TaskQueueForTest task_queue_{"FakeVideoCapturerWithTaskQueue"};
> };
> 
> }  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/fakevideorenderer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/fakevideorenderer.cc
>new file mode 100644
>index 00000000000..fdd789a4b7d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/fakevideorenderer.cc
>@@ -0,0 +1,33 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "media/base/fakevideorenderer.h"
>+
>+namespace cricket {
>+
>+FakeVideoRenderer::FakeVideoRenderer() = default;
>+
>+void FakeVideoRenderer::OnFrame(const webrtc::VideoFrame& frame) {
>+  rtc::CritScope cs(&crit_);
>+  // TODO(zhurunz) Check with VP8 team to see if we can remove this
>+  // tolerance on Y values. Some unit tests produce Y values close
>+  // to 16 rather than close to zero, for supposedly black frames.
>+  // Largest value observed is 34, e.g., running
>+  // PeerConnectionIntegrationTest.SendAndReceive16To9AspectRatio.
>+  black_frame_ = CheckFrameColorYuv(0, 48, 128, 128, 128, 128, &frame);
>+  // Treat unexpected frame size as error.
>+  ++num_rendered_frames_;
>+  width_ = frame.width();
>+  height_ = frame.height();
>+  rotation_ = frame.rotation();
>+  timestamp_us_ = frame.timestamp_us();
>+}
>+
>+}  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/fakevideorenderer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/fakevideorenderer.h
>index fa9aff70f52..f072c729e66 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/fakevideorenderer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/fakevideorenderer.h
>@@ -12,7 +12,7 @@
> #define MEDIA_BASE_FAKEVIDEORENDERER_H_
> 
> #include "api/video/video_frame.h"
>-#include "api/videosinkinterface.h"
>+#include "api/video/video_sink_interface.h"
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/logging.h"
> 
>@@ -21,30 +21,9 @@ namespace cricket {
> // Faked video renderer that has a callback for actions on rendering.
> class FakeVideoRenderer : public rtc::VideoSinkInterface<webrtc::VideoFrame> {
>  public:
>-  FakeVideoRenderer()
>-      : errors_(0),
>-        width_(0),
>-        height_(0),
>-        rotation_(webrtc::kVideoRotation_0),
>-        timestamp_us_(0),
>-        num_rendered_frames_(0),
>-        black_frame_(false) {}
>+  FakeVideoRenderer();
> 
>-  virtual void OnFrame(const webrtc::VideoFrame& frame) {
>-    rtc::CritScope cs(&crit_);
>-    // TODO(zhurunz) Check with VP8 team to see if we can remove this
>-    // tolerance on Y values. Some unit tests produce Y values close
>-    // to 16 rather than close to zero, for supposedly black frames.
>-    // Largest value observed is 34, e.g., running
>-    // PeerConnectionIntegrationTest.SendAndReceive16To9AspectRatio.
>-    black_frame_ = CheckFrameColorYuv(0, 48, 128, 128, 128, 128, &frame);
>-    // Treat unexpected frame size as error.
>-    ++num_rendered_frames_;
>-    width_ = frame.width();
>-    height_ = frame.height();
>-    rotation_ = frame.rotation();
>-    timestamp_us_ = frame.timestamp_us();
>-  }
>+  void OnFrame(const webrtc::VideoFrame& frame) override;
> 
>   int errors() const { return errors_; }
>   int width() const {
>@@ -127,13 +106,13 @@ class FakeVideoRenderer : public rtc::VideoSinkInterface<webrtc::VideoFrame> {
>     return true;
>   }
> 
>-  int errors_;
>-  int width_;
>-  int height_;
>-  webrtc::VideoRotation rotation_;
>-  int64_t timestamp_us_;
>-  int num_rendered_frames_;
>-  bool black_frame_;
>+  int errors_ = 0;
>+  int width_ = 0;
>+  int height_ = 0;
>+  webrtc::VideoRotation rotation_ = webrtc::kVideoRotation_0;
>+  int64_t timestamp_us_ = 0;
>+  int num_rendered_frames_ = 0;
>+  bool black_frame_ = false;
>   rtc::CriticalSection crit_;
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/h264_profile_level_id.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/h264_profile_level_id.cc
>index 4731c18f404..65a6cf0cc91 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/h264_profile_level_id.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/h264_profile_level_id.cc
>@@ -112,19 +112,19 @@ static constexpr LevelConstraint kLevelConstraints[] = {
>     {245760, 8192, webrtc::H264::kLevel4_1},
>     {522240, 8704, webrtc::H264::kLevel4_2},
>     {589824, 22080, webrtc::H264::kLevel5},
>-    {983040, 3684, webrtc::H264::kLevel5_1},
>-    {2073600, 3684, webrtc::H264::kLevel5_2},
>+    {983040, 36864, webrtc::H264::kLevel5_1},
>+    {2073600, 36864, webrtc::H264::kLevel5_2},
> };
> 
> }  // anonymous namespace
> 
>-rtc::Optional<ProfileLevelId> ParseProfileLevelId(const char* str) {
>+absl::optional<ProfileLevelId> ParseProfileLevelId(const char* str) {
>   // The string should consist of 3 bytes in hexadecimal format.
>   if (strlen(str) != 6u)
>-    return rtc::nullopt;
>+    return absl::nullopt;
>   const uint32_t profile_level_id_numeric = strtol(str, nullptr, 16);
>   if (profile_level_id_numeric == 0)
>-    return rtc::nullopt;
>+    return absl::nullopt;
> 
>   // Separate into three bytes.
>   const uint8_t level_idc =
>@@ -159,7 +159,7 @@ rtc::Optional<ProfileLevelId> ParseProfileLevelId(const char* str) {
>       break;
>     default:
>       // Unrecognized level_idc.
>-      return rtc::nullopt;
>+      return absl::nullopt;
>   }
> 
>   // Parse profile_idc/profile_iop into a Profile enum.
>@@ -171,10 +171,10 @@ rtc::Optional<ProfileLevelId> ParseProfileLevelId(const char* str) {
>   }
> 
>   // Unrecognized profile_idc/profile_iop combination.
>-  return rtc::nullopt;
>+  return absl::nullopt;
> }
> 
>-rtc::Optional<Level> SupportedLevel(int max_frame_pixel_count, float max_fps) {
>+absl::optional<Level> SupportedLevel(int max_frame_pixel_count, float max_fps) {
>   static const int kPixelsPerMacroblock = 16 * 16;
> 
>   for (int i = arraysize(kLevelConstraints) - 1; i >= 0; --i) {
>@@ -188,10 +188,10 @@ rtc::Optional<Level> SupportedLevel(int max_frame_pixel_count, float max_fps) {
>   }
> 
>   // No level supported.
>-  return rtc::nullopt;
>+  return absl::nullopt;
> }
> 
>-rtc::Optional<ProfileLevelId> ParseSdpProfileLevelId(
>+absl::optional<ProfileLevelId> ParseSdpProfileLevelId(
>     const CodecParameterMap& params) {
>   // TODO(magjed): The default should really be kProfileBaseline and kLevel1
>   // according to the spec: https://tools.ietf.org/html/rfc6184#section-8.1. In
>@@ -209,7 +209,7 @@ rtc::Optional<ProfileLevelId> ParseSdpProfileLevelId(
>              : ParseProfileLevelId(profile_level_id_it->second.c_str());
> }
> 
>-rtc::Optional<std::string> ProfileLevelIdToString(
>+absl::optional<std::string> ProfileLevelIdToString(
>     const ProfileLevelId& profile_level_id) {
>   // Handle special case level == 1b.
>   if (profile_level_id.level == kLevel1_b) {
>@@ -222,7 +222,7 @@ rtc::Optional<std::string> ProfileLevelIdToString(
>         return {"4d100b"};
>       // Level 1b is not allowed for other profiles.
>       default:
>-        return rtc::nullopt;
>+        return absl::nullopt;
>     }
>   }
> 
>@@ -245,7 +245,7 @@ rtc::Optional<std::string> ProfileLevelIdToString(
>       break;
>     // Unrecognized profile.
>     default:
>-      return rtc::nullopt;
>+      return absl::nullopt;
>   }
> 
>   char str[7];
>@@ -267,9 +267,9 @@ void GenerateProfileLevelIdForAnswer(
>   }
> 
>   // Parse profile-level-ids.
>-  const rtc::Optional<ProfileLevelId> local_profile_level_id =
>+  const absl::optional<ProfileLevelId> local_profile_level_id =
>       ParseSdpProfileLevelId(local_supported_params);
>-  const rtc::Optional<ProfileLevelId> remote_profile_level_id =
>+  const absl::optional<ProfileLevelId> remote_profile_level_id =
>       ParseSdpProfileLevelId(remote_offered_params);
>   // The local and remote codec must have valid and equal H264 Profiles.
>   RTC_DCHECK(local_profile_level_id);
>@@ -297,9 +297,9 @@ void GenerateProfileLevelIdForAnswer(
> 
> bool IsSameH264Profile(const CodecParameterMap& params1,
>                        const CodecParameterMap& params2) {
>-  const rtc::Optional<webrtc::H264::ProfileLevelId> profile_level_id =
>+  const absl::optional<webrtc::H264::ProfileLevelId> profile_level_id =
>       webrtc::H264::ParseSdpProfileLevelId(params1);
>-  const rtc::Optional<webrtc::H264::ProfileLevelId> other_profile_level_id =
>+  const absl::optional<webrtc::H264::ProfileLevelId> other_profile_level_id =
>       webrtc::H264::ParseSdpProfileLevelId(params2);
>   // Compare H264 profiles, but not levels.
>   return profile_level_id && other_profile_level_id &&
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/h264_profile_level_id.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/h264_profile_level_id.h
>index 28899e53d98..b4ff88369c6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/h264_profile_level_id.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/h264_profile_level_id.h
>@@ -14,7 +14,7 @@
> #include <map>
> #include <string>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> #include "common_types.h"  // NOLINT(build/include)
> 
> namespace webrtc {
>@@ -46,7 +46,7 @@ enum Level {
> };
> 
> struct ProfileLevelId {
>-  ProfileLevelId(Profile profile, Level level)
>+  constexpr ProfileLevelId(Profile profile, Level level)
>       : profile(profile), level(level) {}
>   Profile profile;
>   Level level;
>@@ -55,24 +55,24 @@ struct ProfileLevelId {
> // Parse profile level id that is represented as a string of 3 hex bytes.
> // Nothing will be returned if the string is not a recognized H264
> // profile level id.
>-rtc::Optional<ProfileLevelId> ParseProfileLevelId(const char* str);
>+absl::optional<ProfileLevelId> ParseProfileLevelId(const char* str);
> 
> // Parse profile level id that is represented as a string of 3 hex bytes
> // contained in an SDP key-value map. A default profile level id will be
> // returned if the profile-level-id key is missing. Nothing will be returned if
> // the key is present but the string is invalid.
>-rtc::Optional<ProfileLevelId> ParseSdpProfileLevelId(
>+absl::optional<ProfileLevelId> ParseSdpProfileLevelId(
>     const CodecParameterMap& params);
> 
> // Given that a decoder supports up to a given frame size (in pixels) at up to a
> // given number of frames per second, return the highest H.264 level where it
> // can guarantee that it will be able to support all valid encoded streams that
> // are within that level.
>-rtc::Optional<Level> SupportedLevel(int max_frame_pixel_count, float max_fps);
>+absl::optional<Level> SupportedLevel(int max_frame_pixel_count, float max_fps);
> 
> // Returns canonical string representation as three hex bytes of the profile
> // level id, or returns nothing for invalid profile level ids.
>-rtc::Optional<std::string> ProfileLevelIdToString(
>+absl::optional<std::string> ProfileLevelIdToString(
>     const ProfileLevelId& profile_level_id);
> 
> // Generate codec parameters that will be used as answer in an SDP negotiation
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediachannel.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediachannel.cc
>new file mode 100644
>index 00000000000..019739d051c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediachannel.cc
>@@ -0,0 +1,86 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "media/base/mediachannel.h"
>+
>+namespace cricket {
>+
>+VideoOptions::VideoOptions() = default;
>+VideoOptions::~VideoOptions() = default;
>+
>+void MediaChannel::SetInterface(NetworkInterface* iface) {
>+  rtc::CritScope cs(&network_interface_crit_);
>+  network_interface_ = iface;
>+  SetDscp(enable_dscp_ ? PreferredDscp() : rtc::DSCP_DEFAULT);
>+}
>+
>+rtc::DiffServCodePoint MediaChannel::PreferredDscp() const {
>+  return rtc::DSCP_DEFAULT;
>+}
>+
>+int MediaChannel::GetRtpSendTimeExtnId() const {
>+  return -1;
>+}
>+
>+MediaSenderInfo::MediaSenderInfo() = default;
>+MediaSenderInfo::~MediaSenderInfo() = default;
>+
>+MediaReceiverInfo::MediaReceiverInfo() = default;
>+MediaReceiverInfo::~MediaReceiverInfo() = default;
>+
>+VoiceSenderInfo::VoiceSenderInfo() = default;
>+VoiceSenderInfo::~VoiceSenderInfo() = default;
>+
>+VoiceReceiverInfo::VoiceReceiverInfo() = default;
>+VoiceReceiverInfo::~VoiceReceiverInfo() = default;
>+
>+VideoSenderInfo::VideoSenderInfo() = default;
>+VideoSenderInfo::~VideoSenderInfo() = default;
>+
>+VideoReceiverInfo::VideoReceiverInfo() = default;
>+VideoReceiverInfo::~VideoReceiverInfo() = default;
>+
>+VoiceMediaInfo::VoiceMediaInfo() = default;
>+VoiceMediaInfo::~VoiceMediaInfo() = default;
>+
>+VideoMediaInfo::VideoMediaInfo() = default;
>+VideoMediaInfo::~VideoMediaInfo() = default;
>+
>+DataMediaInfo::DataMediaInfo() = default;
>+DataMediaInfo::~DataMediaInfo() = default;
>+
>+AudioSendParameters::AudioSendParameters() = default;
>+AudioSendParameters::~AudioSendParameters() = default;
>+
>+std::map<std::string, std::string> AudioSendParameters::ToStringMap() const {
>+  auto params = RtpSendParameters<AudioCodec>::ToStringMap();
>+  params["options"] = options.ToString();
>+  return params;
>+}
>+
>+VideoSendParameters::VideoSendParameters() = default;
>+VideoSendParameters::~VideoSendParameters() = default;
>+
>+std::map<std::string, std::string> VideoSendParameters::ToStringMap() const {
>+  auto params = RtpSendParameters<VideoCodec>::ToStringMap();
>+  params["conference_mode"] = (conference_mode ? "yes" : "no");
>+  return params;
>+}
>+
>+DataMediaChannel::DataMediaChannel() = default;
>+DataMediaChannel::DataMediaChannel(const MediaConfig& config)
>+    : MediaChannel(config) {}
>+DataMediaChannel::~DataMediaChannel() = default;
>+
>+bool DataMediaChannel::GetStats(DataMediaInfo* info) {
>+  return true;
>+}
>+
>+}  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediachannel.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediachannel.h
>index 89d35b79da5..f0e16583230 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediachannel.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediachannel.h
>@@ -17,40 +17,40 @@
> #include <utility>
> #include <vector>
> 
>+#include "absl/types/optional.h"
> #include "api/audio_codecs/audio_encoder.h"
>-#include "api/optional.h"
>+#include "api/audio_options.h"
>+#include "api/rtcerror.h"
> #include "api/rtpparameters.h"
> #include "api/rtpreceiverinterface.h"
> #include "api/video/video_content_type.h"
>+#include "api/video/video_sink_interface.h"
>+#include "api/video/video_source_interface.h"
> #include "api/video/video_timing.h"
>-#include "api/videosinkinterface.h"
>-#include "api/videosourceinterface.h"
>-#include "call/video_config.h"
>+#include "api/video_codecs/video_encoder_config.h"
> #include "media/base/codec.h"
>+#include "media/base/mediaconfig.h"
> #include "media/base/mediaconstants.h"
> #include "media/base/streamparams.h"
> #include "modules/audio_processing/include/audio_processing_statistics.h"
> #include "rtc_base/asyncpacketsocket.h"
>-#include "rtc_base/basictypes.h"
> #include "rtc_base/buffer.h"
> #include "rtc_base/copyonwritebuffer.h"
> #include "rtc_base/dscp.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/networkroute.h"
>-#include "rtc_base/sigslot.h"
> #include "rtc_base/socket.h"
> #include "rtc_base/stringencode.h"
>-
>+#include "rtc_base/third_party/sigslot/sigslot.h"
> 
> namespace rtc {
>-class RateLimiter;
> class Timing;
> }
> 
> namespace webrtc {
> class AudioSinkInterface;
> class VideoFrame;
>-}
>+}  // namespace webrtc
> 
> namespace cricket {
> 
>@@ -62,7 +62,8 @@ struct VideoFormat;
> const int kScreencastDefaultFps = 5;
> 
> template <class T>
>-static std::string ToStringIfSet(const char* key, const rtc::Optional<T>& val) {
>+static std::string ToStringIfSet(const char* key,
>+                                 const absl::optional<T>& val) {
>   std::string str;
>   if (val) {
>     str = key;
>@@ -75,249 +76,26 @@ static std::string ToStringIfSet(const char* key, const rtc::Optional<T>& val) {
> 
> template <class T>
> static std::string VectorToString(const std::vector<T>& vals) {
>-    std::ostringstream ost;
>-    ost << "[";
>-    for (size_t i = 0; i < vals.size(); ++i) {
>-      if (i > 0) {
>-        ost << ", ";
>-      }
>-      ost << vals[i].ToString();
>+  std::ostringstream ost;  // no-presubmit-check TODO(webrtc:8982)
>+  ost << "[";
>+  for (size_t i = 0; i < vals.size(); ++i) {
>+    if (i > 0) {
>+      ost << ", ";
>     }
>-    ost << "]";
>-    return ost.str();
>-}
>-
>-// Construction-time settings, passed on when creating
>-// MediaChannels.
>-struct MediaConfig {
>-  // Set DSCP value on packets. This flag comes from the
>-  // PeerConnection constraint 'googDscp'.
>-  bool enable_dscp = false;
>-
>-  // Video-specific config.
>-  struct Video {
>-    // Enable WebRTC CPU Overuse Detection. This flag comes from the
>-    // PeerConnection constraint 'googCpuOveruseDetection'.
>-    bool enable_cpu_overuse_detection = true;
>-
>-    // Enable WebRTC suspension of video. No video frames will be sent
>-    // when the bitrate is below the configured minimum bitrate. This
>-    // flag comes from the PeerConnection constraint
>-    // 'googSuspendBelowMinBitrate', and WebRtcVideoChannel copies it
>-    // to VideoSendStream::Config::suspend_below_min_bitrate.
>-    bool suspend_below_min_bitrate = false;
>-
>-    // Set to true if the renderer has an algorithm of frame selection.
>-    // If the value is true, then WebRTC will hand over a frame as soon as
>-    // possible without delay, and rendering smoothness is completely the duty
>-    // of the renderer;
>-    // If the value is false, then WebRTC is responsible to delay frame release
>-    // in order to increase rendering smoothness.
>-    //
>-    // This flag comes from PeerConnection's RtcConfiguration, but is
>-    // currently only set by the command line flag
>-    // 'disable-rtc-smoothness-algorithm'.
>-    // WebRtcVideoChannel::AddRecvStream copies it to the created
>-    // WebRtcVideoReceiveStream, where it is returned by the
>-    // SmoothsRenderedFrames method. This method is used by the
>-    // VideoReceiveStream, where the value is passed on to the
>-    // IncomingVideoStream constructor.
>-    bool disable_prerenderer_smoothing = false;
>-
>-    // Enables periodic bandwidth probing in application-limited region.
>-    bool periodic_alr_bandwidth_probing = false;
>-  } video;
>-
>-  bool operator==(const MediaConfig& o) const {
>-    return enable_dscp == o.enable_dscp &&
>-           video.enable_cpu_overuse_detection ==
>-               o.video.enable_cpu_overuse_detection &&
>-           video.suspend_below_min_bitrate ==
>-               o.video.suspend_below_min_bitrate &&
>-           video.disable_prerenderer_smoothing ==
>-               o.video.disable_prerenderer_smoothing &&
>-           video.periodic_alr_bandwidth_probing ==
>-               o.video.periodic_alr_bandwidth_probing;
>-  }
>-
>-  bool operator!=(const MediaConfig& o) const { return !(*this == o); }
>-};
>-
>-// Options that can be applied to a VoiceMediaChannel or a VoiceMediaEngine.
>-// Used to be flags, but that makes it hard to selectively apply options.
>-// We are moving all of the setting of options to structs like this,
>-// but some things currently still use flags.
>-struct AudioOptions {
>-  void SetAll(const AudioOptions& change) {
>-    SetFrom(&echo_cancellation, change.echo_cancellation);
>-#if defined(WEBRTC_IOS)
>-    SetFrom(&ios_force_software_aec_HACK, change.ios_force_software_aec_HACK);
>-#endif
>-    SetFrom(&auto_gain_control, change.auto_gain_control);
>-    SetFrom(&noise_suppression, change.noise_suppression);
>-    SetFrom(&highpass_filter, change.highpass_filter);
>-    SetFrom(&stereo_swapping, change.stereo_swapping);
>-    SetFrom(&audio_jitter_buffer_max_packets,
>-            change.audio_jitter_buffer_max_packets);
>-    SetFrom(&audio_jitter_buffer_fast_accelerate,
>-            change.audio_jitter_buffer_fast_accelerate);
>-    SetFrom(&typing_detection, change.typing_detection);
>-    SetFrom(&aecm_generate_comfort_noise, change.aecm_generate_comfort_noise);
>-    SetFrom(&experimental_agc, change.experimental_agc);
>-    SetFrom(&extended_filter_aec, change.extended_filter_aec);
>-    SetFrom(&delay_agnostic_aec, change.delay_agnostic_aec);
>-    SetFrom(&experimental_ns, change.experimental_ns);
>-    SetFrom(&intelligibility_enhancer, change.intelligibility_enhancer);
>-    SetFrom(&level_control, change.level_control);
>-    SetFrom(&residual_echo_detector, change.residual_echo_detector);
>-    SetFrom(&tx_agc_target_dbov, change.tx_agc_target_dbov);
>-    SetFrom(&tx_agc_digital_compression_gain,
>-            change.tx_agc_digital_compression_gain);
>-    SetFrom(&tx_agc_limiter, change.tx_agc_limiter);
>-    SetFrom(&combined_audio_video_bwe, change.combined_audio_video_bwe);
>-    SetFrom(&audio_network_adaptor, change.audio_network_adaptor);
>-    SetFrom(&audio_network_adaptor_config, change.audio_network_adaptor_config);
>-    SetFrom(&level_control_initial_peak_level_dbfs,
>-            change.level_control_initial_peak_level_dbfs);
>-  }
>-
>-  bool operator==(const AudioOptions& o) const {
>-    return echo_cancellation == o.echo_cancellation &&
>-#if defined(WEBRTC_IOS)
>-           ios_force_software_aec_HACK == o.ios_force_software_aec_HACK &&
>-#endif
>-           auto_gain_control == o.auto_gain_control &&
>-           noise_suppression == o.noise_suppression &&
>-           highpass_filter == o.highpass_filter &&
>-           stereo_swapping == o.stereo_swapping &&
>-           audio_jitter_buffer_max_packets ==
>-               o.audio_jitter_buffer_max_packets &&
>-           audio_jitter_buffer_fast_accelerate ==
>-               o.audio_jitter_buffer_fast_accelerate &&
>-           typing_detection == o.typing_detection &&
>-           aecm_generate_comfort_noise == o.aecm_generate_comfort_noise &&
>-           experimental_agc == o.experimental_agc &&
>-           extended_filter_aec == o.extended_filter_aec &&
>-           delay_agnostic_aec == o.delay_agnostic_aec &&
>-           experimental_ns == o.experimental_ns &&
>-           intelligibility_enhancer == o.intelligibility_enhancer &&
>-           level_control == o.level_control &&
>-           residual_echo_detector == o.residual_echo_detector &&
>-           tx_agc_target_dbov == o.tx_agc_target_dbov &&
>-           tx_agc_digital_compression_gain ==
>-               o.tx_agc_digital_compression_gain &&
>-           tx_agc_limiter == o.tx_agc_limiter &&
>-           combined_audio_video_bwe == o.combined_audio_video_bwe &&
>-           audio_network_adaptor == o.audio_network_adaptor &&
>-           audio_network_adaptor_config == o.audio_network_adaptor_config &&
>-           level_control_initial_peak_level_dbfs ==
>-               o.level_control_initial_peak_level_dbfs;
>-  }
>-  bool operator!=(const AudioOptions& o) const { return !(*this == o); }
>-
>-  std::string ToString() const {
>-    std::ostringstream ost;
>-    ost << "AudioOptions {";
>-    ost << ToStringIfSet("aec", echo_cancellation);
>-#if defined(WEBRTC_IOS)
>-    ost << ToStringIfSet("ios_force_software_aec_HACK",
>-                         ios_force_software_aec_HACK);
>-#endif
>-    ost << ToStringIfSet("agc", auto_gain_control);
>-    ost << ToStringIfSet("ns", noise_suppression);
>-    ost << ToStringIfSet("hf", highpass_filter);
>-    ost << ToStringIfSet("swap", stereo_swapping);
>-    ost << ToStringIfSet("audio_jitter_buffer_max_packets",
>-                         audio_jitter_buffer_max_packets);
>-    ost << ToStringIfSet("audio_jitter_buffer_fast_accelerate",
>-                         audio_jitter_buffer_fast_accelerate);
>-    ost << ToStringIfSet("typing", typing_detection);
>-    ost << ToStringIfSet("comfort_noise", aecm_generate_comfort_noise);
>-    ost << ToStringIfSet("experimental_agc", experimental_agc);
>-    ost << ToStringIfSet("extended_filter_aec", extended_filter_aec);
>-    ost << ToStringIfSet("delay_agnostic_aec", delay_agnostic_aec);
>-    ost << ToStringIfSet("experimental_ns", experimental_ns);
>-    ost << ToStringIfSet("intelligibility_enhancer", intelligibility_enhancer);
>-    ost << ToStringIfSet("level_control", level_control);
>-    ost << ToStringIfSet("level_control_initial_peak_level_dbfs",
>-                         level_control_initial_peak_level_dbfs);
>-    ost << ToStringIfSet("residual_echo_detector", residual_echo_detector);
>-    ost << ToStringIfSet("tx_agc_target_dbov", tx_agc_target_dbov);
>-    ost << ToStringIfSet("tx_agc_digital_compression_gain",
>-        tx_agc_digital_compression_gain);
>-    ost << ToStringIfSet("tx_agc_limiter", tx_agc_limiter);
>-    ost << ToStringIfSet("combined_audio_video_bwe", combined_audio_video_bwe);
>-    ost << ToStringIfSet("audio_network_adaptor", audio_network_adaptor);
>-    // The adaptor config is a serialized proto buffer and therefore not human
>-    // readable. So we comment out the following line.
>-    // ost << ToStringIfSet("audio_network_adaptor_config",
>-    //     audio_network_adaptor_config);
>-    ost << "}";
>-    return ost.str();
>+    ost << vals[i].ToString();
>   }
>-
>-  // Audio processing that attempts to filter away the output signal from
>-  // later inbound pickup.
>-  rtc::Optional<bool> echo_cancellation;
>-#if defined(WEBRTC_IOS)
>-  // Forces software echo cancellation on iOS. This is a temporary workaround
>-  // (until Apple fixes the bug) for a device with non-functioning AEC. May
>-  // improve performance on that particular device, but will cause unpredictable
>-  // behavior in all other cases. See http://bugs.webrtc.org/8682.
>-  rtc::Optional<bool> ios_force_software_aec_HACK;
>-#endif
>-  // Audio processing to adjust the sensitivity of the local mic dynamically.
>-  rtc::Optional<bool> auto_gain_control;
>-  // Audio processing to filter out background noise.
>-  rtc::Optional<bool> noise_suppression;
>-  // Audio processing to remove background noise of lower frequencies.
>-  rtc::Optional<bool> highpass_filter;
>-  // Audio processing to swap the left and right channels.
>-  rtc::Optional<bool> stereo_swapping;
>-  // Audio receiver jitter buffer (NetEq) max capacity in number of packets.
>-  rtc::Optional<int> audio_jitter_buffer_max_packets;
>-  // Audio receiver jitter buffer (NetEq) fast accelerate mode.
>-  rtc::Optional<bool> audio_jitter_buffer_fast_accelerate;
>-  // Audio processing to detect typing.
>-  rtc::Optional<bool> typing_detection;
>-  rtc::Optional<bool> aecm_generate_comfort_noise;
>-  rtc::Optional<bool> experimental_agc;
>-  rtc::Optional<bool> extended_filter_aec;
>-  rtc::Optional<bool> delay_agnostic_aec;
>-  rtc::Optional<bool> experimental_ns;
>-  rtc::Optional<bool> intelligibility_enhancer;
>-  rtc::Optional<bool> level_control;
>-  // Specifies an optional initialization value for the level controller.
>-  rtc::Optional<float> level_control_initial_peak_level_dbfs;
>-  // Note that tx_agc_* only applies to non-experimental AGC.
>-  rtc::Optional<bool> residual_echo_detector;
>-  rtc::Optional<uint16_t> tx_agc_target_dbov;
>-  rtc::Optional<uint16_t> tx_agc_digital_compression_gain;
>-  rtc::Optional<bool> tx_agc_limiter;
>-  // Enable combined audio+bandwidth BWE.
>-  // TODO(pthatcher): This flag is set from the
>-  // "googCombinedAudioVideoBwe", but not used anywhere. So delete it,
>-  // and check if any other AudioOptions members are unused.
>-  rtc::Optional<bool> combined_audio_video_bwe;
>-  // Enable audio network adaptor.
>-  rtc::Optional<bool> audio_network_adaptor;
>-  // Config string for audio network adaptor.
>-  rtc::Optional<std::string> audio_network_adaptor_config;
>-
>- private:
>-  template <typename T>
>-  static void SetFrom(rtc::Optional<T>* s, const rtc::Optional<T>& o) {
>-    if (o) {
>-      *s = o;
>-    }
>-  }
>-};
>+  ost << "]";
>+  return ost.str();
>+}
> 
> // Options that can be applied to a VideoMediaChannel or a VideoMediaEngine.
> // Used to be flags, but that makes it hard to selectively apply options.
> // We are moving all of the setting of options to structs like this,
> // but some things currently still use flags.
> struct VideoOptions {
>+  VideoOptions();
>+  ~VideoOptions();
>+
>   void SetAll(const VideoOptions& change) {
>     SetFrom(&video_noise_reduction, change.video_noise_reduction);
>     SetFrom(&screencast_min_bitrate_kbps, change.screencast_min_bitrate_kbps);
>@@ -345,20 +123,20 @@ struct VideoOptions {
>   // Enable denoising? This flag comes from the getUserMedia
>   // constraint 'googNoiseReduction', and WebRtcVideoEngine passes it
>   // on to the codec options. Disabled by default.
>-  rtc::Optional<bool> video_noise_reduction;
>+  absl::optional<bool> video_noise_reduction;
>   // Force screencast to use a minimum bitrate. This flag comes from
>   // the PeerConnection constraint 'googScreencastMinBitrate'. It is
>   // copied to the encoder config by WebRtcVideoChannel.
>-  rtc::Optional<int> screencast_min_bitrate_kbps;
>+  absl::optional<int> screencast_min_bitrate_kbps;
>   // Set by screencast sources. Implies selection of encoding settings
>   // suitable for screencast. Most likely not the right way to do
>   // things, e.g., screencast of a text document and screencast of a
>   // youtube video have different needs.
>-  rtc::Optional<bool> is_screencast;
>+  absl::optional<bool> is_screencast;
> 
>  private:
>   template <typename T>
>-  static void SetFrom(rtc::Optional<T>* s, const rtc::Optional<T>& o) {
>+  static void SetFrom(absl::optional<T>* s, const absl::optional<T>& o) {
>     if (o) {
>       *s = o;
>     }
>@@ -392,7 +170,8 @@ class MediaChannel : public sigslot::has_slots<> {
>                             const rtc::PacketOptions& options) = 0;
>     virtual bool SendRtcp(rtc::CopyOnWriteBuffer* packet,
>                           const rtc::PacketOptions& options) = 0;
>-    virtual int SetOption(SocketType type, rtc::Socket::Option opt,
>+    virtual int SetOption(SocketType type,
>+                          rtc::Socket::Option opt,
>                           int option) = 0;
>     virtual ~NetworkInterface() {}
>   };
>@@ -400,17 +179,11 @@ class MediaChannel : public sigslot::has_slots<> {
>   explicit MediaChannel(const MediaConfig& config)
>       : enable_dscp_(config.enable_dscp), network_interface_(NULL) {}
>   MediaChannel() : enable_dscp_(false), network_interface_(NULL) {}
>-  virtual ~MediaChannel() {}
>+  ~MediaChannel() override {}
> 
>   // Sets the abstract interface class for sending RTP/RTCP data.
>-  virtual void SetInterface(NetworkInterface *iface) {
>-    rtc::CritScope cs(&network_interface_crit_);
>-    network_interface_ = iface;
>-    SetDscp(enable_dscp_ ? PreferredDscp() : rtc::DSCP_DEFAULT);
>-  }
>-  virtual rtc::DiffServCodePoint PreferredDscp() const {
>-    return rtc::DSCP_DEFAULT;
>-  }
>+  virtual void SetInterface(NetworkInterface* iface);
>+  virtual rtc::DiffServCodePoint PreferredDscp() const;
>   // Called when a RTP packet is received.
>   virtual void OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
>                                 const rtc::PacketTime& packet_time) = 0;
>@@ -427,11 +200,13 @@ class MediaChannel : public sigslot::has_slots<> {
>   // by sp.
>   virtual bool AddSendStream(const StreamParams& sp) = 0;
>   // Removes an outgoing media stream.
>-  // ssrc must be the first SSRC of the media stream if the stream uses
>-  // multiple SSRCs.
>+  // SSRC must be the first SSRC of the media stream if the stream uses
>+  // multiple SSRCs. In the case of an ssrc of 0, the possibly cached
>+  // StreamParams is removed.
>   virtual bool RemoveSendStream(uint32_t ssrc) = 0;
>-  // Creates a new incoming media stream with SSRCs and CNAME as described
>-  // by sp.
>+  // Creates a new incoming media stream with SSRCs, CNAME as described
>+  // by sp. In the case of a sp without SSRCs, the unsignaled sp is cached
>+  // to be used later for unsignaled streams received.
>   virtual bool AddRecvStream(const StreamParams& sp) = 0;
>   // Removes an incoming media stream.
>   // ssrc must be the first SSRC of the media stream if the stream uses
>@@ -439,9 +214,7 @@ class MediaChannel : public sigslot::has_slots<> {
>   virtual bool RemoveRecvStream(uint32_t ssrc) = 0;
> 
>   // Returns the absoulte sendtime extension id value from media channel.
>-  virtual int GetRtpSendTimeExtnId() const {
>-    return -1;
>-  }
>+  virtual int GetRtpSendTimeExtnId() const;
> 
>   // Base method to send packet using NetworkInterface.
>   bool SendPacket(rtc::CopyOnWriteBuffer* packet,
>@@ -468,13 +241,9 @@ class MediaChannel : public sigslot::has_slots<> {
>   // This method sets DSCP |value| on both RTP and RTCP channels.
>   int SetDscp(rtc::DiffServCodePoint value) {
>     int ret;
>-    ret = SetOption(NetworkInterface::ST_RTP,
>-                    rtc::Socket::OPT_DSCP,
>-                    value);
>+    ret = SetOption(NetworkInterface::ST_RTP, rtc::Socket::OPT_DSCP, value);
>     if (ret == 0) {
>-      ret = SetOption(NetworkInterface::ST_RTCP,
>-                      rtc::Socket::OPT_DSCP,
>-                      value);
>+      ret = SetOption(NetworkInterface::ST_RTCP, rtc::Socket::OPT_DSCP, value);
>     }
>     return ret;
>   }
>@@ -516,9 +285,9 @@ struct SsrcReceiverInfo {
> };
> 
> struct MediaSenderInfo {
>-  void add_ssrc(const SsrcSenderInfo& stat) {
>-    local_stats.push_back(stat);
>-  }
>+  MediaSenderInfo();
>+  ~MediaSenderInfo();
>+  void add_ssrc(const SsrcSenderInfo& stat) { local_stats.push_back(stat); }
>   // Temporary utility function for call sites that only provide SSRC.
>   // As more info is added into SsrcSenderInfo, this function should go away.
>   void add_ssrc(uint32_t ssrc) {
>@@ -535,11 +304,15 @@ struct MediaSenderInfo {
>     }
>     return retval;
>   }
>+  // Returns true if the media has been connected.
>+  bool connected() const { return local_stats.size() > 0; }
>   // Utility accessor for clients that make the assumption only one ssrc
>   // exists per media.
>   // This will eventually go away.
>+  // Call sites that compare this to zero should use connected() instead.
>+  // https://bugs.webrtc.org/8694
>   uint32_t ssrc() const {
>-    if (local_stats.size() > 0) {
>+    if (connected()) {
>       return local_stats[0].ssrc;
>     } else {
>       return 0;
>@@ -551,15 +324,15 @@ struct MediaSenderInfo {
>   float fraction_lost = 0.0f;
>   int64_t rtt_ms = 0;
>   std::string codec_name;
>-  rtc::Optional<int> codec_payload_type;
>+  absl::optional<int> codec_payload_type;
>   std::vector<SsrcSenderInfo> local_stats;
>   std::vector<SsrcReceiverInfo> remote_stats;
> };
> 
> struct MediaReceiverInfo {
>-  void add_ssrc(const SsrcReceiverInfo& stat) {
>-    local_stats.push_back(stat);
>-  }
>+  MediaReceiverInfo();
>+  ~MediaReceiverInfo();
>+  void add_ssrc(const SsrcReceiverInfo& stat) { local_stats.push_back(stat); }
>   // Temporary utility function for call sites that only provide SSRC.
>   // As more info is added into SsrcSenderInfo, this function should go away.
>   void add_ssrc(uint32_t ssrc) {
>@@ -575,11 +348,15 @@ struct MediaReceiverInfo {
>     }
>     return retval;
>   }
>+  // Returns true if the media has been connected.
>+  bool connected() const { return local_stats.size() > 0; }
>   // Utility accessor for clients that make the assumption only one ssrc
>   // exists per media.
>   // This will eventually go away.
>+  // Call sites that compare this to zero should use connected();
>+  // https://bugs.webrtc.org/8694
>   uint32_t ssrc() const {
>-    if (local_stats.size() > 0) {
>+    if (connected()) {
>       return local_stats[0].ssrc;
>     } else {
>       return 0;
>@@ -591,12 +368,14 @@ struct MediaReceiverInfo {
>   int packets_lost = 0;
>   float fraction_lost = 0.0f;
>   std::string codec_name;
>-  rtc::Optional<int> codec_payload_type;
>+  absl::optional<int> codec_payload_type;
>   std::vector<SsrcReceiverInfo> local_stats;
>   std::vector<SsrcSenderInfo> remote_stats;
> };
> 
> struct VoiceSenderInfo : public MediaSenderInfo {
>+  VoiceSenderInfo();
>+  ~VoiceSenderInfo();
>   int ext_seqnum = 0;
>   int jitter_ms = 0;
>   int audio_level = 0;
>@@ -618,6 +397,8 @@ struct VoiceSenderInfo : public MediaSenderInfo {
> };
> 
> struct VoiceReceiverInfo : public MediaReceiverInfo {
>+  VoiceReceiverInfo();
>+  ~VoiceReceiverInfo();
>   int ext_seqnum = 0;
>   int jitter_ms = 0;
>   int jitter_buffer_ms = 0;
>@@ -661,10 +442,11 @@ struct VoiceReceiverInfo : public MediaReceiverInfo {
> };
> 
> struct VideoSenderInfo : public MediaSenderInfo {
>+  VideoSenderInfo();
>+  ~VideoSenderInfo();
>   std::vector<SsrcGroup> ssrc_groups;
>   // TODO(hbos): Move this to |VideoMediaInfo::send_codecs|?
>   std::string encoder_implementation_name;
>-  int packets_cached = 0;
>   int firs_rcvd = 0;
>   int plis_rcvd = 0;
>   int nacks_rcvd = 0;
>@@ -673,18 +455,21 @@ struct VideoSenderInfo : public MediaSenderInfo {
>   int framerate_input = 0;
>   int framerate_sent = 0;
>   int nominal_bitrate = 0;
>-  int preferred_bitrate = 0;
>   int adapt_reason = 0;
>   int adapt_changes = 0;
>   int avg_encode_ms = 0;
>   int encode_usage_percent = 0;
>   uint32_t frames_encoded = 0;
>   bool has_entered_low_resolution = false;
>-  rtc::Optional<uint64_t> qp_sum;
>+  absl::optional<uint64_t> qp_sum;
>   webrtc::VideoContentType content_type = webrtc::VideoContentType::UNSPECIFIED;
>+  // https://w3c.github.io/webrtc-stats/#dom-rtcvideosenderstats-hugeframessent
>+  uint32_t huge_frames_sent = 0;
> };
> 
> struct VideoReceiverInfo : public MediaReceiverInfo {
>+  VideoReceiverInfo();
>+  ~VideoReceiverInfo();
>   std::vector<SsrcGroup> ssrc_groups;
>   // TODO(hbos): Move this to |VideoMediaInfo::receive_codecs|?
>   std::string decoder_implementation_name;
>@@ -704,7 +489,7 @@ struct VideoReceiverInfo : public MediaReceiverInfo {
>   uint32_t frames_received = 0;
>   uint32_t frames_decoded = 0;
>   uint32_t frames_rendered = 0;
>-  rtc::Optional<uint64_t> qp_sum;
>+  absl::optional<uint64_t> qp_sum;
>   int64_t interframe_delay_max_ms = -1;
> 
>   webrtc::VideoContentType content_type = webrtc::VideoContentType::UNSPECIFIED;
>@@ -734,7 +519,7 @@ struct VideoReceiverInfo : public MediaReceiverInfo {
> 
>   // Timing frame info: all important timestamps for a full lifetime of a
>   // single 'timing frame'.
>-  rtc::Optional<webrtc::TimingFrameInfo> timing_frame_info;
>+  absl::optional<webrtc::TimingFrameInfo> timing_frame_info;
> };
> 
> struct DataSenderInfo : public MediaSenderInfo {
>@@ -759,6 +544,8 @@ struct BandwidthEstimationInfo {
> typedef std::map<int, webrtc::RtpCodecParameters> RtpCodecParametersMap;
> 
> struct VoiceMediaInfo {
>+  VoiceMediaInfo();
>+  ~VoiceMediaInfo();
>   void Clear() {
>     senders.clear();
>     receivers.clear();
>@@ -772,6 +559,8 @@ struct VoiceMediaInfo {
> };
> 
> struct VideoMediaInfo {
>+  VideoMediaInfo();
>+  ~VideoMediaInfo();
>   void Clear() {
>     senders.clear();
>     receivers.clear();
>@@ -789,6 +578,8 @@ struct VideoMediaInfo {
> };
> 
> struct DataMediaInfo {
>+  DataMediaInfo();
>+  ~DataMediaInfo();
>   void Clear() {
>     senders.clear();
>     receivers.clear();
>@@ -803,67 +594,71 @@ struct RtcpParameters {
> 
> template <class Codec>
> struct RtpParameters {
>-  virtual std::string ToString() const {
>-    std::ostringstream ost;
>-    ost << "{";
>-    ost << "codecs: " << VectorToString(codecs) << ", ";
>-    ost << "extensions: " << VectorToString(extensions);
>-    ost << "}";
>-    return ost.str();
>-  }
>+  virtual ~RtpParameters() = default;
> 
>   std::vector<Codec> codecs;
>   std::vector<webrtc::RtpExtension> extensions;
>   // TODO(pthatcher): Add streams.
>   RtcpParameters rtcp;
>-  virtual ~RtpParameters() = default;
>-};
> 
>-// TODO(deadbeef): Rename to RtpSenderParameters, since they're intended to
>-// encapsulate all the parameters needed for an RtpSender.
>-template <class Codec>
>-struct RtpSendParameters : RtpParameters<Codec> {
>-  std::string ToString() const override {
>+  std::string ToString() const {
>     std::ostringstream ost;
>     ost << "{";
>-    ost << "codecs: " << VectorToString(this->codecs) << ", ";
>-    ost << "extensions: " << VectorToString(this->extensions) << ", ";
>-    ost << "max_bandwidth_bps: " << max_bandwidth_bps << ", ";
>+    const char* separator = "";
>+    for (const auto& entry : ToStringMap()) {
>+      ost << separator << entry.first << ": " << entry.second;
>+      separator = ", ";
>+    }
>     ost << "}";
>     return ost.str();
>   }
> 
>-  int max_bandwidth_bps = -1;
>+ protected:
>+  virtual std::map<std::string, std::string> ToStringMap() const {
>+    return {{"codecs", VectorToString(codecs)},
>+            {"extensions", VectorToString(extensions)}};
>+  }
> };
> 
>-struct AudioSendParameters : RtpSendParameters<AudioCodec> {
>-  std::string ToString() const override {
>-    std::ostringstream ost;
>-    ost << "{";
>-    ost << "codecs: " << VectorToString(this->codecs) << ", ";
>-    ost << "extensions: " << VectorToString(this->extensions) << ", ";
>-    ost << "max_bandwidth_bps: " << max_bandwidth_bps << ", ";
>-    ost << "options: " << options.ToString();
>-    ost << "}";
>-    return ost.str();
>+// TODO(deadbeef): Rename to RtpSenderParameters, since they're intended to
>+// encapsulate all the parameters needed for an RtpSender.
>+template <class Codec>
>+struct RtpSendParameters : RtpParameters<Codec> {
>+  int max_bandwidth_bps = -1;
>+  // This is the value to be sent in the MID RTP header extension (if the header
>+  // extension in included in the list of extensions).
>+  std::string mid;
>+
>+ protected:
>+  std::map<std::string, std::string> ToStringMap() const override {
>+    auto params = RtpParameters<Codec>::ToStringMap();
>+    params["max_bandwidth_bps"] = rtc::ToString(max_bandwidth_bps);
>+    params["mid"] = (mid.empty() ? "<not set>" : mid);
>+    return params;
>   }
>+};
> 
>+struct AudioSendParameters : RtpSendParameters<AudioCodec> {
>+  AudioSendParameters();
>+  ~AudioSendParameters() override;
>   AudioOptions options;
>-};
> 
>-struct AudioRecvParameters : RtpParameters<AudioCodec> {
>+ protected:
>+  std::map<std::string, std::string> ToStringMap() const override;
> };
> 
>+struct AudioRecvParameters : RtpParameters<AudioCodec> {};
>+
> class VoiceMediaChannel : public MediaChannel {
>  public:
>   VoiceMediaChannel() {}
>   explicit VoiceMediaChannel(const MediaConfig& config)
>       : MediaChannel(config) {}
>-  virtual ~VoiceMediaChannel() {}
>+  ~VoiceMediaChannel() override {}
>   virtual bool SetSendParameters(const AudioSendParameters& params) = 0;
>   virtual bool SetRecvParameters(const AudioRecvParameters& params) = 0;
>   virtual webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const = 0;
>-  virtual bool SetRtpSendParameters(
>+  virtual webrtc::RTCError SetRtpSendParameters(
>       uint32_t ssrc,
>       const webrtc::RtpParameters& parameters) = 0;
>   // Get the receive parameters for the incoming stream identified by |ssrc|.
>@@ -885,11 +680,6 @@ class VoiceMediaChannel : public MediaChannel {
>                             bool enable,
>                             const AudioOptions* options,
>                             AudioSource* source) = 0;
>-  // Gets current energy levels for all incoming streams.
>-  typedef std::vector<std::pair<uint32_t, int>> StreamList;
>-  virtual bool GetActiveStreams(StreamList* actives) = 0;
>-  // Get the current energy level of the stream sent to the speaker.
>-  virtual int GetOutputLevel() = 0;
>   // Set speaker output volume of the specified ssrc.
>   virtual bool SetOutputVolume(uint32_t ssrc, double volume) = 0;
>   // Returns if the telephone-event has been negotiated.
>@@ -912,6 +702,8 @@ class VoiceMediaChannel : public MediaChannel {
> // TODO(deadbeef): Rename to VideoSenderParameters, since they're intended to
> // encapsulate all the parameters needed for a video RtpSender.
> struct VideoSendParameters : RtpSendParameters<VideoCodec> {
>+  VideoSendParameters();
>+  ~VideoSendParameters() override;
>   // Use conference mode? This flag comes from the remote
>   // description's SDP line 'a=x-google-flag:conference', copied over
>   // by VideoChannel::SetRemoteContent_w, and ultimately used by
>@@ -919,24 +711,26 @@ struct VideoSendParameters : RtpSendParameters<VideoCodec> {
>   // WebRtcVideoChannel::WebRtcVideoSendStream::CreateVideoEncoderConfig.
>   // The special screencast behaviour is disabled by default.
>   bool conference_mode = false;
>+
>+ protected:
>+  std::map<std::string, std::string> ToStringMap() const override;
> };
> 
> // TODO(deadbeef): Rename to VideoReceiverParameters, since they're intended to
> // encapsulate all the parameters needed for a video RtpReceiver.
>-struct VideoRecvParameters : RtpParameters<VideoCodec> {
>-};
>+struct VideoRecvParameters : RtpParameters<VideoCodec> {};
> 
> class VideoMediaChannel : public MediaChannel {
>  public:
>   VideoMediaChannel() {}
>   explicit VideoMediaChannel(const MediaConfig& config)
>       : MediaChannel(config) {}
>-  virtual ~VideoMediaChannel() {}
>+  ~VideoMediaChannel() override {}
> 
>   virtual bool SetSendParameters(const VideoSendParameters& params) = 0;
>   virtual bool SetRecvParameters(const VideoRecvParameters& params) = 0;
>   virtual webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const = 0;
>-  virtual bool SetRtpSendParameters(
>+  virtual webrtc::RTCError SetRtpSendParameters(
>       uint32_t ssrc,
>       const webrtc::RtpParameters& parameters) = 0;
>   // Get the receive parameters for the incoming stream identified by |ssrc|.
>@@ -957,7 +751,6 @@ class VideoMediaChannel : public MediaChannel {
>   // The |ssrc| must correspond to a registered send stream.
>   virtual bool SetVideoSend(
>       uint32_t ssrc,
>-      bool enable,
>       const VideoOptions* options,
>       rtc::VideoSourceInterface<webrtc::VideoFrame>* source) = 0;
>   // Sets the sink object to be used for the specified stream.
>@@ -1033,47 +826,34 @@ struct SendDataParams {
> 
> enum SendDataResult { SDR_SUCCESS, SDR_ERROR, SDR_BLOCK };
> 
>-struct DataSendParameters : RtpSendParameters<DataCodec> {
>-  std::string ToString() const {
>-    std::ostringstream ost;
>-    // Options and extensions aren't used.
>-    ost << "{";
>-    ost << "codecs: " << VectorToString(codecs) << ", ";
>-    ost << "max_bandwidth_bps: " << max_bandwidth_bps;
>-    ost << "}";
>-    return ost.str();
>-  }
>-};
>+struct DataSendParameters : RtpSendParameters<DataCodec> {};
> 
>-struct DataRecvParameters : RtpParameters<DataCodec> {
>-};
>+struct DataRecvParameters : RtpParameters<DataCodec> {};
> 
> class DataMediaChannel : public MediaChannel {
>  public:
>-  DataMediaChannel() {}
>-  explicit DataMediaChannel(const MediaConfig& config) : MediaChannel(config) {}
>-  virtual ~DataMediaChannel() {}
>+  DataMediaChannel();
>+  explicit DataMediaChannel(const MediaConfig& config);
>+  ~DataMediaChannel() override;
> 
>   virtual bool SetSendParameters(const DataSendParameters& params) = 0;
>   virtual bool SetRecvParameters(const DataRecvParameters& params) = 0;
> 
>   // TODO(pthatcher): Implement this.
>-  virtual bool GetStats(DataMediaInfo*) { return true; }
>+  virtual bool GetStats(DataMediaInfo* info);
> 
>   virtual bool SetSend(bool send) = 0;
>   virtual bool SetReceive(bool receive) = 0;
> 
>-  virtual void OnNetworkRouteChanged(const std::string&,
>-                                     const rtc::NetworkRoute&) {}
>+  void OnNetworkRouteChanged(const std::string& transport_name,
>+                             const rtc::NetworkRoute& network_route) override {}
> 
>-  virtual bool SendData(
>-      const SendDataParams& params,
>-      const rtc::CopyOnWriteBuffer& payload,
>-      SendDataResult* result = NULL) = 0;
>+  virtual bool SendData(const SendDataParams& params,
>+                        const rtc::CopyOnWriteBuffer& payload,
>+                        SendDataResult* result = NULL) = 0;
>   // Signals when data is received (params, data, len)
>-  sigslot::signal3<const ReceiveDataParams&,
>-                   const char*,
>-                   size_t> SignalDataReceived;
>+  sigslot::signal3<const ReceiveDataParams&, const char*, size_t>
>+      SignalDataReceived;
>   // Signal when the media channel is ready to send the stream. Arguments are:
>   //     writable(bool)
>   sigslot::signal1<bool> SignalReadyToSend;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediaconfig.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediaconfig.h
>new file mode 100644
>index 00000000000..eda387e3191
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediaconfig.h
>@@ -0,0 +1,82 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MEDIA_BASE_MEDIACONFIG_H_
>+#define MEDIA_BASE_MEDIACONFIG_H_
>+
>+namespace cricket {
>+
>+// Construction-time settings, passed on when creating
>+// MediaChannels.
>+struct MediaConfig {
>+  // Set DSCP value on packets. This flag comes from the
>+  // PeerConnection constraint 'googDscp'.
>+  bool enable_dscp = false;
>+
>+  // Video-specific config.
>+  struct Video {
>+    // Enable WebRTC CPU Overuse Detection. This flag comes from the
>+    // PeerConnection constraint 'googCpuOveruseDetection'.
>+    bool enable_cpu_adaptation = true;
>+
>+    // Enable WebRTC suspension of video. No video frames will be sent
>+    // when the bitrate is below the configured minimum bitrate. This
>+    // flag comes from the PeerConnection constraint
>+    // 'googSuspendBelowMinBitrate', and WebRtcVideoChannel copies it
>+    // to VideoSendStream::Config::suspend_below_min_bitrate.
>+    bool suspend_below_min_bitrate = false;
>+
>+    // Set to true if the renderer has an algorithm of frame selection.
>+    // If the value is true, then WebRTC will hand over a frame as soon as
>+    // possible without delay, and rendering smoothness is completely the duty
>+    // of the renderer;
>+    // If the value is false, then WebRTC is responsible to delay frame release
>+    // in order to increase rendering smoothness.
>+    //
>+    // This flag comes from PeerConnection's RtcConfiguration, but is
>+    // currently only set by the command line flag
>+    // 'disable-rtc-smoothness-algorithm'.
>+    // WebRtcVideoChannel::AddRecvStream copies it to the created
>+    // WebRtcVideoReceiveStream, where it is returned by the
>+    // SmoothsRenderedFrames method. This method is used by the
>+    // VideoReceiveStream, where the value is passed on to the
>+    // IncomingVideoStream constructor.
>+    bool enable_prerenderer_smoothing = true;
>+
>+    // Enables periodic bandwidth probing in application-limited region.
>+    bool periodic_alr_bandwidth_probing = false;
>+
>+    // Enables the new method to estimate the cpu load from encoding, used for
>+    // cpu adaptation. This flag is intended to be controlled primarily by a
>+    // Chrome origin-trial.
>+    // TODO(bugs.webrtc.org/8504): If all goes well, the flag will be removed
>+    // together with the old method of estimation.
>+    bool experiment_cpu_load_estimator = false;
>+  } video;
>+
>+  bool operator==(const MediaConfig& o) const {
>+    return enable_dscp == o.enable_dscp &&
>+           video.enable_cpu_adaptation == o.video.enable_cpu_adaptation &&
>+           video.suspend_below_min_bitrate ==
>+               o.video.suspend_below_min_bitrate &&
>+           video.enable_prerenderer_smoothing ==
>+               o.video.enable_prerenderer_smoothing &&
>+           video.periodic_alr_bandwidth_probing ==
>+               o.video.periodic_alr_bandwidth_probing &&
>+           video.experiment_cpu_load_estimator ==
>+               o.video.experiment_cpu_load_estimator;
>+  }
>+
>+  bool operator!=(const MediaConfig& o) const { return !(*this == o); }
>+};
>+
>+}  // namespace cricket
>+
>+#endif  // MEDIA_BASE_MEDIACONFIG_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediaconstants.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediaconstants.cc
>index 9496d9f60ca..522adf3c699 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediaconstants.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediaconstants.cc
>@@ -25,7 +25,7 @@ const float kProcessCpuThreshold = 0.10f;
> const char kRtxCodecName[] = "rtx";
> const char kRedCodecName[] = "red";
> const char kUlpfecCodecName[] = "ulpfec";
>-const char kStereoCodecName[] = "stereo";
>+const char kMultiplexCodecName[] = "multiplex";
> 
> // TODO(brandtr): Change this to 'flexfec' when we are confident that the
> // header format is not changing anymore.
>@@ -39,12 +39,12 @@ const char kCodecParamAssociatedCodecName[] = "acn";
> 
> const char kOpusCodecName[] = "opus";
> const char kIsacCodecName[] = "ISAC";
>-const char kL16CodecName[]  = "L16";
>+const char kL16CodecName[] = "L16";
> const char kG722CodecName[] = "G722";
> const char kIlbcCodecName[] = "ILBC";
> const char kPcmuCodecName[] = "PCMU";
> const char kPcmaCodecName[] = "PCMA";
>-const char kCnCodecName[]   = "CN";
>+const char kCnCodecName[] = "CN";
> const char kDtmfCodecName[] = "telephone-event";
> 
> // draft-spittka-payload-rtp-opus-03.txt
>@@ -86,6 +86,7 @@ const char kRtcpFbParamTransportCc[] = "transport-cc";
> 
> const char kRtcpFbParamCcm[] = "ccm";
> const char kRtcpFbCcmParamFir[] = "fir";
>+const char kRtcpFbParamRrtr[] = "rrtr";
> const char kCodecParamMaxBitrate[] = "x-google-max-bitrate";
> const char kCodecParamMinBitrate[] = "x-google-min-bitrate";
> const char kCodecParamStartBitrate[] = "x-google-start-bitrate";
>@@ -112,4 +113,8 @@ const char kH264FmtpSpropParameterSets[] = "sprop-parameter-sets";
> const char kH264ProfileLevelConstrainedBaseline[] = "42e01f";
> 
> const int kDefaultVideoMaxFramerate = 60;
>+
>+const size_t kConferenceMaxNumSpatialLayers = 3;
>+const size_t kConferenceMaxNumTemporalLayers = 3;
>+const size_t kConferenceDefaultNumTemporalLayers = 3;
> }  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediaconstants.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediaconstants.h
>index 58a66d917a8..14c49baf72c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediaconstants.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediaconstants.h
>@@ -30,7 +30,7 @@ extern const char kRtxCodecName[];
> extern const char kRedCodecName[];
> extern const char kUlpfecCodecName[];
> extern const char kFlexfecCodecName[];
>-extern const char kStereoCodecName[];
>+extern const char kMultiplexCodecName[];
> 
> extern const char kFlexfecFmtpRepairWindow[];
> 
>@@ -101,6 +101,9 @@ extern const char kRtcpFbParamTransportCc[];
> // ccm submessages according to RFC 5104
> extern const char kRtcpFbParamCcm[];
> extern const char kRtcpFbCcmParamFir[];
>+// Receiver reference time report
>+// https://tools.ietf.org/html/rfc3611 section 4.4
>+extern const char kRtcpFbParamRrtr[];
> // Google specific parameters
> extern const char kCodecParamMaxBitrate[];
> extern const char kCodecParamMinBitrate[];
>@@ -134,6 +137,10 @@ extern const char kH264FmtpSpropParameterSets[];
> extern const char kH264ProfileLevelConstrainedBaseline[];
> 
> extern const int kDefaultVideoMaxFramerate;
>+
>+extern const size_t kConferenceMaxNumSpatialLayers;
>+extern const size_t kConferenceMaxNumTemporalLayers;
>+extern const size_t kConferenceDefaultNumTemporalLayers;
> }  // namespace cricket
> 
> #endif  // MEDIA_BASE_MEDIACONSTANTS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediaengine.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediaengine.cc
>index 281ddbb76f7..b0fede3a2a5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediaengine.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediaengine.cc
>@@ -10,25 +10,10 @@
> 
> #include "media/base/mediaengine.h"
> 
>-#if !defined(DISABLE_MEDIA_ENGINE_FACTORY)
>-
> namespace cricket {
> 
>-MediaEngineFactory::MediaEngineCreateFunction
>-    MediaEngineFactory::create_function_ = NULL;
>-
>-MediaEngineFactory::MediaEngineCreateFunction
>-    MediaEngineFactory::SetCreateFunction(MediaEngineCreateFunction function) {
>-  MediaEngineCreateFunction old_function = create_function_;
>-  create_function_ = function;
>-  return old_function;
>-}
>-
>-};  // namespace cricket
>-
>-#endif  // DISABLE_MEDIA_ENGINE_FACTORY
>-
>-namespace cricket {
>+RtpCapabilities::RtpCapabilities() = default;
>+RtpCapabilities::~RtpCapabilities() = default;
> 
> webrtc::RtpParameters CreateRtpParametersWithOneEncoding() {
>   webrtc::RtpParameters parameters;
>@@ -37,4 +22,19 @@ webrtc::RtpParameters CreateRtpParametersWithOneEncoding() {
>   return parameters;
> }
> 
>+webrtc::RtpParameters CreateRtpParametersWithEncodings(StreamParams sp) {
>+  std::vector<uint32_t> primary_ssrcs;
>+  sp.GetPrimarySsrcs(&primary_ssrcs);
>+  size_t encoding_count = primary_ssrcs.size();
>+
>+  std::vector<webrtc::RtpEncodingParameters> encodings(encoding_count);
>+  for (size_t i = 0; i < encodings.size(); ++i) {
>+    encodings[i].ssrc = primary_ssrcs[i];
>+  }
>+  webrtc::RtpParameters parameters;
>+  parameters.encodings = encodings;
>+  parameters.rtcp.cname = sp.cname;
>+  return parameters;
>+}
>+
> };  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediaengine.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediaengine.h
>index 483a96b7e89..57979c27574 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediaengine.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediaengine.h
>@@ -29,20 +29,18 @@
> #include "media/base/videocommon.h"
> #include "rtc_base/platform_file.h"
> 
>-#if defined(GOOGLE_CHROME_BUILD) || defined(CHROMIUM_BUILD)
>-#define DISABLE_MEDIA_ENGINE_FACTORY
>-#endif
>-
> namespace webrtc {
> class AudioDeviceModule;
> class AudioMixer;
> class AudioProcessing;
> class Call;
>-}
>+}  // namespace webrtc
> 
> namespace cricket {
> 
> struct RtpCapabilities {
>+  RtpCapabilities();
>+  ~RtpCapabilities();
>   std::vector<webrtc::RtpExtension> header_extensions;
> };
> 
>@@ -72,9 +70,6 @@ class MediaEngineInterface {
>       const MediaConfig& config,
>       const VideoOptions& options) = 0;
> 
>-  // Gets the current microphone level, as a value between 0 and 10.
>-  virtual int GetInputLevel() = 0;
>-
>   virtual const std::vector<AudioCodec>& audio_send_codecs() = 0;
>   virtual const std::vector<AudioCodec>& audio_recv_codecs() = 0;
>   virtual RtpCapabilities GetAudioCapabilities() = 0;
>@@ -90,25 +85,6 @@ class MediaEngineInterface {
>   virtual void StopAecDump() = 0;
> };
> 
>-
>-#if !defined(DISABLE_MEDIA_ENGINE_FACTORY)
>-class MediaEngineFactory {
>- public:
>-  typedef cricket::MediaEngineInterface* (*MediaEngineCreateFunction)();
>-  // Creates a media engine, using either the compiled system default or the
>-  // creation function specified in SetCreateFunction, if specified.
>-  static MediaEngineInterface* Create();
>-  // Sets the function used when calling Create. If unset, the compiled system
>-  // default will be used. Returns the old create function, or NULL if one
>-  // wasn't set. Likewise, NULL can be used as the |function| parameter to
>-  // reset to the default behavior.
>-  static MediaEngineCreateFunction SetCreateFunction(
>-      MediaEngineCreateFunction function);
>- private:
>-  static MediaEngineCreateFunction create_function_;
>-};
>-#endif
>-
> // CompositeMediaEngine constructs a MediaEngine from separate
> // voice and video engine classes.
> template <class VOICE, class VIDEO>
>@@ -141,7 +117,6 @@ class CompositeMediaEngine : public MediaEngineInterface {
>     return video().CreateChannel(call, config, options);
>   }
> 
>-  virtual int GetInputLevel() { return voice().GetInputLevel(); }
>   virtual const std::vector<AudioCodec>& audio_send_codecs() {
>     return voice().send_codecs();
>   }
>@@ -182,6 +157,7 @@ class DataEngineInterface {
> };
> 
> webrtc::RtpParameters CreateRtpParametersWithOneEncoding();
>+webrtc::RtpParameters CreateRtpParametersWithEncodings(StreamParams sp);
> 
> }  // namespace cricket
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/rtpdataengine.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/rtpdataengine.cc
>index 7cb5fa85859..fc2b73141ed 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/rtpdataengine.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/rtpdataengine.cc
>@@ -17,9 +17,9 @@
> #include "media/base/rtputils.h"
> #include "media/base/streamparams.h"
> #include "rtc_base/copyonwritebuffer.h"
>+#include "rtc_base/data_rate_limiter.h"
> #include "rtc_base/helpers.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/ratelimiter.h"
> #include "rtc_base/sanitizer.h"
> #include "rtc_base/stringutils.h"
> 
>@@ -28,9 +28,7 @@ namespace cricket {
> // We want to avoid IP fragmentation.
> static const size_t kDataMaxRtpPacketLen = 1200U;
> // We reserve space after the RTP header for future wiggle room.
>-static const unsigned char kReservedSpace[] = {
>-  0x00, 0x00, 0x00, 0x00
>-};
>+static const unsigned char kReservedSpace[] = {0x00, 0x00, 0x00, 0x00};
> 
> // Amount of overhead SRTP may take.  We need to leave room in the
> // buffer for it, otherwise SRTP will fail later.  If SRTP ever uses
>@@ -42,8 +40,7 @@ RtpDataEngine::RtpDataEngine() {
>       DataCodec(kGoogleRtpDataCodecPlType, kGoogleRtpDataCodecName));
> }
> 
>-DataMediaChannel* RtpDataEngine::CreateChannel(
>-    const MediaConfig& config) {
>+DataMediaChannel* RtpDataEngine::CreateChannel(const MediaConfig& config) {
>   return new RtpDataMediaChannel(config);
> }
> 
>@@ -64,21 +61,19 @@ RtpDataMediaChannel::RtpDataMediaChannel(const MediaConfig& config)
> void RtpDataMediaChannel::Construct() {
>   sending_ = false;
>   receiving_ = false;
>-  send_limiter_.reset(new rtc::RateLimiter(kDataMaxBandwidth / 8, 1.0));
>+  send_limiter_.reset(new rtc::DataRateLimiter(kDataMaxBandwidth / 8, 1.0));
> }
> 
>-
> RtpDataMediaChannel::~RtpDataMediaChannel() {
>   std::map<uint32_t, RtpClock*>::const_iterator iter;
>   for (iter = rtp_clock_by_send_ssrc_.begin();
>-       iter != rtp_clock_by_send_ssrc_.end();
>-       ++iter) {
>+       iter != rtp_clock_by_send_ssrc_.end(); ++iter) {
>     delete iter->second;
>   }
> }
> 
> void RTC_NO_SANITIZE("float-cast-overflow")  // bugs.webrtc.org/8204
>-RtpClock::Tick(double now, int* seq_num, uint32_t* timestamp) {
>+    RtpClock::Tick(double now, int* seq_num, uint32_t* timestamp) {
>   *seq_num = ++last_seq_num_;
>   *timestamp = timestamp_offset_ + static_cast<uint32_t>(now * clockrate_);
>   // UBSan: 5.92374e+10 is outside the range of representable values of type
>@@ -155,9 +150,9 @@ bool RtpDataMediaChannel::AddSendStream(const StreamParams& stream) {
>   send_streams_.push_back(stream);
>   // TODO(pthatcher): This should be per-stream, not per-ssrc.
>   // And we should probably allow more than one per stream.
>-  rtp_clock_by_send_ssrc_[stream.first_ssrc()] = new RtpClock(
>-      kDataCodecClockrate,
>-      rtc::CreateRandomNonZeroId(), rtc::CreateRandomNonZeroId());
>+  rtp_clock_by_send_ssrc_[stream.first_ssrc()] =
>+      new RtpClock(kDataCodecClockrate, rtc::CreateRandomNonZeroId(),
>+                   rtc::CreateRandomNonZeroId());
> 
>   RTC_LOG(LS_INFO) << "Added data send stream '" << stream.id
>                    << "' with ssrc=" << stream.first_ssrc();
>@@ -198,22 +193,15 @@ bool RtpDataMediaChannel::RemoveRecvStream(uint32_t ssrc) {
>   return true;
> }
> 
>-void RtpDataMediaChannel::OnPacketReceived(
>-    rtc::CopyOnWriteBuffer* packet, const rtc::PacketTime& packet_time) {
>+void RtpDataMediaChannel::OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
>+                                           const rtc::PacketTime& packet_time) {
>   RtpHeader header;
>   if (!GetRtpHeader(packet->cdata(), packet->size(), &header)) {
>-    // Don't want to log for every corrupt packet.
>-    // RTC_LOG(LS_WARNING) << "Could not read rtp header from packet of length "
>-    //                 << packet->length() << ".";
>     return;
>   }
> 
>   size_t header_length;
>   if (!GetRtpHeaderLen(packet->cdata(), packet->size(), &header_length)) {
>-    // Don't want to log for every corrupt packet.
>-    // RTC_LOG(LS_WARNING) << "Could not read rtp header"
>-    //                 << length from packet of length "
>-    //                 << packet->length() << ".";
>     return;
>   }
>   const char* data =
>@@ -227,12 +215,6 @@ void RtpDataMediaChannel::OnPacketReceived(
>   }
> 
>   if (!FindCodecById(recv_codecs_, header.payload_type)) {
>-    // For bundling, this will be logged for every message.
>-    // So disable this logging.
>-    // RTC_LOG(LS_WARNING) << "Not receiving packet "
>-    //                << header.ssrc << ":" << header.seq_num
>-    //                << " (" << data_len << ")"
>-    //                << " because unknown payload id: " << header.payload_type;
>     return;
>   }
> 
>@@ -261,16 +243,15 @@ bool RtpDataMediaChannel::SetMaxSendBandwidth(int bps) {
>   if (bps <= 0) {
>     bps = kDataMaxBandwidth;
>   }
>-  send_limiter_.reset(new rtc::RateLimiter(bps / 8, 1.0));
>+  send_limiter_.reset(new rtc::DataRateLimiter(bps / 8, 1.0));
>   RTC_LOG(LS_INFO) << "RtpDataMediaChannel::SetSendBandwidth to " << bps
>                    << "bps.";
>   return true;
> }
> 
>-bool RtpDataMediaChannel::SendData(
>-    const SendDataParams& params,
>-    const rtc::CopyOnWriteBuffer& payload,
>-    SendDataResult* result) {
>+bool RtpDataMediaChannel::SendData(const SendDataParams& params,
>+                                   const rtc::CopyOnWriteBuffer& payload,
>+                                   SendDataResult* result) {
>   if (result) {
>     // If we return true, we'll set this to SDR_SUCCESS.
>     *result = SDR_ERROR;
>@@ -323,8 +304,8 @@ bool RtpDataMediaChannel::SendData(
>   RtpHeader header;
>   header.payload_type = found_codec->id;
>   header.ssrc = params.ssrc;
>-  rtp_clock_by_send_ssrc_[header.ssrc]->Tick(
>-      now, &header.seq_num, &header.timestamp);
>+  rtp_clock_by_send_ssrc_[header.ssrc]->Tick(now, &header.seq_num,
>+                                             &header.timestamp);
> 
>   rtc::CopyOnWriteBuffer packet(kMinRtpPacketLen, packet_len);
>   if (!SetRtpHeader(packet.data(), packet.size(), header)) {
>@@ -340,7 +321,9 @@ bool RtpDataMediaChannel::SendData(
>                       << ", timestamp=" << header.timestamp
>                       << ", len=" << payload.size();
> 
>-  MediaChannel::SendPacket(&packet, rtc::PacketOptions());
>+  rtc::PacketOptions options;
>+  options.info_signaled_after_sent.packet_type = rtc::PacketType::kData;
>+  MediaChannel::SendPacket(&packet, options);
>   send_limiter_->Use(packet_len, now);
>   if (result) {
>     *result = SDR_SUCCESS;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/rtpdataengine.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/rtpdataengine.h
>index 64e083b0fd9..966d0c81514 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/rtpdataengine.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/rtpdataengine.h
>@@ -20,6 +20,10 @@
> #include "media/base/mediaconstants.h"
> #include "media/base/mediaengine.h"
> 
>+namespace rtc {
>+class DataRateLimiter;
>+}
>+
> namespace cricket {
> 
> struct DataCodec;
>@@ -30,9 +34,7 @@ class RtpDataEngine : public DataEngineInterface {
> 
>   virtual DataMediaChannel* CreateChannel(const MediaConfig& config);
> 
>-  virtual const std::vector<DataCodec>& data_codecs() {
>-    return data_codecs_;
>-  }
>+  virtual const std::vector<DataCodec>& data_codecs() { return data_codecs_; }
> 
>  private:
>   std::vector<DataCodec> data_codecs_;
>@@ -84,10 +86,9 @@ class RtpDataMediaChannel : public DataMediaChannel {
>   virtual void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet,
>                               const rtc::PacketTime& packet_time) {}
>   virtual void OnReadyToSend(bool ready) {}
>-  virtual bool SendData(
>-    const SendDataParams& params,
>-    const rtc::CopyOnWriteBuffer& payload,
>-    SendDataResult* result);
>+  virtual bool SendData(const SendDataParams& params,
>+                        const rtc::CopyOnWriteBuffer& payload,
>+                        SendDataResult* result);
>   virtual rtc::DiffServCodePoint PreferredDscp() const;
> 
>  private:
>@@ -103,7 +104,7 @@ class RtpDataMediaChannel : public DataMediaChannel {
>   std::vector<StreamParams> send_streams_;
>   std::vector<StreamParams> recv_streams_;
>   std::map<uint32_t, RtpClock*> rtp_clock_by_send_ssrc_;
>-  std::unique_ptr<rtc::RateLimiter> send_limiter_;
>+  std::unique_ptr<rtc::DataRateLimiter> send_limiter_;
> };
> 
> }  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/rtpdataengine_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/rtpdataengine_unittest.cc
>index a05c3de2649..c15c55f5e62 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/rtpdataengine_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/rtpdataengine_unittest.cc
>@@ -24,9 +24,9 @@ class FakeDataReceiver : public sigslot::has_slots<> {
>  public:
>   FakeDataReceiver() : has_received_data_(false) {}
> 
>-  void OnDataReceived(
>-      const cricket::ReceiveDataParams& params,
>-      const char* data, size_t len) {
>+  void OnDataReceived(const cricket::ReceiveDataParams& params,
>+                      const char* data,
>+                      size_t len) {
>     has_received_data_ = true;
>     last_received_data_ = std::string(data, len);
>     last_received_data_len_ = len;
>@@ -74,34 +74,24 @@ class RtpDataMediaChannelTest : public testing::Test {
>     cricket::RtpDataMediaChannel* channel =
>         static_cast<cricket::RtpDataMediaChannel*>(dme->CreateChannel(config));
>     channel->SetInterface(iface_.get());
>-    channel->SignalDataReceived.connect(
>-        receiver_.get(), &FakeDataReceiver::OnDataReceived);
>+    channel->SignalDataReceived.connect(receiver_.get(),
>+                                        &FakeDataReceiver::OnDataReceived);
>     return channel;
>   }
> 
>-  FakeDataReceiver* receiver() {
>-    return receiver_.get();
>-  }
>+  FakeDataReceiver* receiver() { return receiver_.get(); }
> 
>-  bool HasReceivedData() {
>-    return receiver_->has_received_data();
>-  }
>+  bool HasReceivedData() { return receiver_->has_received_data(); }
> 
>-  std::string GetReceivedData() {
>-    return receiver_->last_received_data();
>-  }
>+  std::string GetReceivedData() { return receiver_->last_received_data(); }
> 
>-  size_t GetReceivedDataLen() {
>-    return receiver_->last_received_data_len();
>-  }
>+  size_t GetReceivedDataLen() { return receiver_->last_received_data_len(); }
> 
>   cricket::ReceiveDataParams GetReceivedDataParams() {
>     return receiver_->last_received_data_params();
>   }
> 
>-  bool HasSentData(int count) {
>-    return (iface_->NumRtpPackets() > count);
>-  }
>+  bool HasSentData(int count) { return (iface_->NumRtpPackets() > count); }
> 
>   std::string GetSentData(int index) {
>     // Assume RTP header of length 12
>@@ -202,8 +192,7 @@ TEST_F(RtpDataMediaChannelTest, SendData) {
>   unsigned char data[] = "food";
>   rtc::CopyOnWriteBuffer payload(data, 4);
>   unsigned char padded_data[] = {
>-    0x00, 0x00, 0x00, 0x00,
>-    'f', 'o', 'o', 'd',
>+      0x00, 0x00, 0x00, 0x00, 'f', 'o', 'o', 'd',
>   };
>   cricket::SendDataResult result;
> 
>@@ -246,8 +235,7 @@ TEST_F(RtpDataMediaChannelTest, SendData) {
>   EXPECT_EQ(cricket::SDR_SUCCESS, result);
>   ASSERT_TRUE(HasSentData(0));
>   EXPECT_EQ(sizeof(padded_data), GetSentData(0).length());
>-  EXPECT_EQ(0, memcmp(
>-      padded_data, GetSentData(0).data(), sizeof(padded_data)));
>+  EXPECT_EQ(0, memcmp(padded_data, GetSentData(0).data(), sizeof(padded_data)));
>   cricket::RtpHeader header0 = GetSentDataHeader(0);
>   EXPECT_NE(0, header0.seq_num);
>   EXPECT_NE(0U, header0.timestamp);
>@@ -260,8 +248,7 @@ TEST_F(RtpDataMediaChannelTest, SendData) {
>   EXPECT_TRUE(dmc->SendData(params, payload, &result));
>   ASSERT_TRUE(HasSentData(1));
>   EXPECT_EQ(sizeof(padded_data), GetSentData(1).length());
>-  EXPECT_EQ(0, memcmp(
>-      padded_data, GetSentData(1).data(), sizeof(padded_data)));
>+  EXPECT_EQ(0, memcmp(padded_data, GetSentData(1).data(), sizeof(padded_data)));
>   cricket::RtpHeader header1 = GetSentDataHeader(1);
>   EXPECT_EQ(header1.ssrc, 42U);
>   EXPECT_EQ(header1.payload_type, 103);
>@@ -322,11 +309,9 @@ TEST_F(RtpDataMediaChannelTest, SendDataRate) {
> 
> TEST_F(RtpDataMediaChannelTest, ReceiveData) {
>   // PT= 103, SN=2, TS=3, SSRC = 4, data = "abcde"
>-  unsigned char data[] = {
>-    0x80, 0x67, 0x00, 0x02, 0x00, 0x00, 0x00, 0x03, 0x00, 0x00, 0x00, 0x2A,
>-    0x00, 0x00, 0x00, 0x00,
>-    'a', 'b', 'c', 'd', 'e'
>-  };
>+  unsigned char data[] = {0x80, 0x67, 0x00, 0x02, 0x00, 0x00, 0x00,
>+                          0x03, 0x00, 0x00, 0x00, 0x2A, 0x00, 0x00,
>+                          0x00, 0x00, 'a',  'b',  'c',  'd',  'e'};
>   rtc::CopyOnWriteBuffer packet(data, sizeof(data));
> 
>   std::unique_ptr<cricket::RtpDataMediaChannel> dmc(CreateChannel());
>@@ -364,9 +349,7 @@ TEST_F(RtpDataMediaChannelTest, ReceiveData) {
> }
> 
> TEST_F(RtpDataMediaChannelTest, InvalidRtpPackets) {
>-  unsigned char data[] = {
>-    0x80, 0x65, 0x00, 0x02
>-  };
>+  unsigned char data[] = {0x80, 0x65, 0x00, 0x02};
>   rtc::CopyOnWriteBuffer packet(data, sizeof(data));
> 
>   std::unique_ptr<cricket::RtpDataMediaChannel> dmc(CreateChannel());
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/rtputils.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/rtputils.cc
>index d0ba1cf72b0..ee34cb101e6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/rtputils.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/rtputils.cc
>@@ -35,9 +35,8 @@ namespace {
> // Fake auth tag written by the sender when external authentication is enabled.
> // HMAC in packet will be compared against this value before updating packet
> // with actual HMAC value.
>-static const uint8_t kFakeAuthTag[10] = {
>-    0xba, 0xdd, 0xba, 0xdd, 0xba, 0xdd, 0xba, 0xdd, 0xba, 0xdd
>-};
>+static const uint8_t kFakeAuthTag[10] = {0xba, 0xdd, 0xba, 0xdd, 0xba,
>+                                         0xdd, 0xba, 0xdd, 0xba, 0xdd};
> 
> void UpdateAbsSendTimeExtensionValue(uint8_t* extension_data,
>                                      size_t length,
>@@ -99,10 +98,10 @@ void UpdateRtpAuthTag(uint8_t* rtp,
>   size_t auth_required_length = length - tag_length + kRocLength;
> 
>   uint8_t output[64];
>-  size_t result = rtc::ComputeHmac(
>-      rtc::DIGEST_SHA_1, &packet_time_params.srtp_auth_key[0],
>-      packet_time_params.srtp_auth_key.size(), rtp,
>-      auth_required_length, output, sizeof(output));
>+  size_t result =
>+      rtc::ComputeHmac(rtc::DIGEST_SHA_1, &packet_time_params.srtp_auth_key[0],
>+                       packet_time_params.srtp_auth_key.size(), rtp,
>+                       auth_required_length, output, sizeof(output));
> 
>   if (result < tag_length) {
>     RTC_NOTREACHED();
>@@ -205,18 +204,21 @@ bool GetRtpSsrc(const void* data, size_t len, uint32_t* value) {
> }
> 
> bool GetRtpHeaderLen(const void* data, size_t len, size_t* value) {
>-  if (!data || len < kMinRtpPacketLen || !value) return false;
>+  if (!data || len < kMinRtpPacketLen || !value)
>+    return false;
>   const uint8_t* header = static_cast<const uint8_t*>(data);
>   // Get base header size + length of CSRCs (not counting extension yet).
>   size_t header_size = kMinRtpPacketLen + (header[0] & 0xF) * sizeof(uint32_t);
>-  if (len < header_size) return false;
>+  if (len < header_size)
>+    return false;
>   // If there's an extension, read and add in the extension size.
>   if (header[0] & 0x10) {
>     if (len < header_size + sizeof(uint32_t))
>       return false;
>     header_size +=
>         ((rtc::GetBE16(header + header_size + 2) + 1) * sizeof(uint32_t));
>-    if (len < header_size) return false;
>+    if (len < header_size)
>+      return false;
>   }
>   *value = header_size;
>   return true;
>@@ -241,11 +243,14 @@ bool GetRtcpType(const void* data, size_t len, int* value) {
> // to send non-compound packets only to feedback messages.
> bool GetRtcpSsrc(const void* data, size_t len, uint32_t* value) {
>   // Packet should be at least of 8 bytes, to get SSRC from a RTCP packet.
>-  if (!data || len < kMinRtcpPacketLen + 4 || !value) return false;
>+  if (!data || len < kMinRtcpPacketLen + 4 || !value)
>+    return false;
>   int pl_type;
>-  if (!GetRtcpType(data, len, &pl_type)) return false;
>+  if (!GetRtcpType(data, len, &pl_type))
>+    return false;
>   // SDES packet parsing is not supported.
>-  if (pl_type == kRtcpTypeSDES) return false;
>+  if (pl_type == kRtcpTypeSDES)
>+    return false;
>   *value = rtc::GetBE32(static_cast<const uint8_t*>(data) + 4);
>   return true;
> }
>@@ -256,8 +261,8 @@ bool SetRtpSsrc(void* data, size_t len, uint32_t value) {
> 
> // Assumes version 2, no padding, no extensions, no csrcs.
> bool SetRtpHeader(void* data, size_t len, const RtpHeader& header) {
>-  if (!IsValidRtpPayloadType(header.payload_type) ||
>-      header.seq_num < 0 || header.seq_num > UINT16_MAX) {
>+  if (!IsValidRtpPayloadType(header.payload_type) || header.seq_num < 0 ||
>+      header.seq_num > static_cast<int>(UINT16_MAX)) {
>     return false;
>   }
>   return (SetUint8(data, kRtpFlagsOffset, kRtpVersion << 6) &&
>@@ -275,6 +280,16 @@ bool IsRtpPacket(const void* data, size_t len) {
>   return (static_cast<const uint8_t*>(data)[0] >> 6) == kRtpVersion;
> }
> 
>+// Check the RTP payload type. If 63 < payload type < 96, it's RTCP.
>+// For additional details, see http://tools.ietf.org/html/rfc5761.
>+bool IsRtcpPacket(const char* data, size_t len) {
>+  if (len < 2) {
>+    return false;
>+  }
>+  char pt = data[1] & 0x7F;
>+  return (63 < pt) && (pt < 96);
>+}
>+
> bool IsValidRtpPayloadType(int payload_type) {
>   return payload_type >= 0 && payload_type <= 127;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/rtputils.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/rtputils.h
>index 0b7205cf8f5..0fd9a59816b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/rtputils.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/rtputils.h
>@@ -31,13 +31,13 @@ struct RtpHeader {
> };
> 
> enum RtcpTypes {
>-  kRtcpTypeSR = 200,      // Sender report payload type.
>-  kRtcpTypeRR = 201,      // Receiver report payload type.
>-  kRtcpTypeSDES = 202,    // SDES payload type.
>-  kRtcpTypeBye = 203,     // BYE payload type.
>-  kRtcpTypeApp = 204,     // APP payload type.
>-  kRtcpTypeRTPFB = 205,   // Transport layer Feedback message payload type.
>-  kRtcpTypePSFB = 206,    // Payload-specific Feedback message payload type.
>+  kRtcpTypeSR = 200,     // Sender report payload type.
>+  kRtcpTypeRR = 201,     // Receiver report payload type.
>+  kRtcpTypeSDES = 202,   // SDES payload type.
>+  kRtcpTypeBye = 203,    // BYE payload type.
>+  kRtcpTypeApp = 204,    // APP payload type.
>+  kRtcpTypeRTPFB = 205,  // Transport layer Feedback message payload type.
>+  kRtcpTypePSFB = 206,   // Payload-specific Feedback message payload type.
> };
> 
> bool GetRtpPayloadType(const void* data, size_t len, int* value);
>@@ -55,6 +55,7 @@ bool SetRtpHeader(void* data, size_t len, const RtpHeader& header);
> 
> bool IsRtpPacket(const void* data, size_t len);
> 
>+bool IsRtcpPacket(const char* data, size_t len);
> // True if |payload type| is 0-127.
> bool IsValidRtpPayloadType(int payload_type);
> 
>@@ -84,7 +85,6 @@ bool ApplyPacketOptions(uint8_t* data,
>                         const rtc::PacketTimeUpdateParams& packet_time_params,
>                         uint64_t time_us);
> 
>-
> }  // namespace cricket
> 
> #endif  // MEDIA_BASE_RTPUTILS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/rtputils_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/rtputils_unittest.cc
>index a71eac7a076..08b4d4f4bd4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/rtputils_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/rtputils_unittest.cc
>@@ -10,78 +10,68 @@
> 
> #include <vector>
> 
>-#include "media/base/rtputils.h"
> #include "media/base/fakertp.h"
>+#include "media/base/rtputils.h"
> #include "rtc_base/asyncpacketsocket.h"
> #include "rtc_base/gunit.h"
> 
> namespace cricket {
> 
> static const uint8_t kRtpPacketWithMarker[] = {
>-    0x80, 0x80, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01
>-};
>+    0x80, 0x80, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01};
> // 3 CSRCs (0x01020304, 0x12345678, 0xAABBCCDD)
> // Extension (0xBEDE, 0x1122334455667788)
> static const uint8_t kRtpPacketWithMarkerAndCsrcAndExtension[] = {
>     0x93, 0x80, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01,
>     0x01, 0x02, 0x03, 0x04, 0x12, 0x34, 0x56, 0x78, 0xAA, 0xBB, 0xCC, 0xDD,
>-    0xBE, 0xDE, 0x00, 0x02, 0x11, 0x22, 0x33, 0x44, 0x55, 0x66, 0x77, 0x88
>-};
>-static const uint8_t kInvalidPacket[] = { 0x80, 0x00 };
>+    0xBE, 0xDE, 0x00, 0x02, 0x11, 0x22, 0x33, 0x44, 0x55, 0x66, 0x77, 0x88};
>+static const uint8_t kInvalidPacket[] = {0x80, 0x00};
> static const uint8_t kInvalidPacketWithCsrc[] = {
>     0x83, 0x00, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01,
>-    0x01, 0x02, 0x03, 0x04, 0x12, 0x34, 0x56, 0x78, 0xAA, 0xBB, 0xCC
>-};
>+    0x01, 0x02, 0x03, 0x04, 0x12, 0x34, 0x56, 0x78, 0xAA, 0xBB, 0xCC};
> static const uint8_t kInvalidPacketWithCsrcAndExtension1[] = {
>-    0x93, 0x80, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01,
>-    0x01, 0x02, 0x03, 0x04, 0x12, 0x34, 0x56, 0x78, 0xAA, 0xBB, 0xCC, 0xDD,
>-    0xBE, 0xDE, 0x00
>-};
>+    0x93, 0x80, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00,
>+    0x00, 0x00, 0x01, 0x01, 0x02, 0x03, 0x04, 0x12, 0x34,
>+    0x56, 0x78, 0xAA, 0xBB, 0xCC, 0xDD, 0xBE, 0xDE, 0x00};
> static const uint8_t kInvalidPacketWithCsrcAndExtension2[] = {
>     0x93, 0x80, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x01,
>     0x01, 0x02, 0x03, 0x04, 0x12, 0x34, 0x56, 0x78, 0xAA, 0xBB, 0xCC, 0xDD,
>-    0xBE, 0xDE, 0x00, 0x02, 0x11, 0x22, 0x33, 0x44, 0x55, 0x66, 0x77
>-};
>+    0xBE, 0xDE, 0x00, 0x02, 0x11, 0x22, 0x33, 0x44, 0x55, 0x66, 0x77};
> 
> // PT = 206, FMT = 1, Sender SSRC  = 0x1111, Media SSRC = 0x1111
> // No FCI information is needed for PLI.
> static const uint8_t kNonCompoundRtcpPliFeedbackPacket[] = {
>-    0x81, 0xCE, 0x00, 0x0C, 0x00, 0x00, 0x11, 0x11, 0x00, 0x00, 0x11, 0x11
>-};
>+    0x81, 0xCE, 0x00, 0x0C, 0x00, 0x00, 0x11, 0x11, 0x00, 0x00, 0x11, 0x11};
> 
> // Packet has only mandatory fixed RTCP header
> // PT = 204, SSRC = 0x1111
>-static const uint8_t kNonCompoundRtcpAppPacket[] = {
>-    0x81, 0xCC, 0x00, 0x0C, 0x00, 0x00, 0x11, 0x11
>-};
>+static const uint8_t kNonCompoundRtcpAppPacket[] = {0x81, 0xCC, 0x00, 0x0C,
>+                                                    0x00, 0x00, 0x11, 0x11};
> 
> // PT = 202, Source count = 0
>-static const uint8_t kNonCompoundRtcpSDESPacket[] = {
>-    0x80, 0xCA, 0x00, 0x00
>-};
>+static const uint8_t kNonCompoundRtcpSDESPacket[] = {0x80, 0xCA, 0x00, 0x00};
> 
>-static uint8_t kFakeTag[4] = { 0xba, 0xdd, 0xba, 0xdd };
>+static uint8_t kFakeTag[4] = {0xba, 0xdd, 0xba, 0xdd};
> static uint8_t kTestKey[] = "12345678901234567890";
>-static uint8_t kTestAstValue[3] = { 0xaa, 0xbb, 0xcc };
>+static uint8_t kTestAstValue[3] = {0xaa, 0xbb, 0xcc};
> 
> // Valid rtp Message with 2 byte header extension.
> static uint8_t kRtpMsgWith2ByteExtnHeader[] = {
>+    // clang-format off
>+    // clang formatting doesn't respect inline comments.
>   0x90, 0x00, 0x00, 0x00,
>   0x00, 0x00, 0x00, 0x00,
>   0xAA, 0xBB, 0xCC, 0XDD,  // SSRC
>   0x10, 0x00, 0x00, 0x01,  // 2 Byte header extension
>   0x01, 0x00, 0x00, 0x00
>+    // clang-format on
> };
> 
> // RTP packet with single byte extension header of length 4 bytes.
> // Extension id = 3 and length = 3
> static uint8_t kRtpMsgWithAbsSendTimeExtension[] = {
>-  0x90, 0x00, 0x00, 0x00,
>-  0x00, 0x00, 0x00, 0x00,
>-  0x00, 0x00, 0x00, 0x00,
>-  0xBE, 0xDE, 0x00, 0x02,
>-  0x22, 0x00, 0x02, 0x1c,
>-  0x32, 0xaa, 0xbb, 0xcc,
>+    0x90, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
>+    0xBE, 0xDE, 0x00, 0x02, 0x22, 0x00, 0x02, 0x1c, 0x32, 0xaa, 0xbb, 0xcc,
> };
> 
> // Index of AbsSendTimeExtn data in message |kRtpMsgWithAbsSendTimeExtension|.
>@@ -123,11 +113,10 @@ TEST(RtpUtilsTest, GetRtp) {
> }
> 
> TEST(RtpUtilsTest, SetRtpHeader) {
>-  uint8_t packet[] = {
>-    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00
>-  };
>+  uint8_t packet[] = {0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
>+                      0x00, 0x00, 0x00, 0x00, 0x00, 0x00};
> 
>-  RtpHeader header = { 9, 1111, 2222u, 3333u };
>+  RtpHeader header = {9, 1111, 2222u, 3333u};
>   EXPECT_TRUE(SetRtpHeader(packet, sizeof(packet), header));
> 
>   // Bits: 10 0 0 0000
>@@ -172,33 +161,31 @@ TEST(RtpUtilsTest, GetRtcp) {
> 
>   uint32_t ssrc;
>   EXPECT_TRUE(GetRtcpSsrc(kNonCompoundRtcpPliFeedbackPacket,
>-                          sizeof(kNonCompoundRtcpPliFeedbackPacket),
>-                          &ssrc));
>+                          sizeof(kNonCompoundRtcpPliFeedbackPacket), &ssrc));
>   EXPECT_TRUE(GetRtcpSsrc(kNonCompoundRtcpAppPacket,
>-                          sizeof(kNonCompoundRtcpAppPacket),
>-                          &ssrc));
>+                          sizeof(kNonCompoundRtcpAppPacket), &ssrc));
>   EXPECT_FALSE(GetRtcpSsrc(kNonCompoundRtcpSDESPacket,
>-                           sizeof(kNonCompoundRtcpSDESPacket),
>-                           &ssrc));
>+                           sizeof(kNonCompoundRtcpSDESPacket), &ssrc));
> }
> 
> // Invalid RTP packets.
> TEST(RtpUtilsTest, InvalidRtpHeader) {
>   // Rtp message with invalid length.
>   const uint8_t kRtpMsgWithInvalidLength[] = {
>+      // clang-format off
>+      // clang formatting doesn't respect inline comments.
>       0x94, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
>       0xAA, 0xBB, 0xCC, 0XDD,  // SSRC
>       0xDD, 0xCC, 0xBB, 0xAA,  // Only 1 CSRC, but CC count is 4.
>+      // clang-format on
>   };
>   EXPECT_FALSE(ValidateRtpHeader(kRtpMsgWithInvalidLength,
>                                  sizeof(kRtpMsgWithInvalidLength), nullptr));
> 
>   // Rtp message with single byte header extension, invalid extension length.
>   const uint8_t kRtpMsgWithInvalidExtnLength[] = {
>-      0x90, 0x00, 0x00, 0x00,
>-      0x00, 0x00, 0x00, 0x00,
>-      0x00, 0x00, 0x00, 0x00,
>-      0xBE, 0xDE, 0x0A, 0x00,  // Extn length - 0x0A00
>+      0x90, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
>+      0x00, 0x00, 0x00, 0x00, 0xBE, 0xDE, 0x0A, 0x00,  // Extn length - 0x0A00
>   };
>   EXPECT_FALSE(ValidateRtpHeader(kRtpMsgWithInvalidExtnLength,
>                                  sizeof(kRtpMsgWithInvalidExtnLength),
>@@ -232,6 +219,8 @@ TEST(RtpUtilsTest, UpdateAbsSendTimeExtensionInTurnSendIndication) {
>   // A valid STUN indication message with a valid RTP header in data attribute
>   // payload field and no extension bit set.
>   uint8_t message_without_extension[] = {
>+      // clang-format off
>+      // clang formatting doesn't respect inline comments.
>       0x00, 0x16, 0x00, 0x18,  // length of
>       0x21, 0x12, 0xA4, 0x42,  // magic cookie
>       '0',  '1',  '2',  '3',   // transaction id
>@@ -243,6 +232,7 @@ TEST(RtpUtilsTest, UpdateAbsSendTimeExtensionInTurnSendIndication) {
>       0x80, 0x00, 0x00, 0x00,  // RTP packet.
>       0x00, 0x00, 0x00, 0x00,
>       0x00, 0x00, 0x00, 0x00,
>+      // clang-format on
>   };
>   EXPECT_TRUE(UpdateRtpAbsSendTimeExtension(
>       message_without_extension, sizeof(message_without_extension), 3, 0));
>@@ -250,6 +240,8 @@ TEST(RtpUtilsTest, UpdateAbsSendTimeExtensionInTurnSendIndication) {
>   // A valid STUN indication message with a valid RTP header and a extension
>   // header.
>   uint8_t message[] = {
>+      // clang-format off
>+      // clang formatting doesn't respect inline comments.
>       0x00, 0x16, 0x00, 0x24,  // length of
>       0x21, 0x12, 0xA4, 0x42,  // magic cookie
>       '0',  '1',  '2',  '3',   // transaction id
>@@ -261,6 +253,7 @@ TEST(RtpUtilsTest, UpdateAbsSendTimeExtensionInTurnSendIndication) {
>       0x90, 0x00, 0x00, 0x00,  // RTP packet.
>       0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0xBE, 0xDE,
>       0x00, 0x02, 0x22, 0xaa, 0xbb, 0xcc, 0x32, 0xaa, 0xbb, 0xcc,
>+      // clang-format on
>   };
>   EXPECT_TRUE(UpdateRtpAbsSendTimeExtension(message, sizeof(message), 3, 0));
> }
>@@ -349,5 +342,4 @@ TEST(RtpUtilsTest, ApplyPacketOptionsWithAuthParamsAndAbsSendTime) {
>                       sizeof(kExpectedTimestamp)));
> }
> 
>-
> }  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/streamparams.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/streamparams.cc
>index fd61a87ffd7..154baf69470 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/streamparams.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/streamparams.cc
>@@ -11,7 +11,9 @@
> #include "media/base/streamparams.h"
> 
> #include <list>
>-#include <sstream>
>+
>+#include "rtc_base/checks.h"
>+#include "rtc_base/strings/string_builder.h"
> 
> namespace cricket {
> namespace {
>@@ -20,7 +22,7 @@ namespace {
> void AddStream(std::vector<StreamParams>* streams, const StreamParams& stream) {
>   streams->push_back(stream);
> }
>-}
>+}  // namespace
> 
> const char kFecSsrcGroupSemantics[] = "FEC";
> const char kFecFrSsrcGroupSemantics[] = "FEC-FR";
>@@ -36,18 +38,21 @@ bool GetStream(const StreamParamsVec& streams,
>   return found != nullptr;
> }
> 
>-bool MediaStreams::GetAudioStream(
>-    const StreamSelector& selector, StreamParams* stream) {
>+MediaStreams::MediaStreams() = default;
>+MediaStreams::~MediaStreams() = default;
>+
>+bool MediaStreams::GetAudioStream(const StreamSelector& selector,
>+                                  StreamParams* stream) {
>   return GetStream(audio_, selector, stream);
> }
> 
>-bool MediaStreams::GetVideoStream(
>-    const StreamSelector& selector, StreamParams* stream) {
>+bool MediaStreams::GetVideoStream(const StreamSelector& selector,
>+                                  StreamParams* stream) {
>   return GetStream(video_, selector, stream);
> }
> 
>-bool MediaStreams::GetDataStream(
>-    const StreamSelector& selector, StreamParams* stream) {
>+bool MediaStreams::GetDataStream(const StreamSelector& selector,
>+                                 StreamParams* stream) {
>   return GetStream(data_, selector, stream);
> }
> 
>@@ -69,81 +74,98 @@ void MediaStreams::AddDataStream(const StreamParams& stream) {
>   AddStream(&data_, stream);
> }
> 
>-bool MediaStreams::RemoveAudioStream(
>-    const StreamSelector& selector) {
>+bool MediaStreams::RemoveAudioStream(const StreamSelector& selector) {
>   return RemoveStream(&audio_, selector);
> }
> 
>-bool MediaStreams::RemoveVideoStream(
>-    const StreamSelector& selector) {
>+bool MediaStreams::RemoveVideoStream(const StreamSelector& selector) {
>   return RemoveStream(&video_, selector);
> }
> 
>-bool MediaStreams::RemoveDataStream(
>-    const StreamSelector& selector) {
>+bool MediaStreams::RemoveDataStream(const StreamSelector& selector) {
>   return RemoveStream(&data_, selector);
> }
> 
> static std::string SsrcsToString(const std::vector<uint32_t>& ssrcs) {
>-  std::ostringstream ost;
>-  ost << "ssrcs:[";
>+  char buf[1024];
>+  rtc::SimpleStringBuilder sb(buf);
>+  sb << "ssrcs:[";
>   for (std::vector<uint32_t>::const_iterator it = ssrcs.begin();
>        it != ssrcs.end(); ++it) {
>     if (it != ssrcs.begin()) {
>-      ost << ",";
>+      sb << ",";
>     }
>-    ost << *it;
>+    sb << *it;
>   }
>-  ost << "]";
>-  return ost.str();
>+  sb << "]";
>+  return sb.str();
> }
> 
>+SsrcGroup::SsrcGroup(const std::string& usage,
>+                     const std::vector<uint32_t>& ssrcs)
>+    : semantics(usage), ssrcs(ssrcs) {}
>+SsrcGroup::SsrcGroup(const SsrcGroup&) = default;
>+SsrcGroup::SsrcGroup(SsrcGroup&&) = default;
>+SsrcGroup::~SsrcGroup() = default;
>+
>+SsrcGroup& SsrcGroup::operator=(const SsrcGroup&) = default;
>+SsrcGroup& SsrcGroup::operator=(SsrcGroup&&) = default;
>+
> bool SsrcGroup::has_semantics(const std::string& semantics_in) const {
>   return (semantics == semantics_in && ssrcs.size() > 0);
> }
> 
> std::string SsrcGroup::ToString() const {
>-  std::ostringstream ost;
>-  ost << "{";
>-  ost << "semantics:" << semantics << ";";
>-  ost << SsrcsToString(ssrcs);
>-  ost << "}";
>-  return ost.str();
>+  char buf[1024];
>+  rtc::SimpleStringBuilder sb(buf);
>+  sb << "{";
>+  sb << "semantics:" << semantics << ";";
>+  sb << SsrcsToString(ssrcs);
>+  sb << "}";
>+  return sb.str();
> }
> 
>+StreamParams::StreamParams() = default;
>+StreamParams::StreamParams(const StreamParams&) = default;
>+StreamParams::StreamParams(StreamParams&&) = default;
>+StreamParams::~StreamParams() = default;
>+StreamParams& StreamParams::operator=(const StreamParams&) = default;
>+StreamParams& StreamParams::operator=(StreamParams&&) = default;
>+
> std::string StreamParams::ToString() const {
>-  std::ostringstream ost;
>-  ost << "{";
>+  char buf[2 * 1024];
>+  rtc::SimpleStringBuilder sb(buf);
>+  sb << "{";
>   if (!groupid.empty()) {
>-    ost << "groupid:" << groupid << ";";
>+    sb << "groupid:" << groupid << ";";
>   }
>   if (!id.empty()) {
>-    ost << "id:" << id << ";";
>+    sb << "id:" << id << ";";
>   }
>-  ost << SsrcsToString(ssrcs) << ";";
>-  ost << "ssrc_groups:";
>+  sb << SsrcsToString(ssrcs) << ";";
>+  sb << "ssrc_groups:";
>   for (std::vector<SsrcGroup>::const_iterator it = ssrc_groups.begin();
>        it != ssrc_groups.end(); ++it) {
>     if (it != ssrc_groups.begin()) {
>-      ost << ",";
>+      sb << ",";
>     }
>-    ost << it->ToString();
>-  }
>-  ost << ";";
>-  if (!type.empty()) {
>-    ost << "type:" << type << ";";
>-  }
>-  if (!display.empty()) {
>-    ost << "display:" << display << ";";
>+    sb << it->ToString();
>   }
>+  sb << ";";
>   if (!cname.empty()) {
>-    ost << "cname:" << cname << ";";
>+    sb << "cname:" << cname << ";";
>   }
>-  if (!sync_label.empty()) {
>-    ost << "sync_label:" << sync_label;
>+  sb << "stream_ids:";
>+  for (std::vector<std::string>::const_iterator it = stream_ids_.begin();
>+       it != stream_ids_.end(); ++it) {
>+    if (it != stream_ids_.begin()) {
>+      sb << ",";
>+    }
>+    sb << *it;
>   }
>-  ost << "}";
>-  return ost.str();
>+  sb << ";";
>+  sb << "}";
>+  return sb.str();
> }
> void StreamParams::GetPrimarySsrcs(std::vector<uint32_t>* ssrcs) const {
>   const SsrcGroup* sim_group = get_ssrc_group(kSimSsrcGroupSemantics);
>@@ -187,9 +209,8 @@ bool StreamParams::GetSecondarySsrc(const std::string& semantics,
>                                     uint32_t* secondary_ssrc) const {
>   for (std::vector<SsrcGroup>::const_iterator it = ssrc_groups.begin();
>        it != ssrc_groups.end(); ++it) {
>-    if (it->has_semantics(semantics) &&
>-          it->ssrcs.size() >= 2 &&
>-          it->ssrcs[0] == primary_ssrc) {
>+    if (it->has_semantics(semantics) && it->ssrcs.size() >= 2 &&
>+        it->ssrcs[0] == primary_ssrc) {
>       *secondary_ssrc = it->ssrcs[1];
>       return true;
>     }
>@@ -197,6 +218,18 @@ bool StreamParams::GetSecondarySsrc(const std::string& semantics,
>   return false;
> }
> 
>+std::vector<std::string> StreamParams::stream_ids() const {
>+  return stream_ids_;
>+}
>+
>+void StreamParams::set_stream_ids(const std::vector<std::string>& stream_ids) {
>+  stream_ids_ = stream_ids;
>+}
>+
>+std::string StreamParams::first_stream_id() const {
>+  return stream_ids_.empty() ? "" : stream_ids_[0];
>+}
>+
> bool IsOneSsrcStream(const StreamParams& sp) {
>   if (sp.ssrcs.size() == 1 && sp.ssrc_groups.empty()) {
>     return true;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/streamparams.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/streamparams.h
>index 1b2ebfa8711..488b61d0404 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/streamparams.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/streamparams.h
>@@ -43,25 +43,38 @@ extern const char kFidSsrcGroupSemantics[];
> extern const char kSimSsrcGroupSemantics[];
> 
> struct SsrcGroup {
>-  SsrcGroup(const std::string& usage, const std::vector<uint32_t>& ssrcs)
>-      : semantics(usage), ssrcs(ssrcs) {}
>+  SsrcGroup(const std::string& usage, const std::vector<uint32_t>& ssrcs);
>+  SsrcGroup(const SsrcGroup&);
>+  SsrcGroup(SsrcGroup&&);
>+  ~SsrcGroup();
>+  SsrcGroup& operator=(const SsrcGroup&);
>+  SsrcGroup& operator=(SsrcGroup&&);
> 
>   bool operator==(const SsrcGroup& other) const {
>     return (semantics == other.semantics && ssrcs == other.ssrcs);
>   }
>-  bool operator!=(const SsrcGroup &other) const {
>-    return !(*this == other);
>-  }
>+  bool operator!=(const SsrcGroup& other) const { return !(*this == other); }
> 
>   bool has_semantics(const std::string& semantics) const;
> 
>   std::string ToString() const;
> 
>-  std::string semantics;  // e.g FIX, FEC, SIM.
>+  std::string semantics;        // e.g FIX, FEC, SIM.
>   std::vector<uint32_t> ssrcs;  // SSRCs of this type.
> };
> 
>+// StreamParams is used to represent a sender/track in a SessionDescription.
>+// In Plan B, this means that multiple StreamParams can exist within one
>+// MediaContentDescription, while in UnifiedPlan this means that there is one
>+// StreamParams per MediaContentDescription.
> struct StreamParams {
>+  StreamParams();
>+  StreamParams(const StreamParams&);
>+  StreamParams(StreamParams&&);
>+  ~StreamParams();
>+  StreamParams& operator=(const StreamParams&);
>+  StreamParams& operator=(StreamParams&&);
>+
>   static StreamParams CreateLegacy(uint32_t ssrc) {
>     StreamParams stream;
>     stream.ssrcs.push_back(ssrc);
>@@ -69,18 +82,11 @@ struct StreamParams {
>   }
> 
>   bool operator==(const StreamParams& other) const {
>-    return (groupid == other.groupid &&
>-            id == other.id &&
>-            ssrcs == other.ssrcs &&
>-            ssrc_groups == other.ssrc_groups &&
>-            type == other.type &&
>-            display == other.display &&
>-            cname == other.cname &&
>-            sync_label == other.sync_label);
>-  }
>-  bool operator!=(const StreamParams &other) const {
>-    return !(*this == other);
>+    return (groupid == other.groupid && id == other.id &&
>+            ssrcs == other.ssrcs && ssrc_groups == other.ssrc_groups &&
>+            cname == other.cname && stream_ids_ == other.stream_ids_);
>   }
>+  bool operator!=(const StreamParams& other) const { return !(*this == other); }
> 
>   uint32_t first_ssrc() const {
>     if (ssrcs.empty()) {
>@@ -89,16 +95,12 @@ struct StreamParams {
> 
>     return ssrcs[0];
>   }
>-  bool has_ssrcs() const {
>-    return !ssrcs.empty();
>-  }
>+  bool has_ssrcs() const { return !ssrcs.empty(); }
>   bool has_ssrc(uint32_t ssrc) const {
>     return std::find(ssrcs.begin(), ssrcs.end(), ssrc) != ssrcs.end();
>   }
>   void add_ssrc(uint32_t ssrc) { ssrcs.push_back(ssrc); }
>-  bool has_ssrc_groups() const {
>-    return !ssrc_groups.empty();
>-  }
>+  bool has_ssrc_groups() const { return !ssrc_groups.empty(); }
>   bool has_ssrc_group(const std::string& semantics) const {
>     return (get_ssrc_group(semantics) != NULL);
>   }
>@@ -146,22 +148,30 @@ struct StreamParams {
>   void GetFidSsrcs(const std::vector<uint32_t>& primary_ssrcs,
>                    std::vector<uint32_t>* fid_ssrcs) const;
> 
>+  // Stream ids serialized to SDP.
>+  std::vector<std::string> stream_ids() const;
>+  void set_stream_ids(const std::vector<std::string>& stream_ids);
>+
>+  // Returns the first stream id or "" if none exist. This method exists only
>+  // as temporary backwards compatibility with the old sync_label.
>+  std::string first_stream_id() const;
>+
>   std::string ToString() const;
> 
>   // Resource of the MUC jid of the participant of with this stream.
>   // For 1:1 calls, should be left empty (which means remote streams
>-  // and local streams should not be mixed together).
>+  // and local streams should not be mixed together). This is not used
>+  // internally and should be deprecated.
>   std::string groupid;
>-  // Unique per-groupid, not across all groupids
>+  // A unique identifier of the StreamParams object. When the SDP is created,
>+  // this comes from the track ID of the sender that the StreamParams object
>+  // is associated with.
>   std::string id;
>+  // There may be no SSRCs stored in unsignaled case when stream_ids are
>+  // signaled with a=msid lines.
>   std::vector<uint32_t> ssrcs;         // All SSRCs for this source
>   std::vector<SsrcGroup> ssrc_groups;  // e.g. FID, FEC, SIM
>-  // Examples: "camera", "screencast"
>-  std::string type;
>-  // Friendly name describing stream
>-  std::string display;
>-  std::string cname;  // RTCP CNAME
>-  std::string sync_label;  // Friendly name of cname.
>+  std::string cname;                   // RTCP CNAME
> 
>  private:
>   bool AddSecondarySsrc(const std::string& semantics,
>@@ -170,18 +180,22 @@ struct StreamParams {
>   bool GetSecondarySsrc(const std::string& semantics,
>                         uint32_t primary_ssrc,
>                         uint32_t* secondary_ssrc) const;
>+
>+  // The stream IDs of the sender that the StreamParams object is associated
>+  // with. In Plan B this should always be size of 1, while in Unified Plan this
>+  // could be none or multiple stream IDs.
>+  std::vector<std::string> stream_ids_;
> };
> 
> // A Stream can be selected by either groupid+id or ssrc.
> struct StreamSelector {
>   explicit StreamSelector(uint32_t ssrc) : ssrc(ssrc) {}
> 
>-  StreamSelector(const std::string& groupid,
>-                 const std::string& streamid) :
>-      ssrc(0),
>-      groupid(groupid),
>-      streamid(streamid) {
>-  }
>+  StreamSelector(const std::string& groupid, const std::string& streamid)
>+      : ssrc(0), groupid(groupid), streamid(streamid) {}
>+
>+  explicit StreamSelector(const std::string& streamid)
>+      : ssrc(0), streamid(streamid) {}
> 
>   bool Matches(const StreamParams& stream) const {
>     if (ssrc == 0) {
>@@ -206,7 +220,8 @@ typedef std::vector<StreamParams> StreamParamsVec;
> // See https://code.google.com/p/webrtc/issues/detail?id=4107
> struct MediaStreams {
>  public:
>-  MediaStreams() {}
>+  MediaStreams();
>+  ~MediaStreams();
>   void CopyFrom(const MediaStreams& sources);
> 
>   bool empty() const {
>@@ -221,12 +236,9 @@ struct MediaStreams {
>   const std::vector<StreamParams>& data() const { return data_; }
> 
>   // Gets a stream, returning true if found.
>-  bool GetAudioStream(
>-      const StreamSelector& selector, StreamParams* stream);
>-  bool GetVideoStream(
>-      const StreamSelector& selector, StreamParams* stream);
>-  bool GetDataStream(
>-      const StreamSelector& selector, StreamParams* stream);
>+  bool GetAudioStream(const StreamSelector& selector, StreamParams* stream);
>+  bool GetVideoStream(const StreamSelector& selector, StreamParams* stream);
>+  bool GetDataStream(const StreamSelector& selector, StreamParams* stream);
>   // Adds a stream.
>   void AddAudioStream(const StreamParams& stream);
>   void AddVideoStream(const StreamParams& stream);
>@@ -259,10 +271,15 @@ StreamParams* GetStream(StreamParamsVec& streams, Condition condition) {
>   return found == streams.end() ? nullptr : &(*found);
> }
> 
>+inline bool HasStreamWithNoSsrcs(const StreamParamsVec& streams) {
>+  return GetStream(streams,
>+                   [](const StreamParams& sp) { return !sp.has_ssrcs(); });
>+}
>+
> inline const StreamParams* GetStreamBySsrc(const StreamParamsVec& streams,
>                                            uint32_t ssrc) {
>-  return GetStream(streams,
>-      [&ssrc](const StreamParams& sp) { return sp.has_ssrc(ssrc); });
>+  return GetStream(
>+      streams, [&ssrc](const StreamParams& sp) { return sp.has_ssrc(ssrc); });
> }
> 
> inline const StreamParams* GetStreamByIds(const StreamParamsVec& streams,
>@@ -276,16 +293,16 @@ inline const StreamParams* GetStreamByIds(const StreamParamsVec& streams,
> inline StreamParams* GetStreamByIds(StreamParamsVec& streams,
>                                     const std::string& groupid,
>                                     const std::string& id) {
>-  return GetStream(streams,
>-      [&groupid, &id](const StreamParams& sp) {
>-        return sp.groupid == groupid && sp.id == id;
>-      });
>+  return GetStream(streams, [&groupid, &id](const StreamParams& sp) {
>+    return sp.groupid == groupid && sp.id == id;
>+  });
> }
> 
> inline const StreamParams* GetStream(const StreamParamsVec& streams,
>                                      const StreamSelector& selector) {
>-  return GetStream(streams,
>-      [&selector](const StreamParams& sp) { return selector.Matches(sp); });
>+  return GetStream(streams, [&selector](const StreamParams& sp) {
>+    return selector.Matches(sp);
>+  });
> }
> 
> template <class Condition>
>@@ -300,21 +317,21 @@ bool RemoveStream(StreamParamsVec* streams, Condition condition) {
> // Removes the stream from streams. Returns true if a stream is
> // found and removed.
> inline bool RemoveStream(StreamParamsVec* streams,
>-                  const StreamSelector& selector) {
>-  return RemoveStream(streams,
>-      [&selector](const StreamParams& sp) { return selector.Matches(sp); });
>+                         const StreamSelector& selector) {
>+  return RemoveStream(streams, [&selector](const StreamParams& sp) {
>+    return selector.Matches(sp);
>+  });
> }
> inline bool RemoveStreamBySsrc(StreamParamsVec* streams, uint32_t ssrc) {
>-  return RemoveStream(streams,
>-      [&ssrc](const StreamParams& sp) { return sp.has_ssrc(ssrc); });
>+  return RemoveStream(
>+      streams, [&ssrc](const StreamParams& sp) { return sp.has_ssrc(ssrc); });
> }
> inline bool RemoveStreamByIds(StreamParamsVec* streams,
>                               const std::string& groupid,
>                               const std::string& id) {
>-  return RemoveStream(streams,
>-      [&groupid, &id](const StreamParams& sp) {
>-        return sp.groupid == groupid && sp.id == id;
>-      });
>+  return RemoveStream(streams, [&groupid, &id](const StreamParams& sp) {
>+    return sp.groupid == groupid && sp.id == id;
>+  });
> }
> 
> // Checks if |sp| defines parameters for a single primary stream. There may
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/streamparams_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/streamparams_unittest.cc
>index 6e934ae7d6f..c43a955fb6f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/streamparams_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/streamparams_unittest.cc
>@@ -32,10 +32,10 @@ static cricket::StreamParams CreateStreamParamsWithSsrcGroup(
> 
> TEST(SsrcGroup, EqualNotEqual) {
>   cricket::SsrcGroup ssrc_groups[] = {
>-    cricket::SsrcGroup("ABC", MAKE_VECTOR(kSsrcs1)),
>-    cricket::SsrcGroup("ABC", MAKE_VECTOR(kSsrcs2)),
>-    cricket::SsrcGroup("Abc", MAKE_VECTOR(kSsrcs2)),
>-    cricket::SsrcGroup("abc", MAKE_VECTOR(kSsrcs2)),
>+      cricket::SsrcGroup("ABC", MAKE_VECTOR(kSsrcs1)),
>+      cricket::SsrcGroup("ABC", MAKE_VECTOR(kSsrcs2)),
>+      cricket::SsrcGroup("Abc", MAKE_VECTOR(kSsrcs2)),
>+      cricket::SsrcGroup("abc", MAKE_VECTOR(kSsrcs2)),
>   };
> 
>   for (size_t i = 0; i < arraysize(ssrc_groups); ++i) {
>@@ -69,7 +69,7 @@ TEST(StreamParams, CreateLegacy) {
>   EXPECT_EQ(ssrc, one_sp.first_ssrc());
>   EXPECT_TRUE(one_sp.has_ssrcs());
>   EXPECT_TRUE(one_sp.has_ssrc(ssrc));
>-  EXPECT_FALSE(one_sp.has_ssrc(ssrc+1));
>+  EXPECT_FALSE(one_sp.has_ssrc(ssrc + 1));
>   EXPECT_FALSE(one_sp.has_ssrc_groups());
>   EXPECT_EQ(0U, one_sp.ssrc_groups.size());
> }
>@@ -96,6 +96,17 @@ TEST(StreamParams, GetSsrcGroup) {
>   EXPECT_EQ(&sp.ssrc_groups[0], sp.get_ssrc_group("XYZ"));
> }
> 
>+TEST(StreamParams, HasStreamWithNoSsrcs) {
>+  cricket::StreamParams sp_1 = cricket::StreamParams::CreateLegacy(kSsrcs1[0]);
>+  cricket::StreamParams sp_2 = cricket::StreamParams::CreateLegacy(kSsrcs2[0]);
>+  std::vector<cricket::StreamParams> streams({sp_1, sp_2});
>+  EXPECT_FALSE(HasStreamWithNoSsrcs(streams));
>+
>+  cricket::StreamParams unsignaled_stream;
>+  streams.push_back(unsignaled_stream);
>+  EXPECT_TRUE(HasStreamWithNoSsrcs(streams));
>+}
>+
> TEST(StreamParams, EqualNotEqual) {
>   cricket::StreamParams l1 = cricket::StreamParams::CreateLegacy(1);
>   cricket::StreamParams l2 = cricket::StreamParams::CreateLegacy(2);
>@@ -138,7 +149,7 @@ TEST(StreamParams, FidFunctions) {
>   std::vector<uint32_t> fid_vector;
>   fid_vector.push_back(13);
>   cricket::SsrcGroup invalid_fid_group(cricket::kFidSsrcGroupSemantics,
>-                                        fid_vector);
>+                                       fid_vector);
>   cricket::StreamParams sp_invalid;
>   sp_invalid.add_ssrc(13);
>   sp_invalid.ssrc_groups.push_back(invalid_fid_group);
>@@ -208,8 +219,11 @@ TEST(StreamParams, FecFrFunctions) {
> TEST(StreamParams, ToString) {
>   cricket::StreamParams sp =
>       CreateStreamParamsWithSsrcGroup("XYZ", kSsrcs2, arraysize(kSsrcs2));
>-  EXPECT_STREQ("{ssrcs:[1,2];ssrc_groups:{semantics:XYZ;ssrcs:[1,2]};}",
>-               sp.ToString().c_str());
>+  sp.set_stream_ids({"stream_id"});
>+  EXPECT_STREQ(
>+      "{ssrcs:[1,2];ssrc_groups:{semantics:XYZ;ssrcs:[1,2]};stream_ids:stream_"
>+      "id;}",
>+      sp.ToString().c_str());
> }
> 
> TEST(StreamParams, TestIsOneSsrcStream_LegacyStream) {
>@@ -247,10 +261,8 @@ TEST(StreamParams, TestIsOneSsrcStream_SimulcastStream) {
> }
> 
> TEST(StreamParams, TestIsOneSsrcStream_SimRtxStream) {
>-  cricket::StreamParams stream =
>-      cricket::CreateSimWithRtxStreamParams("cname",
>-                                            MAKE_VECTOR(kSsrcs3),
>-                                            MAKE_VECTOR(kRtxSsrcs3));
>+  cricket::StreamParams stream = cricket::CreateSimWithRtxStreamParams(
>+      "cname", MAKE_VECTOR(kSsrcs3), MAKE_VECTOR(kRtxSsrcs3));
>   EXPECT_FALSE(cricket::IsOneSsrcStream(stream));
> }
> 
>@@ -274,19 +286,15 @@ TEST(StreamParams, TestIsSimulcastStream_SimulcastStream) {
> }
> 
> TEST(StreamParams, TestIsSimulcastStream_SimRtxStream) {
>-  cricket::StreamParams stream =
>-      cricket::CreateSimWithRtxStreamParams("cname",
>-                                            MAKE_VECTOR(kSsrcs3),
>-                                            MAKE_VECTOR(kRtxSsrcs3));
>+  cricket::StreamParams stream = cricket::CreateSimWithRtxStreamParams(
>+      "cname", MAKE_VECTOR(kSsrcs3), MAKE_VECTOR(kRtxSsrcs3));
>   EXPECT_TRUE(cricket::IsSimulcastStream(stream));
> }
> 
> TEST(StreamParams, TestIsSimulcastStream_InvalidStreams) {
>   // stream1 has extra non-sim, non-fid ssrc.
>-  cricket::StreamParams stream1 =
>-      cricket::CreateSimWithRtxStreamParams("cname",
>-                                            MAKE_VECTOR(kSsrcs3),
>-                                            MAKE_VECTOR(kRtxSsrcs3));
>+  cricket::StreamParams stream1 = cricket::CreateSimWithRtxStreamParams(
>+      "cname", MAKE_VECTOR(kSsrcs3), MAKE_VECTOR(kRtxSsrcs3));
>   stream1.add_ssrc(25);
>   EXPECT_FALSE(cricket::IsSimulcastStream(stream1));
> 
>@@ -294,7 +302,9 @@ TEST(StreamParams, TestIsSimulcastStream_InvalidStreams) {
>   cricket::StreamParams stream2;
>   stream2.add_ssrc(13);
>   EXPECT_TRUE(stream2.AddFidSsrc(13, 14));
>-  std::remove(stream2.ssrcs.begin(), stream2.ssrcs.end(), 13u);
>+  stream2.ssrcs.erase(
>+      std::remove(stream2.ssrcs.begin(), stream2.ssrcs.end(), 13u),
>+      stream2.ssrcs.end());
>   EXPECT_FALSE(cricket::IsSimulcastStream(stream2));
> 
>   // stream3 has two SIM groups.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/test/mock_mediachannel.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/test/mock_mediachannel.h
>deleted file mode 100644
>index fdfbf3440e7..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/test/mock_mediachannel.h
>+++ /dev/null
>@@ -1,35 +0,0 @@
>-/*
>- *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MEDIA_BASE_TEST_MOCK_MEDIACHANNEL_H_
>-#define MEDIA_BASE_TEST_MOCK_MEDIACHANNEL_H_
>-
>-#include "media/base/fakemediaengine.h"
>-#include "test/gmock.h"
>-
>-namespace webrtc {
>-
>-class MockVideoMediaChannel : public cricket::FakeVideoMediaChannel {
>- public:
>-  MockVideoMediaChannel()
>-      : cricket::FakeVideoMediaChannel(nullptr, cricket::VideoOptions()) {}
>-  MOCK_METHOD1(GetStats, bool(cricket::VideoMediaInfo*));
>-};
>-
>-class MockVoiceMediaChannel : public cricket::FakeVoiceMediaChannel {
>- public:
>-  MockVoiceMediaChannel()
>-      : cricket::FakeVoiceMediaChannel(nullptr, cricket::AudioOptions()) {}
>-  MOCK_METHOD1(GetStats, bool(cricket::VoiceMediaInfo*));
>-};
>-
>-}  // namespace webrtc
>-
>-#endif  // MEDIA_BASE_TEST_MOCK_MEDIACHANNEL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/testutils.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/testutils.cc
>index f92d4013eb1..93c2bc98aef 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/testutils.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/testutils.cc
>@@ -29,7 +29,8 @@ namespace cricket {
> /////////////////////////////////////////////////////////////////////////
> void RawRtpPacket::WriteToByteBuffer(uint32_t in_ssrc,
>                                      rtc::ByteBufferWriter* buf) const {
>-  if (!buf) return;
>+  if (!buf)
>+    return;
> 
>   buf->WriteUInt8(ver_to_cc);
>   buf->WriteUInt8(m_to_pt);
>@@ -40,7 +41,8 @@ void RawRtpPacket::WriteToByteBuffer(uint32_t in_ssrc,
> }
> 
> bool RawRtpPacket::ReadFromByteBuffer(rtc::ByteBufferReader* buf) {
>-  if (!buf) return false;
>+  if (!buf)
>+    return false;
> 
>   bool ret = true;
>   ret &= buf->ReadUInt8(&ver_to_cc);
>@@ -56,19 +58,17 @@ bool RawRtpPacket::SameExceptSeqNumTimestampSsrc(const RawRtpPacket& packet,
>                                                  uint16_t seq,
>                                                  uint32_t ts,
>                                                  uint32_t ssc) const {
>-  return sequence_number == seq &&
>-      timestamp == ts &&
>-      ver_to_cc == packet.ver_to_cc &&
>-      m_to_pt == packet.m_to_pt &&
>-      ssrc == ssc &&
>-      0 == memcmp(payload, packet.payload, sizeof(payload));
>+  return sequence_number == seq && timestamp == ts &&
>+         ver_to_cc == packet.ver_to_cc && m_to_pt == packet.m_to_pt &&
>+         ssrc == ssc && 0 == memcmp(payload, packet.payload, sizeof(payload));
> }
> 
> /////////////////////////////////////////////////////////////////////////
> // Implementation of RawRtcpPacket
> /////////////////////////////////////////////////////////////////////////
>-void RawRtcpPacket::WriteToByteBuffer(rtc::ByteBufferWriter *buf) const {
>-  if (!buf) return;
>+void RawRtcpPacket::WriteToByteBuffer(rtc::ByteBufferWriter* buf) const {
>+  if (!buf)
>+    return;
> 
>   buf->WriteUInt8(ver_to_count);
>   buf->WriteUInt8(type);
>@@ -77,7 +77,8 @@ void RawRtcpPacket::WriteToByteBuffer(rtc::ByteBufferWriter *buf) const {
> }
> 
> bool RawRtcpPacket::ReadFromByteBuffer(rtc::ByteBufferReader* buf) {
>-  if (!buf) return false;
>+  if (!buf)
>+    return false;
> 
>   bool ret = true;
>   ret &= buf->ReadUInt8(&ver_to_count);
>@@ -88,10 +89,9 @@ bool RawRtcpPacket::ReadFromByteBuffer(rtc::ByteBufferReader* buf) {
> }
> 
> bool RawRtcpPacket::EqualsTo(const RawRtcpPacket& packet) const {
>-  return ver_to_count == packet.ver_to_count &&
>-      type == packet.type &&
>-      length == packet.length &&
>-      0 == memcmp(payload, packet.payload, sizeof(payload));
>+  return ver_to_count == packet.ver_to_count && type == packet.type &&
>+         length == packet.length &&
>+         0 == memcmp(payload, packet.payload, sizeof(payload));
> }
> 
> // Implementation of VideoCaptureListener.
>@@ -103,7 +103,7 @@ VideoCapturerListener::VideoCapturerListener(VideoCapturer* capturer)
>       frame_height_(0),
>       resolution_changed_(false) {
>   capturer->SignalStateChange.connect(this,
>-      &VideoCapturerListener::OnStateChange);
>+                                      &VideoCapturerListener::OnStateChange);
>   capturer->AddOrUpdateSink(this, rtc::VideoSinkWants());
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/testutils.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/testutils.h
>index 2cee107b303..c84696236b4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/testutils.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/testutils.h
>@@ -18,14 +18,13 @@
> #include "media/base/videocapturer.h"
> #include "media/base/videocommon.h"
> #include "rtc_base/arraysize.h"
>-#include "rtc_base/basictypes.h"
>-#include "rtc_base/sigslot.h"
>+#include "rtc_base/third_party/sigslot/sigslot.h"
> 
> namespace rtc {
> class ByteBufferReader;
> class ByteBufferWriter;
> class StreamInterface;
>-}
>+}  // namespace rtc
> 
> namespace webrtc {
> class VideoFrame;
>@@ -38,7 +37,8 @@ namespace cricket {
> // Returns size of ARGB image.
> #define ARGB_SIZE(w, h) (w * h * 4)
> 
>-template <class T> inline std::vector<T> MakeVector(const T a[], size_t s) {
>+template <class T>
>+inline std::vector<T> MakeVector(const T a[], size_t s) {
>   return std::vector<T>(a, a + s);
> }
> #define MAKE_VECTOR(a) cricket::MakeVector(a, arraysize(a))
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/turnutils_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/turnutils_unittest.cc
>index ca1282760bf..e953c2519c1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/turnutils_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/turnutils_unittest.cc
>@@ -50,12 +50,15 @@ TEST(TurnUtilsTest, InvalidTurnSendIndicationMessages) {
> 
>   // Stun Send Indication message with no DATA attribute in message.
>   const uint8_t kTurnSendIndicatinMsgWithNoDataAttribute[] = {
>+      // clang-format off
>+      // clang formatting doesn't respect inline comments.
>       0x00, 0x16, 0x00, 0x08,  // length of
>       0x21, 0x12, 0xA4, 0x42,  // magic cookie
>       '0',  '1',  '2',  '3',   // transaction id
>       '4',  '5',  '6',  '7',  '8',  '9', 'a',  'b',
>       0x00, 0x20, 0x00, 0x04,  // Mapped address.
>       0x00, 0x00, 0x00, 0x00,
>+      // clang-format on
>   };
>   EXPECT_FALSE(
>       UnwrapTurnPacket(kTurnSendIndicatinMsgWithNoDataAttribute,
>@@ -72,6 +75,8 @@ TEST(TurnUtilsTest, ValidTurnSendIndicationMessage) {
>   // A valid STUN indication message with a valid RTP header in data attribute
>   // payload field and no extension bit set.
>   const uint8_t kTurnSendIndicationMsgWithoutRtpExtension[] = {
>+      // clang-format off
>+      // clang formatting doesn't respect inline comments.
>       0x00, 0x16, 0x00, 0x18,  // length of
>       0x21, 0x12, 0xA4, 0x42,  // magic cookie
>       '0',  '1',  '2',  '3',   // transaction id
>@@ -81,11 +86,12 @@ TEST(TurnUtilsTest, ValidTurnSendIndicationMessage) {
>       0x00, 0x13, 0x00, 0x0C,  // Data attribute.
>       0x80, 0x00, 0x00, 0x00,  // RTP packet.
>       0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
>+      // clang-format on
>   };
>-  EXPECT_TRUE(UnwrapTurnPacket(
>-      kTurnSendIndicationMsgWithoutRtpExtension,
>-      sizeof(kTurnSendIndicationMsgWithoutRtpExtension), &content_pos,
>-      &content_size));
>+  EXPECT_TRUE(
>+      UnwrapTurnPacket(kTurnSendIndicationMsgWithoutRtpExtension,
>+                       sizeof(kTurnSendIndicationMsgWithoutRtpExtension),
>+                       &content_pos, &content_size));
>   EXPECT_EQ(12U, content_size);
>   EXPECT_EQ(32U, content_pos);
> }
>@@ -93,15 +99,18 @@ TEST(TurnUtilsTest, ValidTurnSendIndicationMessage) {
> // Verify that parsing of valid TURN Channel Messages.
> TEST(TurnUtilsTest, ValidTurnChannelMessages) {
>   const uint8_t kTurnChannelMsgWithRtpPacket[] = {
>+      // clang-format off
>+      // clang formatting doesn't respect inline comments.
>       0x40, 0x00, 0x00, 0x0C,
>       0x80, 0x00, 0x00, 0x00,  // RTP packet.
>       0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
>+      // clang-format on
>   };
> 
>   size_t content_pos = 0, content_size = 0;
>-  EXPECT_TRUE(UnwrapTurnPacket(
>-      kTurnChannelMsgWithRtpPacket,
>-      sizeof(kTurnChannelMsgWithRtpPacket), &content_pos, &content_size));
>+  EXPECT_TRUE(UnwrapTurnPacket(kTurnChannelMsgWithRtpPacket,
>+                               sizeof(kTurnChannelMsgWithRtpPacket),
>+                               &content_pos, &content_size));
>   EXPECT_EQ(12U, content_size);
>   EXPECT_EQ(4U, content_pos);
> }
>@@ -113,8 +122,8 @@ TEST(TurnUtilsTest, ChannelMessageZeroLength) {
>   EXPECT_TRUE(UnwrapTurnPacket(kTurnChannelMsgWithZeroLength,
>                                sizeof(kTurnChannelMsgWithZeroLength),
>                                &content_pos, &content_size));
>-  EXPECT_EQ(4, content_pos);
>-  EXPECT_EQ(0, content_size);
>+  EXPECT_EQ(4u, content_pos);
>+  EXPECT_EQ(0u, content_size);
> }
> 
> }  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videoadapter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videoadapter.cc
>index 553c2c320ba..c966713848a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videoadapter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videoadapter.cc
>@@ -16,7 +16,7 @@
> #include <limits>
> #include <utility>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> #include "media/base/mediaconstants.h"
> #include "media/base/videocommon.h"
> #include "rtc_base/arraysize.h"
>@@ -194,8 +194,8 @@ bool VideoAdapter::AdaptFrameResolution(int in_width,
>   }
> 
>   // Calculate how the input should be cropped.
>-  if (!requested_format_ ||
>-      requested_format_->width == 0 || requested_format_->height == 0) {
>+  if (!requested_format_ || requested_format_->width == 0 ||
>+      requested_format_->height == 0) {
>     *cropped_width = in_width;
>     *cropped_height = in_height;
>   } else {
>@@ -236,8 +236,8 @@ bool VideoAdapter::AdaptFrameResolution(int in_width,
>   if (scale.numerator != scale.denominator)
>     ++frames_scaled_;
> 
>-  if (previous_width_ && (previous_width_ != *out_width ||
>-                          previous_height_ != *out_height)) {
>+  if (previous_width_ &&
>+      (previous_width_ != *out_width || previous_height_ != *out_height)) {
>     ++adaption_changes_;
>     RTC_LOG(LS_INFO) << "Frame size changed: scaled " << frames_scaled_
>                      << " / out " << frames_out_ << " / in " << frames_in_
>@@ -255,14 +255,15 @@ bool VideoAdapter::AdaptFrameResolution(int in_width,
>   return true;
> }
> 
>-void VideoAdapter::OnOutputFormatRequest(const VideoFormat& format) {
>+void VideoAdapter::OnOutputFormatRequest(
>+    const absl::optional<VideoFormat>& format) {
>   rtc::CritScope cs(&critical_section_);
>   requested_format_ = format;
>-  next_frame_timestamp_ns_ = rtc::nullopt;
>+  next_frame_timestamp_ns_ = absl::nullopt;
> }
> 
> void VideoAdapter::OnResolutionFramerateRequest(
>-    const rtc::Optional<int>& target_pixel_count,
>+    const absl::optional<int>& target_pixel_count,
>     int max_pixel_count,
>     int max_framerate_fps) {
>   rtc::CritScope cs(&critical_section_);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videoadapter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videoadapter.h
>index 5b3d13841b3..4bb65748812 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videoadapter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videoadapter.h
>@@ -11,7 +11,7 @@
> #ifndef MEDIA_BASE_VIDEOADAPTER_H_
> #define MEDIA_BASE_VIDEOADAPTER_H_
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> #include "media/base/videocommon.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/criticalsection.h"
>@@ -48,7 +48,7 @@ class VideoAdapter {
>   // requested aspect ratio is orientation agnostic and will be adjusted to
>   // maintain the input orientation, so it doesn't matter if e.g. 1280x720 or
>   // 720x1280 is requested.
>-  void OnOutputFormatRequest(const VideoFormat& format);
>+  void OnOutputFormatRequest(const absl::optional<VideoFormat>& format);
> 
>   // Requests the output frame size from |AdaptFrameResolution| to have as close
>   // as possible to |target_pixel_count| pixels (if set) but no more than
>@@ -58,7 +58,7 @@ class VideoAdapter {
>   // Set |max_pixel_count| and/or |max_framerate_fps| to
>   // std::numeric_limit<int>::max() if no upper limit is desired.
>   void OnResolutionFramerateRequest(
>-      const rtc::Optional<int>& target_pixel_count,
>+      const absl::optional<int>& target_pixel_count,
>       int max_pixel_count,
>       int max_framerate_fps);
> 
>@@ -75,13 +75,13 @@ class VideoAdapter {
>   // Resolution must be divisible by this factor.
>   const int required_resolution_alignment_;
>   // The target timestamp for the next frame based on requested format.
>-  rtc::Optional<int64_t> next_frame_timestamp_ns_
>+  absl::optional<int64_t> next_frame_timestamp_ns_
>       RTC_GUARDED_BY(critical_section_);
> 
>   // Max number of pixels requested via calls to OnOutputFormatRequest,
>   // OnResolutionRequest respectively.
>   // The adapted output format is the minimum of these.
>-  rtc::Optional<VideoFormat> requested_format_
>+  absl::optional<VideoFormat> requested_format_
>       RTC_GUARDED_BY(critical_section_);
>   int resolution_request_target_pixel_count_ RTC_GUARDED_BY(critical_section_);
>   int resolution_request_max_pixel_count_ RTC_GUARDED_BY(critical_section_);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videoadapter_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videoadapter_unittest.cc
>index 039d1da6366..c5e05b70d69 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videoadapter_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videoadapter_unittest.cc
>@@ -15,7 +15,8 @@
> #include <string>
> #include <vector>
> 
>-#include "media/base/fakevideocapturer.h"
>+#include "absl/memory/memory.h"
>+#include "media/base/fakeframesource.h"
> #include "media/base/mediachannel.h"
> #include "media/base/testutils.h"
> #include "media/base/videoadapter.h"
>@@ -24,29 +25,26 @@
> 
> namespace cricket {
> namespace {
>+const int kWidth = 1280;
>+const int kHeight = 720;
> const int kDefaultFps = 30;
> }  // namespace
> 
> class VideoAdapterTest : public testing::Test {
>  public:
>-  virtual void SetUp() {
>-    capturer_.reset(new FakeVideoCapturer);
>-    capture_format_ = capturer_->GetSupportedFormats()->at(0);
>-    capture_format_.interval = VideoFormat::FpsToInterval(kDefaultFps);
>-
>-    listener_.reset(new VideoCapturerListener(&adapter_));
>-    capturer_->AddOrUpdateSink(listener_.get(), rtc::VideoSinkWants());
>-  }
>-
>-  virtual void TearDown() {
>-    // Explicitly disconnect the VideoCapturer before to avoid data races
>-    // (frames delivered to VideoCapturerListener while it's being destructed).
>-    capturer_->RemoveSink(listener_.get());
>+  void SetUp() override {
>+    capture_format_ = {kWidth, kHeight, VideoFormat::FpsToInterval(kDefaultFps),
>+                       cricket::FOURCC_I420};
>+    frame_source_ = absl::make_unique<FakeFrameSource>(
>+        kWidth, kHeight,
>+        VideoFormat::FpsToInterval(kDefaultFps) / rtc::kNumNanosecsPerMicrosec);
>+
>+    adapter_wrapper_ = absl::make_unique<VideoAdapterWrapper>(&adapter_);
>   }
> 
>  protected:
>-  class VideoCapturerListener
>-      : public rtc::VideoSinkInterface<webrtc::VideoFrame> {
>+  // Wrap a VideoAdapter and collect stats.
>+  class VideoAdapterWrapper {
>    public:
>     struct Stats {
>       int captured_frames;
>@@ -59,7 +57,7 @@ class VideoAdapterTest : public testing::Test {
>       int out_height;
>     };
> 
>-    explicit VideoCapturerListener(VideoAdapter* adapter)
>+    explicit VideoAdapterWrapper(VideoAdapter* adapter)
>         : video_adapter_(adapter),
>           cropped_width_(0),
>           cropped_height_(0),
>@@ -69,8 +67,7 @@ class VideoAdapterTest : public testing::Test {
>           dropped_frames_(0),
>           last_adapt_was_no_op_(false) {}
> 
>-    void OnFrame(const webrtc::VideoFrame& frame) {
>-      rtc::CritScope lock(&crit_);
>+    void AdaptFrame(const webrtc::VideoFrame& frame) {
>       const int in_width = frame.width();
>       const int in_height = frame.height();
>       int cropped_width;
>@@ -95,7 +92,6 @@ class VideoAdapterTest : public testing::Test {
>     }
> 
>     Stats GetStats() {
>-      rtc::CritScope lock(&crit_);
>       Stats stats;
>       stats.captured_frames = captured_frames_;
>       stats.dropped_frames = dropped_frames_;
>@@ -108,7 +104,6 @@ class VideoAdapterTest : public testing::Test {
>     }
> 
>    private:
>-    rtc::CriticalSection crit_;
>     VideoAdapter* video_adapter_;
>     int cropped_width_;
>     int cropped_height_;
>@@ -119,8 +114,7 @@ class VideoAdapterTest : public testing::Test {
>     bool last_adapt_was_no_op_;
>   };
> 
>-
>-  void VerifyAdaptedResolution(const VideoCapturerListener::Stats& stats,
>+  void VerifyAdaptedResolution(const VideoAdapterWrapper::Stats& stats,
>                                int cropped_width,
>                                int cropped_height,
>                                int out_width,
>@@ -131,25 +125,24 @@ class VideoAdapterTest : public testing::Test {
>     EXPECT_EQ(out_height, stats.out_height);
>   }
> 
>-  std::unique_ptr<FakeVideoCapturer> capturer_;
>+  std::unique_ptr<FakeFrameSource> frame_source_;
>   VideoAdapter adapter_;
>   int cropped_width_;
>   int cropped_height_;
>   int out_width_;
>   int out_height_;
>-  std::unique_ptr<VideoCapturerListener> listener_;
>+  std::unique_ptr<VideoAdapterWrapper> adapter_wrapper_;
>   VideoFormat capture_format_;
> };
> 
> // Do not adapt the frame rate or the resolution. Expect no frame drop, no
> // cropping, and no resolution change.
> TEST_F(VideoAdapterTest, AdaptNothing) {
>-  EXPECT_EQ(CS_RUNNING, capturer_->Start(capture_format_));
>   for (int i = 0; i < 10; ++i)
>-    capturer_->CaptureFrame();
>+    adapter_wrapper_->AdaptFrame(frame_source_->GetFrame());
> 
>   // Verify no frame drop and no resolution change.
>-  VideoCapturerListener::Stats stats = listener_->GetStats();
>+  VideoAdapterWrapper::Stats stats = adapter_wrapper_->GetStats();
>   EXPECT_GE(stats.captured_frames, 10);
>   EXPECT_EQ(0, stats.dropped_frames);
>   VerifyAdaptedResolution(stats, capture_format_.width, capture_format_.height,
>@@ -158,15 +151,14 @@ TEST_F(VideoAdapterTest, AdaptNothing) {
> }
> 
> TEST_F(VideoAdapterTest, AdaptZeroInterval) {
>-  VideoFormat format = capturer_->GetSupportedFormats()->at(0);
>+  VideoFormat format = capture_format_;
>   format.interval = 0;
>   adapter_.OnOutputFormatRequest(format);
>-  EXPECT_EQ(CS_RUNNING, capturer_->Start(capture_format_));
>   for (int i = 0; i < 10; ++i)
>-    capturer_->CaptureFrame();
>+    adapter_wrapper_->AdaptFrame(frame_source_->GetFrame());
> 
>   // Verify no crash and that frames aren't dropped.
>-  VideoCapturerListener::Stats stats = listener_->GetStats();
>+  VideoAdapterWrapper::Stats stats = adapter_wrapper_->GetStats();
>   EXPECT_GE(stats.captured_frames, 10);
>   EXPECT_EQ(0, stats.dropped_frames);
>   VerifyAdaptedResolution(stats, capture_format_.width, capture_format_.height,
>@@ -179,49 +171,48 @@ TEST_F(VideoAdapterTest, AdaptFramerateToHalf) {
>   VideoFormat request_format = capture_format_;
>   request_format.interval *= 2;
>   adapter_.OnOutputFormatRequest(request_format);
>-  EXPECT_EQ(CS_RUNNING, capturer_->Start(capture_format_));
> 
>   // Capture 10 frames and verify that every other frame is dropped. The first
>   // frame should not be dropped.
>-  capturer_->CaptureFrame();
>-  EXPECT_GE(listener_->GetStats().captured_frames, 1);
>-  EXPECT_EQ(0, listener_->GetStats().dropped_frames);
>+  adapter_wrapper_->AdaptFrame(frame_source_->GetFrame());
>+  EXPECT_GE(adapter_wrapper_->GetStats().captured_frames, 1);
>+  EXPECT_EQ(0, adapter_wrapper_->GetStats().dropped_frames);
> 
>-  capturer_->CaptureFrame();
>-  EXPECT_GE(listener_->GetStats().captured_frames, 2);
>-  EXPECT_EQ(1, listener_->GetStats().dropped_frames);
>+  adapter_wrapper_->AdaptFrame(frame_source_->GetFrame());
>+  EXPECT_GE(adapter_wrapper_->GetStats().captured_frames, 2);
>+  EXPECT_EQ(1, adapter_wrapper_->GetStats().dropped_frames);
> 
>-  capturer_->CaptureFrame();
>-  EXPECT_GE(listener_->GetStats().captured_frames, 3);
>-  EXPECT_EQ(1, listener_->GetStats().dropped_frames);
>+  adapter_wrapper_->AdaptFrame(frame_source_->GetFrame());
>+  EXPECT_GE(adapter_wrapper_->GetStats().captured_frames, 3);
>+  EXPECT_EQ(1, adapter_wrapper_->GetStats().dropped_frames);
> 
>-  capturer_->CaptureFrame();
>-  EXPECT_GE(listener_->GetStats().captured_frames, 4);
>-  EXPECT_EQ(2, listener_->GetStats().dropped_frames);
>+  adapter_wrapper_->AdaptFrame(frame_source_->GetFrame());
>+  EXPECT_GE(adapter_wrapper_->GetStats().captured_frames, 4);
>+  EXPECT_EQ(2, adapter_wrapper_->GetStats().dropped_frames);
> 
>-  capturer_->CaptureFrame();
>-  EXPECT_GE(listener_->GetStats().captured_frames, 5);
>-  EXPECT_EQ(2, listener_->GetStats().dropped_frames);
>+  adapter_wrapper_->AdaptFrame(frame_source_->GetFrame());
>+  EXPECT_GE(adapter_wrapper_->GetStats().captured_frames, 5);
>+  EXPECT_EQ(2, adapter_wrapper_->GetStats().dropped_frames);
> 
>-  capturer_->CaptureFrame();
>-  EXPECT_GE(listener_->GetStats().captured_frames, 6);
>-  EXPECT_EQ(3, listener_->GetStats().dropped_frames);
>+  adapter_wrapper_->AdaptFrame(frame_source_->GetFrame());
>+  EXPECT_GE(adapter_wrapper_->GetStats().captured_frames, 6);
>+  EXPECT_EQ(3, adapter_wrapper_->GetStats().dropped_frames);
> 
>-  capturer_->CaptureFrame();
>-  EXPECT_GE(listener_->GetStats().captured_frames, 7);
>-  EXPECT_EQ(3, listener_->GetStats().dropped_frames);
>+  adapter_wrapper_->AdaptFrame(frame_source_->GetFrame());
>+  EXPECT_GE(adapter_wrapper_->GetStats().captured_frames, 7);
>+  EXPECT_EQ(3, adapter_wrapper_->GetStats().dropped_frames);
> 
>-  capturer_->CaptureFrame();
>-  EXPECT_GE(listener_->GetStats().captured_frames, 8);
>-  EXPECT_EQ(4, listener_->GetStats().dropped_frames);
>+  adapter_wrapper_->AdaptFrame(frame_source_->GetFrame());
>+  EXPECT_GE(adapter_wrapper_->GetStats().captured_frames, 8);
>+  EXPECT_EQ(4, adapter_wrapper_->GetStats().dropped_frames);
> 
>-  capturer_->CaptureFrame();
>-  EXPECT_GE(listener_->GetStats().captured_frames, 9);
>-  EXPECT_EQ(4, listener_->GetStats().dropped_frames);
>+  adapter_wrapper_->AdaptFrame(frame_source_->GetFrame());
>+  EXPECT_GE(adapter_wrapper_->GetStats().captured_frames, 9);
>+  EXPECT_EQ(4, adapter_wrapper_->GetStats().dropped_frames);
> 
>-  capturer_->CaptureFrame();
>-  EXPECT_GE(listener_->GetStats().captured_frames, 10);
>-  EXPECT_EQ(5, listener_->GetStats().dropped_frames);
>+  adapter_wrapper_->AdaptFrame(frame_source_->GetFrame());
>+  EXPECT_GE(adapter_wrapper_->GetStats().captured_frames, 10);
>+  EXPECT_EQ(5, adapter_wrapper_->GetStats().dropped_frames);
> }
> 
> // Adapt the frame rate to be two thirds of the capture rate at the beginning.
>@@ -231,49 +222,48 @@ TEST_F(VideoAdapterTest, AdaptFramerateToTwoThirds) {
>   VideoFormat request_format = capture_format_;
>   request_format.interval = request_format.interval * 3 / 2;
>   adapter_.OnOutputFormatRequest(request_format);
>-  EXPECT_EQ(CS_RUNNING, capturer_->Start(capture_format_));
> 
>   // Capture 10 frames and verify that every third frame is dropped. The first
>   // frame should not be dropped.
>-  capturer_->CaptureFrame();
>-  EXPECT_GE(listener_->GetStats().captured_frames, 1);
>-  EXPECT_EQ(0, listener_->GetStats().dropped_frames);
>+  adapter_wrapper_->AdaptFrame(frame_source_->GetFrame());
>+  EXPECT_GE(adapter_wrapper_->GetStats().captured_frames, 1);
>+  EXPECT_EQ(0, adapter_wrapper_->GetStats().dropped_frames);
> 
>-  capturer_->CaptureFrame();
>-  EXPECT_GE(listener_->GetStats().captured_frames, 2);
>-  EXPECT_EQ(0, listener_->GetStats().dropped_frames);
>+  adapter_wrapper_->AdaptFrame(frame_source_->GetFrame());
>+  EXPECT_GE(adapter_wrapper_->GetStats().captured_frames, 2);
>+  EXPECT_EQ(0, adapter_wrapper_->GetStats().dropped_frames);
> 
>-  capturer_->CaptureFrame();
>-  EXPECT_GE(listener_->GetStats().captured_frames, 3);
>-  EXPECT_EQ(1, listener_->GetStats().dropped_frames);
>+  adapter_wrapper_->AdaptFrame(frame_source_->GetFrame());
>+  EXPECT_GE(adapter_wrapper_->GetStats().captured_frames, 3);
>+  EXPECT_EQ(1, adapter_wrapper_->GetStats().dropped_frames);
> 
>-  capturer_->CaptureFrame();
>-  EXPECT_GE(listener_->GetStats().captured_frames, 4);
>-  EXPECT_EQ(1, listener_->GetStats().dropped_frames);
>+  adapter_wrapper_->AdaptFrame(frame_source_->GetFrame());
>+  EXPECT_GE(adapter_wrapper_->GetStats().captured_frames, 4);
>+  EXPECT_EQ(1, adapter_wrapper_->GetStats().dropped_frames);
> 
>-  capturer_->CaptureFrame();
>-  EXPECT_GE(listener_->GetStats().captured_frames, 5);
>-  EXPECT_EQ(1, listener_->GetStats().dropped_frames);
>+  adapter_wrapper_->AdaptFrame(frame_source_->GetFrame());
>+  EXPECT_GE(adapter_wrapper_->GetStats().captured_frames, 5);
>+  EXPECT_EQ(1, adapter_wrapper_->GetStats().dropped_frames);
> 
>-  capturer_->CaptureFrame();
>-  EXPECT_GE(listener_->GetStats().captured_frames, 6);
>-  EXPECT_EQ(2, listener_->GetStats().dropped_frames);
>+  adapter_wrapper_->AdaptFrame(frame_source_->GetFrame());
>+  EXPECT_GE(adapter_wrapper_->GetStats().captured_frames, 6);
>+  EXPECT_EQ(2, adapter_wrapper_->GetStats().dropped_frames);
> 
>-  capturer_->CaptureFrame();
>-  EXPECT_GE(listener_->GetStats().captured_frames, 7);
>-  EXPECT_EQ(2, listener_->GetStats().dropped_frames);
>+  adapter_wrapper_->AdaptFrame(frame_source_->GetFrame());
>+  EXPECT_GE(adapter_wrapper_->GetStats().captured_frames, 7);
>+  EXPECT_EQ(2, adapter_wrapper_->GetStats().dropped_frames);
> 
>-  capturer_->CaptureFrame();
>-  EXPECT_GE(listener_->GetStats().captured_frames, 8);
>-  EXPECT_EQ(2, listener_->GetStats().dropped_frames);
>+  adapter_wrapper_->AdaptFrame(frame_source_->GetFrame());
>+  EXPECT_GE(adapter_wrapper_->GetStats().captured_frames, 8);
>+  EXPECT_EQ(2, adapter_wrapper_->GetStats().dropped_frames);
> 
>-  capturer_->CaptureFrame();
>-  EXPECT_GE(listener_->GetStats().captured_frames, 9);
>-  EXPECT_EQ(3, listener_->GetStats().dropped_frames);
>+  adapter_wrapper_->AdaptFrame(frame_source_->GetFrame());
>+  EXPECT_GE(adapter_wrapper_->GetStats().captured_frames, 9);
>+  EXPECT_EQ(3, adapter_wrapper_->GetStats().dropped_frames);
> 
>-  capturer_->CaptureFrame();
>-  EXPECT_GE(listener_->GetStats().captured_frames, 10);
>-  EXPECT_EQ(3, listener_->GetStats().dropped_frames);
>+  adapter_wrapper_->AdaptFrame(frame_source_->GetFrame());
>+  EXPECT_GE(adapter_wrapper_->GetStats().captured_frames, 10);
>+  EXPECT_EQ(3, adapter_wrapper_->GetStats().dropped_frames);
> }
> 
> // Request frame rate twice as high as captured frame rate. Expect no frame
>@@ -282,12 +272,11 @@ TEST_F(VideoAdapterTest, AdaptFramerateHighLimit) {
>   VideoFormat request_format = capture_format_;
>   request_format.interval /= 2;
>   adapter_.OnOutputFormatRequest(request_format);
>-  EXPECT_EQ(CS_RUNNING, capturer_->Start(capture_format_));
>   for (int i = 0; i < 10; ++i)
>-    capturer_->CaptureFrame();
>+    adapter_wrapper_->AdaptFrame(frame_source_->GetFrame());
> 
>   // Verify no frame drop.
>-  EXPECT_EQ(0, listener_->GetStats().dropped_frames);
>+  EXPECT_EQ(0, adapter_wrapper_->GetStats().dropped_frames);
> }
> 
> // After the first timestamp, add a big offset to the timestamps. Expect that
>@@ -370,37 +359,35 @@ TEST_F(VideoAdapterTest, AdaptFramerateTimestampJitter) {
> TEST_F(VideoAdapterTest, AdaptFramerateOntheFly) {
>   VideoFormat request_format = capture_format_;
>   adapter_.OnOutputFormatRequest(request_format);
>-  EXPECT_EQ(CS_RUNNING, capturer_->Start(capture_format_));
>   for (int i = 0; i < 10; ++i)
>-    capturer_->CaptureFrame();
>+    adapter_wrapper_->AdaptFrame(frame_source_->GetFrame());
> 
>   // Verify no frame drop before adaptation.
>-  EXPECT_EQ(0, listener_->GetStats().dropped_frames);
>+  EXPECT_EQ(0, adapter_wrapper_->GetStats().dropped_frames);
> 
>   // Adapat the frame rate.
>   request_format.interval *= 2;
>   adapter_.OnOutputFormatRequest(request_format);
> 
>   for (int i = 0; i < 20; ++i)
>-    capturer_->CaptureFrame();
>+    adapter_wrapper_->AdaptFrame(frame_source_->GetFrame());
> 
>   // Verify frame drop after adaptation.
>-  EXPECT_GT(listener_->GetStats().dropped_frames, 0);
>+  EXPECT_GT(adapter_wrapper_->GetStats().dropped_frames, 0);
> }
> 
> // Do not adapt the frame rate or the resolution. Expect no frame drop, no
> // cropping, and no resolution change.
>-TEST_F(VideoAdapterTest, OnFramerateRequestMax) {
>-  adapter_.OnResolutionFramerateRequest(rtc::nullopt,
>+TEST_F(VideoAdapterTest, AdaptFramerateRequestMax) {
>+  adapter_.OnResolutionFramerateRequest(absl::nullopt,
>                                         std::numeric_limits<int>::max(),
>                                         std::numeric_limits<int>::max());
> 
>-  EXPECT_EQ(CS_RUNNING, capturer_->Start(capture_format_));
>   for (int i = 0; i < 10; ++i)
>-    capturer_->CaptureFrame();
>+    adapter_wrapper_->AdaptFrame(frame_source_->GetFrame());
> 
>   // Verify no frame drop and no resolution change.
>-  VideoCapturerListener::Stats stats = listener_->GetStats();
>+  VideoAdapterWrapper::Stats stats = adapter_wrapper_->GetStats();
>   EXPECT_GE(stats.captured_frames, 10);
>   EXPECT_EQ(0, stats.dropped_frames);
>   VerifyAdaptedResolution(stats, capture_format_.width, capture_format_.height,
>@@ -408,30 +395,28 @@ TEST_F(VideoAdapterTest, OnFramerateRequestMax) {
>   EXPECT_TRUE(stats.last_adapt_was_no_op);
> }
> 
>-TEST_F(VideoAdapterTest, OnFramerateRequestZero) {
>-  adapter_.OnResolutionFramerateRequest(rtc::nullopt,
>+TEST_F(VideoAdapterTest, AdaptFramerateRequestZero) {
>+  adapter_.OnResolutionFramerateRequest(absl::nullopt,
>                                         std::numeric_limits<int>::max(), 0);
>-  EXPECT_EQ(CS_RUNNING, capturer_->Start(capture_format_));
>   for (int i = 0; i < 10; ++i)
>-    capturer_->CaptureFrame();
>+    adapter_wrapper_->AdaptFrame(frame_source_->GetFrame());
> 
>   // Verify no crash and that frames aren't dropped.
>-  VideoCapturerListener::Stats stats = listener_->GetStats();
>+  VideoAdapterWrapper::Stats stats = adapter_wrapper_->GetStats();
>   EXPECT_GE(stats.captured_frames, 10);
>   EXPECT_EQ(10, stats.dropped_frames);
> }
> 
> // Adapt the frame rate to be half of the capture rate at the beginning. Expect
> // the number of dropped frames to be half of the number the captured frames.
>-TEST_F(VideoAdapterTest, OnFramerateRequestHalf) {
>+TEST_F(VideoAdapterTest, AdaptFramerateRequestHalf) {
>   adapter_.OnResolutionFramerateRequest(
>-      rtc::nullopt, std::numeric_limits<int>::max(), kDefaultFps / 2);
>-  EXPECT_EQ(CS_RUNNING, capturer_->Start(capture_format_));
>+      absl::nullopt, std::numeric_limits<int>::max(), kDefaultFps / 2);
>   for (int i = 0; i < 10; ++i)
>-    capturer_->CaptureFrame();
>+    adapter_wrapper_->AdaptFrame(frame_source_->GetFrame());
> 
>   // Verify no crash and that frames aren't dropped.
>-  VideoCapturerListener::Stats stats = listener_->GetStats();
>+  VideoAdapterWrapper::Stats stats = adapter_wrapper_->GetStats();
>   EXPECT_GE(stats.captured_frames, 10);
>   EXPECT_EQ(5, stats.dropped_frames);
>   VerifyAdaptedResolution(stats, capture_format_.width, capture_format_.height,
>@@ -505,12 +490,11 @@ TEST_F(VideoAdapterTest, AdaptResolution) {
>   request_format.width /= 2;
>   request_format.height /= 2;
>   adapter_.OnOutputFormatRequest(request_format);
>-  EXPECT_EQ(CS_RUNNING, capturer_->Start(capture_format_));
>   for (int i = 0; i < 10; ++i)
>-    capturer_->CaptureFrame();
>+    adapter_wrapper_->AdaptFrame(frame_source_->GetFrame());
> 
>   // Verify no frame drop, no cropping, and resolution change.
>-  VideoCapturerListener::Stats stats = listener_->GetStats();
>+  VideoAdapterWrapper::Stats stats = adapter_wrapper_->GetStats();
>   EXPECT_EQ(0, stats.dropped_frames);
>   VerifyAdaptedResolution(stats, capture_format_.width, capture_format_.height,
>                           request_format.width, request_format.height);
>@@ -522,38 +506,36 @@ TEST_F(VideoAdapterTest, AdaptResolution) {
> TEST_F(VideoAdapterTest, AdaptResolutionOnTheFly) {
>   VideoFormat request_format = capture_format_;
>   adapter_.OnOutputFormatRequest(request_format);
>-  EXPECT_EQ(CS_RUNNING, capturer_->Start(capture_format_));
>   for (int i = 0; i < 10; ++i)
>-    capturer_->CaptureFrame();
>+    adapter_wrapper_->AdaptFrame(frame_source_->GetFrame());
> 
>   // Verify no resolution change before adaptation.
>-  VerifyAdaptedResolution(listener_->GetStats(),
>-                          capture_format_.width, capture_format_.height,
>-                          request_format.width, request_format.height);
>+  VerifyAdaptedResolution(adapter_wrapper_->GetStats(), capture_format_.width,
>+                          capture_format_.height, request_format.width,
>+                          request_format.height);
> 
>   // Adapt the frame resolution.
>   request_format.width /= 2;
>   request_format.height /= 2;
>   adapter_.OnOutputFormatRequest(request_format);
>   for (int i = 0; i < 10; ++i)
>-    capturer_->CaptureFrame();
>+    adapter_wrapper_->AdaptFrame(frame_source_->GetFrame());
> 
>   // Verify resolution change after adaptation.
>-  VerifyAdaptedResolution(listener_->GetStats(),
>-                          capture_format_.width, capture_format_.height,
>-                          request_format.width, request_format.height);
>+  VerifyAdaptedResolution(adapter_wrapper_->GetStats(), capture_format_.width,
>+                          capture_format_.height, request_format.width,
>+                          request_format.height);
> }
> 
> // Drop all frames.
> TEST_F(VideoAdapterTest, DropAllFrames) {
>   VideoFormat format;  // with resolution 0x0.
>   adapter_.OnOutputFormatRequest(format);
>-  EXPECT_EQ(CS_RUNNING, capturer_->Start(capture_format_));
>   for (int i = 0; i < 10; ++i)
>-    capturer_->CaptureFrame();
>+    adapter_wrapper_->AdaptFrame(frame_source_->GetFrame());
> 
>   // Verify all frames are dropped.
>-  VideoCapturerListener::Stats stats = listener_->GetStats();
>+  VideoAdapterWrapper::Stats stats = adapter_wrapper_->GetStats();
>   EXPECT_GE(stats.captured_frames, 10);
>   EXPECT_EQ(stats.captured_frames, stats.dropped_frames);
> }
>@@ -750,7 +732,7 @@ TEST_F(VideoAdapterTest, TestOnResolutionRequestInSmallSteps) {
>   EXPECT_EQ(720, out_height_);
> 
>   // Adapt down one step.
>-  adapter_.OnResolutionFramerateRequest(rtc::nullopt, 1280 * 720 - 1,
>+  adapter_.OnResolutionFramerateRequest(absl::nullopt, 1280 * 720 - 1,
>                                         std::numeric_limits<int>::max());
>   EXPECT_TRUE(adapter_.AdaptFrameResolution(1280, 720, 0,
>                                             &cropped_width_, &cropped_height_,
>@@ -761,7 +743,7 @@ TEST_F(VideoAdapterTest, TestOnResolutionRequestInSmallSteps) {
>   EXPECT_EQ(540, out_height_);
> 
>   // Adapt down one step more.
>-  adapter_.OnResolutionFramerateRequest(rtc::nullopt, 960 * 540 - 1,
>+  adapter_.OnResolutionFramerateRequest(absl::nullopt, 960 * 540 - 1,
>                                         std::numeric_limits<int>::max());
>   EXPECT_TRUE(adapter_.AdaptFrameResolution(1280, 720, 0,
>                                             &cropped_width_, &cropped_height_,
>@@ -772,7 +754,7 @@ TEST_F(VideoAdapterTest, TestOnResolutionRequestInSmallSteps) {
>   EXPECT_EQ(360, out_height_);
> 
>   // Adapt down one step more.
>-  adapter_.OnResolutionFramerateRequest(rtc::nullopt, 640 * 360 - 1,
>+  adapter_.OnResolutionFramerateRequest(absl::nullopt, 640 * 360 - 1,
>                                         std::numeric_limits<int>::max());
>   EXPECT_TRUE(adapter_.AdaptFrameResolution(1280, 720, 0,
>                                             &cropped_width_, &cropped_height_,
>@@ -828,7 +810,7 @@ TEST_F(VideoAdapterTest, TestOnResolutionRequestMaxZero) {
>   EXPECT_EQ(1280, out_width_);
>   EXPECT_EQ(720, out_height_);
> 
>-  adapter_.OnResolutionFramerateRequest(rtc::nullopt, 0,
>+  adapter_.OnResolutionFramerateRequest(absl::nullopt, 0,
>                                         std::numeric_limits<int>::max());
>   EXPECT_FALSE(adapter_.AdaptFrameResolution(1280, 720, 0,
>                                              &cropped_width_, &cropped_height_,
>@@ -837,7 +819,7 @@ TEST_F(VideoAdapterTest, TestOnResolutionRequestMaxZero) {
> 
> TEST_F(VideoAdapterTest, TestOnResolutionRequestInLargeSteps) {
>   // Large step down.
>-  adapter_.OnResolutionFramerateRequest(rtc::nullopt, 640 * 360 - 1,
>+  adapter_.OnResolutionFramerateRequest(absl::nullopt, 640 * 360 - 1,
>                                         std::numeric_limits<int>::max());
>   EXPECT_TRUE(adapter_.AdaptFrameResolution(1280, 720, 0,
>                                             &cropped_width_, &cropped_height_,
>@@ -860,7 +842,7 @@ TEST_F(VideoAdapterTest, TestOnResolutionRequestInLargeSteps) {
> }
> 
> TEST_F(VideoAdapterTest, TestOnOutputFormatRequestCapsMaxResolution) {
>-  adapter_.OnResolutionFramerateRequest(rtc::nullopt, 640 * 360 - 1,
>+  adapter_.OnResolutionFramerateRequest(absl::nullopt, 640 * 360 - 1,
>                                         std::numeric_limits<int>::max());
>   EXPECT_TRUE(adapter_.AdaptFrameResolution(1280, 720, 0,
>                                             &cropped_width_, &cropped_height_,
>@@ -880,7 +862,7 @@ TEST_F(VideoAdapterTest, TestOnOutputFormatRequestCapsMaxResolution) {
>   EXPECT_EQ(480, out_width_);
>   EXPECT_EQ(270, out_height_);
> 
>-  adapter_.OnResolutionFramerateRequest(rtc::nullopt, 960 * 720,
>+  adapter_.OnResolutionFramerateRequest(absl::nullopt, 960 * 720,
>                                         std::numeric_limits<int>::max());
>   EXPECT_TRUE(adapter_.AdaptFrameResolution(1280, 720, 0,
>                                             &cropped_width_, &cropped_height_,
>@@ -900,7 +882,7 @@ TEST_F(VideoAdapterTest, TestOnResolutionRequestReset) {
>   EXPECT_EQ(1280, out_width_);
>   EXPECT_EQ(720, out_height_);
> 
>-  adapter_.OnResolutionFramerateRequest(rtc::nullopt, 640 * 360 - 1,
>+  adapter_.OnResolutionFramerateRequest(absl::nullopt, 640 * 360 - 1,
>                                         std::numeric_limits<int>::max());
>   EXPECT_TRUE(adapter_.AdaptFrameResolution(1280, 720, 0,
>                                             &cropped_width_, &cropped_height_,
>@@ -910,7 +892,7 @@ TEST_F(VideoAdapterTest, TestOnResolutionRequestReset) {
>   EXPECT_EQ(480, out_width_);
>   EXPECT_EQ(270, out_height_);
> 
>-  adapter_.OnResolutionFramerateRequest(rtc::nullopt,
>+  adapter_.OnResolutionFramerateRequest(absl::nullopt,
>                                         std::numeric_limits<int>::max(),
>                                         std::numeric_limits<int>::max());
>   EXPECT_TRUE(adapter_.AdaptFrameResolution(1280, 720, 0,
>@@ -936,7 +918,7 @@ TEST_F(VideoAdapterTest, TestCroppingWithResolutionRequest) {
>   EXPECT_EQ(360, out_height_);
> 
>   // Adapt down one step.
>-  adapter_.OnResolutionFramerateRequest(rtc::nullopt, 640 * 360 - 1,
>+  adapter_.OnResolutionFramerateRequest(absl::nullopt, 640 * 360 - 1,
>                                         std::numeric_limits<int>::max());
>   // Expect cropping to 16:9 format and 3/4 scaling.
>   EXPECT_TRUE(adapter_.AdaptFrameResolution(640, 480, 0,
>@@ -948,7 +930,7 @@ TEST_F(VideoAdapterTest, TestCroppingWithResolutionRequest) {
>   EXPECT_EQ(270, out_height_);
> 
>   // Adapt down one step more.
>-  adapter_.OnResolutionFramerateRequest(rtc::nullopt, 480 * 270 - 1,
>+  adapter_.OnResolutionFramerateRequest(absl::nullopt, 480 * 270 - 1,
>                                         std::numeric_limits<int>::max());
>   // Expect cropping to 16:9 format and 1/2 scaling.
>   EXPECT_TRUE(adapter_.AdaptFrameResolution(640, 480, 0,
>@@ -1000,7 +982,7 @@ TEST_F(VideoAdapterTest, TestCroppingOddResolution) {
>   // Ask for 640x360 (16:9 aspect), with 3/16 scaling.
>   adapter_.OnOutputFormatRequest(
>       VideoFormat(640, 360, 0, FOURCC_I420));
>-  adapter_.OnResolutionFramerateRequest(rtc::nullopt,
>+  adapter_.OnResolutionFramerateRequest(absl::nullopt,
>                                         640 * 360 * 3 / 16 * 3 / 16,
>                                         std::numeric_limits<int>::max());
> 
>@@ -1022,8 +1004,7 @@ TEST_F(VideoAdapterTest, TestAdaptToVerySmallResolution) {
>   const int w = 1920;
>   const int h = 1080;
>   adapter_.OnOutputFormatRequest(VideoFormat(w, h, 0, FOURCC_I420));
>-  adapter_.OnResolutionFramerateRequest(rtc::nullopt,
>-                                        w * h * 1 / 16 * 1 / 16,
>+  adapter_.OnResolutionFramerateRequest(absl::nullopt, w * h * 1 / 16 * 1 / 16,
>                                         std::numeric_limits<int>::max());
> 
>   // Send 1920x1080 (16:9 aspect).
>@@ -1070,7 +1051,7 @@ TEST_F(VideoAdapterTest, AdaptFrameResolutionDropWithResolutionRequest) {
>       &cropped_width_, &cropped_height_,
>       &out_width_, &out_height_));
> 
>-  adapter_.OnResolutionFramerateRequest(rtc::nullopt, 640 * 480 - 1,
>+  adapter_.OnResolutionFramerateRequest(absl::nullopt, 640 * 480 - 1,
>                                         std::numeric_limits<int>::max());
> 
>   // Still expect all frames to be dropped
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videobroadcaster.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videobroadcaster.cc
>index d2a9c54116d..a16a327b072 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videobroadcaster.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videobroadcaster.cc
>@@ -21,6 +21,7 @@ namespace rtc {
> VideoBroadcaster::VideoBroadcaster() {
>   thread_checker_.DetachFromThread();
> }
>+VideoBroadcaster::~VideoBroadcaster() = default;
> 
> void VideoBroadcaster::AddOrUpdateSink(
>     VideoSinkInterface<webrtc::VideoFrame>* sink,
>@@ -64,9 +65,9 @@ void VideoBroadcaster::OnFrame(const webrtc::VideoFrame& frame) {
>       continue;
>     }
>     if (sink_pair.wants.black_frames) {
>-      sink_pair.sink->OnFrame(webrtc::VideoFrame(
>-          GetBlackFrameBuffer(frame.width(), frame.height()), frame.rotation(),
>-          frame.timestamp_us()));
>+      sink_pair.sink->OnFrame(
>+          webrtc::VideoFrame(GetBlackFrameBuffer(frame.width(), frame.height()),
>+                             frame.rotation(), frame.timestamp_us()));
>     } else {
>       sink_pair.sink->OnFrame(frame);
>     }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videobroadcaster.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videobroadcaster.h
>index c19f3f75192..119769d2cf7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videobroadcaster.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videobroadcaster.h
>@@ -16,7 +16,7 @@
> #include <vector>
> 
> #include "api/video/video_frame.h"
>-#include "api/videosinkinterface.h"
>+#include "api/video/video_sink_interface.h"
> #include "media/base/videosourcebase.h"
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/thread_checker.h"
>@@ -33,6 +33,7 @@ class VideoBroadcaster : public VideoSourceBase,
>                          public VideoSinkInterface<webrtc::VideoFrame> {
>  public:
>   VideoBroadcaster();
>+  ~VideoBroadcaster() override;
>   void AddOrUpdateSink(VideoSinkInterface<webrtc::VideoFrame>* sink,
>                        const VideoSinkWants& wants) override;
>   void RemoveSink(VideoSinkInterface<webrtc::VideoFrame>* sink) override;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videobroadcaster_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videobroadcaster_unittest.cc
>index 0f2057ebe0e..a8c38848204 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videobroadcaster_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videobroadcaster_unittest.cc
>@@ -10,17 +10,16 @@
> 
> #include <limits>
> 
>-#include "media/base/videobroadcaster.h"
> #include "api/video/i420_buffer.h"
> #include "api/video/video_frame.h"
> #include "media/base/fakevideorenderer.h"
>+#include "media/base/videobroadcaster.h"
> #include "rtc_base/gunit.h"
> 
> using rtc::VideoBroadcaster;
> using rtc::VideoSinkWants;
> using cricket::FakeVideoRenderer;
> 
>-
> TEST(VideoBroadcasterTest, frame_wanted) {
>   VideoBroadcaster broadcaster;
>   EXPECT_FALSE(broadcaster.frame_wanted());
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videocapturer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videocapturer.cc
>index f6e05ba5d51..f8420dc1ad6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videocapturer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videocapturer.cc
>@@ -27,8 +27,6 @@ static const int64_t kMaxDistance = ~(static_cast<int64_t>(1) << 63);
> #ifdef WEBRTC_LINUX
> static const int kYU12Penalty = 16;  // Needs to be higher than MJPG index.
> #endif
>-static const char* kSimulcastScreenshareFieldTrialName =
>-    "WebRTC-SimulcastScreenshare";
> 
> }  // namespace
> 
>@@ -40,6 +38,8 @@ VideoCapturer::VideoCapturer() : apply_rotation_(false) {
>   Construct();
> }
> 
>+VideoCapturer::~VideoCapturer() {}
>+
> void VideoCapturer::Construct() {
>   enable_camera_list_ = false;
>   capture_state_ = CS_STOPPED;
>@@ -65,6 +65,10 @@ bool VideoCapturer::StartCapturing(const VideoFormat& capture_format) {
>   return true;
> }
> 
>+bool VideoCapturer::apply_rotation() {
>+  return apply_rotation_;
>+}
>+
> void VideoCapturer::SetSupportedFormats(
>     const std::vector<VideoFormat>& formats) {
>   // This method is OK to call during initialization on a separate thread.
>@@ -178,10 +182,7 @@ bool VideoCapturer::AdaptFrame(int width,
>     return false;
>   }
> 
>-  bool simulcast_screenshare_enabled =
>-      webrtc::field_trial::IsEnabled(kSimulcastScreenshareFieldTrialName);
>-  if (enable_video_adapter_ &&
>-      (!IsScreencast() || simulcast_screenshare_enabled)) {
>+  if (enable_video_adapter_) {
>     if (!video_adapter_.AdaptFrameResolution(
>             width, height, camera_time_us * rtc::kNumNanosecsPerMicrosec,
>             crop_width, crop_height, out_width, out_height)) {
>@@ -319,9 +320,10 @@ int64_t VideoCapturer::GetFormatDistance(const VideoFormat& desired,
>   // Require camera fps to be at least 96% of what is requested, or higher,
>   // if resolution differs. 96% allows for slight variations in fps. e.g. 29.97
>   if (delta_fps < 0) {
>-    float min_desirable_fps = delta_w ?
>-    VideoFormat::IntervalToFpsFloat(desired.interval) * 28.f / 30.f :
>-    VideoFormat::IntervalToFpsFloat(desired.interval) * 23.f / 30.f;
>+    float min_desirable_fps =
>+        delta_w
>+            ? VideoFormat::IntervalToFpsFloat(desired.interval) * 28.f / 30.f
>+            : VideoFormat::IntervalToFpsFloat(desired.interval) * 23.f / 30.f;
>     delta_fps = -delta_fps;
>     if (supported_fps < min_desirable_fps) {
>       distance |= static_cast<int64_t>(1) << 62;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videocapturer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videocapturer.h
>index 219e03643fb..826e38fa5a1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videocapturer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videocapturer.h
>@@ -20,13 +20,13 @@
> #include <string>
> #include <vector>
> 
>-#include "api/videosourceinterface.h"
>+#include "api/video/video_source_interface.h"
> #include "media/base/videoadapter.h"
> #include "media/base/videobroadcaster.h"
> #include "media/base/videocommon.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/criticalsection.h"
>-#include "rtc_base/sigslot.h"
>+#include "rtc_base/third_party/sigslot/sigslot.h"
> #include "rtc_base/thread_checker.h"
> #include "rtc_base/timestampaligner.h"
> 
>@@ -38,12 +38,12 @@ namespace cricket {
> 
> // Current state of the capturer.
> enum CaptureState {
>-  CS_STOPPED,    // The capturer has been stopped or hasn't started yet.
>-  CS_STARTING,   // The capturer is in the process of starting. Note, it may
>-                 // still fail to start.
>-  CS_RUNNING,    // The capturer has been started successfully and is now
>-                 // capturing.
>-  CS_FAILED,     // The capturer failed to start.
>+  CS_STOPPED,   // The capturer has been stopped or hasn't started yet.
>+  CS_STARTING,  // The capturer is in the process of starting. Note, it may
>+                // still fail to start.
>+  CS_RUNNING,   // The capturer has been started successfully and is now
>+                // capturing.
>+  CS_FAILED,    // The capturer failed to start.
> };
> 
> // VideoCapturer is an abstract class that defines the interfaces for video
>@@ -77,7 +77,7 @@ class VideoCapturer : public sigslot::has_slots<>,
>  public:
>   VideoCapturer();
> 
>-  virtual ~VideoCapturer() {}
>+  ~VideoCapturer() override;
> 
>   // Gets the id of the underlying device, which is available after the capturer
>   // is initialized. Can be used to determine if two capturers reference the
>@@ -126,19 +126,15 @@ class VideoCapturer : public sigslot::has_slots<>,
>   // Note that the width and height of the captured frames may differ from the
>   // capture format. For example, the capture format is HD but the captured
>   // frames may be smaller than HD.
>-  const VideoFormat* GetCaptureFormat() const {
>-    return capture_format_.get();
>-  }
>+  const VideoFormat* GetCaptureFormat() const { return capture_format_.get(); }
> 
>   // Stop the video capturer.
>   virtual void Stop() = 0;
>   // Check if the video capturer is running.
>   virtual bool IsRunning() = 0;
>-  CaptureState capture_state() const {
>-    return capture_state_;
>-  }
>+  CaptureState capture_state() const { return capture_state_; }
> 
>-  virtual bool apply_rotation() { return apply_rotation_; }
>+  virtual bool apply_rotation();
> 
>   // Returns true if the capturer is screencasting. This can be used to
>   // implement screencast specific behavior.
>@@ -154,9 +150,7 @@ class VideoCapturer : public sigslot::has_slots<>,
>   void set_enable_camera_list(bool enable_camera_list) {
>     enable_camera_list_ = enable_camera_list;
>   }
>-  bool enable_camera_list() {
>-    return enable_camera_list_;
>-  }
>+  bool enable_camera_list() { return enable_camera_list_; }
> 
>   // Signal all capture state changes that are not a direct result of calling
>   // Start().
>@@ -227,9 +221,7 @@ class VideoCapturer : public sigslot::has_slots<>,
>   virtual bool GetPreferredFourccs(std::vector<uint32_t>* fourccs) = 0;
> 
>   // mutators to set private attributes
>-  void SetId(const std::string& id) {
>-    id_ = id;
>-  }
>+  void SetId(const std::string& id) { id_ = id; }
> 
>   void SetCaptureFormat(const VideoFormat* format) {
>     capture_format_.reset(format ? new VideoFormat(*format) : NULL);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videocapturer_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videocapturer_unittest.cc
>index 7450694d218..080f4a08cae 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videocapturer_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videocapturer_unittest.cc
>@@ -21,7 +21,7 @@
> #include "rtc_base/logging.h"
> #include "rtc_base/thread.h"
> 
>-using cricket::FakeVideoCapturer;
>+using cricket::FakeVideoCapturerWithTaskQueue;
> 
> namespace {
> 
>@@ -31,9 +31,7 @@ const int kMinHdHeight = 720;
> 
> }  // namespace
> 
>-class VideoCapturerTest
>-    : public sigslot::has_slots<>,
>-      public testing::Test {
>+class VideoCapturerTest : public sigslot::has_slots<>, public testing::Test {
>  public:
>   VideoCapturerTest()
>       : capture_state_(cricket::CS_STOPPED), num_state_changes_(0) {
>@@ -42,8 +40,8 @@ class VideoCapturerTest
> 
>  protected:
>   void InitCapturer(bool is_screencast) {
>-    capturer_ = std::unique_ptr<FakeVideoCapturer>(
>-        new FakeVideoCapturer(is_screencast));
>+    capturer_ = std::unique_ptr<FakeVideoCapturerWithTaskQueue>(
>+        new FakeVideoCapturerWithTaskQueue(is_screencast));
>     capturer_->SignalStateChange.connect(this,
>                                          &VideoCapturerTest::OnStateChange);
>     capturer_->AddOrUpdateSink(&renderer_, rtc::VideoSinkWants());
>@@ -58,7 +56,7 @@ class VideoCapturerTest
>   cricket::CaptureState capture_state() { return capture_state_; }
>   int num_state_changes() { return num_state_changes_; }
> 
>-  std::unique_ptr<cricket::FakeVideoCapturer> capturer_;
>+  std::unique_ptr<FakeVideoCapturerWithTaskQueue> capturer_;
>   cricket::CaptureState capture_state_;
>   int num_state_changes_;
>   cricket::FakeVideoRenderer renderer_;
>@@ -67,11 +65,10 @@ class VideoCapturerTest
> 
> TEST_F(VideoCapturerTest, CaptureState) {
>   EXPECT_TRUE(capturer_->enable_video_adapter());
>-  EXPECT_EQ(cricket::CS_RUNNING, capturer_->Start(cricket::VideoFormat(
>-      640,
>-      480,
>-      cricket::VideoFormat::FpsToInterval(30),
>-      cricket::FOURCC_I420)));
>+  EXPECT_EQ(cricket::CS_RUNNING,
>+            capturer_->Start(cricket::VideoFormat(
>+                640, 480, cricket::VideoFormat::FpsToInterval(30),
>+                cricket::FOURCC_I420)));
>   EXPECT_TRUE(capturer_->IsRunning());
>   EXPECT_EQ_WAIT(cricket::CS_RUNNING, capture_state(), kMsCallbackWait);
>   EXPECT_EQ(1, num_state_changes());
>@@ -327,9 +324,8 @@ TEST_F(VideoCapturerTest, SinkWantsMaxPixelAndMaxPixelCountStepUp) {
> }
> 
> TEST_F(VideoCapturerTest, TestFourccMatch) {
>-  cricket::VideoFormat desired(640, 480,
>-                               cricket::VideoFormat::FpsToInterval(30),
>-                               cricket::FOURCC_ANY);
>+  cricket::VideoFormat desired(
>+      640, 480, cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_ANY);
>   cricket::VideoFormat best;
>   EXPECT_TRUE(capturer_->GetBestCaptureFormat(desired, &best));
>   EXPECT_EQ(640, best.width);
>@@ -344,9 +340,8 @@ TEST_F(VideoCapturerTest, TestFourccMatch) {
> }
> 
> TEST_F(VideoCapturerTest, TestResolutionMatch) {
>-  cricket::VideoFormat desired(1920, 1080,
>-                               cricket::VideoFormat::FpsToInterval(30),
>-                               cricket::FOURCC_ANY);
>+  cricket::VideoFormat desired(
>+      1920, 1080, cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_ANY);
>   cricket::VideoFormat best;
>   // Ask for 1920x1080. Get HD 1280x720 which is the highest.
>   EXPECT_TRUE(capturer_->GetBestCaptureFormat(desired, &best));
>@@ -390,21 +385,22 @@ TEST_F(VideoCapturerTest, TestResolutionMatch) {
> TEST_F(VideoCapturerTest, TestHDResolutionMatch) {
>   // Add some HD formats typical of a mediocre HD webcam.
>   std::vector<cricket::VideoFormat> formats;
>-  formats.push_back(cricket::VideoFormat(320, 240,
>-      cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>-  formats.push_back(cricket::VideoFormat(640, 480,
>-      cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>-  formats.push_back(cricket::VideoFormat(960, 544,
>-      cricket::VideoFormat::FpsToInterval(24), cricket::FOURCC_I420));
>-  formats.push_back(cricket::VideoFormat(1280, 720,
>-      cricket::VideoFormat::FpsToInterval(15), cricket::FOURCC_I420));
>+  formats.push_back(cricket::VideoFormat(
>+      320, 240, cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>+  formats.push_back(cricket::VideoFormat(
>+      640, 480, cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>+  formats.push_back(cricket::VideoFormat(
>+      960, 544, cricket::VideoFormat::FpsToInterval(24), cricket::FOURCC_I420));
>+  formats.push_back(
>+      cricket::VideoFormat(1280, 720, cricket::VideoFormat::FpsToInterval(15),
>+                           cricket::FOURCC_I420));
>   formats.push_back(cricket::VideoFormat(2592, 1944,
>-      cricket::VideoFormat::FpsToInterval(7), cricket::FOURCC_I420));
>+                                         cricket::VideoFormat::FpsToInterval(7),
>+                                         cricket::FOURCC_I420));
>   capturer_->ResetSupportedFormats(formats);
> 
>-  cricket::VideoFormat desired(960, 720,
>-                               cricket::VideoFormat::FpsToInterval(30),
>-                               cricket::FOURCC_ANY);
>+  cricket::VideoFormat desired(
>+      960, 720, cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_ANY);
>   cricket::VideoFormat best;
>   // Ask for 960x720 30 fps. Get qHD 24 fps
>   EXPECT_TRUE(capturer_->GetBestCaptureFormat(desired, &best));
>@@ -484,19 +480,19 @@ TEST_F(VideoCapturerTest, TestHDResolutionMatch) {
> // Some cameras support 320x240 and 320x640. Verify we choose 320x240.
> TEST_F(VideoCapturerTest, TestStrangeFormats) {
>   std::vector<cricket::VideoFormat> supported_formats;
>-  supported_formats.push_back(cricket::VideoFormat(320, 240,
>-      cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>-  supported_formats.push_back(cricket::VideoFormat(320, 640,
>-      cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>+  supported_formats.push_back(cricket::VideoFormat(
>+      320, 240, cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>+  supported_formats.push_back(cricket::VideoFormat(
>+      320, 640, cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>   capturer_->ResetSupportedFormats(supported_formats);
> 
>   std::vector<cricket::VideoFormat> required_formats;
>-  required_formats.push_back(cricket::VideoFormat(320, 240,
>-      cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>-  required_formats.push_back(cricket::VideoFormat(320, 200,
>-      cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>-  required_formats.push_back(cricket::VideoFormat(320, 180,
>-      cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>+  required_formats.push_back(cricket::VideoFormat(
>+      320, 240, cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>+  required_formats.push_back(cricket::VideoFormat(
>+      320, 200, cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>+  required_formats.push_back(cricket::VideoFormat(
>+      320, 180, cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>   cricket::VideoFormat best;
>   for (size_t i = 0; i < required_formats.size(); ++i) {
>     EXPECT_TRUE(capturer_->GetBestCaptureFormat(required_formats[i], &best));
>@@ -505,10 +501,10 @@ TEST_F(VideoCapturerTest, TestStrangeFormats) {
>   }
> 
>   supported_formats.clear();
>-  supported_formats.push_back(cricket::VideoFormat(320, 640,
>-      cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>-  supported_formats.push_back(cricket::VideoFormat(320, 240,
>-      cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>+  supported_formats.push_back(cricket::VideoFormat(
>+      320, 640, cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>+  supported_formats.push_back(cricket::VideoFormat(
>+      320, 240, cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>   capturer_->ResetSupportedFormats(supported_formats);
> 
>   for (size_t i = 0; i < required_formats.size(); ++i) {
>@@ -522,19 +518,19 @@ TEST_F(VideoCapturerTest, TestStrangeFormats) {
> TEST_F(VideoCapturerTest, TestPoorFpsFormats) {
>   // all formats are low framerate
>   std::vector<cricket::VideoFormat> supported_formats;
>-  supported_formats.push_back(cricket::VideoFormat(320, 240,
>-      cricket::VideoFormat::FpsToInterval(10), cricket::FOURCC_I420));
>-  supported_formats.push_back(cricket::VideoFormat(640, 480,
>-      cricket::VideoFormat::FpsToInterval(7), cricket::FOURCC_I420));
>-  supported_formats.push_back(cricket::VideoFormat(1280, 720,
>-      cricket::VideoFormat::FpsToInterval(2), cricket::FOURCC_I420));
>+  supported_formats.push_back(cricket::VideoFormat(
>+      320, 240, cricket::VideoFormat::FpsToInterval(10), cricket::FOURCC_I420));
>+  supported_formats.push_back(cricket::VideoFormat(
>+      640, 480, cricket::VideoFormat::FpsToInterval(7), cricket::FOURCC_I420));
>+  supported_formats.push_back(cricket::VideoFormat(
>+      1280, 720, cricket::VideoFormat::FpsToInterval(2), cricket::FOURCC_I420));
>   capturer_->ResetSupportedFormats(supported_formats);
> 
>   std::vector<cricket::VideoFormat> required_formats;
>-  required_formats.push_back(cricket::VideoFormat(320, 240,
>-      cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>-  required_formats.push_back(cricket::VideoFormat(640, 480,
>-      cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>+  required_formats.push_back(cricket::VideoFormat(
>+      320, 240, cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>+  required_formats.push_back(cricket::VideoFormat(
>+      640, 480, cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>   cricket::VideoFormat best;
>   for (size_t i = 0; i < required_formats.size(); ++i) {
>     EXPECT_TRUE(capturer_->GetBestCaptureFormat(required_formats[i], &best));
>@@ -544,12 +540,12 @@ TEST_F(VideoCapturerTest, TestPoorFpsFormats) {
> 
>   // Increase framerate of 320x240. Expect low fps VGA avoided.
>   supported_formats.clear();
>-  supported_formats.push_back(cricket::VideoFormat(320, 240,
>-      cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>-  supported_formats.push_back(cricket::VideoFormat(640, 480,
>-      cricket::VideoFormat::FpsToInterval(7), cricket::FOURCC_I420));
>-  supported_formats.push_back(cricket::VideoFormat(1280, 720,
>-      cricket::VideoFormat::FpsToInterval(2), cricket::FOURCC_I420));
>+  supported_formats.push_back(cricket::VideoFormat(
>+      320, 240, cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>+  supported_formats.push_back(cricket::VideoFormat(
>+      640, 480, cricket::VideoFormat::FpsToInterval(7), cricket::FOURCC_I420));
>+  supported_formats.push_back(cricket::VideoFormat(
>+      1280, 720, cricket::VideoFormat::FpsToInterval(2), cricket::FOURCC_I420));
>   capturer_->ResetSupportedFormats(supported_formats);
> 
>   for (size_t i = 0; i < required_formats.size(); ++i) {
>@@ -563,12 +559,12 @@ TEST_F(VideoCapturerTest, TestPoorFpsFormats) {
> // the frame rate properly.
> TEST_F(VideoCapturerTest, TestSameSizeDifferentFpsFormats) {
>   std::vector<cricket::VideoFormat> supported_formats;
>-  supported_formats.push_back(cricket::VideoFormat(320, 240,
>-      cricket::VideoFormat::FpsToInterval(10), cricket::FOURCC_I420));
>-  supported_formats.push_back(cricket::VideoFormat(320, 240,
>-      cricket::VideoFormat::FpsToInterval(20), cricket::FOURCC_I420));
>-  supported_formats.push_back(cricket::VideoFormat(320, 240,
>-      cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>+  supported_formats.push_back(cricket::VideoFormat(
>+      320, 240, cricket::VideoFormat::FpsToInterval(10), cricket::FOURCC_I420));
>+  supported_formats.push_back(cricket::VideoFormat(
>+      320, 240, cricket::VideoFormat::FpsToInterval(20), cricket::FOURCC_I420));
>+  supported_formats.push_back(cricket::VideoFormat(
>+      320, 240, cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>   capturer_->ResetSupportedFormats(supported_formats);
> 
>   std::vector<cricket::VideoFormat> required_formats = supported_formats;
>@@ -586,23 +582,24 @@ TEST_F(VideoCapturerTest, TestSameSizeDifferentFpsFormats) {
> TEST_F(VideoCapturerTest, TestFpsFormats) {
>   // We have VGA but low fps. Choose VGA, not HD
>   std::vector<cricket::VideoFormat> supported_formats;
>-  supported_formats.push_back(cricket::VideoFormat(1280, 720,
>-      cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>-  supported_formats.push_back(cricket::VideoFormat(640, 480,
>-      cricket::VideoFormat::FpsToInterval(15), cricket::FOURCC_I420));
>-  supported_formats.push_back(cricket::VideoFormat(640, 400,
>-      cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>-  supported_formats.push_back(cricket::VideoFormat(640, 360,
>-      cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>+  supported_formats.push_back(
>+      cricket::VideoFormat(1280, 720, cricket::VideoFormat::FpsToInterval(30),
>+                           cricket::FOURCC_I420));
>+  supported_formats.push_back(cricket::VideoFormat(
>+      640, 480, cricket::VideoFormat::FpsToInterval(15), cricket::FOURCC_I420));
>+  supported_formats.push_back(cricket::VideoFormat(
>+      640, 400, cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>+  supported_formats.push_back(cricket::VideoFormat(
>+      640, 360, cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>   capturer_->ResetSupportedFormats(supported_formats);
> 
>   std::vector<cricket::VideoFormat> required_formats;
>-  required_formats.push_back(cricket::VideoFormat(640, 480,
>-      cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_ANY));
>-  required_formats.push_back(cricket::VideoFormat(640, 480,
>-      cricket::VideoFormat::FpsToInterval(20), cricket::FOURCC_ANY));
>-  required_formats.push_back(cricket::VideoFormat(640, 480,
>-      cricket::VideoFormat::FpsToInterval(10), cricket::FOURCC_ANY));
>+  required_formats.push_back(cricket::VideoFormat(
>+      640, 480, cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_ANY));
>+  required_formats.push_back(cricket::VideoFormat(
>+      640, 480, cricket::VideoFormat::FpsToInterval(20), cricket::FOURCC_ANY));
>+  required_formats.push_back(cricket::VideoFormat(
>+      640, 480, cricket::VideoFormat::FpsToInterval(10), cricket::FOURCC_ANY));
>   cricket::VideoFormat best;
> 
>   // Expect 30 fps to choose 30 fps format.
>@@ -625,16 +622,17 @@ TEST_F(VideoCapturerTest, TestFpsFormats) {
> 
>   // We have VGA 60 fps and 15 fps. Choose best fps.
>   supported_formats.clear();
>-  supported_formats.push_back(cricket::VideoFormat(1280, 720,
>-      cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>-  supported_formats.push_back(cricket::VideoFormat(640, 480,
>-      cricket::VideoFormat::FpsToInterval(60), cricket::FOURCC_MJPG));
>-  supported_formats.push_back(cricket::VideoFormat(640, 480,
>-      cricket::VideoFormat::FpsToInterval(15), cricket::FOURCC_I420));
>-  supported_formats.push_back(cricket::VideoFormat(640, 400,
>-      cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>-  supported_formats.push_back(cricket::VideoFormat(640, 360,
>-      cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>+  supported_formats.push_back(
>+      cricket::VideoFormat(1280, 720, cricket::VideoFormat::FpsToInterval(30),
>+                           cricket::FOURCC_I420));
>+  supported_formats.push_back(cricket::VideoFormat(
>+      640, 480, cricket::VideoFormat::FpsToInterval(60), cricket::FOURCC_MJPG));
>+  supported_formats.push_back(cricket::VideoFormat(
>+      640, 480, cricket::VideoFormat::FpsToInterval(15), cricket::FOURCC_I420));
>+  supported_formats.push_back(cricket::VideoFormat(
>+      640, 400, cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>+  supported_formats.push_back(cricket::VideoFormat(
>+      640, 360, cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>   capturer_->ResetSupportedFormats(supported_formats);
> 
>   // Expect 30 fps to choose 60 fps format and will set best fps to 60.
>@@ -659,12 +657,12 @@ TEST_F(VideoCapturerTest, TestFpsFormats) {
> TEST_F(VideoCapturerTest, TestRequest16x10_9) {
>   std::vector<cricket::VideoFormat> supported_formats;
>   // We do not support HD, expect 4x3 for 4x3, 16x10, and 16x9 requests.
>-  supported_formats.push_back(cricket::VideoFormat(640, 480,
>-      cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>-  supported_formats.push_back(cricket::VideoFormat(640, 400,
>-      cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>-  supported_formats.push_back(cricket::VideoFormat(640, 360,
>-      cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>+  supported_formats.push_back(cricket::VideoFormat(
>+      640, 480, cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>+  supported_formats.push_back(cricket::VideoFormat(
>+      640, 400, cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>+  supported_formats.push_back(cricket::VideoFormat(
>+      640, 360, cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>   capturer_->ResetSupportedFormats(supported_formats);
> 
>   std::vector<cricket::VideoFormat> required_formats = supported_formats;
>@@ -678,14 +676,14 @@ TEST_F(VideoCapturerTest, TestRequest16x10_9) {
> 
>   // We do not support 16x9 HD, expect 4x3 for 4x3, 16x10, and 16x9 requests.
>   supported_formats.clear();
>-  supported_formats.push_back(cricket::VideoFormat(960, 720,
>-      cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>-  supported_formats.push_back(cricket::VideoFormat(640, 480,
>-      cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>-  supported_formats.push_back(cricket::VideoFormat(640, 400,
>-      cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>-  supported_formats.push_back(cricket::VideoFormat(640, 360,
>-      cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>+  supported_formats.push_back(cricket::VideoFormat(
>+      960, 720, cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>+  supported_formats.push_back(cricket::VideoFormat(
>+      640, 480, cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>+  supported_formats.push_back(cricket::VideoFormat(
>+      640, 400, cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>+  supported_formats.push_back(cricket::VideoFormat(
>+      640, 360, cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>   capturer_->ResetSupportedFormats(supported_formats);
> 
>   // Expect 4x3, 16x10, and 16x9 requests are respected.
>@@ -697,14 +695,15 @@ TEST_F(VideoCapturerTest, TestRequest16x10_9) {
> 
>   // We support 16x9HD, Expect 4x3, 16x10, and 16x9 requests are respected.
>   supported_formats.clear();
>-  supported_formats.push_back(cricket::VideoFormat(1280, 720,
>-      cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>-  supported_formats.push_back(cricket::VideoFormat(640, 480,
>-      cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>-  supported_formats.push_back(cricket::VideoFormat(640, 400,
>-      cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>-  supported_formats.push_back(cricket::VideoFormat(640, 360,
>-      cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>+  supported_formats.push_back(
>+      cricket::VideoFormat(1280, 720, cricket::VideoFormat::FpsToInterval(30),
>+                           cricket::FOURCC_I420));
>+  supported_formats.push_back(cricket::VideoFormat(
>+      640, 480, cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>+  supported_formats.push_back(cricket::VideoFormat(
>+      640, 400, cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>+  supported_formats.push_back(cricket::VideoFormat(
>+      640, 360, cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>   capturer_->ResetSupportedFormats(supported_formats);
> 
>   // Expect 4x3 for 4x3 and 16x10 requests.
>@@ -722,7 +721,8 @@ TEST_F(VideoCapturerTest, TestRequest16x10_9) {
> 
> bool HdFormatInList(const std::vector<cricket::VideoFormat>& formats) {
>   for (std::vector<cricket::VideoFormat>::const_iterator found =
>-           formats.begin(); found != formats.end(); ++found) {
>+           formats.begin();
>+       found != formats.end(); ++found) {
>     if (found->height >= kMinHdHeight) {
>       return true;
>     }
>@@ -734,13 +734,11 @@ TEST_F(VideoCapturerTest, Whitelist) {
>   // The definition of HD only applies to the height. Set the HD width to the
>   // smallest legal number to document this fact in this test.
>   const int kMinHdWidth = 1;
>-  cricket::VideoFormat hd_format(kMinHdWidth,
>-                                 kMinHdHeight,
>+  cricket::VideoFormat hd_format(kMinHdWidth, kMinHdHeight,
>                                  cricket::VideoFormat::FpsToInterval(30),
>                                  cricket::FOURCC_I420);
>-  cricket::VideoFormat vga_format(640, 480,
>-                                  cricket::VideoFormat::FpsToInterval(30),
>-                                  cricket::FOURCC_I420);
>+  cricket::VideoFormat vga_format(
>+      640, 480, cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420);
>   std::vector<cricket::VideoFormat> formats = *capturer_->GetSupportedFormats();
>   formats.push_back(hd_format);
> 
>@@ -760,15 +758,16 @@ TEST_F(VideoCapturerTest, Whitelist) {
> }
> 
> TEST_F(VideoCapturerTest, BlacklistAllFormats) {
>-  cricket::VideoFormat vga_format(640, 480,
>-                                  cricket::VideoFormat::FpsToInterval(30),
>-                                  cricket::FOURCC_I420);
>+  cricket::VideoFormat vga_format(
>+      640, 480, cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420);
>   std::vector<cricket::VideoFormat> supported_formats;
>   // Mock a device that only supports HD formats.
>-  supported_formats.push_back(cricket::VideoFormat(1280, 720,
>-      cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>-  supported_formats.push_back(cricket::VideoFormat(1920, 1080,
>-      cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>+  supported_formats.push_back(
>+      cricket::VideoFormat(1280, 720, cricket::VideoFormat::FpsToInterval(30),
>+                           cricket::FOURCC_I420));
>+  supported_formats.push_back(
>+      cricket::VideoFormat(1920, 1080, cricket::VideoFormat::FpsToInterval(30),
>+                           cricket::FOURCC_I420));
>   capturer_->ResetSupportedFormats(supported_formats);
>   EXPECT_EQ(2u, capturer_->GetSupportedFormats()->size());
>   // Now, enable the list, which would exclude both formats. However, since
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videocommon.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videocommon.cc
>index e5168b55ca4..9def96c4697 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videocommon.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videocommon.cc
>@@ -12,9 +12,9 @@
> 
> #include <limits.h>  // For INT_MAX
> #include <math.h>
>-#include <sstream>
> 
> #include "rtc_base/arraysize.h"
>+#include "rtc_base/strings/string_builder.h"
> 
> namespace cricket {
> 
>@@ -24,20 +24,20 @@ struct FourCCAliasEntry {
> };
> 
> static const FourCCAliasEntry kFourCCAliases[] = {
>-  {FOURCC_IYUV, FOURCC_I420},
>-  {FOURCC_YU16, FOURCC_I422},
>-  {FOURCC_YU24, FOURCC_I444},
>-  {FOURCC_YUYV, FOURCC_YUY2},
>-  {FOURCC_YUVS, FOURCC_YUY2},
>-  {FOURCC_HDYC, FOURCC_UYVY},
>-  {FOURCC_2VUY, FOURCC_UYVY},
>-  {FOURCC_JPEG, FOURCC_MJPG},  // Note: JPEG has DHT while MJPG does not.
>-  {FOURCC_DMB1, FOURCC_MJPG},
>-  {FOURCC_BA81, FOURCC_BGGR},
>-  {FOURCC_RGB3, FOURCC_RAW},
>-  {FOURCC_BGR3, FOURCC_24BG},
>-  {FOURCC_CM32, FOURCC_BGRA},
>-  {FOURCC_CM24, FOURCC_RAW},
>+    {FOURCC_IYUV, FOURCC_I420},
>+    {FOURCC_YU16, FOURCC_I422},
>+    {FOURCC_YU24, FOURCC_I444},
>+    {FOURCC_YUYV, FOURCC_YUY2},
>+    {FOURCC_YUVS, FOURCC_YUY2},
>+    {FOURCC_HDYC, FOURCC_UYVY},
>+    {FOURCC_2VUY, FOURCC_UYVY},
>+    {FOURCC_JPEG, FOURCC_MJPG},  // Note: JPEG has DHT while MJPG does not.
>+    {FOURCC_DMB1, FOURCC_MJPG},
>+    {FOURCC_BA81, FOURCC_BGGR},
>+    {FOURCC_RGB3, FOURCC_RAW},
>+    {FOURCC_BGR3, FOURCC_24BG},
>+    {FOURCC_CM32, FOURCC_BGRA},
>+    {FOURCC_CM24, FOURCC_RAW},
> };
> 
> uint32_t CanonicalFourCC(uint32_t fourcc) {
>@@ -62,7 +62,7 @@ const int64_t VideoFormat::kMinimumInterval;  // Initialized in header.
> std::string VideoFormat::ToString() const {
>   std::string fourcc_name = GetFourccName(fourcc) + " ";
>   for (std::string::const_iterator i = fourcc_name.begin();
>-      i < fourcc_name.end(); ++i) {
>+       i < fourcc_name.end(); ++i) {
>     // Test character is printable; Avoid isprint() which asserts on negatives.
>     if (*i < 32 || *i >= 127) {
>       fourcc_name = "";
>@@ -70,10 +70,11 @@ std::string VideoFormat::ToString() const {
>     }
>   }
> 
>-  std::ostringstream ss;
>-  ss << fourcc_name << width << "x" << height << "x"
>+  char buf[256];
>+  rtc::SimpleStringBuilder sb(buf);
>+  sb << fourcc_name << width << "x" << height << "x"
>      << IntervalToFpsFloat(interval);
>-  return ss.str();
>+  return sb.str();
> }
> 
> }  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videocommon.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videocommon.h
>index 7415883eca4..0ffd989cb65 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videocommon.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videocommon.h
>@@ -30,8 +30,7 @@ const uint32_t kDummyVideoSsrc = 0xFFFFFFFF;
> 
> // Minimum interval is 10k fps.
> #define FPS_TO_INTERVAL(fps) \
>-    (fps ? rtc::kNumNanosecsPerSec / fps : \
>-    rtc::kNumNanosecsPerSec / 10000)
>+  (fps ? rtc::kNumNanosecsPerSec / fps : rtc::kNumNanosecsPerSec / 10000)
> 
> //////////////////////////////////////////////////////////////////////////////
> // Definition of FourCC codes
>@@ -39,7 +38,7 @@ const uint32_t kDummyVideoSsrc = 0xFFFFFFFF;
> // Convert four characters to a FourCC code.
> // Needs to be a macro otherwise the OS X compiler complains when the kFormat*
> // constants are used in a switch.
>-#define CRICKET_FOURCC(a, b, c, d)                                 \
>+#define CRICKET_FOURCC(a, b, c, d)                                \
>   ((static_cast<uint32_t>(a)) | (static_cast<uint32_t>(b) << 8) | \
>    (static_cast<uint32_t>(c) << 16) | (static_cast<uint32_t>(d) << 24))
> // Some pages discussing FourCC codes:
>@@ -73,7 +72,7 @@ enum FourCC {
>   FOURCC_BGRA = CRICKET_FOURCC('B', 'G', 'R', 'A'),
>   FOURCC_ABGR = CRICKET_FOURCC('A', 'B', 'G', 'R'),
>   FOURCC_24BG = CRICKET_FOURCC('2', '4', 'B', 'G'),
>-  FOURCC_RAW  = CRICKET_FOURCC('r', 'a', 'w', ' '),
>+  FOURCC_RAW = CRICKET_FOURCC('r', 'a', 'w', ' '),
>   FOURCC_RGBA = CRICKET_FOURCC('R', 'G', 'B', 'A'),
>   FOURCC_RGBP = CRICKET_FOURCC('R', 'G', 'B', 'P'),  // bgr565.
>   FOURCC_RGBO = CRICKET_FOURCC('R', 'G', 'B', 'O'),  // abgr1555.
>@@ -144,8 +143,8 @@ inline std::string GetFourccName(uint32_t fourcc) {
> 
> // VideoFormat with Plain Old Data for global variables.
> struct VideoFormatPod {
>-  int width;  // Number of pixels.
>-  int height;  // Number of pixels.
>+  int width;         // Number of pixels.
>+  int height;        // Number of pixels.
>   int64_t interval;  // Nanoseconds.
>   uint32_t fourcc;  // Color space. FOURCC_ANY means that any color space is OK.
> };
>@@ -154,9 +153,7 @@ struct VideoFormat : VideoFormatPod {
>   static const int64_t kMinimumInterval =
>       rtc::kNumNanosecsPerSec / 10000;  // 10k fps.
> 
>-  VideoFormat() {
>-    Construct(0, 0, 0, 0);
>-  }
>+  VideoFormat() { Construct(0, 0, 0, 0); }
> 
>   VideoFormat(int w, int h, int64_t interval_ns, uint32_t cc) {
>     Construct(w, h, interval_ns, cc);
>@@ -189,12 +186,12 @@ struct VideoFormat : VideoFormatPod {
>       return 0.f;
>     }
>     return static_cast<float>(rtc::kNumNanosecsPerSec) /
>-        static_cast<float>(interval);
>+           static_cast<float>(interval);
>   }
> 
>   bool operator==(const VideoFormat& format) const {
>     return width == format.width && height == format.height &&
>-        interval == format.interval && fourcc == format.fourcc;
>+           interval == format.interval && fourcc == format.fourcc;
>   }
> 
>   bool operator!=(const VideoFormat& format) const {
>@@ -203,10 +200,10 @@ struct VideoFormat : VideoFormatPod {
> 
>   bool operator<(const VideoFormat& format) const {
>     return (fourcc < format.fourcc) ||
>-        (fourcc == format.fourcc && width < format.width) ||
>-        (fourcc == format.fourcc && width == format.width &&
>+           (fourcc == format.fourcc && width < format.width) ||
>+           (fourcc == format.fourcc && width == format.width &&
>             height < format.height) ||
>-        (fourcc == format.fourcc && width == format.width &&
>+           (fourcc == format.fourcc && width == format.width &&
>             height == format.height && interval > format.interval);
>   }
> 
>@@ -219,7 +216,7 @@ struct VideoFormat : VideoFormatPod {
>   // and frame rate.
>   bool IsPixelRateLess(const VideoFormat& format) const {
>     return width * height * framerate() <
>-        format.width * format.height * format.framerate();
>+           format.width * format.height * format.framerate();
>   }
> 
>   // Get a string presentation in the form of "fourcc width x height x fps"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videoengine_unittest.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videoengine_unittest.h
>deleted file mode 100644
>index acbfc8d69a2..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videoengine_unittest.h
>+++ /dev/null
>@@ -1,944 +0,0 @@
>-/*
>- *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MEDIA_BASE_VIDEOENGINE_UNITTEST_H_  // NOLINT
>-#define MEDIA_BASE_VIDEOENGINE_UNITTEST_H_
>-
>-#include <memory>
>-#include <string>
>-#include <vector>
>-
>-#include "call/call.h"
>-#include "logging/rtc_event_log/rtc_event_log.h"
>-#include "media/base/fakenetworkinterface.h"
>-#include "media/base/fakevideocapturer.h"
>-#include "media/base/fakevideorenderer.h"
>-#include "media/base/mediachannel.h"
>-#include "media/base/streamparams.h"
>-#include "media/engine/fakewebrtccall.h"
>-#include "rtc_base/bytebuffer.h"
>-#include "rtc_base/gunit.h"
>-#include "rtc_base/timeutils.h"
>-
>-namespace cricket {
>-class WebRtcVideoEncoderFactory;
>-class WebRtcVideoDecoderFactory;
>-}  // namespace cricket
>-
>-#define EXPECT_FRAME_WAIT(c, w, h, t) \
>-  EXPECT_EQ_WAIT((c), renderer_.num_rendered_frames(), (t)); \
>-  EXPECT_EQ((w), renderer_.width()); \
>-  EXPECT_EQ((h), renderer_.height()); \
>-  EXPECT_EQ(0, renderer_.errors()); \
>-
>-#define EXPECT_FRAME_ON_RENDERER_WAIT(r, c, w, h, t) \
>-  EXPECT_EQ_WAIT((c), (r).num_rendered_frames(), (t)); \
>-  EXPECT_EQ((w), (r).width()); \
>-  EXPECT_EQ((h), (r).height()); \
>-  EXPECT_EQ(0, (r).errors()); \
>-
>-#define EXPECT_GT_FRAME_ON_RENDERER_WAIT(r, c, w, h, t) \
>-  EXPECT_TRUE_WAIT((r).num_rendered_frames() >= (c) && \
>-                   (w) == (r).width() && \
>-                   (h) == (r).height(), (t)); \
>-  EXPECT_EQ(0, (r).errors());
>-
>-static const uint32_t kTimeout = 5000U;
>-static const uint32_t kDefaultReceiveSsrc = 0;
>-static const uint32_t kSsrc = 1234u;
>-static const uint32_t kRtxSsrc = 4321u;
>-static const uint32_t kSsrcs4[] = {1, 2, 3, 4};
>-static const int kVideoWidth = 640;
>-static const int kVideoHeight = 360;
>-static const int kFramerate = 30;
>-
>-inline bool IsEqualCodec(const cricket::VideoCodec& a,
>-                         const cricket::VideoCodec& b) {
>-  return a.id == b.id && a.name == b.name;
>-}
>-
>-template<class E, class C>
>-class VideoMediaChannelTest : public testing::Test,
>-                              public sigslot::has_slots<> {
>- protected:
>-  VideoMediaChannelTest<E, C>()
>-      : call_(webrtc::Call::Create(webrtc::Call::Config(&event_log_))),
>-        engine_(std::unique_ptr<cricket::WebRtcVideoEncoderFactory>(),
>-                std::unique_ptr<cricket::WebRtcVideoDecoderFactory>()) {}
>-
>-  virtual cricket::VideoCodec DefaultCodec() = 0;
>-
>-  virtual cricket::StreamParams DefaultSendStreamParams() {
>-    return cricket::StreamParams::CreateLegacy(kSsrc);
>-  }
>-
>-  virtual void SetUp() {
>-    cricket::MediaConfig media_config;
>-    // Disabling cpu overuse detection actually disables quality scaling too; it
>-    // implies DegradationPreference kMaintainResolution. Automatic scaling
>-    // needs to be disabled, otherwise, tests which check the size of received
>-    // frames become flaky.
>-    media_config.video.enable_cpu_overuse_detection = false;
>-    channel_.reset(engine_.CreateChannel(call_.get(), media_config,
>-                                         cricket::VideoOptions()));
>-    channel_->OnReadyToSend(true);
>-    EXPECT_TRUE(channel_.get() != NULL);
>-    network_interface_.SetDestination(channel_.get());
>-    channel_->SetInterface(&network_interface_);
>-    cricket::VideoRecvParameters parameters;
>-    parameters.codecs = engine_.codecs();
>-    channel_->SetRecvParameters(parameters);
>-    EXPECT_TRUE(channel_->AddSendStream(DefaultSendStreamParams()));
>-    video_capturer_.reset(CreateFakeVideoCapturer());
>-    cricket::VideoFormat format(640, 480,
>-                                cricket::VideoFormat::FpsToInterval(kFramerate),
>-                                cricket::FOURCC_I420);
>-    EXPECT_EQ(cricket::CS_RUNNING, video_capturer_->Start(format));
>-    EXPECT_TRUE(
>-        channel_->SetVideoSend(kSsrc, true, nullptr, video_capturer_.get()));
>-  }
>-
>-  virtual cricket::FakeVideoCapturer* CreateFakeVideoCapturer() {
>-    return new cricket::FakeVideoCapturer();
>-  }
>-
>-  // Utility method to setup an additional stream to send and receive video.
>-  // Used to test send and recv between two streams.
>-  void SetUpSecondStream() {
>-    SetUpSecondStreamWithNoRecv();
>-    // Setup recv for second stream.
>-    EXPECT_TRUE(channel_->AddRecvStream(
>-        cricket::StreamParams::CreateLegacy(kSsrc + 2)));
>-    // Make the second renderer available for use by a new stream.
>-    EXPECT_TRUE(channel_->SetSink(kSsrc + 2, &renderer2_));
>-  }
>-  // Setup an additional stream just to send video. Defer add recv stream.
>-  // This is required if you want to test unsignalled recv of video rtp packets.
>-  void SetUpSecondStreamWithNoRecv() {
>-    // SetUp() already added kSsrc make sure duplicate SSRCs cant be added.
>-    EXPECT_TRUE(channel_->AddRecvStream(
>-        cricket::StreamParams::CreateLegacy(kSsrc)));
>-    EXPECT_TRUE(channel_->SetSink(kSsrc, &renderer_));
>-    EXPECT_FALSE(channel_->AddSendStream(
>-        cricket::StreamParams::CreateLegacy(kSsrc)));
>-    EXPECT_TRUE(channel_->AddSendStream(
>-        cricket::StreamParams::CreateLegacy(kSsrc + 2)));
>-    // We dont add recv for the second stream.
>-
>-    // Setup the receive and renderer for second stream after send.
>-    video_capturer_2_.reset(CreateFakeVideoCapturer());
>-    cricket::VideoFormat format(640, 480,
>-                                cricket::VideoFormat::FpsToInterval(kFramerate),
>-                                cricket::FOURCC_I420);
>-    EXPECT_EQ(cricket::CS_RUNNING, video_capturer_2_->Start(format));
>-
>-    EXPECT_TRUE(channel_->SetVideoSend(kSsrc + 2, true, nullptr,
>-                                       video_capturer_2_.get()));
>-  }
>-  virtual void TearDown() {
>-    channel_.reset();
>-  }
>-  bool SetDefaultCodec() {
>-    return SetOneCodec(DefaultCodec());
>-  }
>-
>-  bool SetOneCodec(int pt, const char* name) {
>-    return SetOneCodec(cricket::VideoCodec(pt, name));
>-  }
>-  bool SetOneCodec(const cricket::VideoCodec& codec) {
>-    cricket::VideoFormat capture_format(
>-        kVideoWidth, kVideoHeight,
>-        cricket::VideoFormat::FpsToInterval(kFramerate), cricket::FOURCC_I420);
>-
>-    if (video_capturer_) {
>-      EXPECT_EQ(cricket::CS_RUNNING, video_capturer_->Start(capture_format));
>-    }
>-    if (video_capturer_2_) {
>-      EXPECT_EQ(cricket::CS_RUNNING, video_capturer_2_->Start(capture_format));
>-    }
>-
>-    bool sending = channel_->sending();
>-    bool success = SetSend(false);
>-    if (success) {
>-      cricket::VideoSendParameters parameters;
>-      parameters.codecs.push_back(codec);
>-      success = channel_->SetSendParameters(parameters);
>-    }
>-    if (success) {
>-      success = SetSend(sending);
>-    }
>-    return success;
>-  }
>-  bool SetSend(bool send) {
>-    return channel_->SetSend(send);
>-  }
>-  int DrainOutgoingPackets() {
>-    int packets = 0;
>-    do {
>-      packets = NumRtpPackets();
>-      // 100 ms should be long enough.
>-      rtc::Thread::Current()->ProcessMessages(100);
>-    } while (NumRtpPackets() > packets);
>-    return NumRtpPackets();
>-  }
>-  bool SendFrame() {
>-    if (video_capturer_2_) {
>-      video_capturer_2_->CaptureFrame();
>-    }
>-    return video_capturer_.get() &&
>-        video_capturer_->CaptureFrame();
>-  }
>-  bool WaitAndSendFrame(int wait_ms) {
>-    bool ret = rtc::Thread::Current()->ProcessMessages(wait_ms);
>-    ret &= SendFrame();
>-    return ret;
>-  }
>-  // Sends frames and waits for the decoder to be fully initialized.
>-  // Returns the number of frames that were sent.
>-  int WaitForDecoder() {
>-#if defined(HAVE_OPENMAX)
>-    // Send enough frames for the OpenMAX decoder to continue processing, and
>-    // return the number of frames sent.
>-    // Send frames for a full kTimeout's worth of 15fps video.
>-    int frame_count = 0;
>-    while (frame_count < static_cast<int>(kTimeout) / 66) {
>-      EXPECT_TRUE(WaitAndSendFrame(66));
>-      ++frame_count;
>-    }
>-    return frame_count;
>-#else
>-    return 0;
>-#endif
>-  }
>-  bool SendCustomVideoFrame(int w, int h) {
>-    if (!video_capturer_.get()) return false;
>-    return video_capturer_->CaptureCustomFrame(w, h, cricket::FOURCC_I420);
>-  }
>-  int NumRtpBytes() {
>-    return network_interface_.NumRtpBytes();
>-  }
>-  int NumRtpBytes(uint32_t ssrc) {
>-    return network_interface_.NumRtpBytes(ssrc);
>-  }
>-  int NumRtpPackets() {
>-    return network_interface_.NumRtpPackets();
>-  }
>-  int NumRtpPackets(uint32_t ssrc) {
>-    return network_interface_.NumRtpPackets(ssrc);
>-  }
>-  int NumSentSsrcs() {
>-    return network_interface_.NumSentSsrcs();
>-  }
>-  const rtc::CopyOnWriteBuffer* GetRtpPacket(int index) {
>-    return network_interface_.GetRtpPacket(index);
>-  }
>-  int NumRtcpPackets() {
>-    return network_interface_.NumRtcpPackets();
>-  }
>-  const rtc::CopyOnWriteBuffer* GetRtcpPacket(int index) {
>-    return network_interface_.GetRtcpPacket(index);
>-  }
>-  static int GetPayloadType(const rtc::CopyOnWriteBuffer* p) {
>-    int pt = -1;
>-    ParseRtpPacket(p, NULL, &pt, NULL, NULL, NULL, NULL);
>-    return pt;
>-  }
>-  static bool ParseRtpPacket(const rtc::CopyOnWriteBuffer* p,
>-                             bool* x,
>-                             int* pt,
>-                             int* seqnum,
>-                             uint32_t* tstamp,
>-                             uint32_t* ssrc,
>-                             std::string* payload) {
>-    rtc::ByteBufferReader buf(p->data<char>(), p->size());
>-    uint8_t u08 = 0;
>-    uint16_t u16 = 0;
>-    uint32_t u32 = 0;
>-
>-    // Read X and CC fields.
>-    if (!buf.ReadUInt8(&u08)) return false;
>-    bool extension = ((u08 & 0x10) != 0);
>-    uint8_t cc = (u08 & 0x0F);
>-    if (x) *x = extension;
>-
>-    // Read PT field.
>-    if (!buf.ReadUInt8(&u08)) return false;
>-    if (pt) *pt = (u08 & 0x7F);
>-
>-    // Read Sequence Number field.
>-    if (!buf.ReadUInt16(&u16)) return false;
>-    if (seqnum) *seqnum = u16;
>-
>-    // Read Timestamp field.
>-    if (!buf.ReadUInt32(&u32)) return false;
>-    if (tstamp) *tstamp = u32;
>-
>-    // Read SSRC field.
>-    if (!buf.ReadUInt32(&u32)) return false;
>-    if (ssrc) *ssrc = u32;
>-
>-    // Skip CSRCs.
>-    for (uint8_t i = 0; i < cc; ++i) {
>-      if (!buf.ReadUInt32(&u32)) return false;
>-    }
>-
>-    // Skip extension header.
>-    if (extension) {
>-      // Read Profile-specific extension header ID
>-      if (!buf.ReadUInt16(&u16)) return false;
>-
>-      // Read Extension header length
>-      if (!buf.ReadUInt16(&u16)) return false;
>-      uint16_t ext_header_len = u16;
>-
>-      // Read Extension header
>-      for (uint16_t i = 0; i < ext_header_len; ++i) {
>-        if (!buf.ReadUInt32(&u32)) return false;
>-      }
>-    }
>-
>-    if (payload) {
>-      return buf.ReadString(payload, buf.Length());
>-    }
>-    return true;
>-  }
>-
>-  // Parse all RTCP packet, from start_index to stop_index, and count how many
>-  // FIR (PT=206 and FMT=4 according to RFC 5104). If successful, set the count
>-  // and return true.
>-  bool CountRtcpFir(int start_index, int stop_index, int* fir_count) {
>-    int count = 0;
>-    for (int i = start_index; i < stop_index; ++i) {
>-      std::unique_ptr<const rtc::CopyOnWriteBuffer> p(GetRtcpPacket(i));
>-      rtc::ByteBufferReader buf(p->data<char>(), p->size());
>-      size_t total_len = 0;
>-      // The packet may be a compound RTCP packet.
>-      while (total_len < p->size()) {
>-        // Read FMT, type and length.
>-        uint8_t fmt = 0;
>-        uint8_t type = 0;
>-        uint16_t length = 0;
>-        if (!buf.ReadUInt8(&fmt)) return false;
>-        fmt &= 0x1F;
>-        if (!buf.ReadUInt8(&type)) return false;
>-        if (!buf.ReadUInt16(&length)) return false;
>-        buf.Consume(length * 4);  // Skip RTCP data.
>-        total_len += (length + 1) * 4;
>-        if ((192 == type) || ((206 == type) && (4 == fmt))) {
>-          ++count;
>-        }
>-      }
>-    }
>-
>-    if (fir_count) {
>-      *fir_count = count;
>-    }
>-    return true;
>-  }
>-
>-  // Test that SetSend works.
>-  void SetSend() {
>-    EXPECT_FALSE(channel_->sending());
>-    EXPECT_TRUE(
>-        channel_->SetVideoSend(kSsrc, true, nullptr, video_capturer_.get()));
>-    EXPECT_TRUE(SetOneCodec(DefaultCodec()));
>-    EXPECT_FALSE(channel_->sending());
>-    EXPECT_TRUE(SetSend(true));
>-    EXPECT_TRUE(channel_->sending());
>-    EXPECT_TRUE(SendFrame());
>-    EXPECT_TRUE_WAIT(NumRtpPackets() > 0, kTimeout);
>-    EXPECT_TRUE(SetSend(false));
>-    EXPECT_FALSE(channel_->sending());
>-  }
>-  // Test that SetSend fails without codecs being set.
>-  void SetSendWithoutCodecs() {
>-    EXPECT_FALSE(channel_->sending());
>-    EXPECT_FALSE(SetSend(true));
>-    EXPECT_FALSE(channel_->sending());
>-  }
>-  // Test that we properly set the send and recv buffer sizes by the time
>-  // SetSend is called.
>-  void SetSendSetsTransportBufferSizes() {
>-    EXPECT_TRUE(SetOneCodec(DefaultCodec()));
>-    EXPECT_TRUE(SetSend(true));
>-    EXPECT_EQ(64 * 1024, network_interface_.sendbuf_size());
>-    EXPECT_EQ(64 * 1024, network_interface_.recvbuf_size());
>-  }
>-  // Tests that we can send frames and the right payload type is used.
>-  void Send(const cricket::VideoCodec& codec) {
>-    EXPECT_TRUE(SetOneCodec(codec));
>-    EXPECT_TRUE(SetSend(true));
>-    EXPECT_TRUE(SendFrame());
>-    EXPECT_TRUE_WAIT(NumRtpPackets() > 0, kTimeout);
>-    std::unique_ptr<const rtc::CopyOnWriteBuffer> p(GetRtpPacket(0));
>-    EXPECT_EQ(codec.id, GetPayloadType(p.get()));
>-  }
>-  // Tests that we can send and receive frames.
>-  void SendAndReceive(const cricket::VideoCodec& codec) {
>-    EXPECT_TRUE(SetOneCodec(codec));
>-    EXPECT_TRUE(SetSend(true));
>-    EXPECT_TRUE(channel_->SetSink(kDefaultReceiveSsrc, &renderer_));
>-    EXPECT_EQ(0, renderer_.num_rendered_frames());
>-    EXPECT_TRUE(SendFrame());
>-    EXPECT_FRAME_WAIT(1, kVideoWidth, kVideoHeight, kTimeout);
>-    std::unique_ptr<const rtc::CopyOnWriteBuffer> p(GetRtpPacket(0));
>-    EXPECT_EQ(codec.id, GetPayloadType(p.get()));
>-  }
>-  void SendReceiveManyAndGetStats(const cricket::VideoCodec& codec,
>-                                  int duration_sec, int fps) {
>-    EXPECT_TRUE(SetOneCodec(codec));
>-    EXPECT_TRUE(SetSend(true));
>-    EXPECT_TRUE(channel_->SetSink(kDefaultReceiveSsrc, &renderer_));
>-    EXPECT_EQ(0, renderer_.num_rendered_frames());
>-    for (int i = 0; i < duration_sec; ++i) {
>-      for (int frame = 1; frame <= fps; ++frame) {
>-        EXPECT_TRUE(WaitAndSendFrame(1000 / fps));
>-        EXPECT_FRAME_WAIT(frame + i * fps, kVideoWidth, kVideoHeight, kTimeout);
>-      }
>-    }
>-    std::unique_ptr<const rtc::CopyOnWriteBuffer> p(GetRtpPacket(0));
>-    EXPECT_EQ(codec.id, GetPayloadType(p.get()));
>-  }
>-
>-  // Test that stats work properly for a 1-1 call.
>-  void GetStats() {
>-    const int kDurationSec = 3;
>-    const int kFps = 10;
>-    SendReceiveManyAndGetStats(DefaultCodec(), kDurationSec, kFps);
>-
>-    cricket::VideoMediaInfo info;
>-    EXPECT_TRUE(channel_->GetStats(&info));
>-
>-    ASSERT_EQ(1U, info.senders.size());
>-    // TODO(whyuan): bytes_sent and bytes_rcvd are different. Are both payload?
>-    // For webrtc, bytes_sent does not include the RTP header length.
>-    EXPECT_GT(info.senders[0].bytes_sent, 0);
>-    EXPECT_EQ(NumRtpPackets(), info.senders[0].packets_sent);
>-    EXPECT_EQ(0.0, info.senders[0].fraction_lost);
>-    ASSERT_TRUE(info.senders[0].codec_payload_type);
>-    EXPECT_EQ(DefaultCodec().id, *info.senders[0].codec_payload_type);
>-    EXPECT_EQ(0, info.senders[0].firs_rcvd);
>-    EXPECT_EQ(0, info.senders[0].plis_rcvd);
>-    EXPECT_EQ(0, info.senders[0].nacks_rcvd);
>-    EXPECT_EQ(kVideoWidth, info.senders[0].send_frame_width);
>-    EXPECT_EQ(kVideoHeight, info.senders[0].send_frame_height);
>-    EXPECT_GT(info.senders[0].framerate_input, 0);
>-    EXPECT_GT(info.senders[0].framerate_sent, 0);
>-
>-    EXPECT_EQ(1U, info.send_codecs.count(DefaultCodec().id));
>-    EXPECT_EQ(DefaultCodec().ToCodecParameters(),
>-              info.send_codecs[DefaultCodec().id]);
>-
>-    ASSERT_EQ(1U, info.receivers.size());
>-    EXPECT_EQ(1U, info.senders[0].ssrcs().size());
>-    EXPECT_EQ(1U, info.receivers[0].ssrcs().size());
>-    EXPECT_EQ(info.senders[0].ssrcs()[0], info.receivers[0].ssrcs()[0]);
>-    ASSERT_TRUE(info.receivers[0].codec_payload_type);
>-    EXPECT_EQ(DefaultCodec().id, *info.receivers[0].codec_payload_type);
>-    EXPECT_EQ(NumRtpBytes(), info.receivers[0].bytes_rcvd);
>-    EXPECT_EQ(NumRtpPackets(), info.receivers[0].packets_rcvd);
>-    EXPECT_EQ(0.0, info.receivers[0].fraction_lost);
>-    EXPECT_EQ(0, info.receivers[0].packets_lost);
>-    // TODO(asapersson): Not set for webrtc. Handle missing stats.
>-    // EXPECT_EQ(0, info.receivers[0].packets_concealed);
>-    EXPECT_EQ(0, info.receivers[0].firs_sent);
>-    EXPECT_EQ(0, info.receivers[0].plis_sent);
>-    EXPECT_EQ(0, info.receivers[0].nacks_sent);
>-    EXPECT_EQ(kVideoWidth, info.receivers[0].frame_width);
>-    EXPECT_EQ(kVideoHeight, info.receivers[0].frame_height);
>-    EXPECT_GT(info.receivers[0].framerate_rcvd, 0);
>-    EXPECT_GT(info.receivers[0].framerate_decoded, 0);
>-    EXPECT_GT(info.receivers[0].framerate_output, 0);
>-
>-    EXPECT_EQ(1U, info.receive_codecs.count(DefaultCodec().id));
>-    EXPECT_EQ(DefaultCodec().ToCodecParameters(),
>-              info.receive_codecs[DefaultCodec().id]);
>-  }
>-
>-  cricket::VideoSenderInfo GetSenderStats(size_t i) {
>-    cricket::VideoMediaInfo info;
>-    EXPECT_TRUE(channel_->GetStats(&info));
>-    return info.senders[i];
>-  }
>-
>-  cricket::VideoReceiverInfo GetReceiverStats(size_t i) {
>-    cricket::VideoMediaInfo info;
>-    EXPECT_TRUE(channel_->GetStats(&info));
>-    return info.receivers[i];
>-  }
>-
>-  // Test that stats work properly for a conf call with multiple recv streams.
>-  void GetStatsMultipleRecvStreams() {
>-    cricket::FakeVideoRenderer renderer1, renderer2;
>-    EXPECT_TRUE(SetOneCodec(DefaultCodec()));
>-    cricket::VideoSendParameters parameters;
>-    parameters.codecs.push_back(DefaultCodec());
>-    parameters.conference_mode = true;
>-    EXPECT_TRUE(channel_->SetSendParameters(parameters));
>-    EXPECT_TRUE(SetSend(true));
>-    EXPECT_TRUE(channel_->AddRecvStream(
>-        cricket::StreamParams::CreateLegacy(1)));
>-    EXPECT_TRUE(channel_->AddRecvStream(
>-        cricket::StreamParams::CreateLegacy(2)));
>-    EXPECT_TRUE(channel_->SetSink(1, &renderer1));
>-    EXPECT_TRUE(channel_->SetSink(2, &renderer2));
>-    EXPECT_EQ(0, renderer1.num_rendered_frames());
>-    EXPECT_EQ(0, renderer2.num_rendered_frames());
>-    std::vector<uint32_t> ssrcs;
>-    ssrcs.push_back(1);
>-    ssrcs.push_back(2);
>-    network_interface_.SetConferenceMode(true, ssrcs);
>-    EXPECT_TRUE(SendFrame());
>-    EXPECT_FRAME_ON_RENDERER_WAIT(renderer1, 1, kVideoWidth, kVideoHeight,
>-                                  kTimeout);
>-    EXPECT_FRAME_ON_RENDERER_WAIT(renderer2, 1, kVideoWidth, kVideoHeight,
>-                                  kTimeout);
>-
>-    EXPECT_TRUE(channel_->SetSend(false));
>-
>-    cricket::VideoMediaInfo info;
>-    EXPECT_TRUE(channel_->GetStats(&info));
>-    ASSERT_EQ(1U, info.senders.size());
>-    // TODO(whyuan): bytes_sent and bytes_rcvd are different. Are both payload?
>-    // For webrtc, bytes_sent does not include the RTP header length.
>-    EXPECT_GT(GetSenderStats(0).bytes_sent, 0);
>-    EXPECT_EQ_WAIT(NumRtpPackets(), GetSenderStats(0).packets_sent, kTimeout);
>-    EXPECT_EQ(kVideoWidth, GetSenderStats(0).send_frame_width);
>-    EXPECT_EQ(kVideoHeight, GetSenderStats(0).send_frame_height);
>-
>-    ASSERT_EQ(2U, info.receivers.size());
>-    for (size_t i = 0; i < info.receivers.size(); ++i) {
>-      EXPECT_EQ(1U, GetReceiverStats(i).ssrcs().size());
>-      EXPECT_EQ(i + 1, GetReceiverStats(i).ssrcs()[0]);
>-      EXPECT_EQ_WAIT(NumRtpBytes(), GetReceiverStats(i).bytes_rcvd, kTimeout);
>-      EXPECT_EQ_WAIT(NumRtpPackets(), GetReceiverStats(i).packets_rcvd,
>-                     kTimeout);
>-      EXPECT_EQ_WAIT(kVideoWidth, GetReceiverStats(i).frame_width, kTimeout);
>-      EXPECT_EQ_WAIT(kVideoHeight, GetReceiverStats(i).frame_height, kTimeout);
>-    }
>-  }
>-  // Test that stats work properly for a conf call with multiple send streams.
>-  void GetStatsMultipleSendStreams() {
>-    // Normal setup; note that we set the SSRC explicitly to ensure that
>-    // it will come first in the senders map.
>-    EXPECT_TRUE(SetOneCodec(DefaultCodec()));
>-    cricket::VideoSendParameters parameters;
>-    parameters.codecs.push_back(DefaultCodec());
>-    parameters.conference_mode = true;
>-    EXPECT_TRUE(channel_->SetSendParameters(parameters));
>-    EXPECT_TRUE(channel_->AddRecvStream(
>-        cricket::StreamParams::CreateLegacy(kSsrc)));
>-    EXPECT_TRUE(channel_->SetSink(kSsrc, &renderer_));
>-    EXPECT_TRUE(SetSend(true));
>-    EXPECT_TRUE(SendFrame());
>-    EXPECT_TRUE_WAIT(NumRtpPackets() > 0, kTimeout);
>-    EXPECT_FRAME_WAIT(1, kVideoWidth, kVideoHeight, kTimeout);
>-
>-    // Add an additional capturer, and hook up a renderer to receive it.
>-    cricket::FakeVideoRenderer renderer2;
>-    std::unique_ptr<cricket::FakeVideoCapturer> capturer(
>-        CreateFakeVideoCapturer());
>-    const int kTestWidth = 160;
>-    const int kTestHeight = 120;
>-    cricket::VideoFormat format(kTestWidth, kTestHeight,
>-                                cricket::VideoFormat::FpsToInterval(5),
>-                                cricket::FOURCC_I420);
>-    EXPECT_EQ(cricket::CS_RUNNING, capturer->Start(format));
>-    EXPECT_TRUE(channel_->AddSendStream(
>-        cricket::StreamParams::CreateLegacy(5678)));
>-    EXPECT_TRUE(channel_->SetVideoSend(5678, true, nullptr, capturer.get()));
>-    EXPECT_TRUE(channel_->AddRecvStream(
>-        cricket::StreamParams::CreateLegacy(5678)));
>-    EXPECT_TRUE(channel_->SetSink(5678, &renderer2));
>-    EXPECT_TRUE(capturer->CaptureCustomFrame(
>-        kTestWidth, kTestHeight, cricket::FOURCC_I420));
>-    EXPECT_FRAME_ON_RENDERER_WAIT(
>-        renderer2, 1, kTestWidth, kTestHeight, kTimeout);
>-
>-    // Get stats, and make sure they are correct for two senders. We wait until
>-    // the number of expected packets have been sent to avoid races where we
>-    // check stats before it has been updated.
>-    cricket::VideoMediaInfo info;
>-    for (uint32_t i = 0; i < kTimeout; ++i) {
>-      rtc::Thread::Current()->ProcessMessages(1);
>-      EXPECT_TRUE(channel_->GetStats(&info));
>-      ASSERT_EQ(2U, info.senders.size());
>-      if (info.senders[0].packets_sent + info.senders[1].packets_sent ==
>-          NumRtpPackets()) {
>-        // Stats have been updated for both sent frames, expectations can be
>-        // checked now.
>-        break;
>-      }
>-    }
>-    EXPECT_EQ(NumRtpPackets(),
>-              info.senders[0].packets_sent + info.senders[1].packets_sent)
>-        << "Timed out while waiting for packet counts for all sent packets.";
>-    EXPECT_EQ(1U, info.senders[0].ssrcs().size());
>-    EXPECT_EQ(1234U, info.senders[0].ssrcs()[0]);
>-    EXPECT_EQ(kVideoWidth, info.senders[0].send_frame_width);
>-    EXPECT_EQ(kVideoHeight, info.senders[0].send_frame_height);
>-    EXPECT_EQ(1U, info.senders[1].ssrcs().size());
>-    EXPECT_EQ(5678U, info.senders[1].ssrcs()[0]);
>-    EXPECT_EQ(kTestWidth, info.senders[1].send_frame_width);
>-    EXPECT_EQ(kTestHeight, info.senders[1].send_frame_height);
>-    // The capturer must be unregistered here as it runs out of it's scope next.
>-    channel_->SetVideoSend(5678, true, nullptr, nullptr);
>-  }
>-
>-  // Test that we can set the bandwidth.
>-  void SetSendBandwidth() {
>-    cricket::VideoSendParameters parameters;
>-    parameters.codecs.push_back(DefaultCodec());
>-    parameters.max_bandwidth_bps = -1;  // <= 0 means unlimited.
>-    EXPECT_TRUE(channel_->SetSendParameters(parameters));
>-    parameters.max_bandwidth_bps = 128 * 1024;
>-    EXPECT_TRUE(channel_->SetSendParameters(parameters));
>-  }
>-  // Test that we can set the SSRC for the default send source.
>-  void SetSendSsrc() {
>-    EXPECT_TRUE(SetDefaultCodec());
>-    EXPECT_TRUE(SetSend(true));
>-    EXPECT_TRUE(SendFrame());
>-    EXPECT_TRUE_WAIT(NumRtpPackets() > 0, kTimeout);
>-    uint32_t ssrc = 0;
>-    std::unique_ptr<const rtc::CopyOnWriteBuffer> p(GetRtpPacket(0));
>-    ParseRtpPacket(p.get(), NULL, NULL, NULL, NULL, &ssrc, NULL);
>-    EXPECT_EQ(kSsrc, ssrc);
>-    // Packets are being paced out, so these can mismatch between the first and
>-    // second call to NumRtpPackets until pending packets are paced out.
>-    EXPECT_EQ_WAIT(NumRtpPackets(), NumRtpPackets(ssrc), kTimeout);
>-    EXPECT_EQ_WAIT(NumRtpBytes(), NumRtpBytes(ssrc), kTimeout);
>-    EXPECT_EQ(1, NumSentSsrcs());
>-    EXPECT_EQ(0, NumRtpPackets(kSsrc - 1));
>-    EXPECT_EQ(0, NumRtpBytes(kSsrc - 1));
>-  }
>-  // Test that we can set the SSRC even after codecs are set.
>-  void SetSendSsrcAfterSetCodecs() {
>-    // Remove stream added in Setup.
>-    EXPECT_TRUE(channel_->RemoveSendStream(kSsrc));
>-    EXPECT_TRUE(SetDefaultCodec());
>-    EXPECT_TRUE(channel_->AddSendStream(
>-        cricket::StreamParams::CreateLegacy(999)));
>-    EXPECT_TRUE(
>-        channel_->SetVideoSend(999u, true, nullptr, video_capturer_.get()));
>-    EXPECT_TRUE(SetSend(true));
>-    EXPECT_TRUE(WaitAndSendFrame(0));
>-    EXPECT_TRUE_WAIT(NumRtpPackets() > 0, kTimeout);
>-    uint32_t ssrc = 0;
>-    std::unique_ptr<const rtc::CopyOnWriteBuffer> p(GetRtpPacket(0));
>-    ParseRtpPacket(p.get(), NULL, NULL, NULL, NULL, &ssrc, NULL);
>-    EXPECT_EQ(999u, ssrc);
>-    // Packets are being paced out, so these can mismatch between the first and
>-    // second call to NumRtpPackets until pending packets are paced out.
>-    EXPECT_EQ_WAIT(NumRtpPackets(), NumRtpPackets(ssrc), kTimeout);
>-    EXPECT_EQ_WAIT(NumRtpBytes(), NumRtpBytes(ssrc), kTimeout);
>-    EXPECT_EQ(1, NumSentSsrcs());
>-    EXPECT_EQ(0, NumRtpPackets(kSsrc));
>-    EXPECT_EQ(0, NumRtpBytes(kSsrc));
>-  }
>-  // Test that we can set the default video renderer before and after
>-  // media is received.
>-  void SetSink() {
>-    uint8_t data1[] = {
>-        0x80, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00};
>-
>-    rtc::CopyOnWriteBuffer packet1(data1, sizeof(data1));
>-    rtc::SetBE32(packet1.data() + 8, kSsrc);
>-    channel_->SetSink(kDefaultReceiveSsrc, NULL);
>-    EXPECT_TRUE(SetDefaultCodec());
>-    EXPECT_TRUE(SetSend(true));
>-    EXPECT_EQ(0, renderer_.num_rendered_frames());
>-    channel_->OnPacketReceived(&packet1, rtc::PacketTime());
>-    EXPECT_TRUE(channel_->SetSink(kDefaultReceiveSsrc, &renderer_));
>-    EXPECT_TRUE(SendFrame());
>-    EXPECT_FRAME_WAIT(1, kVideoWidth, kVideoHeight, kTimeout);
>-  }
>-
>-  // Tests empty StreamParams is rejected.
>-  void RejectEmptyStreamParams() {
>-    // Remove the send stream that was added during Setup.
>-    EXPECT_TRUE(channel_->RemoveSendStream(kSsrc));
>-
>-    cricket::StreamParams empty;
>-    EXPECT_FALSE(channel_->AddSendStream(empty));
>-    EXPECT_TRUE(channel_->AddSendStream(
>-        cricket::StreamParams::CreateLegacy(789u)));
>-  }
>-
>-  // Tests setting up and configuring a send stream.
>-  void AddRemoveSendStreams() {
>-    EXPECT_TRUE(SetOneCodec(DefaultCodec()));
>-    EXPECT_TRUE(SetSend(true));
>-    EXPECT_TRUE(channel_->SetSink(kDefaultReceiveSsrc, &renderer_));
>-    EXPECT_TRUE(SendFrame());
>-    EXPECT_FRAME_WAIT(1, kVideoWidth, kVideoHeight, kTimeout);
>-    EXPECT_GT(NumRtpPackets(), 0);
>-    uint32_t ssrc = 0;
>-    size_t last_packet = NumRtpPackets() - 1;
>-    std::unique_ptr<const rtc::CopyOnWriteBuffer>
>-        p(GetRtpPacket(static_cast<int>(last_packet)));
>-    ParseRtpPacket(p.get(), NULL, NULL, NULL, NULL, &ssrc, NULL);
>-    EXPECT_EQ(kSsrc, ssrc);
>-
>-    // Remove the send stream that was added during Setup.
>-    EXPECT_TRUE(channel_->RemoveSendStream(kSsrc));
>-    int rtp_packets = NumRtpPackets();
>-
>-    EXPECT_TRUE(channel_->AddSendStream(
>-        cricket::StreamParams::CreateLegacy(789u)));
>-    EXPECT_TRUE(
>-        channel_->SetVideoSend(789u, true, nullptr, video_capturer_.get()));
>-    EXPECT_EQ(rtp_packets, NumRtpPackets());
>-    // Wait 30ms to guarantee the engine does not drop the frame.
>-    EXPECT_TRUE(WaitAndSendFrame(30));
>-    EXPECT_TRUE_WAIT(NumRtpPackets() > rtp_packets, kTimeout);
>-
>-    last_packet = NumRtpPackets() - 1;
>-    p.reset(GetRtpPacket(static_cast<int>(last_packet)));
>-    ParseRtpPacket(p.get(), NULL, NULL, NULL, NULL, &ssrc, NULL);
>-    EXPECT_EQ(789u, ssrc);
>-  }
>-
>-  // Tests the behavior of incoming streams in a conference scenario.
>-  void SimulateConference() {
>-    cricket::FakeVideoRenderer renderer1, renderer2;
>-    EXPECT_TRUE(SetDefaultCodec());
>-    cricket::VideoSendParameters parameters;
>-    parameters.codecs.push_back(DefaultCodec());
>-    parameters.conference_mode = true;
>-    EXPECT_TRUE(channel_->SetSendParameters(parameters));
>-    EXPECT_TRUE(SetSend(true));
>-    EXPECT_TRUE(channel_->AddRecvStream(
>-        cricket::StreamParams::CreateLegacy(1)));
>-    EXPECT_TRUE(channel_->AddRecvStream(
>-        cricket::StreamParams::CreateLegacy(2)));
>-    EXPECT_TRUE(channel_->SetSink(1, &renderer1));
>-    EXPECT_TRUE(channel_->SetSink(2, &renderer2));
>-    EXPECT_EQ(0, renderer1.num_rendered_frames());
>-    EXPECT_EQ(0, renderer2.num_rendered_frames());
>-    std::vector<uint32_t> ssrcs;
>-    ssrcs.push_back(1);
>-    ssrcs.push_back(2);
>-    network_interface_.SetConferenceMode(true, ssrcs);
>-    EXPECT_TRUE(SendFrame());
>-    EXPECT_FRAME_ON_RENDERER_WAIT(renderer1, 1, kVideoWidth, kVideoHeight,
>-                                  kTimeout);
>-    EXPECT_FRAME_ON_RENDERER_WAIT(renderer2, 1, kVideoWidth, kVideoHeight,
>-                                  kTimeout);
>-
>-    std::unique_ptr<const rtc::CopyOnWriteBuffer> p(GetRtpPacket(0));
>-    EXPECT_EQ(DefaultCodec().id, GetPayloadType(p.get()));
>-    EXPECT_EQ(kVideoWidth, renderer1.width());
>-    EXPECT_EQ(kVideoHeight, renderer1.height());
>-    EXPECT_EQ(kVideoWidth, renderer2.width());
>-    EXPECT_EQ(kVideoHeight, renderer2.height());
>-    EXPECT_TRUE(channel_->RemoveRecvStream(2));
>-    EXPECT_TRUE(channel_->RemoveRecvStream(1));
>-  }
>-
>-  // Tests that we can add and remove capturers and frames are sent out properly
>-  void AddRemoveCapturer() {
>-    cricket::VideoCodec codec = DefaultCodec();
>-    const int time_between_send_ms =
>-        cricket::VideoFormat::FpsToInterval(kFramerate);
>-    EXPECT_TRUE(SetOneCodec(codec));
>-    EXPECT_TRUE(SetSend(true));
>-    EXPECT_TRUE(channel_->SetSink(kDefaultReceiveSsrc, &renderer_));
>-    EXPECT_EQ(0, renderer_.num_rendered_frames());
>-    EXPECT_TRUE(SendFrame());
>-    EXPECT_FRAME_WAIT(1, kVideoWidth, kVideoHeight, kTimeout);
>-    std::unique_ptr<cricket::FakeVideoCapturer> capturer(
>-        CreateFakeVideoCapturer());
>-
>-    // TODO(nisse): This testcase fails if we don't configure
>-    // screencast. It's unclear why, I see nothing obvious in this
>-    // test which is related to screencast logic.
>-    cricket::VideoOptions video_options;
>-    video_options.is_screencast = true;
>-    channel_->SetVideoSend(kSsrc, true, &video_options, nullptr);
>-
>-    cricket::VideoFormat format(480, 360,
>-                                cricket::VideoFormat::FpsToInterval(30),
>-                                cricket::FOURCC_I420);
>-    EXPECT_EQ(cricket::CS_RUNNING, capturer->Start(format));
>-    // All capturers start generating frames with the same timestamp. ViE does
>-    // not allow the same timestamp to be used. Capture one frame before
>-    // associating the capturer with the channel.
>-    EXPECT_TRUE(capturer->CaptureCustomFrame(format.width, format.height,
>-                                             cricket::FOURCC_I420));
>-
>-    int captured_frames = 1;
>-    for (int iterations = 0; iterations < 2; ++iterations) {
>-      EXPECT_TRUE(channel_->SetVideoSend(kSsrc, true, nullptr, capturer.get()));
>-      rtc::Thread::Current()->ProcessMessages(time_between_send_ms);
>-      EXPECT_TRUE(capturer->CaptureCustomFrame(format.width, format.height,
>-                                               cricket::FOURCC_I420));
>-      ++captured_frames;
>-      // Wait until frame of right size is captured.
>-      EXPECT_TRUE_WAIT(renderer_.num_rendered_frames() >= captured_frames &&
>-                       format.width == renderer_.width() &&
>-                       format.height == renderer_.height() &&
>-                       !renderer_.black_frame(), kTimeout);
>-      EXPECT_GE(renderer_.num_rendered_frames(), captured_frames);
>-      EXPECT_EQ(format.width, renderer_.width());
>-      EXPECT_EQ(format.height, renderer_.height());
>-      captured_frames = renderer_.num_rendered_frames() + 1;
>-      EXPECT_FALSE(renderer_.black_frame());
>-      EXPECT_TRUE(channel_->SetVideoSend(kSsrc, true, nullptr, nullptr));
>-      // Make sure a black frame is generated within the specified timeout.
>-      // The black frame should be the resolution of the previous frame to
>-      // prevent expensive encoder reconfigurations.
>-      EXPECT_TRUE_WAIT(renderer_.num_rendered_frames() >= captured_frames &&
>-                       format.width == renderer_.width() &&
>-                       format.height == renderer_.height() &&
>-                       renderer_.black_frame(), kTimeout);
>-      EXPECT_GE(renderer_.num_rendered_frames(), captured_frames);
>-      EXPECT_EQ(format.width, renderer_.width());
>-      EXPECT_EQ(format.height, renderer_.height());
>-      EXPECT_TRUE(renderer_.black_frame());
>-
>-      // The black frame has the same timestamp as the next frame since it's
>-      // timestamp is set to the last frame's timestamp + interval. WebRTC will
>-      // not render a frame with the same timestamp so capture another frame
>-      // with the frame capturer to increment the next frame's timestamp.
>-      EXPECT_TRUE(capturer->CaptureCustomFrame(format.width, format.height,
>-                                               cricket::FOURCC_I420));
>-    }
>-  }
>-
>-  // Tests that if SetVideoSend is called with a NULL capturer after the
>-  // capturer was already removed, the application doesn't crash (and no black
>-  // frame is sent).
>-  void RemoveCapturerWithoutAdd() {
>-    EXPECT_TRUE(SetOneCodec(DefaultCodec()));
>-    EXPECT_TRUE(SetSend(true));
>-    EXPECT_TRUE(channel_->SetSink(kDefaultReceiveSsrc, &renderer_));
>-    EXPECT_EQ(0, renderer_.num_rendered_frames());
>-    EXPECT_TRUE(SendFrame());
>-    EXPECT_FRAME_WAIT(1, kVideoWidth, kVideoHeight, kTimeout);
>-    // Wait for one frame so they don't get dropped because we send frames too
>-    // tightly.
>-    rtc::Thread::Current()->ProcessMessages(30);
>-    // Remove the capturer.
>-    EXPECT_TRUE(channel_->SetVideoSend(kSsrc, true, nullptr, nullptr));
>-
>-    // No capturer was added, so this SetVideoSend shouldn't do anything.
>-    EXPECT_TRUE(channel_->SetVideoSend(kSsrc, true, nullptr, nullptr));
>-    rtc::Thread::Current()->ProcessMessages(300);
>-    // Verify no more frames were sent.
>-    EXPECT_EQ(1, renderer_.num_rendered_frames());
>-  }
>-
>-  // Tests that we can add and remove capturer as unique sources.
>-  void AddRemoveCapturerMultipleSources() {
>-    // WebRTC implementation will drop frames if pushed to quickly. Wait the
>-    // interval time to avoid that.
>-    // WebRTC implementation will drop frames if pushed to quickly. Wait the
>-    // interval time to avoid that.
>-    // Set up the stream associated with the engine.
>-    EXPECT_TRUE(channel_->AddRecvStream(
>-        cricket::StreamParams::CreateLegacy(kSsrc)));
>-    EXPECT_TRUE(channel_->SetSink(kSsrc, &renderer_));
>-    cricket::VideoFormat capture_format;  // default format
>-    capture_format.interval = cricket::VideoFormat::FpsToInterval(kFramerate);
>-    // Set up additional stream 1.
>-    cricket::FakeVideoRenderer renderer1;
>-    EXPECT_FALSE(channel_->SetSink(1, &renderer1));
>-    EXPECT_TRUE(channel_->AddRecvStream(
>-        cricket::StreamParams::CreateLegacy(1)));
>-    EXPECT_TRUE(channel_->SetSink(1, &renderer1));
>-    EXPECT_TRUE(channel_->AddSendStream(
>-        cricket::StreamParams::CreateLegacy(1)));
>-    std::unique_ptr<cricket::FakeVideoCapturer> capturer1(
>-        CreateFakeVideoCapturer());
>-    EXPECT_EQ(cricket::CS_RUNNING, capturer1->Start(capture_format));
>-    // Set up additional stream 2.
>-    cricket::FakeVideoRenderer renderer2;
>-    EXPECT_FALSE(channel_->SetSink(2, &renderer2));
>-    EXPECT_TRUE(channel_->AddRecvStream(
>-        cricket::StreamParams::CreateLegacy(2)));
>-    EXPECT_TRUE(channel_->SetSink(2, &renderer2));
>-    EXPECT_TRUE(channel_->AddSendStream(
>-        cricket::StreamParams::CreateLegacy(2)));
>-    std::unique_ptr<cricket::FakeVideoCapturer> capturer2(
>-        CreateFakeVideoCapturer());
>-    EXPECT_EQ(cricket::CS_RUNNING, capturer2->Start(capture_format));
>-    // State for all the streams.
>-    EXPECT_TRUE(SetOneCodec(DefaultCodec()));
>-    // A limitation in the lmi implementation requires that SetVideoSend() is
>-    // called after SetOneCodec().
>-    // TODO(hellner): this seems like an unnecessary constraint, fix it.
>-    EXPECT_TRUE(channel_->SetVideoSend(1, true, nullptr, capturer1.get()));
>-    EXPECT_TRUE(channel_->SetVideoSend(2, true, nullptr, capturer2.get()));
>-    EXPECT_TRUE(SetSend(true));
>-    // Test capturer associated with engine.
>-    const int kTestWidth = 160;
>-    const int kTestHeight = 120;
>-    EXPECT_TRUE(capturer1->CaptureCustomFrame(
>-        kTestWidth, kTestHeight, cricket::FOURCC_I420));
>-    EXPECT_FRAME_ON_RENDERER_WAIT(
>-        renderer1, 1, kTestWidth, kTestHeight, kTimeout);
>-    // Capture a frame with additional capturer2, frames should be received
>-    EXPECT_TRUE(capturer2->CaptureCustomFrame(
>-        kTestWidth, kTestHeight, cricket::FOURCC_I420));
>-    EXPECT_FRAME_ON_RENDERER_WAIT(
>-        renderer2, 1, kTestWidth, kTestHeight, kTimeout);
>-    // Successfully remove the capturer.
>-    EXPECT_TRUE(channel_->SetVideoSend(kSsrc, true, nullptr, nullptr));
>-    // The capturers must be unregistered here as it runs out of it's scope
>-    // next.
>-    EXPECT_TRUE(channel_->SetVideoSend(1, true, nullptr, nullptr));
>-    EXPECT_TRUE(channel_->SetVideoSend(2, true, nullptr, nullptr));
>-  }
>-
>-  // Test that multiple send streams can be created and deleted properly.
>-  void MultipleSendStreams() {
>-    // Remove stream added in Setup. I.e. remove stream corresponding to default
>-    // channel.
>-    EXPECT_TRUE(channel_->RemoveSendStream(kSsrc));
>-    const unsigned int kSsrcsSize = sizeof(kSsrcs4)/sizeof(kSsrcs4[0]);
>-    for (unsigned int i = 0; i < kSsrcsSize; ++i) {
>-      EXPECT_TRUE(channel_->AddSendStream(
>-          cricket::StreamParams::CreateLegacy(kSsrcs4[i])));
>-    }
>-    // Delete one of the non default channel streams, let the destructor delete
>-    // the remaining ones.
>-    EXPECT_TRUE(channel_->RemoveSendStream(kSsrcs4[kSsrcsSize - 1]));
>-    // Stream should already be deleted.
>-    EXPECT_FALSE(channel_->RemoveSendStream(kSsrcs4[kSsrcsSize - 1]));
>-  }
>-
>-  // Two streams one channel tests.
>-
>-  // Tests that we can send and receive frames.
>-  void TwoStreamsSendAndReceive(const cricket::VideoCodec& codec) {
>-    SetUpSecondStream();
>-    // Test sending and receiving on first stream.
>-    SendAndReceive(codec);
>-    // Test sending and receiving on second stream.
>-    EXPECT_EQ_WAIT(1, renderer2_.num_rendered_frames(), kTimeout);
>-    EXPECT_GT(NumRtpPackets(), 0);
>-    EXPECT_EQ(1, renderer2_.num_rendered_frames());
>-  }
>-
>-  webrtc::RtcEventLogNullImpl event_log_;
>-  const std::unique_ptr<webrtc::Call> call_;
>-  E engine_;
>-  std::unique_ptr<cricket::FakeVideoCapturer> video_capturer_;
>-  std::unique_ptr<cricket::FakeVideoCapturer> video_capturer_2_;
>-  std::unique_ptr<C> channel_;
>-  cricket::FakeNetworkInterface network_interface_;
>-  cricket::FakeVideoRenderer renderer_;
>-
>-  // Used by test cases where 2 streams are run on the same channel.
>-  cricket::FakeVideoRenderer renderer2_;
>-};
>-
>-#endif  // MEDIA_BASE_VIDEOENGINE_UNITTEST_H_  NOLINT
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videosourcebase.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videosourcebase.cc
>index 299795fca6f..b193b9ff65d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videosourcebase.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videosourcebase.cc
>@@ -17,6 +17,7 @@ namespace rtc {
> VideoSourceBase::VideoSourceBase() {
>   thread_checker_.DetachFromThread();
> }
>+VideoSourceBase::~VideoSourceBase() = default;
> 
> void VideoSourceBase::AddOrUpdateSink(
>     VideoSinkInterface<webrtc::VideoFrame>* sink,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videosourcebase.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videosourcebase.h
>index 35f85bb5304..74ad280343a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videosourcebase.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/videosourcebase.h
>@@ -14,7 +14,7 @@
> #include <vector>
> 
> #include "api/video/video_frame.h"
>-#include "api/videosourceinterface.h"
>+#include "api/video/video_source_interface.h"
> #include "rtc_base/thread_checker.h"
> 
> namespace rtc {
>@@ -23,6 +23,7 @@ namespace rtc {
> class VideoSourceBase : public VideoSourceInterface<webrtc::VideoFrame> {
>  public:
>   VideoSourceBase();
>+  ~VideoSourceBase() override;
>   void AddOrUpdateSink(VideoSinkInterface<webrtc::VideoFrame>* sink,
>                        const VideoSinkWants& wants) override;
>   void RemoveSink(VideoSinkInterface<webrtc::VideoFrame>* sink) override;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/vp9_profile.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/vp9_profile.cc
>new file mode 100644
>index 00000000000..a28a912762a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/vp9_profile.cc
>@@ -0,0 +1,63 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "media/base/vp9_profile.h"
>+
>+#include "rtc_base/string_to_number.h"
>+
>+namespace webrtc {
>+
>+// Profile information for VP9 video.
>+const char kVP9FmtpProfileId[] = "x-google-profile-id";
>+
>+std::string VP9ProfileToString(VP9Profile profile) {
>+  switch (profile) {
>+    case VP9Profile::kProfile0:
>+      return "0";
>+    case VP9Profile::kProfile2:
>+      return "2";
>+  }
>+  return "0";
>+}
>+
>+absl::optional<VP9Profile> StringToVP9Profile(const std::string& str) {
>+  const absl::optional<int> i = rtc::StringToNumber<int>(str);
>+  if (!i.has_value())
>+    return absl::nullopt;
>+
>+  switch (i.value()) {
>+    case 0:
>+      return VP9Profile::kProfile0;
>+    case 2:
>+      return VP9Profile::kProfile2;
>+    default:
>+      return absl::nullopt;
>+  }
>+  return absl::nullopt;
>+}
>+
>+absl::optional<VP9Profile> ParseSdpForVP9Profile(
>+    const SdpVideoFormat::Parameters& params) {
>+  const auto profile_it = params.find(kVP9FmtpProfileId);
>+  if (profile_it == params.end())
>+    return VP9Profile::kProfile0;
>+  const std::string& profile_str = profile_it->second;
>+  return StringToVP9Profile(profile_str);
>+}
>+
>+bool IsSameVP9Profile(const SdpVideoFormat::Parameters& params1,
>+                      const SdpVideoFormat::Parameters& params2) {
>+  const absl::optional<VP9Profile> profile = ParseSdpForVP9Profile(params1);
>+  const absl::optional<VP9Profile> other_profile =
>+      ParseSdpForVP9Profile(params2);
>+  return profile && other_profile && profile == other_profile;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/vp9_profile.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/vp9_profile.h
>new file mode 100644
>index 00000000000..5cc1d166a87
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/vp9_profile.h
>@@ -0,0 +1,53 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MEDIA_BASE_VP9_PROFILE_H_
>+#define MEDIA_BASE_VP9_PROFILE_H_
>+
>+#include <map>
>+#include <string>
>+
>+#include "absl/types/optional.h"
>+#include "api/video_codecs/sdp_video_format.h"
>+#include "common_types.h"  // NOLINT(build/include)
>+
>+namespace webrtc {
>+
>+// Profile information for VP9 video.
>+extern const char kVP9FmtpProfileId[];
>+
>+enum class VP9Profile {
>+  kProfile0,
>+  kProfile2,
>+};
>+
>+// Helper functions to convert VP9Profile to std::string. Returns "0" by
>+// default.
>+std::string VP9ProfileToString(VP9Profile profile);
>+
>+// Helper functions to convert std::string to VP9Profile. Returns null if given
>+// an invalid profile string.
>+absl::optional<VP9Profile> StringToVP9Profile(const std::string& str);
>+
>+// Parse profile that is represented as a string of single digit contained in an
>+// SDP key-value map. A default profile(kProfile0) will be returned if the
>+// profile key is missing. Nothing will be returned if the key is present but
>+// the string is invalid.
>+absl::optional<VP9Profile> ParseSdpForVP9Profile(
>+    const SdpVideoFormat::Parameters& params);
>+
>+// Returns true if the parameters have the same VP9 profile, or neither contains
>+// VP9 profile.
>+bool IsSameVP9Profile(const SdpVideoFormat::Parameters& params1,
>+                      const SdpVideoFormat::Parameters& params2);
>+
>+}  // namespace webrtc
>+
>+#endif  // MEDIA_BASE_VP9_PROFILE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/OWNERS
>new file mode 100644
>index 00000000000..421e88bf3ae
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/OWNERS
>@@ -0,0 +1 @@
>+per-file simulcast*=sprang@webrtc.org
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/adm_helpers.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/adm_helpers.cc
>index 119cc644510..7ec6575413b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/adm_helpers.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/adm_helpers.cc
>@@ -27,7 +27,7 @@ namespace adm_helpers {
> // the ADM functions, depending on the ID type.
> #if defined(WEBRTC_WIN)
> #define AUDIO_DEVICE_ID \
>-    (AudioDeviceModule::WindowsDeviceType::kDefaultCommunicationDevice)
>+  (AudioDeviceModule::WindowsDeviceType::kDefaultCommunicationDevice)
> #else
> #define AUDIO_DEVICE_ID (0u)
> #endif  // defined(WEBRTC_WIN)
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/apm_helpers.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/apm_helpers.cc
>index cf5dde27ddf..ea37cb44f46 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/apm_helpers.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/apm_helpers.cc
>@@ -31,7 +31,8 @@ void Init(AudioProcessing* apm) {
>   GainControl* gc = apm->gain_control();
>   if (gc->set_analog_level_limits(kMinVolumeLevel, kMaxVolumeLevel) != 0) {
>     RTC_DLOG(LS_ERROR) << "Failed to set analog level limits with minimum: "
>-        << kMinVolumeLevel << " and maximum: " << kMaxVolumeLevel;
>+                       << kMinVolumeLevel
>+                       << " and maximum: " << kMaxVolumeLevel;
>   }
> }
> 
>@@ -44,8 +45,7 @@ AgcConfig GetAgcConfig(AudioProcessing* apm) {
>   return result;
> }
> 
>-void SetAgcConfig(AudioProcessing* apm,
>-                  const AgcConfig& config) {
>+void SetAgcConfig(AudioProcessing* apm, const AgcConfig& config) {
>   RTC_DCHECK(apm);
>   GainControl* gc = apm->gain_control();
>   if (gc->set_target_level_dbfs(config.targetLeveldBOv) != 0) {
>@@ -62,8 +62,7 @@ void SetAgcConfig(AudioProcessing* apm,
>   }
> }
> 
>-void SetAgcStatus(AudioProcessing* apm,
>-                  bool enable) {
>+void SetAgcStatus(AudioProcessing* apm, bool enable) {
>   RTC_DCHECK(apm);
> #if defined(WEBRTC_IOS) || defined(WEBRTC_ANDROID)
>   GainControl::Mode agc_mode = GainControl::kFixedDigital;
>@@ -82,9 +81,7 @@ void SetAgcStatus(AudioProcessing* apm,
>   RTC_LOG(LS_INFO) << "AGC set to " << enable << " with mode " << agc_mode;
> }
> 
>-void SetEcStatus(AudioProcessing* apm,
>-                 bool enable,
>-                 EcModes mode) {
>+void SetEcStatus(AudioProcessing* apm, bool enable, EcModes mode) {
>   RTC_DCHECK(apm);
>   RTC_DCHECK(mode == kEcConference || mode == kEcAecm) << "mode: " << mode;
>   EchoCancellation* ec = apm->echo_cancellation();
>@@ -99,8 +96,7 @@ void SetEcStatus(AudioProcessing* apm,
>       RTC_LOG(LS_ERROR) << "Failed to enable/disable AEC: " << enable;
>       return;
>     }
>-    if (ec->set_suppression_level(EchoCancellation::kHighSuppression)
>-        != 0) {
>+    if (ec->set_suppression_level(EchoCancellation::kHighSuppression) != 0) {
>       RTC_LOG(LS_ERROR) << "Failed to set high AEC aggressiveness.";
>       return;
>     }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/apm_helpers.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/apm_helpers.h
>index 42465fcec07..f775d8a4736 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/apm_helpers.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/apm_helpers.h
>@@ -18,8 +18,8 @@ namespace webrtc {
> class AudioProcessing;
> 
> enum EcModes {
>-  kEcConference,     // Conferencing default (aggressive AEC).
>-  kEcAecm,           // AEC mobile.
>+  kEcConference,  // Conferencing default (aggressive AEC).
>+  kEcAecm,        // AEC mobile.
> };
> 
> struct AgcConfig {
>@@ -32,13 +32,9 @@ namespace apm_helpers {
> 
> void Init(AudioProcessing* apm);
> AgcConfig GetAgcConfig(AudioProcessing* apm);
>-void SetAgcConfig(AudioProcessing* apm,
>-                  const AgcConfig& config);
>-void SetAgcStatus(AudioProcessing* apm,
>-                  bool enable);
>-void SetEcStatus(AudioProcessing* apm,
>-                 bool enable,
>-                 EcModes mode);
>+void SetAgcConfig(AudioProcessing* apm, const AgcConfig& config);
>+void SetAgcStatus(AudioProcessing* apm, bool enable);
>+void SetEcStatus(AudioProcessing* apm, bool enable, EcModes mode);
> void SetEcMetricsStatus(AudioProcessing* apm, bool enable);
> void SetAecmMode(AudioProcessing* apm, bool enable_cng);
> void SetNsStatus(AudioProcessing* apm, bool enable);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/apm_helpers_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/apm_helpers_unittest.cc
>index bad24af83bb..1f22963152f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/apm_helpers_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/apm_helpers_unittest.cc
>@@ -18,7 +18,7 @@
> namespace webrtc {
> namespace {
> 
>-constexpr AgcConfig kDefaultAgcConfig = { 3, 9, true };
>+constexpr AgcConfig kDefaultAgcConfig = {3, 9, true};
> 
> struct TestHelper {
>   TestHelper() {
>@@ -60,8 +60,7 @@ struct TestHelper {
> 
> TEST(ApmHelpersTest, AgcConfig_DefaultConfiguration) {
>   TestHelper helper;
>-  AgcConfig agc_config =
>-      apm_helpers::GetAgcConfig(helper.apm());
>+  AgcConfig agc_config = apm_helpers::GetAgcConfig(helper.apm());
> 
>   EXPECT_EQ(kDefaultAgcConfig.targetLeveldBOv, agc_config.targetLeveldBOv);
>   EXPECT_EQ(kDefaultAgcConfig.digitalCompressionGaindB,
>@@ -70,19 +69,16 @@ TEST(ApmHelpersTest, AgcConfig_DefaultConfiguration) {
> }
> 
> TEST(ApmHelpersTest, AgcConfig_GetAndSet) {
>-  const AgcConfig agc_config = { 11, 17, false };
>+  const AgcConfig agc_config = {11, 17, false};
> 
>   TestHelper helper;
>   apm_helpers::SetAgcConfig(helper.apm(), agc_config);
>-  AgcConfig actual_config =
>-      apm_helpers::GetAgcConfig(helper.apm());
>+  AgcConfig actual_config = apm_helpers::GetAgcConfig(helper.apm());
> 
>   EXPECT_EQ(agc_config.digitalCompressionGaindB,
>             actual_config.digitalCompressionGaindB);
>-  EXPECT_EQ(agc_config.limiterEnable,
>-            actual_config.limiterEnable);
>-  EXPECT_EQ(agc_config.targetLeveldBOv,
>-            actual_config.targetLeveldBOv);
>+  EXPECT_EQ(agc_config.limiterEnable, actual_config.limiterEnable);
>+  EXPECT_EQ(agc_config.targetLeveldBOv, actual_config.targetLeveldBOv);
> }
> 
> TEST(ApmHelpersTest, AgcStatus_DefaultMode) {
>@@ -189,7 +185,7 @@ TEST(ApmHelpersTest, AecmMode_DefaultMode) {
>   TestHelper helper;
>   EchoControlMobile* ecm = helper.apm()->echo_control_mobile();
>   EXPECT_EQ(EchoControlMobile::kSpeakerphone, ecm->routing_mode());
>-  EXPECT_TRUE(ecm->is_comfort_noise_enabled());
>+  EXPECT_FALSE(ecm->is_comfort_noise_enabled());
> }
> 
> TEST(ApmHelpersTest, AecmMode_EnableDisableCng) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/convert_legacy_video_factory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/convert_legacy_video_factory.cc
>index a5b46f81ab5..e6447bcc9f3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/convert_legacy_video_factory.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/convert_legacy_video_factory.cc
>@@ -13,21 +13,21 @@
> #include <utility>
> #include <vector>
> 
>+#include "absl/memory/memory.h"
> #include "api/video_codecs/video_decoder_factory.h"
>+#include "api/video_codecs/video_decoder_software_fallback_wrapper.h"
> #include "api/video_codecs/video_encoder_factory.h"
>+#include "api/video_codecs/video_encoder_software_fallback_wrapper.h"
> #include "media/base/h264_profile_level_id.h"
> #include "media/engine/internaldecoderfactory.h"
> #include "media/engine/internalencoderfactory.h"
> #include "media/engine/scopedvideodecoder.h"
> #include "media/engine/scopedvideoencoder.h"
> #include "media/engine/simulcast_encoder_adapter.h"
>-#include "media/engine/videodecodersoftwarefallbackwrapper.h"
>-#include "media/engine/videoencodersoftwarefallbackwrapper.h"
> #include "media/engine/vp8_encoder_simulcast_proxy.h"
> #include "media/engine/webrtcvideodecoderfactory.h"
> #include "media/engine/webrtcvideoencoderfactory.h"
> #include "rtc_base/checks.h"
>-#include "rtc_base/ptr_util.h"
> 
> namespace cricket {
> 
>@@ -98,9 +98,14 @@ class EncoderAdapter : public webrtc::VideoEncoderFactory {
>       std::unique_ptr<WebRtcVideoEncoderFactory> external_encoder_factory)
>       : internal_encoder_factory_(new webrtc::InternalEncoderFactory()),
>         external_encoder_factory_(
>-            rtc::MakeUnique<CricketToWebRtcEncoderFactory>(
>+            absl::make_unique<CricketToWebRtcEncoderFactory>(
>                 std::move(external_encoder_factory))) {}
> 
>+  explicit EncoderAdapter(
>+      std::unique_ptr<webrtc::VideoEncoderFactory> external_encoder_factory)
>+      : internal_encoder_factory_(new webrtc::InternalEncoderFactory()),
>+        external_encoder_factory_(std::move(external_encoder_factory)) {}
>+
>   webrtc::VideoEncoderFactory::CodecInfo QueryVideoEncoder(
>       const webrtc::SdpVideoFormat& format) const override {
>     if (IsFormatSupported(external_encoder_factory_->GetSupportedFormats(),
>@@ -125,8 +130,8 @@ class EncoderAdapter : public webrtc::VideoEncoderFactory {
>                           format)) {
>       internal_encoder =
>           CodecNamesEq(format.name.c_str(), kVp8CodecName)
>-              ? rtc::MakeUnique<webrtc::VP8EncoderSimulcastProxy>(
>-                    internal_encoder_factory_.get())
>+              ? absl::make_unique<webrtc::VP8EncoderSimulcastProxy>(
>+                    internal_encoder_factory_.get(), format)
>               : internal_encoder_factory_->CreateVideoEncoder(format);
>     }
> 
>@@ -136,15 +141,15 @@ class EncoderAdapter : public webrtc::VideoEncoderFactory {
>                           format)) {
>       external_encoder =
>           CodecNamesEq(format.name.c_str(), kVp8CodecName)
>-              ? rtc::MakeUnique<webrtc::SimulcastEncoderAdapter>(
>-                    external_encoder_factory_.get())
>+              ? absl::make_unique<webrtc::SimulcastEncoderAdapter>(
>+                    external_encoder_factory_.get(), format)
>               : external_encoder_factory_->CreateVideoEncoder(format);
>     }
> 
>     if (internal_encoder && external_encoder) {
>       // Both internal SW encoder and external HW encoder available - create
>       // fallback encoder.
>-      return rtc::MakeUnique<webrtc::VideoEncoderSoftwareFallbackWrapper>(
>+      return webrtc::CreateVideoEncoderSoftwareFallbackWrapper(
>           std::move(internal_encoder), std::move(external_encoder));
>     }
>     return external_encoder ? std::move(external_encoder)
>@@ -171,51 +176,90 @@ class EncoderAdapter : public webrtc::VideoEncoderFactory {
>   const std::unique_ptr<webrtc::VideoEncoderFactory> external_encoder_factory_;
> };
> 
>+// Converts the cricket::WebRtcVideoDecoderFactory to a
>+// webrtc::VideoDecoderFactory (without adding any SW fallback).
>+class CricketToWebRtcDecoderFactory : public webrtc::VideoDecoderFactory {
>+ public:
>+  explicit CricketToWebRtcDecoderFactory(
>+      std::unique_ptr<WebRtcVideoDecoderFactory> external_decoder_factory)
>+      : external_decoder_factory_(std::move(external_decoder_factory)) {}
>+
>+  std::unique_ptr<webrtc::VideoDecoder> CreateVideoDecoder(
>+      const webrtc::SdpVideoFormat& format) override {
>+    if (external_decoder_factory_ != nullptr) {
>+      return CreateScopedVideoDecoder(external_decoder_factory_.get(),
>+                                      VideoCodec(format), {});
>+    }
>+    return nullptr;
>+  }
>+
>+  std::vector<webrtc::SdpVideoFormat> GetSupportedFormats() const override {
>+    // This is not implemented for the legacy decoder factory.
>+    RTC_NOTREACHED();
>+    return std::vector<webrtc::SdpVideoFormat>();
>+  }
>+
>+ private:
>+  const std::unique_ptr<WebRtcVideoDecoderFactory> external_decoder_factory_;
>+};
>+
> // This class combines an external factory with the internal factory and adds
>-// internal SW codecs, simulcast, and SW fallback wrappers.
>+// internal SW codecs and SW fallback wrappers.
> class DecoderAdapter : public webrtc::VideoDecoderFactory {
>  public:
>   explicit DecoderAdapter(
>       std::unique_ptr<WebRtcVideoDecoderFactory> external_decoder_factory)
>-      : external_decoder_factory_(std::move(external_decoder_factory)) {}
>+      : internal_decoder_factory_(new webrtc::InternalDecoderFactory()),
>+        external_decoder_factory_(
>+            absl::make_unique<CricketToWebRtcDecoderFactory>(
>+                std::move(external_decoder_factory))) {}
>+
>+  explicit DecoderAdapter(
>+      std::unique_ptr<webrtc::VideoDecoderFactory> external_decoder_factory)
>+      : internal_decoder_factory_(new webrtc::InternalDecoderFactory()),
>+        external_decoder_factory_(std::move(external_decoder_factory)) {}
> 
>   std::unique_ptr<webrtc::VideoDecoder> CreateVideoDecoder(
>       const webrtc::SdpVideoFormat& format) override {
>     std::unique_ptr<webrtc::VideoDecoder> internal_decoder;
>-    webrtc::InternalDecoderFactory internal_decoder_factory;
>-    if (IsFormatSupported(internal_decoder_factory.GetSupportedFormats(),
>+    if (IsFormatSupported(internal_decoder_factory_->GetSupportedFormats(),
>                           format)) {
>-      internal_decoder = internal_decoder_factory.CreateVideoDecoder(format);
>+      internal_decoder = internal_decoder_factory_->CreateVideoDecoder(format);
>     }
> 
>-    const VideoCodec codec(format);
>-    const VideoDecoderParams params = {};
>+    std::unique_ptr<webrtc::VideoDecoder> external_decoder = nullptr;
>     if (external_decoder_factory_ != nullptr) {
>-      std::unique_ptr<webrtc::VideoDecoder> external_decoder =
>-          CreateScopedVideoDecoder(external_decoder_factory_.get(), codec,
>-                                   params);
>-      if (external_decoder) {
>-        if (!internal_decoder)
>-          return external_decoder;
>-        // Both external and internal decoder available - create fallback
>-        // wrapper.
>-        return std::unique_ptr<webrtc::VideoDecoder>(
>-            new webrtc::VideoDecoderSoftwareFallbackWrapper(
>-                std::move(internal_decoder), std::move(external_decoder)));
>-      }
>+      external_decoder = external_decoder_factory_->CreateVideoDecoder(format);
>     }
> 
>-    return internal_decoder;
>+    if (internal_decoder && external_decoder) {
>+      // Both internal SW decoder and external HW decoder available - create
>+      // fallback decoder.
>+      return webrtc::CreateVideoDecoderSoftwareFallbackWrapper(
>+          std::move(internal_decoder), std::move(external_decoder));
>+    }
>+    return external_decoder ? std::move(external_decoder)
>+                            : std::move(internal_decoder);
>   }
> 
>   std::vector<webrtc::SdpVideoFormat> GetSupportedFormats() const override {
>-    // This is not implemented for the legacy decoder factory.
>-    RTC_NOTREACHED();
>-    return std::vector<webrtc::SdpVideoFormat>();
>+    std::vector<webrtc::SdpVideoFormat> formats =
>+        internal_decoder_factory_->GetSupportedFormats();
>+
>+    // Add external codecs.
>+    for (const webrtc::SdpVideoFormat& format :
>+         external_decoder_factory_->GetSupportedFormats()) {
>+      // Don't add same codec twice.
>+      if (!IsFormatSupported(formats, format))
>+        formats.push_back(format);
>+    }
>+
>+    return formats;
>   }
> 
>  private:
>-  const std::unique_ptr<WebRtcVideoDecoderFactory> external_decoder_factory_;
>+  const std::unique_ptr<webrtc::VideoDecoderFactory> internal_decoder_factory_;
>+  const std::unique_ptr<webrtc::VideoDecoderFactory> external_decoder_factory_;
> };
> 
> }  // namespace
>@@ -226,10 +270,22 @@ std::unique_ptr<webrtc::VideoEncoderFactory> ConvertVideoEncoderFactory(
>       new EncoderAdapter(std::move(external_encoder_factory)));
> }
> 
>+std::unique_ptr<webrtc::VideoEncoderFactory> ConvertVideoEncoderFactory(
>+    std::unique_ptr<webrtc::VideoEncoderFactory> external_encoder_factory) {
>+  return std::unique_ptr<webrtc::VideoEncoderFactory>(
>+      new EncoderAdapter(std::move(external_encoder_factory)));
>+}
>+
> std::unique_ptr<webrtc::VideoDecoderFactory> ConvertVideoDecoderFactory(
>     std::unique_ptr<WebRtcVideoDecoderFactory> external_decoder_factory) {
>   return std::unique_ptr<webrtc::VideoDecoderFactory>(
>       new DecoderAdapter(std::move(external_decoder_factory)));
> }
> 
>+std::unique_ptr<webrtc::VideoDecoderFactory> ConvertVideoDecoderFactory(
>+    std::unique_ptr<webrtc::VideoDecoderFactory> external_decoder_factory) {
>+  return std::unique_ptr<webrtc::VideoDecoderFactory>(
>+      new DecoderAdapter(std::move(external_decoder_factory)));
>+}
>+
> }  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/convert_legacy_video_factory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/convert_legacy_video_factory.h
>index 5bd3580a6f1..9d0d154910e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/convert_legacy_video_factory.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/convert_legacy_video_factory.h
>@@ -30,9 +30,15 @@ class WebRtcVideoDecoderFactory;
> std::unique_ptr<webrtc::VideoEncoderFactory> ConvertVideoEncoderFactory(
>     std::unique_ptr<WebRtcVideoEncoderFactory> external_encoder_factory);
> 
>+std::unique_ptr<webrtc::VideoEncoderFactory> ConvertVideoEncoderFactory(
>+    std::unique_ptr<webrtc::VideoEncoderFactory> external_encoder_factory);
>+
> std::unique_ptr<webrtc::VideoDecoderFactory> ConvertVideoDecoderFactory(
>     std::unique_ptr<WebRtcVideoDecoderFactory> external_decoder_factory);
> 
>+std::unique_ptr<webrtc::VideoDecoderFactory> ConvertVideoDecoderFactory(
>+    std::unique_ptr<webrtc::VideoDecoderFactory> external_decoder_factory);
>+
> }  // namespace cricket
> 
> #endif  // MEDIA_ENGINE_CONVERT_LEGACY_VIDEO_FACTORY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/fakewebrtccall.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/fakewebrtccall.cc
>index 5dcfd762e9d..24ae019d08b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/fakewebrtccall.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/fakewebrtccall.cc
>@@ -90,7 +90,7 @@ bool FakeAudioReceiveStream::VerifyLastPacket(const uint8_t* data,
> 
> bool FakeAudioReceiveStream::DeliverRtp(const uint8_t* packet,
>                                         size_t length,
>-                                        const webrtc::PacketTime& packet_time) {
>+                                        int64_t /* packet_time_us */) {
>   ++received_packets_;
>   last_packet_.SetData(packet, length);
>   return true;
>@@ -123,7 +123,7 @@ FakeVideoSendStream::FakeVideoSendStream(
>       framerate_scaling_enabled_(false),
>       source_(nullptr),
>       num_swapped_frames_(0) {
>-  RTC_DCHECK(config.encoder_settings.encoder != NULL);
>+  RTC_DCHECK(config.encoder_settings.encoder_factory != nullptr);
>   ReconfigureVideoEncoder(std::move(encoder_config));
> }
> 
>@@ -156,7 +156,7 @@ bool FakeVideoSendStream::GetVp8Settings(
>     return false;
>   }
> 
>-  *settings = vpx_settings_.vp8;
>+  *settings = codec_specific_settings_.vp8;
>   return true;
> }
> 
>@@ -166,7 +166,17 @@ bool FakeVideoSendStream::GetVp9Settings(
>     return false;
>   }
> 
>-  *settings = vpx_settings_.vp9;
>+  *settings = codec_specific_settings_.vp9;
>+  return true;
>+}
>+
>+bool FakeVideoSendStream::GetH264Settings(
>+    webrtc::VideoCodecH264* settings) const {
>+  if (!codec_settings_set_) {
>+    return false;
>+  }
>+
>+  *settings = codec_specific_settings_.h264;
>   return true;
> }
> 
>@@ -224,24 +234,31 @@ void FakeVideoSendStream::ReconfigureVideoEncoder(
>   } else {
>     width = height = 0;
>   }
>-  video_streams_ = config.video_stream_factory->CreateEncoderStreams(
>-      width, height, config);
>+  video_streams_ =
>+      config.video_stream_factory->CreateEncoderStreams(width, height, config);
>   if (config.encoder_specific_settings != NULL) {
>-    if (config_.encoder_settings.payload_name == "VP8") {
>-      config.encoder_specific_settings->FillVideoCodecVp8(&vpx_settings_.vp8);
>+    const unsigned char num_temporal_layers = static_cast<unsigned char>(
>+        video_streams_.back().num_temporal_layers.value_or(1));
>+    if (config_.rtp.payload_name == "VP8") {
>+      config.encoder_specific_settings->FillVideoCodecVp8(
>+          &codec_specific_settings_.vp8);
>       if (!video_streams_.empty()) {
>-        vpx_settings_.vp8.numberOfTemporalLayers = static_cast<unsigned char>(
>-            video_streams_.back().temporal_layer_thresholds_bps.size() + 1);
>+        codec_specific_settings_.vp8.numberOfTemporalLayers =
>+            num_temporal_layers;
>       }
>-    } else if (config_.encoder_settings.payload_name == "VP9") {
>-      config.encoder_specific_settings->FillVideoCodecVp9(&vpx_settings_.vp9);
>+    } else if (config_.rtp.payload_name == "VP9") {
>+      config.encoder_specific_settings->FillVideoCodecVp9(
>+          &codec_specific_settings_.vp9);
>       if (!video_streams_.empty()) {
>-        vpx_settings_.vp9.numberOfTemporalLayers = static_cast<unsigned char>(
>-            video_streams_.back().temporal_layer_thresholds_bps.size() + 1);
>+        codec_specific_settings_.vp9.numberOfTemporalLayers =
>+            num_temporal_layers;
>       }
>+    } else if (config_.rtp.payload_name == "H264") {
>+      config.encoder_specific_settings->FillVideoCodecH264(
>+          &codec_specific_settings_.h264);
>     } else {
>       ADD_FAILURE() << "Unsupported encoder payload: "
>-                    << config_.encoder_settings.payload_name;
>+                    << config_.rtp.payload_name;
>     }
>   }
>   codec_settings_set_ = config.encoder_specific_settings != NULL;
>@@ -249,6 +266,17 @@ void FakeVideoSendStream::ReconfigureVideoEncoder(
>   ++num_encoder_reconfigurations_;
> }
> 
>+void FakeVideoSendStream::UpdateActiveSimulcastLayers(
>+    const std::vector<bool> active_layers) {
>+  sending_ = false;
>+  for (const bool active_layer : active_layers) {
>+    if (active_layer) {
>+      sending_ = true;
>+      break;
>+    }
>+  }
>+}
>+
> void FakeVideoSendStream::Start() {
>   sending_ = true;
> }
>@@ -259,26 +287,24 @@ void FakeVideoSendStream::Stop() {
> 
> void FakeVideoSendStream::SetSource(
>     rtc::VideoSourceInterface<webrtc::VideoFrame>* source,
>-    const webrtc::VideoSendStream::DegradationPreference&
>-        degradation_preference) {
>-  RTC_DCHECK(source != source_);
>+    const webrtc::DegradationPreference& degradation_preference) {
>   if (source_)
>     source_->RemoveSink(this);
>   source_ = source;
>   switch (degradation_preference) {
>-    case DegradationPreference::kMaintainFramerate:
>+    case webrtc::DegradationPreference::MAINTAIN_FRAMERATE:
>       resolution_scaling_enabled_ = true;
>       framerate_scaling_enabled_ = false;
>       break;
>-    case DegradationPreference::kMaintainResolution:
>+    case webrtc::DegradationPreference::MAINTAIN_RESOLUTION:
>       resolution_scaling_enabled_ = false;
>       framerate_scaling_enabled_ = true;
>       break;
>-    case DegradationPreference::kBalanced:
>+    case webrtc::DegradationPreference::BALANCED:
>       resolution_scaling_enabled_ = true;
>       framerate_scaling_enabled_ = true;
>       break;
>-    case DegradationPreference::kDegradationDisabled:
>+    case webrtc::DegradationPreference::DISABLED:
>       resolution_scaling_enabled_ = false;
>       framerate_scaling_enabled_ = false;
>       break;
>@@ -297,7 +323,10 @@ void FakeVideoSendStream::InjectVideoSinkWants(
> 
> FakeVideoReceiveStream::FakeVideoReceiveStream(
>     webrtc::VideoReceiveStream::Config config)
>-    : config_(std::move(config)), receiving_(false) {}
>+    : config_(std::move(config)),
>+      receiving_(false),
>+      num_added_secondary_sinks_(0),
>+      num_removed_secondary_sinks_(0) {}
> 
> const webrtc::VideoReceiveStream::Config& FakeVideoReceiveStream::GetConfig()
>     const {
>@@ -335,10 +364,22 @@ void FakeVideoReceiveStream::EnableEncodedFrameRecording(rtc::PlatformFile file,
> }
> 
> void FakeVideoReceiveStream::AddSecondarySink(
>-    webrtc::RtpPacketSinkInterface* sink) {}
>+    webrtc::RtpPacketSinkInterface* sink) {
>+  ++num_added_secondary_sinks_;
>+}
> 
> void FakeVideoReceiveStream::RemoveSecondarySink(
>-    const webrtc::RtpPacketSinkInterface* sink) {}
>+    const webrtc::RtpPacketSinkInterface* sink) {
>+  ++num_removed_secondary_sinks_;
>+}
>+
>+int FakeVideoReceiveStream::GetNumAddedSecondarySinks() const {
>+  return num_added_secondary_sinks_;
>+}
>+
>+int FakeVideoReceiveStream::GetNumRemovedSecondarySinks() const {
>+  return num_removed_secondary_sinks_;
>+}
> 
> FakeFlexfecReceiveStream::FakeFlexfecReceiveStream(
>     const webrtc::FlexfecReceiveStream::Config& config)
>@@ -358,9 +399,8 @@ void FakeFlexfecReceiveStream::OnRtpPacket(const webrtc::RtpPacketReceived&) {
>   RTC_NOTREACHED() << "Not implemented.";
> }
> 
>-FakeCall::FakeCall(const webrtc::Call::Config& config)
>-    : config_(config),
>-      audio_network_state_(webrtc::kNetworkUp),
>+FakeCall::FakeCall()
>+    : audio_network_state_(webrtc::kNetworkUp),
>       video_network_state_(webrtc::kNetworkUp),
>       num_created_send_streams_(0),
>       num_created_receive_streams_(0),
>@@ -374,10 +414,6 @@ FakeCall::~FakeCall() {
>   EXPECT_EQ(0u, audio_receive_streams_.size());
> }
> 
>-webrtc::Call::Config FakeCall::GetConfig() const {
>-  return config_;
>-}
>-
> const std::vector<FakeVideoSendStream*>& FakeCall::GetVideoSendStreams() {
>   return video_send_streams_;
> }
>@@ -546,10 +582,9 @@ webrtc::PacketReceiver* FakeCall::Receiver() {
>   return this;
> }
> 
>-FakeCall::DeliveryStatus FakeCall::DeliverPacket(
>-    webrtc::MediaType media_type,
>-    rtc::CopyOnWriteBuffer packet,
>-    const webrtc::PacketTime& packet_time) {
>+FakeCall::DeliveryStatus FakeCall::DeliverPacket(webrtc::MediaType media_type,
>+                                                 rtc::CopyOnWriteBuffer packet,
>+                                                 int64_t packet_time_us) {
>   EXPECT_GE(packet.size(), 12u);
>   RTC_DCHECK(media_type == webrtc::MediaType::AUDIO ||
>              media_type == webrtc::MediaType::VIDEO);
>@@ -567,7 +602,7 @@ FakeCall::DeliveryStatus FakeCall::DeliverPacket(
>   if (media_type == webrtc::MediaType::AUDIO) {
>     for (auto receiver : audio_receive_streams_) {
>       if (receiver->GetConfig().rtp.remote_ssrc == ssrc) {
>-        receiver->DeliverRtp(packet.cdata(), packet.size(), packet_time);
>+        receiver->DeliverRtp(packet.cdata(), packet.size(), packet_time_us);
>         return DELIVERY_OK;
>       }
>     }
>@@ -591,16 +626,6 @@ webrtc::Call::Stats FakeCall::GetStats() const {
>   return stats_;
> }
> 
>-void FakeCall::SetBitrateConfig(
>-    const webrtc::Call::Config::BitrateConfig& bitrate_config) {
>-  config_.bitrate_config = bitrate_config;
>-}
>-
>-void FakeCall::SetBitrateConfigMask(
>-    const webrtc::Call::Config::BitrateConfigMask& mask) {
>-  // TODO(zstein): not implemented
>-}
>-
> void FakeCall::SetBitrateAllocationStrategy(
>     std::unique_ptr<rtc::BitrateAllocationStrategy>
>         bitrate_allocation_strategy) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/fakewebrtccall.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/fakewebrtccall.h
>index cd9f1eb2cbf..8f8270915c6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/fakewebrtccall.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/fakewebrtccall.h
>@@ -29,10 +29,11 @@
> #include "call/audio_send_stream.h"
> #include "call/call.h"
> #include "call/flexfec_receive_stream.h"
>-#include "modules/rtp_rtcp/source/rtp_packet_received.h"
>-#include "rtc_base/buffer.h"
>+#include "call/test/mock_rtp_transport_controller_send.h"
> #include "call/video_receive_stream.h"
> #include "call/video_send_stream.h"
>+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
>+#include "rtc_base/buffer.h"
> 
> namespace cricket {
> class FakeAudioSendStream final : public webrtc::AudioSendStream {
>@@ -44,8 +45,8 @@ class FakeAudioSendStream final : public webrtc::AudioSendStream {
>     int duration_ms = 0;
>   };
> 
>-  explicit FakeAudioSendStream(
>-      int id, const webrtc::AudioSendStream::Config& config);
>+  explicit FakeAudioSendStream(int id,
>+                               const webrtc::AudioSendStream::Config& config);
> 
>   int id() const { return id_; }
>   const webrtc::AudioSendStream::Config& GetConfig() const override;
>@@ -61,7 +62,9 @@ class FakeAudioSendStream final : public webrtc::AudioSendStream {
>   void Stop() override { sending_ = false; }
>   void SendAudioData(std::unique_ptr<webrtc::AudioFrame> audio_frame) override {
>   }
>-  bool SendTelephoneEvent(int payload_type, int payload_frequency, int event,
>+  bool SendTelephoneEvent(int payload_type,
>+                          int payload_frequency,
>+                          int event,
>                           int duration_ms) override;
>   void SetMuted(bool muted) override;
>   webrtc::AudioSendStream::Stats GetStats() const override;
>@@ -79,7 +82,8 @@ class FakeAudioSendStream final : public webrtc::AudioSendStream {
> class FakeAudioReceiveStream final : public webrtc::AudioReceiveStream {
>  public:
>   explicit FakeAudioReceiveStream(
>-      int id, const webrtc::AudioReceiveStream::Config& config);
>+      int id,
>+      const webrtc::AudioReceiveStream::Config& config);
> 
>   int id() const { return id_; }
>   const webrtc::AudioReceiveStream::Config& GetConfig() const;
>@@ -88,9 +92,7 @@ class FakeAudioReceiveStream final : public webrtc::AudioReceiveStream {
>   bool VerifyLastPacket(const uint8_t* data, size_t length) const;
>   const webrtc::AudioSinkInterface* sink() const { return sink_; }
>   float gain() const { return gain_; }
>-  bool DeliverRtp(const uint8_t* packet,
>-                  size_t length,
>-                  const webrtc::PacketTime& packet_time);
>+  bool DeliverRtp(const uint8_t* packet, size_t length, int64_t packet_time_us);
>   bool started() const { return started_; }
> 
>  private:
>@@ -100,7 +102,6 @@ class FakeAudioReceiveStream final : public webrtc::AudioReceiveStream {
>   void Stop() override { started_ = false; }
> 
>   webrtc::AudioReceiveStream::Stats GetStats() const override;
>-  int GetOutputLevel() const override { return 0; }
>   void SetSink(webrtc::AudioSinkInterface* sink) override;
>   void SetGain(float gain) override;
>   std::vector<webrtc::RtpSource> GetSources() const override {
>@@ -131,6 +132,7 @@ class FakeVideoSendStream final
>   bool IsSending() const;
>   bool GetVp8Settings(webrtc::VideoCodecVP8* settings) const;
>   bool GetVp9Settings(webrtc::VideoCodecVP9* settings) const;
>+  bool GetH264Settings(webrtc::VideoCodecH264* settings) const;
> 
>   int GetNumberOfSwappedFrames() const;
>   int GetLastWidth() const;
>@@ -159,11 +161,13 @@ class FakeVideoSendStream final
>   void OnFrame(const webrtc::VideoFrame& frame) override;
> 
>   // webrtc::VideoSendStream implementation.
>+  void UpdateActiveSimulcastLayers(
>+      const std::vector<bool> active_layers) override;
>   void Start() override;
>   void Stop() override;
>-  void SetSource(rtc::VideoSourceInterface<webrtc::VideoFrame>* source,
>-                 const webrtc::VideoSendStream::DegradationPreference&
>-                     degradation_preference) override;
>+  void SetSource(
>+      rtc::VideoSourceInterface<webrtc::VideoFrame>* source,
>+      const webrtc::DegradationPreference& degradation_preference) override;
>   webrtc::VideoSendStream::Stats GetStats() override;
>   void ReconfigureVideoEncoder(webrtc::VideoEncoderConfig config) override;
> 
>@@ -174,15 +178,16 @@ class FakeVideoSendStream final
>   rtc::VideoSinkWants sink_wants_;
> 
>   bool codec_settings_set_;
>-  union VpxSettings {
>+  union CodecSpecificSettings {
>     webrtc::VideoCodecVP8 vp8;
>     webrtc::VideoCodecVP9 vp9;
>-  } vpx_settings_;
>+    webrtc::VideoCodecH264 h264;
>+  } codec_specific_settings_;
>   bool resolution_scaling_enabled_;
>   bool framerate_scaling_enabled_;
>   rtc::VideoSourceInterface<webrtc::VideoFrame>* source_;
>   int num_swapped_frames_;
>-  rtc::Optional<webrtc::VideoFrame> last_frame_;
>+  absl::optional<webrtc::VideoFrame> last_frame_;
>   webrtc::VideoSendStream::Stats stats_;
>   int num_encoder_reconfigurations_ = 0;
> };
>@@ -205,6 +210,9 @@ class FakeVideoReceiveStream final : public webrtc::VideoReceiveStream {
>   void AddSecondarySink(webrtc::RtpPacketSinkInterface* sink) override;
>   void RemoveSecondarySink(const webrtc::RtpPacketSinkInterface* sink) override;
> 
>+  int GetNumAddedSecondarySinks() const;
>+  int GetNumRemovedSecondarySinks() const;
>+
>  private:
>   // webrtc::VideoReceiveStream implementation.
>   void Start() override;
>@@ -215,6 +223,9 @@ class FakeVideoReceiveStream final : public webrtc::VideoReceiveStream {
>   webrtc::VideoReceiveStream::Config config_;
>   bool receiving_;
>   webrtc::VideoReceiveStream::Stats stats_;
>+
>+  int num_added_secondary_sinks_;
>+  int num_removed_secondary_sinks_;
> };
> 
> class FakeFlexfecReceiveStream final : public webrtc::FlexfecReceiveStream {
>@@ -234,10 +245,13 @@ class FakeFlexfecReceiveStream final : public webrtc::FlexfecReceiveStream {
> 
> class FakeCall final : public webrtc::Call, public webrtc::PacketReceiver {
>  public:
>-  explicit FakeCall(const webrtc::Call::Config& config);
>+  FakeCall();
>   ~FakeCall() override;
> 
>-  webrtc::Call::Config GetConfig() const;
>+  webrtc::MockRtpTransportControllerSend* GetMockTransportControllerSend() {
>+    return &transport_controller_send_;
>+  }
>+
>   const std::vector<FakeVideoSendStream*>& GetVideoSendStreams();
>   const std::vector<FakeVideoReceiveStream*>& GetVideoReceiveStreams();
> 
>@@ -290,26 +304,28 @@ class FakeCall final : public webrtc::Call, public webrtc::PacketReceiver {
> 
>   DeliveryStatus DeliverPacket(webrtc::MediaType media_type,
>                                rtc::CopyOnWriteBuffer packet,
>-                               const webrtc::PacketTime& packet_time) override;
>+                               int64_t packet_time_us) override;
>+
>+  webrtc::RtpTransportControllerSendInterface* GetTransportControllerSend()
>+      override {
>+    return &transport_controller_send_;
>+  }
> 
>   webrtc::Call::Stats GetStats() const override;
> 
>-  void SetBitrateConfig(
>-      const webrtc::Call::Config::BitrateConfig& bitrate_config) override;
>-  void SetBitrateConfigMask(
>-      const webrtc::Call::Config::BitrateConfigMask& mask) override;
>   void SetBitrateAllocationStrategy(
>       std::unique_ptr<rtc::BitrateAllocationStrategy>
>           bitrate_allocation_strategy) override;
>-  void OnNetworkRouteChanged(const std::string& transport_name,
>-                             const rtc::NetworkRoute& network_route) override {}
>+
>   void SignalChannelNetworkState(webrtc::MediaType media,
>                                  webrtc::NetworkState state) override;
>   void OnTransportOverheadChanged(webrtc::MediaType media,
>                                   int transport_overhead_per_packet) override;
>   void OnSentPacket(const rtc::SentPacket& sent_packet) override;
> 
>-  webrtc::Call::Config config_;
>+  testing::NiceMock<webrtc::MockRtpTransportControllerSend>
>+      transport_controller_send_;
>+
>   webrtc::NetworkState audio_network_state_;
>   webrtc::NetworkState video_network_state_;
>   rtc::SentPacket last_sent_packet_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/fakewebrtcdeviceinfo.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/fakewebrtcdeviceinfo.h
>index 51a5b8e50cb..7103b4c3437 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/fakewebrtcdeviceinfo.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/fakewebrtcdeviceinfo.h
>@@ -33,9 +33,10 @@ class FakeWebRtcDeviceInfo : public webrtc::VideoCaptureModule::DeviceInfo {
>   }
>   void AddCapability(const std::string& device_id,
>                      const webrtc::VideoCaptureCapability& cap) {
>-    Device* dev = GetDeviceById(
>-        reinterpret_cast<const char*>(device_id.c_str()));
>-    if (!dev) return;
>+    Device* dev =
>+        GetDeviceById(reinterpret_cast<const char*>(device_id.c_str()));
>+    if (!dev)
>+      return;
>     dev->caps.push_back(cap);
>   }
>   virtual uint32_t NumberOfDevices() {
>@@ -49,28 +50,32 @@ class FakeWebRtcDeviceInfo : public webrtc::VideoCaptureModule::DeviceInfo {
>                                 char* product_id,
>                                 uint32_t product_id_len) {
>     Device* dev = GetDeviceByIndex(device_num);
>-    if (!dev) return -1;
>+    if (!dev)
>+      return -1;
>     rtc::strcpyn(reinterpret_cast<char*>(device_name), device_name_len,
>-                       dev->name.c_str());
>+                 dev->name.c_str());
>     rtc::strcpyn(reinterpret_cast<char*>(device_id), device_id_len,
>-                       dev->id.c_str());
>+                 dev->id.c_str());
>     if (product_id) {
>       rtc::strcpyn(reinterpret_cast<char*>(product_id), product_id_len,
>-                         dev->product.c_str());
>+                   dev->product.c_str());
>     }
>     return 0;
>   }
>   virtual int32_t NumberOfCapabilities(const char* device_id) {
>     Device* dev = GetDeviceById(device_id);
>-    if (!dev) return -1;
>+    if (!dev)
>+      return -1;
>     return static_cast<int32_t>(dev->caps.size());
>   }
>   virtual int32_t GetCapability(const char* device_id,
>                                 const uint32_t device_cap_num,
>                                 webrtc::VideoCaptureCapability& cap) {
>     Device* dev = GetDeviceById(device_id);
>-    if (!dev) return -1;
>-    if (device_cap_num >= dev->caps.size()) return -1;
>+    if (!dev)
>+      return -1;
>+    if (device_cap_num >= dev->caps.size())
>+      return -1;
>     cap = dev->caps[device_cap_num];
>     return 0;
>   }
>@@ -84,9 +89,11 @@ class FakeWebRtcDeviceInfo : public webrtc::VideoCaptureModule::DeviceInfo {
>       webrtc::VideoCaptureCapability& resulting) {
>     return -1;  // not implemented
>   }
>-  virtual int32_t DisplayCaptureSettingsDialogBox(
>-      const char* device_id, const char* dialog_title,
>-      void* parent, uint32_t x, uint32_t y) {
>+  virtual int32_t DisplayCaptureSettingsDialogBox(const char* device_id,
>+                                                  const char* dialog_title,
>+                                                  void* parent,
>+                                                  uint32_t x,
>+                                                  uint32_t y) {
>     return -1;  // not implemented
>   }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/fakewebrtcvcmfactory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/fakewebrtcvcmfactory.h
>index 70931e129a1..41ec98df8b7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/fakewebrtcvcmfactory.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/fakewebrtcvcmfactory.h
>@@ -23,9 +23,10 @@ class FakeWebRtcVcmFactory : public cricket::WebRtcVcmFactoryInterface {
>  public:
>   virtual rtc::scoped_refptr<webrtc::VideoCaptureModule> Create(
>       const char* device_id) {
>-    if (!device_info.GetDeviceById(device_id)) return NULL;
>+    if (!device_info.GetDeviceById(device_id))
>+      return NULL;
>     rtc::scoped_refptr<FakeWebRtcVideoCaptureModule> module(
>-        new rtc::RefCountedObject<FakeWebRtcVideoCaptureModule>(this));
>+        new rtc::RefCountedObject<FakeWebRtcVideoCaptureModule>());
>     modules.push_back(module);
>     return module;
>   }
>@@ -34,16 +35,8 @@ class FakeWebRtcVcmFactory : public cricket::WebRtcVcmFactoryInterface {
>   }
>   virtual void DestroyDeviceInfo(webrtc::VideoCaptureModule::DeviceInfo* info) {
>   }
>-  void OnDestroyed(webrtc::VideoCaptureModule* module) {
>-    std::remove(modules.begin(), modules.end(), module);
>-  }
>   FakeWebRtcDeviceInfo device_info;
>   std::vector<rtc::scoped_refptr<FakeWebRtcVideoCaptureModule>> modules;
> };
> 
>-FakeWebRtcVideoCaptureModule::~FakeWebRtcVideoCaptureModule() {
>-  if (factory_)
>-    factory_->OnDestroyed(this);
>-}
>-
> #endif  // MEDIA_ENGINE_FAKEWEBRTCVCMFACTORY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/fakewebrtcvideocapturemodule.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/fakewebrtcvideocapturemodule.h
>index bf23a112774..47f1393c3f3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/fakewebrtcvideocapturemodule.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/fakewebrtcvideocapturemodule.h
>@@ -16,22 +16,22 @@
> #include "api/video/i420_buffer.h"
> #include "media/base/testutils.h"
> #include "media/engine/webrtcvideocapturer.h"
>-
>-class FakeWebRtcVcmFactory;
>+#include "rtc_base/task_queue_for_test.h"
> 
> // Fake class for mocking out webrtc::VideoCaptureModule.
> class FakeWebRtcVideoCaptureModule : public webrtc::VideoCaptureModule {
>  public:
>-  explicit FakeWebRtcVideoCaptureModule(FakeWebRtcVcmFactory* factory)
>-      : factory_(factory), callback_(NULL), running_(false) {}
>-  ~FakeWebRtcVideoCaptureModule();
>+  FakeWebRtcVideoCaptureModule()
>+      : callback_(NULL), running_(false) {}
>+  ~FakeWebRtcVideoCaptureModule() {}
>   void RegisterCaptureDataCallback(
>       rtc::VideoSinkInterface<webrtc::VideoFrame>* callback) override {
>     callback_ = callback;
>   }
>   void DeRegisterCaptureDataCallback() override { callback_ = NULL; }
>   int32_t StartCapture(const webrtc::VideoCaptureCapability& cap) override {
>-    if (running_) return -1;
>+    if (running_)
>+      return -1;
>     cap_ = cap;
>     running_ = true;
>     return 0;
>@@ -45,7 +45,8 @@ class FakeWebRtcVideoCaptureModule : public webrtc::VideoCaptureModule {
>   }
>   bool CaptureStarted() override { return running_; }
>   int32_t CaptureSettings(webrtc::VideoCaptureCapability& settings) override {
>-    if (!running_) return -1;
>+    if (!running_)
>+      return -1;
>     settings = cap_;
>     return 0;
>   }
>@@ -60,25 +61,24 @@ class FakeWebRtcVideoCaptureModule : public webrtc::VideoCaptureModule {
>     return true;  // Rotation compensation is turned on.
>   }
>   void SendFrame(int w, int h) {
>-    if (!running_) return;
>+    if (!running_ || !callback_)
>+      return;
> 
>-    rtc::scoped_refptr<webrtc::I420Buffer> buffer =
>-        webrtc::I420Buffer::Create(w, h);
>-    // Initialize memory to satisfy DrMemory tests. See
>-    // https://bugs.chromium.org/p/libyuv/issues/detail?id=377
>-    buffer->InitializeData();
>-    if (callback_) {
>-      callback_->OnFrame(
>-          webrtc::VideoFrame(buffer, 0, 0, webrtc::kVideoRotation_0));
>-    }
>+    task_queue_.SendTask([this, w, h]() {
>+      rtc::scoped_refptr<webrtc::I420Buffer> buffer =
>+          webrtc::I420Buffer::Create(w, h);
>+      // Initialize memory to satisfy DrMemory tests. See
>+      // https://bugs.chromium.org/p/libyuv/issues/detail?id=377
>+      buffer->InitializeData();
>+      callback_->OnFrame(webrtc::VideoFrame(buffer, webrtc::kVideoRotation_0,
>+                                            0 /* timestamp_us */));
>+    });
>   }
> 
>-  const webrtc::VideoCaptureCapability& cap() const {
>-    return cap_;
>-  }
>+  const webrtc::VideoCaptureCapability& cap() const { return cap_; }
> 
>  private:
>-  FakeWebRtcVcmFactory* factory_;
>+  rtc::test::TaskQueueForTest task_queue_{"FakeWebRtcVideoCaptureModule"};
>   rtc::VideoSinkInterface<webrtc::VideoFrame>* callback_;
>   bool running_;
>   webrtc::VideoCaptureCapability cap_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/fakewebrtcvideoengine.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/fakewebrtcvideoengine.cc
>new file mode 100644
>index 00000000000..7081b64933a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/fakewebrtcvideoengine.cc
>@@ -0,0 +1,296 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "media/engine/fakewebrtcvideoengine.h"
>+
>+#include "media/base/codec.h"
>+#include "media/engine/simulcast_encoder_adapter.h"
>+#include "media/engine/webrtcvideodecoderfactory.h"
>+#include "media/engine/webrtcvideoencoderfactory.h"
>+#include "modules/video_coding/include/video_error_codes.h"
>+#include "rtc_base/gunit.h"
>+#include "rtc_base/stringutils.h"
>+
>+namespace cricket {
>+
>+namespace {
>+
>+static const int kEventTimeoutMs = 10000;
>+
>+bool IsFormatSupported(
>+    const std::vector<webrtc::SdpVideoFormat>& supported_formats,
>+    const webrtc::SdpVideoFormat& format) {
>+  for (const webrtc::SdpVideoFormat& supported_format : supported_formats) {
>+    if (IsSameCodec(format.name, format.parameters, supported_format.name,
>+                    supported_format.parameters)) {
>+      return true;
>+    }
>+  }
>+  return false;
>+}
>+
>+}  // namespace
>+
>+// Decoder.
>+FakeWebRtcVideoDecoder::FakeWebRtcVideoDecoder(
>+    FakeWebRtcVideoDecoderFactory* factory)
>+    : num_frames_received_(0), factory_(factory) {}
>+
>+FakeWebRtcVideoDecoder::~FakeWebRtcVideoDecoder() {
>+  if (factory_) {
>+    factory_->DecoderDestroyed(this);
>+  }
>+}
>+
>+int32_t FakeWebRtcVideoDecoder::InitDecode(const webrtc::VideoCodec*, int32_t) {
>+  return WEBRTC_VIDEO_CODEC_OK;
>+}
>+
>+int32_t FakeWebRtcVideoDecoder::Decode(const webrtc::EncodedImage&,
>+                                       bool,
>+                                       const webrtc::CodecSpecificInfo*,
>+                                       int64_t) {
>+  num_frames_received_++;
>+  return WEBRTC_VIDEO_CODEC_OK;
>+}
>+
>+int32_t FakeWebRtcVideoDecoder::RegisterDecodeCompleteCallback(
>+    webrtc::DecodedImageCallback*) {
>+  return WEBRTC_VIDEO_CODEC_OK;
>+}
>+
>+int32_t FakeWebRtcVideoDecoder::Release() {
>+  return WEBRTC_VIDEO_CODEC_OK;
>+}
>+
>+int FakeWebRtcVideoDecoder::GetNumFramesReceived() const {
>+  return num_frames_received_;
>+}
>+
>+// Decoder factory.
>+FakeWebRtcVideoDecoderFactory::FakeWebRtcVideoDecoderFactory()
>+    : num_created_decoders_(0) {}
>+
>+std::vector<webrtc::SdpVideoFormat>
>+FakeWebRtcVideoDecoderFactory::GetSupportedFormats() const {
>+  std::vector<webrtc::SdpVideoFormat> formats;
>+
>+  for (const webrtc::SdpVideoFormat& format : supported_codec_formats_) {
>+    // Don't add same codec twice.
>+    if (!IsFormatSupported(formats, format))
>+      formats.push_back(format);
>+  }
>+
>+  return formats;
>+}
>+
>+std::unique_ptr<webrtc::VideoDecoder>
>+FakeWebRtcVideoDecoderFactory::CreateVideoDecoder(
>+    const webrtc::SdpVideoFormat& format) {
>+  if (IsFormatSupported(supported_codec_formats_, format)) {
>+    num_created_decoders_++;
>+    std::unique_ptr<FakeWebRtcVideoDecoder> decoder =
>+        absl::make_unique<FakeWebRtcVideoDecoder>(this);
>+    decoders_.push_back(decoder.get());
>+    return decoder;
>+  }
>+
>+  return nullptr;
>+}
>+
>+void FakeWebRtcVideoDecoderFactory::DecoderDestroyed(
>+    FakeWebRtcVideoDecoder* decoder) {
>+  decoders_.erase(std::remove(decoders_.begin(), decoders_.end(), decoder),
>+                  decoders_.end());
>+}
>+
>+void FakeWebRtcVideoDecoderFactory::AddSupportedVideoCodecType(
>+    const webrtc::SdpVideoFormat& format) {
>+  supported_codec_formats_.push_back(format);
>+}
>+
>+int FakeWebRtcVideoDecoderFactory::GetNumCreatedDecoders() {
>+  return num_created_decoders_;
>+}
>+
>+const std::vector<FakeWebRtcVideoDecoder*>&
>+FakeWebRtcVideoDecoderFactory::decoders() {
>+  return decoders_;
>+}
>+
>+// Encoder.
>+FakeWebRtcVideoEncoder::FakeWebRtcVideoEncoder(
>+    FakeWebRtcVideoEncoderFactory* factory)
>+    : init_encode_event_(false, false),
>+      num_frames_encoded_(0),
>+      factory_(factory) {}
>+
>+FakeWebRtcVideoEncoder::~FakeWebRtcVideoEncoder() {
>+  if (factory_) {
>+    factory_->EncoderDestroyed(this);
>+  }
>+}
>+
>+int32_t FakeWebRtcVideoEncoder::InitEncode(
>+    const webrtc::VideoCodec* codecSettings,
>+    int32_t numberOfCores,
>+    size_t maxPayloadSize) {
>+  rtc::CritScope lock(&crit_);
>+  codec_settings_ = *codecSettings;
>+  init_encode_event_.Set();
>+  return WEBRTC_VIDEO_CODEC_OK;
>+}
>+
>+int32_t FakeWebRtcVideoEncoder::Encode(
>+    const webrtc::VideoFrame& inputImage,
>+    const webrtc::CodecSpecificInfo* codecSpecificInfo,
>+    const std::vector<webrtc::FrameType>* frame_types) {
>+  rtc::CritScope lock(&crit_);
>+  ++num_frames_encoded_;
>+  init_encode_event_.Set();
>+  return WEBRTC_VIDEO_CODEC_OK;
>+}
>+
>+int32_t FakeWebRtcVideoEncoder::RegisterEncodeCompleteCallback(
>+    webrtc::EncodedImageCallback* callback) {
>+  return WEBRTC_VIDEO_CODEC_OK;
>+}
>+
>+int32_t FakeWebRtcVideoEncoder::Release() {
>+  return WEBRTC_VIDEO_CODEC_OK;
>+}
>+
>+int32_t FakeWebRtcVideoEncoder::SetChannelParameters(uint32_t packetLoss,
>+                                                     int64_t rtt) {
>+  return WEBRTC_VIDEO_CODEC_OK;
>+}
>+
>+int32_t FakeWebRtcVideoEncoder::SetRateAllocation(
>+    const webrtc::VideoBitrateAllocation& allocation,
>+    uint32_t framerate) {
>+  return WEBRTC_VIDEO_CODEC_OK;
>+}
>+
>+bool FakeWebRtcVideoEncoder::WaitForInitEncode() {
>+  return init_encode_event_.Wait(kEventTimeoutMs);
>+}
>+
>+webrtc::VideoCodec FakeWebRtcVideoEncoder::GetCodecSettings() {
>+  rtc::CritScope lock(&crit_);
>+  return codec_settings_;
>+}
>+
>+int FakeWebRtcVideoEncoder::GetNumEncodedFrames() {
>+  rtc::CritScope lock(&crit_);
>+  return num_frames_encoded_;
>+}
>+
>+// Video encoder factory.
>+FakeWebRtcVideoEncoderFactory::FakeWebRtcVideoEncoderFactory()
>+    : created_video_encoder_event_(false, false),
>+      num_created_encoders_(0),
>+      encoders_have_internal_sources_(false),
>+      vp8_factory_mode_(false) {}
>+
>+std::vector<webrtc::SdpVideoFormat>
>+FakeWebRtcVideoEncoderFactory::GetSupportedFormats() const {
>+  std::vector<webrtc::SdpVideoFormat> formats;
>+
>+  for (const webrtc::SdpVideoFormat& format : formats_) {
>+    // Don't add same codec twice.
>+    if (!IsFormatSupported(formats, format))
>+      formats.push_back(format);
>+  }
>+
>+  return formats;
>+}
>+
>+std::unique_ptr<webrtc::VideoEncoder>
>+FakeWebRtcVideoEncoderFactory::CreateVideoEncoder(
>+    const webrtc::SdpVideoFormat& format) {
>+  rtc::CritScope lock(&crit_);
>+  std::unique_ptr<webrtc::VideoEncoder> encoder;
>+  if (IsFormatSupported(formats_, format)) {
>+    if (CodecNamesEq(format.name.c_str(), kVp8CodecName) &&
>+        !vp8_factory_mode_) {
>+      // The simulcast adapter will ask this factory for multiple VP8
>+      // encoders. Enter vp8_factory_mode so that we now create these encoders
>+      // instead of more adapters.
>+      vp8_factory_mode_ = true;
>+      encoder =
>+          absl::make_unique<webrtc::SimulcastEncoderAdapter>(this, format);
>+    } else {
>+      num_created_encoders_++;
>+      created_video_encoder_event_.Set();
>+      encoder = absl::make_unique<FakeWebRtcVideoEncoder>(this);
>+      encoders_.push_back(static_cast<FakeWebRtcVideoEncoder*>(encoder.get()));
>+    }
>+  }
>+  return encoder;
>+}
>+
>+webrtc::VideoEncoderFactory::CodecInfo
>+FakeWebRtcVideoEncoderFactory::QueryVideoEncoder(
>+    const webrtc::SdpVideoFormat& format) const {
>+  webrtc::VideoEncoderFactory::CodecInfo info;
>+  info.has_internal_source = encoders_have_internal_sources_;
>+  info.is_hardware_accelerated = true;
>+  return info;
>+}
>+
>+bool FakeWebRtcVideoEncoderFactory::WaitForCreatedVideoEncoders(
>+    int num_encoders) {
>+  int64_t start_offset_ms = rtc::TimeMillis();
>+  int64_t wait_time = kEventTimeoutMs;
>+  do {
>+    if (GetNumCreatedEncoders() >= num_encoders)
>+      return true;
>+    wait_time = kEventTimeoutMs - (rtc::TimeMillis() - start_offset_ms);
>+  } while (wait_time > 0 && created_video_encoder_event_.Wait(wait_time));
>+  return false;
>+}
>+
>+void FakeWebRtcVideoEncoderFactory::EncoderDestroyed(
>+    FakeWebRtcVideoEncoder* encoder) {
>+  rtc::CritScope lock(&crit_);
>+  encoders_.erase(std::remove(encoders_.begin(), encoders_.end(), encoder),
>+                  encoders_.end());
>+}
>+
>+void FakeWebRtcVideoEncoderFactory::set_encoders_have_internal_sources(
>+    bool internal_source) {
>+  encoders_have_internal_sources_ = internal_source;
>+}
>+
>+void FakeWebRtcVideoEncoderFactory::AddSupportedVideoCodec(
>+    const webrtc::SdpVideoFormat& format) {
>+  formats_.push_back(format);
>+}
>+
>+void FakeWebRtcVideoEncoderFactory::AddSupportedVideoCodecType(
>+    const std::string& name) {
>+  // This is to match the default H264 params of cricket::VideoCodec.
>+  cricket::VideoCodec video_codec(name);
>+  formats_.push_back(
>+      webrtc::SdpVideoFormat(video_codec.name, video_codec.params));
>+}
>+
>+int FakeWebRtcVideoEncoderFactory::GetNumCreatedEncoders() {
>+  rtc::CritScope lock(&crit_);
>+  return num_created_encoders_;
>+}
>+
>+const std::vector<FakeWebRtcVideoEncoder*>
>+FakeWebRtcVideoEncoderFactory::encoders() {
>+  rtc::CritScope lock(&crit_);
>+  return encoders_;
>+}
>+
>+}  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/fakewebrtcvideoengine.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/fakewebrtcvideoengine.h
>index 2153f6be386..f4287a31d26 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/fakewebrtcvideoengine.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/fakewebrtcvideoengine.h
>@@ -11,248 +11,124 @@
> #ifndef MEDIA_ENGINE_FAKEWEBRTCVIDEOENGINE_H_
> #define MEDIA_ENGINE_FAKEWEBRTCVIDEOENGINE_H_
> 
>-#include <map>
>-#include <set>
>-#include <vector>
>+#include <memory>
> #include <string>
>+#include <vector>
> 
>+#include "absl/memory/memory.h"
> #include "api/video_codecs/video_decoder.h"
>+#include "api/video_codecs/video_decoder_factory.h"
> #include "api/video_codecs/video_encoder.h"
>-#include "media/base/codec.h"
>-#include "media/engine/webrtcvideodecoderfactory.h"
>-#include "media/engine/webrtcvideoencoderfactory.h"
>-#include "modules/video_coding/include/video_error_codes.h"
>-#include "rtc_base/basictypes.h"
>+#include "api/video_codecs/video_encoder_factory.h"
> #include "rtc_base/criticalsection.h"
>-#include "rtc_base/gunit.h"
>-#include "rtc_base/stringutils.h"
>+#include "rtc_base/event.h"
> #include "rtc_base/thread_annotations.h"
> 
> namespace cricket {
>-static const int kEventTimeoutMs = 10000;
>+
>+class FakeWebRtcVideoDecoderFactory;
>+class FakeWebRtcVideoEncoderFactory;
> 
> // Fake class for mocking out webrtc::VideoDecoder
> class FakeWebRtcVideoDecoder : public webrtc::VideoDecoder {
>  public:
>-  FakeWebRtcVideoDecoder() : num_frames_received_(0) {}
>-
>-  virtual int32_t InitDecode(const webrtc::VideoCodec*, int32_t) {
>-    return WEBRTC_VIDEO_CODEC_OK;
>-  }
>+  explicit FakeWebRtcVideoDecoder(FakeWebRtcVideoDecoderFactory* factory);
>+  ~FakeWebRtcVideoDecoder();
> 
>-  virtual int32_t Decode(const webrtc::EncodedImage&,
>-                         bool,
>-                         const webrtc::RTPFragmentationHeader*,
>-                         const webrtc::CodecSpecificInfo*,
>-                         int64_t) {
>-    num_frames_received_++;
>-    return WEBRTC_VIDEO_CODEC_OK;
>-  }
>+  int32_t InitDecode(const webrtc::VideoCodec*, int32_t) override;
>+  int32_t Decode(const webrtc::EncodedImage&,
>+                 bool,
>+                 const webrtc::CodecSpecificInfo*,
>+                 int64_t) override;
>+  int32_t RegisterDecodeCompleteCallback(
>+      webrtc::DecodedImageCallback*) override;
>+  int32_t Release() override;
> 
>-  virtual int32_t RegisterDecodeCompleteCallback(
>-      webrtc::DecodedImageCallback*) {
>-    return WEBRTC_VIDEO_CODEC_OK;
>-  }
>-
>-  virtual int32_t Release() { return WEBRTC_VIDEO_CODEC_OK; }
>-
>-  int GetNumFramesReceived() const {
>-    return num_frames_received_;
>-  }
>+  int GetNumFramesReceived() const;
> 
>  private:
>   int num_frames_received_;
>+  FakeWebRtcVideoDecoderFactory* factory_;
> };
> 
>-// Fake class for mocking out WebRtcVideoDecoderFactory.
>-class FakeWebRtcVideoDecoderFactory : public WebRtcVideoDecoderFactory {
>+// Fake class for mocking out webrtc::VideoDecoderFactory.
>+class FakeWebRtcVideoDecoderFactory : public webrtc::VideoDecoderFactory {
>  public:
>-  FakeWebRtcVideoDecoderFactory()
>-      : num_created_decoders_(0) {
>-  }
>-
>-  virtual webrtc::VideoDecoder* CreateVideoDecoder(
>-      webrtc::VideoCodecType type) {
>-    if (supported_codec_types_.count(type) == 0) {
>-      return NULL;
>-    }
>-    FakeWebRtcVideoDecoder* decoder = new FakeWebRtcVideoDecoder();
>-    decoders_.push_back(decoder);
>-    num_created_decoders_++;
>-    return decoder;
>-  }
>+  FakeWebRtcVideoDecoderFactory();
> 
>-  virtual webrtc::VideoDecoder* CreateVideoDecoderWithParams(
>-      webrtc::VideoCodecType type,
>-      VideoDecoderParams params) {
>-    params_.push_back(params);
>-    return CreateVideoDecoder(type);
>-  }
>+  std::vector<webrtc::SdpVideoFormat> GetSupportedFormats() const override;
>+  std::unique_ptr<webrtc::VideoDecoder> CreateVideoDecoder(
>+      const webrtc::SdpVideoFormat& format) override;
> 
>-  virtual void DestroyVideoDecoder(webrtc::VideoDecoder* decoder) {
>-    decoders_.erase(
>-        std::remove(decoders_.begin(), decoders_.end(), decoder),
>-        decoders_.end());
>-    delete decoder;
>-  }
>-
>-  void AddSupportedVideoCodecType(webrtc::VideoCodecType type) {
>-    supported_codec_types_.insert(type);
>-  }
>-
>-  int GetNumCreatedDecoders() {
>-    return num_created_decoders_;
>-  }
>-
>-  const std::vector<FakeWebRtcVideoDecoder*>& decoders() {
>-    return decoders_;
>-  }
>-
>-  const std::vector<VideoDecoderParams>& params() { return params_; }
>+  void DecoderDestroyed(FakeWebRtcVideoDecoder* decoder);
>+  void AddSupportedVideoCodecType(const webrtc::SdpVideoFormat& format);
>+  int GetNumCreatedDecoders();
>+  const std::vector<FakeWebRtcVideoDecoder*>& decoders();
> 
>  private:
>-  std::set<webrtc::VideoCodecType> supported_codec_types_;
>+  std::vector<webrtc::SdpVideoFormat> supported_codec_formats_;
>   std::vector<FakeWebRtcVideoDecoder*> decoders_;
>   int num_created_decoders_;
>-  std::vector<VideoDecoderParams> params_;
> };
> 
> // Fake class for mocking out webrtc::VideoEnoder
> class FakeWebRtcVideoEncoder : public webrtc::VideoEncoder {
>  public:
>-  FakeWebRtcVideoEncoder()
>-      : init_encode_event_(false, false), num_frames_encoded_(0) {}
>+  explicit FakeWebRtcVideoEncoder(FakeWebRtcVideoEncoderFactory* factory);
>+  ~FakeWebRtcVideoEncoder();
> 
>   int32_t InitEncode(const webrtc::VideoCodec* codecSettings,
>                      int32_t numberOfCores,
>-                     size_t maxPayloadSize) override {
>-    rtc::CritScope lock(&crit_);
>-    codec_settings_ = *codecSettings;
>-    init_encode_event_.Set();
>-    return WEBRTC_VIDEO_CODEC_OK;
>-  }
>-
>-  bool WaitForInitEncode() { return init_encode_event_.Wait(kEventTimeoutMs); }
>-
>-  webrtc::VideoCodec GetCodecSettings() {
>-    rtc::CritScope lock(&crit_);
>-    return codec_settings_;
>-  }
>-
>+                     size_t maxPayloadSize) override;
>   int32_t Encode(const webrtc::VideoFrame& inputImage,
>                  const webrtc::CodecSpecificInfo* codecSpecificInfo,
>-                 const std::vector<webrtc::FrameType>* frame_types) override {
>-    rtc::CritScope lock(&crit_);
>-    ++num_frames_encoded_;
>-    init_encode_event_.Set();
>-    return WEBRTC_VIDEO_CODEC_OK;
>-  }
>-
>+                 const std::vector<webrtc::FrameType>* frame_types) override;
>   int32_t RegisterEncodeCompleteCallback(
>-      webrtc::EncodedImageCallback* callback) override {
>-    return WEBRTC_VIDEO_CODEC_OK;
>-  }
>-
>-  int32_t Release() override { return WEBRTC_VIDEO_CODEC_OK; }
>-
>-  int32_t SetChannelParameters(uint32_t packetLoss, int64_t rtt) override {
>-    return WEBRTC_VIDEO_CODEC_OK;
>-  }
>-
>-  int32_t SetRateAllocation(const webrtc::BitrateAllocation& allocation,
>-                            uint32_t framerate) override {
>-    return WEBRTC_VIDEO_CODEC_OK;
>-  }
>+      webrtc::EncodedImageCallback* callback) override;
>+  int32_t Release() override;
>+  int32_t SetChannelParameters(uint32_t packetLoss, int64_t rtt) override;
>+  int32_t SetRateAllocation(const webrtc::VideoBitrateAllocation& allocation,
>+                            uint32_t framerate) override;
> 
>-  int GetNumEncodedFrames() {
>-    rtc::CritScope lock(&crit_);
>-    return num_frames_encoded_;
>-  }
>+  bool WaitForInitEncode();
>+  webrtc::VideoCodec GetCodecSettings();
>+  int GetNumEncodedFrames();
> 
>  private:
>   rtc::CriticalSection crit_;
>   rtc::Event init_encode_event_;
>   int num_frames_encoded_ RTC_GUARDED_BY(crit_);
>   webrtc::VideoCodec codec_settings_ RTC_GUARDED_BY(crit_);
>+  FakeWebRtcVideoEncoderFactory* factory_;
> };
> 
>-// Fake class for mocking out WebRtcVideoEncoderFactory.
>-class FakeWebRtcVideoEncoderFactory : public WebRtcVideoEncoderFactory {
>+// Fake class for mocking out webrtc::VideoEncoderFactory.
>+class FakeWebRtcVideoEncoderFactory : public webrtc::VideoEncoderFactory {
>  public:
>-  FakeWebRtcVideoEncoderFactory()
>-      : created_video_encoder_event_(false, false),
>-        num_created_encoders_(0),
>-        encoders_have_internal_sources_(false) {}
>-
>-  webrtc::VideoEncoder* CreateVideoEncoder(
>-      const cricket::VideoCodec& codec) override {
>-    rtc::CritScope lock(&crit_);
>-    if (!FindMatchingCodec(codecs_, codec))
>-      return nullptr;
>-    FakeWebRtcVideoEncoder* encoder = new FakeWebRtcVideoEncoder();
>-    encoders_.push_back(encoder);
>-    num_created_encoders_++;
>-    created_video_encoder_event_.Set();
>-    return encoder;
>-  }
>-
>-  bool WaitForCreatedVideoEncoders(int num_encoders) {
>-    int64_t start_offset_ms = rtc::TimeMillis();
>-    int64_t wait_time = kEventTimeoutMs;
>-    do {
>-      if (GetNumCreatedEncoders() >= num_encoders)
>-        return true;
>-      wait_time = kEventTimeoutMs - (rtc::TimeMillis() - start_offset_ms);
>-    } while (wait_time > 0 && created_video_encoder_event_.Wait(wait_time));
>-    return false;
>-  }
>-
>-  void DestroyVideoEncoder(webrtc::VideoEncoder* encoder) override {
>-    rtc::CritScope lock(&crit_);
>-    encoders_.erase(
>-        std::remove(encoders_.begin(), encoders_.end(), encoder),
>-        encoders_.end());
>-    delete encoder;
>-  }
>-
>-  const std::vector<cricket::VideoCodec>& supported_codecs() const override {
>-    return codecs_;
>-  }
>-
>-  bool EncoderTypeHasInternalSource(
>-      webrtc::VideoCodecType type) const override {
>-    return encoders_have_internal_sources_;
>-  }
>-
>-  void set_encoders_have_internal_sources(bool internal_source) {
>-    encoders_have_internal_sources_ = internal_source;
>-  }
>-
>-  void AddSupportedVideoCodec(const cricket::VideoCodec& codec) {
>-    codecs_.push_back(codec);
>-  }
>-
>-  void AddSupportedVideoCodecType(const std::string& name) {
>-    codecs_.push_back(cricket::VideoCodec(name));
>-  }
>-
>-  int GetNumCreatedEncoders() {
>-    rtc::CritScope lock(&crit_);
>-    return num_created_encoders_;
>-  }
>-
>-  const std::vector<FakeWebRtcVideoEncoder*> encoders() {
>-    rtc::CritScope lock(&crit_);
>-    return encoders_;
>-  }
>+  FakeWebRtcVideoEncoderFactory();
>+
>+  std::vector<webrtc::SdpVideoFormat> GetSupportedFormats() const override;
>+  std::unique_ptr<webrtc::VideoEncoder> CreateVideoEncoder(
>+      const webrtc::SdpVideoFormat& format) override;
>+  CodecInfo QueryVideoEncoder(
>+      const webrtc::SdpVideoFormat& format) const override;
>+
>+  bool WaitForCreatedVideoEncoders(int num_encoders);
>+  void EncoderDestroyed(FakeWebRtcVideoEncoder* encoder);
>+  void set_encoders_have_internal_sources(bool internal_source);
>+  void AddSupportedVideoCodec(const webrtc::SdpVideoFormat& format);
>+  void AddSupportedVideoCodecType(const std::string& name);
>+  int GetNumCreatedEncoders();
>+  const std::vector<FakeWebRtcVideoEncoder*> encoders();
> 
>  private:
>   rtc::CriticalSection crit_;
>   rtc::Event created_video_encoder_event_;
>-  std::vector<cricket::VideoCodec> codecs_;
>+  std::vector<webrtc::SdpVideoFormat> formats_;
>   std::vector<FakeWebRtcVideoEncoder*> encoders_ RTC_GUARDED_BY(crit_);
>   int num_created_encoders_ RTC_GUARDED_BY(crit_);
>   bool encoders_have_internal_sources_;
>+  bool vp8_factory_mode_;
> };
> 
> }  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/internaldecoderfactory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/internaldecoderfactory.cc
>index 57daca256c4..df74773c86d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/internaldecoderfactory.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/internaldecoderfactory.cc
>@@ -10,6 +10,7 @@
> 
> #include "media/engine/internaldecoderfactory.h"
> 
>+#include "api/video_codecs/sdp_video_format.h"
> #include "media/base/mediaconstants.h"
> #include "modules/video_coding/codecs/h264/include/h264.h"
> #include "modules/video_coding/codecs/vp8/include/vp8.h"
>@@ -19,13 +20,29 @@
> 
> namespace webrtc {
> 
>+namespace {
>+
>+bool IsFormatSupported(
>+    const std::vector<webrtc::SdpVideoFormat>& supported_formats,
>+    const webrtc::SdpVideoFormat& format) {
>+  for (const webrtc::SdpVideoFormat& supported_format : supported_formats) {
>+    if (cricket::IsSameCodec(format.name, format.parameters,
>+                             supported_format.name,
>+                             supported_format.parameters)) {
>+      return true;
>+    }
>+  }
>+  return false;
>+}
>+
>+}  // namespace
>+
> std::vector<SdpVideoFormat> InternalDecoderFactory::GetSupportedFormats()
>     const {
>   std::vector<SdpVideoFormat> formats;
>-  if (VP8Decoder::IsSupported())
>-    formats.push_back(SdpVideoFormat(cricket::kVp8CodecName));
>-  if (VP9Decoder::IsSupported())
>-    formats.push_back(SdpVideoFormat(cricket::kVp9CodecName));
>+  formats.push_back(SdpVideoFormat(cricket::kVp8CodecName));
>+  for (const SdpVideoFormat& format : SupportedVP9Codecs())
>+    formats.push_back(format);
>   for (const SdpVideoFormat& h264_format : SupportedH264Codecs())
>     formats.push_back(h264_format);
>   return formats;
>@@ -33,18 +50,19 @@ std::vector<SdpVideoFormat> InternalDecoderFactory::GetSupportedFormats()
> 
> std::unique_ptr<VideoDecoder> InternalDecoderFactory::CreateVideoDecoder(
>     const SdpVideoFormat& format) {
>+  if (!IsFormatSupported(GetSupportedFormats(), format)) {
>+    RTC_LOG(LS_ERROR) << "Trying to create decoder for unsupported format";
>+    return nullptr;
>+  }
>+
>   if (cricket::CodecNamesEq(format.name, cricket::kVp8CodecName))
>     return VP8Decoder::Create();
>-
>-  if (cricket::CodecNamesEq(format.name, cricket::kVp9CodecName)) {
>-    RTC_DCHECK(VP9Decoder::IsSupported());
>+  if (cricket::CodecNamesEq(format.name, cricket::kVp9CodecName))
>     return VP9Decoder::Create();
>-  }
>-
>   if (cricket::CodecNamesEq(format.name, cricket::kH264CodecName))
>     return H264Decoder::Create();
> 
>-  RTC_LOG(LS_ERROR) << "Trying to create decoder for unsupported format";
>+  RTC_NOTREACHED();
>   return nullptr;
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/internalencoderfactory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/internalencoderfactory.cc
>index 0390646add4..e6c3c2e72fd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/internalencoderfactory.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/internalencoderfactory.cc
>@@ -12,6 +12,7 @@
> 
> #include <utility>
> 
>+#include "api/video_codecs/sdp_video_format.h"
> #include "modules/video_coding/codecs/h264/include/h264.h"
> #include "modules/video_coding/codecs/vp8/include/vp8.h"
> #include "modules/video_coding/codecs/vp9/include/vp9.h"
>@@ -22,14 +23,11 @@ namespace webrtc {
> std::vector<SdpVideoFormat> InternalEncoderFactory::GetSupportedFormats()
>     const {
>   std::vector<SdpVideoFormat> supported_codecs;
>-  if (VP8Encoder::IsSupported())
>-    supported_codecs.push_back(SdpVideoFormat(cricket::kVp8CodecName));
>-  if (webrtc::VP9Encoder::IsSupported())
>-    supported_codecs.push_back(SdpVideoFormat(cricket::kVp9CodecName));
>-
>+  supported_codecs.push_back(SdpVideoFormat(cricket::kVp8CodecName));
>+  for (const webrtc::SdpVideoFormat& format : webrtc::SupportedVP9Codecs())
>+    supported_codecs.push_back(format);
>   for (const webrtc::SdpVideoFormat& format : webrtc::SupportedH264Codecs())
>     supported_codecs.push_back(format);
>-
>   return supported_codecs;
> }
> 
>@@ -45,13 +43,10 @@ std::unique_ptr<VideoEncoder> InternalEncoderFactory::CreateVideoEncoder(
>     const SdpVideoFormat& format) {
>   if (cricket::CodecNamesEq(format.name, cricket::kVp8CodecName))
>     return VP8Encoder::Create();
>-
>   if (cricket::CodecNamesEq(format.name, cricket::kVp9CodecName))
>-    return VP9Encoder::Create();
>-
>+    return VP9Encoder::Create(cricket::VideoCodec(format));
>   if (cricket::CodecNamesEq(format.name, cricket::kH264CodecName))
>     return H264Encoder::Create(cricket::VideoCodec(format));
>-
>   RTC_LOG(LS_ERROR) << "Trying to created encoder of unsupported format "
>                     << format.name;
>   return nullptr;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/multiplexcodecfactory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/multiplexcodecfactory.cc
>new file mode 100644
>index 00000000000..236a2e81acc
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/multiplexcodecfactory.cc
>@@ -0,0 +1,118 @@
>+/*
>+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "media/engine/multiplexcodecfactory.h"
>+
>+#include <utility>
>+
>+#include "api/video_codecs/sdp_video_format.h"
>+#include "media/base/codec.h"
>+#include "media/base/mediaconstants.h"
>+#include "modules/video_coding/codecs/multiplex/include/multiplex_decoder_adapter.h"
>+#include "modules/video_coding/codecs/multiplex/include/multiplex_encoder_adapter.h"
>+#include "rtc_base/logging.h"
>+
>+namespace {
>+
>+bool IsMultiplexCodec(const cricket::VideoCodec& codec) {
>+  return cricket::CodecNamesEq(codec.name.c_str(),
>+                               cricket::kMultiplexCodecName);
>+}
>+
>+}  // anonymous namespace
>+
>+namespace webrtc {
>+
>+constexpr const char* kMultiplexAssociatedCodecName = cricket::kVp9CodecName;
>+
>+MultiplexEncoderFactory::MultiplexEncoderFactory(
>+    std::unique_ptr<VideoEncoderFactory> factory,
>+    bool supports_augmenting_data)
>+    : factory_(std::move(factory)),
>+      supports_augmenting_data_(supports_augmenting_data) {}
>+
>+std::vector<SdpVideoFormat> MultiplexEncoderFactory::GetSupportedFormats()
>+    const {
>+  std::vector<SdpVideoFormat> formats = factory_->GetSupportedFormats();
>+  for (const auto& format : formats) {
>+    if (cricket::CodecNamesEq(format.name, kMultiplexAssociatedCodecName)) {
>+      SdpVideoFormat multiplex_format = format;
>+      multiplex_format.parameters[cricket::kCodecParamAssociatedCodecName] =
>+          format.name;
>+      multiplex_format.name = cricket::kMultiplexCodecName;
>+      formats.push_back(multiplex_format);
>+      break;
>+    }
>+  }
>+  return formats;
>+}
>+
>+VideoEncoderFactory::CodecInfo MultiplexEncoderFactory::QueryVideoEncoder(
>+    const SdpVideoFormat& format) const {
>+  if (!IsMultiplexCodec(cricket::VideoCodec(format)))
>+    return factory_->QueryVideoEncoder(format);
>+  return factory_->QueryVideoEncoder(
>+      SdpVideoFormat(kMultiplexAssociatedCodecName));
>+}
>+
>+std::unique_ptr<VideoEncoder> MultiplexEncoderFactory::CreateVideoEncoder(
>+    const SdpVideoFormat& format) {
>+  if (!IsMultiplexCodec(cricket::VideoCodec(format)))
>+    return factory_->CreateVideoEncoder(format);
>+  const auto& it =
>+      format.parameters.find(cricket::kCodecParamAssociatedCodecName);
>+  if (it == format.parameters.end()) {
>+    RTC_LOG(LS_ERROR) << "No assicated codec for multiplex.";
>+    return nullptr;
>+  }
>+  SdpVideoFormat associated_format = format;
>+  associated_format.name = it->second;
>+  return std::unique_ptr<VideoEncoder>(new MultiplexEncoderAdapter(
>+      factory_.get(), associated_format, supports_augmenting_data_));
>+}
>+
>+MultiplexDecoderFactory::MultiplexDecoderFactory(
>+    std::unique_ptr<VideoDecoderFactory> factory,
>+    bool supports_augmenting_data)
>+    : factory_(std::move(factory)),
>+      supports_augmenting_data_(supports_augmenting_data) {}
>+
>+std::vector<SdpVideoFormat> MultiplexDecoderFactory::GetSupportedFormats()
>+    const {
>+  std::vector<SdpVideoFormat> formats = factory_->GetSupportedFormats();
>+  for (const auto& format : formats) {
>+    if (cricket::CodecNamesEq(format.name, kMultiplexAssociatedCodecName)) {
>+      SdpVideoFormat multiplex_format = format;
>+      multiplex_format.parameters[cricket::kCodecParamAssociatedCodecName] =
>+          format.name;
>+      multiplex_format.name = cricket::kMultiplexCodecName;
>+      formats.push_back(multiplex_format);
>+    }
>+  }
>+  return formats;
>+}
>+
>+std::unique_ptr<VideoDecoder> MultiplexDecoderFactory::CreateVideoDecoder(
>+    const SdpVideoFormat& format) {
>+  if (!IsMultiplexCodec(cricket::VideoCodec(format)))
>+    return factory_->CreateVideoDecoder(format);
>+  const auto& it =
>+      format.parameters.find(cricket::kCodecParamAssociatedCodecName);
>+  if (it == format.parameters.end()) {
>+    RTC_LOG(LS_ERROR) << "No assicated codec for multiplex.";
>+    return nullptr;
>+  }
>+  SdpVideoFormat associated_format = format;
>+  associated_format.name = it->second;
>+  return std::unique_ptr<VideoDecoder>(new MultiplexDecoderAdapter(
>+      factory_.get(), associated_format, supports_augmenting_data_));
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/multiplexcodecfactory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/multiplexcodecfactory.h
>new file mode 100644
>index 00000000000..030904f1b84
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/multiplexcodecfactory.h
>@@ -0,0 +1,61 @@
>+/*
>+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MEDIA_ENGINE_MULTIPLEXCODECFACTORY_H_
>+#define MEDIA_ENGINE_MULTIPLEXCODECFACTORY_H_
>+
>+#include <memory>
>+#include <vector>
>+
>+#include "api/video_codecs/video_decoder_factory.h"
>+#include "api/video_codecs/video_encoder_factory.h"
>+
>+namespace webrtc {
>+
>+class MultiplexEncoderFactory : public VideoEncoderFactory {
>+ public:
>+  // supports_augmenting_data defines if the encoder would support augmenting
>+  // data in that case the encoder expects video frame buffer of type
>+  // AugmentedVideoFrameBuffer the encoder would encode the attached buffer and
>+  // data together if the flag is not set any frame buffer can be passed in
>+  MultiplexEncoderFactory(std::unique_ptr<VideoEncoderFactory> factory,
>+                          bool supports_augmenting_data = false);
>+
>+  std::vector<SdpVideoFormat> GetSupportedFormats() const override;
>+  CodecInfo QueryVideoEncoder(const SdpVideoFormat& format) const override;
>+  std::unique_ptr<VideoEncoder> CreateVideoEncoder(
>+      const SdpVideoFormat& format) override;
>+
>+ private:
>+  std::unique_ptr<VideoEncoderFactory> factory_;
>+  const bool supports_augmenting_data_;
>+};
>+
>+class MultiplexDecoderFactory : public VideoDecoderFactory {
>+ public:
>+  // supports_augmenting_data defines if the decoder would support augmenting
>+  // data in that case the decoder expects the encoded video frame to contain
>+  // augmenting_data it is expected that the sender is using MultiplexEncoder
>+  // with supports_augmenting_data set
>+  MultiplexDecoderFactory(std::unique_ptr<VideoDecoderFactory> factory,
>+                          bool supports_augmenting_data = false);
>+
>+  std::vector<SdpVideoFormat> GetSupportedFormats() const override;
>+  std::unique_ptr<VideoDecoder> CreateVideoDecoder(
>+      const SdpVideoFormat& format) override;
>+
>+ private:
>+  std::unique_ptr<VideoDecoderFactory> factory_;
>+  const bool supports_augmenting_data_;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MEDIA_ENGINE_MULTIPLEXCODECFACTORY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/stereocodecfactory_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/multiplexcodecfactory_unittest.cc
>similarity index 79%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/stereocodecfactory_unittest.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/multiplexcodecfactory_unittest.cc
>index a86eff04aca..bc30f60c6d4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/stereocodecfactory_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/multiplexcodecfactory_unittest.cc
>@@ -8,7 +8,7 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "media/engine/stereocodecfactory.h"
>+#include "media/engine/multiplexcodecfactory.h"
> 
> #include <utility>
> 
>@@ -22,24 +22,24 @@
> 
> namespace webrtc {
> 
>-TEST(StereoDecoderFactory, CreateVideoDecoder) {
>+TEST(MultiplexDecoderFactory, CreateVideoDecoder) {
>   std::unique_ptr<VideoDecoderFactory> internal_factory(
>       new InternalDecoderFactory());
>-  StereoDecoderFactory factory(std::move(internal_factory));
>+  MultiplexDecoderFactory factory(std::move(internal_factory));
>   std::unique_ptr<VideoDecoder> decoder =
>       factory.CreateVideoDecoder(SdpVideoFormat(
>-          cricket::kStereoCodecName,
>+          cricket::kMultiplexCodecName,
>           {{cricket::kCodecParamAssociatedCodecName, cricket::kVp9CodecName}}));
>   EXPECT_TRUE(decoder);
> }
> 
>-TEST(StereoEncoderFactory, CreateVideoEncoder) {
>+TEST(MultiplexEncoderFactory, CreateVideoEncoder) {
>   std::unique_ptr<VideoEncoderFactory> internal_factory(
>       new InternalEncoderFactory());
>-  StereoEncoderFactory factory(std::move(internal_factory));
>+  MultiplexEncoderFactory factory(std::move(internal_factory));
>   std::unique_ptr<VideoEncoder> encoder =
>       factory.CreateVideoEncoder(SdpVideoFormat(
>-          cricket::kStereoCodecName,
>+          cricket::kMultiplexCodecName,
>           {{cricket::kCodecParamAssociatedCodecName, cricket::kVp9CodecName}}));
>   EXPECT_TRUE(encoder);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/nullwebrtcvideoengine.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/nullwebrtcvideoengine.h
>index 0ff997d7ad3..9af0f9b130c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/nullwebrtcvideoengine.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/nullwebrtcvideoengine.h
>@@ -22,7 +22,6 @@ class Call;
> 
> }  // namespace webrtc
> 
>-
> namespace cricket {
> 
> class VideoMediaChannel;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/payload_type_mapper.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/payload_type_mapper.cc
>index 1927f43a00e..5d2fb6757c4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/payload_type_mapper.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/payload_type_mapper.cc
>@@ -30,52 +30,55 @@ PayloadTypeMapper::PayloadTypeMapper()
>     // mapping within the exclusive range.
>     : next_unused_payload_type_(96),
>       max_payload_type_(127),
>-      mappings_({
>-          // Static payload type assignments according to RFC 3551.
>-          {{"PCMU",   8000, 1}, 0},
>-          {{"GSM",    8000, 1}, 3},
>-          {{"G723",   8000, 1}, 4},
>-          {{"DVI4",   8000, 1}, 5},
>-          {{"DVI4",  16000, 1}, 6},
>-          {{"LPC",    8000, 1}, 7},
>-          {{"PCMA",   8000, 1}, 8},
>-          {{"G722",   8000, 1}, 9},
>-          {{"L16",   44100, 2}, 10},
>-          {{"L16",   44100, 1}, 11},
>-          {{"QCELP",  8000, 1}, 12},
>-          {{"CN",     8000, 1}, 13},
>-          // RFC 4566 is a bit ambiguous on the contents of the "encoding
>-          // parameters" field, which, for audio, encodes the number of
>-          // channels. It is "optional and may be omitted if the number of
>-          // channels is one". Does that necessarily imply that an omitted
>-          // encoding parameter means one channel?  Since RFC 3551 doesn't
>-          // specify a value for this parameter for MPA, I've included both 0
>-          // and 1 here, to increase the chances it will be correctly used if
>-          // someone implements an MPEG audio encoder/decoder.
>-          {{"MPA",   90000, 0}, 14},
>-          {{"MPA",   90000, 1}, 14},
>-          {{"G728",   8000, 1}, 15},
>-          {{"DVI4",  11025, 1}, 16},
>-          {{"DVI4",  22050, 1}, 17},
>-          {{"G729",   8000, 1}, 18},
>-
>-          // Payload type assignments currently used by WebRTC.
>-          // Includes data to reduce collisions (and thus reassignments)
>-          {{kGoogleRtpDataCodecName, 0, 0}, kGoogleRtpDataCodecPlType},
>-          {{kIlbcCodecName,    8000, 1}, 102},
>-          {{kIsacCodecName,   16000, 1}, 103},
>-          {{kIsacCodecName,   32000, 1}, 104},
>-          {{kCnCodecName,     16000, 1}, 105},
>-          {{kCnCodecName,     32000, 1}, 106},
>-          {{kGoogleSctpDataCodecName, 0, 0}, kGoogleSctpDataCodecPlType},
>-          {{kOpusCodecName,   48000, 2,
>-              {{"minptime", "10"}, {"useinbandfec", "1"}}}, 111},
>-          // TODO(solenberg): Remove the hard coded 16k,32k,48k DTMF once we
>-          // assign payload types dynamically for send side as well.
>-          {{kDtmfCodecName,   48000, 1}, 110},
>-          {{kDtmfCodecName,   32000, 1}, 112},
>-          {{kDtmfCodecName,   16000, 1}, 113},
>-          {{kDtmfCodecName,    8000, 1}, 126}}) {
>+      mappings_(
>+          {// Static payload type assignments according to RFC 3551.
>+           {{"PCMU", 8000, 1}, 0},
>+           {{"GSM", 8000, 1}, 3},
>+           {{"G723", 8000, 1}, 4},
>+           {{"DVI4", 8000, 1}, 5},
>+           {{"DVI4", 16000, 1}, 6},
>+           {{"LPC", 8000, 1}, 7},
>+           {{"PCMA", 8000, 1}, 8},
>+           {{"G722", 8000, 1}, 9},
>+           {{"L16", 44100, 2}, 10},
>+           {{"L16", 44100, 1}, 11},
>+           {{"QCELP", 8000, 1}, 12},
>+           {{"CN", 8000, 1}, 13},
>+           // RFC 4566 is a bit ambiguous on the contents of the "encoding
>+           // parameters" field, which, for audio, encodes the number of
>+           // channels. It is "optional and may be omitted if the number of
>+           // channels is one". Does that necessarily imply that an omitted
>+           // encoding parameter means one channel?  Since RFC 3551 doesn't
>+           // specify a value for this parameter for MPA, I've included both 0
>+           // and 1 here, to increase the chances it will be correctly used if
>+           // someone implements an MPEG audio encoder/decoder.
>+           {{"MPA", 90000, 0}, 14},
>+           {{"MPA", 90000, 1}, 14},
>+           {{"G728", 8000, 1}, 15},
>+           {{"DVI4", 11025, 1}, 16},
>+           {{"DVI4", 22050, 1}, 17},
>+           {{"G729", 8000, 1}, 18},
>+
>+           // Payload type assignments currently used by WebRTC.
>+           // Includes data to reduce collisions (and thus reassignments)
>+           {{kGoogleRtpDataCodecName, 0, 0}, kGoogleRtpDataCodecPlType},
>+           {{kIlbcCodecName, 8000, 1}, 102},
>+           {{kIsacCodecName, 16000, 1}, 103},
>+           {{kIsacCodecName, 32000, 1}, 104},
>+           {{kCnCodecName, 16000, 1}, 105},
>+           {{kCnCodecName, 32000, 1}, 106},
>+           {{kGoogleSctpDataCodecName, 0, 0}, kGoogleSctpDataCodecPlType},
>+           {{kOpusCodecName,
>+             48000,
>+             2,
>+             {{"minptime", "10"}, {"useinbandfec", "1"}}},
>+            111},
>+           // TODO(solenberg): Remove the hard coded 16k,32k,48k DTMF once we
>+           // assign payload types dynamically for send side as well.
>+           {{kDtmfCodecName, 48000, 1}, 110},
>+           {{kDtmfCodecName, 32000, 1}, 112},
>+           {{kDtmfCodecName, 16000, 1}, 113},
>+           {{kDtmfCodecName, 8000, 1}, 126}}) {
>   // TODO(ossu): Try to keep this as change-proof as possible until we're able
>   // to remove the payload type constants from everywhere in the code.
>   for (const auto& mapping : mappings_) {
>@@ -85,7 +88,7 @@ PayloadTypeMapper::PayloadTypeMapper()
> 
> PayloadTypeMapper::~PayloadTypeMapper() = default;
> 
>-rtc::Optional<int> PayloadTypeMapper::GetMappingFor(
>+absl::optional<int> PayloadTypeMapper::GetMappingFor(
>     const webrtc::SdpAudioFormat& format) {
>   auto iter = mappings_.find(format);
>   if (iter != mappings_.end())
>@@ -102,19 +105,19 @@ rtc::Optional<int> PayloadTypeMapper::GetMappingFor(
>     }
>   }
> 
>-  return rtc::nullopt;
>+  return absl::nullopt;
> }
> 
>-rtc::Optional<int> PayloadTypeMapper::FindMappingFor(
>+absl::optional<int> PayloadTypeMapper::FindMappingFor(
>     const webrtc::SdpAudioFormat& format) const {
>   auto iter = mappings_.find(format);
>   if (iter != mappings_.end())
>     return iter->second;
> 
>-  return rtc::nullopt;
>+  return absl::nullopt;
> }
> 
>-rtc::Optional<AudioCodec> PayloadTypeMapper::ToAudioCodec(
>+absl::optional<AudioCodec> PayloadTypeMapper::ToAudioCodec(
>     const webrtc::SdpAudioFormat& format) {
>   // TODO(ossu): We can safely set bitrate to zero here, since that field is
>   // not presented in the SDP. It is used to ferry around some target bitrate
>@@ -129,7 +132,7 @@ rtc::Optional<AudioCodec> PayloadTypeMapper::ToAudioCodec(
>     return std::move(codec);
>   }
> 
>-  return rtc::nullopt;
>+  return absl::nullopt;
> }
> 
> bool PayloadTypeMapper::SdpAudioFormatOrdering::operator()(
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/payload_type_mapper.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/payload_type_mapper.h
>index 914c08c97ba..d8ab4a4261f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/payload_type_mapper.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/payload_type_mapper.h
>@@ -14,8 +14,8 @@
> #include <map>
> #include <set>
> 
>+#include "absl/types/optional.h"
> #include "api/audio_codecs/audio_format.h"
>-#include "api/optional.h"
> #include "media/base/codec.h"
> 
> namespace cricket {
>@@ -30,15 +30,16 @@ class PayloadTypeMapper {
>   // Finds the current payload type for |format| or assigns a new one, if no
>   // current mapping exists. Will return an empty value if it was unable to
>   // create a mapping, i.e. if all dynamic payload type ids have been used up.
>-  rtc::Optional<int> GetMappingFor(const webrtc::SdpAudioFormat& format);
>+  absl::optional<int> GetMappingFor(const webrtc::SdpAudioFormat& format);
> 
>   // Finds the current payload type for |format|, if any. Returns an empty value
>   // if no payload type mapping exists for the format.
>-  rtc::Optional<int> FindMappingFor(const webrtc::SdpAudioFormat& format) const;
>+  absl::optional<int> FindMappingFor(
>+      const webrtc::SdpAudioFormat& format) const;
> 
>   // Like GetMappingFor, but fills in an AudioCodec structure with the necessary
>   // information instead.
>-  rtc::Optional<AudioCodec> ToAudioCodec(const webrtc::SdpAudioFormat& format);
>+  absl::optional<AudioCodec> ToAudioCodec(const webrtc::SdpAudioFormat& format);
> 
>  private:
>   struct SdpAudioFormatOrdering {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/payload_type_mapper_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/payload_type_mapper_unittest.cc
>index 96d56c28d2f..979ce61dccc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/payload_type_mapper_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/payload_type_mapper_unittest.cc
>@@ -22,42 +22,45 @@ class PayloadTypeMapperTest : public testing::Test {
> };
> 
> TEST_F(PayloadTypeMapperTest, StaticPayloadTypes) {
>-  EXPECT_EQ(0,  mapper_.FindMappingFor({"pcmu",   8000, 1}));
>-  EXPECT_EQ(3,  mapper_.FindMappingFor({"gsm",    8000, 1}));
>-  EXPECT_EQ(4,  mapper_.FindMappingFor({"g723",   8000, 1}));
>-  EXPECT_EQ(5,  mapper_.FindMappingFor({"dvi4",   8000, 1}));
>-  EXPECT_EQ(6,  mapper_.FindMappingFor({"dvi4",  16000, 1}));
>-  EXPECT_EQ(7,  mapper_.FindMappingFor({"lpc",    8000, 1}));
>-  EXPECT_EQ(8,  mapper_.FindMappingFor({"pcma",   8000, 1}));
>-  EXPECT_EQ(9,  mapper_.FindMappingFor({"g722",   8000, 1}));
>-  EXPECT_EQ(10, mapper_.FindMappingFor({"l16",   44100, 2}));
>-  EXPECT_EQ(11, mapper_.FindMappingFor({"l16",   44100, 1}));
>-  EXPECT_EQ(12, mapper_.FindMappingFor({"qcelp",  8000, 1}));
>-  EXPECT_EQ(13, mapper_.FindMappingFor({"cn",     8000, 1}));
>-  EXPECT_EQ(14, mapper_.FindMappingFor({"mpa",   90000, 0}));
>-  EXPECT_EQ(14, mapper_.FindMappingFor({"mpa",   90000, 1}));
>-  EXPECT_EQ(15, mapper_.FindMappingFor({"g728",   8000, 1}));
>-  EXPECT_EQ(16, mapper_.FindMappingFor({"dvi4",  11025, 1}));
>-  EXPECT_EQ(17, mapper_.FindMappingFor({"dvi4",  22050, 1}));
>-  EXPECT_EQ(18, mapper_.FindMappingFor({"g729",   8000, 1}));
>+  EXPECT_EQ(0, mapper_.FindMappingFor({"pcmu", 8000, 1}));
>+  EXPECT_EQ(3, mapper_.FindMappingFor({"gsm", 8000, 1}));
>+  EXPECT_EQ(4, mapper_.FindMappingFor({"g723", 8000, 1}));
>+  EXPECT_EQ(5, mapper_.FindMappingFor({"dvi4", 8000, 1}));
>+  EXPECT_EQ(6, mapper_.FindMappingFor({"dvi4", 16000, 1}));
>+  EXPECT_EQ(7, mapper_.FindMappingFor({"lpc", 8000, 1}));
>+  EXPECT_EQ(8, mapper_.FindMappingFor({"pcma", 8000, 1}));
>+  EXPECT_EQ(9, mapper_.FindMappingFor({"g722", 8000, 1}));
>+  EXPECT_EQ(10, mapper_.FindMappingFor({"l16", 44100, 2}));
>+  EXPECT_EQ(11, mapper_.FindMappingFor({"l16", 44100, 1}));
>+  EXPECT_EQ(12, mapper_.FindMappingFor({"qcelp", 8000, 1}));
>+  EXPECT_EQ(13, mapper_.FindMappingFor({"cn", 8000, 1}));
>+  EXPECT_EQ(14, mapper_.FindMappingFor({"mpa", 90000, 0}));
>+  EXPECT_EQ(14, mapper_.FindMappingFor({"mpa", 90000, 1}));
>+  EXPECT_EQ(15, mapper_.FindMappingFor({"g728", 8000, 1}));
>+  EXPECT_EQ(16, mapper_.FindMappingFor({"dvi4", 11025, 1}));
>+  EXPECT_EQ(17, mapper_.FindMappingFor({"dvi4", 22050, 1}));
>+  EXPECT_EQ(18, mapper_.FindMappingFor({"g729", 8000, 1}));
> }
> 
> TEST_F(PayloadTypeMapperTest, WebRTCPayloadTypes) {
>   // Tests that the payload mapper knows about the audio and data formats we've
>   // been using in WebRTC, with their hard coded values.
>-  auto data_mapping = [this] (const char *name) {
>+  auto data_mapping = [this](const char* name) {
>     return mapper_.FindMappingFor({name, 0, 0});
>   };
>   EXPECT_EQ(kGoogleRtpDataCodecPlType, data_mapping(kGoogleRtpDataCodecName));
>   EXPECT_EQ(kGoogleSctpDataCodecPlType, data_mapping(kGoogleSctpDataCodecName));
> 
>-  EXPECT_EQ(102, mapper_.FindMappingFor({kIlbcCodecName,  8000, 1}));
>+  EXPECT_EQ(102, mapper_.FindMappingFor({kIlbcCodecName, 8000, 1}));
>   EXPECT_EQ(103, mapper_.FindMappingFor({kIsacCodecName, 16000, 1}));
>   EXPECT_EQ(104, mapper_.FindMappingFor({kIsacCodecName, 32000, 1}));
>-  EXPECT_EQ(105, mapper_.FindMappingFor({kCnCodecName,   16000, 1}));
>-  EXPECT_EQ(106, mapper_.FindMappingFor({kCnCodecName,   32000, 1}));
>-  EXPECT_EQ(111, mapper_.FindMappingFor({kOpusCodecName, 48000, 2,
>-        {{"minptime", "10"}, {"useinbandfec", "1"}}}));
>+  EXPECT_EQ(105, mapper_.FindMappingFor({kCnCodecName, 16000, 1}));
>+  EXPECT_EQ(106, mapper_.FindMappingFor({kCnCodecName, 32000, 1}));
>+  EXPECT_EQ(111, mapper_.FindMappingFor(
>+                     {kOpusCodecName,
>+                      48000,
>+                      2,
>+                      {{"minptime", "10"}, {"useinbandfec", "1"}}}));
>   // TODO(solenberg): Remove 16k, 32k, 48k DTMF checks once these payload types
>   // are dynamically assigned.
>   EXPECT_EQ(110, mapper_.FindMappingFor({kDtmfCodecName, 48000, 1}));
>@@ -83,7 +86,7 @@ TEST_F(PayloadTypeMapperTest, ValidDynamicPayloadTypes) {
>   std::set<int> used_payload_types;
>   for (int i = 0; i != 256; ++i) {
>     std::string format_name = "unknown_format_" + std::to_string(i);
>-    webrtc::SdpAudioFormat format(format_name.c_str(), i*100, (i % 2) + 1);
>+    webrtc::SdpAudioFormat format(format_name.c_str(), i * 100, (i % 2) + 1);
>     auto opt_payload_type = mapper_.GetMappingFor(format);
>     bool mapper_is_full = false;
> 
>@@ -124,7 +127,7 @@ TEST_F(PayloadTypeMapperTest, ToAudioCodec) {
>   EXPECT_TRUE(opt_audio_codec);
> 
>   if (opt_payload_type && opt_audio_codec) {
>-    int payload_type        = *opt_payload_type;
>+    int payload_type = *opt_payload_type;
>     const AudioCodec& codec = *opt_audio_codec;
> 
>     EXPECT_EQ(codec.id, payload_type);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/scopedvideodecoder.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/scopedvideodecoder.cc
>index a9f85378ab5..733132b0801 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/scopedvideodecoder.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/scopedvideodecoder.cc
>@@ -30,7 +30,6 @@ class ScopedVideoDecoder : public webrtc::VideoDecoder {
>   int32_t Release() override;
>   int32_t Decode(const webrtc::EncodedImage& input_image,
>                  bool missing_frames,
>-                 const webrtc::RTPFragmentationHeader* fragmentation,
>                  const webrtc::CodecSpecificInfo* codec_specific_info,
>                  int64_t render_time_ms) override;
>   bool PrefersLateDecoding() const override;
>@@ -64,11 +63,10 @@ int32_t ScopedVideoDecoder::Release() {
> int32_t ScopedVideoDecoder::Decode(
>     const webrtc::EncodedImage& input_image,
>     bool missing_frames,
>-    const webrtc::RTPFragmentationHeader* fragmentation,
>     const webrtc::CodecSpecificInfo* codec_specific_info,
>     int64_t render_time_ms) {
>-  return decoder_->Decode(input_image, missing_frames, fragmentation,
>-                          codec_specific_info, render_time_ms);
>+  return decoder_->Decode(input_image, missing_frames, codec_specific_info,
>+                          render_time_ms);
> }
> 
> bool ScopedVideoDecoder::PrefersLateDecoding() const {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/scopedvideoencoder.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/scopedvideoencoder.cc
>index 0f563641f63..026654e0720 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/scopedvideoencoder.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/scopedvideoencoder.cc
>@@ -12,6 +12,7 @@
> 
> #include <vector>
> 
>+#include "api/video/video_bitrate_allocation.h"
> #include "api/video_codecs/video_encoder.h"
> 
> namespace cricket {
>@@ -34,10 +35,9 @@ class ScopedVideoEncoder : public webrtc::VideoEncoder {
>                  const std::vector<webrtc::FrameType>* frame_types) override;
>   int32_t SetChannelParameters(uint32_t packet_loss, int64_t rtt) override;
>   int32_t SetRates(uint32_t bitrate, uint32_t framerate) override;
>-  int32_t SetRateAllocation(const webrtc::BitrateAllocation& allocation,
>+  int32_t SetRateAllocation(const webrtc::VideoBitrateAllocation& allocation,
>                             uint32_t framerate) override;
>   ScalingSettings GetScalingSettings() const override;
>-  int32_t SetPeriodicKeyFrames(bool enable) override;
>   bool SupportsNativeHandle() const override;
>   const char* ImplementationName() const override;
> 
>@@ -85,7 +85,7 @@ int32_t ScopedVideoEncoder::SetRates(uint32_t bitrate, uint32_t framerate) {
> }
> 
> int32_t ScopedVideoEncoder::SetRateAllocation(
>-    const webrtc::BitrateAllocation& allocation,
>+    const webrtc::VideoBitrateAllocation& allocation,
>     uint32_t framerate) {
>   return encoder_->SetRateAllocation(allocation, framerate);
> }
>@@ -95,10 +95,6 @@ webrtc::VideoEncoder::ScalingSettings ScopedVideoEncoder::GetScalingSettings()
>   return encoder_->GetScalingSettings();
> }
> 
>-int32_t ScopedVideoEncoder::SetPeriodicKeyFrames(bool enable) {
>-  return encoder_->SetPeriodicKeyFrames(enable);
>-}
>-
> bool ScopedVideoEncoder::SupportsNativeHandle() const {
>   return encoder_->SupportsNativeHandle();
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/simulcast.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/simulcast.cc
>index 6d68453f7b6..dc05081acea 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/simulcast.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/simulcast.cc
>@@ -15,12 +15,26 @@
> #include "media/base/streamparams.h"
> #include "media/engine/constants.h"
> #include "media/engine/simulcast.h"
>+#include "modules/video_coding/utility/simulcast_rate_allocator.h"
> #include "rtc_base/arraysize.h"
> #include "rtc_base/logging.h"
> #include "system_wrappers/include/field_trial.h"
> 
> namespace cricket {
> 
>+namespace {
>+
>+// Limits for legacy conference screensharing mode. Currently used for the
>+// lower of the two simulcast streams.
>+constexpr int kScreenshareDefaultTl0BitrateKbps = 200;
>+constexpr int kScreenshareDefaultTl1BitrateKbps = 1000;
>+
>+// Max bitrate for the higher one of the two simulcast stream used for screen
>+// content.
>+constexpr int kScreenshareHighStreamMaxBitrateBps = 1600000;
>+
>+}  // namespace
>+
> struct SimulcastFormat {
>   int width;
>   int height;
>@@ -41,6 +55,7 @@ struct SimulcastFormat {
> // These tables describe from which resolution we can use how many
> // simulcast layers at what bitrates (maximum, target, and minimum).
> // Important!! Keep this table from high resolution to low resolution.
>+// clang-format off
> const SimulcastFormat kSimulcastFormats[] = {
>   {1920, 1080, 3, 5000, 4000, 800},
>   {1280, 720, 3,  2500, 2500, 600},
>@@ -50,23 +65,34 @@ const SimulcastFormat kSimulcastFormats[] = {
>   {320, 180, 1, 200, 150, 30},
>   {0, 0, 1, 200, 150, 30}
> };
>+// clang-format on
>+
>+const int kMaxScreenshareSimulcastLayers = 2;
> 
>-const int kMaxScreenshareSimulcastStreams = 2;
>-
>-// Multiway: Number of temporal layers for each simulcast stream, for maximum
>-// possible number of simulcast streams |kMaxSimulcastStreams|. The array
>-// goes from lowest resolution at position 0 to highest resolution.
>-// For example, first three elements correspond to say: QVGA, VGA, WHD.
>-static const int
>-    kDefaultConferenceNumberOfTemporalLayers[webrtc::kMaxSimulcastStreams] =
>-    {3, 3, 3, 3};
>-
>-void GetSimulcastSsrcs(const StreamParams& sp, std::vector<uint32_t>* ssrcs) {
>-  const SsrcGroup* sim_group = sp.get_ssrc_group(kSimSsrcGroupSemantics);
>-  if (sim_group) {
>-    ssrcs->insert(
>-        ssrcs->end(), sim_group->ssrcs.begin(), sim_group->ssrcs.end());
>+// Multiway: Number of temporal layers for each simulcast stream.
>+int DefaultNumberOfTemporalLayers(int simulcast_id) {
>+  RTC_CHECK_GE(simulcast_id, 0);
>+  RTC_CHECK_LT(simulcast_id, webrtc::kMaxSimulcastStreams);
>+
>+  const int kDefaultNumTemporalLayers = 3;
>+
>+  const std::string group_name =
>+      webrtc::field_trial::FindFullName("WebRTC-VP8ConferenceTemporalLayers");
>+  if (group_name.empty())
>+    return kDefaultNumTemporalLayers;
>+
>+  int num_temporal_layers = kDefaultNumTemporalLayers;
>+  if (sscanf(group_name.c_str(), "%d", &num_temporal_layers) == 1 &&
>+      num_temporal_layers > 0 &&
>+      num_temporal_layers <= webrtc::kMaxTemporalStreams) {
>+    return num_temporal_layers;
>   }
>+
>+  RTC_LOG(LS_WARNING) << "Attempt to set number of temporal layers to "
>+                         "incorrect value: "
>+                      << group_name;
>+
>+  return kDefaultNumTemporalLayers;
> }
> 
> int FindSimulcastFormatIndex(int width, int height) {
>@@ -98,7 +124,7 @@ int FindSimulcastFormatIndex(int width, int height, size_t max_layers) {
> }
> 
> // Simulcast stream width and height must both be dividable by
>-// |2 ^ simulcast_layers - 1|.
>+// |2 ^ (simulcast_layers - 1)|.
> int NormalizeSimulcastSize(int size, size_t simulcast_layers) {
>   const int base2_exponent = static_cast<int>(simulcast_layers) - 1;
>   return ((size >> base2_exponent) << base2_exponent);
>@@ -132,181 +158,206 @@ void SlotSimulcastMaxResolution(size_t max_layers, int* width, int* height) {
>                    << " height:" << *height;
> }
> 
>-int GetTotalMaxBitrateBps(const std::vector<webrtc::VideoStream>& streams) {
>+void BoostMaxSimulcastLayer(int max_bitrate_bps,
>+                            std::vector<webrtc::VideoStream>* layers) {
>+  if (layers->empty())
>+    return;
>+
>+  // Spend additional bits to boost the max layer.
>+  int bitrate_left_bps = max_bitrate_bps - GetTotalMaxBitrateBps(*layers);
>+  if (bitrate_left_bps > 0) {
>+    layers->back().max_bitrate_bps += bitrate_left_bps;
>+  }
>+}
>+
>+int GetTotalMaxBitrateBps(const std::vector<webrtc::VideoStream>& layers) {
>+  if (layers.empty())
>+    return 0;
>+
>   int total_max_bitrate_bps = 0;
>-  for (size_t s = 0; s < streams.size() - 1; ++s) {
>-    total_max_bitrate_bps += streams[s].target_bitrate_bps;
>+  for (size_t s = 0; s < layers.size() - 1; ++s) {
>+    total_max_bitrate_bps += layers[s].target_bitrate_bps;
>   }
>-  total_max_bitrate_bps += streams.back().max_bitrate_bps;
>+  total_max_bitrate_bps += layers.back().max_bitrate_bps;
>   return total_max_bitrate_bps;
> }
> 
>-std::vector<webrtc::VideoStream> GetSimulcastConfig(size_t max_streams,
>-                                                    int width,
>-                                                    int height,
>-                                                    int max_bitrate_bps,
>-                                                    double bitrate_priority,
>-                                                    int max_qp,
>-                                                    int max_framerate,
>-                                                    bool is_screencast) {
>-  size_t num_simulcast_layers;
>-  if (is_screencast) {
>-    if (UseSimulcastScreenshare()) {
>-      num_simulcast_layers =
>-          std::min<int>(max_streams, kMaxScreenshareSimulcastStreams);
>-    } else {
>-      num_simulcast_layers = 1;
>-    }
>+std::vector<webrtc::VideoStream> GetSimulcastConfig(
>+    size_t max_layers,
>+    int width,
>+    int height,
>+    int /*max_bitrate_bps*/,
>+    double bitrate_priority,
>+    int max_qp,
>+    int max_framerate,
>+    bool is_screenshare,
>+    bool temporal_layers_supported) {
>+  if (is_screenshare) {
>+    return GetScreenshareLayers(max_layers, width, height, bitrate_priority,
>+                                max_qp, max_framerate,
>+                                temporal_layers_supported);
>   } else {
>-    num_simulcast_layers = FindSimulcastMaxLayers(width, height);
>+    return GetNormalSimulcastLayers(max_layers, width, height, bitrate_priority,
>+                                    max_qp, max_framerate,
>+                                    temporal_layers_supported);
>   }
>+}
> 
>-  if (num_simulcast_layers > max_streams) {
>-    // If the number of SSRCs in the group differs from our target
>-    // number of simulcast streams for current resolution, switch down
>-    // to a resolution that matches our number of SSRCs.
>-    SlotSimulcastMaxResolution(max_streams, &width, &height);
>-    num_simulcast_layers = max_streams;
>+std::vector<webrtc::VideoStream> GetNormalSimulcastLayers(
>+    size_t max_layers,
>+    int width,
>+    int height,
>+    double bitrate_priority,
>+    int max_qp,
>+    int max_framerate,
>+    bool temporal_layers_supported) {
>+  // TODO(bugs.webrtc.org/8785): Currently if the resolution isn't large enough
>+  // (defined in kSimulcastFormats) we scale down the number of simulcast
>+  // layers. Consider changing this so that the application can have more
>+  // control over exactly how many simulcast layers are used.
>+  size_t num_simulcast_layers = FindSimulcastMaxLayers(width, height);
>+  if (num_simulcast_layers > max_layers) {
>+    // TODO(bugs.webrtc.org/8486): This scales down the resolution if the
>+    // number of simulcast layers created by the application isn't sufficient
>+    // (defined in kSimulcastFormats). For example if the input frame's
>+    // resolution is HD, but there are only 2 simulcast layers, the
>+    // resolution gets scaled down to VGA. Consider taking this logic out to
>+    // allow the application more control over the resolutions.
>+    SlotSimulcastMaxResolution(max_layers, &width, &height);
>+    num_simulcast_layers = max_layers;
>   }
>-  std::vector<webrtc::VideoStream> streams;
>-  streams.resize(num_simulcast_layers);
>-
>-  if (is_screencast) {
>-    ScreenshareLayerConfig config = ScreenshareLayerConfig::GetDefault();
>-    // For legacy screenshare in conference mode, tl0 and tl1 bitrates are
>-    // piggybacked on the VideoCodec struct as target and max bitrates,
>-    // respectively. See eg. webrtc::VP8EncoderImpl::SetRates().
>-    streams[0].width = width;
>-    streams[0].height = height;
>-    streams[0].max_qp = max_qp;
>-    streams[0].max_framerate = 5;
>-    streams[0].min_bitrate_bps = kMinVideoBitrateBps;
>-    streams[0].target_bitrate_bps = config.tl0_bitrate_kbps * 1000;
>-    streams[0].max_bitrate_bps = config.tl1_bitrate_kbps * 1000;
>-    streams[0].temporal_layer_thresholds_bps.clear();
>-    streams[0].temporal_layer_thresholds_bps.push_back(config.tl0_bitrate_kbps *
>-                                                       1000);
>-
>-    // With simulcast enabled, add another spatial layer. This one will have a
>-    // more normal layout, with the regular 3 temporal layer pattern and no fps
>-    // restrictions. The base simulcast stream will still use legacy setup.
>-    if (num_simulcast_layers == kMaxScreenshareSimulcastStreams) {
>-      // Add optional upper simulcast layer.
>-      // Lowest temporal layers of a 3 layer setup will have 40% of the total
>-      // bitrate allocation for that stream. Make sure the gap between the
>-      // target of the lower stream and first temporal layer of the higher one
>-      // is at most 2x the bitrate, so that upswitching is not hampered by
>-      // stalled bitrate estimates.
>-      int max_bitrate_bps = 2 * ((streams[0].target_bitrate_bps * 10) / 4);
>-      // Cap max bitrate so it isn't overly high for the given resolution.
>-      max_bitrate_bps = std::min<int>(
>-          max_bitrate_bps, FindSimulcastMaxBitrateBps(width, height));
>-
>-      streams[1].width = width;
>-      streams[1].height = height;
>-      streams[1].max_qp = max_qp;
>-      streams[1].max_framerate = max_framerate;
>-      // Three temporal layers means two thresholds.
>-      streams[1].temporal_layer_thresholds_bps.resize(2);
>-      streams[1].min_bitrate_bps = streams[0].target_bitrate_bps * 2;
>-      streams[1].target_bitrate_bps = max_bitrate_bps;
>-      streams[1].max_bitrate_bps = max_bitrate_bps;
>-    }
>-  } else {
>-    // Format width and height has to be divisible by |2 ^ number_streams - 1|.
>-    width = NormalizeSimulcastSize(width, num_simulcast_layers);
>-    height = NormalizeSimulcastSize(height, num_simulcast_layers);
>-
>-    // Add simulcast sub-streams from lower resolution to higher resolutions.
>-    // Add simulcast streams, from highest resolution (|s| = number_streams -1)
>-    // to lowest resolution at |s| = 0.
>-    for (size_t s = num_simulcast_layers - 1;; --s) {
>-      streams[s].width = width;
>-      streams[s].height = height;
>-      // TODO(pbos): Fill actual temporal-layer bitrate thresholds.
>-      streams[s].max_qp = max_qp;
>-      streams[s].temporal_layer_thresholds_bps.resize(
>-          kDefaultConferenceNumberOfTemporalLayers[s] - 1);
>-      streams[s].max_bitrate_bps = FindSimulcastMaxBitrateBps(width, height);
>-      streams[s].target_bitrate_bps =
>-          FindSimulcastTargetBitrateBps(width, height);
>-      streams[s].min_bitrate_bps = FindSimulcastMinBitrateBps(width, height);
>-      streams[s].max_framerate = max_framerate;
>-
>-      width /= 2;
>-      height /= 2;
>-
>-      if (s == 0)
>-        break;
>-    }
>-
>-    // Spend additional bits to boost the max stream.
>-    int bitrate_left_bps = max_bitrate_bps - GetTotalMaxBitrateBps(streams);
>-    if (bitrate_left_bps > 0) {
>-      streams.back().max_bitrate_bps += bitrate_left_bps;
>+  std::vector<webrtc::VideoStream> layers(num_simulcast_layers);
>+
>+  // Format width and height has to be divisible by |2 ^ num_simulcast_layers -
>+  // 1|.
>+  width = NormalizeSimulcastSize(width, num_simulcast_layers);
>+  height = NormalizeSimulcastSize(height, num_simulcast_layers);
>+  // Add simulcast streams, from highest resolution (|s| = num_simulcast_layers
>+  // -1) to lowest resolution at |s| = 0.
>+  for (size_t s = num_simulcast_layers - 1;; --s) {
>+    layers[s].width = width;
>+    layers[s].height = height;
>+    // TODO(pbos): Fill actual temporal-layer bitrate thresholds.
>+    layers[s].max_qp = max_qp;
>+    layers[s].num_temporal_layers =
>+        temporal_layers_supported ? DefaultNumberOfTemporalLayers(s)
>+                                  : 0;
>+    layers[s].max_bitrate_bps = FindSimulcastMaxBitrateBps(width, height);
>+    layers[s].target_bitrate_bps = FindSimulcastTargetBitrateBps(width, height);
>+    int num_temporal_layers = DefaultNumberOfTemporalLayers(s);
>+    if (s == 0) {
>+      // If alternative number temporal layers is selected, adjust the
>+      // bitrate of the lowest simulcast stream so that absolute bitrate for
>+      // the base temporal layer matches the bitrate for the base temporal
>+      // layer with the default 3 simulcast streams. Otherwise we risk a
>+      // higher threshold for receiving a feed at all.
>+      float rate_factor = 1.0;
>+      if (num_temporal_layers == 3) {
>+        if (webrtc::field_trial::IsEnabled("WebRTC-UseShortVP8TL3Pattern")) {
>+          // Shortened pattern increases TL0 bitrate from 40% to 60%.
>+          rate_factor = 0.4 / 0.6;
>+        }
>+      } else {
>+        rate_factor =
>+            webrtc::SimulcastRateAllocator::GetTemporalRateAllocation(3, 0) /
>+            webrtc::SimulcastRateAllocator::GetTemporalRateAllocation(
>+                num_temporal_layers, 0);
>+      }
>+
>+      layers[s].max_bitrate_bps =
>+          static_cast<int>(layers[s].max_bitrate_bps * rate_factor);
>+      layers[s].target_bitrate_bps =
>+          static_cast<int>(layers[s].target_bitrate_bps * rate_factor);
>     }
>-  }
>-
>-  // The bitrate priority currently implemented on a per-sender level, so we
>-  // just set it for the first video stream.
>-  streams[0].bitrate_priority = bitrate_priority;
>-  return streams;
>-}
>-
>-static const int kScreenshareMinBitrateKbps = 50;
>-static const int kScreenshareMaxBitrateKbps = 6000;
>-static const int kScreenshareDefaultTl0BitrateKbps = 200;
>-static const int kScreenshareDefaultTl1BitrateKbps = 1000;
>-
>-static const char* kScreencastLayerFieldTrialName =
>-    "WebRTC-ScreenshareLayerRates";
>-static const char* kSimulcastScreenshareFieldTrialName =
>-    "WebRTC-SimulcastScreenshare";
>-
>-ScreenshareLayerConfig::ScreenshareLayerConfig(int tl0_bitrate, int tl1_bitrate)
>-    : tl0_bitrate_kbps(tl0_bitrate), tl1_bitrate_kbps(tl1_bitrate) {
>-}
>+    layers[s].min_bitrate_bps = FindSimulcastMinBitrateBps(width, height);
>+    layers[s].max_framerate = max_framerate;
> 
>-ScreenshareLayerConfig ScreenshareLayerConfig::GetDefault() {
>-  std::string group =
>-      webrtc::field_trial::FindFullName(kScreencastLayerFieldTrialName);
>+    width /= 2;
>+    height /= 2;
> 
>-  ScreenshareLayerConfig config(kScreenshareDefaultTl0BitrateKbps,
>-                                kScreenshareDefaultTl1BitrateKbps);
>-  if (!group.empty() && !FromFieldTrialGroup(group, &config)) {
>-    RTC_LOG(LS_WARNING) << "Unable to parse WebRTC-ScreenshareLayerRates"
>-                           " field trial group: '"
>-                        << group << "'.";
>+    if (s == 0) {
>+      break;
>+    }
>   }
>-  return config;
>+  // Currently the relative bitrate priority of the sender is controlled by
>+  // the value of the lowest VideoStream.
>+  // TODO(bugs.webrtc.org/8630): The web specification describes being able to
>+  // control relative bitrate for each individual simulcast layer, but this
>+  // is currently just implemented per rtp sender.
>+  layers[0].bitrate_priority = bitrate_priority;
>+  return layers;
> }
> 
>-bool ScreenshareLayerConfig::FromFieldTrialGroup(
>-    const std::string& group,
>-    ScreenshareLayerConfig* config) {
>-  // Parse field trial group name, containing bitrates for tl0 and tl1.
>-  int tl0_bitrate;
>-  int tl1_bitrate;
>-  if (sscanf(group.c_str(), "%d-%d", &tl0_bitrate, &tl1_bitrate) != 2) {
>-    return false;
>-  }
>+std::vector<webrtc::VideoStream> GetScreenshareLayers(
>+    size_t max_layers,
>+    int width,
>+    int height,
>+    double bitrate_priority,
>+    int max_qp,
>+    int max_framerate,
>+    bool temporal_layers_supported) {
>+  size_t num_simulcast_layers =
>+      std::min<int>(max_layers, kMaxScreenshareSimulcastLayers);
>+
>+  std::vector<webrtc::VideoStream> layers(num_simulcast_layers);
>+  // For legacy screenshare in conference mode, tl0 and tl1 bitrates are
>+  // piggybacked on the VideoCodec struct as target and max bitrates,
>+  // respectively. See eg. webrtc::LibvpxVp8Encoder::SetRates().
>+  layers[0].width = width;
>+  layers[0].height = height;
>+  layers[0].max_qp = max_qp;
>+  layers[0].max_framerate = 5;
>+  layers[0].min_bitrate_bps = kMinVideoBitrateBps;
>+  layers[0].target_bitrate_bps = kScreenshareDefaultTl0BitrateKbps * 1000;
>+  layers[0].max_bitrate_bps = kScreenshareDefaultTl1BitrateKbps * 1000;
>+  layers[0].num_temporal_layers = temporal_layers_supported ? 2 : 0;
>+
>+  // With simulcast enabled, add another spatial layer. This one will have a
>+  // more normal layout, with the regular 3 temporal layer pattern and no fps
>+  // restrictions. The base simulcast layer will still use legacy setup.
>+  if (num_simulcast_layers == kMaxScreenshareSimulcastLayers) {
>+    // Add optional upper simulcast layer.
>+    const int num_temporal_layers = DefaultNumberOfTemporalLayers(1);
>+    int max_bitrate_bps;
>+    if (!temporal_layers_supported) {
>+      // Set the max bitrate to where the base layer would have been if temporal
>+      // layer were enabled.
>+      max_bitrate_bps = static_cast<int>(
>+          kScreenshareHighStreamMaxBitrateBps *
>+          webrtc::SimulcastRateAllocator::GetTemporalRateAllocation(
>+              num_temporal_layers, 0));
>+    } else if (DefaultNumberOfTemporalLayers(1) != 3 ||
>+               webrtc::field_trial::IsEnabled("WebRTC-UseShortVP8TL3Pattern")) {
>+      // Experimental temporal layer mode used, use increased max bitrate.
>+      max_bitrate_bps = kScreenshareHighStreamMaxBitrateBps;
>+    } else {
>+      // Keep current bitrates with default 3tl/8 frame settings.
>+      // Lowest temporal layers of a 3 layer setup will have 40% of the total
>+      // bitrate allocation for that simulcast layer. Make sure the gap between
>+      // the target of the lower simulcast layer and first temporal layer of the
>+      // higher one is at most 2x the bitrate, so that upswitching is not
>+      // hampered by stalled bitrate estimates.
>+      max_bitrate_bps = 2 * ((layers[0].target_bitrate_bps * 10) / 4);
>+    }
> 
>-  // Sanity check.
>-  if (tl0_bitrate < kScreenshareMinBitrateKbps ||
>-      tl0_bitrate > kScreenshareMaxBitrateKbps ||
>-      tl1_bitrate < kScreenshareMinBitrateKbps ||
>-      tl1_bitrate > kScreenshareMaxBitrateKbps || tl0_bitrate > tl1_bitrate) {
>-    return false;
>+    // Cap max bitrate so it isn't overly high for the given resolution.
>+    max_bitrate_bps = std::min<int>(max_bitrate_bps,
>+                                    FindSimulcastMaxBitrateBps(width, height));
>+    layers[1].width = width;
>+    layers[1].height = height;
>+    layers[1].max_qp = max_qp;
>+    layers[1].max_framerate = max_framerate;
>+    layers[1].num_temporal_layers =
>+        temporal_layers_supported ? DefaultNumberOfTemporalLayers(1) : 0;
>+    layers[1].min_bitrate_bps = layers[0].target_bitrate_bps * 2;
>+    layers[1].target_bitrate_bps = max_bitrate_bps;
>+    layers[1].max_bitrate_bps = max_bitrate_bps;
>   }
> 
>-  config->tl0_bitrate_kbps = tl0_bitrate;
>-  config->tl1_bitrate_kbps = tl1_bitrate;
>-
>-  return true;
>-}
>-
>-bool UseSimulcastScreenshare() {
>-  return webrtc::field_trial::IsEnabled(kSimulcastScreenshareFieldTrialName);
>+  // The bitrate priority currently implemented on a per-sender level, so we
>+  // just set it for the first simulcast layer.
>+  layers[0].bitrate_priority = bitrate_priority;
>+  return layers;
> }
> 
> }  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/simulcast.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/simulcast.h
>index 31fdcae80b5..84583bca36c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/simulcast.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/simulcast.h
>@@ -11,54 +11,52 @@
> #ifndef MEDIA_ENGINE_SIMULCAST_H_
> #define MEDIA_ENGINE_SIMULCAST_H_
> 
>-#include <string>
> #include <vector>
> 
>-#include "call/video_config.h"
>-#include "rtc_base/basictypes.h"
>+#include "api/video_codecs/video_encoder_config.h"
> 
> namespace cricket {
>-struct StreamParams;
>-
>-// TODO(sprang): Remove this, as we're moving away from temporal layer mode.
>-// Config for use with screen cast when temporal layers are enabled.
>-struct ScreenshareLayerConfig {
>- public:
>-  ScreenshareLayerConfig(int tl0_bitrate, int tl1_bitrate);
>-
>-  // Bitrates, for temporal layers 0 and 1.
>-  int tl0_bitrate_kbps;
>-  int tl1_bitrate_kbps;
>-
>-  static ScreenshareLayerConfig GetDefault();
>-
>-  // Parse bitrate from group name on format "(tl0_bitrate)-(tl1_bitrate)",
>-  // eg. "100-1000" for the default rates.
>-  static bool FromFieldTrialGroup(const std::string& group,
>-                                  ScreenshareLayerConfig* config);
>-};
>-
>-// TODO(pthatcher): Write unit tests just for these functions,
>-// independent of WebrtcVideoEngine.
> 
>+// Gets the total maximum bitrate for the |streams|.
> int GetTotalMaxBitrateBps(const std::vector<webrtc::VideoStream>& streams);
> 
>-// Get the ssrcs of the SIM group from the stream params.
>-void GetSimulcastSsrcs(const StreamParams& sp, std::vector<uint32_t>* ssrcs);
>-
>-// Get simulcast settings.
>-// TODO(sprang): Remove default parameter when it's not longer referenced.
>-std::vector<webrtc::VideoStream> GetSimulcastConfig(size_t max_streams,
>-                                                    int width,
>-                                                    int height,
>-                                                    int max_bitrate_bps,
>-                                                    double bitrate_priority,
>-                                                    int max_qp,
>-                                                    int max_framerate,
>-                                                    bool is_screencast = false);
>-
>-bool UseSimulcastScreenshare();
>-
>+// Adds any bitrate of |max_bitrate_bps| that is above the total maximum bitrate
>+// for the |layers| to the highest quality layer.
>+void BoostMaxSimulcastLayer(int max_bitrate_bps,
>+                            std::vector<webrtc::VideoStream>* layers);
>+
>+// Gets simulcast settings.
>+// TODO(asapersson): Remove max_bitrate_bps.
>+std::vector<webrtc::VideoStream> GetSimulcastConfig(
>+    size_t max_layers,
>+    int width,
>+    int height,
>+    int /*max_bitrate_bps*/,
>+    double bitrate_priority,
>+    int max_qp,
>+    int max_framerate,
>+    bool is_screenshare,
>+    bool temporal_layers_supported = true);
>+
>+// Gets the simulcast config layers for a non-screensharing case.
>+std::vector<webrtc::VideoStream> GetNormalSimulcastLayers(
>+    size_t max_layers,
>+    int width,
>+    int height,
>+    double bitrate_priority,
>+    int max_qp,
>+    int max_framerate,
>+    bool temporal_layers_supported = true);
>+
>+// Gets simulcast config layers for screenshare settings.
>+std::vector<webrtc::VideoStream> GetScreenshareLayers(
>+    size_t max_layers,
>+    int width,
>+    int height,
>+    double bitrate_priority,
>+    int max_qp,
>+    int max_framerate,
>+    bool temporal_layers_supported = true);
> }  // namespace cricket
> 
> #endif  // MEDIA_ENGINE_SIMULCAST_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/simulcast_encoder_adapter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/simulcast_encoder_adapter.cc
>index ea54e662938..d58a7a7d2a2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/simulcast_encoder_adapter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/simulcast_encoder_adapter.cc
>@@ -12,12 +12,11 @@
> 
> #include <algorithm>
> 
>-
> #include "api/video/i420_buffer.h"
>+#include "api/video/video_bitrate_allocation.h"
> #include "api/video_codecs/video_encoder_factory.h"
> #include "media/engine/scopedvideoencoder.h"
>-#include "modules/video_coding/codecs/vp8/screenshare_layers.h"
>-#include "modules/video_coding/codecs/vp8/simulcast_rate_allocator.h"
>+#include "modules/video_coding/utility/simulcast_rate_allocator.h"
> #include "rtc_base/checks.h"
> #include "system_wrappers/include/clock.h"
> #include "third_party/libyuv/include/libyuv/scale.h"
>@@ -76,7 +75,8 @@ int VerifyCodec(const webrtc::VideoCodec* inst) {
>   if (inst->width <= 1 || inst->height <= 1) {
>     return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
>   }
>-  if (inst->VP8().automaticResizeOn && inst->numberOfSimulcastStreams > 1) {
>+  if (inst->codecType == webrtc::kVideoCodecVP8 &&
>+      inst->VP8().automaticResizeOn && inst->numberOfSimulcastStreams > 1) {
>     return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
>   }
>   return WEBRTC_VIDEO_CODEC_OK;
>@@ -103,57 +103,19 @@ class AdapterEncodedImageCallback : public webrtc::EncodedImageCallback {
>   webrtc::SimulcastEncoderAdapter* const adapter_;
>   const size_t stream_idx_;
> };
>-
>-// Utility class used to adapt the simulcast id as reported by the temporal
>-// layers factory, since each sub-encoder will report stream 0.
>-class TemporalLayersFactoryAdapter : public webrtc::TemporalLayersFactory {
>- public:
>-  TemporalLayersFactoryAdapter(int adapted_simulcast_id,
>-                               const TemporalLayersFactory& tl_factory)
>-      : adapted_simulcast_id_(adapted_simulcast_id), tl_factory_(tl_factory) {}
>-  ~TemporalLayersFactoryAdapter() override {}
>-  webrtc::TemporalLayers* Create(int simulcast_id,
>-                                 int temporal_layers,
>-                                 uint8_t initial_tl0_pic_idx) const override {
>-    return tl_factory_.Create(adapted_simulcast_id_, temporal_layers,
>-                              initial_tl0_pic_idx);
>-  }
>-  std::unique_ptr<webrtc::TemporalLayersChecker> CreateChecker(
>-      int simulcast_id,
>-      int temporal_layers,
>-      uint8_t initial_tl0_pic_idx) const override {
>-    return tl_factory_.CreateChecker(adapted_simulcast_id_, temporal_layers,
>-                                     initial_tl0_pic_idx);
>-  }
>-
>-  const int adapted_simulcast_id_;
>-  const TemporalLayersFactory& tl_factory_;
>-};
>-
> }  // namespace
> 
> namespace webrtc {
> 
>-SimulcastEncoderAdapter::SimulcastEncoderAdapter(VideoEncoderFactory* factory)
>+SimulcastEncoderAdapter::SimulcastEncoderAdapter(VideoEncoderFactory* factory,
>+                                                 const SdpVideoFormat& format)
>     : inited_(0),
>       factory_(factory),
>-      cricket_factory_(nullptr),
>+      video_format_(format),
>       encoded_complete_callback_(nullptr),
>       implementation_name_("SimulcastEncoderAdapter") {
>-  // The adapter is typically created on the worker thread, but operated on
>-  // the encoder task queue.
>-  encoder_queue_.Detach();
>+  RTC_DCHECK(factory_);
> 
>-  memset(&codec_, 0, sizeof(webrtc::VideoCodec));
>-}
>-
>-SimulcastEncoderAdapter::SimulcastEncoderAdapter(
>-    cricket::WebRtcVideoEncoderFactory* factory)
>-    : inited_(0),
>-      factory_(nullptr),
>-      cricket_factory_(factory),
>-      encoded_complete_callback_(nullptr),
>-      implementation_name_("SimulcastEncoderAdapter") {
>   // The adapter is typically created on the worker thread, but operated on
>   // the encoder task queue.
>   encoder_queue_.Detach();
>@@ -217,8 +179,8 @@ int SimulcastEncoderAdapter::InitEncode(const VideoCodec* inst,
>   }
> 
>   codec_ = *inst;
>-  SimulcastRateAllocator rate_allocator(codec_, nullptr);
>-  BitrateAllocation allocation = rate_allocator.GetAllocation(
>+  SimulcastRateAllocator rate_allocator(codec_);
>+  VideoBitrateAllocation allocation = rate_allocator.GetAllocation(
>       codec_.startBitrate * 1000, codec_.maxFramerate);
>   std::vector<uint32_t> start_bitrates;
>   for (int i = 0; i < kMaxSimulcastStreams; ++i) {
>@@ -243,11 +205,8 @@ int SimulcastEncoderAdapter::InitEncode(const VideoCodec* inst,
>       PopulateStreamCodec(codec_, i, start_bitrate_kbps,
>                           highest_resolution_stream, &stream_codec);
>     }
>-    TemporalLayersFactoryAdapter tl_factory_adapter(i,
>-                                                    *codec_.VP8()->tl_factory);
>-    stream_codec.VP8()->tl_factory = &tl_factory_adapter;
> 
>-    // TODO(ronghuawu): Remove once this is handled in VP8EncoderImpl.
>+    // TODO(ronghuawu): Remove once this is handled in LibvpxVp8Encoder.
>     if (stream_codec.qpMax < kDefaultMinQp) {
>       stream_codec.qpMax = kDefaultMaxQp;
>     }
>@@ -260,9 +219,8 @@ int SimulcastEncoderAdapter::InitEncode(const VideoCodec* inst,
>       encoder = std::move(stored_encoders_.top());
>       stored_encoders_.pop();
>     } else {
>-      encoder = factory_ ? factory_->CreateVideoEncoder(SdpVideoFormat("VP8"))
>-                         : CreateScopedVideoEncoder(cricket_factory_,
>-                                                    cricket::VideoCodec("VP8"));
>+      encoder = factory_->CreateVideoEncoder(SdpVideoFormat(
>+          codec_.codecType == webrtc::kVideoCodecVP8 ? "VP8" : "H264"));
>     }
> 
>     ret = encoder->InitEncode(&stream_codec, number_of_cores, max_payload_size);
>@@ -412,8 +370,9 @@ int SimulcastEncoderAdapter::SetChannelParameters(uint32_t packet_loss,
>   return WEBRTC_VIDEO_CODEC_OK;
> }
> 
>-int SimulcastEncoderAdapter::SetRateAllocation(const BitrateAllocation& bitrate,
>-                                               uint32_t new_framerate) {
>+int SimulcastEncoderAdapter::SetRateAllocation(
>+    const VideoBitrateAllocation& bitrate,
>+    uint32_t new_framerate) {
>   RTC_DCHECK_CALLED_SEQUENTIALLY(&encoder_queue_);
> 
>   if (!Initialized()) {
>@@ -455,7 +414,7 @@ int SimulcastEncoderAdapter::SetRateAllocation(const BitrateAllocation& bitrate,
> 
>     // Slice the temporal layers out of the full allocation and pass it on to
>     // the encoder handling the current simulcast stream.
>-    BitrateAllocation stream_allocation;
>+    VideoBitrateAllocation stream_allocation;
>     for (int i = 0; i < kMaxTemporalStreams; ++i) {
>       if (bitrate.HasBitrate(stream_idx, i)) {
>         stream_allocation.SetBitrate(0, i, bitrate.GetBitrate(stream_idx, i));
>@@ -477,8 +436,11 @@ EncodedImageCallback::Result SimulcastEncoderAdapter::OnEncodedImage(
>     const RTPFragmentationHeader* fragmentation) {
>   CodecSpecificInfo stream_codec_specific = *codecSpecificInfo;
>   stream_codec_specific.codec_name = implementation_name_.c_str();
>-  CodecSpecificInfoVP8* vp8Info = &(stream_codec_specific.codecSpecific.VP8);
>-  vp8Info->simulcastIdx = stream_idx;
>+  if (stream_codec_specific.codecType == webrtc::kVideoCodecVP8) {
>+    stream_codec_specific.codecSpecific.VP8.simulcastIdx = stream_idx;
>+  } else if (stream_codec_specific.codecType == webrtc::kVideoCodecH264) {
>+    stream_codec_specific.codecSpecific.H264.simulcast_idx = stream_idx;
>+  }
> 
>   return encoded_complete_callback_->OnEncodedImage(
>       encodedImage, &stream_codec_specific, fragmentation);
>@@ -493,8 +455,6 @@ void SimulcastEncoderAdapter::PopulateStreamCodec(
>   *stream_codec = inst;
> 
>   // Stream specific settings.
>-  stream_codec->VP8()->numberOfTemporalLayers =
>-      inst.simulcastStream[stream_index].numberOfTemporalLayers;
>   stream_codec->numberOfSimulcastStreams = 0;
>   stream_codec->width = inst.simulcastStream[stream_index].width;
>   stream_codec->height = inst.simulcastStream[stream_index].height;
>@@ -503,19 +463,24 @@ void SimulcastEncoderAdapter::PopulateStreamCodec(
>   stream_codec->qpMax = inst.simulcastStream[stream_index].qpMax;
>   // Settings that are based on stream/resolution.
>   const bool lowest_resolution_stream = (stream_index == 0);
>-  if (lowest_resolution_stream) {
>+  if (lowest_resolution_stream && inst.mode != VideoCodecMode::kScreensharing) {
>     // Settings for lowest spatial resolutions.
>     stream_codec->qpMax = kLowestResMaxQp;
>   }
>-  if (!highest_resolution_stream) {
>-    // For resolutions below CIF, set the codec |complexity| parameter to
>-    // kComplexityHigher, which maps to cpu_used = -4.
>-    int pixels_per_frame = stream_codec->width * stream_codec->height;
>-    if (pixels_per_frame < 352 * 288) {
>-      stream_codec->VP8()->complexity = webrtc::kComplexityHigher;
>+  if (inst.codecType == webrtc::kVideoCodecVP8) {
>+    stream_codec->VP8()->numberOfTemporalLayers =
>+        inst.simulcastStream[stream_index].numberOfTemporalLayers;
>+    if (!highest_resolution_stream) {
>+      // For resolutions below CIF, set the codec |complexity| parameter to
>+      // kComplexityHigher, which maps to cpu_used = -4.
>+      int pixels_per_frame = stream_codec->width * stream_codec->height;
>+      if (pixels_per_frame < 352 * 288) {
>+        stream_codec->VP8()->complexity =
>+            webrtc::VideoCodecComplexity::kComplexityHigher;
>+      }
>+      // Turn off denoising for all streams but the highest resolution.
>+      stream_codec->VP8()->denoisingOn = false;
>     }
>-    // Turn off denoising for all streams but the highest resolution.
>-    stream_codec->VP8()->denoisingOn = false;
>   }
>   // TODO(ronghuawu): what to do with targetBitrate.
> 
>@@ -550,7 +515,7 @@ VideoEncoder::ScalingSettings SimulcastEncoderAdapter::GetScalingSettings()
>   // RTC_DCHECK_CALLED_SEQUENTIALLY(&encoder_queue_);
>   // Turn off quality scaling for simulcast.
>   if (!Initialized() || NumberOfStreams(codec_) != 1) {
>-    return VideoEncoder::ScalingSettings(false);
>+    return VideoEncoder::ScalingSettings::kOff;
>   }
>   return streaminfos_[0].encoder->GetScalingSettings();
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/simulcast_encoder_adapter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/simulcast_encoder_adapter.h
>index 9650cd39a8b..af830d7ab80 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/simulcast_encoder_adapter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/simulcast_encoder_adapter.h
>@@ -19,7 +19,7 @@
> #include <vector>
> 
> #include "media/engine/webrtcvideoencoderfactory.h"
>-#include "modules/video_coding/codecs/vp8/include/vp8.h"
>+#include "modules/video_coding/include/video_codec_interface.h"
> #include "rtc_base/atomicops.h"
> #include "rtc_base/sequenced_task_checker.h"
> 
>@@ -32,11 +32,10 @@ class VideoEncoderFactory;
> // webrtc::VideoEncoder instances with the given VideoEncoderFactory.
> // The object is created and destroyed on the worker thread, but all public
> // interfaces should be called from the encoder task queue.
>-class SimulcastEncoderAdapter : public VP8Encoder {
>+class SimulcastEncoderAdapter : public VideoEncoder {
>  public:
>-  explicit SimulcastEncoderAdapter(VideoEncoderFactory* factory);
>-  // Deprecated - use webrtc::VideoEncoderFactory instead.
>-  explicit SimulcastEncoderAdapter(cricket::WebRtcVideoEncoderFactory* factory);
>+  explicit SimulcastEncoderAdapter(VideoEncoderFactory* factory,
>+                                   const SdpVideoFormat& format);
>   virtual ~SimulcastEncoderAdapter();
> 
>   // Implements VideoEncoder.
>@@ -49,7 +48,7 @@ class SimulcastEncoderAdapter : public VP8Encoder {
>              const std::vector<FrameType>* frame_types) override;
>   int RegisterEncodeCompleteCallback(EncodedImageCallback* callback) override;
>   int SetChannelParameters(uint32_t packet_loss, int64_t rtt) override;
>-  int SetRateAllocation(const BitrateAllocation& bitrate,
>+  int SetRateAllocation(const VideoBitrateAllocation& bitrate,
>                         uint32_t new_framerate) override;
> 
>   // Eventual handler for the contained encoders' EncodedImageCallbacks, but
>@@ -100,7 +99,7 @@ class SimulcastEncoderAdapter : public VP8Encoder {
> 
>   volatile int inited_;  // Accessed atomically.
>   VideoEncoderFactory* const factory_;
>-  cricket::WebRtcVideoEncoderFactory* const cricket_factory_;
>+  const SdpVideoFormat video_format_;
>   VideoCodec codec_;
>   std::vector<StreamInfo> streaminfos_;
>   EncodedImageCallback* encoded_complete_callback_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/simulcast_encoder_adapter_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/simulcast_encoder_adapter_unittest.cc
>index 8087d7be0bb..5f1eebe0d03 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/simulcast_encoder_adapter_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/simulcast_encoder_adapter_unittest.cc
>@@ -12,85 +12,136 @@
> #include <memory>
> #include <vector>
> 
>+#include "absl/memory/memory.h"
>+#include "api/test/create_simulcast_test_fixture.h"
>+#include "api/test/simulcast_test_fixture.h"
> #include "api/video_codecs/sdp_video_format.h"
> #include "api/video_codecs/video_encoder_factory.h"
> #include "common_video/include/video_frame_buffer.h"
> #include "media/engine/internalencoderfactory.h"
> #include "media/engine/simulcast_encoder_adapter.h"
>-#include "modules/video_coding/codecs/vp8/simulcast_test_utility.h"
>+#include "modules/video_coding/codecs/vp8/include/vp8.h"
> #include "modules/video_coding/include/video_codec_interface.h"
>-#include "rtc_base/ptr_util.h"
>+#include "modules/video_coding/utility/simulcast_test_fixture_impl.h"
>+#include "test/function_video_decoder_factory.h"
>+#include "test/function_video_encoder_factory.h"
> #include "test/gmock.h"
>+#include "test/gtest.h"
>+
>+using ::testing::_;
>+using ::testing::Return;
> 
> namespace webrtc {
>-namespace testing {
>+namespace test {
> 
>-class TestSimulcastEncoderAdapter : public TestVp8Simulcast {
>- public:
>-  TestSimulcastEncoderAdapter() : factory_(new InternalEncoderFactory()) {}
>+namespace {
> 
>- protected:
>-  std::unique_ptr<VP8Encoder> CreateEncoder() override {
>-    return rtc::MakeUnique<SimulcastEncoderAdapter>(factory_.get());
>-  }
>-  std::unique_ptr<VP8Decoder> CreateDecoder() override {
>-    return VP8Decoder::Create();
>-  }
>+constexpr int kDefaultWidth = 1280;
>+constexpr int kDefaultHeight = 720;
> 
>- private:
>-  std::unique_ptr<VideoEncoderFactory> factory_;
>-};
>+std::unique_ptr<SimulcastTestFixture> CreateSpecificSimulcastTestFixture(
>+    VideoEncoderFactory* internal_encoder_factory) {
>+  std::unique_ptr<VideoEncoderFactory> encoder_factory =
>+      absl::make_unique<FunctionVideoEncoderFactory>(
>+          [internal_encoder_factory]() {
>+            return absl::make_unique<SimulcastEncoderAdapter>(
>+                internal_encoder_factory,
>+                SdpVideoFormat(cricket::kVp8CodecName));
>+          });
>+  std::unique_ptr<VideoDecoderFactory> decoder_factory =
>+      absl::make_unique<FunctionVideoDecoderFactory>(
>+          []() { return VP8Decoder::Create(); });
>+  return CreateSimulcastTestFixture(std::move(encoder_factory),
>+                                    std::move(decoder_factory),
>+                                    SdpVideoFormat(cricket::kVp8CodecName));
>+}
>+
>+}  // namespace
>+
>+TEST(SimulcastEncoderAdapterSimulcastTest, TestKeyFrameRequestsOnAllStreams) {
>+  InternalEncoderFactory internal_encoder_factory;
>+  auto fixture = CreateSpecificSimulcastTestFixture(&internal_encoder_factory);
>+  fixture->TestKeyFrameRequestsOnAllStreams();
>+}
> 
>-TEST_F(TestSimulcastEncoderAdapter, TestKeyFrameRequestsOnAllStreams) {
>-  TestVp8Simulcast::TestKeyFrameRequestsOnAllStreams();
>+TEST(SimulcastEncoderAdapterSimulcastTest, TestPaddingAllStreams) {
>+  InternalEncoderFactory internal_encoder_factory;
>+  auto fixture = CreateSpecificSimulcastTestFixture(&internal_encoder_factory);
>+  fixture->TestPaddingAllStreams();
> }
> 
>-TEST_F(TestSimulcastEncoderAdapter, TestPaddingAllStreams) {
>-  TestVp8Simulcast::TestPaddingAllStreams();
>+TEST(SimulcastEncoderAdapterSimulcastTest, TestPaddingTwoStreams) {
>+  InternalEncoderFactory internal_encoder_factory;
>+  auto fixture = CreateSpecificSimulcastTestFixture(&internal_encoder_factory);
>+  fixture->TestPaddingTwoStreams();
> }
> 
>-TEST_F(TestSimulcastEncoderAdapter, TestPaddingTwoStreams) {
>-  TestVp8Simulcast::TestPaddingTwoStreams();
>+TEST(SimulcastEncoderAdapterSimulcastTest, TestPaddingTwoStreamsOneMaxedOut) {
>+  InternalEncoderFactory internal_encoder_factory;
>+  auto fixture = CreateSpecificSimulcastTestFixture(&internal_encoder_factory);
>+  fixture->TestPaddingTwoStreamsOneMaxedOut();
> }
> 
>-TEST_F(TestSimulcastEncoderAdapter, TestPaddingTwoStreamsOneMaxedOut) {
>-  TestVp8Simulcast::TestPaddingTwoStreamsOneMaxedOut();
>+TEST(SimulcastEncoderAdapterSimulcastTest, TestPaddingOneStream) {
>+  InternalEncoderFactory internal_encoder_factory;
>+  auto fixture = CreateSpecificSimulcastTestFixture(&internal_encoder_factory);
>+  fixture->TestPaddingOneStream();
> }
> 
>-TEST_F(TestSimulcastEncoderAdapter, TestPaddingOneStream) {
>-  TestVp8Simulcast::TestPaddingOneStream();
>+TEST(SimulcastEncoderAdapterSimulcastTest, TestPaddingOneStreamTwoMaxedOut) {
>+  InternalEncoderFactory internal_encoder_factory;
>+  auto fixture = CreateSpecificSimulcastTestFixture(&internal_encoder_factory);
>+  fixture->TestPaddingOneStreamTwoMaxedOut();
> }
> 
>-TEST_F(TestSimulcastEncoderAdapter, TestPaddingOneStreamTwoMaxedOut) {
>-  TestVp8Simulcast::TestPaddingOneStreamTwoMaxedOut();
>+TEST(SimulcastEncoderAdapterSimulcastTest, TestSendAllStreams) {
>+  InternalEncoderFactory internal_encoder_factory;
>+  auto fixture = CreateSpecificSimulcastTestFixture(&internal_encoder_factory);
>+  fixture->TestSendAllStreams();
> }
> 
>-TEST_F(TestSimulcastEncoderAdapter, TestSendAllStreams) {
>-  TestVp8Simulcast::TestSendAllStreams();
>+TEST(SimulcastEncoderAdapterSimulcastTest, TestDisablingStreams) {
>+  InternalEncoderFactory internal_encoder_factory;
>+  auto fixture = CreateSpecificSimulcastTestFixture(&internal_encoder_factory);
>+  fixture->TestDisablingStreams();
> }
> 
>-TEST_F(TestSimulcastEncoderAdapter, TestDisablingStreams) {
>-  TestVp8Simulcast::TestDisablingStreams();
>+TEST(SimulcastEncoderAdapterSimulcastTest, TestActiveStreams) {
>+  InternalEncoderFactory internal_encoder_factory;
>+  auto fixture = CreateSpecificSimulcastTestFixture(&internal_encoder_factory);
>+  fixture->TestActiveStreams();
> }
> 
>-TEST_F(TestSimulcastEncoderAdapter, TestSwitchingToOneStream) {
>-  TestVp8Simulcast::TestSwitchingToOneStream();
>+TEST(SimulcastEncoderAdapterSimulcastTest, TestSwitchingToOneStream) {
>+  InternalEncoderFactory internal_encoder_factory;
>+  auto fixture = CreateSpecificSimulcastTestFixture(&internal_encoder_factory);
>+  fixture->TestSwitchingToOneStream();
> }
> 
>-TEST_F(TestSimulcastEncoderAdapter, TestSwitchingToOneOddStream) {
>-  TestVp8Simulcast::TestSwitchingToOneOddStream();
>+TEST(SimulcastEncoderAdapterSimulcastTest, TestSwitchingToOneOddStream) {
>+  InternalEncoderFactory internal_encoder_factory;
>+  auto fixture = CreateSpecificSimulcastTestFixture(&internal_encoder_factory);
>+  fixture->TestSwitchingToOneOddStream();
> }
> 
>-TEST_F(TestSimulcastEncoderAdapter, TestStrideEncodeDecode) {
>-  TestVp8Simulcast::TestStrideEncodeDecode();
>+TEST(SimulcastEncoderAdapterSimulcastTest, TestStrideEncodeDecode) {
>+  InternalEncoderFactory internal_encoder_factory;
>+  auto fixture = CreateSpecificSimulcastTestFixture(&internal_encoder_factory);
>+  fixture->TestStrideEncodeDecode();
> }
> 
>-TEST_F(TestSimulcastEncoderAdapter, TestSaptioTemporalLayers333PatternEncoder) {
>-  TestVp8Simulcast::TestSaptioTemporalLayers333PatternEncoder();
>+TEST(SimulcastEncoderAdapterSimulcastTest,
>+     TestSpatioTemporalLayers333PatternEncoder) {
>+  InternalEncoderFactory internal_encoder_factory;
>+  auto fixture = CreateSpecificSimulcastTestFixture(&internal_encoder_factory);
>+  fixture->TestSpatioTemporalLayers333PatternEncoder();
> }
> 
>-TEST_F(TestSimulcastEncoderAdapter, TestSpatioTemporalLayers321PatternEncoder) {
>-  TestVp8Simulcast::TestSpatioTemporalLayers321PatternEncoder();
>+TEST(SimulcastEncoderAdapterSimulcastTest,
>+     TestSpatioTemporalLayers321PatternEncoder) {
>+  InternalEncoderFactory internal_encoder_factory;
>+  auto fixture = CreateSpecificSimulcastTestFixture(&internal_encoder_factory);
>+  fixture->TestSpatioTemporalLayers321PatternEncoder();
> }
> 
> class MockVideoEncoder;
>@@ -119,7 +170,7 @@ class MockVideoEncoderFactory : public VideoEncoderFactory {
> class MockVideoEncoder : public VideoEncoder {
>  public:
>   explicit MockVideoEncoder(MockVideoEncoderFactory* factory)
>-      : factory_(factory) {}
>+      : factory_(factory), callback_(nullptr) {}
> 
>   // TODO(nisse): Valid overrides commented out, because the gmock
>   // methods don't use any override declarations, and we want to avoid
>@@ -146,7 +197,7 @@ class MockVideoEncoder : public VideoEncoder {
> 
>   MOCK_METHOD0(Release, int32_t());
> 
>-  int32_t SetRateAllocation(const BitrateAllocation& bitrate_allocation,
>+  int32_t SetRateAllocation(const VideoBitrateAllocation& bitrate_allocation,
>                             uint32_t framerate) {
>     last_set_bitrate_ = bitrate_allocation;
>     return 0;
>@@ -169,6 +220,7 @@ class MockVideoEncoder : public VideoEncoder {
>     image._encodedHeight = height;
>     CodecSpecificInfo codec_specific_info;
>     memset(&codec_specific_info, 0, sizeof(codec_specific_info));
>+    codec_specific_info.codecType = webrtc::kVideoCodecVP8;
>     callback_->OnEncodedImage(image, &codec_specific_info, nullptr);
>   }
> 
>@@ -180,7 +232,7 @@ class MockVideoEncoder : public VideoEncoder {
>     init_encode_return_value_ = value;
>   }
> 
>-  BitrateAllocation last_set_bitrate() const { return last_set_bitrate_; }
>+  VideoBitrateAllocation last_set_bitrate() const { return last_set_bitrate_; }
> 
>   MOCK_CONST_METHOD0(ImplementationName, const char*());
> 
>@@ -188,7 +240,7 @@ class MockVideoEncoder : public VideoEncoder {
>   MockVideoEncoderFactory* const factory_;
>   bool supports_native_handle_ = false;
>   int32_t init_encode_return_value_ = 0;
>-  BitrateAllocation last_set_bitrate_;
>+  VideoBitrateAllocation last_set_bitrate_;
> 
>   VideoCodec codec_;
>   EncodedImageCallback* callback_;
>@@ -246,8 +298,8 @@ class TestSimulcastEncoderAdapterFakeHelper {
> 
>   // Can only be called once as the SimulcastEncoderAdapter will take the
>   // ownership of |factory_|.
>-  VP8Encoder* CreateMockEncoderAdapter() {
>-    return new SimulcastEncoderAdapter(factory_.get());
>+  VideoEncoder* CreateMockEncoderAdapter() {
>+    return new SimulcastEncoderAdapter(factory_.get(), SdpVideoFormat("VP8"));
>   }
> 
>   void ExpectCallSetChannelParameters(uint32_t packetLoss, int64_t rtt) {
>@@ -307,11 +359,10 @@ class TestSimulcastEncoderAdapterFake : public ::testing::Test,
>   }
> 
>   void SetupCodec() {
>-    TestVp8Simulcast::DefaultSettings(
>-        &codec_, static_cast<const int*>(kTestTemporalLayerProfile));
>-    rate_allocator_.reset(new SimulcastRateAllocator(codec_, nullptr));
>-    tl_factory_.SetListener(rate_allocator_.get());
>-    codec_.VP8()->tl_factory = &tl_factory_;
>+    SimulcastTestFixtureImpl::DefaultSettings(
>+        &codec_, static_cast<const int*>(kTestTemporalLayerProfile),
>+        kVideoCodecVP8);
>+    rate_allocator_.reset(new SimulcastRateAllocator(codec_));
>     EXPECT_EQ(0, adapter_->InitEncode(&codec_, 1, 1200));
>     adapter_->RegisterEncodeCompleteCallback(this);
>   }
>@@ -320,7 +371,6 @@ class TestSimulcastEncoderAdapterFake : public ::testing::Test,
>     const VideoCodec& target =
>         helper_->factory()->encoders()[stream_index]->codec();
>     EXPECT_EQ(ref.codecType, target.codecType);
>-    EXPECT_EQ(0, strcmp(ref.plName, target.plName));
>     EXPECT_EQ(ref.plType, target.plType);
>     EXPECT_EQ(ref.width, target.width);
>     EXPECT_EQ(ref.height, target.height);
>@@ -328,14 +378,10 @@ class TestSimulcastEncoderAdapterFake : public ::testing::Test,
>     EXPECT_EQ(ref.maxBitrate, target.maxBitrate);
>     EXPECT_EQ(ref.minBitrate, target.minBitrate);
>     EXPECT_EQ(ref.maxFramerate, target.maxFramerate);
>-    EXPECT_EQ(ref.VP8().pictureLossIndicationOn,
>-              target.VP8().pictureLossIndicationOn);
>     EXPECT_EQ(ref.VP8().complexity, target.VP8().complexity);
>-    EXPECT_EQ(ref.VP8().resilience, target.VP8().resilience);
>     EXPECT_EQ(ref.VP8().numberOfTemporalLayers,
>               target.VP8().numberOfTemporalLayers);
>     EXPECT_EQ(ref.VP8().denoisingOn, target.VP8().denoisingOn);
>-    EXPECT_EQ(ref.VP8().errorConcealmentOn, target.VP8().errorConcealmentOn);
>     EXPECT_EQ(ref.VP8().automaticResizeOn, target.VP8().automaticResizeOn);
>     EXPECT_EQ(ref.VP8().frameDroppingOn, target.VP8().frameDroppingOn);
>     EXPECT_EQ(ref.VP8().keyFrameInterval, target.VP8().keyFrameInterval);
>@@ -351,7 +397,6 @@ class TestSimulcastEncoderAdapterFake : public ::testing::Test,
>     *ref_codec = codec_;
>     ref_codec->VP8()->numberOfTemporalLayers =
>         kTestTemporalLayerProfile[stream_index];
>-    ref_codec->VP8()->tl_factory = &tl_factory_;
>     ref_codec->width = codec_.simulcastStream[stream_index].width;
>     ref_codec->height = codec_.simulcastStream[stream_index].height;
>     ref_codec->maxBitrate = codec_.simulcastStream[stream_index].maxBitrate;
>@@ -366,7 +411,8 @@ class TestSimulcastEncoderAdapterFake : public ::testing::Test,
>     // stream 0, the lowest resolution stream.
>     InitRefCodec(0, &ref_codec);
>     ref_codec.qpMax = 45;
>-    ref_codec.VP8()->complexity = webrtc::kComplexityHigher;
>+    ref_codec.VP8()->complexity =
>+        webrtc::VideoCodecComplexity::kComplexityHigher;
>     ref_codec.VP8()->denoisingOn = false;
>     ref_codec.startBitrate = 100;  // Should equal to the target bitrate.
>     VerifyCodec(ref_codec, 0);
>@@ -390,12 +436,11 @@ class TestSimulcastEncoderAdapterFake : public ::testing::Test,
> 
>  protected:
>   std::unique_ptr<TestSimulcastEncoderAdapterFakeHelper> helper_;
>-  std::unique_ptr<VP8Encoder> adapter_;
>+  std::unique_ptr<VideoEncoder> adapter_;
>   VideoCodec codec_;
>   int last_encoded_image_width_;
>   int last_encoded_image_height_;
>   int last_encoded_image_simulcast_index_;
>-  TemporalLayersFactory tl_factory_;
>   std::unique_ptr<SimulcastRateAllocator> rate_allocator_;
> };
> 
>@@ -464,11 +509,10 @@ TEST_F(TestSimulcastEncoderAdapterFake, EncodedCallbackForDifferentEncoders) {
> // with the lowest stream.
> TEST_F(TestSimulcastEncoderAdapterFake, ReusesEncodersInOrder) {
>   // Set up common settings for three streams.
>-  TestVp8Simulcast::DefaultSettings(
>-      &codec_, static_cast<const int*>(kTestTemporalLayerProfile));
>-  rate_allocator_.reset(new SimulcastRateAllocator(codec_, nullptr));
>-  tl_factory_.SetListener(rate_allocator_.get());
>-  codec_.VP8()->tl_factory = &tl_factory_;
>+  SimulcastTestFixtureImpl::DefaultSettings(
>+      &codec_, static_cast<const int*>(kTestTemporalLayerProfile),
>+      kVideoCodecVP8);
>+  rate_allocator_.reset(new SimulcastRateAllocator(codec_));
>   adapter_->RegisterEncodeCompleteCallback(this);
> 
>   // Input data.
>@@ -664,9 +708,9 @@ TEST_F(TestSimulcastEncoderAdapterFake, ReinitDoesNotReorderFrameSimulcastIdx) {
> }
> 
> TEST_F(TestSimulcastEncoderAdapterFake, SupportsNativeHandleForSingleStreams) {
>-  TestVp8Simulcast::DefaultSettings(
>-      &codec_, static_cast<const int*>(kTestTemporalLayerProfile));
>-  codec_.VP8()->tl_factory = &tl_factory_;
>+  SimulcastTestFixtureImpl::DefaultSettings(
>+      &codec_, static_cast<const int*>(kTestTemporalLayerProfile),
>+      kVideoCodecVP8);
>   codec_.numberOfSimulcastStreams = 1;
>   EXPECT_EQ(0, adapter_->InitEncode(&codec_, 1, 1200));
>   adapter_->RegisterEncodeCompleteCallback(this);
>@@ -678,39 +722,39 @@ TEST_F(TestSimulcastEncoderAdapterFake, SupportsNativeHandleForSingleStreams) {
> }
> 
> TEST_F(TestSimulcastEncoderAdapterFake, SetRatesUnderMinBitrate) {
>-  TestVp8Simulcast::DefaultSettings(
>-      &codec_, static_cast<const int*>(kTestTemporalLayerProfile));
>-  codec_.VP8()->tl_factory = &tl_factory_;
>+  SimulcastTestFixtureImpl::DefaultSettings(
>+      &codec_, static_cast<const int*>(kTestTemporalLayerProfile),
>+      kVideoCodecVP8);
>   codec_.minBitrate = 50;
>   codec_.numberOfSimulcastStreams = 1;
>   EXPECT_EQ(0, adapter_->InitEncode(&codec_, 1, 1200));
>-  rate_allocator_.reset(new SimulcastRateAllocator(codec_, nullptr));
>+  rate_allocator_.reset(new SimulcastRateAllocator(codec_));
> 
>   // Above min should be respected.
>-  BitrateAllocation target_bitrate =
>+  VideoBitrateAllocation target_bitrate =
>       rate_allocator_->GetAllocation(codec_.minBitrate * 1000, 30);
>   adapter_->SetRateAllocation(target_bitrate, 30);
>   EXPECT_EQ(target_bitrate,
>             helper_->factory()->encoders()[0]->last_set_bitrate());
> 
>   // Below min but non-zero should be replaced with the min bitrate.
>-  BitrateAllocation too_low_bitrate =
>+  VideoBitrateAllocation too_low_bitrate =
>       rate_allocator_->GetAllocation((codec_.minBitrate - 1) * 1000, 30);
>   adapter_->SetRateAllocation(too_low_bitrate, 30);
>   EXPECT_EQ(target_bitrate,
>             helper_->factory()->encoders()[0]->last_set_bitrate());
> 
>   // Zero should be passed on as is, since it means "pause".
>-  adapter_->SetRateAllocation(BitrateAllocation(), 30);
>-  EXPECT_EQ(BitrateAllocation(),
>+  adapter_->SetRateAllocation(VideoBitrateAllocation(), 30);
>+  EXPECT_EQ(VideoBitrateAllocation(),
>             helper_->factory()->encoders()[0]->last_set_bitrate());
> }
> 
> TEST_F(TestSimulcastEncoderAdapterFake, SupportsImplementationName) {
>   EXPECT_STREQ("SimulcastEncoderAdapter", adapter_->ImplementationName());
>-  TestVp8Simulcast::DefaultSettings(
>-      &codec_, static_cast<const int*>(kTestTemporalLayerProfile));
>-  codec_.VP8()->tl_factory = &tl_factory_;
>+  SimulcastTestFixtureImpl::DefaultSettings(
>+      &codec_, static_cast<const int*>(kTestTemporalLayerProfile),
>+      kVideoCodecVP8);
>   std::vector<const char*> encoder_names;
>   encoder_names.push_back("codec1");
>   encoder_names.push_back("codec2");
>@@ -731,9 +775,9 @@ TEST_F(TestSimulcastEncoderAdapterFake, SupportsImplementationName) {
> 
> TEST_F(TestSimulcastEncoderAdapterFake,
>        SupportsNativeHandleForMultipleStreams) {
>-  TestVp8Simulcast::DefaultSettings(
>-      &codec_, static_cast<const int*>(kTestTemporalLayerProfile));
>-  codec_.VP8()->tl_factory = &tl_factory_;
>+  SimulcastTestFixtureImpl::DefaultSettings(
>+      &codec_, static_cast<const int*>(kTestTemporalLayerProfile),
>+      kVideoCodecVP8);
>   codec_.numberOfSimulcastStreams = 3;
>   EXPECT_EQ(0, adapter_->InitEncode(&codec_, 1, 1200));
>   adapter_->RegisterEncodeCompleteCallback(this);
>@@ -749,9 +793,10 @@ TEST_F(TestSimulcastEncoderAdapterFake,
> }
> 
> // TODO(nisse): Reuse definition in webrtc/test/fake_texture_handle.h.
>-class FakeNativeBuffer : public VideoFrameBuffer {
>+class FakeNativeBufferNoI420 : public VideoFrameBuffer {
>  public:
>-  FakeNativeBuffer(int width, int height) : width_(width), height_(height) {}
>+  FakeNativeBufferNoI420(int width, int height)
>+      : width_(width), height_(height) {}
> 
>   Type type() const override { return Type::kNative; }
>   int width() const override { return width_; }
>@@ -769,9 +814,9 @@ class FakeNativeBuffer : public VideoFrameBuffer {
> 
> TEST_F(TestSimulcastEncoderAdapterFake,
>        NativeHandleForwardingForMultipleStreams) {
>-  TestVp8Simulcast::DefaultSettings(
>-      &codec_, static_cast<const int*>(kTestTemporalLayerProfile));
>-  codec_.VP8()->tl_factory = &tl_factory_;
>+  SimulcastTestFixtureImpl::DefaultSettings(
>+      &codec_, static_cast<const int*>(kTestTemporalLayerProfile),
>+      kVideoCodecVP8);
>   codec_.numberOfSimulcastStreams = 3;
>   // High start bitrate, so all streams are enabled.
>   codec_.startBitrate = 3000;
>@@ -783,7 +828,7 @@ TEST_F(TestSimulcastEncoderAdapterFake,
>   EXPECT_TRUE(adapter_->SupportsNativeHandle());
> 
>   rtc::scoped_refptr<VideoFrameBuffer> buffer(
>-      new rtc::RefCountedObject<FakeNativeBuffer>(1280, 720));
>+      new rtc::RefCountedObject<FakeNativeBufferNoI420>(1280, 720));
>   VideoFrame input_frame(buffer, 100, 1000, kVideoRotation_180);
>   // Expect calls with the given video frame verbatim, since it's a texture
>   // frame and can't otherwise be modified/resized.
>@@ -794,9 +839,9 @@ TEST_F(TestSimulcastEncoderAdapterFake,
> }
> 
> TEST_F(TestSimulcastEncoderAdapterFake, TestFailureReturnCodesFromEncodeCalls) {
>-  TestVp8Simulcast::DefaultSettings(
>-      &codec_, static_cast<const int*>(kTestTemporalLayerProfile));
>-  codec_.VP8()->tl_factory = &tl_factory_;
>+  SimulcastTestFixtureImpl::DefaultSettings(
>+      &codec_, static_cast<const int*>(kTestTemporalLayerProfile),
>+      kVideoCodecVP8);
>   codec_.numberOfSimulcastStreams = 3;
>   EXPECT_EQ(0, adapter_->InitEncode(&codec_, 1, 1200));
>   adapter_->RegisterEncodeCompleteCallback(this);
>@@ -816,9 +861,9 @@ TEST_F(TestSimulcastEncoderAdapterFake, TestFailureReturnCodesFromEncodeCalls) {
> }
> 
> TEST_F(TestSimulcastEncoderAdapterFake, TestInitFailureCleansUpEncoders) {
>-  TestVp8Simulcast::DefaultSettings(
>-      &codec_, static_cast<const int*>(kTestTemporalLayerProfile));
>-  codec_.VP8()->tl_factory = &tl_factory_;
>+  SimulcastTestFixtureImpl::DefaultSettings(
>+      &codec_, static_cast<const int*>(kTestTemporalLayerProfile),
>+      kVideoCodecVP8);
>   codec_.numberOfSimulcastStreams = 3;
>   helper_->factory()->set_init_encode_return_value(
>       WEBRTC_VIDEO_CODEC_FALLBACK_SOFTWARE);
>@@ -827,5 +872,36 @@ TEST_F(TestSimulcastEncoderAdapterFake, TestInitFailureCleansUpEncoders) {
>   EXPECT_TRUE(helper_->factory()->encoders().empty());
> }
> 
>-}  // namespace testing
>+TEST_F(TestSimulcastEncoderAdapterFake, DoesNotAlterMaxQpForScreenshare) {
>+  const int kHighMaxQp = 56;
>+  const int kLowMaxQp = 46;
>+
>+  SimulcastTestFixtureImpl::DefaultSettings(
>+      &codec_, static_cast<const int*>(kTestTemporalLayerProfile),
>+      kVideoCodecVP8);
>+  codec_.numberOfSimulcastStreams = 3;
>+  codec_.simulcastStream[0].qpMax = kHighMaxQp;
>+  codec_.mode = VideoCodecMode::kScreensharing;
>+
>+  EXPECT_EQ(0, adapter_->InitEncode(&codec_, 1, 1200));
>+  EXPECT_EQ(3u, helper_->factory()->encoders().size());
>+
>+  // Just check the lowest stream, which is the one that where the adapter
>+  // might alter the max qp setting.
>+  VideoCodec ref_codec;
>+  InitRefCodec(0, &ref_codec);
>+  ref_codec.qpMax = kHighMaxQp;
>+  ref_codec.VP8()->complexity = webrtc::VideoCodecComplexity::kComplexityHigher;
>+  ref_codec.VP8()->denoisingOn = false;
>+  ref_codec.startBitrate = 100;  // Should equal to the target bitrate.
>+  VerifyCodec(ref_codec, 0);
>+
>+  // Change the max qp and try again.
>+  codec_.simulcastStream[0].qpMax = kLowMaxQp;
>+  EXPECT_EQ(0, adapter_->InitEncode(&codec_, 1, 1200));
>+  EXPECT_EQ(3u, helper_->factory()->encoders().size());
>+  ref_codec.qpMax = kLowMaxQp;
>+  VerifyCodec(ref_codec, 0);
>+}
>+}  // namespace test
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/simulcast_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/simulcast_unittest.cc
>index 1433c5c335d..29226d12114 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/simulcast_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/simulcast_unittest.cc
>@@ -1,5 +1,5 @@
> /*
>- *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>  *
>  *  Use of this source code is governed by a BSD-style license
>  *  that can be found in the LICENSE file in the root of the source
>@@ -8,49 +8,159 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include <string>
>-
> #include "media/engine/simulcast.h"
>+
>+#include "media/engine/constants.h"
>+#include "test/field_trial.h"
> #include "test/gtest.h"
> 
>-namespace cricket {
>+namespace webrtc {
>+namespace {
>+constexpr int kQpMax = 55;
>+constexpr double kBitratePriority = 2.0;
>+constexpr int kMaxFps = 33;
>+constexpr int kMaxBitrateBps = 0;
>+constexpr bool kScreenshare = true;
>+constexpr int kDefaultTemporalLayers = 3;  // Value from simulcast.cc.
>+
>+// Values from kSimulcastConfigs in simulcast.cc.
>+const std::vector<VideoStream> GetSimulcastBitrates720p() {
>+  std::vector<VideoStream> streams(3);
>+  streams[0].min_bitrate_bps = 30000;
>+  streams[0].target_bitrate_bps = 150000;
>+  streams[0].max_bitrate_bps = 200000;
>+  streams[1].min_bitrate_bps = 150000;
>+  streams[1].target_bitrate_bps = 500000;
>+  streams[1].max_bitrate_bps = 700000;
>+  streams[2].min_bitrate_bps = 600000;
>+  streams[2].target_bitrate_bps = 2500000;
>+  streams[2].max_bitrate_bps = 2500000;
>+  return streams;
>+}
>+}  // namespace
>+
>+TEST(SimulcastTest, TotalMaxBitrateIsZeroForNoStreams) {
>+  std::vector<VideoStream> streams;
>+  EXPECT_EQ(0, cricket::GetTotalMaxBitrateBps(streams));
>+}
>+
>+TEST(SimulcastTest, GetTotalMaxBitrateForSingleStream) {
>+  std::vector<VideoStream> streams(1);
>+  streams[0].max_bitrate_bps = 100000;
>+  EXPECT_EQ(100000, cricket::GetTotalMaxBitrateBps(streams));
>+}
>+
>+TEST(SimulcastTest, GetTotalMaxBitrateForMultipleStreams) {
>+  std::vector<VideoStream> streams(3);
>+  streams[0].target_bitrate_bps = 100000;
>+  streams[1].target_bitrate_bps = 200000;
>+  streams[2].max_bitrate_bps = 400000;
>+  EXPECT_EQ(700000, cricket::GetTotalMaxBitrateBps(streams));
>+}
>+
>+TEST(SimulcastTest, BandwidthAboveTotalMaxBitrateGivenToHighestStream) {
>+  std::vector<VideoStream> streams(3);
>+  streams[0].target_bitrate_bps = 100000;
>+  streams[1].target_bitrate_bps = 200000;
>+  streams[2].max_bitrate_bps = 400000;
>+
>+  // No bitrate above the total max to give to the highest stream.
>+  const int kMaxTotalBps = cricket::GetTotalMaxBitrateBps(streams);
>+  cricket::BoostMaxSimulcastLayer(kMaxTotalBps, &streams);
>+  EXPECT_EQ(400000, streams[2].max_bitrate_bps);
>+  EXPECT_EQ(kMaxTotalBps, cricket::GetTotalMaxBitrateBps(streams));
> 
>-class ScreenshareLayerConfigTest : public testing::Test,
>-                                   protected ScreenshareLayerConfig {
>- public:
>-  ScreenshareLayerConfigTest() : ScreenshareLayerConfig(0, 0) {}
>+  // The bitrate above the total max should be given to the highest stream.
>+  cricket::BoostMaxSimulcastLayer(kMaxTotalBps + 1, &streams);
>+  EXPECT_EQ(400000 + 1, streams[2].max_bitrate_bps);
>+  EXPECT_EQ(kMaxTotalBps + 1, cricket::GetTotalMaxBitrateBps(streams));
>+}
>+
>+TEST(SimulcastTest, GetConfig) {
>+  const std::vector<VideoStream> kExpected = GetSimulcastBitrates720p();
>+
>+  const size_t kMaxLayers = 3;
>+  std::vector<VideoStream> streams = cricket::GetSimulcastConfig(
>+      kMaxLayers, 1280, 720, kMaxBitrateBps, kBitratePriority, kQpMax, kMaxFps,
>+      !kScreenshare);
> 
>-  void ExpectParsingFails(const std::string& group) {
>-    ScreenshareLayerConfig config(100, 1000);
>-    EXPECT_FALSE(FromFieldTrialGroup(group, &config));
>+  EXPECT_EQ(kMaxLayers, streams.size());
>+  EXPECT_EQ(320u, streams[0].width);
>+  EXPECT_EQ(180u, streams[0].height);
>+  EXPECT_EQ(640u, streams[1].width);
>+  EXPECT_EQ(360u, streams[1].height);
>+  EXPECT_EQ(1280u, streams[2].width);
>+  EXPECT_EQ(720u, streams[2].height);
>+
>+  for (size_t i = 0; i < streams.size(); ++i) {
>+    EXPECT_EQ(size_t{kDefaultTemporalLayers}, streams[i].num_temporal_layers);
>+    EXPECT_EQ(kMaxFps, streams[i].max_framerate);
>+    EXPECT_EQ(kQpMax, streams[i].max_qp);
>+    EXPECT_EQ(kExpected[i].min_bitrate_bps, streams[i].min_bitrate_bps);
>+    EXPECT_EQ(kExpected[i].target_bitrate_bps, streams[i].target_bitrate_bps);
>+    EXPECT_EQ(kExpected[i].max_bitrate_bps, streams[i].max_bitrate_bps);
>+    EXPECT_TRUE(streams[i].active);
>   }
>-};
>+  // Currently set on lowest stream.
>+  EXPECT_EQ(kBitratePriority, streams[0].bitrate_priority);
>+  EXPECT_FALSE(streams[1].bitrate_priority);
>+  EXPECT_FALSE(streams[2].bitrate_priority);
>+}
>+
>+TEST(SimulcastTest, GetConfigWithLimitedMaxLayers) {
>+  const size_t kMaxLayers = 2;
>+  std::vector<VideoStream> streams = cricket::GetSimulcastConfig(
>+      kMaxLayers, 1280, 720, kMaxBitrateBps, kBitratePriority, kQpMax, kMaxFps,
>+      !kScreenshare);
> 
>-TEST_F(ScreenshareLayerConfigTest, UsesDefaultBitrateConfigForDefaultGroup) {
>-  ExpectParsingFails("");
>+  EXPECT_EQ(kMaxLayers, streams.size());
>+  EXPECT_EQ(320u, streams[0].width);
>+  EXPECT_EQ(180u, streams[0].height);
>+  EXPECT_EQ(640u, streams[1].width);
>+  EXPECT_EQ(360u, streams[1].height);
> }
> 
>-TEST_F(ScreenshareLayerConfigTest, UsesDefaultConfigForInvalidBitrates) {
>-  ExpectParsingFails("-");
>-  ExpectParsingFails("1-");
>-  ExpectParsingFails("-1");
>-  ExpectParsingFails("-12");
>-  ExpectParsingFails("12-");
>-  ExpectParsingFails("booh!");
>-  ExpectParsingFails("1-b");
>-  ExpectParsingFails("a-2");
>-  ExpectParsingFails("49-1000");
>-  ExpectParsingFails("50-6001");
>-  ExpectParsingFails("100-99");
>-  ExpectParsingFails("1002003004005006-99");
>-  ExpectParsingFails("99-1002003004005006");
>+TEST(SimulcastTest, GetConfigWithNormalizedResolution) {
>+  const size_t kMaxLayers = 2;
>+  std::vector<VideoStream> streams = cricket::GetSimulcastConfig(
>+      kMaxLayers, 640 + 1, 360 + 1, kMaxBitrateBps, kBitratePriority, kQpMax,
>+      kMaxFps, !kScreenshare);
>+
>+  // Must be dividable by |2 ^ (num_layers - 1)|.
>+  EXPECT_EQ(kMaxLayers, streams.size());
>+  EXPECT_EQ(320u, streams[0].width);
>+  EXPECT_EQ(180u, streams[0].height);
>+  EXPECT_EQ(640u, streams[1].width);
>+  EXPECT_EQ(360u, streams[1].height);
>+}
>+
>+TEST(SimulcastTest, GetConfigForScreenshareSimulcast) {
>+  const size_t kMaxLayers = 3;
>+  std::vector<VideoStream> streams = cricket::GetSimulcastConfig(
>+      kMaxLayers, 1400, 800, kMaxBitrateBps, kBitratePriority, kQpMax, kMaxFps,
>+      kScreenshare);
>+
>+  EXPECT_GT(streams.size(), 1u);
>+  for (size_t i = 0; i < streams.size(); ++i) {
>+    EXPECT_EQ(1400u, streams[i].width) << "Screen content never scaled.";
>+    EXPECT_EQ(800u, streams[i].height) << "Screen content never scaled.";
>+    EXPECT_EQ(kQpMax, streams[i].max_qp);
>+    EXPECT_TRUE(streams[i].active);
>+    EXPECT_GT(streams[i].num_temporal_layers, size_t{1});
>+    EXPECT_GT(streams[i].max_framerate, 0);
>+    EXPECT_GT(streams[i].min_bitrate_bps, 0);
>+    EXPECT_GT(streams[i].target_bitrate_bps, streams[i].min_bitrate_bps);
>+    EXPECT_GE(streams[i].max_bitrate_bps, streams[i].target_bitrate_bps);
>+  }
> }
> 
>-TEST_F(ScreenshareLayerConfigTest, ParsesValidBitrateConfig) {
>-  ScreenshareLayerConfig config(100, 1000);
>-  EXPECT_TRUE(ScreenshareLayerConfig::FromFieldTrialGroup("101-1001", &config));
>-  EXPECT_EQ(101, config.tl0_bitrate_kbps);
>-  EXPECT_EQ(1001, config.tl1_bitrate_kbps);
>+TEST(SimulcastTest, GetConfigForScreenshareSimulcastWithLimitedMaxLayers) {
>+  const size_t kMaxLayers = 1;
>+  std::vector<VideoStream> streams = cricket::GetSimulcastConfig(
>+      kMaxLayers, 1400, 800, kMaxBitrateBps, kBitratePriority, kQpMax, kMaxFps,
>+      kScreenshare);
>+
>+  EXPECT_EQ(kMaxLayers, streams.size());
> }
> 
>-}  // namespace cricket
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/stereocodecfactory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/stereocodecfactory.cc
>deleted file mode 100644
>index 5ffb0c406dd..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/stereocodecfactory.cc
>+++ /dev/null
>@@ -1,111 +0,0 @@
>-/*
>- *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "media/engine/stereocodecfactory.h"
>-
>-#include <utility>
>-
>-#include "api/video_codecs/sdp_video_format.h"
>-#include "media/base/codec.h"
>-#include "media/base/mediaconstants.h"
>-#include "modules/video_coding/codecs/stereo/include/stereo_decoder_adapter.h"
>-#include "modules/video_coding/codecs/stereo/include/stereo_encoder_adapter.h"
>-#include "rtc_base/logging.h"
>-
>-namespace {
>-
>-bool IsStereoCodec(const cricket::VideoCodec& codec) {
>-  return cricket::CodecNamesEq(codec.name.c_str(), cricket::kStereoCodecName);
>-}
>-
>-}  // anonymous namespace
>-
>-namespace webrtc {
>-
>-constexpr const char* kStereoAssociatedCodecName = cricket::kVp9CodecName;
>-
>-StereoEncoderFactory::StereoEncoderFactory(
>-    std::unique_ptr<VideoEncoderFactory> factory)
>-    : factory_(std::move(factory)) {}
>-
>-std::vector<SdpVideoFormat> StereoEncoderFactory::GetSupportedFormats() const {
>-  std::vector<SdpVideoFormat> formats = factory_->GetSupportedFormats();
>-  for (const auto& format : formats) {
>-    if (cricket::CodecNamesEq(format.name, kStereoAssociatedCodecName)) {
>-      SdpVideoFormat stereo_format = format;
>-      stereo_format.parameters[cricket::kCodecParamAssociatedCodecName] =
>-          format.name;
>-      stereo_format.name = cricket::kStereoCodecName;
>-      formats.push_back(stereo_format);
>-      break;
>-    }
>-  }
>-  return formats;
>-}
>-
>-VideoEncoderFactory::CodecInfo StereoEncoderFactory::QueryVideoEncoder(
>-    const SdpVideoFormat& format) const {
>-  if (!IsStereoCodec(cricket::VideoCodec(format)))
>-    return factory_->QueryVideoEncoder(format);
>-  return factory_->QueryVideoEncoder(
>-      SdpVideoFormat(kStereoAssociatedCodecName));
>-}
>-
>-std::unique_ptr<VideoEncoder> StereoEncoderFactory::CreateVideoEncoder(
>-    const SdpVideoFormat& format) {
>-  if (!IsStereoCodec(cricket::VideoCodec(format)))
>-    return factory_->CreateVideoEncoder(format);
>-  const auto& it =
>-      format.parameters.find(cricket::kCodecParamAssociatedCodecName);
>-  if (it == format.parameters.end()) {
>-    RTC_LOG(LS_ERROR) << "No assicated codec for stereo.";
>-    return nullptr;
>-  }
>-  SdpVideoFormat associated_format = format;
>-  associated_format.name = it->second;
>-  return std::unique_ptr<VideoEncoder>(
>-      new StereoEncoderAdapter(factory_.get(), associated_format));
>-}
>-
>-StereoDecoderFactory::StereoDecoderFactory(
>-    std::unique_ptr<VideoDecoderFactory> factory)
>-    : factory_(std::move(factory)) {}
>-
>-std::vector<SdpVideoFormat> StereoDecoderFactory::GetSupportedFormats() const {
>-  std::vector<SdpVideoFormat> formats = factory_->GetSupportedFormats();
>-  for (const auto& format : formats) {
>-    if (cricket::CodecNamesEq(format.name, kStereoAssociatedCodecName)) {
>-      SdpVideoFormat stereo_format = format;
>-      stereo_format.parameters[cricket::kCodecParamAssociatedCodecName] =
>-          format.name;
>-      stereo_format.name = cricket::kStereoCodecName;
>-      formats.push_back(stereo_format);
>-    }
>-  }
>-  return formats;
>-}
>-
>-std::unique_ptr<VideoDecoder> StereoDecoderFactory::CreateVideoDecoder(
>-    const SdpVideoFormat& format) {
>-  if (!IsStereoCodec(cricket::VideoCodec(format)))
>-    return factory_->CreateVideoDecoder(format);
>-  const auto& it =
>-      format.parameters.find(cricket::kCodecParamAssociatedCodecName);
>-  if (it == format.parameters.end()) {
>-    RTC_LOG(LS_ERROR) << "No assicated codec for stereo.";
>-    return nullptr;
>-  }
>-  SdpVideoFormat associated_format = format;
>-  associated_format.name = it->second;
>-  return std::unique_ptr<VideoDecoder>(
>-      new StereoDecoderAdapter(factory_.get(), associated_format));
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/stereocodecfactory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/stereocodecfactory.h
>deleted file mode 100644
>index 468d2faf0b6..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/stereocodecfactory.h
>+++ /dev/null
>@@ -1,49 +0,0 @@
>-/*
>- *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MEDIA_ENGINE_STEREOCODECFACTORY_H_
>-#define MEDIA_ENGINE_STEREOCODECFACTORY_H_
>-
>-#include <memory>
>-#include <vector>
>-
>-#include "api/video_codecs/video_decoder_factory.h"
>-#include "api/video_codecs/video_encoder_factory.h"
>-
>-namespace webrtc {
>-
>-class StereoEncoderFactory : public VideoEncoderFactory {
>- public:
>-  explicit StereoEncoderFactory(std::unique_ptr<VideoEncoderFactory> factory);
>-
>-  std::vector<SdpVideoFormat> GetSupportedFormats() const override;
>-  CodecInfo QueryVideoEncoder(const SdpVideoFormat& format) const override;
>-  std::unique_ptr<VideoEncoder> CreateVideoEncoder(
>-      const SdpVideoFormat& format) override;
>-
>- private:
>-  std::unique_ptr<VideoEncoderFactory> factory_;
>-};
>-
>-class StereoDecoderFactory : public VideoDecoderFactory {
>- public:
>-  explicit StereoDecoderFactory(std::unique_ptr<VideoDecoderFactory> factory);
>-
>-  std::vector<SdpVideoFormat> GetSupportedFormats() const override;
>-  std::unique_ptr<VideoDecoder> CreateVideoDecoder(
>-      const SdpVideoFormat& format) override;
>-
>- private:
>-  std::unique_ptr<VideoDecoderFactory> factory_;
>-};
>-
>-}  // namespace webrtc
>-
>-#endif  // MEDIA_ENGINE_STEREOCODECFACTORY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/videodecodersoftwarefallbackwrapper.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/videodecodersoftwarefallbackwrapper.cc
>deleted file mode 100644
>index 87fb62239a2..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/videodecodersoftwarefallbackwrapper.cc
>+++ /dev/null
>@@ -1,141 +0,0 @@
>-/*
>- *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "media/engine/videodecodersoftwarefallbackwrapper.h"
>-
>-#include <string>
>-#include <utility>
>-
>-#include "media/engine/internaldecoderfactory.h"
>-#include "modules/video_coding/include/video_error_codes.h"
>-#include "rtc_base/checks.h"
>-#include "rtc_base/logging.h"
>-#include "rtc_base/trace_event.h"
>-
>-namespace webrtc {
>-
>-VideoDecoderSoftwareFallbackWrapper::VideoDecoderSoftwareFallbackWrapper(
>-    std::unique_ptr<VideoDecoder> sw_fallback_decoder,
>-    std::unique_ptr<VideoDecoder> hw_decoder)
>-    : use_hw_decoder_(true),
>-      hw_decoder_(std::move(hw_decoder)),
>-      hw_decoder_initialized_(false),
>-      fallback_decoder_(std::move(sw_fallback_decoder)),
>-      fallback_implementation_name_(
>-          std::string(fallback_decoder_->ImplementationName()) +
>-          " (fallback from: " + hw_decoder_->ImplementationName() + ")"),
>-      callback_(nullptr) {}
>-
>-int32_t VideoDecoderSoftwareFallbackWrapper::InitDecode(
>-    const VideoCodec* codec_settings,
>-    int32_t number_of_cores) {
>-  // Always try to use the HW decoder in this state.
>-  use_hw_decoder_ = true;
>-  codec_settings_ = *codec_settings;
>-  number_of_cores_ = number_of_cores;
>-  int32_t ret = hw_decoder_->InitDecode(codec_settings, number_of_cores);
>-  if (ret == WEBRTC_VIDEO_CODEC_OK) {
>-    hw_decoder_initialized_ = true;
>-    return ret;
>-  }
>-  hw_decoder_initialized_ = false;
>-
>-  // Try to initialize fallback decoder.
>-  if (InitFallbackDecoder())
>-    return WEBRTC_VIDEO_CODEC_OK;
>-
>-  return ret;
>-}
>-
>-bool VideoDecoderSoftwareFallbackWrapper::InitFallbackDecoder() {
>-  RTC_LOG(LS_WARNING) << "Decoder falling back to software decoding.";
>-  if (fallback_decoder_->InitDecode(&codec_settings_, number_of_cores_) !=
>-      WEBRTC_VIDEO_CODEC_OK) {
>-    RTC_LOG(LS_ERROR) << "Failed to initialize software-decoder fallback.";
>-    use_hw_decoder_ = true;
>-    return false;
>-  }
>-  if (callback_)
>-    fallback_decoder_->RegisterDecodeCompleteCallback(callback_);
>-  use_hw_decoder_ = false;
>-  return true;
>-}
>-
>-int32_t VideoDecoderSoftwareFallbackWrapper::Decode(
>-    const EncodedImage& input_image,
>-    bool missing_frames,
>-    const RTPFragmentationHeader* fragmentation,
>-    const CodecSpecificInfo* codec_specific_info,
>-    int64_t render_time_ms) {
>-    TRACE_EVENT0("webrtc", "VideoDecoderSoftwareFallbackWrapper::Decode");
>-  // Try initializing and decoding with the provided decoder on every keyframe
>-  // or when there's no fallback decoder. This is the normal case.
>-  if (use_hw_decoder_ || input_image._frameType == kVideoFrameKey) {
>-    int32_t ret = WEBRTC_VIDEO_CODEC_FALLBACK_SOFTWARE;
>-    // Try reinitializing the decoder if it had failed before.
>-    if (!hw_decoder_initialized_) {
>-      hw_decoder_initialized_ =
>-          hw_decoder_->InitDecode(&codec_settings_, number_of_cores_) ==
>-          WEBRTC_VIDEO_CODEC_OK;
>-    }
>-    if (hw_decoder_initialized_) {
>-      ret = hw_decoder_->Decode(input_image, missing_frames, fragmentation,
>-                             codec_specific_info, render_time_ms);
>-    }
>-    if (ret == WEBRTC_VIDEO_CODEC_OK) {
>-      if (!use_hw_decoder_) {
>-        // Decode OK -> stop using fallback decoder.
>-        RTC_LOG(LS_WARNING)
>-            << "Decode OK, no longer using the software fallback decoder.";
>-        use_hw_decoder_ = true;
>-        return WEBRTC_VIDEO_CODEC_OK;
>-      }
>-    }
>-    if (ret != WEBRTC_VIDEO_CODEC_FALLBACK_SOFTWARE)
>-      return ret;
>-    if (use_hw_decoder_) {
>-      // Try to initialize fallback decoder.
>-      if (!InitFallbackDecoder())
>-        return ret;
>-    }
>-  }
>-  return fallback_decoder_->Decode(input_image, missing_frames, fragmentation,
>-                                   codec_specific_info, render_time_ms);
>-}
>-
>-int32_t VideoDecoderSoftwareFallbackWrapper::RegisterDecodeCompleteCallback(
>-    DecodedImageCallback* callback) {
>-  callback_ = callback;
>-  int32_t ret = hw_decoder_->RegisterDecodeCompleteCallback(callback);
>-  if (!use_hw_decoder_)
>-    return fallback_decoder_->RegisterDecodeCompleteCallback(callback);
>-  return ret;
>-}
>-
>-int32_t VideoDecoderSoftwareFallbackWrapper::Release() {
>-  if (!use_hw_decoder_) {
>-    RTC_LOG(LS_INFO) << "Releasing software fallback decoder.";
>-    fallback_decoder_->Release();
>-  }
>-  hw_decoder_initialized_ = false;
>-  return hw_decoder_->Release();
>-}
>-
>-bool VideoDecoderSoftwareFallbackWrapper::PrefersLateDecoding() const {
>-  return use_hw_decoder_ ? hw_decoder_->PrefersLateDecoding()
>-                         : fallback_decoder_->PrefersLateDecoding();
>-}
>-
>-const char* VideoDecoderSoftwareFallbackWrapper::ImplementationName() const {
>-  return use_hw_decoder_ ? hw_decoder_->ImplementationName()
>-                         : fallback_implementation_name_.c_str();
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/videodecodersoftwarefallbackwrapper.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/videodecodersoftwarefallbackwrapper.h
>deleted file mode 100644
>index 97953ddc826..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/videodecodersoftwarefallbackwrapper.h
>+++ /dev/null
>@@ -1,64 +0,0 @@
>-/*
>- *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MEDIA_ENGINE_VIDEODECODERSOFTWAREFALLBACKWRAPPER_H_
>-#define MEDIA_ENGINE_VIDEODECODERSOFTWAREFALLBACKWRAPPER_H_
>-
>-#include <memory>
>-#include <string>
>-
>-#include "api/video_codecs/video_decoder.h"
>-
>-namespace webrtc {
>-
>-// Class used to wrap external VideoDecoders to provide a fallback option on
>-// software decoding when a hardware decoder fails to decode a stream due to
>-// hardware restrictions, such as max resolution.
>-class VideoDecoderSoftwareFallbackWrapper : public VideoDecoder {
>- public:
>-  VideoDecoderSoftwareFallbackWrapper(
>-      std::unique_ptr<VideoDecoder> sw_fallback_decoder,
>-      std::unique_ptr<VideoDecoder> hw_decoder);
>-
>-  int32_t InitDecode(const VideoCodec* codec_settings,
>-                     int32_t number_of_cores) override;
>-
>-  int32_t Decode(const EncodedImage& input_image,
>-                 bool missing_frames,
>-                 const RTPFragmentationHeader* fragmentation,
>-                 const CodecSpecificInfo* codec_specific_info,
>-                 int64_t render_time_ms) override;
>-
>-  int32_t RegisterDecodeCompleteCallback(
>-      DecodedImageCallback* callback) override;
>-
>-  int32_t Release() override;
>-  bool PrefersLateDecoding() const override;
>-
>-  const char* ImplementationName() const override;
>-
>- private:
>-  bool InitFallbackDecoder();
>-
>-  // Determines if we are trying to use the HW or SW decoder.
>-  bool use_hw_decoder_;
>-  std::unique_ptr<VideoDecoder> hw_decoder_;
>-  bool hw_decoder_initialized_;
>-
>-  VideoCodec codec_settings_;
>-  int32_t number_of_cores_;
>-  const std::unique_ptr<VideoDecoder> fallback_decoder_;
>-  const std::string fallback_implementation_name_;
>-  DecodedImageCallback* callback_;
>-};
>-
>-}  // namespace webrtc
>-
>-#endif  // MEDIA_ENGINE_VIDEODECODERSOFTWAREFALLBACKWRAPPER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/videoencodersoftwarefallbackwrapper.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/videoencodersoftwarefallbackwrapper.h
>deleted file mode 100644
>index a9a349c221f..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/videoencodersoftwarefallbackwrapper.h
>+++ /dev/null
>@@ -1,102 +0,0 @@
>-/*
>- *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MEDIA_ENGINE_VIDEOENCODERSOFTWAREFALLBACKWRAPPER_H_
>-#define MEDIA_ENGINE_VIDEOENCODERSOFTWAREFALLBACKWRAPPER_H_
>-
>-#include <memory>
>-#include <string>
>-#include <vector>
>-
>-#include "api/video_codecs/video_encoder.h"
>-#include "media/base/codec.h"
>-
>-namespace webrtc {
>-
>-// Class used to wrap external VideoEncoders to provide a fallback option on
>-// software encoding when a hardware encoder fails to encode a stream due to
>-// hardware restrictions, such as max resolution.
>-class VideoEncoderSoftwareFallbackWrapper : public VideoEncoder {
>- public:
>-  VideoEncoderSoftwareFallbackWrapper(
>-      std::unique_ptr<webrtc::VideoEncoder> sw_encoder,
>-      std::unique_ptr<webrtc::VideoEncoder> hw_encoder);
>-
>-  int32_t InitEncode(const VideoCodec* codec_settings,
>-                     int32_t number_of_cores,
>-                     size_t max_payload_size) override;
>-
>-  int32_t RegisterEncodeCompleteCallback(
>-      EncodedImageCallback* callback) override;
>-
>-  int32_t Release() override;
>-  int32_t Encode(const VideoFrame& frame,
>-                 const CodecSpecificInfo* codec_specific_info,
>-                 const std::vector<FrameType>* frame_types) override;
>-  int32_t SetChannelParameters(uint32_t packet_loss, int64_t rtt) override;
>-  int32_t SetRateAllocation(const BitrateAllocation& bitrate_allocation,
>-                            uint32_t framerate) override;
>-  bool SupportsNativeHandle() const override;
>-  ScalingSettings GetScalingSettings() const override;
>-  const char *ImplementationName() const override;
>-
>- private:
>-  bool InitFallbackEncoder();
>-
>-  // If |forced_fallback_possible_| is true:
>-  // The forced fallback is requested if the resolution is less than or equal to
>-  // |max_pixels_|. The resolution is allowed to be scaled down to
>-  // |min_pixels_|.
>-  class ForcedFallbackParams {
>-   public:
>-    bool IsValid(const VideoCodec& codec) const {
>-      return codec.width * codec.height <= max_pixels_;
>-    }
>-
>-    bool active_ = false;
>-    int min_pixels_ = 320 * 180;
>-    int max_pixels_ = 320 * 240;
>-  };
>-
>-  bool TryInitForcedFallbackEncoder();
>-  bool TryReInitForcedFallbackEncoder();
>-  void ValidateSettingsForForcedFallback();
>-  bool IsForcedFallbackActive() const;
>-  void MaybeModifyCodecForFallback();
>-
>-  // Settings used in the last InitEncode call and used if a dynamic fallback to
>-  // software is required.
>-  VideoCodec codec_settings_;
>-  int32_t number_of_cores_;
>-  size_t max_payload_size_;
>-
>-  // The last bitrate/framerate set, and a flag for noting they are set.
>-  bool rates_set_;
>-  BitrateAllocation bitrate_allocation_;
>-  uint32_t framerate_;
>-
>-  // The last channel parameters set, and a flag for noting they are set.
>-  bool channel_parameters_set_;
>-  uint32_t packet_loss_;
>-  int64_t rtt_;
>-
>-  bool use_fallback_encoder_;
>-  const std::unique_ptr<webrtc::VideoEncoder> encoder_;
>-
>-  const std::unique_ptr<webrtc::VideoEncoder> fallback_encoder_;
>-  EncodedImageCallback* callback_;
>-
>-  bool forced_fallback_possible_;
>-  ForcedFallbackParams forced_fallback_;
>-};
>-
>-}  // namespace webrtc
>-
>-#endif  // MEDIA_ENGINE_VIDEOENCODERSOFTWAREFALLBACKWRAPPER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/vp8_encoder_simulcast_proxy.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/vp8_encoder_simulcast_proxy.cc
>index b97e74e5efb..4d858630b58 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/vp8_encoder_simulcast_proxy.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/vp8_encoder_simulcast_proxy.cc
>@@ -15,11 +15,15 @@
> #include "rtc_base/checks.h"
> 
> namespace webrtc {
>-VP8EncoderSimulcastProxy::VP8EncoderSimulcastProxy(VideoEncoderFactory* factory)
>-    : factory_(factory), callback_(nullptr) {
>-  encoder_ = factory_->CreateVideoEncoder(SdpVideoFormat("VP8"));
>+VP8EncoderSimulcastProxy::VP8EncoderSimulcastProxy(VideoEncoderFactory* factory,
>+                                                   const SdpVideoFormat& format)
>+    : factory_(factory), video_format_(format), callback_(nullptr) {
>+  encoder_ = factory_->CreateVideoEncoder(format);
> }
> 
>+VP8EncoderSimulcastProxy::VP8EncoderSimulcastProxy(VideoEncoderFactory* factory)
>+    : VP8EncoderSimulcastProxy(factory, SdpVideoFormat("VP8")) {}
>+
> VP8EncoderSimulcastProxy::~VP8EncoderSimulcastProxy() {}
> 
> int VP8EncoderSimulcastProxy::Release() {
>@@ -31,7 +35,7 @@ int VP8EncoderSimulcastProxy::InitEncode(const VideoCodec* inst,
>                                          size_t max_payload_size) {
>   int ret = encoder_->InitEncode(inst, number_of_cores, max_payload_size);
>   if (ret == WEBRTC_VIDEO_CODEC_ERR_SIMULCAST_PARAMETERS_NOT_SUPPORTED) {
>-    encoder_.reset(new SimulcastEncoderAdapter(factory_));
>+    encoder_.reset(new SimulcastEncoderAdapter(factory_, video_format_));
>     if (callback_) {
>       encoder_->RegisterEncodeCompleteCallback(callback_);
>     }
>@@ -59,7 +63,7 @@ int VP8EncoderSimulcastProxy::SetChannelParameters(uint32_t packet_loss,
> }
> 
> int VP8EncoderSimulcastProxy::SetRateAllocation(
>-    const BitrateAllocation& bitrate,
>+    const VideoBitrateAllocation& bitrate,
>     uint32_t new_framerate) {
>   return encoder_->SetRateAllocation(bitrate, new_framerate);
> }
>@@ -69,10 +73,6 @@ VideoEncoder::ScalingSettings VP8EncoderSimulcastProxy::GetScalingSettings()
>   return encoder_->GetScalingSettings();
> }
> 
>-int32_t VP8EncoderSimulcastProxy::SetPeriodicKeyFrames(bool enable) {
>-  return encoder_->SetPeriodicKeyFrames(enable);
>-}
>-
> bool VP8EncoderSimulcastProxy::SupportsNativeHandle() const {
>   return encoder_->SupportsNativeHandle();
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/vp8_encoder_simulcast_proxy.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/vp8_encoder_simulcast_proxy.h
>index 5833124c508..11ab0c55cee 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/vp8_encoder_simulcast_proxy.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/vp8_encoder_simulcast_proxy.h
>@@ -15,6 +15,7 @@
> #include <memory>
> #include <vector>
> 
>+#include "api/video_codecs/sdp_video_format.h"
> #include "api/video_codecs/video_encoder_factory.h"
> #include "modules/video_coding/codecs/vp8/include/vp8.h"
> 
>@@ -24,8 +25,13 @@ namespace webrtc {
> // doesn't support simulcast for provided settings.
> class VP8EncoderSimulcastProxy : public VP8Encoder {
>  public:
>+  VP8EncoderSimulcastProxy(VideoEncoderFactory* factory,
>+                           const SdpVideoFormat& format);
>+  // Deprecated. Remove once all clients use constructor with both factory and
>+  // SdpVideoFormat;
>   explicit VP8EncoderSimulcastProxy(VideoEncoderFactory* factory);
>-  virtual ~VP8EncoderSimulcastProxy();
>+
>+  ~VP8EncoderSimulcastProxy() override;
> 
>   // Implements VideoEncoder.
>   int Release() override;
>@@ -37,17 +43,17 @@ class VP8EncoderSimulcastProxy : public VP8Encoder {
>              const std::vector<FrameType>* frame_types) override;
>   int RegisterEncodeCompleteCallback(EncodedImageCallback* callback) override;
>   int SetChannelParameters(uint32_t packet_loss, int64_t rtt) override;
>-  int SetRateAllocation(const BitrateAllocation& bitrate,
>+  int SetRateAllocation(const VideoBitrateAllocation& bitrate,
>                         uint32_t new_framerate) override;
> 
>   VideoEncoder::ScalingSettings GetScalingSettings() const override;
> 
>-  int32_t SetPeriodicKeyFrames(bool enable) override;
>   bool SupportsNativeHandle() const override;
>   const char* ImplementationName() const override;
> 
>  private:
>   VideoEncoderFactory* const factory_;
>+  SdpVideoFormat video_format_;
>   std::unique_ptr<VideoEncoder> encoder_;
>   EncodedImageCallback* callback_;
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/vp8_encoder_simulcast_proxy_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/vp8_encoder_simulcast_proxy_unittest.cc
>index 3e2761cfb60..dfa35cbaf6a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/vp8_encoder_simulcast_proxy_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/vp8_encoder_simulcast_proxy_unittest.cc
>@@ -63,8 +63,6 @@ TEST(VP8EncoderSimulcastProxy, ChoosesCorrectImplementation) {
>       "SimulcastEncoderAdapter (Fake, Fake, Fake)";
>   VideoCodec codec_settings;
>   webrtc::test::CodecSettings(kVideoCodecVP8, &codec_settings);
>-  TemporalLayersFactory tl_factory;
>-  codec_settings.VP8()->tl_factory = &tl_factory;
>   codec_settings.simulcastStream[0] = {
>       test::kTestWidth, test::kTestHeight, 2, 2000, 1000, 1000, 56};
>   codec_settings.simulcastStream[1] = {
>@@ -85,7 +83,8 @@ TEST(VP8EncoderSimulcastProxy, ChoosesCorrectImplementation) {
>       .Times(1)
>       .WillOnce(Return(mock_encoder));
> 
>-  VP8EncoderSimulcastProxy simulcast_enabled_proxy(&simulcast_factory);
>+  VP8EncoderSimulcastProxy simulcast_enabled_proxy(&simulcast_factory,
>+                                                   SdpVideoFormat("VP8"));
>   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>             simulcast_enabled_proxy.InitEncode(&codec_settings, 4, 1200));
>   EXPECT_EQ(kImplementationName, simulcast_enabled_proxy.ImplementationName());
>@@ -124,7 +123,8 @@ TEST(VP8EncoderSimulcastProxy, ChoosesCorrectImplementation) {
>       .WillOnce(Return(mock_encoder3))
>       .WillOnce(Return(mock_encoder4));
> 
>-  VP8EncoderSimulcastProxy simulcast_disabled_proxy(&nonsimulcast_factory);
>+  VP8EncoderSimulcastProxy simulcast_disabled_proxy(&nonsimulcast_factory,
>+                                                    SdpVideoFormat("VP8"));
>   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>             simulcast_disabled_proxy.InitEncode(&codec_settings, 4, 1200));
>   EXPECT_EQ(kSimulcastAdaptedImplementationName,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcmediaengine.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcmediaengine.cc
>index 6c6464afcd9..82badc10440 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcmediaengine.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcmediaengine.cc
>@@ -27,6 +27,7 @@
> 
> namespace cricket {
> 
>+#if defined(USE_BUILTIN_SW_CODECS)
> namespace {
> 
> MediaEngineInterface* CreateWebRtcMediaEngine(
>@@ -86,6 +87,7 @@ MediaEngineInterface* WebRtcMediaEngineFactory::Create(
>       adm, audio_encoder_factory, audio_decoder_factory, video_encoder_factory,
>       video_decoder_factory, audio_mixer, audio_processing);
> }
>+#endif
> 
> std::unique_ptr<MediaEngineInterface> WebRtcMediaEngineFactory::Create(
>     rtc::scoped_refptr<webrtc::AudioDeviceModule> adm,
>@@ -173,12 +175,12 @@ std::vector<webrtc::RtpExtension> FilterRtpExtensions(
>   // Sort by name, ascending (prioritise encryption), so that we don't reset
>   // extensions if they were specified in a different order (also allows us
>   // to use std::unique below).
>-  std::sort(result.begin(), result.end(),
>-            [](const webrtc::RtpExtension& rhs,
>-               const webrtc::RtpExtension& lhs) {
>-                return rhs.encrypt == lhs.encrypt ? rhs.uri < lhs.uri
>-                                                  : rhs.encrypt > lhs.encrypt;
>-              });
>+  std::sort(
>+      result.begin(), result.end(),
>+      [](const webrtc::RtpExtension& rhs, const webrtc::RtpExtension& lhs) {
>+        return rhs.encrypt == lhs.encrypt ? rhs.uri < lhs.uri
>+                                          : rhs.encrypt > lhs.encrypt;
>+      });
> 
>   // Remove unnecessary extensions (used on send side).
>   if (filter_redundant_extensions) {
>@@ -200,9 +202,8 @@ std::vector<webrtc::RtpExtension> FilterRtpExtensions(
>   return result;
> }
> 
>-webrtc::Call::Config::BitrateConfig GetBitrateConfigForCodec(
>-    const Codec& codec) {
>-  webrtc::Call::Config::BitrateConfig config;
>+webrtc::BitrateConstraints GetBitrateConfigForCodec(const Codec& codec) {
>+  webrtc::BitrateConstraints config;
>   int bitrate_kbps = 0;
>   if (codec.GetParam(kCodecParamMinBitrate, &bitrate_kbps) &&
>       bitrate_kbps > 0) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcmediaengine.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcmediaengine.h
>index 632e7304782..b9927b1a9d2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcmediaengine.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcmediaengine.h
>@@ -25,11 +25,11 @@ class AudioMixer;
> class AudioProcessing;
> class VideoDecoderFactory;
> class VideoEncoderFactory;
>-}
>+}  // namespace webrtc
> namespace cricket {
> class WebRtcVideoDecoderFactory;
> class WebRtcVideoEncoderFactory;
>-}
>+}  // namespace cricket
> 
> namespace cricket {
> 
>@@ -85,8 +85,7 @@ std::vector<webrtc::RtpExtension> FilterRtpExtensions(
>     bool (*supported)(const std::string&),
>     bool filter_redundant_extensions);
> 
>-webrtc::Call::Config::BitrateConfig GetBitrateConfigForCodec(
>-    const Codec& codec);
>+webrtc::BitrateConstraints GetBitrateConfigForCodec(const Codec& codec);
> 
> }  // namespace cricket
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcmediaengine_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcmediaengine_unittest.cc
>index 85170e24e5c..446ad24e77c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcmediaengine_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcmediaengine_unittest.cc
>@@ -12,7 +12,10 @@
> 
> #include "api/audio_codecs/builtin_audio_decoder_factory.h"
> #include "api/audio_codecs/builtin_audio_encoder_factory.h"
>+#include "api/video_codecs/builtin_video_decoder_factory.h"
>+#include "api/video_codecs/builtin_video_encoder_factory.h"
> #include "media/engine/webrtcmediaengine.h"
>+#include "modules/audio_processing/include/audio_processing.h"
> #include "test/gtest.h"
> 
> using webrtc::RtpExtension;
>@@ -101,14 +104,14 @@ TEST(WebRtcMediaEngineTest, FilterRtpExtensions_EmptyList) {
>   std::vector<RtpExtension> extensions;
>   std::vector<webrtc::RtpExtension> filtered =
>       FilterRtpExtensions(extensions, SupportedExtensions1, true);
>-  EXPECT_EQ(0, filtered.size());
>+  EXPECT_EQ(0u, filtered.size());
> }
> 
> TEST(WebRtcMediaEngineTest, FilterRtpExtensions_IncludeOnlySupported) {
>   std::vector<RtpExtension> extensions = MakeUniqueExtensions();
>   std::vector<webrtc::RtpExtension> filtered =
>       FilterRtpExtensions(extensions, SupportedExtensions1, false);
>-  EXPECT_EQ(2, filtered.size());
>+  EXPECT_EQ(2u, filtered.size());
>   EXPECT_EQ("c", filtered[0].uri);
>   EXPECT_EQ("i", filtered[1].uri);
> }
>@@ -117,7 +120,7 @@ TEST(WebRtcMediaEngineTest, FilterRtpExtensions_SortedByName_1) {
>   std::vector<RtpExtension> extensions = MakeUniqueExtensions();
>   std::vector<webrtc::RtpExtension> filtered =
>       FilterRtpExtensions(extensions, SupportedExtensions2, false);
>-  EXPECT_EQ(12, filtered.size());
>+  EXPECT_EQ(12u, filtered.size());
>   EXPECT_TRUE(IsSorted(filtered));
> }
> 
>@@ -125,7 +128,7 @@ TEST(WebRtcMediaEngineTest, FilterRtpExtensions_SortedByName_2) {
>   std::vector<RtpExtension> extensions = MakeUniqueExtensions();
>   std::vector<webrtc::RtpExtension> filtered =
>       FilterRtpExtensions(extensions, SupportedExtensions2, true);
>-  EXPECT_EQ(12, filtered.size());
>+  EXPECT_EQ(12u, filtered.size());
>   EXPECT_TRUE(IsSorted(filtered));
> }
> 
>@@ -133,7 +136,7 @@ TEST(WebRtcMediaEngineTest, FilterRtpExtensions_DontRemoveRedundant) {
>   std::vector<RtpExtension> extensions = MakeRedundantExtensions();
>   std::vector<webrtc::RtpExtension> filtered =
>       FilterRtpExtensions(extensions, SupportedExtensions2, false);
>-  EXPECT_EQ(12, filtered.size());
>+  EXPECT_EQ(12u, filtered.size());
>   EXPECT_TRUE(IsSorted(filtered));
>   EXPECT_EQ(filtered[0].uri, filtered[1].uri);
> }
>@@ -142,7 +145,7 @@ TEST(WebRtcMediaEngineTest, FilterRtpExtensions_RemoveRedundant) {
>   std::vector<RtpExtension> extensions = MakeRedundantExtensions();
>   std::vector<webrtc::RtpExtension> filtered =
>       FilterRtpExtensions(extensions, SupportedExtensions2, true);
>-  EXPECT_EQ(6, filtered.size());
>+  EXPECT_EQ(6u, filtered.size());
>   EXPECT_TRUE(IsSorted(filtered));
>   EXPECT_NE(filtered[0].uri, filtered[1].uri);
> }
>@@ -155,7 +158,7 @@ TEST(WebRtcMediaEngineTest, FilterRtpExtensions_RemoveRedundantEncrypted_1) {
>   extensions.push_back(webrtc::RtpExtension("b", 4));
>   std::vector<webrtc::RtpExtension> filtered =
>       FilterRtpExtensions(extensions, SupportedExtensions2, true);
>-  EXPECT_EQ(3, filtered.size());
>+  EXPECT_EQ(3u, filtered.size());
>   EXPECT_TRUE(IsSorted(filtered));
>   EXPECT_EQ(filtered[0].uri, filtered[1].uri);
>   EXPECT_NE(filtered[0].encrypt, filtered[1].encrypt);
>@@ -171,7 +174,7 @@ TEST(WebRtcMediaEngineTest, FilterRtpExtensions_RemoveRedundantEncrypted_2) {
>   extensions.push_back(webrtc::RtpExtension("b", 4));
>   std::vector<webrtc::RtpExtension> filtered =
>       FilterRtpExtensions(extensions, SupportedExtensions2, true);
>-  EXPECT_EQ(3, filtered.size());
>+  EXPECT_EQ(3u, filtered.size());
>   EXPECT_TRUE(IsSorted(filtered));
>   EXPECT_EQ(filtered[0].uri, filtered[1].uri);
>   EXPECT_NE(filtered[0].encrypt, filtered[1].encrypt);
>@@ -190,7 +193,7 @@ TEST(WebRtcMediaEngineTest, FilterRtpExtensions_RemoveRedundantBwe_1) {
>   extensions.push_back(RtpExtension(RtpExtension::kTimestampOffsetUri, 14));
>   std::vector<webrtc::RtpExtension> filtered =
>       FilterRtpExtensions(extensions, SupportedExtensions2, true);
>-  EXPECT_EQ(1, filtered.size());
>+  EXPECT_EQ(1u, filtered.size());
>   EXPECT_EQ(RtpExtension::kTransportSequenceNumberUri, filtered[0].uri);
> }
> 
>@@ -209,7 +212,7 @@ TEST(WebRtcMediaEngineTest, FilterRtpExtensions_RemoveRedundantBweEncrypted_1) {
>   extensions.push_back(RtpExtension(RtpExtension::kTimestampOffsetUri, 14));
>   std::vector<webrtc::RtpExtension> filtered =
>       FilterRtpExtensions(extensions, SupportedExtensions2, true);
>-  EXPECT_EQ(2, filtered.size());
>+  EXPECT_EQ(2u, filtered.size());
>   EXPECT_EQ(RtpExtension::kTransportSequenceNumberUri, filtered[0].uri);
>   EXPECT_EQ(RtpExtension::kTransportSequenceNumberUri, filtered[1].uri);
>   EXPECT_NE(filtered[0].encrypt, filtered[1].encrypt);
>@@ -222,7 +225,7 @@ TEST(WebRtcMediaEngineTest, FilterRtpExtensions_RemoveRedundantBwe_2) {
>   extensions.push_back(RtpExtension(RtpExtension::kTimestampOffsetUri, 7));
>   std::vector<webrtc::RtpExtension> filtered =
>       FilterRtpExtensions(extensions, SupportedExtensions2, true);
>-  EXPECT_EQ(1, filtered.size());
>+  EXPECT_EQ(1u, filtered.size());
>   EXPECT_EQ(RtpExtension::kAbsSendTimeUri, filtered[0].uri);
> }
> 
>@@ -232,14 +235,17 @@ TEST(WebRtcMediaEngineTest, FilterRtpExtensions_RemoveRedundantBwe_3) {
>   extensions.push_back(RtpExtension(RtpExtension::kTimestampOffsetUri, 14));
>   std::vector<webrtc::RtpExtension> filtered =
>       FilterRtpExtensions(extensions, SupportedExtensions2, true);
>-  EXPECT_EQ(1, filtered.size());
>+  EXPECT_EQ(1u, filtered.size());
>   EXPECT_EQ(RtpExtension::kTimestampOffsetUri, filtered[0].uri);
> }
> 
> TEST(WebRtcMediaEngineFactoryTest, CreateWithBuiltinDecoders) {
>   std::unique_ptr<MediaEngineInterface> engine(WebRtcMediaEngineFactory::Create(
>-      nullptr, webrtc::CreateBuiltinAudioEncoderFactory(),
>-      webrtc::CreateBuiltinAudioDecoderFactory(), nullptr, nullptr));
>+      nullptr /* adm */, webrtc::CreateBuiltinAudioEncoderFactory(),
>+      webrtc::CreateBuiltinAudioDecoderFactory(),
>+      webrtc::CreateBuiltinVideoEncoderFactory(),
>+      webrtc::CreateBuiltinVideoDecoderFactory(), nullptr /* audio_mixer */,
>+      webrtc::AudioProcessingBuilder().Create()));
>   EXPECT_TRUE(engine);
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideocapturer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideocapturer.cc
>index cf9eab6e6c5..f8a9f748a80 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideocapturer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideocapturer.cc
>@@ -20,7 +20,10 @@
> #include "rtc_base/timeutils.h"
> 
> #include "modules/video_capture/video_capture_factory.h"
>+
>+#if defined(WEBRTC_WIN)
> #include "rtc_base/win32.h"  // Need this to #include the impl files.
>+#endif                       // WEBRTC_WIN
> #include "system_wrappers/include/field_trial.h"
> 
> namespace cricket {
>@@ -323,8 +326,7 @@ bool WebRtcVideoCapturer::GetPreferredFourccs(std::vector<uint32_t>* fourccs) {
>   return true;
> }
> 
>-void WebRtcVideoCapturer::OnFrame(
>-    const webrtc::VideoFrame& sample) {
>+void WebRtcVideoCapturer::OnFrame(const webrtc::VideoFrame& sample) {
>   // This can only happen between Start() and Stop().
>   RTC_DCHECK(start_thread_);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideocapturer_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideocapturer_unittest.cc
>index 8770960c4c7..e98f1570e73 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideocapturer_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideocapturer_unittest.cc
>@@ -81,8 +81,7 @@ TEST_F(WebRtcVideoCapturerTest, TestInitVcm) {
> TEST_F(WebRtcVideoCapturerTest, TestCapture) {
>   EXPECT_TRUE(capturer_->Init(cricket::Device(kTestDeviceName, kTestDeviceId)));
>   cricket::VideoCapturerListener listener(capturer_.get());
>-  cricket::VideoFormat format(
>-      capturer_->GetSupportedFormats()->at(0));
>+  cricket::VideoFormat format(capturer_->GetSupportedFormats()->at(0));
>   EXPECT_EQ(cricket::CS_STARTING, capturer_->Start(format));
>   EXPECT_TRUE(capturer_->IsRunning());
>   ASSERT_TRUE(capturer_->GetCaptureFormat() != NULL);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideocapturerfactory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideocapturerfactory.cc
>index c948101ea89..333df6f165a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideocapturerfactory.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideocapturerfactory.cc
>@@ -19,8 +19,7 @@ namespace cricket {
> std::unique_ptr<VideoCapturer> WebRtcVideoDeviceCapturerFactory::Create(
>     const Device& device) {
> #ifdef HAVE_WEBRTC_VIDEO
>-  std::unique_ptr<WebRtcVideoCapturer> capturer(
>-      new WebRtcVideoCapturer());
>+  std::unique_ptr<WebRtcVideoCapturer> capturer(new WebRtcVideoCapturer());
>   if (!capturer->Init(device)) {
>     return std::unique_ptr<VideoCapturer>();
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideoengine.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideoengine.cc
>index d6c51d1f327..3a72cde891e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideoengine.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideoengine.cc
>@@ -25,7 +25,9 @@
> #include "call/call.h"
> #include "common_video/h264/profile_level_id.h"
> #include "media/engine/constants.h"
>-#include "media/engine/convert_legacy_video_factory.h"
>+#if defined(USE_BUILTIN_SW_CODECS)
>+#include "media/engine/convert_legacy_video_factory.h"  // nogncheck
>+#endif
> #include "media/engine/simulcast.h"
> #include "media/engine/webrtcmediaengine.h"
> #include "media/engine/webrtcvoiceengine.h"
>@@ -37,8 +39,6 @@
> #include "rtc_base/trace_event.h"
> #include "system_wrappers/include/field_trial.h"
> 
>-using DegradationPreference = webrtc::VideoSendStream::DegradationPreference;
>-
> namespace cricket {
> 
> // Hack in order to pass in |receive_stream_id| to legacy clients.
>@@ -46,6 +46,7 @@ namespace cricket {
> // webrtc:7925 is fixed.
> class DecoderFactoryAdapter {
>  public:
>+#if defined(USE_BUILTIN_SW_CODECS)
>   explicit DecoderFactoryAdapter(
>       std::unique_ptr<WebRtcVideoDecoderFactory> external_video_decoder_factory)
>       : cricket_decoder_with_params_(new CricketDecoderWithParams(
>@@ -53,6 +54,7 @@ class DecoderFactoryAdapter {
>         decoder_factory_(ConvertVideoDecoderFactory(
>             std::unique_ptr<WebRtcVideoDecoderFactory>(
>                 cricket_decoder_with_params_))) {}
>+#endif
> 
>   explicit DecoderFactoryAdapter(
>       std::unique_ptr<webrtc::VideoDecoderFactory> video_decoder_factory)
>@@ -136,7 +138,6 @@ class NullVideoDecoder : public webrtc::VideoDecoder {
> 
>   int32_t Decode(const webrtc::EncodedImage& input_image,
>                  bool missing_frames,
>-                 const webrtc::RTPFragmentationHeader* fragmentation,
>                  const webrtc::CodecSpecificInfo* codec_specific_info,
>                  int64_t render_time_ms) override {
>     RTC_LOG(LS_ERROR) << "The NullVideoDecoder doesn't support decoding.";
>@@ -186,7 +187,6 @@ void AddDefaultFeedbackParams(VideoCodec* codec) {
>   codec->AddFeedbackParam(FeedbackParam(kRtcpFbParamNack, kRtcpFbNackParamPli));
> }
> 
>-
> // This function will assign dynamic payload types (in the range [96, 127]) to
> // the input codecs, and also add ULPFEC, RED, FlexFEC, and associated RTX
> // codecs for recognized codecs (VP8, VP9, H264, and RED). It will also add
>@@ -221,23 +221,23 @@ std::vector<VideoCodec> AssignPayloadTypesAndDefaultCodecs(
> 
>     // Increment payload type.
>     ++payload_type;
>-    if (payload_type > kLastDynamicPayloadType)
>+    if (payload_type > kLastDynamicPayloadType) {
>+      RTC_LOG(LS_ERROR) << "Out of dynamic payload types, skipping the rest.";
>       break;
>+    }
> 
>-    // Add associated RTX codec for recognized codecs.
>-    // TODO(deadbeef): Should we add RTX codecs for external codecs whose names
>-    // we don't recognize?
>-    if (CodecNamesEq(codec.name, kVp8CodecName) ||
>-        CodecNamesEq(codec.name, kVp9CodecName) ||
>-        CodecNamesEq(codec.name, kH264CodecName) ||
>-        CodecNamesEq(codec.name, kRedCodecName)) {
>+    // Add associated RTX codec for non-FEC codecs.
>+    if (!CodecNamesEq(codec.name, kUlpfecCodecName) &&
>+        !CodecNamesEq(codec.name, kFlexfecCodecName)) {
>       output_codecs.push_back(
>           VideoCodec::CreateRtxCodec(payload_type, codec.id));
> 
>       // Increment payload type.
>       ++payload_type;
>-      if (payload_type > kLastDynamicPayloadType)
>+      if (payload_type > kLastDynamicPayloadType) {
>+        RTC_LOG(LS_ERROR) << "Out of dynamic payload types, skipping the rest.";
>         break;
>+      }
>     }
>   }
>   return output_codecs;
>@@ -318,8 +318,10 @@ static bool ValidateStreamParams(const StreamParams& sp) {
> 
> // Returns true if the given codec is disallowed from doing simulcast.
> bool IsCodecBlacklistedForSimulcast(const std::string& codec_name) {
>-  return CodecNamesEq(codec_name, kH264CodecName) ||
>-         CodecNamesEq(codec_name, kVp9CodecName);
>+  return webrtc::field_trial::IsEnabled("WebRTC-H264Simulcast")
>+    ? CodecNamesEq(codec_name, kVp9CodecName)
>+    : CodecNamesEq(codec_name, kH264CodecName) ||
>+      CodecNamesEq(codec_name, kVp9CodecName);
> }
> 
> // The selected thresholds for QVGA and VGA corresponded to a QP around 10.
>@@ -336,67 +338,67 @@ static int GetMaxDefaultVideoBitrateKbps(int width, int height) {
>   }
> }
> 
>-bool GetVp9LayersFromFieldTrialGroup(int* num_spatial_layers,
>-                                     int* num_temporal_layers) {
>+bool GetVp9LayersFromFieldTrialGroup(size_t* num_spatial_layers,
>+                                     size_t* num_temporal_layers) {
>   std::string group = webrtc::field_trial::FindFullName("WebRTC-SupportVP9SVC");
>   if (group.empty())
>     return false;
> 
>-  if (sscanf(group.c_str(), "EnabledByFlag_%dSL%dTL", num_spatial_layers,
>+  if (sscanf(group.c_str(), "EnabledByFlag_%zuSL%zuTL", num_spatial_layers,
>              num_temporal_layers) != 2) {
>     return false;
>   }
>-  const int kMaxSpatialLayers = 2;
>+  const size_t kMaxSpatialLayers = 3;
>   if (*num_spatial_layers > kMaxSpatialLayers || *num_spatial_layers < 1)
>     return false;
> 
>-  const int kMaxTemporalLayers = 3;
>+  const size_t kMaxTemporalLayers = 3;
>   if (*num_temporal_layers > kMaxTemporalLayers || *num_temporal_layers < 1)
>     return false;
> 
>   return true;
> }
> 
>-int GetDefaultVp9SpatialLayers() {
>-  int num_sl;
>-  int num_tl;
>+absl::optional<size_t> GetVp9SpatialLayersFromFieldTrial() {
>+  size_t num_sl;
>+  size_t num_tl;
>   if (GetVp9LayersFromFieldTrialGroup(&num_sl, &num_tl)) {
>     return num_sl;
>   }
>-  return 1;
>+  return absl::nullopt;
> }
> 
>-int GetDefaultVp9TemporalLayers() {
>-  int num_sl;
>-  int num_tl;
>+absl::optional<size_t> GetVp9TemporalLayersFromFieldTrial() {
>+  size_t num_sl;
>+  size_t num_tl;
>   if (GetVp9LayersFromFieldTrialGroup(&num_sl, &num_tl)) {
>     return num_tl;
>   }
>-  return 1;
>+  return absl::nullopt;
> }
> 
> const char kForcedFallbackFieldTrial[] =
>     "WebRTC-VP8-Forced-Fallback-Encoder-v2";
> 
>-rtc::Optional<int> GetFallbackMinBpsFromFieldTrial() {
>+absl::optional<int> GetFallbackMinBpsFromFieldTrial() {
>   if (!webrtc::field_trial::IsEnabled(kForcedFallbackFieldTrial))
>-    return rtc::nullopt;
>+    return absl::nullopt;
> 
>   std::string group =
>       webrtc::field_trial::FindFullName(kForcedFallbackFieldTrial);
>   if (group.empty())
>-    return rtc::nullopt;
>+    return absl::nullopt;
> 
>   int min_pixels;
>   int max_pixels;
>   int min_bps;
>   if (sscanf(group.c_str(), "Enabled-%d,%d,%d", &min_pixels, &max_pixels,
>              &min_bps) != 3) {
>-    return rtc::nullopt;
>+    return absl::nullopt;
>   }
> 
>   if (min_bps <= 0)
>-    return rtc::nullopt;
>+    return absl::nullopt;
> 
>   return min_bps;
> }
>@@ -454,17 +456,32 @@ WebRtcVideoChannel::WebRtcVideoSendStream::ConfigureVideoEncoderSettings(
>   if (CodecNamesEq(codec.name, kVp9CodecName)) {
>     webrtc::VideoCodecVP9 vp9_settings =
>         webrtc::VideoEncoder::GetDefaultVp9Settings();
>-    if (is_screencast) {
>-      // TODO(asapersson): Set to 2 for now since there is a DCHECK in
>-      // VideoSendStream::ReconfigureVideoEncoder.
>-      vp9_settings.numberOfSpatialLayers = 2;
>-    } else {
>-      vp9_settings.numberOfSpatialLayers = GetDefaultVp9SpatialLayers();
>-    }
>+    const size_t default_num_spatial_layers =
>+        parameters_.config.rtp.ssrcs.size();
>+    const size_t num_spatial_layers =
>+        GetVp9SpatialLayersFromFieldTrial().value_or(
>+            default_num_spatial_layers);
>+
>+    const size_t default_num_temporal_layers =
>+        num_spatial_layers > 1 ? kConferenceDefaultNumTemporalLayers : 1;
>+    const size_t num_temporal_layers =
>+        GetVp9TemporalLayersFromFieldTrial().value_or(
>+            default_num_temporal_layers);
>+
>+    vp9_settings.numberOfSpatialLayers = std::min<unsigned char>(
>+        num_spatial_layers, kConferenceMaxNumSpatialLayers);
>+    vp9_settings.numberOfTemporalLayers = std::min<unsigned char>(
>+        num_temporal_layers, kConferenceMaxNumTemporalLayers);
>+
>     // VP9 denoising is disabled by default.
>     vp9_settings.denoisingOn = codec_default_denoising ? true : denoising;
>-    vp9_settings.frameDroppingOn = frame_dropping;
>     vp9_settings.automaticResizeOn = automatic_resize;
>+    // Ensure frame dropping is always enabled.
>+    RTC_DCHECK(vp9_settings.frameDroppingOn);
>+    if (!is_screencast) {
>+      // Limit inter-layer prediction to key pictures.
>+      vp9_settings.interLayerPred = webrtc::InterLayerPredMode::kOnKeyPic;
>+    }
>     return new rtc::RefCountedObject<
>         webrtc::VideoEncoderConfig::Vp9EncoderSpecificSettings>(vp9_settings);
>   }
>@@ -477,7 +494,7 @@ DefaultUnsignalledSsrcHandler::DefaultUnsignalledSsrcHandler()
> UnsignalledSsrcHandler::Action DefaultUnsignalledSsrcHandler::OnUnsignalledSsrc(
>     WebRtcVideoChannel* channel,
>     uint32_t ssrc) {
>-  rtc::Optional<uint32_t> default_recv_ssrc =
>+  absl::optional<uint32_t> default_recv_ssrc =
>       channel->GetDefaultReceiveStreamSsrc();
> 
>   if (default_recv_ssrc) {
>@@ -486,8 +503,9 @@ UnsignalledSsrcHandler::Action DefaultUnsignalledSsrcHandler::OnUnsignalledSsrc(
>     channel->RemoveRecvStream(*default_recv_ssrc);
>   }
> 
>-  StreamParams sp;
>+  StreamParams sp = channel->unsignaled_stream_params();
>   sp.ssrcs.push_back(ssrc);
>+
>   RTC_LOG(LS_INFO) << "Creating default receive stream for SSRC=" << ssrc
>                    << ".";
>   if (!channel->AddRecvStream(sp, true)) {
>@@ -507,13 +525,14 @@ void DefaultUnsignalledSsrcHandler::SetDefaultSink(
>     WebRtcVideoChannel* channel,
>     rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) {
>   default_sink_ = sink;
>-  rtc::Optional<uint32_t> default_recv_ssrc =
>+  absl::optional<uint32_t> default_recv_ssrc =
>       channel->GetDefaultReceiveStreamSsrc();
>   if (default_recv_ssrc) {
>     channel->SetSink(*default_recv_ssrc, default_sink_);
>   }
> }
> 
>+#if defined(USE_BUILTIN_SW_CODECS)
> WebRtcVideoEngine::WebRtcVideoEngine(
>     std::unique_ptr<WebRtcVideoEncoderFactory> external_video_encoder_factory,
>     std::unique_ptr<WebRtcVideoDecoderFactory> external_video_decoder_factory)
>@@ -523,6 +542,7 @@ WebRtcVideoEngine::WebRtcVideoEngine(
>           std::move(external_video_encoder_factory))) {
>   RTC_LOG(LS_INFO) << "WebRtcVideoEngine::WebRtcVideoEngine()";
> }
>+#endif
> 
> WebRtcVideoEngine::WebRtcVideoEngine(
>     std::unique_ptr<webrtc::VideoEncoderFactory> video_encoder_factory,
>@@ -571,8 +591,12 @@ RtpCapabilities WebRtcVideoEngine::GetCapabilities() const {
>       webrtc::RtpExtension(webrtc::RtpExtension::kVideoContentTypeUri,
>                            webrtc::RtpExtension::kVideoContentTypeDefaultId));
>   capabilities.header_extensions.push_back(
>-        webrtc::RtpExtension(webrtc::RtpExtension::kVideoTimingUri,
>-                             webrtc::RtpExtension::kVideoTimingDefaultId));
>+      webrtc::RtpExtension(webrtc::RtpExtension::kVideoTimingUri,
>+                           webrtc::RtpExtension::kVideoTimingDefaultId));
>+  // TODO(bugs.webrtc.org/4050): Add MID header extension as capability once MID
>+  // demuxing is completed.
>+  // capabilities.header_extensions.push_back(webrtc::RtpExtension(
>+  //     webrtc::RtpExtension::kMidUri, webrtc::RtpExtension::kMidDefaultId));
>   return capabilities;
> }
> 
>@@ -606,7 +630,7 @@ WebRtcVideoChannel::~WebRtcVideoChannel() {
>     delete kv.second;
> }
> 
>-rtc::Optional<WebRtcVideoChannel::VideoCodecSettings>
>+absl::optional<WebRtcVideoChannel::VideoCodecSettings>
> WebRtcVideoChannel::SelectSendVideoCodec(
>     const std::vector<VideoCodecSettings>& remote_mapped_codecs) const {
>   const std::vector<VideoCodec> local_supported_codecs =
>@@ -622,7 +646,7 @@ WebRtcVideoChannel::SelectSendVideoCodec(
>       return remote_mapped_codec;
>   }
>   // No remote codec was supported.
>-  return rtc::nullopt;
>+  return absl::nullopt;
> }
> 
> bool WebRtcVideoChannel::NonFlexfecReceiveCodecsHaveChanged(
>@@ -640,10 +664,10 @@ bool WebRtcVideoChannel::NonFlexfecReceiveCodecsHaveChanged(
>   // to support munging the SDP in this way without recreating receive
>   // streams, we ignore the order of the received codecs so that
>   // changing the order doesn't cause this "blink".
>-  auto comparison =
>-      [](const VideoCodecSettings& codec1, const VideoCodecSettings& codec2) {
>-        return codec1.codec.id > codec2.codec.id;
>-      };
>+  auto comparison = [](const VideoCodecSettings& codec1,
>+                       const VideoCodecSettings& codec2) {
>+    return codec1.codec.id > codec2.codec.id;
>+  };
>   std::sort(before.begin(), before.end(), comparison);
>   std::sort(after.begin(), after.end(), comparison);
> 
>@@ -663,7 +687,7 @@ bool WebRtcVideoChannel::GetChangedSendParameters(
>   }
> 
>   // Select one of the remote codecs that will be used as send codec.
>-  rtc::Optional<VideoCodecSettings> selected_send_codec =
>+  absl::optional<VideoCodecSettings> selected_send_codec =
>       SelectSendVideoCodec(MapCodecs(params.codecs));
> 
>   if (!selected_send_codec) {
>@@ -689,7 +713,11 @@ bool WebRtcVideoChannel::GetChangedSendParameters(
>       params.extensions, webrtc::RtpExtension::IsSupportedForVideo, true);
>   if (!send_rtp_extensions_ || (*send_rtp_extensions_ != filtered_extensions)) {
>     changed_params->rtp_header_extensions =
>-        rtc::Optional<std::vector<webrtc::RtpExtension>>(filtered_extensions);
>+        absl::optional<std::vector<webrtc::RtpExtension>>(filtered_extensions);
>+  }
>+
>+  if (params.mid != send_params_.mid) {
>+    changed_params->mid = params.mid;
>   }
> 
>   // Handle max bitrate.
>@@ -765,12 +793,13 @@ bool WebRtcVideoChannel::SetSendParameters(const VideoSendParameters& params) {
>       // codec target bitrate.
>       // TODO(pbos): Figure out whether b=AS means max bitrate for this
>       // WebRtcVideoChannel (in which case we're good), or per sender (SSRC),
>-      // in which case this should not set a Call::BitrateConfig but rather
>+      // in which case this should not set a BitrateConstraints but rather
>       // reconfigure all senders.
>       bitrate_config_.max_bitrate_bps =
>           params.max_bandwidth_bps == 0 ? -1 : params.max_bandwidth_bps;
>     }
>-    call_->SetBitrateConfig(bitrate_config_);
>+    call_->GetTransportControllerSend()->SetSdpBitrateParameters(
>+        bitrate_config_);
>   }
> 
>   {
>@@ -816,7 +845,7 @@ webrtc::RtpParameters WebRtcVideoChannel::GetRtpSendParameters(
>   return rtp_params;
> }
> 
>-bool WebRtcVideoChannel::SetRtpSendParameters(
>+webrtc::RTCError WebRtcVideoChannel::SetRtpSendParameters(
>     uint32_t ssrc,
>     const webrtc::RtpParameters& parameters) {
>   TRACE_EVENT0("webrtc", "WebRtcVideoChannel::SetRtpSendParameters");
>@@ -825,7 +854,7 @@ bool WebRtcVideoChannel::SetRtpSendParameters(
>   if (it == send_streams_.end()) {
>     RTC_LOG(LS_ERROR) << "Attempting to set RTP send parameters for stream "
>                       << "with ssrc " << ssrc << " which doesn't exist.";
>-    return false;
>+    return webrtc::RTCError(webrtc::RTCErrorType::INTERNAL_ERROR);
>   }
> 
>   // TODO(deadbeef): Handle setting parameters with a list of codecs in a
>@@ -834,7 +863,7 @@ bool WebRtcVideoChannel::SetRtpSendParameters(
>   if (current_parameters.codecs != parameters.codecs) {
>     RTC_LOG(LS_ERROR) << "Using SetParameters to change the set of codecs "
>                       << "is not currently supported.";
>-    return false;
>+    return webrtc::RTCError(webrtc::RTCErrorType::INTERNAL_ERROR);
>   }
> 
>   return it->second->SetRtpParameters(parameters);
>@@ -862,15 +891,14 @@ webrtc::RtpParameters WebRtcVideoChannel::GetRtpReceiveParameters(
>           << "with SSRC " << ssrc << " which doesn't exist.";
>       return webrtc::RtpParameters();
>     }
>-    // TODO(deadbeef): Return stream-specific parameters, beyond just SSRC.
>-    rtp_params.encodings.emplace_back();
>-    rtp_params.encodings[0].ssrc = it->second->GetFirstPrimarySsrc();
>+    rtp_params = it->second->GetRtpParameters();
>   }
> 
>   // Add codecs, which any stream is prepared to receive.
>   for (const VideoCodec& codec : recv_params_.codecs) {
>     rtp_params.codecs.push_back(codec.ToCodecParameters());
>   }
>+
>   return rtp_params;
> }
> 
>@@ -938,7 +966,7 @@ bool WebRtcVideoChannel::GetChangedRecvParameters(
> 
>   if (NonFlexfecReceiveCodecsHaveChanged(recv_codecs_, mapped_codecs)) {
>     changed_params->codec_settings =
>-        rtc::Optional<std::vector<VideoCodecSettings>>(mapped_codecs);
>+        absl::optional<std::vector<VideoCodecSettings>>(mapped_codecs);
>   }
> 
>   // Handle RTP header extensions.
>@@ -946,7 +974,7 @@ bool WebRtcVideoChannel::GetChangedRecvParameters(
>       params.extensions, webrtc::RtpExtension::IsSupportedForVideo, false);
>   if (filtered_extensions != recv_rtp_extensions_) {
>     changed_params->rtp_header_extensions =
>-        rtc::Optional<std::vector<webrtc::RtpExtension>>(filtered_extensions);
>+        absl::optional<std::vector<webrtc::RtpExtension>>(filtered_extensions);
>   }
> 
>   int flexfec_payload_type = mapped_codecs.front().flexfec_payload_type;
>@@ -1031,18 +1059,13 @@ bool WebRtcVideoChannel::SetSend(bool send) {
>   return true;
> }
> 
>-// TODO(nisse): The enable argument was used for mute logic which has
>-// been moved to VideoBroadcaster. So remove the argument from this
>-// method.
> bool WebRtcVideoChannel::SetVideoSend(
>     uint32_t ssrc,
>-    bool enable,
>     const VideoOptions* options,
>     rtc::VideoSourceInterface<webrtc::VideoFrame>* source) {
>   TRACE_EVENT0("webrtc", "SetVideoSend");
>   RTC_DCHECK(ssrc != 0);
>-  RTC_LOG(LS_INFO) << "SetVideoSend (ssrc= " << ssrc << ", enable = " << enable
>-                   << ", options: "
>+  RTC_LOG(LS_INFO) << "SetVideoSend (ssrc= " << ssrc << ", options: "
>                    << (options ? options->ToString() : "nullptr")
>                    << ", source = " << (source ? "(source)" : "nullptr") << ")";
> 
>@@ -1055,7 +1078,7 @@ bool WebRtcVideoChannel::SetVideoSend(
>     return false;
>   }
> 
>-  return kv->second->SetVideoSend(enable, options, source);
>+  return kv->second->SetVideoSend(options, source);
> }
> 
> bool WebRtcVideoChannel::ValidateSendSsrcAvailability(
>@@ -1099,11 +1122,14 @@ bool WebRtcVideoChannel::AddSendStream(const StreamParams& sp) {
>   config.suspend_below_min_bitrate = video_config_.suspend_below_min_bitrate;
>   config.periodic_alr_bandwidth_probing =
>       video_config_.periodic_alr_bandwidth_probing;
>+  config.encoder_settings.experiment_cpu_load_estimator =
>+      video_config_.experiment_cpu_load_estimator;
>+  config.encoder_settings.encoder_factory = encoder_factory_;
>+
>   WebRtcVideoSendStream* stream = new WebRtcVideoSendStream(
>-      call_, sp, std::move(config), default_send_options_, encoder_factory_,
>-      video_config_.enable_cpu_overuse_detection,
>-      bitrate_config_.max_bitrate_bps, send_codec_, send_rtp_extensions_,
>-      send_params_);
>+      call_, sp, std::move(config), default_send_options_,
>+      video_config_.enable_cpu_adaptation, bitrate_config_.max_bitrate_bps,
>+      send_codec_, send_rtp_extensions_, send_params_);
> 
>   uint32_t ssrc = sp.first_ssrc();
>   RTC_DCHECK(ssrc != 0);
>@@ -1179,6 +1205,13 @@ bool WebRtcVideoChannel::AddRecvStream(const StreamParams& sp,
>   RTC_LOG(LS_INFO) << "AddRecvStream"
>                    << (default_stream ? " (default stream)" : "") << ": "
>                    << sp.ToString();
>+  if (!sp.has_ssrcs()) {
>+    // This is a StreamParam with unsignaled SSRCs. Store it, so it can be used
>+    // later when we know the SSRC on the first packet arrival.
>+    unsignaled_stream_params_ = sp;
>+    return true;
>+  }
>+
>   if (!ValidateStreamParams(sp))
>     return false;
> 
>@@ -1208,9 +1241,12 @@ bool WebRtcVideoChannel::AddRecvStream(const StreamParams& sp,
>   webrtc::FlexfecReceiveStream::Config flexfec_config(this);
>   ConfigureReceiverRtp(&config, &flexfec_config, sp);
> 
>+  // TODO(nisse): Rename config variable to avoid negation.
>   config.disable_prerenderer_smoothing =
>-      video_config_.disable_prerenderer_smoothing;
>-  config.sync_group = sp.sync_label;
>+      !video_config_.enable_prerenderer_smoothing;
>+  if (!sp.stream_ids().empty()) {
>+    config.sync_group = sp.stream_ids()[0];
>+  }
> 
>   receive_streams_[ssrc] = new WebRtcVideoReceiveStream(
>       call_, sp, std::move(config), decoder_factory_, default_stream,
>@@ -1274,8 +1310,10 @@ void WebRtcVideoChannel::ConfigureReceiverRtp(
> bool WebRtcVideoChannel::RemoveRecvStream(uint32_t ssrc) {
>   RTC_LOG(LS_INFO) << "RemoveRecvStream: " << ssrc;
>   if (ssrc == 0) {
>-    RTC_LOG(LS_ERROR) << "RemoveRecvStream with 0 ssrc is not supported.";
>-    return false;
>+    // This indicates that we need to remove the unsignaled stream parameters
>+    // that are cached.
>+    unsignaled_stream_params_ = StreamParams();
>+    return true;
>   }
> 
>   rtc::CritScope stream_lock(&stream_crit_);
>@@ -1346,7 +1384,7 @@ bool WebRtcVideoChannel::GetStats(VideoMediaInfo* info) {
> }
> 
> void WebRtcVideoChannel::FillSenderStats(VideoMediaInfo* video_media_info,
>-                                          bool log_stats) {
>+                                         bool log_stats) {
>   rtc::CritScope stream_lock(&stream_crit_);
>   for (std::map<uint32_t, WebRtcVideoSendStream*>::iterator it =
>            send_streams_.begin();
>@@ -1357,7 +1395,7 @@ void WebRtcVideoChannel::FillSenderStats(VideoMediaInfo* video_media_info,
> }
> 
> void WebRtcVideoChannel::FillReceiverStats(VideoMediaInfo* video_media_info,
>-                                            bool log_stats) {
>+                                           bool log_stats) {
>   rtc::CritScope stream_lock(&stream_crit_);
>   for (std::map<uint32_t, WebRtcVideoReceiveStream*>::iterator it =
>            receive_streams_.begin();
>@@ -1390,14 +1428,11 @@ void WebRtcVideoChannel::FillSendAndReceiveCodecStats(
>   }
> }
> 
>-void WebRtcVideoChannel::OnPacketReceived(
>-    rtc::CopyOnWriteBuffer* packet,
>-    const rtc::PacketTime& packet_time) {
>-  const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp,
>-                                              packet_time.not_before);
>+void WebRtcVideoChannel::OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
>+                                          const rtc::PacketTime& packet_time) {
>   const webrtc::PacketReceiver::DeliveryStatus delivery_result =
>       call_->Receiver()->DeliverPacket(webrtc::MediaType::VIDEO, *packet,
>-                                       webrtc_packet_time);
>+                                       packet_time.timestamp);
>   switch (delivery_result) {
>     case webrtc::PacketReceiver::DELIVERY_OK:
>       return;
>@@ -1441,24 +1476,21 @@ void WebRtcVideoChannel::OnPacketReceived(
>   }
> 
>   if (call_->Receiver()->DeliverPacket(webrtc::MediaType::VIDEO, *packet,
>-                                       webrtc_packet_time) !=
>+                                       packet_time.timestamp) !=
>       webrtc::PacketReceiver::DELIVERY_OK) {
>     RTC_LOG(LS_WARNING) << "Failed to deliver RTP packet on re-delivery.";
>     return;
>   }
> }
> 
>-void WebRtcVideoChannel::OnRtcpReceived(
>-    rtc::CopyOnWriteBuffer* packet,
>-    const rtc::PacketTime& packet_time) {
>-  const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp,
>-                                              packet_time.not_before);
>+void WebRtcVideoChannel::OnRtcpReceived(rtc::CopyOnWriteBuffer* packet,
>+                                        const rtc::PacketTime& packet_time) {
>   // TODO(pbos): Check webrtc::PacketReceiver::DELIVERY_OK once we deliver
>   // for both audio and video on the same path. Since BundleFilter doesn't
>   // filter RTCP anymore incoming RTCP packets could've been going to audio (so
>   // logging failures spam the log).
>   call_->Receiver()->DeliverPacket(webrtc::MediaType::VIDEO, *packet,
>-                                   webrtc_packet_time);
>+                                   packet_time.timestamp);
> }
> 
> void WebRtcVideoChannel::OnReadyToSend(bool ready) {
>@@ -1471,8 +1503,8 @@ void WebRtcVideoChannel::OnReadyToSend(bool ready) {
> void WebRtcVideoChannel::OnNetworkRouteChanged(
>     const std::string& transport_name,
>     const rtc::NetworkRoute& network_route) {
>-  // TODO(zhihaung): Merge these two callbacks.
>-  call_->OnNetworkRouteChanged(transport_name, network_route);
>+  call_->GetTransportControllerSend()->OnNetworkRouteChanged(transport_name,
>+                                                             network_route);
>   call_->OnTransportOverheadChanged(webrtc::MediaType::VIDEO,
>                                     network_route.packet_overhead);
> }
>@@ -1480,22 +1512,20 @@ void WebRtcVideoChannel::OnNetworkRouteChanged(
> void WebRtcVideoChannel::SetInterface(NetworkInterface* iface) {
>   MediaChannel::SetInterface(iface);
>   // Set the RTP recv/send buffer to a bigger size
>-  MediaChannel::SetOption(NetworkInterface::ST_RTP,
>-                          rtc::Socket::OPT_RCVBUF,
>+  MediaChannel::SetOption(NetworkInterface::ST_RTP, rtc::Socket::OPT_RCVBUF,
>                           kVideoRtpBufferSize);
> 
>   // Speculative change to increase the outbound socket buffer size.
>   // In b/15152257, we are seeing a significant number of packets discarded
>   // due to lack of socket buffer space, although it's not yet clear what the
>   // ideal value should be.
>-  MediaChannel::SetOption(NetworkInterface::ST_RTP,
>-                          rtc::Socket::OPT_SNDBUF,
>+  MediaChannel::SetOption(NetworkInterface::ST_RTP, rtc::Socket::OPT_SNDBUF,
>                           kVideoRtpBufferSize);
> }
> 
>-rtc::Optional<uint32_t> WebRtcVideoChannel::GetDefaultReceiveStreamSsrc() {
>+absl::optional<uint32_t> WebRtcVideoChannel::GetDefaultReceiveStreamSsrc() {
>   rtc::CritScope stream_lock(&stream_crit_);
>-  rtc::Optional<uint32_t> ssrc;
>+  absl::optional<uint32_t> ssrc;
>   for (auto it = receive_streams_.begin(); it != receive_streams_.end(); ++it) {
>     if (it->second->IsDefaultStream()) {
>       ssrc.emplace(it->first);
>@@ -1524,7 +1554,7 @@ WebRtcVideoChannel::WebRtcVideoSendStream::VideoSendStreamParameters::
>         webrtc::VideoSendStream::Config config,
>         const VideoOptions& options,
>         int max_bitrate_bps,
>-        const rtc::Optional<VideoCodecSettings>& codec_settings)
>+        const absl::optional<VideoCodecSettings>& codec_settings)
>     : config(std::move(config)),
>       options(options),
>       max_bitrate_bps(max_bitrate_bps),
>@@ -1536,11 +1566,10 @@ WebRtcVideoChannel::WebRtcVideoSendStream::WebRtcVideoSendStream(
>     const StreamParams& sp,
>     webrtc::VideoSendStream::Config config,
>     const VideoOptions& options,
>-    webrtc::VideoEncoderFactory* encoder_factory,
>     bool enable_cpu_overuse_detection,
>     int max_bitrate_bps,
>-    const rtc::Optional<VideoCodecSettings>& codec_settings,
>-    const rtc::Optional<std::vector<webrtc::RtpExtension>>& rtp_extensions,
>+    const absl::optional<VideoCodecSettings>& codec_settings,
>+    const absl::optional<std::vector<webrtc::RtpExtension>>& rtp_extensions,
>     // TODO(deadbeef): Don't duplicate information between send_params,
>     // rtp_extensions, options, etc.
>     const VideoSendParameters& send_params)
>@@ -1550,11 +1579,10 @@ WebRtcVideoChannel::WebRtcVideoSendStream::WebRtcVideoSendStream(
>       call_(call),
>       enable_cpu_overuse_detection_(enable_cpu_overuse_detection),
>       source_(nullptr),
>-      encoder_factory_(encoder_factory),
>       stream_(nullptr),
>       encoder_sink_(nullptr),
>       parameters_(std::move(config), options, max_bitrate_bps, codec_settings),
>-      rtp_parameters_(CreateRtpParametersWithOneEncoding()),
>+      rtp_parameters_(CreateRtpParametersWithEncodings(sp)),
>       sending_(false) {
>   parameters_.config.rtp.max_packet_size = kVideoMtu;
>   parameters_.conference_mode = send_params.conference_mode;
>@@ -1598,13 +1626,17 @@ WebRtcVideoChannel::WebRtcVideoSendStream::WebRtcVideoSendStream(
>   parameters_.config.track_id = sp.id;
>   if (rtp_extensions) {
>     parameters_.config.rtp.extensions = *rtp_extensions;
>+    rtp_parameters_.header_extensions = *rtp_extensions;
>   }
>   parameters_.config.rtp.rtcp_mode = send_params.rtcp.reduced_size
>                                          ? webrtc::RtcpMode::kReducedSize
>                                          : webrtc::RtcpMode::kCompound;
>+  parameters_.config.rtp.mid = send_params.mid;
>+
>+  rtp_parameters_.rtcp.reduced_size = send_params.rtcp.reduced_size;
>+
>   if (codec_settings) {
>-    bool force_encoder_allocation = false;
>-    SetCodec(*codec_settings, force_encoder_allocation);
>+    SetCodec(*codec_settings);
>   }
> }
> 
>@@ -1612,21 +1644,15 @@ WebRtcVideoChannel::WebRtcVideoSendStream::~WebRtcVideoSendStream() {
>   if (stream_ != NULL) {
>     call_->DestroyVideoSendStream(stream_);
>   }
>-  // Release |allocated_encoder_|.
>-  allocated_encoder_.reset();
> }
> 
> bool WebRtcVideoChannel::WebRtcVideoSendStream::SetVideoSend(
>-    bool enable,
>     const VideoOptions* options,
>     rtc::VideoSourceInterface<webrtc::VideoFrame>* source) {
>   TRACE_EVENT0("webrtc", "WebRtcVideoSendStream::SetVideoSend");
>   RTC_DCHECK_RUN_ON(&thread_checker_);
> 
>-  // Ignore |options| pointer if |enable| is false.
>-  bool options_present = enable && options;
>-
>-  if (options_present) {
>+  if (options) {
>     VideoOptions old_options = parameters_.options;
>     parameters_.options.SetAll(*options);
>     if (parameters_.options.is_screencast.value_or(false) !=
>@@ -1635,8 +1661,7 @@ bool WebRtcVideoChannel::WebRtcVideoSendStream::SetVideoSend(
>       // If screen content settings change, we may need to recreate the codec
>       // instance so that the correct type is used.
> 
>-      bool force_encoder_allocation = true;
>-      SetCodec(*parameters_.codec_settings, force_encoder_allocation);
>+      SetCodec(*parameters_.codec_settings);
>       // Mark screenshare parameter as being updated, then test for any other
>       // changes that may require codec reconfiguration.
>       old_options.is_screencast = options->is_screencast;
>@@ -1647,7 +1672,7 @@ bool WebRtcVideoChannel::WebRtcVideoSendStream::SetVideoSend(
>   }
> 
>   if (source_ && stream_) {
>-    stream_->SetSource(nullptr, DegradationPreference::kDegradationDisabled);
>+    stream_->SetSource(nullptr, webrtc::DegradationPreference::DISABLED);
>   }
>   // Switch to the new source.
>   source_ = source;
>@@ -1657,24 +1682,30 @@ bool WebRtcVideoChannel::WebRtcVideoSendStream::SetVideoSend(
>   return true;
> }
> 
>-webrtc::VideoSendStream::DegradationPreference
>+webrtc::DegradationPreference
> WebRtcVideoChannel::WebRtcVideoSendStream::GetDegradationPreference() const {
>   // Do not adapt resolution for screen content as this will likely
>   // result in blurry and unreadable text.
>   // |this| acts like a VideoSource to make sure SinkWants are handled on the
>   // correct thread.
>-  DegradationPreference degradation_preference;
>-  if (!enable_cpu_overuse_detection_) {
>-    degradation_preference = DegradationPreference::kDegradationDisabled;
>+  webrtc::DegradationPreference degradation_preference;
>+  if (rtp_parameters_.degradation_preference !=
>+      webrtc::DegradationPreference::BALANCED) {
>+    // If the degradationPreference is different from the default value, assume
>+    // it is what we want, regardless of trials or other internal settings.
>+    degradation_preference = rtp_parameters_.degradation_preference;
>+  } else if (!enable_cpu_overuse_detection_) {
>+    degradation_preference = webrtc::DegradationPreference::DISABLED;
>+  } else if (parameters_.options.is_screencast.value_or(false)) {
>+    degradation_preference = webrtc::DegradationPreference::MAINTAIN_RESOLUTION;
>+  } else if (webrtc::field_trial::IsEnabled(
>+                 "WebRTC-Video-BalancedDegradation")) {
>+    degradation_preference = webrtc::DegradationPreference::BALANCED;
>   } else {
>-    if (parameters_.options.is_screencast.value_or(false)) {
>-      degradation_preference = DegradationPreference::kMaintainResolution;
>-    } else if (webrtc::field_trial::IsEnabled(
>-                   "WebRTC-Video-BalancedDegradation")) {
>-      degradation_preference = DegradationPreference::kBalanced;
>-    } else {
>-      degradation_preference = DegradationPreference::kMaintainFramerate;
>-    }
>+    // TODO(orphis): The default should be BALANCED as the standard mandates.
>+    // Right now, there is no way to set it to BALANCED as it would change
>+    // the behavior for any project expecting MAINTAIN_FRAMERATE by default.
>+    degradation_preference = webrtc::DegradationPreference::MAINTAIN_FRAMERATE;
>   }
>   return degradation_preference;
> }
>@@ -1685,35 +1716,13 @@ WebRtcVideoChannel::WebRtcVideoSendStream::GetSsrcs() const {
> }
> 
> void WebRtcVideoChannel::WebRtcVideoSendStream::SetCodec(
>-    const VideoCodecSettings& codec_settings,
>-    bool force_encoder_allocation) {
>+    const VideoCodecSettings& codec_settings) {
>   RTC_DCHECK_RUN_ON(&thread_checker_);
>   parameters_.encoder_config = CreateVideoEncoderConfig(codec_settings.codec);
>   RTC_DCHECK_GT(parameters_.encoder_config.number_of_streams, 0);
> 
>-  // Do not re-create encoders of the same type. We can't overwrite
>-  // |allocated_encoder_| immediately, because we need to release it after the
>-  // RecreateWebRtcStream() call.
>-  std::unique_ptr<webrtc::VideoEncoder> new_encoder;
>-  if (force_encoder_allocation || !allocated_encoder_ ||
>-      allocated_codec_ != codec_settings.codec) {
>-    const webrtc::SdpVideoFormat format(codec_settings.codec.name,
>-                                        codec_settings.codec.params);
>-    new_encoder = encoder_factory_->CreateVideoEncoder(format);
>-
>-    parameters_.config.encoder_settings.encoder = new_encoder.get();
>-
>-    const webrtc::VideoEncoderFactory::CodecInfo info =
>-        encoder_factory_->QueryVideoEncoder(format);
>-    parameters_.config.encoder_settings.full_overuse_time =
>-        info.is_hardware_accelerated;
>-    parameters_.config.encoder_settings.internal_source =
>-        info.has_internal_source;
>-  } else {
>-    new_encoder = std::move(allocated_encoder_);
>-  }
>-  parameters_.config.encoder_settings.payload_name = codec_settings.codec.name;
>-  parameters_.config.encoder_settings.payload_type = codec_settings.codec.id;
>+  parameters_.config.rtp.payload_name = codec_settings.codec.name;
>+  parameters_.config.rtp.payload_type = codec_settings.codec.id;
>   parameters_.config.rtp.ulpfec = codec_settings.ulpfec;
>   parameters_.config.rtp.flexfec.payload_type =
>       codec_settings.flexfec_payload_type;
>@@ -1735,10 +1744,10 @@ void WebRtcVideoChannel::WebRtcVideoSendStream::SetCodec(
> 
>   parameters_.codec_settings = codec_settings;
> 
>+  // TODO(nisse): Avoid recreation, it should be enough to call
>+  // ReconfigureEncoder.
>   RTC_LOG(LS_INFO) << "RecreateWebRtcStream (send) because of SetCodec.";
>   RecreateWebRtcStream();
>-  allocated_encoder_ = std::move(new_encoder);
>-  allocated_codec_ = codec_settings.codec;
> }
> 
> void WebRtcVideoChannel::WebRtcVideoSendStream::SetSendParameters(
>@@ -1749,10 +1758,17 @@ void WebRtcVideoChannel::WebRtcVideoSendStream::SetSendParameters(
>   bool recreate_stream = false;
>   if (params.rtcp_mode) {
>     parameters_.config.rtp.rtcp_mode = *params.rtcp_mode;
>+    rtp_parameters_.rtcp.reduced_size =
>+        parameters_.config.rtp.rtcp_mode == webrtc::RtcpMode::kReducedSize;
>     recreate_stream = true;
>   }
>   if (params.rtp_header_extensions) {
>     parameters_.config.rtp.extensions = *params.rtp_header_extensions;
>+    rtp_parameters_.header_extensions = *params.rtp_header_extensions;
>+    recreate_stream = true;
>+  }
>+  if (params.mid) {
>+    parameters_.config.rtp.mid = *params.mid;
>     recreate_stream = true;
>   }
>   if (params.max_bandwidth_bps) {
>@@ -1765,12 +1781,10 @@ void WebRtcVideoChannel::WebRtcVideoSendStream::SetSendParameters(
> 
>   // Set codecs and options.
>   if (params.codec) {
>-    bool force_encoder_allocation = false;
>-    SetCodec(*params.codec, force_encoder_allocation);
>+    SetCodec(*params.codec);
>     recreate_stream = false;  // SetCodec has already recreated the stream.
>   } else if (params.conference_mode && parameters_.codec_settings) {
>-    bool force_encoder_allocation = false;
>-    SetCodec(*parameters_.codec_settings, force_encoder_allocation);
>+    SetCodec(*parameters_.codec_settings);
>     recreate_stream = false;  // SetCodec has already recreated the stream.
>   }
>   if (recreate_stream) {
>@@ -1780,26 +1794,60 @@ void WebRtcVideoChannel::WebRtcVideoSendStream::SetSendParameters(
>   }
> }
> 
>-bool WebRtcVideoChannel::WebRtcVideoSendStream::SetRtpParameters(
>+webrtc::RTCError WebRtcVideoChannel::WebRtcVideoSendStream::SetRtpParameters(
>     const webrtc::RtpParameters& new_parameters) {
>   RTC_DCHECK_RUN_ON(&thread_checker_);
>-  if (!ValidateRtpParameters(new_parameters)) {
>-    return false;
>+  webrtc::RTCError error = ValidateRtpParameters(new_parameters);
>+  if (!error.ok()) {
>+    return error;
>+  }
>+
>+  bool new_bitrate = false;
>+  for (size_t i = 0; i < rtp_parameters_.encodings.size(); ++i) {
>+    if ((new_parameters.encodings[i].min_bitrate_bps !=
>+         rtp_parameters_.encodings[i].min_bitrate_bps) ||
>+        (new_parameters.encodings[i].max_bitrate_bps !=
>+         rtp_parameters_.encodings[i].max_bitrate_bps)) {
>+      new_bitrate = true;
>+    }
>   }
> 
>-  bool reconfigure_encoder = (new_parameters.encodings[0].max_bitrate_bps !=
>-                              rtp_parameters_.encodings[0].max_bitrate_bps) ||
>-                             (new_parameters.encodings[0].bitrate_priority !=
>-                              rtp_parameters_.encodings[0].bitrate_priority);
>+  bool new_degradation_preference = false;
>+  if (new_parameters.degradation_preference !=
>+      rtp_parameters_.degradation_preference) {
>+    new_degradation_preference = true;
>+  }
>+
>+  // TODO(bugs.webrtc.org/8807): The bitrate priority really doesn't require an
>+  // entire encoder reconfiguration, it just needs to update the bitrate
>+  // allocator.
>+  bool reconfigure_encoder =
>+      new_bitrate || (new_parameters.encodings[0].bitrate_priority !=
>+                      rtp_parameters_.encodings[0].bitrate_priority);
>+
>+  // TODO(bugs.webrtc.org/8807): The active field as well should not require
>+  // a full encoder reconfiguration, but it needs to update both the bitrate
>+  // allocator and the video bitrate allocator.
>+  bool new_send_state = false;
>+  for (size_t i = 0; i < rtp_parameters_.encodings.size(); ++i) {
>+    if (new_parameters.encodings[i].active !=
>+        rtp_parameters_.encodings[i].active) {
>+      new_send_state = true;
>+    }
>+  }
>   rtp_parameters_ = new_parameters;
>   // Codecs are currently handled at the WebRtcVideoChannel level.
>   rtp_parameters_.codecs.clear();
>-  if (reconfigure_encoder) {
>+  if (reconfigure_encoder || new_send_state) {
>     ReconfigureEncoder();
>   }
>-  // Encoding may have been activated/deactivated.
>-  UpdateSendState();
>-  return true;
>+  if (new_send_state) {
>+    UpdateSendState();
>+  }
>+  if (new_degradation_preference) {
>+    stream_->SetSource(this, GetDegradationPreference());
>+  }
>+  return webrtc::RTCError::OK();
> }
> 
> webrtc::RtpParameters
>@@ -1808,33 +1856,60 @@ WebRtcVideoChannel::WebRtcVideoSendStream::GetRtpParameters() const {
>   return rtp_parameters_;
> }
> 
>-bool WebRtcVideoChannel::WebRtcVideoSendStream::ValidateRtpParameters(
>+webrtc::RTCError
>+WebRtcVideoChannel::WebRtcVideoSendStream::ValidateRtpParameters(
>     const webrtc::RtpParameters& rtp_parameters) {
>+  using webrtc::RTCErrorType;
>   RTC_DCHECK_RUN_ON(&thread_checker_);
>-  if (rtp_parameters.encodings.size() != 1) {
>-    RTC_LOG(LS_ERROR)
>-        << "Attempted to set RtpParameters without exactly one encoding";
>-    return false;
>+  if (rtp_parameters.encodings.size() != rtp_parameters_.encodings.size()) {
>+    LOG_AND_RETURN_ERROR(
>+        RTCErrorType::INVALID_MODIFICATION,
>+        "Attempted to set RtpParameters with different encoding count");
>+  }
>+  if (rtp_parameters.rtcp != rtp_parameters_.rtcp) {
>+    LOG_AND_RETURN_ERROR(
>+        RTCErrorType::INVALID_MODIFICATION,
>+        "Attempted to set RtpParameters with modified RTCP parameters");
>+  }
>+  if (rtp_parameters.header_extensions != rtp_parameters_.header_extensions) {
>+    LOG_AND_RETURN_ERROR(
>+        RTCErrorType::INVALID_MODIFICATION,
>+        "Attempted to set RtpParameters with modified header extensions");
>   }
>   if (rtp_parameters.encodings[0].ssrc != rtp_parameters_.encodings[0].ssrc) {
>-    RTC_LOG(LS_ERROR) << "Attempted to set RtpParameters with modified SSRC";
>-    return false;
>+    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_MODIFICATION,
>+                         "Attempted to set RtpParameters with modified SSRC");
>   }
>   if (rtp_parameters.encodings[0].bitrate_priority <= 0) {
>-    RTC_LOG(LS_ERROR) << "Attempted to set RtpParameters bitrate_priority to "
>-                         "an invalid number. bitrate_priority must be > 0.";
>-    return false;
>+    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_RANGE,
>+                         "Attempted to set RtpParameters bitrate_priority to "
>+                         "an invalid number. bitrate_priority must be > 0.");
>+  }
>+  for (size_t i = 0; i < rtp_parameters.encodings.size(); ++i) {
>+    if (rtp_parameters.encodings[i].min_bitrate_bps &&
>+        rtp_parameters.encodings[i].max_bitrate_bps) {
>+      if (*rtp_parameters.encodings[i].max_bitrate_bps <
>+          *rtp_parameters.encodings[i].min_bitrate_bps) {
>+        LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_RANGE,
>+                             "Attempted to set RtpParameters min bitrate "
>+                             "larger than max bitrate.");
>+      }
>+    }
>   }
>-  return true;
>+  return webrtc::RTCError::OK();
> }
> 
> void WebRtcVideoChannel::WebRtcVideoSendStream::UpdateSendState() {
>   RTC_DCHECK_RUN_ON(&thread_checker_);
>-  // TODO(deadbeef): Need to handle more than one encoding in the future.
>-  RTC_DCHECK(rtp_parameters_.encodings.size() == 1u);
>-  if (sending_ && rtp_parameters_.encodings[0].active) {
>+  if (sending_) {
>     RTC_DCHECK(stream_ != nullptr);
>-    stream_->Start();
>+    std::vector<bool> active_layers(rtp_parameters_.encodings.size());
>+    for (size_t i = 0; i < active_layers.size(); ++i) {
>+      active_layers[i] = rtp_parameters_.encodings[i].active;
>+    }
>+    // This updates what simulcast layers are sending, and possibly starts
>+    // or stops the VideoSendStream.
>+    stream_->UpdateActiveSimulcastLayers(active_layers);
>   } else {
>     if (stream_ != nullptr) {
>       stream_->Stop();
>@@ -1847,6 +1922,10 @@ WebRtcVideoChannel::WebRtcVideoSendStream::CreateVideoEncoderConfig(
>     const VideoCodec& codec) const {
>   RTC_DCHECK_RUN_ON(&thread_checker_);
>   webrtc::VideoEncoderConfig encoder_config;
>+  encoder_config.codec_type = webrtc::PayloadStringToCodecType(codec.name);
>+  encoder_config.video_format =
>+      webrtc::SdpVideoFormat(codec.name, codec.params);
>+
>   bool is_screencast = parameters_.options.is_screencast.value_or(false);
>   if (is_screencast) {
>     encoder_config.min_transmit_bitrate_bps =
>@@ -1865,20 +1944,34 @@ WebRtcVideoChannel::WebRtcVideoSendStream::CreateVideoEncoderConfig(
>   // configure a single stream.
>   encoder_config.number_of_streams = parameters_.config.rtp.ssrcs.size();
>   if (IsCodecBlacklistedForSimulcast(codec.name) ||
>-      (is_screencast &&
>-       (!UseSimulcastScreenshare() || !parameters_.conference_mode))) {
>+      (is_screencast && !parameters_.conference_mode)) {
>     encoder_config.number_of_streams = 1;
>   }
> 
>+  // parameters_.max_bitrate comes from the max bitrate set at the SDP
>+  // (m-section) level with the attribute "b=AS." Note that we override this
>+  // value below if the RtpParameters max bitrate set with
>+  // RtpSender::SetParameters has a lower value.
>   int stream_max_bitrate = parameters_.max_bitrate_bps;
>-  if (rtp_parameters_.encodings[0].max_bitrate_bps) {
>+  // When simulcast is enabled (when there are multiple encodings),
>+  // encodings[i].max_bitrate_bps will be enforced by
>+  // encoder_config.simulcast_layers[i].max_bitrate_bps. Otherwise, it's
>+  // enforced by stream_max_bitrate, taking the minimum of the two maximums
>+  // (one coming from SDP, the other coming from RtpParameters).
>+  if (rtp_parameters_.encodings[0].max_bitrate_bps &&
>+      rtp_parameters_.encodings.size() == 1) {
>     stream_max_bitrate =
>         webrtc::MinPositive(*(rtp_parameters_.encodings[0].max_bitrate_bps),
>                             parameters_.max_bitrate_bps);
>   }
> 
>+  // The codec max bitrate comes from the "x-google-max-bitrate" parameter
>+  // attribute set in the SDP for a specific codec. As done in
>+  // WebRtcVideoChannel::SetSendParameters, this value does not override the
>+  // stream max_bitrate set above.
>   int codec_max_bitrate_kbps;
>-  if (codec.GetParam(kCodecParamMaxBitrate, &codec_max_bitrate_kbps)) {
>+  if (codec.GetParam(kCodecParamMaxBitrate, &codec_max_bitrate_kbps) &&
>+      stream_max_bitrate == -1) {
>     stream_max_bitrate = codec_max_bitrate_kbps * 1000;
>   }
>   encoder_config.max_bitrate_bps = stream_max_bitrate;
>@@ -1890,6 +1983,28 @@ WebRtcVideoChannel::WebRtcVideoSendStream::CreateVideoEncoderConfig(
>   encoder_config.bitrate_priority =
>       rtp_parameters_.encodings[0].bitrate_priority;
> 
>+  // Application-controlled state is held in the encoder_config's
>+  // simulcast_layers. Currently this is used to control which simulcast layers
>+  // are active and for configuring the min/max bitrate.
>+  // The encoder_config's simulcast_layers is also used for non-simulcast (when
>+  // there is a single layer).
>+  RTC_DCHECK_GE(rtp_parameters_.encodings.size(),
>+                encoder_config.number_of_streams);
>+  RTC_DCHECK_GT(encoder_config.number_of_streams, 0);
>+  encoder_config.simulcast_layers.resize(encoder_config.number_of_streams);
>+  for (size_t i = 0; i < encoder_config.simulcast_layers.size(); ++i) {
>+    encoder_config.simulcast_layers[i].active =
>+        rtp_parameters_.encodings[i].active;
>+    if (rtp_parameters_.encodings[i].min_bitrate_bps) {
>+      encoder_config.simulcast_layers[i].min_bitrate_bps =
>+          *rtp_parameters_.encodings[i].min_bitrate_bps;
>+    }
>+    if (rtp_parameters_.encodings[i].max_bitrate_bps) {
>+      encoder_config.simulcast_layers[i].max_bitrate_bps =
>+          *rtp_parameters_.encodings[i].max_bitrate_bps;
>+    }
>+  }
>+
>   int max_qp = kDefaultQpMax;
>   codec.GetParam(kCodecParamMaxQuantization, &max_qp);
>   encoder_config.video_stream_factory =
>@@ -1915,8 +2030,8 @@ void WebRtcVideoChannel::WebRtcVideoSendStream::ReconfigureEncoder() {
>   webrtc::VideoEncoderConfig encoder_config =
>       CreateVideoEncoderConfig(codec_settings.codec);
> 
>-  encoder_config.encoder_specific_settings = ConfigureVideoEncoderSettings(
>-      codec_settings.codec);
>+  encoder_config.encoder_specific_settings =
>+      ConfigureVideoEncoderSettings(codec_settings.codec);
> 
>   stream_->ReconfigureVideoEncoder(encoder_config.Copy());
> 
>@@ -2006,9 +2121,9 @@ VideoSenderInfo WebRtcVideoChannel::WebRtcVideoSendStream::GetVideoSenderInfo(
>   info.qp_sum = stats.qp_sum;
> 
>   info.nominal_bitrate = stats.media_bitrate_bps;
>-  info.preferred_bitrate = stats.preferred_media_bitrate_bps;
> 
>   info.content_type = stats.content_type;
>+  info.huge_frames_sent = stats.huge_frames_sent;
> 
>   info.send_frame_width = 0;
>   info.send_frame_height = 0;
>@@ -2135,7 +2250,7 @@ WebRtcVideoChannel::WebRtcVideoReceiveStream::GetSsrcs() const {
>   return stream_params_.ssrcs;
> }
> 
>-rtc::Optional<uint32_t>
>+absl::optional<uint32_t>
> WebRtcVideoChannel::WebRtcVideoReceiveStream::GetFirstPrimarySsrc() const {
>   std::vector<uint32_t> primary_ssrcs;
>   stream_params_.GetPrimarySsrcs(&primary_ssrcs);
>@@ -2143,12 +2258,22 @@ WebRtcVideoChannel::WebRtcVideoReceiveStream::GetFirstPrimarySsrc() const {
>   if (primary_ssrcs.empty()) {
>     RTC_LOG(LS_WARNING)
>         << "Empty primary ssrcs vector, returning empty optional";
>-    return rtc::nullopt;
>+    return absl::nullopt;
>   } else {
>     return primary_ssrcs[0];
>   }
> }
> 
>+webrtc::RtpParameters
>+WebRtcVideoChannel::WebRtcVideoReceiveStream::GetRtpParameters() const {
>+  webrtc::RtpParameters rtp_parameters;
>+  rtp_parameters.encodings.emplace_back();
>+  rtp_parameters.encodings[0].ssrc = GetFirstPrimarySsrc();
>+  rtp_parameters.header_extensions = config_.rtp.extensions;
>+
>+  return rtp_parameters;
>+}
>+
> void WebRtcVideoChannel::WebRtcVideoReceiveStream::ConfigureCodecs(
>     const std::vector<VideoCodecSettings>& recv_codecs,
>     DecoderMap* old_decoders) {
>@@ -2161,6 +2286,13 @@ void WebRtcVideoChannel::WebRtcVideoReceiveStream::ConfigureCodecs(
>                                         recv_codec.codec.params);
>     std::unique_ptr<webrtc::VideoDecoder> new_decoder;
> 
>+    if (allocated_decoders_.count(video_format) > 0) {
>+      RTC_LOG(LS_WARNING)
>+          << "VideoReceiveStream configured with duplicate codecs: "
>+          << video_format.name;
>+      continue;
>+    }
>+
>     auto it = old_decoders->find(video_format);
>     if (it != old_decoders->end()) {
>       new_decoder = std::move(it->second);
>@@ -2201,6 +2333,7 @@ void WebRtcVideoChannel::WebRtcVideoReceiveStream::ConfigureCodecs(
>   config_.rtp.red_payload_type = codec.ulpfec.red_payload_type;
> 
>   config_.rtp.nack.rtp_history_ms = HasNack(codec.codec) ? kNackHistoryMs : 0;
>+  config_.rtp.rtcp_xr.receiver_reference_time_report = HasRrtr(codec.codec);
>   if (codec.ulpfec.red_rtx_payload_type != -1) {
>     config_.rtp
>         .rtx_associated_payload_types[codec.ulpfec.red_rtx_payload_type] =
>@@ -2219,7 +2352,7 @@ void WebRtcVideoChannel::WebRtcVideoReceiveStream::SetLocalSsrc(
>   // should not be able to create a sender with the same SSRC as a receiver, but
>   // right now this can't be done due to unittests depending on receiving what
>   // they are sending from the same MediaChannel.
>-  if (local_ssrc == config_.rtp.remote_ssrc) {
>+  if (local_ssrc == config_.rtp.local_ssrc) {
>     RTC_LOG(LS_INFO) << "Ignoring call to SetLocalSsrc because parameters are "
>                         "unchanged; local_ssrc="
>                      << local_ssrc;
>@@ -2299,8 +2432,7 @@ void WebRtcVideoChannel::WebRtcVideoReceiveStream::SetRecvParameters(
>   }
> }
> 
>-void WebRtcVideoChannel::WebRtcVideoReceiveStream::
>-    RecreateWebRtcVideoStream() {
>+void WebRtcVideoChannel::WebRtcVideoReceiveStream::RecreateWebRtcVideoStream() {
>   if (stream_) {
>     MaybeDissociateFlexfecFromVideo();
>     call_->DestroyVideoReceiveStream(stream_);
>@@ -2420,8 +2552,8 @@ WebRtcVideoChannel::WebRtcVideoReceiveStream::GetVideoReceiverInfo(
>   info.jitter_buffer_ms = stats.jitter_buffer_ms;
>   info.min_playout_delay_ms = stats.min_playout_delay_ms;
>   info.render_delay_ms = stats.render_delay_ms;
>-  info.frames_received = stats.frame_counts.key_frames +
>-                         stats.frame_counts.delta_frames;
>+  info.frames_received =
>+      stats.frame_counts.key_frames + stats.frame_counts.delta_frames;
>   info.frames_decoded = stats.frames_decoded;
>   info.frames_rendered = stats.frames_rendered;
>   info.qp_sum = stats.qp_sum;
>@@ -2540,8 +2672,7 @@ WebRtcVideoChannel::MapCodecs(const std::vector<VideoCodec>& codecs) {
>   RTC_DCHECK(!video_codecs.empty());
> 
>   for (std::map<int, int>::const_iterator it = rtx_mapping.begin();
>-       it != rtx_mapping.end();
>-       ++it) {
>+       it != rtx_mapping.end(); ++it) {
>     if (!payload_used[it->first]) {
>       RTC_LOG(LS_ERROR) << "RTX mapped to payload not in codec list.";
>       return std::vector<VideoCodecSettings>();
>@@ -2571,32 +2702,88 @@ WebRtcVideoChannel::MapCodecs(const std::vector<VideoCodec>& codecs) {
>   return video_codecs;
> }
> 
>-EncoderStreamFactory::EncoderStreamFactory(std::string codec_name,
>-                                           int max_qp,
>-                                           int max_framerate,
>-                                           bool is_screencast,
>-                                           bool conference_mode)
>+// TODO(bugs.webrtc.org/8785): Consider removing max_qp and max_framerate
>+// as members of EncoderStreamFactory and instead set these values individually
>+// for each stream in the VideoEncoderConfig.simulcast_layers.
>+EncoderStreamFactory::EncoderStreamFactory(
>+    std::string codec_name,
>+    int max_qp,
>+    int max_framerate,
>+    bool is_screenshare,
>+    bool screenshare_config_explicitly_enabled)
>+
>     : codec_name_(codec_name),
>       max_qp_(max_qp),
>       max_framerate_(max_framerate),
>-      is_screencast_(is_screencast),
>-      conference_mode_(conference_mode) {}
>+      is_screenshare_(is_screenshare),
>+      screenshare_config_explicitly_enabled_(
>+          screenshare_config_explicitly_enabled) {}
> 
> std::vector<webrtc::VideoStream> EncoderStreamFactory::CreateEncoderStreams(
>     int width,
>     int height,
>     const webrtc::VideoEncoderConfig& encoder_config) {
>-  if (is_screencast_ &&
>-      (!conference_mode_ || !cricket::UseSimulcastScreenshare())) {
>+  if (is_screenshare_ && !screenshare_config_explicitly_enabled_) {
>     RTC_DCHECK_EQ(1, encoder_config.number_of_streams);
>   }
>+  RTC_DCHECK_GT(encoder_config.number_of_streams, 0);
>+  RTC_DCHECK_EQ(encoder_config.simulcast_layers.size(),
>+                encoder_config.number_of_streams);
>+  std::vector<webrtc::VideoStream> layers;
>+
>   if (encoder_config.number_of_streams > 1 ||
>-      (CodecNamesEq(codec_name_, kVp8CodecName) && is_screencast_ &&
>-       conference_mode_)) {
>-    return GetSimulcastConfig(encoder_config.number_of_streams, width, height,
>-                              encoder_config.max_bitrate_bps,
>-                              encoder_config.bitrate_priority, max_qp_,
>-                              max_framerate_, is_screencast_);
>+     ((CodecNamesEq(codec_name_, kVp8CodecName) ||
>+        CodecNamesEq(codec_name_, kH264CodecName)) &&
>+       is_screenshare_ && screenshare_config_explicitly_enabled_)) {
>+    bool temporal_layers_supported = CodecNamesEq(codec_name_, kVp8CodecName);
>+    layers = GetSimulcastConfig(encoder_config.number_of_streams, width, height,
>+                                0 /*not used*/, encoder_config.bitrate_priority,
>+                                max_qp_, max_framerate_, is_screenshare_,
>+                                temporal_layers_supported);
>+    // Update the active simulcast layers and configured bitrates.
>+    bool is_highest_layer_max_bitrate_configured = false;
>+    for (size_t i = 0; i < layers.size(); ++i) {
>+      layers[i].active = encoder_config.simulcast_layers[i].active;
>+      // Update simulcast bitrates with configured min and max bitrate.
>+      if (encoder_config.simulcast_layers[i].min_bitrate_bps > 0) {
>+        layers[i].min_bitrate_bps =
>+            encoder_config.simulcast_layers[i].min_bitrate_bps;
>+      }
>+      if (encoder_config.simulcast_layers[i].max_bitrate_bps > 0) {
>+        layers[i].max_bitrate_bps =
>+            encoder_config.simulcast_layers[i].max_bitrate_bps;
>+      }
>+      if (encoder_config.simulcast_layers[i].min_bitrate_bps > 0 &&
>+          encoder_config.simulcast_layers[i].max_bitrate_bps > 0) {
>+        // Min and max bitrate are configured.
>+        // Set target to 3/4 of the max bitrate (or to max if below min).
>+        layers[i].target_bitrate_bps = layers[i].max_bitrate_bps * 3 / 4;
>+        if (layers[i].target_bitrate_bps < layers[i].min_bitrate_bps)
>+          layers[i].target_bitrate_bps = layers[i].max_bitrate_bps;
>+      } else if (encoder_config.simulcast_layers[i].min_bitrate_bps > 0) {
>+        // Only min bitrate is configured, make sure target/max are above min.
>+        layers[i].target_bitrate_bps =
>+            std::max(layers[i].target_bitrate_bps, layers[i].min_bitrate_bps);
>+        layers[i].max_bitrate_bps =
>+            std::max(layers[i].max_bitrate_bps, layers[i].min_bitrate_bps);
>+      } else if (encoder_config.simulcast_layers[i].max_bitrate_bps > 0) {
>+        // Only max bitrate is configured, make sure min/target are below max.
>+        layers[i].min_bitrate_bps =
>+            std::min(layers[i].min_bitrate_bps, layers[i].max_bitrate_bps);
>+        layers[i].target_bitrate_bps =
>+            std::min(layers[i].target_bitrate_bps, layers[i].max_bitrate_bps);
>+      }
>+      if (i == layers.size() - 1) {
>+        is_highest_layer_max_bitrate_configured =
>+            encoder_config.simulcast_layers[i].max_bitrate_bps > 0;
>+      }
>+    }
>+    if (!is_screenshare_ && !is_highest_layer_max_bitrate_configured) {
>+      // No application-configured maximum for the largest layer.
>+      // If there is bitrate leftover, give it to the largest layer.
>+      BoostMaxSimulcastLayer(encoder_config.max_bitrate_bps, &layers);
>+    }
>+    return layers;
>   }
> 
>   // For unset max bitrates set default bitrate for non-simulcast.
>@@ -2605,23 +2792,38 @@ std::vector<webrtc::VideoStream> EncoderStreamFactory::CreateEncoderStreams(
>           ? encoder_config.max_bitrate_bps
>           : GetMaxDefaultVideoBitrateKbps(width, height) * 1000;
> 
>-  webrtc::VideoStream stream;
>-  stream.width = width;
>-  stream.height = height;
>-  stream.max_framerate = max_framerate_;
>-  stream.min_bitrate_bps = GetMinVideoBitrateBps();
>-  stream.target_bitrate_bps = stream.max_bitrate_bps = max_bitrate_bps;
>-  stream.max_qp = max_qp_;
>-  stream.bitrate_priority = encoder_config.bitrate_priority;
>-
>-  if (CodecNamesEq(codec_name_, kVp9CodecName) && !is_screencast_) {
>-    stream.temporal_layer_thresholds_bps.resize(GetDefaultVp9TemporalLayers() -
>-                                                1);
>-  }
>-
>-  std::vector<webrtc::VideoStream> streams;
>-  streams.push_back(stream);
>-  return streams;
>+  int min_bitrate_bps = GetMinVideoBitrateBps();
>+  if (encoder_config.simulcast_layers[0].min_bitrate_bps > 0) {
>+    // Use set min bitrate.
>+    min_bitrate_bps = encoder_config.simulcast_layers[0].min_bitrate_bps;
>+    // If only min bitrate is configured, make sure max is above min.
>+    if (encoder_config.max_bitrate_bps <= 0)
>+      max_bitrate_bps = std::max(min_bitrate_bps, max_bitrate_bps);
>+  }
>+
>+  webrtc::VideoStream layer;
>+  layer.width = width;
>+  layer.height = height;
>+  layer.max_framerate = max_framerate_;
>+
>+  // In the case that the application sets a max bitrate that's lower than the
>+  // min bitrate, we adjust it down (see bugs.webrtc.org/9141).
>+  layer.min_bitrate_bps = std::min(min_bitrate_bps, max_bitrate_bps);
>+  layer.target_bitrate_bps = layer.max_bitrate_bps = max_bitrate_bps;
>+  layer.max_qp = max_qp_;
>+  layer.bitrate_priority = encoder_config.bitrate_priority;
>+
>+  if (CodecNamesEq(codec_name_, kVp9CodecName)) {
>+    RTC_DCHECK(encoder_config.encoder_specific_settings);
>+    // Use VP9 SVC layering from codec settings which might be initialized
>+    // though field trial in ConfigureVideoEncoderSettings.
>+    webrtc::VideoCodecVP9 vp9_settings;
>+    encoder_config.encoder_specific_settings->FillVideoCodecVp9(&vp9_settings);
>+    layer.num_temporal_layers = vp9_settings.numberOfTemporalLayers;
>+  }
>+
>+  layers.push_back(layer);
>+  return layers;
> }
> 
> }  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideoengine.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideoengine.h
>index 56c7586de9f..f00ebee4f88 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideoengine.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideoengine.h
>@@ -17,12 +17,12 @@
> #include <string>
> #include <vector>
> 
>+#include "absl/types/optional.h"
> #include "api/call/transport.h"
>-#include "api/optional.h"
> #include "api/video/video_frame.h"
>+#include "api/video/video_sink_interface.h"
>+#include "api/video/video_source_interface.h"
> #include "api/video_codecs/sdp_video_format.h"
>-#include "api/videosinkinterface.h"
>-#include "api/videosourceinterface.h"
> #include "call/call.h"
> #include "call/flexfec_receive_stream.h"
> #include "call/video_receive_stream.h"
>@@ -42,7 +42,7 @@ class VideoDecoderFactory;
> class VideoEncoder;
> class VideoEncoderFactory;
> struct MediaConfig;
>-}
>+}  // namespace webrtc
> 
> namespace rtc {
> class Thread;
>@@ -51,19 +51,7 @@ class Thread;
> namespace cricket {
> 
> class DecoderFactoryAdapter;
>-class VideoCapturer;
>-class VideoProcessor;
>-class VideoRenderer;
>-class VoiceMediaChannel;
>-class WebRtcDecoderObserver;
>-class WebRtcEncoderObserver;
>-class WebRtcLocalStreamInfo;
>-class WebRtcRenderAdapter;
> class WebRtcVideoChannel;
>-class WebRtcVideoChannelRecvInfo;
>-class WebRtcVideoChannelSendInfo;
>-class WebRtcVoiceEngine;
>-class WebRtcVoiceMediaChannel;
> 
> class UnsignalledSsrcHandler {
>  public:
>@@ -80,8 +68,7 @@ class UnsignalledSsrcHandler {
> class DefaultUnsignalledSsrcHandler : public UnsignalledSsrcHandler {
>  public:
>   DefaultUnsignalledSsrcHandler();
>-  Action OnUnsignalledSsrc(WebRtcVideoChannel* channel,
>-                           uint32_t ssrc) override;
>+  Action OnUnsignalledSsrc(WebRtcVideoChannel* channel, uint32_t ssrc) override;
> 
>   rtc::VideoSinkInterface<webrtc::VideoFrame>* GetDefaultSink() const;
>   void SetDefaultSink(WebRtcVideoChannel* channel,
>@@ -96,11 +83,13 @@ class DefaultUnsignalledSsrcHandler : public UnsignalledSsrcHandler {
> // WebRtcVideoEngine is used for the new native WebRTC Video API (webrtc:1667).
> class WebRtcVideoEngine {
>  public:
>+#if defined(USE_BUILTIN_SW_CODECS)
>   // Internal SW video codecs will be added on top of the external codecs.
>   WebRtcVideoEngine(
>       std::unique_ptr<WebRtcVideoEncoderFactory> external_video_encoder_factory,
>       std::unique_ptr<WebRtcVideoDecoderFactory>
>           external_video_decoder_factory);
>+#endif
> 
>   // These video codec factories represents all video codecs, i.e. both software
>   // and external hardware codecs.
>@@ -137,8 +126,9 @@ class WebRtcVideoChannel : public VideoMediaChannel, public webrtc::Transport {
>   bool SetSendParameters(const VideoSendParameters& params) override;
>   bool SetRecvParameters(const VideoRecvParameters& params) override;
>   webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const override;
>-  bool SetRtpSendParameters(uint32_t ssrc,
>-                            const webrtc::RtpParameters& parameters) override;
>+  webrtc::RTCError SetRtpSendParameters(
>+      uint32_t ssrc,
>+      const webrtc::RtpParameters& parameters) override;
>   webrtc::RtpParameters GetRtpReceiveParameters(uint32_t ssrc) const override;
>   bool SetRtpReceiveParameters(
>       uint32_t ssrc,
>@@ -147,7 +137,6 @@ class WebRtcVideoChannel : public VideoMediaChannel, public webrtc::Transport {
>   bool SetSend(bool send) override;
>   bool SetVideoSend(
>       uint32_t ssrc,
>-      bool enable,
>       const VideoOptions* options,
>       rtc::VideoSourceInterface<webrtc::VideoFrame>* source) override;
>   bool AddSendStream(const StreamParams& sp) override;
>@@ -172,7 +161,9 @@ class WebRtcVideoChannel : public VideoMediaChannel, public webrtc::Transport {
>   // Implemented for VideoMediaChannelTest.
>   bool sending() const { return sending_; }
> 
>-  rtc::Optional<uint32_t> GetDefaultReceiveStreamSsrc();
>+  absl::optional<uint32_t> GetDefaultReceiveStreamSsrc();
>+
>+  StreamParams unsignaled_stream_params() { return unsignaled_stream_params_; }
> 
>   // AdaptReason is used for expressing why a WebRtcVideoSendStream request
>   // a lower input frame size than the currently configured camera input frame
>@@ -208,21 +199,22 @@ class WebRtcVideoChannel : public VideoMediaChannel, public webrtc::Transport {
> 
>   struct ChangedSendParameters {
>     // These optionals are unset if not changed.
>-    rtc::Optional<VideoCodecSettings> codec;
>-    rtc::Optional<std::vector<webrtc::RtpExtension>> rtp_header_extensions;
>-    rtc::Optional<int> max_bandwidth_bps;
>-    rtc::Optional<bool> conference_mode;
>-    rtc::Optional<webrtc::RtcpMode> rtcp_mode;
>+    absl::optional<VideoCodecSettings> codec;
>+    absl::optional<std::vector<webrtc::RtpExtension>> rtp_header_extensions;
>+    absl::optional<std::string> mid;
>+    absl::optional<int> max_bandwidth_bps;
>+    absl::optional<bool> conference_mode;
>+    absl::optional<webrtc::RtcpMode> rtcp_mode;
>   };
> 
>   struct ChangedRecvParameters {
>     // These optionals are unset if not changed.
>-    rtc::Optional<std::vector<VideoCodecSettings>> codec_settings;
>-    rtc::Optional<std::vector<webrtc::RtpExtension>> rtp_header_extensions;
>+    absl::optional<std::vector<VideoCodecSettings>> codec_settings;
>+    absl::optional<std::vector<webrtc::RtpExtension>> rtp_header_extensions;
>     // Keep track of the FlexFEC payload type separately from |codec_settings|.
>     // This allows us to recreate the FlexfecReceiveStream separately from the
>     // VideoReceiveStream when the FlexFEC payload type is changed.
>-    rtc::Optional<int> flexfec_payload_type;
>+    absl::optional<int> flexfec_payload_type;
>   };
> 
>   bool GetChangedSendParameters(const VideoSendParameters& params,
>@@ -255,16 +247,15 @@ class WebRtcVideoChannel : public VideoMediaChannel, public webrtc::Transport {
>         const StreamParams& sp,
>         webrtc::VideoSendStream::Config config,
>         const VideoOptions& options,
>-        webrtc::VideoEncoderFactory* encoder_factory,
>         bool enable_cpu_overuse_detection,
>         int max_bitrate_bps,
>-        const rtc::Optional<VideoCodecSettings>& codec_settings,
>-        const rtc::Optional<std::vector<webrtc::RtpExtension>>& rtp_extensions,
>+        const absl::optional<VideoCodecSettings>& codec_settings,
>+        const absl::optional<std::vector<webrtc::RtpExtension>>& rtp_extensions,
>         const VideoSendParameters& send_params);
>     virtual ~WebRtcVideoSendStream();
> 
>     void SetSendParameters(const ChangedSendParameters& send_params);
>-    bool SetRtpParameters(const webrtc::RtpParameters& parameters);
>+    webrtc::RTCError SetRtpParameters(const webrtc::RtpParameters& parameters);
>     webrtc::RtpParameters GetRtpParameters() const;
> 
>     // Implements rtc::VideoSourceInterface<webrtc::VideoFrame>.
>@@ -275,8 +266,7 @@ class WebRtcVideoChannel : public VideoMediaChannel, public webrtc::Transport {
>                          const rtc::VideoSinkWants& wants) override;
>     void RemoveSink(rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) override;
> 
>-    bool SetVideoSend(bool mute,
>-                      const VideoOptions* options,
>+    bool SetVideoSend(const VideoOptions* options,
>                       rtc::VideoSourceInterface<webrtc::VideoFrame>* source);
> 
>     void SetSend(bool send);
>@@ -295,12 +285,12 @@ class WebRtcVideoChannel : public VideoMediaChannel, public webrtc::Transport {
>           webrtc::VideoSendStream::Config config,
>           const VideoOptions& options,
>           int max_bitrate_bps,
>-          const rtc::Optional<VideoCodecSettings>& codec_settings);
>+          const absl::optional<VideoCodecSettings>& codec_settings);
>       webrtc::VideoSendStream::Config config;
>       VideoOptions options;
>       int max_bitrate_bps;
>       bool conference_mode;
>-      rtc::Optional<VideoCodecSettings> codec_settings;
>+      absl::optional<VideoCodecSettings> codec_settings;
>       // Sent resolutions + bitrates etc. by the underlying VideoSendStream,
>       // typically changes when setting a new resolution or reconfiguring
>       // bitrates.
>@@ -309,51 +299,46 @@ class WebRtcVideoChannel : public VideoMediaChannel, public webrtc::Transport {
> 
>     rtc::scoped_refptr<webrtc::VideoEncoderConfig::EncoderSpecificSettings>
>     ConfigureVideoEncoderSettings(const VideoCodec& codec);
>-    void SetCodec(const VideoCodecSettings& codec,
>-                  bool force_encoder_allocation);
>+    void SetCodec(const VideoCodecSettings& codec);
>     void RecreateWebRtcStream();
>     webrtc::VideoEncoderConfig CreateVideoEncoderConfig(
>         const VideoCodec& codec) const;
>     void ReconfigureEncoder();
>-    bool ValidateRtpParameters(const webrtc::RtpParameters& parameters);
>+    webrtc::RTCError ValidateRtpParameters(
>+        const webrtc::RtpParameters& parameters);
> 
>     // Calls Start or Stop according to whether or not |sending_| is true,
>     // and whether or not the encoding in |rtp_parameters_| is active.
>     void UpdateSendState();
> 
>-    webrtc::VideoSendStream::DegradationPreference GetDegradationPreference()
>-        const RTC_EXCLUSIVE_LOCKS_REQUIRED(&thread_checker_);
>+    webrtc::DegradationPreference GetDegradationPreference() const
>+        RTC_EXCLUSIVE_LOCKS_REQUIRED(&thread_checker_);
> 
>     rtc::ThreadChecker thread_checker_;
>     rtc::AsyncInvoker invoker_;
>     rtc::Thread* worker_thread_;
>-    const std::vector<uint32_t> ssrcs_ RTC_ACCESS_ON(&thread_checker_);
>-    const std::vector<SsrcGroup> ssrc_groups_ RTC_ACCESS_ON(&thread_checker_);
>+    const std::vector<uint32_t> ssrcs_ RTC_GUARDED_BY(&thread_checker_);
>+    const std::vector<SsrcGroup> ssrc_groups_ RTC_GUARDED_BY(&thread_checker_);
>     webrtc::Call* const call_;
>     const bool enable_cpu_overuse_detection_;
>     rtc::VideoSourceInterface<webrtc::VideoFrame>* source_
>-        RTC_ACCESS_ON(&thread_checker_);
>-    webrtc::VideoEncoderFactory* const encoder_factory_
>-        RTC_ACCESS_ON(&thread_checker_);
>+        RTC_GUARDED_BY(&thread_checker_);
> 
>-    webrtc::VideoSendStream* stream_ RTC_ACCESS_ON(&thread_checker_);
>+    webrtc::VideoSendStream* stream_ RTC_GUARDED_BY(&thread_checker_);
>     rtc::VideoSinkInterface<webrtc::VideoFrame>* encoder_sink_
>-        RTC_ACCESS_ON(&thread_checker_);
>+        RTC_GUARDED_BY(&thread_checker_);
>     // Contains settings that are the same for all streams in the MediaChannel,
>     // such as codecs, header extensions, and the global bitrate limit for the
>     // entire channel.
>-    VideoSendStreamParameters parameters_ RTC_ACCESS_ON(&thread_checker_);
>+    VideoSendStreamParameters parameters_ RTC_GUARDED_BY(&thread_checker_);
>     // Contains settings that are unique for each stream, such as max_bitrate.
>     // Does *not* contain codecs, however.
>     // TODO(skvlad): Move ssrcs_ and ssrc_groups_ into rtp_parameters_.
>     // TODO(skvlad): Combine parameters_ and rtp_parameters_ once we have only
>     // one stream per MediaChannel.
>-    webrtc::RtpParameters rtp_parameters_ RTC_ACCESS_ON(&thread_checker_);
>-    std::unique_ptr<webrtc::VideoEncoder> allocated_encoder_
>-        RTC_ACCESS_ON(&thread_checker_);
>-    VideoCodec allocated_codec_ RTC_ACCESS_ON(&thread_checker_);
>+    webrtc::RtpParameters rtp_parameters_ RTC_GUARDED_BY(&thread_checker_);
> 
>-    bool sending_ RTC_ACCESS_ON(&thread_checker_);
>+    bool sending_ RTC_GUARDED_BY(&thread_checker_);
>   };
> 
>   // Wrapper for the receiver part, contains configs etc. that are needed to
>@@ -372,7 +357,9 @@ class WebRtcVideoChannel : public VideoMediaChannel, public webrtc::Transport {
>     ~WebRtcVideoReceiveStream();
> 
>     const std::vector<uint32_t>& GetSsrcs() const;
>-    rtc::Optional<uint32_t> GetFirstPrimarySsrc() const;
>+
>+    // Does not return codecs, they are filled by the owning WebRtcVideoChannel.
>+    webrtc::RtpParameters GetRtpParameters() const;
> 
>     void SetLocalSsrc(uint32_t local_ssrc);
>     // TODO(deadbeef): Move these feedback parameters into the recv parameters.
>@@ -414,6 +401,8 @@ class WebRtcVideoChannel : public VideoMediaChannel, public webrtc::Transport {
> 
>     std::string GetCodecNameFromPayloadType(int payload_type);
> 
>+    absl::optional<uint32_t> GetFirstPrimarySsrc() const;
>+
>     webrtc::Call* const call_;
>     StreamParams stream_params_;
> 
>@@ -454,7 +443,7 @@ class WebRtcVideoChannel : public VideoMediaChannel, public webrtc::Transport {
>   // Select what video codec will be used for sending, i.e. what codec is used
>   // for local encoding, based on supported remote codecs. The first remote
>   // codec that is supported locally will be selected.
>-  rtc::Optional<VideoCodecSettings> SelectSendVideoCodec(
>+  absl::optional<VideoCodecSettings> SelectSendVideoCodec(
>       const std::vector<VideoCodecSettings>& remote_mapped_codecs) const;
> 
>   static bool NonFlexfecReceiveCodecsHaveChanged(
>@@ -487,8 +476,8 @@ class WebRtcVideoChannel : public VideoMediaChannel, public webrtc::Transport {
>   std::set<uint32_t> send_ssrcs_ RTC_GUARDED_BY(stream_crit_);
>   std::set<uint32_t> receive_ssrcs_ RTC_GUARDED_BY(stream_crit_);
> 
>-  rtc::Optional<VideoCodecSettings> send_codec_;
>-  rtc::Optional<std::vector<webrtc::RtpExtension>> send_rtp_extensions_;
>+  absl::optional<VideoCodecSettings> send_codec_;
>+  absl::optional<std::vector<webrtc::RtpExtension>> send_rtp_extensions_;
> 
>   webrtc::VideoEncoderFactory* const encoder_factory_;
>   DecoderFactoryAdapter* const decoder_factory_;
>@@ -497,13 +486,18 @@ class WebRtcVideoChannel : public VideoMediaChannel, public webrtc::Transport {
>   // See reason for keeping track of the FlexFEC payload type separately in
>   // comment in WebRtcVideoChannel::ChangedRecvParameters.
>   int recv_flexfec_payload_type_;
>-  webrtc::Call::Config::BitrateConfig bitrate_config_;
>+  webrtc::BitrateConstraints bitrate_config_;
>   // TODO(deadbeef): Don't duplicate information between
>   // send_params/recv_params, rtp_extensions, options, etc.
>   VideoSendParameters send_params_;
>   VideoOptions default_send_options_;
>   VideoRecvParameters recv_params_;
>   int64_t last_stats_log_ms_;
>+  // This is a stream param that comes from the remote description, but wasn't
>+  // signaled with any a=ssrc lines. It holds information that was signaled
>+  // before the unsignaled receive stream is created when the first packet is
>+  // received.
>+  StreamParams unsignaled_stream_params_;
> };
> 
> class EncoderStreamFactory
>@@ -512,8 +506,8 @@ class EncoderStreamFactory
>   EncoderStreamFactory(std::string codec_name,
>                        int max_qp,
>                        int max_framerate,
>-                       bool is_screencast,
>-                       bool conference_mode);
>+                       bool is_screenshare,
>+                       bool screenshare_config_explicitly_enabled);
> 
>  private:
>   std::vector<webrtc::VideoStream> CreateEncoderStreams(
>@@ -524,8 +518,10 @@ class EncoderStreamFactory
>   const std::string codec_name_;
>   const int max_qp_;
>   const int max_framerate_;
>-  const bool is_screencast_;
>-  const bool conference_mode_;
>+  const bool is_screenshare_;
>+  // Allows a screenshare specific configuration, which enables temporal
>+  // layering and allows simulcast.
>+  const bool screenshare_config_explicitly_enabled_;
> };
> 
> }  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideoengine_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideoengine_unittest.cc
>index 82154a8c0d7..5361a95f4db 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideoengine_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvideoengine_unittest.cc
>@@ -17,6 +17,8 @@
> #include "api/rtpparameters.h"
> #include "api/test/mock_video_decoder_factory.h"
> #include "api/test/mock_video_encoder_factory.h"
>+#include "api/video_codecs/builtin_video_decoder_factory.h"
>+#include "api/video_codecs/builtin_video_encoder_factory.h"
> #include "api/video_codecs/sdp_video_format.h"
> #include "api/video_codecs/video_decoder_factory.h"
> #include "api/video_codecs/video_encoder.h"
>@@ -24,30 +26,44 @@
> #include "call/flexfec_receive_stream.h"
> #include "common_video/h264/profile_level_id.h"
> #include "logging/rtc_event_log/rtc_event_log.h"
>+#include "media/base/fakenetworkinterface.h"
>+#include "media/base/fakevideocapturer.h"
>+#include "media/base/fakevideorenderer.h"
> #include "media/base/mediaconstants.h"
> #include "media/base/rtputils.h"
> #include "media/base/testutils.h"
>-#include "media/base/videoengine_unittest.h"
> #include "media/engine/constants.h"
> #include "media/engine/fakewebrtccall.h"
> #include "media/engine/fakewebrtcvideoengine.h"
>-#include "media/engine/internalencoderfactory.h"
> #include "media/engine/simulcast.h"
> #include "media/engine/webrtcvideoengine.h"
> #include "media/engine/webrtcvoiceengine.h"
>+#include "modules/rtp_rtcp/include/rtp_header_parser.h"
> #include "rtc_base/arraysize.h"
> #include "rtc_base/gunit.h"
>+#include "rtc_base/numerics/safe_conversions.h"
> #include "rtc_base/stringutils.h"
> #include "test/field_trial.h"
> #include "test/gmock.h"
> 
>+using cricket::FakeVideoCapturerWithTaskQueue;
>+using webrtc::BitrateConstraints;
> using webrtc::RtpExtension;
>+using testing::Field;
> 
> namespace {
> static const int kDefaultQpMax = 56;
> 
> static const uint8_t kRedRtxPayloadType = 125;
> 
>+static const uint32_t kTimeout = 5000U;
>+static const uint32_t kDefaultReceiveSsrc = 0;
>+static const uint32_t kSsrc = 1234u;
>+static const uint32_t kSsrcs4[] = {1, 2, 3, 4};
>+static const int kVideoWidth = 640;
>+static const int kVideoHeight = 360;
>+static const int kFramerate = 30;
>+
> static const uint32_t kSsrcs1[] = {1};
> static const uint32_t kSsrcs3[] = {1, 2, 3};
> static const uint32_t kRtxSsrcs1[] = {4};
>@@ -60,7 +76,7 @@ static const char kUnsupportedExtensionName[] =
> 
> cricket::VideoCodec RemoveFeedbackParams(cricket::VideoCodec&& codec) {
>   codec.feedback_params = cricket::FeedbackParams();
>-  return codec;
>+  return std::move(codec);
> }
> 
> void VerifyCodecHasDefaultFeedbackParams(const cricket::VideoCodec& codec) {
>@@ -101,9 +117,8 @@ const int* FindKeyByValue(const std::map<int, int>& m, int v) {
>   return nullptr;
> }
> 
>-bool HasRtxReceiveAssociation(
>-    const webrtc::VideoReceiveStream::Config& config,
>-    int payload_type) {
>+bool HasRtxReceiveAssociation(const webrtc::VideoReceiveStream::Config& config,
>+                              int payload_type) {
>   return FindKeyByValue(config.rtp.rtx_associated_payload_types,
>                         payload_type) != nullptr;
> }
>@@ -130,7 +145,7 @@ rtc::scoped_refptr<webrtc::VideoFrameBuffer> CreateBlackFrameBuffer(
> void VerifySendStreamHasRtxTypes(const webrtc::VideoSendStream::Config& config,
>                                  const std::map<int, int>& rtx_types) {
>   std::map<int, int>::const_iterator it;
>-  it = rtx_types.find(config.encoder_settings.payload_type);
>+  it = rtx_types.find(config.rtp.payload_type);
>   EXPECT_TRUE(it != rtx_types.end() &&
>               it->second == config.rtp.rtx.payload_type);
> 
>@@ -143,12 +158,46 @@ void VerifySendStreamHasRtxTypes(const webrtc::VideoSendStream::Config& config,
> 
> cricket::MediaConfig GetMediaConfig() {
>   cricket::MediaConfig media_config;
>-  media_config.video.enable_cpu_overuse_detection = false;
>+  media_config.video.enable_cpu_adaptation = false;
>   return media_config;
> }
> 
>+// Values from GetMaxDefaultVideoBitrateKbps in webrtcvideoengine.cc.
>+int GetMaxDefaultBitrateBps(size_t width, size_t height) {
>+  if (width * height <= 320 * 240) {
>+    return 600000;
>+  } else if (width * height <= 640 * 480) {
>+    return 1700000;
>+  } else if (width * height <= 960 * 540) {
>+    return 2000000;
>+  } else {
>+    return 2500000;
>+  }
>+}
>+
>+class MockVideoSource : public rtc::VideoSourceInterface<webrtc::VideoFrame> {
>+ public:
>+  MOCK_METHOD2(AddOrUpdateSink,
>+               void(rtc::VideoSinkInterface<webrtc::VideoFrame>* sink,
>+                    const rtc::VideoSinkWants& wants));
>+  MOCK_METHOD1(RemoveSink,
>+               void(rtc::VideoSinkInterface<webrtc::VideoFrame>* sink));
>+};
>+
> }  // namespace
> 
>+#define EXPECT_FRAME_WAIT(c, w, h, t)                        \
>+  EXPECT_EQ_WAIT((c), renderer_.num_rendered_frames(), (t)); \
>+  EXPECT_EQ((w), renderer_.width());                         \
>+  EXPECT_EQ((h), renderer_.height());                        \
>+  EXPECT_EQ(0, renderer_.errors());
>+
>+#define EXPECT_FRAME_ON_RENDERER_WAIT(r, c, w, h, t)   \
>+  EXPECT_EQ_WAIT((c), (r).num_rendered_frames(), (t)); \
>+  EXPECT_EQ((w), (r).width());                         \
>+  EXPECT_EQ((h), (r).height());                        \
>+  EXPECT_EQ(0, (r).errors());
>+
> namespace cricket {
> class WebRtcVideoEngineTest : public ::testing::Test {
>  public:
>@@ -158,30 +207,15 @@ class WebRtcVideoEngineTest : public ::testing::Test {
>         call_(webrtc::Call::Create(webrtc::Call::Config(&event_log_))),
>         encoder_factory_(new cricket::FakeWebRtcVideoEncoderFactory),
>         decoder_factory_(new cricket::FakeWebRtcVideoDecoderFactory),
>-        engine_(std::unique_ptr<cricket::WebRtcVideoEncoderFactory>(
>+        engine_(std::unique_ptr<cricket::FakeWebRtcVideoEncoderFactory>(
>                     encoder_factory_),
>-                std::unique_ptr<cricket::WebRtcVideoDecoderFactory>(
>-                    decoder_factory_)) {
>-    std::vector<VideoCodec> engine_codecs = engine_.codecs();
>-    RTC_DCHECK(!engine_codecs.empty());
>-    bool codec_set = false;
>-    for (const cricket::VideoCodec& codec : engine_codecs) {
>-      if (codec.name == "rtx") {
>-        int associated_payload_type;
>-        if (codec.GetParam(kCodecParamAssociatedPayloadType,
>-                           &associated_payload_type)) {
>-          default_apt_rtx_types_[associated_payload_type] = codec.id;
>-        }
>-      } else if (!codec_set && codec.name != "red" && codec.name != "ulpfec") {
>-        default_codec_ = codec;
>-        codec_set = true;
>-      }
>-    }
>-
>-    RTC_DCHECK(codec_set);
>-  }
>+                std::unique_ptr<cricket::FakeWebRtcVideoDecoderFactory>(
>+                    decoder_factory_)) {}
> 
>  protected:
>+  void AssignDefaultAptRtxTypes();
>+  void AssignDefaultCodec();
>+
>   // Find the index of the codec in the engine with the given name. The codec
>   // must be present.
>   size_t GetEngineCodecIndex(const std::string& name) const;
>@@ -190,9 +224,9 @@ class WebRtcVideoEngineTest : public ::testing::Test {
>   // present.
>   cricket::VideoCodec GetEngineCodec(const std::string& name) const;
> 
>-  VideoMediaChannel* SetUpForExternalEncoderFactory();
>+  VideoMediaChannel* SetSendParamsWithAllSupportedCodecs();
> 
>-  VideoMediaChannel* SetUpForExternalDecoderFactory(
>+  VideoMediaChannel* SetRecvParamsWithSupportedCodecs(
>       const std::vector<VideoCodec>& codecs);
> 
>   void TestExtendedEncoderOveruse(bool use_external_encoder);
>@@ -202,10 +236,6 @@ class WebRtcVideoEngineTest : public ::testing::Test {
>   // Used in WebRtcVideoEngineVoiceTest, but defined here so it's properly
>   // initialized when the constructor is called.
>   std::unique_ptr<webrtc::Call> call_;
>-  // TODO(magjed): Update all tests to use new video codec factories once the
>-  // old factories are deprecated,
>-  // https://bugs.chromium.org/p/webrtc/issues/detail?id=7925.
>-  // These factories are owned by the video engine.
>   cricket::FakeWebRtcVideoEncoderFactory* encoder_factory_;
>   cricket::FakeWebRtcVideoDecoderFactory* decoder_factory_;
>   WebRtcVideoEngine engine_;
>@@ -213,22 +243,10 @@ class WebRtcVideoEngineTest : public ::testing::Test {
>   std::map<int, int> default_apt_rtx_types_;
> };
> 
>-TEST_F(WebRtcVideoEngineTest, AnnouncesVp9AccordingToBuildFlags) {
>-  bool claims_vp9_support = false;
>-  for (const cricket::VideoCodec& codec : engine_.codecs()) {
>-    if (codec.name == "VP9") {
>-      claims_vp9_support = true;
>-      break;
>-    }
>-  }
>-#if defined(RTC_DISABLE_VP9)
>-  EXPECT_FALSE(claims_vp9_support);
>-#else
>-  EXPECT_TRUE(claims_vp9_support);
>-#endif  // defined(RTC_DISABLE_VP9)
>-}
>-
> TEST_F(WebRtcVideoEngineTest, DefaultRtxCodecHasAssociatedPayloadTypeSet) {
>+  encoder_factory_->AddSupportedVideoCodecType("VP8");
>+  AssignDefaultCodec();
>+
>   std::vector<VideoCodec> engine_codecs = engine_.codecs();
>   for (size_t i = 0; i < engine_codecs.size(); ++i) {
>     if (engine_codecs[i].name != kRtxCodecName)
>@@ -291,13 +309,14 @@ TEST_F(WebRtcVideoEngineTest, SupportsVideoRotationHeaderExtension) {
> }
> 
> TEST_F(WebRtcVideoEngineTest, CVOSetHeaderExtensionBeforeCapturer) {
>-  // Allocate the capturer first to prevent early destruction before channel's
>+  // Allocate the source first to prevent early destruction before channel's
>   // dtor is called.
>-  cricket::FakeVideoCapturer capturer;
>+  testing::NiceMock<MockVideoSource> video_source;
> 
>   encoder_factory_->AddSupportedVideoCodecType("VP8");
> 
>-  std::unique_ptr<VideoMediaChannel> channel(SetUpForExternalEncoderFactory());
>+  std::unique_ptr<VideoMediaChannel> channel(
>+      SetSendParamsWithAllSupportedCodecs());
>   EXPECT_TRUE(channel->AddSendStream(StreamParams::CreateLegacy(kSsrc)));
> 
>   // Add CVO extension.
>@@ -308,26 +327,35 @@ TEST_F(WebRtcVideoEngineTest, CVOSetHeaderExtensionBeforeCapturer) {
>       RtpExtension(RtpExtension::kVideoRotationUri, id));
>   EXPECT_TRUE(channel->SetSendParameters(parameters));
> 
>+  EXPECT_CALL(
>+      video_source,
>+      AddOrUpdateSink(testing::_,
>+                      Field(&rtc::VideoSinkWants::rotation_applied, false)));
>   // Set capturer.
>-  EXPECT_TRUE(channel->SetVideoSend(kSsrc, true, nullptr, &capturer));
>+  EXPECT_TRUE(channel->SetVideoSend(kSsrc, nullptr, &video_source));
> 
>   // Verify capturer has turned off applying rotation.
>-  EXPECT_FALSE(capturer.apply_rotation());
>+  testing::Mock::VerifyAndClear(&video_source);
> 
>   // Verify removing header extension turns on applying rotation.
>   parameters.extensions.clear();
>+  EXPECT_CALL(
>+      video_source,
>+      AddOrUpdateSink(testing::_,
>+                      Field(&rtc::VideoSinkWants::rotation_applied, true)));
>+
>   EXPECT_TRUE(channel->SetSendParameters(parameters));
>-  EXPECT_TRUE(capturer.apply_rotation());
> }
> 
> TEST_F(WebRtcVideoEngineTest, CVOSetHeaderExtensionBeforeAddSendStream) {
>-  // Allocate the capturer first to prevent early destruction before channel's
>+  // Allocate the source first to prevent early destruction before channel's
>   // dtor is called.
>-  cricket::FakeVideoCapturer capturer;
>+  testing::NiceMock<MockVideoSource> video_source;
> 
>   encoder_factory_->AddSupportedVideoCodecType("VP8");
> 
>-  std::unique_ptr<VideoMediaChannel> channel(SetUpForExternalEncoderFactory());
>+  std::unique_ptr<VideoMediaChannel> channel(
>+      SetSendParamsWithAllSupportedCodecs());
>   // Add CVO extension.
>   const int id = 1;
>   cricket::VideoSendParameters parameters;
>@@ -337,27 +365,33 @@ TEST_F(WebRtcVideoEngineTest, CVOSetHeaderExtensionBeforeAddSendStream) {
>   EXPECT_TRUE(channel->SetSendParameters(parameters));
>   EXPECT_TRUE(channel->AddSendStream(StreamParams::CreateLegacy(kSsrc)));
> 
>-  // Set capturer.
>-  EXPECT_TRUE(channel->SetVideoSend(kSsrc, true, nullptr, &capturer));
>-
>-  // Verify capturer has turned off applying rotation.
>-  EXPECT_FALSE(capturer.apply_rotation());
>+  // Set source.
>+  EXPECT_CALL(
>+      video_source,
>+      AddOrUpdateSink(testing::_,
>+                      Field(&rtc::VideoSinkWants::rotation_applied, false)));
>+  EXPECT_TRUE(channel->SetVideoSend(kSsrc, nullptr, &video_source));
> }
> 
> TEST_F(WebRtcVideoEngineTest, CVOSetHeaderExtensionAfterCapturer) {
>-  cricket::FakeVideoCapturer capturer;
>+  testing::NiceMock<MockVideoSource> video_source;
> 
>   encoder_factory_->AddSupportedVideoCodecType("VP8");
>   encoder_factory_->AddSupportedVideoCodecType("VP9");
> 
>-  std::unique_ptr<VideoMediaChannel> channel(SetUpForExternalEncoderFactory());
>+  std::unique_ptr<VideoMediaChannel> channel(
>+      SetSendParamsWithAllSupportedCodecs());
>   EXPECT_TRUE(channel->AddSendStream(StreamParams::CreateLegacy(kSsrc)));
> 
>   // Set capturer.
>-  EXPECT_TRUE(channel->SetVideoSend(kSsrc, true, nullptr, &capturer));
>+  EXPECT_CALL(
>+      video_source,
>+      AddOrUpdateSink(testing::_,
>+                      Field(&rtc::VideoSinkWants::rotation_applied, true)));
>+  EXPECT_TRUE(channel->SetVideoSend(kSsrc, nullptr, &video_source));
> 
>   // Verify capturer has turned on applying rotation.
>-  EXPECT_TRUE(capturer.apply_rotation());
>+  testing::Mock::VerifyAndClear(&video_source);
> 
>   // Add CVO extension.
>   const int id = 1;
>@@ -368,18 +402,27 @@ TEST_F(WebRtcVideoEngineTest, CVOSetHeaderExtensionAfterCapturer) {
>       RtpExtension(RtpExtension::kVideoRotationUri, id));
>   // Also remove the first codec to trigger a codec change as well.
>   parameters.codecs.erase(parameters.codecs.begin());
>+  EXPECT_CALL(
>+      video_source,
>+      AddOrUpdateSink(testing::_,
>+                      Field(&rtc::VideoSinkWants::rotation_applied, false)));
>   EXPECT_TRUE(channel->SetSendParameters(parameters));
> 
>   // Verify capturer has turned off applying rotation.
>-  EXPECT_FALSE(capturer.apply_rotation());
>+  testing::Mock::VerifyAndClear(&video_source);
> 
>   // Verify removing header extension turns on applying rotation.
>   parameters.extensions.clear();
>+  EXPECT_CALL(
>+      video_source,
>+      AddOrUpdateSink(testing::_,
>+                      Field(&rtc::VideoSinkWants::rotation_applied, true)));
>   EXPECT_TRUE(channel->SetSendParameters(parameters));
>-  EXPECT_TRUE(capturer.apply_rotation());
> }
> 
> TEST_F(WebRtcVideoEngineTest, SetSendFailsBeforeSettingCodecs) {
>+  encoder_factory_->AddSupportedVideoCodecType("VP8");
>+
>   std::unique_ptr<VideoMediaChannel> channel(
>       engine_.CreateChannel(call_.get(), GetMediaConfig(), VideoOptions()));
> 
>@@ -392,6 +435,8 @@ TEST_F(WebRtcVideoEngineTest, SetSendFailsBeforeSettingCodecs) {
> }
> 
> TEST_F(WebRtcVideoEngineTest, GetStatsWithoutSendCodecsSetDoesNotCrash) {
>+  encoder_factory_->AddSupportedVideoCodecType("VP8");
>+
>   std::unique_ptr<VideoMediaChannel> channel(
>       engine_.CreateChannel(call_.get(), GetMediaConfig(), VideoOptions()));
>   EXPECT_TRUE(channel->AddSendStream(StreamParams::CreateLegacy(123)));
>@@ -399,18 +444,19 @@ TEST_F(WebRtcVideoEngineTest, GetStatsWithoutSendCodecsSetDoesNotCrash) {
>   channel->GetStats(&info);
> }
> 
>-TEST_F(WebRtcVideoEngineTest, UseExternalFactoryForVp8WhenSupported) {
>+TEST_F(WebRtcVideoEngineTest, UseFactoryForVp8WhenSupported) {
>   encoder_factory_->AddSupportedVideoCodecType("VP8");
> 
>-  std::unique_ptr<VideoMediaChannel> channel(SetUpForExternalEncoderFactory());
>+  std::unique_ptr<VideoMediaChannel> channel(
>+      SetSendParamsWithAllSupportedCodecs());
>   channel->OnReadyToSend(true);
> 
>   EXPECT_TRUE(
>       channel->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrc)));
>   EXPECT_EQ(0, encoder_factory_->GetNumCreatedEncoders());
>   EXPECT_TRUE(channel->SetSend(true));
>-  cricket::FakeVideoCapturer capturer;
>-  EXPECT_TRUE(channel->SetVideoSend(kSsrc, true, nullptr, &capturer));
>+  FakeVideoCapturerWithTaskQueue capturer;
>+  EXPECT_TRUE(channel->SetVideoSend(kSsrc, nullptr, &capturer));
>   EXPECT_EQ(cricket::CS_RUNNING,
>             capturer.Start(capturer.GetSupportedFormats()->front()));
>   EXPECT_TRUE(capturer.CaptureFrame());
>@@ -434,25 +480,24 @@ TEST_F(WebRtcVideoEngineTest, UseExternalFactoryForVp8WhenSupported) {
>   EXPECT_EQ(0u, encoder_factory_->encoders().size());
> }
> 
>-// Test that when an external encoder factory supports a codec we don't
>-// internally support, we still add an RTX codec for it.
>-// TODO(deadbeef): Currently this test is only effective if WebRTC is
>-// built with no internal H264 support. This test should be updated
>-// if/when we start adding RTX codecs for unrecognized codec names.
>-TEST_F(WebRtcVideoEngineTest, RtxCodecAddedForExternalCodec) {
>+// Test that when an encoder factory supports H264, we add an RTX
>+// codec for it.
>+// TODO(deadbeef): This test should be updated if/when we start
>+// adding RTX codecs for unrecognized codec names.
>+TEST_F(WebRtcVideoEngineTest, RtxCodecAddedForH264Codec) {
>   using webrtc::H264::ProfileLevelIdToString;
>   using webrtc::H264::ProfileLevelId;
>   using webrtc::H264::kLevel1;
>-  cricket::VideoCodec h264_constrained_baseline("H264");
>-  h264_constrained_baseline.params[kH264FmtpProfileLevelId] =
>+  webrtc::SdpVideoFormat h264_constrained_baseline("H264");
>+  h264_constrained_baseline.parameters[kH264FmtpProfileLevelId] =
>       *ProfileLevelIdToString(
>           ProfileLevelId(webrtc::H264::kProfileConstrainedBaseline, kLevel1));
>-  cricket::VideoCodec h264_constrained_high("H264");
>-  h264_constrained_high.params[kH264FmtpProfileLevelId] =
>+  webrtc::SdpVideoFormat h264_constrained_high("H264");
>+  h264_constrained_high.parameters[kH264FmtpProfileLevelId] =
>       *ProfileLevelIdToString(
>           ProfileLevelId(webrtc::H264::kProfileConstrainedHigh, kLevel1));
>-  cricket::VideoCodec h264_high("H264");
>-  h264_high.params[kH264FmtpProfileLevelId] = *ProfileLevelIdToString(
>+  webrtc::SdpVideoFormat h264_high("H264");
>+  h264_high.parameters[kH264FmtpProfileLevelId] = *ProfileLevelIdToString(
>       ProfileLevelId(webrtc::H264::kProfileHigh, kLevel1));
> 
>   encoder_factory_->AddSupportedVideoCodec(h264_constrained_baseline);
>@@ -464,52 +509,23 @@ TEST_F(WebRtcVideoEngineTest, RtxCodecAddedForExternalCodec) {
>   // Now search for RTX codecs for them. Expect that they all have associated
>   // RTX codecs.
>   EXPECT_TRUE(HasRtxCodec(
>-      codecs, FindMatchingCodec(codecs, h264_constrained_baseline)->id));
>+      codecs,
>+      FindMatchingCodec(codecs, cricket::VideoCodec(h264_constrained_baseline))
>+          ->id));
>   EXPECT_TRUE(HasRtxCodec(
>-      codecs, FindMatchingCodec(codecs, h264_constrained_high)->id));
>+      codecs,
>+      FindMatchingCodec(codecs, cricket::VideoCodec(h264_constrained_high))
>+          ->id));
>   EXPECT_TRUE(HasRtxCodec(
>-      codecs, FindMatchingCodec(codecs, h264_high)->id));
>-}
>-
>-void WebRtcVideoEngineTest::TestExtendedEncoderOveruse(
>-    bool use_external_encoder) {
>-  std::unique_ptr<VideoMediaChannel> channel;
>-  FakeCall* fake_call = new FakeCall(webrtc::Call::Config(&event_log_));
>-  call_.reset(fake_call);
>-  if (use_external_encoder) {
>-    encoder_factory_->AddSupportedVideoCodecType("VP8");
>-    channel.reset(SetUpForExternalEncoderFactory());
>-  } else {
>-    channel.reset(
>-        engine_.CreateChannel(call_.get(), GetMediaConfig(), VideoOptions()));
>-  }
>-  ASSERT_TRUE(
>-      channel->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrc)));
>-  cricket::VideoSendParameters parameters;
>-  parameters.codecs.push_back(GetEngineCodec("VP8"));
>-  EXPECT_TRUE(channel->SetSendParameters(parameters));
>-  EXPECT_TRUE(channel->SetSend(true));
>-  FakeVideoSendStream* stream = fake_call->GetVideoSendStreams()[0];
>-
>-  EXPECT_EQ(use_external_encoder,
>-            stream->GetConfig().encoder_settings.full_overuse_time);
>-  // Remove stream previously added to free the external encoder instance.
>-  EXPECT_TRUE(channel->RemoveSendStream(kSsrc));
>-}
>-
>-TEST_F(WebRtcVideoEngineTest, EnablesFullEncoderTimeForExternalEncoders) {
>-  TestExtendedEncoderOveruse(true);
>-}
>-
>-TEST_F(WebRtcVideoEngineTest, DisablesFullEncoderTimeForNonExternalEncoders) {
>-  TestExtendedEncoderOveruse(false);
>+      codecs, FindMatchingCodec(codecs, cricket::VideoCodec(h264_high))->id));
> }
> 
> #if !defined(RTC_DISABLE_VP9)
> TEST_F(WebRtcVideoEngineTest, CanConstructDecoderForVp9EncoderFactory) {
>   encoder_factory_->AddSupportedVideoCodecType("VP9");
> 
>-  std::unique_ptr<VideoMediaChannel> channel(SetUpForExternalEncoderFactory());
>+  std::unique_ptr<VideoMediaChannel> channel(
>+      SetSendParamsWithAllSupportedCodecs());
> 
>   EXPECT_TRUE(
>       channel->AddRecvStream(cricket::StreamParams::CreateLegacy(kSsrc)));
>@@ -518,15 +534,16 @@ TEST_F(WebRtcVideoEngineTest, CanConstructDecoderForVp9EncoderFactory) {
> 
> TEST_F(WebRtcVideoEngineTest, PropagatesInputFrameTimestamp) {
>   encoder_factory_->AddSupportedVideoCodecType("VP8");
>-  FakeCall* fake_call = new FakeCall(webrtc::Call::Config(&event_log_));
>+  FakeCall* fake_call = new FakeCall();
>   call_.reset(fake_call);
>-  std::unique_ptr<VideoMediaChannel> channel(SetUpForExternalEncoderFactory());
>+  std::unique_ptr<VideoMediaChannel> channel(
>+      SetSendParamsWithAllSupportedCodecs());
> 
>   EXPECT_TRUE(
>       channel->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrc)));
> 
>-  FakeVideoCapturer capturer;
>-  EXPECT_TRUE(channel->SetVideoSend(kSsrc, true, nullptr, &capturer));
>+  FakeVideoCapturerWithTaskQueue capturer;
>+  EXPECT_TRUE(channel->SetVideoSend(kSsrc, nullptr, &capturer));
>   capturer.Start(cricket::VideoFormat(1280, 720,
>                                       cricket::VideoFormat::FpsToInterval(60),
>                                       cricket::FOURCC_I420));
>@@ -570,6 +587,35 @@ TEST_F(WebRtcVideoEngineTest, PropagatesInputFrameTimestamp) {
>   EXPECT_TRUE(channel->RemoveSendStream(kSsrc));
> }
> 
>+void WebRtcVideoEngineTest::AssignDefaultAptRtxTypes() {
>+  std::vector<VideoCodec> engine_codecs = engine_.codecs();
>+  RTC_DCHECK(!engine_codecs.empty());
>+  for (const cricket::VideoCodec& codec : engine_codecs) {
>+    if (codec.name == "rtx") {
>+      int associated_payload_type;
>+      if (codec.GetParam(kCodecParamAssociatedPayloadType,
>+                         &associated_payload_type)) {
>+        default_apt_rtx_types_[associated_payload_type] = codec.id;
>+      }
>+    }
>+  }
>+}
>+
>+void WebRtcVideoEngineTest::AssignDefaultCodec() {
>+  std::vector<VideoCodec> engine_codecs = engine_.codecs();
>+  RTC_DCHECK(!engine_codecs.empty());
>+  bool codec_set = false;
>+  for (const cricket::VideoCodec& codec : engine_codecs) {
>+    if (!codec_set && codec.name != "rtx" && codec.name != "red" &&
>+        codec.name != "ulpfec") {
>+      default_codec_ = codec;
>+      codec_set = true;
>+    }
>+  }
>+
>+  RTC_DCHECK(codec_set);
>+}
>+
> size_t WebRtcVideoEngineTest::GetEngineCodecIndex(
>     const std::string& name) const {
>   const std::vector<cricket::VideoCodec> codecs = engine_.codecs();
>@@ -580,7 +626,7 @@ size_t WebRtcVideoEngineTest::GetEngineCodecIndex(
>     // The tests only use H264 Constrained Baseline. Make sure we don't return
>     // an internal H264 codec from the engine with a different H264 profile.
>     if (CodecNamesEq(name.c_str(), kH264CodecName)) {
>-      const rtc::Optional<webrtc::H264::ProfileLevelId> profile_level_id =
>+      const absl::optional<webrtc::H264::ProfileLevelId> profile_level_id =
>           webrtc::H264::ParseSdpProfileLevelId(engine_codec.params);
>       if (profile_level_id->profile !=
>           webrtc::H264::kProfileConstrainedBaseline) {
>@@ -599,20 +645,27 @@ cricket::VideoCodec WebRtcVideoEngineTest::GetEngineCodec(
>   return engine_.codecs()[GetEngineCodecIndex(name)];
> }
> 
>-VideoMediaChannel* WebRtcVideoEngineTest::SetUpForExternalEncoderFactory() {
>+VideoMediaChannel*
>+WebRtcVideoEngineTest::SetSendParamsWithAllSupportedCodecs() {
>   VideoMediaChannel* channel =
>       engine_.CreateChannel(call_.get(), GetMediaConfig(), VideoOptions());
>   cricket::VideoSendParameters parameters;
>   // We need to look up the codec in the engine to get the correct payload type.
>-  for (const VideoCodec& codec : encoder_factory_->supported_codecs())
>-    parameters.codecs.push_back(GetEngineCodec(codec.name));
>+  for (const webrtc::SdpVideoFormat& format :
>+       encoder_factory_->GetSupportedFormats()) {
>+    cricket::VideoCodec engine_codec = GetEngineCodec(format.name);
>+    if (std::find(parameters.codecs.begin(), parameters.codecs.end(),
>+                  engine_codec) == parameters.codecs.end()) {
>+      parameters.codecs.push_back(engine_codec);
>+    }
>+  }
> 
>   EXPECT_TRUE(channel->SetSendParameters(parameters));
> 
>   return channel;
> }
> 
>-VideoMediaChannel* WebRtcVideoEngineTest::SetUpForExternalDecoderFactory(
>+VideoMediaChannel* WebRtcVideoEngineTest::SetRecvParamsWithSupportedCodecs(
>     const std::vector<VideoCodec>& codecs) {
>   VideoMediaChannel* channel =
>       engine_.CreateChannel(call_.get(), GetMediaConfig(), VideoOptions());
>@@ -626,16 +679,16 @@ VideoMediaChannel* WebRtcVideoEngineTest::SetUpForExternalDecoderFactory(
> TEST_F(WebRtcVideoEngineTest, UsesSimulcastAdapterForVp8Factories) {
>   encoder_factory_->AddSupportedVideoCodecType("VP8");
> 
>-  std::unique_ptr<VideoMediaChannel> channel(SetUpForExternalEncoderFactory());
>+  std::unique_ptr<VideoMediaChannel> channel(
>+      SetSendParamsWithAllSupportedCodecs());
> 
>   std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs3);
> 
>-  EXPECT_TRUE(
>-      channel->AddSendStream(CreateSimStreamParams("cname", ssrcs)));
>+  EXPECT_TRUE(channel->AddSendStream(CreateSimStreamParams("cname", ssrcs)));
>   EXPECT_TRUE(channel->SetSend(true));
> 
>-  cricket::FakeVideoCapturer capturer;
>-  EXPECT_TRUE(channel->SetVideoSend(ssrcs.front(), true, nullptr, &capturer));
>+  FakeVideoCapturerWithTaskQueue capturer;
>+  EXPECT_TRUE(channel->SetVideoSend(ssrcs.front(), nullptr, &capturer));
>   EXPECT_EQ(cricket::CS_RUNNING,
>             capturer.Start(capturer.GetSupportedFormats()->front()));
>   EXPECT_TRUE(capturer.CaptureFrame());
>@@ -654,41 +707,43 @@ TEST_F(WebRtcVideoEngineTest, UsesSimulcastAdapterForVp8Factories) {
>     prev_width = codec_settings.width;
>   }
> 
>-  EXPECT_TRUE(channel->SetVideoSend(ssrcs.front(), true, nullptr, nullptr));
>+  EXPECT_TRUE(channel->SetVideoSend(ssrcs.front(), nullptr, nullptr));
> 
>   channel.reset();
>   ASSERT_EQ(0u, encoder_factory_->encoders().size());
> }
> 
>-TEST_F(WebRtcVideoEngineTest, ChannelWithExternalH264CanChangeToInternalVp8) {
>+TEST_F(WebRtcVideoEngineTest, ChannelWithH264CanChangeToVp8) {
>+  encoder_factory_->AddSupportedVideoCodecType("VP8");
>   encoder_factory_->AddSupportedVideoCodecType("H264");
> 
>-  std::unique_ptr<VideoMediaChannel> channel(SetUpForExternalEncoderFactory());
>-
>-  EXPECT_TRUE(
>-      channel->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrc)));
>-  ASSERT_EQ(1u, encoder_factory_->encoders().size());
>-
>-  cricket::VideoSendParameters parameters;
>-  parameters.codecs.push_back(GetEngineCodec("VP8"));
>-  EXPECT_TRUE(channel->SetSendParameters(parameters));
>-  ASSERT_EQ(0u, encoder_factory_->encoders().size());
>-}
>-
>-TEST_F(WebRtcVideoEngineTest,
>-       DontUseExternalEncoderFactoryForUnsupportedCodecs) {
>-  encoder_factory_->AddSupportedVideoCodecType("H264");
>+  // Set capturer.
>+  FakeVideoCapturerWithTaskQueue capturer;
>+  EXPECT_EQ(cricket::CS_RUNNING,
>+            capturer.Start(capturer.GetSupportedFormats()->front()));
> 
>   std::unique_ptr<VideoMediaChannel> channel(
>       engine_.CreateChannel(call_.get(), GetMediaConfig(), VideoOptions()));
>   cricket::VideoSendParameters parameters;
>-  parameters.codecs.push_back(GetEngineCodec("VP8"));
>+  parameters.codecs.push_back(GetEngineCodec("H264"));
>   EXPECT_TRUE(channel->SetSendParameters(parameters));
> 
>   EXPECT_TRUE(
>       channel->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrc)));
>-  // Make sure DestroyVideoEncoder was called on the factory.
>-  ASSERT_EQ(0u, encoder_factory_->encoders().size());
>+  EXPECT_TRUE(channel->SetVideoSend(kSsrc, nullptr, &capturer));
>+  // Sending one frame will have allocate the encoder.
>+  EXPECT_TRUE(capturer.CaptureFrame());
>+
>+  ASSERT_EQ_WAIT(1u, encoder_factory_->encoders().size(), kTimeout);
>+
>+  cricket::VideoSendParameters new_parameters;
>+  new_parameters.codecs.push_back(GetEngineCodec("VP8"));
>+  EXPECT_TRUE(channel->SetSendParameters(new_parameters));
>+
>+  // Sending one frame will switch encoder.
>+  EXPECT_TRUE(capturer.CaptureFrame());
>+
>+  EXPECT_EQ_WAIT(1u, encoder_factory_->encoders().size(), kTimeout);
> }
> 
> TEST_F(WebRtcVideoEngineTest,
>@@ -704,15 +759,14 @@ TEST_F(WebRtcVideoEngineTest,
> 
>   std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs3);
> 
>-  EXPECT_TRUE(
>-      channel->AddSendStream(CreateSimStreamParams("cname", ssrcs)));
>+  EXPECT_TRUE(channel->AddSendStream(CreateSimStreamParams("cname", ssrcs)));
>   EXPECT_TRUE(channel->SetSend(true));
> 
>   // Send a fake frame, or else the media engine will configure the simulcast
>   // encoder adapter at a low-enough size that it'll only create a single
>   // encoder layer.
>-  cricket::FakeVideoCapturer capturer;
>-  EXPECT_TRUE(channel->SetVideoSend(ssrcs.front(), true, nullptr, &capturer));
>+  FakeVideoCapturerWithTaskQueue capturer;
>+  EXPECT_TRUE(channel->SetVideoSend(ssrcs.front(), nullptr, &capturer));
>   EXPECT_EQ(cricket::CS_RUNNING,
>             capturer.Start(capturer.GetSupportedFormats()->front()));
>   EXPECT_TRUE(capturer.CaptureFrame());
>@@ -740,15 +794,16 @@ TEST_F(WebRtcVideoEngineTest,
> 
>   EXPECT_TRUE(
>       channel->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrc)));
>-  ASSERT_EQ(1u, encoder_factory_->encoders().size());
> 
>   // Send a frame of 720p. This should trigger a "real" encoder initialization.
>   cricket::VideoFormat format(
>       1280, 720, cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420);
>-  cricket::FakeVideoCapturer capturer;
>-  EXPECT_TRUE(channel->SetVideoSend(kSsrc, true, nullptr, &capturer));
>+  FakeVideoCapturerWithTaskQueue capturer;
>+  EXPECT_TRUE(channel->SetVideoSend(kSsrc, nullptr, &capturer));
>   EXPECT_EQ(cricket::CS_RUNNING, capturer.Start(format));
>   EXPECT_TRUE(capturer.CaptureFrame());
>+  ASSERT_TRUE(encoder_factory_->WaitForCreatedVideoEncoders(1));
>+  ASSERT_EQ(1u, encoder_factory_->encoders().size());
>   ASSERT_TRUE(encoder_factory_->encoders()[0]->WaitForInitEncode());
>   EXPECT_EQ(webrtc::kVideoCodecH264,
>             encoder_factory_->encoders()[0]->GetCodecSettings().codecType);
>@@ -758,10 +813,16 @@ TEST_F(WebRtcVideoEngineTest,
>   ASSERT_EQ(0u, encoder_factory_->encoders().size());
> }
> 
>-TEST_F(WebRtcVideoEngineTest, SimulcastDisabledForH264) {
>+TEST_F(WebRtcVideoEngineTest, SimulcastEnabledForH264BehindFieldTrial) {
>+  webrtc::test::ScopedFieldTrials override_field_trials_(
>+      "WebRTC-H264Simulcast/Enabled/");
>   encoder_factory_->AddSupportedVideoCodecType("H264");
> 
>-  std::unique_ptr<VideoMediaChannel> channel(SetUpForExternalEncoderFactory());
>+  std::unique_ptr<VideoMediaChannel> channel(
>+      engine_.CreateChannel(call_.get(), GetMediaConfig(), VideoOptions()));
>+  cricket::VideoSendParameters parameters;
>+  parameters.codecs.push_back(GetEngineCodec("H264"));
>+  EXPECT_TRUE(channel->SetSendParameters(parameters));
> 
>   const std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs3);
>   EXPECT_TRUE(
>@@ -770,17 +831,18 @@ TEST_F(WebRtcVideoEngineTest, SimulcastDisabledForH264) {
>   // Send a frame of 720p. This should trigger a "real" encoder initialization.
>   cricket::VideoFormat format(
>       1280, 720, cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420);
>-  cricket::FakeVideoCapturer capturer;
>-  EXPECT_TRUE(channel->SetVideoSend(ssrcs[0], true, nullptr, &capturer));
>+  FakeVideoCapturerWithTaskQueue capturer;
>+  EXPECT_TRUE(channel->SetVideoSend(ssrcs[0], nullptr, &capturer));
>   EXPECT_EQ(cricket::CS_RUNNING, capturer.Start(format));
>   EXPECT_TRUE(capturer.CaptureFrame());
> 
>+  ASSERT_TRUE(encoder_factory_->WaitForCreatedVideoEncoders(1));
>   ASSERT_EQ(1u, encoder_factory_->encoders().size());
>   FakeWebRtcVideoEncoder* encoder = encoder_factory_->encoders()[0];
>   ASSERT_TRUE(encoder_factory_->encoders()[0]->WaitForInitEncode());
>   EXPECT_EQ(webrtc::kVideoCodecH264, encoder->GetCodecSettings().codecType);
>-  EXPECT_EQ(1u, encoder->GetCodecSettings().numberOfSimulcastStreams);
>-  EXPECT_TRUE(channel->SetVideoSend(ssrcs[0], true, nullptr, nullptr));
>+  EXPECT_LT(1u, encoder->GetCodecSettings().numberOfSimulcastStreams);
>+  EXPECT_TRUE(channel->SetVideoSend(ssrcs[0], nullptr, nullptr));
> }
> 
> // Test that the FlexFEC field trial properly alters the output of
>@@ -789,6 +851,8 @@ TEST_F(WebRtcVideoEngineTest, SimulcastDisabledForH264) {
> // TODO(brandtr): Remove this test, when the FlexFEC field trial is gone.
> TEST_F(WebRtcVideoEngineTest,
>        Flexfec03SupportedAsInternalCodecBehindFieldTrial) {
>+  encoder_factory_->AddSupportedVideoCodecType("VP8");
>+
>   auto is_flexfec = [](const VideoCodec& codec) {
>     if (codec.name == "flexfec-03")
>       return true;
>@@ -808,21 +872,21 @@ TEST_F(WebRtcVideoEngineTest,
>             std::find_if(codecs_after.begin(), codecs_after.end(), is_flexfec));
> }
> 
>-// Test that external codecs are added after internal SW codecs.
>-TEST_F(WebRtcVideoEngineTest, ReportSupportedExternalCodecs) {
>-  const char* kFakeExternalCodecName = "FakeExternalCodec";
>-  encoder_factory_->AddSupportedVideoCodecType(kFakeExternalCodecName);
>+// Test that codecs are added in the order they are reported from the factory.
>+TEST_F(WebRtcVideoEngineTest, ReportSupportedCodecs) {
>+  encoder_factory_->AddSupportedVideoCodecType("VP8");
>+  const char* kFakeCodecName = "FakeCodec";
>+  encoder_factory_->AddSupportedVideoCodecType(kFakeCodecName);
> 
>-  // The external codec should appear after the internal codec in the vector.
>+  // The last reported codec should appear after the first codec in the vector.
>   const size_t vp8_index = GetEngineCodecIndex("VP8");
>-  const size_t fake_external_codec_index =
>-      GetEngineCodecIndex(kFakeExternalCodecName);
>-  EXPECT_LT(vp8_index, fake_external_codec_index);
>+  const size_t fake_codec_index = GetEngineCodecIndex(kFakeCodecName);
>+  EXPECT_LT(vp8_index, fake_codec_index);
> }
> 
>-// Test that an external codec that was added after the engine was initialized
>+// Test that a codec that was added after the engine was initialized
> // does show up in the codec list after it was added.
>-TEST_F(WebRtcVideoEngineTest, ReportSupportedExternalCodecsWithAddedCodec) {
>+TEST_F(WebRtcVideoEngineTest, ReportSupportedAddedCodec) {
>   const char* kFakeExternalCodecName1 = "FakeExternalCodec1";
>   const char* kFakeExternalCodecName2 = "FakeExternalCodec2";
> 
>@@ -834,7 +898,8 @@ TEST_F(WebRtcVideoEngineTest, ReportSupportedExternalCodecsWithAddedCodec) {
>   // Add second codec.
>   encoder_factory_->AddSupportedVideoCodecType(kFakeExternalCodecName2);
>   std::vector<cricket::VideoCodec> codecs_after(engine_.codecs());
>-  EXPECT_EQ(codecs_before.size() + 1, codecs_after.size());
>+  // The codec itself and RTX should have been added.
>+  EXPECT_EQ(codecs_before.size() + 2, codecs_after.size());
> 
>   // Check that both fake codecs are present and that the second fake codec
>   // appears after the first fake codec.
>@@ -843,13 +908,22 @@ TEST_F(WebRtcVideoEngineTest, ReportSupportedExternalCodecsWithAddedCodec) {
>   EXPECT_LT(fake_codec_index1, fake_codec_index2);
> }
> 
>-TEST_F(WebRtcVideoEngineTest, RegisterExternalDecodersIfSupported) {
>-  decoder_factory_->AddSupportedVideoCodecType(webrtc::kVideoCodecVP8);
>+TEST_F(WebRtcVideoEngineTest, ReportRtxForExternalCodec) {
>+  const char* kFakeCodecName = "FakeCodec";
>+  encoder_factory_->AddSupportedVideoCodecType(kFakeCodecName);
>+
>+  const size_t fake_codec_index = GetEngineCodecIndex(kFakeCodecName);
>+  EXPECT_EQ("rtx", engine_.codecs().at(fake_codec_index + 1).name);
>+}
>+
>+TEST_F(WebRtcVideoEngineTest, RegisterDecodersIfSupported) {
>+  encoder_factory_->AddSupportedVideoCodecType("VP8");
>+  decoder_factory_->AddSupportedVideoCodecType(webrtc::SdpVideoFormat("VP8"));
>   cricket::VideoRecvParameters parameters;
>   parameters.codecs.push_back(GetEngineCodec("VP8"));
> 
>   std::unique_ptr<VideoMediaChannel> channel(
>-      SetUpForExternalDecoderFactory(parameters.codecs));
>+      SetRecvParamsWithSupportedCodecs(parameters.codecs));
> 
>   EXPECT_TRUE(
>       channel->AddRecvStream(cricket::StreamParams::CreateLegacy(kSsrc)));
>@@ -864,19 +938,19 @@ TEST_F(WebRtcVideoEngineTest, RegisterExternalDecodersIfSupported) {
>   EXPECT_EQ(0u, decoder_factory_->decoders().size());
> }
> 
>-// Verifies that we can set up decoders that are not internally supported.
>-TEST_F(WebRtcVideoEngineTest, RegisterExternalH264DecoderIfSupported) {
>+// Verifies that we can set up decoders.
>+TEST_F(WebRtcVideoEngineTest, RegisterH264DecoderIfSupported) {
>   // TODO(pbos): Do not assume that encoder/decoder support is symmetric. We
>   // can't even query the WebRtcVideoDecoderFactory for supported codecs.
>   // For now we add a FakeWebRtcVideoEncoderFactory to add H264 to supported
>   // codecs.
>   encoder_factory_->AddSupportedVideoCodecType("H264");
>-  decoder_factory_->AddSupportedVideoCodecType(webrtc::kVideoCodecH264);
>+  decoder_factory_->AddSupportedVideoCodecType(webrtc::SdpVideoFormat("H264"));
>   std::vector<cricket::VideoCodec> codecs;
>   codecs.push_back(GetEngineCodec("H264"));
> 
>   std::unique_ptr<VideoMediaChannel> channel(
>-      SetUpForExternalDecoderFactory(codecs));
>+      SetRecvParamsWithSupportedCodecs(codecs));
> 
>   EXPECT_TRUE(
>       channel->AddRecvStream(cricket::StreamParams::CreateLegacy(kSsrc)));
>@@ -962,12 +1036,16 @@ TEST(WebRtcVideoEngineNewVideoCodecFactoryTest, Vp8) {
>   const webrtc::SdpVideoFormat format("VP8");
>   EXPECT_CALL(*encoder_factory, QueryVideoEncoder(format))
>       .WillRepeatedly(testing::Return(codec_info));
>-  FakeWebRtcVideoEncoder* const encoder = new FakeWebRtcVideoEncoder();
>+  FakeWebRtcVideoEncoder* const encoder = new FakeWebRtcVideoEncoder(nullptr);
>+  rtc::Event encoder_created(false, false);
>   EXPECT_CALL(*encoder_factory, CreateVideoEncoderProxy(format))
>-      .WillOnce(testing::Return(encoder));
>+      .WillOnce(
>+          ::testing::DoAll(::testing::InvokeWithoutArgs(
>+                               [&encoder_created]() { encoder_created.Set(); }),
>+                           ::testing::Return(encoder)));
> 
>   // Mock decoder creation. |engine| take ownership of the decoder.
>-  FakeWebRtcVideoDecoder* const decoder = new FakeWebRtcVideoDecoder();
>+  FakeWebRtcVideoDecoder* const decoder = new FakeWebRtcVideoDecoder(nullptr);
>   EXPECT_CALL(*decoder_factory, CreateVideoDecoderProxy(format))
>       .WillOnce(testing::Return(decoder));
> 
>@@ -988,6 +1066,15 @@ TEST(WebRtcVideoEngineNewVideoCodecFactoryTest, Vp8) {
>       send_channel->AddSendStream(StreamParams::CreateLegacy(send_ssrc)));
>   EXPECT_TRUE(send_channel->SetSend(true));
> 
>+  // Set capturer.
>+  FakeVideoCapturerWithTaskQueue capturer;
>+  EXPECT_EQ(cricket::CS_RUNNING,
>+            capturer.Start(capturer.GetSupportedFormats()->front()));
>+  EXPECT_TRUE(send_channel->SetVideoSend(send_ssrc, nullptr, &capturer));
>+  // Sending one frame will allocate the encoder.
>+  EXPECT_TRUE(capturer.CaptureFrame());
>+  encoder_created.Wait(kTimeout);
>+
>   // Create recv channel.
>   const int recv_ssrc = 321;
>   std::unique_ptr<VideoMediaChannel> recv_channel(
>@@ -1043,48 +1130,11 @@ TEST(WebRtcVideoEngineNewVideoCodecFactoryTest, NullDecoder) {
>   EXPECT_TRUE(recv_channel->RemoveRecvStream(recv_ssrc));
> }
> 
>-class WebRtcVideoChannelBaseTest
>-    : public VideoMediaChannelTest<WebRtcVideoEngine, WebRtcVideoChannel> {
>- protected:
>-  typedef VideoMediaChannelTest<WebRtcVideoEngine, WebRtcVideoChannel> Base;
>-
>-  cricket::VideoCodec GetEngineCodec(const std::string& name) {
>-    for (const cricket::VideoCodec& engine_codec : engine_.codecs()) {
>-      if (CodecNamesEq(name, engine_codec.name))
>-        return engine_codec;
>-    }
>-    // This point should never be reached.
>-    ADD_FAILURE() << "Unrecognized codec name: " << name;
>-    return cricket::VideoCodec();
>-  }
>-
>-  cricket::VideoCodec DefaultCodec() override { return GetEngineCodec("VP8"); }
>-};
>-
>-// Verifies that id given in stream params is passed to the decoder factory.
>-TEST_F(WebRtcVideoEngineTest, StreamParamsIdPassedToDecoderFactory) {
>-  decoder_factory_->AddSupportedVideoCodecType(webrtc::kVideoCodecVP8);
>-  cricket::VideoRecvParameters parameters;
>-  parameters.codecs.push_back(GetEngineCodec("VP8"));
>-
>-  std::unique_ptr<VideoMediaChannel> channel(
>-      SetUpForExternalDecoderFactory(parameters.codecs));
>-
>-  StreamParams sp = cricket::StreamParams::CreateLegacy(kSsrc);
>-  sp.id = "FakeStreamParamsId";
>-  EXPECT_TRUE(channel->AddRecvStream(sp));
>-  EXPECT_EQ(1u, decoder_factory_->decoders().size());
>-
>-  std::vector<cricket::VideoDecoderParams> params = decoder_factory_->params();
>-  ASSERT_EQ(1u, params.size());
>-  EXPECT_EQ(sp.id, params[0].receive_stream_id);
>-}
>-
> TEST_F(WebRtcVideoEngineTest, DISABLED_RecreatesEncoderOnContentTypeChange) {
>   encoder_factory_->AddSupportedVideoCodecType("VP8");
>-  std::unique_ptr<FakeCall> fake_call(
>-      new FakeCall(webrtc::Call::Config(&event_log_)));
>-  std::unique_ptr<VideoMediaChannel> channel(SetUpForExternalEncoderFactory());
>+  std::unique_ptr<FakeCall> fake_call(new FakeCall());
>+  std::unique_ptr<VideoMediaChannel> channel(
>+      SetSendParamsWithAllSupportedCodecs());
>   ASSERT_TRUE(
>       channel->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrc)));
>   cricket::VideoCodec codec = GetEngineCodec("VP8");
>@@ -1094,45 +1144,45 @@ TEST_F(WebRtcVideoEngineTest, DISABLED_RecreatesEncoderOnContentTypeChange) {
>   channel->SetSend(true);
>   ASSERT_TRUE(channel->SetSendParameters(parameters));
> 
>-  cricket::FakeVideoCapturer capturer;
>+  FakeVideoCapturerWithTaskQueue capturer;
>   VideoOptions options;
>-  EXPECT_TRUE(channel->SetVideoSend(kSsrc, true, &options, &capturer));
>+  EXPECT_TRUE(channel->SetVideoSend(kSsrc, &options, &capturer));
> 
>   EXPECT_EQ(cricket::CS_RUNNING,
>             capturer.Start(capturer.GetSupportedFormats()->front()));
>   EXPECT_TRUE(capturer.CaptureFrame());
>   ASSERT_TRUE(encoder_factory_->WaitForCreatedVideoEncoders(1));
>-  EXPECT_EQ(webrtc::kRealtimeVideo,
>+  EXPECT_EQ(webrtc::VideoCodecMode::kRealtimeVideo,
>             encoder_factory_->encoders().back()->GetCodecSettings().mode);
> 
>-  EXPECT_TRUE(channel->SetVideoSend(kSsrc, true, &options, &capturer));
>+  EXPECT_TRUE(channel->SetVideoSend(kSsrc, &options, &capturer));
>   EXPECT_TRUE(capturer.CaptureFrame());
>   // No change in content type, keep current encoder.
>   EXPECT_EQ(1, encoder_factory_->GetNumCreatedEncoders());
> 
>   options.is_screencast.emplace(true);
>-  EXPECT_TRUE(channel->SetVideoSend(kSsrc, true, &options, &capturer));
>+  EXPECT_TRUE(channel->SetVideoSend(kSsrc, &options, &capturer));
>   EXPECT_TRUE(capturer.CaptureFrame());
>   // Change to screen content, recreate encoder. For the simulcast encoder
>   // adapter case, this will result in two calls since InitEncode triggers a
>   // a new instance.
>   ASSERT_TRUE(encoder_factory_->WaitForCreatedVideoEncoders(2));
>-  EXPECT_EQ(webrtc::kScreensharing,
>+  EXPECT_EQ(webrtc::VideoCodecMode::kScreensharing,
>             encoder_factory_->encoders().back()->GetCodecSettings().mode);
> 
>-  EXPECT_TRUE(channel->SetVideoSend(kSsrc, true, &options, &capturer));
>+  EXPECT_TRUE(channel->SetVideoSend(kSsrc, &options, &capturer));
>   EXPECT_TRUE(capturer.CaptureFrame());
>   // Still screen content, no need to update encoder.
>   EXPECT_EQ(2, encoder_factory_->GetNumCreatedEncoders());
> 
>   options.is_screencast.emplace(false);
>   options.video_noise_reduction.emplace(false);
>-  EXPECT_TRUE(channel->SetVideoSend(kSsrc, true, &options, &capturer));
>+  EXPECT_TRUE(channel->SetVideoSend(kSsrc, &options, &capturer));
>   // Change back to regular video content, update encoder. Also change
>   // a non |is_screencast| option just to verify it doesn't affect recreation.
>   EXPECT_TRUE(capturer.CaptureFrame());
>   ASSERT_TRUE(encoder_factory_->WaitForCreatedVideoEncoders(3));
>-  EXPECT_EQ(webrtc::kRealtimeVideo,
>+  EXPECT_EQ(webrtc::VideoCodecMode::kRealtimeVideo,
>             encoder_factory_->encoders().back()->GetCodecSettings().mode);
> 
>   // Remove stream previously added to free the external encoder instance.
>@@ -1140,65 +1190,739 @@ TEST_F(WebRtcVideoEngineTest, DISABLED_RecreatesEncoderOnContentTypeChange) {
>   EXPECT_EQ(0u, encoder_factory_->encoders().size());
> }
> 
>+class WebRtcVideoChannelBaseTest : public testing::Test {
>+ protected:
>+  WebRtcVideoChannelBaseTest()
>+      : call_(webrtc::Call::Create(webrtc::Call::Config(&event_log_))),
>+        engine_(webrtc::CreateBuiltinVideoEncoderFactory(),
>+                webrtc::CreateBuiltinVideoDecoderFactory()) {}
>+
>+  virtual void SetUp() {
>+    cricket::MediaConfig media_config;
>+    // Disabling cpu overuse detection actually disables quality scaling too; it
>+    // implies DegradationPreference kMaintainResolution. Automatic scaling
>+    // needs to be disabled, otherwise, tests which check the size of received
>+    // frames become flaky.
>+    media_config.video.enable_cpu_adaptation = false;
>+    channel_.reset(engine_.CreateChannel(call_.get(), media_config,
>+                                         cricket::VideoOptions()));
>+    channel_->OnReadyToSend(true);
>+    EXPECT_TRUE(channel_.get() != NULL);
>+    network_interface_.SetDestination(channel_.get());
>+    channel_->SetInterface(&network_interface_);
>+    cricket::VideoRecvParameters parameters;
>+    parameters.codecs = engine_.codecs();
>+    channel_->SetRecvParameters(parameters);
>+    EXPECT_TRUE(channel_->AddSendStream(DefaultSendStreamParams()));
>+    video_capturer_.reset(CreateFakeVideoCapturer());
>+    cricket::VideoFormat format(640, 480,
>+                                cricket::VideoFormat::FpsToInterval(kFramerate),
>+                                cricket::FOURCC_I420);
>+    EXPECT_EQ(cricket::CS_RUNNING, video_capturer_->Start(format));
>+    EXPECT_TRUE(channel_->SetVideoSend(kSsrc, nullptr, video_capturer_.get()));
>+  }
>+
>+  virtual cricket::FakeVideoCapturerWithTaskQueue* CreateFakeVideoCapturer() {
>+    return new cricket::FakeVideoCapturerWithTaskQueue();
>+  }
>+
>+  // Utility method to setup an additional stream to send and receive video.
>+  // Used to test send and recv between two streams.
>+  void SetUpSecondStream() {
>+    SetUpSecondStreamWithNoRecv();
>+    // Setup recv for second stream.
>+    EXPECT_TRUE(channel_->AddRecvStream(
>+        cricket::StreamParams::CreateLegacy(kSsrc + 2)));
>+    // Make the second renderer available for use by a new stream.
>+    EXPECT_TRUE(channel_->SetSink(kSsrc + 2, &renderer2_));
>+  }
>+  // Setup an additional stream just to send video. Defer add recv stream.
>+  // This is required if you want to test unsignalled recv of video rtp packets.
>+  void SetUpSecondStreamWithNoRecv() {
>+    // SetUp() already added kSsrc make sure duplicate SSRCs cant be added.
>+    EXPECT_TRUE(
>+        channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(kSsrc)));
>+    EXPECT_TRUE(channel_->SetSink(kSsrc, &renderer_));
>+    EXPECT_FALSE(
>+        channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrc)));
>+    EXPECT_TRUE(channel_->AddSendStream(
>+        cricket::StreamParams::CreateLegacy(kSsrc + 2)));
>+    // We dont add recv for the second stream.
>+
>+    // Setup the receive and renderer for second stream after send.
>+    video_capturer_2_.reset(CreateFakeVideoCapturer());
>+    cricket::VideoFormat format(640, 480,
>+                                cricket::VideoFormat::FpsToInterval(kFramerate),
>+                                cricket::FOURCC_I420);
>+    EXPECT_EQ(cricket::CS_RUNNING, video_capturer_2_->Start(format));
>+
>+    EXPECT_TRUE(
>+        channel_->SetVideoSend(kSsrc + 2, nullptr, video_capturer_2_.get()));
>+  }
>+  virtual void TearDown() { channel_.reset(); }
>+  bool SetDefaultCodec() { return SetOneCodec(DefaultCodec()); }
>+
>+  bool SetOneCodec(const cricket::VideoCodec& codec) {
>+    cricket::VideoFormat capture_format(
>+        kVideoWidth, kVideoHeight,
>+        cricket::VideoFormat::FpsToInterval(kFramerate), cricket::FOURCC_I420);
>+
>+    if (video_capturer_) {
>+      EXPECT_EQ(cricket::CS_RUNNING, video_capturer_->Start(capture_format));
>+    }
>+    if (video_capturer_2_) {
>+      EXPECT_EQ(cricket::CS_RUNNING, video_capturer_2_->Start(capture_format));
>+    }
>+
>+    bool sending = channel_->sending();
>+    bool success = SetSend(false);
>+    if (success) {
>+      cricket::VideoSendParameters parameters;
>+      parameters.codecs.push_back(codec);
>+      success = channel_->SetSendParameters(parameters);
>+    }
>+    if (success) {
>+      success = SetSend(sending);
>+    }
>+    return success;
>+  }
>+  bool SetSend(bool send) { return channel_->SetSend(send); }
>+  bool SendFrame() {
>+    if (video_capturer_2_) {
>+      video_capturer_2_->CaptureFrame();
>+    }
>+    return video_capturer_.get() && video_capturer_->CaptureFrame();
>+  }
>+  bool WaitAndSendFrame(int wait_ms) {
>+    bool ret = rtc::Thread::Current()->ProcessMessages(wait_ms);
>+    ret &= SendFrame();
>+    return ret;
>+  }
>+  int NumRtpBytes() { return network_interface_.NumRtpBytes(); }
>+  int NumRtpBytes(uint32_t ssrc) {
>+    return network_interface_.NumRtpBytes(ssrc);
>+  }
>+  int NumRtpPackets() { return network_interface_.NumRtpPackets(); }
>+  int NumRtpPackets(uint32_t ssrc) {
>+    return network_interface_.NumRtpPackets(ssrc);
>+  }
>+  int NumSentSsrcs() { return network_interface_.NumSentSsrcs(); }
>+  const rtc::CopyOnWriteBuffer* GetRtpPacket(int index) {
>+    return network_interface_.GetRtpPacket(index);
>+  }
>+  static int GetPayloadType(const rtc::CopyOnWriteBuffer* p) {
>+    webrtc::RTPHeader header;
>+    EXPECT_TRUE(ParseRtpPacket(p, &header));
>+    return header.payloadType;
>+  }
>+
>+  static bool ParseRtpPacket(const rtc::CopyOnWriteBuffer* p,
>+                             webrtc::RTPHeader* header) {
>+    std::unique_ptr<webrtc::RtpHeaderParser> parser(
>+        webrtc::RtpHeaderParser::Create());
>+    return parser->Parse(p->cdata(), p->size(), header);
>+  }
>+
>+  // Tests that we can send and receive frames.
>+  void SendAndReceive(const cricket::VideoCodec& codec) {
>+    EXPECT_TRUE(SetOneCodec(codec));
>+    EXPECT_TRUE(SetSend(true));
>+    EXPECT_TRUE(channel_->SetSink(kDefaultReceiveSsrc, &renderer_));
>+    EXPECT_EQ(0, renderer_.num_rendered_frames());
>+    EXPECT_TRUE(SendFrame());
>+    EXPECT_FRAME_WAIT(1, kVideoWidth, kVideoHeight, kTimeout);
>+    std::unique_ptr<const rtc::CopyOnWriteBuffer> p(GetRtpPacket(0));
>+    EXPECT_EQ(codec.id, GetPayloadType(p.get()));
>+  }
>+
>+  void SendReceiveManyAndGetStats(const cricket::VideoCodec& codec,
>+                                  int duration_sec,
>+                                  int fps) {
>+    EXPECT_TRUE(SetOneCodec(codec));
>+    EXPECT_TRUE(SetSend(true));
>+    EXPECT_TRUE(channel_->SetSink(kDefaultReceiveSsrc, &renderer_));
>+    EXPECT_EQ(0, renderer_.num_rendered_frames());
>+    for (int i = 0; i < duration_sec; ++i) {
>+      for (int frame = 1; frame <= fps; ++frame) {
>+        EXPECT_TRUE(WaitAndSendFrame(1000 / fps));
>+        EXPECT_FRAME_WAIT(frame + i * fps, kVideoWidth, kVideoHeight, kTimeout);
>+      }
>+    }
>+    std::unique_ptr<const rtc::CopyOnWriteBuffer> p(GetRtpPacket(0));
>+    EXPECT_EQ(codec.id, GetPayloadType(p.get()));
>+  }
>+
>+  cricket::VideoSenderInfo GetSenderStats(size_t i) {
>+    cricket::VideoMediaInfo info;
>+    EXPECT_TRUE(channel_->GetStats(&info));
>+    return info.senders[i];
>+  }
>+
>+  cricket::VideoReceiverInfo GetReceiverStats(size_t i) {
>+    cricket::VideoMediaInfo info;
>+    EXPECT_TRUE(channel_->GetStats(&info));
>+    return info.receivers[i];
>+  }
>+
>+  // Two streams one channel tests.
>+
>+  // Tests that we can send and receive frames.
>+  void TwoStreamsSendAndReceive(const cricket::VideoCodec& codec) {
>+    SetUpSecondStream();
>+    // Test sending and receiving on first stream.
>+    SendAndReceive(codec);
>+    // Test sending and receiving on second stream.
>+    EXPECT_EQ_WAIT(1, renderer2_.num_rendered_frames(), kTimeout);
>+    EXPECT_GT(NumRtpPackets(), 0);
>+    EXPECT_EQ(1, renderer2_.num_rendered_frames());
>+  }
>+
>+  cricket::VideoCodec GetEngineCodec(const std::string& name) {
>+    for (const cricket::VideoCodec& engine_codec : engine_.codecs()) {
>+      if (CodecNamesEq(name, engine_codec.name))
>+        return engine_codec;
>+    }
>+    // This point should never be reached.
>+    ADD_FAILURE() << "Unrecognized codec name: " << name;
>+    return cricket::VideoCodec();
>+  }
>+
>+  cricket::VideoCodec DefaultCodec() { return GetEngineCodec("VP8"); }
>+
>+  cricket::StreamParams DefaultSendStreamParams() {
>+    return cricket::StreamParams::CreateLegacy(kSsrc);
>+  }
>+
>+  webrtc::RtcEventLogNullImpl event_log_;
>+  const std::unique_ptr<webrtc::Call> call_;
>+  WebRtcVideoEngine engine_;
>+  std::unique_ptr<cricket::FakeVideoCapturerWithTaskQueue> video_capturer_;
>+  std::unique_ptr<cricket::FakeVideoCapturerWithTaskQueue> video_capturer_2_;
>+  std::unique_ptr<WebRtcVideoChannel> channel_;
>+  cricket::FakeNetworkInterface network_interface_;
>+  cricket::FakeVideoRenderer renderer_;
>+
>+  // Used by test cases where 2 streams are run on the same channel.
>+  cricket::FakeVideoRenderer renderer2_;
>+};
>+
>+// Test that SetSend works.
> TEST_F(WebRtcVideoChannelBaseTest, SetSend) {
>-  Base::SetSend();
>-}
>+  EXPECT_FALSE(channel_->sending());
>+  EXPECT_TRUE(channel_->SetVideoSend(kSsrc, nullptr, video_capturer_.get()));
>+  EXPECT_TRUE(SetOneCodec(DefaultCodec()));
>+  EXPECT_FALSE(channel_->sending());
>+  EXPECT_TRUE(SetSend(true));
>+  EXPECT_TRUE(channel_->sending());
>+  EXPECT_TRUE(SendFrame());
>+  EXPECT_TRUE_WAIT(NumRtpPackets() > 0, kTimeout);
>+  EXPECT_TRUE(SetSend(false));
>+  EXPECT_FALSE(channel_->sending());
>+}
>+
>+// Test that SetSend fails without codecs being set.
> TEST_F(WebRtcVideoChannelBaseTest, SetSendWithoutCodecs) {
>-  Base::SetSendWithoutCodecs();
>+  EXPECT_FALSE(channel_->sending());
>+  EXPECT_FALSE(SetSend(true));
>+  EXPECT_FALSE(channel_->sending());
> }
>+
>+// Test that we properly set the send and recv buffer sizes by the time
>+// SetSend is called.
> TEST_F(WebRtcVideoChannelBaseTest, SetSendSetsTransportBufferSizes) {
>-  Base::SetSendSetsTransportBufferSizes();
>+  EXPECT_TRUE(SetOneCodec(DefaultCodec()));
>+  EXPECT_TRUE(SetSend(true));
>+  EXPECT_EQ(64 * 1024, network_interface_.sendbuf_size());
>+  EXPECT_EQ(64 * 1024, network_interface_.recvbuf_size());
> }
> 
>+// Test that stats work properly for a 1-1 call.
> TEST_F(WebRtcVideoChannelBaseTest, GetStats) {
>-  Base::GetStats();
>-}
>+  const int kDurationSec = 3;
>+  const int kFps = 10;
>+  SendReceiveManyAndGetStats(DefaultCodec(), kDurationSec, kFps);
>+
>+  cricket::VideoMediaInfo info;
>+  EXPECT_TRUE(channel_->GetStats(&info));
>+
>+  ASSERT_EQ(1U, info.senders.size());
>+  // TODO(whyuan): bytes_sent and bytes_rcvd are different. Are both payload?
>+  // For webrtc, bytes_sent does not include the RTP header length.
>+  EXPECT_GT(info.senders[0].bytes_sent, 0);
>+  EXPECT_EQ(NumRtpPackets(), info.senders[0].packets_sent);
>+  EXPECT_EQ(0.0, info.senders[0].fraction_lost);
>+  ASSERT_TRUE(info.senders[0].codec_payload_type);
>+  EXPECT_EQ(DefaultCodec().id, *info.senders[0].codec_payload_type);
>+  EXPECT_EQ(0, info.senders[0].firs_rcvd);
>+  EXPECT_EQ(0, info.senders[0].plis_rcvd);
>+  EXPECT_EQ(0, info.senders[0].nacks_rcvd);
>+  EXPECT_EQ(kVideoWidth, info.senders[0].send_frame_width);
>+  EXPECT_EQ(kVideoHeight, info.senders[0].send_frame_height);
>+  EXPECT_GT(info.senders[0].framerate_input, 0);
>+  EXPECT_GT(info.senders[0].framerate_sent, 0);
>+
>+  EXPECT_EQ(1U, info.send_codecs.count(DefaultCodec().id));
>+  EXPECT_EQ(DefaultCodec().ToCodecParameters(),
>+            info.send_codecs[DefaultCodec().id]);
>+
>+  ASSERT_EQ(1U, info.receivers.size());
>+  EXPECT_EQ(1U, info.senders[0].ssrcs().size());
>+  EXPECT_EQ(1U, info.receivers[0].ssrcs().size());
>+  EXPECT_EQ(info.senders[0].ssrcs()[0], info.receivers[0].ssrcs()[0]);
>+  ASSERT_TRUE(info.receivers[0].codec_payload_type);
>+  EXPECT_EQ(DefaultCodec().id, *info.receivers[0].codec_payload_type);
>+  EXPECT_EQ(NumRtpBytes(), info.receivers[0].bytes_rcvd);
>+  EXPECT_EQ(NumRtpPackets(), info.receivers[0].packets_rcvd);
>+  EXPECT_EQ(0.0, info.receivers[0].fraction_lost);
>+  EXPECT_EQ(0, info.receivers[0].packets_lost);
>+  // TODO(asapersson): Not set for webrtc. Handle missing stats.
>+  // EXPECT_EQ(0, info.receivers[0].packets_concealed);
>+  EXPECT_EQ(0, info.receivers[0].firs_sent);
>+  EXPECT_EQ(0, info.receivers[0].plis_sent);
>+  EXPECT_EQ(0, info.receivers[0].nacks_sent);
>+  EXPECT_EQ(kVideoWidth, info.receivers[0].frame_width);
>+  EXPECT_EQ(kVideoHeight, info.receivers[0].frame_height);
>+  EXPECT_GT(info.receivers[0].framerate_rcvd, 0);
>+  EXPECT_GT(info.receivers[0].framerate_decoded, 0);
>+  EXPECT_GT(info.receivers[0].framerate_output, 0);
>+
>+  EXPECT_EQ(1U, info.receive_codecs.count(DefaultCodec().id));
>+  EXPECT_EQ(DefaultCodec().ToCodecParameters(),
>+            info.receive_codecs[DefaultCodec().id]);
>+}
>+
>+// Test that stats work properly for a conf call with multiple recv streams.
> TEST_F(WebRtcVideoChannelBaseTest, GetStatsMultipleRecvStreams) {
>-  Base::GetStatsMultipleRecvStreams();
>-}
>-TEST_F(WebRtcVideoChannelBaseTest, GetStatsMultipleSendStreams) {
>-  Base::GetStatsMultipleSendStreams();
>+  cricket::FakeVideoRenderer renderer1, renderer2;
>+  EXPECT_TRUE(SetOneCodec(DefaultCodec()));
>+  cricket::VideoSendParameters parameters;
>+  parameters.codecs.push_back(DefaultCodec());
>+  parameters.conference_mode = true;
>+  EXPECT_TRUE(channel_->SetSendParameters(parameters));
>+  EXPECT_TRUE(SetSend(true));
>+  EXPECT_TRUE(channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(1)));
>+  EXPECT_TRUE(channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(2)));
>+  EXPECT_TRUE(channel_->SetSink(1, &renderer1));
>+  EXPECT_TRUE(channel_->SetSink(2, &renderer2));
>+  EXPECT_EQ(0, renderer1.num_rendered_frames());
>+  EXPECT_EQ(0, renderer2.num_rendered_frames());
>+  std::vector<uint32_t> ssrcs;
>+  ssrcs.push_back(1);
>+  ssrcs.push_back(2);
>+  network_interface_.SetConferenceMode(true, ssrcs);
>+  EXPECT_TRUE(SendFrame());
>+  EXPECT_FRAME_ON_RENDERER_WAIT(renderer1, 1, kVideoWidth, kVideoHeight,
>+                                kTimeout);
>+  EXPECT_FRAME_ON_RENDERER_WAIT(renderer2, 1, kVideoWidth, kVideoHeight,
>+                                kTimeout);
>+
>+  EXPECT_TRUE(channel_->SetSend(false));
>+
>+  cricket::VideoMediaInfo info;
>+  EXPECT_TRUE(channel_->GetStats(&info));
>+  ASSERT_EQ(1U, info.senders.size());
>+  // TODO(whyuan): bytes_sent and bytes_rcvd are different. Are both payload?
>+  // For webrtc, bytes_sent does not include the RTP header length.
>+  EXPECT_GT(GetSenderStats(0).bytes_sent, 0);
>+  EXPECT_EQ_WAIT(NumRtpPackets(), GetSenderStats(0).packets_sent, kTimeout);
>+  EXPECT_EQ(kVideoWidth, GetSenderStats(0).send_frame_width);
>+  EXPECT_EQ(kVideoHeight, GetSenderStats(0).send_frame_height);
>+
>+  ASSERT_EQ(2U, info.receivers.size());
>+  for (size_t i = 0; i < info.receivers.size(); ++i) {
>+    EXPECT_EQ(1U, GetReceiverStats(i).ssrcs().size());
>+    EXPECT_EQ(i + 1, GetReceiverStats(i).ssrcs()[0]);
>+    EXPECT_EQ_WAIT(NumRtpBytes(), GetReceiverStats(i).bytes_rcvd, kTimeout);
>+    EXPECT_EQ_WAIT(NumRtpPackets(), GetReceiverStats(i).packets_rcvd, kTimeout);
>+    EXPECT_EQ_WAIT(kVideoWidth, GetReceiverStats(i).frame_width, kTimeout);
>+    EXPECT_EQ_WAIT(kVideoHeight, GetReceiverStats(i).frame_height, kTimeout);
>+  }
> }
> 
>+// Test that stats work properly for a conf call with multiple send streams.
>+TEST_F(WebRtcVideoChannelBaseTest, GetStatsMultipleSendStreams) {
>+  // Normal setup; note that we set the SSRC explicitly to ensure that
>+  // it will come first in the senders map.
>+  EXPECT_TRUE(SetOneCodec(DefaultCodec()));
>+  cricket::VideoSendParameters parameters;
>+  parameters.codecs.push_back(DefaultCodec());
>+  parameters.conference_mode = true;
>+  EXPECT_TRUE(channel_->SetSendParameters(parameters));
>+  EXPECT_TRUE(
>+      channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(kSsrc)));
>+  EXPECT_TRUE(channel_->SetSink(kSsrc, &renderer_));
>+  EXPECT_TRUE(SetSend(true));
>+  EXPECT_TRUE(SendFrame());
>+  EXPECT_TRUE_WAIT(NumRtpPackets() > 0, kTimeout);
>+  EXPECT_FRAME_WAIT(1, kVideoWidth, kVideoHeight, kTimeout);
>+
>+  // Add an additional capturer, and hook up a renderer to receive it.
>+  cricket::FakeVideoRenderer renderer2;
>+  std::unique_ptr<cricket::FakeVideoCapturerWithTaskQueue> capturer(
>+      CreateFakeVideoCapturer());
>+  const int kTestWidth = 160;
>+  const int kTestHeight = 120;
>+  cricket::VideoFormat format(kTestWidth, kTestHeight,
>+                              cricket::VideoFormat::FpsToInterval(5),
>+                              cricket::FOURCC_I420);
>+  EXPECT_EQ(cricket::CS_RUNNING, capturer->Start(format));
>+  EXPECT_TRUE(
>+      channel_->AddSendStream(cricket::StreamParams::CreateLegacy(5678)));
>+  EXPECT_TRUE(channel_->SetVideoSend(5678, nullptr, capturer.get()));
>+  EXPECT_TRUE(
>+      channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(5678)));
>+  EXPECT_TRUE(channel_->SetSink(5678, &renderer2));
>+  EXPECT_TRUE(capturer->CaptureCustomFrame(kTestWidth, kTestHeight));
>+  EXPECT_FRAME_ON_RENDERER_WAIT(renderer2, 1, kTestWidth, kTestHeight,
>+                                kTimeout);
>+
>+  // Get stats, and make sure they are correct for two senders. We wait until
>+  // the number of expected packets have been sent to avoid races where we
>+  // check stats before it has been updated.
>+  cricket::VideoMediaInfo info;
>+  for (uint32_t i = 0; i < kTimeout; ++i) {
>+    rtc::Thread::Current()->ProcessMessages(1);
>+    EXPECT_TRUE(channel_->GetStats(&info));
>+    ASSERT_EQ(2U, info.senders.size());
>+    if (info.senders[0].packets_sent + info.senders[1].packets_sent ==
>+        NumRtpPackets()) {
>+      // Stats have been updated for both sent frames, expectations can be
>+      // checked now.
>+      break;
>+    }
>+  }
>+  EXPECT_EQ(NumRtpPackets(),
>+            info.senders[0].packets_sent + info.senders[1].packets_sent)
>+      << "Timed out while waiting for packet counts for all sent packets.";
>+  EXPECT_EQ(1U, info.senders[0].ssrcs().size());
>+  EXPECT_EQ(1234U, info.senders[0].ssrcs()[0]);
>+  EXPECT_EQ(kVideoWidth, info.senders[0].send_frame_width);
>+  EXPECT_EQ(kVideoHeight, info.senders[0].send_frame_height);
>+  EXPECT_EQ(1U, info.senders[1].ssrcs().size());
>+  EXPECT_EQ(5678U, info.senders[1].ssrcs()[0]);
>+  EXPECT_EQ(kTestWidth, info.senders[1].send_frame_width);
>+  EXPECT_EQ(kTestHeight, info.senders[1].send_frame_height);
>+  // The capturer must be unregistered here as it runs out of it's scope next.
>+  channel_->SetVideoSend(5678, nullptr, nullptr);
>+}
>+
>+// Test that we can set the bandwidth.
> TEST_F(WebRtcVideoChannelBaseTest, SetSendBandwidth) {
>-  Base::SetSendBandwidth();
>+  cricket::VideoSendParameters parameters;
>+  parameters.codecs.push_back(DefaultCodec());
>+  parameters.max_bandwidth_bps = -1;  // <= 0 means unlimited.
>+  EXPECT_TRUE(channel_->SetSendParameters(parameters));
>+  parameters.max_bandwidth_bps = 128 * 1024;
>+  EXPECT_TRUE(channel_->SetSendParameters(parameters));
> }
> 
>+// Test that we can set the SSRC for the default send source.
> TEST_F(WebRtcVideoChannelBaseTest, SetSendSsrc) {
>-  Base::SetSendSsrc();
>-}
>+  EXPECT_TRUE(SetDefaultCodec());
>+  EXPECT_TRUE(SetSend(true));
>+  EXPECT_TRUE(SendFrame());
>+  EXPECT_TRUE_WAIT(NumRtpPackets() > 0, kTimeout);
>+  webrtc::RTPHeader header;
>+  std::unique_ptr<const rtc::CopyOnWriteBuffer> p(GetRtpPacket(0));
>+  EXPECT_TRUE(ParseRtpPacket(p.get(), &header));
>+  EXPECT_EQ(kSsrc, header.ssrc);
>+
>+  // Packets are being paced out, so these can mismatch between the first and
>+  // second call to NumRtpPackets until pending packets are paced out.
>+  EXPECT_EQ_WAIT(NumRtpPackets(), NumRtpPackets(header.ssrc), kTimeout);
>+  EXPECT_EQ_WAIT(NumRtpBytes(), NumRtpBytes(header.ssrc), kTimeout);
>+  EXPECT_EQ(1, NumSentSsrcs());
>+  EXPECT_EQ(0, NumRtpPackets(kSsrc - 1));
>+  EXPECT_EQ(0, NumRtpBytes(kSsrc - 1));
>+}
>+
>+// Test that we can set the SSRC even after codecs are set.
> TEST_F(WebRtcVideoChannelBaseTest, SetSendSsrcAfterSetCodecs) {
>-  Base::SetSendSsrcAfterSetCodecs();
>-}
>-
>+  // Remove stream added in Setup.
>+  EXPECT_TRUE(channel_->RemoveSendStream(kSsrc));
>+  EXPECT_TRUE(SetDefaultCodec());
>+  EXPECT_TRUE(
>+      channel_->AddSendStream(cricket::StreamParams::CreateLegacy(999)));
>+  EXPECT_TRUE(channel_->SetVideoSend(999u, nullptr, video_capturer_.get()));
>+  EXPECT_TRUE(SetSend(true));
>+  EXPECT_TRUE(WaitAndSendFrame(0));
>+  EXPECT_TRUE_WAIT(NumRtpPackets() > 0, kTimeout);
>+  webrtc::RTPHeader header;
>+  std::unique_ptr<const rtc::CopyOnWriteBuffer> p(GetRtpPacket(0));
>+  EXPECT_TRUE(ParseRtpPacket(p.get(), &header));
>+  EXPECT_EQ(999u, header.ssrc);
>+  // Packets are being paced out, so these can mismatch between the first and
>+  // second call to NumRtpPackets until pending packets are paced out.
>+  EXPECT_EQ_WAIT(NumRtpPackets(), NumRtpPackets(header.ssrc), kTimeout);
>+  EXPECT_EQ_WAIT(NumRtpBytes(), NumRtpBytes(header.ssrc), kTimeout);
>+  EXPECT_EQ(1, NumSentSsrcs());
>+  EXPECT_EQ(0, NumRtpPackets(kSsrc));
>+  EXPECT_EQ(0, NumRtpBytes(kSsrc));
>+}
>+
>+// Test that we can set the default video renderer before and after
>+// media is received.
> TEST_F(WebRtcVideoChannelBaseTest, SetSink) {
>-  Base::SetSink();
>-}
>-
>+  uint8_t data1[] = {0x80, 0x00, 0x00, 0x00, 0x00, 0x00,
>+                     0x00, 0x00, 0x00, 0x00, 0x00, 0x00};
>+
>+  rtc::CopyOnWriteBuffer packet1(data1, sizeof(data1));
>+  rtc::SetBE32(packet1.data() + 8, kSsrc);
>+  channel_->SetSink(kDefaultReceiveSsrc, NULL);
>+  EXPECT_TRUE(SetDefaultCodec());
>+  EXPECT_TRUE(SetSend(true));
>+  EXPECT_EQ(0, renderer_.num_rendered_frames());
>+  channel_->OnPacketReceived(&packet1, rtc::PacketTime());
>+  EXPECT_TRUE(channel_->SetSink(kDefaultReceiveSsrc, &renderer_));
>+  EXPECT_TRUE(SendFrame());
>+  EXPECT_FRAME_WAIT(1, kVideoWidth, kVideoHeight, kTimeout);
>+}
>+
>+// Tests setting up and configuring a send stream.
> TEST_F(WebRtcVideoChannelBaseTest, AddRemoveSendStreams) {
>-  Base::AddRemoveSendStreams();
>-}
>+  EXPECT_TRUE(SetOneCodec(DefaultCodec()));
>+  EXPECT_TRUE(SetSend(true));
>+  EXPECT_TRUE(channel_->SetSink(kDefaultReceiveSsrc, &renderer_));
>+  EXPECT_TRUE(SendFrame());
>+  EXPECT_FRAME_WAIT(1, kVideoWidth, kVideoHeight, kTimeout);
>+  EXPECT_GT(NumRtpPackets(), 0);
>+  webrtc::RTPHeader header;
>+  size_t last_packet = NumRtpPackets() - 1;
>+  std::unique_ptr<const rtc::CopyOnWriteBuffer> p(
>+      GetRtpPacket(static_cast<int>(last_packet)));
>+  EXPECT_TRUE(ParseRtpPacket(p.get(), &header));
>+  EXPECT_EQ(kSsrc, header.ssrc);
>+
>+  // Remove the send stream that was added during Setup.
>+  EXPECT_TRUE(channel_->RemoveSendStream(kSsrc));
>+  int rtp_packets = NumRtpPackets();
> 
>-TEST_F(WebRtcVideoChannelBaseTest, SimulateConference) {
>-  Base::SimulateConference();
>+  EXPECT_TRUE(
>+      channel_->AddSendStream(cricket::StreamParams::CreateLegacy(789u)));
>+  EXPECT_TRUE(channel_->SetVideoSend(789u, nullptr, video_capturer_.get()));
>+  EXPECT_EQ(rtp_packets, NumRtpPackets());
>+  // Wait 30ms to guarantee the engine does not drop the frame.
>+  EXPECT_TRUE(WaitAndSendFrame(30));
>+  EXPECT_TRUE_WAIT(NumRtpPackets() > rtp_packets, kTimeout);
>+
>+  last_packet = NumRtpPackets() - 1;
>+  p.reset(GetRtpPacket(static_cast<int>(last_packet)));
>+  EXPECT_TRUE(ParseRtpPacket(p.get(), &header));
>+  EXPECT_EQ(789u, header.ssrc);
> }
> 
>+// Tests the behavior of incoming streams in a conference scenario.
>+TEST_F(WebRtcVideoChannelBaseTest, SimulateConference) {
>+  cricket::FakeVideoRenderer renderer1, renderer2;
>+  EXPECT_TRUE(SetDefaultCodec());
>+  cricket::VideoSendParameters parameters;
>+  parameters.codecs.push_back(DefaultCodec());
>+  parameters.conference_mode = true;
>+  EXPECT_TRUE(channel_->SetSendParameters(parameters));
>+  EXPECT_TRUE(SetSend(true));
>+  EXPECT_TRUE(channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(1)));
>+  EXPECT_TRUE(channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(2)));
>+  EXPECT_TRUE(channel_->SetSink(1, &renderer1));
>+  EXPECT_TRUE(channel_->SetSink(2, &renderer2));
>+  EXPECT_EQ(0, renderer1.num_rendered_frames());
>+  EXPECT_EQ(0, renderer2.num_rendered_frames());
>+  std::vector<uint32_t> ssrcs;
>+  ssrcs.push_back(1);
>+  ssrcs.push_back(2);
>+  network_interface_.SetConferenceMode(true, ssrcs);
>+  EXPECT_TRUE(SendFrame());
>+  EXPECT_FRAME_ON_RENDERER_WAIT(renderer1, 1, kVideoWidth, kVideoHeight,
>+                                kTimeout);
>+  EXPECT_FRAME_ON_RENDERER_WAIT(renderer2, 1, kVideoWidth, kVideoHeight,
>+                                kTimeout);
>+
>+  std::unique_ptr<const rtc::CopyOnWriteBuffer> p(GetRtpPacket(0));
>+  EXPECT_EQ(DefaultCodec().id, GetPayloadType(p.get()));
>+  EXPECT_EQ(kVideoWidth, renderer1.width());
>+  EXPECT_EQ(kVideoHeight, renderer1.height());
>+  EXPECT_EQ(kVideoWidth, renderer2.width());
>+  EXPECT_EQ(kVideoHeight, renderer2.height());
>+  EXPECT_TRUE(channel_->RemoveRecvStream(2));
>+  EXPECT_TRUE(channel_->RemoveRecvStream(1));
>+}
>+
>+// Tests that we can add and remove capturers and frames are sent out properly
> TEST_F(WebRtcVideoChannelBaseTest, DISABLED_AddRemoveCapturer) {
>-  Base::AddRemoveCapturer();
>+  using cricket::VideoCodec;
>+  using cricket::VideoOptions;
>+  using cricket::VideoFormat;
>+  using cricket::FOURCC_I420;
>+
>+  VideoCodec codec = DefaultCodec();
>+  const int time_between_send_ms = VideoFormat::FpsToInterval(kFramerate);
>+  EXPECT_TRUE(SetOneCodec(codec));
>+  EXPECT_TRUE(SetSend(true));
>+  EXPECT_TRUE(channel_->SetSink(kDefaultReceiveSsrc, &renderer_));
>+  EXPECT_EQ(0, renderer_.num_rendered_frames());
>+  EXPECT_TRUE(SendFrame());
>+  EXPECT_FRAME_WAIT(1, kVideoWidth, kVideoHeight, kTimeout);
>+  std::unique_ptr<cricket::FakeVideoCapturerWithTaskQueue> capturer(
>+      CreateFakeVideoCapturer());
>+
>+  // TODO(nisse): This testcase fails if we don't configure
>+  // screencast. It's unclear why, I see nothing obvious in this
>+  // test which is related to screencast logic.
>+  VideoOptions video_options;
>+  video_options.is_screencast = true;
>+  channel_->SetVideoSend(kSsrc, &video_options, nullptr);
>+
>+  VideoFormat format(480, 360, VideoFormat::FpsToInterval(30), FOURCC_I420);
>+  EXPECT_EQ(cricket::CS_RUNNING, capturer->Start(format));
>+  // All capturers start generating frames with the same timestamp. ViE does
>+  // not allow the same timestamp to be used. Capture one frame before
>+  // associating the capturer with the channel.
>+  EXPECT_TRUE(capturer->CaptureCustomFrame(format.width, format.height));
>+
>+  int captured_frames = 1;
>+  for (int iterations = 0; iterations < 2; ++iterations) {
>+    EXPECT_TRUE(channel_->SetVideoSend(kSsrc, nullptr, capturer.get()));
>+    rtc::Thread::Current()->ProcessMessages(time_between_send_ms);
>+    EXPECT_TRUE(capturer->CaptureCustomFrame(format.width, format.height));
>+    ++captured_frames;
>+    // Wait until frame of right size is captured.
>+    EXPECT_TRUE_WAIT(renderer_.num_rendered_frames() >= captured_frames &&
>+                         format.width == renderer_.width() &&
>+                         format.height == renderer_.height() &&
>+                         !renderer_.black_frame(),
>+                     kTimeout);
>+    EXPECT_GE(renderer_.num_rendered_frames(), captured_frames);
>+    EXPECT_EQ(format.width, renderer_.width());
>+    EXPECT_EQ(format.height, renderer_.height());
>+    captured_frames = renderer_.num_rendered_frames() + 1;
>+    EXPECT_FALSE(renderer_.black_frame());
>+    EXPECT_TRUE(channel_->SetVideoSend(kSsrc, nullptr, nullptr));
>+    // Make sure a black frame is generated within the specified timeout.
>+    // The black frame should be the resolution of the previous frame to
>+    // prevent expensive encoder reconfigurations.
>+    EXPECT_TRUE_WAIT(renderer_.num_rendered_frames() >= captured_frames &&
>+                         format.width == renderer_.width() &&
>+                         format.height == renderer_.height() &&
>+                         renderer_.black_frame(),
>+                     kTimeout);
>+    EXPECT_GE(renderer_.num_rendered_frames(), captured_frames);
>+    EXPECT_EQ(format.width, renderer_.width());
>+    EXPECT_EQ(format.height, renderer_.height());
>+    EXPECT_TRUE(renderer_.black_frame());
>+
>+    // The black frame has the same timestamp as the next frame since it's
>+    // timestamp is set to the last frame's timestamp + interval. WebRTC will
>+    // not render a frame with the same timestamp so capture another frame
>+    // with the frame capturer to increment the next frame's timestamp.
>+    EXPECT_TRUE(capturer->CaptureCustomFrame(format.width, format.height));
>+  }
> }
> 
>+// Tests that if SetVideoSend is called with a NULL capturer after the
>+// capturer was already removed, the application doesn't crash (and no black
>+// frame is sent).
> TEST_F(WebRtcVideoChannelBaseTest, RemoveCapturerWithoutAdd) {
>-  Base::RemoveCapturerWithoutAdd();
>-}
>-
>+  EXPECT_TRUE(SetOneCodec(DefaultCodec()));
>+  EXPECT_TRUE(SetSend(true));
>+  EXPECT_TRUE(channel_->SetSink(kDefaultReceiveSsrc, &renderer_));
>+  EXPECT_EQ(0, renderer_.num_rendered_frames());
>+  EXPECT_TRUE(SendFrame());
>+  EXPECT_FRAME_WAIT(1, kVideoWidth, kVideoHeight, kTimeout);
>+  // Wait for one frame so they don't get dropped because we send frames too
>+  // tightly.
>+  rtc::Thread::Current()->ProcessMessages(30);
>+  // Remove the capturer.
>+  EXPECT_TRUE(channel_->SetVideoSend(kSsrc, nullptr, nullptr));
>+
>+  // No capturer was added, so this SetVideoSend shouldn't do anything.
>+  EXPECT_TRUE(channel_->SetVideoSend(kSsrc, nullptr, nullptr));
>+  rtc::Thread::Current()->ProcessMessages(300);
>+  // Verify no more frames were sent.
>+  EXPECT_EQ(1, renderer_.num_rendered_frames());
>+}
>+
>+// Tests that we can add and remove capturer as unique sources.
> TEST_F(WebRtcVideoChannelBaseTest, AddRemoveCapturerMultipleSources) {
>-  Base::AddRemoveCapturerMultipleSources();
>-}
>+  // WebRTC implementation will drop frames if pushed to quickly. Wait the
>+  // interval time to avoid that.
>+  // WebRTC implementation will drop frames if pushed to quickly. Wait the
>+  // interval time to avoid that.
>+  // Set up the stream associated with the engine.
>+  EXPECT_TRUE(
>+      channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(kSsrc)));
>+  EXPECT_TRUE(channel_->SetSink(kSsrc, &renderer_));
>+  cricket::VideoFormat capture_format(
>+      kVideoWidth, kVideoHeight,
>+      cricket::VideoFormat::FpsToInterval(kFramerate), cricket::FOURCC_I420);
>+  // Set up additional stream 1.
>+  cricket::FakeVideoRenderer renderer1;
>+  EXPECT_FALSE(channel_->SetSink(1, &renderer1));
>+  EXPECT_TRUE(channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(1)));
>+  EXPECT_TRUE(channel_->SetSink(1, &renderer1));
>+  EXPECT_TRUE(channel_->AddSendStream(cricket::StreamParams::CreateLegacy(1)));
>+  std::unique_ptr<cricket::FakeVideoCapturerWithTaskQueue> capturer1(
>+      CreateFakeVideoCapturer());
>+  EXPECT_EQ(cricket::CS_RUNNING, capturer1->Start(capture_format));
>+  // Set up additional stream 2.
>+  cricket::FakeVideoRenderer renderer2;
>+  EXPECT_FALSE(channel_->SetSink(2, &renderer2));
>+  EXPECT_TRUE(channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(2)));
>+  EXPECT_TRUE(channel_->SetSink(2, &renderer2));
>+  EXPECT_TRUE(channel_->AddSendStream(cricket::StreamParams::CreateLegacy(2)));
>+  std::unique_ptr<cricket::FakeVideoCapturerWithTaskQueue> capturer2(
>+      CreateFakeVideoCapturer());
>+  EXPECT_EQ(cricket::CS_RUNNING, capturer2->Start(capture_format));
>+  // State for all the streams.
>+  EXPECT_TRUE(SetOneCodec(DefaultCodec()));
>+  // A limitation in the lmi implementation requires that SetVideoSend() is
>+  // called after SetOneCodec().
>+  // TODO(hellner): this seems like an unnecessary constraint, fix it.
>+  EXPECT_TRUE(channel_->SetVideoSend(1, nullptr, capturer1.get()));
>+  EXPECT_TRUE(channel_->SetVideoSend(2, nullptr, capturer2.get()));
>+  EXPECT_TRUE(SetSend(true));
>+  // Test capturer associated with engine.
>+  const int kTestWidth = 160;
>+  const int kTestHeight = 120;
>+  EXPECT_TRUE(capturer1->CaptureCustomFrame(kTestWidth, kTestHeight));
>+  EXPECT_FRAME_ON_RENDERER_WAIT(renderer1, 1, kTestWidth, kTestHeight,
>+                                kTimeout);
>+  // Capture a frame with additional capturer2, frames should be received
>+  EXPECT_TRUE(capturer2->CaptureCustomFrame(kTestWidth, kTestHeight));
>+  EXPECT_FRAME_ON_RENDERER_WAIT(renderer2, 1, kTestWidth, kTestHeight,
>+                                kTimeout);
>+  // Successfully remove the capturer.
>+  EXPECT_TRUE(channel_->SetVideoSend(kSsrc, nullptr, nullptr));
>+  // The capturers must be unregistered here as it runs out of it's scope
>+  // next.
>+  EXPECT_TRUE(channel_->SetVideoSend(1, nullptr, nullptr));
>+  EXPECT_TRUE(channel_->SetVideoSend(2, nullptr, nullptr));
>+}
>+
>+// Tests empty StreamParams is rejected.
> TEST_F(WebRtcVideoChannelBaseTest, RejectEmptyStreamParams) {
>-  Base::RejectEmptyStreamParams();
>+  // Remove the send stream that was added during Setup.
>+  EXPECT_TRUE(channel_->RemoveSendStream(kSsrc));
>+
>+  cricket::StreamParams empty;
>+  EXPECT_FALSE(channel_->AddSendStream(empty));
>+  EXPECT_TRUE(
>+      channel_->AddSendStream(cricket::StreamParams::CreateLegacy(789u)));
> }
>+
>+// Test that multiple send streams can be created and deleted properly.
> TEST_F(WebRtcVideoChannelBaseTest, MultipleSendStreams) {
>-  Base::MultipleSendStreams();
>+  // Remove stream added in Setup. I.e. remove stream corresponding to default
>+  // channel.
>+  EXPECT_TRUE(channel_->RemoveSendStream(kSsrc));
>+  const unsigned int kSsrcsSize = sizeof(kSsrcs4) / sizeof(kSsrcs4[0]);
>+  for (unsigned int i = 0; i < kSsrcsSize; ++i) {
>+    EXPECT_TRUE(channel_->AddSendStream(
>+        cricket::StreamParams::CreateLegacy(kSsrcs4[i])));
>+  }
>+  // Delete one of the non default channel streams, let the destructor delete
>+  // the remaining ones.
>+  EXPECT_TRUE(channel_->RemoveSendStream(kSsrcs4[kSsrcsSize - 1]));
>+  // Stream should already be deleted.
>+  EXPECT_FALSE(channel_->RemoveSendStream(kSsrcs4[kSsrcsSize - 1]));
> }
> 
> TEST_F(WebRtcVideoChannelBaseTest, SendAndReceiveVp8Vga) {
>@@ -1221,7 +1945,7 @@ TEST_F(WebRtcVideoChannelBaseTest, TwoStreamsSendAndReceive) {
>   // at QVGA, then verify that instead.
>   cricket::VideoCodec codec = GetEngineCodec("VP8");
>   codec.params[kCodecParamStartBitrate] = "1000000";
>-  Base::TwoStreamsSendAndReceive(codec);
>+  TwoStreamsSendAndReceive(codec);
> }
> 
> class WebRtcVideoChannelTest : public WebRtcVideoEngineTest {
>@@ -1230,7 +1954,13 @@ class WebRtcVideoChannelTest : public WebRtcVideoEngineTest {
>   explicit WebRtcVideoChannelTest(const char* field_trials)
>       : WebRtcVideoEngineTest(field_trials), last_ssrc_(0) {}
>   void SetUp() override {
>-    fake_call_.reset(new FakeCall(webrtc::Call::Config(&event_log_)));
>+    encoder_factory_->AddSupportedVideoCodecType("VP8");
>+    encoder_factory_->AddSupportedVideoCodecType("VP9");
>+#if defined(WEBRTC_USE_H264)
>+    encoder_factory_->AddSupportedVideoCodecType("H264");
>+#endif
>+
>+    fake_call_.reset(new FakeCall());
>     channel_.reset(engine_.CreateChannel(fake_call_.get(), GetMediaConfig(),
>                                          VideoOptions()));
>     channel_->OnReadyToSend(true);
>@@ -1277,6 +2007,9 @@ class WebRtcVideoChannelTest : public WebRtcVideoEngineTest {
>                                           int expected_start_bitrate_bps,
>                                           const char* max_bitrate_kbps,
>                                           int expected_max_bitrate_bps) {
>+    ExpectSetBitrateParameters(expected_min_bitrate_bps,
>+                               expected_start_bitrate_bps,
>+                               expected_max_bitrate_bps);
>     auto& codecs = send_parameters_.codecs;
>     codecs.clear();
>     codecs.push_back(GetEngineCodec("VP8"));
>@@ -1284,13 +2017,23 @@ class WebRtcVideoChannelTest : public WebRtcVideoEngineTest {
>     codecs[0].params[kCodecParamStartBitrate] = start_bitrate_kbps;
>     codecs[0].params[kCodecParamMaxBitrate] = max_bitrate_kbps;
>     EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
>+  }
> 
>-    EXPECT_EQ(expected_min_bitrate_bps,
>-              fake_call_->GetConfig().bitrate_config.min_bitrate_bps);
>-    EXPECT_EQ(expected_start_bitrate_bps,
>-              fake_call_->GetConfig().bitrate_config.start_bitrate_bps);
>-    EXPECT_EQ(expected_max_bitrate_bps,
>-              fake_call_->GetConfig().bitrate_config.max_bitrate_bps);
>+  void ExpectSetBitrateParameters(int min_bitrate_bps,
>+                                  int start_bitrate_bps,
>+                                  int max_bitrate_bps) {
>+    EXPECT_CALL(
>+        *fake_call_->GetMockTransportControllerSend(),
>+        SetSdpBitrateParameters(AllOf(
>+            Field(&BitrateConstraints::min_bitrate_bps, min_bitrate_bps),
>+            Field(&BitrateConstraints::start_bitrate_bps, start_bitrate_bps),
>+            Field(&BitrateConstraints::max_bitrate_bps, max_bitrate_bps))));
>+  }
>+
>+  void ExpectSetMaxBitrate(int max_bitrate_bps) {
>+    EXPECT_CALL(*fake_call_->GetMockTransportControllerSend(),
>+                SetSdpBitrateParameters(Field(
>+                    &BitrateConstraints::max_bitrate_bps, max_bitrate_bps)));
>   }
> 
>   void TestSetSendRtpHeaderExtensions(const std::string& ext_uri) {
>@@ -1399,11 +2142,11 @@ class WebRtcVideoChannelTest : public WebRtcVideoEngineTest {
> 
>   FakeVideoSendStream* SetDenoisingOption(
>       uint32_t ssrc,
>-      cricket::FakeVideoCapturer* capturer,
>+      FakeVideoCapturerWithTaskQueue* capturer,
>       bool enabled) {
>     cricket::VideoOptions options;
>     options.video_noise_reduction = enabled;
>-    EXPECT_TRUE(channel_->SetVideoSend(ssrc, true, &options, capturer));
>+    EXPECT_TRUE(channel_->SetVideoSend(ssrc, &options, capturer));
>     // Options only take effect on the next frame.
>     EXPECT_TRUE(capturer->CaptureFrame());
> 
>@@ -1435,7 +2178,7 @@ class WebRtcVideoChannelTest : public WebRtcVideoEngineTest {
>         fake_call_->GetVideoSendStreams();
>     EXPECT_EQ(1u, streams.size());
>     FakeVideoSendStream* stream = streams[streams.size() - 1];
>-    EXPECT_EQ(1, stream->GetEncoderConfig().number_of_streams);
>+    EXPECT_EQ(1u, stream->GetEncoderConfig().number_of_streams);
>     return stream->GetVideoStreams()[0].max_bitrate_bps;
>   }
> 
>@@ -1449,7 +2192,7 @@ class WebRtcVideoChannelTest : public WebRtcVideoEngineTest {
>         channel_->GetRtpSendParameters(last_ssrc_);
>     EXPECT_EQ(1UL, parameters.encodings.size());
>     parameters.encodings[0].max_bitrate_bps = stream_max;
>-    EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters));
>+    EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
>     // Read back the parameteres and verify they have the correct value
>     parameters = channel_->GetRtpSendParameters(last_ssrc_);
>     EXPECT_EQ(1UL, parameters.encodings.size());
>@@ -1458,6 +2201,21 @@ class WebRtcVideoChannelTest : public WebRtcVideoEngineTest {
>     EXPECT_EQ(expected_encoder_bitrate, GetMaxEncoderBitrate());
>   }
> 
>+  // Values from kSimulcastConfigs in simulcast.cc.
>+  const std::vector<webrtc::VideoStream> GetSimulcastBitrates720p() const {
>+    std::vector<webrtc::VideoStream> layers(3);
>+    layers[0].min_bitrate_bps = 30000;
>+    layers[0].target_bitrate_bps = 150000;
>+    layers[0].max_bitrate_bps = 200000;
>+    layers[1].min_bitrate_bps = 150000;
>+    layers[1].target_bitrate_bps = 500000;
>+    layers[1].max_bitrate_bps = 700000;
>+    layers[2].min_bitrate_bps = 600000;
>+    layers[2].target_bitrate_bps = 2500000;
>+    layers[2].max_bitrate_bps = 2500000;
>+    return layers;
>+  }
>+
>   std::unique_ptr<FakeCall> fake_call_;
>   std::unique_ptr<VideoMediaChannel> channel_;
>   cricket::VideoSendParameters send_parameters_;
>@@ -1470,10 +2228,10 @@ TEST_F(WebRtcVideoChannelTest, SetsSyncGroupFromSyncLabel) {
>   const std::string kSyncLabel = "AvSyncLabel";
> 
>   cricket::StreamParams sp = cricket::StreamParams::CreateLegacy(kVideoSsrc);
>-  sp.sync_label = kSyncLabel;
>+  sp.set_stream_ids({kSyncLabel});
>   EXPECT_TRUE(channel_->AddRecvStream(sp));
> 
>-  EXPECT_EQ(1, fake_call_->GetVideoReceiveStreams().size());
>+  EXPECT_EQ(1u, fake_call_->GetVideoReceiveStreams().size());
>   EXPECT_EQ(kSyncLabel,
>             fake_call_->GetVideoReceiveStreams()[0]->GetConfig().sync_group)
>       << "SyncGroup should be set based on sync_label";
>@@ -1505,7 +2263,7 @@ TEST_F(WebRtcVideoChannelTest, RecvStreamWithSimAndRtx) {
>   EXPECT_TRUE(VerifyRtxReceiveAssociations(recv_stream->GetConfig()))
>       << "RTX should be mapped for all decoders/payload types.";
>   EXPECT_TRUE(HasRtxReceiveAssociation(recv_stream->GetConfig(),
>-                                          GetEngineCodec("red").id))
>+                                       GetEngineCodec("red").id))
>       << "RTX should be mapped for the RED payload type";
> 
>   EXPECT_EQ(rtx_ssrcs[0], recv_stream->GetConfig().rtp.rtx_ssrc);
>@@ -1522,7 +2280,7 @@ TEST_F(WebRtcVideoChannelTest, RecvStreamWithRtx) {
>   EXPECT_TRUE(VerifyRtxReceiveAssociations(recv_stream->GetConfig()))
>       << "RTX should be mapped for all decoders/payload types.";
>   EXPECT_TRUE(HasRtxReceiveAssociation(recv_stream->GetConfig(),
>-                                          GetEngineCodec("red").id))
>+                                       GetEngineCodec("red").id))
>       << "RTX should be mapped for the RED payload type";
> }
> 
>@@ -1812,6 +2570,7 @@ TEST_F(WebRtcVideoChannelTest, TransportCcCanBeEnabledAndDisabled) {
> }
> 
> TEST_F(WebRtcVideoChannelTest, NackIsEnabledByDefault) {
>+  AssignDefaultCodec();
>   VerifyCodecHasDefaultFeedbackParams(default_codec_);
> 
>   cricket::VideoSendParameters parameters;
>@@ -1879,18 +2638,19 @@ TEST_F(WebRtcVideoChannelTest, ReconfiguresEncodersWhenNotSending) {
>   EXPECT_EQ(0u, streams[0].width);
>   EXPECT_EQ(0u, streams[0].height);
> 
>-  cricket::FakeVideoCapturer capturer;
>-  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, true, nullptr, &capturer));
>+  FakeVideoCapturerWithTaskQueue capturer;
>+  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, &capturer));
>   VideoFormat capture_format = capturer.GetSupportedFormats()->front();
>   EXPECT_EQ(cricket::CS_RUNNING, capturer.Start(capture_format));
>   EXPECT_TRUE(capturer.CaptureFrame());
> 
>   // Frame entered, should be reconfigured to new dimensions.
>   streams = stream->GetVideoStreams();
>-  EXPECT_EQ(capture_format.width, streams[0].width);
>-  EXPECT_EQ(capture_format.height, streams[0].height);
>+  EXPECT_EQ(rtc::checked_cast<size_t>(capture_format.width), streams[0].width);
>+  EXPECT_EQ(rtc::checked_cast<size_t>(capture_format.height),
>+            streams[0].height);
> 
>-  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, true, nullptr, nullptr));
>+  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
> }
> 
> TEST_F(WebRtcVideoChannelTest, UsesCorrectSettingsForScreencast) {
>@@ -1901,11 +2661,11 @@ TEST_F(WebRtcVideoChannelTest, UsesCorrectSettingsForScreencast) {
>   EXPECT_TRUE(channel_->SetSendParameters(parameters));
>   AddSendStream();
> 
>-  cricket::FakeVideoCapturer capturer;
>+  FakeVideoCapturerWithTaskQueue capturer;
>   VideoOptions min_bitrate_options;
>   min_bitrate_options.screencast_min_bitrate_kbps = kScreenshareMinBitrateKbps;
>-  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, true, &min_bitrate_options,
>-                                     &capturer));
>+  EXPECT_TRUE(
>+      channel_->SetVideoSend(last_ssrc_, &min_bitrate_options, &capturer));
>   cricket::VideoFormat capture_format_hd =
>       capturer.GetSupportedFormats()->front();
>   EXPECT_EQ(1280, capture_format_hd.width);
>@@ -1926,17 +2686,19 @@ TEST_F(WebRtcVideoChannelTest, UsesCorrectSettingsForScreencast) {
>   EXPECT_EQ(webrtc::VideoEncoderConfig::ContentType::kRealtimeVideo,
>             encoder_config.content_type);
>   std::vector<webrtc::VideoStream> streams = send_stream->GetVideoStreams();
>-  EXPECT_EQ(capture_format_hd.width, streams.front().width);
>-  EXPECT_EQ(capture_format_hd.height, streams.front().height);
>+  EXPECT_EQ(rtc::checked_cast<size_t>(capture_format_hd.width),
>+            streams.front().width);
>+  EXPECT_EQ(rtc::checked_cast<size_t>(capture_format_hd.height),
>+            streams.front().height);
>   EXPECT_EQ(0, encoder_config.min_transmit_bitrate_bps)
>       << "Non-screenshare shouldn't use min-transmit bitrate.";
> 
>-  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, true, nullptr, nullptr));
>+  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
>   EXPECT_EQ(1, send_stream->GetNumberOfSwappedFrames());
>   VideoOptions screencast_options;
>   screencast_options.is_screencast = true;
>   EXPECT_TRUE(
>-      channel_->SetVideoSend(last_ssrc_, true, &screencast_options, &capturer));
>+      channel_->SetVideoSend(last_ssrc_, &screencast_options, &capturer));
>   EXPECT_TRUE(capturer.CaptureFrame());
>   // Send stream recreated after option change.
>   ASSERT_EQ(2, fake_call_->GetNumCreatedSendStreams());
>@@ -1951,24 +2713,25 @@ TEST_F(WebRtcVideoChannelTest, UsesCorrectSettingsForScreencast) {
>             encoder_config.min_transmit_bitrate_bps);
> 
>   streams = send_stream->GetVideoStreams();
>-  EXPECT_EQ(capture_format_hd.width, streams.front().width);
>-  EXPECT_EQ(capture_format_hd.height, streams.front().height);
>-  EXPECT_TRUE(streams[0].temporal_layer_thresholds_bps.empty());
>-  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, true, nullptr, nullptr));
>+  EXPECT_EQ(rtc::checked_cast<size_t>(capture_format_hd.width),
>+            streams.front().width);
>+  EXPECT_EQ(rtc::checked_cast<size_t>(capture_format_hd.height),
>+            streams.front().height);
>+  EXPECT_FALSE(streams[0].num_temporal_layers.has_value());
>+  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
> }
> 
> TEST_F(WebRtcVideoChannelTest,
>        ConferenceModeScreencastConfiguresTemporalLayer) {
>-  static const int kConferenceScreencastTemporalBitrateBps =
>-      ScreenshareLayerConfig::GetDefault().tl0_bitrate_kbps * 1000;
>+  static const int kConferenceScreencastTemporalBitrateBps = 200 * 1000;
>   send_parameters_.conference_mode = true;
>   channel_->SetSendParameters(send_parameters_);
> 
>   AddSendStream();
>   VideoOptions options;
>   options.is_screencast = true;
>-  cricket::FakeVideoCapturer capturer;
>-  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, true, &options, &capturer));
>+  FakeVideoCapturerWithTaskQueue capturer;
>+  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &capturer));
>   cricket::VideoFormat capture_format_hd =
>       capturer.GetSupportedFormats()->front();
>   EXPECT_EQ(cricket::CS_RUNNING, capturer.Start(capture_format_hd));
>@@ -1989,11 +2752,11 @@ TEST_F(WebRtcVideoChannelTest,
> 
>   std::vector<webrtc::VideoStream> streams = send_stream->GetVideoStreams();
>   ASSERT_EQ(1u, streams.size());
>-  ASSERT_EQ(1u, streams[0].temporal_layer_thresholds_bps.size());
>+  ASSERT_EQ(2u, streams[0].num_temporal_layers);
>   EXPECT_EQ(kConferenceScreencastTemporalBitrateBps,
>-            streams[0].temporal_layer_thresholds_bps[0]);
>+            streams[0].target_bitrate_bps);
> 
>-  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, true, nullptr, nullptr));
>+  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
> }
> 
> TEST_F(WebRtcVideoChannelTest, SuspendBelowMinBitrateDisabledByDefault) {
>@@ -2042,10 +2805,10 @@ TEST_F(WebRtcVideoChannelTest, VerifyVp8SpecificSettings) {
>   // both RTX and regular SSRCs).
>   FakeVideoSendStream* stream = SetUpSimulcast(false, true);
> 
>-  cricket::FakeVideoCapturer capturer;
>+  FakeVideoCapturerWithTaskQueue capturer;
>   EXPECT_EQ(cricket::CS_RUNNING,
>             capturer.Start(capturer.GetSupportedFormats()->front()));
>-  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, true, nullptr, &capturer));
>+  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, &capturer));
>   channel_->SetSend(true);
> 
>   EXPECT_TRUE(capturer.CaptureFrame());
>@@ -2069,13 +2832,13 @@ TEST_F(WebRtcVideoChannelTest, VerifyVp8SpecificSettings) {
>   EXPECT_TRUE(vp8_settings.automaticResizeOn);
>   EXPECT_TRUE(vp8_settings.frameDroppingOn);
> 
>-  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, true, nullptr, nullptr));
>+  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
>   stream = SetUpSimulcast(true, false);
>-  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, true, nullptr, &capturer));
>+  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, &capturer));
>   channel_->SetSend(true);
>   EXPECT_TRUE(capturer.CaptureFrame());
> 
>-  EXPECT_EQ(3, stream->GetVideoStreams().size());
>+  EXPECT_EQ(3u, stream->GetVideoStreams().size());
>   ASSERT_TRUE(stream->GetVp8Settings(&vp8_settings)) << "No VP8 config set.";
>   // Autmatic resize off when using simulcast.
>   EXPECT_FALSE(vp8_settings.automaticResizeOn);
>@@ -2084,11 +2847,11 @@ TEST_F(WebRtcVideoChannelTest, VerifyVp8SpecificSettings) {
>   // In screen-share mode, denoising is forced off and simulcast disabled.
>   VideoOptions options;
>   options.is_screencast = true;
>-  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, true, &options, &capturer));
>+  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &capturer));
> 
>   stream = SetDenoisingOption(last_ssrc_, &capturer, false);
> 
>-  EXPECT_EQ(1, stream->GetVideoStreams().size());
>+  EXPECT_EQ(1u, stream->GetVideoStreams().size());
>   ASSERT_TRUE(stream->GetVp8Settings(&vp8_settings)) << "No VP8 config set.";
>   EXPECT_FALSE(vp8_settings.denoisingOn);
>   // Resizing and frame dropping always off for screen sharing.
>@@ -2102,14 +2865,14 @@ TEST_F(WebRtcVideoChannelTest, VerifyVp8SpecificSettings) {
>   EXPECT_FALSE(vp8_settings.automaticResizeOn);
>   EXPECT_FALSE(vp8_settings.frameDroppingOn);
> 
>-  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, true, nullptr, nullptr));
>+  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
> }
> 
> // Test that setting the same options doesn't result in the encoder being
> // reconfigured.
> TEST_F(WebRtcVideoChannelTest, SetIdenticalOptionsDoesntReconfigureEncoder) {
>   VideoOptions options;
>-  cricket::FakeVideoCapturer capturer;
>+  FakeVideoCapturerWithTaskQueue capturer;
> 
>   AddSendStream();
>   EXPECT_EQ(cricket::CS_RUNNING,
>@@ -2119,22 +2882,22 @@ TEST_F(WebRtcVideoChannelTest, SetIdenticalOptionsDoesntReconfigureEncoder) {
>   ASSERT_TRUE(channel_->SetSendParameters(parameters));
>   FakeVideoSendStream* send_stream = fake_call_->GetVideoSendStreams().front();
> 
>-  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, true, &options, &capturer));
>-  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, true, &options, &capturer));
>+  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &capturer));
>+  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &capturer));
>   EXPECT_TRUE(capturer.CaptureFrame());
>   // Expect 1 reconfigurations at this point from the initial configuration.
>   EXPECT_EQ(1, send_stream->num_encoder_reconfigurations());
> 
>   // Set the options one more time and expect no additional reconfigurations.
>-  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, true, &options, &capturer));
>+  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &capturer));
>   EXPECT_EQ(1, send_stream->num_encoder_reconfigurations());
> 
>   // Change |options| and expect 2 reconfigurations.
>   options.video_noise_reduction = true;
>-  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, true, &options, &capturer));
>+  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &capturer));
>   EXPECT_EQ(2, send_stream->num_encoder_reconfigurations());
> 
>-  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, true, nullptr, nullptr));
>+  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
> }
> 
> class Vp9SettingsTest : public WebRtcVideoChannelTest {
>@@ -2161,10 +2924,10 @@ TEST_F(Vp9SettingsTest, VerifyVp9SpecificSettings) {
> 
>   FakeVideoSendStream* stream = SetUpSimulcast(false, false);
> 
>-  cricket::FakeVideoCapturer capturer;
>+  FakeVideoCapturerWithTaskQueue capturer;
>   EXPECT_EQ(cricket::CS_RUNNING,
>             capturer.Start(capturer.GetSupportedFormats()->front()));
>-  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, true, nullptr, &capturer));
>+  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, &capturer));
>   channel_->SetSend(true);
> 
>   EXPECT_TRUE(capturer.CaptureFrame());
>@@ -2190,22 +2953,54 @@ TEST_F(Vp9SettingsTest, VerifyVp9SpecificSettings) {
>   // In screen-share mode, denoising is forced off.
>   VideoOptions options;
>   options.is_screencast = true;
>-  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, true, &options, &capturer));
>+  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &capturer));
> 
>   stream = SetDenoisingOption(last_ssrc_, &capturer, false);
> 
>   ASSERT_TRUE(stream->GetVp9Settings(&vp9_settings)) << "No VP9 config set.";
>   EXPECT_FALSE(vp9_settings.denoisingOn);
>-  // Frame dropping always off for screen sharing.
>-  EXPECT_FALSE(vp9_settings.frameDroppingOn);
>+  // Frame dropping always on for screen sharing.
>+  EXPECT_TRUE(vp9_settings.frameDroppingOn);
> 
>   stream = SetDenoisingOption(last_ssrc_, &capturer, false);
> 
>   ASSERT_TRUE(stream->GetVp9Settings(&vp9_settings)) << "No VP9 config set.";
>   EXPECT_FALSE(vp9_settings.denoisingOn);
>-  EXPECT_FALSE(vp9_settings.frameDroppingOn);
>+  EXPECT_TRUE(vp9_settings.frameDroppingOn);
>+
>+  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
>+}
>+
>+TEST_F(Vp9SettingsTest, MultipleSsrcsEnablesSvc) {
>+  cricket::VideoSendParameters parameters;
>+  parameters.codecs.push_back(GetEngineCodec("VP9"));
>+  ASSERT_TRUE(channel_->SetSendParameters(parameters));
> 
>-  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, true, nullptr, nullptr));
>+  std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs3);
>+
>+  FakeVideoSendStream* stream =
>+      AddSendStream(CreateSimStreamParams("cname", ssrcs));
>+
>+  webrtc::VideoSendStream::Config config = stream->GetConfig().Copy();
>+  EXPECT_EQ(ssrcs.size(), config.rtp.ssrcs.size());
>+
>+  FakeVideoCapturerWithTaskQueue capturer;
>+  EXPECT_EQ(cricket::CS_RUNNING,
>+            capturer.Start(capturer.GetSupportedFormats()->front()));
>+  EXPECT_TRUE(channel_->SetVideoSend(ssrcs[0], nullptr, &capturer));
>+  channel_->SetSend(true);
>+
>+  EXPECT_TRUE(capturer.CaptureFrame());
>+
>+  webrtc::VideoCodecVP9 vp9_settings;
>+  ASSERT_TRUE(stream->GetVp9Settings(&vp9_settings)) << "No VP9 config set.";
>+
>+  const size_t kNumSpatialLayers = ssrcs.size();
>+  const size_t kNumTemporalLayers = 3;
>+  EXPECT_EQ(vp9_settings.numberOfSpatialLayers, kNumSpatialLayers);
>+  EXPECT_EQ(vp9_settings.numberOfTemporalLayers, kNumTemporalLayers);
>+
>+  EXPECT_TRUE(channel_->SetVideoSend(ssrcs[0], nullptr, nullptr));
> }
> 
> class Vp9SettingsTestWithFieldTrial : public Vp9SettingsTest {
>@@ -2221,10 +3016,10 @@ class Vp9SettingsTestWithFieldTrial : public Vp9SettingsTest {
> 
>     FakeVideoSendStream* stream = SetUpSimulcast(false, false);
> 
>-    cricket::FakeVideoCapturer capturer;
>+    FakeVideoCapturerWithTaskQueue capturer;
>     EXPECT_EQ(cricket::CS_RUNNING,
>               capturer.Start(capturer.GetSupportedFormats()->front()));
>-    EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, true, nullptr, &capturer));
>+    EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, &capturer));
>     channel_->SetSend(true);
> 
>     EXPECT_TRUE(capturer.CaptureFrame());
>@@ -2234,7 +3029,7 @@ class Vp9SettingsTestWithFieldTrial : public Vp9SettingsTest {
>     EXPECT_EQ(num_spatial_layers, vp9_settings.numberOfSpatialLayers);
>     EXPECT_EQ(num_temporal_layers, vp9_settings.numberOfTemporalLayers);
> 
>-    EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, true, nullptr, nullptr));
>+    EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
>   }
> };
> 
>@@ -2331,8 +3126,8 @@ TEST_F(WebRtcVideoChannelTest, AdaptsOnOveruseAndChangeResolution) {
> 
>   AddSendStream();
> 
>-  cricket::FakeVideoCapturer capturer;
>-  ASSERT_TRUE(channel_->SetVideoSend(last_ssrc_, true, nullptr, &capturer));
>+  FakeVideoCapturerWithTaskQueue capturer;
>+  ASSERT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, &capturer));
>   ASSERT_EQ(cricket::CS_RUNNING,
>             capturer.Start(capturer.GetSupportedFormats()->front()));
>   ASSERT_TRUE(channel_->SetSend(true));
>@@ -2340,7 +3135,7 @@ TEST_F(WebRtcVideoChannelTest, AdaptsOnOveruseAndChangeResolution) {
>   ASSERT_EQ(1u, fake_call_->GetVideoSendStreams().size());
>   FakeVideoSendStream* send_stream = fake_call_->GetVideoSendStreams().front();
> 
>-  EXPECT_TRUE(capturer.CaptureCustomFrame(1280, 720, cricket::FOURCC_I420));
>+  EXPECT_TRUE(capturer.CaptureCustomFrame(1280, 720));
>   EXPECT_EQ(1, send_stream->GetNumberOfSwappedFrames());
>   EXPECT_EQ(1280, send_stream->GetLastWidth());
>   EXPECT_EQ(720, send_stream->GetLastHeight());
>@@ -2350,7 +3145,7 @@ TEST_F(WebRtcVideoChannelTest, AdaptsOnOveruseAndChangeResolution) {
>   wants.max_pixel_count =
>       send_stream->GetLastWidth() * send_stream->GetLastHeight() - 1;
>   send_stream->InjectVideoSinkWants(wants);
>-  EXPECT_TRUE(capturer.CaptureCustomFrame(1280, 720, cricket::FOURCC_I420));
>+  EXPECT_TRUE(capturer.CaptureCustomFrame(1280, 720));
>   EXPECT_EQ(2, send_stream->GetNumberOfSwappedFrames());
>   EXPECT_EQ(1280 * 3 / 4, send_stream->GetLastWidth());
>   EXPECT_EQ(720 * 3 / 4, send_stream->GetLastHeight());
>@@ -2359,13 +3154,13 @@ TEST_F(WebRtcVideoChannelTest, AdaptsOnOveruseAndChangeResolution) {
>   wants.max_pixel_count =
>       send_stream->GetLastWidth() * send_stream->GetLastHeight() - 1;
>   send_stream->InjectVideoSinkWants(wants);
>-  EXPECT_TRUE(capturer.CaptureCustomFrame(1280, 720, cricket::FOURCC_I420));
>+  EXPECT_TRUE(capturer.CaptureCustomFrame(1280, 720));
>   EXPECT_EQ(3, send_stream->GetNumberOfSwappedFrames());
>   EXPECT_EQ(1280 * 2 / 4, send_stream->GetLastWidth());
>   EXPECT_EQ(720 * 2 / 4, send_stream->GetLastHeight());
> 
>   // Change input resolution.
>-  EXPECT_TRUE(capturer.CaptureCustomFrame(1284, 724, cricket::FOURCC_I420));
>+  EXPECT_TRUE(capturer.CaptureCustomFrame(1284, 724));
>   EXPECT_EQ(4, send_stream->GetNumberOfSwappedFrames());
>   EXPECT_EQ(1284 / 2, send_stream->GetLastWidth());
>   EXPECT_EQ(724 / 2, send_stream->GetLastHeight());
>@@ -2379,7 +3174,7 @@ TEST_F(WebRtcVideoChannelTest, AdaptsOnOveruseAndChangeResolution) {
>   // Default step down is 3/5 pixel count, so go up by 5/3.
>   wants.target_pixel_count = (current_pixel_count * 5 / 3);
>   send_stream->InjectVideoSinkWants(wants);
>-  EXPECT_TRUE(capturer.CaptureCustomFrame(1284, 724, cricket::FOURCC_I420));
>+  EXPECT_TRUE(capturer.CaptureCustomFrame(1284, 724));
>   EXPECT_EQ(5, send_stream->GetNumberOfSwappedFrames());
>   EXPECT_EQ(1284 * 3 / 4, send_stream->GetLastWidth());
>   EXPECT_EQ(724 * 3 / 4, send_stream->GetLastHeight());
>@@ -2390,12 +3185,12 @@ TEST_F(WebRtcVideoChannelTest, AdaptsOnOveruseAndChangeResolution) {
>   wants.max_pixel_count = current_pixel_count * 4;
>   wants.target_pixel_count = (current_pixel_count * 5 / 3);
>   send_stream->InjectVideoSinkWants(wants);
>-  EXPECT_TRUE(capturer.CaptureCustomFrame(1284, 724, cricket::FOURCC_I420));
>+  EXPECT_TRUE(capturer.CaptureCustomFrame(1284, 724));
>   EXPECT_EQ(6, send_stream->GetNumberOfSwappedFrames());
>   EXPECT_EQ(1284, send_stream->GetLastWidth());
>   EXPECT_EQ(724, send_stream->GetLastHeight());
> 
>-  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, true, nullptr, nullptr));
>+  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
> }
> 
> TEST_F(WebRtcVideoChannelTest, PreviousAdaptationDoesNotApplyToScreenshare) {
>@@ -2404,7 +3199,7 @@ TEST_F(WebRtcVideoChannelTest, PreviousAdaptationDoesNotApplyToScreenshare) {
>   parameters.codecs.push_back(codec);
> 
>   MediaConfig media_config = GetMediaConfig();
>-  media_config.video.enable_cpu_overuse_detection = true;
>+  media_config.video.enable_cpu_adaptation = true;
>   channel_.reset(
>       engine_.CreateChannel(fake_call_.get(), media_config, VideoOptions()));
>   channel_->OnReadyToSend(true);
>@@ -2412,19 +3207,18 @@ TEST_F(WebRtcVideoChannelTest, PreviousAdaptationDoesNotApplyToScreenshare) {
> 
>   AddSendStream();
> 
>-  cricket::FakeVideoCapturer capturer;
>+  FakeVideoCapturerWithTaskQueue capturer;
>   ASSERT_EQ(cricket::CS_RUNNING,
>             capturer.Start(capturer.GetSupportedFormats()->front()));
>   ASSERT_TRUE(channel_->SetSend(true));
>   cricket::VideoOptions camera_options;
>   camera_options.is_screencast = false;
>-  channel_->SetVideoSend(last_ssrc_, true /* enable */, &camera_options,
>-                         &capturer);
>+  channel_->SetVideoSend(last_ssrc_, &camera_options, &capturer);
> 
>   ASSERT_EQ(1u, fake_call_->GetVideoSendStreams().size());
>   FakeVideoSendStream* send_stream = fake_call_->GetVideoSendStreams().front();
> 
>-  EXPECT_TRUE(capturer.CaptureCustomFrame(1280, 720, cricket::FOURCC_I420));
>+  EXPECT_TRUE(capturer.CaptureCustomFrame(1280, 720));
>   EXPECT_EQ(1, send_stream->GetNumberOfSwappedFrames());
>   EXPECT_EQ(1280, send_stream->GetLastWidth());
>   EXPECT_EQ(720, send_stream->GetLastHeight());
>@@ -2434,20 +3228,19 @@ TEST_F(WebRtcVideoChannelTest, PreviousAdaptationDoesNotApplyToScreenshare) {
>   wants.max_pixel_count =
>       send_stream->GetLastWidth() * send_stream->GetLastHeight() - 1;
>   send_stream->InjectVideoSinkWants(wants);
>-  EXPECT_TRUE(capturer.CaptureCustomFrame(1280, 720, cricket::FOURCC_I420));
>+  EXPECT_TRUE(capturer.CaptureCustomFrame(1280, 720));
>   EXPECT_EQ(2, send_stream->GetNumberOfSwappedFrames());
>   EXPECT_EQ(1280 * 3 / 4, send_stream->GetLastWidth());
>   EXPECT_EQ(720 * 3 / 4, send_stream->GetLastHeight());
> 
>   // Switch to screen share. Expect no CPU adaptation.
>-  cricket::FakeVideoCapturer screen_share(true);
>+  FakeVideoCapturerWithTaskQueue screen_share(true);
>   ASSERT_EQ(cricket::CS_RUNNING,
>             screen_share.Start(screen_share.GetSupportedFormats()->front()));
>   cricket::VideoOptions screenshare_options;
>   screenshare_options.is_screencast = true;
>-  channel_->SetVideoSend(last_ssrc_, true /* enable */, &screenshare_options,
>-                         &screen_share);
>-  EXPECT_TRUE(screen_share.CaptureCustomFrame(1284, 724, cricket::FOURCC_I420));
>+  channel_->SetVideoSend(last_ssrc_, &screenshare_options, &screen_share);
>+  EXPECT_TRUE(screen_share.CaptureCustomFrame(1284, 724));
>   ASSERT_EQ(2, fake_call_->GetNumCreatedSendStreams());
>   send_stream = fake_call_->GetVideoSendStreams().front();
>   EXPECT_EQ(1, send_stream->GetNumberOfSwappedFrames());
>@@ -2455,21 +3248,20 @@ TEST_F(WebRtcVideoChannelTest, PreviousAdaptationDoesNotApplyToScreenshare) {
>   EXPECT_EQ(724, send_stream->GetLastHeight());
> 
>   // Switch back to the normal capturer. Expect the frame to be CPU adapted.
>-  channel_->SetVideoSend(last_ssrc_, true /* enable */, &camera_options,
>-                         &capturer);
>+  channel_->SetVideoSend(last_ssrc_, &camera_options, &capturer);
>   send_stream = fake_call_->GetVideoSendStreams().front();
>   // We have a new fake send stream, so it doesn't remember the old sink wants.
>   // In practice, it will be populated from
>   // VideoStreamEncoder::VideoSourceProxy::SetSource(), so simulate that here.
>   send_stream->InjectVideoSinkWants(wants);
>-  EXPECT_TRUE(capturer.CaptureCustomFrame(1280, 720, cricket::FOURCC_I420));
>+  EXPECT_TRUE(capturer.CaptureCustomFrame(1280, 720));
>   ASSERT_EQ(3, fake_call_->GetNumCreatedSendStreams());
>   send_stream = fake_call_->GetVideoSendStreams().front();
>   EXPECT_EQ(1, send_stream->GetNumberOfSwappedFrames());
>   EXPECT_EQ(1280 * 3 / 4, send_stream->GetLastWidth());
>   EXPECT_EQ(720 * 3 / 4, send_stream->GetLastHeight());
> 
>-  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, true, nullptr, nullptr));
>+  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
> }
> 
> // TODO(asapersson): Remove this test when the balanced field trial is removed.
>@@ -2481,7 +3273,7 @@ void WebRtcVideoChannelTest::TestDegradationPreference(
>   parameters.codecs.push_back(codec);
> 
>   MediaConfig media_config = GetMediaConfig();
>-  media_config.video.enable_cpu_overuse_detection = true;
>+  media_config.video.enable_cpu_adaptation = true;
>   channel_.reset(
>       engine_.CreateChannel(fake_call_.get(), media_config, VideoOptions()));
>   channel_->OnReadyToSend(true);
>@@ -2490,9 +3282,9 @@ void WebRtcVideoChannelTest::TestDegradationPreference(
> 
>   AddSendStream();
> 
>-  cricket::FakeVideoCapturer capturer;
>+  FakeVideoCapturerWithTaskQueue capturer;
>   VideoOptions options;
>-  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, true, &options, &capturer));
>+  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &capturer));
>   cricket::VideoFormat capture_format = capturer.GetSupportedFormats()->front();
>   EXPECT_EQ(cricket::CS_RUNNING, capturer.Start(capture_format));
> 
>@@ -2503,7 +3295,7 @@ void WebRtcVideoChannelTest::TestDegradationPreference(
>             send_stream->resolution_scaling_enabled());
>   EXPECT_EQ(fps_scaling_enabled, send_stream->framerate_scaling_enabled());
> 
>-  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, true, nullptr, nullptr));
>+  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
> }
> 
> void WebRtcVideoChannelTest::TestCpuAdaptation(bool enable_overuse,
>@@ -2515,7 +3307,7 @@ void WebRtcVideoChannelTest::TestCpuAdaptation(bool enable_overuse,
> 
>   MediaConfig media_config = GetMediaConfig();
>   if (enable_overuse) {
>-    media_config.video.enable_cpu_overuse_detection = true;
>+    media_config.video.enable_cpu_adaptation = true;
>   }
>   channel_.reset(
>       engine_.CreateChannel(fake_call_.get(), media_config, VideoOptions()));
>@@ -2525,10 +3317,10 @@ void WebRtcVideoChannelTest::TestCpuAdaptation(bool enable_overuse,
> 
>   AddSendStream();
> 
>-  cricket::FakeVideoCapturer capturer;
>+  FakeVideoCapturerWithTaskQueue capturer;
>   VideoOptions options;
>   options.is_screencast = is_screenshare;
>-  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, true, &options, &capturer));
>+  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &capturer));
>   cricket::VideoFormat capture_format = capturer.GetSupportedFormats()->front();
>   capture_format.interval = rtc::kNumNanosecsPerSec / kDefaultFps;
>   EXPECT_EQ(cricket::CS_RUNNING, capturer.Start(capture_format));
>@@ -2548,7 +3340,7 @@ void WebRtcVideoChannelTest::TestCpuAdaptation(bool enable_overuse,
>     EXPECT_EQ(capture_format.width, send_stream->GetLastWidth());
>     EXPECT_EQ(capture_format.height, send_stream->GetLastHeight());
> 
>-    EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, true, nullptr, nullptr));
>+    EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
>     return;
>   }
> 
>@@ -2609,7 +3401,7 @@ void WebRtcVideoChannelTest::TestCpuAdaptation(bool enable_overuse,
>   EXPECT_EQ(capture_format.width, send_stream->GetLastWidth());
>   EXPECT_EQ(capture_format.height, send_stream->GetLastHeight());
> 
>-  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, true, nullptr, nullptr));
>+  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
> }
> 
> TEST_F(WebRtcVideoChannelTest, EstimatesNtpStartTimeCorrectly) {
>@@ -2649,6 +3441,7 @@ TEST_F(WebRtcVideoChannelTest, EstimatesNtpStartTimeCorrectly) {
> }
> 
> TEST_F(WebRtcVideoChannelTest, SetDefaultSendCodecs) {
>+  AssignDefaultAptRtxTypes();
>   ASSERT_TRUE(channel_->SetSendParameters(send_parameters_));
> 
>   VideoCodec codec;
>@@ -2677,8 +3470,7 @@ TEST_F(WebRtcVideoChannelTest, SetDefaultSendCodecs) {
> // when the field trials are not enabled.
> // TODO(brandtr): Remove or update these tests when FlexFEC _is_ enabled by
> // default.
>-TEST_F(WebRtcVideoChannelTest,
>-       FlexfecSendCodecWithoutSsrcNotExposedByDefault) {
>+TEST_F(WebRtcVideoChannelTest, FlexfecSendCodecWithoutSsrcNotExposedByDefault) {
>   FakeVideoSendStream* stream = AddSendStream();
>   webrtc::VideoSendStream::Config config = stream->GetConfig().Copy();
> 
>@@ -2697,8 +3489,7 @@ TEST_F(WebRtcVideoChannelTest, FlexfecSendCodecWithSsrcNotExposedByDefault) {
>   EXPECT_TRUE(config.rtp.flexfec.protected_media_ssrcs.empty());
> }
> 
>-TEST_F(WebRtcVideoChannelTest,
>-       FlexfecRecvCodecWithoutSsrcNotExposedByDefault) {
>+TEST_F(WebRtcVideoChannelTest, FlexfecRecvCodecWithoutSsrcNotExposedByDefault) {
>   AddRecvStream();
> 
>   const std::vector<FakeFlexfecReceiveStream*>& streams =
>@@ -2736,6 +3527,13 @@ TEST_F(WebRtcVideoChannelFlexfecRecvTest, SetDefaultRecvCodecsWithoutSsrc) {
>   const std::vector<FakeFlexfecReceiveStream*>& streams =
>       fake_call_->GetFlexfecReceiveStreams();
>   EXPECT_TRUE(streams.empty());
>+
>+  const std::vector<FakeVideoReceiveStream*>& video_streams =
>+      fake_call_->GetVideoReceiveStreams();
>+  ASSERT_EQ(1U, video_streams.size());
>+  const FakeVideoReceiveStream& video_stream = *video_streams.front();
>+  EXPECT_EQ(0, video_stream.GetNumAddedSecondarySinks());
>+  EXPECT_EQ(0, video_stream.GetNumRemovedSecondarySinks());
> }
> 
> TEST_F(WebRtcVideoChannelFlexfecRecvTest, SetDefaultRecvCodecsWithSsrc) {
>@@ -2755,8 +3553,10 @@ TEST_F(WebRtcVideoChannelFlexfecRecvTest, SetDefaultRecvCodecsWithSsrc) {
>   const std::vector<FakeVideoReceiveStream*>& video_streams =
>       fake_call_->GetVideoReceiveStreams();
>   ASSERT_EQ(1U, video_streams.size());
>+  const FakeVideoReceiveStream& video_stream = *video_streams.front();
>+  EXPECT_EQ(1, video_stream.GetNumAddedSecondarySinks());
>   const webrtc::VideoReceiveStream::Config& video_config =
>-      video_streams.front()->GetConfig();
>+      video_stream.GetConfig();
>   EXPECT_TRUE(video_config.rtp.protected_by_flexfec);
> }
> 
>@@ -2769,7 +3569,12 @@ TEST_F(WebRtcVideoChannelFlexfecRecvTest,
>   AddRecvStream(
>       CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], kFlexfecSsrc));
>   EXPECT_EQ(1, fake_call_->GetNumCreatedReceiveStreams());
>-  EXPECT_EQ(1U, fake_call_->GetVideoReceiveStreams().size());
>+  const std::vector<FakeVideoReceiveStream*>& video_streams =
>+      fake_call_->GetVideoReceiveStreams();
>+  ASSERT_EQ(1U, video_streams.size());
>+  const FakeVideoReceiveStream& video_stream = *video_streams.front();
>+  EXPECT_EQ(0, video_stream.GetNumAddedSecondarySinks());
>+  EXPECT_EQ(0, video_stream.GetNumRemovedSecondarySinks());
> 
>   // Enable FlexFEC.
>   recv_parameters.codecs.push_back(GetEngineCodec("flexfec-03"));
>@@ -2780,6 +3585,8 @@ TEST_F(WebRtcVideoChannelFlexfecRecvTest,
>       << "Enabling FlexFEC should not create VideoReceiveStream.";
>   EXPECT_EQ(1U, fake_call_->GetFlexfecReceiveStreams().size())
>       << "Enabling FlexFEC should create a single FlexfecReceiveStream.";
>+  EXPECT_EQ(1, video_stream.GetNumAddedSecondarySinks());
>+  EXPECT_EQ(0, video_stream.GetNumRemovedSecondarySinks());
> }
> 
> TEST_F(WebRtcVideoChannelFlexfecRecvTest,
>@@ -2792,8 +3599,13 @@ TEST_F(WebRtcVideoChannelFlexfecRecvTest,
>   AddRecvStream(
>       CreatePrimaryWithFecFrStreamParams("cname", kSsrcs1[0], kFlexfecSsrc));
>   EXPECT_EQ(2, fake_call_->GetNumCreatedReceiveStreams());
>-  EXPECT_EQ(1U, fake_call_->GetVideoReceiveStreams().size());
>   EXPECT_EQ(1U, fake_call_->GetFlexfecReceiveStreams().size());
>+  const std::vector<FakeVideoReceiveStream*>& video_streams =
>+      fake_call_->GetVideoReceiveStreams();
>+  ASSERT_EQ(1U, video_streams.size());
>+  const FakeVideoReceiveStream& video_stream = *video_streams.front();
>+  EXPECT_EQ(1, video_stream.GetNumAddedSecondarySinks());
>+  EXPECT_EQ(0, video_stream.GetNumRemovedSecondarySinks());
> 
>   // Disable FlexFEC.
>   recv_parameters.codecs.clear();
>@@ -2805,6 +3617,8 @@ TEST_F(WebRtcVideoChannelFlexfecRecvTest,
>       << "Disabling FlexFEC should not destroy VideoReceiveStream.";
>   EXPECT_TRUE(fake_call_->GetFlexfecReceiveStreams().empty())
>       << "Disabling FlexFEC should destroy FlexfecReceiveStream.";
>+  EXPECT_EQ(1, video_stream.GetNumAddedSecondarySinks());
>+  EXPECT_EQ(1, video_stream.GetNumRemovedSecondarySinks());
> }
> 
> // TODO(brandtr): When FlexFEC is no longer behind a field trial, merge all
>@@ -2818,8 +3632,7 @@ class WebRtcVideoChannelFlexfecSendRecvTest : public WebRtcVideoChannelTest {
>   }
> };
> 
>-TEST_F(WebRtcVideoChannelFlexfecSendRecvTest,
>-       SetDefaultSendCodecsWithoutSsrc) {
>+TEST_F(WebRtcVideoChannelFlexfecSendRecvTest, SetDefaultSendCodecsWithoutSsrc) {
>   FakeVideoSendStream* stream = AddSendStream();
>   webrtc::VideoSendStream::Config config = stream->GetConfig().Copy();
> 
>@@ -2915,7 +3728,7 @@ TEST_F(WebRtcVideoChannelFlexfecRecvTest,
>   webrtc::VideoSendStream::Config config = stream->GetConfig().Copy();
> 
>   EXPECT_EQ(-1, config.rtp.flexfec.payload_type);
>-  EXPECT_EQ(0, config.rtp.flexfec.ssrc);
>+  EXPECT_EQ(0u, config.rtp.flexfec.ssrc);
>   EXPECT_TRUE(config.rtp.flexfec.protected_media_ssrcs.empty());
> }
> 
>@@ -2931,7 +3744,7 @@ TEST_F(WebRtcVideoChannelFlexfecRecvTest,
>   webrtc::VideoSendStream::Config config = stream->GetConfig().Copy();
> 
>   EXPECT_EQ(-1, config.rtp.flexfec.payload_type);
>-  EXPECT_EQ(0, config.rtp.flexfec.ssrc);
>+  EXPECT_EQ(0u, config.rtp.flexfec.ssrc);
>   EXPECT_TRUE(config.rtp.flexfec.protected_media_ssrcs.empty());
> }
> 
>@@ -3064,8 +3877,8 @@ TEST_F(WebRtcVideoChannelTest, SetSendCodecsChangesExistingStreams) {
>   channel_->SetSend(true);
> 
>   FakeVideoSendStream* stream = AddSendStream();
>-  cricket::FakeVideoCapturer capturer;
>-  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, true, nullptr, &capturer));
>+  FakeVideoCapturerWithTaskQueue capturer;
>+  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, &capturer));
> 
>   std::vector<webrtc::VideoStream> streams = stream->GetVideoStreams();
>   EXPECT_EQ(kDefaultQpMax, streams[0].max_qp);
>@@ -3076,7 +3889,7 @@ TEST_F(WebRtcVideoChannelTest, SetSendCodecsChangesExistingStreams) {
>   ASSERT_TRUE(channel_->SetSendParameters(parameters));
>   streams = fake_call_->GetVideoSendStreams()[0]->GetVideoStreams();
>   EXPECT_EQ(kDefaultQpMax + 1, streams[0].max_qp);
>-  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, true, nullptr, nullptr));
>+  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
> }
> 
> TEST_F(WebRtcVideoChannelTest, SetSendCodecsWithBitrates) {
>@@ -3093,8 +3906,7 @@ TEST_F(WebRtcVideoChannelTest, SetSendCodecsWithHighMaxBitrate) {
> 
> TEST_F(WebRtcVideoChannelTest,
>        SetSendCodecsWithoutBitratesUsesCorrectDefaults) {
>-  SetSendCodecsShouldWorkForBitrates(
>-      "", 0, "", -1, "", -1);
>+  SetSendCodecsShouldWorkForBitrates("", 0, "", -1, "", -1);
> }
> 
> TEST_F(WebRtcVideoChannelTest, SetSendCodecsCapsMinAndStartBitrate) {
>@@ -3114,74 +3926,144 @@ TEST_F(WebRtcVideoChannelTest, SetSendCodecsWithBitratesAndMaxSendBandwidth) {
>   send_parameters_.codecs[0].params[kCodecParamStartBitrate] = "200";
>   send_parameters_.codecs[0].params[kCodecParamMaxBitrate] = "300";
>   send_parameters_.max_bandwidth_bps = 400000;
>-  EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
>-  EXPECT_EQ(100000, fake_call_->GetConfig().bitrate_config.min_bitrate_bps);
>-  EXPECT_EQ(200000, fake_call_->GetConfig().bitrate_config.start_bitrate_bps);
>   // We expect max_bandwidth_bps to take priority, if set.
>-  EXPECT_EQ(400000, fake_call_->GetConfig().bitrate_config.max_bitrate_bps);
>+  ExpectSetBitrateParameters(100000, 200000, 400000);
>+  EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
>+  // Since the codec isn't changing, start_bitrate_bps should be -1.
>+  ExpectSetBitrateParameters(100000, -1, 350000);
> 
>   // Decrease max_bandwidth_bps.
>   send_parameters_.max_bandwidth_bps = 350000;
>   EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
>-  EXPECT_EQ(100000, fake_call_->GetConfig().bitrate_config.min_bitrate_bps);
>-  // Since the codec isn't changing, start_bitrate_bps should be -1.
>-  EXPECT_EQ(-1, fake_call_->GetConfig().bitrate_config.start_bitrate_bps);
>-  EXPECT_EQ(350000, fake_call_->GetConfig().bitrate_config.max_bitrate_bps);
> 
>   // Now try again with the values flipped around.
>   send_parameters_.codecs[0].params[kCodecParamMaxBitrate] = "400";
>   send_parameters_.max_bandwidth_bps = 300000;
>+  ExpectSetBitrateParameters(100000, 200000, 300000);
>   EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
>-  EXPECT_EQ(100000, fake_call_->GetConfig().bitrate_config.min_bitrate_bps);
>-  EXPECT_EQ(200000, fake_call_->GetConfig().bitrate_config.start_bitrate_bps);
>-  EXPECT_EQ(300000, fake_call_->GetConfig().bitrate_config.max_bitrate_bps);
> 
>   // If we change the codec max, max_bandwidth_bps should still apply.
>   send_parameters_.codecs[0].params[kCodecParamMaxBitrate] = "350";
>+  ExpectSetBitrateParameters(100000, 200000, 300000);
>   EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
>-  EXPECT_EQ(100000, fake_call_->GetConfig().bitrate_config.min_bitrate_bps);
>-  EXPECT_EQ(200000, fake_call_->GetConfig().bitrate_config.start_bitrate_bps);
>-  EXPECT_EQ(300000, fake_call_->GetConfig().bitrate_config.max_bitrate_bps);
> }
> 
>-TEST_F(WebRtcVideoChannelTest,
>-       SetMaxSendBandwidthShouldPreserveOtherBitrates) {
>+TEST_F(WebRtcVideoChannelTest, SetMaxSendBandwidthShouldPreserveOtherBitrates) {
>   SetSendCodecsShouldWorkForBitrates("100", 100000, "150", 150000, "200",
>                                      200000);
>   send_parameters_.max_bandwidth_bps = 300000;
>+  // Setting max bitrate should keep previous min bitrate.
>+  // Setting max bitrate should not reset start bitrate.
>+  ExpectSetBitrateParameters(100000, -1, 300000);
>   EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
>-  EXPECT_EQ(100000, fake_call_->GetConfig().bitrate_config.min_bitrate_bps)
>-      << "Setting max bitrate should keep previous min bitrate.";
>-  EXPECT_EQ(-1, fake_call_->GetConfig().bitrate_config.start_bitrate_bps)
>-      << "Setting max bitrate should not reset start bitrate.";
>-  EXPECT_EQ(300000, fake_call_->GetConfig().bitrate_config.max_bitrate_bps);
> }
> 
> TEST_F(WebRtcVideoChannelTest, SetMaxSendBandwidthShouldBeRemovable) {
>   send_parameters_.max_bandwidth_bps = 300000;
>+  ExpectSetMaxBitrate(300000);
>   EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
>-  EXPECT_EQ(300000, fake_call_->GetConfig().bitrate_config.max_bitrate_bps);
>   // -1 means to disable max bitrate (set infinite).
>   send_parameters_.max_bandwidth_bps = -1;
>+  ExpectSetMaxBitrate(-1);
>   EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
>-  EXPECT_EQ(-1, fake_call_->GetConfig().bitrate_config.max_bitrate_bps)
>-      << "Setting zero max bitrate did not reset start bitrate.";
> }
> 
>-TEST_F(WebRtcVideoChannelTest, SetMaxSendBandwidthAndAddSendStream) {
>+TEST_F(WebRtcVideoChannelTest, SetMaxSendBandwidthAndAddSendStream) {
>+  send_parameters_.max_bandwidth_bps = 99999;
>+  FakeVideoSendStream* stream = AddSendStream();
>+  ExpectSetMaxBitrate(send_parameters_.max_bandwidth_bps);
>+  ASSERT_TRUE(channel_->SetSendParameters(send_parameters_));
>+  ASSERT_EQ(1u, stream->GetVideoStreams().size());
>+  EXPECT_EQ(send_parameters_.max_bandwidth_bps,
>+            stream->GetVideoStreams()[0].max_bitrate_bps);
>+
>+  send_parameters_.max_bandwidth_bps = 77777;
>+  ExpectSetMaxBitrate(send_parameters_.max_bandwidth_bps);
>+  ASSERT_TRUE(channel_->SetSendParameters(send_parameters_));
>+  EXPECT_EQ(send_parameters_.max_bandwidth_bps,
>+            stream->GetVideoStreams()[0].max_bitrate_bps);
>+}
>+
>+// Tests that when the codec specific max bitrate and VideoSendParameters
>+// max_bandwidth_bps are used, that it sets the VideoStream's max bitrate
>+// appropriately.
>+TEST_F(WebRtcVideoChannelTest,
>+       MaxBitratePrioritizesVideoSendParametersOverCodecMaxBitrate) {
>+  send_parameters_.codecs[0].params[kCodecParamMinBitrate] = "100";
>+  send_parameters_.codecs[0].params[kCodecParamStartBitrate] = "200";
>+  send_parameters_.codecs[0].params[kCodecParamMaxBitrate] = "300";
>+  send_parameters_.max_bandwidth_bps = -1;
>+  AddSendStream();
>+  ExpectSetMaxBitrate(300000);
>+  ASSERT_TRUE(channel_->SetSendParameters(send_parameters_));
>+
>+  std::vector<FakeVideoSendStream*> video_send_streams = GetFakeSendStreams();
>+  ASSERT_EQ(1u, video_send_streams.size());
>+  FakeVideoSendStream* video_send_stream = video_send_streams[0];
>+  ASSERT_EQ(1u, video_send_streams[0]->GetVideoStreams().size());
>+  // First the max bitrate is set based upon the codec param.
>+  EXPECT_EQ(300000,
>+            video_send_streams[0]->GetVideoStreams()[0].max_bitrate_bps);
>+
>+  // The VideoSendParameters max bitrate overrides the codec's.
>+  send_parameters_.max_bandwidth_bps = 500000;
>+  ExpectSetMaxBitrate(send_parameters_.max_bandwidth_bps);
>+  ASSERT_TRUE(channel_->SetSendParameters(send_parameters_));
>+  ASSERT_EQ(1u, video_send_stream->GetVideoStreams().size());
>+  EXPECT_EQ(500000, video_send_stream->GetVideoStreams()[0].max_bitrate_bps);
>+}
>+
>+// Tests that when the codec specific max bitrate and RtpParameters
>+// max_bitrate_bps are used, that it sets the VideoStream's max bitrate
>+// appropriately.
>+TEST_F(WebRtcVideoChannelTest,
>+       MaxBitratePrioritizesRtpParametersOverCodecMaxBitrate) {
>+  send_parameters_.codecs[0].params[kCodecParamMinBitrate] = "100";
>+  send_parameters_.codecs[0].params[kCodecParamStartBitrate] = "200";
>+  send_parameters_.codecs[0].params[kCodecParamMaxBitrate] = "300";
>+  send_parameters_.max_bandwidth_bps = -1;
>+  AddSendStream();
>+  ExpectSetMaxBitrate(300000);
>+  ASSERT_TRUE(channel_->SetSendParameters(send_parameters_));
>+
>+  std::vector<FakeVideoSendStream*> video_send_streams = GetFakeSendStreams();
>+  ASSERT_EQ(1u, video_send_streams.size());
>+  FakeVideoSendStream* video_send_stream = video_send_streams[0];
>+  ASSERT_EQ(1u, video_send_stream->GetVideoStreams().size());
>+  // First the max bitrate is set based upon the codec param.
>+  EXPECT_EQ(300000, video_send_stream->GetVideoStreams()[0].max_bitrate_bps);
>+
>+  // The RtpParameter max bitrate overrides the codec's.
>+  webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
>+  ASSERT_EQ(1u, parameters.encodings.size());
>+  parameters.encodings[0].max_bitrate_bps = 500000;
>+  EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
>+  ASSERT_EQ(1u, video_send_stream->GetVideoStreams().size());
>+  EXPECT_EQ(parameters.encodings[0].max_bitrate_bps,
>+            video_send_stream->GetVideoStreams()[0].max_bitrate_bps);
>+}
>+
>+TEST_F(WebRtcVideoChannelTest,
>+       MaxBitrateIsMinimumOfMaxSendBandwidthAndMaxEncodingBitrate) {
>   send_parameters_.max_bandwidth_bps = 99999;
>   FakeVideoSendStream* stream = AddSendStream();
>+  ExpectSetMaxBitrate(send_parameters_.max_bandwidth_bps);
>   ASSERT_TRUE(channel_->SetSendParameters(send_parameters_));
>-  EXPECT_EQ(send_parameters_.max_bandwidth_bps,
>-            fake_call_->GetConfig().bitrate_config.max_bitrate_bps);
>   ASSERT_EQ(1u, stream->GetVideoStreams().size());
>   EXPECT_EQ(send_parameters_.max_bandwidth_bps,
>             stream->GetVideoStreams()[0].max_bitrate_bps);
> 
>-  send_parameters_.max_bandwidth_bps = 77777;
>-  ASSERT_TRUE(channel_->SetSendParameters(send_parameters_));
>-  EXPECT_EQ(send_parameters_.max_bandwidth_bps,
>-            fake_call_->GetConfig().bitrate_config.max_bitrate_bps);
>+  // Get and set the rtp encoding parameters.
>+  webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
>+  EXPECT_EQ(1u, parameters.encodings.size());
>+
>+  parameters.encodings[0].max_bitrate_bps = 99999 - 1;
>+  EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
>+  EXPECT_EQ(parameters.encodings[0].max_bitrate_bps,
>+            stream->GetVideoStreams()[0].max_bitrate_bps);
>+
>+  parameters.encodings[0].max_bitrate_bps = 99999 + 1;
>+  EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
>   EXPECT_EQ(send_parameters_.max_bandwidth_bps,
>             stream->GetVideoStreams()[0].max_bitrate_bps);
> }
>@@ -3194,8 +4076,8 @@ TEST_F(WebRtcVideoChannelTest, SetMaxSendBitrateCanIncreaseSenderBitrate) {
> 
>   FakeVideoSendStream* stream = AddSendStream();
> 
>-  cricket::FakeVideoCapturer capturer;
>-  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, true, nullptr, &capturer));
>+  FakeVideoCapturerWithTaskQueue capturer;
>+  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, &capturer));
>   EXPECT_EQ(cricket::CS_RUNNING,
>             capturer.Start(capturer.GetSupportedFormats()->front()));
> 
>@@ -3209,7 +4091,7 @@ TEST_F(WebRtcVideoChannelTest, SetMaxSendBitrateCanIncreaseSenderBitrate) {
>   EXPECT_TRUE(capturer.CaptureFrame());
>   streams = stream->GetVideoStreams();
>   EXPECT_EQ(initial_max_bitrate_bps * 2, streams[0].max_bitrate_bps);
>-  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, true, nullptr, nullptr));
>+  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
> }
> 
> TEST_F(WebRtcVideoChannelTest,
>@@ -3223,8 +4105,8 @@ TEST_F(WebRtcVideoChannelTest,
>       cricket::CreateSimStreamParams("cname", MAKE_VECTOR(kSsrcs3)));
> 
>   // Send a frame to make sure this scales up to >1 stream (simulcast).
>-  cricket::FakeVideoCapturer capturer;
>-  EXPECT_TRUE(channel_->SetVideoSend(kSsrcs3[0], true, nullptr, &capturer));
>+  FakeVideoCapturerWithTaskQueue capturer;
>+  EXPECT_TRUE(channel_->SetVideoSend(kSsrcs3[0], nullptr, &capturer));
>   EXPECT_EQ(cricket::CS_RUNNING,
>             capturer.Start(capturer.GetSupportedFormats()->front()));
>   EXPECT_TRUE(capturer.CaptureFrame());
>@@ -3243,7 +4125,7 @@ TEST_F(WebRtcVideoChannelTest,
>   int increased_max_bitrate_bps = GetTotalMaxBitrateBps(streams);
>   EXPECT_EQ(initial_max_bitrate_bps * 2, increased_max_bitrate_bps);
> 
>-  EXPECT_TRUE(channel_->SetVideoSend(kSsrcs3[0], true, nullptr, nullptr));
>+  EXPECT_TRUE(channel_->SetVideoSend(kSsrcs3[0], nullptr, nullptr));
> }
> 
> TEST_F(WebRtcVideoChannelTest, SetSendCodecsWithMaxQuantization) {
>@@ -3252,7 +4134,7 @@ TEST_F(WebRtcVideoChannelTest, SetSendCodecsWithMaxQuantization) {
>   parameters.codecs.push_back(GetEngineCodec("VP8"));
>   parameters.codecs[0].params[kCodecParamMaxQuantization] = kMaxQuantization;
>   EXPECT_TRUE(channel_->SetSendParameters(parameters));
>-  EXPECT_EQ(static_cast<unsigned int>(atoi(kMaxQuantization)),
>+  EXPECT_EQ(atoi(kMaxQuantization),
>             AddSendStream()->GetVideoStreams().back().max_qp);
> 
>   VideoCodec codec;
>@@ -3333,9 +4215,9 @@ TEST_F(WebRtcVideoChannelTest, SetRecvCodecsWithRtx) {
>   rtx_codec2.SetParam("apt", rtx_codec.id);
>   parameters.codecs.push_back(rtx_codec2);
> 
>-  EXPECT_FALSE(channel_->SetRecvParameters(parameters)) <<
>-      "RTX codec with another RTX as associated payload type should be "
>-      "rejected.";
>+  EXPECT_FALSE(channel_->SetRecvParameters(parameters))
>+      << "RTX codec with another RTX as associated payload type should be "
>+         "rejected.";
> }
> 
> TEST_F(WebRtcVideoChannelTest, SetRecvCodecsWithChangedRtxPayloadType) {
>@@ -3632,12 +4514,17 @@ TEST_F(WebRtcVideoChannelTest, TestSetSendRtcpReducedSize) {
>   // Create stream, expecting that default mode is "compound".
>   FakeVideoSendStream* stream1 = AddSendStream();
>   EXPECT_EQ(webrtc::RtcpMode::kCompound, stream1->GetConfig().rtp.rtcp_mode);
>+  webrtc::RtpParameters rtp_parameters =
>+      channel_->GetRtpSendParameters(last_ssrc_);
>+  EXPECT_FALSE(rtp_parameters.rtcp.reduced_size);
> 
>   // Now enable reduced size mode.
>   send_parameters_.rtcp.reduced_size = true;
>   EXPECT_TRUE(channel_->SetSendParameters(send_parameters_));
>   stream1 = fake_call_->GetVideoSendStreams()[0];
>   EXPECT_EQ(webrtc::RtcpMode::kReducedSize, stream1->GetConfig().rtp.rtcp_mode);
>+  rtp_parameters = channel_->GetRtpSendParameters(last_ssrc_);
>+  EXPECT_TRUE(rtp_parameters.rtcp.reduced_size);
> 
>   // Create a new stream and ensure it picks up the reduced size mode.
>   FakeVideoSendStream* stream2 = AddSendStream();
>@@ -3760,18 +4647,6 @@ TEST_F(WebRtcVideoChannelTest, GetStatsReportsUpperResolution) {
>   EXPECT_EQ(90, info.senders[0].send_frame_height);
> }
> 
>-TEST_F(WebRtcVideoChannelTest, GetStatsReportsPreferredBitrate) {
>-  FakeVideoSendStream* stream = AddSendStream();
>-  webrtc::VideoSendStream::Stats stats;
>-  stats.preferred_media_bitrate_bps = 5;
>-  stream->SetStats(stats);
>-
>-  cricket::VideoMediaInfo info;
>-  ASSERT_TRUE(channel_->GetStats(&info));
>-  ASSERT_EQ(1u, info.senders.size());
>-  EXPECT_EQ(5, info.senders[0].preferred_bitrate);
>-}
>-
> TEST_F(WebRtcVideoChannelTest, GetStatsReportsCpuAdaptationStats) {
>   FakeVideoSendStream* stream = AddSendStream();
>   webrtc::VideoSendStream::Stats stats;
>@@ -3817,8 +4692,7 @@ TEST_F(WebRtcVideoChannelTest,
>             info.senders[0].adapt_reason);
> }
> 
>-TEST_F(WebRtcVideoChannelTest,
>-       GetStatsTranslatesSendRtcpPacketTypesCorrectly) {
>+TEST_F(WebRtcVideoChannelTest, GetStatsTranslatesSendRtcpPacketTypesCorrectly) {
>   FakeVideoSendStream* stream = AddSendStream();
>   webrtc::VideoSendStream::Stats stats;
>   stats.substreams[17].rtcp_packet_type_counts.fir_packets = 2;
>@@ -3850,11 +4724,11 @@ TEST_F(WebRtcVideoChannelTest,
>   cricket::VideoMediaInfo info;
>   ASSERT_TRUE(channel_->GetStats(&info));
>   EXPECT_EQ(stats.rtcp_packet_type_counts.fir_packets,
>-            info.receivers[0].firs_sent);
>+            rtc::checked_cast<unsigned int>(info.receivers[0].firs_sent));
>   EXPECT_EQ(stats.rtcp_packet_type_counts.nack_packets,
>-            info.receivers[0].nacks_sent);
>+            rtc::checked_cast<unsigned int>(info.receivers[0].nacks_sent));
>   EXPECT_EQ(stats.rtcp_packet_type_counts.pli_packets,
>-            info.receivers[0].plis_sent);
>+            rtc::checked_cast<unsigned int>(info.receivers[0].plis_sent));
> }
> 
> TEST_F(WebRtcVideoChannelTest, GetStatsTranslatesDecodeStatsCorrectly) {
>@@ -3890,7 +4764,8 @@ TEST_F(WebRtcVideoChannelTest, GetStatsTranslatesDecodeStatsCorrectly) {
>   EXPECT_EQ(stats.render_delay_ms, info.receivers[0].render_delay_ms);
>   EXPECT_EQ(stats.width, info.receivers[0].frame_width);
>   EXPECT_EQ(stats.height, info.receivers[0].frame_height);
>-  EXPECT_EQ(stats.frame_counts.key_frames + stats.frame_counts.delta_frames,
>+  EXPECT_EQ(rtc::checked_cast<unsigned int>(stats.frame_counts.key_frames +
>+                                            stats.frame_counts.delta_frames),
>             info.receivers[0].frames_received);
>   EXPECT_EQ(stats.frames_rendered, info.receivers[0].frames_rendered);
>   EXPECT_EQ(stats.frames_decoded, info.receivers[0].frames_decoded);
>@@ -3913,11 +4788,11 @@ TEST_F(WebRtcVideoChannelTest, GetStatsTranslatesReceivePacketStatsCorrectly) {
>   EXPECT_EQ(stats.rtp_stats.transmitted.payload_bytes +
>                 stats.rtp_stats.transmitted.header_bytes +
>                 stats.rtp_stats.transmitted.padding_bytes,
>-            info.receivers[0].bytes_rcvd);
>+            rtc::checked_cast<size_t>(info.receivers[0].bytes_rcvd));
>   EXPECT_EQ(stats.rtp_stats.transmitted.packets,
>-            info.receivers[0].packets_rcvd);
>+            rtc::checked_cast<unsigned int>(info.receivers[0].packets_rcvd));
>   EXPECT_EQ(stats.rtcp_stats.packets_lost, info.receivers[0].packets_lost);
>-  EXPECT_EQ(static_cast<float>(stats.rtcp_stats.fraction_lost) / (1 << 8),
>+  EXPECT_EQ(rtc::checked_cast<float>(stats.rtcp_stats.fraction_lost) / (1 << 8),
>             info.receivers[0].fraction_lost);
> }
> 
>@@ -4006,7 +4881,7 @@ TEST_F(WebRtcVideoChannelTest, DefaultReceiveStreamReconfiguresToUseRtx) {
>   EXPECT_TRUE(VerifyRtxReceiveAssociations(recv_stream->GetConfig()))
>       << "RTX should be mapped for all decoders/payload types.";
>   EXPECT_TRUE(HasRtxReceiveAssociation(recv_stream->GetConfig(),
>-                                          GetEngineCodec("red").id))
>+                                       GetEngineCodec("red").id))
>       << "RTX should be mapped also for the RED payload type";
>   EXPECT_EQ(rtx_ssrcs[0], recv_stream->GetConfig().rtp.rtx_ssrc);
> }
>@@ -4128,6 +5003,44 @@ TEST_F(WebRtcVideoChannelTest, MapsReceivedPayloadTypeToCodecName) {
>   EXPECT_STREQ("", info.receivers[0].codec_name.c_str());
> }
> 
>+// Tests that when we add a stream without SSRCs, but contains a stream_id
>+// that it is stored and its stream id is later used when the first packet
>+// arrives to properly create a receive stream with a sync label.
>+TEST_F(WebRtcVideoChannelTest, RecvUnsignaledSsrcWithSignaledStreamId) {
>+  const char kSyncLabel[] = "sync_label";
>+  cricket::StreamParams unsignaled_stream;
>+  unsignaled_stream.set_stream_ids({kSyncLabel});
>+  ASSERT_TRUE(channel_->AddRecvStream(unsignaled_stream));
>+  // The stream shouldn't have been created at this point because it doesn't
>+  // have any SSRCs.
>+  EXPECT_EQ(0u, fake_call_->GetVideoReceiveStreams().size());
>+
>+  // Create and deliver packet.
>+  const size_t kDataLength = 12;
>+  uint8_t data[kDataLength];
>+  memset(data, 0, sizeof(data));
>+  rtc::SetBE32(&data[8], kIncomingUnsignalledSsrc);
>+  rtc::CopyOnWriteBuffer packet(data, kDataLength);
>+  rtc::PacketTime packet_time;
>+  channel_->OnPacketReceived(&packet, packet_time);
>+
>+  // The stream should now be created with the appropriate sync label.
>+  EXPECT_EQ(1u, fake_call_->GetVideoReceiveStreams().size());
>+  EXPECT_EQ(kSyncLabel,
>+            fake_call_->GetVideoReceiveStreams()[0]->GetConfig().sync_group);
>+
>+  // Removing the unsignaled stream should clear the cache. This time when
>+  // a default video receive stream is created it won't have a sync_group.
>+  ASSERT_TRUE(channel_->RemoveRecvStream(0));
>+  ASSERT_TRUE(channel_->RemoveRecvStream(kIncomingUnsignalledSsrc));
>+  EXPECT_EQ(0u, fake_call_->GetVideoReceiveStreams().size());
>+
>+  channel_->OnPacketReceived(&packet, packet_time);
>+  EXPECT_EQ(1u, fake_call_->GetVideoReceiveStreams().size());
>+  EXPECT_TRUE(
>+      fake_call_->GetVideoReceiveStreams()[0]->GetConfig().sync_group.empty());
>+}
>+
> void WebRtcVideoChannelTest::TestReceiveUnsignaledSsrcPacket(
>     uint8_t payload_type,
>     bool expect_created_receive_stream) {
>@@ -4173,6 +5086,7 @@ TEST_F(WebRtcVideoChannelTest, Vp9PacketCreatesUnsignalledStream) {
> }
> 
> TEST_F(WebRtcVideoChannelTest, RtxPacketDoesntCreateUnsignalledStream) {
>+  AssignDefaultAptRtxTypes();
>   const cricket::VideoCodec vp8 = GetEngineCodec("VP8");
>   const int rtx_vp8_payload_type = default_apt_rtx_types_[vp8.id];
>   TestReceiveUnsignaledSsrcPacket(rtx_vp8_payload_type,
>@@ -4220,15 +5134,14 @@ TEST_F(WebRtcVideoChannelTest, ReceiveDifferentUnsignaledSsrc) {
>   cricket::RtpHeader rtpHeader;
>   rtpHeader.payload_type = GetEngineCodec("VP8").id;
>   rtpHeader.seq_num = rtpHeader.timestamp = 0;
>-  rtpHeader.ssrc = kIncomingUnsignalledSsrc+1;
>+  rtpHeader.ssrc = kIncomingUnsignalledSsrc + 1;
>   cricket::SetRtpHeader(data, sizeof(data), rtpHeader);
>   rtc::CopyOnWriteBuffer packet(data, sizeof(data));
>   rtc::PacketTime packet_time;
>   channel_->OnPacketReceived(&packet, packet_time);
>   // VP8 packet should create default receive stream.
>   ASSERT_EQ(1u, fake_call_->GetVideoReceiveStreams().size());
>-  FakeVideoReceiveStream* recv_stream =
>-    fake_call_->GetVideoReceiveStreams()[0];
>+  FakeVideoReceiveStream* recv_stream = fake_call_->GetVideoReceiveStreams()[0];
>   EXPECT_EQ(rtpHeader.ssrc, recv_stream->GetConfig().rtp.remote_ssrc);
>   // Verify that the receive stream sinks to a renderer.
>   webrtc::VideoFrame video_frame(CreateBlackFrameBuffer(4, 4), 100, 0,
>@@ -4238,7 +5151,7 @@ TEST_F(WebRtcVideoChannelTest, ReceiveDifferentUnsignaledSsrc) {
> 
>   // Receive VP9 packet on second SSRC.
>   rtpHeader.payload_type = GetEngineCodec("VP9").id;
>-  rtpHeader.ssrc = kIncomingUnsignalledSsrc+2;
>+  rtpHeader.ssrc = kIncomingUnsignalledSsrc + 2;
>   cricket::SetRtpHeader(data, sizeof(data), rtpHeader);
>   rtc::CopyOnWriteBuffer packet2(data, sizeof(data));
>   channel_->OnPacketReceived(&packet2, packet_time);
>@@ -4255,7 +5168,7 @@ TEST_F(WebRtcVideoChannelTest, ReceiveDifferentUnsignaledSsrc) {
> #if defined(WEBRTC_USE_H264)
>   // Receive H264 packet on third SSRC.
>   rtpHeader.payload_type = 126;
>-  rtpHeader.ssrc = kIncomingUnsignalledSsrc+3;
>+  rtpHeader.ssrc = kIncomingUnsignalledSsrc + 3;
>   cricket::SetRtpHeader(data, sizeof(data), rtpHeader);
>   rtc::CopyOnWriteBuffer packet3(data, sizeof(data));
>   channel_->OnPacketReceived(&packet3, packet_time);
>@@ -4320,11 +5233,11 @@ TEST_F(WebRtcVideoChannelTest,
>   EXPECT_EQ(kSsrcs3[1], recv_stream1->GetConfig().rtp.remote_ssrc);
> }
> 
>-TEST_F(WebRtcVideoChannelTest, CanSentMaxBitrateForExistingStream) {
>+TEST_F(WebRtcVideoChannelTest, CanSetMaxBitrateForExistingStream) {
>   AddSendStream();
> 
>-  cricket::FakeVideoCapturer capturer;
>-  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, true, nullptr, &capturer));
>+  FakeVideoCapturerWithTaskQueue capturer;
>+  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, &capturer));
>   cricket::VideoFormat capture_format_hd =
>       capturer.GetSupportedFormats()->front();
>   EXPECT_EQ(1280, capture_format_hd.width);
>@@ -4349,34 +5262,119 @@ TEST_F(WebRtcVideoChannelTest, CanSentMaxBitrateForExistingStream) {
>   SetAndExpectMaxBitrate(0, 800, 800);
>   SetAndExpectMaxBitrate(0, 0, default_encoder_bitrate);
> 
>-  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, true, nullptr, nullptr));
>+  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
> }
> 
> TEST_F(WebRtcVideoChannelTest, CannotSetMaxBitrateForNonexistentStream) {
>   webrtc::RtpParameters nonexistent_parameters =
>       channel_->GetRtpSendParameters(last_ssrc_);
>-  EXPECT_EQ(0, nonexistent_parameters.encodings.size());
>+  EXPECT_EQ(0u, nonexistent_parameters.encodings.size());
> 
>   nonexistent_parameters.encodings.push_back(webrtc::RtpEncodingParameters());
>   EXPECT_FALSE(
>-      channel_->SetRtpSendParameters(last_ssrc_, nonexistent_parameters));
>+      channel_->SetRtpSendParameters(last_ssrc_, nonexistent_parameters).ok());
> }
> 
> TEST_F(WebRtcVideoChannelTest,
>-       CannotSetRtpSendParametersWithIncorrectNumberOfEncodings) {
>-  // This test verifies that setting RtpParameters succeeds only if
>-  // the structure contains exactly one encoding.
>-  // TODO(skvlad): Update this test when we start supporting setting parameters
>-  // for each encoding individually.
>+       SetLowMaxBitrateOverwritesVideoStreamMinBitrate) {
>+  FakeVideoSendStream* stream = AddSendStream();
>+
>+  webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
>+  EXPECT_EQ(1UL, parameters.encodings.size());
>+  EXPECT_FALSE(parameters.encodings[0].max_bitrate_bps.has_value());
>+  EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
>+
>+  // Note that this is testing the behavior of the FakeVideoSendStream, which
>+  // also calls to CreateEncoderStreams to get the VideoStreams, so essentially
>+  // we are just testing the behavior of
>+  // EncoderStreamFactory::CreateEncoderStreams.
>+  ASSERT_EQ(1UL, stream->GetVideoStreams().size());
>+  EXPECT_EQ(kMinVideoBitrateBps, stream->GetVideoStreams()[0].min_bitrate_bps);
>+
>+  // Set a low max bitrate & check that VideoStream.min_bitrate_bps is limited
>+  // by this amount.
>+  parameters = channel_->GetRtpSendParameters(last_ssrc_);
>+  int low_max_bitrate_bps = kMinVideoBitrateBps - 1000;
>+  parameters.encodings[0].max_bitrate_bps = low_max_bitrate_bps;
>+  EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
>+
>+  ASSERT_EQ(1UL, stream->GetVideoStreams().size());
>+  EXPECT_EQ(low_max_bitrate_bps, stream->GetVideoStreams()[0].min_bitrate_bps);
>+  EXPECT_EQ(low_max_bitrate_bps, stream->GetVideoStreams()[0].max_bitrate_bps);
>+}
>+
>+TEST_F(WebRtcVideoChannelTest,
>+       SetHighMinBitrateOverwritesVideoStreamMaxBitrate) {
>+  FakeVideoSendStream* stream = AddSendStream();
>+
>+  // Note that this is testing the behavior of the FakeVideoSendStream, which
>+  // also calls to CreateEncoderStreams to get the VideoStreams, so essentially
>+  // we are just testing the behavior of
>+  // EncoderStreamFactory::CreateEncoderStreams.
>+  ASSERT_EQ(1UL, stream->GetVideoStreams().size());
>+  int high_min_bitrate_bps = stream->GetVideoStreams()[0].max_bitrate_bps + 1;
>+
>+  // Set a high min bitrate and check that max_bitrate_bps is adjusted up.
>+  webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
>+  EXPECT_EQ(1UL, parameters.encodings.size());
>+  parameters.encodings[0].min_bitrate_bps = high_min_bitrate_bps;
>+  EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
>+
>+  ASSERT_EQ(1UL, stream->GetVideoStreams().size());
>+  EXPECT_EQ(high_min_bitrate_bps, stream->GetVideoStreams()[0].min_bitrate_bps);
>+  EXPECT_EQ(high_min_bitrate_bps, stream->GetVideoStreams()[0].max_bitrate_bps);
>+}
>+
>+TEST_F(WebRtcVideoChannelTest,
>+       SetMinBitrateAboveMaxBitrateLimitAdjustsMinBitrateDown) {
>+  send_parameters_.max_bandwidth_bps = 99999;
>+  FakeVideoSendStream* stream = AddSendStream();
>+  ExpectSetMaxBitrate(send_parameters_.max_bandwidth_bps);
>+  ASSERT_TRUE(channel_->SetSendParameters(send_parameters_));
>+  ASSERT_EQ(1UL, stream->GetVideoStreams().size());
>+  EXPECT_EQ(kMinVideoBitrateBps, stream->GetVideoStreams()[0].min_bitrate_bps);
>+  EXPECT_EQ(send_parameters_.max_bandwidth_bps,
>+            stream->GetVideoStreams()[0].max_bitrate_bps);
>+
>+  // Set min bitrate above global max bitrate and check that min_bitrate_bps is
>+  // adjusted down.
>+  webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
>+  EXPECT_EQ(1UL, parameters.encodings.size());
>+  parameters.encodings[0].min_bitrate_bps = 99999 + 1;
>+  EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
>+  ASSERT_EQ(1UL, stream->GetVideoStreams().size());
>+  EXPECT_EQ(send_parameters_.max_bandwidth_bps,
>+            stream->GetVideoStreams()[0].min_bitrate_bps);
>+  EXPECT_EQ(send_parameters_.max_bandwidth_bps,
>+            stream->GetVideoStreams()[0].max_bitrate_bps);
>+}
> 
>+TEST_F(WebRtcVideoChannelTest,
>+       CannotSetRtpSendParametersWithIncorrectNumberOfEncodings) {
>   AddSendStream();
>   webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
>   // Two or more encodings should result in failure.
>   parameters.encodings.push_back(webrtc::RtpEncodingParameters());
>-  EXPECT_FALSE(channel_->SetRtpSendParameters(last_ssrc_, parameters));
>+  EXPECT_FALSE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
>+  // Zero encodings should also fail.
>+  parameters.encodings.clear();
>+  EXPECT_FALSE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
>+}
>+
>+TEST_F(WebRtcVideoChannelTest,
>+       CannotSetSimulcastRtpSendParametersWithIncorrectNumberOfEncodings) {
>+  std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs3);
>+  StreamParams sp = CreateSimStreamParams("cname", ssrcs);
>+  AddSendStream(sp);
>+
>+  webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
>+
>+  // Additional encodings should result in failure.
>+  parameters.encodings.push_back(webrtc::RtpEncodingParameters());
>+  EXPECT_FALSE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
>   // Zero encodings should also fail.
>   parameters.encodings.clear();
>-  EXPECT_FALSE(channel_->SetRtpSendParameters(last_ssrc_, parameters));
>+  EXPECT_FALSE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
> }
> 
> // Changing the SSRC through RtpParameters is not allowed.
>@@ -4384,7 +5382,7 @@ TEST_F(WebRtcVideoChannelTest, CannotSetSsrcInRtpSendParameters) {
>   AddSendStream();
>   webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
>   parameters.encodings[0].ssrc = 0xdeadbeef;
>-  EXPECT_FALSE(channel_->SetRtpSendParameters(last_ssrc_, parameters));
>+  EXPECT_FALSE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
> }
> 
> // Tests that when RTCRtpEncodingParameters.bitrate_priority gets set to
>@@ -4397,9 +5395,9 @@ TEST_F(WebRtcVideoChannelTest, SetRtpSendParametersInvalidBitratePriority) {
>             parameters.encodings[0].bitrate_priority);
> 
>   parameters.encodings[0].bitrate_priority = 0;
>-  EXPECT_FALSE(channel_->SetRtpSendParameters(last_ssrc_, parameters));
>+  EXPECT_FALSE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
>   parameters.encodings[0].bitrate_priority = -2;
>-  EXPECT_FALSE(channel_->SetRtpSendParameters(last_ssrc_, parameters));
>+  EXPECT_FALSE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
> }
> 
> // Tests when the the RTCRtpEncodingParameters.bitrate_priority gets set
>@@ -4414,7 +5412,7 @@ TEST_F(WebRtcVideoChannelTest, SetRtpSendParametersPriorityOneStream) {
>   // Change the value and set it on the VideoChannel.
>   double new_bitrate_priority = 2.0;
>   parameters.encodings[0].bitrate_priority = new_bitrate_priority;
>-  EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters));
>+  EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
> 
>   // Verify that the encoding parameters bitrate_priority is set for the
>   // VideoChannel.
>@@ -4434,7 +5432,7 @@ TEST_F(WebRtcVideoChannelTest, SetRtpSendParametersPriorityOneStream) {
>   // Check that the vector of VideoStreams also was propagated correctly. Note
>   // that this is testing the behavior of the FakeVideoSendStream, which mimics
>   // the calls to CreateEncoderStreams to get the VideoStreams.
>-  EXPECT_EQ(rtc::Optional<double>(new_bitrate_priority),
>+  EXPECT_EQ(absl::optional<double>(new_bitrate_priority),
>             video_send_stream->GetVideoStreams()[0].bitrate_priority);
> }
> 
>@@ -4443,18 +5441,18 @@ TEST_F(WebRtcVideoChannelTest, SetRtpSendParametersPriorityOneStream) {
> // streams.
> TEST_F(WebRtcVideoChannelTest, SetRtpSendParametersPrioritySimulcastStreams) {
>   // Create the stream params with multiple ssrcs for simulcast.
>-  const int kNumSimulcastStreams = 3;
>+  const size_t kNumSimulcastStreams = 3;
>   std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs3);
>   StreamParams stream_params = CreateSimStreamParams("cname", ssrcs);
>   AddSendStream(stream_params);
>   uint32_t primary_ssrc = stream_params.first_ssrc();
> 
>-  // Using the FakeVideoCapturer, we manually send a full size frame. This
>-  // creates multiple VideoStreams for all simulcast layers when reconfiguring,
>-  // and allows us to test this behavior.
>-  cricket::FakeVideoCapturer capturer;
>+  // Using the FakeVideoCapturerWithTaskQueue, we manually send a full size
>+  // frame. This creates multiple VideoStreams for all simulcast layers when
>+  // reconfiguring, and allows us to test this behavior.
>+  FakeVideoCapturerWithTaskQueue capturer;
>   VideoOptions options;
>-  EXPECT_TRUE(channel_->SetVideoSend(primary_ssrc, true, &options, &capturer));
>+  EXPECT_TRUE(channel_->SetVideoSend(primary_ssrc, &options, &capturer));
>   EXPECT_EQ(cricket::CS_RUNNING,
>             capturer.Start(cricket::VideoFormat(
>                 1920, 1080, cricket::VideoFormat::FpsToInterval(30),
>@@ -4464,17 +5462,17 @@ TEST_F(WebRtcVideoChannelTest, SetRtpSendParametersPrioritySimulcastStreams) {
>   // Get and set the rtp encoding parameters.
>   webrtc::RtpParameters parameters =
>       channel_->GetRtpSendParameters(primary_ssrc);
>-  EXPECT_EQ(1UL, parameters.encodings.size());
>+  EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
>   EXPECT_EQ(webrtc::kDefaultBitratePriority,
>             parameters.encodings[0].bitrate_priority);
>   // Change the value and set it on the VideoChannel.
>   double new_bitrate_priority = 2.0;
>   parameters.encodings[0].bitrate_priority = new_bitrate_priority;
>-  EXPECT_TRUE(channel_->SetRtpSendParameters(primary_ssrc, parameters));
>+  EXPECT_TRUE(channel_->SetRtpSendParameters(primary_ssrc, parameters).ok());
> 
>   // Verify that the encoding parameters priority is set on the VideoChannel.
>   parameters = channel_->GetRtpSendParameters(primary_ssrc);
>-  EXPECT_EQ(1UL, parameters.encodings.size());
>+  EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
>   EXPECT_EQ(new_bitrate_priority, parameters.encodings[0].bitrate_priority);
> 
>   // Verify that the new value propagated down to the encoder.
>@@ -4492,22 +5490,394 @@ TEST_F(WebRtcVideoChannelTest, SetRtpSendParametersPrioritySimulcastStreams) {
>   // FakeVideoSendStream calls CreateEncoderStreams, and we are testing that
>   // these are created appropriately for the simulcast case.
>   EXPECT_EQ(kNumSimulcastStreams, video_send_stream->GetVideoStreams().size());
>-  EXPECT_EQ(rtc::Optional<double>(new_bitrate_priority),
>+  EXPECT_EQ(absl::optional<double>(new_bitrate_priority),
>             video_send_stream->GetVideoStreams()[0].bitrate_priority);
>   // Since we are only setting bitrate priority per-sender, the other
>   // VideoStreams should have a bitrate priority of 0.
>-  EXPECT_EQ(rtc::nullopt,
>+  EXPECT_EQ(absl::nullopt,
>             video_send_stream->GetVideoStreams()[1].bitrate_priority);
>-  EXPECT_EQ(rtc::nullopt,
>+  EXPECT_EQ(absl::nullopt,
>             video_send_stream->GetVideoStreams()[2].bitrate_priority);
>-  EXPECT_TRUE(channel_->SetVideoSend(primary_ssrc, true, nullptr, nullptr));
>+  EXPECT_TRUE(channel_->SetVideoSend(primary_ssrc, nullptr, nullptr));
>+}
>+
>+TEST_F(WebRtcVideoChannelTest, GetAndSetRtpSendParametersMinAndMaxBitrate) {
>+  const size_t kNumSimulcastStreams = 3;
>+  SetUpSimulcast(true, false);
>+
>+  // Get and set the rtp encoding parameters.
>+  webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
>+  EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
>+  for (const auto& encoding : parameters.encodings) {
>+    EXPECT_FALSE(encoding.min_bitrate_bps);
>+    EXPECT_FALSE(encoding.max_bitrate_bps);
>+  }
>+
>+  // Change the value and set it on the VideoChannel.
>+  parameters.encodings[0].min_bitrate_bps = 100000;
>+  parameters.encodings[0].max_bitrate_bps = 200000;
>+  parameters.encodings[1].min_bitrate_bps = 300000;
>+  parameters.encodings[1].max_bitrate_bps = 400000;
>+  parameters.encodings[2].min_bitrate_bps = 500000;
>+  parameters.encodings[2].max_bitrate_bps = 600000;
>+  EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
>+
>+  // Verify that the bitrates are set on the VideoChannel.
>+  parameters = channel_->GetRtpSendParameters(last_ssrc_);
>+  EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
>+  EXPECT_EQ(100000, parameters.encodings[0].min_bitrate_bps);
>+  EXPECT_EQ(200000, parameters.encodings[0].max_bitrate_bps);
>+  EXPECT_EQ(300000, parameters.encodings[1].min_bitrate_bps);
>+  EXPECT_EQ(400000, parameters.encodings[1].max_bitrate_bps);
>+  EXPECT_EQ(500000, parameters.encodings[2].min_bitrate_bps);
>+  EXPECT_EQ(600000, parameters.encodings[2].max_bitrate_bps);
>+}
>+
>+TEST_F(WebRtcVideoChannelTest, SetRtpSendParametersFailsWithIncorrectBitrate) {
>+  const size_t kNumSimulcastStreams = 3;
>+  SetUpSimulcast(true, false);
>+
>+  // Get and set the rtp encoding parameters.
>+  webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
>+  EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
>+
>+  // Max bitrate lower than min bitrate should fail.
>+  parameters.encodings[2].min_bitrate_bps = 100000;
>+  parameters.encodings[2].max_bitrate_bps = 100000 - 1;
>+  EXPECT_EQ(webrtc::RTCErrorType::INVALID_RANGE,
>+            channel_->SetRtpSendParameters(last_ssrc_, parameters).type());
>+}
>+
>+// Test that min and max bitrate values set via RtpParameters are correctly
>+// propagated to the underlying encoder, and that the target is set to 3/4 of
>+// the maximum (3/4 was chosen because it's similar to the simulcast defaults
>+// that are used if no min/max are specified).
>+TEST_F(WebRtcVideoChannelTest, MinAndMaxSimulcastBitratePropagatedToEncoder) {
>+  const size_t kNumSimulcastStreams = 3;
>+  FakeVideoSendStream* stream = SetUpSimulcast(true, false);
>+
>+  // Send a full size frame so all simulcast layers are used when reconfiguring.
>+  FakeVideoCapturerWithTaskQueue capturer;
>+  VideoOptions options;
>+  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &capturer));
>+  EXPECT_EQ(cricket::CS_RUNNING,
>+            capturer.Start(capturer.GetSupportedFormats()->front()));
>+  channel_->SetSend(true);
>+  EXPECT_TRUE(capturer.CaptureFrame());
>+
>+  // Get and set the rtp encoding parameters.
>+  // Change the value and set it on the VideoChannel.
>+  webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
>+  EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
>+  parameters.encodings[0].min_bitrate_bps = 100000;
>+  parameters.encodings[0].max_bitrate_bps = 200000;
>+  parameters.encodings[1].min_bitrate_bps = 300000;
>+  parameters.encodings[1].max_bitrate_bps = 400000;
>+  parameters.encodings[2].min_bitrate_bps = 500000;
>+  parameters.encodings[2].max_bitrate_bps = 600000;
>+  EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
>+
>+  // Verify that the new value propagated down to the encoder.
>+  // Check that WebRtcVideoSendStream updates VideoEncoderConfig correctly.
>+  EXPECT_EQ(2, stream->num_encoder_reconfigurations());
>+  webrtc::VideoEncoderConfig encoder_config = stream->GetEncoderConfig().Copy();
>+  EXPECT_EQ(kNumSimulcastStreams, encoder_config.number_of_streams);
>+  EXPECT_EQ(kNumSimulcastStreams, encoder_config.simulcast_layers.size());
>+  EXPECT_EQ(100000, encoder_config.simulcast_layers[0].min_bitrate_bps);
>+  EXPECT_EQ(200000, encoder_config.simulcast_layers[0].max_bitrate_bps);
>+  EXPECT_EQ(300000, encoder_config.simulcast_layers[1].min_bitrate_bps);
>+  EXPECT_EQ(400000, encoder_config.simulcast_layers[1].max_bitrate_bps);
>+  EXPECT_EQ(500000, encoder_config.simulcast_layers[2].min_bitrate_bps);
>+  EXPECT_EQ(600000, encoder_config.simulcast_layers[2].max_bitrate_bps);
>+
>+  // FakeVideoSendStream calls CreateEncoderStreams, test that the vector of
>+  // VideoStreams are created appropriately for the simulcast case.
>+  EXPECT_EQ(kNumSimulcastStreams, stream->GetVideoStreams().size());
>+  // Target bitrate: 200000 * 3 / 4 = 150000.
>+  EXPECT_EQ(100000, stream->GetVideoStreams()[0].min_bitrate_bps);
>+  EXPECT_EQ(150000, stream->GetVideoStreams()[0].target_bitrate_bps);
>+  EXPECT_EQ(200000, stream->GetVideoStreams()[0].max_bitrate_bps);
>+  // Target bitrate: 400000 * 3 / 4 = 300000.
>+  EXPECT_EQ(300000, stream->GetVideoStreams()[1].min_bitrate_bps);
>+  EXPECT_EQ(300000, stream->GetVideoStreams()[1].target_bitrate_bps);
>+  EXPECT_EQ(400000, stream->GetVideoStreams()[1].max_bitrate_bps);
>+  // Target bitrate: 600000 * 3 / 4 = 450000, less than min -> max.
>+  EXPECT_EQ(500000, stream->GetVideoStreams()[2].min_bitrate_bps);
>+  EXPECT_EQ(600000, stream->GetVideoStreams()[2].target_bitrate_bps);
>+  EXPECT_EQ(600000, stream->GetVideoStreams()[2].max_bitrate_bps);
>+
>+  // No parameter changed, encoder should not be reconfigured.
>+  EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
>+  EXPECT_EQ(2, stream->num_encoder_reconfigurations());
>+
>+  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
>+}
>+
>+// Test to only specify the min or max bitrate value for a layer via
>+// RtpParameters. The unspecified min/max and target value should be set to the
>+// simulcast default that is used if no min/max are specified.
>+TEST_F(WebRtcVideoChannelTest, MinOrMaxSimulcastBitratePropagatedToEncoder) {
>+  const size_t kNumSimulcastStreams = 3;
>+  const std::vector<webrtc::VideoStream> kDefault = GetSimulcastBitrates720p();
>+  FakeVideoSendStream* stream = SetUpSimulcast(true, false);
>+
>+  // Send a full size frame so all simulcast layers are used when reconfiguring.
>+  FakeVideoCapturerWithTaskQueue capturer;
>+  VideoOptions options;
>+  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &capturer));
>+  EXPECT_EQ(cricket::CS_RUNNING,
>+            capturer.Start(capturer.GetSupportedFormats()->front()));
>+  channel_->SetSend(true);
>+  EXPECT_TRUE(capturer.CaptureFrame());
>+
>+  // Get and set the rtp encoding parameters.
>+  webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
>+  EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
>+
>+  // Change the value and set it on the VideoChannel.
>+  // Layer 0: only configure min bitrate.
>+  const int kMinBpsLayer0 = kDefault[0].min_bitrate_bps + 1;
>+  parameters.encodings[0].min_bitrate_bps = kMinBpsLayer0;
>+  // Layer 1: only configure max bitrate.
>+  const int kMaxBpsLayer1 = kDefault[1].max_bitrate_bps - 1;
>+  parameters.encodings[1].max_bitrate_bps = kMaxBpsLayer1;
>+  EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
>+
>+  // Verify that the new value propagated down to the encoder.
>+  // Check that WebRtcVideoSendStream updates VideoEncoderConfig correctly.
>+  webrtc::VideoEncoderConfig encoder_config = stream->GetEncoderConfig().Copy();
>+  EXPECT_EQ(kNumSimulcastStreams, encoder_config.number_of_streams);
>+  EXPECT_EQ(kNumSimulcastStreams, encoder_config.simulcast_layers.size());
>+  EXPECT_EQ(kMinBpsLayer0, encoder_config.simulcast_layers[0].min_bitrate_bps);
>+  EXPECT_EQ(-1, encoder_config.simulcast_layers[0].max_bitrate_bps);
>+  EXPECT_EQ(-1, encoder_config.simulcast_layers[1].min_bitrate_bps);
>+  EXPECT_EQ(kMaxBpsLayer1, encoder_config.simulcast_layers[1].max_bitrate_bps);
>+  EXPECT_EQ(-1, encoder_config.simulcast_layers[2].min_bitrate_bps);
>+  EXPECT_EQ(-1, encoder_config.simulcast_layers[2].max_bitrate_bps);
>+
>+  // FakeVideoSendStream calls CreateEncoderStreams, test that the vector of
>+  // VideoStreams are created appropriately for the simulcast case.
>+  EXPECT_EQ(kNumSimulcastStreams, stream->GetVideoStreams().size());
>+  // Layer 0: min configured bitrate should overwrite min default.
>+  EXPECT_EQ(kMinBpsLayer0, stream->GetVideoStreams()[0].min_bitrate_bps);
>+  EXPECT_EQ(kDefault[0].target_bitrate_bps,
>+            stream->GetVideoStreams()[0].target_bitrate_bps);
>+  EXPECT_EQ(kDefault[0].max_bitrate_bps,
>+            stream->GetVideoStreams()[0].max_bitrate_bps);
>+  // Layer 1: max configured bitrate should overwrite max default.
>+  EXPECT_EQ(kDefault[1].min_bitrate_bps,
>+            stream->GetVideoStreams()[1].min_bitrate_bps);
>+  EXPECT_EQ(kDefault[1].target_bitrate_bps,
>+            stream->GetVideoStreams()[1].target_bitrate_bps);
>+  EXPECT_EQ(kMaxBpsLayer1, stream->GetVideoStreams()[1].max_bitrate_bps);
>+  // Layer 2: min and max bitrate not configured, default expected.
>+  EXPECT_EQ(kDefault[2].min_bitrate_bps,
>+            stream->GetVideoStreams()[2].min_bitrate_bps);
>+  EXPECT_EQ(kDefault[2].target_bitrate_bps,
>+            stream->GetVideoStreams()[2].target_bitrate_bps);
>+  EXPECT_EQ(kDefault[2].max_bitrate_bps,
>+            stream->GetVideoStreams()[2].max_bitrate_bps);
>+
>+  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
>+}
>+
>+// Test that specifying the min (or max) bitrate value for a layer via
>+// RtpParameters above (or below) the simulcast default max (or min) adjusts the
>+// unspecified values accordingly.
>+TEST_F(WebRtcVideoChannelTest, SetMinAndMaxSimulcastBitrateAboveBelowDefault) {
>+  const size_t kNumSimulcastStreams = 3;
>+  const std::vector<webrtc::VideoStream> kDefault = GetSimulcastBitrates720p();
>+  FakeVideoSendStream* stream = SetUpSimulcast(true, false);
>+
>+  // Send a full size frame so all simulcast layers are used when reconfiguring.
>+  FakeVideoCapturerWithTaskQueue capturer;
>+  VideoOptions options;
>+  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &capturer));
>+  EXPECT_EQ(cricket::CS_RUNNING,
>+            capturer.Start(capturer.GetSupportedFormats()->front()));
>+  channel_->SetSend(true);
>+  EXPECT_TRUE(capturer.CaptureFrame());
>+
>+  // Get and set the rtp encoding parameters.
>+  webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
>+  EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
>+
>+  // Change the value and set it on the VideoChannel.
>+  // For layer 0, set the min bitrate above the default max.
>+  const int kMinBpsLayer0 = kDefault[0].max_bitrate_bps + 1;
>+  parameters.encodings[0].min_bitrate_bps = kMinBpsLayer0;
>+  // For layer 1, set the max bitrate below the default min.
>+  const int kMaxBpsLayer1 = kDefault[1].min_bitrate_bps - 1;
>+  parameters.encodings[1].max_bitrate_bps = kMaxBpsLayer1;
>+  EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
>+
>+  // Verify that the new value propagated down to the encoder.
>+  // FakeVideoSendStream calls CreateEncoderStreams, test that the vector of
>+  // VideoStreams are created appropriately for the simulcast case.
>+  EXPECT_EQ(kNumSimulcastStreams, stream->GetVideoStreams().size());
>+  // Layer 0: Min bitrate above default max (target/max should be adjusted).
>+  EXPECT_EQ(kMinBpsLayer0, stream->GetVideoStreams()[0].min_bitrate_bps);
>+  EXPECT_EQ(kMinBpsLayer0, stream->GetVideoStreams()[0].target_bitrate_bps);
>+  EXPECT_EQ(kMinBpsLayer0, stream->GetVideoStreams()[0].max_bitrate_bps);
>+  // Layer 1: Max bitrate below default min (min/target should be adjusted).
>+  EXPECT_EQ(kMaxBpsLayer1, stream->GetVideoStreams()[1].min_bitrate_bps);
>+  EXPECT_EQ(kMaxBpsLayer1, stream->GetVideoStreams()[1].target_bitrate_bps);
>+  EXPECT_EQ(kMaxBpsLayer1, stream->GetVideoStreams()[1].max_bitrate_bps);
>+  // Layer 2: min and max bitrate not configured, default expected.
>+  EXPECT_EQ(kDefault[2].min_bitrate_bps,
>+            stream->GetVideoStreams()[2].min_bitrate_bps);
>+  EXPECT_EQ(kDefault[2].target_bitrate_bps,
>+            stream->GetVideoStreams()[2].target_bitrate_bps);
>+  EXPECT_EQ(kDefault[2].max_bitrate_bps,
>+            stream->GetVideoStreams()[2].max_bitrate_bps);
>+
>+  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
>+}
>+
>+TEST_F(WebRtcVideoChannelTest, BandwidthAboveTotalMaxBitrateGivenToMaxLayer) {
>+  const size_t kNumSimulcastStreams = 3;
>+  const std::vector<webrtc::VideoStream> kDefault = GetSimulcastBitrates720p();
>+  FakeVideoSendStream* stream = SetUpSimulcast(true, false);
>+
>+  // Send a full size frame so all simulcast layers are used when reconfiguring.
>+  FakeVideoCapturerWithTaskQueue capturer;
>+  VideoOptions options;
>+  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &capturer));
>+  EXPECT_EQ(cricket::CS_RUNNING,
>+            capturer.Start(capturer.GetSupportedFormats()->front()));
>+  channel_->SetSend(true);
>+  EXPECT_TRUE(capturer.CaptureFrame());
>+
>+  // Set max bitrate for all but the highest layer.
>+  webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
>+  EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
>+  parameters.encodings[0].max_bitrate_bps = kDefault[0].max_bitrate_bps;
>+  parameters.encodings[1].max_bitrate_bps = kDefault[1].max_bitrate_bps;
>+  EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
>+
>+  // Set max bandwidth equal to total max bitrate.
>+  send_parameters_.max_bandwidth_bps =
>+      GetTotalMaxBitrateBps(stream->GetVideoStreams());
>+  ExpectSetMaxBitrate(send_parameters_.max_bandwidth_bps);
>+  ASSERT_TRUE(channel_->SetSendParameters(send_parameters_));
>+
>+  // No bitrate above the total max to give to the highest layer.
>+  EXPECT_EQ(kNumSimulcastStreams, stream->GetVideoStreams().size());
>+  EXPECT_EQ(kDefault[2].max_bitrate_bps,
>+            stream->GetVideoStreams()[2].max_bitrate_bps);
>+
>+  // Set max bandwidth above the total max bitrate.
>+  send_parameters_.max_bandwidth_bps =
>+      GetTotalMaxBitrateBps(stream->GetVideoStreams()) + 1;
>+  ExpectSetMaxBitrate(send_parameters_.max_bandwidth_bps);
>+  ASSERT_TRUE(channel_->SetSendParameters(send_parameters_));
>+
>+  // The highest layer has no max bitrate set -> the bitrate above the total
>+  // max should be given to the highest layer.
>+  EXPECT_EQ(kNumSimulcastStreams, stream->GetVideoStreams().size());
>+  EXPECT_EQ(send_parameters_.max_bandwidth_bps,
>+            GetTotalMaxBitrateBps(stream->GetVideoStreams()));
>+  EXPECT_EQ(kDefault[2].max_bitrate_bps + 1,
>+            stream->GetVideoStreams()[2].max_bitrate_bps);
>+
>+  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
>+}
>+
>+TEST_F(WebRtcVideoChannelTest,
>+       BandwidthAboveTotalMaxBitrateNotGivenToMaxLayerIfMaxBitrateSet) {
>+  const size_t kNumSimulcastStreams = 3;
>+  const std::vector<webrtc::VideoStream> kDefault = GetSimulcastBitrates720p();
>+  EXPECT_EQ(kNumSimulcastStreams, kDefault.size());
>+  FakeVideoSendStream* stream = SetUpSimulcast(true, false);
>+
>+  // Send a full size frame so all simulcast layers are used when reconfiguring.
>+  FakeVideoCapturerWithTaskQueue capturer;
>+  VideoOptions options;
>+  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, &options, &capturer));
>+  EXPECT_EQ(cricket::CS_RUNNING,
>+            capturer.Start(capturer.GetSupportedFormats()->front()));
>+  channel_->SetSend(true);
>+  EXPECT_TRUE(capturer.CaptureFrame());
>+
>+  // Set max bitrate for the highest layer.
>+  webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
>+  EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
>+  parameters.encodings[2].max_bitrate_bps = kDefault[2].max_bitrate_bps;
>+  EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
>+
>+  // Set max bandwidth above the total max bitrate.
>+  send_parameters_.max_bandwidth_bps =
>+      GetTotalMaxBitrateBps(stream->GetVideoStreams()) + 1;
>+  ExpectSetMaxBitrate(send_parameters_.max_bandwidth_bps);
>+  ASSERT_TRUE(channel_->SetSendParameters(send_parameters_));
>+
>+  // The highest layer has the max bitrate set -> the bitrate above the total
>+  // max should not be given to the highest layer.
>+  EXPECT_EQ(kNumSimulcastStreams, stream->GetVideoStreams().size());
>+  EXPECT_EQ(*parameters.encodings[2].max_bitrate_bps,
>+            stream->GetVideoStreams()[2].max_bitrate_bps);
>+
>+  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
>+}
>+
>+// Test that min and max bitrate values set via RtpParameters are correctly
>+// propagated to the underlying encoder for a single stream.
>+TEST_F(WebRtcVideoChannelTest, MinAndMaxBitratePropagatedToEncoder) {
>+  FakeVideoSendStream* stream = AddSendStream();
>+  EXPECT_TRUE(channel_->SetSend(true));
>+  EXPECT_TRUE(stream->IsSending());
>+
>+  // Set min and max bitrate.
>+  webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(last_ssrc_);
>+  EXPECT_EQ(1u, parameters.encodings.size());
>+  parameters.encodings[0].min_bitrate_bps = 80000;
>+  parameters.encodings[0].max_bitrate_bps = 150000;
>+  EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
>+
>+  // Check that WebRtcVideoSendStream updates VideoEncoderConfig correctly.
>+  webrtc::VideoEncoderConfig encoder_config = stream->GetEncoderConfig().Copy();
>+  EXPECT_EQ(1u, encoder_config.number_of_streams);
>+  EXPECT_EQ(1u, encoder_config.simulcast_layers.size());
>+  EXPECT_EQ(80000, encoder_config.simulcast_layers[0].min_bitrate_bps);
>+  EXPECT_EQ(150000, encoder_config.simulcast_layers[0].max_bitrate_bps);
>+
>+  // FakeVideoSendStream calls CreateEncoderStreams, test that the vector of
>+  // VideoStreams are created appropriately.
>+  EXPECT_EQ(1u, stream->GetVideoStreams().size());
>+  EXPECT_EQ(80000, stream->GetVideoStreams()[0].min_bitrate_bps);
>+  EXPECT_EQ(150000, stream->GetVideoStreams()[0].target_bitrate_bps);
>+  EXPECT_EQ(150000, stream->GetVideoStreams()[0].max_bitrate_bps);
>+}
>+
>+// Test the default min and max bitrate value are correctly propagated to the
>+// underlying encoder for a single stream (when the values are not set via
>+// RtpParameters).
>+TEST_F(WebRtcVideoChannelTest, DefaultMinAndMaxBitratePropagatedToEncoder) {
>+  FakeVideoSendStream* stream = AddSendStream();
>+  EXPECT_TRUE(channel_->SetSend(true));
>+  EXPECT_TRUE(stream->IsSending());
>+
>+  // Check that WebRtcVideoSendStream updates VideoEncoderConfig correctly.
>+  webrtc::VideoEncoderConfig encoder_config = stream->GetEncoderConfig().Copy();
>+  EXPECT_EQ(1u, encoder_config.number_of_streams);
>+  EXPECT_EQ(1u, encoder_config.simulcast_layers.size());
>+  EXPECT_EQ(-1, encoder_config.simulcast_layers[0].min_bitrate_bps);
>+  EXPECT_EQ(-1, encoder_config.simulcast_layers[0].max_bitrate_bps);
>+
>+  // FakeVideoSendStream calls CreateEncoderStreams, test that the vector of
>+  // VideoStreams are created appropriately.
>+  EXPECT_EQ(1u, stream->GetVideoStreams().size());
>+  EXPECT_EQ(cricket::kMinVideoBitrateBps,
>+            stream->GetVideoStreams()[0].min_bitrate_bps);
>+  EXPECT_GT(stream->GetVideoStreams()[0].max_bitrate_bps,
>+            stream->GetVideoStreams()[0].min_bitrate_bps);
>+  EXPECT_EQ(stream->GetVideoStreams()[0].max_bitrate_bps,
>+            stream->GetVideoStreams()[0].target_bitrate_bps);
> }
> 
> // Test that a stream will not be sending if its encoding is made inactive
> // through SetRtpSendParameters.
>-// TODO(deadbeef): Update this test when we start supporting setting parameters
>-// for each encoding individually.
>-TEST_F(WebRtcVideoChannelTest, SetRtpSendParametersEncodingsActive) {
>+TEST_F(WebRtcVideoChannelTest, SetRtpSendParametersOneEncodingActive) {
>   FakeVideoSendStream* stream = AddSendStream();
>   EXPECT_TRUE(channel_->SetSend(true));
>   EXPECT_TRUE(stream->IsSending());
>@@ -4517,15 +5887,91 @@ TEST_F(WebRtcVideoChannelTest, SetRtpSendParametersEncodingsActive) {
>   ASSERT_EQ(1u, parameters.encodings.size());
>   ASSERT_TRUE(parameters.encodings[0].active);
>   parameters.encodings[0].active = false;
>-  EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters));
>+  EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
>   EXPECT_FALSE(stream->IsSending());
> 
>   // Now change it back to active and verify we resume sending.
>   parameters.encodings[0].active = true;
>-  EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters));
>+  EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
>   EXPECT_TRUE(stream->IsSending());
> }
> 
>+// Tests that when active is updated for any simulcast layer then the send
>+// stream's sending state will be updated and it will be reconfigured with the
>+// new appropriate active simulcast streams.
>+TEST_F(WebRtcVideoChannelTest, SetRtpSendParametersMultipleEncodingsActive) {
>+  // Create the stream params with multiple ssrcs for simulcast.
>+  const size_t kNumSimulcastStreams = 3;
>+  std::vector<uint32_t> ssrcs = MAKE_VECTOR(kSsrcs3);
>+  StreamParams stream_params = CreateSimStreamParams("cname", ssrcs);
>+  FakeVideoSendStream* fake_video_send_stream = AddSendStream(stream_params);
>+  uint32_t primary_ssrc = stream_params.first_ssrc();
>+
>+  // Using the FakeVideoCapturerWithTaskQueue, we manually send a full size
>+  // frame. This allows us to test that ReconfigureEncoder is called
>+  // appropriately.
>+  cricket::FakeVideoCapturerWithTaskQueue capturer;
>+  VideoOptions options;
>+  EXPECT_TRUE(channel_->SetVideoSend(primary_ssrc, &options, &capturer));
>+  EXPECT_EQ(cricket::CS_RUNNING,
>+            capturer.Start(cricket::VideoFormat(
>+                1920, 1080, cricket::VideoFormat::FpsToInterval(30),
>+                cricket::FOURCC_I420)));
>+  channel_->SetSend(true);
>+  EXPECT_TRUE(capturer.CaptureFrame());
>+
>+  // Check that all encodings are initially active.
>+  webrtc::RtpParameters parameters =
>+      channel_->GetRtpSendParameters(primary_ssrc);
>+  EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
>+  EXPECT_TRUE(parameters.encodings[0].active);
>+  EXPECT_TRUE(parameters.encodings[1].active);
>+  EXPECT_TRUE(parameters.encodings[2].active);
>+  EXPECT_TRUE(fake_video_send_stream->IsSending());
>+
>+  // Only turn on only the middle stream.
>+  parameters.encodings[0].active = false;
>+  parameters.encodings[1].active = true;
>+  parameters.encodings[2].active = false;
>+  EXPECT_TRUE(channel_->SetRtpSendParameters(primary_ssrc, parameters).ok());
>+  // Verify that the active fields are set on the VideoChannel.
>+  parameters = channel_->GetRtpSendParameters(primary_ssrc);
>+  EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
>+  EXPECT_FALSE(parameters.encodings[0].active);
>+  EXPECT_TRUE(parameters.encodings[1].active);
>+  EXPECT_FALSE(parameters.encodings[2].active);
>+  // Check that the VideoSendStream is updated appropriately. This means its
>+  // send state was updated and it was reconfigured.
>+  EXPECT_TRUE(fake_video_send_stream->IsSending());
>+  std::vector<webrtc::VideoStream> simulcast_streams =
>+      fake_video_send_stream->GetVideoStreams();
>+  EXPECT_EQ(kNumSimulcastStreams, simulcast_streams.size());
>+  EXPECT_FALSE(simulcast_streams[0].active);
>+  EXPECT_TRUE(simulcast_streams[1].active);
>+  EXPECT_FALSE(simulcast_streams[2].active);
>+
>+  // Turn off all streams.
>+  parameters.encodings[0].active = false;
>+  parameters.encodings[1].active = false;
>+  parameters.encodings[2].active = false;
>+  EXPECT_TRUE(channel_->SetRtpSendParameters(primary_ssrc, parameters).ok());
>+  // Verify that the active fields are set on the VideoChannel.
>+  parameters = channel_->GetRtpSendParameters(primary_ssrc);
>+  EXPECT_EQ(kNumSimulcastStreams, parameters.encodings.size());
>+  EXPECT_FALSE(parameters.encodings[0].active);
>+  EXPECT_FALSE(parameters.encodings[1].active);
>+  EXPECT_FALSE(parameters.encodings[2].active);
>+  // Check that the VideoSendStream is off.
>+  EXPECT_FALSE(fake_video_send_stream->IsSending());
>+  simulcast_streams = fake_video_send_stream->GetVideoStreams();
>+  EXPECT_EQ(kNumSimulcastStreams, simulcast_streams.size());
>+  EXPECT_FALSE(simulcast_streams[0].active);
>+  EXPECT_FALSE(simulcast_streams[1].active);
>+  EXPECT_FALSE(simulcast_streams[2].active);
>+
>+  EXPECT_TRUE(channel_->SetVideoSend(primary_ssrc, nullptr, nullptr));
>+}
>+
> // Test that if a stream is reconfigured (due to a codec change or other
> // change) while its encoding is still inactive, it doesn't start sending.
> TEST_F(WebRtcVideoChannelTest,
>@@ -4547,7 +5993,7 @@ TEST_F(WebRtcVideoChannelTest,
>   parameters.encodings[0].active = false;
>   EXPECT_EQ(1u, GetFakeSendStreams().size());
>   EXPECT_EQ(1, fake_call_->GetNumCreatedSendStreams());
>-  EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters));
>+  EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, parameters).ok());
>   EXPECT_FALSE(stream->IsSending());
> 
>   // Reorder the codec list, causing the stream to be reconfigured.
>@@ -4583,6 +6029,16 @@ TEST_F(WebRtcVideoChannelTest, GetRtpSendParametersCodecs) {
>             rtp_parameters.codecs[1]);
> }
> 
>+// Test that GetRtpSendParameters returns the currently configured RTCP CNAME.
>+TEST_F(WebRtcVideoChannelTest, GetRtpSendParametersRtcpCname) {
>+  StreamParams params = StreamParams::CreateLegacy(kSsrc);
>+  params.cname = "rtcpcname";
>+  AddSendStream(params);
>+
>+  webrtc::RtpParameters rtp_parameters = channel_->GetRtpSendParameters(kSsrc);
>+  EXPECT_STREQ("rtcpcname", rtp_parameters.rtcp.cname.c_str());
>+}
>+
> // Test that RtpParameters for send stream has one encoding and it has
> // the correct SSRC.
> TEST_F(WebRtcVideoChannelTest, GetRtpSendParametersSsrc) {
>@@ -4594,6 +6050,44 @@ TEST_F(WebRtcVideoChannelTest, GetRtpSendParametersSsrc) {
>   EXPECT_EQ(last_ssrc_, rtp_parameters.encodings[0].ssrc);
> }
> 
>+TEST_F(WebRtcVideoChannelTest, DetectRtpSendParameterHeaderExtensionsChange) {
>+  AddSendStream();
>+
>+  webrtc::RtpParameters rtp_parameters =
>+      channel_->GetRtpSendParameters(last_ssrc_);
>+  rtp_parameters.header_extensions.emplace_back();
>+
>+  EXPECT_NE(0u, rtp_parameters.header_extensions.size());
>+
>+  webrtc::RTCError result =
>+      channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters);
>+  EXPECT_EQ(webrtc::RTCErrorType::INVALID_MODIFICATION, result.type());
>+}
>+
>+TEST_F(WebRtcVideoChannelTest, GetRtpSendParametersDegradationPreference) {
>+  AddSendStream();
>+
>+  FakeVideoCapturerWithTaskQueue capturer;
>+  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, &capturer));
>+
>+  webrtc::RtpParameters rtp_parameters =
>+      channel_->GetRtpSendParameters(last_ssrc_);
>+  EXPECT_EQ(rtp_parameters.degradation_preference,
>+            webrtc::DegradationPreference::BALANCED);
>+  rtp_parameters.degradation_preference =
>+      webrtc::DegradationPreference::MAINTAIN_FRAMERATE;
>+
>+  EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, rtp_parameters).ok());
>+
>+  webrtc::RtpParameters updated_rtp_parameters =
>+      channel_->GetRtpSendParameters(last_ssrc_);
>+  EXPECT_EQ(updated_rtp_parameters.degradation_preference,
>+            webrtc::DegradationPreference::MAINTAIN_FRAMERATE);
>+
>+  // Remove the source since it will be destroyed before the channel
>+  EXPECT_TRUE(channel_->SetVideoSend(last_ssrc_, nullptr, nullptr));
>+}
>+
> // Test that if we set/get parameters multiple times, we get the same results.
> TEST_F(WebRtcVideoChannelTest, SetAndGetRtpSendParameters) {
>   AddSendStream();
>@@ -4606,7 +6100,7 @@ TEST_F(WebRtcVideoChannelTest, SetAndGetRtpSendParameters) {
>       channel_->GetRtpSendParameters(last_ssrc_);
> 
>   // We should be able to set the params we just got.
>-  EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, initial_params));
>+  EXPECT_TRUE(channel_->SetRtpSendParameters(last_ssrc_, initial_params).ok());
> 
>   // ... And this shouldn't change the params returned by GetRtpSendParameters.
>   EXPECT_EQ(initial_params, channel_->GetRtpSendParameters(last_ssrc_));
>@@ -4768,15 +6262,13 @@ void WebRtcVideoChannelTest::TestReceiverLocalSsrcConfiguration(
>   // local send stream's SSRC.
>   AddSendStream(StreamParams::CreateLegacy(kSecondSenderSsrc));
>   ASSERT_TRUE(channel_->RemoveSendStream(kSenderSsrc));
>-  receive_streams =
>-      fake_call_->GetVideoReceiveStreams();
>+  receive_streams = fake_call_->GetVideoReceiveStreams();
>   ASSERT_EQ(1u, receive_streams.size());
>   EXPECT_EQ(kSecondSenderSsrc, receive_streams[0]->GetConfig().rtp.local_ssrc);
> 
>   // Removing the last sender should fall back to default local SSRC.
>   ASSERT_TRUE(channel_->RemoveSendStream(kSecondSenderSsrc));
>-  receive_streams =
>-      fake_call_->GetVideoReceiveStreams();
>+  receive_streams = fake_call_->GetVideoReceiveStreams();
>   ASSERT_EQ(1u, receive_streams.size());
>   EXPECT_EQ(kExpectedDefaultReceiverSsrc,
>             receive_streams[0]->GetConfig().rtp.local_ssrc);
>@@ -4793,16 +6285,17 @@ TEST_F(WebRtcVideoChannelTest, ConfiguresLocalSsrcOnExistingReceivers) {
> class WebRtcVideoChannelSimulcastTest : public testing::Test {
>  public:
>   WebRtcVideoChannelSimulcastTest()
>-      : fake_call_(webrtc::Call::Config(&event_log_)),
>+      : fake_call_(),
>         encoder_factory_(new cricket::FakeWebRtcVideoEncoderFactory),
>         decoder_factory_(new cricket::FakeWebRtcVideoDecoderFactory),
>-        engine_(std::unique_ptr<cricket::WebRtcVideoEncoderFactory>(
>+        engine_(std::unique_ptr<cricket::FakeWebRtcVideoEncoderFactory>(
>                     encoder_factory_),
>-                std::unique_ptr<cricket::WebRtcVideoDecoderFactory>(
>+                std::unique_ptr<cricket::FakeWebRtcVideoDecoderFactory>(
>                     decoder_factory_)),
>         last_ssrc_(0) {}
> 
>   void SetUp() override {
>+    encoder_factory_->AddSupportedVideoCodecType("VP8");
>     channel_.reset(
>         engine_.CreateChannel(&fake_call_, GetMediaConfig(), VideoOptions()));
>     channel_->OnReadyToSend(true);
>@@ -4829,12 +6322,11 @@ class WebRtcVideoChannelSimulcastTest : public testing::Test {
>     AddSendStream(CreateSimStreamParams("cname", ssrcs));
>     // Send a full-size frame to trigger a stream reconfiguration to use all
>     // expected simulcast layers.
>-    cricket::FakeVideoCapturer capturer;
>+    FakeVideoCapturerWithTaskQueue capturer;
>     VideoOptions options;
>     if (screenshare)
>       options.is_screencast = screenshare;
>-    EXPECT_TRUE(
>-        channel_->SetVideoSend(ssrcs.front(), true, &options, &capturer));
>+    EXPECT_TRUE(channel_->SetVideoSend(ssrcs.front(), &options, &capturer));
>     // Fetch the latest stream since SetVideoSend() may recreate it if the
>     // screen content setting is changed.
>     FakeVideoSendStream* stream = fake_call_.GetVideoSendStreams().front();
>@@ -4845,6 +6337,9 @@ class WebRtcVideoChannelSimulcastTest : public testing::Test {
>     channel_->SetSend(true);
>     EXPECT_TRUE(capturer.CaptureFrame());
> 
>+    auto rtp_parameters = channel_->GetRtpSendParameters(kSsrcs3[0]);
>+    EXPECT_EQ(num_configured_streams, rtp_parameters.encodings.size());
>+
>     std::vector<webrtc::VideoStream> video_streams = stream->GetVideoStreams();
>     ASSERT_EQ(expected_num_streams, video_streams.size());
> 
>@@ -4853,12 +6348,12 @@ class WebRtcVideoChannelSimulcastTest : public testing::Test {
>       expected_streams = GetSimulcastConfig(
>           num_configured_streams, capture_width, capture_height, 0,
>           webrtc::kDefaultBitratePriority, kDefaultQpMax,
>-          kDefaultVideoMaxFramerate, screenshare);
>+          kDefaultVideoMaxFramerate, screenshare, true);
>       if (screenshare) {
>         for (const webrtc::VideoStream& stream : expected_streams) {
>           // Never scale screen content.
>-          EXPECT_EQ(stream.width, capture_width);
>-          EXPECT_EQ(stream.height, capture_height);
>+          EXPECT_EQ(stream.width, rtc::checked_cast<size_t>(capture_width));
>+          EXPECT_EQ(stream.height, rtc::checked_cast<size_t>(capture_height));
>         }
>       }
>     } else {
>@@ -4867,18 +6362,8 @@ class WebRtcVideoChannelSimulcastTest : public testing::Test {
>       stream.height = capture_height;
>       stream.max_framerate = kDefaultVideoMaxFramerate;
>       stream.min_bitrate_bps = cricket::kMinVideoBitrateBps;
>-      int max_bitrate_kbps;
>-      if (capture_width * capture_height <= 320 * 240) {
>-        max_bitrate_kbps = 600;
>-      } else if (capture_width * capture_height <= 640 * 480) {
>-        max_bitrate_kbps = 1700;
>-      } else if (capture_width * capture_height <= 960 * 540) {
>-        max_bitrate_kbps = 2000;
>-      } else {
>-        max_bitrate_kbps = 2500;
>-      }
>       stream.target_bitrate_bps = stream.max_bitrate_bps =
>-          max_bitrate_kbps * 1000;
>+          GetMaxDefaultBitrateBps(capture_width, capture_height);
>       stream.max_qp = kDefaultQpMax;
>       expected_streams.push_back(stream);
>     }
>@@ -4910,10 +6395,13 @@ class WebRtcVideoChannelSimulcastTest : public testing::Test {
>       EXPECT_GT(video_streams[i].max_qp, 0);
>       EXPECT_EQ(expected_streams[i].max_qp, video_streams[i].max_qp);
> 
>-      EXPECT_EQ(!conference_mode,
>-                expected_streams[i].temporal_layer_thresholds_bps.empty());
>-      EXPECT_EQ(expected_streams[i].temporal_layer_thresholds_bps,
>-                video_streams[i].temporal_layer_thresholds_bps);
>+      EXPECT_EQ(conference_mode,
>+                expected_streams[i].num_temporal_layers.has_value());
>+
>+      if (conference_mode) {
>+        EXPECT_EQ(expected_streams[i].num_temporal_layers,
>+                  video_streams[i].num_temporal_layers);
>+      }
> 
>       if (i == num_streams - 1) {
>         total_max_bitrate_bps += video_streams[i].max_bitrate_bps;
>@@ -4922,7 +6410,7 @@ class WebRtcVideoChannelSimulcastTest : public testing::Test {
>       }
>     }
> 
>-    EXPECT_TRUE(channel_->SetVideoSend(ssrcs.front(), true, nullptr, nullptr));
>+    EXPECT_TRUE(channel_->SetVideoSend(ssrcs.front(), nullptr, nullptr));
>   }
> 
>   FakeVideoSendStream* AddSendStream() {
>@@ -4930,8 +6418,7 @@ class WebRtcVideoChannelSimulcastTest : public testing::Test {
>   }
> 
>   FakeVideoSendStream* AddSendStream(const StreamParams& sp) {
>-    size_t num_streams =
>-        fake_call_.GetVideoSendStreams().size();
>+    size_t num_streams = fake_call_.GetVideoSendStreams().size();
>     EXPECT_TRUE(channel_->AddSendStream(sp));
>     std::vector<FakeVideoSendStream*> streams =
>         fake_call_.GetVideoSendStreams();
>@@ -4948,8 +6435,7 @@ class WebRtcVideoChannelSimulcastTest : public testing::Test {
>   }
> 
>   FakeVideoReceiveStream* AddRecvStream(const StreamParams& sp) {
>-    size_t num_streams =
>-        fake_call_.GetVideoReceiveStreams().size();
>+    size_t num_streams = fake_call_.GetVideoReceiveStreams().size();
>     EXPECT_TRUE(channel_->AddRecvStream(sp));
>     std::vector<FakeVideoReceiveStream*> streams =
>         fake_call_.GetVideoReceiveStreams();
>@@ -4982,28 +6468,13 @@ TEST_F(WebRtcVideoChannelSimulcastTest, SetSendCodecsWithOddSizeInSimulcast) {
>                           true);
> }
> 
>-TEST_F(WebRtcVideoChannelSimulcastTest, SetSendCodecsForScreenshare) {
>-  VerifySimulcastSettings(cricket::VideoCodec("VP8"), 1280, 720, 3, 1, true,
>-                          false);
>-}
>-
>-TEST_F(WebRtcVideoChannelSimulcastTest,
>-       SetSendCodecsForConferenceModeScreenshare) {
>-  VerifySimulcastSettings(cricket::VideoCodec("VP8"), 1280, 720, 3, 1, true,
>-                          true);
>-}
>-
> TEST_F(WebRtcVideoChannelSimulcastTest, SetSendCodecsForSimulcastScreenshare) {
>-  webrtc::test::ScopedFieldTrials override_field_trials_(
>-      "WebRTC-SimulcastScreenshare/Enabled/");
>   VerifySimulcastSettings(cricket::VideoCodec("VP8"), 1280, 720, 3, 2, true,
>                           true);
> }
> 
> TEST_F(WebRtcVideoChannelSimulcastTest,
>        NoSimulcastScreenshareWithoutConference) {
>-  webrtc::test::ScopedFieldTrials override_field_trials_(
>-      "WebRTC-SimulcastScreenshare/Enabled/");
>   VerifySimulcastSettings(cricket::VideoCodec("VP8"), 1280, 720, 3, 1, true,
>                           false);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvoiceengine.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvoiceengine.cc
>index a1cbf9204f3..f8aa5936338 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvoiceengine.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvoiceengine.cc
>@@ -19,6 +19,7 @@
> #include <utility>
> #include <vector>
> 
>+#include "api/audio_codecs/audio_codec_pair_id.h"
> #include "api/call/audio_sink.h"
> #include "media/base/audiosource.h"
> #include "media/base/mediaconstants.h"
>@@ -32,14 +33,15 @@
> #include "modules/audio_processing/aec_dump/aec_dump_factory.h"
> #include "modules/audio_processing/include/audio_processing.h"
> #include "rtc_base/arraysize.h"
>-#include "rtc_base/base64.h"
> #include "rtc_base/byteorder.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/helpers.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/race_checker.h"
> #include "rtc_base/stringencode.h"
>+#include "rtc_base/strings/audio_format_to_string.h"
> #include "rtc_base/stringutils.h"
>+#include "rtc_base/third_party/base64/base64.h"
> #include "rtc_base/trace_event.h"
> #include "system_wrappers/include/field_trial.h"
> #include "system_wrappers/include/metrics.h"
>@@ -68,7 +70,7 @@ const int kOpusBitrateFbBps = 32000;
> // See also http://tools.ietf.org/html/draft-jennings-rtcweb-qos-00
> const rtc::DiffServCodePoint kAudioDscpValue = rtc::DSCP_EF;
> 
>-const int kMinTelephoneEventCode = 0;           // RFC4733 (Section 2.3.1)
>+const int kMinTelephoneEventCode = 0;  // RFC4733 (Section 2.3.1)
> const int kMaxTelephoneEventCode = 255;
> 
> const int kMinPayloadType = 0;
>@@ -145,7 +147,7 @@ bool VerifyUniquePayloadTypes(const std::vector<AudioCodec>& codecs) {
>   return it == payload_types.end();
> }
> 
>-rtc::Optional<std::string> GetAudioNetworkAdaptorConfig(
>+absl::optional<std::string> GetAudioNetworkAdaptorConfig(
>     const AudioOptions& options) {
>   if (options.audio_network_adaptor && *options.audio_network_adaptor &&
>       options.audio_network_adaptor_config) {
>@@ -153,14 +155,14 @@ rtc::Optional<std::string> GetAudioNetworkAdaptorConfig(
>     // equals true and |options_.audio_network_adaptor_config| has a value.
>     return options.audio_network_adaptor_config;
>   }
>-  return rtc::nullopt;
>+  return absl::nullopt;
> }
> 
> // |max_send_bitrate_bps| is the bitrate from "b=" in SDP.
> // |rtp_max_bitrate_bps| is the bitrate from RtpSender::SetParameters.
>-rtc::Optional<int> ComputeSendBitrate(int max_send_bitrate_bps,
>-                                      rtc::Optional<int> rtp_max_bitrate_bps,
>-                                      const webrtc::AudioCodecSpec& spec) {
>+absl::optional<int> ComputeSendBitrate(int max_send_bitrate_bps,
>+                                       absl::optional<int> rtp_max_bitrate_bps,
>+                                       const webrtc::AudioCodecSpec& spec) {
>   // If application-configured bitrate is set, take minimum of that and SDP
>   // bitrate.
>   const int bps =
>@@ -179,7 +181,7 @@ rtc::Optional<int> ComputeSendBitrate(int max_send_bitrate_bps,
>                       << " to bitrate " << bps << " bps"
>                       << ", requires at least " << spec.info.min_bitrate_bps
>                       << " bps.";
>-    return rtc::nullopt;
>+    return absl::nullopt;
>   }
> 
>   if (spec.info.HasFixedBitrate()) {
>@@ -295,7 +297,6 @@ void WebRtcVoiceEngine::Init() {
>     options.delay_agnostic_aec = false;
>     options.experimental_ns = false;
>     options.intelligibility_enhancer = false;
>-    options.level_control = false;
>     options.residual_echo_detector = true;
>     bool error = ApplyOptions(options);
>     RTC_DCHECK(error);
>@@ -304,8 +305,8 @@ void WebRtcVoiceEngine::Init() {
>   initialized_ = true;
> }
> 
>-rtc::scoped_refptr<webrtc::AudioState>
>-    WebRtcVoiceEngine::GetAudioState() const {
>+rtc::scoped_refptr<webrtc::AudioState> WebRtcVoiceEngine::GetAudioState()
>+    const {
>   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>   return audio_state_;
> }
>@@ -435,8 +436,8 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
>             << "Disabling EC since built-in EC will be used instead";
>       }
>     }
>-    webrtc::apm_helpers::SetEcStatus(
>-        apm(), *options.echo_cancellation, ec_mode);
>+    webrtc::apm_helpers::SetEcStatus(apm(), *options.echo_cancellation,
>+                                     ec_mode);
> #if !defined(WEBRTC_ANDROID)
>     webrtc::apm_helpers::SetEcMetricsStatus(apm(), *options.echo_cancellation);
> #endif
>@@ -523,8 +524,8 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
>   if (options.typing_detection) {
>     RTC_LOG(LS_INFO) << "Typing detection is enabled? "
>                      << *options.typing_detection;
>-    webrtc::apm_helpers::SetTypingDetectionStatus(
>-        apm(), *options.typing_detection);
>+    webrtc::apm_helpers::SetTypingDetectionStatus(apm(),
>+                                                  *options.typing_detection);
>   }
> 
>   webrtc::Config config;
>@@ -564,22 +565,8 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
>         new webrtc::Intelligibility(*intelligibility_enhancer_));
>   }
> 
>-  if (options.level_control) {
>-    level_control_ = options.level_control;
>-  }
>-
>   webrtc::AudioProcessing::Config apm_config = apm()->GetConfig();
> 
>-  RTC_LOG(LS_INFO) << "Level control: "
>-                   << (!!level_control_ ? *level_control_ : -1);
>-  if (level_control_) {
>-    apm_config.level_controller.enabled = *level_control_;
>-    if (options.level_control_initial_peak_level_dbfs) {
>-      apm_config.level_controller.initial_peak_level_dbfs =
>-          *options.level_control_initial_peak_level_dbfs;
>-    }
>-  }
>-
>   if (options.highpass_filter) {
>     apm_config.high_pass_filter.enabled = *options.highpass_filter;
>   }
>@@ -593,12 +580,6 @@ bool WebRtcVoiceEngine::ApplyOptions(const AudioOptions& options_in) {
>   return true;
> }
> 
>-// TODO(solenberg): Remove, once AudioMonitor is gone.
>-int WebRtcVoiceEngine::GetInputLevel() {
>-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  return audio_state()->GetAudioInputStats().quantized_audio_level;
>-}
>-
> const std::vector<AudioCodec>& WebRtcVoiceEngine::send_codecs() const {
>   RTC_DCHECK(signal_thread_checker_.CalledOnValidThread());
>   return send_codecs_;
>@@ -615,11 +596,16 @@ RtpCapabilities WebRtcVoiceEngine::GetCapabilities() const {
>   capabilities.header_extensions.push_back(
>       webrtc::RtpExtension(webrtc::RtpExtension::kAudioLevelUri,
>                            webrtc::RtpExtension::kAudioLevelDefaultId));
>-  if (webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe")) {
>+  if (webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe") &&
>+      !webrtc::field_trial::IsEnabled("WebRTC-Audio-ABWENoTWCC")) {
>     capabilities.header_extensions.push_back(webrtc::RtpExtension(
>         webrtc::RtpExtension::kTransportSequenceNumberUri,
>         webrtc::RtpExtension::kTransportSequenceNumberDefaultId));
>   }
>+  // TODO(bugs.webrtc.org/4050): Add MID header extension as capability once MID
>+  // demuxing is completed.
>+  // capabilities.header_extensions.push_back(webrtc::RtpExtension(
>+  //     webrtc::RtpExtension::kMidUri, webrtc::RtpExtension::kMidDefaultId));
>   return capabilities;
> }
> 
>@@ -687,25 +673,22 @@ AudioCodecs WebRtcVoiceEngine::CollectCodecs(
>   AudioCodecs out;
> 
>   // Only generate CN payload types for these clockrates:
>-  std::map<int, bool, std::greater<int>> generate_cn = {{ 8000,  false },
>-                                                        { 16000, false },
>-                                                        { 32000, false }};
>+  std::map<int, bool, std::greater<int>> generate_cn = {
>+      {8000, false}, {16000, false}, {32000, false}};
>   // Only generate telephone-event payload types for these clockrates:
>-  std::map<int, bool, std::greater<int>> generate_dtmf = {{ 8000,  false },
>-                                                          { 16000, false },
>-                                                          { 32000, false },
>-                                                          { 48000, false }};
>+  std::map<int, bool, std::greater<int>> generate_dtmf = {
>+      {8000, false}, {16000, false}, {32000, false}, {48000, false}};
> 
>   auto map_format = [&mapper](const webrtc::SdpAudioFormat& format,
>                               AudioCodecs* out) {
>-    rtc::Optional<AudioCodec> opt_codec = mapper.ToAudioCodec(format);
>+    absl::optional<AudioCodec> opt_codec = mapper.ToAudioCodec(format);
>     if (opt_codec) {
>       if (out) {
>         out->push_back(*opt_codec);
>       }
>     } else {
>       RTC_LOG(LS_ERROR) << "Unable to assign payload type to format: "
>-                        << format;
>+                        << rtc::ToString(format);
>     }
> 
>     return opt_codec;
>@@ -713,7 +696,7 @@ AudioCodecs WebRtcVoiceEngine::CollectCodecs(
> 
>   for (const auto& spec : specs) {
>     // We need to do some extra stuff before adding the main codecs to out.
>-    rtc::Optional<AudioCodec> opt_codec = map_format(spec.format, nullptr);
>+    absl::optional<AudioCodec> opt_codec = map_format(spec.format, nullptr);
>     if (opt_codec) {
>       AudioCodec& codec = *opt_codec;
>       if (spec.info.supports_network_adaption) {
>@@ -762,16 +745,18 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
>  public:
>   WebRtcAudioSendStream(
>       uint32_t ssrc,
>+      const std::string& mid,
>       const std::string& c_name,
>       const std::string track_id,
>-      const rtc::Optional<webrtc::AudioSendStream::Config::SendCodecSpec>&
>+      const absl::optional<webrtc::AudioSendStream::Config::SendCodecSpec>&
>           send_codec_spec,
>       const std::vector<webrtc::RtpExtension>& extensions,
>       int max_send_bitrate_bps,
>-      const rtc::Optional<std::string>& audio_network_adaptor_config,
>+      const absl::optional<std::string>& audio_network_adaptor_config,
>       webrtc::Call* call,
>       webrtc::Transport* send_transport,
>-      const rtc::scoped_refptr<webrtc::AudioEncoderFactory>& encoder_factory)
>+      const rtc::scoped_refptr<webrtc::AudioEncoderFactory>& encoder_factory,
>+      const absl::optional<webrtc::AudioCodecPairId> codec_pair_id)
>       : call_(call),
>         config_(send_transport),
>         send_side_bwe_with_overhead_(
>@@ -781,12 +766,16 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
>     RTC_DCHECK(call);
>     RTC_DCHECK(encoder_factory);
>     config_.rtp.ssrc = ssrc;
>+    config_.rtp.mid = mid;
>     config_.rtp.c_name = c_name;
>     config_.rtp.extensions = extensions;
>     config_.audio_network_adaptor_config = audio_network_adaptor_config;
>     config_.encoder_factory = encoder_factory;
>+    config_.codec_pair_id = codec_pair_id;
>     config_.track_id = track_id;
>     rtp_parameters_.encodings[0].ssrc = ssrc;
>+    rtp_parameters_.rtcp.cname = c_name;
>+    rtp_parameters_.header_extensions = extensions;
> 
>     if (send_codec_spec) {
>       UpdateSendCodecSpec(*send_codec_spec);
>@@ -810,11 +799,21 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
>   void SetRtpExtensions(const std::vector<webrtc::RtpExtension>& extensions) {
>     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>     config_.rtp.extensions = extensions;
>+    rtp_parameters_.header_extensions = extensions;
>+    ReconfigureAudioSendStream();
>+  }
>+
>+  void SetMid(const std::string& mid) {
>+    RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>+    if (config_.rtp.mid == mid) {
>+      return;
>+    }
>+    config_.rtp.mid = mid;
>     ReconfigureAudioSendStream();
>   }
> 
>   void SetAudioNetworkAdaptorConfig(
>-      const rtc::Optional<std::string>& audio_network_adaptor_config) {
>+      const absl::optional<std::string>& audio_network_adaptor_config) {
>     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>     if (config_.audio_network_adaptor_config == audio_network_adaptor_config) {
>       return;
>@@ -844,7 +843,9 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
>     return true;
>   }
> 
>-  bool SendTelephoneEvent(int payload_type, int payload_freq, int event,
>+  bool SendTelephoneEvent(int payload_type,
>+                          int payload_freq,
>+                          int event,
>                           int duration_ms) {
>     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>     RTC_DCHECK(stream_);
>@@ -915,13 +916,10 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
>     RTC_CHECK_RUNS_SERIALIZED(&audio_capture_race_checker_);
>     RTC_DCHECK(stream_);
>     std::unique_ptr<webrtc::AudioFrame> audio_frame(new webrtc::AudioFrame());
>-    audio_frame->UpdateFrame(audio_frame->timestamp_,
>-                             static_cast<const int16_t*>(audio_data),
>-                             number_of_frames,
>-                             sample_rate,
>-                             audio_frame->speech_type_,
>-                             audio_frame->vad_activity_,
>-                             number_of_channels);
>+    audio_frame->UpdateFrame(
>+        audio_frame->timestamp_, static_cast<const int16_t*>(audio_data),
>+        number_of_frames, sample_rate, audio_frame->speech_type_,
>+        audio_frame->vad_activity_, number_of_channels);
>     stream_->SendAudioData(std::move(audio_frame));
>   }
> 
>@@ -939,40 +937,53 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
>     return rtp_parameters_;
>   }
> 
>-  bool ValidateRtpParameters(const webrtc::RtpParameters& rtp_parameters) {
>-    if (rtp_parameters.encodings.size() != 1) {
>-      RTC_LOG(LS_ERROR)
>-          << "Attempted to set RtpParameters without exactly one encoding";
>-      return false;
>+  webrtc::RTCError ValidateRtpParameters(
>+      const webrtc::RtpParameters& rtp_parameters) {
>+    using webrtc::RTCErrorType;
>+    if (rtp_parameters.encodings.size() != rtp_parameters_.encodings.size()) {
>+      LOG_AND_RETURN_ERROR(
>+          RTCErrorType::INVALID_MODIFICATION,
>+          "Attempted to set RtpParameters with different encoding count");
>+    }
>+    if (rtp_parameters.rtcp != rtp_parameters_.rtcp) {
>+      LOG_AND_RETURN_ERROR(
>+          RTCErrorType::INVALID_MODIFICATION,
>+          "Attempted to set RtpParameters with modified RTCP parameters");
>+    }
>+    if (rtp_parameters.header_extensions != rtp_parameters_.header_extensions) {
>+      LOG_AND_RETURN_ERROR(
>+          RTCErrorType::INVALID_MODIFICATION,
>+          "Attempted to set RtpParameters with modified header extensions");
>     }
>     if (rtp_parameters.encodings[0].ssrc != rtp_parameters_.encodings[0].ssrc) {
>-      RTC_LOG(LS_ERROR) << "Attempted to set RtpParameters with modified SSRC";
>-      return false;
>+      LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_MODIFICATION,
>+                           "Attempted to set RtpParameters with modified SSRC");
>     }
>     if (rtp_parameters.encodings[0].bitrate_priority <= 0) {
>-      RTC_LOG(LS_ERROR) << "Attempted to set RtpParameters bitrate_priority to "
>-                           "an invalid number.";
>-      return false;
>+      LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_RANGE,
>+                           "Attempted to set RtpParameters bitrate_priority to "
>+                           "an invalid number.");
>     }
>-    return true;
>+    return webrtc::RTCError::OK();
>   }
> 
>-  bool SetRtpParameters(const webrtc::RtpParameters& parameters) {
>-    if (!ValidateRtpParameters(parameters)) {
>-      return false;
>+  webrtc::RTCError SetRtpParameters(const webrtc::RtpParameters& parameters) {
>+    webrtc::RTCError error = ValidateRtpParameters(parameters);
>+    if (!error.ok()) {
>+      return error;
>     }
> 
>-    rtc::Optional<int> send_rate;
>+    absl::optional<int> send_rate;
>     if (audio_codec_spec_) {
>       send_rate = ComputeSendBitrate(max_send_bitrate_bps_,
>                                      parameters.encodings[0].max_bitrate_bps,
>                                      *audio_codec_spec_);
>       if (!send_rate) {
>-        return false;
>+        return webrtc::RTCError(webrtc::RTCErrorType::INTERNAL_ERROR);
>       }
>     }
> 
>-    const rtc::Optional<int> old_rtp_max_bitrate =
>+    const absl::optional<int> old_rtp_max_bitrate =
>         rtp_parameters_.encodings[0].max_bitrate_bps;
>     double old_priority = rtp_parameters_.encodings[0].bitrate_priority;
>     rtp_parameters_ = parameters;
>@@ -991,9 +1002,13 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
>     if (reconfigure_send_stream) {
>       ReconfigureAudioSendStream();
>     }
>+
>+    rtp_parameters_.rtcp.cname = config_.rtp.c_name;
>+    rtp_parameters_.rtcp.reduced_size = false;
>+
>     // parameters.encodings[0].active could have changed.
>     UpdateSendState();
>-    return true;
>+    return webrtc::RTCError::OK();
>   }
> 
>  private:
>@@ -1104,7 +1119,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
>   bool muted_ = false;
>   int max_send_bitrate_bps_;
>   webrtc::RtpParameters rtp_parameters_;
>-  rtc::Optional<webrtc::AudioCodecSpec> audio_codec_spec_;
>+  absl::optional<webrtc::AudioCodecSpec> audio_codec_spec_;
> 
>   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioSendStream);
> };
>@@ -1116,12 +1131,13 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
>       uint32_t local_ssrc,
>       bool use_transport_cc,
>       bool use_nack,
>-      const std::string& sync_group,
>+      const std::vector<std::string>& stream_ids,
>       const std::vector<webrtc::RtpExtension>& extensions,
>       webrtc::Call* call,
>       webrtc::Transport* rtcp_send_transport,
>       const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory,
>       const std::map<int, webrtc::SdpAudioFormat>& decoder_map,
>+      absl::optional<webrtc::AudioCodecPairId> codec_pair_id,
>       size_t jitter_buffer_max_packets,
>       bool jitter_buffer_fast_accelerate)
>       : call_(call), config_() {
>@@ -1134,9 +1150,12 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
>     config_.rtcp_send_transport = rtcp_send_transport;
>     config_.jitter_buffer_max_packets = jitter_buffer_max_packets;
>     config_.jitter_buffer_fast_accelerate = jitter_buffer_fast_accelerate;
>-    config_.sync_group = sync_group;
>+    if (!stream_ids.empty()) {
>+      config_.sync_group = stream_ids[0];
>+    }
>     config_.decoder_factory = decoder_factory;
>     config_.decoder_map = decoder_map;
>+    config_.codec_pair_id = codec_pair_id;
>     RecreateAudioReceiveStream();
>   }
> 
>@@ -1173,8 +1192,13 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
>     ReconfigureAudioReceiveStream();
>   }
> 
>-  void MaybeRecreateAudioReceiveStream(const std::string& sync_group) {
>+  void MaybeRecreateAudioReceiveStream(
>+      const std::vector<std::string>& stream_ids) {
>     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>+    std::string sync_group;
>+    if (!stream_ids.empty()) {
>+      sync_group = stream_ids[0];
>+    }
>     if (config_.sync_group != sync_group) {
>       RTC_LOG(LS_INFO) << "Recreating AudioReceiveStream for SSRC="
>                        << config_.rtp.remote_ssrc
>@@ -1190,20 +1214,17 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
>     return stream_->GetStats();
>   }
> 
>-  int GetOutputLevel() const {
>-    RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-    RTC_DCHECK(stream_);
>-    return stream_->GetOutputLevel();
>-  }
>-
>   void SetRawAudioSink(std::unique_ptr<webrtc::AudioSinkInterface> sink) {
>     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>+    // Need to update the stream's sink first; once raw_audio_sink_ is
>+    // reassigned, whatever was in there before is destroyed.
>+    stream_->SetSink(sink.get());
>     raw_audio_sink_ = std::move(sink);
>-    stream_->SetSink(raw_audio_sink_.get());
>   }
> 
>   void SetOutputVolume(double volume) {
>     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>+    output_volume_ = volume;
>     stream_->SetGain(volume);
>   }
> 
>@@ -1224,6 +1245,15 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
>     return stream_->GetSources();
>   }
> 
>+  webrtc::RtpParameters GetRtpParameters() const {
>+    webrtc::RtpParameters rtp_parameters;
>+    rtp_parameters.encodings.emplace_back();
>+    rtp_parameters.encodings[0].ssrc = config_.rtp.remote_ssrc;
>+    rtp_parameters.header_extensions = config_.rtp.extensions;
>+
>+    return rtp_parameters;
>+  }
>+
>  private:
>   void RecreateAudioReceiveStream() {
>     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>@@ -1232,6 +1262,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
>     }
>     stream_ = call_->CreateAudioReceiveStream(config_);
>     RTC_CHECK(stream_);
>+    stream_->SetGain(output_volume_);
>     SetPlayout(playout_);
>     stream_->SetSink(raw_audio_sink_.get());
>   }
>@@ -1249,6 +1280,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
>   // configuration changes.
>   webrtc::AudioReceiveStream* stream_ = nullptr;
>   bool playout_ = false;
>+  float output_volume_ = 1.0;
>   std::unique_ptr<webrtc::AudioSinkInterface> raw_audio_sink_;
> 
>   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioReceiveStream);
>@@ -1300,15 +1332,20 @@ bool WebRtcVoiceMediaChannel::SetSendParameters(
>   if (!ValidateRtpExtensions(params.extensions)) {
>     return false;
>   }
>-  std::vector<webrtc::RtpExtension> filtered_extensions =
>-      FilterRtpExtensions(params.extensions,
>-                          webrtc::RtpExtension::IsSupportedForAudio, true);
>+  std::vector<webrtc::RtpExtension> filtered_extensions = FilterRtpExtensions(
>+      params.extensions, webrtc::RtpExtension::IsSupportedForAudio, true);
>   if (send_rtp_extensions_ != filtered_extensions) {
>     send_rtp_extensions_.swap(filtered_extensions);
>     for (auto& it : send_streams_) {
>       it.second->SetRtpExtensions(send_rtp_extensions_);
>     }
>   }
>+  if (!params.mid.empty()) {
>+    mid_ = params.mid;
>+    for (auto& it : send_streams_) {
>+      it.second->SetMid(params.mid);
>+    }
>+  }
> 
>   if (!SetMaxSendBitrate(params.max_bandwidth_bps)) {
>     return false;
>@@ -1332,9 +1369,8 @@ bool WebRtcVoiceMediaChannel::SetRecvParameters(
>   if (!ValidateRtpExtensions(params.extensions)) {
>     return false;
>   }
>-  std::vector<webrtc::RtpExtension> filtered_extensions =
>-      FilterRtpExtensions(params.extensions,
>-                          webrtc::RtpExtension::IsSupportedForAudio, false);
>+  std::vector<webrtc::RtpExtension> filtered_extensions = FilterRtpExtensions(
>+      params.extensions, webrtc::RtpExtension::IsSupportedForAudio, false);
>   if (recv_rtp_extensions_ != filtered_extensions) {
>     recv_rtp_extensions_.swap(filtered_extensions);
>     for (auto& it : recv_streams_) {
>@@ -1363,7 +1399,7 @@ webrtc::RtpParameters WebRtcVoiceMediaChannel::GetRtpSendParameters(
>   return rtp_params;
> }
> 
>-bool WebRtcVoiceMediaChannel::SetRtpSendParameters(
>+webrtc::RTCError WebRtcVoiceMediaChannel::SetRtpSendParameters(
>     uint32_t ssrc,
>     const webrtc::RtpParameters& parameters) {
>   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>@@ -1371,7 +1407,7 @@ bool WebRtcVoiceMediaChannel::SetRtpSendParameters(
>   if (it == send_streams_.end()) {
>     RTC_LOG(LS_WARNING) << "Attempting to set RTP send parameters for stream "
>                         << "with ssrc " << ssrc << " which doesn't exist.";
>-    return false;
>+    return webrtc::RTCError(webrtc::RTCErrorType::INTERNAL_ERROR);
>   }
> 
>   // TODO(deadbeef): Handle setting parameters with a list of codecs in a
>@@ -1380,7 +1416,7 @@ bool WebRtcVoiceMediaChannel::SetRtpSendParameters(
>   if (current_parameters.codecs != parameters.codecs) {
>     RTC_LOG(LS_ERROR) << "Using SetParameters to change the set of codecs "
>                       << "is not currently supported.";
>-    return false;
>+    return webrtc::RTCError(webrtc::RTCErrorType::UNSUPPORTED_PARAMETER);
>   }
> 
>   // TODO(minyue): The following legacy actions go into
>@@ -1419,9 +1455,7 @@ webrtc::RtpParameters WebRtcVoiceMediaChannel::GetRtpReceiveParameters(
>           << "with ssrc " << ssrc << " which doesn't exist.";
>       return webrtc::RtpParameters();
>     }
>-    rtp_params.encodings.emplace_back();
>-    // TODO(deadbeef): Return stream-specific parameters.
>-    rtp_params.encodings[0].ssrc = ssrc;
>+    rtp_params = it->second->GetRtpParameters();
>   }
> 
>   for (const AudioCodec& codec : recv_codecs_) {
>@@ -1476,7 +1510,7 @@ bool WebRtcVoiceMediaChannel::SetOptions(const AudioOptions& options) {
>     return false;
>   }
> 
>-  rtc::Optional<std::string> audio_network_adaptor_config =
>+  absl::optional<std::string> audio_network_adaptor_config =
>       GetAudioNetworkAdaptorConfig(options_);
>   for (auto& it : send_streams_) {
>     it.second->SetAudioNetworkAdaptorConfig(audio_network_adaptor_config);
>@@ -1515,7 +1549,7 @@ bool WebRtcVoiceMediaChannel::SetRecvCodecs(
>     auto format = AudioCodecToSdpAudioFormat(codec);
>     if (!IsCodec(codec, "cn") && !IsCodec(codec, "telephone-event") &&
>         !engine()->decoder_factory_->IsSupportedDecoder(format)) {
>-      RTC_LOG(LS_ERROR) << "Unsupported codec: " << format;
>+      RTC_LOG(LS_ERROR) << "Unsupported codec: " << rtc::ToString(format);
>       return false;
>     }
>     // We allow adding new codecs but don't allow changing the payload type of
>@@ -1569,7 +1603,7 @@ bool WebRtcVoiceMediaChannel::SetRecvCodecs(
> bool WebRtcVoiceMediaChannel::SetSendCodecs(
>     const std::vector<AudioCodec>& codecs) {
>   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  dtmf_payload_type_ = rtc::nullopt;
>+  dtmf_payload_type_ = absl::nullopt;
>   dtmf_payload_freq_ = -1;
> 
>   // Validate supplied codecs list.
>@@ -1599,9 +1633,10 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
>   }
> 
>   // Scan through the list to figure out the codec to use for sending.
>-  rtc::Optional<webrtc::AudioSendStream::Config::SendCodecSpec> send_codec_spec;
>-  webrtc::Call::Config::BitrateConfig bitrate_config;
>-  rtc::Optional<webrtc::AudioCodecInfo> voice_codec_info;
>+  absl::optional<webrtc::AudioSendStream::Config::SendCodecSpec>
>+      send_codec_spec;
>+  webrtc::BitrateConstraints bitrate_config;
>+  absl::optional<webrtc::AudioCodecInfo> voice_codec_info;
>   for (const AudioCodec& voice_codec : codecs) {
>     if (!(IsCodec(voice_codec, kCnCodecName) ||
>           IsCodec(voice_codec, kDtmfCodecName) ||
>@@ -1674,7 +1709,7 @@ bool WebRtcVoiceMediaChannel::SetSendCodecs(
>     // "unchanged" so that BWE isn't affected.
>     bitrate_config.start_bitrate_bps = -1;
>   }
>-  call_->SetBitrateConfig(bitrate_config);
>+  call_->GetTransportControllerSend()->SetSdpBitrateParameters(bitrate_config);
> 
>   // Check if the transport cc feedback or NACK status has changed on the
>   // preferred send codec, and in that case reconfigure all receive streams.
>@@ -1772,19 +1807,12 @@ bool WebRtcVoiceMediaChannel::AddSendStream(const StreamParams& sp) {
>     return false;
>   }
> 
>-  rtc::Optional<std::string> audio_network_adaptor_config =
>+  absl::optional<std::string> audio_network_adaptor_config =
>       GetAudioNetworkAdaptorConfig(options_);
>   WebRtcAudioSendStream* stream = new WebRtcAudioSendStream(
>-      ssrc,
>-      sp.cname,
>-      sp.id,
>-      send_codec_spec_,
>-      send_rtp_extensions_,
>-      max_send_bitrate_bps_,
>-      audio_network_adaptor_config,
>-      call_,
>-      this,
>-      engine()->encoder_factory_);
>+      ssrc, mid_, sp.cname, sp.id, send_codec_spec_, send_rtp_extensions_,
>+      max_send_bitrate_bps_, audio_network_adaptor_config, call_, this,
>+      engine()->encoder_factory_, codec_pair_id_);
>   send_streams_.insert(std::make_pair(ssrc, stream));
> 
>   // At this point the stream's local SSRC has been updated. If it is the first
>@@ -1834,6 +1862,13 @@ bool WebRtcVoiceMediaChannel::AddRecvStream(const StreamParams& sp) {
>   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>   RTC_LOG(LS_INFO) << "AddRecvStream: " << sp.ToString();
> 
>+  if (!sp.has_ssrcs()) {
>+    // This is a StreamParam with unsignaled SSRCs. Store it, so it can be used
>+    // later when we know the SSRCs on the first packet arrival.
>+    unsignaled_stream_params_ = sp;
>+    return true;
>+  }
>+
>   if (!ValidateStreamParams(sp)) {
>     return false;
>   }
>@@ -1845,9 +1880,9 @@ bool WebRtcVoiceMediaChannel::AddRecvStream(const StreamParams& sp) {
>   }
> 
>   // If this stream was previously received unsignaled, we promote it, possibly
>-  // recreating the AudioReceiveStream, if sync_label has changed.
>+  // recreating the AudioReceiveStream, if stream ids have changed.
>   if (MaybeDeregisterUnsignaledRecvStream(ssrc)) {
>-    recv_streams_[ssrc]->MaybeRecreateAudioReceiveStream(sp.sync_label);
>+    recv_streams_[ssrc]->MaybeRecreateAudioReceiveStream(sp.stream_ids());
>     return true;
>   }
> 
>@@ -1858,20 +1893,12 @@ bool WebRtcVoiceMediaChannel::AddRecvStream(const StreamParams& sp) {
> 
>   // Create a new channel for receiving audio data.
>   recv_streams_.insert(std::make_pair(
>-      ssrc,
>-      new WebRtcAudioReceiveStream(
>-          ssrc,
>-          receiver_reports_ssrc_,
>-          recv_transport_cc_enabled_,
>-          recv_nack_enabled_,
>-          sp.sync_label,
>-          recv_rtp_extensions_,
>-          call_,
>-          this,
>-          engine()->decoder_factory_,
>-          decoder_map_,
>-          engine()->audio_jitter_buffer_max_packets_,
>-          engine()->audio_jitter_buffer_fast_accelerate_)));
>+      ssrc, new WebRtcAudioReceiveStream(
>+                ssrc, receiver_reports_ssrc_, recv_transport_cc_enabled_,
>+                recv_nack_enabled_, sp.stream_ids(), recv_rtp_extensions_,
>+                call_, this, engine()->decoder_factory_, decoder_map_,
>+                codec_pair_id_, engine()->audio_jitter_buffer_max_packets_,
>+                engine()->audio_jitter_buffer_fast_accelerate_)));
>   recv_streams_[ssrc]->SetPlayout(playout_);
> 
>   return true;
>@@ -1882,6 +1909,13 @@ bool WebRtcVoiceMediaChannel::RemoveRecvStream(uint32_t ssrc) {
>   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>   RTC_LOG(LS_INFO) << "RemoveRecvStream: " << ssrc;
> 
>+  if (ssrc == 0) {
>+    // This indicates that we need to remove the unsignaled stream parameters
>+    // that are cached.
>+    unsignaled_stream_params_ = StreamParams();
>+    return true;
>+  }
>+
>   const auto it = recv_streams_.find(ssrc);
>   if (it == recv_streams_.end()) {
>     RTC_LOG(LS_WARNING) << "Try to remove stream with ssrc " << ssrc
>@@ -1920,30 +1954,6 @@ bool WebRtcVoiceMediaChannel::SetLocalSource(uint32_t ssrc,
>   return true;
> }
> 
>-// TODO(solenberg): Remove, once AudioMonitor is gone.
>-bool WebRtcVoiceMediaChannel::GetActiveStreams(
>-    AudioInfo::StreamList* actives) {
>-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  actives->clear();
>-  for (const auto& ch : recv_streams_) {
>-    int level = ch.second->GetOutputLevel();
>-    if (level > 0) {
>-      actives->push_back(std::make_pair(ch.first, level));
>-    }
>-  }
>-  return true;
>-}
>-
>-// TODO(solenberg): Remove, once AudioMonitor is gone.
>-int WebRtcVoiceMediaChannel::GetOutputLevel() {
>-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  int highest = 0;
>-  for (const auto& ch : recv_streams_) {
>-    highest = std::max(ch.second->GetOutputLevel(), highest);
>-  }
>-  return highest;
>-}
>-
> bool WebRtcVoiceMediaChannel::SetOutputVolume(uint32_t ssrc, double volume) {
>   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>   std::vector<uint32_t> ssrcs(1, ssrc);
>@@ -1966,14 +1976,15 @@ bool WebRtcVoiceMediaChannel::SetOutputVolume(uint32_t ssrc, double volume) {
> }
> 
> bool WebRtcVoiceMediaChannel::CanInsertDtmf() {
>-  return dtmf_payload_type_ ? true : false;
>+  return dtmf_payload_type_.has_value() && send_;
> }
> 
>-bool WebRtcVoiceMediaChannel::InsertDtmf(uint32_t ssrc, int event,
>+bool WebRtcVoiceMediaChannel::InsertDtmf(uint32_t ssrc,
>+                                         int event,
>                                          int duration) {
>   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>   RTC_LOG(LS_INFO) << "WebRtcVoiceMediaChannel::InsertDtmf";
>-  if (!dtmf_payload_type_) {
>+  if (!CanInsertDtmf()) {
>     return false;
>   }
> 
>@@ -1983,8 +1994,7 @@ bool WebRtcVoiceMediaChannel::InsertDtmf(uint32_t ssrc, int event,
>     RTC_LOG(LS_WARNING) << "The specified ssrc " << ssrc << " is not in use.";
>     return false;
>   }
>-  if (event < kMinTelephoneEventCode ||
>-      event > kMaxTelephoneEventCode) {
>+  if (event < kMinTelephoneEventCode || event > kMaxTelephoneEventCode) {
>     RTC_LOG(LS_WARNING) << "DTMF event code " << event << " out of range.";
>     return false;
>   }
>@@ -1994,14 +2004,13 @@ bool WebRtcVoiceMediaChannel::InsertDtmf(uint32_t ssrc, int event,
> }
> 
> void WebRtcVoiceMediaChannel::OnPacketReceived(
>-    rtc::CopyOnWriteBuffer* packet, const rtc::PacketTime& packet_time) {
>+    rtc::CopyOnWriteBuffer* packet,
>+    const rtc::PacketTime& packet_time) {
>   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
> 
>-  const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp,
>-                                              packet_time.not_before);
>   webrtc::PacketReceiver::DeliveryStatus delivery_result =
>       call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO, *packet,
>-                                       webrtc_packet_time);
>+                                       packet_time.timestamp);
>   if (delivery_result != webrtc::PacketReceiver::DELIVERY_UNKNOWN_SSRC) {
>     return;
>   }
>@@ -2014,10 +2023,11 @@ void WebRtcVoiceMediaChannel::OnPacketReceived(
>     return;
>   }
>   RTC_DCHECK(std::find(unsignaled_recv_ssrcs_.begin(),
>-      unsignaled_recv_ssrcs_.end(), ssrc) == unsignaled_recv_ssrcs_.end());
>+                       unsignaled_recv_ssrcs_.end(),
>+                       ssrc) == unsignaled_recv_ssrcs_.end());
> 
>   // Add new stream.
>-  StreamParams sp;
>+  StreamParams sp = unsignaled_stream_params_;
>   sp.ssrcs.push_back(ssrc);
>   RTC_LOG(LS_INFO) << "Creating unsignaled receive stream for SSRC=" << ssrc;
>   if (!AddRecvStream(sp)) {
>@@ -2025,9 +2035,8 @@ void WebRtcVoiceMediaChannel::OnPacketReceived(
>     return;
>   }
>   unsignaled_recv_ssrcs_.push_back(ssrc);
>-  RTC_HISTOGRAM_COUNTS_LINEAR(
>-      "WebRTC.Audio.NumOfUnsignaledStreams", unsignaled_recv_ssrcs_.size(), 1,
>-      100, 101);
>+  RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.NumOfUnsignaledStreams",
>+                              unsignaled_recv_ssrcs_.size(), 1, 100, 101);
> 
>   // Remove oldest unsignaled stream, if we have too many.
>   if (unsignaled_recv_ssrcs_.size() > kMaxUnsignaledRecvStreams) {
>@@ -2054,27 +2063,26 @@ void WebRtcVoiceMediaChannel::OnPacketReceived(
>   }
> 
>   delivery_result = call_->Receiver()->DeliverPacket(
>-      webrtc::MediaType::AUDIO, *packet, webrtc_packet_time);
>+      webrtc::MediaType::AUDIO, *packet, packet_time.timestamp);
>   RTC_DCHECK_NE(webrtc::PacketReceiver::DELIVERY_UNKNOWN_SSRC, delivery_result);
> }
> 
> void WebRtcVoiceMediaChannel::OnRtcpReceived(
>-    rtc::CopyOnWriteBuffer* packet, const rtc::PacketTime& packet_time) {
>+    rtc::CopyOnWriteBuffer* packet,
>+    const rtc::PacketTime& packet_time) {
>   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
> 
>   // Forward packet to Call as well.
>-  const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp,
>-                                              packet_time.not_before);
>   call_->Receiver()->DeliverPacket(webrtc::MediaType::AUDIO, *packet,
>-                                   webrtc_packet_time);
>+                                   packet_time.timestamp);
> }
> 
> void WebRtcVoiceMediaChannel::OnNetworkRouteChanged(
>     const std::string& transport_name,
>     const rtc::NetworkRoute& network_route) {
>   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  // TODO(zhihaung): Merge these two callbacks.
>-  call_->OnNetworkRouteChanged(transport_name, network_route);
>+  call_->GetTransportControllerSend()->OnNetworkRouteChanged(transport_name,
>+                                                             network_route);
>   call_->OnTransportOverheadChanged(webrtc::MediaType::AUDIO,
>                                     network_route.packet_overhead);
> }
>@@ -2260,12 +2268,11 @@ std::vector<webrtc::RtpSource> WebRtcVoiceMediaChannel::GetSources(
>   return it->second->GetSources();
> }
> 
>-bool WebRtcVoiceMediaChannel::
>-    MaybeDeregisterUnsignaledRecvStream(uint32_t ssrc) {
>+bool WebRtcVoiceMediaChannel::MaybeDeregisterUnsignaledRecvStream(
>+    uint32_t ssrc) {
>   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>   auto it = std::find(unsignaled_recv_ssrcs_.begin(),
>-                      unsignaled_recv_ssrcs_.end(),
>-                      ssrc);
>+                      unsignaled_recv_ssrcs_.end(), ssrc);
>   if (it != unsignaled_recv_ssrcs_.end()) {
>     unsignaled_recv_ssrcs_.erase(it);
>     return true;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvoiceengine.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvoiceengine.h
>index 47aba8a1e31..cd0c55c0118 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvoiceengine.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvoiceengine.h
>@@ -42,6 +42,7 @@ class WebRtcVoiceMediaChannel;
> // It uses the WebRtc VoiceEngine library for audio handling.
> class WebRtcVoiceEngine final {
>   friend class WebRtcVoiceMediaChannel;
>+
>  public:
>   WebRtcVoiceEngine(
>       webrtc::AudioDeviceModule* adm,
>@@ -59,8 +60,6 @@ class WebRtcVoiceEngine final {
>                                    const MediaConfig& config,
>                                    const AudioOptions& options);
> 
>-  int GetInputLevel();
>-
>   const std::vector<AudioCodec>& send_codecs() const;
>   const std::vector<AudioCodec>& recv_codecs() const;
>   RtpCapabilities GetCapabilities() const;
>@@ -122,15 +121,14 @@ class WebRtcVoiceEngine final {
> 
>   webrtc::AgcConfig default_agc_config_;
>   // Cache received extended_filter_aec, delay_agnostic_aec, experimental_ns
>-  // level controller, and intelligibility_enhancer values, and apply them
>+  // and intelligibility_enhancer values, and apply them
>   // in case they are missing in the audio options. We need to do this because
>   // SetExtraOptions() will revert to defaults for options which are not
>   // provided.
>-  rtc::Optional<bool> extended_filter_aec_;
>-  rtc::Optional<bool> delay_agnostic_aec_;
>-  rtc::Optional<bool> experimental_ns_;
>-  rtc::Optional<bool> intelligibility_enhancer_;
>-  rtc::Optional<bool> level_control_;
>+  absl::optional<bool> extended_filter_aec_;
>+  absl::optional<bool> delay_agnostic_aec_;
>+  absl::optional<bool> experimental_ns_;
>+  absl::optional<bool> intelligibility_enhancer_;
>   // Jitter buffer settings for new streams.
>   size_t audio_jitter_buffer_max_packets_ = 50;
>   bool audio_jitter_buffer_fast_accelerate_ = false;
>@@ -156,8 +154,9 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
>   bool SetSendParameters(const AudioSendParameters& params) override;
>   bool SetRecvParameters(const AudioRecvParameters& params) override;
>   webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const override;
>-  bool SetRtpSendParameters(uint32_t ssrc,
>-                            const webrtc::RtpParameters& parameters) override;
>+  webrtc::RTCError SetRtpSendParameters(
>+      uint32_t ssrc,
>+      const webrtc::RtpParameters& parameters) override;
>   webrtc::RtpParameters GetRtpReceiveParameters(uint32_t ssrc) const override;
>   bool SetRtpReceiveParameters(
>       uint32_t ssrc,
>@@ -173,8 +172,6 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
>   bool RemoveSendStream(uint32_t ssrc) override;
>   bool AddRecvStream(const StreamParams& sp) override;
>   bool RemoveRecvStream(uint32_t ssrc) override;
>-  bool GetActiveStreams(StreamList* actives) override;
>-  int GetOutputLevel() override;
>   // SSRC=0 will apply the new volume to current and future unsignaled streams.
>   bool SetOutputVolume(uint32_t ssrc, double volume) override;
> 
>@@ -210,7 +207,8 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
> 
>   bool SendRtcp(const uint8_t* data, size_t len) override {
>     rtc::CopyOnWriteBuffer packet(data, len, kMaxRtpPacketLen);
>-    return VoiceMediaChannel::SendRtcp(&packet, rtc::PacketOptions());
>+    rtc::PacketOptions rtc_options;
>+    return VoiceMediaChannel::SendRtcp(&packet, rtc_options);
>   }
> 
>  private:
>@@ -225,7 +223,8 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
>   int CreateVoEChannel();
>   bool DeleteVoEChannel(int channel);
>   bool SetMaxSendBitrate(int bps);
>-  bool ValidateRtpParameters(const webrtc::RtpParameters& parameters);
>+  webrtc::RTCError ValidateRtpParameters(
>+      const webrtc::RtpParameters& parameters);
>   void SetupRecording();
>   // Check if 'ssrc' is an unsignaled stream, and if so mark it as not being
>   // unsignaled anymore (i.e. it is now removed, or signaled), and return true.
>@@ -243,7 +242,7 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
> 
>   int max_send_bitrate_bps_ = 0;
>   AudioOptions options_;
>-  rtc::Optional<int> dtmf_payload_type_;
>+  absl::optional<int> dtmf_payload_type_;
>   int dtmf_payload_freq_ = -1;
>   bool recv_transport_cc_enabled_ = false;
>   bool recv_nack_enabled_ = false;
>@@ -255,6 +254,12 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
>   // Queue of unsignaled SSRCs; oldest at the beginning.
>   std::vector<uint32_t> unsignaled_recv_ssrcs_;
> 
>+  // This is a stream param that comes from the remote description, but wasn't
>+  // signaled with any a=ssrc lines. It holds the information that was signaled
>+  // before the unsignaled receive stream is created when the first packet is
>+  // received.
>+  StreamParams unsignaled_stream_params_;
>+
>   // Volume for unsignaled streams, which may be set before the stream exists.
>   double default_recv_volume_ = 1.0;
>   // Sink for latest unsignaled stream - may be set before the stream exists.
>@@ -267,14 +272,19 @@ class WebRtcVoiceMediaChannel final : public VoiceMediaChannel,
>   class WebRtcAudioSendStream;
>   std::map<uint32_t, WebRtcAudioSendStream*> send_streams_;
>   std::vector<webrtc::RtpExtension> send_rtp_extensions_;
>+  std::string mid_;
> 
>   class WebRtcAudioReceiveStream;
>   std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_;
>   std::vector<webrtc::RtpExtension> recv_rtp_extensions_;
> 
>-  rtc::Optional<webrtc::AudioSendStream::Config::SendCodecSpec>
>+  absl::optional<webrtc::AudioSendStream::Config::SendCodecSpec>
>       send_codec_spec_;
> 
>+  // TODO(kwiberg): Per-SSRC codec pair IDs?
>+  const webrtc::AudioCodecPairId codec_pair_id_ =
>+      webrtc::AudioCodecPairId::Create();
>+
>   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel);
> };
> }  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvoiceengine_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvoiceengine_unittest.cc
>index 9b56dbf8267..4e20dfc16a2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvoiceengine_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/webrtcvoiceengine_unittest.cc
>@@ -36,12 +36,14 @@
> 
> using testing::_;
> using testing::ContainerEq;
>+using testing::Field;
> using testing::Return;
> using testing::ReturnPointee;
> using testing::SaveArg;
> using testing::StrictMock;
> 
> namespace {
>+using webrtc::BitrateConstraints;
> 
> constexpr uint32_t kMaxUnsignaledRecvStreams = 4;
> 
>@@ -52,10 +54,16 @@ const cricket::AudioCodec kG722CodecVoE(9, "G722", 16000, 64000, 1);
> const cricket::AudioCodec kG722CodecSdp(9, "G722", 8000, 64000, 1);
> const cricket::AudioCodec kCn8000Codec(13, "CN", 8000, 0, 1);
> const cricket::AudioCodec kCn16000Codec(105, "CN", 16000, 0, 1);
>-const cricket::AudioCodec
>-    kTelephoneEventCodec1(106, "telephone-event", 8000, 0, 1);
>-const cricket::AudioCodec
>-    kTelephoneEventCodec2(107, "telephone-event", 32000, 0, 1);
>+const cricket::AudioCodec kTelephoneEventCodec1(106,
>+                                                "telephone-event",
>+                                                8000,
>+                                                0,
>+                                                1);
>+const cricket::AudioCodec kTelephoneEventCodec2(107,
>+                                                "telephone-event",
>+                                                32000,
>+                                                0,
>+                                                1);
> 
> const uint32_t kSsrc0 = 0;
> const uint32_t kSsrc1 = 1;
>@@ -63,7 +71,7 @@ const uint32_t kSsrcX = 0x99;
> const uint32_t kSsrcY = 0x17;
> const uint32_t kSsrcZ = 0x42;
> const uint32_t kSsrcW = 0x02;
>-const uint32_t kSsrcs4[] = { 11, 200, 30, 44 };
>+const uint32_t kSsrcs4[] = {11, 200, 30, 44};
> 
> constexpr int kRtpHistoryMs = 5000;
> 
>@@ -85,10 +93,11 @@ void AdmSetupExpectations(webrtc::test::MockAudioDeviceModule* adm) {
>   EXPECT_CALL(*adm, Init()).WillOnce(Return(0));
>   EXPECT_CALL(*adm, RegisterAudioCallback(_)).WillOnce(Return(0));
> #if defined(WEBRTC_WIN)
>-  EXPECT_CALL(*adm, SetPlayoutDevice(
>-      testing::Matcher<webrtc::AudioDeviceModule::WindowsDeviceType>(
>-          webrtc::AudioDeviceModule::kDefaultCommunicationDevice)))
>-              .WillOnce(Return(0));
>+  EXPECT_CALL(
>+      *adm, SetPlayoutDevice(
>+                testing::Matcher<webrtc::AudioDeviceModule::WindowsDeviceType>(
>+                    webrtc::AudioDeviceModule::kDefaultCommunicationDevice)))
>+      .WillOnce(Return(0));
> #else
>   EXPECT_CALL(*adm, SetPlayoutDevice(0)).WillOnce(Return(0));
> #endif  // #if defined(WEBRTC_WIN)
>@@ -96,10 +105,11 @@ void AdmSetupExpectations(webrtc::test::MockAudioDeviceModule* adm) {
>   EXPECT_CALL(*adm, StereoPlayoutIsAvailable(testing::_)).WillOnce(Return(0));
>   EXPECT_CALL(*adm, SetStereoPlayout(false)).WillOnce(Return(0));
> #if defined(WEBRTC_WIN)
>-  EXPECT_CALL(*adm, SetRecordingDevice(
>-      testing::Matcher<webrtc::AudioDeviceModule::WindowsDeviceType>(
>-          webrtc::AudioDeviceModule::kDefaultCommunicationDevice)))
>-              .WillOnce(Return(0));
>+  EXPECT_CALL(
>+      *adm, SetRecordingDevice(
>+                testing::Matcher<webrtc::AudioDeviceModule::WindowsDeviceType>(
>+                    webrtc::AudioDeviceModule::kDefaultCommunicationDevice)))
>+      .WillOnce(Return(0));
> #else
>   EXPECT_CALL(*adm, SetRecordingDevice(0)).WillOnce(Return(0));
> #endif  // #if defined(WEBRTC_WIN)
>@@ -115,7 +125,8 @@ void AdmSetupExpectations(webrtc::test::MockAudioDeviceModule* adm) {
>   EXPECT_CALL(*adm, StopRecording()).WillOnce(Return(0));
>   EXPECT_CALL(*adm, RegisterAudioCallback(nullptr)).WillOnce(Return(0));
>   EXPECT_CALL(*adm, Terminate()).WillOnce(Return(0));
>-  EXPECT_CALL(*adm, Release()).Times(3)
>+  EXPECT_CALL(*adm, Release())
>+      .Times(3)
>       .WillRepeatedly(Return(rtc::RefCountReleaseStatus::kDroppedLastRef));
> }
> }  // namespace
>@@ -160,7 +171,7 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
>         apm_ec_(*apm_->echo_cancellation()),
>         apm_ns_(*apm_->noise_suppression()),
>         apm_vd_(*apm_->voice_detection()),
>-        call_(webrtc::Call::Config(&event_log_)),
>+        call_(),
>         override_field_trials_(field_trials) {
>     // AudioDeviceModule.
>     AdmSetupExpectations(&adm_);
>@@ -215,10 +226,14 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
>   }
> 
>   bool SetupSendStream() {
>+    return SetupSendStream(cricket::StreamParams::CreateLegacy(kSsrcX));
>+  }
>+
>+  bool SetupSendStream(const cricket::StreamParams& sp) {
>     if (!SetupChannel()) {
>       return false;
>     }
>-    if (!channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcX))) {
>+    if (!channel_->AddSendStream(sp)) {
>       return false;
>     }
>     EXPECT_CALL(*apm_, set_output_will_be_muted(false));
>@@ -244,9 +259,7 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
>     channel_->OnPacketReceived(&packet, rtc::PacketTime());
>   }
> 
>-  void TearDown() override {
>-    delete channel_;
>-  }
>+  void TearDown() override { delete channel_; }
> 
>   const cricket::FakeAudioSendStream& GetSendStream(uint32_t ssrc) {
>     const auto* send_stream = call_.GetAudioSendStream(ssrc);
>@@ -285,7 +298,9 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
>     EXPECT_TRUE(channel_->SetSendParameters(params));
>   }
> 
>-  void SetAudioSend(uint32_t ssrc, bool enable, cricket::AudioSource* source,
>+  void SetAudioSend(uint32_t ssrc,
>+                    bool enable,
>+                    cricket::AudioSource* source,
>                     const cricket::AudioOptions* options = nullptr) {
>     EXPECT_CALL(*apm_, set_output_will_be_muted(!enable));
>     ASSERT_TRUE(channel_);
>@@ -295,14 +310,15 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
>     EXPECT_TRUE(channel_->SetAudioSend(ssrc, enable, options, source));
>   }
> 
>-  void TestInsertDtmf(uint32_t ssrc, bool caller,
>+  void TestInsertDtmf(uint32_t ssrc,
>+                      bool caller,
>                       const cricket::AudioCodec& codec) {
>     EXPECT_TRUE(SetupChannel());
>     if (caller) {
>       // If this is a caller, local description will be applied and add the
>       // send stream.
>-      EXPECT_TRUE(channel_->AddSendStream(
>-          cricket::StreamParams::CreateLegacy(kSsrcX)));
>+      EXPECT_TRUE(
>+          channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcX)));
>     }
> 
>     // Test we can only InsertDtmf when the other side supports telephone-event.
>@@ -317,8 +333,8 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
>     if (!caller) {
>       // If this is callee, there's no active send channel yet.
>       EXPECT_FALSE(channel_->InsertDtmf(ssrc, 2, 123));
>-      EXPECT_TRUE(channel_->AddSendStream(
>-          cricket::StreamParams::CreateLegacy(kSsrcX)));
>+      EXPECT_TRUE(
>+          channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcX)));
>     }
> 
>     // Check we fail if the ssrc is invalid.
>@@ -362,7 +378,7 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
>     EXPECT_EQ(1UL, parameters.encodings.size());
> 
>     parameters.encodings[0].max_bitrate_bps = bitrate;
>-    return channel_->SetRtpSendParameters(ssrc, parameters);
>+    return channel_->SetRtpSendParameters(ssrc, parameters).ok();
>   }
> 
>   void SetGlobalMaxBitrate(const cricket::AudioCodec& codec, int bitrate) {
>@@ -380,11 +396,12 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
>     EXPECT_EQ(expected_bitrate, spec->target_bitrate_bps);
>   }
> 
>-  rtc::Optional<int> GetCodecBitrate(int32_t ssrc) {
>+  absl::optional<int> GetCodecBitrate(int32_t ssrc) {
>     return GetSendStreamConfig(ssrc).send_codec_spec->target_bitrate_bps;
>   }
> 
>-  const rtc::Optional<std::string>& GetAudioNetworkAdaptorConfig(int32_t ssrc) {
>+  const absl::optional<std::string>& GetAudioNetworkAdaptorConfig(
>+      int32_t ssrc) {
>     return GetSendStreamConfig(ssrc).audio_network_adaptor_config;
>   }
> 
>@@ -425,14 +442,16 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
>     codecs[0].params[cricket::kCodecParamMinBitrate] = min_bitrate_kbps;
>     codecs[0].params[cricket::kCodecParamStartBitrate] = start_bitrate_kbps;
>     codecs[0].params[cricket::kCodecParamMaxBitrate] = max_bitrate_kbps;
>-    SetSendParameters(send_parameters_);
>+    EXPECT_CALL(*call_.GetMockTransportControllerSend(),
>+                SetSdpBitrateParameters(
>+                    AllOf(Field(&BitrateConstraints::min_bitrate_bps,
>+                                expected_min_bitrate_bps),
>+                          Field(&BitrateConstraints::start_bitrate_bps,
>+                                expected_start_bitrate_bps),
>+                          Field(&BitrateConstraints::max_bitrate_bps,
>+                                expected_max_bitrate_bps))));
> 
>-    EXPECT_EQ(expected_min_bitrate_bps,
>-              call_.GetConfig().bitrate_config.min_bitrate_bps);
>-    EXPECT_EQ(expected_start_bitrate_bps,
>-              call_.GetConfig().bitrate_config.start_bitrate_bps);
>-    EXPECT_EQ(expected_max_bitrate_bps,
>-              call_.GetConfig().bitrate_config.max_bitrate_bps);
>+    SetSendParameters(send_parameters_);
>   }
> 
>   void TestSetSendRtpHeaderExtensions(const std::string& ext) {
>@@ -461,8 +480,8 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
>     EXPECT_EQ(id, GetSendStreamConfig(kSsrcX).rtp.extensions[0].id);
> 
>     // Ensure extension is set properly on new stream.
>-    EXPECT_TRUE(channel_->AddSendStream(
>-        cricket::StreamParams::CreateLegacy(kSsrcY)));
>+    EXPECT_TRUE(
>+        channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcY)));
>     EXPECT_NE(call_.GetAudioSendStream(kSsrcX),
>               call_.GetAudioSendStream(kSsrcY));
>     EXPECT_EQ(1u, GetSendStreamConfig(kSsrcY).rtp.extensions.size());
>@@ -639,17 +658,22 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
>     const auto stats = GetAudioReceiveStreamStats();
>     EXPECT_EQ(info.ssrc(), stats.remote_ssrc);
>     EXPECT_EQ(info.bytes_rcvd, stats.bytes_rcvd);
>-    EXPECT_EQ(info.packets_rcvd, stats.packets_rcvd);
>-    EXPECT_EQ(info.packets_lost, stats.packets_lost);
>+    EXPECT_EQ(rtc::checked_cast<unsigned int>(info.packets_rcvd),
>+              stats.packets_rcvd);
>+    EXPECT_EQ(rtc::checked_cast<unsigned int>(info.packets_lost),
>+              stats.packets_lost);
>     EXPECT_EQ(info.fraction_lost, stats.fraction_lost);
>     EXPECT_EQ(info.codec_name, stats.codec_name);
>     EXPECT_EQ(info.codec_payload_type, stats.codec_payload_type);
>-    EXPECT_EQ(info.ext_seqnum, stats.ext_seqnum);
>-    EXPECT_EQ(info.jitter_ms, stats.jitter_ms);
>-    EXPECT_EQ(info.jitter_buffer_ms, stats.jitter_buffer_ms);
>-    EXPECT_EQ(info.jitter_buffer_preferred_ms,
>+    EXPECT_EQ(rtc::checked_cast<unsigned int>(info.ext_seqnum),
>+              stats.ext_seqnum);
>+    EXPECT_EQ(rtc::checked_cast<unsigned int>(info.jitter_ms), stats.jitter_ms);
>+    EXPECT_EQ(rtc::checked_cast<unsigned int>(info.jitter_buffer_ms),
>+              stats.jitter_buffer_ms);
>+    EXPECT_EQ(rtc::checked_cast<unsigned int>(info.jitter_buffer_preferred_ms),
>               stats.jitter_buffer_preferred_ms);
>-    EXPECT_EQ(info.delay_estimate_ms, stats.delay_estimate_ms);
>+    EXPECT_EQ(rtc::checked_cast<unsigned int>(info.delay_estimate_ms),
>+              stats.delay_estimate_ms);
>     EXPECT_EQ(info.audio_level, stats.audio_level);
>     EXPECT_EQ(info.total_samples_received, stats.total_samples_received);
>     EXPECT_EQ(info.concealed_samples, stats.concealed_samples);
>@@ -698,7 +722,6 @@ class WebRtcVoiceEngineTestFake : public testing::Test {
>   webrtc::test::MockEchoCancellation& apm_ec_;
>   webrtc::test::MockNoiseSuppression& apm_ns_;
>   webrtc::test::MockVoiceDetection& apm_vd_;
>-  webrtc::RtcEventLogNullImpl event_log_;
>   cricket::FakeCall call_;
>   std::unique_ptr<cricket::WebRtcVoiceEngine> engine_;
>   cricket::VoiceMediaChannel* channel_ = nullptr;
>@@ -875,7 +898,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SetRecvCodecsAfterAddingStreams) {
>   EXPECT_TRUE(channel_->SetRecvParameters(parameters));
> 
>   const auto& dm = GetRecvStreamConfig(kSsrcX).decoder_map;
>-  ASSERT_EQ(1, dm.count(106));
>+  ASSERT_EQ(1u, dm.count(106));
>   EXPECT_EQ(webrtc::SdpAudioFormat("isac", 16000, 1), dm.at(106));
> }
> 
>@@ -978,8 +1001,8 @@ TEST_F(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthMultiRateAsCallee) {
>   parameters.max_bandwidth_bps = kDesiredBitrate;
>   SetSendParameters(parameters);
> 
>-  EXPECT_TRUE(channel_->AddSendStream(
>-      cricket::StreamParams::CreateLegacy(kSsrcX)));
>+  EXPECT_TRUE(
>+      channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcX)));
> 
>   EXPECT_EQ(kDesiredBitrate, GetCodecBitrate(kSsrcX));
> }
>@@ -1031,10 +1054,11 @@ TEST_F(WebRtcVoiceEngineTestFake, CannotSetMaxBitrateForNonexistentStream) {
>   EXPECT_TRUE(SetupChannel());
>   webrtc::RtpParameters nonexistent_parameters =
>       channel_->GetRtpSendParameters(kSsrcX);
>-  EXPECT_EQ(0, nonexistent_parameters.encodings.size());
>+  EXPECT_EQ(0u, nonexistent_parameters.encodings.size());
> 
>   nonexistent_parameters.encodings.push_back(webrtc::RtpEncodingParameters());
>-  EXPECT_FALSE(channel_->SetRtpSendParameters(kSsrcX, nonexistent_parameters));
>+  EXPECT_FALSE(
>+      channel_->SetRtpSendParameters(kSsrcX, nonexistent_parameters).ok());
> }
> 
> TEST_F(WebRtcVoiceEngineTestFake,
>@@ -1048,10 +1072,10 @@ TEST_F(WebRtcVoiceEngineTestFake,
>   webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(kSsrcX);
>   // Two or more encodings should result in failure.
>   parameters.encodings.push_back(webrtc::RtpEncodingParameters());
>-  EXPECT_FALSE(channel_->SetRtpSendParameters(kSsrcX, parameters));
>+  EXPECT_FALSE(channel_->SetRtpSendParameters(kSsrcX, parameters).ok());
>   // Zero encodings should also fail.
>   parameters.encodings.clear();
>-  EXPECT_FALSE(channel_->SetRtpSendParameters(kSsrcX, parameters));
>+  EXPECT_FALSE(channel_->SetRtpSendParameters(kSsrcX, parameters).ok());
> }
> 
> // Changing the SSRC through RtpParameters is not allowed.
>@@ -1059,7 +1083,7 @@ TEST_F(WebRtcVoiceEngineTestFake, CannotSetSsrcInRtpSendParameters) {
>   EXPECT_TRUE(SetupSendStream());
>   webrtc::RtpParameters parameters = channel_->GetRtpSendParameters(kSsrcX);
>   parameters.encodings[0].ssrc = 0xdeadbeef;
>-  EXPECT_FALSE(channel_->SetRtpSendParameters(kSsrcX, parameters));
>+  EXPECT_FALSE(channel_->SetRtpSendParameters(kSsrcX, parameters).ok());
> }
> 
> // Test that a stream will not be sending if its encoding is made
>@@ -1073,14 +1097,14 @@ TEST_F(WebRtcVoiceEngineTestFake, SetRtpParametersEncodingsActive) {
>   ASSERT_EQ(1u, parameters.encodings.size());
>   ASSERT_TRUE(parameters.encodings[0].active);
>   parameters.encodings[0].active = false;
>-  EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrcX, parameters));
>+  EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrcX, parameters).ok());
>   EXPECT_FALSE(GetSendStream(kSsrcX).IsSending());
> 
>   // Now change it back to active and verify we resume sending.
>   // This should occur even when other parameters are updated.
>   parameters.encodings[0].active = true;
>-  parameters.encodings[0].max_bitrate_bps = rtc::Optional<int>(6000);
>-  EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrcX, parameters));
>+  parameters.encodings[0].max_bitrate_bps = absl::optional<int>(6000);
>+  EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrcX, parameters).ok());
>   EXPECT_TRUE(GetSendStream(kSsrcX).IsSending());
> }
> 
>@@ -1127,6 +1151,30 @@ TEST_F(WebRtcVoiceEngineTestFake, GetRtpSendParametersCodecs) {
>   EXPECT_EQ(kPcmuCodec.ToCodecParameters(), rtp_parameters.codecs[1]);
> }
> 
>+// Test that GetRtpSendParameters returns the currently configured RTCP CNAME.
>+TEST_F(WebRtcVoiceEngineTestFake, GetRtpSendParametersRtcpCname) {
>+  cricket::StreamParams params = cricket::StreamParams::CreateLegacy(kSsrcX);
>+  params.cname = "rtcpcname";
>+  EXPECT_TRUE(SetupSendStream(params));
>+
>+  webrtc::RtpParameters rtp_parameters = channel_->GetRtpSendParameters(kSsrcX);
>+  EXPECT_STREQ("rtcpcname", rtp_parameters.rtcp.cname.c_str());
>+}
>+
>+TEST_F(WebRtcVoiceEngineTestFake,
>+       DetectRtpSendParameterHeaderExtensionsChange) {
>+  EXPECT_TRUE(SetupSendStream());
>+
>+  webrtc::RtpParameters rtp_parameters = channel_->GetRtpSendParameters(kSsrcX);
>+  rtp_parameters.header_extensions.emplace_back();
>+
>+  EXPECT_NE(0u, rtp_parameters.header_extensions.size());
>+
>+  webrtc::RTCError result =
>+      channel_->SetRtpSendParameters(kSsrcX, rtp_parameters);
>+  EXPECT_EQ(webrtc::RTCErrorType::INVALID_MODIFICATION, result.type());
>+}
>+
> // Test that GetRtpSendParameters returns an SSRC.
> TEST_F(WebRtcVoiceEngineTestFake, GetRtpSendParametersSsrc) {
>   EXPECT_TRUE(SetupSendStream());
>@@ -1146,7 +1194,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SetAndGetRtpSendParameters) {
>   webrtc::RtpParameters initial_params = channel_->GetRtpSendParameters(kSsrcX);
> 
>   // We should be able to set the params we just got.
>-  EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrcX, initial_params));
>+  EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrcX, initial_params).ok());
> 
>   // ... And this shouldn't change the params returned by GetRtpSendParameters.
>   webrtc::RtpParameters new_params = channel_->GetRtpSendParameters(kSsrcX);
>@@ -1169,7 +1217,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SetRtpSendParameterUpdatesMaxBitrate) {
> 
>   constexpr int kMaxBitrateBps = 6000;
>   rtp_parameters.encodings[0].max_bitrate_bps = kMaxBitrateBps;
>-  EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrcX, rtp_parameters));
>+  EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrcX, rtp_parameters).ok());
> 
>   const int max_bitrate = GetSendStreamConfig(kSsrcX).max_bitrate_bps;
>   EXPECT_EQ(max_bitrate, kMaxBitrateBps);
>@@ -1185,9 +1233,9 @@ TEST_F(WebRtcVoiceEngineTestFake, SetRtpSendParameterInvalidBitratePriority) {
>             rtp_parameters.encodings[0].bitrate_priority);
> 
>   rtp_parameters.encodings[0].bitrate_priority = 0;
>-  EXPECT_FALSE(channel_->SetRtpSendParameters(kSsrcX, rtp_parameters));
>+  EXPECT_FALSE(channel_->SetRtpSendParameters(kSsrcX, rtp_parameters).ok());
>   rtp_parameters.encodings[0].bitrate_priority = -1.0;
>-  EXPECT_FALSE(channel_->SetRtpSendParameters(kSsrcX, rtp_parameters));
>+  EXPECT_FALSE(channel_->SetRtpSendParameters(kSsrcX, rtp_parameters).ok());
> }
> 
> // Test that the bitrate_priority in the send stream config gets updated when
>@@ -1201,7 +1249,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SetRtpSendParameterUpdatesBitratePriority) {
>             rtp_parameters.encodings[0].bitrate_priority);
>   double new_bitrate_priority = 2.0;
>   rtp_parameters.encodings[0].bitrate_priority = new_bitrate_priority;
>-  EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrcX, rtp_parameters));
>+  EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrcX, rtp_parameters).ok());
> 
>   // The priority should get set for both the audio channel's rtp parameters
>   // and the audio send stream's audio config.
>@@ -1307,7 +1355,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecs) {
>   EXPECT_EQ(22000, send_codec_spec.target_bitrate_bps);
>   EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str());
>   EXPECT_NE(send_codec_spec.format.clockrate_hz, 8000);
>-  EXPECT_EQ(rtc::nullopt, send_codec_spec.cng_payload_type);
>+  EXPECT_EQ(absl::nullopt, send_codec_spec.cng_payload_type);
>   EXPECT_FALSE(channel_->CanInsertDtmf());
> }
> 
>@@ -1471,7 +1519,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecOpusGoodNBitrateNoStereo) {
>   EXPECT_EQ(111, spec.payload_type);
>   EXPECT_EQ(96000, spec.target_bitrate_bps);
>   EXPECT_EQ("opus", spec.format.name);
>-  EXPECT_EQ(2, spec.format.num_channels);
>+  EXPECT_EQ(2u, spec.format.num_channels);
>   EXPECT_EQ(48000, spec.format.clockrate_hz);
> }
> 
>@@ -1525,17 +1573,18 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCapsMinAndStartBitrate) {
>   SetSendCodecsShouldWorkForBitrates("-1", 0, "-100", -1, "", -1);
> }
> 
>-TEST_F(WebRtcVoiceEngineTestFake,
>-       SetMaxSendBandwidthForAudioDoesntAffectBwe) {
>+TEST_F(WebRtcVoiceEngineTestFake, SetMaxSendBandwidthForAudioDoesntAffectBwe) {
>   SetSendCodecsShouldWorkForBitrates("100", 100000, "150", 150000, "200",
>                                      200000);
>   send_parameters_.max_bandwidth_bps = 100000;
>+  // Setting max bitrate should keep previous min bitrate
>+  // Setting max bitrate should not reset start bitrate.
>+  EXPECT_CALL(*call_.GetMockTransportControllerSend(),
>+              SetSdpBitrateParameters(
>+                  AllOf(Field(&BitrateConstraints::min_bitrate_bps, 100000),
>+                        Field(&BitrateConstraints::start_bitrate_bps, -1),
>+                        Field(&BitrateConstraints::max_bitrate_bps, 200000))));
>   SetSendParameters(send_parameters_);
>-  EXPECT_EQ(100000, call_.GetConfig().bitrate_config.min_bitrate_bps)
>-      << "Setting max bitrate should keep previous min bitrate.";
>-  EXPECT_EQ(-1, call_.GetConfig().bitrate_config.start_bitrate_bps)
>-      << "Setting max bitrate should not reset start bitrate.";
>-  EXPECT_EQ(200000, call_.GetConfig().bitrate_config.max_bitrate_bps);
> }
> 
> // Test that we can enable NACK with opus as caller.
>@@ -1543,9 +1592,8 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecEnableNackAsCaller) {
>   EXPECT_TRUE(SetupSendStream());
>   cricket::AudioSendParameters parameters;
>   parameters.codecs.push_back(kOpusCodec);
>-  parameters.codecs[0].AddFeedbackParam(
>-      cricket::FeedbackParam(cricket::kRtcpFbParamNack,
>-                             cricket::kParamValueEmpty));
>+  parameters.codecs[0].AddFeedbackParam(cricket::FeedbackParam(
>+      cricket::kRtcpFbParamNack, cricket::kParamValueEmpty));
>   EXPECT_EQ(0, GetSendStreamConfig(kSsrcX).rtp.nack.rtp_history_ms);
>   SetSendParameters(parameters);
>   EXPECT_EQ(kRtpHistoryMs, GetSendStreamConfig(kSsrcX).rtp.nack.rtp_history_ms);
>@@ -1556,16 +1604,15 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecEnableNackAsCallee) {
>   EXPECT_TRUE(SetupRecvStream());
>   cricket::AudioSendParameters parameters;
>   parameters.codecs.push_back(kOpusCodec);
>-  parameters.codecs[0].AddFeedbackParam(
>-      cricket::FeedbackParam(cricket::kRtcpFbParamNack,
>-                             cricket::kParamValueEmpty));
>+  parameters.codecs[0].AddFeedbackParam(cricket::FeedbackParam(
>+      cricket::kRtcpFbParamNack, cricket::kParamValueEmpty));
>   EXPECT_EQ(0, GetRecvStreamConfig(kSsrcX).rtp.nack.rtp_history_ms);
>   SetSendParameters(parameters);
>   // NACK should be enabled even with no send stream.
>   EXPECT_EQ(kRtpHistoryMs, GetRecvStreamConfig(kSsrcX).rtp.nack.rtp_history_ms);
> 
>-  EXPECT_TRUE(channel_->AddSendStream(
>-      cricket::StreamParams::CreateLegacy(kSsrcX)));
>+  EXPECT_TRUE(
>+      channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcX)));
>   EXPECT_EQ(kRtpHistoryMs, GetSendStreamConfig(kSsrcX).rtp.nack.rtp_history_ms);
> }
> 
>@@ -1575,9 +1622,8 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecEnableNackRecvStreams) {
>   EXPECT_TRUE(AddRecvStream(kSsrcY));
>   cricket::AudioSendParameters parameters;
>   parameters.codecs.push_back(kOpusCodec);
>-  parameters.codecs[0].AddFeedbackParam(
>-      cricket::FeedbackParam(cricket::kRtcpFbParamNack,
>-                             cricket::kParamValueEmpty));
>+  parameters.codecs[0].AddFeedbackParam(cricket::FeedbackParam(
>+      cricket::kRtcpFbParamNack, cricket::kParamValueEmpty));
>   EXPECT_EQ(0, GetSendStreamConfig(kSsrcX).rtp.nack.rtp_history_ms);
>   EXPECT_EQ(0, GetRecvStreamConfig(kSsrcY).rtp.nack.rtp_history_ms);
>   SetSendParameters(parameters);
>@@ -1590,9 +1636,8 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecDisableNack) {
>   EXPECT_TRUE(SetupSendStream());
>   cricket::AudioSendParameters parameters;
>   parameters.codecs.push_back(kOpusCodec);
>-  parameters.codecs[0].AddFeedbackParam(
>-      cricket::FeedbackParam(cricket::kRtcpFbParamNack,
>-                             cricket::kParamValueEmpty));
>+  parameters.codecs[0].AddFeedbackParam(cricket::FeedbackParam(
>+      cricket::kRtcpFbParamNack, cricket::kParamValueEmpty));
>   SetSendParameters(parameters);
>   EXPECT_EQ(kRtpHistoryMs, GetSendStreamConfig(kSsrcX).rtp.nack.rtp_history_ms);
> 
>@@ -1608,9 +1653,8 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecDisableNackRecvStreams) {
>   EXPECT_TRUE(AddRecvStream(kSsrcY));
>   cricket::AudioSendParameters parameters;
>   parameters.codecs.push_back(kOpusCodec);
>-  parameters.codecs[0].AddFeedbackParam(
>-      cricket::FeedbackParam(cricket::kRtcpFbParamNack,
>-                             cricket::kParamValueEmpty));
>+  parameters.codecs[0].AddFeedbackParam(cricket::FeedbackParam(
>+      cricket::kRtcpFbParamNack, cricket::kParamValueEmpty));
>   SetSendParameters(parameters);
>   EXPECT_EQ(kRtpHistoryMs, GetSendStreamConfig(kSsrcX).rtp.nack.rtp_history_ms);
>   EXPECT_EQ(kRtpHistoryMs, GetRecvStreamConfig(kSsrcY).rtp.nack.rtp_history_ms);
>@@ -1628,9 +1672,8 @@ TEST_F(WebRtcVoiceEngineTestFake, AddRecvStreamEnableNack) {
>   cricket::AudioSendParameters parameters;
>   parameters.codecs.push_back(kIsacCodec);
>   parameters.codecs.push_back(kCn16000Codec);
>-  parameters.codecs[0].AddFeedbackParam(
>-      cricket::FeedbackParam(cricket::kRtcpFbParamNack,
>-                             cricket::kParamValueEmpty));
>+  parameters.codecs[0].AddFeedbackParam(cricket::FeedbackParam(
>+      cricket::kRtcpFbParamNack, cricket::kParamValueEmpty));
>   SetSendParameters(parameters);
>   EXPECT_EQ(kRtpHistoryMs, GetSendStreamConfig(kSsrcX).rtp.nack.rtp_history_ms);
> 
>@@ -1707,7 +1750,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsBitrate) {
>     EXPECT_EQ(32000, spec.target_bitrate_bps);
>   }
> 
>-  parameters.codecs[0].bitrate = 0;         // bitrate == default
>+  parameters.codecs[0].bitrate = 0;  // bitrate == default
>   SetSendParameters(parameters);
>   {
>     const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
>@@ -1715,7 +1758,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsBitrate) {
>     EXPECT_STRCASEEQ("ISAC", spec.format.name.c_str());
>     EXPECT_EQ(32000, spec.target_bitrate_bps);
>   }
>-  parameters.codecs[0].bitrate = 28000;     // bitrate == 28000
>+  parameters.codecs[0].bitrate = 28000;  // bitrate == 28000
>   SetSendParameters(parameters);
>   {
>     const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
>@@ -1724,7 +1767,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsBitrate) {
>     EXPECT_EQ(28000, spec.target_bitrate_bps);
>   }
> 
>-  parameters.codecs[0] = kPcmuCodec;        // bitrate == 64000
>+  parameters.codecs[0] = kPcmuCodec;  // bitrate == 64000
>   SetSendParameters(parameters);
>   {
>     const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
>@@ -1733,7 +1776,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsBitrate) {
>     EXPECT_EQ(64000, spec.target_bitrate_bps);
>   }
> 
>-  parameters.codecs[0].bitrate = 0;         // bitrate == default
>+  parameters.codecs[0].bitrate = 0;  // bitrate == default
>   SetSendParameters(parameters);
>   {
>     const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
>@@ -1743,7 +1786,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsBitrate) {
>   }
> 
>   parameters.codecs[0] = kOpusCodec;
>-  parameters.codecs[0].bitrate = 0;         // bitrate == default
>+  parameters.codecs[0].bitrate = 0;  // bitrate == default
>   SetSendParameters(parameters);
>   {
>     const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
>@@ -1774,7 +1817,24 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsDTMFOnTop) {
>   const auto& spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
>   EXPECT_EQ(96, spec.payload_type);
>   EXPECT_STRCASEEQ("ISAC", spec.format.name.c_str());
>+  SetSend(true);
>+  EXPECT_TRUE(channel_->CanInsertDtmf());
>+}
>+
>+// Test that CanInsertDtmf() is governed by the send flag
>+TEST_F(WebRtcVoiceEngineTestFake, DTMFControlledBySendFlag) {
>+  EXPECT_TRUE(SetupSendStream());
>+  cricket::AudioSendParameters parameters;
>+  parameters.codecs.push_back(kTelephoneEventCodec1);
>+  parameters.codecs.push_back(kPcmuCodec);
>+  parameters.codecs[0].id = 98;  // DTMF
>+  parameters.codecs[1].id = 96;
>+  SetSendParameters(parameters);
>+  EXPECT_FALSE(channel_->CanInsertDtmf());
>+  SetSend(true);
>   EXPECT_TRUE(channel_->CanInsertDtmf());
>+  SetSend(false);
>+  EXPECT_FALSE(channel_->CanInsertDtmf());
> }
> 
> // Test that payload type range is limited for telephone-event codec.
>@@ -1786,6 +1846,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsDTMFPayloadTypeOutOfRange) {
>   parameters.codecs[0].id = 0;  // DTMF
>   parameters.codecs[1].id = 96;
>   SetSendParameters(parameters);
>+  SetSend(true);
>   EXPECT_TRUE(channel_->CanInsertDtmf());
>   parameters.codecs[0].id = 128;  // DTMF
>   EXPECT_FALSE(channel_->SetSendParameters(parameters));
>@@ -1832,8 +1893,9 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCNandDTMFAsCaller) {
>   const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
>   EXPECT_EQ(96, send_codec_spec.payload_type);
>   EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str());
>-  EXPECT_EQ(1, send_codec_spec.format.num_channels);
>+  EXPECT_EQ(1u, send_codec_spec.format.num_channels);
>   EXPECT_EQ(97, send_codec_spec.cng_payload_type);
>+  SetSend(true);
>   EXPECT_TRUE(channel_->CanInsertDtmf());
> }
> 
>@@ -1851,14 +1913,15 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCNandDTMFAsCallee) {
>   parameters.codecs[2].id = 97;  // wideband CN
>   parameters.codecs[4].id = 98;  // DTMF
>   SetSendParameters(parameters);
>-  EXPECT_TRUE(channel_->AddSendStream(
>-      cricket::StreamParams::CreateLegacy(kSsrcX)));
>+  EXPECT_TRUE(
>+      channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcX)));
> 
>   const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
>   EXPECT_EQ(96, send_codec_spec.payload_type);
>   EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str());
>-  EXPECT_EQ(1, send_codec_spec.format.num_channels);
>+  EXPECT_EQ(1u, send_codec_spec.format.num_channels);
>   EXPECT_EQ(97, send_codec_spec.cng_payload_type);
>+  SetSend(true);
>   EXPECT_TRUE(channel_->CanInsertDtmf());
> }
> 
>@@ -1875,7 +1938,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCNNoMatch) {
>   {
>     const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
>     EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str());
>-    EXPECT_EQ(1, send_codec_spec.format.num_channels);
>+    EXPECT_EQ(1u, send_codec_spec.format.num_channels);
>     EXPECT_EQ(97, send_codec_spec.cng_payload_type);
>   }
>   // Set PCMU(8K) and CN(16K). VAD should not be activated.
>@@ -1884,7 +1947,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCNNoMatch) {
>   {
>     const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
>     EXPECT_STRCASEEQ("PCMU", send_codec_spec.format.name.c_str());
>-    EXPECT_EQ(rtc::nullopt, send_codec_spec.cng_payload_type);
>+    EXPECT_EQ(absl::nullopt, send_codec_spec.cng_payload_type);
>   }
>   // Set PCMU(8K) and CN(8K). VAD should be activated.
>   parameters.codecs[1] = kCn8000Codec;
>@@ -1892,7 +1955,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCNNoMatch) {
>   {
>     const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
>     EXPECT_STRCASEEQ("PCMU", send_codec_spec.format.name.c_str());
>-    EXPECT_EQ(1, send_codec_spec.format.num_channels);
>+    EXPECT_EQ(1u, send_codec_spec.format.num_channels);
>     EXPECT_EQ(13, send_codec_spec.cng_payload_type);
>   }
>   // Set ISAC(16K) and CN(8K). VAD should not be activated.
>@@ -1901,7 +1964,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCNNoMatch) {
>   {
>     const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
>     EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str());
>-    EXPECT_EQ(rtc::nullopt, send_codec_spec.cng_payload_type);
>+    EXPECT_EQ(absl::nullopt, send_codec_spec.cng_payload_type);
>   }
> }
> 
>@@ -1922,8 +1985,9 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsCaseInsensitive) {
>   const auto& send_codec_spec = *GetSendStreamConfig(kSsrcX).send_codec_spec;
>   EXPECT_EQ(96, send_codec_spec.payload_type);
>   EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str());
>-  EXPECT_EQ(1, send_codec_spec.format.num_channels);
>+  EXPECT_EQ(1u, send_codec_spec.format.num_channels);
>   EXPECT_EQ(97, send_codec_spec.cng_payload_type);
>+  SetSend(true);
>   EXPECT_TRUE(channel_->CanInsertDtmf());
> }
> 
>@@ -2043,8 +2107,8 @@ TEST_F(WebRtcVoiceEngineTestFake, CreateAndDeleteMultipleSendStreams) {
>   SetSend(true);
> 
>   for (uint32_t ssrc : kSsrcs4) {
>-    EXPECT_TRUE(channel_->AddSendStream(
>-        cricket::StreamParams::CreateLegacy(ssrc)));
>+    EXPECT_TRUE(
>+        channel_->AddSendStream(cricket::StreamParams::CreateLegacy(ssrc)));
>     SetAudioSend(ssrc, true, &fake_source_);
>     // Verify that we are in a sending state for all the created streams.
>     EXPECT_TRUE(GetSendStream(ssrc).IsSending());
>@@ -2066,8 +2130,8 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsWithMultipleSendStreams) {
> 
>   // Create send streams.
>   for (uint32_t ssrc : kSsrcs4) {
>-    EXPECT_TRUE(channel_->AddSendStream(
>-        cricket::StreamParams::CreateLegacy(ssrc)));
>+    EXPECT_TRUE(
>+        channel_->AddSendStream(cricket::StreamParams::CreateLegacy(ssrc)));
>   }
> 
>   cricket::AudioSendParameters parameters;
>@@ -2083,7 +2147,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsWithMultipleSendStreams) {
>     const auto& send_codec_spec =
>         *call_.GetAudioSendStream(ssrc)->GetConfig().send_codec_spec;
>     EXPECT_STRCASEEQ("ISAC", send_codec_spec.format.name.c_str());
>-    EXPECT_EQ(1, send_codec_spec.format.num_channels);
>+    EXPECT_EQ(1u, send_codec_spec.format.num_channels);
>     EXPECT_EQ(97, send_codec_spec.cng_payload_type);
>   }
> 
>@@ -2095,7 +2159,7 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendCodecsWithMultipleSendStreams) {
>     const auto& send_codec_spec =
>         *call_.GetAudioSendStream(ssrc)->GetConfig().send_codec_spec;
>     EXPECT_STRCASEEQ("PCMU", send_codec_spec.format.name.c_str());
>-    EXPECT_EQ(rtc::nullopt, send_codec_spec.cng_payload_type);
>+    EXPECT_EQ(absl::nullopt, send_codec_spec.cng_payload_type);
>   }
> }
> 
>@@ -2105,8 +2169,8 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendWithMultipleSendStreams) {
> 
>   // Create the send channels and they should be a "not sending" date.
>   for (uint32_t ssrc : kSsrcs4) {
>-    EXPECT_TRUE(channel_->AddSendStream(
>-        cricket::StreamParams::CreateLegacy(ssrc)));
>+    EXPECT_TRUE(
>+        channel_->AddSendStream(cricket::StreamParams::CreateLegacy(ssrc)));
>     SetAudioSend(ssrc, true, &fake_source_);
>     EXPECT_FALSE(GetSendStream(ssrc).IsSending());
>   }
>@@ -2132,8 +2196,8 @@ TEST_F(WebRtcVoiceEngineTestFake, GetStatsWithMultipleSendStreams) {
> 
>   // Create send streams.
>   for (uint32_t ssrc : kSsrcs4) {
>-    EXPECT_TRUE(channel_->AddSendStream(
>-        cricket::StreamParams::CreateLegacy(ssrc)));
>+    EXPECT_TRUE(
>+        channel_->AddSendStream(cricket::StreamParams::CreateLegacy(ssrc)));
>   }
> 
>   // Create a receive stream to check that none of the send streams end up in
>@@ -2159,7 +2223,7 @@ TEST_F(WebRtcVoiceEngineTestFake, GetStatsWithMultipleSendStreams) {
> 
>     // We have added one receive stream. We should see empty stats.
>     EXPECT_EQ(info.receivers.size(), 1u);
>-    EXPECT_EQ(info.receivers[0].ssrc(), 0);
>+    EXPECT_EQ(info.receivers[0].ssrc(), 0u);
>   }
> 
>   // Remove the kSsrcY stream. No receiver stats.
>@@ -2262,7 +2326,7 @@ TEST_F(WebRtcVoiceEngineTestFake, AudioSendResetAudioNetworkAdaptor) {
>   cricket::AudioOptions options;
>   options.audio_network_adaptor = false;
>   SetAudioSend(kSsrcX, true, nullptr, &options);
>-  EXPECT_EQ(rtc::nullopt, GetAudioNetworkAdaptorConfig(kSsrcX));
>+  EXPECT_EQ(absl::nullopt, GetAudioNetworkAdaptorConfig(kSsrcX));
> }
> 
> TEST_F(WebRtcVoiceEngineTestFake, AudioNetworkAdaptorNotGetOverridden) {
>@@ -2274,7 +2338,7 @@ TEST_F(WebRtcVoiceEngineTestFake, AudioNetworkAdaptorNotGetOverridden) {
>             GetAudioNetworkAdaptorConfig(kSsrcX));
>   const int initial_num = call_.GetNumCreatedSendStreams();
>   cricket::AudioOptions options;
>-  options.audio_network_adaptor = rtc::nullopt;
>+  options.audio_network_adaptor = absl::nullopt;
>   // Unvalued |options.audio_network_adaptor|.should not reset audio network
>   // adaptor.
>   SetAudioSend(kSsrcX, true, nullptr, &options);
>@@ -2344,7 +2408,7 @@ TEST_F(WebRtcVoiceEngineWithSendSideBweWithOverheadTest,
> 
>   constexpr int kMaxBitrateBps = 6000;
>   rtp_parameters.encodings[0].max_bitrate_bps = kMaxBitrateBps;
>-  EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrcX, rtp_parameters));
>+  EXPECT_TRUE(channel_->SetRtpSendParameters(kSsrcX, rtp_parameters).ok());
> 
>   const int max_bitrate = GetSendStreamConfig(kSsrcX).max_bitrate_bps;
> #if WEBRTC_OPUS_SUPPORT_120MS_PTIME
>@@ -2382,7 +2446,7 @@ TEST_F(WebRtcVoiceEngineTestFake, GetStats) {
>     VerifyVoiceSenderInfo(info.senders[0], false);
>     // We have added one receive stream. We should see empty stats.
>     EXPECT_EQ(info.receivers.size(), 1u);
>-    EXPECT_EQ(info.receivers[0].ssrc(), 0);
>+    EXPECT_EQ(info.receivers[0].ssrc(), 0u);
>   }
> 
>   // Start sending - this affects some reported stats.
>@@ -2431,8 +2495,8 @@ TEST_F(WebRtcVoiceEngineTestFake, SetSendSsrcWithMultipleStreams) {
> TEST_F(WebRtcVoiceEngineTestFake, SetSendSsrcAfterCreatingReceiveChannel) {
>   EXPECT_TRUE(SetupChannel());
>   EXPECT_TRUE(AddRecvStream(kSsrcY));
>-  EXPECT_TRUE(channel_->AddSendStream(
>-      cricket::StreamParams::CreateLegacy(kSsrcX)));
>+  EXPECT_TRUE(
>+      channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcX)));
>   EXPECT_TRUE(call_.GetAudioSendStream(kSsrcX));
>   EXPECT_EQ(kSsrcX, GetRecvStreamConfig(kSsrcY).rtp.local_ssrc);
> }
>@@ -2443,8 +2507,8 @@ TEST_F(WebRtcVoiceEngineTestFake, Recv) {
>   EXPECT_TRUE(AddRecvStream(1));
>   DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
> 
>-  EXPECT_TRUE(GetRecvStream(1).VerifyLastPacket(kPcmuFrame,
>-                                                sizeof(kPcmuFrame)));
>+  EXPECT_TRUE(
>+      GetRecvStream(1).VerifyLastPacket(kPcmuFrame, sizeof(kPcmuFrame)));
> }
> 
> // Test that we can properly receive packets on multiple streams.
>@@ -2502,13 +2566,46 @@ TEST_F(WebRtcVoiceEngineTestFake, RecvWithMultipleStreams) {
> // Test that receiving on an unsignaled stream works (a stream is created).
> TEST_F(WebRtcVoiceEngineTestFake, RecvUnsignaled) {
>   EXPECT_TRUE(SetupChannel());
>-  EXPECT_EQ(0, call_.GetAudioReceiveStreams().size());
>+  EXPECT_EQ(0u, call_.GetAudioReceiveStreams().size());
> 
>   DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
> 
>-  EXPECT_EQ(1, call_.GetAudioReceiveStreams().size());
>-  EXPECT_TRUE(GetRecvStream(kSsrc1).VerifyLastPacket(kPcmuFrame,
>-                                                     sizeof(kPcmuFrame)));
>+  EXPECT_EQ(1u, call_.GetAudioReceiveStreams().size());
>+  EXPECT_TRUE(
>+      GetRecvStream(kSsrc1).VerifyLastPacket(kPcmuFrame, sizeof(kPcmuFrame)));
>+}
>+
>+// Tests that when we add a stream without SSRCs, but contains a stream_id
>+// that it is stored and its stream id is later used when the first packet
>+// arrives to properly create a receive stream with a sync label.
>+TEST_F(WebRtcVoiceEngineTestFake, RecvUnsignaledSsrcWithSignaledStreamId) {
>+  const char kSyncLabel[] = "sync_label";
>+  EXPECT_TRUE(SetupChannel());
>+  cricket::StreamParams unsignaled_stream;
>+  unsignaled_stream.set_stream_ids({kSyncLabel});
>+  ASSERT_TRUE(channel_->AddRecvStream(unsignaled_stream));
>+  // The stream shouldn't have been created at this point because it doesn't
>+  // have any SSRCs.
>+  EXPECT_EQ(0u, call_.GetAudioReceiveStreams().size());
>+
>+  DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
>+
>+  EXPECT_EQ(1u, call_.GetAudioReceiveStreams().size());
>+  EXPECT_TRUE(
>+      GetRecvStream(kSsrc1).VerifyLastPacket(kPcmuFrame, sizeof(kPcmuFrame)));
>+  EXPECT_EQ(kSyncLabel, GetRecvStream(kSsrc1).GetConfig().sync_group);
>+
>+  // Removing the unsignaled stream clears the cached parameters. If a new
>+  // default unsignaled receive stream is created it will not have a sync group.
>+  channel_->RemoveRecvStream(0);
>+  channel_->RemoveRecvStream(kSsrc1);
>+
>+  DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
>+
>+  EXPECT_EQ(1u, call_.GetAudioReceiveStreams().size());
>+  EXPECT_TRUE(
>+      GetRecvStream(kSsrc1).VerifyLastPacket(kPcmuFrame, sizeof(kPcmuFrame)));
>+  EXPECT_TRUE(GetRecvStream(kSsrc1).GetConfig().sync_group.empty());
> }
> 
> // Test that receiving N unsignaled stream works (streams will be created), and
>@@ -2569,18 +2666,18 @@ TEST_F(WebRtcVoiceEngineTestFake, RecvUnsignaledAfterSignaled) {
>   rtc::SetBE32(&packet[8], signaled_ssrc);
>   EXPECT_TRUE(AddRecvStream(signaled_ssrc));
>   DeliverPacket(packet, sizeof(packet));
>-  EXPECT_TRUE(GetRecvStream(signaled_ssrc).VerifyLastPacket(
>-      packet, sizeof(packet)));
>-  EXPECT_EQ(1, call_.GetAudioReceiveStreams().size());
>+  EXPECT_TRUE(
>+      GetRecvStream(signaled_ssrc).VerifyLastPacket(packet, sizeof(packet)));
>+  EXPECT_EQ(1u, call_.GetAudioReceiveStreams().size());
> 
>   // Note that the first unknown SSRC cannot be 0, because we only support
>   // creating receive streams for SSRC!=0.
>   const uint32_t unsignaled_ssrc = 7011;
>   rtc::SetBE32(&packet[8], unsignaled_ssrc);
>   DeliverPacket(packet, sizeof(packet));
>-  EXPECT_TRUE(GetRecvStream(unsignaled_ssrc).VerifyLastPacket(
>-      packet, sizeof(packet)));
>-  EXPECT_EQ(2, call_.GetAudioReceiveStreams().size());
>+  EXPECT_TRUE(
>+      GetRecvStream(unsignaled_ssrc).VerifyLastPacket(packet, sizeof(packet)));
>+  EXPECT_EQ(2u, call_.GetAudioReceiveStreams().size());
> 
>   DeliverPacket(packet, sizeof(packet));
>   EXPECT_EQ(2, GetRecvStream(unsignaled_ssrc).received_packets());
>@@ -2588,7 +2685,7 @@ TEST_F(WebRtcVoiceEngineTestFake, RecvUnsignaledAfterSignaled) {
>   rtc::SetBE32(&packet[8], signaled_ssrc);
>   DeliverPacket(packet, sizeof(packet));
>   EXPECT_EQ(2, GetRecvStream(signaled_ssrc).received_packets());
>-  EXPECT_EQ(2, call_.GetAudioReceiveStreams().size());
>+  EXPECT_EQ(2u, call_.GetAudioReceiveStreams().size());
> }
> 
> // Two tests to verify that adding a receive stream with the same SSRC as a
>@@ -2599,17 +2696,17 @@ TEST_F(WebRtcVoiceEngineTestFake, AddRecvStreamAfterUnsignaled_NoRecreate) {
> 
>   // Spawn unsignaled stream with SSRC=1.
>   DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
>-  EXPECT_EQ(1, call_.GetAudioReceiveStreams().size());
>-  EXPECT_TRUE(GetRecvStream(1).VerifyLastPacket(kPcmuFrame,
>-                                                sizeof(kPcmuFrame)));
>+  EXPECT_EQ(1u, call_.GetAudioReceiveStreams().size());
>+  EXPECT_TRUE(
>+      GetRecvStream(1).VerifyLastPacket(kPcmuFrame, sizeof(kPcmuFrame)));
> 
>   // Verify that the underlying stream object in Call is not recreated when a
>   // stream with SSRC=1 is added.
>   const auto& streams = call_.GetAudioReceiveStreams();
>-  EXPECT_EQ(1, streams.size());
>+  EXPECT_EQ(1u, streams.size());
>   int audio_receive_stream_id = streams.front()->id();
>   EXPECT_TRUE(AddRecvStream(1));
>-  EXPECT_EQ(1, streams.size());
>+  EXPECT_EQ(1u, streams.size());
>   EXPECT_EQ(audio_receive_stream_id, streams.front()->id());
> }
> 
>@@ -2618,20 +2715,20 @@ TEST_F(WebRtcVoiceEngineTestFake, AddRecvStreamAfterUnsignaled_Recreate) {
> 
>   // Spawn unsignaled stream with SSRC=1.
>   DeliverPacket(kPcmuFrame, sizeof(kPcmuFrame));
>-  EXPECT_EQ(1, call_.GetAudioReceiveStreams().size());
>-  EXPECT_TRUE(GetRecvStream(1).VerifyLastPacket(kPcmuFrame,
>-                                                sizeof(kPcmuFrame)));
>+  EXPECT_EQ(1u, call_.GetAudioReceiveStreams().size());
>+  EXPECT_TRUE(
>+      GetRecvStream(1).VerifyLastPacket(kPcmuFrame, sizeof(kPcmuFrame)));
> 
>   // Verify that the underlying stream object in Call *is* recreated when a
>   // stream with SSRC=1 is added, and which has changed stream parameters.
>   const auto& streams = call_.GetAudioReceiveStreams();
>-  EXPECT_EQ(1, streams.size());
>+  EXPECT_EQ(1u, streams.size());
>   int audio_receive_stream_id = streams.front()->id();
>   cricket::StreamParams stream_params;
>   stream_params.ssrcs.push_back(1);
>-  stream_params.sync_label = "sync_label";
>+  stream_params.set_stream_ids({"stream_id"});
>   EXPECT_TRUE(channel_->AddRecvStream(stream_params));
>-  EXPECT_EQ(1, streams.size());
>+  EXPECT_EQ(1u, streams.size());
>   EXPECT_NE(audio_receive_stream_id, streams.front()->id());
> }
> 
>@@ -2663,12 +2760,12 @@ TEST_F(WebRtcVoiceEngineTestFake, StreamCleanup) {
>   EXPECT_TRUE(AddRecvStream(1));
>   EXPECT_TRUE(AddRecvStream(2));
> 
>-  EXPECT_EQ(1, call_.GetAudioSendStreams().size());
>-  EXPECT_EQ(2, call_.GetAudioReceiveStreams().size());
>+  EXPECT_EQ(1u, call_.GetAudioSendStreams().size());
>+  EXPECT_EQ(2u, call_.GetAudioReceiveStreams().size());
>   delete channel_;
>   channel_ = NULL;
>-  EXPECT_EQ(0, call_.GetAudioSendStreams().size());
>-  EXPECT_EQ(0, call_.GetAudioReceiveStreams().size());
>+  EXPECT_EQ(0u, call_.GetAudioSendStreams().size());
>+  EXPECT_EQ(0u, call_.GetAudioReceiveStreams().size());
> }
> 
> TEST_F(WebRtcVoiceEngineTestFake, TestAddRecvStreamFailWithZeroSsrc) {
>@@ -2705,21 +2802,24 @@ TEST_F(WebRtcVoiceEngineTestFake, InsertDtmfOnSendStreamAsCallee) {
> TEST_F(WebRtcVoiceEngineTestFake, SetAudioOptions) {
>   EXPECT_TRUE(SetupSendStream());
>   EXPECT_TRUE(AddRecvStream(kSsrcY));
>-  EXPECT_CALL(adm_,
>-              BuiltInAECIsAvailable()).Times(9).WillRepeatedly(Return(false));
>-  EXPECT_CALL(adm_,
>-              BuiltInAGCIsAvailable()).Times(4).WillRepeatedly(Return(false));
>-  EXPECT_CALL(adm_,
>-              BuiltInNSIsAvailable()).Times(2).WillRepeatedly(Return(false));
>-
>-  EXPECT_EQ(50, GetRecvStreamConfig(kSsrcY).jitter_buffer_max_packets);
>+  EXPECT_CALL(adm_, BuiltInAECIsAvailable())
>+      .Times(9)
>+      .WillRepeatedly(Return(false));
>+  EXPECT_CALL(adm_, BuiltInAGCIsAvailable())
>+      .Times(4)
>+      .WillRepeatedly(Return(false));
>+  EXPECT_CALL(adm_, BuiltInNSIsAvailable())
>+      .Times(2)
>+      .WillRepeatedly(Return(false));
>+
>+  EXPECT_EQ(50u, GetRecvStreamConfig(kSsrcY).jitter_buffer_max_packets);
>   EXPECT_FALSE(GetRecvStreamConfig(kSsrcY).jitter_buffer_fast_accelerate);
> 
>   // Nothing set in AudioOptions, so everything should be as default.
>   send_parameters_.options = cricket::AudioOptions();
>   SetSendParameters(send_parameters_);
>   EXPECT_TRUE(IsHighPassFilterEnabled());
>-  EXPECT_EQ(50, GetRecvStreamConfig(kSsrcY).jitter_buffer_max_packets);
>+  EXPECT_EQ(50u, GetRecvStreamConfig(kSsrcY).jitter_buffer_max_packets);
>   EXPECT_FALSE(GetRecvStreamConfig(kSsrcY).jitter_buffer_fast_accelerate);
> 
>   // Turn echo cancellation off
>@@ -2800,14 +2900,18 @@ TEST_F(WebRtcVoiceEngineTestFake, SetAudioOptions) {
> 
> TEST_F(WebRtcVoiceEngineTestFake, SetOptionOverridesViaChannels) {
>   EXPECT_TRUE(SetupSendStream());
>-  EXPECT_CALL(adm_,
>-              BuiltInAECIsAvailable()).Times(8).WillRepeatedly(Return(false));
>-  EXPECT_CALL(adm_,
>-              BuiltInAGCIsAvailable()).Times(8).WillRepeatedly(Return(false));
>-  EXPECT_CALL(adm_,
>-              BuiltInNSIsAvailable()).Times(8).WillRepeatedly(Return(false));
>-  EXPECT_CALL(adm_,
>-              RecordingIsInitialized()).Times(2).WillRepeatedly(Return(false));
>+  EXPECT_CALL(adm_, BuiltInAECIsAvailable())
>+      .Times(8)
>+      .WillRepeatedly(Return(false));
>+  EXPECT_CALL(adm_, BuiltInAGCIsAvailable())
>+      .Times(8)
>+      .WillRepeatedly(Return(false));
>+  EXPECT_CALL(adm_, BuiltInNSIsAvailable())
>+      .Times(8)
>+      .WillRepeatedly(Return(false));
>+  EXPECT_CALL(adm_, RecordingIsInitialized())
>+      .Times(2)
>+      .WillRepeatedly(Return(false));
>   EXPECT_CALL(adm_, Recording()).Times(2).WillRepeatedly(Return(false));
>   EXPECT_CALL(adm_, InitRecording()).Times(2).WillRepeatedly(Return(0));
>   webrtc::AudioProcessing::Config apm_config;
>@@ -3008,26 +3112,26 @@ TEST_F(WebRtcVoiceEngineTestFake, SetOutputVolumeUnsignaledRecvStream) {
>   EXPECT_DOUBLE_EQ(4, GetRecvStream(kSsrcX).gain());
> }
> 
>-TEST_F(WebRtcVoiceEngineTestFake, SetsSyncGroupFromSyncLabel) {
>+TEST_F(WebRtcVoiceEngineTestFake, SetsSyncGroupFromStreamId) {
>   const uint32_t kAudioSsrc = 123;
>-  const std::string kSyncLabel = "AvSyncLabel";
>+  const std::string kStreamId = "AvSyncLabel";
> 
>   EXPECT_TRUE(SetupSendStream());
>   cricket::StreamParams sp = cricket::StreamParams::CreateLegacy(kAudioSsrc);
>-  sp.sync_label = kSyncLabel;
>+  sp.set_stream_ids({kStreamId});
>   // Creating two channels to make sure that sync label is set properly for both
>   // the default voice channel and following ones.
>   EXPECT_TRUE(channel_->AddRecvStream(sp));
>   sp.ssrcs[0] += 1;
>   EXPECT_TRUE(channel_->AddRecvStream(sp));
> 
>-  ASSERT_EQ(2, call_.GetAudioReceiveStreams().size());
>-  EXPECT_EQ(kSyncLabel,
>+  ASSERT_EQ(2u, call_.GetAudioReceiveStreams().size());
>+  EXPECT_EQ(kStreamId,
>             call_.GetAudioReceiveStream(kAudioSsrc)->GetConfig().sync_group)
>-      << "SyncGroup should be set based on sync_label";
>-  EXPECT_EQ(kSyncLabel,
>+      << "SyncGroup should be set based on stream id";
>+  EXPECT_EQ(kStreamId,
>             call_.GetAudioReceiveStream(kAudioSsrc + 1)->GetConfig().sync_group)
>-      << "SyncGroup should be set based on sync_label";
>+      << "SyncGroup should be set based on stream id";
> }
> 
> // TODO(solenberg): Remove, once recv streams are configured through Call.
>@@ -3042,15 +3146,15 @@ TEST_F(WebRtcVoiceEngineTestFake, ConfiguresAudioReceiveStreamRtpExtensions) {
>   EXPECT_TRUE(SetupSendStream());
>   SetSendParameters(send_parameters_);
>   for (uint32_t ssrc : ssrcs) {
>-    EXPECT_TRUE(channel_->AddRecvStream(
>-        cricket::StreamParams::CreateLegacy(ssrc)));
>+    EXPECT_TRUE(
>+        channel_->AddRecvStream(cricket::StreamParams::CreateLegacy(ssrc)));
>   }
> 
>-  EXPECT_EQ(2, call_.GetAudioReceiveStreams().size());
>+  EXPECT_EQ(2u, call_.GetAudioReceiveStreams().size());
>   for (uint32_t ssrc : ssrcs) {
>     const auto* s = call_.GetAudioReceiveStream(ssrc);
>     EXPECT_NE(nullptr, s);
>-    EXPECT_EQ(0, s->GetConfig().rtp.extensions.size());
>+    EXPECT_EQ(0u, s->GetConfig().rtp.extensions.size());
>   }
> 
>   // Set up receive extensions.
>@@ -3058,7 +3162,7 @@ TEST_F(WebRtcVoiceEngineTestFake, ConfiguresAudioReceiveStreamRtpExtensions) {
>   cricket::AudioRecvParameters recv_parameters;
>   recv_parameters.extensions = capabilities.header_extensions;
>   channel_->SetRecvParameters(recv_parameters);
>-  EXPECT_EQ(2, call_.GetAudioReceiveStreams().size());
>+  EXPECT_EQ(2u, call_.GetAudioReceiveStreams().size());
>   for (uint32_t ssrc : ssrcs) {
>     const auto* s = call_.GetAudioReceiveStream(ssrc);
>     EXPECT_NE(nullptr, s);
>@@ -3078,7 +3182,7 @@ TEST_F(WebRtcVoiceEngineTestFake, ConfiguresAudioReceiveStreamRtpExtensions) {
>   for (uint32_t ssrc : ssrcs) {
>     const auto* s = call_.GetAudioReceiveStream(ssrc);
>     EXPECT_NE(nullptr, s);
>-    EXPECT_EQ(0, s->GetConfig().rtp.extensions.size());
>+    EXPECT_EQ(0u, s->GetConfig().rtp.extensions.size());
>   }
> }
> 
>@@ -3087,11 +3191,9 @@ TEST_F(WebRtcVoiceEngineTestFake, DeliverAudioPacket_Call) {
>   const uint32_t kAudioSsrc = 1;
>   rtc::CopyOnWriteBuffer kPcmuPacket(kPcmuFrame, sizeof(kPcmuFrame));
>   static const unsigned char kRtcp[] = {
>-    0x80, 0xc9, 0x00, 0x01, 0x00, 0x00, 0x00, 0x02,
>-    0x00, 0x00, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00,
>-    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
>-    0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00
>-  };
>+      0x80, 0xc9, 0x00, 0x01, 0x00, 0x00, 0x00, 0x02, 0x00, 0x00, 0x00,
>+      0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
>+      0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00};
>   rtc::CopyOnWriteBuffer kRtcpPacket(kRtcp, sizeof(kRtcp));
> 
>   EXPECT_TRUE(SetupSendStream());
>@@ -3101,7 +3203,7 @@ TEST_F(WebRtcVoiceEngineTestFake, DeliverAudioPacket_Call) {
>   EXPECT_TRUE(media_channel->AddRecvStream(
>       cricket::StreamParams::CreateLegacy(kAudioSsrc)));
> 
>-  EXPECT_EQ(1, call_.GetAudioReceiveStreams().size());
>+  EXPECT_EQ(1u, call_.GetAudioReceiveStreams().size());
>   const cricket::FakeAudioReceiveStream* s =
>       call_.GetAudioReceiveStream(kAudioSsrc);
>   EXPECT_EQ(0, s->received_packets());
>@@ -3117,8 +3219,8 @@ TEST_F(WebRtcVoiceEngineTestFake, AssociateFirstSendChannel_SendCreatedFirst) {
>   EXPECT_TRUE(SetupSendStream());
>   EXPECT_TRUE(AddRecvStream(kSsrcY));
>   EXPECT_EQ(kSsrcX, GetRecvStreamConfig(kSsrcY).rtp.local_ssrc);
>-  EXPECT_TRUE(channel_->AddSendStream(
>-      cricket::StreamParams::CreateLegacy(kSsrcZ)));
>+  EXPECT_TRUE(
>+      channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcZ)));
>   EXPECT_EQ(kSsrcX, GetRecvStreamConfig(kSsrcY).rtp.local_ssrc);
>   EXPECT_TRUE(AddRecvStream(kSsrcW));
>   EXPECT_EQ(kSsrcX, GetRecvStreamConfig(kSsrcW).rtp.local_ssrc);
>@@ -3127,13 +3229,13 @@ TEST_F(WebRtcVoiceEngineTestFake, AssociateFirstSendChannel_SendCreatedFirst) {
> TEST_F(WebRtcVoiceEngineTestFake, AssociateFirstSendChannel_RecvCreatedFirst) {
>   EXPECT_TRUE(SetupRecvStream());
>   EXPECT_EQ(0xFA17FA17u, GetRecvStreamConfig(kSsrcX).rtp.local_ssrc);
>-  EXPECT_TRUE(channel_->AddSendStream(
>-      cricket::StreamParams::CreateLegacy(kSsrcY)));
>+  EXPECT_TRUE(
>+      channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcY)));
>   EXPECT_EQ(kSsrcY, GetRecvStreamConfig(kSsrcX).rtp.local_ssrc);
>   EXPECT_TRUE(AddRecvStream(kSsrcZ));
>   EXPECT_EQ(kSsrcY, GetRecvStreamConfig(kSsrcZ).rtp.local_ssrc);
>-  EXPECT_TRUE(channel_->AddSendStream(
>-      cricket::StreamParams::CreateLegacy(kSsrcW)));
>+  EXPECT_TRUE(
>+      channel_->AddSendStream(cricket::StreamParams::CreateLegacy(kSsrcW)));
>   EXPECT_EQ(kSsrcY, GetRecvStreamConfig(kSsrcX).rtp.local_ssrc);
>   EXPECT_EQ(kSsrcY, GetRecvStreamConfig(kSsrcZ).rtp.local_ssrc);
> }
>@@ -3343,9 +3445,9 @@ TEST(WebRtcVoiceEngineTest, HasCorrectPayloadTypeMapping) {
>       EXPECT_EQ(9, codec.id);
>     } else if (is_codec("telephone-event", 8000)) {
>       EXPECT_EQ(126, codec.id);
>-    // TODO(solenberg): 16k, 32k, 48k DTMF should be dynamically assigned.
>-    // Remove these checks once both send and receive side assigns payload types
>-    // dynamically.
>+      // TODO(solenberg): 16k, 32k, 48k DTMF should be dynamically assigned.
>+      // Remove these checks once both send and receive side assigns payload
>+      // types dynamically.
>     } else if (is_codec("telephone-event", 16000)) {
>       EXPECT_EQ(113, codec.id);
>     } else if (is_codec("telephone-event", 32000)) {
>@@ -3376,7 +3478,7 @@ TEST(WebRtcVoiceEngineTest, Has32Channels) {
>       webrtc::Call::Create(webrtc::Call::Config(&event_log)));
> 
>   cricket::VoiceMediaChannel* channels[32];
>-  int num_channels = 0;
>+  size_t num_channels = 0;
>   while (num_channels < arraysize(channels)) {
>     cricket::VoiceMediaChannel* channel = engine.CreateChannel(
>         call.get(), cricket::MediaConfig(), cricket::AudioOptions());
>@@ -3385,7 +3487,7 @@ TEST(WebRtcVoiceEngineTest, Has32Channels) {
>     channels[num_channels++] = channel;
>   }
> 
>-  int expected = arraysize(channels);
>+  size_t expected = arraysize(channels);
>   EXPECT_EQ(expected, num_channels);
> 
>   while (num_channels > 0) {
>@@ -3451,17 +3553,16 @@ TEST(WebRtcVoiceEngineTest, CollectRecvCodecs) {
>                                     mock_decoder_factory, nullptr, apm);
>   engine.Init();
>   auto codecs = engine.recv_codecs();
>-  EXPECT_EQ(11, codecs.size());
>+  EXPECT_EQ(11u, codecs.size());
> 
>   // Rather than just ASSERTing that there are enough codecs, ensure that we can
>   // check the actual values safely, to provide better test results.
>-  auto get_codec =
>-      [&codecs](size_t index) -> const cricket::AudioCodec& {
>-        static const cricket::AudioCodec missing_codec(0, "<missing>", 0, 0, 0);
>-        if (codecs.size() > index)
>-          return codecs[index];
>-        return missing_codec;
>-      };
>+  auto get_codec = [&codecs](size_t index) -> const cricket::AudioCodec& {
>+    static const cricket::AudioCodec missing_codec(0, "<missing>", 0, 0, 0);
>+    if (codecs.size() > index)
>+      return codecs[index];
>+    return missing_codec;
>+  };
> 
>   // Ensure the general codecs are generated first and in order.
>   for (size_t i = 0; i != specs.size(); ++i) {
>@@ -3473,18 +3574,17 @@ TEST(WebRtcVoiceEngineTest, CollectRecvCodecs) {
> 
>   // Find the index of a codec, or -1 if not found, so that we can easily check
>   // supplementary codecs are ordered after the general codecs.
>-  auto find_codec =
>-      [&codecs](const webrtc::SdpAudioFormat& format) -> int {
>-        for (size_t i = 0; i != codecs.size(); ++i) {
>-          const cricket::AudioCodec& codec = codecs[i];
>-          if (STR_CASE_CMP(codec.name.c_str(), format.name.c_str()) == 0 &&
>-              codec.clockrate == format.clockrate_hz &&
>-              codec.channels == format.num_channels) {
>-            return rtc::checked_cast<int>(i);
>-          }
>-        }
>-        return -1;
>-      };
>+  auto find_codec = [&codecs](const webrtc::SdpAudioFormat& format) -> int {
>+    for (size_t i = 0; i != codecs.size(); ++i) {
>+      const cricket::AudioCodec& codec = codecs[i];
>+      if (STR_CASE_CMP(codec.name.c_str(), format.name.c_str()) == 0 &&
>+          codec.clockrate == format.clockrate_hz &&
>+          codec.channels == format.num_channels) {
>+        return rtc::checked_cast<int>(i);
>+      }
>+    }
>+    return -1;
>+  };
> 
>   // Ensure all supplementary codecs are generated last. Their internal ordering
>   // is not important.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/sctp/sctptransport.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/sctp/sctptransport.cc
>index e893c03e3dd..eaaf63fb3a6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/sctp/sctptransport.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/sctp/sctptransport.cc
>@@ -16,15 +16,15 @@ enum PreservedErrno {
>   SCTP_EINPROGRESS = EINPROGRESS,
>   SCTP_EWOULDBLOCK = EWOULDBLOCK
> };
>-}
>+}  // namespace
> 
> #include "media/sctp/sctptransport.h"
> 
> #include <stdarg.h>
> #include <stdio.h>
> 
>+#include <algorithm>
> #include <memory>
>-#include <sstream>
> 
> #include "media/base/codec.h"
> #include "media/base/mediaconstants.h"
>@@ -36,6 +36,7 @@ enum PreservedErrno {
> #include "rtc_base/helpers.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/numerics/safe_conversions.h"
>+#include "rtc_base/stringutils.h"
> #include "rtc_base/thread_checker.h"
> #include "rtc_base/trace_event.h"
> #include "usrsctplib/usrsctp.h"
>@@ -72,65 +73,6 @@ enum PayloadProtocolIdentifier {
>   PPID_TEXT_LAST = 51
> };
> 
>-typedef std::set<uint32_t> StreamSet;
>-
>-// Returns a comma-separated, human-readable list of the stream IDs in 's'
>-std::string ListStreams(const StreamSet& s) {
>-  std::stringstream result;
>-  bool first = true;
>-  for (StreamSet::const_iterator it = s.begin(); it != s.end(); ++it) {
>-    if (!first) {
>-      result << ", " << *it;
>-    } else {
>-      result << *it;
>-      first = false;
>-    }
>-  }
>-  return result.str();
>-}
>-
>-// Returns a pipe-separated, human-readable list of the SCTP_STREAM_RESET
>-// flags in 'flags'
>-std::string ListFlags(int flags) {
>-  std::stringstream result;
>-  bool first = true;
>-// Skip past the first 12 chars (strlen("SCTP_STREAM_"))
>-#define MAKEFLAG(X) \
>-  { X, #X + 12 }
>-  struct flaginfo_t {
>-    int value;
>-    const char* name;
>-  } flaginfo[] = {MAKEFLAG(SCTP_STREAM_RESET_INCOMING_SSN),
>-                  MAKEFLAG(SCTP_STREAM_RESET_OUTGOING_SSN),
>-                  MAKEFLAG(SCTP_STREAM_RESET_DENIED),
>-                  MAKEFLAG(SCTP_STREAM_RESET_FAILED),
>-                  MAKEFLAG(SCTP_STREAM_CHANGE_DENIED)};
>-#undef MAKEFLAG
>-  for (uint32_t i = 0; i < arraysize(flaginfo); ++i) {
>-    if (flags & flaginfo[i].value) {
>-      if (!first)
>-        result << " | ";
>-      result << flaginfo[i].name;
>-      first = false;
>-    }
>-  }
>-  return result.str();
>-}
>-
>-// Returns a comma-separated, human-readable list of the integers in 'array'.
>-// All 'num_elems' of them.
>-std::string ListArray(const uint16_t* array, int num_elems) {
>-  std::stringstream result;
>-  for (int i = 0; i < num_elems; ++i) {
>-    if (i) {
>-      result << ", " << array[i];
>-    } else {
>-      result << array[i];
>-    }
>-  }
>-  return result.str();
>-}
>-
> // Helper for logging SCTP messages.
> #if defined(__GNUC__)
> __attribute__((__format__(__printf__, 1, 2)))
>@@ -312,8 +254,8 @@ class SctpTransport::UsrSctpWrapper {
>     SctpTransport* transport = static_cast<SctpTransport*>(addr);
>     RTC_LOG(LS_VERBOSE) << "global OnSctpOutboundPacket():"
>                         << "addr: " << addr << "; length: " << length
>-                        << "; tos: " << std::hex << static_cast<int>(tos)
>-                        << "; set_df: " << std::hex << static_cast<int>(set_df);
>+                        << "; tos: " << rtc::ToHex(tos)
>+                        << "; set_df: " << rtc::ToHex(set_df);
> 
>     VerboseLogPacket(data, length, SCTP_DUMP_OUTBOUND);
>     // Note: We have to copy the data; the caller will delete it.
>@@ -349,22 +291,44 @@ class SctpTransport::UsrSctpWrapper {
>       // It's neither a notification nor a recognized data packet.  Drop it.
>       RTC_LOG(LS_ERROR) << "Received an unknown PPID " << ppid
>                         << " on an SCTP packet.  Dropping.";
>+      free(data);
>     } else {
>-      rtc::CopyOnWriteBuffer buffer;
>       ReceiveDataParams params;
>-      buffer.SetData(reinterpret_cast<uint8_t*>(data), length);
>+
>+      // Expect only continuation messages belonging to the same sid, usrsctp
>+      // ensures this.
>+      RTC_CHECK(transport->partial_message_.size() == 0 ||
>+                rcv.rcv_sid == transport->partial_message_sid_);
>+
>+      transport->partial_message_.AppendData(reinterpret_cast<uint8_t*>(data),
>+                                             length);
>+      transport->partial_message_sid_ = rcv.rcv_sid;
>+
>+      free(data);
>+
>+      // Merge partial messages until they exceed the maximum send buffer size.
>+      // This enables messages from a single send to be delivered in a single
>+      // callback. Larger messages (originating from other implementations) will
>+      // still be delivered in chunks.
>+      if (!(flags & MSG_EOR) &&
>+          (transport->partial_message_.size() < kSendBufferSize)) {
>+        return 1;
>+      }
>+
>       params.sid = rcv.rcv_sid;
>       params.seq_num = rcv.rcv_ssn;
>       params.timestamp = rcv.rcv_tsn;
>       params.type = type;
>+
>       // The ownership of the packet transfers to |invoker_|. Using
>       // CopyOnWriteBuffer is the most convenient way to do this.
>       transport->invoker_.AsyncInvoke<void>(
>           RTC_FROM_HERE, transport->network_thread_,
>-          rtc::Bind(&SctpTransport::OnInboundPacketFromSctpToChannel, transport,
>-                    buffer, params, flags));
>+          rtc::Bind(&SctpTransport::OnInboundPacketFromSctpToTransport,
>+                    transport, transport->partial_message_, params, flags));
>+
>+      transport->partial_message_.Clear();
>     }
>-    free(data);
>     return 1;
>   }
> 
>@@ -405,14 +369,14 @@ class SctpTransport::UsrSctpWrapper {
> };
> 
> SctpTransport::SctpTransport(rtc::Thread* network_thread,
>-                             rtc::PacketTransportInternal* channel)
>+                             rtc::PacketTransportInternal* transport)
>     : network_thread_(network_thread),
>-      transport_channel_(channel),
>-      was_ever_writable_(channel->writable()) {
>+      transport_(transport),
>+      was_ever_writable_(transport->writable()) {
>   RTC_DCHECK(network_thread_);
>-  RTC_DCHECK(transport_channel_);
>+  RTC_DCHECK(transport_);
>   RTC_DCHECK_RUN_ON(network_thread_);
>-  ConnectTransportChannelSignals();
>+  ConnectTransportSignals();
> }
> 
> SctpTransport::~SctpTransport() {
>@@ -420,15 +384,14 @@ SctpTransport::~SctpTransport() {
>   CloseSctpSocket();
> }
> 
>-void SctpTransport::SetTransportChannel(rtc::PacketTransportInternal* channel) {
>+void SctpTransport::SetDtlsTransport(rtc::PacketTransportInternal* transport) {
>   RTC_DCHECK_RUN_ON(network_thread_);
>-  RTC_DCHECK(channel);
>-  DisconnectTransportChannelSignals();
>-  transport_channel_ = channel;
>-  ConnectTransportChannelSignals();
>-  if (!was_ever_writable_ && channel->writable()) {
>+  DisconnectTransportSignals();
>+  transport_ = transport;
>+  ConnectTransportSignals();
>+  if (!was_ever_writable_ && transport && transport->writable()) {
>     was_ever_writable_ = true;
>-    // New channel is writable, now we can start the SCTP connection if Start
>+    // New transport is writable, now we can start the SCTP connection if Start
>     // was called already.
>     if (started_) {
>       RTC_DCHECK(!sock_);
>@@ -457,7 +420,7 @@ bool SctpTransport::Start(int local_sctp_port, int remote_sctp_port) {
>   remote_port_ = remote_sctp_port;
>   started_ = true;
>   RTC_DCHECK(!sock_);
>-  // Only try to connect if the DTLS channel has been writable before
>+  // Only try to connect if the DTLS transport has been writable before
>   // (indicating that the DTLS handshake is complete).
>   if (was_ever_writable_) {
>     return Connect();
>@@ -472,43 +435,40 @@ bool SctpTransport::OpenStream(int sid) {
>                         << "Not adding data stream "
>                         << "with sid=" << sid << " because sid is too high.";
>     return false;
>-  } else if (open_streams_.find(sid) != open_streams_.end()) {
>+  }
>+  auto it = stream_status_by_sid_.find(sid);
>+  if (it == stream_status_by_sid_.end()) {
>+    stream_status_by_sid_[sid] = StreamStatus();
>+    return true;
>+  }
>+  if (it->second.is_open()) {
>     RTC_LOG(LS_WARNING) << debug_name_ << "->OpenStream(...): "
>                         << "Not adding data stream "
>                         << "with sid=" << sid
>                         << " because stream is already open.";
>     return false;
>-  } else if (queued_reset_streams_.find(sid) != queued_reset_streams_.end() ||
>-             sent_reset_streams_.find(sid) != sent_reset_streams_.end()) {
>+  } else {
>     RTC_LOG(LS_WARNING) << debug_name_ << "->OpenStream(...): "
>                         << "Not adding data stream "
>                         << " with sid=" << sid
>                         << " because stream is still closing.";
>     return false;
>   }
>-
>-  open_streams_.insert(sid);
>-  return true;
> }
> 
> bool SctpTransport::ResetStream(int sid) {
>   RTC_DCHECK_RUN_ON(network_thread_);
>-  StreamSet::iterator found = open_streams_.find(sid);
>-  if (found == open_streams_.end()) {
>-    RTC_LOG(LS_WARNING) << debug_name_ << "->ResetStream(" << sid << "): "
>-                        << "stream not found.";
>+
>+  auto it = stream_status_by_sid_.find(sid);
>+  if (it == stream_status_by_sid_.end() || !it->second.is_open()) {
>+    RTC_LOG(LS_WARNING) << debug_name_ << "->ResetStream(" << sid
>+                        << "): stream not open.";
>     return false;
>-  } else {
>-    RTC_LOG(LS_VERBOSE) << debug_name_ << "->ResetStream(" << sid << "): "
>-                        << "Removing and queuing RE-CONFIG chunk.";
>-    open_streams_.erase(found);
>   }
> 
>-  // SCTP won't let you have more than one stream reset pending at a time, but
>-  // you can close multiple streams in a single reset.  So, we keep an internal
>-  // queue of streams-to-reset, and send them as one reset message in
>-  // SendQueuedStreamResets().
>-  queued_reset_streams_.insert(sid);
>+  RTC_LOG(LS_VERBOSE) << debug_name_ << "->ResetStream(" << sid << "): "
>+                      << "Queuing RE-CONFIG chunk.";
>+  it->second.closure_initiated = true;
> 
>   // Signal our stream-reset logic that it should try to send now, if it can.
>   SendQueuedStreamResets();
>@@ -534,12 +494,15 @@ bool SctpTransport::SendData(const SendDataParams& params,
>     return false;
>   }
> 
>-  if (params.type != DMT_CONTROL &&
>-      open_streams_.find(params.sid) == open_streams_.end()) {
>-    RTC_LOG(LS_WARNING) << debug_name_ << "->SendData(...): "
>-                        << "Not sending data because sid is unknown: "
>-                        << params.sid;
>-    return false;
>+  if (params.type != DMT_CONTROL) {
>+    auto it = stream_status_by_sid_.find(params.sid);
>+    if (it == stream_status_by_sid_.end() || !it->second.is_open()) {
>+      RTC_LOG(LS_WARNING)
>+          << debug_name_ << "->SendData(...): "
>+          << "Not sending data because sid is unknown or closing: "
>+          << params.sid;
>+      return false;
>+    }
>   }
> 
>   // Send data using SCTP.
>@@ -548,6 +511,7 @@ bool SctpTransport::SendData(const SendDataParams& params,
>   spa.sendv_flags |= SCTP_SEND_SNDINFO_VALID;
>   spa.sendv_sndinfo.snd_sid = params.sid;
>   spa.sendv_sndinfo.snd_ppid = rtc::HostToNetwork32(GetPpid(params.type));
>+  spa.sendv_sndinfo.snd_flags |= SCTP_EOR;
> 
>   // Ordered implies reliable.
>   if (!params.ordered) {
>@@ -591,18 +555,23 @@ bool SctpTransport::ReadyToSendData() {
>   return ready_to_send_data_;
> }
> 
>-void SctpTransport::ConnectTransportChannelSignals() {
>+void SctpTransport::ConnectTransportSignals() {
>   RTC_DCHECK_RUN_ON(network_thread_);
>-  transport_channel_->SignalWritableState.connect(
>-      this, &SctpTransport::OnWritableState);
>-  transport_channel_->SignalReadPacket.connect(this,
>-                                               &SctpTransport::OnPacketRead);
>+  if (!transport_) {
>+    return;
>+  }
>+  transport_->SignalWritableState.connect(this,
>+                                          &SctpTransport::OnWritableState);
>+  transport_->SignalReadPacket.connect(this, &SctpTransport::OnPacketRead);
> }
> 
>-void SctpTransport::DisconnectTransportChannelSignals() {
>+void SctpTransport::DisconnectTransportSignals() {
>   RTC_DCHECK_RUN_ON(network_thread_);
>-  transport_channel_->SignalWritableState.disconnect(this);
>-  transport_channel_->SignalReadPacket.disconnect(this);
>+  if (!transport_) {
>+    return;
>+  }
>+  transport_->SignalWritableState.disconnect(this);
>+  transport_->SignalReadPacket.disconnect(this);
> }
> 
> bool SctpTransport::Connect() {
>@@ -652,7 +621,9 @@ bool SctpTransport::Connect() {
>   sctp_paddrparams params = {{0}};
>   memcpy(&params.spp_address, &remote_sconn, sizeof(remote_sconn));
>   params.spp_flags = SPP_PMTUD_DISABLE;
>-  params.spp_pathmtu = kSctpMtu;
>+  // The MTU value provided specifies the space available for chunks in the
>+  // packet, so we subtract the SCTP header size.
>+  params.spp_pathmtu = kSctpMtu - sizeof(struct sctp_common_header);
>   if (usrsctp_setsockopt(sock_, IPPROTO_SCTP, SCTP_PEER_ADDR_PARAMS, &params,
>                          sizeof(params))) {
>     RTC_LOG_ERRNO(LS_ERROR) << debug_name_ << "->Connect(): "
>@@ -746,6 +717,15 @@ bool SctpTransport::ConfigureSctpSocket() {
>     return false;
>   }
> 
>+  // Explicit EOR.
>+  uint32_t eor = 1;
>+  if (usrsctp_setsockopt(sock_, IPPROTO_SCTP, SCTP_EXPLICIT_EOR, &eor,
>+                         sizeof(eor))) {
>+    RTC_LOG_ERRNO(LS_ERROR) << debug_name_ << "->ConfigureSctpSocket(): "
>+                            << "Failed to set SCTP_EXPLICIT_EOR.";
>+    return false;
>+  }
>+
>   // Subscribe to SCTP event notifications.
>   int event_types[] = {SCTP_ASSOC_CHANGE, SCTP_PEER_ADDR_CHANGE,
>                        SCTP_SEND_FAILED_EVENT, SCTP_SENDER_DRY_EVENT,
>@@ -783,46 +763,63 @@ void SctpTransport::CloseSctpSocket() {
> 
> bool SctpTransport::SendQueuedStreamResets() {
>   RTC_DCHECK_RUN_ON(network_thread_);
>-  if (!sent_reset_streams_.empty() || queued_reset_streams_.empty()) {
>+
>+  // Figure out how many streams need to be reset. We need to do this so we can
>+  // allocate the right amount of memory for the sctp_reset_streams structure.
>+  size_t num_streams = std::count_if(
>+      stream_status_by_sid_.begin(), stream_status_by_sid_.end(),
>+      [](const std::map<uint32_t, StreamStatus>::value_type& stream) {
>+        return stream.second.need_outgoing_reset();
>+      });
>+  if (num_streams == 0) {
>+    // Nothing to reset.
>     return true;
>   }
> 
>   RTC_LOG(LS_VERBOSE) << "SendQueuedStreamResets[" << debug_name_
>-                      << "]: Sending [" << ListStreams(queued_reset_streams_)
>-                      << "], Open: [" << ListStreams(open_streams_)
>-                      << "], Sent: [" << ListStreams(sent_reset_streams_)
>-                      << "]";
>+                      << "]: Resetting " << num_streams << " outgoing streams.";
> 
>-  const size_t num_streams = queued_reset_streams_.size();
>   const size_t num_bytes =
>       sizeof(struct sctp_reset_streams) + (num_streams * sizeof(uint16_t));
>-
>   std::vector<uint8_t> reset_stream_buf(num_bytes, 0);
>   struct sctp_reset_streams* resetp =
>       reinterpret_cast<sctp_reset_streams*>(&reset_stream_buf[0]);
>   resetp->srs_assoc_id = SCTP_ALL_ASSOC;
>-  resetp->srs_flags = SCTP_STREAM_RESET_INCOMING | SCTP_STREAM_RESET_OUTGOING;
>+  resetp->srs_flags = SCTP_STREAM_RESET_OUTGOING;
>   resetp->srs_number_streams = rtc::checked_cast<uint16_t>(num_streams);
>   int result_idx = 0;
>-  for (StreamSet::iterator it = queued_reset_streams_.begin();
>-       it != queued_reset_streams_.end(); ++it) {
>-    resetp->srs_stream_list[result_idx++] = *it;
>+
>+  for (const std::map<uint32_t, StreamStatus>::value_type& stream :
>+       stream_status_by_sid_) {
>+    if (!stream.second.need_outgoing_reset()) {
>+      continue;
>+    }
>+    resetp->srs_stream_list[result_idx++] = stream.first;
>   }
> 
>   int ret =
>       usrsctp_setsockopt(sock_, IPPROTO_SCTP, SCTP_RESET_STREAMS, resetp,
>                          rtc::checked_cast<socklen_t>(reset_stream_buf.size()));
>   if (ret < 0) {
>-    RTC_LOG_ERRNO(LS_ERROR) << debug_name_
>-                            << "->SendQueuedStreamResets(): "
>-                               "Failed to send a stream reset for "
>-                            << num_streams << " streams";
>+    // Note that usrsctp only lets us have one reset in progress at a time
>+    // (even though multiple streams can be reset at once). If this happens,
>+    // SendQueuedStreamResets will end up called after the current in-progress
>+    // reset finishes, in OnStreamResetEvent.
>+    RTC_LOG_ERRNO(LS_WARNING) << debug_name_
>+                              << "->SendQueuedStreamResets(): "
>+                                 "Failed to send a stream reset for "
>+                              << num_streams << " streams";
>     return false;
>   }
> 
>-  // sent_reset_streams_ is empty, and all the queued_reset_streams_ go into
>-  // it now.
>-  queued_reset_streams_.swap(sent_reset_streams_);
>+  // Since the usrsctp call completed successfully, update our stream status
>+  // map to note that we started the outgoing reset.
>+  for (auto it = stream_status_by_sid_.begin();
>+       it != stream_status_by_sid_.end(); ++it) {
>+    if (it->second.need_outgoing_reset()) {
>+      it->second.outgoing_reset_initiated = true;
>+    }
>+  }
>   return true;
> }
> 
>@@ -836,7 +833,7 @@ void SctpTransport::SetReadyToSendData() {
> 
> void SctpTransport::OnWritableState(rtc::PacketTransportInternal* transport) {
>   RTC_DCHECK_RUN_ON(network_thread_);
>-  RTC_DCHECK_EQ(transport_channel_, transport);
>+  RTC_DCHECK_EQ(transport_, transport);
>   if (!was_ever_writable_ && transport->writable()) {
>     was_ever_writable_ = true;
>     if (started_) {
>@@ -852,7 +849,7 @@ void SctpTransport::OnPacketRead(rtc::PacketTransportInternal* transport,
>                                  const rtc::PacketTime& packet_time,
>                                  int flags) {
>   RTC_DCHECK_RUN_ON(network_thread_);
>-  RTC_DCHECK_EQ(transport_channel_, transport);
>+  RTC_DCHECK_EQ(transport_, transport);
>   TRACE_EVENT0("webrtc", "SctpTransport::OnPacketRead");
> 
>   if (flags & PF_SRTP_BYPASS) {
>@@ -906,24 +903,24 @@ void SctpTransport::OnPacketFromSctpToNetwork(
>   }
>   TRACE_EVENT0("webrtc", "SctpTransport::OnPacketFromSctpToNetwork");
> 
>-  // Don't create noise by trying to send a packet when the DTLS channel isn't
>+  // Don't create noise by trying to send a packet when the DTLS transport isn't
>   // even writable.
>-  if (!transport_channel_->writable()) {
>+  if (!transport_ || !transport_->writable()) {
>     return;
>   }
> 
>   // Bon voyage.
>-  transport_channel_->SendPacket(buffer.data<char>(), buffer.size(),
>-                                 rtc::PacketOptions(), PF_NORMAL);
>+  transport_->SendPacket(buffer.data<char>(), buffer.size(),
>+                         rtc::PacketOptions(), PF_NORMAL);
> }
> 
>-void SctpTransport::OnInboundPacketFromSctpToChannel(
>+void SctpTransport::OnInboundPacketFromSctpToTransport(
>     const rtc::CopyOnWriteBuffer& buffer,
>     ReceiveDataParams params,
>     int flags) {
>   RTC_DCHECK_RUN_ON(network_thread_);
>   RTC_LOG(LS_VERBOSE) << debug_name_
>-                      << "->OnInboundPacketFromSctpToChannel(...): "
>+                      << "->OnInboundPacketFromSctpToTransport(...): "
>                       << "Received SCTP data:"
>                       << " sid=" << params.sid
>                       << " notification: " << (flags & MSG_NOTIFICATION)
>@@ -932,22 +929,22 @@ void SctpTransport::OnInboundPacketFromSctpToChannel(
>   // connection" message. This sets sock_ = NULL;
>   if (!buffer.size() || !buffer.data()) {
>     RTC_LOG(LS_INFO) << debug_name_
>-                     << "->OnInboundPacketFromSctpToChannel(...): "
>+                     << "->OnInboundPacketFromSctpToTransport(...): "
>                         "No data, closing.";
>     return;
>   }
>   if (flags & MSG_NOTIFICATION) {
>     OnNotificationFromSctp(buffer);
>   } else {
>-    OnDataFromSctpToChannel(params, buffer);
>+    OnDataFromSctpToTransport(params, buffer);
>   }
> }
> 
>-void SctpTransport::OnDataFromSctpToChannel(
>+void SctpTransport::OnDataFromSctpToTransport(
>     const ReceiveDataParams& params,
>     const rtc::CopyOnWriteBuffer& buffer) {
>   RTC_DCHECK_RUN_ON(network_thread_);
>-  RTC_LOG(LS_VERBOSE) << debug_name_ << "->OnDataFromSctpToChannel(...): "
>+  RTC_LOG(LS_VERBOSE) << debug_name_ << "->OnDataFromSctpToTransport(...): "
>                       << "Posting with length: " << buffer.size()
>                       << " on stream " << params.sid;
>   // Reports all received messages to upper layers, no matter whether the sid
>@@ -1042,78 +1039,73 @@ void SctpTransport::OnNotificationAssocChange(const sctp_assoc_change& change) {
> void SctpTransport::OnStreamResetEvent(
>     const struct sctp_stream_reset_event* evt) {
>   RTC_DCHECK_RUN_ON(network_thread_);
>-  // A stream reset always involves two RE-CONFIG chunks for us -- we always
>-  // simultaneously reset a sid's sequence number in both directions.  The
>-  // requesting side transmits a RE-CONFIG chunk and waits for the peer to send
>-  // one back.  Both sides get this SCTP_STREAM_RESET_EVENT when they receive
>-  // RE-CONFIGs.
>+
>+  // This callback indicates that a reset is complete for incoming and/or
>+  // outgoing streams. The reset may have been initiated by us or the remote
>+  // side.
>   const int num_sids = (evt->strreset_length - sizeof(*evt)) /
>                        sizeof(evt->strreset_stream_list[0]);
>-  RTC_LOG(LS_VERBOSE) << "SCTP_STREAM_RESET_EVENT(" << debug_name_
>-                      << "): Flags = 0x" << std::hex << evt->strreset_flags
>-                      << " (" << ListFlags(evt->strreset_flags) << ")";
>-  RTC_LOG(LS_VERBOSE) << "Assoc = " << evt->strreset_assoc_id << ", Streams = ["
>-                      << ListArray(evt->strreset_stream_list, num_sids)
>-                      << "], Open: [" << ListStreams(open_streams_)
>-                      << "], Q'd: [" << ListStreams(queued_reset_streams_)
>-                      << "], Sent: [" << ListStreams(sent_reset_streams_)
>-                      << "]";
>-
>-  // If both sides try to reset some streams at the same time (even if they're
>-  // disjoint sets), we can get reset failures.
>+
>   if (evt->strreset_flags & SCTP_STREAM_RESET_FAILED) {
>-    // OK, just try again.  The stream IDs sent over when the RESET_FAILED flag
>-    // is set seem to be garbage values.  Ignore them.
>-    queued_reset_streams_.insert(sent_reset_streams_.begin(),
>-                                 sent_reset_streams_.end());
>-    sent_reset_streams_.clear();
>-
>-  } else if (evt->strreset_flags & SCTP_STREAM_RESET_INCOMING_SSN) {
>-    // Each side gets an event for each direction of a stream.  That is,
>-    // closing sid k will make each side receive INCOMING and OUTGOING reset
>-    // events for k.  As per RFC6525, Section 5, paragraph 2, each side will
>-    // get an INCOMING event first.
>-    for (int i = 0; i < num_sids; i++) {
>-      const int stream_id = evt->strreset_stream_list[i];
>-
>-      // See if this stream ID was closed by our peer or ourselves.
>-      StreamSet::iterator it = sent_reset_streams_.find(stream_id);
>-
>-      // The reset was requested locally.
>-      if (it != sent_reset_streams_.end()) {
>-        RTC_LOG(LS_VERBOSE) << "SCTP_STREAM_RESET_EVENT(" << debug_name_
>-                            << "): local sid " << stream_id << " acknowledged.";
>-        sent_reset_streams_.erase(it);
>-
>-      } else if ((it = open_streams_.find(stream_id)) != open_streams_.end()) {
>-        // The peer requested the reset.
>-        RTC_LOG(LS_VERBOSE) << "SCTP_STREAM_RESET_EVENT(" << debug_name_
>-                            << "): closing sid " << stream_id;
>-        open_streams_.erase(it);
>-        SignalStreamClosedRemotely(stream_id);
>-
>-      } else if ((it = queued_reset_streams_.find(stream_id)) !=
>-                 queued_reset_streams_.end()) {
>-        // The peer requested the reset, but there was a local reset
>-        // queued.
>-        RTC_LOG(LS_VERBOSE) << "SCTP_STREAM_RESET_EVENT(" << debug_name_
>-                            << "): double-sided close for sid " << stream_id;
>-        // Both sides want the stream closed, and the peer got to send the
>-        // RE-CONFIG first.  Treat it like the local Remove(Send|Recv)Stream
>-        // finished quickly.
>-        queued_reset_streams_.erase(it);
>-
>-      } else {
>-        // This stream is unknown.  Sometimes this can be from an
>-        // RESET_FAILED-related retransmit.
>-        RTC_LOG(LS_VERBOSE) << "SCTP_STREAM_RESET_EVENT(" << debug_name_
>-                            << "): Unknown sid " << stream_id;
>+    // OK, just try sending any previously sent stream resets again. The stream
>+    // IDs sent over when the RESET_FIALED flag is set seem to be garbage
>+    // values. Ignore them.
>+    for (std::map<uint32_t, StreamStatus>::value_type& stream :
>+         stream_status_by_sid_) {
>+      stream.second.outgoing_reset_initiated = false;
>+    }
>+    SendQueuedStreamResets();
>+    // TODO(deadbeef): If this happens, the entire SCTP association is in quite
>+    // crippled state. The SCTP session should be dismantled, and the WebRTC
>+    // connectivity errored because is clear that the distant party is not
>+    // playing ball: malforms the transported data.
>+    return;
>+  }
>+
>+  // Loop over the received events and properly update the StreamStatus map.
>+  for (int i = 0; i < num_sids; i++) {
>+    const uint32_t sid = evt->strreset_stream_list[i];
>+    auto it = stream_status_by_sid_.find(sid);
>+    if (it == stream_status_by_sid_.end()) {
>+      // This stream is unknown. Sometimes this can be from a
>+      // RESET_FAILED-related retransmit.
>+      RTC_LOG(LS_VERBOSE) << "SCTP_STREAM_RESET_EVENT(" << debug_name_
>+                          << "): Unknown sid " << sid;
>+      continue;
>+    }
>+    StreamStatus& status = it->second;
>+
>+    if (evt->strreset_flags & SCTP_STREAM_RESET_INCOMING_SSN) {
>+      RTC_LOG(LS_VERBOSE) << "SCTP_STREAM_RESET_INCOMING_SSN(" << debug_name_
>+                          << "): sid " << sid;
>+      status.incoming_reset_complete = true;
>+      // If we receive an incoming stream reset and we haven't started the
>+      // closing procedure ourselves, this means the remote side started the
>+      // closing procedure; fire a signal so that the relevant data channel
>+      // can change to "closing" (we still need to reset the outgoing stream
>+      // before it changes to "closed").
>+      if (!status.closure_initiated) {
>+        SignalClosingProcedureStartedRemotely(sid);
>       }
>     }
>+    if (evt->strreset_flags & SCTP_STREAM_RESET_OUTGOING_SSN) {
>+      RTC_LOG(LS_VERBOSE) << "SCTP_STREAM_RESET_OUTGOING_SSN(" << debug_name_
>+                          << "): sid " << sid;
>+      status.outgoing_reset_complete = true;
>+    }
>+
>+    // If this reset completes the closing procedure, remove the stream from
>+    // our map so we can consider it closed, and fire a signal such that the
>+    // relevant DataChannel will change its state to "closed" and its ID can be
>+    // re-used.
>+    if (status.reset_complete()) {
>+      stream_status_by_sid_.erase(it);
>+      SignalClosingProcedureComplete(sid);
>+    }
>   }
> 
>-  // Always try to send the queued RESET because this call indicates that the
>-  // last local RESET or remote RESET has made some progress.
>+  // Always try to send any queued resets because this call indicates that the
>+  // last outgoing or incoming reset has made some progress.
>   SendQueuedStreamResets();
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/sctp/sctptransport.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/sctp/sctptransport.h
>index e99784dbe37..85c66271119 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/sctp/sctptransport.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/sctp/sctptransport.h
>@@ -13,6 +13,7 @@
> 
> #include <errno.h>
> 
>+#include <map>
> #include <memory>  // for unique_ptr.
> #include <set>
> #include <string>
>@@ -21,7 +22,7 @@
> #include "rtc_base/asyncinvoker.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/copyonwritebuffer.h"
>-#include "rtc_base/sigslot.h"
>+#include "rtc_base/third_party/sigslot/sigslot.h"
> #include "rtc_base/thread.h"
> // For SendDataParams/ReceiveDataParams.
> #include "media/base/mediachannel.h"
>@@ -46,15 +47,15 @@ struct SctpInboundPacket;
> //  3.  OnSctpOutboundPacket(wrapped_data)
> // [sctp thread returns having async invoked on the network thread]
> //  4.  SctpTransport::OnPacketFromSctpToNetwork(wrapped_data)
>-//  5.  TransportChannel::SendPacket(wrapped_data)
>+//  5.  DtlsTransport::SendPacket(wrapped_data)
> //  6.  ... across network ... a packet is sent back ...
> //  7.  SctpTransport::OnPacketReceived(wrapped_data)
> //  8.  usrsctp_conninput(wrapped_data)
> // [network thread returns; sctp thread then calls the following]
> //  9.  OnSctpInboundData(data)
> // [sctp thread returns having async invoked on the network thread]
>-//  10. SctpTransport::OnInboundPacketFromSctpToChannel(inboundpacket)
>-//  11. SctpTransport::OnDataFromSctpToChannel(data)
>+//  10. SctpTransport::OnInboundPacketFromSctpToTransport(inboundpacket)
>+//  11. SctpTransport::OnDataFromSctpToTransport(data)
> //  12. SctpTransport::SignalDataReceived(data)
> // [from the same thread, methods registered/connected to
> //  SctpTransport are called with the recieved data]
>@@ -71,7 +72,7 @@ class SctpTransport : public SctpTransportInternal,
>   ~SctpTransport() override;
> 
>   // SctpTransportInternal overrides (see sctptransportinternal.h for comments).
>-  void SetTransportChannel(rtc::PacketTransportInternal* channel) override;
>+  void SetDtlsTransport(rtc::PacketTransportInternal* transport) override;
>   bool Start(int local_port, int remote_port) override;
>   bool OpenStream(int sid) override;
>   bool ResetStream(int sid) override;
>@@ -88,8 +89,8 @@ class SctpTransport : public SctpTransportInternal,
>   rtc::Thread* network_thread() const { return network_thread_; }
> 
>  private:
>-  void ConnectTransportChannelSignals();
>-  void DisconnectTransportChannelSignals();
>+  void ConnectTransportSignals();
>+  void DisconnectTransportSignals();
> 
>   // Creates the socket and connects.
>   bool Connect();
>@@ -124,11 +125,11 @@ class SctpTransport : public SctpTransportInternal,
>   // Called using |invoker_| to decide what to do with the packet.
>   // The |flags| parameter is used by SCTP to distinguish notification packets
>   // from other types of packets.
>-  void OnInboundPacketFromSctpToChannel(const rtc::CopyOnWriteBuffer& buffer,
>-                                        ReceiveDataParams params,
>-                                        int flags);
>-  void OnDataFromSctpToChannel(const ReceiveDataParams& params,
>-                               const rtc::CopyOnWriteBuffer& buffer);
>+  void OnInboundPacketFromSctpToTransport(const rtc::CopyOnWriteBuffer& buffer,
>+                                          ReceiveDataParams params,
>+                                          int flags);
>+  void OnDataFromSctpToTransport(const ReceiveDataParams& params,
>+                                 const rtc::CopyOnWriteBuffer& buffer);
>   void OnNotificationFromSctp(const rtc::CopyOnWriteBuffer& buffer);
>   void OnNotificationAssocChange(const sctp_assoc_change& change);
> 
>@@ -140,7 +141,13 @@ class SctpTransport : public SctpTransportInternal,
>   // Helps pass inbound/outbound packets asynchronously to the network thread.
>   rtc::AsyncInvoker invoker_;
>   // Underlying DTLS channel.
>-  rtc::PacketTransportInternal* transport_channel_;
>+  rtc::PacketTransportInternal* transport_ = nullptr;
>+
>+  // Track the data received from usrsctp between callbacks until the EOR bit
>+  // arrives.
>+  rtc::CopyOnWriteBuffer partial_message_;
>+  int partial_message_sid_;
>+
>   bool was_ever_writable_ = false;
>   int local_port_ = kSctpDefaultPort;
>   int remote_port_ = kSctpDefaultPort;
>@@ -149,21 +156,51 @@ class SctpTransport : public SctpTransportInternal,
>   // Has Start been called? Don't create SCTP socket until it has.
>   bool started_ = false;
>   // Are we ready to queue data (SCTP socket created, and not blocked due to
>-  // congestion control)? Different than |transport_channel_|'s "ready to
>-  // send".
>+  // congestion control)? Different than |transport_|'s "ready to send".
>   bool ready_to_send_data_ = false;
> 
>-  typedef std::set<uint32_t> StreamSet;
>-  // When a data channel opens a stream, it goes into open_streams_.  When we
>-  // want to close it, the stream's ID goes into queued_reset_streams_.  When
>-  // we actually transmit a RE-CONFIG chunk with that stream ID, the ID goes
>-  // into sent_reset_streams_.  When we get a response RE-CONFIG chunk back
>-  // acknowledging the reset, we remove the stream ID from
>-  // sent_reset_streams_.  We use sent_reset_streams_ to differentiate
>-  // between acknowledgment RE-CONFIG and peer-initiated RE-CONFIGs.
>-  StreamSet open_streams_;
>-  StreamSet queued_reset_streams_;
>-  StreamSet sent_reset_streams_;
>+  // Used to keep track of the status of each stream (or rather, each pair of
>+  // incoming/outgoing streams with matching IDs). It's specifically used to
>+  // keep track of the status of resets, but more information could be put here
>+  // later.
>+  //
>+  // See datachannel.h for a summary of the closing procedure.
>+  struct StreamStatus {
>+    // Closure initiated by application via ResetStream? Note that
>+    // this may be true while outgoing_reset_initiated is false if the outgoing
>+    // reset needed to be queued.
>+    bool closure_initiated = false;
>+    // Whether we've initiated the outgoing stream reset via
>+    // SCTP_RESET_STREAMS.
>+    bool outgoing_reset_initiated = false;
>+    // Whether usrsctp has indicated that the incoming/outgoing streams have
>+    // been reset. It's expected that the peer will reset its outgoing stream
>+    // (our incoming stream) after receiving the reset for our outgoing stream,
>+    // though older versions of chromium won't do this. See crbug.com/559394
>+    // for context.
>+    bool outgoing_reset_complete = false;
>+    bool incoming_reset_complete = false;
>+
>+    // Some helper methods to improve code readability.
>+    bool is_open() const {
>+      return !closure_initiated && !incoming_reset_complete &&
>+             !outgoing_reset_complete;
>+    }
>+    // We need to send an outgoing reset if the application has closed the data
>+    // channel, or if we received a reset of the incoming stream from the
>+    // remote endpoint, indicating the data channel was closed remotely.
>+    bool need_outgoing_reset() const {
>+      return (incoming_reset_complete || closure_initiated) &&
>+             !outgoing_reset_initiated;
>+    }
>+    bool reset_complete() const {
>+      return outgoing_reset_complete && incoming_reset_complete;
>+    }
>+  };
>+
>+  // Entries should only be removed from this map if |reset_complete| is
>+  // true.
>+  std::map<uint32_t, StreamStatus> stream_status_by_sid_;
> 
>   // A static human-readable name for debugging messages.
>   const char* debug_name_ = "SctpTransport";
>@@ -179,9 +216,9 @@ class SctpTransportFactory : public SctpTransportInternalFactory {
>       : network_thread_(network_thread) {}
> 
>   std::unique_ptr<SctpTransportInternal> CreateSctpTransport(
>-      rtc::PacketTransportInternal* channel) override {
>+      rtc::PacketTransportInternal* transport) override {
>     return std::unique_ptr<SctpTransportInternal>(
>-        new SctpTransport(network_thread_, channel));
>+        new SctpTransport(network_thread_, transport));
>   }
> 
>  private:
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/sctp/sctptransport_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/sctp/sctptransport_unittest.cc
>index 32e439a1883..75c2796cf81 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/sctp/sctptransport_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/sctp/sctptransport_unittest.cc
>@@ -31,7 +31,7 @@ static const int kDefaultTimeout = 10000;  // 10 seconds.
> // Use ports other than the default 5000 for testing.
> static const int kTransport1Port = 5001;
> static const int kTransport2Port = 5002;
>-}
>+}  // namespace
> 
> namespace cricket {
> 
>@@ -67,56 +67,54 @@ class SctpFakeDataReceiver : public sigslot::has_slots<> {
>   ReceiveDataParams last_params_;
> };
> 
>-class SignalReadyToSendObserver : public sigslot::has_slots<> {
>+class SctpTransportObserver : public sigslot::has_slots<> {
>  public:
>-  SignalReadyToSendObserver() : signaled_(false) {}
>-
>-  void OnSignaled() { signaled_ = true; }
>-
>-  bool IsSignaled() { return signaled_; }
>-
>- private:
>-  bool signaled_;
>-};
>-
>-class SignalTransportClosedObserver : public sigslot::has_slots<> {
>- public:
>-  SignalTransportClosedObserver() {}
>-  void BindSelf(SctpTransport* transport) {
>-    transport->SignalStreamClosedRemotely.connect(
>-        this, &SignalTransportClosedObserver::OnStreamClosed);
>+  explicit SctpTransportObserver(SctpTransport* transport) {
>+    transport->SignalClosingProcedureComplete.connect(
>+        this, &SctpTransportObserver::OnClosingProcedureComplete);
>+    transport->SignalReadyToSendData.connect(
>+        this, &SctpTransportObserver::OnReadyToSend);
>   }
>-  void OnStreamClosed(int stream) { streams_.push_back(stream); }
> 
>   int StreamCloseCount(int stream) {
>-    return std::count(streams_.begin(), streams_.end(), stream);
>+    return std::count(closed_streams_.begin(), closed_streams_.end(), stream);
>   }
> 
>   bool WasStreamClosed(int stream) {
>-    return std::find(streams_.begin(), streams_.end(), stream) !=
>-           streams_.end();
>+    return std::find(closed_streams_.begin(), closed_streams_.end(), stream) !=
>+           closed_streams_.end();
>   }
> 
>+  bool ReadyToSend() { return ready_to_send_; }
>+
>  private:
>-  std::vector<int> streams_;
>+  void OnClosingProcedureComplete(int stream) {
>+    closed_streams_.push_back(stream);
>+  }
>+  void OnReadyToSend() { ready_to_send_ = true; }
>+
>+  std::vector<int> closed_streams_;
>+  bool ready_to_send_ = false;
> };
> 
>+// Helper class used to immediately attempt to reopen a stream as soon as it's
>+// been closed.
> class SignalTransportClosedReopener : public sigslot::has_slots<> {
>  public:
>   SignalTransportClosedReopener(SctpTransport* transport, SctpTransport* peer)
>       : transport_(transport), peer_(peer) {}
> 
>+  int StreamCloseCount(int stream) {
>+    return std::count(streams_.begin(), streams_.end(), stream);
>+  }
>+
>+ private:
>   void OnStreamClosed(int stream) {
>     transport_->OpenStream(stream);
>     peer_->OpenStream(stream);
>     streams_.push_back(stream);
>   }
> 
>-  int StreamCloseCount(int stream) {
>-    return std::count(streams_.begin(), streams_.end(), stream);
>-  }
>-
>- private:
>   SctpTransport* transport_;
>   SctpTransport* peer_;
>   std::vector<int> streams_;
>@@ -238,16 +236,16 @@ class SctpTransportTest : public testing::Test, public sigslot::has_slots<> {
> };
> 
> // Test that data can be sent end-to-end when an SCTP transport starts with one
>-// transport channel (which is unwritable), and then switches to another
>-// channel. A common scenario due to how BUNDLE works.
>-TEST_F(SctpTransportTest, SwitchTransportChannel) {
>+// transport (which is unwritable), and then switches to another transport. A
>+// common scenario due to how BUNDLE works.
>+TEST_F(SctpTransportTest, SwitchDtlsTransport) {
>   FakeDtlsTransport black_hole("black hole", 0);
>   FakeDtlsTransport fake_dtls1("fake dtls 1", 0);
>   FakeDtlsTransport fake_dtls2("fake dtls 2", 0);
>   SctpFakeDataReceiver recv1;
>   SctpFakeDataReceiver recv2;
> 
>-  // Construct transport1 with the "black hole" channel.
>+  // Construct transport1 with the "black hole" transport.
>   std::unique_ptr<SctpTransport> transport1(
>       CreateTransport(&black_hole, &recv1));
>   std::unique_ptr<SctpTransport> transport2(
>@@ -261,10 +259,10 @@ TEST_F(SctpTransportTest, SwitchTransportChannel) {
>   transport1->Start(kTransport1Port, kTransport2Port);
>   transport2->Start(kTransport2Port, kTransport1Port);
> 
>-  // Switch transport1_ to the normal fake_dtls1_ channel.
>-  transport1->SetTransportChannel(&fake_dtls1);
>+  // Switch transport1_ to the normal fake_dtls1_ transport.
>+  transport1->SetDtlsTransport(&fake_dtls1);
> 
>-  // Connect the two fake DTLS channels.
>+  // Connect the two fake DTLS transports.
>   bool asymmetric = false;
>   fake_dtls1.SetDestination(&fake_dtls2, asymmetric);
> 
>@@ -274,6 +272,10 @@ TEST_F(SctpTransportTest, SwitchTransportChannel) {
>   ASSERT_TRUE(SendData(transport2.get(), 1, "bar", &result));
>   EXPECT_TRUE_WAIT(ReceivedData(&recv2, 1, "foo"), kDefaultTimeout);
>   EXPECT_TRUE_WAIT(ReceivedData(&recv1, 1, "bar"), kDefaultTimeout);
>+
>+  // Setting a null DtlsTransport should work. This could happen when an SCTP
>+  // data section is rejected.
>+  transport1->SetDtlsTransport(nullptr);
> }
> 
> // Calling Start twice shouldn't do anything bad, if with the same parameters.
>@@ -317,7 +319,7 @@ TEST_F(SctpTransportTest, NegativeOnePortTreatedAsDefault) {
>   transport1->Start(kSctpDefaultPort, kSctpDefaultPort);
>   transport2->Start(-1, -1);
> 
>-  // Connect the two fake DTLS channels.
>+  // Connect the two fake DTLS transports.
>   bool asymmetric = false;
>   fake_dtls1.SetDestination(&fake_dtls2, asymmetric);
> 
>@@ -347,19 +349,16 @@ TEST_F(SctpTransportTest, ResetStreamWithAlreadyResetStreamFails) {
> }
> 
> // Test that SignalReadyToSendData is fired after Start has been called and the
>-// DTLS channel is writable.
>+// DTLS transport is writable.
> TEST_F(SctpTransportTest, SignalReadyToSendDataAfterDtlsWritable) {
>   FakeDtlsTransport fake_dtls("fake dtls", 0);
>   SctpFakeDataReceiver recv;
>   std::unique_ptr<SctpTransport> transport(CreateTransport(&fake_dtls, &recv));
>-
>-  SignalReadyToSendObserver signal_observer;
>-  transport->SignalReadyToSendData.connect(
>-      &signal_observer, &SignalReadyToSendObserver::OnSignaled);
>+  SctpTransportObserver observer(transport.get());
> 
>   transport->Start(kSctpDefaultPort, kSctpDefaultPort);
>   fake_dtls.SetWritable(true);
>-  EXPECT_TRUE_WAIT(signal_observer.IsSignaled(), kDefaultTimeout);
>+  EXPECT_TRUE_WAIT(observer.ReadyToSend(), kDefaultTimeout);
> }
> 
> // Test that after an SCTP socket's buffer is filled, SignalReadyToSendData
>@@ -472,10 +471,8 @@ TEST_F(SctpTransportTest, SendDataHighPorts) {
> 
> TEST_F(SctpTransportTest, ClosesRemoteStream) {
>   SetupConnectedTransportsWithTwoStreams();
>-  SignalTransportClosedObserver transport1_sig_receiver,
>-      transport2_sig_receiver;
>-  transport1_sig_receiver.BindSelf(transport1());
>-  transport2_sig_receiver.BindSelf(transport2());
>+  SctpTransportObserver transport1_observer(transport1());
>+  SctpTransportObserver transport2_observer(transport2());
> 
>   SendDataResult result;
>   ASSERT_TRUE(SendData(transport1(), 1, "hello?", &result));
>@@ -486,18 +483,16 @@ TEST_F(SctpTransportTest, ClosesRemoteStream) {
>   EXPECT_TRUE_WAIT(ReceivedData(receiver1(), 2, "hi transport1"),
>                    kDefaultTimeout);
> 
>-  // Close transport 1.  Transport 2 should notify us.
>+  // Close stream 1 on transport 1. Transport 2 should notify us.
>   transport1()->ResetStream(1);
>-  EXPECT_TRUE_WAIT(transport2_sig_receiver.WasStreamClosed(1), kDefaultTimeout);
>+  EXPECT_TRUE_WAIT(transport2_observer.WasStreamClosed(1), kDefaultTimeout);
> }
> 
> TEST_F(SctpTransportTest, ClosesTwoRemoteStreams) {
>   SetupConnectedTransportsWithTwoStreams();
>   AddStream(3);
>-  SignalTransportClosedObserver transport1_sig_receiver,
>-      transport2_sig_receiver;
>-  transport1_sig_receiver.BindSelf(transport1());
>-  transport2_sig_receiver.BindSelf(transport2());
>+  SctpTransportObserver transport1_observer(transport1());
>+  SctpTransportObserver transport2_observer(transport2());
> 
>   SendDataResult result;
>   ASSERT_TRUE(SendData(transport1(), 1, "hello?", &result));
>@@ -511,18 +506,16 @@ TEST_F(SctpTransportTest, ClosesTwoRemoteStreams) {
>   // Close two streams on one side.
>   transport2()->ResetStream(2);
>   transport2()->ResetStream(3);
>-  EXPECT_TRUE_WAIT(transport1_sig_receiver.WasStreamClosed(2), kDefaultTimeout);
>-  EXPECT_TRUE_WAIT(transport1_sig_receiver.WasStreamClosed(3), kDefaultTimeout);
>+  EXPECT_TRUE_WAIT(transport2_observer.WasStreamClosed(2), kDefaultTimeout);
>+  EXPECT_TRUE_WAIT(transport2_observer.WasStreamClosed(3), kDefaultTimeout);
> }
> 
> TEST_F(SctpTransportTest, ClosesStreamsOnBothSides) {
>   SetupConnectedTransportsWithTwoStreams();
>   AddStream(3);
>   AddStream(4);
>-  SignalTransportClosedObserver transport1_sig_receiver,
>-      transport2_sig_receiver;
>-  transport1_sig_receiver.BindSelf(transport1());
>-  transport2_sig_receiver.BindSelf(transport2());
>+  SctpTransportObserver transport1_observer(transport1());
>+  SctpTransportObserver transport2_observer(transport2());
> 
>   SendDataResult result;
>   ASSERT_TRUE(SendData(transport1(), 1, "hello?", &result));
>@@ -541,10 +534,10 @@ TEST_F(SctpTransportTest, ClosesStreamsOnBothSides) {
>   transport2()->ResetStream(2);
>   transport2()->ResetStream(3);
>   transport2()->ResetStream(4);
>-  EXPECT_TRUE_WAIT(transport2_sig_receiver.WasStreamClosed(1), kDefaultTimeout);
>-  EXPECT_TRUE_WAIT(transport1_sig_receiver.WasStreamClosed(2), kDefaultTimeout);
>-  EXPECT_TRUE_WAIT(transport1_sig_receiver.WasStreamClosed(3), kDefaultTimeout);
>-  EXPECT_TRUE_WAIT(transport1_sig_receiver.WasStreamClosed(4), kDefaultTimeout);
>+  EXPECT_TRUE_WAIT(transport2_observer.WasStreamClosed(1), kDefaultTimeout);
>+  EXPECT_TRUE_WAIT(transport1_observer.WasStreamClosed(2), kDefaultTimeout);
>+  EXPECT_TRUE_WAIT(transport1_observer.WasStreamClosed(3), kDefaultTimeout);
>+  EXPECT_TRUE_WAIT(transport1_observer.WasStreamClosed(4), kDefaultTimeout);
> }
> 
> TEST_F(SctpTransportTest, RefusesHighNumberedTransports) {
>@@ -553,20 +546,18 @@ TEST_F(SctpTransportTest, RefusesHighNumberedTransports) {
>   EXPECT_FALSE(AddStream(kMaxSctpSid + 1));
> }
> 
>-// Flaky, see webrtc:4453.
>-TEST_F(SctpTransportTest, DISABLED_ReusesAStream) {
>+TEST_F(SctpTransportTest, ReusesAStream) {
>   // Shut down transport 1, then open it up again for reuse.
>   SetupConnectedTransportsWithTwoStreams();
>   SendDataResult result;
>-  SignalTransportClosedObserver transport2_sig_receiver;
>-  transport2_sig_receiver.BindSelf(transport2());
>+  SctpTransportObserver transport2_observer(transport2());
> 
>   ASSERT_TRUE(SendData(transport1(), 1, "hello?", &result));
>   EXPECT_EQ(SDR_SUCCESS, result);
>   EXPECT_TRUE_WAIT(ReceivedData(receiver2(), 1, "hello?"), kDefaultTimeout);
> 
>   transport1()->ResetStream(1);
>-  EXPECT_TRUE_WAIT(transport2_sig_receiver.WasStreamClosed(1), kDefaultTimeout);
>+  EXPECT_TRUE_WAIT(transport2_observer.WasStreamClosed(1), kDefaultTimeout);
>   // Transport 1 is gone now.
> 
>   // Create a new transport 1.
>@@ -575,8 +566,7 @@ TEST_F(SctpTransportTest, DISABLED_ReusesAStream) {
>   EXPECT_EQ(SDR_SUCCESS, result);
>   EXPECT_TRUE_WAIT(ReceivedData(receiver2(), 1, "hi?"), kDefaultTimeout);
>   transport1()->ResetStream(1);
>-  EXPECT_TRUE_WAIT(transport2_sig_receiver.StreamCloseCount(1) == 2,
>-                   kDefaultTimeout);
>+  EXPECT_EQ_WAIT(2, transport2_observer.StreamCloseCount(1), kDefaultTimeout);
> }
> 
> }  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/sctp/sctptransportinternal.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/sctp/sctptransportinternal.h
>index c8c606ed115..0380a870b4c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/sctp/sctptransportinternal.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/sctp/sctptransportinternal.h
>@@ -48,17 +48,15 @@ constexpr uint16_t kMinSctpSid = 0;
> // usrsctp.h)
> const int kSctpDefaultPort = 5000;
> 
>-// Abstract SctpTransport interface for use internally (by
>-// PeerConnection/WebRtcSession/etc.). Exists to allow mock/fake SctpTransports
>-// to be created.
>+// Abstract SctpTransport interface for use internally (by PeerConnection etc.).
>+// Exists to allow mock/fake SctpTransports to be created.
> class SctpTransportInternal {
>  public:
>   virtual ~SctpTransportInternal() {}
> 
>-  // Changes what underlying DTLS channel is uses. Used when switching which
>+  // Changes what underlying DTLS transport is uses. Used when switching which
>   // bundled transport the SctpTransport uses.
>-  // Assumes |channel| is non-null.
>-  virtual void SetTransportChannel(rtc::PacketTransportInternal* channel) = 0;
>+  virtual void SetDtlsTransport(rtc::PacketTransportInternal* transport) = 0;
> 
>   // When Start is called, connects as soon as possible; this can be called
>   // before DTLS completes, in which case the connection will begin when DTLS
>@@ -87,10 +85,10 @@ class SctpTransportInternal {
>   // used" part. See:
>   // https://bugs.chromium.org/p/chromium/issues/detail?id=619849
>   virtual bool OpenStream(int sid) = 0;
>-  // The inverse of OpenStream. When this method returns, the reset process may
>-  // have not finished but it will have begun.
>-  // TODO(deadbeef): We need a way to tell when it's done. See:
>-  // https://bugs.chromium.org/p/webrtc/issues/detail?id=4453
>+  // The inverse of OpenStream. Begins the closing procedure, which will
>+  // eventually result in SignalClosingProcedureComplete on the side that
>+  // initiates it, and both SignalClosingProcedureStartedRemotely and
>+  // SignalClosingProcedureComplete on the other side.
>   virtual bool ResetStream(int sid) = 0;
>   // Send data down this channel (will be wrapped as SCTP packets then given to
>   // usrsctp that will then post the network interface).
>@@ -113,8 +111,14 @@ class SctpTransportInternal {
>   // contains message payload.
>   sigslot::signal2<const ReceiveDataParams&, const rtc::CopyOnWriteBuffer&>
>       SignalDataReceived;
>-  // Parameter is SID of closed stream.
>-  sigslot::signal1<int> SignalStreamClosedRemotely;
>+  // Parameter is SID; fired when we receive an incoming stream reset on an
>+  // open stream, indicating that the other side started the closing procedure.
>+  // After resetting the outgoing stream, SignalClosingProcedureComplete will
>+  // fire too.
>+  sigslot::signal1<int> SignalClosingProcedureStartedRemotely;
>+  // Parameter is SID; fired when closing procedure is complete (both incoming
>+  // and outgoing streams reset).
>+  sigslot::signal1<int> SignalClosingProcedureComplete;
> 
>   // Helper for debugging.
>   virtual void set_debug_name_for_testing(const char* debug_name) = 0;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/BUILD.gn
>index 95bb5b7a874..e1c1f3a3e6e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/BUILD.gn
>@@ -17,7 +17,6 @@ group("modules") {
>     "audio_processing",
>     "bitrate_controller",
>     "congestion_controller",
>-    "media_file",
>     "pacing",
>     "remote_bitrate_estimator",
>     "rtp_rtcp",
>@@ -37,8 +36,7 @@ rtc_source_set("module_api_public") {
>   ]
>   deps = [
>     "..:webrtc_common",
>-    "../:typedefs",
>-    "../api:optional",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
> 
>@@ -50,15 +48,24 @@ rtc_source_set("module_api") {
>   ]
>   deps = [
>     ":module_api_public",
>+    ":module_fec_api",
>     "..:webrtc_common",
>-    "../:typedefs",
>     "../api:libjingle_peerconnection_api",
>-    "../api:optional",
>-    "../api:video_frame_api",
>-    "../api:video_frame_api_i420",
>+    "../api/transport:network_control",
>+    "../api/video:video_frame",
>+    "../api/video:video_frame_i420",
>+    "../modules/rtp_rtcp:rtp_video_header",
>     "../rtc_base:deprecation",
>     "../rtc_base:rtc_base_approved",
>     "video_coding:codec_globals_headers",
>+    "//third_party/abseil-cpp/absl/types:optional",
>+  ]
>+}
>+
>+rtc_source_set("module_fec_api") {
>+  visibility = [ "*" ]
>+  sources = [
>+    "include/module_fec_types.h",
>   ]
> }
> 
>@@ -67,7 +74,6 @@ if (rtc_include_tests) {
>     "../resources/audio_coding/testfile32kHz.pcm",
>     "../resources/audio_coding/teststereo32kHz.pcm",
>     "../resources/foreman_cif.yuv",
>-    "../resources/paris_qcif.yuv",
>   ]
> 
>   if (is_ios) {
>@@ -89,7 +95,6 @@ if (rtc_include_tests) {
>       "audio_coding:audio_coding_modules_tests",
>       "rtp_rtcp:rtp_rtcp_modules_tests",
>       "video_coding:video_coding_modules_tests",
>-      "//testing/gmock",
>       "//testing/gtest",
>     ]
> 
>@@ -113,19 +118,6 @@ if (rtc_include_tests) {
>     if (is_ios) {
>       deps += [ ":modules_tests_bundle_data" ]
>     }
>-
>-    if (!build_with_chromium && is_clang) {
>-      # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>-      suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-    }
>-    if (is_win) {
>-      cflags = [
>-        # TODO(phoglund): get rid of 4373 supression when
>-        # http://code.google.com/p/webrtc/issues/detail?id=261 is solved.
>-        # legacy warning for ignoring const / volatile in signatures.
>-        "/wd4373",
>-      ]
>-    }
>   }
> 
>   modules_unittests_resources = [
>@@ -246,11 +238,6 @@ if (rtc_include_tests) {
>       "module_common_types_unittest.cc",
>     ]
> 
>-    if (!build_with_chromium && is_clang) {
>-      # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>-      suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-    }
>-
>     deps = [
>       ":module_api",
>       "../test:test_main",
>@@ -258,9 +245,9 @@ if (rtc_include_tests) {
>       "audio_device:audio_device_unittests",
>       "audio_mixer:audio_mixer_unittests",
>       "audio_processing:audio_processing_unittests",
>+      "audio_processing/aec3:aec3_unittests",
>       "bitrate_controller:bitrate_controller_unittests",
>       "congestion_controller:congestion_controller_unittests",
>-      "media_file:media_file_unittests",
>       "pacing:pacing_unittests",
>       "remote_bitrate_estimator:remote_bitrate_estimator_unittests",
>       "rtp_rtcp:rtp_rtcp_unittests",
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/BUILD.gn
>index cc5036a3ca2..2829db44ae3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/BUILD.gn
>@@ -8,7 +8,6 @@
> 
> import("../../webrtc.gni")
> import("audio_coding.gni")
>-import("//build/config/arm.gni")
> if (!build_with_mozilla) {
>   import("//third_party/protobuf/proto_library.gni")
> }
>@@ -50,11 +49,11 @@ rtc_static_library("audio_format_conversion") {
>   deps = [
>     "../..:webrtc_common",
>     "../../api:array_view",
>-    "../../api:optional",
>     "../../api/audio_codecs:audio_codecs_api",
>     "../../rtc_base:checks",
>     "../../rtc_base:rtc_base_approved",
>     "../../rtc_base:sanitizer",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
> 
>@@ -67,10 +66,9 @@ rtc_static_library("rent_a_codec") {
>     "acm2/rent_a_codec.h",
>   ]
>   deps = [
>-           "../..:typedefs",
>            "../../rtc_base:checks",
>            "../../api:array_view",
>-           "../../api:optional",
>+           "//third_party/abseil-cpp/absl/types:optional",
>            "../../api/audio_codecs:audio_codecs_api",
>            "../..:webrtc_common",
>            "../../rtc_base:protobuf_utils",
>@@ -85,26 +83,18 @@ rtc_static_library("rent_a_codec") {
>   defines = audio_codec_defines
> }
> 
>-config("audio_coding_config") {
>-  include_dirs = [
>-    "include",
>-    "../include",
>-  ]
>-}
>-
> rtc_source_set("audio_coding_module_typedefs") {
>   sources = [
>     "include/audio_coding_module_typedefs.h",
>   ]
>   deps = [
>-    "..:module_api",
>-    "../..:typedefs",
>     "../..:webrtc_common",
>   ]
> }
> 
> rtc_static_library("audio_coding") {
>   visibility += [ "*" ]
>+  allow_poison = [ "audio_codecs" ]  # TODO(bugs.webrtc.org/8396): Remove.
>   sources = [
>     "acm2/acm_receiver.cc",
>     "acm2/acm_receiver.h",
>@@ -120,35 +110,27 @@ rtc_static_library("audio_coding") {
> 
>   defines = []
> 
>-  public_configs = [ ":audio_coding_config" ]
>-
>   if (rtc_include_opus) {
>     public_deps = [
>       ":webrtc_opus",
>     ]
>   }
> 
>-  if (is_win) {
>-    cflags = [
>-      # TODO(kjellander): Bug 261: fix this warning.
>-      "/wd4373",  # virtual function override.
>-    ]
>-  }
>-
>   deps = audio_coding_deps + [
>-           "../..:typedefs",
>+           "../../api/audio:audio_frame_api",
>+           "..:module_api",
>+           "../../common_audio:common_audio_c",
>            "../../rtc_base:deprecation",
>            "../../rtc_base:checks",
>            "../../system_wrappers:metrics_api",
>-           "..:module_api",
>            "../../api:array_view",
>            "../../api/audio_codecs:audio_codecs_api",
>-           "../../api/audio_codecs:builtin_audio_decoder_factory",
>            ":audio_coding_module_typedefs",
>            ":neteq",
>            ":rent_a_codec",
>+           "../../rtc_base:audio_format_to_string",
>            "../../rtc_base:rtc_base_approved",
>-           "../../api:optional",
>+           "//third_party/abseil-cpp/absl/types:optional",
>            "../../logging:rtc_event_log_api",
>          ]
>   defines = audio_coding_defines
>@@ -166,10 +148,6 @@ rtc_static_library("legacy_encoded_audio_frame") {
>   ]
> }
> 
>-config("cng_config") {
>-  include_dirs = [ "codecs/cng/include" ]
>-}
>-
> rtc_static_library("cng") {
>   visibility += [ "*" ]
>   sources = [
>@@ -179,22 +157,16 @@ rtc_static_library("cng") {
>     "codecs/cng/webrtc_cng.h",
>   ]
> 
>-  public_configs = [ ":cng_config" ]
>-
>   deps = [
>-    "../..:typedefs",
>     "../..:webrtc_common",
>     "../../api:array_view",
>     "../../api/audio_codecs:audio_codecs_api",
>     "../../common_audio",
>+    "../../common_audio:common_audio_c",
>     "../../rtc_base:rtc_base_approved",
>   ]
> }
> 
>-config("red_config") {
>-  include_dirs = [ "codecs/red" ]
>-}
>-
> rtc_static_library("red") {
>   visibility += [ "*" ]
>   sources = [
>@@ -202,8 +174,6 @@ rtc_static_library("red") {
>     "codecs/red/audio_encoder_copy_red.h",
>   ]
> 
>-  public_configs = [ ":red_config" ]
>-
>   deps = [
>     "../../api/audio_codecs:audio_codecs_api",
>     "../../common_audio",
>@@ -212,12 +182,9 @@ rtc_static_library("red") {
>   ]
> }
> 
>-config("g711_config") {
>-  include_dirs = [ "codecs/g711/include" ]
>-}
>-
> rtc_static_library("g711") {
>   visibility += [ "*" ]
>+  poisonous = [ "audio_codecs" ]
>   sources = [
>     "codecs/g711/audio_decoder_pcm.cc",
>     "codecs/g711/audio_decoder_pcm.h",
>@@ -225,8 +192,6 @@ rtc_static_library("g711") {
>     "codecs/g711/audio_encoder_pcm.h",
>   ]
> 
>-  public_configs = [ ":g711_config" ]
>-
>   deps = [
>     ":legacy_encoded_audio_frame",
>     "../..:webrtc_common",
>@@ -240,24 +205,20 @@ rtc_static_library("g711") {
> }
> 
> rtc_source_set("g711_c") {
>+  poisonous = [ "audio_codecs" ]
>   sources = [
>-    "codecs/g711/g711.c",
>-    "codecs/g711/g711.h",
>     "codecs/g711/g711_interface.c",
>     "codecs/g711/g711_interface.h",
>   ]
>   deps = [
>-    "../..:typedefs",
>     "../..:webrtc_common",
>+    "../third_party/g711:g711_3p",
>   ]
> }
> 
>-config("g722_config") {
>-  include_dirs = [ "codecs/g722/include" ]
>-}
>-
> rtc_static_library("g722") {
>   visibility += [ "*" ]
>+  poisonous = [ "audio_codecs" ]
>   sources = [
>     "codecs/g722/audio_decoder_g722.cc",
>     "codecs/g722/audio_decoder_g722.h",
>@@ -265,8 +226,6 @@ rtc_static_library("g722") {
>     "codecs/g722/audio_encoder_g722.h",
>   ]
> 
>-  public_configs = [ ":g722_config" ]
>-
>   deps = [
>     ":legacy_encoded_audio_frame",
>     "../..:webrtc_common",
>@@ -281,25 +240,20 @@ rtc_static_library("g722") {
> }
> 
> rtc_source_set("g722_c") {
>+  poisonous = [ "audio_codecs" ]
>   sources = [
>-    "codecs/g722/g722_decode.c",
>-    "codecs/g722/g722_enc_dec.h",
>-    "codecs/g722/g722_encode.c",
>     "codecs/g722/g722_interface.c",
>     "codecs/g722/g722_interface.h",
>   ]
>   deps = [
>-    "../..:typedefs",
>     "../..:webrtc_common",
>+    "../third_party/g722:g722_3p",
>   ]
> }
> 
>-config("ilbc_config") {
>-  include_dirs = [ "codecs/ilbc/include" ]
>-}
>-
> rtc_static_library("ilbc") {
>   visibility += webrtc_default_visibility
>+  poisonous = [ "audio_codecs" ]
>   sources = [
>     "codecs/ilbc/audio_decoder_ilbc.cc",
>     "codecs/ilbc/audio_decoder_ilbc.h",
>@@ -307,8 +261,6 @@ rtc_static_library("ilbc") {
>     "codecs/ilbc/audio_encoder_ilbc.h",
>   ]
> 
>-  public_configs = [ ":ilbc_config" ]
>-
>   deps = [
>     ":legacy_encoded_audio_frame",
>     "../..:webrtc_common",
>@@ -324,6 +276,7 @@ rtc_static_library("ilbc") {
> }
> 
> rtc_source_set("ilbc_c") {
>+  poisonous = [ "audio_codecs" ]
>   sources = [
>     "codecs/ilbc/abs_quant.c",
>     "codecs/ilbc/abs_quant.h",
>@@ -466,45 +419,42 @@ rtc_source_set("ilbc_c") {
>     "codecs/ilbc/xcorr_coef.h",
>   ]
> 
>-  public_configs = [ ":ilbc_config" ]
>-
>   deps = [
>-    "../..:typedefs",
>     "../..:webrtc_common",
>     "../../api/audio_codecs:audio_codecs_api",
>     "../../common_audio",
>+    "../../common_audio:common_audio_c",
>     "../../rtc_base:checks",
>     "../../rtc_base:rtc_base_approved",
>     "../../rtc_base:sanitizer",
>+    "../../rtc_base/system:arch",
>+    "../../rtc_base/system:unused",
>   ]
> }
> 
> rtc_static_library("isac_common") {
>+  poisonous = [ "audio_codecs" ]
>   sources = [
>     "codecs/isac/audio_decoder_isac_t.h",
>     "codecs/isac/audio_decoder_isac_t_impl.h",
>     "codecs/isac/audio_encoder_isac_t.h",
>     "codecs/isac/audio_encoder_isac_t_impl.h",
>-    "codecs/isac/bandwidth_info.h",
>     "codecs/isac/locked_bandwidth_info.cc",
>     "codecs/isac/locked_bandwidth_info.h",
>   ]
>   deps = [
>-    "../..:typedefs",
>+    ":isac_bwinfo",
>     "../..:webrtc_common",
>-    "../../api:optional",
>     "../../api/audio_codecs:audio_codecs_api",
>     "../../rtc_base:checks",
>     "../../rtc_base:rtc_base_approved",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
> 
>-config("isac_config") {
>-  include_dirs = [ "codecs/isac/main/include" ]
>-}
>-
> rtc_static_library("isac") {
>   visibility += [ "*" ]
>+  poisonous = [ "audio_codecs" ]
>   sources = [
>     "codecs/isac/main/include/audio_decoder_isac.h",
>     "codecs/isac/main/include/audio_encoder_isac.h",
>@@ -521,7 +471,39 @@ rtc_static_library("isac") {
>   ]
> }
> 
>+rtc_source_set("isac_bwinfo") {
>+  sources = [
>+    "codecs/isac/bandwidth_info.h",
>+  ]
>+  deps = []
>+}
>+
>+rtc_source_set("isac_vad") {
>+  visibility += webrtc_default_visibility
>+  sources = [
>+    "codecs/isac/main/source/filter_functions.c",
>+    "codecs/isac/main/source/filter_functions.h",
>+    "codecs/isac/main/source/isac_vad.c",
>+    "codecs/isac/main/source/isac_vad.h",
>+    "codecs/isac/main/source/os_specific_inline.h",
>+    "codecs/isac/main/source/pitch_estimator.c",
>+    "codecs/isac/main/source/pitch_estimator.h",
>+    "codecs/isac/main/source/pitch_filter.c",
>+    "codecs/isac/main/source/pitch_filter.h",
>+    "codecs/isac/main/source/settings.h",
>+    "codecs/isac/main/source/structs.h",
>+  ]
>+  deps = [
>+    ":isac_bwinfo",
>+    "../../rtc_base:compile_assert_c",
>+    "../../rtc_base/system:arch",
>+    "../../rtc_base/system:ignore_warnings",
>+    "../third_party/fft",
>+  ]
>+}
>+
> rtc_static_library("isac_c") {
>+  poisonous = [ "audio_codecs" ]
>   sources = [
>     "codecs/isac/main/include/isac.h",
>     "codecs/isac/main/source/arith_routines.c",
>@@ -540,11 +522,6 @@ rtc_static_library("isac_c") {
>     "codecs/isac/main/source/encode_lpc_swb.h",
>     "codecs/isac/main/source/entropy_coding.c",
>     "codecs/isac/main/source/entropy_coding.h",
>-    "codecs/isac/main/source/fft.c",
>-    "codecs/isac/main/source/fft.h",
>-    "codecs/isac/main/source/filter_functions.c",
>-    "codecs/isac/main/source/filterbank_tables.c",
>-    "codecs/isac/main/source/filterbank_tables.h",
>     "codecs/isac/main/source/filterbanks.c",
>     "codecs/isac/main/source/intialize.c",
>     "codecs/isac/main/source/isac.c",
>@@ -560,18 +537,12 @@ rtc_static_library("isac_c") {
>     "codecs/isac/main/source/lpc_shape_swb16_tables.h",
>     "codecs/isac/main/source/lpc_tables.c",
>     "codecs/isac/main/source/lpc_tables.h",
>-    "codecs/isac/main/source/os_specific_inline.h",
>-    "codecs/isac/main/source/pitch_estimator.c",
>-    "codecs/isac/main/source/pitch_estimator.h",
>-    "codecs/isac/main/source/pitch_filter.c",
>     "codecs/isac/main/source/pitch_gain_tables.c",
>     "codecs/isac/main/source/pitch_gain_tables.h",
>     "codecs/isac/main/source/pitch_lag_tables.c",
>     "codecs/isac/main/source/pitch_lag_tables.h",
>-    "codecs/isac/main/source/settings.h",
>     "codecs/isac/main/source/spectrum_ar_model_tables.c",
>     "codecs/isac/main/source/spectrum_ar_model_tables.h",
>-    "codecs/isac/main/source/structs.h",
>     "codecs/isac/main/source/transform.c",
>   ]
> 
>@@ -579,32 +550,28 @@ rtc_static_library("isac_c") {
>     libs = [ "m" ]
>   }
> 
>-  public_configs = [ ":isac_config" ]
>-
>   deps = [
>-    ":isac_common",
>-    "../..:typedefs",
>+    ":isac_bwinfo",
>+    ":isac_vad",
>     "../..:webrtc_common",
>     "../../common_audio",
>+    "../../common_audio:common_audio_c",
>     "../../rtc_base:checks",
>     "../../rtc_base:compile_assert_c",
>     "../../rtc_base:rtc_base_approved",
>+    "../../rtc_base/system:arch",
>+    "../third_party/fft",
>   ]
> }
> 
>-config("isac_fix_config") {
>-  include_dirs = [ "codecs/isac/fix/include" ]
>-}
>-
> rtc_static_library("isac_fix") {
>   visibility += [ "*" ]
>+  poisonous = [ "audio_codecs" ]
>   sources = [
>     "codecs/isac/fix/source/audio_decoder_isacfix.cc",
>     "codecs/isac/fix/source/audio_encoder_isacfix.cc",
>   ]
> 
>-  public_configs = [ ":isac_fix_config" ]
>-
>   deps = [
>     ":isac_common",
>     "../../api/audio_codecs:audio_codecs_api",
>@@ -621,30 +588,41 @@ rtc_static_library("isac_fix") {
> }
> 
> rtc_source_set("isac_fix_common") {
>+  poisonous = [ "audio_codecs" ]
>   sources = [
>     "codecs/isac/fix/source/codec.h",
>+    "codecs/isac/fix/source/entropy_coding.h",
>     "codecs/isac/fix/source/fft.c",
>     "codecs/isac/fix/source/fft.h",
>     "codecs/isac/fix/source/filterbank_internal.h",
>     "codecs/isac/fix/source/settings.h",
>+    "codecs/isac/fix/source/structs.h",
>+    "codecs/isac/fix/source/transform_tables.c",
>   ]
>-  public_configs = [ ":isac_fix_config" ]
>   deps = [
>-    "../..:typedefs",
>+    ":isac_bwinfo",
>+    "../../common_audio",
>+    "../../common_audio:common_audio_c",
>   ]
> }
> 
> rtc_source_set("isac_fix_c_arm_asm") {
>+  poisonous = [ "audio_codecs" ]
>   sources = []
>   if (current_cpu == "arm" && arm_version >= 7) {
>     sources += [
>       "codecs/isac/fix/source/lattice_armv7.S",
>       "codecs/isac/fix/source/pitch_filter_armv6.S",
>     ]
>+    deps = [
>+      ":isac_fix_common",
>+      "../../rtc_base/system:asm_defines",
>+    ]
>   }
> }
> 
> rtc_source_set("isac_fix_c") {
>+  poisonous = [ "audio_codecs" ]
>   sources = [
>     "codecs/isac/fix/include/audio_decoder_isacfix.h",
>     "codecs/isac/fix/include/audio_encoder_isacfix.h",
>@@ -660,7 +638,6 @@ rtc_source_set("isac_fix_c") {
>     "codecs/isac/fix/source/decode_plc.c",
>     "codecs/isac/fix/source/encode.c",
>     "codecs/isac/fix/source/entropy_coding.c",
>-    "codecs/isac/fix/source/entropy_coding.h",
>     "codecs/isac/fix/source/filterbank_tables.c",
>     "codecs/isac/fix/source/filterbank_tables.h",
>     "codecs/isac/fix/source/filterbanks.c",
>@@ -685,24 +662,22 @@ rtc_source_set("isac_fix_c") {
>     "codecs/isac/fix/source/pitch_lag_tables.h",
>     "codecs/isac/fix/source/spectrum_ar_model_tables.c",
>     "codecs/isac/fix/source/spectrum_ar_model_tables.h",
>-    "codecs/isac/fix/source/structs.h",
>     "codecs/isac/fix/source/transform.c",
>-    "codecs/isac/fix/source/transform_tables.c",
>   ]
> 
>-  public_configs = [ ":isac_fix_config" ]
>-
>   deps = [
>+    ":isac_bwinfo",
>     ":isac_common",
>-    "../..:typedefs",
>     "../..:webrtc_common",
>     "../../api/audio_codecs:audio_codecs_api",
>     "../../common_audio",
>+    "../../common_audio:common_audio_c",
>     "../../rtc_base:checks",
>     "../../rtc_base:compile_assert_c",
>     "../../rtc_base:rtc_base_approved",
>     "../../rtc_base:sanitizer",
>     "../../system_wrappers:cpu_features_api",
>+    "../third_party/fft",
>   ]
> 
>   public_deps = [
>@@ -711,6 +686,13 @@ rtc_source_set("isac_fix_c") {
> 
>   if (rtc_build_with_neon) {
>     deps += [ ":isac_neon" ]
>+
>+    # TODO(bugs.webrtc.org/9579): Consider moving the usage of NEON from
>+    # pitch_estimator_c.c into the "isac_neon" target and delete this flag:
>+    if (current_cpu != "arm64") {
>+      suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
>+      cflags = [ "-mfpu=neon" ]
>+    }
>   }
> 
>   if (current_cpu == "arm" && arm_version >= 7) {
>@@ -748,6 +730,7 @@ rtc_source_set("isac_fix_c") {
> 
> if (rtc_build_with_neon) {
>   rtc_static_library("isac_neon") {
>+    poisonous = [ "audio_codecs" ]
>     sources = [
>       "codecs/isac/fix/source/entropy_coding_neon.c",
>       "codecs/isac/fix/source/filterbanks_neon.c",
>@@ -757,9 +740,7 @@ if (rtc_build_with_neon) {
>     ]
> 
>     if (current_cpu != "arm64") {
>-      # Enable compilation for the NEON instruction set. This is needed
>-      # since //build/config/arm.gni only enables NEON for iOS, not Android.
>-      # This provides the same functionality as webrtc/build/arm_neon.gypi.
>+      # Enable compilation for the NEON instruction set.
>       suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
>       cflags = [ "-mfpu=neon" ]
>     }
>@@ -776,18 +757,16 @@ if (rtc_build_with_neon) {
>     deps = [
>       ":isac_fix_common",
>       "../../common_audio",
>+      "../../common_audio:common_audio_c",
>       "../../rtc_base:checks",
>       "../../rtc_base:rtc_base_approved",
>     ]
>   }
> }
> 
>-config("pcm16b_config") {
>-  include_dirs = [ "codecs/pcm16b/include" ]
>-}
>-
> rtc_static_library("pcm16b") {
>   visibility += [ "*" ]
>+  poisonous = [ "audio_codecs" ]
>   sources = [
>     "codecs/pcm16b/audio_decoder_pcm16b.cc",
>     "codecs/pcm16b/audio_decoder_pcm16b.h",
>@@ -808,24 +787,23 @@ rtc_static_library("pcm16b") {
>   public_deps = [
>     ":pcm16b_c",
>   ]
>-  public_configs = [ ":pcm16b_config" ]
> }
> 
> rtc_source_set("pcm16b_c") {
>+  poisonous = [ "audio_codecs" ]
>   sources = [
>     "codecs/pcm16b/pcm16b.c",
>     "codecs/pcm16b/pcm16b.h",
>   ]
> 
>-  public_configs = [ ":pcm16b_config" ]
>   deps = [
>-    "../..:typedefs",
>     "../..:webrtc_common",
>   ]
> }
> 
> rtc_static_library("webrtc_opus") {
>   visibility += webrtc_default_visibility
>+  poisonous = [ "audio_codecs" ]
>   sources = [
>     "codecs/opus/audio_decoder_opus.cc",
>     "codecs/opus/audio_decoder_opus.h",
>@@ -836,14 +814,16 @@ rtc_static_library("webrtc_opus") {
>   deps = [
>     ":audio_network_adaptor",
>     "../..:webrtc_common",
>-    "../../api:optional",
>     "../../api/audio_codecs:audio_codecs_api",
>     "../../api/audio_codecs/opus:audio_encoder_opus_config",
>     "../../common_audio",
>     "../../rtc_base:checks",
>     "../../rtc_base:rtc_base_approved",
>     "../../rtc_base:rtc_numerics",
>+    "../../rtc_base:safe_minmax",
>     "../../system_wrappers:field_trial_api",
>+    "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
>   public_deps = [
>     ":webrtc_opus_c",
>@@ -860,6 +840,7 @@ rtc_static_library("webrtc_opus") {
> }
> 
> rtc_source_set("webrtc_opus_c") {
>+  poisonous = [ "audio_codecs" ]
>   sources = [
>     "codecs/opus/opus_inst.h",
>     "codecs/opus/opus_interface.c",
>@@ -877,7 +858,6 @@ rtc_source_set("webrtc_opus_c") {
>   }
> 
>   deps = [
>-    "../..:typedefs",
>     "../..:webrtc_common",
>     "../../rtc_base:checks",
>     "../../rtc_base:rtc_base_approved",
>@@ -910,7 +890,7 @@ rtc_static_library("audio_network_adaptor_config") {
>     "audio_network_adaptor/include/audio_network_adaptor_config.h",
>   ]
>   deps = [
>-    "../../api:optional",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
> 
>@@ -949,15 +929,18 @@ rtc_static_library("audio_network_adaptor") {
> 
>   deps = [
>     "../..:webrtc_common",
>-    "../../api:optional",
>     "../../api/audio_codecs:audio_codecs_api",
>     "../../common_audio",
>+    "../../logging:rtc_event_audio",
>     "../../logging:rtc_event_log_api",
>     "../../rtc_base:checks",
>     "../../rtc_base:protobuf_utils",
>     "../../rtc_base:rtc_base_approved",
>+    "../../rtc_base/system:file_wrapper",
>     "../../system_wrappers",
>     "../../system_wrappers:field_trial_api",
>+    "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> 
>   if (rtc_enable_protobuf) {
>@@ -979,9 +962,9 @@ rtc_source_set("neteq_decoder_enum") {
>     "neteq/neteq_decoder_enum.h",
>   ]
>   deps = [
>-    "../../api:optional",
>     "../../api/audio_codecs:audio_codecs_api",
>     "../../rtc_base:rtc_base_approved",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
> 
>@@ -1004,10 +987,6 @@ rtc_static_library("neteq") {
>     "neteq/cross_correlation.h",
>     "neteq/decision_logic.cc",
>     "neteq/decision_logic.h",
>-    "neteq/decision_logic_fax.cc",
>-    "neteq/decision_logic_fax.h",
>-    "neteq/decision_logic_normal.cc",
>-    "neteq/decision_logic_normal.h",
>     "neteq/decoder_database.cc",
>     "neteq/decoder_database.h",
>     "neteq/defines.h",
>@@ -1023,6 +1002,8 @@ rtc_static_library("neteq") {
>     "neteq/dtmf_tone_generator.h",
>     "neteq/expand.cc",
>     "neteq/expand.h",
>+    "neteq/expand_uma_logger.cc",
>+    "neteq/expand_uma_logger.h",
>     "neteq/include/neteq.h",
>     "neteq/merge.cc",
>     "neteq/merge.h",
>@@ -1064,18 +1045,22 @@ rtc_static_library("neteq") {
>     ":cng",
>     ":neteq_decoder_enum",
>     "..:module_api",
>-    "../..:typedefs",
>     "../..:webrtc_common",
>     "../../api:libjingle_peerconnection_api",
>-    "../../api:optional",
>+    "../../api/audio:audio_frame_api",
>     "../../api/audio_codecs:audio_codecs_api",
>     "../../common_audio",
>+    "../../common_audio:common_audio_c",
>+    "../../rtc_base:audio_format_to_string",
>     "../../rtc_base:checks",
>     "../../rtc_base:gtest_prod",
>     "../../rtc_base:rtc_base_approved",
>+    "../../rtc_base:safe_minmax",
>     "../../rtc_base:sanitizer",
>+    "../../rtc_base/system:fallthrough",
>     "../../system_wrappers:field_trial_api",
>     "../../system_wrappers:metrics_api",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
> 
>@@ -1100,23 +1085,17 @@ rtc_source_set("neteq_tools_minimal") {
>     "neteq/tools/packet_source.h",
>   ]
> 
>-  if (!build_with_chromium && is_clang) {
>-    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>-    suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-  }
>-
>   deps = [
>     ":neteq",
>-    "..:module_api",
>-    "../..:typedefs",
>     "../..:webrtc_common",
>     "../../api:libjingle_peerconnection_api",
>-    "../../api:optional",
>+    "../../api/audio:audio_frame_api",
>     "../../api/audio_codecs:audio_codecs_api",
>     "../../api/audio_codecs:builtin_audio_decoder_factory",
>     "../../rtc_base:checks",
>     "../../rtc_base:rtc_base_approved",
>     "../rtp_rtcp",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
> 
>@@ -1129,6 +1108,8 @@ rtc_source_set("neteq_test_tools") {
>     "neteq/tools/audio_loop.h",
>     "neteq/tools/constant_pcm_packet_source.cc",
>     "neteq/tools/constant_pcm_packet_source.h",
>+    "neteq/tools/neteq_packet_source_input.cc",
>+    "neteq/tools/neteq_packet_source_input.h",
>     "neteq/tools/output_audio_file.h",
>     "neteq/tools/output_wav_file.h",
>     "neteq/tools/rtp_file_source.cc",
>@@ -1137,8 +1118,6 @@ rtc_source_set("neteq_test_tools") {
>     "neteq/tools/rtp_generator.h",
>   ]
> 
>-  public_configs = [ ":neteq_tools_config" ]
>-
>   if (!build_with_chromium && is_clang) {
>     # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>     suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>@@ -1146,8 +1125,6 @@ rtc_source_set("neteq_test_tools") {
> 
>   deps = [
>     ":pcm16b",
>-    "..:module_api",
>-    "../..:typedefs",
>     "../..:webrtc_common",
>     "../../api:array_view",
>     "../../api:libjingle_peerconnection_api",
>@@ -1156,6 +1133,7 @@ rtc_source_set("neteq_test_tools") {
>     "../../rtc_base:rtc_base",
>     "../../rtc_base:rtc_base_approved",
>     "../../rtc_base:rtc_base_tests_utils",
>+    "../../rtc_base/system:arch",
>     "../../test:rtp_test_utils",
>     "../rtp_rtcp",
>     "../rtp_rtcp:rtp_rtcp_format",
>@@ -1168,17 +1146,13 @@ rtc_source_set("neteq_test_tools") {
> 
>   if (rtc_enable_protobuf) {
>     sources += [
>-      "neteq/tools/neteq_packet_source_input.cc",
>-      "neteq/tools/neteq_packet_source_input.h",
>+      "neteq/tools/neteq_event_log_input.cc",
>+      "neteq/tools/neteq_event_log_input.h",
>     ]
>     deps += [ ":rtc_event_log_source" ]
>   }
> }
> 
>-config("neteq_tools_config") {
>-  include_dirs = [ "tools" ]
>-}
>-
> rtc_source_set("neteq_tools") {
>   visibility += webrtc_default_visibility
>   sources = [
>@@ -1188,25 +1162,25 @@ rtc_source_set("neteq_tools") {
>     "neteq/tools/neteq_delay_analyzer.h",
>     "neteq/tools/neteq_replacement_input.cc",
>     "neteq/tools/neteq_replacement_input.h",
>+    "neteq/tools/neteq_stats_getter.cc",
>+    "neteq/tools/neteq_stats_getter.h",
>   ]
> 
>-  public_configs = [ ":neteq_tools_config" ]
>-
>   if (!build_with_chromium && is_clang) {
>     # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>     suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>   }
> 
>   deps = [
>-    "../..:typedefs",
>+    "..:module_api",
>     "../..:webrtc_common",
>     "../../api:array_view",
>-    "../../api:optional",
>     "../../api/audio_codecs:audio_codecs_api",
>     "../../rtc_base:checks",
>     "../../rtc_base:rtc_base_approved",
>     "../rtp_rtcp",
>     "../rtp_rtcp:rtp_rtcp_format",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> 
>   public_deps = [
>@@ -1225,7 +1199,6 @@ rtc_source_set("neteq_input_audio_tools") {
>   ]
> 
>   deps = [
>-    "../..:typedefs",
>     "../..:webrtc_common",
>     "../../common_audio",
>     "../../rtc_base:checks",
>@@ -1289,7 +1262,6 @@ if (rtc_include_tests) {
>       ":g711_test",
>       ":g722_test",
>       ":ilbc_test",
>-      ":insert_packet_with_timing",
>       ":isac_api_test",
>       ":isac_fix_test",
>       ":isac_switch_samprate_test",
>@@ -1297,6 +1269,7 @@ if (rtc_include_tests) {
>       ":neteq_ilbc_quality_test",
>       ":neteq_isac_quality_test",
>       ":neteq_opus_quality_test",
>+      ":neteq_pcm16b_quality_test",
>       ":neteq_pcmu_quality_test",
>       ":neteq_speed_test",
>       ":rtp_analyze",
>@@ -1316,8 +1289,6 @@ if (rtc_include_tests) {
> 
>     sources = [
>       "test/ACMTest.h",
>-      "test/APITest.cc",
>-      "test/APITest.h",
>       "test/Channel.cc",
>       "test/Channel.h",
>       "test/EncodeDecodeTest.cc",
>@@ -1353,21 +1324,19 @@ if (rtc_include_tests) {
>       ":audio_format_conversion",
>       ":pcm16b_c",
>       "..:module_api",
>-      "../..:typedefs",
>       "../..:webrtc_common",
>-      "../../api:optional",
>+      "../../api/audio:audio_frame_api",
>       "../../api/audio_codecs:builtin_audio_decoder_factory",
>+      "../../api/audio_codecs:builtin_audio_encoder_factory",
>+      "../../rtc_base:checks",
>       "../../rtc_base:rtc_base_approved",
>+      "../../rtc_base/synchronization:rw_lock_wrapper",
>       "../../system_wrappers",
>+      "../../test:fileutils",
>       "../../test:test_support",
>+      "//third_party/abseil-cpp/absl/types:optional",
>     ]
>     defines = audio_coding_defines
>-    if (is_win) {
>-      cflags = [
>-        # TODO(kjellander): bugs.webrtc.org/261: Fix this warning.
>-        "/wd4373",  # virtual function override.
>-      ]
>-    }
>     if (!build_with_chromium && is_clang) {
>       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>@@ -1385,13 +1354,13 @@ if (rtc_include_tests) {
>     deps = [
>       ":neteq_test_support",
>       ":neteq_test_tools",
>-      "../..:typedefs",
>       "../..:webrtc_common",
>       "../../api/audio_codecs/opus:audio_encoder_opus",
>       "../../rtc_base:protobuf_utils",
>       "../../rtc_base:rtc_base_approved",
>       "../../system_wrappers",
>       "../../system_wrappers:field_trial_api",
>+      "../../test:fileutils",
>       "../../test:perf_test",
>       "../../test:test_support",
>     ]
>@@ -1412,6 +1381,7 @@ if (rtc_include_tests) {
>     defines = audio_coding_defines
> 
>     deps = audio_coding_deps + [
>+             "..:module_api",
>              ":audio_coding",
>              ":audio_format_conversion",
>              "../../api/audio_codecs:audio_codecs_api",
>@@ -1433,9 +1403,11 @@ if (rtc_include_tests) {
>     defines = audio_coding_defines
> 
>     deps = audio_coding_deps + [
>+             "../../api/audio:audio_frame_api",
>              "../../rtc_base:checks",
>              ":audio_coding",
>              ":neteq_tools",
>+             "../../api/audio_codecs:builtin_audio_decoder_factory",
>              "../../api/audio_codecs:audio_codecs_api",
>              "../../rtc_base:rtc_base_approved",
>              "../../test:test_support",
>@@ -1460,49 +1432,21 @@ if (rtc_include_tests) {
>       ":audio_coding_module_typedefs",
>       ":audio_format_conversion",
>       "..:module_api",
>-      "../..:typedefs",
>       "../../:webrtc_common",
>-      "../../api:optional",
>+      "../../api/audio:audio_frame_api",
>+      "../../api/audio_codecs:builtin_audio_decoder_factory",
>+      "../../rtc_base:checks",
>       "../../rtc_base:rtc_base_approved",
>       "../../system_wrappers",
>       "../../system_wrappers:system_wrappers_default",
>+      "../../test:fileutils",
>       "../../test:test_support",
>       "../rtp_rtcp",
>       "//testing/gtest",
>+      "//third_party/abseil-cpp/absl/types:optional",
>     ]
>   }  # delay_test
> 
>-  rtc_executable("insert_packet_with_timing") {
>-    testonly = true
>-    sources = [
>-      "test/Channel.cc",
>-      "test/Channel.h",
>-      "test/PCMFile.cc",
>-      "test/PCMFile.h",
>-      "test/insert_packet_with_timing.cc",
>-    ]
>-
>-    if (!build_with_chromium && is_clang) {
>-      # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>-      suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-    }
>-
>-    deps = [
>-      ":audio_coding",
>-      ":audio_format_conversion",
>-      "..:module_api",
>-      "../..:typedefs",
>-      "../../:webrtc_common",
>-      "../../api:optional",
>-      "../../rtc_base:rtc_base_approved",
>-      "../../system_wrappers",
>-      "../../system_wrappers:system_wrappers_default",
>-      "../../test:test_support",
>-      "../rtp_rtcp",
>-      "//testing/gtest",
>-    ]
>-  }  # insert_packet_with_timing
>-
>   audio_decoder_unittests_resources =
>       [ "../../resources/audio_coding/testfile32kHz.pcm" ]
> 
>@@ -1522,29 +1466,23 @@ if (rtc_include_tests) {
>       "neteq/audio_decoder_unittest.cc",
>     ]
> 
>-    if (!build_with_chromium && is_clang) {
>-      # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>-      suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-    }
>-
>-    deps = []
>-
>     defines = neteq_defines
> 
>-    deps += audio_coding_deps
>-    deps += [
>-      ":ilbc",
>-      ":isac",
>-      ":isac_fix",
>-      ":neteq",
>-      ":neteq_tools",
>-      "../../api/audio_codecs:audio_codecs_api",
>-      "../../api/audio_codecs/opus:audio_encoder_opus",
>-      "../../common_audio",
>-      "../../rtc_base:protobuf_utils",
>-      "../../test:test_main",
>-      "//testing/gtest",
>-    ]
>+    deps = [
>+             ":ilbc",
>+             ":isac",
>+             ":isac_fix",
>+             ":neteq",
>+             ":neteq_tools",
>+             "../../test:fileutils",
>+             "../../api/audio_codecs:audio_codecs_api",
>+             "../../api/audio_codecs/opus:audio_encoder_opus",
>+             "../../common_audio",
>+             "../../rtc_base:protobuf_utils",
>+             "../../rtc_base/system:arch",
>+             "../../test:test_main",
>+             "//testing/gtest",
>+           ] + audio_coding_deps
> 
>     data = audio_decoder_unittests_resources
> 
>@@ -1555,7 +1493,7 @@ if (rtc_include_tests) {
>     if (is_ios) {
>       deps += [ ":audio_decoder_unittests_bundle_data" ]
>     }
>-  }  # audio_decoder_unittests
>+  }
> 
>   if (rtc_enable_protobuf) {
>     proto_library("neteq_unittest_proto") {
>@@ -1569,9 +1507,9 @@ if (rtc_include_tests) {
>       testonly = true
>       defines = []
>       deps = [
>-        "..:module_api",
>-        "../..:typedefs",
>         "../../rtc_base:checks",
>+        "../../test:field_trial",
>+        "../../test:fileutils",
>       ]
>       sources = [
>         "neteq/tools/neteq_rtpplay.cc",
>@@ -1582,13 +1520,6 @@ if (rtc_include_tests) {
>         suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>       }
> 
>-      if (is_win) {
>-        cflags = [
>-          # TODO(kjellander): bugs.webrtc.org/261: Fix this warning.
>-          "/wd4373",  # virtual function override.
>-        ]
>-      }
>-
>       deps += [
>         ":neteq",
>         ":neteq_test_tools",
>@@ -1620,7 +1551,7 @@ if (rtc_include_tests) {
>     testonly = true
>     defines = []
>     deps = [
>-      "../..:typedefs",
>+      "../../test:fileutils",
>     ]
>     sources = [
>       "codecs/isac/fix/test/isac_speed_test.cc",
>@@ -1678,14 +1609,14 @@ if (rtc_include_tests) {
>       ":neteq",
>       ":neteq_test_tools",
>       ":pcm16b",
>-      "..:module_api",
>-      "../..:typedefs",
>       "../..:webrtc_common",
>+      "../../api/audio:audio_frame_api",
>       "../../api/audio_codecs:audio_codecs_api",
>       "../../api/audio_codecs:builtin_audio_decoder_factory",
>       "../../rtc_base:checks",
>       "../../rtc_base:rtc_base_approved",
>       "../../system_wrappers",
>+      "../../test:fileutils",
>       "../../test:test_support",
>       "//testing/gtest",
>     ]
>@@ -1706,12 +1637,11 @@ if (rtc_include_tests) {
>     deps = [
>       ":neteq",
>       ":neteq_test_tools",
>-      "..:module_api",
>-      "../..:typedefs",
>       "../..:webrtc_common",
>       "../../api/audio_codecs:builtin_audio_decoder_factory",
>       "../../rtc_base:checks",
>       "../../rtc_base:rtc_base_approved",
>+      "../../test:fileutils",
>       "../../test:test_support",
>       "//testing/gtest",
>     ]
>@@ -1721,9 +1651,10 @@ if (rtc_include_tests) {
>     testonly = true
> 
>     deps = audio_coding_deps + [
>-             "../..:typedefs",
>+             "//third_party/abseil-cpp/absl/memory",
>              ":audio_coding",
>              ":neteq_input_audio_tools",
>+             "../../api/audio:audio_frame_api",
>              "../../api/audio_codecs/g711:audio_encoder_g711",
>              "../../api/audio_codecs/L16:audio_encoder_L16",
>              "../../api/audio_codecs/g722:audio_encoder_g722",
>@@ -1745,7 +1676,6 @@ if (rtc_include_tests) {
>     testonly = true
> 
>     deps = audio_coding_deps + [
>-             "../..:typedefs",
>              "../../system_wrappers:system_wrappers_default",
>              "../rtp_rtcp:rtp_rtcp_format",
>              "../../api:array_view",
>@@ -1790,11 +1720,6 @@ if (rtc_include_tests) {
>       "../../system_wrappers:system_wrappers_default",
>       "//testing/gtest",
>     ]
>-
>-    if (!build_with_chromium && is_clang) {
>-      # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>-      suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-    }
>   }
> 
>   rtc_executable("neteq_opus_quality_test") {
>@@ -1825,10 +1750,10 @@ if (rtc_include_tests) {
>     deps = [
>       ":neteq",
>       ":neteq_test_support",
>-      "../..:typedefs",
>       "../..:webrtc_common",
>       "../../rtc_base:rtc_base_approved",
>       "../../system_wrappers:system_wrappers_default",
>+      "../../test:fileutils",
>       "../../test:test_support",
>     ]
>   }
>@@ -1849,6 +1774,7 @@ if (rtc_include_tests) {
>       "../../rtc_base:checks",
>       "../../rtc_base:rtc_base_approved",
>       "../../system_wrappers:system_wrappers_default",
>+      "../../test:fileutils",
>       "../../test:test_main",
>       "//testing/gtest",
>     ]
>@@ -1884,6 +1810,26 @@ if (rtc_include_tests) {
>       ":neteq_quality_test_support",
>       "../../rtc_base:checks",
>       "../../rtc_base:rtc_base_approved",
>+      "../../test:fileutils",
>+      "../../test:test_main",
>+      "//testing/gtest",
>+    ]
>+  }
>+
>+  rtc_executable("neteq_pcm16b_quality_test") {
>+    testonly = true
>+
>+    sources = [
>+      "neteq/test/neteq_pcm16b_quality_test.cc",
>+    ]
>+
>+    deps = [
>+      ":neteq",
>+      ":neteq_quality_test_support",
>+      ":pcm16b",
>+      "../../rtc_base:checks",
>+      "../../rtc_base:rtc_base_approved",
>+      "../../test:fileutils",
>       "../../test:test_main",
>       "//testing/gtest",
>     ]
>@@ -1905,23 +1851,6 @@ if (rtc_include_tests) {
>     data = [
>       "../../resources/speech_and_misc_wb.pcm",
>     ]
>-
>-    if (is_win) {
>-      cflags = [
>-        # Disable warnings to enable Win64 build, issue 1323.
>-        "/wd4267",  # size_t to int truncation
>-      ]
>-    }
>-  }
>-
>-  config("isac_test_warnings_config") {
>-    if (is_win && is_clang) {
>-      cflags = [
>-        # Disable warnings failing when compiling with Clang on Windows.
>-        # https://bugs.chromium.org/p/webrtc/issues/detail?id=5366
>-        "-Wno-format",
>-      ]
>-    }
>   }
> 
>   rtc_source_set("isac_test_util") {
>@@ -1939,19 +1868,11 @@ if (rtc_include_tests) {
>       "codecs/isac/main/test/simpleKenny.c",
>     ]
> 
>-    include_dirs = [
>-      "codecs/isac/main/include",
>-      "codecs/isac/main/test",
>-      "codecs/isac/main/util",
>-    ]
>-
>     deps = [
>       ":isac",
>       ":isac_test_util",
>       "../../rtc_base:rtc_base_approved",
>     ]
>-
>-    configs += [ ":isac_test_warnings_config" ]
>   }
> 
>   rtc_executable("g711_test") {
>@@ -1975,7 +1896,6 @@ if (rtc_include_tests) {
> 
>     deps = [
>       ":g722",
>-      "../..:typedefs",
>       "../..:webrtc_common",
>     ]
>   }
>@@ -1992,12 +1912,6 @@ if (rtc_include_tests) {
>       ":isac_test_util",
>       "../../rtc_base:rtc_base_approved",
>     ]
>-
>-    include_dirs = [
>-      "codecs/isac/main/include",
>-      "codecs/isac/main/test",
>-      "codecs/isac/main/util",
>-    ]
>   }
> 
>   rtc_executable("isac_switch_samprate_test") {
>@@ -2010,13 +1924,8 @@ if (rtc_include_tests) {
>     deps = [
>       ":isac",
>       ":isac_test_util",
>-    ]
>-
>-    include_dirs = [
>-      "codecs/isac/main/include",
>-      "codecs/isac/main/test",
>-      "codecs/isac/main/util",
>-      "../../common_audio/signal_processing/include",
>+      "../../common_audio",
>+      "../../common_audio:common_audio_c",
>     ]
>   }
> 
>@@ -2043,14 +1952,10 @@ if (rtc_include_tests) {
>       ":webrtc_opus",
>       "../../common_audio",
>       "../../rtc_base:rtc_base_approved",
>+      "../../test:fileutils",
>       "../../test:test_main",
>       "//testing/gtest",
>     ]
>-
>-    if (!build_with_chromium && is_clang) {
>-      # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>-      suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-    }
>   }
> 
>   rtc_source_set("audio_coding_unittests") {
>@@ -2158,16 +2063,18 @@ if (rtc_include_tests) {
>       ":rent_a_codec",
>       ":webrtc_opus",
>       "..:module_api",
>-      "../..:typedefs",
>       "../..:webrtc_common",
>+      "../../api/audio:audio_frame_api",
>       "../../api/audio_codecs:audio_codecs_api",
>       "../../api/audio_codecs:builtin_audio_decoder_factory",
>       "../../api/audio_codecs:builtin_audio_encoder_factory",
>       "../../api/audio_codecs/opus:audio_decoder_opus",
>       "../../api/audio_codecs/opus:audio_encoder_opus",
>       "../../common_audio",
>+      "../../common_audio:common_audio_c",
>       "../../common_audio:mock_common_audio",
>       "../../logging:mocks",
>+      "../../logging:rtc_event_audio",
>       "../../logging:rtc_event_log_api",
>       "../../rtc_base:checks",
>       "../../rtc_base:protobuf_utils",
>@@ -2175,15 +2082,17 @@ if (rtc_include_tests) {
>       "../../rtc_base:rtc_base_approved",
>       "../../rtc_base:rtc_base_tests_utils",
>       "../../rtc_base:sanitizer",
>+      "../../rtc_base/system:arch",
>       "../../system_wrappers",
>       "../../system_wrappers:cpu_features_api",
>       "../../test:audio_codec_mocks",
>       "../../test:field_trial",
>+      "../../test:fileutils",
>       "../../test:rtp_test_utils",
>       "../../test:test_common",
>       "../../test:test_support",
>-      "//testing/gmock",
>       "//testing/gtest",
>+      "//third_party/abseil-cpp/absl/memory",
>     ]
> 
>     defines = audio_coding_defines
>@@ -2228,27 +2137,3 @@ rtc_source_set("audio_encoder_interface") {
>     "../../api/audio_codecs:audio_codecs_api",
>   ]
> }
>-
>-# For backwards compatibility only! Use
>-# webrtc/api/audio_codecs:builtin_audio_decoder_factory instead.
>-# TODO(kwiberg): Remove this.
>-rtc_source_set("builtin_audio_decoder_factory") {
>-  sources = [
>-    "codecs/builtin_audio_decoder_factory.h",
>-  ]
>-  deps = [
>-    "../../api/audio_codecs:builtin_audio_decoder_factory",
>-  ]
>-}
>-
>-# For backwards compatibility only! Use
>-# webrtc/api/audio_codecs:builtin_audio_decoder_factory instead.
>-# TODO(ossu): Remove this.
>-rtc_source_set("builtin_audio_encoder_factory") {
>-  sources = [
>-    "codecs/builtin_audio_encoder_factory.h",
>-  ]
>-  deps = [
>-    "../../api/audio_codecs:builtin_audio_encoder_factory",
>-  ]
>-}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_codec_database.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_codec_database.cc
>index 4553b52c62d..a322c952101 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_codec_database.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_codec_database.cc
>@@ -42,7 +42,7 @@ bool IsILBCRateValid(int rate, int frame_size_samples) {
>       (rate == 13300)) {
>     return true;
>   } else if (((frame_size_samples == 160) || (frame_size_samples == 320)) &&
>-      (rate == 15200)) {
>+             (rate == 15200)) {
>     return true;
>   } else {
>     return false;
>@@ -62,55 +62,54 @@ bool IsOpusRateValid(int rate) {
> 
> const CodecInst ACMCodecDB::database_[] = {
> #if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
>-  {103, "ISAC", 16000, 480, 1, 32000},
>-# if (defined(WEBRTC_CODEC_ISAC))
>-  {104, "ISAC", 32000, 960, 1, 56000},
>-# endif
>+    {103, "ISAC", 16000, 480, 1, 32000},
>+#if (defined(WEBRTC_CODEC_ISAC))
>+    {104, "ISAC", 32000, 960, 1, 56000},
> #endif
>-  // Mono
>-  {107, "L16", 8000, 80, 1, 128000},
>-  {108, "L16", 16000, 160, 1, 256000},
>-  {109, "L16", 32000, 320, 1, 512000},
>-  // Stereo
>-  {111, "L16", 8000, 80, 2, 128000},
>-  {112, "L16", 16000, 160, 2, 256000},
>-  {113, "L16", 32000, 320, 2, 512000},
>-  // G.711, PCM mu-law and A-law.
>-  // Mono
>-  {0, "PCMU", 8000, 160, 1, 64000},
>-  {8, "PCMA", 8000, 160, 1, 64000},
>-  // Stereo
>-  {110, "PCMU", 8000, 160, 2, 64000},
>-  {118, "PCMA", 8000, 160, 2, 64000},
>+#endif
>+    // Mono
>+    {107, "L16", 8000, 80, 1, 128000},
>+    {108, "L16", 16000, 160, 1, 256000},
>+    {109, "L16", 32000, 320, 1, 512000},
>+    // Stereo
>+    {111, "L16", 8000, 80, 2, 128000},
>+    {112, "L16", 16000, 160, 2, 256000},
>+    {113, "L16", 32000, 320, 2, 512000},
>+    // G.711, PCM mu-law and A-law.
>+    // Mono
>+    {0, "PCMU", 8000, 160, 1, 64000},
>+    {8, "PCMA", 8000, 160, 1, 64000},
>+    // Stereo
>+    {110, "PCMU", 8000, 160, 2, 64000},
>+    {118, "PCMA", 8000, 160, 2, 64000},
> #ifdef WEBRTC_CODEC_ILBC
>-  {102, "ILBC", 8000, 240, 1, 13300},
>+    {102, "ILBC", 8000, 240, 1, 13300},
> #endif
>-  // Mono
>-  {9, "G722", 16000, 320, 1, 64000},
>-  // Stereo
>-  {119, "G722", 16000, 320, 2, 64000},
>+    // Mono
>+    {9, "G722", 16000, 320, 1, 64000},
>+    // Stereo
>+    {119, "G722", 16000, 320, 2, 64000},
> #ifdef WEBRTC_CODEC_OPUS
>-  // Opus internally supports 48, 24, 16, 12, 8 kHz.
>-  // Mono and stereo.
>-  {120, "opus", 48000, 960, 2, 64000},
>+    // Opus internally supports 48, 24, 16, 12, 8 kHz.
>+    // Mono and stereo.
>+    {120, "opus", 48000, 960, 2, 64000},
> #endif
>-  // Comfort noise for four different sampling frequencies.
>-  {13, "CN", 8000, 240, 1, 0},
>-  {98, "CN", 16000, 480, 1, 0},
>-  {99, "CN", 32000, 960, 1, 0},
>+    // Comfort noise for four different sampling frequencies.
>+    {13, "CN", 8000, 240, 1, 0},
>+    {98, "CN", 16000, 480, 1, 0},
>+    {99, "CN", 32000, 960, 1, 0},
> #ifdef ENABLE_48000_HZ
>-  {100, "CN", 48000, 1440, 1, 0},
>+    {100, "CN", 48000, 1440, 1, 0},
> #endif
>-  {106, "telephone-event", 8000, 240, 1, 0},
>-  {114, "telephone-event", 16000, 240, 1, 0},
>-  {115, "telephone-event", 32000, 240, 1, 0},
>-  {116, "telephone-event", 48000, 240, 1, 0},
>+    {106, "telephone-event", 8000, 240, 1, 0},
>+    {114, "telephone-event", 16000, 240, 1, 0},
>+    {115, "telephone-event", 32000, 240, 1, 0},
>+    {116, "telephone-event", 48000, 240, 1, 0},
> #ifdef WEBRTC_CODEC_RED
>-  {127, "red", 8000, 0, 1, 0},
>+    {127, "red", 8000, 0, 1, 0},
> #endif
>-  // To prevent compile errors due to trailing commas.
>-  {-1, "Null", -1, -1, 0, -1}
>-};
>+    // To prevent compile errors due to trailing commas.
>+    {-1, "Null", -1, -1, 0, -1}};
> 
> // Create database with all codec settings at compile time.
> // Each entry needs the following parameters in the given order:
>@@ -119,9 +118,9 @@ const CodecInst ACMCodecDB::database_[] = {
> const ACMCodecDB::CodecSettings ACMCodecDB::codec_settings_[] = {
> #if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
>     {2, {480, 960}, 0, 1},
>-# if (defined(WEBRTC_CODEC_ISAC))
>+#if (defined(WEBRTC_CODEC_ISAC))
>     {1, {960}, 0, 1},
>-# endif
>+#endif
> #endif
>     // Mono
>     {4, {80, 160, 240, 320}, 0, 2},
>@@ -146,9 +145,9 @@ const ACMCodecDB::CodecSettings ACMCodecDB::codec_settings_[] = {
>     // Stereo
>     {6, {160, 320, 480, 640, 800, 960}, 0, 2},
> #ifdef WEBRTC_CODEC_OPUS
>-    // Opus supports frames shorter than 10ms,
>-    // but it doesn't help us to use them.
>-    // Mono and stereo.
>+// Opus supports frames shorter than 10ms,
>+// but it doesn't help us to use them.
>+// Mono and stereo.
> #if WEBRTC_OPUS_SUPPORT_120MS_PTIME
>     {5, {480, 960, 1920, 2880, 5760}, 0, 2},
> #else
>@@ -171,16 +170,15 @@ const ACMCodecDB::CodecSettings ACMCodecDB::codec_settings_[] = {
>     {1, {0}, 0, 1},
> #endif
>     // To prevent compile errors due to trailing commas.
>-    {-1, {-1}, -1, 0}
>-};
>+    {-1, {-1}, -1, 0}};
> 
> // Create a database of all NetEQ decoders at compile time.
> const NetEqDecoder ACMCodecDB::neteq_decoders_[] = {
> #if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
>     NetEqDecoder::kDecoderISAC,
>-# if (defined(WEBRTC_CODEC_ISAC))
>+#if (defined(WEBRTC_CODEC_ISAC))
>     NetEqDecoder::kDecoderISACswb,
>-# endif
>+#endif
> #endif
>     // Mono
>     NetEqDecoder::kDecoderPCM16B, NetEqDecoder::kDecoderPCM16Bwb,
>@@ -210,10 +208,8 @@ const NetEqDecoder ACMCodecDB::neteq_decoders_[] = {
> #ifdef ENABLE_48000_HZ
>     NetEqDecoder::kDecoderCNGswb48kHz,
> #endif
>-    NetEqDecoder::kDecoderAVT,
>-    NetEqDecoder::kDecoderAVT16kHz,
>-    NetEqDecoder::kDecoderAVT32kHz,
>-    NetEqDecoder::kDecoderAVT48kHz,
>+    NetEqDecoder::kDecoderAVT, NetEqDecoder::kDecoderAVT16kHz,
>+    NetEqDecoder::kDecoderAVT32kHz, NetEqDecoder::kDecoderAVT48kHz,
> #ifdef WEBRTC_CODEC_RED
>     NetEqDecoder::kDecoderRED,
> #endif
>@@ -260,8 +256,7 @@ int ACMCodecDB::CodecNumber(const CodecInst& codec_inst) {
>     int i;
>     int packet_size_samples;
>     for (i = 0; i < codec_settings_[codec_id].num_packet_sizes; i++) {
>-      packet_size_samples =
>-          codec_settings_[codec_id].packet_sizes_samples[i];
>+      packet_size_samples = codec_settings_[codec_id].packet_sizes_samples[i];
>       if (codec_inst.pacsize == packet_size_samples) {
>         packet_size_ok = true;
>         break;
>@@ -282,11 +277,10 @@ int ACMCodecDB::CodecNumber(const CodecInst& codec_inst) {
>   if (STR_CASE_CMP("isac", codec_inst.plname) == 0) {
>     return IsISACRateValid(codec_inst.rate) ? codec_id : kInvalidRate;
>   } else if (STR_CASE_CMP("ilbc", codec_inst.plname) == 0) {
>-    return IsILBCRateValid(codec_inst.rate, codec_inst.pacsize)
>-        ? codec_id : kInvalidRate;
>+    return IsILBCRateValid(codec_inst.rate, codec_inst.pacsize) ? codec_id
>+                                                                : kInvalidRate;
>   } else if (STR_CASE_CMP("opus", codec_inst.plname) == 0) {
>-    return IsOpusRateValid(codec_inst.rate)
>-        ? codec_id : kInvalidRate;
>+    return IsOpusRateValid(codec_inst.rate) ? codec_id : kInvalidRate;
>   }
> 
>   return database_[codec_id].rate == codec_inst.rate ? codec_id : kInvalidRate;
>@@ -298,8 +292,7 @@ int ACMCodecDB::CodecNumber(const CodecInst& codec_inst) {
> // Does not check other codec settings, such as payload type and packet size.
> // Returns the id of the codec, or -1 if no match is found.
> int ACMCodecDB::CodecId(const CodecInst& codec_inst) {
>-  return (CodecId(codec_inst.plname, codec_inst.plfreq,
>-                  codec_inst.channels));
>+  return (CodecId(codec_inst.plname, codec_inst.plfreq, codec_inst.channels));
> }
> 
> int ACMCodecDB::CodecId(const char* payload_name,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_codec_database.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_codec_database.h
>index 81cd4be05ae..ee6bb46faa3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_codec_database.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_codec_database.h
>@@ -18,7 +18,6 @@
> 
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/audio_coding/acm2/rent_a_codec.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -31,7 +30,7 @@ class ACMCodecDB {
>   //                 build.
>   // kMaxNumPacketSize - Maximum number of allowed packet sizes for one codec.
>   // These might need to be increased if adding a new codec to the database
>-  static const int kMaxNumCodecs =  50;
>+  static const int kMaxNumCodecs = 50;
>   static const int kMaxNumPacketSize = 6;
> 
>   // Codec specific settings
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_receive_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_receive_test.cc
>index 082506ab56b..ba8937e6eab 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_receive_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_receive_test.cc
>@@ -21,6 +21,7 @@
> #include "modules/audio_coding/neteq/tools/audio_sink.h"
> #include "modules/audio_coding/neteq/tools/packet.h"
> #include "modules/audio_coding/neteq/tools/packet_source.h"
>+#include "modules/include/module_common_types.h"
> #include "test/gtest.h"
> 
> namespace webrtc {
>@@ -155,8 +156,7 @@ void AcmReceiveTestOldApi::RegisterNetEqTestCodecs() {
>       continue;
>     }
> 
>-    if (RemapPltypeAndUseThisCodec(my_codec_param.plname,
>-                                   my_codec_param.plfreq,
>+    if (RemapPltypeAndUseThisCodec(my_codec_param.plname, my_codec_param.plfreq,
>                                    my_codec_param.channels,
>                                    &my_codec_param.pltype)) {
>       ASSERT_EQ(true,
>@@ -199,12 +199,10 @@ void AcmReceiveTestOldApi::Run() {
>     WebRtcRTPHeader header;
>     header.header = packet->header();
>     header.frameType = kAudioFrameSpeech;
>-    memset(&header.type.Audio, 0, sizeof(RTPAudioHeader));
>     EXPECT_EQ(0,
>               acm_->IncomingPacket(
>                   packet->payload(),
>-                  static_cast<int32_t>(packet->payload_length_bytes()),
>-                  header))
>+                  static_cast<int32_t>(packet->payload_length_bytes()), header))
>         << "Failure when inserting packet:" << std::endl
>         << "  PT = " << static_cast<int>(header.header.payloadType) << std::endl
>         << "  TS = " << header.header.timestamp << std::endl
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_receive_test.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_receive_test.h
>index c7e7da6e36b..83ffcb3a905 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_receive_test.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_receive_test.h
>@@ -11,7 +11,7 @@
> #ifndef MODULES_AUDIO_CODING_ACM2_ACM_RECEIVE_TEST_H_
> #define MODULES_AUDIO_CODING_ACM2_ACM_RECEIVE_TEST_H_
> 
>-#include <stddef.h> // for size_t
>+#include <stddef.h>  // for size_t
> #include <memory>
> #include <string>
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_receiver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_receiver.cc
>index 0d5dcaea096..0a88e70ec93 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_receiver.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_receiver.cc
>@@ -22,10 +22,12 @@
> #include "modules/audio_coding/acm2/call_statistics.h"
> #include "modules/audio_coding/acm2/rent_a_codec.h"
> #include "modules/audio_coding/neteq/include/neteq.h"
>+#include "modules/include/module_common_types.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/format_macros.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/numerics/safe_conversions.h"
>+#include "rtc_base/strings/audio_format_to_string.h"
> #include "system_wrappers/include/clock.h"
> 
> namespace webrtc {
>@@ -38,7 +40,8 @@ AcmReceiver::AcmReceiver(const AudioCodingModule::Config& config)
>       clock_(config.clock),
>       resampled_last_output_frame_(true) {
>   RTC_DCHECK(clock_);
>-  memset(last_audio_buffer_.get(), 0, AudioFrame::kMaxDataSizeSamples);
>+  memset(last_audio_buffer_.get(), 0,
>+         sizeof(int16_t) * AudioFrame::kMaxDataSizeSamples);
> }
> 
> AcmReceiver::~AcmReceiver() = default;
>@@ -61,7 +64,7 @@ int AcmReceiver::LeastRequiredDelayMs() const {
>   return neteq_->LeastRequiredDelayMs();
> }
> 
>-rtc::Optional<int> AcmReceiver::last_packet_sample_rate_hz() const {
>+absl::optional<int> AcmReceiver::last_packet_sample_rate_hz() const {
>   rtc::CritScope lock(&crit_sect_);
>   return last_packet_sample_rate_hz_;
> }
>@@ -83,7 +86,7 @@ int AcmReceiver::InsertPacket(const WebRtcRTPHeader& rtp_header,
>   {
>     rtc::CritScope lock(&crit_sect_);
> 
>-    const rtc::Optional<CodecInst> ci =
>+    const absl::optional<CodecInst> ci =
>         RtpHeaderToDecoder(*header, incoming_payload[0]);
>     if (!ci) {
>       RTC_LOG_F(LS_ERROR) << "Payload-type "
>@@ -199,15 +202,15 @@ int32_t AcmReceiver::AddCodec(int acm_codec_id,
>   const auto neteq_decoder = [acm_codec_id, channels]() -> NetEqDecoder {
>     if (acm_codec_id == -1)
>       return NetEqDecoder::kDecoderArbitrary;  // External decoder.
>-    const rtc::Optional<RentACodec::CodecId> cid =
>+    const absl::optional<RentACodec::CodecId> cid =
>         RentACodec::CodecIdFromIndex(acm_codec_id);
>     RTC_DCHECK(cid) << "Invalid codec index: " << acm_codec_id;
>-    const rtc::Optional<NetEqDecoder> ned =
>+    const absl::optional<NetEqDecoder> ned =
>         RentACodec::NetEqDecoderFromCodecId(*cid, channels);
>     RTC_DCHECK(ned) << "Invalid codec ID: " << static_cast<int>(*cid);
>     return *ned;
>   }();
>-  const rtc::Optional<SdpAudioFormat> new_format =
>+  const absl::optional<SdpAudioFormat> new_format =
>       NetEqDecoderToSdpAudioFormat(neteq_decoder);
> 
>   rtc::CritScope lock(&crit_sect_);
>@@ -228,8 +231,8 @@ int32_t AcmReceiver::AddCodec(int acm_codec_id,
>   if (!audio_decoder) {
>     ret_val = neteq_->RegisterPayloadType(neteq_decoder, name, payload_type);
>   } else {
>-    ret_val = neteq_->RegisterExternalDecoder(
>-        audio_decoder, neteq_decoder, name, payload_type);
>+    ret_val = neteq_->RegisterExternalDecoder(audio_decoder, neteq_decoder,
>+                                              name, payload_type);
>   }
>   if (ret_val != NetEq::kOK) {
>     RTC_LOG(LERROR) << "AcmReceiver::AddCodec " << acm_codec_id
>@@ -260,7 +263,8 @@ bool AcmReceiver::AddCodec(int rtp_payload_type,
>       neteq_->RegisterPayloadType(rtp_payload_type, audio_format);
>   if (!success) {
>     RTC_LOG(LERROR) << "AcmReceiver::AddCodec failed for payload type "
>-                    << rtp_payload_type << ", decoder format " << audio_format;
>+                    << rtp_payload_type << ", decoder format "
>+                    << rtc::ToString(audio_format);
>   }
>   return success;
> }
>@@ -272,9 +276,9 @@ void AcmReceiver::FlushBuffers() {
> void AcmReceiver::RemoveAllCodecs() {
>   rtc::CritScope lock(&crit_sect_);
>   neteq_->RemoveAllPayloadTypes();
>-  last_audio_decoder_ = rtc::nullopt;
>-  last_audio_format_ = rtc::nullopt;
>-  last_packet_sample_rate_hz_ = rtc::nullopt;
>+  last_audio_decoder_ = absl::nullopt;
>+  last_audio_format_ = absl::nullopt;
>+  last_packet_sample_rate_hz_ = absl::nullopt;
> }
> 
> int AcmReceiver::RemoveCodec(uint8_t payload_type) {
>@@ -285,14 +289,14 @@ int AcmReceiver::RemoveCodec(uint8_t payload_type) {
>     return -1;
>   }
>   if (last_audio_decoder_ && payload_type == last_audio_decoder_->pltype) {
>-    last_audio_decoder_ = rtc::nullopt;
>-    last_audio_format_ = rtc::nullopt;
>-    last_packet_sample_rate_hz_ = rtc::nullopt;
>+    last_audio_decoder_ = absl::nullopt;
>+    last_audio_format_ = absl::nullopt;
>+    last_packet_sample_rate_hz_ = absl::nullopt;
>   }
>   return 0;
> }
> 
>-rtc::Optional<uint32_t> AcmReceiver::GetPlayoutTimestamp() {
>+absl::optional<uint32_t> AcmReceiver::GetPlayoutTimestamp() {
>   return neteq_->GetPlayoutTimestamp();
> }
> 
>@@ -313,7 +317,7 @@ int AcmReceiver::LastAudioCodec(CodecInst* codec) const {
>   return 0;
> }
> 
>-rtc::Optional<SdpAudioFormat> AcmReceiver::LastAudioFormat() const {
>+absl::optional<SdpAudioFormat> AcmReceiver::LastAudioFormat() const {
>   rtc::CritScope lock(&crit_sect_);
>   return last_audio_format_;
> }
>@@ -350,7 +354,7 @@ void AcmReceiver::GetNetworkStatistics(NetworkStatistics* acm_stat) {
> int AcmReceiver::DecoderByPayloadType(uint8_t payload_type,
>                                       CodecInst* codec) const {
>   rtc::CritScope lock(&crit_sect_);
>-  const rtc::Optional<CodecInst> ci = neteq_->GetDecoder(payload_type);
>+  const absl::optional<CodecInst> ci = neteq_->GetDecoder(payload_type);
>   if (ci) {
>     *codec = *ci;
>     return 0;
>@@ -380,10 +384,10 @@ void AcmReceiver::ResetInitialDelay() {
>   // TODO(turajs): Should NetEq Buffer be flushed?
> }
> 
>-const rtc::Optional<CodecInst> AcmReceiver::RtpHeaderToDecoder(
>+const absl::optional<CodecInst> AcmReceiver::RtpHeaderToDecoder(
>     const RTPHeader& rtp_header,
>     uint8_t first_payload_byte) const {
>-  const rtc::Optional<CodecInst> ci =
>+  const absl::optional<CodecInst> ci =
>       neteq_->GetDecoder(rtp_header.payloadType);
>   if (ci && STR_CASE_CMP(ci->plname, "red") == 0) {
>     // This is a RED packet. Get the payload of the audio codec.
>@@ -398,10 +402,9 @@ uint32_t AcmReceiver::NowInTimestamp(int decoder_sampling_rate) const {
>   // the least significant bits. (32-6) bits cover 2^(32-6) = 67108864 ms.
>   // We masked 6 most significant bits of 32-bit so there is no overflow in
>   // the conversion from milliseconds to timestamp.
>-  const uint32_t now_in_ms = static_cast<uint32_t>(
>-      clock_->TimeInMilliseconds() & 0x03ffffff);
>-  return static_cast<uint32_t>(
>-      (decoder_sampling_rate / 1000) * now_in_ms);
>+  const uint32_t now_in_ms =
>+      static_cast<uint32_t>(clock_->TimeInMilliseconds() & 0x03ffffff);
>+  return static_cast<uint32_t>((decoder_sampling_rate / 1000) * now_in_ms);
> }
> 
> void AcmReceiver::GetDecodingCallStatistics(
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_receiver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_receiver.h
>index 5c6b36fdb0f..f133a873c26 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_receiver.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_receiver.h
>@@ -16,17 +16,16 @@
> #include <string>
> #include <vector>
> 
>+#include "absl/types/optional.h"
> #include "api/array_view.h"
>-#include "api/optional.h"
>+#include "api/audio/audio_frame.h"
> #include "common_audio/vad/include/webrtc_vad.h"
> #include "modules/audio_coding/acm2/acm_resampler.h"
> #include "modules/audio_coding/acm2/call_statistics.h"
> #include "modules/audio_coding/include/audio_coding_module.h"
> #include "modules/audio_coding/neteq/include/neteq.h"
>-#include "modules/include/module_common_types.h"
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/thread_annotations.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -159,7 +158,7 @@ class AcmReceiver {
>   // packet. If no packet of a registered non-CNG codec has been received, the
>   // return value is empty. Also, if the decoder was unregistered since the last
>   // packet was inserted, the return value is empty.
>-  rtc::Optional<int> last_packet_sample_rate_hz() const;
>+  absl::optional<int> last_packet_sample_rate_hz() const;
> 
>   // Returns last_output_sample_rate_hz from the NetEq instance.
>   int last_output_sample_rate_hz() const;
>@@ -195,7 +194,7 @@ class AcmReceiver {
> 
>   // Returns the RTP timestamp for the last sample delivered by GetAudio().
>   // The return value will be empty if no valid timestamp is available.
>-  rtc::Optional<uint32_t> GetPlayoutTimestamp();
>+  absl::optional<uint32_t> GetPlayoutTimestamp();
> 
>   // Returns the current total delay from NetEq (packet buffer and sync buffer)
>   // in ms, with smoothing applied to even out short-time fluctuations due to
>@@ -215,7 +214,7 @@ class AcmReceiver {
>   //
>   int LastAudioCodec(CodecInst* codec) const;
> 
>-  rtc::Optional<SdpAudioFormat> LastAudioFormat() const;
>+  absl::optional<SdpAudioFormat> LastAudioFormat() const;
> 
>   //
>   // Get a decoder given its registered payload-type.
>@@ -273,22 +272,23 @@ class AcmReceiver {
>     int sample_rate_hz;
>   };
> 
>-  const rtc::Optional<CodecInst> RtpHeaderToDecoder(const RTPHeader& rtp_header,
>-                                                    uint8_t first_payload_byte)
>-      const RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
>+  const absl::optional<CodecInst> RtpHeaderToDecoder(
>+      const RTPHeader& rtp_header,
>+      uint8_t first_payload_byte) const
>+      RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
> 
>   uint32_t NowInTimestamp(int decoder_sampling_rate) const;
> 
>   rtc::CriticalSection crit_sect_;
>-  rtc::Optional<CodecInst> last_audio_decoder_ RTC_GUARDED_BY(crit_sect_);
>-  rtc::Optional<SdpAudioFormat> last_audio_format_ RTC_GUARDED_BY(crit_sect_);
>+  absl::optional<CodecInst> last_audio_decoder_ RTC_GUARDED_BY(crit_sect_);
>+  absl::optional<SdpAudioFormat> last_audio_format_ RTC_GUARDED_BY(crit_sect_);
>   ACMResampler resampler_ RTC_GUARDED_BY(crit_sect_);
>   std::unique_ptr<int16_t[]> last_audio_buffer_ RTC_GUARDED_BY(crit_sect_);
>   CallStatistics call_stats_ RTC_GUARDED_BY(crit_sect_);
>   const std::unique_ptr<NetEq> neteq_;  // NetEq is thread-safe; no lock needed.
>   const Clock* const clock_;
>   bool resampled_last_output_frame_ RTC_GUARDED_BY(crit_sect_);
>-  rtc::Optional<int> last_packet_sample_rate_hz_ RTC_GUARDED_BY(crit_sect_);
>+  absl::optional<int> last_packet_sample_rate_hz_ RTC_GUARDED_BY(crit_sect_);
> };
> 
> }  // namespace acm2
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_receiver_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_receiver_unittest.cc
>index 8d0b2f100e9..457ea1d4d21 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_receiver_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_receiver_unittest.cc
>@@ -17,6 +17,7 @@
> #include "modules/audio_coding/acm2/rent_a_codec.h"
> #include "modules/audio_coding/include/audio_coding_module.h"
> #include "modules/audio_coding/neteq/tools/rtp_generator.h"
>+#include "modules/include/module_common_types.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/numerics/safe_conversions.h"
> #include "system_wrappers/include/clock.h"
>@@ -29,12 +30,11 @@ namespace acm2 {
> namespace {
> 
> bool CodecsEqual(const CodecInst& codec_a, const CodecInst& codec_b) {
>-    if (strcmp(codec_a.plname, codec_b.plname) != 0 ||
>-        codec_a.plfreq != codec_b.plfreq ||
>-        codec_a.pltype != codec_b.pltype ||
>-        codec_b.channels != codec_a.channels)
>-      return false;
>-    return true;
>+  if (strcmp(codec_a.plname, codec_b.plname) != 0 ||
>+      codec_a.plfreq != codec_b.plfreq || codec_a.pltype != codec_b.pltype ||
>+      codec_b.channels != codec_a.channels)
>+    return false;
>+  return true;
> }
> 
> struct CodecIdInst {
>@@ -82,7 +82,6 @@ class AcmReceiverTestOldApi : public AudioPacketizationCallback,
>     rtp_header_.header.numCSRCs = 0;
>     rtp_header_.header.payloadType = 0;
>     rtp_header_.frameType = kAudioFrameSpeech;
>-    rtp_header_.type.Audio.isCNG = false;
>   }
> 
>   void TearDown() override {}
>@@ -114,7 +113,7 @@ class AcmReceiverTestOldApi : public AudioPacketizationCallback,
>   }
> 
>   template <size_t N>
>-  void AddSetOfCodecs(const RentACodec::CodecId(&ids)[N]) {
>+  void AddSetOfCodecs(const RentACodec::CodecId (&ids)[N]) {
>     for (auto id : ids) {
>       const auto i = RentACodec::CodecIndexFromId(id);
>       ASSERT_TRUE(i);
>@@ -135,10 +134,6 @@ class AcmReceiverTestOldApi : public AudioPacketizationCallback,
> 
>     rtp_header_.header.payloadType = payload_type;
>     rtp_header_.frameType = frame_type;
>-    if (frame_type == kAudioFrameSpeech)
>-      rtp_header_.type.Audio.isCNG = false;
>-    else
>-      rtp_header_.type.Audio.isCNG = true;
>     rtp_header_.header.timestamp = timestamp;
> 
>     int ret_val = receiver_->InsertPacket(
>@@ -185,13 +180,13 @@ TEST_F(AcmReceiverTestOldApi, MAYBE_AddCodecGetCodec) {
>     CodecInst my_codec;
>     if (n & 0x1) {
>       // Codecs with odd index should match the reference.
>-      EXPECT_EQ(0, receiver_->DecoderByPayloadType(codecs_[n].pltype,
>-                                                   &my_codec));
>+      EXPECT_EQ(0,
>+                receiver_->DecoderByPayloadType(codecs_[n].pltype, &my_codec));
>       EXPECT_TRUE(CodecsEqual(codecs_[n], my_codec));
>     } else {
>       // Codecs with even index are not registered.
>-      EXPECT_EQ(-1, receiver_->DecoderByPayloadType(codecs_[n].pltype,
>-                                                    &my_codec));
>+      EXPECT_EQ(-1,
>+                receiver_->DecoderByPayloadType(codecs_[n].pltype, &my_codec));
>     }
>   }
> }
>@@ -297,7 +292,7 @@ TEST_F(AcmReceiverTestOldApi, MAYBE_SampleRate) {
> class AcmReceiverTestFaxModeOldApi : public AcmReceiverTestOldApi {
>  protected:
>   AcmReceiverTestFaxModeOldApi() {
>-    config_.neteq_config.playout_mode = kPlayoutFax;
>+    config_.neteq_config.for_test_no_time_stretching = true;
>   }
> 
>   void RunVerifyAudioFrame(RentACodec::CodecId codec_id) {
>@@ -306,7 +301,7 @@ class AcmReceiverTestFaxModeOldApi : public AcmReceiverTestOldApi {
>     // timestamp increments predictable; in normal mode, NetEq may decide to do
>     // accelerate or pre-emptive expand operations after some time, offsetting
>     // the timestamp.
>-    EXPECT_EQ(kPlayoutFax, config_.neteq_config.playout_mode);
>+    EXPECT_TRUE(config_.neteq_config.for_test_no_time_stretching);
> 
>     const RentACodec::CodecId kCodecId[] = {codec_id};
>     AddSetOfCodecs(kCodecId);
>@@ -325,7 +320,8 @@ class AcmReceiverTestFaxModeOldApi : public AcmReceiverTestOldApi {
>     // Expect the first output timestamp to be 5*fs/8000 samples before the
>     // first inserted timestamp (because of NetEq's look-ahead). (This value is
>     // defined in Expand::overlap_length_.)
>-    uint32_t expected_output_ts = last_packet_send_timestamp_ -
>+    uint32_t expected_output_ts =
>+        last_packet_send_timestamp_ -
>         rtc::CheckedDivExact(5 * output_sample_rate_hz, 8000);
> 
>     AudioFrame frame;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_resampler.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_resampler.cc
>index b97ced23ded..c0b20644537 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_resampler.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_resampler.cc
>@@ -19,11 +19,9 @@
> namespace webrtc {
> namespace acm2 {
> 
>-ACMResampler::ACMResampler() {
>-}
>+ACMResampler::ACMResampler() {}
> 
>-ACMResampler::~ACMResampler() {
>-}
>+ACMResampler::~ACMResampler() {}
> 
> int ACMResampler::Resample10Msec(const int16_t* in_audio,
>                                  int in_freq_hz,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_resampler.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_resampler.h
>index d7d7bcf1658..904ea524f88 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_resampler.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_resampler.h
>@@ -12,7 +12,6 @@
> #define MODULES_AUDIO_CODING_ACM2_ACM_RESAMPLER_H_
> 
> #include "common_audio/resampler/include/push_resampler.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> namespace acm2 {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_send_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_send_test.cc
>index 307c9066b1b..b1a3e9811d7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_send_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_send_test.cc
>@@ -15,6 +15,7 @@
> #include <string.h>
> 
> #include "api/audio_codecs/audio_encoder.h"
>+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
> #include "modules/audio_coding/include/audio_coding_module.h"
> #include "modules/audio_coding/neteq/tools/input_audio_file.h"
> #include "modules/audio_coding/neteq/tools/packet.h"
>@@ -28,7 +29,12 @@ AcmSendTestOldApi::AcmSendTestOldApi(InputAudioFile* audio_source,
>                                      int source_rate_hz,
>                                      int test_duration_ms)
>     : clock_(0),
>-      acm_(webrtc::AudioCodingModule::Create(&clock_)),
>+      acm_(webrtc::AudioCodingModule::Create([this] {
>+        AudioCodingModule::Config config;
>+        config.clock = &clock_;
>+        config.decoder_factory = CreateBuiltinAudioDecoderFactory();
>+        return config;
>+      }())),
>       audio_source_(audio_source),
>       source_rate_hz_(source_rate_hz),
>       input_block_size_samples_(
>@@ -89,10 +95,9 @@ std::unique_ptr<Packet> AcmSendTestOldApi::NextPacket() {
>     RTC_CHECK(audio_source_->Read(input_block_size_samples_,
>                                   input_frame_.mutable_data()));
>     if (input_frame_.num_channels_ > 1) {
>-      InputAudioFile::DuplicateInterleaved(input_frame_.data(),
>-                                           input_block_size_samples_,
>-                                           input_frame_.num_channels_,
>-                                           input_frame_.mutable_data());
>+      InputAudioFile::DuplicateInterleaved(
>+          input_frame_.data(), input_block_size_samples_,
>+          input_frame_.num_channels_, input_frame_.mutable_data());
>     }
>     data_to_send_ = false;
>     RTC_CHECK_GE(acm_->Add10MsData(input_frame_), 0);
>@@ -132,7 +137,7 @@ std::unique_ptr<Packet> AcmSendTestOldApi::CreatePacket() {
>   packet_memory[0] = 0x80;
>   packet_memory[1] = static_cast<uint8_t>(payload_type_);
>   packet_memory[2] = (sequence_number_ >> 8) & 0xFF;
>-  packet_memory[3] = (sequence_number_) & 0xFF;
>+  packet_memory[3] = (sequence_number_)&0xFF;
>   packet_memory[4] = (timestamp_ >> 24) & 0xFF;
>   packet_memory[5] = (timestamp_ >> 16) & 0xFF;
>   packet_memory[6] = (timestamp_ >> 8) & 0xFF;
>@@ -146,8 +151,7 @@ std::unique_ptr<Packet> AcmSendTestOldApi::CreatePacket() {
>   ++sequence_number_;
> 
>   // Copy the payload data.
>-  memcpy(packet_memory + kRtpHeaderSize,
>-         &last_payload_vec_[0],
>+  memcpy(packet_memory + kRtpHeaderSize, &last_payload_vec_[0],
>          last_payload_vec_.size());
>   std::unique_ptr<Packet> packet(
>       new Packet(packet_memory, allocated_bytes, clock_.TimeInMilliseconds()));
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_send_test.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_send_test.h
>index 6aea0f1b520..68ba9e1fb64 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_send_test.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/acm_send_test.h
>@@ -14,6 +14,7 @@
> #include <memory>
> #include <vector>
> 
>+#include "api/audio/audio_frame.h"
> #include "modules/audio_coding/include/audio_coding_module.h"
> #include "modules/audio_coding/neteq/tools/packet_source.h"
> #include "rtc_base/constructormagic.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/audio_coding_module.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
>index a2a3e2d987c..ef4250eb40e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
>@@ -12,11 +12,11 @@
> 
> #include <algorithm>
> 
>-#include "api/audio_codecs/builtin_audio_decoder_factory.h"
> #include "modules/audio_coding/acm2/acm_receiver.h"
> #include "modules/audio_coding/acm2/acm_resampler.h"
> #include "modules/audio_coding/acm2/codec_manager.h"
> #include "modules/audio_coding/acm2/rent_a_codec.h"
>+#include "modules/include/module_common_types.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/numerics/safe_conversions.h"
>@@ -54,7 +54,7 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
>       rtc::FunctionView<void(const AudioEncoder*)> query) override;
> 
>   // Get current send codec.
>-  rtc::Optional<CodecInst> SendCodec() const override;
>+  absl::optional<CodecInst> SendCodec() const override;
> 
>   // Get current send frequency.
>   int SendFrequency() const override;
>@@ -142,7 +142,7 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
>   // Get current received codec.
>   int ReceiveCodec(CodecInst* current_codec) const override;
> 
>-  rtc::Optional<SdpAudioFormat> ReceiveFormat() const override;
>+  absl::optional<SdpAudioFormat> ReceiveFormat() const override;
> 
>   // Incoming packet from network parsed and ready for decode.
>   int IncomingPacket(const uint8_t* incoming_payload,
>@@ -160,7 +160,7 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
> 
>   RTC_DEPRECATED int32_t PlayoutTimestamp(uint32_t* timestamp) override;
> 
>-  rtc::Optional<uint32_t> PlayoutTimestamp() override;
>+  absl::optional<uint32_t> PlayoutTimestamp() override;
> 
>   int FilteredCurrentDelayMs() const override;
> 
>@@ -323,9 +323,10 @@ int DownMix(const AudioFrame& frame,
>   if (!frame.muted()) {
>     const int16_t* frame_data = frame.data();
>     for (size_t n = 0; n < frame.samples_per_channel_; ++n) {
>-      out_buff[n] = static_cast<int16_t>(
>-          (static_cast<int32_t>(frame_data[2 * n]) +
>-           static_cast<int32_t>(frame_data[2 * n + 1])) >> 1);
>+      out_buff[n] =
>+          static_cast<int16_t>((static_cast<int32_t>(frame_data[2 * n]) +
>+                                static_cast<int32_t>(frame_data[2 * n + 1])) >>
>+                               1);
>     }
>   } else {
>     std::fill(out_buff, out_buff + frame.samples_per_channel_, 0);
>@@ -472,7 +473,7 @@ int32_t AudioCodingModuleImpl::Encode(const InputData& input_data) {
>   if (!HaveValidEncoder("Process"))
>     return -1;
> 
>-  if(!first_frame_) {
>+  if (!first_frame_) {
>     RTC_DCHECK(IsNewerTimestamp(input_data.input_timestamp, last_timestamp_))
>         << "Time should not move backwards";
>   }
>@@ -493,9 +494,10 @@ int32_t AudioCodingModuleImpl::Encode(const InputData& input_data) {
>   // Clear the buffer before reuse - encoded data will get appended.
>   encode_buffer_.Clear();
>   encoded_info = encoder_stack_->Encode(
>-      rtp_timestamp, rtc::ArrayView<const int16_t>(
>-                         input_data.audio, input_data.audio_channel *
>-                                               input_data.length_per_channel),
>+      rtp_timestamp,
>+      rtc::ArrayView<const int16_t>(
>+          input_data.audio,
>+          input_data.audio_channel * input_data.length_per_channel),
>       &encode_buffer_);
> 
>   bitrate_logger_.MaybeLog(encoder_stack_->GetTargetBitrate() / 1000);
>@@ -603,7 +605,7 @@ void AudioCodingModuleImpl::QueryEncoder(
> }
> 
> // Get current send codec.
>-rtc::Optional<CodecInst> AudioCodingModuleImpl::SendCodec() const {
>+absl::optional<CodecInst> AudioCodingModuleImpl::SendCodec() const {
>   rtc::CritScope lock(&acm_crit_sect_);
>   if (encoder_factory_) {
>     auto* ci = encoder_factory_->codec_manager.GetCodecInst();
>@@ -616,12 +618,12 @@ rtc::Optional<CodecInst> AudioCodingModuleImpl::SendCodec() const {
>     if (enc) {
>       return acm2::CodecManager::ForgeCodecInst(enc.get());
>     }
>-    return rtc::nullopt;
>+    return absl::nullopt;
>   } else {
>     return encoder_stack_
>-               ? rtc::Optional<CodecInst>(
>+               ? absl::optional<CodecInst>(
>                      acm2::CodecManager::ForgeCodecInst(encoder_stack_.get()))
>-               : rtc::nullopt;
>+               : absl::nullopt;
>   }
> }
> 
>@@ -640,7 +642,7 @@ int AudioCodingModuleImpl::SendFrequency() const {
> void AudioCodingModuleImpl::SetBitRate(int bitrate_bps) {
>   rtc::CritScope lock(&acm_crit_sect_);
>   if (encoder_stack_) {
>-    encoder_stack_->OnReceivedUplinkBandwidth(bitrate_bps, rtc::nullopt);
>+    encoder_stack_->OnReceivedUplinkBandwidth(bitrate_bps, absl::nullopt);
>   }
> }
> 
>@@ -767,7 +769,6 @@ int AudioCodingModuleImpl::PreprocessToAddData(const AudioFrame& in_frame,
>     expected_in_ts_ = in_frame.timestamp_;
>   }
> 
>-
>   if (!down_mix && !resample) {
>     // No pre-processing is required.
>     if (expected_in_ts_ == expected_codec_ts_) {
>@@ -793,8 +794,8 @@ int AudioCodingModuleImpl::PreprocessToAddData(const AudioFrame& in_frame,
>   if (down_mix) {
>     // If a resampling is required the output of a down-mix is written into a
>     // local buffer, otherwise, it will be written to the output frame.
>-    int16_t* dest_ptr_audio = resample ?
>-        audio : preprocess_frame_.mutable_data();
>+    int16_t* dest_ptr_audio =
>+        resample ? audio : preprocess_frame_.mutable_data();
>     if (DownMix(in_frame, WEBRTC_10MS_PCM_AUDIO, dest_ptr_audio) < 0)
>       return -1;
>     preprocess_frame_.num_channels_ = 1;
>@@ -912,7 +913,8 @@ int AudioCodingModuleImpl::SetVAD(bool enable_dtx,
> }
> 
> // Get VAD/DTX settings.
>-int AudioCodingModuleImpl::VAD(bool* dtx_enabled, bool* vad_enabled,
>+int AudioCodingModuleImpl::VAD(bool* dtx_enabled,
>+                               bool* vad_enabled,
>                                ACMVADMode* mode) const {
>   rtc::CritScope lock(&acm_crit_sect_);
>   const auto* sp = encoder_factory_->codec_manager.GetStackParams();
>@@ -1062,7 +1064,7 @@ int AudioCodingModuleImpl::ReceiveCodec(CodecInst* current_codec) const {
>   return receiver_.LastAudioCodec(current_codec);
> }
> 
>-rtc::Optional<SdpAudioFormat> AudioCodingModuleImpl::ReceiveFormat() const {
>+absl::optional<SdpAudioFormat> AudioCodingModuleImpl::ReceiveFormat() const {
>   rtc::CritScope lock(&acm_crit_sect_);
>   return receiver_.LastAudioFormat();
> }
>@@ -1180,14 +1182,14 @@ int AudioCodingModuleImpl::DisableOpusDtx() {
> }
> 
> int32_t AudioCodingModuleImpl::PlayoutTimestamp(uint32_t* timestamp) {
>-  rtc::Optional<uint32_t> ts = PlayoutTimestamp();
>+  absl::optional<uint32_t> ts = PlayoutTimestamp();
>   if (!ts)
>     return -1;
>   *timestamp = *ts;
>   return 0;
> }
> 
>-rtc::Optional<uint32_t> AudioCodingModuleImpl::PlayoutTimestamp() {
>+absl::optional<uint32_t> AudioCodingModuleImpl::PlayoutTimestamp() {
>   return receiver_.GetPlayoutTimestamp();
> }
> 
>@@ -1229,7 +1231,7 @@ int AudioCodingModuleImpl::LeastRequiredDelayMs() const {
> }
> 
> void AudioCodingModuleImpl::GetDecodingCallStatistics(
>-      AudioDecodingCallStats* call_stats) const {
>+    AudioDecodingCallStats* call_stats) const {
>   receiver_.GetDecodingCallStatistics(call_stats);
> }
> 
>@@ -1243,8 +1245,11 @@ ANAStats AudioCodingModuleImpl::GetANAStats() const {
> 
> }  // namespace
> 
>-AudioCodingModule::Config::Config()
>-    : neteq_config(), clock(Clock::GetRealTimeClock()) {
>+AudioCodingModule::Config::Config(
>+    rtc::scoped_refptr<AudioDecoderFactory> decoder_factory)
>+    : neteq_config(),
>+      clock(Clock::GetRealTimeClock()),
>+      decoder_factory(decoder_factory) {
>   // Post-decode VAD is disabled by default in NetEq, however, Audio
>   // Conference Mixer relies on VAD decisions and fails without them.
>   neteq_config.enable_post_decode_vad = true;
>@@ -1253,34 +1258,7 @@ AudioCodingModule::Config::Config()
> AudioCodingModule::Config::Config(const Config&) = default;
> AudioCodingModule::Config::~Config() = default;
> 
>-AudioCodingModule* AudioCodingModule::Create(int id) {
>-  RTC_UNUSED(id);
>-  return Create();
>-}
>-
>-// Create module
>-AudioCodingModule* AudioCodingModule::Create() {
>-  Config config;
>-  config.clock = Clock::GetRealTimeClock();
>-  config.decoder_factory = CreateBuiltinAudioDecoderFactory();
>-  return Create(config);
>-}
>-
>-AudioCodingModule* AudioCodingModule::Create(Clock* clock) {
>-  Config config;
>-  config.clock = clock;
>-  config.decoder_factory = CreateBuiltinAudioDecoderFactory();
>-  return Create(config);
>-}
>-
> AudioCodingModule* AudioCodingModule::Create(const Config& config) {
>-  if (!config.decoder_factory) {
>-    // TODO(ossu): Backwards compatibility. Will be removed after a deprecation
>-    // cycle.
>-    Config config_copy = config;
>-    config_copy.decoder_factory = CreateBuiltinAudioDecoderFactory();
>-    return new AudioCodingModuleImpl(config_copy);
>-  }
>   return new AudioCodingModuleImpl(config);
> }
> 
>@@ -1303,7 +1281,7 @@ int AudioCodingModule::Codec(const char* payload_name,
>                              CodecInst* codec,
>                              int sampling_freq_hz,
>                              size_t channels) {
>-  rtc::Optional<CodecInst> ci = acm2::RentACodec::CodecInstByParams(
>+  absl::optional<CodecInst> ci = acm2::RentACodec::CodecInstByParams(
>       payload_name, sampling_freq_hz, channels);
>   if (ci) {
>     *codec = *ci;
>@@ -1323,12 +1301,12 @@ int AudioCodingModule::Codec(const char* payload_name,
> int AudioCodingModule::Codec(const char* payload_name,
>                              int sampling_freq_hz,
>                              size_t channels) {
>-  rtc::Optional<acm2::RentACodec::CodecId> ci =
>+  absl::optional<acm2::RentACodec::CodecId> ci =
>       acm2::RentACodec::CodecIdByParams(payload_name, sampling_freq_hz,
>                                         channels);
>   if (!ci)
>     return -1;
>-  rtc::Optional<int> i = acm2::RentACodec::CodecIndexFromId(*ci);
>+  absl::optional<int> i = acm2::RentACodec::CodecIndexFromId(*ci);
>   return i ? *i : -1;
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest.cc
>index cc37b81cef7..01f83a796dd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest.cc
>@@ -32,12 +32,12 @@
> #include "modules/audio_coding/neteq/tools/output_wav_file.h"
> #include "modules/audio_coding/neteq/tools/packet.h"
> #include "modules/audio_coding/neteq/tools/rtp_file_source.h"
>-#include "modules/include/module_common_types.h"
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/messagedigest.h"
> #include "rtc_base/numerics/safe_conversions.h"
> #include "rtc_base/platform_thread.h"
> #include "rtc_base/refcountedobject.h"
>+#include "rtc_base/system/arch.h"
> #include "rtc_base/thread_annotations.h"
> #include "system_wrappers/include/clock.h"
> #include "system_wrappers/include/event_wrapper.h"
>@@ -79,8 +79,6 @@ class RtpUtility {
>     rtp_header->frameType = kAudioFrameSpeech;
> 
>     rtp_header->header.payload_type_frequency = kSampleRateHz;
>-    rtp_header->type.Audio.channel = 1;
>-    rtp_header->type.Audio.isCNG = false;
>   }
> 
>   void Forward(WebRtcRTPHeader* rtp_header) {
>@@ -166,7 +164,12 @@ class AudioCodingModuleTestOldApi : public ::testing::Test {
>   void TearDown() {}
> 
>   void SetUp() {
>-    acm_.reset(AudioCodingModule::Create(clock_));
>+    acm_.reset(AudioCodingModule::Create([this] {
>+      AudioCodingModule::Config config;
>+      config.clock = clock_;
>+      config.decoder_factory = CreateBuiltinAudioDecoderFactory();
>+      return config;
>+    }()));
> 
>     rtp_utility_->Populate(&rtp_header_);
> 
>@@ -237,7 +240,7 @@ class AudioCodingModuleTestOldApi : public ::testing::Test {
> 
>   // These two have to be kept in sync for now. In the future, we'll be able to
>   // eliminate the CodecInst and keep only the SdpAudioFormat.
>-  rtc::Optional<SdpAudioFormat> audio_format_;
>+  absl::optional<SdpAudioFormat> audio_format_;
>   CodecInst codec_;
> 
>   Clock* clock_;
>@@ -421,19 +424,12 @@ class AudioCodingModuleTestWithComfortNoiseOldApi
>     const struct {
>       int ix;
>       FrameType type;
>-    } expectation[] = {{2, kAudioFrameCN},
>-                       {5, kEmptyFrame},
>-                       {8, kEmptyFrame},
>-                       {11, kAudioFrameCN},
>-                       {14, kEmptyFrame},
>-                       {17, kEmptyFrame},
>-                       {20, kAudioFrameCN},
>-                       {23, kEmptyFrame},
>-                       {26, kEmptyFrame},
>-                       {29, kEmptyFrame},
>-                       {32, kAudioFrameCN},
>-                       {35, kEmptyFrame},
>-                       {38, kEmptyFrame}};
>+    } expectation[] = {
>+        {2, kAudioFrameCN},  {5, kEmptyFrame},    {8, kEmptyFrame},
>+        {11, kAudioFrameCN}, {14, kEmptyFrame},   {17, kEmptyFrame},
>+        {20, kAudioFrameCN}, {23, kEmptyFrame},   {26, kEmptyFrame},
>+        {29, kEmptyFrame},   {32, kAudioFrameCN}, {35, kEmptyFrame},
>+        {38, kEmptyFrame}};
>     for (int i = 0; i < kLoops; ++i) {
>       int num_calls_before = packet_cb_.num_calls();
>       EXPECT_EQ(i / blocks_per_packet, num_calls_before);
>@@ -682,10 +678,8 @@ class AcmIsacMtTestOldApi : public AudioCodingModuleMtTestOldApi {
>       last_packet_number_ = num_calls;
>     }
>     ASSERT_GT(last_payload_vec_.size(), 0u);
>-    ASSERT_EQ(
>-        0,
>-        acm_->IncomingPacket(
>-            &last_payload_vec_[0], last_payload_vec_.size(), rtp_header_));
>+    ASSERT_EQ(0, acm_->IncomingPacket(&last_payload_vec_[0],
>+                                      last_payload_vec_.size(), rtp_header_));
>   }
> 
>   void InsertAudio() override {
>@@ -815,9 +809,8 @@ class AcmReRegisterIsacMtTestOldApi : public AudioCodingModuleTestOldApi {
>       // Encode new frame.
>       uint32_t input_timestamp = rtp_header_.header.timestamp;
>       while (info.encoded_bytes == 0) {
>-        info =
>-            isac_encoder_->Encode(input_timestamp, audio_loop_.GetNextBlock(),
>-                                  &encoded);
>+        info = isac_encoder_->Encode(input_timestamp,
>+                                     audio_loop_.GetNextBlock(), &encoded);
>         input_timestamp += 160;  // 10 ms at 16 kHz.
>       }
>       EXPECT_EQ(rtp_header_.header.timestamp + kPacketSizeSamples,
>@@ -980,35 +973,35 @@ class AcmReceiverBitExactnessOldApi : public ::testing::Test {
> #if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX)) && \
>     defined(WEBRTC_CODEC_ILBC)
> TEST_F(AcmReceiverBitExactnessOldApi, 8kHzOutput) {
>-  Run(8000, PlatformChecksum("2adede965c6f87de7142c51552111d08",
>-                             "028c0fc414b1c9ab7e582dccdf381e98",
>-                             "36c95170c1393d4b765d1c17b61ef977",
>+  Run(8000, PlatformChecksum("7294941b62293e143d6d6c84955923fd",
>+                             "f26b8c9aa8257c7185925fa5b102f46a",
>+                             "65e5ef5c3de9c2abf3c8d0989772e9fc",
>                              "4598140b5e4f7ee66c5adad609e65a3e",
>-                             "bac5db6dff44323be401060f1279a532"));
>+                             "04a1d3e735730b6d7dbd5df10f717d27"));
> }
> 
> TEST_F(AcmReceiverBitExactnessOldApi, 16kHzOutput) {
>-  Run(16000, PlatformChecksum("c2550a3db7632de409e8db0093df1c12",
>-                              "edd31f4b6665cd5b9041fb93f2316594",
>-                              "22128bca51650cb61c80bed63b595603",
>+  Run(16000, PlatformChecksum("de8143dd3cc23241f1e1d5cf14e04b8a",
>+                              "eada3f321120d8d5afffbe4170a55d2f",
>+                              "135d8c3c7b92aa13d45cad7c91068e3e",
>                               "f2aad418af974a3b1694d5ae5cc2c3c7",
>-                              "61c3cb9386b9503feebcb829c9be54bd"));
>+                              "728b69068332efade35b1a9c32029e1b"));
> }
> 
> TEST_F(AcmReceiverBitExactnessOldApi, 32kHzOutput) {
>-  Run(32000, PlatformChecksum("85e28d7950132d56f90b099c90f82153",
>-                              "7b903f5c89997f271b405e63c245ef45",
>-                              "8b8fc6c6fd1dcdcfb3dd90e1ce597f10",
>+  Run(32000, PlatformChecksum("521d336237bdcc9ab44050e9da8917fc",
>+                              "73d44a7bedb6dfa7c70cf997223d8c10",
>+                              "f3ee2f14b03fb8e98f526f82583f84d9",
>                               "100869c8dcde51346c2073e52a272d98",
>-                              "fdec5301dc649a47d407382b587e14da"));
>+                              "5f338b4bc38707d0a14d75a357e1546e"));
> }
> 
> TEST_F(AcmReceiverBitExactnessOldApi, 48kHzOutput) {
>-  Run(48000, PlatformChecksum("ab611510e8fd6d5210a23cc04d3f0e8e",
>-                              "d8609bc9b495d81f29779344c68bcc47",
>-                              "ec5ebb90cda0ea5bb89e79d698af65de",
>+  Run(48000, PlatformChecksum("5955e31373828969de7fb308fb58a84e",
>+                              "83c0eca235b1a806426ff6ca8655cdf7",
>+                              "1126a8c03d1ebc6aa7348b9c541e2082",
>                               "bd44bf97e7899186532f91235cef444d",
>-                              "0baae2972cca142027d4af44f95f0bd5"));
>+                              "9d092dbc96e7ef6870b78c1056e87315"));
> }
> 
> TEST_F(AcmReceiverBitExactnessOldApi, 48kHzOutputExternalDecoder) {
>@@ -1053,9 +1046,11 @@ TEST_F(AcmReceiverBitExactnessOldApi, 48kHzOutputExternalDecoder) {
>                                        : fact_->IsSupportedDecoder(format);
>     }
>     std::unique_ptr<AudioDecoder> MakeAudioDecoder(
>-        const SdpAudioFormat& format) override {
>-      return format.name == "MockPCMu" ? std::move(mock_decoder_)
>-                                       : fact_->MakeAudioDecoder(format);
>+        const SdpAudioFormat& format,
>+        absl::optional<AudioCodecPairId> codec_pair_id) override {
>+      return format.name == "MockPCMu"
>+                 ? std::move(mock_decoder_)
>+                 : fact_->MakeAudioDecoder(format, codec_pair_id);
>     }
> 
>    private:
>@@ -1088,11 +1083,12 @@ TEST_F(AcmReceiverBitExactnessOldApi, 48kHzOutputExternalDecoder) {
> 
>   rtc::scoped_refptr<rtc::RefCountedObject<ADFactory>> factory(
>       new rtc::RefCountedObject<ADFactory>);
>-  Run(48000, PlatformChecksum("ab611510e8fd6d5210a23cc04d3f0e8e",
>-                              "d8609bc9b495d81f29779344c68bcc47",
>-                              "ec5ebb90cda0ea5bb89e79d698af65de",
>-                              "bd44bf97e7899186532f91235cef444d",
>-                              "0baae2972cca142027d4af44f95f0bd5"),
>+  Run(48000,
>+      PlatformChecksum("5955e31373828969de7fb308fb58a84e",
>+                       "83c0eca235b1a806426ff6ca8655cdf7",
>+                       "1126a8c03d1ebc6aa7348b9c541e2082",
>+                       "bd44bf97e7899186532f91235cef444d",
>+                       "9d092dbc96e7ef6870b78c1056e87315"),
>       factory, [](AudioCodingModule* acm) {
>         acm->RegisterReceiveCodec(0, {"MockPCMu", 8000, 1});
>       });
>@@ -1148,11 +1144,8 @@ class AcmSenderBitExactnessOldApi : public ::testing::Test,
>                          int frame_size_rtp_timestamps) {
>     payload_type_ = payload_type;
>     frame_size_rtp_timestamps_ = frame_size_rtp_timestamps;
>-    return send_test_->RegisterCodec(payload_name,
>-                                     sampling_freq_hz,
>-                                     channels,
>-                                     payload_type,
>-                                     frame_size_samples);
>+    return send_test_->RegisterCodec(payload_name, sampling_freq_hz, channels,
>+                                     payload_type, frame_size_samples);
>   }
> 
>   bool RegisterExternalSendCodec(AudioEncoder* external_speech_encoder,
>@@ -1251,11 +1244,8 @@ class AcmSenderBitExactnessOldApi : public ::testing::Test,
>                  int codec_frame_size_samples,
>                  int codec_frame_size_rtp_timestamps) {
>     ASSERT_TRUE(SetUpSender());
>-    ASSERT_TRUE(RegisterSendCodec(codec_name,
>-                                  codec_sample_rate_hz,
>-                                  channels,
>-                                  payload_type,
>-                                  codec_frame_size_samples,
>+    ASSERT_TRUE(RegisterSendCodec(codec_name, codec_sample_rate_hz, channels,
>+                                  payload_type, codec_frame_size_samples,
>                                   codec_frame_size_rtp_timestamps));
>   }
> 
>@@ -1336,82 +1326,62 @@ TEST_F(AcmSenderBitExactnessOldApi, MAYBE_IsacSwb30ms) {
> 
> TEST_F(AcmSenderBitExactnessOldApi, Pcm16_8000khz_10ms) {
>   ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 8000, 1, 107, 80, 80));
>-  Run("de4a98e1406f8b798d99cd0704e862e2",
>-      "c1edd36339ce0326cc4550041ad719a0",
>-      100,
>-      test::AcmReceiveTestOldApi::kMonoOutput);
>+  Run("de4a98e1406f8b798d99cd0704e862e2", "c1edd36339ce0326cc4550041ad719a0",
>+      100, test::AcmReceiveTestOldApi::kMonoOutput);
> }
> 
> TEST_F(AcmSenderBitExactnessOldApi, Pcm16_16000khz_10ms) {
>   ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 16000, 1, 108, 160, 160));
>-  Run("ae646d7b68384a1269cc080dd4501916",
>-      "ad786526383178b08d80d6eee06e9bad",
>-      100,
>-      test::AcmReceiveTestOldApi::kMonoOutput);
>+  Run("ae646d7b68384a1269cc080dd4501916", "ad786526383178b08d80d6eee06e9bad",
>+      100, test::AcmReceiveTestOldApi::kMonoOutput);
> }
> 
> TEST_F(AcmSenderBitExactnessOldApi, Pcm16_32000khz_10ms) {
>   ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 32000, 1, 109, 320, 320));
>-  Run("7fe325e8fbaf755e3c5df0b11a4774fb",
>-      "5ef82ea885e922263606c6fdbc49f651",
>-      100,
>-      test::AcmReceiveTestOldApi::kMonoOutput);
>+  Run("7fe325e8fbaf755e3c5df0b11a4774fb", "5ef82ea885e922263606c6fdbc49f651",
>+      100, test::AcmReceiveTestOldApi::kMonoOutput);
> }
> 
> TEST_F(AcmSenderBitExactnessOldApi, Pcm16_stereo_8000khz_10ms) {
>   ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 8000, 2, 111, 80, 80));
>-  Run("fb263b74e7ac3de915474d77e4744ceb",
>-      "62ce5adb0d4965d0a52ec98ae7f98974",
>-      100,
>-      test::AcmReceiveTestOldApi::kStereoOutput);
>+  Run("fb263b74e7ac3de915474d77e4744ceb", "62ce5adb0d4965d0a52ec98ae7f98974",
>+      100, test::AcmReceiveTestOldApi::kStereoOutput);
> }
> 
> TEST_F(AcmSenderBitExactnessOldApi, Pcm16_stereo_16000khz_10ms) {
>   ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 16000, 2, 112, 160, 160));
>-  Run("d09e9239553649d7ac93e19d304281fd",
>-      "41ca8edac4b8c71cd54fd9f25ec14870",
>-      100,
>-      test::AcmReceiveTestOldApi::kStereoOutput);
>+  Run("d09e9239553649d7ac93e19d304281fd", "41ca8edac4b8c71cd54fd9f25ec14870",
>+      100, test::AcmReceiveTestOldApi::kStereoOutput);
> }
> 
> TEST_F(AcmSenderBitExactnessOldApi, Pcm16_stereo_32000khz_10ms) {
>   ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 32000, 2, 113, 320, 320));
>-  Run("5f025d4f390982cc26b3d92fe02e3044",
>-      "50e58502fb04421bf5b857dda4c96879",
>-      100,
>-      test::AcmReceiveTestOldApi::kStereoOutput);
>+  Run("5f025d4f390982cc26b3d92fe02e3044", "50e58502fb04421bf5b857dda4c96879",
>+      100, test::AcmReceiveTestOldApi::kStereoOutput);
> }
> 
> TEST_F(AcmSenderBitExactnessOldApi, Pcmu_20ms) {
>   ASSERT_NO_FATAL_FAILURE(SetUpTest("PCMU", 8000, 1, 0, 160, 160));
>-  Run("81a9d4c0bb72e9becc43aef124c981e9",
>-      "8f9b8750bd80fe26b6cbf6659b89f0f9",
>-      50,
>-      test::AcmReceiveTestOldApi::kMonoOutput);
>+  Run("81a9d4c0bb72e9becc43aef124c981e9", "8f9b8750bd80fe26b6cbf6659b89f0f9",
>+      50, test::AcmReceiveTestOldApi::kMonoOutput);
> }
> 
> TEST_F(AcmSenderBitExactnessOldApi, Pcma_20ms) {
>   ASSERT_NO_FATAL_FAILURE(SetUpTest("PCMA", 8000, 1, 8, 160, 160));
>-  Run("39611f798969053925a49dc06d08de29",
>-      "6ad745e55aa48981bfc790d0eeef2dd1",
>-      50,
>-      test::AcmReceiveTestOldApi::kMonoOutput);
>+  Run("39611f798969053925a49dc06d08de29", "6ad745e55aa48981bfc790d0eeef2dd1",
>+      50, test::AcmReceiveTestOldApi::kMonoOutput);
> }
> 
> TEST_F(AcmSenderBitExactnessOldApi, Pcmu_stereo_20ms) {
>   ASSERT_NO_FATAL_FAILURE(SetUpTest("PCMU", 8000, 2, 110, 160, 160));
>-  Run("437bec032fdc5cbaa0d5175430af7b18",
>-      "60b6f25e8d1e74cb679cfe756dd9bca5",
>-      50,
>-      test::AcmReceiveTestOldApi::kStereoOutput);
>+  Run("437bec032fdc5cbaa0d5175430af7b18", "60b6f25e8d1e74cb679cfe756dd9bca5",
>+      50, test::AcmReceiveTestOldApi::kStereoOutput);
> }
> 
> TEST_F(AcmSenderBitExactnessOldApi, Pcma_stereo_20ms) {
>   ASSERT_NO_FATAL_FAILURE(SetUpTest("PCMA", 8000, 2, 118, 160, 160));
>-  Run("a5c6d83c5b7cedbeff734238220a4b0c",
>-      "92b282c83efd20e7eeef52ba40842cf7",
>-      50,
>-      test::AcmReceiveTestOldApi::kStereoOutput);
>+  Run("a5c6d83c5b7cedbeff734238220a4b0c", "92b282c83efd20e7eeef52ba40842cf7",
>+      50, test::AcmReceiveTestOldApi::kStereoOutput);
> }
> 
> #if defined(WEBRTC_ANDROID)
>@@ -1585,13 +1555,14 @@ class AcmSetBitRateTest : public ::testing::Test {
>     return send_test_->RegisterExternalCodec(external_speech_encoder);
>   }
> 
>-  void RunInner(int expected_total_bits) {
>+  void RunInner(int min_expected_total_bits, int max_expected_total_bits) {
>     int nr_bytes = 0;
>     while (std::unique_ptr<test::Packet> next_packet =
>                send_test_->NextPacket()) {
>       nr_bytes += rtc::checked_cast<int>(next_packet->payload_length_bytes());
>     }
>-    EXPECT_EQ(expected_total_bits, nr_bytes * 8);
>+    EXPECT_LE(min_expected_total_bits, nr_bytes * 8);
>+    EXPECT_GE(max_expected_total_bits, nr_bytes * 8);
>   }
> 
>   void SetUpTest(const char* codec_name,
>@@ -1614,10 +1585,12 @@ class AcmSetBitRateOldApi : public AcmSetBitRateTest {
>  protected:
>   // Runs the test. SetUpSender() must have been called and a codec must be set
>   // up before calling this method.
>-  void Run(int target_bitrate_bps, int expected_total_bits) {
>+  void Run(int target_bitrate_bps,
>+           int min_expected_total_bits,
>+           int max_expected_total_bits) {
>     ASSERT_TRUE(send_test_->acm());
>     send_test_->acm()->SetBitRate(target_bitrate_bps);
>-    RunInner(expected_total_bits);
>+    RunInner(min_expected_total_bits, max_expected_total_bits);
>   }
> };
> 
>@@ -1625,16 +1598,14 @@ class AcmSetBitRateNewApi : public AcmSetBitRateTest {
>  protected:
>   // Runs the test. SetUpSender() must have been called and a codec must be set
>   // up before calling this method.
>-  void Run(int expected_total_bits) { RunInner(expected_total_bits); }
>+  void Run(int min_expected_total_bits, int max_expected_total_bits) {
>+    RunInner(min_expected_total_bits, max_expected_total_bits);
>+  }
> };
> 
> TEST_F(AcmSetBitRateOldApi, Opus_48khz_20ms_10kbps) {
>   ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
>-#if defined(WEBRTC_ANDROID)
>-  Run(10000, 8640);
>-#else
>-  Run(10000, 8680);
>-#endif  // WEBRTC_ANDROID
>+  Run(10000, 8000, 12000);
> }
> 
> TEST_F(AcmSetBitRateNewApi, OpusFromFormat_48khz_20ms_10kbps) {
>@@ -1643,20 +1614,12 @@ TEST_F(AcmSetBitRateNewApi, OpusFromFormat_48khz_20ms_10kbps) {
>   const auto encoder = AudioEncoderOpus::MakeAudioEncoder(*config, 107);
>   ASSERT_TRUE(SetUpSender());
>   ASSERT_TRUE(RegisterExternalSendCodec(encoder.get(), 107));
>-#if defined(WEBRTC_ANDROID)
>-  RunInner(8640);
>-#else
>-  RunInner(8680);
>-#endif  // WEBRTC_ANDROID
>+  RunInner(8000, 12000);
> }
> 
> TEST_F(AcmSetBitRateOldApi, Opus_48khz_20ms_50kbps) {
>   ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
>-#if defined(WEBRTC_ANDROID)
>-  Run(50000, 45792);
>-#else
>-  Run(50000, 45520);
>-#endif  // WEBRTC_ANDROID
>+  Run(50000, 40000, 60000);
> }
> 
> TEST_F(AcmSetBitRateNewApi, OpusFromFormat_48khz_20ms_50kbps) {
>@@ -1665,11 +1628,7 @@ TEST_F(AcmSetBitRateNewApi, OpusFromFormat_48khz_20ms_50kbps) {
>   const auto encoder = AudioEncoderOpus::MakeAudioEncoder(*config, 107);
>   ASSERT_TRUE(SetUpSender());
>   ASSERT_TRUE(RegisterExternalSendCodec(encoder.get(), 107));
>-#if defined(WEBRTC_ANDROID)
>-  RunInner(45792);
>-#else
>-  RunInner(45520);
>-#endif  // WEBRTC_ANDROID
>+  RunInner(40000, 60000);
> }
> 
> // The result on the Android platforms is inconsistent for this test case.
>@@ -1685,7 +1644,7 @@ TEST_F(AcmSetBitRateNewApi, OpusFromFormat_48khz_20ms_50kbps) {
> #endif
> TEST_F(AcmSetBitRateOldApi, MAYBE_Opus_48khz_20ms_100kbps) {
>   ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
>-  Run(100000, 100832);
>+  Run(100000, 80000, 120000);
> }
> 
> TEST_F(AcmSetBitRateNewApi, MAYBE_OpusFromFormat_48khz_20ms_100kbps) {
>@@ -1694,18 +1653,18 @@ TEST_F(AcmSetBitRateNewApi, MAYBE_OpusFromFormat_48khz_20ms_100kbps) {
>   const auto encoder = AudioEncoderOpus::MakeAudioEncoder(*config, 107);
>   ASSERT_TRUE(SetUpSender());
>   ASSERT_TRUE(RegisterExternalSendCodec(encoder.get(), 107));
>-  RunInner(100832);
>+  RunInner(80000, 120000);
> }
> 
> // These next 2 tests ensure that the SetBitRate function has no effect on PCM
> TEST_F(AcmSetBitRateOldApi, Pcm16_8khz_10ms_8kbps) {
>   ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 8000, 1, 107, 80, 80));
>-  Run(8000, 128000);
>+  Run(8000, 128000, 128000);
> }
> 
> TEST_F(AcmSetBitRateOldApi, Pcm16_8khz_10ms_32kbps) {
>   ASSERT_NO_FATAL_FAILURE(SetUpTest("L16", 8000, 1, 107, 80, 80));
>-  Run(32000, 128000);
>+  Run(32000, 128000, 128000);
> }
> 
> // This test is for verifying the SetBitRate function. The bitrate is changed
>@@ -1745,15 +1704,18 @@ class AcmChangeBitRateOldApi : public AcmSetBitRateOldApi {
>       if (packet_counter == nr_packets / 2)
>         send_test_->acm()->SetBitRate(target_bitrate_bps);
>       if (packet_counter < nr_packets / 2)
>-        nr_bytes_before += rtc::checked_cast<int>(
>-            next_packet->payload_length_bytes());
>+        nr_bytes_before +=
>+            rtc::checked_cast<int>(next_packet->payload_length_bytes());
>       else
>-        nr_bytes_after += rtc::checked_cast<int>(
>-            next_packet->payload_length_bytes());
>+        nr_bytes_after +=
>+            rtc::checked_cast<int>(next_packet->payload_length_bytes());
>       packet_counter++;
>     }
>-    EXPECT_EQ(expected_before_switch_bits, nr_bytes_before * 8);
>-    EXPECT_EQ(expected_after_switch_bits, nr_bytes_after * 8);
>+    // Check that bitrate is 80-120 percent of expected value.
>+    EXPECT_GE(expected_before_switch_bits, nr_bytes_before * 8 * 8 / 10);
>+    EXPECT_LE(expected_before_switch_bits, nr_bytes_before * 8 * 12 / 10);
>+    EXPECT_GE(expected_after_switch_bits, nr_bytes_after * 8 * 8 / 10);
>+    EXPECT_LE(expected_after_switch_bits, nr_bytes_after * 8 * 12 / 10);
>   }
> 
>   uint32_t sampling_freq_hz_;
>@@ -1762,33 +1724,17 @@ class AcmChangeBitRateOldApi : public AcmSetBitRateOldApi {
> 
> TEST_F(AcmChangeBitRateOldApi, Opus_48khz_20ms_10kbps_2) {
>   ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
>-#if defined(WEBRTC_ANDROID)
>-  Run(10000, 29512, 4800);
>-#else
>-  Run(10000, 32200, 5368);
>-#endif  // WEBRTC_ANDROID
>+  Run(10000, 32200, 5208);
> }
> 
> TEST_F(AcmChangeBitRateOldApi, Opus_48khz_20ms_50kbps_2) {
>   ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
>-#if defined(WEBRTC_ANDROID)
>-  Run(50000, 29512, 23304);
>-#else
>-  Run(50000, 32200, 23920);
>-#endif  // WEBRTC_ANDROID
>+  Run(50000, 32200, 23928);
> }
> 
> TEST_F(AcmChangeBitRateOldApi, Opus_48khz_20ms_100kbps_2) {
>   ASSERT_NO_FATAL_FAILURE(SetUpTest("opus", 48000, 1, 107, 960, 960));
>-#if defined(WEBRTC_ANDROID)
>-  #if defined(WEBRTC_ARCH_ARM64)
>-    Run(100000, 29512, 50440);
>-  #else
>-    Run(100000, 29512, 50496);
>-  #endif  // WEBRTC_ARCH_ARM64
>-#else
>   Run(100000, 32200, 50448);
>-#endif  // WEBRTC_ANDROID
> }
> 
> // These next 2 tests ensure that the SetBitRate function has no effect on PCM
>@@ -1811,7 +1757,6 @@ TEST_F(AcmSenderBitExactnessOldApi, External_Pcmu_20ms) {
>   MockAudioEncoder mock_encoder;
>   // Set expectations on the mock encoder and also delegate the calls to the
>   // real encoder.
>-  EXPECT_CALL(mock_encoder, Die());
>   EXPECT_CALL(mock_encoder, SampleRateHz())
>       .Times(AtLeast(1))
>       .WillRepeatedly(Invoke(&encoder, &AudioEncoderPcmU::SampleRateHz));
>@@ -1830,12 +1775,10 @@ TEST_F(AcmSenderBitExactnessOldApi, External_Pcmu_20ms) {
>       .WillRepeatedly(Invoke(&encoder, &AudioEncoderPcmU::GetTargetBitrate));
>   EXPECT_CALL(mock_encoder, EncodeImpl(_, _, _))
>       .Times(AtLeast(1))
>-      .WillRepeatedly(Invoke(&encoder,
>-                             static_cast<
>-                             AudioEncoder::EncodedInfo(AudioEncoder::*)(
>-                                 uint32_t,
>-                                 rtc::ArrayView<const int16_t>,
>-                                 rtc::Buffer*)>(&AudioEncoderPcmU::Encode)));
>+      .WillRepeatedly(Invoke(
>+          &encoder, static_cast<AudioEncoder::EncodedInfo (AudioEncoder::*)(
>+                        uint32_t, rtc::ArrayView<const int16_t>, rtc::Buffer*)>(
>+                        &AudioEncoderPcmU::Encode)));
>   EXPECT_CALL(mock_encoder, SetFec(_))
>       .Times(AtLeast(1))
>       .WillRepeatedly(Invoke(&encoder, &AudioEncoderPcmU::SetFec));
>@@ -1885,11 +1828,7 @@ class AcmSwitchingOutputFrequencyOldApi : public ::testing::Test,
>     // this class.
>     test::AudioSinkFork output(this, &output_file);
>     test::AcmReceiveTestToggleOutputFreqOldApi receive_test(
>-        this,
>-        &output,
>-        output_freq_1,
>-        output_freq_2,
>-        toggle_period_ms,
>+        this, &output, output_freq_1, output_freq_2, toggle_period_ms,
>         test::AcmReceiveTestOldApi::kMonoOutput);
>     ASSERT_NO_FATAL_FAILURE(receive_test.RegisterDefaultCodecs());
>     output_freq_2_ = output_freq_2;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/call_statistics.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/call_statistics.h
>index 9dd052f003a..9dced6475a8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/call_statistics.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/call_statistics.h
>@@ -11,8 +11,8 @@
> #ifndef MODULES_AUDIO_CODING_ACM2_CALL_STATISTICS_H_
> #define MODULES_AUDIO_CODING_ACM2_CALL_STATISTICS_H_
> 
>+#include "api/audio/audio_frame.h"
> #include "common_types.h"  // NOLINT(build/include)
>-#include "modules/include/module_common_types.h"
> 
> //
> // This class is for book keeping of calls to ACM. It is not useful to log API
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/call_statistics_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/call_statistics_unittest.cc
>index 77c3863ea86..528708f2fb6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/call_statistics_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/call_statistics_unittest.cc
>@@ -52,6 +52,3 @@ TEST(CallStatisticsTest, AllCalls) {
> }  // namespace acm2
> 
> }  // namespace webrtc
>-
>-
>-
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/codec_manager.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/codec_manager.cc
>index a101d3debb4..f29e0f1a1e2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/codec_manager.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/codec_manager.cc
>@@ -14,7 +14,6 @@
> //#include "rtc_base/format_macros.h"
> #include "modules/audio_coding/acm2/rent_a_codec.h"
> #include "rtc_base/logging.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> namespace acm2 {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/codec_manager.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/codec_manager.h
>index 7485426cea4..ffbad967088 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/codec_manager.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/codec_manager.h
>@@ -13,7 +13,7 @@
> 
> #include <map>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/audio_coding/acm2/rent_a_codec.h"
> #include "modules/audio_coding/include/audio_coding_module.h"
>@@ -43,7 +43,7 @@ class CodecManager final {
>     return send_codec_inst_ ? &*send_codec_inst_ : nullptr;
>   }
> 
>-  void UnsetCodecInst() { send_codec_inst_ = rtc::nullopt; }
>+  void UnsetCodecInst() { send_codec_inst_ = absl::nullopt; }
> 
>   const RentACodec::StackParameters* GetStackParams() const {
>     return &codec_stack_params_;
>@@ -63,7 +63,7 @@ class CodecManager final {
> 
>  private:
>   rtc::ThreadChecker thread_checker_;
>-  rtc::Optional<CodecInst> send_codec_inst_;
>+  absl::optional<CodecInst> send_codec_inst_;
>   RentACodec::StackParameters codec_stack_params_;
>   bool recreate_encoder_ = true;  // Need to recreate encoder?
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/codec_manager_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/codec_manager_unittest.cc
>index e041b5e97a3..6a5ea5f52e2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/codec_manager_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/codec_manager_unittest.cc
>@@ -28,7 +28,6 @@ std::unique_ptr<MockAudioEncoder> CreateMockEncoder() {
>   EXPECT_CALL(*enc, SampleRateHz()).WillRepeatedly(Return(8000));
>   EXPECT_CALL(*enc, NumChannels()).WillRepeatedly(Return(1));
>   EXPECT_CALL(*enc, Max10MsFramesInAPacket()).WillRepeatedly(Return(1));
>-  EXPECT_CALL(*enc, Die());
>   return enc;
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/rent_a_codec.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/rent_a_codec.cc
>index 4a1746e3da9..7eff320b723 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/rent_a_codec.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/rent_a_codec.cc
>@@ -44,7 +44,7 @@
> namespace webrtc {
> namespace acm2 {
> 
>-rtc::Optional<RentACodec::CodecId> RentACodec::CodecIdByParams(
>+absl::optional<RentACodec::CodecId> RentACodec::CodecIdByParams(
>     const char* payload_name,
>     int sampling_freq_hz,
>     size_t channels) {
>@@ -52,25 +52,25 @@ rtc::Optional<RentACodec::CodecId> RentACodec::CodecIdByParams(
>       ACMCodecDB::CodecId(payload_name, sampling_freq_hz, channels));
> }
> 
>-rtc::Optional<CodecInst> RentACodec::CodecInstById(CodecId codec_id) {
>-  rtc::Optional<int> mi = CodecIndexFromId(codec_id);
>-  return mi ? rtc::Optional<CodecInst>(Database()[*mi])
>-            : rtc::nullopt;
>+absl::optional<CodecInst> RentACodec::CodecInstById(CodecId codec_id) {
>+  absl::optional<int> mi = CodecIndexFromId(codec_id);
>+  return mi ? absl::optional<CodecInst>(Database()[*mi]) : absl::nullopt;
> }
> 
>-rtc::Optional<RentACodec::CodecId> RentACodec::CodecIdByInst(
>+absl::optional<RentACodec::CodecId> RentACodec::CodecIdByInst(
>     const CodecInst& codec_inst) {
>   return CodecIdFromIndex(ACMCodecDB::CodecNumber(codec_inst));
> }
> 
>-rtc::Optional<CodecInst> RentACodec::CodecInstByParams(const char* payload_name,
>-                                                       int sampling_freq_hz,
>-                                                       size_t channels) {
>-  rtc::Optional<CodecId> codec_id =
>+absl::optional<CodecInst> RentACodec::CodecInstByParams(
>+    const char* payload_name,
>+    int sampling_freq_hz,
>+    size_t channels) {
>+  absl::optional<CodecId> codec_id =
>       CodecIdByParams(payload_name, sampling_freq_hz, channels);
>   if (!codec_id)
>-    return rtc::nullopt;
>-  rtc::Optional<CodecInst> ci = CodecInstById(*codec_id);
>+    return absl::nullopt;
>+  absl::optional<CodecInst> ci = CodecInstById(*codec_id);
>   RTC_DCHECK(ci);
> 
>   // Keep the number of channels from the function call. For most codecs it
>@@ -84,13 +84,13 @@ bool RentACodec::IsCodecValid(const CodecInst& codec_inst) {
>   return ACMCodecDB::CodecNumber(codec_inst) >= 0;
> }
> 
>-rtc::Optional<bool> RentACodec::IsSupportedNumChannels(CodecId codec_id,
>-                                                       size_t num_channels) {
>+absl::optional<bool> RentACodec::IsSupportedNumChannels(CodecId codec_id,
>+                                                        size_t num_channels) {
>   auto i = CodecIndexFromId(codec_id);
>-  return i ? rtc::Optional<bool>(
>+  return i ? absl::optional<bool>(
>                  ACMCodecDB::codec_settings_[*i].channel_support >=
>                  num_channels)
>-           : rtc::nullopt;
>+           : absl::nullopt;
> }
> 
> rtc::ArrayView<const CodecInst> RentACodec::Database() {
>@@ -98,12 +98,12 @@ rtc::ArrayView<const CodecInst> RentACodec::Database() {
>                                          NumberOfCodecs());
> }
> 
>-rtc::Optional<NetEqDecoder> RentACodec::NetEqDecoderFromCodecId(
>+absl::optional<NetEqDecoder> RentACodec::NetEqDecoderFromCodecId(
>     CodecId codec_id,
>     size_t num_channels) {
>-  rtc::Optional<int> i = CodecIndexFromId(codec_id);
>+  absl::optional<int> i = CodecIndexFromId(codec_id);
>   if (!i)
>-    return rtc::nullopt;
>+    return absl::nullopt;
>   const NetEqDecoder ned = ACMCodecDB::neteq_decoders_[*i];
>   return (ned == NetEqDecoder::kDecoderOpus && num_channels == 2)
>              ? NetEqDecoder::kDecoderOpus_2ch
>@@ -276,8 +276,7 @@ std::unique_ptr<AudioEncoder> RentACodec::RentEncoderStack(
> 
>   auto pt = [&param](const std::map<int, int>& m) {
>     auto it = m.find(param->speech_encoder->SampleRateHz());
>-    return it == m.end() ? rtc::nullopt
>-                         : rtc::Optional<int>(it->second);
>+    return it == m.end() ? absl::nullopt : absl::optional<int>(it->second);
>   };
>   auto cng_pt = pt(param->cng_payload_types);
>   param->use_cng =
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/rent_a_codec.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/rent_a_codec.h
>index f8fac4c01b0..5de78156821 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/rent_a_codec.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/rent_a_codec.h
>@@ -15,15 +15,14 @@
> #include <map>
> #include <memory>
> 
>+#include "absl/types/optional.h"
> #include "api/array_view.h"
> #include "api/audio_codecs/audio_decoder.h"
> #include "api/audio_codecs/audio_encoder.h"
>-#include "api/optional.h"
> #include "modules/audio_coding/include/audio_coding_module_typedefs.h"
> #include "modules/audio_coding/neteq/neteq_decoder_enum.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/scoped_ref_ptr.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -107,28 +106,28 @@ class RentACodec {
>     return static_cast<size_t>(CodecId::kNumCodecs);
>   }
> 
>-  static inline rtc::Optional<int> CodecIndexFromId(CodecId codec_id) {
>+  static inline absl::optional<int> CodecIndexFromId(CodecId codec_id) {
>     const int i = static_cast<int>(codec_id);
>     return i >= 0 && i < static_cast<int>(NumberOfCodecs())
>-               ? rtc::Optional<int>(i)
>-               : rtc::nullopt;
>+               ? absl::optional<int>(i)
>+               : absl::nullopt;
>   }
> 
>-  static inline rtc::Optional<CodecId> CodecIdFromIndex(int codec_index) {
>+  static inline absl::optional<CodecId> CodecIdFromIndex(int codec_index) {
>     return static_cast<size_t>(codec_index) < NumberOfCodecs()
>-               ? rtc::Optional<RentACodec::CodecId>(
>+               ? absl::optional<RentACodec::CodecId>(
>                      static_cast<RentACodec::CodecId>(codec_index))
>-               : rtc::nullopt;
>+               : absl::nullopt;
>   }
> 
>-  static rtc::Optional<CodecId> CodecIdByParams(const char* payload_name,
>-                                                int sampling_freq_hz,
>-                                                size_t channels);
>-  static rtc::Optional<CodecInst> CodecInstById(CodecId codec_id);
>-  static rtc::Optional<CodecId> CodecIdByInst(const CodecInst& codec_inst);
>-  static rtc::Optional<CodecInst> CodecInstByParams(const char* payload_name,
>-                                                    int sampling_freq_hz,
>-                                                    size_t channels);
>+  static absl::optional<CodecId> CodecIdByParams(const char* payload_name,
>+                                                 int sampling_freq_hz,
>+                                                 size_t channels);
>+  static absl::optional<CodecInst> CodecInstById(CodecId codec_id);
>+  static absl::optional<CodecId> CodecIdByInst(const CodecInst& codec_inst);
>+  static absl::optional<CodecInst> CodecInstByParams(const char* payload_name,
>+                                                     int sampling_freq_hz,
>+                                                     size_t channels);
>   static bool IsCodecValid(const CodecInst& codec_inst);
> 
>   static inline bool IsPayloadTypeValid(int payload_type) {
>@@ -137,10 +136,10 @@ class RentACodec {
> 
>   static rtc::ArrayView<const CodecInst> Database();
> 
>-  static rtc::Optional<bool> IsSupportedNumChannels(CodecId codec_id,
>-                                                    size_t num_channels);
>+  static absl::optional<bool> IsSupportedNumChannels(CodecId codec_id,
>+                                                     size_t num_channels);
> 
>-  static rtc::Optional<NetEqDecoder> NetEqDecoderFromCodecId(
>+  static absl::optional<NetEqDecoder> NetEqDecoderFromCodecId(
>       CodecId codec_id,
>       size_t num_channels);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/rent_a_codec_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/rent_a_codec_unittest.cc
>index 9eded20d23e..fd3329cfb11 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/rent_a_codec_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/rent_a_codec_unittest.cc
>@@ -10,6 +10,7 @@
> 
> #include <memory>
> 
>+#include "common_types.h"
> #include "modules/audio_coding/acm2/rent_a_codec.h"
> #include "rtc_base/arraysize.h"
> #include "test/gtest.h"
>@@ -53,8 +54,7 @@ class RentACodecTestF : public ::testing::Test {
>                        int expected_send_even_if_empty) {
>     rtc::Buffer out;
>     AudioEncoder::EncodedInfo encoded_info;
>-    encoded_info =
>-        encoder_->Encode(timestamp_, kZeroData, &out);
>+    encoded_info = encoder_->Encode(timestamp_, kZeroData, &out);
>     timestamp_ += kDataLengthSamples;
>     EXPECT_TRUE(encoded_info.redundant.empty());
>     EXPECT_EQ(expected_out_length, encoded_info.encoded_bytes);
>@@ -131,13 +131,11 @@ TEST(RentACodecTest, ExternalEncoder) {
>   {
>     ::testing::InSequence s;
>     info.encoded_timestamp = 0;
>-    EXPECT_CALL(
>-        *external_encoder,
>-        EncodeImpl(0, rtc::ArrayView<const int16_t>(audio), &encoded))
>+    EXPECT_CALL(*external_encoder,
>+                EncodeImpl(0, rtc::ArrayView<const int16_t>(audio), &encoded))
>         .WillOnce(Return(info));
>     EXPECT_CALL(marker, Mark("A"));
>     EXPECT_CALL(marker, Mark("B"));
>-    EXPECT_CALL(*external_encoder, Die());
>     EXPECT_CALL(marker, Mark("C"));
>   }
> 
>@@ -180,11 +178,9 @@ void TestCngAndRedResetSpeechEncoder(bool use_cng, bool use_red) {
>   {
>     ::testing::InSequence s;
>     EXPECT_CALL(marker, Mark("disabled"));
>-    EXPECT_CALL(*speech_encoder1, Die());
>     EXPECT_CALL(marker, Mark("enabled"));
>     if (use_cng || use_red)
>       EXPECT_CALL(*speech_encoder2, Reset());
>-    EXPECT_CALL(*speech_encoder2, Die());
>   }
> 
>   RentACodec::StackParameters param1, param2;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc
>index 55e53094c4b..c4832a36de4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.cc
>@@ -123,7 +123,7 @@ AudioEncoderRuntimeConfig AudioNetworkAdaptorImpl::GetEncoderRuntimeConfig() {
>     controller->MakeDecision(&config);
> 
>   // Update ANA stats.
>-  auto increment_opt = [](rtc::Optional<uint32_t>& a) {
>+  auto increment_opt = [](absl::optional<uint32_t>& a) {
>     a = a.value_or(0) + 1;
>   };
>   if (prev_config_) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h
>index 14000fed142..e208ed2607c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h
>@@ -75,7 +75,7 @@ class AudioNetworkAdaptorImpl final : public AudioNetworkAdaptor {
> 
>   Controller::NetworkMetrics last_metrics_;
> 
>-  rtc::Optional<AudioEncoderRuntimeConfig> prev_config_;
>+  absl::optional<AudioEncoderRuntimeConfig> prev_config_;
> 
>   ANAStats stats_;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc
>index c4379182630..5948ac36440 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc
>@@ -102,8 +102,7 @@ AudioNetworkAdaptorStates CreateAudioNetworkAdaptor() {
>   config.event_log = states.event_log.get();
>   // AudioNetworkAdaptorImpl governs the lifetime of controller manager.
>   states.audio_network_adaptor.reset(new AudioNetworkAdaptorImpl(
>-      config,
>-      std::move(controller_manager), std::move(debug_dump_writer)));
>+      config, std::move(controller_manager), std::move(debug_dump_writer)));
> 
>   return states;
> }
>@@ -192,7 +191,7 @@ TEST(AudioNetworkAdaptorImplTest,
>       "WebRTC-Audio-BitrateAdaptation/Enabled/WebRTC-Audio-FecAdaptation/"
>       "Enabled/");
>   rtc::ScopedFakeClock fake_clock;
>-  fake_clock.AdvanceTime(rtc::TimeDelta::FromMilliseconds(kClockInitialTimeMs));
>+  fake_clock.AdvanceTime(TimeDelta::ms(kClockInitialTimeMs));
>   auto states = CreateAudioNetworkAdaptor();
>   AudioEncoderRuntimeConfig config;
>   config.bitrate_bps = 32000;
>@@ -210,7 +209,7 @@ TEST(AudioNetworkAdaptorImplTest,
> TEST(AudioNetworkAdaptorImplTest,
>      DumpNetworkMetricsIsCalledOnSetNetworkMetrics) {
>   rtc::ScopedFakeClock fake_clock;
>-  fake_clock.AdvanceTime(rtc::TimeDelta::FromMilliseconds(kClockInitialTimeMs));
>+  fake_clock.AdvanceTime(TimeDelta::ms(kClockInitialTimeMs));
> 
>   auto states = CreateAudioNetworkAdaptor();
> 
>@@ -229,14 +228,14 @@ TEST(AudioNetworkAdaptorImplTest,
>               DumpNetworkMetrics(NetworkMetricsIs(check), timestamp_check));
>   states.audio_network_adaptor->SetUplinkBandwidth(kBandwidth);
> 
>-  fake_clock.AdvanceTime(rtc::TimeDelta::FromMilliseconds(100));
>+  fake_clock.AdvanceTime(TimeDelta::ms(100));
>   timestamp_check += 100;
>   check.uplink_packet_loss_fraction = kPacketLoss;
>   EXPECT_CALL(*states.mock_debug_dump_writer,
>               DumpNetworkMetrics(NetworkMetricsIs(check), timestamp_check));
>   states.audio_network_adaptor->SetUplinkPacketLossFraction(kPacketLoss);
> 
>-  fake_clock.AdvanceTime(rtc::TimeDelta::FromMilliseconds(50));
>+  fake_clock.AdvanceTime(TimeDelta::ms(50));
>   timestamp_check += 50;
>   check.uplink_recoverable_packet_loss_fraction = kRecoverablePacketLoss;
>   EXPECT_CALL(*states.mock_debug_dump_writer,
>@@ -244,21 +243,21 @@ TEST(AudioNetworkAdaptorImplTest,
>   states.audio_network_adaptor->SetUplinkRecoverablePacketLossFraction(
>       kRecoverablePacketLoss);
> 
>-  fake_clock.AdvanceTime(rtc::TimeDelta::FromMilliseconds(200));
>+  fake_clock.AdvanceTime(TimeDelta::ms(200));
>   timestamp_check += 200;
>   check.rtt_ms = kRtt;
>   EXPECT_CALL(*states.mock_debug_dump_writer,
>               DumpNetworkMetrics(NetworkMetricsIs(check), timestamp_check));
>   states.audio_network_adaptor->SetRtt(kRtt);
> 
>-  fake_clock.AdvanceTime(rtc::TimeDelta::FromMilliseconds(150));
>+  fake_clock.AdvanceTime(TimeDelta::ms(150));
>   timestamp_check += 150;
>   check.target_audio_bitrate_bps = kTargetAudioBitrate;
>   EXPECT_CALL(*states.mock_debug_dump_writer,
>               DumpNetworkMetrics(NetworkMetricsIs(check), timestamp_check));
>   states.audio_network_adaptor->SetTargetAudioBitrate(kTargetAudioBitrate);
> 
>-  fake_clock.AdvanceTime(rtc::TimeDelta::FromMilliseconds(50));
>+  fake_clock.AdvanceTime(TimeDelta::ms(50));
>   timestamp_check += 50;
>   check.overhead_bytes_per_packet = kOverhead;
>   EXPECT_CALL(*states.mock_debug_dump_writer,
>@@ -314,12 +313,12 @@ TEST(AudioNetworkAdaptorImplTest, TestANAStats) {
> 
>   auto ana_stats = states.audio_network_adaptor->GetStats();
> 
>-  EXPECT_EQ(ana_stats.bitrate_action_counter, 2);
>-  EXPECT_EQ(ana_stats.channel_action_counter, 2);
>-  EXPECT_EQ(ana_stats.dtx_action_counter, 2);
>-  EXPECT_EQ(ana_stats.fec_action_counter, 2);
>-  EXPECT_EQ(ana_stats.frame_length_increase_counter, 1);
>-  EXPECT_EQ(ana_stats.frame_length_decrease_counter, 1);
>+  EXPECT_EQ(ana_stats.bitrate_action_counter, 2u);
>+  EXPECT_EQ(ana_stats.channel_action_counter, 2u);
>+  EXPECT_EQ(ana_stats.dtx_action_counter, 2u);
>+  EXPECT_EQ(ana_stats.fec_action_counter, 2u);
>+  EXPECT_EQ(ana_stats.frame_length_increase_counter, 1u);
>+  EXPECT_EQ(ana_stats.frame_length_decrease_counter, 1u);
>   EXPECT_EQ(ana_stats.uplink_packet_loss_fraction, 0.1f);
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.h
>index 601f794b6aa..282f599b6da 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.h
>@@ -43,8 +43,8 @@ class BitrateController final : public Controller {
>   const Config config_;
>   int bitrate_bps_;
>   int frame_length_ms_;
>-  rtc::Optional<int> target_audio_bitrate_bps_;
>-  rtc::Optional<size_t> overhead_bytes_per_packet_;
>+  absl::optional<int> target_audio_bitrate_bps_;
>+  absl::optional<size_t> overhead_bytes_per_packet_;
>   RTC_DISALLOW_COPY_AND_ASSIGN(BitrateController);
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc
>index 9864511c482..f07735712dd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller_unittest.cc
>@@ -20,8 +20,8 @@ namespace {
> 
> void UpdateNetworkMetrics(
>     BitrateController* controller,
>-    const rtc::Optional<int>& target_audio_bitrate_bps,
>-    const rtc::Optional<size_t>& overhead_bytes_per_packet) {
>+    const absl::optional<int>& target_audio_bitrate_bps,
>+    const absl::optional<size_t>& overhead_bytes_per_packet) {
>   // UpdateNetworkMetrics can accept multiple network metric updates at once.
>   // However, currently, the most used case is to update one metric at a time.
>   // To reflect this fact, we separate the calls.
>@@ -38,7 +38,7 @@ void UpdateNetworkMetrics(
> }
> 
> void CheckDecision(BitrateController* controller,
>-                   const rtc::Optional<int>& frame_length_ms,
>+                   const absl::optional<int>& frame_length_ms,
>                    int expected_bitrate_bps) {
>   AudioEncoderRuntimeConfig config;
>   config.frame_length_ms = frame_length_ms;
>@@ -58,7 +58,7 @@ TEST(AnaBitrateControllerTest, OutputInitValueWhenTargetBitrateUnknown) {
>   constexpr size_t kOverheadBytesPerPacket = 64;
>   BitrateController controller(BitrateController::Config(
>       kInitialBitrateBps, kInitialFrameLengthMs, 0, 0));
>-  UpdateNetworkMetrics(&controller, rtc::nullopt, kOverheadBytesPerPacket);
>+  UpdateNetworkMetrics(&controller, absl::nullopt, kOverheadBytesPerPacket);
>   CheckDecision(&controller, kInitialFrameLengthMs * 2, kInitialBitrateBps);
> }
> 
>@@ -68,7 +68,7 @@ TEST(AnaBitrateControllerTest, OutputInitValueWhenOverheadUnknown) {
>   constexpr int kTargetBitrateBps = 48000;
>   BitrateController controller(BitrateController::Config(
>       kInitialBitrateBps, kInitialFrameLengthMs, 0, 0));
>-  UpdateNetworkMetrics(&controller, kTargetBitrateBps, rtc::nullopt);
>+  UpdateNetworkMetrics(&controller, kTargetBitrateBps, absl::nullopt);
>   CheckDecision(&controller, kInitialFrameLengthMs * 2, kInitialBitrateBps);
> }
> 
>@@ -80,9 +80,9 @@ TEST(AnaBitrateControllerTest, ChangeBitrateOnTargetBitrateChanged) {
>       BitrateController::Config(32000, kInitialFrameLengthMs, 0, 0));
>   constexpr int kTargetBitrateBps = 48000;
>   constexpr size_t kOverheadBytesPerPacket = 64;
>-  constexpr int kBitrateBps =
>-      kTargetBitrateBps -
>-      kOverheadBytesPerPacket * 8 * 1000 / kInitialFrameLengthMs;
>+  constexpr int kBitrateBps = kTargetBitrateBps - kOverheadBytesPerPacket * 8 *
>+                                                      1000 /
>+                                                      kInitialFrameLengthMs;
>   // Frame length unchanged, bitrate changes in accordance with
>   // |metrics.target_audio_bitrate_bps| and |metrics.overhead_bytes_per_packet|.
>   UpdateNetworkMetrics(&controller, kTargetBitrateBps, kOverheadBytesPerPacket);
>@@ -104,9 +104,9 @@ TEST(AnaBitrateControllerTest, UpdateMultipleNetworkMetricsAtOnce) {
>       BitrateController::Config(32000, kInitialFrameLengthMs, 0, 0));
>   constexpr int kTargetBitrateBps = 48000;
>   constexpr size_t kOverheadBytesPerPacket = 64;
>-  constexpr int kBitrateBps =
>-      kTargetBitrateBps -
>-      kOverheadBytesPerPacket * 8 * 1000 / kInitialFrameLengthMs;
>+  constexpr int kBitrateBps = kTargetBitrateBps - kOverheadBytesPerPacket * 8 *
>+                                                      1000 /
>+                                                      kInitialFrameLengthMs;
>   Controller::NetworkMetrics network_metrics;
>   network_metrics.target_audio_bitrate_bps = kTargetBitrateBps;
>   network_metrics.overhead_bytes_per_packet = kOverheadBytesPerPacket;
>@@ -122,11 +122,11 @@ TEST(AnaBitrateControllerTest, TreatUnknownFrameLengthAsFrameLengthUnchanged) {
>       BitrateController::Config(32000, kInitialFrameLengthMs, 0, 0));
>   constexpr int kTargetBitrateBps = 48000;
>   constexpr size_t kOverheadBytesPerPacket = 64;
>-  constexpr int kBitrateBps =
>-      kTargetBitrateBps -
>-      kOverheadBytesPerPacket * 8 * 1000 / kInitialFrameLengthMs;
>+  constexpr int kBitrateBps = kTargetBitrateBps - kOverheadBytesPerPacket * 8 *
>+                                                      1000 /
>+                                                      kInitialFrameLengthMs;
>   UpdateNetworkMetrics(&controller, kTargetBitrateBps, kOverheadBytesPerPacket);
>-  CheckDecision(&controller, rtc::nullopt, kBitrateBps);
>+  CheckDecision(&controller, absl::nullopt, kBitrateBps);
> }
> 
> TEST(AnaBitrateControllerTest, IncreaseBitrateOnFrameLengthIncreased) {
>@@ -138,11 +138,11 @@ TEST(AnaBitrateControllerTest, IncreaseBitrateOnFrameLengthIncreased) {
> 
>   constexpr int kTargetBitrateBps = 48000;
>   constexpr size_t kOverheadBytesPerPacket = 64;
>-  constexpr int kBitrateBps =
>-      kTargetBitrateBps -
>-      kOverheadBytesPerPacket * 8 * 1000 / kInitialFrameLengthMs;
>+  constexpr int kBitrateBps = kTargetBitrateBps - kOverheadBytesPerPacket * 8 *
>+                                                      1000 /
>+                                                      kInitialFrameLengthMs;
>   UpdateNetworkMetrics(&controller, kTargetBitrateBps, kOverheadBytesPerPacket);
>-  CheckDecision(&controller, rtc::nullopt, kBitrateBps);
>+  CheckDecision(&controller, absl::nullopt, kBitrateBps);
> 
>   constexpr int kFrameLengthMs = 60;
>   constexpr size_t kPacketOverheadRateDiff =
>@@ -162,11 +162,11 @@ TEST(AnaBitrateControllerTest, DecreaseBitrateOnFrameLengthDecreased) {
> 
>   constexpr int kTargetBitrateBps = 48000;
>   constexpr size_t kOverheadBytesPerPacket = 64;
>-  constexpr int kBitrateBps =
>-      kTargetBitrateBps -
>-      kOverheadBytesPerPacket * 8 * 1000 / kInitialFrameLengthMs;
>+  constexpr int kBitrateBps = kTargetBitrateBps - kOverheadBytesPerPacket * 8 *
>+                                                      1000 /
>+                                                      kInitialFrameLengthMs;
>   UpdateNetworkMetrics(&controller, kTargetBitrateBps, kOverheadBytesPerPacket);
>-  CheckDecision(&controller, rtc::nullopt, kBitrateBps);
>+  CheckDecision(&controller, absl::nullopt, kBitrateBps);
> 
>   constexpr int kFrameLengthMs = 20;
>   constexpr size_t kPacketOverheadRateDiff =
>@@ -213,9 +213,9 @@ TEST(AnaBitrateControllerTest, CheckBehaviorOnChangingCondition) {
> 
>   // Next: change frame length.
>   frame_length_ms = 60;
>-  current_bitrate += rtc::checked_cast<int>(
>-      overhead_bytes_per_packet * 8 * 1000 / 20 -
>-      overhead_bytes_per_packet * 8 * 1000 / 60);
>+  current_bitrate +=
>+      rtc::checked_cast<int>(overhead_bytes_per_packet * 8 * 1000 / 20 -
>+                             overhead_bytes_per_packet * 8 * 1000 / 60);
>   UpdateNetworkMetrics(&controller, overall_bitrate, overhead_bytes_per_packet);
>   CheckDecision(&controller, frame_length_ms, current_bitrate);
> 
>@@ -227,9 +227,9 @@ TEST(AnaBitrateControllerTest, CheckBehaviorOnChangingCondition) {
> 
>   // Next: change frame length.
>   frame_length_ms = 20;
>-  current_bitrate -= rtc::checked_cast<int>(
>-      overhead_bytes_per_packet * 8 * 1000 / 20 -
>-      overhead_bytes_per_packet * 8 * 1000 / 60);
>+  current_bitrate -=
>+      rtc::checked_cast<int>(overhead_bytes_per_packet * 8 * 1000 / 20 -
>+                             overhead_bytes_per_packet * 8 * 1000 / 60);
>   UpdateNetworkMetrics(&controller, overall_bitrate, overhead_bytes_per_packet);
>   CheckDecision(&controller, frame_length_ms, current_bitrate);
> 
>@@ -237,9 +237,9 @@ TEST(AnaBitrateControllerTest, CheckBehaviorOnChangingCondition) {
>   overall_bitrate -= 100;
>   current_bitrate -= 100;
>   frame_length_ms = 60;
>-  current_bitrate += rtc::checked_cast<int>(
>-      overhead_bytes_per_packet * 8 * 1000 / 20 -
>-      overhead_bytes_per_packet * 8 * 1000 / 60);
>+  current_bitrate +=
>+      rtc::checked_cast<int>(overhead_bytes_per_packet * 8 * 1000 / 20 -
>+                             overhead_bytes_per_packet * 8 * 1000 / 60);
> 
>   UpdateNetworkMetrics(&controller, overall_bitrate, overhead_bytes_per_packet);
>   CheckDecision(&controller, frame_length_ms, current_bitrate);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/channel_controller.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/channel_controller.h
>index f53ddd6b197..23cbef663b4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/channel_controller.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/channel_controller.h
>@@ -44,7 +44,7 @@ class ChannelController final : public Controller {
>  private:
>   const Config config_;
>   size_t channels_to_encode_;
>-  rtc::Optional<int> uplink_bandwidth_bps_;
>+  absl::optional<int> uplink_bandwidth_bps_;
>   RTC_DISALLOW_COPY_AND_ASSIGN(ChannelController);
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/channel_controller_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/channel_controller_unittest.cc
>index 64e5daee2bc..bfa6f01b98f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/channel_controller_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/channel_controller_unittest.cc
>@@ -32,7 +32,7 @@ std::unique_ptr<ChannelController> CreateChannelController(int init_channels) {
> }
> 
> void CheckDecision(ChannelController* controller,
>-                   const rtc::Optional<int>& uplink_bandwidth_bps,
>+                   const absl::optional<int>& uplink_bandwidth_bps,
>                    size_t expected_num_channels) {
>   if (uplink_bandwidth_bps) {
>     Controller::NetworkMetrics network_metrics;
>@@ -49,7 +49,7 @@ void CheckDecision(ChannelController* controller,
> TEST(ChannelControllerTest, OutputInitValueWhenUplinkBandwidthUnknown) {
>   constexpr int kInitChannels = 2;
>   auto controller = CreateChannelController(kInitChannels);
>-  CheckDecision(controller.get(), rtc::nullopt, kInitChannels);
>+  CheckDecision(controller.get(), absl::nullopt, kInitChannels);
> }
> 
> TEST(ChannelControllerTest, SwitchTo2ChannelsOnHighUplinkBandwidth) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/controller.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/controller.h
>index af2f569f895..19d8599d5e2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/controller.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/controller.h
>@@ -11,7 +11,7 @@
> #ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CONTROLLER_H_
> #define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_CONTROLLER_H_
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> #include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
> 
> namespace webrtc {
>@@ -21,12 +21,12 @@ class Controller {
>   struct NetworkMetrics {
>     NetworkMetrics();
>     ~NetworkMetrics();
>-    rtc::Optional<int> uplink_bandwidth_bps;
>-    rtc::Optional<float> uplink_packet_loss_fraction;
>-    rtc::Optional<float> uplink_recoverable_packet_loss_fraction;
>-    rtc::Optional<int> target_audio_bitrate_bps;
>-    rtc::Optional<int> rtt_ms;
>-    rtc::Optional<size_t> overhead_bytes_per_packet;
>+    absl::optional<int> uplink_bandwidth_bps;
>+    absl::optional<float> uplink_packet_loss_fraction;
>+    absl::optional<float> uplink_recoverable_packet_loss_fraction;
>+    absl::optional<int> target_audio_bitrate_bps;
>+    absl::optional<int> rtt_ms;
>+    absl::optional<size_t> overhead_bytes_per_packet;
>   };
> 
>   virtual ~Controller() = default;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.cc
>index 313aa628c7c..32f9fcb07be 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.cc
>@@ -296,9 +296,9 @@ std::unique_ptr<ControllerManager> ControllerManagerImpl::Create(
>   }
> 
>   if (scoring_points.size() == 0) {
>-    return std::unique_ptr<ControllerManagerImpl>(new ControllerManagerImpl(
>-        ControllerManagerImpl::Config(0, 0), std::move(controllers),
>-        scoring_points));
>+    return std::unique_ptr<ControllerManagerImpl>(
>+        new ControllerManagerImpl(ControllerManagerImpl::Config(0, 0),
>+                                  std::move(controllers), scoring_points));
>   } else {
>     RTC_CHECK(controller_manager_config.has_min_reordering_time_ms());
>     RTC_CHECK(controller_manager_config.has_min_reordering_squared_distance());
>@@ -327,7 +327,7 @@ ControllerManagerImpl::ControllerManagerImpl(
>     const std::map<const Controller*, std::pair<int, float>>& scoring_points)
>     : config_(config),
>       controllers_(std::move(controllers)),
>-      last_reordering_time_ms_(rtc::nullopt),
>+      last_reordering_time_ms_(absl::nullopt),
>       last_scoring_point_(0, 0.0) {
>   for (auto& controller : controllers_)
>     default_sorted_controllers_.push_back(controller.get());
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.h
>index 5c63f2f5142..1ff9bbf5f23 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.h
>@@ -104,7 +104,7 @@ class ControllerManagerImpl final : public ControllerManager {
> 
>   std::vector<std::unique_ptr<Controller>> controllers_;
> 
>-  rtc::Optional<int64_t> last_reordering_time_ms_;
>+  absl::optional<int64_t> last_reordering_time_ms_;
>   ScoringPoint last_scoring_point_;
> 
>   std::vector<Controller*> default_sorted_controllers_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc
>index 576661cd35f..061e4aa8c6a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager_unittest.cc
>@@ -87,8 +87,8 @@ ControllerManagerStates CreateControllerManager() {
> // exists in the vector.
> void CheckControllersOrder(
>     ControllerManagerStates* states,
>-    const rtc::Optional<int>& uplink_bandwidth_bps,
>-    const rtc::Optional<float>& uplink_packet_loss_fraction,
>+    const absl::optional<int>& uplink_bandwidth_bps,
>+    const absl::optional<float>& uplink_packet_loss_fraction,
>     const std::vector<int>& expected_order) {
>   RTC_DCHECK_EQ(kNumControllers, expected_order.size());
>   Controller::NetworkMetrics metrics;
>@@ -124,8 +124,7 @@ TEST(ControllerManagerTest, ControllersInDefaultOrderOnEmptyNetworkMetrics) {
>   auto states = CreateControllerManager();
>   // |network_metrics| are empty, and the controllers are supposed to follow the
>   // default order.
>-  CheckControllersOrder(&states, rtc::nullopt, rtc::nullopt,
>-                        {0, 1, 2, 3});
>+  CheckControllersOrder(&states, absl::nullopt, absl::nullopt, {0, 1, 2, 3});
> }
> 
> TEST(ControllerManagerTest, ControllersWithoutCharPointAtEndAndInDefaultOrder) {
>@@ -148,8 +147,7 @@ TEST(ControllerManagerTest, DoNotReorderBeforeMinReordingTime) {
>   CheckControllersOrder(&states, kChracteristicBandwithBps[0],
>                         kChracteristicPacketLossFraction[0],
>                         {kNumControllers - 2, kNumControllers - 1, 0, 1});
>-  fake_clock.AdvanceTime(
>-      rtc::TimeDelta::FromMilliseconds(kMinReorderingTimeMs - 1));
>+  fake_clock.AdvanceTime(TimeDelta::ms(kMinReorderingTimeMs - 1));
>   // Move uplink bandwidth and packet loss fraction to the other controller's
>   // characteristic point, which would cause controller manager to reorder the
>   // controllers if time had reached min reordering time.
>@@ -170,8 +168,7 @@ TEST(ControllerManagerTest, ReorderBeyondMinReordingTimeAndMinDistance) {
>   // of two controllers.
>   CheckControllersOrder(&states, kBandwidthBps, kPacketLossFraction,
>                         {kNumControllers - 2, kNumControllers - 1, 0, 1});
>-  fake_clock.AdvanceTime(
>-      rtc::TimeDelta::FromMilliseconds(kMinReorderingTimeMs));
>+  fake_clock.AdvanceTime(TimeDelta::ms(kMinReorderingTimeMs));
>   // Then let network metrics move a little towards the other controller.
>   CheckControllersOrder(&states, kBandwidthBps - kMinBandwithChangeBps - 1,
>                         kPacketLossFraction,
>@@ -190,8 +187,7 @@ TEST(ControllerManagerTest, DoNotReorderIfNetworkMetricsChangeTooSmall) {
>   // of two controllers.
>   CheckControllersOrder(&states, kBandwidthBps, kPacketLossFraction,
>                         {kNumControllers - 2, kNumControllers - 1, 0, 1});
>-  fake_clock.AdvanceTime(
>-      rtc::TimeDelta::FromMilliseconds(kMinReorderingTimeMs));
>+  fake_clock.AdvanceTime(TimeDelta::ms(kMinReorderingTimeMs));
>   // Then let network metrics move a little towards the other controller.
>   CheckControllersOrder(&states, kBandwidthBps - kMinBandwithChangeBps + 1,
>                         kPacketLossFraction,
>@@ -351,7 +347,7 @@ TEST(ControllerManagerTest, DebugDumpLoggedWhenCreateFromConfigString) {
> 
>   constexpr int64_t kClockInitialTimeMs = 12345678;
>   rtc::ScopedFakeClock fake_clock;
>-  fake_clock.AdvanceTime(rtc::TimeDelta::FromMilliseconds(kClockInitialTimeMs));
>+  fake_clock.AdvanceTime(TimeDelta::ms(kClockInitialTimeMs));
>   auto debug_dump_writer =
>       std::unique_ptr<MockDebugDumpWriter>(new NiceMock<MockDebugDumpWriter>());
>   EXPECT_CALL(*debug_dump_writer, Die());
>@@ -448,8 +444,7 @@ TEST(ControllerManagerTest, CreateFromConfigStringAndCheckReordering) {
> 
>   metrics.uplink_bandwidth_bps = kChracteristicBandwithBps[1];
>   metrics.uplink_packet_loss_fraction = kChracteristicPacketLossFraction[1];
>-  fake_clock.AdvanceTime(
>-      rtc::TimeDelta::FromMilliseconds(kMinReorderingTimeMs - 1));
>+  fake_clock.AdvanceTime(TimeDelta::ms(kMinReorderingTimeMs - 1));
>   controllers = states.controller_manager->GetSortedControllers(metrics);
>   // Should not reorder since min reordering time is not met.
>   CheckControllersOrder(controllers,
>@@ -458,7 +453,7 @@ TEST(ControllerManagerTest, CreateFromConfigStringAndCheckReordering) {
>                             ControllerType::CHANNEL, ControllerType::DTX,
>                             ControllerType::BIT_RATE});
> 
>-  fake_clock.AdvanceTime(rtc::TimeDelta::FromMilliseconds(1));
>+  fake_clock.AdvanceTime(TimeDelta::ms(1));
>   controllers = states.controller_manager->GetSortedControllers(metrics);
>   // Reorder now.
>   CheckControllersOrder(controllers,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.h
>index e40c8328af0..b464eb6eefd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.h
>@@ -17,7 +17,7 @@
> #include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/ignore_wundef.h"
>-#include "system_wrappers/include/file_wrapper.h"
>+#include "rtc_base/system/file_wrapper.h"
> #if WEBRTC_ENABLE_PROTOBUF
> RTC_PUSH_IGNORING_WUNDEF()
> #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/dtx_controller.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/dtx_controller.h
>index 8a2427ed7c8..fb40db28b20 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/dtx_controller.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/dtx_controller.h
>@@ -40,7 +40,7 @@ class DtxController final : public Controller {
>  private:
>   const Config config_;
>   bool dtx_enabled_;
>-  rtc::Optional<int> uplink_bandwidth_bps_;
>+  absl::optional<int> uplink_bandwidth_bps_;
>   RTC_DISALLOW_COPY_AND_ASSIGN(DtxController);
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/dtx_controller_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/dtx_controller_unittest.cc
>index e38e65d8f57..67bf9e5af1d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/dtx_controller_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/dtx_controller_unittest.cc
>@@ -30,7 +30,7 @@ std::unique_ptr<DtxController> CreateController(int initial_dtx_enabled) {
> }
> 
> void CheckDecision(DtxController* controller,
>-                   const rtc::Optional<int>& uplink_bandwidth_bps,
>+                   const absl::optional<int>& uplink_bandwidth_bps,
>                    bool expected_dtx_enabled) {
>   if (uplink_bandwidth_bps) {
>     Controller::NetworkMetrics network_metrics;
>@@ -47,7 +47,7 @@ void CheckDecision(DtxController* controller,
> TEST(DtxControllerTest, OutputInitValueWhenUplinkBandwidthUnknown) {
>   constexpr bool kInitialDtxEnabled = true;
>   auto controller = CreateController(kInitialDtxEnabled);
>-  CheckDecision(controller.get(), rtc::nullopt, kInitialDtxEnabled);
>+  CheckDecision(controller.get(), absl::nullopt, kInitialDtxEnabled);
> }
> 
> TEST(DtxControllerTest, TurnOnDtxForLowUplinkBandwidth) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.cc
>index 9cdbc54c67f..4a92343d968 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer.cc
>@@ -12,10 +12,10 @@
> 
> #include <algorithm>
> 
>+#include "absl/memory/memory.h"
> #include "logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h"
> #include "logging/rtc_event_log/rtc_event_log.h"
> #include "modules/audio_coding/audio_network_adaptor/event_log_writer.h"
>-#include "rtc_base/ptr_util.h"
> 
> namespace webrtc {
> 
>@@ -63,9 +63,9 @@ void EventLogWriter::MaybeLogEncoderConfig(
> }
> 
> void EventLogWriter::LogEncoderConfig(const AudioEncoderRuntimeConfig& config) {
>-  auto config_copy = rtc::MakeUnique<AudioEncoderRuntimeConfig>(config);
>-  event_log_->Log(
>-      rtc::MakeUnique<RtcEventAudioNetworkAdaptation>(std::move(config_copy)));
>+  auto config_copy = absl::make_unique<AudioEncoderRuntimeConfig>(config);
>+  event_log_->Log(absl::make_unique<RtcEventAudioNetworkAdaptation>(
>+      std::move(config_copy)));
>   last_logged_config_ = config;
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.cc
>index 62f356dd078..7ab72c9437e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.cc
>@@ -23,7 +23,7 @@ class NullSmoothingFilter final : public SmoothingFilter {
>  public:
>   void AddSample(float sample) override { last_sample_ = sample; }
> 
>-  rtc::Optional<float> GetAverage() override { return last_sample_; }
>+  absl::optional<float> GetAverage() override { return last_sample_; }
> 
>   bool SetTimeConstantMs(int time_constant_ms) override {
>     RTC_NOTREACHED();
>@@ -31,9 +31,9 @@ class NullSmoothingFilter final : public SmoothingFilter {
>   }
> 
>  private:
>-  rtc::Optional<float> last_sample_;
>+  absl::optional<float> last_sample_;
> };
>-}
>+}  // namespace
> 
> FecControllerPlrBased::Config::Config(
>     bool initial_fec_enabled,
>@@ -89,7 +89,7 @@ void FecControllerPlrBased::MakeDecision(AudioEncoderRuntimeConfig* config) {
> }
> 
> bool FecControllerPlrBased::FecEnablingDecision(
>-    const rtc::Optional<float>& packet_loss) const {
>+    const absl::optional<float>& packet_loss) const {
>   if (!uplink_bandwidth_bps_ || !packet_loss) {
>     return false;
>   } else {
>@@ -100,7 +100,7 @@ bool FecControllerPlrBased::FecEnablingDecision(
> }
> 
> bool FecControllerPlrBased::FecDisablingDecision(
>-    const rtc::Optional<float>& packet_loss) const {
>+    const absl::optional<float>& packet_loss) const {
>   if (!uplink_bandwidth_bps_ || !packet_loss) {
>     return false;
>   } else {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h
>index c2735370d06..b66883ef552 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h
>@@ -56,12 +56,12 @@ class FecControllerPlrBased final : public Controller {
>   void MakeDecision(AudioEncoderRuntimeConfig* config) override;
> 
>  private:
>-  bool FecEnablingDecision(const rtc::Optional<float>& packet_loss) const;
>-  bool FecDisablingDecision(const rtc::Optional<float>& packet_loss) const;
>+  bool FecEnablingDecision(const absl::optional<float>& packet_loss) const;
>+  bool FecDisablingDecision(const absl::optional<float>& packet_loss) const;
> 
>   const Config config_;
>   bool fec_enabled_;
>-  rtc::Optional<int> uplink_bandwidth_bps_;
>+  absl::optional<int> uplink_bandwidth_bps_;
>   const std::unique_ptr<SmoothingFilter> packet_loss_smoother_;
> 
>   RTC_DISALLOW_COPY_AND_ASSIGN(FecControllerPlrBased);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based_unittest.cc
>index 8636aa96b19..de66717fba8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based_unittest.cc
>@@ -78,8 +78,8 @@ FecControllerPlrBasedTestStates CreateFecControllerPlrBased(
> }
> 
> void UpdateNetworkMetrics(FecControllerPlrBasedTestStates* states,
>-                          const rtc::Optional<int>& uplink_bandwidth_bps,
>-                          const rtc::Optional<float>& uplink_packet_loss) {
>+                          const absl::optional<int>& uplink_bandwidth_bps,
>+                          const absl::optional<float>& uplink_packet_loss) {
>   // UpdateNetworkMetrics can accept multiple network metric updates at once.
>   // However, currently, the most used case is to update one metric at a time.
>   // To reflect this fact, we separate the calls.
>@@ -131,7 +131,7 @@ TEST(FecControllerPlrBasedTest, OutputInitValueWhenUplinkBandwidthUnknown) {
>           kEnablingPacketLossAtLowBw - kEpsilon, kEnablingPacketLossAtLowBw,
>           kEnablingPacketLossAtLowBw + kEpsilon}) {
>       auto states = CreateFecControllerPlrBased(initial_fec_enabled);
>-      UpdateNetworkMetrics(&states, rtc::nullopt, packet_loss);
>+      UpdateNetworkMetrics(&states, absl::nullopt, packet_loss);
>       CheckDecision(&states, initial_fec_enabled, packet_loss);
>     }
>   }
>@@ -146,7 +146,7 @@ TEST(FecControllerPlrBasedTest,
>                           kDisablingBandwidthLow + 1, kEnablingBandwidthLow - 1,
>                           kEnablingBandwidthLow, kEnablingBandwidthLow + 1}) {
>       auto states = CreateFecControllerPlrBased(initial_fec_enabled);
>-      UpdateNetworkMetrics(&states, bandwidth, rtc::nullopt);
>+      UpdateNetworkMetrics(&states, bandwidth, absl::nullopt);
>       CheckDecision(&states, initial_fec_enabled, 0.0);
>     }
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based.h
>index ade55aea5d3..9a3c37c3d1d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based.h
>@@ -55,8 +55,8 @@ class FecControllerRplrBased final : public Controller {
> 
>   const Config config_;
>   bool fec_enabled_;
>-  rtc::Optional<int> uplink_bandwidth_bps_;
>-  rtc::Optional<float> uplink_recoverable_packet_loss_;
>+  absl::optional<int> uplink_bandwidth_bps_;
>+  absl::optional<float> uplink_recoverable_packet_loss_;
> 
>   RTC_DISALLOW_COPY_AND_ASSIGN(FecControllerRplrBased);
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based_unittest.cc
>index 0fc003beab8..8e8704ee738 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based_unittest.cc
>@@ -44,14 +44,14 @@ constexpr float kEnablingRecoverablePacketLossAtHighBw = 0.05f;
> 
> constexpr float kEpsilon = 1e-5f;
> 
>-rtc::Optional<float> GetRandomProbabilityOrUnknown() {
>+absl::optional<float> GetRandomProbabilityOrUnknown() {
>   std::random_device rd;
>   std::mt19937 generator(rd());
>   std::uniform_real_distribution<> distribution(0, 1);
> 
>   return (distribution(generator) < 0.2)
>-             ? rtc::nullopt
>-             : rtc::Optional<float>(distribution(generator));
>+             ? absl::nullopt
>+             : absl::optional<float>(distribution(generator));
> }
> 
> std::unique_ptr<FecControllerRplrBased> CreateFecControllerRplrBased(
>@@ -70,9 +70,9 @@ std::unique_ptr<FecControllerRplrBased> CreateFecControllerRplrBased(
> 
> void UpdateNetworkMetrics(
>     FecControllerRplrBased* controller,
>-    const rtc::Optional<int>& uplink_bandwidth_bps,
>-    const rtc::Optional<float>& uplink_packet_loss,
>-    const rtc::Optional<float>& uplink_recoveralbe_packet_loss) {
>+    const absl::optional<int>& uplink_bandwidth_bps,
>+    const absl::optional<float>& uplink_packet_loss,
>+    const absl::optional<float>& uplink_recoveralbe_packet_loss) {
>   // UpdateNetworkMetrics can accept multiple network metric updates at once.
>   // However, currently, the most used case is to update one metric at a time.
>   // To reflect this fact, we separate the calls.
>@@ -96,8 +96,8 @@ void UpdateNetworkMetrics(
> 
> void UpdateNetworkMetrics(
>     FecControllerRplrBased* controller,
>-    const rtc::Optional<int>& uplink_bandwidth_bps,
>-    const rtc::Optional<float>& uplink_recoveralbe_packet_loss) {
>+    const absl::optional<int>& uplink_bandwidth_bps,
>+    const absl::optional<float>& uplink_recoveralbe_packet_loss) {
>   // FecControllerRplrBased doesn't currently use the PLR (general packet-loss
>   // rate) at all. (This might be changed in the future.) The unit-tests will
>   // use a random value (including unknown), to show this does not interfere.
>@@ -148,7 +148,7 @@ TEST(FecControllerRplrBasedTest, OutputInitValueWhenUplinkBandwidthUnknown) {
>           kEnablingRecoverablePacketLossAtHighBw,
>           kEnablingRecoverablePacketLossAtHighBw + kEpsilon}) {
>       auto controller = CreateFecControllerRplrBased(initial_fec_enabled);
>-      UpdateNetworkMetrics(controller.get(), rtc::nullopt,
>+      UpdateNetworkMetrics(controller.get(), absl::nullopt,
>                            recoverable_packet_loss);
>       CheckDecision(controller.get(), initial_fec_enabled,
>                     recoverable_packet_loss);
>@@ -165,7 +165,7 @@ TEST(FecControllerRplrBasedTest,
>                           kDisablingBandwidthLow + 1, kEnablingBandwidthLow - 1,
>                           kEnablingBandwidthLow, kEnablingBandwidthLow + 1}) {
>       auto controller = CreateFecControllerRplrBased(initial_fec_enabled);
>-      UpdateNetworkMetrics(controller.get(), bandwidth, rtc::nullopt);
>+      UpdateNetworkMetrics(controller.get(), bandwidth, absl::nullopt);
>       CheckDecision(controller.get(), initial_fec_enabled, 0.0);
>     }
>   }
>@@ -209,11 +209,11 @@ TEST(FecControllerRplrBasedTest, EnableFecForMediumBandwidth) {
>   auto controller = CreateFecControllerRplrBased(false);
>   constexpr float kRecoverablePacketLoss =
>       (kEnablingRecoverablePacketLossAtLowBw +
>-       kEnablingRecoverablePacketLossAtHighBw) / 2.0;
>-  UpdateNetworkMetrics(
>-      controller.get(),
>-      (kEnablingBandwidthHigh + kEnablingBandwidthLow) / 2,
>-      kRecoverablePacketLoss);
>+       kEnablingRecoverablePacketLossAtHighBw) /
>+      2.0;
>+  UpdateNetworkMetrics(controller.get(),
>+                       (kEnablingBandwidthHigh + kEnablingBandwidthLow) / 2,
>+                       kRecoverablePacketLoss);
>   CheckDecision(controller.get(), true, kRecoverablePacketLoss);
> }
> 
>@@ -274,11 +274,12 @@ TEST(FecControllerRplrBasedTest, DisableFecOnMediumBandwidth) {
>   auto controller = CreateFecControllerRplrBased(true);
>   constexpr float kRecoverablePacketLoss =
>       ((kDisablingRecoverablePacketLossAtLowBw +
>-        kDisablingRecoverablePacketLossAtHighBw) / 2.0f) - kEpsilon;
>-  UpdateNetworkMetrics(
>-      controller.get(),
>-      (kDisablingBandwidthHigh + kDisablingBandwidthLow) / 2,
>-      kRecoverablePacketLoss);
>+        kDisablingRecoverablePacketLossAtHighBw) /
>+       2.0f) -
>+      kEpsilon;
>+  UpdateNetworkMetrics(controller.get(),
>+                       (kDisablingBandwidthHigh + kDisablingBandwidthLow) / 2,
>+                       kRecoverablePacketLoss);
>   CheckDecision(controller.get(), false, kRecoverablePacketLoss);
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller.cc
>index 6c3cae07bc9..40e97cb371e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller.cc
>@@ -25,7 +25,7 @@ int OverheadRateBps(size_t overhead_bytes_per_packet, int frame_length_ms) {
>   return static_cast<int>(overhead_bytes_per_packet * 8 * 1000 /
>                           frame_length_ms);
> }
>-}
>+}  // namespace
> 
> FrameLengthController::Config::Config(
>     const std::vector<int>& encoder_frame_lengths_ms,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller.h
>index c254b3dcf66..f084fd0382f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller.h
>@@ -73,11 +73,11 @@ class FrameLengthController final : public Controller {
> 
>   std::vector<int>::const_iterator frame_length_ms_;
> 
>-  rtc::Optional<int> uplink_bandwidth_bps_;
>+  absl::optional<int> uplink_bandwidth_bps_;
> 
>-  rtc::Optional<float> uplink_packet_loss_fraction_;
>+  absl::optional<float> uplink_packet_loss_fraction_;
> 
>-  rtc::Optional<size_t> overhead_bytes_per_packet_;
>+  absl::optional<size_t> overhead_bytes_per_packet_;
> 
>   // True if the previous frame length decision was an increase, otherwise
>   // false.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller_unittest.cc
>index 1f984470c0d..f97ad4f96ca 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller_unittest.cc
>@@ -78,9 +78,9 @@ CreateChangeCriteriaFor20ms60msAnd120ms() {
> 
> void UpdateNetworkMetrics(
>     FrameLengthController* controller,
>-    const rtc::Optional<int>& uplink_bandwidth_bps,
>-    const rtc::Optional<float>& uplink_packet_loss_fraction,
>-    const rtc::Optional<size_t>& overhead_bytes_per_packet) {
>+    const absl::optional<int>& uplink_bandwidth_bps,
>+    const absl::optional<float>& uplink_packet_loss_fraction,
>+    const absl::optional<size_t>& overhead_bytes_per_packet) {
>   // UpdateNetworkMetrics can accept multiple network metric updates at once.
>   // However, currently, the most used case is to update one metric at a time.
>   // To reflect this fact, we separate the calls.
>@@ -114,14 +114,14 @@ TEST(FrameLengthControllerTest, DecreaseTo20MsOnHighUplinkBandwidth) {
>   auto controller =
>       CreateController(CreateChangeCriteriaFor20msAnd60ms(), {20, 60}, 60);
>   UpdateNetworkMetrics(controller.get(), kFl60msTo20msBandwidthBps,
>-                       rtc::nullopt, kOverheadBytesPerPacket);
>+                       absl::nullopt, kOverheadBytesPerPacket);
>   CheckDecision(controller.get(), 20);
> }
> 
> TEST(FrameLengthControllerTest, DecreaseTo20MsOnHighUplinkPacketLossFraction) {
>   auto controller =
>       CreateController(CreateChangeCriteriaFor20msAnd60ms(), {20, 60}, 60);
>-  UpdateNetworkMetrics(controller.get(), rtc::nullopt,
>+  UpdateNetworkMetrics(controller.get(), absl::nullopt,
>                        kFlDecreasingPacketLossFraction,
>                        kOverheadBytesPerPacket);
>   CheckDecision(controller.get(), 20);
>@@ -252,16 +252,12 @@ TEST(FrameLengthControllerTest, From120MsTo20MsOnHighUplinkBandwidth) {
>   auto controller = CreateController(CreateChangeCriteriaFor20ms60msAnd120ms(),
>                                      {20, 60, 120}, 120);
>   // It takes two steps for frame length to go from 120ms to 20ms.
>-  UpdateNetworkMetrics(controller.get(),
>-                       kFl60msTo20msBandwidthBps,
>-                       rtc::nullopt,
>-                       kOverheadBytesPerPacket);
>+  UpdateNetworkMetrics(controller.get(), kFl60msTo20msBandwidthBps,
>+                       absl::nullopt, kOverheadBytesPerPacket);
>   CheckDecision(controller.get(), 60);
> 
>-  UpdateNetworkMetrics(controller.get(),
>-                       kFl60msTo20msBandwidthBps,
>-                       rtc::nullopt,
>-                       kOverheadBytesPerPacket);
>+  UpdateNetworkMetrics(controller.get(), kFl60msTo20msBandwidthBps,
>+                       absl::nullopt, kOverheadBytesPerPacket);
>   CheckDecision(controller.get(), 20);
> }
> 
>@@ -269,12 +265,12 @@ TEST(FrameLengthControllerTest, From120MsTo20MsOnHighUplinkPacketLossFraction) {
>   auto controller = CreateController(CreateChangeCriteriaFor20ms60msAnd120ms(),
>                                      {20, 60, 120}, 120);
>   // It takes two steps for frame length to go from 120ms to 20ms.
>-  UpdateNetworkMetrics(controller.get(), rtc::nullopt,
>+  UpdateNetworkMetrics(controller.get(), absl::nullopt,
>                        kFlDecreasingPacketLossFraction,
>                        kOverheadBytesPerPacket);
>   CheckDecision(controller.get(), 60);
> 
>-  UpdateNetworkMetrics(controller.get(), rtc::nullopt,
>+  UpdateNetworkMetrics(controller.get(), absl::nullopt,
>                        kFlDecreasingPacketLossFraction,
>                        kOverheadBytesPerPacket);
>   CheckDecision(controller.get(), 20);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h
>index 7687446cacc..c11279ea90b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h
>@@ -11,8 +11,8 @@
> #ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_H_
> #define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_H_
> 
>+#include "absl/types/optional.h"
> #include "api/audio_codecs/audio_encoder.h"
>-#include "api/optional.h"
> #include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h
>index 874fc97eb24..257a79af18d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor_config.h
>@@ -11,7 +11,7 @@
> #ifndef MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_CONFIG_H_
> #define MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_CONFIG_H_
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> 
> namespace webrtc {
> 
>@@ -21,18 +21,18 @@ struct AudioEncoderRuntimeConfig {
>   ~AudioEncoderRuntimeConfig();
>   AudioEncoderRuntimeConfig& operator=(const AudioEncoderRuntimeConfig& other);
>   bool operator==(const AudioEncoderRuntimeConfig& other) const;
>-  rtc::Optional<int> bitrate_bps;
>-  rtc::Optional<int> frame_length_ms;
>+  absl::optional<int> bitrate_bps;
>+  absl::optional<int> frame_length_ms;
>   // Note: This is what we tell the encoder. It doesn't have to reflect
>   // the actual NetworkMetrics; it's subject to our decision.
>-  rtc::Optional<float> uplink_packet_loss_fraction;
>-  rtc::Optional<bool> enable_fec;
>-  rtc::Optional<bool> enable_dtx;
>+  absl::optional<float> uplink_packet_loss_fraction;
>+  absl::optional<bool> enable_fec;
>+  absl::optional<bool> enable_dtx;
> 
>   // Some encoders can encode fewer channels than the actual input to make
>   // better use of the bandwidth. |num_channels| sets the number of channels
>   // to encode.
>-  rtc::Optional<size_t> num_channels;
>+  absl::optional<size_t> num_channels;
> 
>   // This is true if the last frame length change was an increase, and otherwise
>   // false.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/parse_ana_dump.py b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/parse_ana_dump.py
>index a52b065f12c..1bd5f0c288a 100755
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/parse_ana_dump.py
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/audio_network_adaptor/parse_ana_dump.py
>@@ -66,9 +66,9 @@ def ParseAnaDump(dump_file_to_parse):
>     first_time_stamp = None
>     while True:
>       event = GetNextMessageFromFile(file_to_parse)
>-      if event == None:
>+      if event is None:
>         break
>-      if first_time_stamp == None:
>+      if first_time_stamp is None:
>         first_time_stamp = event.timestamp
>       if event.type == debug_dump_pb2.Event.ENCODER_RUNTIME_CONFIG:
>         for decision in event.encoder_runtime_config.DESCRIPTOR.fields:
>@@ -110,7 +110,7 @@ def main():
>       action='append')
> 
>   options = parser.parse_args()[0]
>-  if options.dump_file_to_parse == None:
>+  if options.dump_file_to_parse is None:
>     print "No dump file to parse is set.\n"
>     parser.print_help()
>     exit()
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/audio_format_conversion.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/audio_format_conversion.cc
>index a99a28cf2b6..e38aa333350 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/audio_format_conversion.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/audio_format_conversion.cc
>@@ -12,8 +12,8 @@
> 
> #include <string.h>
> 
>+#include "absl/types/optional.h"
> #include "api/array_view.h"
>-#include "api/optional.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/numerics/safe_conversions.h"
> #include "rtc_base/sanitizer.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h
>deleted file mode 100644
>index 7494ac0e6d7..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h
>+++ /dev/null
>@@ -1,20 +0,0 @@
>-/*
>- *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-// This file is for backwards compatibility only! Use
>-// webrtc/api/audio_codecs/builtin_audio_decoder_factory.h instead!
>-// TODO(kwiberg): Remove it.
>-
>-#ifndef MODULES_AUDIO_CODING_CODECS_BUILTIN_AUDIO_DECODER_FACTORY_H_
>-#define MODULES_AUDIO_CODING_CODECS_BUILTIN_AUDIO_DECODER_FACTORY_H_
>-
>-#include "api/audio_codecs/builtin_audio_decoder_factory.h"
>-
>-#endif  // MODULES_AUDIO_CODING_CODECS_BUILTIN_AUDIO_DECODER_FACTORY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc
>index 27b89b6b34c..9b36dfdd14c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory_unittest.cc
>@@ -19,7 +19,8 @@ TEST(AudioDecoderFactoryTest, CreateUnknownDecoder) {
>   rtc::scoped_refptr<AudioDecoderFactory> adf =
>       CreateBuiltinAudioDecoderFactory();
>   ASSERT_TRUE(adf);
>-  EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("rey", 8000, 1)));
>+  EXPECT_FALSE(
>+      adf->MakeAudioDecoder(SdpAudioFormat("rey", 8000, 1), absl::nullopt));
> }
> 
> TEST(AudioDecoderFactoryTest, CreatePcmu) {
>@@ -27,11 +28,16 @@ TEST(AudioDecoderFactoryTest, CreatePcmu) {
>       CreateBuiltinAudioDecoderFactory();
>   ASSERT_TRUE(adf);
>   // PCMu supports 8 kHz, and any number of channels.
>-  EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("pcmu", 8000, 0)));
>-  EXPECT_TRUE(adf->MakeAudioDecoder(SdpAudioFormat("pcmu", 8000, 1)));
>-  EXPECT_TRUE(adf->MakeAudioDecoder(SdpAudioFormat("pcmu", 8000, 2)));
>-  EXPECT_TRUE(adf->MakeAudioDecoder(SdpAudioFormat("pcmu", 8000, 3)));
>-  EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("pcmu", 16000, 1)));
>+  EXPECT_FALSE(
>+      adf->MakeAudioDecoder(SdpAudioFormat("pcmu", 8000, 0), absl::nullopt));
>+  EXPECT_TRUE(
>+      adf->MakeAudioDecoder(SdpAudioFormat("pcmu", 8000, 1), absl::nullopt));
>+  EXPECT_TRUE(
>+      adf->MakeAudioDecoder(SdpAudioFormat("pcmu", 8000, 2), absl::nullopt));
>+  EXPECT_TRUE(
>+      adf->MakeAudioDecoder(SdpAudioFormat("pcmu", 8000, 3), absl::nullopt));
>+  EXPECT_FALSE(
>+      adf->MakeAudioDecoder(SdpAudioFormat("pcmu", 16000, 1), absl::nullopt));
> }
> 
> TEST(AudioDecoderFactoryTest, CreatePcma) {
>@@ -39,11 +45,16 @@ TEST(AudioDecoderFactoryTest, CreatePcma) {
>       CreateBuiltinAudioDecoderFactory();
>   ASSERT_TRUE(adf);
>   // PCMa supports 8 kHz, and any number of channels.
>-  EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("pcma", 8000, 0)));
>-  EXPECT_TRUE(adf->MakeAudioDecoder(SdpAudioFormat("pcma", 8000, 1)));
>-  EXPECT_TRUE(adf->MakeAudioDecoder(SdpAudioFormat("pcma", 8000, 2)));
>-  EXPECT_TRUE(adf->MakeAudioDecoder(SdpAudioFormat("pcma", 8000, 3)));
>-  EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("pcma", 16000, 1)));
>+  EXPECT_FALSE(
>+      adf->MakeAudioDecoder(SdpAudioFormat("pcma", 8000, 0), absl::nullopt));
>+  EXPECT_TRUE(
>+      adf->MakeAudioDecoder(SdpAudioFormat("pcma", 8000, 1), absl::nullopt));
>+  EXPECT_TRUE(
>+      adf->MakeAudioDecoder(SdpAudioFormat("pcma", 8000, 2), absl::nullopt));
>+  EXPECT_TRUE(
>+      adf->MakeAudioDecoder(SdpAudioFormat("pcma", 8000, 3), absl::nullopt));
>+  EXPECT_FALSE(
>+      adf->MakeAudioDecoder(SdpAudioFormat("pcma", 16000, 1), absl::nullopt));
> }
> 
> TEST(AudioDecoderFactoryTest, CreateIlbc) {
>@@ -51,12 +62,16 @@ TEST(AudioDecoderFactoryTest, CreateIlbc) {
>       CreateBuiltinAudioDecoderFactory();
>   ASSERT_TRUE(adf);
>   // iLBC supports 8 kHz, 1 channel.
>-  EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("ilbc", 8000, 0)));
>+  EXPECT_FALSE(
>+      adf->MakeAudioDecoder(SdpAudioFormat("ilbc", 8000, 0), absl::nullopt));
> #ifdef WEBRTC_CODEC_ILBC
>-  EXPECT_TRUE(adf->MakeAudioDecoder(SdpAudioFormat("ilbc", 8000, 1)));
>+  EXPECT_TRUE(
>+      adf->MakeAudioDecoder(SdpAudioFormat("ilbc", 8000, 1), absl::nullopt));
> #endif
>-  EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("ilbc", 8000, 2)));
>-  EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("ilbc", 16000, 1)));
>+  EXPECT_FALSE(
>+      adf->MakeAudioDecoder(SdpAudioFormat("ilbc", 8000, 2), absl::nullopt));
>+  EXPECT_FALSE(
>+      adf->MakeAudioDecoder(SdpAudioFormat("ilbc", 16000, 1), absl::nullopt));
> }
> 
> TEST(AudioDecoderFactoryTest, CreateIsac) {
>@@ -65,15 +80,22 @@ TEST(AudioDecoderFactoryTest, CreateIsac) {
>   ASSERT_TRUE(adf);
>   // iSAC supports 16 kHz, 1 channel. The float implementation additionally
>   // supports 32 kHz, 1 channel.
>-  EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("isac", 16000, 0)));
>-  EXPECT_TRUE(adf->MakeAudioDecoder(SdpAudioFormat("isac", 16000, 1)));
>-  EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("isac", 16000, 2)));
>-  EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("isac", 8000, 1)));
>-  EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("isac", 48000, 1)));
>+  EXPECT_FALSE(
>+      adf->MakeAudioDecoder(SdpAudioFormat("isac", 16000, 0), absl::nullopt));
>+  EXPECT_TRUE(
>+      adf->MakeAudioDecoder(SdpAudioFormat("isac", 16000, 1), absl::nullopt));
>+  EXPECT_FALSE(
>+      adf->MakeAudioDecoder(SdpAudioFormat("isac", 16000, 2), absl::nullopt));
>+  EXPECT_FALSE(
>+      adf->MakeAudioDecoder(SdpAudioFormat("isac", 8000, 1), absl::nullopt));
>+  EXPECT_FALSE(
>+      adf->MakeAudioDecoder(SdpAudioFormat("isac", 48000, 1), absl::nullopt));
> #ifdef WEBRTC_ARCH_ARM
>-  EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("isac", 32000, 1)));
>+  EXPECT_FALSE(
>+      adf->MakeAudioDecoder(SdpAudioFormat("isac", 32000, 1), absl::nullopt));
> #else
>-  EXPECT_TRUE(adf->MakeAudioDecoder(SdpAudioFormat("isac", 32000, 1)));
>+  EXPECT_TRUE(
>+      adf->MakeAudioDecoder(SdpAudioFormat("isac", 32000, 1), absl::nullopt));
> #endif
> }
> 
>@@ -85,10 +107,11 @@ TEST(AudioDecoderFactoryTest, CreateL16) {
>   const int clockrates[] = {8000, 16000, 32000, 48000};
>   const int num_channels[] = {1, 2, 3, 4711};
>   for (int clockrate : clockrates) {
>-    EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("l16", clockrate, 0)));
>+    EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("l16", clockrate, 0),
>+                                       absl::nullopt));
>     for (int channels : num_channels) {
>-      EXPECT_TRUE(
>-          adf->MakeAudioDecoder(SdpAudioFormat("l16", clockrate, channels)));
>+      EXPECT_TRUE(adf->MakeAudioDecoder(
>+          SdpAudioFormat("l16", clockrate, channels), absl::nullopt));
>     }
>   }
> }
>@@ -98,16 +121,22 @@ TEST(AudioDecoderFactoryTest, CreateG722) {
>       CreateBuiltinAudioDecoderFactory();
>   ASSERT_TRUE(adf);
>   // g722 supports 8 kHz, 1-2 channels.
>-  EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("g722", 8000, 0)));
>-  EXPECT_TRUE(adf->MakeAudioDecoder(SdpAudioFormat("g722", 8000, 1)));
>-  EXPECT_TRUE(adf->MakeAudioDecoder(SdpAudioFormat("g722", 8000, 2)));
>-  EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("g722", 8000, 3)));
>-  EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("g722", 16000, 1)));
>-  EXPECT_FALSE(adf->MakeAudioDecoder(SdpAudioFormat("g722", 32000, 1)));
>+  EXPECT_FALSE(
>+      adf->MakeAudioDecoder(SdpAudioFormat("g722", 8000, 0), absl::nullopt));
>+  EXPECT_TRUE(
>+      adf->MakeAudioDecoder(SdpAudioFormat("g722", 8000, 1), absl::nullopt));
>+  EXPECT_TRUE(
>+      adf->MakeAudioDecoder(SdpAudioFormat("g722", 8000, 2), absl::nullopt));
>+  EXPECT_FALSE(
>+      adf->MakeAudioDecoder(SdpAudioFormat("g722", 8000, 3), absl::nullopt));
>+  EXPECT_FALSE(
>+      adf->MakeAudioDecoder(SdpAudioFormat("g722", 16000, 1), absl::nullopt));
>+  EXPECT_FALSE(
>+      adf->MakeAudioDecoder(SdpAudioFormat("g722", 32000, 1), absl::nullopt));
> 
>   // g722 actually uses a 16 kHz sample rate instead of the nominal 8 kHz.
>   std::unique_ptr<AudioDecoder> dec =
>-      adf->MakeAudioDecoder(SdpAudioFormat("g722", 8000, 1));
>+      adf->MakeAudioDecoder(SdpAudioFormat("g722", 8000, 1), absl::nullopt);
>   EXPECT_EQ(16000, dec->SampleRateHz());
> }
> 
>@@ -126,8 +155,10 @@ TEST(AudioDecoderFactoryTest, CreateOpus) {
>         }
>         const bool good = (hz == 48000 && channels == 2 &&
>                            (stereo == "XX" || stereo == "0" || stereo == "1"));
>-        EXPECT_EQ(good, static_cast<bool>(adf->MakeAudioDecoder(SdpAudioFormat(
>-                            "opus", hz, channels, std::move(params)))));
>+        EXPECT_EQ(good,
>+                  static_cast<bool>(adf->MakeAudioDecoder(
>+                      SdpAudioFormat("opus", hz, channels, std::move(params)),
>+                      absl::nullopt)));
>       }
>     }
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/builtin_audio_encoder_factory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/builtin_audio_encoder_factory.h
>deleted file mode 100644
>index 6ec765ab977..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/builtin_audio_encoder_factory.h
>+++ /dev/null
>@@ -1,20 +0,0 @@
>-/*
>- *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-// This file is for backwards compatibility only! Use
>-// webrtc/api/audio_codecs/builtin_audio_decoder_factory.h instead!
>-// TODO(ossu): Remove it.
>-
>-#ifndef MODULES_AUDIO_CODING_CODECS_BUILTIN_AUDIO_ENCODER_FACTORY_H_
>-#define MODULES_AUDIO_CODING_CODECS_BUILTIN_AUDIO_ENCODER_FACTORY_H_
>-
>-#include "api/audio_codecs/builtin_audio_encoder_factory.h"
>-
>-#endif  // MODULES_AUDIO_CODING_CODECS_BUILTIN_AUDIO_ENCODER_FACTORY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/builtin_audio_encoder_factory_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/builtin_audio_encoder_factory_unittest.cc
>index 984bfd24f69..64f0159ea7d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/builtin_audio_encoder_factory_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/builtin_audio_encoder_factory_unittest.cc
>@@ -43,7 +43,7 @@ TEST_P(AudioEncoderFactoryTest, CanConstructAllSupportedEncoders) {
>   auto supported_encoders = factory->GetSupportedEncoders();
>   for (const auto& spec : supported_encoders) {
>     auto info = factory->QueryAudioEncoder(spec.format);
>-    auto encoder = factory->MakeAudioEncoder(127, spec.format);
>+    auto encoder = factory->MakeAudioEncoder(127, spec.format, absl::nullopt);
>     EXPECT_TRUE(encoder);
>     EXPECT_EQ(encoder->SampleRateHz(), info->sample_rate_hz);
>     EXPECT_EQ(encoder->NumChannels(), info->num_channels);
>@@ -56,7 +56,8 @@ TEST_P(AudioEncoderFactoryTest, CanRunAllSupportedEncoders) {
>   auto factory = GetParam();
>   auto supported_encoders = factory->GetSupportedEncoders();
>   for (const auto& spec : supported_encoders) {
>-    auto encoder = factory->MakeAudioEncoder(kTestPayloadType, spec.format);
>+    auto encoder =
>+        factory->MakeAudioEncoder(kTestPayloadType, spec.format, absl::nullopt);
>     EXPECT_TRUE(encoder);
>     encoder->Reset();
>     const int num_samples = rtc::checked_cast<int>(
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
>index 5a043ecbccb..055190497a5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
>@@ -11,8 +11,8 @@
> #include "modules/audio_coding/codecs/cng/audio_encoder_cng.h"
> 
> #include <algorithm>
>-#include <memory>
> #include <limits>
>+#include <memory>
> #include <utility>
> 
> namespace webrtc {
>@@ -44,19 +44,19 @@ bool AudioEncoderCng::Config::IsOk() const {
> }
> 
> AudioEncoderCng::AudioEncoderCng(Config&& config)
>-    : speech_encoder_(
>-          ([&] { RTC_CHECK(config.IsOk()) << "Invalid configuration."; }(),
>-           std::move(config.speech_encoder))),
>+    : speech_encoder_((static_cast<void>([&] {
>+                         RTC_CHECK(config.IsOk()) << "Invalid configuration.";
>+                       }()),
>+                       std::move(config.speech_encoder))),
>       cng_payload_type_(config.payload_type),
>       num_cng_coefficients_(config.num_cng_coefficients),
>       sid_frame_interval_ms_(config.sid_frame_interval_ms),
>       last_frame_active_(true),
>       vad_(config.vad ? std::unique_ptr<Vad>(config.vad)
>-           : CreateVad(config.vad_mode)),
>+                      : CreateVad(config.vad_mode)),
>       cng_encoder_(new ComfortNoiseEncoder(SampleRateHz(),
>                                            sid_frame_interval_ms_,
>-                                           num_cng_coefficients_)) {
>-}
>+                                           num_cng_coefficients_)) {}
> 
> AudioEncoderCng::~AudioEncoderCng() = default;
> 
>@@ -158,9 +158,8 @@ void AudioEncoderCng::Reset() {
>   rtp_timestamps_.clear();
>   last_frame_active_ = true;
>   vad_->Reset();
>-  cng_encoder_.reset(
>-      new ComfortNoiseEncoder(SampleRateHz(), sid_frame_interval_ms_,
>-                              num_cng_coefficients_));
>+  cng_encoder_.reset(new ComfortNoiseEncoder(
>+      SampleRateHz(), sid_frame_interval_ms_, num_cng_coefficients_));
> }
> 
> bool AudioEncoderCng::SetFec(bool enable) {
>@@ -198,7 +197,7 @@ void AudioEncoderCng::OnReceivedUplinkRecoverablePacketLossFraction(
> 
> void AudioEncoderCng::OnReceivedUplinkBandwidth(
>     int target_audio_bitrate_bps,
>-    rtc::Optional<int64_t> bwe_period_ms) {
>+    absl::optional<int64_t> bwe_period_ms) {
>   speech_encoder_->OnReceivedUplinkBandwidth(target_audio_bitrate_bps,
>                                              bwe_period_ms);
> }
>@@ -217,11 +216,10 @@ AudioEncoder::EncodedInfo AudioEncoderCng::EncodePassive(
>     // that value, in which case we don't want to overwrite any value from
>     // an earlier iteration.
>     size_t encoded_bytes_tmp =
>-        cng_encoder_->Encode(
>-            rtc::ArrayView<const int16_t>(
>-                &speech_buffer_[i * samples_per_10ms_frame],
>-                samples_per_10ms_frame),
>-            force_sid, encoded);
>+        cng_encoder_->Encode(rtc::ArrayView<const int16_t>(
>+                                 &speech_buffer_[i * samples_per_10ms_frame],
>+                                 samples_per_10ms_frame),
>+                             force_sid, encoded);
> 
>     if (encoded_bytes_tmp > 0) {
>       RTC_CHECK(!output_produced);
>@@ -238,9 +236,8 @@ AudioEncoder::EncodedInfo AudioEncoderCng::EncodePassive(
>   return info;
> }
> 
>-AudioEncoder::EncodedInfo AudioEncoderCng::EncodeActive(
>-    size_t frames_to_encode,
>-    rtc::Buffer* encoded) {
>+AudioEncoder::EncodedInfo AudioEncoderCng::EncodeActive(size_t frames_to_encode,
>+                                                        rtc::Buffer* encoded) {
>   const size_t samples_per_10ms_frame = SamplesPer10msFrame();
>   AudioEncoder::EncodedInfo info;
>   for (size_t i = 0; i < frames_to_encode; ++i) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h
>index 4491289f284..e4c65077651 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h
>@@ -69,7 +69,7 @@ class AudioEncoderCng final : public AudioEncoder {
>       float uplink_recoverable_packet_loss_fraction) override;
>   void OnReceivedUplinkBandwidth(
>       int target_audio_bitrate_bps,
>-      rtc::Optional<int64_t> bwe_period_ms) override;
>+      absl::optional<int64_t> bwe_period_ms) override;
> 
>  private:
>   EncodedInfo EncodePassive(size_t frames_to_encode,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
>index 6e246c9a55e..a76dcbda298 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
>@@ -30,7 +30,7 @@ namespace {
> static const size_t kMaxNumSamples = 48 * 10 * 2;  // 10 ms @ 48 kHz stereo.
> static const size_t kMockReturnEncodedBytes = 17;
> static const int kCngPayloadType = 18;
>-}
>+}  // namespace
> 
> class AudioEncoderCngTest : public ::testing::Test {
>  protected:
>@@ -43,7 +43,6 @@ class AudioEncoderCngTest : public ::testing::Test {
>         sample_rate_hz_(8000) {
>     memset(audio_, 0, kMaxNumSamples * 2);
>     EXPECT_CALL(*mock_encoder_, NumChannels()).WillRepeatedly(Return(1));
>-    EXPECT_CALL(*mock_encoder_, Die()).Times(1);
>   }
> 
>   void TearDown() override {
>@@ -95,8 +94,7 @@ class AudioEncoderCngTest : public ::testing::Test {
>     InSequence s;
>     AudioEncoder::EncodedInfo info;
>     for (size_t j = 0; j < num_calls - 1; ++j) {
>-      EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
>-          .WillOnce(Return(info));
>+      EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _)).WillOnce(Return(info));
>     }
>     info.encoded_bytes = kMockReturnEncodedBytes;
>     EXPECT_CALL(*mock_encoder_, EncodeImpl(_, _, _))
>@@ -156,12 +154,14 @@ class AudioEncoderCngTest : public ::testing::Test {
>     EXPECT_CALL(
>         *mock_vad_,
>         VoiceActivity(_, expected_first_block_size_ms * sample_rate_hz_ / 1000,
>-                      sample_rate_hz_)).WillOnce(Return(Vad::kPassive));
>+                      sample_rate_hz_))
>+        .WillOnce(Return(Vad::kPassive));
>     if (expected_second_block_size_ms > 0) {
>       EXPECT_CALL(*mock_vad_,
>                   VoiceActivity(
>                       _, expected_second_block_size_ms * sample_rate_hz_ / 1000,
>-                      sample_rate_hz_)).WillOnce(Return(Vad::kPassive));
>+                      sample_rate_hz_))
>+          .WillOnce(Return(Vad::kPassive));
>     }
> 
>     // With this call to Encode(), |mock_vad_| should be called according to the
>@@ -221,8 +221,8 @@ TEST_F(AudioEncoderCngTest, CheckFrameSizePropagation) {
> TEST_F(AudioEncoderCngTest, CheckTargetAudioBitratePropagation) {
>   CreateCng(MakeCngConfig());
>   EXPECT_CALL(*mock_encoder_,
>-              OnReceivedUplinkBandwidth(4711, rtc::Optional<int64_t>()));
>-  cng_->OnReceivedUplinkBandwidth(4711, rtc::nullopt);
>+              OnReceivedUplinkBandwidth(4711, absl::optional<int64_t>()));
>+  cng_->OnReceivedUplinkBandwidth(4711, absl::nullopt);
> }
> 
> TEST_F(AudioEncoderCngTest, CheckPacketLossFractionPropagation) {
>@@ -430,9 +430,7 @@ class AudioEncoderCngDeathTest : public AudioEncoderCngTest {
>   // Override AudioEncoderCngTest::TearDown, since that one expects a call to
>   // the destructor of |mock_vad_|. In this case, that object is already
>   // deleted.
>-  void TearDown() override {
>-    cng_.reset();
>-  }
>+  void TearDown() override { cng_.reset(); }
> 
>   AudioEncoderCng::Config MakeCngConfig() {
>     // Don't provide a Vad mock object, since it would leak when the test dies.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/cng_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/cng_unittest.cc
>index 54e5189d30a..81688b1f472 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/cng_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/cng_unittest.cc
>@@ -29,10 +29,7 @@ enum : size_t {
>   kCNGNumParamsTooHigh = WEBRTC_CNG_MAX_LPC_ORDER + 1
> };
> 
>-enum {
>-  kNoSid,
>-  kForceSid
>-};
>+enum { kNoSid, kForceSid };
> 
> class CngTest : public ::testing::Test {
>  protected:
>@@ -46,11 +43,11 @@ class CngTest : public ::testing::Test {
> void CngTest::SetUp() {
>   FILE* input_file;
>   const std::string file_name =
>-        webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
>+      webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
>   input_file = fopen(file_name.c_str(), "rb");
>   ASSERT_TRUE(input_file != NULL);
>-  ASSERT_EQ(640, static_cast<int32_t>(fread(speech_data_, sizeof(int16_t),
>-                                             640, input_file)));
>+  ASSERT_EQ(640, static_cast<int32_t>(
>+                     fread(speech_data_, sizeof(int16_t), 640, input_file)));
>   fclose(input_file);
>   input_file = NULL;
> }
>@@ -74,11 +71,18 @@ void CngTest::TestCngEncode(int sample_rate_hz, int quality) {
> // Create CNG encoder, init with faulty values, free CNG encoder.
> TEST_F(CngTest, CngInitFail) {
>   // Call with too few parameters.
>-  EXPECT_DEATH({ ComfortNoiseEncoder(8000, kSidNormalIntervalUpdate,
>-                                     kCNGNumParamsLow); }, "");
>+  EXPECT_DEATH(
>+      {
>+        ComfortNoiseEncoder(8000, kSidNormalIntervalUpdate, kCNGNumParamsLow);
>+      },
>+      "");
>   // Call with too many parameters.
>-  EXPECT_DEATH({ ComfortNoiseEncoder(8000, kSidNormalIntervalUpdate,
>-                                     kCNGNumParamsTooHigh); }, "");
>+  EXPECT_DEATH(
>+      {
>+        ComfortNoiseEncoder(8000, kSidNormalIntervalUpdate,
>+                            kCNGNumParamsTooHigh);
>+      },
>+      "");
> }
> 
> // Encode Cng with too long input vector.
>@@ -209,13 +213,15 @@ TEST_F(CngTest, CngAutoSid) {
> 
>   // Normal Encode, 100 msec, where no SID data should be generated.
>   for (int i = 0; i < 10; i++) {
>-    EXPECT_EQ(0U, cng_encoder.Encode(
>-        rtc::ArrayView<const int16_t>(speech_data_, 160), kNoSid, &sid_data));
>+    EXPECT_EQ(
>+        0U, cng_encoder.Encode(rtc::ArrayView<const int16_t>(speech_data_, 160),
>+                               kNoSid, &sid_data));
>   }
> 
>   // We have reached 100 msec, and SID data should be generated.
>-  EXPECT_EQ(kCNGNumParamsNormal + 1, cng_encoder.Encode(
>-      rtc::ArrayView<const int16_t>(speech_data_, 160), kNoSid, &sid_data));
>+  EXPECT_EQ(kCNGNumParamsNormal + 1,
>+            cng_encoder.Encode(rtc::ArrayView<const int16_t>(speech_data_, 160),
>+                               kNoSid, &sid_data));
> }
> 
> // Test automatic SID, with very short interval.
>@@ -228,13 +234,16 @@ TEST_F(CngTest, CngAutoSidShort) {
>   ComfortNoiseDecoder cng_decoder;
> 
>   // First call will never generate SID, unless forced to.
>-  EXPECT_EQ(0U, cng_encoder.Encode(
>-      rtc::ArrayView<const int16_t>(speech_data_, 160), kNoSid, &sid_data));
>+  EXPECT_EQ(0U,
>+            cng_encoder.Encode(rtc::ArrayView<const int16_t>(speech_data_, 160),
>+                               kNoSid, &sid_data));
> 
>   // Normal Encode, 100 msec, SID data should be generated all the time.
>   for (int i = 0; i < 10; i++) {
>-    EXPECT_EQ(kCNGNumParamsNormal + 1, cng_encoder.Encode(
>-        rtc::ArrayView<const int16_t>(speech_data_, 160), kNoSid, &sid_data));
>+    EXPECT_EQ(
>+        kCNGNumParamsNormal + 1,
>+        cng_encoder.Encode(rtc::ArrayView<const int16_t>(speech_data_, 160),
>+                           kNoSid, &sid_data));
>   }
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.cc
>index bd17a614bc4..a07b093b3c7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.cc
>@@ -25,28 +25,26 @@ const size_t kCngMaxOutsizeOrder = 640;
> void WebRtcCng_K2a16(int16_t* k, int useOrder, int16_t* a);
> 
> const int32_t WebRtcCng_kDbov[94] = {
>-  1081109975, 858756178, 682134279, 541838517, 430397633, 341876992,
>-  271562548,  215709799, 171344384, 136103682, 108110997, 85875618,
>-  68213428,   54183852,  43039763,  34187699,  27156255,  21570980,
>-  17134438,   13610368,  10811100,  8587562,   6821343,   5418385,
>-  4303976,    3418770,   2715625,   2157098,   1713444,   1361037,
>-  1081110,    858756,    682134,    541839,    430398,    341877,
>-  271563,     215710,    171344,    136104,    108111,    85876,
>-  68213,      54184,     43040,     34188,     27156,     21571,
>-  17134,      13610,     10811,     8588,      6821,      5418,
>-  4304,       3419,      2716,      2157,      1713,      1361,
>-  1081,       859,       682,       542,       430,       342,
>-  272,        216,       171,       136,       108,       86,
>-  68,         54,        43,        34,        27,        22,
>-  17,         14,        11,        9,         7,         5,
>-  4,          3,         3,         2,         2,         1,
>-  1,          1,         1,         1
>-};
>+    1081109975, 858756178, 682134279, 541838517, 430397633, 341876992,
>+    271562548,  215709799, 171344384, 136103682, 108110997, 85875618,
>+    68213428,   54183852,  43039763,  34187699,  27156255,  21570980,
>+    17134438,   13610368,  10811100,  8587562,   6821343,   5418385,
>+    4303976,    3418770,   2715625,   2157098,   1713444,   1361037,
>+    1081110,    858756,    682134,    541839,    430398,    341877,
>+    271563,     215710,    171344,    136104,    108111,    85876,
>+    68213,      54184,     43040,     34188,     27156,     21571,
>+    17134,      13610,     10811,     8588,      6821,      5418,
>+    4304,       3419,      2716,      2157,      1713,      1361,
>+    1081,       859,       682,       542,       430,       342,
>+    272,        216,       171,       136,       108,       86,
>+    68,         54,        43,        34,        27,        22,
>+    17,         14,        11,        9,         7,         5,
>+    4,          3,         3,         2,         2,         1,
>+    1,          1,         1,         1};
> 
> const int16_t WebRtcCng_kCorrWindow[WEBRTC_CNG_MAX_LPC_ORDER] = {
>-  32702, 32636, 32570, 32505, 32439, 32374,
>-  32309, 32244, 32179, 32114, 32049, 31985
>-};
>+    32702, 32636, 32570, 32505, 32439, 32374,
>+    32309, 32244, 32179, 32114, 32049, 31985};
> 
> }  // namespace
> 
>@@ -57,7 +55,7 @@ ComfortNoiseDecoder::ComfortNoiseDecoder() {
> }
> 
> void ComfortNoiseDecoder::Reset() {
>-  dec_seed_ = 7777;  /* For debugging only. */
>+  dec_seed_ = 7777; /* For debugging only. */
>   dec_target_energy_ = 0;
>   dec_used_energy_ = 0;
>   for (auto& c : dec_target_reflCoefs_)
>@@ -115,11 +113,11 @@ bool ComfortNoiseDecoder::Generate(rtc::ArrayView<int16_t> out_data,
>   int16_t excitation[kCngMaxOutsizeOrder];
>   int16_t low[kCngMaxOutsizeOrder];
>   int16_t lpPoly[WEBRTC_CNG_MAX_LPC_ORDER + 1];
>-  int16_t ReflBetaStd = 26214;  /* 0.8 in q15. */
>-  int16_t ReflBetaCompStd = 6553;  /* 0.2 in q15. */
>-  int16_t ReflBetaNewP = 19661;  /* 0.6 in q15. */
>-  int16_t ReflBetaCompNewP = 13107;  /* 0.4 in q15. */
>-  int16_t Beta, BetaC;  /* These are in Q15. */
>+  int16_t ReflBetaStd = 26214;      /* 0.8 in q15. */
>+  int16_t ReflBetaCompStd = 6553;   /* 0.2 in q15. */
>+  int16_t ReflBetaNewP = 19661;     /* 0.6 in q15. */
>+  int16_t ReflBetaCompNewP = 13107; /* 0.4 in q15. */
>+  int16_t Beta, BetaC;              /* These are in Q15. */
>   int32_t targetEnergy;
>   int16_t En;
>   int16_t temp16;
>@@ -139,30 +137,28 @@ bool ComfortNoiseDecoder::Generate(rtc::ArrayView<int16_t> out_data,
>   }
> 
>   /* Calculate new scale factor in Q13 */
>-  dec_used_scale_factor_ =
>-      rtc::checked_cast<int16_t>(
>-          WEBRTC_SPL_MUL_16_16_RSFT(dec_used_scale_factor_, Beta >> 2, 13) +
>-          WEBRTC_SPL_MUL_16_16_RSFT(dec_target_scale_factor_, BetaC >> 2, 13));
>+  dec_used_scale_factor_ = rtc::checked_cast<int16_t>(
>+      WEBRTC_SPL_MUL_16_16_RSFT(dec_used_scale_factor_, Beta >> 2, 13) +
>+      WEBRTC_SPL_MUL_16_16_RSFT(dec_target_scale_factor_, BetaC >> 2, 13));
> 
>-  dec_used_energy_  = dec_used_energy_ >> 1;
>+  dec_used_energy_ = dec_used_energy_ >> 1;
>   dec_used_energy_ += dec_target_energy_ >> 1;
> 
>   /* Do the same for the reflection coeffs, albeit in Q15. */
>   for (size_t i = 0; i < WEBRTC_CNG_MAX_LPC_ORDER; i++) {
>-    dec_used_reflCoefs_[i] = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(
>-        dec_used_reflCoefs_[i], Beta, 15);
>-    dec_used_reflCoefs_[i] += (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(
>-        dec_target_reflCoefs_[i], BetaC, 15);
>+    dec_used_reflCoefs_[i] =
>+        (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(dec_used_reflCoefs_[i], Beta, 15);
>+    dec_used_reflCoefs_[i] +=
>+        (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(dec_target_reflCoefs_[i], BetaC, 15);
>   }
> 
>   /* Compute the polynomial coefficients. */
>   WebRtcCng_K2a16(dec_used_reflCoefs_, WEBRTC_CNG_MAX_LPC_ORDER, lpPoly);
> 
>-
>   targetEnergy = dec_used_energy_;
> 
>   /* Calculate scaling factor based on filter energy. */
>-  En = 8192;  /* 1.0 in Q13. */
>+  En = 8192; /* 1.0 in Q13. */
>   for (size_t i = 0; i < (WEBRTC_CNG_MAX_LPC_ORDER); i++) {
>     /* Floating point value for reference.
>        E *= 1.0 - (dec_used_reflCoefs_[i] / 32768.0) *
>@@ -171,11 +167,11 @@ bool ComfortNoiseDecoder::Generate(rtc::ArrayView<int16_t> out_data,
> 
>     /* Same in fixed point. */
>     /* K(i).^2 in Q15. */
>-    temp16 = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(
>-        dec_used_reflCoefs_[i], dec_used_reflCoefs_[i], 15);
>+    temp16 = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(dec_used_reflCoefs_[i],
>+                                                dec_used_reflCoefs_[i], 15);
>     /* 1 - K(i).^2 in Q15. */
>     temp16 = 0x7fff - temp16;
>-    En = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(En, temp16, 15);
>+    En = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(En, temp16, 15);
>   }
> 
>   /* float scaling= sqrt(E * dec_target_energy_ / (1 << 24)); */
>@@ -183,8 +179,8 @@ bool ComfortNoiseDecoder::Generate(rtc::ArrayView<int16_t> out_data,
>   /* Calculate sqrt(En * target_energy / excitation energy) */
>   targetEnergy = WebRtcSpl_Sqrt(dec_used_energy_);
> 
>-  En = (int16_t) WebRtcSpl_Sqrt(En) << 6;
>-  En = (En * 3) >> 1;  /* 1.5 estimates sqrt(2). */
>+  En = (int16_t)WebRtcSpl_Sqrt(En) << 6;
>+  En = (En * 3) >> 1; /* 1.5 estimates sqrt(2). */
>   dec_used_scale_factor_ = (int16_t)((En * targetEnergy) >> 12);
> 
>   /* Generate excitation. */
>@@ -217,7 +213,7 @@ ComfortNoiseEncoder::ComfortNoiseEncoder(int fs, int interval, int quality)
>       enc_Energy_(0),
>       enc_reflCoefs_{0},
>       enc_corrVector_{0},
>-      enc_seed_(7777)  /* For debugging only. */ {
>+      enc_seed_(7777) /* For debugging only. */ {
>   RTC_CHECK_GT(quality, 0);
>   RTC_CHECK_LE(quality, WEBRTC_CNG_MAX_LPC_ORDER);
>   /* Needed to get the right function pointers in SPLIB. */
>@@ -236,7 +232,7 @@ void ComfortNoiseEncoder::Reset(int fs, int interval, int quality) {
>     c = 0;
>   for (auto& c : enc_corrVector_)
>     c = 0;
>-  enc_seed_ = 7777;  /* For debugging only. */
>+  enc_seed_ = 7777; /* For debugging only. */
> }
> 
> size_t ComfortNoiseEncoder::Encode(rtc::ArrayView<const int16_t> speech,
>@@ -312,20 +308,19 @@ size_t ComfortNoiseEncoder::Encode(rtc::ArrayView<const int16_t> speech,
>       if (negate)
>         *bptr = -*bptr;
> 
>-      blo = (int32_t) * aptr * (*bptr & 0xffff);
>-      bhi = ((blo >> 16) & 0xffff)
>-          + ((int32_t)(*aptr++) * ((*bptr >> 16) & 0xffff));
>+      blo = (int32_t)*aptr * (*bptr & 0xffff);
>+      bhi = ((blo >> 16) & 0xffff) +
>+            ((int32_t)(*aptr++) * ((*bptr >> 16) & 0xffff));
>       blo = (blo & 0xffff) | ((bhi & 0xffff) << 16);
> 
>-      *bptr = (((bhi >> 16) & 0x7fff) << 17) | ((uint32_t) blo >> 15);
>+      *bptr = (((bhi >> 16) & 0x7fff) << 17) | ((uint32_t)blo >> 15);
>       if (negate)
>         *bptr = -*bptr;
>       bptr++;
>     }
>     /* End of bandwidth expansion. */
> 
>-    stab = WebRtcSpl_LevinsonDurbin(corrVector, arCoefs, refCs,
>-                                    enc_nrOfCoefs_);
>+    stab = WebRtcSpl_LevinsonDurbin(corrVector, arCoefs, refCs, enc_nrOfCoefs_);
> 
>     if (!stab) {
>       /* Disregard from this frame */
>@@ -345,13 +340,12 @@ size_t ComfortNoiseEncoder::Encode(rtc::ArrayView<const int16_t> speech,
>   } else {
>     /* Average history with new values. */
>     for (i = 0; i < enc_nrOfCoefs_; i++) {
>-      enc_reflCoefs_[i] = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(
>-          enc_reflCoefs_[i], ReflBeta, 15);
>+      enc_reflCoefs_[i] =
>+          (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(enc_reflCoefs_[i], ReflBeta, 15);
>       enc_reflCoefs_[i] +=
>-          (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(refCs[i], ReflBetaComp, 15);
>+          (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(refCs[i], ReflBetaComp, 15);
>     }
>-    enc_Energy_ =
>-        (outEnergy >> 2) + (enc_Energy_ >> 1) + (enc_Energy_ >> 2);
>+    enc_Energy_ = (outEnergy >> 2) + (enc_Energy_ >> 1) + (enc_Energy_ >> 2);
>   }
> 
>   if (enc_Energy_ < 1) {
>@@ -372,25 +366,25 @@ size_t ComfortNoiseEncoder::Encode(rtc::ArrayView<const int16_t> speech,
>       index = 94;
> 
>     const size_t output_coefs = enc_nrOfCoefs_ + 1;
>-    output->AppendData(output_coefs, [&] (rtc::ArrayView<uint8_t> output) {
>-        output[0] = (uint8_t)index;
>-
>-        /* Quantize coefficients with tweak for WebRtc implementation of
>-         * RFC3389. */
>-        if (enc_nrOfCoefs_ == WEBRTC_CNG_MAX_LPC_ORDER) {
>-          for (i = 0; i < enc_nrOfCoefs_; i++) {
>-            /* Q15 to Q7 with rounding. */
>-            output[i + 1] = ((enc_reflCoefs_[i] + 128) >> 8);
>-          }
>-        } else {
>-          for (i = 0; i < enc_nrOfCoefs_; i++) {
>-            /* Q15 to Q7 with rounding. */
>-            output[i + 1] = (127 + ((enc_reflCoefs_[i] + 128) >> 8));
>-          }
>+    output->AppendData(output_coefs, [&](rtc::ArrayView<uint8_t> output) {
>+      output[0] = (uint8_t)index;
>+
>+      /* Quantize coefficients with tweak for WebRtc implementation of
>+       * RFC3389. */
>+      if (enc_nrOfCoefs_ == WEBRTC_CNG_MAX_LPC_ORDER) {
>+        for (i = 0; i < enc_nrOfCoefs_; i++) {
>+          /* Q15 to Q7 with rounding. */
>+          output[i + 1] = ((enc_reflCoefs_[i] + 128) >> 8);
>+        }
>+      } else {
>+        for (i = 0; i < enc_nrOfCoefs_; i++) {
>+          /* Q15 to Q7 with rounding. */
>+          output[i + 1] = (127 + ((enc_reflCoefs_[i] + 128) >> 8));
>         }
>+      }
> 
>-        return output_coefs;
>-      });
>+      return output_coefs;
>+    });
> 
>     enc_msSinceSid_ =
>         static_cast<int16_t>((1000 * num_samples) / enc_sampfreq_);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h
>index 5e21b8fd4f3..3f8fadcd71b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h
>@@ -8,7 +8,6 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-
> #ifndef MODULES_AUDIO_CODING_CODECS_CNG_WEBRTC_CNG_H_
> #define MODULES_AUDIO_CODING_CODECS_CNG_WEBRTC_CNG_H_
> 
>@@ -16,7 +15,6 @@
> 
> #include "api/array_view.h"
> #include "rtc_base/buffer.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> #define WEBRTC_CNG_MAX_LPC_ORDER 12
> 
>@@ -54,8 +52,8 @@ class ComfortNoiseDecoder {
>   int16_t dec_filtstate_[WEBRTC_CNG_MAX_LPC_ORDER + 1];
>   int16_t dec_filtstateLow_[WEBRTC_CNG_MAX_LPC_ORDER + 1];
>   uint16_t dec_order_;
>-  int16_t dec_target_scale_factor_;  /* Q29 */
>-  int16_t dec_used_scale_factor_;  /* Q29 */
>+  int16_t dec_target_scale_factor_; /* Q29 */
>+  int16_t dec_used_scale_factor_;   /* Q29 */
> };
> 
> class ComfortNoiseEncoder {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc
>index a620a3ed7b6..25f495f39d8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g711/audio_decoder_pcm.cc
>@@ -10,8 +10,8 @@
> 
> #include "modules/audio_coding/codecs/g711/audio_decoder_pcm.h"
> 
>-#include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
> #include "modules/audio_coding/codecs/g711/g711_interface.h"
>+#include "modules/audio_coding/codecs/legacy_encoded_audio_frame.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
>index 9fb94fd5c1b..c14287ed487 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.cc
>@@ -42,8 +42,8 @@ AudioEncoderPcm::AudioEncoderPcm(const Config& config, int sample_rate_hz)
>       payload_type_(config.payload_type),
>       num_10ms_frames_per_packet_(
>           static_cast<size_t>(config.frame_size_ms / 10)),
>-      full_frame_samples_(
>-          config.num_channels * config.frame_size_ms * sample_rate_hz / 1000),
>+      full_frame_samples_(config.num_channels * config.frame_size_ms *
>+                          sample_rate_hz / 1000),
>       first_timestamp_in_buffer_(0) {
>   RTC_CHECK_GT(sample_rate_hz, 0) << "Sample rate must be larger than 0 Hz";
>   RTC_CHECK_EQ(config.frame_size_ms % 10, 0)
>@@ -70,8 +70,8 @@ size_t AudioEncoderPcm::Max10MsFramesInAPacket() const {
> }
> 
> int AudioEncoderPcm::GetTargetBitrate() const {
>-  return static_cast<int>(
>-      8 * BytesPerSample() * SampleRateHz() * NumChannels());
>+  return static_cast<int>(8 * BytesPerSample() * SampleRateHz() *
>+                          NumChannels());
> }
> 
> AudioEncoder::EncodedInfo AudioEncoderPcm::EncodeImpl(
>@@ -89,13 +89,12 @@ AudioEncoder::EncodedInfo AudioEncoderPcm::EncodeImpl(
>   EncodedInfo info;
>   info.encoded_timestamp = first_timestamp_in_buffer_;
>   info.payload_type = payload_type_;
>-  info.encoded_bytes =
>-      encoded->AppendData(full_frame_samples_ * BytesPerSample(),
>-                          [&] (rtc::ArrayView<uint8_t> encoded) {
>-                            return EncodeCall(&speech_buffer_[0],
>-                                              full_frame_samples_,
>-                                              encoded.data());
>-                          });
>+  info.encoded_bytes = encoded->AppendData(
>+      full_frame_samples_ * BytesPerSample(),
>+      [&](rtc::ArrayView<uint8_t> encoded) {
>+        return EncodeCall(&speech_buffer_[0], full_frame_samples_,
>+                          encoded.data());
>+      });
>   speech_buffer_.clear();
>   info.encoder_type = GetCodecType();
>   return info;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g711/g711_interface.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g711/g711_interface.c
>index b5dacd6894e..5fe1692ccbc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g711/g711_interface.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g711/g711_interface.c
>@@ -7,10 +7,11 @@
>  *  in the file PATENTS.  All contributing project authors may
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
>+
> #include <string.h>
>-#include "g711.h"
>-#include "g711_interface.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>+
>+#include "modules/third_party/g711/g711.h"
>+#include "modules/audio_coding/codecs/g711/g711_interface.h"
> 
> size_t WebRtcG711_EncodeA(const int16_t* speechIn,
>                           size_t len,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g711/g711_interface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g711/g711_interface.h
>index 1f23da62de8..83f9d378ed1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g711/g711_interface.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g711/g711_interface.h
>@@ -11,7 +11,7 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_G711_G711_INTERFACE_H_
> #define MODULES_AUDIO_CODING_CODECS_G711_G711_INTERFACE_H_
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include <stdint.h>
> 
> // Comfort noise constants
> #define G711_WEBRTC_SPEECH 1
>@@ -112,19 +112,19 @@ size_t WebRtcG711_DecodeU(const uint8_t* encoded,
>                           int16_t* speechType);
> 
> /**********************************************************************
>-* WebRtcG711_Version(...)
>-*
>-* This function gives the version string of the G.711 codec.
>-*
>-* Input:
>-*      - lenBytes:     the size of Allocated space (in Bytes) where
>-*                      the version number is written to (in string format).
>-*
>-* Output:
>-*      - version:      Pointer to a buffer where the version number is
>-*                      written to.
>-*
>-*/
>+ * WebRtcG711_Version(...)
>+ *
>+ * This function gives the version string of the G.711 codec.
>+ *
>+ * Input:
>+ *      - lenBytes:     the size of Allocated space (in Bytes) where
>+ *                      the version number is written to (in string format).
>+ *
>+ * Output:
>+ *      - version:      Pointer to a buffer where the version number is
>+ *                      written to.
>+ *
>+ */
> 
> int16_t WebRtcG711_Version(char* version, int16_t lenBytes);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g711/test/testG711.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g711/test/testG711.cc
>index 98f3925faa5..f3a42f5d79a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g711/test/testG711.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g711/test/testG711.cc
>@@ -69,7 +69,6 @@ int main(int argc, char* argv[]) {
>     printf("outfile    : Speech output file\n\n");
>     printf("outbits    : Output bitstream file [optional]\n\n");
>     exit(0);
>-
>   }
> 
>   /* Get version and print */
>@@ -80,8 +79,8 @@ int main(int argc, char* argv[]) {
>   /* Get frame length */
>   int framelength_int = atoi(argv[1]);
>   if (framelength_int < 0) {
>-      printf("  G.722: Invalid framelength %d.\n", framelength_int);
>-      exit(1);
>+    printf("  G.722: Invalid framelength %d.\n", framelength_int);
>+    exit(1);
>   }
>   framelength = static_cast<size_t>(framelength_int);
> 
>@@ -112,7 +111,7 @@ int main(int argc, char* argv[]) {
>     printf("\nBitfile:  %s\n", bitname);
>   }
> 
>-  starttime = clock() / (double) CLOCKS_PER_SEC_G711; /* Runtime statistics */
>+  starttime = clock() / (double)CLOCKS_PER_SEC_G711; /* Runtime statistics */
> 
>   /* Initialize encoder and decoder */
>   framecnt = 0;
>@@ -155,11 +154,10 @@ int main(int argc, char* argv[]) {
>     }
>   }
> 
>-  runtime = (double)(clock() / (double) CLOCKS_PER_SEC_G711 - starttime);
>-  length_file = ((double) framecnt * (double) framelength / 8000);
>+  runtime = (double)(clock() / (double)CLOCKS_PER_SEC_G711 - starttime);
>+  length_file = ((double)framecnt * (double)framelength / 8000);
>   printf("\n\nLength of speech file: %.1f s\n", length_file);
>-  printf("Time to run G.711:      %.2f s (%.2f %% of realtime)\n\n",
>-         runtime,
>+  printf("Time to run G.711:      %.2f s (%.2f %% of realtime)\n\n", runtime,
>          (100 * runtime / length_file));
>   printf("---------------------END----------------------\n");
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
>index ec97ee33212..cb96c3c17ef 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
>@@ -123,7 +123,7 @@ AudioEncoder::EncodedInfo AudioEncoderG722Impl::EncodeImpl(
>   const size_t bytes_to_encode = samples_per_channel / 2 * num_channels_;
>   EncodedInfo info;
>   info.encoded_bytes = encoded->AppendData(
>-      bytes_to_encode, [&] (rtc::ArrayView<uint8_t> encoded) {
>+      bytes_to_encode, [&](rtc::ArrayView<uint8_t> encoded) {
>         // Interleave the encoded bytes of the different channels. Each separate
>         // channel and the interleaved stream encodes two samples per byte, most
>         // significant half first.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h
>index 1f4b943f664..3cf143914f4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h
>@@ -46,8 +46,8 @@ class AudioEncoderG722Impl final : public AudioEncoder {
>   // The encoder state for one channel.
>   struct EncoderState {
>     G722EncInst* encoder;
>-    std::unique_ptr<int16_t[]> speech_buffer;   // Queued up for encoding.
>-    rtc::Buffer encoded_buffer;                 // Already encoded.
>+    std::unique_ptr<int16_t[]> speech_buffer;  // Queued up for encoding.
>+    rtc::Buffer encoded_buffer;                // Already encoded.
>     EncoderState();
>     ~EncoderState();
>   };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/g722_enc_dec.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/g722_enc_dec.h
>deleted file mode 100644
>index ccda09baf7a..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/g722_enc_dec.h
>+++ /dev/null
>@@ -1,160 +0,0 @@
>-/*
>- * SpanDSP - a series of DSP components for telephony
>- *
>- * g722.h - The ITU G.722 codec.
>- *
>- * Written by Steve Underwood <steveu@coppice.org>
>- *
>- * Copyright (C) 2005 Steve Underwood
>- *
>- *  Despite my general liking of the GPL, I place my own contributions 
>- *  to this code in the public domain for the benefit of all mankind -
>- *  even the slimy ones who might try to proprietize my work and use it
>- *  to my detriment.
>- *
>- * Based on a single channel G.722 codec which is:
>- *
>- *****    Copyright (c) CMU    1993      *****
>- * Computer Science, Speech Group
>- * Chengxiang Lu and Alex Hauptmann
>- *
>- * $Id: g722.h,v 1.10 2006/06/16 12:45:53 steveu Exp $
>- *
>- * Modifications for WebRtc, 2011/04/28, by tlegrand:
>- * -Changed to use WebRtc types
>- * -Added new defines for minimum and maximum values of short int
>- */
>-
>-
>-/*! \file */
>-
>-#if !defined(_G722_ENC_DEC_H_)
>-#define _G722_ENC_DEC_H_
>-
>-#include "typedefs.h"  // NOLINT(build/include)
>-
>-/*! \page g722_page G.722 encoding and decoding
>-\section g722_page_sec_1 What does it do?
>-The G.722 module is a bit exact implementation of the ITU G.722 specification for all three
>-specified bit rates - 64000bps, 56000bps and 48000bps. It passes the ITU tests.
>-
>-To allow fast and flexible interworking with narrow band telephony, the encoder and decoder
>-support an option for the linear audio to be an 8k samples/second stream. In this mode the
>-codec is considerably faster, and still fully compatible with wideband terminals using G.722.
>-
>-\section g722_page_sec_2 How does it work?
>-???.
>-*/
>-
>-#define WEBRTC_INT16_MAX 32767
>-#define WEBRTC_INT16_MIN -32768
>-
>-enum
>-{
>-    G722_SAMPLE_RATE_8000 = 0x0001,
>-    G722_PACKED = 0x0002
>-};
>-
>-typedef struct
>-{
>-    /*! TRUE if the operating in the special ITU test mode, with the band split filters
>-             disabled. */
>-    int itu_test_mode;
>-    /*! TRUE if the G.722 data is packed */
>-    int packed;
>-    /*! TRUE if encode from 8k samples/second */
>-    int eight_k;
>-    /*! 6 for 48000kbps, 7 for 56000kbps, or 8 for 64000kbps. */
>-    int bits_per_sample;
>-
>-    /*! Signal history for the QMF */
>-    int x[24];
>-
>-    struct
>-    {
>-        int s;
>-        int sp;
>-        int sz;
>-        int r[3];
>-        int a[3];
>-        int ap[3];
>-        int p[3];
>-        int d[7];
>-        int b[7];
>-        int bp[7];
>-        int sg[7];
>-        int nb;
>-        int det;
>-    } band[2];
>-
>-    unsigned int in_buffer;
>-    int in_bits;
>-    unsigned int out_buffer;
>-    int out_bits;
>-} G722EncoderState;
>-
>-typedef struct
>-{
>-    /*! TRUE if the operating in the special ITU test mode, with the band split filters
>-             disabled. */
>-    int itu_test_mode;
>-    /*! TRUE if the G.722 data is packed */
>-    int packed;
>-    /*! TRUE if decode to 8k samples/second */
>-    int eight_k;
>-    /*! 6 for 48000kbps, 7 for 56000kbps, or 8 for 64000kbps. */
>-    int bits_per_sample;
>-
>-    /*! Signal history for the QMF */
>-    int x[24];
>-
>-    struct
>-    {
>-        int s;
>-        int sp;
>-        int sz;
>-        int r[3];
>-        int a[3];
>-        int ap[3];
>-        int p[3];
>-        int d[7];
>-        int b[7];
>-        int bp[7];
>-        int sg[7];
>-        int nb;
>-        int det;
>-    } band[2];
>-    
>-    unsigned int in_buffer;
>-    int in_bits;
>-    unsigned int out_buffer;
>-    int out_bits;
>-} G722DecoderState;
>-
>-#ifdef __cplusplus
>-extern "C" {
>-#endif
>-
>-G722EncoderState* WebRtc_g722_encode_init(G722EncoderState* s,
>-                                          int rate,
>-                                          int options);
>-int WebRtc_g722_encode_release(G722EncoderState *s);
>-size_t WebRtc_g722_encode(G722EncoderState *s,
>-                          uint8_t g722_data[],
>-                          const int16_t amp[],
>-                          size_t len);
>-
>-G722DecoderState* WebRtc_g722_decode_init(G722DecoderState* s,
>-                                          int rate,
>-                                          int options);
>-int WebRtc_g722_decode_release(G722DecoderState *s);
>-size_t WebRtc_g722_decode(G722DecoderState *s,
>-                          int16_t amp[],
>-                          const uint8_t g722_data[],
>-                          size_t len);
>-
>-#ifdef __cplusplus
>-}
>-#endif
>-
>-#endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/g722_interface.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/g722_interface.c
>index b6dccb454f3..36ee6d92bea 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/g722_interface.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/g722_interface.c
>@@ -8,13 +8,11 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-
>-
> #include <stdlib.h>
> #include <string.h>
>-#include "g722_enc_dec.h"
>-#include "g722_interface.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>+
>+#include "modules/audio_coding/codecs/g722/g722_interface.h"
>+#include "modules/third_party/g722/g722_enc_dec.h"
> 
> int16_t WebRtcG722_CreateEncoder(G722EncInst **G722enc_inst)
> {
>@@ -104,4 +102,3 @@ int16_t WebRtcG722_Version(char *versionStr, short len)
>         return -1;
>     }
> }
>-
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/g722_interface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/g722_interface.h
>index d95722314f8..85c1cd02a08 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/g722_interface.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/g722_interface.h
>@@ -11,27 +11,26 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_G722_G722_INTERFACE_H_
> #define MODULES_AUDIO_CODING_CODECS_G722_G722_INTERFACE_H_
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include <stdint.h>
> 
> /*
>  * Solution to support multiple instances
>  */
> 
>-typedef struct WebRtcG722EncInst    G722EncInst;
>-typedef struct WebRtcG722DecInst    G722DecInst;
>+typedef struct WebRtcG722EncInst G722EncInst;
>+typedef struct WebRtcG722DecInst G722DecInst;
> 
> /*
>  * Comfort noise constants
>  */
> 
>-#define G722_WEBRTC_SPEECH     1
>-#define G722_WEBRTC_CNG        2
>+#define G722_WEBRTC_SPEECH 1
>+#define G722_WEBRTC_CNG 2
> 
> #ifdef __cplusplus
> extern "C" {
> #endif
> 
>-
> /****************************************************************************
>  * WebRtcG722_CreateEncoder(...)
>  *
>@@ -43,8 +42,7 @@ extern "C" {
>  * Return value               :  0 - Ok
>  *                              -1 - Error
>  */
>-int16_t WebRtcG722_CreateEncoder(G722EncInst **G722enc_inst);
>-
>+int16_t WebRtcG722_CreateEncoder(G722EncInst** G722enc_inst);
> 
> /****************************************************************************
>  * WebRtcG722_EncoderInit(...)
>@@ -59,8 +57,7 @@ int16_t WebRtcG722_CreateEncoder(G722EncInst **G722enc_inst);
>  *                              -1 - Error
>  */
> 
>-int16_t WebRtcG722_EncoderInit(G722EncInst *G722enc_inst);
>-
>+int16_t WebRtcG722_EncoderInit(G722EncInst* G722enc_inst);
> 
> /****************************************************************************
>  * WebRtcG722_FreeEncoder(...)
>@@ -73,9 +70,7 @@ int16_t WebRtcG722_EncoderInit(G722EncInst *G722enc_inst);
>  * Return value               :  0 - Ok
>  *                              -1 - Error
>  */
>-int WebRtcG722_FreeEncoder(G722EncInst *G722enc_inst);
>-
>-
>+int WebRtcG722_FreeEncoder(G722EncInst* G722enc_inst);
> 
> /****************************************************************************
>  * WebRtcG722_Encode(...)
>@@ -99,7 +94,6 @@ size_t WebRtcG722_Encode(G722EncInst* G722enc_inst,
>                          size_t len,
>                          uint8_t* encoded);
> 
>-
> /****************************************************************************
>  * WebRtcG722_CreateDecoder(...)
>  *
>@@ -111,7 +105,7 @@ size_t WebRtcG722_Encode(G722EncInst* G722enc_inst,
>  * Return value               :  0 - Ok
>  *                              -1 - Error
>  */
>-int16_t WebRtcG722_CreateDecoder(G722DecInst **G722dec_inst);
>+int16_t WebRtcG722_CreateDecoder(G722DecInst** G722dec_inst);
> 
> /****************************************************************************
>  * WebRtcG722_DecoderInit(...)
>@@ -136,8 +130,7 @@ void WebRtcG722_DecoderInit(G722DecInst* inst);
>  *                              -1 - Error
>  */
> 
>-int WebRtcG722_FreeDecoder(G722DecInst *G722dec_inst);
>-
>+int WebRtcG722_FreeDecoder(G722DecInst* G722dec_inst);
> 
> /****************************************************************************
>  * WebRtcG722_Decode(...)
>@@ -159,11 +152,11 @@ int WebRtcG722_FreeDecoder(G722DecInst *G722dec_inst);
>  * Return value             : Samples in decoded vector
>  */
> 
>-size_t WebRtcG722_Decode(G722DecInst *G722dec_inst,
>+size_t WebRtcG722_Decode(G722DecInst* G722dec_inst,
>                          const uint8_t* encoded,
>                          size_t len,
>-                         int16_t *decoded,
>-                         int16_t *speechType);
>+                         int16_t* decoded,
>+                         int16_t* speechType);
> 
> /****************************************************************************
>  * WebRtcG722_Version(...)
>@@ -171,12 +164,10 @@ size_t WebRtcG722_Decode(G722DecInst *G722dec_inst,
>  * Get a string with the current version of the codec
>  */
> 
>-int16_t WebRtcG722_Version(char *versionStr, short len);
>-
>+int16_t WebRtcG722_Version(char* versionStr, short len);
> 
> #ifdef __cplusplus
> }
> #endif
> 
>-
> #endif /* MODULES_AUDIO_CODING_CODECS_G722_G722_INTERFACE_H_ */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/test/testG722.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/test/testG722.cc
>index e0281f2b655..9f2155d0f7f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/test/testG722.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/test/testG722.cc
>@@ -15,144 +15,141 @@
> #include <stdio.h>
> #include <stdlib.h>
> #include <string.h>
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> /* include API */
> #include "modules/audio_coding/codecs/g722/g722_interface.h"
> 
> /* Runtime statistics */
> #include <time.h>
>-#define CLOCKS_PER_SEC_G722  100000
>+#define CLOCKS_PER_SEC_G722 100000
> 
> // Forward declaration
>-typedef struct WebRtcG722EncInst    G722EncInst;
>-typedef struct WebRtcG722DecInst    G722DecInst;
>+typedef struct WebRtcG722EncInst G722EncInst;
>+typedef struct WebRtcG722DecInst G722DecInst;
> 
> /* function for reading audio data from PCM file */
>-bool readframe(int16_t *data, FILE *inp, size_t length)
>-{
>-    size_t rlen = fread(data, sizeof(int16_t), length, inp);
>-    if (rlen >= length)
>-      return false;
>-    memset(data + rlen, 0, (length - rlen) * sizeof(int16_t));
>-    return true;
>+bool readframe(int16_t* data, FILE* inp, size_t length) {
>+  size_t rlen = fread(data, sizeof(int16_t), length, inp);
>+  if (rlen >= length)
>+    return false;
>+  memset(data + rlen, 0, (length - rlen) * sizeof(int16_t));
>+  return true;
> }
> 
>-int main(int argc, char* argv[])
>-{
>-    char inname[60], outbit[40], outname[40];
>-    FILE *inp, *outbitp, *outp;
>-
>-    int framecnt;
>-    bool endfile;
>-    size_t framelength = 160;
>-    G722EncInst *G722enc_inst;
>-    G722DecInst *G722dec_inst;
>-
>-    /* Runtime statistics */
>-    double starttime;
>-    double runtime = 0;
>-    double length_file;
>-
>-    size_t stream_len = 0;
>-    int16_t shortdata[960];
>-    int16_t decoded[960];
>-    uint8_t streamdata[80 * 6];
>-    int16_t speechType[1];
>-
>-    /* handling wrong input arguments in the command line */
>-    if (argc!=5)  {
>-        printf("\n\nWrong number of arguments or flag values.\n\n");
>-
>-        printf("\n");
>-        printf("Usage:\n\n");
>-        printf("./testG722.exe framelength infile outbitfile outspeechfile \n\n");
>-        printf("with:\n");
>-        printf("framelength  :    Framelength in samples.\n\n");
>-        printf("infile       :    Normal speech input file\n\n");
>-        printf("outbitfile   :    Bitstream output file\n\n");
>-        printf("outspeechfile:    Speech output file\n\n");
>-        exit(0);
>-
>-    }
>-
>-    /* Get frame length */
>-    int framelength_int = atoi(argv[1]);
>-    if (framelength_int < 0) {
>-        printf("  G.722: Invalid framelength %d.\n", framelength_int);
>-        exit(1);
>-    }
>-    framelength = static_cast<size_t>(framelength_int);
>-
>-    /* Get Input and Output files */
>-    sscanf(argv[2], "%s", inname);
>-    sscanf(argv[3], "%s", outbit);
>-    sscanf(argv[4], "%s", outname);
>-
>-    if ((inp = fopen(inname,"rb")) == NULL) {
>-        printf("  G.722: Cannot read file %s.\n", inname);
>-        exit(1);
>-    }
>-    if ((outbitp = fopen(outbit,"wb")) == NULL) {
>-        printf("  G.722: Cannot write file %s.\n", outbit);
>-        exit(1);
>-    }
>-    if ((outp = fopen(outname,"wb")) == NULL) {
>-        printf("  G.722: Cannot write file %s.\n", outname);
>-        exit(1);
>+int main(int argc, char* argv[]) {
>+  char inname[60], outbit[40], outname[40];
>+  FILE *inp, *outbitp, *outp;
>+
>+  int framecnt;
>+  bool endfile;
>+  size_t framelength = 160;
>+  G722EncInst* G722enc_inst;
>+  G722DecInst* G722dec_inst;
>+
>+  /* Runtime statistics */
>+  double starttime;
>+  double runtime = 0;
>+  double length_file;
>+
>+  size_t stream_len = 0;
>+  int16_t shortdata[960];
>+  int16_t decoded[960];
>+  uint8_t streamdata[80 * 6];
>+  int16_t speechType[1];
>+
>+  /* handling wrong input arguments in the command line */
>+  if (argc != 5) {
>+    printf("\n\nWrong number of arguments or flag values.\n\n");
>+
>+    printf("\n");
>+    printf("Usage:\n\n");
>+    printf("./testG722.exe framelength infile outbitfile outspeechfile \n\n");
>+    printf("with:\n");
>+    printf("framelength  :    Framelength in samples.\n\n");
>+    printf("infile       :    Normal speech input file\n\n");
>+    printf("outbitfile   :    Bitstream output file\n\n");
>+    printf("outspeechfile:    Speech output file\n\n");
>+    exit(0);
>+  }
>+
>+  /* Get frame length */
>+  int framelength_int = atoi(argv[1]);
>+  if (framelength_int < 0) {
>+    printf("  G.722: Invalid framelength %d.\n", framelength_int);
>+    exit(1);
>+  }
>+  framelength = static_cast<size_t>(framelength_int);
>+
>+  /* Get Input and Output files */
>+  sscanf(argv[2], "%s", inname);
>+  sscanf(argv[3], "%s", outbit);
>+  sscanf(argv[4], "%s", outname);
>+
>+  if ((inp = fopen(inname, "rb")) == NULL) {
>+    printf("  G.722: Cannot read file %s.\n", inname);
>+    exit(1);
>+  }
>+  if ((outbitp = fopen(outbit, "wb")) == NULL) {
>+    printf("  G.722: Cannot write file %s.\n", outbit);
>+    exit(1);
>+  }
>+  if ((outp = fopen(outname, "wb")) == NULL) {
>+    printf("  G.722: Cannot write file %s.\n", outname);
>+    exit(1);
>+  }
>+  printf("\nInput:%s\nOutput bitstream:%s\nOutput:%s\n", inname, outbit,
>+         outname);
>+
>+  /* Create and init */
>+  WebRtcG722_CreateEncoder((G722EncInst**)&G722enc_inst);
>+  WebRtcG722_CreateDecoder((G722DecInst**)&G722dec_inst);
>+  WebRtcG722_EncoderInit((G722EncInst*)G722enc_inst);
>+  WebRtcG722_DecoderInit((G722DecInst*)G722dec_inst);
>+
>+  /* Initialize encoder and decoder */
>+  framecnt = 0;
>+  endfile = false;
>+  while (!endfile) {
>+    framecnt++;
>+
>+    /* Read speech block */
>+    endfile = readframe(shortdata, inp, framelength);
>+
>+    /* Start clock before call to encoder and decoder */
>+    starttime = clock() / (double)CLOCKS_PER_SEC_G722;
>+
>+    /* G.722 encoding + decoding */
>+    stream_len = WebRtcG722_Encode((G722EncInst*)G722enc_inst, shortdata,
>+                                   framelength, streamdata);
>+    WebRtcG722_Decode(G722dec_inst, streamdata, stream_len, decoded,
>+                      speechType);
>+
>+    /* Stop clock after call to encoder and decoder */
>+    runtime += (double)((clock() / (double)CLOCKS_PER_SEC_G722) - starttime);
>+
>+    /* Write coded bits to file */
>+    if (fwrite(streamdata, sizeof(short), stream_len / 2, outbitp) !=
>+        stream_len / 2) {
>+      return -1;
>     }
>-    printf("\nInput:%s\nOutput bitstream:%s\nOutput:%s\n", inname, outbit, outname);
>-
>-    /* Create and init */
>-    WebRtcG722_CreateEncoder((G722EncInst **)&G722enc_inst);
>-    WebRtcG722_CreateDecoder((G722DecInst **)&G722dec_inst);
>-    WebRtcG722_EncoderInit((G722EncInst *)G722enc_inst);
>-    WebRtcG722_DecoderInit((G722DecInst *)G722dec_inst);
>-
>-
>-    /* Initialize encoder and decoder */
>-    framecnt = 0;
>-    endfile = false;
>-    while (!endfile) {
>-        framecnt++;
>-
>-        /* Read speech block */
>-        endfile = readframe(shortdata, inp, framelength);
>-
>-        /* Start clock before call to encoder and decoder */
>-        starttime = clock()/(double)CLOCKS_PER_SEC_G722;
>-
>-        /* G.722 encoding + decoding */
>-        stream_len = WebRtcG722_Encode((G722EncInst *)G722enc_inst, shortdata, framelength, streamdata);
>-        WebRtcG722_Decode(G722dec_inst, streamdata, stream_len, decoded,
>-                          speechType);
>-
>-        /* Stop clock after call to encoder and decoder */
>-        runtime += (double)((clock()/(double)CLOCKS_PER_SEC_G722)-starttime);
>-
>-        /* Write coded bits to file */
>-        if (fwrite(streamdata, sizeof(short), stream_len / 2, outbitp) !=
>-            stream_len / 2) {
>-          return -1;
>-        }
>-        /* Write coded speech to file */
>-        if (fwrite(decoded, sizeof(short), framelength, outp) !=
>-            framelength) {
>-          return -1;
>-        }
>+    /* Write coded speech to file */
>+    if (fwrite(decoded, sizeof(short), framelength, outp) != framelength) {
>+      return -1;
>     }
>+  }
> 
>-    WebRtcG722_FreeEncoder((G722EncInst *)G722enc_inst);
>-    WebRtcG722_FreeDecoder((G722DecInst *)G722dec_inst);
>+  WebRtcG722_FreeEncoder((G722EncInst*)G722enc_inst);
>+  WebRtcG722_FreeDecoder((G722DecInst*)G722dec_inst);
> 
>-    length_file = ((double)framecnt*(double)framelength/16000);
>-    printf("\n\nLength of speech file: %.1f s\n", length_file);
>-    printf("Time to run G.722:      %.2f s (%.2f %% of realtime)\n\n", runtime, (100*runtime/length_file));
>-    printf("---------------------END----------------------\n");
>+  length_file = ((double)framecnt * (double)framelength / 16000);
>+  printf("\n\nLength of speech file: %.1f s\n", length_file);
>+  printf("Time to run G.722:      %.2f s (%.2f %% of realtime)\n\n", runtime,
>+         (100 * runtime / length_file));
>+  printf("---------------------END----------------------\n");
> 
>-    fclose(inp);
>-    fclose(outbitp);
>-    fclose(outp);
>+  fclose(inp);
>+  fclose(outbitp);
>+  fclose(outp);
> 
>-    return 0;
>+  return 0;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/abs_quant.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/abs_quant.c
>index 263749ad2a0..308902f574a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/abs_quant.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/abs_quant.c
>@@ -16,9 +16,9 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>-#include "constants.h"
>-#include "abs_quant_loop.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
>+#include "modules/audio_coding/codecs/ilbc/constants.h"
>+#include "modules/audio_coding/codecs/ilbc/abs_quant_loop.h"
> 
> 
> /*----------------------------------------------------------------*
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/abs_quant.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/abs_quant.h
>index 090ed73f123..331921cc639 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/abs_quant.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/abs_quant.h
>@@ -19,7 +19,7 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ABS_QUANT_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ABS_QUANT_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  *  predictive noise shaping encoding of scaled start state
>@@ -27,13 +27,13 @@
>  *---------------------------------------------------------------*/
> 
> void WebRtcIlbcfix_AbsQuant(
>-    IlbcEncoder *iLBCenc_inst,
>+    IlbcEncoder* iLBCenc_inst,
>     /* (i) Encoder instance */
>-    iLBC_bits *iLBC_encbits, /* (i/o) Encoded bits (outputs idxForMax
>+    iLBC_bits* iLBC_encbits, /* (i/o) Encoded bits (outputs idxForMax
>                                    and idxVec, uses state_first as
>                                    input) */
>-    int16_t *in,     /* (i) vector to encode */
>-    int16_t *weightDenum   /* (i) denominator of synthesis filter */
>-                            );
>+    int16_t* in,             /* (i) vector to encode */
>+    int16_t* weightDenum     /* (i) denominator of synthesis filter */
>+    );
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/abs_quant_loop.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/abs_quant_loop.c
>index 4b764534467..2d8a9980ebb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/abs_quant_loop.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/abs_quant_loop.c
>@@ -16,9 +16,9 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>-#include "constants.h"
>-#include "sort_sq.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
>+#include "modules/audio_coding/codecs/ilbc/constants.h"
>+#include "modules/audio_coding/codecs/ilbc/sort_sq.h"
> 
> void WebRtcIlbcfix_AbsQuantLoop(int16_t *syntOutIN, int16_t *in_weightedIN,
>                                 int16_t *weightDenumIN, size_t *quantLenIN,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/abs_quant_loop.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/abs_quant_loop.h
>index 92f6fff8474..a193a07cda1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/abs_quant_loop.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/abs_quant_loop.h
>@@ -19,15 +19,17 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ABS_QUANT_LOOP_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ABS_QUANT_LOOP_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  *  predictive noise shaping encoding of scaled start state
>  *  (subrutine for WebRtcIlbcfix_StateSearch)
>  *---------------------------------------------------------------*/
> 
>-void WebRtcIlbcfix_AbsQuantLoop(int16_t *syntOutIN, int16_t *in_weightedIN,
>-                                int16_t *weightDenumIN, size_t *quantLenIN,
>-                                int16_t *idxVecIN);
>+void WebRtcIlbcfix_AbsQuantLoop(int16_t* syntOutIN,
>+                                int16_t* in_weightedIN,
>+                                int16_t* weightDenumIN,
>+                                size_t* quantLenIN,
>+                                int16_t* idxVecIN);
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc
>index 08d21f4202e..9e58ce03ca4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/audio_decoder_ilbc.cc
>@@ -33,10 +33,10 @@ bool AudioDecoderIlbcImpl::HasDecodePlc() const {
> }
> 
> int AudioDecoderIlbcImpl::DecodeInternal(const uint8_t* encoded,
>-                                     size_t encoded_len,
>-                                     int sample_rate_hz,
>-                                     int16_t* decoded,
>-                                     SpeechType* speech_type) {
>+                                         size_t encoded_len,
>+                                         int sample_rate_hz,
>+                                         int16_t* decoded,
>+                                         SpeechType* speech_type) {
>   RTC_DCHECK_EQ(sample_rate_hz, 8000);
>   int16_t temp_type = 1;  // Default is speech.
>   int ret = WebRtcIlbcfix_Decode(dec_state_, encoded, encoded_len, decoded,
>@@ -86,10 +86,9 @@ std::vector<AudioDecoder::ParseResult> AudioDecoderIlbcImpl::ParsePayload(
>   } else {
>     size_t byte_offset;
>     uint32_t timestamp_offset;
>-    for (byte_offset = 0, timestamp_offset = 0;
>-         byte_offset < payload.size();
>+    for (byte_offset = 0, timestamp_offset = 0; byte_offset < payload.size();
>          byte_offset += bytes_per_frame,
>-             timestamp_offset += timestamps_per_frame) {
>+        timestamp_offset += timestamps_per_frame) {
>       std::unique_ptr<EncodedAudioFrame> frame(new LegacyEncodedAudioFrame(
>           this, rtc::Buffer(payload.data() + byte_offset, bytes_per_frame)));
>       results.emplace_back(timestamp + timestamp_offset, 0, std::move(frame));
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
>index e108c412ceb..dae956b30c0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
>@@ -89,7 +89,6 @@ AudioEncoder::EncodedInfo AudioEncoderIlbcImpl::EncodeImpl(
>     uint32_t rtp_timestamp,
>     rtc::ArrayView<const int16_t> audio,
>     rtc::Buffer* encoded) {
>-
>   // Save timestamp if starting a new packet.
>   if (num_10ms_frames_buffered_ == 0)
>     first_timestamp_in_buffer_ = rtp_timestamp;
>@@ -107,19 +106,15 @@ AudioEncoder::EncodedInfo AudioEncoderIlbcImpl::EncodeImpl(
>   // Encode buffered input.
>   RTC_DCHECK_EQ(num_10ms_frames_buffered_, num_10ms_frames_per_packet_);
>   num_10ms_frames_buffered_ = 0;
>-  size_t encoded_bytes =
>-      encoded->AppendData(
>-          RequiredOutputSizeBytes(),
>-          [&] (rtc::ArrayView<uint8_t> encoded) {
>-            const int r = WebRtcIlbcfix_Encode(
>-                encoder_,
>-                input_buffer_,
>-                kSampleRateHz / 100 * num_10ms_frames_per_packet_,
>-                encoded.data());
>-            RTC_CHECK_GE(r, 0);
>-
>-            return static_cast<size_t>(r);
>-          });
>+  size_t encoded_bytes = encoded->AppendData(
>+      RequiredOutputSizeBytes(), [&](rtc::ArrayView<uint8_t> encoded) {
>+        const int r = WebRtcIlbcfix_Encode(
>+            encoder_, input_buffer_,
>+            kSampleRateHz / 100 * num_10ms_frames_per_packet_, encoded.data());
>+        RTC_CHECK_GE(r, 0);
>+
>+        return static_cast<size_t>(r);
>+      });
> 
>   RTC_DCHECK_EQ(encoded_bytes, RequiredOutputSizeBytes());
> 
>@@ -135,20 +130,24 @@ void AudioEncoderIlbcImpl::Reset() {
>   if (encoder_)
>     RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderFree(encoder_));
>   RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderCreate(&encoder_));
>-  const int encoder_frame_size_ms = frame_size_ms_ > 30
>-                                        ? frame_size_ms_ / 2
>-                                        : frame_size_ms_;
>+  const int encoder_frame_size_ms =
>+      frame_size_ms_ > 30 ? frame_size_ms_ / 2 : frame_size_ms_;
>   RTC_CHECK_EQ(0, WebRtcIlbcfix_EncoderInit(encoder_, encoder_frame_size_ms));
>   num_10ms_frames_buffered_ = 0;
> }
> 
> size_t AudioEncoderIlbcImpl::RequiredOutputSizeBytes() const {
>   switch (num_10ms_frames_per_packet_) {
>-    case 2:   return 38;
>-    case 3:   return 50;
>-    case 4:   return 2 * 38;
>-    case 6:   return 2 * 50;
>-    default:  RTC_FATAL();
>+    case 2:
>+      return 38;
>+    case 3:
>+      return 50;
>+    case 4:
>+      return 2 * 38;
>+    case 6:
>+      return 2 * 50;
>+    default:
>+      RTC_FATAL();
>   }
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.c
>index 1a3735fc3db..77b0f7f7336 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.c
>@@ -16,9 +16,9 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>-#include "constants.h"
>-#include "augmented_cb_corr.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
>+#include "modules/audio_coding/codecs/ilbc/constants.h"
>+#include "modules/audio_coding/codecs/ilbc/augmented_cb_corr.h"
> 
> void WebRtcIlbcfix_AugmentedCbCorr(
>     int16_t *target,   /* (i) Target vector */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.h
>index e8eb0696d22..646e5649b98 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/augmented_cb_corr.h
>@@ -19,23 +19,23 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_AUGMENTED_CB_CORR_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_AUGMENTED_CB_CORR_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  *  Calculate correlation between target and Augmented codebooks
>  *---------------------------------------------------------------*/
> 
> void WebRtcIlbcfix_AugmentedCbCorr(
>-    int16_t *target,   /* (i) Target vector */
>-    int16_t *buffer,   /* (i) Memory buffer */
>-    int16_t *interpSamples, /* (i) buffer with
>+    int16_t* target,        /* (i) Target vector */
>+    int16_t* buffer,        /* (i) Memory buffer */
>+    int16_t* interpSamples, /* (i) buffer with
>                                            interpolated samples */
>-    int32_t *crossDot,  /* (o) The cross correlation between
>-                                           the target and the Augmented
>-                                           vector */
>-    size_t low,    /* (i) Lag to start from (typically
>-                                                   20) */
>-    size_t high,   /* (i) Lag to end at (typically 39 */
>-    int scale);   /* (i) Scale factor to use for the crossDot */
>+    int32_t* crossDot,      /* (o) The cross correlation between
>+                                               the target and the Augmented
>+                                               vector */
>+    size_t low,             /* (i) Lag to start from (typically
>+                                                            20) */
>+    size_t high,            /* (i) Lag to end at (typically 39 */
>+    int scale);             /* (i) Scale factor to use for the crossDot */
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/bw_expand.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/bw_expand.c
>index b02abab3f25..566af7dd332 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/bw_expand.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/bw_expand.c
>@@ -16,7 +16,7 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  *  lpc bandwidth expansion
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/bw_expand.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/bw_expand.h
>index 6268baa9fa5..d25325c7f26 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/bw_expand.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/bw_expand.h
>@@ -19,18 +19,18 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_BW_EXPAND_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_BW_EXPAND_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  *  lpc bandwidth expansion
>  *---------------------------------------------------------------*/
> 
> void WebRtcIlbcfix_BwExpand(
>-    int16_t *out, /* (o) the bandwidth expanded lpc coefficients */
>-    int16_t *in,  /* (i) the lpc coefficients before bandwidth
>-                                   expansion */
>-    int16_t *coef, /* (i) the bandwidth expansion factor Q15 */
>+    int16_t* out,  /* (o) the bandwidth expanded lpc coefficients */
>+    int16_t* in,   /* (i) the lpc coefficients before bandwidth
>+                                    expansion */
>+    int16_t* coef, /* (i) the bandwidth expansion factor Q15 */
>     int16_t length /* (i) the length of lpc coefficient vectors */
>-                            );
>+    );
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_construct.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_construct.c
>index c19818ee37e..1e9a7040c7d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_construct.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_construct.c
>@@ -16,11 +16,20 @@
> 
> ******************************************************************/
> 
>-#include "cb_construct.h"
>+#include "modules/audio_coding/codecs/ilbc/cb_construct.h"
> 
>-#include "defines.h"
>-#include "gain_dequant.h"
>-#include "get_cd_vec.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
>+#include "modules/audio_coding/codecs/ilbc/gain_dequant.h"
>+#include "modules/audio_coding/codecs/ilbc/get_cd_vec.h"
>+#include "rtc_base/sanitizer.h"
>+
>+// An arithmetic operation that is allowed to overflow. (It's still undefined
>+// behavior, so not a good idea; this just makes UBSan ignore the violation, so
>+// that our old code can continue to do what it's always been doing.)
>+static inline int32_t RTC_NO_SANITIZE("signed-integer-overflow")
>+    OverflowingAddS32S32ToS32(int32_t a, int32_t b) {
>+  return a + b;
>+}
> 
> /*----------------------------------------------------------------*
>  *  Construct decoded vector from codebook and gains.
>@@ -62,7 +71,7 @@ bool WebRtcIlbcfix_CbConstruct(
>   for (j=0;j<veclen;j++) {
>     a32 = (*gainPtr++) * cbvec0[j];
>     a32 += (*gainPtr++) * cbvec1[j];
>-    a32 += (*gainPtr) * cbvec2[j];
>+    a32 = OverflowingAddS32S32ToS32(a32, (*gainPtr) * cbvec2[j]);
>     gainPtr -= 2;
>     decvector[j] = (int16_t)((a32 + 8192) >> 14);
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_construct.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_construct.h
>index aeb00da5f1c..f4df38728c9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_construct.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_construct.h
>@@ -20,7 +20,10 @@
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_CONSTRUCT_H_
> 
> #include <stdbool.h>
>-#include "defines.h"
>+#include <stdint.h>
>+
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
>+#include "rtc_base/system/unused.h"
> 
> /*----------------------------------------------------------------*
>  *  Construct decoded vector from codebook and gains.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.c
>index 6ad2f8eb7ad..9304a916805 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.c
>@@ -16,9 +16,9 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>-#include "constants.h"
>-#include "cb_mem_energy_calc.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
>+#include "modules/audio_coding/codecs/ilbc/constants.h"
>+#include "modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h"
> 
> /*----------------------------------------------------------------*
>  *  Function WebRtcIlbcfix_CbMemEnergy computes the energy of all
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.h
>index e8e2fe979ab..894f5d051cf 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy.h
>@@ -21,14 +21,14 @@
> 
> void WebRtcIlbcfix_CbMemEnergy(
>     size_t range,
>-    int16_t *CB,   /* (i) The CB memory (1:st section) */
>-    int16_t *filteredCB,  /* (i) The filtered CB memory (2:nd section) */
>-    size_t lMem,   /* (i) Length of the CB memory */
>-    size_t lTarget,   /* (i) Length of the target vector */
>-    int16_t *energyW16,  /* (o) Energy in the CB vectors */
>-    int16_t *energyShifts, /* (o) Shift value of the energy */
>-    int scale,   /* (i) The scaling of all energy values */
>-    size_t base_size  /* (i) Index to where energy values should be stored */
>-                               );
>+    int16_t* CB,           /* (i) The CB memory (1:st section) */
>+    int16_t* filteredCB,   /* (i) The filtered CB memory (2:nd section) */
>+    size_t lMem,           /* (i) Length of the CB memory */
>+    size_t lTarget,        /* (i) Length of the target vector */
>+    int16_t* energyW16,    /* (o) Energy in the CB vectors */
>+    int16_t* energyShifts, /* (o) Shift value of the energy */
>+    int scale,             /* (i) The scaling of all energy values */
>+    size_t base_size /* (i) Index to where energy values should be stored */
>+    );
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.c
>index acd6b9c1181..df9ff45be94 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.c
>@@ -16,8 +16,8 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>-#include "constants.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
>+#include "modules/audio_coding/codecs/ilbc/constants.h"
> 
> void WebRtcIlbcfix_CbMemEnergyAugmentation(
>     int16_t *interpSamples, /* (i) The interpolated samples */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.h
>index 00eb0179f1d..b7b972f9118 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.h
>@@ -20,12 +20,12 @@
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_MEM_ENERGY_AUGMENTATION_H_
> 
> void WebRtcIlbcfix_CbMemEnergyAugmentation(
>-    int16_t *interpSamples, /* (i) The interpolated samples */
>-    int16_t *CBmem,   /* (i) The CB memory */
>-    int scale,   /* (i) The scaling of all energy values */
>-    size_t base_size,  /* (i) Index to where energy values should be stored */
>-    int16_t *energyW16,  /* (o) Energy in the CB vectors */
>-    int16_t *energyShifts /* (o) Shift value of the energy */
>-                                           );
>+    int16_t* interpSamples, /* (i) The interpolated samples */
>+    int16_t* CBmem,         /* (i) The CB memory */
>+    int scale,              /* (i) The scaling of all energy values */
>+    size_t base_size,   /* (i) Index to where energy values should be stored */
>+    int16_t* energyW16, /* (o) Energy in the CB vectors */
>+    int16_t* energyShifts /* (o) Shift value of the energy */
>+    );
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.c
>index f2415febc0d..35d3ce4bc04 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.c
>@@ -16,7 +16,7 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /* Compute the energy of the rest of the cb memory
>  * by step wise adding and subtracting the next
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h
>index af8e658a02b..5511ef1f3f2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h
>@@ -20,14 +20,14 @@
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_MEM_ENERGY_CALC_H_
> 
> void WebRtcIlbcfix_CbMemEnergyCalc(
>-    int32_t energy,   /* (i) input start energy */
>-    size_t range,   /* (i) number of iterations */
>-    int16_t *ppi,   /* (i) input pointer 1 */
>-    int16_t *ppo,   /* (i) input pointer 2 */
>-    int16_t *energyW16,  /* (o) Energy in the CB vectors */
>-    int16_t *energyShifts, /* (o) Shift value of the energy */
>-    int scale,   /* (i) The scaling of all energy values */
>-    size_t base_size  /* (i) Index to where energy values should be stored */
>-                                   );
>+    int32_t energy,        /* (i) input start energy */
>+    size_t range,          /* (i) number of iterations */
>+    int16_t* ppi,          /* (i) input pointer 1 */
>+    int16_t* ppo,          /* (i) input pointer 2 */
>+    int16_t* energyW16,    /* (o) Energy in the CB vectors */
>+    int16_t* energyShifts, /* (o) Shift value of the energy */
>+    int scale,             /* (i) The scaling of all energy values */
>+    size_t base_size /* (i) Index to where energy values should be stored */
>+    );
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_search.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_search.c
>index be949514496..88b2f01f997 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_search.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_search.c
>@@ -16,18 +16,18 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>-#include "gain_quant.h"
>-#include "filtered_cb_vecs.h"
>-#include "constants.h"
>-#include "cb_mem_energy.h"
>-#include "interpolate_samples.h"
>-#include "cb_mem_energy_augmentation.h"
>-#include "cb_search_core.h"
>-#include "energy_inverse.h"
>-#include "augmented_cb_corr.h"
>-#include "cb_update_best_index.h"
>-#include "create_augmented_vec.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
>+#include "modules/audio_coding/codecs/ilbc/gain_quant.h"
>+#include "modules/audio_coding/codecs/ilbc/filtered_cb_vecs.h"
>+#include "modules/audio_coding/codecs/ilbc/constants.h"
>+#include "modules/audio_coding/codecs/ilbc/cb_mem_energy.h"
>+#include "modules/audio_coding/codecs/ilbc/interpolate_samples.h"
>+#include "modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.h"
>+#include "modules/audio_coding/codecs/ilbc/cb_search_core.h"
>+#include "modules/audio_coding/codecs/ilbc/energy_inverse.h"
>+#include "modules/audio_coding/codecs/ilbc/augmented_cb_corr.h"
>+#include "modules/audio_coding/codecs/ilbc/cb_update_best_index.h"
>+#include "modules/audio_coding/codecs/ilbc/create_augmented_vec.h"
> 
> /*----------------------------------------------------------------*
>  *  Search routine for codebook encoding and gain quantization.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_search.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_search.h
>index c8626c5153c..393a2de1263 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_search.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_search.h
>@@ -20,16 +20,16 @@
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_SEARCH_H_
> 
> void WebRtcIlbcfix_CbSearch(
>-    IlbcEncoder *iLBCenc_inst,
>+    IlbcEncoder* iLBCenc_inst,
>     /* (i) the encoder state structure */
>-    int16_t *index,  /* (o) Codebook indices */
>-    int16_t *gain_index, /* (o) Gain quantization indices */
>-    int16_t *intarget, /* (i) Target vector for encoding */
>-    int16_t *decResidual,/* (i) Decoded residual for codebook construction */
>-    size_t lMem,  /* (i) Length of buffer */
>-    size_t lTarget,  /* (i) Length of vector */
>-    int16_t *weightDenum,/* (i) weighting filter coefficients in Q12 */
>-    size_t block  /* (i) the subblock number */
>-                            );
>+    int16_t* index,       /* (o) Codebook indices */
>+    int16_t* gain_index,  /* (o) Gain quantization indices */
>+    int16_t* intarget,    /* (i) Target vector for encoding */
>+    int16_t* decResidual, /* (i) Decoded residual for codebook construction */
>+    size_t lMem,          /* (i) Length of buffer */
>+    size_t lTarget,       /* (i) Length of vector */
>+    int16_t* weightDenum, /* (i) weighting filter coefficients in Q12 */
>+    size_t block          /* (i) the subblock number */
>+    );
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_search_core.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_search_core.c
>index fafa39f69bc..09d26d35d9a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_search_core.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_search_core.c
>@@ -16,8 +16,8 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>-#include "constants.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
>+#include "modules/audio_coding/codecs/ilbc/constants.h"
> 
> void WebRtcIlbcfix_CbSearchCore(
>     int32_t *cDot,    /* (i) Cross Correlation */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_search_core.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_search_core.h
>index 1db7f70c54f..af5a1db5e0b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_search_core.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_search_core.h
>@@ -19,22 +19,22 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_SEARCH_CORE_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_SEARCH_CORE_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> void WebRtcIlbcfix_CbSearchCore(
>-    int32_t *cDot,    /* (i) Cross Correlation */
>-    size_t range,    /* (i) Search range */
>-    int16_t stage,    /* (i) Stage of this search */
>-    int16_t *inverseEnergy,  /* (i) Inversed energy */
>-    int16_t *inverseEnergyShift, /* (i) Shifts of inversed energy
>+    int32_t* cDot,               /* (i) Cross Correlation */
>+    size_t range,                /* (i) Search range */
>+    int16_t stage,               /* (i) Stage of this search */
>+    int16_t* inverseEnergy,      /* (i) Inversed energy */
>+    int16_t* inverseEnergyShift, /* (i) Shifts of inversed energy
>                                           with the offset 2*16-29 */
>-    int32_t *Crit,    /* (o) The criteria */
>-    size_t *bestIndex,   /* (o) Index that corresponds to
>-                                   maximum criteria (in this
>-                                   vector) */
>-    int32_t *bestCrit,   /* (o) Value of critera for the
>-                                  chosen index */
>-    int16_t *bestCritSh);  /* (o) The domain of the chosen
>-                                    criteria */
>+    int32_t* Crit,               /* (o) The criteria */
>+    size_t* bestIndex,           /* (o) Index that corresponds to
>+                                           maximum criteria (in this
>+                                           vector) */
>+    int32_t* bestCrit, /* (o) Value of critera for the
>+                                chosen index */
>+    int16_t* bestCritSh); /* (o) The domain of the chosen
>+                                   criteria */
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_update_best_index.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_update_best_index.c
>index fc27ea9f6ce..ed20c468e46 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_update_best_index.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_update_best_index.c
>@@ -16,9 +16,9 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>-#include "cb_update_best_index.h"
>-#include "constants.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
>+#include "modules/audio_coding/codecs/ilbc/cb_update_best_index.h"
>+#include "modules/audio_coding/codecs/ilbc/constants.h"
> 
> void WebRtcIlbcfix_CbUpdateBestIndex(
>     int32_t CritNew,    /* (i) New Potentially best Criteria */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_update_best_index.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_update_best_index.h
>index 4929b640c45..3f57d4845fe 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_update_best_index.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/cb_update_best_index.h
>@@ -19,20 +19,20 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_UPDATE_BEST_INDEX_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_UPDATE_BEST_INDEX_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> void WebRtcIlbcfix_CbUpdateBestIndex(
>-    int32_t CritNew,    /* (i) New Potentially best Criteria */
>-    int16_t CritNewSh,   /* (i) Shift value of above Criteria */
>-    size_t IndexNew,   /* (i) Index of new Criteria */
>-    int32_t cDotNew,    /* (i) Cross dot of new index */
>-    int16_t invEnergyNew,  /* (i) Inversed energy new index */
>-    int16_t energyShiftNew,  /* (i) Energy shifts of new index */
>-    int32_t *CritMax,   /* (i/o) Maximum Criteria (so far) */
>-    int16_t *shTotMax,   /* (i/o) Shifts of maximum criteria */
>-    size_t *bestIndex,   /* (i/o) Index that corresponds to
>-                                   maximum criteria */
>-    int16_t *bestGain);   /* (i/o) Gain in Q14 that corresponds
>-                                   to maximum criteria */
>+    int32_t CritNew,        /* (i) New Potentially best Criteria */
>+    int16_t CritNewSh,      /* (i) Shift value of above Criteria */
>+    size_t IndexNew,        /* (i) Index of new Criteria */
>+    int32_t cDotNew,        /* (i) Cross dot of new index */
>+    int16_t invEnergyNew,   /* (i) Inversed energy new index */
>+    int16_t energyShiftNew, /* (i) Energy shifts of new index */
>+    int32_t* CritMax,       /* (i/o) Maximum Criteria (so far) */
>+    int16_t* shTotMax,      /* (i/o) Shifts of maximum criteria */
>+    size_t* bestIndex,      /* (i/o) Index that corresponds to
>+                                      maximum criteria */
>+    int16_t* bestGain);     /* (i/o) Gain in Q14 that corresponds
>+                                     to maximum criteria */
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/chebyshev.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/chebyshev.c
>index 21a8f40df6f..38a3069d9e0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/chebyshev.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/chebyshev.c
>@@ -16,8 +16,8 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>-#include "constants.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
>+#include "modules/audio_coding/codecs/ilbc/constants.h"
> 
> /*------------------------------------------------------------------*
>  *  Calculate the Chevyshev polynomial series
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/chebyshev.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/chebyshev.h
>index f990d0b9dac..64b2f490796 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/chebyshev.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/chebyshev.h
>@@ -19,7 +19,7 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CHEBYSHEV_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CHEBYSHEV_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*------------------------------------------------------------------*
>  *  Calculate the Chevyshev polynomial series
>@@ -30,8 +30,8 @@
> 
> int16_t WebRtcIlbcfix_Chebyshev(
>     /* (o) Result of C(x) */
>-    int16_t x,  /* (i) Value to the Chevyshev polynomial */
>-    int16_t *f  /* (i) The coefficients in the polynomial */
>-                                      );
>+    int16_t x, /* (i) Value to the Chevyshev polynomial */
>+    int16_t* f /* (i) The coefficients in the polynomial */
>+    );
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/comp_corr.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/comp_corr.c
>index 7653cb0c25d..b43f2fc10fb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/comp_corr.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/comp_corr.c
>@@ -16,7 +16,7 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  *  Compute cross correlation and pitch gain for pitch prediction
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/comp_corr.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/comp_corr.h
>index 6f6c5ec11c2..1e6b296eabe 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/comp_corr.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/comp_corr.h
>@@ -19,21 +19,20 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_COMP_CORR_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_COMP_CORR_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  *  Compute cross correlation and pitch gain for pitch prediction
>  *  of last subframe at given lag.
>  *---------------------------------------------------------------*/
> 
>-void WebRtcIlbcfix_CompCorr(
>-    int32_t *corr, /* (o) cross correlation */
>-    int32_t *ener, /* (o) energy */
>-    int16_t *buffer, /* (i) signal buffer */
>-    size_t lag,  /* (i) pitch lag */
>-    size_t bLen, /* (i) length of buffer */
>-    size_t sRange, /* (i) correlation search length */
>-    int16_t scale /* (i) number of rightshifts to use */
>+void WebRtcIlbcfix_CompCorr(int32_t* corr,   /* (o) cross correlation */
>+                            int32_t* ener,   /* (o) energy */
>+                            int16_t* buffer, /* (i) signal buffer */
>+                            size_t lag,      /* (i) pitch lag */
>+                            size_t bLen,     /* (i) length of buffer */
>+                            size_t sRange,   /* (i) correlation search length */
>+                            int16_t scale /* (i) number of rightshifts to use */
>                             );
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/constants.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/constants.c
>index 9e341942e66..8efa6aea9cc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/constants.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/constants.c
>@@ -16,8 +16,8 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>-#include "constants.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
>+#include "modules/audio_coding/codecs/ilbc/constants.h"
> 
> /* HP Filters {b[0] b[1] b[2] -a[1] -a[2]} */
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/constants.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/constants.h
>index 506b588f6a3..07369a39e0a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/constants.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/constants.h
>@@ -19,8 +19,7 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CONSTANTS_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CONSTANTS_H_
> 
>-#include "defines.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /* high pass filters */
> 
>@@ -79,7 +78,8 @@ extern const int16_t WebRtcIlbcfix_kAlpha[];
> 
> /* enhancer definitions */
> 
>-extern const int16_t WebRtcIlbcfix_kEnhPolyPhaser[ENH_UPS0][ENH_FLO_MULT2_PLUS1];
>+extern const int16_t WebRtcIlbcfix_kEnhPolyPhaser[ENH_UPS0]
>+                                                 [ENH_FLO_MULT2_PLUS1];
> extern const int16_t WebRtcIlbcfix_kEnhWt[];
> extern const size_t WebRtcIlbcfix_kEnhPlocs[];
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/create_augmented_vec.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/create_augmented_vec.c
>index adc428df7bf..6a4d058eb74 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/create_augmented_vec.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/create_augmented_vec.c
>@@ -16,8 +16,8 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>-#include "constants.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
>+#include "modules/audio_coding/codecs/ilbc/constants.h"
> #include "rtc_base/sanitizer.h"
> #include "common_audio/signal_processing/include/signal_processing_library.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/create_augmented_vec.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/create_augmented_vec.h
>index 0616e2e0b22..28c940006e1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/create_augmented_vec.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/create_augmented_vec.h
>@@ -19,7 +19,7 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CREATE_AUGMENTED_VEC_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CREATE_AUGMENTED_VEC_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  *  Recreate a specific codebook vector from the augmented part.
>@@ -27,8 +27,8 @@
>  *----------------------------------------------------------------*/
> 
> void WebRtcIlbcfix_CreateAugmentedVec(
>-    size_t index,          /* (i) Index for the augmented vector to be
>-                              created */
>+    size_t index, /* (i) Index for the augmented vector to be
>+                     created */
>     const int16_t* buffer, /* (i) Pointer to the end of the codebook memory
>                               that is used for creation of the augmented
>                               codebook */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/decode.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/decode.c
>index a6e045dc675..3848bc7d7b2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/decode.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/decode.c
>@@ -16,21 +16,24 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>-#include "simple_lsf_dequant.h"
>-#include "decoder_interpolate_lsf.h"
>-#include "index_conv_dec.h"
>-#include "do_plc.h"
>-#include "constants.h"
>-#include "enhancer_interface.h"
>-#include "xcorr_coef.h"
>-#include "lsf_check.h"
>-#include "decode_residual.h"
>-#include "unpack_bits.h"
>-#include "hp_output.h"
>-#include "init_decode.h"
>+// Defines WEBRTC_ARCH_BIG_ENDIAN, used below.
>+#include "rtc_base/system/arch.h"
>+
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
>+#include "modules/audio_coding/codecs/ilbc/simple_lsf_dequant.h"
>+#include "modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.h"
>+#include "modules/audio_coding/codecs/ilbc/index_conv_dec.h"
>+#include "modules/audio_coding/codecs/ilbc/do_plc.h"
>+#include "modules/audio_coding/codecs/ilbc/constants.h"
>+#include "modules/audio_coding/codecs/ilbc/enhancer_interface.h"
>+#include "modules/audio_coding/codecs/ilbc/xcorr_coef.h"
>+#include "modules/audio_coding/codecs/ilbc/lsf_check.h"
>+#include "modules/audio_coding/codecs/ilbc/decode_residual.h"
>+#include "modules/audio_coding/codecs/ilbc/unpack_bits.h"
>+#include "modules/audio_coding/codecs/ilbc/hp_output.h"
>+#include "modules/audio_coding/codecs/ilbc/init_decode.h"
> #ifndef WEBRTC_ARCH_BIG_ENDIAN
>-#include "swap_bytes.h"
>+#include "modules/audio_coding/codecs/ilbc/swap_bytes.h"
> #endif
> 
> /*----------------------------------------------------------------*
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/decode.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/decode.h
>index b35be969b99..d73f79880b6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/decode.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/decode.h
>@@ -19,7 +19,10 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DECODE_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DECODE_H_
> 
>-#include "defines.h"
>+#include <stdint.h>
>+
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
>+#include "rtc_base/system/unused.h"
> 
> /*----------------------------------------------------------------*
>  *  main decoder function
>@@ -31,8 +34,8 @@ int WebRtcIlbcfix_DecodeImpl(
>     const uint16_t* bytes,     /* (i) encoded signal bits */
>     IlbcDecoder* iLBCdec_inst, /* (i/o) the decoder state
>                                            structure */
>-    int16_t mode               /* (i) 0: bad packet, PLC,
>-                                      1: normal */
>+    int16_t mode /* (i) 0: bad packet, PLC,
>+                        1: normal */
>     ) RTC_WARN_UNUSED_RESULT;
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/decode_residual.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/decode_residual.c
>index dd5c353cb53..3c113aeb25a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/decode_residual.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/decode_residual.c
>@@ -16,19 +16,19 @@
> 
> ******************************************************************/
> 
>-#include "decode_residual.h"
>+#include "modules/audio_coding/codecs/ilbc/decode_residual.h"
> 
> #include <string.h>
> 
>-#include "defines.h"
>-#include "state_construct.h"
>-#include "cb_construct.h"
>-#include "index_conv_dec.h"
>-#include "do_plc.h"
>-#include "constants.h"
>-#include "enhancer_interface.h"
>-#include "xcorr_coef.h"
>-#include "lsf_check.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
>+#include "modules/audio_coding/codecs/ilbc/state_construct.h"
>+#include "modules/audio_coding/codecs/ilbc/cb_construct.h"
>+#include "modules/audio_coding/codecs/ilbc/index_conv_dec.h"
>+#include "modules/audio_coding/codecs/ilbc/do_plc.h"
>+#include "modules/audio_coding/codecs/ilbc/constants.h"
>+#include "modules/audio_coding/codecs/ilbc/enhancer_interface.h"
>+#include "modules/audio_coding/codecs/ilbc/xcorr_coef.h"
>+#include "modules/audio_coding/codecs/ilbc/lsf_check.h"
> 
> /*----------------------------------------------------------------*
>  *  frame residual decoder function (subrutine to iLBC_decode)
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/decode_residual.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/decode_residual.h
>index 048671e0d6c..d54aada29a9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/decode_residual.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/decode_residual.h
>@@ -20,7 +20,10 @@
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DECODE_RESIDUAL_H_
> 
> #include <stdbool.h>
>-#include "defines.h"
>+#include <stdint.h>
>+
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
>+#include "rtc_base/system/unused.h"
> 
> /*----------------------------------------------------------------*
>  *  frame residual decoder function (subrutine to iLBC_decode)
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.c
>index 06ab2e7190a..8413a73766e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.c
>@@ -16,10 +16,10 @@
> 
> ******************************************************************/
> 
>-#include "lsf_interpolate_to_poly_dec.h"
>-#include "bw_expand.h"
>-#include "defines.h"
>-#include "constants.h"
>+#include "modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.h"
>+#include "modules/audio_coding/codecs/ilbc/bw_expand.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
>+#include "modules/audio_coding/codecs/ilbc/constants.h"
> 
> /*----------------------------------------------------------------*
>  *  obtain synthesis and weighting filters form lsf coefficients
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.h
>index d83ad475387..48d43ec2fa1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.h
>@@ -19,20 +19,20 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DECODER_INTERPOLATE_LSF_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DECODER_INTERPOLATE_LSF_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  *  obtain synthesis and weighting filters form lsf coefficients
>  *---------------------------------------------------------------*/
> 
> void WebRtcIlbcfix_DecoderInterpolateLsp(
>-    int16_t *syntdenum,  /* (o) synthesis filter coefficients */
>-    int16_t *weightdenum, /* (o) weighting denumerator
>+    int16_t* syntdenum, /* (o) synthesis filter coefficients */
>+    int16_t* weightdenum, /* (o) weighting denumerator
>                                    coefficients */
>-    int16_t *lsfdeq,   /* (i) dequantized lsf coefficients */
>-    int16_t length,   /* (i) length of lsf coefficient vector */
>-    IlbcDecoder *iLBCdec_inst
>+    int16_t* lsfdeq, /* (i) dequantized lsf coefficients */
>+    int16_t length,  /* (i) length of lsf coefficient vector */
>+    IlbcDecoder* iLBCdec_inst
>     /* (i) the decoder state structure */
>-                                          );
>+    );
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/defines.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/defines.h
>index 604e337aa68..43948a20c2b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/defines.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/defines.h
>@@ -19,108 +19,114 @@
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DEFINES_H_
> 
> #include <string.h>
>-#include "signal_processing_library.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>+
>+#include "common_audio/signal_processing/include/signal_processing_library.h"
> 
> /* general codec settings */
> 
>-#define FS       8000
>-#define BLOCKL_20MS     160
>-#define BLOCKL_30MS     240
>-#define BLOCKL_MAX     240
>-#define NSUB_20MS     4
>-#define NSUB_30MS     6
>-#define NSUB_MAX     6
>-#define NASUB_20MS     2
>-#define NASUB_30MS     4
>-#define NASUB_MAX     4
>-#define SUBL      40
>-#define STATE_LEN     80
>-#define STATE_SHORT_LEN_30MS  58
>-#define STATE_SHORT_LEN_20MS  57
>+#define FS 8000
>+#define BLOCKL_20MS 160
>+#define BLOCKL_30MS 240
>+#define BLOCKL_MAX 240
>+#define NSUB_20MS 4
>+#define NSUB_30MS 6
>+#define NSUB_MAX 6
>+#define NASUB_20MS 2
>+#define NASUB_30MS 4
>+#define NASUB_MAX 4
>+#define SUBL 40
>+#define STATE_LEN 80
>+#define STATE_SHORT_LEN_30MS 58
>+#define STATE_SHORT_LEN_20MS 57
> 
> /* LPC settings */
> 
>-#define LPC_FILTERORDER    10
>-#define LPC_LOOKBACK    60
>-#define LPC_N_20MS     1
>-#define LPC_N_30MS     2
>-#define LPC_N_MAX     2
>-#define LPC_ASYMDIFF    20
>-#define LSF_NSPLIT     3
>-#define LSF_NUMBER_OF_STEPS   4
>-#define LPC_HALFORDER    5
>+#define LPC_FILTERORDER 10
>+#define LPC_LOOKBACK 60
>+#define LPC_N_20MS 1
>+#define LPC_N_30MS 2
>+#define LPC_N_MAX 2
>+#define LPC_ASYMDIFF 20
>+#define LSF_NSPLIT 3
>+#define LSF_NUMBER_OF_STEPS 4
>+#define LPC_HALFORDER 5
> #define COS_GRID_POINTS 60
> 
> /* cb settings */
> 
>-#define CB_NSTAGES     3
>-#define CB_EXPAND     2
>-#define CB_MEML      147
>-#define CB_FILTERLEN    (2*4)
>-#define CB_HALFFILTERLEN   4
>-#define CB_RESRANGE     34
>-#define CB_MAXGAIN_FIXQ6   83 /* error = -0.24% */
>-#define CB_MAXGAIN_FIXQ14   21299
>+#define CB_NSTAGES 3
>+#define CB_EXPAND 2
>+#define CB_MEML 147
>+#define CB_FILTERLEN (2 * 4)
>+#define CB_HALFFILTERLEN 4
>+#define CB_RESRANGE 34
>+#define CB_MAXGAIN_FIXQ6 83 /* error = -0.24% */
>+#define CB_MAXGAIN_FIXQ14 21299
> 
> /* enhancer */
> 
>-#define ENH_BLOCKL     80  /* block length */
>-#define ENH_BLOCKL_HALF    (ENH_BLOCKL/2)
>-#define ENH_HL      3  /* 2*ENH_HL+1 is number blocks
>-                                                                           in said second sequence */
>-#define ENH_SLOP     2  /* max difference estimated and
>-                                                                           correct pitch period */
>-#define ENH_PLOCSL     8  /* pitch-estimates and
>-                                                                           pitch-locations buffer length */
>-#define ENH_OVERHANG    2
>-#define ENH_UPS0     4  /* upsampling rate */
>-#define ENH_FL0      3  /* 2*FLO+1 is the length of each filter */
>-#define ENH_FLO_MULT2_PLUS1   7
>-#define ENH_VECTL     (ENH_BLOCKL+2*ENH_FL0)
>-#define ENH_CORRDIM     (2*ENH_SLOP+1)
>-#define ENH_NBLOCKS     (BLOCKL/ENH_BLOCKL)
>-#define ENH_NBLOCKS_EXTRA   5
>-#define ENH_NBLOCKS_TOT    8 /* ENH_NBLOCKS+ENH_NBLOCKS_EXTRA */
>-#define ENH_BUFL     (ENH_NBLOCKS_TOT)*ENH_BLOCKL
>-#define ENH_BUFL_FILTEROVERHEAD  3
>-#define ENH_A0      819   /* Q14 */
>-#define ENH_A0_MINUS_A0A0DIV4  848256041 /* Q34 */
>-#define ENH_A0DIV2     26843546 /* Q30 */
>+#define ENH_BLOCKL 80 /* block length */
>+#define ENH_BLOCKL_HALF (ENH_BLOCKL / 2)
>+#define ENH_HL                                                         \
>+  3 /* 2*ENH_HL+1 is number blocks                                     \
>+                                                        in said second \
>+       sequence */
>+#define ENH_SLOP                    \
>+  2 /* max difference estimated and \
>+                                                       correct pitch period */
>+#define ENH_PLOCSL                                                          \
>+  8 /* pitch-estimates and                                                  \
>+                                                     pitch-locations buffer \
>+       length */
>+#define ENH_OVERHANG 2
>+#define ENH_UPS0 4 /* upsampling rate */
>+#define ENH_FL0 3  /* 2*FLO+1 is the length of each filter */
>+#define ENH_FLO_MULT2_PLUS1 7
>+#define ENH_VECTL (ENH_BLOCKL + 2 * ENH_FL0)
>+#define ENH_CORRDIM (2 * ENH_SLOP + 1)
>+#define ENH_NBLOCKS (BLOCKL / ENH_BLOCKL)
>+#define ENH_NBLOCKS_EXTRA 5
>+#define ENH_NBLOCKS_TOT 8 /* ENH_NBLOCKS+ENH_NBLOCKS_EXTRA */
>+#define ENH_BUFL (ENH_NBLOCKS_TOT) * ENH_BLOCKL
>+#define ENH_BUFL_FILTEROVERHEAD 3
>+#define ENH_A0 819                      /* Q14 */
>+#define ENH_A0_MINUS_A0A0DIV4 848256041 /* Q34 */
>+#define ENH_A0DIV2 26843546             /* Q30 */
> 
> /* PLC */
> 
> /* Down sampling */
> 
>-#define FILTERORDER_DS_PLUS1  7
>-#define DELAY_DS     3
>-#define FACTOR_DS     2
>+#define FILTERORDER_DS_PLUS1 7
>+#define DELAY_DS 3
>+#define FACTOR_DS 2
> 
> /* bit stream defs */
> 
>-#define NO_OF_BYTES_20MS   38
>-#define NO_OF_BYTES_30MS   50
>-#define NO_OF_WORDS_20MS   19
>-#define NO_OF_WORDS_30MS   25
>-#define STATE_BITS     3
>-#define BYTE_LEN     8
>-#define ULP_CLASSES     3
>+#define NO_OF_BYTES_20MS 38
>+#define NO_OF_BYTES_30MS 50
>+#define NO_OF_WORDS_20MS 19
>+#define NO_OF_WORDS_30MS 25
>+#define STATE_BITS 3
>+#define BYTE_LEN 8
>+#define ULP_CLASSES 3
> 
> /* help parameters */
> 
>-#define TWO_PI_FIX     25736 /* Q12 */
>+#define TWO_PI_FIX 25736 /* Q12 */
> 
> /* Constants for codebook search and creation */
> 
>-#define ST_MEM_L_TBL  85
>-#define MEM_LF_TBL  147
>-
>+#define ST_MEM_L_TBL 85
>+#define MEM_LF_TBL 147
> 
> /* Struct for the bits */
> typedef struct iLBC_bits_t_ {
>-  int16_t lsf[LSF_NSPLIT*LPC_N_MAX];
>-  int16_t cb_index[CB_NSTAGES*(NASUB_MAX+1)];  /* First CB_NSTAGES values contains extra CB index */
>-  int16_t gain_index[CB_NSTAGES*(NASUB_MAX+1)]; /* First CB_NSTAGES values contains extra CB gain */
>+  int16_t lsf[LSF_NSPLIT * LPC_N_MAX];
>+  int16_t cb_index[CB_NSTAGES * (NASUB_MAX + 1)];   /* First CB_NSTAGES values
>+                                                       contains extra CB index */
>+  int16_t gain_index[CB_NSTAGES * (NASUB_MAX + 1)]; /* First CB_NSTAGES values
>+                                                       contains extra CB gain */
>   size_t idxForMax;
>   int16_t state_first;
>   int16_t idxVec[STATE_SHORT_LEN_30MS];
>@@ -130,7 +136,6 @@ typedef struct iLBC_bits_t_ {
> 
> /* type definition encoder instance */
> typedef struct IlbcEncoder_ {
>-
>   /* flag for frame size mode */
>   int16_t mode;
> 
>@@ -171,7 +176,6 @@ typedef struct IlbcEncoder_ {
> 
> /* type definition decoder instance */
> typedef struct IlbcDecoder_ {
>-
>   /* flag for frame size mode */
>   int16_t mode;
> 
>@@ -198,13 +202,13 @@ typedef struct IlbcDecoder_ {
> 
>   int16_t prevScale, prevPLI;
>   size_t prevLag;
>-  int16_t prevLpc[LPC_FILTERORDER+1];
>-  int16_t prevResidual[NSUB_MAX*SUBL];
>+  int16_t prevLpc[LPC_FILTERORDER + 1];
>+  int16_t prevResidual[NSUB_MAX * SUBL];
>   int16_t seed;
> 
>   /* previous synthesis filter parameters */
> 
>-  int16_t old_syntdenum[(LPC_FILTERORDER + 1)*NSUB_MAX];
>+  int16_t old_syntdenum[(LPC_FILTERORDER + 1) * NSUB_MAX];
> 
>   /* state of output HP filter */
>   int16_t hpimemx[2];
>@@ -212,7 +216,7 @@ typedef struct IlbcDecoder_ {
> 
>   /* enhancer state information */
>   int use_enhancer;
>-  int16_t enh_buf[ENH_BUFL+ENH_BUFL_FILTEROVERHEAD];
>+  int16_t enh_buf[ENH_BUFL + ENH_BUFL_FILTEROVERHEAD];
>   size_t enh_period[ENH_NBLOCKS_TOT];
> 
> } IlbcDecoder;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/do_plc.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/do_plc.c
>index ebddec3f5ee..26ec03f80b8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/do_plc.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/do_plc.c
>@@ -16,10 +16,10 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>-#include "constants.h"
>-#include "comp_corr.h"
>-#include "bw_expand.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
>+#include "modules/audio_coding/codecs/ilbc/constants.h"
>+#include "modules/audio_coding/codecs/ilbc/comp_corr.h"
>+#include "modules/audio_coding/codecs/ilbc/bw_expand.h"
> 
> /*----------------------------------------------------------------*
>  *  Packet loss concealment routine. Conceals a residual signal
>@@ -40,6 +40,7 @@ void WebRtcIlbcfix_DoThePlc(
>   size_t i;
>   int32_t cross, ener, cross_comp, ener_comp = 0;
>   int32_t measure, maxMeasure, energy;
>+  int32_t noise_energy_threshold_30dB;
>   int16_t max, crossSquareMax, crossSquare;
>   size_t j, lag, randlag;
>   int16_t tmp1, tmp2;
>@@ -231,8 +232,8 @@ void WebRtcIlbcfix_DoThePlc(
>     }
> 
>     /* compute concealed residual */
>+    noise_energy_threshold_30dB = (int32_t)iLBCdec_inst->blockl * 900;
>     energy = 0;
>-
>     for (i=0; i<iLBCdec_inst->blockl; i++) {
> 
>       /* noise component -  52 < randlagFIX < 117 */
>@@ -268,15 +269,14 @@ void WebRtcIlbcfix_DoThePlc(
>           ((pitchfact * PLCresidual[i] + (32767 - pitchfact) * randvec[i] +
>               16384) >> 15)) >> 15);
> 
>-      /* Shifting down the result one step extra to ensure that no overflow
>-         will occur */
>-      energy += (PLCresidual[i] * PLCresidual[i]) >>
>-          (iLBCdec_inst->prevScale + 1);
>+      /* Compute energy until threshold for noise energy is reached */
>+      if (energy < noise_energy_threshold_30dB) {
>+        energy += PLCresidual[i] * PLCresidual[i];
>+      }
>     }
> 
>     /* less than 30 dB, use only noise */
>-    if (energy < (WEBRTC_SPL_SHIFT_W32(((int32_t)iLBCdec_inst->blockl*900),-(iLBCdec_inst->prevScale+1)))) {
>-      energy = 0;
>+    if (energy < noise_energy_threshold_30dB) {
>       for (i=0; i<iLBCdec_inst->blockl; i++) {
>         PLCresidual[i] = randvec[i];
>       }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/do_plc.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/do_plc.h
>index 76a598a9c6c..2fbae1de838 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/do_plc.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/do_plc.h
>@@ -19,7 +19,7 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DO_PLC_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DO_PLC_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  *  Packet loss concealment routine. Conceals a residual signal
>@@ -27,15 +27,15 @@
>  *---------------------------------------------------------------*/
> 
> void WebRtcIlbcfix_DoThePlc(
>-    int16_t *PLCresidual,  /* (o) concealed residual */
>-    int16_t *PLClpc,    /* (o) concealed LP parameters */
>-    int16_t PLI,     /* (i) packet loss indicator
>-                                                           0 - no PL, 1 = PL */
>-    int16_t *decresidual,  /* (i) decoded residual */
>-    int16_t *lpc,    /* (i) decoded LPC (only used for no PL) */
>-    size_t inlag,    /* (i) pitch lag */
>-    IlbcDecoder *iLBCdec_inst
>+    int16_t* PLCresidual, /* (o) concealed residual */
>+    int16_t* PLClpc,      /* (o) concealed LP parameters */
>+    int16_t PLI,          /* (i) packet loss indicator
>+                                                                0 - no PL, 1 = PL */
>+    int16_t* decresidual, /* (i) decoded residual */
>+    int16_t* lpc,         /* (i) decoded LPC (only used for no PL) */
>+    size_t inlag,         /* (i) pitch lag */
>+    IlbcDecoder* iLBCdec_inst
>     /* (i/o) decoder instance */
>-                            );
>+    );
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/encode.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/encode.c
>index 72543555be3..912e23ca8b0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/encode.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/encode.c
>@@ -18,25 +18,28 @@
> 
> #include <string.h>
> 
>-#include "defines.h"
>-#include "lpc_encode.h"
>-#include "frame_classify.h"
>-#include "state_search.h"
>-#include "state_construct.h"
>-#include "constants.h"
>-#include "cb_search.h"
>-#include "cb_construct.h"
>-#include "index_conv_enc.h"
>-#include "pack_bits.h"
>-#include "hp_input.h"
>+// Defines WEBRTC_ARCH_BIG_ENDIAN, used below.
>+#include "rtc_base/system/arch.h"
>+
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
>+#include "modules/audio_coding/codecs/ilbc/lpc_encode.h"
>+#include "modules/audio_coding/codecs/ilbc/frame_classify.h"
>+#include "modules/audio_coding/codecs/ilbc/state_search.h"
>+#include "modules/audio_coding/codecs/ilbc/state_construct.h"
>+#include "modules/audio_coding/codecs/ilbc/constants.h"
>+#include "modules/audio_coding/codecs/ilbc/cb_search.h"
>+#include "modules/audio_coding/codecs/ilbc/cb_construct.h"
>+#include "modules/audio_coding/codecs/ilbc/index_conv_enc.h"
>+#include "modules/audio_coding/codecs/ilbc/pack_bits.h"
>+#include "modules/audio_coding/codecs/ilbc/hp_input.h"
> #include "rtc_base/checks.h"
> 
> #ifdef SPLIT_10MS
>-#include "unpack_bits.h"
>-#include "index_conv_dec.h"
>+#include "modules/audio_coding/codecs/ilbc/unpack_bits.h"
>+#include "modules/audio_coding/codecs/ilbc/index_conv_dec.h"
> #endif
> #ifndef WEBRTC_ARCH_BIG_ENDIAN
>-#include "swap_bytes.h"
>+#include "modules/audio_coding/codecs/ilbc/swap_bytes.h"
> #endif
> 
> /*----------------------------------------------------------------*
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/encode.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/encode.h
>index 0692a345680..db00e2c46e9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/encode.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/encode.h
>@@ -19,17 +19,17 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENCODE_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENCODE_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  *  main encoder function
>  *---------------------------------------------------------------*/
> 
> void WebRtcIlbcfix_EncodeImpl(
>-    uint16_t *bytes,     /* (o) encoded data bits iLBC */
>-    const int16_t *block, /* (i) speech vector to encode */
>-    IlbcEncoder *iLBCenc_inst /* (i/o) the general encoder
>+    uint16_t* bytes,      /* (o) encoded data bits iLBC */
>+    const int16_t* block, /* (i) speech vector to encode */
>+    IlbcEncoder* iLBCenc_inst /* (i/o) the general encoder
>                                            state */
>-                          );
>+    );
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/energy_inverse.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/energy_inverse.c
>index b2bdcfffc3f..7f00254aea5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/energy_inverse.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/energy_inverse.c
>@@ -18,7 +18,7 @@
> 
> /* Inverses the in vector in into Q29 domain */
> 
>-#include "energy_inverse.h"
>+#include "modules/audio_coding/codecs/ilbc/energy_inverse.h"
> 
> void WebRtcIlbcfix_EnergyInverse(
>     int16_t *energy,    /* (i/o) Energy and inverse
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/energy_inverse.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/energy_inverse.h
>index cf5f88be578..359a9e20123 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/energy_inverse.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/energy_inverse.h
>@@ -19,14 +19,15 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENERGY_INVERSE_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENERGY_INVERSE_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /* Inverses the in vector in into Q29 domain */
> 
> void WebRtcIlbcfix_EnergyInverse(
>-    int16_t *energy,     /* (i/o) Energy and inverse
>-                                                                   energy (in Q29) */
>-    size_t noOfEnergies);   /* (i)   The length of the energy
>-                                   vector */
>+    int16_t*
>+        energy, /* (i/o) Energy and inverse
>+                                                          energy (in Q29) */
>+    size_t noOfEnergies); /* (i)   The length of the energy
>+                                 vector */
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/enh_upsample.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/enh_upsample.c
>index 929c630c680..0a5f044dbc1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/enh_upsample.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/enh_upsample.c
>@@ -16,8 +16,8 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>-#include "constants.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
>+#include "modules/audio_coding/codecs/ilbc/constants.h"
> 
> /*----------------------------------------------------------------*
>  * upsample finite array assuming zeros outside bounds
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/enh_upsample.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/enh_upsample.h
>index 62f69234c10..b427eca50a7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/enh_upsample.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/enh_upsample.h
>@@ -19,15 +19,15 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENH_UPSAMPLE_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENH_UPSAMPLE_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  * upsample finite array assuming zeros outside bounds
>  *---------------------------------------------------------------*/
> 
> void WebRtcIlbcfix_EnhUpsample(
>-    int32_t *useq1, /* (o) upsampled output sequence */
>-    int16_t *seq1 /* (i) unupsampled sequence */
>-                                );
>+    int32_t* useq1, /* (o) upsampled output sequence */
>+    int16_t* seq1   /* (i) unupsampled sequence */
>+    );
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/enhancer.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/enhancer.c
>index 521d00441c2..d5cd977ea07 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/enhancer.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/enhancer.c
>@@ -16,10 +16,10 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>-#include "constants.h"
>-#include "get_sync_seq.h"
>-#include "smooth.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
>+#include "modules/audio_coding/codecs/ilbc/constants.h"
>+#include "modules/audio_coding/codecs/ilbc/get_sync_seq.h"
>+#include "modules/audio_coding/codecs/ilbc/smooth.h"
> 
> /*----------------------------------------------------------------*
>  * perform enhancement on idata+centerStartPos through
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/enhancer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/enhancer.h
>index 5b4a30b0c22..1a6131b91a5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/enhancer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/enhancer.h
>@@ -19,7 +19,7 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENHANCER_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENHANCER_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  * perform enhancement on idata+centerStartPos through
>@@ -27,13 +27,13 @@
>  *---------------------------------------------------------------*/
> 
> void WebRtcIlbcfix_Enhancer(
>-    int16_t *odata,   /* (o) smoothed block, dimension blockl */
>-    int16_t *idata,   /* (i) data buffer used for enhancing */
>-    size_t idatal,   /* (i) dimension idata */
>+    int16_t* odata,        /* (o) smoothed block, dimension blockl */
>+    int16_t* idata,        /* (i) data buffer used for enhancing */
>+    size_t idatal,         /* (i) dimension idata */
>     size_t centerStartPos, /* (i) first sample current block within idata */
>-    size_t *period,   /* (i) pitch period array (pitch bward-in time) */
>-    const size_t *plocs,   /* (i) locations where period array values valid */
>-    size_t periodl   /* (i) dimension of period and plocs */
>-                            );
>+    size_t* period,        /* (i) pitch period array (pitch bward-in time) */
>+    const size_t* plocs,   /* (i) locations where period array values valid */
>+    size_t periodl         /* (i) dimension of period and plocs */
>+    );
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.c
>index 1bd4b266bd0..f85df6d7ce4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.c
>@@ -18,11 +18,11 @@
> 
> #include <string.h>
> 
>-#include "defines.h"
>-#include "constants.h"
>-#include "xcorr_coef.h"
>-#include "enhancer.h"
>-#include "hp_output.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
>+#include "modules/audio_coding/codecs/ilbc/constants.h"
>+#include "modules/audio_coding/codecs/ilbc/xcorr_coef.h"
>+#include "modules/audio_coding/codecs/ilbc/enhancer.h"
>+#include "modules/audio_coding/codecs/ilbc/hp_output.h"
> 
> 
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.h
>index 17a0354b7c6..de4571514ff 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/enhancer_interface.h
>@@ -19,16 +19,15 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENHANCER_INTERFACE_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENHANCER_INTERFACE_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  * interface for enhancer
>  *---------------------------------------------------------------*/
> 
> size_t  // (o) Estimated lag in end of in[]
>-    WebRtcIlbcfix_EnhancerInterface(
>-        int16_t* out,                // (o) enhanced signal
>-        const int16_t* in,           // (i) unenhanced signal
>-        IlbcDecoder* iLBCdec_inst);  // (i) buffers etc
>+WebRtcIlbcfix_EnhancerInterface(int16_t* out,       // (o) enhanced signal
>+                                const int16_t* in,  // (i) unenhanced signal
>+                                IlbcDecoder* iLBCdec_inst);  // (i) buffers etc
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.c
>index 04d17a67ef5..4624211912b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.c
>@@ -16,8 +16,8 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>-#include "constants.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
>+#include "modules/audio_coding/codecs/ilbc/constants.h"
> 
> /*----------------------------------------------------------------*
>  *  Construct an additional codebook vector by filtering the
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.h
>index 947c462342a..c51ac39a437 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.h
>@@ -19,7 +19,7 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_FILTERED_CB_VECS_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_FILTERED_CB_VECS_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  *  Construct an additional codebook vector by filtering the
>@@ -28,11 +28,11 @@
>  *---------------------------------------------------------------*/
> 
> void WebRtcIlbcfix_FilteredCbVecs(
>-    int16_t *cbvectors, /* (o) Codebook vector for the higher section */
>-    int16_t *CBmem,  /* (i) Codebook memory that is filtered to create a
>-                                           second CB section */
>-    size_t lMem,  /* (i) Length of codebook memory */
>-    size_t samples    /* (i) Number of samples to filter */
>-                                  );
>+    int16_t* cbvectors, /* (o) Codebook vector for the higher section */
>+    int16_t* CBmem,     /* (i) Codebook memory that is filtered to create a
>+                                              second CB section */
>+    size_t lMem,        /* (i) Length of codebook memory */
>+    size_t samples      /* (i) Number of samples to filter */
>+    );
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/frame_classify.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/frame_classify.c
>index 48332808e4e..6edf921650b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/frame_classify.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/frame_classify.c
>@@ -16,8 +16,8 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>-#include "constants.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
>+#include "modules/audio_coding/codecs/ilbc/constants.h"
> 
> /*----------------------------------------------------------------*
>  *  Classification of subframes to localize start state
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/frame_classify.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/frame_classify.h
>index 60b32495059..43c6e57a5da 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/frame_classify.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/frame_classify.h
>@@ -21,9 +21,9 @@
> 
> size_t WebRtcIlbcfix_FrameClassify(
>     /* (o) Index to the max-energy sub frame */
>-    IlbcEncoder *iLBCenc_inst,
>+    IlbcEncoder* iLBCenc_inst,
>     /* (i/o) the encoder state structure */
>-    int16_t *residualFIX /* (i) lpc residual signal */
>-                                                );
>+    int16_t* residualFIX /* (i) lpc residual signal */
>+    );
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/gain_dequant.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/gain_dequant.c
>index 07e09bb887a..cb405aebb3b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/gain_dequant.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/gain_dequant.c
>@@ -16,8 +16,8 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>-#include "constants.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
>+#include "modules/audio_coding/codecs/ilbc/constants.h"
> 
> /*----------------------------------------------------------------*
>  *  decoder for quantized gains in the gain-shape coding of
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/gain_dequant.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/gain_dequant.h
>index efa3c7a32eb..86cc7870708 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/gain_dequant.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/gain_dequant.h
>@@ -19,7 +19,7 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GAIN_DEQUANT_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GAIN_DEQUANT_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  *  decoder for quantized gains in the gain-shape coding of
>@@ -30,7 +30,7 @@ int16_t WebRtcIlbcfix_GainDequant(
>     /* (o) quantized gain value (Q14) */
>     int16_t index, /* (i) quantization index */
>     int16_t maxIn, /* (i) maximum of unquantized gain (Q14) */
>-    int16_t stage /* (i) The stage of the search */
>-                                         );
>+    int16_t stage  /* (i) The stage of the search */
>+    );
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/gain_quant.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/gain_quant.c
>index 90b4114d3b6..2472239ad1e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/gain_quant.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/gain_quant.c
>@@ -16,8 +16,8 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>-#include "constants.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
>+#include "modules/audio_coding/codecs/ilbc/constants.h"
> 
> /*----------------------------------------------------------------*
>  *  quantizer for the gain in the gain-shape coding of residual
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/gain_quant.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/gain_quant.h
>index 5780bee366a..51c0bc9a5ca 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/gain_quant.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/gain_quant.h
>@@ -19,17 +19,18 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GAIN_QUANT_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GAIN_QUANT_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  *  quantizer for the gain in the gain-shape coding of residual
>  *---------------------------------------------------------------*/
> 
>-int16_t WebRtcIlbcfix_GainQuant( /* (o) quantized gain value */
>-    int16_t gain, /* (i) gain value Q14 */
>-    int16_t maxIn, /* (i) maximum of gain value Q14 */
>-    int16_t stage, /* (i) The stage of the search */
>-    int16_t *index /* (o) quantization index */
>-                                       );
>+int16_t
>+WebRtcIlbcfix_GainQuant(               /* (o) quantized gain value */
>+                        int16_t gain,  /* (i) gain value Q14 */
>+                        int16_t maxIn, /* (i) maximum of gain value Q14 */
>+                        int16_t stage, /* (i) The stage of the search */
>+                        int16_t* index /* (o) quantization index */
>+                        );
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/get_cd_vec.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/get_cd_vec.c
>index 272b49a71c9..d3479ec0bc7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/get_cd_vec.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/get_cd_vec.c
>@@ -16,11 +16,11 @@
> 
> ******************************************************************/
> 
>-#include "get_cd_vec.h"
>+#include "modules/audio_coding/codecs/ilbc/get_cd_vec.h"
> 
>-#include "defines.h"
>-#include "constants.h"
>-#include "create_augmented_vec.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
>+#include "modules/audio_coding/codecs/ilbc/constants.h"
>+#include "modules/audio_coding/codecs/ilbc/create_augmented_vec.h"
> 
> /*----------------------------------------------------------------*
>  *  Construct codebook vector for given index.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/get_cd_vec.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/get_cd_vec.h
>index f16789ec111..647b0634a0e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/get_cd_vec.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/get_cd_vec.h
>@@ -20,8 +20,11 @@
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_CD_VEC_H_
> 
> #include <stdbool.h>
>+#include <stddef.h>
>+#include <stdint.h>
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
>+#include "rtc_base/system/unused.h"
> 
> // Returns true on success, false on failure. In case of failure, the decoder
> // state may be corrupted and needs resetting.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/get_lsp_poly.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/get_lsp_poly.c
>index a8375afb609..ecf57701ab0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/get_lsp_poly.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/get_lsp_poly.c
>@@ -16,7 +16,7 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  * Construct the polynomials F1(z) and F2(z) from the LSP
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/get_lsp_poly.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/get_lsp_poly.h
>index 537b7715061..d469409af99 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/get_lsp_poly.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/get_lsp_poly.h
>@@ -19,7 +19,7 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_LSP_POLY_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_LSP_POLY_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  * Construct the polynomials F1(z) and F2(z) from the LSP
>@@ -40,8 +40,7 @@
>  * }
>  *---------------------------------------------------------------*/
> 
>-void WebRtcIlbcfix_GetLspPoly(
>-    int16_t *lsp, /* (i) LSP in Q15 */
>-    int32_t *f);  /* (o) polonymial in Q24 */
>+void WebRtcIlbcfix_GetLspPoly(int16_t* lsp, /* (i) LSP in Q15 */
>+                              int32_t* f);  /* (o) polonymial in Q24 */
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/get_sync_seq.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/get_sync_seq.c
>index a98a96cdf10..c5b11f14dd0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/get_sync_seq.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/get_sync_seq.c
>@@ -16,10 +16,10 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>-#include "constants.h"
>-#include "refiner.h"
>-#include "nearest_neighbor.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
>+#include "modules/audio_coding/codecs/ilbc/constants.h"
>+#include "modules/audio_coding/codecs/ilbc/refiner.h"
>+#include "modules/audio_coding/codecs/ilbc/nearest_neighbor.h"
> 
> /*----------------------------------------------------------------*
>  * get the pitch-synchronous sample sequence
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/get_sync_seq.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/get_sync_seq.h
>index 9e97e60359f..2281b06726c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/get_sync_seq.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/get_sync_seq.h
>@@ -19,22 +19,22 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_SYNC_SEQ_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_SYNC_SEQ_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  * get the pitch-synchronous sample sequence
>  *---------------------------------------------------------------*/
> 
> void WebRtcIlbcfix_GetSyncSeq(
>-    int16_t *idata,   /* (i) original data */
>-    size_t idatal,   /* (i) dimension of data */
>+    int16_t* idata,        /* (i) original data */
>+    size_t idatal,         /* (i) dimension of data */
>     size_t centerStartPos, /* (i) where current block starts */
>-    size_t *period,   /* (i) rough-pitch-period array       (Q-2) */
>-    const size_t *plocs, /* (i) where periods of period array are taken (Q-2) */
>-    size_t periodl,   /* (i) dimension period array */
>-    size_t hl,    /* (i) 2*hl+1 is the number of sequences */
>-    int16_t *surround  /* (i/o) The contribution from this sequence
>-                                summed with earlier contributions */
>-                              );
>+    size_t* period,        /* (i) rough-pitch-period array       (Q-2) */
>+    const size_t* plocs, /* (i) where periods of period array are taken (Q-2) */
>+    size_t periodl,      /* (i) dimension period array */
>+    size_t hl,           /* (i) 2*hl+1 is the number of sequences */
>+    int16_t* surround    /* (i/o) The contribution from this sequence
>+                                  summed with earlier contributions */
>+    );
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/hp_input.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/hp_input.c
>index 5d8a86050e8..dd6e20b8ae6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/hp_input.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/hp_input.c
>@@ -16,7 +16,7 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  *  high-pass filter of input with *0.5 and saturation
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/hp_input.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/hp_input.h
>index fdc0c374978..ac0d26b6db5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/hp_input.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/hp_input.h
>@@ -19,16 +19,16 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_HP_INPUT_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_HP_INPUT_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> void WebRtcIlbcfix_HpInput(
>-    int16_t *signal,     /* (i/o) signal vector */
>-    int16_t *ba,      /* (i)   B- and A-coefficients (2:nd order)
>-                                                                   {b[0] b[1] b[2] -a[1] -a[2]} a[0]
>-                                                                   is assumed to be 1.0 */
>-    int16_t *y,      /* (i/o) Filter state yhi[n-1] ylow[n-1]
>-                                                                   yhi[n-2] ylow[n-2] */
>-    int16_t *x,      /* (i/o) Filter state x[n-1] x[n-2] */
>+    int16_t* signal, /* (i/o) signal vector */
>+    int16_t* ba,     /* (i)   B- and A-coefficients (2:nd order)
>+                              {b[0] b[1] b[2] -a[1] -a[2]}
>+                              a[0] is assumed to be 1.0 */
>+    int16_t* y,      /* (i/o) Filter state yhi[n-1] ylow[n-1]
>+                              yhi[n-2] ylow[n-2] */
>+    int16_t* x,      /* (i/o) Filter state x[n-1] x[n-2] */
>     size_t len);     /* (i)   Number of samples to filter */
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/hp_output.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/hp_output.c
>index 8b18c047b93..0628e58746d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/hp_output.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/hp_output.c
>@@ -16,7 +16,7 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  *  high-pass filter of output and *2 with saturation
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/hp_output.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/hp_output.h
>index 6f6a56dd194..88ecdb5ba18 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/hp_output.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/hp_output.h
>@@ -19,16 +19,16 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_HP_OUTPUT_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_HP_OUTPUT_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> void WebRtcIlbcfix_HpOutput(
>-    int16_t *signal,     /* (i/o) signal vector */
>-    int16_t *ba,      /* (i)   B- and A-coefficients (2:nd order)
>-                               {b[0] b[1] b[2] -a[1] -a[2]} a[0]
>-                               is assumed to be 1.0 */
>-    int16_t *y,      /* (i/o) Filter state yhi[n-1] ylow[n-1]
>+    int16_t* signal, /* (i/o) signal vector */
>+    int16_t* ba,     /* (i)   B- and A-coefficients (2:nd order)
>+                              {b[0] b[1] b[2] -a[1] -a[2]} a[0]
>+                              is assumed to be 1.0 */
>+    int16_t* y,      /* (i/o) Filter state yhi[n-1] ylow[n-1]
>                               yhi[n-2] ylow[n-2] */
>-    int16_t *x,      /* (i/o) Filter state x[n-1] x[n-2] */
>-    size_t len);      /* (i)   Number of samples to filter */
>+    int16_t* x,      /* (i/o) Filter state x[n-1] x[n-2] */
>+    size_t len);     /* (i)   Number of samples to filter */
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/ilbc.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/ilbc.c
>index 08677d9b854..21fc3f8ad0d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/ilbc.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/ilbc.c
>@@ -16,15 +16,16 @@
> 
> ******************************************************************/
> 
>-#include "ilbc.h"
>-#include "defines.h"
>-#include "init_encode.h"
>-#include "encode.h"
>-#include "init_decode.h"
>-#include "decode.h"
>-#include "rtc_base/checks.h"
> #include <stdlib.h>
> 
>+#include "modules/audio_coding/codecs/ilbc/ilbc.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
>+#include "modules/audio_coding/codecs/ilbc/init_encode.h"
>+#include "modules/audio_coding/codecs/ilbc/encode.h"
>+#include "modules/audio_coding/codecs/ilbc/init_decode.h"
>+#include "modules/audio_coding/codecs/ilbc/decode.h"
>+#include "rtc_base/checks.h"
>+
> int16_t WebRtcIlbcfix_EncoderAssign(IlbcEncoderInstance** iLBC_encinst,
>                                     int16_t* ILBCENC_inst_Addr,
>                                     int16_t* size) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/ilbc.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/ilbc.h
>index 78364890827..de8cfde1110 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/ilbc.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/ilbc.h
>@@ -19,12 +19,7 @@
> #define MODULES_AUDIO_CODING_CODECS_ILBC_ILBC_H_
> 
> #include <stddef.h>
>-
>-/*
>- * Define the fixpoint numeric formats
>- */
>-
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include <stdint.h>
> 
> /*
>  * Solution to support multiple instances
>@@ -40,216 +35,214 @@ typedef struct iLBC_decinst_t_ IlbcDecoderInstance;
>  */
> 
> #define ILBC_SPEECH 1
>-#define ILBC_CNG  2
>+#define ILBC_CNG 2
> 
> #ifdef __cplusplus
> extern "C" {
> #endif
> 
>-  /****************************************************************************
>-   * WebRtcIlbcfix_XxxAssign(...)
>-   *
>-   * These functions assigns the encoder/decoder instance to the specified
>-   * memory location
>-   *
>-   * Input:
>-   *     - XXX_xxxinst       : Pointer to created instance that should be
>-   *                           assigned
>-   *     - ILBCXXX_inst_Addr : Pointer to the desired memory space
>-   *     - size              : The size that this structure occupies (in Word16)
>-   *
>-   * Return value             :  0 - Ok
>-   *                            -1 - Error
>-   */
>-
>-  int16_t WebRtcIlbcfix_EncoderAssign(IlbcEncoderInstance **iLBC_encinst,
>-                                      int16_t *ILBCENC_inst_Addr,
>-                                      int16_t *size);
>-  int16_t WebRtcIlbcfix_DecoderAssign(IlbcDecoderInstance **iLBC_decinst,
>-                                      int16_t *ILBCDEC_inst_Addr,
>-                                      int16_t *size);
>+/****************************************************************************
>+ * WebRtcIlbcfix_XxxAssign(...)
>+ *
>+ * These functions assigns the encoder/decoder instance to the specified
>+ * memory location
>+ *
>+ * Input:
>+ *     - XXX_xxxinst       : Pointer to created instance that should be
>+ *                           assigned
>+ *     - ILBCXXX_inst_Addr : Pointer to the desired memory space
>+ *     - size              : The size that this structure occupies (in Word16)
>+ *
>+ * Return value             :  0 - Ok
>+ *                            -1 - Error
>+ */
> 
>+int16_t WebRtcIlbcfix_EncoderAssign(IlbcEncoderInstance** iLBC_encinst,
>+                                    int16_t* ILBCENC_inst_Addr,
>+                                    int16_t* size);
>+int16_t WebRtcIlbcfix_DecoderAssign(IlbcDecoderInstance** iLBC_decinst,
>+                                    int16_t* ILBCDEC_inst_Addr,
>+                                    int16_t* size);
> 
>-  /****************************************************************************
>-   * WebRtcIlbcfix_XxxAssign(...)
>-   *
>-   * These functions create a instance to the specified structure
>-   *
>-   * Input:
>-   *      - XXX_inst        : Pointer to created instance that should be created
>-   *
>-   * Return value           :  0 - Ok
>-   *                          -1 - Error
>-   */
>+/****************************************************************************
>+ * WebRtcIlbcfix_XxxAssign(...)
>+ *
>+ * These functions create a instance to the specified structure
>+ *
>+ * Input:
>+ *      - XXX_inst        : Pointer to created instance that should be created
>+ *
>+ * Return value           :  0 - Ok
>+ *                          -1 - Error
>+ */
> 
>-  int16_t WebRtcIlbcfix_EncoderCreate(IlbcEncoderInstance **iLBC_encinst);
>-  int16_t WebRtcIlbcfix_DecoderCreate(IlbcDecoderInstance **iLBC_decinst);
>+int16_t WebRtcIlbcfix_EncoderCreate(IlbcEncoderInstance** iLBC_encinst);
>+int16_t WebRtcIlbcfix_DecoderCreate(IlbcDecoderInstance** iLBC_decinst);
> 
>-  /****************************************************************************
>-   * WebRtcIlbcfix_XxxFree(...)
>-   *
>-   * These functions frees the dynamic memory of a specified instance
>-   *
>-   * Input:
>-   *      - XXX_inst          : Pointer to created instance that should be freed
>-   *
>-   * Return value             :  0 - Ok
>-   *                            -1 - Error
>-   */
>+/****************************************************************************
>+ * WebRtcIlbcfix_XxxFree(...)
>+ *
>+ * These functions frees the dynamic memory of a specified instance
>+ *
>+ * Input:
>+ *      - XXX_inst          : Pointer to created instance that should be freed
>+ *
>+ * Return value             :  0 - Ok
>+ *                            -1 - Error
>+ */
> 
>-  int16_t WebRtcIlbcfix_EncoderFree(IlbcEncoderInstance *iLBC_encinst);
>-  int16_t WebRtcIlbcfix_DecoderFree(IlbcDecoderInstance *iLBC_decinst);
>+int16_t WebRtcIlbcfix_EncoderFree(IlbcEncoderInstance* iLBC_encinst);
>+int16_t WebRtcIlbcfix_DecoderFree(IlbcDecoderInstance* iLBC_decinst);
> 
>+/****************************************************************************
>+ * WebRtcIlbcfix_EncoderInit(...)
>+ *
>+ * This function initializes a iLBC instance
>+ *
>+ * Input:
>+ *      - iLBCenc_inst      : iLBC instance, i.e. the user that should receive
>+ *                            be initialized
>+ *      - frameLen          : The frame length of the codec 20/30 (ms)
>+ *
>+ * Return value             :  0 - Ok
>+ *                            -1 - Error
>+ */
> 
>-  /****************************************************************************
>-   * WebRtcIlbcfix_EncoderInit(...)
>-   *
>-   * This function initializes a iLBC instance
>-   *
>-   * Input:
>-   *      - iLBCenc_inst      : iLBC instance, i.e. the user that should receive
>-   *                            be initialized
>-   *      - frameLen          : The frame length of the codec 20/30 (ms)
>-   *
>-   * Return value             :  0 - Ok
>-   *                            -1 - Error
>-   */
>+int16_t WebRtcIlbcfix_EncoderInit(IlbcEncoderInstance* iLBCenc_inst,
>+                                  int16_t frameLen);
> 
>-  int16_t WebRtcIlbcfix_EncoderInit(IlbcEncoderInstance *iLBCenc_inst,
>-                                    int16_t frameLen);
>+/****************************************************************************
>+ * WebRtcIlbcfix_Encode(...)
>+ *
>+ * This function encodes one iLBC frame. Input speech length has be a
>+ * multiple of the frame length.
>+ *
>+ * Input:
>+ *      - iLBCenc_inst      : iLBC instance, i.e. the user that should encode
>+ *                            a package
>+ *      - speechIn          : Input speech vector
>+ *      - len               : Samples in speechIn (160, 240, 320 or 480)
>+ *
>+ * Output:
>+ *  - encoded               : The encoded data vector
>+ *
>+ * Return value             : >0 - Length (in bytes) of coded data
>+ *                            -1 - Error
>+ */
> 
>-  /****************************************************************************
>-   * WebRtcIlbcfix_Encode(...)
>-   *
>-   * This function encodes one iLBC frame. Input speech length has be a
>-   * multiple of the frame length.
>-   *
>-   * Input:
>-   *      - iLBCenc_inst      : iLBC instance, i.e. the user that should encode
>-   *                            a package
>-   *      - speechIn          : Input speech vector
>-   *      - len               : Samples in speechIn (160, 240, 320 or 480)
>-   *
>-   * Output:
>-   *  - encoded               : The encoded data vector
>-   *
>-   * Return value             : >0 - Length (in bytes) of coded data
>-   *                            -1 - Error
>-   */
>+int WebRtcIlbcfix_Encode(IlbcEncoderInstance* iLBCenc_inst,
>+                         const int16_t* speechIn,
>+                         size_t len,
>+                         uint8_t* encoded);
> 
>-  int WebRtcIlbcfix_Encode(IlbcEncoderInstance *iLBCenc_inst,
>-                           const int16_t *speechIn,
>-                           size_t len,
>-                           uint8_t* encoded);
>+/****************************************************************************
>+ * WebRtcIlbcfix_DecoderInit(...)
>+ *
>+ * This function initializes a iLBC instance with either 20 or 30 ms frames
>+ * Alternatively the WebRtcIlbcfix_DecoderInit_XXms can be used. Then it's
>+ * not needed to specify the frame length with a variable.
>+ *
>+ * Input:
>+ *      - IlbcDecoderInstance : iLBC decoder instance
>+ *      - frameLen            : The frame length of the codec 20/30 (ms)
>+ *
>+ * Return value               :  0 - Ok
>+ *                              -1 - Error
>+ */
> 
>-  /****************************************************************************
>-   * WebRtcIlbcfix_DecoderInit(...)
>-   *
>-   * This function initializes a iLBC instance with either 20 or 30 ms frames
>-   * Alternatively the WebRtcIlbcfix_DecoderInit_XXms can be used. Then it's
>-   * not needed to specify the frame length with a variable.
>-   *
>-   * Input:
>-   *      - IlbcDecoderInstance : iLBC decoder instance
>-   *      - frameLen            : The frame length of the codec 20/30 (ms)
>-   *
>-   * Return value               :  0 - Ok
>-   *                              -1 - Error
>-   */
>+int16_t WebRtcIlbcfix_DecoderInit(IlbcDecoderInstance* iLBCdec_inst,
>+                                  int16_t frameLen);
>+void WebRtcIlbcfix_DecoderInit20Ms(IlbcDecoderInstance* iLBCdec_inst);
>+void WebRtcIlbcfix_Decoderinit30Ms(IlbcDecoderInstance* iLBCdec_inst);
> 
>-  int16_t WebRtcIlbcfix_DecoderInit(IlbcDecoderInstance *iLBCdec_inst,
>-                                    int16_t frameLen);
>-  void WebRtcIlbcfix_DecoderInit20Ms(IlbcDecoderInstance* iLBCdec_inst);
>-  void WebRtcIlbcfix_Decoderinit30Ms(IlbcDecoderInstance* iLBCdec_inst);
>+/****************************************************************************
>+ * WebRtcIlbcfix_Decode(...)
>+ *
>+ * This function decodes a packet with iLBC frame(s). Output speech length
>+ * will be a multiple of 160 or 240 samples ((160 or 240)*frames/packet).
>+ *
>+ * Input:
>+ *      - iLBCdec_inst      : iLBC instance, i.e. the user that should decode
>+ *                            a packet
>+ *      - encoded           : Encoded iLBC frame(s)
>+ *      - len               : Bytes in encoded vector
>+ *
>+ * Output:
>+ *      - decoded           : The decoded vector
>+ *      - speechType        : 1 normal, 2 CNG
>+ *
>+ * Return value             : >0 - Samples in decoded vector
>+ *                            -1 - Error
>+ */
> 
>-  /****************************************************************************
>-   * WebRtcIlbcfix_Decode(...)
>-   *
>-   * This function decodes a packet with iLBC frame(s). Output speech length
>-   * will be a multiple of 160 or 240 samples ((160 or 240)*frames/packet).
>-   *
>-   * Input:
>-   *      - iLBCdec_inst      : iLBC instance, i.e. the user that should decode
>-   *                            a packet
>-   *      - encoded           : Encoded iLBC frame(s)
>-   *      - len               : Bytes in encoded vector
>-   *
>-   * Output:
>-   *      - decoded           : The decoded vector
>-   *      - speechType        : 1 normal, 2 CNG
>-   *
>-   * Return value             : >0 - Samples in decoded vector
>-   *                            -1 - Error
>-   */
>+int WebRtcIlbcfix_Decode(IlbcDecoderInstance* iLBCdec_inst,
>+                         const uint8_t* encoded,
>+                         size_t len,
>+                         int16_t* decoded,
>+                         int16_t* speechType);
>+int WebRtcIlbcfix_Decode20Ms(IlbcDecoderInstance* iLBCdec_inst,
>+                             const uint8_t* encoded,
>+                             size_t len,
>+                             int16_t* decoded,
>+                             int16_t* speechType);
>+int WebRtcIlbcfix_Decode30Ms(IlbcDecoderInstance* iLBCdec_inst,
>+                             const uint8_t* encoded,
>+                             size_t len,
>+                             int16_t* decoded,
>+                             int16_t* speechType);
>+
>+/****************************************************************************
>+ * WebRtcIlbcfix_DecodePlc(...)
>+ *
>+ * This function conducts PLC for iLBC frame(s). Output speech length
>+ * will be a multiple of 160 or 240 samples.
>+ *
>+ * Input:
>+ *      - iLBCdec_inst      : iLBC instance, i.e. the user that should perform
>+ *                            a PLC
>+ *      - noOfLostFrames    : Number of PLC frames to produce
>+ *
>+ * Output:
>+ *      - decoded           : The "decoded" vector
>+ *
>+ * Return value             : Samples in decoded PLC vector
>+ */
> 
>-  int WebRtcIlbcfix_Decode(IlbcDecoderInstance* iLBCdec_inst,
>-                           const uint8_t* encoded,
>-                           size_t len,
>-                           int16_t* decoded,
>-                           int16_t* speechType);
>-  int WebRtcIlbcfix_Decode20Ms(IlbcDecoderInstance* iLBCdec_inst,
>-                               const uint8_t* encoded,
>-                               size_t len,
>-                               int16_t* decoded,
>-                               int16_t* speechType);
>-  int WebRtcIlbcfix_Decode30Ms(IlbcDecoderInstance* iLBCdec_inst,
>-                               const uint8_t* encoded,
>-                               size_t len,
>+size_t WebRtcIlbcfix_DecodePlc(IlbcDecoderInstance* iLBCdec_inst,
>                                int16_t* decoded,
>-                               int16_t* speechType);
>-
>-  /****************************************************************************
>-   * WebRtcIlbcfix_DecodePlc(...)
>-   *
>-   * This function conducts PLC for iLBC frame(s). Output speech length
>-   * will be a multiple of 160 or 240 samples.
>-   *
>-   * Input:
>-   *      - iLBCdec_inst      : iLBC instance, i.e. the user that should perform
>-   *                            a PLC
>-   *      - noOfLostFrames    : Number of PLC frames to produce
>-   *
>-   * Output:
>-   *      - decoded           : The "decoded" vector
>-   *
>-   * Return value             : Samples in decoded PLC vector
>-   */
>+                               size_t noOfLostFrames);
> 
>-  size_t WebRtcIlbcfix_DecodePlc(IlbcDecoderInstance *iLBCdec_inst,
>-                                 int16_t *decoded,
>-                                 size_t noOfLostFrames);
>-
>-  /****************************************************************************
>-   * WebRtcIlbcfix_NetEqPlc(...)
>-   *
>-   * This function updates the decoder when a packet loss has occured, but it
>-   * does not produce any PLC data. Function can be used if another PLC method
>-   * is used (i.e NetEq).
>-   *
>-   * Input:
>-   *      - iLBCdec_inst      : iLBC instance that should be updated
>-   *      - noOfLostFrames    : Number of lost frames
>-   *
>-   * Output:
>-   *      - decoded           : The "decoded" vector (nothing in this case)
>-   *
>-   * Return value             : Samples in decoded PLC vector
>-   */
>+/****************************************************************************
>+ * WebRtcIlbcfix_NetEqPlc(...)
>+ *
>+ * This function updates the decoder when a packet loss has occured, but it
>+ * does not produce any PLC data. Function can be used if another PLC method
>+ * is used (i.e NetEq).
>+ *
>+ * Input:
>+ *      - iLBCdec_inst      : iLBC instance that should be updated
>+ *      - noOfLostFrames    : Number of lost frames
>+ *
>+ * Output:
>+ *      - decoded           : The "decoded" vector (nothing in this case)
>+ *
>+ * Return value             : Samples in decoded PLC vector
>+ */
> 
>-  size_t WebRtcIlbcfix_NetEqPlc(IlbcDecoderInstance *iLBCdec_inst,
>-                                int16_t *decoded,
>-                                size_t noOfLostFrames);
>+size_t WebRtcIlbcfix_NetEqPlc(IlbcDecoderInstance* iLBCdec_inst,
>+                              int16_t* decoded,
>+                              size_t noOfLostFrames);
> 
>-  /****************************************************************************
>-   * WebRtcIlbcfix_version(...)
>-   *
>-   * This function returns the version number of iLBC
>-   *
>-   * Output:
>-   *      - version           : Version number of iLBC (maximum 20 char)
>-   */
>+/****************************************************************************
>+ * WebRtcIlbcfix_version(...)
>+ *
>+ * This function returns the version number of iLBC
>+ *
>+ * Output:
>+ *      - version           : Version number of iLBC (maximum 20 char)
>+ */
> 
>-  void WebRtcIlbcfix_version(char *version);
>+void WebRtcIlbcfix_version(char* version);
> 
> #ifdef __cplusplus
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/ilbc_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/ilbc_unittest.cc
>index b8d3c7c55a8..5ec1219b3b5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/ilbc_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/ilbc_unittest.cc
>@@ -71,7 +71,7 @@ class SplitIlbcTest : public ::testing::TestWithParam<std::pair<int, int> > {
> TEST_P(SplitIlbcTest, NumFrames) {
>   AudioDecoderIlbcImpl decoder;
>   const size_t frame_length_samples = frame_length_ms_ * 8;
>-  const auto generate_payload = [] (size_t payload_length_bytes) {
>+  const auto generate_payload = [](size_t payload_length_bytes) {
>     rtc::Buffer payload(payload_length_bytes);
>     // Fill payload with increasing integers {0, 1, 2, ...}.
>     for (size_t i = 0; i < payload.size(); ++i) {
>@@ -104,7 +104,8 @@ TEST_P(SplitIlbcTest, NumFrames) {
> // The maximum is defined by the largest payload length that can be uniquely
> // resolved to a frame size of either 38 bytes (20 ms) or 50 bytes (30 ms).
> INSTANTIATE_TEST_CASE_P(
>-    IlbcTest, SplitIlbcTest,
>+    IlbcTest,
>+    SplitIlbcTest,
>     ::testing::Values(std::pair<int, int>(1, 20),  // 1 frame, 20 ms.
>                       std::pair<int, int>(2, 20),  // 2 frames, 20 ms.
>                       std::pair<int, int>(3, 20),  // And so on.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/index_conv_dec.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/index_conv_dec.c
>index 033a0a35847..ad12cee39ca 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/index_conv_dec.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/index_conv_dec.c
>@@ -16,7 +16,7 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> void WebRtcIlbcfix_IndexConvDec(
>     int16_t *index   /* (i/o) Codebook indexes */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/index_conv_dec.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/index_conv_dec.h
>index 4b61d31617b..4f08ce04dfb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/index_conv_dec.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/index_conv_dec.h
>@@ -19,10 +19,9 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INDEX_CONV_DEC_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INDEX_CONV_DEC_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
>-void WebRtcIlbcfix_IndexConvDec(
>-    int16_t *index   /* (i/o) Codebook indexes */
>+void WebRtcIlbcfix_IndexConvDec(int16_t* index /* (i/o) Codebook indexes */
>                                 );
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/index_conv_enc.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/index_conv_enc.c
>index 0e7701b4f69..6cf164e7ee7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/index_conv_enc.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/index_conv_enc.c
>@@ -16,7 +16,7 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> /*----------------------------------------------------------------*
>  *  Convert the codebook indexes to make the search easier
>  *---------------------------------------------------------------*/
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/index_conv_enc.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/index_conv_enc.h
>index 935928a55cd..f89949949cd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/index_conv_enc.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/index_conv_enc.h
>@@ -19,14 +19,13 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INDEX_CONV_ENC_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INDEX_CONV_ENC_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  *  Convert the codebook indexes to make the search easier
>  *---------------------------------------------------------------*/
> 
>-void WebRtcIlbcfix_IndexConvEnc(
>-    int16_t *index   /* (i/o) Codebook indexes */
>+void WebRtcIlbcfix_IndexConvEnc(int16_t* index /* (i/o) Codebook indexes */
>                                 );
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/init_decode.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/init_decode.c
>index 1f92480d9fd..c63cc7ce812 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/init_decode.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/init_decode.c
>@@ -16,8 +16,8 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>-#include "constants.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
>+#include "modules/audio_coding/codecs/ilbc/constants.h"
> 
> /*----------------------------------------------------------------*
>  *  Initiation of decoder instance.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/init_decode.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/init_decode.h
>index 6f3a17becba..fdcf9f0b1ab 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/init_decode.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/init_decode.h
>@@ -19,17 +19,18 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INIT_DECODE_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INIT_DECODE_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  *  Initiation of decoder instance.
>  *---------------------------------------------------------------*/
> 
>-int WebRtcIlbcfix_InitDecode(  /* (o) Number of decoded samples */
>-    IlbcDecoder *iLBCdec_inst, /* (i/o) Decoder instance */
>-    int16_t mode,     /* (i) frame size mode */
>-    int use_enhancer           /* (i) 1 to use enhancer
>-                                  0 to run without enhancer */
>-                                         );
>+int WebRtcIlbcfix_InitDecode(/* (o) Number of decoded samples */
>+                             IlbcDecoder*
>+                                 iLBCdec_inst, /* (i/o) Decoder instance */
>+                             int16_t mode,     /* (i) frame size mode */
>+                             int use_enhancer  /* (i) 1 to use enhancer
>+                                                  0 to run without enhancer */
>+                             );
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/init_encode.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/init_encode.c
>index f559d8441f9..b21f77efa61 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/init_encode.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/init_encode.c
>@@ -16,8 +16,8 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>-#include "constants.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
>+#include "modules/audio_coding/codecs/ilbc/constants.h"
> 
> /*----------------------------------------------------------------*
>  *  Initiation of encoder instance.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/init_encode.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/init_encode.h
>index fc11cc7d20f..f91a9b0add0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/init_encode.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/init_encode.h
>@@ -19,15 +19,16 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INIT_ENCODE_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INIT_ENCODE_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  *  Initiation of encoder instance.
>  *---------------------------------------------------------------*/
> 
>-int WebRtcIlbcfix_InitEncode(  /* (o) Number of bytes encoded */
>-    IlbcEncoder *iLBCenc_inst, /* (i/o) Encoder instance */
>-    int16_t mode     /* (i) frame size mode */
>-                                         );
>+int WebRtcIlbcfix_InitEncode(/* (o) Number of bytes encoded */
>+                             IlbcEncoder*
>+                                 iLBCenc_inst, /* (i/o) Encoder instance */
>+                             int16_t mode      /* (i) frame size mode */
>+                             );
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/interpolate.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/interpolate.c
>index d0869a5dc20..3ce480eaa69 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/interpolate.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/interpolate.c
>@@ -16,8 +16,8 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>-#include "constants.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
>+#include "modules/audio_coding/codecs/ilbc/constants.h"
> 
> /*----------------------------------------------------------------*
>  *  interpolation between vectors
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/interpolate.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/interpolate.h
>index befe2416f49..9f0323649b3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/interpolate.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/interpolate.h
>@@ -19,17 +19,17 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INTERPOLATE_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INTERPOLATE_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  *  interpolation between vectors
>  *---------------------------------------------------------------*/
> 
> void WebRtcIlbcfix_Interpolate(
>-    int16_t *out, /* (o) output vector */
>-    int16_t *in1, /* (i) first input vector */
>-    int16_t *in2, /* (i) second input vector */
>-    int16_t coef, /* (i) weight coefficient in Q14 */
>+    int16_t* out,    /* (o) output vector */
>+    int16_t* in1,    /* (i) first input vector */
>+    int16_t* in2,    /* (i) second input vector */
>+    int16_t coef,    /* (i) weight coefficient in Q14 */
>     int16_t length); /* (i) number of sample is vectors */
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/interpolate_samples.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/interpolate_samples.c
>index 376dbbb668a..9ca38a4a34f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/interpolate_samples.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/interpolate_samples.c
>@@ -16,8 +16,8 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>-#include "constants.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
>+#include "modules/audio_coding/codecs/ilbc/constants.h"
> 
> void WebRtcIlbcfix_InterpolateSamples(
>     int16_t *interpSamples, /* (o) The interpolated samples */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/interpolate_samples.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/interpolate_samples.h
>index 8a2827385f7..264a101aa4e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/interpolate_samples.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/interpolate_samples.h
>@@ -19,16 +19,16 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INTERPOLATE_SAMPLES_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INTERPOLATE_SAMPLES_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  *  Construct the interpolated samples for the Augmented CB
>  *---------------------------------------------------------------*/
> 
> void WebRtcIlbcfix_InterpolateSamples(
>-    int16_t *interpSamples, /* (o) The interpolated samples */
>-    int16_t *CBmem,   /* (i) The CB memory */
>-    size_t lMem    /* (i) Length of the CB memory */
>-                                      );
>+    int16_t* interpSamples, /* (o) The interpolated samples */
>+    int16_t* CBmem,         /* (i) The CB memory */
>+    size_t lMem             /* (i) Length of the CB memory */
>+    );
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lpc_encode.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lpc_encode.c
>index 258b6b15520..9b2a0c0d41a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lpc_encode.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lpc_encode.c
>@@ -16,12 +16,12 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>-#include "simple_lpc_analysis.h"
>-#include "simple_interpolate_lsf.h"
>-#include "simple_lsf_quant.h"
>-#include "lsf_check.h"
>-#include "constants.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
>+#include "modules/audio_coding/codecs/ilbc/simple_lpc_analysis.h"
>+#include "modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.h"
>+#include "modules/audio_coding/codecs/ilbc/simple_lsf_quant.h"
>+#include "modules/audio_coding/codecs/ilbc/lsf_check.h"
>+#include "modules/audio_coding/codecs/ilbc/constants.h"
> 
> /*----------------------------------------------------------------*
>  *  lpc encoder
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lpc_encode.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lpc_encode.h
>index 776b3a14cdd..256fa492bba 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lpc_encode.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lpc_encode.h
>@@ -19,21 +19,21 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LPC_ENCODE_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LPC_ENCODE_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  *  lpc encoder
>  *---------------------------------------------------------------*/
> 
> void WebRtcIlbcfix_LpcEncode(
>-    int16_t *syntdenum,  /* (i/o) synthesis filter coefficients
>-                                  before/after encoding */
>-    int16_t *weightdenum, /* (i/o) weighting denumerator coefficients
>+    int16_t* syntdenum,   /* (i/o) synthesis filter coefficients
>                                    before/after encoding */
>-    int16_t *lsf_index,  /* (o) lsf quantization index */
>-    int16_t *data,   /* (i) Speech to do LPC analysis on */
>-    IlbcEncoder *iLBCenc_inst
>+    int16_t* weightdenum, /* (i/o) weighting denumerator coefficients
>+                                   before/after encoding */
>+    int16_t* lsf_index,   /* (o) lsf quantization index */
>+    int16_t* data,        /* (i) Speech to do LPC analysis on */
>+    IlbcEncoder* iLBCenc_inst
>     /* (i/o) the encoder state structure */
>-                             );
>+    );
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lsf_check.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lsf_check.c
>index 0e3bd8485a8..684b2ce8cec 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lsf_check.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lsf_check.c
>@@ -16,8 +16,8 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>-#include "constants.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
>+#include "modules/audio_coding/codecs/ilbc/constants.h"
> 
> /*----------------------------------------------------------------*
>  *  check for stability of lsf coefficients
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lsf_check.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lsf_check.h
>index bfcea083aa8..d367c1d09fb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lsf_check.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lsf_check.h
>@@ -19,15 +19,14 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_CHECK_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_CHECK_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  *  check for stability of lsf coefficients
>  *---------------------------------------------------------------*/
> 
>-int WebRtcIlbcfix_LsfCheck(
>-    int16_t *lsf, /* LSF parameters */
>-    int dim, /* dimension of LSF */
>-    int NoAn); /* No of analysis per frame */
>+int WebRtcIlbcfix_LsfCheck(int16_t* lsf, /* LSF parameters */
>+                           int dim,      /* dimension of LSF */
>+                           int NoAn);    /* No of analysis per frame */
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.c
>index 66bbde8e7f3..e501f3cb6ee 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.c
>@@ -16,9 +16,9 @@
> 
> ******************************************************************/
> 
>-#include "interpolate.h"
>-#include "lsf_to_poly.h"
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/interpolate.h"
>+#include "modules/audio_coding/codecs/ilbc/lsf_to_poly.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  *  interpolation of lsf coefficients for the decoder
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.h
>index 45305284978..016897a144c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.h
>@@ -19,19 +19,19 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_INTERPOLATE_TO_POLY_DEC_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_INTERPOLATE_TO_POLY_DEC_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  *  interpolation of lsf coefficients for the decoder
>  *---------------------------------------------------------------*/
> 
> void WebRtcIlbcfix_LspInterpolate2PolyDec(
>-    int16_t *a,   /* (o) lpc coefficients Q12 */
>-    int16_t *lsf1,  /* (i) first set of lsf coefficients Q13 */
>-    int16_t *lsf2,  /* (i) second set of lsf coefficients Q13 */
>+    int16_t* a,    /* (o) lpc coefficients Q12 */
>+    int16_t* lsf1, /* (i) first set of lsf coefficients Q13 */
>+    int16_t* lsf2, /* (i) second set of lsf coefficients Q13 */
>     int16_t coef,  /* (i) weighting coefficient to use between
>                                    lsf1 and lsf2 Q14 */
>-    int16_t length  /* (i) length of coefficient vectors */
>-                                          );
>+    int16_t length /* (i) length of coefficient vectors */
>+    );
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.c
>index cf67ecc3be7..a660c4f2ffd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.c
>@@ -16,9 +16,9 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>-#include "interpolate.h"
>-#include "lsf_to_poly.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
>+#include "modules/audio_coding/codecs/ilbc/interpolate.h"
>+#include "modules/audio_coding/codecs/ilbc/lsf_to_poly.h"
> 
> /*----------------------------------------------------------------*
>  *  lsf interpolator and conversion from lsf to a coefficients
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.h
>index a2d2db61a29..9cb0dd9f429 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.h
>@@ -19,7 +19,7 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_INTERPOLATE_TO_POLY_ENC_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_INTERPOLATE_TO_POLY_ENC_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  *  lsf interpolator and conversion from lsf to a coefficients
>@@ -27,12 +27,12 @@
>  *---------------------------------------------------------------*/
> 
> void WebRtcIlbcfix_LsfInterpolate2PloyEnc(
>-    int16_t *a,  /* (o) lpc coefficients Q12 */
>-    int16_t *lsf1, /* (i) first set of lsf coefficients Q13 */
>-    int16_t *lsf2, /* (i) second set of lsf coefficients Q13 */
>-    int16_t coef, /* (i) weighting coefficient to use between
>-                           lsf1 and lsf2 Q14 */
>+    int16_t* a,    /* (o) lpc coefficients Q12 */
>+    int16_t* lsf1, /* (i) first set of lsf coefficients Q13 */
>+    int16_t* lsf2, /* (i) second set of lsf coefficients Q13 */
>+    int16_t coef,  /* (i) weighting coefficient to use between
>+                            lsf1 and lsf2 Q14 */
>     int16_t length /* (i) length of coefficient vectors */
>-                                          );
>+    );
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lsf_to_lsp.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lsf_to_lsp.c
>index cfab013ab7e..8767e2dec22 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lsf_to_lsp.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lsf_to_lsp.c
>@@ -16,8 +16,8 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>-#include "constants.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
>+#include "modules/audio_coding/codecs/ilbc/constants.h"
> 
> /*----------------------------------------------------------------*
>  *  conversion from lsf to lsp coefficients
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lsf_to_lsp.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lsf_to_lsp.h
>index ace29dec00b..921101ac9e2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lsf_to_lsp.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lsf_to_lsp.h
>@@ -19,16 +19,16 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_TO_LSP_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_TO_LSP_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  *  conversion from lsf to lsp coefficients
>  *---------------------------------------------------------------*/
> 
> void WebRtcIlbcfix_Lsf2Lsp(
>-    int16_t *lsf, /* (i) lsf in Q13 values between 0 and pi */
>-    int16_t *lsp, /* (o) lsp in Q15 values between -1 and 1 */
>+    int16_t* lsf, /* (i) lsf in Q13 values between 0 and pi */
>+    int16_t* lsp, /* (o) lsp in Q15 values between -1 and 1 */
>     int16_t m     /* (i) number of coefficients */
>-                           );
>+    );
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lsf_to_poly.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lsf_to_poly.c
>index c3a34cab00e..4dbf96dc0be 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lsf_to_poly.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lsf_to_poly.c
>@@ -16,10 +16,10 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>-#include "lsf_to_lsp.h"
>-#include "get_lsp_poly.h"
>-#include "constants.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
>+#include "modules/audio_coding/codecs/ilbc/lsf_to_lsp.h"
>+#include "modules/audio_coding/codecs/ilbc/get_lsp_poly.h"
>+#include "modules/audio_coding/codecs/ilbc/constants.h"
> 
> void WebRtcIlbcfix_Lsf2Poly(
>     int16_t *a,     /* (o) predictor coefficients (order = 10) in Q12 */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lsf_to_poly.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lsf_to_poly.h
>index 782ef9f6c34..e551836f529 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lsf_to_poly.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lsf_to_poly.h
>@@ -19,15 +19,15 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_TO_POLY_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_TO_POLY_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  *  Convert from LSF coefficients to A coefficients
>  *---------------------------------------------------------------*/
> 
> void WebRtcIlbcfix_Lsf2Poly(
>-    int16_t *a,     /* (o) predictor coefficients (order = 10) in Q12 */
>-    int16_t *lsf    /* (i) line spectral frequencies in Q13 */
>-                            );
>+    int16_t* a,  /* (o) predictor coefficients (order = 10) in Q12 */
>+    int16_t* lsf /* (i) line spectral frequencies in Q13 */
>+    );
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lsp_to_lsf.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lsp_to_lsf.c
>index be95de7a2f7..db11cfed2e9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lsp_to_lsf.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lsp_to_lsf.c
>@@ -16,8 +16,8 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>-#include "constants.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
>+#include "modules/audio_coding/codecs/ilbc/constants.h"
> 
> /*----------------------------------------------------------------*
>  *  conversion from LSP coefficients to LSF coefficients
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lsp_to_lsf.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lsp_to_lsf.h
>index 11739853fbf..358786e7520 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lsp_to_lsf.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/lsp_to_lsf.h
>@@ -19,17 +19,17 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSP_TO_LSF_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSP_TO_LSF_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  *  conversion from LSP coefficients to LSF coefficients
>  *---------------------------------------------------------------*/
> 
> void WebRtcIlbcfix_Lsp2Lsf(
>-    int16_t *lsp, /* (i) lsp vector -1...+1 in Q15 */
>-    int16_t *lsf, /* (o) Lsf vector 0...Pi in Q13
>+    int16_t* lsp, /* (i) lsp vector -1...+1 in Q15 */
>+    int16_t* lsf, /* (o) Lsf vector 0...Pi in Q13
>                            (ordered, so that lsf[i]<lsf[i+1]) */
>-    int16_t m  /* (i) Number of coefficients */
>-                           );
>+    int16_t m     /* (i) Number of coefficients */
>+    );
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/my_corr.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/my_corr.c
>index 80847b6099c..2f2a058874d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/my_corr.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/my_corr.c
>@@ -16,7 +16,7 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  * compute cross correlation between sequences
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/my_corr.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/my_corr.h
>index 1d0ac85bde1..21deea52312 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/my_corr.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/my_corr.h
>@@ -19,18 +19,17 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_MY_CORR_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_MY_CORR_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  * compute cross correlation between sequences
>  *---------------------------------------------------------------*/
> 
>-void WebRtcIlbcfix_MyCorr(
>-    int32_t* corr,  /* (o) correlation of seq1 and seq2 */
>-    const int16_t* seq1,  /* (i) first sequence */
>-    size_t dim1,  /* (i) dimension first seq1 */
>-    const int16_t* seq2, /* (i) second sequence */
>-    size_t dim2   /* (i) dimension seq2 */
>+void WebRtcIlbcfix_MyCorr(int32_t* corr, /* (o) correlation of seq1 and seq2 */
>+                          const int16_t* seq1, /* (i) first sequence */
>+                          size_t dim1,         /* (i) dimension first seq1 */
>+                          const int16_t* seq2, /* (i) second sequence */
>+                          size_t dim2          /* (i) dimension seq2 */
>                           );
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/nearest_neighbor.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/nearest_neighbor.c
>index 2b58abc4f94..9d78528ba81 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/nearest_neighbor.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/nearest_neighbor.c
>@@ -16,7 +16,7 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> void WebRtcIlbcfix_NearestNeighbor(size_t* index,
>                                    const size_t* array,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/nearest_neighbor.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/nearest_neighbor.h
>index 8ed52bc04df..68b5c59b405 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/nearest_neighbor.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/nearest_neighbor.h
>@@ -19,7 +19,7 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_NEAREST_NEIGHBOR_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_NEAREST_NEIGHBOR_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  * Find index in array such that the array element with said
>@@ -27,10 +27,10 @@
>  *---------------------------------------------------------------*/
> 
> void WebRtcIlbcfix_NearestNeighbor(
>-    size_t* index, /* (o) index of array element closest to value */
>+    size_t* index,       /* (o) index of array element closest to value */
>     const size_t* array, /* (i) data array (Q2) */
>-    size_t value, /* (i) value (Q2) */
>-    size_t arlength /* (i) dimension of data array (==ENH_NBLOCKS_TOT) */
>-                                   );
>+    size_t value,        /* (i) value (Q2) */
>+    size_t arlength      /* (i) dimension of data array (==ENH_NBLOCKS_TOT) */
>+    );
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/pack_bits.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/pack_bits.c
>index 9475ecba01a..507e25e3ea2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/pack_bits.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/pack_bits.c
>@@ -16,7 +16,7 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  *  unpacking of bits from bitstream, i.e., vector of bytes
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/pack_bits.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/pack_bits.h
>index 66df7fc8035..8dcf41ce089 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/pack_bits.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/pack_bits.h
>@@ -19,16 +19,16 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_PACK_BITS_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_PACK_BITS_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  *  unpacking of bits from bitstream, i.e., vector of bytes
>  *---------------------------------------------------------------*/
> 
>-void WebRtcIlbcfix_PackBits( 
>-    uint16_t *bitstream,   /* (o) The packetized bitstream */
>-    iLBC_bits *enc_bits,  /* (i) Encoded bits */
>-    int16_t mode     /* (i) Codec mode (20 or 30) */
>-                             );
>+void WebRtcIlbcfix_PackBits(
>+    uint16_t* bitstream, /* (o) The packetized bitstream */
>+    iLBC_bits* enc_bits, /* (i) Encoded bits */
>+    int16_t mode         /* (i) Codec mode (20 or 30) */
>+    );
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/poly_to_lsf.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/poly_to_lsf.c
>index df8a78b1ed8..92aa16538d9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/poly_to_lsf.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/poly_to_lsf.c
>@@ -16,10 +16,10 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>-#include "constants.h"
>-#include "poly_to_lsp.h"
>-#include "lsp_to_lsf.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
>+#include "modules/audio_coding/codecs/ilbc/constants.h"
>+#include "modules/audio_coding/codecs/ilbc/poly_to_lsp.h"
>+#include "modules/audio_coding/codecs/ilbc/lsp_to_lsf.h"
> 
> void WebRtcIlbcfix_Poly2Lsf(
>     int16_t *lsf,   /* (o) lsf coefficients (Q13) */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/poly_to_lsf.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/poly_to_lsf.h
>index 3f2a2a01158..8a68d07d650 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/poly_to_lsf.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/poly_to_lsf.h
>@@ -19,15 +19,14 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_POLY_TO_LSF_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_POLY_TO_LSF_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  *  conversion from lpc coefficients to lsf coefficients
>  *---------------------------------------------------------------*/
> 
>-void WebRtcIlbcfix_Poly2Lsf(
>-    int16_t *lsf,   /* (o) lsf coefficients (Q13) */
>-    int16_t *a    /* (i) A coefficients (Q12) */
>+void WebRtcIlbcfix_Poly2Lsf(int16_t* lsf, /* (o) lsf coefficients (Q13) */
>+                            int16_t* a    /* (i) A coefficients (Q12) */
>                             );
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/poly_to_lsp.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/poly_to_lsp.c
>index ca376b34deb..88df6396e46 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/poly_to_lsp.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/poly_to_lsp.c
>@@ -16,9 +16,9 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>-#include "constants.h"
>-#include "chebyshev.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
>+#include "modules/audio_coding/codecs/ilbc/constants.h"
>+#include "modules/audio_coding/codecs/ilbc/chebyshev.h"
> 
> /*----------------------------------------------------------------*
>  * conversion from lpc coefficients to lsp coefficients
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/poly_to_lsp.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/poly_to_lsp.h
>index cc275076fa9..76378f29344 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/poly_to_lsp.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/poly_to_lsp.h
>@@ -19,7 +19,7 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_POLY_TO_LSP_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_POLY_TO_LSP_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  * conversion from lpc coefficients to lsp coefficients
>@@ -27,10 +27,10 @@
>  *---------------------------------------------------------------*/
> 
> void WebRtcIlbcfix_Poly2Lsp(
>-    int16_t *a,  /* (o) A coefficients in Q12 */
>-    int16_t *lsp, /* (i) LSP coefficients in Q15 */
>-    int16_t *old_lsp /* (i) old LSP coefficients that are used if the new
>+    int16_t* a,      /* (o) A coefficients in Q12 */
>+    int16_t* lsp,    /* (i) LSP coefficients in Q15 */
>+    int16_t* old_lsp /* (i) old LSP coefficients that are used if the new
>                               coefficients turn out to be unstable */
>-                            );
>+    );
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/refiner.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/refiner.c
>index 3c1265e5d50..d8a9bfb5799 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/refiner.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/refiner.c
>@@ -16,10 +16,10 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>-#include "constants.h"
>-#include "enh_upsample.h"
>-#include "my_corr.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
>+#include "modules/audio_coding/codecs/ilbc/constants.h"
>+#include "modules/audio_coding/codecs/ilbc/enh_upsample.h"
>+#include "modules/audio_coding/codecs/ilbc/my_corr.h"
> 
> /*----------------------------------------------------------------*
>  * find segment starting near idata+estSegPos that has highest
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/refiner.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/refiner.h
>index 92b81be1caf..87d0de78161 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/refiner.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/refiner.h
>@@ -19,7 +19,7 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_REFINER_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_REFINER_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  * find segment starting near idata+estSegPos that has highest
>@@ -30,14 +30,14 @@
>  *---------------------------------------------------------------*/
> 
> void WebRtcIlbcfix_Refiner(
>-    size_t *updStartPos, /* (o) updated start point (Q-2) */
>-    int16_t *idata,   /* (i) original data buffer */
>-    size_t idatal,   /* (i) dimension of idata */
>+    size_t* updStartPos,   /* (o) updated start point (Q-2) */
>+    int16_t* idata,        /* (i) original data buffer */
>+    size_t idatal,         /* (i) dimension of idata */
>     size_t centerStartPos, /* (i) beginning center segment */
>-    size_t estSegPos,  /* (i) estimated beginning other segment (Q-2) */
>-    int16_t *surround,  /* (i/o) The contribution from this sequence
>-                                 summed with earlier contributions */
>-    int16_t gain    /* (i) Gain to use for this sequence */
>-                           );
>+    size_t estSegPos,      /* (i) estimated beginning other segment (Q-2) */
>+    int16_t* surround,     /* (i/o) The contribution from this sequence
>+                                    summed with earlier contributions */
>+    int16_t gain           /* (i) Gain to use for this sequence */
>+    );
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.c
>index e63dda8c8f7..df41b281b9a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.c
>@@ -16,10 +16,10 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>-#include "lsf_interpolate_to_poly_enc.h"
>-#include "bw_expand.h"
>-#include "constants.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
>+#include "modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.h"
>+#include "modules/audio_coding/codecs/ilbc/bw_expand.h"
>+#include "modules/audio_coding/codecs/ilbc/constants.h"
> 
> /*----------------------------------------------------------------*
>  *  lsf interpolator (subrutine to LPCencode)
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.h
>index ccf66d32954..317f613d6c4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.h
>@@ -19,28 +19,28 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_INTERPOLATE_LSF_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_INTERPOLATE_LSF_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  *  lsf interpolator (subrutine to LPCencode)
>  *---------------------------------------------------------------*/
> 
> void WebRtcIlbcfix_SimpleInterpolateLsf(
>-    int16_t *syntdenum, /* (o) the synthesis filter denominator
>-                                   resulting from the quantized
>-                                   interpolated lsf Q12 */
>-    int16_t *weightdenum, /* (o) the weighting filter denominator
>+    int16_t* syntdenum,   /* (o) the synthesis filter denominator
>+                                     resulting from the quantized
>+                                     interpolated lsf Q12 */
>+    int16_t* weightdenum, /* (o) the weighting filter denominator
>                                    resulting from the unquantized
>                                    interpolated lsf Q12 */
>-    int16_t *lsf,  /* (i) the unquantized lsf coefficients Q13 */
>-    int16_t *lsfdeq,  /* (i) the dequantized lsf coefficients Q13 */
>-    int16_t *lsfold,  /* (i) the unquantized lsf coefficients of
>-                                           the previous signal frame Q13 */
>-    int16_t *lsfdeqold, /* (i) the dequantized lsf coefficients of the
>-                                   previous signal frame Q13 */
>-    int16_t length,  /* (i) should equate FILTERORDER */
>-    IlbcEncoder *iLBCenc_inst
>+    int16_t* lsf,         /* (i) the unquantized lsf coefficients Q13 */
>+    int16_t* lsfdeq,      /* (i) the dequantized lsf coefficients Q13 */
>+    int16_t* lsfold,      /* (i) the unquantized lsf coefficients of
>+                                               the previous signal frame Q13 */
>+    int16_t* lsfdeqold,   /* (i) the dequantized lsf coefficients of the
>+                                     previous signal frame Q13 */
>+    int16_t length,       /* (i) should equate FILTERORDER */
>+    IlbcEncoder* iLBCenc_inst
>     /* (i/o) the encoder state structure */
>-                                        );
>+    );
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.c
>index 72d80e04306..09e64ac6dcf 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.c
>@@ -16,11 +16,11 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>-#include "window32_w32.h"
>-#include "bw_expand.h"
>-#include "poly_to_lsf.h"
>-#include "constants.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
>+#include "modules/audio_coding/codecs/ilbc/window32_w32.h"
>+#include "modules/audio_coding/codecs/ilbc/bw_expand.h"
>+#include "modules/audio_coding/codecs/ilbc/poly_to_lsf.h"
>+#include "modules/audio_coding/codecs/ilbc/constants.h"
> 
> /*----------------------------------------------------------------*
>  *  lpc analysis (subrutine to LPCencode)
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.h
>index eb5b80302e5..3b0548d057a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.h
>@@ -19,17 +19,17 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_LPC_ANALYSIS_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_LPC_ANALYSIS_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  *  lpc analysis (subrutine to LPCencode)
>  *---------------------------------------------------------------*/
> 
> void WebRtcIlbcfix_SimpleLpcAnalysis(
>-    int16_t *lsf,   /* (o) lsf coefficients */
>-    int16_t *data,   /* (i) new block of speech */
>-    IlbcEncoder *iLBCenc_inst
>+    int16_t* lsf,  /* (o) lsf coefficients */
>+    int16_t* data, /* (i) new block of speech */
>+    IlbcEncoder* iLBCenc_inst
>     /* (i/o) the encoder state structure */
>-                                     );
>+    );
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.c
>index 9fa0d61ded2..90673a2a8b6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.c
>@@ -16,8 +16,8 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>-#include "constants.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
>+#include "modules/audio_coding/codecs/ilbc/constants.h"
> 
> /*----------------------------------------------------------------*
>  *  obtain dequantized lsf coefficients from quantization index
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.h
>index 660f8fe15dd..ee184864404 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.h
>@@ -19,16 +19,16 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_LSF_DEQUANT_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_LSF_DEQUANT_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  *  obtain dequantized lsf coefficients from quantization index
>  *---------------------------------------------------------------*/
> 
> void WebRtcIlbcfix_SimpleLsfDeQ(
>-    int16_t *lsfdeq,  /* (o) dequantized lsf coefficients */
>-    int16_t *index,  /* (i) quantization index */
>-    int16_t lpc_n  /* (i) number of LPCs */
>-                                );
>+    int16_t* lsfdeq, /* (o) dequantized lsf coefficients */
>+    int16_t* index,  /* (i) quantization index */
>+    int16_t lpc_n    /* (i) number of LPCs */
>+    );
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/simple_lsf_quant.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/simple_lsf_quant.c
>index 8daba501504..45373a943bf 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/simple_lsf_quant.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/simple_lsf_quant.c
>@@ -16,9 +16,9 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>-#include "split_vq.h"
>-#include "constants.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
>+#include "modules/audio_coding/codecs/ilbc/split_vq.h"
>+#include "modules/audio_coding/codecs/ilbc/constants.h"
> 
> /*----------------------------------------------------------------*
>  *  lsf quantizer (subrutine to LPCencode)
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/simple_lsf_quant.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/simple_lsf_quant.h
>index 6dc57196bf8..74fb0be5977 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/simple_lsf_quant.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/simple_lsf_quant.h
>@@ -19,19 +19,19 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_LSF_QUANT_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_LSF_QUANT_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  *  lsf quantizer (subrutine to LPCencode)
>  *---------------------------------------------------------------*/
> 
> void WebRtcIlbcfix_SimpleLsfQ(
>-    int16_t *lsfdeq, /* (o) dequantized lsf coefficients
>+    int16_t* lsfdeq, /* (o) dequantized lsf coefficients
>                                    (dimension FILTERORDER) Q13 */
>-    int16_t *index, /* (o) quantization index */
>-    int16_t *lsf, /* (i) the lsf coefficient vector to be
>-                           quantized (dimension FILTERORDER) Q13 */
>-    int16_t lpc_n /* (i) number of lsf sets to quantize */
>-                              );
>+    int16_t* index,  /* (o) quantization index */
>+    int16_t* lsf,    /* (i) the lsf coefficient vector to be
>+                              quantized (dimension FILTERORDER) Q13 */
>+    int16_t lpc_n    /* (i) number of lsf sets to quantize */
>+    );
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/smooth.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/smooth.c
>index ed879327c09..edafb0c9351 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/smooth.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/smooth.c
>@@ -16,9 +16,9 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>-#include "constants.h"
>-#include "smooth_out_data.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
>+#include "modules/audio_coding/codecs/ilbc/constants.h"
>+#include "modules/audio_coding/codecs/ilbc/smooth_out_data.h"
> 
> /*----------------------------------------------------------------*
>  * find the smoothed output data
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/smooth.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/smooth.h
>index 351543229af..52e7ff97b67 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/smooth.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/smooth.h
>@@ -19,18 +19,17 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SMOOTH_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SMOOTH_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  * find the smoothed output data
>  *---------------------------------------------------------------*/
> 
>-void WebRtcIlbcfix_Smooth(
>-    int16_t *odata,   /* (o) smoothed output */
>-    int16_t *current,  /* (i) the un enhanced residual for
>-                                this block */
>-    int16_t *surround  /* (i) The approximation from the
>-                                surrounding sequences */
>+void WebRtcIlbcfix_Smooth(int16_t* odata,   /* (o) smoothed output */
>+                          int16_t* current, /* (i) the un enhanced residual for
>+                                                     this block */
>+                          int16_t* surround /* (i) The approximation from the
>+                                                     surrounding sequences */
>                           );
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/smooth_out_data.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/smooth_out_data.c
>index 9c41e0422b0..72b3a477909 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/smooth_out_data.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/smooth_out_data.c
>@@ -16,8 +16,18 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>-#include "constants.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
>+#include "modules/audio_coding/codecs/ilbc/constants.h"
>+#include "rtc_base/sanitizer.h"
>+
>+// An s32 + s32 -> s32 addition that's allowed to overflow. (It's still
>+// undefined behavior, so not a good idea; this just makes UBSan ignore the
>+// violation, so that our old code can continue to do what it's always been
>+// doing.)
>+static inline int32_t RTC_NO_SANITIZE("signed-integer-overflow")
>+    OverflowingAdd_S32_S32_To_S32(int32_t a, int32_t b) {
>+  return a + b;
>+}
> 
> int32_t WebRtcIlbcfix_Smooth_odata(
>     int16_t *odata,
>@@ -37,7 +47,7 @@ int32_t WebRtcIlbcfix_Smooth_odata(
>   errs=0;
>   for(i=0;i<80;i++) {
>     err = (psseq[i] - odata[i]) >> 3;
>-    errs += err * err;  /* errs in Q-6 */
>+    errs = OverflowingAdd_S32_S32_To_S32(errs, err * err);  // errs in Q-6
>   }
> 
>   return errs;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/smooth_out_data.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/smooth_out_data.h
>index 7ed1a055c90..df946e3a728 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/smooth_out_data.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/smooth_out_data.h
>@@ -19,17 +19,15 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SMOOTH_OUT_DATA_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SMOOTH_OUT_DATA_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  * help function to WebRtcIlbcfix_Smooth()
>  *---------------------------------------------------------------*/
> 
>-int32_t WebRtcIlbcfix_Smooth_odata(
>-    int16_t *odata,
>-    int16_t *psseq,
>-    int16_t *surround,
>-    int16_t C);
>-
>+int32_t WebRtcIlbcfix_Smooth_odata(int16_t* odata,
>+                                   int16_t* psseq,
>+                                   int16_t* surround,
>+                                   int16_t C);
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/sort_sq.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/sort_sq.c
>index c51bf6dc637..dd3ca8057e5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/sort_sq.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/sort_sq.c
>@@ -16,7 +16,7 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  *  scalar quantization
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/sort_sq.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/sort_sq.h
>index 436e1185c01..1fe7fbf2372 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/sort_sq.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/sort_sq.h
>@@ -19,18 +19,18 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SORT_SQ_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SORT_SQ_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  *  scalar quantization
>  *---------------------------------------------------------------*/
> 
> void WebRtcIlbcfix_SortSq(
>-    int16_t *xq,   /* (o) the quantized value */
>-    int16_t *index,  /* (o) the quantization index */
>-    int16_t x,   /* (i) the value to quantize */
>-    const int16_t *cb, /* (i) the quantization codebook */
>-    int16_t cb_size  /* (i) the size of the quantization codebook */
>-                           );
>+    int16_t* xq,       /* (o) the quantized value */
>+    int16_t* index,    /* (o) the quantization index */
>+    int16_t x,         /* (i) the value to quantize */
>+    const int16_t* cb, /* (i) the quantization codebook */
>+    int16_t cb_size    /* (i) the size of the quantization codebook */
>+    );
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/split_vq.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/split_vq.c
>index 39b6e1b56ac..2f218ed5924 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/split_vq.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/split_vq.c
>@@ -16,10 +16,10 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>-#include "constants.h"
>-#include "vq3.h"
>-#include "vq4.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
>+#include "modules/audio_coding/codecs/ilbc/constants.h"
>+#include "modules/audio_coding/codecs/ilbc/vq3.h"
>+#include "modules/audio_coding/codecs/ilbc/vq4.h"
> 
> /*----------------------------------------------------------------*
>  *  split vector quantization
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/split_vq.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/split_vq.h
>index 3b303bd1563..6bc2db67e3a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/split_vq.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/split_vq.h
>@@ -19,20 +19,20 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SPLIT_VQ_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SPLIT_VQ_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  *  split vector quantization
>  *---------------------------------------------------------------*/
> 
> void WebRtcIlbcfix_SplitVq(
>-    int16_t *qX,  /* (o) the quantized vector in Q13 */
>-    int16_t *index, /* (o) a vector of indexes for all vector
>+    int16_t* qX,    /* (o) the quantized vector in Q13 */
>+    int16_t* index, /* (o) a vector of indexes for all vector
>                                    codebooks in the split */
>-    int16_t *X,  /* (i) the vector to quantize */
>-    int16_t *CB,  /* (i) the quantizer codebook in Q13 */
>-    int16_t *dim, /* (i) the dimension of X and qX */
>-    int16_t *cbsize /* (i) the number of vectors in the codebook */
>-                           );
>+    int16_t* X,     /* (i) the vector to quantize */
>+    int16_t* CB,    /* (i) the quantizer codebook in Q13 */
>+    int16_t* dim,   /* (i) the dimension of X and qX */
>+    int16_t* cbsize /* (i) the number of vectors in the codebook */
>+    );
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/state_construct.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/state_construct.c
>index 29fe91b87ea..753415b9c80 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/state_construct.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/state_construct.c
>@@ -16,8 +16,8 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>-#include "constants.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
>+#include "modules/audio_coding/codecs/ilbc/constants.h"
> 
> /*----------------------------------------------------------------*
>  *  decoding of the start state
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/state_construct.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/state_construct.h
>index 9339f656e4f..0dadf488684 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/state_construct.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/state_construct.h
>@@ -26,10 +26,10 @@
> void WebRtcIlbcfix_StateConstruct(
>     size_t idxForMax,   /* (i) 6-bit index for the quantization of
>                                            max amplitude */
>-    int16_t *idxVec,   /* (i) vector of quantization indexes */
>-    int16_t *syntDenum,  /* (i) synthesis filter denumerator */
>-    int16_t *Out_fix,  /* (o) the decoded state vector */
>-    size_t len    /* (i) length of a state vector */
>-                                  );
>+    int16_t* idxVec,    /* (i) vector of quantization indexes */
>+    int16_t* syntDenum, /* (i) synthesis filter denumerator */
>+    int16_t* Out_fix,   /* (o) the decoded state vector */
>+    size_t len          /* (i) length of a state vector */
>+    );
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/state_search.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/state_search.c
>index 295c543d84e..5e8a2f55ab3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/state_search.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/state_search.c
>@@ -16,9 +16,9 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>-#include "constants.h"
>-#include "abs_quant.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
>+#include "modules/audio_coding/codecs/ilbc/constants.h"
>+#include "modules/audio_coding/codecs/ilbc/abs_quant.h"
> 
> /*----------------------------------------------------------------*
>  *  encoding of start state
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/state_search.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/state_search.h
>index 94a9d092662..1ad27cec026 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/state_search.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/state_search.h
>@@ -19,20 +19,20 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_STATE_SEARCH_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_STATE_SEARCH_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  *  encoding of start state
>  *---------------------------------------------------------------*/
> 
> void WebRtcIlbcfix_StateSearch(
>-    IlbcEncoder *iLBCenc_inst,
>+    IlbcEncoder* iLBCenc_inst,
>     /* (i) Encoder instance */
>-    iLBC_bits *iLBC_encbits,/* (i/o) Encoded bits (output idxForMax
>-                               and idxVec, input state_first) */
>-    int16_t *residual,   /* (i) target residual vector */
>-    int16_t *syntDenum,  /* (i) lpc synthesis filter */
>-    int16_t *weightDenum  /* (i) weighting filter denuminator */
>-                               );
>+    iLBC_bits* iLBC_encbits, /* (i/o) Encoded bits (output idxForMax
>+                                and idxVec, input state_first) */
>+    int16_t* residual,       /* (i) target residual vector */
>+    int16_t* syntDenum,      /* (i) lpc synthesis filter */
>+    int16_t* weightDenum     /* (i) weighting filter denuminator */
>+    );
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/swap_bytes.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/swap_bytes.c
>index b795e56ac44..806cc2a39a8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/swap_bytes.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/swap_bytes.c
>@@ -16,7 +16,7 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  * Swap bytes (to simplify operations on Little Endian machines)
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/swap_bytes.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/swap_bytes.h
>index 95182993ca1..381b73ae4f7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/swap_bytes.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/swap_bytes.h
>@@ -19,16 +19,16 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SWAP_BYTES_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SWAP_BYTES_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  * Swap bytes (to simplify operations on Little Endian machines)
>  *---------------------------------------------------------------*/
> 
> void WebRtcIlbcfix_SwapBytes(
>-    const uint16_t* input,   /* (i) the sequence to swap */
>-    size_t wordLength,      /* (i) number or uint16_t to swap */
>-    uint16_t* output         /* (o) the swapped sequence */
>-                              );
>+    const uint16_t* input, /* (i) the sequence to swap */
>+    size_t wordLength,     /* (i) number or uint16_t to swap */
>+    uint16_t* output       /* (o) the swapped sequence */
>+    );
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDSDPUtils.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/test/empty.cc
>similarity index 100%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDSDPUtils.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/test/empty.cc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/unpack_bits.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/unpack_bits.c
>index fae06f63397..ad6a7ee1dab 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/unpack_bits.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/unpack_bits.c
>@@ -16,7 +16,7 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  *  unpacking of bits from bitstream, i.e., vector of bytes
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/unpack_bits.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/unpack_bits.h
>index 01cf398004b..4fd0a80b4bb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/unpack_bits.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/unpack_bits.h
>@@ -19,16 +19,19 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_UNPACK_BITS_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_UNPACK_BITS_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  *  unpacking of bits from bitstream, i.e., vector of bytes
>  *---------------------------------------------------------------*/
> 
>-int16_t WebRtcIlbcfix_UnpackBits( /* (o) "Empty" frame indicator */
>-    const uint16_t *bitstream,    /* (i) The packatized bitstream */
>-    iLBC_bits *enc_bits,  /* (o) Paramerers from bitstream */
>-    int16_t mode     /* (i) Codec mode (20 or 30) */
>-                                        );
>+int16_t
>+WebRtcIlbcfix_UnpackBits(/* (o) "Empty" frame indicator */
>+                         const uint16_t*
>+                             bitstream, /* (i) The packatized bitstream */
>+                         iLBC_bits*
>+                             enc_bits, /* (o) Paramerers from bitstream */
>+                         int16_t mode  /* (i) Codec mode (20 or 30) */
>+                         );
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/vq3.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/vq3.c
>index ee3b26e0394..b63a7a84797 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/vq3.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/vq3.c
>@@ -16,8 +16,8 @@
> 
> ******************************************************************/
> 
>-#include "vq3.h"
>-#include "constants.h"
>+#include "modules/audio_coding/codecs/ilbc/vq3.h"
>+#include "modules/audio_coding/codecs/ilbc/constants.h"
> 
> /*----------------------------------------------------------------*
>  *  vector quantization
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/vq3.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/vq3.h
>index 6d3dc3ac62b..c946478a1a5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/vq3.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/vq3.h
>@@ -19,18 +19,18 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_VQ3_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_VQ3_H_
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include <stdint.h>
> 
> /*----------------------------------------------------------------*
>  *  Vector quantization of order 3 (based on MSE)
>  *---------------------------------------------------------------*/
> 
> void WebRtcIlbcfix_Vq3(
>-    int16_t *Xq,  /* (o) the quantized vector (Q13) */
>-    int16_t *index, /* (o) the quantization index */
>-    int16_t *CB,  /* (i) the vector quantization codebook (Q13) */
>-    int16_t *X,  /* (i) the vector to quantize (Q13) */
>-    int16_t n_cb  /* (i) the number of vectors in the codebook */
>-                       );
>+    int16_t* Xq,    /* (o) the quantized vector (Q13) */
>+    int16_t* index, /* (o) the quantization index */
>+    int16_t* CB,    /* (i) the vector quantization codebook (Q13) */
>+    int16_t* X,     /* (i) the vector to quantize (Q13) */
>+    int16_t n_cb    /* (i) the number of vectors in the codebook */
>+    );
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/vq4.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/vq4.c
>index bc209521324..2522ac28e6f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/vq4.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/vq4.c
>@@ -16,8 +16,8 @@
> 
> ******************************************************************/
> 
>-#include "vq4.h"
>-#include "constants.h"
>+#include "modules/audio_coding/codecs/ilbc/vq4.h"
>+#include "modules/audio_coding/codecs/ilbc/constants.h"
> 
> /*----------------------------------------------------------------*
>  *  vector quantization
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/vq4.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/vq4.h
>index c7f5271fe5f..6d14830c03e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/vq4.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/vq4.h
>@@ -19,18 +19,18 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_VQ4_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_VQ4_H_
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include <stdint.h>
> 
> /*----------------------------------------------------------------*
>  *  Vector quantization of order 4 (based on MSE)
>  *---------------------------------------------------------------*/
> 
> void WebRtcIlbcfix_Vq4(
>-    int16_t *Xq,  /* (o) the quantized vector (Q13) */
>-    int16_t *index, /* (o) the quantization index */
>-    int16_t *CB,  /* (i) the vector quantization codebook (Q13) */
>-    int16_t *X,  /* (i) the vector to quantize (Q13) */
>-    int16_t n_cb  /* (i) the number of vectors in the codebook */
>-                       );
>+    int16_t* Xq,    /* (o) the quantized vector (Q13) */
>+    int16_t* index, /* (o) the quantization index */
>+    int16_t* CB,    /* (i) the vector quantization codebook (Q13) */
>+    int16_t* X,     /* (i) the vector to quantize (Q13) */
>+    int16_t n_cb    /* (i) the number of vectors in the codebook */
>+    );
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/window32_w32.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/window32_w32.c
>index dc12a5a7c43..fe19de68502 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/window32_w32.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/window32_w32.c
>@@ -16,7 +16,7 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  *  window multiplication
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/window32_w32.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/window32_w32.h
>index f53592e87b6..0cef0848f8c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/window32_w32.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/window32_w32.h
>@@ -19,17 +19,16 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_WINDOW32_W32_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_WINDOW32_W32_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  *  window multiplication
>  *---------------------------------------------------------------*/
> 
>-void WebRtcIlbcfix_Window32W32(
>-    int32_t *z,    /* Output */
>-    int32_t *x,    /* Input (same domain as Output)*/
>-    const int32_t  *y,  /* Q31 Window */
>-    size_t N     /* length to process */
>+void WebRtcIlbcfix_Window32W32(int32_t* z, /* Output */
>+                               int32_t* x, /* Input (same domain as Output)*/
>+                               const int32_t* y, /* Q31 Window */
>+                               size_t N          /* length to process */
>                                );
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/xcorr_coef.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/xcorr_coef.c
>index 0d898c54a48..f6375dfbff5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/xcorr_coef.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/xcorr_coef.c
>@@ -16,7 +16,7 @@
> 
> ******************************************************************/
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  * cross correlation which finds the optimal lag for the
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/xcorr_coef.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/xcorr_coef.h
>index e2d511a6f9a..e6c3d3f6060 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/xcorr_coef.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/xcorr_coef.h
>@@ -19,7 +19,7 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_XCORR_COEF_H_
> #define MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_XCORR_COEF_H_
> 
>-#include "defines.h"
>+#include "modules/audio_coding/codecs/ilbc/defines.h"
> 
> /*----------------------------------------------------------------*
>  * cross correlation which finds the optimal lag for the
>@@ -27,12 +27,12 @@
>  *---------------------------------------------------------------*/
> 
> size_t WebRtcIlbcfix_XcorrCoef(
>-    int16_t *target,  /* (i) first array */
>-    int16_t *regressor, /* (i) second array */
>-    size_t subl,  /* (i) dimension arrays */
>-    size_t searchLen, /* (i) the search lenght */
>-    size_t offset,  /* (i) samples offset between arrays */
>-    int16_t step   /* (i) +1 or -1 */
>-                            );
>+    int16_t* target,    /* (i) first array */
>+    int16_t* regressor, /* (i) second array */
>+    size_t subl,        /* (i) dimension arrays */
>+    size_t searchLen,   /* (i) the search lenght */
>+    size_t offset,      /* (i) samples offset between arrays */
>+    int16_t step        /* (i) +1 or -1 */
>+    );
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h
>index eda1cfa545e..f840ffa1644 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/audio_decoder_isac_t.h
>@@ -13,8 +13,8 @@
> 
> #include <vector>
> 
>+#include "absl/types/optional.h"
> #include "api/audio_codecs/audio_decoder.h"
>-#include "api/optional.h"
> #include "modules/audio_coding/codecs/isac/locked_bandwidth_info.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/scoped_ref_ptr.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
>index 696b799a41f..cbf15fcdc51 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
>@@ -90,9 +90,8 @@ size_t AudioEncoderIsacT<T>::NumChannels() const {
> template <typename T>
> size_t AudioEncoderIsacT<T>::Num10MsFramesInNextPacket() const {
>   const int samples_in_next_packet = T::GetNewFrameLen(isac_state_);
>-  return static_cast<size_t>(
>-      rtc::CheckedDivExact(samples_in_next_packet,
>-                           rtc::CheckedDivExact(SampleRateHz(), 100)));
>+  return static_cast<size_t>(rtc::CheckedDivExact(
>+      samples_in_next_packet, rtc::CheckedDivExact(SampleRateHz(), 100)));
> }
> 
> template <typename T>
>@@ -123,8 +122,7 @@ AudioEncoder::EncodedInfo AudioEncoderIsacT<T>::EncodeImpl(
>   }
> 
>   size_t encoded_bytes = encoded->AppendData(
>-      kSufficientEncodeBufferSizeBytes,
>-      [&] (rtc::ArrayView<uint8_t> encoded) {
>+      kSufficientEncodeBufferSizeBytes, [&](rtc::ArrayView<uint8_t> encoded) {
>         int r = T::Encode(isac_state_, audio.data(), encoded.data());
> 
>         RTC_CHECK_GE(r, 0) << "Encode failed (error code "
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/bandwidth_info.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/bandwidth_info.h
>index 05397804f51..c3830a5f7c6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/bandwidth_info.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/bandwidth_info.h
>@@ -11,7 +11,7 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ISAC_BANDWIDTH_INFO_H_
> #define MODULES_AUDIO_CODING_CODECS_ISAC_BANDWIDTH_INFO_H_
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include <stdint.h>
> 
> typedef struct {
>   int in_use;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDSDPUtils.m b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/empty.cc
>similarity index 100%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ARDSDPUtils.m
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/empty.cc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/include/isacfix.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/include/isacfix.h
>index ef194ca5280..8fcfebb21ef 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/include/isacfix.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/include/isacfix.h
>@@ -14,624 +14,592 @@
> #include <stddef.h>
> 
> #include "modules/audio_coding/codecs/isac/bandwidth_info.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>-
>-typedef struct {
>-  void *dummy;
>-} ISACFIX_MainStruct;
> 
>+typedef struct { void* dummy; } ISACFIX_MainStruct;
> 
> #if defined(__cplusplus)
> extern "C" {
> #endif
> 
>+/**************************************************************************
>+ * WebRtcIsacfix_AssignSize(...)
>+ *
>+ *  Functions used when malloc is not allowed
>+ *  Output the number of bytes needed to allocate for iSAC struct.
>+ *
>+ */
>+
>+int16_t WebRtcIsacfix_AssignSize(int* sizeinbytes);
>+
>+/**************************************************************************
>+ * WebRtcIsacfix_Assign(...)
>+ *
>+ * Functions used when malloc is not allowed, it
>+ * places a struct at the given address.
>+ *
>+ * Input:
>+ *      - *ISAC_main_inst   : a pointer to the coder instance.
>+ *      - ISACFIX_inst_Addr : address of the memory where a space is
>+ *                            for iSAC structure.
>+ *
>+ * Return value             : 0 - Ok
>+ *                           -1 - Error
>+ */
>+
>+int16_t WebRtcIsacfix_Assign(ISACFIX_MainStruct** inst,
>+                             void* ISACFIX_inst_Addr);
>+
>+/****************************************************************************
>+ * WebRtcIsacfix_Create(...)
>+ *
>+ * This function creates an ISAC instance, which will contain the state
>+ * information for one coding/decoding channel.
>+ *
>+ * Input:
>+ *      - *ISAC_main_inst   : a pointer to the coder instance.
>+ *
>+ * Return value             : 0 - Ok
>+ *                           -1 - Error
>+ */
>+
>+int16_t WebRtcIsacfix_Create(ISACFIX_MainStruct** ISAC_main_inst);
>+
>+/****************************************************************************
>+ * WebRtcIsacfix_Free(...)
>+ *
>+ * This function frees the ISAC instance created at the beginning.
>+ *
>+ * Input:
>+ *      - ISAC_main_inst    : a ISAC instance.
>+ *
>+ * Return value             :  0 - Ok
>+ *                            -1 - Error
>+ */
>+
>+int16_t WebRtcIsacfix_Free(ISACFIX_MainStruct* ISAC_main_inst);
>+
>+/****************************************************************************
>+ * WebRtcIsacfix_EncoderInit(...)
>+ *
>+ * This function initializes an ISAC instance prior to the encoder calls.
>+ *
>+ * Input:
>+ *     - ISAC_main_inst     : ISAC instance.
>+ *     - CodingMode         : 0 - Bit rate and frame length are automatically
>+ *                                adjusted to available bandwidth on
>+ *                                transmission channel.
>+ *                            1 - User sets a frame length and a target bit
>+ *                                rate which is taken as the maximum short-term
>+ *                                average bit rate.
>+ *
>+ * Return value             :  0 - Ok
>+ *                            -1 - Error
>+ */
>+
>+int16_t WebRtcIsacfix_EncoderInit(ISACFIX_MainStruct* ISAC_main_inst,
>+                                  int16_t CodingMode);
>+
>+/****************************************************************************
>+ * WebRtcIsacfix_Encode(...)
>+ *
>+ * This function encodes 10ms frame(s) and inserts it into a package.
>+ * Input speech length has to be 160 samples (10ms). The encoder buffers those
>+ * 10ms frames until it reaches the chosen Framesize (480 or 960 samples
>+ * corresponding to 30 or 60 ms frames), and then proceeds to the encoding.
>+ *
>+ * Input:
>+ *      - ISAC_main_inst    : ISAC instance.
>+ *      - speechIn          : input speech vector.
>+ *
>+ * Output:
>+ *      - encoded           : the encoded data vector
>+ *
>+ * Return value             : >0 - Length (in bytes) of coded data
>+ *                             0 - The buffer didn't reach the chosen framesize
>+ *                                 so it keeps buffering speech samples.
>+ *                            -1 - Error
>+ */
> 
>-  /**************************************************************************
>-   * WebRtcIsacfix_AssignSize(...)
>-   *
>-   *  Functions used when malloc is not allowed
>-   *  Output the number of bytes needed to allocate for iSAC struct.
>-   *
>-   */
>-
>-  int16_t WebRtcIsacfix_AssignSize(int *sizeinbytes);
>-
>-  /**************************************************************************
>-   * WebRtcIsacfix_Assign(...)
>-   *
>-   * Functions used when malloc is not allowed, it
>-   * places a struct at the given address.
>-   *
>-   * Input:
>-   *      - *ISAC_main_inst   : a pointer to the coder instance.
>-   *      - ISACFIX_inst_Addr : address of the memory where a space is
>-   *                            for iSAC structure.
>-   *
>-   * Return value             : 0 - Ok
>-   *                           -1 - Error
>-   */
>-
>-  int16_t WebRtcIsacfix_Assign(ISACFIX_MainStruct **inst,
>-                                     void *ISACFIX_inst_Addr);
>-
>-  /****************************************************************************
>-   * WebRtcIsacfix_Create(...)
>-   *
>-   * This function creates an ISAC instance, which will contain the state
>-   * information for one coding/decoding channel.
>-   *
>-   * Input:
>-   *      - *ISAC_main_inst   : a pointer to the coder instance.
>-   *
>-   * Return value             : 0 - Ok
>-   *                           -1 - Error
>-   */
>-
>-  int16_t WebRtcIsacfix_Create(ISACFIX_MainStruct **ISAC_main_inst);
>-
>-
>-  /****************************************************************************
>-   * WebRtcIsacfix_Free(...)
>-   *
>-   * This function frees the ISAC instance created at the beginning.
>-   *
>-   * Input:
>-   *      - ISAC_main_inst    : a ISAC instance.
>-   *
>-   * Return value             :  0 - Ok
>-   *                            -1 - Error
>-   */
>-
>-  int16_t WebRtcIsacfix_Free(ISACFIX_MainStruct *ISAC_main_inst);
>-
>-
>-  /****************************************************************************
>-   * WebRtcIsacfix_EncoderInit(...)
>-   *
>-   * This function initializes an ISAC instance prior to the encoder calls.
>-   *
>-   * Input:
>-   *     - ISAC_main_inst     : ISAC instance.
>-   *     - CodingMode         : 0 - Bit rate and frame length are automatically
>-   *                                adjusted to available bandwidth on
>-   *                                transmission channel.
>-   *                            1 - User sets a frame length and a target bit
>-   *                                rate which is taken as the maximum short-term
>-   *                                average bit rate.
>-   *
>-   * Return value             :  0 - Ok
>-   *                            -1 - Error
>-   */
>-
>-  int16_t WebRtcIsacfix_EncoderInit(ISACFIX_MainStruct *ISAC_main_inst,
>-                                    int16_t  CodingMode);
>-
>-
>-  /****************************************************************************
>-   * WebRtcIsacfix_Encode(...)
>-   *
>-   * This function encodes 10ms frame(s) and inserts it into a package.
>-   * Input speech length has to be 160 samples (10ms). The encoder buffers those
>-   * 10ms frames until it reaches the chosen Framesize (480 or 960 samples
>-   * corresponding to 30 or 60 ms frames), and then proceeds to the encoding.
>-   *
>-   * Input:
>-   *      - ISAC_main_inst    : ISAC instance.
>-   *      - speechIn          : input speech vector.
>-   *
>-   * Output:
>-   *      - encoded           : the encoded data vector
>-   *
>-   * Return value             : >0 - Length (in bytes) of coded data
>-   *                             0 - The buffer didn't reach the chosen framesize
>-   *                                 so it keeps buffering speech samples.
>-   *                            -1 - Error
>-   */
>-
>-  int WebRtcIsacfix_Encode(ISACFIX_MainStruct *ISAC_main_inst,
>-                           const int16_t *speechIn,
>-                           uint8_t* encoded);
>-
>-
>-
>-  /****************************************************************************
>-   * WebRtcIsacfix_EncodeNb(...)
>-   *
>-   * This function encodes 10ms narrow band (8 kHz sampling) frame(s) and inserts
>-   * it into a package. Input speech length has to be 80 samples (10ms). The encoder
>-   * interpolates into wide-band (16 kHz sampling) buffers those
>-   * 10ms frames until it reaches the chosen Framesize (480 or 960 wide-band samples
>-   * corresponding to 30 or 60 ms frames), and then proceeds to the encoding.
>-   *
>-   * The function is enabled if WEBRTC_ISAC_FIX_NB_CALLS_ENABLED is defined
>-   *
>-   * Input:
>-   *      - ISAC_main_inst    : ISAC instance.
>-   *      - speechIn          : input speech vector.
>-   *
>-   * Output:
>-   *      - encoded           : the encoded data vector
>-   *
>-   * Return value             : >0 - Length (in bytes) of coded data
>-   *                             0 - The buffer didn't reach the chosen framesize
>-   *                                 so it keeps buffering speech samples.
>-   *                            -1 - Error
>-   */
>+int WebRtcIsacfix_Encode(ISACFIX_MainStruct* ISAC_main_inst,
>+                         const int16_t* speechIn,
>+                         uint8_t* encoded);
> 
>+/****************************************************************************
>+ * WebRtcIsacfix_EncodeNb(...)
>+ *
>+ * This function encodes 10ms narrow band (8 kHz sampling) frame(s) and inserts
>+ * it into a package. Input speech length has to be 80 samples (10ms). The
>+ * encoder interpolates into wide-band (16 kHz sampling) buffers those 10ms
>+ * frames until it reaches the chosen Framesize (480 or 960 wide-band samples
>+ * corresponding to 30 or 60 ms frames), and then proceeds to the encoding.
>+ *
>+ * The function is enabled if WEBRTC_ISAC_FIX_NB_CALLS_ENABLED is defined
>+ *
>+ * Input:
>+ *      - ISAC_main_inst    : ISAC instance.
>+ *      - speechIn          : input speech vector.
>+ *
>+ * Output:
>+ *      - encoded           : the encoded data vector
>+ *
>+ * Return value             : >0 - Length (in bytes) of coded data
>+ *                             0 - The buffer didn't reach the chosen framesize
>+ *                                 so it keeps buffering speech samples.
>+ *                            -1 - Error
>+ */
> 
> #ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
>-  int16_t WebRtcIsacfix_EncodeNb(ISACFIX_MainStruct *ISAC_main_inst,
>-                                 const int16_t *speechIn,
>-                                 int16_t *encoded);
>-#endif //  WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
>-
>-
>-
>-  /****************************************************************************
>-   * WebRtcIsacfix_DecoderInit(...)
>-   *
>-   * This function initializes an ISAC instance prior to the decoder calls.
>-   *
>-   * Input:
>-   *  - ISAC_main_inst : ISAC instance.
>-   */
>-
>-  void WebRtcIsacfix_DecoderInit(ISACFIX_MainStruct* ISAC_main_inst);
>-
>-  /****************************************************************************
>-   * WebRtcIsacfix_UpdateBwEstimate1(...)
>-   *
>-   * This function updates the estimate of the bandwidth.
>-   *
>-   * Input:
>-   *      - ISAC_main_inst    : ISAC instance.
>-   *      - encoded           : encoded ISAC frame(s).
>-   *      - packet_size       : size of the packet in bytes.
>-   *      - rtp_seq_number    : the RTP number of the packet.
>-   *      - arr_ts            : the arrival time of the packet (from NetEq)
>-   *                            in samples.
>-   *
>-   * Return value             : 0 - Ok
>-   *                           -1 - Error
>-   */
>-
>-  int16_t WebRtcIsacfix_UpdateBwEstimate1(ISACFIX_MainStruct *ISAC_main_inst,
>-                                          const uint8_t* encoded,
>-                                          size_t packet_size,
>-                                          uint16_t rtp_seq_number,
>-                                          uint32_t arr_ts);
>-
>-  /****************************************************************************
>-   * WebRtcIsacfix_UpdateBwEstimate(...)
>-   *
>-   * This function updates the estimate of the bandwidth.
>-   *
>-   * Input:
>-   *      - ISAC_main_inst    : ISAC instance.
>-   *      - encoded           : encoded ISAC frame(s).
>-   *      - packet_size       : size of the packet in bytes.
>-   *      - rtp_seq_number    : the RTP number of the packet.
>-   *      - send_ts           : the send time of the packet from RTP header,
>-   *                            in samples.
>-   *      - arr_ts            : the arrival time of the packet (from NetEq)
>-   *                            in samples.
>-   *
>-   * Return value             :  0 - Ok
>-   *                            -1 - Error
>-   */
>-
>-  int16_t WebRtcIsacfix_UpdateBwEstimate(ISACFIX_MainStruct *ISAC_main_inst,
>-                                         const uint8_t* encoded,
>-                                         size_t packet_size,
>-                                         uint16_t rtp_seq_number,
>-                                         uint32_t send_ts,
>-                                         uint32_t arr_ts);
>-
>-  /****************************************************************************
>-   * WebRtcIsacfix_Decode(...)
>-   *
>-   * This function decodes an ISAC frame. Output speech length
>-   * will be a multiple of 480 samples: 480 or 960 samples,
>-   * depending on the framesize (30 or 60 ms).
>-   *
>-   * Input:
>-   *      - ISAC_main_inst    : ISAC instance.
>-   *      - encoded           : encoded ISAC frame(s)
>-   *      - len               : bytes in encoded vector
>-   *
>-   * Output:
>-   *      - decoded           : The decoded vector
>-   *
>-   * Return value             : >0 - number of samples in decoded vector
>-   *                            -1 - Error
>-   */
>-
>-  int WebRtcIsacfix_Decode(ISACFIX_MainStruct *ISAC_main_inst,
>-                           const uint8_t* encoded,
>-                           size_t len,
>-                           int16_t *decoded,
>-                           int16_t *speechType);
>-
>-
>-  /****************************************************************************
>-   * WebRtcIsacfix_DecodeNb(...)
>-   *
>-   * This function decodes a ISAC frame in narrow-band (8 kHz sampling).
>-   * Output speech length will be a multiple of 240 samples: 240 or 480 samples,
>-   * depending on the framesize (30 or 60 ms).
>-   *
>-   * The function is enabled if WEBRTC_ISAC_FIX_NB_CALLS_ENABLED is defined
>-   *
>-   * Input:
>-   *      - ISAC_main_inst    : ISAC instance.
>-   *      - encoded           : encoded ISAC frame(s)
>-   *      - len               : bytes in encoded vector
>-   *
>-   * Output:
>-   *      - decoded           : The decoded vector
>-   *
>-   * Return value             : >0 - number of samples in decoded vector
>-   *                            -1 - Error
>-   */
>+int16_t WebRtcIsacfix_EncodeNb(ISACFIX_MainStruct* ISAC_main_inst,
>+                               const int16_t* speechIn,
>+                               int16_t* encoded);
>+#endif  //  WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
>+
>+/****************************************************************************
>+ * WebRtcIsacfix_DecoderInit(...)
>+ *
>+ * This function initializes an ISAC instance prior to the decoder calls.
>+ *
>+ * Input:
>+ *  - ISAC_main_inst : ISAC instance.
>+ */
>+
>+void WebRtcIsacfix_DecoderInit(ISACFIX_MainStruct* ISAC_main_inst);
>+
>+/****************************************************************************
>+ * WebRtcIsacfix_UpdateBwEstimate1(...)
>+ *
>+ * This function updates the estimate of the bandwidth.
>+ *
>+ * Input:
>+ *      - ISAC_main_inst    : ISAC instance.
>+ *      - encoded           : encoded ISAC frame(s).
>+ *      - packet_size       : size of the packet in bytes.
>+ *      - rtp_seq_number    : the RTP number of the packet.
>+ *      - arr_ts            : the arrival time of the packet (from NetEq)
>+ *                            in samples.
>+ *
>+ * Return value             : 0 - Ok
>+ *                           -1 - Error
>+ */
>+
>+int16_t WebRtcIsacfix_UpdateBwEstimate1(ISACFIX_MainStruct* ISAC_main_inst,
>+                                        const uint8_t* encoded,
>+                                        size_t packet_size,
>+                                        uint16_t rtp_seq_number,
>+                                        uint32_t arr_ts);
>+
>+/****************************************************************************
>+ * WebRtcIsacfix_UpdateBwEstimate(...)
>+ *
>+ * This function updates the estimate of the bandwidth.
>+ *
>+ * Input:
>+ *      - ISAC_main_inst    : ISAC instance.
>+ *      - encoded           : encoded ISAC frame(s).
>+ *      - packet_size       : size of the packet in bytes.
>+ *      - rtp_seq_number    : the RTP number of the packet.
>+ *      - send_ts           : the send time of the packet from RTP header,
>+ *                            in samples.
>+ *      - arr_ts            : the arrival time of the packet (from NetEq)
>+ *                            in samples.
>+ *
>+ * Return value             :  0 - Ok
>+ *                            -1 - Error
>+ */
>+
>+int16_t WebRtcIsacfix_UpdateBwEstimate(ISACFIX_MainStruct* ISAC_main_inst,
>+                                       const uint8_t* encoded,
>+                                       size_t packet_size,
>+                                       uint16_t rtp_seq_number,
>+                                       uint32_t send_ts,
>+                                       uint32_t arr_ts);
>+
>+/****************************************************************************
>+ * WebRtcIsacfix_Decode(...)
>+ *
>+ * This function decodes an ISAC frame. Output speech length
>+ * will be a multiple of 480 samples: 480 or 960 samples,
>+ * depending on the framesize (30 or 60 ms).
>+ *
>+ * Input:
>+ *      - ISAC_main_inst    : ISAC instance.
>+ *      - encoded           : encoded ISAC frame(s)
>+ *      - len               : bytes in encoded vector
>+ *
>+ * Output:
>+ *      - decoded           : The decoded vector
>+ *
>+ * Return value             : >0 - number of samples in decoded vector
>+ *                            -1 - Error
>+ */
>+
>+int WebRtcIsacfix_Decode(ISACFIX_MainStruct* ISAC_main_inst,
>+                         const uint8_t* encoded,
>+                         size_t len,
>+                         int16_t* decoded,
>+                         int16_t* speechType);
>+
>+/****************************************************************************
>+ * WebRtcIsacfix_DecodeNb(...)
>+ *
>+ * This function decodes a ISAC frame in narrow-band (8 kHz sampling).
>+ * Output speech length will be a multiple of 240 samples: 240 or 480 samples,
>+ * depending on the framesize (30 or 60 ms).
>+ *
>+ * The function is enabled if WEBRTC_ISAC_FIX_NB_CALLS_ENABLED is defined
>+ *
>+ * Input:
>+ *      - ISAC_main_inst    : ISAC instance.
>+ *      - encoded           : encoded ISAC frame(s)
>+ *      - len               : bytes in encoded vector
>+ *
>+ * Output:
>+ *      - decoded           : The decoded vector
>+ *
>+ * Return value             : >0 - number of samples in decoded vector
>+ *                            -1 - Error
>+ */
> 
> #ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
>-  int WebRtcIsacfix_DecodeNb(ISACFIX_MainStruct *ISAC_main_inst,
>-                             const uint16_t *encoded,
>-                             size_t len,
>-                             int16_t *decoded,
>-                             int16_t *speechType);
>-#endif //  WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
>-
>-
>-  /****************************************************************************
>-   * WebRtcIsacfix_DecodePlcNb(...)
>-   *
>-   * This function conducts PLC for ISAC frame(s) in narrow-band (8kHz sampling).
>-   * Output speech length  will be "240*noOfLostFrames" samples
>-   * that equevalent of "30*noOfLostFrames" millisecond.
>-   *
>-   * The function is enabled if WEBRTC_ISAC_FIX_NB_CALLS_ENABLED is defined
>-   *
>-   * Input:
>-   *      - ISAC_main_inst    : ISAC instance.
>-   *      - noOfLostFrames    : Number of PLC frames (240 sample=30ms) to produce
>-   *                            NOTE! Maximum number is 2 (480 samples = 60ms)
>-   *
>-   * Output:
>-   *      - decoded           : The decoded vector
>-   *
>-   * Return value             : Number of samples in decoded PLC vector
>-   */
>+int WebRtcIsacfix_DecodeNb(ISACFIX_MainStruct* ISAC_main_inst,
>+                           const uint16_t* encoded,
>+                           size_t len,
>+                           int16_t* decoded,
>+                           int16_t* speechType);
>+#endif  //  WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
>+
>+/****************************************************************************
>+ * WebRtcIsacfix_DecodePlcNb(...)
>+ *
>+ * This function conducts PLC for ISAC frame(s) in narrow-band (8kHz sampling).
>+ * Output speech length  will be "240*noOfLostFrames" samples
>+ * that equevalent of "30*noOfLostFrames" millisecond.
>+ *
>+ * The function is enabled if WEBRTC_ISAC_FIX_NB_CALLS_ENABLED is defined
>+ *
>+ * Input:
>+ *      - ISAC_main_inst    : ISAC instance.
>+ *      - noOfLostFrames    : Number of PLC frames (240 sample=30ms) to produce
>+ *                            NOTE! Maximum number is 2 (480 samples = 60ms)
>+ *
>+ * Output:
>+ *      - decoded           : The decoded vector
>+ *
>+ * Return value             : Number of samples in decoded PLC vector
>+ */
> 
> #ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
>-  size_t WebRtcIsacfix_DecodePlcNb(ISACFIX_MainStruct *ISAC_main_inst,
>-                                   int16_t *decoded,
>-                                   size_t noOfLostFrames);
>-#endif // WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
>-
>-
>-
>-
>-  /****************************************************************************
>-   * WebRtcIsacfix_DecodePlc(...)
>-   *
>-   * This function conducts PLC for ISAC frame(s) in wide-band (16kHz sampling).
>-   * Output speech length  will be "480*noOfLostFrames" samples
>-   * that is equevalent of "30*noOfLostFrames" millisecond.
>-   *
>-   * Input:
>-   *      - ISAC_main_inst    : ISAC instance.
>-   *      - noOfLostFrames    : Number of PLC frames (480sample = 30ms)
>-   *                            to produce
>-   *                            NOTE! Maximum number is 2 (960 samples = 60ms)
>-   *
>-   * Output:
>-   *      - decoded           : The decoded vector
>-   *
>-   * Return value             : Number of samples in decoded PLC vector
>-   */
>-
>-  size_t WebRtcIsacfix_DecodePlc(ISACFIX_MainStruct *ISAC_main_inst,
>-                                 int16_t *decoded,
>-                                 size_t noOfLostFrames );
>-
>-
>-  /****************************************************************************
>-   * WebRtcIsacfix_ReadFrameLen(...)
>-   *
>-   * This function returns the length of the frame represented in the packet.
>-   *
>-   * Input:
>-   *      - encoded           : Encoded bitstream
>-   *      - encoded_len_bytes : Length of the bitstream in bytes.
>-   *
>-   * Output:
>-   *      - frameLength       : Length of frame in packet (in samples)
>-   *
>-   */
>-
>-  int16_t WebRtcIsacfix_ReadFrameLen(const uint8_t* encoded,
>-                                     size_t encoded_len_bytes,
>-                                     size_t* frameLength);
>-
>-  /****************************************************************************
>-   * WebRtcIsacfix_Control(...)
>-   *
>-   * This function sets the limit on the short-term average bit rate and the
>-   * frame length. Should be used only in Instantaneous mode.
>-   *
>-   * Input:
>-   *      - ISAC_main_inst    : ISAC instance.
>-   *      - rate              : limit on the short-term average bit rate,
>-   *                            in bits/second (between 10000 and 32000)
>-   *      - framesize         : number of milliseconds per frame (30 or 60)
>-   *
>-   * Return value             : 0  - ok
>-   *                           -1 - Error
>-   */
>-
>-  int16_t WebRtcIsacfix_Control(ISACFIX_MainStruct *ISAC_main_inst,
>-                                int16_t rate,
>-                                int framesize);
>-
>-  void WebRtcIsacfix_SetInitialBweBottleneck(ISACFIX_MainStruct* ISAC_main_inst,
>-                                             int bottleneck_bits_per_second);
>-
>-  /****************************************************************************
>-   * WebRtcIsacfix_ControlBwe(...)
>-   *
>-   * This function sets the initial values of bottleneck and frame-size if
>-   * iSAC is used in channel-adaptive mode. Through this API, users can
>-   * enforce a frame-size for all values of bottleneck. Then iSAC will not
>-   * automatically change the frame-size.
>-   *
>-   *
>-   * Input:
>-   *      - ISAC_main_inst    : ISAC instance.
>-   *      - rateBPS           : initial value of bottleneck in bits/second
>-   *                            10000 <= rateBPS <= 32000 is accepted
>-   *      - frameSizeMs       : number of milliseconds per frame (30 or 60)
>-   *      - enforceFrameSize  : 1 to enforce the given frame-size through out
>-   *                            the adaptation process, 0 to let iSAC change
>-   *                            the frame-size if required.
>-   *
>-   * Return value             : 0  - ok
>-   *                           -1 - Error
>-   */
>-
>-  int16_t WebRtcIsacfix_ControlBwe(ISACFIX_MainStruct *ISAC_main_inst,
>-                                   int16_t rateBPS,
>-                                   int frameSizeMs,
>-                                   int16_t enforceFrameSize);
>-
>-
>-
>-  /****************************************************************************
>-   * WebRtcIsacfix_version(...)
>-   *
>-   * This function returns the version number.
>-   *
>-   * Output:
>-   *      - version      : Pointer to character string
>-   *
>-   */
>-
>-  void WebRtcIsacfix_version(char *version);
>-
>-
>-  /****************************************************************************
>-   * WebRtcIsacfix_GetErrorCode(...)
>-   *
>-   * This function can be used to check the error code of an iSAC instance. When
>-   * a function returns -1 a error code will be set for that instance. The
>-   * function below extract the code of the last error that occured in the
>-   * specified instance.
>-   *
>-   * Input:
>-   *  - ISAC_main_inst        : ISAC instance
>-   *
>-   * Return value             : Error code
>-   */
>-
>-  int16_t WebRtcIsacfix_GetErrorCode(ISACFIX_MainStruct *ISAC_main_inst);
>-
>-
>-  /****************************************************************************
>-   * WebRtcIsacfix_GetUplinkBw(...)
>-   *
>-   * This function return iSAC send bitrate
>-   *
>-   * Input:
>-   *      - ISAC_main_inst    : iSAC instance
>-   *
>-   * Return value             : <0 Error code
>-   *                            else bitrate
>-   */
>-
>-  int32_t WebRtcIsacfix_GetUplinkBw(ISACFIX_MainStruct *ISAC_main_inst);
>-
>-
>-  /****************************************************************************
>-   * WebRtcIsacfix_SetMaxPayloadSize(...)
>-   *
>-   * This function sets a limit for the maximum payload size of iSAC. The same
>-   * value is used both for 30 and 60 msec packets.
>-   * The absolute max will be valid until next time the function is called.
>-   * NOTE! This function may override the function WebRtcIsacfix_SetMaxRate()
>-   *
>-   * Input:
>-   *      - ISAC_main_inst    : iSAC instance
>-   *      - maxPayloadBytes   : maximum size of the payload in bytes
>-   *                            valid values are between 100 and 400 bytes
>-   *
>-   *
>-   * Return value             : 0 if sucessful
>-   *                           -1 if error happens
>-   */
>-
>-  int16_t WebRtcIsacfix_SetMaxPayloadSize(ISACFIX_MainStruct *ISAC_main_inst,
>-                                          int16_t maxPayloadBytes);
>-
>-
>-  /****************************************************************************
>-   * WebRtcIsacfix_SetMaxRate(...)
>-   *
>-   * This function sets the maximum rate which the codec may not exceed for a
>-   * singel packet. The maximum rate is set in bits per second.
>-   * The codec has an absolute maximum rate of 53400 bits per second (200 bytes
>-   * per 30 msec).
>-   * It is possible to set a maximum rate between 32000 and 53400 bits per second.
>-   *
>-   * The rate limit is valid until next time the function is called.
>-   *
>-   * NOTE! Packet size will never go above the value set if calling
>-   * WebRtcIsacfix_SetMaxPayloadSize() (default max packet size is 400 bytes).
>-   *
>-   * Input:
>-   *      - ISAC_main_inst    : iSAC instance
>-   *      - maxRateInBytes    : maximum rate in bits per second,
>-   *                            valid values are 32000 to 53400 bits
>-   *
>-   * Return value             : 0 if sucessful
>-   *                           -1 if error happens
>-   */
>-
>-  int16_t WebRtcIsacfix_SetMaxRate(ISACFIX_MainStruct *ISAC_main_inst,
>-                                   int32_t maxRate);
>-
>-  /****************************************************************************
>-   * WebRtcIsacfix_CreateInternal(...)
>-   *
>-   * This function creates the memory that is used to store data in the encoder
>-   *
>-   * Input:
>-   *      - *ISAC_main_inst   : a pointer to the coder instance.
>-   *
>-   * Return value             : 0 - Ok
>-   *                           -1 - Error
>-   */
>-
>-  int16_t WebRtcIsacfix_CreateInternal(ISACFIX_MainStruct *ISAC_main_inst);
>-
>-
>-  /****************************************************************************
>-   * WebRtcIsacfix_FreeInternal(...)
>-   *
>-   * This function frees the internal memory for storing encoder data.
>-   *
>-   * Input:
>-   *      - ISAC_main_inst        : an ISAC instance.
>-   *
>-   * Return value                 :  0 - Ok
>-   *                                -1 - Error
>-   */
>-
>-  int16_t WebRtcIsacfix_FreeInternal(ISACFIX_MainStruct *ISAC_main_inst);
>-
>-
>-  /****************************************************************************
>-   * WebRtcIsacfix_GetNewBitStream(...)
>-   *
>-   * This function returns encoded data, with the recieved bwe-index in the
>-   * stream. It should always return a complete packet, i.e. only called once
>-   * even for 60 msec frames
>-   *
>-   * Input:
>-   *      - ISAC_main_inst    : ISAC instance.
>-   *      - bweIndex          : index of bandwidth estimate to put in new bitstream
>-   *      - scale             : factor for rate change (0.4 ~=> half the rate, 1 no change).
>-   *
>-   * Output:
>-   *      - encoded           : the encoded data vector
>-   *
>-   * Return value             : >0 - Length (in bytes) of coded data
>-   *                            -1 - Error
>-   */
>-
>-  int16_t WebRtcIsacfix_GetNewBitStream(ISACFIX_MainStruct *ISAC_main_inst,
>-                                        int16_t          bweIndex,
>-                                        float              scale,
>-                                        uint8_t* encoded);
>-
>-
>-  /****************************************************************************
>-   * WebRtcIsacfix_GetDownLinkBwIndex(...)
>-   *
>-   * This function returns index representing the Bandwidth estimate from
>-   * other side to this side.
>-   *
>-   * Input:
>-   *      - ISAC_main_inst    : iSAC struct
>-   *
>-   * Output:
>-   *      - rateIndex         : Bandwidth estimate to transmit to other side.
>-   *
>-   */
>-
>-  int16_t WebRtcIsacfix_GetDownLinkBwIndex(ISACFIX_MainStruct* ISAC_main_inst,
>-                                           int16_t*     rateIndex);
>-
>-
>-  /****************************************************************************
>-   * WebRtcIsacfix_UpdateUplinkBw(...)
>-   *
>-   * This function takes an index representing the Bandwidth estimate from
>-   * this side to other side and updates BWE.
>-   *
>-   * Input:
>-   *      - ISAC_main_inst    : iSAC struct
>-   *      - rateIndex         : Bandwidth estimate from other side.
>-   *
>-   */
>-
>-  int16_t WebRtcIsacfix_UpdateUplinkBw(ISACFIX_MainStruct* ISAC_main_inst,
>-                                       int16_t     rateIndex);
>-
>-
>-  /****************************************************************************
>-   * WebRtcIsacfix_ReadBwIndex(...)
>-   *
>-   * This function returns the index of the Bandwidth estimate from the bitstream.
>-   *
>-   * Input:
>-   *      - encoded           : Encoded bitstream
>-   *      - encoded_len_bytes : Length of the bitstream in bytes.
>-   *
>-   * Output:
>-   *      - rateIndex         : Bandwidth estimate in bitstream
>-   *
>-   */
>-
>-  int16_t WebRtcIsacfix_ReadBwIndex(const uint8_t* encoded,
>-                                    size_t encoded_len_bytes,
>-                                    int16_t* rateIndex);
>-
>-
>-  /****************************************************************************
>-   * WebRtcIsacfix_GetNewFrameLen(...)
>-   *
>-   * This function return the next frame length (in samples) of iSAC.
>-   *
>-   * Input:
>-   *      -ISAC_main_inst     : iSAC instance
>-   *
>-   * Return value             : frame lenght in samples
>-   */
>-
>-  int16_t WebRtcIsacfix_GetNewFrameLen(ISACFIX_MainStruct *ISAC_main_inst);
>-
>-  /* Fills in an IsacBandwidthInfo struct. */
>-  void WebRtcIsacfix_GetBandwidthInfo(ISACFIX_MainStruct* ISAC_main_inst,
>-                                      IsacBandwidthInfo* bwinfo);
>-
>-  /* Uses the values from an IsacBandwidthInfo struct. */
>-  void WebRtcIsacfix_SetBandwidthInfo(ISACFIX_MainStruct* ISAC_main_inst,
>-                                      const IsacBandwidthInfo* bwinfo);
>+size_t WebRtcIsacfix_DecodePlcNb(ISACFIX_MainStruct* ISAC_main_inst,
>+                                 int16_t* decoded,
>+                                 size_t noOfLostFrames);
>+#endif  // WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
>+
>+/****************************************************************************
>+ * WebRtcIsacfix_DecodePlc(...)
>+ *
>+ * This function conducts PLC for ISAC frame(s) in wide-band (16kHz sampling).
>+ * Output speech length  will be "480*noOfLostFrames" samples
>+ * that is equevalent of "30*noOfLostFrames" millisecond.
>+ *
>+ * Input:
>+ *      - ISAC_main_inst    : ISAC instance.
>+ *      - noOfLostFrames    : Number of PLC frames (480sample = 30ms)
>+ *                            to produce
>+ *                            NOTE! Maximum number is 2 (960 samples = 60ms)
>+ *
>+ * Output:
>+ *      - decoded           : The decoded vector
>+ *
>+ * Return value             : Number of samples in decoded PLC vector
>+ */
>+
>+size_t WebRtcIsacfix_DecodePlc(ISACFIX_MainStruct* ISAC_main_inst,
>+                               int16_t* decoded,
>+                               size_t noOfLostFrames);
>+
>+/****************************************************************************
>+ * WebRtcIsacfix_ReadFrameLen(...)
>+ *
>+ * This function returns the length of the frame represented in the packet.
>+ *
>+ * Input:
>+ *      - encoded           : Encoded bitstream
>+ *      - encoded_len_bytes : Length of the bitstream in bytes.
>+ *
>+ * Output:
>+ *      - frameLength       : Length of frame in packet (in samples)
>+ *
>+ */
>+
>+int16_t WebRtcIsacfix_ReadFrameLen(const uint8_t* encoded,
>+                                   size_t encoded_len_bytes,
>+                                   size_t* frameLength);
>+
>+/****************************************************************************
>+ * WebRtcIsacfix_Control(...)
>+ *
>+ * This function sets the limit on the short-term average bit rate and the
>+ * frame length. Should be used only in Instantaneous mode.
>+ *
>+ * Input:
>+ *      - ISAC_main_inst    : ISAC instance.
>+ *      - rate              : limit on the short-term average bit rate,
>+ *                            in bits/second (between 10000 and 32000)
>+ *      - framesize         : number of milliseconds per frame (30 or 60)
>+ *
>+ * Return value             : 0  - ok
>+ *                           -1 - Error
>+ */
>+
>+int16_t WebRtcIsacfix_Control(ISACFIX_MainStruct* ISAC_main_inst,
>+                              int16_t rate,
>+                              int framesize);
>+
>+void WebRtcIsacfix_SetInitialBweBottleneck(ISACFIX_MainStruct* ISAC_main_inst,
>+                                           int bottleneck_bits_per_second);
>+
>+/****************************************************************************
>+ * WebRtcIsacfix_ControlBwe(...)
>+ *
>+ * This function sets the initial values of bottleneck and frame-size if
>+ * iSAC is used in channel-adaptive mode. Through this API, users can
>+ * enforce a frame-size for all values of bottleneck. Then iSAC will not
>+ * automatically change the frame-size.
>+ *
>+ *
>+ * Input:
>+ *      - ISAC_main_inst    : ISAC instance.
>+ *      - rateBPS           : initial value of bottleneck in bits/second
>+ *                            10000 <= rateBPS <= 32000 is accepted
>+ *      - frameSizeMs       : number of milliseconds per frame (30 or 60)
>+ *      - enforceFrameSize  : 1 to enforce the given frame-size through out
>+ *                            the adaptation process, 0 to let iSAC change
>+ *                            the frame-size if required.
>+ *
>+ * Return value             : 0  - ok
>+ *                           -1 - Error
>+ */
>+
>+int16_t WebRtcIsacfix_ControlBwe(ISACFIX_MainStruct* ISAC_main_inst,
>+                                 int16_t rateBPS,
>+                                 int frameSizeMs,
>+                                 int16_t enforceFrameSize);
>+
>+/****************************************************************************
>+ * WebRtcIsacfix_version(...)
>+ *
>+ * This function returns the version number.
>+ *
>+ * Output:
>+ *      - version      : Pointer to character string
>+ *
>+ */
>+
>+void WebRtcIsacfix_version(char* version);
>+
>+/****************************************************************************
>+ * WebRtcIsacfix_GetErrorCode(...)
>+ *
>+ * This function can be used to check the error code of an iSAC instance. When
>+ * a function returns -1 a error code will be set for that instance. The
>+ * function below extract the code of the last error that occured in the
>+ * specified instance.
>+ *
>+ * Input:
>+ *  - ISAC_main_inst        : ISAC instance
>+ *
>+ * Return value             : Error code
>+ */
>+
>+int16_t WebRtcIsacfix_GetErrorCode(ISACFIX_MainStruct* ISAC_main_inst);
>+
>+/****************************************************************************
>+ * WebRtcIsacfix_GetUplinkBw(...)
>+ *
>+ * This function return iSAC send bitrate
>+ *
>+ * Input:
>+ *      - ISAC_main_inst    : iSAC instance
>+ *
>+ * Return value             : <0 Error code
>+ *                            else bitrate
>+ */
>+
>+int32_t WebRtcIsacfix_GetUplinkBw(ISACFIX_MainStruct* ISAC_main_inst);
>+
>+/****************************************************************************
>+ * WebRtcIsacfix_SetMaxPayloadSize(...)
>+ *
>+ * This function sets a limit for the maximum payload size of iSAC. The same
>+ * value is used both for 30 and 60 msec packets.
>+ * The absolute max will be valid until next time the function is called.
>+ * NOTE! This function may override the function WebRtcIsacfix_SetMaxRate()
>+ *
>+ * Input:
>+ *      - ISAC_main_inst    : iSAC instance
>+ *      - maxPayloadBytes   : maximum size of the payload in bytes
>+ *                            valid values are between 100 and 400 bytes
>+ *
>+ *
>+ * Return value             : 0 if sucessful
>+ *                           -1 if error happens
>+ */
>+
>+int16_t WebRtcIsacfix_SetMaxPayloadSize(ISACFIX_MainStruct* ISAC_main_inst,
>+                                        int16_t maxPayloadBytes);
>+
>+/****************************************************************************
>+ * WebRtcIsacfix_SetMaxRate(...)
>+ *
>+ * This function sets the maximum rate which the codec may not exceed for a
>+ * singel packet. The maximum rate is set in bits per second.
>+ * The codec has an absolute maximum rate of 53400 bits per second (200 bytes
>+ * per 30 msec).
>+ * It is possible to set a maximum rate between 32000 and 53400 bits per second.
>+ *
>+ * The rate limit is valid until next time the function is called.
>+ *
>+ * NOTE! Packet size will never go above the value set if calling
>+ * WebRtcIsacfix_SetMaxPayloadSize() (default max packet size is 400 bytes).
>+ *
>+ * Input:
>+ *      - ISAC_main_inst    : iSAC instance
>+ *      - maxRateInBytes    : maximum rate in bits per second,
>+ *                            valid values are 32000 to 53400 bits
>+ *
>+ * Return value             : 0 if sucessful
>+ *                           -1 if error happens
>+ */
>+
>+int16_t WebRtcIsacfix_SetMaxRate(ISACFIX_MainStruct* ISAC_main_inst,
>+                                 int32_t maxRate);
>+
>+/****************************************************************************
>+ * WebRtcIsacfix_CreateInternal(...)
>+ *
>+ * This function creates the memory that is used to store data in the encoder
>+ *
>+ * Input:
>+ *      - *ISAC_main_inst   : a pointer to the coder instance.
>+ *
>+ * Return value             : 0 - Ok
>+ *                           -1 - Error
>+ */
>+
>+int16_t WebRtcIsacfix_CreateInternal(ISACFIX_MainStruct* ISAC_main_inst);
>+
>+/****************************************************************************
>+ * WebRtcIsacfix_FreeInternal(...)
>+ *
>+ * This function frees the internal memory for storing encoder data.
>+ *
>+ * Input:
>+ *      - ISAC_main_inst        : an ISAC instance.
>+ *
>+ * Return value                 :  0 - Ok
>+ *                                -1 - Error
>+ */
>+
>+int16_t WebRtcIsacfix_FreeInternal(ISACFIX_MainStruct* ISAC_main_inst);
>+
>+/****************************************************************************
>+ * WebRtcIsacfix_GetNewBitStream(...)
>+ *
>+ * This function returns encoded data, with the recieved bwe-index in the
>+ * stream. It should always return a complete packet, i.e. only called once
>+ * even for 60 msec frames
>+ *
>+ * Input:
>+ *      - ISAC_main_inst    : ISAC instance.
>+ *      - bweIndex          : index of bandwidth estimate to put in new
>+ * bitstream - scale             : factor for rate change (0.4 ~=> half the
>+ * rate, 1 no change).
>+ *
>+ * Output:
>+ *      - encoded           : the encoded data vector
>+ *
>+ * Return value             : >0 - Length (in bytes) of coded data
>+ *                            -1 - Error
>+ */
>+
>+int16_t WebRtcIsacfix_GetNewBitStream(ISACFIX_MainStruct* ISAC_main_inst,
>+                                      int16_t bweIndex,
>+                                      float scale,
>+                                      uint8_t* encoded);
>+
>+/****************************************************************************
>+ * WebRtcIsacfix_GetDownLinkBwIndex(...)
>+ *
>+ * This function returns index representing the Bandwidth estimate from
>+ * other side to this side.
>+ *
>+ * Input:
>+ *      - ISAC_main_inst    : iSAC struct
>+ *
>+ * Output:
>+ *      - rateIndex         : Bandwidth estimate to transmit to other side.
>+ *
>+ */
>+
>+int16_t WebRtcIsacfix_GetDownLinkBwIndex(ISACFIX_MainStruct* ISAC_main_inst,
>+                                         int16_t* rateIndex);
>+
>+/****************************************************************************
>+ * WebRtcIsacfix_UpdateUplinkBw(...)
>+ *
>+ * This function takes an index representing the Bandwidth estimate from
>+ * this side to other side and updates BWE.
>+ *
>+ * Input:
>+ *      - ISAC_main_inst    : iSAC struct
>+ *      - rateIndex         : Bandwidth estimate from other side.
>+ *
>+ */
>+
>+int16_t WebRtcIsacfix_UpdateUplinkBw(ISACFIX_MainStruct* ISAC_main_inst,
>+                                     int16_t rateIndex);
>+
>+/****************************************************************************
>+ * WebRtcIsacfix_ReadBwIndex(...)
>+ *
>+ * This function returns the index of the Bandwidth estimate from the bitstream.
>+ *
>+ * Input:
>+ *      - encoded           : Encoded bitstream
>+ *      - encoded_len_bytes : Length of the bitstream in bytes.
>+ *
>+ * Output:
>+ *      - rateIndex         : Bandwidth estimate in bitstream
>+ *
>+ */
>+
>+int16_t WebRtcIsacfix_ReadBwIndex(const uint8_t* encoded,
>+                                  size_t encoded_len_bytes,
>+                                  int16_t* rateIndex);
>+
>+/****************************************************************************
>+ * WebRtcIsacfix_GetNewFrameLen(...)
>+ *
>+ * This function return the next frame length (in samples) of iSAC.
>+ *
>+ * Input:
>+ *      -ISAC_main_inst     : iSAC instance
>+ *
>+ * Return value             : frame lenght in samples
>+ */
>+
>+int16_t WebRtcIsacfix_GetNewFrameLen(ISACFIX_MainStruct* ISAC_main_inst);
>+
>+/* Fills in an IsacBandwidthInfo struct. */
>+void WebRtcIsacfix_GetBandwidthInfo(ISACFIX_MainStruct* ISAC_main_inst,
>+                                    IsacBandwidthInfo* bwinfo);
>+
>+/* Uses the values from an IsacBandwidthInfo struct. */
>+void WebRtcIsacfix_SetBandwidthInfo(ISACFIX_MainStruct* ISAC_main_inst,
>+                                    const IsacBandwidthInfo* bwinfo);
> 
> #if defined(__cplusplus)
> }
> #endif
> 
>-
>-
> #endif /* MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INCLUDE_ISACFIX_H_ */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines.c
>index 5925d68e353..eaeef50f047 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines.c
>@@ -16,7 +16,7 @@
>  *
>  */
> 
>-#include "arith_routins.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/arith_routins.h"
> 
> 
> /****************************************************************************
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_hist.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_hist.c
>index f1eb5ae8f75..cad3056b376 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_hist.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_hist.c
>@@ -15,7 +15,7 @@
>  *
>  */
> 
>-#include "arith_routins.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/arith_routins.h"
> 
> 
> /****************************************************************************
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_logist.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_logist.c
>index 808aeb7fd9d..8e979604613 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_logist.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routines_logist.c
>@@ -15,7 +15,7 @@
>  *
>  */
> 
>-#include "arith_routins.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/arith_routins.h"
> 
> /* Tables for piecewise linear cdf functions: y = k*x */
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routins.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routins.h
>index 830c52fcaf1..cc4ed555cf8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routins.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/arith_routins.h
>@@ -18,8 +18,7 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ARITH_ROUTINS_H_
> #define MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ARITH_ROUTINS_H_
> 
>-#include "structs.h"
>-
>+#include "modules/audio_coding/codecs/isac/fix/source/structs.h"
> 
> /****************************************************************************
>  * WebRtcIsacfix_EncLogisticMulti2(...)
>@@ -36,12 +35,10 @@
>  * Return value             :  0 if ok,
>  *                             <0 otherwise.
>  */
>-int WebRtcIsacfix_EncLogisticMulti2(
>-    Bitstr_enc *streamData,
>-    int16_t *dataQ7,
>-    const uint16_t *env,
>-    const int16_t lenData);
>-
>+int WebRtcIsacfix_EncLogisticMulti2(Bitstr_enc* streamData,
>+                                    int16_t* dataQ7,
>+                                    const uint16_t* env,
>+                                    const int16_t lenData);
> 
> /****************************************************************************
>  * WebRtcIsacfix_EncTerminate(...)
>@@ -54,8 +51,7 @@ int WebRtcIsacfix_EncLogisticMulti2(
>  *
>  * Return value             : number of bytes in the stream
>  */
>-int16_t WebRtcIsacfix_EncTerminate(Bitstr_enc *streamData);
>-
>+int16_t WebRtcIsacfix_EncTerminate(Bitstr_enc* streamData);
> 
> /****************************************************************************
>  * WebRtcIsacfix_DecLogisticMulti2(...)
>@@ -74,12 +70,10 @@ int16_t WebRtcIsacfix_EncTerminate(Bitstr_enc *streamData);
>  * Return value             : number of bytes in the stream so far
>  *                            <0 if error detected
>  */
>-int WebRtcIsacfix_DecLogisticMulti2(
>-    int16_t *data,
>-    Bitstr_dec *streamData,
>-    const int32_t *env,
>-    const int16_t lenData);
>-
>+int WebRtcIsacfix_DecLogisticMulti2(int16_t* data,
>+                                    Bitstr_dec* streamData,
>+                                    const int32_t* env,
>+                                    const int16_t lenData);
> 
> /****************************************************************************
>  * WebRtcIsacfix_EncHistMulti(...)
>@@ -95,12 +89,10 @@ int WebRtcIsacfix_DecLogisticMulti2(
>  * Return value             : 0 if ok
>  *                            <0 if error detected
>  */
>-int WebRtcIsacfix_EncHistMulti(
>-    Bitstr_enc *streamData,
>-    const int16_t *data,
>-    const uint16_t *const *cdf,
>-    const int16_t lenData);
>-
>+int WebRtcIsacfix_EncHistMulti(Bitstr_enc* streamData,
>+                               const int16_t* data,
>+                               const uint16_t* const* cdf,
>+                               const int16_t lenData);
> 
> /****************************************************************************
>  * WebRtcIsacfix_DecHistBisectMulti(...)
>@@ -122,13 +114,11 @@ int WebRtcIsacfix_EncHistMulti(
>  * Return value             : number of bytes in the stream
>  *                            <0 if error detected
>  */
>-int16_t WebRtcIsacfix_DecHistBisectMulti(
>-    int16_t *data,
>-    Bitstr_dec *streamData,
>-    const uint16_t *const *cdf,
>-    const uint16_t *cdfSize,
>-    const int16_t lenData);
>-
>+int16_t WebRtcIsacfix_DecHistBisectMulti(int16_t* data,
>+                                         Bitstr_dec* streamData,
>+                                         const uint16_t* const* cdf,
>+                                         const uint16_t* cdfSize,
>+                                         const int16_t lenData);
> 
> /****************************************************************************
>  * WebRtcIsacfix_DecHistOneStepMulti(...)
>@@ -150,11 +140,10 @@ int16_t WebRtcIsacfix_DecHistBisectMulti(
>  * Return value             : number of bytes in original stream
>  *                            <0 if error detected
>  */
>-int16_t WebRtcIsacfix_DecHistOneStepMulti(
>-    int16_t *data,
>-    Bitstr_dec *streamData,
>-    const uint16_t *const *cdf,
>-    const uint16_t *initIndex,
>-    const int16_t lenData);
>+int16_t WebRtcIsacfix_DecHistOneStepMulti(int16_t* data,
>+                                          Bitstr_dec* streamData,
>+                                          const uint16_t* const* cdf,
>+                                          const uint16_t* initIndex,
>+                                          const int16_t lenData);
> 
> #endif /* MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ARITH_ROUTINS_H_ */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.c
>index d926f47e3aa..354ae3b5710 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.c
>@@ -18,9 +18,9 @@
>  *
>  */
> 
>-#include "bandwidth_estimator.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.h"
> 
>-#include "settings.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/settings.h"
> #include "rtc_base/checks.h"
> 
> /* array of quantization levels for bottle neck info; Matlab code: */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.h
>index f80a5876fab..f8ac1efae74 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.h
>@@ -19,8 +19,7 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_BANDWIDTH_ESTIMATOR_H_
> #define MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_BANDWIDTH_ESTIMATOR_H_
> 
>-#include "structs.h"
>-
>+#include "modules/audio_coding/codecs/isac/fix/source/structs.h"
> 
> /****************************************************************************
>  * WebRtcIsacfix_InitBandwidthEstimator(...)
>@@ -33,8 +32,7 @@
>  * Return value            : 0
>  */
> 
>-int32_t WebRtcIsacfix_InitBandwidthEstimator(BwEstimatorstr *bwest_str);
>-
>+int32_t WebRtcIsacfix_InitBandwidthEstimator(BwEstimatorstr* bwest_str);
> 
> /****************************************************************************
>  * WebRtcIsacfix_UpdateUplinkBwImpl(...)
>@@ -57,16 +55,17 @@ int32_t WebRtcIsacfix_InitBandwidthEstimator(BwEstimatorstr *bwest_str);
>  *                           -1 otherwise
>  */
> 
>-int32_t WebRtcIsacfix_UpdateUplinkBwImpl(BwEstimatorstr       *bwest_str,
>-                                         const uint16_t        rtp_number,
>-                                         const int16_t         frameSize,
>-                                         const uint32_t        send_ts,
>-                                         const uint32_t        arr_ts,
>-                                         const size_t          pksize,
>-                                         const uint16_t        Index);
>-
>-/* Update receiving estimates. Used when we only receive BWE index, no iSAC data packet. */
>-int16_t WebRtcIsacfix_UpdateUplinkBwRec(BwEstimatorstr *bwest_str,
>+int32_t WebRtcIsacfix_UpdateUplinkBwImpl(BwEstimatorstr* bwest_str,
>+                                         const uint16_t rtp_number,
>+                                         const int16_t frameSize,
>+                                         const uint32_t send_ts,
>+                                         const uint32_t arr_ts,
>+                                         const size_t pksize,
>+                                         const uint16_t Index);
>+
>+/* Update receiving estimates. Used when we only receive BWE index, no iSAC data
>+ * packet. */
>+int16_t WebRtcIsacfix_UpdateUplinkBwRec(BwEstimatorstr* bwest_str,
>                                         const int16_t Index);
> 
> /****************************************************************************
>@@ -81,19 +80,19 @@ int16_t WebRtcIsacfix_UpdateUplinkBwRec(BwEstimatorstr *bwest_str,
>  * Return:
>  *      bandwith and jitter index (0..23)
>  */
>-uint16_t WebRtcIsacfix_GetDownlinkBwIndexImpl(BwEstimatorstr *bwest_str);
>+uint16_t WebRtcIsacfix_GetDownlinkBwIndexImpl(BwEstimatorstr* bwest_str);
> 
> /* Returns the bandwidth estimation (in bps) */
>-uint16_t WebRtcIsacfix_GetDownlinkBandwidth(const BwEstimatorstr *bwest_str);
>+uint16_t WebRtcIsacfix_GetDownlinkBandwidth(const BwEstimatorstr* bwest_str);
> 
> /* Returns the bandwidth that iSAC should send with in bps */
>-int16_t WebRtcIsacfix_GetUplinkBandwidth(const BwEstimatorstr *bwest_str);
>+int16_t WebRtcIsacfix_GetUplinkBandwidth(const BwEstimatorstr* bwest_str);
> 
> /* Returns the max delay (in ms) */
>-int16_t WebRtcIsacfix_GetDownlinkMaxDelay(const BwEstimatorstr *bwest_str);
>+int16_t WebRtcIsacfix_GetDownlinkMaxDelay(const BwEstimatorstr* bwest_str);
> 
> /* Returns the max delay value from the other side in ms */
>-int16_t WebRtcIsacfix_GetUplinkMaxDelay(const BwEstimatorstr *bwest_str);
>+int16_t WebRtcIsacfix_GetUplinkMaxDelay(const BwEstimatorstr* bwest_str);
> 
> /* Fills in an IsacExternalBandwidthInfo struct. */
> void WebRtcIsacfixBw_GetBandwidthInfo(BwEstimatorstr* bwest_str,
>@@ -107,29 +106,31 @@ void WebRtcIsacfixBw_SetBandwidthInfo(BwEstimatorstr* bwest_str,
>  * update amount of data in bottle neck buffer and burst handling
>  * returns minimum payload size (bytes)
>  */
>-uint16_t WebRtcIsacfix_GetMinBytes(RateModel *State,
>-                                   int16_t StreamSize,     /* bytes in bitstream */
>-                                   const int16_t FrameLen,    /* ms per frame */
>-                                   const int16_t BottleNeck,        /* bottle neck rate; excl headers (bps) */
>-                                   const int16_t DelayBuildUp);     /* max delay from bottle neck buffering (ms) */
>+uint16_t WebRtcIsacfix_GetMinBytes(
>+    RateModel* State,
>+    int16_t StreamSize,          /* bytes in bitstream */
>+    const int16_t FrameLen,      /* ms per frame */
>+    const int16_t BottleNeck,    /* bottle neck rate; excl headers (bps) */
>+    const int16_t DelayBuildUp); /* max delay from bottle neck buffering (ms) */
> 
> /*
>  * update long-term average bitrate and amount of data in buffer
>  */
>-void WebRtcIsacfix_UpdateRateModel(RateModel *State,
>-                                   int16_t StreamSize,    /* bytes in bitstream */
>-                                   const int16_t FrameSamples,  /* samples per frame */
>-                                   const int16_t BottleNeck);       /* bottle neck rate; excl headers (bps) */
>+void WebRtcIsacfix_UpdateRateModel(
>+    RateModel* State,
>+    int16_t StreamSize,         /* bytes in bitstream */
>+    const int16_t FrameSamples, /* samples per frame */
>+    const int16_t BottleNeck);  /* bottle neck rate; excl headers (bps) */
> 
>-
>-void WebRtcIsacfix_InitRateModel(RateModel *State);
>+void WebRtcIsacfix_InitRateModel(RateModel* State);
> 
> /* Returns the new framelength value (input argument: bottle_neck) */
>-int16_t WebRtcIsacfix_GetNewFrameLength(int16_t bottle_neck, int16_t current_framelength);
>+int16_t WebRtcIsacfix_GetNewFrameLength(int16_t bottle_neck,
>+                                        int16_t current_framelength);
> 
> /* Returns the new SNR value (input argument: bottle_neck) */
>-//returns snr in Q10
>+// returns snr in Q10
> int16_t WebRtcIsacfix_GetSnr(int16_t bottle_neck, int16_t framesamples);
> 
>-
>-#endif /*  MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_BANDWIDTH_ESTIMATOR_H_ */
>+#endif /*  MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_BANDWIDTH_ESTIMATOR_H_ \
>+        */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h
>index c1a4b1832eb..c95b53ff64e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/codec.h
>@@ -19,7 +19,7 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_CODEC_H_
> #define MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_CODEC_H_
> 
>-#include "structs.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/structs.h"
> 
> #ifdef __cplusplus
> extern "C" {
>@@ -38,7 +38,7 @@ int WebRtcIsacfix_DecodeImpl(int16_t* signal_out16,
> 
> void WebRtcIsacfix_DecodePlcImpl(int16_t* decoded,
>                                  IsacFixDecoderInstance* ISACdec_obj,
>-                                 size_t* current_framesample );
>+                                 size_t* current_framesample);
> 
> int WebRtcIsacfix_EncodeImpl(int16_t* in,
>                              IsacFixEncoderInstance* ISACenc_obj,
>@@ -64,7 +64,6 @@ void WebRtcIsacfix_InitPitchAnalysis(PitchAnalysisStruct* State);
> 
> void WebRtcIsacfix_InitPlc(PLCstr* State);
> 
>-
> /* transform functions */
> 
> void WebRtcIsacfix_InitTransform(void);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/decode.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/decode.c
>index cf2d507cb57..144208818a5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/decode.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/decode.c
>@@ -17,12 +17,12 @@
> 
> #include <string.h>
> 
>-#include "bandwidth_estimator.h"
>-#include "codec.h"
>-#include "entropy_coding.h"
>-#include "pitch_estimator.h"
>-#include "settings.h"
>-#include "structs.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/codec.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/entropy_coding.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/settings.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/structs.h"
> 
> 
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/decode_bwe.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/decode_bwe.c
>index 316f59a5e2f..99676504cdd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/decode_bwe.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/decode_bwe.c
>@@ -16,10 +16,10 @@
>  */
> 
> 
>-#include "bandwidth_estimator.h"
>-#include "codec.h"
>-#include "entropy_coding.h"
>-#include "structs.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/codec.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/entropy_coding.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/structs.h"
> 
> 
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/decode_plc.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/decode_plc.c
>index e907f2b6a6b..873cf951ba3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/decode_plc.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/decode_plc.c
>@@ -17,12 +17,12 @@
> 
> #include <string.h>
> 
>-#include "settings.h"
>-#include "entropy_coding.h"
>-#include "pitch_estimator.h"
>-#include "bandwidth_estimator.h"
>-#include "structs.h"
>-#include "codec.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/settings.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/entropy_coding.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/bandwidth_estimator.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/structs.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/codec.h"
> 
> 
> #define NO_OF_PRIMES 8
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c
>index ed816c0ad70..cb154456d4e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c
>@@ -18,14 +18,14 @@
> 
> #include <stddef.h>
> 
>-#include "arith_routins.h"
>-#include "spectrum_ar_model_tables.h"
>-#include "pitch_gain_tables.h"
>-#include "pitch_lag_tables.h"
>-#include "entropy_coding.h"
>-#include "lpc_tables.h"
>-#include "settings.h"
>-#include "signal_processing_library.h"
>+#include "common_audio/signal_processing/include/signal_processing_library.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/arith_routins.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/entropy_coding.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/lpc_tables.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/pitch_gain_tables.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/pitch_lag_tables.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/settings.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/spectrum_ar_model_tables.h"
> #include "rtc_base/sanitizer.h"
> 
> /*
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.h
>index 31f044de0b6..b4251cee1ea 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.h
>@@ -19,94 +19,82 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ENTROPY_CODING_H_
> #define MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ENTROPY_CODING_H_
> 
>-#include "structs.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/structs.h"
> 
> /* decode complex spectrum (return number of bytes in stream) */
>-int WebRtcIsacfix_DecodeSpec(Bitstr_dec  *streamdata,
>-                             int16_t *frQ7,
>-                             int16_t *fiQ7,
>+int WebRtcIsacfix_DecodeSpec(Bitstr_dec* streamdata,
>+                             int16_t* frQ7,
>+                             int16_t* fiQ7,
>                              int16_t AvgPitchGain_Q12);
> 
> /* encode complex spectrum */
>-int WebRtcIsacfix_EncodeSpec(const int16_t *fr,
>-                             const int16_t *fi,
>-                             Bitstr_enc *streamdata,
>+int WebRtcIsacfix_EncodeSpec(const int16_t* fr,
>+                             const int16_t* fi,
>+                             Bitstr_enc* streamdata,
>                              int16_t AvgPitchGain_Q12);
> 
>-
> /* decode & dequantize LPC Coef */
>-int WebRtcIsacfix_DecodeLpcCoef(Bitstr_dec  *streamdata,
>-                                int32_t *LPCCoefQ17,
>-                                int32_t *gain_lo_hiQ17,
>-                                int16_t *outmodel);
>+int WebRtcIsacfix_DecodeLpcCoef(Bitstr_dec* streamdata,
>+                                int32_t* LPCCoefQ17,
>+                                int32_t* gain_lo_hiQ17,
>+                                int16_t* outmodel);
> 
>-int WebRtcIsacfix_DecodeLpc(int32_t *gain_lo_hiQ17,
>-                            int16_t *LPCCoef_loQ15,
>-                            int16_t *LPCCoef_hiQ15,
>-                            Bitstr_dec  *streamdata,
>-                            int16_t *outmodel);
>+int WebRtcIsacfix_DecodeLpc(int32_t* gain_lo_hiQ17,
>+                            int16_t* LPCCoef_loQ15,
>+                            int16_t* LPCCoef_hiQ15,
>+                            Bitstr_dec* streamdata,
>+                            int16_t* outmodel);
> 
> /* quantize & code LPC Coef */
>-int WebRtcIsacfix_EncodeLpc(int32_t *gain_lo_hiQ17,
>-                            int16_t *LPCCoef_loQ15,
>-                            int16_t *LPCCoef_hiQ15,
>-                            int16_t *model,
>-                            int32_t *sizeQ11,
>-                            Bitstr_enc *streamdata,
>+int WebRtcIsacfix_EncodeLpc(int32_t* gain_lo_hiQ17,
>+                            int16_t* LPCCoef_loQ15,
>+                            int16_t* LPCCoef_hiQ15,
>+                            int16_t* model,
>+                            int32_t* sizeQ11,
>+                            Bitstr_enc* streamdata,
>                             IsacSaveEncoderData* encData,
>-                            transcode_obj *transcodeParam);
>+                            transcode_obj* transcodeParam);
> 
>-int WebRtcIsacfix_EstCodeLpcGain(int32_t *gain_lo_hiQ17,
>-                                 Bitstr_enc *streamdata,
>+int WebRtcIsacfix_EstCodeLpcGain(int32_t* gain_lo_hiQ17,
>+                                 Bitstr_enc* streamdata,
>                                  IsacSaveEncoderData* encData);
> /* decode & dequantize RC */
>-int WebRtcIsacfix_DecodeRcCoef(Bitstr_dec *streamdata,
>-                               int16_t *RCQ15);
>+int WebRtcIsacfix_DecodeRcCoef(Bitstr_dec* streamdata, int16_t* RCQ15);
> 
> /* quantize & code RC */
>-int WebRtcIsacfix_EncodeRcCoef(int16_t *RCQ15,
>-                               Bitstr_enc *streamdata);
>+int WebRtcIsacfix_EncodeRcCoef(int16_t* RCQ15, Bitstr_enc* streamdata);
> 
> /* decode & dequantize squared Gain */
>-int WebRtcIsacfix_DecodeGain2(Bitstr_dec *streamdata,
>-                              int32_t *Gain2);
>+int WebRtcIsacfix_DecodeGain2(Bitstr_dec* streamdata, int32_t* Gain2);
> 
> /* quantize & code squared Gain (input is squared gain) */
>-int WebRtcIsacfix_EncodeGain2(int32_t *gain2,
>-                              Bitstr_enc *streamdata);
>+int WebRtcIsacfix_EncodeGain2(int32_t* gain2, Bitstr_enc* streamdata);
> 
>-int WebRtcIsacfix_EncodePitchGain(int16_t *PitchGains_Q12,
>-                                  Bitstr_enc *streamdata,
>+int WebRtcIsacfix_EncodePitchGain(int16_t* PitchGains_Q12,
>+                                  Bitstr_enc* streamdata,
>                                   IsacSaveEncoderData* encData);
> 
>-int WebRtcIsacfix_EncodePitchLag(int16_t *PitchLagQ7,
>-                                 int16_t *PitchGain_Q12,
>-                                 Bitstr_enc *streamdata,
>+int WebRtcIsacfix_EncodePitchLag(int16_t* PitchLagQ7,
>+                                 int16_t* PitchGain_Q12,
>+                                 Bitstr_enc* streamdata,
>                                  IsacSaveEncoderData* encData);
> 
>-int WebRtcIsacfix_DecodePitchGain(Bitstr_dec *streamdata,
>-                                  int16_t *PitchGain_Q12);
>-
>-int WebRtcIsacfix_DecodePitchLag(Bitstr_dec *streamdata,
>-                                 int16_t *PitchGain_Q12,
>-                                 int16_t *PitchLagQ7);
>-
>-int WebRtcIsacfix_DecodeFrameLen(Bitstr_dec *streamdata,
>-                                 size_t *framelength);
>+int WebRtcIsacfix_DecodePitchGain(Bitstr_dec* streamdata,
>+                                  int16_t* PitchGain_Q12);
> 
>+int WebRtcIsacfix_DecodePitchLag(Bitstr_dec* streamdata,
>+                                 int16_t* PitchGain_Q12,
>+                                 int16_t* PitchLagQ7);
> 
>-int WebRtcIsacfix_EncodeFrameLen(int16_t framelength,
>-                                 Bitstr_enc *streamdata);
>+int WebRtcIsacfix_DecodeFrameLen(Bitstr_dec* streamdata, size_t* framelength);
> 
>-int WebRtcIsacfix_DecodeSendBandwidth(Bitstr_dec *streamdata,
>-                                      int16_t *BWno);
>+int WebRtcIsacfix_EncodeFrameLen(int16_t framelength, Bitstr_enc* streamdata);
> 
>+int WebRtcIsacfix_DecodeSendBandwidth(Bitstr_dec* streamdata, int16_t* BWno);
> 
>-int WebRtcIsacfix_EncodeReceiveBandwidth(int16_t *BWno,
>-                                         Bitstr_enc *streamdata);
>+int WebRtcIsacfix_EncodeReceiveBandwidth(int16_t* BWno, Bitstr_enc* streamdata);
> 
>-void WebRtcIsacfix_TranscodeLpcCoef(int32_t *tmpcoeffs_gQ6,
>-                                    int16_t *index_gQQ);
>+void WebRtcIsacfix_TranscodeLpcCoef(int32_t* tmpcoeffs_gQ6, int16_t* index_gQQ);
> 
> // Pointer functions for LPC transforms.
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding_neon.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding_neon.c
>index 2eccdfaa611..02005678800 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding_neon.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding_neon.c
>@@ -14,12 +14,11 @@
>  * generic platforms.
>  */
> 
>-#include "entropy_coding.h"
>-
> #include <arm_neon.h>
> #include <stddef.h>
> 
>-#include "signal_processing_library.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/entropy_coding.h"
>+#include "common_audio/signal_processing/include/signal_processing_library.h"
> #include "rtc_base/checks.h"
> 
> void WebRtcIsacfix_MatrixProduct1Neon(const int16_t matrix0[],
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/fft.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/fft.c
>index 21566f1718b..a0ed3f83ce9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/fft.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/fft.c
>@@ -16,7 +16,7 @@
>  */
> 
> 
>-#include "fft.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/fft.h"
> 
> static const int16_t kSortTabFft[240] = {
>   0, 60, 120, 180, 20, 80, 140, 200, 40, 100, 160, 220,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/fft.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/fft.h
>index 83062eafd42..4fe9b96be4f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/fft.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/fft.h
>@@ -30,12 +30,10 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_FFT_H_
> #define MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_FFT_H_
> 
>+#include "modules/audio_coding/codecs/isac/fix/source/structs.h"
> 
>-#include "structs.h"
>-
>-
>-int16_t WebRtcIsacfix_FftRadix16Fastest(int16_t RexQx[], int16_t ImxQx[], int16_t iSign);
>-
>-
>+int16_t WebRtcIsacfix_FftRadix16Fastest(int16_t RexQx[],
>+                                        int16_t ImxQx[],
>+                                        int16_t iSign);
> 
> #endif /* MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_FFT_H_ */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbank_internal.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbank_internal.h
>index 1c349697dd8..6b99914b645 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbank_internal.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbank_internal.h
>@@ -11,7 +11,7 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_FILTERBANK_INTERNAL_H_
> #define MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_FILTERBANK_INTERNAL_H_
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include <stdint.h>
> 
> #if defined(__cplusplus) || defined(c_plusplus)
> extern "C" {
>@@ -42,44 +42,41 @@ void WebRtcIsacfix_HighpassFilterFixDec32MIPS(int16_t* io,
> #endif
> 
> typedef void (*AllpassFilter2FixDec16)(
>-    int16_t *data_ch1,           // Input and output in channel 1, in Q0
>-    int16_t *data_ch2,           // Input and output in channel 2, in Q0
>-    const int16_t *factor_ch1,   // Scaling factor for channel 1, in Q15
>-    const int16_t *factor_ch2,   // Scaling factor for channel 2, in Q15
>+    int16_t* data_ch1,           // Input and output in channel 1, in Q0
>+    int16_t* data_ch2,           // Input and output in channel 2, in Q0
>+    const int16_t* factor_ch1,   // Scaling factor for channel 1, in Q15
>+    const int16_t* factor_ch2,   // Scaling factor for channel 2, in Q15
>     const int length,            // Length of the data buffers
>-    int32_t *filter_state_ch1,   // Filter state for channel 1, in Q16
>-    int32_t *filter_state_ch2);  // Filter state for channel 2, in Q16
>+    int32_t* filter_state_ch1,   // Filter state for channel 1, in Q16
>+    int32_t* filter_state_ch2);  // Filter state for channel 2, in Q16
> extern AllpassFilter2FixDec16 WebRtcIsacfix_AllpassFilter2FixDec16;
> 
>-void WebRtcIsacfix_AllpassFilter2FixDec16C(
>-   int16_t *data_ch1,
>-   int16_t *data_ch2,
>-   const int16_t *factor_ch1,
>-   const int16_t *factor_ch2,
>-   const int length,
>-   int32_t *filter_state_ch1,
>-   int32_t *filter_state_ch2);
>+void WebRtcIsacfix_AllpassFilter2FixDec16C(int16_t* data_ch1,
>+                                           int16_t* data_ch2,
>+                                           const int16_t* factor_ch1,
>+                                           const int16_t* factor_ch2,
>+                                           const int length,
>+                                           int32_t* filter_state_ch1,
>+                                           int32_t* filter_state_ch2);
> 
> #if defined(WEBRTC_HAS_NEON)
>-void WebRtcIsacfix_AllpassFilter2FixDec16Neon(
>-   int16_t *data_ch1,
>-   int16_t *data_ch2,
>-   const int16_t *factor_ch1,
>-   const int16_t *factor_ch2,
>-   const int length,
>-   int32_t *filter_state_ch1,
>-   int32_t *filter_state_ch2);
>+void WebRtcIsacfix_AllpassFilter2FixDec16Neon(int16_t* data_ch1,
>+                                              int16_t* data_ch2,
>+                                              const int16_t* factor_ch1,
>+                                              const int16_t* factor_ch2,
>+                                              const int length,
>+                                              int32_t* filter_state_ch1,
>+                                              int32_t* filter_state_ch2);
> #endif
> 
> #if defined(MIPS_DSP_R1_LE)
>-void WebRtcIsacfix_AllpassFilter2FixDec16MIPS(
>-   int16_t *data_ch1,
>-   int16_t *data_ch2,
>-   const int16_t *factor_ch1,
>-   const int16_t *factor_ch2,
>-   const int length,
>-   int32_t *filter_state_ch1,
>-   int32_t *filter_state_ch2);
>+void WebRtcIsacfix_AllpassFilter2FixDec16MIPS(int16_t* data_ch1,
>+                                              int16_t* data_ch2,
>+                                              const int16_t* factor_ch1,
>+                                              const int16_t* factor_ch2,
>+                                              const int length,
>+                                              int32_t* filter_state_ch1,
>+                                              int32_t* filter_state_ch2);
> #endif
> 
> #if defined(__cplusplus) || defined(c_plusplus)
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbank_tables.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbank_tables.c
>index 732611b643d..f2dec79c2d0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbank_tables.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbank_tables.c
>@@ -16,7 +16,7 @@
>  *
>  */
> 
>-#include "filterbank_tables.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/filterbank_tables.h"
> 
> /* HPstcoeff_in_Q14 = {a1, a2, b1 - b0 * a1, b2 - b0 * a2};
>  * In float, they are: {-1.94895953203325f, 0.94984516000000f,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbank_tables.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbank_tables.h
>index 55e1f44a015..01e5a7ba855 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbank_tables.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbank_tables.h
>@@ -19,7 +19,7 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_FILTERBANK_TABLES_H_
> #define MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_FILTERBANK_TABLES_H_
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include <stdint.h>
> 
> #if defined(__cplusplus) || defined(c_plusplus)
> extern "C" {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks.c
>index b4583e95d15..6aba8b6a54a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks.c
>@@ -18,11 +18,11 @@
>  *
>  */
> 
>-#include "filterbank_internal.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/filterbank_internal.h"
> 
>-#include "codec.h"
>-#include "filterbank_tables.h"
>-#include "settings.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/codec.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/filterbank_tables.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/settings.h"
> #include "rtc_base/checks.h"
> 
> // Declare a function pointer.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks_neon.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks_neon.c
>index fd29ccb3ff0..a31cea60017 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks_neon.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks_neon.c
>@@ -15,6 +15,7 @@
> 
> #include <arm_neon.h>
> 
>+#include "modules/audio_coding/codecs/isac/fix/source/filterbank_internal.h"
> #include "rtc_base/checks.h"
> 
> void WebRtcIsacfix_AllpassFilter2FixDec16Neon(
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks_unittest.cc
>index d17f4a5f9de..b7456a70563 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/filterbanks_unittest.cc
>@@ -15,14 +15,13 @@
> #include "rtc_base/sanitizer.h"
> #include "system_wrappers/include/cpu_features_wrapper.h"
> #include "test/gtest.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> class FilterBanksTest : public testing::Test {
>  protected:
>   // Pass a function pointer to the Tester function.
>   void RTC_NO_SANITIZE("signed-integer-overflow")  // bugs.webrtc.org/5513
>-  CalculateResidualEnergyTester(AllpassFilter2FixDec16
>-                                AllpassFilter2FixDec16Function) {
>+      CalculateResidualEnergyTester(
>+          AllpassFilter2FixDec16 AllpassFilter2FixDec16Function) {
>     const int kSamples = QLOOKAHEAD;
>     const int kState = 2;
>     int16_t data_ch1[kSamples] = {0};
>@@ -31,12 +30,14 @@ class FilterBanksTest : public testing::Test {
>     int32_t state_ch2[kState] = {0};
>     const int32_t out_state_ch1[kState] = {-809122714, 1645972152};
>     const int32_t out_state_ch2[kState] = {428019288, 1057309936};
>-    const int32_t out_data_ch1[kSamples] = {0, 0, 347, 10618, 16718, -7089,
>-        32767, 16913, 27042, 8377, -22973, -28372, -27603, -14804, 398, -25332,
>-        -11200, 18044, 25223, -6839, 1116, -23984, 32717, 7364};
>-    const int32_t out_data_ch2[kSamples] = {0, 0, 3010, 22351, 21106, 16969,
>-        -2095, -664, 3513, -30980, 32767, -23839, 13335, 20289, -6831, 339,
>-        -17207, 32767, 4959, 6177, 32767, 16599, -4747, 20504};
>+    const int32_t out_data_ch1[kSamples] = {
>+        0,      0,     347,    10618,  16718,  -7089,  32767, 16913,
>+        27042,  8377,  -22973, -28372, -27603, -14804, 398,   -25332,
>+        -11200, 18044, 25223,  -6839,  1116,   -23984, 32717, 7364};
>+    const int32_t out_data_ch2[kSamples] = {
>+        0,      0,      3010,  22351,  21106, 16969, -2095, -664,
>+        3513,   -30980, 32767, -23839, 13335, 20289, -6831, 339,
>+        -17207, 32767,  4959,  6177,   32767, 16599, -4747, 20504};
>     int sign = 1;
> 
>     for (int i = 0; i < kSamples; i++) {
>@@ -46,13 +47,9 @@ class FilterBanksTest : public testing::Test {
>       // UBSan: -1 * -2147483648 cannot be represented in type 'int'
>     };
> 
>-    AllpassFilter2FixDec16Function(data_ch1,
>-                                   data_ch2,
>-                                   WebRtcIsacfix_kUpperApFactorsQ15,
>-                                   WebRtcIsacfix_kLowerApFactorsQ15,
>-                                   kSamples,
>-                                   state_ch1,
>-                                   state_ch2);
>+    AllpassFilter2FixDec16Function(
>+        data_ch1, data_ch2, WebRtcIsacfix_kUpperApFactorsQ15,
>+        WebRtcIsacfix_kLowerApFactorsQ15, kSamples, state_ch1, state_ch2);
> 
>     for (int i = 0; i < kSamples; i++) {
>       EXPECT_EQ(out_data_ch1[i], data_ch1[i]);
>@@ -77,13 +74,13 @@ TEST_F(FilterBanksTest, HighpassFilterFixDec32Test) {
>   int16_t in[kSamples];
>   int32_t state[2] = {12345, 987654};
> #ifdef WEBRTC_ARCH_ARM_V7
>-  int32_t out[kSamples] = {-1040, -1035, -22875, -1397, -27604, 20018, 7917,
>-    -1279, -8552, -14494, -7558, -23537, -27258, -30554, -32768, -3432, -32768,
>-    25215, -27536, 22436};
>+  int32_t out[kSamples] = {-1040,  -1035, -22875, -1397, -27604, 20018,  7917,
>+                           -1279,  -8552, -14494, -7558, -23537, -27258, -30554,
>+                           -32768, -3432, -32768, 25215, -27536, 22436};
> #else
>-  int32_t out[kSamples] = {-1040, -1035, -22875, -1397, -27604, 20017, 7915,
>-    -1280, -8554, -14496, -7561, -23541, -27263, -30560, -32768, -3441, -32768,
>-    25203, -27550, 22419};
>+  int32_t out[kSamples] = {-1040,  -1035, -22875, -1397, -27604, 20017,  7915,
>+                           -1280,  -8554, -14496, -7561, -23541, -27263, -30560,
>+                           -32768, -3441, -32768, 25203, -27550, 22419};
> #endif
>   HighpassFilterFixDec32 WebRtcIsacfix_HighpassFilterFixDec32;
> #if defined(MIPS_DSP_R1_LE)
>@@ -98,7 +95,7 @@ TEST_F(FilterBanksTest, HighpassFilterFixDec32Test) {
>   }
> 
>   WebRtcIsacfix_HighpassFilterFixDec32(in, kSamples,
>-      WebRtcIsacfix_kHPStCoeffOut1Q30, state);
>+                                       WebRtcIsacfix_kHPStCoeffOut1Q30, state);
> 
>   for (int i = 0; i < kSamples; i++) {
>     EXPECT_EQ(out[i], in[i]);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/filters_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/filters_unittest.cc
>index fa529863c06..471fa57474a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/filters_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/filters_unittest.cc
>@@ -11,7 +11,6 @@
> #include "modules/audio_coding/codecs/isac/fix/source/codec.h"
> #include "system_wrappers/include/cpu_features_wrapper.h"
> #include "test/gtest.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> class FiltersTest : public testing::Test {
>  protected:
>@@ -23,34 +22,37 @@ class FiltersTest : public testing::Test {
>     int32_t r_buffer[kOrder + 2] = {0};
> 
>     // Test an overflow case.
>-    const int16_t x_buffer_0[kBuffer] = {0, 0, 3010, 22351, 21106, 16969, -2095,
>-        -664, 3513, -30980, 32767, -23839, 13335, 20289, -6831, 339, -17207,
>-        32767, 4959, 6177, 32767, 16599, -4747, 20504, 3513, -30980, 32767,
>-        -23839, 13335, 20289, 0, -16969, -2095, -664, 3513, 31981, 32767,
>-        -13839, 23336, 30281};
>-    const int32_t r_expected_0[kOrder + 2] = {1872498461, -224288754, 203789985,
>-        483400487, -208272635, 2436500, 137785322, 266600814, -208486262,
>-        329510080, 137949184, -161738972, -26894267, 237630192};
>+    const int16_t x_buffer_0[kBuffer] = {
>+        0,      0,      3010,  22351,  21106, 16969,  -2095, -664,
>+        3513,   -30980, 32767, -23839, 13335, 20289,  -6831, 339,
>+        -17207, 32767,  4959,  6177,   32767, 16599,  -4747, 20504,
>+        3513,   -30980, 32767, -23839, 13335, 20289,  0,     -16969,
>+        -2095,  -664,   3513,  31981,  32767, -13839, 23336, 30281};
>+    const int32_t r_expected_0[kOrder + 2] = {
>+        1872498461, -224288754, 203789985, 483400487,  -208272635,
>+        2436500,    137785322,  266600814, -208486262, 329510080,
>+        137949184,  -161738972, -26894267, 237630192};
> 
>-    WebRtcIsacfix_AutocorrFixFunction(r_buffer, x_buffer_0,
>-                                      kBuffer, kOrder + 1, &scale);
>+    WebRtcIsacfix_AutocorrFixFunction(r_buffer, x_buffer_0, kBuffer, kOrder + 1,
>+                                      &scale);
>     for (int i = 0; i < kOrder + 2; i++) {
>       EXPECT_EQ(r_expected_0[i], r_buffer[i]);
>     }
>     EXPECT_EQ(3, scale);
> 
>     // Test a no-overflow case.
>-    const int16_t x_buffer_1[kBuffer] = {0, 0, 300, 21, 206, 169, -295,
>-        -664, 3513, -300, 327, -29, 15, 289, -6831, 339, -107,
>-        37, 59, 6177, 327, 169, -4747, 204, 313, -980, 767,
>-        -9, 135, 289, 0, -6969, -2095, -664, 0, 1, 7,
>-        -39, 236, 281};
>-    const int32_t r_expected_1[kOrder + 2] = {176253864, 8126617, 1983287,
>-        -26196788, -3487363, -42839676, -24644043, 3469813, 30559879, 31905045,
>-        5101567, 29328896, -55787438, -13163978};
>+    const int16_t x_buffer_1[kBuffer] = {
>+        0,   0,     300,   21,   206,   169,  -295, -664, 3513, -300,
>+        327, -29,   15,    289,  -6831, 339,  -107, 37,   59,   6177,
>+        327, 169,   -4747, 204,  313,   -980, 767,  -9,   135,  289,
>+        0,   -6969, -2095, -664, 0,     1,    7,    -39,  236,  281};
>+    const int32_t r_expected_1[kOrder + 2] = {
>+        176253864, 8126617,   1983287,   -26196788, -3487363,
>+        -42839676, -24644043, 3469813,   30559879,  31905045,
>+        5101567,   29328896,  -55787438, -13163978};
> 
>-    WebRtcIsacfix_AutocorrFixFunction(r_buffer, x_buffer_1,
>-                                      kBuffer, kOrder + 1, &scale);
>+    WebRtcIsacfix_AutocorrFixFunction(r_buffer, x_buffer_1, kBuffer, kOrder + 1,
>+                                      &scale);
>     for (int i = 0; i < kOrder + 2; i++) {
>       EXPECT_EQ(r_expected_1[i], r_buffer[i]);
>     }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/initialize.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/initialize.c
>index 4481f981220..1b82958883f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/initialize.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/initialize.c
>@@ -15,9 +15,9 @@
>  *
>  */
> 
>-#include "codec.h"
>-#include "structs.h"
>-#include "pitch_estimator.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/codec.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/structs.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h"
> 
> 
> void WebRtcIsacfix_InitMaskingEnc(MaskFiltstr_enc *maskdata) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice.c
>index 81369991d5d..7bbf4e054a9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice.c
>@@ -15,8 +15,8 @@
>  *
>  */
> 
>-#include "codec.h"
>-#include "settings.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/codec.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/settings.h"
> #include "rtc_base/sanitizer.h"
> 
> #define LATTICE_MUL_32_32_RSFT16(a32a, a32b, b32)                  \
>@@ -279,8 +279,7 @@ void WebRtcIsacfix_NormLatticeFilterAr(size_t orderCoef,
> 
>     for (i=0;i<HALF_SUBFRAMELEN;i++)
>     {
>-
>-      tmp32 = lat_inQ25[i + temp1] * (1 << 1);  // Q25->Q26
>+      tmp32 = OverflowingLShiftS32(lat_inQ25[i + temp1], 1);  // Q25->Q26
>       tmp32 = WEBRTC_SPL_MUL_16_32_RSFT16(inv_gain16, tmp32); //lat_in[]*inv_gain in (Q(18-sh)*Q26)>>16 = Q(28-sh)
>       tmp32 = WEBRTC_SPL_SHIFT_W32(tmp32, -(28-sh)); // lat_in[]*inv_gain in Q0
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice_armv7.S b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice_armv7.S
>index e2d43185416..4c63227d7dd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice_armv7.S
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice_armv7.S
>@@ -25,8 +25,8 @@
> @ r12: constant #16384
> @ r6, r7, r8, r10, r11: scratch
> 
>-#include "system_wrappers/include/asm_defines.h"
>-#include "settings.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/settings.h"
>+#include "rtc_base/system/asm_defines.h"
> 
> GLOBAL_FUNCTION WebRtcIsacfix_FilterArLoop
> .align  2
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice_c.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice_c.c
>index d2356af5144..43406612e81 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice_c.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice_c.c
>@@ -14,9 +14,8 @@
>  *
>  */
> 
>-#include "settings.h"
>-#include "signal_processing_library.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include "common_audio/signal_processing/include/signal_processing_library.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/settings.h"
> 
> /* Filter ar_g_Q0[] and ar_f_Q0[] through an AR filter with coefficients
>  * cth_Q15[] and sth_Q15[].
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice_mips.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice_mips.c
>index 613c07c9bd7..31897266291 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice_mips.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/lattice_mips.c
>@@ -9,9 +9,9 @@
>  */
> 
> #include <stddef.h>
>+#include <stdint.h>
> 
> #include "modules/audio_coding/codecs/isac/fix/source/settings.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> // Filter ar_g_Q0[] and ar_f_Q0[] through an AR filter with coefficients
> // cth_Q15[] and sth_Q15[].
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.c
>index 97e4ce03a43..b538085d14d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.c
>@@ -18,9 +18,10 @@
> #include "lpc_masking_model.h"
> 
> #include <limits.h>  /* For LLONG_MAX and LLONG_MIN. */
>-#include "codec.h"
>-#include "entropy_coding.h"
>-#include "settings.h"
>+
>+#include "modules/audio_coding/codecs/isac/fix/source/codec.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/entropy_coding.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/settings.h"
> 
> /* The conversion is implemented by the step-down algorithm */
> void WebRtcSpl_AToK_JSK(
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.h
>index 97d291aca27..40a99e8a77a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.h
>@@ -22,21 +22,21 @@
> extern "C" {
> #endif
> 
>-#include "structs.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/structs.h"
> 
>-void WebRtcIsacfix_GetVars(const int16_t *input,
>-                           const int16_t *pitchGains_Q12,
>-                           uint32_t *oldEnergy,
>-                           int16_t *varscale);
>+void WebRtcIsacfix_GetVars(const int16_t* input,
>+                           const int16_t* pitchGains_Q12,
>+                           uint32_t* oldEnergy,
>+                           int16_t* varscale);
> 
>-void WebRtcIsacfix_GetLpcCoef(int16_t *inLoQ0,
>-                              int16_t *inHiQ0,
>-                              MaskFiltstr_enc *maskdata,
>+void WebRtcIsacfix_GetLpcCoef(int16_t* inLoQ0,
>+                              int16_t* inHiQ0,
>+                              MaskFiltstr_enc* maskdata,
>                               int16_t snrQ10,
>-                              const int16_t *pitchGains_Q12,
>-                              int32_t *gain_lo_hiQ17,
>-                              int16_t *lo_coeffQ15,
>-                              int16_t *hi_coeffQ15);
>+                              const int16_t* pitchGains_Q12,
>+                              int32_t* gain_lo_hiQ17,
>+                              int16_t* lo_coeffQ15,
>+                              int16_t* hi_coeffQ15);
> 
> typedef int32_t (*CalculateResidualEnergy)(int lpc_order,
>                                            int32_t q_val_corr,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model_unittest.cc
>index 1604cc41445..fab0a043cce 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model_unittest.cc
>@@ -11,26 +11,25 @@
> #include "modules/audio_coding/codecs/isac/fix/source/lpc_masking_model.h"
> #include "system_wrappers/include/cpu_features_wrapper.h"
> #include "test/gtest.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> class LpcMaskingModelTest : public testing::Test {
>  protected:
>   // Pass a function pointer to the Tester function.
>-  void CalculateResidualEnergyTester(CalculateResidualEnergy
>-                                     CalculateResidualEnergyFunction) {
>+  void CalculateResidualEnergyTester(
>+      CalculateResidualEnergy CalculateResidualEnergyFunction) {
>     const int kIntOrder = 10;
>     const int32_t kInt32QDomain = 5;
>     const int kIntShift = 11;
>-    int16_t a[kIntOrder + 1] = {32760, 122, 7, 0, -32760, -3958,
>-        -48, 18745, 498, 9, 23456};
>-    int32_t corr[kIntOrder + 1] = {11443647, -27495, 0,
>-        98745, -11443600, 1, 1, 498, 9, 888, 23456};
>+    int16_t a[kIntOrder + 1] = {32760, 122,   7,   0, -32760, -3958,
>+                                -48,   18745, 498, 9, 23456};
>+    int32_t corr[kIntOrder + 1] = {11443647, -27495, 0, 98745, -11443600, 1,
>+                                   1,        498,    9, 888,   23456};
>     int q_shift_residual = 0;
>     int32_t residual_energy = 0;
> 
>     // Test the code path where (residual_energy >= 0x10000).
>-    residual_energy = CalculateResidualEnergyFunction(kIntOrder,
>-        kInt32QDomain, kIntShift, a, corr, &q_shift_residual);
>+    residual_energy = CalculateResidualEnergyFunction(
>+        kIntOrder, kInt32QDomain, kIntShift, a, corr, &q_shift_residual);
>     EXPECT_EQ(1789023310, residual_energy);
>     EXPECT_EQ(2, q_shift_residual);
> 
>@@ -40,8 +39,8 @@ class LpcMaskingModelTest : public testing::Test {
>       a[i] = 24575 >> i;
>       corr[i] = i;
>     }
>-    residual_energy = CalculateResidualEnergyFunction(kIntOrder,
>-        kInt32QDomain, kIntShift, a, corr, &q_shift_residual);
>+    residual_energy = CalculateResidualEnergyFunction(
>+        kIntOrder, kInt32QDomain, kIntShift, a, corr, &q_shift_residual);
>     EXPECT_EQ(1595279092, residual_energy);
>     EXPECT_EQ(26, q_shift_residual);
> 
>@@ -49,8 +48,8 @@ class LpcMaskingModelTest : public testing::Test {
>     for (int i = 0; i < kIntOrder + 1; i++) {
>       a[i] = 2457 >> i;
>     }
>-    residual_energy = CalculateResidualEnergyFunction(kIntOrder,
>-        kInt32QDomain, kIntShift, a, corr, &q_shift_residual);
>+    residual_energy = CalculateResidualEnergyFunction(
>+        kIntOrder, kInt32QDomain, kIntShift, a, corr, &q_shift_residual);
>     EXPECT_EQ(2029266944, residual_energy);
>     EXPECT_EQ(33, q_shift_residual);
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_tables.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_tables.c
>index bc0f129711f..df723fdd0b2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_tables.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_tables.c
>@@ -16,8 +16,8 @@
>  */
> 
> 
>-#include "settings.h"
>-#include "lpc_tables.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/settings.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/lpc_tables.h"
> 
> /* indices of KLT coefficients used */
> const uint16_t WebRtcIsacfix_kSelIndGain[12] = {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_tables.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_tables.h
>index 05c53dd94c5..2d68fa6c6e5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_tables.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_tables.h
>@@ -18,7 +18,7 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_LPC_TABLES_H_
> #define MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_LPC_TABLES_H_
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include <stdint.h>
> 
> /* indices of KLT coefficients used */
> extern const uint16_t WebRtcIsacfix_kSelIndGain[12];
>@@ -26,10 +26,10 @@ extern const uint16_t WebRtcIsacfix_kSelIndGain[12];
> extern const uint16_t WebRtcIsacfix_kSelIndShape[108];
> 
> /* cdf array for model indicator */
>-extern const uint16_t WebRtcIsacfix_kModelCdf[KLT_NUM_MODELS+1];
>+extern const uint16_t WebRtcIsacfix_kModelCdf[KLT_NUM_MODELS + 1];
> 
> /* pointer to cdf array for model indicator */
>-extern const uint16_t *WebRtcIsacfix_kModelCdfPtr[1];
>+extern const uint16_t* WebRtcIsacfix_kModelCdfPtr[1];
> 
> /* initial cdf index for decoder of model indicator */
> extern const uint16_t WebRtcIsacfix_kModelInitIndex[1];
>@@ -70,9 +70,9 @@ extern const uint16_t WebRtcIsacfix_kCdfGain[1212];
> extern const uint16_t WebRtcIsacfix_kCdfShape[2059];
> 
> /* pointers to cdf tables for quantizer indices */
>-extern const uint16_t *WebRtcIsacfix_kCdfGainPtr[KLT_NUM_MODELS][12];
>+extern const uint16_t* WebRtcIsacfix_kCdfGainPtr[KLT_NUM_MODELS][12];
> 
>-extern const uint16_t *WebRtcIsacfix_kCdfShapePtr[KLT_NUM_MODELS][108];
>+extern const uint16_t* WebRtcIsacfix_kCdfShapePtr[KLT_NUM_MODELS][108];
> 
> /* code length for all coefficients using different models */
> extern const int16_t WebRtcIsacfix_kCodeLenGainQ11[392];
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h
>index 0aa727dd412..4303c82711d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_estimator.h
>@@ -18,23 +18,24 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_ESTIMATOR_H_
> #define MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_ESTIMATOR_H_
> 
>-#include "structs.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/structs.h"
> 
>-void WebRtcIsacfix_PitchAnalysis(const int16_t *in,               /* PITCH_FRAME_LEN samples */
>-                                 int16_t *outQ0,                  /* PITCH_FRAME_LEN+QLOOKAHEAD samples */
>-                                 PitchAnalysisStruct *State,
>-                                 int16_t *lagsQ7,
>-                                 int16_t *PitchGains_Q12);
>+void WebRtcIsacfix_PitchAnalysis(
>+    const int16_t* in, /* PITCH_FRAME_LEN samples */
>+    int16_t* outQ0,    /* PITCH_FRAME_LEN+QLOOKAHEAD samples */
>+    PitchAnalysisStruct* State,
>+    int16_t* lagsQ7,
>+    int16_t* PitchGains_Q12);
> 
>-void WebRtcIsacfix_InitialPitch(const int16_t *in,
>-                                PitchAnalysisStruct *State,
>-                                int16_t *qlags);
>+void WebRtcIsacfix_InitialPitch(const int16_t* in,
>+                                PitchAnalysisStruct* State,
>+                                int16_t* qlags);
> 
>-void WebRtcIsacfix_PitchFilter(int16_t *indatFix,
>-                               int16_t *outdatQQ,
>-                               PitchFiltstr *pfp,
>-                               int16_t *lagsQ7,
>-                               int16_t *gainsQ12,
>+void WebRtcIsacfix_PitchFilter(int16_t* indatFix,
>+                               int16_t* outdatQQ,
>+                               PitchFiltstr* pfp,
>+                               int16_t* lagsQ7,
>+                               int16_t* gainsQ12,
>                                int16_t type);
> 
> void WebRtcIsacfix_PitchFilterCore(int loopNumber,
>@@ -48,17 +49,18 @@ void WebRtcIsacfix_PitchFilterCore(int loopNumber,
>                                    int16_t* outputBuf,
>                                    int* index2);
> 
>-void WebRtcIsacfix_PitchFilterGains(const int16_t *indatQ0,
>-                                    PitchFiltstr *pfp,
>-                                    int16_t *lagsQ7,
>-                                    int16_t *gainsQ12);
>+void WebRtcIsacfix_PitchFilterGains(const int16_t* indatQ0,
>+                                    PitchFiltstr* pfp,
>+                                    int16_t* lagsQ7,
>+                                    int16_t* gainsQ12);
> 
>-void WebRtcIsacfix_DecimateAllpass32(const int16_t *in,
>-                                     int32_t *state_in,        /* array of size: 2*ALLPASSSECTIONS+1 */
>-                                     int16_t N,                   /* number of input samples */
>-                                     int16_t *out);             /* array of size N/2 */
>+void WebRtcIsacfix_DecimateAllpass32(
>+    const int16_t* in,
>+    int32_t* state_in, /* array of size: 2*ALLPASSSECTIONS+1 */
>+    int16_t N,         /* number of input samples */
>+    int16_t* out);     /* array of size N/2 */
> 
>-int32_t WebRtcIsacfix_Log2Q8( uint32_t x );
>+int32_t WebRtcIsacfix_Log2Q8(uint32_t x);
> 
> void WebRtcIsacfix_PCorr2Q32(const int16_t* in, int32_t* logcorQ8);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_filter_armv6.S b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_filter_armv6.S
>index 44fcac86f87..065946856f8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_filter_armv6.S
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_filter_armv6.S
>@@ -13,8 +13,8 @@
> @
> @ Output is bit-exact with the reference C code in pitch_filter.c.
> 
>-#include "system_wrappers/include/asm_defines.h"
>-#include "settings.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/settings.h"
>+#include "rtc_base/system/asm_defines.h"
> 
> GLOBAL_FUNCTION WebRtcIsacfix_PitchFilterCore
> .align  2
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_gain_tables.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_gain_tables.c
>index bfd83b72581..bfbab1950d4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_gain_tables.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_gain_tables.c
>@@ -15,7 +15,7 @@
>  *
>  */
> 
>-#include "pitch_gain_tables.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/pitch_gain_tables.h"
> 
> 
> /********************* Pitch Filter Gain Coefficient Tables ************************/
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_gain_tables.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_gain_tables.h
>index fe4d2883d43..59e1738bce1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_gain_tables.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_gain_tables.h
>@@ -11,16 +11,18 @@
> /*
>  * pitch_gain_tables.h
>  *
>- * This file contains tables for the pitch filter side-info in the entropy coder.
>+ * This file contains tables for the pitch filter side-info in the entropy
>+ * coder.
>  *
>  */
> 
> #ifndef MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_GAIN_TABLES_H_
> #define MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_GAIN_TABLES_H_
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include <stdint.h>
> 
>-/********************* Pitch Filter Gain Coefficient Tables ************************/
>+/********************* Pitch Filter Gain Coefficient Tables
>+ * ************************/
> /* cdf for quantized pitch filter gains */
> extern const uint16_t WebRtcIsacfix_kPitchGainCdf[255];
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_lag_tables.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_lag_tables.c
>index 4566b6eb780..894716e7398 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_lag_tables.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_lag_tables.c
>@@ -15,8 +15,8 @@
>  *
>  */
> 
>-#include "settings.h"
>-#include "pitch_lag_tables.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/settings.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/pitch_lag_tables.h"
> 
> 
> /********************* Pitch Filter Gain Coefficient Tables ************************/
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_lag_tables.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_lag_tables.h
>index a8c0c3ae7e6..228da26731f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_lag_tables.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/pitch_lag_tables.h
>@@ -11,16 +11,18 @@
> /*
>  * pitch_lag_tables.h
>  *
>- * This file contains tables for the pitch filter side-info in the entropy coder.
>+ * This file contains tables for the pitch filter side-info in the entropy
>+ * coder.
>  *
>  */
> 
> #ifndef MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_LAG_TABLES_H_
> #define MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_LAG_TABLES_H_
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include <stdint.h>
> 
>-/********************* Pitch Filter Lag Coefficient Tables ************************/
>+/********************* Pitch Filter Lag Coefficient Tables
>+ * ************************/
> 
> /* tables for use with small pitch gain */
> 
>@@ -30,7 +32,7 @@ extern const uint16_t WebRtcIsacfix_kPitchLagCdf2Lo[20];
> extern const uint16_t WebRtcIsacfix_kPitchLagCdf3Lo[2];
> extern const uint16_t WebRtcIsacfix_kPitchLagCdf4Lo[10];
> 
>-extern const uint16_t *WebRtcIsacfix_kPitchLagPtrLo[4];
>+extern const uint16_t* WebRtcIsacfix_kPitchLagPtrLo[4];
> 
> /* size of first cdf table */
> extern const uint16_t WebRtcIsacfix_kPitchLagSizeLo[1];
>@@ -46,8 +48,6 @@ extern const uint16_t WebRtcIsacfix_kInitIndLo[3];
> extern const int16_t WebRtcIsacfix_kMeanLag2Lo[19];
> extern const int16_t WebRtcIsacfix_kMeanLag4Lo[9];
> 
>-
>-
> /* tables for use with medium pitch gain */
> 
> /* cdfs for quantized pitch lags */
>@@ -56,7 +56,7 @@ extern const uint16_t WebRtcIsacfix_kPitchLagCdf2Mid[36];
> extern const uint16_t WebRtcIsacfix_kPitchLagCdf3Mid[2];
> extern const uint16_t WebRtcIsacfix_kPitchLagCdf4Mid[20];
> 
>-extern const uint16_t *WebRtcIsacfix_kPitchLagPtrMid[4];
>+extern const uint16_t* WebRtcIsacfix_kPitchLagPtrMid[4];
> 
> /* size of first cdf table */
> extern const uint16_t WebRtcIsacfix_kPitchLagSizeMid[1];
>@@ -72,7 +72,6 @@ extern const uint16_t WebRtcIsacfix_kInitIndMid[3];
> extern const int16_t WebRtcIsacfix_kMeanLag2Mid[35];
> extern const int16_t WebRtcIsacfix_kMeanLag4Mid[19];
> 
>-
> /* tables for use with large pitch gain */
> 
> /* cdfs for quantized pitch lags */
>@@ -81,7 +80,7 @@ extern const uint16_t WebRtcIsacfix_kPitchLagCdf2Hi[68];
> extern const uint16_t WebRtcIsacfix_kPitchLagCdf3Hi[2];
> extern const uint16_t WebRtcIsacfix_kPitchLagCdf4Hi[35];
> 
>-extern const uint16_t *WebRtcIsacfix_kPitchLagPtrHi[4];
>+extern const uint16_t* WebRtcIsacfix_kPitchLagPtrHi[4];
> 
> /* size of first cdf table */
> extern const uint16_t WebRtcIsacfix_kPitchLagSizeHi[1];
>@@ -97,5 +96,4 @@ extern const uint16_t WebRtcIsacfix_kInitIndHi[3];
> extern const int16_t WebRtcIsacfix_kMeanLag2Hi[67];
> extern const int16_t WebRtcIsacfix_kMeanLag4Hi[34];
> 
>-
> #endif /* MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_LAG_TABLES_H_ */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/settings.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/settings.h
>index 34c0efefd0a..03a2d05457d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/settings.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/settings.h
>@@ -18,84 +18,82 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_SETTINGS_H_
> #define MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_SETTINGS_H_
> 
>-
> /* sampling frequency (Hz) */
>-#define FS                                      16000
>+#define FS 16000
> /* 1.5 times Sampling frequency */
>-#define FS_1_HALF        (uint32_t) 24000
>+#define FS_1_HALF (uint32_t)24000
> /* Three times Sampling frequency */
>-#define FS3          (uint32_t) 48000
>+#define FS3 (uint32_t)48000
> /* Eight times Sampling frequency */
>-#define FS8          (uint32_t) 128000
>+#define FS8 (uint32_t)128000
> 
> /* number of samples per frame (either 480 (30ms) or 960 (60ms)) */
>-#define INITIAL_FRAMESAMPLES     960
>+#define INITIAL_FRAMESAMPLES 960
> 
> /* miliseconds */
>-#define FRAMESIZE                               30
>+#define FRAMESIZE 30
> /* number of samples per frame processed in the encoder (30ms) */
>-#define FRAMESAMPLES                            480     /* ((FRAMESIZE*FS)/1000) */
>-#define FRAMESAMPLES_HALF       240
>+#define FRAMESAMPLES 480 /* ((FRAMESIZE*FS)/1000) */
>+#define FRAMESAMPLES_HALF 240
> /* max number of samples per frame (= 60 ms frame) */
>-#define MAX_FRAMESAMPLES      960
>+#define MAX_FRAMESAMPLES 960
> /* number of samples per 10ms frame */
>-#define FRAMESAMPLES_10ms                       160      /* ((10*FS)/1000) */
>+#define FRAMESAMPLES_10ms 160 /* ((10*FS)/1000) */
> /* Number of samples per 1 ms */
>-#define SAMPLES_PER_MSEC      16
>+#define SAMPLES_PER_MSEC 16
> /* number of subframes */
>-#define SUBFRAMES                               6
>+#define SUBFRAMES 6
> /* length of a subframe */
>-#define UPDATE                                  80
>+#define UPDATE 80
> /* length of half a subframe (low/high band) */
>-#define HALF_SUBFRAMELEN                        40    /* (UPDATE/2) */
>-/* samples of look ahead (in a half-band, so actually half the samples of look ahead @ FS) */
>-#define QLOOKAHEAD                              24    /* 3 ms */
>+#define HALF_SUBFRAMELEN 40 /* (UPDATE/2) */
>+/* samples of look ahead (in a half-band, so actually half the samples of look
>+ * ahead @ FS) */
>+#define QLOOKAHEAD 24 /* 3 ms */
> 
> /* order of AR model in spectral entropy coder */
>-#define AR_ORDER                                6
>-#define MAX_ORDER                               13
>-#define LEVINSON_MAX_ORDER                  12
>+#define AR_ORDER 6
>+#define MAX_ORDER 13
>+#define LEVINSON_MAX_ORDER 12
> 
> /* window length (masking analysis) */
>-#define WINLEN                                  256
>+#define WINLEN 256
> /* order of low-band pole filter used to approximate masking curve */
>-#define ORDERLO                                 12
>+#define ORDERLO 12
> /* order of hi-band pole filter used to approximate masking curve */
>-#define ORDERHI                                 6
>-
>-#define KLT_NUM_AVG_GAIN                        0
>-#define KLT_NUM_AVG_SHAPE                       0
>-#define KLT_NUM_MODELS                          3
>-#define LPC_SHAPE_ORDER                         18    /* (ORDERLO + ORDERHI) */
>-
>-#define KLT_ORDER_GAIN                          12    /* (2 * SUBFRAMES) */
>-#define KLT_ORDER_SHAPE                         108   /*  (LPC_SHAPE_ORDER * SUBFRAMES) */
>+#define ORDERHI 6
> 
>+#define KLT_NUM_AVG_GAIN 0
>+#define KLT_NUM_AVG_SHAPE 0
>+#define KLT_NUM_MODELS 3
>+#define LPC_SHAPE_ORDER 18 /* (ORDERLO + ORDERHI) */
> 
>+#define KLT_ORDER_GAIN 12   /* (2 * SUBFRAMES) */
>+#define KLT_ORDER_SHAPE 108 /*  (LPC_SHAPE_ORDER * SUBFRAMES) */
> 
> /* order for post_filter_bank */
>-#define POSTQORDER                              3
>+#define POSTQORDER 3
> /* order for pre-filterbank */
>-#define QORDER                                  3
>+#define QORDER 3
> /* for decimator */
>-#define ALLPASSSECTIONS                         2
>+#define ALLPASSSECTIONS 2
> /* The number of composite all-pass filter factors */
>-#define NUMBEROFCOMPOSITEAPSECTIONS             4
>+#define NUMBEROFCOMPOSITEAPSECTIONS 4
> 
> /* The number of all-pass filter factors in an upper or lower channel*/
>-#define NUMBEROFCHANNELAPSECTIONS               2
>-
>-
>-
>-#define DPMIN_Q10                            -10240   /* -10.00 in Q10 */
>-#define DPMAX_Q10                             10240   /* 10.00 in Q10 */
>-#define MINBITS_Q10                           10240   /* 10.0 in Q10 */
>+#define NUMBEROFCHANNELAPSECTIONS 2
> 
>+#define DPMIN_Q10 -10240  /* -10.00 in Q10 */
>+#define DPMAX_Q10 10240   /* 10.00 in Q10 */
>+#define MINBITS_Q10 10240 /* 10.0 in Q10 */
> 
> /* array size for byte stream in number of Word16. */
>-#define STREAM_MAXW16       300 /* The old maximum size still needed for the decoding */
>-#define STREAM_MAXW16_30MS  100 /* 100 Word16 = 200 bytes = 53.4 kbit/s @ 30 ms.framelength */
>-#define STREAM_MAXW16_60MS  200 /* 200 Word16 = 400 bytes = 53.4 kbit/s @ 60 ms.framelength */
>+#define STREAM_MAXW16 \
>+  300 /* The old maximum size still needed for the decoding */
>+#define STREAM_MAXW16_30MS \
>+  100 /* 100 Word16 = 200 bytes = 53.4 kbit/s @ 30 ms.framelength */
>+#define STREAM_MAXW16_60MS \
>+  200 /* 200 Word16 = 400 bytes = 53.4 kbit/s @ 60 ms.framelength */
> /* This is used only at the decoder bit-stream struct.
>  * - The encoder and decoder bitstream containers are of different size because
>  *   old iSAC limited the encoded bitstream to 600 bytes. But newer versions
>@@ -110,106 +108,104 @@
> /* storage size for bit counts */
> //#define BIT_COUNTER_SIZE                        30
> /* maximum order of any AR model or filter */
>-#define MAX_AR_MODEL_ORDER                      12
>+#define MAX_AR_MODEL_ORDER 12
> 
> /* Maximum number of iterations allowed to limit payload size */
>-#define MAX_PAYLOAD_LIMIT_ITERATION           1
>+#define MAX_PAYLOAD_LIMIT_ITERATION 1
> 
> /* Bandwidth estimator */
> 
>-#define MIN_ISAC_BW                           10000     /* Minimum bandwidth in bits per sec */
>-#define MAX_ISAC_BW                           32000     /* Maxmum bandwidth in bits per sec */
>-#define MIN_ISAC_MD                           5         /* Minimum Max Delay in ?? */
>-#define MAX_ISAC_MD                           25        /* Maxmum Max Delay in ?? */
>-#define DELAY_CORRECTION_MAX      717
>-#define DELAY_CORRECTION_MED      819
>-#define Thld_30_60         18000
>-#define Thld_60_30         27000
>-
>-/* assumed header size; we don't know the exact number (header compression may be used) */
>-#define HEADER_SIZE                           35       /* bytes */
>-#define INIT_FRAME_LEN                        60
>-#define INIT_BN_EST                           20000
>-#define INIT_BN_EST_Q7                        2560000  /* 20 kbps in Q7 */
>-#define INIT_REC_BN_EST_Q5                    789312   /* INIT_BN_EST + INIT_HDR_RATE in Q5 */
>+#define MIN_ISAC_BW 10000 /* Minimum bandwidth in bits per sec */
>+#define MAX_ISAC_BW 32000 /* Maxmum bandwidth in bits per sec */
>+#define MIN_ISAC_MD 5     /* Minimum Max Delay in ?? */
>+#define MAX_ISAC_MD 25    /* Maxmum Max Delay in ?? */
>+#define DELAY_CORRECTION_MAX 717
>+#define DELAY_CORRECTION_MED 819
>+#define Thld_30_60 18000
>+#define Thld_60_30 27000
>+
>+/* assumed header size; we don't know the exact number (header compression may
>+ * be used) */
>+#define HEADER_SIZE 35 /* bytes */
>+#define INIT_FRAME_LEN 60
>+#define INIT_BN_EST 20000
>+#define INIT_BN_EST_Q7 2560000    /* 20 kbps in Q7 */
>+#define INIT_REC_BN_EST_Q5 789312 /* INIT_BN_EST + INIT_HDR_RATE in Q5 */
> 
> /* 8738 in Q18 is ~ 1/30 */
>-/* #define INIT_HDR_RATE (((HEADER_SIZE * 8 * 1000) * 8738) >> NUM_BITS_TO_SHIFT (INIT_FRAME_LEN)) */
>-#define INIT_HDR_RATE                    4666
>+/* #define INIT_HDR_RATE (((HEADER_SIZE * 8 * 1000) * 8738) >> NUM_BITS_TO_SHIFT
>+ * (INIT_FRAME_LEN)) */
>+#define INIT_HDR_RATE 4666
> /* number of packets in a row for a high rate burst */
>-#define BURST_LEN                             3
>+#define BURST_LEN 3
> /* ms, max time between two full bursts */
>-#define BURST_INTERVAL                        800
>+#define BURST_INTERVAL 800
> /* number of packets in a row for initial high rate burst */
>-#define INIT_BURST_LEN                        5
>+#define INIT_BURST_LEN 5
> /* bits/s, rate for the first BURST_LEN packets */
>-#define INIT_RATE                             10240000 /* INIT_BN_EST in Q9 */
>-
>+#define INIT_RATE 10240000 /* INIT_BN_EST in Q9 */
> 
> /* For pitch analysis */
>-#define PITCH_FRAME_LEN                         240  /* (FRAMESAMPLES/2) 30 ms  */
>-#define PITCH_MAX_LAG                           140       /* 57 Hz  */
>-#define PITCH_MIN_LAG                           20                /* 400 Hz */
>-#define PITCH_MIN_LAG_Q8                        5120 /* 256 * PITCH_MIN_LAG */
>-#define OFFSET_Q8                               768  /* 256 * 3 */
>-
>-#define PITCH_MAX_GAIN_Q12      1843                  /* 0.45 */
>-#define PITCH_LAG_SPAN2                         65   /* (PITCH_MAX_LAG/2-PITCH_MIN_LAG/2+5) */
>-#define PITCH_CORR_LEN2                         60     /* 15 ms  */
>-#define PITCH_CORR_STEP2                        60   /* (PITCH_FRAME_LEN/4) */
>-#define PITCH_SUBFRAMES                         4
>-#define PITCH_SUBFRAME_LEN                      60   /* (PITCH_FRAME_LEN/PITCH_SUBFRAMES) */
>+#define PITCH_FRAME_LEN 240   /* (FRAMESAMPLES/2) 30 ms  */
>+#define PITCH_MAX_LAG 140     /* 57 Hz  */
>+#define PITCH_MIN_LAG 20      /* 400 Hz */
>+#define PITCH_MIN_LAG_Q8 5120 /* 256 * PITCH_MIN_LAG */
>+#define OFFSET_Q8 768         /* 256 * 3 */
>+
>+#define PITCH_MAX_GAIN_Q12 1843 /* 0.45 */
>+#define PITCH_LAG_SPAN2 65      /* (PITCH_MAX_LAG/2-PITCH_MIN_LAG/2+5) */
>+#define PITCH_CORR_LEN2 60      /* 15 ms  */
>+#define PITCH_CORR_STEP2 60     /* (PITCH_FRAME_LEN/4) */
>+#define PITCH_SUBFRAMES 4
>+#define PITCH_SUBFRAME_LEN 60 /* (PITCH_FRAME_LEN/PITCH_SUBFRAMES) */
> 
> /* For pitch filter */
>-#define PITCH_BUFFSIZE                   190  /* (PITCH_MAX_LAG + 50) Extra 50 for fraction and LP filters */
>-#define PITCH_INTBUFFSIZE               430  /* (PITCH_FRAME_LEN+PITCH_BUFFSIZE) */
>-#define PITCH_FRACS                             8
>-#define PITCH_FRACORDER                         9
>-#define PITCH_DAMPORDER                         5
>-
>+#define PITCH_BUFFSIZE \
>+  190 /* (PITCH_MAX_LAG + 50) Extra 50 for fraction and LP filters */
>+#define PITCH_INTBUFFSIZE 430 /* (PITCH_FRAME_LEN+PITCH_BUFFSIZE) */
>+#define PITCH_FRACS 8
>+#define PITCH_FRACORDER 9
>+#define PITCH_DAMPORDER 5
> 
> /* Order of high pass filter */
>-#define HPORDER                                 2
>-
>+#define HPORDER 2
> 
> /* PLC */
>-#define DECAY_RATE               10               /* Q15, 20% of decay every lost frame apllied linearly sample by sample*/
>-#define PLC_WAS_USED              1
>-#define PLC_NOT_USED              3
>-#define RECOVERY_OVERLAP         80
>-#define RESAMP_RES              256
>-#define RESAMP_RES_BIT            8
>-
>-
>+#define DECAY_RATE \
>+  10 /* Q15, 20% of decay every lost frame apllied linearly sample by sample*/
>+#define PLC_WAS_USED 1
>+#define PLC_NOT_USED 3
>+#define RECOVERY_OVERLAP 80
>+#define RESAMP_RES 256
>+#define RESAMP_RES_BIT 8
> 
> /* Define Error codes */
> /* 6000 General */
>-#define ISAC_MEMORY_ALLOCATION_FAILED    6010
>-#define ISAC_MODE_MISMATCH       6020
>-#define ISAC_DISALLOWED_BOTTLENECK     6030
>-#define ISAC_DISALLOWED_FRAME_LENGTH    6040
>+#define ISAC_MEMORY_ALLOCATION_FAILED 6010
>+#define ISAC_MODE_MISMATCH 6020
>+#define ISAC_DISALLOWED_BOTTLENECK 6030
>+#define ISAC_DISALLOWED_FRAME_LENGTH 6040
> /* 6200 Bandwidth estimator */
>-#define ISAC_RANGE_ERROR_BW_ESTIMATOR    6240
>+#define ISAC_RANGE_ERROR_BW_ESTIMATOR 6240
> /* 6400 Encoder */
>-#define ISAC_ENCODER_NOT_INITIATED     6410
>-#define ISAC_DISALLOWED_CODING_MODE     6420
>-#define ISAC_DISALLOWED_FRAME_MODE_ENCODER   6430
>-#define ISAC_DISALLOWED_BITSTREAM_LENGTH            6440
>-#define ISAC_PAYLOAD_LARGER_THAN_LIMIT              6450
>+#define ISAC_ENCODER_NOT_INITIATED 6410
>+#define ISAC_DISALLOWED_CODING_MODE 6420
>+#define ISAC_DISALLOWED_FRAME_MODE_ENCODER 6430
>+#define ISAC_DISALLOWED_BITSTREAM_LENGTH 6440
>+#define ISAC_PAYLOAD_LARGER_THAN_LIMIT 6450
> /* 6600 Decoder */
>-#define ISAC_DECODER_NOT_INITIATED     6610
>-#define ISAC_EMPTY_PACKET       6620
>+#define ISAC_DECODER_NOT_INITIATED 6610
>+#define ISAC_EMPTY_PACKET 6620
> #define ISAC_PACKET_TOO_SHORT 6625
>-#define ISAC_DISALLOWED_FRAME_MODE_DECODER   6630
>-#define ISAC_RANGE_ERROR_DECODE_FRAME_LENGTH  6640
>-#define ISAC_RANGE_ERROR_DECODE_BANDWIDTH   6650
>-#define ISAC_RANGE_ERROR_DECODE_PITCH_GAIN   6660
>-#define ISAC_RANGE_ERROR_DECODE_PITCH_LAG   6670
>-#define ISAC_RANGE_ERROR_DECODE_LPC     6680
>-#define ISAC_RANGE_ERROR_DECODE_SPECTRUM   6690
>-#define ISAC_LENGTH_MISMATCH      6730
>+#define ISAC_DISALLOWED_FRAME_MODE_DECODER 6630
>+#define ISAC_RANGE_ERROR_DECODE_FRAME_LENGTH 6640
>+#define ISAC_RANGE_ERROR_DECODE_BANDWIDTH 6650
>+#define ISAC_RANGE_ERROR_DECODE_PITCH_GAIN 6660
>+#define ISAC_RANGE_ERROR_DECODE_PITCH_LAG 6670
>+#define ISAC_RANGE_ERROR_DECODE_LPC 6680
>+#define ISAC_RANGE_ERROR_DECODE_SPECTRUM 6690
>+#define ISAC_LENGTH_MISMATCH 6730
> /* 6800 Call setup formats */
>-#define ISAC_INCOMPATIBLE_FORMATS     6810
>-
>+#define ISAC_INCOMPATIBLE_FORMATS 6810
> 
> #endif /* MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_SETTINGS_H_ */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/spectrum_ar_model_tables.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/spectrum_ar_model_tables.c
>index cf2dea7507e..4ef9a338ccb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/spectrum_ar_model_tables.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/spectrum_ar_model_tables.c
>@@ -16,8 +16,8 @@
>  *
>  */
> 
>-#include "spectrum_ar_model_tables.h"
>-#include "settings.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/spectrum_ar_model_tables.h"
>+#include "modules/audio_coding/codecs/isac/fix/source/settings.h"
> 
> /********************* AR Coefficient Tables ************************/
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/spectrum_ar_model_tables.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/spectrum_ar_model_tables.h
>index 9a7e9463969..2282a369cb6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/spectrum_ar_model_tables.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/spectrum_ar_model_tables.h
>@@ -19,8 +19,9 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_SPECTRUM_AR_MODEL_TABLES_H_
> #define MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_SPECTRUM_AR_MODEL_TABLES_H_
> 
>-#include "settings.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include <stdint.h>
>+
>+#include "modules/audio_coding/codecs/isac/fix/source/settings.h"
> 
> /********************* AR Coefficient Tables ************************/
> /* cdf for quantized reflection coefficient 1 */
>@@ -62,15 +63,15 @@ extern const int16_t WebRtcIsacfix_kRc6Levels[11];
> /* quantization boundary levels for reflection coefficients */
> extern const int16_t WebRtcIsacfix_kRcBound[12];
> 
>-/* initial indices for AR reflection coefficient quantizer and cdf table search */
>+/* initial indices for AR reflection coefficient quantizer and cdf table search
>+ */
> extern const uint16_t WebRtcIsacfix_kRcInitInd[AR_ORDER];
> 
> /* pointers to AR cdf tables */
>-extern const uint16_t *WebRtcIsacfix_kRcCdfPtr[AR_ORDER];
>+extern const uint16_t* WebRtcIsacfix_kRcCdfPtr[AR_ORDER];
> 
> /* pointers to AR representation levels tables */
>-extern const int16_t *WebRtcIsacfix_kRcLevPtr[AR_ORDER];
>-
>+extern const int16_t* WebRtcIsacfix_kRcLevPtr[AR_ORDER];
> 
> /******************** GAIN Coefficient Tables ***********************/
> /* cdf for Gain coefficient */
>@@ -83,7 +84,7 @@ extern const int32_t WebRtcIsacfix_kGain2Lev[18];
> extern const int32_t WebRtcIsacfix_kGain2Bound[19];
> 
> /* pointer to Gain cdf table */
>-extern const uint16_t *WebRtcIsacfix_kGainPtr[1];
>+extern const uint16_t* WebRtcIsacfix_kGainPtr[1];
> 
> /* Gain initial index for gain quantizer and cdf table search */
> extern const uint16_t WebRtcIsacfix_kGainInitInd[1];
>@@ -92,4 +93,5 @@ extern const uint16_t WebRtcIsacfix_kGainInitInd[1];
> /* Cosine table */
> extern const int16_t WebRtcIsacfix_kCos[6][60];
> 
>-#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_SPECTRUM_AR_MODEL_TABLES_H_ */
>+#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_SPECTRUM_AR_MODEL_TABLES_H_ \
>+        */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/structs.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/structs.h
>index 7a14e5c6390..59226ac24b3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/structs.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/structs.h
>@@ -18,80 +18,64 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_STRUCTS_H_
> #define MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_STRUCTS_H_
> 
>-
> #include "common_audio/signal_processing/include/signal_processing_library.h"
> #include "modules/audio_coding/codecs/isac/bandwidth_info.h"
> #include "modules/audio_coding/codecs/isac/fix/source/settings.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> /* Bitstream struct for decoder */
> typedef struct Bitstreamstruct_dec {
>-
>-  uint16_t  stream[INTERNAL_STREAM_SIZE_W16];  /* Array bytestream to decode */
>-  uint32_t  W_upper;          /* Upper boundary of interval W */
>-  uint32_t  streamval;
>-  uint16_t  stream_index;     /* Index to the current position in bytestream */
>-  int16_t   full;             /* 0 - first byte in memory filled, second empty*/
>+  uint16_t stream[INTERNAL_STREAM_SIZE_W16]; /* Array bytestream to decode */
>+  uint32_t W_upper;                          /* Upper boundary of interval W */
>+  uint32_t streamval;
>+  uint16_t stream_index; /* Index to the current position in bytestream */
>+  int16_t full;          /* 0 - first byte in memory filled, second empty*/
>   /* 1 - both bytes are empty (we just filled the previous memory */
> 
>-  size_t stream_size;  /* The size of stream in bytes. */
>+  size_t stream_size; /* The size of stream in bytes. */
> } Bitstr_dec;
> 
> /* Bitstream struct for encoder */
> typedef struct Bitstreamstruct_enc {
>-
>-  uint16_t  stream[STREAM_MAXW16_60MS];   /* Vector for adding encoded bytestream */
>-  uint32_t  W_upper;          /* Upper boundary of interval W */
>-  uint32_t  streamval;
>-  uint16_t  stream_index;     /* Index to the current position in bytestream */
>-  int16_t   full;             /* 0 - first byte in memory filled, second empty*/
>+  uint16_t
>+      stream[STREAM_MAXW16_60MS]; /* Vector for adding encoded bytestream */
>+  uint32_t W_upper;               /* Upper boundary of interval W */
>+  uint32_t streamval;
>+  uint16_t stream_index; /* Index to the current position in bytestream */
>+  int16_t full;          /* 0 - first byte in memory filled, second empty*/
>   /* 1 - both bytes are empty (we just filled the previous memory */
> 
> } Bitstr_enc;
> 
>-
> typedef struct {
>-
>   int16_t DataBufferLoQ0[WINLEN];
>   int16_t DataBufferHiQ0[WINLEN];
> 
>-  int32_t CorrBufLoQQ[ORDERLO+1];
>-  int32_t CorrBufHiQQ[ORDERHI+1];
>+  int32_t CorrBufLoQQ[ORDERLO + 1];
>+  int32_t CorrBufHiQQ[ORDERHI + 1];
> 
>-  int16_t CorrBufLoQdom[ORDERLO+1];
>-  int16_t CorrBufHiQdom[ORDERHI+1];
>+  int16_t CorrBufLoQdom[ORDERLO + 1];
>+  int16_t CorrBufHiQdom[ORDERHI + 1];
> 
>-  int32_t PreStateLoGQ15[ORDERLO+1];
>-  int32_t PreStateHiGQ15[ORDERHI+1];
>+  int32_t PreStateLoGQ15[ORDERLO + 1];
>+  int32_t PreStateHiGQ15[ORDERHI + 1];
> 
>   uint32_t OldEnergy;
> 
> } MaskFiltstr_enc;
> 
>-
>-
> typedef struct {
>-
>-  int16_t PostStateLoGQ0[ORDERLO+1];
>-  int16_t PostStateHiGQ0[ORDERHI+1];
>+  int16_t PostStateLoGQ0[ORDERLO + 1];
>+  int16_t PostStateHiGQ0[ORDERHI + 1];
> 
>   uint32_t OldEnergy;
> 
> } MaskFiltstr_dec;
> 
>-
>-
>-
>-
>-
>-
>-
> typedef struct {
>+  // state vectors for each of the two analysis filters
> 
>-  //state vectors for each of the two analysis filters
>-
>-  int32_t INSTAT1_fix[2*(QORDER-1)];
>-  int32_t INSTAT2_fix[2*(QORDER-1)];
>+  int32_t INSTAT1_fix[2 * (QORDER - 1)];
>+  int32_t INSTAT2_fix[2 * (QORDER - 1)];
>   int16_t INLABUF1_fix[QLOOKAHEAD];
>   int16_t INLABUF2_fix[QLOOKAHEAD];
> 
>@@ -100,12 +84,10 @@ typedef struct {
> 
> } PreFiltBankstr;
> 
>-
> typedef struct {
>-
>-  //state vectors for each of the two analysis filters
>-  int32_t STATE_0_LOWER_fix[2*POSTQORDER];
>-  int32_t STATE_0_UPPER_fix[2*POSTQORDER];
>+  // state vectors for each of the two analysis filters
>+  int32_t STATE_0_LOWER_fix[2 * POSTQORDER];
>+  int32_t STATE_0_UPPER_fix[2 * POSTQORDER];
> 
>   /* High pass filter */
> 
>@@ -115,8 +97,6 @@ typedef struct {
> } PostFiltBankstr;
> 
> typedef struct {
>-
>-
>   /* data buffer for pitch filter */
>   int16_t ubufQQ[PITCH_BUFFSIZE];
> 
>@@ -129,42 +109,35 @@ typedef struct {
> 
> } PitchFiltstr;
> 
>-
>-
> typedef struct {
>+  // for inital estimator
>+  int16_t dec_buffer16[PITCH_CORR_LEN2 + PITCH_CORR_STEP2 + PITCH_MAX_LAG / 2 -
>+                       PITCH_FRAME_LEN / 2 + 2];
>+  int32_t decimator_state32[2 * ALLPASSSECTIONS + 1];
>+  int16_t inbuf[QLOOKAHEAD];
> 
>-  //for inital estimator
>-  int16_t   dec_buffer16[PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2-PITCH_FRAME_LEN/2+2];
>-  int32_t   decimator_state32[2*ALLPASSSECTIONS+1];
>-  int16_t   inbuf[QLOOKAHEAD];
>-
>-  PitchFiltstr  PFstr_wght;
>-  PitchFiltstr  PFstr;
>-
>+  PitchFiltstr PFstr_wght;
>+  PitchFiltstr PFstr;
> 
> } PitchAnalysisStruct;
> 
>-
> typedef struct {
>   /* Parameters used in PLC to avoid re-computation       */
> 
>   /* --- residual signals --- */
>-  int16_t prevPitchInvIn[FRAMESAMPLES/2];
>-  int16_t prevPitchInvOut[PITCH_MAX_LAG + 10];            // [FRAMESAMPLES/2]; save 90
>-  int32_t prevHP[PITCH_MAX_LAG + 10];                     // [FRAMESAMPLES/2]; save 90
>-
>+  int16_t prevPitchInvIn[FRAMESAMPLES / 2];
>+  int16_t prevPitchInvOut[PITCH_MAX_LAG + 10];  // [FRAMESAMPLES/2]; save 90
>+  int32_t prevHP[PITCH_MAX_LAG + 10];           // [FRAMESAMPLES/2]; save 90
> 
>   int16_t decayCoeffPriodic; /* how much to supress a sample */
>   int16_t decayCoeffNoise;
>-  int16_t used;       /* if PLC is used */
>-
>-
>-  int16_t *lastPitchLP;                                  // [FRAMESAMPLES/2]; saved 240;
>+  int16_t used; /* if PLC is used */
> 
>+  int16_t* lastPitchLP;  // [FRAMESAMPLES/2]; saved 240;
> 
>   /* --- LPC side info --- */
>-  int16_t lofilt_coefQ15[ ORDERLO ];
>-  int16_t hifilt_coefQ15[ ORDERHI ];
>+  int16_t lofilt_coefQ15[ORDERLO];
>+  int16_t hifilt_coefQ15[ORDERHI];
>   int32_t gain_lo_hiQ17[2];
> 
>   /* --- LTP side info --- */
>@@ -173,95 +146,101 @@ typedef struct {
>   int16_t lastPitchLag_Q7;
> 
>   /* --- Add-overlap in recovery packet --- */
>-  int16_t overlapLP[ RECOVERY_OVERLAP ];                 // [FRAMESAMPLES/2]; saved 160
>+  int16_t overlapLP[RECOVERY_OVERLAP];  // [FRAMESAMPLES/2]; saved 160
> 
>   int16_t pitchCycles;
>   int16_t A;
>   int16_t B;
>   size_t pitchIndex;
>   size_t stretchLag;
>-  int16_t *prevPitchLP;                                  // [ FRAMESAMPLES/2 ]; saved 240
>+  int16_t* prevPitchLP;  // [ FRAMESAMPLES/2 ]; saved 240
>   int16_t seed;
> 
>   int16_t std;
> } PLCstr;
> 
>-
>-
> /* Have instance of struct together with other iSAC structs */
> typedef struct {
>-
>-  int16_t   prevFrameSizeMs;      /* Previous frame size (in ms) */
>-  uint16_t  prevRtpNumber;      /* Previous RTP timestamp from received packet */
>+  int16_t prevFrameSizeMs; /* Previous frame size (in ms) */
>+  uint16_t prevRtpNumber;  /* Previous RTP timestamp from received packet */
>   /* (in samples relative beginning)  */
>-  uint32_t  prevSendTime;   /* Send time for previous packet, from RTP header */
>-  uint32_t  prevArrivalTime;      /* Arrival time for previous packet (in ms using timeGetTime()) */
>-  uint16_t  prevRtpRate;          /* rate of previous packet, derived from RTP timestamps (in bits/s) */
>-  uint32_t  lastUpdate;           /* Time since the last update of the Bottle Neck estimate (in samples) */
>-  uint32_t  lastReduction;        /* Time sinse the last reduction (in samples) */
>-  int32_t   countUpdates;         /* How many times the estimate was update in the beginning */
>-
>-  /* The estimated bottle neck rate from there to here (in bits/s)                */
>-  uint32_t  recBw;
>-  uint32_t  recBwInv;
>-  uint32_t  recBwAvg;
>-  uint32_t  recBwAvgQ;
>-
>-  uint32_t  minBwInv;
>-  uint32_t  maxBwInv;
>-
>-  /* The estimated mean absolute jitter value, as seen on this side (in ms)       */
>-  int32_t   recJitter;
>-  int32_t   recJitterShortTerm;
>-  int32_t   recJitterShortTermAbs;
>-  int32_t   recMaxDelay;
>-  int32_t   recMaxDelayAvgQ;
>-
>-
>-  int16_t   recHeaderRate;         /* (assumed) bitrate for headers (bps) */
>-
>-  uint32_t  sendBwAvg;           /* The estimated bottle neck rate from here to there (in bits/s) */
>-  int32_t   sendMaxDelayAvg;    /* The estimated mean absolute jitter value, as seen on the other siee (in ms)  */
>-
>-
>-  int16_t   countRecPkts;          /* number of packets received since last update */
>-  int16_t   highSpeedRec;        /* flag for marking that a high speed network has been detected downstream */
>+  uint32_t prevSendTime;    /* Send time for previous packet, from RTP header */
>+  uint32_t prevArrivalTime; /* Arrival time for previous packet (in ms using
>+                               timeGetTime()) */
>+  uint16_t
>+      prevRtpRate; /* rate of previous packet, derived from RTP timestamps (in
>+                      bits/s) */
>+  uint32_t
>+      lastUpdate;         /* Time since the last update of the Bottle Neck estimate (in
>+                             samples) */
>+  uint32_t lastReduction; /* Time sinse the last reduction (in samples) */
>+  int32_t countUpdates;   /* How many times the estimate was update in the
>+                             beginning */
>+
>+  /* The estimated bottle neck rate from there to here (in bits/s) */
>+  uint32_t recBw;
>+  uint32_t recBwInv;
>+  uint32_t recBwAvg;
>+  uint32_t recBwAvgQ;
>+
>+  uint32_t minBwInv;
>+  uint32_t maxBwInv;
>+
>+  /* The estimated mean absolute jitter value, as seen on this side (in ms) */
>+  int32_t recJitter;
>+  int32_t recJitterShortTerm;
>+  int32_t recJitterShortTermAbs;
>+  int32_t recMaxDelay;
>+  int32_t recMaxDelayAvgQ;
>+
>+  int16_t recHeaderRate; /* (assumed) bitrate for headers (bps) */
>+
>+  uint32_t sendBwAvg; /* The estimated bottle neck rate from here to there (in
>+                         bits/s) */
>+  int32_t
>+      sendMaxDelayAvg; /* The estimated mean absolute jitter value, as seen on
>+                          the other siee (in ms)  */
>+
>+  int16_t countRecPkts; /* number of packets received since last update */
>+  int16_t highSpeedRec; /* flag for marking that a high speed network has been
>+                           detected downstream */
> 
>   /* number of consecutive pkts sent during which the bwe estimate has
>-     remained at a value greater than the downstream threshold for determining highspeed network */
>-  int16_t   countHighSpeedRec;
>+     remained at a value greater than the downstream threshold for determining
>+     highspeed network */
>+  int16_t countHighSpeedRec;
> 
>-  /* flag indicating bwe should not adjust down immediately for very late pckts */
>-  int16_t   inWaitPeriod;
>+  /* flag indicating bwe should not adjust down immediately for very late pckts
>+   */
>+  int16_t inWaitPeriod;
> 
>   /* variable holding the time of the start of a window of time when
>      bwe should not adjust down immediately for very late pckts */
>-  uint32_t  startWaitPeriod;
>+  uint32_t startWaitPeriod;
> 
>   /* number of consecutive pkts sent during which the bwe estimate has
>-     remained at a value greater than the upstream threshold for determining highspeed network */
>-  int16_t   countHighSpeedSent;
>+     remained at a value greater than the upstream threshold for determining
>+     highspeed network */
>+  int16_t countHighSpeedSent;
> 
>-  /* flag indicated the desired number of packets over threshold rate have been sent and
>-     bwe will assume the connection is over broadband network */
>-  int16_t   highSpeedSend;
>+  /* flag indicated the desired number of packets over threshold rate have been
>+     sent and bwe will assume the connection is over broadband network */
>+  int16_t highSpeedSend;
> 
>   IsacBandwidthInfo external_bw_info;
> } BwEstimatorstr;
> 
>-
> typedef struct {
>-
>   /* boolean, flags if previous packet exceeded B.N. */
>-  int16_t    PrevExceed;
>+  int16_t PrevExceed;
>   /* ms */
>-  int16_t    ExceedAgo;
>+  int16_t ExceedAgo;
>   /* packets left to send in current burst */
>-  int16_t    BurstCounter;
>+  int16_t BurstCounter;
>   /* packets */
>-  int16_t    InitCounter;
>+  int16_t InitCounter;
>   /* ms remaining in buffer when next packet will be sent */
>-  int16_t    StillBuffered;
>+  int16_t StillBuffered;
> 
> } RateModel;
> 
>@@ -271,112 +250,107 @@ typedef struct {
>    handle 60 ms of data.
> */
> typedef struct {
>-
>   /* Used to keep track of if it is first or second part of 60 msec packet */
>-  int     startIdx;
>+  int startIdx;
> 
>   /* Frame length in samples */
>-  int16_t         framelength;
>+  int16_t framelength;
> 
>   /* Pitch Gain */
>-  int16_t   pitchGain_index[2];
>+  int16_t pitchGain_index[2];
> 
>   /* Pitch Lag */
>-  int32_t   meanGain[2];
>-  int16_t   pitchIndex[PITCH_SUBFRAMES*2];
>+  int32_t meanGain[2];
>+  int16_t pitchIndex[PITCH_SUBFRAMES * 2];
> 
>   /* LPC */
>-  int32_t         LPCcoeffs_g[12*2]; /* KLT_ORDER_GAIN = 12 */
>-  int16_t   LPCindex_s[108*2]; /* KLT_ORDER_SHAPE = 108 */
>-  int16_t   LPCindex_g[12*2];  /* KLT_ORDER_GAIN = 12 */
>+  int32_t LPCcoeffs_g[12 * 2]; /* KLT_ORDER_GAIN = 12 */
>+  int16_t LPCindex_s[108 * 2]; /* KLT_ORDER_SHAPE = 108 */
>+  int16_t LPCindex_g[12 * 2];  /* KLT_ORDER_GAIN = 12 */
> 
>   /* Encode Spec */
>-  int16_t   fre[FRAMESAMPLES];
>-  int16_t   fim[FRAMESAMPLES];
>-  int16_t   AvgPitchGain[2];
>+  int16_t fre[FRAMESAMPLES];
>+  int16_t fim[FRAMESAMPLES];
>+  int16_t AvgPitchGain[2];
> 
>   /* Used in adaptive mode only */
>-  int     minBytes;
>+  int minBytes;
> 
> } IsacSaveEncoderData;
> 
> typedef struct {
>-
>-  Bitstr_enc          bitstr_obj;
>-  MaskFiltstr_enc     maskfiltstr_obj;
>-  PreFiltBankstr      prefiltbankstr_obj;
>-  PitchFiltstr        pitchfiltstr_obj;
>+  Bitstr_enc bitstr_obj;
>+  MaskFiltstr_enc maskfiltstr_obj;
>+  PreFiltBankstr prefiltbankstr_obj;
>+  PitchFiltstr pitchfiltstr_obj;
>   PitchAnalysisStruct pitchanalysisstr_obj;
>-  RateModel           rate_data_obj;
>+  RateModel rate_data_obj;
> 
>-  int16_t         buffer_index;
>-  int16_t         current_framesamples;
>+  int16_t buffer_index;
>+  int16_t current_framesamples;
> 
>-  int16_t      data_buffer_fix[FRAMESAMPLES]; // the size was MAX_FRAMESAMPLES
>+  int16_t data_buffer_fix[FRAMESAMPLES];  // the size was MAX_FRAMESAMPLES
> 
>-  int16_t         frame_nb;
>-  int16_t         BottleNeck;
>-  int16_t         MaxDelay;
>-  int16_t         new_framelength;
>-  int16_t         s2nr;
>-  uint16_t        MaxBits;
>+  int16_t frame_nb;
>+  int16_t BottleNeck;
>+  int16_t MaxDelay;
>+  int16_t new_framelength;
>+  int16_t s2nr;
>+  uint16_t MaxBits;
> 
>-  int16_t         bitstr_seed;
>+  int16_t bitstr_seed;
> #ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
>-  PostFiltBankstr     interpolatorstr_obj;
>+  PostFiltBankstr interpolatorstr_obj;
> #endif
> 
>-  IsacSaveEncoderData *SaveEnc_ptr;
>-  int16_t         payloadLimitBytes30; /* Maximum allowed number of bits for a 30 msec packet */
>-  int16_t         payloadLimitBytes60; /* Maximum allowed number of bits for a 30 msec packet */
>-  int16_t         maxPayloadBytes;     /* Maximum allowed number of bits for both 30 and 60 msec packet */
>-  int16_t         maxRateInBytes;      /* Maximum allowed rate in bytes per 30 msec packet */
>-  int16_t         enforceFrameSize;    /* If set iSAC will never change packet size */
>+  IsacSaveEncoderData* SaveEnc_ptr;
>+  int16_t payloadLimitBytes30; /* Maximum allowed number of bits for a 30 msec
>+                                  packet */
>+  int16_t payloadLimitBytes60; /* Maximum allowed number of bits for a 30 msec
>+                                  packet */
>+  int16_t maxPayloadBytes;     /* Maximum allowed number of bits for both 30 and 60
>+                                  msec packet */
>+  int16_t maxRateInBytes; /* Maximum allowed rate in bytes per 30 msec packet */
>+  int16_t enforceFrameSize; /* If set iSAC will never change packet size */
> 
> } IsacFixEncoderInstance;
> 
>-
> typedef struct {
>-
>-  Bitstr_dec          bitstr_obj;
>-  MaskFiltstr_dec     maskfiltstr_obj;
>-  PostFiltBankstr     postfiltbankstr_obj;
>-  PitchFiltstr        pitchfiltstr_obj;
>-  PLCstr              plcstr_obj;               /* TS; for packet loss concealment */
>+  Bitstr_dec bitstr_obj;
>+  MaskFiltstr_dec maskfiltstr_obj;
>+  PostFiltBankstr postfiltbankstr_obj;
>+  PitchFiltstr pitchfiltstr_obj;
>+  PLCstr plcstr_obj; /* TS; for packet loss concealment */
> 
> #ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
>-  PreFiltBankstr      decimatorstr_obj;
>+  PreFiltBankstr decimatorstr_obj;
> #endif
> 
> } IsacFixDecoderInstance;
> 
>-
>-
> typedef struct {
>-
>   IsacFixEncoderInstance ISACenc_obj;
>   IsacFixDecoderInstance ISACdec_obj;
>-  BwEstimatorstr     bwestimator_obj;
>-  int16_t         CodingMode;       /* 0 = adaptive; 1 = instantaneous */
>-  int16_t   errorcode;
>-  int16_t   initflag;  /* 0 = nothing initiated; 1 = encoder or decoder */
>+  BwEstimatorstr bwestimator_obj;
>+  int16_t CodingMode; /* 0 = adaptive; 1 = instantaneous */
>+  int16_t errorcode;
>+  int16_t initflag; /* 0 = nothing initiated; 1 = encoder or decoder */
>   /* not initiated; 2 = all initiated */
> } ISACFIX_SubStruct;
> 
>-
> typedef struct {
>-  int32_t   lpcGains[12];     /* 6 lower-band & 6 upper-band we may need to double it for 60*/
>+  int32_t lpcGains
>+      [12]; /* 6 lower-band & 6 upper-band we may need to double it for 60*/
>   /* */
>-  uint32_t  W_upper;          /* Upper boundary of interval W */
>-  uint32_t  streamval;
>-  uint16_t  stream_index;     /* Index to the current position in bytestream */
>-  int16_t   full;             /* 0 - first byte in memory filled, second empty*/
>+  uint32_t W_upper; /* Upper boundary of interval W */
>+  uint32_t streamval;
>+  uint16_t stream_index; /* Index to the current position in bytestream */
>+  int16_t full;          /* 0 - first byte in memory filled, second empty*/
>   /* 1 - both bytes are empty (we just filled the previous memory */
>-  uint16_t  beforeLastWord;
>-  uint16_t  lastWord;
>+  uint16_t beforeLastWord;
>+  uint16_t lastWord;
> } transcode_obj;
> 
>+// Bitstr_enc myBitStr;
> 
>-//Bitstr_enc myBitStr;
>-
>-#endif  /* MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_STRUCTS_H_ */
>+#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_STRUCTS_H_ */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/transform.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/transform.c
>index 2f1275da5c1..80b244b5f14 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/transform.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/transform.c
>@@ -18,6 +18,7 @@
> #include "modules/audio_coding/codecs/isac/fix/source/codec.h"
> #include "modules/audio_coding/codecs/isac/fix/source/fft.h"
> #include "modules/audio_coding/codecs/isac/fix/source/settings.h"
>+#include "modules/third_party/fft/fft.h"
> 
> /* Tables are defined in transform_tables.c file or ARM assembly files. */
> /* Cosine table 1 in Q14 */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/transform_tables.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/transform_tables.c
>index eed88e4a3ba..e661effddef 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/transform_tables.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/transform_tables.c
>@@ -13,8 +13,9 @@
>  * transform functions WebRtcIsacfix_Time2Spec and WebRtcIsacfix_Spec2Time.
>  */
> 
>+#include <stdint.h>
>+
> #include "modules/audio_coding/codecs/isac/fix/source/settings.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> /* Cosine table 1 in Q14. */
> const int16_t WebRtcIsacfix_kCosTab1[FRAMESAMPLES/2] = {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/transform_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/transform_unittest.cc
>index 347b0492e9b..a058530c255 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/transform_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/source/transform_unittest.cc
>@@ -12,147 +12,156 @@
> #include "system_wrappers/include/cpu_features_wrapper.h"
> #include "test/gtest.h"
> 
>-static const int kSamples = FRAMESAMPLES/2;
>+static const int kSamples = FRAMESAMPLES / 2;
> static const int32_t spec2time_out_expected_1[kSamples] = {
>-  -3366470, -2285227,
>-  -3415765, -2310215, -3118030, -2222470, -3030254, -2192091, -3423170,
>-  -2216041, -3305541, -2171936, -3195767, -2095779, -3153304, -2157560,
>-  -3071167, -2032108, -3101190, -1972016, -3103824, -2089118, -3139811,
>-  -1898337, -3102801, -2055082, -3029665, -1854140, -2962586, -1966454,
>-  -3071167, -1894588, -2851743, -1917315, -2848087, -1594932, -2799242,
>-  -1462184, -2845887, -1437599, -2691776, -1329637, -2770659, -1268491,
>-  -2625161, -1578991, -2460299, -1186385, -2365613, -1039354, -2322608,
>-  -958518, -2271749, -789860, -2254538, -850308, -2384436, -850959, -2133734,
>-  -587678, -2093316, -495115, -1973364, -475177, -1801282, -173507,
>-  -1848516, -158015, -1792018, -62648, -1643313, 214746, -1500758, 267077,
>-  -1450193, 560521, -1521579, 675283, -1345408, 857559, -1300822, 1116332,
>-  -1294533, 1241117, -1070027, 1263503, -983816, 1529821, -1019586,
>-  1910421, -955420, 2073688, -836459, 2401105, -653905, 2690474, -731425,
>-  2930131, -935234, 3299500, -875978, 3523432, -878906, 3924822, -1081630,
>-  4561267, -1203023, 5105274, -1510983, 6052762, -2294646, 7021597,
>-  -3108053, 8826736, -4935222, 11678789, -8442713, 18725700, -21526692,
>-  25420577, 19589811, -28108666, 12634054, -14483066, 6263217, -9979706,
>-  3665661, -7909736, 2531530, -6434896, 1700772, -5525393, 1479473,
>-  -4894262, 1231760, -4353044, 1032940, -3786590, 941152, -3331614,
>-  665090, -2851619, 830696, -2762201, 958007, -2483118, 788233, -2184965,
>-  804825, -1967306, 1007255, -1862474, 920889, -1457506, 755406, -1405841,
>-  890230, -1302124, 1161599, -701867, 1154163, -1083366, 1204743, -513581,
>-  1547264, -650636, 1493384, -285543, 1771863, -277906, 1841343, -9078,
>-  1751863, 230222, 1819578, 207170, 1978972, 398137, 2106468, 552155,
>-  1997624, 685213, 2129520, 601078, 2238736, 944591, 2441879, 1194178,
>-  2355280, 986124, 2393328, 1049005, 2417944, 1208368, 2489516, 1352023,
>-  2572118, 1445283, 2856081, 1532997, 2742279, 1615877, 2915274, 1808036,
>-  2856871, 1806936, 3241747, 1622461, 2978558, 1841297, 3010378, 1923666,
>-  3271367, 2126700, 3070935, 1956958, 3107588, 2128405, 3288872, 2114911,
>-  3315952, 2406651, 3344038, 2370199, 3368980, 2144361, 3305030, 2183803,
>-  3401450, 2523102, 3405463, 2452475, 3463355, 2421678, 3551968, 2431949,
>-  3477251, 2148125, 3244489, 2174090};
>+    -3366470, -2285227, -3415765,  -2310215, -3118030,  -2222470, -3030254,
>+    -2192091, -3423170, -2216041,  -3305541, -2171936,  -3195767, -2095779,
>+    -3153304, -2157560, -3071167,  -2032108, -3101190,  -1972016, -3103824,
>+    -2089118, -3139811, -1898337,  -3102801, -2055082,  -3029665, -1854140,
>+    -2962586, -1966454, -3071167,  -1894588, -2851743,  -1917315, -2848087,
>+    -1594932, -2799242, -1462184,  -2845887, -1437599,  -2691776, -1329637,
>+    -2770659, -1268491, -2625161,  -1578991, -2460299,  -1186385, -2365613,
>+    -1039354, -2322608, -958518,   -2271749, -789860,   -2254538, -850308,
>+    -2384436, -850959,  -2133734,  -587678,  -2093316,  -495115,  -1973364,
>+    -475177,  -1801282, -173507,   -1848516, -158015,   -1792018, -62648,
>+    -1643313, 214746,   -1500758,  267077,   -1450193,  560521,   -1521579,
>+    675283,   -1345408, 857559,    -1300822, 1116332,   -1294533, 1241117,
>+    -1070027, 1263503,  -983816,   1529821,  -1019586,  1910421,  -955420,
>+    2073688,  -836459,  2401105,   -653905,  2690474,   -731425,  2930131,
>+    -935234,  3299500,  -875978,   3523432,  -878906,   3924822,  -1081630,
>+    4561267,  -1203023, 5105274,   -1510983, 6052762,   -2294646, 7021597,
>+    -3108053, 8826736,  -4935222,  11678789, -8442713,  18725700, -21526692,
>+    25420577, 19589811, -28108666, 12634054, -14483066, 6263217,  -9979706,
>+    3665661,  -7909736, 2531530,   -6434896, 1700772,   -5525393, 1479473,
>+    -4894262, 1231760,  -4353044,  1032940,  -3786590,  941152,   -3331614,
>+    665090,   -2851619, 830696,    -2762201, 958007,    -2483118, 788233,
>+    -2184965, 804825,   -1967306,  1007255,  -1862474,  920889,   -1457506,
>+    755406,   -1405841, 890230,    -1302124, 1161599,   -701867,  1154163,
>+    -1083366, 1204743,  -513581,   1547264,  -650636,   1493384,  -285543,
>+    1771863,  -277906,  1841343,   -9078,    1751863,   230222,   1819578,
>+    207170,   1978972,  398137,    2106468,  552155,    1997624,  685213,
>+    2129520,  601078,   2238736,   944591,   2441879,   1194178,  2355280,
>+    986124,   2393328,  1049005,   2417944,  1208368,   2489516,  1352023,
>+    2572118,  1445283,  2856081,   1532997,  2742279,   1615877,  2915274,
>+    1808036,  2856871,  1806936,   3241747,  1622461,   2978558,  1841297,
>+    3010378,  1923666,  3271367,   2126700,  3070935,   1956958,  3107588,
>+    2128405,  3288872,  2114911,   3315952,  2406651,   3344038,  2370199,
>+    3368980,  2144361,  3305030,   2183803,  3401450,   2523102,  3405463,
>+    2452475,  3463355,  2421678,   3551968,  2431949,   3477251,  2148125,
>+    3244489,  2174090};
> static const int32_t spec2time_out_expected_2[kSamples] = {
>-  1691694, -2499988, -2035547,
>-  1060469, 988634, -2044502, -306271, 2041000, 201454, -2289456, 93694,
>-  2129427, -369152, -1887834, 860796, 2089102, -929424, -1673956, 1395291,
>-  1785651, -1619673, -1380109, 1963449, 1093311, -2111007, -840456,
>-  2372786, 578119, -2242702, 89774, 2463304, -132717, -2121480, 643634,
>-  2277636, -1125999, -1995858, 1543748, 2227861, -1483779, -1495491,
>-  2102642, 1833876, -1920568, -958378, 2485101, 772261, -2454257, -24942,
>-  2918714, 136838, -2500453, 816118, 3039735, -746560, -2365815, 1586396,
>-  2714951, -1511696, -1942334, 2571792, 2182827, -2325335, -1311543,
>-  3055970, 1367220, -2737182, -110626, 3889222, 631008, -3280879, 853066,
>-  4122279, -706638, -3334449, 2148311, 3993512, -1846301, -3004894,
>-  3426779, 3329522, -3165264, -2242423, 4756866, 2557711, -4131280,
>-  -805259, 5702711, 1120592, -4852821, 743664, 6476444, -621186, -5465828,
>-  2815787, 6768835, -3017442, -5338409, 5658126, 6838454, -5492288,
>-  -4682382, 8874947, 6153814, -8832561, -2649251, 12817398, 4237692,
>-  -13000247, 1190661, 18986363, -115738, -19693978, 9908367, 30660381,
>-  -10632635, -37962068, 47022884, 89744622, -42087632, 40279224,
>-  -88869341, -47542383, 38572364, 10441576, -30339718, -9926740, 19896578,
>-  28009, -18886612, -1124047, 13232498, -4150304, -12770551, 2637074,
>-  9051831, -6162211, -8713972, 4557937, 5489716, -6862312, -5532349,
>-  5415449, 2791310, -6999367, -2790102, 5375806, 546222, -6486452,
>-  -821261, 4994973, -1278840, -5645501, 1060484, 3996285, -2503954,
>-  -4653629, 2220549, 3036977, -3282133, -3318585, 2780636, 1789880,
>-  -4004589, -2041031, 3105373, 574819, -3992722, -971004, 3001703,
>-  -676739, -3841508, 417284, 2897970, -1427018, -3058480, 1189948,
>-  2210960, -2268992, -2603272, 1949785, 1576172, -2720404, -1891738,
>-  2309456, 769178, -2975646, -707150, 2424652, -88039, -2966660, -65452,
>-  2320780, -957557, -2798978, 744640, 1879794, -1672081, -2365319,
>-  1253309, 1366383, -2204082, -1544367, 1801452, 613828, -2531994,
>-  -983847, 2064842, 118326, -2613790, -203220, 2219635, -730341, -2641861,
>-  563557, 1765434, -1329916, -2272927, 1037138, 1266725, -1939220,
>-  -1588643, 1754528, 816552, -2376303, -1099167, 1864999, 122477,
>-  -2422762, -400027, 1889228, -579916, -2490353, 287139, 2011318,
>-  -1176657, -2502978, 812896, 1116502, -1940211};
>+    1691694,   -2499988, -2035547,  1060469,   988634,    -2044502, -306271,
>+    2041000,   201454,   -2289456,  93694,     2129427,   -369152,  -1887834,
>+    860796,    2089102,  -929424,   -1673956,  1395291,   1785651,  -1619673,
>+    -1380109,  1963449,  1093311,   -2111007,  -840456,   2372786,  578119,
>+    -2242702,  89774,    2463304,   -132717,   -2121480,  643634,   2277636,
>+    -1125999,  -1995858, 1543748,   2227861,   -1483779,  -1495491, 2102642,
>+    1833876,   -1920568, -958378,   2485101,   772261,    -2454257, -24942,
>+    2918714,   136838,   -2500453,  816118,    3039735,   -746560,  -2365815,
>+    1586396,   2714951,  -1511696,  -1942334,  2571792,   2182827,  -2325335,
>+    -1311543,  3055970,  1367220,   -2737182,  -110626,   3889222,  631008,
>+    -3280879,  853066,   4122279,   -706638,   -3334449,  2148311,  3993512,
>+    -1846301,  -3004894, 3426779,   3329522,   -3165264,  -2242423, 4756866,
>+    2557711,   -4131280, -805259,   5702711,   1120592,   -4852821, 743664,
>+    6476444,   -621186,  -5465828,  2815787,   6768835,   -3017442, -5338409,
>+    5658126,   6838454,  -5492288,  -4682382,  8874947,   6153814,  -8832561,
>+    -2649251,  12817398, 4237692,   -13000247, 1190661,   18986363, -115738,
>+    -19693978, 9908367,  30660381,  -10632635, -37962068, 47022884, 89744622,
>+    -42087632, 40279224, -88869341, -47542383, 38572364,  10441576, -30339718,
>+    -9926740,  19896578, 28009,     -18886612, -1124047,  13232498, -4150304,
>+    -12770551, 2637074,  9051831,   -6162211,  -8713972,  4557937,  5489716,
>+    -6862312,  -5532349, 5415449,   2791310,   -6999367,  -2790102, 5375806,
>+    546222,    -6486452, -821261,   4994973,   -1278840,  -5645501, 1060484,
>+    3996285,   -2503954, -4653629,  2220549,   3036977,   -3282133, -3318585,
>+    2780636,   1789880,  -4004589,  -2041031,  3105373,   574819,   -3992722,
>+    -971004,   3001703,  -676739,   -3841508,  417284,    2897970,  -1427018,
>+    -3058480,  1189948,  2210960,   -2268992,  -2603272,  1949785,  1576172,
>+    -2720404,  -1891738, 2309456,   769178,    -2975646,  -707150,  2424652,
>+    -88039,    -2966660, -65452,    2320780,   -957557,   -2798978, 744640,
>+    1879794,   -1672081, -2365319,  1253309,   1366383,   -2204082, -1544367,
>+    1801452,   613828,   -2531994,  -983847,   2064842,   118326,   -2613790,
>+    -203220,   2219635,  -730341,   -2641861,  563557,    1765434,  -1329916,
>+    -2272927,  1037138,  1266725,   -1939220,  -1588643,  1754528,  816552,
>+    -2376303,  -1099167, 1864999,   122477,    -2422762,  -400027,  1889228,
>+    -579916,   -2490353, 287139,    2011318,   -1176657,  -2502978, 812896,
>+    1116502,   -1940211};
> static const int16_t time2spec_out_expected_1[kSamples] = {
>-  20342, 23889, -10063, -9419,
>-  3242, 7280, -2012, -5029, 332, 4478, -97, -3244, -891, 3117, 773, -2204,
>-  -1335, 2009, 1236, -1469, -1562, 1277, 1366, -815, -1619, 599, 1449, -177,
>-  -1507, 116, 1294, 263, -1338, -244, 1059, 553, -1045, -549, 829, 826,
>-  -731, -755, 516, 909, -427, -853, 189, 1004, -184, -828, -108, 888, 72,
>-  -700, -280, 717, 342, -611, -534, 601, 534, -374, -646, 399, 567, -171,
>-  -720, 234, 645, -11, -712, -26, 593, 215, -643, -172, 536, 361, -527,
>-  -403, 388, 550, -361, -480, 208, 623, -206, -585, 41, 578, 12, -504,
>-  -182, 583, 218, -437, -339, 499, 263, -354, -450, 347, 456, -193, -524,
>-  212, 475, -74, -566, 94, 511, 112, -577, -201, 408, 217, -546, -295, 338,
>-  387, -13, 4, -46, 2, -76, 103, -83, 108, -55, 100, -150, 131, -156, 141,
>-  -171, 179, -190, 128, -227, 172, -214, 215, -189, 265, -244, 322, -335,
>-  337, -352, 358, -368, 362, -355, 366, -381, 403, -395, 411, -392, 446,
>-  -458, 504, -449, 507, -464, 452, -491, 481, -534, 486, -516, 560, -535,
>-  525, -537, 559, -554, 570, -616, 591, -585, 627, -509, 588, -584, 547,
>-  -610, 580, -614, 635, -620, 655, -554, 546, -591, 642, -590, 660, -656,
>-  629, -604, 620, -580, 617, -645, 648, -573, 612, -604, 584, -571, 597,
>-  -562, 627, -550, 560, -606, 529, -584, 568, -503, 532, -463, 512, -440,
>-  399, -457, 437, -349, 278, -317, 257, -220, 163, -8, -61, 18, -161, 367,
>-  -1306};
>+    20342, 23889, -10063, -9419, 3242,  7280,  -2012, -5029, 332,   4478,
>+    -97,   -3244, -891,   3117,  773,   -2204, -1335, 2009,  1236,  -1469,
>+    -1562, 1277,  1366,   -815,  -1619, 599,   1449,  -177,  -1507, 116,
>+    1294,  263,   -1338,  -244,  1059,  553,   -1045, -549,  829,   826,
>+    -731,  -755,  516,    909,   -427,  -853,  189,   1004,  -184,  -828,
>+    -108,  888,   72,     -700,  -280,  717,   342,   -611,  -534,  601,
>+    534,   -374,  -646,   399,   567,   -171,  -720,  234,   645,   -11,
>+    -712,  -26,   593,    215,   -643,  -172,  536,   361,   -527,  -403,
>+    388,   550,   -361,   -480,  208,   623,   -206,  -585,  41,    578,
>+    12,    -504,  -182,   583,   218,   -437,  -339,  499,   263,   -354,
>+    -450,  347,   456,    -193,  -524,  212,   475,   -74,   -566,  94,
>+    511,   112,   -577,   -201,  408,   217,   -546,  -295,  338,   387,
>+    -13,   4,     -46,    2,     -76,   103,   -83,   108,   -55,   100,
>+    -150,  131,   -156,   141,   -171,  179,   -190,  128,   -227,  172,
>+    -214,  215,   -189,   265,   -244,  322,   -335,  337,   -352,  358,
>+    -368,  362,   -355,   366,   -381,  403,   -395,  411,   -392,  446,
>+    -458,  504,   -449,   507,   -464,  452,   -491,  481,   -534,  486,
>+    -516,  560,   -535,   525,   -537,  559,   -554,  570,   -616,  591,
>+    -585,  627,   -509,   588,   -584,  547,   -610,  580,   -614,  635,
>+    -620,  655,   -554,   546,   -591,  642,   -590,  660,   -656,  629,
>+    -604,  620,   -580,   617,   -645,  648,   -573,  612,   -604,  584,
>+    -571,  597,   -562,   627,   -550,  560,   -606,  529,   -584,  568,
>+    -503,  532,   -463,   512,   -440,  399,   -457,  437,   -349,  278,
>+    -317,  257,   -220,   163,   -8,    -61,   18,    -161,  367,   -1306};
> static const int16_t time2spec_out_expected_2[kSamples] = {
>-  14283, -11552, -15335, 6626,
>-  7554, -2150, -6309, 1307, 4523, -4, -3908, -314, 3001, 914, -2715, -1042,
>-  2094, 1272, -1715, -1399, 1263, 1508, -1021, -1534, 735, 1595, -439, -1447,
>-  155, 1433, 22, -1325, -268, 1205, 424, -1030, -608, 950, 643, -733, -787,
>-  661, 861, -502, -888, 331, 852, -144, -849, 19, 833, 99, -826, -154,
>-  771, 368, -735, -459, 645, 513, -491, -604, 431, 630, -314, -598, 183,
>-  622, -78, -612, -48, 641, 154, -645, -257, 610, 281, -529, -444, 450,
>-  441, -327, -506, 274, 476, -232, -570, 117, 554, -86, -531, -21, 572,
>-  151, -606, -221, 496, 322, -407, -388, 407, 394, -268, -428, 280, 505,
>-  -115, -588, 19, 513, -29, -539, -109, 468, 173, -501, -242, 442, 278,
>-  -478, -680, 656, -659, 656, -669, 602, -688, 612, -667, 612, -642, 627,
>-  -648, 653, -676, 596, -680, 655, -649, 678, -672, 587, -608, 637, -645,
>-  637, -620, 556, -580, 553, -635, 518, -599, 583, -501, 536, -544, 473,
>-  -552, 583, -511, 541, -532, 563, -486, 461, -453, 486, -388, 424, -416,
>-  432, -374, 399, -462, 364, -346, 293, -329, 331, -313, 281, -247, 309,
>-  -337, 241, -190, 207, -194, 179, -163, 155, -156, 117, -135, 107, -126,
>-  29, -22, 81, -8, 17, -61, -10, 8, -37, 80, -44, 72, -88, 65, -89, 130,
>-  -114, 181, -215, 189, -245, 260, -288, 294, -339, 344, -396, 407, -429,
>-  438, -439, 485, -556, 629, -612, 637, -645, 661, -737, 829, -830, 831,
>-  -1041};
>+    14283, -11552, -15335, 6626,  7554,  -2150, -6309, 1307,  4523,  -4,
>+    -3908, -314,   3001,   914,   -2715, -1042, 2094,  1272,  -1715, -1399,
>+    1263,  1508,   -1021,  -1534, 735,   1595,  -439,  -1447, 155,   1433,
>+    22,    -1325,  -268,   1205,  424,   -1030, -608,  950,   643,   -733,
>+    -787,  661,    861,    -502,  -888,  331,   852,   -144,  -849,  19,
>+    833,   99,     -826,   -154,  771,   368,   -735,  -459,  645,   513,
>+    -491,  -604,   431,    630,   -314,  -598,  183,   622,   -78,   -612,
>+    -48,   641,    154,    -645,  -257,  610,   281,   -529,  -444,  450,
>+    441,   -327,   -506,   274,   476,   -232,  -570,  117,   554,   -86,
>+    -531,  -21,    572,    151,   -606,  -221,  496,   322,   -407,  -388,
>+    407,   394,    -268,   -428,  280,   505,   -115,  -588,  19,    513,
>+    -29,   -539,   -109,   468,   173,   -501,  -242,  442,   278,   -478,
>+    -680,  656,    -659,   656,   -669,  602,   -688,  612,   -667,  612,
>+    -642,  627,    -648,   653,   -676,  596,   -680,  655,   -649,  678,
>+    -672,  587,    -608,   637,   -645,  637,   -620,  556,   -580,  553,
>+    -635,  518,    -599,   583,   -501,  536,   -544,  473,   -552,  583,
>+    -511,  541,    -532,   563,   -486,  461,   -453,  486,   -388,  424,
>+    -416,  432,    -374,   399,   -462,  364,   -346,  293,   -329,  331,
>+    -313,  281,    -247,   309,   -337,  241,   -190,  207,   -194,  179,
>+    -163,  155,    -156,   117,   -135,  107,   -126,  29,    -22,   81,
>+    -8,    17,     -61,    -10,   8,     -37,   80,    -44,   72,    -88,
>+    65,    -89,    130,    -114,  181,   -215,  189,   -245,  260,   -288,
>+    294,   -339,   344,    -396,  407,   -429,  438,   -439,  485,   -556,
>+    629,   -612,   637,    -645,  661,   -737,  829,   -830,  831,   -1041};
> 
> class TransformTest : public testing::Test {
>  protected:
>-   TransformTest() {
>-     WebRtcSpl_Init();
>-   }
>+  TransformTest() { WebRtcSpl_Init(); }
> 
>-   // Pass a function pointer to the Tester function.
>-   void Time2SpecTester(Time2Spec Time2SpecFunction) {
>-     // WebRtcIsacfix_Time2Spec functions hard coded the buffer lengths. It's a
>-     // large buffer but we have to test it here.
>-     int16_t data_in_1[kSamples] = {0};
>-     int16_t data_in_2[kSamples] = {0};
>-     int16_t data_out_1[kSamples] = {0};
>-     int16_t data_out_2[kSamples] = {0};
>+  // Pass a function pointer to the Tester function.
>+  void Time2SpecTester(Time2Spec Time2SpecFunction) {
>+    // WebRtcIsacfix_Time2Spec functions hard coded the buffer lengths. It's a
>+    // large buffer but we have to test it here.
>+    int16_t data_in_1[kSamples] = {0};
>+    int16_t data_in_2[kSamples] = {0};
>+    int16_t data_out_1[kSamples] = {0};
>+    int16_t data_out_2[kSamples] = {0};
> 
>-     for(int i = 0; i < kSamples; i++) {
>-       data_in_1[i] = i * i + 1777;
>-       data_in_2[i] = WEBRTC_SPL_WORD16_MAX / (i + 1) + 17;
>-     }
>+    for (int i = 0; i < kSamples; i++) {
>+      data_in_1[i] = i * i + 1777;
>+      data_in_2[i] = WEBRTC_SPL_WORD16_MAX / (i + 1) + 17;
>+    }
> 
>-     Time2SpecFunction(data_in_1, data_in_2, data_out_1, data_out_2);
>+    Time2SpecFunction(data_in_1, data_in_2, data_out_1, data_out_2);
> 
>-     for (int i = 0; i < kSamples; i++) {
>-       // We don't require bit-exact for ARM assembly code.
>-       EXPECT_LE(abs(time2spec_out_expected_1[i] - data_out_1[i]), 1);
>-       EXPECT_LE(abs(time2spec_out_expected_2[i] - data_out_2[i]), 1);
>-     }
>-   }
>+    for (int i = 0; i < kSamples; i++) {
>+      // We don't require bit-exact for ARM assembly code.
>+      EXPECT_LE(abs(time2spec_out_expected_1[i] - data_out_1[i]), 1);
>+      EXPECT_LE(abs(time2spec_out_expected_2[i] - data_out_2[i]), 1);
>+    }
>+  }
> 
>   // Pass a function pointer to the Tester function.
>   void Spec2TimeTester(Spec2Time Spec2TimeFunction) {
>@@ -162,7 +171,7 @@ class TransformTest : public testing::Test {
>     int16_t data_in_2[kSamples] = {0};
>     int32_t data_out_1[kSamples] = {0};
>     int32_t data_out_2[kSamples] = {0};
>-    for(int i = 0; i < kSamples; i++) {
>+    for (int i = 0; i < kSamples; i++) {
>       data_in_1[i] = i * i + 1777;
>       data_in_2[i] = WEBRTC_SPL_WORD16_MAX / (i + 1) + 17;
>     }
>@@ -175,7 +184,6 @@ class TransformTest : public testing::Test {
>       EXPECT_LE(abs(spec2time_out_expected_2[i] - data_out_2[i]), 16);
>     }
>   }
>-
> };
> 
> TEST_F(TransformTest, Time2SpecTest) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc
>index fc779d81850..aeca2e8e319 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/test/isac_speed_test.cc
>@@ -25,19 +25,21 @@ class IsacSpeedTest : public AudioCodecSpeedTest {
>   IsacSpeedTest();
>   void SetUp() override;
>   void TearDown() override;
>-  float EncodeABlock(int16_t* in_data, uint8_t* bit_stream,
>-                     size_t max_bytes, size_t* encoded_bytes) override;
>-  float DecodeABlock(const uint8_t* bit_stream, size_t encoded_bytes,
>+  float EncodeABlock(int16_t* in_data,
>+                     uint8_t* bit_stream,
>+                     size_t max_bytes,
>+                     size_t* encoded_bytes) override;
>+  float DecodeABlock(const uint8_t* bit_stream,
>+                     size_t encoded_bytes,
>                      int16_t* out_data) override;
>-  ISACFIX_MainStruct *ISACFIX_main_inst_;
>+  ISACFIX_MainStruct* ISACFIX_main_inst_;
> };
> 
> IsacSpeedTest::IsacSpeedTest()
>     : AudioCodecSpeedTest(kIsacBlockDurationMs,
>                           kIsacInputSamplingKhz,
>                           kIsacOutputSamplingKhz),
>-      ISACFIX_main_inst_(NULL) {
>-}
>+      ISACFIX_main_inst_(NULL) {}
> 
> void IsacSpeedTest::SetUp() {
>   AudioCodecSpeedTest::SetUp();
>@@ -60,8 +62,10 @@ void IsacSpeedTest::TearDown() {
>   EXPECT_EQ(0, WebRtcIsacfix_Free(ISACFIX_main_inst_));
> }
> 
>-float IsacSpeedTest::EncodeABlock(int16_t* in_data, uint8_t* bit_stream,
>-                                  size_t max_bytes, size_t* encoded_bytes) {
>+float IsacSpeedTest::EncodeABlock(int16_t* in_data,
>+                                  uint8_t* bit_stream,
>+                                  size_t max_bytes,
>+                                  size_t* encoded_bytes) {
>   // ISAC takes 10 ms everycall
>   const int subblocks = block_duration_ms_ / 10;
>   const int subblock_length = 10 * input_sampling_khz_;
>@@ -70,8 +74,8 @@ float IsacSpeedTest::EncodeABlock(int16_t* in_data, uint8_t* bit_stream,
>   clock_t clocks = clock();
>   size_t pointer = 0;
>   for (int idx = 0; idx < subblocks; idx++, pointer += subblock_length) {
>-    value = WebRtcIsacfix_Encode(ISACFIX_main_inst_, &in_data[pointer],
>-                                 bit_stream);
>+    value =
>+        WebRtcIsacfix_Encode(ISACFIX_main_inst_, &in_data[pointer], bit_stream);
>     if (idx == subblocks - 1)
>       EXPECT_GT(value, 0);
>     else
>@@ -108,7 +112,6 @@ const coding_param param_set[] = {
>                     string("pcm"),
>                     true)};
> 
>-INSTANTIATE_TEST_CASE_P(AllTest, IsacSpeedTest,
>-                        ::testing::ValuesIn(param_set));
>+INSTANTIATE_TEST_CASE_P(AllTest, IsacSpeedTest, ::testing::ValuesIn(param_set));
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/test/kenny.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/test/kenny.cc
>index b11c6702cec..fb64a2b597a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/test/kenny.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/fix/test/kenny.cc
>@@ -8,11 +8,11 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>+#include <ctype.h>
> #include <stdio.h>
> #include <stdlib.h>
> #include <string.h>
> #include <time.h>
>-#include <ctype.h>
> 
> #include "modules/audio_coding/codecs/isac/fix/include/isacfix.h"
> #include "test/gtest.h"
>@@ -22,14 +22,16 @@
> // separate encoder and decoder.
> 
> /* Defines */
>-#define SEED_FILE "randseed.txt"  /* Used when running decoder on garbage data */
>-#define MAX_FRAMESAMPLES    960   /* max number of samples per frame (= 60 ms frame) */
>-#define FRAMESAMPLES_10ms 160   /* number of samples per 10ms frame */
>-#define FS           16000 /* sampling frequency (Hz) */
>+#define SEED_FILE                                             \
>+  "randseed.txt" /* Used when running decoder on garbage data \
>+                  */
>+#define MAX_FRAMESAMPLES \
>+  960 /* max number of samples per frame (= 60 ms frame) */
>+#define FRAMESAMPLES_10ms 160 /* number of samples per 10ms frame */
>+#define FS 16000              /* sampling frequency (Hz) */
> 
> /* Function for reading audio data from PCM file */
>-int readframe(int16_t *data, FILE *inp, int length) {
>-
>+int readframe(int16_t* data, FILE* inp, int length) {
>   short k, rlen, status = 0;
> 
>   rlen = fread(data, sizeof(int16_t), length, inp);
>@@ -45,25 +47,24 @@ int readframe(int16_t *data, FILE *inp, int length) {
> // Globals needed because gtest does not provide access to argv.
> // This should be reworked to use flags.
> static int global_argc;
>-static char **global_argv;
>+static char** global_argv;
> 
> /* Struct for bottleneck model */
> typedef struct {
>-  uint32_t send_time;            /* samples */
>-  uint32_t arrival_time;         /* samples */
>-  uint32_t sample_count;         /* samples */
>+  uint32_t send_time;    /* samples */
>+  uint32_t arrival_time; /* samples */
>+  uint32_t sample_count; /* samples */
>   uint16_t rtp_number;
> } BottleNeckModel;
> 
>-void get_arrival_time(int current_framesamples,   /* samples */
>-                      size_t packet_size,         /* bytes */
>-                      int bottleneck,             /* excluding headers; bits/s */
>-                      BottleNeckModel *BN_data)
>-{
>+void get_arrival_time(int current_framesamples, /* samples */
>+                      size_t packet_size,       /* bytes */
>+                      int bottleneck,           /* excluding headers; bits/s */
>+                      BottleNeckModel* BN_data) {
>   const int HeaderSize = 35;
>   int HeaderRate;
> 
>-  HeaderRate = HeaderSize * 8 * FS / current_framesamples;     /* bits/s */
>+  HeaderRate = HeaderSize * 8 * FS / current_framesamples; /* bits/s */
> 
>   /* everything in samples */
>   BN_data->sample_count = BN_data->sample_count + current_framesamples;
>@@ -80,37 +81,37 @@ void get_arrival_time(int current_framesamples,   /* samples */
> 
> void get_arrival_time2(int current_framesamples,
>                        int current_delay,
>-                       BottleNeckModel *BN_data)
>-{
>+                       BottleNeckModel* BN_data) {
>   if (current_delay == -1)
>-    //dropped packet
>+  // dropped packet
>   {
>     BN_data->arrival_time += current_framesamples;
>-  }
>-  else if (current_delay != -2)
>-  {
>+  } else if (current_delay != -2) {
>     //
>-    BN_data->arrival_time += (current_framesamples + ((FS/1000) * current_delay));
>+    BN_data->arrival_time +=
>+        (current_framesamples + ((FS / 1000) * current_delay));
>   }
>-  //else
>-  //current packet has same timestamp as previous packet
>+  // else
>+  // current packet has same timestamp as previous packet
> 
>   BN_data->rtp_number++;
> }
> 
> TEST(IsacFixTest, Kenny) {
>   int argc = global_argc;
>-  char **argv = global_argv;
>+  char** argv = global_argv;
> 
>-  char inname[100], outname[100],  outbitsname[100], bottleneck_file[100];
>+  char inname[100], outname[100], outbitsname[100], bottleneck_file[100];
>   FILE *inp, *outp, *f_bn, *outbits;
>   int endfile;
> 
>-  size_t i;
>+  const char* chartjson_result_file = NULL;
>+
>+  int i;
>   int errtype, h = 0, k, packetLossPercent = 0;
>   int16_t CodingMode;
>   int16_t bottleneck;
>-  int framesize = 30;           /* ms */
>+  int framesize = 30; /* ms */
>   int cur_framesmpls, err = 0, lostPackets = 0;
> 
>   /* Runtime statistics */
>@@ -131,16 +132,16 @@ TEST(IsacFixTest, Kenny) {
>   int32_t payloadRate = 0;
>   int setControlBWE = 0;
>   int readLoss;
>-  FILE  *plFile = NULL;
>+  FILE* plFile = NULL;
> 
>   char version_number[20];
>   char tmpBit[5] = ".bit";
> 
>-  int totalbits =0;
>-  int totalsmpls =0;
>+  int totalbits = 0;
>+  int totalsmpls = 0;
>   int16_t testNum, testCE;
> 
>-  FILE *fp_gns = NULL;
>+  FILE* fp_gns = NULL;
>   int gns = 0;
>   int cur_delay = 0;
>   char gns_file[100];
>@@ -149,20 +150,20 @@ TEST(IsacFixTest, Kenny) {
>   int16_t lostFrame;
>   float scale = (float)0.7;
>   /* only one structure used for ISAC encoder */
>-  ISACFIX_MainStruct *ISAC_main_inst = NULL;
>+  ISACFIX_MainStruct* ISAC_main_inst = NULL;
> 
>   /* For fault test 10, garbage data */
>-  FILE *seedfile;
>-  unsigned int random_seed = (unsigned int) time(NULL);//1196764538
>+  FILE* seedfile;
>+  unsigned int random_seed = (unsigned int)time(NULL);  // 1196764538
> 
>-  BottleNeckModel       BN_data;
>-  f_bn  = NULL;
>+  BottleNeckModel BN_data;
>+  f_bn = NULL;
> 
>   readLoss = 0;
>   packetLossPercent = 0;
> 
>   /* Handling wrong input arguments in the command line */
>-  if ((argc<3) || (argc>21))  {
>+  if ((argc < 3) || (argc > 22)) {
>     printf("\n\nWrong number of arguments or flag values.\n\n");
> 
>     printf("\n");
>@@ -170,7 +171,7 @@ TEST(IsacFixTest, Kenny) {
>     printf("iSAC version %s \n\n", version_number);
> 
>     printf("Usage:\n\n");
>-    printf("%s [-F num][-I] bottleneck_value infile outfile \n\n", argv[0]);
>+    printf("%s [-I] bottleneck_value infile outfile [-F num]\n\n", argv[0]);
>     printf("with:\n");
>     printf("[-I]             :if -I option is specified, the coder will use\n");
>     printf("                  an instantaneous Bottleneck value. If not, it\n");
>@@ -178,60 +179,77 @@ TEST(IsacFixTest, Kenny) {
>     printf("bottleneck_value :the value of the bottleneck provided either\n");
>     printf("                  as a fixed value (e.g. 25000) or\n");
>     printf("                  read from a file (e.g. bottleneck.txt)\n\n");
>-    printf("[-INITRATE num]  :Set a new value for initial rate. Note! Only used"
>-           " in adaptive mode.\n\n");
>-    printf("[-FL num]        :Set (initial) frame length in msec. Valid length"
>-           " are 30 and 60 msec.\n\n");
>+    printf("infile           :Normal speech input file\n\n");
>+    printf("outfile          :Speech output file\n\n");
>+    printf(
>+        "[-INITRATE num]  :Set a new value for initial rate. Note! Only used"
>+        " in adaptive mode.\n\n");
>+    printf(
>+        "[-FL num]        :Set (initial) frame length in msec. Valid length"
>+        " are 30 and 60 msec.\n\n");
>     printf("[-FIXED_FL]      :Frame length to be fixed to initial value.\n\n");
>-    printf("[-MAX num]       :Set the limit for the payload size of iSAC"
>-           " in bytes. \n");
>+    printf(
>+        "[-MAX num]       :Set the limit for the payload size of iSAC"
>+        " in bytes. \n");
>     printf("                  Minimum 100, maximum 400.\n\n");
>     printf("[-MAXRATE num]   :Set the maxrate for iSAC in bits per second. \n");
>     printf("                  Minimum 32000, maximum 53400.\n\n");
>     printf("[-F num]         :if -F option is specified, the test function\n");
>-    printf("                  will run the iSAC API fault scenario specified"
>-           " by the\n");
>+    printf(
>+        "                  will run the iSAC API fault scenario specified"
>+        " by the\n");
>     printf("                  supplied number.\n");
>     printf("                  F 1 - Call encoder prior to init encoder call\n");
>     printf("                  F 2 - Call decoder prior to init decoder call\n");
>     printf("                  F 3 - Call decoder prior to encoder call\n");
>-    printf("                  F 4 - Call decoder with a too short coded"
>-           " sequence\n");
>-    printf("                  F 5 - Call decoder with a too long coded"
>-           " sequence\n");
>+    printf(
>+        "                  F 4 - Call decoder with a too short coded"
>+        " sequence\n");
>+    printf(
>+        "                  F 5 - Call decoder with a too long coded"
>+        " sequence\n");
>     printf("                  F 6 - Call decoder with random bit stream\n");
>-    printf("                  F 7 - Call init encoder/decoder at random"
>-           " during a call\n");
>-    printf("                  F 8 - Call encoder/decoder without having"
>-           " allocated memory for \n");
>+    printf(
>+        "                  F 7 - Call init encoder/decoder at random"
>+        " during a call\n");
>+    printf(
>+        "                  F 8 - Call encoder/decoder without having"
>+        " allocated memory for \n");
>     printf("                        encoder/decoder instance\n");
>     printf("                  F 9 - Call decodeB without calling decodeA\n");
>     printf("                  F 10 - Call decodeB with garbage data\n");
>-    printf("[-PL num]       : if -PL option is specified 0<num<100 will "
>-           "specify the\n");
>+    printf(
>+        "[-PL num]        :if -PL option is specified 0<num<100 will "
>+        "specify the\n");
>     printf("                  percentage of packet loss\n\n");
>-    printf("[-G file]       : if -G option is specified the file given is"
>-           " a .gns file\n");
>+    printf(
>+        "[-G file]        :if -G option is specified the file given is"
>+        " a .gns file\n");
>     printf("                  that represents a network profile\n\n");
>-    printf("[-NB num]       : if -NB option, use the narrowband interfaces\n");
>-    printf("                  num=1 => encode with narrowband encoder"
>-           " (infile is narrowband)\n");
>-    printf("                  num=2 => decode with narrowband decoder"
>-           " (outfile is narrowband)\n\n");
>-    printf("[-CE num]       : Test of APIs used by Conference Engine.\n");
>-    printf("                  CE 1 - createInternal, freeInternal,"
>-           " getNewBitstream \n");
>+    printf("[-NB num]        :if -NB option, use the narrowband interfaces\n");
>+    printf(
>+        "                  num=1 => encode with narrowband encoder"
>+        " (infile is narrowband)\n");
>+    printf(
>+        "                  num=2 => decode with narrowband decoder"
>+        " (outfile is narrowband)\n\n");
>+    printf("[-CE num]        :Test of APIs used by Conference Engine.\n");
>+    printf(
>+        "                  CE 1 - createInternal, freeInternal,"
>+        " getNewBitstream \n");
>     printf("                  CE 2 - transcode, getBWE \n");
>     printf("                  CE 3 - getSendBWE, setSendBWE.  \n\n");
>-    printf("[-RTP_INIT num] : if -RTP_INIT option is specified num will be"
>-           " the initial\n");
>+    printf(
>+        "[-RTP_INIT num]  :if -RTP_INIT option is specified num will be"
>+        " the initial\n");
>     printf("                  value of the rtp sequence number.\n\n");
>-    printf("infile          : Normal speech input file\n\n");
>-    printf("outfile         : Speech output file\n\n");
>-    printf("Example usage   : \n\n");
>+    printf("[--isolated-script-test-perf-output=file]\n");
>+    printf(
>+        "                 :If this option is specified, perf values will be"
>+        " written to this file in a JSON format.\n\n");
>+    printf("Example usage    :\n\n");
>     printf("%s -I bottleneck.txt speechIn.pcm speechOut.pcm\n\n", argv[0]);
>     exit(1);
>-
>   }
> 
>   /* Print version number */
>@@ -242,21 +260,36 @@ TEST(IsacFixTest, Kenny) {
>   CodingMode = 0;
>   testNum = 0;
>   testCE = 0;
>-  for (i = 1; i + 2 < static_cast<size_t>(argc); i++) {
>-    /* Instantaneous mode */
>-    if (!strcmp ("-I", argv[i])) {
>-      printf("\nInstantaneous BottleNeck\n");
>-      CodingMode = 1;
>-      i++;
>-    }
>+  i = 1;
>+
>+  /* Instantaneous mode */
>+  if (!strcmp("-I", argv[i])) {
>+    printf("\nInstantaneous BottleNeck\n");
>+    CodingMode = 1;
>+    i++;
>+  }
>+
>+  /* Bottleneck value is processed after the for */
>+  i++;
> 
>+  /* Get Input and Output files */
>+  sscanf(argv[i++], "%s", inname);
>+  sscanf(argv[i++], "%s", outname);
>+
>+  for (; i < argc; i++) {
>     /* Set (initial) bottleneck value */
>-    if (!strcmp ("-INITRATE", argv[i])) {
>+    if (!strcmp("-INITRATE", argv[i])) {
>+      if (i + 1 >= argc) {
>+        printf("-INITRATE requires a parameter.\n");
>+        exit(1);
>+      }
>       rateBPS = atoi(argv[i + 1]);
>       setControlBWE = 1;
>       if ((rateBPS < 10000) || (rateBPS > 32000)) {
>-        printf("\n%d is not a initial rate. "
>-               "Valid values are in the range 10000 to 32000.\n", rateBPS);
>+        printf(
>+            "\n%d is not a initial rate. "
>+            "Valid values are in the range 10000 to 32000.\n",
>+            rateBPS);
>         exit(1);
>       }
>       printf("\nNew initial rate: %d\n", rateBPS);
>@@ -264,11 +297,17 @@ TEST(IsacFixTest, Kenny) {
>     }
> 
>     /* Set (initial) framelength */
>-    if (!strcmp ("-FL", argv[i])) {
>+    if (!strcmp("-FL", argv[i])) {
>+      if (i + 1 >= argc) {
>+        printf("-FL requires a parameter.\n");
>+        exit(1);
>+      }
>       framesize = atoi(argv[i + 1]);
>       if ((framesize != 30) && (framesize != 60)) {
>-        printf("\n%d is not a valid frame length. "
>-               "Valid length are 30 and 60 msec.\n", framesize);
>+        printf(
>+            "\n%d is not a valid frame length. "
>+            "Valid length are 30 and 60 msec.\n",
>+            framesize);
>         exit(1);
>       }
>       printf("\nFrame Length: %d\n", framesize);
>@@ -276,75 +315,99 @@ TEST(IsacFixTest, Kenny) {
>     }
> 
>     /* Fixed frame length */
>-    if (!strcmp ("-FIXED_FL", argv[i])) {
>+    if (!strcmp("-FIXED_FL", argv[i])) {
>       fixedFL = 1;
>       setControlBWE = 1;
>     }
> 
>     /* Set maximum allowed payload size in bytes */
>-    if (!strcmp ("-MAX", argv[i])) {
>+    if (!strcmp("-MAX", argv[i])) {
>+      if (i + 1 >= argc) {
>+        printf("-MAX requires a parameter.\n");
>+        exit(1);
>+      }
>       payloadSize = atoi(argv[i + 1]);
>       printf("Maximum Payload Size: %d\n", payloadSize);
>       i++;
>     }
> 
>     /* Set maximum rate in bytes */
>-    if (!strcmp ("-MAXRATE", argv[i])) {
>+    if (!strcmp("-MAXRATE", argv[i])) {
>+      if (i + 1 >= argc) {
>+        printf("-MAXRATE requires a parameter.\n");
>+        exit(1);
>+      }
>       payloadRate = atoi(argv[i + 1]);
>       printf("Maximum Rate in kbps: %d\n", payloadRate);
>       i++;
>     }
> 
>     /* Test of fault scenarious */
>-    if (!strcmp ("-F", argv[i])) {
>+    if (!strcmp("-F", argv[i])) {
>+      if (i + 1 >= argc) {
>+        printf("-F requires a parameter.");
>+        exit(1);
>+      }
>       testNum = atoi(argv[i + 1]);
>       printf("\nFault test: %d\n", testNum);
>       if (testNum < 1 || testNum > 10) {
>-        printf("\n%d is not a valid Fault Scenario number."
>-               " Valid Fault Scenarios are numbered 1-10.\n", testNum);
>+        printf(
>+            "\n%d is not a valid Fault Scenario number."
>+            " Valid Fault Scenarios are numbered 1-10.\n",
>+            testNum);
>         exit(1);
>       }
>       i++;
>     }
> 
>     /* Packet loss test */
>-    if (!strcmp ("-PL", argv[i])) {
>-      if( isdigit( *argv[i+1] ) ) {
>-        packetLossPercent = atoi( argv[i+1] );
>-        if( (packetLossPercent < 0) | (packetLossPercent > 100) ) {
>-          printf( "\nInvalid packet loss perentage \n" );
>-          exit( 1 );
>+    if (!strcmp("-PL", argv[i])) {
>+      if (i + 1 >= argc) {
>+        printf("-PL requires a parameter.\n");
>+        exit(1);
>+      }
>+      if (isdigit(*argv[i + 1])) {
>+        packetLossPercent = atoi(argv[i + 1]);
>+        if ((packetLossPercent < 0) | (packetLossPercent > 100)) {
>+          printf("\nInvalid packet loss perentage \n");
>+          exit(1);
>         }
>-        if( packetLossPercent > 0 ) {
>-          printf( "\nSimulating %d %% of independent packet loss\n",
>-                  packetLossPercent );
>+        if (packetLossPercent > 0) {
>+          printf("\nSimulating %d %% of independent packet loss\n",
>+                 packetLossPercent);
>         } else {
>-          printf( "\nNo Packet Loss Is Simulated \n" );
>+          printf("\nNo Packet Loss Is Simulated \n");
>         }
>         readLoss = 0;
>       } else {
>         readLoss = 1;
>-        plFile = fopen( argv[i+1], "rb" );
>-        if( plFile == NULL ) {
>-          FAIL() << "Couldn't open the frameloss file: " << argv[i+1];
>+        plFile = fopen(argv[i + 1], "rb");
>+        if (plFile == NULL) {
>+          FAIL() << "Couldn't open the frameloss file: " << argv[i + 1];
>         }
>-        printf( "\nSimulating packet loss through the given "
>-                "channel file: %s\n", argv[i+1] );
>+        printf(
>+            "\nSimulating packet loss through the given "
>+            "channel file: %s\n",
>+            argv[i + 1]);
>       }
>       i++;
>     }
> 
>     /* Random packetlosses */
>-    if (!strcmp ("-rnd", argv[i])) {
>-      srand(time(NULL) );
>-      printf( "\n Random pattern in lossed packets \n" );
>+    if (!strcmp("-rnd", argv[i])) {
>+      srand(time(NULL));
>+      printf("\n Random pattern in lossed packets \n");
>     }
> 
>     /* Use gns file */
>-    if (!strcmp ("-G", argv[i])) {
>+    if (!strcmp("-G", argv[i])) {
>+      if (i + 1 >= argc) {
>+        printf("-G requires a parameter.\n");
>+        exit(1);
>+      }
>       sscanf(argv[i + 1], "%s", gns_file);
>       fp_gns = fopen(gns_file, "rb");
>-      if (fp_gns  == NULL) {
>+      if (fp_gns == NULL) {
>         FAIL() << "Cannot read file " << gns_file << ".";
>       }
>       gns = 1;
>@@ -352,43 +415,66 @@ TEST(IsacFixTest, Kenny) {
>     }
> 
>     /* Run Narrowband interfaces (either encoder or decoder) */
>-    if (!strcmp ("-NB", argv[i])) {
>+    if (!strcmp("-NB", argv[i])) {
>+      if (i + 1 >= argc) {
>+        printf("-NB requires a parameter.\n");
>+        exit(1);
>+      }
>       nbTest = atoi(argv[i + 1]);
>       i++;
>     }
> 
>     /* Run Conference Engine APIs */
>-    if (!strcmp ("-CE", argv[i])) {
>+    if (!strcmp("-CE", argv[i])) {
>+      if (i + 1 >= argc) {
>+        printf("-CE requires a parameter.\n");
>+        exit(1);
>+      }
>       testCE = atoi(argv[i + 1]);
>-      if (testCE==1 || testCE==2) {
>+      if (testCE == 1 || testCE == 2) {
>         i++;
>-        scale = (float)atof( argv[i+1] );
>+        scale = (float)atof(argv[i + 1]);
>       } else if (testCE < 1 || testCE > 3) {
>-        printf("\n%d is not a valid CE-test number, valid Fault "
>-               "Scenarios are numbered 1-3\n", testCE);
>+        printf(
>+            "\n%d is not a valid CE-test number, valid Fault "
>+            "Scenarios are numbered 1-3\n",
>+            testCE);
>         exit(1);
>       }
>       i++;
>     }
> 
>     /* Set initial RTP number */
>-    if (!strcmp ("-RTP_INIT", argv[i])) {
>+    if (!strcmp("-RTP_INIT", argv[i])) {
>+      if (i + 1 >= argc) {
>+        printf("-RTP_INIT requires a parameter.\n");
>+        exit(1);
>+      }
>       i++;
>     }
>+
>+    if (strstr(argv[i], "--isolated-script-test-perf-output") == argv[i]) {
>+      const char* filename_start = strstr(argv[i], "=");
>+      if (!filename_start || strlen(filename_start) < 2) {
>+        printf("Expected --isolated-script-test-perf-output=/some/filename\n");
>+        exit(1);
>+      }
>+      chartjson_result_file = filename_start + 1;
>+    }
>   }
> 
>   /* Get Bottleneck value                                                   */
>   /* Gns files and bottleneck should not and can not be used simultaneously */
>-  bottleneck = atoi(argv[CodingMode+1]);
>+  bottleneck = atoi(argv[CodingMode + 1]);
>   if (bottleneck == 0 && gns == 0) {
>-    sscanf(argv[CodingMode+1], "%s", bottleneck_file);
>+    sscanf(argv[CodingMode + 1], "%s", bottleneck_file);
>     f_bn = fopen(bottleneck_file, "rb");
>-    if (f_bn  == NULL) {
>+    if (f_bn == NULL) {
>       printf("No value provided for BottleNeck\n");
>       FAIL() << "Cannot read file " << bottleneck_file;
>     } else {
>       int aux_var;
>-      printf("reading bottleneck rates from file %s\n\n",bottleneck_file);
>+      printf("reading bottleneck rates from file %s\n\n", bottleneck_file);
>       if (fscanf(f_bn, "%d", &aux_var) == EOF) {
>         /* Set pointer to beginning of file */
>         fseek(f_bn, 0L, SEEK_SET);
>@@ -413,27 +499,23 @@ TEST(IsacFixTest, Kenny) {
>     printf("\nAdaptive BottleNeck\n");
>   }
> 
>-  /* Get Input and Output files */
>-  sscanf(argv[argc-2], "%s", inname);
>-  sscanf(argv[argc-1], "%s", outname);
>-
>   /* Add '.bit' to output bitstream file */
>   while ((int)outname[h] != 0) {
>     outbitsname[h] = outname[h];
>     h++;
>   }
>-  for (k=0; k<5; k++) {
>+  for (k = 0; k < 5; k++) {
>     outbitsname[h] = tmpBit[k];
>     h++;
>   }
>-  if ((inp = fopen(inname,"rb")) == NULL) {
>+  if ((inp = fopen(inname, "rb")) == NULL) {
>     FAIL() << "  iSAC: Cannot read file " << inname;
>   }
>-  if ((outp = fopen(outname,"wb")) == NULL) {
>+  if ((outp = fopen(outname, "wb")) == NULL) {
>     FAIL() << "  iSAC: Cannot write file " << outname;
>   }
> 
>-  if ((outbits = fopen(outbitsname,"wb")) == NULL) {
>+  if ((outbits = fopen(outbitsname, "wb")) == NULL) {
>     FAIL() << "  iSAC: Cannot write file " << outbitsname;
>   }
>   printf("\nInput:%s\nOutput:%s\n\n", inname, outname);
>@@ -443,30 +525,28 @@ TEST(IsacFixTest, Kenny) {
>     /* Test to run decoder with garbage data */
>     srand(random_seed);
> 
>-    if ( (seedfile = fopen(SEED_FILE, "a+t") ) == NULL ) {
>+    if ((seedfile = fopen(SEED_FILE, "a+t")) == NULL) {
>       printf("Error: Could not open file %s\n", SEED_FILE);
>-    }
>-    else {
>+    } else {
>       fprintf(seedfile, "%u\n", random_seed);
>       fclose(seedfile);
>     }
>   }
> 
>   /* Runtime statistics */
>-  starttime = clock()/(double)CLOCKS_PER_SEC;
>+  starttime = clock() / (double)CLOCKS_PER_SEC;
> 
>   /* Initialize the ISAC and BN structs */
>-  if (testNum != 8)
>-  {
>-    if(1){
>-      err =WebRtcIsacfix_Create(&ISAC_main_inst);
>-    }else{
>+  if (testNum != 8) {
>+    if (1) {
>+      err = WebRtcIsacfix_Create(&ISAC_main_inst);
>+    } else {
>       /* Test the Assign functions */
>       int sss;
>-      void *ppp;
>-      err =WebRtcIsacfix_AssignSize(&sss);
>-      ppp=malloc(sss);
>-      err =WebRtcIsacfix_Assign(&ISAC_main_inst,ppp);
>+      void* ppp;
>+      err = WebRtcIsacfix_AssignSize(&sss);
>+      ppp = malloc(sss);
>+      err = WebRtcIsacfix_Assign(&ISAC_main_inst, ppp);
>     }
>     /* Error check */
>     if (err < 0) {
>@@ -482,13 +562,13 @@ TEST(IsacFixTest, Kenny) {
>   }
> 
>   /* Init of bandwidth data */
>-  BN_data.send_time     = 0;
>-  BN_data.arrival_time  = 0;
>-  BN_data.sample_count  = 0;
>-  BN_data.rtp_number    = 0;
>+  BN_data.send_time = 0;
>+  BN_data.arrival_time = 0;
>+  BN_data.sample_count = 0;
>+  BN_data.rtp_number = 0;
> 
>   /* Initialize encoder and decoder */
>-  framecnt= 0;
>+  framecnt = 0;
>   endfile = 0;
>   if (testNum != 1) {
>     WebRtcIsacfix_EncoderInit(ISAC_main_inst, CodingMode);
>@@ -501,10 +581,10 @@ TEST(IsacFixTest, Kenny) {
>     err = WebRtcIsacfix_Control(ISAC_main_inst, bottleneck, framesize);
>     if (err < 0) {
>       /* exit if returned with error */
>-      errtype=WebRtcIsacfix_GetErrorCode(ISAC_main_inst);
>+      errtype = WebRtcIsacfix_GetErrorCode(ISAC_main_inst);
>       printf("\n\n Error in control: %d.\n\n", errtype);
>     }
>-  } else if(setControlBWE == 1) {
>+  } else if (setControlBWE == 1) {
>     err = WebRtcIsacfix_ControlBwe(ISAC_main_inst, rateBPS, framesize, fixedFL);
>   }
> 
>@@ -512,7 +592,7 @@ TEST(IsacFixTest, Kenny) {
>     err = WebRtcIsacfix_SetMaxPayloadSize(ISAC_main_inst, payloadSize);
>     if (err < 0) {
>       /* exit if returned with error */
>-      errtype=WebRtcIsacfix_GetErrorCode(ISAC_main_inst);
>+      errtype = WebRtcIsacfix_GetErrorCode(ISAC_main_inst);
>       FAIL() << "Error in SetMaxPayloadSize: " << errtype;
>     }
>   }
>@@ -520,35 +600,32 @@ TEST(IsacFixTest, Kenny) {
>     err = WebRtcIsacfix_SetMaxRate(ISAC_main_inst, payloadRate);
>     if (err < 0) {
>       /* exit if returned with error */
>-      errtype=WebRtcIsacfix_GetErrorCode(ISAC_main_inst);
>+      errtype = WebRtcIsacfix_GetErrorCode(ISAC_main_inst);
>       FAIL() << "Error in SetMaxRateInBytes: " << errtype;
>     }
>   }
> 
>   *speechType = 1;
> 
>-
>   while (endfile == 0) {
>-
>-    if(testNum == 7 && (rand()%2 == 0)) {
>+    if (testNum == 7 && (rand() % 2 == 0)) {
>       err = WebRtcIsacfix_EncoderInit(ISAC_main_inst, CodingMode);
>       /* Error check */
>       if (err < 0) {
>-        errtype=WebRtcIsacfix_GetErrorCode(ISAC_main_inst);
>+        errtype = WebRtcIsacfix_GetErrorCode(ISAC_main_inst);
>         printf("\n\n Error in encoderinit: %d.\n\n", errtype);
>       }
> 
>       WebRtcIsacfix_DecoderInit(ISAC_main_inst);
>     }
> 
>-
>     cur_framesmpls = 0;
>     while (1) {
>       /* Read 10 ms speech block */
>       if (nbTest != 1) {
>         endfile = readframe(shortdata, inp, FRAMESAMPLES_10ms);
>       } else {
>-        endfile = readframe(shortdata, inp, (FRAMESAMPLES_10ms/2));
>+        endfile = readframe(shortdata, inp, (FRAMESAMPLES_10ms / 2));
>       }
> 
>       if (testNum == 7) {
>@@ -561,22 +638,18 @@ TEST(IsacFixTest, Kenny) {
>           short bwe;
> 
>           /* Encode */
>-          stream_len_int = WebRtcIsacfix_Encode(ISAC_main_inst,
>-                                                shortdata,
>+          stream_len_int = WebRtcIsacfix_Encode(ISAC_main_inst, shortdata,
>                                                 (uint8_t*)streamdata);
> 
>           /* If packet is ready, and CE testing, call the different API
>              functions from the internal API. */
>-          if (stream_len_int>0) {
>+          if (stream_len_int > 0) {
>             if (testCE == 1) {
>               err = WebRtcIsacfix_ReadBwIndex(
>                   reinterpret_cast<const uint8_t*>(streamdata),
>-                  static_cast<size_t>(stream_len_int),
>-                  &bwe);
>+                  static_cast<size_t>(stream_len_int), &bwe);
>               stream_len_int = WebRtcIsacfix_GetNewBitStream(
>-                  ISAC_main_inst,
>-                  bwe,
>-                  scale,
>+                  ISAC_main_inst, bwe, scale,
>                   reinterpret_cast<uint8_t*>(streamdata));
>             } else if (testCE == 2) {
>               /* transcode function not supported */
>@@ -587,37 +660,33 @@ TEST(IsacFixTest, Kenny) {
>               err = WebRtcIsacfix_GetDownLinkBwIndex(ISAC_main_inst, &bwe);
>               /* Error Check */
>               if (err < 0) {
>-                errtype=WebRtcIsacfix_GetErrorCode(ISAC_main_inst);
>+                errtype = WebRtcIsacfix_GetErrorCode(ISAC_main_inst);
>                 printf("\nError in getSendBWE: %d.\n", errtype);
>               }
> 
>               err = WebRtcIsacfix_UpdateUplinkBw(ISAC_main_inst, bwe);
>               /* Error Check */
>               if (err < 0) {
>-                errtype=WebRtcIsacfix_GetErrorCode(ISAC_main_inst);
>+                errtype = WebRtcIsacfix_GetErrorCode(ISAC_main_inst);
>                 printf("\nError in setBWE: %d.\n", errtype);
>               }
>-
>             }
>           }
>         } else {
> #ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
>-          stream_len_int = WebRtcIsacfix_EncodeNb(ISAC_main_inst,
>-                                                  shortdata,
>-                                                  streamdata);
>+          stream_len_int =
>+              WebRtcIsacfix_EncodeNb(ISAC_main_inst, shortdata, streamdata);
> #else
>           stream_len_int = -1;
> #endif
>         }
>-      }
>-      else
>-      {
>+      } else {
>         break;
>       }
> 
>       if (stream_len_int < 0 || err < 0) {
>         /* exit if returned with error */
>-        errtype=WebRtcIsacfix_GetErrorCode(ISAC_main_inst);
>+        errtype = WebRtcIsacfix_GetErrorCode(ISAC_main_inst);
>         printf("\nError in encoder: %d.\n", errtype);
>       } else {
>         stream_len = static_cast<size_t>(stream_len_int);
>@@ -646,7 +715,8 @@ TEST(IsacFixTest, Kenny) {
>       }
> 
>       /* exit encoder loop if the encoder returned a bitstream */
>-      if (stream_len != 0) break;
>+      if (stream_len != 0)
>+        break;
>     }
> 
>     /* make coded sequence to short be inreasing */
>@@ -663,7 +733,7 @@ TEST(IsacFixTest, Kenny) {
> 
>     if (testNum == 6) {
>       srand(time(NULL));
>-      for (i = 0; i < stream_len; i++ ) {
>+      for (i = 0; i < static_cast<int>(stream_len); i++) {
>         streamdata[i] = rand();
>       }
>     }
>@@ -681,8 +751,7 @@ TEST(IsacFixTest, Kenny) {
>     /* simulate packet handling through NetEq and the modem */
>     if (!(testNum == 3 && framecnt == 0)) {
>       if (gns == 0) {
>-        get_arrival_time(cur_framesmpls, stream_len, bottleneck,
>-                         &BN_data);
>+        get_arrival_time(cur_framesmpls, stream_len, bottleneck, &BN_data);
>       } else {
>         get_arrival_time2(cur_framesmpls, cur_delay, &BN_data);
>       }
>@@ -690,44 +759,38 @@ TEST(IsacFixTest, Kenny) {
> 
>     /* packet not dropped */
>     if (cur_delay != -1) {
>-
>       /* Error test number 10, garbage data */
>       if (testNum == 10) {
>-        for ( i = 0; i < stream_len; i++) {
>-          streamdata[i] = (short) (streamdata[i] + (short) rand());
>+        for (i = 0; i < static_cast<int>(stream_len); i++) {
>+          streamdata[i] = (short)(streamdata[i] + (short)rand());
>         }
>       }
> 
>       if (testNum != 9) {
>         err = WebRtcIsacfix_UpdateBwEstimate(
>-            ISAC_main_inst,
>-            reinterpret_cast<const uint8_t*>(streamdata),
>-            stream_len,
>-            BN_data.rtp_number,
>-            BN_data.send_time,
>+            ISAC_main_inst, reinterpret_cast<const uint8_t*>(streamdata),
>+            stream_len, BN_data.rtp_number, BN_data.send_time,
>             BN_data.arrival_time);
> 
>         if (err < 0) {
>           /* exit if returned with error */
>-          errtype=WebRtcIsacfix_GetErrorCode(ISAC_main_inst);
>+          errtype = WebRtcIsacfix_GetErrorCode(ISAC_main_inst);
>           printf("\nError in decoder: %d.\n", errtype);
>         }
>       }
> 
>-      if( readLoss == 1 ) {
>-        if( fread( &lostFrame, sizeof(int16_t), 1, plFile ) != 1 ) {
>-          rewind( plFile );
>+      if (readLoss == 1) {
>+        if (fread(&lostFrame, sizeof(int16_t), 1, plFile) != 1) {
>+          rewind(plFile);
>         }
>         lostFrame = !lostFrame;
>       } else {
>-        lostFrame = (rand()%100 < packetLossPercent);
>+        lostFrame = (rand() % 100 < packetLossPercent);
>       }
> 
>-
>-
>       /* iSAC decoding */
>-      if( lostFrame && framecnt >  0) {
>-        if (nbTest !=2) {
>+      if (lostFrame && framecnt > 0) {
>+        if (nbTest != 2) {
>           declen = static_cast<int>(
>               WebRtcIsacfix_DecodePlc(ISAC_main_inst, decoded, prevFrameSize));
>         } else {
>@@ -740,32 +803,29 @@ TEST(IsacFixTest, Kenny) {
>         }
>         lostPackets++;
>       } else {
>-        if (nbTest !=2 ) {
>+        if (nbTest != 2) {
>           size_t FL;
>           /* Call getFramelen, only used here for function test */
>           err = WebRtcIsacfix_ReadFrameLen(
>               reinterpret_cast<const uint8_t*>(streamdata), stream_len, &FL);
>           declen = WebRtcIsacfix_Decode(
>-              ISAC_main_inst,
>-              reinterpret_cast<const uint8_t*>(streamdata),
>-              stream_len,
>-              decoded,
>-              speechType);
>+              ISAC_main_inst, reinterpret_cast<const uint8_t*>(streamdata),
>+              stream_len, decoded, speechType);
>           /* Error check */
>           if (err < 0 || declen < 0 || FL != static_cast<size_t>(declen)) {
>-            errtype=WebRtcIsacfix_GetErrorCode(ISAC_main_inst);
>+            errtype = WebRtcIsacfix_GetErrorCode(ISAC_main_inst);
>             printf(
>                 "\nError %d in ReadFrameLen (%s), Decode (%s), with FL %zu and "
>                 "declen %d.\n",
>                 errtype, err < 0 ? "yes" : "no", declen < 0 ? "yes" : "no", FL,
>                 declen);
>           }
>-          prevFrameSize = static_cast<size_t>(declen/480);
>+          prevFrameSize = static_cast<size_t>(declen / 480);
> 
>         } else {
> #ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
>-          declen = WebRtcIsacfix_DecodeNb( ISAC_main_inst, streamdata,
>-                                           stream_len, decoded, speechType );
>+          declen = WebRtcIsacfix_DecodeNb(ISAC_main_inst, streamdata,
>+                                          stream_len, decoded, speechType);
> #else
>           declen = -1;
> #endif
>@@ -775,13 +835,12 @@ TEST(IsacFixTest, Kenny) {
> 
>       if (declen <= 0) {
>         /* exit if returned with error */
>-        errtype=WebRtcIsacfix_GetErrorCode(ISAC_main_inst);
>+        errtype = WebRtcIsacfix_GetErrorCode(ISAC_main_inst);
>         printf("\nError in decoder: %d.\n", errtype);
>       }
> 
>       /* Write decoded speech frame to file */
>-      if (fwrite(decoded, sizeof(int16_t),
>-                 declen, outp) != (size_t)declen) {
>+      if (fwrite(decoded, sizeof(int16_t), declen, outp) != (size_t)declen) {
>         FAIL();
>       }
>       //   fprintf( ratefile, "%f \n", stream_len / ( ((double)declen)/
>@@ -796,41 +855,43 @@ TEST(IsacFixTest, Kenny) {
> 
>     /* Error test number 10, garbage data */
>     if (testNum == 10) {
>-      if ( (seedfile = fopen(SEED_FILE, "a+t") ) == NULL ) {
>-        printf( "Error: Could not open file %s\n", SEED_FILE);
>-      }
>-      else {
>+      if ((seedfile = fopen(SEED_FILE, "a+t")) == NULL) {
>+        printf("Error: Could not open file %s\n", SEED_FILE);
>+      } else {
>         fprintf(seedfile, "ok\n\n");
>         fclose(seedfile);
>       }
>     }
>   }
>   printf("\nLost Frames %d ~ %4.1f%%\n", lostPackets,
>-         (double)lostPackets/(double)framecnt*100.0 );
>+         (double)lostPackets / (double)framecnt * 100.0);
>   printf("\n\ntotal bits                          = %d bits", totalbits);
>   printf("\nmeasured average bitrate              = %0.3f kbits/s",
>-         (double)totalbits *(FS/1000) / totalsmpls);
>+         (double)totalbits * (FS / 1000) / totalsmpls);
>   printf("\n");
> 
>   /* Runtime statistics */
> 
>-
>-  runtime = (double)(((double)clock()/(double)CLOCKS_PER_SEC)-starttime);
>-  length_file = ((double)framecnt*(double)declen/FS);
>+  runtime = (double)(((double)clock() / (double)CLOCKS_PER_SEC) - starttime);
>+  length_file = ((double)framecnt * (double)declen / FS);
>   printf("\n\nLength of speech file: %.1f s\n", length_file);
>-  printf("Time to run iSAC:      %.2f s (%.2f %% of realtime)\n\n",
>-         runtime, (100*runtime/length_file));
>+  printf("Time to run iSAC:      %.2f s (%.2f %% of realtime)\n\n", runtime,
>+         (100 * runtime / length_file));
>   printf("\n\n_______________________________________________\n");
> 
>   // Record the results with Perf test tools.
>   webrtc::test::PrintResult("isac", "", "time_per_10ms_frame",
>                             (runtime * 10000) / length_file, "us", false);
> 
>+  if (chartjson_result_file) {
>+    webrtc::test::WritePerfResults(chartjson_result_file);
>+  }
>+
>   fclose(inp);
>   fclose(outp);
>   fclose(outbits);
> 
>-  if ( testCE == 1) {
>+  if (testCE == 1) {
>     WebRtcIsacfix_FreeInternal(ISAC_main_inst);
>   }
>   WebRtcIsacfix_Free(ISAC_main_inst);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/include/isac.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/include/isac.h
>index e1ee818059a..6cd70a64195 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/include/isac.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/include/isac.h
>@@ -14,711 +14,648 @@
> #include <stddef.h>
> 
> #include "modules/audio_coding/codecs/isac/bandwidth_info.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
>-typedef struct WebRtcISACStruct    ISACStruct;
>+typedef struct WebRtcISACStruct ISACStruct;
> 
> #if defined(__cplusplus)
> extern "C" {
> #endif
> 
>-  /******************************************************************************
>-   * WebRtcIsac_AssignSize(...)
>-   *
>-   * This function returns the size of the ISAC instance, so that the instance
>-   * can be created outside iSAC.
>-   *
>-   * Input:
>-   *        - samplingRate      : sampling rate of the input/output audio.
>-   *
>-   * Output:
>-   *        - sizeinbytes       : number of bytes needed to allocate for the
>-   *                              instance.
>-   *
>-   * Return value               : 0 - Ok
>-   *                             -1 - Error
>-   */
>-
>-  int16_t WebRtcIsac_AssignSize(
>-      int* sizeinbytes);
>-
>-
>-  /******************************************************************************
>-   * WebRtcIsac_Assign(...)
>-   *
>-   * This function assignes the memory already created to the ISAC instance.
>-   *
>-   * Input:
>-   *        - *ISAC_main_inst   : a pointer to the coder instance.
>-   *        - samplingRate      : sampling rate of the input/output audio.
>-   *        - ISAC_inst_Addr    : the already allocated memory, where we put the
>-   *                              iSAC structure.
>-   *
>-   * Return value               : 0 - Ok
>-   *                             -1 - Error
>-   */
>-
>-  int16_t WebRtcIsac_Assign(
>-      ISACStruct** ISAC_main_inst,
>-      void*        ISAC_inst_Addr);
>-
>-
>-  /******************************************************************************
>-   * WebRtcIsac_Create(...)
>-   *
>-   * This function creates an ISAC instance, which will contain the state
>-   * information for one coding/decoding channel.
>-   *
>-   * Input:
>-   *        - *ISAC_main_inst   : a pointer to the coder instance.
>-   *
>-   * Return value               : 0 - Ok
>-   *                             -1 - Error
>-   */
>-
>-  int16_t WebRtcIsac_Create(
>-      ISACStruct** ISAC_main_inst);
>-
>-
>-  /******************************************************************************
>-   * WebRtcIsac_Free(...)
>-   *
>-   * This function frees the ISAC instance created at the beginning.
>-   *
>-   * Input:
>-   *        - ISAC_main_inst    : an ISAC instance.
>-   *
>-   * Return value               : 0 - Ok
>-   *                             -1 - Error
>-   */
>-
>-  int16_t WebRtcIsac_Free(
>-      ISACStruct* ISAC_main_inst);
>-
>-
>-  /******************************************************************************
>-   * WebRtcIsac_EncoderInit(...)
>-   *
>-   * This function initializes an ISAC instance prior to the encoder calls.
>-   *
>-   * Input:
>-   *        - ISAC_main_inst    : ISAC instance.
>-   *        - CodingMode        : 0 -> Bit rate and frame length are
>-   *                                automatically adjusted to available bandwidth
>-   *                                on transmission channel, just valid if codec
>-   *                                is created to work in wideband mode.
>-   *                              1 -> User sets a frame length and a target bit
>-   *                                rate which is taken as the maximum
>-   *                                short-term average bit rate.
>-   *
>-   * Return value               : 0 - Ok
>-   *                             -1 - Error
>-   */
>-
>-  int16_t WebRtcIsac_EncoderInit(
>-      ISACStruct* ISAC_main_inst,
>-      int16_t CodingMode);
>-
>-
>-  /******************************************************************************
>-   * WebRtcIsac_Encode(...)
>-   *
>-   * This function encodes 10ms audio blocks and inserts it into a package.
>-   * Input speech length has 160 samples if operating at 16 kHz sampling
>-   * rate, or 320 if operating at 32 kHz sampling rate. The encoder buffers the
>-   * input audio until the whole frame is buffered then proceeds with encoding.
>-   *
>-   *
>-   * Input:
>-   *        - ISAC_main_inst    : ISAC instance.
>-   *        - speechIn          : input speech vector.
>-   *
>-   * Output:
>-   *        - encoded           : the encoded data vector
>-   *
>-   * Return value:
>-   *                            : >0 - Length (in bytes) of coded data
>-   *                            :  0 - The buffer didn't reach the chosen
>-   *                               frame-size so it keeps buffering speech
>-   *                               samples.
>-   *                            : -1 - Error
>-   */
>-
>-  int WebRtcIsac_Encode(
>-      ISACStruct*        ISAC_main_inst,
>-      const int16_t* speechIn,
>-      uint8_t* encoded);
>-
>-
>-  /******************************************************************************
>-   * WebRtcIsac_DecoderInit(...)
>-   *
>-   * This function initializes an ISAC instance prior to the decoder calls.
>-   *
>-   * Input:
>-   *        - ISAC_main_inst    : ISAC instance.
>-   */
>-
>-  void WebRtcIsac_DecoderInit(ISACStruct* ISAC_main_inst);
>-
>-  /******************************************************************************
>-   * WebRtcIsac_UpdateBwEstimate(...)
>-   *
>-   * This function updates the estimate of the bandwidth.
>-   *
>-   * Input:
>-   *        - ISAC_main_inst    : ISAC instance.
>-   *        - encoded           : encoded ISAC frame(s).
>-   *        - packet_size       : size of the packet.
>-   *        - rtp_seq_number    : the RTP number of the packet.
>-   *        - send_ts           : the RTP send timestamp, given in samples
>-   *        - arr_ts            : the arrival time of the packet (from NetEq)
>-   *                              in samples.
>-   *
>-   * Return value               : 0 - Ok
>-   *                             -1 - Error
>-   */
>-
>-  int16_t WebRtcIsac_UpdateBwEstimate(
>-      ISACStruct*         ISAC_main_inst,
>-      const uint8_t* encoded,
>-      size_t         packet_size,
>-      uint16_t        rtp_seq_number,
>-      uint32_t        send_ts,
>-      uint32_t        arr_ts);
>-
>-
>-  /******************************************************************************
>-   * WebRtcIsac_Decode(...)
>-   *
>-   * This function decodes an ISAC frame. At 16 kHz sampling rate, the length
>-   * of the output audio could be either 480 or 960 samples, equivalent to
>-   * 30 or 60 ms respectively. At 32 kHz sampling rate, the length of the
>-   * output audio is 960 samples, which is 30 ms.
>-   *
>-   * Input:
>-   *        - ISAC_main_inst    : ISAC instance.
>-   *        - encoded           : encoded ISAC frame(s).
>-   *        - len               : bytes in encoded vector.
>-   *
>-   * Output:
>-   *        - decoded           : The decoded vector.
>-   *
>-   * Return value               : >0 - number of samples in decoded vector.
>-   *                              -1 - Error.
>-   */
>-
>-  int WebRtcIsac_Decode(
>-      ISACStruct*           ISAC_main_inst,
>-      const uint8_t* encoded,
>-      size_t         len,
>-      int16_t*        decoded,
>-      int16_t*        speechType);
>-
>-
>-  /******************************************************************************
>-   * WebRtcIsac_DecodePlc(...)
>-   *
>-   * This function conducts PLC for ISAC frame(s). Output speech length
>-   * will be a multiple of frames, i.e. multiples of 30 ms audio. Therefore,
>-   * the output is multiple of 480 samples if operating at 16 kHz and multiple
>-   * of 960 if operating at 32 kHz.
>-   *
>-   * Input:
>-   *        - ISAC_main_inst    : ISAC instance.
>-   *        - noOfLostFrames    : Number of PLC frames to produce.
>-   *
>-   * Output:
>-   *        - decoded           : The decoded vector.
>-   *
>-   * Return value               : Number of samples in decoded PLC vector
>-   */
>-
>-  size_t WebRtcIsac_DecodePlc(
>-      ISACStruct*  ISAC_main_inst,
>-      int16_t* decoded,
>-      size_t  noOfLostFrames);
>-
>-
>-  /******************************************************************************
>-   * WebRtcIsac_Control(...)
>-   *
>-   * This function sets the limit on the short-term average bit-rate and the
>-   * frame length. Should be used only in Instantaneous mode. At 16 kHz sampling
>-   * rate, an average bit-rate between 10000 to 32000 bps is valid and a
>-   * frame-size of 30 or 60 ms is acceptable. At 32 kHz, an average bit-rate
>-   * between 10000 to 56000 is acceptable, and the valid frame-size is 30 ms.
>-   *
>-   * Input:
>-   *        - ISAC_main_inst    : ISAC instance.
>-   *        - rate              : limit on the short-term average bit rate,
>-   *                              in bits/second.
>-   *        - framesize         : frame-size in millisecond.
>-   *
>-   * Return value               : 0  - ok
>-   *                             -1 - Error
>-   */
>-
>-  int16_t WebRtcIsac_Control(
>-      ISACStruct*   ISAC_main_inst,
>-      int32_t rate,
>-      int framesize);
>-
>-  void WebRtcIsac_SetInitialBweBottleneck(ISACStruct* ISAC_main_inst,
>-                                          int bottleneck_bits_per_second);
>-
>-  /******************************************************************************
>-   * WebRtcIsac_ControlBwe(...)
>-   *
>-   * This function sets the initial values of bottleneck and frame-size if
>-   * iSAC is used in channel-adaptive mode. Therefore, this API is not
>-   * applicable if the codec is created to operate in super-wideband mode.
>-   *
>-   * Through this API, users can enforce a frame-size for all values of
>-   * bottleneck. Then iSAC will not automatically change the frame-size.
>-   *
>-   *
>-   * Input:
>-   *        - ISAC_main_inst    : ISAC instance.
>-   *        - rateBPS           : initial value of bottleneck in bits/second
>-   *                              10000 <= rateBPS <= 56000 is accepted
>-   *                              For default bottleneck set rateBPS = 0
>-   *        - frameSizeMs       : number of milliseconds per frame (30 or 60)
>-   *        - enforceFrameSize  : 1 to enforce the given frame-size through
>-   *                              out the adaptation process, 0 to let iSAC
>-   *                              change the frame-size if required.
>-   *
>-   * Return value               : 0  - ok
>-   *                             -1 - Error
>-   */
>-
>-  int16_t WebRtcIsac_ControlBwe(
>-      ISACStruct* ISAC_main_inst,
>-      int32_t rateBPS,
>-      int frameSizeMs,
>-      int16_t enforceFrameSize);
>-
>-
>-  /******************************************************************************
>-   * WebRtcIsac_ReadFrameLen(...)
>-   *
>-   * This function returns the length of the frame represented in the packet.
>-   *
>-   * Input:
>-   *        - encoded           : Encoded bit-stream
>-   *
>-   * Output:
>-   *        - frameLength       : Length of frame in packet (in samples)
>-   *
>-   */
>-
>-  int16_t WebRtcIsac_ReadFrameLen(
>-      ISACStruct*          ISAC_main_inst,
>-      const uint8_t* encoded,
>-      int16_t*       frameLength);
>-
>-
>-  /******************************************************************************
>-   * WebRtcIsac_version(...)
>-   *
>-   * This function returns the version number.
>-   *
>-   * Output:
>-   *        - version      : Pointer to character string
>-   *
>-   */
>-
>-  void WebRtcIsac_version(
>-      char *version);
>-
>-
>-  /******************************************************************************
>-   * WebRtcIsac_GetErrorCode(...)
>-   *
>-   * This function can be used to check the error code of an iSAC instance. When
>-   * a function returns -1 a error code will be set for that instance. The
>-   * function below extract the code of the last error that occurred in the
>-   * specified instance.
>-   *
>-   * Input:
>-   *        - ISAC_main_inst    : ISAC instance
>-   *
>-   * Return value               : Error code
>-   */
>-
>-  int16_t WebRtcIsac_GetErrorCode(
>-      ISACStruct* ISAC_main_inst);
>-
>-
>-  /****************************************************************************
>-   * WebRtcIsac_GetUplinkBw(...)
>-   *
>-   * This function outputs the target bottleneck of the codec. In
>-   * channel-adaptive mode, the target bottleneck is specified through in-band
>-   * signalling retreived by bandwidth estimator.
>-   * In channel-independent, also called instantaneous mode, the target
>-   * bottleneck is provided to the encoder by calling xxx_control(...). If
>-   * xxx_control is never called the default values is returned. The default
>-   * value for bottleneck at 16 kHz encoder sampling rate is 32000 bits/sec,
>-   * and it is 56000 bits/sec for 32 kHz sampling rate.
>-   * Note that the output is the iSAC internal operating bottleneck which might
>-   * differ slightly from the one provided through xxx_control().
>-   *
>-   * Input:
>-   *        - ISAC_main_inst    : iSAC instance
>-   *
>-   * Output:
>-   *        - *bottleneck       : bottleneck in bits/sec
>-   *
>-   * Return value               : -1 if error happens
>-   *                               0 bit-rates computed correctly.
>-   */
>-
>-  int16_t WebRtcIsac_GetUplinkBw(
>-      ISACStruct*    ISAC_main_inst,
>-      int32_t* bottleneck);
>-
>-
>-  /******************************************************************************
>-   * WebRtcIsac_SetMaxPayloadSize(...)
>-   *
>-   * This function sets a limit for the maximum payload size of iSAC. The same
>-   * value is used both for 30 and 60 ms packets. If the encoder sampling rate
>-   * is 16 kHz the maximum payload size is between 120 and 400 bytes. If the
>-   * encoder sampling rate is 32 kHz the maximum payload size is between 120
>-   * and 600 bytes.
>-   *
>-   * If an out of range limit is used, the function returns -1, but the closest
>-   * valid value will be applied.
>-   *
>-   * ---------------
>-   * IMPORTANT NOTES
>-   * ---------------
>-   * The size of a packet is limited to the minimum of 'max-payload-size' and
>-   * 'max-rate.' For instance, let's assume the max-payload-size is set to
>-   * 170 bytes, and max-rate is set to 40 kbps. Note that a limit of 40 kbps
>-   * translates to 150 bytes for 30ms frame-size & 300 bytes for 60ms
>-   * frame-size. Then a packet with a frame-size of 30 ms is limited to 150,
>-   * i.e. min(170, 150), and a packet with 60 ms frame-size is limited to
>-   * 170 bytes, i.e. min(170, 300).
>-   *
>-   * Input:
>-   *        - ISAC_main_inst    : iSAC instance
>-   *        - maxPayloadBytes   : maximum size of the payload in bytes
>-   *                              valid values are between 120 and 400 bytes
>-   *                              if encoder sampling rate is 16 kHz. For
>-   *                              32 kHz encoder sampling rate valid values
>-   *                              are between 120 and 600 bytes.
>-   *
>-   * Return value               : 0 if successful
>-   *                             -1 if error happens
>-   */
>-
>-  int16_t WebRtcIsac_SetMaxPayloadSize(
>-      ISACStruct* ISAC_main_inst,
>-      int16_t maxPayloadBytes);
>-
>-
>-  /******************************************************************************
>-   * WebRtcIsac_SetMaxRate(...)
>-   *
>-   * This function sets the maximum rate which the codec may not exceed for
>-   * any signal packet. The maximum rate is defined and payload-size per
>-   * frame-size in bits per second.
>-   *
>-   * The codec has a maximum rate of 53400 bits per second (200 bytes per 30
>-   * ms) if the encoder sampling rate is 16kHz, and 160 kbps (600 bytes/30 ms)
>-   * if the encoder sampling rate is 32 kHz.
>-   *
>-   * It is possible to set a maximum rate between 32000 and 53400 bits/sec
>-   * in wideband mode, and 32000 to 160000 bits/sec in super-wideband mode.
>-   *
>-   * If an out of range limit is used, the function returns -1, but the closest
>-   * valid value will be applied.
>-   *
>-   * ---------------
>-   * IMPORTANT NOTES
>-   * ---------------
>-   * The size of a packet is limited to the minimum of 'max-payload-size' and
>-   * 'max-rate.' For instance, let's assume the max-payload-size is set to
>-   * 170 bytes, and max-rate is set to 40 kbps. Note that a limit of 40 kbps
>-   * translates to 150 bytes for 30ms frame-size & 300 bytes for 60ms
>-   * frame-size. Then a packet with a frame-size of 30 ms is limited to 150,
>-   * i.e. min(170, 150), and a packet with 60 ms frame-size is limited to
>-   * 170 bytes, min(170, 300).
>-   *
>-   * Input:
>-   *        - ISAC_main_inst    : iSAC instance
>-   *        - maxRate           : maximum rate in bits per second,
>-   *                              valid values are 32000 to 53400 bits/sec in
>-   *                              wideband mode, and 32000 to 160000 bits/sec in
>-   *                              super-wideband mode.
>-   *
>-   * Return value               : 0 if successful
>-   *                             -1 if error happens
>-   */
>-
>-  int16_t WebRtcIsac_SetMaxRate(
>-      ISACStruct* ISAC_main_inst,
>-      int32_t maxRate);
>-
>-
>-  /******************************************************************************
>-   * WebRtcIsac_DecSampRate()
>-   * Return the sampling rate of the decoded audio.
>-   *
>-   * Input:
>-   *        - ISAC_main_inst    : iSAC instance
>-   *
>-   * Return value               : sampling frequency in Hertz.
>-   *
>-   */
>-
>-  uint16_t WebRtcIsac_DecSampRate(ISACStruct* ISAC_main_inst);
>-
>-
>-  /******************************************************************************
>-   * WebRtcIsac_EncSampRate()
>-   *
>-   * Input:
>-   *        - ISAC_main_inst    : iSAC instance
>-   *
>-   * Return value               : sampling rate in Hertz.
>-   *
>-   */
>-
>-  uint16_t WebRtcIsac_EncSampRate(ISACStruct* ISAC_main_inst);
>-
>-
>-  /******************************************************************************
>-   * WebRtcIsac_SetDecSampRate()
>-   * Set the sampling rate of the decoder.  Initialization of the decoder WILL
>-   * NOT overwrite the sampling rate of the encoder. The default value is 16 kHz
>-   * which is set when the instance is created.
>-   *
>-   * Input:
>-   *        - ISAC_main_inst    : iSAC instance
>-   *        - sampRate          : sampling rate in Hertz.
>-   *
>-   * Return value               : 0 if successful
>-   *                             -1 if failed.
>-   */
>-
>-  int16_t WebRtcIsac_SetDecSampRate(ISACStruct* ISAC_main_inst,
>-                                          uint16_t samp_rate_hz);
>-
>-
>-  /******************************************************************************
>-   * WebRtcIsac_SetEncSampRate()
>-   * Set the sampling rate of the encoder. Initialization of the encoder WILL
>-   * NOT overwrite the sampling rate of the encoder. The default value is 16 kHz
>-   * which is set when the instance is created. The encoding-mode and the
>-   * bottleneck remain unchanged by this call, however, the maximum rate and
>-   * maximum payload-size will reset to their default value.
>-   *
>-   * Input:
>-   *        - ISAC_main_inst    : iSAC instance
>-   *        - sampRate          : sampling rate in Hertz.
>-   *
>-   * Return value               : 0 if successful
>-   *                             -1 if failed.
>-   */
>-
>-  int16_t WebRtcIsac_SetEncSampRate(ISACStruct* ISAC_main_inst,
>-                                          uint16_t sample_rate_hz);
>-
>-
>-
>-  /******************************************************************************
>-   * WebRtcIsac_GetNewBitStream(...)
>-   *
>-   * This function returns encoded data, with the recieved bwe-index in the
>-   * stream. If the rate is set to a value less than bottleneck of codec
>-   * the new bistream will be re-encoded with the given target rate.
>-   * It should always return a complete packet, i.e. only called once
>-   * even for 60 msec frames.
>-   *
>-   * NOTE 1! This function does not write in the ISACStruct, it is not allowed.
>-   * NOTE 2! Currently not implemented for SWB mode.
>-   * NOTE 3! Rates larger than the bottleneck of the codec will be limited
>-   *         to the current bottleneck.
>-   *
>-   * Input:
>-   *        - ISAC_main_inst    : ISAC instance.
>-   *        - bweIndex          : Index of bandwidth estimate to put in new
>-   *                              bitstream
>-   *        - rate              : target rate of the transcoder is bits/sec.
>-   *                              Valid values are the accepted rate in iSAC,
>-   *                              i.e. 10000 to 56000.
>-   *        - isRCU                       : if the new bit-stream is an RCU stream.
>-   *                              Note that the rate parameter always indicates
>-   *                              the target rate of the main payload, regardless
>-   *                              of 'isRCU' value.
>-   *
>-   * Output:
>-   *        - encoded           : The encoded data vector
>-   *
>-   * Return value               : >0 - Length (in bytes) of coded data
>-   *                              -1 - Error  or called in SWB mode
>-   *                                 NOTE! No error code is written to
>-   *                                 the struct since it is only allowed to read
>-   *                                 the struct.
>-   */
>-  int16_t WebRtcIsac_GetNewBitStream(
>-      ISACStruct*    ISAC_main_inst,
>-      int16_t  bweIndex,
>-      int16_t  jitterInfo,
>-      int32_t  rate,
>-      uint8_t* encoded,
>-      int16_t  isRCU);
>-
>-
>-
>-  /****************************************************************************
>-   * WebRtcIsac_GetDownLinkBwIndex(...)
>-   *
>-   * This function returns index representing the Bandwidth estimate from
>-   * other side to this side.
>-   *
>-   * Input:
>-   *        - ISAC_main_inst    : iSAC struct
>-   *
>-   * Output:
>-   *        - bweIndex          : Bandwidth estimate to transmit to other side.
>-   *
>-   */
>-
>-  int16_t WebRtcIsac_GetDownLinkBwIndex(
>-      ISACStruct*  ISAC_main_inst,
>-      int16_t* bweIndex,
>-      int16_t* jitterInfo);
>-
>-
>-  /****************************************************************************
>-   * WebRtcIsac_UpdateUplinkBw(...)
>-   *
>-   * This function takes an index representing the Bandwidth estimate from
>-   * this side to other side and updates BWE.
>-   *
>-   * Input:
>-   *        - ISAC_main_inst    : iSAC struct
>-   *        - bweIndex          : Bandwidth estimate from other side.
>-   *
>-   */
>-
>-  int16_t WebRtcIsac_UpdateUplinkBw(
>-      ISACStruct* ISAC_main_inst,
>-      int16_t bweIndex);
>-
>-
>-  /****************************************************************************
>-   * WebRtcIsac_ReadBwIndex(...)
>-   *
>-   * This function returns the index of the Bandwidth estimate from the bitstream.
>-   *
>-   * Input:
>-   *        - encoded           : Encoded bitstream
>-   *
>-   * Output:
>-   *        - frameLength       : Length of frame in packet (in samples)
>-   *        - bweIndex         : Bandwidth estimate in bitstream
>-   *
>-   */
>-
>-  int16_t WebRtcIsac_ReadBwIndex(
>-      const uint8_t* encoded,
>-      int16_t*       bweIndex);
>-
>-
>-
>-  /*******************************************************************************
>-   * WebRtcIsac_GetNewFrameLen(...)
>-   *
>-   * returns the frame lenght (in samples) of the next packet. In the case of channel-adaptive
>-   * mode, iSAC decides on its frame lenght based on the estimated bottleneck
>-   * this allows a user to prepare for the next packet (at the encoder)
>-   *
>-   * The primary usage is in CE to make the iSAC works in channel-adaptive mode
>-   *
>-   * Input:
>-   *        - ISAC_main_inst     : iSAC struct
>-   *
>-   * Return Value                : frame lenght in samples
>-   *
>-   */
>-
>-  int16_t WebRtcIsac_GetNewFrameLen(
>-      ISACStruct* ISAC_main_inst);
>-
>-
>-  /****************************************************************************
>-   *  WebRtcIsac_GetRedPayload(...)
>-   *
>-   *  Populates "encoded" with the redundant payload of the recently encoded
>-   *  frame. This function has to be called once that WebRtcIsac_Encode(...)
>-   *  returns a positive value. Regardless of the frame-size this function will
>-   *  be called only once after encoding is completed.
>-   *
>-   * Input:
>-   *      - ISAC_main_inst    : iSAC struct
>-   *
>-   * Output:
>-   *        - encoded            : the encoded data vector
>-   *
>-   *
>-   * Return value:
>-   *                              : >0 - Length (in bytes) of coded data
>-   *                              : -1 - Error
>-   *
>-   *
>-   */
>-  int16_t WebRtcIsac_GetRedPayload(
>-      ISACStruct*    ISAC_main_inst,
>-      uint8_t* encoded);
>-
>-
>-  /****************************************************************************
>-   * WebRtcIsac_DecodeRcu(...)
>-   *
>-   * This function decodes a redundant (RCU) iSAC frame. Function is called in
>-   * NetEq with a stored RCU payload i case of packet loss. Output speech length
>-   * will be a multiple of 480 samples: 480 or 960 samples,
>-   * depending on the framesize (30 or 60 ms).
>-   *
>-   * Input:
>-   *      - ISAC_main_inst     : ISAC instance.
>-   *      - encoded            : encoded ISAC RCU frame(s)
>-   *      - len                : bytes in encoded vector
>-   *
>-   * Output:
>-   *      - decoded            : The decoded vector
>-   *
>-   * Return value              : >0 - number of samples in decoded vector
>-   *                             -1 - Error
>-   */
>-  int WebRtcIsac_DecodeRcu(
>-      ISACStruct*           ISAC_main_inst,
>-      const uint8_t* encoded,
>-      size_t         len,
>-      int16_t*        decoded,
>-      int16_t*        speechType);
>-
>-  /* Fills in an IsacBandwidthInfo struct. |inst| should be a decoder. */
>-  void WebRtcIsac_GetBandwidthInfo(ISACStruct* inst, IsacBandwidthInfo* bwinfo);
>-
>-  /* Uses the values from an IsacBandwidthInfo struct. |inst| should be an
>-     encoder. */
>-  void WebRtcIsac_SetBandwidthInfo(ISACStruct* inst,
>-                                   const IsacBandwidthInfo* bwinfo);
>-
>-  /* If |inst| is a decoder but not an encoder: tell it what sample rate the
>-     encoder is using, for bandwidth estimation purposes. */
>-  void WebRtcIsac_SetEncSampRateInDecoder(ISACStruct* inst, int sample_rate_hz);
>+/******************************************************************************
>+ * WebRtcIsac_AssignSize(...)
>+ *
>+ * This function returns the size of the ISAC instance, so that the instance
>+ * can be created outside iSAC.
>+ *
>+ * Input:
>+ *        - samplingRate      : sampling rate of the input/output audio.
>+ *
>+ * Output:
>+ *        - sizeinbytes       : number of bytes needed to allocate for the
>+ *                              instance.
>+ *
>+ * Return value               : 0 - Ok
>+ *                             -1 - Error
>+ */
>+
>+int16_t WebRtcIsac_AssignSize(int* sizeinbytes);
>+
>+/******************************************************************************
>+ * WebRtcIsac_Assign(...)
>+ *
>+ * This function assignes the memory already created to the ISAC instance.
>+ *
>+ * Input:
>+ *        - *ISAC_main_inst   : a pointer to the coder instance.
>+ *        - samplingRate      : sampling rate of the input/output audio.
>+ *        - ISAC_inst_Addr    : the already allocated memory, where we put the
>+ *                              iSAC structure.
>+ *
>+ * Return value               : 0 - Ok
>+ *                             -1 - Error
>+ */
>+
>+int16_t WebRtcIsac_Assign(ISACStruct** ISAC_main_inst, void* ISAC_inst_Addr);
>+
>+/******************************************************************************
>+ * WebRtcIsac_Create(...)
>+ *
>+ * This function creates an ISAC instance, which will contain the state
>+ * information for one coding/decoding channel.
>+ *
>+ * Input:
>+ *        - *ISAC_main_inst   : a pointer to the coder instance.
>+ *
>+ * Return value               : 0 - Ok
>+ *                             -1 - Error
>+ */
>+
>+int16_t WebRtcIsac_Create(ISACStruct** ISAC_main_inst);
>+
>+/******************************************************************************
>+ * WebRtcIsac_Free(...)
>+ *
>+ * This function frees the ISAC instance created at the beginning.
>+ *
>+ * Input:
>+ *        - ISAC_main_inst    : an ISAC instance.
>+ *
>+ * Return value               : 0 - Ok
>+ *                             -1 - Error
>+ */
>+
>+int16_t WebRtcIsac_Free(ISACStruct* ISAC_main_inst);
>+
>+/******************************************************************************
>+ * WebRtcIsac_EncoderInit(...)
>+ *
>+ * This function initializes an ISAC instance prior to the encoder calls.
>+ *
>+ * Input:
>+ *        - ISAC_main_inst    : ISAC instance.
>+ *        - CodingMode        : 0 -> Bit rate and frame length are
>+ *                                automatically adjusted to available bandwidth
>+ *                                on transmission channel, just valid if codec
>+ *                                is created to work in wideband mode.
>+ *                              1 -> User sets a frame length and a target bit
>+ *                                rate which is taken as the maximum
>+ *                                short-term average bit rate.
>+ *
>+ * Return value               : 0 - Ok
>+ *                             -1 - Error
>+ */
>+
>+int16_t WebRtcIsac_EncoderInit(ISACStruct* ISAC_main_inst, int16_t CodingMode);
>+
>+/******************************************************************************
>+ * WebRtcIsac_Encode(...)
>+ *
>+ * This function encodes 10ms audio blocks and inserts it into a package.
>+ * Input speech length has 160 samples if operating at 16 kHz sampling
>+ * rate, or 320 if operating at 32 kHz sampling rate. The encoder buffers the
>+ * input audio until the whole frame is buffered then proceeds with encoding.
>+ *
>+ *
>+ * Input:
>+ *        - ISAC_main_inst    : ISAC instance.
>+ *        - speechIn          : input speech vector.
>+ *
>+ * Output:
>+ *        - encoded           : the encoded data vector
>+ *
>+ * Return value:
>+ *                            : >0 - Length (in bytes) of coded data
>+ *                            :  0 - The buffer didn't reach the chosen
>+ *                               frame-size so it keeps buffering speech
>+ *                               samples.
>+ *                            : -1 - Error
>+ */
>+
>+int WebRtcIsac_Encode(ISACStruct* ISAC_main_inst,
>+                      const int16_t* speechIn,
>+                      uint8_t* encoded);
>+
>+/******************************************************************************
>+ * WebRtcIsac_DecoderInit(...)
>+ *
>+ * This function initializes an ISAC instance prior to the decoder calls.
>+ *
>+ * Input:
>+ *        - ISAC_main_inst    : ISAC instance.
>+ */
>+
>+void WebRtcIsac_DecoderInit(ISACStruct* ISAC_main_inst);
>+
>+/******************************************************************************
>+ * WebRtcIsac_UpdateBwEstimate(...)
>+ *
>+ * This function updates the estimate of the bandwidth.
>+ *
>+ * Input:
>+ *        - ISAC_main_inst    : ISAC instance.
>+ *        - encoded           : encoded ISAC frame(s).
>+ *        - packet_size       : size of the packet.
>+ *        - rtp_seq_number    : the RTP number of the packet.
>+ *        - send_ts           : the RTP send timestamp, given in samples
>+ *        - arr_ts            : the arrival time of the packet (from NetEq)
>+ *                              in samples.
>+ *
>+ * Return value               : 0 - Ok
>+ *                             -1 - Error
>+ */
>+
>+int16_t WebRtcIsac_UpdateBwEstimate(ISACStruct* ISAC_main_inst,
>+                                    const uint8_t* encoded,
>+                                    size_t packet_size,
>+                                    uint16_t rtp_seq_number,
>+                                    uint32_t send_ts,
>+                                    uint32_t arr_ts);
>+
>+/******************************************************************************
>+ * WebRtcIsac_Decode(...)
>+ *
>+ * This function decodes an ISAC frame. At 16 kHz sampling rate, the length
>+ * of the output audio could be either 480 or 960 samples, equivalent to
>+ * 30 or 60 ms respectively. At 32 kHz sampling rate, the length of the
>+ * output audio is 960 samples, which is 30 ms.
>+ *
>+ * Input:
>+ *        - ISAC_main_inst    : ISAC instance.
>+ *        - encoded           : encoded ISAC frame(s).
>+ *        - len               : bytes in encoded vector.
>+ *
>+ * Output:
>+ *        - decoded           : The decoded vector.
>+ *
>+ * Return value               : >0 - number of samples in decoded vector.
>+ *                              -1 - Error.
>+ */
>+
>+int WebRtcIsac_Decode(ISACStruct* ISAC_main_inst,
>+                      const uint8_t* encoded,
>+                      size_t len,
>+                      int16_t* decoded,
>+                      int16_t* speechType);
>+
>+/******************************************************************************
>+ * WebRtcIsac_DecodePlc(...)
>+ *
>+ * This function conducts PLC for ISAC frame(s). Output speech length
>+ * will be a multiple of frames, i.e. multiples of 30 ms audio. Therefore,
>+ * the output is multiple of 480 samples if operating at 16 kHz and multiple
>+ * of 960 if operating at 32 kHz.
>+ *
>+ * Input:
>+ *        - ISAC_main_inst    : ISAC instance.
>+ *        - noOfLostFrames    : Number of PLC frames to produce.
>+ *
>+ * Output:
>+ *        - decoded           : The decoded vector.
>+ *
>+ * Return value               : Number of samples in decoded PLC vector
>+ */
>+
>+size_t WebRtcIsac_DecodePlc(ISACStruct* ISAC_main_inst,
>+                            int16_t* decoded,
>+                            size_t noOfLostFrames);
>+
>+/******************************************************************************
>+ * WebRtcIsac_Control(...)
>+ *
>+ * This function sets the limit on the short-term average bit-rate and the
>+ * frame length. Should be used only in Instantaneous mode. At 16 kHz sampling
>+ * rate, an average bit-rate between 10000 to 32000 bps is valid and a
>+ * frame-size of 30 or 60 ms is acceptable. At 32 kHz, an average bit-rate
>+ * between 10000 to 56000 is acceptable, and the valid frame-size is 30 ms.
>+ *
>+ * Input:
>+ *        - ISAC_main_inst    : ISAC instance.
>+ *        - rate              : limit on the short-term average bit rate,
>+ *                              in bits/second.
>+ *        - framesize         : frame-size in millisecond.
>+ *
>+ * Return value               : 0  - ok
>+ *                             -1 - Error
>+ */
>+
>+int16_t WebRtcIsac_Control(ISACStruct* ISAC_main_inst,
>+                           int32_t rate,
>+                           int framesize);
>+
>+void WebRtcIsac_SetInitialBweBottleneck(ISACStruct* ISAC_main_inst,
>+                                        int bottleneck_bits_per_second);
>+
>+/******************************************************************************
>+ * WebRtcIsac_ControlBwe(...)
>+ *
>+ * This function sets the initial values of bottleneck and frame-size if
>+ * iSAC is used in channel-adaptive mode. Therefore, this API is not
>+ * applicable if the codec is created to operate in super-wideband mode.
>+ *
>+ * Through this API, users can enforce a frame-size for all values of
>+ * bottleneck. Then iSAC will not automatically change the frame-size.
>+ *
>+ *
>+ * Input:
>+ *        - ISAC_main_inst    : ISAC instance.
>+ *        - rateBPS           : initial value of bottleneck in bits/second
>+ *                              10000 <= rateBPS <= 56000 is accepted
>+ *                              For default bottleneck set rateBPS = 0
>+ *        - frameSizeMs       : number of milliseconds per frame (30 or 60)
>+ *        - enforceFrameSize  : 1 to enforce the given frame-size through
>+ *                              out the adaptation process, 0 to let iSAC
>+ *                              change the frame-size if required.
>+ *
>+ * Return value               : 0  - ok
>+ *                             -1 - Error
>+ */
>+
>+int16_t WebRtcIsac_ControlBwe(ISACStruct* ISAC_main_inst,
>+                              int32_t rateBPS,
>+                              int frameSizeMs,
>+                              int16_t enforceFrameSize);
>+
>+/******************************************************************************
>+ * WebRtcIsac_ReadFrameLen(...)
>+ *
>+ * This function returns the length of the frame represented in the packet.
>+ *
>+ * Input:
>+ *        - encoded           : Encoded bit-stream
>+ *
>+ * Output:
>+ *        - frameLength       : Length of frame in packet (in samples)
>+ *
>+ */
>+
>+int16_t WebRtcIsac_ReadFrameLen(ISACStruct* ISAC_main_inst,
>+                                const uint8_t* encoded,
>+                                int16_t* frameLength);
>+
>+/******************************************************************************
>+ * WebRtcIsac_version(...)
>+ *
>+ * This function returns the version number.
>+ *
>+ * Output:
>+ *        - version      : Pointer to character string
>+ *
>+ */
>+
>+void WebRtcIsac_version(char* version);
>+
>+/******************************************************************************
>+ * WebRtcIsac_GetErrorCode(...)
>+ *
>+ * This function can be used to check the error code of an iSAC instance. When
>+ * a function returns -1 a error code will be set for that instance. The
>+ * function below extract the code of the last error that occurred in the
>+ * specified instance.
>+ *
>+ * Input:
>+ *        - ISAC_main_inst    : ISAC instance
>+ *
>+ * Return value               : Error code
>+ */
>+
>+int16_t WebRtcIsac_GetErrorCode(ISACStruct* ISAC_main_inst);
>+
>+/****************************************************************************
>+ * WebRtcIsac_GetUplinkBw(...)
>+ *
>+ * This function outputs the target bottleneck of the codec. In
>+ * channel-adaptive mode, the target bottleneck is specified through in-band
>+ * signalling retreived by bandwidth estimator.
>+ * In channel-independent, also called instantaneous mode, the target
>+ * bottleneck is provided to the encoder by calling xxx_control(...). If
>+ * xxx_control is never called the default values is returned. The default
>+ * value for bottleneck at 16 kHz encoder sampling rate is 32000 bits/sec,
>+ * and it is 56000 bits/sec for 32 kHz sampling rate.
>+ * Note that the output is the iSAC internal operating bottleneck which might
>+ * differ slightly from the one provided through xxx_control().
>+ *
>+ * Input:
>+ *        - ISAC_main_inst    : iSAC instance
>+ *
>+ * Output:
>+ *        - *bottleneck       : bottleneck in bits/sec
>+ *
>+ * Return value               : -1 if error happens
>+ *                               0 bit-rates computed correctly.
>+ */
>+
>+int16_t WebRtcIsac_GetUplinkBw(ISACStruct* ISAC_main_inst, int32_t* bottleneck);
>+
>+/******************************************************************************
>+ * WebRtcIsac_SetMaxPayloadSize(...)
>+ *
>+ * This function sets a limit for the maximum payload size of iSAC. The same
>+ * value is used both for 30 and 60 ms packets. If the encoder sampling rate
>+ * is 16 kHz the maximum payload size is between 120 and 400 bytes. If the
>+ * encoder sampling rate is 32 kHz the maximum payload size is between 120
>+ * and 600 bytes.
>+ *
>+ * If an out of range limit is used, the function returns -1, but the closest
>+ * valid value will be applied.
>+ *
>+ * ---------------
>+ * IMPORTANT NOTES
>+ * ---------------
>+ * The size of a packet is limited to the minimum of 'max-payload-size' and
>+ * 'max-rate.' For instance, let's assume the max-payload-size is set to
>+ * 170 bytes, and max-rate is set to 40 kbps. Note that a limit of 40 kbps
>+ * translates to 150 bytes for 30ms frame-size & 300 bytes for 60ms
>+ * frame-size. Then a packet with a frame-size of 30 ms is limited to 150,
>+ * i.e. min(170, 150), and a packet with 60 ms frame-size is limited to
>+ * 170 bytes, i.e. min(170, 300).
>+ *
>+ * Input:
>+ *        - ISAC_main_inst    : iSAC instance
>+ *        - maxPayloadBytes   : maximum size of the payload in bytes
>+ *                              valid values are between 120 and 400 bytes
>+ *                              if encoder sampling rate is 16 kHz. For
>+ *                              32 kHz encoder sampling rate valid values
>+ *                              are between 120 and 600 bytes.
>+ *
>+ * Return value               : 0 if successful
>+ *                             -1 if error happens
>+ */
>+
>+int16_t WebRtcIsac_SetMaxPayloadSize(ISACStruct* ISAC_main_inst,
>+                                     int16_t maxPayloadBytes);
>+
>+/******************************************************************************
>+ * WebRtcIsac_SetMaxRate(...)
>+ *
>+ * This function sets the maximum rate which the codec may not exceed for
>+ * any signal packet. The maximum rate is defined and payload-size per
>+ * frame-size in bits per second.
>+ *
>+ * The codec has a maximum rate of 53400 bits per second (200 bytes per 30
>+ * ms) if the encoder sampling rate is 16kHz, and 160 kbps (600 bytes/30 ms)
>+ * if the encoder sampling rate is 32 kHz.
>+ *
>+ * It is possible to set a maximum rate between 32000 and 53400 bits/sec
>+ * in wideband mode, and 32000 to 160000 bits/sec in super-wideband mode.
>+ *
>+ * If an out of range limit is used, the function returns -1, but the closest
>+ * valid value will be applied.
>+ *
>+ * ---------------
>+ * IMPORTANT NOTES
>+ * ---------------
>+ * The size of a packet is limited to the minimum of 'max-payload-size' and
>+ * 'max-rate.' For instance, let's assume the max-payload-size is set to
>+ * 170 bytes, and max-rate is set to 40 kbps. Note that a limit of 40 kbps
>+ * translates to 150 bytes for 30ms frame-size & 300 bytes for 60ms
>+ * frame-size. Then a packet with a frame-size of 30 ms is limited to 150,
>+ * i.e. min(170, 150), and a packet with 60 ms frame-size is limited to
>+ * 170 bytes, min(170, 300).
>+ *
>+ * Input:
>+ *        - ISAC_main_inst    : iSAC instance
>+ *        - maxRate           : maximum rate in bits per second,
>+ *                              valid values are 32000 to 53400 bits/sec in
>+ *                              wideband mode, and 32000 to 160000 bits/sec in
>+ *                              super-wideband mode.
>+ *
>+ * Return value               : 0 if successful
>+ *                             -1 if error happens
>+ */
>+
>+int16_t WebRtcIsac_SetMaxRate(ISACStruct* ISAC_main_inst, int32_t maxRate);
>+
>+/******************************************************************************
>+ * WebRtcIsac_DecSampRate()
>+ * Return the sampling rate of the decoded audio.
>+ *
>+ * Input:
>+ *        - ISAC_main_inst    : iSAC instance
>+ *
>+ * Return value               : sampling frequency in Hertz.
>+ *
>+ */
>+
>+uint16_t WebRtcIsac_DecSampRate(ISACStruct* ISAC_main_inst);
>+
>+/******************************************************************************
>+ * WebRtcIsac_EncSampRate()
>+ *
>+ * Input:
>+ *        - ISAC_main_inst    : iSAC instance
>+ *
>+ * Return value               : sampling rate in Hertz.
>+ *
>+ */
>+
>+uint16_t WebRtcIsac_EncSampRate(ISACStruct* ISAC_main_inst);
>+
>+/******************************************************************************
>+ * WebRtcIsac_SetDecSampRate()
>+ * Set the sampling rate of the decoder.  Initialization of the decoder WILL
>+ * NOT overwrite the sampling rate of the encoder. The default value is 16 kHz
>+ * which is set when the instance is created.
>+ *
>+ * Input:
>+ *        - ISAC_main_inst    : iSAC instance
>+ *        - sampRate          : sampling rate in Hertz.
>+ *
>+ * Return value               : 0 if successful
>+ *                             -1 if failed.
>+ */
>+
>+int16_t WebRtcIsac_SetDecSampRate(ISACStruct* ISAC_main_inst,
>+                                  uint16_t samp_rate_hz);
>+
>+/******************************************************************************
>+ * WebRtcIsac_SetEncSampRate()
>+ * Set the sampling rate of the encoder. Initialization of the encoder WILL
>+ * NOT overwrite the sampling rate of the encoder. The default value is 16 kHz
>+ * which is set when the instance is created. The encoding-mode and the
>+ * bottleneck remain unchanged by this call, however, the maximum rate and
>+ * maximum payload-size will reset to their default value.
>+ *
>+ * Input:
>+ *        - ISAC_main_inst    : iSAC instance
>+ *        - sampRate          : sampling rate in Hertz.
>+ *
>+ * Return value               : 0 if successful
>+ *                             -1 if failed.
>+ */
>+
>+int16_t WebRtcIsac_SetEncSampRate(ISACStruct* ISAC_main_inst,
>+                                  uint16_t sample_rate_hz);
>+
>+/******************************************************************************
>+ * WebRtcIsac_GetNewBitStream(...)
>+ *
>+ * This function returns encoded data, with the recieved bwe-index in the
>+ * stream. If the rate is set to a value less than bottleneck of codec
>+ * the new bistream will be re-encoded with the given target rate.
>+ * It should always return a complete packet, i.e. only called once
>+ * even for 60 msec frames.
>+ *
>+ * NOTE 1! This function does not write in the ISACStruct, it is not allowed.
>+ * NOTE 2! Currently not implemented for SWB mode.
>+ * NOTE 3! Rates larger than the bottleneck of the codec will be limited
>+ *         to the current bottleneck.
>+ *
>+ * Input:
>+ *        - ISAC_main_inst    : ISAC instance.
>+ *        - bweIndex          : Index of bandwidth estimate to put in new
>+ *                              bitstream
>+ *        - rate              : target rate of the transcoder is bits/sec.
>+ *                              Valid values are the accepted rate in iSAC,
>+ *                              i.e. 10000 to 56000.
>+ *        - isRCU                       : if the new bit-stream is an RCU
>+ * stream. Note that the rate parameter always indicates the target rate of the
>+ * main payload, regardless of 'isRCU' value.
>+ *
>+ * Output:
>+ *        - encoded           : The encoded data vector
>+ *
>+ * Return value               : >0 - Length (in bytes) of coded data
>+ *                              -1 - Error  or called in SWB mode
>+ *                                 NOTE! No error code is written to
>+ *                                 the struct since it is only allowed to read
>+ *                                 the struct.
>+ */
>+int16_t WebRtcIsac_GetNewBitStream(ISACStruct* ISAC_main_inst,
>+                                   int16_t bweIndex,
>+                                   int16_t jitterInfo,
>+                                   int32_t rate,
>+                                   uint8_t* encoded,
>+                                   int16_t isRCU);
>+
>+/****************************************************************************
>+ * WebRtcIsac_GetDownLinkBwIndex(...)
>+ *
>+ * This function returns index representing the Bandwidth estimate from
>+ * other side to this side.
>+ *
>+ * Input:
>+ *        - ISAC_main_inst    : iSAC struct
>+ *
>+ * Output:
>+ *        - bweIndex          : Bandwidth estimate to transmit to other side.
>+ *
>+ */
>+
>+int16_t WebRtcIsac_GetDownLinkBwIndex(ISACStruct* ISAC_main_inst,
>+                                      int16_t* bweIndex,
>+                                      int16_t* jitterInfo);
>+
>+/****************************************************************************
>+ * WebRtcIsac_UpdateUplinkBw(...)
>+ *
>+ * This function takes an index representing the Bandwidth estimate from
>+ * this side to other side and updates BWE.
>+ *
>+ * Input:
>+ *        - ISAC_main_inst    : iSAC struct
>+ *        - bweIndex          : Bandwidth estimate from other side.
>+ *
>+ */
>+
>+int16_t WebRtcIsac_UpdateUplinkBw(ISACStruct* ISAC_main_inst, int16_t bweIndex);
>+
>+/****************************************************************************
>+ * WebRtcIsac_ReadBwIndex(...)
>+ *
>+ * This function returns the index of the Bandwidth estimate from the bitstream.
>+ *
>+ * Input:
>+ *        - encoded           : Encoded bitstream
>+ *
>+ * Output:
>+ *        - frameLength       : Length of frame in packet (in samples)
>+ *        - bweIndex         : Bandwidth estimate in bitstream
>+ *
>+ */
>+
>+int16_t WebRtcIsac_ReadBwIndex(const uint8_t* encoded, int16_t* bweIndex);
>+
>+/*******************************************************************************
>+ * WebRtcIsac_GetNewFrameLen(...)
>+ *
>+ * returns the frame lenght (in samples) of the next packet. In the case of
>+ * channel-adaptive mode, iSAC decides on its frame lenght based on the
>+ * estimated bottleneck this allows a user to prepare for the next packet (at
>+ * the encoder)
>+ *
>+ * The primary usage is in CE to make the iSAC works in channel-adaptive mode
>+ *
>+ * Input:
>+ *        - ISAC_main_inst     : iSAC struct
>+ *
>+ * Return Value                : frame lenght in samples
>+ *
>+ */
>+
>+int16_t WebRtcIsac_GetNewFrameLen(ISACStruct* ISAC_main_inst);
>+
>+/****************************************************************************
>+ *  WebRtcIsac_GetRedPayload(...)
>+ *
>+ *  Populates "encoded" with the redundant payload of the recently encoded
>+ *  frame. This function has to be called once that WebRtcIsac_Encode(...)
>+ *  returns a positive value. Regardless of the frame-size this function will
>+ *  be called only once after encoding is completed.
>+ *
>+ * Input:
>+ *      - ISAC_main_inst    : iSAC struct
>+ *
>+ * Output:
>+ *        - encoded            : the encoded data vector
>+ *
>+ *
>+ * Return value:
>+ *                              : >0 - Length (in bytes) of coded data
>+ *                              : -1 - Error
>+ *
>+ *
>+ */
>+int16_t WebRtcIsac_GetRedPayload(ISACStruct* ISAC_main_inst, uint8_t* encoded);
>+
>+/****************************************************************************
>+ * WebRtcIsac_DecodeRcu(...)
>+ *
>+ * This function decodes a redundant (RCU) iSAC frame. Function is called in
>+ * NetEq with a stored RCU payload i case of packet loss. Output speech length
>+ * will be a multiple of 480 samples: 480 or 960 samples,
>+ * depending on the framesize (30 or 60 ms).
>+ *
>+ * Input:
>+ *      - ISAC_main_inst     : ISAC instance.
>+ *      - encoded            : encoded ISAC RCU frame(s)
>+ *      - len                : bytes in encoded vector
>+ *
>+ * Output:
>+ *      - decoded            : The decoded vector
>+ *
>+ * Return value              : >0 - number of samples in decoded vector
>+ *                             -1 - Error
>+ */
>+int WebRtcIsac_DecodeRcu(ISACStruct* ISAC_main_inst,
>+                         const uint8_t* encoded,
>+                         size_t len,
>+                         int16_t* decoded,
>+                         int16_t* speechType);
>+
>+/* Fills in an IsacBandwidthInfo struct. |inst| should be a decoder. */
>+void WebRtcIsac_GetBandwidthInfo(ISACStruct* inst, IsacBandwidthInfo* bwinfo);
>+
>+/* Uses the values from an IsacBandwidthInfo struct. |inst| should be an
>+   encoder. */
>+void WebRtcIsac_SetBandwidthInfo(ISACStruct* inst,
>+                                 const IsacBandwidthInfo* bwinfo);
>+
>+/* If |inst| is a decoder but not an encoder: tell it what sample rate the
>+   encoder is using, for bandwidth estimation purposes. */
>+void WebRtcIsac_SetEncSampRateInDecoder(ISACStruct* inst, int sample_rate_hz);
> 
> #if defined(__cplusplus)
> }
> #endif
> 
>-
>-
> #endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INCLUDE_ISAC_H_ */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/arith_routines.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/arith_routines.c
>index 5c901bba005..9d5c6930b1b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/arith_routines.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/arith_routines.c
>@@ -8,8 +8,8 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "arith_routines.h"
>-#include "settings.h"
>+#include "modules/audio_coding/codecs/isac/main/source/arith_routines.h"
>+#include "modules/audio_coding/codecs/isac/main/source/settings.h"
> 
> 
> /*
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/arith_routines.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/arith_routines.h
>index 73d0c2306e7..6e7ea1da5ed 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/arith_routines.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/arith_routines.h
>@@ -18,46 +18,50 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ARITH_ROUTINES_H_
> #define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ARITH_ROUTINES_H_
> 
>-#include "structs.h"
>-
>+#include "modules/audio_coding/codecs/isac/main/source/structs.h"
> 
> int WebRtcIsac_EncLogisticMulti2(
>-    Bitstr *streamdata,              /* in-/output struct containing bitstream */
>-    int16_t *dataQ7,           /* input: data vector */
>-    const uint16_t *env,       /* input: side info vector defining the width of the pdf */
>-    const int N,                     /* input: data vector length */
>+    Bitstr* streamdata, /* in-/output struct containing bitstream */
>+    int16_t* dataQ7,    /* input: data vector */
>+    const uint16_t*
>+        env,     /* input: side info vector defining the width of the pdf */
>+    const int N, /* input: data vector length */
>     const int16_t isSWB12kHz); /* if the codec is working in 12kHz bandwidth */
> 
> /* returns the number of bytes in the stream */
>-int WebRtcIsac_EncTerminate(Bitstr *streamdata); /* in-/output struct containing bitstream */
>+int WebRtcIsac_EncTerminate(
>+    Bitstr* streamdata); /* in-/output struct containing bitstream */
> 
> /* returns the number of bytes in the stream so far */
> int WebRtcIsac_DecLogisticMulti2(
>-    int16_t *data,             /* output: data vector */
>-    Bitstr *streamdata,              /* in-/output struct containing bitstream */
>-    const uint16_t *env,       /* input: side info vector defining the width of the pdf */
>-    const int16_t *dither,     /* input: dither vector */
>-    const int N,                     /* input: data vector length */
>+    int16_t* data,      /* output: data vector */
>+    Bitstr* streamdata, /* in-/output struct containing bitstream */
>+    const uint16_t*
>+        env, /* input: side info vector defining the width of the pdf */
>+    const int16_t* dither,     /* input: dither vector */
>+    const int N,               /* input: data vector length */
>     const int16_t isSWB12kHz); /* if the codec is working in 12kHz bandwidth */
> 
> void WebRtcIsac_EncHistMulti(
>-    Bitstr *streamdata,         /* in-/output struct containing bitstream */
>-    const int *data,            /* input: data vector */
>-    const uint16_t *const *cdf, /* input: array of cdf arrays */
>+    Bitstr* streamdata,         /* in-/output struct containing bitstream */
>+    const int* data,            /* input: data vector */
>+    const uint16_t* const* cdf, /* input: array of cdf arrays */
>     const int N);               /* input: data vector length */
> 
> int WebRtcIsac_DecHistBisectMulti(
>-    int *data,                      /* output: data vector */
>-    Bitstr *streamdata,             /* in-/output struct containing bitstream */
>-    const uint16_t *const *cdf,     /* input: array of cdf arrays */
>-    const uint16_t *cdf_size, /* input: array of cdf table sizes+1 (power of two: 2^k) */
>-    const int N);                   /* input: data vector length */
>+    int* data,                  /* output: data vector */
>+    Bitstr* streamdata,         /* in-/output struct containing bitstream */
>+    const uint16_t* const* cdf, /* input: array of cdf arrays */
>+    const uint16_t*
>+        cdf_size, /* input: array of cdf table sizes+1 (power of two: 2^k) */
>+    const int N); /* input: data vector length */
> 
> int WebRtcIsac_DecHistOneStepMulti(
>-    int *data,                       /* output: data vector */
>-    Bitstr *streamdata,              /* in-/output struct containing bitstream */
>-    const uint16_t *const *cdf,      /* input: array of cdf arrays */
>-    const uint16_t *init_index,/* input: vector of initial cdf table search entries */
>-    const int N);                    /* input: data vector length */
>+    int* data,                  /* output: data vector */
>+    Bitstr* streamdata,         /* in-/output struct containing bitstream */
>+    const uint16_t* const* cdf, /* input: array of cdf arrays */
>+    const uint16_t*
>+        init_index, /* input: vector of initial cdf table search entries */
>+    const int N);   /* input: data vector length */
> 
> #endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ARITH_ROUTINES_H_ */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/arith_routines_hist.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/arith_routines_hist.c
>index b229c679fc0..e948979fd74 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/arith_routines_hist.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/arith_routines_hist.c
>@@ -8,8 +8,8 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "settings.h"
>-#include "arith_routines.h"
>+#include "modules/audio_coding/codecs/isac/main/source/settings.h"
>+#include "modules/audio_coding/codecs/isac/main/source/arith_routines.h"
> 
> 
> /*
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/arith_routines_logist.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/arith_routines_logist.c
>index a463d4853a2..777780f54fa 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/arith_routines_logist.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/arith_routines_logist.c
>@@ -17,7 +17,7 @@
>  */
> 
> 
>-#include "arith_routines.h"
>+#include "modules/audio_coding/codecs/isac/main/source/arith_routines.h"
> 
> 
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/audio_encoder_isac_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/audio_encoder_isac_unittest.cc
>index 333ab52c00f..87ae0e0bcb9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/audio_encoder_isac_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/audio_encoder_isac_unittest.cc
>@@ -29,7 +29,11 @@ void TestGoodConfig(const AudioEncoderIsacFloatImpl::Config& config) {
> // Wrap subroutine calls that test things in this, so that the error messages
> // will be accompanied by stack traces that make it possible to tell which
> // subroutine invocation caused the failure.
>-#define S(x) do { SCOPED_TRACE(#x); x; } while (0)
>+#define S(x)          \
>+  do {                \
>+    SCOPED_TRACE(#x); \
>+    x;                \
>+  } while (0)
> 
> }  // namespace
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.c
>index 4a16932341b..673f10d446a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.c
>@@ -16,14 +16,14 @@
>  *
>  */
> 
>-#include "bandwidth_estimator.h"
>-#include "settings.h"
>-#include "isac.h"
>-#include "rtc_base/checks.h"
>-
> #include <math.h>
> #include <string.h>
> 
>+#include "modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.h"
>+#include "modules/audio_coding/codecs/isac/main/source/settings.h"
>+#include "modules/audio_coding/codecs/isac/main/include/isac.h"
>+#include "rtc_base/checks.h"
>+
> /* array of quantization levels for bottle neck info; Matlab code: */
> /* sprintf('%4.1ff, ', logspace(log10(5000), log10(40000), 12)) */
> static const float kQRateTableWb[12] =
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.h
>index ddebdcd3420..d80ff73bf5d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.h
>@@ -19,166 +19,156 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_BANDWIDTH_ESTIMATOR_H_
> #define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_BANDWIDTH_ESTIMATOR_H_
> 
>-#include "structs.h"
>-#include "settings.h"
>+#include <stddef.h>
> 
>+#include "modules/audio_coding/codecs/isac/main/source/settings.h"
>+#include "modules/audio_coding/codecs/isac/main/source/structs.h"
> 
>-#define MIN_ISAC_BW     10000
>-#define MIN_ISAC_BW_LB  10000
>-#define MIN_ISAC_BW_UB  25000
>+#define MIN_ISAC_BW 10000
>+#define MIN_ISAC_BW_LB 10000
>+#define MIN_ISAC_BW_UB 25000
> 
>-#define MAX_ISAC_BW     56000
>-#define MAX_ISAC_BW_UB  32000
>-#define MAX_ISAC_BW_LB  32000
>+#define MAX_ISAC_BW 56000
>+#define MAX_ISAC_BW_UB 32000
>+#define MAX_ISAC_BW_LB 32000
> 
>-#define MIN_ISAC_MD     5
>-#define MAX_ISAC_MD     25
>+#define MIN_ISAC_MD 5
>+#define MAX_ISAC_MD 25
> 
> // assumed header size, in bytes; we don't know the exact number
> // (header compression may be used)
>-#define HEADER_SIZE        35
>+#define HEADER_SIZE 35
> 
> // Initial Frame-Size, in ms, for Wideband & Super-Wideband Mode
>-#define INIT_FRAME_LEN_WB  60
>+#define INIT_FRAME_LEN_WB 60
> #define INIT_FRAME_LEN_SWB 30
> 
> // Initial Bottleneck Estimate, in bits/sec, for
> // Wideband & Super-wideband mode
>-#define INIT_BN_EST_WB     20e3f
>-#define INIT_BN_EST_SWB    56e3f
>+#define INIT_BN_EST_WB 20e3f
>+#define INIT_BN_EST_SWB 56e3f
> 
> // Initial Header rate (header rate depends on frame-size),
> // in bits/sec, for Wideband & Super-Wideband mode.
>-#define INIT_HDR_RATE_WB                                                \
>+#define INIT_HDR_RATE_WB \
>   ((float)HEADER_SIZE * 8.0f * 1000.0f / (float)INIT_FRAME_LEN_WB)
>-#define INIT_HDR_RATE_SWB                                               \
>+#define INIT_HDR_RATE_SWB \
>   ((float)HEADER_SIZE * 8.0f * 1000.0f / (float)INIT_FRAME_LEN_SWB)
> 
> // number of packets in a row for a high rate burst
>-#define BURST_LEN       3
>+#define BURST_LEN 3
> 
> // ms, max time between two full bursts
>-#define BURST_INTERVAL  500
>+#define BURST_INTERVAL 500
> 
> // number of packets in a row for initial high rate burst
>-#define INIT_BURST_LEN  5
>+#define INIT_BURST_LEN 5
> 
> // bits/s, rate for the first BURST_LEN packets
>-#define INIT_RATE_WB       INIT_BN_EST_WB
>-#define INIT_RATE_SWB      INIT_BN_EST_SWB
>-
>+#define INIT_RATE_WB INIT_BN_EST_WB
>+#define INIT_RATE_SWB INIT_BN_EST_SWB
> 
> #if defined(__cplusplus)
> extern "C" {
> #endif
> 
>-  /* This function initializes the struct                    */
>-  /* to be called before using the struct for anything else  */
>-  /* returns 0 if everything went fine, -1 otherwise         */
>-  int32_t WebRtcIsac_InitBandwidthEstimator(
>-      BwEstimatorstr*           bwest_str,
>-      enum IsacSamplingRate encoderSampRate,
>-      enum IsacSamplingRate decoderSampRate);
>-
>-  /* This function updates the receiving estimate                                                      */
>-  /* Parameters:                                                                                       */
>-  /* rtp_number    - value from RTP packet, from NetEq                                                 */
>-  /* frame length  - length of signal frame in ms, from iSAC decoder                                   */
>-  /* send_ts       - value in RTP header giving send time in samples                                   */
>-  /* arr_ts        - value given by timeGetTime() time of arrival in samples of packet from NetEq      */
>-  /* pksize        - size of packet in bytes, from NetEq                                               */
>-  /* Index         - integer (range 0...23) indicating bottle neck & jitter as estimated by other side */
>-  /* returns 0 if everything went fine, -1 otherwise                                                   */
>-  int16_t WebRtcIsac_UpdateBandwidthEstimator(
>-      BwEstimatorstr* bwest_str,
>-      const uint16_t rtp_number,
>-      const int32_t frame_length,
>-      const uint32_t send_ts,
>-      const uint32_t arr_ts,
>-      const size_t pksize);
>-
>-  /* Update receiving estimates. Used when we only receive BWE index, no iSAC data packet. */
>-  int16_t WebRtcIsac_UpdateUplinkBwImpl(
>-      BwEstimatorstr*           bwest_str,
>-      int16_t               Index,
>-      enum IsacSamplingRate encoderSamplingFreq);
>-
>-  /* Returns the bandwidth/jitter estimation code (integer 0...23) to put in the sending iSAC payload */
>-  void WebRtcIsac_GetDownlinkBwJitIndexImpl(
>-      BwEstimatorstr* bwest_str,
>-      int16_t* bottleneckIndex,
>-      int16_t* jitterInfo,
>-      enum IsacSamplingRate decoderSamplingFreq);
>-
>-  /* Returns the bandwidth estimation (in bps) */
>-  int32_t WebRtcIsac_GetDownlinkBandwidth(
>-      const BwEstimatorstr *bwest_str);
>-
>-  /* Returns the max delay (in ms) */
>-  int32_t WebRtcIsac_GetDownlinkMaxDelay(
>-      const BwEstimatorstr *bwest_str);
>-
>-  /* Returns the bandwidth that iSAC should send with in bps */
>-  int32_t WebRtcIsac_GetUplinkBandwidth(const BwEstimatorstr* bwest_str);
>-
>-  /* Returns the max delay value from the other side in ms */
>-  int32_t WebRtcIsac_GetUplinkMaxDelay(
>-      const BwEstimatorstr *bwest_str);
>-
>-  /* Fills in an IsacExternalBandwidthInfo struct. */
>-  void WebRtcIsacBw_GetBandwidthInfo(
>-      BwEstimatorstr* bwest_str,
>-      enum IsacSamplingRate decoder_sample_rate_hz,
>-      IsacBandwidthInfo* bwinfo);
>-
>-  /* Uses the values from an IsacExternalBandwidthInfo struct. */
>-  void WebRtcIsacBw_SetBandwidthInfo(BwEstimatorstr* bwest_str,
>-                                     const IsacBandwidthInfo* bwinfo);
>-
>-  /*
>-   * update amount of data in bottle neck buffer and burst handling
>-   * returns minimum payload size (bytes)
>-   */
>-  int WebRtcIsac_GetMinBytes(
>-      RateModel*         State,
>-      int                StreamSize,    /* bytes in bitstream */
>-      const int          FrameLen,      /* ms per frame */
>-      const double       BottleNeck,    /* bottle neck rate; excl headers (bps) */
>-      const double       DelayBuildUp,  /* max delay from bottleneck buffering (ms) */
>-      enum ISACBandwidth bandwidth
>-      /*,int16_t        frequentLargePackets*/);
>-
>-  /*
>-   * update long-term average bitrate and amount of data in buffer
>-   */
>-  void WebRtcIsac_UpdateRateModel(
>-      RateModel*   State,
>-      int          StreamSize,                /* bytes in bitstream */
>-      const int    FrameSamples,        /* samples per frame */
>-      const double BottleNeck);       /* bottle neck rate; excl headers (bps) */
>-
>-
>-  void WebRtcIsac_InitRateModel(
>-      RateModel *State);
>-
>-  /* Returns the new framelength value (input argument: bottle_neck) */
>-  int WebRtcIsac_GetNewFrameLength(
>-      double bottle_neck,
>-      int    current_framelength);
>-
>-  /* Returns the new SNR value (input argument: bottle_neck) */
>-  double WebRtcIsac_GetSnr(
>-      double bottle_neck,
>-      int    new_framelength);
>-
>-
>-  int16_t WebRtcIsac_UpdateUplinkJitter(
>-      BwEstimatorstr*              bwest_str,
>-      int32_t                  index);
>+/* This function initializes the struct                    */
>+/* to be called before using the struct for anything else  */
>+/* returns 0 if everything went fine, -1 otherwise         */
>+int32_t WebRtcIsac_InitBandwidthEstimator(
>+    BwEstimatorstr* bwest_str,
>+    enum IsacSamplingRate encoderSampRate,
>+    enum IsacSamplingRate decoderSampRate);
>+
>+/* This function updates the receiving estimate */
>+/* Parameters: */
>+/* rtp_number    - value from RTP packet, from NetEq */
>+/* frame length  - length of signal frame in ms, from iSAC decoder */
>+/* send_ts       - value in RTP header giving send time in samples */
>+/* arr_ts        - value given by timeGetTime() time of arrival in samples of
>+ * packet from NetEq      */
>+/* pksize        - size of packet in bytes, from NetEq */
>+/* Index         - integer (range 0...23) indicating bottle neck & jitter as
>+ * estimated by other side */
>+/* returns 0 if everything went fine, -1 otherwise */
>+int16_t WebRtcIsac_UpdateBandwidthEstimator(BwEstimatorstr* bwest_str,
>+                                            const uint16_t rtp_number,
>+                                            const int32_t frame_length,
>+                                            const uint32_t send_ts,
>+                                            const uint32_t arr_ts,
>+                                            const size_t pksize);
>+
>+/* Update receiving estimates. Used when we only receive BWE index, no iSAC data
>+ * packet. */
>+int16_t WebRtcIsac_UpdateUplinkBwImpl(
>+    BwEstimatorstr* bwest_str,
>+    int16_t Index,
>+    enum IsacSamplingRate encoderSamplingFreq);
>+
>+/* Returns the bandwidth/jitter estimation code (integer 0...23) to put in the
>+ * sending iSAC payload */
>+void WebRtcIsac_GetDownlinkBwJitIndexImpl(
>+    BwEstimatorstr* bwest_str,
>+    int16_t* bottleneckIndex,
>+    int16_t* jitterInfo,
>+    enum IsacSamplingRate decoderSamplingFreq);
>+
>+/* Returns the bandwidth estimation (in bps) */
>+int32_t WebRtcIsac_GetDownlinkBandwidth(const BwEstimatorstr* bwest_str);
>+
>+/* Returns the max delay (in ms) */
>+int32_t WebRtcIsac_GetDownlinkMaxDelay(const BwEstimatorstr* bwest_str);
>+
>+/* Returns the bandwidth that iSAC should send with in bps */
>+int32_t WebRtcIsac_GetUplinkBandwidth(const BwEstimatorstr* bwest_str);
>+
>+/* Returns the max delay value from the other side in ms */
>+int32_t WebRtcIsac_GetUplinkMaxDelay(const BwEstimatorstr* bwest_str);
>+
>+/* Fills in an IsacExternalBandwidthInfo struct. */
>+void WebRtcIsacBw_GetBandwidthInfo(BwEstimatorstr* bwest_str,
>+                                   enum IsacSamplingRate decoder_sample_rate_hz,
>+                                   IsacBandwidthInfo* bwinfo);
>+
>+/* Uses the values from an IsacExternalBandwidthInfo struct. */
>+void WebRtcIsacBw_SetBandwidthInfo(BwEstimatorstr* bwest_str,
>+                                   const IsacBandwidthInfo* bwinfo);
>+
>+/*
>+ * update amount of data in bottle neck buffer and burst handling
>+ * returns minimum payload size (bytes)
>+ */
>+int WebRtcIsac_GetMinBytes(
>+    RateModel* State,
>+    int StreamSize,            /* bytes in bitstream */
>+    const int FrameLen,        /* ms per frame */
>+    const double BottleNeck,   /* bottle neck rate; excl headers (bps) */
>+    const double DelayBuildUp, /* max delay from bottleneck buffering (ms) */
>+    enum ISACBandwidth bandwidth
>+    /*,int16_t        frequentLargePackets*/);
>+
>+/*
>+ * update long-term average bitrate and amount of data in buffer
>+ */
>+void WebRtcIsac_UpdateRateModel(
>+    RateModel* State,
>+    int StreamSize,           /* bytes in bitstream */
>+    const int FrameSamples,   /* samples per frame */
>+    const double BottleNeck); /* bottle neck rate; excl headers (bps) */
>+
>+void WebRtcIsac_InitRateModel(RateModel* State);
>+
>+/* Returns the new framelength value (input argument: bottle_neck) */
>+int WebRtcIsac_GetNewFrameLength(double bottle_neck, int current_framelength);
>+
>+/* Returns the new SNR value (input argument: bottle_neck) */
>+double WebRtcIsac_GetSnr(double bottle_neck, int new_framelength);
>+
>+int16_t WebRtcIsac_UpdateUplinkJitter(BwEstimatorstr* bwest_str, int32_t index);
> 
> #if defined(__cplusplus)
> }
> #endif
> 
>-
>-#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_BANDWIDTH_ESTIMATOR_H_ */
>+#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_BANDWIDTH_ESTIMATOR_H_ \
>+        */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/codec.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/codec.h
>index d938725b921..a7c7ddc14a6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/codec.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/codec.h
>@@ -19,15 +19,19 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CODEC_H_
> #define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CODEC_H_
> 
>-#include "structs.h"
>+#include <stddef.h>
> 
>+#include "modules/audio_coding/codecs/isac/main/source/structs.h"
>+#include "modules/third_party/fft/fft.h"
> 
> void WebRtcIsac_ResetBitstream(Bitstr* bit_stream);
> 
>-int WebRtcIsac_EstimateBandwidth(BwEstimatorstr* bwest_str, Bitstr* streamdata,
>+int WebRtcIsac_EstimateBandwidth(BwEstimatorstr* bwest_str,
>+                                 Bitstr* streamdata,
>                                  size_t packet_size,
>                                  uint16_t rtp_seq_number,
>-                                 uint32_t send_ts, uint32_t arr_ts,
>+                                 uint32_t send_ts,
>+                                 uint32_t arr_ts,
>                                  enum IsacSamplingRate encoderSampRate,
>                                  enum IsacSamplingRate decoderSampRate);
> 
>@@ -37,7 +41,8 @@ int WebRtcIsac_DecodeLb(const TransformTables* transform_tables,
>                         int16_t* current_framesamples,
>                         int16_t isRCUPayload);
> 
>-int WebRtcIsac_DecodeRcuLb(float* signal_out, ISACLBDecStruct* ISACdec_obj,
>+int WebRtcIsac_DecodeRcuLb(float* signal_out,
>+                           ISACLBDecStruct* ISACdec_obj,
>                            int16_t* current_framesamples);
> 
> int WebRtcIsac_EncodeLb(const TransformTables* transform_tables,
>@@ -47,15 +52,20 @@ int WebRtcIsac_EncodeLb(const TransformTables* transform_tables,
>                         int16_t bottleneckIndex);
> 
> int WebRtcIsac_EncodeStoredDataLb(const IsacSaveEncoderData* ISACSavedEnc_obj,
>-                                  Bitstr* ISACBitStr_obj, int BWnumber,
>+                                  Bitstr* ISACBitStr_obj,
>+                                  int BWnumber,
>                                   float scale);
> 
> int WebRtcIsac_EncodeStoredDataUb(
>-    const ISACUBSaveEncDataStruct* ISACSavedEnc_obj, Bitstr* bitStream,
>-    int32_t jitterInfo, float scale, enum ISACBandwidth bandwidth);
>+    const ISACUBSaveEncDataStruct* ISACSavedEnc_obj,
>+    Bitstr* bitStream,
>+    int32_t jitterInfo,
>+    float scale,
>+    enum ISACBandwidth bandwidth);
> 
> int16_t WebRtcIsac_GetRedPayloadUb(
>-    const ISACUBSaveEncDataStruct* ISACSavedEncObj, Bitstr* bitStreamObj,
>+    const ISACUBSaveEncDataStruct* ISACSavedEncObj,
>+    Bitstr* bitStreamObj,
>     enum ISACBandwidth bandwidth);
> 
> /******************************************************************************
>@@ -81,7 +91,6 @@ int16_t WebRtcIsac_RateAllocation(int32_t inRateBitPerSec,
>                                   double* rateUBBitPerSec,
>                                   enum ISACBandwidth* bandwidthKHz);
> 
>-
> /******************************************************************************
>  * WebRtcIsac_DecodeUb16()
>  *
>@@ -166,15 +175,8 @@ int WebRtcIsac_EncodeUb12(const TransformTables* transform_tables,
> 
> void WebRtcIsac_InitMasking(MaskFiltstr* maskdata);
> 
>-void WebRtcIsac_InitPreFilterbank(PreFiltBankstr* prefiltdata);
>-
> void WebRtcIsac_InitPostFilterbank(PostFiltBankstr* postfiltdata);
> 
>-void WebRtcIsac_InitPitchFilter(PitchFiltstr* pitchfiltdata);
>-
>-void WebRtcIsac_InitPitchAnalysis(PitchAnalysisStruct* State);
>-
>-
> /**************************** transform functions ****************************/
> 
> void WebRtcIsac_InitTransform(TransformTables* tables);
>@@ -193,41 +195,29 @@ void WebRtcIsac_Spec2time(const TransformTables* tables,
>                           double* outre2,
>                           FFTstr* fftstr_obj);
> 
>-/******************************* filter functions ****************************/
>-
>-void WebRtcIsac_AllPoleFilter(double* InOut, double* Coef, size_t lengthInOut,
>-                              int orderCoef);
>-
>-void WebRtcIsac_AllZeroFilter(double* In, double* Coef, size_t lengthInOut,
>-                              int orderCoef, double* Out);
>-
>-void WebRtcIsac_ZeroPoleFilter(double* In, double* ZeroCoef, double* PoleCoef,
>-                               size_t lengthInOut, int orderCoef, double* Out);
>-
>-
> /***************************** filterbank functions **************************/
> 
>-void WebRtcIsac_SplitAndFilterFloat(float* in, float* LP, float* HP,
>-                                    double* LP_la, double* HP_la,
>-                                    PreFiltBankstr* prefiltdata);
>-
>-
>-void WebRtcIsac_FilterAndCombineFloat(float* InLP, float* InHP, float* Out,
>+void WebRtcIsac_FilterAndCombineFloat(float* InLP,
>+                                      float* InHP,
>+                                      float* Out,
>                                       PostFiltBankstr* postfiltdata);
> 
>-
> /************************* normalized lattice filters ************************/
> 
>-void WebRtcIsac_NormLatticeFilterMa(int orderCoef, float* stateF, float* stateG,
>-                                    float* lat_in, double* filtcoeflo,
>+void WebRtcIsac_NormLatticeFilterMa(int orderCoef,
>+                                    float* stateF,
>+                                    float* stateG,
>+                                    float* lat_in,
>+                                    double* filtcoeflo,
>                                     double* lat_out);
> 
>-void WebRtcIsac_NormLatticeFilterAr(int orderCoef, float* stateF, float* stateG,
>-                                    double* lat_in, double* lo_filt_coef,
>+void WebRtcIsac_NormLatticeFilterAr(int orderCoef,
>+                                    float* stateF,
>+                                    float* stateG,
>+                                    double* lat_in,
>+                                    double* lo_filt_coef,
>                                     float* lat_out);
> 
> void WebRtcIsac_Dir2Lat(double* a, int orderCoef, float* sth, float* cth);
> 
>-void WebRtcIsac_AutoCorr(double* r, const double* x, size_t N, size_t order);
>-
> #endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CODEC_H_ */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/crc.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/crc.c
>index 2419e240339..1bb0827031c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/crc.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/crc.c
>@@ -8,9 +8,10 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "crc.h"
> #include <stdlib.h>
>-#include "signal_processing_library.h"
>+
>+#include "modules/audio_coding/codecs/isac/main/source/crc.h"
>+#include "common_audio/signal_processing/include/signal_processing_library.h"
> 
> #define POLYNOMIAL 0x04c11db7L
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/crc.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/crc.h
>index b3197a1a7db..f031019ed37 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/crc.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/crc.h
>@@ -18,7 +18,7 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CRC_H_
> #define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CRC_H_
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include <stdint.h>
> 
> /****************************************************************************
>  * WebRtcIsac_GetCrc(...)
>@@ -36,11 +36,6 @@
>  *                   -1 - Error
>  */
> 
>-int WebRtcIsac_GetCrc(
>-    const int16_t* encoded,
>-    int no_of_word8s,
>-    uint32_t* crc);
>-
>-
>+int WebRtcIsac_GetCrc(const int16_t* encoded, int no_of_word8s, uint32_t* crc);
> 
> #endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CRC_H_ */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/decode.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/decode.c
>index e925efba6e0..6e114e4a2be 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/decode.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/decode.c
>@@ -18,18 +18,17 @@
>  *
>  */
> 
>-
>-#include "codec.h"
>-#include "entropy_coding.h"
>-#include "pitch_estimator.h"
>-#include "bandwidth_estimator.h"
>-#include "structs.h"
>-#include "settings.h"
>-
> #include <stdio.h>
> #include <stdlib.h>
> #include <string.h>
> 
>+#include "modules/audio_coding/codecs/isac/main/source/codec.h"
>+#include "modules/audio_coding/codecs/isac/main/source/entropy_coding.h"
>+#include "modules/audio_coding/codecs/isac/main/source/pitch_estimator.h"
>+#include "modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.h"
>+#include "modules/audio_coding/codecs/isac/main/source/structs.h"
>+#include "modules/audio_coding/codecs/isac/main/source/settings.h"
>+#include "modules/audio_coding/codecs/isac/main/source/pitch_filter.h"
> 
> /*
>  * function to decode the bitstream
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/decode_bwe.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/decode_bwe.c
>index 019cc895288..89d970fc75e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/decode_bwe.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/decode_bwe.c
>@@ -8,10 +8,10 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "structs.h"
>-#include "bandwidth_estimator.h"
>-#include "entropy_coding.h"
>-#include "codec.h"
>+#include "modules/audio_coding/codecs/isac/main/source/structs.h"
>+#include "modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.h"
>+#include "modules/audio_coding/codecs/isac/main/source/entropy_coding.h"
>+#include "modules/audio_coding/codecs/isac/main/source/codec.h"
> 
> 
> int
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/encode.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/encode.c
>index 3f1912b6d37..bf92d02c533 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/encode.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/encode.c
>@@ -21,20 +21,22 @@
> #include <string.h>
> #include <stdio.h>
> 
>-#include "structs.h"
>-#include "codec.h"
>-#include "pitch_estimator.h"
>-#include "entropy_coding.h"
>-#include "arith_routines.h"
>-#include "pitch_gain_tables.h"
>-#include "pitch_lag_tables.h"
>-#include "spectrum_ar_model_tables.h"
>-#include "lpc_tables.h"
>-#include "lpc_analysis.h"
>-#include "bandwidth_estimator.h"
>-#include "lpc_shape_swb12_tables.h"
>-#include "lpc_shape_swb16_tables.h"
>-#include "lpc_gain_swb_tables.h"
>+#include "modules/audio_coding/codecs/isac/main/source/structs.h"
>+#include "modules/audio_coding/codecs/isac/main/source/codec.h"
>+#include "modules/audio_coding/codecs/isac/main/source/pitch_estimator.h"
>+#include "modules/audio_coding/codecs/isac/main/source/entropy_coding.h"
>+#include "modules/audio_coding/codecs/isac/main/source/arith_routines.h"
>+#include "modules/audio_coding/codecs/isac/main/source/pitch_gain_tables.h"
>+#include "modules/audio_coding/codecs/isac/main/source/pitch_lag_tables.h"
>+#include "modules/audio_coding/codecs/isac/main/source/spectrum_ar_model_tables.h"
>+#include "modules/audio_coding/codecs/isac/main/source/lpc_tables.h"
>+#include "modules/audio_coding/codecs/isac/main/source/lpc_analysis.h"
>+#include "modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.h"
>+#include "modules/audio_coding/codecs/isac/main/source/lpc_shape_swb12_tables.h"
>+#include "modules/audio_coding/codecs/isac/main/source/lpc_shape_swb16_tables.h"
>+#include "modules/audio_coding/codecs/isac/main/source/lpc_gain_swb_tables.h"
>+#include "modules/audio_coding/codecs/isac/main/source/isac_vad.h"
>+#include "modules/audio_coding/codecs/isac/main/source/pitch_filter.h"
> 
> 
> #define UB_LOOKAHEAD 24
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/encode_lpc_swb.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/encode_lpc_swb.c
>index 64cd4a75ec8..7b02e64a014 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/encode_lpc_swb.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/encode_lpc_swb.c
>@@ -16,17 +16,15 @@
>  *
>  */
> 
>-#include "encode_lpc_swb.h"
>-
> #include <math.h>
> #include <stdio.h>
> #include <string.h>
> 
>-#include "lpc_gain_swb_tables.h"
>-#include "lpc_shape_swb12_tables.h"
>-#include "lpc_shape_swb16_tables.h"
>-#include "settings.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include "modules/audio_coding/codecs/isac/main/source/encode_lpc_swb.h"
>+#include "modules/audio_coding/codecs/isac/main/source/lpc_gain_swb_tables.h"
>+#include "modules/audio_coding/codecs/isac/main/source/lpc_shape_swb12_tables.h"
>+#include "modules/audio_coding/codecs/isac/main/source/lpc_shape_swb16_tables.h"
>+#include "modules/audio_coding/codecs/isac/main/source/settings.h"
> 
> /******************************************************************************
>  * WebRtcIsac_RemoveLarMean()
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/encode_lpc_swb.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/encode_lpc_swb.h
>index 6609644b0c5..8bc3d752c30 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/encode_lpc_swb.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/encode_lpc_swb.h
>@@ -19,9 +19,8 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ENCODE_LPC_SWB_H_
> #define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ENCODE_LPC_SWB_H_
> 
>-#include "settings.h"
>-#include "structs.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include "modules/audio_coding/codecs/isac/main/source/settings.h"
>+#include "modules/audio_coding/codecs/isac/main/source/structs.h"
> 
> /******************************************************************************
>  * WebRtcIsac_RemoveLarMean()
>@@ -39,9 +38,7 @@
>  *
>  *
>  */
>-int16_t WebRtcIsac_RemoveLarMean(
>-    double*     lar,
>-    int16_t bandwidth);
>+int16_t WebRtcIsac_RemoveLarMean(double* lar, int16_t bandwidth);
> 
> /******************************************************************************
>  * WebRtcIsac_DecorrelateIntraVec()
>@@ -59,11 +56,9 @@ int16_t WebRtcIsac_RemoveLarMean(
>  * Output:
>  *      -out                : decorrelated LAR vectors.
>  */
>-int16_t WebRtcIsac_DecorrelateIntraVec(
>-    const double* inLAR,
>-    double*       out,
>-    int16_t   bandwidth);
>-
>+int16_t WebRtcIsac_DecorrelateIntraVec(const double* inLAR,
>+                                       double* out,
>+                                       int16_t bandwidth);
> 
> /******************************************************************************
>  * WebRtcIsac_DecorrelateInterVec()
>@@ -82,11 +77,9 @@ int16_t WebRtcIsac_DecorrelateIntraVec(
>  * Output:
>  *      -out                : decorrelated LAR vectors.
>  */
>-int16_t WebRtcIsac_DecorrelateInterVec(
>-    const double* data,
>-    double*       out,
>-    int16_t   bandwidth);
>-
>+int16_t WebRtcIsac_DecorrelateInterVec(const double* data,
>+                                       double* out,
>+                                       int16_t bandwidth);
> 
> /******************************************************************************
>  * WebRtcIsac_QuantizeUncorrLar()
>@@ -102,11 +95,7 @@ int16_t WebRtcIsac_DecorrelateInterVec(
>  *      -data               : quantized version of the input.
>  *      -idx                : pointer to quantization indices.
>  */
>-double WebRtcIsac_QuantizeUncorrLar(
>-    double*     data,
>-    int*        idx,
>-    int16_t bandwidth);
>-
>+double WebRtcIsac_QuantizeUncorrLar(double* data, int* idx, int16_t bandwidth);
> 
> /******************************************************************************
>  * WebRtcIsac_CorrelateIntraVec()
>@@ -121,11 +110,9 @@ double WebRtcIsac_QuantizeUncorrLar(
>  * Output:
>  *      -out                : correlated parametrs.
>  */
>-int16_t WebRtcIsac_CorrelateIntraVec(
>-    const double* data,
>-    double*       out,
>-    int16_t   bandwidth);
>-
>+int16_t WebRtcIsac_CorrelateIntraVec(const double* data,
>+                                     double* out,
>+                                     int16_t bandwidth);
> 
> /******************************************************************************
>  * WebRtcIsac_CorrelateInterVec()
>@@ -140,17 +127,15 @@ int16_t WebRtcIsac_CorrelateIntraVec(
>  * Output:
>  *      -out                : correlated parametrs.
>  */
>-int16_t WebRtcIsac_CorrelateInterVec(
>-    const double* data,
>-    double*       out,
>-    int16_t   bandwidth);
>-
>+int16_t WebRtcIsac_CorrelateInterVec(const double* data,
>+                                     double* out,
>+                                     int16_t bandwidth);
> 
> /******************************************************************************
>  * WebRtcIsac_AddLarMean()
>  *
>  * This is the inverse of WebRtcIsac_RemoveLarMean()
>- * 
>+ *
>  * Input:
>  *      -data               : pointer to mean-removed LAR:s.
>  *      -bandwidth          : indicates if the given LAR vectors belong
>@@ -159,10 +144,7 @@ int16_t WebRtcIsac_CorrelateInterVec(
>  * Output:
>  *      -data               : pointer to LARs.
>  */
>-int16_t WebRtcIsac_AddLarMean(
>-    double*     data,
>-    int16_t bandwidth);
>-
>+int16_t WebRtcIsac_AddLarMean(double* data, int16_t bandwidth);
> 
> /******************************************************************************
>  * WebRtcIsac_DequantizeLpcParam()
>@@ -177,11 +159,9 @@ int16_t WebRtcIsac_AddLarMean(
>  * Output:
>  *      -out                : pointer to quantized values.
>  */
>-int16_t WebRtcIsac_DequantizeLpcParam(
>-    const int*  idx,
>-    double*     out,
>-    int16_t bandwidth);
>-
>+int16_t WebRtcIsac_DequantizeLpcParam(const int* idx,
>+                                      double* out,
>+                                      int16_t bandwidth);
> 
> /******************************************************************************
>  * WebRtcIsac_ToLogDomainRemoveMean()
>@@ -194,9 +174,7 @@ int16_t WebRtcIsac_DequantizeLpcParam(
>  * Output:
>  *      -lpcGain            : mean-removed in log domain.
>  */
>-int16_t WebRtcIsac_ToLogDomainRemoveMean(
>-    double* lpGains);
>-
>+int16_t WebRtcIsac_ToLogDomainRemoveMean(double* lpGains);
> 
> /******************************************************************************
>  * WebRtcIsac_DecorrelateLPGain()
>@@ -210,16 +188,13 @@ int16_t WebRtcIsac_ToLogDomainRemoveMean(
>  * Output:
>  *      -out                : decorrelated parameters.
>  */
>-int16_t WebRtcIsac_DecorrelateLPGain(
>-    const double* data,
>-    double*       out);
>-
>+int16_t WebRtcIsac_DecorrelateLPGain(const double* data, double* out);
> 
> /******************************************************************************
>  * WebRtcIsac_QuantizeLpcGain()
>  *
>  * Quantize the decorrelated log-domain gains.
>- * 
>+ *
>  * Input:
>  *      -lpcGain            : uncorrelated LPC gains.
>  *
>@@ -227,10 +202,7 @@ int16_t WebRtcIsac_DecorrelateLPGain(
>  *      -idx                : quantization indices
>  *      -lpcGain            : quantized value of the inpt.
>  */
>-double WebRtcIsac_QuantizeLpcGain(
>-    double* lpGains,
>-    int*    idx);
>-
>+double WebRtcIsac_QuantizeLpcGain(double* lpGains, int* idx);
> 
> /******************************************************************************
>  * WebRtcIsac_DequantizeLpcGain()
>@@ -243,10 +215,7 @@ double WebRtcIsac_QuantizeLpcGain(
>  * Output:
>  *      -lpcGains           : quantized values of the given parametes.
>  */
>-int16_t WebRtcIsac_DequantizeLpcGain(
>-    const int* idx,
>-    double*    lpGains);
>-
>+int16_t WebRtcIsac_DequantizeLpcGain(const int* idx, double* lpGains);
> 
> /******************************************************************************
>  * WebRtcIsac_CorrelateLpcGain()
>@@ -259,10 +228,7 @@ int16_t WebRtcIsac_DequantizeLpcGain(
>  * Output:
>  *      -out                : correlated parameters.
>  */
>-int16_t WebRtcIsac_CorrelateLpcGain(
>-    const double* data,
>-    double*       out);
>-
>+int16_t WebRtcIsac_CorrelateLpcGain(const double* data, double* out);
> 
> /******************************************************************************
>  * WebRtcIsac_AddMeanToLinearDomain()
>@@ -275,8 +241,6 @@ int16_t WebRtcIsac_CorrelateLpcGain(
>  * Output:
>  *      -lpcGain            : LPC gain in normal domain.
>  */
>-int16_t WebRtcIsac_AddMeanToLinearDomain(
>-    double* lpcGains);
>-
>+int16_t WebRtcIsac_AddMeanToLinearDomain(double* lpcGains);
> 
> #endif  // MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ENCODE_LPC_SWB_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/entropy_coding.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/entropy_coding.c
>index b494cc287af..28767afc633 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/entropy_coding.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/entropy_coding.c
>@@ -17,19 +17,19 @@
>  */
> 
> 
>-#include "entropy_coding.h"
>-#include "settings.h"
>-#include "arith_routines.h"
>-#include "signal_processing_library.h"
>-#include "spectrum_ar_model_tables.h"
>-#include "lpc_tables.h"
>-#include "pitch_gain_tables.h"
>-#include "pitch_lag_tables.h"
>-#include "encode_lpc_swb.h"
>-#include "lpc_shape_swb12_tables.h"
>-#include "lpc_shape_swb16_tables.h"
>-#include "lpc_gain_swb_tables.h"
>-#include "os_specific_inline.h"
>+#include "common_audio/signal_processing/include/signal_processing_library.h"
>+#include "modules/audio_coding/codecs/isac/main/source/entropy_coding.h"
>+#include "modules/audio_coding/codecs/isac/main/source/settings.h"
>+#include "modules/audio_coding/codecs/isac/main/source/arith_routines.h"
>+#include "modules/audio_coding/codecs/isac/main/source/spectrum_ar_model_tables.h"
>+#include "modules/audio_coding/codecs/isac/main/source/lpc_tables.h"
>+#include "modules/audio_coding/codecs/isac/main/source/pitch_gain_tables.h"
>+#include "modules/audio_coding/codecs/isac/main/source/pitch_lag_tables.h"
>+#include "modules/audio_coding/codecs/isac/main/source/encode_lpc_swb.h"
>+#include "modules/audio_coding/codecs/isac/main/source/lpc_shape_swb12_tables.h"
>+#include "modules/audio_coding/codecs/isac/main/source/lpc_shape_swb16_tables.h"
>+#include "modules/audio_coding/codecs/isac/main/source/lpc_gain_swb_tables.h"
>+#include "modules/audio_coding/codecs/isac/main/source/os_specific_inline.h"
> 
> #include <math.h>
> #include <string.h>
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/entropy_coding.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/entropy_coding.h
>index 2ccd96b620e..6c2b8d3cc1a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/entropy_coding.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/entropy_coding.h
>@@ -19,8 +19,8 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ENTROPY_CODING_H_
> #define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ENTROPY_CODING_H_
> 
>-#include "settings.h"
>-#include "structs.h"
>+#include "modules/audio_coding/codecs/isac/main/source/settings.h"
>+#include "modules/audio_coding/codecs/isac/main/source/structs.h"
> 
> /******************************************************************************
>  * WebRtcIsac_DecodeSpec()
>@@ -46,8 +46,11 @@
>  * Return value             : < 0 if an error occures
>  *                              0 if succeeded.
>  */
>-int WebRtcIsac_DecodeSpec(Bitstr* streamdata, int16_t AvgPitchGain_Q12,
>-                          enum ISACBand band, double* fr, double* fi);
>+int WebRtcIsac_DecodeSpec(Bitstr* streamdata,
>+                          int16_t AvgPitchGain_Q12,
>+                          enum ISACBand band,
>+                          double* fr,
>+                          double* fi);
> 
> /******************************************************************************
>  * WebRtcIsac_EncodeSpec()
>@@ -72,24 +75,31 @@ int WebRtcIsac_DecodeSpec(Bitstr* streamdata, int16_t AvgPitchGain_Q12,
>  * Return value             : < 0 if an error occures
>  *                              0 if succeeded.
>  */
>-int WebRtcIsac_EncodeSpec(const int16_t* fr, const int16_t* fi,
>-                          int16_t AvgPitchGain_Q12, enum ISACBand band,
>+int WebRtcIsac_EncodeSpec(const int16_t* fr,
>+                          const int16_t* fi,
>+                          int16_t AvgPitchGain_Q12,
>+                          enum ISACBand band,
>                           Bitstr* streamdata);
> 
> /* decode & dequantize LPC Coef */
> int WebRtcIsac_DecodeLpcCoef(Bitstr* streamdata, double* LPCCoef);
>-int WebRtcIsac_DecodeLpcCoefUB(Bitstr* streamdata, double* lpcVecs,
>+int WebRtcIsac_DecodeLpcCoefUB(Bitstr* streamdata,
>+                               double* lpcVecs,
>                                double* percepFilterGains,
>                                int16_t bandwidth);
> 
>-int WebRtcIsac_DecodeLpc(Bitstr* streamdata, double* LPCCoef_lo,
>+int WebRtcIsac_DecodeLpc(Bitstr* streamdata,
>+                         double* LPCCoef_lo,
>                          double* LPCCoef_hi);
> 
> /* quantize & code LPC Coef */
>-void WebRtcIsac_EncodeLpcLb(double* LPCCoef_lo, double* LPCCoef_hi,
>-                            Bitstr* streamdata, IsacSaveEncoderData* encData);
>+void WebRtcIsac_EncodeLpcLb(double* LPCCoef_lo,
>+                            double* LPCCoef_hi,
>+                            Bitstr* streamdata,
>+                            IsacSaveEncoderData* encData);
> 
>-void WebRtcIsac_EncodeLpcGainLb(double* LPCCoef_lo, double* LPCCoef_hi,
>+void WebRtcIsac_EncodeLpcGainLb(double* LPCCoef_lo,
>+                                double* LPCCoef_hi,
>                                 Bitstr* streamdata,
>                                 IsacSaveEncoderData* encData);
> 
>@@ -126,7 +136,8 @@ void WebRtcIsac_EncodeLpcGainLb(double* LPCCoef_lo, double* LPCCoef_hi,
>  * Return value             : 0 if encoding is successful,
>  *                           <0 if failed to encode.
>  */
>-int16_t WebRtcIsac_EncodeLpcUB(double* lpcCoeff, Bitstr* streamdata,
>+int16_t WebRtcIsac_EncodeLpcUB(double* lpcCoeff,
>+                               Bitstr* streamdata,
>                                double* interpolLPCCoeff,
>                                int16_t bandwidth,
>                                ISACUBSaveEncDataStruct* encData);
>@@ -184,9 +195,9 @@ void WebRtcIsac_EncodePitchLag(double* PitchLags,
>                                Bitstr* streamdata,
>                                IsacSaveEncoderData* encData);
> 
>-int WebRtcIsac_DecodePitchGain(Bitstr* streamdata,
>-                               int16_t* PitchGain_Q12);
>-int WebRtcIsac_DecodePitchLag(Bitstr* streamdata, int16_t* PitchGain_Q12,
>+int WebRtcIsac_DecodePitchGain(Bitstr* streamdata, int16_t* PitchGain_Q12);
>+int WebRtcIsac_DecodePitchLag(Bitstr* streamdata,
>+                              int16_t* PitchGain_Q12,
>                               double* PitchLag);
> 
> int WebRtcIsac_DecodeFrameLen(Bitstr* streamdata, int16_t* framelength);
>@@ -200,10 +211,10 @@ void WebRtcIsac_Poly2Rc(double* a, int N, double* RC);
> /* Step-up */
> void WebRtcIsac_Rc2Poly(double* RC, int N, double* a);
> 
>-void WebRtcIsac_TranscodeLPCCoef(double* LPCCoef_lo, double* LPCCoef_hi,
>+void WebRtcIsac_TranscodeLPCCoef(double* LPCCoef_lo,
>+                                 double* LPCCoef_hi,
>                                  int* index_g);
> 
>-
> /******************************************************************************
>  * WebRtcIsac_EncodeLpcGainUb()
>  * Encode LPC gains of sub-Frames.
>@@ -220,10 +231,10 @@ void WebRtcIsac_TranscodeLPCCoef(double* LPCCoef_lo, double* LPCCoef_hi,
>  *  - lpcGainIndex          : quantization indices for lpc gains, these will
>  *                            be stored to be used  for FEC.
>  */
>-void WebRtcIsac_EncodeLpcGainUb(double* lpGains, Bitstr* streamdata,
>+void WebRtcIsac_EncodeLpcGainUb(double* lpGains,
>+                                Bitstr* streamdata,
>                                 int* lpcGainIndex);
> 
>-
> /******************************************************************************
>  * WebRtcIsac_EncodeLpcGainUb()
>  * Store LPC gains of sub-Frames in 'streamdata'.
>@@ -239,7 +250,6 @@ void WebRtcIsac_EncodeLpcGainUb(double* lpGains, Bitstr* streamdata,
>  */
> void WebRtcIsac_StoreLpcGainUb(double* lpGains, Bitstr* streamdata);
> 
>-
> /******************************************************************************
>  * WebRtcIsac_DecodeLpcGainUb()
>  * Decode the LPC gain of sub-frames.
>@@ -257,7 +267,6 @@ void WebRtcIsac_StoreLpcGainUb(double* lpGains, Bitstr* streamdata);
>  */
> int16_t WebRtcIsac_DecodeLpcGainUb(double* lpGains, Bitstr* streamdata);
> 
>-
> /******************************************************************************
>  * WebRtcIsac_EncodeBandwidth()
>  * Encode if the bandwidth of encoded audio is 0-12 kHz or 0-16 kHz.
>@@ -277,7 +286,6 @@ int16_t WebRtcIsac_DecodeLpcGainUb(double* lpGains, Bitstr* streamdata);
> int16_t WebRtcIsac_EncodeBandwidth(enum ISACBandwidth bandwidth,
>                                    Bitstr* streamData);
> 
>-
> /******************************************************************************
>  * WebRtcIsac_DecodeBandwidth()
>  * Decode the bandwidth of the encoded audio, i.e. if the bandwidth is 0-12 kHz
>@@ -298,7 +306,6 @@ int16_t WebRtcIsac_EncodeBandwidth(enum ISACBandwidth bandwidth,
> int16_t WebRtcIsac_DecodeBandwidth(Bitstr* streamData,
>                                    enum ISACBandwidth* bandwidth);
> 
>-
> /******************************************************************************
>  * WebRtcIsac_EncodeJitterInfo()
>  * Decode the jitter information.
>@@ -316,9 +323,7 @@ int16_t WebRtcIsac_DecodeBandwidth(Bitstr* streamData,
>  * Return value             : 0 if succeeded.
>  *                           <0 if failed.
>  */
>-int16_t WebRtcIsac_EncodeJitterInfo(int32_t jitterIndex,
>-                                    Bitstr* streamData);
>-
>+int16_t WebRtcIsac_EncodeJitterInfo(int32_t jitterIndex, Bitstr* streamData);
> 
> /******************************************************************************
>  * WebRtcIsac_DecodeJitterInfo()
>@@ -337,7 +342,6 @@ int16_t WebRtcIsac_EncodeJitterInfo(int32_t jitterIndex,
>  * Return value             : 0 if succeeded.
>  *                           <0 if failed.
>  */
>-int16_t WebRtcIsac_DecodeJitterInfo(Bitstr* streamData,
>-                                    int32_t* jitterInfo);
>+int16_t WebRtcIsac_DecodeJitterInfo(Bitstr* streamData, int32_t* jitterInfo);
> 
> #endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ENTROPY_CODING_H_ */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/filter_functions.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/filter_functions.c
>index d47eb1fa664..a4f297c5a17 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/filter_functions.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/filter_functions.c
>@@ -13,16 +13,14 @@
> #ifdef WEBRTC_ANDROID
> #include <stdlib.h>
> #endif
>-#include "pitch_estimator.h"
>-#include "lpc_analysis.h"
>-#include "codec.h"
> 
>+#include "modules/audio_coding/codecs/isac/main/source/pitch_estimator.h"
>+#include "modules/audio_coding/codecs/isac/main/source/isac_vad.h"
> 
>-
>-void WebRtcIsac_AllPoleFilter(double* InOut,
>-                              double* Coef,
>-                              size_t lengthInOut,
>-                              int orderCoef) {
>+static void WebRtcIsac_AllPoleFilter(double* InOut,
>+                                     double* Coef,
>+                                     size_t lengthInOut,
>+                                     int orderCoef) {
>   /* the state of filter is assumed to be in InOut[-1] to InOut[-orderCoef] */
>   double scal;
>   double sum;
>@@ -55,12 +53,11 @@ void WebRtcIsac_AllPoleFilter(double* InOut,
>   }
> }
> 
>-
>-void WebRtcIsac_AllZeroFilter(double* In,
>-                              double* Coef,
>-                              size_t lengthInOut,
>-                              int orderCoef,
>-                              double* Out) {
>+static void WebRtcIsac_AllZeroFilter(double* In,
>+                                     double* Coef,
>+                                     size_t lengthInOut,
>+                                     int orderCoef,
>+                                     double* Out) {
>   /* the state of filter is assumed to be in In[-1] to In[-orderCoef] */
> 
>   size_t n;
>@@ -80,13 +77,12 @@ void WebRtcIsac_AllZeroFilter(double* In,
>   }
> }
> 
>-
>-void WebRtcIsac_ZeroPoleFilter(double* In,
>-                               double* ZeroCoef,
>-                               double* PoleCoef,
>-                               size_t lengthInOut,
>-                               int orderCoef,
>-                               double* Out) {
>+static void WebRtcIsac_ZeroPoleFilter(double* In,
>+                                      double* ZeroCoef,
>+                                      double* PoleCoef,
>+                                      size_t lengthInOut,
>+                                      int orderCoef,
>+                                      double* Out) {
>   /* the state of the zero section is assumed to be in In[-1] to In[-orderCoef] */
>   /* the state of the pole section is assumed to be in Out[-1] to Out[-orderCoef] */
> 
>@@ -115,8 +111,10 @@ void WebRtcIsac_AutoCorr(double* r, const double* x, size_t N, size_t order) {
> 
> }
> 
>-
>-void WebRtcIsac_BwExpand(double* out, double* in, double coef, size_t length) {
>+static void WebRtcIsac_BwExpand(double* out,
>+                                double* in,
>+                                double coef,
>+                                size_t length) {
>   size_t i;
>   double  chirp;
> 
>@@ -195,69 +193,3 @@ void WebRtcIsac_WeightingFilter(const double* in,
>   memcpy(weiout, weoutbuf+PITCH_WLPCORDER, sizeof(double) * PITCH_FRAME_LEN);
>   memcpy(whiout, whoutbuf+PITCH_WLPCORDER, sizeof(double) * PITCH_FRAME_LEN);
> }
>-
>-
>-static const double APupper[ALLPASSSECTIONS] = {0.0347, 0.3826};
>-static const double APlower[ALLPASSSECTIONS] = {0.1544, 0.744};
>-
>-
>-void WebRtcIsac_AllpassFilterForDec(double* InOut,
>-                                    const double* APSectionFactors,
>-                                    size_t lengthInOut,
>-                                    double* FilterState) {
>-  //This performs all-pass filtering--a series of first order all-pass sections are used
>-  //to filter the input in a cascade manner.
>-  size_t n,j;
>-  double temp;
>-  for (j=0; j<ALLPASSSECTIONS; j++){
>-    for (n=0;n<lengthInOut;n+=2){
>-      temp = InOut[n]; //store input
>-      InOut[n] = FilterState[j] + APSectionFactors[j]*temp;
>-      FilterState[j] = -APSectionFactors[j]*InOut[n] + temp;
>-    }
>-  }
>-}
>-
>-void WebRtcIsac_DecimateAllpass(const double* in,
>-                                double* state_in,
>-                                size_t N,
>-                                double* out) {
>-  size_t n;
>-  double data_vec[PITCH_FRAME_LEN];
>-
>-  /* copy input */
>-  memcpy(data_vec+1, in, sizeof(double) * (N-1));
>-
>-  data_vec[0] = state_in[2*ALLPASSSECTIONS];   //the z^(-1) state
>-  state_in[2*ALLPASSSECTIONS] = in[N-1];
>-
>-  WebRtcIsac_AllpassFilterForDec(data_vec+1, APupper, N, state_in);
>-  WebRtcIsac_AllpassFilterForDec(data_vec, APlower, N, state_in+ALLPASSSECTIONS);
>-
>-  for (n=0;n<N/2;n++)
>-    out[n] = data_vec[2*n] + data_vec[2*n+1];
>-
>-}
>-
>-
>-/* create high-pass filter ocefficients
>- * z = 0.998 * exp(j*2*pi*35/8000);
>- * p = 0.94 * exp(j*2*pi*140/8000);
>- * HP_b = [1, -2*real(z), abs(z)^2];
>- * HP_a = [1, -2*real(p), abs(p)^2]; */
>-static const double a_coef[2] = { 1.86864659625574, -0.88360000000000};
>-static const double b_coef[2] = {-1.99524591718270,  0.99600400000000};
>-
>-/* second order high-pass filter */
>-void WebRtcIsac_Highpass(const double* in,
>-                         double* out,
>-                         double* state,
>-                         size_t N) {
>-  size_t k;
>-
>-  for (k=0; k<N; k++) {
>-    *out = *in + state[1];
>-    state[1] = state[0] + b_coef[0] * *in + a_coef[0] * *out;
>-    state[0] = b_coef[1] * *in++ + a_coef[1] * *out++;
>-  }
>-}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/filter_functions.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/filter_functions.h
>new file mode 100644
>index 00000000000..48a9b7426bb
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/filter_functions.h
>@@ -0,0 +1,23 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_FILTER_FUNCTIONS_H_
>+#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_FILTER_FUNCTIONS_H_
>+
>+#include "modules/audio_coding/codecs/isac/main/source/structs.h"
>+
>+void WebRtcIsac_AutoCorr(double* r, const double* x, size_t N, size_t order);
>+
>+void WebRtcIsac_WeightingFilter(const double* in,
>+                                double* weiout,
>+                                double* whiout,
>+                                WeightFiltstr* wfdata);
>+
>+#endif  // MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_FILTER_FUNCTIONS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/filterbank_tables.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/filterbank_tables.c
>deleted file mode 100644
>index 0f844af9d24..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/filterbank_tables.c
>+++ /dev/null
>@@ -1,37 +0,0 @@
>-/*
>- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-/* filterbank_tables.c*/
>-/* This file contains variables that are used in filterbanks.c*/
>-
>-#include "filterbank_tables.h"
>-#include "settings.h"
>-
>-/* The composite all-pass filter factors */
>-const float WebRtcIsac_kCompositeApFactorsFloat[4] = {
>- 0.03470000000000f,  0.15440000000000f,  0.38260000000000f,  0.74400000000000f};
>-
>-/* The upper channel all-pass filter factors */
>-const float WebRtcIsac_kUpperApFactorsFloat[2] = {
>- 0.03470000000000f,  0.38260000000000f};
>-
>-/* The lower channel all-pass filter factors */
>-const float WebRtcIsac_kLowerApFactorsFloat[2] = {
>- 0.15440000000000f,  0.74400000000000f};
>-
>-/* The matrix for transforming the backward composite state to upper channel state */
>-const float WebRtcIsac_kTransform1Float[8] = {
>-  -0.00158678506084f,  0.00127157815343f, -0.00104805672709f,  0.00084837248079f,
>-  0.00134467983258f, -0.00107756549387f,  0.00088814793277f, -0.00071893072525f};
>-
>-/* The matrix for transforming the backward composite state to lower channel state */
>-const float WebRtcIsac_kTransform2Float[8] = {
>- -0.00170686041697f,  0.00136780109829f, -0.00112736532350f,  0.00091257055385f,
>-  0.00103094281812f, -0.00082615076557f,  0.00068092756088f, -0.00055119165484f};
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/filterbank_tables.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/filterbank_tables.h
>deleted file mode 100644
>index 714e2270813..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/filterbank_tables.h
>+++ /dev/null
>@@ -1,46 +0,0 @@
>-/*
>- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-/*
>- * filterbank_tables.h
>- *
>- * Header file for variables that are defined in
>- * filterbank_tables.c.
>- *
>- */
>-
>-#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_FILTERBANK_TABLES_H_
>-#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_FILTERBANK_TABLES_H_
>-
>-#include "structs.h"
>-
>-/********************* Coefficient Tables ************************/
>-/* The number of composite all-pass filter factors */
>-#define NUMBEROFCOMPOSITEAPSECTIONS 4
>-
>-/* The number of all-pass filter factors in an upper or lower channel*/
>-#define NUMBEROFCHANNELAPSECTIONS 2
>-
>-/* The composite all-pass filter factors */
>-extern const float WebRtcIsac_kCompositeApFactorsFloat[4];
>-
>-/* The upper channel all-pass filter factors */
>-extern const float WebRtcIsac_kUpperApFactorsFloat[2];
>-
>-/* The lower channel all-pass filter factors */
>-extern const float WebRtcIsac_kLowerApFactorsFloat[2];
>-
>-/* The matrix for transforming the backward composite state to upper channel state */
>-extern const float WebRtcIsac_kTransform1Float[8];
>-
>-/* The matrix for transforming the backward composite state to lower channel state */
>-extern const float WebRtcIsac_kTransform2Float[8];
>-
>-#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_FILTERBANK_TABLES_H_ */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/filterbanks.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/filterbanks.c
>index 671fd321cee..d57b55022d3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/filterbanks.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/filterbanks.c
>@@ -18,241 +18,9 @@
>  *
>  */
> 
>-#include "settings.h"
>-#include "filterbank_tables.h"
>-#include "codec.h"
>-
>-/* This function performs all-pass filtering--a series of first order all-pass
>- * sections are used to filter the input in a cascade manner.
>- * The input is overwritten!!
>- */
>-static void WebRtcIsac_AllPassFilter2Float(float *InOut, const float *APSectionFactors,
>-                                           int lengthInOut, int NumberOfSections,
>-                                           float *FilterState)
>-{
>-  int n, j;
>-  float temp;
>-  for (j=0; j<NumberOfSections; j++){
>-    for (n=0;n<lengthInOut;n++){
>-      temp = FilterState[j] + APSectionFactors[j] * InOut[n];
>-      FilterState[j] = -APSectionFactors[j] * temp + InOut[n];
>-      InOut[n] = temp;
>-    }
>-  }
>-}
>-
>-/* HPstcoeff_in = {a1, a2, b1 - b0 * a1, b2 - b0 * a2}; */
>-static const float kHpStCoefInFloat[4] =
>-{-1.94895953203325f, 0.94984516000000f, -0.05101826139794f, 0.05015484000000f};
>-
>-/* Function WebRtcIsac_SplitAndFilter
>- * This function creates low-pass and high-pass decimated versions of part of
>- the input signal, and part of the signal in the input 'lookahead buffer'.
>-
>- INPUTS:
>- in: a length FRAMESAMPLES array of input samples
>- prefiltdata: input data structure containing the filterbank states
>- and lookahead samples from the previous encoding
>- iteration.
>- OUTPUTS:
>- LP: a FRAMESAMPLES_HALF array of low-pass filtered samples that
>- have been phase equalized.  The first QLOOKAHEAD samples are
>- based on the samples in the two prefiltdata->INLABUFx arrays
>- each of length QLOOKAHEAD.
>- The remaining FRAMESAMPLES_HALF-QLOOKAHEAD samples are based
>- on the first FRAMESAMPLES_HALF-QLOOKAHEAD samples of the input
>- array in[].
>- HP: a FRAMESAMPLES_HALF array of high-pass filtered samples that
>- have been phase equalized.  The first QLOOKAHEAD samples are
>- based on the samples in the two prefiltdata->INLABUFx arrays
>- each of length QLOOKAHEAD.
>- The remaining FRAMESAMPLES_HALF-QLOOKAHEAD samples are based
>- on the first FRAMESAMPLES_HALF-QLOOKAHEAD samples of the input
>- array in[].
>-
>- LP_la: a FRAMESAMPLES_HALF array of low-pass filtered samples.
>- These samples are not phase equalized. They are computed
>- from the samples in the in[] array.
>- HP_la: a FRAMESAMPLES_HALF array of high-pass filtered samples
>- that are not phase equalized. They are computed from
>- the in[] vector.
>- prefiltdata: this input data structure's filterbank state and
>- lookahead sample buffers are updated for the next
>- encoding iteration.
>-*/
>-void WebRtcIsac_SplitAndFilterFloat(float *pin, float *LP, float *HP,
>-                                    double *LP_la, double *HP_la,
>-                                    PreFiltBankstr *prefiltdata)
>-{
>-  int k,n;
>-  float CompositeAPFilterState[NUMBEROFCOMPOSITEAPSECTIONS];
>-  float ForTransform_CompositeAPFilterState[NUMBEROFCOMPOSITEAPSECTIONS];
>-  float ForTransform_CompositeAPFilterState2[NUMBEROFCOMPOSITEAPSECTIONS];
>-  float tempinoutvec[FRAMESAMPLES+MAX_AR_MODEL_ORDER];
>-  float tempin_ch1[FRAMESAMPLES+MAX_AR_MODEL_ORDER];
>-  float tempin_ch2[FRAMESAMPLES+MAX_AR_MODEL_ORDER];
>-  float in[FRAMESAMPLES];
>-  float ftmp;
>-
>-
>-  /* High pass filter */
>-
>-  for (k=0;k<FRAMESAMPLES;k++) {
>-    in[k] = pin[k] + kHpStCoefInFloat[2] * prefiltdata->HPstates_float[0] +
>-        kHpStCoefInFloat[3] * prefiltdata->HPstates_float[1];
>-    ftmp = pin[k] - kHpStCoefInFloat[0] * prefiltdata->HPstates_float[0] -
>-        kHpStCoefInFloat[1] * prefiltdata->HPstates_float[1];
>-    prefiltdata->HPstates_float[1] = prefiltdata->HPstates_float[0];
>-    prefiltdata->HPstates_float[0] = ftmp;
>-  }
>-
>-  /*
>-    % backwards all-pass filtering to obtain zero-phase
>-    [tmp1(N2+LA:-1:LA+1, 1), state1] = filter(Q.coef, Q.coef(end:-1:1), in(N:-2:2));
>-    tmp1(LA:-1:1) = filter(Q.coef, Q.coef(end:-1:1), Q.LookAheadBuf1, state1);
>-    Q.LookAheadBuf1 = in(N:-2:N-2*LA+2);
>-  */
>-  /*Backwards all-pass filter the odd samples of the input (upper channel)
>-    to eventually obtain zero phase.  The composite all-pass filter (comprised of both
>-    the upper and lower channel all-pass filsters in series) is used for the
>-    filtering. */
>-
>-  /* First Channel */
>-
>-  /*initial state of composite filter is zero */
>-  for (k=0;k<NUMBEROFCOMPOSITEAPSECTIONS;k++){
>-    CompositeAPFilterState[k] = 0.0;
>-  }
>-  /* put every other sample of input into a temporary vector in reverse (backward) order*/
>-  for (k=0;k<FRAMESAMPLES_HALF;k++) {
>-    tempinoutvec[k] = in[FRAMESAMPLES-1-2*k];
>-  }
>-
>-  /* now all-pass filter the backwards vector.  Output values overwrite the input vector. */
>-  WebRtcIsac_AllPassFilter2Float(tempinoutvec, WebRtcIsac_kCompositeApFactorsFloat,
>-                                 FRAMESAMPLES_HALF, NUMBEROFCOMPOSITEAPSECTIONS, CompositeAPFilterState);
>-
>-  /* save the backwards filtered output for later forward filtering,
>-     but write it in forward order*/
>-  for (k=0;k<FRAMESAMPLES_HALF;k++) {
>-    tempin_ch1[FRAMESAMPLES_HALF+QLOOKAHEAD-1-k] = tempinoutvec[k];
>-  }
>-
>-  /* save the backwards filter state  becaue it will be transformed
>-     later into a forward state */
>-  for (k=0; k<NUMBEROFCOMPOSITEAPSECTIONS; k++) {
>-    ForTransform_CompositeAPFilterState[k] = CompositeAPFilterState[k];
>-  }
>-
>-  /* now backwards filter the samples in the lookahead buffer. The samples were
>-     placed there in the encoding of the previous frame.  The output samples
>-     overwrite the input samples */
>-  WebRtcIsac_AllPassFilter2Float(prefiltdata->INLABUF1_float,
>-                                 WebRtcIsac_kCompositeApFactorsFloat, QLOOKAHEAD,
>-                                 NUMBEROFCOMPOSITEAPSECTIONS, CompositeAPFilterState);
>-
>-  /* save the output, but write it in forward order */
>-  /* write the lookahead samples for the next encoding iteration. Every other
>-     sample at the end of the input frame is written in reverse order for the
>-     lookahead length. Exported in the prefiltdata structure. */
>-  for (k=0;k<QLOOKAHEAD;k++) {
>-    tempin_ch1[QLOOKAHEAD-1-k]=prefiltdata->INLABUF1_float[k];
>-    prefiltdata->INLABUF1_float[k]=in[FRAMESAMPLES-1-2*k];
>-  }
>-
>-  /* Second Channel.  This is exactly like the first channel, except that the
>-     even samples are now filtered instead (lower channel). */
>-  for (k=0;k<NUMBEROFCOMPOSITEAPSECTIONS;k++){
>-    CompositeAPFilterState[k] = 0.0;
>-  }
>-
>-  for (k=0;k<FRAMESAMPLES_HALF;k++) {
>-    tempinoutvec[k] = in[FRAMESAMPLES-2-2*k];
>-  }
>-
>-  WebRtcIsac_AllPassFilter2Float(tempinoutvec, WebRtcIsac_kCompositeApFactorsFloat,
>-                                 FRAMESAMPLES_HALF, NUMBEROFCOMPOSITEAPSECTIONS, CompositeAPFilterState);
>-
>-  for (k=0;k<FRAMESAMPLES_HALF;k++) {
>-    tempin_ch2[FRAMESAMPLES_HALF+QLOOKAHEAD-1-k] = tempinoutvec[k];
>-  }
>-
>-  for (k=0; k<NUMBEROFCOMPOSITEAPSECTIONS; k++) {
>-    ForTransform_CompositeAPFilterState2[k] = CompositeAPFilterState[k];
>-  }
>-
>-
>-  WebRtcIsac_AllPassFilter2Float(prefiltdata->INLABUF2_float,
>-                                 WebRtcIsac_kCompositeApFactorsFloat, QLOOKAHEAD,NUMBEROFCOMPOSITEAPSECTIONS,
>-                                 CompositeAPFilterState);
>-
>-  for (k=0;k<QLOOKAHEAD;k++) {
>-    tempin_ch2[QLOOKAHEAD-1-k]=prefiltdata->INLABUF2_float[k];
>-    prefiltdata->INLABUF2_float[k]=in[FRAMESAMPLES-2-2*k];
>-  }
>-
>-  /* Transform filter states from backward to forward */
>-  /*At this point, each of the states of the backwards composite filters for the
>-    two channels are transformed into forward filtering states for the corresponding
>-    forward channel filters.  Each channel's forward filtering state from the previous
>-    encoding iteration is added to the transformed state to get a proper forward state */
>-
>-  /* So the existing NUMBEROFCOMPOSITEAPSECTIONS x 1 (4x1) state vector is multiplied by a
>-     NUMBEROFCHANNELAPSECTIONSxNUMBEROFCOMPOSITEAPSECTIONS (2x4) transform matrix to get the
>-     new state that is added to the previous 2x1 input state */
>-
>-  for (k=0;k<NUMBEROFCHANNELAPSECTIONS;k++){ /* k is row variable */
>-    for (n=0; n<NUMBEROFCOMPOSITEAPSECTIONS;n++){/* n is column variable */
>-      prefiltdata->INSTAT1_float[k] += ForTransform_CompositeAPFilterState[n]*
>-          WebRtcIsac_kTransform1Float[k*NUMBEROFCHANNELAPSECTIONS+n];
>-      prefiltdata->INSTAT2_float[k] += ForTransform_CompositeAPFilterState2[n]*
>-          WebRtcIsac_kTransform2Float[k*NUMBEROFCHANNELAPSECTIONS+n];
>-    }
>-  }
>-
>-  /*obtain polyphase components by forward all-pass filtering through each channel */
>-  /* the backward filtered samples are now forward filtered with the corresponding channel filters */
>-  /* The all pass filtering automatically updates the filter states which are exported in the
>-     prefiltdata structure */
>-  WebRtcIsac_AllPassFilter2Float(tempin_ch1,WebRtcIsac_kUpperApFactorsFloat,
>-                                 FRAMESAMPLES_HALF, NUMBEROFCHANNELAPSECTIONS, prefiltdata->INSTAT1_float);
>-  WebRtcIsac_AllPassFilter2Float(tempin_ch2,WebRtcIsac_kLowerApFactorsFloat,
>-                                 FRAMESAMPLES_HALF, NUMBEROFCHANNELAPSECTIONS, prefiltdata->INSTAT2_float);
>-
>-  /* Now Construct low-pass and high-pass signals as combinations of polyphase components */
>-  for (k=0; k<FRAMESAMPLES_HALF; k++) {
>-    LP[k] = 0.5f*(tempin_ch1[k] + tempin_ch2[k]);/* low pass signal*/
>-    HP[k] = 0.5f*(tempin_ch1[k] - tempin_ch2[k]);/* high pass signal*/
>-  }
>-
>-  /* Lookahead LP and HP signals */
>-  /* now create low pass and high pass signals of the input vector.  However, no
>-     backwards filtering is performed, and hence no phase equalization is involved.
>-     Also, the input contains some samples that are lookahead samples.  The high pass
>-     and low pass signals that are created are used outside this function for analysis
>-     (not encoding) purposes */
>-
>-  /* set up input */
>-  for (k=0; k<FRAMESAMPLES_HALF; k++) {
>-    tempin_ch1[k]=in[2*k+1];
>-    tempin_ch2[k]=in[2*k];
>-  }
>-
>-  /* the input filter states are passed in and updated by the all-pass filtering routine and
>-     exported in the prefiltdata structure*/
>-  WebRtcIsac_AllPassFilter2Float(tempin_ch1,WebRtcIsac_kUpperApFactorsFloat,
>-                                 FRAMESAMPLES_HALF, NUMBEROFCHANNELAPSECTIONS, prefiltdata->INSTATLA1_float);
>-  WebRtcIsac_AllPassFilter2Float(tempin_ch2,WebRtcIsac_kLowerApFactorsFloat,
>-                                 FRAMESAMPLES_HALF, NUMBEROFCHANNELAPSECTIONS, prefiltdata->INSTATLA2_float);
>-
>-  for (k=0; k<FRAMESAMPLES_HALF; k++) {
>-    LP_la[k] = (float)(0.5f*(tempin_ch1[k] + tempin_ch2[k])); /*low pass */
>-    HP_la[k] = (double)(0.5f*(tempin_ch1[k] - tempin_ch2[k])); /* high pass */
>-  }
>-
>-
>-}/*end of WebRtcIsac_SplitAndFilter */
>-
>+#include "modules/audio_coding/codecs/isac/main/source/settings.h"
>+#include "modules/audio_coding/codecs/isac/main/source/codec.h"
>+#include "modules/audio_coding/codecs/isac/main/source/isac_vad.h"
> 
> /* Combining */
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/intialize.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/intialize.c
>index 01e683c573f..5c951f6e9d1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/intialize.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/intialize.c
>@@ -10,12 +10,12 @@
> 
> /* encode.c  - Encoding function for the iSAC coder */
> 
>-#include "structs.h"
>-#include "codec.h"
>-#include "pitch_estimator.h"
>-
> #include <math.h>
> 
>+#include "modules/audio_coding/codecs/isac/main/source/structs.h"
>+#include "modules/audio_coding/codecs/isac/main/source/codec.h"
>+#include "modules/audio_coding/codecs/isac/main/source/pitch_estimator.h"
>+
> void WebRtcIsac_InitMasking(MaskFiltstr *maskdata) {
> 
>   int k;
>@@ -43,39 +43,6 @@ void WebRtcIsac_InitMasking(MaskFiltstr *maskdata) {
>   return;
> }
> 
>-void WebRtcIsac_InitPreFilterbank(PreFiltBankstr *prefiltdata)
>-{
>-  int k;
>-
>-  for (k = 0; k < QLOOKAHEAD; k++) {
>-    prefiltdata->INLABUF1[k] = 0;
>-    prefiltdata->INLABUF2[k] = 0;
>-
>-    prefiltdata->INLABUF1_float[k] = 0;
>-    prefiltdata->INLABUF2_float[k] = 0;
>-  }
>-  for (k = 0; k < 2*(QORDER-1); k++) {
>-    prefiltdata->INSTAT1[k] = 0;
>-    prefiltdata->INSTAT2[k] = 0;
>-    prefiltdata->INSTATLA1[k] = 0;
>-    prefiltdata->INSTATLA2[k] = 0;
>-
>-    prefiltdata->INSTAT1_float[k] = 0;
>-    prefiltdata->INSTAT2_float[k] = 0;
>-    prefiltdata->INSTATLA1_float[k] = 0;
>-    prefiltdata->INSTATLA2_float[k] = 0;
>-  }
>-
>-  /* High pass filter states */
>-  prefiltdata->HPstates[0] = 0.0;
>-  prefiltdata->HPstates[1] = 0.0;
>-
>-  prefiltdata->HPstates_float[0] = 0.0f;
>-  prefiltdata->HPstates_float[1] = 0.0f;
>-
>-  return;
>-}
>-
> void WebRtcIsac_InitPostFilterbank(PostFiltBankstr *postfiltdata)
> {
>   int k;
>@@ -103,69 +70,3 @@ void WebRtcIsac_InitPostFilterbank(PostFiltBankstr *postfiltdata)
> 
>   return;
> }
>-
>-
>-void WebRtcIsac_InitPitchFilter(PitchFiltstr *pitchfiltdata)
>-{
>-  int k;
>-
>-  for (k = 0; k < PITCH_BUFFSIZE; k++) {
>-    pitchfiltdata->ubuf[k] = 0.0;
>-  }
>-  pitchfiltdata->ystate[0] = 0.0;
>-  for (k = 1; k < (PITCH_DAMPORDER); k++) {
>-    pitchfiltdata->ystate[k] = 0.0;
>-  }
>-  pitchfiltdata->oldlagp[0] = 50.0;
>-  pitchfiltdata->oldgainp[0] = 0.0;
>-}
>-
>-void WebRtcIsac_InitWeightingFilter(WeightFiltstr *wfdata)
>-{
>-  int k;
>-  double t, dtmp, dtmp2, denum, denum2;
>-
>-  for (k=0;k<PITCH_WLPCBUFLEN;k++)
>-    wfdata->buffer[k]=0.0;
>-
>-  for (k=0;k<PITCH_WLPCORDER;k++) {
>-    wfdata->istate[k]=0.0;
>-    wfdata->weostate[k]=0.0;
>-    wfdata->whostate[k]=0.0;
>-  }
>-
>-  /* next part should be in Matlab, writing to a global table */
>-  t = 0.5;
>-  denum = 1.0 / ((double) PITCH_WLPCWINLEN);
>-  denum2 = denum * denum;
>-  for (k=0;k<PITCH_WLPCWINLEN;k++) {
>-    dtmp = PITCH_WLPCASYM * t * denum + (1-PITCH_WLPCASYM) * t * t * denum2;
>-    dtmp *= 3.14159265;
>-    dtmp2 = sin(dtmp);
>-    wfdata->window[k] = dtmp2 * dtmp2;
>-    t++;
>-  }
>-}
>-
>-/* clear all buffers */
>-void WebRtcIsac_InitPitchAnalysis(PitchAnalysisStruct *State)
>-{
>-  int k;
>-
>-  for (k = 0; k < PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2-PITCH_FRAME_LEN/2+2; k++)
>-    State->dec_buffer[k] = 0.0;
>-  for (k = 0; k < 2*ALLPASSSECTIONS+1; k++)
>-    State->decimator_state[k] = 0.0;
>-  for (k = 0; k < 2; k++)
>-    State->hp_state[k] = 0.0;
>-  for (k = 0; k < QLOOKAHEAD; k++)
>-    State->whitened_buf[k] = 0.0;
>-  for (k = 0; k < QLOOKAHEAD; k++)
>-    State->inbuf[k] = 0.0;
>-
>-  WebRtcIsac_InitPitchFilter(&(State->PFstr_wght));
>-
>-  WebRtcIsac_InitPitchFilter(&(State->PFstr));
>-
>-  WebRtcIsac_InitWeightingFilter(&(State->Wghtstr));
>-}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c
>index 79dc7e2b394..866959691c2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/isac.c
>@@ -31,6 +31,8 @@
> #include "modules/audio_coding/codecs/isac/main/source/lpc_shape_swb16_tables.h"
> #include "modules/audio_coding/codecs/isac/main/source/os_specific_inline.h"
> #include "modules/audio_coding/codecs/isac/main/source/structs.h"
>+#include "modules/audio_coding/codecs/isac/main/source/isac_vad.h"
>+#include "rtc_base/system/arch.h"
> 
> #define BIT_MASK_DEC_INIT 0x0001
> #define BIT_MASK_ENC_INIT 0x0002
>@@ -1266,8 +1268,10 @@ static int Decode(ISACStruct* ISAC_main_inst,
> 
>         /* It might be less due to garbage. */
>         if ((numDecodedBytesUB != lenNextStream) &&
>-            (numDecodedBytesUB != (lenNextStream -
>-                encoded[numDecodedBytesLB + 1 + numDecodedBytesUB]))) {
>+            (numDecodedBytesLB + 1 + numDecodedBytesUB >= lenEncodedBytes ||
>+             numDecodedBytesUB !=
>+                 (lenNextStream -
>+                  encoded[numDecodedBytesLB + 1 + numDecodedBytesUB]))) {
>           instISAC->errorCode = ISAC_LENGTH_MISMATCH;
>           return -1;
>         }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/isac_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/isac_unittest.cc
>index 727f0f61e04..3ec28cca74d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/isac_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/isac_unittest.cc
>@@ -35,15 +35,13 @@ class IsacTest : public ::testing::Test {
>   uint8_t bitstream_small_[7];  // Simulate sync packets.
> };
> 
>-IsacTest::IsacTest()
>-    : isac_codec_(NULL) {
>-}
>+IsacTest::IsacTest() : isac_codec_(NULL) {}
> 
> void IsacTest::SetUp() {
>   // Read some samples from a speech file, to be used in the encode test.
>   FILE* input_file;
>   const std::string file_name =
>-        webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
>+      webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
>   input_file = fopen(file_name.c_str(), "rb");
>   ASSERT_TRUE(input_file != NULL);
>   ASSERT_EQ(kIsacNumberOfSamples,
>@@ -69,7 +67,8 @@ TEST_F(IsacTest, IsacFreeFail) {
> TEST_F(IsacTest, IsacCreateFree) {
>   EXPECT_EQ(0, WebRtcIsac_Create(&isac_codec_));
>   EXPECT_TRUE(isac_codec_ != NULL);
>-  EXPECT_EQ(0, WebRtcIsac_Free(isac_codec_));}
>+  EXPECT_EQ(0, WebRtcIsac_Free(isac_codec_));
>+}
> 
> TEST_F(IsacTest, IsacUpdateBWE) {
>   // Create encoder memory.
>@@ -86,17 +85,17 @@ TEST_F(IsacTest, IsacUpdateBWE) {
>                                             12345, 56789));
> 
>   // Encode 60 ms of data (needed to create a first packet).
>-  encoded_bytes =  WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
>+  encoded_bytes = WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
>   EXPECT_EQ(0, encoded_bytes);
>-  encoded_bytes =  WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
>+  encoded_bytes = WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
>   EXPECT_EQ(0, encoded_bytes);
>-  encoded_bytes =  WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
>+  encoded_bytes = WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
>   EXPECT_EQ(0, encoded_bytes);
>-  encoded_bytes =  WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
>+  encoded_bytes = WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
>   EXPECT_EQ(0, encoded_bytes);
>-  encoded_bytes =  WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
>+  encoded_bytes = WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
>   EXPECT_EQ(0, encoded_bytes);
>-  encoded_bytes =  WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
>+  encoded_bytes = WebRtcIsac_Encode(isac_codec_, speech_data_, bitstream_);
>   EXPECT_GT(encoded_bytes, 0);
> 
>   // Call to update bandwidth estimator with real data.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/isac_vad.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/isac_vad.c
>new file mode 100644
>index 00000000000..57cf0c39da5
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/isac_vad.c
>@@ -0,0 +1,409 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_coding/codecs/isac/main/source/isac_vad.h"
>+
>+#include <math.h>
>+
>+void WebRtcIsac_InitPitchFilter(PitchFiltstr* pitchfiltdata) {
>+  int k;
>+
>+  for (k = 0; k < PITCH_BUFFSIZE; k++) {
>+    pitchfiltdata->ubuf[k] = 0.0;
>+  }
>+  pitchfiltdata->ystate[0] = 0.0;
>+  for (k = 1; k < (PITCH_DAMPORDER); k++) {
>+    pitchfiltdata->ystate[k] = 0.0;
>+  }
>+  pitchfiltdata->oldlagp[0] = 50.0;
>+  pitchfiltdata->oldgainp[0] = 0.0;
>+}
>+
>+static void WebRtcIsac_InitWeightingFilter(WeightFiltstr* wfdata) {
>+  int k;
>+  double t, dtmp, dtmp2, denum, denum2;
>+
>+  for (k = 0; k < PITCH_WLPCBUFLEN; k++)
>+    wfdata->buffer[k] = 0.0;
>+
>+  for (k = 0; k < PITCH_WLPCORDER; k++) {
>+    wfdata->istate[k] = 0.0;
>+    wfdata->weostate[k] = 0.0;
>+    wfdata->whostate[k] = 0.0;
>+  }
>+
>+  /* next part should be in Matlab, writing to a global table */
>+  t = 0.5;
>+  denum = 1.0 / ((double)PITCH_WLPCWINLEN);
>+  denum2 = denum * denum;
>+  for (k = 0; k < PITCH_WLPCWINLEN; k++) {
>+    dtmp = PITCH_WLPCASYM * t * denum + (1 - PITCH_WLPCASYM) * t * t * denum2;
>+    dtmp *= 3.14159265;
>+    dtmp2 = sin(dtmp);
>+    wfdata->window[k] = dtmp2 * dtmp2;
>+    t++;
>+  }
>+}
>+
>+void WebRtcIsac_InitPitchAnalysis(PitchAnalysisStruct* State) {
>+  int k;
>+
>+  for (k = 0; k < PITCH_CORR_LEN2 + PITCH_CORR_STEP2 + PITCH_MAX_LAG / 2 -
>+                      PITCH_FRAME_LEN / 2 + 2;
>+       k++)
>+    State->dec_buffer[k] = 0.0;
>+  for (k = 0; k < 2 * ALLPASSSECTIONS + 1; k++)
>+    State->decimator_state[k] = 0.0;
>+  for (k = 0; k < 2; k++)
>+    State->hp_state[k] = 0.0;
>+  for (k = 0; k < QLOOKAHEAD; k++)
>+    State->whitened_buf[k] = 0.0;
>+  for (k = 0; k < QLOOKAHEAD; k++)
>+    State->inbuf[k] = 0.0;
>+
>+  WebRtcIsac_InitPitchFilter(&(State->PFstr_wght));
>+
>+  WebRtcIsac_InitPitchFilter(&(State->PFstr));
>+
>+  WebRtcIsac_InitWeightingFilter(&(State->Wghtstr));
>+}
>+
>+void WebRtcIsac_InitPreFilterbank(PreFiltBankstr* prefiltdata) {
>+  int k;
>+
>+  for (k = 0; k < QLOOKAHEAD; k++) {
>+    prefiltdata->INLABUF1[k] = 0;
>+    prefiltdata->INLABUF2[k] = 0;
>+
>+    prefiltdata->INLABUF1_float[k] = 0;
>+    prefiltdata->INLABUF2_float[k] = 0;
>+  }
>+  for (k = 0; k < 2 * (QORDER - 1); k++) {
>+    prefiltdata->INSTAT1[k] = 0;
>+    prefiltdata->INSTAT2[k] = 0;
>+    prefiltdata->INSTATLA1[k] = 0;
>+    prefiltdata->INSTATLA2[k] = 0;
>+
>+    prefiltdata->INSTAT1_float[k] = 0;
>+    prefiltdata->INSTAT2_float[k] = 0;
>+    prefiltdata->INSTATLA1_float[k] = 0;
>+    prefiltdata->INSTATLA2_float[k] = 0;
>+  }
>+
>+  /* High pass filter states */
>+  prefiltdata->HPstates[0] = 0.0;
>+  prefiltdata->HPstates[1] = 0.0;
>+
>+  prefiltdata->HPstates_float[0] = 0.0f;
>+  prefiltdata->HPstates_float[1] = 0.0f;
>+
>+  return;
>+}
>+
>+double WebRtcIsac_LevDurb(double* a, double* k, double* r, size_t order) {
>+  const double LEVINSON_EPS = 1.0e-10;
>+
>+  double sum, alpha;
>+  size_t m, m_h, i;
>+  alpha = 0;  // warning -DH
>+  a[0] = 1.0;
>+  if (r[0] < LEVINSON_EPS) { /* if r[0] <= 0, set LPC coeff. to zero */
>+    for (i = 0; i < order; i++) {
>+      k[i] = 0;
>+      a[i + 1] = 0;
>+    }
>+  } else {
>+    a[1] = k[0] = -r[1] / r[0];
>+    alpha = r[0] + r[1] * k[0];
>+    for (m = 1; m < order; m++) {
>+      sum = r[m + 1];
>+      for (i = 0; i < m; i++) {
>+        sum += a[i + 1] * r[m - i];
>+      }
>+      k[m] = -sum / alpha;
>+      alpha += k[m] * sum;
>+      m_h = (m + 1) >> 1;
>+      for (i = 0; i < m_h; i++) {
>+        sum = a[i + 1] + k[m] * a[m - i];
>+        a[m - i] += k[m] * a[i + 1];
>+        a[i + 1] = sum;
>+      }
>+      a[m + 1] = k[m];
>+    }
>+  }
>+  return alpha;
>+}
>+
>+/* The upper channel all-pass filter factors */
>+const float WebRtcIsac_kUpperApFactorsFloat[2] = {0.03470000000000f,
>+                                                  0.38260000000000f};
>+
>+/* The lower channel all-pass filter factors */
>+const float WebRtcIsac_kLowerApFactorsFloat[2] = {0.15440000000000f,
>+                                                  0.74400000000000f};
>+
>+/* This function performs all-pass filtering--a series of first order all-pass
>+ * sections are used to filter the input in a cascade manner.
>+ * The input is overwritten!!
>+ */
>+void WebRtcIsac_AllPassFilter2Float(float* InOut,
>+                                    const float* APSectionFactors,
>+                                    int lengthInOut,
>+                                    int NumberOfSections,
>+                                    float* FilterState) {
>+  int n, j;
>+  float temp;
>+  for (j = 0; j < NumberOfSections; j++) {
>+    for (n = 0; n < lengthInOut; n++) {
>+      temp = FilterState[j] + APSectionFactors[j] * InOut[n];
>+      FilterState[j] = -APSectionFactors[j] * temp + InOut[n];
>+      InOut[n] = temp;
>+    }
>+  }
>+}
>+
>+/* The number of composite all-pass filter factors */
>+#define NUMBEROFCOMPOSITEAPSECTIONS 4
>+
>+/* Function WebRtcIsac_SplitAndFilter
>+ * This function creates low-pass and high-pass decimated versions of part of
>+ the input signal, and part of the signal in the input 'lookahead buffer'.
>+
>+ INPUTS:
>+ in: a length FRAMESAMPLES array of input samples
>+ prefiltdata: input data structure containing the filterbank states
>+ and lookahead samples from the previous encoding
>+ iteration.
>+ OUTPUTS:
>+ LP: a FRAMESAMPLES_HALF array of low-pass filtered samples that
>+ have been phase equalized.  The first QLOOKAHEAD samples are
>+ based on the samples in the two prefiltdata->INLABUFx arrays
>+ each of length QLOOKAHEAD.
>+ The remaining FRAMESAMPLES_HALF-QLOOKAHEAD samples are based
>+ on the first FRAMESAMPLES_HALF-QLOOKAHEAD samples of the input
>+ array in[].
>+ HP: a FRAMESAMPLES_HALF array of high-pass filtered samples that
>+ have been phase equalized.  The first QLOOKAHEAD samples are
>+ based on the samples in the two prefiltdata->INLABUFx arrays
>+ each of length QLOOKAHEAD.
>+ The remaining FRAMESAMPLES_HALF-QLOOKAHEAD samples are based
>+ on the first FRAMESAMPLES_HALF-QLOOKAHEAD samples of the input
>+ array in[].
>+
>+ LP_la: a FRAMESAMPLES_HALF array of low-pass filtered samples.
>+ These samples are not phase equalized. They are computed
>+ from the samples in the in[] array.
>+ HP_la: a FRAMESAMPLES_HALF array of high-pass filtered samples
>+ that are not phase equalized. They are computed from
>+ the in[] vector.
>+ prefiltdata: this input data structure's filterbank state and
>+ lookahead sample buffers are updated for the next
>+ encoding iteration.
>+*/
>+void WebRtcIsac_SplitAndFilterFloat(float* pin,
>+                                    float* LP,
>+                                    float* HP,
>+                                    double* LP_la,
>+                                    double* HP_la,
>+                                    PreFiltBankstr* prefiltdata) {
>+  int k, n;
>+  float CompositeAPFilterState[NUMBEROFCOMPOSITEAPSECTIONS];
>+  float ForTransform_CompositeAPFilterState[NUMBEROFCOMPOSITEAPSECTIONS];
>+  float ForTransform_CompositeAPFilterState2[NUMBEROFCOMPOSITEAPSECTIONS];
>+  float tempinoutvec[FRAMESAMPLES + MAX_AR_MODEL_ORDER];
>+  float tempin_ch1[FRAMESAMPLES + MAX_AR_MODEL_ORDER];
>+  float tempin_ch2[FRAMESAMPLES + MAX_AR_MODEL_ORDER];
>+  float in[FRAMESAMPLES];
>+  float ftmp;
>+
>+  /* HPstcoeff_in = {a1, a2, b1 - b0 * a1, b2 - b0 * a2}; */
>+  static const float kHpStCoefInFloat[4] = {
>+      -1.94895953203325f, 0.94984516000000f, -0.05101826139794f,
>+      0.05015484000000f};
>+
>+  /* The composite all-pass filter factors */
>+  static const float WebRtcIsac_kCompositeApFactorsFloat[4] = {
>+      0.03470000000000f, 0.15440000000000f, 0.38260000000000f,
>+      0.74400000000000f};
>+
>+  // The matrix for transforming the backward composite state to upper channel
>+  // state.
>+  static const float WebRtcIsac_kTransform1Float[8] = {
>+      -0.00158678506084f, 0.00127157815343f, -0.00104805672709f,
>+      0.00084837248079f,  0.00134467983258f, -0.00107756549387f,
>+      0.00088814793277f,  -0.00071893072525f};
>+
>+  // The matrix for transforming the backward composite state to lower channel
>+  // state.
>+  static const float WebRtcIsac_kTransform2Float[8] = {
>+      -0.00170686041697f, 0.00136780109829f, -0.00112736532350f,
>+      0.00091257055385f,  0.00103094281812f, -0.00082615076557f,
>+      0.00068092756088f,  -0.00055119165484f};
>+
>+  /* High pass filter */
>+
>+  for (k = 0; k < FRAMESAMPLES; k++) {
>+    in[k] = pin[k] + kHpStCoefInFloat[2] * prefiltdata->HPstates_float[0] +
>+            kHpStCoefInFloat[3] * prefiltdata->HPstates_float[1];
>+    ftmp = pin[k] - kHpStCoefInFloat[0] * prefiltdata->HPstates_float[0] -
>+           kHpStCoefInFloat[1] * prefiltdata->HPstates_float[1];
>+    prefiltdata->HPstates_float[1] = prefiltdata->HPstates_float[0];
>+    prefiltdata->HPstates_float[0] = ftmp;
>+  }
>+
>+  /* First Channel */
>+
>+  /*initial state of composite filter is zero */
>+  for (k = 0; k < NUMBEROFCOMPOSITEAPSECTIONS; k++) {
>+    CompositeAPFilterState[k] = 0.0;
>+  }
>+  /* put every other sample of input into a temporary vector in reverse
>+   * (backward) order*/
>+  for (k = 0; k < FRAMESAMPLES_HALF; k++) {
>+    tempinoutvec[k] = in[FRAMESAMPLES - 1 - 2 * k];
>+  }
>+
>+  /* now all-pass filter the backwards vector.  Output values overwrite the
>+   * input vector. */
>+  WebRtcIsac_AllPassFilter2Float(
>+      tempinoutvec, WebRtcIsac_kCompositeApFactorsFloat, FRAMESAMPLES_HALF,
>+      NUMBEROFCOMPOSITEAPSECTIONS, CompositeAPFilterState);
>+
>+  /* save the backwards filtered output for later forward filtering,
>+     but write it in forward order*/
>+  for (k = 0; k < FRAMESAMPLES_HALF; k++) {
>+    tempin_ch1[FRAMESAMPLES_HALF + QLOOKAHEAD - 1 - k] = tempinoutvec[k];
>+  }
>+
>+  /* save the backwards filter state  becaue it will be transformed
>+     later into a forward state */
>+  for (k = 0; k < NUMBEROFCOMPOSITEAPSECTIONS; k++) {
>+    ForTransform_CompositeAPFilterState[k] = CompositeAPFilterState[k];
>+  }
>+
>+  /* now backwards filter the samples in the lookahead buffer. The samples were
>+     placed there in the encoding of the previous frame.  The output samples
>+     overwrite the input samples */
>+  WebRtcIsac_AllPassFilter2Float(
>+      prefiltdata->INLABUF1_float, WebRtcIsac_kCompositeApFactorsFloat,
>+      QLOOKAHEAD, NUMBEROFCOMPOSITEAPSECTIONS, CompositeAPFilterState);
>+
>+  /* save the output, but write it in forward order */
>+  /* write the lookahead samples for the next encoding iteration. Every other
>+     sample at the end of the input frame is written in reverse order for the
>+     lookahead length. Exported in the prefiltdata structure. */
>+  for (k = 0; k < QLOOKAHEAD; k++) {
>+    tempin_ch1[QLOOKAHEAD - 1 - k] = prefiltdata->INLABUF1_float[k];
>+    prefiltdata->INLABUF1_float[k] = in[FRAMESAMPLES - 1 - 2 * k];
>+  }
>+
>+  /* Second Channel.  This is exactly like the first channel, except that the
>+     even samples are now filtered instead (lower channel). */
>+  for (k = 0; k < NUMBEROFCOMPOSITEAPSECTIONS; k++) {
>+    CompositeAPFilterState[k] = 0.0;
>+  }
>+
>+  for (k = 0; k < FRAMESAMPLES_HALF; k++) {
>+    tempinoutvec[k] = in[FRAMESAMPLES - 2 - 2 * k];
>+  }
>+
>+  WebRtcIsac_AllPassFilter2Float(
>+      tempinoutvec, WebRtcIsac_kCompositeApFactorsFloat, FRAMESAMPLES_HALF,
>+      NUMBEROFCOMPOSITEAPSECTIONS, CompositeAPFilterState);
>+
>+  for (k = 0; k < FRAMESAMPLES_HALF; k++) {
>+    tempin_ch2[FRAMESAMPLES_HALF + QLOOKAHEAD - 1 - k] = tempinoutvec[k];
>+  }
>+
>+  for (k = 0; k < NUMBEROFCOMPOSITEAPSECTIONS; k++) {
>+    ForTransform_CompositeAPFilterState2[k] = CompositeAPFilterState[k];
>+  }
>+
>+  WebRtcIsac_AllPassFilter2Float(
>+      prefiltdata->INLABUF2_float, WebRtcIsac_kCompositeApFactorsFloat,
>+      QLOOKAHEAD, NUMBEROFCOMPOSITEAPSECTIONS, CompositeAPFilterState);
>+
>+  for (k = 0; k < QLOOKAHEAD; k++) {
>+    tempin_ch2[QLOOKAHEAD - 1 - k] = prefiltdata->INLABUF2_float[k];
>+    prefiltdata->INLABUF2_float[k] = in[FRAMESAMPLES - 2 - 2 * k];
>+  }
>+
>+  /* Transform filter states from backward to forward */
>+  /*At this point, each of the states of the backwards composite filters for the
>+    two channels are transformed into forward filtering states for the
>+    corresponding forward channel filters.  Each channel's forward filtering
>+    state from the previous
>+    encoding iteration is added to the transformed state to get a proper forward
>+    state */
>+
>+  /* So the existing NUMBEROFCOMPOSITEAPSECTIONS x 1 (4x1) state vector is
>+     multiplied by a NUMBEROFCHANNELAPSECTIONSxNUMBEROFCOMPOSITEAPSECTIONS (2x4)
>+     transform matrix to get the new state that is added to the previous 2x1
>+     input state */
>+
>+  for (k = 0; k < NUMBEROFCHANNELAPSECTIONS; k++) { /* k is row variable */
>+    for (n = 0; n < NUMBEROFCOMPOSITEAPSECTIONS;
>+         n++) { /* n is column variable */
>+      prefiltdata->INSTAT1_float[k] +=
>+          ForTransform_CompositeAPFilterState[n] *
>+          WebRtcIsac_kTransform1Float[k * NUMBEROFCHANNELAPSECTIONS + n];
>+      prefiltdata->INSTAT2_float[k] +=
>+          ForTransform_CompositeAPFilterState2[n] *
>+          WebRtcIsac_kTransform2Float[k * NUMBEROFCHANNELAPSECTIONS + n];
>+    }
>+  }
>+
>+  /*obtain polyphase components by forward all-pass filtering through each
>+   * channel */
>+  /* the backward filtered samples are now forward filtered with the
>+   * corresponding channel filters */
>+  /* The all pass filtering automatically updates the filter states which are
>+     exported in the prefiltdata structure */
>+  WebRtcIsac_AllPassFilter2Float(tempin_ch1, WebRtcIsac_kUpperApFactorsFloat,
>+                                 FRAMESAMPLES_HALF, NUMBEROFCHANNELAPSECTIONS,
>+                                 prefiltdata->INSTAT1_float);
>+  WebRtcIsac_AllPassFilter2Float(tempin_ch2, WebRtcIsac_kLowerApFactorsFloat,
>+                                 FRAMESAMPLES_HALF, NUMBEROFCHANNELAPSECTIONS,
>+                                 prefiltdata->INSTAT2_float);
>+
>+  /* Now Construct low-pass and high-pass signals as combinations of polyphase
>+   * components */
>+  for (k = 0; k < FRAMESAMPLES_HALF; k++) {
>+    LP[k] = 0.5f * (tempin_ch1[k] + tempin_ch2[k]); /* low pass signal*/
>+    HP[k] = 0.5f * (tempin_ch1[k] - tempin_ch2[k]); /* high pass signal*/
>+  }
>+
>+  /* Lookahead LP and HP signals */
>+  /* now create low pass and high pass signals of the input vector.  However, no
>+     backwards filtering is performed, and hence no phase equalization is
>+     involved. Also, the input contains some samples that are lookahead samples.
>+     The high pass and low pass signals that are created are used outside this
>+     function for analysis (not encoding) purposes */
>+
>+  /* set up input */
>+  for (k = 0; k < FRAMESAMPLES_HALF; k++) {
>+    tempin_ch1[k] = in[2 * k + 1];
>+    tempin_ch2[k] = in[2 * k];
>+  }
>+
>+  /* the input filter states are passed in and updated by the all-pass filtering
>+     routine and exported in the prefiltdata structure*/
>+  WebRtcIsac_AllPassFilter2Float(tempin_ch1, WebRtcIsac_kUpperApFactorsFloat,
>+                                 FRAMESAMPLES_HALF, NUMBEROFCHANNELAPSECTIONS,
>+                                 prefiltdata->INSTATLA1_float);
>+  WebRtcIsac_AllPassFilter2Float(tempin_ch2, WebRtcIsac_kLowerApFactorsFloat,
>+                                 FRAMESAMPLES_HALF, NUMBEROFCHANNELAPSECTIONS,
>+                                 prefiltdata->INSTATLA2_float);
>+
>+  for (k = 0; k < FRAMESAMPLES_HALF; k++) {
>+    LP_la[k] = (float)(0.5f * (tempin_ch1[k] + tempin_ch2[k]));  /*low pass */
>+    HP_la[k] = (double)(0.5f * (tempin_ch1[k] - tempin_ch2[k])); /* high pass */
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/isac_vad.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/isac_vad.h
>new file mode 100644
>index 00000000000..1aecfc40467
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/isac_vad.h
>@@ -0,0 +1,45 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ISAC_VAD_H_
>+#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ISAC_VAD_H_
>+
>+#include <stddef.h>
>+
>+#include "modules/audio_coding/codecs/isac/main/source/structs.h"
>+
>+void WebRtcIsac_InitPitchFilter(PitchFiltstr* pitchfiltdata);
>+void WebRtcIsac_InitPitchAnalysis(PitchAnalysisStruct* state);
>+void WebRtcIsac_InitPreFilterbank(PreFiltBankstr* prefiltdata);
>+
>+double WebRtcIsac_LevDurb(double* a, double* k, double* r, size_t order);
>+
>+/* The number of all-pass filter factors in an upper or lower channel*/
>+#define NUMBEROFCHANNELAPSECTIONS 2
>+
>+/* The upper channel all-pass filter factors */
>+extern const float WebRtcIsac_kUpperApFactorsFloat[2];
>+
>+/* The lower channel all-pass filter factors */
>+extern const float WebRtcIsac_kLowerApFactorsFloat[2];
>+
>+void WebRtcIsac_AllPassFilter2Float(float* InOut,
>+                                    const float* APSectionFactors,
>+                                    int lengthInOut,
>+                                    int NumberOfSections,
>+                                    float* FilterState);
>+void WebRtcIsac_SplitAndFilterFloat(float* in,
>+                                    float* LP,
>+                                    float* HP,
>+                                    double* LP_la,
>+                                    double* HP_la,
>+                                    PreFiltBankstr* prefiltdata);
>+
>+#endif  // MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ISAC_VAD_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/lattice.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/lattice.c
>index eabe7080457..d9d2d656659 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/lattice.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/lattice.c
>@@ -14,8 +14,6 @@
>  * contains the normalized lattice filter routines (MA and AR) for iSAC codec
>  *
>  */
>-#include "settings.h"
>-#include "codec.h"
> 
> #include <math.h>
> #include <memory.h>
>@@ -24,6 +22,9 @@
> #include <stdlib.h>
> #endif
> 
>+#include "modules/audio_coding/codecs/isac/main/source/settings.h"
>+#include "modules/audio_coding/codecs/isac/main/source/codec.h"
>+
> /* filter the signal using normalized lattice filter */
> /* MA filter */
> void WebRtcIsac_NormLatticeFilterMa(int orderCoef,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_analysis.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_analysis.c
>index 60fc25b98bb..0fda73bda63 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_analysis.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_analysis.c
>@@ -8,16 +8,15 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "lpc_analysis.h"
>-#include "settings.h"
>-#include "codec.h"
>-#include "entropy_coding.h"
>-
> #include <math.h>
> #include <string.h>
> 
>-#define LEVINSON_EPS    1.0e-10
>-
>+#include "modules/audio_coding/codecs/isac/main/source/lpc_analysis.h"
>+#include "modules/audio_coding/codecs/isac/main/source/settings.h"
>+#include "modules/audio_coding/codecs/isac/main/source/codec.h"
>+#include "modules/audio_coding/codecs/isac/main/source/entropy_coding.h"
>+#include "modules/audio_coding/codecs/isac/main/source/filter_functions.h"
>+#include "modules/audio_coding/codecs/isac/main/source/isac_vad.h"
> 
> /* window */
> /* Matlab generation code:
>@@ -75,45 +74,10 @@ static const double kLpcCorrWindow[WINLEN] = {
>   0.00155690, 0.00124918, 0.00094895, 0.00066112, 0.00039320, 0.00015881
> };
> 
>-double WebRtcIsac_LevDurb(double *a, double *k, double *r, size_t order)
>-{
>-
>-  double sum, alpha;
>-  size_t m, m_h, i;
>-  alpha = 0; //warning -DH
>-  a[0] = 1.0;
>-  if (r[0] < LEVINSON_EPS) { /* if r[0] <= 0, set LPC coeff. to zero */
>-    for (i = 0; i < order; i++) {
>-      k[i] = 0;
>-      a[i+1] = 0;
>-    }
>-  } else {
>-    a[1] = k[0] = -r[1]/r[0];
>-    alpha = r[0] + r[1] * k[0];
>-    for (m = 1; m < order; m++){
>-      sum = r[m + 1];
>-      for (i = 0; i < m; i++){
>-        sum += a[i+1] * r[m - i];
>-      }
>-      k[m] = -sum / alpha;
>-      alpha += k[m] * sum;
>-      m_h = (m + 1) >> 1;
>-      for (i = 0; i < m_h; i++){
>-        sum = a[i+1] + k[m] * a[m - i];
>-        a[m - i] += k[m] * a[i+1];
>-        a[i+1] = sum;
>-      }
>-      a[m+1] = k[m];
>-    }
>-  }
>-  return alpha;
>-}
>-
>-
>-//was static before, but didn't work with MEX file
>-void WebRtcIsac_GetVars(const double *input, const int16_t *pitchGains_Q12,
>-                       double *oldEnergy, double *varscale)
>-{
>+static void WebRtcIsac_GetVars(const double* input,
>+                               const int16_t* pitchGains_Q12,
>+                               double* oldEnergy,
>+                               double* varscale) {
>   double nrg[4], chng, pg;
>   int k;
> 
>@@ -162,12 +126,9 @@ void WebRtcIsac_GetVars(const double *input, const int16_t *pitchGains_Q12,
>   *oldEnergy = nrg[3];
> }
> 
>-void
>-WebRtcIsac_GetVarsUB(
>-    const double* input,
>-    double*       oldEnergy,
>-    double*       varscale)
>-{
>+static void WebRtcIsac_GetVarsUB(const double* input,
>+                                 double* oldEnergy,
>+                                 double* varscale) {
>   double nrg[4], chng;
>   int k;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_analysis.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_analysis.h
>index 1566ab0d40b..5503e2d49b7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_analysis.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_analysis.h
>@@ -18,33 +18,29 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_ANALYSIS_H_
> #define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_ANALYSIS_H_
> 
>-#include "settings.h"
>-#include "structs.h"
>-
>-double WebRtcIsac_LevDurb(double *a, double *k, double *r, size_t order);
>-
>-void WebRtcIsac_GetVars(const double *input, const int16_t *pitchGains_Q12,
>-                       double *oldEnergy, double *varscale);
>-
>-void WebRtcIsac_GetLpcCoefLb(double *inLo, double *inHi, MaskFiltstr *maskdata,
>-                             double signal_noise_ratio, const int16_t *pitchGains_Q12,
>-                             double *lo_coeff, double *hi_coeff);
>-
>-
>-void WebRtcIsac_GetLpcGain(
>-    double         signal_noise_ratio,
>-    const double*  filtCoeffVecs,
>-    int            numVecs,
>-    double*        gain,
>-    double         corrLo[][UB_LPC_ORDER + 1],
>-    const double*  varscale);
>-
>-void WebRtcIsac_GetLpcCoefUb(
>-    double*      inSignal,
>-    MaskFiltstr* maskdata,
>-    double*      lpCoeff,
>-    double       corr[][UB_LPC_ORDER + 1],
>-    double*      varscale,
>-    int16_t  bandwidth);
>+#include "modules/audio_coding/codecs/isac/main/source/settings.h"
>+#include "modules/audio_coding/codecs/isac/main/source/structs.h"
>+
>+void WebRtcIsac_GetLpcCoefLb(double* inLo,
>+                             double* inHi,
>+                             MaskFiltstr* maskdata,
>+                             double signal_noise_ratio,
>+                             const int16_t* pitchGains_Q12,
>+                             double* lo_coeff,
>+                             double* hi_coeff);
>+
>+void WebRtcIsac_GetLpcGain(double signal_noise_ratio,
>+                           const double* filtCoeffVecs,
>+                           int numVecs,
>+                           double* gain,
>+                           double corrLo[][UB_LPC_ORDER + 1],
>+                           const double* varscale);
>+
>+void WebRtcIsac_GetLpcCoefUb(double* inSignal,
>+                             MaskFiltstr* maskdata,
>+                             double* lpCoeff,
>+                             double corr[][UB_LPC_ORDER + 1],
>+                             double* varscale,
>+                             int16_t bandwidth);
> 
> #endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_ANALYIS_H_ */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_gain_swb_tables.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_gain_swb_tables.c
>index d9073d5e9d8..670754065f4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_gain_swb_tables.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_gain_swb_tables.c
>@@ -16,9 +16,8 @@
>  *
>  */
> 
>-#include "lpc_gain_swb_tables.h"
>-#include "settings.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include "modules/audio_coding/codecs/isac/main/source/lpc_gain_swb_tables.h"
>+#include "modules/audio_coding/codecs/isac/main/source/settings.h"
> 
> const double WebRtcIsac_kQSizeLpcGain = 0.100000;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_gain_swb_tables.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_gain_swb_tables.h
>index 5ea537575d0..39c4a24ef40 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_gain_swb_tables.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_gain_swb_tables.h
>@@ -19,8 +19,9 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_GAIN_SWB_TABLES_H_
> #define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_GAIN_SWB_TABLES_H_
> 
>-#include "settings.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include <stdint.h>
>+
>+#include "modules/audio_coding/codecs/isac/main/source/settings.h"
> 
> extern const double WebRtcIsac_kQSizeLpcGain;
> 
>@@ -46,4 +47,4 @@ extern const uint16_t* WebRtcIsac_kLpcGainCdfMat[SUBFRAMES];
> 
> extern const double WebRtcIsac_kLpcGainDecorrMat[SUBFRAMES][SUBFRAMES];
> 
>-#endif // MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_GAIN_SWB_TABLES_H_
>+#endif  // MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_GAIN_SWB_TABLES_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb12_tables.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb12_tables.c
>index 6e64e414790..e3600a7fab4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb12_tables.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb12_tables.c
>@@ -16,9 +16,8 @@
>  *
>  */
> 
>-#include "lpc_shape_swb12_tables.h"
>-#include "settings.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include "modules/audio_coding/codecs/isac/main/source/lpc_shape_swb12_tables.h"
>+#include "modules/audio_coding/codecs/isac/main/source/settings.h"
> 
> /*
> * Mean value of LAR
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb12_tables.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb12_tables.h
>index c703a6cb4ba..7448a1e76be 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb12_tables.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb12_tables.h
>@@ -19,29 +19,30 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_SHAPE_SWB12_TABLES_H_
> #define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_SHAPE_SWB12_TABLES_H_
> 
>-#include "settings.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include <stdint.h>
>+
>+#include "modules/audio_coding/codecs/isac/main/source/settings.h"
> 
> extern const double WebRtcIsac_kMeanLarUb12[UB_LPC_ORDER];
> 
> extern const double WebRtcIsac_kMeanLpcGain;
> 
>-extern const double WebRtcIsac_kIntraVecDecorrMatUb12[UB_LPC_ORDER][UB_LPC_ORDER];
>+extern const double WebRtcIsac_kIntraVecDecorrMatUb12[UB_LPC_ORDER]
>+                                                     [UB_LPC_ORDER];
> 
>-extern const double WebRtcIsac_kInterVecDecorrMatUb12
>-[UB_LPC_VEC_PER_FRAME][UB_LPC_VEC_PER_FRAME];
>+extern const double WebRtcIsac_kInterVecDecorrMatUb12[UB_LPC_VEC_PER_FRAME]
>+                                                     [UB_LPC_VEC_PER_FRAME];
> 
> extern const double WebRtcIsac_kLpcShapeQStepSizeUb12;
> 
>-extern const double WebRtcIsac_kLpcShapeLeftRecPointUb12
>-[UB_LPC_ORDER*UB_LPC_VEC_PER_FRAME];
>-
>+extern const double
>+    WebRtcIsac_kLpcShapeLeftRecPointUb12[UB_LPC_ORDER * UB_LPC_VEC_PER_FRAME];
> 
>-extern const int16_t WebRtcIsac_kLpcShapeNumRecPointUb12
>-[UB_LPC_ORDER * UB_LPC_VEC_PER_FRAME];
>+extern const int16_t
>+    WebRtcIsac_kLpcShapeNumRecPointUb12[UB_LPC_ORDER * UB_LPC_VEC_PER_FRAME];
> 
>-extern const uint16_t WebRtcIsac_kLpcShapeEntropySearchUb12
>-[UB_LPC_ORDER * UB_LPC_VEC_PER_FRAME];
>+extern const uint16_t
>+    WebRtcIsac_kLpcShapeEntropySearchUb12[UB_LPC_ORDER * UB_LPC_VEC_PER_FRAME];
> 
> extern const uint16_t WebRtcIsac_kLpcShapeCdfVec0Ub12[14];
> 
>@@ -59,7 +60,7 @@ extern const uint16_t WebRtcIsac_kLpcShapeCdfVec6Ub12[33];
> 
> extern const uint16_t WebRtcIsac_kLpcShapeCdfVec7Ub12[49];
> 
>-extern const uint16_t* WebRtcIsac_kLpcShapeCdfMatUb12
>-[UB_LPC_ORDER * UB_LPC_VEC_PER_FRAME];
>+extern const uint16_t*
>+    WebRtcIsac_kLpcShapeCdfMatUb12[UB_LPC_ORDER * UB_LPC_VEC_PER_FRAME];
> 
>-#endif // MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_SHAPE_SWB12_TABLES_H_
>+#endif  // MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_SHAPE_SWB12_TABLES_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb16_tables.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb16_tables.c
>index 852d07fd736..59617fd2746 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb16_tables.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb16_tables.c
>@@ -16,9 +16,8 @@
>  *
>  */
> 
>-#include "lpc_shape_swb16_tables.h"
>-#include "settings.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include "modules/audio_coding/codecs/isac/main/source/lpc_shape_swb16_tables.h"
>+#include "modules/audio_coding/codecs/isac/main/source/settings.h"
> 
> /*
> * Mean value of LAR
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb16_tables.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb16_tables.h
>index 17a77d7c1d3..51101db936b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb16_tables.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_shape_swb16_tables.h
>@@ -19,15 +19,17 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_SHAPE_SWB16_TABLES_H_
> #define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_SHAPE_SWB16_TABLES_H_
> 
>-#include "settings.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include <stdint.h>
>+
>+#include "modules/audio_coding/codecs/isac/main/source/settings.h"
> 
> extern const double WebRtcIsac_kMeanLarUb16[UB_LPC_ORDER];
> 
>-extern const double WebRtcIsac_kIintraVecDecorrMatUb16[UB_LPC_ORDER][UB_LPC_ORDER];
>+extern const double WebRtcIsac_kIintraVecDecorrMatUb16[UB_LPC_ORDER]
>+                                                      [UB_LPC_ORDER];
> 
>-extern const double WebRtcIsac_kInterVecDecorrMatUb16
>-[UB16_LPC_VEC_PER_FRAME][UB16_LPC_VEC_PER_FRAME];
>+extern const double WebRtcIsac_kInterVecDecorrMatUb16[UB16_LPC_VEC_PER_FRAME]
>+                                                     [UB16_LPC_VEC_PER_FRAME];
> 
> extern const uint16_t WebRtcIsac_kLpcShapeCdfVec01Ub16[14];
> 
>@@ -61,18 +63,19 @@ extern const uint16_t WebRtcIsac_kLpcShapeCdfVec01Ub165[34];
> 
> extern const uint16_t WebRtcIsac_kLpcShapeCdfVec01Ub166[71];
> 
>-extern const uint16_t* WebRtcIsac_kLpcShapeCdfMatUb16
>-[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME];
>+extern const uint16_t*
>+    WebRtcIsac_kLpcShapeCdfMatUb16[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME];
> 
>-extern const double WebRtcIsac_kLpcShapeLeftRecPointUb16
>-[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME];
>+extern const double
>+    WebRtcIsac_kLpcShapeLeftRecPointUb16[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME];
> 
>-extern const int16_t WebRtcIsac_kLpcShapeNumRecPointUb16
>-[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME];
>+extern const int16_t
>+    WebRtcIsac_kLpcShapeNumRecPointUb16[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME];
> 
>-extern const uint16_t WebRtcIsac_kLpcShapeEntropySearchUb16
>-[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME];
>+extern const uint16_t
>+    WebRtcIsac_kLpcShapeEntropySearchUb16[UB_LPC_ORDER *
>+                                          UB16_LPC_VEC_PER_FRAME];
> 
> extern const double WebRtcIsac_kLpcShapeQStepSizeUb16;
> 
>-#endif // MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_SHAPE_SWB16_TABLES_H_
>+#endif  // MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_SHAPE_SWB16_TABLES_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_tables.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_tables.c
>index 909809bc678..461b92eb8a4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_tables.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_tables.c
>@@ -10,8 +10,8 @@
> 
> /* coding tables for the KLT coefficients */
> 
>-#include "lpc_tables.h"
>-#include "settings.h"
>+#include "modules/audio_coding/codecs/isac/main/source/lpc_tables.h"
>+#include "modules/audio_coding/codecs/isac/main/source/settings.h"
> 
> /* cdf array for model indicator */
> const uint16_t WebRtcIsac_kQKltModelCdf[4] = {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_tables.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_tables.h
>index 9412f2e389b..2d92dfaaa55 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_tables.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/lpc_tables.h
>@@ -18,31 +18,31 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_TABLES_H_
> #define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_TABLES_H_
> 
>-#include "structs.h"
>+#include "modules/audio_coding/codecs/isac/main/source/structs.h"
> 
>-#include "settings.h"
>+#include "modules/audio_coding/codecs/isac/main/source/settings.h"
> 
>-#define KLT_STEPSIZE         1.00000000
>-#define KLT_NUM_AVG_GAIN     0
>-#define KLT_NUM_AVG_SHAPE    0
>-#define KLT_NUM_MODELS  3
>-#define LPC_GAIN_SCALE     4.000f
>-#define LPC_LOBAND_SCALE   2.100f
>-#define LPC_LOBAND_ORDER   ORDERLO
>-#define LPC_HIBAND_SCALE   0.450f
>-#define LPC_HIBAND_ORDER   ORDERHI
>-#define LPC_GAIN_ORDER     2
>+#define KLT_STEPSIZE 1.00000000
>+#define KLT_NUM_AVG_GAIN 0
>+#define KLT_NUM_AVG_SHAPE 0
>+#define KLT_NUM_MODELS 3
>+#define LPC_GAIN_SCALE 4.000f
>+#define LPC_LOBAND_SCALE 2.100f
>+#define LPC_LOBAND_ORDER ORDERLO
>+#define LPC_HIBAND_SCALE 0.450f
>+#define LPC_HIBAND_ORDER ORDERHI
>+#define LPC_GAIN_ORDER 2
> 
>-#define LPC_SHAPE_ORDER    (LPC_LOBAND_ORDER + LPC_HIBAND_ORDER)
>+#define LPC_SHAPE_ORDER (LPC_LOBAND_ORDER + LPC_HIBAND_ORDER)
> 
>-#define KLT_ORDER_GAIN     (LPC_GAIN_ORDER * SUBFRAMES)
>-#define KLT_ORDER_SHAPE    (LPC_SHAPE_ORDER * SUBFRAMES)
>+#define KLT_ORDER_GAIN (LPC_GAIN_ORDER * SUBFRAMES)
>+#define KLT_ORDER_SHAPE (LPC_SHAPE_ORDER * SUBFRAMES)
> 
> /* cdf array for model indicator */
>-extern const uint16_t WebRtcIsac_kQKltModelCdf[KLT_NUM_MODELS+1];
>+extern const uint16_t WebRtcIsac_kQKltModelCdf[KLT_NUM_MODELS + 1];
> 
> /* pointer to cdf array for model indicator */
>-extern const uint16_t *WebRtcIsac_kQKltModelCdfPtr[1];
>+extern const uint16_t* WebRtcIsac_kQKltModelCdfPtr[1];
> 
> /* initial cdf index for decoder of model indicator */
> extern const uint16_t WebRtcIsac_kQKltModelInitIndex[1];
>@@ -78,9 +78,9 @@ extern const uint16_t WebRtcIsac_kQKltCdfGain[404];
> extern const uint16_t WebRtcIsac_kQKltCdfShape[686];
> 
> /* pointers to cdf tables for quantizer indices */
>-extern const uint16_t *WebRtcIsac_kQKltCdfPtrGain[12];
>+extern const uint16_t* WebRtcIsac_kQKltCdfPtrGain[12];
> 
>-extern const uint16_t *WebRtcIsac_kQKltCdfPtrShape[108];
>+extern const uint16_t* WebRtcIsac_kQKltCdfPtrShape[108];
> 
> /* left KLT transforms */
> extern const double WebRtcIsac_kKltT1Gain[4];
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/os_specific_inline.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/os_specific_inline.h
>index 597dc21d1cf..fe9afa4ba29 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/os_specific_inline.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/os_specific_inline.h
>@@ -8,12 +8,12 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-
> #ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_OS_SPECIFIC_INLINE_H_
> #define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_OS_SPECIFIC_INLINE_H_
> 
> #include <math.h>
>-#include "typedefs.h"  // NOLINT(build/include)
>+
>+#include "rtc_base/system/arch.h"
> 
> #if defined(WEBRTC_POSIX)
> #define WebRtcIsac_lrint lrint
>@@ -24,11 +24,12 @@ static __inline long int WebRtcIsac_lrint(double x_dbl) {
>   __asm {
>     fld x_dbl
>     fistp x_int
>-  };
>+  }
>+  ;
> 
>   return x_int;
> }
>-#else // Do a slow but correct implementation of lrint
>+#else  // Do a slow but correct implementation of lrint
> 
> static __inline long int WebRtcIsac_lrint(double x_dbl) {
>   long int x_int;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/pitch_estimator.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/pitch_estimator.c
>index 090b94ca00c..8a19ac17109 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/pitch_estimator.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/pitch_estimator.c
>@@ -8,7 +8,7 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "pitch_estimator.h"
>+#include "modules/audio_coding/codecs/isac/main/source/pitch_estimator.h"
> 
> #include <math.h>
> #include <memory.h>
>@@ -17,6 +17,10 @@
> #include <stdlib.h>
> #endif
> 
>+#include "modules/audio_coding/codecs/isac/main/source/filter_functions.h"
>+#include "modules/audio_coding/codecs/isac/main/source/pitch_filter.h"
>+#include "rtc_base/system/ignore_warnings.h"
>+
> static const double kInterpolWin[8] = {-0.00067556028640,  0.02184247643159, -0.12203175715679,  0.60086484101160,
>                                        0.60086484101160, -0.12203175715679,  0.02184247643159, -0.00067556028640};
> 
>@@ -122,13 +126,56 @@ static void PCorr(const double *in, double *outcorr)
>   }
> }
> 
>+static void WebRtcIsac_AllpassFilterForDec(double* InOut,
>+                                           const double* APSectionFactors,
>+                                           size_t lengthInOut,
>+                                           double* FilterState) {
>+  // This performs all-pass filtering--a series of first order all-pass
>+  // sections are used to filter the input in a cascade manner.
>+  size_t n, j;
>+  double temp;
>+  for (j = 0; j < ALLPASSSECTIONS; j++) {
>+    for (n = 0; n < lengthInOut; n += 2) {
>+      temp = InOut[n];  // store input
>+      InOut[n] = FilterState[j] + APSectionFactors[j] * temp;
>+      FilterState[j] = -APSectionFactors[j] * InOut[n] + temp;
>+    }
>+  }
>+}
> 
>-void WebRtcIsac_InitializePitch(const double *in,
>-                                const double old_lag,
>-                                const double old_gain,
>-                                PitchAnalysisStruct *State,
>-                                double *lags)
>-{
>+static void WebRtcIsac_DecimateAllpass(
>+    const double* in,
>+    double* state_in,  // array of size: 2*ALLPASSSECTIONS+1
>+    size_t N,          // number of input samples
>+    double* out) {     // array of size N/2
>+
>+  static const double APupper[ALLPASSSECTIONS] = {0.0347, 0.3826};
>+  static const double APlower[ALLPASSSECTIONS] = {0.1544, 0.744};
>+
>+  size_t n;
>+  double data_vec[PITCH_FRAME_LEN];
>+
>+  /* copy input */
>+  memcpy(data_vec + 1, in, sizeof(double) * (N - 1));
>+
>+  data_vec[0] = state_in[2 * ALLPASSSECTIONS];  // the z^(-1) state
>+  state_in[2 * ALLPASSSECTIONS] = in[N - 1];
>+
>+  WebRtcIsac_AllpassFilterForDec(data_vec + 1, APupper, N, state_in);
>+  WebRtcIsac_AllpassFilterForDec(data_vec, APlower, N,
>+                                 state_in + ALLPASSSECTIONS);
>+
>+  for (n = 0; n < N / 2; n++)
>+    out[n] = data_vec[2 * n] + data_vec[2 * n + 1];
>+}
>+
>+RTC_PUSH_IGNORING_WFRAME_LARGER_THAN()
>+
>+static void WebRtcIsac_InitializePitch(const double* in,
>+                                       const double old_lag,
>+                                       const double old_gain,
>+                                       PitchAnalysisStruct* State,
>+                                       double* lags) {
>   double buf_dec[PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2+2];
>   double ratio, log_lag, gain_bias;
>   double bias;
>@@ -449,7 +496,7 @@ void WebRtcIsac_InitializePitch(const double *in,
>   }
> }
> 
>-
>+RTC_POP_IGNORING_WFRAME_LARGER_THAN()
> 
> /* create weighting matrix by orthogonalizing a basis of polynomials of increasing order
>  * t = (0:4)';
>@@ -464,6 +511,29 @@ static const double kWeight[5][5] = {
>   { 0.01714285714286,   0.05142857142857,  -0.05714285714286,  -0.30857142857143,  0.29714285714286}
> };
> 
>+/* second order high-pass filter */
>+static void WebRtcIsac_Highpass(const double* in,
>+                         double* out,
>+                         double* state,
>+                         size_t N) {
>+  /* create high-pass filter ocefficients
>+   * z = 0.998 * exp(j*2*pi*35/8000);
>+   * p = 0.94 * exp(j*2*pi*140/8000);
>+   * HP_b = [1, -2*real(z), abs(z)^2];
>+   * HP_a = [1, -2*real(p), abs(p)^2]; */
>+  static const double a_coef[2] = { 1.86864659625574, -0.88360000000000};
>+  static const double b_coef[2] = {-1.99524591718270,  0.99600400000000};
>+
>+  size_t k;
>+
>+  for (k=0; k<N; k++) {
>+    *out = *in + state[1];
>+    state[1] = state[0] + b_coef[0] * *in + a_coef[0] * *out;
>+    state[0] = b_coef[1] * *in++ + a_coef[1] * *out++;
>+  }
>+}
>+
>+RTC_PUSH_IGNORING_WFRAME_LARGER_THAN()
> 
> void WebRtcIsac_PitchAnalysis(const double *in,               /* PITCH_FRAME_LEN samples */
>                               double *out,                    /* PITCH_FRAME_LEN+QLOOKAHEAD samples */
>@@ -621,3 +691,5 @@ void WebRtcIsac_PitchAnalysis(const double *in,               /* PITCH_FRAME_LEN
>   for (k = 0; k < QLOOKAHEAD; k++)
>     State->inbuf[k] = inbuf[k + PITCH_FRAME_LEN];
> }
>+
>+RTC_POP_IGNORING_WFRAME_LARGER_THAN()
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/pitch_estimator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/pitch_estimator.h
>index 96e1e17ee42..4ab78c20ad3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/pitch_estimator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/pitch_estimator.h
>@@ -18,58 +18,15 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_ESTIMATOR_H_
> #define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_ESTIMATOR_H_
> 
>-#include "structs.h"
>+#include <stddef.h>
> 
>+#include "modules/audio_coding/codecs/isac/main/source/structs.h"
> 
>-
>-void WebRtcIsac_PitchAnalysis(const double *in,               /* PITCH_FRAME_LEN samples */
>-                              double *out,                    /* PITCH_FRAME_LEN+QLOOKAHEAD samples */
>-                              PitchAnalysisStruct *State,
>-                              double *lags,
>-                              double *gains);
>-
>-void WebRtcIsac_InitializePitch(const double *in,
>-                                const double old_lag,
>-                                const double old_gain,
>-                                PitchAnalysisStruct *State,
>-                                double *lags);
>-
>-void WebRtcIsac_PitchfilterPre(double *indat,
>-                               double *outdat,
>-                               PitchFiltstr *pfp,
>-                               double *lags,
>-                               double *gains);
>-
>-void WebRtcIsac_PitchfilterPost(double *indat,
>-                                double *outdat,
>-                                PitchFiltstr *pfp,
>-                                double *lags,
>-                                double *gains);
>-
>-void WebRtcIsac_PitchfilterPre_la(double *indat,
>-                                  double *outdat,
>-                                  PitchFiltstr *pfp,
>-                                  double *lags,
>-                                  double *gains);
>-
>-void WebRtcIsac_PitchfilterPre_gains(double *indat,
>-                                     double *outdat,
>-                                     double out_dG[][PITCH_FRAME_LEN + QLOOKAHEAD],
>-                                     PitchFiltstr *pfp,
>-                                     double *lags,
>-                                     double *gains);
>-
>-void WebRtcIsac_WeightingFilter(const double *in, double *weiout, double *whiout, WeightFiltstr *wfdata);
>-
>-void WebRtcIsac_Highpass(const double *in,
>-                         double *out,
>-                         double *state,
>-                         size_t N);
>-
>-void WebRtcIsac_DecimateAllpass(const double *in,
>-                                double *state_in,  /* array of size:
>-                                                    *     2*ALLPASSSECTIONS+1 */
>-                                size_t N,          /* number of input samples */
>-                                double *out);      /* array of size N/2 */
>+void WebRtcIsac_PitchAnalysis(
>+    const double* in, /* PITCH_FRAME_LEN samples */
>+    double* out,      /* PITCH_FRAME_LEN+QLOOKAHEAD samples */
>+    PitchAnalysisStruct* State,
>+    double* lags,
>+    double* gains);
> 
> #endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_ESTIMATOR_H_ */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/pitch_filter.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/pitch_filter.c
>index 24ec63acdf1..61cd533ab35 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/pitch_filter.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/pitch_filter.c
>@@ -8,14 +8,12 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "pitch_estimator.h"
>-
> #include <math.h>
> #include <memory.h>
> #include <stdlib.h>
> 
>-#include "os_specific_inline.h"
>-
>+#include "modules/audio_coding/codecs/isac/main/source/pitch_estimator.h"
>+#include "modules/audio_coding/codecs/isac/main/source/os_specific_inline.h"
> #include "rtc_base/compile_assert_c.h"
> 
> /*
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/pitch_filter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/pitch_filter.h
>new file mode 100644
>index 00000000000..9a232de87b8
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/pitch_filter.h
>@@ -0,0 +1,42 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_FILTER_H_
>+#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_FILTER_H_
>+
>+#include "modules/audio_coding/codecs/isac/main/source/structs.h"
>+
>+void WebRtcIsac_PitchfilterPre(double* indat,
>+                               double* outdat,
>+                               PitchFiltstr* pfp,
>+                               double* lags,
>+                               double* gains);
>+
>+void WebRtcIsac_PitchfilterPost(double* indat,
>+                                double* outdat,
>+                                PitchFiltstr* pfp,
>+                                double* lags,
>+                                double* gains);
>+
>+void WebRtcIsac_PitchfilterPre_la(double* indat,
>+                                  double* outdat,
>+                                  PitchFiltstr* pfp,
>+                                  double* lags,
>+                                  double* gains);
>+
>+void WebRtcIsac_PitchfilterPre_gains(
>+    double* indat,
>+    double* outdat,
>+    double out_dG[][PITCH_FRAME_LEN + QLOOKAHEAD],
>+    PitchFiltstr* pfp,
>+    double* lags,
>+    double* gains);
>+
>+#endif  // MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_FILTER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/pitch_gain_tables.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/pitch_gain_tables.c
>index 947d3e76362..080432c3a52 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/pitch_gain_tables.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/pitch_gain_tables.c
>@@ -8,9 +8,8 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "pitch_gain_tables.h"
>-
>-#include "settings.h"
>+#include "modules/audio_coding/codecs/isac/main/source/pitch_gain_tables.h"
>+#include "modules/audio_coding/codecs/isac/main/source/settings.h"
> 
> /* header file for coding tables for the pitch filter side-info in the entropy coder */
> /********************* Pitch Filter Gain Coefficient Tables ************************/
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/pitch_gain_tables.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/pitch_gain_tables.h
>index fe506ee284b..145fd4e6aae 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/pitch_gain_tables.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/pitch_gain_tables.h
>@@ -11,17 +11,20 @@
> /*
>  * pitch_gain_tables.h
>  *
>- * This file contains tables for the pitch filter side-info in the entropy coder.
>+ * This file contains tables for the pitch filter side-info in the entropy
>+ * coder.
>  *
>  */
> 
> #ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_GAIN_TABLES_H_
> #define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_GAIN_TABLES_H_
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include <stdint.h>
> 
>-/* header file for coding tables for the pitch filter side-info in the entropy coder */
>-/********************* Pitch Filter Gain Coefficient Tables ************************/
>+/* header file for coding tables for the pitch filter side-info in the entropy
>+ * coder */
>+/********************* Pitch Filter Gain Coefficient Tables
>+ * ************************/
> /* cdf for quantized pitch filter gains */
> extern const uint16_t WebRtcIsac_kQPitchGainCdf[255];
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/pitch_lag_tables.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/pitch_lag_tables.c
>index f845a225503..57d12021acb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/pitch_lag_tables.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/pitch_lag_tables.c
>@@ -8,8 +8,8 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "pitch_lag_tables.h"
>-#include "settings.h"
>+#include "modules/audio_coding/codecs/isac/main/source/pitch_lag_tables.h"
>+#include "modules/audio_coding/codecs/isac/main/source/settings.h"
> 
> /* header file for coding tables for the pitch filter side-info in the entropy coder */
> /********************* Pitch Filter Gain Coefficient Tables ************************/
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/pitch_lag_tables.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/pitch_lag_tables.h
>index 6a57c872648..b48e358a5a8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/pitch_lag_tables.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/pitch_lag_tables.h
>@@ -11,16 +11,20 @@
> /*
>  * pitch_lag_tables.h
>  *
>- * This file contains tables for the pitch filter side-info in the entropy coder.
>+ * This file contains tables for the pitch filter side-info in the entropy
>+ * coder.
>  *
>  */
> 
> #ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_LAG_TABLES_H_
> #define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_LAG_TABLES_H_
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>-/* header file for coding tables for the pitch filter side-info in the entropy coder */
>-/********************* Pitch Filter Lag Coefficient Tables ************************/
>+#include <stdint.h>
>+
>+/* header file for coding tables for the pitch filter side-info in the entropy
>+ * coder */
>+/********************* Pitch Filter Lag Coefficient Tables
>+ * ************************/
> 
> /* tables for use with small pitch gain */
> 
>@@ -30,7 +34,7 @@ extern const uint16_t WebRtcIsac_kQPitchLagCdf2Lo[20];
> extern const uint16_t WebRtcIsac_kQPitchLagCdf3Lo[2];
> extern const uint16_t WebRtcIsac_kQPitchLagCdf4Lo[10];
> 
>-extern const uint16_t *WebRtcIsac_kQPitchLagCdfPtrLo[4];
>+extern const uint16_t* WebRtcIsac_kQPitchLagCdfPtrLo[4];
> 
> /* size of first cdf table */
> extern const uint16_t WebRtcIsac_kQPitchLagCdfSizeLo[1];
>@@ -49,7 +53,6 @@ extern const double WebRtcIsac_kQMeanLag4Lo[9];
> 
> extern const double WebRtcIsac_kQPitchLagStepsizeLo;
> 
>-
> /* tables for use with medium pitch gain */
> 
> /* cdfs for quantized pitch lags */
>@@ -58,7 +61,7 @@ extern const uint16_t WebRtcIsac_kQPitchLagCdf2Mid[36];
> extern const uint16_t WebRtcIsac_kQPitchLagCdf3Mid[2];
> extern const uint16_t WebRtcIsac_kQPitchLagCdf4Mid[20];
> 
>-extern const uint16_t *WebRtcIsac_kQPitchLagCdfPtrMid[4];
>+extern const uint16_t* WebRtcIsac_kQPitchLagCdfPtrMid[4];
> 
> /* size of first cdf table */
> extern const uint16_t WebRtcIsac_kQPitchLagCdfSizeMid[1];
>@@ -77,7 +80,6 @@ extern const double WebRtcIsac_kQMeanLag4Mid[19];
> 
> extern const double WebRtcIsac_kQPitchLagStepsizeMid;
> 
>-
> /* tables for use with large pitch gain */
> 
> /* cdfs for quantized pitch lags */
>@@ -86,7 +88,7 @@ extern const uint16_t WebRtcIsac_kQPitchLagCdf2Hi[68];
> extern const uint16_t WebRtcIsac_kQPitchLagCdf3Hi[2];
> extern const uint16_t WebRtcIsac_kQPitchLagCdf4Hi[35];
> 
>-extern const uint16_t *WebRtcIsac_kQPitchLagCdfPtrHi[4];
>+extern const uint16_t* WebRtcIsac_kQPitchLagCdfPtrHi[4];
> 
> /* size of first cdf table */
> extern const uint16_t WebRtcIsac_kQPitchLagCdfSizeHi[1];
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/settings.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/settings.h
>index c08d72ffe39..abce90c4f58 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/settings.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/settings.h
>@@ -19,187 +19,178 @@
> #define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_SETTINGS_H_
> 
> /* sampling frequency (Hz) */
>-#define FS                                      16000
>+#define FS 16000
> 
> /* number of samples per frame (either 320 (20ms), 480 (30ms) or 960 (60ms)) */
>-#define INITIAL_FRAMESAMPLES     960
>-
>-
>-#define MAXFFTSIZE 2048
>-#define NFACTOR 11
>-
>-
>+#define INITIAL_FRAMESAMPLES 960
> 
> /* do not modify the following; this will have to be modified if we
>  * have a 20ms framesize option */
> /**********************************************************************/
> /* miliseconds */
>-#define FRAMESIZE                               30
>+#define FRAMESIZE 30
> /* number of samples per frame processed in the encoder, 480 */
>-#define FRAMESAMPLES                            480 /* ((FRAMESIZE*FS)/1000) */
>-#define FRAMESAMPLES_HALF      240
>-#define FRAMESAMPLES_QUARTER                    120
>+#define FRAMESAMPLES 480 /* ((FRAMESIZE*FS)/1000) */
>+#define FRAMESAMPLES_HALF 240
>+#define FRAMESAMPLES_QUARTER 120
> /**********************************************************************/
> 
>-
>-
> /* max number of samples per frame (= 60 ms frame) */
>-#define MAX_FRAMESAMPLES      960
>-#define MAX_SWBFRAMESAMPLES                     (MAX_FRAMESAMPLES * 2)
>+#define MAX_FRAMESAMPLES 960
>+#define MAX_SWBFRAMESAMPLES (MAX_FRAMESAMPLES * 2)
> /* number of samples per 10ms frame */
>-#define FRAMESAMPLES_10ms                       ((10*FS)/1000)
>-#define SWBFRAMESAMPLES_10ms                    (FRAMESAMPLES_10ms * 2)
>+#define FRAMESAMPLES_10ms ((10 * FS) / 1000)
>+#define SWBFRAMESAMPLES_10ms (FRAMESAMPLES_10ms * 2)
> /* number of samples in 30 ms frame */
>-#define FRAMESAMPLES_30ms            480
>+#define FRAMESAMPLES_30ms 480
> /* number of subframes */
>-#define SUBFRAMES                               6
>+#define SUBFRAMES 6
> /* length of a subframe */
>-#define UPDATE                                  80
>+#define UPDATE 80
> /* length of half a subframe (low/high band) */
>-#define HALF_SUBFRAMELEN                        (UPDATE/2)
>+#define HALF_SUBFRAMELEN (UPDATE / 2)
> /* samples of look ahead (in a half-band, so actually
>  * half the samples of look ahead @ FS) */
>-#define QLOOKAHEAD                              24    /* 3 ms */
>+#define QLOOKAHEAD 24 /* 3 ms */
> /* order of AR model in spectral entropy coder */
>-#define AR_ORDER                                6
>+#define AR_ORDER 6
> /* order of LP model in spectral entropy coder */
>-#define LP_ORDER                                0
>+#define LP_ORDER 0
> 
> /* window length (masking analysis) */
>-#define WINLEN                                  256
>+#define WINLEN 256
> /* order of low-band pole filter used to approximate masking curve */
>-#define ORDERLO                                 12
>+#define ORDERLO 12
> /* order of hi-band pole filter used to approximate masking curve */
>-#define ORDERHI                                 6
>-
>-#define UB_LPC_ORDER                            4
>-#define UB_LPC_VEC_PER_FRAME                    2
>-#define UB16_LPC_VEC_PER_FRAME                  4
>-#define UB_ACTIVE_SUBFRAMES                     2
>-#define UB_MAX_LPC_ORDER                        6
>-#define UB_INTERPOL_SEGMENTS                    1
>-#define UB16_INTERPOL_SEGMENTS                  3
>-#define LB_TOTAL_DELAY_SAMPLES                 48
>-enum ISACBandwidth {isac8kHz = 8, isac12kHz = 12, isac16kHz = 16};
>-enum ISACBand {kIsacLowerBand = 0, kIsacUpperBand12 = 1, kIsacUpperBand16 = 2};
>-enum IsacSamplingRate {kIsacWideband = 16,  kIsacSuperWideband = 32};
>-#define UB_LPC_GAIN_DIM                 SUBFRAMES
>-#define FB_STATE_SIZE_WORD32                    6
>-
>+#define ORDERHI 6
>+
>+#define UB_LPC_ORDER 4
>+#define UB_LPC_VEC_PER_FRAME 2
>+#define UB16_LPC_VEC_PER_FRAME 4
>+#define UB_ACTIVE_SUBFRAMES 2
>+#define UB_MAX_LPC_ORDER 6
>+#define UB_INTERPOL_SEGMENTS 1
>+#define UB16_INTERPOL_SEGMENTS 3
>+#define LB_TOTAL_DELAY_SAMPLES 48
>+enum ISACBandwidth { isac8kHz = 8, isac12kHz = 12, isac16kHz = 16 };
>+enum ISACBand {
>+  kIsacLowerBand = 0,
>+  kIsacUpperBand12 = 1,
>+  kIsacUpperBand16 = 2
>+};
>+enum IsacSamplingRate { kIsacWideband = 16, kIsacSuperWideband = 32 };
>+#define UB_LPC_GAIN_DIM SUBFRAMES
>+#define FB_STATE_SIZE_WORD32 6
> 
> /* order for post_filter_bank */
>-#define POSTQORDER                              3
>+#define POSTQORDER 3
> /* order for pre-filterbank */
>-#define QORDER                                  3
>+#define QORDER 3
> /* another order */
>-#define QORDER_ALL                              (POSTQORDER+QORDER-1)
>+#define QORDER_ALL (POSTQORDER + QORDER - 1)
> /* for decimator */
>-#define ALLPASSSECTIONS                         2
>-
>+#define ALLPASSSECTIONS 2
> 
> /* array size for byte stream in number of bytes. */
> /* The old maximum size still needed for the decoding */
>-#define STREAM_SIZE_MAX     600
>-#define STREAM_SIZE_MAX_30  200 /* 200 bytes=53.4 kbps @ 30 ms.framelength */
>-#define STREAM_SIZE_MAX_60  400 /* 400 bytes=53.4 kbps @ 60 ms.framelength */
>+#define STREAM_SIZE_MAX 600
>+#define STREAM_SIZE_MAX_30 200 /* 200 bytes=53.4 kbps @ 30 ms.framelength */
>+#define STREAM_SIZE_MAX_60 400 /* 400 bytes=53.4 kbps @ 60 ms.framelength */
> 
> /* storage size for bit counts */
>-#define BIT_COUNTER_SIZE                        30
>+#define BIT_COUNTER_SIZE 30
> /* maximum order of any AR model or filter */
>-#define MAX_AR_MODEL_ORDER                      12//50
>-
>+#define MAX_AR_MODEL_ORDER 12  // 50
> 
> /* For pitch analysis */
>-#define PITCH_FRAME_LEN                         (FRAMESAMPLES_HALF) /* 30 ms  */
>-#define PITCH_MAX_LAG                           140     /* 57 Hz  */
>-#define PITCH_MIN_LAG                           20              /* 400 Hz */
>-#define PITCH_MAX_GAIN                          0.45
>-#define PITCH_MAX_GAIN_06                       0.27  /* PITCH_MAX_GAIN*0.6 */
>-#define PITCH_MAX_GAIN_Q12      1843
>-#define PITCH_LAG_SPAN2                     (PITCH_MAX_LAG/2-PITCH_MIN_LAG/2+5)
>-#define PITCH_CORR_LEN2                         60     /* 15 ms  */
>-#define PITCH_CORR_STEP2                        (PITCH_FRAME_LEN/4)
>-#define PITCH_BW        11     /* half the band width of correlation surface */
>-#define PITCH_SUBFRAMES                         4
>-#define PITCH_GRAN_PER_SUBFRAME                 5
>-#define PITCH_SUBFRAME_LEN        (PITCH_FRAME_LEN/PITCH_SUBFRAMES)
>-#define PITCH_UPDATE              (PITCH_SUBFRAME_LEN/PITCH_GRAN_PER_SUBFRAME)
>+#define PITCH_FRAME_LEN (FRAMESAMPLES_HALF) /* 30 ms  */
>+#define PITCH_MAX_LAG 140                   /* 57 Hz  */
>+#define PITCH_MIN_LAG 20                    /* 400 Hz */
>+#define PITCH_MAX_GAIN 0.45
>+#define PITCH_MAX_GAIN_06 0.27 /* PITCH_MAX_GAIN*0.6 */
>+#define PITCH_MAX_GAIN_Q12 1843
>+#define PITCH_LAG_SPAN2 (PITCH_MAX_LAG / 2 - PITCH_MIN_LAG / 2 + 5)
>+#define PITCH_CORR_LEN2 60 /* 15 ms  */
>+#define PITCH_CORR_STEP2 (PITCH_FRAME_LEN / 4)
>+#define PITCH_BW 11 /* half the band width of correlation surface */
>+#define PITCH_SUBFRAMES 4
>+#define PITCH_GRAN_PER_SUBFRAME 5
>+#define PITCH_SUBFRAME_LEN (PITCH_FRAME_LEN / PITCH_SUBFRAMES)
>+#define PITCH_UPDATE (PITCH_SUBFRAME_LEN / PITCH_GRAN_PER_SUBFRAME)
> /* maximum number of peaks to be examined in correlation surface */
>-#define PITCH_MAX_NUM_PEAKS                  10
>-#define PITCH_PEAK_DECAY               0.85
>+#define PITCH_MAX_NUM_PEAKS 10
>+#define PITCH_PEAK_DECAY 0.85
> /* For weighting filter */
>-#define PITCH_WLPCORDER                   6
>-#define PITCH_WLPCWINLEN               PITCH_FRAME_LEN
>-#define PITCH_WLPCASYM                   0.3         /* asymmetry parameter */
>-#define PITCH_WLPCBUFLEN               PITCH_WLPCWINLEN
>+#define PITCH_WLPCORDER 6
>+#define PITCH_WLPCWINLEN PITCH_FRAME_LEN
>+#define PITCH_WLPCASYM 0.3 /* asymmetry parameter */
>+#define PITCH_WLPCBUFLEN PITCH_WLPCWINLEN
> /* For pitch filter */
> /* Extra 50 for fraction and LP filters */
>-#define PITCH_BUFFSIZE                   (PITCH_MAX_LAG + 50)
>-#define PITCH_INTBUFFSIZE               (PITCH_FRAME_LEN+PITCH_BUFFSIZE)
>+#define PITCH_BUFFSIZE (PITCH_MAX_LAG + 50)
>+#define PITCH_INTBUFFSIZE (PITCH_FRAME_LEN + PITCH_BUFFSIZE)
> /* Max rel. step for interpolation */
>-#define PITCH_UPSTEP                1.5
>+#define PITCH_UPSTEP 1.5
> /* Max rel. step for interpolation */
>-#define PITCH_DOWNSTEP                   0.67
>-#define PITCH_FRACS                             8
>-#define PITCH_FRACORDER                         9
>-#define PITCH_DAMPORDER                         5
>-#define PITCH_FILTDELAY                         1.5f
>+#define PITCH_DOWNSTEP 0.67
>+#define PITCH_FRACS 8
>+#define PITCH_FRACORDER 9
>+#define PITCH_DAMPORDER 5
>+#define PITCH_FILTDELAY 1.5f
> /* stepsize for quantization of the pitch Gain */
>-#define PITCH_GAIN_STEPSIZE                     0.125
>-
>-
>+#define PITCH_GAIN_STEPSIZE 0.125
> 
> /* Order of high pass filter */
>-#define HPORDER                                 2
>+#define HPORDER 2
> 
> /* some mathematical constants */
> /* log2(exp) */
>-#define LOG2EXP                                 1.44269504088896
>-#define PI                                      3.14159265358979
>+#define LOG2EXP 1.44269504088896
>+#define PI 3.14159265358979
> 
> /* Maximum number of iterations allowed to limit payload size */
>-#define MAX_PAYLOAD_LIMIT_ITERATION             5
>+#define MAX_PAYLOAD_LIMIT_ITERATION 5
> 
> /* Redundant Coding */
>-#define RCU_BOTTLENECK_BPS                      16000
>-#define RCU_TRANSCODING_SCALE                   0.40f
>-#define RCU_TRANSCODING_SCALE_INVERSE           2.5f
>+#define RCU_BOTTLENECK_BPS 16000
>+#define RCU_TRANSCODING_SCALE 0.40f
>+#define RCU_TRANSCODING_SCALE_INVERSE 2.5f
> 
>-#define RCU_TRANSCODING_SCALE_UB                0.50f
>-#define RCU_TRANSCODING_SCALE_UB_INVERSE        2.0f
>+#define RCU_TRANSCODING_SCALE_UB 0.50f
>+#define RCU_TRANSCODING_SCALE_UB_INVERSE 2.0f
> 
> /* Define Error codes */
> /* 6000 General */
>-#define ISAC_MEMORY_ALLOCATION_FAILED    6010
>-#define ISAC_MODE_MISMATCH       6020
>-#define ISAC_DISALLOWED_BOTTLENECK     6030
>-#define ISAC_DISALLOWED_FRAME_LENGTH    6040
>-#define ISAC_UNSUPPORTED_SAMPLING_FREQUENCY         6050
>+#define ISAC_MEMORY_ALLOCATION_FAILED 6010
>+#define ISAC_MODE_MISMATCH 6020
>+#define ISAC_DISALLOWED_BOTTLENECK 6030
>+#define ISAC_DISALLOWED_FRAME_LENGTH 6040
>+#define ISAC_UNSUPPORTED_SAMPLING_FREQUENCY 6050
> 
> /* 6200 Bandwidth estimator */
>-#define ISAC_RANGE_ERROR_BW_ESTIMATOR    6240
>+#define ISAC_RANGE_ERROR_BW_ESTIMATOR 6240
> /* 6400 Encoder */
>-#define ISAC_ENCODER_NOT_INITIATED     6410
>-#define ISAC_DISALLOWED_CODING_MODE     6420
>-#define ISAC_DISALLOWED_FRAME_MODE_ENCODER   6430
>-#define ISAC_DISALLOWED_BITSTREAM_LENGTH            6440
>-#define ISAC_PAYLOAD_LARGER_THAN_LIMIT              6450
>-#define ISAC_DISALLOWED_ENCODER_BANDWIDTH           6460
>+#define ISAC_ENCODER_NOT_INITIATED 6410
>+#define ISAC_DISALLOWED_CODING_MODE 6420
>+#define ISAC_DISALLOWED_FRAME_MODE_ENCODER 6430
>+#define ISAC_DISALLOWED_BITSTREAM_LENGTH 6440
>+#define ISAC_PAYLOAD_LARGER_THAN_LIMIT 6450
>+#define ISAC_DISALLOWED_ENCODER_BANDWIDTH 6460
> /* 6600 Decoder */
>-#define ISAC_DECODER_NOT_INITIATED     6610
>-#define ISAC_EMPTY_PACKET       6620
>-#define ISAC_DISALLOWED_FRAME_MODE_DECODER   6630
>-#define ISAC_RANGE_ERROR_DECODE_FRAME_LENGTH  6640
>-#define ISAC_RANGE_ERROR_DECODE_BANDWIDTH   6650
>-#define ISAC_RANGE_ERROR_DECODE_PITCH_GAIN   6660
>-#define ISAC_RANGE_ERROR_DECODE_PITCH_LAG   6670
>-#define ISAC_RANGE_ERROR_DECODE_LPC     6680
>-#define ISAC_RANGE_ERROR_DECODE_SPECTRUM   6690
>-#define ISAC_LENGTH_MISMATCH      6730
>-#define ISAC_RANGE_ERROR_DECODE_BANDWITH            6740
>-#define ISAC_DISALLOWED_BANDWIDTH_MODE_DECODER      6750
>-#define ISAC_DISALLOWED_LPC_MODEL                   6760
>+#define ISAC_DECODER_NOT_INITIATED 6610
>+#define ISAC_EMPTY_PACKET 6620
>+#define ISAC_DISALLOWED_FRAME_MODE_DECODER 6630
>+#define ISAC_RANGE_ERROR_DECODE_FRAME_LENGTH 6640
>+#define ISAC_RANGE_ERROR_DECODE_BANDWIDTH 6650
>+#define ISAC_RANGE_ERROR_DECODE_PITCH_GAIN 6660
>+#define ISAC_RANGE_ERROR_DECODE_PITCH_LAG 6670
>+#define ISAC_RANGE_ERROR_DECODE_LPC 6680
>+#define ISAC_RANGE_ERROR_DECODE_SPECTRUM 6690
>+#define ISAC_LENGTH_MISMATCH 6730
>+#define ISAC_RANGE_ERROR_DECODE_BANDWITH 6740
>+#define ISAC_DISALLOWED_BANDWIDTH_MODE_DECODER 6750
>+#define ISAC_DISALLOWED_LPC_MODEL 6760
> /* 6800 Call setup formats */
>-#define ISAC_INCOMPATIBLE_FORMATS     6810
>+#define ISAC_INCOMPATIBLE_FORMATS 6810
> 
> #endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_SETTINGS_H_ */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/spectrum_ar_model_tables.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/spectrum_ar_model_tables.c
>index 0f6d889225d..839d5d45868 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/spectrum_ar_model_tables.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/spectrum_ar_model_tables.c
>@@ -8,8 +8,8 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "spectrum_ar_model_tables.h"
>-#include "settings.h"
>+#include "modules/audio_coding/codecs/isac/main/source/spectrum_ar_model_tables.h"
>+#include "modules/audio_coding/codecs/isac/main/source/settings.h"
> 
> /********************* AR Coefficient Tables ************************/
> /* cdf for quantized reflection coefficient 1 */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/spectrum_ar_model_tables.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/spectrum_ar_model_tables.h
>index 82ab363adbd..d272be0dc38 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/spectrum_ar_model_tables.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/spectrum_ar_model_tables.h
>@@ -11,7 +11,7 @@
> /*
>  * spectrum_ar_model_tables.h
>  *
>- * This file contains definitions of tables with AR coefficients, 
>+ * This file contains definitions of tables with AR coefficients,
>  * Gain coefficients and cosine tables.
>  *
>  */
>@@ -19,7 +19,7 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_SPECTRUM_AR_MODEL_TABLES_H_
> #define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_SPECTRUM_AR_MODEL_TABLES_H_
> 
>-#include "structs.h"
>+#include "modules/audio_coding/codecs/isac/main/source/structs.h"
> 
> #define NUM_AR_RC_QUANT_BAUNDARY 12
> 
>@@ -45,15 +45,15 @@ extern const uint16_t WebRtcIsac_kQArRc6Cdf[NUM_AR_RC_QUANT_BAUNDARY];
> /* quantization boundary levels for reflection coefficients */
> extern const int16_t WebRtcIsac_kQArBoundaryLevels[NUM_AR_RC_QUANT_BAUNDARY];
> 
>-/* initial indices for AR reflection coefficient quantizer and cdf table search */
>+/* initial indices for AR reflection coefficient quantizer and cdf table search
>+ */
> extern const uint16_t WebRtcIsac_kQArRcInitIndex[AR_ORDER];
> 
> /* pointers to AR cdf tables */
>-extern const uint16_t *WebRtcIsac_kQArRcCdfPtr[AR_ORDER];
>+extern const uint16_t* WebRtcIsac_kQArRcCdfPtr[AR_ORDER];
> 
> /* pointers to AR representation levels tables */
>-extern const int16_t *WebRtcIsac_kQArRcLevelsPtr[AR_ORDER];
>-
>+extern const int16_t* WebRtcIsac_kQArRcLevelsPtr[AR_ORDER];
> 
> /******************** GAIN Coefficient Tables ***********************/
> /* cdf for Gain coefficient */
>@@ -66,7 +66,7 @@ extern const int32_t WebRtcIsac_kQGain2Levels[18];
> extern const int32_t WebRtcIsac_kQGain2BoundaryLevels[19];
> 
> /* pointer to Gain cdf table */
>-extern const uint16_t *WebRtcIsac_kQGainCdf_ptr[1];
>+extern const uint16_t* WebRtcIsac_kQGainCdf_ptr[1];
> 
> /* Gain initial index for gain quantizer and cdf table search */
> extern const uint16_t WebRtcIsac_kQGainInitIndex[1];
>@@ -75,4 +75,5 @@ extern const uint16_t WebRtcIsac_kQGainInitIndex[1];
> /* Cosine table */
> extern const int16_t WebRtcIsac_kCos[6][60];
> 
>-#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_SPECTRUM_AR_MODEL_TABLES_H_ */
>+#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_SPECTRUM_AR_MODEL_TABLES_H_ \
>+        */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/structs.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/structs.h
>index ef4282b5214..1255d5b16c7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/structs.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/structs.h
>@@ -19,183 +19,170 @@
> #define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_STRUCTS_H_
> 
> #include "modules/audio_coding/codecs/isac/bandwidth_info.h"
>-#include "modules/audio_coding/codecs/isac/main/include/isac.h"
> #include "modules/audio_coding/codecs/isac/main/source/settings.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include "modules/third_party/fft/fft.h"
> 
> typedef struct Bitstreamstruct {
>-
>-  uint8_t   stream[STREAM_SIZE_MAX];
>-  uint32_t  W_upper;
>-  uint32_t  streamval;
>-  uint32_t  stream_index;
>+  uint8_t stream[STREAM_SIZE_MAX];
>+  uint32_t W_upper;
>+  uint32_t streamval;
>+  uint32_t stream_index;
> 
> } Bitstr;
> 
> typedef struct {
>+  double DataBufferLo[WINLEN];
>+  double DataBufferHi[WINLEN];
> 
>-  double    DataBufferLo[WINLEN];
>-  double    DataBufferHi[WINLEN];
>-
>-  double    CorrBufLo[ORDERLO+1];
>-  double    CorrBufHi[ORDERHI+1];
>+  double CorrBufLo[ORDERLO + 1];
>+  double CorrBufHi[ORDERHI + 1];
> 
>-  float    PreStateLoF[ORDERLO+1];
>-  float    PreStateLoG[ORDERLO+1];
>-  float    PreStateHiF[ORDERHI+1];
>-  float    PreStateHiG[ORDERHI+1];
>-  float    PostStateLoF[ORDERLO+1];
>-  float    PostStateLoG[ORDERLO+1];
>-  float    PostStateHiF[ORDERHI+1];
>-  float    PostStateHiG[ORDERHI+1];
>+  float PreStateLoF[ORDERLO + 1];
>+  float PreStateLoG[ORDERLO + 1];
>+  float PreStateHiF[ORDERHI + 1];
>+  float PreStateHiG[ORDERHI + 1];
>+  float PostStateLoF[ORDERLO + 1];
>+  float PostStateLoG[ORDERLO + 1];
>+  float PostStateHiF[ORDERHI + 1];
>+  float PostStateHiG[ORDERHI + 1];
> 
>-  double    OldEnergy;
>+  double OldEnergy;
> 
> } MaskFiltstr;
> 
>-
> typedef struct {
>-
>-  //state vectors for each of the two analysis filters
>-  double    INSTAT1[2*(QORDER-1)];
>-  double    INSTAT2[2*(QORDER-1)];
>-  double    INSTATLA1[2*(QORDER-1)];
>-  double    INSTATLA2[2*(QORDER-1)];
>-  double    INLABUF1[QLOOKAHEAD];
>-  double    INLABUF2[QLOOKAHEAD];
>-
>-  float    INSTAT1_float[2*(QORDER-1)];
>-  float    INSTAT2_float[2*(QORDER-1)];
>-  float    INSTATLA1_float[2*(QORDER-1)];
>-  float    INSTATLA2_float[2*(QORDER-1)];
>-  float    INLABUF1_float[QLOOKAHEAD];
>-  float    INLABUF2_float[QLOOKAHEAD];
>+  // state vectors for each of the two analysis filters
>+  double INSTAT1[2 * (QORDER - 1)];
>+  double INSTAT2[2 * (QORDER - 1)];
>+  double INSTATLA1[2 * (QORDER - 1)];
>+  double INSTATLA2[2 * (QORDER - 1)];
>+  double INLABUF1[QLOOKAHEAD];
>+  double INLABUF2[QLOOKAHEAD];
>+
>+  float INSTAT1_float[2 * (QORDER - 1)];
>+  float INSTAT2_float[2 * (QORDER - 1)];
>+  float INSTATLA1_float[2 * (QORDER - 1)];
>+  float INSTATLA2_float[2 * (QORDER - 1)];
>+  float INLABUF1_float[QLOOKAHEAD];
>+  float INLABUF2_float[QLOOKAHEAD];
> 
>   /* High pass filter */
>-  double    HPstates[HPORDER];
>-  float    HPstates_float[HPORDER];
>+  double HPstates[HPORDER];
>+  float HPstates_float[HPORDER];
> 
> } PreFiltBankstr;
> 
>-
> typedef struct {
>-
>-  //state vectors for each of the two analysis filters
>-  double    STATE_0_LOWER[2*POSTQORDER];
>-  double    STATE_0_UPPER[2*POSTQORDER];
>+  // state vectors for each of the two analysis filters
>+  double STATE_0_LOWER[2 * POSTQORDER];
>+  double STATE_0_UPPER[2 * POSTQORDER];
> 
>   /* High pass filter */
>-  double    HPstates1[HPORDER];
>-  double    HPstates2[HPORDER];
>+  double HPstates1[HPORDER];
>+  double HPstates2[HPORDER];
> 
>-  float    STATE_0_LOWER_float[2*POSTQORDER];
>-  float    STATE_0_UPPER_float[2*POSTQORDER];
>+  float STATE_0_LOWER_float[2 * POSTQORDER];
>+  float STATE_0_UPPER_float[2 * POSTQORDER];
> 
>-  float    HPstates1_float[HPORDER];
>-  float    HPstates2_float[HPORDER];
>+  float HPstates1_float[HPORDER];
>+  float HPstates2_float[HPORDER];
> 
> } PostFiltBankstr;
> 
> typedef struct {
>+  // data buffer for pitch filter
>+  double ubuf[PITCH_BUFFSIZE];
> 
>-  //data buffer for pitch filter
>-  double    ubuf[PITCH_BUFFSIZE];
>-
>-  //low pass state vector
>-  double    ystate[PITCH_DAMPORDER];
>+  // low pass state vector
>+  double ystate[PITCH_DAMPORDER];
> 
>-  //old lag and gain
>-  double    oldlagp[1];
>-  double    oldgainp[1];
>+  // old lag and gain
>+  double oldlagp[1];
>+  double oldgainp[1];
> 
> } PitchFiltstr;
> 
> typedef struct {
>+  // data buffer
>+  double buffer[PITCH_WLPCBUFLEN];
> 
>-  //data buffer
>-  double    buffer[PITCH_WLPCBUFLEN];
>+  // state vectors
>+  double istate[PITCH_WLPCORDER];
>+  double weostate[PITCH_WLPCORDER];
>+  double whostate[PITCH_WLPCORDER];
> 
>-  //state vectors
>-  double    istate[PITCH_WLPCORDER];
>-  double    weostate[PITCH_WLPCORDER];
>-  double    whostate[PITCH_WLPCORDER];
>-
>-  //LPC window   -> should be a global array because constant
>-  double    window[PITCH_WLPCWINLEN];
>+  // LPC window   -> should be a global array because constant
>+  double window[PITCH_WLPCWINLEN];
> 
> } WeightFiltstr;
> 
> typedef struct {
>+  // for inital estimator
>+  double dec_buffer[PITCH_CORR_LEN2 + PITCH_CORR_STEP2 + PITCH_MAX_LAG / 2 -
>+                    PITCH_FRAME_LEN / 2 + 2];
>+  double decimator_state[2 * ALLPASSSECTIONS + 1];
>+  double hp_state[2];
> 
>-  //for inital estimator
>-  double         dec_buffer[PITCH_CORR_LEN2 + PITCH_CORR_STEP2 +
>-                            PITCH_MAX_LAG/2 - PITCH_FRAME_LEN/2+2];
>-  double        decimator_state[2*ALLPASSSECTIONS+1];
>-  double        hp_state[2];
>-
>-  double        whitened_buf[QLOOKAHEAD];
>+  double whitened_buf[QLOOKAHEAD];
> 
>-  double        inbuf[QLOOKAHEAD];
>+  double inbuf[QLOOKAHEAD];
> 
>-  PitchFiltstr  PFstr_wght;
>-  PitchFiltstr  PFstr;
>+  PitchFiltstr PFstr_wght;
>+  PitchFiltstr PFstr;
>   WeightFiltstr Wghtstr;
> 
> } PitchAnalysisStruct;
> 
>-
>-
> /* Have instance of struct together with other iSAC structs */
> typedef struct {
>-
>   /* Previous frame length (in ms)                                    */
>-  int32_t    prev_frame_length;
>+  int32_t prev_frame_length;
> 
>   /* Previous RTP timestamp from received
>      packet (in samples relative beginning)                           */
>-  int32_t    prev_rec_rtp_number;
>+  int32_t prev_rec_rtp_number;
> 
>   /* Send timestamp for previous packet (in ms using timeGetTime())   */
>-  uint32_t    prev_rec_send_ts;
>+  uint32_t prev_rec_send_ts;
> 
>   /* Arrival time for previous packet (in ms using timeGetTime())     */
>-  uint32_t    prev_rec_arr_ts;
>+  uint32_t prev_rec_arr_ts;
> 
>   /* rate of previous packet, derived from RTP timestamps (in bits/s) */
>-  float   prev_rec_rtp_rate;
>+  float prev_rec_rtp_rate;
> 
>   /* Time sinse the last update of the BN estimate (in ms)            */
>-  uint32_t    last_update_ts;
>+  uint32_t last_update_ts;
> 
>   /* Time sinse the last reduction (in ms)                            */
>-  uint32_t    last_reduction_ts;
>+  uint32_t last_reduction_ts;
> 
>   /* How many times the estimate was update in the beginning          */
>-  int32_t    count_tot_updates_rec;
>+  int32_t count_tot_updates_rec;
> 
>   /* The estimated bottle neck rate from there to here (in bits/s)    */
>-  int32_t  rec_bw;
>-  float   rec_bw_inv;
>-  float   rec_bw_avg;
>-  float   rec_bw_avg_Q;
>+  int32_t rec_bw;
>+  float rec_bw_inv;
>+  float rec_bw_avg;
>+  float rec_bw_avg_Q;
> 
>   /* The estimated mean absolute jitter value,
>      as seen on this side (in ms)                                     */
>-  float   rec_jitter;
>-  float   rec_jitter_short_term;
>-  float   rec_jitter_short_term_abs;
>-  float   rec_max_delay;
>-  float   rec_max_delay_avg_Q;
>+  float rec_jitter;
>+  float rec_jitter_short_term;
>+  float rec_jitter_short_term_abs;
>+  float rec_max_delay;
>+  float rec_max_delay_avg_Q;
> 
>   /* (assumed) bitrate for headers (bps)                              */
>-  float   rec_header_rate;
>+  float rec_header_rate;
> 
>   /* The estimated bottle neck rate from here to there (in bits/s)    */
>-  float    send_bw_avg;
>+  float send_bw_avg;
> 
>   /* The estimated mean absolute jitter value, as seen on
>      the other siee (in ms)                                           */
>-  float   send_max_delay_avg;
>+  float send_max_delay_avg;
> 
>   // number of packets received since last update
>   int num_pkts_rec;
>@@ -218,72 +205,54 @@ typedef struct {
> 
>   int change_to_WB;
> 
>-  uint32_t                 senderTimestamp;
>-  uint32_t                 receiverTimestamp;
>-  //enum IsacSamplingRate incomingStreamSampFreq;
>-  uint16_t                 numConsecLatePkts;
>-  float                        consecLatency;
>-  int16_t                  inWaitLatePkts;
>+  uint32_t senderTimestamp;
>+  uint32_t receiverTimestamp;
>+  // enum IsacSamplingRate incomingStreamSampFreq;
>+  uint16_t numConsecLatePkts;
>+  float consecLatency;
>+  int16_t inWaitLatePkts;
> 
>   IsacBandwidthInfo external_bw_info;
> } BwEstimatorstr;
> 
>-
> typedef struct {
>-
>   /* boolean, flags if previous packet exceeded B.N. */
>-  int    PrevExceed;
>+  int PrevExceed;
>   /* ms */
>-  int    ExceedAgo;
>+  int ExceedAgo;
>   /* packets left to send in current burst */
>-  int    BurstCounter;
>+  int BurstCounter;
>   /* packets */
>-  int    InitCounter;
>+  int InitCounter;
>   /* ms remaining in buffer when next packet will be sent */
>   double StillBuffered;
> 
> } RateModel;
> 
>-
>-typedef struct {
>-
>-  unsigned int SpaceAlloced;
>-  unsigned int MaxPermAlloced;
>-  double Tmp0[MAXFFTSIZE];
>-  double Tmp1[MAXFFTSIZE];
>-  double Tmp2[MAXFFTSIZE];
>-  double Tmp3[MAXFFTSIZE];
>-  int Perm[MAXFFTSIZE];
>-  int factor [NFACTOR];
>-
>-} FFTstr;
>-
>-
> /* The following strutc is used to store data from encoding, to make it
>    fast and easy to construct a new bitstream with a different Bandwidth
>    estimate. All values (except framelength and minBytes) is double size to
>    handle 60 ms of data.
> */
> typedef struct {
>-
>   /* Used to keep track of if it is first or second part of 60 msec packet */
>-  int         startIdx;
>+  int startIdx;
> 
>   /* Frame length in samples */
>   int16_t framelength;
> 
>   /* Pitch Gain */
>-  int         pitchGain_index[2];
>+  int pitchGain_index[2];
> 
>   /* Pitch Lag */
>-  double      meanGain[2];
>-  int         pitchIndex[PITCH_SUBFRAMES*2];
>+  double meanGain[2];
>+  int pitchIndex[PITCH_SUBFRAMES * 2];
> 
>   /* LPC */
>-  int         LPCindex_s[108*2]; /* KLT_ORDER_SHAPE = 108 */
>-  int         LPCindex_g[12*2];  /* KLT_ORDER_GAIN = 12 */
>-  double      LPCcoeffs_lo[(ORDERLO+1)*SUBFRAMES*2];
>-  double      LPCcoeffs_hi[(ORDERHI+1)*SUBFRAMES*2];
>+  int LPCindex_s[108 * 2]; /* KLT_ORDER_SHAPE = 108 */
>+  int LPCindex_g[12 * 2];  /* KLT_ORDER_GAIN = 12 */
>+  double LPCcoeffs_lo[(ORDERLO + 1) * SUBFRAMES * 2];
>+  double LPCcoeffs_hi[(ORDERHI + 1) * SUBFRAMES * 2];
> 
>   /* Encode Spec */
>   int16_t fre[FRAMESAMPLES];
>@@ -291,59 +260,54 @@ typedef struct {
>   int16_t AvgPitchGain[2];
> 
>   /* Used in adaptive mode only */
>-  int         minBytes;
>+  int minBytes;
> 
> } IsacSaveEncoderData;
> 
>-
> typedef struct {
>+  int indexLPCShape[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME];
>+  double lpcGain[SUBFRAMES << 1];
>+  int lpcGainIndex[SUBFRAMES << 1];
> 
>-  int         indexLPCShape[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME];
>-  double      lpcGain[SUBFRAMES<<1];
>-  int         lpcGainIndex[SUBFRAMES<<1];
>-
>-  Bitstr      bitStreamObj;
>+  Bitstr bitStreamObj;
> 
>   int16_t realFFT[FRAMESAMPLES_HALF];
>   int16_t imagFFT[FRAMESAMPLES_HALF];
> } ISACUBSaveEncDataStruct;
> 
>-
>-
> typedef struct {
>-
>-  Bitstr              bitstr_obj;
>-  MaskFiltstr         maskfiltstr_obj;
>-  PreFiltBankstr      prefiltbankstr_obj;
>-  PitchFiltstr        pitchfiltstr_obj;
>+  Bitstr bitstr_obj;
>+  MaskFiltstr maskfiltstr_obj;
>+  PreFiltBankstr prefiltbankstr_obj;
>+  PitchFiltstr pitchfiltstr_obj;
>   PitchAnalysisStruct pitchanalysisstr_obj;
>-  FFTstr              fftstr_obj;
>+  FFTstr fftstr_obj;
>   IsacSaveEncoderData SaveEnc_obj;
> 
>-  int                 buffer_index;
>-  int16_t         current_framesamples;
>+  int buffer_index;
>+  int16_t current_framesamples;
> 
>-  float               data_buffer_float[FRAMESAMPLES_30ms];
>+  float data_buffer_float[FRAMESAMPLES_30ms];
> 
>-  int                 frame_nb;
>-  double              bottleneck;
>-  int16_t         new_framelength;
>-  double              s2nr;
>+  int frame_nb;
>+  double bottleneck;
>+  int16_t new_framelength;
>+  double s2nr;
> 
>   /* Maximum allowed number of bits for a 30 msec packet */
>-  int16_t         payloadLimitBytes30;
>+  int16_t payloadLimitBytes30;
>   /* Maximum allowed number of bits for a 30 msec packet */
>-  int16_t         payloadLimitBytes60;
>+  int16_t payloadLimitBytes60;
>   /* Maximum allowed number of bits for both 30 and 60 msec packet */
>-  int16_t         maxPayloadBytes;
>+  int16_t maxPayloadBytes;
>   /* Maximum allowed rate in bytes per 30 msec packet */
>-  int16_t         maxRateInBytes;
>+  int16_t maxRateInBytes;
> 
>   /*---
>     If set to 1 iSAC will not addapt the frame-size, if used in
>     channel-adaptive mode. The initial value will be used for all rates.
>     ---*/
>-  int16_t         enforceFrameSize;
>+  int16_t enforceFrameSize;
> 
>   /*-----
>     This records the BWE index the encoder injected into the bit-stream.
>@@ -352,64 +316,53 @@ typedef struct {
>     a recursive procedure (WebRtcIsac_GetDownlinkBwJitIndexImpl) and has to be
>     called only once per each encode.
>     -----*/
>-  int16_t         lastBWIdx;
>+  int16_t lastBWIdx;
> } ISACLBEncStruct;
> 
> typedef struct {
>-
>-  Bitstr                  bitstr_obj;
>-  MaskFiltstr             maskfiltstr_obj;
>-  PreFiltBankstr          prefiltbankstr_obj;
>-  FFTstr                  fftstr_obj;
>+  Bitstr bitstr_obj;
>+  MaskFiltstr maskfiltstr_obj;
>+  PreFiltBankstr prefiltbankstr_obj;
>+  FFTstr fftstr_obj;
>   ISACUBSaveEncDataStruct SaveEnc_obj;
> 
>-  int                     buffer_index;
>-  float                   data_buffer_float[MAX_FRAMESAMPLES +
>-                                            LB_TOTAL_DELAY_SAMPLES];
>-  double                  bottleneck;
>+  int buffer_index;
>+  float data_buffer_float[MAX_FRAMESAMPLES + LB_TOTAL_DELAY_SAMPLES];
>+  double bottleneck;
>   /* Maximum allowed number of bits for a 30 msec packet */
>-  //int16_t        payloadLimitBytes30;
>+  // int16_t        payloadLimitBytes30;
>   /* Maximum allowed number of bits for both 30 and 60 msec packet */
>-  //int16_t        maxPayloadBytes;
>-  int16_t             maxPayloadSizeBytes;
>+  // int16_t        maxPayloadBytes;
>+  int16_t maxPayloadSizeBytes;
> 
>-  double                  lastLPCVec[UB_LPC_ORDER];
>-  int16_t             numBytesUsed;
>-  int16_t             lastJitterInfo;
>+  double lastLPCVec[UB_LPC_ORDER];
>+  int16_t numBytesUsed;
>+  int16_t lastJitterInfo;
> } ISACUBEncStruct;
> 
>-
>-
> typedef struct {
>-
>-  Bitstr          bitstr_obj;
>-  MaskFiltstr     maskfiltstr_obj;
>+  Bitstr bitstr_obj;
>+  MaskFiltstr maskfiltstr_obj;
>   PostFiltBankstr postfiltbankstr_obj;
>-  PitchFiltstr    pitchfiltstr_obj;
>-  FFTstr          fftstr_obj;
>+  PitchFiltstr pitchfiltstr_obj;
>+  FFTstr fftstr_obj;
> 
> } ISACLBDecStruct;
> 
> typedef struct {
>-
>-  Bitstr          bitstr_obj;
>-  MaskFiltstr     maskfiltstr_obj;
>+  Bitstr bitstr_obj;
>+  MaskFiltstr maskfiltstr_obj;
>   PostFiltBankstr postfiltbankstr_obj;
>-  FFTstr          fftstr_obj;
>+  FFTstr fftstr_obj;
> 
> } ISACUBDecStruct;
> 
>-
>-
> typedef struct {
>-
>   ISACLBEncStruct ISACencLB_obj;
>   ISACLBDecStruct ISACdecLB_obj;
> } ISACLBStruct;
> 
>-
> typedef struct {
>-
>   ISACUBEncStruct ISACencUB_obj;
>   ISACUBDecStruct ISACdecUB_obj;
> } ISACUBStruct;
>@@ -421,14 +374,14 @@ typedef struct {
> */
> typedef struct {
>   /* 6 lower-band & 6 upper-band */
>-  double       loFiltGain[SUBFRAMES];
>-  double       hiFiltGain[SUBFRAMES];
>+  double loFiltGain[SUBFRAMES];
>+  double hiFiltGain[SUBFRAMES];
>   /* Upper boundary of interval W */
>   uint32_t W_upper;
>   uint32_t streamval;
>   /* Index to the current position in bytestream */
>   uint32_t stream_index;
>-  uint8_t  stream[3];
>+  uint8_t stream[3];
> } transcode_obj;
> 
> typedef struct {
>@@ -444,46 +397,46 @@ typedef struct {
> 
> typedef struct {
>   // lower-band codec instance
>-  ISACLBStruct              instLB;
>+  ISACLBStruct instLB;
>   // upper-band codec instance
>-  ISACUBStruct              instUB;
>+  ISACUBStruct instUB;
> 
>   // Bandwidth Estimator and model for the rate.
>-  BwEstimatorstr            bwestimator_obj;
>-  RateModel                 rate_data_obj;
>-  double                    MaxDelay;
>+  BwEstimatorstr bwestimator_obj;
>+  RateModel rate_data_obj;
>+  double MaxDelay;
> 
>   /* 0 = adaptive; 1 = instantaneous */
>-  int16_t               codingMode;
>+  int16_t codingMode;
> 
>   // overall bottleneck of the codec
>-  int32_t               bottleneck;
>+  int32_t bottleneck;
> 
>   // QMF Filter state
>-  int32_t               analysisFBState1[FB_STATE_SIZE_WORD32];
>-  int32_t               analysisFBState2[FB_STATE_SIZE_WORD32];
>-  int32_t               synthesisFBState1[FB_STATE_SIZE_WORD32];
>-  int32_t               synthesisFBState2[FB_STATE_SIZE_WORD32];
>+  int32_t analysisFBState1[FB_STATE_SIZE_WORD32];
>+  int32_t analysisFBState2[FB_STATE_SIZE_WORD32];
>+  int32_t synthesisFBState1[FB_STATE_SIZE_WORD32];
>+  int32_t synthesisFBState2[FB_STATE_SIZE_WORD32];
> 
>   // Error Code
>-  int16_t               errorCode;
>+  int16_t errorCode;
> 
>   // bandwidth of the encoded audio 8, 12 or 16 kHz
>-  enum ISACBandwidth        bandwidthKHz;
>+  enum ISACBandwidth bandwidthKHz;
>   // Sampling rate of audio, encoder and decode,  8 or 16 kHz
>   enum IsacSamplingRate encoderSamplingRateKHz;
>   enum IsacSamplingRate decoderSamplingRateKHz;
>   // Flag to keep track of initializations, lower & upper-band
>   // encoder and decoder.
>-  int16_t               initFlag;
>+  int16_t initFlag;
> 
>   // Flag to to indicate signal bandwidth switch
>-  int16_t               resetFlag_8kHz;
>+  int16_t resetFlag_8kHz;
> 
>   // Maximum allowed rate, measured in Bytes per 30 ms.
>-  int16_t               maxRateBytesPer30Ms;
>+  int16_t maxRateBytesPer30Ms;
>   // Maximum allowed payload-size, measured in Bytes.
>-  int16_t               maxPayloadSizeBytes;
>+  int16_t maxPayloadSizeBytes;
>   /* The expected sampling rate of the input signal. Valid values are 16000
>    * and 32000. This is not the operation sampling rate of the codec. */
>   uint16_t in_sample_rate_hz;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/transform.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/transform.c
>index 8992897f454..082ad941c4d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/transform.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/transform.c
>@@ -8,12 +8,13 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "settings.h"
>-#include "fft.h"
>-#include "codec.h"
>-#include "os_specific_inline.h"
> #include <math.h>
> 
>+#include "modules/audio_coding/codecs/isac/main/source/settings.h"
>+#include "modules/audio_coding/codecs/isac/main/source/codec.h"
>+#include "modules/audio_coding/codecs/isac/main/source/os_specific_inline.h"
>+#include "modules/third_party/fft/fft.h"
>+
> void WebRtcIsac_InitTransform(TransformTables* tables) {
>   int k;
>   double fact, phase;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
>index 33d9efcda1b..35a8832462b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
>@@ -11,23 +11,26 @@
> // ReleaseTest-API.cpp : Defines the entry point for the console application.
> //
> 
>+#include <ctype.h>
> #include <stdio.h>
> #include <stdlib.h>
> #include <string.h>
> #include <time.h>
>-#include <ctype.h>
> #include <iostream>
> 
> /* include API */
>-#include "isac.h"
>-#include "utility.h"
>+#include "modules/audio_coding/codecs/isac/main/include/isac.h"
>+#include "modules/audio_coding/codecs/isac/main/util/utility.h"
> #include "rtc_base/format_macros.h"
> 
> /* Defines */
>-#define SEED_FILE "randseed.txt" /* Used when running decoder on garbage data */
>-#define MAX_FRAMESAMPLES 960     /* max number of samples per frame
>-                                    (= 60 ms frame & 16 kHz) or
>-                                    (= 30 ms frame & 32 kHz) */
>+#define SEED_FILE                                             \
>+  "randseed.txt" /* Used when running decoder on garbage data \
>+                  */
>+#define MAX_FRAMESAMPLES                                         \
>+  960                         /* max number of samples per frame \
>+                                 (= 60 ms frame & 16 kHz) or     \
>+                                 (= 30 ms frame & 32 kHz) */
> #define FRAMESAMPLES_10ms 160 /* number of samples per 10ms frame */
> #define SWBFRAMESAMPLES_10ms 320
> //#define FS 16000 /* sampling frequency (Hz) */
>@@ -42,7 +45,7 @@ using namespace std;
> 
> int main(int argc, char* argv[]) {
>   char inname[100], outname[100], bottleneck_file[100], vadfile[100];
>-  FILE* inp, *outp, * f_bn = NULL, * vadp = NULL, *bandwidthp;
>+  FILE *inp, *outp, *f_bn = NULL, *vadp = NULL, *bandwidthp;
>   int framecnt, endfile;
> 
>   size_t i;
>@@ -230,8 +233,10 @@ int main(int argc, char* argv[]) {
>       rateBPS = atoi(argv[i + 1]);
>       setControlBWE = 1;
>       if ((rateBPS < 10000) || (rateBPS > 32000)) {
>-        printf("\n%d is not a initial rate. Valid values are in the range "
>-               "10000 to 32000.\n", rateBPS);
>+        printf(
>+            "\n%d is not a initial rate. Valid values are in the range "
>+            "10000 to 32000.\n",
>+            rateBPS);
>         exit(0);
>       }
>       printf("New initial rate: %d\n", rateBPS);
>@@ -242,8 +247,10 @@ int main(int argc, char* argv[]) {
>     if (!strcmp("-FL", argv[i])) {
>       framesize = atoi(argv[i + 1]);
>       if ((framesize != 30) && (framesize != 60)) {
>-        printf("\n%d is not a valid frame length. Valid length are 30 and 60 "
>-               "msec.\n", framesize);
>+        printf(
>+            "\n%d is not a valid frame length. Valid length are 30 and 60 "
>+            "msec.\n",
>+            framesize);
>         exit(0);
>       }
>       setControlBWE = 1;
>@@ -277,8 +284,10 @@ int main(int argc, char* argv[]) {
>       testNum = atoi(argv[i + 1]);
>       printf("Fault test: %d\n", testNum);
>       if (testNum < 1 || testNum > 10) {
>-        printf("\n%d is not a valid Fault Scenario number. Valid Fault "
>-               "Scenarios are numbered 1-10.\n", testNum);
>+        printf(
>+            "\n%d is not a valid Fault Scenario number. Valid Fault "
>+            "Scenarios are numbered 1-10.\n",
>+            testNum);
>         exit(0);
>       }
>       i++;
>@@ -336,8 +345,10 @@ int main(int argc, char* argv[]) {
>         sscanf(argv[i], "%s", bottleneck_file);
>         f_bn = fopen(bottleneck_file, "rb");
>         if (f_bn == NULL) {
>-          printf("Error No value provided for BottleNeck and cannot read file "
>-                 "%s.\n", bottleneck_file);
>+          printf(
>+              "Error No value provided for BottleNeck and cannot read file "
>+              "%s.\n",
>+              bottleneck_file);
>           exit(0);
>         } else {
>           printf("reading bottleneck rates from file %s\n\n", bottleneck_file);
>@@ -637,8 +648,8 @@ int main(int argc, char* argv[]) {
>             }
> 
>             if (fwrite(streamDataTransCoding, sizeof(uint8_t),
>-                       streamLenTransCoding, transcodingBitstream) !=
>-                streamLenTransCoding) {
>+                       streamLenTransCoding,
>+                       transcodingBitstream) != streamLenTransCoding) {
>               return -1;
>             }
> 
>@@ -718,8 +729,7 @@ int main(int argc, char* argv[]) {
>           fprintf(stderr, "Error in RED trans-coding\n");
>           exit(0);
>         }
>-        streamLenTransCoding =
>-            static_cast<size_t>(streamLenTransCoding_int);
>+        streamLenTransCoding = static_cast<size_t>(streamLenTransCoding_int);
>       }
>     }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc
>index 4194478de76..59a3ade5dde 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/test/SwitchingSampRate/SwitchingSampRate.cc
>@@ -13,20 +13,19 @@
> //
> 
> #include <iostream>
>-#include "isac.h"
>-#include "utility.h"
>-#include "signal_processing_library.h"
> 
>-#define MAX_FILE_NAME  500
>-#define MAX_NUM_CLIENTS 2
>+#include "common_audio/signal_processing/include/signal_processing_library.h"
>+#include "modules/audio_coding/codecs/isac/main/include/isac.h"
>+#include "modules/audio_coding/codecs/isac/main/util/utility.h"
> 
>+#define MAX_FILE_NAME 500
>+#define MAX_NUM_CLIENTS 2
> 
> #define NUM_CLIENTS 2
> 
> using namespace std;
> 
>-int main(int argc, char* argv[])
>-{
>+int main(int argc, char* argv[]) {
>   char fileNameWB[MAX_FILE_NAME];
>   char fileNameSWB[MAX_FILE_NAME];
> 
>@@ -67,21 +66,18 @@ int main(int argc, char* argv[])
>   printf("    iSAC-swb version %s\n", versionNumber);
>   printf("____________________________________________\n");
> 
>-
>-  fileNameWB[0]  = '\0';
>+  fileNameWB[0] = '\0';
>   fileNameSWB[0] = '\0';
> 
>   char myFlag[20];
>   strcpy(myFlag, "-wb");
>   // READ THE WIDEBAND AND SUPER-WIDEBAND FILE NAMES
>-  if(readParamString(argc, argv, myFlag, fileNameWB, MAX_FILE_NAME) <= 0)
>-  {
>+  if (readParamString(argc, argv, myFlag, fileNameWB, MAX_FILE_NAME) <= 0) {
>     printf("No wideband file is specified");
>   }
> 
>   strcpy(myFlag, "-swb");
>-  if(readParamString(argc, argv, myFlag, fileNameSWB, MAX_FILE_NAME) <= 0)
>-  {
>+  if (readParamString(argc, argv, myFlag, fileNameSWB, MAX_FILE_NAME) <= 0) {
>     printf("No super-wideband file is specified");
>   }
> 
>@@ -96,16 +92,15 @@ int main(int argc, char* argv[])
>   strcpy(myFlag, "-I");
>   short codingMode = readSwitch(argc, argv, myFlag);
> 
>-  for(clientCntr = 0; clientCntr < NUM_CLIENTS; clientCntr++)
>-  {
>+  for (clientCntr = 0; clientCntr < NUM_CLIENTS; clientCntr++) {
>     codecInstance[clientCntr] = NULL;
> 
>     printf("\n");
>     printf("Client %d\n", clientCntr + 1);
>     printf("---------\n");
>-    printf("Starting %s",
>-           (encoderSampRate[clientCntr] == 16000)
>-           ? "wideband":"super-wideband");
>+    printf("Starting %s", (encoderSampRate[clientCntr] == 16000)
>+                              ? "wideband"
>+                              : "super-wideband");
> 
>     // Open output File Name
>     OPEN_FILE_WB(outFile[clientCntr], outFileName[clientCntr]);
>@@ -113,30 +108,27 @@ int main(int argc, char* argv[])
> 
>     samplesIn10ms[clientCntr] = encoderSampRate[clientCntr] * 10;
> 
>-    if(codingMode == 1)
>-    {
>-      bottleneck[clientCntr] = (clientCntr)? bnSWB:bnWB;
>-    }
>-    else
>-    {
>-      bottleneck[clientCntr] = (clientCntr)? minBn:maxBn;
>+    if (codingMode == 1) {
>+      bottleneck[clientCntr] = (clientCntr) ? bnSWB : bnWB;
>+    } else {
>+      bottleneck[clientCntr] = (clientCntr) ? minBn : maxBn;
>     }
> 
>     printf("Bottleneck....................... %0.3f kbits/sec \n",
>            bottleneck[clientCntr] / 1000.0);
> 
>     // coding-mode
>-    printf("Encoding Mode.................... %s\n",
>-           (codingMode == 1)? "Channel-Independent (Instantaneous)":"Adaptive");
>+    printf(
>+        "Encoding Mode.................... %s\n",
>+        (codingMode == 1) ? "Channel-Independent (Instantaneous)" : "Adaptive");
> 
>     lenEncodedInBytes[clientCntr] = 0;
>     lenAudioIn10ms[clientCntr] = 0;
>     lenEncodedInBytesTmp[clientCntr] = 0;
>     lenAudioIn10msTmp[clientCntr] = 0;
> 
>-    packetData[clientCntr] = (BottleNeckModel*)new(BottleNeckModel);
>-    if(packetData[clientCntr] == NULL)
>-    {
>+    packetData[clientCntr] = (BottleNeckModel*)new (BottleNeckModel);
>+    if (packetData[clientCntr] == NULL) {
>       printf("Could not allocate memory for packetData \n");
>       return -1;
>     }
>@@ -144,24 +136,22 @@ int main(int argc, char* argv[])
>     memset(resamplerState[clientCntr], 0, sizeof(int32_t) * 8);
>   }
> 
>-  for(clientCntr = 0; clientCntr < NUM_CLIENTS; clientCntr++)
>-  {
>+  for (clientCntr = 0; clientCntr < NUM_CLIENTS; clientCntr++) {
>     // Create
>-    if(WebRtcIsac_Create(&codecInstance[clientCntr]))
>-    {
>+    if (WebRtcIsac_Create(&codecInstance[clientCntr])) {
>       printf("Could not creat client %d\n", clientCntr + 1);
>       return -1;
>     }
> 
>-    WebRtcIsac_SetEncSampRate(codecInstance[clientCntr], encoderSampRate[clientCntr]);
>+    WebRtcIsac_SetEncSampRate(codecInstance[clientCntr],
>+                              encoderSampRate[clientCntr]);
> 
>-    WebRtcIsac_SetDecSampRate(codecInstance[clientCntr],
>-                              encoderSampRate[clientCntr + (1 - ((clientCntr & 1)<<1))]);
>+    WebRtcIsac_SetDecSampRate(
>+        codecInstance[clientCntr],
>+        encoderSampRate[clientCntr + (1 - ((clientCntr & 1) << 1))]);
> 
>     // Initialize Encoder
>-    if(WebRtcIsac_EncoderInit(codecInstance[clientCntr],
>-                              codingMode) < 0)
>-    {
>+    if (WebRtcIsac_EncoderInit(codecInstance[clientCntr], codingMode) < 0) {
>       printf("Could not initialize client, %d\n", clientCntr + 1);
>       return -1;
>     }
>@@ -169,12 +159,10 @@ int main(int argc, char* argv[])
>     WebRtcIsac_DecoderInit(codecInstance[clientCntr]);
> 
>     // setup Rate if in Instantaneous mode
>-    if(codingMode != 0)
>-    {
>+    if (codingMode != 0) {
>       // ONLY Clients who are not in Adaptive mode
>-      if(WebRtcIsac_Control(codecInstance[clientCntr],
>-                            bottleneck[clientCntr], 30) < 0)
>-      {
>+      if (WebRtcIsac_Control(codecInstance[clientCntr], bottleneck[clientCntr],
>+                             30) < 0) {
>         printf("Could not setup bottleneck and frame-size for client %d\n",
>                clientCntr + 1);
>         return -1;
>@@ -182,7 +170,6 @@ int main(int argc, char* argv[])
>     }
>   }
> 
>-
>   size_t streamLen;
>   short numSamplesRead;
>   size_t lenDecodedAudio;
>@@ -191,7 +178,7 @@ int main(int argc, char* argv[])
> 
>   printf("\n");
>   short num10ms[MAX_NUM_CLIENTS];
>-  memset(num10ms, 0, sizeof(short)*MAX_NUM_CLIENTS);
>+  memset(num10ms, 0, sizeof(short) * MAX_NUM_CLIENTS);
>   FILE* arrivalTimeFile1 = fopen("arrivalTime1.dat", "wb");
>   FILE* arrivalTimeFile2 = fopen("arrivalTime2.dat", "wb");
>   short numPrint[MAX_NUM_CLIENTS];
>@@ -204,61 +191,60 @@ int main(int argc, char* argv[])
>   short audioBuff60ms[60 * 32];
>   short resampledAudio60ms[60 * 32];
> 
>-  unsigned short bitStream[600+600];
>+  unsigned short bitStream[600 + 600];
>   short speechType[1];
> 
>   short numSampFreqChanged = 0;
>-  while(numSampFreqChanged < 10)
>-  {
>-    for(clientCntr = 0; clientCntr < NUM_CLIENTS; clientCntr++)
>-    {
>+  while (numSampFreqChanged < 10) {
>+    for (clientCntr = 0; clientCntr < NUM_CLIENTS; clientCntr++) {
>       // Encoding/decoding for this pair of clients, if there is
>       // audio for any of them
>-      //if(audioLeft[clientCntr] || audioLeft[clientCntr + 1])
>+      // if(audioLeft[clientCntr] || audioLeft[clientCntr + 1])
>       //{
>-      //for(pairCntr = 0; pairCntr < 2; pairCntr++)
>+      // for(pairCntr = 0; pairCntr < 2; pairCntr++)
>       //{
>-      senderIdx = clientCntr; // + pairCntr;
>-      receiverIdx = 1 - clientCntr;//  + (1 - pairCntr);
>+      senderIdx = clientCntr;        // + pairCntr;
>+      receiverIdx = 1 - clientCntr;  //  + (1 - pairCntr);
> 
>-      //if(num10ms[senderIdx] > 6)
>+      // if(num10ms[senderIdx] > 6)
>       //{
>       //    printf("Too many frames read for client %d",
>       //        senderIdx + 1);
>       //    return -1;
>       //}
> 
>-      numSamplesRead = (short)fread(audioBuff10ms, sizeof(short),
>-                                    samplesIn10ms[senderIdx], inFile[senderIdx]);
>-      if(numSamplesRead != samplesIn10ms[senderIdx])
>-      {
>+      numSamplesRead =
>+          (short)fread(audioBuff10ms, sizeof(short), samplesIn10ms[senderIdx],
>+                       inFile[senderIdx]);
>+      if (numSamplesRead != samplesIn10ms[senderIdx]) {
>         // file finished switch encoder sampling frequency.
>-        printf("Changing Encoder Sampling frequency in client %d to ", senderIdx+1);
>+        printf("Changing Encoder Sampling frequency in client %d to ",
>+               senderIdx + 1);
>         fclose(inFile[senderIdx]);
>         numSampFreqChanged++;
>-        if(encoderSampRate[senderIdx] == 16000)
>-        {
>+        if (encoderSampRate[senderIdx] == 16000) {
>           printf("super-wideband.\n");
>           OPEN_FILE_RB(inFile[senderIdx], fileNameSWB);
>           encoderSampRate[senderIdx] = 32000;
>-        }
>-        else
>-        {
>+        } else {
>           printf("wideband.\n");
>           OPEN_FILE_RB(inFile[senderIdx], fileNameWB);
>           encoderSampRate[senderIdx] = 16000;
>         }
>-        WebRtcIsac_SetEncSampRate(codecInstance[senderIdx], encoderSampRate[senderIdx]);
>-        WebRtcIsac_SetDecSampRate(codecInstance[receiverIdx], encoderSampRate[senderIdx]);
>+        WebRtcIsac_SetEncSampRate(codecInstance[senderIdx],
>+                                  encoderSampRate[senderIdx]);
>+        WebRtcIsac_SetDecSampRate(codecInstance[receiverIdx],
>+                                  encoderSampRate[senderIdx]);
> 
>         samplesIn10ms[clientCntr] = encoderSampRate[clientCntr] * 10;
> 
>-        numSamplesRead = (short)fread(audioBuff10ms, sizeof(short),
>-                                      samplesIn10ms[senderIdx], inFile[senderIdx]);
>-        if(numSamplesRead != samplesIn10ms[senderIdx])
>-        {
>+        numSamplesRead =
>+            (short)fread(audioBuff10ms, sizeof(short), samplesIn10ms[senderIdx],
>+                         inFile[senderIdx]);
>+        if (numSamplesRead != samplesIn10ms[senderIdx]) {
>           printf(" File %s for client %d has not enough audio\n",
>-                 (encoderSampRate[senderIdx]==16000)? "wideband":"super-wideband",
>+                 (encoderSampRate[senderIdx] == 16000) ? "wideband"
>+                                                       : "super-wideband",
>                  senderIdx + 1);
>           return -1;
>         }
>@@ -266,39 +252,34 @@ int main(int argc, char* argv[])
>       num10ms[senderIdx]++;
> 
>       // sanity check
>-      //if(num10ms[senderIdx] > 6)
>+      // if(num10ms[senderIdx] > 6)
>       //{
>-      //    printf("Client %d has got more than 60 ms audio and encoded no packet.\n",
>+      //    printf("Client %d has got more than 60 ms audio and encoded no
>+      //    packet.\n",
>       //        senderIdx);
>       //    return -1;
>       //}
> 
>       // Encode
> 
>-
>       int streamLen_int = WebRtcIsac_Encode(codecInstance[senderIdx],
>-                                            audioBuff10ms,
>-                                            (uint8_t*)bitStream);
>+                                            audioBuff10ms, (uint8_t*)bitStream);
>       int16_t ggg;
>       if (streamLen_int > 0) {
>         if ((WebRtcIsac_ReadFrameLen(
>                 codecInstance[receiverIdx],
>-                reinterpret_cast<const uint8_t*>(bitStream),
>-                &ggg)) < 0)
>+                reinterpret_cast<const uint8_t*>(bitStream), &ggg)) < 0)
>           printf("ERROR\n");
>       }
> 
>       // Sanity check
>-      if(streamLen_int < 0)
>-      {
>+      if (streamLen_int < 0) {
>         printf(" Encoder error in client %d \n", senderIdx + 1);
>         return -1;
>       }
>       streamLen = static_cast<size_t>(streamLen_int);
> 
>-
>-      if(streamLen > 0)
>-      {
>+      if (streamLen > 0) {
>         // Packet generated; model sending through a channel, do bandwidth
>         // estimation at the receiver and decode.
>         lenEncodedInBytes[senderIdx] += streamLen;
>@@ -307,32 +288,30 @@ int main(int argc, char* argv[])
>         lenAudioIn10msTmp[senderIdx] += (unsigned int)num10ms[senderIdx];
> 
>         // Print after ~5 sec.
>-        if(lenAudioIn10msTmp[senderIdx] >= 100)
>-        {
>+        if (lenAudioIn10msTmp[senderIdx] >= 100) {
>           numPrint[senderIdx]++;
>-          printf("  %d,  %6.3f => %6.3f ", senderIdx+1,
>+          printf("  %d,  %6.3f => %6.3f ", senderIdx + 1,
>                  bottleneck[senderIdx] / 1000.0,
>                  lenEncodedInBytesTmp[senderIdx] * 0.8 /
>-                 lenAudioIn10msTmp[senderIdx]);
>+                     lenAudioIn10msTmp[senderIdx]);
> 
>-          if(codingMode == 0)
>-          {
>+          if (codingMode == 0) {
>             int32_t bn;
>             WebRtcIsac_GetUplinkBw(codecInstance[senderIdx], &bn);
>             printf("[%d] ", bn);
>           }
>-          //int16_t rateIndexLB;
>-          //int16_t rateIndexUB;
>-          //WebRtcIsac_GetDownLinkBwIndex(codecInstance[receiverIdx],
>+          // int16_t rateIndexLB;
>+          // int16_t rateIndexUB;
>+          // WebRtcIsac_GetDownLinkBwIndex(codecInstance[receiverIdx],
>           //    &rateIndexLB, &rateIndexUB);
>-          //printf(" (%2d, %2d) ", rateIndexLB, rateIndexUB);
>+          // printf(" (%2d, %2d) ", rateIndexLB, rateIndexUB);
> 
>           cout << flush;
>           lenEncodedInBytesTmp[senderIdx] = 0;
>-          lenAudioIn10msTmp[senderIdx]    = 0;
>-          //if(senderIdx == (NUM_CLIENTS - 1))
>+          lenAudioIn10msTmp[senderIdx] = 0;
>+          // if(senderIdx == (NUM_CLIENTS - 1))
>           //{
>-          printf("  %0.1f \n", lenAudioIn10ms[senderIdx] * 10. /1000);
>+          printf("  %0.1f \n", lenAudioIn10ms[senderIdx] * 10. / 1000);
>           //}
> 
>           // After ~20 sec change the bottleneck.
>@@ -384,23 +363,20 @@ int main(int argc, char* argv[])
> 
>         // model a channel of given bottleneck, to get the receive timestamp
>         get_arrival_time(num10ms[senderIdx] * samplesIn10ms[senderIdx],
>-                         streamLen, bottleneck[senderIdx], packetData[senderIdx],
>-                         encoderSampRate[senderIdx]*1000, encoderSampRate[senderIdx]*1000);
>+                         streamLen, bottleneck[senderIdx],
>+                         packetData[senderIdx],
>+                         encoderSampRate[senderIdx] * 1000,
>+                         encoderSampRate[senderIdx] * 1000);
> 
>         // Write the arrival time.
>-        if(senderIdx == 0)
>-        {
>+        if (senderIdx == 0) {
>           if (fwrite(&(packetData[senderIdx]->arrival_time),
>-                     sizeof(unsigned int),
>-                     1, arrivalTimeFile1) != 1) {
>+                     sizeof(unsigned int), 1, arrivalTimeFile1) != 1) {
>             return -1;
>           }
>-        }
>-        else
>-        {
>+        } else {
>           if (fwrite(&(packetData[senderIdx]->arrival_time),
>-                     sizeof(unsigned int),
>-                     1, arrivalTimeFile2) != 1) {
>+                     sizeof(unsigned int), 1, arrivalTimeFile2) != 1) {
>             return -1;
>           }
>         }
>@@ -408,8 +384,7 @@ int main(int argc, char* argv[])
>         // BWE
>         if (WebRtcIsac_UpdateBwEstimate(
>                 codecInstance[receiverIdx],
>-                reinterpret_cast<const uint8_t*>(bitStream),
>-                streamLen,
>+                reinterpret_cast<const uint8_t*>(bitStream), streamLen,
>                 packetData[senderIdx]->rtp_number,
>                 packetData[senderIdx]->sample_count,
>                 packetData[senderIdx]->arrival_time) < 0) {
>@@ -418,34 +393,27 @@ int main(int argc, char* argv[])
>         }
>         /**/
>         // Decode
>-        int lenDecodedAudio_int = WebRtcIsac_Decode(
>-            codecInstance[receiverIdx],
>-            reinterpret_cast<const uint8_t*>(bitStream),
>-            streamLen,
>-            audioBuff60ms,
>-            speechType);
>-        if(lenDecodedAudio_int < 0)
>-        {
>+        int lenDecodedAudio_int =
>+            WebRtcIsac_Decode(codecInstance[receiverIdx],
>+                              reinterpret_cast<const uint8_t*>(bitStream),
>+                              streamLen, audioBuff60ms, speechType);
>+        if (lenDecodedAudio_int < 0) {
>           printf(" Decoder error in client %d \n", receiverIdx + 1);
>           return -1;
>         }
>         lenDecodedAudio = static_cast<size_t>(lenDecodedAudio_int);
> 
>-        if(encoderSampRate[senderIdx] == 16000)
>-        {
>-          WebRtcSpl_UpsampleBy2(audioBuff60ms, lenDecodedAudio, resampledAudio60ms,
>+        if (encoderSampRate[senderIdx] == 16000) {
>+          WebRtcSpl_UpsampleBy2(audioBuff60ms, lenDecodedAudio,
>+                                resampledAudio60ms,
>                                 resamplerState[receiverIdx]);
>           if (fwrite(resampledAudio60ms, sizeof(short), lenDecodedAudio << 1,
>-                     outFile[receiverIdx]) !=
>-              lenDecodedAudio << 1) {
>+                     outFile[receiverIdx]) != lenDecodedAudio << 1) {
>             return -1;
>           }
>-        }
>-        else
>-        {
>+        } else {
>           if (fwrite(audioBuff60ms, sizeof(short), lenDecodedAudio,
>-                     outFile[receiverIdx]) !=
>-              lenDecodedAudio) {
>+                     outFile[receiverIdx]) != lenDecodedAudio) {
>             return -1;
>           }
>         }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/test/simpleKenny.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/test/simpleKenny.c
>index 0d2d2cb375e..4b48b5033a5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/test/simpleKenny.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/test/simpleKenny.c
>@@ -26,8 +26,8 @@
> #include <math.h>
> 
> /* include API */
>-#include "isac.h"
>-#include "utility.h"
>+#include "modules/audio_coding/codecs/isac/main/include/isac.h"
>+#include "modules/audio_coding/codecs/isac/main/util/utility.h"
> #include "rtc_base/format_macros.h"
> 
> /* max number of samples per frame (= 60 ms frame) */
>@@ -416,7 +416,7 @@ int main(int argc, char* argv[]) {
>   printf("Rate Limit.................. %d bits/sec \n", rateLimit);
> 
> #ifdef WIN32
>-  fprintf(bitrateFile, "%d  %10u     %d     %6.3f  %6.3f    %6.3f\n",
>+  fprintf(bitrateFile, "%d  %10lu     %d     %6.3f  %6.3f    %6.3f\n",
>           sampFreqKHz, framecnt, bottleneck, rateLB, rateUB, rate);
>   fclose(bitrateFile);
> #endif  // WIN32
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/util/utility.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/util/utility.c
>index d9c4332123c..56547b11c2d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/util/utility.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/util/utility.c
>@@ -12,7 +12,8 @@
> #include <stdio.h>
> #include <string.h>
> #include <math.h>
>-#include "utility.h"
>+
>+#include "modules/audio_coding/codecs/isac/main/util/utility.h"
> 
> /* function for reading audio data from PCM file */
> int
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/util/utility.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/util/utility.h
>index b5882a57fb3..1acc54251bd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/util/utility.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/util/utility.h
>@@ -11,134 +11,98 @@
> #ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_UTIL_UTILITY_H_
> #define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_UTIL_UTILITY_H_
> 
>-#include <stdlib.h>
> #include <stdio.h>
>+#include <stdlib.h>
> 
> #if defined(__cplusplus)
> extern "C" {
> #endif
> 
>-#define OPEN_FILE_WB(filePtr, fullPath)                         \
>-  do                                                            \
>-  {                                                             \
>-    if(fullPath != NULL)                                        \
>-    {                                                           \
>-      filePtr = fopen(fullPath, "wb");                          \
>-      if(filePtr == NULL)                                       \
>-      {                                                         \
>-        printf("could not open %s to write to.", fullPath);     \
>-        return -1;                                              \
>-      }                                                         \
>-    }                                                           \
>-    else                                                        \
>-    {                                                           \
>-      filePtr = NULL;                                           \
>-    }                                                           \
>-  }while(0)
>-
>-#define OPEN_FILE_AB(filePtr, fullPath)                         \
>-  do                                                            \
>-  {                                                             \
>-    if(fullPath != NULL)                                        \
>-    {                                                           \
>-      filePtr = fopen(fullPath, "ab");                          \
>-      if(filePtr == NULL)                                       \
>-      {                                                         \
>-        printf("could not open %s to write to.", fullPath);     \
>-        return -1;                                              \
>-      }                                                         \
>-    }                                                           \
>-    else                                                        \
>-    {                                                           \
>-      filePtr = NULL;                                           \
>-    }                                                           \
>-  }while(0)
>-
>-#define OPEN_FILE_RB(filePtr, fullPath)                         \
>-  do                                                            \
>-  {                                                             \
>-    if(fullPath != NULL)                                        \
>-    {                                                           \
>-      filePtr = fopen(fullPath, "rb");                          \
>-      if(filePtr == NULL)                                       \
>-      {                                                         \
>-        printf("could not open %s to read from.", fullPath);    \
>-        return -1;                                              \
>-      }                                                         \
>-    }                                                           \
>-    else                                                        \
>-    {                                                           \
>-      filePtr = NULL;                                           \
>-    }                                                           \
>-  }while(0)
>-
>-#define WRITE_FILE_D(bufferPtr, len, filePtr)           \
>-  do                                                    \
>-  {                                                     \
>-    if(filePtr != NULL)                                 \
>-    {                                                   \
>-      double dummy[1000];                               \
>-      int cntr;                                         \
>-      for(cntr = 0; cntr < (len); cntr++)               \
>-      {                                                 \
>-        dummy[cntr] = (double)bufferPtr[cntr];          \
>-      }                                                 \
>-      fwrite(dummy, sizeof(double), len, filePtr);      \
>-      fflush(filePtr);                                  \
>-    }                                                   \
>-  } while(0)
>-
>-  typedef struct {
>-    unsigned int whenPackGeneratedMs;
>-    unsigned int whenPrevPackLeftMs;
>-    unsigned int sendTimeMs ;          /* milisecond */
>-    unsigned int arrival_time;         /* samples */
>-    unsigned int sample_count;         /* samples, also used as "send time stamp" */
>-    unsigned int rtp_number;
>-  } BottleNeckModel;
>-
>-  void get_arrival_time(
>-      int              current_framesamples,   /* samples */
>-      size_t           packet_size,            /* bytes */
>-      int              bottleneck,             /* excluding headers; bits/s */
>-      BottleNeckModel* BN_data,
>-      short            senderSampFreqHz,
>-      short            receiverSampFreqHz);
>-
>-  /* function for reading audio data from PCM file */
>-  int readframe(
>-      short* data,
>-      FILE*  inp,
>-      int    length);
>-
>-  short readSwitch(
>-      int   argc,
>-      char* argv[],
>-      char* strID);
>-
>-  double readParamDouble(
>-      int    argc,
>-      char*  argv[],
>-      char*  strID,
>-      double defaultVal);
>-
>-  int readParamInt(
>-      int   argc,
>-      char* argv[],
>-      char* strID,
>-      int   defaultVal);
>-
>-  int readParamString(
>-      int   argc,
>-      char* argv[],
>-      char* strID,
>-      char* stringParam,
>-      int   maxSize);
>+#define OPEN_FILE_WB(filePtr, fullPath)                     \
>+  do {                                                      \
>+    if (fullPath != NULL) {                                 \
>+      filePtr = fopen(fullPath, "wb");                      \
>+      if (filePtr == NULL) {                                \
>+        printf("could not open %s to write to.", fullPath); \
>+        return -1;                                          \
>+      }                                                     \
>+    } else {                                                \
>+      filePtr = NULL;                                       \
>+    }                                                       \
>+  } while (0)
>+
>+#define OPEN_FILE_AB(filePtr, fullPath)                     \
>+  do {                                                      \
>+    if (fullPath != NULL) {                                 \
>+      filePtr = fopen(fullPath, "ab");                      \
>+      if (filePtr == NULL) {                                \
>+        printf("could not open %s to write to.", fullPath); \
>+        return -1;                                          \
>+      }                                                     \
>+    } else {                                                \
>+      filePtr = NULL;                                       \
>+    }                                                       \
>+  } while (0)
>+
>+#define OPEN_FILE_RB(filePtr, fullPath)                      \
>+  do {                                                       \
>+    if (fullPath != NULL) {                                  \
>+      filePtr = fopen(fullPath, "rb");                       \
>+      if (filePtr == NULL) {                                 \
>+        printf("could not open %s to read from.", fullPath); \
>+        return -1;                                           \
>+      }                                                      \
>+    } else {                                                 \
>+      filePtr = NULL;                                        \
>+    }                                                        \
>+  } while (0)
>+
>+#define WRITE_FILE_D(bufferPtr, len, filePtr)      \
>+  do {                                             \
>+    if (filePtr != NULL) {                         \
>+      double dummy[1000];                          \
>+      int cntr;                                    \
>+      for (cntr = 0; cntr < (len); cntr++) {       \
>+        dummy[cntr] = (double)bufferPtr[cntr];     \
>+      }                                            \
>+      fwrite(dummy, sizeof(double), len, filePtr); \
>+      fflush(filePtr);                             \
>+    }                                              \
>+  } while (0)
>+
>+typedef struct {
>+  unsigned int whenPackGeneratedMs;
>+  unsigned int whenPrevPackLeftMs;
>+  unsigned int sendTimeMs;   /* milisecond */
>+  unsigned int arrival_time; /* samples */
>+  unsigned int sample_count; /* samples, also used as "send time stamp" */
>+  unsigned int rtp_number;
>+} BottleNeckModel;
>+
>+void get_arrival_time(int current_framesamples, /* samples */
>+                      size_t packet_size,       /* bytes */
>+                      int bottleneck,           /* excluding headers; bits/s */
>+                      BottleNeckModel* BN_data,
>+                      short senderSampFreqHz,
>+                      short receiverSampFreqHz);
>+
>+/* function for reading audio data from PCM file */
>+int readframe(short* data, FILE* inp, int length);
>+
>+short readSwitch(int argc, char* argv[], char* strID);
>+
>+double readParamDouble(int argc, char* argv[], char* strID, double defaultVal);
>+
>+int readParamInt(int argc, char* argv[], char* strID, int defaultVal);
>+
>+int readParamString(int argc,
>+                    char* argv[],
>+                    char* strID,
>+                    char* stringParam,
>+                    int maxSize);
> 
> #if defined(__cplusplus)
> }
> #endif
> 
>-
>-
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.cc
>index 341336ee145..0bf3b19a3b1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.cc
>@@ -27,7 +27,7 @@ size_t LegacyEncodedAudioFrame::Duration() const {
>   return (ret < 0) ? 0 : static_cast<size_t>(ret);
> }
> 
>-rtc::Optional<AudioDecoder::EncodedAudioFrame::DecodeResult>
>+absl::optional<AudioDecoder::EncodedAudioFrame::DecodeResult>
> LegacyEncodedAudioFrame::Decode(rtc::ArrayView<int16_t> decoded) const {
>   AudioDecoder::SpeechType speech_type = AudioDecoder::kSpeech;
>   const int ret = decoder_->Decode(
>@@ -35,7 +35,7 @@ LegacyEncodedAudioFrame::Decode(rtc::ArrayView<int16_t> decoded) const {
>       decoded.size() * sizeof(int16_t), decoded.data(), &speech_type);
> 
>   if (ret < 0)
>-    return rtc::nullopt;
>+    return absl::nullopt;
> 
>   return DecodeResult{static_cast<size_t>(ret), speech_type};
> }
>@@ -68,10 +68,9 @@ std::vector<AudioDecoder::ParseResult> LegacyEncodedAudioFrame::SplitBySamples(
>         split_size_bytes * timestamps_per_ms / bytes_per_ms);
>     size_t byte_offset;
>     uint32_t timestamp_offset;
>-    for (byte_offset = 0, timestamp_offset = 0;
>-         byte_offset < payload.size();
>+    for (byte_offset = 0, timestamp_offset = 0; byte_offset < payload.size();
>          byte_offset += split_size_bytes,
>-             timestamp_offset += timestamps_per_chunk) {
>+        timestamp_offset += timestamps_per_chunk) {
>       split_size_bytes =
>           std::min(split_size_bytes, payload.size() - byte_offset);
>       rtc::Buffer new_payload(payload.data() + byte_offset, split_size_bytes);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h
>index 275576e628f..05d4fe489c9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame.h
>@@ -32,7 +32,7 @@ class LegacyEncodedAudioFrame final : public AudioDecoder::EncodedAudioFrame {
> 
>   size_t Duration() const override;
> 
>-  rtc::Optional<DecodeResult> Decode(
>+  absl::optional<DecodeResult> Decode(
>       rtc::ArrayView<int16_t> decoded) const override;
> 
>   // For testing:
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame_unittest.cc
>index e2dd4454e10..9079bcd1ddd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/legacy_encoded_audio_frame_unittest.cc
>@@ -97,20 +97,15 @@ TEST_P(SplitBySamplesTest, PayloadSizes) {
>   // 40 ms -> 20 + 20 ms
>   // 50 ms -> 25 + 25 ms
>   // 60 ms -> 30 + 30 ms
>-  ExpectedSplit expected_splits[] = {
>-    {10, 1, {10}},
>-    {20, 1, {20}},
>-    {30, 1, {30}},
>-    {40, 2, {20, 20}},
>-    {50, 2, {25, 25}},
>-    {60, 2, {30, 30}}
>-  };
>+  ExpectedSplit expected_splits[] = {{10, 1, {10}},     {20, 1, {20}},
>+                                     {30, 1, {30}},     {40, 2, {20, 20}},
>+                                     {50, 2, {25, 25}}, {60, 2, {30, 30}}};
> 
>   for (const auto& expected_split : expected_splits) {
>     // The payload values are set to steadily increase (modulo 256), so that the
>     // resulting frames can be checked and we can be reasonably certain no
>     // sample was missed or repeated.
>-    const auto generate_payload = [] (size_t num_bytes) {
>+    const auto generate_payload = [](size_t num_bytes) {
>       rtc::Buffer payload(num_bytes);
>       uint8_t value = 0;
>       // Allow wrap-around of value in counter below.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
>index 3d10b6fdbea..302b7140319 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/audio_decoder_opus.cc
>@@ -36,7 +36,9 @@ class OpusFrame : public AudioDecoder::EncodedAudioFrame {
>     return (ret < 0) ? 0 : static_cast<size_t>(ret);
>   }
> 
>-  rtc::Optional<DecodeResult> Decode(
>+  bool IsDtxPacket() const override { return payload_.size() <= 2; }
>+
>+  absl::optional<DecodeResult> Decode(
>       rtc::ArrayView<int16_t> decoded) const override {
>     AudioDecoder::SpeechType speech_type = AudioDecoder::kSpeech;
>     int ret;
>@@ -51,7 +53,7 @@ class OpusFrame : public AudioDecoder::EncodedAudioFrame {
>     }
> 
>     if (ret < 0)
>-      return rtc::nullopt;
>+      return absl::nullopt;
> 
>     return DecodeResult{static_cast<size_t>(ret), speech_type};
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
>index caac4ae34c3..e6240e69826 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc
>@@ -14,6 +14,7 @@
> #include <iterator>
> #include <utility>
> 
>+#include "absl/memory/memory.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h"
> #include "modules/audio_coding/audio_network_adaptor/controller_manager.h"
>@@ -25,7 +26,6 @@
> #include "rtc_base/numerics/safe_conversions.h"
> #include "rtc_base/numerics/safe_minmax.h"
> #include "rtc_base/protobuf_utils.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/string_to_number.h"
> #include "rtc_base/timeutils.h"
> #include "system_wrappers/include/field_trial.h"
>@@ -105,18 +105,18 @@ float OptimizePacketLossRate(float new_loss_rate, float old_loss_rate) {
>   }
> }
> 
>-rtc::Optional<std::string> GetFormatParameter(const SdpAudioFormat& format,
>-                                              const std::string& param) {
>+absl::optional<std::string> GetFormatParameter(const SdpAudioFormat& format,
>+                                               const std::string& param) {
>   auto it = format.parameters.find(param);
>   if (it == format.parameters.end())
>-    return rtc::nullopt;
>+    return absl::nullopt;
> 
>   return it->second;
> }
> 
> template <typename T>
>-rtc::Optional<T> GetFormatParameter(const SdpAudioFormat& format,
>-                                    const std::string& param) {
>+absl::optional<T> GetFormatParameter(const SdpAudioFormat& format,
>+                                     const std::string& param) {
>   return rtc::StringToNumber<T>(GetFormatParameter(format, param).value_or(""));
> }
> 
>@@ -139,7 +139,7 @@ int CalculateDefaultBitrate(int max_playback_rate, size_t num_channels) {
> // out how invalid it is and accurately log invalid values.
> int CalculateBitrate(int max_playback_rate_hz,
>                      size_t num_channels,
>-                     rtc::Optional<std::string> bitrate_param) {
>+                     absl::optional<std::string> bitrate_param) {
>   const int default_bitrate =
>       CalculateDefaultBitrate(max_playback_rate_hz, num_channels);
> 
>@@ -239,10 +239,10 @@ std::unique_ptr<AudioEncoder> AudioEncoderOpusImpl::MakeAudioEncoder(
>     const AudioEncoderOpusConfig& config,
>     int payload_type) {
>   RTC_DCHECK(config.IsOk());
>-  return rtc::MakeUnique<AudioEncoderOpusImpl>(config, payload_type);
>+  return absl::make_unique<AudioEncoderOpusImpl>(config, payload_type);
> }
> 
>-rtc::Optional<AudioCodecInfo> AudioEncoderOpusImpl::QueryAudioEncoder(
>+absl::optional<AudioCodecInfo> AudioEncoderOpusImpl::QueryAudioEncoder(
>     const SdpAudioFormat& format) {
>   if (STR_CASE_CMP(format.name.c_str(), GetPayloadName()) == 0 &&
>       format.clockrate_hz == 48000 && format.num_channels == 2) {
>@@ -258,7 +258,7 @@ rtc::Optional<AudioCodecInfo> AudioEncoderOpusImpl::QueryAudioEncoder(
> 
>     return info;
>   }
>-  return rtc::nullopt;
>+  return absl::nullopt;
> }
> 
> AudioEncoderOpusConfig AudioEncoderOpusImpl::CreateConfig(
>@@ -274,11 +274,11 @@ AudioEncoderOpusConfig AudioEncoderOpusImpl::CreateConfig(
>   return config;
> }
> 
>-rtc::Optional<AudioEncoderOpusConfig> AudioEncoderOpusImpl::SdpToConfig(
>+absl::optional<AudioEncoderOpusConfig> AudioEncoderOpusImpl::SdpToConfig(
>     const SdpAudioFormat& format) {
>   if (STR_CASE_CMP(format.name.c_str(), "opus") != 0 ||
>       format.clockrate_hz != 48000 || format.num_channels != 2) {
>-    return rtc::nullopt;
>+    return absl::nullopt;
>   }
> 
>   AudioEncoderOpusConfig config;
>@@ -313,7 +313,7 @@ rtc::Optional<AudioEncoderOpusConfig> AudioEncoderOpusImpl::SdpToConfig(
>   return config;
> }
> 
>-rtc::Optional<int> AudioEncoderOpusImpl::GetNewComplexity(
>+absl::optional<int> AudioEncoderOpusImpl::GetNewComplexity(
>     const AudioEncoderOpusConfig& config) {
>   RTC_DCHECK(config.IsOk());
>   const int bitrate_bps = GetBitrateBps(config);
>@@ -322,7 +322,7 @@ rtc::Optional<int> AudioEncoderOpusImpl::GetNewComplexity(
>       bitrate_bps <= config.complexity_threshold_bps +
>                          config.complexity_threshold_window_bps) {
>     // Within the hysteresis window; make no change.
>-    return rtc::nullopt;
>+    return absl::nullopt;
>   } else {
>     return bitrate_bps <= config.complexity_threshold_bps
>                ? config.low_rate_complexity
>@@ -330,7 +330,7 @@ rtc::Optional<int> AudioEncoderOpusImpl::GetNewComplexity(
>   }
> }
> 
>-rtc::Optional<int> AudioEncoderOpusImpl::GetNewBandwidth(
>+absl::optional<int> AudioEncoderOpusImpl::GetNewBandwidth(
>     const AudioEncoderOpusConfig& config,
>     OpusEncInst* inst) {
>   constexpr int kMinWidebandBitrate = 8000;
>@@ -339,17 +339,17 @@ rtc::Optional<int> AudioEncoderOpusImpl::GetNewBandwidth(
>   RTC_DCHECK(config.IsOk());
>   const int bitrate = GetBitrateBps(config);
>   if (bitrate > kAutomaticThreshold) {
>-    return rtc::Optional<int>(OPUS_AUTO);
>+    return absl::optional<int>(OPUS_AUTO);
>   }
>   const int bandwidth = WebRtcOpus_GetBandwidth(inst);
>   RTC_DCHECK_GE(bandwidth, 0);
>   if (bitrate > kMaxNarrowbandBitrate && bandwidth < OPUS_BANDWIDTH_WIDEBAND) {
>-    return rtc::Optional<int>(OPUS_BANDWIDTH_WIDEBAND);
>+    return absl::optional<int>(OPUS_BANDWIDTH_WIDEBAND);
>   } else if (bitrate < kMinWidebandBitrate &&
>              bandwidth > OPUS_BANDWIDTH_NARROWBAND) {
>-    return rtc::Optional<int>(OPUS_BANDWIDTH_NARROWBAND);
>+    return absl::optional<int>(OPUS_BANDWIDTH_NARROWBAND);
>   }
>-  return rtc::Optional<int>();
>+  return absl::optional<int>();
> }
> 
> class AudioEncoderOpusImpl::PacketLossFractionSmoother {
>@@ -388,7 +388,7 @@ AudioEncoderOpusImpl::AudioEncoderOpusImpl(const AudioEncoderOpusConfig& config,
>             return DefaultAudioNetworkAdaptorCreator(config_string, event_log);
>           },
>           // We choose 5sec as initial time constant due to empirical data.
>-          rtc::MakeUnique<SmoothingFilterImpl>(5000)) {}
>+          absl::make_unique<SmoothingFilterImpl>(5000)) {}
> 
> AudioEncoderOpusImpl::AudioEncoderOpusImpl(
>     const AudioEncoderOpusConfig& config,
>@@ -529,7 +529,7 @@ void AudioEncoderOpusImpl::OnReceivedUplinkRecoverablePacketLossFraction(
> 
> void AudioEncoderOpusImpl::OnReceivedUplinkBandwidth(
>     int target_audio_bitrate_bps,
>-    rtc::Optional<int64_t> bwe_period_ms) {
>+    absl::optional<int64_t> bwe_period_ms) {
>   if (audio_network_adaptor_) {
>     audio_network_adaptor_->SetTargetAudioBitrate(target_audio_bitrate_bps);
>     // We give smoothed bitrate allocation to audio network adaptor as
>@@ -613,20 +613,17 @@ AudioEncoder::EncodedInfo AudioEncoderOpusImpl::EncodeImpl(
> 
>   const size_t max_encoded_bytes = SufficientOutputBufferSize();
>   EncodedInfo info;
>-  info.encoded_bytes =
>-      encoded->AppendData(
>-          max_encoded_bytes, [&] (rtc::ArrayView<uint8_t> encoded) {
>-            int status = WebRtcOpus_Encode(
>-                inst_, &input_buffer_[0],
>-                rtc::CheckedDivExact(input_buffer_.size(),
>-                                     config_.num_channels),
>-                rtc::saturated_cast<int16_t>(max_encoded_bytes),
>-                encoded.data());
>-
>-            RTC_CHECK_GE(status, 0);  // Fails only if fed invalid data.
>-
>-            return static_cast<size_t>(status);
>-          });
>+  info.encoded_bytes = encoded->AppendData(
>+      max_encoded_bytes, [&](rtc::ArrayView<uint8_t> encoded) {
>+        int status = WebRtcOpus_Encode(
>+            inst_, &input_buffer_[0],
>+            rtc::CheckedDivExact(input_buffer_.size(), config_.num_channels),
>+            rtc::saturated_cast<int16_t>(max_encoded_bytes), encoded.data());
>+
>+        RTC_CHECK_GE(status, 0);  // Fails only if fed invalid data.
>+
>+        return static_cast<size_t>(status);
>+      });
>   input_buffer_.clear();
> 
>   bool dtx_frame = (info.encoded_bytes <= 2);
>@@ -801,7 +798,7 @@ void AudioEncoderOpusImpl::MaybeUpdateUplinkBandwidth() {
>     if (!bitrate_smoother_last_update_time_ ||
>         now_ms - *bitrate_smoother_last_update_time_ >=
>             config_.uplink_bandwidth_update_interval_ms) {
>-      rtc::Optional<float> smoothed_bitrate = bitrate_smoother_->GetAverage();
>+      absl::optional<float> smoothed_bitrate = bitrate_smoother_->GetAverage();
>       if (smoothed_bitrate)
>         audio_network_adaptor_->SetUplinkBandwidth(*smoothed_bitrate);
>       bitrate_smoother_last_update_time_ = now_ms;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
>index 49c5207740a..ea4b2650bbb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h
>@@ -16,10 +16,10 @@
> #include <string>
> #include <vector>
> 
>+#include "absl/types/optional.h"
> #include "api/audio_codecs/audio_encoder.h"
> #include "api/audio_codecs/audio_format.h"
> #include "api/audio_codecs/opus/audio_encoder_opus_config.h"
>-#include "api/optional.h"
> #include "common_audio/smoothing_filter.h"
> #include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
> #include "modules/audio_coding/codecs/opus/opus_interface.h"
>@@ -40,13 +40,13 @@ class AudioEncoderOpusImpl final : public AudioEncoder {
>   // defined by complexity_threshold_bps +/- complexity_threshold_window_bps.
>   // Otherwise, returns the current complexity depending on whether the
>   // current bitrate is above or below complexity_threshold_bps.
>-  static rtc::Optional<int> GetNewComplexity(
>+  static absl::optional<int> GetNewComplexity(
>       const AudioEncoderOpusConfig& config);
> 
>   // Returns OPUS_AUTO if the the current bitrate is above wideband threshold.
>   // Returns empty if it is below, but bandwidth coincides with the desired one.
>   // Otherwise returns the desired bandwidth.
>-  static rtc::Optional<int> GetNewBandwidth(
>+  static absl::optional<int> GetNewBandwidth(
>       const AudioEncoderOpusConfig& config,
>       OpusEncInst* inst);
> 
>@@ -69,7 +69,7 @@ class AudioEncoderOpusImpl final : public AudioEncoder {
> 
>   // Static interface for use by BuiltinAudioEncoderFactory.
>   static constexpr const char* GetPayloadName() { return "opus"; }
>-  static rtc::Optional<AudioCodecInfo> QueryAudioEncoder(
>+  static absl::optional<AudioCodecInfo> QueryAudioEncoder(
>       const SdpAudioFormat& format);
> 
>   int SampleRateHz() const override;
>@@ -98,7 +98,7 @@ class AudioEncoderOpusImpl final : public AudioEncoder {
>       float uplink_recoverable_packet_loss_fraction) override;
>   void OnReceivedUplinkBandwidth(
>       int target_audio_bitrate_bps,
>-      rtc::Optional<int64_t> bwe_period_ms) override;
>+      absl::optional<int64_t> bwe_period_ms) override;
>   void OnReceivedRtt(int rtt_ms) override;
>   void OnReceivedOverhead(size_t overhead_bytes_per_packet) override;
>   void SetReceiverFrameLengthRange(int min_frame_length_ms,
>@@ -125,7 +125,7 @@ class AudioEncoderOpusImpl final : public AudioEncoder {
>  private:
>   class PacketLossFractionSmoother;
> 
>-  static rtc::Optional<AudioEncoderOpusConfig> SdpToConfig(
>+  static absl::optional<AudioEncoderOpusConfig> SdpToConfig(
>       const SdpAudioFormat& format);
>   static void AppendSupportedEncoders(std::vector<AudioCodecSpec>* specs);
>   static AudioCodecInfo QueryAudioEncoder(const AudioEncoderOpusConfig& config);
>@@ -167,9 +167,9 @@ class AudioEncoderOpusImpl final : public AudioEncoder {
>   std::unique_ptr<PacketLossFractionSmoother> packet_loss_fraction_smoother_;
>   const AudioNetworkAdaptorCreator audio_network_adaptor_creator_;
>   std::unique_ptr<AudioNetworkAdaptor> audio_network_adaptor_;
>-  rtc::Optional<size_t> overhead_bytes_per_packet_;
>+  absl::optional<size_t> overhead_bytes_per_packet_;
>   const std::unique_ptr<SmoothingFilter> bitrate_smoother_;
>-  rtc::Optional<int64_t> bitrate_smoother_last_update_time_;
>+  absl::optional<int64_t> bitrate_smoother_last_update_time_;
>   int consecutive_dtx_frames_;
> 
>   friend struct AudioEncoderOpus;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
>index dfef6821ad8..7a6d5fda4bb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
>@@ -12,6 +12,7 @@
> #include <memory>
> #include <utility>
> 
>+#include "absl/memory/memory.h"
> #include "api/audio_codecs/opus/audio_encoder_opus.h"
> #include "common_audio/mocks/mock_smoothing_filter.h"
> #include "common_types.h"  // NOLINT(build/include)
>@@ -54,43 +55,39 @@ AudioEncoderOpusConfig CreateConfigWithParameters(
> }
> 
> struct AudioEncoderOpusStates {
>-  std::shared_ptr<MockAudioNetworkAdaptor*> mock_audio_network_adaptor;
>+  MockAudioNetworkAdaptor* mock_audio_network_adaptor;
>   MockSmoothingFilter* mock_bitrate_smoother;
>   std::unique_ptr<AudioEncoderOpusImpl> encoder;
>   std::unique_ptr<rtc::ScopedFakeClock> fake_clock;
>   AudioEncoderOpusConfig config;
> };
> 
>-AudioEncoderOpusStates CreateCodec(size_t num_channels) {
>-  AudioEncoderOpusStates states;
>-  states.mock_audio_network_adaptor =
>-      std::make_shared<MockAudioNetworkAdaptor*>(nullptr);
>-  states.fake_clock.reset(new rtc::ScopedFakeClock());
>-  states.fake_clock->SetTimeMicros(kInitialTimeUs);
>-  std::weak_ptr<MockAudioNetworkAdaptor*> mock_ptr(
>-      states.mock_audio_network_adaptor);
>+std::unique_ptr<AudioEncoderOpusStates> CreateCodec(size_t num_channels) {
>+  std::unique_ptr<AudioEncoderOpusStates> states =
>+      absl::make_unique<AudioEncoderOpusStates>();
>+  states->mock_audio_network_adaptor = nullptr;
>+  states->fake_clock.reset(new rtc::ScopedFakeClock());
>+  states->fake_clock->SetTimeMicros(kInitialTimeUs);
>+
>+  MockAudioNetworkAdaptor** mock_ptr = &states->mock_audio_network_adaptor;
>   AudioEncoderOpusImpl::AudioNetworkAdaptorCreator creator =
>       [mock_ptr](const std::string&, RtcEventLog* event_log) {
>         std::unique_ptr<MockAudioNetworkAdaptor> adaptor(
>             new NiceMock<MockAudioNetworkAdaptor>());
>         EXPECT_CALL(*adaptor, Die());
>-        if (auto sp = mock_ptr.lock()) {
>-          *sp = adaptor.get();
>-        } else {
>-          RTC_NOTREACHED();
>-        }
>+        *mock_ptr = adaptor.get();
>         return adaptor;
>       };
> 
>   CodecInst codec_inst = kDefaultOpusSettings;
>   codec_inst.channels = num_channels;
>-  states.config = CreateConfig(codec_inst);
>+  states->config = CreateConfig(codec_inst);
>   std::unique_ptr<MockSmoothingFilter> bitrate_smoother(
>       new MockSmoothingFilter());
>-  states.mock_bitrate_smoother = bitrate_smoother.get();
>+  states->mock_bitrate_smoother = bitrate_smoother.get();
> 
>-  states.encoder.reset(new AudioEncoderOpusImpl(
>-      states.config, codec_inst.pltype, std::move(creator),
>+  states->encoder.reset(new AudioEncoderOpusImpl(
>+      states->config, codec_inst.pltype, std::move(creator),
>       std::move(bitrate_smoother)));
>   return states;
> }
>@@ -145,77 +142,77 @@ std::unique_ptr<test::AudioLoop> Create10msAudioBlocks(
> TEST(AudioEncoderOpusTest, DefaultApplicationModeMono) {
>   auto states = CreateCodec(1);
>   EXPECT_EQ(AudioEncoderOpusConfig::ApplicationMode::kVoip,
>-            states.encoder->application());
>+            states->encoder->application());
> }
> 
> TEST(AudioEncoderOpusTest, DefaultApplicationModeStereo) {
>   auto states = CreateCodec(2);
>   EXPECT_EQ(AudioEncoderOpusConfig::ApplicationMode::kAudio,
>-            states.encoder->application());
>+            states->encoder->application());
> }
> 
> TEST(AudioEncoderOpusTest, ChangeApplicationMode) {
>   auto states = CreateCodec(2);
>   EXPECT_TRUE(
>-      states.encoder->SetApplication(AudioEncoder::Application::kSpeech));
>+      states->encoder->SetApplication(AudioEncoder::Application::kSpeech));
>   EXPECT_EQ(AudioEncoderOpusConfig::ApplicationMode::kVoip,
>-            states.encoder->application());
>+            states->encoder->application());
> }
> 
> TEST(AudioEncoderOpusTest, ResetWontChangeApplicationMode) {
>   auto states = CreateCodec(2);
> 
>   // Trigger a reset.
>-  states.encoder->Reset();
>+  states->encoder->Reset();
>   // Verify that the mode is still kAudio.
>   EXPECT_EQ(AudioEncoderOpusConfig::ApplicationMode::kAudio,
>-            states.encoder->application());
>+            states->encoder->application());
> 
>   // Now change to kVoip.
>   EXPECT_TRUE(
>-      states.encoder->SetApplication(AudioEncoder::Application::kSpeech));
>+      states->encoder->SetApplication(AudioEncoder::Application::kSpeech));
>   EXPECT_EQ(AudioEncoderOpusConfig::ApplicationMode::kVoip,
>-            states.encoder->application());
>+            states->encoder->application());
> 
>   // Trigger a reset again.
>-  states.encoder->Reset();
>+  states->encoder->Reset();
>   // Verify that the mode is still kVoip.
>   EXPECT_EQ(AudioEncoderOpusConfig::ApplicationMode::kVoip,
>-            states.encoder->application());
>+            states->encoder->application());
> }
> 
> TEST(AudioEncoderOpusTest, ToggleDtx) {
>   auto states = CreateCodec(2);
>   // Enable DTX
>-  EXPECT_TRUE(states.encoder->SetDtx(true));
>-  EXPECT_TRUE(states.encoder->GetDtx());
>+  EXPECT_TRUE(states->encoder->SetDtx(true));
>+  EXPECT_TRUE(states->encoder->GetDtx());
>   // Turn off DTX.
>-  EXPECT_TRUE(states.encoder->SetDtx(false));
>-  EXPECT_FALSE(states.encoder->GetDtx());
>+  EXPECT_TRUE(states->encoder->SetDtx(false));
>+  EXPECT_FALSE(states->encoder->GetDtx());
> }
> 
> TEST(AudioEncoderOpusTest,
>      OnReceivedUplinkBandwidthWithoutAudioNetworkAdaptor) {
>   auto states = CreateCodec(1);
>-  // Constants are replicated from audio_states.encoderopus.cc.
>+  // Constants are replicated from audio_states->encoderopus.cc.
>   const int kMinBitrateBps = 6000;
>   const int kMaxBitrateBps = 510000;
>   // Set a too low bitrate.
>-  states.encoder->OnReceivedUplinkBandwidth(kMinBitrateBps - 1, rtc::nullopt);
>-  EXPECT_EQ(kMinBitrateBps, states.encoder->GetTargetBitrate());
>+  states->encoder->OnReceivedUplinkBandwidth(kMinBitrateBps - 1, absl::nullopt);
>+  EXPECT_EQ(kMinBitrateBps, states->encoder->GetTargetBitrate());
>   // Set a too high bitrate.
>-  states.encoder->OnReceivedUplinkBandwidth(kMaxBitrateBps + 1, rtc::nullopt);
>-  EXPECT_EQ(kMaxBitrateBps, states.encoder->GetTargetBitrate());
>+  states->encoder->OnReceivedUplinkBandwidth(kMaxBitrateBps + 1, absl::nullopt);
>+  EXPECT_EQ(kMaxBitrateBps, states->encoder->GetTargetBitrate());
>   // Set the minimum rate.
>-  states.encoder->OnReceivedUplinkBandwidth(kMinBitrateBps, rtc::nullopt);
>-  EXPECT_EQ(kMinBitrateBps, states.encoder->GetTargetBitrate());
>+  states->encoder->OnReceivedUplinkBandwidth(kMinBitrateBps, absl::nullopt);
>+  EXPECT_EQ(kMinBitrateBps, states->encoder->GetTargetBitrate());
>   // Set the maximum rate.
>-  states.encoder->OnReceivedUplinkBandwidth(kMaxBitrateBps, rtc::nullopt);
>-  EXPECT_EQ(kMaxBitrateBps, states.encoder->GetTargetBitrate());
>+  states->encoder->OnReceivedUplinkBandwidth(kMaxBitrateBps, absl::nullopt);
>+  EXPECT_EQ(kMaxBitrateBps, states->encoder->GetTargetBitrate());
>   // Set rates from kMaxBitrateBps up to 32000 bps.
>   for (int rate = kMinBitrateBps; rate <= 32000; rate += 1000) {
>-    states.encoder->OnReceivedUplinkBandwidth(rate, rtc::nullopt);
>-    EXPECT_EQ(rate, states.encoder->GetTargetBitrate());
>+    states->encoder->OnReceivedUplinkBandwidth(rate, absl::nullopt);
>+    EXPECT_EQ(rate, states->encoder->GetTargetBitrate());
>   }
> }
> 
>@@ -237,7 +234,7 @@ std::vector<float> IntervalSteps(float a, float b, size_t n) {
> // Sets the packet loss rate to each number in the vector in turn, and verifies
> // that the loss rate as reported by the encoder is |expected_return| for all
> // of them.
>-void TestSetPacketLossRate(AudioEncoderOpusStates* states,
>+void TestSetPacketLossRate(const AudioEncoderOpusStates* states,
>                            const std::vector<float>& losses,
>                            float expected_return) {
>   // |kSampleIntervalMs| is chosen to ease the calculation since
>@@ -246,8 +243,7 @@ void TestSetPacketLossRate(AudioEncoderOpusStates* states,
>   constexpr int64_t kSampleIntervalMs = 184198;
>   for (float loss : losses) {
>     states->encoder->OnReceivedUplinkPacketLossFraction(loss);
>-    states->fake_clock->AdvanceTime(
>-        rtc::TimeDelta::FromMilliseconds(kSampleIntervalMs));
>+    states->fake_clock->AdvanceTime(TimeDelta::ms(kSampleIntervalMs));
>     EXPECT_FLOAT_EQ(expected_return, states->encoder->packet_loss_rate());
>   }
> }
>@@ -262,17 +258,17 @@ TEST(AudioEncoderOpusTest, PacketLossRateOptimized) {
>   // Note that the order of the following calls is critical.
> 
>   // clang-format off
>-  TestSetPacketLossRate(&states, I(0.00f      , 0.01f - eps), 0.00f);
>-  TestSetPacketLossRate(&states, I(0.01f + eps, 0.06f - eps), 0.01f);
>-  TestSetPacketLossRate(&states, I(0.06f + eps, 0.11f - eps), 0.05f);
>-  TestSetPacketLossRate(&states, I(0.11f + eps, 0.22f - eps), 0.10f);
>-  TestSetPacketLossRate(&states, I(0.22f + eps, 1.00f      ), 0.20f);
>-
>-  TestSetPacketLossRate(&states, I(1.00f      , 0.18f + eps), 0.20f);
>-  TestSetPacketLossRate(&states, I(0.18f - eps, 0.09f + eps), 0.10f);
>-  TestSetPacketLossRate(&states, I(0.09f - eps, 0.04f + eps), 0.05f);
>-  TestSetPacketLossRate(&states, I(0.04f - eps, 0.01f + eps), 0.01f);
>-  TestSetPacketLossRate(&states, I(0.01f - eps, 0.00f      ), 0.00f);
>+  TestSetPacketLossRate(states.get(), I(0.00f      , 0.01f - eps), 0.00f);
>+  TestSetPacketLossRate(states.get(), I(0.01f + eps, 0.06f - eps), 0.01f);
>+  TestSetPacketLossRate(states.get(), I(0.06f + eps, 0.11f - eps), 0.05f);
>+  TestSetPacketLossRate(states.get(), I(0.11f + eps, 0.22f - eps), 0.10f);
>+  TestSetPacketLossRate(states.get(), I(0.22f + eps, 1.00f      ), 0.20f);
>+
>+  TestSetPacketLossRate(states.get(), I(1.00f      , 0.18f + eps), 0.20f);
>+  TestSetPacketLossRate(states.get(), I(0.18f - eps, 0.09f + eps), 0.10f);
>+  TestSetPacketLossRate(states.get(), I(0.09f - eps, 0.04f + eps), 0.05f);
>+  TestSetPacketLossRate(states.get(), I(0.04f - eps, 0.01f + eps), 0.01f);
>+  TestSetPacketLossRate(states.get(), I(0.01f - eps, 0.00f      ), 0.00f);
>   // clang-format on
> }
> 
>@@ -282,85 +278,85 @@ TEST(AudioEncoderOpusTest, SetReceiverFrameLengthRange) {
>   // |supported_frame_lengths_ms| should contain only the frame length being
>   // used.
>   using ::testing::ElementsAre;
>-  EXPECT_THAT(states.encoder->supported_frame_lengths_ms(),
>-              ElementsAre(states.encoder->next_frame_length_ms()));
>-  states.encoder->SetReceiverFrameLengthRange(0, 12345);
>-  states.encoder->SetReceiverFrameLengthRange(21, 60);
>-  EXPECT_THAT(states.encoder->supported_frame_lengths_ms(), ElementsAre(60));
>-  states.encoder->SetReceiverFrameLengthRange(20, 59);
>-  EXPECT_THAT(states.encoder->supported_frame_lengths_ms(), ElementsAre(20));
>+  EXPECT_THAT(states->encoder->supported_frame_lengths_ms(),
>+              ElementsAre(states->encoder->next_frame_length_ms()));
>+  states->encoder->SetReceiverFrameLengthRange(0, 12345);
>+  states->encoder->SetReceiverFrameLengthRange(21, 60);
>+  EXPECT_THAT(states->encoder->supported_frame_lengths_ms(), ElementsAre(60));
>+  states->encoder->SetReceiverFrameLengthRange(20, 59);
>+  EXPECT_THAT(states->encoder->supported_frame_lengths_ms(), ElementsAre(20));
> }
> 
> TEST(AudioEncoderOpusTest,
>      InvokeAudioNetworkAdaptorOnReceivedUplinkPacketLossFraction) {
>   auto states = CreateCodec(2);
>-  states.encoder->EnableAudioNetworkAdaptor("", nullptr);
>+  states->encoder->EnableAudioNetworkAdaptor("", nullptr);
> 
>   auto config = CreateEncoderRuntimeConfig();
>-  EXPECT_CALL(**states.mock_audio_network_adaptor, GetEncoderRuntimeConfig())
>+  EXPECT_CALL(*states->mock_audio_network_adaptor, GetEncoderRuntimeConfig())
>       .WillOnce(Return(config));
> 
>   // Since using mock audio network adaptor, any packet loss fraction is fine.
>   constexpr float kUplinkPacketLoss = 0.1f;
>-  EXPECT_CALL(**states.mock_audio_network_adaptor,
>+  EXPECT_CALL(*states->mock_audio_network_adaptor,
>               SetUplinkPacketLossFraction(kUplinkPacketLoss));
>-  states.encoder->OnReceivedUplinkPacketLossFraction(kUplinkPacketLoss);
>+  states->encoder->OnReceivedUplinkPacketLossFraction(kUplinkPacketLoss);
> 
>-  CheckEncoderRuntimeConfig(states.encoder.get(), config);
>+  CheckEncoderRuntimeConfig(states->encoder.get(), config);
> }
> 
> TEST(AudioEncoderOpusTest, InvokeAudioNetworkAdaptorOnReceivedUplinkBandwidth) {
>   auto states = CreateCodec(2);
>-  states.encoder->EnableAudioNetworkAdaptor("", nullptr);
>+  states->encoder->EnableAudioNetworkAdaptor("", nullptr);
> 
>   auto config = CreateEncoderRuntimeConfig();
>-  EXPECT_CALL(**states.mock_audio_network_adaptor, GetEncoderRuntimeConfig())
>+  EXPECT_CALL(*states->mock_audio_network_adaptor, GetEncoderRuntimeConfig())
>       .WillOnce(Return(config));
> 
>   // Since using mock audio network adaptor, any target audio bitrate is fine.
>   constexpr int kTargetAudioBitrate = 30000;
>   constexpr int64_t kProbingIntervalMs = 3000;
>-  EXPECT_CALL(**states.mock_audio_network_adaptor,
>+  EXPECT_CALL(*states->mock_audio_network_adaptor,
>               SetTargetAudioBitrate(kTargetAudioBitrate));
>-  EXPECT_CALL(*states.mock_bitrate_smoother,
>+  EXPECT_CALL(*states->mock_bitrate_smoother,
>               SetTimeConstantMs(kProbingIntervalMs * 4));
>-  EXPECT_CALL(*states.mock_bitrate_smoother, AddSample(kTargetAudioBitrate));
>-  states.encoder->OnReceivedUplinkBandwidth(kTargetAudioBitrate,
>-                                            kProbingIntervalMs);
>+  EXPECT_CALL(*states->mock_bitrate_smoother, AddSample(kTargetAudioBitrate));
>+  states->encoder->OnReceivedUplinkBandwidth(kTargetAudioBitrate,
>+                                             kProbingIntervalMs);
> 
>-  CheckEncoderRuntimeConfig(states.encoder.get(), config);
>+  CheckEncoderRuntimeConfig(states->encoder.get(), config);
> }
> 
> TEST(AudioEncoderOpusTest, InvokeAudioNetworkAdaptorOnReceivedRtt) {
>   auto states = CreateCodec(2);
>-  states.encoder->EnableAudioNetworkAdaptor("", nullptr);
>+  states->encoder->EnableAudioNetworkAdaptor("", nullptr);
> 
>   auto config = CreateEncoderRuntimeConfig();
>-  EXPECT_CALL(**states.mock_audio_network_adaptor, GetEncoderRuntimeConfig())
>+  EXPECT_CALL(*states->mock_audio_network_adaptor, GetEncoderRuntimeConfig())
>       .WillOnce(Return(config));
> 
>   // Since using mock audio network adaptor, any rtt is fine.
>   constexpr int kRtt = 30;
>-  EXPECT_CALL(**states.mock_audio_network_adaptor, SetRtt(kRtt));
>-  states.encoder->OnReceivedRtt(kRtt);
>+  EXPECT_CALL(*states->mock_audio_network_adaptor, SetRtt(kRtt));
>+  states->encoder->OnReceivedRtt(kRtt);
> 
>-  CheckEncoderRuntimeConfig(states.encoder.get(), config);
>+  CheckEncoderRuntimeConfig(states->encoder.get(), config);
> }
> 
> TEST(AudioEncoderOpusTest, InvokeAudioNetworkAdaptorOnReceivedOverhead) {
>   auto states = CreateCodec(2);
>-  states.encoder->EnableAudioNetworkAdaptor("", nullptr);
>+  states->encoder->EnableAudioNetworkAdaptor("", nullptr);
> 
>   auto config = CreateEncoderRuntimeConfig();
>-  EXPECT_CALL(**states.mock_audio_network_adaptor, GetEncoderRuntimeConfig())
>+  EXPECT_CALL(*states->mock_audio_network_adaptor, GetEncoderRuntimeConfig())
>       .WillOnce(Return(config));
> 
>   // Since using mock audio network adaptor, any overhead is fine.
>   constexpr size_t kOverhead = 64;
>-  EXPECT_CALL(**states.mock_audio_network_adaptor, SetOverhead(kOverhead));
>-  states.encoder->OnReceivedOverhead(kOverhead);
>+  EXPECT_CALL(*states->mock_audio_network_adaptor, SetOverhead(kOverhead));
>+  states->encoder->OnReceivedOverhead(kOverhead);
> 
>-  CheckEncoderRuntimeConfig(states.encoder.get(), config);
>+  CheckEncoderRuntimeConfig(states->encoder.get(), config);
> }
> 
> TEST(AudioEncoderOpusTest,
>@@ -376,18 +372,17 @@ TEST(AudioEncoderOpusTest,
>   constexpr int64_t kSecondSampleTimeMs = 6931;
> 
>   // First time, no filtering.
>-  states.encoder->OnReceivedUplinkPacketLossFraction(kPacketLossFraction_1);
>-  EXPECT_FLOAT_EQ(0.01f, states.encoder->packet_loss_rate());
>+  states->encoder->OnReceivedUplinkPacketLossFraction(kPacketLossFraction_1);
>+  EXPECT_FLOAT_EQ(0.01f, states->encoder->packet_loss_rate());
> 
>-  states.fake_clock->AdvanceTime(
>-      rtc::TimeDelta::FromMilliseconds(kSecondSampleTimeMs));
>-  states.encoder->OnReceivedUplinkPacketLossFraction(kPacketLossFraction_2);
>+  states->fake_clock->AdvanceTime(TimeDelta::ms(kSecondSampleTimeMs));
>+  states->encoder->OnReceivedUplinkPacketLossFraction(kPacketLossFraction_2);
> 
>   // Now the output of packet loss fraction smoother should be
>   // (0.02 + 0.198) / 2 = 0.109, which reach the threshold for the optimized
>   // packet loss rate to increase to 0.05. If no smoothing has been made, the
>   // optimized packet loss rate should have been increase to 0.1.
>-  EXPECT_FLOAT_EQ(0.05f, states.encoder->packet_loss_rate());
>+  EXPECT_FLOAT_EQ(0.05f, states->encoder->packet_loss_rate());
> }
> 
> TEST(AudioEncoderOpusTest, DoNotInvokeSetTargetBitrateIfOverheadUnknown) {
>@@ -396,12 +391,12 @@ TEST(AudioEncoderOpusTest, DoNotInvokeSetTargetBitrateIfOverheadUnknown) {
> 
>   auto states = CreateCodec(2);
> 
>-  states.encoder->OnReceivedUplinkBandwidth(kDefaultOpusSettings.rate * 2,
>-                                            rtc::nullopt);
>+  states->encoder->OnReceivedUplinkBandwidth(kDefaultOpusSettings.rate * 2,
>+                                             absl::nullopt);
> 
>   // Since |OnReceivedOverhead| has not been called, the codec bitrate should
>   // not change.
>-  EXPECT_EQ(kDefaultOpusSettings.rate, states.encoder->GetTargetBitrate());
>+  EXPECT_EQ(kDefaultOpusSettings.rate, states->encoder->GetTargetBitrate());
> }
> 
> TEST(AudioEncoderOpusTest, OverheadRemovedFromTargetAudioBitrate) {
>@@ -411,15 +406,15 @@ TEST(AudioEncoderOpusTest, OverheadRemovedFromTargetAudioBitrate) {
>   auto states = CreateCodec(2);
> 
>   constexpr size_t kOverheadBytesPerPacket = 64;
>-  states.encoder->OnReceivedOverhead(kOverheadBytesPerPacket);
>+  states->encoder->OnReceivedOverhead(kOverheadBytesPerPacket);
> 
>   constexpr int kTargetBitrateBps = 40000;
>-  states.encoder->OnReceivedUplinkBandwidth(kTargetBitrateBps, rtc::nullopt);
>+  states->encoder->OnReceivedUplinkBandwidth(kTargetBitrateBps, absl::nullopt);
> 
>   int packet_rate = rtc::CheckedDivExact(48000, kDefaultOpusSettings.pacsize);
>   EXPECT_EQ(kTargetBitrateBps -
>                 8 * static_cast<int>(kOverheadBytesPerPacket) * packet_rate,
>-            states.encoder->GetTargetBitrate());
>+            states->encoder->GetTargetBitrate());
> }
> 
> TEST(AudioEncoderOpusTest, BitrateBounded) {
>@@ -432,7 +427,7 @@ TEST(AudioEncoderOpusTest, BitrateBounded) {
>   auto states = CreateCodec(2);
> 
>   constexpr size_t kOverheadBytesPerPacket = 64;
>-  states.encoder->OnReceivedOverhead(kOverheadBytesPerPacket);
>+  states->encoder->OnReceivedOverhead(kOverheadBytesPerPacket);
> 
>   int packet_rate = rtc::CheckedDivExact(48000, kDefaultOpusSettings.pacsize);
> 
>@@ -440,15 +435,15 @@ TEST(AudioEncoderOpusTest, BitrateBounded) {
>   // subtracted. The eventual codec rate should be bounded by |kMinBitrateBps|.
>   int target_bitrate =
>       kOverheadBytesPerPacket * 8 * packet_rate + kMinBitrateBps - 1;
>-  states.encoder->OnReceivedUplinkBandwidth(target_bitrate, rtc::nullopt);
>-  EXPECT_EQ(kMinBitrateBps, states.encoder->GetTargetBitrate());
>+  states->encoder->OnReceivedUplinkBandwidth(target_bitrate, absl::nullopt);
>+  EXPECT_EQ(kMinBitrateBps, states->encoder->GetTargetBitrate());
> 
>   // Set a target rate that is greater than |kMaxBitrateBps| when overhead is
>   // subtracted. The eventual codec rate should be bounded by |kMaxBitrateBps|.
>   target_bitrate =
>       kOverheadBytesPerPacket * 8 * packet_rate + kMaxBitrateBps + 1;
>-  states.encoder->OnReceivedUplinkBandwidth(target_bitrate, rtc::nullopt);
>-  EXPECT_EQ(kMaxBitrateBps, states.encoder->GetTargetBitrate());
>+  states->encoder->OnReceivedUplinkBandwidth(target_bitrate, absl::nullopt);
>+  EXPECT_EQ(kMaxBitrateBps, states->encoder->GetTargetBitrate());
> }
> 
> // Verifies that the complexity adaptation in the config works as intended.
>@@ -459,7 +454,7 @@ TEST(AudioEncoderOpusTest, ConfigComplexityAdaptation) {
> 
>   // Bitrate within hysteresis window. Expect empty output.
>   config.bitrate_bps = 12500;
>-  EXPECT_EQ(rtc::nullopt, AudioEncoderOpusImpl::GetNewComplexity(config));
>+  EXPECT_EQ(absl::nullopt, AudioEncoderOpusImpl::GetNewComplexity(config));
> 
>   // Bitrate below hysteresis window. Expect higher complexity.
>   config.bitrate_bps = 10999;
>@@ -467,7 +462,7 @@ TEST(AudioEncoderOpusTest, ConfigComplexityAdaptation) {
> 
>   // Bitrate within hysteresis window. Expect empty output.
>   config.bitrate_bps = 12500;
>-  EXPECT_EQ(rtc::nullopt, AudioEncoderOpusImpl::GetNewComplexity(config));
>+  EXPECT_EQ(absl::nullopt, AudioEncoderOpusImpl::GetNewComplexity(config));
> 
>   // Bitrate above hysteresis window. Expect lower complexity.
>   config.bitrate_bps = 14001;
>@@ -493,9 +488,9 @@ TEST(AudioEncoderOpusTest, ConfigBandwidthAdaptation) {
>                        : 1));
> 
>   // Bitrate below minmum wideband. Expect narrowband.
>-  config.bitrate_bps = rtc::Optional<int>(7999);
>+  config.bitrate_bps = absl::optional<int>(7999);
>   auto bandwidth = AudioEncoderOpusImpl::GetNewBandwidth(config, inst);
>-  EXPECT_EQ(rtc::Optional<int>(OPUS_BANDWIDTH_NARROWBAND), bandwidth);
>+  EXPECT_EQ(absl::optional<int>(OPUS_BANDWIDTH_NARROWBAND), bandwidth);
>   WebRtcOpus_SetBandwidth(inst, *bandwidth);
>   // It is necessary to encode here because Opus has some logic in the encoder
>   // that goes from the user-set bandwidth to the used and returned one.
>@@ -504,14 +499,14 @@ TEST(AudioEncoderOpusTest, ConfigBandwidthAdaptation) {
>                     kMaxBytes, bitstream);
> 
>   // Bitrate not yet above maximum narrowband. Expect empty.
>-  config.bitrate_bps = rtc::Optional<int>(9000);
>+  config.bitrate_bps = absl::optional<int>(9000);
>   bandwidth = AudioEncoderOpusImpl::GetNewBandwidth(config, inst);
>-  EXPECT_EQ(rtc::Optional<int>(), bandwidth);
>+  EXPECT_EQ(absl::optional<int>(), bandwidth);
> 
>   // Bitrate above maximum narrowband. Expect wideband.
>-  config.bitrate_bps = rtc::Optional<int>(9001);
>+  config.bitrate_bps = absl::optional<int>(9001);
>   bandwidth = AudioEncoderOpusImpl::GetNewBandwidth(config, inst);
>-  EXPECT_EQ(rtc::Optional<int>(OPUS_BANDWIDTH_WIDEBAND), bandwidth);
>+  EXPECT_EQ(absl::optional<int>(OPUS_BANDWIDTH_WIDEBAND), bandwidth);
>   WebRtcOpus_SetBandwidth(inst, *bandwidth);
>   // It is necessary to encode here because Opus has some logic in the encoder
>   // that goes from the user-set bandwidth to the used and returned one.
>@@ -520,64 +515,64 @@ TEST(AudioEncoderOpusTest, ConfigBandwidthAdaptation) {
>                     kMaxBytes, bitstream);
> 
>   // Bitrate not yet below minimum wideband. Expect empty.
>-  config.bitrate_bps = rtc::Optional<int>(8000);
>+  config.bitrate_bps = absl::optional<int>(8000);
>   bandwidth = AudioEncoderOpusImpl::GetNewBandwidth(config, inst);
>-  EXPECT_EQ(rtc::Optional<int>(), bandwidth);
>+  EXPECT_EQ(absl::optional<int>(), bandwidth);
> 
>   // Bitrate above automatic threshold. Expect automatic.
>-  config.bitrate_bps = rtc::Optional<int>(12001);
>+  config.bitrate_bps = absl::optional<int>(12001);
>   bandwidth = AudioEncoderOpusImpl::GetNewBandwidth(config, inst);
>-  EXPECT_EQ(rtc::Optional<int>(OPUS_AUTO), bandwidth);
>+  EXPECT_EQ(absl::optional<int>(OPUS_AUTO), bandwidth);
> 
>   EXPECT_EQ(0, WebRtcOpus_EncoderFree(inst));
> }
> 
> TEST(AudioEncoderOpusTest, EmptyConfigDoesNotAffectEncoderSettings) {
>   auto states = CreateCodec(2);
>-  states.encoder->EnableAudioNetworkAdaptor("", nullptr);
>+  states->encoder->EnableAudioNetworkAdaptor("", nullptr);
> 
>   auto config = CreateEncoderRuntimeConfig();
>   AudioEncoderRuntimeConfig empty_config;
> 
>-  EXPECT_CALL(**states.mock_audio_network_adaptor, GetEncoderRuntimeConfig())
>+  EXPECT_CALL(*states->mock_audio_network_adaptor, GetEncoderRuntimeConfig())
>       .WillOnce(Return(config))
>       .WillOnce(Return(empty_config));
> 
>   constexpr size_t kOverhead = 64;
>-  EXPECT_CALL(**states.mock_audio_network_adaptor, SetOverhead(kOverhead))
>+  EXPECT_CALL(*states->mock_audio_network_adaptor, SetOverhead(kOverhead))
>       .Times(2);
>-  states.encoder->OnReceivedOverhead(kOverhead);
>-  states.encoder->OnReceivedOverhead(kOverhead);
>+  states->encoder->OnReceivedOverhead(kOverhead);
>+  states->encoder->OnReceivedOverhead(kOverhead);
> 
>-  CheckEncoderRuntimeConfig(states.encoder.get(), config);
>+  CheckEncoderRuntimeConfig(states->encoder.get(), config);
> }
> 
> TEST(AudioEncoderOpusTest, UpdateUplinkBandwidthInAudioNetworkAdaptor) {
>   auto states = CreateCodec(2);
>-  states.encoder->EnableAudioNetworkAdaptor("", nullptr);
>+  states->encoder->EnableAudioNetworkAdaptor("", nullptr);
>   std::array<int16_t, 480 * 2> audio;
>   audio.fill(0);
>   rtc::Buffer encoded;
>-  EXPECT_CALL(*states.mock_bitrate_smoother, GetAverage())
>+  EXPECT_CALL(*states->mock_bitrate_smoother, GetAverage())
>       .WillOnce(Return(50000));
>-  EXPECT_CALL(**states.mock_audio_network_adaptor, SetUplinkBandwidth(50000));
>-  states.encoder->Encode(
>+  EXPECT_CALL(*states->mock_audio_network_adaptor, SetUplinkBandwidth(50000));
>+  states->encoder->Encode(
>       0, rtc::ArrayView<const int16_t>(audio.data(), audio.size()), &encoded);
> 
>   // Repeat update uplink bandwidth tests.
>   for (int i = 0; i < 5; i++) {
>     // Don't update till it is time to update again.
>-    states.fake_clock->AdvanceTime(rtc::TimeDelta::FromMilliseconds(
>-        states.config.uplink_bandwidth_update_interval_ms - 1));
>-    states.encoder->Encode(
>+    states->fake_clock->AdvanceTime(
>+        TimeDelta::ms(states->config.uplink_bandwidth_update_interval_ms - 1));
>+    states->encoder->Encode(
>         0, rtc::ArrayView<const int16_t>(audio.data(), audio.size()), &encoded);
> 
>     // Update when it is time to update.
>-    EXPECT_CALL(*states.mock_bitrate_smoother, GetAverage())
>+    EXPECT_CALL(*states->mock_bitrate_smoother, GetAverage())
>         .WillOnce(Return(40000));
>-    EXPECT_CALL(**states.mock_audio_network_adaptor, SetUplinkBandwidth(40000));
>-    states.fake_clock->AdvanceTime(rtc::TimeDelta::FromMilliseconds(1));
>-    states.encoder->Encode(
>+    EXPECT_CALL(*states->mock_audio_network_adaptor, SetUplinkBandwidth(40000));
>+    states->fake_clock->AdvanceTime(TimeDelta::ms(1));
>+    states->encoder->Encode(
>         0, rtc::ArrayView<const int16_t>(audio.data(), audio.size()), &encoded);
>   }
> }
>@@ -586,25 +581,25 @@ TEST(AudioEncoderOpusTest, EncodeAtMinBitrate) {
>   auto states = CreateCodec(1);
>   constexpr int kNumPacketsToEncode = 2;
>   auto audio_frames =
>-      Create10msAudioBlocks(states.encoder, kNumPacketsToEncode * 20);
>+      Create10msAudioBlocks(states->encoder, kNumPacketsToEncode * 20);
>   ASSERT_TRUE(audio_frames) << "Create10msAudioBlocks failed";
>   rtc::Buffer encoded;
>   uint32_t rtp_timestamp = 12345;  // Just a number not important to this test.
> 
>-  states.encoder->OnReceivedUplinkBandwidth(0, rtc::nullopt);
>+  states->encoder->OnReceivedUplinkBandwidth(0, absl::nullopt);
>   for (int packet_index = 0; packet_index < kNumPacketsToEncode;
>        packet_index++) {
>     // Make sure we are not encoding before we have enough data for
>     // a 20ms packet.
>     for (int index = 0; index < 1; index++) {
>-      states.encoder->Encode(rtp_timestamp, audio_frames->GetNextBlock(),
>-                             &encoded);
>+      states->encoder->Encode(rtp_timestamp, audio_frames->GetNextBlock(),
>+                              &encoded);
>       EXPECT_EQ(0u, encoded.size());
>     }
> 
>     // Should encode now.
>-    states.encoder->Encode(rtp_timestamp, audio_frames->GetNextBlock(),
>-                           &encoded);
>+    states->encoder->Encode(rtp_timestamp, audio_frames->GetNextBlock(),
>+                            &encoded);
>     EXPECT_GT(encoded.size(), 0u);
>     encoded.Clear();
>   }
>@@ -758,8 +753,8 @@ TEST(AudioEncoderOpusTest, SetMaxPlaybackRateNb) {
>   EXPECT_EQ(8000, config.max_playback_rate_hz);
>   EXPECT_EQ(12000, config.bitrate_bps);
> 
>-  config = CreateConfigWithParameters({{"maxplaybackrate", "8000"},
>-                                       {"stereo", "1"}});
>+  config = CreateConfigWithParameters(
>+      {{"maxplaybackrate", "8000"}, {"stereo", "1"}});
>   EXPECT_EQ(8000, config.max_playback_rate_hz);
>   EXPECT_EQ(24000, config.bitrate_bps);
> }
>@@ -770,8 +765,8 @@ TEST(AudioEncoderOpusTest, SetMaxPlaybackRateMb) {
>   EXPECT_EQ(8001, config.max_playback_rate_hz);
>   EXPECT_EQ(20000, config.bitrate_bps);
> 
>-  config = CreateConfigWithParameters({{"maxplaybackrate", "8001"},
>-                                       {"stereo", "1"}});
>+  config = CreateConfigWithParameters(
>+      {{"maxplaybackrate", "8001"}, {"stereo", "1"}});
>   EXPECT_EQ(8001, config.max_playback_rate_hz);
>   EXPECT_EQ(40000, config.bitrate_bps);
> }
>@@ -782,8 +777,8 @@ TEST(AudioEncoderOpusTest, SetMaxPlaybackRateWb) {
>   EXPECT_EQ(12001, config.max_playback_rate_hz);
>   EXPECT_EQ(20000, config.bitrate_bps);
> 
>-  config = CreateConfigWithParameters({{"maxplaybackrate", "12001"},
>-                                       {"stereo", "1"}});
>+  config = CreateConfigWithParameters(
>+      {{"maxplaybackrate", "12001"}, {"stereo", "1"}});
>   EXPECT_EQ(12001, config.max_playback_rate_hz);
>   EXPECT_EQ(40000, config.bitrate_bps);
> }
>@@ -794,8 +789,8 @@ TEST(AudioEncoderOpusTest, SetMaxPlaybackRateSwb) {
>   EXPECT_EQ(16001, config.max_playback_rate_hz);
>   EXPECT_EQ(32000, config.bitrate_bps);
> 
>-  config = CreateConfigWithParameters({{"maxplaybackrate", "16001"},
>-                                       {"stereo", "1"}});
>+  config = CreateConfigWithParameters(
>+      {{"maxplaybackrate", "16001"}, {"stereo", "1"}});
>   EXPECT_EQ(16001, config.max_playback_rate_hz);
>   EXPECT_EQ(64000, config.bitrate_bps);
> }
>@@ -806,8 +801,8 @@ TEST(AudioEncoderOpusTest, SetMaxPlaybackRateFb) {
>   EXPECT_EQ(24001, config.max_playback_rate_hz);
>   EXPECT_EQ(32000, config.bitrate_bps);
> 
>-  config = CreateConfigWithParameters({{"maxplaybackrate", "24001"},
>-                                       {"stereo", "1"}});
>+  config = CreateConfigWithParameters(
>+      {{"maxplaybackrate", "24001"}, {"stereo", "1"}});
>   EXPECT_EQ(24001, config.max_playback_rate_hz);
>   EXPECT_EQ(64000, config.bitrate_bps);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/opus_bandwidth_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/opus_bandwidth_unittest.cc
>index 4394949f965..7f09c2aa1a9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/opus_bandwidth_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/opus_bandwidth_unittest.cc
>@@ -105,12 +105,12 @@ TEST(BandwidthAdaptationTest, BandwidthAdaptationTest) {
>   test::ScopedFieldTrials override_field_trials(
>       "WebRTC-AdjustOpusBandwidth/Enabled/");
> 
>-  constexpr float kMaxNarrowbandRatio = 0.003f;
>+  constexpr float kMaxNarrowbandRatio = 0.0035f;
>   constexpr float kMinWidebandRatio = 0.03f;
> 
>   // Create encoder.
>   AudioEncoderOpusConfig enc_config;
>-  enc_config.bitrate_bps = rtc::Optional<int>(7999);
>+  enc_config.bitrate_bps = absl::optional<int>(7999);
>   enc_config.num_channels = kNumChannels;
>   constexpr int payload_type = 17;
>   auto encoder = AudioEncoderOpus::MakeAudioEncoder(enc_config, payload_type);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc
>index 4e0a17ecc13..4fab8a75a35 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc
>@@ -38,8 +38,8 @@ class OpusFecTest : public TestWithParam<coding_param> {
>  protected:
>   OpusFecTest();
> 
>-  virtual void SetUp();
>-  virtual void TearDown();
>+  void SetUp() override;
>+  void TearDown() override;
> 
>   virtual void EncodeABlock();
> 
>@@ -83,8 +83,8 @@ void OpusFecTest::SetUp() {
>   rewind(fp);
> 
>   // Allocate memory to contain the whole file.
>-  in_data_.reset(new int16_t[loop_length_samples_ +
>-      block_length_sample_ * channels_]);
>+  in_data_.reset(
>+      new int16_t[loop_length_samples_ + block_length_sample_ * channels_]);
> 
>   // Copy the file into the buffer.
>   ASSERT_EQ(fread(&in_data_[0], sizeof(int16_t), loop_length_samples_, fp),
>@@ -130,14 +130,12 @@ OpusFecTest::OpusFecTest()
>       max_bytes_(0),
>       encoded_bytes_(0),
>       opus_encoder_(NULL),
>-      opus_decoder_(NULL) {
>-}
>+      opus_decoder_(NULL) {}
> 
> void OpusFecTest::EncodeABlock() {
>-  int value = WebRtcOpus_Encode(opus_encoder_,
>-                                &in_data_[data_pointer_],
>-                                block_length_sample_,
>-                                max_bytes_, &bit_stream_[0]);
>+  int value =
>+      WebRtcOpus_Encode(opus_encoder_, &in_data_[data_pointer_],
>+                        block_length_sample_, max_bytes_, &bit_stream_[0]);
>   EXPECT_GT(value, 0);
> 
>   encoded_bytes_ = static_cast<size_t>(value);
>@@ -151,9 +149,9 @@ void OpusFecTest::DecodeABlock(bool lost_previous, bool lost_current) {
>     // Decode previous frame.
>     if (!lost_current &&
>         WebRtcOpus_PacketHasFec(&bit_stream_[0], encoded_bytes_) == 1) {
>-      value_1 = WebRtcOpus_DecodeFec(opus_decoder_, &bit_stream_[0],
>-                                     encoded_bytes_, &out_data_[0],
>-                                     &audio_type);
>+      value_1 =
>+          WebRtcOpus_DecodeFec(opus_decoder_, &bit_stream_[0], encoded_bytes_,
>+                               &out_data_[0], &audio_type);
>     } else {
>       value_1 = WebRtcOpus_DecodePlc(opus_decoder_, &out_data_[0], 1);
>     }
>@@ -173,16 +171,14 @@ TEST_P(OpusFecTest, RandomPacketLossTest) {
>   int time_now_ms, fec_frames;
>   int actual_packet_loss_rate;
>   bool lost_current, lost_previous;
>-  mode mode_set[3] = {{true, 0},
>-                      {false, 0},
>-                      {true, 50}};
>+  mode mode_set[3] = {{true, 0}, {false, 0}, {true, 50}};
> 
>   lost_current = false;
>   for (int i = 0; i < 3; i++) {
>     if (mode_set[i].fec) {
>       EXPECT_EQ(0, WebRtcOpus_EnableFec(opus_encoder_));
>-      EXPECT_EQ(0, WebRtcOpus_SetPacketLossRate(opus_encoder_,
>-          mode_set[i].target_packet_loss_rate));
>+      EXPECT_EQ(0, WebRtcOpus_SetPacketLossRate(
>+                       opus_encoder_, mode_set[i].target_packet_loss_rate));
>       printf("FEC is ON, target at packet loss rate %d percent.\n",
>              mode_set[i].target_packet_loss_rate);
>     } else {
>@@ -218,7 +214,7 @@ TEST_P(OpusFecTest, RandomPacketLossTest) {
>       // |data_pointer_| is incremented and wrapped across
>       // |loop_length_samples_|.
>       data_pointer_ = (data_pointer_ + block_length_sample_ * channels_) %
>-        loop_length_samples_;
>+                      loop_length_samples_;
>     }
>     if (mode_set[i].fec) {
>       printf("%.2f percent frames has FEC.\n",
>@@ -242,7 +238,6 @@ const coding_param param_set[] = {
>                     string("pcm"))};
> 
> // 64 kbps, stereo
>-INSTANTIATE_TEST_CASE_P(AllTest, OpusFecTest,
>-                        ::testing::ValuesIn(param_set));
>+INSTANTIATE_TEST_CASE_P(AllTest, OpusFecTest, ::testing::ValuesIn(param_set));
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/opus_inst.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/opus_inst.h
>index 066fa2267f2..2473a5c7538 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/opus_inst.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/opus_inst.h
>@@ -32,5 +32,4 @@ struct WebRtcOpusDecInst {
>   int in_dtx_mode;
> };
> 
>-
> #endif  // MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INST_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/opus_interface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/opus_interface.h
>index 4b8e892344a..ddb4ff9781f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/opus_interface.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/opus_interface.h
>@@ -12,9 +12,9 @@
> #define MODULES_AUDIO_CODING_CODECS_OPUS_OPUS_INTERFACE_H_
> 
> #include <stddef.h>
>+#include <stdint.h>
> 
> #include "modules/audio_coding/codecs/opus/opus_inst.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> #ifdef __cplusplus
> extern "C" {
>@@ -318,8 +318,10 @@ void WebRtcOpus_DecoderInit(OpusDecInst* inst);
>  * Return value              : >0 - Samples per channel in decoded vector
>  *                             -1 - Error
>  */
>-int WebRtcOpus_Decode(OpusDecInst* inst, const uint8_t* encoded,
>-                      size_t encoded_bytes, int16_t* decoded,
>+int WebRtcOpus_Decode(OpusDecInst* inst,
>+                      const uint8_t* encoded,
>+                      size_t encoded_bytes,
>+                      int16_t* decoded,
>                       int16_t* audio_type);
> 
> /****************************************************************************
>@@ -336,7 +338,8 @@ int WebRtcOpus_Decode(OpusDecInst* inst, const uint8_t* encoded,
>  * Return value                   : >0 - number of samples in decoded PLC vector
>  *                                  -1 - Error
>  */
>-int WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
>+int WebRtcOpus_DecodePlc(OpusDecInst* inst,
>+                         int16_t* decoded,
>                          int number_of_lost_frames);
> 
> /****************************************************************************
>@@ -357,8 +360,10 @@ int WebRtcOpus_DecodePlc(OpusDecInst* inst, int16_t* decoded,
>  *                              0 - No FEC data in the packet
>  *                             -1 - Error
>  */
>-int WebRtcOpus_DecodeFec(OpusDecInst* inst, const uint8_t* encoded,
>-                         size_t encoded_bytes, int16_t* decoded,
>+int WebRtcOpus_DecodeFec(OpusDecInst* inst,
>+                         const uint8_t* encoded,
>+                         size_t encoded_bytes,
>+                         int16_t* decoded,
>                          int16_t* audio_type);
> 
> /****************************************************************************
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/opus_speed_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/opus_speed_test.cc
>index ca46aa1b16c..03b59ed32ea 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/opus_speed_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/opus_speed_test.cc
>@@ -23,9 +23,12 @@ class OpusSpeedTest : public AudioCodecSpeedTest {
>   OpusSpeedTest();
>   void SetUp() override;
>   void TearDown() override;
>-  float EncodeABlock(int16_t* in_data, uint8_t* bit_stream,
>-                     size_t max_bytes, size_t* encoded_bytes) override;
>-  float DecodeABlock(const uint8_t* bit_stream, size_t encoded_bytes,
>+  float EncodeABlock(int16_t* in_data,
>+                     uint8_t* bit_stream,
>+                     size_t max_bytes,
>+                     size_t* encoded_bytes) override;
>+  float DecodeABlock(const uint8_t* bit_stream,
>+                     size_t encoded_bytes,
>                      int16_t* out_data) override;
>   WebRtcOpusEncInst* opus_encoder_;
>   WebRtcOpusDecInst* opus_decoder_;
>@@ -36,8 +39,7 @@ OpusSpeedTest::OpusSpeedTest()
>                           kOpusSamplingKhz,
>                           kOpusSamplingKhz),
>       opus_encoder_(NULL),
>-      opus_decoder_(NULL) {
>-}
>+      opus_decoder_(NULL) {}
> 
> void OpusSpeedTest::SetUp() {
>   AudioCodecSpeedTest::SetUp();
>@@ -57,12 +59,13 @@ void OpusSpeedTest::TearDown() {
>   EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
> }
> 
>-float OpusSpeedTest::EncodeABlock(int16_t* in_data, uint8_t* bit_stream,
>-                                  size_t max_bytes, size_t* encoded_bytes) {
>+float OpusSpeedTest::EncodeABlock(int16_t* in_data,
>+                                  uint8_t* bit_stream,
>+                                  size_t max_bytes,
>+                                  size_t* encoded_bytes) {
>   clock_t clocks = clock();
>-  int value = WebRtcOpus_Encode(opus_encoder_, in_data,
>-                                input_length_sample_, max_bytes,
>-                                bit_stream);
>+  int value = WebRtcOpus_Encode(opus_encoder_, in_data, input_length_sample_,
>+                                max_bytes, bit_stream);
>   clocks = clock() - clocks;
>   EXPECT_GT(value, 0);
>   *encoded_bytes = static_cast<size_t>(value);
>@@ -70,7 +73,8 @@ float OpusSpeedTest::EncodeABlock(int16_t* in_data, uint8_t* bit_stream,
> }
> 
> float OpusSpeedTest::DecodeABlock(const uint8_t* bit_stream,
>-                                  size_t encoded_bytes, int16_t* out_data) {
>+                                  size_t encoded_bytes,
>+                                  int16_t* out_data) {
>   int value;
>   int16_t audio_type;
>   clock_t clocks = clock();
>@@ -84,13 +88,13 @@ float OpusSpeedTest::DecodeABlock(const uint8_t* bit_stream,
> /* Test audio length in second. */
> constexpr size_t kDurationSec = 400;
> 
>-#define ADD_TEST(complexity) \
>-TEST_P(OpusSpeedTest, OpusSetComplexityTest##complexity) { \
>-  /* Set complexity. */ \
>-  printf("Setting complexity to %d ...\n", complexity); \
>-  EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_encoder_, complexity)); \
>-  EncodeDecode(kDurationSec); \
>-}
>+#define ADD_TEST(complexity)                                           \
>+  TEST_P(OpusSpeedTest, OpusSetComplexityTest##complexity) {           \
>+    /* Set complexity. */                                              \
>+    printf("Setting complexity to %d ...\n", complexity);              \
>+    EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_encoder_, complexity)); \
>+    EncodeDecode(kDurationSec);                                        \
>+  }
> 
> ADD_TEST(10);
> ADD_TEST(9);
>@@ -136,7 +140,6 @@ const coding_param param_set[] = {
>                     string("pcm"),
>                     true)};
> 
>-INSTANTIATE_TEST_CASE_P(AllTest, OpusSpeedTest,
>-                        ::testing::ValuesIn(param_set));
>+INSTANTIATE_TEST_CASE_P(AllTest, OpusSpeedTest, ::testing::ValuesIn(param_set));
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc
>index 12a15853c94..034f8cd2037 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc
>@@ -58,9 +58,12 @@ class OpusTest : public TestWithParam<::testing::tuple<int, int>> {
>                    int16_t* audio_type);
> 
>   void SetMaxPlaybackRate(WebRtcOpusEncInst* encoder,
>-                          opus_int32 expect, int32_t set);
>+                          opus_int32 expect,
>+                          int32_t set);
> 
>-  void CheckAudioBounded(const int16_t* audio, size_t samples, size_t channels,
>+  void CheckAudioBounded(const int16_t* audio,
>+                         size_t samples,
>+                         size_t channels,
>                          uint16_t bound) const;
> 
>   WebRtcOpusEncInst* opus_encoder_;
>@@ -78,15 +81,15 @@ OpusTest::OpusTest()
>       opus_decoder_(NULL),
>       encoded_bytes_(0),
>       channels_(static_cast<size_t>(::testing::get<0>(GetParam()))),
>-      application_(::testing::get<1>(GetParam())) {
>-}
>+      application_(::testing::get<1>(GetParam())) {}
> 
>-void OpusTest::PrepareSpeechData(size_t channel, int block_length_ms,
>+void OpusTest::PrepareSpeechData(size_t channel,
>+                                 int block_length_ms,
>                                  int loop_length_ms) {
>-  const std::string file_name =
>-        webrtc::test::ResourcePath((channel == 1) ?
>-            "audio_coding/testfile32kHz" :
>-            "audio_coding/teststereo32kHz", "pcm");
>+  const std::string file_name = webrtc::test::ResourcePath(
>+      (channel == 1) ? "audio_coding/testfile32kHz"
>+                     : "audio_coding/teststereo32kHz",
>+      "pcm");
>   if (loop_length_ms < block_length_ms) {
>     loop_length_ms = block_length_ms;
>   }
>@@ -100,13 +103,14 @@ void OpusTest::SetMaxPlaybackRate(WebRtcOpusEncInst* encoder,
>                                   int32_t set) {
>   opus_int32 bandwidth;
>   EXPECT_EQ(0, WebRtcOpus_SetMaxPlaybackRate(opus_encoder_, set));
>-  opus_encoder_ctl(opus_encoder_->encoder,
>-                   OPUS_GET_MAX_BANDWIDTH(&bandwidth));
>+  opus_encoder_ctl(opus_encoder_->encoder, OPUS_GET_MAX_BANDWIDTH(&bandwidth));
>   EXPECT_EQ(expect, bandwidth);
> }
> 
>-void OpusTest::CheckAudioBounded(const int16_t* audio, size_t samples,
>-                                 size_t channels, uint16_t bound) const {
>+void OpusTest::CheckAudioBounded(const int16_t* audio,
>+                                 size_t samples,
>+                                 size_t channels,
>+                                 uint16_t bound) const {
>   for (size_t i = 0; i < samples; ++i) {
>     for (size_t c = 0; c < channels; ++c) {
>       ASSERT_GE(audio[i * channels + c], -bound);
>@@ -120,16 +124,15 @@ int OpusTest::EncodeDecode(WebRtcOpusEncInst* encoder,
>                            WebRtcOpusDecInst* decoder,
>                            int16_t* output_audio,
>                            int16_t* audio_type) {
>-  int encoded_bytes_int = WebRtcOpus_Encode(
>-      encoder, input_audio.data(),
>-      rtc::CheckedDivExact(input_audio.size(), channels_),
>-      kMaxBytes, bitstream_);
>+  int encoded_bytes_int =
>+      WebRtcOpus_Encode(encoder, input_audio.data(),
>+                        rtc::CheckedDivExact(input_audio.size(), channels_),
>+                        kMaxBytes, bitstream_);
>   EXPECT_GE(encoded_bytes_int, 0);
>   encoded_bytes_ = static_cast<size_t>(encoded_bytes_int);
>   int est_len = WebRtcOpus_DurationEst(decoder, bitstream_, encoded_bytes_);
>-  int act_len = WebRtcOpus_Decode(decoder, bitstream_,
>-                                  encoded_bytes_, output_audio,
>-                                  audio_type);
>+  int act_len = WebRtcOpus_Decode(decoder, bitstream_, encoded_bytes_,
>+                                  output_audio, audio_type);
>   EXPECT_EQ(est_len, act_len);
>   return act_len;
> }
>@@ -141,30 +144,28 @@ void OpusTest::TestDtxEffect(bool dtx, int block_length_ms) {
>   const size_t samples = kOpusRateKhz * block_length_ms;
> 
>   // Create encoder memory.
>-  EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_,
>-                                        channels_,
>-                                        application_));
>+  EXPECT_EQ(0,
>+            WebRtcOpus_EncoderCreate(&opus_encoder_, channels_, application_));
>   EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_, channels_));
> 
>   // Set bitrate.
>-  EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_,
>-                                     channels_ == 1 ? 32000 : 64000));
>+  EXPECT_EQ(
>+      0, WebRtcOpus_SetBitRate(opus_encoder_, channels_ == 1 ? 32000 : 64000));
> 
>   // Set input audio as silence.
>   std::vector<int16_t> silence(samples * channels_, 0);
> 
>   // Setting DTX.
>-  EXPECT_EQ(0, dtx ? WebRtcOpus_EnableDtx(opus_encoder_) :
>-      WebRtcOpus_DisableDtx(opus_encoder_));
>+  EXPECT_EQ(0, dtx ? WebRtcOpus_EnableDtx(opus_encoder_)
>+                   : WebRtcOpus_DisableDtx(opus_encoder_));
> 
>   int16_t audio_type;
>   int16_t* output_data_decode = new int16_t[samples * channels_];
> 
>   for (int i = 0; i < 100; ++i) {
>-    EXPECT_EQ(samples,
>-              static_cast<size_t>(EncodeDecode(
>-                  opus_encoder_, speech_data_.GetNextBlock(), opus_decoder_,
>-                  output_data_decode, &audio_type)));
>+    EXPECT_EQ(samples, static_cast<size_t>(EncodeDecode(
>+                           opus_encoder_, speech_data_.GetNextBlock(),
>+                           opus_decoder_, output_data_decode, &audio_type)));
>     // If not DTX, it should never enter DTX mode. If DTX, we do not care since
>     // whether it enters DTX depends on the signal type.
>     if (!dtx) {
>@@ -178,10 +179,9 @@ void OpusTest::TestDtxEffect(bool dtx, int block_length_ms) {
>   // We input some silent segments. In DTX mode, the encoder will stop sending.
>   // However, DTX may happen after a while.
>   for (int i = 0; i < 30; ++i) {
>-    EXPECT_EQ(samples,
>-              static_cast<size_t>(EncodeDecode(
>-                  opus_encoder_, silence, opus_decoder_, output_data_decode,
>-                  &audio_type)));
>+    EXPECT_EQ(samples, static_cast<size_t>(
>+                           EncodeDecode(opus_encoder_, silence, opus_decoder_,
>+                                        output_data_decode, &audio_type)));
>     if (!dtx) {
>       EXPECT_GT(encoded_bytes_, 1U);
>       EXPECT_EQ(0, opus_encoder_->in_dtx_mode);
>@@ -227,10 +227,9 @@ void OpusTest::TestDtxEffect(bool dtx, int block_length_ms) {
>     int i = 0;
>     for (; i < max_dtx_frames; ++i) {
>       time += block_length_ms;
>-      EXPECT_EQ(samples,
>-                static_cast<size_t>(EncodeDecode(
>-                    opus_encoder_, silence, opus_decoder_, output_data_decode,
>-                    &audio_type)));
>+      EXPECT_EQ(samples, static_cast<size_t>(
>+                             EncodeDecode(opus_encoder_, silence, opus_decoder_,
>+                                          output_data_decode, &audio_type)));
>       if (dtx) {
>         if (encoded_bytes_ > 1)
>           break;
>@@ -263,10 +262,9 @@ void OpusTest::TestDtxEffect(bool dtx, int block_length_ms) {
> 
>     // Enters DTX again immediately.
>     time += block_length_ms;
>-    EXPECT_EQ(samples,
>-              static_cast<size_t>(EncodeDecode(
>-                  opus_encoder_, silence, opus_decoder_, output_data_decode,
>-                  &audio_type)));
>+    EXPECT_EQ(samples, static_cast<size_t>(
>+                           EncodeDecode(opus_encoder_, silence, opus_decoder_,
>+                                        output_data_decode, &audio_type)));
>     if (dtx) {
>       EXPECT_EQ(1U, encoded_bytes_);  // Send 1 byte.
>       EXPECT_EQ(1, opus_encoder_->in_dtx_mode);
>@@ -287,10 +285,9 @@ void OpusTest::TestDtxEffect(bool dtx, int block_length_ms) {
>   silence[0] = 10000;
>   if (dtx) {
>     // Verify that encoder/decoder can jump out from DTX mode.
>-    EXPECT_EQ(samples,
>-              static_cast<size_t>(EncodeDecode(
>-                  opus_encoder_, silence, opus_decoder_, output_data_decode,
>-                  &audio_type)));
>+    EXPECT_EQ(samples, static_cast<size_t>(
>+                           EncodeDecode(opus_encoder_, silence, opus_decoder_,
>+                                        output_data_decode, &audio_type)));
>     EXPECT_GT(encoded_bytes_, 1U);
>     EXPECT_EQ(0, opus_encoder_->in_dtx_mode);
>     EXPECT_EQ(0, opus_decoder_->in_dtx_mode);
>@@ -375,9 +372,8 @@ TEST(OpusTest, OpusFreeFail) {
> 
> // Test normal Create and Free.
> TEST_P(OpusTest, OpusCreateFree) {
>-  EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_,
>-                                        channels_,
>-                                        application_));
>+  EXPECT_EQ(0,
>+            WebRtcOpus_EncoderCreate(&opus_encoder_, channels_, application_));
>   EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_, channels_));
>   EXPECT_TRUE(opus_encoder_ != NULL);
>   EXPECT_TRUE(opus_decoder_ != NULL);
>@@ -390,23 +386,20 @@ TEST_P(OpusTest, OpusEncodeDecode) {
>   PrepareSpeechData(channels_, 20, 20);
> 
>   // Create encoder memory.
>-  EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_,
>-                                        channels_,
>-                                        application_));
>-  EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_,
>-                                        channels_));
>+  EXPECT_EQ(0,
>+            WebRtcOpus_EncoderCreate(&opus_encoder_, channels_, application_));
>+  EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_, channels_));
> 
>   // Set bitrate.
>-  EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_,
>-                                     channels_ == 1 ? 32000 : 64000));
>+  EXPECT_EQ(
>+      0, WebRtcOpus_SetBitRate(opus_encoder_, channels_ == 1 ? 32000 : 64000));
> 
>   // Check number of channels for decoder.
>   EXPECT_EQ(channels_, WebRtcOpus_DecoderChannels(opus_decoder_));
> 
>   // Check application mode.
>   opus_int32 app;
>-  opus_encoder_ctl(opus_encoder_->encoder,
>-                   OPUS_GET_APPLICATION(&app));
>+  opus_encoder_ctl(opus_encoder_->encoder, OPUS_GET_APPLICATION(&app));
>   EXPECT_EQ(application_ == 0 ? OPUS_APPLICATION_VOIP : OPUS_APPLICATION_AUDIO,
>             app);
> 
>@@ -429,9 +422,8 @@ TEST_P(OpusTest, OpusSetBitRate) {
>   EXPECT_EQ(-1, WebRtcOpus_SetBitRate(opus_encoder_, 60000));
> 
>   // Create encoder memory, try with different bitrates.
>-  EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_,
>-                                        channels_,
>-                                        application_));
>+  EXPECT_EQ(0,
>+            WebRtcOpus_EncoderCreate(&opus_encoder_, channels_, application_));
>   EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_, 30000));
>   EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_, 60000));
>   EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_, 300000));
>@@ -446,9 +438,8 @@ TEST_P(OpusTest, OpusSetComplexity) {
>   EXPECT_EQ(-1, WebRtcOpus_SetComplexity(opus_encoder_, 9));
> 
>   // Create encoder memory, try with different complexities.
>-  EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_,
>-                                        channels_,
>-                                        application_));
>+  EXPECT_EQ(0,
>+            WebRtcOpus_EncoderCreate(&opus_encoder_, channels_, application_));
> 
>   EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_encoder_, 0));
>   EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_encoder_, 10));
>@@ -524,9 +515,8 @@ TEST_P(OpusTest, OpusDecodeInit) {
>   PrepareSpeechData(channels_, 20, 20);
> 
>   // Create encoder memory.
>-  EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_,
>-                                        channels_,
>-                                        application_));
>+  EXPECT_EQ(0,
>+            WebRtcOpus_EncoderCreate(&opus_encoder_, channels_, application_));
>   EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_, channels_));
> 
>   // Encode & decode.
>@@ -540,9 +530,9 @@ TEST_P(OpusTest, OpusDecodeInit) {
>   WebRtcOpus_DecoderInit(opus_decoder_);
> 
>   EXPECT_EQ(kOpus20msFrameSamples,
>-            static_cast<size_t>(WebRtcOpus_Decode(
>-                opus_decoder_, bitstream_, encoded_bytes_, output_data_decode,
>-                &audio_type)));
>+            static_cast<size_t>(
>+                WebRtcOpus_Decode(opus_decoder_, bitstream_, encoded_bytes_,
>+                                  output_data_decode, &audio_type)));
> 
>   // Free memory.
>   delete[] output_data_decode;
>@@ -556,9 +546,8 @@ TEST_P(OpusTest, OpusEnableDisableFec) {
>   EXPECT_EQ(-1, WebRtcOpus_DisableFec(opus_encoder_));
> 
>   // Create encoder memory.
>-  EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_,
>-                                        channels_,
>-                                        application_));
>+  EXPECT_EQ(0,
>+            WebRtcOpus_EncoderCreate(&opus_encoder_, channels_, application_));
> 
>   EXPECT_EQ(0, WebRtcOpus_EnableFec(opus_encoder_));
>   EXPECT_EQ(0, WebRtcOpus_DisableFec(opus_encoder_));
>@@ -573,30 +562,25 @@ TEST_P(OpusTest, OpusEnableDisableDtx) {
>   EXPECT_EQ(-1, WebRtcOpus_DisableDtx(opus_encoder_));
> 
>   // Create encoder memory.
>-  EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_,
>-                                        channels_,
>-                                        application_));
>+  EXPECT_EQ(0,
>+            WebRtcOpus_EncoderCreate(&opus_encoder_, channels_, application_));
> 
>   opus_int32 dtx;
> 
>   // DTX is off by default.
>-  opus_encoder_ctl(opus_encoder_->encoder,
>-                   OPUS_GET_DTX(&dtx));
>+  opus_encoder_ctl(opus_encoder_->encoder, OPUS_GET_DTX(&dtx));
>   EXPECT_EQ(0, dtx);
> 
>   // Test to enable DTX.
>   EXPECT_EQ(0, WebRtcOpus_EnableDtx(opus_encoder_));
>-  opus_encoder_ctl(opus_encoder_->encoder,
>-                   OPUS_GET_DTX(&dtx));
>+  opus_encoder_ctl(opus_encoder_->encoder, OPUS_GET_DTX(&dtx));
>   EXPECT_EQ(1, dtx);
> 
>   // Test to disable DTX.
>   EXPECT_EQ(0, WebRtcOpus_DisableDtx(opus_encoder_));
>-  opus_encoder_ctl(opus_encoder_->encoder,
>-                   OPUS_GET_DTX(&dtx));
>+  opus_encoder_ctl(opus_encoder_->encoder, OPUS_GET_DTX(&dtx));
>   EXPECT_EQ(0, dtx);
> 
>-
>   // Free memory.
>   EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
> }
>@@ -630,9 +614,8 @@ TEST_P(OpusTest, OpusSetPacketLossRate) {
>   EXPECT_EQ(-1, WebRtcOpus_SetPacketLossRate(opus_encoder_, 50));
> 
>   // Create encoder memory.
>-  EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_,
>-                                        channels_,
>-                                        application_));
>+  EXPECT_EQ(0,
>+            WebRtcOpus_EncoderCreate(&opus_encoder_, channels_, application_));
> 
>   EXPECT_EQ(0, WebRtcOpus_SetPacketLossRate(opus_encoder_, 50));
>   EXPECT_EQ(-1, WebRtcOpus_SetPacketLossRate(opus_encoder_, -1));
>@@ -647,9 +630,8 @@ TEST_P(OpusTest, OpusSetMaxPlaybackRate) {
>   EXPECT_EQ(-1, WebRtcOpus_SetMaxPlaybackRate(opus_encoder_, 20000));
> 
>   // Create encoder memory.
>-  EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_,
>-                                        channels_,
>-                                        application_));
>+  EXPECT_EQ(0,
>+            WebRtcOpus_EncoderCreate(&opus_encoder_, channels_, application_));
> 
>   SetMaxPlaybackRate(opus_encoder_, OPUS_BANDWIDTH_FULLBAND, 48000);
>   SetMaxPlaybackRate(opus_encoder_, OPUS_BANDWIDTH_FULLBAND, 24001);
>@@ -671,14 +653,13 @@ TEST_P(OpusTest, OpusDecodePlc) {
>   PrepareSpeechData(channels_, 20, 20);
> 
>   // Create encoder memory.
>-  EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_,
>-                                        channels_,
>-                                        application_));
>+  EXPECT_EQ(0,
>+            WebRtcOpus_EncoderCreate(&opus_encoder_, channels_, application_));
>   EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_, channels_));
> 
>   // Set bitrate.
>-  EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_,
>-                                     channels_== 1 ? 32000 : 64000));
>+  EXPECT_EQ(
>+      0, WebRtcOpus_SetBitRate(opus_encoder_, channels_ == 1 ? 32000 : 64000));
> 
>   // Check number of channels for decoder.
>   EXPECT_EQ(channels_, WebRtcOpus_DecoderChannels(opus_decoder_));
>@@ -693,9 +674,8 @@ TEST_P(OpusTest, OpusDecodePlc) {
> 
>   // Call decoder PLC.
>   int16_t* plc_buffer = new int16_t[kOpus20msFrameSamples * channels_];
>-  EXPECT_EQ(kOpus20msFrameSamples,
>-            static_cast<size_t>(WebRtcOpus_DecodePlc(
>-                opus_decoder_, plc_buffer, 1)));
>+  EXPECT_EQ(kOpus20msFrameSamples, static_cast<size_t>(WebRtcOpus_DecodePlc(
>+                                       opus_decoder_, plc_buffer, 1)));
> 
>   // Free memory.
>   delete[] plc_buffer;
>@@ -709,34 +689,33 @@ TEST_P(OpusTest, OpusDurationEstimation) {
>   PrepareSpeechData(channels_, 20, 20);
> 
>   // Create.
>-  EXPECT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_,
>-                                        channels_,
>-                                        application_));
>+  EXPECT_EQ(0,
>+            WebRtcOpus_EncoderCreate(&opus_encoder_, channels_, application_));
>   EXPECT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_, channels_));
> 
>   // 10 ms. We use only first 10 ms of a 20 ms block.
>   auto speech_block = speech_data_.GetNextBlock();
>   int encoded_bytes_int = WebRtcOpus_Encode(
>       opus_encoder_, speech_block.data(),
>-      rtc::CheckedDivExact(speech_block.size(), 2 * channels_),
>-      kMaxBytes, bitstream_);
>+      rtc::CheckedDivExact(speech_block.size(), 2 * channels_), kMaxBytes,
>+      bitstream_);
>   EXPECT_GE(encoded_bytes_int, 0);
>-  EXPECT_EQ(kOpus10msFrameSamples,
>-            static_cast<size_t>(WebRtcOpus_DurationEst(
>-                opus_decoder_, bitstream_,
>-                static_cast<size_t>(encoded_bytes_int))));
>+  EXPECT_EQ(
>+      kOpus10msFrameSamples,
>+      static_cast<size_t>(WebRtcOpus_DurationEst(
>+          opus_decoder_, bitstream_, static_cast<size_t>(encoded_bytes_int))));
> 
>   // 20 ms
>   speech_block = speech_data_.GetNextBlock();
>-  encoded_bytes_int = WebRtcOpus_Encode(
>-      opus_encoder_, speech_block.data(),
>-      rtc::CheckedDivExact(speech_block.size(), channels_),
>-      kMaxBytes, bitstream_);
>+  encoded_bytes_int =
>+      WebRtcOpus_Encode(opus_encoder_, speech_block.data(),
>+                        rtc::CheckedDivExact(speech_block.size(), channels_),
>+                        kMaxBytes, bitstream_);
>   EXPECT_GE(encoded_bytes_int, 0);
>-  EXPECT_EQ(kOpus20msFrameSamples,
>-            static_cast<size_t>(WebRtcOpus_DurationEst(
>-                opus_decoder_, bitstream_,
>-                static_cast<size_t>(encoded_bytes_int))));
>+  EXPECT_EQ(
>+      kOpus20msFrameSamples,
>+      static_cast<size_t>(WebRtcOpus_DurationEst(
>+          opus_decoder_, bitstream_, static_cast<size_t>(encoded_bytes_int))));
> 
>   // Free memory.
>   EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
>@@ -749,15 +728,13 @@ TEST_P(OpusTest, OpusDecodeRepacketized) {
>   PrepareSpeechData(channels_, 20, 20 * kPackets);
> 
>   // Create encoder memory.
>-  ASSERT_EQ(0, WebRtcOpus_EncoderCreate(&opus_encoder_,
>-                                        channels_,
>-                                        application_));
>-  ASSERT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_,
>-                                        channels_));
>+  ASSERT_EQ(0,
>+            WebRtcOpus_EncoderCreate(&opus_encoder_, channels_, application_));
>+  ASSERT_EQ(0, WebRtcOpus_DecoderCreate(&opus_decoder_, channels_));
> 
>   // Set bitrate.
>-  EXPECT_EQ(0, WebRtcOpus_SetBitRate(opus_encoder_,
>-                                     channels_ == 1 ? 32000 : 64000));
>+  EXPECT_EQ(
>+      0, WebRtcOpus_SetBitRate(opus_encoder_, channels_ == 1 ? 32000 : 64000));
> 
>   // Check number of channels for decoder.
>   EXPECT_EQ(channels_, WebRtcOpus_DecoderChannels(opus_decoder_));
>@@ -776,9 +753,9 @@ TEST_P(OpusTest, OpusDecodeRepacketized) {
>         WebRtcOpus_Encode(opus_encoder_, speech_block.data(),
>                           rtc::CheckedDivExact(speech_block.size(), channels_),
>                           kMaxBytes, bitstream_);
>-    if (opus_repacketizer_cat(
>-            rp, bitstream_,
>-            rtc::checked_cast<opus_int32>(encoded_bytes_)) == OPUS_OK) {
>+    if (opus_repacketizer_cat(rp, bitstream_,
>+                              rtc::checked_cast<opus_int32>(encoded_bytes_)) ==
>+        OPUS_OK) {
>       ++num_packets;
>       if (num_packets == kPackets) {
>         break;
>@@ -798,9 +775,9 @@ TEST_P(OpusTest, OpusDecodeRepacketized) {
>                 opus_decoder_, bitstream_, encoded_bytes_)));
> 
>   EXPECT_EQ(kOpus20msFrameSamples * kPackets,
>-            static_cast<size_t>(WebRtcOpus_Decode(
>-                opus_decoder_, bitstream_, encoded_bytes_,
>-                output_data_decode.get(), &audio_type)));
>+            static_cast<size_t>(
>+                WebRtcOpus_Decode(opus_decoder_, bitstream_, encoded_bytes_,
>+                                  output_data_decode.get(), &audio_type)));
> 
>   // Free memory.
>   opus_repacketizer_destroy(rp);
>@@ -812,5 +789,4 @@ INSTANTIATE_TEST_CASE_P(VariousMode,
>                         OpusTest,
>                         Combine(Values(1, 2), Values(0, 1)));
> 
>-
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.c
>index 54aa84d0e98..2f6dce5f416 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.c
>@@ -8,9 +8,7 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "pcm16b.h"
>-
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
> 
> size_t WebRtcPcm16b_Encode(const int16_t* speech,
>                            size_t len,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h
>index 041701a4b58..75d1efda3b0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/pcm16b/pcm16b.h
>@@ -15,8 +15,7 @@
>  */
> 
> #include <stddef.h>
>-
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include <stdint.h>
> 
> #ifdef __cplusplus
> extern "C" {
>@@ -38,9 +37,7 @@ extern "C" {
>  *                                Always equal to twice the len input parameter.
>  */
> 
>-size_t WebRtcPcm16b_Encode(const int16_t* speech,
>-                           size_t len,
>-                           uint8_t* encoded);
>+size_t WebRtcPcm16b_Encode(const int16_t* speech, size_t len, uint8_t* encoded);
> 
> /****************************************************************************
>  * WebRtcPcm16b_Decode(...)
>@@ -57,9 +54,7 @@ size_t WebRtcPcm16b_Encode(const int16_t* speech,
>  * Returned value               : Samples in speech
>  */
> 
>-size_t WebRtcPcm16b_Decode(const uint8_t* encoded,
>-                           size_t len,
>-                           int16_t* speech);
>+size_t WebRtcPcm16b_Decode(const uint8_t* encoded, size_t len, int16_t* speech);
> 
> #ifdef __cplusplus
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
>index 4b9df6e0e6c..2601f26e44f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.cc
>@@ -58,10 +58,8 @@ AudioEncoder::EncodedInfo AudioEncoderCopyRed::EncodeImpl(
>     uint32_t rtp_timestamp,
>     rtc::ArrayView<const int16_t> audio,
>     rtc::Buffer* encoded) {
>-
>   const size_t primary_offset = encoded->size();
>-  EncodedInfo info =
>-      speech_encoder_->Encode(rtp_timestamp, audio, encoded);
>+  EncodedInfo info = speech_encoder_->Encode(rtp_timestamp, audio, encoded);
> 
>   RTC_CHECK(info.redundant.empty()) << "Cannot use nested redundant encoders.";
>   RTC_DCHECK_EQ(encoded->size() - primary_offset, info.encoded_bytes);
>@@ -134,7 +132,7 @@ void AudioEncoderCopyRed::OnReceivedUplinkRecoverablePacketLossFraction(
> 
> void AudioEncoderCopyRed::OnReceivedUplinkBandwidth(
>     int target_audio_bitrate_bps,
>-    rtc::Optional<int64_t> bwe_period_ms) {
>+    absl::optional<int64_t> bwe_period_ms) {
>   speech_encoder_->OnReceivedUplinkBandwidth(target_audio_bitrate_bps,
>                                              bwe_period_ms);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
>index e625c50d962..492ee3afb47 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
>@@ -57,7 +57,7 @@ class AudioEncoderCopyRed final : public AudioEncoder {
>       float uplink_recoverable_packet_loss_fraction) override;
>   void OnReceivedUplinkBandwidth(
>       int target_audio_bitrate_bps,
>-      rtc::Optional<int64_t> bwe_period_ms) override;
>+      absl::optional<int64_t> bwe_period_ms) override;
> 
>  protected:
>   EncodedInfo EncodeImpl(uint32_t rtp_timestamp,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
>index 79d205e256f..0f5a811badf 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
>@@ -48,10 +48,7 @@ class AudioEncoderCopyRedTest : public ::testing::Test {
>         .WillRepeatedly(Return(sample_rate_hz_));
>   }
> 
>-  void TearDown() override {
>-    EXPECT_CALL(*mock_encoder_, Die()).Times(1);
>-    red_.reset();
>-  }
>+  void TearDown() override { red_.reset(); }
> 
>   void Encode() {
>     ASSERT_TRUE(red_.get() != NULL);
>@@ -74,8 +71,7 @@ class AudioEncoderCopyRedTest : public ::testing::Test {
>   const int red_payload_type_;
> };
> 
>-TEST_F(AudioEncoderCopyRedTest, CreateAndDestroy) {
>-}
>+TEST_F(AudioEncoderCopyRedTest, CreateAndDestroy) {}
> 
> TEST_F(AudioEncoderCopyRedTest, CheckSampleRatePropagation) {
>   EXPECT_CALL(*mock_encoder_, SampleRateHz()).WillOnce(Return(17));
>@@ -100,8 +96,8 @@ TEST_F(AudioEncoderCopyRedTest, CheckMaxFrameSizePropagation) {
> 
> TEST_F(AudioEncoderCopyRedTest, CheckTargetAudioBitratePropagation) {
>   EXPECT_CALL(*mock_encoder_,
>-              OnReceivedUplinkBandwidth(4711, rtc::Optional<int64_t>()));
>-  red_->OnReceivedUplinkBandwidth(4711, rtc::nullopt);
>+              OnReceivedUplinkBandwidth(4711, absl::optional<int64_t>()));
>+  red_->OnReceivedUplinkBandwidth(4711, absl::nullopt);
> }
> 
> TEST_F(AudioEncoderCopyRedTest, CheckPacketLossFractionPropagation) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc
>index d3749c142e3..c539152212e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc
>@@ -34,8 +34,7 @@ AudioCodecSpeedTest::AudioCodecSpeedTest(int block_duration_ms,
>       encoded_bytes_(0),
>       encoding_time_ms_(0.0),
>       decoding_time_ms_(0.0),
>-      out_file_(NULL) {
>-}
>+      out_file_(NULL) {}
> 
> void AudioCodecSpeedTest::SetUp() {
>   channels_ = get<0>(GetParam());
>@@ -52,8 +51,8 @@ void AudioCodecSpeedTest::SetUp() {
>   rewind(fp);
> 
>   // Allocate memory to contain the whole file.
>-  in_data_.reset(new int16_t[loop_length_samples_ +
>-      input_length_sample_ * channels_]);
>+  in_data_.reset(
>+      new int16_t[loop_length_samples_ + input_length_sample_ * channels_]);
> 
>   data_pointer_ = 0;
> 
>@@ -111,11 +110,11 @@ void AudioCodecSpeedTest::EncodeDecode(size_t audio_duration_sec) {
>     time_ms = DecodeABlock(&bit_stream_[0], encoded_bytes_, &out_data_[0]);
>     decoding_time_ms_ += time_ms;
>     if (save_out_data_) {
>-      fwrite(&out_data_[0], sizeof(int16_t),
>-             output_length_sample_ * channels_, out_file_);
>+      fwrite(&out_data_[0], sizeof(int16_t), output_length_sample_ * channels_,
>+             out_file_);
>     }
>     data_pointer_ = (data_pointer_ + input_length_sample_ * channels_) %
>-        loop_length_samples_;
>+                    loop_length_samples_;
>     time_now_ms += block_duration_ms_;
>   }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h
>index 9e616e789b0..c626bfa0ed9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h
>@@ -15,7 +15,6 @@
> #include <string>
> 
> #include "test/gtest.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -36,15 +35,18 @@ class AudioCodecSpeedTest : public testing::TestWithParam<coding_param> {
>   // 2. save the bit stream to |bit_stream| of |max_bytes| bytes in size,
>   // 3. assign |encoded_bytes| with the length of the bit stream (in bytes),
>   // 4. return the cost of time (in millisecond) spent on actual encoding.
>-  virtual float EncodeABlock(int16_t* in_data, uint8_t* bit_stream,
>-                             size_t max_bytes, size_t* encoded_bytes) = 0;
>+  virtual float EncodeABlock(int16_t* in_data,
>+                             uint8_t* bit_stream,
>+                             size_t max_bytes,
>+                             size_t* encoded_bytes) = 0;
> 
>   // DecodeABlock(...) does the following:
>   // 1. decodes the bit stream in |bit_stream| with a length of |encoded_bytes|
>   // (in bytes),
>   // 2. save the decoded audio in |out_data|,
>   // 3. return the cost of time (in millisecond) spent on actual decoding.
>-  virtual float DecodeABlock(const uint8_t* bit_stream, size_t encoded_bytes,
>+  virtual float DecodeABlock(const uint8_t* bit_stream,
>+                             size_t encoded_bytes,
>                              int16_t* out_data) = 0;
> 
>   // Encoding and decode an audio of |audio_duration| (in seconds) and
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/include/audio_coding_module.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/include/audio_coding_module.h
>index 12c98eeec8f..41745abc7b3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/include/audio_coding_module.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/include/audio_coding_module.h
>@@ -15,17 +15,15 @@
> #include <string>
> #include <vector>
> 
>+#include "absl/types/optional.h"
> #include "api/audio_codecs/audio_decoder_factory.h"
> #include "api/audio_codecs/audio_encoder.h"
>-#include "api/optional.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/audio_coding/include/audio_coding_module_typedefs.h"
> #include "modules/audio_coding/neteq/include/neteq.h"
>-#include "modules/include/module.h"
> #include "rtc_base/deprecation.h"
> #include "rtc_base/function_view.h"
> #include "system_wrappers/include/clock.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -66,7 +64,8 @@ class AudioCodingModule {
> 
>  public:
>   struct Config {
>-    Config();
>+    explicit Config(
>+        rtc::scoped_refptr<AudioDecoderFactory> decoder_factory = nullptr);
>     Config(const Config&);
>     ~Config();
> 
>@@ -75,17 +74,6 @@ class AudioCodingModule {
>     rtc::scoped_refptr<AudioDecoderFactory> decoder_factory;
>   };
> 
>-  ///////////////////////////////////////////////////////////////////////////
>-  // Creation and destruction of a ACM.
>-  //
>-  // The second method is used for testing where a simulated clock can be
>-  // injected into ACM. ACM will take the ownership of the object clock and
>-  // delete it when destroyed.
>-  //
>-  // TODO(solenberg): Remove once downstream projects are updated.
>-  RTC_DEPRECATED static AudioCodingModule* Create(int id);
>-  static AudioCodingModule* Create();
>-  static AudioCodingModule* Create(Clock* clock);
>   static AudioCodingModule* Create(const Config& config);
>   virtual ~AudioCodingModule() = default;
> 
>@@ -138,8 +126,10 @@ class AudioCodingModule {
>   //   -1 if no codec matches the given parameters.
>   //    0 if succeeded.
>   //
>-  static int Codec(const char* payload_name, CodecInst* codec,
>-                   int sampling_freq_hz, size_t channels);
>+  static int Codec(const char* payload_name,
>+                   CodecInst* codec,
>+                   int sampling_freq_hz,
>+                   size_t channels);
> 
>   ///////////////////////////////////////////////////////////////////////////
>   // int32_t Codec()
>@@ -157,7 +147,8 @@ class AudioCodingModule {
>   //   if the codec is found, the index of the codec in the list,
>   //   -1 if the codec is not found.
>   //
>-  static int Codec(const char* payload_name, int sampling_freq_hz,
>+  static int Codec(const char* payload_name,
>+                   int sampling_freq_hz,
>                    size_t channels);
> 
>   ///////////////////////////////////////////////////////////////////////////
>@@ -239,7 +230,7 @@ class AudioCodingModule {
>   // Return value:
>   //   The send codec, or nothing if we don't have one
>   //
>-  virtual rtc::Optional<CodecInst> SendCodec() const = 0;
>+  virtual absl::optional<CodecInst> SendCodec() const = 0;
> 
>   ///////////////////////////////////////////////////////////////////////////
>   // int32_t SendFrequency()
>@@ -288,9 +279,7 @@ class AudioCodingModule {
>   //
>   // Input:
>   //   -audio_frame        : the input audio frame, containing raw audio
>-  //                         sampling frequency etc.,
>-  //                         c.f. module_common_types.h for definition of
>-  //                         AudioFrame.
>+  //                         sampling frequency etc.
>   //
>   // Return value:
>   //   >= 0   number of bytes encoded.
>@@ -411,8 +400,8 @@ class AudioCodingModule {
>   //    0 if succeeded.
>   //
>   virtual int32_t SetVAD(const bool enable_dtx = true,
>-                               const bool enable_vad = false,
>-                               const ACMVADMode vad_mode = VADNormal) = 0;
>+                         const bool enable_vad = false,
>+                         const ACMVADMode vad_mode = VADNormal) = 0;
> 
>   ///////////////////////////////////////////////////////////////////////////
>   // int32_t VAD()
>@@ -429,8 +418,9 @@ class AudioCodingModule {
>   //   -1 if fails to retrieve the setting of DTX/VAD,
>   //    0 if succeeded.
>   //
>-  virtual int32_t VAD(bool* dtx_enabled, bool* vad_enabled,
>-                            ACMVADMode* vad_mode) const = 0;
>+  virtual int32_t VAD(bool* dtx_enabled,
>+                      bool* vad_enabled,
>+                      ACMVADMode* vad_mode) const = 0;
> 
>   ///////////////////////////////////////////////////////////////////////////
>   // int32_t RegisterVADCallback()
>@@ -540,8 +530,7 @@ class AudioCodingModule {
>   //   -1 if fails to unregister.
>   //    0 if the given codec is successfully unregistered.
>   //
>-  virtual int UnregisterReceiveCodec(
>-      uint8_t payload_type) = 0;
>+  virtual int UnregisterReceiveCodec(uint8_t payload_type) = 0;
> 
>   ///////////////////////////////////////////////////////////////////////////
>   // int32_t ReceiveCodec()
>@@ -559,7 +548,7 @@ class AudioCodingModule {
>   virtual int32_t ReceiveCodec(CodecInst* curr_receive_codec) const = 0;
> 
>   ///////////////////////////////////////////////////////////////////////////
>-  // rtc::Optional<SdpAudioFormat> ReceiveFormat()
>+  // absl::optional<SdpAudioFormat> ReceiveFormat()
>   // Get the format associated with last received payload.
>   //
>   // Return value:
>@@ -567,7 +556,7 @@ class AudioCodingModule {
>   //    received payload.
>   //    An empty Optional if no payload has yet been received.
>   //
>-  virtual rtc::Optional<SdpAudioFormat> ReceiveFormat() const = 0;
>+  virtual absl::optional<SdpAudioFormat> ReceiveFormat() const = 0;
> 
>   ///////////////////////////////////////////////////////////////////////////
>   // int32_t IncomingPacket()
>@@ -644,7 +633,7 @@ class AudioCodingModule {
>   // the latest audio obtained by calling PlayoutData10ms(), or empty if no
>   // valid timestamp is available.
>   //
>-  virtual rtc::Optional<uint32_t> PlayoutTimestamp() = 0;
>+  virtual absl::optional<uint32_t> PlayoutTimestamp() = 0;
> 
>   ///////////////////////////////////////////////////////////////////////////
>   // int FilteredCurrentDelayMs()
>@@ -673,9 +662,7 @@ class AudioCodingModule {
>   //
>   // Output:
>   //   -audio_frame        : output audio frame which contains raw audio data
>-  //                         and other relevant parameters, c.f.
>-  //                         module_common_types.h for the definition of
>-  //                         AudioFrame.
>+  //                         and other relevant parameters.
>   //   -muted              : if true, the sample data in audio_frame is not
>   //                         populated, and must be interpreted as all zero.
>   //
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h
>index ad71ef1f970..cd4351b07ff 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h
>@@ -13,9 +13,6 @@
> 
> #include <map>
> 
>-#include "modules/include/module_common_types.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>-
> namespace webrtc {
> 
> ///////////////////////////////////////////////////////////////////////////
>@@ -42,8 +39,8 @@ enum ACMVADMode {
> // kAudio             : optimized for non-voice signals like music.
> //
> enum OpusApplicationMode {
>- kVoip = 0,
>- kAudio = 1,
>+  kVoip = 0,
>+  kAudio = 1,
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/accelerate.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/accelerate.h
>index bf4f0f72669..b0bab325030 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/accelerate.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/accelerate.h
>@@ -16,7 +16,6 @@
> #include "modules/audio_coding/neteq/audio_multi_vector.h"
> #include "modules/audio_coding/neteq/time_stretch.h"
> #include "rtc_base/constructormagic.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -29,10 +28,10 @@ class BackgroundNoise;
> // Accelerate are implemented.
> class Accelerate : public TimeStretch {
>  public:
>-  Accelerate(int sample_rate_hz, size_t num_channels,
>+  Accelerate(int sample_rate_hz,
>+             size_t num_channels,
>              const BackgroundNoise& background_noise)
>-      : TimeStretch(sample_rate_hz, num_channels, background_noise) {
>-  }
>+      : TimeStretch(sample_rate_hz, num_channels, background_noise) {}
> 
>   // This method performs the actual Accelerate operation. The samples are
>   // read from |input|, of length |input_length| elements, and are written to
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
>index 3181d6fc1c6..1ad48350bbb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/audio_decoder_unittest.cc
>@@ -100,21 +100,21 @@ class AudioDecoderTest : public ::testing::Test {
>         payload_type_(17),
>         decoder_(NULL) {}
> 
>-  virtual ~AudioDecoderTest() {}
>+  ~AudioDecoderTest() override {}
> 
>-  virtual void SetUp() {
>+  void SetUp() override {
>     if (audio_encoder_)
>       codec_input_rate_hz_ = audio_encoder_->SampleRateHz();
>     // Create arrays.
>     ASSERT_GT(data_length_, 0u) << "The test must set data_length_ > 0";
>   }
> 
>-  virtual void TearDown() {
>+  void TearDown() override {
>     delete decoder_;
>     decoder_ = NULL;
>   }
> 
>-  virtual void InitEncoder() { }
>+  virtual void InitEncoder() {}
> 
>   // TODO(henrik.lundin) Change return type to size_t once most/all overriding
>   // implementations are gone.
>@@ -136,12 +136,13 @@ class AudioDecoderTest : public ::testing::Test {
>                                                  samples_per_10ms, channels_,
>                                                  interleaved_input.get());
> 
>-      encoded_info = audio_encoder_->Encode(
>-          0, rtc::ArrayView<const int16_t>(interleaved_input.get(),
>-                                           audio_encoder_->NumChannels() *
>-                                               audio_encoder_->SampleRateHz() /
>-                                               100),
>-          output);
>+      encoded_info =
>+          audio_encoder_->Encode(0,
>+                                 rtc::ArrayView<const int16_t>(
>+                                     interleaved_input.get(),
>+                                     audio_encoder_->NumChannels() *
>+                                         audio_encoder_->SampleRateHz() / 100),
>+                                 output);
>     }
>     EXPECT_EQ(payload_type_, encoded_info.payload_type);
>     return static_cast<int>(encoded_info.encoded_bytes);
>@@ -152,11 +153,14 @@ class AudioDecoderTest : public ::testing::Test {
>   // with |mse|. The encoded stream should contain |expected_bytes|. For stereo
>   // audio, the absolute difference between the two channels is compared vs
>   // |channel_diff_tolerance|.
>-  void EncodeDecodeTest(size_t expected_bytes, int tolerance, double mse,
>-                        int delay = 0, int channel_diff_tolerance = 0) {
>+  void EncodeDecodeTest(size_t expected_bytes,
>+                        int tolerance,
>+                        double mse,
>+                        int delay = 0,
>+                        int channel_diff_tolerance = 0) {
>     ASSERT_GE(tolerance, 0) << "Test must define a tolerance >= 0";
>-    ASSERT_GE(channel_diff_tolerance, 0) <<
>-        "Test must define a channel_diff_tolerance >= 0";
>+    ASSERT_GE(channel_diff_tolerance, 0)
>+        << "Test must define a channel_diff_tolerance >= 0";
>     size_t processed_samples = 0u;
>     rtc::Buffer encoded;
>     size_t encoded_bytes = 0u;
>@@ -168,10 +172,10 @@ class AudioDecoderTest : public ::testing::Test {
>       input.resize(input.size() + frame_size_, 0);
>       // Read from input file.
>       ASSERT_GE(input.size() - processed_samples, frame_size_);
>-      ASSERT_TRUE(input_audio_.Read(
>-          frame_size_, codec_input_rate_hz_, &input[processed_samples]));
>-      size_t enc_len = EncodeFrame(
>-          &input[processed_samples], frame_size_, &encoded);
>+      ASSERT_TRUE(input_audio_.Read(frame_size_, codec_input_rate_hz_,
>+                                    &input[processed_samples]));
>+      size_t enc_len =
>+          EncodeFrame(&input[processed_samples], frame_size_, &encoded);
>       // Make sure that frame_size_ * channels_ samples are allocated and free.
>       decoded.resize((processed_samples + frame_size_) * channels_, 0);
>       AudioDecoder::SpeechType speech_type;
>@@ -189,11 +193,11 @@ class AudioDecoderTest : public ::testing::Test {
>     if (expected_bytes) {
>       EXPECT_EQ(expected_bytes, encoded_bytes);
>     }
>-    CompareInputOutput(
>-        input, decoded, processed_samples, channels_, tolerance, delay);
>+    CompareInputOutput(input, decoded, processed_samples, channels_, tolerance,
>+                       delay);
>     if (channels_ == 2)
>-      CompareTwoChannels(
>-          decoded, processed_samples, channels_, channel_diff_tolerance);
>+      CompareTwoChannels(decoded, processed_samples, channels_,
>+                         channel_diff_tolerance);
>     EXPECT_LE(
>         MseInputOutput(input, decoded, processed_samples, channels_, delay),
>         mse);
>@@ -242,10 +246,9 @@ class AudioDecoderTest : public ::testing::Test {
>     AudioDecoder::SpeechType speech_type;
>     decoder_->Reset();
>     std::unique_ptr<int16_t[]> output(new int16_t[frame_size_ * channels_]);
>-    size_t dec_len = decoder_->Decode(encoded.data(), enc_len,
>-                                      codec_input_rate_hz_,
>-                                      frame_size_ * channels_ * sizeof(int16_t),
>-                                      output.get(), &speech_type);
>+    size_t dec_len = decoder_->Decode(
>+        encoded.data(), enc_len, codec_input_rate_hz_,
>+        frame_size_ * channels_ * sizeof(int16_t), output.get(), &speech_type);
>     EXPECT_EQ(frame_size_ * channels_, dec_len);
>     // Call DecodePlc and verify that we get one frame of data.
>     // (Overwrite the output from the above Decode call, but that does not
>@@ -332,10 +335,9 @@ class AudioDecoderIlbcTest : public AudioDecoderTest {
>     AudioDecoder::SpeechType speech_type;
>     decoder_->Reset();
>     std::unique_ptr<int16_t[]> output(new int16_t[frame_size_ * channels_]);
>-    size_t dec_len = decoder_->Decode(encoded.data(), enc_len,
>-                                      codec_input_rate_hz_,
>-                                      frame_size_ * channels_ * sizeof(int16_t),
>-                                      output.get(), &speech_type);
>+    size_t dec_len = decoder_->Decode(
>+        encoded.data(), enc_len, codec_input_rate_hz_,
>+        frame_size_ * channels_ * sizeof(int16_t), output.get(), &speech_type);
>     EXPECT_EQ(frame_size_, dec_len);
>     // Simply call DecodePlc and verify that we get 0 as return value.
>     EXPECT_EQ(0U, decoder_->DecodePlc(1, output.get()));
>@@ -462,7 +464,7 @@ TEST_F(AudioDecoderPcmUTest, EncodeDecode) {
> 
> namespace {
> int SetAndGetTargetBitrate(AudioEncoder* audio_encoder, int rate) {
>-  audio_encoder->OnReceivedUplinkBandwidth(rate, rtc::nullopt);
>+  audio_encoder->OnReceivedUplinkBandwidth(rate, absl::nullopt);
>   return audio_encoder->GetTargetBitrate();
> }
> void TestSetAndGetTargetBitratesWithFixedCodec(AudioEncoder* audio_encoder,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/audio_multi_vector.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/audio_multi_vector.cc
>index c3e623fddea..874633f389a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/audio_multi_vector.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/audio_multi_vector.cc
>@@ -15,13 +15,13 @@
> #include <algorithm>
> 
> #include "rtc_base/checks.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
> AudioMultiVector::AudioMultiVector(size_t N) {
>   assert(N > 0);
>-  if (N < 1) N = 1;
>+  if (N < 1)
>+    N = 1;
>   for (size_t n = 0; n < N; ++n) {
>     channels_.push_back(new AudioVector);
>   }
>@@ -30,7 +30,8 @@ AudioMultiVector::AudioMultiVector(size_t N) {
> 
> AudioMultiVector::AudioMultiVector(size_t N, size_t initial_size) {
>   assert(N > 0);
>-  if (N < 1) N = 1;
>+  if (N < 1)
>+    N = 1;
>   for (size_t n = 0; n < N; ++n) {
>     channels_.push_back(new AudioVector(initial_size));
>   }
>@@ -86,7 +87,7 @@ void AudioMultiVector::PushBackInterleaved(const int16_t* append_this,
>     }
>     channels_[channel]->PushBack(temp_array, length_per_channel);
>   }
>-  delete [] temp_array;
>+  delete[] temp_array;
> }
> 
> void AudioMultiVector::PushBack(const AudioMultiVector& append_this) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/audio_multi_vector.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/audio_multi_vector.h
>index f54c98b9427..4a9ed489e8e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/audio_multi_vector.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/audio_multi_vector.h
>@@ -17,7 +17,6 @@
> 
> #include "modules/audio_coding/neteq/audio_vector.h"
> #include "rtc_base/constructormagic.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/audio_multi_vector_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/audio_multi_vector_unittest.cc
>index f05aee026c9..3f3283eb4f6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/audio_multi_vector_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/audio_multi_vector_unittest.cc
>@@ -17,7 +17,6 @@
> 
> #include "rtc_base/numerics/safe_conversions.h"
> #include "test/gtest.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -37,9 +36,7 @@ class AudioMultiVectorTest : public ::testing::TestWithParam<size_t> {
>     array_interleaved_ = new int16_t[num_channels_ * array_length()];
>   }
> 
>-  ~AudioMultiVectorTest() {
>-    delete [] array_interleaved_;
>-  }
>+  ~AudioMultiVectorTest() { delete[] array_interleaved_; }
> 
>   virtual void SetUp() {
>     // Populate test arrays.
>@@ -58,9 +55,7 @@ class AudioMultiVectorTest : public ::testing::TestWithParam<size_t> {
>     }
>   }
> 
>-  size_t array_length() const {
>-    return sizeof(array_) / sizeof(array_[0]);
>-  }
>+  size_t array_length() const { return sizeof(array_) / sizeof(array_[0]); }
> 
>   const size_t num_channels_;
>   size_t interleaved_length_;
>@@ -168,8 +163,9 @@ TEST_P(AudioMultiVectorTest, PushBackFromIndex) {
>   ASSERT_EQ(2u, vec2.Size());
>   for (size_t channel = 0; channel < num_channels_; ++channel) {
>     for (size_t i = 0; i < 2; ++i) {
>-      EXPECT_EQ(array_interleaved_[channel + num_channels_ *
>-                  (array_length() - 2 + i)], vec2[channel][i]);
>+      EXPECT_EQ(array_interleaved_[channel +
>+                                   num_channels_ * (array_length() - 2 + i)],
>+                vec2[channel][i]);
>     }
>   }
> }
>@@ -206,7 +202,7 @@ TEST_P(AudioMultiVectorTest, ReadInterleaved) {
>   EXPECT_EQ(0,
>             memcmp(array_interleaved_, output, read_samples * sizeof(int16_t)));
> 
>-  delete [] output;
>+  delete[] output;
> }
> 
> // Test the PopFront method.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/audio_vector.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/audio_vector.cc
>index 93cd1fb31cd..b3ad48fc436 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/audio_vector.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/audio_vector.cc
>@@ -16,12 +16,10 @@
> #include <memory>
> 
> #include "rtc_base/checks.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>-AudioVector::AudioVector()
>-    : AudioVector(kDefaultInitialSize) {
>+AudioVector::AudioVector() : AudioVector(kDefaultInitialSize) {
>   Clear();
> }
> 
>@@ -47,16 +45,15 @@ void AudioVector::CopyTo(AudioVector* copy_to) const {
>   copy_to->end_index_ = Size();
> }
> 
>-void AudioVector::CopyTo(
>-    size_t length, size_t position, int16_t* copy_to) const {
>+void AudioVector::CopyTo(size_t length,
>+                         size_t position,
>+                         int16_t* copy_to) const {
>   if (length == 0)
>     return;
>   length = std::min(length, Size() - position);
>   const size_t copy_index = (begin_index_ + position) % capacity_;
>-  const size_t first_chunk_length =
>-      std::min(length, capacity_ - copy_index);
>-  memcpy(copy_to, &array_[copy_index],
>-         first_chunk_length * sizeof(int16_t));
>+  const size_t first_chunk_length = std::min(length, capacity_ - copy_index);
>+  memcpy(copy_to, &array_[copy_index], first_chunk_length * sizeof(int16_t));
>   const size_t remaining_length = length - first_chunk_length;
>   if (remaining_length > 0) {
>     memcpy(&copy_to[first_chunk_length], array_.get(),
>@@ -102,8 +99,9 @@ void AudioVector::PushBack(const AudioVector& append_this) {
>   PushBack(append_this, append_this.Size(), 0);
> }
> 
>-void AudioVector::PushBack(
>-    const AudioVector& append_this, size_t length, size_t position) {
>+void AudioVector::PushBack(const AudioVector& append_this,
>+                           size_t length,
>+                           size_t position) {
>   RTC_DCHECK_LE(position, append_this.Size());
>   RTC_DCHECK_LE(length, append_this.Size() - position);
> 
>@@ -116,8 +114,8 @@ void AudioVector::PushBack(
> 
>   const size_t start_index =
>       (append_this.begin_index_ + position) % append_this.capacity_;
>-  const size_t first_chunk_length = std::min(
>-      length, append_this.capacity_ - start_index);
>+  const size_t first_chunk_length =
>+      std::min(length, append_this.capacity_ - start_index);
>   PushBack(&append_this.array_[start_index], first_chunk_length);
> 
>   const size_t remaining_length = length - first_chunk_length;
>@@ -179,8 +177,7 @@ void AudioVector::InsertAt(const int16_t* insert_this,
>   }
> }
> 
>-void AudioVector::InsertZerosAt(size_t length,
>-                                size_t position) {
>+void AudioVector::InsertZerosAt(size_t length, size_t position) {
>   if (length == 0)
>     return;
>   // Cap the insert position at the current array length.
>@@ -265,7 +262,8 @@ void AudioVector::CrossFade(const AudioVector& append_this,
>     alpha -= alpha_step;
>     array_[(position + i) % capacity_] =
>         (alpha * array_[(position + i) % capacity_] +
>-            (16384 - alpha) * append_this[i] + 8192) >> 14;
>+         (16384 - alpha) * append_this[i] + 8192) >>
>+        14;
>   }
>   assert(alpha >= 0);  // Verify that the slope was correct.
>   // Append what is left of |append_this|.
>@@ -319,8 +317,8 @@ void AudioVector::InsertByPushBack(const int16_t* insert_this,
> }
> 
> void AudioVector::InsertByPushFront(const int16_t* insert_this,
>-                                   size_t length,
>-                                   size_t position) {
>+                                    size_t length,
>+                                    size_t position) {
>   std::unique_ptr<int16_t[]> temp_array(nullptr);
>   if (position > 0) {
>     // TODO(minyue): see if it is possible to avoid copying to a buffer.
>@@ -335,8 +333,7 @@ void AudioVector::InsertByPushFront(const int16_t* insert_this,
>     PushFront(temp_array.get(), position);
> }
> 
>-void AudioVector::InsertZerosByPushBack(size_t length,
>-                                        size_t position) {
>+void AudioVector::InsertZerosByPushBack(size_t length, size_t position) {
>   const size_t move_chunk_length = Size() - position;
>   std::unique_ptr<int16_t[]> temp_array(nullptr);
>   if (move_chunk_length > 0) {
>@@ -359,8 +356,7 @@ void AudioVector::InsertZerosByPushBack(size_t length,
>     PushBack(temp_array.get(), move_chunk_length);
> }
> 
>-void AudioVector::InsertZerosByPushFront(size_t length,
>-                                         size_t position) {
>+void AudioVector::InsertZerosByPushFront(size_t length, size_t position) {
>   std::unique_ptr<int16_t[]> temp_array(nullptr);
>   if (position > 0) {
>     temp_array.reset(new int16_t[position]);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/audio_vector.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/audio_vector.h
>index 754a9fdbe61..825a3bceac9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/audio_vector.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/audio_vector.h
>@@ -16,7 +16,6 @@
> 
> #include "rtc_base/checks.h"
> #include "rtc_base/constructormagic.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -75,7 +74,8 @@ class AudioVector {
>   // them at |position|. The length of the AudioVector is increased by |length|.
>   // |position| = 0 means that the new values are prepended to the vector.
>   // |position| = Size() means that the new values are appended to the vector.
>-  virtual void InsertAt(const int16_t* insert_this, size_t length,
>+  virtual void InsertAt(const int16_t* insert_this,
>+                        size_t length,
>                         size_t position);
> 
>   // Like InsertAt, but inserts |length| zero elements at |position|.
>@@ -140,10 +140,12 @@ class AudioVector {
> 
>   void Reserve(size_t n);
> 
>-  void InsertByPushBack(const int16_t* insert_this, size_t length,
>+  void InsertByPushBack(const int16_t* insert_this,
>+                        size_t length,
>                         size_t position);
> 
>-  void InsertByPushFront(const int16_t* insert_this, size_t length,
>+  void InsertByPushFront(const int16_t* insert_this,
>+                         size_t length,
>                          size_t position);
> 
>   void InsertZerosByPushBack(size_t length, size_t position);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/audio_vector_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/audio_vector_unittest.cc
>index 1b54abca5c8..1a16bb387ab 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/audio_vector_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/audio_vector_unittest.cc
>@@ -17,7 +17,6 @@
> 
> #include "rtc_base/numerics/safe_conversions.h"
> #include "test/gtest.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -30,9 +29,7 @@ class AudioVectorTest : public ::testing::Test {
>     }
>   }
> 
>-  size_t array_length() const {
>-    return sizeof(array_) / sizeof(array_[0]);
>-  }
>+  size_t array_length() const { return sizeof(array_) / sizeof(array_[0]); }
> 
>   int16_t array_[10];
> };
>@@ -283,8 +280,8 @@ TEST_F(AudioVectorTest, InsertBeyondEnd) {
>   for (int i = 0; i < kNewLength; ++i) {
>     new_array[i] = 100 + i;
>   }
>-  int insert_position = rtc::checked_cast<int>(
>-      array_length() + 10); // Too large.
>+  int insert_position =
>+      rtc::checked_cast<int>(array_length() + 10);  // Too large.
>   vec.InsertAt(new_array, kNewLength, insert_position);
>   // Verify that the vector looks as follows:
>   // {0, 1, ..., kLength - 1, 100, 101, ..., 100 + kNewLength - 1 }.
>@@ -375,7 +372,7 @@ TEST_F(AudioVectorTest, CrossFade) {
>     EXPECT_EQ(0, vec1[i]);
>   }
>   // Check mixing zone.
>-  for (size_t i = 0 ; i < kFadeLength; ++i) {
>+  for (size_t i = 0; i < kFadeLength; ++i) {
>     EXPECT_NEAR((i + 1) * 100 / (kFadeLength + 1),
>                 vec1[kLength - kFadeLength + i], 1);
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/background_noise.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/background_noise.cc
>index eda5c7514de..08c278e14a7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/background_noise.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/background_noise.cc
>@@ -27,8 +27,7 @@ const size_t BackgroundNoise::kMaxLpcOrder;
> 
> BackgroundNoise::BackgroundNoise(size_t num_channels)
>     : num_channels_(num_channels),
>-      channel_parameters_(new ChannelParameters[num_channels_]),
>-      mode_(NetEq::kBgnOn) {
>+      channel_parameters_(new ChannelParameters[num_channels_]) {
>   Reset();
> }
> 
>@@ -39,7 +38,6 @@ void BackgroundNoise::Reset() {
>   for (size_t channel = 0; channel < num_channels_; ++channel) {
>     channel_parameters_[channel].Reset();
>   }
>-  // Keep _bgnMode as it is.
> }
> 
> void BackgroundNoise::Update(const AudioMultiVector& input,
>@@ -60,11 +58,11 @@ void BackgroundNoise::Update(const AudioMultiVector& input,
>     int16_t temp_signal_array[kVecLen + kMaxLpcOrder] = {0};
>     int16_t* temp_signal = &temp_signal_array[kMaxLpcOrder];
>     input[channel_ix].CopyTo(kVecLen, input.Size() - kVecLen, temp_signal);
>-    int32_t sample_energy = CalculateAutoCorrelation(temp_signal, kVecLen,
>-                                                     auto_correlation);
>+    int32_t sample_energy =
>+        CalculateAutoCorrelation(temp_signal, kVecLen, auto_correlation);
> 
>     if ((!vad.running() &&
>-        sample_energy < parameters.energy_update_threshold) ||
>+         sample_energy < parameters.energy_update_threshold) ||
>         (vad.running() && !vad.active_speech())) {
>       // Generate LPC coefficients.
>       if (auto_correlation[0] > 0) {
>@@ -93,10 +91,8 @@ void BackgroundNoise::Update(const AudioMultiVector& input,
>       WebRtcSpl_FilterMAFastQ12(temp_signal + kVecLen - kResidualLength,
>                                 fiter_output, lpc_coefficients,
>                                 kMaxLpcOrder + 1, kResidualLength);
>-      int32_t residual_energy = WebRtcSpl_DotProductWithScale(fiter_output,
>-                                                              fiter_output,
>-                                                              kResidualLength,
>-                                                              0);
>+      int32_t residual_energy = WebRtcSpl_DotProductWithScale(
>+          fiter_output, fiter_output, kResidualLength, 0);
> 
>       // Check spectral flatness.
>       // Comparing the residual variance with the input signal variance tells
>@@ -148,7 +144,8 @@ const int16_t* BackgroundNoise::FilterState(size_t channel) const {
>   return channel_parameters_[channel].filter_state;
> }
> 
>-void BackgroundNoise::SetFilterState(size_t channel, const int16_t* input,
>+void BackgroundNoise::SetFilterState(size_t channel,
>+                                     const int16_t* input,
>                                      size_t length) {
>   assert(channel < num_channels_);
>   length = std::min(length, kMaxLpcOrder);
>@@ -166,7 +163,9 @@ int16_t BackgroundNoise::ScaleShift(size_t channel) const {
> }
> 
> int32_t BackgroundNoise::CalculateAutoCorrelation(
>-    const int16_t* signal, size_t length, int32_t* auto_correlation) const {
>+    const int16_t* signal,
>+    size_t length,
>+    int32_t* auto_correlation) const {
>   static const int kCorrelationStep = -1;
>   const int correlation_scale =
>       CrossCorrelationWithAutoShift(signal, signal, length, kMaxLpcOrder + 1,
>@@ -187,15 +186,16 @@ void BackgroundNoise::IncrementEnergyThreshold(size_t channel,
>   assert(channel < num_channels_);
>   ChannelParameters& parameters = channel_parameters_[channel];
>   int32_t temp_energy =
>-    (kThresholdIncrement * parameters.low_energy_update_threshold) >> 16;
>-  temp_energy += kThresholdIncrement *
>-      (parameters.energy_update_threshold & 0xFF);
>-  temp_energy += (kThresholdIncrement *
>-      ((parameters.energy_update_threshold>>8) & 0xFF)) << 8;
>+      (kThresholdIncrement * parameters.low_energy_update_threshold) >> 16;
>+  temp_energy +=
>+      kThresholdIncrement * (parameters.energy_update_threshold & 0xFF);
>+  temp_energy +=
>+      (kThresholdIncrement * ((parameters.energy_update_threshold >> 8) & 0xFF))
>+      << 8;
>   parameters.low_energy_update_threshold += temp_energy;
> 
>-  parameters.energy_update_threshold += kThresholdIncrement *
>-      (parameters.energy_update_threshold>>16);
>+  parameters.energy_update_threshold +=
>+      kThresholdIncrement * (parameters.energy_update_threshold >> 16);
>   parameters.energy_update_threshold +=
>       parameters.low_energy_update_threshold >> 16;
>   parameters.low_energy_update_threshold =
>@@ -203,8 +203,7 @@ void BackgroundNoise::IncrementEnergyThreshold(size_t channel,
> 
>   // Update maximum energy.
>   // Decrease by a factor 1/1024 each time.
>-  parameters.max_energy = parameters.max_energy -
>-      (parameters.max_energy >> 10);
>+  parameters.max_energy = parameters.max_energy - (parameters.max_energy >> 10);
>   if (sample_energy > parameters.max_energy) {
>     parameters.max_energy = sample_energy;
>   }
>@@ -225,9 +224,8 @@ void BackgroundNoise::SaveParameters(size_t channel,
>   assert(channel < num_channels_);
>   ChannelParameters& parameters = channel_parameters_[channel];
>   memcpy(parameters.filter, lpc_coefficients,
>-         (kMaxLpcOrder+1) * sizeof(int16_t));
>-  memcpy(parameters.filter_state, filter_state,
>-         kMaxLpcOrder * sizeof(int16_t));
>+         (kMaxLpcOrder + 1) * sizeof(int16_t));
>+  memcpy(parameters.filter_state, filter_state, kMaxLpcOrder * sizeof(int16_t));
>   // Save energy level and update energy threshold levels.
>   // Never get under 1.0 in average sample energy.
>   parameters.energy = std::max(sample_energy, 1);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/background_noise.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/background_noise.h
>index da3c1f66648..58eecaa705c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/background_noise.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/background_noise.h
>@@ -17,7 +17,6 @@
> #include "modules/audio_coding/neteq/audio_multi_vector.h"
> #include "modules/audio_coding/neteq/include/neteq.h"
> #include "rtc_base/constructormagic.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -38,8 +37,7 @@ class BackgroundNoise {
> 
>   // Updates the parameter estimates based on the signal currently in the
>   // |sync_buffer|, and on the latest decision in |vad| if it is running.
>-  void Update(const AudioMultiVector& sync_buffer,
>-              const PostDecodeVad& vad);
>+  void Update(const AudioMultiVector& sync_buffer, const PostDecodeVad& vad);
> 
>   // Returns |energy_| for |channel|.
>   int32_t Energy(size_t channel) const;
>@@ -68,11 +66,6 @@ class BackgroundNoise {
> 
>   // Accessors.
>   bool initialized() const { return initialized_; }
>-  NetEq::BackgroundNoiseMode mode() const { return mode_; }
>-
>-  // Sets the mode of the background noise playout for cases when there is long
>-  // duration of packet loss.
>-  void set_mode(NetEq::BackgroundNoiseMode mode) { mode_ = mode; }
> 
>  private:
>   static const int kThresholdIncrement = 229;  // 0.0035 in Q16.
>@@ -83,9 +76,7 @@ class BackgroundNoise {
> 
>   struct ChannelParameters {
>     // Constructor.
>-    ChannelParameters() {
>-      Reset();
>-    }
>+    ChannelParameters() { Reset(); }
> 
>     void Reset() {
>       energy = 2500;
>@@ -95,7 +86,7 @@ class BackgroundNoise {
>       memset(filter_state, 0, sizeof(filter_state));
>       memset(filter, 0, sizeof(filter));
>       filter[0] = 4096;
>-      mute_factor = 0,
>+      mute_factor = 0;
>       scale = 20000;
>       scale_shift = 24;
>     }
>@@ -128,7 +119,6 @@ class BackgroundNoise {
>   size_t num_channels_;
>   std::unique_ptr<ChannelParameters[]> channel_parameters_;
>   bool initialized_;
>-  NetEq::BackgroundNoiseMode mode_;
> 
>   RTC_DISALLOW_COPY_AND_ASSIGN(BackgroundNoise);
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/buffer_level_filter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/buffer_level_filter.cc
>index 6005de6989a..4d015b6827f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/buffer_level_filter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/buffer_level_filter.cc
>@@ -31,7 +31,8 @@ void BufferLevelFilter::Update(size_t buffer_size_packets,
>   //                            (1 - |level_factor_|) * |buffer_size_packets|
>   // |level_factor_| and |filtered_current_level_| are in Q8.
>   // |buffer_size_packets| is in Q0.
>-  filtered_current_level_ = ((level_factor_ * filtered_current_level_) >> 8) +
>+  filtered_current_level_ =
>+      ((level_factor_ * filtered_current_level_) >> 8) +
>       ((256 - level_factor_) * static_cast<int>(buffer_size_packets));
> 
>   // Account for time-scale operations (accelerate and pre-emptive expand).
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/buffer_level_filter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/buffer_level_filter.h
>index 7a48c72b742..c8d27dcd1f3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/buffer_level_filter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/buffer_level_filter.h
>@@ -28,7 +28,8 @@ class BufferLevelFilter {
>   // corresponding number of packets, and is subtracted from the filtered
>   // value (thus bypassing the filter operation). |packet_len_samples| is the
>   // number of audio samples carried in each incoming packet.
>-  virtual void Update(size_t buffer_size_packets, int time_stretched_samples,
>+  virtual void Update(size_t buffer_size_packets,
>+                      int time_stretched_samples,
>                       size_t packet_len_samples);
> 
>   // Set the current target buffer level (obtained from
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/buffer_level_filter_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/buffer_level_filter_unittest.cc
>index 72c87274291..b6dcd2a98a8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/buffer_level_filter_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/buffer_level_filter_unittest.cc
>@@ -39,8 +39,7 @@ TEST(BufferLevelFilter, ConvergenceTest) {
>       }
>       // Expect the filtered value to be (theoretically)
>       // (1 - (251/256) ^ |times|) * |value|.
>-      double expected_value_double =
>-          (1 - pow(251.0 / 256.0, times)) * value;
>+      double expected_value_double = (1 - pow(251.0 / 256.0, times)) * value;
>       int expected_value = static_cast<int>(expected_value_double);
>       // filtered_current_level() returns the value in Q8.
>       // The actual value may differ slightly from the expected value due to
>@@ -94,7 +93,6 @@ TEST(BufferLevelFilter, FilterFactor) {
>   EXPECT_EQ(expected_value, filter.filtered_current_level() >> 8);
> }
> 
>-
> TEST(BufferLevelFilter, TimeStretchedSamples) {
>   BufferLevelFilter filter;
>   filter.SetTargetBufferLevel(1);  // Makes filter coefficient 251/256.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/comfort_noise.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/comfort_noise.cc
>index 5e0a87526ee..b341acd5a50 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/comfort_noise.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/comfort_noise.cc
>@@ -35,10 +35,9 @@ int ComfortNoise::UpdateParameters(const Packet& packet) {
>   return kOK;
> }
> 
>-int ComfortNoise::Generate(size_t requested_length,
>-                           AudioMultiVector* output) {
>+int ComfortNoise::Generate(size_t requested_length, AudioMultiVector* output) {
>   // TODO(hlundin): Change to an enumerator and skip assert.
>-  assert(fs_hz_ == 8000 || fs_hz_ == 16000 || fs_hz_ ==  32000 ||
>+  assert(fs_hz_ == 8000 || fs_hz_ == 16000 || fs_hz_ == 32000 ||
>          fs_hz_ == 48000);
>   // Not adapted for multi-channel yet.
>   if (output->Channels() != 1) {
>@@ -63,8 +62,7 @@ int ComfortNoise::Generate(size_t requested_length,
> 
>   std::unique_ptr<int16_t[]> temp(new int16_t[number_of_samples]);
>   if (!cng_decoder->Generate(
>-          rtc::ArrayView<int16_t>(temp.get(), number_of_samples),
>-          new_period)) {
>+          rtc::ArrayView<int16_t>(temp.get(), number_of_samples), new_period)) {
>     // Error returned.
>     output->Zeros(requested_length);
>     RTC_LOG(LS_ERROR)
>@@ -75,9 +73,9 @@ int ComfortNoise::Generate(size_t requested_length,
> 
>   if (first_call_) {
>     // Set tapering window parameters. Values are in Q15.
>-    int16_t muting_window;  // Mixing factor for overlap data.
>-    int16_t muting_window_increment;  // Mixing factor increment (negative).
>-    int16_t unmuting_window;  // Mixing factor for comfort noise.
>+    int16_t muting_window;              // Mixing factor for overlap data.
>+    int16_t muting_window_increment;    // Mixing factor increment (negative).
>+    int16_t unmuting_window;            // Mixing factor for comfort noise.
>     int16_t unmuting_window_increment;  // Mixing factor increment.
>     if (fs_hz_ == 8000) {
>       muting_window = DspHelper::kMuteFactorStart8kHz;
>@@ -109,7 +107,8 @@ int ComfortNoise::Generate(size_t requested_length,
>       // channel.
>       (*sync_buffer_)[0][start_ix + i] =
>           (((*sync_buffer_)[0][start_ix + i] * muting_window) +
>-              ((*output)[0][i] * unmuting_window) + 16384) >> 15;
>+           ((*output)[0][i] * unmuting_window) + 16384) >>
>+          15;
>       muting_window += muting_window_increment;
>       unmuting_window += unmuting_window_increment;
>     }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/comfort_noise.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/comfort_noise.h
>index 18800ad6441..5169124040f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/comfort_noise.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/comfort_noise.h
>@@ -13,7 +13,6 @@
> 
> #include "modules/audio_coding/neteq/audio_multi_vector.h"
> #include "rtc_base/constructormagic.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -32,14 +31,14 @@ class ComfortNoise {
>     kMultiChannelNotSupported
>   };
> 
>-  ComfortNoise(int fs_hz, DecoderDatabase* decoder_database,
>+  ComfortNoise(int fs_hz,
>+               DecoderDatabase* decoder_database,
>                SyncBuffer* sync_buffer)
>       : fs_hz_(fs_hz),
>         first_call_(true),
>         overlap_length_(5 * fs_hz_ / 8000),
>         decoder_database_(decoder_database),
>-        sync_buffer_(sync_buffer) {
>-  }
>+        sync_buffer_(sync_buffer) {}
> 
>   // Resets the state. Should be called before each new comfort noise period.
>   void Reset();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/cross_correlation.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/cross_correlation.cc
>index da9c9132492..2a03d4af263 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/cross_correlation.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/cross_correlation.cc
>@@ -48,8 +48,9 @@ int CrossCorrelationWithAutoShift(const int16_t* sequence_1,
>   // There are some corner cases that 2) is not satisfied, e.g.,
>   // max_1 = 17, max_2 = 30848, sequence_1_length = 4095, in such case,
>   // optimal scaling is 0, while the following calculation results in 1.
>-  const int32_t factor = (max_1 * max_2) / (std::numeric_limits<int32_t>::max()
>-      / static_cast<int32_t>(sequence_1_length));
>+  const int32_t factor =
>+      (max_1 * max_2) / (std::numeric_limits<int32_t>::max() /
>+                         static_cast<int32_t>(sequence_1_length));
>   const int scaling = factor == 0 ? 0 : 31 - WebRtcSpl_NormW32(factor);
> 
>   WebRtcSpl_CrossCorrelation(cross_correlation, sequence_1, sequence_2,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/decision_logic.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/decision_logic.cc
>index 966d5c376da..e24ca6283a0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/decision_logic.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/decision_logic.cc
>@@ -10,46 +10,37 @@
> 
> #include "modules/audio_coding/neteq/decision_logic.h"
> 
>+#include <assert.h>
> #include <algorithm>
>+#include <limits>
> 
> #include "modules/audio_coding/neteq/buffer_level_filter.h"
>-#include "modules/audio_coding/neteq/decision_logic_fax.h"
>-#include "modules/audio_coding/neteq/decision_logic_normal.h"
>+#include "modules/audio_coding/neteq/decoder_database.h"
> #include "modules/audio_coding/neteq/delay_manager.h"
> #include "modules/audio_coding/neteq/expand.h"
> #include "modules/audio_coding/neteq/packet_buffer.h"
> #include "modules/audio_coding/neteq/sync_buffer.h"
>+#include "modules/include/module_common_types.h"
>+#include "system_wrappers/include/field_trial.h"
> 
> namespace webrtc {
> 
> DecisionLogic* DecisionLogic::Create(int fs_hz,
>                                      size_t output_size_samples,
>-                                     NetEqPlayoutMode playout_mode,
>+                                     bool disallow_time_stretching,
>                                      DecoderDatabase* decoder_database,
>                                      const PacketBuffer& packet_buffer,
>                                      DelayManager* delay_manager,
>                                      BufferLevelFilter* buffer_level_filter,
>                                      const TickTimer* tick_timer) {
>-  switch (playout_mode) {
>-    case kPlayoutOn:
>-    case kPlayoutStreaming:
>-      return new DecisionLogicNormal(
>-          fs_hz, output_size_samples, playout_mode, decoder_database,
>-          packet_buffer, delay_manager, buffer_level_filter, tick_timer);
>-    case kPlayoutFax:
>-    case kPlayoutOff:
>-      return new DecisionLogicFax(
>-          fs_hz, output_size_samples, playout_mode, decoder_database,
>-          packet_buffer, delay_manager, buffer_level_filter, tick_timer);
>-  }
>-  // This line cannot be reached, but must be here to avoid compiler errors.
>-  assert(false);
>-  return NULL;
>+  return new DecisionLogic(fs_hz, output_size_samples, disallow_time_stretching,
>+                           decoder_database, packet_buffer, delay_manager,
>+                           buffer_level_filter, tick_timer);
> }
> 
> DecisionLogic::DecisionLogic(int fs_hz,
>                              size_t output_size_samples,
>-                             NetEqPlayoutMode playout_mode,
>+                             bool disallow_time_stretching,
>                              DecoderDatabase* decoder_database,
>                              const PacketBuffer& packet_buffer,
>                              DelayManager* delay_manager,
>@@ -64,11 +55,13 @@ DecisionLogic::DecisionLogic(int fs_hz,
>       packet_length_samples_(0),
>       sample_memory_(0),
>       prev_time_scale_(false),
>+      disallow_time_stretching_(disallow_time_stretching),
>       timescale_countdown_(
>           tick_timer_->GetNewCountdown(kMinTimescaleInterval + 1)),
>       num_consecutive_expands_(0),
>-      playout_mode_(playout_mode) {
>-  delay_manager_->set_streaming_mode(playout_mode_ == kPlayoutStreaming);
>+      postpone_decoding_after_expand_(field_trial::IsEnabled(
>+          "WebRTC-Audio-NetEqPostponeDecodingAfterExpand")) {
>+  delay_manager_->set_streaming_mode(false);
>   SetSampleRate(fs_hz, output_size_samples);
> }
> 
>@@ -94,7 +87,7 @@ void DecisionLogic::SoftReset() {
> 
> void DecisionLogic::SetSampleRate(int fs_hz, size_t output_size_samples) {
>   // TODO(hlundin): Change to an enumerator and skip assert.
>-  assert(fs_hz == 8000 || fs_hz == 16000 || fs_hz ==  32000 || fs_hz == 48000);
>+  assert(fs_hz == 8000 || fs_hz == 16000 || fs_hz == 32000 || fs_hz == 48000);
>   fs_mult_ = fs_hz / 8000;
>   output_size_samples_ = output_size_samples;
> }
>@@ -121,17 +114,79 @@ Operations DecisionLogic::GetDecision(const SyncBuffer& sync_buffer,
>   const size_t cur_size_samples =
>       samples_left + packet_buffer_.NumSamplesInBuffer(decoder_frame_length);
> 
>-  prev_time_scale_ = prev_time_scale_ &&
>-      (prev_mode == kModeAccelerateSuccess ||
>-          prev_mode == kModeAccelerateLowEnergy ||
>-          prev_mode == kModePreemptiveExpandSuccess ||
>-          prev_mode == kModePreemptiveExpandLowEnergy);
>+  prev_time_scale_ =
>+      prev_time_scale_ && (prev_mode == kModeAccelerateSuccess ||
>+                           prev_mode == kModeAccelerateLowEnergy ||
>+                           prev_mode == kModePreemptiveExpandSuccess ||
>+                           prev_mode == kModePreemptiveExpandLowEnergy);
> 
>   FilterBufferLevel(cur_size_samples, prev_mode);
> 
>-  return GetDecisionSpecialized(sync_buffer, expand, decoder_frame_length,
>-                                next_packet, prev_mode, play_dtmf,
>-                                reset_decoder, generated_noise_samples);
>+  // Guard for errors, to avoid getting stuck in error mode.
>+  if (prev_mode == kModeError) {
>+    if (!next_packet) {
>+      return kExpand;
>+    } else {
>+      return kUndefined;  // Use kUndefined to flag for a reset.
>+    }
>+  }
>+
>+  uint32_t target_timestamp = sync_buffer.end_timestamp();
>+  uint32_t available_timestamp = 0;
>+  bool is_cng_packet = false;
>+  if (next_packet) {
>+    available_timestamp = next_packet->timestamp;
>+    is_cng_packet =
>+        decoder_database_->IsComfortNoise(next_packet->payload_type);
>+  }
>+
>+  if (is_cng_packet) {
>+    return CngOperation(prev_mode, target_timestamp, available_timestamp,
>+                        generated_noise_samples);
>+  }
>+
>+  // Handle the case with no packet at all available (except maybe DTMF).
>+  if (!next_packet) {
>+    return NoPacket(play_dtmf);
>+  }
>+
>+  // If the expand period was very long, reset NetEQ since it is likely that the
>+  // sender was restarted.
>+  if (num_consecutive_expands_ > kReinitAfterExpands) {
>+    *reset_decoder = true;
>+    return kNormal;
>+  }
>+
>+  // Make sure we don't restart audio too soon after an expansion to avoid
>+  // running out of data right away again. We should only wait if there are no
>+  // DTX or CNG packets in the buffer (otherwise we should just play out what we
>+  // have, since we cannot know the exact duration of DTX or CNG packets), and
>+  // if the mute factor is low enough (otherwise the expansion was short enough
>+  // to not be noticable).
>+  // Note that the MuteFactor is in Q14, so a value of 16384 corresponds to 1.
>+  if (postpone_decoding_after_expand_ && prev_mode == kModeExpand &&
>+      !packet_buffer_.ContainsDtxOrCngPacket(decoder_database_) &&
>+      cur_size_samples<static_cast<size_t>(delay_manager_->TargetLevel() *
>+                                           packet_length_samples_)>> 8 &&
>+      expand.MuteFactor(0) < 16384 / 2) {
>+    return kExpand;
>+  }
>+
>+  const uint32_t five_seconds_samples =
>+      static_cast<uint32_t>(5 * 8000 * fs_mult_);
>+  // Check if the required packet is available.
>+  if (target_timestamp == available_timestamp) {
>+    return ExpectedPacketAvailable(prev_mode, play_dtmf);
>+  } else if (!PacketBuffer::IsObsoleteTimestamp(
>+                 available_timestamp, target_timestamp, five_seconds_samples)) {
>+    return FuturePacketAvailable(
>+        sync_buffer, expand, decoder_frame_length, prev_mode, target_timestamp,
>+        available_timestamp, play_dtmf, generated_noise_samples);
>+  } else {
>+    // This implies that available_timestamp < target_timestamp, which can
>+    // happen when a new stream or codec is received. Signal for a reset.
>+    return kUndefined;
>+  }
> }
> 
> void DecisionLogic::ExpandDecision(Operations operation) {
>@@ -167,4 +222,152 @@ void DecisionLogic::FilterBufferLevel(size_t buffer_size_samples,
>   }
> }
> 
>+Operations DecisionLogic::CngOperation(Modes prev_mode,
>+                                       uint32_t target_timestamp,
>+                                       uint32_t available_timestamp,
>+                                       size_t generated_noise_samples) {
>+  // Signed difference between target and available timestamp.
>+  int32_t timestamp_diff = static_cast<int32_t>(
>+      static_cast<uint32_t>(generated_noise_samples + target_timestamp) -
>+      available_timestamp);
>+  int32_t optimal_level_samp = static_cast<int32_t>(
>+      (delay_manager_->TargetLevel() * packet_length_samples_) >> 8);
>+  const int64_t excess_waiting_time_samp =
>+      -static_cast<int64_t>(timestamp_diff) - optimal_level_samp;
>+
>+  if (excess_waiting_time_samp > optimal_level_samp / 2) {
>+    // The waiting time for this packet will be longer than 1.5
>+    // times the wanted buffer delay. Apply fast-forward to cut the
>+    // waiting time down to the optimal.
>+    noise_fast_forward_ = rtc::dchecked_cast<size_t>(noise_fast_forward_ +
>+                                                     excess_waiting_time_samp);
>+    timestamp_diff =
>+        rtc::saturated_cast<int32_t>(timestamp_diff + excess_waiting_time_samp);
>+  }
>+
>+  if (timestamp_diff < 0 && prev_mode == kModeRfc3389Cng) {
>+    // Not time to play this packet yet. Wait another round before using this
>+    // packet. Keep on playing CNG from previous CNG parameters.
>+    return kRfc3389CngNoPacket;
>+  } else {
>+    // Otherwise, go for the CNG packet now.
>+    noise_fast_forward_ = 0;
>+    return kRfc3389Cng;
>+  }
>+}
>+
>+Operations DecisionLogic::NoPacket(bool play_dtmf) {
>+  if (cng_state_ == kCngRfc3389On) {
>+    // Keep on playing comfort noise.
>+    return kRfc3389CngNoPacket;
>+  } else if (cng_state_ == kCngInternalOn) {
>+    // Keep on playing codec internal comfort noise.
>+    return kCodecInternalCng;
>+  } else if (play_dtmf) {
>+    return kDtmf;
>+  } else {
>+    // Nothing to play, do expand.
>+    return kExpand;
>+  }
>+}
>+
>+Operations DecisionLogic::ExpectedPacketAvailable(Modes prev_mode,
>+                                                  bool play_dtmf) {
>+  if (!disallow_time_stretching_ && prev_mode != kModeExpand && !play_dtmf) {
>+    // Check criterion for time-stretching.
>+    int low_limit, high_limit;
>+    delay_manager_->BufferLimits(&low_limit, &high_limit);
>+    if (buffer_level_filter_->filtered_current_level() >= high_limit << 2)
>+      return kFastAccelerate;
>+    if (TimescaleAllowed()) {
>+      if (buffer_level_filter_->filtered_current_level() >= high_limit)
>+        return kAccelerate;
>+      if (buffer_level_filter_->filtered_current_level() < low_limit)
>+        return kPreemptiveExpand;
>+    }
>+  }
>+  return kNormal;
>+}
>+
>+Operations DecisionLogic::FuturePacketAvailable(
>+    const SyncBuffer& sync_buffer,
>+    const Expand& expand,
>+    size_t decoder_frame_length,
>+    Modes prev_mode,
>+    uint32_t target_timestamp,
>+    uint32_t available_timestamp,
>+    bool play_dtmf,
>+    size_t generated_noise_samples) {
>+  // Required packet is not available, but a future packet is.
>+  // Check if we should continue with an ongoing expand because the new packet
>+  // is too far into the future.
>+  uint32_t timestamp_leap = available_timestamp - target_timestamp;
>+  if ((prev_mode == kModeExpand) && !ReinitAfterExpands(timestamp_leap) &&
>+      !MaxWaitForPacket() && PacketTooEarly(timestamp_leap) &&
>+      UnderTargetLevel()) {
>+    if (play_dtmf) {
>+      // Still have DTMF to play, so do not do expand.
>+      return kDtmf;
>+    } else {
>+      // Nothing to play.
>+      return kExpand;
>+    }
>+  }
>+
>+  const size_t samples_left =
>+      sync_buffer.FutureLength() - expand.overlap_length();
>+  const size_t cur_size_samples =
>+      samples_left + packet_buffer_.NumPacketsInBuffer() * decoder_frame_length;
>+
>+  // If previous was comfort noise, then no merge is needed.
>+  if (prev_mode == kModeRfc3389Cng || prev_mode == kModeCodecInternalCng) {
>+    // Keep the same delay as before the CNG, but make sure that the number of
>+    // samples in buffer is no higher than 4 times the optimal level. (Note that
>+    // TargetLevel() is in Q8.)
>+    if (static_cast<uint32_t>(generated_noise_samples + target_timestamp) >=
>+            available_timestamp ||
>+        cur_size_samples >
>+            ((delay_manager_->TargetLevel() * packet_length_samples_) >> 8) *
>+                4) {
>+      // Time to play this new packet.
>+      return kNormal;
>+    } else {
>+      // Too early to play this new packet; keep on playing comfort noise.
>+      if (prev_mode == kModeRfc3389Cng) {
>+        return kRfc3389CngNoPacket;
>+      } else {  // prevPlayMode == kModeCodecInternalCng.
>+        return kCodecInternalCng;
>+      }
>+    }
>+  }
>+  // Do not merge unless we have done an expand before.
>+  if (prev_mode == kModeExpand) {
>+    return kMerge;
>+  } else if (play_dtmf) {
>+    // Play DTMF instead of expand.
>+    return kDtmf;
>+  } else {
>+    return kExpand;
>+  }
>+}
>+
>+bool DecisionLogic::UnderTargetLevel() const {
>+  return buffer_level_filter_->filtered_current_level() <=
>+         delay_manager_->TargetLevel();
>+}
>+
>+bool DecisionLogic::ReinitAfterExpands(uint32_t timestamp_leap) const {
>+  return timestamp_leap >=
>+         static_cast<uint32_t>(output_size_samples_ * kReinitAfterExpands);
>+}
>+
>+bool DecisionLogic::PacketTooEarly(uint32_t timestamp_leap) const {
>+  return timestamp_leap >
>+         static_cast<uint32_t>(output_size_samples_ * num_consecutive_expands_);
>+}
>+
>+bool DecisionLogic::MaxWaitForPacket() const {
>+  return num_consecutive_expands_ >= kMaxWaitForPacket;
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/decision_logic.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/decision_logic.h
>index 5b67196c439..00b862008c9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/decision_logic.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/decision_logic.h
>@@ -15,7 +15,6 @@
> #include "modules/audio_coding/neteq/include/neteq.h"
> #include "modules/audio_coding/neteq/tick_timer.h"
> #include "rtc_base/constructormagic.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -28,32 +27,34 @@ class PacketBuffer;
> class SyncBuffer;
> struct Packet;
> 
>-// This is the base class for the decision tree implementations. Derived classes
>-// must implement the method GetDecisionSpecialized().
>-class DecisionLogic {
>+// This is the class for the decision tree implementation.
>+class DecisionLogic final {
>  public:
>   // Static factory function which creates different types of objects depending
>   // on the |playout_mode|.
>   static DecisionLogic* Create(int fs_hz,
>                                size_t output_size_samples,
>-                               NetEqPlayoutMode playout_mode,
>+                               bool disallow_time_stretching,
>                                DecoderDatabase* decoder_database,
>                                const PacketBuffer& packet_buffer,
>                                DelayManager* delay_manager,
>                                BufferLevelFilter* buffer_level_filter,
>                                const TickTimer* tick_timer);
> 
>+  static const int kReinitAfterExpands = 100;
>+  static const int kMaxWaitForPacket = 10;
>+
>   // Constructor.
>   DecisionLogic(int fs_hz,
>                 size_t output_size_samples,
>-                NetEqPlayoutMode playout_mode,
>+                bool disallow_time_stretching,
>                 DecoderDatabase* decoder_database,
>                 const PacketBuffer& packet_buffer,
>                 DelayManager* delay_manager,
>                 BufferLevelFilter* buffer_level_filter,
>                 const TickTimer* tick_timer);
> 
>-  virtual ~DecisionLogic();
>+  ~DecisionLogic();
> 
>   // Resets object to a clean state.
>   void Reset();
>@@ -94,12 +95,10 @@ class DecisionLogic {
>   // not. Note that this is necessary, since an expand decision can be changed
>   // to kNormal in NetEqImpl::GetDecision if there is still enough data in the
>   // sync buffer.
>-  virtual void ExpandDecision(Operations operation);
>+  void ExpandDecision(Operations operation);
> 
>   // Adds |value| to |sample_memory_|.
>-  void AddSampleMemory(int32_t value) {
>-    sample_memory_ += value;
>-  }
>+  void AddSampleMemory(int32_t value) { sample_memory_ += value; }
> 
>   // Accessors and mutators.
>   void set_sample_memory(int32_t value) { sample_memory_ = value; }
>@@ -109,40 +108,63 @@ class DecisionLogic {
>     packet_length_samples_ = value;
>   }
>   void set_prev_time_scale(bool value) { prev_time_scale_ = value; }
>-  NetEqPlayoutMode playout_mode() const { return playout_mode_; }
> 
>- protected:
>+ private:
>   // The value 5 sets maximum time-stretch rate to about 100 ms/s.
>   static const int kMinTimescaleInterval = 5;
> 
>-  enum CngState {
>-    kCngOff,
>-    kCngRfc3389On,
>-    kCngInternalOn
>-  };
>-
>-  // Returns the operation that should be done next. |sync_buffer| and |expand|
>-  // are provided for reference. |decoder_frame_length| is the number of samples
>-  // obtained from the last decoded frame. If there is a packet available, it
>-  // should be supplied in |next_packet|; otherwise it should be NULL. The mode
>-  // resulting from the last call to NetEqImpl::GetAudio is supplied in
>-  // |prev_mode|. If there is a DTMF event to play, |play_dtmf| should be set to
>-  // true. The output variable |reset_decoder| will be set to true if a reset is
>-  // required; otherwise it is left unchanged (i.e., it can remain true if it
>-  // was true before the call).  Should be implemented by derived classes.
>-  virtual Operations GetDecisionSpecialized(const SyncBuffer& sync_buffer,
>-                                            const Expand& expand,
>-                                            size_t decoder_frame_length,
>-                                            const Packet* next_packet,
>-                                            Modes prev_mode,
>-                                            bool play_dtmf,
>-                                            bool* reset_decoder,
>-                                            size_t generated_noise_samples) = 0;
>+  enum CngState { kCngOff, kCngRfc3389On, kCngInternalOn };
> 
>   // Updates the |buffer_level_filter_| with the current buffer level
>   // |buffer_size_packets|.
>   void FilterBufferLevel(size_t buffer_size_packets, Modes prev_mode);
> 
>+  // Returns the operation given that the next available packet is a comfort
>+  // noise payload (RFC 3389 only, not codec-internal).
>+  Operations CngOperation(Modes prev_mode,
>+                          uint32_t target_timestamp,
>+                          uint32_t available_timestamp,
>+                          size_t generated_noise_samples);
>+
>+  // Returns the operation given that no packets are available (except maybe
>+  // a DTMF event, flagged by setting |play_dtmf| true).
>+  Operations NoPacket(bool play_dtmf);
>+
>+  // Returns the operation to do given that the expected packet is available.
>+  Operations ExpectedPacketAvailable(Modes prev_mode, bool play_dtmf);
>+
>+  // Returns the operation to do given that the expected packet is not
>+  // available, but a packet further into the future is at hand.
>+  Operations FuturePacketAvailable(const SyncBuffer& sync_buffer,
>+                                   const Expand& expand,
>+                                   size_t decoder_frame_length,
>+                                   Modes prev_mode,
>+                                   uint32_t target_timestamp,
>+                                   uint32_t available_timestamp,
>+                                   bool play_dtmf,
>+                                   size_t generated_noise_samples);
>+
>+  // Checks if enough time has elapsed since the last successful timescale
>+  // operation was done (i.e., accelerate or preemptive expand).
>+  bool TimescaleAllowed() const {
>+    return !timescale_countdown_ || timescale_countdown_->Finished();
>+  }
>+
>+  // Checks if the current (filtered) buffer level is under the target level.
>+  bool UnderTargetLevel() const;
>+
>+  // Checks if |timestamp_leap| is so long into the future that a reset due
>+  // to exceeding kReinitAfterExpands will be done.
>+  bool ReinitAfterExpands(uint32_t timestamp_leap) const;
>+
>+  // Checks if we still have not done enough expands to cover the distance from
>+  // the last decoded packet to the next available packet, the distance beeing
>+  // conveyed in |timestamp_leap|.
>+  bool PacketTooEarly(uint32_t timestamp_leap) const;
>+
>+  // Checks if num_consecutive_expands_ >= kMaxWaitForPacket.
>+  bool MaxWaitForPacket() const;
>+
>   DecoderDatabase* decoder_database_;
>   const PacketBuffer& packet_buffer_;
>   DelayManager* delay_manager_;
>@@ -156,11 +178,11 @@ class DecisionLogic {
>   size_t packet_length_samples_;
>   int sample_memory_;
>   bool prev_time_scale_;
>+  bool disallow_time_stretching_;
>   std::unique_ptr<TickTimer::Countdown> timescale_countdown_;
>   int num_consecutive_expands_;
>-  const NetEqPlayoutMode playout_mode_;
>+  const bool postpone_decoding_after_expand_;
> 
>- private:
>   RTC_DISALLOW_COPY_AND_ASSIGN(DecisionLogic);
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/decision_logic_fax.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/decision_logic_fax.cc
>deleted file mode 100644
>index cc21ee9debf..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/decision_logic_fax.cc
>+++ /dev/null
>@@ -1,103 +0,0 @@
>-/*
>- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "modules/audio_coding/neteq/decision_logic_fax.h"
>-
>-#include <assert.h>
>-
>-#include <algorithm>
>-
>-#include "modules/audio_coding/neteq/decoder_database.h"
>-#include "modules/audio_coding/neteq/sync_buffer.h"
>-
>-namespace webrtc {
>-
>-Operations DecisionLogicFax::GetDecisionSpecialized(
>-    const SyncBuffer& sync_buffer,
>-    const Expand& expand,
>-    size_t decoder_frame_length,
>-    const Packet* next_packet,
>-    Modes prev_mode,
>-    bool play_dtmf,
>-    bool* reset_decoder,
>-    size_t generated_noise_samples) {
>-  assert(playout_mode_ == kPlayoutFax || playout_mode_ == kPlayoutOff);
>-  uint32_t target_timestamp = sync_buffer.end_timestamp();
>-  uint32_t available_timestamp = 0;
>-  int is_cng_packet = 0;
>-  if (next_packet) {
>-    available_timestamp = next_packet->timestamp;
>-    is_cng_packet =
>-        decoder_database_->IsComfortNoise(next_packet->payload_type);
>-  }
>-  if (is_cng_packet) {
>-    if (static_cast<int32_t>((generated_noise_samples + target_timestamp)
>-        - available_timestamp) >= 0) {
>-      // Time to play this packet now.
>-      return kRfc3389Cng;
>-    } else {
>-      // Wait before playing this packet.
>-      return kRfc3389CngNoPacket;
>-    }
>-  }
>-  if (!next_packet) {
>-    // No packet. If in CNG mode, play as usual. Otherwise, use other method to
>-    // generate data.
>-    if (cng_state_ == kCngRfc3389On) {
>-      // Continue playing comfort noise.
>-      return kRfc3389CngNoPacket;
>-    } else if (cng_state_ == kCngInternalOn) {
>-      // Continue playing codec-internal comfort noise.
>-      return kCodecInternalCng;
>-    } else {
>-      // Nothing to play. Generate some data to play out.
>-      switch (playout_mode_) {
>-        case kPlayoutOff:
>-          return kAlternativePlc;
>-        case kPlayoutFax:
>-          return kAudioRepetition;
>-        default:
>-          assert(false);
>-          return kUndefined;
>-      }
>-    }
>-  } else if (target_timestamp == available_timestamp) {
>-    return kNormal;
>-  } else {
>-    if (static_cast<int32_t>((generated_noise_samples + target_timestamp)
>-        - available_timestamp) >= 0) {
>-      return kNormal;
>-    } else {
>-      // If currently playing comfort noise, continue with that. Do not
>-      // increase the timestamp counter since generated_noise_stopwatch_ in
>-      // NetEqImpl will take care of the time-keeping.
>-      if (cng_state_ == kCngRfc3389On) {
>-        return kRfc3389CngNoPacket;
>-      } else if (cng_state_ == kCngInternalOn) {
>-        return kCodecInternalCng;
>-      } else {
>-        // Otherwise, do packet-loss concealment and increase the
>-        // timestamp while waiting for the time to play this packet.
>-        switch (playout_mode_) {
>-          case kPlayoutOff:
>-            return kAlternativePlcIncreaseTimestamp;
>-          case kPlayoutFax:
>-            return kAudioRepetitionIncreaseTimestamp;
>-          default:
>-            assert(0);
>-            return kUndefined;
>-        }
>-      }
>-    }
>-  }
>-}
>-
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/decision_logic_fax.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/decision_logic_fax.h
>deleted file mode 100644
>index cefd8e4e426..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/decision_logic_fax.h
>+++ /dev/null
>@@ -1,57 +0,0 @@
>-/*
>- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_FAX_H_
>-#define MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_FAX_H_
>-
>-#include "modules/audio_coding/neteq/decision_logic.h"
>-#include "rtc_base/constructormagic.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>-
>-namespace webrtc {
>-
>-// Implementation of the DecisionLogic class for playout modes kPlayoutFax and
>-// kPlayoutOff.
>-class DecisionLogicFax : public DecisionLogic {
>- public:
>-  // Constructor.
>-  DecisionLogicFax(int fs_hz,
>-                   size_t output_size_samples,
>-                   NetEqPlayoutMode playout_mode,
>-                   DecoderDatabase* decoder_database,
>-                   const PacketBuffer& packet_buffer,
>-                   DelayManager* delay_manager,
>-                   BufferLevelFilter* buffer_level_filter,
>-                   const TickTimer* tick_timer)
>-      : DecisionLogic(fs_hz,
>-                      output_size_samples,
>-                      playout_mode,
>-                      decoder_database,
>-                      packet_buffer,
>-                      delay_manager,
>-                      buffer_level_filter,
>-                      tick_timer) {}
>-
>- protected:
>-  Operations GetDecisionSpecialized(const SyncBuffer& sync_buffer,
>-                                    const Expand& expand,
>-                                    size_t decoder_frame_length,
>-                                    const Packet* next_packet,
>-                                    Modes prev_mode,
>-                                    bool play_dtmf,
>-                                    bool* reset_decoder,
>-                                    size_t generated_noise_samples) override;
>-
>- private:
>-  RTC_DISALLOW_COPY_AND_ASSIGN(DecisionLogicFax);
>-};
>-
>-}  // namespace webrtc
>-#endif  // MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_FAX_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/decision_logic_normal.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/decision_logic_normal.cc
>deleted file mode 100644
>index 10f501ac9ab..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/decision_logic_normal.cc
>+++ /dev/null
>@@ -1,241 +0,0 @@
>-/*
>- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "modules/audio_coding/neteq/decision_logic_normal.h"
>-
>-#include <assert.h>
>-
>-#include <algorithm>
>-
>-#include "modules/audio_coding/neteq/buffer_level_filter.h"
>-#include "modules/audio_coding/neteq/decoder_database.h"
>-#include "modules/audio_coding/neteq/delay_manager.h"
>-#include "modules/audio_coding/neteq/expand.h"
>-#include "modules/audio_coding/neteq/packet_buffer.h"
>-#include "modules/audio_coding/neteq/sync_buffer.h"
>-#include "modules/include/module_common_types.h"
>-
>-namespace webrtc {
>-
>-Operations DecisionLogicNormal::GetDecisionSpecialized(
>-    const SyncBuffer& sync_buffer,
>-    const Expand& expand,
>-    size_t decoder_frame_length,
>-    const Packet* next_packet,
>-    Modes prev_mode,
>-    bool play_dtmf,
>-    bool* reset_decoder,
>-    size_t generated_noise_samples) {
>-  assert(playout_mode_ == kPlayoutOn || playout_mode_ == kPlayoutStreaming);
>-  // Guard for errors, to avoid getting stuck in error mode.
>-  if (prev_mode == kModeError) {
>-    if (!next_packet) {
>-      return kExpand;
>-    } else {
>-      return kUndefined;  // Use kUndefined to flag for a reset.
>-    }
>-  }
>-
>-  uint32_t target_timestamp = sync_buffer.end_timestamp();
>-  uint32_t available_timestamp = 0;
>-  bool is_cng_packet = false;
>-  if (next_packet) {
>-    available_timestamp = next_packet->timestamp;
>-    is_cng_packet =
>-        decoder_database_->IsComfortNoise(next_packet->payload_type);
>-  }
>-
>-  if (is_cng_packet) {
>-    return CngOperation(prev_mode, target_timestamp, available_timestamp,
>-                        generated_noise_samples);
>-  }
>-
>-  // Handle the case with no packet at all available (except maybe DTMF).
>-  if (!next_packet) {
>-    return NoPacket(play_dtmf);
>-  }
>-
>-  // If the expand period was very long, reset NetEQ since it is likely that the
>-  // sender was restarted.
>-  if (num_consecutive_expands_ > kReinitAfterExpands) {
>-    *reset_decoder = true;
>-    return kNormal;
>-  }
>-
>-  const uint32_t five_seconds_samples =
>-      static_cast<uint32_t>(5 * 8000 * fs_mult_);
>-  // Check if the required packet is available.
>-  if (target_timestamp == available_timestamp) {
>-    return ExpectedPacketAvailable(prev_mode, play_dtmf);
>-  } else if (!PacketBuffer::IsObsoleteTimestamp(
>-                 available_timestamp, target_timestamp, five_seconds_samples)) {
>-    return FuturePacketAvailable(sync_buffer, expand, decoder_frame_length,
>-                                 prev_mode, target_timestamp,
>-                                 available_timestamp, play_dtmf,
>-                                 generated_noise_samples);
>-  } else {
>-    // This implies that available_timestamp < target_timestamp, which can
>-    // happen when a new stream or codec is received. Signal for a reset.
>-    return kUndefined;
>-  }
>-}
>-
>-Operations DecisionLogicNormal::CngOperation(Modes prev_mode,
>-                                             uint32_t target_timestamp,
>-                                             uint32_t available_timestamp,
>-                                             size_t generated_noise_samples) {
>-  // Signed difference between target and available timestamp.
>-  int32_t timestamp_diff = static_cast<int32_t>(
>-      static_cast<uint32_t>(generated_noise_samples + target_timestamp) -
>-      available_timestamp);
>-  int32_t optimal_level_samp = static_cast<int32_t>(
>-      (delay_manager_->TargetLevel() * packet_length_samples_) >> 8);
>-  const int64_t excess_waiting_time_samp =
>-      -static_cast<int64_t>(timestamp_diff) - optimal_level_samp;
>-
>-  if (excess_waiting_time_samp > optimal_level_samp / 2) {
>-    // The waiting time for this packet will be longer than 1.5
>-    // times the wanted buffer delay. Apply fast-forward to cut the
>-    // waiting time down to the optimal.
>-    noise_fast_forward_ = rtc::dchecked_cast<size_t>(noise_fast_forward_ +
>-                                                     excess_waiting_time_samp);
>-    timestamp_diff =
>-        rtc::saturated_cast<int32_t>(timestamp_diff + excess_waiting_time_samp);
>-  }
>-
>-  if (timestamp_diff < 0 && prev_mode == kModeRfc3389Cng) {
>-    // Not time to play this packet yet. Wait another round before using this
>-    // packet. Keep on playing CNG from previous CNG parameters.
>-    return kRfc3389CngNoPacket;
>-  } else {
>-    // Otherwise, go for the CNG packet now.
>-    noise_fast_forward_ = 0;
>-    return kRfc3389Cng;
>-  }
>-}
>-
>-Operations DecisionLogicNormal::NoPacket(bool play_dtmf) {
>-  if (cng_state_ == kCngRfc3389On) {
>-    // Keep on playing comfort noise.
>-    return kRfc3389CngNoPacket;
>-  } else if (cng_state_ == kCngInternalOn) {
>-    // Keep on playing codec internal comfort noise.
>-    return kCodecInternalCng;
>-  } else if (play_dtmf) {
>-    return kDtmf;
>-  } else {
>-    // Nothing to play, do expand.
>-    return kExpand;
>-  }
>-}
>-
>-Operations DecisionLogicNormal::ExpectedPacketAvailable(Modes prev_mode,
>-                                                        bool play_dtmf) {
>-  if (prev_mode != kModeExpand && !play_dtmf) {
>-    // Check criterion for time-stretching.
>-    int low_limit, high_limit;
>-    delay_manager_->BufferLimits(&low_limit, &high_limit);
>-    if (buffer_level_filter_->filtered_current_level() >= high_limit << 2)
>-      return kFastAccelerate;
>-    if (TimescaleAllowed()) {
>-      if (buffer_level_filter_->filtered_current_level() >= high_limit)
>-        return kAccelerate;
>-      if (buffer_level_filter_->filtered_current_level() < low_limit)
>-        return kPreemptiveExpand;
>-    }
>-  }
>-  return kNormal;
>-}
>-
>-Operations DecisionLogicNormal::FuturePacketAvailable(
>-    const SyncBuffer& sync_buffer,
>-    const Expand& expand,
>-    size_t decoder_frame_length,
>-    Modes prev_mode,
>-    uint32_t target_timestamp,
>-    uint32_t available_timestamp,
>-    bool play_dtmf,
>-    size_t generated_noise_samples) {
>-  // Required packet is not available, but a future packet is.
>-  // Check if we should continue with an ongoing expand because the new packet
>-  // is too far into the future.
>-  uint32_t timestamp_leap = available_timestamp - target_timestamp;
>-  if ((prev_mode == kModeExpand) &&
>-      !ReinitAfterExpands(timestamp_leap) &&
>-      !MaxWaitForPacket() &&
>-      PacketTooEarly(timestamp_leap) &&
>-      UnderTargetLevel()) {
>-    if (play_dtmf) {
>-      // Still have DTMF to play, so do not do expand.
>-      return kDtmf;
>-    } else {
>-      // Nothing to play.
>-      return kExpand;
>-    }
>-  }
>-
>-  const size_t samples_left =
>-      sync_buffer.FutureLength() - expand.overlap_length();
>-  const size_t cur_size_samples = samples_left +
>-      packet_buffer_.NumPacketsInBuffer() * decoder_frame_length;
>-
>-  // If previous was comfort noise, then no merge is needed.
>-  if (prev_mode == kModeRfc3389Cng ||
>-      prev_mode == kModeCodecInternalCng) {
>-    // Keep the same delay as before the CNG, but make sure that the number of
>-    // samples in buffer is no higher than 4 times the optimal level. (Note that
>-    // TargetLevel() is in Q8.)
>-    if (static_cast<uint32_t>(generated_noise_samples + target_timestamp) >=
>-            available_timestamp ||
>-        cur_size_samples >
>-            ((delay_manager_->TargetLevel() * packet_length_samples_) >> 8) *
>-            4) {
>-      // Time to play this new packet.
>-      return kNormal;
>-    } else {
>-      // Too early to play this new packet; keep on playing comfort noise.
>-      if (prev_mode == kModeRfc3389Cng) {
>-        return kRfc3389CngNoPacket;
>-      } else {  // prevPlayMode == kModeCodecInternalCng.
>-        return kCodecInternalCng;
>-      }
>-    }
>-  }
>-  // Do not merge unless we have done an expand before.
>-  if (prev_mode == kModeExpand) {
>-    return kMerge;
>-  } else if (play_dtmf) {
>-    // Play DTMF instead of expand.
>-    return kDtmf;
>-  } else {
>-    return kExpand;
>-  }
>-}
>-
>-bool DecisionLogicNormal::UnderTargetLevel() const {
>-  return buffer_level_filter_->filtered_current_level() <=
>-      delay_manager_->TargetLevel();
>-}
>-
>-bool DecisionLogicNormal::ReinitAfterExpands(uint32_t timestamp_leap) const {
>-  return timestamp_leap >=
>-      static_cast<uint32_t>(output_size_samples_ * kReinitAfterExpands);
>-}
>-
>-bool DecisionLogicNormal::PacketTooEarly(uint32_t timestamp_leap) const {
>-  return timestamp_leap >
>-      static_cast<uint32_t>(output_size_samples_ * num_consecutive_expands_);
>-}
>-
>-bool DecisionLogicNormal::MaxWaitForPacket() const {
>-  return num_consecutive_expands_ >= kMaxWaitForPacket;
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/decision_logic_normal.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/decision_logic_normal.h
>deleted file mode 100644
>index 366d10380d0..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/decision_logic_normal.h
>+++ /dev/null
>@@ -1,107 +0,0 @@
>-/*
>- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_NORMAL_H_
>-#define MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_NORMAL_H_
>-
>-#include "modules/audio_coding/neteq/decision_logic.h"
>-#include "rtc_base/constructormagic.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>-
>-namespace webrtc {
>-
>-// Implementation of the DecisionLogic class for playout modes kPlayoutOn and
>-// kPlayoutStreaming.
>-class DecisionLogicNormal : public DecisionLogic {
>- public:
>-  // Constructor.
>-  DecisionLogicNormal(int fs_hz,
>-                      size_t output_size_samples,
>-                      NetEqPlayoutMode playout_mode,
>-                      DecoderDatabase* decoder_database,
>-                      const PacketBuffer& packet_buffer,
>-                      DelayManager* delay_manager,
>-                      BufferLevelFilter* buffer_level_filter,
>-                      const TickTimer* tick_timer)
>-      : DecisionLogic(fs_hz,
>-                      output_size_samples,
>-                      playout_mode,
>-                      decoder_database,
>-                      packet_buffer,
>-                      delay_manager,
>-                      buffer_level_filter,
>-                      tick_timer) {}
>-
>- protected:
>-  static const int kReinitAfterExpands = 100;
>-  static const int kMaxWaitForPacket = 10;
>-
>-  Operations GetDecisionSpecialized(const SyncBuffer& sync_buffer,
>-                                    const Expand& expand,
>-                                    size_t decoder_frame_length,
>-                                    const Packet* next_packet,
>-                                    Modes prev_mode,
>-                                    bool play_dtmf,
>-                                    bool* reset_decoder,
>-                                    size_t generated_noise_samples) override;
>-
>-  // Returns the operation to do given that the expected packet is not
>-  // available, but a packet further into the future is at hand.
>-  virtual Operations FuturePacketAvailable(
>-      const SyncBuffer& sync_buffer,
>-      const Expand& expand,
>-      size_t decoder_frame_length,
>-      Modes prev_mode,
>-      uint32_t target_timestamp,
>-      uint32_t available_timestamp,
>-      bool play_dtmf,
>-      size_t generated_noise_samples);
>-
>-  // Returns the operation to do given that the expected packet is available.
>-  virtual Operations ExpectedPacketAvailable(Modes prev_mode, bool play_dtmf);
>-
>-  // Returns the operation given that no packets are available (except maybe
>-  // a DTMF event, flagged by setting |play_dtmf| true).
>-  virtual Operations NoPacket(bool play_dtmf);
>-
>- private:
>-  // Returns the operation given that the next available packet is a comfort
>-  // noise payload (RFC 3389 only, not codec-internal).
>-  Operations CngOperation(Modes prev_mode,
>-                          uint32_t target_timestamp,
>-                          uint32_t available_timestamp,
>-                          size_t generated_noise_samples);
>-
>-  // Checks if enough time has elapsed since the last successful timescale
>-  // operation was done (i.e., accelerate or preemptive expand).
>-  bool TimescaleAllowed() const {
>-    return !timescale_countdown_ || timescale_countdown_->Finished();
>-  }
>-
>-  // Checks if the current (filtered) buffer level is under the target level.
>-  bool UnderTargetLevel() const;
>-
>-  // Checks if |timestamp_leap| is so long into the future that a reset due
>-  // to exceeding kReinitAfterExpands will be done.
>-  bool ReinitAfterExpands(uint32_t timestamp_leap) const;
>-
>-  // Checks if we still have not done enough expands to cover the distance from
>-  // the last decoded packet to the next available packet, the distance beeing
>-  // conveyed in |timestamp_leap|.
>-  bool PacketTooEarly(uint32_t timestamp_leap) const;
>-
>-  // Checks if num_consecutive_expands_ >= kMaxWaitForPacket.
>-  bool MaxWaitForPacket() const;
>-
>-  RTC_DISALLOW_COPY_AND_ASSIGN(DecisionLogicNormal);
>-};
>-
>-}  // namespace webrtc
>-#endif  // MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_NORMAL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/decision_logic_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/decision_logic_unittest.cc
>index 1a7bab93da0..6929daadfff 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/decision_logic_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/decision_logic_unittest.cc
>@@ -26,26 +26,14 @@ TEST(DecisionLogic, CreateAndDestroy) {
>   int fs_hz = 8000;
>   int output_size_samples = fs_hz / 100;  // Samples per 10 ms.
>   DecoderDatabase decoder_database(
>-      new rtc::RefCountedObject<MockAudioDecoderFactory>);
>+      new rtc::RefCountedObject<MockAudioDecoderFactory>, absl::nullopt);
>   TickTimer tick_timer;
>   PacketBuffer packet_buffer(10, &tick_timer);
>   DelayPeakDetector delay_peak_detector(&tick_timer);
>   DelayManager delay_manager(240, &delay_peak_detector, &tick_timer);
>   BufferLevelFilter buffer_level_filter;
>   DecisionLogic* logic = DecisionLogic::Create(
>-      fs_hz, output_size_samples, kPlayoutOn, &decoder_database, packet_buffer,
>-      &delay_manager, &buffer_level_filter, &tick_timer);
>-  delete logic;
>-  logic = DecisionLogic::Create(
>-      fs_hz, output_size_samples, kPlayoutStreaming, &decoder_database,
>-      packet_buffer, &delay_manager, &buffer_level_filter, &tick_timer);
>-  delete logic;
>-  logic = DecisionLogic::Create(
>-      fs_hz, output_size_samples, kPlayoutFax, &decoder_database, packet_buffer,
>-      &delay_manager, &buffer_level_filter, &tick_timer);
>-  delete logic;
>-  logic = DecisionLogic::Create(
>-      fs_hz, output_size_samples, kPlayoutOff, &decoder_database, packet_buffer,
>+      fs_hz, output_size_samples, false, &decoder_database, packet_buffer,
>       &delay_manager, &buffer_level_filter, &tick_timer);
>   delete logic;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/decoder_database.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/decoder_database.cc
>index d289b45cc04..1fd8c03ca24 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/decoder_database.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/decoder_database.cc
>@@ -15,40 +15,51 @@
> #include "api/audio_codecs/audio_decoder.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>+#include "rtc_base/strings/audio_format_to_string.h"
> 
> namespace webrtc {
> 
> DecoderDatabase::DecoderDatabase(
>-    const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory)
>+    const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory,
>+    absl::optional<AudioCodecPairId> codec_pair_id)
>     : active_decoder_type_(-1),
>       active_cng_decoder_type_(-1),
>-      decoder_factory_(decoder_factory) {}
>+      decoder_factory_(decoder_factory),
>+      codec_pair_id_(codec_pair_id) {}
> 
> DecoderDatabase::~DecoderDatabase() = default;
> 
>-DecoderDatabase::DecoderInfo::DecoderInfo(const SdpAudioFormat& audio_format,
>-                                          AudioDecoderFactory* factory,
>-                                          const std::string& codec_name)
>+DecoderDatabase::DecoderInfo::DecoderInfo(
>+    const SdpAudioFormat& audio_format,
>+    absl::optional<AudioCodecPairId> codec_pair_id,
>+    AudioDecoderFactory* factory,
>+    const std::string& codec_name)
>     : name_(codec_name),
>       audio_format_(audio_format),
>+      codec_pair_id_(codec_pair_id),
>       factory_(factory),
>       external_decoder_(nullptr),
>       cng_decoder_(CngDecoder::Create(audio_format)),
>       subtype_(SubtypeFromFormat(audio_format)) {}
> 
>-DecoderDatabase::DecoderInfo::DecoderInfo(const SdpAudioFormat& audio_format,
>-                                          AudioDecoderFactory* factory)
>-    : DecoderInfo(audio_format, factory, audio_format.name) {}
>+DecoderDatabase::DecoderInfo::DecoderInfo(
>+    const SdpAudioFormat& audio_format,
>+    absl::optional<AudioCodecPairId> codec_pair_id,
>+    AudioDecoderFactory* factory)
>+    : DecoderInfo(audio_format, codec_pair_id, factory, audio_format.name) {}
> 
>-DecoderDatabase::DecoderInfo::DecoderInfo(NetEqDecoder ct,
>-                                          AudioDecoderFactory* factory)
>-    : DecoderInfo(*NetEqDecoderToSdpAudioFormat(ct), factory) {}
>+DecoderDatabase::DecoderInfo::DecoderInfo(
>+    NetEqDecoder ct,
>+    absl::optional<AudioCodecPairId> codec_pair_id,
>+    AudioDecoderFactory* factory)
>+    : DecoderInfo(*NetEqDecoderToSdpAudioFormat(ct), codec_pair_id, factory) {}
> 
> DecoderDatabase::DecoderInfo::DecoderInfo(const SdpAudioFormat& audio_format,
>                                           AudioDecoder* ext_dec,
>                                           const std::string& codec_name)
>     : name_(codec_name),
>       audio_format_(audio_format),
>+      codec_pair_id_(absl::nullopt),
>       factory_(nullptr),
>       external_decoder_(ext_dec),
>       subtype_(Subtype::kNormal) {
>@@ -83,9 +94,9 @@ AudioDecoder* DecoderDatabase::DecoderInfo::GetDecoder() const {
>     // TODO(ossu): Keep a check here for now, since a number of tests create
>     // DecoderInfos without factories.
>     RTC_DCHECK(factory_);
>-    decoder_ = factory_->MakeAudioDecoder(audio_format_);
>+    decoder_ = factory_->MakeAudioDecoder(audio_format_, codec_pair_id_);
>   }
>-  RTC_DCHECK(decoder_) << "Failed to create: " << audio_format_;
>+  RTC_DCHECK(decoder_) << "Failed to create: " << rtc::ToString(audio_format_);
>   return decoder_.get();
> }
> 
>@@ -97,7 +108,7 @@ bool DecoderDatabase::DecoderInfo::IsType(const std::string& name) const {
>   return IsType(name.c_str());
> }
> 
>-rtc::Optional<DecoderDatabase::DecoderInfo::CngDecoder>
>+absl::optional<DecoderDatabase::DecoderInfo::CngDecoder>
> DecoderDatabase::DecoderInfo::CngDecoder::Create(const SdpAudioFormat& format) {
>   if (STR_CASE_CMP(format.name.c_str(), "CN") == 0) {
>     // CN has a 1:1 RTP clock rate to sample rate ratio.
>@@ -106,7 +117,7 @@ DecoderDatabase::DecoderInfo::CngDecoder::Create(const SdpAudioFormat& format) {
>                sample_rate_hz == 32000 || sample_rate_hz == 48000);
>     return DecoderDatabase::DecoderInfo::CngDecoder{sample_rate_hz};
>   } else {
>-    return rtc::nullopt;
>+    return absl::nullopt;
>   }
> }
> 
>@@ -123,9 +134,13 @@ DecoderDatabase::DecoderInfo::SubtypeFromFormat(const SdpAudioFormat& format) {
>   return Subtype::kNormal;
> }
> 
>-bool DecoderDatabase::Empty() const { return decoders_.empty(); }
>+bool DecoderDatabase::Empty() const {
>+  return decoders_.empty();
>+}
> 
>-int DecoderDatabase::Size() const { return static_cast<int>(decoders_.size()); }
>+int DecoderDatabase::Size() const {
>+  return static_cast<int>(decoders_.size());
>+}
> 
> void DecoderDatabase::Reset() {
>   decoders_.clear();
>@@ -156,7 +171,8 @@ std::vector<int> DecoderDatabase::SetCodecs(
>     RTC_DCHECK_LE(rtp_payload_type, 0x7f);
>     if (decoders_.count(rtp_payload_type) == 0) {
>       decoders_.insert(std::make_pair(
>-          rtp_payload_type, DecoderInfo(audio_format, decoder_factory_.get())));
>+          rtp_payload_type,
>+          DecoderInfo(audio_format, codec_pair_id_, decoder_factory_.get())));
>     } else {
>       // The mapping for this payload type hasn't changed.
>     }
>@@ -178,7 +194,7 @@ int DecoderDatabase::RegisterPayload(uint8_t rtp_payload_type,
>   if (!opt_format) {
>     return kCodecNotSupported;
>   }
>-  DecoderInfo info(*opt_format, decoder_factory_, name);
>+  DecoderInfo info(*opt_format, codec_pair_id_, decoder_factory_, name);
>   if (!info.CanGetDecoder()) {
>     return kCodecNotSupported;
>   }
>@@ -197,7 +213,8 @@ int DecoderDatabase::RegisterPayload(int rtp_payload_type,
>     return kInvalidRtpPayloadType;
>   }
>   const auto ret = decoders_.insert(std::make_pair(
>-      rtp_payload_type, DecoderInfo(audio_format, decoder_factory_.get())));
>+      rtp_payload_type,
>+      DecoderInfo(audio_format, codec_pair_id_, decoder_factory_.get())));
>   if (ret.second == false) {
>     // Database already contains a decoder with type |rtp_payload_type|.
>     return kDecoderExists;
>@@ -217,7 +234,8 @@ int DecoderDatabase::InsertExternal(uint8_t rtp_payload_type,
>   }
> 
>   const auto opt_db_format = NetEqDecoderToSdpAudioFormat(codec_type);
>-  const SdpAudioFormat format = opt_db_format.value_or({"arbitrary", 0, 0});
>+  const SdpAudioFormat format =
>+      opt_db_format.value_or(SdpAudioFormat("arbitrary", 0, 0));
> 
>   std::pair<DecoderMap::iterator, bool> ret;
>   DecoderInfo info(format, decoder, codec_name);
>@@ -262,7 +280,7 @@ const DecoderDatabase::DecoderInfo* DecoderDatabase::GetDecoderInfo(
> int DecoderDatabase::SetActiveDecoder(uint8_t rtp_payload_type,
>                                       bool* new_decoder) {
>   // Check that |rtp_payload_type| exists in the database.
>-  const DecoderInfo *info = GetDecoderInfo(rtp_payload_type);
>+  const DecoderInfo* info = GetDecoderInfo(rtp_payload_type);
>   if (!info) {
>     // Decoder not found.
>     return kDecoderNotFound;
>@@ -275,7 +293,7 @@ int DecoderDatabase::SetActiveDecoder(uint8_t rtp_payload_type,
>     *new_decoder = true;
>   } else if (active_decoder_type_ != rtp_payload_type) {
>     // Moving from one active decoder to another. Delete the first one.
>-    const DecoderInfo *old_info = GetDecoderInfo(active_decoder_type_);
>+    const DecoderInfo* old_info = GetDecoderInfo(active_decoder_type_);
>     RTC_DCHECK(old_info);
>     old_info->DropDecoder();
>     *new_decoder = true;
>@@ -294,7 +312,7 @@ AudioDecoder* DecoderDatabase::GetActiveDecoder() const {
> 
> int DecoderDatabase::SetActiveCngDecoder(uint8_t rtp_payload_type) {
>   // Check that |rtp_payload_type| exists in the database.
>-  const DecoderInfo *info = GetDecoderInfo(rtp_payload_type);
>+  const DecoderInfo* info = GetDecoderInfo(rtp_payload_type);
>   if (!info) {
>     // Decoder not found.
>     return kDecoderNotFound;
>@@ -321,7 +339,7 @@ ComfortNoiseDecoder* DecoderDatabase::GetActiveCngDecoder() const {
> }
> 
> AudioDecoder* DecoderDatabase::GetDecoder(uint8_t rtp_payload_type) const {
>-  const DecoderInfo *info = GetDecoderInfo(rtp_payload_type);
>+  const DecoderInfo* info = GetDecoderInfo(rtp_payload_type);
>   return info ? info->GetDecoder() : nullptr;
> }
> 
>@@ -336,17 +354,17 @@ bool DecoderDatabase::IsType(uint8_t rtp_payload_type,
> }
> 
> bool DecoderDatabase::IsComfortNoise(uint8_t rtp_payload_type) const {
>-  const DecoderInfo *info = GetDecoderInfo(rtp_payload_type);
>+  const DecoderInfo* info = GetDecoderInfo(rtp_payload_type);
>   return info && info->IsComfortNoise();
> }
> 
> bool DecoderDatabase::IsDtmf(uint8_t rtp_payload_type) const {
>-  const DecoderInfo *info = GetDecoderInfo(rtp_payload_type);
>+  const DecoderInfo* info = GetDecoderInfo(rtp_payload_type);
>   return info && info->IsDtmf();
> }
> 
> bool DecoderDatabase::IsRed(uint8_t rtp_payload_type) const {
>-  const DecoderInfo *info = GetDecoderInfo(rtp_payload_type);
>+  const DecoderInfo* info = GetDecoderInfo(rtp_payload_type);
>   return info && info->IsRed();
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/decoder_database.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/decoder_database.h
>index 5f0d173806b..131e769079f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/decoder_database.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/decoder_database.h
>@@ -23,7 +23,6 @@
> #include "modules/audio_coding/neteq/packet.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/scoped_ref_ptr.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -43,11 +42,14 @@ class DecoderDatabase {
>   class DecoderInfo {
>    public:
>     DecoderInfo(const SdpAudioFormat& audio_format,
>+                absl::optional<AudioCodecPairId> codec_pair_id,
>                 AudioDecoderFactory* factory,
>                 const std::string& codec_name);
>     explicit DecoderInfo(const SdpAudioFormat& audio_format,
>+                         absl::optional<AudioCodecPairId> codec_pair_id,
>                          AudioDecoderFactory* factory = nullptr);
>     explicit DecoderInfo(NetEqDecoder ct,
>+                         absl::optional<AudioCodecPairId> codec_pair_id,
>                          AudioDecoderFactory* factory = nullptr);
>     DecoderInfo(const SdpAudioFormat& audio_format,
>                 AudioDecoder* ext_dec,
>@@ -85,14 +87,10 @@ class DecoderDatabase {
>     }
> 
>     // Returns true if the decoder's format is DTMF.
>-    bool IsDtmf() const {
>-      return subtype_ == Subtype::kDtmf;
>-    }
>+    bool IsDtmf() const { return subtype_ == Subtype::kDtmf; }
> 
>     // Returns true if the decoder's format is RED.
>-    bool IsRed() const {
>-      return subtype_ == Subtype::kRed;
>-    }
>+    bool IsRed() const { return subtype_ == Subtype::kRed; }
> 
>     // Returns true if the decoder's format is named |name|.
>     bool IsType(const char* name) const;
>@@ -108,6 +106,7 @@ class DecoderDatabase {
>     const std::string name_;
> 
>     const SdpAudioFormat audio_format_;
>+    const absl::optional<AudioCodecPairId> codec_pair_id_;
>     AudioDecoderFactory* const factory_;
>     mutable std::unique_ptr<AudioDecoder> decoder_;
> 
>@@ -116,17 +115,12 @@ class DecoderDatabase {
> 
>     // Set iff this is a comfort noise decoder.
>     struct CngDecoder {
>-      static rtc::Optional<CngDecoder> Create(const SdpAudioFormat& format);
>+      static absl::optional<CngDecoder> Create(const SdpAudioFormat& format);
>       int sample_rate_hz;
>     };
>-    const rtc::Optional<CngDecoder> cng_decoder_;
>+    const absl::optional<CngDecoder> cng_decoder_;
> 
>-    enum class Subtype : int8_t {
>-      kNormal,
>-      kComfortNoise,
>-      kDtmf,
>-      kRed
>-    };
>+    enum class Subtype : int8_t { kNormal, kComfortNoise, kDtmf, kRed };
> 
>     static Subtype SubtypeFromFormat(const SdpAudioFormat& format);
> 
>@@ -138,7 +132,8 @@ class DecoderDatabase {
>   static const uint8_t kRtpPayloadTypeError = 0xFF;
> 
>   DecoderDatabase(
>-      const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory);
>+      const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory,
>+      absl::optional<AudioCodecPairId> codec_pair_id);
> 
>   virtual ~DecoderDatabase();
> 
>@@ -242,6 +237,7 @@ class DecoderDatabase {
>   int active_cng_decoder_type_;
>   mutable std::unique_ptr<ComfortNoiseDecoder> active_cng_decoder_;
>   rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
>+  const absl::optional<AudioCodecPairId> codec_pair_id_;
> 
>   RTC_DISALLOW_COPY_AND_ASSIGN(DecoderDatabase);
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/decoder_database_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/decoder_database_unittest.cc
>index 53fbadf116c..10043e0bba6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/decoder_database_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/decoder_database_unittest.cc
>@@ -28,7 +28,8 @@ using testing::Invoke;
> namespace webrtc {
> 
> TEST(DecoderDatabase, CreateAndDestroy) {
>-  DecoderDatabase db(new rtc::RefCountedObject<MockAudioDecoderFactory>);
>+  DecoderDatabase db(new rtc::RefCountedObject<MockAudioDecoderFactory>,
>+                     absl::nullopt);
>   EXPECT_EQ(0, db.Size());
>   EXPECT_TRUE(db.Empty());
> }
>@@ -41,7 +42,7 @@ TEST(DecoderDatabase, InsertAndRemove) {
>         EXPECT_EQ("pcmu", format.name);
>         return true;
>       }));
>-  DecoderDatabase db(factory);
>+  DecoderDatabase db(factory, absl::nullopt);
>   const uint8_t kPayloadType = 0;
>   const std::string kCodecName = "Robert\'); DROP TABLE Students;";
>   EXPECT_EQ(
>@@ -66,7 +67,7 @@ TEST(DecoderDatabase, InsertAndRemoveAll) {
>         EXPECT_EQ("pcma", format.name);
>         return true;
>       }));
>-  DecoderDatabase db(factory);
>+  DecoderDatabase db(factory, absl::nullopt);
>   const std::string kCodecName1 = "Robert\'); DROP TABLE Students;";
>   const std::string kCodecName2 = "https://xkcd.com/327/";
>   EXPECT_EQ(DecoderDatabase::kOK,
>@@ -89,13 +90,14 @@ TEST(DecoderDatabase, GetDecoderInfo) {
>         return true;
>       }));
>   auto* decoder = new MockAudioDecoder;
>-  EXPECT_CALL(*factory, MakeAudioDecoderMock(_, _))
>+  EXPECT_CALL(*factory, MakeAudioDecoderMock(_, _, _))
>       .WillOnce(Invoke([decoder](const SdpAudioFormat& format,
>+                                 absl::optional<AudioCodecPairId> codec_pair_id,
>                                  std::unique_ptr<AudioDecoder>* dec) {
>         EXPECT_EQ("pcmu", format.name);
>         dec->reset(decoder);
>       }));
>-  DecoderDatabase db(factory);
>+  DecoderDatabase db(factory, absl::nullopt);
>   const uint8_t kPayloadType = 0;
>   const std::string kCodecName = "Robert\'); DROP TABLE Students;";
>   EXPECT_EQ(
>@@ -108,11 +110,11 @@ TEST(DecoderDatabase, GetDecoderInfo) {
>   EXPECT_EQ(kCodecName, info->get_name());
>   EXPECT_EQ(decoder, db.GetDecoder(kPayloadType));
>   info = db.GetDecoderInfo(kPayloadType + 1);  // Other payload type.
>-  EXPECT_TRUE(info == NULL);  // Should not be found.
>+  EXPECT_TRUE(info == NULL);                   // Should not be found.
> }
> 
> TEST(DecoderDatabase, GetDecoder) {
>-  DecoderDatabase db(CreateBuiltinAudioDecoderFactory());
>+  DecoderDatabase db(CreateBuiltinAudioDecoderFactory(), absl::nullopt);
>   const uint8_t kPayloadType = 0;
>   const std::string kCodecName = "Robert\'); DROP TABLE Students;";
>   EXPECT_EQ(DecoderDatabase::kOK,
>@@ -130,7 +132,7 @@ TEST(DecoderDatabase, TypeTests) {
>         EXPECT_EQ("pcmu", format.name);
>         return true;
>       }));
>-  DecoderDatabase db(factory);
>+  DecoderDatabase db(factory, absl::nullopt);
>   const uint8_t kPayloadTypePcmU = 0;
>   const uint8_t kPayloadTypeCng = 13;
>   const uint8_t kPayloadTypeDtmf = 100;
>@@ -165,7 +167,8 @@ TEST(DecoderDatabase, TypeTests) {
> }
> 
> TEST(DecoderDatabase, ExternalDecoder) {
>-  DecoderDatabase db(new rtc::RefCountedObject<MockAudioDecoderFactory>);
>+  DecoderDatabase db(new rtc::RefCountedObject<MockAudioDecoderFactory>,
>+                     absl::nullopt);
>   const uint8_t kPayloadType = 0;
>   const std::string kCodecName = "Robert\'); DROP TABLE Students;";
>   MockAudioDecoder decoder;
>@@ -202,7 +205,7 @@ TEST(DecoderDatabase, CheckPayloadTypes) {
>         EXPECT_EQ("pcmu", format.name);
>         return true;
>       }));
>-  DecoderDatabase db(factory);
>+  DecoderDatabase db(factory, absl::nullopt);
>   // Load a number of payloads into the database. Payload types are 0, 1, ...,
>   // while the decoder type is the same for all payload types (this does not
>   // matter for the test).
>@@ -242,7 +245,7 @@ TEST(DecoderDatabase, CheckPayloadTypes) {
> 
> // Test the methods for setting and getting active speech and CNG decoders.
> TEST(DecoderDatabase, IF_ISAC(ActiveDecoders)) {
>-  DecoderDatabase db(CreateBuiltinAudioDecoderFactory());
>+  DecoderDatabase db(CreateBuiltinAudioDecoderFactory(), absl::nullopt);
>   // Load payload types.
>   ASSERT_EQ(DecoderDatabase::kOK,
>             db.RegisterPayload(0, NetEqDecoder::kDecoderPCMu, "pcmu"));
>@@ -289,7 +292,6 @@ TEST(DecoderDatabase, IF_ISAC(ActiveDecoders)) {
>   // Try to set non-existing codecs as active.
>   EXPECT_EQ(DecoderDatabase::kDecoderNotFound,
>             db.SetActiveDecoder(17, &changed));
>-  EXPECT_EQ(DecoderDatabase::kDecoderNotFound,
>-            db.SetActiveCngDecoder(17));
>+  EXPECT_EQ(DecoderDatabase::kDecoderNotFound, db.SetActiveCngDecoder(17));
> }
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/defines.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/defines.h
>index 496a36dfa57..768f0b955e1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/defines.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/defines.h
>@@ -24,10 +24,6 @@ enum Operations {
>   kRfc3389CngNoPacket,
>   kCodecInternalCng,
>   kDtmf,
>-  kAlternativePlc,
>-  kAlternativePlcIncreaseTimestamp,
>-  kAudioRepetition,
>-  kAudioRepetitionIncreaseTimestamp,
>   kUndefined = -1
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/delay_manager.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/delay_manager.cc
>index b70131d9ed4..a945cdcad84 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/delay_manager.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/delay_manager.cc
>@@ -155,8 +155,9 @@ void DelayManager::UpdateCumulativeSums(int packet_len_ms,
>       (packet_iat_stopwatch_->ElapsedMs() << 8) / packet_len_ms;
>   // Calculate cumulative sum IAT with sequence number compensation. The sum
>   // is zero if there is no clock-drift.
>-  iat_cumulative_sum_ += (iat_packets_q8 -
>-      (static_cast<int>(sequence_number - last_seq_no_) << 8));
>+  iat_cumulative_sum_ +=
>+      (iat_packets_q8 -
>+       (static_cast<int>(sequence_number - last_seq_no_) << 8));
>   // Subtract drift term.
>   iat_cumulative_sum_ -= kCumulativeSumDrift;
>   // Ensure not negative.
>@@ -189,8 +190,8 @@ void DelayManager::UpdateHistogram(size_t iat_packets) {
>   assert(iat_packets < iat_vector_.size());
>   int vector_sum = 0;  // Sum up the vector elements as they are processed.
>   // Multiply each element in |iat_vector_| with |iat_factor_|.
>-  for (IATVector::iterator it = iat_vector_.begin();
>-      it != iat_vector_.end(); ++it) {
>+  for (IATVector::iterator it = iat_vector_.begin(); it != iat_vector_.end();
>+       ++it) {
>     *it = (static_cast<int64_t>(*it) * iat_factor_) >> 15;
>     vector_sum += *it;
>   }
>@@ -236,7 +237,7 @@ void DelayManager::LimitTargetLevel() {
>   least_required_delay_ms_ = (target_level_ * packet_len_ms_) >> 8;
> 
>   if (packet_len_ms_ > 0 && minimum_delay_ms_ > 0) {
>-    int minimum_delay_packet_q8 =  (minimum_delay_ms_ << 8) / packet_len_ms_;
>+    int minimum_delay_packet_q8 = (minimum_delay_ms_ << 8) / packet_len_ms_;
>     target_level_ = std::max(target_level_, minimum_delay_packet_q8);
>   }
> 
>@@ -269,8 +270,8 @@ int DelayManager::CalculateTargetLevel(int iat_packets) {
>   // (in Q30) by definition, and since the solution is often a low value for
>   // |iat_index|, it is more efficient to start with |sum| = 1 and subtract
>   // elements from the start of the histogram.
>-  size_t index = 0;  // Start from the beginning of |iat_vector_|.
>-  int sum = 1 << 30;  // Assign to 1 in Q30.
>+  size_t index = 0;           // Start from the beginning of |iat_vector_|.
>+  int sum = 1 << 30;          // Assign to 1 in Q30.
>   sum -= iat_vector_[index];  // Ensure that target level is >= 1.
> 
>   do {
>@@ -313,13 +314,12 @@ int DelayManager::SetPacketAudioLength(int length_ms) {
>   return 0;
> }
> 
>-
> void DelayManager::Reset() {
>   packet_len_ms_ = 0;  // Packet size unknown.
>   streaming_mode_ = false;
>   peak_detector_.Reset();
>   ResetHistogram();  // Resets target levels too.
>-  iat_factor_ = 0;  // Adapt the histogram faster for the first few packets.
>+  iat_factor_ = 0;   // Adapt the histogram faster for the first few packets.
>   packet_iat_stopwatch_ = tick_timer_->GetNewStopwatch();
>   max_iat_stopwatch_ = tick_timer_->GetNewStopwatch();
>   iat_cumulative_sum_ = 0;
>@@ -471,8 +471,12 @@ int DelayManager::least_required_delay_ms() const {
>   return least_required_delay_ms_;
> }
> 
>-int DelayManager::base_target_level() const { return base_target_level_; }
>-void DelayManager::set_streaming_mode(bool value) { streaming_mode_ = value; }
>+int DelayManager::base_target_level() const {
>+  return base_target_level_;
>+}
>+void DelayManager::set_streaming_mode(bool value) {
>+  streaming_mode_ = value;
>+}
> int DelayManager::last_pack_cng_or_dtmf() const {
>   return last_pack_cng_or_dtmf_;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/delay_manager.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/delay_manager.h
>index 0d082c8788f..d75d5b47c02 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/delay_manager.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/delay_manager.h
>@@ -18,7 +18,6 @@
> 
> #include "modules/audio_coding/neteq/tick_timer.h"
> #include "rtc_base/constructormagic.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -117,9 +116,9 @@ class DelayManager {
>   virtual void set_last_pack_cng_or_dtmf(int value);
> 
>  private:
>-  static const int kLimitProbability = 53687091;  // 1/20 in Q30.
>+  static const int kLimitProbability = 53687091;         // 1/20 in Q30.
>   static const int kLimitProbabilityStreaming = 536871;  // 1/2000 in Q30.
>-  static const int kMaxStreamingPeakPeriodMs = 600000;  // 10 minutes in ms.
>+  static const int kMaxStreamingPeakPeriodMs = 600000;   // 10 minutes in ms.
>   static const int kCumulativeSumDrift = 2;  // Drift term for cumulative sum
>                                              // |iat_cumulative_sum_|.
>   // Steady-state forgetting factor for |iat_vector_|, 0.9993 in Q15.
>@@ -146,28 +145,29 @@ class DelayManager {
> 
>   bool first_packet_received_;
>   const size_t max_packets_in_buffer_;  // Capacity of the packet buffer.
>-  IATVector iat_vector_;  // Histogram of inter-arrival times.
>+  IATVector iat_vector_;                // Histogram of inter-arrival times.
>   int iat_factor_;  // Forgetting factor for updating the IAT histogram (Q15).
>   const TickTimer* tick_timer_;
>   // Time elapsed since last packet.
>   std::unique_ptr<TickTimer::Stopwatch> packet_iat_stopwatch_;
>-  int base_target_level_;   // Currently preferred buffer level before peak
>-                            // detection and streaming mode (Q0).
>+  int base_target_level_;  // Currently preferred buffer level before peak
>+                           // detection and streaming mode (Q0).
>   // TODO(turajs) change the comment according to the implementation of
>   // minimum-delay.
>-  int target_level_;  // Currently preferred buffer level in (fractions)
>-                      // of packets (Q8), before adding any extra delay.
>+  int target_level_;   // Currently preferred buffer level in (fractions)
>+                       // of packets (Q8), before adding any extra delay.
>   int packet_len_ms_;  // Length of audio in each incoming packet [ms].
>   bool streaming_mode_;
>-  uint16_t last_seq_no_;  // Sequence number for last received packet.
>-  uint32_t last_timestamp_;  // Timestamp for the last received packet.
>-  int minimum_delay_ms_;  // Externally set minimum delay.
>+  uint16_t last_seq_no_;         // Sequence number for last received packet.
>+  uint32_t last_timestamp_;      // Timestamp for the last received packet.
>+  int minimum_delay_ms_;         // Externally set minimum delay.
>   int least_required_delay_ms_;  // Smallest preferred buffer level (same unit
>-                              // as |target_level_|), before applying
>-                              // |minimum_delay_ms_| and/or |maximum_delay_ms_|.
>-  int maximum_delay_ms_;  // Externally set maximum allowed delay.
>-  int iat_cumulative_sum_;  // Cumulative sum of delta inter-arrival times.
>-  int max_iat_cumulative_sum_;  // Max of |iat_cumulative_sum_|.
>+                                 // as |target_level_|), before applying
>+                                 // |minimum_delay_ms_| and/or
>+                                 // |maximum_delay_ms_|.
>+  int maximum_delay_ms_;         // Externally set maximum allowed delay.
>+  int iat_cumulative_sum_;       // Cumulative sum of delta inter-arrival times.
>+  int max_iat_cumulative_sum_;   // Max of |iat_cumulative_sum_|.
>   // Time elapsed since maximum was observed.
>   std::unique_ptr<TickTimer::Stopwatch> max_iat_stopwatch_;
>   DelayPeakDetector& peak_detector_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/delay_manager_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/delay_manager_unittest.cc
>index 953bc6b4002..f9c56800178 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/delay_manager_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/delay_manager_unittest.cc
>@@ -49,8 +49,7 @@ DelayManagerTest::DelayManagerTest()
>     : dm_(NULL), detector_(&tick_timer_), seq_no_(0x1234), ts_(0x12345678) {}
> 
> void DelayManagerTest::SetUp() {
>-  EXPECT_CALL(detector_, Reset())
>-            .Times(1);
>+  EXPECT_CALL(detector_, Reset()).Times(1);
>   dm_ = new DelayManager(kMaxNumberOfPackets, &detector_, &tick_timer_);
> }
> 
>@@ -94,8 +93,7 @@ TEST_F(DelayManagerTest, VectorInitialization) {
> TEST_F(DelayManagerTest, SetPacketAudioLength) {
>   const int kLengthMs = 30;
>   // Expect DelayManager to pass on the new length to the detector object.
>-  EXPECT_CALL(detector_, SetPacketAudioLength(kLengthMs))
>-      .Times(1);
>+  EXPECT_CALL(detector_, SetPacketAudioLength(kLengthMs)).Times(1);
>   EXPECT_EQ(0, dm_->SetPacketAudioLength(kLengthMs));
>   EXPECT_EQ(-1, dm_->SetPacketAudioLength(-1));  // Illegal parameter value.
> }
>@@ -121,8 +119,7 @@ TEST_F(DelayManagerTest, UpdateNormal) {
>   // Expect detector update method to be called once with inter-arrival time
>   // equal to 1 packet, and (base) target level equal to 1 as well.
>   // Return false to indicate no peaks found.
>-  EXPECT_CALL(detector_, Update(1, 1))
>-      .WillOnce(Return(false));
>+  EXPECT_CALL(detector_, Update(1, 1)).WillOnce(Return(false));
>   InsertNextPacket();
>   EXPECT_EQ(1 << 8, dm_->TargetLevel());  // In Q8.
>   EXPECT_EQ(1, dm_->base_target_level());
>@@ -145,8 +142,7 @@ TEST_F(DelayManagerTest, UpdateLongInterArrivalTime) {
>   // Expect detector update method to be called once with inter-arrival time
>   // equal to 1 packet, and (base) target level equal to 1 as well.
>   // Return false to indicate no peaks found.
>-  EXPECT_CALL(detector_, Update(2, 2))
>-      .WillOnce(Return(false));
>+  EXPECT_CALL(detector_, Update(2, 2)).WillOnce(Return(false));
>   InsertNextPacket();
>   EXPECT_EQ(2 << 8, dm_->TargetLevel());  // In Q8.
>   EXPECT_EQ(2, dm_->base_target_level());
>@@ -169,10 +165,8 @@ TEST_F(DelayManagerTest, UpdatePeakFound) {
>   // Expect detector update method to be called once with inter-arrival time
>   // equal to 1 packet, and (base) target level equal to 1 as well.
>   // Return true to indicate that peaks are found. Let the peak height be 5.
>-  EXPECT_CALL(detector_, Update(1, 1))
>-      .WillOnce(Return(true));
>-  EXPECT_CALL(detector_, MaxPeakHeight())
>-      .WillOnce(Return(5));
>+  EXPECT_CALL(detector_, Update(1, 1)).WillOnce(Return(true));
>+  EXPECT_CALL(detector_, MaxPeakHeight()).WillOnce(Return(5));
>   InsertNextPacket();
>   EXPECT_EQ(5 << 8, dm_->TargetLevel());
>   EXPECT_EQ(1, dm_->base_target_level());  // Base target level is w/o peaks.
>@@ -193,8 +187,7 @@ TEST_F(DelayManagerTest, TargetDelay) {
>   // Expect detector update method to be called once with inter-arrival time
>   // equal to 1 packet, and (base) target level equal to 1 as well.
>   // Return false to indicate no peaks found.
>-  EXPECT_CALL(detector_, Update(1, 1))
>-      .WillOnce(Return(false));
>+  EXPECT_CALL(detector_, Update(1, 1)).WillOnce(Return(false));
>   InsertNextPacket();
>   const int kExpectedTarget = 1;
>   EXPECT_EQ(kExpectedTarget << 8, dm_->TargetLevel());  // In Q8.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/delay_peak_detector_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/delay_peak_detector_unittest.cc
>index 058ba663621..fd4dded6089 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/delay_peak_detector_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/delay_peak_detector_unittest.cc
>@@ -65,8 +65,8 @@ TEST(DelayPeakDetector, TriggerPeakMode) {
>   int next = 1;  // Start with the second packet to get a proper IAT.
>   while (next < kNumPackets) {
>     while (next < kNumPackets && arrival_times_ms[next] <= time) {
>-      int iat_packets = (arrival_times_ms[next] - arrival_times_ms[next - 1]) /
>-          kPacketSizeMs;
>+      int iat_packets =
>+          (arrival_times_ms[next] - arrival_times_ms[next - 1]) / kPacketSizeMs;
>       const int kTargetBufferLevel = 1;  // Define peaks to be iat > 2.
>       if (time < peak_mode_start_ms || time > peak_mode_end_ms) {
>         EXPECT_FALSE(detector.Update(iat_packets, kTargetBufferLevel));
>@@ -112,8 +112,8 @@ TEST(DelayPeakDetector, DoNotTriggerPeakMode) {
>   int next = 1;  // Start with the second packet to get a proper IAT.
>   while (next < kNumPackets) {
>     while (next < kNumPackets && arrival_times_ms[next] <= time) {
>-      int iat_packets = (arrival_times_ms[next] - arrival_times_ms[next - 1]) /
>-          kPacketSizeMs;
>+      int iat_packets =
>+          (arrival_times_ms[next] - arrival_times_ms[next - 1]) / kPacketSizeMs;
>       const int kTargetBufferLevel = 2;  // Define peaks to be iat > 4.
>       EXPECT_FALSE(detector.Update(iat_packets, kTargetBufferLevel));
>       ++next;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/dsp_helper.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/dsp_helper.cc
>index 2a1d81ba0dc..05b0f70bcf2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/dsp_helper.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/dsp_helper.cc
>@@ -21,41 +21,29 @@ namespace webrtc {
> 
> // Table of constants used in method DspHelper::ParabolicFit().
> const int16_t DspHelper::kParabolaCoefficients[17][3] = {
>-    { 120, 32, 64 },
>-    { 140, 44, 75 },
>-    { 150, 50, 80 },
>-    { 160, 57, 85 },
>-    { 180, 72, 96 },
>-    { 200, 89, 107 },
>-    { 210, 98, 112 },
>-    { 220, 108, 117 },
>-    { 240, 128, 128 },
>-    { 260, 150, 139 },
>-    { 270, 162, 144 },
>-    { 280, 174, 149 },
>-    { 300, 200, 160 },
>-    { 320, 228, 171 },
>-    { 330, 242, 176 },
>-    { 340, 257, 181 },
>-    { 360, 288, 192 } };
>+    {120, 32, 64},   {140, 44, 75},   {150, 50, 80},   {160, 57, 85},
>+    {180, 72, 96},   {200, 89, 107},  {210, 98, 112},  {220, 108, 117},
>+    {240, 128, 128}, {260, 150, 139}, {270, 162, 144}, {280, 174, 149},
>+    {300, 200, 160}, {320, 228, 171}, {330, 242, 176}, {340, 257, 181},
>+    {360, 288, 192}};
> 
> // Filter coefficients used when downsampling from the indicated sample rates
> // (8, 16, 32, 48 kHz) to 4 kHz. Coefficients are in Q12. The corresponding Q0
> // values are provided in the comments before each array.
> 
> // Q0 values: {0.3, 0.4, 0.3}.
>-const int16_t DspHelper::kDownsample8kHzTbl[3] = { 1229, 1638, 1229 };
>+const int16_t DspHelper::kDownsample8kHzTbl[3] = {1229, 1638, 1229};
> 
> // Q0 values: {0.15, 0.2, 0.3, 0.2, 0.15}.
>-const int16_t DspHelper::kDownsample16kHzTbl[5] = { 614, 819, 1229, 819, 614 };
>+const int16_t DspHelper::kDownsample16kHzTbl[5] = {614, 819, 1229, 819, 614};
> 
> // Q0 values: {0.1425, 0.1251, 0.1525, 0.1628, 0.1525, 0.1251, 0.1425}.
>-const int16_t DspHelper::kDownsample32kHzTbl[7] = {
>-    584, 512, 625, 667, 625, 512, 584 };
>+const int16_t DspHelper::kDownsample32kHzTbl[7] = {584, 512, 625, 667,
>+                                                   625, 512, 584};
> 
> // Q0 values: {0.2487, 0.0952, 0.1042, 0.1074, 0.1042, 0.0952, 0.2487}.
>-const int16_t DspHelper::kDownsample48kHzTbl[7] = {
>-    1019, 390, 427, 440, 427, 390, 1019 };
>+const int16_t DspHelper::kDownsample48kHzTbl[7] = {1019, 390, 427, 440,
>+                                                   427,  390, 1019};
> 
> int DspHelper::RampSignal(const int16_t* input,
>                           size_t length,
>@@ -115,9 +103,12 @@ int DspHelper::RampSignal(AudioMultiVector* signal,
>   return end_factor;
> }
> 
>-void DspHelper::PeakDetection(int16_t* data, size_t data_length,
>-                              size_t num_peaks, int fs_mult,
>-                              size_t* peak_index, int16_t* peak_value) {
>+void DspHelper::PeakDetection(int16_t* data,
>+                              size_t data_length,
>+                              size_t num_peaks,
>+                              int fs_mult,
>+                              size_t* peak_index,
>+                              int16_t* peak_value) {
>   size_t min_index = 0;
>   size_t max_index = 0;
> 
>@@ -163,8 +154,10 @@ void DspHelper::PeakDetection(int16_t* data, size_t data_length,
>   }
> }
> 
>-void DspHelper::ParabolicFit(int16_t* signal_points, int fs_mult,
>-                             size_t* peak_index, int16_t* peak_value) {
>+void DspHelper::ParabolicFit(int16_t* signal_points,
>+                             int fs_mult,
>+                             size_t* peak_index,
>+                             int16_t* peak_value) {
>   uint16_t fit_index[13];
>   if (fs_mult == 1) {
>     fit_index[0] = 0;
>@@ -204,23 +197,26 @@ void DspHelper::ParabolicFit(int16_t* signal_points, int fs_mult,
> 
>   //  num = -3 * signal_points[0] + 4 * signal_points[1] - signal_points[2];
>   //  den =      signal_points[0] - 2 * signal_points[1] + signal_points[2];
>-  int32_t num = (signal_points[0] * -3) + (signal_points[1] * 4)
>-      - signal_points[2];
>+  int32_t num =
>+      (signal_points[0] * -3) + (signal_points[1] * 4) - signal_points[2];
>   int32_t den = signal_points[0] + (signal_points[1] * -2) + signal_points[2];
>   int32_t temp = num * 120;
>   int flag = 1;
>-  int16_t stp = kParabolaCoefficients[fit_index[fs_mult]][0]
>-      - kParabolaCoefficients[fit_index[fs_mult - 1]][0];
>-  int16_t strt = (kParabolaCoefficients[fit_index[fs_mult]][0]
>-      + kParabolaCoefficients[fit_index[fs_mult - 1]][0]) / 2;
>+  int16_t stp = kParabolaCoefficients[fit_index[fs_mult]][0] -
>+                kParabolaCoefficients[fit_index[fs_mult - 1]][0];
>+  int16_t strt = (kParabolaCoefficients[fit_index[fs_mult]][0] +
>+                  kParabolaCoefficients[fit_index[fs_mult - 1]][0]) /
>+                 2;
>   int16_t lmt;
>   if (temp < -den * strt) {
>     lmt = strt - stp;
>     while (flag) {
>       if ((flag == fs_mult) || (temp > -den * lmt)) {
>-        *peak_value = (den * kParabolaCoefficients[fit_index[fs_mult - flag]][1]
>-            + num * kParabolaCoefficients[fit_index[fs_mult - flag]][2]
>-            + signal_points[0] * 256) / 256;
>+        *peak_value =
>+            (den * kParabolaCoefficients[fit_index[fs_mult - flag]][1] +
>+             num * kParabolaCoefficients[fit_index[fs_mult - flag]][2] +
>+             signal_points[0] * 256) /
>+            256;
>         *peak_index = *peak_index * 2 * fs_mult - flag;
>         flag = 0;
>       } else {
>@@ -233,9 +229,9 @@ void DspHelper::ParabolicFit(int16_t* signal_points, int fs_mult,
>     while (flag) {
>       if ((flag == fs_mult) || (temp < -den * lmt)) {
>         int32_t temp_term_1 =
>-            den * kParabolaCoefficients[fit_index[fs_mult+flag]][1];
>+            den * kParabolaCoefficients[fit_index[fs_mult + flag]][1];
>         int32_t temp_term_2 =
>-            num * kParabolaCoefficients[fit_index[fs_mult+flag]][2];
>+            num * kParabolaCoefficients[fit_index[fs_mult + flag]][2];
>         int32_t temp_term_3 = signal_points[0] * 256;
>         *peak_value = (temp_term_1 + temp_term_2 + temp_term_3) / 256;
>         *peak_index = *peak_index * 2 * fs_mult + flag;
>@@ -251,8 +247,10 @@ void DspHelper::ParabolicFit(int16_t* signal_points, int fs_mult,
>   }
> }
> 
>-size_t DspHelper::MinDistortion(const int16_t* signal, size_t min_lag,
>-                                size_t max_lag, size_t length,
>+size_t DspHelper::MinDistortion(const int16_t* signal,
>+                                size_t min_lag,
>+                                size_t max_lag,
>+                                size_t length,
>                                 int32_t* distortion_value) {
>   size_t best_index = 0;
>   int32_t min_distortion = WEBRTC_SPL_WORD32_MAX;
>@@ -273,9 +271,12 @@ size_t DspHelper::MinDistortion(const int16_t* signal, size_t min_lag,
>   return best_index;
> }
> 
>-void DspHelper::CrossFade(const int16_t* input1, const int16_t* input2,
>-                          size_t length, int16_t* mix_factor,
>-                          int16_t factor_decrement, int16_t* output) {
>+void DspHelper::CrossFade(const int16_t* input1,
>+                          const int16_t* input2,
>+                          size_t length,
>+                          int16_t* mix_factor,
>+                          int16_t factor_decrement,
>+                          int16_t* output) {
>   int16_t factor = *mix_factor;
>   int16_t complement_factor = 16384 - factor;
>   for (size_t i = 0; i < length; i++) {
>@@ -287,8 +288,10 @@ void DspHelper::CrossFade(const int16_t* input1, const int16_t* input2,
>   *mix_factor = factor;
> }
> 
>-void DspHelper::UnmuteSignal(const int16_t* input, size_t length,
>-                             int16_t* factor, int increment,
>+void DspHelper::UnmuteSignal(const int16_t* input,
>+                             size_t length,
>+                             int16_t* factor,
>+                             int increment,
>                              int16_t* output) {
>   uint16_t factor_16b = *factor;
>   int32_t factor_32b = (static_cast<int32_t>(factor_16b) << 6) + 32;
>@@ -308,17 +311,20 @@ void DspHelper::MuteSignal(int16_t* signal, int mute_slope, size_t length) {
>   }
> }
> 
>-int DspHelper::DownsampleTo4kHz(const int16_t* input, size_t input_length,
>-                                size_t output_length, int input_rate_hz,
>-                                bool compensate_delay, int16_t* output) {
>+int DspHelper::DownsampleTo4kHz(const int16_t* input,
>+                                size_t input_length,
>+                                size_t output_length,
>+                                int input_rate_hz,
>+                                bool compensate_delay,
>+                                int16_t* output) {
>   // Set filter parameters depending on input frequency.
>   // NOTE: The phase delay values are wrong compared to the true phase delay
>   // of the filters. However, the error is preserved (through the +1 term) for
>   // consistency.
>   const int16_t* filter_coefficients;  // Filter coefficients.
>-  size_t filter_length;  // Number of coefficients.
>-  size_t filter_delay;  // Phase delay in samples.
>-  int16_t factor;  // Conversion rate (inFsHz / 8000).
>+  size_t filter_length;                // Number of coefficients.
>+  size_t filter_delay;                 // Phase delay in samples.
>+  int16_t factor;                      // Conversion rate (inFsHz / 8000).
>   switch (input_rate_hz) {
>     case 8000: {
>       filter_length = 3;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/dsp_helper.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/dsp_helper.h
>index 7ceb66f57af..efa2f9ce064 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/dsp_helper.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/dsp_helper.h
>@@ -15,7 +15,6 @@
> 
> #include "modules/audio_coding/neteq/audio_multi_vector.h"
> #include "rtc_base/constructormagic.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -85,9 +84,12 @@ class DspHelper {
>   // locations and values are written to the arrays |peak_index| and
>   // |peak_value|, respectively. Both arrays must hold at least |num_peaks|
>   // elements.
>-  static void PeakDetection(int16_t* data, size_t data_length,
>-                            size_t num_peaks, int fs_mult,
>-                            size_t* peak_index, int16_t* peak_value);
>+  static void PeakDetection(int16_t* data,
>+                            size_t data_length,
>+                            size_t num_peaks,
>+                            int fs_mult,
>+                            size_t* peak_index,
>+                            int16_t* peak_value);
> 
>   // Estimates the height and location of a maximum. The three values in the
>   // array |signal_points| are used as basis for a parabolic fit, which is then
>@@ -95,30 +97,40 @@ class DspHelper {
>   // assumed to be from a 4 kHz signal, while the maximum, written to
>   // |peak_index| and |peak_value| is given in the full sample rate, as
>   // indicated by the sample rate multiplier |fs_mult|.
>-  static void ParabolicFit(int16_t* signal_points, int fs_mult,
>-                           size_t* peak_index, int16_t* peak_value);
>+  static void ParabolicFit(int16_t* signal_points,
>+                           int fs_mult,
>+                           size_t* peak_index,
>+                           int16_t* peak_value);
> 
>   // Calculates the sum-abs-diff for |signal| when compared to a displaced
>   // version of itself. Returns the displacement lag that results in the minimum
>   // distortion. The resulting distortion is written to |distortion_value|.
>   // The values of |min_lag| and |max_lag| are boundaries for the search.
>-  static size_t MinDistortion(const int16_t* signal, size_t min_lag,
>-                           size_t max_lag, size_t length,
>-                           int32_t* distortion_value);
>+  static size_t MinDistortion(const int16_t* signal,
>+                              size_t min_lag,
>+                              size_t max_lag,
>+                              size_t length,
>+                              int32_t* distortion_value);
> 
>   // Mixes |length| samples from |input1| and |input2| together and writes the
>   // result to |output|. The gain for |input1| starts at |mix_factor| (Q14) and
>   // is decreased by |factor_decrement| (Q14) for each sample. The gain for
>   // |input2| is the complement 16384 - mix_factor.
>-  static void CrossFade(const int16_t* input1, const int16_t* input2,
>-                        size_t length, int16_t* mix_factor,
>-                        int16_t factor_decrement, int16_t* output);
>+  static void CrossFade(const int16_t* input1,
>+                        const int16_t* input2,
>+                        size_t length,
>+                        int16_t* mix_factor,
>+                        int16_t factor_decrement,
>+                        int16_t* output);
> 
>   // Scales |input| with an increasing gain. Applies |factor| (Q14) to the first
>   // sample and increases the gain by |increment| (Q20) for each sample. The
>   // result is written to |output|. |length| samples are processed.
>-  static void UnmuteSignal(const int16_t* input, size_t length, int16_t* factor,
>-                           int increment, int16_t* output);
>+  static void UnmuteSignal(const int16_t* input,
>+                           size_t length,
>+                           int16_t* factor,
>+                           int increment,
>+                           int16_t* output);
> 
>   // Starts at unity gain and gradually fades out |signal|. For each sample,
>   // the gain is reduced by |mute_slope| (Q14). |length| samples are processed.
>@@ -129,9 +141,12 @@ class DspHelper {
>   // samples to |output|. Compensates for the phase delay of the downsampling
>   // filters if |compensate_delay| is true. Returns -1 if the input is too short
>   // to produce |output_length| samples, otherwise 0.
>-  static int DownsampleTo4kHz(const int16_t* input, size_t input_length,
>-                              size_t output_length, int input_rate_hz,
>-                              bool compensate_delay, int16_t* output);
>+  static int DownsampleTo4kHz(const int16_t* input,
>+                              size_t input_length,
>+                              size_t output_length,
>+                              int input_rate_hz,
>+                              bool compensate_delay,
>+                              int16_t* output);
> 
>  private:
>   // Table of constants used in method DspHelper::ParabolicFit().
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/dsp_helper_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/dsp_helper_unittest.cc
>index 98ae2a2c2d1..ec434a4b193 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/dsp_helper_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/dsp_helper_unittest.cc
>@@ -12,7 +12,6 @@
> 
> #include "modules/audio_coding/neteq/audio_multi_vector.h"
> #include "test/gtest.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -30,8 +29,8 @@ TEST(DspHelper, RampSignalArray) {
>   int increment = (16384 << 6) / kLen;
> 
>   // Test first method.
>-  int stop_factor = DspHelper::RampSignal(input, kLen, start_factor, increment,
>-                                          output);
>+  int stop_factor =
>+      DspHelper::RampSignal(input, kLen, start_factor, increment, output);
>   EXPECT_EQ(16383, stop_factor);  // Almost reach 1 in Q14.
>   for (int i = 0; i < kLen; ++i) {
>     EXPECT_EQ(1000 * i / kLen, output[i]);
>@@ -63,8 +62,8 @@ TEST(DspHelper, RampSignalAudioMultiVector) {
>   // Q20, while the factor is in Q14, hence the shift by 6.
>   int increment = (16384 << 6) / kLen;
> 
>-  int stop_factor = DspHelper::RampSignal(&input, start_index, kLen,
>-                                          start_factor, increment);
>+  int stop_factor =
>+      DspHelper::RampSignal(&input, start_index, kLen, start_factor, increment);
>   EXPECT_EQ(16383, stop_factor);  // Almost reach 1 in Q14.
>   // Verify that the first |kLen| samples are left untouched.
>   int i;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/dtmf_buffer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/dtmf_buffer.cc
>index 370de42bbe7..656cff9606b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/dtmf_buffer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/dtmf_buffer.cc
>@@ -98,9 +98,8 @@ int DtmfBuffer::ParseEvent(uint32_t rtp_timestamp,
> // already in the buffer. If so, the new event is simply merged with the
> // existing one.
> int DtmfBuffer::InsertEvent(const DtmfEvent& event) {
>-  if (event.event_no < 0 || event.event_no > 15 ||
>-      event.volume < 0 || event.volume > 63 ||
>-      event.duration <= 0 || event.duration > 65535) {
>+  if (event.event_no < 0 || event.event_no > 15 || event.volume < 0 ||
>+      event.volume > 63 || event.duration <= 0 || event.duration > 65535) {
>     RTC_LOG(LS_WARNING) << "InsertEvent invalid parameters";
>     return kInvalidEventParameters;
>   }
>@@ -142,8 +141,8 @@ bool DtmfBuffer::GetEvent(uint32_t current_timestamp, DtmfEvent* event) {
> #endif
>       }
>     }
>-    if (current_timestamp >= it->timestamp
>-        && current_timestamp <= event_end) {  // TODO(hlundin): Change to <.
>+    if (current_timestamp >= it->timestamp &&
>+        current_timestamp <= event_end) {  // TODO(hlundin): Change to <.
>       // Found a matching event.
>       if (event) {
>         event->event_no = it->event_no;
>@@ -153,16 +152,15 @@ bool DtmfBuffer::GetEvent(uint32_t current_timestamp, DtmfEvent* event) {
>         event->timestamp = it->timestamp;
>       }
> #ifdef LEGACY_BITEXACT
>-      if (it->end_bit &&
>-          current_timestamp + frame_len_samples_ >= event_end) {
>+      if (it->end_bit && current_timestamp + frame_len_samples_ >= event_end) {
>         // We are done playing this. Erase the event.
>         buffer_.erase(it);
>       }
> #endif
>       return true;
>     } else if (current_timestamp > event_end) {  // TODO(hlundin): Change to >=.
>-      // Erase old event. Operation returns a valid pointer to the next element
>-      // in the list.
>+// Erase old event. Operation returns a valid pointer to the next element
>+// in the list.
> #ifdef LEGACY_BITEXACT
>       if (!next_available) {
>         if (event) {
>@@ -196,10 +194,7 @@ bool DtmfBuffer::Empty() const {
> }
> 
> int DtmfBuffer::SetSampleRate(int fs_hz) {
>-  if (fs_hz != 8000 &&
>-      fs_hz != 16000 &&
>-      fs_hz != 32000 &&
>-      fs_hz != 48000) {
>+  if (fs_hz != 8000 && fs_hz != 16000 && fs_hz != 32000 && fs_hz != 48000) {
>     return kInvalidSampleRate;
>   }
>   max_extrapolation_samples_ = 7 * fs_hz / 100;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/dtmf_buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/dtmf_buffer.h
>index 87a56554d52..6de512772f2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/dtmf_buffer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/dtmf_buffer.h
>@@ -15,7 +15,6 @@
> #include <string>  // size_t
> 
> #include "rtc_base/constructormagic.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -28,19 +27,9 @@ struct DtmfEvent {
> 
>   // Constructors
>   DtmfEvent()
>-      : timestamp(0),
>-        event_no(0),
>-        volume(0),
>-        duration(0),
>-        end_bit(false) {
>-  }
>+      : timestamp(0), event_no(0), volume(0), duration(0), end_bit(false) {}
>   DtmfEvent(uint32_t ts, int ev, int vol, int dur, bool end)
>-      : timestamp(ts),
>-        event_no(ev),
>-        volume(vol),
>-        duration(dur),
>-        end_bit(end) {
>-  }
>+      : timestamp(ts), event_no(ev), volume(vol), duration(dur), end_bit(end) {}
> };
> 
> // This is the buffer holding DTMF events while waiting for them to be played.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/dtmf_buffer_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/dtmf_buffer_unittest.cc
>index 7bcf1e0f4ee..607a5ec6fa9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/dtmf_buffer_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/dtmf_buffer_unittest.cc
>@@ -31,11 +31,11 @@ static int sample_rate_hz = 8000;
> 
> static uint32_t MakeDtmfPayload(int event, bool end, int volume, int duration) {
>   uint32_t payload = 0;
>-//  0                   1                   2                   3
>-//  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
>-// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>-// |     event     |E|R| volume    |          duration             |
>-// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>+  //  0                   1                   2                   3
>+  //  0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
>+  // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>+  // |     event     |E|R| volume    |          duration             |
>+  // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>   payload |= (event & 0x00FF) << 24;
>   payload |= (end ? 0x00800000 : 0x00000000);
>   payload |= (volume & 0x003F) << 16;
>@@ -44,13 +44,10 @@ static uint32_t MakeDtmfPayload(int event, bool end, int volume, int duration) {
>   return payload;
> }
> 
>-static bool EqualEvents(const DtmfEvent& a,
>-                        const DtmfEvent& b) {
>-  return (a.duration == b.duration
>-      && a.end_bit == b.end_bit
>-      && a.event_no == b.event_no
>-      && a.timestamp == b.timestamp
>-      && a.volume == b.volume);
>+static bool EqualEvents(const DtmfEvent& a, const DtmfEvent& b) {
>+  return (a.duration == b.duration && a.end_bit == b.end_bit &&
>+          a.event_no == b.event_no && a.timestamp == b.timestamp &&
>+          a.volume == b.volume);
> }
> 
> TEST(DtmfBuffer, CreateAndDestroy) {
>@@ -68,9 +65,8 @@ TEST(DtmfBuffer, ParseEvent) {
>   uint32_t payload = MakeDtmfPayload(event_no, end_bit, volume, duration);
>   uint8_t* payload_ptr = reinterpret_cast<uint8_t*>(&payload);
>   DtmfEvent event;
>-  EXPECT_EQ(DtmfBuffer::kOK,
>-            DtmfBuffer::ParseEvent(timestamp, payload_ptr, sizeof(payload),
>-                                   &event));
>+  EXPECT_EQ(DtmfBuffer::kOK, DtmfBuffer::ParseEvent(timestamp, payload_ptr,
>+                                                    sizeof(payload), &event));
>   EXPECT_EQ(duration, event.duration);
>   EXPECT_EQ(end_bit, event.end_bit);
>   EXPECT_EQ(event_no, event.event_no);
>@@ -107,7 +103,7 @@ TEST(DtmfBuffer, SimpleInsertAndGet) {
>   EXPECT_TRUE(EqualEvents(event, out_event));
>   EXPECT_EQ(1u, buffer.Length());
>   EXPECT_FALSE(buffer.Empty());
>-  // Give a "current" timestamp after the event has ended.
>+// Give a "current" timestamp after the event has ended.
> #ifdef LEGACY_BITEXACT
>   EXPECT_TRUE(buffer.GetEvent(timestamp + duration + 10, &out_event));
> #endif
>@@ -171,17 +167,17 @@ TEST(DtmfBuffer, OverlappingEvents) {
>   // Expect to get the long event.
>   EXPECT_TRUE(buffer.GetEvent(timestamp, &out_event));
>   EXPECT_TRUE(EqualEvents(long_event, out_event));
>-  // Expect no more events.
>+// Expect no more events.
> #ifdef LEGACY_BITEXACT
>-  EXPECT_TRUE(buffer.GetEvent(timestamp + long_event.duration + 10,
>-                              &out_event));
>+  EXPECT_TRUE(
>+      buffer.GetEvent(timestamp + long_event.duration + 10, &out_event));
>   EXPECT_TRUE(EqualEvents(long_event, out_event));
>-  EXPECT_TRUE(buffer.GetEvent(timestamp + long_event.duration + 10,
>-                              &out_event));
>+  EXPECT_TRUE(
>+      buffer.GetEvent(timestamp + long_event.duration + 10, &out_event));
>   EXPECT_TRUE(EqualEvents(short_event, out_event));
> #else
>-  EXPECT_FALSE(buffer.GetEvent(timestamp + long_event.duration + 10,
>-                               &out_event));
>+  EXPECT_FALSE(
>+      buffer.GetEvent(timestamp + long_event.duration + 10, &out_event));
> #endif
>   EXPECT_TRUE(buffer.Empty());
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/dtmf_tone_generator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/dtmf_tone_generator.cc
>index b848c600d07..6fdb95aacad 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/dtmf_tone_generator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/dtmf_tone_generator.cc
>@@ -39,72 +39,69 @@ namespace webrtc {
> // sample rates fs = {8000, 16000, 32000, 48000} Hz, and events 0 through 15.
> // Values are in Q14.
> const int DtmfToneGenerator::kCoeff1[4][16] = {
>-    { 24219, 27980, 27980, 27980, 26956, 26956, 26956, 25701, 25701, 25701,
>-      24219, 24219, 27980, 26956, 25701, 24219 },
>-    { 30556, 31548, 31548, 31548, 31281, 31281, 31281, 30951, 30951, 30951,
>-      30556, 30556, 31548, 31281, 30951, 30556 },
>-    { 32210, 32462, 32462, 32462, 32394, 32394, 32394, 32311, 32311, 32311,
>-      32210, 32210, 32462, 32394, 32311, 32210 },
>-    { 32520, 32632, 32632, 32632, 32602, 32602, 32602, 32564, 32564, 32564,
>-      32520, 32520, 32632, 32602, 32564, 32520 } };
>+    {24219, 27980, 27980, 27980, 26956, 26956, 26956, 25701, 25701, 25701,
>+     24219, 24219, 27980, 26956, 25701, 24219},
>+    {30556, 31548, 31548, 31548, 31281, 31281, 31281, 30951, 30951, 30951,
>+     30556, 30556, 31548, 31281, 30951, 30556},
>+    {32210, 32462, 32462, 32462, 32394, 32394, 32394, 32311, 32311, 32311,
>+     32210, 32210, 32462, 32394, 32311, 32210},
>+    {32520, 32632, 32632, 32632, 32602, 32602, 32602, 32564, 32564, 32564,
>+     32520, 32520, 32632, 32602, 32564, 32520}};
> 
> // The filter coefficient a = 2*cos(2*pi*f/fs) for the high frequency tone, for
> // sample rates fs = {8000, 16000, 32000, 48000} Hz, and events 0 through 15.
> // Values are in Q14.
> const int DtmfToneGenerator::kCoeff2[4][16] = {
>-    { 16325, 19073, 16325, 13085, 19073, 16325, 13085, 19073, 16325, 13085,
>-      19073, 13085, 9315, 9315, 9315, 9315},
>-    { 28361, 29144, 28361, 27409, 29144, 28361, 27409, 29144, 28361, 27409,
>-      29144, 27409, 26258, 26258, 26258, 26258},
>-    { 31647, 31849, 31647, 31400, 31849, 31647, 31400, 31849, 31647, 31400,
>-      31849, 31400, 31098, 31098, 31098, 31098},
>-    { 32268, 32359, 32268, 32157, 32359, 32268, 32157, 32359, 32268, 32157,
>-      32359, 32157, 32022, 32022, 32022, 32022} };
>+    {16325, 19073, 16325, 13085, 19073, 16325, 13085, 19073, 16325, 13085,
>+     19073, 13085, 9315, 9315, 9315, 9315},
>+    {28361, 29144, 28361, 27409, 29144, 28361, 27409, 29144, 28361, 27409,
>+     29144, 27409, 26258, 26258, 26258, 26258},
>+    {31647, 31849, 31647, 31400, 31849, 31647, 31400, 31849, 31647, 31400,
>+     31849, 31400, 31098, 31098, 31098, 31098},
>+    {32268, 32359, 32268, 32157, 32359, 32268, 32157, 32359, 32268, 32157,
>+     32359, 32157, 32022, 32022, 32022, 32022}};
> 
> // The initialization value x[-2] = sin(2*pi*f/fs) for the low frequency tone,
> // for sample rates fs = {8000, 16000, 32000, 48000} Hz, and events 0-15.
> // Values are in Q14.
> const int DtmfToneGenerator::kInitValue1[4][16] = {
>-    { 11036, 8528, 8528, 8528, 9315, 9315, 9315, 10163, 10163, 10163, 11036,
>-      11036, 8528, 9315, 10163, 11036},
>-    { 5918, 4429, 4429, 4429, 4879, 4879, 4879, 5380, 5380, 5380, 5918, 5918,
>-      4429, 4879, 5380, 5918},
>-    { 3010, 2235, 2235, 2235, 2468, 2468, 2468, 2728, 2728, 2728, 3010, 3010,
>-      2235, 2468, 2728, 3010},
>-    { 2013, 1493, 1493, 1493, 1649, 1649, 1649, 1823, 1823, 1823, 2013, 2013,
>-      1493, 1649, 1823, 2013 } };
>+    {11036, 8528, 8528, 8528, 9315, 9315, 9315, 10163, 10163, 10163, 11036,
>+     11036, 8528, 9315, 10163, 11036},
>+    {5918, 4429, 4429, 4429, 4879, 4879, 4879, 5380, 5380, 5380, 5918, 5918,
>+     4429, 4879, 5380, 5918},
>+    {3010, 2235, 2235, 2235, 2468, 2468, 2468, 2728, 2728, 2728, 3010, 3010,
>+     2235, 2468, 2728, 3010},
>+    {2013, 1493, 1493, 1493, 1649, 1649, 1649, 1823, 1823, 1823, 2013, 2013,
>+     1493, 1649, 1823, 2013}};
> 
> // The initialization value x[-2] = sin(2*pi*f/fs) for the high frequency tone,
> // for sample rates fs = {8000, 16000, 32000, 48000} Hz, and events 0-15.
> // Values are in Q14.
> const int DtmfToneGenerator::kInitValue2[4][16] = {
>-    { 14206, 13323, 14206, 15021, 13323, 14206, 15021, 13323, 14206, 15021,
>-      13323, 15021, 15708, 15708, 15708, 15708},
>-    { 8207, 7490, 8207, 8979, 7490, 8207, 8979, 7490, 8207, 8979, 7490, 8979,
>-      9801, 9801, 9801, 9801},
>-    { 4249, 3853, 4249, 4685, 3853, 4249, 4685, 3853, 4249, 4685, 3853, 4685,
>-      5164, 5164, 5164, 5164},
>-    { 2851, 2582, 2851, 3148, 2582, 2851, 3148, 2582, 2851, 3148, 2582, 3148,
>-      3476, 3476, 3476, 3476} };
>+    {14206, 13323, 14206, 15021, 13323, 14206, 15021, 13323, 14206, 15021,
>+     13323, 15021, 15708, 15708, 15708, 15708},
>+    {8207, 7490, 8207, 8979, 7490, 8207, 8979, 7490, 8207, 8979, 7490, 8979,
>+     9801, 9801, 9801, 9801},
>+    {4249, 3853, 4249, 4685, 3853, 4249, 4685, 3853, 4249, 4685, 3853, 4685,
>+     5164, 5164, 5164, 5164},
>+    {2851, 2582, 2851, 3148, 2582, 2851, 3148, 2582, 2851, 3148, 2582, 3148,
>+     3476, 3476, 3476, 3476}};
> 
> // Amplitude multipliers for volume values 0 through 63, corresponding to
> // 0 dBm0 through -63 dBm0. Values are in Q14.
> // for a in range(0, 64):
> //   print round(16141.0 * 10**(-float(a)/20))
> const int DtmfToneGenerator::kAmplitude[64] = {
>-    16141, 14386, 12821, 11427, 10184, 9077, 8090, 7210, 6426, 5727, 5104, 4549,
>-    4054, 3614, 3221, 2870, 2558, 2280, 2032, 1811, 1614, 1439, 1282, 1143,
>-    1018, 908, 809, 721, 643, 573, 510, 455, 405, 361, 322, 287, 256, 228, 203,
>-    181, 161, 144, 128, 114, 102, 91, 81, 72, 64, 57, 51, 45, 41, 36, 32, 29,
>-    26, 23, 20, 18, 16, 14, 13, 11 };
>+    16141, 14386, 12821, 11427, 10184, 9077, 8090, 7210, 6426, 5727, 5104,
>+    4549,  4054,  3614,  3221,  2870,  2558, 2280, 2032, 1811, 1614, 1439,
>+    1282,  1143,  1018,  908,   809,   721,  643,  573,  510,  455,  405,
>+    361,   322,   287,   256,   228,   203,  181,  161,  144,  128,  114,
>+    102,   91,    81,    72,    64,    57,   51,   45,   41,   36,   32,
>+    29,    26,    23,    20,    18,    16,   14,   13,   11};
> 
> // Constructor.
> DtmfToneGenerator::DtmfToneGenerator()
>-    : initialized_(false),
>-      coeff1_(0),
>-      coeff2_(0),
>-      amplitude_(0) {
>-}
>+    : initialized_(false), coeff1_(0), coeff2_(0), amplitude_(0) {}
> 
> // Initialize the DTMF generator with sample rate fs Hz (8000, 16000, 32000,
> // 48000), event (0-15) and attenuation (0-36 dB).
>@@ -170,8 +167,7 @@ void DtmfToneGenerator::Reset() {
> }
> 
> // Generate num_samples of DTMF signal and write to |output|.
>-int DtmfToneGenerator::Generate(size_t num_samples,
>-                                AudioMultiVector* output) {
>+int DtmfToneGenerator::Generate(size_t num_samples, AudioMultiVector* output) {
>   if (!initialized_) {
>     return kNotInitialized;
>   }
>@@ -183,10 +179,10 @@ int DtmfToneGenerator::Generate(size_t num_samples,
>   output->AssertSize(num_samples);
>   for (size_t i = 0; i < num_samples; ++i) {
>     // Use recursion formula y[n] = a * y[n - 1] - y[n - 2].
>-    int16_t temp_val_low = ((coeff1_ * sample_history1_[1] + 8192) >> 14)
>-        - sample_history1_[0];
>-    int16_t temp_val_high = ((coeff2_ * sample_history2_[1] + 8192) >> 14)
>-        - sample_history2_[0];
>+    int16_t temp_val_low =
>+        ((coeff1_ * sample_history1_[1] + 8192) >> 14) - sample_history1_[0];
>+    int16_t temp_val_high =
>+        ((coeff2_ * sample_history2_[1] + 8192) >> 14) - sample_history2_[0];
> 
>     // Update recursion memory.
>     sample_history1_[0] = sample_history1_[1];
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/dtmf_tone_generator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/dtmf_tone_generator.h
>index faad6a25b18..a773ff353d3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/dtmf_tone_generator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/dtmf_tone_generator.h
>@@ -13,7 +13,6 @@
> 
> #include "modules/audio_coding/neteq/audio_multi_vector.h"
> #include "rtc_base/constructormagic.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -37,7 +36,7 @@ class DtmfToneGenerator {
>   static const int kCoeff2[4][16];  // 2nd oscillator model coefficient table.
>   static const int kInitValue1[4][16];  // Initialization for 1st oscillator.
>   static const int kInitValue2[4][16];  // Initialization for 2nd oscillator.
>-  static const int kAmplitude[64];  // Amplitude for 0 through -63 dBm0.
>+  static const int kAmplitude[64];      // Amplitude for 0 through -63 dBm0.
>   static const int16_t kAmpMultiplier = 23171;  // 3 dB attenuation (in Q15).
> 
>   bool initialized_;            // True if generator is initialized properly.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/dtmf_tone_generator_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/dtmf_tone_generator_unittest.cc
>index 437af0d3459..11a0ac6a6b1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/dtmf_tone_generator_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/dtmf_tone_generator_unittest.cc
>@@ -14,6 +14,7 @@
> 
> #include <math.h>
> 
>+#include "common_audio/include/audio_util.h"
> #include "modules/audio_coding/neteq/audio_multi_vector.h"
> #include "test/gtest.h"
> 
>@@ -79,12 +80,11 @@ class DtmfToneGeneratorTest : public ::testing::Test {
>         EXPECT_EQ(kNumSamples, tone_gen_.Generate(kNumSamples, &signal));
>         for (int n = 0; n < kNumSamples; ++n) {
>           double attenuation_factor =
>-              pow(10, -static_cast<double>(attenuation) / 20);
>+              DbToRatio(-static_cast<float>(attenuation));
>           // Verify that the attenuation is correct.
>           for (int channel = 0; channel < channels; ++channel) {
>             EXPECT_NEAR(attenuation_factor * ref_signal[channel][n],
>-                        signal[channel][n],
>-                        2);
>+                        signal[channel][n], 2);
>           }
>         }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/expand.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/expand.cc
>index 3fb09ead494..5f671ada6b6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/expand.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/expand.cc
>@@ -14,7 +14,7 @@
> #include <string.h>  // memset
> 
> #include <algorithm>  // min, max
>-#include <limits>  // numeric_limits<T>
>+#include <limits>     // numeric_limits<T>
> 
> #include "common_audio/signal_processing/include/signal_processing_library.h"
> #include "modules/audio_coding/neteq/background_noise.h"
>@@ -94,7 +94,6 @@ int Expand::Process(AudioMultiVector* output) {
>     GenerateRandomVector(2, rand_length, random_vector);
>   }
> 
>-
>   // Generate signal.
>   UpdateLagIndex();
> 
>@@ -103,8 +102,8 @@ int Expand::Process(AudioMultiVector* output) {
>   size_t expansion_vector_length = max_lag_ + overlap_length_;
>   size_t current_lag = expand_lags_[current_lag_index_];
>   // Copy lag+overlap data.
>-  size_t expansion_vector_position = expansion_vector_length - current_lag -
>-      overlap_length_;
>+  size_t expansion_vector_position =
>+      expansion_vector_length - current_lag - overlap_length_;
>   size_t temp_length = current_lag + overlap_length_;
>   for (size_t channel_ix = 0; channel_ix < num_channels_; ++channel_ix) {
>     ChannelParameters& parameters = channel_parameters_[channel_ix];
>@@ -175,8 +174,10 @@ int Expand::Process(AudioMultiVector* output) {
>         // Do overlap add between new vector and overlap.
>         (*sync_buffer_)[channel_ix][start_ix + i] =
>             (((*sync_buffer_)[channel_ix][start_ix + i] * muting_window) +
>-                (((parameters.mute_factor * voiced_vector_storage[i]) >> 14) *
>-                    unmuting_window) + 16384) >> 15;
>+             (((parameters.mute_factor * voiced_vector_storage[i]) >> 14) *
>+              unmuting_window) +
>+             16384) >>
>+            15;
>         muting_window += muting_window_increment;
>         unmuting_window += unmuting_window_increment;
>       }
>@@ -188,10 +189,10 @@ int Expand::Process(AudioMultiVector* output) {
>       // parameters.expand_vector0 and parameters.expand_vector1 no longer
>       // match with expand_lags_, causing invalid reads and writes. Is it a good
>       // idea to enable this again, and solve the vector size problem?
>-//      max_lag_ = fs_mult * 120;
>-//      expand_lags_[0] = fs_mult * 120;
>-//      expand_lags_[1] = fs_mult * 120;
>-//      expand_lags_[2] = fs_mult * 120;
>+      //      max_lag_ = fs_mult * 120;
>+      //      expand_lags_[0] = fs_mult * 120;
>+      //      expand_lags_[1] = fs_mult * 120;
>+      //      expand_lags_[2] = fs_mult * 120;
>     }
> 
>     // Unvoiced part.
>@@ -204,8 +205,7 @@ int Expand::Process(AudioMultiVector* output) {
>     }
>     WebRtcSpl_AffineTransformVector(scaled_random_vector, random_vector,
>                                     parameters.ar_gain, add_constant,
>-                                    parameters.ar_gain_scale,
>-                                    current_lag);
>+                                    parameters.ar_gain_scale, current_lag);
>     WebRtcSpl_FilterARFastQ12(scaled_random_vector, unvoiced_vector,
>                               parameters.ar_filter, kUnvoicedLpcOrder + 1,
>                               current_lag);
>@@ -230,8 +230,9 @@ int Expand::Process(AudioMultiVector* output) {
> 
>     // Create combined signal by shifting in more and more of unvoiced part.
>     temp_shift = 8 - temp_shift;  // = getbits(mix_factor_increment).
>-    size_t temp_length = (parameters.current_voice_mix_factor -
>-        parameters.voice_mix_factor) >> temp_shift;
>+    size_t temp_length =
>+        (parameters.current_voice_mix_factor - parameters.voice_mix_factor) >>
>+        temp_shift;
>     temp_length = std::min(temp_length, current_lag);
>     DspHelper::CrossFade(voiced_vector, unvoiced_vector, temp_length,
>                          &parameters.current_voice_mix_factor,
>@@ -266,9 +267,8 @@ int Expand::Process(AudioMultiVector* output) {
>     // Mute segment according to slope value.
>     if ((consecutive_expands_ != 0) || !parameters.onset) {
>       // Mute to the previous level, then continue with the muting.
>-      WebRtcSpl_AffineTransformVector(temp_data, temp_data,
>-                                      parameters.mute_factor, 8192,
>-                                      14, current_lag);
>+      WebRtcSpl_AffineTransformVector(
>+          temp_data, temp_data, parameters.mute_factor, 8192, 14, current_lag);
> 
>       if (!stop_muting_) {
>         DspHelper::MuteSignal(temp_data, parameters.mute_slope, current_lag);
>@@ -276,8 +276,8 @@ int Expand::Process(AudioMultiVector* output) {
>         // Shift by 6 to go from Q20 to Q14.
>         // TODO(hlundin): Adding 8192 before shifting 6 steps seems wrong.
>         // Legacy.
>-        int16_t gain = static_cast<int16_t>(16384 -
>-            (((current_lag * parameters.mute_slope) + 8192) >> 6));
>+        int16_t gain = static_cast<int16_t>(
>+            16384 - (((current_lag * parameters.mute_slope) + 8192) >> 6));
>         gain = ((gain * parameters.mute_factor) + 8192) >> 14;
> 
>         // Guard against getting stuck with very small (but sometimes audible)
>@@ -291,12 +291,9 @@ int Expand::Process(AudioMultiVector* output) {
>     }
> 
>     // Background noise part.
>-    GenerateBackgroundNoise(random_vector,
>-                            channel_ix,
>-                            channel_parameters_[channel_ix].mute_slope,
>-                            TooManyExpands(),
>-                            current_lag,
>-                            unvoiced_array_memory);
>+    GenerateBackgroundNoise(
>+        random_vector, channel_ix, channel_parameters_[channel_ix].mute_slope,
>+        TooManyExpands(), current_lag, unvoiced_array_memory);
> 
>     // Add background noise to the combined voiced-unvoiced signal.
>     for (size_t i = 0; i < current_lag; i++) {
>@@ -311,8 +308,9 @@ int Expand::Process(AudioMultiVector* output) {
>   }
> 
>   // Increase call number and cap it.
>-  consecutive_expands_ = consecutive_expands_ >= kMaxConsecutiveExpands ?
>-      kMaxConsecutiveExpands : consecutive_expands_ + 1;
>+  consecutive_expands_ = consecutive_expands_ >= kMaxConsecutiveExpands
>+                             ? kMaxConsecutiveExpands
>+                             : consecutive_expands_ + 1;
>   expand_duration_samples_ += output->Size();
>   // Clamp the duration counter at 2 seconds.
>   expand_duration_samples_ = std::min(expand_duration_samples_,
>@@ -329,7 +327,7 @@ void Expand::SetParametersForNormalAfterExpand() {
> }
> 
> void Expand::SetParametersForMergeAfterExpand() {
>-  current_lag_index_ = -1; /* out of the 3 possible ones */
>+  current_lag_index_ = -1;  /* out of the 3 possible ones */
>   lag_index_direction_ = 1; /* make sure we get the "optimal" lag */
>   stop_muting_ = true;
> }
>@@ -357,7 +355,7 @@ void Expand::InitializeForAnExpandPeriod() {
>   consecutive_expands_ = 0;
>   for (size_t ix = 0; ix < num_channels_; ++ix) {
>     channel_parameters_[ix].current_voice_mix_factor = 16384;  // 1.0 in Q14.
>-    channel_parameters_[ix].mute_factor = 16384;  // 1.0 in Q14.
>+    channel_parameters_[ix].mute_factor = 16384;               // 1.0 in Q14.
>     // Start with 0 gain for background noise.
>     background_noise_->SetMuteFactor(ix, 0);
>   }
>@@ -420,10 +418,10 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
>   // Calculate distortion around the |kNumCorrelationCandidates| best lags.
>   int distortion_scale = 0;
>   for (size_t i = 0; i < kNumCorrelationCandidates; i++) {
>-    size_t min_index = std::max(fs_mult_20,
>-                                best_correlation_index[i] - fs_mult_4);
>-    size_t max_index = std::min(fs_mult_120 - 1,
>-                                best_correlation_index[i] + fs_mult_4);
>+    size_t min_index =
>+        std::max(fs_mult_20, best_correlation_index[i] - fs_mult_4);
>+    size_t max_index =
>+        std::min(fs_mult_120 - 1, best_correlation_index[i] + fs_mult_4);
>     best_distortion_index[i] = DspHelper::MinDistortion(
>         &(audio_history[signal_length - fs_mult_dist_len]), min_index,
>         max_index, fs_mult_dist_len, &best_distortion_w32[i]);
>@@ -459,23 +457,23 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
> 
>   // Calculate the exact best correlation in the range between
>   // |correlation_lag| and |distortion_lag|.
>-  correlation_length =
>-      std::max(std::min(distortion_lag + 10, fs_mult_120),
>-               static_cast<size_t>(60 * fs_mult));
>+  correlation_length = std::max(std::min(distortion_lag + 10, fs_mult_120),
>+                                static_cast<size_t>(60 * fs_mult));
> 
>   size_t start_index = std::min(distortion_lag, correlation_lag);
>   size_t correlation_lags = static_cast<size_t>(
>-      WEBRTC_SPL_ABS_W16((distortion_lag-correlation_lag)) + 1);
>+      WEBRTC_SPL_ABS_W16((distortion_lag - correlation_lag)) + 1);
>   assert(correlation_lags <= static_cast<size_t>(99 * fs_mult + 1));
> 
>   for (size_t channel_ix = 0; channel_ix < num_channels_; ++channel_ix) {
>     ChannelParameters& parameters = channel_parameters_[channel_ix];
>     // Calculate suitable scaling.
>     int16_t signal_max = WebRtcSpl_MaxAbsValueW16(
>-        &audio_history[signal_length - correlation_length - start_index
>-                       - correlation_lags],
>-                       correlation_length + start_index + correlation_lags - 1);
>-    int correlation_scale = (31 - WebRtcSpl_NormW32(signal_max * signal_max)) +
>+        &audio_history[signal_length - correlation_length - start_index -
>+                       correlation_lags],
>+        correlation_length + start_index + correlation_lags - 1);
>+    int correlation_scale =
>+        (31 - WebRtcSpl_NormW32(signal_max * signal_max)) +
>         (31 - WebRtcSpl_NormW32(static_cast<int32_t>(correlation_length))) - 31;
>     correlation_scale = std::max(0, correlation_scale);
> 
>@@ -520,8 +518,8 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
>       // Calculate max_correlation / sqrt(energy1 * energy2) in Q14.
>       int cc_shift = 14 - (energy1_scale + energy2_scale) / 2;
>       max_correlation = WEBRTC_SPL_SHIFT_W32(max_correlation, cc_shift);
>-      corr_coefficient = WebRtcSpl_DivW32W16(max_correlation,
>-                                             sqrt_energy_product);
>+      corr_coefficient =
>+          WebRtcSpl_DivW32W16(max_correlation, sqrt_energy_product);
>       // Cap at 1.0 in Q14.
>       corr_coefficient = std::min(16384, corr_coefficient);
>     } else {
>@@ -547,9 +545,9 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
>       int32_t scaled_energy2 = std::max(16 - WebRtcSpl_NormW32(energy2), 0);
>       int32_t scaled_energy1 = scaled_energy2 - 13;
>       // Calculate scaled_energy1 / scaled_energy2 in Q13.
>-      int32_t energy_ratio = WebRtcSpl_DivW32W16(
>-          WEBRTC_SPL_SHIFT_W32(energy1, -scaled_energy1),
>-          static_cast<int16_t>(energy2 >> scaled_energy2));
>+      int32_t energy_ratio =
>+          WebRtcSpl_DivW32W16(WEBRTC_SPL_SHIFT_W32(energy1, -scaled_energy1),
>+                              static_cast<int16_t>(energy2 >> scaled_energy2));
>       // Calculate sqrt ratio in Q13 (sqrt of en1/en2 in Q26).
>       amplitude_ratio =
>           static_cast<int16_t>(WebRtcSpl_SqrtFloor(energy_ratio << 13));
>@@ -558,16 +556,13 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
>       parameters.expand_vector0.PushBack(vector1, expansion_length);
>       parameters.expand_vector1.Clear();
>       if (parameters.expand_vector1.Size() < expansion_length) {
>-        parameters.expand_vector1.Extend(
>-            expansion_length - parameters.expand_vector1.Size());
>+        parameters.expand_vector1.Extend(expansion_length -
>+                                         parameters.expand_vector1.Size());
>       }
>       std::unique_ptr<int16_t[]> temp_1(new int16_t[expansion_length]);
>-      WebRtcSpl_AffineTransformVector(temp_1.get(),
>-                                      const_cast<int16_t*>(vector2),
>-                                      amplitude_ratio,
>-                                      4096,
>-                                      13,
>-                                      expansion_length);
>+      WebRtcSpl_AffineTransformVector(
>+          temp_1.get(), const_cast<int16_t*>(vector2), amplitude_ratio, 4096,
>+          13, expansion_length);
>       parameters.expand_vector1.OverwriteAt(temp_1.get(), expansion_length, 0);
>     } else {
>       // Energy change constraint not fulfilled. Only use last vector.
>@@ -606,11 +601,11 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
>     // Calculate the LPC and the gain of the filters.
> 
>     // Calculate kUnvoicedLpcOrder + 1 lags of the auto-correlation function.
>-    size_t temp_index = signal_length - fs_mult_lpc_analysis_len -
>-        kUnvoicedLpcOrder;
>+    size_t temp_index =
>+        signal_length - fs_mult_lpc_analysis_len - kUnvoicedLpcOrder;
>     // Copy signal to temporary vector to be able to pad with leading zeros.
>-    int16_t* temp_signal = new int16_t[fs_mult_lpc_analysis_len
>-                                       + kUnvoicedLpcOrder];
>+    int16_t* temp_signal =
>+        new int16_t[fs_mult_lpc_analysis_len + kUnvoicedLpcOrder];
>     memset(temp_signal, 0,
>            sizeof(int16_t) * (fs_mult_lpc_analysis_len + kUnvoicedLpcOrder));
>     memcpy(&temp_signal[kUnvoicedLpcOrder],
>@@ -619,16 +614,15 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
>     CrossCorrelationWithAutoShift(
>         &temp_signal[kUnvoicedLpcOrder], &temp_signal[kUnvoicedLpcOrder],
>         fs_mult_lpc_analysis_len, kUnvoicedLpcOrder + 1, -1, auto_correlation);
>-    delete [] temp_signal;
>+    delete[] temp_signal;
> 
>     // Verify that variance is positive.
>     if (auto_correlation[0] > 0) {
>       // Estimate AR filter parameters using Levinson-Durbin algorithm;
>       // kUnvoicedLpcOrder + 1 filter coefficients.
>-      int16_t stability = WebRtcSpl_LevinsonDurbin(auto_correlation,
>-                                                   parameters.ar_filter,
>-                                                   reflection_coeff,
>-                                                   kUnvoicedLpcOrder);
>+      int16_t stability =
>+          WebRtcSpl_LevinsonDurbin(auto_correlation, parameters.ar_filter,
>+                                   reflection_coeff, kUnvoicedLpcOrder);
> 
>       // Keep filter parameters only if filter is stable.
>       if (stability != 1) {
>@@ -671,10 +665,8 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
>            &(audio_history[signal_length - 128 - kUnvoicedLpcOrder]),
>            sizeof(int16_t) * kUnvoicedLpcOrder);
>     WebRtcSpl_FilterMAFastQ12(&audio_history[signal_length - 128],
>-                              unvoiced_vector,
>-                              parameters.ar_filter,
>-                              kUnvoicedLpcOrder + 1,
>-                              128);
>+                              unvoiced_vector, parameters.ar_filter,
>+                              kUnvoicedLpcOrder + 1, 128);
>     const int unvoiced_max_abs = [&] {
>       const int16_t max_abs = WebRtcSpl_MaxAbsValueW16(unvoiced_vector, 128);
>       // Since WebRtcSpl_MaxAbsValueW16 returns 2^15 - 1 when the input contains
>@@ -689,10 +681,8 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
>     int unvoiced_prescale =
>         std::max(0, 2 * WebRtcSpl_GetSizeInBits(unvoiced_max_abs) - 24);
> 
>-    int32_t unvoiced_energy = WebRtcSpl_DotProductWithScale(unvoiced_vector,
>-                                                            unvoiced_vector,
>-                                                            128,
>-                                                            unvoiced_prescale);
>+    int32_t unvoiced_energy = WebRtcSpl_DotProductWithScale(
>+        unvoiced_vector, unvoiced_vector, 128, unvoiced_prescale);
> 
>     // Normalize |unvoiced_energy| to 28 or 29 bits to preserve sqrt() accuracy.
>     int16_t unvoiced_scale = WebRtcSpl_NormW32(unvoiced_energy) - 3;
>@@ -703,8 +693,8 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
>     unvoiced_energy = WEBRTC_SPL_SHIFT_W32(unvoiced_energy, unvoiced_scale);
>     int16_t unvoiced_gain =
>         static_cast<int16_t>(WebRtcSpl_SqrtFloor(unvoiced_energy));
>-    parameters.ar_gain_scale = 13
>-        + (unvoiced_scale + 7 - unvoiced_prescale) / 2;
>+    parameters.ar_gain_scale =
>+        13 + (unvoiced_scale + 7 - unvoiced_prescale) / 2;
>     parameters.ar_gain = unvoiced_gain;
> 
>     // Calculate voice_mix_factor from corr_coefficient.
>@@ -717,17 +707,17 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
>       int16_t x1, x2, x3;
>       // |corr_coefficient| is in Q14.
>       x1 = static_cast<int16_t>(corr_coefficient);
>-      x2 = (x1 * x1) >> 14;   // Shift 14 to keep result in Q14.
>+      x2 = (x1 * x1) >> 14;  // Shift 14 to keep result in Q14.
>       x3 = (x1 * x2) >> 14;
>-      static const int kCoefficients[4] = { -5179, 19931, -16422, 5776 };
>+      static const int kCoefficients[4] = {-5179, 19931, -16422, 5776};
>       int32_t temp_sum = kCoefficients[0] * 16384;
>       temp_sum += kCoefficients[1] * x1;
>       temp_sum += kCoefficients[2] * x2;
>       temp_sum += kCoefficients[3] * x3;
>       parameters.voice_mix_factor =
>           static_cast<int16_t>(std::min(temp_sum / 4096, 16384));
>-      parameters.voice_mix_factor = std::max(parameters.voice_mix_factor,
>-                                             static_cast<int16_t>(0));
>+      parameters.voice_mix_factor =
>+          std::max(parameters.voice_mix_factor, static_cast<int16_t>(0));
>     } else {
>       parameters.voice_mix_factor = 0;
>     }
>@@ -743,9 +733,9 @@ void Expand::AnalyzeSignal(int16_t* random_vector) {
>       // the division.
>       // Shift the denominator from Q13 to Q5 before the division. The result of
>       // the division will then be in Q20.
>-      int temp_ratio = WebRtcSpl_DivW32W16(
>-          (slope - 8192) << 12,
>-          static_cast<int16_t>((distortion_lag * slope) >> 8));
>+      int16_t denom =
>+          rtc::saturated_cast<int16_t>((distortion_lag * slope) >> 8);
>+      int temp_ratio = WebRtcSpl_DivW32W16((slope - 8192) << 12, denom);
>       if (slope > 14746) {
>         // slope > 1.8.
>         // Divide by 2, with proper rounding.
>@@ -816,8 +806,8 @@ void Expand::Correlation(const int16_t* input,
>   static const size_t kNumCorrelationLags = 54;
>   static const size_t kCorrelationLength = 60;
>   // Downsample to 4 kHz sample rate.
>-  static const size_t kDownsampledLength = kCorrelationStartLag
>-      + kNumCorrelationLags + kCorrelationLength;
>+  static const size_t kDownsampledLength =
>+      kCorrelationStartLag + kNumCorrelationLags + kCorrelationLength;
>   int16_t downsampled_input[kDownsampledLength];
>   static const size_t kFilterDelay = 0;
>   WebRtcSpl_DownsampleFast(
>@@ -827,8 +817,8 @@ void Expand::Correlation(const int16_t* input,
>       downsampling_factor, kFilterDelay);
> 
>   // Normalize |downsampled_input| to using all 16 bits.
>-  int16_t max_value = WebRtcSpl_MaxAbsValueW16(downsampled_input,
>-                                               kDownsampledLength);
>+  int16_t max_value =
>+      WebRtcSpl_MaxAbsValueW16(downsampled_input, kDownsampledLength);
>   int16_t norm_shift = 16 - WebRtcSpl_NormW32(max_value);
>   WebRtcSpl_VectorBitShiftW16(downsampled_input, kDownsampledLength,
>                               downsampled_input, norm_shift);
>@@ -836,13 +826,13 @@ void Expand::Correlation(const int16_t* input,
>   int32_t correlation[kNumCorrelationLags];
>   CrossCorrelationWithAutoShift(
>       &downsampled_input[kDownsampledLength - kCorrelationLength],
>-      &downsampled_input[kDownsampledLength - kCorrelationLength
>-          - kCorrelationStartLag],
>+      &downsampled_input[kDownsampledLength - kCorrelationLength -
>+                         kCorrelationStartLag],
>       kCorrelationLength, kNumCorrelationLags, -1, correlation);
> 
>   // Normalize and move data from 32-bit to 16-bit vector.
>-  int32_t max_correlation = WebRtcSpl_MaxAbsValueW32(correlation,
>-                                                     kNumCorrelationLags);
>+  int32_t max_correlation =
>+      WebRtcSpl_MaxAbsValueW32(correlation, kNumCorrelationLags);
>   int16_t norm_shift2 = static_cast<int16_t>(
>       std::max(18 - WebRtcSpl_NormW32(max_correlation), 0));
>   WebRtcSpl_VectorBitShiftW32ToW16(output, kNumCorrelationLags, correlation,
>@@ -894,63 +884,23 @@ void Expand::GenerateBackgroundNoise(int16_t* random_vector,
> 
>     // Scale random vector to correct energy level.
>     WebRtcSpl_AffineTransformVector(
>-        scaled_random_vector, random_vector,
>-        background_noise_->Scale(channel), dc_offset,
>-        background_noise_->ScaleShift(channel),
>-        num_noise_samples);
>+        scaled_random_vector, random_vector, background_noise_->Scale(channel),
>+        dc_offset, background_noise_->ScaleShift(channel), num_noise_samples);
> 
>     WebRtcSpl_FilterARFastQ12(scaled_random_vector, noise_samples,
>                               background_noise_->Filter(channel),
>-                              kNoiseLpcOrder + 1,
>-                              num_noise_samples);
>+                              kNoiseLpcOrder + 1, num_noise_samples);
> 
>     background_noise_->SetFilterState(
>-        channel,
>-        &(noise_samples[num_noise_samples - kNoiseLpcOrder]),
>+        channel, &(noise_samples[num_noise_samples - kNoiseLpcOrder]),
>         kNoiseLpcOrder);
> 
>     // Unmute the background noise.
>     int16_t bgn_mute_factor = background_noise_->MuteFactor(channel);
>-    NetEq::BackgroundNoiseMode bgn_mode = background_noise_->mode();
>-    if (bgn_mode == NetEq::kBgnFade && too_many_expands &&
>-        bgn_mute_factor > 0) {
>-      // Fade BGN to zero.
>-      // Calculate muting slope, approximately -2^18 / fs_hz.
>-      int mute_slope;
>-      if (fs_hz_ == 8000) {
>-        mute_slope = -32;
>-      } else if (fs_hz_ == 16000) {
>-        mute_slope = -16;
>-      } else if (fs_hz_ == 32000) {
>-        mute_slope = -8;
>-      } else {
>-        mute_slope = -5;
>-      }
>-      // Use UnmuteSignal function with negative slope.
>-      // |bgn_mute_factor| is in Q14. |mute_slope| is in Q20.
>-      DspHelper::UnmuteSignal(noise_samples,
>-                              num_noise_samples,
>-                              &bgn_mute_factor,
>-                              mute_slope,
>-                              noise_samples);
>-    } else if (bgn_mute_factor < 16384) {
>-      // If mode is kBgnOn, or if kBgnFade has started fading,
>-      // use regular |mute_slope|.
>-      if (!stop_muting_ && bgn_mode != NetEq::kBgnOff &&
>-          !(bgn_mode == NetEq::kBgnFade && too_many_expands)) {
>-        DspHelper::UnmuteSignal(noise_samples,
>-                                static_cast<int>(num_noise_samples),
>-                                &bgn_mute_factor,
>-                                mute_slope,
>-                                noise_samples);
>-      } else {
>-        // kBgnOn and stop muting, or
>-        // kBgnOff (mute factor is always 0), or
>-        // kBgnFade has reached 0.
>-        WebRtcSpl_AffineTransformVector(noise_samples, noise_samples,
>-                                        bgn_mute_factor, 8192, 14,
>-                                        num_noise_samples);
>-      }
>+    if (bgn_mute_factor < 16384) {
>+      WebRtcSpl_AffineTransformVector(noise_samples, noise_samples,
>+                                      bgn_mute_factor, 8192, 14,
>+                                      num_noise_samples);
>     }
>     // Update mute_factor in BackgroundNoise class.
>     background_noise_->SetMuteFactor(channel, bgn_mute_factor);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/expand.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/expand.h
>index 39249f13c20..4cfe7b9d8cb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/expand.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/expand.h
>@@ -16,7 +16,6 @@
> 
> #include "modules/audio_coding/neteq/audio_multi_vector.h"
> #include "rtc_base/constructormagic.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -57,7 +56,7 @@ class Expand {
>   virtual void SetParametersForMergeAfterExpand();
> 
>   // Returns the mute factor for |channel|.
>-  int16_t MuteFactor(size_t channel) {
>+  int16_t MuteFactor(size_t channel) const {
>     assert(channel < num_channels_);
>     return channel_parameters_[channel].mute_factor;
>   }
>@@ -114,7 +113,7 @@ class Expand {
>     int16_t ar_filter_state[kUnvoicedLpcOrder];
>     int16_t ar_gain;
>     int16_t ar_gain_scale;
>-    int16_t voice_mix_factor; /* Q14 */
>+    int16_t voice_mix_factor;         /* Q14 */
>     int16_t current_voice_mix_factor; /* Q14 */
>     AudioVector expand_vector0;
>     AudioVector expand_vector1;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/expand_uma_logger.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/expand_uma_logger.cc
>new file mode 100644
>index 00000000000..01c2dabe638
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/expand_uma_logger.cc
>@@ -0,0 +1,69 @@
>+/*  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_coding/neteq/expand_uma_logger.h"
>+#include "rtc_base/checks.h"
>+#include "system_wrappers/include/metrics.h"
>+
>+namespace webrtc {
>+namespace {
>+std::unique_ptr<TickTimer::Countdown> GetNewCountdown(
>+    const TickTimer& tick_timer,
>+    int logging_period_s) {
>+  return tick_timer.GetNewCountdown((logging_period_s * 1000) /
>+                                    tick_timer.ms_per_tick());
>+}
>+}  // namespace
>+
>+ExpandUmaLogger::ExpandUmaLogger(std::string uma_name,
>+                                 int logging_period_s,
>+                                 const TickTimer* tick_timer)
>+    : uma_name_(uma_name),
>+      logging_period_s_(logging_period_s),
>+      tick_timer_(*tick_timer),
>+      timer_(GetNewCountdown(tick_timer_, logging_period_s_)) {
>+  RTC_DCHECK(tick_timer);
>+  RTC_DCHECK_GT(logging_period_s_, 0);
>+}
>+
>+ExpandUmaLogger::~ExpandUmaLogger() = default;
>+
>+void ExpandUmaLogger::UpdateSampleCounter(uint64_t samples,
>+                                          int sample_rate_hz) {
>+  if ((last_logged_value_ && *last_logged_value_ > samples) ||
>+      sample_rate_hz_ != sample_rate_hz) {
>+    // Sanity checks. The incremental counter moved backwards, or sample rate
>+    // changed.
>+    last_logged_value_.reset();
>+  }
>+  last_value_ = samples;
>+  sample_rate_hz_ = sample_rate_hz;
>+  if (!last_logged_value_) {
>+    last_logged_value_ = absl::optional<uint64_t>(samples);
>+  }
>+
>+  if (!timer_->Finished()) {
>+    // Not yet time to log.
>+    return;
>+  }
>+
>+  RTC_DCHECK(last_logged_value_);
>+  RTC_DCHECK_GE(last_value_, *last_logged_value_);
>+  const uint64_t diff = last_value_ - *last_logged_value_;
>+  last_logged_value_ = absl::optional<uint64_t>(last_value_);
>+  // Calculate rate in percent.
>+  RTC_DCHECK_GT(sample_rate_hz, 0);
>+  const int rate = (100 * diff) / (sample_rate_hz * logging_period_s_);
>+  RTC_DCHECK_GE(rate, 0);
>+  RTC_DCHECK_LE(rate, 100);
>+  RTC_HISTOGRAM_PERCENTAGE_SPARSE(uma_name_, rate);
>+  timer_ = GetNewCountdown(tick_timer_, logging_period_s_);
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/expand_uma_logger.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/expand_uma_logger.h
>new file mode 100644
>index 00000000000..00907d4e5a8
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/expand_uma_logger.h
>@@ -0,0 +1,54 @@
>+/*  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_CODING_NETEQ_EXPAND_UMA_LOGGER_H_
>+#define MODULES_AUDIO_CODING_NETEQ_EXPAND_UMA_LOGGER_H_
>+
>+#include <memory>
>+#include <string>
>+
>+#include "absl/types/optional.h"
>+#include "modules/audio_coding/neteq/tick_timer.h"
>+#include "rtc_base/constructormagic.h"
>+
>+namespace webrtc {
>+
>+// This class is used to periodically log values to a UMA histogram. The caller
>+// is expected to update this class with an incremental sample counter which
>+// counts expand samples. At the end of each logging period, the class will
>+// calculate the fraction of samples that were expand samples during that period
>+// and report that in percent. The logging period must be strictly positive.
>+// Does not take ownership of tick_timer and the pointer must refer to a valid
>+// object that outlives the one constructed.
>+class ExpandUmaLogger {
>+ public:
>+  ExpandUmaLogger(std::string uma_name,
>+                  int logging_period_s,
>+                  const TickTimer* tick_timer);
>+
>+  ~ExpandUmaLogger();
>+
>+  // In this call, value should be an incremental sample counter. The sample
>+  // rate must be strictly positive.
>+  void UpdateSampleCounter(uint64_t value, int sample_rate_hz);
>+
>+ private:
>+  const std::string uma_name_;
>+  const int logging_period_s_;
>+  const TickTimer& tick_timer_;
>+  std::unique_ptr<TickTimer::Countdown> timer_;
>+  absl::optional<uint64_t> last_logged_value_;
>+  uint64_t last_value_ = 0;
>+  int sample_rate_hz_ = 0;
>+
>+  RTC_DISALLOW_COPY_AND_ASSIGN(ExpandUmaLogger);
>+};
>+
>+}  // namespace webrtc
>+#endif  // MODULES_AUDIO_CODING_NETEQ_EXPAND_UMA_LOGGER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/include/neteq.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/include/neteq.h
>index 0d47f21a065..731dbe684ab 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/include/neteq.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/include/neteq.h
>@@ -16,14 +16,14 @@
> #include <string>
> #include <vector>
> 
>+#include "absl/types/optional.h"
>+#include "api/audio_codecs/audio_codec_pair_id.h"
> #include "api/audio_codecs/audio_decoder.h"
>-#include "api/optional.h"
> #include "api/rtp_headers.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/audio_coding/neteq/neteq_decoder_enum.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/scoped_ref_ptr.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -32,25 +32,25 @@ class AudioFrame;
> class AudioDecoderFactory;
> 
> struct NetEqNetworkStatistics {
>-  uint16_t current_buffer_size_ms;  // Current jitter buffer size in ms.
>+  uint16_t current_buffer_size_ms;    // Current jitter buffer size in ms.
>   uint16_t preferred_buffer_size_ms;  // Target buffer size in ms.
>-  uint16_t jitter_peaks_found;  // 1 if adding extra delay due to peaky
>-                                // jitter; 0 otherwise.
>-  uint16_t packet_loss_rate;  // Loss rate (network + late) in Q14.
>-  uint16_t expand_rate;  // Fraction (of original stream) of synthesized
>-                         // audio inserted through expansion (in Q14).
>+  uint16_t jitter_peaks_found;        // 1 if adding extra delay due to peaky
>+                                      // jitter; 0 otherwise.
>+  uint16_t packet_loss_rate;          // Loss rate (network + late) in Q14.
>+  uint16_t expand_rate;         // Fraction (of original stream) of synthesized
>+                                // audio inserted through expansion (in Q14).
>   uint16_t speech_expand_rate;  // Fraction (of original stream) of synthesized
>                                 // speech inserted through expansion (in Q14).
>-  uint16_t preemptive_rate;  // Fraction of data inserted through pre-emptive
>-                             // expansion (in Q14).
>-  uint16_t accelerate_rate;  // Fraction of data removed through acceleration
>-                             // (in Q14).
>-  uint16_t secondary_decoded_rate;  // Fraction of data coming from FEC/RED
>-                                    // decoding (in Q14).
>+  uint16_t preemptive_rate;     // Fraction of data inserted through pre-emptive
>+                                // expansion (in Q14).
>+  uint16_t accelerate_rate;     // Fraction of data removed through acceleration
>+                                // (in Q14).
>+  uint16_t secondary_decoded_rate;    // Fraction of data coming from FEC/RED
>+                                      // decoding (in Q14).
>   uint16_t secondary_discarded_rate;  // Fraction of discarded FEC/RED data (in
>                                       // Q14).
>-  int32_t clockdrift_ppm;  // Average clock-drift in parts-per-million
>-                           // (positive or negative).
>+  int32_t clockdrift_ppm;     // Average clock-drift in parts-per-million
>+                              // (positive or negative).
>   size_t added_zero_samples;  // Number of zero samples added in "off" mode.
>   // Statistics for packet waiting times, i.e., the time between a packet
>   // arrives until it is decoded.
>@@ -69,52 +69,34 @@ struct NetEqLifetimeStatistics {
>   uint64_t concealed_samples = 0;
>   uint64_t concealment_events = 0;
>   uint64_t jitter_buffer_delay_ms = 0;
>-};
>-
>-enum NetEqPlayoutMode {
>-  kPlayoutOn,
>-  kPlayoutOff,
>-  kPlayoutFax,
>-  kPlayoutStreaming
>+  // Below stat is not part of the spec.
>+  uint64_t voice_concealed_samples = 0;
> };
> 
> // This is the interface class for NetEq.
> class NetEq {
>  public:
>-  enum BackgroundNoiseMode {
>-    kBgnOn,    // Default behavior with eternal noise.
>-    kBgnFade,  // Noise fades to zero after some time.
>-    kBgnOff    // Background noise is always zero.
>-  };
>-
>   struct Config {
>-    Config()
>-        : sample_rate_hz(16000),
>-          enable_post_decode_vad(false),
>-          max_packets_in_buffer(50),
>-          // |max_delay_ms| has the same effect as calling SetMaximumDelay().
>-          max_delay_ms(2000),
>-          background_noise_mode(kBgnOff),
>-          playout_mode(kPlayoutOn),
>-          enable_fast_accelerate(false) {}
>+    Config();
>+    Config(const Config&);
>+    Config(Config&&);
>+    ~Config();
>+    Config& operator=(const Config&);
>+    Config& operator=(Config&&);
> 
>     std::string ToString() const;
> 
>-    int sample_rate_hz;  // Initial value. Will change with input data.
>-    bool enable_post_decode_vad;
>-    size_t max_packets_in_buffer;
>-    int max_delay_ms;
>-    BackgroundNoiseMode background_noise_mode;
>-    NetEqPlayoutMode playout_mode;
>-    bool enable_fast_accelerate;
>+    int sample_rate_hz = 16000;  // Initial value. Will change with input data.
>+    bool enable_post_decode_vad = false;
>+    size_t max_packets_in_buffer = 50;
>+    int max_delay_ms = 2000;
>+    bool enable_fast_accelerate = false;
>     bool enable_muted_state = false;
>+    absl::optional<AudioCodecPairId> codec_pair_id;
>+    bool for_test_no_time_stretching = false;  // Use only for testing.
>   };
> 
>-  enum ReturnCodes {
>-    kOK = 0,
>-    kFail = -1,
>-    kNotImplemented = -2
>-  };
>+  enum ReturnCodes { kOK = 0, kFail = -1, kNotImplemented = -2 };
> 
>   // Creates a new NetEq object, with parameters set in |config|. The |config|
>   // object will only have to be valid for the duration of the call to this
>@@ -219,16 +201,6 @@ class NetEq {
>   // The packet buffer part of the delay is not updated during DTX/CNG periods.
>   virtual int FilteredCurrentDelayMs() const = 0;
> 
>-  // Sets the playout mode to |mode|.
>-  // Deprecated. Set the mode in the Config struct passed to the constructor.
>-  // TODO(henrik.lundin) Delete.
>-  virtual void SetPlayoutMode(NetEqPlayoutMode mode) = 0;
>-
>-  // Returns the current playout mode.
>-  // Deprecated.
>-  // TODO(henrik.lundin) Delete.
>-  virtual NetEqPlayoutMode PlayoutMode() const = 0;
>-
>   // Writes the current network statistics to |stats|. The statistics are reset
>   // after the call.
>   virtual int NetworkStatistics(NetEqNetworkStatistics* stats) = 0;
>@@ -253,7 +225,7 @@ class NetEq {
> 
>   // Returns the RTP timestamp for the last sample delivered by GetAudio().
>   // The return value will be empty if no valid timestamp is available.
>-  virtual rtc::Optional<uint32_t> GetPlayoutTimestamp() const = 0;
>+  virtual absl::optional<uint32_t> GetPlayoutTimestamp() const = 0;
> 
>   // Returns the sample rate in Hz of the audio produced in the last GetAudio
>   // call. If GetAudio has not been called yet, the configured sample rate
>@@ -262,11 +234,11 @@ class NetEq {
> 
>   // Returns info about the decoder for the given payload type, or an empty
>   // value if we have no decoder for that payload type.
>-  virtual rtc::Optional<CodecInst> GetDecoder(int payload_type) const = 0;
>+  virtual absl::optional<CodecInst> GetDecoder(int payload_type) const = 0;
> 
>   // Returns the decoder format for the given payload type. Returns empty if no
>   // such payload type was registered.
>-  virtual rtc::Optional<SdpAudioFormat> GetDecoderFormat(
>+  virtual absl::optional<SdpAudioFormat> GetDecoderFormat(
>       int payload_type) const = 0;
> 
>   // Not implemented.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/merge.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/merge.cc
>index 9bc509b9073..3c9ad19d058 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/merge.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/merge.cc
>@@ -43,11 +43,11 @@ Merge::Merge(int fs_hz,
> 
> Merge::~Merge() = default;
> 
>-size_t Merge::Process(int16_t* input, size_t input_length,
>-                      int16_t* external_mute_factor_array,
>+size_t Merge::Process(int16_t* input,
>+                      size_t input_length,
>                       AudioMultiVector* output) {
>   // TODO(hlundin): Change to an enumerator and skip assert.
>-  assert(fs_hz_ == 8000 || fs_hz_ == 16000 || fs_hz_ ==  32000 ||
>+  assert(fs_hz_ == 8000 || fs_hz_ == 16000 || fs_hz_ == 32000 ||
>          fs_hz_ == 48000);
>   assert(fs_hz_ <= kMaxSampleRate);  // Should not be possible.
> 
>@@ -69,24 +69,13 @@ size_t Merge::Process(int16_t* input, size_t input_length,
>       new int16_t[input_length_per_channel]);
>   std::unique_ptr<int16_t[]> expanded_channel(new int16_t[expanded_length]);
>   for (size_t channel = 0; channel < num_channels_; ++channel) {
>-    input_vector[channel].CopyTo(
>-        input_length_per_channel, 0, input_channel.get());
>+    input_vector[channel].CopyTo(input_length_per_channel, 0,
>+                                 input_channel.get());
>     expanded_[channel].CopyTo(expanded_length, 0, expanded_channel.get());
> 
>-    int16_t new_mute_factor = SignalScaling(
>-        input_channel.get(), input_length_per_channel, expanded_channel.get());
>-
>-    // Adjust muting factor (product of "main" muting factor and expand muting
>-    // factor).
>-    int16_t* external_mute_factor = &external_mute_factor_array[channel];
>-    *external_mute_factor =
>-        (*external_mute_factor * expand_->MuteFactor(channel)) >> 14;
>-
>-    // Update |external_mute_factor| if it is lower than |new_mute_factor|.
>-    if (new_mute_factor > *external_mute_factor) {
>-      *external_mute_factor = std::min(new_mute_factor,
>-                                       static_cast<int16_t>(16384));
>-    }
>+    const int16_t new_mute_factor = std::min<int16_t>(
>+        16384, SignalScaling(input_channel.get(), input_length_per_channel,
>+                             expanded_channel.get()));
> 
>     if (channel == 0) {
>       // Downsample, correlate, and find strongest correlation period for the
>@@ -105,23 +94,29 @@ size_t Merge::Process(int16_t* input, size_t input_length,
> 
>     // Mute the new decoded data if needed (and unmute it linearly).
>     // This is the overlapping part of expanded_signal.
>-    size_t interpolation_length = std::min(
>-        kMaxCorrelationLength * fs_mult_,
>-        expanded_length - best_correlation_index);
>-    interpolation_length = std::min(interpolation_length,
>-                                    input_length_per_channel);
>-    if (*external_mute_factor < 16384) {
>+    size_t interpolation_length =
>+        std::min(kMaxCorrelationLength * fs_mult_,
>+                 expanded_length - best_correlation_index);
>+    interpolation_length =
>+        std::min(interpolation_length, input_length_per_channel);
>+
>+    RTC_DCHECK_LE(new_mute_factor, 16384);
>+    int16_t mute_factor =
>+        std::max(expand_->MuteFactor(channel), new_mute_factor);
>+    RTC_DCHECK_GE(mute_factor, 0);
>+
>+    if (mute_factor < 16384) {
>       // Set a suitable muting slope (Q20). 0.004 for NB, 0.002 for WB,
>-      // and so on.
>-      int increment = 4194 / fs_mult_;
>-      *external_mute_factor =
>-          static_cast<int16_t>(DspHelper::RampSignal(input_channel.get(),
>-                                                     interpolation_length,
>-                                                     *external_mute_factor,
>-                                                     increment));
>+      // and so on, or as fast as it takes to come back to full gain within the
>+      // frame length.
>+      const int back_to_fullscale_inc = static_cast<int>(
>+          ((16384 - mute_factor) << 6) / input_length_per_channel);
>+      const int increment = std::max(4194 / fs_mult_, back_to_fullscale_inc);
>+      mute_factor = static_cast<int16_t>(DspHelper::RampSignal(
>+          input_channel.get(), interpolation_length, mute_factor, increment));
>       DspHelper::UnmuteSignal(&input_channel[interpolation_length],
>                               input_length_per_channel - interpolation_length,
>-                              external_mute_factor, increment,
>+                              &mute_factor, increment,
>                               &decoded_output[interpolation_length]);
>     } else {
>       // No muting needed.
>@@ -134,12 +129,12 @@ size_t Merge::Process(int16_t* input, size_t input_length,
>     // Do overlap and mix linearly.
>     int16_t increment =
>         static_cast<int16_t>(16384 / (interpolation_length + 1));  // In Q14.
>-    int16_t mute_factor = 16384 - increment;
>+    int16_t local_mute_factor = 16384 - increment;
>     memmove(temp_data_.data(), expanded_channel.get(),
>             sizeof(int16_t) * best_correlation_index);
>     DspHelper::CrossFade(&expanded_channel[best_correlation_index],
>                          input_channel.get(), interpolation_length,
>-                         &mute_factor, increment, decoded_output);
>+                         &local_mute_factor, increment, decoded_output);
> 
>     output_length = best_correlation_index + input_length_per_channel;
>     if (channel == 0) {
>@@ -209,30 +204,28 @@ size_t Merge::GetExpandedSignal(size_t* old_length, size_t* expand_period) {
>   return required_length;
> }
> 
>-int16_t Merge::SignalScaling(const int16_t* input, size_t input_length,
>+int16_t Merge::SignalScaling(const int16_t* input,
>+                             size_t input_length,
>                              const int16_t* expanded_signal) const {
>   // Adjust muting factor if new vector is more or less of the BGN energy.
>   const auto mod_input_length = rtc::SafeMin<size_t>(
>       64 * rtc::dchecked_cast<size_t>(fs_mult_), input_length);
>   const int16_t expanded_max =
>       WebRtcSpl_MaxAbsValueW16(expanded_signal, mod_input_length);
>-  int32_t factor = (expanded_max * expanded_max) /
>-      (std::numeric_limits<int32_t>::max() /
>-          static_cast<int32_t>(mod_input_length));
>+  int32_t factor =
>+      (expanded_max * expanded_max) / (std::numeric_limits<int32_t>::max() /
>+                                       static_cast<int32_t>(mod_input_length));
>   const int expanded_shift = factor == 0 ? 0 : 31 - WebRtcSpl_NormW32(factor);
>-  int32_t energy_expanded = WebRtcSpl_DotProductWithScale(expanded_signal,
>-                                                          expanded_signal,
>-                                                          mod_input_length,
>-                                                          expanded_shift);
>+  int32_t energy_expanded = WebRtcSpl_DotProductWithScale(
>+      expanded_signal, expanded_signal, mod_input_length, expanded_shift);
> 
>   // Calculate energy of input signal.
>   const int16_t input_max = WebRtcSpl_MaxAbsValueW16(input, mod_input_length);
>   factor = (input_max * input_max) / (std::numeric_limits<int32_t>::max() /
>-      static_cast<int32_t>(mod_input_length));
>+                                      static_cast<int32_t>(mod_input_length));
>   const int input_shift = factor == 0 ? 0 : 31 - WebRtcSpl_NormW32(factor);
>-  int32_t energy_input = WebRtcSpl_DotProductWithScale(input, input,
>-                                                       mod_input_length,
>-                                                       input_shift);
>+  int32_t energy_input = WebRtcSpl_DotProductWithScale(
>+      input, input, mod_input_length, input_shift);
> 
>   // Align to the same Q-domain.
>   if (input_shift > expanded_shift) {
>@@ -263,8 +256,10 @@ int16_t Merge::SignalScaling(const int16_t* input, size_t input_length,
> 
> // TODO(hlundin): There are some parameter values in this method that seem
> // strange. Compare with Expand::Correlation.
>-void Merge::Downsample(const int16_t* input, size_t input_length,
>-                       const int16_t* expanded_signal, size_t expanded_length) {
>+void Merge::Downsample(const int16_t* input,
>+                       size_t input_length,
>+                       const int16_t* expanded_signal,
>+                       size_t expanded_length) {
>   const int16_t* filter_coefficients;
>   size_t num_coefficients;
>   int decimation_factor = fs_hz_ / 4000;
>@@ -284,14 +279,18 @@ void Merge::Downsample(const int16_t* input, size_t input_length,
>     num_coefficients = 7;
>   }
>   size_t signal_offset = num_coefficients - 1;
>-  WebRtcSpl_DownsampleFast(&expanded_signal[signal_offset],
>-                           expanded_length - signal_offset,
>-                           expanded_downsampled_, kExpandDownsampLength,
>-                           filter_coefficients, num_coefficients,
>-                           decimation_factor, kCompensateDelay);
>+  WebRtcSpl_DownsampleFast(
>+      &expanded_signal[signal_offset], expanded_length - signal_offset,
>+      expanded_downsampled_, kExpandDownsampLength, filter_coefficients,
>+      num_coefficients, decimation_factor, kCompensateDelay);
>   if (input_length <= length_limit) {
>     // Not quite long enough, so we have to cheat a bit.
>-    size_t temp_len = input_length - signal_offset;
>+    // If the input is really short, we'll just use the input length as is, and
>+    // won't bother with correcting for the offset. This is clearly a
>+    // pathological case, and the signal quality will suffer.
>+    const size_t temp_len = input_length > signal_offset
>+                                ? input_length - signal_offset
>+                                : input_length;
>     // TODO(hlundin): Should |downsamp_temp_len| be corrected for round-off
>     // errors? I.e., (temp_len + decimation_factor - 1) / decimation_factor?
>     size_t downsamp_temp_len = temp_len / decimation_factor;
>@@ -302,15 +301,15 @@ void Merge::Downsample(const int16_t* input, size_t input_length,
>     memset(&input_downsampled_[downsamp_temp_len], 0,
>            sizeof(int16_t) * (kInputDownsampLength - downsamp_temp_len));
>   } else {
>-    WebRtcSpl_DownsampleFast(&input[signal_offset],
>-                             input_length - signal_offset, input_downsampled_,
>-                             kInputDownsampLength, filter_coefficients,
>-                             num_coefficients, decimation_factor,
>-                             kCompensateDelay);
>+    WebRtcSpl_DownsampleFast(
>+        &input[signal_offset], input_length - signal_offset, input_downsampled_,
>+        kInputDownsampLength, filter_coefficients, num_coefficients,
>+        decimation_factor, kCompensateDelay);
>   }
> }
> 
>-size_t Merge::CorrelateAndPeakSearch(size_t start_position, size_t input_length,
>+size_t Merge::CorrelateAndPeakSearch(size_t start_position,
>+                                     size_t input_length,
>                                      size_t expand_period) const {
>   // Calculate correlation without any normalization.
>   const size_t max_corr_length = kMaxCorrelationLength;
>@@ -329,8 +328,8 @@ size_t Merge::CorrelateAndPeakSearch(size_t start_position, size_t input_length,
>       new int16_t[correlation_buffer_size]);
>   memset(correlation16.get(), 0, correlation_buffer_size * sizeof(int16_t));
>   int16_t* correlation_ptr = &correlation16[pad_length];
>-  int32_t max_correlation = WebRtcSpl_MaxAbsValueW32(correlation,
>-                                                     stop_position_downsamp);
>+  int32_t max_correlation =
>+      WebRtcSpl_MaxAbsValueW32(correlation, stop_position_downsamp);
>   int norm_shift = std::max(0, 17 - WebRtcSpl_NormW32(max_correlation));
>   WebRtcSpl_VectorBitShiftW32ToW16(correlation_ptr, stop_position_downsamp,
>                                    correlation, norm_shift);
>@@ -367,7 +366,7 @@ size_t Merge::CorrelateAndPeakSearch(size_t start_position, size_t input_length,
>   while (((best_correlation_index + input_length) <
>           (timestamps_per_call_ + expand_->overlap_length())) ||
>          ((best_correlation_index + input_length) < start_position)) {
>-    assert(false);  // Should never happen.
>+    assert(false);                            // Should never happen.
>     best_correlation_index += expand_period;  // Jump one lag ahead.
>   }
>   return best_correlation_index;
>@@ -377,5 +376,4 @@ size_t Merge::RequiredFutureSamples() {
>   return fs_hz_ / 100 * num_channels_;  // 10 ms.
> }
> 
>-
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/merge.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/merge.h
>index e027cd7963a..235c07f5b10 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/merge.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/merge.h
>@@ -15,7 +15,6 @@
> 
> #include "modules/audio_coding/neteq/audio_multi_vector.h"
> #include "rtc_base/constructormagic.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -43,11 +42,9 @@ class Merge {
>   // |input|, having |input_length| samples in total for all channels
>   // (interleaved). The result is written to |output|. The number of channels
>   // allocated in |output| defines the number of channels that will be used when
>-  // de-interleaving |input|. The values in |external_mute_factor_array| (Q14)
>-  // will be used to scale the audio, and is updated in the process. The array
>-  // must have |num_channels_| elements.
>-  virtual size_t Process(int16_t* input, size_t input_length,
>-                         int16_t* external_mute_factor_array,
>+  // de-interleaving |input|.
>+  virtual size_t Process(int16_t* input,
>+                         size_t input_length,
>                          AudioMultiVector* output);
> 
>   virtual size_t RequiredFutureSamples();
>@@ -71,19 +68,23 @@ class Merge {
> 
>   // Analyzes |input| and |expanded_signal| and returns muting factor (Q14) to
>   // be used on the new data.
>-  int16_t SignalScaling(const int16_t* input, size_t input_length,
>+  int16_t SignalScaling(const int16_t* input,
>+                        size_t input_length,
>                         const int16_t* expanded_signal) const;
> 
>   // Downsamples |input| (|input_length| samples) and |expanded_signal| to
>   // 4 kHz sample rate. The downsampled signals are written to
>   // |input_downsampled_| and |expanded_downsampled_|, respectively.
>-  void Downsample(const int16_t* input, size_t input_length,
>-                  const int16_t* expanded_signal, size_t expanded_length);
>+  void Downsample(const int16_t* input,
>+                  size_t input_length,
>+                  const int16_t* expanded_signal,
>+                  size_t expanded_length);
> 
>   // Calculates cross-correlation between |input_downsampled_| and
>   // |expanded_downsampled_|, and finds the correlation maximum. The maximizing
>   // lag is returned.
>-  size_t CorrelateAndPeakSearch(size_t start_position, size_t input_length,
>+  size_t CorrelateAndPeakSearch(size_t start_position,
>+                                size_t input_length,
>                                 size_t expand_period) const;
> 
>   const int fs_mult_;  // fs_hz_ / 8000.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/mock/mock_buffer_level_filter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/mock/mock_buffer_level_filter.h
>index f662fb6ec0e..bf9fd59c9ac 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/mock/mock_buffer_level_filter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/mock/mock_buffer_level_filter.h
>@@ -20,17 +20,14 @@ namespace webrtc {
> class MockBufferLevelFilter : public BufferLevelFilter {
>  public:
>   virtual ~MockBufferLevelFilter() { Die(); }
>-  MOCK_METHOD0(Die,
>-      void());
>-  MOCK_METHOD0(Reset,
>-      void());
>+  MOCK_METHOD0(Die, void());
>+  MOCK_METHOD0(Reset, void());
>   MOCK_METHOD3(Update,
>-      void(size_t buffer_size_packets, int time_stretched_samples,
>-           size_t packet_len_samples));
>-  MOCK_METHOD1(SetTargetBufferLevel,
>-      void(int target_buffer_level));
>-  MOCK_CONST_METHOD0(filtered_current_level,
>-      int());
>+               void(size_t buffer_size_packets,
>+                    int time_stretched_samples,
>+                    size_t packet_len_samples));
>+  MOCK_METHOD1(SetTargetBufferLevel, void(int target_buffer_level));
>+  MOCK_CONST_METHOD0(filtered_current_level, int());
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h
>index 049b6932107..b1d815141e8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h
>@@ -23,18 +23,16 @@ class MockDecoderDatabase : public DecoderDatabase {
>  public:
>   explicit MockDecoderDatabase(
>       rtc::scoped_refptr<AudioDecoderFactory> factory = nullptr)
>-      : DecoderDatabase(factory) {}
>+      : DecoderDatabase(factory, absl::nullopt) {}
>   virtual ~MockDecoderDatabase() { Die(); }
>   MOCK_METHOD0(Die, void());
>-  MOCK_CONST_METHOD0(Empty,
>-      bool());
>-  MOCK_CONST_METHOD0(Size,
>-      int());
>-  MOCK_METHOD0(Reset,
>-      void());
>+  MOCK_CONST_METHOD0(Empty, bool());
>+  MOCK_CONST_METHOD0(Size, int());
>+  MOCK_METHOD0(Reset, void());
>   MOCK_METHOD3(RegisterPayload,
>-      int(uint8_t rtp_payload_type, NetEqDecoder codec_type,
>-          const std::string& name));
>+               int(uint8_t rtp_payload_type,
>+                   NetEqDecoder codec_type,
>+                   const std::string& name));
>   MOCK_METHOD2(RegisterPayload,
>                int(int rtp_payload_type, const SdpAudioFormat& audio_format));
>   MOCK_METHOD4(InsertExternal,
>@@ -42,19 +40,15 @@ class MockDecoderDatabase : public DecoderDatabase {
>                    NetEqDecoder codec_type,
>                    const std::string& codec_name,
>                    AudioDecoder* decoder));
>-  MOCK_METHOD1(Remove,
>-      int(uint8_t rtp_payload_type));
>+  MOCK_METHOD1(Remove, int(uint8_t rtp_payload_type));
>   MOCK_METHOD0(RemoveAll, void());
>   MOCK_CONST_METHOD1(GetDecoderInfo,
>-      const DecoderInfo*(uint8_t rtp_payload_type));
>+                     const DecoderInfo*(uint8_t rtp_payload_type));
>   MOCK_METHOD2(SetActiveDecoder,
>-      int(uint8_t rtp_payload_type, bool* new_decoder));
>-  MOCK_CONST_METHOD0(GetActiveDecoder,
>-      AudioDecoder*());
>-  MOCK_METHOD1(SetActiveCngDecoder,
>-      int(uint8_t rtp_payload_type));
>-  MOCK_CONST_METHOD0(GetActiveCngDecoder,
>-      ComfortNoiseDecoder*());
>+               int(uint8_t rtp_payload_type, bool* new_decoder));
>+  MOCK_CONST_METHOD0(GetActiveDecoder, AudioDecoder*());
>+  MOCK_METHOD1(SetActiveCngDecoder, int(uint8_t rtp_payload_type));
>+  MOCK_CONST_METHOD0(GetActiveCngDecoder, ComfortNoiseDecoder*());
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/mock/mock_delay_manager.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/mock/mock_delay_manager.h
>index 61f209d5c49..9b2ed498d9d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/mock/mock_delay_manager.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/mock/mock_delay_manager.h
>@@ -25,37 +25,25 @@ class MockDelayManager : public DelayManager {
>       : DelayManager(max_packets_in_buffer, peak_detector, tick_timer) {}
>   virtual ~MockDelayManager() { Die(); }
>   MOCK_METHOD0(Die, void());
>-  MOCK_CONST_METHOD0(iat_vector,
>-      const IATVector&());
>+  MOCK_CONST_METHOD0(iat_vector, const IATVector&());
>   MOCK_METHOD3(Update,
>-      int(uint16_t sequence_number, uint32_t timestamp, int sample_rate_hz));
>-  MOCK_METHOD1(CalculateTargetLevel,
>-      int(int iat_packets));
>-  MOCK_METHOD1(SetPacketAudioLength,
>-      int(int length_ms));
>-  MOCK_METHOD0(Reset,
>-      void());
>-  MOCK_CONST_METHOD0(PeakFound,
>-      bool());
>-  MOCK_METHOD1(UpdateCounters,
>-      void(int elapsed_time_ms));
>-  MOCK_METHOD0(ResetPacketIatCount,
>-      void());
>-  MOCK_CONST_METHOD2(BufferLimits,
>-      void(int* lower_limit, int* higher_limit));
>-  MOCK_CONST_METHOD0(TargetLevel,
>-      int());
>+               int(uint16_t sequence_number,
>+                   uint32_t timestamp,
>+                   int sample_rate_hz));
>+  MOCK_METHOD1(CalculateTargetLevel, int(int iat_packets));
>+  MOCK_METHOD1(SetPacketAudioLength, int(int length_ms));
>+  MOCK_METHOD0(Reset, void());
>+  MOCK_CONST_METHOD0(PeakFound, bool());
>+  MOCK_METHOD1(UpdateCounters, void(int elapsed_time_ms));
>+  MOCK_METHOD0(ResetPacketIatCount, void());
>+  MOCK_CONST_METHOD2(BufferLimits, void(int* lower_limit, int* higher_limit));
>+  MOCK_CONST_METHOD0(TargetLevel, int());
>   MOCK_METHOD0(RegisterEmptyPacket, void());
>-  MOCK_METHOD1(set_extra_delay_ms,
>-      void(int16_t delay));
>-  MOCK_CONST_METHOD0(base_target_level,
>-      int());
>-  MOCK_METHOD1(set_streaming_mode,
>-      void(bool value));
>-  MOCK_CONST_METHOD0(last_pack_cng_or_dtmf,
>-      int());
>-  MOCK_METHOD1(set_last_pack_cng_or_dtmf,
>-      void(int value));
>+  MOCK_METHOD1(set_extra_delay_ms, void(int16_t delay));
>+  MOCK_CONST_METHOD0(base_target_level, int());
>+  MOCK_METHOD1(set_streaming_mode, void(bool value));
>+  MOCK_CONST_METHOD0(last_pack_cng_or_dtmf, int());
>+  MOCK_METHOD1(set_last_pack_cng_or_dtmf, void(int value));
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/mock/mock_dtmf_buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/mock/mock_dtmf_buffer.h
>index 153a4d716aa..11b571f86c4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/mock/mock_dtmf_buffer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/mock/mock_dtmf_buffer.h
>@@ -22,16 +22,11 @@ class MockDtmfBuffer : public DtmfBuffer {
>   MockDtmfBuffer(int fs) : DtmfBuffer(fs) {}
>   virtual ~MockDtmfBuffer() { Die(); }
>   MOCK_METHOD0(Die, void());
>-  MOCK_METHOD0(Flush,
>-      void());
>-  MOCK_METHOD1(InsertEvent,
>-      int(const DtmfEvent& event));
>-  MOCK_METHOD2(GetEvent,
>-      bool(uint32_t current_timestamp, DtmfEvent* event));
>-  MOCK_CONST_METHOD0(Length,
>-      size_t());
>-  MOCK_CONST_METHOD0(Empty,
>-      bool());
>+  MOCK_METHOD0(Flush, void());
>+  MOCK_METHOD1(InsertEvent, int(const DtmfEvent& event));
>+  MOCK_METHOD2(GetEvent, bool(uint32_t current_timestamp, DtmfEvent* event));
>+  MOCK_CONST_METHOD0(Length, size_t());
>+  MOCK_CONST_METHOD0(Empty, bool());
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/mock/mock_dtmf_tone_generator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/mock/mock_dtmf_tone_generator.h
>index 2cb59808912..be4b7b5f610 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/mock/mock_dtmf_tone_generator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/mock/mock_dtmf_tone_generator.h
>@@ -21,14 +21,10 @@ class MockDtmfToneGenerator : public DtmfToneGenerator {
>  public:
>   virtual ~MockDtmfToneGenerator() { Die(); }
>   MOCK_METHOD0(Die, void());
>-  MOCK_METHOD3(Init,
>-      int(int fs, int event, int attenuation));
>-  MOCK_METHOD0(Reset,
>-      void());
>-  MOCK_METHOD2(Generate,
>-      int(size_t num_samples, AudioMultiVector* output));
>-  MOCK_CONST_METHOD0(initialized,
>-      bool());
>+  MOCK_METHOD3(Init, int(int fs, int event, int attenuation));
>+  MOCK_METHOD0(Reset, void());
>+  MOCK_METHOD2(Generate, int(size_t num_samples, AudioMultiVector* output));
>+  MOCK_CONST_METHOD0(initialized, bool());
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/mock/mock_expand.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/mock/mock_expand.h
>index 05fdaec750d..aed01646f51 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/mock/mock_expand.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/mock/mock_expand.h
>@@ -33,16 +33,11 @@ class MockExpand : public Expand {
>                num_channels) {}
>   virtual ~MockExpand() { Die(); }
>   MOCK_METHOD0(Die, void());
>-  MOCK_METHOD0(Reset,
>-      void());
>-  MOCK_METHOD1(Process,
>-      int(AudioMultiVector* output));
>-  MOCK_METHOD0(SetParametersForNormalAfterExpand,
>-      void());
>-  MOCK_METHOD0(SetParametersForMergeAfterExpand,
>-      void());
>-  MOCK_CONST_METHOD0(overlap_length,
>-      size_t());
>+  MOCK_METHOD0(Reset, void());
>+  MOCK_METHOD1(Process, int(AudioMultiVector* output));
>+  MOCK_METHOD0(SetParametersForNormalAfterExpand, void());
>+  MOCK_METHOD0(SetParametersForMergeAfterExpand, void());
>+  MOCK_CONST_METHOD0(overlap_length, size_t());
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h
>index b3152405b5a..1fb3db5f965 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h
>@@ -15,7 +15,6 @@
> #include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
> #include "rtc_base/constructormagic.h"
> #include "test/gmock.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -75,17 +74,16 @@ class MockExternalPcm16B : public AudioDecoder {
>                    int sample_rate_hz,
>                    int16_t* decoded,
>                    SpeechType* speech_type));
>-  MOCK_CONST_METHOD0(HasDecodePlc,
>-      bool());
>-  MOCK_METHOD2(DecodePlc,
>-      size_t(size_t num_frames, int16_t* decoded));
>+  MOCK_CONST_METHOD0(HasDecodePlc, bool());
>+  MOCK_METHOD2(DecodePlc, size_t(size_t num_frames, int16_t* decoded));
>   MOCK_METHOD0(Reset, void());
>   MOCK_METHOD5(IncomingPacket,
>-      int(const uint8_t* payload, size_t payload_len,
>-          uint16_t rtp_sequence_number, uint32_t rtp_timestamp,
>-          uint32_t arrival_timestamp));
>-  MOCK_METHOD0(ErrorCode,
>-      int());
>+               int(const uint8_t* payload,
>+                   size_t payload_len,
>+                   uint16_t rtp_sequence_number,
>+                   uint32_t rtp_timestamp,
>+                   uint32_t arrival_timestamp));
>+  MOCK_METHOD0(ErrorCode, int());
> 
>   int SampleRateHz() const /* override */ { return real_.SampleRateHz(); }
>   size_t Channels() const /* override */ { return real_.Channels(); }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/mock/mock_packet_buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/mock/mock_packet_buffer.h
>index ac7d9b7a250..b477b1ab6b4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/mock/mock_packet_buffer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/mock/mock_packet_buffer.h
>@@ -38,8 +38,8 @@ class MockPacketBuffer : public PacketBuffer {
>   MOCK_METHOD5(InsertPacketList,
>                int(PacketList* packet_list,
>                    const DecoderDatabase& decoder_database,
>-                   rtc::Optional<uint8_t>* current_rtp_payload_type,
>-                   rtc::Optional<uint8_t>* current_cng_rtp_payload_type,
>+                   absl::optional<uint8_t>* current_rtp_payload_type,
>+                   absl::optional<uint8_t>* current_cng_rtp_payload_type,
>                    StatisticsCalculator* stats));
>   MOCK_CONST_METHOD1(NextTimestamp,
>       int(uint32_t* next_timestamp));
>@@ -47,8 +47,7 @@ class MockPacketBuffer : public PacketBuffer {
>       int(uint32_t timestamp, uint32_t* next_timestamp));
>   MOCK_CONST_METHOD0(PeekNextPacket,
>       const Packet*());
>-  MOCK_METHOD0(GetNextPacket,
>-      rtc::Optional<Packet>());
>+  MOCK_METHOD0(GetNextPacket, absl::optional<Packet>());
>   MOCK_METHOD1(DiscardNextPacket, int(StatisticsCalculator* stats));
>   MOCK_METHOD3(DiscardOldPackets,
>                void(uint32_t timestamp_limit,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/mock/mock_red_payload_splitter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/mock/mock_red_payload_splitter.h
>index 27a22766644..426c467afa4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/mock/mock_red_payload_splitter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/mock/mock_red_payload_splitter.h
>@@ -21,8 +21,8 @@ class MockRedPayloadSplitter : public RedPayloadSplitter {
>  public:
>   MOCK_METHOD1(SplitRed, bool(PacketList* packet_list));
>   MOCK_METHOD2(CheckRedPayloads,
>-               int(PacketList* packet_list,
>-                   const DecoderDatabase& decoder_database));
>+               void(PacketList* packet_list,
>+                    const DecoderDatabase& decoder_database));
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/nack_tracker.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/nack_tracker.cc
>index d187883916d..c62cdf88e4d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/nack_tracker.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/nack_tracker.cc
>@@ -14,7 +14,6 @@
> 
> #include <algorithm>  // For std::max.
> 
>-#include "modules/include/module_common_types.h"
> #include "rtc_base/checks.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/nack_tracker.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/nack_tracker.h
>index 4f88d917c20..1936a944ccb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/nack_tracker.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/nack_tracker.h
>@@ -11,10 +11,11 @@
> #ifndef MODULES_AUDIO_CODING_NETEQ_NACK_TRACKER_H_
> #define MODULES_AUDIO_CODING_NETEQ_NACK_TRACKER_H_
> 
>-#include <vector>
> #include <map>
>+#include <vector>
> 
> #include "modules/audio_coding/include/audio_coding_module_typedefs.h"
>+#include "modules/include/module_common_types.h"
> #include "rtc_base/gtest_prod_util.h"
> 
> //
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/nack_tracker_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/nack_tracker_unittest.cc
>index 19adf30bf6b..a44f41ba412 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/nack_tracker_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/nack_tracker_unittest.cc
>@@ -17,7 +17,6 @@
> 
> #include "modules/audio_coding/include/audio_coding_module_typedefs.h"
> #include "test/gtest.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> namespace {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq.cc
>index 8b74973afff..cf1c6aa6653 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq.cc
>@@ -11,23 +11,28 @@
> #include "modules/audio_coding/neteq/include/neteq.h"
> 
> #include <memory>
>-#include <sstream>
> 
> #include "modules/audio_coding/neteq/neteq_impl.h"
>+#include "rtc_base/strings/string_builder.h"
> 
> namespace webrtc {
> 
>+NetEq::Config::Config() = default;
>+NetEq::Config::Config(const Config&) = default;
>+NetEq::Config::Config(Config&&) = default;
>+NetEq::Config::~Config() = default;
>+NetEq::Config& NetEq::Config::operator=(const Config&) = default;
>+NetEq::Config& NetEq::Config::operator=(Config&&) = default;
>+
> std::string NetEq::Config::ToString() const {
>-  std::stringstream ss;
>-  ss << "sample_rate_hz=" << sample_rate_hz
>-     << ", enable_post_decode_vad="
>+  char buf[1024];
>+  rtc::SimpleStringBuilder ss(buf);
>+  ss << "sample_rate_hz=" << sample_rate_hz << ", enable_post_decode_vad="
>      << (enable_post_decode_vad ? "true" : "false")
>      << ", max_packets_in_buffer=" << max_packets_in_buffer
>-     << ", background_noise_mode=" << background_noise_mode
>-     << ", playout_mode=" << playout_mode
>      << ", enable_fast_accelerate="
>-     << (enable_fast_accelerate ? " true": "false")
>-     << ", enable_muted_state=" << (enable_muted_state ? " true": "false");
>+     << (enable_fast_accelerate ? " true" : "false")
>+     << ", enable_muted_state=" << (enable_muted_state ? " true" : "false");
>   return ss.str();
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq_decoder_enum.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq_decoder_enum.cc
>index 8d66c2a25ca..e3b633e85d2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq_decoder_enum.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq_decoder_enum.cc
>@@ -15,7 +15,7 @@
> 
> namespace webrtc {
> 
>-rtc::Optional<SdpAudioFormat> NetEqDecoderToSdpAudioFormat(NetEqDecoder nd) {
>+absl::optional<SdpAudioFormat> NetEqDecoderToSdpAudioFormat(NetEqDecoder nd) {
>   switch (nd) {
>     case NetEqDecoder::kDecoderPCMu:
>       return SdpAudioFormat("pcmu", 8000, 1);
>@@ -78,7 +78,7 @@ rtc::Optional<SdpAudioFormat> NetEqDecoderToSdpAudioFormat(NetEqDecoder nd) {
>     case NetEqDecoder::kDecoderCNGswb48kHz:
>       return SdpAudioFormat("cn", 48000, 1);
>     default:
>-      return rtc::nullopt;
>+      return absl::nullopt;
>   }
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq_decoder_enum.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq_decoder_enum.h
>index 024f03cd57f..00629bc8316 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq_decoder_enum.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq_decoder_enum.h
>@@ -11,8 +11,8 @@
> #ifndef MODULES_AUDIO_CODING_NETEQ_NETEQ_DECODER_ENUM_H_
> #define MODULES_AUDIO_CODING_NETEQ_NETEQ_DECODER_ENUM_H_
> 
>+#include "absl/types/optional.h"
> #include "api/audio_codecs/audio_format.h"
>-#include "api/optional.h"
> 
> namespace webrtc {
> 
>@@ -49,7 +49,7 @@ enum class NetEqDecoder {
>   kDecoderOpus_2ch,
> };
> 
>-rtc::Optional<SdpAudioFormat> NetEqDecoderToSdpAudioFormat(NetEqDecoder nd);
>+absl::optional<SdpAudioFormat> NetEqDecoderToSdpAudioFormat(NetEqDecoder nd);
> 
> }  // namespace webrtc
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc
>index ec166277f81..5c350bb47a0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc
>@@ -12,13 +12,13 @@
> 
> #include <memory>
> 
>+#include "api/audio/audio_frame.h"
> #include "api/audio_codecs/builtin_audio_decoder_factory.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h"
> #include "modules/audio_coding/neteq/tools/input_audio_file.h"
> #include "modules/audio_coding/neteq/tools/neteq_external_decoder_test.h"
> #include "modules/audio_coding/neteq/tools/rtp_generator.h"
>-#include "modules/include/module_common_types.h"
> #include "test/gmock.h"
> #include "test/testsupport/fileutils.h"
> 
>@@ -55,8 +55,8 @@ class NetEqExternalDecoderUnitTest : public test::NetEqExternalDecoderTest {
>   }
> 
>   virtual ~NetEqExternalDecoderUnitTest() {
>-    delete [] input_;
>-    delete [] encoded_;
>+    delete[] input_;
>+    delete[] encoded_;
>     // ~NetEqExternalDecoderTest() will delete |external_decoder_|, so expecting
>     // Die() to be called.
>     EXPECT_CALL(*external_decoder_, Die()).Times(1);
>@@ -75,8 +75,8 @@ class NetEqExternalDecoderUnitTest : public test::NetEqExternalDecoderTest {
>     if (!input_file_->Read(frame_size_samples_, input_)) {
>       return -1;
>     }
>-    payload_size_bytes_ = WebRtcPcm16b_Encode(input_, frame_size_samples_,
>-                                              encoded_);
>+    payload_size_bytes_ =
>+        WebRtcPcm16b_Encode(input_, frame_size_samples_, encoded_);
> 
>     int next_send_time = rtp_generator_->GetRtpHeader(
>         kPayloadType, frame_size_samples_, &rtp_header_);
>@@ -111,9 +111,10 @@ class NetEqExternalDecoderUnitTest : public test::NetEqExternalDecoderTest {
>     uint32_t time_now = 0;
>     for (int k = 0; k < num_loops; ++k) {
>       while (time_now >= next_arrival_time) {
>-        InsertPacket(rtp_header_, rtc::ArrayView<const uint8_t>(
>-                                      encoded_, payload_size_bytes_),
>-                     next_arrival_time);
>+        InsertPacket(
>+            rtp_header_,
>+            rtc::ArrayView<const uint8_t>(encoded_, payload_size_bytes_),
>+            next_arrival_time);
>         // Get next input packet.
>         do {
>           next_send_time = GetNewPacket();
>@@ -148,6 +149,7 @@ class NetEqExternalDecoderUnitTest : public test::NetEqExternalDecoderTest {
>   }
> 
>   int samples_per_ms() const { return samples_per_ms_; }
>+
>  private:
>   std::unique_ptr<MockExternalPcm16B> external_decoder_;
>   int samples_per_ms_;
>@@ -337,11 +339,9 @@ TEST_F(LargeTimestampJumpTest, JumpLongerThanHalfRange) {
>       static_cast<uint32_t>(kJumpToTimestamp - kJumpFromTimestamp) > 0x7FFFFFFF,
>       "jump should be larger than half range");
>   // Replace the default RTP generator with one that jumps in timestamp.
>-  ResetRtpGenerator(new test::TimestampJumpRtpGenerator(samples_per_ms(),
>-                                                        kStartSeqeunceNumber,
>-                                                        kStartTimestamp,
>-                                                        kJumpFromTimestamp,
>-                                                        kJumpToTimestamp));
>+  ResetRtpGenerator(new test::TimestampJumpRtpGenerator(
>+      samples_per_ms(), kStartSeqeunceNumber, kStartTimestamp,
>+      kJumpFromTimestamp, kJumpToTimestamp));
> 
>   RunTest(130);  // Run 130 laps @ 10 ms each in the test loop.
>   EXPECT_EQ(kRecovered, test_state_);
>@@ -361,11 +361,9 @@ TEST_F(LargeTimestampJumpTest, JumpLongerThanHalfRangeAndWrap) {
>       static_cast<uint32_t>(kJumpToTimestamp - kJumpFromTimestamp) > 0x7FFFFFFF,
>       "jump should be larger than half range");
>   // Replace the default RTP generator with one that jumps in timestamp.
>-  ResetRtpGenerator(new test::TimestampJumpRtpGenerator(samples_per_ms(),
>-                                                        kStartSeqeunceNumber,
>-                                                        kStartTimestamp,
>-                                                        kJumpFromTimestamp,
>-                                                        kJumpToTimestamp));
>+  ResetRtpGenerator(new test::TimestampJumpRtpGenerator(
>+      samples_per_ms(), kStartSeqeunceNumber, kStartTimestamp,
>+      kJumpFromTimestamp, kJumpToTimestamp));
> 
>   RunTest(130);  // Run 130 laps @ 10 ms each in the test loop.
>   EXPECT_EQ(kRecovered, test_state_);
>@@ -420,11 +418,9 @@ TEST_F(ShortTimestampJumpTest, JumpShorterThanHalfRange) {
>       static_cast<uint32_t>(kJumpToTimestamp - kJumpFromTimestamp) < 0x7FFFFFFF,
>       "jump should be smaller than half range");
>   // Replace the default RTP generator with one that jumps in timestamp.
>-  ResetRtpGenerator(new test::TimestampJumpRtpGenerator(samples_per_ms(),
>-                                                        kStartSeqeunceNumber,
>-                                                        kStartTimestamp,
>-                                                        kJumpFromTimestamp,
>-                                                        kJumpToTimestamp));
>+  ResetRtpGenerator(new test::TimestampJumpRtpGenerator(
>+      samples_per_ms(), kStartSeqeunceNumber, kStartTimestamp,
>+      kJumpFromTimestamp, kJumpToTimestamp));
> 
>   RunTest(130);  // Run 130 laps @ 10 ms each in the test loop.
>   EXPECT_EQ(kRecovered, test_state_);
>@@ -444,11 +440,9 @@ TEST_F(ShortTimestampJumpTest, JumpShorterThanHalfRangeAndWrap) {
>       static_cast<uint32_t>(kJumpToTimestamp - kJumpFromTimestamp) < 0x7FFFFFFF,
>       "jump should be smaller than half range");
>   // Replace the default RTP generator with one that jumps in timestamp.
>-  ResetRtpGenerator(new test::TimestampJumpRtpGenerator(samples_per_ms(),
>-                                                        kStartSeqeunceNumber,
>-                                                        kStartTimestamp,
>-                                                        kJumpFromTimestamp,
>-                                                        kJumpToTimestamp));
>+  ResetRtpGenerator(new test::TimestampJumpRtpGenerator(
>+      samples_per_ms(), kStartSeqeunceNumber, kStartTimestamp,
>+      kJumpFromTimestamp, kJumpToTimestamp));
> 
>   RunTest(130);  // Run 130 laps @ 10 ms each in the test loop.
>   EXPECT_EQ(kRecovered, test_state_);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq_impl.cc
>index 6836c7e59a2..0ef32638d23 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq_impl.cc
>@@ -41,11 +41,12 @@
> #include "modules/audio_coding/neteq/sync_buffer.h"
> #include "modules/audio_coding/neteq/tick_timer.h"
> #include "modules/audio_coding/neteq/timestamp_scaler.h"
>-#include "modules/include/module_common_types.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/numerics/safe_conversions.h"
> #include "rtc_base/sanitizer.h"
>+#include "rtc_base/strings/audio_format_to_string.h"
>+#include "rtc_base/system/fallthrough.h"
> #include "rtc_base/trace_event.h"
> #include "system_wrappers/include/field_trial.h"
> 
>@@ -56,7 +57,8 @@ NetEqImpl::Dependencies::Dependencies(
>     const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory)
>     : tick_timer(new TickTimer),
>       buffer_level_filter(new BufferLevelFilter),
>-      decoder_database(new DecoderDatabase(decoder_factory)),
>+      decoder_database(
>+          new DecoderDatabase(decoder_factory, config.codec_pair_id)),
>       delay_peak_detector(new DelayPeakDetector(tick_timer.get())),
>       delay_manager(new DelayManager(config.max_packets_in_buffer,
>                                      delay_peak_detector.get(),
>@@ -99,13 +101,16 @@ NetEqImpl::NetEqImpl(const NetEq::Config& config,
>       reset_decoder_(false),
>       ssrc_(0),
>       first_packet_(true),
>-      background_noise_mode_(config.background_noise_mode),
>-      playout_mode_(config.playout_mode),
>       enable_fast_accelerate_(config.enable_fast_accelerate),
>       nack_enabled_(false),
>       enable_muted_state_(config.enable_muted_state),
>-      use_dtx_delay_fix_(
>-          field_trial::IsEnabled("WebRTC-NetEqOpusDtxDelayFix")) {
>+      expand_uma_logger_("WebRTC.Audio.ExpandRatePercent",
>+                         10,  // Report once every 10 s.
>+                         tick_timer_.get()),
>+      speech_expand_uma_logger_("WebRTC.Audio.SpeechExpandRatePercent",
>+                                10,  // Report once every 10 s.
>+                                tick_timer_.get()),
>+      no_time_stretching_(config.for_test_no_time_stretching) {
>   RTC_LOG(LS_INFO) << "NetEq config: " << config.ToString();
>   int fs = config.sample_rate_hz;
>   if (fs != 8000 && fs != 16000 && fs != 32000 && fs != 48000) {
>@@ -262,7 +267,8 @@ int NetEqImpl::RegisterExternalDecoder(AudioDecoder* decoder,
> bool NetEqImpl::RegisterPayloadType(int rtp_payload_type,
>                                     const SdpAudioFormat& audio_format) {
>   RTC_LOG(LS_VERBOSE) << "NetEqImpl::RegisterPayloadType: payload type "
>-                      << rtp_payload_type << ", codec " << audio_format;
>+                      << rtp_payload_type << ", codec "
>+                      << rtc::ToString(audio_format);
>   rtc::CritScope lock(&crit_sect_);
>   return decoder_database_->RegisterPayload(rtp_payload_type, audio_format) ==
>          DecoderDatabase::kOK;
>@@ -352,23 +358,6 @@ int NetEqImpl::FilteredCurrentDelayMs() const {
>   return static_cast<int>(delay_samples) / rtc::CheckedDivExact(fs_hz_, 1000);
> }
> 
>-// Deprecated.
>-// TODO(henrik.lundin) Delete.
>-void NetEqImpl::SetPlayoutMode(NetEqPlayoutMode mode) {
>-  rtc::CritScope lock(&crit_sect_);
>-  if (mode != playout_mode_) {
>-    playout_mode_ = mode;
>-    CreateDecisionLogic();
>-  }
>-}
>-
>-// Deprecated.
>-// TODO(henrik.lundin) Delete.
>-NetEqPlayoutMode NetEqImpl::PlayoutMode() const {
>-  rtc::CritScope lock(&crit_sect_);
>-  return playout_mode_;
>-}
>-
> int NetEqImpl::NetworkStatistics(NetEqNetworkStatistics* stats) {
>   rtc::CritScope lock(&crit_sect_);
>   assert(decoder_database_.get());
>@@ -416,14 +405,14 @@ void NetEqImpl::DisableVad() {
>   vad_->Disable();
> }
> 
>-rtc::Optional<uint32_t> NetEqImpl::GetPlayoutTimestamp() const {
>+absl::optional<uint32_t> NetEqImpl::GetPlayoutTimestamp() const {
>   rtc::CritScope lock(&crit_sect_);
>   if (first_packet_ || last_mode_ == kModeRfc3389Cng ||
>       last_mode_ == kModeCodecInternalCng) {
>     // We don't have a valid RTP timestamp until we have decoded our first
>     // RTP packet. Also, the RTP timestamp is not accurate while playing CNG,
>     // which is indicated by returning an empty value.
>-    return rtc::nullopt;
>+    return absl::nullopt;
>   }
>   return timestamp_scaler_->ToExternal(playout_timestamp_);
> }
>@@ -433,12 +422,12 @@ int NetEqImpl::last_output_sample_rate_hz() const {
>   return last_output_sample_rate_hz_;
> }
> 
>-rtc::Optional<CodecInst> NetEqImpl::GetDecoder(int payload_type) const {
>+absl::optional<CodecInst> NetEqImpl::GetDecoder(int payload_type) const {
>   rtc::CritScope lock(&crit_sect_);
>   const DecoderDatabase::DecoderInfo* di =
>       decoder_database_->GetDecoderInfo(payload_type);
>   if (!di) {
>-    return rtc::nullopt;
>+    return absl::nullopt;
>   }
> 
>   // Create a CodecInst with some fields set. The remaining fields are zeroed,
>@@ -454,13 +443,13 @@ rtc::Optional<CodecInst> NetEqImpl::GetDecoder(int payload_type) const {
>   return ci;
> }
> 
>-rtc::Optional<SdpAudioFormat> NetEqImpl::GetDecoderFormat(
>+absl::optional<SdpAudioFormat> NetEqImpl::GetDecoderFormat(
>     int payload_type) const {
>   rtc::CritScope lock(&crit_sect_);
>   const DecoderDatabase::DecoderInfo* const di =
>       decoder_database_->GetDecoderInfo(payload_type);
>   if (!di) {
>-    return rtc::nullopt;  // Payload type not registered.
>+    return absl::nullopt;  // Payload type not registered.
>   }
>   return di->GetFormat();
> }
>@@ -623,6 +612,9 @@ int NetEqImpl::InsertPacketInternal(const RTPHeader& rtp_header,
>     // Only accept a few RED payloads of the same type as the main data,
>     // DTMF events and CNG.
>     red_payload_splitter_->CheckRedPayloads(&packet_list, *decoder_database_);
>+    if (packet_list.empty()) {
>+      return kRedundancySplitError;
>+    }
>   }
> 
>   // Check payload types.
>@@ -675,8 +667,7 @@ int NetEqImpl::InsertPacketInternal(const RTPHeader& rtp_header,
>     decoder->IncomingPacket(packet_list.front().payload.data(),
>                             packet_list.front().payload.size(),
>                             packet_list.front().sequence_number,
>-                            packet_list.front().timestamp,
>-                            receive_timestamp);
>+                            packet_list.front().timestamp, receive_timestamp);
>   }
> 
>   PacketList parsed_packet_list;
>@@ -697,7 +688,7 @@ int NetEqImpl::InsertPacketInternal(const RTPHeader& rtp_header,
>       const auto sequence_number = packet.sequence_number;
>       const auto payload_type = packet.payload_type;
>       const Packet::Priority original_priority = packet.priority;
>-      auto packet_from_result = [&] (AudioDecoder::ParseResult& result) {
>+      auto packet_from_result = [&](AudioDecoder::ParseResult& result) {
>         Packet new_packet;
>         new_packet.sequence_number = sequence_number;
>         new_packet.payload_type = payload_type;
>@@ -782,8 +773,7 @@ int NetEqImpl::InsertPacketInternal(const RTPHeader& rtp_header,
>     assert(decoder_info);
>     if (decoder_info->SampleRateHz() != fs_hz_ ||
>         channels != algorithm_buffer_->Channels()) {
>-      SetSampleRateAndChannels(decoder_info->SampleRateHz(),
>-                               channels);
>+      SetSampleRateAndChannels(decoder_info->SampleRateHz(), channels);
>     }
>     if (nack_enabled_) {
>       RTC_DCHECK(nack_);
>@@ -835,6 +825,11 @@ int NetEqImpl::GetAudioInternal(AudioFrame* audio_frame, bool* muted) {
>   last_decoded_timestamps_.clear();
>   tick_timer_->Increment();
>   stats_.IncreaseCounter(output_size_samples_, fs_hz_);
>+  const auto lifetime_stats = stats_.GetLifetimeStatistics();
>+  expand_uma_logger_.UpdateSampleCounter(lifetime_stats.concealed_samples,
>+                                         fs_hz_);
>+  speech_expand_uma_logger_.UpdateSampleCounter(
>+      lifetime_stats.voice_concealed_samples, fs_hz_);
> 
>   // Check for muted state.
>   if (enable_muted_state_ && expand_->Muted() && packet_buffer_->Empty()) {
>@@ -855,8 +850,8 @@ int NetEqImpl::GetAudioInternal(AudioFrame* audio_frame, bool* muted) {
>     return 0;
>   }
> 
>-  int return_value = GetDecision(&operation, &packet_list, &dtmf_event,
>-                                 &play_dtmf);
>+  int return_value =
>+      GetDecision(&operation, &packet_list, &dtmf_event, &play_dtmf);
>   if (return_value != 0) {
>     last_mode_ = kModeError;
>     return return_value;
>@@ -865,21 +860,19 @@ int NetEqImpl::GetAudioInternal(AudioFrame* audio_frame, bool* muted) {
>   AudioDecoder::SpeechType speech_type;
>   int length = 0;
>   const size_t start_num_packets = packet_list.size();
>-  int decode_return_value = Decode(&packet_list, &operation,
>-                                   &length, &speech_type);
>+  int decode_return_value =
>+      Decode(&packet_list, &operation, &length, &speech_type);
> 
>   assert(vad_.get());
>-  bool sid_frame_available =
>-      (operation == kRfc3389Cng && !packet_list.empty());
>+  bool sid_frame_available = (operation == kRfc3389Cng && !packet_list.empty());
>   vad_->Update(decoded_buffer_.get(), static_cast<size_t>(length), speech_type,
>                sid_frame_available, fs_hz_);
> 
>   // This is the criterion that we did decode some data through the speech
>   // decoder, and the operation resulted in comfort noise.
>   const bool codec_internal_sid_frame =
>-      use_dtx_delay_fix_ ? (speech_type == AudioDecoder::kComfortNoise &&
>-                            start_num_packets > packet_list.size())
>-                         : (speech_type == AudioDecoder::kComfortNoise);
>+      (speech_type == AudioDecoder::kComfortNoise &&
>+       start_num_packets > packet_list.size());
> 
>   if (sid_frame_available || codec_internal_sid_frame) {
>     // Start a new stopwatch since we are decoding a new CNG packet.
>@@ -930,33 +923,6 @@ int NetEqImpl::GetAudioInternal(AudioFrame* audio_frame, bool* muted) {
>       return_value = DoDtmf(dtmf_event, &play_dtmf);
>       break;
>     }
>-    case kAlternativePlc: {
>-      // TODO(hlundin): Write test for this.
>-      DoAlternativePlc(false);
>-      break;
>-    }
>-    case kAlternativePlcIncreaseTimestamp: {
>-      // TODO(hlundin): Write test for this.
>-      DoAlternativePlc(true);
>-      break;
>-    }
>-    case kAudioRepetitionIncreaseTimestamp: {
>-      // TODO(hlundin): Write test for this.
>-      sync_buffer_->IncreaseEndTimestamp(
>-          static_cast<uint32_t>(output_size_samples_));
>-      // Skipping break on purpose. Execution should move on into the
>-      // next case.
>-      FALLTHROUGH();
>-    }
>-    case kAudioRepetition: {
>-      // TODO(hlundin): Write test for this.
>-      // Copy last |output_size_samples_| from |sync_buffer_| to
>-      // |algorithm_buffer|.
>-      algorithm_buffer_->PushBackFromIndex(
>-          *sync_buffer_, sync_buffer_->Size() - output_size_samples_);
>-      expand_->Reset();
>-      break;
>-    }
>     case kUndefined: {
>       RTC_LOG(LS_ERROR) << "Invalid operation kUndefined.";
>       assert(false);  // This should not happen.
>@@ -1023,8 +989,7 @@ int NetEqImpl::GetAudioInternal(AudioFrame* audio_frame, bool* muted) {
>   // Update the background noise parameters if last operation wrote data
>   // straight from the decoder to the |sync_buffer_|. That is, none of the
>   // operations that modify the signal can be followed by a parameter update.
>-  if ((last_mode_ == kModeNormal) ||
>-      (last_mode_ == kModeAccelerateFail) ||
>+  if ((last_mode_ == kModeNormal) || (last_mode_ == kModeAccelerateFail) ||
>       (last_mode_ == kModePreemptiveExpandFail) ||
>       (last_mode_ == kModeRfc3389Cng) ||
>       (last_mode_ == kModeCodecInternalCng)) {
>@@ -1041,7 +1006,8 @@ int NetEqImpl::GetAudioInternal(AudioFrame* audio_frame, bool* muted) {
>     // If last operation was not expand, calculate the |playout_timestamp_| from
>     // the |sync_buffer_|. However, do not update the |playout_timestamp_| if it
>     // would be moved "backwards".
>-    uint32_t temp_timestamp = sync_buffer_->end_timestamp() -
>+    uint32_t temp_timestamp =
>+        sync_buffer_->end_timestamp() -
>         static_cast<uint32_t>(sync_buffer_->FutureLength());
>     if (static_cast<int32_t>(temp_timestamp - playout_timestamp_) > 0) {
>       playout_timestamp_ = temp_timestamp;
>@@ -1060,13 +1026,13 @@ int NetEqImpl::GetAudioInternal(AudioFrame* audio_frame, bool* muted) {
>           : timestamp_scaler_->ToExternal(playout_timestamp_) -
>                 static_cast<uint32_t>(audio_frame->samples_per_channel_);
> 
>-  if (!(last_mode_ == kModeRfc3389Cng ||
>-      last_mode_ == kModeCodecInternalCng ||
>-      last_mode_ == kModeExpand)) {
>+  if (!(last_mode_ == kModeRfc3389Cng || last_mode_ == kModeCodecInternalCng ||
>+        last_mode_ == kModeExpand)) {
>     generated_noise_stopwatch_.reset();
>   }
> 
>-  if (decode_return_value) return decode_return_value;
>+  if (decode_return_value)
>+    return decode_return_value;
>   return return_value;
> }
> 
>@@ -1090,11 +1056,10 @@ int NetEqImpl::GetDecision(Operations* operation,
>   RTC_DCHECK(!generated_noise_stopwatch_ ||
>              generated_noise_stopwatch_->ElapsedTicks() >= 1);
>   uint64_t generated_noise_samples =
>-      generated_noise_stopwatch_
>-          ? (generated_noise_stopwatch_->ElapsedTicks() - 1) *
>-                    output_size_samples_ +
>-                decision_logic_->noise_fast_forward()
>-          : 0;
>+      generated_noise_stopwatch_ ? (generated_noise_stopwatch_->ElapsedTicks() -
>+                                    1) * output_size_samples_ +
>+                                       decision_logic_->noise_fast_forward()
>+                                 : 0;
> 
>   if (decision_logic_->CngRfc3389On() || last_mode_ == kModeRfc3389Cng) {
>     // Because of timestamp peculiarities, we have to "manually" disallow using
>@@ -1117,7 +1082,7 @@ int NetEqImpl::GetDecision(Operations* operation,
> 
>   assert(expand_.get());
>   const int samples_left = static_cast<int>(sync_buffer_->FutureLength() -
>-      expand_->overlap_length());
>+                                            expand_->overlap_length());
>   if (last_mode_ == kModeAccelerateSuccess ||
>       last_mode_ == kModeAccelerateLowEnergy ||
>       last_mode_ == kModePreemptiveExpandSuccess ||
>@@ -1129,9 +1094,8 @@ int NetEqImpl::GetDecision(Operations* operation,
> 
>   // Check if it is time to play a DTMF event.
>   if (dtmf_buffer_->GetEvent(
>-      static_cast<uint32_t>(
>-          end_timestamp + generated_noise_samples),
>-      dtmf_event)) {
>+          static_cast<uint32_t>(end_timestamp + generated_noise_samples),
>+          dtmf_event)) {
>     *play_dtmf = true;
>   }
> 
>@@ -1233,12 +1197,12 @@ int NetEqImpl::GetDecision(Operations* operation,
>         decision_logic_->set_prev_time_scale(true);
>         return 0;
>       } else if (samples_left >= static_cast<int>(samples_10_ms) &&
>-          decoder_frame_length_ >= samples_30_ms) {
>+                 decoder_frame_length_ >= samples_30_ms) {
>         // Avoid decoding more data as it might overflow the playout buffer.
>         *operation = kNormal;
>         return 0;
>       } else if (samples_left < static_cast<int>(samples_20_ms) &&
>-          decoder_frame_length_ < samples_30_ms) {
>+                 decoder_frame_length_ < samples_30_ms) {
>         // Build up decoded data by decoding at least 20 ms of audio data. Do
>         // not perform accelerate yet, but wait until we only need to do one
>         // decoding.
>@@ -1257,7 +1221,7 @@ int NetEqImpl::GetDecision(Operations* operation,
>       // audio data.
>       if ((samples_left >= static_cast<int>(samples_30_ms)) ||
>           (samples_left >= static_cast<int>(samples_10_ms) &&
>-              decoder_frame_length_ >= samples_30_ms)) {
>+           decoder_frame_length_ >= samples_30_ms)) {
>         // Already have enough data, so we do not need to extract any more.
>         // Or, avoid decoding more data as it might overflow the playout buffer.
>         // Still try preemptive expand, though.
>@@ -1286,10 +1250,7 @@ int NetEqImpl::GetDecision(Operations* operation,
> 
>   // Get packets from buffer.
>   int extracted_samples = 0;
>-  if (packet && *operation != kAlternativePlc &&
>-      *operation != kAlternativePlcIncreaseTimestamp &&
>-      *operation != kAudioRepetition &&
>-      *operation != kAudioRepetitionIncreaseTimestamp) {
>+  if (packet) {
>     sync_buffer_->IncreaseEndTimestamp(packet->timestamp - end_timestamp);
>     if (decision_logic_->CngOff()) {
>       // Adjustment of timestamp only corresponds to an actual packet loss
>@@ -1329,7 +1290,8 @@ int NetEqImpl::GetDecision(Operations* operation,
>   return 0;
> }
> 
>-int NetEqImpl::Decode(PacketList* packet_list, Operations* operation,
>+int NetEqImpl::Decode(PacketList* packet_list,
>+                      Operations* operation,
>                       int* decoded_length,
>                       AudioDecoder::SpeechType* speech_type) {
>   *speech_type = AudioDecoder::kSpeech;
>@@ -1354,8 +1316,8 @@ int NetEqImpl::Decode(PacketList* packet_list, Operations* operation,
>       decoder_database_->SetActiveDecoder(payload_type, &decoder_changed);
>       if (decoder_changed) {
>         // We have a new decoder. Re-init some values.
>-        const DecoderDatabase::DecoderInfo* decoder_info = decoder_database_
>-            ->GetDecoderInfo(payload_type);
>+        const DecoderDatabase::DecoderInfo* decoder_info =
>+            decoder_database_->GetDecoderInfo(payload_type);
>         assert(decoder_info);
>         if (!decoder_info) {
>           RTC_LOG(LS_WARNING)
>@@ -1401,8 +1363,8 @@ int NetEqImpl::Decode(PacketList* packet_list, Operations* operation,
>     RTC_DCHECK(packet_list->empty());
>     return_value = DecodeCng(decoder, decoded_length, speech_type);
>   } else {
>-    return_value = DecodeLoop(packet_list, *operation, decoder,
>-                              decoded_length, speech_type);
>+    return_value = DecodeLoop(packet_list, *operation, decoder, decoded_length,
>+                              speech_type);
>   }
> 
>   if (*decoded_length < 0) {
>@@ -1436,7 +1398,8 @@ int NetEqImpl::Decode(PacketList* packet_list, Operations* operation,
>   return return_value;
> }
> 
>-int NetEqImpl::DecodeCng(AudioDecoder* decoder, int* decoded_length,
>+int NetEqImpl::DecodeCng(AudioDecoder* decoder,
>+                         int* decoded_length,
>                          AudioDecoder::SpeechType* speech_type) {
>   if (!decoder) {
>     // This happens when active decoder is not defined.
>@@ -1446,9 +1409,9 @@ int NetEqImpl::DecodeCng(AudioDecoder* decoder, int* decoded_length,
> 
>   while (*decoded_length < rtc::dchecked_cast<int>(output_size_samples_)) {
>     const int length = decoder->Decode(
>-            nullptr, 0, fs_hz_,
>-            (decoded_buffer_length_ - *decoded_length) * sizeof(int16_t),
>-            &decoded_buffer_[*decoded_length], speech_type);
>+        nullptr, 0, fs_hz_,
>+        (decoded_buffer_length_ - *decoded_length) * sizeof(int16_t),
>+        &decoded_buffer_[*decoded_length], speech_type);
>     if (length > 0) {
>       *decoded_length += length;
>     } else {
>@@ -1466,15 +1429,16 @@ int NetEqImpl::DecodeCng(AudioDecoder* decoder, int* decoded_length,
>   return 0;
> }
> 
>-int NetEqImpl::DecodeLoop(PacketList* packet_list, const Operations& operation,
>-                          AudioDecoder* decoder, int* decoded_length,
>+int NetEqImpl::DecodeLoop(PacketList* packet_list,
>+                          const Operations& operation,
>+                          AudioDecoder* decoder,
>+                          int* decoded_length,
>                           AudioDecoder::SpeechType* speech_type) {
>   RTC_DCHECK(last_decoded_timestamps_.empty());
> 
>   // Do decoding.
>-  while (
>-      !packet_list->empty() &&
>-      !decoder_database_->IsComfortNoise(packet_list->front().payload_type)) {
>+  while (!packet_list->empty() && !decoder_database_->IsComfortNoise(
>+                                      packet_list->front().payload_type)) {
>     assert(decoder);  // At this point, we must have a decoder object.
>     // The number of channels in the |sync_buffer_| should be the same as the
>     // number decoder channels.
>@@ -1516,27 +1480,26 @@ int NetEqImpl::DecodeLoop(PacketList* packet_list, const Operations& operation,
> 
>   // If the list is not empty at this point, either a decoding error terminated
>   // the while-loop, or list must hold exactly one CNG packet.
>-  assert(
>-      packet_list->empty() || *decoded_length < 0 ||
>-      (packet_list->size() == 1 &&
>-       decoder_database_->IsComfortNoise(packet_list->front().payload_type)));
>+  assert(packet_list->empty() || *decoded_length < 0 ||
>+         (packet_list->size() == 1 && decoder_database_->IsComfortNoise(
>+                                          packet_list->front().payload_type)));
>   return 0;
> }
> 
>-void NetEqImpl::DoNormal(const int16_t* decoded_buffer, size_t decoded_length,
>-                         AudioDecoder::SpeechType speech_type, bool play_dtmf) {
>+void NetEqImpl::DoNormal(const int16_t* decoded_buffer,
>+                         size_t decoded_length,
>+                         AudioDecoder::SpeechType speech_type,
>+                         bool play_dtmf) {
>   assert(normal_.get());
>-  assert(mute_factor_array_.get());
>   normal_->Process(decoded_buffer, decoded_length, last_mode_,
>-                   mute_factor_array_.get(), algorithm_buffer_.get());
>+                   algorithm_buffer_.get());
>   if (decoded_length != 0) {
>     last_mode_ = kModeNormal;
>   }
> 
>   // If last packet was decoded as an inband CNG, set mode to CNG instead.
>-  if ((speech_type == AudioDecoder::kComfortNoise)
>-      || ((last_mode_ == kModeCodecInternalCng)
>-          && (decoded_length == 0))) {
>+  if ((speech_type == AudioDecoder::kComfortNoise) ||
>+      ((last_mode_ == kModeCodecInternalCng) && (decoded_length == 0))) {
>     // TODO(hlundin): Remove second part of || statement above.
>     last_mode_ = kModeCodecInternalCng;
>   }
>@@ -1546,13 +1509,13 @@ void NetEqImpl::DoNormal(const int16_t* decoded_buffer, size_t decoded_length,
>   }
> }
> 
>-void NetEqImpl::DoMerge(int16_t* decoded_buffer, size_t decoded_length,
>-                        AudioDecoder::SpeechType speech_type, bool play_dtmf) {
>-  assert(mute_factor_array_.get());
>+void NetEqImpl::DoMerge(int16_t* decoded_buffer,
>+                        size_t decoded_length,
>+                        AudioDecoder::SpeechType speech_type,
>+                        bool play_dtmf) {
>   assert(merge_.get());
>-  size_t new_length = merge_->Process(decoded_buffer, decoded_length,
>-                                      mute_factor_array_.get(),
>-                                      algorithm_buffer_.get());
>+  size_t new_length =
>+      merge_->Process(decoded_buffer, decoded_length, algorithm_buffer_.get());
>   // Correction can be negative.
>   int expand_length_correction =
>       rtc::dchecked_cast<int>(new_length) -
>@@ -1580,7 +1543,7 @@ void NetEqImpl::DoMerge(int16_t* decoded_buffer, size_t decoded_length,
> 
> int NetEqImpl::DoExpand(bool play_dtmf) {
>   while ((sync_buffer_->FutureLength() - expand_->overlap_length()) <
>-      output_size_samples_) {
>+         output_size_samples_) {
>     algorithm_buffer_->Clear();
>     int return_value = expand_->Process(algorithm_buffer_.get());
>     size_t length = algorithm_buffer_->Size();
>@@ -1628,11 +1591,10 @@ int NetEqImpl::DoAccelerate(int16_t* decoded_buffer,
>   size_t decoded_length_per_channel = decoded_length / num_channels;
>   if (decoded_length_per_channel < required_samples) {
>     // Must move data from the |sync_buffer_| in order to get 30 ms.
>-    borrowed_samples_per_channel = static_cast<int>(required_samples -
>-        decoded_length_per_channel);
>+    borrowed_samples_per_channel =
>+        static_cast<int>(required_samples - decoded_length_per_channel);
>     memmove(&decoded_buffer[borrowed_samples_per_channel * num_channels],
>-            decoded_buffer,
>-            sizeof(int16_t) * decoded_length);
>+            decoded_buffer, sizeof(int16_t) * decoded_length);
>     sync_buffer_->ReadInterleavedFromEnd(borrowed_samples_per_channel,
>                                          decoded_buffer);
>     decoded_length = required_samples * num_channels;
>@@ -1665,17 +1627,16 @@ int NetEqImpl::DoAccelerate(int16_t* decoded_buffer,
>     if (length < borrowed_samples_per_channel) {
>       // This destroys the beginning of the buffer, but will not cause any
>       // problems.
>-      sync_buffer_->ReplaceAtIndex(*algorithm_buffer_,
>-                                   sync_buffer_->Size() -
>-                                   borrowed_samples_per_channel);
>+      sync_buffer_->ReplaceAtIndex(
>+          *algorithm_buffer_,
>+          sync_buffer_->Size() - borrowed_samples_per_channel);
>       sync_buffer_->PushFrontZeros(borrowed_samples_per_channel - length);
>       algorithm_buffer_->PopFront(length);
>       assert(algorithm_buffer_->Empty());
>     } else {
>-      sync_buffer_->ReplaceAtIndex(*algorithm_buffer_,
>-                                   borrowed_samples_per_channel,
>-                                   sync_buffer_->Size() -
>-                                   borrowed_samples_per_channel);
>+      sync_buffer_->ReplaceAtIndex(
>+          *algorithm_buffer_, borrowed_samples_per_channel,
>+          sync_buffer_->Size() - borrowed_samples_per_channel);
>       algorithm_buffer_->PopFront(borrowed_samples_per_channel);
>     }
>   }
>@@ -1707,11 +1668,11 @@ int NetEqImpl::DoPreemptiveExpand(int16_t* decoded_buffer,
>         required_samples - decoded_length_per_channel;
>     // Calculate how many of these were already played out.
>     old_borrowed_samples_per_channel =
>-        (borrowed_samples_per_channel > sync_buffer_->FutureLength()) ?
>-        (borrowed_samples_per_channel - sync_buffer_->FutureLength()) : 0;
>+        (borrowed_samples_per_channel > sync_buffer_->FutureLength())
>+            ? (borrowed_samples_per_channel - sync_buffer_->FutureLength())
>+            : 0;
>     memmove(&decoded_buffer[borrowed_samples_per_channel * num_channels],
>-            decoded_buffer,
>-            sizeof(int16_t) * decoded_length);
>+            decoded_buffer, sizeof(int16_t) * decoded_length);
>     sync_buffer_->ReadInterleavedFromEnd(borrowed_samples_per_channel,
>                                          decoded_buffer);
>     decoded_length = required_samples * num_channels;
>@@ -1719,8 +1680,7 @@ int NetEqImpl::DoPreemptiveExpand(int16_t* decoded_buffer,
> 
>   size_t samples_added;
>   PreemptiveExpand::ReturnCodes return_code = preemptive_expand_->Process(
>-      decoded_buffer, decoded_length,
>-      old_borrowed_samples_per_channel,
>+      decoded_buffer, decoded_length, old_borrowed_samples_per_channel,
>       algorithm_buffer_.get(), &samples_added);
>   stats_.PreemptiveExpandedSamples(samples_added);
>   switch (return_code) {
>@@ -1773,8 +1733,8 @@ int NetEqImpl::DoRfc3389Cng(PacketList* packet_list, bool play_dtmf) {
>       return -comfort_noise_->internal_error_code();
>     }
>   }
>-  int cn_return = comfort_noise_->Generate(output_size_samples_,
>-                                           algorithm_buffer_.get());
>+  int cn_return =
>+      comfort_noise_->Generate(output_size_samples_, algorithm_buffer_.get());
>   expand_->Reset();
>   last_mode_ = kModeRfc3389Cng;
>   if (!play_dtmf) {
>@@ -1793,9 +1753,8 @@ int NetEqImpl::DoRfc3389Cng(PacketList* packet_list, bool play_dtmf) {
> void NetEqImpl::DoCodecInternalCng(const int16_t* decoded_buffer,
>                                    size_t decoded_length) {
>   RTC_DCHECK(normal_.get());
>-  RTC_DCHECK(mute_factor_array_.get());
>   normal_->Process(decoded_buffer, decoded_length, last_mode_,
>-                   mute_factor_array_.get(), algorithm_buffer_.get());
>+                   algorithm_buffer_.get());
>   last_mode_ = kModeCodecInternalCng;
>   expand_->Reset();
> }
>@@ -1880,39 +1839,17 @@ int NetEqImpl::DoDtmf(const DtmfEvent& dtmf_event, bool* play_dtmf) {
>   return 0;
> }
> 
>-void NetEqImpl::DoAlternativePlc(bool increase_timestamp) {
>-  AudioDecoder* decoder = decoder_database_->GetActiveDecoder();
>-  size_t length;
>-  if (decoder && decoder->HasDecodePlc()) {
>-    // Use the decoder's packet-loss concealment.
>-    // TODO(hlundin): Will probably need a longer buffer for multi-channel.
>-    int16_t decoded_buffer[kMaxFrameSize];
>-    length = decoder->DecodePlc(1, decoded_buffer);
>-    if (length > 0)
>-      algorithm_buffer_->PushBackInterleaved(decoded_buffer, length);
>-  } else {
>-    // Do simple zero-stuffing.
>-    length = output_size_samples_;
>-    algorithm_buffer_->Zeros(length);
>-    // By not advancing the timestamp, NetEq inserts samples.
>-    stats_.AddZeros(length);
>-  }
>-  if (increase_timestamp) {
>-    sync_buffer_->IncreaseEndTimestamp(static_cast<uint32_t>(length));
>-  }
>-  expand_->Reset();
>-}
>-
>-int NetEqImpl::DtmfOverdub(const DtmfEvent& dtmf_event, size_t num_channels,
>+int NetEqImpl::DtmfOverdub(const DtmfEvent& dtmf_event,
>+                           size_t num_channels,
>                            int16_t* output) const {
>   size_t out_index = 0;
>   size_t overdub_length = output_size_samples_;  // Default value.
> 
>   if (sync_buffer_->dtmf_index() > sync_buffer_->next_index()) {
>     // Special operation for transition from "DTMF only" to "DTMF overdub".
>-    out_index = std::min(
>-        sync_buffer_->dtmf_index() - sync_buffer_->next_index(),
>-        output_size_samples_);
>+    out_index =
>+        std::min(sync_buffer_->dtmf_index() - sync_buffer_->next_index(),
>+                 output_size_samples_);
>     overdub_length = output_size_samples_ - out_index;
>   }
> 
>@@ -1923,8 +1860,8 @@ int NetEqImpl::DtmfOverdub(const DtmfEvent& dtmf_event, size_t num_channels,
>                                                    dtmf_event.volume);
>   }
>   if (dtmf_return_value == 0) {
>-    dtmf_return_value = dtmf_tone_generator_->Generate(overdub_length,
>-                                                       &dtmf_output);
>+    dtmf_return_value =
>+        dtmf_tone_generator_->Generate(overdub_length, &dtmf_output);
>     assert(overdub_length == dtmf_output.Size());
>   }
>   dtmf_output.ReadInterleaved(overdub_length, &output[out_index]);
>@@ -1951,7 +1888,7 @@ int NetEqImpl::ExtractPackets(size_t required_samples,
>   // Packet extraction loop.
>   do {
>     timestamp_ = next_packet->timestamp;
>-    rtc::Optional<Packet> packet = packet_buffer_->GetNextPacket();
>+    absl::optional<Packet> packet = packet_buffer_->GetNextPacket();
>     // |next_packet| may be invalid after the |packet_buffer_| operation.
>     next_packet = nullptr;
>     if (!packet) {
>@@ -2003,7 +1940,7 @@ int NetEqImpl::ExtractPackets(size_t required_samples,
>     stats_.JitterBufferDelay(extracted_samples, waiting_time_ms);
> 
>     packet_list->push_back(std::move(*packet));  // Store packet in list.
>-    packet = rtc::nullopt;  // Ensure it's never used after the move.
>+    packet = absl::nullopt;  // Ensure it's never used after the move.
> 
>     // Check what packet is available next.
>     next_packet = packet_buffer_->PeekNextPacket();
>@@ -2045,7 +1982,7 @@ void NetEqImpl::SetSampleRateAndChannels(int fs_hz, size_t channels) {
>   RTC_LOG(LS_VERBOSE) << "SetSampleRateAndChannels " << fs_hz << " "
>                       << channels;
>   // TODO(hlundin): Change to an enumerator and skip assert.
>-  assert(fs_hz == 8000 || fs_hz == 16000 || fs_hz ==  32000 || fs_hz == 48000);
>+  assert(fs_hz == 8000 || fs_hz == 16000 || fs_hz == 32000 || fs_hz == 48000);
>   assert(channels > 0);
> 
>   fs_hz_ = fs_hz;
>@@ -2055,12 +1992,6 @@ void NetEqImpl::SetSampleRateAndChannels(int fs_hz, size_t channels) {
> 
>   last_mode_ = kModeNormal;
> 
>-  // Create a new array of mute factors and set all to 1.
>-  mute_factor_array_.reset(new int16_t[channels]);
>-  for (size_t i = 0; i < channels; ++i) {
>-    mute_factor_array_[i] = 16384;  // 1.0 in Q14.
>-  }
>-
>   ComfortNoiseDecoder* cng_decoder = decoder_database_->GetActiveCngDecoder();
>   if (cng_decoder)
>     cng_decoder->Reset();
>@@ -2077,7 +2008,6 @@ void NetEqImpl::SetSampleRateAndChannels(int fs_hz, size_t channels) {
> 
>   // Delete BackgroundNoise object and create a new one.
>   background_noise_.reset(new BackgroundNoise(channels));
>-  background_noise_->set_mode(background_noise_mode_);
> 
>   // Reset random vector.
>   random_vector_.Reset();
>@@ -2086,7 +2016,7 @@ void NetEqImpl::SetSampleRateAndChannels(int fs_hz, size_t channels) {
> 
>   // Move index so that we create a small set of future samples (all 0).
>   sync_buffer_->set_next_index(sync_buffer_->next_index() -
>-      expand_->overlap_length());
>+                               expand_->overlap_length());
> 
>   normal_.reset(new Normal(fs_hz, decoder_database_.get(), *background_noise_,
>                            expand_.get()));
>@@ -2096,8 +2026,8 @@ void NetEqImpl::SetSampleRateAndChannels(int fs_hz, size_t channels) {
>       fs_hz, channels, *background_noise_, expand_->overlap_length()));
> 
>   // Delete ComfortNoise object and create a new one.
>-  comfort_noise_.reset(new ComfortNoise(fs_hz, decoder_database_.get(),
>-                                        sync_buffer_.get()));
>+  comfort_noise_.reset(
>+      new ComfortNoise(fs_hz, decoder_database_.get(), sync_buffer_.get()));
> 
>   // Verify that |decoded_buffer_| is long enough.
>   if (decoded_buffer_length_ < kMaxFrameSize * channels) {
>@@ -2133,8 +2063,8 @@ NetEqImpl::OutputType NetEqImpl::LastOutputType() {
> 
> void NetEqImpl::CreateDecisionLogic() {
>   decision_logic_.reset(DecisionLogic::Create(
>-      fs_hz_, output_size_samples_, playout_mode_, decoder_database_.get(),
>-      *packet_buffer_.get(), delay_manager_.get(), buffer_level_filter_.get(),
>-      tick_timer_.get()));
>+      fs_hz_, output_size_samples_, no_time_stretching_,
>+      decoder_database_.get(), *packet_buffer_.get(), delay_manager_.get(),
>+      buffer_level_filter_.get(), tick_timer_.get()));
> }
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq_impl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq_impl.h
>index a922bcdcd64..e0e0c138822 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq_impl.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq_impl.h
>@@ -14,20 +14,20 @@
> #include <memory>
> #include <string>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
>+#include "api/audio/audio_frame.h"
> #include "modules/audio_coding/neteq/audio_multi_vector.h"
> #include "modules/audio_coding/neteq/defines.h"
>+#include "modules/audio_coding/neteq/expand_uma_logger.h"
> #include "modules/audio_coding/neteq/include/neteq.h"
> #include "modules/audio_coding/neteq/packet.h"  // Declare PacketList.
> #include "modules/audio_coding/neteq/random_vector.h"
> #include "modules/audio_coding/neteq/rtcp.h"
> #include "modules/audio_coding/neteq/statistics_calculator.h"
> #include "modules/audio_coding/neteq/tick_timer.h"
>-#include "modules/include/module_common_types.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/thread_annotations.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -167,16 +167,6 @@ class NetEqImpl : public webrtc::NetEq {
> 
>   int FilteredCurrentDelayMs() const override;
> 
>-  // Sets the playout mode to |mode|.
>-  // Deprecated.
>-  // TODO(henrik.lundin) Delete.
>-  void SetPlayoutMode(NetEqPlayoutMode mode) override;
>-
>-  // Returns the current playout mode.
>-  // Deprecated.
>-  // TODO(henrik.lundin) Delete.
>-  NetEqPlayoutMode PlayoutMode() const override;
>-
>   // Writes the current network statistics to |stats|. The statistics are reset
>   // after the call.
>   int NetworkStatistics(NetEqNetworkStatistics* stats) override;
>@@ -197,13 +187,13 @@ class NetEqImpl : public webrtc::NetEq {
>   // Disables post-decode VAD.
>   void DisableVad() override;
> 
>-  rtc::Optional<uint32_t> GetPlayoutTimestamp() const override;
>+  absl::optional<uint32_t> GetPlayoutTimestamp() const override;
> 
>   int last_output_sample_rate_hz() const override;
> 
>-  rtc::Optional<CodecInst> GetDecoder(int payload_type) const override;
>+  absl::optional<CodecInst> GetDecoder(int payload_type) const override;
> 
>-  rtc::Optional<SdpAudioFormat> GetDecoderFormat(
>+  absl::optional<SdpAudioFormat> GetDecoderFormat(
>       int payload_type) const override;
> 
>   int SetTargetNumberOfChannels() override;
>@@ -335,12 +325,6 @@ class NetEqImpl : public webrtc::NetEq {
>   int DoDtmf(const DtmfEvent& dtmf_event, bool* play_dtmf)
>       RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
> 
>-  // Produces packet-loss concealment using alternative methods. If the codec
>-  // has an internal PLC, it is called to generate samples. Otherwise, the
>-  // method performs zero-stuffing.
>-  void DoAlternativePlc(bool increase_timestamp)
>-      RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
>-
>   // Overdub DTMF on top of |output|.
>   int DtmfOverdub(const DtmfEvent& dtmf_event,
>                   size_t num_channels,
>@@ -417,20 +401,17 @@ class NetEqImpl : public webrtc::NetEq {
>   size_t decoder_frame_length_ RTC_GUARDED_BY(crit_sect_);
>   Modes last_mode_ RTC_GUARDED_BY(crit_sect_);
>   Operations last_operation_ RTC_GUARDED_BY(crit_sect_);
>-  std::unique_ptr<int16_t[]> mute_factor_array_ RTC_GUARDED_BY(crit_sect_);
>   size_t decoded_buffer_length_ RTC_GUARDED_BY(crit_sect_);
>   std::unique_ptr<int16_t[]> decoded_buffer_ RTC_GUARDED_BY(crit_sect_);
>   uint32_t playout_timestamp_ RTC_GUARDED_BY(crit_sect_);
>   bool new_codec_ RTC_GUARDED_BY(crit_sect_);
>   uint32_t timestamp_ RTC_GUARDED_BY(crit_sect_);
>   bool reset_decoder_ RTC_GUARDED_BY(crit_sect_);
>-  rtc::Optional<uint8_t> current_rtp_payload_type_ RTC_GUARDED_BY(crit_sect_);
>-  rtc::Optional<uint8_t> current_cng_rtp_payload_type_
>+  absl::optional<uint8_t> current_rtp_payload_type_ RTC_GUARDED_BY(crit_sect_);
>+  absl::optional<uint8_t> current_cng_rtp_payload_type_
>       RTC_GUARDED_BY(crit_sect_);
>   uint32_t ssrc_ RTC_GUARDED_BY(crit_sect_);
>   bool first_packet_ RTC_GUARDED_BY(crit_sect_);
>-  const BackgroundNoiseMode background_noise_mode_ RTC_GUARDED_BY(crit_sect_);
>-  NetEqPlayoutMode playout_mode_ RTC_GUARDED_BY(crit_sect_);
>   bool enable_fast_accelerate_ RTC_GUARDED_BY(crit_sect_);
>   std::unique_ptr<NackTracker> nack_ RTC_GUARDED_BY(crit_sect_);
>   bool nack_enabled_ RTC_GUARDED_BY(crit_sect_);
>@@ -440,7 +421,9 @@ class NetEqImpl : public webrtc::NetEq {
>   std::unique_ptr<TickTimer::Stopwatch> generated_noise_stopwatch_
>       RTC_GUARDED_BY(crit_sect_);
>   std::vector<uint32_t> last_decoded_timestamps_ RTC_GUARDED_BY(crit_sect_);
>-  const bool use_dtx_delay_fix_ RTC_GUARDED_BY(crit_sect_);
>+  ExpandUmaLogger expand_uma_logger_ RTC_GUARDED_BY(crit_sect_);
>+  ExpandUmaLogger speech_expand_uma_logger_ RTC_GUARDED_BY(crit_sect_);
>+  bool no_time_stretching_ RTC_GUARDED_BY(crit_sect_);  // Only used for test.
> 
>  private:
>   RTC_DISALLOW_COPY_AND_ASSIGN(NetEqImpl);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
>index 3088b37c364..b772dfa71de 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
>@@ -27,7 +27,6 @@
> #include "modules/audio_coding/neteq/preemptive_expand.h"
> #include "modules/audio_coding/neteq/sync_buffer.h"
> #include "modules/audio_coding/neteq/timestamp_scaler.h"
>-#include "modules/include/module_common_types.h"
> #include "rtc_base/numerics/safe_conversions.h"
> #include "test/gmock.h"
> #include "test/gtest.h"
>@@ -313,8 +312,9 @@ TEST_F(NetEqImplTest, InsertPacket) {
> 
>   rtc::scoped_refptr<MockAudioDecoderFactory> mock_decoder_factory(
>       new rtc::RefCountedObject<MockAudioDecoderFactory>);
>-  EXPECT_CALL(*mock_decoder_factory, MakeAudioDecoderMock(_, _))
>+  EXPECT_CALL(*mock_decoder_factory, MakeAudioDecoderMock(_, _, _))
>       .WillOnce(Invoke([&](const SdpAudioFormat& format,
>+                           absl::optional<AudioCodecPairId> codec_pair_id,
>                            std::unique_ptr<AudioDecoder>* dec) {
>         EXPECT_EQ("pcmu", format.name);
> 
>@@ -334,7 +334,7 @@ TEST_F(NetEqImplTest, InsertPacket) {
> 
>         *dec = std::move(mock_decoder);
>       }));
>-  DecoderDatabase::DecoderInfo info(NetEqDecoder::kDecoderPCMu,
>+  DecoderDatabase::DecoderInfo info(NetEqDecoder::kDecoderPCMu, absl::nullopt,
>                                     mock_decoder_factory);
> 
>   // Expectations for decoder database.
>@@ -518,10 +518,10 @@ TEST_F(NetEqImplTest, VerifyTimestampPropagation) {
>   // The value of the last of the output samples is the same as the number of
>   // samples played from the decoded packet. Thus, this number + the RTP
>   // timestamp should match the playout timestamp.
>-  // Wrap the expected value in an rtc::Optional to compare them as such.
>+  // Wrap the expected value in an absl::optional to compare them as such.
>   EXPECT_EQ(
>-      rtc::Optional<uint32_t>(rtp_header.timestamp +
>-                              output.data()[output.samples_per_channel_ - 1]),
>+      absl::optional<uint32_t>(rtp_header.timestamp +
>+                               output.data()[output.samples_per_channel_ - 1]),
>       neteq_->GetPlayoutTimestamp());
> 
>   // Check the timestamp for the last value in the sync buffer. This should
>@@ -783,12 +783,12 @@ TEST_F(NetEqImplTest, CodecInternalCng) {
> 
>   bool muted;
>   EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output, &muted));
>-  rtc::Optional<uint32_t> last_timestamp = neteq_->GetPlayoutTimestamp();
>+  absl::optional<uint32_t> last_timestamp = neteq_->GetPlayoutTimestamp();
>   ASSERT_TRUE(last_timestamp);
> 
>   // Lambda for verifying the timestamps.
>   auto verify_timestamp = [&last_timestamp, &expected_timestamp_increment](
>-      rtc::Optional<uint32_t> ts, size_t i) {
>+                              absl::optional<uint32_t> ts, size_t i) {
>     if (expected_timestamp_increment[i] == -1) {
>       // Expect to get an empty timestamp value during CNG and PLC.
>       EXPECT_FALSE(ts) << "i = " << i;
>@@ -1377,32 +1377,6 @@ class NetEqImplTest120ms : public NetEqImplTest {
>   uint16_t sequence_number_ = 1;
> };
> 
>-TEST_F(NetEqImplTest120ms, AudioRepetition) {
>-  config_.playout_mode = kPlayoutFax;
>-  CreateInstanceNoMocks();
>-  Register120msCodec(AudioDecoder::kSpeech);
>-
>-  InsertPacket(first_timestamp());
>-  GetFirstPacket();
>-
>-  bool muted;
>-  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output_, &muted));
>-  EXPECT_EQ(kAudioRepetition, neteq_->last_operation_for_test());
>-}
>-
>-TEST_F(NetEqImplTest120ms, AlternativePlc) {
>-  config_.playout_mode = kPlayoutOff;
>-  CreateInstanceNoMocks();
>-  Register120msCodec(AudioDecoder::kSpeech);
>-
>-  InsertPacket(first_timestamp());
>-  GetFirstPacket();
>-
>-  bool muted;
>-  EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output_, &muted));
>-  EXPECT_EQ(kAlternativePlc, neteq_->last_operation_for_test());
>-}
>-
> TEST_F(NetEqImplTest120ms, CodecInternalCng) {
>   CreateInstanceNoMocks();
>   Register120msCodec(AudioDecoder::kComfortNoise);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
>index 334715f8d16..57fc682f73c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq_network_stats_unittest.cc
>@@ -10,10 +10,10 @@
> 
> #include <memory>
> 
>+#include "api/audio/audio_frame.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/audio_coding/neteq/tools/neteq_external_decoder_test.h"
> #include "modules/audio_coding/neteq/tools/rtp_generator.h"
>-#include "modules/include/module_common_types.h"
> #include "test/gmock.h"
> 
> namespace webrtc {
>@@ -46,7 +46,7 @@ class MockAudioDecoder final : public AudioDecoder {
> 
>     size_t Duration() const override { return kPacketDuration; }
> 
>-    rtc::Optional<DecodeResult> Decode(
>+    absl::optional<DecodeResult> Decode(
>         rtc::ArrayView<int16_t> decoded) const override {
>       const size_t output_size =
>           sizeof(int16_t) * kPacketDuration * num_channels_;
>@@ -57,7 +57,7 @@ class MockAudioDecoder final : public AudioDecoder {
>       } else {
>         ADD_FAILURE() << "Expected decoded.size() to be >= output_size ("
>                       << decoded.size() << " vs. " << output_size << ")";
>-        return rtc::nullopt;
>+        return absl::nullopt;
>       }
>     }
> 
>@@ -86,8 +86,8 @@ class MockAudioDecoder final : public AudioDecoder {
>     return kPacketDuration;
>   }
> 
>-  bool PacketHasFec(
>-      const uint8_t* encoded, size_t encoded_len) const /* override */ {
>+  bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const
>+  /* override */ {
>     ADD_FAILURE() << "Since going through ParsePayload, PacketHasFec should "
>                      "never get called.";
>     return fec_enabled_;
>@@ -123,40 +123,40 @@ class NetEqNetworkStatsTest : public NetEqExternalDecoderTest {
>   static const int kPayloadSizeByte = 30;
>   static const int kFrameSizeMs = 20;
> 
>-enum logic {
>-  kIgnore,
>-  kEqual,
>-  kSmallerThan,
>-  kLargerThan,
>-};
>+  enum logic {
>+    kIgnore,
>+    kEqual,
>+    kSmallerThan,
>+    kLargerThan,
>+  };
> 
>-struct NetEqNetworkStatsCheck {
>-  logic current_buffer_size_ms;
>-  logic preferred_buffer_size_ms;
>-  logic jitter_peaks_found;
>-  logic packet_loss_rate;
>-  logic expand_rate;
>-  logic speech_expand_rate;
>-  logic preemptive_rate;
>-  logic accelerate_rate;
>-  logic secondary_decoded_rate;
>-  logic secondary_discarded_rate;
>-  logic clockdrift_ppm;
>-  logic added_zero_samples;
>-  NetEqNetworkStatistics stats_ref;
>-};
>+  struct NetEqNetworkStatsCheck {
>+    logic current_buffer_size_ms;
>+    logic preferred_buffer_size_ms;
>+    logic jitter_peaks_found;
>+    logic packet_loss_rate;
>+    logic expand_rate;
>+    logic speech_expand_rate;
>+    logic preemptive_rate;
>+    logic accelerate_rate;
>+    logic secondary_decoded_rate;
>+    logic secondary_discarded_rate;
>+    logic clockdrift_ppm;
>+    logic added_zero_samples;
>+    NetEqNetworkStatistics stats_ref;
>+  };
> 
>-NetEqNetworkStatsTest(NetEqDecoder codec,
>-                      int sample_rate_hz,
>-                      MockAudioDecoder* decoder)
>-    : NetEqExternalDecoderTest(codec, sample_rate_hz, decoder),
>-      external_decoder_(decoder),
>-      samples_per_ms_(sample_rate_hz / 1000),
>-      frame_size_samples_(kFrameSizeMs * samples_per_ms_),
>-      rtp_generator_(new test::RtpGenerator(samples_per_ms_)),
>-      last_lost_time_(0),
>-      packet_loss_interval_(0xffffffff) {
>-  Init();
>+  NetEqNetworkStatsTest(NetEqDecoder codec,
>+                        int sample_rate_hz,
>+                        MockAudioDecoder* decoder)
>+      : NetEqExternalDecoderTest(codec, sample_rate_hz, decoder),
>+        external_decoder_(decoder),
>+        samples_per_ms_(sample_rate_hz / 1000),
>+        frame_size_samples_(kFrameSizeMs * samples_per_ms_),
>+        rtp_generator_(new test::RtpGenerator(samples_per_ms_)),
>+        last_lost_time_(0),
>+        packet_loss_interval_(0xffffffff) {
>+    Init();
>   }
> 
>   bool Lost(uint32_t send_time) {
>@@ -168,8 +168,9 @@ NetEqNetworkStatsTest(NetEqDecoder codec,
>   }
> 
>   void SetPacketLossRate(double loss_rate) {
>-      packet_loss_interval_ = (loss_rate >= 1e-3 ?
>-          static_cast<double>(kFrameSizeMs) / loss_rate : 0xffffffff);
>+    packet_loss_interval_ =
>+        (loss_rate >= 1e-3 ? static_cast<double>(kFrameSizeMs) / loss_rate
>+                           : 0xffffffff);
>   }
> 
>   // |stats_ref|
>@@ -181,19 +182,19 @@ NetEqNetworkStatsTest(NetEqDecoder codec,
>     NetEqNetworkStatistics stats;
>     neteq()->NetworkStatistics(&stats);
> 
>-#define CHECK_NETEQ_NETWORK_STATS(x)\
>-  switch (expects.x) {\
>-    case kEqual:\
>-      EXPECT_EQ(stats.x, expects.stats_ref.x);\
>-      break;\
>-    case kSmallerThan:\
>-      EXPECT_LT(stats.x, expects.stats_ref.x);\
>-      break;\
>-    case kLargerThan:\
>-      EXPECT_GT(stats.x, expects.stats_ref.x);\
>-      break;\
>-    default:\
>-      break;\
>+#define CHECK_NETEQ_NETWORK_STATS(x)           \
>+  switch (expects.x) {                         \
>+    case kEqual:                               \
>+      EXPECT_EQ(stats.x, expects.stats_ref.x); \
>+      break;                                   \
>+    case kSmallerThan:                         \
>+      EXPECT_LT(stats.x, expects.stats_ref.x); \
>+      break;                                   \
>+    case kLargerThan:                          \
>+      EXPECT_GT(stats.x, expects.stats_ref.x); \
>+      break;                                   \
>+    default:                                   \
>+      break;                                   \
>   }
> 
>     CHECK_NETEQ_NETWORK_STATS(current_buffer_size_ms);
>@@ -220,15 +221,13 @@ NetEqNetworkStatsTest(NetEqDecoder codec,
>     uint32_t next_send_time;
> 
>     // Initiate |last_lost_time_|.
>-    time_now = next_send_time = last_lost_time_ =
>-        rtp_generator_->GetRtpHeader(kPayloadType, frame_size_samples_,
>-                                     &rtp_header_);
>+    time_now = next_send_time = last_lost_time_ = rtp_generator_->GetRtpHeader(
>+        kPayloadType, frame_size_samples_, &rtp_header_);
>     for (int k = 0; k < num_loops; ++k) {
>       // Delay by one frame such that the FEC can come in.
>       while (time_now + kFrameSizeMs >= next_send_time) {
>-        next_send_time = rtp_generator_->GetRtpHeader(kPayloadType,
>-                                                      frame_size_samples_,
>-                                                      &rtp_header_);
>+        next_send_time = rtp_generator_->GetRtpHeader(
>+            kPayloadType, frame_size_samples_, &rtp_header_);
>         if (!Lost(next_send_time)) {
>           static const uint8_t payload[kPayloadSizeByte] = {0};
>           InsertPacket(rtp_header_, payload, next_send_time);
>@@ -243,21 +242,19 @@ NetEqNetworkStatsTest(NetEqDecoder codec,
> 
>   void DecodeFecTest() {
>     external_decoder_->set_fec_enabled(false);
>-    NetEqNetworkStatsCheck expects = {
>-      kIgnore,  // current_buffer_size_ms
>-      kIgnore,  // preferred_buffer_size_ms
>-      kIgnore,  // jitter_peaks_found
>-      kEqual,  // packet_loss_rate
>-      kEqual,  // expand_rate
>-      kEqual,  // voice_expand_rate
>-      kIgnore,  // preemptive_rate
>-      kEqual,  // accelerate_rate
>-      kEqual,  // decoded_fec_rate
>-      kEqual,  // discarded_fec_rate
>-      kIgnore,  // clockdrift_ppm
>-      kEqual,  // added_zero_samples
>-      {0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0}
>-    };
>+    NetEqNetworkStatsCheck expects = {kIgnore,  // current_buffer_size_ms
>+                                      kIgnore,  // preferred_buffer_size_ms
>+                                      kIgnore,  // jitter_peaks_found
>+                                      kEqual,   // packet_loss_rate
>+                                      kEqual,   // expand_rate
>+                                      kEqual,   // voice_expand_rate
>+                                      kIgnore,  // preemptive_rate
>+                                      kEqual,   // accelerate_rate
>+                                      kEqual,   // decoded_fec_rate
>+                                      kEqual,   // discarded_fec_rate
>+                                      kIgnore,  // clockdrift_ppm
>+                                      kEqual,   // added_zero_samples
>+                                      {0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0}};
>     RunTest(50, expects);
> 
>     // Next we introduce packet losses.
>@@ -277,21 +274,19 @@ NetEqNetworkStatsTest(NetEqDecoder codec,
>   }
> 
>   void NoiseExpansionTest() {
>-    NetEqNetworkStatsCheck expects = {
>-      kIgnore,  // current_buffer_size_ms
>-      kIgnore,  // preferred_buffer_size_ms
>-      kIgnore,  // jitter_peaks_found
>-      kEqual,  // packet_loss_rate
>-      kEqual,  // expand_rate
>-      kEqual,  // speech_expand_rate
>-      kIgnore,  // preemptive_rate
>-      kEqual,  // accelerate_rate
>-      kEqual,  // decoded_fec_rate
>-      kEqual,  // discard_fec_rate
>-      kIgnore,  // clockdrift_ppm
>-      kEqual,  // added_zero_samples
>-      {0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0}
>-    };
>+    NetEqNetworkStatsCheck expects = {kIgnore,  // current_buffer_size_ms
>+                                      kIgnore,  // preferred_buffer_size_ms
>+                                      kIgnore,  // jitter_peaks_found
>+                                      kEqual,   // packet_loss_rate
>+                                      kEqual,   // expand_rate
>+                                      kEqual,   // speech_expand_rate
>+                                      kIgnore,  // preemptive_rate
>+                                      kEqual,   // accelerate_rate
>+                                      kEqual,   // decoded_fec_rate
>+                                      kEqual,   // discard_fec_rate
>+                                      kIgnore,  // clockdrift_ppm
>+                                      kEqual,   // added_zero_samples
>+                                      {0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0}};
>     RunTest(50, expects);
> 
>     SetPacketLossRate(1);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc
>index 1bef9c83a2d..ef4c235e6e3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc
>@@ -11,17 +11,17 @@
> // Test to verify correct stereo and multi-channel operation.
> 
> #include <algorithm>
>+#include <list>
> #include <memory>
> #include <string>
>-#include <list>
> 
>+#include "api/audio/audio_frame.h"
> #include "api/audio_codecs/builtin_audio_decoder_factory.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
> #include "modules/audio_coding/neteq/include/neteq.h"
> #include "modules/audio_coding/neteq/tools/input_audio_file.h"
> #include "modules/audio_coding/neteq/tools/rtp_generator.h"
>-#include "modules/include/module_common_types.h"
> #include "test/gtest.h"
> #include "test/testsupport/fileutils.h"
> 
>@@ -72,17 +72,17 @@ class NetEqStereoTest : public ::testing::TestWithParam<TestParameters> {
>     input_ = new int16_t[frame_size_samples_];
>     encoded_ = new uint8_t[2 * frame_size_samples_];
>     input_multi_channel_ = new int16_t[frame_size_samples_ * num_channels_];
>-    encoded_multi_channel_ = new uint8_t[frame_size_samples_ * 2 *
>-                                         num_channels_];
>+    encoded_multi_channel_ =
>+        new uint8_t[frame_size_samples_ * 2 * num_channels_];
>   }
> 
>   ~NetEqStereoTest() {
>     delete neteq_mono_;
>     delete neteq_;
>-    delete [] input_;
>-    delete [] encoded_;
>-    delete [] input_multi_channel_;
>-    delete [] encoded_multi_channel_;
>+    delete[] input_;
>+    delete[] encoded_;
>+    delete[] input_multi_channel_;
>+    delete[] encoded_multi_channel_;
>   }
> 
>   virtual void SetUp() {
>@@ -142,17 +142,15 @@ class NetEqStereoTest : public ::testing::TestWithParam<TestParameters> {
>     if (!input_file_->Read(frame_size_samples_, input_)) {
>       return -1;
>     }
>-    payload_size_bytes_ = WebRtcPcm16b_Encode(input_, frame_size_samples_,
>-                                             encoded_);
>+    payload_size_bytes_ =
>+        WebRtcPcm16b_Encode(input_, frame_size_samples_, encoded_);
>     if (frame_size_samples_ * 2 != payload_size_bytes_) {
>       return -1;
>     }
>-    int next_send_time = rtp_generator_mono_.GetRtpHeader(kPayloadTypeMono,
>-                                                          frame_size_samples_,
>-                                                          &rtp_header_mono_);
>-    test::InputAudioFile::DuplicateInterleaved(input_, frame_size_samples_,
>-                                               num_channels_,
>-                                               input_multi_channel_);
>+    int next_send_time = rtp_generator_mono_.GetRtpHeader(
>+        kPayloadTypeMono, frame_size_samples_, &rtp_header_mono_);
>+    test::InputAudioFile::DuplicateInterleaved(
>+        input_, frame_size_samples_, num_channels_, input_multi_channel_);
>     multi_payload_size_bytes_ = WebRtcPcm16b_Encode(
>         input_multi_channel_, frame_size_samples_ * num_channels_,
>         encoded_multi_channel_);
>@@ -267,8 +265,7 @@ class NetEqStereoTest : public ::testing::TestWithParam<TestParameters> {
> 
> class NetEqStereoTestNoJitter : public NetEqStereoTest {
>  protected:
>-  NetEqStereoTestNoJitter()
>-      : NetEqStereoTest() {
>+  NetEqStereoTestNoJitter() : NetEqStereoTest() {
>     // Start the sender 100 ms before the receiver to pre-fill the buffer.
>     // This is to avoid doing preemptive expand early in the test.
>     // TODO(hlundin): Mock the decision making instead to control the modes.
>@@ -282,17 +279,15 @@ TEST_P(NetEqStereoTestNoJitter, RunTest) {
> 
> class NetEqStereoTestPositiveDrift : public NetEqStereoTest {
>  protected:
>-  NetEqStereoTestPositiveDrift()
>-      : NetEqStereoTest(),
>-        drift_factor(0.9) {
>+  NetEqStereoTestPositiveDrift() : NetEqStereoTest(), drift_factor(0.9) {
>     // Start the sender 100 ms before the receiver to pre-fill the buffer.
>     // This is to avoid doing preemptive expand early in the test.
>     // TODO(hlundin): Mock the decision making instead to control the modes.
>     last_arrival_time_ = -100;
>   }
>   virtual int GetArrivalTime(int send_time) {
>-    int arrival_time = last_arrival_time_ +
>-        drift_factor * (send_time - last_send_time_);
>+    int arrival_time =
>+        last_arrival_time_ + drift_factor * (send_time - last_send_time_);
>     last_send_time_ = send_time;
>     last_arrival_time_ = arrival_time;
>     return arrival_time;
>@@ -307,8 +302,7 @@ TEST_P(NetEqStereoTestPositiveDrift, RunTest) {
> 
> class NetEqStereoTestNegativeDrift : public NetEqStereoTestPositiveDrift {
>  protected:
>-  NetEqStereoTestNegativeDrift()
>-      : NetEqStereoTestPositiveDrift() {
>+  NetEqStereoTestNegativeDrift() : NetEqStereoTestPositiveDrift() {
>     drift_factor = 1.1;
>     last_arrival_time_ = 0;
>   }
>@@ -322,10 +316,7 @@ class NetEqStereoTestDelays : public NetEqStereoTest {
>  protected:
>   static const int kDelayInterval = 10;
>   static const int kDelay = 1000;
>-  NetEqStereoTestDelays()
>-      : NetEqStereoTest(),
>-        frame_index_(0) {
>-  }
>+  NetEqStereoTestDelays() : NetEqStereoTest(), frame_index_(0) {}
> 
>   virtual int GetArrivalTime(int send_time) {
>     // Deliver immediately, unless we have a back-log.
>@@ -349,22 +340,16 @@ TEST_P(NetEqStereoTestDelays, RunTest) {
> class NetEqStereoTestLosses : public NetEqStereoTest {
>  protected:
>   static const int kLossInterval = 10;
>-  NetEqStereoTestLosses()
>-      : NetEqStereoTest(),
>-        frame_index_(0) {
>-  }
>+  NetEqStereoTestLosses() : NetEqStereoTest(), frame_index_(0) {}
> 
>-  virtual bool Lost() {
>-    return (++frame_index_) % kLossInterval == 0;
>-  }
>+  virtual bool Lost() { return (++frame_index_) % kLossInterval == 0; }
> 
>   // TODO(hlundin): NetEq is not giving bitexact results for these cases.
>   virtual void VerifyOutput(size_t num_samples) {
>     for (size_t i = 0; i < num_samples; ++i) {
>       const int16_t* output_data = output_.data();
>       const int16_t* output_multi_channel_data = output_multi_channel_.data();
>-      auto first_channel_sample =
>-          output_multi_channel_data[i * num_channels_];
>+      auto first_channel_sample = output_multi_channel_data[i * num_channels_];
>       for (size_t j = 0; j < num_channels_; ++j) {
>         const int kErrorMargin = 200;
>         EXPECT_NEAR(output_data[i],
>@@ -384,7 +369,6 @@ TEST_P(NetEqStereoTestLosses, RunTest) {
>   RunTest(100);
> }
> 
>-
> // Creates a list of parameter sets.
> std::list<TestParameters> GetTestParameters() {
>   std::list<TestParameters> l;
>@@ -412,9 +396,9 @@ std::list<TestParameters> GetTestParameters() {
> 
> // Pretty-printing the test parameters in case of an error.
> void PrintTo(const TestParameters& p, ::std::ostream* os) {
>-  *os << "{frame_size = " << p.frame_size <<
>-      ", num_channels = " << p.num_channels <<
>-      ", sample_rate = " << p.sample_rate << "}";
>+  *os << "{frame_size = " << p.frame_size
>+      << ", num_channels = " << p.num_channels
>+      << ", sample_rate = " << p.sample_rate << "}";
> }
> 
> // Instantiate the tests. Each test is instantiated using the function above,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
>index ef31f4b56e3..96a392bff28 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
>@@ -20,21 +20,23 @@
> #include <string>
> #include <vector>
> 
>+#include "api/audio/audio_frame.h"
> #include "api/audio_codecs/builtin_audio_decoder_factory.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
> #include "modules/audio_coding/neteq/tools/audio_loop.h"
>+#include "modules/audio_coding/neteq/tools/neteq_packet_source_input.h"
>+#include "modules/audio_coding/neteq/tools/neteq_test.h"
> #include "modules/audio_coding/neteq/tools/rtp_file_source.h"
>-#include "modules/include/module_common_types.h"
> #include "rtc_base/ignore_wundef.h"
> #include "rtc_base/messagedigest.h"
> #include "rtc_base/numerics/safe_conversions.h"
> #include "rtc_base/protobuf_utils.h"
> #include "rtc_base/stringencode.h"
>+#include "rtc_base/system/arch.h"
> #include "test/field_trial.h"
> #include "test/gtest.h"
> #include "test/testsupport/fileutils.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> // This must come after test/gtest.h
> #include "rtc_base/flags.h"  // NOLINT(build/include)
>@@ -61,17 +63,17 @@ const std::string& PlatformChecksum(const std::string& checksum_general,
>                                     const std::string& checksum_win_32,
>                                     const std::string& checksum_win_64) {
> #if defined(WEBRTC_ANDROID)
>-  #ifdef WEBRTC_ARCH_64_BITS
>-    return checksum_android_64;
>-  #else
>-    return checksum_android_32;
>-  #endif  // WEBRTC_ARCH_64_BITS
>+#ifdef WEBRTC_ARCH_64_BITS
>+  return checksum_android_64;
>+#else
>+  return checksum_android_32;
>+#endif  // WEBRTC_ARCH_64_BITS
> #elif defined(WEBRTC_WIN)
>-  #ifdef WEBRTC_ARCH_64_BITS
>-    return checksum_win_64;
>-  #else
>-    return checksum_win_32;
>-  #endif  // WEBRTC_ARCH_64_BITS
>+#ifdef WEBRTC_ARCH_64_BITS
>+  return checksum_win_64;
>+#else
>+  return checksum_win_32;
>+#endif  // WEBRTC_ARCH_64_BITS
> #else
>   return checksum_general;
> #endif  // WEBRTC_WIN
>@@ -107,7 +109,8 @@ void Convert(const webrtc::RtcpStatistics& stats_raw,
>   stats->set_jitter(stats_raw.jitter);
> }
> 
>-void AddMessage(FILE* file, rtc::MessageDigest* digest,
>+void AddMessage(FILE* file,
>+                rtc::MessageDigest* digest,
>                 const std::string& message) {
>   int32_t size = message.length();
>   if (file)
>@@ -164,7 +167,8 @@ class ResultSink {
>   explicit ResultSink(const std::string& output_file);
>   ~ResultSink();
> 
>-  template<typename T> void AddResult(const T* test_results, size_t length);
>+  template <typename T>
>+  void AddResult(const T* test_results, size_t length);
> 
>   void AddResult(const NetEqNetworkStatistics& stats);
>   void AddResult(const RtcpStatistics& stats);
>@@ -190,7 +194,7 @@ ResultSink::~ResultSink() {
>     fclose(output_fp_);
> }
> 
>-template<typename T>
>+template <typename T>
> void ResultSink::AddResult(const T* test_results, size_t length) {
>   if (output_fp_) {
>     ASSERT_EQ(length, fwrite(test_results, sizeof(T), length, output_fp_));
>@@ -247,7 +251,7 @@ class NetEqDecodingTest : public ::testing::Test {
>   virtual void SetUp();
>   virtual void TearDown();
>   void SelectDecoders(NetEqDecoder* used_codec);
>-  void OpenInputFile(const std::string &rtp_file);
>+  void OpenInputFile(const std::string& rtp_file);
>   void Process();
> 
>   void DecodeAndCompare(const std::string& rtp_file,
>@@ -265,9 +269,11 @@ class NetEqDecodingTest : public ::testing::Test {
>                           uint8_t* payload,
>                           size_t* payload_len);
> 
>-  void WrapTest(uint16_t start_seq_no, uint32_t start_timestamp,
>+  void WrapTest(uint16_t start_seq_no,
>+                uint32_t start_timestamp,
>                 const std::set<uint16_t>& drop_seq_numbers,
>-                bool expect_seq_no_wrap, bool expect_timestamp_wrap);
>+                bool expect_seq_no_wrap,
>+                bool expect_timestamp_wrap);
> 
>   void LongCngWithClockDrift(double drift_factor,
>                              double network_freeze_ms,
>@@ -316,7 +322,7 @@ void NetEqDecodingTest::TearDown() {
>   delete neteq_;
> }
> 
>-void NetEqDecodingTest::OpenInputFile(const std::string &rtp_file) {
>+void NetEqDecodingTest::OpenInputFile(const std::string& rtp_file) {
>   rtp_source_.reset(test::RtpFileSource::Create(rtp_file));
> }
> 
>@@ -384,8 +390,8 @@ void NetEqDecodingTest::DecodeAndCompare(
>     ss << "Lap number " << i++ << " in DecodeAndCompare while loop";
>     SCOPED_TRACE(ss.str());  // Print out the parameter values on failure.
>     ASSERT_NO_FATAL_FAILURE(Process());
>-    ASSERT_NO_FATAL_FAILURE(output.AddResult(
>-        out_frame_.data(), out_frame_.samples_per_channel_));
>+    ASSERT_NO_FATAL_FAILURE(
>+        output.AddResult(out_frame_.data(), out_frame_.samples_per_channel_));
> 
>     // Query the network statistics API once per second
>     if (sim_clock_ % 1000 == 0) {
>@@ -447,7 +453,7 @@ void NetEqDecodingTest::PopulateCng(int frame_index,
>   rtp_info->ssrc = 0x1234;     // Just an arbitrary SSRC.
>   rtp_info->payloadType = 98;  // WB CNG.
>   rtp_info->markerBit = 0;
>-  payload[0] = 64;  // Noise level -64 dBov, quite arbitrarily chosen.
>+  payload[0] = 64;   // Noise level -64 dBov, quite arbitrarily chosen.
>   *payload_len = 1;  // Only noise level, no spectral parameters.
> }
> 
>@@ -462,36 +468,29 @@ TEST_F(NetEqDecodingTest, MAYBE_TestBitExactness) {
>   const std::string input_rtp_file =
>       webrtc::test::ResourcePath("audio_coding/neteq_universal_new", "rtp");
> 
>-  const std::string output_checksum = PlatformChecksum(
>-      "09fa7646e2ad032a0b156177b95f09012430f81f",
>-      "1c64eb8b55ce8878676c6a1e6ddd78f48de0668b",
>-      "not used",
>-      "09fa7646e2ad032a0b156177b95f09012430f81f",
>-      "759fef89a5de52bd17e733dc255c671ce86be909");
>+  const std::string output_checksum =
>+      PlatformChecksum("0c6dc227f781c81a229970f8fceda1a012498cba",
>+                       "15c4a2202877a414515e218bdb7992f0ad53e5af", "not used",
>+                       "0c6dc227f781c81a229970f8fceda1a012498cba",
>+                       "25fc4c863caa499aa447a5b8d059f5452cbcc500");
> 
>   const std::string network_stats_checksum =
>-      PlatformChecksum("5b4262ca328e5f066af5d34f3380521583dd20de",
>-                       "80235b6d727281203acb63b98f9a9e85d95f7ec0",
>-                       "not used",
>-                       "5b4262ca328e5f066af5d34f3380521583dd20de",
>-                       "5b4262ca328e5f066af5d34f3380521583dd20de");
>-
>-  const std::string rtcp_stats_checksum = PlatformChecksum(
>-      "b8880bf9fed2487efbddcb8d94b9937a29ae521d",
>-      "f3f7b3d3e71d7e635240b5373b57df6a7e4ce9d4",
>-      "not used",
>-      "b8880bf9fed2487efbddcb8d94b9937a29ae521d",
>-      "b8880bf9fed2487efbddcb8d94b9937a29ae521d");
>-
>-  DecodeAndCompare(input_rtp_file,
>-                   output_checksum,
>-                   network_stats_checksum,
>-                   rtcp_stats_checksum,
>-                   FLAG_gen_ref);
>+      PlatformChecksum("4b2370f5c794741d2a46be5c7935c66ef3fb53e9",
>+                       "e339cb2adf5ab3dfc21cb7205d670a34751e8336", "not used",
>+                       "4b2370f5c794741d2a46be5c7935c66ef3fb53e9",
>+                       "4b2370f5c794741d2a46be5c7935c66ef3fb53e9");
>+
>+  const std::string rtcp_stats_checksum =
>+      PlatformChecksum("b8880bf9fed2487efbddcb8d94b9937a29ae521d",
>+                       "f3f7b3d3e71d7e635240b5373b57df6a7e4ce9d4", "not used",
>+                       "b8880bf9fed2487efbddcb8d94b9937a29ae521d",
>+                       "b8880bf9fed2487efbddcb8d94b9937a29ae521d");
>+
>+  DecodeAndCompare(input_rtp_file, output_checksum, network_stats_checksum,
>+                   rtcp_stats_checksum, FLAG_gen_ref);
> }
> 
>-#if !defined(WEBRTC_IOS) &&                                         \
>-    defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) &&                      \
>+#if !defined(WEBRTC_IOS) && defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) && \
>     defined(WEBRTC_CODEC_OPUS)
> #define MAYBE_TestOpusBitExactness TestOpusBitExactness
> #else
>@@ -501,56 +500,38 @@ TEST_F(NetEqDecodingTest, MAYBE_TestOpusBitExactness) {
>   const std::string input_rtp_file =
>       webrtc::test::ResourcePath("audio_coding/neteq_opus", "rtp");
> 
>-  const std::string output_checksum = PlatformChecksum(
>-      "7ea28d7edf9395f4ac8e8d8dd3a9e5c620b1bf48",
>-      "5b1e691ab1c4465c742d6d944bc71e3b1c0e4c0e",
>-      "b096114dd8c233eaf2b0ce9802ac95af13933772",
>-      "7ea28d7edf9395f4ac8e8d8dd3a9e5c620b1bf48",
>-      "7ea28d7edf9395f4ac8e8d8dd3a9e5c620b1bf48");
>+  const std::string output_checksum =
>+      PlatformChecksum("14a63b3c7b925c82296be4bafc71bec85f2915c2",
>+                       "b7b7ed802b0e18ee416973bf3b9ae98599b0181d",
>+                       "5876e52dda90d5ca433c3726555b907b97c86374",
>+                       "14a63b3c7b925c82296be4bafc71bec85f2915c2",
>+                       "14a63b3c7b925c82296be4bafc71bec85f2915c2");
> 
>   const std::string network_stats_checksum =
>-      PlatformChecksum("9e72233c78baf685e500dd6c94212b30a4c5f27d",
>-                       "9a37270e4242fbd31e80bb47dc5e7ab82cf2d557",
>-                       "4f1e9734bc80a290faaf9d611efcb8d7802dbc4f",
>-                       "9e72233c78baf685e500dd6c94212b30a4c5f27d",
>-                       "9e72233c78baf685e500dd6c94212b30a4c5f27d");
>-
>-  const std::string rtcp_stats_checksum = PlatformChecksum(
>-      "e37c797e3de6a64dda88c9ade7a013d022a2e1e0",
>-      "e37c797e3de6a64dda88c9ade7a013d022a2e1e0",
>-      "e37c797e3de6a64dda88c9ade7a013d022a2e1e0",
>-      "e37c797e3de6a64dda88c9ade7a013d022a2e1e0",
>-      "e37c797e3de6a64dda88c9ade7a013d022a2e1e0");
>-
>-  DecodeAndCompare(input_rtp_file,
>-                   output_checksum,
>-                   network_stats_checksum,
>-                   rtcp_stats_checksum,
>-                   FLAG_gen_ref);
>-}
>+      PlatformChecksum("adb3272498e436d1c019cbfd71610e9510c54497",
>+                       "fa935a91abc7291db47428a2d7c5361b98713a92",
>+                       "42106aa5267300f709f63737707ef07afd9dac61",
>+                       "adb3272498e436d1c019cbfd71610e9510c54497",
>+                       "adb3272498e436d1c019cbfd71610e9510c54497");
> 
>-// This test fixture is identical to NetEqDecodingTest, except that it enables
>-// the WebRTC-NetEqOpusDtxDelayFix field trial.
>-// TODO(bugs.webrtc.org/8488): When the field trial is over and the feature is
>-// default enabled, remove this fixture class and let the
>-// TestOpusDtxBitExactness test build directly on NetEqDecodingTest.
>-class NetEqDecodingTestWithOpusDtxFieldTrial : public NetEqDecodingTest {
>- public:
>-  NetEqDecodingTestWithOpusDtxFieldTrial()
>-      : override_field_trials_("WebRTC-NetEqOpusDtxDelayFix/Enabled/") {}
>+  const std::string rtcp_stats_checksum =
>+      PlatformChecksum("e37c797e3de6a64dda88c9ade7a013d022a2e1e0",
>+                       "e37c797e3de6a64dda88c9ade7a013d022a2e1e0",
>+                       "e37c797e3de6a64dda88c9ade7a013d022a2e1e0",
>+                       "e37c797e3de6a64dda88c9ade7a013d022a2e1e0",
>+                       "e37c797e3de6a64dda88c9ade7a013d022a2e1e0");
> 
>- private:
>-  test::ScopedFieldTrials override_field_trials_;
>-};
>+  DecodeAndCompare(input_rtp_file, output_checksum, network_stats_checksum,
>+                   rtcp_stats_checksum, FLAG_gen_ref);
>+}
> 
>-#if !defined(WEBRTC_IOS) &&                                         \
>-    defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) &&                      \
>+#if !defined(WEBRTC_IOS) && defined(WEBRTC_NETEQ_UNITTEST_BITEXACT) && \
>     defined(WEBRTC_CODEC_OPUS)
> #define MAYBE_TestOpusDtxBitExactness TestOpusDtxBitExactness
> #else
> #define MAYBE_TestOpusDtxBitExactness DISABLED_TestOpusDtxBitExactness
> #endif
>-TEST_F(NetEqDecodingTestWithOpusDtxFieldTrial, MAYBE_TestOpusDtxBitExactness) {
>+TEST_F(NetEqDecodingTest, MAYBE_TestOpusDtxBitExactness) {
>   const std::string input_rtp_file =
>       webrtc::test::ResourcePath("audio_coding/neteq_opus_dtx", "rtp");
> 
>@@ -576,7 +557,7 @@ TEST_F(NetEqDecodingTestWithOpusDtxFieldTrial, MAYBE_TestOpusDtxBitExactness) {
> class NetEqDecodingTestFaxMode : public NetEqDecodingTest {
>  protected:
>   NetEqDecodingTestFaxMode() : NetEqDecodingTest() {
>-    config_.playout_mode = kPlayoutFax;
>+    config_.for_test_no_time_stretching = true;
>   }
>   void TestJitterBufferDelay(bool apply_packet_loss);
> };
>@@ -712,7 +693,7 @@ void NetEqDecodingTest::LongCngWithClockDrift(double drift_factor,
>   }
> 
>   EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
>-  rtc::Optional<uint32_t> playout_timestamp = neteq_->GetPlayoutTimestamp();
>+  absl::optional<uint32_t> playout_timestamp = neteq_->GetPlayoutTimestamp();
>   ASSERT_TRUE(playout_timestamp);
>   int32_t delay_before = timestamp - *playout_timestamp;
> 
>@@ -819,10 +800,8 @@ TEST_F(NetEqDecodingTest, LongCngWithNegativeClockDrift) {
>   const bool kGetAudioDuringFreezeRecovery = false;
>   const int kDelayToleranceMs = 20;
>   const int kMaxTimeToSpeechMs = 100;
>-  LongCngWithClockDrift(kDriftFactor,
>-                        kNetworkFreezeTimeMs,
>-                        kGetAudioDuringFreezeRecovery,
>-                        kDelayToleranceMs,
>+  LongCngWithClockDrift(kDriftFactor, kNetworkFreezeTimeMs,
>+                        kGetAudioDuringFreezeRecovery, kDelayToleranceMs,
>                         kMaxTimeToSpeechMs);
> }
> 
>@@ -833,10 +812,8 @@ TEST_F(NetEqDecodingTest, LongCngWithPositiveClockDrift) {
>   const bool kGetAudioDuringFreezeRecovery = false;
>   const int kDelayToleranceMs = 20;
>   const int kMaxTimeToSpeechMs = 100;
>-  LongCngWithClockDrift(kDriftFactor,
>-                        kNetworkFreezeTimeMs,
>-                        kGetAudioDuringFreezeRecovery,
>-                        kDelayToleranceMs,
>+  LongCngWithClockDrift(kDriftFactor, kNetworkFreezeTimeMs,
>+                        kGetAudioDuringFreezeRecovery, kDelayToleranceMs,
>                         kMaxTimeToSpeechMs);
> }
> 
>@@ -847,10 +824,8 @@ TEST_F(NetEqDecodingTest, LongCngWithNegativeClockDriftNetworkFreeze) {
>   const bool kGetAudioDuringFreezeRecovery = false;
>   const int kDelayToleranceMs = 50;
>   const int kMaxTimeToSpeechMs = 200;
>-  LongCngWithClockDrift(kDriftFactor,
>-                        kNetworkFreezeTimeMs,
>-                        kGetAudioDuringFreezeRecovery,
>-                        kDelayToleranceMs,
>+  LongCngWithClockDrift(kDriftFactor, kNetworkFreezeTimeMs,
>+                        kGetAudioDuringFreezeRecovery, kDelayToleranceMs,
>                         kMaxTimeToSpeechMs);
> }
> 
>@@ -861,10 +836,8 @@ TEST_F(NetEqDecodingTest, LongCngWithPositiveClockDriftNetworkFreeze) {
>   const bool kGetAudioDuringFreezeRecovery = false;
>   const int kDelayToleranceMs = 20;
>   const int kMaxTimeToSpeechMs = 100;
>-  LongCngWithClockDrift(kDriftFactor,
>-                        kNetworkFreezeTimeMs,
>-                        kGetAudioDuringFreezeRecovery,
>-                        kDelayToleranceMs,
>+  LongCngWithClockDrift(kDriftFactor, kNetworkFreezeTimeMs,
>+                        kGetAudioDuringFreezeRecovery, kDelayToleranceMs,
>                         kMaxTimeToSpeechMs);
> }
> 
>@@ -875,10 +848,8 @@ TEST_F(NetEqDecodingTest, LongCngWithPositiveClockDriftNetworkFreezeExtraPull) {
>   const bool kGetAudioDuringFreezeRecovery = true;
>   const int kDelayToleranceMs = 20;
>   const int kMaxTimeToSpeechMs = 100;
>-  LongCngWithClockDrift(kDriftFactor,
>-                        kNetworkFreezeTimeMs,
>-                        kGetAudioDuringFreezeRecovery,
>-                        kDelayToleranceMs,
>+  LongCngWithClockDrift(kDriftFactor, kNetworkFreezeTimeMs,
>+                        kGetAudioDuringFreezeRecovery, kDelayToleranceMs,
>                         kMaxTimeToSpeechMs);
> }
> 
>@@ -888,10 +859,8 @@ TEST_F(NetEqDecodingTest, LongCngWithoutClockDrift) {
>   const bool kGetAudioDuringFreezeRecovery = false;
>   const int kDelayToleranceMs = 10;
>   const int kMaxTimeToSpeechMs = 50;
>-  LongCngWithClockDrift(kDriftFactor,
>-                        kNetworkFreezeTimeMs,
>-                        kGetAudioDuringFreezeRecovery,
>-                        kDelayToleranceMs,
>+  LongCngWithClockDrift(kDriftFactor, kNetworkFreezeTimeMs,
>+                        kGetAudioDuringFreezeRecovery, kDelayToleranceMs,
>                         kMaxTimeToSpeechMs);
> }
> 
>@@ -964,9 +933,6 @@ TEST_F(NetEqDecodingTest, GetAudioBeforeInsertPacket) {
> 
> class NetEqBgnTest : public NetEqDecodingTest {
>  protected:
>-  virtual void TestCondition(double sum_squared_noise,
>-                             bool should_be_faded) = 0;
>-
>   void CheckBgn(int sampling_rate_hz) {
>     size_t expected_samples_per_channel = 0;
>     uint8_t payload_type = 0xFF;  // Invalid.
>@@ -1019,11 +985,11 @@ class NetEqBgnTest : public NetEqDecodingTest {
>       ASSERT_EQ(AudioFrame::kNormalSpeech, output.speech_type_);
> 
>       // Next packet.
>-      rtp_info.timestamp += rtc::checked_cast<uint32_t>(
>-          expected_samples_per_channel);
>+      rtp_info.timestamp +=
>+          rtc::checked_cast<uint32_t>(expected_samples_per_channel);
>       rtp_info.sequenceNumber++;
>-      receive_timestamp += rtc::checked_cast<uint32_t>(
>-          expected_samples_per_channel);
>+      receive_timestamp +=
>+          rtc::checked_cast<uint32_t>(expected_samples_per_channel);
>     }
> 
>     output.Reset();
>@@ -1058,7 +1024,7 @@ class NetEqBgnTest : public NetEqDecodingTest {
>         for (size_t k = 0;
>              k < output.num_channels_ * output.samples_per_channel_; ++k)
>           sum_squared += output_data[k] * output_data[k];
>-        TestCondition(sum_squared, n > kFadingThreshold);
>+        EXPECT_EQ(0, sum_squared);
>       } else {
>         EXPECT_EQ(AudioFrame::kPLC, output.speech_type_);
>       }
>@@ -1067,53 +1033,7 @@ class NetEqBgnTest : public NetEqDecodingTest {
>   }
> };
> 
>-class NetEqBgnTestOn : public NetEqBgnTest {
>- protected:
>-  NetEqBgnTestOn() : NetEqBgnTest() {
>-    config_.background_noise_mode = NetEq::kBgnOn;
>-  }
>-
>-  void TestCondition(double sum_squared_noise, bool /*should_be_faded*/) {
>-    EXPECT_NE(0, sum_squared_noise);
>-  }
>-};
>-
>-class NetEqBgnTestOff : public NetEqBgnTest {
>- protected:
>-  NetEqBgnTestOff() : NetEqBgnTest() {
>-    config_.background_noise_mode = NetEq::kBgnOff;
>-  }
>-
>-  void TestCondition(double sum_squared_noise, bool /*should_be_faded*/) {
>-    EXPECT_EQ(0, sum_squared_noise);
>-  }
>-};
>-
>-class NetEqBgnTestFade : public NetEqBgnTest {
>- protected:
>-  NetEqBgnTestFade() : NetEqBgnTest() {
>-    config_.background_noise_mode = NetEq::kBgnFade;
>-  }
>-
>-  void TestCondition(double sum_squared_noise, bool should_be_faded) {
>-    if (should_be_faded)
>-      EXPECT_EQ(0, sum_squared_noise);
>-  }
>-};
>-
>-TEST_F(NetEqBgnTestOn, RunTest) {
>-  CheckBgn(8000);
>-  CheckBgn(16000);
>-  CheckBgn(32000);
>-}
>-
>-TEST_F(NetEqBgnTestOff, RunTest) {
>-  CheckBgn(8000);
>-  CheckBgn(16000);
>-  CheckBgn(32000);
>-}
>-
>-TEST_F(NetEqBgnTestFade, RunTest) {
>+TEST_F(NetEqBgnTest, RunTest) {
>   CheckBgn(8000);
>   CheckBgn(16000);
>   CheckBgn(32000);
>@@ -1162,8 +1082,8 @@ void NetEqDecodingTest::WrapTest(uint16_t start_seq_no,
>       if (packets_inserted > 4) {
>         // Expect preferred and actual buffer size to be no more than 2 frames.
>         EXPECT_LE(network_stats.preferred_buffer_size_ms, kFrameSizeMs * 2);
>-        EXPECT_LE(network_stats.current_buffer_size_ms, kFrameSizeMs * 2 +
>-                  algorithmic_delay_ms_);
>+        EXPECT_LE(network_stats.current_buffer_size_ms,
>+                  kFrameSizeMs * 2 + algorithmic_delay_ms_);
>       }
>       last_seq_no = seq_no;
>       last_timestamp = timestamp;
>@@ -1184,7 +1104,7 @@ void NetEqDecodingTest::WrapTest(uint16_t start_seq_no,
>     ASSERT_EQ(1u, output.num_channels_);
> 
>     // Expect delay (in samples) to be less than 2 packets.
>-    rtc::Optional<uint32_t> playout_timestamp = neteq_->GetPlayoutTimestamp();
>+    absl::optional<uint32_t> playout_timestamp = neteq_->GetPlayoutTimestamp();
>     ASSERT_TRUE(playout_timestamp);
>     EXPECT_LE(timestamp - *playout_timestamp,
>               static_cast<uint32_t>(kSamples * 2));
>@@ -1229,8 +1149,8 @@ void NetEqDecodingTest::DuplicateCng() {
>   const int kSamples = kFrameSizeMs * kSampleRateKhz;
>   const size_t kPayloadBytes = kSamples * 2;
> 
>-  const int algorithmic_delay_samples = std::max(
>-      algorithmic_delay_ms_ * kSampleRateKhz, 5 * kSampleRateKhz / 8);
>+  const int algorithmic_delay_samples =
>+      std::max(algorithmic_delay_ms_ * kSampleRateKhz, 5 * kSampleRateKhz / 8);
>   // Insert three speech packets. Three are needed to get the frame length
>   // correct.
>   uint8_t payload[kPayloadBytes] = {0};
>@@ -1296,13 +1216,15 @@ void NetEqDecodingTest::DuplicateCng() {
>   ASSERT_EQ(0, neteq_->GetAudio(&out_frame_, &muted));
>   ASSERT_EQ(kBlockSize16kHz, out_frame_.samples_per_channel_);
>   EXPECT_EQ(AudioFrame::kNormalSpeech, out_frame_.speech_type_);
>-  rtc::Optional<uint32_t> playout_timestamp = neteq_->GetPlayoutTimestamp();
>+  absl::optional<uint32_t> playout_timestamp = neteq_->GetPlayoutTimestamp();
>   ASSERT_TRUE(playout_timestamp);
>   EXPECT_EQ(timestamp + kSamples - algorithmic_delay_samples,
>             *playout_timestamp);
> }
> 
>-TEST_F(NetEqDecodingTest, DiscardDuplicateCng) { DuplicateCng(); }
>+TEST_F(NetEqDecodingTest, DiscardDuplicateCng) {
>+  DuplicateCng();
>+}
> 
> TEST_F(NetEqDecodingTest, CngFirst) {
>   uint16_t seq_no = 0;
>@@ -1556,25 +1478,25 @@ namespace {
>     return ::testing::AssertionFailure() << "timestamp_ diff (" << a.timestamp_
>                                          << " != " << b.timestamp_ << ")";
>   if (a.sample_rate_hz_ != b.sample_rate_hz_)
>-    return ::testing::AssertionFailure() << "sample_rate_hz_ diff ("
>-                                         << a.sample_rate_hz_
>-                                         << " != " << b.sample_rate_hz_ << ")";
>+    return ::testing::AssertionFailure()
>+           << "sample_rate_hz_ diff (" << a.sample_rate_hz_
>+           << " != " << b.sample_rate_hz_ << ")";
>   if (a.samples_per_channel_ != b.samples_per_channel_)
>     return ::testing::AssertionFailure()
>            << "samples_per_channel_ diff (" << a.samples_per_channel_
>            << " != " << b.samples_per_channel_ << ")";
>   if (a.num_channels_ != b.num_channels_)
>-    return ::testing::AssertionFailure() << "num_channels_ diff ("
>-                                         << a.num_channels_
>-                                         << " != " << b.num_channels_ << ")";
>+    return ::testing::AssertionFailure()
>+           << "num_channels_ diff (" << a.num_channels_
>+           << " != " << b.num_channels_ << ")";
>   if (a.speech_type_ != b.speech_type_)
>-    return ::testing::AssertionFailure() << "speech_type_ diff ("
>-                                         << a.speech_type_
>-                                         << " != " << b.speech_type_ << ")";
>+    return ::testing::AssertionFailure()
>+           << "speech_type_ diff (" << a.speech_type_
>+           << " != " << b.speech_type_ << ")";
>   if (a.vad_activity_ != b.vad_activity_)
>-    return ::testing::AssertionFailure() << "vad_activity_ diff ("
>-                                         << a.vad_activity_
>-                                         << " != " << b.vad_activity_ << ")";
>+    return ::testing::AssertionFailure()
>+           << "vad_activity_ diff (" << a.vad_activity_
>+           << " != " << b.vad_activity_ << ")";
>   return ::testing::AssertionSuccess();
> }
> 
>@@ -1583,9 +1505,9 @@ namespace {
>   ::testing::AssertionResult res = AudioFramesEqualExceptData(a, b);
>   if (!res)
>     return res;
>-  if (memcmp(
>-      a.data(), b.data(),
>-      a.samples_per_channel_ * a.num_channels_ * sizeof(*a.data())) != 0) {
>+  if (memcmp(a.data(), b.data(),
>+             a.samples_per_channel_ * a.num_channels_ * sizeof(*a.data())) !=
>+      0) {
>     return ::testing::AssertionFailure() << "data_ diff";
>   }
>   return ::testing::AssertionSuccess();
>@@ -1803,4 +1725,32 @@ TEST_F(NetEqDecodingTestFaxMode, TestJitterBufferDelayWithLoss) {
>   TestJitterBufferDelay(true);
> }
> 
>+namespace test {
>+TEST(NetEqNoTimeStretchingMode, RunTest) {
>+  NetEq::Config config;
>+  config.for_test_no_time_stretching = true;
>+  auto codecs = NetEqTest::StandardDecoderMap();
>+  NetEqTest::ExtDecoderMap ext_codecs;
>+  NetEqPacketSourceInput::RtpHeaderExtensionMap rtp_ext_map = {
>+      {1, kRtpExtensionAudioLevel},
>+      {3, kRtpExtensionAbsoluteSendTime},
>+      {5, kRtpExtensionTransportSequenceNumber},
>+      {7, kRtpExtensionVideoContentType},
>+      {8, kRtpExtensionVideoTiming}};
>+  std::unique_ptr<NetEqInput> input(new NetEqRtpDumpInput(
>+      webrtc::test::ResourcePath("audio_coding/neteq_universal_new", "rtp"),
>+      rtp_ext_map));
>+  std::unique_ptr<TimeLimitedNetEqInput> input_time_limit(
>+      new TimeLimitedNetEqInput(std::move(input), 20000));
>+  std::unique_ptr<AudioSink> output(new VoidAudioSink);
>+  NetEqTest::Callbacks callbacks;
>+  NetEqTest test(config, codecs, ext_codecs, std::move(input_time_limit),
>+                 std::move(output), callbacks);
>+  test.Run();
>+  const auto stats = test.SimulationStats();
>+  EXPECT_EQ(0, stats.accelerate_rate);
>+  EXPECT_EQ(0, stats.preemptive_rate);
>+}
>+
>+}  // namespace test
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/normal.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/normal.cc
>index 48d723a81b0..83f7616d8e2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/normal.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/normal.cc
>@@ -27,7 +27,6 @@ namespace webrtc {
> int Normal::Process(const int16_t* input,
>                     size_t length,
>                     Modes last_mode,
>-                    int16_t* external_mute_factor_array,
>                     AudioMultiVector* output) {
>   if (length == 0) {
>     // Nothing to process.
>@@ -66,10 +65,8 @@ int Normal::Process(const int16_t* input,
>     size_t length_per_channel = length / output->Channels();
>     std::unique_ptr<int16_t[]> signal(new int16_t[length_per_channel]);
>     for (size_t channel_ix = 0; channel_ix < output->Channels(); ++channel_ix) {
>-      // Adjust muting factor (main muting factor times expand muting factor).
>-      external_mute_factor_array[channel_ix] = static_cast<int16_t>(
>-          (external_mute_factor_array[channel_ix] *
>-          expand_->MuteFactor(channel_ix)) >> 14);
>+      // Set muting factor to the same as expand muting factor.
>+      int16_t mute_factor = expand_->MuteFactor(channel_ix);
> 
>       (*output)[channel_ix].CopyTo(length_per_channel, 0, signal.get());
> 
>@@ -79,8 +76,7 @@ int Normal::Process(const int16_t* input,
>       // Adjust muting factor if needed (to BGN level).
>       size_t energy_length =
>           std::min(static_cast<size_t>(fs_mult * 64), length_per_channel);
>-      int scaling = 6 + fs_shift
>-          - WebRtcSpl_NormW32(decoded_max * decoded_max);
>+      int scaling = 6 + fs_shift - WebRtcSpl_NormW32(decoded_max * decoded_max);
>       scaling = std::max(scaling, 0);  // |scaling| should always be >= 0.
>       int32_t energy = WebRtcSpl_DotProductWithScale(signal.get(), signal.get(),
>                                                      energy_length, scaling);
>@@ -92,9 +88,8 @@ int Normal::Process(const int16_t* input,
>         energy = 0;
>       }
> 
>-      int mute_factor;
>-      if ((energy != 0) &&
>-          (energy > background_noise_.Energy(channel_ix))) {
>+      int local_mute_factor = 16384;  // 1.0 in Q14.
>+      if ((energy != 0) && (energy > background_noise_.Energy(channel_ix))) {
>         // Normalize new frame energy to 15 bits.
>         scaling = WebRtcSpl_NormW32(energy) - 16;
>         // We want background_noise_.energy() / energy in Q14.
>@@ -103,29 +98,30 @@ int Normal::Process(const int16_t* input,
>         int16_t energy_scaled =
>             static_cast<int16_t>(WEBRTC_SPL_SHIFT_W32(energy, scaling));
>         int32_t ratio = WebRtcSpl_DivW32W16(bgn_energy, energy_scaled);
>-        mute_factor = WebRtcSpl_SqrtFloor(ratio << 14);
>-      } else {
>-        mute_factor = 16384;  // 1.0 in Q14.
>-      }
>-      if (mute_factor > external_mute_factor_array[channel_ix]) {
>-        external_mute_factor_array[channel_ix] =
>-            static_cast<int16_t>(std::min(mute_factor, 16384));
>+        local_mute_factor =
>+            std::min(local_mute_factor, WebRtcSpl_SqrtFloor(ratio << 14));
>       }
>-
>-      // If muted increase by 0.64 for every 20 ms (NB/WB 0.0040/0.0020 in Q14).
>-      int increment = 64 / fs_mult;
>+      mute_factor = std::max<int16_t>(mute_factor, local_mute_factor);
>+      RTC_DCHECK_LE(mute_factor, 16384);
>+      RTC_DCHECK_GE(mute_factor, 0);
>+
>+      // If muted increase by 0.64 for every 20 ms (NB/WB 0.0040/0.0020 in Q14),
>+      // or as fast as it takes to come back to full gain within the frame
>+      // length.
>+      const int back_to_fullscale_inc =
>+          static_cast<int>((16384 - mute_factor) / length_per_channel);
>+      const int increment = std::max(64 / fs_mult, back_to_fullscale_inc);
>       for (size_t i = 0; i < length_per_channel; i++) {
>         // Scale with mute factor.
>         RTC_DCHECK_LT(channel_ix, output->Channels());
>         RTC_DCHECK_LT(i, output->Size());
>-        int32_t scaled_signal = (*output)[channel_ix][i] *
>-            external_mute_factor_array[channel_ix];
>+        int32_t scaled_signal = (*output)[channel_ix][i] * mute_factor;
>         // Shift 14 with proper rounding.
>         (*output)[channel_ix][i] =
>             static_cast<int16_t>((scaled_signal + 8192) >> 14);
>         // Increase mute_factor towards 16384.
>-        external_mute_factor_array[channel_ix] = static_cast<int16_t>(std::min(
>-            external_mute_factor_array[channel_ix] + increment, 16384));
>+        mute_factor =
>+            static_cast<int16_t>(std::min(mute_factor + increment, 16384));
>       }
> 
>       // Interpolate the expanded data into the new vector.
>@@ -153,8 +149,6 @@ int Normal::Process(const int16_t* input,
>     static const size_t kCngLength = 48;
>     RTC_DCHECK_LE(8 * fs_mult, kCngLength);
>     int16_t cng_output[kCngLength];
>-    // Reset mute factor and start up fresh.
>-    external_mute_factor_array[0] = 16384;
>     ComfortNoiseDecoder* cng_decoder = decoder_database_->GetActiveCngDecoder();
> 
>     if (cng_decoder) {
>@@ -186,28 +180,6 @@ int Normal::Process(const int16_t* input,
>     }
>     RTC_DCHECK_GT(win_up_Q14,
>                   (1 << 14) - 32);  // Worst case rouding is a length of 34
>-  } else if (external_mute_factor_array[0] < 16384) {
>-    // Previous was neither of Expand, FadeToBGN or RFC3389_CNG, but we are
>-    // still ramping up from previous muting.
>-    // If muted increase by 0.64 for every 20 ms (NB/WB 0.0040/0.0020 in Q14).
>-    int increment = 64 / fs_mult;
>-    size_t length_per_channel = length / output->Channels();
>-    for (size_t i = 0; i < length_per_channel; i++) {
>-      for (size_t channel_ix = 0; channel_ix < output->Channels();
>-          ++channel_ix) {
>-        // Scale with mute factor.
>-        RTC_DCHECK_LT(channel_ix, output->Channels());
>-        RTC_DCHECK_LT(i, output->Size());
>-        int32_t scaled_signal = (*output)[channel_ix][i] *
>-            external_mute_factor_array[channel_ix];
>-        // Shift 14 with proper rounding.
>-        (*output)[channel_ix][i] =
>-            static_cast<int16_t>((scaled_signal + 8192) >> 14);
>-        // Increase mute_factor towards 16384.
>-        external_mute_factor_array[channel_ix] = static_cast<int16_t>(std::min(
>-            16384, external_mute_factor_array[channel_ix] + increment));
>-      }
>-    }
>   }
> 
>   return static_cast<int>(length);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/normal.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/normal.h
>index ab02217a374..b13f16e50dc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/normal.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/normal.h
>@@ -20,7 +20,6 @@
> #include "rtc_base/checks.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/numerics/safe_conversions.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -53,11 +52,10 @@ class Normal {
>   // result is written to |output|. The number of channels allocated in
>   // |output| defines the number of channels that will be used when
>   // de-interleaving |input|. |last_mode| contains the mode used in the previous
>-  // GetAudio call (i.e., not the current one), and |external_mute_factor| is
>-  // a pointer to the mute factor in the NetEqImpl class.
>-  int Process(const int16_t* input, size_t length,
>+  // GetAudio call (i.e., not the current one).
>+  int Process(const int16_t* input,
>+              size_t length,
>               Modes last_mode,
>-              int16_t* external_mute_factor_array,
>               AudioMultiVector* output);
> 
>  private:
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/normal_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/normal_unittest.cc
>index b0655d9a819..106762afbb1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/normal_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/normal_unittest.cc
>@@ -39,7 +39,7 @@ int ExpandProcess120ms(AudioMultiVector* output) {
>   return 0;
> }
> 
>-} // namespace
>+}  // namespace
> 
> TEST(Normal, CreateAndDestroy) {
>   MockDecoderDatabase db;
>@@ -68,16 +68,10 @@ TEST(Normal, AvoidDivideByZero) {
>   Normal normal(fs, &db, bgn, &expand);
> 
>   int16_t input[1000] = {0};
>-  std::unique_ptr<int16_t[]> mute_factor_array(new int16_t[channels]);
>-  for (size_t i = 0; i < channels; ++i) {
>-    mute_factor_array[i] = 16384;
>-  }
>   AudioMultiVector output(channels);
> 
>   // Zero input length.
>-  EXPECT_EQ(
>-      0,
>-      normal.Process(input, 0, kModeExpand, mute_factor_array.get(), &output));
>+  EXPECT_EQ(0, normal.Process(input, 0, kModeExpand, &output));
>   EXPECT_EQ(0u, output.Size());
> 
>   // Try to make energy_length >> scaling = 0;
>@@ -90,11 +84,7 @@ TEST(Normal, AvoidDivideByZero) {
>   // and using this as a denominator would lead to problems.
>   int input_size_samples = 63;
>   EXPECT_EQ(input_size_samples,
>-            normal.Process(input,
>-                           input_size_samples,
>-                           kModeExpand,
>-                           mute_factor_array.get(),
>-                           &output));
>+            normal.Process(input, input_size_samples, kModeExpand, &output));
> 
>   EXPECT_CALL(db, Die());      // Called when |db| goes out of scope.
>   EXPECT_CALL(expand, Die());  // Called when |expand| goes out of scope.
>@@ -114,18 +104,11 @@ TEST(Normal, InputLengthAndChannelsDoNotMatch) {
>   Normal normal(fs, &db, bgn, &expand);
> 
>   int16_t input[1000] = {0};
>-  std::unique_ptr<int16_t[]> mute_factor_array(new int16_t[channels]);
>-  for (size_t i = 0; i < channels; ++i) {
>-    mute_factor_array[i] = 16384;
>-  }
>   AudioMultiVector output(channels);
> 
>   // Let the number of samples be one sample less than 80 samples per channel.
>   size_t input_len = 80 * channels - 1;
>-  EXPECT_EQ(
>-      0,
>-      normal.Process(
>-          input, input_len, kModeExpand, mute_factor_array.get(), &output));
>+  EXPECT_EQ(0, normal.Process(input, input_len, kModeExpand, &output));
>   EXPECT_EQ(0u, output.Size());
> 
>   EXPECT_CALL(db, Die());      // Called when |db| goes out of scope.
>@@ -147,23 +130,13 @@ TEST(Normal, LastModeExpand120msPacket) {
>   Normal normal(kFs, &db, bgn, &expand);
> 
>   int16_t input[kPacketsizeBytes] = {0};
>-
>-  std::unique_ptr<int16_t[]> mute_factor_array(new int16_t[kChannels]);
>-  for (size_t i = 0; i < kChannels; ++i) {
>-    mute_factor_array[i] = 16384;
>-  }
>-
>   AudioMultiVector output(kChannels);
> 
>   EXPECT_CALL(expand, SetParametersForNormalAfterExpand());
>   EXPECT_CALL(expand, Process(_)).WillOnce(Invoke(ExpandProcess120ms));
>   EXPECT_CALL(expand, Reset());
>   EXPECT_EQ(static_cast<int>(kPacketsizeBytes),
>-            normal.Process(input,
>-                           kPacketsizeBytes,
>-                           kModeExpand,
>-                           mute_factor_array.get(),
>-                           &output));
>+            normal.Process(input, kPacketsizeBytes, kModeExpand, &output));
> 
>   EXPECT_EQ(kPacketsizeBytes, output.Size());
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/packet.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/packet.h
>index 61b014429ab..45c56e1b686 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/packet.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/packet.h
>@@ -17,7 +17,6 @@
> #include "api/audio_codecs/audio_decoder.h"
> #include "modules/audio_coding/neteq/tick_timer.h"
> #include "rtc_base/buffer.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/packet_buffer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/packet_buffer.cc
>index dfffebdae64..c04534eaa66 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/packet_buffer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/packet_buffer.cc
>@@ -29,11 +29,8 @@ namespace {
> class NewTimestampIsLarger {
>  public:
>   explicit NewTimestampIsLarger(const Packet& new_packet)
>-      : new_packet_(new_packet) {
>-  }
>-  bool operator()(const Packet& packet) {
>-    return (new_packet_ >= packet);
>-  }
>+      : new_packet_(new_packet) {}
>+  bool operator()(const Packet& packet) { return (new_packet_ >= packet); }
> 
>  private:
>   const Packet& new_packet_;
>@@ -102,8 +99,7 @@ int PacketBuffer::InsertPacket(Packet&& packet, StatisticsCalculator* stats) {
>   // should be inserted. The list is searched from the back, since the most
>   // likely case is that the new packet should be near the end of the list.
>   PacketList::reverse_iterator rit = std::find_if(
>-      buffer_.rbegin(), buffer_.rend(),
>-      NewTimestampIsLarger(packet));
>+      buffer_.rbegin(), buffer_.rend(), NewTimestampIsLarger(packet));
> 
>   // The new packet is to be inserted to the right of |rit|. If it has the same
>   // timestamp as |rit|, which has a higher priority, do not insert the new
>@@ -118,7 +114,7 @@ int PacketBuffer::InsertPacket(Packet&& packet, StatisticsCalculator* stats) {
>   // packet.
>   PacketList::iterator it = rit.base();
>   if (it != buffer_.end() && packet.timestamp == it->timestamp) {
>-    LogPacketDiscarded(packet.priority.codec_level, stats);
>+    LogPacketDiscarded(it->priority.codec_level, stats);
>     it = buffer_.erase(it);
>   }
>   buffer_.insert(it, std::move(packet));  // Insert the packet at that position.
>@@ -129,8 +125,8 @@ int PacketBuffer::InsertPacket(Packet&& packet, StatisticsCalculator* stats) {
> int PacketBuffer::InsertPacketList(
>     PacketList* packet_list,
>     const DecoderDatabase& decoder_database,
>-    rtc::Optional<uint8_t>* current_rtp_payload_type,
>-    rtc::Optional<uint8_t>* current_cng_rtp_payload_type,
>+    absl::optional<uint8_t>* current_rtp_payload_type,
>+    absl::optional<uint8_t>* current_cng_rtp_payload_type,
>     StatisticsCalculator* stats) {
>   RTC_DCHECK(stats);
>   bool flushed = false;
>@@ -139,7 +135,7 @@ int PacketBuffer::InsertPacketList(
>       if (*current_cng_rtp_payload_type &&
>           **current_cng_rtp_payload_type != packet.payload_type) {
>         // New CNG payload type implies new codec type.
>-        *current_rtp_payload_type = rtc::nullopt;
>+        *current_rtp_payload_type = absl::nullopt;
>         Flush();
>         flushed = true;
>       }
>@@ -152,7 +148,7 @@ int PacketBuffer::InsertPacketList(
>            !EqualSampleRates(packet.payload_type,
>                              **current_cng_rtp_payload_type,
>                              decoder_database))) {
>-        *current_cng_rtp_payload_type = rtc::nullopt;
>+        *current_cng_rtp_payload_type = absl::nullopt;
>         Flush();
>         flushed = true;
>       }
>@@ -206,13 +202,13 @@ const Packet* PacketBuffer::PeekNextPacket() const {
>   return buffer_.empty() ? nullptr : &buffer_.front();
> }
> 
>-rtc::Optional<Packet> PacketBuffer::GetNextPacket() {
>+absl::optional<Packet> PacketBuffer::GetNextPacket() {
>   if (Empty()) {
>     // Buffer is empty.
>-    return rtc::nullopt;
>+    return absl::nullopt;
>   }
> 
>-  rtc::Optional<Packet> packet(std::move(buffer_.front()));
>+  absl::optional<Packet> packet(std::move(buffer_.front()));
>   // Assert that the packet sanity checks in InsertPacket method works.
>   RTC_DCHECK(!packet->empty());
>   buffer_.pop_front();
>@@ -285,6 +281,18 @@ size_t PacketBuffer::NumSamplesInBuffer(size_t last_decoded_length) const {
>   return num_samples;
> }
> 
>+bool PacketBuffer::ContainsDtxOrCngPacket(
>+    const DecoderDatabase* decoder_database) const {
>+  RTC_DCHECK(decoder_database);
>+  for (const Packet& packet : buffer_) {
>+    if ((packet.frame && packet.frame->IsDtxPacket()) ||
>+        decoder_database->IsComfortNoise(packet.payload_type)) {
>+      return true;
>+    }
>+  }
>+  return false;
>+}
>+
> void PacketBuffer::BufferStat(int* num_packets, int* max_num_packets) const {
>   *num_packets = static_cast<int>(buffer_.size());
>   *max_num_packets = static_cast<int>(max_number_of_packets_);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/packet_buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/packet_buffer.h
>index c83bf89c7de..7e34c1e9fce 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/packet_buffer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/packet_buffer.h
>@@ -11,11 +11,11 @@
> #ifndef MODULES_AUDIO_CODING_NETEQ_PACKET_BUFFER_H_
> #define MODULES_AUDIO_CODING_NETEQ_PACKET_BUFFER_H_
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
>+#include "modules/audio_coding/neteq/decoder_database.h"
> #include "modules/audio_coding/neteq/packet.h"
> #include "modules/include/module_common_types.h"
> #include "rtc_base/constructormagic.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -65,8 +65,8 @@ class PacketBuffer {
>   virtual int InsertPacketList(
>       PacketList* packet_list,
>       const DecoderDatabase& decoder_database,
>-      rtc::Optional<uint8_t>* current_rtp_payload_type,
>-      rtc::Optional<uint8_t>* current_cng_rtp_payload_type,
>+      absl::optional<uint8_t>* current_rtp_payload_type,
>+      absl::optional<uint8_t>* current_cng_rtp_payload_type,
>       StatisticsCalculator* stats);
> 
>   // Gets the timestamp for the first packet in the buffer and writes it to the
>@@ -89,7 +89,7 @@ class PacketBuffer {
> 
>   // Extracts the first packet in the buffer and returns it.
>   // Returns an empty optional if the buffer is empty.
>-  virtual rtc::Optional<Packet> GetNextPacket();
>+  virtual absl::optional<Packet> GetNextPacket();
> 
>   // Discards the first packet in the buffer. The packet is deleted.
>   // Returns PacketBuffer::kBufferEmpty if the buffer is empty,
>@@ -121,6 +121,10 @@ class PacketBuffer {
>   // duplicate and redundant packets.
>   virtual size_t NumSamplesInBuffer(size_t last_decoded_length) const;
> 
>+  // Returns true if the packet buffer contains any DTX or CNG packets.
>+  virtual bool ContainsDtxOrCngPacket(
>+      const DecoderDatabase* decoder_database) const;
>+
>   virtual void BufferStat(int* num_packets, int* max_num_packets) const;
> 
>   // Static method returning true if |timestamp| is older than |timestamp_limit|
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/packet_buffer_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/packet_buffer_unittest.cc
>index 0ddeb8a5854..cb40d7da32d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/packet_buffer_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/packet_buffer_unittest.cc
>@@ -175,21 +175,22 @@ TEST(PacketBuffer, InsertPacketList) {
> 
>   MockDecoderDatabase decoder_database;
>   auto factory = CreateBuiltinAudioDecoderFactory();
>-  const DecoderDatabase::DecoderInfo info(NetEqDecoder::kDecoderPCMu, factory);
>+  const DecoderDatabase::DecoderInfo info(NetEqDecoder::kDecoderPCMu,
>+                                          absl::nullopt, factory);
>   EXPECT_CALL(decoder_database, GetDecoderInfo(0))
>       .WillRepeatedly(Return(&info));
> 
>   StrictMock<MockStatisticsCalculator> mock_stats;
> 
>-  rtc::Optional<uint8_t> current_pt;
>-  rtc::Optional<uint8_t> current_cng_pt;
>+  absl::optional<uint8_t> current_pt;
>+  absl::optional<uint8_t> current_cng_pt;
>   EXPECT_EQ(PacketBuffer::kOK,
>             buffer.InsertPacketList(&list, decoder_database, &current_pt,
>                                     &current_cng_pt, &mock_stats));
>   EXPECT_TRUE(list.empty());  // The PacketBuffer should have depleted the list.
>   EXPECT_EQ(10u, buffer.NumPacketsInBuffer());
>   EXPECT_EQ(0, current_pt);      // Current payload type changed to 0.
>-  EXPECT_EQ(rtc::nullopt, current_cng_pt);  // CNG payload type not changed.
>+  EXPECT_EQ(absl::nullopt, current_cng_pt);  // CNG payload type not changed.
> 
>   buffer.Flush();  // Clean up.
> 
>@@ -219,24 +220,26 @@ TEST(PacketBuffer, InsertPacketListChangePayloadType) {
> 
>   MockDecoderDatabase decoder_database;
>   auto factory = CreateBuiltinAudioDecoderFactory();
>-  const DecoderDatabase::DecoderInfo info0(NetEqDecoder::kDecoderPCMu, factory);
>+  const DecoderDatabase::DecoderInfo info0(NetEqDecoder::kDecoderPCMu,
>+                                           absl::nullopt, factory);
>   EXPECT_CALL(decoder_database, GetDecoderInfo(0))
>       .WillRepeatedly(Return(&info0));
>-  const DecoderDatabase::DecoderInfo info1(NetEqDecoder::kDecoderPCMa, factory);
>+  const DecoderDatabase::DecoderInfo info1(NetEqDecoder::kDecoderPCMa,
>+                                           absl::nullopt, factory);
>   EXPECT_CALL(decoder_database, GetDecoderInfo(1))
>       .WillRepeatedly(Return(&info1));
> 
>   StrictMock<MockStatisticsCalculator> mock_stats;
> 
>-  rtc::Optional<uint8_t> current_pt;
>-  rtc::Optional<uint8_t> current_cng_pt;
>+  absl::optional<uint8_t> current_pt;
>+  absl::optional<uint8_t> current_cng_pt;
>   EXPECT_EQ(PacketBuffer::kFlushed,
>             buffer.InsertPacketList(&list, decoder_database, &current_pt,
>                                     &current_cng_pt, &mock_stats));
>   EXPECT_TRUE(list.empty());  // The PacketBuffer should have depleted the list.
>   EXPECT_EQ(1u, buffer.NumPacketsInBuffer());  // Only the last packet.
>   EXPECT_EQ(1, current_pt);      // Current payload type changed to 1.
>-  EXPECT_EQ(rtc::nullopt, current_cng_pt);  // CNG payload type not changed.
>+  EXPECT_EQ(absl::nullopt, current_cng_pt);  // CNG payload type not changed.
> 
>   buffer.Flush();  // Clean up.
> 
>@@ -310,7 +313,7 @@ TEST(PacketBuffer, ExtractOrderRedundancy) {
>   EXPECT_EQ(kExpectPacketsInBuffer, buffer.NumPacketsInBuffer());
> 
>   for (size_t i = 0; i < kExpectPacketsInBuffer; ++i) {
>-    const rtc::Optional<Packet> packet = buffer.GetNextPacket();
>+    const absl::optional<Packet> packet = buffer.GetNextPacket();
>     EXPECT_EQ(packet, expect_order[i]);  // Compare contents.
>   }
>   EXPECT_TRUE(buffer.Empty());
>@@ -404,11 +407,12 @@ TEST(PacketBuffer, Reordering) {
> 
>   MockDecoderDatabase decoder_database;
>   auto factory = CreateBuiltinAudioDecoderFactory();
>-  const DecoderDatabase::DecoderInfo info(NetEqDecoder::kDecoderPCMu, factory);
>+  const DecoderDatabase::DecoderInfo info(NetEqDecoder::kDecoderPCMu,
>+                                          absl::nullopt, factory);
>   EXPECT_CALL(decoder_database, GetDecoderInfo(0))
>       .WillRepeatedly(Return(&info));
>-  rtc::Optional<uint8_t> current_pt;
>-  rtc::Optional<uint8_t> current_cng_pt;
>+  absl::optional<uint8_t> current_pt;
>+  absl::optional<uint8_t> current_cng_pt;
> 
>   StrictMock<MockStatisticsCalculator> mock_stats;
> 
>@@ -420,7 +424,7 @@ TEST(PacketBuffer, Reordering) {
>   // Extract them and make sure that come out in the right order.
>   uint32_t current_ts = start_ts;
>   for (int i = 0; i < 10; ++i) {
>-    const rtc::Optional<Packet> packet = buffer.GetNextPacket();
>+    const absl::optional<Packet> packet = buffer.GetNextPacket();
>     ASSERT_TRUE(packet);
>     EXPECT_EQ(current_ts, packet->timestamp);
>     current_ts += ts_increment;
>@@ -444,11 +448,11 @@ TEST(PacketBuffer, CngFirstThenSpeechWithNewSampleRate) {
>   MockDecoderDatabase decoder_database;
>   auto factory = CreateBuiltinAudioDecoderFactory();
>   const DecoderDatabase::DecoderInfo info_cng(NetEqDecoder::kDecoderCNGnb,
>-                                              factory);
>+                                              absl::nullopt, factory);
>   EXPECT_CALL(decoder_database, GetDecoderInfo(kCngPt))
>       .WillRepeatedly(Return(&info_cng));
>   const DecoderDatabase::DecoderInfo info_speech(NetEqDecoder::kDecoderPCM16Bwb,
>-                                                 factory);
>+                                                 absl::nullopt, factory);
>   EXPECT_CALL(decoder_database, GetDecoderInfo(kSpeechPt))
>       .WillRepeatedly(Return(&info_speech));
> 
>@@ -456,8 +460,8 @@ TEST(PacketBuffer, CngFirstThenSpeechWithNewSampleRate) {
>   PacketGenerator gen(0, 0, kCngPt, 10);
>   PacketList list;
>   list.push_back(gen.NextPacket(kPayloadLen));
>-  rtc::Optional<uint8_t> current_pt;
>-  rtc::Optional<uint8_t> current_cng_pt;
>+  absl::optional<uint8_t> current_pt;
>+  absl::optional<uint8_t> current_cng_pt;
> 
>   StrictMock<MockStatisticsCalculator> mock_stats;
> 
>@@ -468,7 +472,7 @@ TEST(PacketBuffer, CngFirstThenSpeechWithNewSampleRate) {
>   EXPECT_EQ(1u, buffer.NumPacketsInBuffer());
>   ASSERT_TRUE(buffer.PeekNextPacket());
>   EXPECT_EQ(kCngPt, buffer.PeekNextPacket()->payload_type);
>-  EXPECT_EQ(current_pt, rtc::nullopt);  // Current payload type not set.
>+  EXPECT_EQ(current_pt, absl::nullopt);  // Current payload type not set.
>   EXPECT_EQ(kCngPt, current_cng_pt);  // CNG payload type set.
> 
>   // Insert second packet, which is wide-band speech.
>@@ -488,7 +492,7 @@ TEST(PacketBuffer, CngFirstThenSpeechWithNewSampleRate) {
>   EXPECT_EQ(kSpeechPt, buffer.PeekNextPacket()->payload_type);
> 
>   EXPECT_EQ(kSpeechPt, current_pt);  // Current payload type set.
>-  EXPECT_EQ(rtc::nullopt, current_cng_pt);  // CNG payload type reset.
>+  EXPECT_EQ(absl::nullopt, current_cng_pt);  // CNG payload type reset.
> 
>   buffer.Flush();                        // Clean up.
>   EXPECT_CALL(decoder_database, Die());  // Called when object is deleted.
>@@ -545,11 +549,12 @@ TEST(PacketBuffer, Failures) {
>   list.push_back(gen.NextPacket(payload_len));  // Valid packet.
>   MockDecoderDatabase decoder_database;
>   auto factory = CreateBuiltinAudioDecoderFactory();
>-  const DecoderDatabase::DecoderInfo info(NetEqDecoder::kDecoderPCMu, factory);
>+  const DecoderDatabase::DecoderInfo info(NetEqDecoder::kDecoderPCMu,
>+                                          absl::nullopt, factory);
>   EXPECT_CALL(decoder_database, GetDecoderInfo(0))
>       .WillRepeatedly(Return(&info));
>-  rtc::Optional<uint8_t> current_pt;
>-  rtc::Optional<uint8_t> current_cng_pt;
>+  absl::optional<uint8_t> current_pt;
>+  absl::optional<uint8_t> current_cng_pt;
>   EXPECT_EQ(PacketBuffer::kInvalidPacket,
>             buffer->InsertPacketList(&list, decoder_database, &current_pt,
>                                      &current_cng_pt, &mock_stats));
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/post_decode_vad.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/post_decode_vad.cc
>index a09d18ffd84..9999d6764bd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/post_decode_vad.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/post_decode_vad.cc
>@@ -45,7 +45,8 @@ void PostDecodeVad::Init() {
>   }
> }
> 
>-void PostDecodeVad::Update(int16_t* signal, size_t length,
>+void PostDecodeVad::Update(int16_t* signal,
>+                           size_t length,
>                            AudioDecoder::SpeechType speech_type,
>                            bool sid_frame,
>                            int fs_hz) {
>@@ -72,13 +73,13 @@ void PostDecodeVad::Update(int16_t* signal, size_t length,
>     active_speech_ = false;
>     // Loop through frame sizes 30, 20, and 10 ms.
>     for (int vad_frame_size_ms = 30; vad_frame_size_ms >= 10;
>-        vad_frame_size_ms -= 10) {
>+         vad_frame_size_ms -= 10) {
>       size_t vad_frame_size_samples =
>           static_cast<size_t>(vad_frame_size_ms * fs_hz / 1000);
>       while (length - vad_sample_index >= vad_frame_size_samples) {
>-        int vad_return = WebRtcVad_Process(
>-            vad_instance_, fs_hz, &signal[vad_sample_index],
>-            vad_frame_size_samples);
>+        int vad_return =
>+            WebRtcVad_Process(vad_instance_, fs_hz, &signal[vad_sample_index],
>+                              vad_frame_size_samples);
>         active_speech_ |= (vad_return == 1);
>         vad_sample_index += vad_frame_size_samples;
>       }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/post_decode_vad.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/post_decode_vad.h
>index 7b67bbe9b88..ad4f0823bb7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/post_decode_vad.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/post_decode_vad.h
>@@ -19,7 +19,6 @@
> #include "modules/audio_coding/neteq/defines.h"
> #include "modules/audio_coding/neteq/packet.h"
> #include "rtc_base/constructormagic.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -30,8 +29,7 @@ class PostDecodeVad {
>         running_(false),
>         active_speech_(true),
>         sid_interval_counter_(0),
>-        vad_instance_(NULL) {
>-  }
>+        vad_instance_(NULL) {}
> 
>   virtual ~PostDecodeVad();
> 
>@@ -46,8 +44,11 @@ class PostDecodeVad {
> 
>   // Updates post-decode VAD with the audio data in |signal| having |length|
>   // samples. The data is of type |speech_type|, at the sample rate |fs_hz|.
>-  void Update(int16_t* signal, size_t length,
>-              AudioDecoder::SpeechType speech_type, bool sid_frame, int fs_hz);
>+  void Update(int16_t* signal,
>+              size_t length,
>+              AudioDecoder::SpeechType speech_type,
>+              bool sid_frame,
>+              int fs_hz);
> 
>   // Accessors.
>   bool enabled() const { return enabled_; }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/preemptive_expand.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/preemptive_expand.cc
>index bc753895b18..4702078d672 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/preemptive_expand.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/preemptive_expand.cc
>@@ -50,8 +50,7 @@ void PreemptiveExpand::SetParametersForPassiveSpeech(size_t len,
>   // but we must ensure that best_correlation is not larger than the length of
>   // the new data.
>   // but we must ensure that best_correlation is not larger than the new data.
>-  *peak_index = std::min(*peak_index,
>-                         len - old_data_length_per_channel_);
>+  *peak_index = std::min(*peak_index, len - old_data_length_per_channel_);
> }
> 
> PreemptiveExpand::ReturnCodes PreemptiveExpand::CheckCriteriaAndStretch(
>@@ -68,13 +67,13 @@ PreemptiveExpand::ReturnCodes PreemptiveExpand::CheckCriteriaAndStretch(
>   // Check for strong correlation (>0.9 in Q14) and at least 15 ms new data,
>   // or passive speech.
>   if (((best_correlation > kCorrelationThreshold) &&
>-      (old_data_length_per_channel_ <= fs_mult_120)) ||
>+       (old_data_length_per_channel_ <= fs_mult_120)) ||
>       !active_speech) {
>     // Do accelerate operation by overlap add.
> 
>     // Set length of the first part, not to be modified.
>-    size_t unmodified_length = std::max(old_data_length_per_channel_,
>-                                        fs_mult_120);
>+    size_t unmodified_length =
>+        std::max(old_data_length_per_channel_, fs_mult_120);
>     // Copy first part, including cross-fade region.
>     output->PushBackInterleaved(
>         input, (unmodified_length + peak_index) * num_channels_);
>@@ -107,8 +106,8 @@ PreemptiveExpand* PreemptiveExpandFactory::Create(
>     size_t num_channels,
>     const BackgroundNoise& background_noise,
>     size_t overlap_samples) const {
>-  return new PreemptiveExpand(
>-      sample_rate_hz, num_channels, background_noise, overlap_samples);
>+  return new PreemptiveExpand(sample_rate_hz, num_channels, background_noise,
>+                              overlap_samples);
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/preemptive_expand.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/preemptive_expand.h
>index 303501db124..4d72616251c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/preemptive_expand.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/preemptive_expand.h
>@@ -16,7 +16,6 @@
> #include "modules/audio_coding/neteq/audio_multi_vector.h"
> #include "modules/audio_coding/neteq/time_stretch.h"
> #include "rtc_base/constructormagic.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -35,15 +34,14 @@ class PreemptiveExpand : public TimeStretch {
>                    size_t overlap_samples)
>       : TimeStretch(sample_rate_hz, num_channels, background_noise),
>         old_data_length_per_channel_(0),
>-        overlap_samples_(overlap_samples) {
>-  }
>+        overlap_samples_(overlap_samples) {}
> 
>   // This method performs the actual PreemptiveExpand operation. The samples are
>   // read from |input|, of length |input_length| elements, and are written to
>   // |output|. The number of samples added through time-stretching is
>   // is provided in the output |length_change_samples|. The method returns
>   // the outcome of the operation as an enumerator value.
>-  ReturnCodes Process(const int16_t *pw16_decoded,
>+  ReturnCodes Process(const int16_t* pw16_decoded,
>                       size_t len,
>                       size_t old_data_len,
>                       AudioMultiVector* output,
>@@ -77,11 +75,10 @@ struct PreemptiveExpandFactory {
>   PreemptiveExpandFactory() {}
>   virtual ~PreemptiveExpandFactory() {}
> 
>-  virtual PreemptiveExpand* Create(
>-      int sample_rate_hz,
>-      size_t num_channels,
>-      const BackgroundNoise& background_noise,
>-      size_t overlap_samples) const;
>+  virtual PreemptiveExpand* Create(int sample_rate_hz,
>+                                   size_t num_channels,
>+                                   const BackgroundNoise& background_noise,
>+                                   size_t overlap_samples) const;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/random_vector.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/random_vector.cc
>index c2df8cf80a4..ada175831c8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/random_vector.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/random_vector.cc
>@@ -13,29 +13,35 @@
> namespace webrtc {
> 
> const int16_t RandomVector::kRandomTable[RandomVector::kRandomTableSize] = {
>-    2680, 5532, 441, 5520, 16170, -5146, -1024, -8733, 3115, 9598, -10380,
>-    -4959, -1280, -21716, 7133, -1522, 13458, -3902, 2789, -675, 3441, 5016,
>-    -13599, -4003, -2739, 3922, -7209, 13352, -11617, -7241, 12905, -2314, 5426,
>-    10121, -9702, 11207, -13542, 1373, 816, -5934, -12504, 4798, 1811, 4112,
>-    -613, 201, -10367, -2960, -2419, 3442, 4299, -6116, -6092, 1552, -1650,
>-    -480, -1237, 18720, -11858, -8303, -8212, 865, -2890, -16968, 12052, -5845,
>-    -5912, 9777, -5665, -6294, 5426, -4737, -6335, 1652, 761, 3832, 641, -8552,
>-    -9084, -5753, 8146, 12156, -4915, 15086, -1231, -1869, 11749, -9319, -6403,
>-    11407, 6232, -1683, 24340, -11166, 4017, -10448, 3153, -2936, 6212, 2891,
>-    -866, -404, -4807, -2324, -1917, -2388, -6470, -3895, -10300, 5323, -5403,
>-    2205, 4640, 7022, -21186, -6244, -882, -10031, -3395, -12885, 7155, -5339,
>-    5079, -2645, -9515, 6622, 14651, 15852, 359, 122, 8246, -3502, -6696, -3679,
>-    -13535, -1409, -704, -7403, -4007, 1798, 279, -420, -12796, -14219, 1141,
>-    3359, 11434, 7049, -6684, -7473, 14283, -4115, -9123, -8969, 4152, 4117,
>-    13792, 5742, 16168, 8661, -1609, -6095, 1881, 14380, -5588, 6758, -6425,
>-    -22969, -7269, 7031, 1119, -1611, -5850, -11281, 3559, -8952, -10146, -4667,
>-    -16251, -1538, 2062, -1012, -13073, 227, -3142, -5265, 20, 5770, -7559,
>-    4740, -4819, 992, -8208, -7130, -4652, 6725, 7369, -1036, 13144, -1588,
>-    -5304, -2344, -449, -5705, -8894, 5205, -17904, -11188, -1022, 4852, 10101,
>-    -5255, -4200, -752, 7941, -1543, 5959, 14719, 13346, 17045, -15605, -1678,
>-    -1600, -9230, 68, 23348, 1172, 7750, 11212, -18227, 9956, 4161, 883, 3947,
>-    4341, 1014, -4889, -2603, 1246, -5630, -3596, -870, -1298, 2784, -3317,
>-    -6612, -20541, 4166, 4181, -8625, 3562, 12890, 4761, 3205, -12259, -8579 };
>+    2680,   5532,   441,    5520,   16170,  -5146,  -1024,  -8733,  3115,
>+    9598,   -10380, -4959,  -1280,  -21716, 7133,   -1522,  13458,  -3902,
>+    2789,   -675,   3441,   5016,   -13599, -4003,  -2739,  3922,   -7209,
>+    13352,  -11617, -7241,  12905,  -2314,  5426,   10121,  -9702,  11207,
>+    -13542, 1373,   816,    -5934,  -12504, 4798,   1811,   4112,   -613,
>+    201,    -10367, -2960,  -2419,  3442,   4299,   -6116,  -6092,  1552,
>+    -1650,  -480,   -1237,  18720,  -11858, -8303,  -8212,  865,    -2890,
>+    -16968, 12052,  -5845,  -5912,  9777,   -5665,  -6294,  5426,   -4737,
>+    -6335,  1652,   761,    3832,   641,    -8552,  -9084,  -5753,  8146,
>+    12156,  -4915,  15086,  -1231,  -1869,  11749,  -9319,  -6403,  11407,
>+    6232,   -1683,  24340,  -11166, 4017,   -10448, 3153,   -2936,  6212,
>+    2891,   -866,   -404,   -4807,  -2324,  -1917,  -2388,  -6470,  -3895,
>+    -10300, 5323,   -5403,  2205,   4640,   7022,   -21186, -6244,  -882,
>+    -10031, -3395,  -12885, 7155,   -5339,  5079,   -2645,  -9515,  6622,
>+    14651,  15852,  359,    122,    8246,   -3502,  -6696,  -3679,  -13535,
>+    -1409,  -704,   -7403,  -4007,  1798,   279,    -420,   -12796, -14219,
>+    1141,   3359,   11434,  7049,   -6684,  -7473,  14283,  -4115,  -9123,
>+    -8969,  4152,   4117,   13792,  5742,   16168,  8661,   -1609,  -6095,
>+    1881,   14380,  -5588,  6758,   -6425,  -22969, -7269,  7031,   1119,
>+    -1611,  -5850,  -11281, 3559,   -8952,  -10146, -4667,  -16251, -1538,
>+    2062,   -1012,  -13073, 227,    -3142,  -5265,  20,     5770,   -7559,
>+    4740,   -4819,  992,    -8208,  -7130,  -4652,  6725,   7369,   -1036,
>+    13144,  -1588,  -5304,  -2344,  -449,   -5705,  -8894,  5205,   -17904,
>+    -11188, -1022,  4852,   10101,  -5255,  -4200,  -752,   7941,   -1543,
>+    5959,   14719,  13346,  17045,  -15605, -1678,  -1600,  -9230,  68,
>+    23348,  1172,   7750,   11212,  -18227, 9956,   4161,   883,    3947,
>+    4341,   1014,   -4889,  -2603,  1246,   -5630,  -3596,  -870,   -1298,
>+    2784,   -3317,  -6612,  -20541, 4166,   4181,   -8625,  3562,   12890,
>+    4761,   3205,   -12259, -8579};
> 
> void RandomVector::Reset() {
>   seed_ = 777;
>@@ -51,7 +57,7 @@ void RandomVector::Generate(size_t length, int16_t* output) {
> }
> 
> void RandomVector::IncreaseSeedIncrement(int16_t increase_by) {
>-  seed_increment_+= increase_by;
>+  seed_increment_ += increase_by;
>   seed_increment_ &= kRandomTableSize - 1;
> }
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/random_vector.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/random_vector.h
>index 18adbe095c1..e8c7ba847c0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/random_vector.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/random_vector.h
>@@ -11,10 +11,10 @@
> #ifndef MODULES_AUDIO_CODING_NETEQ_RANDOM_VECTOR_H_
> #define MODULES_AUDIO_CODING_NETEQ_RANDOM_VECTOR_H_
> 
>-#include <string.h>  // size_t
>+#include <stddef.h>
>+#include <stdint.h>
> 
> #include "rtc_base/constructormagic.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -24,10 +24,7 @@ class RandomVector {
>   static const size_t kRandomTableSize = 256;
>   static const int16_t kRandomTable[kRandomTableSize];
> 
>-  RandomVector()
>-      : seed_(777),
>-        seed_increment_(1) {
>-  }
>+  RandomVector() : seed_(777), seed_increment_(1) {}
> 
>   void Reset();
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/red_payload_splitter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/red_payload_splitter.cc
>index 85e399c8637..f5435e8eab4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/red_payload_splitter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/red_payload_splitter.cc
>@@ -130,13 +130,16 @@ bool RedPayloadSplitter::SplitRed(PacketList* packet_list) {
>   return ret;
> }
> 
>-int RedPayloadSplitter::CheckRedPayloads(
>+void RedPayloadSplitter::CheckRedPayloads(
>     PacketList* packet_list,
>     const DecoderDatabase& decoder_database) {
>   int main_payload_type = -1;
>-  int num_deleted_packets = 0;
>   for (auto it = packet_list->begin(); it != packet_list->end(); /* */) {
>     uint8_t this_payload_type = it->payload_type;
>+    if (decoder_database.IsRed(this_payload_type)) {
>+      it = packet_list->erase(it);
>+      continue;
>+    }
>     if (!decoder_database.IsDtmf(this_payload_type) &&
>         !decoder_database.IsComfortNoise(this_payload_type)) {
>       if (main_payload_type == -1) {
>@@ -149,14 +152,12 @@ int RedPayloadSplitter::CheckRedPayloads(
>           // moves the iterator |it| to the next packet in the list. Thus, we
>           // do not have to increment it manually.
>           it = packet_list->erase(it);
>-          ++num_deleted_packets;
>           continue;
>         }
>       }
>     }
>     ++it;
>   }
>-  return num_deleted_packets;
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/red_payload_splitter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/red_payload_splitter.h
>index 1475b1b4a19..5e239ddc708 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/red_payload_splitter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/red_payload_splitter.h
>@@ -38,10 +38,9 @@ class RedPayloadSplitter {
> 
>   // Checks all packets in |packet_list|. Packets that are DTMF events or
>   // comfort noise payloads are kept. Except that, only one single payload type
>-  // is accepted. Any packet with another payload type is discarded.  Returns
>-  // the number of discarded packets.
>-  virtual int CheckRedPayloads(PacketList* packet_list,
>-                               const DecoderDatabase& decoder_database);
>+  // is accepted. Any packet with another payload type is discarded.
>+  virtual void CheckRedPayloads(PacketList* packet_list,
>+                                const DecoderDatabase& decoder_database);
> 
>  private:
>   RTC_DISALLOW_COPY_AND_ASSIGN(RedPayloadSplitter);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/red_payload_splitter_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/red_payload_splitter_unittest.cc
>index 4f511ad643c..52fc1ba1779 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/red_payload_splitter_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/red_payload_splitter_unittest.cc
>@@ -100,8 +100,8 @@ Packet CreateRedPayload(size_t num_payloads,
>     // Not the last block; set F = 1.
>     *payload_ptr |= 0x80;
>     ++payload_ptr;
>-    int this_offset = rtc::checked_cast<int>(
>-        (num_payloads - i - 1) * timestamp_offset);
>+    int this_offset =
>+        rtc::checked_cast<int>((num_payloads - i - 1) * timestamp_offset);
>     *payload_ptr = this_offset >> 6;
>     ++payload_ptr;
>     assert(kPayloadLength <= 1023);  // Max length described by 10 bits.
>@@ -299,7 +299,7 @@ TEST(RedPayloadSplitter, CheckRedPayloads) {
>   // easier to just register the payload types and let the actual implementation
>   // do its job.
>   DecoderDatabase decoder_database(
>-      new rtc::RefCountedObject<MockAudioDecoderFactory>);
>+      new rtc::RefCountedObject<MockAudioDecoderFactory>, absl::nullopt);
>   decoder_database.RegisterPayload(0, NetEqDecoder::kDecoderCNGnb, "cng-nb");
>   decoder_database.RegisterPayload(1, NetEqDecoder::kDecoderPCMu, "pcmu");
>   decoder_database.RegisterPayload(2, NetEqDecoder::kDecoderAVT, "avt");
>@@ -319,6 +319,30 @@ TEST(RedPayloadSplitter, CheckRedPayloads) {
>   EXPECT_TRUE(packet_list.empty());
> }
> 
>+// This test creates a RED packet where the payloads also have the payload type
>+// for RED. That is, some kind of weird nested RED packet. This is not supported
>+// and the splitter should discard all packets.
>+TEST(RedPayloadSplitter, CheckRedPayloadsRecursiveRed) {
>+  PacketList packet_list;
>+  for (uint8_t i = 0; i <= 3; ++i) {
>+    // Create packet with RED payload type, payload length 10 bytes, all 0.
>+    packet_list.push_back(CreatePacket(kRedPayloadType, 10, 0));
>+  }
>+
>+  // Use a real DecoderDatabase object here instead of a mock, since it is
>+  // easier to just register the payload types and let the actual implementation
>+  // do its job.
>+  DecoderDatabase decoder_database(
>+      new rtc::RefCountedObject<MockAudioDecoderFactory>, absl::nullopt);
>+  decoder_database.RegisterPayload(kRedPayloadType, NetEqDecoder::kDecoderRED,
>+                                   "red");
>+
>+  RedPayloadSplitter splitter;
>+  splitter.CheckRedPayloads(&packet_list, decoder_database);
>+
>+  EXPECT_TRUE(packet_list.empty());  // Should have dropped all packets.
>+}
>+
> // Packet A is split into A1, A2 and A3. But the length parameter is off, so
> // the last payloads should be discarded.
> TEST(RedPayloadSplitter, WrongPayloadLength) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/rtcp.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/rtcp.cc
>index 2885398584d..551eb5f75b4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/rtcp.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/rtcp.cc
>@@ -15,8 +15,6 @@
> 
> #include <algorithm>
> 
>-#include "modules/include/module_common_types.h"
>-
> namespace webrtc {
> 
> void Rtcp::Init(uint16_t start_sequence_number) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/rtcp.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/rtcp.h
>index ce2035b219c..b1de7eb7f48 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/rtcp.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/rtcp.h
>@@ -13,7 +13,6 @@
> 
> #include "modules/audio_coding/neteq/include/neteq.h"
> #include "rtc_base/constructormagic.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -22,9 +21,7 @@ struct RTPHeader;
> 
> class Rtcp {
>  public:
>-  Rtcp() {
>-    Init(0);
>-  }
>+  Rtcp() { Init(0); }
> 
>   ~Rtcp() {}
> 
>@@ -39,17 +36,17 @@ class Rtcp {
>   void GetStatistics(bool no_reset, RtcpStatistics* stats);
> 
>  private:
>-  uint16_t cycles_;  // The number of wrap-arounds for the sequence number.
>-  uint16_t max_seq_no_;  // The maximum sequence number received. Starts over
>-                         // from 0 after wrap-around.
>+  uint16_t cycles_;       // The number of wrap-arounds for the sequence number.
>+  uint16_t max_seq_no_;   // The maximum sequence number received. Starts over
>+                          // from 0 after wrap-around.
>   uint16_t base_seq_no_;  // The sequence number of the first received packet.
>   uint32_t received_packets_;  // The number of packets that have been received.
>   uint32_t received_packets_prior_;  // Number of packets received when last
>                                      // report was generated.
>   uint32_t expected_prior_;  // Expected number of packets, at the time of the
>                              // last report.
>-  int64_t jitter_;  // Current jitter value in Q4.
>-  int32_t transit_;  // Clock difference for previous packet.
>+  int64_t jitter_;           // Current jitter value in Q4.
>+  int32_t transit_;          // Clock difference for previous packet.
> 
>   RTC_DISALLOW_COPY_AND_ASSIGN(Rtcp);
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/statistics_calculator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/statistics_calculator.cc
>index 7228a5778e5..3d5744cb29f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/statistics_calculator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/statistics_calculator.cc
>@@ -42,8 +42,7 @@ StatisticsCalculator::PeriodicUmaLogger::PeriodicUmaLogger(
>     : uma_name_(uma_name),
>       report_interval_ms_(report_interval_ms),
>       max_value_(max_value),
>-      timer_(0) {
>-}
>+      timer_(0) {}
> 
> StatisticsCalculator::PeriodicUmaLogger::~PeriodicUmaLogger() = default;
> 
>@@ -66,8 +65,7 @@ StatisticsCalculator::PeriodicUmaCount::PeriodicUmaCount(
>     const std::string& uma_name,
>     int report_interval_ms,
>     int max_value)
>-    : PeriodicUmaLogger(uma_name, report_interval_ms, max_value) {
>-}
>+    : PeriodicUmaLogger(uma_name, report_interval_ms, max_value) {}
> 
> StatisticsCalculator::PeriodicUmaCount::~PeriodicUmaCount() {
>   // Log the count for the current (incomplete) interval.
>@@ -90,8 +88,7 @@ StatisticsCalculator::PeriodicUmaAverage::PeriodicUmaAverage(
>     const std::string& uma_name,
>     int report_interval_ms,
>     int max_value)
>-    : PeriodicUmaLogger(uma_name, report_interval_ms, max_value) {
>-}
>+    : PeriodicUmaLogger(uma_name, report_interval_ms, max_value) {}
> 
> StatisticsCalculator::PeriodicUmaAverage::~PeriodicUmaAverage() {
>   // Log the average for the current (incomplete) interval.
>@@ -153,34 +150,38 @@ void StatisticsCalculator::ResetMcu() {
> void StatisticsCalculator::ExpandedVoiceSamples(size_t num_samples,
>                                                 bool is_new_concealment_event) {
>   expanded_speech_samples_ += num_samples;
>-  ConcealedSamplesCorrection(rtc::dchecked_cast<int>(num_samples));
>+  ConcealedSamplesCorrection(rtc::dchecked_cast<int>(num_samples), true);
>   lifetime_stats_.concealment_events += is_new_concealment_event;
> }
> 
> void StatisticsCalculator::ExpandedNoiseSamples(size_t num_samples,
>                                                 bool is_new_concealment_event) {
>   expanded_noise_samples_ += num_samples;
>-  ConcealedSamplesCorrection(rtc::dchecked_cast<int>(num_samples));
>+  ConcealedSamplesCorrection(rtc::dchecked_cast<int>(num_samples), false);
>   lifetime_stats_.concealment_events += is_new_concealment_event;
> }
> 
> void StatisticsCalculator::ExpandedVoiceSamplesCorrection(int num_samples) {
>   expanded_speech_samples_ =
>       AddIntToSizeTWithLowerCap(num_samples, expanded_speech_samples_);
>-  ConcealedSamplesCorrection(num_samples);
>+  ConcealedSamplesCorrection(num_samples, true);
> }
> 
> void StatisticsCalculator::ExpandedNoiseSamplesCorrection(int num_samples) {
>   expanded_noise_samples_ =
>       AddIntToSizeTWithLowerCap(num_samples, expanded_noise_samples_);
>-  ConcealedSamplesCorrection(num_samples);
>+  ConcealedSamplesCorrection(num_samples, false);
> }
> 
>-void StatisticsCalculator::ConcealedSamplesCorrection(int num_samples) {
>+void StatisticsCalculator::ConcealedSamplesCorrection(int num_samples,
>+                                                      bool is_voice) {
>   if (num_samples < 0) {
>     // Store negative correction to subtract from future positive additions.
>     // See also the function comment in the header file.
>     concealed_samples_correction_ -= num_samples;
>+    if (is_voice) {
>+      voice_concealed_samples_correction_ -= num_samples;
>+    }
>     return;
>   }
> 
>@@ -188,6 +189,13 @@ void StatisticsCalculator::ConcealedSamplesCorrection(int num_samples) {
>       std::min(static_cast<size_t>(num_samples), concealed_samples_correction_);
>   concealed_samples_correction_ -= canceled_out;
>   lifetime_stats_.concealed_samples += num_samples - canceled_out;
>+
>+  if (is_voice) {
>+    const size_t voice_canceled_out = std::min(
>+        static_cast<size_t>(num_samples), voice_concealed_samples_correction_);
>+    voice_concealed_samples_correction_ -= voice_canceled_out;
>+    lifetime_stats_.voice_concealed_samples += num_samples - voice_canceled_out;
>+  }
> }
> 
> void StatisticsCalculator::PreemptiveExpandedSamples(size_t num_samples) {
>@@ -255,11 +263,10 @@ void StatisticsCalculator::StoreWaitingTime(int waiting_time_ms) {
>   waiting_times_.push_back(waiting_time_ms);
> }
> 
>-void StatisticsCalculator::GetNetworkStatistics(
>-    int fs_hz,
>-    size_t num_samples_in_buffers,
>-    size_t samples_per_packet,
>-    NetEqNetworkStatistics *stats) {
>+void StatisticsCalculator::GetNetworkStatistics(int fs_hz,
>+                                                size_t num_samples_in_buffers,
>+                                                size_t samples_per_packet,
>+                                                NetEqNetworkStatistics* stats) {
>   RTC_DCHECK_GT(fs_hz, 0);
>   RTC_DCHECK(stats);
> 
>@@ -280,20 +287,18 @@ void StatisticsCalculator::GetNetworkStatistics(
>       CalculateQ14Ratio(expanded_speech_samples_ + expanded_noise_samples_,
>                         timestamps_since_last_report_);
> 
>-  stats->speech_expand_rate =
>-      CalculateQ14Ratio(expanded_speech_samples_,
>-                        timestamps_since_last_report_);
>+  stats->speech_expand_rate = CalculateQ14Ratio(expanded_speech_samples_,
>+                                                timestamps_since_last_report_);
> 
>-  stats->secondary_decoded_rate =
>-      CalculateQ14Ratio(secondary_decoded_samples_,
>-                        timestamps_since_last_report_);
>+  stats->secondary_decoded_rate = CalculateQ14Ratio(
>+      secondary_decoded_samples_, timestamps_since_last_report_);
> 
>   const size_t discarded_secondary_samples =
>       discarded_secondary_packets_ * samples_per_packet;
>-  stats->secondary_discarded_rate = CalculateQ14Ratio(
>-      discarded_secondary_samples,
>-      static_cast<uint32_t>(discarded_secondary_samples +
>-        secondary_decoded_samples_));
>+  stats->secondary_discarded_rate =
>+      CalculateQ14Ratio(discarded_secondary_samples,
>+                        static_cast<uint32_t>(discarded_secondary_samples +
>+                                              secondary_decoded_samples_));
> 
>   if (waiting_times_.size() == 0) {
>     stats->mean_waiting_time_ms = -1;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/statistics_calculator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/statistics_calculator.h
>index c3d5c86cffe..5f7d06a23b3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/statistics_calculator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/statistics_calculator.h
>@@ -16,7 +16,6 @@
> 
> #include "modules/audio_coding/neteq/include/neteq.h"
> #include "rtc_base/constructormagic.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -98,7 +97,7 @@ class StatisticsCalculator {
>   void GetNetworkStatistics(int fs_hz,
>                             size_t num_samples_in_buffers,
>                             size_t samples_per_packet,
>-                            NetEqNetworkStatistics *stats);
>+                            NetEqNetworkStatistics* stats);
> 
>   // Populates |preferred_buffer_size_ms|, |jitter_peaks_found| and
>   // |clockdrift_ppm| in |stats|. This is a convenience method, and does not
>@@ -173,13 +172,14 @@ class StatisticsCalculator {
>   // If the correction is negative, it is cached and will be subtracted against
>   // future additions to the counter. This is meant to be called from
>   // Expanded{Voice,Noise}Samples{Correction}.
>-  void ConcealedSamplesCorrection(int num_samples);
>+  void ConcealedSamplesCorrection(int num_samples, bool is_voice);
> 
>   // Calculates numerator / denominator, and returns the value in Q14.
>   static uint16_t CalculateQ14Ratio(size_t numerator, uint32_t denominator);
> 
>   NetEqLifetimeStatistics lifetime_stats_;
>   size_t concealed_samples_correction_ = 0;
>+  size_t voice_concealed_samples_correction_ = 0;
>   size_t preemptive_samples_;
>   size_t accelerate_samples_;
>   size_t added_zero_samples_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/sync_buffer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/sync_buffer.cc
>index 28d7649fdb3..82ca16fe602 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/sync_buffer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/sync_buffer.cc
>@@ -27,7 +27,7 @@ void SyncBuffer::PushBack(const AudioMultiVector& append_this) {
>     next_index_ -= samples_added;
>   } else {
>     // This means that we are pushing out future data that was never used.
>-//    assert(false);
>+    //    assert(false);
>     // TODO(hlundin): This assert must be disabled to support 60 ms frames.
>     // This should not happen even for 60 ms frames, but it does. Investigate
>     // why.
>@@ -75,9 +75,8 @@ void SyncBuffer::GetNextAudioInterleaved(size_t requested_len,
>   RTC_DCHECK(output);
>   const size_t samples_to_read = std::min(FutureLength(), requested_len);
>   output->ResetWithoutMuting();
>-  const size_t tot_samples_read =
>-      ReadInterleavedFromIndex(next_index_, samples_to_read,
>-                               output->mutable_data());
>+  const size_t tot_samples_read = ReadInterleavedFromIndex(
>+      next_index_, samples_to_read, output->mutable_data());
>   const size_t samples_read_per_channel = tot_samples_read / Channels();
>   next_index_ += samples_read_per_channel;
>   output->num_channels_ = Channels();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/sync_buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/sync_buffer.h
>index ab9ff525ef1..3833cb29855 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/sync_buffer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/sync_buffer.h
>@@ -11,10 +11,9 @@
> #ifndef MODULES_AUDIO_CODING_NETEQ_SYNC_BUFFER_H_
> #define MODULES_AUDIO_CODING_NETEQ_SYNC_BUFFER_H_
> 
>+#include "api/audio/audio_frame.h"
> #include "modules/audio_coding/neteq/audio_multi_vector.h"
>-#include "modules/include/module_common_types.h"
> #include "rtc_base/constructormagic.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -92,7 +91,7 @@ class SyncBuffer : public AudioMultiVector {
>  private:
>   size_t next_index_;
>   uint32_t end_timestamp_;  // The timestamp of the last sample in the buffer.
>-  size_t dtmf_index_;  // Index to the first non-DTMF sample in the buffer.
>+  size_t dtmf_index_;       // Index to the first non-DTMF sample in the buffer.
> 
>   RTC_DISALLOW_COPY_AND_ASSIGN(SyncBuffer);
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/test/neteq_ilbc_quality_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/test/neteq_ilbc_quality_test.cc
>index bca401a5d22..ad61235cf28 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/test/neteq_ilbc_quality_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/test/neteq_ilbc_quality_test.cc
>@@ -52,7 +52,8 @@ class NetEqIlbcQualityTest : public NetEqQualityTest {
> 
>   int EncodeBlock(int16_t* in_data,
>                   size_t block_size_samples,
>-                  rtc::Buffer* payload, size_t max_bytes) override {
>+                  rtc::Buffer* payload,
>+                  size_t max_bytes) override {
>     const size_t kFrameSizeSamples = 80;  // Samples per 10 ms.
>     size_t encoded_samples = 0;
>     uint32_t dummy_timestamp = 0;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc
>index d88f789f02b..94984b87e6f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/test/neteq_isac_quality_test.cc
>@@ -30,8 +30,11 @@ class NetEqIsacQualityTest : public NetEqQualityTest {
>   NetEqIsacQualityTest();
>   void SetUp() override;
>   void TearDown() override;
>-  int EncodeBlock(int16_t* in_data, size_t block_size_samples,
>-                  rtc::Buffer* payload, size_t max_bytes) override;
>+  int EncodeBlock(int16_t* in_data,
>+                  size_t block_size_samples,
>+                  rtc::Buffer* payload,
>+                  size_t max_bytes) override;
>+
>  private:
>   ISACFIX_MainStruct* isac_encoder_;
>   int bit_rate_kbps_;
>@@ -44,10 +47,10 @@ NetEqIsacQualityTest::NetEqIsacQualityTest()
>                        NetEqDecoder::kDecoderISAC),
>       isac_encoder_(NULL),
>       bit_rate_kbps_(FLAG_bit_rate_kbps) {
>-    // Flag validation
>-    RTC_CHECK(FLAG_bit_rate_kbps >= 10 && FLAG_bit_rate_kbps <= 32)
>-        << "Invalid bit rate, should be between 10 and 32 kbps.";
>-  }
>+  // Flag validation
>+  RTC_CHECK(FLAG_bit_rate_kbps >= 10 && FLAG_bit_rate_kbps <= 32)
>+      << "Invalid bit rate, should be between 10 and 32 kbps.";
>+}
> 
> void NetEqIsacQualityTest::SetUp() {
>   ASSERT_EQ(1u, channels_) << "iSAC supports only mono audio.";
>@@ -69,7 +72,8 @@ void NetEqIsacQualityTest::TearDown() {
> 
> int NetEqIsacQualityTest::EncodeBlock(int16_t* in_data,
>                                       size_t block_size_samples,
>-                                      rtc::Buffer* payload, size_t max_bytes) {
>+                                      rtc::Buffer* payload,
>+                                      size_t max_bytes) {
>   // ISAC takes 10 ms for every call.
>   const int subblocks = kIsacBlockDurationMs / 10;
>   const int subblock_length = 10 * kIsacInputSamplingKhz;
>@@ -80,11 +84,11 @@ int NetEqIsacQualityTest::EncodeBlock(int16_t* in_data,
>     // The Isac encoder does not perform encoding (and returns 0) until it
>     // receives a sequence of sub-blocks that amount to the frame duration.
>     EXPECT_EQ(0, value);
>-    payload->AppendData(max_bytes, [&] (rtc::ArrayView<uint8_t> payload) {
>-        value = WebRtcIsacfix_Encode(isac_encoder_, &in_data[pointer],
>-                                     payload.data());
>-        return (value >= 0) ? static_cast<size_t>(value) : 0;
>-      });
>+    payload->AppendData(max_bytes, [&](rtc::ArrayView<uint8_t> payload) {
>+      value = WebRtcIsacfix_Encode(isac_encoder_, &in_data[pointer],
>+                                   payload.data());
>+      return (value >= 0) ? static_cast<size_t>(value) : 0;
>+    });
>   }
>   EXPECT_GT(value, 0);
>   return value;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc
>index c2542b69376..6861e4c55aa 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/test/neteq_opus_quality_test.cc
>@@ -8,8 +8,8 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "modules/audio_coding/codecs/opus/opus_interface.h"
> #include "modules/audio_coding/codecs/opus/opus_inst.h"
>+#include "modules/audio_coding/codecs/opus/opus_interface.h"
> #include "modules/audio_coding/neteq/tools/neteq_quality_test.h"
> #include "rtc_base/flags.h"
> 
>@@ -24,8 +24,10 @@ static const int kOpusSamplingKhz = 48;
> 
> DEFINE_int(bit_rate_kbps, 32, "Target bit rate (kbps).");
> 
>-DEFINE_int(complexity, 10, "Complexity: 0 ~ 10 -- defined as in Opus"
>-    "specification.");
>+DEFINE_int(complexity,
>+           10,
>+           "Complexity: 0 ~ 10 -- defined as in Opus"
>+           "specification.");
> 
> DEFINE_int(maxplaybackrate, 48000, "Maximum playback rate (Hz).");
> 
>@@ -46,8 +48,11 @@ class NetEqOpusQualityTest : public NetEqQualityTest {
>   NetEqOpusQualityTest();
>   void SetUp() override;
>   void TearDown() override;
>-  int EncodeBlock(int16_t* in_data, size_t block_size_samples,
>-                  rtc::Buffer* payload, size_t max_bytes) override;
>+  int EncodeBlock(int16_t* in_data,
>+                  size_t block_size_samples,
>+                  rtc::Buffer* payload,
>+                  size_t max_bytes) override;
>+
>  private:
>   WebRtcOpusEncInst* opus_encoder_;
>   OpusRepacketizer* repacketizer_;
>@@ -120,8 +125,7 @@ void NetEqOpusQualityTest::SetUp() {
>   }
>   EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_encoder_, complexity_));
>   EXPECT_EQ(0, WebRtcOpus_SetMaxPlaybackRate(opus_encoder_, maxplaybackrate_));
>-  EXPECT_EQ(0, WebRtcOpus_SetPacketLossRate(opus_encoder_,
>-                                            target_loss_rate_));
>+  EXPECT_EQ(0, WebRtcOpus_SetPacketLossRate(opus_encoder_, target_loss_rate_));
>   NetEqQualityTest::SetUp();
> }
> 
>@@ -134,26 +138,25 @@ void NetEqOpusQualityTest::TearDown() {
> 
> int NetEqOpusQualityTest::EncodeBlock(int16_t* in_data,
>                                       size_t block_size_samples,
>-                                      rtc::Buffer* payload, size_t max_bytes) {
>+                                      rtc::Buffer* payload,
>+                                      size_t max_bytes) {
>   EXPECT_EQ(block_size_samples, sub_block_size_samples_ * sub_packets_);
>   int16_t* pointer = in_data;
>   int value;
>   opus_repacketizer_init(repacketizer_);
>   for (int idx = 0; idx < sub_packets_; idx++) {
>-    payload->AppendData(max_bytes, [&] (rtc::ArrayView<uint8_t> payload) {
>-        value = WebRtcOpus_Encode(opus_encoder_,
>-                                  pointer, sub_block_size_samples_,
>-                                  max_bytes, payload.data());
>+    payload->AppendData(max_bytes, [&](rtc::ArrayView<uint8_t> payload) {
>+      value = WebRtcOpus_Encode(opus_encoder_, pointer, sub_block_size_samples_,
>+                                max_bytes, payload.data());
> 
>-        Log() << "Encoded a frame with Opus mode "
>-              << (value == 0 ? 0 : payload[0] >> 3)
>-              << std::endl;
>+      Log() << "Encoded a frame with Opus mode "
>+            << (value == 0 ? 0 : payload[0] >> 3) << std::endl;
> 
>-        return (value >= 0) ? static_cast<size_t>(value) : 0;
>-      });
>+      return (value >= 0) ? static_cast<size_t>(value) : 0;
>+    });
> 
>-    if (OPUS_OK != opus_repacketizer_cat(repacketizer_,
>-                                         payload->data(), value)) {
>+    if (OPUS_OK !=
>+        opus_repacketizer_cat(repacketizer_, payload->data(), value)) {
>       opus_repacketizer_init(repacketizer_);
>       // If the repacketization fails, we discard this frame.
>       return 0;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/test/neteq_pcm16b_quality_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/test/neteq_pcm16b_quality_test.cc
>new file mode 100644
>index 00000000000..8872b94a063
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/test/neteq_pcm16b_quality_test.cc
>@@ -0,0 +1,82 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <memory>
>+
>+#include "modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
>+
>+#include "modules/audio_coding/neteq/tools/neteq_quality_test.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/flags.h"
>+#include "rtc_base/numerics/safe_conversions.h"
>+#include "test/testsupport/fileutils.h"
>+
>+using testing::InitGoogleTest;
>+
>+namespace webrtc {
>+namespace test {
>+namespace {
>+static const int kInputSampleRateKhz = 48;
>+static const int kOutputSampleRateKhz = 48;
>+
>+DEFINE_int(frame_size_ms, 20, "Codec frame size (milliseconds).");
>+
>+}  // namespace
>+
>+class NetEqPcm16bQualityTest : public NetEqQualityTest {
>+ protected:
>+  NetEqPcm16bQualityTest()
>+      : NetEqQualityTest(FLAG_frame_size_ms,
>+                         kInputSampleRateKhz,
>+                         kOutputSampleRateKhz,
>+                         NetEqDecoder::kDecoderPCM16Bswb48kHz) {
>+    // Flag validation
>+    RTC_CHECK(FLAG_frame_size_ms >= 10 && FLAG_frame_size_ms <= 60 &&
>+              (FLAG_frame_size_ms % 10) == 0)
>+        << "Invalid frame size, should be 10, 20, ..., 60 ms.";
>+  }
>+
>+  void SetUp() override {
>+    AudioEncoderPcm16B::Config config;
>+    config.frame_size_ms = FLAG_frame_size_ms;
>+    config.sample_rate_hz = 48000;
>+    config.num_channels = channels_;
>+    encoder_.reset(new AudioEncoderPcm16B(config));
>+    NetEqQualityTest::SetUp();
>+  }
>+
>+  int EncodeBlock(int16_t* in_data,
>+                  size_t block_size_samples,
>+                  rtc::Buffer* payload,
>+                  size_t max_bytes) override {
>+    const size_t kFrameSizeSamples = 480;  // Samples per 10 ms.
>+    size_t encoded_samples = 0;
>+    uint32_t dummy_timestamp = 0;
>+    AudioEncoder::EncodedInfo info;
>+    do {
>+      info = encoder_->Encode(dummy_timestamp,
>+                              rtc::ArrayView<const int16_t>(
>+                                  in_data + encoded_samples, kFrameSizeSamples),
>+                              payload);
>+      encoded_samples += kFrameSizeSamples;
>+    } while (info.encoded_bytes == 0);
>+    return rtc::checked_cast<int>(info.encoded_bytes);
>+  }
>+
>+ private:
>+  std::unique_ptr<AudioEncoderPcm16B> encoder_;
>+};
>+
>+TEST_F(NetEqPcm16bQualityTest, Test) {
>+  Simulate();
>+}
>+
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc
>index bc3c1682e43..54ff849739a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/test/neteq_pcmu_quality_test.cc
>@@ -52,7 +52,8 @@ class NetEqPcmuQualityTest : public NetEqQualityTest {
> 
>   int EncodeBlock(int16_t* in_data,
>                   size_t block_size_samples,
>-                  rtc::Buffer* payload, size_t max_bytes) override {
>+                  rtc::Buffer* payload,
>+                  size_t max_bytes) override {
>     const size_t kFrameSizeSamples = 80;  // Samples per 10 ms.
>     size_t encoded_samples = 0;
>     uint32_t dummy_timestamp = 0;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/test/neteq_performance_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/test/neteq_performance_unittest.cc
>index 0510af8004e..90a205e7c1f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/test/neteq_performance_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/test/neteq_performance_unittest.cc
>@@ -9,10 +9,9 @@
>  */
> 
> #include "modules/audio_coding/neteq/tools/neteq_performance_test.h"
>+#include "system_wrappers/include/field_trial.h"
> #include "test/gtest.h"
> #include "test/testsupport/perf_test.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>-#include "system_wrappers/include/field_trial.h"
> 
> // Runs a test with 10% packet losses and 10% clock drift, to exercise
> // both loss concealment and time-stretching code.
>@@ -27,8 +26,8 @@ TEST(NetEqPerformanceTest, Run) {
>           : kSimulationTimeMs,
>       kLossPeriod, kDriftFactor);
>   ASSERT_GT(runtime, 0);
>-  webrtc::test::PrintResult(
>-      "neteq_performance", "", "10_pl_10_drift", runtime, "ms", true);
>+  webrtc::test::PrintResult("neteq_performance", "", "10_pl_10_drift", runtime,
>+                            "ms", true);
> }
> 
> // Runs a test with neither packet losses nor clock drift, to put
>@@ -37,7 +36,7 @@ TEST(NetEqPerformanceTest, Run) {
> TEST(NetEqPerformanceTest, RunClean) {
>   const int kSimulationTimeMs = 10000000;
>   const int kQuickSimulationTimeMs = 100000;
>-  const int kLossPeriod = 0;  // No losses.
>+  const int kLossPeriod = 0;        // No losses.
>   const double kDriftFactor = 0.0;  // No clock drift.
>   int64_t runtime = webrtc::test::NetEqPerformanceTest::Run(
>       webrtc::field_trial::IsEnabled("WebRTC-QuickPerfTest")
>@@ -45,6 +44,6 @@ TEST(NetEqPerformanceTest, RunClean) {
>           : kSimulationTimeMs,
>       kLossPeriod, kDriftFactor);
>   ASSERT_GT(runtime, 0);
>-  webrtc::test::PrintResult(
>-      "neteq_performance", "", "0_pl_0_drift", runtime, "ms", true);
>+  webrtc::test::PrintResult("neteq_performance", "", "0_pl_0_drift", runtime,
>+                            "ms", true);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/test/neteq_speed_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/test/neteq_speed_test.cc
>index ad123fe9261..c486ac60bd7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/test/neteq_speed_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/test/neteq_speed_test.cc
>@@ -15,27 +15,27 @@
> #include "modules/audio_coding/neteq/tools/neteq_performance_test.h"
> #include "rtc_base/flags.h"
> #include "test/testsupport/fileutils.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> // Define command line flags.
> DEFINE_int(runtime_ms, 10000, "Simulated runtime in ms.");
>-DEFINE_int(lossrate, 10,
>-           "Packet lossrate; drop every N packets.");
>-DEFINE_float(drift, 0.1f,
>-             "Clockdrift factor.");
>+DEFINE_int(lossrate, 10, "Packet lossrate; drop every N packets.");
>+DEFINE_float(drift, 0.1f, "Clockdrift factor.");
> DEFINE_bool(help, false, "Print this message.");
> 
> int main(int argc, char* argv[]) {
>   std::string program_name = argv[0];
>-  std::string usage = "Tool for measuring the speed of NetEq.\n"
>-      "Usage: " + program_name + " [options]\n\n"
>+  std::string usage =
>+      "Tool for measuring the speed of NetEq.\n"
>+      "Usage: " +
>+      program_name +
>+      " [options]\n\n"
>       "  --runtime_ms=N         runtime in ms; default is 10000 ms\n"
>       "  --lossrate=N           drop every N packets; default is 10\n"
>       "  --drift=F              clockdrift factor between 0.0 and 1.0; "
>       "default is 0.1\n";
>   webrtc::test::SetExecutablePath(argv[0]);
>-  if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) ||
>-      FLAG_help || argc != 1) {
>+  if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) || FLAG_help ||
>+      argc != 1) {
>     printf("%s", usage.c_str());
>     if (FLAG_help) {
>       rtc::FlagList::Print(nullptr, false);
>@@ -47,9 +47,8 @@ int main(int argc, char* argv[]) {
>   RTC_CHECK_GE(FLAG_lossrate, 0);
>   RTC_CHECK(FLAG_drift >= 0.0 && FLAG_drift < 1.0);
> 
>-  int64_t result =
>-      webrtc::test::NetEqPerformanceTest::Run(FLAG_runtime_ms, FLAG_lossrate,
>-                                              FLAG_drift);
>+  int64_t result = webrtc::test::NetEqPerformanceTest::Run(
>+      FLAG_runtime_ms, FLAG_lossrate, FLAG_drift);
>   if (result <= 0) {
>     std::cout << "There was an error" << std::endl;
>     return -1;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tick_timer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tick_timer.h
>index 4a9ade5cb34..520099e3466 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tick_timer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tick_timer.h
>@@ -15,7 +15,6 @@
> 
> #include "rtc_base/checks.h"
> #include "rtc_base/constructormagic.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/time_stretch.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/time_stretch.cc
>index 8a1bfa28351..560d9be56d6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/time_stretch.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/time_stretch.cc
>@@ -80,7 +80,7 @@ TimeStretch::ReturnCodes TimeStretch::Process(const int16_t* input,
>   // Calculate scaling to ensure that |peak_index| samples can be square-summed
>   // without overflowing.
>   int scaling = 31 - WebRtcSpl_NormW32(max_input_value_ * max_input_value_) -
>-      WebRtcSpl_NormW32(static_cast<int32_t>(peak_index));
>+                WebRtcSpl_NormW32(static_cast<int32_t>(peak_index));
>   scaling = std::max(0, scaling);
> 
>   // |vec1| starts at 15 ms minus one pitch period.
>@@ -99,8 +99,8 @@ TimeStretch::ReturnCodes TimeStretch::Process(const int16_t* input,
>       WebRtcSpl_DotProductWithScale(vec1, vec2, peak_index, scaling);
> 
>   // Check if the signal seems to be active speech or not (simple VAD).
>-  bool active_speech = SpeechDetection(vec1_energy, vec2_energy, peak_index,
>-                                       scaling);
>+  bool active_speech =
>+      SpeechDetection(vec1_energy, vec2_energy, peak_index, scaling);
> 
>   int16_t best_correlation;
>   if (!active_speech) {
>@@ -126,8 +126,8 @@ TimeStretch::ReturnCodes TimeStretch::Process(const int16_t* input,
>         static_cast<int16_t>(vec2_energy >> energy2_scale);
> 
>     // Calculate square-root of energy product.
>-    int16_t sqrt_energy_prod = WebRtcSpl_SqrtFloor(vec1_energy_int16 *
>-                                                   vec2_energy_int16);
>+    int16_t sqrt_energy_prod =
>+        WebRtcSpl_SqrtFloor(vec1_energy_int16 * vec2_energy_int16);
> 
>     // Calculate cross_corr / sqrt(en1*en2) in Q14.
>     int temp_scale = 14 - (energy1_scale + energy2_scale) / 2;
>@@ -138,7 +138,6 @@ TimeStretch::ReturnCodes TimeStretch::Process(const int16_t* input,
>     best_correlation = std::min(static_cast<int16_t>(16384), best_correlation);
>   }
> 
>-
>   // Check accelerate criteria and stretch the signal.
>   ReturnCodes return_value =
>       CheckCriteriaAndStretch(input, input_len, peak_index, best_correlation,
>@@ -172,8 +171,10 @@ void TimeStretch::AutoCorrelation() {
>                                    auto_corr, scaling);
> }
> 
>-bool TimeStretch::SpeechDetection(int32_t vec1_energy, int32_t vec2_energy,
>-                                  size_t peak_index, int scaling) const {
>+bool TimeStretch::SpeechDetection(int32_t vec1_energy,
>+                                  int32_t vec2_energy,
>+                                  size_t peak_index,
>+                                  int scaling) const {
>   // Check if the signal seems to be active speech or not (simple VAD).
>   // If (vec1_energy + vec2_energy) / (2 * peak_index) <=
>   // 8 * background_noise_energy, then we say that the signal contains no
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/time_stretch.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/time_stretch.h
>index ace10cd0553..13ad2c8554d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/time_stretch.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/time_stretch.h
>@@ -16,7 +16,6 @@
> 
> #include "modules/audio_coding/neteq/audio_multi_vector.h"
> #include "rtc_base/constructormagic.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -35,7 +34,8 @@ class TimeStretch {
>     kError = -1
>   };
> 
>-  TimeStretch(int sample_rate_hz, size_t num_channels,
>+  TimeStretch(int sample_rate_hz,
>+              size_t num_channels,
>               const BackgroundNoise& background_noise)
>       : sample_rate_hz_(sample_rate_hz),
>         fs_mult_(sample_rate_hz / 8000),
>@@ -43,10 +43,8 @@ class TimeStretch {
>         master_channel_(0),  // First channel is master.
>         background_noise_(background_noise),
>         max_input_value_(0) {
>-    assert(sample_rate_hz_ == 8000 ||
>-           sample_rate_hz_ == 16000 ||
>-           sample_rate_hz_ == 32000 ||
>-           sample_rate_hz_ == 48000);
>+    assert(sample_rate_hz_ == 8000 || sample_rate_hz_ == 16000 ||
>+           sample_rate_hz_ == 32000 || sample_rate_hz_ == 48000);
>     assert(num_channels_ > 0);
>     assert(master_channel_ < num_channels_);
>     memset(auto_correlation_, 0, sizeof(auto_correlation_));
>@@ -106,8 +104,10 @@ class TimeStretch {
>   void AutoCorrelation();
> 
>   // Performs a simple voice-activity detection based on the input parameters.
>-  bool SpeechDetection(int32_t vec1_energy, int32_t vec2_energy,
>-                       size_t peak_index, int scaling) const;
>+  bool SpeechDetection(int32_t vec1_energy,
>+                       int32_t vec2_energy,
>+                       size_t peak_index,
>+                       int scaling) const;
> 
>   RTC_DISALLOW_COPY_AND_ASSIGN(TimeStretch);
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/time_stretch_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/time_stretch_unittest.cc
>index 8d0f4d4a642..c96c7d4ec2a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/time_stretch_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/time_stretch_unittest.cc
>@@ -34,8 +34,8 @@ TEST(TimeStretch, CreateAndDestroy) {
>   const int kOverlapSamples = 5 * kSampleRate / 8000;
>   BackgroundNoise bgn(kNumChannels);
>   Accelerate accelerate(kSampleRate, kNumChannels, bgn);
>-  PreemptiveExpand preemptive_expand(
>-      kSampleRate, kNumChannels, bgn, kOverlapSamples);
>+  PreemptiveExpand preemptive_expand(kSampleRate, kNumChannels, bgn,
>+                                     kOverlapSamples);
> }
> 
> TEST(TimeStretch, CreateUsingFactory) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/timestamp_scaler.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/timestamp_scaler.cc
>index d7aa9fe8683..07d945edce0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/timestamp_scaler.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/timestamp_scaler.cc
>@@ -70,7 +70,6 @@ uint32_t TimestampScaler::ToInternal(uint32_t external_timestamp,
>   }
> }
> 
>-
> uint32_t TimestampScaler::ToExternal(uint32_t internal_timestamp) const {
>   if (!first_packet_received_ || (numerator_ == denominator_)) {
>     // Not initialized, or scale factor is 1.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/timestamp_scaler.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/timestamp_scaler.h
>index f0b05d634c6..62f535c464c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/timestamp_scaler.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/timestamp_scaler.h
>@@ -13,7 +13,6 @@
> 
> #include "modules/audio_coding/neteq/packet.h"
> #include "rtc_base/constructormagic.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/timestamp_scaler_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/timestamp_scaler_unittest.cc
>index b3c1bb0ee00..1f1445af051 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/timestamp_scaler_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/timestamp_scaler_unittest.cc
>@@ -8,10 +8,10 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>+#include "modules/audio_coding/neteq/timestamp_scaler.h"
> #include "api/audio_codecs/builtin_audio_decoder_factory.h"
> #include "modules/audio_coding/neteq/mock/mock_decoder_database.h"
> #include "modules/audio_coding/neteq/packet.h"
>-#include "modules/audio_coding/neteq/timestamp_scaler.h"
> #include "test/gmock.h"
> #include "test/gtest.h"
> 
>@@ -25,7 +25,8 @@ TEST(TimestampScaler, TestNoScaling) {
>   MockDecoderDatabase db;
>   auto factory = CreateBuiltinAudioDecoderFactory();
>   // Use PCMu, because it doesn't use scaled timestamps.
>-  const DecoderDatabase::DecoderInfo info(NetEqDecoder::kDecoderPCMu, factory);
>+  const DecoderDatabase::DecoderInfo info(NetEqDecoder::kDecoderPCMu,
>+                                          absl::nullopt, factory);
>   static const uint8_t kRtpPayloadType = 0;
>   EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
>       .WillRepeatedly(Return(&info));
>@@ -46,7 +47,8 @@ TEST(TimestampScaler, TestNoScalingLargeStep) {
>   MockDecoderDatabase db;
>   auto factory = CreateBuiltinAudioDecoderFactory();
>   // Use PCMu, because it doesn't use scaled timestamps.
>-  const DecoderDatabase::DecoderInfo info(NetEqDecoder::kDecoderPCMu, factory);
>+  const DecoderDatabase::DecoderInfo info(NetEqDecoder::kDecoderPCMu,
>+                                          absl::nullopt, factory);
>   static const uint8_t kRtpPayloadType = 0;
>   EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
>       .WillRepeatedly(Return(&info));
>@@ -58,7 +60,7 @@ TEST(TimestampScaler, TestNoScalingLargeStep) {
>   // |external_timestamp| will be a large positive value.
>   start_timestamp = start_timestamp - 5 * kStep;
>   for (uint32_t timestamp = start_timestamp; timestamp != 5 * kStep;
>-      timestamp += kStep) {
>+       timestamp += kStep) {
>     // Scale to internal timestamp.
>     EXPECT_EQ(timestamp, scaler.ToInternal(timestamp, kRtpPayloadType));
>     // Scale back.
>@@ -72,7 +74,8 @@ TEST(TimestampScaler, TestG722) {
>   MockDecoderDatabase db;
>   auto factory = CreateBuiltinAudioDecoderFactory();
>   // Use G722, which has a factor 2 scaling.
>-  const DecoderDatabase::DecoderInfo info(NetEqDecoder::kDecoderG722, factory);
>+  const DecoderDatabase::DecoderInfo info(NetEqDecoder::kDecoderG722,
>+                                          absl::nullopt, factory);
>   static const uint8_t kRtpPayloadType = 17;
>   EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
>       .WillRepeatedly(Return(&info));
>@@ -97,7 +100,8 @@ TEST(TimestampScaler, TestG722LargeStep) {
>   MockDecoderDatabase db;
>   auto factory = CreateBuiltinAudioDecoderFactory();
>   // Use G722, which has a factor 2 scaling.
>-  const DecoderDatabase::DecoderInfo info(NetEqDecoder::kDecoderG722, factory);
>+  const DecoderDatabase::DecoderInfo info(NetEqDecoder::kDecoderG722,
>+                                          absl::nullopt, factory);
>   static const uint8_t kRtpPayloadType = 17;
>   EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
>       .WillRepeatedly(Return(&info));
>@@ -127,9 +131,9 @@ TEST(TimestampScaler, TestG722WithCng) {
>   auto factory = CreateBuiltinAudioDecoderFactory();
>   // Use G722, which has a factor 2 scaling.
>   const DecoderDatabase::DecoderInfo info_g722(NetEqDecoder::kDecoderG722,
>-                                               factory);
>+                                               absl::nullopt, factory);
>   const DecoderDatabase::DecoderInfo info_cng(NetEqDecoder::kDecoderCNGwb,
>-                                              factory);
>+                                              absl::nullopt, factory);
>   static const uint8_t kRtpPayloadTypeG722 = 17;
>   static const uint8_t kRtpPayloadTypeCng = 13;
>   EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadTypeG722))
>@@ -170,7 +174,8 @@ TEST(TimestampScaler, TestG722Packet) {
>   MockDecoderDatabase db;
>   auto factory = CreateBuiltinAudioDecoderFactory();
>   // Use G722, which has a factor 2 scaling.
>-  const DecoderDatabase::DecoderInfo info(NetEqDecoder::kDecoderG722, factory);
>+  const DecoderDatabase::DecoderInfo info(NetEqDecoder::kDecoderG722,
>+                                          absl::nullopt, factory);
>   static const uint8_t kRtpPayloadType = 17;
>   EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
>       .WillRepeatedly(Return(&info));
>@@ -199,7 +204,8 @@ TEST(TimestampScaler, TestG722PacketList) {
>   MockDecoderDatabase db;
>   auto factory = CreateBuiltinAudioDecoderFactory();
>   // Use G722, which has a factor 2 scaling.
>-  const DecoderDatabase::DecoderInfo info(NetEqDecoder::kDecoderG722, factory);
>+  const DecoderDatabase::DecoderInfo info(NetEqDecoder::kDecoderG722,
>+                                          absl::nullopt, factory);
>   static const uint8_t kRtpPayloadType = 17;
>   EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
>       .WillRepeatedly(Return(&info));
>@@ -232,7 +238,8 @@ TEST(TimestampScaler, TestG722Reset) {
>   MockDecoderDatabase db;
>   auto factory = CreateBuiltinAudioDecoderFactory();
>   // Use G722, which has a factor 2 scaling.
>-  const DecoderDatabase::DecoderInfo info(NetEqDecoder::kDecoderG722, factory);
>+  const DecoderDatabase::DecoderInfo info(NetEqDecoder::kDecoderG722,
>+                                          absl::nullopt, factory);
>   static const uint8_t kRtpPayloadType = 17;
>   EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
>       .WillRepeatedly(Return(&info));
>@@ -272,7 +279,8 @@ TEST(TimestampScaler, TestG722Reset) {
> TEST(TimestampScaler, TestOpusLargeStep) {
>   MockDecoderDatabase db;
>   auto factory = CreateBuiltinAudioDecoderFactory();
>-  const DecoderDatabase::DecoderInfo info(NetEqDecoder::kDecoderOpus, factory);
>+  const DecoderDatabase::DecoderInfo info(NetEqDecoder::kDecoderOpus,
>+                                          absl::nullopt, factory);
>   static const uint8_t kRtpPayloadType = 17;
>   EXPECT_CALL(db, GetDecoderInfo(kRtpPayloadType))
>       .WillRepeatedly(Return(&info));
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/audio_checksum.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/audio_checksum.h
>index db67edf4e00..ee8c5823806 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/audio_checksum.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/audio_checksum.h
>@@ -19,7 +19,7 @@
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/messagedigest.h"
> #include "rtc_base/stringencode.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include "rtc_base/system/arch.h"
> 
> namespace webrtc {
> namespace test {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/audio_loop.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/audio_loop.cc
>index b5ad8817fe3..972921b51ab 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/audio_loop.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/audio_loop.cc
>@@ -21,16 +21,18 @@ bool AudioLoop::Init(const std::string file_name,
>                      size_t max_loop_length_samples,
>                      size_t block_length_samples) {
>   FILE* fp = fopen(file_name.c_str(), "rb");
>-  if (!fp) return false;
>+  if (!fp)
>+    return false;
> 
>-  audio_array_.reset(new int16_t[max_loop_length_samples +
>-                                 block_length_samples]);
>-  size_t samples_read = fread(audio_array_.get(), sizeof(int16_t),
>-                              max_loop_length_samples, fp);
>+  audio_array_.reset(
>+      new int16_t[max_loop_length_samples + block_length_samples]);
>+  size_t samples_read =
>+      fread(audio_array_.get(), sizeof(int16_t), max_loop_length_samples, fp);
>   fclose(fp);
> 
>   // Block length must be shorter than the loop length.
>-  if (block_length_samples > samples_read) return false;
>+  if (block_length_samples > samples_read)
>+    return false;
> 
>   // Add an extra block length of samples to the end of the array, starting
>   // over again from the beginning of the array. This is done to simplify
>@@ -54,6 +56,5 @@ rtc::ArrayView<const int16_t> AudioLoop::GetNextBlock() {
>   return rtc::ArrayView<const int16_t>(output_ptr, block_length_samples_);
> }
> 
>-
> }  // namespace test
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/audio_loop.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/audio_loop.h
>index abb1a36378a..c7788a693ab 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/audio_loop.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/audio_loop.h
>@@ -16,7 +16,6 @@
> 
> #include "api/array_view.h"
> #include "rtc_base/constructormagic.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> namespace test {
>@@ -26,10 +25,7 @@ namespace test {
> class AudioLoop {
>  public:
>   AudioLoop()
>-      : next_index_(0),
>-        loop_length_samples_(0),
>-        block_length_samples_(0) {
>-  }
>+      : next_index_(0), loop_length_samples_(0), block_length_samples_(0) {}
> 
>   virtual ~AudioLoop() {}
> 
>@@ -38,7 +34,8 @@ class AudioLoop {
>   // greater. Otherwise, the loop length is the same as the file length.
>   // The audio will be delivered in blocks of |block_length_samples|.
>   // Returns false if the initialization failed, otherwise true.
>-  bool Init(const std::string file_name, size_t max_loop_length_samples,
>+  bool Init(const std::string file_name,
>+            size_t max_loop_length_samples,
>             size_t block_length_samples);
> 
>   // Returns a (pointer,size) pair for the next block of audio. The size is
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/audio_sink.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/audio_sink.h
>index ecec51b895c..be2a315350a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/audio_sink.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/audio_sink.h
>@@ -11,9 +11,8 @@
> #ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_SINK_H_
> #define MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_SINK_H_
> 
>-#include "modules/include/module_common_types.h"
>+#include "api/audio/audio_frame.h"
> #include "rtc_base/constructormagic.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> namespace test {
>@@ -32,9 +31,8 @@ class AudioSink {
>   // Writes |audio_frame| to the AudioSink. Returns true if successful,
>   // otherwise false.
>   bool WriteAudioFrame(const AudioFrame& audio_frame) {
>-    return WriteArray(
>-        audio_frame.data(),
>-        audio_frame.samples_per_channel_ * audio_frame.num_channels_);
>+    return WriteArray(audio_frame.data(), audio_frame.samples_per_channel_ *
>+                                              audio_frame.num_channels_);
>   }
> 
>  private:
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/encode_neteq_input.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/encode_neteq_input.cc
>index a8d1bdfe0ad..c5766706581 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/encode_neteq_input.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/encode_neteq_input.cc
>@@ -27,12 +27,14 @@ EncodeNetEqInput::EncodeNetEqInput(std::unique_ptr<Generator> generator,
>   CreatePacket();
> }
> 
>-rtc::Optional<int64_t> EncodeNetEqInput::NextPacketTime() const {
>+EncodeNetEqInput::~EncodeNetEqInput() = default;
>+
>+absl::optional<int64_t> EncodeNetEqInput::NextPacketTime() const {
>   RTC_DCHECK(packet_data_);
>   return static_cast<int64_t>(packet_data_->time_ms);
> }
> 
>-rtc::Optional<int64_t> EncodeNetEqInput::NextOutputEventTime() const {
>+absl::optional<int64_t> EncodeNetEqInput::NextOutputEventTime() const {
>   return next_output_event_ms_;
> }
> 
>@@ -50,7 +52,11 @@ void EncodeNetEqInput::AdvanceOutputEvent() {
>   next_output_event_ms_ += kOutputPeriodMs;
> }
> 
>-rtc::Optional<RTPHeader> EncodeNetEqInput::NextHeader() const {
>+bool EncodeNetEqInput::ended() const {
>+  return next_output_event_ms_ <= input_duration_ms_;
>+}
>+
>+absl::optional<RTPHeader> EncodeNetEqInput::NextHeader() const {
>   RTC_DCHECK(packet_data_);
>   return packet_data_->header;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/encode_neteq_input.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/encode_neteq_input.h
>index b44d4ac08c0..caa9ac76f41 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/encode_neteq_input.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/encode_neteq_input.h
>@@ -15,7 +15,6 @@
> 
> #include "api/audio_codecs/audio_encoder.h"
> #include "modules/audio_coding/neteq/tools/neteq_input.h"
>-#include "modules/include/module_common_types.h"
> 
> namespace webrtc {
> namespace test {
>@@ -36,20 +35,19 @@ class EncodeNetEqInput : public NetEqInput {
>   EncodeNetEqInput(std::unique_ptr<Generator> generator,
>                    std::unique_ptr<AudioEncoder> encoder,
>                    int64_t input_duration_ms);
>+  ~EncodeNetEqInput() override;
> 
>-  rtc::Optional<int64_t> NextPacketTime() const override;
>+  absl::optional<int64_t> NextPacketTime() const override;
> 
>-  rtc::Optional<int64_t> NextOutputEventTime() const override;
>+  absl::optional<int64_t> NextOutputEventTime() const override;
> 
>   std::unique_ptr<PacketData> PopPacket() override;
> 
>   void AdvanceOutputEvent() override;
> 
>-  bool ended() const override {
>-    return next_output_event_ms_ <= input_duration_ms_;
>-  }
>+  bool ended() const override;
> 
>-  rtc::Optional<RTPHeader> NextHeader() const override;
>+  absl::optional<RTPHeader> NextHeader() const override;
> 
>  private:
>   static constexpr int64_t kOutputPeriodMs = 10;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.cc
>index f4376088b89..a51dac3fe0d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.cc
>@@ -22,34 +22,24 @@ int FakeDecodeFromFile::DecodeInternal(const uint8_t* encoded,
>                                        int sample_rate_hz,
>                                        int16_t* decoded,
>                                        SpeechType* speech_type) {
>+  RTC_DCHECK_EQ(sample_rate_hz, SampleRateHz());
>+
>+  const int samples_to_decode = PacketDuration(encoded, encoded_len);
>+  const int total_samples_to_decode = samples_to_decode * (stereo_ ? 2 : 1);
>+
>   if (encoded_len == 0) {
>     // Decoder is asked to produce codec-internal comfort noise.
>     RTC_DCHECK(!encoded);  // NetEq always sends nullptr in this case.
>     RTC_DCHECK(cng_mode_);
>-    RTC_DCHECK_GT(last_decoded_length_, 0);
>-    std::fill_n(decoded, last_decoded_length_, 0);
>+    RTC_DCHECK_GT(total_samples_to_decode, 0);
>+    std::fill_n(decoded, total_samples_to_decode, 0);
>     *speech_type = kComfortNoise;
>-    return rtc::dchecked_cast<int>(last_decoded_length_);
>+    return rtc::dchecked_cast<int>(total_samples_to_decode);
>   }
> 
>   RTC_CHECK_GE(encoded_len, 12);
>   uint32_t timestamp_to_decode =
>       ByteReader<uint32_t>::ReadLittleEndian(encoded);
>-  uint32_t samples_to_decode =
>-      ByteReader<uint32_t>::ReadLittleEndian(&encoded[4]);
>-  if (samples_to_decode == 0) {
>-    // Number of samples in packet is unknown.
>-    if (last_decoded_length_ > 0) {
>-      // Use length of last decoded packet, but since this is the total for all
>-      // channels, we have to divide by 2 in the stereo case.
>-      samples_to_decode = rtc::dchecked_cast<int>(rtc::CheckedDivExact(
>-          last_decoded_length_, static_cast<size_t>(stereo_ ? 2uL : 1uL)));
>-    } else {
>-      // This is the first packet to decode, and we do not know the length of
>-      // it. Set it to 10 ms.
>-      samples_to_decode = rtc::CheckedDivExact(sample_rate_hz, 100);
>-    }
>-  }
> 
>   if (next_timestamp_from_input_ &&
>       timestamp_to_decode != *next_timestamp_from_input_) {
>@@ -65,11 +55,11 @@ int FakeDecodeFromFile::DecodeInternal(const uint8_t* encoded,
>       ByteReader<uint32_t>::ReadLittleEndian(&encoded[8]);
>   if (original_payload_size_bytes == 1) {
>     // This is a comfort noise payload.
>-    RTC_DCHECK_GT(last_decoded_length_, 0);
>-    std::fill_n(decoded, last_decoded_length_, 0);
>+    RTC_DCHECK_GT(total_samples_to_decode, 0);
>+    std::fill_n(decoded, total_samples_to_decode, 0);
>     *speech_type = kComfortNoise;
>     cng_mode_ = true;
>-    return rtc::dchecked_cast<int>(last_decoded_length_);
>+    return rtc::dchecked_cast<int>(total_samples_to_decode);
>   }
> 
>   cng_mode_ = false;
>@@ -78,12 +68,39 @@ int FakeDecodeFromFile::DecodeInternal(const uint8_t* encoded,
>   if (stereo_) {
>     InputAudioFile::DuplicateInterleaved(decoded, samples_to_decode, 2,
>                                          decoded);
>-    samples_to_decode *= 2;
>   }
> 
>   *speech_type = kSpeech;
>   last_decoded_length_ = samples_to_decode;
>-  return rtc::dchecked_cast<int>(last_decoded_length_);
>+  return rtc::dchecked_cast<int>(total_samples_to_decode);
>+}
>+
>+int FakeDecodeFromFile::PacketDuration(const uint8_t* encoded,
>+                                       size_t encoded_len) const {
>+  const uint32_t original_payload_size_bytes =
>+      encoded_len < 8 + sizeof(uint32_t)
>+          ? 0
>+          : ByteReader<uint32_t>::ReadLittleEndian(&encoded[8]);
>+  const uint32_t samples_to_decode =
>+      encoded_len < 4 + sizeof(uint32_t)
>+          ? 0
>+          : ByteReader<uint32_t>::ReadLittleEndian(&encoded[4]);
>+  if (  // Decoder is asked to produce codec-internal comfort noise
>+      encoded_len == 0 ||
>+      // Comfort noise payload
>+      original_payload_size_bytes == 1 || samples_to_decode == 0 ||
>+      // Erroneous duration since it is not a multiple of 10ms
>+      samples_to_decode % rtc::CheckedDivExact(SampleRateHz(), 100) != 0) {
>+    if (last_decoded_length_ > 0) {
>+      // Use length of last decoded packet.
>+      return rtc::dchecked_cast<int>(last_decoded_length_);
>+    } else {
>+      // This is the first packet to decode, and we do not know the length of
>+      // it. Set it to 10 ms.
>+      return rtc::CheckedDivExact(SampleRateHz(), 100);
>+    }
>+  }
>+  return samples_to_decode;
> }
> 
> void FakeDecodeFromFile::PrepareEncoded(uint32_t timestamp,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.h
>index 7aa8e6e40f6..c266d38daa8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.h
>@@ -13,9 +13,9 @@
> 
> #include <memory>
> 
>+#include "absl/types/optional.h"
> #include "api/array_view.h"
> #include "api/audio_codecs/audio_decoder.h"
>-#include "api/optional.h"
> #include "modules/audio_coding/neteq/tools/input_audio_file.h"
> 
> namespace webrtc {
>@@ -50,6 +50,8 @@ class FakeDecodeFromFile : public AudioDecoder {
>                      int16_t* decoded,
>                      SpeechType* speech_type) override;
> 
>+  int PacketDuration(const uint8_t* encoded, size_t encoded_len) const override;
>+
>   // Helper method. Writes |timestamp|, |samples| and
>   // |original_payload_size_bytes| to |encoded| in a format that the
>   // FakeDecodeFromFile decoder will understand. |encoded| must be at least 12
>@@ -61,7 +63,7 @@ class FakeDecodeFromFile : public AudioDecoder {
> 
>  private:
>   std::unique_ptr<InputAudioFile> input_;
>-  rtc::Optional<uint32_t> next_timestamp_from_input_;
>+  absl::optional<uint32_t> next_timestamp_from_input_;
>   const int sample_rate_hz_;
>   const bool stereo_;
>   size_t last_decoded_length_ = 0;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/input_audio_file.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/input_audio_file.cc
>index 31ebf983424..6d11064724c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/input_audio_file.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/input_audio_file.cc
>@@ -20,7 +20,9 @@ InputAudioFile::InputAudioFile(const std::string file_name, bool loop_at_end)
>   fp_ = fopen(file_name.c_str(), "rb");
> }
> 
>-InputAudioFile::~InputAudioFile() { fclose(fp_); }
>+InputAudioFile::~InputAudioFile() {
>+  fclose(fp_);
>+}
> 
> bool InputAudioFile::Read(size_t samples, int16_t* destination) {
>   if (!fp_) {
>@@ -56,19 +58,25 @@ bool InputAudioFile::Seek(int samples) {
>   RTC_CHECK_NE(EOF, file_size) << "Error returned when getting file position.";
>   // Find new position.
>   long new_pos = current_pos + sizeof(int16_t) * samples;  // Samples to bytes.
>-  RTC_CHECK_GE(new_pos, 0)
>-      << "Trying to move to before the beginning of the file";
>   if (loop_at_end_) {
>     new_pos = new_pos % file_size;  // Wrap around the end of the file.
>+    if (new_pos < 0) {
>+      // For negative values of new_pos, newpos % file_size will also be
>+      // negative. To get the correct result it's needed to add file_size.
>+      new_pos += file_size;
>+    }
>   } else {
>     new_pos = new_pos > file_size ? file_size : new_pos;  // Don't loop.
>   }
>+  RTC_CHECK_GE(new_pos, 0)
>+      << "Trying to move to before the beginning of the file";
>   // Move to new position relative to the beginning of the file.
>   RTC_CHECK_EQ(0, fseek(fp_, new_pos, SEEK_SET));
>   return true;
> }
> 
>-void InputAudioFile::DuplicateInterleaved(const int16_t* source, size_t samples,
>+void InputAudioFile::DuplicateInterleaved(const int16_t* source,
>+                                          size_t samples,
>                                           size_t channels,
>                                           int16_t* destination) {
>   // Start from the end of |source| and |destination|, and work towards the
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/input_audio_file.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/input_audio_file.h
>index 6bfa369b38a..b36dc24e1af 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/input_audio_file.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/input_audio_file.h
>@@ -16,7 +16,6 @@
> #include <string>
> 
> #include "rtc_base/constructormagic.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> namespace test {
>@@ -45,8 +44,10 @@ class InputAudioFile {
>   // channels are identical. The output |destination| must have the capacity to
>   // hold samples * channels elements. Note that |source| and |destination| can
>   // be the same array (i.e., point to the same address).
>-  static void DuplicateInterleaved(const int16_t* source, size_t samples,
>-                                   size_t channels, int16_t* destination);
>+  static void DuplicateInterleaved(const int16_t* source,
>+                                   size_t samples,
>+                                   size_t channels,
>+                                   int16_t* destination);
> 
>  private:
>   FILE* fp_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_delay_analyzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_delay_analyzer.cc
>index 882f82321ce..e5bd7652eff 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_delay_analyzer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_delay_analyzer.cc
>@@ -17,11 +17,19 @@
> #include <limits>
> #include <utility>
> 
>+#include "modules/include/module_common_types.h"
> #include "rtc_base/checks.h"
> 
> namespace webrtc {
> namespace test {
> namespace {
>+std::string kArrivalDelayX = "arrival_delay_x";
>+std::string kArrivalDelayY = "arrival_delay_y";
>+std::string kTargetDelayX = "target_delay_x";
>+std::string kTargetDelayY = "target_delay_y";
>+std::string kPlayoutDelayX = "playout_delay_x";
>+std::string kPlayoutDelayY = "playout_delay_y";
>+
> // Helper function for NetEqDelayAnalyzer::CreateGraphs. Returns the
> // interpolated value of a function at the point x. Vector x_vec contains the
> // sample points, and y_vec contains the function values at these points. The
>@@ -53,6 +61,26 @@ double LinearInterpolate(double x,
>   }
>   return y;
> }
>+
>+void PrintDelays(const NetEqDelayAnalyzer::Delays& delays,
>+                 int64_t ref_time_ms,
>+                 const std::string& var_name_x,
>+                 const std::string& var_name_y,
>+                 std::ofstream& output,
>+                 const std::string& terminator = "") {
>+  output << var_name_x << " = [ ";
>+  for (const std::pair<int64_t, float>& delay : delays) {
>+    output << (delay.first - ref_time_ms) / 1000.f << ", ";
>+  }
>+  output << "]" << terminator << std::endl;
>+
>+  output << var_name_y << " = [ ";
>+  for (const std::pair<int64_t, float>& delay : delays) {
>+    output << delay.second << ", ";
>+  }
>+  output << "]" << terminator << std::endl;
>+}
>+
> }  // namespace
> 
> void NetEqDelayAnalyzer::AfterInsertPacket(
>@@ -96,12 +124,10 @@ void NetEqDelayAnalyzer::AfterGetAudio(int64_t time_now_ms,
>   ++get_audio_count_;
> }
> 
>-void NetEqDelayAnalyzer::CreateGraphs(
>-    std::vector<float>* send_time_s,
>-    std::vector<float>* arrival_delay_ms,
>-    std::vector<float>* corrected_arrival_delay_ms,
>-    std::vector<rtc::Optional<float>>* playout_delay_ms,
>-    std::vector<rtc::Optional<float>>* target_delay_ms) const {
>+void NetEqDelayAnalyzer::CreateGraphs(Delays* arrival_delay_ms,
>+                                      Delays* corrected_arrival_delay_ms,
>+                                      Delays* playout_delay_ms,
>+                                      Delays* target_delay_ms) const {
>   if (get_audio_time_ms_.empty()) {
>     return;
>   }
>@@ -122,111 +148,76 @@ void NetEqDelayAnalyzer::CreateGraphs(
>   // calculates the base offset.
>   for (auto& d : data_) {
>     rtp_timestamps_ms.push_back(
>-        unwrapper.Unwrap(d.first) /
>+        static_cast<double>(unwrapper.Unwrap(d.first)) /
>         rtc::CheckedDivExact(last_sample_rate_hz_, 1000));
>     offset =
>         std::min(offset, d.second.arrival_time_ms - rtp_timestamps_ms.back());
>   }
> 
>-  // Calculate send times in seconds for each packet. This is the (unwrapped)
>-  // RTP timestamp in ms divided by 1000.
>-  send_time_s->resize(rtp_timestamps_ms.size());
>-  std::transform(rtp_timestamps_ms.begin(), rtp_timestamps_ms.end(),
>-                 send_time_s->begin(), [rtp_timestamps_ms](double x) {
>-                   return (x - rtp_timestamps_ms[0]) / 1000.f;
>-                 });
>-  RTC_DCHECK_EQ(send_time_s->size(), rtp_timestamps_ms.size());
>-
>   // This loop traverses the data again and populates the graph vectors. The
>   // reason to have two loops and traverse twice is that the offset cannot be
>   // known until the first traversal is done. Meanwhile, the final offset must
>   // be known already at the start of this second loop.
>-  auto data_it = data_.cbegin();
>-  for (size_t i = 0; i < send_time_s->size(); ++i, ++data_it) {
>-    RTC_DCHECK(data_it != data_.end());
>-    const double offset_send_time_ms = rtp_timestamps_ms[i] + offset;
>-    const auto& timing = data_it->second;
>-    corrected_arrival_delay_ms->push_back(
>+  size_t i = 0;
>+  for (const auto& data : data_) {
>+    const double offset_send_time_ms = rtp_timestamps_ms[i++] + offset;
>+    const auto& timing = data.second;
>+    corrected_arrival_delay_ms->push_back(std::make_pair(
>+        timing.arrival_time_ms,
>         LinearInterpolate(timing.arrival_time_ms, get_audio_time_ms_,
>                           nominal_get_audio_time_ms) -
>-        offset_send_time_ms);
>-    arrival_delay_ms->push_back(timing.arrival_time_ms - offset_send_time_ms);
>+            offset_send_time_ms));
>+    arrival_delay_ms->push_back(std::make_pair(
>+        timing.arrival_time_ms, timing.arrival_time_ms - offset_send_time_ms));
> 
>     if (timing.decode_get_audio_count) {
>       // This packet was decoded.
>       RTC_DCHECK(timing.sync_delay_ms);
>-      const float playout_ms = *timing.decode_get_audio_count * 10 +
>-                               get_audio_time_ms_[0] + *timing.sync_delay_ms -
>-                               offset_send_time_ms;
>-      playout_delay_ms->push_back(playout_ms);
>+      const int64_t get_audio_time =
>+          *timing.decode_get_audio_count * 10 + get_audio_time_ms_[0];
>+      const float playout_ms =
>+          get_audio_time + *timing.sync_delay_ms - offset_send_time_ms;
>+      playout_delay_ms->push_back(std::make_pair(get_audio_time, playout_ms));
>       RTC_DCHECK(timing.target_delay_ms);
>       RTC_DCHECK(timing.current_delay_ms);
>       const float target =
>           playout_ms - *timing.current_delay_ms + *timing.target_delay_ms;
>-      target_delay_ms->push_back(target);
>-    } else {
>-      // This packet was never decoded. Mark target and playout delays as empty.
>-      playout_delay_ms->push_back(rtc::nullopt);
>-      target_delay_ms->push_back(rtc::nullopt);
>+      target_delay_ms->push_back(std::make_pair(get_audio_time, target));
>     }
>   }
>-  RTC_DCHECK(data_it == data_.end());
>-  RTC_DCHECK_EQ(send_time_s->size(), corrected_arrival_delay_ms->size());
>-  RTC_DCHECK_EQ(send_time_s->size(), playout_delay_ms->size());
>-  RTC_DCHECK_EQ(send_time_s->size(), target_delay_ms->size());
> }
> 
> void NetEqDelayAnalyzer::CreateMatlabScript(
>     const std::string& script_name) const {
>-  std::vector<float> send_time_s;
>-  std::vector<float> arrival_delay_ms;
>-  std::vector<float> corrected_arrival_delay_ms;
>-  std::vector<rtc::Optional<float>> playout_delay_ms;
>-  std::vector<rtc::Optional<float>> target_delay_ms;
>-  CreateGraphs(&send_time_s, &arrival_delay_ms, &corrected_arrival_delay_ms,
>+  Delays arrival_delay_ms;
>+  Delays corrected_arrival_delay_ms;
>+  Delays playout_delay_ms;
>+  Delays target_delay_ms;
>+  CreateGraphs(&arrival_delay_ms, &corrected_arrival_delay_ms,
>                &playout_delay_ms, &target_delay_ms);
> 
>+  // Maybe better to find the actually smallest timestamp, to surely avoid
>+  // x-axis starting from negative.
>+  const int64_t ref_time_ms = arrival_delay_ms.front().first;
>+
>   // Create an output file stream to Matlab script file.
>   std::ofstream output(script_name);
>-  // The iterator is used to batch-output comma-separated values from vectors.
>-  std::ostream_iterator<float> output_iterator(output, ",");
>-
>-  output << "send_time_s = [ ";
>-  std::copy(send_time_s.begin(), send_time_s.end(), output_iterator);
>-  output << "];" << std::endl;
>-
>-  output << "arrival_delay_ms = [ ";
>-  std::copy(arrival_delay_ms.begin(), arrival_delay_ms.end(), output_iterator);
>-  output << "];" << std::endl;
>-
>-  output << "corrected_arrival_delay_ms = [ ";
>-  std::copy(corrected_arrival_delay_ms.begin(),
>-            corrected_arrival_delay_ms.end(), output_iterator);
>-  output << "];" << std::endl;
>-
>-  output << "playout_delay_ms = [ ";
>-  for (const auto& v : playout_delay_ms) {
>-    if (!v) {
>-      output << "nan, ";
>-    } else {
>-      output << *v << ", ";
>-    }
>-  }
>-  output << "];" << std::endl;
>-
>-  output << "target_delay_ms = [ ";
>-  for (const auto& v : target_delay_ms) {
>-    if (!v) {
>-      output << "nan, ";
>-    } else {
>-      output << *v << ", ";
>-    }
>-  }
>-  output << "];" << std::endl;
> 
>-  output << "h=plot(send_time_s, arrival_delay_ms, "
>-         << "send_time_s, target_delay_ms, 'g.', "
>-         << "send_time_s, playout_delay_ms);" << std::endl;
>+  PrintDelays(corrected_arrival_delay_ms, ref_time_ms, kArrivalDelayX,
>+              kArrivalDelayY, output, ";");
>+
>+  // PrintDelays(corrected_arrival_delay_x, kCorrectedArrivalDelayX,
>+  // kCorrectedArrivalDelayY, output);
>+
>+  PrintDelays(playout_delay_ms, ref_time_ms, kPlayoutDelayX, kPlayoutDelayY,
>+              output, ";");
>+
>+  PrintDelays(target_delay_ms, ref_time_ms, kTargetDelayX, kTargetDelayY,
>+              output, ";");
>+
>+  output << "h=plot(" << kArrivalDelayX << ", " << kArrivalDelayY << ", "
>+         << kTargetDelayX << ", " << kTargetDelayY << ", 'g.', "
>+         << kPlayoutDelayX << ", " << kPlayoutDelayY << ");" << std::endl;
>   output << "set(h(1),'color',0.75*[1 1 1]);" << std::endl;
>   output << "set(h(2),'markersize',6);" << std::endl;
>   output << "set(h(3),'linew',1.5);" << std::endl;
>@@ -234,7 +225,7 @@ void NetEqDelayAnalyzer::CreateMatlabScript(
>   output << "axis tight" << std::endl;
>   output << "ax2=axis;" << std::endl;
>   output << "axis([ax2(1:3) ax1(4)])" << std::endl;
>-  output << "xlabel('send time [s]');" << std::endl;
>+  output << "xlabel('time [s]');" << std::endl;
>   output << "ylabel('relative delay [ms]');" << std::endl;
>   if (!ssrcs_.empty()) {
>     auto ssrc_it = ssrcs_.cbegin();
>@@ -254,65 +245,45 @@ void NetEqDelayAnalyzer::CreateMatlabScript(
> 
> void NetEqDelayAnalyzer::CreatePythonScript(
>     const std::string& script_name) const {
>-  std::vector<float> send_time_s;
>-  std::vector<float> arrival_delay_ms;
>-  std::vector<float> corrected_arrival_delay_ms;
>-  std::vector<rtc::Optional<float>> playout_delay_ms;
>-  std::vector<rtc::Optional<float>> target_delay_ms;
>-  CreateGraphs(&send_time_s, &arrival_delay_ms, &corrected_arrival_delay_ms,
>+  Delays arrival_delay_ms;
>+  Delays corrected_arrival_delay_ms;
>+  Delays playout_delay_ms;
>+  Delays target_delay_ms;
>+  CreateGraphs(&arrival_delay_ms, &corrected_arrival_delay_ms,
>                &playout_delay_ms, &target_delay_ms);
> 
>+  // Maybe better to find the actually smallest timestamp, to surely avoid
>+  // x-axis starting from negative.
>+  const int64_t ref_time_ms = arrival_delay_ms.front().first;
>+
>   // Create an output file stream to the python script file.
>   std::ofstream output(script_name);
>-  // The iterator is used to batch-output comma-separated values from vectors.
>-  std::ostream_iterator<float> output_iterator(output, ",");
> 
>   // Necessary includes
>   output << "import numpy as np" << std::endl;
>   output << "import matplotlib.pyplot as plt" << std::endl;
> 
>-  output << "send_time_s = [";
>-  std::copy(send_time_s.begin(), send_time_s.end(), output_iterator);
>-  output << "]" << std::endl;
>-
>-  output << "arrival_delay_ms = [";
>-  std::copy(arrival_delay_ms.begin(), arrival_delay_ms.end(), output_iterator);
>-  output << "]" << std::endl;
>-
>-  output << "corrected_arrival_delay_ms = [";
>-  std::copy(corrected_arrival_delay_ms.begin(),
>-            corrected_arrival_delay_ms.end(), output_iterator);
>-  output << "]" << std::endl;
>-
>-  output << "playout_delay_ms = [";
>-  for (const auto& v : playout_delay_ms) {
>-    if (!v) {
>-      output << "float('nan'), ";
>-    } else {
>-      output << *v << ", ";
>-    }
>-  }
>-  output << "]" << std::endl;
>-
>-  output << "target_delay_ms = [";
>-  for (const auto& v : target_delay_ms) {
>-    if (!v) {
>-      output << "float('nan'), ";
>-    } else {
>-      output << *v << ", ";
>-    }
>-  }
>-  output << "]" << std::endl;
>+  PrintDelays(corrected_arrival_delay_ms, ref_time_ms, kArrivalDelayX,
>+              kArrivalDelayY, output);
>+
>+  // PrintDelays(corrected_arrival_delay_x, kCorrectedArrivalDelayX,
>+  // kCorrectedArrivalDelayY, output);
>+
>+  PrintDelays(playout_delay_ms, ref_time_ms, kPlayoutDelayX, kPlayoutDelayY,
>+              output);
>+
>+  PrintDelays(target_delay_ms, ref_time_ms, kTargetDelayX, kTargetDelayY,
>+              output);
> 
>   output << "if __name__ == '__main__':" << std::endl;
>-  output << "  h=plt.plot(send_time_s, arrival_delay_ms, "
>-         << "send_time_s, target_delay_ms, 'g.', "
>-         << "send_time_s, playout_delay_ms)" << std::endl;
>+  output << "  h=plt.plot(" << kArrivalDelayX << ", " << kArrivalDelayY << ", "
>+         << kTargetDelayX << ", " << kTargetDelayY << ", 'g.', "
>+         << kPlayoutDelayX << ", " << kPlayoutDelayY << ")" << std::endl;
>   output << "  plt.setp(h[0],'color',[.75, .75, .75])" << std::endl;
>   output << "  plt.setp(h[1],'markersize',6)" << std::endl;
>   output << "  plt.setp(h[2],'linewidth',1.5)" << std::endl;
>   output << "  plt.axis('tight')" << std::endl;
>-  output << "  plt.xlabel('send time [s]')" << std::endl;
>+  output << "  plt.xlabel('time [s]')" << std::endl;
>   output << "  plt.ylabel('relative delay [ms]')" << std::endl;
>   if (!ssrcs_.empty()) {
>     auto ssrc_it = ssrcs_.cbegin();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_delay_analyzer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_delay_analyzer.h
>index e6d6913f13f..f6b895aae06 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_delay_analyzer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_delay_analyzer.h
>@@ -16,10 +16,9 @@
> #include <string>
> #include <vector>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> #include "modules/audio_coding/neteq/tools/neteq_input.h"
> #include "modules/audio_coding/neteq/tools/neteq_test.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> namespace test {
>@@ -37,11 +36,11 @@ class NetEqDelayAnalyzer : public test::NetEqPostInsertPacket,
>                      bool muted,
>                      NetEq* neteq) override;
> 
>-  void CreateGraphs(std::vector<float>* send_times_s,
>-                    std::vector<float>* arrival_delay_ms,
>-                    std::vector<float>* corrected_arrival_delay_ms,
>-                    std::vector<rtc::Optional<float>>* playout_delay_ms,
>-                    std::vector<rtc::Optional<float>>* target_delay_ms) const;
>+  using Delays = std::vector<std::pair<int64_t, float>>;
>+  void CreateGraphs(Delays* arrival_delay_ms,
>+                    Delays* corrected_arrival_delay_ms,
>+                    Delays* playout_delay_ms,
>+                    Delays* target_delay_ms) const;
> 
>   // Creates a matlab script with file name script_name. When executed in
>   // Matlab, the script will generate graphs with the same timing information
>@@ -55,12 +54,12 @@ class NetEqDelayAnalyzer : public test::NetEqPostInsertPacket,
> 
>  private:
>   struct TimingData {
>-    explicit TimingData(double at) : arrival_time_ms(at) {}
>-    double arrival_time_ms;
>-    rtc::Optional<int64_t> decode_get_audio_count;
>-    rtc::Optional<int64_t> sync_delay_ms;
>-    rtc::Optional<int> target_delay_ms;
>-    rtc::Optional<int> current_delay_ms;
>+    explicit TimingData(int64_t at) : arrival_time_ms(at) {}
>+    int64_t arrival_time_ms;
>+    absl::optional<int64_t> decode_get_audio_count;
>+    absl::optional<int64_t> sync_delay_ms;
>+    absl::optional<int> target_delay_ms;
>+    absl::optional<int> current_delay_ms;
>   };
>   std::map<uint32_t, TimingData> data_;
>   std::vector<int64_t> get_audio_time_ms_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_event_log_input.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_event_log_input.cc
>new file mode 100644
>index 00000000000..21c5f9ef022
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_event_log_input.cc
>@@ -0,0 +1,47 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_coding/neteq/tools/neteq_event_log_input.h"
>+
>+#include <limits>
>+
>+#include "modules/audio_coding/neteq/tools/rtc_event_log_source.h"
>+#include "rtc_base/checks.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+NetEqEventLogInput::NetEqEventLogInput(const std::string& file_name,
>+                                       const RtpHeaderExtensionMap& hdr_ext_map)
>+    : source_(RtcEventLogSource::Create(file_name)) {
>+  for (const auto& ext_pair : hdr_ext_map) {
>+    source_->RegisterRtpHeaderExtension(ext_pair.second, ext_pair.first);
>+  }
>+  LoadNextPacket();
>+  AdvanceOutputEvent();
>+}
>+
>+absl::optional<int64_t> NetEqEventLogInput::NextOutputEventTime() const {
>+  return next_output_event_ms_;
>+}
>+
>+void NetEqEventLogInput::AdvanceOutputEvent() {
>+  next_output_event_ms_ = source_->NextAudioOutputEventMs();
>+  if (*next_output_event_ms_ == std::numeric_limits<int64_t>::max()) {
>+    next_output_event_ms_ = absl::nullopt;
>+  }
>+}
>+
>+PacketSource* NetEqEventLogInput::source() {
>+  return source_.get();
>+}
>+
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_event_log_input.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_event_log_input.h
>new file mode 100644
>index 00000000000..86cf9f2f02a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_event_log_input.h
>@@ -0,0 +1,45 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_EVENT_LOG_INPUT_H_
>+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_EVENT_LOG_INPUT_H_
>+
>+#include <map>
>+#include <memory>
>+#include <string>
>+
>+#include "modules/audio_coding/neteq/tools/neteq_packet_source_input.h"
>+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+class RtcEventLogSource;
>+
>+// Implementation of NetEqPacketSourceInput to be used with an
>+// RtcEventLogSource.
>+class NetEqEventLogInput final : public NetEqPacketSourceInput {
>+ public:
>+  NetEqEventLogInput(const std::string& file_name,
>+                     const RtpHeaderExtensionMap& hdr_ext_map);
>+
>+  absl::optional<int64_t> NextOutputEventTime() const override;
>+  void AdvanceOutputEvent() override;
>+
>+ protected:
>+  PacketSource* source() override;
>+
>+ private:
>+  std::unique_ptr<RtcEventLogSource> source_;
>+};
>+
>+}  // namespace test
>+}  // namespace webrtc
>+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_EVENT_LOG_INPUT_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.cc
>index 68dde5296f2..3bd218b8b8e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.cc
>@@ -8,9 +8,9 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-
> #include "modules/audio_coding/neteq/tools/neteq_external_decoder_test.h"
> 
>+#include "api/audio/audio_frame.h"
> #include "api/audio_codecs/builtin_audio_decoder_factory.h"
> #include "rtc_base/format_macros.h"
> #include "test/gtest.h"
>@@ -31,9 +31,8 @@ NetEqExternalDecoderTest::NetEqExternalDecoderTest(NetEqDecoder codec,
> }
> 
> void NetEqExternalDecoderTest::Init() {
>-  ASSERT_EQ(NetEq::kOK,
>-            neteq_->RegisterExternalDecoder(decoder_, codec_, name_,
>-                                            kPayloadType));
>+  ASSERT_EQ(NetEq::kOK, neteq_->RegisterExternalDecoder(decoder_, codec_, name_,
>+                                                        kPayloadType));
> }
> 
> void NetEqExternalDecoderTest::InsertPacket(
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h
>index aefa62e4d17..78f0085e036 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h
>@@ -17,7 +17,6 @@
> #include "api/audio_codecs/audio_decoder.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/audio_coding/neteq/include/neteq.h"
>-#include "modules/include/module_common_types.h"
> 
> namespace webrtc {
> namespace test {
>@@ -32,7 +31,7 @@ class NetEqExternalDecoderTest {
>                            int sample_rate_hz,
>                            AudioDecoder* decoder);
> 
>-  virtual ~NetEqExternalDecoderTest() { }
>+  virtual ~NetEqExternalDecoderTest() {}
> 
>   // In Init(), we register the external decoder.
>   void Init();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_input.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_input.cc
>index 44513ab15b5..bb2a01e0e17 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_input.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_input.cc
>@@ -15,6 +15,9 @@
> namespace webrtc {
> namespace test {
> 
>+NetEqInput::PacketData::PacketData() = default;
>+NetEqInput::PacketData::~PacketData() = default;
>+
> std::string NetEqInput::PacketData::ToString() const {
>   std::stringstream ss;
>   ss << "{"
>@@ -28,5 +31,52 @@ std::string NetEqInput::PacketData::ToString() const {
>   return ss.str();
> }
> 
>+TimeLimitedNetEqInput::TimeLimitedNetEqInput(std::unique_ptr<NetEqInput> input,
>+                                             int64_t duration_ms)
>+    : input_(std::move(input)),
>+      start_time_ms_(input_->NextEventTime()),
>+      duration_ms_(duration_ms) {}
>+
>+TimeLimitedNetEqInput::~TimeLimitedNetEqInput() = default;
>+
>+absl::optional<int64_t> TimeLimitedNetEqInput::NextPacketTime() const {
>+  return ended_ ? absl::nullopt : input_->NextPacketTime();
>+}
>+
>+absl::optional<int64_t> TimeLimitedNetEqInput::NextOutputEventTime() const {
>+  return ended_ ? absl::nullopt : input_->NextOutputEventTime();
>+}
>+
>+std::unique_ptr<NetEqInput::PacketData> TimeLimitedNetEqInput::PopPacket() {
>+  if (ended_) {
>+    return std::unique_ptr<PacketData>();
>+  }
>+  auto packet = input_->PopPacket();
>+  MaybeSetEnded();
>+  return packet;
>+}
>+
>+void TimeLimitedNetEqInput::AdvanceOutputEvent() {
>+  if (!ended_) {
>+    input_->AdvanceOutputEvent();
>+    MaybeSetEnded();
>+  }
>+}
>+
>+bool TimeLimitedNetEqInput::ended() const {
>+  return ended_ || input_->ended();
>+}
>+
>+absl::optional<RTPHeader> TimeLimitedNetEqInput::NextHeader() const {
>+  return ended_ ? absl::nullopt : input_->NextHeader();
>+}
>+
>+void TimeLimitedNetEqInput::MaybeSetEnded() {
>+  if (NextEventTime() && start_time_ms_ &&
>+      *NextEventTime() - *start_time_ms_ > duration_ms_) {
>+    ended_ = true;
>+  }
>+}
>+
> }  // namespace test
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_input.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_input.h
>index 88d9eb94139..b9cba098d86 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_input.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_input.h
>@@ -15,7 +15,7 @@
> #include <memory>
> #include <string>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/audio_coding/neteq/tools/packet.h"
> #include "modules/audio_coding/neteq/tools/packet_source.h"
>@@ -28,33 +28,35 @@ namespace test {
> class NetEqInput {
>  public:
>   struct PacketData {
>+    PacketData();
>+    ~PacketData();
>     std::string ToString() const;
> 
>     RTPHeader header;
>     rtc::Buffer payload;
>-    double time_ms;
>+    int64_t time_ms;
>   };
> 
>   virtual ~NetEqInput() = default;
> 
>   // Returns at what time (in ms) NetEq::InsertPacket should be called next, or
>   // empty if the source is out of packets.
>-  virtual rtc::Optional<int64_t> NextPacketTime() const = 0;
>+  virtual absl::optional<int64_t> NextPacketTime() const = 0;
> 
>   // Returns at what time (in ms) NetEq::GetAudio should be called next, or
>   // empty if no more output events are available.
>-  virtual rtc::Optional<int64_t> NextOutputEventTime() const = 0;
>+  virtual absl::optional<int64_t> NextOutputEventTime() const = 0;
> 
>   // Returns the time (in ms) for the next event from either NextPacketTime()
>   // or NextOutputEventTime(), or empty if both are out of events.
>-  rtc::Optional<int64_t> NextEventTime() const {
>+  absl::optional<int64_t> NextEventTime() const {
>     const auto a = NextPacketTime();
>     const auto b = NextOutputEventTime();
>     // Return the minimum of non-empty |a| and |b|, or empty if both are empty.
>     if (a) {
>       return b ? std::min(*a, *b) : a;
>     }
>-    return b ? b : rtc::nullopt;
>+    return b ? b : absl::nullopt;
>   }
> 
>   // Returns the next packet to be inserted into NetEq. The packet following the
>@@ -75,7 +77,30 @@ class NetEqInput {
> 
>   // Returns the RTP header for the next packet, i.e., the packet that will be
>   // delivered next by PopPacket().
>-  virtual rtc::Optional<RTPHeader> NextHeader() const = 0;
>+  virtual absl::optional<RTPHeader> NextHeader() const = 0;
>+};
>+
>+// Wrapper class to impose a time limit on a NetEqInput object, typically
>+// another time limit than what the object itself provides. For example, an
>+// input taken from a file can be cut shorter by wrapping it in this class.
>+class TimeLimitedNetEqInput : public NetEqInput {
>+ public:
>+  TimeLimitedNetEqInput(std::unique_ptr<NetEqInput> input, int64_t duration_ms);
>+  ~TimeLimitedNetEqInput() override;
>+  absl::optional<int64_t> NextPacketTime() const override;
>+  absl::optional<int64_t> NextOutputEventTime() const override;
>+  std::unique_ptr<PacketData> PopPacket() override;
>+  void AdvanceOutputEvent() override;
>+  bool ended() const override;
>+  absl::optional<RTPHeader> NextHeader() const override;
>+
>+ private:
>+  void MaybeSetEnded();
>+
>+  std::unique_ptr<NetEqInput> input_;
>+  const absl::optional<int64_t> start_time_ms_;
>+  const int64_t duration_ms_;
>+  bool ended_ = false;
> };
> 
> }  // namespace test
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_packet_source_input.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_packet_source_input.cc
>index 0741d7c88dd..a86cf6afa59 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_packet_source_input.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_packet_source_input.cc
>@@ -13,7 +13,6 @@
> #include <algorithm>
> #include <limits>
> 
>-#include "modules/audio_coding/neteq/tools/rtc_event_log_source.h"
> #include "modules/audio_coding/neteq/tools/rtp_file_source.h"
> #include "rtc_base/checks.h"
> 
>@@ -22,15 +21,14 @@ namespace test {
> 
> NetEqPacketSourceInput::NetEqPacketSourceInput() : next_output_event_ms_(0) {}
> 
>-rtc::Optional<int64_t> NetEqPacketSourceInput::NextPacketTime() const {
>+absl::optional<int64_t> NetEqPacketSourceInput::NextPacketTime() const {
>   return packet_
>-             ? rtc::Optional<int64_t>(static_cast<int64_t>(packet_->time_ms()))
>-             : rtc::nullopt;
>+             ? absl::optional<int64_t>(static_cast<int64_t>(packet_->time_ms()))
>+             : absl::nullopt;
> }
> 
>-rtc::Optional<RTPHeader> NetEqPacketSourceInput::NextHeader() const {
>-  return packet_ ? rtc::Optional<RTPHeader>(packet_->header())
>-                 : rtc::nullopt;
>+absl::optional<RTPHeader> NetEqPacketSourceInput::NextHeader() const {
>+  return packet_ ? absl::optional<RTPHeader>(packet_->header()) : absl::nullopt;
> }
> 
> void NetEqPacketSourceInput::LoadNextPacket() {
>@@ -60,6 +58,12 @@ std::unique_ptr<NetEqInput::PacketData> NetEqPacketSourceInput::PopPacket() {
>   return packet_data;
> }
> 
>+void NetEqPacketSourceInput::SelectSsrc(uint32_t ssrc) {
>+  source()->SelectSsrc(ssrc);
>+  if (packet_ && packet_->header().ssrc != ssrc)
>+    LoadNextPacket();
>+}
>+
> NetEqRtpDumpInput::NetEqRtpDumpInput(const std::string& file_name,
>                                      const RtpHeaderExtensionMap& hdr_ext_map)
>     : source_(RtpFileSource::Create(file_name)) {
>@@ -69,7 +73,7 @@ NetEqRtpDumpInput::NetEqRtpDumpInput(const std::string& file_name,
>   LoadNextPacket();
> }
> 
>-rtc::Optional<int64_t> NetEqRtpDumpInput::NextOutputEventTime() const {
>+absl::optional<int64_t> NetEqRtpDumpInput::NextOutputEventTime() const {
>   return next_output_event_ms_;
> }
> 
>@@ -78,7 +82,7 @@ void NetEqRtpDumpInput::AdvanceOutputEvent() {
>     *next_output_event_ms_ += kOutputPeriodMs;
>   }
>   if (!NextPacketTime()) {
>-    next_output_event_ms_ = rtc::nullopt;
>+    next_output_event_ms_ = absl::nullopt;
>   }
> }
> 
>@@ -86,30 +90,5 @@ PacketSource* NetEqRtpDumpInput::source() {
>   return source_.get();
> }
> 
>-NetEqEventLogInput::NetEqEventLogInput(const std::string& file_name,
>-                                       const RtpHeaderExtensionMap& hdr_ext_map)
>-    : source_(RtcEventLogSource::Create(file_name)) {
>-  for (const auto& ext_pair : hdr_ext_map) {
>-    source_->RegisterRtpHeaderExtension(ext_pair.second, ext_pair.first);
>-  }
>-  LoadNextPacket();
>-  AdvanceOutputEvent();
>-}
>-
>-rtc::Optional<int64_t> NetEqEventLogInput::NextOutputEventTime() const {
>-  return next_output_event_ms_;
>-}
>-
>-void NetEqEventLogInput::AdvanceOutputEvent() {
>-  next_output_event_ms_ = source_->NextAudioOutputEventMs();
>-  if (*next_output_event_ms_ == std::numeric_limits<int64_t>::max()) {
>-    next_output_event_ms_ = rtc::nullopt;
>-  }
>-}
>-
>-PacketSource* NetEqEventLogInput::source() {
>-  return source_.get();
>-}
>-
> }  // namespace test
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_packet_source_input.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_packet_source_input.h
>index b482556c5a8..b32c5340851 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_packet_source_input.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_packet_source_input.h
>@@ -21,7 +21,6 @@ namespace webrtc {
> namespace test {
> 
> class RtpFileSource;
>-class RtcEventLogSource;
> 
> // An adapter class to dress up a PacketSource object as a NetEqInput.
> class NetEqPacketSourceInput : public NetEqInput {
>@@ -29,16 +28,17 @@ class NetEqPacketSourceInput : public NetEqInput {
>   using RtpHeaderExtensionMap = std::map<int, webrtc::RTPExtensionType>;
> 
>   NetEqPacketSourceInput();
>-  rtc::Optional<int64_t> NextPacketTime() const override;
>+  absl::optional<int64_t> NextPacketTime() const override;
>   std::unique_ptr<PacketData> PopPacket() override;
>-  rtc::Optional<RTPHeader> NextHeader() const override;
>+  absl::optional<RTPHeader> NextHeader() const override;
>   bool ended() const override { return !next_output_event_ms_; }
>+  void SelectSsrc(uint32_t);
> 
>  protected:
>   virtual PacketSource* source() = 0;
>   void LoadNextPacket();
> 
>-  rtc::Optional<int64_t> next_output_event_ms_;
>+  absl::optional<int64_t> next_output_event_ms_;
> 
>  private:
>   std::unique_ptr<Packet> packet_;
>@@ -50,7 +50,7 @@ class NetEqRtpDumpInput final : public NetEqPacketSourceInput {
>   NetEqRtpDumpInput(const std::string& file_name,
>                     const RtpHeaderExtensionMap& hdr_ext_map);
> 
>-  rtc::Optional<int64_t> NextOutputEventTime() const override;
>+  absl::optional<int64_t> NextOutputEventTime() const override;
>   void AdvanceOutputEvent() override;
> 
>  protected:
>@@ -62,23 +62,6 @@ class NetEqRtpDumpInput final : public NetEqPacketSourceInput {
>   std::unique_ptr<RtpFileSource> source_;
> };
> 
>-// Implementation of NetEqPacketSourceInput to be used with an
>-// RtcEventLogSource.
>-class NetEqEventLogInput final : public NetEqPacketSourceInput {
>- public:
>-  NetEqEventLogInput(const std::string& file_name,
>-                     const RtpHeaderExtensionMap& hdr_ext_map);
>-
>-  rtc::Optional<int64_t> NextOutputEventTime() const override;
>-  void AdvanceOutputEvent() override;
>-
>- protected:
>-  PacketSource* source() override;
>-
>- private:
>-  std::unique_ptr<RtcEventLogSource> source_;
>-};
>-
> }  // namespace test
> }  // namespace webrtc
> #endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_PACKET_SOURCE_INPUT_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc
>index 27ecdf46797..52524fed2c4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc
>@@ -10,17 +10,16 @@
> 
> #include "modules/audio_coding/neteq/tools/neteq_performance_test.h"
> 
>+#include "api/audio/audio_frame.h"
> #include "api/audio_codecs/builtin_audio_decoder_factory.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
> #include "modules/audio_coding/neteq/include/neteq.h"
> #include "modules/audio_coding/neteq/tools/audio_loop.h"
> #include "modules/audio_coding/neteq/tools/rtp_generator.h"
>-#include "modules/include/module_common_types.h"
> #include "rtc_base/checks.h"
> #include "system_wrappers/include/clock.h"
> #include "test/testsupport/fileutils.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> using webrtc::NetEq;
> using webrtc::test::AudioLoop;
>@@ -95,15 +94,14 @@ int64_t NetEqPerformanceTest::Run(int runtime_ms,
>       }
> 
>       // Get next packet.
>-      packet_input_time_ms = rtp_gen.GetRtpHeader(kPayloadType,
>-                                                  kInputBlockSizeSamples,
>-                                                  &rtp_header);
>+      packet_input_time_ms = rtp_gen.GetRtpHeader(
>+          kPayloadType, kInputBlockSizeSamples, &rtp_header);
>       input_samples = audio_loop.GetNextBlock();
>       if (input_samples.empty())
>         return -1;
>       payload_len = WebRtcPcm16b_Encode(input_samples.data(),
>                                         input_samples.size(), input_payload);
>-      assert(payload_len == kInputBlockSizeSamples * sizeof(int16_t));
>+      RTC_DCHECK_EQ(payload_len, kInputBlockSizeSamples * sizeof(int16_t));
>     }
> 
>     // Get output audio, but don't do anything with it.
>@@ -113,8 +111,7 @@ int64_t NetEqPerformanceTest::Run(int runtime_ms,
>     if (error != NetEq::kOK)
>       return -1;
> 
>-    assert(out_frame.samples_per_channel_ ==
>-           static_cast<size_t>(kSampRateHz * 10 / 1000));
>+    RTC_DCHECK_EQ(out_frame.samples_per_channel_, (kSampRateHz * 10) / 1000);
> 
>     static const int kOutputBlockSizeMs = 10;
>     time_now_ms += kOutputBlockSizeMs;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.h
>index dcf0314b4bd..d2212f03a33 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.h
>@@ -11,7 +11,7 @@
> #ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_PERFORMANCE_TEST_H_
> #define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_PERFORMANCE_TEST_H_
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include <stdint.h>
> 
> namespace webrtc {
> namespace test {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc
>index 7df46ff4f3c..faca8959d4d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc
>@@ -47,7 +47,9 @@ static bool ValidateFilename(const std::string& value, bool write) {
>   return true;
> }
> 
>-DEFINE_string(in_filename, DefaultInFilename().c_str(),
>+DEFINE_string(
>+    in_filename,
>+    DefaultInFilename().c_str(),
>     "Filename for input audio (specify sample rate with --input_sample_rate, "
>     "and channels with --channels).");
> 
>@@ -55,41 +57,54 @@ DEFINE_int(input_sample_rate, 16000, "Sample rate of input file in Hz.");
> 
> DEFINE_int(channels, 1, "Number of channels in input audio.");
> 
>-DEFINE_string(out_filename, DefaultOutFilename().c_str(),
>-    "Name of output audio file.");
>+DEFINE_string(out_filename,
>+              DefaultOutFilename().c_str(),
>+              "Name of output audio file.");
> 
> DEFINE_int(runtime_ms, 10000, "Simulated runtime (milliseconds).");
> 
> DEFINE_int(packet_loss_rate, 10, "Percentile of packet loss.");
> 
>-DEFINE_int(random_loss_mode, 1,
>-    "Random loss mode: 0--no loss, 1--uniform loss, 2--Gilbert Elliot loss.");
>+DEFINE_int(random_loss_mode,
>+           kUniformLoss,
>+           "Random loss mode: 0--no loss, 1--uniform loss, 2--Gilbert Elliot "
>+           "loss, 3--fixed loss.");
> 
>-DEFINE_int(burst_length, 30,
>-    "Burst length in milliseconds, only valid for Gilbert Elliot loss.");
>+DEFINE_int(burst_length,
>+           30,
>+           "Burst length in milliseconds, only valid for Gilbert Elliot loss.");
> 
> DEFINE_float(drift_factor, 0.0, "Time drift factor.");
> 
>+DEFINE_int(preload_packets, 0, "Preload the buffer with this many packets.");
>+
>+DEFINE_string(loss_events,
>+              "",
>+              "List of loss events time and duration separated by comma: "
>+              "<first_event_time> <first_event_duration>, <second_event_time> "
>+              "<second_event_duration>, ...");
>+
> // ProbTrans00Solver() is to calculate the transition probability from no-loss
> // state to itself in a modified Gilbert Elliot packet loss model. The result is
> // to achieve the target packet loss rate |loss_rate|, when a packet is not
> // lost only if all |units| drawings within the duration of the packet result in
> // no-loss.
>-static double ProbTrans00Solver(int units, double loss_rate,
>+static double ProbTrans00Solver(int units,
>+                                double loss_rate,
>                                 double prob_trans_10) {
>   if (units == 1)
>     return prob_trans_10 / (1.0f - loss_rate) - prob_trans_10;
>-// 0 == prob_trans_00 ^ (units - 1) + (1 - loss_rate) / prob_trans_10 *
>-//     prob_trans_00 - (1 - loss_rate) * (1 + 1 / prob_trans_10).
>-// There is a unique solution between 0.0 and 1.0, due to the monotonicity and
>-// an opposite sign at 0.0 and 1.0.
>-// For simplicity, we reformulate the equation as
>-//     f(x) = x ^ (units - 1) + a x + b.
>-// Its derivative is
>-//     f'(x) = (units - 1) x ^ (units - 2) + a.
>-// The derivative is strictly greater than 0 when x is between 0 and 1.
>-// We use Newton's method to solve the equation, iteration is
>-//     x(k+1) = x(k) - f(x) / f'(x);
>+  // 0 == prob_trans_00 ^ (units - 1) + (1 - loss_rate) / prob_trans_10 *
>+  //     prob_trans_00 - (1 - loss_rate) * (1 + 1 / prob_trans_10).
>+  // There is a unique solution between 0.0 and 1.0, due to the monotonicity and
>+  // an opposite sign at 0.0 and 1.0.
>+  // For simplicity, we reformulate the equation as
>+  //     f(x) = x ^ (units - 1) + a x + b.
>+  // Its derivative is
>+  //     f'(x) = (units - 1) x ^ (units - 2) + a.
>+  // The derivative is strictly greater than 0 when x is between 0 and 1.
>+  // We use Newton's method to solve the equation, iteration is
>+  //     x(k+1) = x(k) - f(x) / f'(x);
>   const double kPrecision = 0.001f;
>   const int kIterations = 100;
>   const double a = (1.0f - loss_rate) / prob_trans_10;
>@@ -107,7 +122,7 @@ static double ProbTrans00Solver(int units, double loss_rate,
>       x = 0.0f;
>     }
>     f = pow(x, units - 1) + a * x + b;
>-    iter ++;
>+    iter++;
>   }
>   return x;
> }
>@@ -155,8 +170,9 @@ NetEqQualityTest::NetEqQualityTest(int block_duration_ms,
>   RTC_CHECK(FLAG_packet_loss_rate >= 0 && FLAG_packet_loss_rate <= 100)
>       << "Invalid packet loss percentile, should be between 0 and 100.";
> 
>-  RTC_CHECK(FLAG_random_loss_mode >= 0 && FLAG_random_loss_mode <= 2)
>-      << "Invalid random packet loss mode, should be between 0 and 2.";
>+  RTC_CHECK(FLAG_random_loss_mode >= 0 && FLAG_random_loss_mode < kLastLossMode)
>+      << "Invalid random packet loss mode, should be between 0 and "
>+      << kLastLossMode - 1 << ".";
> 
>   RTC_CHECK_GE(FLAG_burst_length, kPacketLossTimeUnitMs)
>       << "Invalid burst length, should be greater than or equal to "
>@@ -165,6 +181,9 @@ NetEqQualityTest::NetEqQualityTest(int block_duration_ms,
>   RTC_CHECK_GT(FLAG_drift_factor, -0.1)
>       << "Invalid drift factor, should be greater than -0.1.";
> 
>+  RTC_CHECK_GE(FLAG_preload_packets, 0)
>+      << "Invalid number of packets to preload; must be non-negative.";
>+
>   const std::string out_filename = FLAG_out_filename;
>   const std::string log_filename = out_filename + ".log";
>   log_file_.open(log_filename.c_str(), std::ofstream::out);
>@@ -192,15 +211,13 @@ NetEqQualityTest::~NetEqQualityTest() {
>   log_file_.close();
> }
> 
>-bool NoLoss::Lost() {
>+bool NoLoss::Lost(int now_ms) {
>   return false;
> }
> 
>-UniformLoss::UniformLoss(double loss_rate)
>-    : loss_rate_(loss_rate) {
>-}
>+UniformLoss::UniformLoss(double loss_rate) : loss_rate_(loss_rate) {}
> 
>-bool UniformLoss::Lost() {
>+bool UniformLoss::Lost(int now_ms) {
>   int drop_this = rand();
>   return (drop_this < loss_rate_ * RAND_MAX);
> }
>@@ -209,23 +226,43 @@ GilbertElliotLoss::GilbertElliotLoss(double prob_trans_11, double prob_trans_01)
>     : prob_trans_11_(prob_trans_11),
>       prob_trans_01_(prob_trans_01),
>       lost_last_(false),
>-      uniform_loss_model_(new UniformLoss(0)) {
>-}
>+      uniform_loss_model_(new UniformLoss(0)) {}
> 
> GilbertElliotLoss::~GilbertElliotLoss() {}
> 
>-bool GilbertElliotLoss::Lost() {
>+bool GilbertElliotLoss::Lost(int now_ms) {
>   // Simulate bursty channel (Gilbert model).
>   // (1st order) Markov chain model with memory of the previous/last
>   // packet state (lost or received).
>   if (lost_last_) {
>     // Previous packet was not received.
>     uniform_loss_model_->set_loss_rate(prob_trans_11_);
>-    return lost_last_ = uniform_loss_model_->Lost();
>+    return lost_last_ = uniform_loss_model_->Lost(now_ms);
>   } else {
>     uniform_loss_model_->set_loss_rate(prob_trans_01_);
>-    return lost_last_ = uniform_loss_model_->Lost();
>+    return lost_last_ = uniform_loss_model_->Lost(now_ms);
>+  }
>+}
>+
>+FixedLossModel::FixedLossModel(
>+    std::set<FixedLossEvent, FixedLossEventCmp> loss_events)
>+    : loss_events_(loss_events) {
>+  loss_events_it_ = loss_events_.begin();
>+}
>+
>+FixedLossModel::~FixedLossModel() {}
>+
>+bool FixedLossModel::Lost(int now_ms) {
>+  if (loss_events_it_ != loss_events_.end() &&
>+      now_ms > loss_events_it_->start_ms) {
>+    if (now_ms <= loss_events_it_->start_ms + loss_events_it_->duration_ms) {
>+      return true;
>+    } else {
>+      ++loss_events_it_;
>+      return false;
>+    }
>   }
>+  return false;
> }
> 
> void NetEqQualityTest::SetUp() {
>@@ -235,19 +272,19 @@ void NetEqQualityTest::SetUp() {
> 
>   int units = block_duration_ms_ / kPacketLossTimeUnitMs;
>   switch (FLAG_random_loss_mode) {
>-    case 1: {
>+    case kUniformLoss: {
>       // |unit_loss_rate| is the packet loss rate for each unit time interval
>       // (kPacketLossTimeUnitMs). Since a packet loss event is generated if any
>       // of |block_duration_ms_ / kPacketLossTimeUnitMs| unit time intervals of
>       // a full packet duration is drawn with a loss, |unit_loss_rate| fulfills
>       // (1 - unit_loss_rate) ^ (block_duration_ms_ / kPacketLossTimeUnitMs) ==
>       // 1 - packet_loss_rate.
>-      double unit_loss_rate = (1.0f - pow(1.0f - 0.01f * packet_loss_rate_,
>-          1.0f / units));
>+      double unit_loss_rate =
>+          (1.0f - pow(1.0f - 0.01f * packet_loss_rate_, 1.0f / units));
>       loss_model_.reset(new UniformLoss(unit_loss_rate));
>       break;
>     }
>-    case 2: {
>+    case kGilbertElliotLoss: {
>       // |FLAG_burst_length| should be integer times of kPacketLossTimeUnitMs.
>       ASSERT_EQ(0, FLAG_burst_length % kPacketLossTimeUnitMs);
> 
>@@ -269,8 +306,27 @@ void NetEqQualityTest::SetUp() {
>       double loss_rate = 0.01f * packet_loss_rate_;
>       double prob_trans_10 = 1.0f * kPacketLossTimeUnitMs / FLAG_burst_length;
>       double prob_trans_00 = ProbTrans00Solver(units, loss_rate, prob_trans_10);
>-      loss_model_.reset(new GilbertElliotLoss(1.0f - prob_trans_10,
>-                                              1.0f - prob_trans_00));
>+      loss_model_.reset(
>+          new GilbertElliotLoss(1.0f - prob_trans_10, 1.0f - prob_trans_00));
>+      break;
>+    }
>+    case kFixedLoss: {
>+      std::istringstream loss_events_stream(FLAG_loss_events);
>+      std::string loss_event_string;
>+      std::set<FixedLossEvent, FixedLossEventCmp> loss_events;
>+      while (std::getline(loss_events_stream, loss_event_string, ',')) {
>+        std::vector<int> loss_event_params;
>+        std::istringstream loss_event_params_stream(loss_event_string);
>+        std::copy(std::istream_iterator<int>(loss_event_params_stream),
>+                  std::istream_iterator<int>(),
>+                  std::back_inserter(loss_event_params));
>+        RTC_CHECK_EQ(loss_event_params.size(), 2);
>+        auto result = loss_events.insert(
>+            FixedLossEvent(loss_event_params[0], loss_event_params[1]));
>+        RTC_CHECK(result.second);
>+      }
>+      RTC_CHECK_GT(loss_events.size(), 0);
>+      loss_model_.reset(new FixedLossModel(loss_events));
>       break;
>     }
>     default: {
>@@ -293,8 +349,8 @@ bool NetEqQualityTest::PacketLost() {
>   // The loop is to make sure that codecs with different block lengths share the
>   // same packet loss profile.
>   bool lost = false;
>-  for (int idx = 0; idx < cycles; idx ++) {
>-    if (loss_model_->Lost()) {
>+  for (int idx = 0; idx < cycles; idx++) {
>+    if (loss_model_->Lost(decoded_time_ms_)) {
>       // The packet will be lost if any of the drawings indicates a loss, but
>       // the loop has to go on to make sure that codecs with different block
>       // lengths keep the same pace.
>@@ -305,14 +361,10 @@ bool NetEqQualityTest::PacketLost() {
> }
> 
> int NetEqQualityTest::Transmit() {
>-  int packet_input_time_ms =
>-      rtp_generator_->GetRtpHeader(kPayloadType, in_size_samples_,
>-                                   &rtp_header_);
>-  Log() << "Packet of size "
>-        << payload_size_bytes_
>-        << " bytes, for frame at "
>-        << packet_input_time_ms
>-        << " ms ";
>+  int packet_input_time_ms = rtp_generator_->GetRtpHeader(
>+      kPayloadType, in_size_samples_, &rtp_header_);
>+  Log() << "Packet of size " << payload_size_bytes_ << " bytes, for frame at "
>+        << packet_input_time_ms << " ms ";
>   if (payload_size_bytes_ > 0) {
>     if (!PacketLost()) {
>       int ret = neteq_->InsertPacket(
>@@ -352,13 +404,13 @@ void NetEqQualityTest::Simulate() {
>   int audio_size_samples;
> 
>   while (decoded_time_ms_ < FLAG_runtime_ms) {
>-    // Assume 10 packets in packets buffer.
>-    while (decodable_time_ms_ - 10 * block_duration_ms_ < decoded_time_ms_) {
>+    // Preload the buffer if needed.
>+    while (decodable_time_ms_ - FLAG_preload_packets * block_duration_ms_ <
>+           decoded_time_ms_) {
>       ASSERT_TRUE(in_file_->Read(in_size_samples_ * channels_, &in_data_[0]));
>       payload_.Clear();
>-      payload_size_bytes_ = EncodeBlock(&in_data_[0],
>-                                        in_size_samples_, &payload_,
>-                                        max_payload_bytes_);
>+      payload_size_bytes_ = EncodeBlock(&in_data_[0], in_size_samples_,
>+                                        &payload_, max_payload_bytes_);
>       total_payload_size_bytes_ += payload_size_bytes_;
>       decodable_time_ms_ = Transmit() + block_duration_ms_;
>     }
>@@ -368,8 +420,7 @@ void NetEqQualityTest::Simulate() {
>     }
>   }
>   Log() << "Average bit rate was "
>-        << 8.0f * total_payload_size_bytes_ / FLAG_runtime_ms
>-        << " kbps"
>+        << 8.0f * total_payload_size_bytes_ / FLAG_runtime_ms << " kbps"
>         << std::endl;
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h
>index 8532f25a0d7..3d7760e46d4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.h
>@@ -19,29 +19,35 @@
> #include "modules/audio_coding/neteq/tools/audio_sink.h"
> #include "modules/audio_coding/neteq/tools/input_audio_file.h"
> #include "modules/audio_coding/neteq/tools/rtp_generator.h"
>-#include "modules/include/module_common_types.h"
> #include "rtc_base/flags.h"
> #include "test/gtest.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> namespace test {
> 
>+enum LossModes {
>+  kNoLoss,
>+  kUniformLoss,
>+  kGilbertElliotLoss,
>+  kFixedLoss,
>+  kLastLossMode
>+};
>+
> class LossModel {
>  public:
>-  virtual ~LossModel() {};
>-  virtual bool Lost() = 0;
>+  virtual ~LossModel(){};
>+  virtual bool Lost(int now_ms) = 0;
> };
> 
> class NoLoss : public LossModel {
>  public:
>-  bool Lost() override;
>+  bool Lost(int now_ms) override;
> };
> 
> class UniformLoss : public LossModel {
>  public:
>   UniformLoss(double loss_rate);
>-  bool Lost() override;
>+  bool Lost(int now_ms) override;
>   void set_loss_rate(double loss_rate) { loss_rate_ = loss_rate; }
> 
>  private:
>@@ -52,7 +58,7 @@ class GilbertElliotLoss : public LossModel {
>  public:
>   GilbertElliotLoss(double prob_trans_11, double prob_trans_01);
>   ~GilbertElliotLoss() override;
>-  bool Lost() override;
>+  bool Lost(int now_ms) override;
> 
>  private:
>   // Prob. of losing current packet, when previous packet is lost.
>@@ -63,6 +69,31 @@ class GilbertElliotLoss : public LossModel {
>   std::unique_ptr<UniformLoss> uniform_loss_model_;
> };
> 
>+struct FixedLossEvent {
>+  int start_ms;
>+  int duration_ms;
>+  FixedLossEvent(int start_ms, int duration_ms)
>+      : start_ms(start_ms), duration_ms(duration_ms) {}
>+};
>+
>+struct FixedLossEventCmp {
>+  bool operator()(const FixedLossEvent& l_event,
>+                  const FixedLossEvent& r_event) const {
>+    return l_event.start_ms < r_event.start_ms;
>+  }
>+};
>+
>+class FixedLossModel : public LossModel {
>+ public:
>+  FixedLossModel(std::set<FixedLossEvent, FixedLossEventCmp> loss_events);
>+  ~FixedLossModel() override;
>+  bool Lost(int now_ms) override;
>+
>+ private:
>+  std::set<FixedLossEvent, FixedLossEventCmp> loss_events_;
>+  std::set<FixedLossEvent, FixedLossEventCmp>::iterator loss_events_it_;
>+};
>+
> class NetEqQualityTest : public ::testing::Test {
>  protected:
>   NetEqQualityTest(int block_duration_ms,
>@@ -78,8 +109,10 @@ class NetEqQualityTest : public ::testing::Test {
>   // |block_size_samples| (samples per channel),
>   // 2. save the bit stream to |payload| of |max_bytes| bytes in size,
>   // 3. returns the length of the payload (in bytes),
>-  virtual int EncodeBlock(int16_t* in_data, size_t block_size_samples,
>-                          rtc::Buffer* payload, size_t max_bytes) = 0;
>+  virtual int EncodeBlock(int16_t* in_data,
>+                          size_t block_size_samples,
>+                          rtc::Buffer* payload,
>+                          size_t max_bytes) = 0;
> 
>   // PacketLost(...) determines weather a packet sent at an indicated time gets
>   // lost or not.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_replacement_input.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_replacement_input.cc
>index 6c846c0d0d6..362eb89de96 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_replacement_input.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_replacement_input.cc
>@@ -28,16 +28,15 @@ NetEqReplacementInput::NetEqReplacementInput(
>   RTC_CHECK(source_);
>   packet_ = source_->PopPacket();
>   ReplacePacket();
>-  RTC_CHECK(packet_);
> }
> 
>-rtc::Optional<int64_t> NetEqReplacementInput::NextPacketTime() const {
>+absl::optional<int64_t> NetEqReplacementInput::NextPacketTime() const {
>   return packet_
>-             ? rtc::Optional<int64_t>(static_cast<int64_t>(packet_->time_ms))
>-             : rtc::nullopt;
>+             ? absl::optional<int64_t>(static_cast<int64_t>(packet_->time_ms))
>+             : absl::nullopt;
> }
> 
>-rtc::Optional<int64_t> NetEqReplacementInput::NextOutputEventTime() const {
>+absl::optional<int64_t> NetEqReplacementInput::NextOutputEventTime() const {
>   return source_->NextOutputEventTime();
> }
> 
>@@ -56,7 +55,7 @@ bool NetEqReplacementInput::ended() const {
>   return source_->ended();
> }
> 
>-rtc::Optional<RTPHeader> NetEqReplacementInput::NextHeader() const {
>+absl::optional<RTPHeader> NetEqReplacementInput::NextHeader() const {
>   return source_->NextHeader();
> }
> 
>@@ -82,7 +81,7 @@ void NetEqReplacementInput::ReplacePacket() {
>     return;
>   }
> 
>-  rtc::Optional<RTPHeader> next_hdr = source_->NextHeader();
>+  absl::optional<RTPHeader> next_hdr = source_->NextHeader();
>   RTC_DCHECK(next_hdr);
>   uint8_t payload[12];
>   RTC_DCHECK_LE(last_frame_size_timestamps_, 120 * 48);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_replacement_input.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_replacement_input.h
>index 3a893997712..9ce9b9dc635 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_replacement_input.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_replacement_input.h
>@@ -28,12 +28,12 @@ class NetEqReplacementInput : public NetEqInput {
>                         const std::set<uint8_t>& comfort_noise_types,
>                         const std::set<uint8_t>& forbidden_types);
> 
>-  rtc::Optional<int64_t> NextPacketTime() const override;
>-  rtc::Optional<int64_t> NextOutputEventTime() const override;
>+  absl::optional<int64_t> NextPacketTime() const override;
>+  absl::optional<int64_t> NextOutputEventTime() const override;
>   std::unique_ptr<PacketData> PopPacket() override;
>   void AdvanceOutputEvent() override;
>   bool ended() const override;
>-  rtc::Optional<RTPHeader> NextHeader() const override;
>+  absl::optional<RTPHeader> NextHeader() const override;
> 
>  private:
>   void ReplacePacket();
>@@ -42,7 +42,7 @@ class NetEqReplacementInput : public NetEqInput {
>   const uint8_t replacement_payload_type_;
>   const std::set<uint8_t> comfort_noise_types_;
>   const std::set<uint8_t> forbidden_types_;
>-  std::unique_ptr<PacketData> packet_;  // The next packet to deliver.
>+  std::unique_ptr<PacketData> packet_;         // The next packet to deliver.
>   uint32_t last_frame_size_timestamps_ = 960;  // Initial guess: 20 ms @ 48 kHz.
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
>index de7ff2ea99b..bf782bf3f6c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
>@@ -13,30 +13,26 @@
> #include <limits.h>  // For ULONG_MAX returned by strtoul.
> #include <stdio.h>
> #include <stdlib.h>  // For strtoul.
>-#include <string.h>
>-
>-#include <algorithm>
>-#include <ios>
> #include <iostream>
> #include <memory>
>-#include <numeric>
> #include <string>
> 
> #include "modules/audio_coding/neteq/include/neteq.h"
> #include "modules/audio_coding/neteq/tools/fake_decode_from_file.h"
> #include "modules/audio_coding/neteq/tools/input_audio_file.h"
> #include "modules/audio_coding/neteq/tools/neteq_delay_analyzer.h"
>+#include "modules/audio_coding/neteq/tools/neteq_event_log_input.h"
> #include "modules/audio_coding/neteq/tools/neteq_packet_source_input.h"
> #include "modules/audio_coding/neteq/tools/neteq_replacement_input.h"
>+#include "modules/audio_coding/neteq/tools/neteq_stats_getter.h"
> #include "modules/audio_coding/neteq/tools/neteq_test.h"
> #include "modules/audio_coding/neteq/tools/output_audio_file.h"
> #include "modules/audio_coding/neteq/tools/output_wav_file.h"
> #include "modules/audio_coding/neteq/tools/rtp_file_source.h"
>-#include "modules/include/module_common_types.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/flags.h"
>+#include "test/field_trial.h"
> #include "test/testsupport/fileutils.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> namespace test {
>@@ -76,7 +72,7 @@ bool ValidatePayloadType(int value) {
> 
> bool ValidateSsrcValue(const std::string& str) {
>   uint32_t dummy_ssrc;
>-  if (ParseSsrc(str, &dummy_ssrc)) // Value is ok.
>+  if (ParseSsrc(str, &dummy_ssrc))  // Value is ok.
>     return true;
>   printf("Invalid SSRC: %s\n", str.c_str());
>   return false;
>@@ -111,10 +107,15 @@ DEFINE_int(cn_nb, 13, "RTP payload type for comfort noise (8 kHz)");
> DEFINE_int(cn_wb, 98, "RTP payload type for comfort noise (16 kHz)");
> DEFINE_int(cn_swb32, 99, "RTP payload type for comfort noise (32 kHz)");
> DEFINE_int(cn_swb48, 100, "RTP payload type for comfort noise (48 kHz)");
>-DEFINE_bool(codec_map, false, "Prints the mapping between RTP payload type and "
>-    "codec");
>-DEFINE_string(replacement_audio_file, "",
>-              "A PCM file that will be used to populate ""dummy"" RTP packets");
>+DEFINE_bool(codec_map,
>+            false,
>+            "Prints the mapping between RTP payload type and "
>+            "codec");
>+DEFINE_string(replacement_audio_file,
>+              "",
>+              "A PCM file that will be used to populate "
>+              "dummy"
>+              " RTP packets");
> DEFINE_string(ssrc,
>               "",
>               "Only use packets with this SSRC (decimal or hex, the latter "
>@@ -122,6 +123,8 @@ DEFINE_string(ssrc,
> DEFINE_int(audio_level, 1, "Extension ID for audio level (RFC 6464)");
> DEFINE_int(abs_send_time, 3, "Extension ID for absolute sender time");
> DEFINE_int(transport_seq_no, 5, "Extension ID for transport sequence number");
>+DEFINE_int(video_content_type, 7, "Extension ID for video content type");
>+DEFINE_int(video_timing, 8, "Extension ID for video timing");
> DEFINE_bool(matlabplot,
>             false,
>             "Generates a matlab script for plotting the delay profile");
>@@ -129,6 +132,13 @@ DEFINE_bool(pythonplot,
>             false,
>             "Generates a python script for plotting the delay profile");
> DEFINE_bool(help, false, "Prints this message");
>+DEFINE_bool(concealment_events, false, "Prints concealment events");
>+DEFINE_string(
>+    force_fieldtrials,
>+    "",
>+    "Field trials control experimental feature code which can be forced. "
>+    "E.g. running with --force_fieldtrials=WebRTC-FooFeature/Enable/"
>+    " will assign the group Enable to field trial WebRTC-FooFeature.");
> 
> // Maps a codec type to a printable name string.
> std::string CodecName(NetEqDecoder codec) {
>@@ -208,7 +218,7 @@ void PrintCodecMapping() {
>   PrintCodecMappingEntry(NetEqDecoder::kDecoderCNGswb48kHz, FLAG_cn_swb48);
> }
> 
>-rtc::Optional<int> CodecSampleRate(uint8_t payload_type) {
>+absl::optional<int> CodecSampleRate(uint8_t payload_type) {
>   if (payload_type == FLAG_pcmu || payload_type == FLAG_pcma ||
>       payload_type == FLAG_ilbc || payload_type == FLAG_pcm16b ||
>       payload_type == FLAG_cn_nb || payload_type == FLAG_avt)
>@@ -225,59 +235,9 @@ rtc::Optional<int> CodecSampleRate(uint8_t payload_type) {
>     return 48000;
>   if (payload_type == FLAG_red)
>     return 0;
>-  return rtc::nullopt;
>+  return absl::nullopt;
> }
> 
>-// Class to let through only the packets with a given SSRC. Should be used as an
>-// outer layer on another NetEqInput object.
>-class FilterSsrcInput : public NetEqInput {
>- public:
>-  FilterSsrcInput(std::unique_ptr<NetEqInput> source, uint32_t ssrc)
>-      : source_(std::move(source)), ssrc_(ssrc) {
>-    FindNextWithCorrectSsrc();
>-    RTC_CHECK(source_->NextHeader()) << "Found no packet with SSRC = 0x"
>-                                     << std::hex << ssrc_;
>-  }
>-
>-  // All methods but PopPacket() simply relay to the |source_| object.
>-  rtc::Optional<int64_t> NextPacketTime() const override {
>-    return source_->NextPacketTime();
>-  }
>-  rtc::Optional<int64_t> NextOutputEventTime() const override {
>-    return source_->NextOutputEventTime();
>-  }
>-
>-  // Returns the next packet, and throws away upcoming packets that do not match
>-  // the desired SSRC.
>-  std::unique_ptr<PacketData> PopPacket() override {
>-    std::unique_ptr<PacketData> packet_to_return = source_->PopPacket();
>-    RTC_DCHECK(!packet_to_return || packet_to_return->header.ssrc == ssrc_);
>-    // Pre-fetch the next packet with correct SSRC. Hence, |source_| will always
>-    // be have a valid packet (or empty if no more packets are available) when
>-    // this method returns.
>-    FindNextWithCorrectSsrc();
>-    return packet_to_return;
>-  }
>-
>-  void AdvanceOutputEvent() override { source_->AdvanceOutputEvent(); }
>-
>-  bool ended() const override { return source_->ended(); }
>-
>-  rtc::Optional<RTPHeader> NextHeader() const override {
>-    return source_->NextHeader();
>-  }
>-
>- private:
>-  void FindNextWithCorrectSsrc() {
>-    while (source_->NextHeader() && source_->NextHeader()->ssrc != ssrc_) {
>-      source_->PopPacket();
>-    }
>-  }
>-
>-  std::unique_ptr<NetEqInput> source_;
>-  uint32_t ssrc_;
>-};
>-
> // A callback class which prints whenver the inserted packet stream changes
> // the SSRC.
> class SsrcSwitchDetector : public NetEqPostInsertPacket {
>@@ -292,8 +252,8 @@ class SsrcSwitchDetector : public NetEqPostInsertPacket {
>                          NetEq* neteq) override {
>     if (last_ssrc_ && packet.header.ssrc != *last_ssrc_) {
>       std::cout << "Changing streams from 0x" << std::hex << *last_ssrc_
>-                << " to 0x" << std::hex << packet.header.ssrc
>-                << std::dec << " (payload type "
>+                << " to 0x" << std::hex << packet.header.ssrc << std::dec
>+                << " (payload type "
>                 << static_cast<int>(packet.header.payloadType) << ")"
>                 << std::endl;
>     }
>@@ -305,125 +265,18 @@ class SsrcSwitchDetector : public NetEqPostInsertPacket {
> 
>  private:
>   NetEqPostInsertPacket* other_callback_;
>-  rtc::Optional<uint32_t> last_ssrc_;
>-};
>-
>-class StatsGetter : public NetEqGetAudioCallback {
>- public:
>-  // This struct is a replica of webrtc::NetEqNetworkStatistics, but with all
>-  // values stored in double precision.
>-  struct Stats {
>-    double current_buffer_size_ms = 0.0;
>-    double preferred_buffer_size_ms = 0.0;
>-    double jitter_peaks_found = 0.0;
>-    double packet_loss_rate = 0.0;
>-    double expand_rate = 0.0;
>-    double speech_expand_rate = 0.0;
>-    double preemptive_rate = 0.0;
>-    double accelerate_rate = 0.0;
>-    double secondary_decoded_rate = 0.0;
>-    double secondary_discarded_rate = 0.0;
>-    double clockdrift_ppm = 0.0;
>-    double added_zero_samples = 0.0;
>-    double mean_waiting_time_ms = 0.0;
>-    double median_waiting_time_ms = 0.0;
>-    double min_waiting_time_ms = 0.0;
>-    double max_waiting_time_ms = 0.0;
>-  };
>-
>-  // Takes a pointer to another callback object, which will be invoked after
>-  // this object finishes. This does not transfer ownership, and null is a
>-  // valid value.
>-  explicit StatsGetter(NetEqGetAudioCallback* other_callback)
>-      : other_callback_(other_callback) {}
>-
>-  void BeforeGetAudio(NetEq* neteq) override {
>-    if (other_callback_) {
>-      other_callback_->BeforeGetAudio(neteq);
>-    }
>-  }
>-
>-  void AfterGetAudio(int64_t time_now_ms,
>-                     const AudioFrame& audio_frame,
>-                     bool muted,
>-                     NetEq* neteq) override {
>-    if (++counter_ >= 100) {
>-      counter_ = 0;
>-      NetEqNetworkStatistics stats;
>-      RTC_CHECK_EQ(neteq->NetworkStatistics(&stats), 0);
>-      stats_.push_back(stats);
>-    }
>-    if (other_callback_) {
>-      other_callback_->AfterGetAudio(time_now_ms, audio_frame, muted, neteq);
>-    }
>-  }
>-
>-  double AverageSpeechExpandRate() const {
>-    double sum_speech_expand =
>-        std::accumulate(stats_.begin(), stats_.end(), double{0.0},
>-                        [](double a, NetEqNetworkStatistics b) {
>-                          return a + static_cast<double>(b.speech_expand_rate);
>-                        });
>-    return sum_speech_expand / 16384.0 / stats_.size();
>-  }
>-
>-  Stats AverageStats() const {
>-    Stats sum_stats = std::accumulate(
>-        stats_.begin(), stats_.end(), Stats(),
>-        [](Stats a, NetEqNetworkStatistics b) {
>-          a.current_buffer_size_ms += b.current_buffer_size_ms;
>-          a.preferred_buffer_size_ms += b.preferred_buffer_size_ms;
>-          a.jitter_peaks_found += b.jitter_peaks_found;
>-          a.packet_loss_rate += b.packet_loss_rate / 16384.0;
>-          a.expand_rate += b.expand_rate / 16384.0;
>-          a.speech_expand_rate += b.speech_expand_rate / 16384.0;
>-          a.preemptive_rate += b.preemptive_rate / 16384.0;
>-          a.accelerate_rate += b.accelerate_rate / 16384.0;
>-          a.secondary_decoded_rate += b.secondary_decoded_rate / 16384.0;
>-          a.secondary_discarded_rate += b.secondary_discarded_rate / 16384.0;
>-          a.clockdrift_ppm += b.clockdrift_ppm;
>-          a.added_zero_samples += b.added_zero_samples;
>-          a.mean_waiting_time_ms += b.mean_waiting_time_ms;
>-          a.median_waiting_time_ms += b.median_waiting_time_ms;
>-          a.min_waiting_time_ms =
>-              std::min(a.min_waiting_time_ms,
>-                       static_cast<double>(b.min_waiting_time_ms));
>-          a.max_waiting_time_ms =
>-              std::max(a.max_waiting_time_ms,
>-                       static_cast<double>(b.max_waiting_time_ms));
>-          return a;
>-        });
>-
>-    sum_stats.current_buffer_size_ms /= stats_.size();
>-    sum_stats.preferred_buffer_size_ms /= stats_.size();
>-    sum_stats.jitter_peaks_found /= stats_.size();
>-    sum_stats.packet_loss_rate /= stats_.size();
>-    sum_stats.expand_rate /= stats_.size();
>-    sum_stats.speech_expand_rate /= stats_.size();
>-    sum_stats.preemptive_rate /= stats_.size();
>-    sum_stats.accelerate_rate /= stats_.size();
>-    sum_stats.secondary_decoded_rate /= stats_.size();
>-    sum_stats.secondary_discarded_rate /= stats_.size();
>-    sum_stats.clockdrift_ppm /= stats_.size();
>-    sum_stats.added_zero_samples /= stats_.size();
>-    sum_stats.mean_waiting_time_ms /= stats_.size();
>-    sum_stats.median_waiting_time_ms /= stats_.size();
>-
>-    return sum_stats;
>-  }
>-
>- private:
>-  NetEqGetAudioCallback* other_callback_;
>-  size_t counter_ = 0;
>-  std::vector<NetEqNetworkStatistics> stats_;
>+  absl::optional<uint32_t> last_ssrc_;
> };
> 
> int RunTest(int argc, char* argv[]) {
>   std::string program_name = argv[0];
>-  std::string usage = "Tool for decoding an RTP dump file using NetEq.\n"
>-      "Run " + program_name + " --help for usage.\n"
>-      "Example usage:\n" + program_name +
>-      " input.rtp output.{pcm, wav}\n";
>+  std::string usage =
>+      "Tool for decoding an RTP dump file using NetEq.\n"
>+      "Run " +
>+      program_name +
>+      " --help for usage.\n"
>+      "Example usage:\n" +
>+      program_name + " input.rtp output.{pcm, wav}\n";
>   if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true)) {
>     return 1;
>   }
>@@ -446,6 +299,10 @@ int RunTest(int argc, char* argv[]) {
>     std::cout << usage;
>     return 0;
>   }
>+
>+  ValidateFieldTrialsStringOrDie(FLAG_force_fieldtrials);
>+  ScopedFieldTrials field_trials(FLAG_force_fieldtrials);
>+
>   RTC_CHECK(ValidatePayloadType(FLAG_pcmu));
>   RTC_CHECK(ValidatePayloadType(FLAG_pcma));
>   RTC_CHECK(ValidatePayloadType(FLAG_ilbc));
>@@ -470,12 +327,16 @@ int RunTest(int argc, char* argv[]) {
>   RTC_CHECK(ValidateExtensionId(FLAG_audio_level));
>   RTC_CHECK(ValidateExtensionId(FLAG_abs_send_time));
>   RTC_CHECK(ValidateExtensionId(FLAG_transport_seq_no));
>+  RTC_CHECK(ValidateExtensionId(FLAG_video_content_type));
>+  RTC_CHECK(ValidateExtensionId(FLAG_video_timing));
> 
>   // Gather RTP header extensions in a map.
>   NetEqPacketSourceInput::RtpHeaderExtensionMap rtp_ext_map = {
>       {FLAG_audio_level, kRtpExtensionAudioLevel},
>       {FLAG_abs_send_time, kRtpExtensionAbsoluteSendTime},
>-      {FLAG_transport_seq_no, kRtpExtensionTransportSequenceNumber}};
>+      {FLAG_transport_seq_no, kRtpExtensionTransportSequenceNumber},
>+      {FLAG_video_content_type, kRtpExtensionVideoContentType},
>+      {FLAG_video_timing, kRtpExtensionVideoTiming}};
> 
>   const std::string input_file_name = argv[1];
>   std::unique_ptr<NetEqInput> input;
>@@ -494,14 +355,13 @@ int RunTest(int argc, char* argv[]) {
>   if (strlen(FLAG_ssrc) > 0) {
>     uint32_t ssrc;
>     RTC_CHECK(ParseSsrc(FLAG_ssrc, &ssrc)) << "Flag verification has failed.";
>-    input.reset(new FilterSsrcInput(std::move(input), ssrc));
>+    static_cast<NetEqPacketSourceInput*>(input.get())->SelectSsrc(ssrc);
>   }
> 
>   // Check the sample rate.
>-  rtc::Optional<int> sample_rate_hz;
>+  absl::optional<int> sample_rate_hz;
>   std::set<std::pair<int, uint32_t>> discarded_pt_and_ssrc;
>-  while (input->NextHeader()) {
>-    rtc::Optional<RTPHeader> first_rtp_header = input->NextHeader();
>+  while (absl::optional<RTPHeader> first_rtp_header = input->NextHeader()) {
>     RTC_DCHECK(first_rtp_header);
>     sample_rate_hz = CodecSampleRate(first_rtp_header->payloadType);
>     if (sample_rate_hz) {
>@@ -548,34 +408,38 @@ int RunTest(int argc, char* argv[]) {
>   std::cout << "Output file: " << output_file_name << std::endl;
> 
>   NetEqTest::DecoderMap codecs = {
>-      {FLAG_pcmu, std::make_pair(NetEqDecoder::kDecoderPCMu, "pcmu")},
>-      {FLAG_pcma, std::make_pair(NetEqDecoder::kDecoderPCMa, "pcma")},
>-      {FLAG_ilbc, std::make_pair(NetEqDecoder::kDecoderILBC, "ilbc")},
>-      {FLAG_isac, std::make_pair(NetEqDecoder::kDecoderISAC, "isac")},
>-      {FLAG_isac_swb,
>-       std::make_pair(NetEqDecoder::kDecoderISACswb, "isac-swb")},
>-      {FLAG_opus, std::make_pair(NetEqDecoder::kDecoderOpus, "opus")},
>-      {FLAG_pcm16b, std::make_pair(NetEqDecoder::kDecoderPCM16B, "pcm16-nb")},
>-      {FLAG_pcm16b_wb,
>-       std::make_pair(NetEqDecoder::kDecoderPCM16Bwb, "pcm16-wb")},
>-      {FLAG_pcm16b_swb32,
>-       std::make_pair(NetEqDecoder::kDecoderPCM16Bswb32kHz, "pcm16-swb32")},
>-      {FLAG_pcm16b_swb48,
>-       std::make_pair(NetEqDecoder::kDecoderPCM16Bswb48kHz, "pcm16-swb48")},
>-      {FLAG_g722, std::make_pair(NetEqDecoder::kDecoderG722, "g722")},
>-      {FLAG_avt, std::make_pair(NetEqDecoder::kDecoderAVT, "avt")},
>-      {FLAG_avt_16, std::make_pair(NetEqDecoder::kDecoderAVT16kHz, "avt-16")},
>-      {FLAG_avt_32,
>-       std::make_pair(NetEqDecoder::kDecoderAVT32kHz, "avt-32")},
>-      {FLAG_avt_48,
>-       std::make_pair(NetEqDecoder::kDecoderAVT48kHz, "avt-48")},
>-      {FLAG_red, std::make_pair(NetEqDecoder::kDecoderRED, "red")},
>-      {FLAG_cn_nb, std::make_pair(NetEqDecoder::kDecoderCNGnb, "cng-nb")},
>-      {FLAG_cn_wb, std::make_pair(NetEqDecoder::kDecoderCNGwb, "cng-wb")},
>-      {FLAG_cn_swb32,
>-       std::make_pair(NetEqDecoder::kDecoderCNGswb32kHz, "cng-swb32")},
>-      {FLAG_cn_swb48,
>-       std::make_pair(NetEqDecoder::kDecoderCNGswb48kHz, "cng-swb48")}};
>+    {FLAG_pcmu, std::make_pair(NetEqDecoder::kDecoderPCMu, "pcmu")},
>+    {FLAG_pcma, std::make_pair(NetEqDecoder::kDecoderPCMa, "pcma")},
>+#ifdef WEBRTC_CODEC_ILBC
>+    {FLAG_ilbc, std::make_pair(NetEqDecoder::kDecoderILBC, "ilbc")},
>+#endif
>+    {FLAG_isac, std::make_pair(NetEqDecoder::kDecoderISAC, "isac")},
>+#if !defined(WEBRTC_ANDROID)
>+    {FLAG_isac_swb, std::make_pair(NetEqDecoder::kDecoderISACswb, "isac-swb")},
>+#endif
>+#ifdef WEBRTC_CODEC_OPUS
>+    {FLAG_opus, std::make_pair(NetEqDecoder::kDecoderOpus, "opus")},
>+#endif
>+    {FLAG_pcm16b, std::make_pair(NetEqDecoder::kDecoderPCM16B, "pcm16-nb")},
>+    {FLAG_pcm16b_wb,
>+     std::make_pair(NetEqDecoder::kDecoderPCM16Bwb, "pcm16-wb")},
>+    {FLAG_pcm16b_swb32,
>+     std::make_pair(NetEqDecoder::kDecoderPCM16Bswb32kHz, "pcm16-swb32")},
>+    {FLAG_pcm16b_swb48,
>+     std::make_pair(NetEqDecoder::kDecoderPCM16Bswb48kHz, "pcm16-swb48")},
>+    {FLAG_g722, std::make_pair(NetEqDecoder::kDecoderG722, "g722")},
>+    {FLAG_avt, std::make_pair(NetEqDecoder::kDecoderAVT, "avt")},
>+    {FLAG_avt_16, std::make_pair(NetEqDecoder::kDecoderAVT16kHz, "avt-16")},
>+    {FLAG_avt_32, std::make_pair(NetEqDecoder::kDecoderAVT32kHz, "avt-32")},
>+    {FLAG_avt_48, std::make_pair(NetEqDecoder::kDecoderAVT48kHz, "avt-48")},
>+    {FLAG_red, std::make_pair(NetEqDecoder::kDecoderRED, "red")},
>+    {FLAG_cn_nb, std::make_pair(NetEqDecoder::kDecoderCNGnb, "cng-nb")},
>+    {FLAG_cn_wb, std::make_pair(NetEqDecoder::kDecoderCNGwb, "cng-wb")},
>+    {FLAG_cn_swb32,
>+     std::make_pair(NetEqDecoder::kDecoderCNGswb32kHz, "cng-swb32")},
>+    {FLAG_cn_swb48,
>+     std::make_pair(NetEqDecoder::kDecoderCNGswb48kHz, "cng-swb48")}
>+  };
> 
>   // Check if a replacement audio file was provided.
>   std::unique_ptr<AudioDecoder> replacement_decoder;
>@@ -599,9 +463,8 @@ int RunTest(int argc, char* argv[]) {
> 
>     std::set<uint8_t> cn_types = std_set_int32_to_uint8(
>         {FLAG_cn_nb, FLAG_cn_wb, FLAG_cn_swb32, FLAG_cn_swb48});
>-    std::set<uint8_t> forbidden_types =
>-        std_set_int32_to_uint8({FLAG_g722, FLAG_red, FLAG_avt,
>-                                FLAG_avt_16, FLAG_avt_32, FLAG_avt_48});
>+    std::set<uint8_t> forbidden_types = std_set_int32_to_uint8(
>+        {FLAG_g722, FLAG_red, FLAG_avt, FLAG_avt_16, FLAG_avt_32, FLAG_avt_48});
>     input.reset(new NetEqReplacementInput(std::move(input), replacement_pt,
>                                           cn_types, forbidden_types));
> 
>@@ -623,7 +486,7 @@ int RunTest(int argc, char* argv[]) {
> 
>   SsrcSwitchDetector ssrc_switch_detector(delay_analyzer.get());
>   callbacks.post_insert_packet = &ssrc_switch_detector;
>-  StatsGetter stats_getter(delay_analyzer.get());
>+  NetEqStatsGetter stats_getter(std::move(delay_analyzer));
>   callbacks.get_audio_callback = &stats_getter;
>   NetEq::Config config;
>   config.sample_rate_hz = *sample_rate_hz;
>@@ -638,7 +501,8 @@ int RunTest(int argc, char* argv[]) {
>                  '_');
>     std::cout << "Creating Matlab plot script " << matlab_script_name + ".m"
>               << std::endl;
>-    delay_analyzer->CreateMatlabScript(matlab_script_name + ".m");
>+    stats_getter.delay_analyzer()->CreateMatlabScript(matlab_script_name +
>+                                                      ".m");
>   }
>   if (FLAG_pythonplot) {
>     auto python_script_name = output_file_name;
>@@ -646,7 +510,8 @@ int RunTest(int argc, char* argv[]) {
>                  '_');
>     std::cout << "Creating Python plot script " << python_script_name + ".py"
>               << std::endl;
>-    delay_analyzer->CreatePythonScript(python_script_name + ".py");
>+    stats_getter.delay_analyzer()->CreatePythonScript(python_script_name +
>+                                                      ".py");
>   }
> 
>   printf("Simulation statistics:\n");
>@@ -668,7 +533,12 @@ int RunTest(int argc, char* argv[]) {
>   printf("  max_waiting_time_ms: %f ms\n", stats.max_waiting_time_ms);
>   printf("  current_buffer_size_ms: %f ms\n", stats.current_buffer_size_ms);
>   printf("  preferred_buffer_size_ms: %f ms\n", stats.preferred_buffer_size_ms);
>-
>+  if (FLAG_concealment_events) {
>+    std::cout << " concealment_events_ms:" << std::endl;
>+    for (auto concealment_event : stats_getter.concealment_events())
>+      std::cout << concealment_event.ToString() << std::endl;
>+    std::cout << " end of concealment_events_ms" << std::endl;
>+  }
>   return 0;
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_stats_getter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_stats_getter.cc
>new file mode 100644
>index 00000000000..1a521760b1f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_stats_getter.cc
>@@ -0,0 +1,145 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_coding/neteq/tools/neteq_stats_getter.h"
>+
>+#include <algorithm>
>+#include <numeric>
>+#include <utility>
>+
>+#include "rtc_base/checks.h"
>+#include "rtc_base/strings/string_builder.h"
>+#include "rtc_base/timeutils.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+std::string NetEqStatsGetter::ConcealmentEvent::ToString() const {
>+  char ss_buf[256];
>+  rtc::SimpleStringBuilder ss(ss_buf);
>+  ss << "ConcealmentEvent duration_ms:" << duration_ms
>+     << " event_number:" << concealment_event_number
>+     << " time_from_previous_event_end_ms:" << time_from_previous_event_end_ms;
>+  return ss.str();
>+}
>+
>+NetEqStatsGetter::NetEqStatsGetter(
>+    std::unique_ptr<NetEqDelayAnalyzer> delay_analyzer)
>+    : delay_analyzer_(std::move(delay_analyzer)) {}
>+
>+void NetEqStatsGetter::BeforeGetAudio(NetEq* neteq) {
>+  if (delay_analyzer_) {
>+    delay_analyzer_->BeforeGetAudio(neteq);
>+  }
>+}
>+
>+void NetEqStatsGetter::AfterGetAudio(int64_t time_now_ms,
>+                                     const AudioFrame& audio_frame,
>+                                     bool muted,
>+                                     NetEq* neteq) {
>+  // TODO(minyue): Get stats should better not be called as a call back after
>+  // get audio. It is called independently from get audio in practice.
>+  const auto lifetime_stat = neteq->GetLifetimeStatistics();
>+  if (last_stats_query_time_ms_ == 0 ||
>+      rtc::TimeDiff(time_now_ms, last_stats_query_time_ms_) >=
>+          stats_query_interval_ms_) {
>+    NetEqNetworkStatistics stats;
>+    RTC_CHECK_EQ(neteq->NetworkStatistics(&stats), 0);
>+    stats_.push_back(std::make_pair(time_now_ms, stats));
>+    lifetime_stats_.push_back(std::make_pair(time_now_ms, lifetime_stat));
>+    last_stats_query_time_ms_ = time_now_ms;
>+  }
>+
>+  if (current_concealment_event_ != lifetime_stat.concealment_events &&
>+      voice_concealed_samples_until_last_event_ <
>+          lifetime_stat.voice_concealed_samples) {
>+    if (last_event_end_time_ms_ > 0) {
>+      // Do not account for the first event to avoid start of the call
>+      // skewing.
>+      ConcealmentEvent concealment_event;
>+      uint64_t last_event_voice_concealed_samples =
>+          lifetime_stat.voice_concealed_samples -
>+          voice_concealed_samples_until_last_event_;
>+      RTC_CHECK_GT(last_event_voice_concealed_samples, 0);
>+      concealment_event.duration_ms = last_event_voice_concealed_samples /
>+                                      (audio_frame.sample_rate_hz_ / 1000);
>+      concealment_event.concealment_event_number = current_concealment_event_;
>+      concealment_event.time_from_previous_event_end_ms =
>+          time_now_ms - last_event_end_time_ms_;
>+      concealment_events_.emplace_back(concealment_event);
>+      voice_concealed_samples_until_last_event_ =
>+          lifetime_stat.voice_concealed_samples;
>+    }
>+    last_event_end_time_ms_ = time_now_ms;
>+    voice_concealed_samples_until_last_event_ =
>+        lifetime_stat.voice_concealed_samples;
>+    current_concealment_event_ = lifetime_stat.concealment_events;
>+  }
>+
>+  if (delay_analyzer_) {
>+    delay_analyzer_->AfterGetAudio(time_now_ms, audio_frame, muted, neteq);
>+  }
>+}
>+
>+double NetEqStatsGetter::AverageSpeechExpandRate() const {
>+  double sum_speech_expand = std::accumulate(
>+      stats_.begin(), stats_.end(), double{0.0},
>+      [](double a, std::pair<int64_t, NetEqNetworkStatistics> b) {
>+        return a + static_cast<double>(b.second.speech_expand_rate);
>+      });
>+  return sum_speech_expand / 16384.0 / stats_.size();
>+}
>+
>+NetEqStatsGetter::Stats NetEqStatsGetter::AverageStats() const {
>+  Stats sum_stats = std::accumulate(
>+      stats_.begin(), stats_.end(), Stats(),
>+      [](Stats a, std::pair<int64_t, NetEqNetworkStatistics> bb) {
>+        const auto& b = bb.second;
>+        a.current_buffer_size_ms += b.current_buffer_size_ms;
>+        a.preferred_buffer_size_ms += b.preferred_buffer_size_ms;
>+        a.jitter_peaks_found += b.jitter_peaks_found;
>+        a.packet_loss_rate += b.packet_loss_rate / 16384.0;
>+        a.expand_rate += b.expand_rate / 16384.0;
>+        a.speech_expand_rate += b.speech_expand_rate / 16384.0;
>+        a.preemptive_rate += b.preemptive_rate / 16384.0;
>+        a.accelerate_rate += b.accelerate_rate / 16384.0;
>+        a.secondary_decoded_rate += b.secondary_decoded_rate / 16384.0;
>+        a.secondary_discarded_rate += b.secondary_discarded_rate / 16384.0;
>+        a.clockdrift_ppm += b.clockdrift_ppm;
>+        a.added_zero_samples += b.added_zero_samples;
>+        a.mean_waiting_time_ms += b.mean_waiting_time_ms;
>+        a.median_waiting_time_ms += b.median_waiting_time_ms;
>+        a.min_waiting_time_ms = std::min(
>+            a.min_waiting_time_ms, static_cast<double>(b.min_waiting_time_ms));
>+        a.max_waiting_time_ms = std::max(
>+            a.max_waiting_time_ms, static_cast<double>(b.max_waiting_time_ms));
>+        return a;
>+      });
>+
>+  sum_stats.current_buffer_size_ms /= stats_.size();
>+  sum_stats.preferred_buffer_size_ms /= stats_.size();
>+  sum_stats.jitter_peaks_found /= stats_.size();
>+  sum_stats.packet_loss_rate /= stats_.size();
>+  sum_stats.expand_rate /= stats_.size();
>+  sum_stats.speech_expand_rate /= stats_.size();
>+  sum_stats.preemptive_rate /= stats_.size();
>+  sum_stats.accelerate_rate /= stats_.size();
>+  sum_stats.secondary_decoded_rate /= stats_.size();
>+  sum_stats.secondary_discarded_rate /= stats_.size();
>+  sum_stats.clockdrift_ppm /= stats_.size();
>+  sum_stats.added_zero_samples /= stats_.size();
>+  sum_stats.mean_waiting_time_ms /= stats_.size();
>+  sum_stats.median_waiting_time_ms /= stats_.size();
>+
>+  return sum_stats;
>+}
>+
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_stats_getter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_stats_getter.h
>new file mode 100644
>index 00000000000..b1b12bb1f8d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_stats_getter.h
>@@ -0,0 +1,106 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_STATS_GETTER_H_
>+#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_STATS_GETTER_H_
>+
>+#include <memory>
>+#include <string>
>+#include <vector>
>+
>+#include "modules/audio_coding/neteq/tools/neteq_delay_analyzer.h"
>+#include "modules/audio_coding/neteq/tools/neteq_test.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+class NetEqStatsGetter : public NetEqGetAudioCallback {
>+ public:
>+  // This struct is a replica of webrtc::NetEqNetworkStatistics, but with all
>+  // values stored in double precision.
>+  struct Stats {
>+    double current_buffer_size_ms = 0.0;
>+    double preferred_buffer_size_ms = 0.0;
>+    double jitter_peaks_found = 0.0;
>+    double packet_loss_rate = 0.0;
>+    double expand_rate = 0.0;
>+    double speech_expand_rate = 0.0;
>+    double preemptive_rate = 0.0;
>+    double accelerate_rate = 0.0;
>+    double secondary_decoded_rate = 0.0;
>+    double secondary_discarded_rate = 0.0;
>+    double clockdrift_ppm = 0.0;
>+    double added_zero_samples = 0.0;
>+    double mean_waiting_time_ms = 0.0;
>+    double median_waiting_time_ms = 0.0;
>+    double min_waiting_time_ms = 0.0;
>+    double max_waiting_time_ms = 0.0;
>+  };
>+
>+  struct ConcealmentEvent {
>+    uint64_t duration_ms;
>+    size_t concealment_event_number;
>+    int64_t time_from_previous_event_end_ms;
>+    std::string ToString() const;
>+  };
>+
>+  // Takes a pointer to another callback object, which will be invoked after
>+  // this object finishes. This does not transfer ownership, and null is a
>+  // valid value.
>+  explicit NetEqStatsGetter(std::unique_ptr<NetEqDelayAnalyzer> delay_analyzer);
>+
>+  void set_stats_query_interval_ms(int64_t stats_query_interval_ms) {
>+    stats_query_interval_ms_ = stats_query_interval_ms;
>+  }
>+
>+  void BeforeGetAudio(NetEq* neteq) override;
>+
>+  void AfterGetAudio(int64_t time_now_ms,
>+                     const AudioFrame& audio_frame,
>+                     bool muted,
>+                     NetEq* neteq) override;
>+
>+  double AverageSpeechExpandRate() const;
>+
>+  NetEqDelayAnalyzer* delay_analyzer() const { return delay_analyzer_.get(); }
>+
>+  const std::vector<ConcealmentEvent>& concealment_events() const {
>+    // Do not account for the last concealment event to avoid potential end
>+    // call skewing.
>+    return concealment_events_;
>+  }
>+
>+  const std::vector<std::pair<int64_t, NetEqNetworkStatistics>>* stats() const {
>+    return &stats_;
>+  }
>+
>+  const std::vector<std::pair<int64_t, NetEqLifetimeStatistics>>*
>+  lifetime_stats() const {
>+    return &lifetime_stats_;
>+  }
>+
>+  Stats AverageStats() const;
>+
>+ private:
>+  std::unique_ptr<NetEqDelayAnalyzer> delay_analyzer_;
>+  int64_t stats_query_interval_ms_ = 1000;
>+  int64_t last_stats_query_time_ms_ = 0;
>+  std::vector<std::pair<int64_t, NetEqNetworkStatistics>> stats_;
>+  std::vector<std::pair<int64_t, NetEqLifetimeStatistics>> lifetime_stats_;
>+  size_t current_concealment_event_ = 1;
>+  uint64_t voice_concealed_samples_until_last_event_ = 0;
>+  std::vector<ConcealmentEvent> concealment_events_;
>+  int64_t last_event_end_time_ms_ = 0;
>+};
>+
>+}  // namespace test
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_STATS_GETTER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_test.cc
>index 39d95495779..3448f4e840f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_test.cc
>@@ -46,6 +46,8 @@ NetEqTest::NetEqTest(const NetEq::Config& config,
>   RegisterExternalDecoders(ext_codecs);
> }
> 
>+NetEqTest::~NetEqTest() = default;
>+
> int64_t NetEqTest::Run() {
>   const int64_t start_time_ms = *input_->NextEventTime();
>   int64_t time_now_ms = start_time_ms;
>@@ -111,6 +113,42 @@ NetEqNetworkStatistics NetEqTest::SimulationStats() {
>   return stats;
> }
> 
>+NetEqLifetimeStatistics NetEqTest::LifetimeStats() const {
>+  return neteq_->GetLifetimeStatistics();
>+}
>+
>+NetEqTest::DecoderMap NetEqTest::StandardDecoderMap() {
>+  DecoderMap codecs = {
>+    {0, std::make_pair(NetEqDecoder::kDecoderPCMu, "pcmu")},
>+    {8, std::make_pair(NetEqDecoder::kDecoderPCMa, "pcma")},
>+#ifdef WEBRTC_CODEC_ILBC
>+    {102, std::make_pair(NetEqDecoder::kDecoderILBC, "ilbc")},
>+#endif
>+    {103, std::make_pair(NetEqDecoder::kDecoderISAC, "isac")},
>+#if !defined(WEBRTC_ANDROID)
>+    {104, std::make_pair(NetEqDecoder::kDecoderISACswb, "isac-swb")},
>+#endif
>+#ifdef WEBRTC_CODEC_OPUS
>+    {111, std::make_pair(NetEqDecoder::kDecoderOpus, "opus")},
>+#endif
>+    {93, std::make_pair(NetEqDecoder::kDecoderPCM16B, "pcm16-nb")},
>+    {94, std::make_pair(NetEqDecoder::kDecoderPCM16Bwb, "pcm16-wb")},
>+    {95, std::make_pair(NetEqDecoder::kDecoderPCM16Bswb32kHz, "pcm16-swb32")},
>+    {96, std::make_pair(NetEqDecoder::kDecoderPCM16Bswb48kHz, "pcm16-swb48")},
>+    {9, std::make_pair(NetEqDecoder::kDecoderG722, "g722")},
>+    {106, std::make_pair(NetEqDecoder::kDecoderAVT, "avt")},
>+    {114, std::make_pair(NetEqDecoder::kDecoderAVT16kHz, "avt-16")},
>+    {115, std::make_pair(NetEqDecoder::kDecoderAVT32kHz, "avt-32")},
>+    {116, std::make_pair(NetEqDecoder::kDecoderAVT48kHz, "avt-48")},
>+    {117, std::make_pair(NetEqDecoder::kDecoderRED, "red")},
>+    {13, std::make_pair(NetEqDecoder::kDecoderCNGnb, "cng-nb")},
>+    {98, std::make_pair(NetEqDecoder::kDecoderCNGwb, "cng-wb")},
>+    {99, std::make_pair(NetEqDecoder::kDecoderCNGswb32kHz, "cng-swb32")},
>+    {100, std::make_pair(NetEqDecoder::kDecoderCNGswb48kHz, "cng-swb48")}
>+  };
>+  return codecs;
>+}
>+
> void NetEqTest::RegisterDecoders(const DecoderMap& codecs) {
>   for (const auto& c : codecs) {
>     RTC_CHECK_EQ(
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_test.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_test.h
>index 2c0a07caada..ada6bab5bce 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_test.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_test.h
>@@ -82,13 +82,16 @@ class NetEqTest {
>             std::unique_ptr<AudioSink> output,
>             Callbacks callbacks);
> 
>-  ~NetEqTest() = default;
>+  ~NetEqTest();
> 
>   // Runs the test. Returns the duration of the produced audio in ms.
>   int64_t Run();
> 
>   // Returns the statistics from NetEq.
>   NetEqNetworkStatistics SimulationStats();
>+  NetEqLifetimeStatistics LifetimeStats() const;
>+
>+  static DecoderMap StandardDecoderMap();
> 
>  private:
>   void RegisterDecoders(const DecoderMap& codecs);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/packet.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/packet.cc
>index 71337b6e68b..b1a9b64193b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/packet.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/packet.cc
>@@ -14,7 +14,6 @@
> 
> #include <memory>
> 
>-#include "modules/include/module_common_types.h"
> #include "modules/rtp_rtcp/include/rtp_header_parser.h"
> #include "rtc_base/checks.h"
> 
>@@ -159,11 +158,10 @@ void Packet::CopyToHeader(RTPHeader* destination) const {
>   destination->paddingLength = header_.paddingLength;
>   destination->headerLength = header_.headerLength;
>   destination->payload_type_frequency = header_.payload_type_frequency;
>-  memcpy(&destination->arrOfCSRCs,
>-         &header_.arrOfCSRCs,
>+  memcpy(&destination->arrOfCSRCs, &header_.arrOfCSRCs,
>          sizeof(header_.arrOfCSRCs));
>-  memcpy(
>-      &destination->extension, &header_.extension, sizeof(header_.extension));
>+  memcpy(&destination->extension, &header_.extension,
>+         sizeof(header_.extension));
> }
> 
> }  // namespace test
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/packet.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/packet.h
>index 94d45c5b2d9..623c5cb6359 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/packet.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/packet.h
>@@ -15,9 +15,8 @@
> #include <memory>
> 
> #include "api/rtp_headers.h"  // NOLINT(build/include)
>-#include "common_types.h"  // NOLINT(build/include)
>+#include "common_types.h"     // NOLINT(build/include)
> #include "rtc_base/constructormagic.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/packet_source.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/packet_source.h
>index 3f98ba16a9c..fb689e33d4e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/packet_source.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/packet_source.h
>@@ -16,7 +16,6 @@
> 
> #include "modules/audio_coding/neteq/tools/packet.h"
> #include "rtc_base/constructormagic.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> namespace test {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/packet_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/packet_unittest.cc
>index ce6a3b96971..7f3d6630c32 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/packet_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/packet_unittest.cc
>@@ -28,7 +28,7 @@ void MakeRtpHeader(int payload_type,
>   rtp_data[0] = 0x80;
>   rtp_data[1] = static_cast<uint8_t>(payload_type);
>   rtp_data[2] = (seq_number >> 8) & 0xFF;
>-  rtp_data[3] = (seq_number) & 0xFF;
>+  rtp_data[3] = (seq_number)&0xFF;
>   rtp_data[4] = timestamp >> 24;
>   rtp_data[5] = (timestamp >> 16) & 0xFF;
>   rtp_data[6] = (timestamp >> 8) & 0xFF;
>@@ -47,8 +47,8 @@ TEST(TestPacket, RegularPacket) {
>   const uint16_t kSequenceNumber = 4711;
>   const uint32_t kTimestamp = 47114711;
>   const uint32_t kSsrc = 0x12345678;
>-  MakeRtpHeader(
>-      kPayloadType, kSequenceNumber, kTimestamp, kSsrc, packet_memory);
>+  MakeRtpHeader(kPayloadType, kSequenceNumber, kTimestamp, kSsrc,
>+                packet_memory);
>   const double kPacketTime = 1.0;
>   // Hand over ownership of |packet_memory| to |packet|.
>   Packet packet(packet_memory, kPacketLengthBytes, kPacketTime);
>@@ -75,13 +75,11 @@ TEST(TestPacket, DummyPacket) {
>   const uint16_t kSequenceNumber = 4711;
>   const uint32_t kTimestamp = 47114711;
>   const uint32_t kSsrc = 0x12345678;
>-  MakeRtpHeader(
>-      kPayloadType, kSequenceNumber, kTimestamp, kSsrc, packet_memory);
>+  MakeRtpHeader(kPayloadType, kSequenceNumber, kTimestamp, kSsrc,
>+                packet_memory);
>   const double kPacketTime = 1.0;
>   // Hand over ownership of |packet_memory| to |packet|.
>-  Packet packet(packet_memory,
>-                kPacketLengthBytes,
>-                kVirtualPacketLengthBytes,
>+  Packet packet(packet_memory, kPacketLengthBytes, kVirtualPacketLengthBytes,
>                 kPacketTime);
>   ASSERT_TRUE(packet.valid_header());
>   EXPECT_EQ(kPayloadType, packet.header().payloadType);
>@@ -140,8 +138,8 @@ TEST(TestPacket, RED) {
>   const uint16_t kSequenceNumber = 4711;
>   const uint32_t kTimestamp = 47114711;
>   const uint32_t kSsrc = 0x12345678;
>-  MakeRtpHeader(
>-      kRedPayloadType, kSequenceNumber, kTimestamp, kSsrc, packet_memory);
>+  MakeRtpHeader(kRedPayloadType, kSequenceNumber, kTimestamp, kSsrc,
>+                packet_memory);
>   // Create four RED headers.
>   // Payload types are just the same as the block index the offset is 100 times
>   // the block index.
>@@ -154,8 +152,8 @@ TEST(TestPacket, RED) {
>     uint32_t timestamp_offset = 100 * i;
>     int block_length = 10 * i;
>     bool last_block = (i == kRedBlocks - 1) ? true : false;
>-    payload_ptr += MakeRedHeader(
>-        payload_type, timestamp_offset, block_length, last_block, payload_ptr);
>+    payload_ptr += MakeRedHeader(payload_type, timestamp_offset, block_length,
>+                                 last_block, payload_ptr);
>   }
>   const double kPacketTime = 1.0;
>   // Hand over ownership of |packet_memory| to |packet|.
>@@ -178,8 +176,7 @@ TEST(TestPacket, RED) {
>   EXPECT_EQ(kRedBlocks, static_cast<int>(red_headers.size()));
>   int block_index = 0;
>   for (std::list<RTPHeader*>::reverse_iterator it = red_headers.rbegin();
>-       it != red_headers.rend();
>-       ++it) {
>+       it != red_headers.rend(); ++it) {
>     // Reading list from the back, since the extraction puts the main payload
>     // (which is the last one on wire) first.
>     RTPHeader* red_block = *it;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h
>index 13c419d969e..4e2e9b0bd93 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h
>@@ -16,7 +16,6 @@
> #include "common_audio/resampler/include/resampler.h"
> #include "modules/audio_coding/neteq/tools/input_audio_file.h"
> #include "rtc_base/constructormagic.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> namespace test {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc
>index d6224ffa0e7..7c97824551e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc
>@@ -40,7 +40,7 @@ std::unique_ptr<Packet> RtcEventLogSource::NextPacket() {
>   for (; rtp_packet_index_ < parsed_stream_.GetNumberOfEvents();
>        rtp_packet_index_++) {
>     if (parsed_stream_.GetEventType(rtp_packet_index_) ==
>-        ParsedRtcEventLog::RTP_EVENT) {
>+        ParsedRtcEventLogNew::EventType::RTP_EVENT) {
>       PacketDirection direction;
>       size_t header_length;
>       size_t packet_length;
>@@ -66,7 +66,7 @@ std::unique_ptr<Packet> RtcEventLogSource::NextPacket() {
>       }
> 
>       if (parsed_stream_.GetMediaType(packet->header().ssrc, direction) !=
>-          webrtc::ParsedRtcEventLog::MediaType::AUDIO) {
>+          ParsedRtcEventLogNew::MediaType::AUDIO) {
>         continue;
>       }
> 
>@@ -84,13 +84,13 @@ std::unique_ptr<Packet> RtcEventLogSource::NextPacket() {
> int64_t RtcEventLogSource::NextAudioOutputEventMs() {
>   while (audio_output_index_ < parsed_stream_.GetNumberOfEvents()) {
>     if (parsed_stream_.GetEventType(audio_output_index_) ==
>-        ParsedRtcEventLog::AUDIO_PLAYOUT_EVENT) {
>-      uint64_t timestamp_us = parsed_stream_.GetTimestamp(audio_output_index_);
>-      // We call GetAudioPlayout only to check that the protobuf event is
>-      // well-formed.
>-      parsed_stream_.GetAudioPlayout(audio_output_index_, nullptr);
>-      audio_output_index_++;
>-      return timestamp_us / 1000;
>+        ParsedRtcEventLogNew::EventType::AUDIO_PLAYOUT_EVENT) {
>+      LoggedAudioPlayoutEvent playout_event =
>+          parsed_stream_.GetAudioPlayout(audio_output_index_);
>+      if (!(use_ssrc_filter_ && playout_event.ssrc != ssrc_)) {
>+        audio_output_index_++;
>+        return playout_event.timestamp_us / 1000;
>+      }
>     }
>     audio_output_index_++;
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h
>index df01e065352..db4eb19e11f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h
>@@ -14,7 +14,7 @@
> #include <memory>
> #include <string>
> 
>-#include "logging/rtc_event_log/rtc_event_log_parser.h"
>+#include "logging/rtc_event_log/rtc_event_log_parser_new.h"
> #include "modules/audio_coding/neteq/tools/packet_source.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> #include "rtc_base/constructormagic.h"
>@@ -53,7 +53,7 @@ class RtcEventLogSource : public PacketSource {
>   size_t rtp_packet_index_ = 0;
>   size_t audio_output_index_ = 0;
> 
>-  ParsedRtcEventLog parsed_stream_;
>+  ParsedRtcEventLogNew parsed_stream_;
>   std::unique_ptr<RtpHeaderParser> parser_;
> 
>   RTC_DISALLOW_COPY_AND_ASSIGN(RtcEventLogSource);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_analyze.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_analyze.cc
>index 12721ccc96d..f9390381cd3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_analyze.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_analyze.cc
>@@ -20,10 +20,14 @@
> 
> // Define command line flags.
> DEFINE_int(red, 117, "RTP payload type for RED");
>-DEFINE_int(audio_level, -1, "Extension ID for audio level (RFC 6464); "
>-                            "-1 not to print audio level");
>-DEFINE_int(abs_send_time, -1, "Extension ID for absolute sender time; "
>-                             "-1 not to print absolute send time");
>+DEFINE_int(audio_level,
>+           -1,
>+           "Extension ID for audio level (RFC 6464); "
>+           "-1 not to print audio level");
>+DEFINE_int(abs_send_time,
>+           -1,
>+           "Extension ID for absolute sender time; "
>+           "-1 not to print absolute send time");
> DEFINE_bool(help, false, "Print this message");
> 
> int main(int argc, char* argv[]) {
>@@ -37,8 +41,8 @@ int main(int argc, char* argv[]) {
>       program_name + " input.rtp output.txt\n\n" +
>       "Output is sent to stdout if no output file is given. " +
>       "Note that this tool can read files with or without payloads.\n";
>-  if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) ||
>-      FLAG_help || (argc != 2 && argc != 3)) {
>+  if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) || FLAG_help ||
>+      (argc != 2 && argc != 3)) {
>     printf("%s", usage.c_str());
>     if (FLAG_help) {
>       rtc::FlagList::Print(nullptr, false);
>@@ -47,10 +51,11 @@ int main(int argc, char* argv[]) {
>     return 1;
>   }
> 
>-  RTC_CHECK(FLAG_red >= 0 && FLAG_red <= 127);  // Payload type
>-  RTC_CHECK(FLAG_audio_level == -1 ||  // Default
>-      (FLAG_audio_level > 0 && FLAG_audio_level <= 255));  // Extension ID
>-  RTC_CHECK(FLAG_abs_send_time == -1 ||  // Default
>+  RTC_CHECK(FLAG_red >= 0 && FLAG_red <= 127);                   // Payload type
>+  RTC_CHECK(FLAG_audio_level == -1 ||                            // Default
>+            (FLAG_audio_level > 0 && FLAG_audio_level <= 255));  // Extension ID
>+  RTC_CHECK(
>+      FLAG_abs_send_time == -1 ||                              // Default
>       (FLAG_abs_send_time > 0 && FLAG_abs_send_time <= 255));  // Extension ID
> 
>   printf("Input file: %s\n", argv[1]);
>@@ -104,19 +109,14 @@ int main(int argc, char* argv[]) {
>     }
>     // Write packet data to file. Use virtual_packet_length_bytes so that the
>     // correct packet sizes are printed also for RTP header-only dumps.
>-    fprintf(out_file,
>-            "%5u %10u %10u %5i %5i %2i %#08X",
>-            packet->header().sequenceNumber,
>-            packet->header().timestamp,
>+    fprintf(out_file, "%5u %10u %10u %5i %5i %2i %#08X",
>+            packet->header().sequenceNumber, packet->header().timestamp,
>             static_cast<unsigned int>(packet->time_ms()),
>             static_cast<int>(packet->virtual_packet_length_bytes()),
>-            packet->header().payloadType,
>-            packet->header().markerBit,
>+            packet->header().payloadType, packet->header().markerBit,
>             packet->header().ssrc);
>     if (print_audio_level && packet->header().extension.hasAudioLevel) {
>-      fprintf(out_file,
>-              " %5u (%1i)",
>-              packet->header().extension.audioLevel,
>+      fprintf(out_file, " %5u (%1i)", packet->header().extension.audioLevel,
>               packet->header().extension.voiceActivity);
>     }
>     if (print_abs_send_time && packet->header().extension.hasAbsoluteSendTime) {
>@@ -156,11 +156,8 @@ int main(int argc, char* argv[]) {
>       while (!red_headers.empty()) {
>         webrtc::RTPHeader* red = red_headers.front();
>         assert(red);
>-        fprintf(out_file,
>-                "* %5u %10u %10u %5i\n",
>-                red->sequenceNumber,
>-                red->timestamp,
>-                static_cast<unsigned int>(packet->time_ms()),
>+        fprintf(out_file, "* %5u %10u %10u %5i\n", red->sequenceNumber,
>+                red->timestamp, static_cast<unsigned int>(packet->time_ms()),
>                 red->payloadType);
>         red_headers.pop_front();
>         delete red;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_encode.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_encode.cc
>index ce0719980ff..5065ca16757 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_encode.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_encode.cc
>@@ -21,6 +21,8 @@
> #include <map>
> #include <string>
> 
>+#include "absl/memory/memory.h"
>+#include "api/audio/audio_frame.h"
> #include "api/audio_codecs/L16/audio_encoder_L16.h"
> #include "api/audio_codecs/g711/audio_encoder_g711.h"
> #include "api/audio_codecs/g722/audio_encoder_g722.h"
>@@ -32,8 +34,6 @@
> #include "modules/audio_coding/neteq/tools/input_audio_file.h"
> #include "rtc_base/flags.h"
> #include "rtc_base/numerics/safe_conversions.h"
>-#include "rtc_base/ptr_util.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> namespace test {
>@@ -246,11 +246,16 @@ AudioEncoderCng::Config GetCngConfig(int sample_rate_hz) {
>   AudioEncoderCng::Config cng_config;
>   const auto default_payload_type = [&] {
>     switch (sample_rate_hz) {
>-      case 8000: return 13;
>-      case 16000: return 98;
>-      case 32000: return 99;
>-      case 48000: return 100;
>-      default: RTC_NOTREACHED();
>+      case 8000:
>+        return 13;
>+      case 16000:
>+        return 98;
>+      case 32000:
>+        return 99;
>+      case 48000:
>+        return 100;
>+      default:
>+        RTC_NOTREACHED();
>     }
>     return 0;
>   };
>@@ -307,7 +312,7 @@ int RunRtpEncode(int argc, char* argv[]) {
>     AudioEncoderCng::Config cng_config = GetCngConfig(codec->SampleRateHz());
>     RTC_DCHECK(codec);
>     cng_config.speech_encoder = std::move(codec);
>-    codec = rtc::MakeUnique<AudioEncoderCng>(std::move(cng_config));
>+    codec = absl::make_unique<AudioEncoderCng>(std::move(cng_config));
>   }
>   RTC_DCHECK(codec);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
>index d1b1008d704..7bf7d52083c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
>@@ -12,9 +12,7 @@
> 
> #include <assert.h>
> #include <string.h>
>-#ifdef WIN32
>-#include <winsock2.h>
>-#else
>+#ifndef WIN32
> #include <netinet/in.h>
> #endif
> 
>@@ -46,8 +44,7 @@ bool RtpFileSource::ValidPcap(const std::string& file_name) {
>   return !!temp_file;
> }
> 
>-RtpFileSource::~RtpFileSource() {
>-}
>+RtpFileSource::~RtpFileSource() {}
> 
> bool RtpFileSource::RegisterRtpHeaderExtension(RTPExtensionType type,
>                                                uint8_t id) {
>@@ -84,8 +81,7 @@ std::unique_ptr<Packet> RtpFileSource::NextPacket() {
> }
> 
> RtpFileSource::RtpFileSource()
>-    : PacketSource(),
>-      parser_(RtpHeaderParser::Create()) {}
>+    : PacketSource(), parser_(RtpHeaderParser::Create()) {}
> 
> bool RtpFileSource::OpenFile(const std::string& file_name) {
>   rtp_reader_.reset(RtpFileReader::Create(RtpFileReader::kRtpDump, file_name));
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h
>index b44bc64d39a..013333b89a3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h
>@@ -39,7 +39,7 @@ class RtpFileSource : public PacketSource {
>   static bool ValidRtpDump(const std::string& file_name);
>   static bool ValidPcap(const std::string& file_name);
> 
>-  virtual ~RtpFileSource();
>+  ~RtpFileSource() override;
> 
>   // Registers an RTP header extension and binds it to |id|.
>   virtual bool RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_generator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_generator.cc
>index cedd7aeb9e4..ab7acdce392 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_generator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_generator.cc
>@@ -32,8 +32,8 @@ uint32_t RtpGenerator::GetRtpHeader(uint8_t payload_type,
> 
>   uint32_t this_send_time = next_send_time_ms_;
>   assert(samples_per_ms_ > 0);
>-  next_send_time_ms_ += ((1.0 + drift_factor_) * payload_length_samples) /
>-      samples_per_ms_;
>+  next_send_time_ms_ +=
>+      ((1.0 + drift_factor_) * payload_length_samples) / samples_per_ms_;
>   return this_send_time;
> }
> 
>@@ -46,8 +46,8 @@ void RtpGenerator::set_drift_factor(double factor) {
> uint32_t TimestampJumpRtpGenerator::GetRtpHeader(uint8_t payload_type,
>                                                  size_t payload_length_samples,
>                                                  RTPHeader* rtp_header) {
>-  uint32_t ret = RtpGenerator::GetRtpHeader(
>-      payload_type, payload_length_samples, rtp_header);
>+  uint32_t ret = RtpGenerator::GetRtpHeader(payload_type,
>+                                            payload_length_samples, rtp_header);
>   if (timestamp_ - static_cast<uint32_t>(payload_length_samples) <=
>           jump_from_timestamp_ &&
>       timestamp_ > jump_from_timestamp_) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_generator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_generator.h
>index 3b3cca939e9..ca7b04fc8cf 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_generator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_generator.h
>@@ -14,7 +14,6 @@
> #include "api/rtp_headers.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "rtc_base/constructormagic.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> namespace test {
>@@ -32,8 +31,7 @@ class RtpGenerator {
>         next_send_time_ms_(start_send_time_ms),
>         ssrc_(ssrc),
>         samples_per_ms_(samples_per_ms),
>-        drift_factor_(0.0) {
>-  }
>+        drift_factor_(0.0) {}
> 
>   virtual ~RtpGenerator() {}
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_jitter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_jitter.cc
>index d92fed06cf5..3c49443036c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_jitter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_jitter.cc
>@@ -18,7 +18,6 @@
> #include "modules/rtp_rtcp/source/byte_io.h"
> #include "rtc_base/buffer.h"
> #include "rtc_base/flags.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> namespace test {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/APITest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/APITest.cc
>deleted file mode 100644
>index dbf5ad45341..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/APITest.cc
>+++ /dev/null
>@@ -1,1106 +0,0 @@
>-/*
>- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "modules/audio_coding/test/APITest.h"
>-
>-#include <ctype.h>
>-#include <stdio.h>
>-#include <stdlib.h>
>-#include <string.h>
>-
>-#include <iostream>
>-#include <ostream>
>-#include <string>
>-
>-#include "common_types.h"  // NOLINT(build/include)
>-#include "modules/audio_coding/codecs/audio_format_conversion.h"
>-#include "modules/audio_coding/test/utility.h"
>-#include "rtc_base/platform_thread.h"
>-#include "rtc_base/timeutils.h"
>-#include "system_wrappers/include/event_wrapper.h"
>-#include "test/gtest.h"
>-#include "test/testsupport/fileutils.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>-
>-namespace webrtc {
>-
>-#define TEST_DURATION_SEC 600
>-#define NUMBER_OF_SENDER_TESTS 6
>-#define MAX_FILE_NAME_LENGTH_BYTE 500
>-
>-void APITest::Wait(uint32_t waitLengthMs) {
>-  if (_randomTest) {
>-    return;
>-  } else {
>-    EventWrapper* myEvent = EventWrapper::Create();
>-    myEvent->Wait(waitLengthMs);
>-    delete myEvent;
>-    return;
>-  }
>-}
>-
>-APITest::APITest()
>-    : _acmA(AudioCodingModule::Create()),
>-      _acmB(AudioCodingModule::Create()),
>-      _channel_A2B(NULL),
>-      _channel_B2A(NULL),
>-      _writeToFile(true),
>-      _pullEventA(NULL),
>-      _pushEventA(NULL),
>-      _processEventA(NULL),
>-      _apiEventA(NULL),
>-      _pullEventB(NULL),
>-      _pushEventB(NULL),
>-      _processEventB(NULL),
>-      _apiEventB(NULL),
>-      _codecCntrA(0),
>-      _codecCntrB(0),
>-      _thereIsEncoderA(false),
>-      _thereIsEncoderB(false),
>-      _thereIsDecoderA(false),
>-      _thereIsDecoderB(false),
>-      _sendVADA(false),
>-      _sendDTXA(false),
>-      _sendVADModeA(VADNormal),
>-      _sendVADB(false),
>-      _sendDTXB(false),
>-      _sendVADModeB(VADNormal),
>-      _minDelayA(0),
>-      _minDelayB(0),
>-      _dotPositionA(0),
>-      _dotMoveDirectionA(1),
>-      _dotPositionB(39),
>-      _dotMoveDirectionB(-1),
>-      _vadCallbackA(NULL),
>-      _vadCallbackB(NULL),
>-      _apiTestRWLock(*RWLockWrapper::CreateRWLock()),
>-      _randomTest(false),
>-      _testNumA(0),
>-      _testNumB(1) {
>-  int n;
>-  for (n = 0; n < 32; n++) {
>-    _payloadUsed[n] = false;
>-  }
>-
>-  _movingDot[40] = '\0';
>-
>-  for (int n = 0; n < 40; n++) {
>-    _movingDot[n] = ' ';
>-  }
>-}
>-
>-APITest::~APITest() {
>-  DELETE_POINTER(_channel_A2B);
>-  DELETE_POINTER(_channel_B2A);
>-
>-  DELETE_POINTER(_pushEventA);
>-  DELETE_POINTER(_pullEventA);
>-  DELETE_POINTER(_processEventA);
>-  DELETE_POINTER(_apiEventA);
>-
>-  DELETE_POINTER(_pushEventB);
>-  DELETE_POINTER(_pullEventB);
>-  DELETE_POINTER(_processEventB);
>-  DELETE_POINTER(_apiEventB);
>-
>-  _inFileA.Close();
>-  _outFileA.Close();
>-
>-  _inFileB.Close();
>-  _outFileB.Close();
>-
>-  DELETE_POINTER(_vadCallbackA);
>-  DELETE_POINTER(_vadCallbackB);
>-
>-  delete &_apiTestRWLock;
>-}
>-
>-int16_t APITest::SetUp() {
>-  CodecInst dummyCodec;
>-  int lastPayloadType = 0;
>-
>-  int16_t numCodecs = _acmA->NumberOfCodecs();
>-  for (uint8_t n = 0; n < numCodecs; n++) {
>-    AudioCodingModule::Codec(n, &dummyCodec);
>-    if ((STR_CASE_CMP(dummyCodec.plname, "CN") == 0)
>-        && (dummyCodec.plfreq == 32000)) {
>-      continue;
>-    }
>-
>-    printf("Register Receive Codec %s  ", dummyCodec.plname);
>-
>-    if ((n != 0) && !FixedPayloadTypeCodec(dummyCodec.plname)) {
>-      // Check registration with an already occupied payload type
>-      int currentPayloadType = dummyCodec.pltype;
>-      dummyCodec.pltype = 97;  //lastPayloadType;
>-      EXPECT_EQ(true, _acmB->RegisterReceiveCodec(dummyCodec.pltype,
>-                                                  CodecInstToSdp(dummyCodec)));
>-      dummyCodec.pltype = currentPayloadType;
>-    }
>-
>-    if ((n < numCodecs - 1) && !FixedPayloadTypeCodec(dummyCodec.plname)) {
>-      // test if re-registration works;
>-      CodecInst nextCodec;
>-      int currentPayloadType = dummyCodec.pltype;
>-      AudioCodingModule::Codec(n + 1, &nextCodec);
>-      dummyCodec.pltype = nextCodec.pltype;
>-      if (!FixedPayloadTypeCodec(nextCodec.plname)) {
>-        _acmB->RegisterReceiveCodec(dummyCodec.pltype,
>-                                    CodecInstToSdp(dummyCodec));
>-      }
>-      dummyCodec.pltype = currentPayloadType;
>-    }
>-
>-    if ((n < numCodecs - 1) && !FixedPayloadTypeCodec(dummyCodec.plname)) {
>-      // test if un-registration works;
>-      CodecInst nextCodec;
>-      AudioCodingModule::Codec(n + 1, &nextCodec);
>-      nextCodec.pltype = dummyCodec.pltype;
>-      if (!FixedPayloadTypeCodec(nextCodec.plname)) {
>-        EXPECT_EQ(true, _acmA->RegisterReceiveCodec(nextCodec.pltype,
>-                                                    CodecInstToSdp(nextCodec)));
>-        CHECK_ERROR_MT(_acmA->UnregisterReceiveCodec(nextCodec.pltype));
>-      }
>-    }
>-
>-    EXPECT_EQ(true, _acmA->RegisterReceiveCodec(dummyCodec.pltype,
>-                                                CodecInstToSdp(dummyCodec)));
>-    printf("   side A done!");
>-    EXPECT_EQ(true, _acmB->RegisterReceiveCodec(dummyCodec.pltype,
>-                                                CodecInstToSdp(dummyCodec)));
>-    printf("   side B done!\n");
>-
>-    if (!strcmp(dummyCodec.plname, "CN")) {
>-      CHECK_ERROR_MT(_acmA->RegisterSendCodec(dummyCodec));
>-      CHECK_ERROR_MT(_acmB->RegisterSendCodec(dummyCodec));
>-    }
>-    lastPayloadType = dummyCodec.pltype;
>-    if ((lastPayloadType >= 96) && (lastPayloadType <= 127)) {
>-      _payloadUsed[lastPayloadType - 96] = true;
>-    }
>-  }
>-  _thereIsDecoderA = true;
>-  _thereIsDecoderB = true;
>-
>-  // Register Send Codec
>-  AudioCodingModule::Codec((uint8_t) _codecCntrA, &dummyCodec);
>-  CHECK_ERROR_MT(_acmA->RegisterSendCodec(dummyCodec));
>-  _thereIsEncoderA = true;
>-  //
>-  AudioCodingModule::Codec((uint8_t) _codecCntrB, &dummyCodec);
>-  CHECK_ERROR_MT(_acmB->RegisterSendCodec(dummyCodec));
>-  _thereIsEncoderB = true;
>-
>-  uint16_t frequencyHz;
>-
>-  printf("\n\nAPI Test\n");
>-  printf("========\n");
>-  printf("Hit enter to accept the default values indicated in []\n\n");
>-
>-  //--- Input A
>-  std::string file_name = webrtc::test::ResourcePath(
>-      "audio_coding/testfile32kHz", "pcm");
>-  frequencyHz = 32000;
>-  printf("Enter input file at side A [%s]: ", file_name.c_str());
>-  PCMFile::ChooseFile(&file_name, 499, &frequencyHz);
>-  _inFileA.Open(file_name, frequencyHz, "rb", true);
>-
>-  //--- Output A
>-  std::string out_file_a = webrtc::test::OutputPath() + "outA.pcm";
>-  printf("Enter output file at side A [%s]: ", out_file_a.c_str());
>-  PCMFile::ChooseFile(&out_file_a, 499, &frequencyHz);
>-  _outFileA.Open(out_file_a, frequencyHz, "wb");
>-
>-  //--- Input B
>-  file_name = webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
>-  printf("\n\nEnter input file at side B [%s]: ", file_name.c_str());
>-  PCMFile::ChooseFile(&file_name, 499, &frequencyHz);
>-  _inFileB.Open(file_name, frequencyHz, "rb", true);
>-
>-  //--- Output B
>-  std::string out_file_b = webrtc::test::OutputPath() + "outB.pcm";
>-  printf("Enter output file at side B [%s]: ", out_file_b.c_str());
>-  PCMFile::ChooseFile(&out_file_b, 499, &frequencyHz);
>-  _outFileB.Open(out_file_b, frequencyHz, "wb");
>-
>-  //--- Set A-to-B channel
>-  _channel_A2B = new Channel(2);
>-  CHECK_ERROR_MT(_acmA->RegisterTransportCallback(_channel_A2B));
>-  _channel_A2B->RegisterReceiverACM(_acmB.get());
>-
>-  //--- Set B-to-A channel
>-  _channel_B2A = new Channel(1);
>-  CHECK_ERROR_MT(_acmB->RegisterTransportCallback(_channel_B2A));
>-  _channel_B2A->RegisterReceiverACM(_acmA.get());
>-
>-  //--- EVENT TIMERS
>-  // A
>-  _pullEventA = EventTimerWrapper::Create();
>-  _pushEventA = EventTimerWrapper::Create();
>-  _processEventA = EventTimerWrapper::Create();
>-  _apiEventA = EventWrapper::Create();
>-  // B
>-  _pullEventB = EventTimerWrapper::Create();
>-  _pushEventB = EventTimerWrapper::Create();
>-  _processEventB = EventTimerWrapper::Create();
>-  _apiEventB = EventWrapper::Create();
>-
>-  //--- I/O params
>-  // A
>-  _outFreqHzA = _outFileA.SamplingFrequency();
>-  // B
>-  _outFreqHzB = _outFileB.SamplingFrequency();
>-
>-  char print[11];
>-
>-  printf("\nRandom Test (y/n)?");
>-  EXPECT_TRUE(fgets(print, 10, stdin) != NULL);
>-  print[10] = '\0';
>-  if (strstr(print, "y") != NULL) {
>-    _randomTest = true;
>-    _verbose = false;
>-    _writeToFile = false;
>-  } else {
>-    _randomTest = false;
>-    printf("\nPrint Tests (y/n)? ");
>-    EXPECT_TRUE(fgets(print, 10, stdin) != NULL);
>-    print[10] = '\0';
>-    if (strstr(print, "y") == NULL) {
>-      EXPECT_TRUE(freopen("APITest_log.txt", "w", stdout) != 0);
>-      _verbose = false;
>-    }
>-  }
>-
>-  _vadCallbackA = new VADCallback;
>-  _vadCallbackB = new VADCallback;
>-
>-  return 0;
>-}
>-
>-bool APITest::PushAudioThreadA(void* obj) {
>-  return static_cast<APITest*>(obj)->PushAudioRunA();
>-}
>-
>-bool APITest::PushAudioThreadB(void* obj) {
>-  return static_cast<APITest*>(obj)->PushAudioRunB();
>-}
>-
>-bool APITest::PullAudioThreadA(void* obj) {
>-  return static_cast<APITest*>(obj)->PullAudioRunA();
>-}
>-
>-bool APITest::PullAudioThreadB(void* obj) {
>-  return static_cast<APITest*>(obj)->PullAudioRunB();
>-}
>-
>-bool APITest::ProcessThreadA(void* obj) {
>-  return static_cast<APITest*>(obj)->ProcessRunA();
>-}
>-
>-bool APITest::ProcessThreadB(void* obj) {
>-  return static_cast<APITest*>(obj)->ProcessRunB();
>-}
>-
>-bool APITest::APIThreadA(void* obj) {
>-  return static_cast<APITest*>(obj)->APIRunA();
>-}
>-
>-bool APITest::APIThreadB(void* obj) {
>-  return static_cast<APITest*>(obj)->APIRunB();
>-}
>-
>-bool APITest::PullAudioRunA() {
>-  _pullEventA->Wait(100);
>-  AudioFrame audioFrame;
>-  bool muted;
>-  if (_acmA->PlayoutData10Ms(_outFreqHzA, &audioFrame, &muted) < 0) {
>-    bool thereIsDecoder;
>-    {
>-      ReadLockScoped rl(_apiTestRWLock);
>-      thereIsDecoder = _thereIsDecoderA;
>-    }
>-    if (thereIsDecoder) {
>-      fprintf(stderr, "\n>>>>>>    cannot pull audio A       <<<<<<<< \n");
>-    }
>-  } else {
>-    if (_writeToFile) {
>-      _outFileA.Write10MsData(audioFrame);
>-    }
>-  }
>-  return true;
>-}
>-
>-bool APITest::PullAudioRunB() {
>-  _pullEventB->Wait(100);
>-  AudioFrame audioFrame;
>-  bool muted;
>-  if (_acmB->PlayoutData10Ms(_outFreqHzB, &audioFrame, &muted) < 0) {
>-    bool thereIsDecoder;
>-    {
>-      ReadLockScoped rl(_apiTestRWLock);
>-      thereIsDecoder = _thereIsDecoderB;
>-    }
>-    if (thereIsDecoder) {
>-      fprintf(stderr, "\n>>>>>>    cannot pull audio B       <<<<<<<< \n");
>-      fprintf(stderr, "%d %d\n", _testNumA, _testNumB);
>-    }
>-  } else {
>-    if (_writeToFile) {
>-      _outFileB.Write10MsData(audioFrame);
>-    }
>-  }
>-  return true;
>-}
>-
>-bool APITest::PushAudioRunA() {
>-  _pushEventA->Wait(100);
>-  AudioFrame audioFrame;
>-  _inFileA.Read10MsData(audioFrame);
>-  if (_acmA->Add10MsData(audioFrame) < 0) {
>-    bool thereIsEncoder;
>-    {
>-      ReadLockScoped rl(_apiTestRWLock);
>-      thereIsEncoder = _thereIsEncoderA;
>-    }
>-    if (thereIsEncoder) {
>-      fprintf(stderr, "\n>>>>        add10MsData at A failed       <<<<\n");
>-    }
>-  }
>-  return true;
>-}
>-
>-bool APITest::PushAudioRunB() {
>-  _pushEventB->Wait(100);
>-  AudioFrame audioFrame;
>-  _inFileB.Read10MsData(audioFrame);
>-  if (_acmB->Add10MsData(audioFrame) < 0) {
>-    bool thereIsEncoder;
>-    {
>-      ReadLockScoped rl(_apiTestRWLock);
>-      thereIsEncoder = _thereIsEncoderB;
>-    }
>-
>-    if (thereIsEncoder) {
>-      fprintf(stderr, "\n>>>>   cannot add audio to B    <<<<");
>-    }
>-  }
>-
>-  return true;
>-}
>-
>-bool APITest::ProcessRunA() {
>-  _processEventA->Wait(100);
>-  return true;
>-}
>-
>-bool APITest::ProcessRunB() {
>-  _processEventB->Wait(100);
>-  return true;
>-}
>-
>-/*/
>- *
>- * In side A we test the APIs which are related to sender Side.
>- *
>-/*/
>-
>-void APITest::RunTest(char thread) {
>-  int testNum;
>-  {
>-    WriteLockScoped cs(_apiTestRWLock);
>-    if (thread == 'A') {
>-      _testNumA = (_testNumB + 1 + (rand() % 3)) % 4;
>-      testNum = _testNumA;
>-
>-      _movingDot[_dotPositionA] = ' ';
>-      if (_dotPositionA == 0) {
>-        _dotMoveDirectionA = 1;
>-      }
>-      if (_dotPositionA == 19) {
>-        _dotMoveDirectionA = -1;
>-      }
>-      _dotPositionA += _dotMoveDirectionA;
>-      _movingDot[_dotPositionA] = (_dotMoveDirectionA > 0) ? '>' : '<';
>-    } else {
>-      _testNumB = (_testNumA + 1 + (rand() % 3)) % 4;
>-      testNum = _testNumB;
>-
>-      _movingDot[_dotPositionB] = ' ';
>-      if (_dotPositionB == 20) {
>-        _dotMoveDirectionB = 1;
>-      }
>-      if (_dotPositionB == 39) {
>-        _dotMoveDirectionB = -1;
>-      }
>-      _dotPositionB += _dotMoveDirectionB;
>-      _movingDot[_dotPositionB] = (_dotMoveDirectionB > 0) ? '>' : '<';
>-    }
>-    //fprintf(stderr, "%c: %d \n", thread, testNum);
>-    //fflush(stderr);
>-  }
>-  switch (testNum) {
>-    case 0:
>-      CurrentCodec('A');
>-      ChangeCodec('A');
>-      break;
>-    case 1:
>-      if (!_randomTest) {
>-        fprintf(stdout, "\nTesting Delay ...\n");
>-      }
>-      TestDelay('A');
>-      break;
>-    case 2:
>-      TestSendVAD('A');
>-      break;
>-    case 3:
>-      TestRegisteration('A');
>-      break;
>-    default:
>-      fprintf(stderr, "Wrong Test Number\n");
>-      getc(stdin);
>-      exit(1);
>-  }
>-}
>-
>-bool APITest::APIRunA() {
>-  _apiEventA->Wait(50);
>-
>-  bool randomTest;
>-  {
>-    ReadLockScoped rl(_apiTestRWLock);
>-    randomTest = _randomTest;
>-  }
>-  if (randomTest) {
>-    RunTest('A');
>-  } else {
>-    CurrentCodec('A');
>-    ChangeCodec('A');
>-    if (_codecCntrA == 0) {
>-      fprintf(stdout, "\nTesting Delay ...\n");
>-      TestDelay('A');
>-    }
>-    // VAD TEST
>-    TestSendVAD('A');
>-    TestRegisteration('A');
>-  }
>-  return true;
>-}
>-
>-bool APITest::APIRunB() {
>-  _apiEventB->Wait(50);
>-  bool randomTest;
>-  {
>-    ReadLockScoped rl(_apiTestRWLock);
>-    randomTest = _randomTest;
>-  }
>-  //_apiEventB->Wait(2000);
>-  if (randomTest) {
>-    RunTest('B');
>-  }
>-
>-  return true;
>-}
>-
>-void APITest::Perform() {
>-  SetUp();
>-
>-  //--- THREADS
>-  // A
>-  // PUSH
>-  rtc::PlatformThread myPushAudioThreadA(PushAudioThreadA, this,
>-                                         "PushAudioThreadA");
>-  myPushAudioThreadA.Start();
>-  // PULL
>-  rtc::PlatformThread myPullAudioThreadA(PullAudioThreadA, this,
>-                                         "PullAudioThreadA");
>-  myPullAudioThreadA.Start();
>-  // Process
>-  rtc::PlatformThread myProcessThreadA(ProcessThreadA, this, "ProcessThreadA");
>-  myProcessThreadA.Start();
>-  // API
>-  rtc::PlatformThread myAPIThreadA(APIThreadA, this, "APIThreadA");
>-  myAPIThreadA.Start();
>-  // B
>-  // PUSH
>-  rtc::PlatformThread myPushAudioThreadB(PushAudioThreadB, this,
>-                                         "PushAudioThreadB");
>-  myPushAudioThreadB.Start();
>-  // PULL
>-  rtc::PlatformThread myPullAudioThreadB(PullAudioThreadB, this,
>-                                         "PullAudioThreadB");
>-  myPullAudioThreadB.Start();
>-  // Process
>-  rtc::PlatformThread myProcessThreadB(ProcessThreadB, this, "ProcessThreadB");
>-  myProcessThreadB.Start();
>-  // API
>-  rtc::PlatformThread myAPIThreadB(APIThreadB, this, "APIThreadB");
>-  myAPIThreadB.Start();
>-
>-  //_apiEventA->StartTimer(true, 5000);
>-  //_apiEventB->StartTimer(true, 5000);
>-
>-  _processEventA->StartTimer(true, 10);
>-  _processEventB->StartTimer(true, 10);
>-
>-  _pullEventA->StartTimer(true, 10);
>-  _pullEventB->StartTimer(true, 10);
>-
>-  _pushEventA->StartTimer(true, 10);
>-  _pushEventB->StartTimer(true, 10);
>-
>-  // Keep main thread waiting for sender/receiver
>-  // threads to complete
>-  EventWrapper* completeEvent = EventWrapper::Create();
>-  uint64_t startTime = rtc::TimeMillis();
>-  uint64_t currentTime;
>-  // Run test in 2 minutes (120000 ms).
>-  do {
>-    {
>-      //ReadLockScoped rl(_apiTestRWLock);
>-      //fprintf(stderr, "\r%s", _movingDot);
>-    }
>-    //fflush(stderr);
>-    completeEvent->Wait(50);
>-    currentTime = rtc::TimeMillis();
>-  } while ((currentTime - startTime) < 120000);
>-
>-  //completeEvent->Wait(0xFFFFFFFF);
>-  //(unsigned long)((unsigned long)TEST_DURATION_SEC * (unsigned long)1000));
>-  delete completeEvent;
>-
>-  myPushAudioThreadA.Stop();
>-  myPullAudioThreadA.Stop();
>-  myProcessThreadA.Stop();
>-  myAPIThreadA.Stop();
>-
>-  myPushAudioThreadB.Stop();
>-  myPullAudioThreadB.Stop();
>-  myProcessThreadB.Stop();
>-  myAPIThreadB.Stop();
>-}
>-
>-void APITest::CheckVADStatus(char side) {
>-
>-  bool dtxEnabled;
>-  bool vadEnabled;
>-  ACMVADMode vadMode;
>-
>-  if (side == 'A') {
>-    _acmA->VAD(&dtxEnabled, &vadEnabled, &vadMode);
>-    _acmA->RegisterVADCallback(NULL);
>-    _vadCallbackA->Reset();
>-    _acmA->RegisterVADCallback(_vadCallbackA);
>-
>-    if (!_randomTest) {
>-      if (_verbose) {
>-        fprintf(stdout, "DTX %3s, VAD %3s, Mode %d", dtxEnabled ? "ON" : "OFF",
>-                vadEnabled ? "ON" : "OFF", (int) vadMode);
>-        Wait(5000);
>-        fprintf(stdout, " => bit-rate %3.0f kbps\n", _channel_A2B->BitRate());
>-      } else {
>-        Wait(5000);
>-        fprintf(stdout, "DTX %3s, VAD %3s, Mode %d => bit-rate %3.0f kbps\n",
>-                dtxEnabled ? "ON" : "OFF", vadEnabled ? "ON" : "OFF",
>-                (int) vadMode, _channel_A2B->BitRate());
>-      }
>-      _vadCallbackA->PrintFrameTypes();
>-    }
>-
>-    if (dtxEnabled != _sendDTXA) {
>-      fprintf(stderr, ">>>   Error Enabling DTX    <<<\n");
>-    }
>-    if ((vadEnabled != _sendVADA) && (!dtxEnabled)) {
>-      fprintf(stderr, ">>>   Error Enabling VAD    <<<\n");
>-    }
>-    if ((vadMode != _sendVADModeA) && vadEnabled) {
>-      fprintf(stderr, ">>>   Error setting VAD-mode    <<<\n");
>-    }
>-  } else {
>-    _acmB->VAD(&dtxEnabled, &vadEnabled, &vadMode);
>-
>-    _acmB->RegisterVADCallback(NULL);
>-    _vadCallbackB->Reset();
>-    _acmB->RegisterVADCallback(_vadCallbackB);
>-
>-    if (!_randomTest) {
>-      if (_verbose) {
>-        fprintf(stdout, "DTX %3s, VAD %3s, Mode %d", dtxEnabled ? "ON" : "OFF",
>-                vadEnabled ? "ON" : "OFF", (int) vadMode);
>-        Wait(5000);
>-        fprintf(stdout, " => bit-rate %3.0f kbps\n", _channel_B2A->BitRate());
>-      } else {
>-        Wait(5000);
>-        fprintf(stdout, "DTX %3s, VAD %3s, Mode %d => bit-rate %3.0f kbps\n",
>-                dtxEnabled ? "ON" : "OFF", vadEnabled ? "ON" : "OFF",
>-                (int) vadMode, _channel_B2A->BitRate());
>-      }
>-      _vadCallbackB->PrintFrameTypes();
>-    }
>-
>-    if (dtxEnabled != _sendDTXB) {
>-      fprintf(stderr, ">>>   Error Enabling DTX    <<<\n");
>-    }
>-    if ((vadEnabled != _sendVADB) && (!dtxEnabled)) {
>-      fprintf(stderr, ">>>   Error Enabling VAD    <<<\n");
>-    }
>-    if ((vadMode != _sendVADModeB) && vadEnabled) {
>-      fprintf(stderr, ">>>   Error setting VAD-mode    <<<\n");
>-    }
>-  }
>-}
>-
>-// Set Min delay, get delay, playout timestamp
>-void APITest::TestDelay(char side) {
>-  AudioCodingModule* myACM;
>-  Channel* myChannel;
>-  int32_t* myMinDelay;
>-  EventTimerWrapper* myEvent = EventTimerWrapper::Create();
>-
>-  uint32_t inTimestamp = 0;
>-  double estimDelay = 0;
>-
>-  double averageEstimDelay = 0;
>-  double averageDelay = 0;
>-
>-  test::CircularBuffer estimDelayCB(100);
>-  estimDelayCB.SetArithMean(true);
>-
>-  if (side == 'A') {
>-    myACM = _acmA.get();
>-    myChannel = _channel_B2A;
>-    myMinDelay = &_minDelayA;
>-  } else {
>-    myACM = _acmB.get();
>-    myChannel = _channel_A2B;
>-    myMinDelay = &_minDelayB;
>-  }
>-
>-  CHECK_ERROR_MT(myACM->SetMinimumPlayoutDelay(*myMinDelay));
>-
>-  inTimestamp = myChannel->LastInTimestamp();
>-  rtc::Optional<uint32_t> outTimestamp = myACM->PlayoutTimestamp();
>-  CHECK_ERROR_MT(outTimestamp ? 0 : -1);
>-
>-  if (!_randomTest) {
>-    myEvent->StartTimer(true, 30);
>-    int n = 0;
>-    int settlePoint = 5000;
>-    while (n < settlePoint + 400) {
>-      myEvent->Wait(1000);
>-
>-      inTimestamp = myChannel->LastInTimestamp();
>-      outTimestamp = myACM->PlayoutTimestamp();
>-      CHECK_ERROR_MT(outTimestamp ? 0 : -1);
>-
>-      //std::cout << outTimestamp << std::endl << std::flush;
>-      estimDelay = (double)((uint32_t)(inTimestamp - *outTimestamp)) /
>-                   ((double)myACM->ReceiveFrequency() / 1000.0);
>-
>-      estimDelayCB.Update(estimDelay);
>-
>-      estimDelayCB.ArithMean(averageEstimDelay);
>-      //printf("\n %6.1f \n", estimDelay);
>-      //std::cout << " " << std::flush;
>-
>-      if (_verbose) {
>-        fprintf(stdout,
>-                "\rExpected: %4d,    retreived: %6.1f,   measured: %6.1f",
>-                *myMinDelay, averageDelay, averageEstimDelay);
>-        std::cout << " " << std::flush;
>-      }
>-      if ((averageDelay > *myMinDelay) && (n < settlePoint)) {
>-        settlePoint = n;
>-      }
>-      n++;
>-    }
>-    myEvent->StopTimer();
>-  }
>-
>-  if ((!_verbose) && (!_randomTest)) {
>-    fprintf(stdout, "\nExpected: %4d,    retreived: %6.1f,   measured: %6.1f",
>-            *myMinDelay, averageDelay, averageEstimDelay);
>-  }
>-
>-  *myMinDelay = (rand() % 1000) + 1;
>-
>-  NetworkStatistics networkStat;
>-  CHECK_ERROR_MT(myACM->GetNetworkStatistics(&networkStat));
>-
>-  if (!_randomTest) {
>-    fprintf(stdout, "\n\nJitter Statistics at Side %c\n", side);
>-    fprintf(stdout, "--------------------------------------\n");
>-    fprintf(stdout, "buffer-size............. %d\n",
>-            networkStat.currentBufferSize);
>-    fprintf(stdout, "Preferred buffer-size... %d\n",
>-            networkStat.preferredBufferSize);
>-    fprintf(stdout, "Peaky jitter mode........%d\n",
>-            networkStat.jitterPeaksFound);
>-    fprintf(stdout, "packet-size rate........ %d\n",
>-            networkStat.currentPacketLossRate);
>-    fprintf(stdout, "expand rate............. %d\n",
>-            networkStat.currentExpandRate);
>-    fprintf(stdout, "speech expand rate...... %d\n",
>-            networkStat.currentSpeechExpandRate);
>-    fprintf(stdout, "Preemptive rate......... %d\n",
>-            networkStat.currentPreemptiveRate);
>-    fprintf(stdout, "Accelerate rate......... %d\n",
>-            networkStat.currentAccelerateRate);
>-    fprintf(stdout, "Secondary decoded rate.. %d\n",
>-            networkStat.currentSecondaryDecodedRate);
>-    fprintf(stdout, "Secondary discarded rate.%d\n",
>-            networkStat.currentSecondaryDiscardedRate);
>-    fprintf(stdout, "Clock-drift............. %d\n", networkStat.clockDriftPPM);
>-    fprintf(stdout, "Mean waiting time....... %d\n",
>-            networkStat.meanWaitingTimeMs);
>-    fprintf(stdout, "Median waiting time..... %d\n",
>-            networkStat.medianWaitingTimeMs);
>-    fprintf(stdout, "Min waiting time........ %d\n",
>-            networkStat.minWaitingTimeMs);
>-    fprintf(stdout, "Max waiting time........ %d\n",
>-            networkStat.maxWaitingTimeMs);
>-  }
>-
>-  CHECK_ERROR_MT(myACM->SetMinimumPlayoutDelay(*myMinDelay));
>-
>-  if (!_randomTest) {
>-    myEvent->Wait(500);
>-    fprintf(stdout, "\n");
>-    fprintf(stdout, "\n");
>-  }
>-  delete myEvent;
>-}
>-
>-// Unregister a codec & register again.
>-void APITest::TestRegisteration(char sendSide) {
>-  AudioCodingModule* sendACM;
>-  AudioCodingModule* receiveACM;
>-  bool* thereIsDecoder;
>-  EventWrapper* myEvent = EventWrapper::Create();
>-
>-  if (!_randomTest) {
>-    fprintf(stdout, "\n\n");
>-    fprintf(stdout,
>-            "---------------------------------------------------------\n");
>-    fprintf(stdout, "           Unregister/register Receive Codec\n");
>-    fprintf(stdout,
>-            "---------------------------------------------------------\n");
>-  }
>-
>-  switch (sendSide) {
>-    case 'A': {
>-      sendACM = _acmA.get();
>-      receiveACM = _acmB.get();
>-      thereIsDecoder = &_thereIsDecoderB;
>-      break;
>-    }
>-    case 'B': {
>-      sendACM = _acmB.get();
>-      receiveACM = _acmA.get();
>-      thereIsDecoder = &_thereIsDecoderA;
>-      break;
>-    }
>-    default:
>-      fprintf(stderr, "Invalid sender-side in TestRegistration(%c)\n",
>-              sendSide);
>-      exit(-1);
>-  }
>-
>-  auto myCodec = sendACM->SendCodec();
>-  if (!myCodec) {
>-    CodecInst ci;
>-    AudioCodingModule::Codec(_codecCntrA, &ci);
>-    myCodec = ci;
>-  }
>-
>-  if (!_randomTest) {
>-    fprintf(stdout, "Unregistering reveive codec, NO AUDIO.\n");
>-    fflush (stdout);
>-  }
>-  {
>-    WriteLockScoped wl(_apiTestRWLock);
>-    *thereIsDecoder = false;
>-  }
>-  //myEvent->Wait(20);
>-  CHECK_ERROR_MT(receiveACM->UnregisterReceiveCodec(myCodec->pltype));
>-  Wait(1000);
>-
>-  int currentPayload = myCodec->pltype;
>-
>-  if (!FixedPayloadTypeCodec(myCodec->plname)) {
>-    int32_t i;
>-    for (i = 0; i < 32; i++) {
>-      if (!_payloadUsed[i]) {
>-        if (!_randomTest) {
>-          fprintf(stdout,
>-                  "Register receive codec with new Payload, AUDIO BACK.\n");
>-        }
>-        //myCodec->pltype = i + 96;
>-        //CHECK_ERROR_MT(receiveACM->RegisterReceiveCodec(*myCodec));
>-        //CHECK_ERROR_MT(sendACM->RegisterSendCodec(*myCodec));
>-        //myEvent->Wait(20);
>-        //{
>-        //    WriteLockScoped wl(_apiTestRWLock);
>-        //    *thereIsDecoder = true;
>-        //}
>-        Wait(1000);
>-
>-        if (!_randomTest) {
>-          fprintf(stdout, "Unregistering reveive codec, NO AUDIO.\n");
>-        }
>-        //{
>-        //    WriteLockScoped wl(_apiTestRWLock);
>-        //    *thereIsDecoder = false;
>-        //}
>-        //myEvent->Wait(20);
>-        //CHECK_ERROR_MT(receiveACM->UnregisterReceiveCodec(myCodec->pltype));
>-        Wait(1000);
>-
>-        myCodec->pltype = currentPayload;
>-        if (!_randomTest) {
>-          fprintf(stdout,
>-                  "Register receive codec with default Payload, AUDIO BACK.\n");
>-          fflush (stdout);
>-        }
>-        EXPECT_EQ(true, receiveACM->RegisterReceiveCodec(
>-                            myCodec->pltype, CodecInstToSdp(*myCodec)));
>-        //CHECK_ERROR_MT(sendACM->RegisterSendCodec(*myCodec));
>-        myEvent->Wait(20);
>-        {
>-          WriteLockScoped wl(_apiTestRWLock);
>-          *thereIsDecoder = true;
>-        }
>-        Wait(1000);
>-
>-        break;
>-      }
>-    }
>-    if (i == 32) {
>-      EXPECT_EQ(true, receiveACM->RegisterReceiveCodec(
>-                          myCodec->pltype, CodecInstToSdp(*myCodec)));
>-      {
>-        WriteLockScoped wl(_apiTestRWLock);
>-        *thereIsDecoder = true;
>-      }
>-    }
>-  } else {
>-    if (!_randomTest) {
>-      fprintf(stdout,
>-              "Register receive codec with fixed Payload, AUDIO BACK.\n");
>-      fflush (stdout);
>-    }
>-    EXPECT_EQ(true, receiveACM->RegisterReceiveCodec(myCodec->pltype,
>-                                                     CodecInstToSdp(*myCodec)));
>-    //CHECK_ERROR_MT(receiveACM->UnregisterReceiveCodec(myCodec->pltype));
>-    //CHECK_ERROR_MT(receiveACM->RegisterReceiveCodec(*myCodec));
>-    myEvent->Wait(20);
>-    {
>-      WriteLockScoped wl(_apiTestRWLock);
>-      *thereIsDecoder = true;
>-    }
>-  }
>-  delete myEvent;
>-  if (!_randomTest) {
>-    fprintf(stdout,
>-            "---------------------------------------------------------\n");
>-  }
>-}
>-
>-void APITest::TestSendVAD(char side) {
>-  if (_randomTest) {
>-    return;
>-  }
>-
>-  bool* vad;
>-  bool* dtx;
>-  ACMVADMode* mode;
>-  Channel* myChannel;
>-  AudioCodingModule* myACM;
>-
>-  CodecInst myCodec;
>-  if (!_randomTest) {
>-    fprintf(stdout, "\n\n");
>-    fprintf(stdout, "-----------------------------------------------\n");
>-    fprintf(stdout, "                Test VAD API\n");
>-    fprintf(stdout, "-----------------------------------------------\n");
>-  }
>-
>-  if (side == 'A') {
>-    AudioCodingModule::Codec(_codecCntrA, &myCodec);
>-    vad = &_sendVADA;
>-    dtx = &_sendDTXA;
>-    mode = &_sendVADModeA;
>-    myChannel = _channel_A2B;
>-    myACM = _acmA.get();
>-  } else {
>-    AudioCodingModule::Codec(_codecCntrB, &myCodec);
>-    vad = &_sendVADB;
>-    dtx = &_sendDTXB;
>-    mode = &_sendVADModeB;
>-    myChannel = _channel_B2A;
>-    myACM = _acmB.get();
>-  }
>-
>-  CheckVADStatus(side);
>-  if (!_randomTest) {
>-    fprintf(stdout, "\n\n");
>-  }
>-
>-  switch (*mode) {
>-    case VADNormal:
>-      *vad = true;
>-      *dtx = true;
>-      *mode = VADAggr;
>-      break;
>-    case VADLowBitrate:
>-      *vad = true;
>-      *dtx = true;
>-      *mode = VADVeryAggr;
>-      break;
>-    case VADAggr:
>-      *vad = true;
>-      *dtx = true;
>-      *mode = VADLowBitrate;
>-      break;
>-    case VADVeryAggr:
>-      *vad = false;
>-      *dtx = false;
>-      *mode = VADNormal;
>-      break;
>-    default:
>-      *mode = VADNormal;
>-  }
>-
>-  *dtx = (myCodec.plfreq == 32000) ? false : *dtx;
>-
>-  CHECK_ERROR_MT(myACM->SetVAD(*dtx, *vad, *mode));
>-  myChannel->ResetStats();
>-
>-  CheckVADStatus(side);
>-  if (!_randomTest) {
>-    fprintf(stdout, "\n");
>-    fprintf(stdout, "-----------------------------------------------\n");
>-  }
>-
>-  // Fault Test
>-  CHECK_PROTECTED_MT(myACM->SetVAD(false, true, (ACMVADMode) - 1));
>-  CHECK_PROTECTED_MT(myACM->SetVAD(false, true, (ACMVADMode) 4));
>-
>-}
>-
>-void APITest::CurrentCodec(char side) {
>-  auto myCodec = (side == 'A' ? _acmA : _acmB)->SendCodec();
>-
>-  if (!_randomTest) {
>-    fprintf(stdout, "\n\n");
>-    fprintf(stdout, "Send codec in Side A\n");
>-    fprintf(stdout, "----------------------------\n");
>-    fprintf(stdout, "Name................. %s\n", myCodec->plname);
>-    fprintf(stdout, "Sampling Frequency... %d\n", myCodec->plfreq);
>-    fprintf(stdout, "Rate................. %d\n", myCodec->rate);
>-    fprintf(stdout, "Payload-type......... %d\n", myCodec->pltype);
>-    fprintf(stdout, "Packet-size.......... %d\n", myCodec->pacsize);
>-  }
>-
>-  Wait(100);
>-}
>-
>-void APITest::ChangeCodec(char side) {
>-  CodecInst myCodec;
>-  AudioCodingModule* myACM;
>-  uint8_t* codecCntr;
>-  bool* thereIsEncoder;
>-  bool* vad;
>-  bool* dtx;
>-  ACMVADMode* mode;
>-  Channel* myChannel;
>-  // Reset and Wait
>-  if (!_randomTest) {
>-    fprintf(stdout, "Reset Encoder Side A \n");
>-  }
>-  if (side == 'A') {
>-    myACM = _acmA.get();
>-    codecCntr = &_codecCntrA;
>-    {
>-      WriteLockScoped wl(_apiTestRWLock);
>-      thereIsEncoder = &_thereIsEncoderA;
>-    }
>-    vad = &_sendVADA;
>-    dtx = &_sendDTXA;
>-    mode = &_sendVADModeA;
>-    myChannel = _channel_A2B;
>-  } else {
>-    myACM = _acmB.get();
>-    codecCntr = &_codecCntrB;
>-    {
>-      WriteLockScoped wl(_apiTestRWLock);
>-      thereIsEncoder = &_thereIsEncoderB;
>-    }
>-    vad = &_sendVADB;
>-    dtx = &_sendDTXB;
>-    mode = &_sendVADModeB;
>-    myChannel = _channel_B2A;
>-  }
>-
>-  Wait(100);
>-
>-  // Register the next codec
>-  do {
>-    *codecCntr =
>-        (*codecCntr < AudioCodingModule::NumberOfCodecs() - 1) ?
>-            (*codecCntr + 1) : 0;
>-
>-    if (*codecCntr == 0) {
>-      //printf("Initialize Sender Side A \n");
>-      {
>-        WriteLockScoped wl(_apiTestRWLock);
>-        *thereIsEncoder = false;
>-      }
>-      // After Initialization CN is lost, re-register them
>-      if (AudioCodingModule::Codec("CN", &myCodec, 8000, 1) >= 0) {
>-        CHECK_ERROR_MT(myACM->RegisterSendCodec(myCodec));
>-      }
>-      if (AudioCodingModule::Codec("CN", &myCodec, 16000, 1) >= 0) {
>-        CHECK_ERROR_MT(myACM->RegisterSendCodec(myCodec));
>-      }
>-      // VAD & DTX are disabled after initialization
>-      *vad = false;
>-      *dtx = false;
>-      _writeToFile = false;
>-    }
>-
>-    AudioCodingModule::Codec(*codecCntr, &myCodec);
>-  } while (!STR_CASE_CMP(myCodec.plname, "CN")
>-      || !STR_CASE_CMP(myCodec.plname, "telephone-event")
>-      || !STR_CASE_CMP(myCodec.plname, "RED"));
>-
>-  if (!_randomTest) {
>-    fprintf(stdout,"\n=====================================================\n");
>-    fprintf(stdout, "      Registering New Codec %s, %d kHz, %d kbps\n",
>-            myCodec.plname, myCodec.plfreq / 1000, myCodec.rate / 1000);
>-  }
>-  //std::cout<< std::flush;
>-
>-  // NO DTX for supe-wideband codec at this point
>-  if (myCodec.plfreq == 32000) {
>-    *dtx = false;
>-    CHECK_ERROR_MT(myACM->SetVAD(*dtx, *vad, *mode));
>-
>-  }
>-
>-  CHECK_ERROR_MT(myACM->RegisterSendCodec(myCodec));
>-  myChannel->ResetStats();
>-  {
>-    WriteLockScoped wl(_apiTestRWLock);
>-    *thereIsEncoder = true;
>-  }
>-  Wait(500);
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/APITest.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/APITest.h
>deleted file mode 100644
>index 652d7c401c9..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/APITest.h
>+++ /dev/null
>@@ -1,162 +0,0 @@
>-/*
>- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MODULES_AUDIO_CODING_TEST_APITEST_H_
>-#define MODULES_AUDIO_CODING_TEST_APITEST_H_
>-
>-#include <memory>
>-
>-#include "modules/audio_coding/include/audio_coding_module.h"
>-#include "modules/audio_coding/test/ACMTest.h"
>-#include "modules/audio_coding/test/Channel.h"
>-#include "modules/audio_coding/test/PCMFile.h"
>-#include "modules/audio_coding/test/utility.h"
>-#include "system_wrappers/include/event_wrapper.h"
>-#include "system_wrappers/include/rw_lock_wrapper.h"
>-
>-namespace webrtc {
>-
>-enum APITESTAction {
>-  TEST_CHANGE_CODEC_ONLY = 0,
>-  DTX_TEST = 1
>-};
>-
>-class APITest : public ACMTest {
>- public:
>-  APITest();
>-  ~APITest();
>-
>-  void Perform();
>- private:
>-  int16_t SetUp();
>-
>-  static bool PushAudioThreadA(void* obj);
>-  static bool PullAudioThreadA(void* obj);
>-  static bool ProcessThreadA(void* obj);
>-  static bool APIThreadA(void* obj);
>-
>-  static bool PushAudioThreadB(void* obj);
>-  static bool PullAudioThreadB(void* obj);
>-  static bool ProcessThreadB(void* obj);
>-  static bool APIThreadB(void* obj);
>-
>-  void CheckVADStatus(char side);
>-
>-  // Set Min delay, get delay, playout timestamp
>-  void TestDelay(char side);
>-
>-  // Unregister a codec & register again.
>-  void TestRegisteration(char side);
>-
>-  // Playout Mode, background noise mode.
>-  // Receiver Frequency, playout frequency.
>-  void TestPlayout(char receiveSide);
>-
>-  //
>-  void TestSendVAD(char side);
>-
>-  void CurrentCodec(char side);
>-
>-  void ChangeCodec(char side);
>-
>-  void Wait(uint32_t waitLengthMs);
>-
>-  void RunTest(char thread);
>-
>-  bool PushAudioRunA();
>-  bool PullAudioRunA();
>-  bool ProcessRunA();
>-  bool APIRunA();
>-
>-  bool PullAudioRunB();
>-  bool PushAudioRunB();
>-  bool ProcessRunB();
>-  bool APIRunB();
>-
>-  //--- ACMs
>-  std::unique_ptr<AudioCodingModule> _acmA;
>-  std::unique_ptr<AudioCodingModule> _acmB;
>-
>-  //--- Channels
>-  Channel* _channel_A2B;
>-  Channel* _channel_B2A;
>-
>-  //--- I/O files
>-  // A
>-  PCMFile _inFileA;
>-  PCMFile _outFileA;
>-  // B
>-  PCMFile _outFileB;
>-  PCMFile _inFileB;
>-
>-  //--- I/O params
>-  // A
>-  int32_t _outFreqHzA;
>-  // B
>-  int32_t _outFreqHzB;
>-
>-  // Should we write to file.
>-  // we might skip writing to file if we
>-  // run the test for a long time.
>-  bool _writeToFile;
>-  //--- Events
>-  // A
>-  EventTimerWrapper* _pullEventA;      // pulling data from ACM
>-  EventTimerWrapper* _pushEventA;      // pushing data to ACM
>-  EventTimerWrapper* _processEventA;   // process
>-  EventWrapper* _apiEventA;       // API calls
>-  // B
>-  EventTimerWrapper* _pullEventB;      // pulling data from ACM
>-  EventTimerWrapper* _pushEventB;      // pushing data to ACM
>-  EventTimerWrapper* _processEventB;   // process
>-  EventWrapper* _apiEventB;       // API calls
>-
>-  // keep track of the codec in either side.
>-  uint8_t _codecCntrA;
>-  uint8_t _codecCntrB;
>-
>-  // Is set to true if there is no encoder in either side
>-  bool _thereIsEncoderA;
>-  bool _thereIsEncoderB;
>-  bool _thereIsDecoderA;
>-  bool _thereIsDecoderB;
>-
>-  bool _sendVADA;
>-  bool _sendDTXA;
>-  ACMVADMode _sendVADModeA;
>-
>-  bool _sendVADB;
>-  bool _sendDTXB;
>-  ACMVADMode _sendVADModeB;
>-
>-  int32_t _minDelayA;
>-  int32_t _minDelayB;
>-  bool _payloadUsed[32];
>-
>-  bool _verbose;
>-
>-  int _dotPositionA;
>-  int _dotMoveDirectionA;
>-  int _dotPositionB;
>-  int _dotMoveDirectionB;
>-
>-  char _movingDot[41];
>-
>-  VADCallback* _vadCallbackA;
>-  VADCallback* _vadCallbackB;
>-  RWLockWrapper& _apiTestRWLock;
>-  bool _randomTest;
>-  int _testNumA;
>-  int _testNumB;
>-};
>-
>-}  // namespace webrtc
>-
>-#endif  // MODULES_AUDIO_CODING_TEST_APITEST_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/Channel.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/Channel.cc
>index 7d5e6e291f2..bb970c1d65f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/Channel.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/Channel.cc
>@@ -30,17 +30,16 @@ int32_t Channel::SendData(FrameType frameType,
> 
>   rtpInfo.header.markerBit = false;
>   rtpInfo.header.ssrc = 0;
>-  rtpInfo.header.sequenceNumber = (external_sequence_number_ < 0) ?
>-      _seqNo++ : static_cast<uint16_t>(external_sequence_number_);
>+  rtpInfo.header.sequenceNumber =
>+      (external_sequence_number_ < 0)
>+          ? _seqNo++
>+          : static_cast<uint16_t>(external_sequence_number_);
>   rtpInfo.header.payloadType = payloadType;
>-  rtpInfo.header.timestamp = (external_send_timestamp_ < 0) ? timeStamp :
>-      static_cast<uint32_t>(external_send_timestamp_);
>+  rtpInfo.header.timestamp =
>+      (external_send_timestamp_ < 0)
>+          ? timeStamp
>+          : static_cast<uint32_t>(external_send_timestamp_);
> 
>-  if (frameType == kAudioFrameCN) {
>-    rtpInfo.type.Audio.isCNG = true;
>-  } else {
>-    rtpInfo.type.Audio.isCNG = false;
>-  }
>   if (frameType == kEmptyFrame) {
>     // When frame is empty, we should not transmit it. The frame size of the
>     // next non-empty frame will be based on the previous frame size.
>@@ -48,7 +47,6 @@ int32_t Channel::SendData(FrameType frameType,
>     return 0;
>   }
> 
>-  rtpInfo.type.Audio.channel = 1;
>   // Treat fragmentation separately
>   if (fragmentation != NULL) {
>     // If silence for too long, send only new data.
>@@ -57,7 +55,7 @@ int32_t Channel::SendData(FrameType frameType,
>       // only 0x80 if we have multiple blocks
>       _payloadData[0] = 0x80 + fragmentation->fragmentationPlType[1];
>       size_t REDheader = (fragmentation->fragmentationTimeDiff[1] << 10) +
>-          fragmentation->fragmentationLength[1];
>+                         fragmentation->fragmentationLength[1];
>       _payloadData[1] = uint8_t((REDheader >> 16) & 0x000000FF);
>       _payloadData[2] = uint8_t((REDheader >> 8) & 0x000000FF);
>       _payloadData[3] = uint8_t(REDheader & 0x000000FF);
>@@ -85,18 +83,16 @@ int32_t Channel::SendData(FrameType frameType,
>       if (_leftChannel) {
>         memcpy(&_rtpInfo, &rtpInfo, sizeof(WebRtcRTPHeader));
>         _leftChannel = false;
>-        rtpInfo.type.Audio.channel = 1;
>       } else {
>         memcpy(&rtpInfo, &_rtpInfo, sizeof(WebRtcRTPHeader));
>         _leftChannel = true;
>-        rtpInfo.type.Audio.channel = 2;
>       }
>     }
>   }
> 
>   _channelCritSect.Enter();
>   if (_saveBitStream) {
>-    //fwrite(payloadData, sizeof(uint8_t), payloadSize, _bitStreamFile);
>+    // fwrite(payloadData, sizeof(uint8_t), payloadSize, _bitStreamFile);
>   }
> 
>   if (!_isStereo) {
>@@ -128,8 +124,8 @@ int32_t Channel::SendData(FrameType frameType,
> // TODO(turajs): rewite this method.
> void Channel::CalcStatistics(WebRtcRTPHeader& rtpInfo, size_t payloadSize) {
>   int n;
>-  if ((rtpInfo.header.payloadType != _lastPayloadType)
>-      && (_lastPayloadType != -1)) {
>+  if ((rtpInfo.header.payloadType != _lastPayloadType) &&
>+      (_lastPayloadType != -1)) {
>     // payload-type is changed.
>     // we have to terminate the calculations on the previous payload type
>     // we ignore the last packet in that payload type just to make things
>@@ -156,14 +152,15 @@ void Channel::CalcStatistics(WebRtcRTPHeader& rtpInfo, size_t payloadSize) {
>   if (!newPayload) {
>     if (!currentPayloadStr->newPacket) {
>       if (!_useLastFrameSize) {
>-        _lastFrameSizeSample = (uint32_t) ((uint32_t) rtpInfo.header.timestamp -
>-            (uint32_t) currentPayloadStr->lastTimestamp);
>+        _lastFrameSizeSample =
>+            (uint32_t)((uint32_t)rtpInfo.header.timestamp -
>+                       (uint32_t)currentPayloadStr->lastTimestamp);
>       }
>       assert(_lastFrameSizeSample > 0);
>       int k = 0;
>       for (; k < MAX_NUM_FRAMESIZES; ++k) {
>         if ((currentPayloadStr->frameSizeStats[k].frameSizeSample ==
>-            _lastFrameSizeSample) ||
>+             _lastFrameSizeSample) ||
>             (currentPayloadStr->frameSizeStats[k].frameSizeSample == 0)) {
>           break;
>         }
>@@ -174,9 +171,9 @@ void Channel::CalcStatistics(WebRtcRTPHeader& rtpInfo, size_t payloadSize) {
>                _lastPayloadType, _lastFrameSizeSample);
>         return;
>       }
>-      ACMTestFrameSizeStats* currentFrameSizeStats = &(currentPayloadStr
>-          ->frameSizeStats[k]);
>-      currentFrameSizeStats->frameSizeSample = (int16_t) _lastFrameSizeSample;
>+      ACMTestFrameSizeStats* currentFrameSizeStats =
>+          &(currentPayloadStr->frameSizeStats[k]);
>+      currentFrameSizeStats->frameSizeSample = (int16_t)_lastFrameSizeSample;
> 
>       // increment the number of encoded samples.
>       currentFrameSizeStats->totalEncodedSamples += _lastFrameSizeSample;
>@@ -185,15 +182,15 @@ void Channel::CalcStatistics(WebRtcRTPHeader& rtpInfo, size_t payloadSize) {
>       // increment the total number of bytes (this is based on
>       // the previous payload we don't know the frame-size of
>       // the current payload.
>-      currentFrameSizeStats->totalPayloadLenByte += currentPayloadStr
>-          ->lastPayloadLenByte;
>+      currentFrameSizeStats->totalPayloadLenByte +=
>+          currentPayloadStr->lastPayloadLenByte;
>       // store the maximum payload-size (this is based on
>       // the previous payload we don't know the frame-size of
>       // the current payload.
>-      if (currentFrameSizeStats->maxPayloadLen
>-          < currentPayloadStr->lastPayloadLenByte) {
>-        currentFrameSizeStats->maxPayloadLen = currentPayloadStr
>-            ->lastPayloadLenByte;
>+      if (currentFrameSizeStats->maxPayloadLen <
>+          currentPayloadStr->lastPayloadLenByte) {
>+        currentFrameSizeStats->maxPayloadLen =
>+            currentPayloadStr->lastPayloadLenByte;
>       }
>       // store the current values for the next time
>       currentPayloadStr->lastTimestamp = rtpInfo.header.timestamp;
>@@ -203,8 +200,8 @@ void Channel::CalcStatistics(WebRtcRTPHeader& rtpInfo, size_t payloadSize) {
>       currentPayloadStr->lastPayloadLenByte = payloadSize;
>       currentPayloadStr->lastTimestamp = rtpInfo.header.timestamp;
>       currentPayloadStr->payloadType = rtpInfo.header.payloadType;
>-      memset(currentPayloadStr->frameSizeStats, 0, MAX_NUM_FRAMESIZES *
>-             sizeof(ACMTestFrameSizeStats));
>+      memset(currentPayloadStr->frameSizeStats, 0,
>+             MAX_NUM_FRAMESIZES * sizeof(ACMTestFrameSizeStats));
>     }
>   } else {
>     n = 0;
>@@ -216,8 +213,8 @@ void Channel::CalcStatistics(WebRtcRTPHeader& rtpInfo, size_t payloadSize) {
>     _payloadStats[n].lastPayloadLenByte = payloadSize;
>     _payloadStats[n].lastTimestamp = rtpInfo.header.timestamp;
>     _payloadStats[n].payloadType = rtpInfo.header.payloadType;
>-    memset(_payloadStats[n].frameSizeStats, 0, MAX_NUM_FRAMESIZES *
>-           sizeof(ACMTestFrameSizeStats));
>+    memset(_payloadStats[n].frameSizeStats, 0,
>+           MAX_NUM_FRAMESIZES * sizeof(ACMTestFrameSizeStats));
>   }
> }
> 
>@@ -262,8 +259,7 @@ Channel::Channel(int16_t chID)
>   }
> }
> 
>-Channel::~Channel() {
>-}
>+Channel::~Channel() {}
> 
> void Channel::RegisterReceiverACM(AudioCodingModule* acm) {
>   _receiverACM = acm;
>@@ -311,13 +307,13 @@ int16_t Channel::Stats(CodecInst& codecInst,
>       _channelCritSect.Leave();
>       return 0;
>     }
>-    payloadStats.frameSizeStats[n].usageLenSec = (double) payloadStats
>-        .frameSizeStats[n].totalEncodedSamples / (double) codecInst.plfreq;
>+    payloadStats.frameSizeStats[n].usageLenSec =
>+        (double)payloadStats.frameSizeStats[n].totalEncodedSamples /
>+        (double)codecInst.plfreq;
> 
>     payloadStats.frameSizeStats[n].rateBitPerSec =
>-        payloadStats.frameSizeStats[n].totalPayloadLenByte * 8
>-            / payloadStats.frameSizeStats[n].usageLenSec;
>-
>+        payloadStats.frameSizeStats[n].totalPayloadLenByte * 8 /
>+        payloadStats.frameSizeStats[n].usageLenSec;
>   }
>   _channelCritSect.Leave();
>   return 0;
>@@ -353,14 +349,14 @@ void Channel::Stats(uint8_t* payloadType, uint32_t* payloadLenByte) {
>     if (_payloadStats[k].payloadType == -1) {
>       break;
>     }
>-    payloadType[k] = (uint8_t) _payloadStats[k].payloadType;
>+    payloadType[k] = (uint8_t)_payloadStats[k].payloadType;
>     payloadLenByte[k] = 0;
>     for (n = 0; n < MAX_NUM_FRAMESIZES; n++) {
>       if (_payloadStats[k].frameSizeStats[n].frameSizeSample == 0) {
>         break;
>       }
>-      payloadLenByte[k] += (uint16_t) _payloadStats[k].frameSizeStats[n]
>-          .totalPayloadLenByte;
>+      payloadLenByte[k] +=
>+          (uint16_t)_payloadStats[k].frameSizeStats[n].totalPayloadLenByte;
>     }
>   }
> 
>@@ -387,18 +383,15 @@ void Channel::PrintStats(CodecInst& codecInst) {
>            payloadStats.frameSizeStats[k].rateBitPerSec);
>     printf("Maximum Payload-Size.......... %" PRIuS " Bytes\n",
>            payloadStats.frameSizeStats[k].maxPayloadLen);
>-    printf(
>-        "Maximum Instantaneous Rate.... %.0f bits/sec\n",
>-        ((double) payloadStats.frameSizeStats[k].maxPayloadLen * 8.0
>-            * (double) codecInst.plfreq)
>-            / (double) payloadStats.frameSizeStats[k].frameSizeSample);
>+    printf("Maximum Instantaneous Rate.... %.0f bits/sec\n",
>+           ((double)payloadStats.frameSizeStats[k].maxPayloadLen * 8.0 *
>+            (double)codecInst.plfreq) /
>+               (double)payloadStats.frameSizeStats[k].frameSizeSample);
>     printf("Number of Packets............. %u\n",
>-           (unsigned int) payloadStats.frameSizeStats[k].numPackets);
>+           (unsigned int)payloadStats.frameSizeStats[k].numPackets);
>     printf("Duration...................... %0.3f sec\n\n",
>            payloadStats.frameSizeStats[k].usageLenSec);
>-
>   }
>-
> }
> 
> uint32_t Channel::LastInTimestamp() {
>@@ -413,7 +406,7 @@ double Channel::BitRate() {
>   double rate;
>   uint64_t currTime = rtc::TimeMillis();
>   _channelCritSect.Enter();
>-  rate = ((double) _totalBytes * 8.0) / (double) (currTime - _beginTime);
>+  rate = ((double)_totalBytes * 8.0) / (double)(currTime - _beginTime);
>   _channelCritSect.Leave();
>   return rate;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/Channel.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/Channel.h
>index e01e33e43b5..fd161fb1399 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/Channel.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/Channel.h
>@@ -16,12 +16,11 @@
> #include "modules/audio_coding/include/audio_coding_module.h"
> #include "modules/include/module_common_types.h"
> #include "rtc_base/criticalsection.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>-#define MAX_NUM_PAYLOADS   50
>-#define MAX_NUM_FRAMESIZES  6
>+#define MAX_NUM_PAYLOADS 50
>+#define MAX_NUM_FRAMESIZES 6
> 
> // TODO(turajs): Write constructor for this structure.
> struct ACMTestFrameSizeStats {
>@@ -45,7 +44,6 @@ struct ACMTestPayloadStats {
> 
> class Channel : public AudioPacketizationCallback {
>  public:
>-
>   Channel(int16_t chID = -1);
>   ~Channel() override;
> 
>@@ -56,7 +54,7 @@ class Channel : public AudioPacketizationCallback {
>                    size_t payloadSize,
>                    const RTPFragmentationHeader* fragmentation) override;
> 
>-  void RegisterReceiverACM(AudioCodingModule *acm);
>+  void RegisterReceiverACM(AudioCodingModule* acm);
> 
>   void ResetStats();
> 
>@@ -68,9 +66,7 @@ class Channel : public AudioPacketizationCallback {
> 
>   void PrintStats(CodecInst& codecInst);
> 
>-  void SetIsStereo(bool isStereo) {
>-    _isStereo = isStereo;
>-  }
>+  void SetIsStereo(bool isStereo) { _isStereo = isStereo; }
> 
>   uint32_t LastInTimestamp();
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/EncodeDecodeTest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/EncodeDecodeTest.cc
>index c765f685022..4e16dc8989c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/EncodeDecodeTest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/EncodeDecodeTest.cc
>@@ -10,11 +10,12 @@
> 
> #include "modules/audio_coding/test/EncodeDecodeTest.h"
> 
>-#include <memory>
>-#include <sstream>
> #include <stdio.h>
> #include <stdlib.h>
>+#include <memory>
>+#include <sstream>  // no-presubmit-check TODO(webrtc:8982)
> 
>+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/audio_coding/codecs/audio_format_conversion.h"
> #include "modules/audio_coding/include/audio_coding_module.h"
>@@ -53,7 +54,6 @@ Sender::Sender()
> void Sender::Setup(AudioCodingModule *acm, RTPStream *rtpStream,
>                    std::string in_file_name, int sample_rate, size_t channels) {
>   struct CodecInst sendCodec;
>-  int noOfCodecs = acm->NumberOfCodecs();
>   int codecNo;
> 
>   // Open input file
>@@ -68,19 +68,7 @@ void Sender::Setup(AudioCodingModule *acm, RTPStream *rtpStream,
>   _pcmFile.FastForward(100);
> 
>   // Set the codec for the current test.
>-  if ((testMode == 0) || (testMode == 1)) {
>-    // Set the codec id.
>-    codecNo = codeId;
>-  } else {
>-    // Choose codec on command line.
>-    printf("List of supported codec.\n");
>-    for (int n = 0; n < noOfCodecs; n++) {
>-      EXPECT_EQ(0, acm->Codec(n, &sendCodec));
>-      printf("%d %s\n", n, sendCodec.plname);
>-    }
>-    printf("Choose your codec:");
>-    ASSERT_GT(scanf("%d", &codecNo), 0);
>-  }
>+  codecNo = codeId;
> 
>   EXPECT_EQ(0, acm->Codec(codecNo, &sendCodec));
> 
>@@ -149,21 +137,8 @@ void Receiver::Setup(AudioCodingModule *acm, RTPStream *rtpStream,
>   file_name = file_stream.str();
>   _rtpStream = rtpStream;
> 
>-  if (testMode == 1) {
>-    playSampFreq = recvCodec.plfreq;
>-    _pcmFile.Open(file_name, recvCodec.plfreq, "wb+");
>-  } else if (testMode == 0) {
>-    playSampFreq = 32000;
>-    _pcmFile.Open(file_name, 32000, "wb+");
>-  } else {
>-    printf("\nValid output frequencies:\n");
>-    printf("8000\n16000\n32000\n-1,");
>-    printf("which means output frequency equal to received signal frequency");
>-    printf("\n\nChoose output sampling frequency: ");
>-    ASSERT_GT(scanf("%d", &playSampFreq), 0);
>-    file_name = webrtc::test::OutputPath() + out_file_name + ".pcm";
>-    _pcmFile.Open(file_name, playSampFreq, "wb+");
>-  }
>+  playSampFreq = 32000;
>+  _pcmFile.Open(file_name, 32000, "wb+");
> 
>   _realPayloadSizeBytes = 0;
>   _playoutBuffer = new int16_t[WEBRTC_10MS_PCM_AUDIO];
>@@ -248,15 +223,10 @@ void Receiver::Run() {
>   }
> }
> 
>-EncodeDecodeTest::EncodeDecodeTest() {
>-  _testMode = 2;
>-}
>-
>-EncodeDecodeTest::EncodeDecodeTest(int testMode) {
>-  //testMode == 0 for autotest
>-  //testMode == 1 for testing all codecs/parameters
>-  //testMode > 1 for specific user-input test (as it was used before)
>-  _testMode = testMode;
>+EncodeDecodeTest::EncodeDecodeTest(int test_mode) {
>+  // There used to be different test modes. The only one still supported is the
>+  // "autotest" mode.
>+  RTC_CHECK_EQ(0, test_mode);
> }
> 
> void EncodeDecodeTest::Perform() {
>@@ -269,38 +239,32 @@ void EncodeDecodeTest::Perform() {
>   codePars[1] = 0;
>   codePars[2] = 0;
> 
>-  std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create());
>+  std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create(
>+      AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory())));
>   struct CodecInst sendCodecTmp;
>   numCodecs = acm->NumberOfCodecs();
> 
>-  if (_testMode != 2) {
>-    for (int n = 0; n < numCodecs; n++) {
>-      EXPECT_EQ(0, acm->Codec(n, &sendCodecTmp));
>-      if (STR_CASE_CMP(sendCodecTmp.plname, "telephone-event") == 0) {
>-        numPars[n] = 0;
>-      } else if (STR_CASE_CMP(sendCodecTmp.plname, "cn") == 0) {
>-        numPars[n] = 0;
>-      } else if (STR_CASE_CMP(sendCodecTmp.plname, "red") == 0) {
>-        numPars[n] = 0;
>-      } else if (sendCodecTmp.channels == 2) {
>-        numPars[n] = 0;
>-      } else {
>-        numPars[n] = 1;
>-      }
>+  for (int n = 0; n < numCodecs; n++) {
>+    EXPECT_EQ(0, acm->Codec(n, &sendCodecTmp));
>+    if (STR_CASE_CMP(sendCodecTmp.plname, "telephone-event") == 0) {
>+      numPars[n] = 0;
>+    } else if (STR_CASE_CMP(sendCodecTmp.plname, "cn") == 0) {
>+      numPars[n] = 0;
>+    } else if (STR_CASE_CMP(sendCodecTmp.plname, "red") == 0) {
>+      numPars[n] = 0;
>+    } else if (sendCodecTmp.channels == 2) {
>+      numPars[n] = 0;
>+    } else {
>+      numPars[n] = 1;
>     }
>-  } else {
>-    numCodecs = 1;
>-    numPars[0] = 1;
>   }
> 
>-  _receiver.testMode = _testMode;
>-
>   // Loop over all mono codecs:
>   for (int codeId = 0; codeId < numCodecs; codeId++) {
>     // Only encode using real mono encoders, not telephone-event and cng.
>     for (int loopPars = 1; loopPars <= numPars[codeId]; loopPars++) {
>       // Encode all data to file.
>-      std::string fileName = EncodeToFile(1, codeId, codePars, _testMode);
>+      std::string fileName = EncodeToFile(1, codeId, codePars);
> 
>       RTPFile rtpFile;
>       rtpFile.Open(fileName.c_str(), "rb");
>@@ -318,9 +282,9 @@ void EncodeDecodeTest::Perform() {
> 
> std::string EncodeDecodeTest::EncodeToFile(int fileType,
>                                            int codeId,
>-                                           int* codePars,
>-                                           int testMode) {
>-  std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create());
>+                                           int* codePars) {
>+  std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create(
>+      AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory())));
>   RTPFile rtpFile;
>   std::string fileName = webrtc::test::TempFilename(webrtc::test::OutputPath(),
>                                                     "encode_decode_rtp");
>@@ -328,7 +292,6 @@ std::string EncodeDecodeTest::EncodeToFile(int fileType,
>   rtpFile.WriteHeader();
> 
>   // Store for auto_test and logging.
>-  _sender.testMode = testMode;
>   _sender.codeId = codeId;
> 
>   _sender.Setup(acm.get(), &rtpFile, "audio_coding/testfile32kHz", 32000, 1);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/EncodeDecodeTest.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/EncodeDecodeTest.h
>index 5351a2a4bf0..7a21cb5eaad 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/EncodeDecodeTest.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/EncodeDecodeTest.h
>@@ -18,7 +18,7 @@
> #include "modules/audio_coding/test/ACMTest.h"
> #include "modules/audio_coding/test/PCMFile.h"
> #include "modules/audio_coding/test/RTPFile.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include "modules/include/module_common_types.h"
> 
> namespace webrtc {
> 
>@@ -53,8 +53,6 @@ class Sender {
>   void Run();
>   bool Add10MsData();
> 
>-  //for auto_test and logging
>-  uint8_t testMode;
>   uint8_t codeId;
> 
>  protected:
>@@ -79,7 +77,6 @@ class Receiver {
> 
>   //for auto_test and logging
>   uint8_t codeId;
>-  uint8_t testMode;
> 
>  private:
>   PCMFile _pcmFile;
>@@ -100,18 +97,13 @@ class Receiver {
> 
> class EncodeDecodeTest : public ACMTest {
>  public:
>-  EncodeDecodeTest();
>-  explicit EncodeDecodeTest(int testMode);
>+  explicit EncodeDecodeTest(int test_mode);
>   void Perform() override;
> 
>   uint16_t _playoutFreq;
>-  uint8_t _testMode;
> 
>  private:
>-  std::string EncodeToFile(int fileType,
>-                           int codeId,
>-                           int* codePars,
>-                           int testMode);
>+  std::string EncodeToFile(int fileType, int codeId, int* codePars);
> 
>  protected:
>   Sender _sender;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/PCMFile.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/PCMFile.cc
>index e9e9430a339..659f9688c72 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/PCMFile.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/PCMFile.cc
>@@ -14,7 +14,7 @@
> #include <stdio.h>
> #include <string.h>
> 
>-#include "modules/include/module_common_types.h"
>+#include "rtc_base/checks.h"
> #include "test/gtest.h"
> 
> namespace webrtc {
>@@ -30,8 +30,8 @@ PCMFile::PCMFile()
>       rewinded_(false),
>       read_stereo_(false),
>       save_stereo_(false) {
>-  timestamp_ = (((uint32_t) rand() & 0x0000FFFF) << 16) |
>-      ((uint32_t) rand() & 0x0000FFFF);
>+  timestamp_ =
>+      (((uint32_t)rand() & 0x0000FFFF) << 16) | ((uint32_t)rand() & 0x0000FFFF);
> }
> 
> PCMFile::PCMFile(uint32_t timestamp)
>@@ -52,7 +52,8 @@ PCMFile::~PCMFile() {
>   }
> }
> 
>-int16_t PCMFile::ChooseFile(std::string* file_name, int16_t max_len,
>+int16_t PCMFile::ChooseFile(std::string* file_name,
>+                            int16_t max_len,
>                             uint16_t* frequency_hz) {
>   char tmp_name[MAX_FILE_NAME_LENGTH_BYTE];
> 
>@@ -61,8 +62,8 @@ int16_t PCMFile::ChooseFile(std::string* file_name, int16_t max_len,
>   int16_t n = 0;
> 
>   // Removing trailing spaces.
>-  while ((isspace(tmp_name[n]) || iscntrl(tmp_name[n])) && (tmp_name[n] != 0)
>-      && (n < MAX_FILE_NAME_LENGTH_BYTE)) {
>+  while ((isspace(tmp_name[n]) || iscntrl(tmp_name[n])) && (tmp_name[n] != 0) &&
>+         (n < MAX_FILE_NAME_LENGTH_BYTE)) {
>     n++;
>   }
>   if (n > 0) {
>@@ -80,7 +81,7 @@ int16_t PCMFile::ChooseFile(std::string* file_name, int16_t max_len,
>     tmp_name[n + 1] = '\0';
>   }
> 
>-  int16_t len = (int16_t) strlen(tmp_name);
>+  int16_t len = (int16_t)strlen(tmp_name);
>   if (len > max_len) {
>     return -1;
>   }
>@@ -91,15 +92,17 @@ int16_t PCMFile::ChooseFile(std::string* file_name, int16_t max_len,
>   printf("Enter the sampling frequency (in Hz) of the above file [%u]: ",
>          *frequency_hz);
>   EXPECT_TRUE(fgets(tmp_name, 10, stdin) != NULL);
>-  uint16_t tmp_frequency = (uint16_t) atoi(tmp_name);
>+  uint16_t tmp_frequency = (uint16_t)atoi(tmp_name);
>   if (tmp_frequency > 0) {
>     *frequency_hz = tmp_frequency;
>   }
>   return 0;
> }
> 
>-void PCMFile::Open(const std::string& file_name, uint16_t frequency,
>-                   const char* mode, bool auto_rewind) {
>+void PCMFile::Open(const std::string& file_name,
>+                   uint16_t frequency,
>+                   const char* mode,
>+                   bool auto_rewind) {
>   if ((pcm_file_ = fopen(file_name.c_str(), mode)) == NULL) {
>     printf("Cannot open file %s.\n", file_name.c_str());
>     ADD_FAILURE() << "Unable to read file";
>@@ -125,9 +128,9 @@ int32_t PCMFile::Read10MsData(AudioFrame& audio_frame) {
>     channels = 2;
>   }
> 
>-  int32_t payload_size = (int32_t) fread(audio_frame.mutable_data(),
>-                                         sizeof(uint16_t),
>-                                         samples_10ms_ * channels, pcm_file_);
>+  int32_t payload_size =
>+      (int32_t)fread(audio_frame.mutable_data(), sizeof(uint16_t),
>+                     samples_10ms_ * channels, pcm_file_);
>   if (payload_size < samples_10ms_ * channels) {
>     int16_t* frame_data = audio_frame.mutable_data();
>     for (int k = payload_size; k < samples_10ms_ * channels; k++) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/PCMFile.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/PCMFile.h
>index bbf25719803..57ed5fb0b70 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/PCMFile.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/PCMFile.h
>@@ -16,9 +16,8 @@
> 
> #include <string>
> 
>-#include "api/optional.h"
>-#include "modules/include/module_common_types.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include "absl/types/optional.h"
>+#include "api/audio/audio_frame.h"
> 
> namespace webrtc {
> 
>@@ -28,26 +27,27 @@ class PCMFile {
>   PCMFile(uint32_t timestamp);
>   ~PCMFile();
> 
>-  void Open(const std::string& filename, uint16_t frequency, const char* mode,
>+  void Open(const std::string& filename,
>+            uint16_t frequency,
>+            const char* mode,
>             bool auto_rewind = false);
> 
>   int32_t Read10MsData(AudioFrame& audio_frame);
> 
>-  void Write10MsData(const int16_t *playout_buffer, size_t length_smpls);
>+  void Write10MsData(const int16_t* playout_buffer, size_t length_smpls);
>   void Write10MsData(const AudioFrame& audio_frame);
> 
>   uint16_t PayloadLength10Ms() const;
>   int32_t SamplingFrequency() const;
>   void Close();
>-  bool EndOfFile() const {
>-    return end_of_file_;
>-  }
>+  bool EndOfFile() const { return end_of_file_; }
>   // Moves forward the specified number of 10 ms blocks. If a limit has been set
>   // with SetNum10MsBlocksToRead, fast-forwarding does not count towards this
>   // limit.
>   void FastForward(int num_10ms_blocks);
>   void Rewind();
>-  static int16_t ChooseFile(std::string* file_name, int16_t max_len,
>+  static int16_t ChooseFile(std::string* file_name,
>+                            int16_t max_len,
>                             uint16_t* frequency_hz);
>   bool Rewinded();
>   void SaveStereo(bool is_stereo = true);
>@@ -67,7 +67,7 @@ class PCMFile {
>   uint32_t timestamp_;
>   bool read_stereo_;
>   bool save_stereo_;
>-  rtc::Optional<int> num_10ms_blocks_to_read_;
>+  absl::optional<int> num_10ms_blocks_to_read_;
>   int blocks_read_ = 0;
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/PacketLossTest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/PacketLossTest.cc
>index a6c56fa7807..c5cb3963b3a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/PacketLossTest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/PacketLossTest.cc
>@@ -12,6 +12,7 @@
> 
> #include <memory>
> 
>+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
> #include "test/gtest.h"
> #include "test/testsupport/fileutils.h"
> 
>@@ -22,11 +23,10 @@ ReceiverWithPacketLoss::ReceiverWithPacketLoss()
>       burst_length_(1),
>       packet_counter_(0),
>       lost_packet_counter_(0),
>-      burst_lost_counter_(burst_length_) {
>-}
>+      burst_lost_counter_(burst_length_) {}
> 
>-void ReceiverWithPacketLoss::Setup(AudioCodingModule *acm,
>-                                   RTPStream *rtpStream,
>+void ReceiverWithPacketLoss::Setup(AudioCodingModule* acm,
>+                                   RTPStream* rtpStream,
>                                    std::string out_file_name,
>                                    int channels,
>                                    int loss_rate,
>@@ -83,13 +83,14 @@ bool ReceiverWithPacketLoss::PacketLost() {
>   return false;
> }
> 
>-SenderWithFEC::SenderWithFEC()
>-    : expected_loss_rate_(0) {
>-}
>+SenderWithFEC::SenderWithFEC() : expected_loss_rate_(0) {}
> 
>-void SenderWithFEC::Setup(AudioCodingModule *acm, RTPStream *rtpStream,
>-                          std::string in_file_name, int sample_rate,
>-                          int channels, int expected_loss_rate) {
>+void SenderWithFEC::Setup(AudioCodingModule* acm,
>+                          RTPStream* rtpStream,
>+                          std::string in_file_name,
>+                          int sample_rate,
>+                          int channels,
>+                          int expected_loss_rate) {
>   Sender::Setup(acm, rtpStream, in_file_name, sample_rate, channels);
>   EXPECT_TRUE(SetFEC(true));
>   EXPECT_TRUE(SetPacketLossRate(expected_loss_rate));
>@@ -110,24 +111,27 @@ bool SenderWithFEC::SetPacketLossRate(int expected_loss_rate) {
>   return false;
> }
> 
>-PacketLossTest::PacketLossTest(int channels, int expected_loss_rate,
>-                               int actual_loss_rate, int burst_length)
>+PacketLossTest::PacketLossTest(int channels,
>+                               int expected_loss_rate,
>+                               int actual_loss_rate,
>+                               int burst_length)
>     : channels_(channels),
>-      in_file_name_(channels_ == 1 ? "audio_coding/testfile32kHz" :
>-                    "audio_coding/teststereo32kHz"),
>+      in_file_name_(channels_ == 1 ? "audio_coding/testfile32kHz"
>+                                   : "audio_coding/teststereo32kHz"),
>       sample_rate_hz_(32000),
>       sender_(new SenderWithFEC),
>       receiver_(new ReceiverWithPacketLoss),
>       expected_loss_rate_(expected_loss_rate),
>       actual_loss_rate_(actual_loss_rate),
>-      burst_length_(burst_length) {
>-}
>+      burst_length_(burst_length) {}
> 
> void PacketLossTest::Perform() {
> #ifndef WEBRTC_CODEC_OPUS
>   return;
> #else
>-  std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create());
>+  AudioCodingModule::Config config;
>+  config.decoder_factory = CreateBuiltinAudioDecoderFactory();
>+  std::unique_ptr<AudioCodingModule> acm(AudioCodingModule::Create(config));
> 
>   int codec_id = acm->Codec("opus", 48000, channels_);
> 
>@@ -139,7 +143,6 @@ void PacketLossTest::Perform() {
>   rtpFile.Open(fileName.c_str(), "wb+");
>   rtpFile.WriteHeader();
> 
>-  sender_->testMode = 0;
>   sender_->codeId = codec_id;
> 
>   sender_->Setup(acm.get(), &rtpFile, in_file_name_, sample_rate_hz_, channels_,
>@@ -154,7 +157,6 @@ void PacketLossTest::Perform() {
>   rtpFile.Open(fileName.c_str(), "rb");
>   rtpFile.ReadHeader();
> 
>-  receiver_->testMode = 0;
>   receiver_->codeId = codec_id;
> 
>   receiver_->Setup(acm.get(), &rtpFile, "packetLoss_out", channels_,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/PacketLossTest.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/PacketLossTest.h
>index 7eab442b592..f6f92db1ade 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/PacketLossTest.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/PacketLossTest.h
>@@ -20,8 +20,11 @@ namespace webrtc {
> class ReceiverWithPacketLoss : public Receiver {
>  public:
>   ReceiverWithPacketLoss();
>-  void Setup(AudioCodingModule *acm, RTPStream *rtpStream,
>-             std::string out_file_name, int channels, int loss_rate,
>+  void Setup(AudioCodingModule* acm,
>+             RTPStream* rtpStream,
>+             std::string out_file_name,
>+             int channels,
>+             int loss_rate,
>              int burst_length);
>   bool IncomingPacket() override;
> 
>@@ -37,20 +40,27 @@ class ReceiverWithPacketLoss : public Receiver {
> class SenderWithFEC : public Sender {
>  public:
>   SenderWithFEC();
>-  void Setup(AudioCodingModule *acm, RTPStream *rtpStream,
>-             std::string in_file_name, int sample_rate, int channels,
>+  void Setup(AudioCodingModule* acm,
>+             RTPStream* rtpStream,
>+             std::string in_file_name,
>+             int sample_rate,
>+             int channels,
>              int expected_loss_rate);
>   bool SetPacketLossRate(int expected_loss_rate);
>   bool SetFEC(bool enable_fec);
>+
>  protected:
>   int expected_loss_rate_;
> };
> 
> class PacketLossTest : public ACMTest {
>  public:
>-  PacketLossTest(int channels, int expected_loss_rate_, int actual_loss_rate,
>+  PacketLossTest(int channels,
>+                 int expected_loss_rate_,
>+                 int actual_loss_rate,
>                  int burst_length);
>   void Perform();
>+
>  protected:
>   int channels_;
>   std::string in_file_name_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/RTPFile.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/RTPFile.cc
>index 38f304ea7c4..cfe93fa4b96 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/RTPFile.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/RTPFile.cc
>@@ -14,34 +14,36 @@
> #include <limits>
> 
> #ifdef WIN32
>-#   include <Winsock2.h>
>+#include <Winsock2.h>
> #else
>-#   include <arpa/inet.h>
>+#include <arpa/inet.h>
> #endif
> 
>-#include "audio_coding_module.h"
>-#include "system_wrappers/include/rw_lock_wrapper.h"
>+#include "modules/include/module_common_types.h"
> // TODO(tlegrand): Consider removing usage of gtest.
> #include "test/gtest.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
> void RTPStream::ParseRTPHeader(WebRtcRTPHeader* rtpInfo,
>                                const uint8_t* rtpHeader) {
>   rtpInfo->header.payloadType = rtpHeader[1];
>-  rtpInfo->header.sequenceNumber = (static_cast<uint16_t>(rtpHeader[2]) << 8) |
>-      rtpHeader[3];
>+  rtpInfo->header.sequenceNumber =
>+      (static_cast<uint16_t>(rtpHeader[2]) << 8) | rtpHeader[3];
>   rtpInfo->header.timestamp = (static_cast<uint32_t>(rtpHeader[4]) << 24) |
>-      (static_cast<uint32_t>(rtpHeader[5]) << 16) |
>-      (static_cast<uint32_t>(rtpHeader[6]) << 8) | rtpHeader[7];
>+                              (static_cast<uint32_t>(rtpHeader[5]) << 16) |
>+                              (static_cast<uint32_t>(rtpHeader[6]) << 8) |
>+                              rtpHeader[7];
>   rtpInfo->header.ssrc = (static_cast<uint32_t>(rtpHeader[8]) << 24) |
>-      (static_cast<uint32_t>(rtpHeader[9]) << 16) |
>-      (static_cast<uint32_t>(rtpHeader[10]) << 8) | rtpHeader[11];
>+                         (static_cast<uint32_t>(rtpHeader[9]) << 16) |
>+                         (static_cast<uint32_t>(rtpHeader[10]) << 8) |
>+                         rtpHeader[11];
> }
> 
>-void RTPStream::MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType,
>-                              int16_t seqNo, uint32_t timeStamp,
>+void RTPStream::MakeRTPheader(uint8_t* rtpHeader,
>+                              uint8_t payloadType,
>+                              int16_t seqNo,
>+                              uint32_t timeStamp,
>                               uint32_t ssrc) {
>   rtpHeader[0] = 0x80;
>   rtpHeader[1] = payloadType;
>@@ -57,8 +59,11 @@ void RTPStream::MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType,
>   rtpHeader[11] = ssrc & 0xFF;
> }
> 
>-RTPPacket::RTPPacket(uint8_t payloadType, uint32_t timeStamp, int16_t seqNo,
>-                     const uint8_t* payloadData, size_t payloadSize,
>+RTPPacket::RTPPacket(uint8_t payloadType,
>+                     uint32_t timeStamp,
>+                     int16_t seqNo,
>+                     const uint8_t* payloadData,
>+                     size_t payloadSize,
>                      uint32_t frequency)
>     : payloadType(payloadType),
>       timeStamp(timeStamp),
>@@ -83,20 +88,25 @@ RTPBuffer::~RTPBuffer() {
>   delete _queueRWLock;
> }
> 
>-void RTPBuffer::Write(const uint8_t payloadType, const uint32_t timeStamp,
>-                      const int16_t seqNo, const uint8_t* payloadData,
>-                      const size_t payloadSize, uint32_t frequency) {
>-  RTPPacket *packet = new RTPPacket(payloadType, timeStamp, seqNo, payloadData,
>+void RTPBuffer::Write(const uint8_t payloadType,
>+                      const uint32_t timeStamp,
>+                      const int16_t seqNo,
>+                      const uint8_t* payloadData,
>+                      const size_t payloadSize,
>+                      uint32_t frequency) {
>+  RTPPacket* packet = new RTPPacket(payloadType, timeStamp, seqNo, payloadData,
>                                     payloadSize, frequency);
>   _queueRWLock->AcquireLockExclusive();
>   _rtpQueue.push(packet);
>   _queueRWLock->ReleaseLockExclusive();
> }
> 
>-size_t RTPBuffer::Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData,
>-                       size_t payloadSize, uint32_t* offset) {
>+size_t RTPBuffer::Read(WebRtcRTPHeader* rtpInfo,
>+                       uint8_t* payloadData,
>+                       size_t payloadSize,
>+                       uint32_t* offset) {
>   _queueRWLock->AcquireLockShared();
>-  RTPPacket *packet = _rtpQueue.front();
>+  RTPPacket* packet = _rtpQueue.front();
>   _rtpQueue.pop();
>   _queueRWLock->ReleaseLockShared();
>   rtpInfo->header.markerBit = 1;
>@@ -121,7 +131,7 @@ bool RTPBuffer::EndOfFile() const {
>   return eof;
> }
> 
>-void RTPFile::Open(const char *filename, const char *mode) {
>+void RTPFile::Open(const char* filename, const char* mode) {
>   if ((_rtpFile = fopen(filename, mode)) == NULL) {
>     printf("Cannot write file %s.\n", filename);
>     ADD_FAILURE() << "Unable to write file";
>@@ -166,9 +176,12 @@ void RTPFile::ReadHeader() {
>   padding = ntohs(padding);
> }
> 
>-void RTPFile::Write(const uint8_t payloadType, const uint32_t timeStamp,
>-                    const int16_t seqNo, const uint8_t* payloadData,
>-                    const size_t payloadSize, uint32_t frequency) {
>+void RTPFile::Write(const uint8_t payloadType,
>+                    const uint32_t timeStamp,
>+                    const int16_t seqNo,
>+                    const uint8_t* payloadData,
>+                    const size_t payloadSize,
>+                    uint32_t frequency) {
>   /* write RTP packet to file */
>   uint8_t rtpHeader[12];
>   MakeRTPheader(rtpHeader, payloadType, seqNo, timeStamp, 0);
>@@ -186,8 +199,10 @@ void RTPFile::Write(const uint8_t payloadType, const uint32_t timeStamp,
>   EXPECT_EQ(payloadSize, fwrite(payloadData, 1, payloadSize, _rtpFile));
> }
> 
>-size_t RTPFile::Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData,
>-                     size_t payloadSize, uint32_t* offset) {
>+size_t RTPFile::Read(WebRtcRTPHeader* rtpInfo,
>+                     uint8_t* payloadData,
>+                     size_t payloadSize,
>+                     uint32_t* offset) {
>   uint16_t lengthBytes;
>   uint16_t plen;
>   uint8_t rtpHeader[12];
>@@ -206,8 +221,6 @@ size_t RTPFile::Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData,
> 
>   EXPECT_EQ(1u, fread(rtpHeader, 12, 1, _rtpFile));
>   ParseRTPHeader(rtpInfo, rtpHeader);
>-  rtpInfo->type.Audio.isCNG = false;
>-  rtpInfo->type.Audio.channel = 1;
>   EXPECT_EQ(lengthBytes, plen + 8);
> 
>   if (plen == 0) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/RTPFile.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/RTPFile.h
>index 34dae63b2aa..1d0ddf840a0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/RTPFile.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/RTPFile.h
>@@ -15,38 +15,46 @@
> #include <queue>
> 
> #include "modules/audio_coding/include/audio_coding_module.h"
>-#include "modules/include/module_common_types.h"
>-#include "system_wrappers/include/rw_lock_wrapper.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include "rtc_base/synchronization/rw_lock_wrapper.h"
> 
> namespace webrtc {
> 
> class RTPStream {
>  public:
>-  virtual ~RTPStream() {
>-  }
>+  virtual ~RTPStream() {}
> 
>-  virtual void Write(const uint8_t payloadType, const uint32_t timeStamp,
>-                     const int16_t seqNo, const uint8_t* payloadData,
>-                     const size_t payloadSize, uint32_t frequency) = 0;
>+  virtual void Write(const uint8_t payloadType,
>+                     const uint32_t timeStamp,
>+                     const int16_t seqNo,
>+                     const uint8_t* payloadData,
>+                     const size_t payloadSize,
>+                     uint32_t frequency) = 0;
> 
>   // Returns the packet's payload size. Zero should be treated as an
>   // end-of-stream (in the case that EndOfFile() is true) or an error.
>-  virtual size_t Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData,
>-                      size_t payloadSize, uint32_t* offset) = 0;
>+  virtual size_t Read(WebRtcRTPHeader* rtpInfo,
>+                      uint8_t* payloadData,
>+                      size_t payloadSize,
>+                      uint32_t* offset) = 0;
>   virtual bool EndOfFile() const = 0;
> 
>  protected:
>-  void MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType, int16_t seqNo,
>-                     uint32_t timeStamp, uint32_t ssrc);
>+  void MakeRTPheader(uint8_t* rtpHeader,
>+                     uint8_t payloadType,
>+                     int16_t seqNo,
>+                     uint32_t timeStamp,
>+                     uint32_t ssrc);
> 
>   void ParseRTPHeader(WebRtcRTPHeader* rtpInfo, const uint8_t* rtpHeader);
> };
> 
> class RTPPacket {
>  public:
>-  RTPPacket(uint8_t payloadType, uint32_t timeStamp, int16_t seqNo,
>-            const uint8_t* payloadData, size_t payloadSize,
>+  RTPPacket(uint8_t payloadType,
>+            uint32_t timeStamp,
>+            int16_t seqNo,
>+            const uint8_t* payloadData,
>+            size_t payloadSize,
>             uint32_t frequency);
> 
>   ~RTPPacket();
>@@ -81,20 +89,16 @@ class RTPBuffer : public RTPStream {
> 
>  private:
>   RWLockWrapper* _queueRWLock;
>-  std::queue<RTPPacket *> _rtpQueue;
>+  std::queue<RTPPacket*> _rtpQueue;
> };
> 
> class RTPFile : public RTPStream {
>  public:
>-  ~RTPFile() {
>-  }
>+  ~RTPFile() {}
> 
>-  RTPFile()
>-      : _rtpFile(NULL),
>-        _rtpEOF(false) {
>-  }
>+  RTPFile() : _rtpFile(NULL), _rtpEOF(false) {}
> 
>-  void Open(const char *outFilename, const char *mode);
>+  void Open(const char* outFilename, const char* mode);
> 
>   void Close();
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/TestAllCodecs.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/TestAllCodecs.cc
>index 3a968c99cc2..781337c1078 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/TestAllCodecs.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/TestAllCodecs.cc
>@@ -14,15 +14,17 @@
> #include <limits>
> #include <string>
> 
>+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
>+#include "api/audio_codecs/builtin_audio_encoder_factory.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/audio_coding/codecs/audio_format_conversion.h"
> #include "modules/audio_coding/include/audio_coding_module.h"
> #include "modules/audio_coding/include/audio_coding_module_typedefs.h"
> #include "modules/audio_coding/test/utility.h"
> #include "rtc_base/logging.h"
>+#include "rtc_base/stringencode.h"
> #include "test/gtest.h"
> #include "test/testsupport/fileutils.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> // Description of the test:
> // In this test we set up a one-way communication channel from a participant
>@@ -45,19 +47,19 @@ TestPack::TestPack()
>       timestamp_diff_(0),
>       last_in_timestamp_(0),
>       total_bytes_(0),
>-      payload_size_(0) {
>-}
>+      payload_size_(0) {}
> 
>-TestPack::~TestPack() {
>-}
>+TestPack::~TestPack() {}
> 
> void TestPack::RegisterReceiverACM(AudioCodingModule* acm) {
>   receiver_acm_ = acm;
>   return;
> }
> 
>-int32_t TestPack::SendData(FrameType frame_type, uint8_t payload_type,
>-                           uint32_t timestamp, const uint8_t* payload_data,
>+int32_t TestPack::SendData(FrameType frame_type,
>+                           uint8_t payload_type,
>+                           uint32_t timestamp,
>+                           const uint8_t* payload_data,
>                            size_t payload_size,
>                            const RTPFragmentationHeader* fragmentation) {
>   WebRtcRTPHeader rtp_info;
>@@ -68,18 +70,13 @@ int32_t TestPack::SendData(FrameType frame_type, uint8_t payload_type,
>   rtp_info.header.sequenceNumber = sequence_number_++;
>   rtp_info.header.payloadType = payload_type;
>   rtp_info.header.timestamp = timestamp;
>-  if (frame_type == kAudioFrameCN) {
>-    rtp_info.type.Audio.isCNG = true;
>-  } else {
>-    rtp_info.type.Audio.isCNG = false;
>-  }
>+
>   if (frame_type == kEmptyFrame) {
>     // Skip this frame.
>     return 0;
>   }
> 
>   // Only run mono for all test cases.
>-  rtp_info.type.Audio.channel = 1;
>   memcpy(payload_data_, payload_data, payload_size);
> 
>   status = receiver_acm_->IncomingPacket(payload_data_, payload_size, rtp_info);
>@@ -104,8 +101,10 @@ void TestPack::reset_payload_size() {
> }
> 
> TestAllCodecs::TestAllCodecs(int test_mode)
>-    : acm_a_(AudioCodingModule::Create()),
>-      acm_b_(AudioCodingModule::Create()),
>+    : acm_a_(AudioCodingModule::Create(
>+          AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))),
>+      acm_b_(AudioCodingModule::Create(
>+          AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))),
>       channel_a_to_b_(NULL),
>       test_count_(0),
>       packet_size_samples_(0),
>@@ -122,8 +121,8 @@ TestAllCodecs::~TestAllCodecs() {
> }
> 
> void TestAllCodecs::Perform() {
>-  const std::string file_name = webrtc::test::ResourcePath(
>-      "audio_coding/testfile32kHz", "pcm");
>+  const std::string file_name =
>+      webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
>   infile_a_.Open(file_name, 32000, "rb");
> 
>   if (test_mode_ == 0) {
>@@ -303,17 +302,17 @@ void TestAllCodecs::Perform() {
>   char codec_opus[] = "OPUS";
>   RegisterSendCodec('A', codec_opus, 48000, 6000, 480, kVariableSize);
>   Run(channel_a_to_b_);
>-  RegisterSendCodec('A', codec_opus, 48000, 20000, 480*2, kVariableSize);
>+  RegisterSendCodec('A', codec_opus, 48000, 20000, 480 * 2, kVariableSize);
>   Run(channel_a_to_b_);
>-  RegisterSendCodec('A', codec_opus, 48000, 32000, 480*4, kVariableSize);
>+  RegisterSendCodec('A', codec_opus, 48000, 32000, 480 * 4, kVariableSize);
>   Run(channel_a_to_b_);
>   RegisterSendCodec('A', codec_opus, 48000, 48000, 480, kVariableSize);
>   Run(channel_a_to_b_);
>-  RegisterSendCodec('A', codec_opus, 48000, 64000, 480*4, kVariableSize);
>+  RegisterSendCodec('A', codec_opus, 48000, 64000, 480 * 4, kVariableSize);
>   Run(channel_a_to_b_);
>-  RegisterSendCodec('A', codec_opus, 48000, 96000, 480*6, kVariableSize);
>+  RegisterSendCodec('A', codec_opus, 48000, 96000, 480 * 6, kVariableSize);
>   Run(channel_a_to_b_);
>-  RegisterSendCodec('A', codec_opus, 48000, 500000, 480*2, kVariableSize);
>+  RegisterSendCodec('A', codec_opus, 48000, 500000, 480 * 2, kVariableSize);
>   Run(channel_a_to_b_);
>   outfile_b_.Close();
> #endif
>@@ -348,9 +347,12 @@ void TestAllCodecs::Perform() {
> //                            used when registering, can be an internal header
> //                            set to kVariableSize if the codec is a variable
> //                            rate codec
>-void TestAllCodecs::RegisterSendCodec(char side, char* codec_name,
>-                                      int32_t sampling_freq_hz, int rate,
>-                                      int packet_size, size_t extra_byte) {
>+void TestAllCodecs::RegisterSendCodec(char side,
>+                                      char* codec_name,
>+                                      int32_t sampling_freq_hz,
>+                                      int rate,
>+                                      int packet_size,
>+                                      size_t extra_byte) {
>   if (test_mode_ != 0) {
>     // Print out codec and settings.
>     printf("codec: %s Freq: %d Rate: %d PackSize: %d\n", codec_name,
>@@ -374,9 +376,11 @@ void TestAllCodecs::RegisterSendCodec(char side, char* codec_name,
>   // packet. If variable rate codec (extra_byte == -1), set to -1.
>   if (extra_byte != kVariableSize) {
>     // Add 0.875 to always round up to a whole byte
>-    packet_size_bytes_ = static_cast<size_t>(
>-        static_cast<float>(packet_size * rate) /
>-        static_cast<float>(sampling_freq_hz * 8) + 0.875) + extra_byte;
>+    packet_size_bytes_ =
>+        static_cast<size_t>(static_cast<float>(packet_size * rate) /
>+                                static_cast<float>(sampling_freq_hz * 8) +
>+                            0.875) +
>+        extra_byte;
>   } else {
>     // Packets will have a variable size.
>     packet_size_bytes_ = kVariableSize;
>@@ -393,9 +397,7 @@ void TestAllCodecs::RegisterSendCodec(char side, char* codec_name,
>       my_acm = acm_b_.get();
>       break;
>     }
>-    default: {
>-      break;
>-    }
>+    default: { break; }
>   }
>   ASSERT_TRUE(my_acm != NULL);
> 
>@@ -405,7 +407,14 @@ void TestAllCodecs::RegisterSendCodec(char side, char* codec_name,
>                                        sampling_freq_hz, 1));
>   my_codec_param.rate = rate;
>   my_codec_param.pacsize = packet_size;
>-  CHECK_ERROR(my_acm->RegisterSendCodec(my_codec_param));
>+
>+  auto factory = CreateBuiltinAudioEncoderFactory();
>+  constexpr int payload_type = 17;
>+  SdpAudioFormat format = CodecInstToSdp(my_codec_param);
>+  format.parameters["ptime"] = rtc::ToString(rtc::CheckedDivExact(
>+      packet_size, rtc::CheckedDivExact(sampling_freq_hz, 1000)));
>+  my_acm->SetEncoder(
>+      factory->MakeAudioEncoder(payload_type, format, absl::nullopt));
> }
> 
> void TestAllCodecs::Run(TestPack* channel) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/TestAllCodecs.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/TestAllCodecs.h
>index e0285e95e0a..03539ffd846 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/TestAllCodecs.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/TestAllCodecs.h
>@@ -16,7 +16,6 @@
> #include "modules/audio_coding/test/ACMTest.h"
> #include "modules/audio_coding/test/Channel.h"
> #include "modules/audio_coding/test/PCMFile.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -60,8 +59,12 @@ class TestAllCodecs : public ACMTest {
>   // codec name, and a sampling frequency matching is not required.
>   // This is useful for codecs which support several sampling frequency.
>   // Note! Only mono mode is tested in this test.
>-  void RegisterSendCodec(char side, char* codec_name, int32_t sampling_freq_hz,
>-                         int rate, int packet_size, size_t extra_byte);
>+  void RegisterSendCodec(char side,
>+                         char* codec_name,
>+                         int32_t sampling_freq_hz,
>+                         int rate,
>+                         int packet_size,
>+                         size_t extra_byte);
> 
>   void Run(TestPack* channel);
>   void OpenOutFile(int test_number);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/TestRedFec.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/TestRedFec.cc
>index 034c5954afe..71d34867ac1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/TestRedFec.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/TestRedFec.cc
>@@ -12,43 +12,34 @@
> 
> #include <assert.h>
> 
>+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/audio_coding/codecs/audio_format_conversion.h"
> #include "modules/audio_coding/include/audio_coding_module_typedefs.h"
> #include "modules/audio_coding/test/utility.h"
> #include "test/testsupport/fileutils.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>-
>-#ifdef SUPPORT_RED_WB
>-#undef SUPPORT_RED_WB
>-#endif
>-
>-#ifdef SUPPORT_RED_SWB
>-#undef SUPPORT_RED_SWB
>-#endif
>-
>-#ifdef SUPPORT_RED_FB
>-#undef SUPPORT_RED_FB
>-#endif
> 
> namespace webrtc {
> 
> namespace {
>-  const char kNameL16[] = "L16";
>-  const char kNamePCMU[] = "PCMU";
>-  const char kNameCN[] = "CN";
>-  const char kNameRED[] = "RED";
>-  const char kNameISAC[] = "ISAC";
>-  const char kNameG722[] = "G722";
>-  const char kNameOPUS[] = "opus";
>-}
>+
>+const char kNameL16[] = "L16";
>+const char kNamePCMU[] = "PCMU";
>+const char kNameCN[] = "CN";
>+const char kNameRED[] = "RED";
>+const char kNameISAC[] = "ISAC";
>+const char kNameG722[] = "G722";
>+const char kNameOPUS[] = "opus";
>+
>+}  // namespace
> 
> TestRedFec::TestRedFec()
>-    : _acmA(AudioCodingModule::Create()),
>-      _acmB(AudioCodingModule::Create()),
>+    : _acmA(AudioCodingModule::Create(
>+          AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))),
>+      _acmB(AudioCodingModule::Create(
>+          AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))),
>       _channelA2B(NULL),
>-      _testCntr(0) {
>-}
>+      _testCntr(0) {}
> 
> TestRedFec::~TestRedFec() {
>   if (_channelA2B != NULL) {
>@@ -103,37 +94,22 @@ void TestRedFec::Perform() {
>   EXPECT_EQ(0, RegisterSendCodec('A', kNameG722, 16000));
>   EXPECT_EQ(0, RegisterSendCodec('A', kNameCN, 16000));
> 
>-#ifdef SUPPORT_RED_WB
>-  // Switch codec, RED should remain.
>-  EXPECT_TRUE(_acmA->REDStatus());
>-#else
>   // Switch to a 16 kHz codec, RED should have been switched off.
>   EXPECT_FALSE(_acmA->REDStatus());
>-#endif
> 
>   OpenOutFile(_testCntr);
>   EXPECT_EQ(0, SetVAD(true, true, VADAggr));
>   EXPECT_EQ(0, _acmA->SetREDStatus(false));
>   EXPECT_FALSE(_acmA->REDStatus());
>   Run();
>-#ifdef SUPPORT_RED_WB
>-  EXPECT_EQ(0, _acmA->SetREDStatus(true));
>-  EXPECT_TRUE(_acmA->REDStatus());
>-#else
>   EXPECT_EQ(-1, _acmA->SetREDStatus(true));
>   EXPECT_FALSE(_acmA->REDStatus());
>-#endif
>   Run();
>   _outFileB.Close();
> 
>   RegisterSendCodec('A', kNameISAC, 16000);
> 
>-#ifdef SUPPORT_RED_WB
>-  // Switch codec, RED should remain.
>-  EXPECT_TRUE(_acmA->REDStatus());
>-#else
>   EXPECT_FALSE(_acmA->REDStatus());
>-#endif
> 
>   OpenOutFile(_testCntr);
>   EXPECT_EQ(0, SetVAD(true, true, VADVeryAggr));
>@@ -142,26 +118,16 @@ void TestRedFec::Perform() {
>   Run();
>   _outFileB.Close();
> 
>-#ifdef SUPPORT_RED_WB
>-  EXPECT_EQ(0, _acmA->SetREDStatus(true));
>-  EXPECT_TRUE(_acmA->REDStatus());
>-#else
>   EXPECT_EQ(-1, _acmA->SetREDStatus(true));
>   EXPECT_FALSE(_acmA->REDStatus());
>-#endif
>   OpenOutFile(_testCntr);
>   Run();
>   _outFileB.Close();
> 
>   RegisterSendCodec('A', kNameISAC, 32000);
> 
>-#if defined(SUPPORT_RED_SWB) && defined(SUPPORT_RED_WB)
>-  // Switch codec, RED should remain.
>-  EXPECT_TRUE(_acmA->REDStatus());
>-#else
>   // Switch to a 32 kHz codec, RED should have been switched off.
>   EXPECT_FALSE(_acmA->REDStatus());
>-#endif
> 
>   OpenOutFile(_testCntr);
>   EXPECT_EQ(0, SetVAD(true, true, VADVeryAggr));
>@@ -170,13 +136,8 @@ void TestRedFec::Perform() {
>   Run();
>   _outFileB.Close();
> 
>-#ifdef SUPPORT_RED_SWB
>-  EXPECT_EQ(0, _acmA->SetREDStatus(true));
>-  EXPECT_TRUE(_acmA->REDStatus());
>-#else
>   EXPECT_EQ(-1, _acmA->SetREDStatus(true));
>   EXPECT_FALSE(_acmA->REDStatus());
>-#endif
>   OpenOutFile(_testCntr);
>   Run();
>   _outFileB.Close();
>@@ -184,28 +145,8 @@ void TestRedFec::Perform() {
>   RegisterSendCodec('A', kNameISAC, 32000);
>   EXPECT_EQ(0, SetVAD(false, false, VADNormal));
> 
>-#if defined(SUPPORT_RED_SWB) && defined(SUPPORT_RED_WB)
>-  OpenOutFile(_testCntr);
>-  EXPECT_EQ(0, _acmA->SetREDStatus(true));
>-  EXPECT_TRUE(_acmA->REDStatus());
>-  Run();
>-
>-  RegisterSendCodec('A', kNameISAC, 16000);
>-  EXPECT_TRUE(_acmA->REDStatus());
>-  Run();
>-
>-  RegisterSendCodec('A', kNameISAC, 32000);
>-  EXPECT_TRUE(_acmA->REDStatus());
>-  Run();
>-
>-  RegisterSendCodec('A', kNameISAC, 16000);
>-  EXPECT_TRUE(_acmA->REDStatus());
>-  Run();
>-  _outFileB.Close();
>-#else
>   EXPECT_EQ(-1, _acmA->SetREDStatus(true));
>   EXPECT_FALSE(_acmA->REDStatus());
>-#endif
> 
>   _channelA2B->SetFECTestWithPacketLoss(true);
>   // Following tests are under packet losses.
>@@ -213,13 +154,8 @@ void TestRedFec::Perform() {
>   EXPECT_EQ(0, RegisterSendCodec('A', kNameG722));
>   EXPECT_EQ(0, RegisterSendCodec('A', kNameCN, 16000));
> 
>-#if defined(SUPPORT_RED_WB) && defined(SUPPORT_RED_SWB)
>-  // Switch codec, RED should remain.
>-  EXPECT_TRUE(_acmA->REDStatus());
>-#else
>   // Switch to a 16 kHz codec, RED should have been switched off.
>   EXPECT_FALSE(_acmA->REDStatus());
>-#endif
> 
>   OpenOutFile(_testCntr);
>   EXPECT_EQ(0, SetVAD(true, true, VADAggr));
>@@ -228,26 +164,16 @@ void TestRedFec::Perform() {
>   Run();
>   _outFileB.Close();
> 
>-#ifdef SUPPORT_RED_WB
>-  EXPECT_EQ(0, _acmA->SetREDStatus(true));
>-  EXPECT_TRUE(_acmA->REDStatus());
>-#else
>   EXPECT_EQ(-1, _acmA->SetREDStatus(true));
>   EXPECT_FALSE(_acmA->REDStatus());
>-#endif
>   OpenOutFile(_testCntr);
>   Run();
>   _outFileB.Close();
> 
>   RegisterSendCodec('A', kNameISAC, 16000);
> 
>-#ifdef SUPPORT_RED_WB
>-  // Switch codec, RED should remain.
>-  EXPECT_TRUE(_acmA->REDStatus());
>-#else
>   // Switch to a 16 kHz codec, RED should have been switched off.
>   EXPECT_FALSE(_acmA->REDStatus());
>-#endif
> 
>   OpenOutFile(_testCntr);
>   EXPECT_EQ(0, SetVAD(true, true, VADVeryAggr));
>@@ -255,66 +181,31 @@ void TestRedFec::Perform() {
>   EXPECT_FALSE(_acmA->REDStatus());
>   Run();
>   _outFileB.Close();
>-#ifdef SUPPORT_RED_WB
>-  EXPECT_EQ(0, _acmA->SetREDStatus(true));
>-  EXPECT_TRUE(_acmA->REDStatus());
>-#else
>   EXPECT_EQ(-1, _acmA->SetREDStatus(true));
>   EXPECT_FALSE(_acmA->REDStatus());
>-#endif
>   OpenOutFile(_testCntr);
>   Run();
>   _outFileB.Close();
> 
>   RegisterSendCodec('A', kNameISAC, 32000);
> 
>-#if defined(SUPPORT_RED_SWB) && defined(SUPPORT_RED_WB)
>-  // Switch codec, RED should remain.
>-  EXPECT_TRUE(_acmA->REDStatus());
>-#else
>   // Switch to a 32 kHz codec, RED should have been switched off.
>   EXPECT_FALSE(_acmA->REDStatus());
>-#endif
> 
>   OpenOutFile(_testCntr);
>   EXPECT_EQ(0, SetVAD(true, true, VADVeryAggr));
>   EXPECT_EQ(0, _acmA->SetREDStatus(false));
>   EXPECT_FALSE(_acmA->REDStatus());
>-#ifdef SUPPORT_RED_SWB
>-  EXPECT_EQ(0, _acmA->SetREDStatus(true));
>-  EXPECT_TRUE(_acmA->REDStatus());
>-#else
>   EXPECT_EQ(-1, _acmA->SetREDStatus(true));
>   EXPECT_FALSE(_acmA->REDStatus());
>-#endif
>   OpenOutFile(_testCntr);
>   Run();
>   _outFileB.Close();
> 
>   RegisterSendCodec('A', kNameISAC, 32000);
>   EXPECT_EQ(0, SetVAD(false, false, VADNormal));
>-#if defined(SUPPORT_RED_SWB) && defined(SUPPORT_RED_WB)
>-  OpenOutFile(_testCntr);
>-  EXPECT_EQ(0, _acmA->SetREDStatus(true));
>-  EXPECT_TRUE(_acmA->REDStatus());
>-  Run();
>-
>-  RegisterSendCodec('A', kNameISAC, 16000);
>-  EXPECT_TRUE(_acmA->REDStatus());
>-  Run();
>-
>-  RegisterSendCodec('A', kNameISAC, 32000);
>-  EXPECT_TRUE(_acmA->REDStatus());
>-  Run();
>-
>-  RegisterSendCodec('A', kNameISAC, 16000);
>-  EXPECT_TRUE(_acmA->REDStatus());
>-  Run();
>-  _outFileB.Close();
>-#else
>   EXPECT_EQ(-1, _acmA->SetREDStatus(true));
>   EXPECT_FALSE(_acmA->REDStatus());
>-#endif
> 
> #ifndef WEBRTC_CODEC_OPUS
>   EXPECT_TRUE(false);
>@@ -324,33 +215,14 @@ void TestRedFec::Perform() {
> 
>   RegisterSendCodec('A', kNameOPUS, 48000);
> 
>-#if defined(SUPPORT_RED_FB) && defined(SUPPORT_RED_SWB) &&\
>-  defined(SUPPORT_RED_WB)
>-  // Switch to codec, RED should remain switched on.
>-  EXPECT_TRUE(_acmA->REDStatus());
>-#else
>   EXPECT_FALSE(_acmA->REDStatus());
>-#endif
> 
>   // _channelA2B imposes 25% packet loss rate.
>   EXPECT_EQ(0, _acmA->SetPacketLossRate(25));
> 
>-#ifdef SUPPORT_RED_FB
>-  EXPECT_EQ(0, _acmA->SetREDStatus(true));
>-  EXPECT_TRUE(_acmA->REDStatus());
>-  // Codec FEC and RED are mutually exclusive.
>-  EXPECT_EQ(-1, _acmA->SetCodecFEC(true));
>-
>-  EXPECT_EQ(0, _acmA->SetREDStatus(false));
>-  EXPECT_EQ(0, _acmA->SetCodecFEC(true));
>-
>-  // Codec FEC and RED are mutually exclusive.
>-  EXPECT_EQ(-1, _acmA->SetREDStatus(true));
>-#else
>   EXPECT_EQ(-1, _acmA->SetREDStatus(true));
>   EXPECT_FALSE(_acmA->REDStatus());
>   EXPECT_EQ(0, _acmA->SetCodecFEC(true));
>-#endif
> 
>   EXPECT_TRUE(_acmA->CodecFEC());
>   OpenOutFile(_testCntr);
>@@ -369,12 +241,7 @@ void TestRedFec::Perform() {
> 
>   // Switch to Opus again.
>   RegisterSendCodec('A', kNameOPUS, 48000);
>-#ifdef SUPPORT_RED_FB
>-  // Switch to codec, RED should remain switched on.
>-  EXPECT_TRUE(_acmA->REDStatus());
>-#else
>   EXPECT_FALSE(_acmA->REDStatus());
>-#endif
>   EXPECT_EQ(0, _acmA->SetREDStatus(false));
>   EXPECT_EQ(0, _acmA->SetCodecFEC(false));
>   Run();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/TestRedFec.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/TestRedFec.h
>index 98aa0080ecc..1d9dead0bc4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/TestRedFec.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/TestRedFec.h
>@@ -26,11 +26,13 @@ class TestRedFec : public ACMTest {
>   ~TestRedFec();
> 
>   void Perform();
>+
>  private:
>   // The default value of '-1' indicates that the registration is based only on
>   // codec name and a sampling frequency matching is not required. This is
>   // useful for codecs which support several sampling frequency.
>-  int16_t RegisterSendCodec(char side, const char* codecName,
>+  int16_t RegisterSendCodec(char side,
>+                            const char* codecName,
>                             int32_t sampFreqHz = -1);
>   void Run();
>   void OpenOutFile(int16_t testNumber);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/TestStereo.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/TestStereo.cc
>index ba86719fe82..f8c983c7f2d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/TestStereo.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/TestStereo.cc
>@@ -14,13 +14,13 @@
> 
> #include <string>
> 
>+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/audio_coding/codecs/audio_format_conversion.h"
> #include "modules/audio_coding/include/audio_coding_module_typedefs.h"
> #include "modules/audio_coding/test/utility.h"
> #include "test/gtest.h"
> #include "test/testsupport/fileutils.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -33,11 +33,9 @@ TestPackStereo::TestPackStereo()
>       total_bytes_(0),
>       payload_size_(0),
>       codec_mode_(kNotSet),
>-      lost_packet_(false) {
>-}
>+      lost_packet_(false) {}
> 
>-TestPackStereo::~TestPackStereo() {
>-}
>+TestPackStereo::~TestPackStereo() {}
> 
> void TestPackStereo::RegisterReceiverACM(AudioCodingModule* acm) {
>   receiver_acm_ = acm;
>@@ -64,15 +62,8 @@ int32_t TestPackStereo::SendData(const FrameType frame_type,
>   }
> 
>   if (lost_packet_ == false) {
>-    if (frame_type != kAudioFrameCN) {
>-      rtp_info.type.Audio.isCNG = false;
>-      rtp_info.type.Audio.channel = static_cast<int>(codec_mode_);
>-    } else {
>-      rtp_info.type.Audio.isCNG = true;
>-      rtp_info.type.Audio.channel = static_cast<int>(kMono);
>-    }
>-    status = receiver_acm_->IncomingPacket(payload_data, payload_size,
>-                                           rtp_info);
>+    status =
>+        receiver_acm_->IncomingPacket(payload_data, payload_size, rtp_info);
> 
>     if (frame_type != kAudioFrameCN) {
>       payload_size_ = static_cast<int>(payload_size);
>@@ -108,8 +99,10 @@ void TestPackStereo::set_lost_packet(bool lost) {
> }
> 
> TestStereo::TestStereo(int test_mode)
>-    : acm_a_(AudioCodingModule::Create()),
>-      acm_b_(AudioCodingModule::Create()),
>+    : acm_a_(AudioCodingModule::Create(
>+          AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))),
>+      acm_b_(AudioCodingModule::Create(
>+          AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))),
>       channel_a2b_(NULL),
>       test_cntr_(0),
>       pack_size_samp_(0),
>@@ -120,11 +113,13 @@ TestStereo::TestStereo(int test_mode)
>       l16_16khz_pltype_(-1),
>       l16_32khz_pltype_(-1)
> #ifdef PCMA_AND_PCMU
>-      , pcma_pltype_(-1)
>-      , pcmu_pltype_(-1)
>+      ,
>+      pcma_pltype_(-1),
>+      pcmu_pltype_(-1)
> #endif
> #ifdef WEBRTC_CODEC_OPUS
>-      , opus_pltype_(-1)
>+      ,
>+      opus_pltype_(-1)
> #endif
> {
>   // test_mode = 0 for silent test (auto test)
>@@ -147,10 +142,10 @@ void TestStereo::Perform() {
>   ACMVADMode vad_mode;
> 
>   // Open both mono and stereo test files in 32 kHz.
>-  const std::string file_name_stereo = webrtc::test::ResourcePath(
>-      "audio_coding/teststereo32kHz", "pcm");
>-  const std::string file_name_mono = webrtc::test::ResourcePath(
>-      "audio_coding/testfile32kHz", "pcm");
>+  const std::string file_name_stereo =
>+      webrtc::test::ResourcePath("audio_coding/teststereo32kHz", "pcm");
>+  const std::string file_name_mono =
>+      webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
>   frequency_hz = 32000;
>   in_file_stereo_ = new PCMFile();
>   in_file_mono_ = new PCMFile();
>@@ -225,22 +220,22 @@ void TestStereo::Perform() {
>   OpenOutFile(test_cntr_);
>   char codec_g722[] = "G722";
>   RegisterSendCodec('A', codec_g722, 16000, 64000, 160, codec_channels,
>-      g722_pltype_);
>+                    g722_pltype_);
>   Run(channel_a2b_, audio_channels, codec_channels);
>   RegisterSendCodec('A', codec_g722, 16000, 64000, 320, codec_channels,
>-      g722_pltype_);
>+                    g722_pltype_);
>   Run(channel_a2b_, audio_channels, codec_channels);
>   RegisterSendCodec('A', codec_g722, 16000, 64000, 480, codec_channels,
>-      g722_pltype_);
>+                    g722_pltype_);
>   Run(channel_a2b_, audio_channels, codec_channels);
>   RegisterSendCodec('A', codec_g722, 16000, 64000, 640, codec_channels,
>-      g722_pltype_);
>+                    g722_pltype_);
>   Run(channel_a2b_, audio_channels, codec_channels);
>   RegisterSendCodec('A', codec_g722, 16000, 64000, 800, codec_channels,
>-      g722_pltype_);
>+                    g722_pltype_);
>   Run(channel_a2b_, audio_channels, codec_channels);
>   RegisterSendCodec('A', codec_g722, 16000, 64000, 960, codec_channels,
>-      g722_pltype_);
>+                    g722_pltype_);
>   Run(channel_a2b_, audio_channels, codec_channels);
>   out_file_.Close();
> 
>@@ -254,16 +249,16 @@ void TestStereo::Perform() {
>   OpenOutFile(test_cntr_);
>   char codec_l16[] = "L16";
>   RegisterSendCodec('A', codec_l16, 8000, 128000, 80, codec_channels,
>-      l16_8khz_pltype_);
>+                    l16_8khz_pltype_);
>   Run(channel_a2b_, audio_channels, codec_channels);
>   RegisterSendCodec('A', codec_l16, 8000, 128000, 160, codec_channels,
>-      l16_8khz_pltype_);
>+                    l16_8khz_pltype_);
>   Run(channel_a2b_, audio_channels, codec_channels);
>   RegisterSendCodec('A', codec_l16, 8000, 128000, 240, codec_channels,
>-      l16_8khz_pltype_);
>+                    l16_8khz_pltype_);
>   Run(channel_a2b_, audio_channels, codec_channels);
>   RegisterSendCodec('A', codec_l16, 8000, 128000, 320, codec_channels,
>-      l16_8khz_pltype_);
>+                    l16_8khz_pltype_);
>   Run(channel_a2b_, audio_channels, codec_channels);
>   out_file_.Close();
> 
>@@ -275,16 +270,16 @@ void TestStereo::Perform() {
>   test_cntr_++;
>   OpenOutFile(test_cntr_);
>   RegisterSendCodec('A', codec_l16, 16000, 256000, 160, codec_channels,
>-      l16_16khz_pltype_);
>+                    l16_16khz_pltype_);
>   Run(channel_a2b_, audio_channels, codec_channels);
>   RegisterSendCodec('A', codec_l16, 16000, 256000, 320, codec_channels,
>-      l16_16khz_pltype_);
>+                    l16_16khz_pltype_);
>   Run(channel_a2b_, audio_channels, codec_channels);
>   RegisterSendCodec('A', codec_l16, 16000, 256000, 480, codec_channels,
>-      l16_16khz_pltype_);
>+                    l16_16khz_pltype_);
>   Run(channel_a2b_, audio_channels, codec_channels);
>   RegisterSendCodec('A', codec_l16, 16000, 256000, 640, codec_channels,
>-      l16_16khz_pltype_);
>+                    l16_16khz_pltype_);
>   Run(channel_a2b_, audio_channels, codec_channels);
>   out_file_.Close();
> 
>@@ -296,10 +291,10 @@ void TestStereo::Perform() {
>   test_cntr_++;
>   OpenOutFile(test_cntr_);
>   RegisterSendCodec('A', codec_l16, 32000, 512000, 320, codec_channels,
>-      l16_32khz_pltype_);
>+                    l16_32khz_pltype_);
>   Run(channel_a2b_, audio_channels, codec_channels);
>   RegisterSendCodec('A', codec_l16, 32000, 512000, 640, codec_channels,
>-      l16_32khz_pltype_);
>+                    l16_32khz_pltype_);
>   Run(channel_a2b_, audio_channels, codec_channels);
>   out_file_.Close();
> #ifdef PCMA_AND_PCMU
>@@ -387,26 +382,26 @@ void TestStereo::Perform() {
>   char codec_opus[] = "opus";
>   // Run Opus with 10 ms frame size.
>   RegisterSendCodec('A', codec_opus, 48000, 64000, 480, codec_channels,
>-      opus_pltype_);
>+                    opus_pltype_);
>   Run(channel_a2b_, audio_channels, codec_channels);
>   // Run Opus with 20 ms frame size.
>-  RegisterSendCodec('A', codec_opus, 48000, 64000, 480*2, codec_channels,
>-      opus_pltype_);
>+  RegisterSendCodec('A', codec_opus, 48000, 64000, 480 * 2, codec_channels,
>+                    opus_pltype_);
>   Run(channel_a2b_, audio_channels, codec_channels);
>   // Run Opus with 40 ms frame size.
>-  RegisterSendCodec('A', codec_opus, 48000, 64000, 480*4, codec_channels,
>-      opus_pltype_);
>+  RegisterSendCodec('A', codec_opus, 48000, 64000, 480 * 4, codec_channels,
>+                    opus_pltype_);
>   Run(channel_a2b_, audio_channels, codec_channels);
>   // Run Opus with 60 ms frame size.
>-  RegisterSendCodec('A', codec_opus, 48000, 64000, 480*6, codec_channels,
>-      opus_pltype_);
>+  RegisterSendCodec('A', codec_opus, 48000, 64000, 480 * 6, codec_channels,
>+                    opus_pltype_);
>   Run(channel_a2b_, audio_channels, codec_channels);
>   // Run Opus with 20 ms frame size and different bitrates.
>   RegisterSendCodec('A', codec_opus, 48000, 40000, 960, codec_channels,
>-      opus_pltype_);
>+                    opus_pltype_);
>   Run(channel_a2b_, audio_channels, codec_channels);
>   RegisterSendCodec('A', codec_opus, 48000, 510000, 960, codec_channels,
>-      opus_pltype_);
>+                    opus_pltype_);
>   Run(channel_a2b_, audio_channels, codec_channels);
>   out_file_.Close();
> #endif
>@@ -425,7 +420,7 @@ void TestStereo::Perform() {
>   channel_a2b_->set_codec_mode(kStereo);
>   OpenOutFile(test_cntr_);
>   RegisterSendCodec('A', codec_g722, 16000, 64000, 160, codec_channels,
>-      g722_pltype_);
>+                    g722_pltype_);
>   Run(channel_a2b_, audio_channels, codec_channels);
>   out_file_.Close();
> 
>@@ -438,7 +433,7 @@ void TestStereo::Perform() {
>   channel_a2b_->set_codec_mode(kStereo);
>   OpenOutFile(test_cntr_);
>   RegisterSendCodec('A', codec_l16, 8000, 128000, 80, codec_channels,
>-      l16_8khz_pltype_);
>+                    l16_8khz_pltype_);
>   Run(channel_a2b_, audio_channels, codec_channels);
>   out_file_.Close();
>   if (test_mode_ != 0) {
>@@ -449,7 +444,7 @@ void TestStereo::Perform() {
>   test_cntr_++;
>   OpenOutFile(test_cntr_);
>   RegisterSendCodec('A', codec_l16, 16000, 256000, 160, codec_channels,
>-      l16_16khz_pltype_);
>+                    l16_16khz_pltype_);
>   Run(channel_a2b_, audio_channels, codec_channels);
>   out_file_.Close();
>   if (test_mode_ != 0) {
>@@ -460,7 +455,7 @@ void TestStereo::Perform() {
>   test_cntr_++;
>   OpenOutFile(test_cntr_);
>   RegisterSendCodec('A', codec_l16, 32000, 512000, 320, codec_channels,
>-      l16_32khz_pltype_);
>+                    l16_32khz_pltype_);
>   Run(channel_a2b_, audio_channels, codec_channels);
>   out_file_.Close();
> #ifdef PCMA_AND_PCMU
>@@ -492,7 +487,7 @@ void TestStereo::Perform() {
>   channel_a2b_->set_codec_mode(kStereo);
>   OpenOutFile(test_cntr_);
>   RegisterSendCodec('A', codec_opus, 48000, 64000, 960, codec_channels,
>-      opus_pltype_);
>+                    opus_pltype_);
>   Run(channel_a2b_, audio_channels, codec_channels);
> 
>   // Encode in mono, decode in stereo mode.
>@@ -517,7 +512,7 @@ void TestStereo::Perform() {
>   test_cntr_++;
>   OpenOutFile(test_cntr_);
>   RegisterSendCodec('A', codec_g722, 16000, 64000, 160, codec_channels,
>-      g722_pltype_);
>+                    g722_pltype_);
> 
>   // Make sure it is possible to set VAD/CNG, now that we are sending mono
>   // again.
>@@ -537,7 +532,7 @@ void TestStereo::Perform() {
>   test_cntr_++;
>   OpenOutFile(test_cntr_);
>   RegisterSendCodec('A', codec_l16, 8000, 128000, 80, codec_channels,
>-      l16_8khz_pltype_);
>+                    l16_8khz_pltype_);
>   Run(channel_a2b_, audio_channels, codec_channels);
>   out_file_.Close();
>   if (test_mode_ != 0) {
>@@ -548,7 +543,7 @@ void TestStereo::Perform() {
>   test_cntr_++;
>   OpenOutFile(test_cntr_);
>   RegisterSendCodec('A', codec_l16, 16000, 256000, 160, codec_channels,
>-      l16_16khz_pltype_);
>+                    l16_16khz_pltype_);
>   Run(channel_a2b_, audio_channels, codec_channels);
>   out_file_.Close();
>   if (test_mode_ != 0) {
>@@ -559,7 +554,7 @@ void TestStereo::Perform() {
>   test_cntr_++;
>   OpenOutFile(test_cntr_);
>   RegisterSendCodec('A', codec_l16, 32000, 512000, 320, codec_channels,
>-      l16_32khz_pltype_);
>+                    l16_32khz_pltype_);
>   Run(channel_a2b_, audio_channels, codec_channels);
>   out_file_.Close();
> #ifdef PCMA_AND_PCMU
>@@ -588,7 +583,7 @@ void TestStereo::Perform() {
>   OpenOutFile(test_cntr_);
>   // Encode and decode in mono.
>   RegisterSendCodec('A', codec_opus, 48000, 32000, 960, codec_channels,
>-      opus_pltype_);
>+                    opus_pltype_);
>   CodecInst opus_codec_param;
>   for (uint8_t n = 0; n < num_encoders; n++) {
>     EXPECT_EQ(0, acm_b_->Codec(n, &opus_codec_param));
>@@ -615,8 +610,10 @@ void TestStereo::Perform() {
>   OpenOutFile(test_cntr_);
>   if (test_mode_ != 0) {
>     // Print out codec and settings
>-    printf("Test number: %d\nCodec: Opus Freq: 48000 Rate :32000 PackSize: 960"
>-        " Decode: mono\n", test_cntr_);
>+    printf(
>+        "Test number: %d\nCodec: Opus Freq: 48000 Rate :32000 PackSize: 960"
>+        " Decode: mono\n",
>+        test_cntr_);
>   }
>   Run(channel_a2b_, audio_channels, codec_channels);
>   out_file_.Close();
>@@ -625,8 +622,10 @@ void TestStereo::Perform() {
>   OpenOutFile(test_cntr_);
>   if (test_mode_ != 0) {
>     // Print out codec and settings
>-    printf("Test number: %d\nCodec: Opus Freq: 48000 Rate :32000 PackSize: 960"
>-        " Decode: stereo\n", test_cntr_);
>+    printf(
>+        "Test number: %d\nCodec: Opus Freq: 48000 Rate :32000 PackSize: 960"
>+        " Decode: stereo\n",
>+        test_cntr_);
>   }
>   opus_codec_param.channels = 2;
>   EXPECT_EQ(true,
>@@ -639,8 +638,10 @@ void TestStereo::Perform() {
>   OpenOutFile(test_cntr_);
>   if (test_mode_ != 0) {
>     // Print out codec and settings
>-    printf("Test number: %d\nCodec: Opus Freq: 48000 Rate :32000 PackSize: 960"
>-        " Decode: mono\n", test_cntr_);
>+    printf(
>+        "Test number: %d\nCodec: Opus Freq: 48000 Rate :32000 PackSize: 960"
>+        " Decode: mono\n",
>+        test_cntr_);
>   }
>   opus_codec_param.channels = 1;
>   EXPECT_EQ(true,
>@@ -660,9 +661,10 @@ void TestStereo::Perform() {
> #ifdef WEBRTC_CODEC_OPUS
>     printf("   Opus\n");
> #endif
>-    printf("\nTo complete the test, listen to the %d number of output "
>-           "files.\n",
>-           test_cntr_);
>+    printf(
>+        "\nTo complete the test, listen to the %d number of output "
>+        "files.\n",
>+        test_cntr_);
>   }
> 
>   // Delete the file pointers.
>@@ -679,9 +681,12 @@ void TestStereo::Perform() {
> //          pack_size        - packet size in samples
> //          channels         - number of channels; 1 for mono, 2 for stereo
> //          payload_type     - payload type for the codec
>-void TestStereo::RegisterSendCodec(char side, char* codec_name,
>-                                   int32_t sampling_freq_hz, int rate,
>-                                   int pack_size, int channels,
>+void TestStereo::RegisterSendCodec(char side,
>+                                   char* codec_name,
>+                                   int32_t sampling_freq_hz,
>+                                   int rate,
>+                                   int pack_size,
>+                                   int channels,
>                                    int payload_type) {
>   if (test_mode_ != 0) {
>     // Print out codec and settings
>@@ -717,7 +722,8 @@ void TestStereo::RegisterSendCodec(char side, char* codec_name,
>   CodecInst my_codec_param;
>   // Get all codec parameters before registering
>   EXPECT_GT(AudioCodingModule::Codec(codec_name, &my_codec_param,
>-                                     sampling_freq_hz, channels), -1);
>+                                     sampling_freq_hz, channels),
>+            -1);
>   my_codec_param.rate = rate;
>   my_codec_param.pacsize = pack_size;
>   EXPECT_EQ(0, my_acm->RegisterSendCodec(my_codec_param));
>@@ -725,7 +731,9 @@ void TestStereo::RegisterSendCodec(char side, char* codec_name,
>   send_codec_name_ = codec_name;
> }
> 
>-void TestStereo::Run(TestPackStereo* channel, int in_channels, int out_channels,
>+void TestStereo::Run(TestPackStereo* channel,
>+                     int in_channels,
>+                     int out_channels,
>                      int percent_loss) {
>   AudioFrame audio_frame;
> 
>@@ -780,8 +788,8 @@ void TestStereo::Run(TestPackStereo* channel, int in_channels, int out_channels,
>         variable_packets++;
>       } else {
>         // For fixed rate codecs, check that packet size is correct.
>-        if ((rec_size != pack_size_bytes_ * out_channels)
>-            && (pack_size_bytes_ < 65535)) {
>+        if ((rec_size != pack_size_bytes_ * out_channels) &&
>+            (pack_size_bytes_ < 65535)) {
>           error_count++;
>         }
>       }
>@@ -826,7 +834,7 @@ void TestStereo::OpenOutFile(int16_t test_number) {
>   std::string file_name;
>   std::stringstream file_stream;
>   file_stream << webrtc::test::OutputPath() << "teststereo_out_" << test_number
>-      << ".pcm";
>+              << ".pcm";
>   file_name = file_stream.str();
>   out_file_.Open(file_name, 32000, "wb");
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/TestStereo.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/TestStereo.h
>index a454f256392..6fd7d6ff078 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/TestStereo.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/TestStereo.h
>@@ -23,11 +23,7 @@
> 
> namespace webrtc {
> 
>-enum StereoMonoMode {
>-  kNotSet,
>-  kMono,
>-  kStereo
>-};
>+enum StereoMonoMode { kNotSet, kMono, kStereo };
> 
> class TestPackStereo : public AudioPacketizationCallback {
>  public:
>@@ -72,11 +68,17 @@ class TestStereo : public ACMTest {
>   // The default value of '-1' indicates that the registration is based only on
>   // codec name and a sampling frequncy matching is not required. This is useful
>   // for codecs which support several sampling frequency.
>-  void RegisterSendCodec(char side, char* codec_name, int32_t samp_freq_hz,
>-                         int rate, int pack_size, int channels,
>+  void RegisterSendCodec(char side,
>+                         char* codec_name,
>+                         int32_t samp_freq_hz,
>+                         int rate,
>+                         int pack_size,
>+                         int channels,
>                          int payload_type);
> 
>-  void Run(TestPackStereo* channel, int in_channels, int out_channels,
>+  void Run(TestPackStereo* channel,
>+           int in_channels,
>+           int out_channels,
>            int percent_loss = 0);
>   void OpenOutFile(int16_t test_number);
>   void DisplaySendReceiveCodec();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/TestVADDTX.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/TestVADDTX.cc
>index 80644489895..79b2d31b920 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/TestVADDTX.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/TestVADDTX.cc
>@@ -12,11 +12,11 @@
> 
> #include <string>
> 
>+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
> #include "modules/audio_coding/codecs/audio_format_conversion.h"
> #include "modules/audio_coding/test/PCMFile.h"
> #include "modules/audio_coding/test/utility.h"
> #include "test/testsupport/fileutils.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -62,8 +62,10 @@ void ActivityMonitor::GetStatistics(uint32_t* counter) {
> }
> 
> TestVadDtx::TestVadDtx()
>-    : acm_send_(AudioCodingModule::Create()),
>-      acm_receive_(AudioCodingModule::Create()),
>+    : acm_send_(AudioCodingModule::Create(
>+          AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))),
>+      acm_receive_(AudioCodingModule::Create(
>+          AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))),
>       channel_(new Channel),
>       monitor_(new ActivityMonitor) {
>   EXPECT_EQ(0, acm_send_->RegisterTransportCallback(channel_.get()));
>@@ -81,8 +83,11 @@ void TestVadDtx::RegisterCodec(CodecInst codec_param) {
> 
> // Encoding a file and see if the numbers that various packets occur follow
> // the expectation.
>-void TestVadDtx::Run(std::string in_filename, int frequency, int channels,
>-                     std::string out_filename, bool append,
>+void TestVadDtx::Run(std::string in_filename,
>+                     int frequency,
>+                     int channels,
>+                     std::string out_filename,
>+                     bool append,
>                      const int* expects) {
>   monitor_->ResetStatistics();
> 
>@@ -143,13 +148,10 @@ void TestVadDtx::Run(std::string in_filename, int frequency, int channels,
> 
> // Following is the implementation of TestWebRtcVadDtx.
> TestWebRtcVadDtx::TestWebRtcVadDtx()
>-    : vad_enabled_(false),
>-      dtx_enabled_(false),
>-      output_file_num_(0) {
>-}
>+    : vad_enabled_(false), dtx_enabled_(false), output_file_num_(0) {}
> 
> void TestWebRtcVadDtx::Perform() {
>-  // Go through various test cases.
>+// Go through various test cases.
> #ifdef WEBRTC_CODEC_ISAC
>   // Register iSAC WB as send codec
>   RegisterCodec(kIsacWb);
>@@ -203,15 +205,14 @@ void TestWebRtcVadDtx::Test(bool new_outfile) {
>     output_file_num_++;
>   }
>   std::stringstream out_filename;
>-  out_filename << webrtc::test::OutputPath()
>-               << "testWebRtcVadDtx_outFile_"
>-               << output_file_num_
>-               << ".pcm";
>-  Run(webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"),
>-      32000, 1, out_filename.str(), !new_outfile, expects);
>+  out_filename << webrtc::test::OutputPath() << "testWebRtcVadDtx_outFile_"
>+               << output_file_num_ << ".pcm";
>+  Run(webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"), 32000, 1,
>+      out_filename.str(), !new_outfile, expects);
> }
> 
>-void TestWebRtcVadDtx::SetVAD(bool enable_dtx, bool enable_vad,
>+void TestWebRtcVadDtx::SetVAD(bool enable_dtx,
>+                              bool enable_vad,
>                               ACMVADMode vad_mode) {
>   ACMVADMode mode;
>   EXPECT_EQ(0, acm_send_->SetVAD(enable_dtx, enable_vad, vad_mode));
>@@ -224,10 +225,10 @@ void TestWebRtcVadDtx::SetVAD(bool enable_dtx, bool enable_vad,
>     enable_dtx = enable_vad = false;
>   }
> 
>-  EXPECT_EQ(dtx_enabled_ , enable_dtx); // DTX should be set as expected.
>+  EXPECT_EQ(dtx_enabled_, enable_dtx);  // DTX should be set as expected.
> 
>   if (dtx_enabled_) {
>-    EXPECT_TRUE(vad_enabled_); // WebRTC DTX cannot run without WebRTC VAD.
>+    EXPECT_TRUE(vad_enabled_);  // WebRTC DTX cannot run without WebRTC VAD.
>   } else {
>     // Using no DTX should not affect setting of VAD.
>     EXPECT_EQ(enable_vad, vad_enabled_);
>@@ -247,19 +248,19 @@ void TestOpusDtx::Perform() {
>   int expects[] = {0, 1, 0, 0, 0};
> 
>   // Register Opus as send codec
>-  std::string out_filename = webrtc::test::OutputPath() +
>-      "testOpusDtx_outFile_mono.pcm";
>+  std::string out_filename =
>+      webrtc::test::OutputPath() + "testOpusDtx_outFile_mono.pcm";
>   RegisterCodec(kOpus);
>   EXPECT_EQ(0, acm_send_->DisableOpusDtx());
> 
>-  Run(webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"),
>-      32000, 1, out_filename, false, expects);
>+  Run(webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"), 32000, 1,
>+      out_filename, false, expects);
> 
>   EXPECT_EQ(0, acm_send_->EnableOpusDtx());
>   expects[kEmptyFrame] = 1;
>   expects[kAudioFrameCN] = 1;
>-  Run(webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"),
>-      32000, 1, out_filename, true, expects);
>+  Run(webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"), 32000, 1,
>+      out_filename, true, expects);
> 
>   // Register stereo Opus as send codec
>   out_filename = webrtc::test::OutputPath() + "testOpusDtx_outFile_stereo.pcm";
>@@ -267,15 +268,15 @@ void TestOpusDtx::Perform() {
>   EXPECT_EQ(0, acm_send_->DisableOpusDtx());
>   expects[kEmptyFrame] = 0;
>   expects[kAudioFrameCN] = 0;
>-  Run(webrtc::test::ResourcePath("audio_coding/teststereo32kHz", "pcm"),
>-      32000, 2, out_filename, false, expects);
>+  Run(webrtc::test::ResourcePath("audio_coding/teststereo32kHz", "pcm"), 32000,
>+      2, out_filename, false, expects);
> 
>   EXPECT_EQ(0, acm_send_->EnableOpusDtx());
> 
>   expects[kEmptyFrame] = 1;
>   expects[kAudioFrameCN] = 1;
>-  Run(webrtc::test::ResourcePath("audio_coding/teststereo32kHz", "pcm"),
>-      32000, 2, out_filename, true, expects);
>+  Run(webrtc::test::ResourcePath("audio_coding/teststereo32kHz", "pcm"), 32000,
>+      2, out_filename, true, expects);
> #endif
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/TestVADDTX.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/TestVADDTX.h
>index 8cd4444f3d4..e3840f795a6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/TestVADDTX.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/TestVADDTX.h
>@@ -28,6 +28,7 @@ class ActivityMonitor : public ACMVADCallback {
>   void PrintStatistics();
>   void ResetStatistics();
>   void GetStatistics(uint32_t* stats);
>+
>  private:
>   // 0 - kEmptyFrame
>   // 1 - kAudioFrameSpeech
>@@ -37,7 +38,6 @@ class ActivityMonitor : public ACMVADCallback {
>   uint32_t counter_[5];
> };
> 
>-
> // TestVadDtx is to verify that VAD/DTX perform as they should. It runs through
> // an audio file and check if the occurrence of various packet types follows
> // expectation. TestVadDtx needs its derived class to implement the Perform()
>@@ -65,8 +65,12 @@ class TestVadDtx : public ACMTest {
>   // 2 - kAudioFrameCN
>   // 3 - kVideoFrameKey (not used by audio)
>   // 4 - kVideoFrameDelta (not used by audio)
>-  void Run(std::string in_filename, int frequency, int channels,
>-           std::string out_filename, bool append, const int* expects);
>+  void Run(std::string in_filename,
>+           int frequency,
>+           int channels,
>+           std::string out_filename,
>+           bool append,
>+           const int* expects);
> 
>   std::unique_ptr<AudioCodingModule> acm_send_;
>   std::unique_ptr<AudioCodingModule> acm_receive_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/Tester.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/Tester.cc
>index 9cd774dd15a..8ce50a413bb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/Tester.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/Tester.cc
>@@ -13,7 +13,6 @@
> #include <vector>
> 
> #include "modules/audio_coding/include/audio_coding_module.h"
>-#include "modules/audio_coding/test/APITest.h"
> #include "modules/audio_coding/test/EncodeDecodeTest.h"
> #include "modules/audio_coding/test/PacketLossTest.h"
> #include "modules/audio_coding/test/TestAllCodecs.h"
>@@ -116,7 +115,7 @@ TEST(AudioCodingModuleTest, TestPacketLossBurst) {
> #if defined(WEBRTC_IOS)
> TEST(AudioCodingModuleTest, DISABLED_TestPacketLossStereo) {
> #else
>-  TEST(AudioCodingModuleTest, TestPacketLossStereo) {
>+TEST(AudioCodingModuleTest, TestPacketLossStereo) {
> #endif
>   webrtc::PacketLossTest(2, 10, 10, 1).Perform();
> }
>@@ -133,7 +132,7 @@ TEST(AudioCodingModuleTest, TestPacketLossStereoBurst) {
> // The full API test is too long to run automatically on bots, but can be used
> // for offline testing. User interaction is needed.
> #ifdef ACM_TEST_FULL_API
>-  TEST(AudioCodingModuleTest, TestAPI) {
>-    webrtc::APITest().Perform();
>-  }
>+TEST(AudioCodingModuleTest, TestAPI) {
>+  webrtc::APITest().Perform();
>+}
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/TwoWayCommunication.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/TwoWayCommunication.cc
>index 8049436cd01..3c5b2fede49 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/TwoWayCommunication.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/TwoWayCommunication.cc
>@@ -27,19 +27,21 @@
> #include "modules/audio_coding/test/utility.h"
> #include "test/gtest.h"
> #include "test/testsupport/fileutils.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
> #define MAX_FILE_NAME_LENGTH_BYTE 500
> 
> TwoWayCommunication::TwoWayCommunication(int testMode)
>-    : _acmA(AudioCodingModule::Create()),
>-      _acmRefA(AudioCodingModule::Create()),
>+    : _acmA(AudioCodingModule::Create(
>+          AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))),
>+      _acmRefA(AudioCodingModule::Create(
>+          AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))),
>       _testMode(testMode) {
>   AudioCodingModule::Config config;
>-  // The clicks will be more obvious in FAX mode. TODO(henrik.lundin) Really?
>-  config.neteq_config.playout_mode = kPlayoutFax;
>+  // The clicks will be more obvious if time-stretching is not allowed.
>+  // TODO(henrik.lundin) Really?
>+  config.neteq_config.for_test_no_time_stretching = true;
>   config.decoder_factory = CreateBuiltinAudioDecoderFactory();
>   _acmB.reset(AudioCodingModule::Create(config));
>   _acmRefB.reset(AudioCodingModule::Create(config));
>@@ -58,9 +60,9 @@ TwoWayCommunication::~TwoWayCommunication() {
>   _outFileRefB.Close();
> }
> 
>-void TwoWayCommunication::ChooseCodec(uint8_t* codecID_A,
>-                                      uint8_t* codecID_B) {
>-  std::unique_ptr<AudioCodingModule> tmpACM(AudioCodingModule::Create());
>+void TwoWayCommunication::ChooseCodec(uint8_t* codecID_A, uint8_t* codecID_B) {
>+  std::unique_ptr<AudioCodingModule> tmpACM(AudioCodingModule::Create(
>+      AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory())));
>   uint8_t noCodec = tmpACM->NumberOfCodecs();
>   CodecInst codecInst;
>   printf("List of Supported Codecs\n");
>@@ -72,11 +74,11 @@ void TwoWayCommunication::ChooseCodec(uint8_t* codecID_A,
>   printf("\nChoose a send codec for side A [0]: ");
>   char myStr[15] = "";
>   EXPECT_TRUE(fgets(myStr, 10, stdin) != NULL);
>-  *codecID_A = (uint8_t) atoi(myStr);
>+  *codecID_A = (uint8_t)atoi(myStr);
> 
>   printf("\nChoose a send codec for side B [0]: ");
>   EXPECT_TRUE(fgets(myStr, 10, stdin) != NULL);
>-  *codecID_B = (uint8_t) atoi(myStr);
>+  *codecID_B = (uint8_t)atoi(myStr);
> 
>   printf("\n");
> }
>@@ -115,8 +117,8 @@ void TwoWayCommunication::SetUp() {
>   uint16_t frequencyHz;
> 
>   //--- Input A
>-  std::string in_file_name = webrtc::test::ResourcePath(
>-      "audio_coding/testfile32kHz", "pcm");
>+  std::string in_file_name =
>+      webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
>   frequencyHz = 32000;
>   printf("Enter input file at side A [%s]: ", in_file_name.c_str());
>   PCMFile::ChooseFile(&in_file_name, 499, &frequencyHz);
>@@ -131,8 +133,8 @@ void TwoWayCommunication::SetUp() {
>   _outFileRefA.Open(ref_file_name, frequencyHz, "wb");
> 
>   //--- Input B
>-  in_file_name = webrtc::test::ResourcePath("audio_coding/testfile32kHz",
>-                                            "pcm");
>+  in_file_name =
>+      webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
>   frequencyHz = 32000;
>   printf("\n\nEnter input file at side B [%s]: ", in_file_name.c_str());
>   PCMFile::ChooseFile(&in_file_name, 499, &frequencyHz);
>@@ -197,8 +199,8 @@ void TwoWayCommunication::SetUpAutotest() {
>   uint16_t frequencyHz;
> 
>   //--- Input A and B
>-  std::string in_file_name = webrtc::test::ResourcePath(
>-      "audio_coding/testfile32kHz", "pcm");
>+  std::string in_file_name =
>+      webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
>   frequencyHz = 16000;
>   _inFileA.Open(in_file_name, frequencyHz, "rb");
>   _inFileB.Open(in_file_name, frequencyHz, "rb");
>@@ -207,16 +209,16 @@ void TwoWayCommunication::SetUpAutotest() {
>   std::string output_file_a = webrtc::test::OutputPath() + "outAutotestA.pcm";
>   frequencyHz = 16000;
>   _outFileA.Open(output_file_a, frequencyHz, "wb");
>-  std::string output_ref_file_a = webrtc::test::OutputPath()
>-      + "ref_outAutotestA.pcm";
>+  std::string output_ref_file_a =
>+      webrtc::test::OutputPath() + "ref_outAutotestA.pcm";
>   _outFileRefA.Open(output_ref_file_a, frequencyHz, "wb");
> 
>   //--- Output B
>   std::string output_file_b = webrtc::test::OutputPath() + "outAutotestB.pcm";
>   frequencyHz = 16000;
>   _outFileB.Open(output_file_b, frequencyHz, "wb");
>-  std::string output_ref_file_b = webrtc::test::OutputPath()
>-      + "ref_outAutotestB.pcm";
>+  std::string output_ref_file_b =
>+      webrtc::test::OutputPath() + "ref_outAutotestB.pcm";
>   _outFileRefB.Open(output_ref_file_b, frequencyHz, "wb");
> 
>   //--- Set A-to-B channel
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/TwoWayCommunication.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/TwoWayCommunication.h
>index fb23275a9aa..f0becaec356 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/TwoWayCommunication.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/TwoWayCommunication.h
>@@ -27,6 +27,7 @@ class TwoWayCommunication : public ACMTest {
>   ~TwoWayCommunication();
> 
>   void Perform();
>+
>  private:
>   void ChooseCodec(uint8_t* codecID_A, uint8_t* codecID_B);
>   void SetUp();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/delay_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/delay_test.cc
>index 407f7095cb3..d49cd3ced7f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/delay_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/delay_test.cc
>@@ -15,6 +15,7 @@
> #include <iostream>
> #include <memory>
> 
>+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/audio_coding/codecs/audio_format_conversion.h"
> #include "modules/audio_coding/include/audio_coding_module.h"
>@@ -26,7 +27,6 @@
> #include "system_wrappers/include/event_wrapper.h"
> #include "test/gtest.h"
> #include "test/testsupport/fileutils.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> DEFINE_string(codec, "isac", "Codec Name");
> DEFINE_int(sample_rate_hz, 16000, "Sampling rate in Hertz.");
>@@ -64,8 +64,10 @@ struct TestSettings {
> class DelayTest {
>  public:
>   DelayTest()
>-      : acm_a_(AudioCodingModule::Create()),
>-        acm_b_(AudioCodingModule::Create()),
>+      : acm_a_(AudioCodingModule::Create(
>+            AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))),
>+        acm_b_(AudioCodingModule::Create(
>+            AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))),
>         channel_a2b_(new Channel),
>         test_cntr_(0),
>         encoding_sample_rate_hz_(8000) {}
>@@ -80,26 +82,25 @@ class DelayTest {
> 
>   void Initialize() {
>     test_cntr_ = 0;
>-    std::string file_name = webrtc::test::ResourcePath(
>-        "audio_coding/testfile32kHz", "pcm");
>+    std::string file_name =
>+        webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
>     if (strlen(FLAG_input_file) > 0)
>       file_name = FLAG_input_file;
>     in_file_a_.Open(file_name, 32000, "rb");
>-    ASSERT_EQ(0, acm_a_->InitializeReceiver()) <<
>-        "Couldn't initialize receiver.\n";
>-    ASSERT_EQ(0, acm_b_->InitializeReceiver()) <<
>-        "Couldn't initialize receiver.\n";
>+    ASSERT_EQ(0, acm_a_->InitializeReceiver())
>+        << "Couldn't initialize receiver.\n";
>+    ASSERT_EQ(0, acm_b_->InitializeReceiver())
>+        << "Couldn't initialize receiver.\n";
> 
>     if (FLAG_delay > 0) {
>-      ASSERT_EQ(0, acm_b_->SetMinimumPlayoutDelay(FLAG_delay)) <<
>-          "Failed to set minimum delay.\n";
>+      ASSERT_EQ(0, acm_b_->SetMinimumPlayoutDelay(FLAG_delay))
>+          << "Failed to set minimum delay.\n";
>     }
> 
>     int num_encoders = acm_a_->NumberOfCodecs();
>     CodecInst my_codec_param;
>     for (int n = 0; n < num_encoders; n++) {
>-      EXPECT_EQ(0, acm_b_->Codec(n, &my_codec_param)) <<
>-          "Failed to get codec.";
>+      EXPECT_EQ(0, acm_b_->Codec(n, &my_codec_param)) << "Failed to get codec.";
>       if (STR_CASE_CMP(my_codec_param.plname, "opus") == 0)
>         my_codec_param.channels = 1;
>       else if (my_codec_param.channels > 1)
>@@ -115,12 +116,14 @@ class DelayTest {
>     }
> 
>     // Create and connect the channel
>-    ASSERT_EQ(0, acm_a_->RegisterTransportCallback(channel_a2b_)) <<
>-        "Couldn't register Transport callback.\n";
>+    ASSERT_EQ(0, acm_a_->RegisterTransportCallback(channel_a2b_))
>+        << "Couldn't register Transport callback.\n";
>     channel_a2b_->RegisterReceiverACM(acm_b_.get());
>   }
> 
>-  void Perform(const TestSettings* config, size_t num_tests, int duration_sec,
>+  void Perform(const TestSettings* config,
>+               size_t num_tests,
>+               int duration_sec,
>                const char* output_prefix) {
>     for (size_t n = 0; n < num_tests; ++n) {
>       ApplyConfig(config[n]);
>@@ -131,14 +134,15 @@ class DelayTest {
>  private:
>   void ApplyConfig(const TestSettings& config) {
>     printf("====================================\n");
>-    printf("Test %d \n"
>-           "Codec: %s, %d kHz, %d channel(s)\n"
>-           "ACM: DTX %s, FEC %s\n"
>-           "Channel: %s\n",
>-           ++test_cntr_, config.codec.name, config.codec.sample_rate_hz,
>-           config.codec.num_channels, config.acm.dtx ? "on" : "off",
>-           config.acm.fec ? "on" : "off",
>-           config.packet_loss ? "with packet-loss" : "no packet-loss");
>+    printf(
>+        "Test %d \n"
>+        "Codec: %s, %d kHz, %d channel(s)\n"
>+        "ACM: DTX %s, FEC %s\n"
>+        "Channel: %s\n",
>+        ++test_cntr_, config.codec.name, config.codec.sample_rate_hz,
>+        config.codec.num_channels, config.acm.dtx ? "on" : "off",
>+        config.acm.fec ? "on" : "off",
>+        config.packet_loss ? "with packet-loss" : "no packet-loss");
>     SendCodec(config.codec);
>     ConfigAcm(config.acm);
>     ConfigChannel(config.packet_loss);
>@@ -146,20 +150,20 @@ class DelayTest {
> 
>   void SendCodec(const CodecSettings& config) {
>     CodecInst my_codec_param;
>-    ASSERT_EQ(0, AudioCodingModule::Codec(
>-              config.name, &my_codec_param, config.sample_rate_hz,
>-              config.num_channels)) << "Specified codec is not supported.\n";
>+    ASSERT_EQ(
>+        0, AudioCodingModule::Codec(config.name, &my_codec_param,
>+                                    config.sample_rate_hz, config.num_channels))
>+        << "Specified codec is not supported.\n";
> 
>     encoding_sample_rate_hz_ = my_codec_param.plfreq;
>-    ASSERT_EQ(0, acm_a_->RegisterSendCodec(my_codec_param)) <<
>-        "Failed to register send-codec.\n";
>+    ASSERT_EQ(0, acm_a_->RegisterSendCodec(my_codec_param))
>+        << "Failed to register send-codec.\n";
>   }
> 
>   void ConfigAcm(const AcmSettings& config) {
>-    ASSERT_EQ(0, acm_a_->SetVAD(config.dtx, config.dtx, VADAggr)) <<
>-        "Failed to set VAD.\n";
>-    ASSERT_EQ(0, acm_a_->SetREDStatus(config.fec)) <<
>-        "Failed to set RED.\n";
>+    ASSERT_EQ(0, acm_a_->SetVAD(config.dtx, config.dtx, VADAggr))
>+        << "Failed to set VAD.\n";
>+    ASSERT_EQ(0, acm_a_->SetREDStatus(config.fec)) << "Failed to set RED.\n";
>   }
> 
>   void ConfigChannel(bool packet_loss) {
>@@ -169,7 +173,8 @@ class DelayTest {
>   void OpenOutFile(const char* output_id) {
>     std::stringstream file_stream;
>     file_stream << "delay_test_" << FLAG_codec << "_" << FLAG_sample_rate_hz
>-        << "Hz" << "_" << FLAG_delay << "ms.pcm";
>+                << "Hz"
>+                << "_" << FLAG_delay << "ms.pcm";
>     std::cout << "Output file: " << file_stream.str() << std::endl << std::endl;
>     std::string file_name = webrtc::test::OutputPath() + file_stream.str();
>     out_file_b_.Open(file_name.c_str(), 32000, "wb");
>@@ -194,14 +199,15 @@ class DelayTest {
>       if ((num_frames & 0x3F) == 0x3F) {
>         NetworkStatistics statistics;
>         acm_b_->GetNetworkStatistics(&statistics);
>-        fprintf(stdout, "delay: min=%3d  max=%3d  mean=%3d  median=%3d"
>+        fprintf(stdout,
>+                "delay: min=%3d  max=%3d  mean=%3d  median=%3d"
>                 " ts-based average = %6.3f, "
>                 "curr buff-lev = %4u opt buff-lev = %4u \n",
>                 statistics.minWaitingTimeMs, statistics.maxWaitingTimeMs,
>                 statistics.meanWaitingTimeMs, statistics.medianWaitingTimeMs,
>                 average_delay, statistics.currentBufferSize,
>                 statistics.preferredBufferSize);
>-        fflush (stdout);
>+        fflush(stdout);
>       }
> 
>       in_file_a_.Read10MsData(audio_frame);
>@@ -214,7 +220,7 @@ class DelayTest {
>           audio_frame.data(),
>           audio_frame.samples_per_channel_ * audio_frame.num_channels_);
>       received_ts = channel_a2b_->LastInTimestamp();
>-      rtc::Optional<uint32_t> playout_timestamp = acm_b_->PlayoutTimestamp();
>+      absl::optional<uint32_t> playout_timestamp = acm_b_->PlayoutTimestamp();
>       ASSERT_TRUE(playout_timestamp);
>       inst_delay_sec = static_cast<uint32_t>(received_ts - *playout_timestamp) /
>                        static_cast<double>(encoding_sample_rate_hz_);
>@@ -253,10 +259,8 @@ int main(int argc, char* argv[]) {
>   webrtc::TestSettings test_setting;
>   strcpy(test_setting.codec.name, FLAG_codec);
> 
>-  if (FLAG_sample_rate_hz != 8000 &&
>-      FLAG_sample_rate_hz != 16000 &&
>-      FLAG_sample_rate_hz != 32000 &&
>-      FLAG_sample_rate_hz != 48000) {
>+  if (FLAG_sample_rate_hz != 8000 && FLAG_sample_rate_hz != 16000 &&
>+      FLAG_sample_rate_hz != 32000 && FLAG_sample_rate_hz != 48000) {
>     std::cout << "Invalid sampling rate.\n";
>     return 1;
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/iSACTest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/iSACTest.cc
>index a44259fbbe9..e9fd867e036 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/iSACTest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/iSACTest.cc
>@@ -23,6 +23,7 @@
> #include <time.h>
> #endif
> 
>+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
> #include "modules/audio_coding/codecs/audio_format_conversion.h"
> #include "modules/audio_coding/test/utility.h"
> #include "system_wrappers/include/event_wrapper.h"
>@@ -40,11 +41,11 @@ void SetISACConfigDefault(ACMTestISACConfig& isacConfig) {
>   return;
> }
> 
>-int16_t SetISAConfig(ACMTestISACConfig& isacConfig, AudioCodingModule* acm,
>+int16_t SetISAConfig(ACMTestISACConfig& isacConfig,
>+                     AudioCodingModule* acm,
>                      int testMode) {
>-
>-  if ((isacConfig.currentRateBitPerSec != 0)
>-      || (isacConfig.currentFrameSizeMsec != 0)) {
>+  if ((isacConfig.currentRateBitPerSec != 0) ||
>+      (isacConfig.currentFrameSizeMsec != 0)) {
>     auto sendCodec = acm->SendCodec();
>     EXPECT_TRUE(sendCodec);
>     if (isacConfig.currentRateBitPerSec < 0) {
>@@ -56,8 +57,8 @@ int16_t SetISAConfig(ACMTestISACConfig& isacConfig, AudioCodingModule* acm,
>         sendCodec->rate = isacConfig.currentRateBitPerSec;
>       }
>       if (isacConfig.currentFrameSizeMsec != 0) {
>-        sendCodec->pacsize = isacConfig.currentFrameSizeMsec
>-            * (sendCodec->plfreq / 1000);
>+        sendCodec->pacsize =
>+            isacConfig.currentFrameSizeMsec * (sendCodec->plfreq / 1000);
>       }
>       EXPECT_EQ(0, acm->RegisterSendCodec(*sendCodec));
>     }
>@@ -67,8 +68,10 @@ int16_t SetISAConfig(ACMTestISACConfig& isacConfig, AudioCodingModule* acm,
> }
> 
> ISACTest::ISACTest(int testMode)
>-    : _acmA(AudioCodingModule::Create()),
>-      _acmB(AudioCodingModule::Create()),
>+    : _acmA(AudioCodingModule::Create(
>+          AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))),
>+      _acmB(AudioCodingModule::Create(
>+          AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))),
>       _testMode(testMode) {}
> 
> ISACTest::~ISACTest() {}
>@@ -78,15 +81,15 @@ void ISACTest::Setup() {
>   CodecInst codecParam;
> 
>   for (codecCntr = 0; codecCntr < AudioCodingModule::NumberOfCodecs();
>-      codecCntr++) {
>+       codecCntr++) {
>     EXPECT_EQ(0, AudioCodingModule::Codec(codecCntr, &codecParam));
>-    if (!STR_CASE_CMP(codecParam.plname, "ISAC")
>-        && codecParam.plfreq == 16000) {
>+    if (!STR_CASE_CMP(codecParam.plname, "ISAC") &&
>+        codecParam.plfreq == 16000) {
>       memcpy(&_paramISAC16kHz, &codecParam, sizeof(CodecInst));
>       _idISAC16kHz = codecCntr;
>     }
>-    if (!STR_CASE_CMP(codecParam.plname, "ISAC")
>-        && codecParam.plfreq == 32000) {
>+    if (!STR_CASE_CMP(codecParam.plname, "ISAC") &&
>+        codecParam.plfreq == 32000) {
>       memcpy(&_paramISAC32kHz, &codecParam, sizeof(CodecInst));
>       _idISAC32kHz = codecCntr;
>     }
>@@ -112,8 +115,8 @@ void ISACTest::Setup() {
>   EXPECT_EQ(0, _acmB->RegisterTransportCallback(_channel_B2A.get()));
>   _channel_B2A->RegisterReceiverACM(_acmA.get());
> 
>-  file_name_swb_ = webrtc::test::ResourcePath("audio_coding/testfile32kHz",
>-                                              "pcm");
>+  file_name_swb_ =
>+      webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
> 
>   EXPECT_EQ(0, _acmB->RegisterSendCodec(_paramISAC16kHz));
>   EXPECT_EQ(0, _acmA->RegisterSendCodec(_paramISAC32kHz));
>@@ -210,7 +213,8 @@ void ISACTest::Run10ms() {
>   _outFileB.Write10MsData(audioFrame);
> }
> 
>-void ISACTest::EncodeDecode(int testNr, ACMTestISACConfig& wbISACConfig,
>+void ISACTest::EncodeDecode(int testNr,
>+                            ACMTestISACConfig& wbISACConfig,
>                             ACMTestISACConfig& swbISACConfig) {
>   // Files in Side A and B
>   _inFileA.Open(file_name_swb_, 32000, "rb", true);
>@@ -238,8 +242,8 @@ void ISACTest::EncodeDecode(int testNr, ACMTestISACConfig& wbISACConfig,
>   SetISAConfig(wbISACConfig, _acmB.get(), _testMode);
> 
>   bool adaptiveMode = false;
>-  if ((swbISACConfig.currentRateBitPerSec == -1)
>-      || (wbISACConfig.currentRateBitPerSec == -1)) {
>+  if ((swbISACConfig.currentRateBitPerSec == -1) ||
>+      (wbISACConfig.currentRateBitPerSec == -1)) {
>     adaptiveMode = true;
>   }
>   _myTimer.Reset();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/iSACTest.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/iSACTest.h
>index d0e7d5954f1..22c85b4aa49 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/iSACTest.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/iSACTest.h
>@@ -23,7 +23,7 @@
> #include "modules/audio_coding/test/utility.h"
> 
> #define MAX_FILE_NAME_LENGTH_BYTE 500
>-#define NO_OF_CLIENTS             15
>+#define NO_OF_CLIENTS 15
> 
> namespace webrtc {
> 
>@@ -42,12 +42,14 @@ class ISACTest : public ACMTest {
>   ~ISACTest();
> 
>   void Perform();
>+
>  private:
>   void Setup();
> 
>   void Run10ms();
> 
>-  void EncodeDecode(int testNr, ACMTestISACConfig& wbISACConfig,
>+  void EncodeDecode(int testNr,
>+                    ACMTestISACConfig& wbISACConfig,
>                     ACMTestISACConfig& swbISACConfig);
> 
>   void SwitchingSamplingRate(int testNr, int maxSampRateChange);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/insert_packet_with_timing.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/insert_packet_with_timing.cc
>deleted file mode 100644
>index 2f9bb681e73..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/insert_packet_with_timing.cc
>+++ /dev/null
>@@ -1,323 +0,0 @@
>-/*
>- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include <stdio.h>
>-#include <string.h>
>-
>-#include <memory>
>-
>-#include "common_types.h"  // NOLINT(build/include)
>-#include "modules/audio_coding/codecs/audio_format_conversion.h"
>-#include "modules/audio_coding/include/audio_coding_module.h"
>-#include "modules/audio_coding/test/Channel.h"
>-#include "modules/audio_coding/test/PCMFile.h"
>-#include "modules/include/module_common_types.h"
>-#include "rtc_base/flags.h"
>-#include "system_wrappers/include/clock.h"
>-#include "test/gtest.h"
>-#include "test/testsupport/fileutils.h"
>-
>-// Codec.
>-DEFINE_string(codec, "opus", "Codec Name");
>-DEFINE_int(codec_sample_rate_hz, 48000, "Sampling rate in Hertz.");
>-DEFINE_int(codec_channels, 1, "Number of channels of the codec.");
>-
>-// PCM input/output.
>-DEFINE_string(input, "", "Input PCM file at 16 kHz.");
>-DEFINE_bool(input_stereo, false, "Input is stereo.");
>-DEFINE_int(input_fs_hz, 32000, "Input sample rate Hz.");
>-DEFINE_string(output, "insert_rtp_with_timing_out.pcm", "OutputFile");
>-DEFINE_int(output_fs_hz, 32000, "Output sample rate Hz");
>-
>-// Timing files
>-DEFINE_string(seq_num, "seq_num", "Sequence number file.");
>-DEFINE_string(send_ts, "send_timestamp", "Send timestamp file.");
>-DEFINE_string(receive_ts, "last_rec_timestamp", "Receive timestamp file");
>-
>-// Delay logging
>-DEFINE_string(delay, "", "Log for delay.");
>-
>-// Other setups
>-DEFINE_bool(verbose, false, "Verbosity.");
>-DEFINE_float(loss_rate, 0, "Rate of packet loss < 1");
>-
>-DEFINE_bool(help, false, "Prints this message.");
>-
>-const int32_t kAudioPlayedOut = 0x00000001;
>-const int32_t kPacketPushedIn = 0x00000001 << 1;
>-const int kPlayoutPeriodMs = 10;
>-
>-namespace webrtc {
>-
>-class InsertPacketWithTiming {
>- public:
>-  InsertPacketWithTiming()
>-      : sender_clock_(new SimulatedClock(0)),
>-        receiver_clock_(new SimulatedClock(0)),
>-        send_acm_(AudioCodingModule::Create(sender_clock_)),
>-        receive_acm_(AudioCodingModule::Create(receiver_clock_)),
>-        channel_(new Channel),
>-        seq_num_fid_(fopen(FLAG_seq_num, "rt")),
>-        send_ts_fid_(fopen(FLAG_send_ts, "rt")),
>-        receive_ts_fid_(fopen(FLAG_receive_ts, "rt")),
>-        pcm_out_fid_(fopen(FLAG_output, "wb")),
>-        samples_in_1ms_(48),
>-        num_10ms_in_codec_frame_(2),  // Typical 20 ms frames.
>-        time_to_insert_packet_ms_(3),  // An arbitrary offset on pushing packet.
>-        next_receive_ts_(0),
>-        time_to_playout_audio_ms_(kPlayoutPeriodMs),
>-        loss_threshold_(0),
>-        playout_timing_fid_(fopen("playout_timing.txt", "wt")) {}
>-
>-  void SetUp() {
>-    ASSERT_TRUE(sender_clock_ != NULL);
>-    ASSERT_TRUE(receiver_clock_ != NULL);
>-
>-    ASSERT_TRUE(send_acm_.get() != NULL);
>-    ASSERT_TRUE(receive_acm_.get() != NULL);
>-    ASSERT_TRUE(channel_ != NULL);
>-
>-    ASSERT_TRUE(seq_num_fid_ != NULL);
>-    ASSERT_TRUE(send_ts_fid_ != NULL);
>-    ASSERT_TRUE(receive_ts_fid_ != NULL);
>-
>-    ASSERT_TRUE(playout_timing_fid_ != NULL);
>-
>-    next_receive_ts_ = ReceiveTimestamp();
>-
>-    CodecInst codec;
>-    ASSERT_EQ(0, AudioCodingModule::Codec(FLAG_codec, &codec,
>-                             FLAG_codec_sample_rate_hz,
>-                             FLAG_codec_channels));
>-    ASSERT_EQ(0, receive_acm_->InitializeReceiver());
>-    ASSERT_EQ(0, send_acm_->RegisterSendCodec(codec));
>-    ASSERT_EQ(true, receive_acm_->RegisterReceiveCodec(codec.pltype,
>-                                                       CodecInstToSdp(codec)));
>-
>-    // Set codec-dependent parameters.
>-    samples_in_1ms_ = codec.plfreq / 1000;
>-    num_10ms_in_codec_frame_ = codec.pacsize / (codec.plfreq / 100);
>-
>-    channel_->RegisterReceiverACM(receive_acm_.get());
>-    send_acm_->RegisterTransportCallback(channel_);
>-
>-    if (strlen(FLAG_input) == 0) {
>-      std::string file_name = test::ResourcePath("audio_coding/testfile32kHz",
>-                                                 "pcm");
>-      pcm_in_fid_.Open(file_name, 32000, "r", true);  // auto-rewind
>-      std::cout << "Input file " << file_name << " 32 kHz mono." << std::endl;
>-    } else {
>-      pcm_in_fid_.Open(FLAG_input, static_cast<uint16_t>(FLAG_input_fs_hz),
>-                    "r", true);  // auto-rewind
>-      std::cout << "Input file " << FLAG_input << "at " << FLAG_input_fs_hz
>-          << " Hz in " << ((FLAG_input_stereo) ? "stereo." : "mono.")
>-          << std::endl;
>-      pcm_in_fid_.ReadStereo(FLAG_input_stereo);
>-    }
>-
>-    ASSERT_TRUE(pcm_out_fid_ != NULL);
>-    std::cout << "Output file " << FLAG_output << " at " << FLAG_output_fs_hz
>-        << " Hz." << std::endl;
>-
>-    // Other setups
>-    if (FLAG_loss_rate > 0)
>-      loss_threshold_ = RAND_MAX * FLAG_loss_rate;
>-    else
>-      loss_threshold_ = 0;
>-  }
>-
>-  void TickOneMillisecond(uint32_t* action) {
>-    // One millisecond passed.
>-    time_to_insert_packet_ms_--;
>-    time_to_playout_audio_ms_--;
>-    sender_clock_->AdvanceTimeMilliseconds(1);
>-    receiver_clock_->AdvanceTimeMilliseconds(1);
>-
>-    // Reset action.
>-    *action = 0;
>-
>-    // Is it time to pull audio?
>-    if (time_to_playout_audio_ms_ == 0) {
>-      time_to_playout_audio_ms_ = kPlayoutPeriodMs;
>-      bool muted;
>-      receive_acm_->PlayoutData10Ms(static_cast<int>(FLAG_output_fs_hz),
>-                                    &frame_, &muted);
>-      ASSERT_FALSE(muted);
>-      fwrite(frame_.data(), sizeof(*frame_.data()),
>-             frame_.samples_per_channel_ * frame_.num_channels_, pcm_out_fid_);
>-      *action |= kAudioPlayedOut;
>-    }
>-
>-    // Is it time to push in next packet?
>-    if (time_to_insert_packet_ms_ <= .5) {
>-      *action |= kPacketPushedIn;
>-
>-      // Update time-to-insert packet.
>-      uint32_t t = next_receive_ts_;
>-      next_receive_ts_ = ReceiveTimestamp();
>-      time_to_insert_packet_ms_ += static_cast<float>(next_receive_ts_ - t) /
>-          samples_in_1ms_;
>-
>-      // Push in just enough audio.
>-      for (int n = 0; n < num_10ms_in_codec_frame_; n++) {
>-        pcm_in_fid_.Read10MsData(frame_);
>-        EXPECT_GE(send_acm_->Add10MsData(frame_), 0);
>-      }
>-
>-      // Set the parameters for the packet to be pushed in receiver ACM right
>-      // now.
>-      uint32_t ts = SendTimestamp();
>-      int seq_num = SequenceNumber();
>-      bool lost = false;
>-      channel_->set_send_timestamp(ts);
>-      channel_->set_sequence_number(seq_num);
>-      if (loss_threshold_ > 0 && rand() < loss_threshold_) {
>-        channel_->set_num_packets_to_drop(1);
>-        lost = true;
>-      }
>-
>-      if (FLAG_verbose) {
>-        if (!lost) {
>-          std::cout << "\nInserting packet number " << seq_num
>-              << " timestamp " << ts << std::endl;
>-        } else {
>-          std::cout << "\nLost packet number " << seq_num
>-              << " timestamp " << ts << std::endl;
>-        }
>-      }
>-    }
>-  }
>-
>-  void TearDown() {
>-    delete channel_;
>-
>-    fclose(seq_num_fid_);
>-    fclose(send_ts_fid_);
>-    fclose(receive_ts_fid_);
>-    fclose(pcm_out_fid_);
>-    pcm_in_fid_.Close();
>-  }
>-
>-  ~InsertPacketWithTiming() {
>-    delete sender_clock_;
>-    delete receiver_clock_;
>-  }
>-
>-  // Are there more info to simulate.
>-  bool HasPackets() {
>-    if (feof(seq_num_fid_) || feof(send_ts_fid_) || feof(receive_ts_fid_))
>-      return false;
>-    return true;
>-  }
>-
>-  // Jitter buffer delay.
>-  void Delay(int* optimal_delay, int* current_delay) {
>-    NetworkStatistics statistics;
>-    receive_acm_->GetNetworkStatistics(&statistics);
>-    *optimal_delay = statistics.preferredBufferSize;
>-    *current_delay = statistics.currentBufferSize;
>-  }
>-
>- private:
>-  uint32_t SendTimestamp() {
>-    uint32_t t;
>-    EXPECT_EQ(1, fscanf(send_ts_fid_, "%u\n", &t));
>-    return t;
>-  }
>-
>-  uint32_t ReceiveTimestamp() {
>-    uint32_t t;
>-    EXPECT_EQ(1, fscanf(receive_ts_fid_, "%u\n", &t));
>-    return t;
>-  }
>-
>-  int SequenceNumber() {
>-    int n;
>-    EXPECT_EQ(1, fscanf(seq_num_fid_, "%d\n", &n));
>-    return n;
>-  }
>-
>-  // This class just creates these pointers, not deleting them. They are deleted
>-  // by the associated ACM.
>-  SimulatedClock* sender_clock_;
>-  SimulatedClock* receiver_clock_;
>-
>-  std::unique_ptr<AudioCodingModule> send_acm_;
>-  std::unique_ptr<AudioCodingModule> receive_acm_;
>-  Channel* channel_;
>-
>-  FILE* seq_num_fid_;  // Input (text), one sequence number per line.
>-  FILE* send_ts_fid_;  // Input (text), one send timestamp per line.
>-  FILE* receive_ts_fid_;  // Input (text), one receive timestamp per line.
>-  FILE* pcm_out_fid_;  // Output PCM16.
>-
>-  PCMFile pcm_in_fid_;  // Input PCM16.
>-
>-  int samples_in_1ms_;
>-
>-  // TODO(turajs): this can be computed from the send timestamp, but there is
>-  // some complication to account for lost and reordered packets.
>-  int num_10ms_in_codec_frame_;
>-
>-  float time_to_insert_packet_ms_;
>-  uint32_t next_receive_ts_;
>-  uint32_t time_to_playout_audio_ms_;
>-
>-  AudioFrame frame_;
>-
>-  double loss_threshold_;
>-
>-  // Output (text), sequence number, playout timestamp, time (ms) of playout,
>-  // per line.
>-  FILE* playout_timing_fid_;
>-};
>-
>-}  // webrtc
>-
>-int main(int argc, char* argv[]) {
>-  if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true)) {
>-    return 1;
>-  }
>-  if (FLAG_help) {
>-    rtc::FlagList::Print(nullptr, false);
>-    return 0;
>-  }
>-
>-  webrtc::InsertPacketWithTiming test;
>-  test.SetUp();
>-
>-  FILE* delay_log = NULL;
>-  if (strlen(FLAG_delay) > 0) {
>-    delay_log = fopen(FLAG_delay, "wt");
>-    if (delay_log == NULL) {
>-      std::cout << "Cannot open the file to log delay values." << std::endl;
>-      exit(1);
>-    }
>-  }
>-
>-  uint32_t action_taken;
>-  int optimal_delay_ms;
>-  int current_delay_ms;
>-  while (test.HasPackets()) {
>-    test.TickOneMillisecond(&action_taken);
>-
>-    if (action_taken != 0) {
>-      test.Delay(&optimal_delay_ms, &current_delay_ms);
>-      if (delay_log != NULL) {
>-        fprintf(delay_log, "%3d %3d\n", optimal_delay_ms, current_delay_ms);
>-      }
>-    }
>-  }
>-  std::cout << std::endl;
>-  test.TearDown();
>-  if (delay_log != NULL)
>-    fclose(delay_log);
>-  return 0;
>-}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/opus_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/opus_test.cc
>index b7acc0f7c64..3c17dba66ef 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/opus_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/opus_test.cc
>@@ -14,6 +14,7 @@
> 
> #include <string>
> 
>+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/audio_coding/codecs/audio_format_conversion.h"
> #include "modules/audio_coding/codecs/opus/opus_interface.h"
>@@ -22,12 +23,12 @@
> #include "modules/audio_coding/test/utility.h"
> #include "test/gtest.h"
> #include "test/testsupport/fileutils.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
> OpusTest::OpusTest()
>-    : acm_receiver_(AudioCodingModule::Create()),
>+    : acm_receiver_(AudioCodingModule::Create(
>+          AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))),
>       channel_a2b_(NULL),
>       counter_(0),
>       payload_type_(255),
>@@ -209,8 +210,11 @@ void OpusTest::Perform() {
> #endif
> }
> 
>-void OpusTest::Run(TestPackStereo* channel, size_t channels, int bitrate,
>-                   size_t frame_length, int percent_loss) {
>+void OpusTest::Run(TestPackStereo* channel,
>+                   size_t channels,
>+                   int bitrate,
>+                   size_t frame_length,
>+                   int percent_loss) {
>   AudioFrame audio_frame;
>   int32_t out_freq_hz_b = out_file_.SamplingFrequency();
>   const size_t kBufferSizeSamples = 480 * 12 * 2;  // 120 ms stereo audio.
>@@ -235,8 +239,8 @@ void OpusTest::Run(TestPackStereo* channel, size_t channels, int bitrate,
>   // default.
>   const int kOpusComplexity5 = 5;
>   EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_mono_encoder_, kOpusComplexity5));
>-  EXPECT_EQ(0, WebRtcOpus_SetComplexity(opus_stereo_encoder_,
>-                                        kOpusComplexity5));
>+  EXPECT_EQ(0,
>+            WebRtcOpus_SetComplexity(opus_stereo_encoder_, kOpusComplexity5));
> #endif
> 
>   // Fast-forward 1 second (100 blocks) since the files start with silence.
>@@ -261,19 +265,16 @@ void OpusTest::Run(TestPackStereo* channel, size_t channels, int bitrate,
>     }
> 
>     // If input audio is sampled at 32 kHz, resampling to 48 kHz is required.
>-    EXPECT_EQ(480,
>-              resampler_.Resample10Msec(audio_frame.data(),
>-                                        audio_frame.sample_rate_hz_,
>-                                        48000,
>-                                        channels,
>-                                        kBufferSizeSamples - written_samples,
>-                                        &audio[written_samples]));
>+    EXPECT_EQ(480, resampler_.Resample10Msec(
>+                       audio_frame.data(), audio_frame.sample_rate_hz_, 48000,
>+                       channels, kBufferSizeSamples - written_samples,
>+                       &audio[written_samples]));
>     written_samples += 480 * channels;
> 
>     // Sometimes we need to loop over the audio vector to produce the right
>     // number of packets.
>-    size_t loop_encode = (written_samples - read_samples) /
>-        (channels * frame_length);
>+    size_t loop_encode =
>+        (written_samples - read_samples) / (channels * frame_length);
> 
>     if (loop_encode > 0) {
>       const size_t kMaxBytes = 1000;  // Maximum number of bytes for one packet.
>@@ -317,9 +318,9 @@ void OpusTest::Run(TestPackStereo* channel, size_t channels, int bitrate,
>                 opus_stereo_decoder_, bitstream, bitstream_len_byte,
>                 &out_audio[decoded_samples * channels], &audio_type);
>           } else {
>-            decoded_samples += WebRtcOpus_DecodePlc(
>-                opus_stereo_decoder_, &out_audio[decoded_samples * channels],
>-                1);
>+            decoded_samples +=
>+                WebRtcOpus_DecodePlc(opus_stereo_decoder_,
>+                                     &out_audio[decoded_samples * channels], 1);
>           }
>         }
> 
>@@ -375,14 +376,14 @@ void OpusTest::Run(TestPackStereo* channel, size_t channels, int bitrate,
> void OpusTest::OpenOutFile(int test_number) {
>   std::string file_name;
>   std::stringstream file_stream;
>-  file_stream << webrtc::test::OutputPath() << "opustest_out_"
>-      << test_number << ".pcm";
>+  file_stream << webrtc::test::OutputPath() << "opustest_out_" << test_number
>+              << ".pcm";
>   file_name = file_stream.str();
>   out_file_.Open(file_name, 48000, "wb");
>   file_stream.str("");
>   file_name = file_stream.str();
>   file_stream << webrtc::test::OutputPath() << "opusstandalone_out_"
>-      << test_number << ".pcm";
>+              << test_number << ".pcm";
>   file_name = file_stream.str();
>   out_file_standalone_.Open(file_name, 48000, "wb");
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/opus_test.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/opus_test.h
>index 3e9d9a71878..1356f272314 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/opus_test.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/opus_test.h
>@@ -15,8 +15,8 @@
> 
> #include <memory>
> 
>-#include "modules/audio_coding/codecs/opus/opus_interface.h"
> #include "modules/audio_coding/acm2/acm_resampler.h"
>+#include "modules/audio_coding/codecs/opus/opus_interface.h"
> #include "modules/audio_coding/test/ACMTest.h"
> #include "modules/audio_coding/test/Channel.h"
> #include "modules/audio_coding/test/PCMFile.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/target_delay_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/target_delay_unittest.cc
>index 03135da6c4c..3b129ea2801 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/target_delay_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/target_delay_unittest.cc
>@@ -10,6 +10,8 @@
> 
> #include <memory>
> 
>+#include "api/audio/audio_frame.h"
>+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/audio_coding/codecs/pcm16b/pcm16b.h"
> #include "modules/audio_coding/include/audio_coding_module.h"
>@@ -22,7 +24,9 @@ namespace webrtc {
> 
> class TargetDelayTest : public ::testing::Test {
>  protected:
>-  TargetDelayTest() : acm_(AudioCodingModule::Create()) {}
>+  TargetDelayTest()
>+      : acm_(AudioCodingModule::Create(
>+            AudioCodingModule::Config(CreateBuiltinAudioDecoderFactory()))) {}
> 
>   ~TargetDelayTest() {}
> 
>@@ -39,8 +43,6 @@ class TargetDelayTest : public ::testing::Test {
>     rtp_info_.header.ssrc = 0x12345678;
>     rtp_info_.header.markerBit = false;
>     rtp_info_.header.sequenceNumber = 0;
>-    rtp_info_.type.Audio.channel = 1;
>-    rtp_info_.type.Audio.isCNG = false;
>     rtp_info_.frameType = kAudioFrameSpeech;
> 
>     int16_t audio[kFrameSizeSamples];
>@@ -69,8 +71,8 @@ class TargetDelayTest : public ::testing::Test {
> 
>   void WithTargetDelayBufferNotChanging() {
>     // A target delay that is one packet larger than jitter.
>-    const int kTargetDelayMs = (kInterarrivalJitterPacket + 1) *
>-        kNum10msPerFrame * 10;
>+    const int kTargetDelayMs =
>+        (kInterarrivalJitterPacket + 1) * kNum10msPerFrame * 10;
>     ASSERT_EQ(0, SetMinimumDelay(kTargetDelayMs));
>     for (int n = 0; n < 30; ++n)  // Run enough iterations to fill the buffer.
>       Run(true);
>@@ -87,8 +89,8 @@ class TargetDelayTest : public ::testing::Test {
>     int clean_optimal_delay = GetCurrentOptimalDelayMs();
> 
>     // A relatively large delay.
>-    const int kTargetDelayMs = (kInterarrivalJitterPacket + 10) *
>-        kNum10msPerFrame * 10;
>+    const int kTargetDelayMs =
>+        (kInterarrivalJitterPacket + 10) * kNum10msPerFrame * 10;
>     ASSERT_EQ(0, SetMinimumDelay(kTargetDelayMs));
>     for (int n = 0; n < 300; ++n)  // Run enough iterations to fill the buffer.
>       Run(true);
>@@ -142,8 +144,8 @@ class TargetDelayTest : public ::testing::Test {
>   void Push() {
>     rtp_info_.header.timestamp += kFrameSizeSamples;
>     rtp_info_.header.sequenceNumber++;
>-    ASSERT_EQ(0, acm_->IncomingPacket(payload_, kFrameSizeSamples * 2,
>-                                      rtp_info_));
>+    ASSERT_EQ(0,
>+              acm_->IncomingPacket(payload_, kFrameSizeSamples * 2, rtp_info_));
>   }
> 
>   // Pull audio equivalent to the amount of audio in one RTP packet.
>@@ -191,9 +193,7 @@ class TargetDelayTest : public ::testing::Test {
>     return stats.preferredBufferSize;
>   }
> 
>-  int RequiredDelay() {
>-    return acm_->LeastRequiredDelayMs();
>-  }
>+  int RequiredDelay() { return acm_->LeastRequiredDelayMs(); }
> 
>   std::unique_ptr<AudioCodingModule> acm_;
>   WebRtcRTPHeader rtp_info_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/utility.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/utility.cc
>index 3c64620d3fb..83c25b537e4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/utility.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/utility.cc
>@@ -23,11 +23,7 @@
> 
> namespace webrtc {
> 
>-ACMTestTimer::ACMTestTimer()
>-    : _msec(0),
>-      _sec(0),
>-      _min(0),
>-      _hour(0) {
>+ACMTestTimer::ACMTestTimer() : _msec(0), _sec(0), _min(0), _hour(0) {
>   return;
> }
> 
>@@ -68,12 +64,14 @@ void ACMTestTimer::Tick1sec() {
> 
> void ACMTestTimer::CurrentTimeHMS(char* currTime) {
>   sprintf(currTime, "%4lu:%02u:%06.3f", _hour, _min,
>-          (double) _sec + (double) _msec / 1000.);
>+          (double)_sec + (double)_msec / 1000.);
>   return;
> }
> 
>-void ACMTestTimer::CurrentTime(unsigned long& h, unsigned char& m,
>-                               unsigned char& s, unsigned short& ms) {
>+void ACMTestTimer::CurrentTime(unsigned long& h,
>+                               unsigned char& m,
>+                               unsigned char& s,
>+                               unsigned short& ms) {
>   h = _hour;
>   m = _min;
>   s = _sec;
>@@ -101,9 +99,8 @@ void ACMTestTimer::Adjust() {
> }
> 
> int16_t ChooseCodec(CodecInst& codecInst) {
>-
>   PrintCodecs();
>-  //AudioCodingModule* tmpACM = AudioCodingModule::Create(0);
>+  // AudioCodingModule* tmpACM = AudioCodingModule::Create(0);
>   uint8_t noCodec = AudioCodingModule::NumberOfCodecs();
>   int8_t codecID;
>   bool outOfRange = false;
>@@ -118,7 +115,7 @@ int16_t ChooseCodec(CodecInst& codecInst) {
>     }
>   } while (outOfRange);
> 
>-  CHECK_ERROR(AudioCodingModule::Codec((uint8_t )codecID, &codecInst));
>+  CHECK_ERROR(AudioCodingModule::Codec((uint8_t)codecID, &codecInst));
>   return 0;
> }
> 
>@@ -132,7 +129,6 @@ void PrintCodecs() {
>     printf("%2d- %-18s %5d   %6d\n", codecCntr, codecInst.plname,
>            codecInst.plfreq, codecInst.rate);
>   }
>-
> }
> 
> namespace test {
>@@ -192,7 +188,7 @@ void CircularBuffer::Update(const double newVal) {
>   if (_calcVar) {
>     // to calculate variance we have to update
>     // the sum of squares
>-    _sumSqr += (double) (newVal - oldVal) * (double) (newVal + oldVal);
>+    _sumSqr += (double)(newVal - oldVal) * (double)(newVal + oldVal);
>   }
> }
> 
>@@ -236,17 +232,15 @@ int16_t CircularBuffer::ArithMean(double& mean) {
>   assert(_buffLen > 0);
> 
>   if (_buffIsFull) {
>-
>-    mean = _sum / (double) _buffLen;
>+    mean = _sum / (double)_buffLen;
>     return 0;
>   } else {
>     if (_idx > 0) {
>-      mean = _sum / (double) _idx;
>+      mean = _sum / (double)_idx;
>       return 0;
>     } else {
>       return -1;
>     }
>-
>   }
> }
> 
>@@ -254,11 +248,11 @@ int16_t CircularBuffer::Variance(double& var) {
>   assert(_buffLen > 0);
> 
>   if (_buffIsFull) {
>-    var = _sumSqr / (double) _buffLen;
>+    var = _sumSqr / (double)_buffLen;
>     return 0;
>   } else {
>     if (_idx > 0) {
>-      var = _sumSqr / (double) _idx;
>+      var = _sumSqr / (double)_idx;
>       return 0;
>     } else {
>       return -1;
>@@ -269,9 +263,9 @@ int16_t CircularBuffer::Variance(double& var) {
> }  // namespace test
> 
> bool FixedPayloadTypeCodec(const char* payloadName) {
>-  char fixPayloadTypeCodecs[NUM_CODECS_WITH_FIXED_PAYLOAD_TYPE][32] = { "PCMU",
>-      "PCMA", "GSM", "G723", "DVI4", "LPC", "PCMA", "G722", "QCELP", "CN",
>-      "MPA", "G728", "G729" };
>+  char fixPayloadTypeCodecs[NUM_CODECS_WITH_FIXED_PAYLOAD_TYPE][32] = {
>+      "PCMU", "PCMA",  "GSM", "G723", "DVI4", "LPC", "PCMA",
>+      "G722", "QCELP", "CN",  "MPA",  "G728", "G729"};
> 
>   for (int n = 0; n < NUM_CODECS_WITH_FIXED_PAYLOAD_TYPE; n++) {
>     if (!STR_CASE_CMP(payloadName, fixPayloadTypeCodecs[n])) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/utility.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/utility.h
>index 07cbe7170d6..6f17df5bdf6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/utility.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/test/utility.h
>@@ -17,48 +17,48 @@
> namespace webrtc {
> 
> //-----------------------------
>-#define CHECK_ERROR(f)                                                         \
>-  do {                                                                         \
>-    EXPECT_GE(f, 0) << "Error Calling API";                                    \
>-  } while(0)
>+#define CHECK_ERROR(f)                      \
>+  do {                                      \
>+    EXPECT_GE(f, 0) << "Error Calling API"; \
>+  } while (0)
> 
> //-----------------------------
>-#define CHECK_PROTECTED(f)                                                     \
>-  do {                                                                         \
>-    if (f >= 0) {                                                              \
>-      ADD_FAILURE() << "Error Calling API";                                    \
>-    } else {                                                                   \
>-      printf("An expected error is caught.\n");                                \
>-    }                                                                          \
>-  } while(0)
>+#define CHECK_PROTECTED(f)                      \
>+  do {                                          \
>+    if (f >= 0) {                               \
>+      ADD_FAILURE() << "Error Calling API";     \
>+    } else {                                    \
>+      printf("An expected error is caught.\n"); \
>+    }                                           \
>+  } while (0)
> 
> //----------------------------
>-#define CHECK_ERROR_MT(f)                                                      \
>-  do {                                                                         \
>-    if (f < 0) {                                                               \
>-      fprintf(stderr, "Error Calling API in file %s at line %d \n",            \
>-              __FILE__, __LINE__);                                             \
>-    }                                                                          \
>-  } while(0)
>+#define CHECK_ERROR_MT(f)                                                     \
>+  do {                                                                        \
>+    if (f < 0) {                                                              \
>+      fprintf(stderr, "Error Calling API in file %s at line %d \n", __FILE__, \
>+              __LINE__);                                                      \
>+    }                                                                         \
>+  } while (0)
> 
> //----------------------------
>-#define CHECK_PROTECTED_MT(f)                                                  \
>-  do {                                                                         \
>-    if (f >= 0) {                                                              \
>-      fprintf(stderr, "Error Calling API in file %s at line %d \n",            \
>-              __FILE__, __LINE__);                                             \
>-    } else {                                                                   \
>-      printf("An expected error is caught.\n");                                \
>-    }                                                                          \
>-  } while(0)
>-
>-#define DELETE_POINTER(p)                                                      \
>-  do {                                                                         \
>-    if (p != NULL) {                                                           \
>-      delete p;                                                                \
>-      p = NULL;                                                                \
>-    }                                                                          \
>-  } while(0)
>+#define CHECK_PROTECTED_MT(f)                                                 \
>+  do {                                                                        \
>+    if (f >= 0) {                                                             \
>+      fprintf(stderr, "Error Calling API in file %s at line %d \n", __FILE__, \
>+              __LINE__);                                                      \
>+    } else {                                                                  \
>+      printf("An expected error is caught.\n");                               \
>+    }                                                                         \
>+  } while (0)
>+
>+#define DELETE_POINTER(p) \
>+  do {                    \
>+    if (p != NULL) {      \
>+      delete p;           \
>+      p = NULL;           \
>+    }                     \
>+  } while (0)
> 
> class ACMTestTimer {
>  public:
>@@ -71,7 +71,9 @@ class ACMTestTimer {
>   void Tick100ms();
>   void Tick1sec();
>   void CurrentTimeHMS(char* currTime);
>-  void CurrentTime(unsigned long& h, unsigned char& m, unsigned char& s,
>+  void CurrentTime(unsigned long& h,
>+                   unsigned char& m,
>+                   unsigned char& s,
>                    unsigned short& ms);
> 
>  private:
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/BUILD.gn
>index aeb9fd3d9a6..6eb02a500ee 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/BUILD.gn
>@@ -13,34 +13,13 @@ if (is_android) {
>   import("//build/config/android/rules.gni")
> }
> 
>-config("audio_device_config") {
>-  include_dirs = [
>-    "../include",
>-    "include",
>-    "dummy",  # Contains dummy audio device implementations.
>-  ]
>-}
>-
> config("audio_device_warnings_config") {
>   if (is_win && is_clang) {
>     cflags = [
>       # Disable warnings failing when compiling with Clang on Windows.
>       # https://bugs.chromium.org/p/webrtc/issues/detail?id=5366
>-      "-Wno-bool-conversion",
>       "-Wno-delete-non-virtual-dtor",
>-      "-Wno-logical-op-parentheses",
>-      "-Wno-microsoft-extra-qualification",
>       "-Wno-microsoft-goto",
>-      "-Wno-missing-braces",
>-      "-Wno-parentheses-equality",
>-      "-Wno-reorder",
>-      "-Wno-shift-overflow",
>-      "-Wno-tautological-compare",
>-
>-      # See https://bugs.chromium.org/p/webrtc/issues/detail?id=6265
>-      # for -Wno-thread-safety-analysis
>-      "-Wno-thread-safety-analysis",
>-      "-Wno-unused-private-field",
>     ]
>   }
> }
>@@ -48,16 +27,21 @@ config("audio_device_warnings_config") {
> rtc_source_set("audio_device") {
>   visibility = [ "*" ]
>   public_deps = [
>-    ":audio_device_generic",
>+    ":audio_device_api",
>+
>+    # Deprecated.
>+    # TODO(webrtc:7452): Remove this public dep. audio_device_impl should
>+    # be depended on directly if needed.
>+    ":audio_device_impl",
>   ]
>-  if (rtc_include_internal_audio_device && is_ios) {
>-    public_deps += [ ":audio_device_ios_objc" ]
>-  }
> }
> 
> if (rtc_include_internal_audio_device && is_ios) {
>   rtc_source_set("audio_device_ios_objc") {
>-    visibility = [ ":audio_device" ]
>+    visibility = [
>+      ":audio_device_impl",
>+      ":audio_device_ios_objc_unittests",
>+    ]
>     sources = [
>       "ios/audio_device_ios.h",
>       "ios/audio_device_ios.mm",
>@@ -77,11 +61,14 @@ if (rtc_include_internal_audio_device && is_ios) {
>       "UIKit.framework",
>     ]
>     deps = [
>+      ":audio_device_api",
>+      ":audio_device_buffer",
>       ":audio_device_generic",
>       "../../api:array_view",
>       "../../rtc_base:checks",
>       "../../rtc_base:gtest_prod",
>       "../../rtc_base:rtc_base",
>+      "../../rtc_base/system:fallthrough",
>       "../../sdk:audio_objc",
>       "../../sdk:common_objc",
>       "../../system_wrappers:metrics_api",
>@@ -93,43 +80,156 @@ if (rtc_include_internal_audio_device && is_ios) {
>   }
> }
> 
>+rtc_source_set("audio_device_api") {
>+  visibility = [ "*" ]
>+  sources = [
>+    "include/audio_device.h",
>+    "include/audio_device_defines.h",
>+  ]
>+  deps = [
>+    "../../rtc_base:checks",
>+    "../../rtc_base:deprecation",
>+    "../../rtc_base:rtc_base_approved",
>+    "../../rtc_base:stringutils",
>+  ]
>+  if (!build_with_chromium && is_clang) {
>+    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>+    suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>+  }
>+}
>+
>+rtc_source_set("audio_device_buffer") {
>+  sources = [
>+    "audio_device_buffer.cc",
>+    "audio_device_buffer.h",
>+    "audio_device_config.h",
>+    "fine_audio_buffer.cc",
>+    "fine_audio_buffer.h",
>+  ]
>+  deps = [
>+    ":audio_device_api",
>+    "../../api:array_view",
>+    "../../common_audio:common_audio_c",
>+    "../../rtc_base:checks",
>+    "../../rtc_base:rtc_base_approved",
>+    "../../rtc_base:rtc_task_queue",
>+    "../../system_wrappers",
>+    "../../system_wrappers:metrics_api",
>+  ]
>+  if (!build_with_chromium && is_clang) {
>+    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>+    suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>+  }
>+}
>+
> rtc_source_set("audio_device_generic") {
>-  visibility = [
>-    ":audio_device",
>-    ":audio_device_ios_objc",
>+  sources = [
>+    "audio_device_generic.cc",
>+    "audio_device_generic.h",
>+  ]
>+  deps = [
>+    ":audio_device_api",
>+    ":audio_device_buffer",
>+    "../../rtc_base:rtc_base_approved",
>   ]
>-  public_configs = [ ":audio_device_config" ]
>+  if (!build_with_chromium && is_clang) {
>+    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>+    suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>+  }
>+}
> 
>+rtc_source_set("audio_device_name") {
>+  sources = [
>+    "audio_device_name.cc",
>+    "audio_device_name.h",
>+  ]
>+}
>+
>+rtc_source_set("windows_core_audio_utility") {
>+  if (is_win && !build_with_chromium) {
>+    sources = [
>+      "win/core_audio_utility_win.cc",
>+      "win/core_audio_utility_win.h",
>+    ]
>+
>+    deps = [
>+      ":audio_device_api",
>+      ":audio_device_name",
>+      "../../api/units:time_delta",
>+      "../../rtc_base:checks",
>+      "../../rtc_base:rtc_base_approved",
>+    ]
>+  }
>+}
>+
>+# An ADM with a dedicated factory method which does not depend on the
>+# audio_device_impl target. The goal is to use this new structure and
>+# gradually phase out the old design.
>+# TODO(henrika): currently only supported on Windows.
>+rtc_source_set("audio_device_module_from_input_and_output") {
>+  visibility = [ ":*" ]
>+
>+  if (is_win && !build_with_chromium) {
>+    sources = [
>+      "include/audio_device_factory.cc",
>+      "include/audio_device_factory.h",
>+    ]
>+    sources += [
>+      "win/audio_device_module_win.cc",
>+      "win/audio_device_module_win.h",
>+      "win/core_audio_base_win.cc",
>+      "win/core_audio_base_win.h",
>+      "win/core_audio_input_win.cc",
>+      "win/core_audio_input_win.h",
>+      "win/core_audio_output_win.cc",
>+      "win/core_audio_output_win.h",
>+    ]
>+
>+    deps = [
>+      ":audio_device_api",
>+      ":audio_device_buffer",
>+      ":windows_core_audio_utility",
>+      "../../rtc_base:checks",
>+      "../../rtc_base:rtc_base_approved",
>+      "//third_party/abseil-cpp/absl/memory",
>+      "//third_party/abseil-cpp/absl/types:optional",
>+    ]
>+  }
>+}
>+
>+# Contains default implementations of webrtc::AudioDeviceModule for Windows,
>+# Linux, Mac, iOS and Android.
>+rtc_source_set("audio_device_impl") {
>+  visibility = [ "*" ]
>   deps = [
>-    "..:module_api",
>+    ":audio_device_api",
>+    ":audio_device_buffer",
>+    ":audio_device_generic",
>     "../..:webrtc_common",
>-    "../../:typedefs",
>     "../../api:array_view",
>     "../../common_audio",
>+    "../../common_audio:common_audio_c",
>     "../../rtc_base:checks",
>     "../../rtc_base:deprecation",
>     "../../rtc_base:rtc_base_approved",
>     "../../rtc_base:rtc_task_queue",
>+    "../../rtc_base/system:file_wrapper",
>     "../../system_wrappers",
>     "../../system_wrappers:metrics_api",
>     "../utility",
>   ]
>+  if (rtc_include_internal_audio_device && is_ios) {
>+    deps += [ ":audio_device_ios_objc" ]
>+  }
> 
>   sources = [
>-    "audio_device_buffer.cc",
>-    "audio_device_buffer.h",
>-    "audio_device_config.h",
>-    "audio_device_generic.cc",
>-    "audio_device_generic.h",
>     "dummy/audio_device_dummy.cc",
>     "dummy/audio_device_dummy.h",
>     "dummy/file_audio_device.cc",
>     "dummy/file_audio_device.h",
>-    "fine_audio_buffer.cc",
>-    "fine_audio_buffer.h",
>-    "include/audio_device.h",
>-    "include/audio_device_defines.h",
>     "include/fake_audio_device.h",
>+    "include/test_audio_device.cc",
>+    "include/test_audio_device.h",
>   ]
> 
>   if (build_with_mozilla) {
>@@ -139,28 +239,17 @@ rtc_source_set("audio_device_generic") {
>     ]
>   }
> 
>-  include_dirs = []
>-  if (is_linux) {
>-    include_dirs += [ "linux" ]
>-  }
>-  if (is_ios) {
>-    include_dirs += [ "ios" ]
>-  }
>-  if (is_mac) {
>-    include_dirs += [ "mac" ]
>-  }
>-  if (is_win) {
>-    include_dirs += [ "win" ]
>-  }
>-  if (is_android) {
>-    include_dirs += [ "android" ]
>-  }
>   defines = []
>   cflags = []
>   if (rtc_audio_device_plays_sinus_tone) {
>     defines += [ "AUDIO_DEVICE_PLAYS_SINUS_TONE" ]
>   }
>+  if (rtc_enable_android_aaudio) {
>+    defines += [ "AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO" ]
>+  }
>   if (rtc_include_internal_audio_device) {
>+    # TODO(bugs.webrtc.org/8850): remove this when the circular dependency will be fixed.
>+    check_includes = false
>     sources += [
>       "audio_device_data_observer.cc",
>       "audio_device_impl.cc",
>@@ -190,6 +279,17 @@ rtc_source_set("audio_device_generic") {
>         "log",
>         "OpenSLES",
>       ]
>+      if (rtc_enable_android_aaudio) {
>+        sources += [
>+          "android/aaudio_player.cc",
>+          "android/aaudio_player.h",
>+          "android/aaudio_recorder.cc",
>+          "android/aaudio_recorder.h",
>+          "android/aaudio_wrapper.cc",
>+          "android/aaudio_wrapper.h",
>+        ]
>+        libs += [ "aaudio" ]
>+      }
> 
>       if (build_with_mozilla) {
>         include_dirs += [
>@@ -238,7 +338,7 @@ rtc_source_set("audio_device_generic") {
>           "mac/audio_mixer_manager_mac.cc",
>           "mac/audio_mixer_manager_mac.h",
>         ]
>-        deps += [ ":mac_portaudio" ]
>+        deps += [ "../third_party/portaudio:mac_portaudio" ]
>         libs = [
>           # Needed for CoreGraphics:
>           "ApplicationServices.framework",
>@@ -283,26 +383,6 @@ rtc_source_set("audio_device_generic") {
>   }
> }
> 
>-rtc_source_set("mac_portaudio") {
>-  visibility = [ ":*" ]  # Only targets in this file can depend on this.
>-  sources = [
>-    "mac/portaudio/pa_memorybarrier.h",
>-    "mac/portaudio/pa_ringbuffer.c",
>-    "mac/portaudio/pa_ringbuffer.h",
>-  ]
>-}
>-
>-config("mock_audio_device_config") {
>-  if (is_win) {
>-    cflags = [
>-      # TODO(phoglund): get rid of 4373 supression when
>-      # http://code.google.com/p/webrtc/issues/detail?id=261 is solved.
>-      # legacy warning for ignoring const / volatile in signatures.
>-      "/wd4373",
>-    ]
>-  }
>-}
>-
> rtc_source_set("mock_audio_device") {
>   testonly = true
>   sources = [
>@@ -312,9 +392,10 @@ rtc_source_set("mock_audio_device") {
>   ]
>   deps = [
>     ":audio_device",
>+    ":audio_device_buffer",
>+    ":audio_device_impl",
>     "../../test:test_support",
>   ]
>-  all_dependent_configs = [ ":mock_audio_device_config" ]
> }
> 
> if (rtc_include_tests) {
>@@ -329,12 +410,15 @@ if (rtc_include_tests) {
>       ]
>       deps = [
>         ":audio_device",
>+        ":audio_device_buffer",
>+        ":audio_device_impl",
>+        ":audio_device_ios_objc",
>         ":mock_audio_device",
>         "../../rtc_base:rtc_base_approved",
>         "../../sdk:audio_objc",
>         "../../system_wrappers",
>+        "../../test:fileutils",
>         "../../test:test_support",
>-        "//testing/gmock",
>         "//third_party/ocmock",
>       ]
>       if (!build_with_chromium && is_clang) {
>@@ -349,22 +433,33 @@ if (rtc_include_tests) {
> 
>     sources = [
>       "fine_audio_buffer_unittest.cc",
>+      "include/test_audio_device_unittest.cc",
>     ]
>     deps = [
>       ":audio_device",
>+      ":audio_device_buffer",
>+      ":audio_device_impl",
>       ":mock_audio_device",
>       "../../api:array_view",
>-      "../../api:optional",
>+      "../../common_audio",
>       "../../rtc_base:checks",
>       "../../rtc_base:rtc_base_approved",
>       "../../system_wrappers",
>+      "../../test:fileutils",
>       "../../test:test_support",
>       "../utility:utility",
>-      "//testing/gmock",
>+      "//third_party/abseil-cpp/absl/types:optional",
>     ]
>     if (is_linux || is_mac || is_win) {
>       sources += [ "audio_device_unittest.cc" ]
>     }
>+    if (is_win) {
>+      sources += [ "win/core_audio_utility_win_unittest.cc" ]
>+      deps += [
>+        ":audio_device_module_from_input_and_output",
>+        ":windows_core_audio_utility",
>+      ]
>+    }
>     if (is_android) {
>       # Need to disable error due to the line in
>       # base/android/jni_android.h triggering it:
>@@ -378,7 +473,7 @@ if (rtc_include_tests) {
>         "android/ensure_initialized.h",
>       ]
>       deps += [
>-        "../../../base",
>+        "../../base",
>         "../../sdk/android:libjingle_peerconnection_java",
>       ]
>     }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/aaudio_player.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/aaudio_player.cc
>new file mode 100644
>index 00000000000..e576a787fad
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/aaudio_player.cc
>@@ -0,0 +1,227 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_device/android/aaudio_player.h"
>+
>+#include "absl/memory/memory.h"
>+#include "api/array_view.h"
>+#include "modules/audio_device/android/audio_manager.h"
>+#include "modules/audio_device/fine_audio_buffer.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/logging.h"
>+
>+namespace webrtc {
>+
>+enum AudioDeviceMessageType : uint32_t {
>+  kMessageOutputStreamDisconnected,
>+};
>+
>+AAudioPlayer::AAudioPlayer(AudioManager* audio_manager)
>+    : main_thread_(rtc::Thread::Current()),
>+      aaudio_(audio_manager, AAUDIO_DIRECTION_OUTPUT, this) {
>+  RTC_LOG(INFO) << "ctor";
>+  thread_checker_aaudio_.DetachFromThread();
>+}
>+
>+AAudioPlayer::~AAudioPlayer() {
>+  RTC_LOG(INFO) << "dtor";
>+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
>+  Terminate();
>+  RTC_LOG(INFO) << "#detected underruns: " << underrun_count_;
>+}
>+
>+int AAudioPlayer::Init() {
>+  RTC_LOG(INFO) << "Init";
>+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
>+  if (aaudio_.audio_parameters().channels() == 2) {
>+    RTC_DLOG(LS_WARNING) << "Stereo mode is enabled";
>+  }
>+  return 0;
>+}
>+
>+int AAudioPlayer::Terminate() {
>+  RTC_LOG(INFO) << "Terminate";
>+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
>+  StopPlayout();
>+  return 0;
>+}
>+
>+int AAudioPlayer::InitPlayout() {
>+  RTC_LOG(INFO) << "InitPlayout";
>+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
>+  RTC_DCHECK(!initialized_);
>+  RTC_DCHECK(!playing_);
>+  if (!aaudio_.Init()) {
>+    return -1;
>+  }
>+  initialized_ = true;
>+  return 0;
>+}
>+
>+bool AAudioPlayer::PlayoutIsInitialized() const {
>+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
>+  return initialized_;
>+}
>+
>+int AAudioPlayer::StartPlayout() {
>+  RTC_LOG(INFO) << "StartPlayout";
>+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
>+  RTC_DCHECK(!playing_);
>+  if (!initialized_) {
>+    RTC_DLOG(LS_WARNING)
>+        << "Playout can not start since InitPlayout must succeed first";
>+    return 0;
>+  }
>+  if (fine_audio_buffer_) {
>+    fine_audio_buffer_->ResetPlayout();
>+  }
>+  if (!aaudio_.Start()) {
>+    return -1;
>+  }
>+  underrun_count_ = aaudio_.xrun_count();
>+  first_data_callback_ = true;
>+  playing_ = true;
>+  return 0;
>+}
>+
>+int AAudioPlayer::StopPlayout() {
>+  RTC_LOG(INFO) << "StopPlayout";
>+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
>+  if (!initialized_ || !playing_) {
>+    return 0;
>+  }
>+  if (!aaudio_.Stop()) {
>+    RTC_LOG(LS_ERROR) << "StopPlayout failed";
>+    return -1;
>+  }
>+  thread_checker_aaudio_.DetachFromThread();
>+  initialized_ = false;
>+  playing_ = false;
>+  return 0;
>+}
>+
>+bool AAudioPlayer::Playing() const {
>+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
>+  return playing_;
>+}
>+
>+void AAudioPlayer::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
>+  RTC_DLOG(INFO) << "AttachAudioBuffer";
>+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
>+  audio_device_buffer_ = audioBuffer;
>+  const AudioParameters audio_parameters = aaudio_.audio_parameters();
>+  audio_device_buffer_->SetPlayoutSampleRate(audio_parameters.sample_rate());
>+  audio_device_buffer_->SetPlayoutChannels(audio_parameters.channels());
>+  RTC_CHECK(audio_device_buffer_);
>+  // Create a modified audio buffer class which allows us to ask for any number
>+  // of samples (and not only multiple of 10ms) to match the optimal buffer
>+  // size per callback used by AAudio.
>+  fine_audio_buffer_ = absl::make_unique<FineAudioBuffer>(audio_device_buffer_);
>+}
>+
>+int AAudioPlayer::SpeakerVolumeIsAvailable(bool& available) {
>+  available = false;
>+  return 0;
>+}
>+
>+void AAudioPlayer::OnErrorCallback(aaudio_result_t error) {
>+  RTC_LOG(LS_ERROR) << "OnErrorCallback: " << AAudio_convertResultToText(error);
>+  // TODO(henrika): investigate if we can use a thread checker here. Initial
>+  // tests shows that this callback can sometimes be called on a unique thread
>+  // but according to the documentation it should be on the same thread as the
>+  // data callback.
>+  // RTC_DCHECK_RUN_ON(&thread_checker_aaudio_);
>+  if (aaudio_.stream_state() == AAUDIO_STREAM_STATE_DISCONNECTED) {
>+    // The stream is disconnected and any attempt to use it will return
>+    // AAUDIO_ERROR_DISCONNECTED.
>+    RTC_LOG(WARNING) << "Output stream disconnected";
>+    // AAudio documentation states: "You should not close or reopen the stream
>+    // from the callback, use another thread instead". A message is therefore
>+    // sent to the main thread to do the restart operation.
>+    RTC_DCHECK(main_thread_);
>+    main_thread_->Post(RTC_FROM_HERE, this, kMessageOutputStreamDisconnected);
>+  }
>+}
>+
>+aaudio_data_callback_result_t AAudioPlayer::OnDataCallback(void* audio_data,
>+                                                           int32_t num_frames) {
>+  RTC_DCHECK_RUN_ON(&thread_checker_aaudio_);
>+  // Log device id in first data callback to ensure that a valid device is
>+  // utilized.
>+  if (first_data_callback_) {
>+    RTC_LOG(INFO) << "--- First output data callback: "
>+                  << "device id=" << aaudio_.device_id();
>+    first_data_callback_ = false;
>+  }
>+
>+  // Check if the underrun count has increased. If it has, increase the buffer
>+  // size by adding the size of a burst. It will reduce the risk of underruns
>+  // at the expense of an increased latency.
>+  // TODO(henrika): enable possibility to disable and/or tune the algorithm.
>+  const int32_t underrun_count = aaudio_.xrun_count();
>+  if (underrun_count > underrun_count_) {
>+    RTC_LOG(LS_ERROR) << "Underrun detected: " << underrun_count;
>+    underrun_count_ = underrun_count;
>+    aaudio_.IncreaseOutputBufferSize();
>+  }
>+
>+  // Estimate latency between writing an audio frame to the output stream and
>+  // the time that same frame is played out on the output audio device.
>+  latency_millis_ = aaudio_.EstimateLatencyMillis();
>+  // TODO(henrika): use for development only.
>+  if (aaudio_.frames_written() % (1000 * aaudio_.frames_per_burst()) == 0) {
>+    RTC_DLOG(INFO) << "output latency: " << latency_millis_
>+                   << ", num_frames: " << num_frames;
>+  }
>+
>+  // Read audio data from the WebRTC source using the FineAudioBuffer object
>+  // and write that data into |audio_data| to be played out by AAudio.
>+  // Prime output with zeros during a short initial phase to avoid distortion.
>+  // TODO(henrika): do more work to figure out of if the initial forced silence
>+  // period is really needed.
>+  if (aaudio_.frames_written() < 50 * aaudio_.frames_per_burst()) {
>+    const size_t num_bytes =
>+        sizeof(int16_t) * aaudio_.samples_per_frame() * num_frames;
>+    memset(audio_data, 0, num_bytes);
>+  } else {
>+    fine_audio_buffer_->GetPlayoutData(
>+        rtc::MakeArrayView(static_cast<int16_t*>(audio_data),
>+                           aaudio_.samples_per_frame() * num_frames),
>+        static_cast<int>(latency_millis_ + 0.5));
>+  }
>+
>+  // TODO(henrika): possibly add trace here to be included in systrace.
>+  // See https://developer.android.com/studio/profile/systrace-commandline.html.
>+  return AAUDIO_CALLBACK_RESULT_CONTINUE;
>+}
>+
>+void AAudioPlayer::OnMessage(rtc::Message* msg) {
>+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
>+  switch (msg->message_id) {
>+    case kMessageOutputStreamDisconnected:
>+      HandleStreamDisconnected();
>+      break;
>+  }
>+}
>+
>+void AAudioPlayer::HandleStreamDisconnected() {
>+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
>+  RTC_DLOG(INFO) << "HandleStreamDisconnected";
>+  if (!initialized_ || !playing_) {
>+    return;
>+  }
>+  // Perform a restart by first closing the disconnected stream and then start
>+  // a new stream; this time using the new (preferred) audio output device.
>+  audio_device_buffer_->NativeAudioPlayoutInterrupted();
>+  StopPlayout();
>+  InitPlayout();
>+  StartPlayout();
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/aaudio_player.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/aaudio_player.h
>new file mode 100644
>index 00000000000..a92d508211e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/aaudio_player.h
>@@ -0,0 +1,146 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_DEVICE_ANDROID_AAUDIO_PLAYER_H_
>+#define MODULES_AUDIO_DEVICE_ANDROID_AAUDIO_PLAYER_H_
>+
>+#include <aaudio/AAudio.h>
>+#include <memory>
>+
>+#include "modules/audio_device/android/aaudio_wrapper.h"
>+#include "modules/audio_device/include/audio_device_defines.h"
>+#include "rtc_base/messagehandler.h"
>+#include "rtc_base/thread.h"
>+#include "rtc_base/thread_annotations.h"
>+#include "rtc_base/thread_checker.h"
>+
>+namespace webrtc {
>+
>+class AudioDeviceBuffer;
>+class FineAudioBuffer;
>+class AudioManager;
>+
>+// Implements low-latency 16-bit mono PCM audio output support for Android
>+// using the C based AAudio API.
>+//
>+// An instance must be created and destroyed on one and the same thread.
>+// All public methods must also be called on the same thread. A thread checker
>+// will DCHECK if any method is called on an invalid thread. Audio buffers
>+// are requested on a dedicated high-priority thread owned by AAudio.
>+//
>+// The existing design forces the user to call InitPlayout() after StopPlayout()
>+// to be able to call StartPlayout() again. This is in line with how the Java-
>+// based implementation works.
>+//
>+// An audio stream can be disconnected, e.g. when an audio device is removed.
>+// This implementation will restart the audio stream using the new preferred
>+// device if such an event happens.
>+//
>+// Also supports automatic buffer-size adjustment based on underrun detections
>+// where the internal AAudio buffer can be increased when needed. It will
>+// reduce the risk of underruns (~glitches) at the expense of an increased
>+// latency.
>+class AAudioPlayer final : public AAudioObserverInterface,
>+                           public rtc::MessageHandler {
>+ public:
>+  explicit AAudioPlayer(AudioManager* audio_manager);
>+  ~AAudioPlayer();
>+
>+  int Init();
>+  int Terminate();
>+
>+  int InitPlayout();
>+  bool PlayoutIsInitialized() const;
>+
>+  int StartPlayout();
>+  int StopPlayout();
>+  bool Playing() const;
>+
>+  void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
>+
>+  // Not implemented in AAudio.
>+  int SpeakerVolumeIsAvailable(bool& available);  // NOLINT
>+  int SetSpeakerVolume(uint32_t volume) { return -1; }
>+  int SpeakerVolume(uint32_t& volume) const { return -1; }        // NOLINT
>+  int MaxSpeakerVolume(uint32_t& maxVolume) const { return -1; }  // NOLINT
>+  int MinSpeakerVolume(uint32_t& minVolume) const { return -1; }  // NOLINT
>+
>+ protected:
>+  // AAudioObserverInterface implementation.
>+
>+  // For an output stream, this function should render and write |num_frames|
>+  // of data in the streams current data format to the |audio_data| buffer.
>+  // Called on a real-time thread owned by AAudio.
>+  aaudio_data_callback_result_t OnDataCallback(void* audio_data,
>+                                               int32_t num_frames) override;
>+  // AAudio calls this functions if any error occurs on a callback thread.
>+  // Called on a real-time thread owned by AAudio.
>+  void OnErrorCallback(aaudio_result_t error) override;
>+
>+  // rtc::MessageHandler used for restart messages from the error-callback
>+  // thread to the main (creating) thread.
>+  void OnMessage(rtc::Message* msg) override;
>+
>+ private:
>+  // Closes the existing stream and starts a new stream.
>+  void HandleStreamDisconnected();
>+
>+  // Ensures that methods are called from the same thread as this object is
>+  // created on.
>+  rtc::ThreadChecker main_thread_checker_;
>+
>+  // Stores thread ID in first call to AAudioPlayer::OnDataCallback from a
>+  // real-time thread owned by AAudio. Detached during construction of this
>+  // object.
>+  rtc::ThreadChecker thread_checker_aaudio_;
>+
>+  // The thread on which this object is created on.
>+  rtc::Thread* main_thread_;
>+
>+  // Wraps all AAudio resources. Contains an output stream using the default
>+  // output audio device. Can be accessed on both the main thread and the
>+  // real-time thread owned by AAudio. See separate AAudio documentation about
>+  // thread safety.
>+  AAudioWrapper aaudio_;
>+
>+  // FineAudioBuffer takes an AudioDeviceBuffer which delivers audio data
>+  // in chunks of 10ms. It then allows for this data to be pulled in
>+  // a finer or coarser granularity. I.e. interacting with this class instead
>+  // of directly with the AudioDeviceBuffer one can ask for any number of
>+  // audio data samples.
>+  // Example: native buffer size can be 192 audio frames at 48kHz sample rate.
>+  // WebRTC will provide 480 audio frames per 10ms but AAudio asks for 192
>+  // in each callback (once every 4th ms). This class can then ask for 192 and
>+  // the FineAudioBuffer will ask WebRTC for new data approximately only every
>+  // second callback and also cache non-utilized audio.
>+  std::unique_ptr<FineAudioBuffer> fine_audio_buffer_;
>+
>+  // Counts number of detected underrun events reported by AAudio.
>+  int32_t underrun_count_ = 0;
>+
>+  // True only for the first data callback in each audio session.
>+  bool first_data_callback_ = true;
>+
>+  // Raw pointer handle provided to us in AttachAudioBuffer(). Owned by the
>+  // AudioDeviceModuleImpl class and set by AudioDeviceModule::Create().
>+  AudioDeviceBuffer* audio_device_buffer_ RTC_GUARDED_BY(main_thread_checker_) =
>+      nullptr;
>+
>+  bool initialized_ RTC_GUARDED_BY(main_thread_checker_) = false;
>+  bool playing_ RTC_GUARDED_BY(main_thread_checker_) = false;
>+
>+  // Estimated latency between writing an audio frame to the output stream and
>+  // the time that same frame is played out on the output audio device.
>+  double latency_millis_ RTC_GUARDED_BY(thread_checker_aaudio_) = 0;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_DEVICE_ANDROID_AAUDIO_PLAYER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/aaudio_recorder.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/aaudio_recorder.cc
>new file mode 100644
>index 00000000000..081dda4c64b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/aaudio_recorder.cc
>@@ -0,0 +1,220 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_device/android/aaudio_recorder.h"
>+
>+#include "absl/memory/memory.h"
>+#include "api/array_view.h"
>+#include "modules/audio_device/android/audio_manager.h"
>+#include "modules/audio_device/fine_audio_buffer.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/logging.h"
>+#include "rtc_base/timeutils.h"
>+#include "system_wrappers/include/sleep.h"
>+
>+namespace webrtc {
>+
>+enum AudioDeviceMessageType : uint32_t {
>+  kMessageInputStreamDisconnected,
>+};
>+
>+AAudioRecorder::AAudioRecorder(AudioManager* audio_manager)
>+    : main_thread_(rtc::Thread::Current()),
>+      aaudio_(audio_manager, AAUDIO_DIRECTION_INPUT, this) {
>+  RTC_LOG(INFO) << "ctor";
>+  thread_checker_aaudio_.DetachFromThread();
>+}
>+
>+AAudioRecorder::~AAudioRecorder() {
>+  RTC_LOG(INFO) << "dtor";
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  Terminate();
>+  RTC_LOG(INFO) << "detected owerflows: " << overflow_count_;
>+}
>+
>+int AAudioRecorder::Init() {
>+  RTC_LOG(INFO) << "Init";
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  if (aaudio_.audio_parameters().channels() == 2) {
>+    RTC_DLOG(LS_WARNING) << "Stereo mode is enabled";
>+  }
>+  return 0;
>+}
>+
>+int AAudioRecorder::Terminate() {
>+  RTC_LOG(INFO) << "Terminate";
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  StopRecording();
>+  return 0;
>+}
>+
>+int AAudioRecorder::InitRecording() {
>+  RTC_LOG(INFO) << "InitRecording";
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  RTC_DCHECK(!initialized_);
>+  RTC_DCHECK(!recording_);
>+  if (!aaudio_.Init()) {
>+    return -1;
>+  }
>+  initialized_ = true;
>+  return 0;
>+}
>+
>+int AAudioRecorder::StartRecording() {
>+  RTC_LOG(INFO) << "StartRecording";
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  RTC_DCHECK(initialized_);
>+  RTC_DCHECK(!recording_);
>+  if (fine_audio_buffer_) {
>+    fine_audio_buffer_->ResetPlayout();
>+  }
>+  if (!aaudio_.Start()) {
>+    return -1;
>+  }
>+  overflow_count_ = aaudio_.xrun_count();
>+  first_data_callback_ = true;
>+  recording_ = true;
>+  return 0;
>+}
>+
>+int AAudioRecorder::StopRecording() {
>+  RTC_LOG(INFO) << "StopRecording";
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  if (!initialized_ || !recording_) {
>+    return 0;
>+  }
>+  if (!aaudio_.Stop()) {
>+    return -1;
>+  }
>+  thread_checker_aaudio_.DetachFromThread();
>+  initialized_ = false;
>+  recording_ = false;
>+  return 0;
>+}
>+
>+void AAudioRecorder::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
>+  RTC_LOG(INFO) << "AttachAudioBuffer";
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  audio_device_buffer_ = audioBuffer;
>+  const AudioParameters audio_parameters = aaudio_.audio_parameters();
>+  audio_device_buffer_->SetRecordingSampleRate(audio_parameters.sample_rate());
>+  audio_device_buffer_->SetRecordingChannels(audio_parameters.channels());
>+  RTC_CHECK(audio_device_buffer_);
>+  // Create a modified audio buffer class which allows us to deliver any number
>+  // of samples (and not only multiples of 10ms which WebRTC uses) to match the
>+  // native AAudio buffer size.
>+  fine_audio_buffer_ = absl::make_unique<FineAudioBuffer>(audio_device_buffer_);
>+}
>+
>+int AAudioRecorder::EnableBuiltInAEC(bool enable) {
>+  RTC_LOG(INFO) << "EnableBuiltInAEC: " << enable;
>+  RTC_LOG(LS_ERROR) << "Not implemented";
>+  return -1;
>+}
>+
>+int AAudioRecorder::EnableBuiltInAGC(bool enable) {
>+  RTC_LOG(INFO) << "EnableBuiltInAGC: " << enable;
>+  RTC_LOG(LS_ERROR) << "Not implemented";
>+  return -1;
>+}
>+
>+int AAudioRecorder::EnableBuiltInNS(bool enable) {
>+  RTC_LOG(INFO) << "EnableBuiltInNS: " << enable;
>+  RTC_LOG(LS_ERROR) << "Not implemented";
>+  return -1;
>+}
>+
>+void AAudioRecorder::OnErrorCallback(aaudio_result_t error) {
>+  RTC_LOG(LS_ERROR) << "OnErrorCallback: " << AAudio_convertResultToText(error);
>+  // RTC_DCHECK(thread_checker_aaudio_.CalledOnValidThread());
>+  if (aaudio_.stream_state() == AAUDIO_STREAM_STATE_DISCONNECTED) {
>+    // The stream is disconnected and any attempt to use it will return
>+    // AAUDIO_ERROR_DISCONNECTED..
>+    RTC_LOG(WARNING) << "Input stream disconnected => restart is required";
>+    // AAudio documentation states: "You should not close or reopen the stream
>+    // from the callback, use another thread instead". A message is therefore
>+    // sent to the main thread to do the restart operation.
>+    RTC_DCHECK(main_thread_);
>+    main_thread_->Post(RTC_FROM_HERE, this, kMessageInputStreamDisconnected);
>+  }
>+}
>+
>+// Read and process |num_frames| of data from the |audio_data| buffer.
>+// TODO(henrika): possibly add trace here to be included in systrace.
>+// See https://developer.android.com/studio/profile/systrace-commandline.html.
>+aaudio_data_callback_result_t AAudioRecorder::OnDataCallback(
>+    void* audio_data,
>+    int32_t num_frames) {
>+  // TODO(henrika): figure out why we sometimes hit this one.
>+  // RTC_DCHECK(thread_checker_aaudio_.CalledOnValidThread());
>+  // RTC_LOG(INFO) << "OnDataCallback: " << num_frames;
>+  // Drain the input buffer at first callback to ensure that it does not
>+  // contain any old data. Will also ensure that the lowest possible latency
>+  // is obtained.
>+  if (first_data_callback_) {
>+    RTC_LOG(INFO) << "--- First input data callback: "
>+                  << "device id=" << aaudio_.device_id();
>+    aaudio_.ClearInputStream(audio_data, num_frames);
>+    first_data_callback_ = false;
>+  }
>+  // Check if the overflow counter has increased and if so log a warning.
>+  // TODO(henrika): possible add UMA stat or capacity extension.
>+  const int32_t overflow_count = aaudio_.xrun_count();
>+  if (overflow_count > overflow_count_) {
>+    RTC_LOG(LS_ERROR) << "Overflow detected: " << overflow_count;
>+    overflow_count_ = overflow_count;
>+  }
>+  // Estimated time between an audio frame was recorded by the input device and
>+  // it can read on the input stream.
>+  latency_millis_ = aaudio_.EstimateLatencyMillis();
>+  // TODO(henrika): use for development only.
>+  if (aaudio_.frames_read() % (1000 * aaudio_.frames_per_burst()) == 0) {
>+    RTC_DLOG(INFO) << "input latency: " << latency_millis_
>+                   << ", num_frames: " << num_frames;
>+  }
>+  // Copy recorded audio in |audio_data| to the WebRTC sink using the
>+  // FineAudioBuffer object.
>+  fine_audio_buffer_->DeliverRecordedData(
>+      rtc::MakeArrayView(static_cast<const int16_t*>(audio_data),
>+                         aaudio_.samples_per_frame() * num_frames),
>+      static_cast<int>(latency_millis_ + 0.5));
>+
>+  return AAUDIO_CALLBACK_RESULT_CONTINUE;
>+}
>+
>+void AAudioRecorder::OnMessage(rtc::Message* msg) {
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  switch (msg->message_id) {
>+    case kMessageInputStreamDisconnected:
>+      HandleStreamDisconnected();
>+      break;
>+    default:
>+      RTC_LOG(LS_ERROR) << "Invalid message id: " << msg->message_id;
>+      break;
>+  }
>+}
>+
>+void AAudioRecorder::HandleStreamDisconnected() {
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  RTC_LOG(INFO) << "HandleStreamDisconnected";
>+  if (!initialized_ || !recording_) {
>+    return;
>+  }
>+  // Perform a restart by first closing the disconnected stream and then start
>+  // a new stream; this time using the new (preferred) audio input device.
>+  // TODO(henrika): resolve issue where a one restart attempt leads to a long
>+  // sequence of new calls to OnErrorCallback().
>+  // See b/73148976 for details.
>+  audio_device_buffer_->NativeAudioRecordingInterrupted();
>+  StopRecording();
>+  InitRecording();
>+  StartRecording();
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/aaudio_recorder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/aaudio_recorder.h
>new file mode 100644
>index 00000000000..1b78a018a13
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/aaudio_recorder.h
>@@ -0,0 +1,128 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_DEVICE_ANDROID_AAUDIO_RECORDER_H_
>+#define MODULES_AUDIO_DEVICE_ANDROID_AAUDIO_RECORDER_H_
>+
>+#include <aaudio/AAudio.h>
>+#include <memory>
>+
>+#include "modules/audio_device/android/aaudio_wrapper.h"
>+#include "modules/audio_device/include/audio_device_defines.h"
>+#include "rtc_base/messagehandler.h"
>+#include "rtc_base/thread.h"
>+#include "rtc_base/thread_checker.h"
>+
>+namespace webrtc {
>+
>+class AudioDeviceBuffer;
>+class FineAudioBuffer;
>+class AudioManager;
>+
>+// Implements low-latency 16-bit mono PCM audio input support for Android
>+// using the C based AAudio API.
>+//
>+// An instance must be created and destroyed on one and the same thread.
>+// All public methods must also be called on the same thread. A thread checker
>+// will RTC_DCHECK if any method is called on an invalid thread. Audio buffers
>+// are delivered on a dedicated high-priority thread owned by AAudio.
>+//
>+// The existing design forces the user to call InitRecording() after
>+// StopRecording() to be able to call StartRecording() again. This is in line
>+// with how the Java- based implementation works.
>+//
>+// TODO(henrika): add comments about device changes and adaptive buffer
>+// management.
>+class AAudioRecorder : public AAudioObserverInterface,
>+                       public rtc::MessageHandler {
>+ public:
>+  explicit AAudioRecorder(AudioManager* audio_manager);
>+  ~AAudioRecorder();
>+
>+  int Init();
>+  int Terminate();
>+
>+  int InitRecording();
>+  bool RecordingIsInitialized() const { return initialized_; }
>+
>+  int StartRecording();
>+  int StopRecording();
>+  bool Recording() const { return recording_; }
>+
>+  void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
>+
>+  double latency_millis() const { return latency_millis_; }
>+
>+  // TODO(henrika): add support using AAudio APIs when available.
>+  int EnableBuiltInAEC(bool enable);
>+  int EnableBuiltInAGC(bool enable);
>+  int EnableBuiltInNS(bool enable);
>+
>+ protected:
>+  // AAudioObserverInterface implementation.
>+
>+  // For an input stream, this function should read |num_frames| of recorded
>+  // data, in the stream's current data format, from the |audio_data| buffer.
>+  // Called on a real-time thread owned by AAudio.
>+  aaudio_data_callback_result_t OnDataCallback(void* audio_data,
>+                                               int32_t num_frames) override;
>+
>+  // AAudio calls this function if any error occurs on a callback thread.
>+  // Called on a real-time thread owned by AAudio.
>+  void OnErrorCallback(aaudio_result_t error) override;
>+
>+  // rtc::MessageHandler used for restart messages.
>+  void OnMessage(rtc::Message* msg) override;
>+
>+ private:
>+  // Closes the existing stream and starts a new stream.
>+  void HandleStreamDisconnected();
>+
>+  // Ensures that methods are called from the same thread as this object is
>+  // created on.
>+  rtc::ThreadChecker thread_checker_;
>+
>+  // Stores thread ID in first call to AAudioPlayer::OnDataCallback from a
>+  // real-time thread owned by AAudio. Detached during construction of this
>+  // object.
>+  rtc::ThreadChecker thread_checker_aaudio_;
>+
>+  // The thread on which this object is created on.
>+  rtc::Thread* main_thread_;
>+
>+  // Wraps all AAudio resources. Contains an input stream using the default
>+  // input audio device.
>+  AAudioWrapper aaudio_;
>+
>+  // Raw pointer handle provided to us in AttachAudioBuffer(). Owned by the
>+  // AudioDeviceModuleImpl class and called by AudioDeviceModule::Create().
>+  AudioDeviceBuffer* audio_device_buffer_ = nullptr;
>+
>+  bool initialized_ = false;
>+  bool recording_ = false;
>+
>+  // Consumes audio of native buffer size and feeds the WebRTC layer with 10ms
>+  // chunks of audio.
>+  std::unique_ptr<FineAudioBuffer> fine_audio_buffer_;
>+
>+  // Counts number of detected overflow events reported by AAudio.
>+  int32_t overflow_count_ = 0;
>+
>+  // Estimated time between an audio frame was recorded by the input device and
>+  // it can read on the input stream.
>+  double latency_millis_ = 0;
>+
>+  // True only for the first data callback in each audio session.
>+  bool first_data_callback_ = true;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_DEVICE_ANDROID_AAUDIO_RECORDER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/aaudio_wrapper.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/aaudio_wrapper.cc
>new file mode 100644
>index 00000000000..d95a9b4c6eb
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/aaudio_wrapper.cc
>@@ -0,0 +1,499 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_device/android/aaudio_wrapper.h"
>+
>+#include "modules/audio_device/android/audio_manager.h"
>+#include "rtc_base/logging.h"
>+#include "rtc_base/strings/string_builder.h"
>+#include "rtc_base/timeutils.h"
>+
>+#define LOG_ON_ERROR(op)                                                      \
>+  do {                                                                        \
>+    aaudio_result_t result = (op);                                            \
>+    if (result != AAUDIO_OK) {                                                \
>+      RTC_LOG(LS_ERROR) << #op << ": " << AAudio_convertResultToText(result); \
>+    }                                                                         \
>+  } while (0)
>+
>+#define RETURN_ON_ERROR(op, ...)                                              \
>+  do {                                                                        \
>+    aaudio_result_t result = (op);                                            \
>+    if (result != AAUDIO_OK) {                                                \
>+      RTC_LOG(LS_ERROR) << #op << ": " << AAudio_convertResultToText(result); \
>+      return __VA_ARGS__;                                                     \
>+    }                                                                         \
>+  } while (0)
>+
>+namespace webrtc {
>+
>+namespace {
>+
>+const char* DirectionToString(aaudio_direction_t direction) {
>+  switch (direction) {
>+    case AAUDIO_DIRECTION_OUTPUT:
>+      return "OUTPUT";
>+    case AAUDIO_DIRECTION_INPUT:
>+      return "INPUT";
>+    default:
>+      return "UNKNOWN";
>+  }
>+}
>+
>+const char* SharingModeToString(aaudio_sharing_mode_t mode) {
>+  switch (mode) {
>+    case AAUDIO_SHARING_MODE_EXCLUSIVE:
>+      return "EXCLUSIVE";
>+    case AAUDIO_SHARING_MODE_SHARED:
>+      return "SHARED";
>+    default:
>+      return "UNKNOWN";
>+  }
>+}
>+
>+const char* PerformanceModeToString(aaudio_performance_mode_t mode) {
>+  switch (mode) {
>+    case AAUDIO_PERFORMANCE_MODE_NONE:
>+      return "NONE";
>+    case AAUDIO_PERFORMANCE_MODE_POWER_SAVING:
>+      return "POWER_SAVING";
>+    case AAUDIO_PERFORMANCE_MODE_LOW_LATENCY:
>+      return "LOW_LATENCY";
>+    default:
>+      return "UNKNOWN";
>+  }
>+}
>+
>+const char* FormatToString(int32_t id) {
>+  switch (id) {
>+    case AAUDIO_FORMAT_INVALID:
>+      return "INVALID";
>+    case AAUDIO_FORMAT_UNSPECIFIED:
>+      return "UNSPECIFIED";
>+    case AAUDIO_FORMAT_PCM_I16:
>+      return "PCM_I16";
>+    case AAUDIO_FORMAT_PCM_FLOAT:
>+      return "FLOAT";
>+    default:
>+      return "UNKNOWN";
>+  }
>+}
>+
>+void ErrorCallback(AAudioStream* stream,
>+                   void* user_data,
>+                   aaudio_result_t error) {
>+  RTC_DCHECK(user_data);
>+  AAudioWrapper* aaudio_wrapper = reinterpret_cast<AAudioWrapper*>(user_data);
>+  RTC_LOG(WARNING) << "ErrorCallback: "
>+                   << DirectionToString(aaudio_wrapper->direction());
>+  RTC_DCHECK(aaudio_wrapper->observer());
>+  aaudio_wrapper->observer()->OnErrorCallback(error);
>+}
>+
>+aaudio_data_callback_result_t DataCallback(AAudioStream* stream,
>+                                           void* user_data,
>+                                           void* audio_data,
>+                                           int32_t num_frames) {
>+  RTC_DCHECK(user_data);
>+  RTC_DCHECK(audio_data);
>+  AAudioWrapper* aaudio_wrapper = reinterpret_cast<AAudioWrapper*>(user_data);
>+  RTC_DCHECK(aaudio_wrapper->observer());
>+  return aaudio_wrapper->observer()->OnDataCallback(audio_data, num_frames);
>+}
>+
>+// Wraps the stream builder object to ensure that it is released properly when
>+// the stream builder goes out of scope.
>+class ScopedStreamBuilder {
>+ public:
>+  ScopedStreamBuilder() {
>+    LOG_ON_ERROR(AAudio_createStreamBuilder(&builder_));
>+    RTC_DCHECK(builder_);
>+  }
>+  ~ScopedStreamBuilder() {
>+    if (builder_) {
>+      LOG_ON_ERROR(AAudioStreamBuilder_delete(builder_));
>+    }
>+  }
>+
>+  AAudioStreamBuilder* get() const { return builder_; }
>+
>+ private:
>+  AAudioStreamBuilder* builder_ = nullptr;
>+};
>+
>+}  // namespace
>+
>+AAudioWrapper::AAudioWrapper(AudioManager* audio_manager,
>+                             aaudio_direction_t direction,
>+                             AAudioObserverInterface* observer)
>+    : direction_(direction), observer_(observer) {
>+  RTC_LOG(INFO) << "ctor";
>+  RTC_DCHECK(observer_);
>+  direction_ == AAUDIO_DIRECTION_OUTPUT
>+      ? audio_parameters_ = audio_manager->GetPlayoutAudioParameters()
>+      : audio_parameters_ = audio_manager->GetRecordAudioParameters();
>+  aaudio_thread_checker_.DetachFromThread();
>+  RTC_LOG(INFO) << audio_parameters_.ToString();
>+}
>+
>+AAudioWrapper::~AAudioWrapper() {
>+  RTC_LOG(INFO) << "dtor";
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  RTC_DCHECK(!stream_);
>+}
>+
>+bool AAudioWrapper::Init() {
>+  RTC_LOG(INFO) << "Init";
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  // Creates a stream builder which can be used to open an audio stream.
>+  ScopedStreamBuilder builder;
>+  // Configures the stream builder using audio parameters given at construction.
>+  SetStreamConfiguration(builder.get());
>+  // Opens a stream based on options in the stream builder.
>+  if (!OpenStream(builder.get())) {
>+    return false;
>+  }
>+  // Ensures that the opened stream could activate the requested settings.
>+  if (!VerifyStreamConfiguration()) {
>+    return false;
>+  }
>+  // Optimizes the buffer scheme for lowest possible latency and creates
>+  // additional buffer logic to match the 10ms buffer size used in WebRTC.
>+  if (!OptimizeBuffers()) {
>+    return false;
>+  }
>+  LogStreamState();
>+  return true;
>+}
>+
>+bool AAudioWrapper::Start() {
>+  RTC_LOG(INFO) << "Start";
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  // TODO(henrika): this state check might not be needed.
>+  aaudio_stream_state_t current_state = AAudioStream_getState(stream_);
>+  if (current_state != AAUDIO_STREAM_STATE_OPEN) {
>+    RTC_LOG(LS_ERROR) << "Invalid state: "
>+                      << AAudio_convertStreamStateToText(current_state);
>+    return false;
>+  }
>+  // Asynchronous request for the stream to start.
>+  RETURN_ON_ERROR(AAudioStream_requestStart(stream_), false);
>+  LogStreamState();
>+  return true;
>+}
>+
>+bool AAudioWrapper::Stop() {
>+  RTC_LOG(INFO) << "Stop: " << DirectionToString(direction());
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  // Asynchronous request for the stream to stop.
>+  RETURN_ON_ERROR(AAudioStream_requestStop(stream_), false);
>+  CloseStream();
>+  aaudio_thread_checker_.DetachFromThread();
>+  return true;
>+}
>+
>+double AAudioWrapper::EstimateLatencyMillis() const {
>+  RTC_DCHECK(stream_);
>+  double latency_millis = 0.0;
>+  if (direction() == AAUDIO_DIRECTION_INPUT) {
>+    // For input streams. Best guess we can do is to use the current burst size
>+    // as delay estimate.
>+    latency_millis = static_cast<double>(frames_per_burst()) / sample_rate() *
>+                     rtc::kNumMillisecsPerSec;
>+  } else {
>+    int64_t existing_frame_index;
>+    int64_t existing_frame_presentation_time;
>+    // Get the time at which a particular frame was presented to audio hardware.
>+    aaudio_result_t result = AAudioStream_getTimestamp(
>+        stream_, CLOCK_MONOTONIC, &existing_frame_index,
>+        &existing_frame_presentation_time);
>+    // Results are only valid when the stream is in AAUDIO_STREAM_STATE_STARTED.
>+    if (result == AAUDIO_OK) {
>+      // Get write index for next audio frame.
>+      int64_t next_frame_index = frames_written();
>+      // Number of frames between next frame and the existing frame.
>+      int64_t frame_index_delta = next_frame_index - existing_frame_index;
>+      // Assume the next frame will be written now.
>+      int64_t next_frame_write_time = rtc::TimeNanos();
>+      // Calculate time when next frame will be presented to the hardware taking
>+      // sample rate into account.
>+      int64_t frame_time_delta =
>+          (frame_index_delta * rtc::kNumNanosecsPerSec) / sample_rate();
>+      int64_t next_frame_presentation_time =
>+          existing_frame_presentation_time + frame_time_delta;
>+      // Derive a latency estimate given results above.
>+      latency_millis = static_cast<double>(next_frame_presentation_time -
>+                                           next_frame_write_time) /
>+                       rtc::kNumNanosecsPerMillisec;
>+    }
>+  }
>+  return latency_millis;
>+}
>+
>+// Returns new buffer size or a negative error value if buffer size could not
>+// be increased.
>+bool AAudioWrapper::IncreaseOutputBufferSize() {
>+  RTC_LOG(INFO) << "IncreaseBufferSize";
>+  RTC_DCHECK(stream_);
>+  RTC_DCHECK(aaudio_thread_checker_.CalledOnValidThread());
>+  RTC_DCHECK_EQ(direction(), AAUDIO_DIRECTION_OUTPUT);
>+  aaudio_result_t buffer_size = AAudioStream_getBufferSizeInFrames(stream_);
>+  // Try to increase size of buffer with one burst to reduce risk of underrun.
>+  buffer_size += frames_per_burst();
>+  // Verify that the new buffer size is not larger than max capacity.
>+  // TODO(henrika): keep track of case when we reach the capacity limit.
>+  const int32_t max_buffer_size = buffer_capacity_in_frames();
>+  if (buffer_size > max_buffer_size) {
>+    RTC_LOG(LS_ERROR) << "Required buffer size (" << buffer_size
>+                      << ") is higher than max: " << max_buffer_size;
>+    return false;
>+  }
>+  RTC_LOG(INFO) << "Updating buffer size to: " << buffer_size
>+                << " (max=" << max_buffer_size << ")";
>+  buffer_size = AAudioStream_setBufferSizeInFrames(stream_, buffer_size);
>+  if (buffer_size < 0) {
>+    RTC_LOG(LS_ERROR) << "Failed to change buffer size: "
>+                      << AAudio_convertResultToText(buffer_size);
>+    return false;
>+  }
>+  RTC_LOG(INFO) << "Buffer size changed to: " << buffer_size;
>+  return true;
>+}
>+
>+void AAudioWrapper::ClearInputStream(void* audio_data, int32_t num_frames) {
>+  RTC_LOG(INFO) << "ClearInputStream";
>+  RTC_DCHECK(stream_);
>+  RTC_DCHECK(aaudio_thread_checker_.CalledOnValidThread());
>+  RTC_DCHECK_EQ(direction(), AAUDIO_DIRECTION_INPUT);
>+  aaudio_result_t cleared_frames = 0;
>+  do {
>+    cleared_frames = AAudioStream_read(stream_, audio_data, num_frames, 0);
>+  } while (cleared_frames > 0);
>+}
>+
>+AAudioObserverInterface* AAudioWrapper::observer() const {
>+  return observer_;
>+}
>+
>+AudioParameters AAudioWrapper::audio_parameters() const {
>+  return audio_parameters_;
>+}
>+
>+int32_t AAudioWrapper::samples_per_frame() const {
>+  RTC_DCHECK(stream_);
>+  return AAudioStream_getSamplesPerFrame(stream_);
>+}
>+
>+int32_t AAudioWrapper::buffer_size_in_frames() const {
>+  RTC_DCHECK(stream_);
>+  return AAudioStream_getBufferSizeInFrames(stream_);
>+}
>+
>+int32_t AAudioWrapper::buffer_capacity_in_frames() const {
>+  RTC_DCHECK(stream_);
>+  return AAudioStream_getBufferCapacityInFrames(stream_);
>+}
>+
>+int32_t AAudioWrapper::device_id() const {
>+  RTC_DCHECK(stream_);
>+  return AAudioStream_getDeviceId(stream_);
>+}
>+
>+int32_t AAudioWrapper::xrun_count() const {
>+  RTC_DCHECK(stream_);
>+  return AAudioStream_getXRunCount(stream_);
>+}
>+
>+int32_t AAudioWrapper::format() const {
>+  RTC_DCHECK(stream_);
>+  return AAudioStream_getFormat(stream_);
>+}
>+
>+int32_t AAudioWrapper::sample_rate() const {
>+  RTC_DCHECK(stream_);
>+  return AAudioStream_getSampleRate(stream_);
>+}
>+
>+int32_t AAudioWrapper::channel_count() const {
>+  RTC_DCHECK(stream_);
>+  return AAudioStream_getChannelCount(stream_);
>+}
>+
>+int32_t AAudioWrapper::frames_per_callback() const {
>+  RTC_DCHECK(stream_);
>+  return AAudioStream_getFramesPerDataCallback(stream_);
>+}
>+
>+aaudio_sharing_mode_t AAudioWrapper::sharing_mode() const {
>+  RTC_DCHECK(stream_);
>+  return AAudioStream_getSharingMode(stream_);
>+}
>+
>+aaudio_performance_mode_t AAudioWrapper::performance_mode() const {
>+  RTC_DCHECK(stream_);
>+  return AAudioStream_getPerformanceMode(stream_);
>+}
>+
>+aaudio_stream_state_t AAudioWrapper::stream_state() const {
>+  RTC_DCHECK(stream_);
>+  return AAudioStream_getState(stream_);
>+}
>+
>+int64_t AAudioWrapper::frames_written() const {
>+  RTC_DCHECK(stream_);
>+  return AAudioStream_getFramesWritten(stream_);
>+}
>+
>+int64_t AAudioWrapper::frames_read() const {
>+  RTC_DCHECK(stream_);
>+  return AAudioStream_getFramesRead(stream_);
>+}
>+
>+void AAudioWrapper::SetStreamConfiguration(AAudioStreamBuilder* builder) {
>+  RTC_LOG(INFO) << "SetStreamConfiguration";
>+  RTC_DCHECK(builder);
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  // Request usage of default primary output/input device.
>+  // TODO(henrika): verify that default device follows Java APIs.
>+  // https://developer.android.com/reference/android/media/AudioDeviceInfo.html.
>+  AAudioStreamBuilder_setDeviceId(builder, AAUDIO_UNSPECIFIED);
>+  // Use preferred sample rate given by the audio parameters.
>+  AAudioStreamBuilder_setSampleRate(builder, audio_parameters().sample_rate());
>+  // Use preferred channel configuration given by the audio parameters.
>+  AAudioStreamBuilder_setChannelCount(builder, audio_parameters().channels());
>+  // Always use 16-bit PCM audio sample format.
>+  AAudioStreamBuilder_setFormat(builder, AAUDIO_FORMAT_PCM_I16);
>+  // TODO(henrika): investigate effect of using AAUDIO_SHARING_MODE_EXCLUSIVE.
>+  // Ask for exclusive mode since this will give us the lowest possible latency.
>+  // If exclusive mode isn't available, shared mode will be used instead.
>+  AAudioStreamBuilder_setSharingMode(builder, AAUDIO_SHARING_MODE_SHARED);
>+  // Use the direction that was given at construction.
>+  AAudioStreamBuilder_setDirection(builder, direction_);
>+  // TODO(henrika): investigate performance using different performance modes.
>+  AAudioStreamBuilder_setPerformanceMode(builder,
>+                                         AAUDIO_PERFORMANCE_MODE_LOW_LATENCY);
>+  // Given that WebRTC applications require low latency, our audio stream uses
>+  // an asynchronous callback function to transfer data to and from the
>+  // application. AAudio executes the callback in a higher-priority thread that
>+  // has better performance.
>+  AAudioStreamBuilder_setDataCallback(builder, DataCallback, this);
>+  // Request that AAudio calls this functions if any error occurs on a callback
>+  // thread.
>+  AAudioStreamBuilder_setErrorCallback(builder, ErrorCallback, this);
>+}
>+
>+bool AAudioWrapper::OpenStream(AAudioStreamBuilder* builder) {
>+  RTC_LOG(INFO) << "OpenStream";
>+  RTC_DCHECK(builder);
>+  AAudioStream* stream = nullptr;
>+  RETURN_ON_ERROR(AAudioStreamBuilder_openStream(builder, &stream), false);
>+  stream_ = stream;
>+  LogStreamConfiguration();
>+  return true;
>+}
>+
>+void AAudioWrapper::CloseStream() {
>+  RTC_LOG(INFO) << "CloseStream";
>+  RTC_DCHECK(stream_);
>+  LOG_ON_ERROR(AAudioStream_close(stream_));
>+  stream_ = nullptr;
>+}
>+
>+void AAudioWrapper::LogStreamConfiguration() {
>+  RTC_DCHECK(stream_);
>+  char ss_buf[1024];
>+  rtc::SimpleStringBuilder ss(ss_buf);
>+  ss << "Stream Configuration: ";
>+  ss << "sample rate=" << sample_rate() << ", channels=" << channel_count();
>+  ss << ", samples per frame=" << samples_per_frame();
>+  ss << ", format=" << FormatToString(format());
>+  ss << ", sharing mode=" << SharingModeToString(sharing_mode());
>+  ss << ", performance mode=" << PerformanceModeToString(performance_mode());
>+  ss << ", direction=" << DirectionToString(direction());
>+  ss << ", device id=" << AAudioStream_getDeviceId(stream_);
>+  ss << ", frames per callback=" << frames_per_callback();
>+  RTC_LOG(INFO) << ss.str();
>+}
>+
>+void AAudioWrapper::LogStreamState() {
>+  RTC_LOG(INFO) << "AAudio stream state: "
>+                << AAudio_convertStreamStateToText(stream_state());
>+}
>+
>+bool AAudioWrapper::VerifyStreamConfiguration() {
>+  RTC_LOG(INFO) << "VerifyStreamConfiguration";
>+  RTC_DCHECK(stream_);
>+  // TODO(henrika): should we verify device ID as well?
>+  if (AAudioStream_getSampleRate(stream_) != audio_parameters().sample_rate()) {
>+    RTC_LOG(LS_ERROR) << "Stream unable to use requested sample rate";
>+    return false;
>+  }
>+  if (AAudioStream_getChannelCount(stream_) !=
>+      static_cast<int32_t>(audio_parameters().channels())) {
>+    RTC_LOG(LS_ERROR) << "Stream unable to use requested channel count";
>+    return false;
>+  }
>+  if (AAudioStream_getFormat(stream_) != AAUDIO_FORMAT_PCM_I16) {
>+    RTC_LOG(LS_ERROR) << "Stream unable to use requested format";
>+    return false;
>+  }
>+  if (AAudioStream_getSharingMode(stream_) != AAUDIO_SHARING_MODE_SHARED) {
>+    RTC_LOG(LS_ERROR) << "Stream unable to use requested sharing mode";
>+    return false;
>+  }
>+  if (AAudioStream_getPerformanceMode(stream_) !=
>+      AAUDIO_PERFORMANCE_MODE_LOW_LATENCY) {
>+    RTC_LOG(LS_ERROR) << "Stream unable to use requested performance mode";
>+    return false;
>+  }
>+  if (AAudioStream_getDirection(stream_) != direction()) {
>+    RTC_LOG(LS_ERROR) << "Stream direction could not be set";
>+    return false;
>+  }
>+  if (AAudioStream_getSamplesPerFrame(stream_) !=
>+      static_cast<int32_t>(audio_parameters().channels())) {
>+    RTC_LOG(LS_ERROR) << "Invalid number of samples per frame";
>+    return false;
>+  }
>+  return true;
>+}
>+
>+bool AAudioWrapper::OptimizeBuffers() {
>+  RTC_LOG(INFO) << "OptimizeBuffers";
>+  RTC_DCHECK(stream_);
>+  // Maximum number of frames that can be filled without blocking.
>+  RTC_LOG(INFO) << "max buffer capacity in frames: "
>+                << buffer_capacity_in_frames();
>+  // Query the number of frames that the application should read or write at
>+  // one time for optimal performance.
>+  int32_t frames_per_burst = AAudioStream_getFramesPerBurst(stream_);
>+  RTC_LOG(INFO) << "frames per burst for optimal performance: "
>+                << frames_per_burst;
>+  frames_per_burst_ = frames_per_burst;
>+  if (direction() == AAUDIO_DIRECTION_INPUT) {
>+    // There is no point in calling setBufferSizeInFrames() for input streams
>+    // since it has no effect on the performance (latency in this case).
>+    return true;
>+  }
>+  // Set buffer size to same as burst size to guarantee lowest possible latency.
>+  // This size might change for output streams if underruns are detected and
>+  // automatic buffer adjustment is enabled.
>+  AAudioStream_setBufferSizeInFrames(stream_, frames_per_burst);
>+  int32_t buffer_size = AAudioStream_getBufferSizeInFrames(stream_);
>+  if (buffer_size != frames_per_burst) {
>+    RTC_LOG(LS_ERROR) << "Failed to use optimal buffer burst size";
>+    return false;
>+  }
>+  // Maximum number of frames that can be filled without blocking.
>+  RTC_LOG(INFO) << "buffer burst size in frames: " << buffer_size;
>+  return true;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/aaudio_wrapper.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/aaudio_wrapper.h
>new file mode 100644
>index 00000000000..4915092149b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/aaudio_wrapper.h
>@@ -0,0 +1,127 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_DEVICE_ANDROID_AAUDIO_WRAPPER_H_
>+#define MODULES_AUDIO_DEVICE_ANDROID_AAUDIO_WRAPPER_H_
>+
>+#include <aaudio/AAudio.h>
>+
>+#include "modules/audio_device/include/audio_device_defines.h"
>+#include "rtc_base/thread_checker.h"
>+
>+namespace webrtc {
>+
>+class AudioManager;
>+
>+// AAudio callback interface for audio transport to/from the AAudio stream.
>+// The interface also contains an error callback method for notifications of
>+// e.g. device changes.
>+class AAudioObserverInterface {
>+ public:
>+  // Audio data will be passed in our out of this function dependning on the
>+  // direction of the audio stream. This callback function will be called on a
>+  // real-time thread owned by AAudio.
>+  virtual aaudio_data_callback_result_t OnDataCallback(void* audio_data,
>+                                                       int32_t num_frames) = 0;
>+  // AAudio will call this functions if any error occurs on a callback thread.
>+  // In response, this function could signal or launch another thread to reopen
>+  // a stream on another device. Do not reopen the stream in this callback.
>+  virtual void OnErrorCallback(aaudio_result_t error) = 0;
>+
>+ protected:
>+  virtual ~AAudioObserverInterface() {}
>+};
>+
>+// Utility class which wraps the C-based AAudio API into a more handy C++ class
>+// where the underlying resources (AAudioStreamBuilder and AAudioStream) are
>+// encapsulated. User must set the direction (in or out) at construction since
>+// it defines the stream type and the direction of the data flow in the
>+// AAudioObserverInterface.
>+//
>+// AAudio is a new Android C API introduced in the Android O (26) release.
>+// It is designed for high-performance audio applications that require low
>+// latency. Applications communicate with AAudio by reading and writing data
>+// to streams.
>+//
>+// Each stream is attached to a single audio device, where each audio device
>+// has a unique ID. The ID can be used to bind an audio stream to a specific
>+// audio device but this implementation lets AAudio choose the default primary
>+// device instead (device selection takes place in Java). A stream can only
>+// move data in one direction. When a stream is opened, Android checks to
>+// ensure that the audio device and stream direction agree.
>+class AAudioWrapper {
>+ public:
>+  AAudioWrapper(AudioManager* audio_manager,
>+                aaudio_direction_t direction,
>+                AAudioObserverInterface* observer);
>+  ~AAudioWrapper();
>+
>+  bool Init();
>+  bool Start();
>+  bool Stop();
>+
>+  // For output streams: estimates latency between writing an audio frame to
>+  // the output stream and the time that same frame is played out on the output
>+  // audio device.
>+  // For input streams: estimates latency between reading an audio frame from
>+  // the input stream and the time that same frame was recorded on the input
>+  // audio device.
>+  double EstimateLatencyMillis() const;
>+
>+  // Increases the internal buffer size for output streams by one burst size to
>+  // reduce the risk of underruns. Can be used while a stream is active.
>+  bool IncreaseOutputBufferSize();
>+
>+  // Drains the recording stream of any existing data by reading from it until
>+  // it's empty. Can be used to clear out old data before starting a new audio
>+  // session.
>+  void ClearInputStream(void* audio_data, int32_t num_frames);
>+
>+  AAudioObserverInterface* observer() const;
>+  AudioParameters audio_parameters() const;
>+  int32_t samples_per_frame() const;
>+  int32_t buffer_size_in_frames() const;
>+  int32_t buffer_capacity_in_frames() const;
>+  int32_t device_id() const;
>+  int32_t xrun_count() const;
>+  int32_t format() const;
>+  int32_t sample_rate() const;
>+  int32_t channel_count() const;
>+  int32_t frames_per_callback() const;
>+  aaudio_sharing_mode_t sharing_mode() const;
>+  aaudio_performance_mode_t performance_mode() const;
>+  aaudio_stream_state_t stream_state() const;
>+  int64_t frames_written() const;
>+  int64_t frames_read() const;
>+  aaudio_direction_t direction() const { return direction_; }
>+  AAudioStream* stream() const { return stream_; }
>+  int32_t frames_per_burst() const { return frames_per_burst_; }
>+
>+ private:
>+  void SetStreamConfiguration(AAudioStreamBuilder* builder);
>+  bool OpenStream(AAudioStreamBuilder* builder);
>+  void CloseStream();
>+  void LogStreamConfiguration();
>+  void LogStreamState();
>+  bool VerifyStreamConfiguration();
>+  bool OptimizeBuffers();
>+
>+  rtc::ThreadChecker thread_checker_;
>+  rtc::ThreadChecker aaudio_thread_checker_;
>+  AudioParameters audio_parameters_;
>+  const aaudio_direction_t direction_;
>+  AAudioObserverInterface* observer_ = nullptr;
>+  AAudioStream* stream_ = nullptr;
>+  int32_t frames_per_burst_ = 0;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_DEVICE_ANDROID_AAUDIO_WRAPPER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/audio_device_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/audio_device_unittest.cc
>index 4a9262b0443..06ae706632b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/audio_device_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/audio_device_unittest.cc
>@@ -92,6 +92,7 @@ class AudioStreamInterface {
>  public:
>   virtual void Write(const void* source, size_t num_frames) = 0;
>   virtual void Read(void* destination, size_t num_frames) = 0;
>+
>  protected:
>   virtual ~AudioStreamInterface() {}
> };
>@@ -100,11 +101,10 @@ class AudioStreamInterface {
> // construction.
> class FileAudioStream : public AudioStreamInterface {
>  public:
>-  FileAudioStream(
>-      size_t num_callbacks, const std::string& file_name, int sample_rate)
>-      : file_size_in_bytes_(0),
>-        sample_rate_(sample_rate),
>-        file_pos_(0) {
>+  FileAudioStream(size_t num_callbacks,
>+                  const std::string& file_name,
>+                  int sample_rate)
>+      : file_size_in_bytes_(0), sample_rate_(sample_rate), file_pos_(0) {
>     file_size_in_bytes_ = test::GetFileSize(file_name);
>     sample_rate_ = sample_rate;
>     EXPECT_GE(file_size_in_callbacks(), num_callbacks)
>@@ -114,8 +114,8 @@ class FileAudioStream : public AudioStreamInterface {
>     file_.reset(new int16_t[num_16bit_samples]);
>     FILE* audio_file = fopen(file_name.c_str(), "rb");
>     EXPECT_NE(audio_file, nullptr);
>-    size_t num_samples_read = fread(
>-        file_.get(), sizeof(int16_t), num_16bit_samples, audio_file);
>+    size_t num_samples_read =
>+        fread(file_.get(), sizeof(int16_t), num_16bit_samples, audio_file);
>     EXPECT_EQ(num_samples_read, num_16bit_samples);
>     fclose(audio_file);
>   }
>@@ -126,15 +126,14 @@ class FileAudioStream : public AudioStreamInterface {
>   // Read samples from file stored in memory (at construction) and copy
>   // |num_frames| (<=> 10ms) to the |destination| byte buffer.
>   void Read(void* destination, size_t num_frames) override {
>-    memcpy(destination,
>-           static_cast<int16_t*> (&file_[file_pos_]),
>+    memcpy(destination, static_cast<int16_t*>(&file_[file_pos_]),
>            num_frames * sizeof(int16_t));
>     file_pos_ += num_frames;
>   }
> 
>   int file_size_in_seconds() const {
>-    return static_cast<int>(
>-        file_size_in_bytes_ / (kBytesPerSample * sample_rate_));
>+    return static_cast<int>(file_size_in_bytes_ /
>+                            (kBytesPerSample * sample_rate_));
>   }
>   size_t file_size_in_callbacks() const {
>     return file_size_in_seconds() * kNumCallbacksPerSecond;
>@@ -167,9 +166,7 @@ class FifoAudioStream : public AudioStreamInterface {
>     EXPECT_NE(fifo_.get(), nullptr);
>   }
> 
>-  ~FifoAudioStream() {
>-    Flush();
>-  }
>+  ~FifoAudioStream() { Flush(); }
> 
>   // Allocate new memory, copy |num_frames| samples from |source| into memory
>   // and add pointer to the memory location to end of the list.
>@@ -181,9 +178,7 @@ class FifoAudioStream : public AudioStreamInterface {
>       return;
>     }
>     int16_t* memory = new int16_t[frames_per_buffer_];
>-    memcpy(static_cast<int16_t*> (&memory[0]),
>-           source,
>-           bytes_per_buffer_);
>+    memcpy(static_cast<int16_t*>(&memory[0]), source, bytes_per_buffer_);
>     rtc::CritScope lock(&lock_);
>     fifo_->push_back(memory);
>     const size_t size = fifo_->size();
>@@ -206,24 +201,20 @@ class FifoAudioStream : public AudioStreamInterface {
>     } else {
>       int16_t* memory = fifo_->front();
>       fifo_->pop_front();
>-      memcpy(destination,
>-             static_cast<int16_t*> (&memory[0]),
>-             bytes_per_buffer_);
>+      memcpy(destination, static_cast<int16_t*>(&memory[0]), bytes_per_buffer_);
>       delete memory;
>     }
>   }
> 
>-  size_t size() const {
>-    return fifo_->size();
>-  }
>+  size_t size() const { return fifo_->size(); }
> 
>-  size_t largest_size() const {
>-    return largest_size_;
>-  }
>+  size_t largest_size() const { return largest_size_; }
> 
>   size_t average_size() const {
>-    return (total_written_elements_ == 0) ? 0.0 : 0.5 + static_cast<float> (
>-      total_written_elements_) / (write_count_ - kNumIgnoreFirstCallbacks);
>+    return (total_written_elements_ == 0)
>+               ? 0.0
>+               : 0.5 + static_cast<float>(total_written_elements_) /
>+                           (write_count_ - kNumIgnoreFirstCallbacks);
>   }
> 
>  private:
>@@ -255,8 +246,7 @@ class LatencyMeasuringAudioStream : public AudioStreamInterface {
>         bytes_per_buffer_(frames_per_buffer_ * sizeof(int16_t)),
>         play_count_(0),
>         rec_count_(0),
>-        pulse_time_(0) {
>-  }
>+        pulse_time_(0) {}
> 
>   // Insert periodic impulses in first two samples of |destination|.
>   void Read(void* destination, size_t num_frames) override {
>@@ -272,7 +262,7 @@ class LatencyMeasuringAudioStream : public AudioStreamInterface {
>       }
>       PRINT(".");
>       const int16_t impulse = std::numeric_limits<int16_t>::max();
>-      int16_t* ptr16 = static_cast<int16_t*> (destination);
>+      int16_t* ptr16 = static_cast<int16_t*>(destination);
>       for (size_t i = 0; i < 2; ++i) {
>         ptr16[i] = impulse;
>       }
>@@ -289,19 +279,18 @@ class LatencyMeasuringAudioStream : public AudioStreamInterface {
>       // been transmitted (sets |pulse_time_| to value larger than zero).
>       return;
>     }
>-    const int16_t* ptr16 = static_cast<const int16_t*> (source);
>+    const int16_t* ptr16 = static_cast<const int16_t*>(source);
>     std::vector<int16_t> vec(ptr16, ptr16 + num_frames);
>     // Find max value in the audio buffer.
>     int max = *std::max_element(vec.begin(), vec.end());
>     // Find index (element position in vector) of the max element.
>-    int index_of_max = std::distance(vec.begin(),
>-                                     std::find(vec.begin(), vec.end(),
>-                                     max));
>+    int index_of_max =
>+        std::distance(vec.begin(), std::find(vec.begin(), vec.end(), max));
>     if (max > kImpulseThreshold) {
>       PRINTD("(%d,%d)", max, index_of_max);
>       int64_t now_time = rtc::TimeMillis();
>-      int extra_delay = IndexToMilliseconds(static_cast<double> (index_of_max));
>-      PRINTD("[%d]", static_cast<int> (now_time - pulse_time_));
>+      int extra_delay = IndexToMilliseconds(static_cast<double>(index_of_max));
>+      PRINTD("[%d]", static_cast<int>(now_time - pulse_time_));
>       PRINTD("[%d]", extra_delay);
>       // Total latency is the difference between transmit time and detection
>       // tome plus the extra delay within the buffer in which we detected the
>@@ -315,9 +304,7 @@ class LatencyMeasuringAudioStream : public AudioStreamInterface {
>     }
>   }
> 
>-  size_t num_latency_values() const {
>-    return latencies_.size();
>-  }
>+  size_t num_latency_values() const { return latencies_.size(); }
> 
>   int min_latency() const {
>     if (latencies_.empty())
>@@ -334,9 +321,9 @@ class LatencyMeasuringAudioStream : public AudioStreamInterface {
>   int average_latency() const {
>     if (latencies_.empty())
>       return 0;
>-    return 0.5 + static_cast<double> (
>-        std::accumulate(latencies_.begin(), latencies_.end(), 0)) /
>-        latencies_.size();
>+    return 0.5 + static_cast<double>(
>+                     std::accumulate(latencies_.begin(), latencies_.end(), 0)) /
>+                     latencies_.size();
>   }
> 
>   void PrintResults() const {
>@@ -345,8 +332,8 @@ class LatencyMeasuringAudioStream : public AudioStreamInterface {
>       PRINT("%d ", *it);
>     }
>     PRINT("\n");
>-    PRINT("%s[min, max, avg]=[%d, %d, %d] ms\n", kTag,
>-        min_latency(), max_latency(), average_latency());
>+    PRINT("%s[min, max, avg]=[%d, %d, %d] ms\n", kTag, min_latency(),
>+          max_latency(), average_latency());
>   }
> 
>   int IndexToMilliseconds(double index) const {
>@@ -405,7 +392,7 @@ class MockAudioTransportAndroid : public test::MockAudioTransport {
>                                       const int32_t clockDrift,
>                                       const uint32_t currentMicLevel,
>                                       const bool keyPressed,
>-                                      uint32_t& newMicLevel) {
>+                                      uint32_t& newMicLevel) {  // NOLINT
>     EXPECT_TRUE(rec_mode()) << "No test is expecting these callbacks.";
>     rec_count_++;
>     // Process the recorded audio stream if an AudioStreamInterface
>@@ -424,7 +411,7 @@ class MockAudioTransportAndroid : public test::MockAudioTransport {
>                                const size_t nChannels,
>                                const uint32_t samplesPerSec,
>                                void* audioSamples,
>-                               size_t& nSamplesOut,
>+                               size_t& nSamplesOut,  // NOLINT
>                                int64_t* elapsed_time_ms,
>                                int64_t* ntp_time_ms) {
>     EXPECT_TRUE(play_mode()) << "No test is expecting these callbacks.";
>@@ -473,8 +460,7 @@ class MockAudioTransportAndroid : public test::MockAudioTransport {
> // AudioDeviceTest test fixture.
> class AudioDeviceTest : public ::testing::Test {
>  protected:
>-  AudioDeviceTest()
>-      : test_is_done_(EventWrapper::Create()) {
>+  AudioDeviceTest() : test_is_done_(EventWrapper::Create()) {
>     // One-time initialization of JVM and application context. Ensures that we
>     // can do calls between C++ and Java. Initializes both Java and OpenSL ES
>     // implementations.
>@@ -487,22 +473,12 @@ class AudioDeviceTest : public ::testing::Test {
>     record_parameters_ = audio_manager()->GetRecordAudioParameters();
>     build_info_.reset(new BuildInfo());
>   }
>-  virtual ~AudioDeviceTest() {
>-    EXPECT_EQ(0, audio_device_->Terminate());
>-  }
>+  virtual ~AudioDeviceTest() { EXPECT_EQ(0, audio_device_->Terminate()); }
> 
>-  int playout_sample_rate() const {
>-    return playout_parameters_.sample_rate();
>-  }
>-  int record_sample_rate() const {
>-    return record_parameters_.sample_rate();
>-  }
>-  size_t playout_channels() const {
>-    return playout_parameters_.channels();
>-  }
>-  size_t record_channels() const {
>-    return record_parameters_.channels();
>-  }
>+  int playout_sample_rate() const { return playout_parameters_.sample_rate(); }
>+  int record_sample_rate() const { return record_parameters_.sample_rate(); }
>+  size_t playout_channels() const { return playout_parameters_.channels(); }
>+  size_t record_channels() const { return record_parameters_.channels(); }
>   size_t playout_frames_per_10ms_buffer() const {
>     return playout_parameters_.frames_per_10ms_buffer();
>   }
>@@ -527,8 +503,8 @@ class AudioDeviceTest : public ::testing::Test {
>   }
> 
>   AudioManager* GetAudioManager(AudioDeviceModule* adm) const {
>-    return static_cast<AudioDeviceModuleImpl*>(adm)->
>-        GetAndroidAudioManagerForTest();
>+    return static_cast<AudioDeviceModuleImpl*>(adm)
>+        ->GetAndroidAudioManagerForTest();
>   }
> 
>   AudioDeviceBuffer* audio_device_buffer() const {
>@@ -546,9 +522,7 @@ class AudioDeviceTest : public ::testing::Test {
>   std::string GetFileName(int sample_rate) {
>     EXPECT_TRUE(sample_rate == 48000 || sample_rate == 44100);
>     char fname[64];
>-    snprintf(fname,
>-             sizeof(fname),
>-             "audio_device/audio_short%d",
>+    snprintf(fname, sizeof(fname), "audio_device/audio_short%d",
>              sample_rate / 1000);
>     std::string file_name(webrtc::test::ResourcePath(fname, "pcm"));
>     EXPECT_TRUE(test::FileExists(file_name));
>@@ -684,8 +658,11 @@ TEST_F(AudioDeviceTest, VerifyDefaultAudioLayer) {
>   const AudioDeviceModule::AudioLayer audio_layer = GetActiveAudioLayer();
>   bool low_latency_output = audio_manager()->IsLowLatencyPlayoutSupported();
>   bool low_latency_input = audio_manager()->IsLowLatencyRecordSupported();
>+  bool aaudio = audio_manager()->IsAAudioSupported();
>   AudioDeviceModule::AudioLayer expected_audio_layer;
>-  if (low_latency_output && low_latency_input) {
>+  if (aaudio) {
>+    expected_audio_layer = AudioDeviceModule::kAndroidAAudioAudio;
>+  } else if (low_latency_output && low_latency_input) {
>     expected_audio_layer = AudioDeviceModule::kAndroidOpenSLESAudio;
>   } else if (low_latency_output && !low_latency_input) {
>     expected_audio_layer =
>@@ -702,16 +679,16 @@ TEST_F(AudioDeviceTest, VerifyDefaultAudioLayer) {
> TEST_F(AudioDeviceTest, CorrectAudioLayerIsUsedForCombinedJavaOpenSLCombo) {
>   AudioDeviceModule::AudioLayer expected_layer =
>       AudioDeviceModule::kAndroidJavaInputAndOpenSLESOutputAudio;
>-  AudioDeviceModule::AudioLayer active_layer = TestActiveAudioLayer(
>-      expected_layer);
>+  AudioDeviceModule::AudioLayer active_layer =
>+      TestActiveAudioLayer(expected_layer);
>   EXPECT_EQ(expected_layer, active_layer);
> }
> 
> TEST_F(AudioDeviceTest, CorrectAudioLayerIsUsedForJavaInBothDirections) {
>   AudioDeviceModule::AudioLayer expected_layer =
>       AudioDeviceModule::kAndroidJavaAudio;
>-  AudioDeviceModule::AudioLayer active_layer = TestActiveAudioLayer(
>-      expected_layer);
>+  AudioDeviceModule::AudioLayer active_layer =
>+      TestActiveAudioLayer(expected_layer);
>   EXPECT_EQ(expected_layer, active_layer);
> }
> 
>@@ -723,6 +700,40 @@ TEST_F(AudioDeviceTest, CorrectAudioLayerIsUsedForOpenSLInBothDirections) {
>   EXPECT_EQ(expected_layer, active_layer);
> }
> 
>+// TODO(bugs.webrtc.org/8914)
>+#if !defined(AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO)
>+#define MAYBE_CorrectAudioLayerIsUsedForAAudioInBothDirections \
>+  DISABLED_CorrectAudioLayerIsUsedForAAudioInBothDirections
>+#else
>+#define MAYBE_CorrectAudioLayerIsUsedForAAudioInBothDirections \
>+  CorrectAudioLayerIsUsedForAAudioInBothDirections
>+#endif
>+TEST_F(AudioDeviceTest,
>+       MAYBE_CorrectAudioLayerIsUsedForAAudioInBothDirections) {
>+  AudioDeviceModule::AudioLayer expected_layer =
>+      AudioDeviceModule::kAndroidAAudioAudio;
>+  AudioDeviceModule::AudioLayer active_layer =
>+      TestActiveAudioLayer(expected_layer);
>+  EXPECT_EQ(expected_layer, active_layer);
>+}
>+
>+// TODO(bugs.webrtc.org/8914)
>+#if !defined(AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO)
>+#define MAYBE_CorrectAudioLayerIsUsedForCombinedJavaAAudioCombo \
>+  DISABLED_CorrectAudioLayerIsUsedForCombinedJavaAAudioCombo
>+#else
>+#define MAYBE_CorrectAudioLayerIsUsedForCombinedJavaAAudioCombo \
>+  CorrectAudioLayerIsUsedForCombinedJavaAAudioCombo
>+#endif
>+TEST_F(AudioDeviceTest,
>+       MAYBE_CorrectAudioLayerIsUsedForCombinedJavaAAudioCombo) {
>+  AudioDeviceModule::AudioLayer expected_layer =
>+      AudioDeviceModule::kAndroidJavaInputAndAAudioOutputAudio;
>+  AudioDeviceModule::AudioLayer active_layer =
>+      TestActiveAudioLayer(expected_layer);
>+  EXPECT_EQ(expected_layer, active_layer);
>+}
>+
> // The Android ADM supports two different delay reporting modes. One for the
> // low-latency output path (in combination with OpenSL ES), and one for the
> // high-latency output path (Java backends in both directions). These two tests
>@@ -737,7 +748,7 @@ TEST_F(AudioDeviceTest, UsesCorrectDelayEstimateForHighLatencyOutputPath) {
> TEST_F(AudioDeviceTest, UsesCorrectDelayEstimateForLowLatencyOutputPath) {
>   EXPECT_EQ(kLowLatencyModeDelayEstimateInMilliseconds,
>             TestDelayOnAudioLayer(
>-      AudioDeviceModule::kAndroidJavaInputAndOpenSLESOutputAudio));
>+                AudioDeviceModule::kAndroidJavaInputAndOpenSLESOutputAudio));
> }
> 
> // Ensure that the ADM internal audio device buffer is configured to use the
>@@ -753,7 +764,6 @@ TEST_F(AudioDeviceTest, VerifyAudioDeviceBufferParameters) {
>             audio_device_buffer()->RecordingChannels());
> }
> 
>-
> TEST_F(AudioDeviceTest, InitTerminate) {
>   // Initialization is part of the test fixture.
>   EXPECT_TRUE(audio_device()->Initialized());
>@@ -859,11 +869,8 @@ TEST_F(AudioDeviceTest, StartPlayoutVerifyCallbacks) {
>   MockAudioTransportAndroid mock(kPlayout);
>   mock.HandleCallbacks(test_is_done_.get(), nullptr, kNumCallbacks);
>   EXPECT_CALL(mock, NeedMorePlayData(playout_frames_per_10ms_buffer(),
>-                                     kBytesPerSample,
>-                                     playout_channels(),
>-                                     playout_sample_rate(),
>-                                     NotNull(),
>-                                     _, _, _))
>+                                     kBytesPerSample, playout_channels(),
>+                                     playout_sample_rate(), NotNull(), _, _, _))
>       .Times(AtLeast(kNumCallbacks));
>   EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
>   StartPlayout();
>@@ -873,19 +880,15 @@ TEST_F(AudioDeviceTest, StartPlayoutVerifyCallbacks) {
> 
> // Start recording and verify that the native audio layer starts feeding real
> // audio samples via the RecordedDataIsAvailable callback.
>+// TODO(henrika): investigate if it is possible to perform a sanity check of
>+// delay estimates as well (argument #6).
> TEST_F(AudioDeviceTest, StartRecordingVerifyCallbacks) {
>   MockAudioTransportAndroid mock(kRecording);
>   mock.HandleCallbacks(test_is_done_.get(), nullptr, kNumCallbacks);
>-  EXPECT_CALL(mock, RecordedDataIsAvailable(NotNull(),
>-                                            record_frames_per_10ms_buffer(),
>-                                            kBytesPerSample,
>-                                            record_channels(),
>-                                            record_sample_rate(),
>-                                            total_delay_ms(),
>-                                            0,
>-                                            0,
>-                                            false,
>-                                            _))
>+  EXPECT_CALL(
>+      mock, RecordedDataIsAvailable(NotNull(), record_frames_per_10ms_buffer(),
>+                                    kBytesPerSample, record_channels(),
>+                                    record_sample_rate(), _, 0, 0, false, _))
>       .Times(AtLeast(kNumCallbacks));
> 
>   EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
>@@ -894,29 +897,19 @@ TEST_F(AudioDeviceTest, StartRecordingVerifyCallbacks) {
>   StopRecording();
> }
> 
>-
> // Start playout and recording (full-duplex audio) and verify that audio is
> // active in both directions.
> TEST_F(AudioDeviceTest, StartPlayoutAndRecordingVerifyCallbacks) {
>   MockAudioTransportAndroid mock(kPlayout | kRecording);
>-  mock.HandleCallbacks(test_is_done_.get(), nullptr,  kNumCallbacks);
>+  mock.HandleCallbacks(test_is_done_.get(), nullptr, kNumCallbacks);
>   EXPECT_CALL(mock, NeedMorePlayData(playout_frames_per_10ms_buffer(),
>-                                     kBytesPerSample,
>-                                     playout_channels(),
>-                                     playout_sample_rate(),
>-                                     NotNull(),
>-                                     _, _, _))
>+                                     kBytesPerSample, playout_channels(),
>+                                     playout_sample_rate(), NotNull(), _, _, _))
>       .Times(AtLeast(kNumCallbacks));
>-  EXPECT_CALL(mock, RecordedDataIsAvailable(NotNull(),
>-                                            record_frames_per_10ms_buffer(),
>-                                            kBytesPerSample,
>-                                            record_channels(),
>-                                            record_sample_rate(),
>-                                            total_delay_ms(),
>-                                            0,
>-                                            0,
>-                                            false,
>-                                            _))
>+  EXPECT_CALL(
>+      mock, RecordedDataIsAvailable(NotNull(), record_frames_per_10ms_buffer(),
>+                                    kBytesPerSample, record_channels(),
>+                                    record_sample_rate(), _, 0, 0, false, _))
>       .Times(AtLeast(kNumCallbacks));
>   EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
>   StartPlayout();
>@@ -937,8 +930,7 @@ TEST_F(AudioDeviceTest, RunPlayoutWithFileAsSource) {
>   std::string file_name = GetFileName(playout_sample_rate());
>   std::unique_ptr<FileAudioStream> file_audio_stream(
>       new FileAudioStream(num_callbacks, file_name, playout_sample_rate()));
>-  mock.HandleCallbacks(test_is_done_.get(),
>-                       file_audio_stream.get(),
>+  mock.HandleCallbacks(test_is_done_.get(), file_audio_stream.get(),
>                        num_callbacks);
>   // SetMaxPlayoutVolume();
>   EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
>@@ -972,15 +964,14 @@ TEST_F(AudioDeviceTest, DISABLED_RunPlayoutAndRecordingInFullDuplex) {
>   NiceMock<MockAudioTransportAndroid> mock(kPlayout | kRecording);
>   std::unique_ptr<FifoAudioStream> fifo_audio_stream(
>       new FifoAudioStream(playout_frames_per_10ms_buffer()));
>-  mock.HandleCallbacks(test_is_done_.get(),
>-                       fifo_audio_stream.get(),
>+  mock.HandleCallbacks(test_is_done_.get(), fifo_audio_stream.get(),
>                        kFullDuplexTimeInSec * kNumCallbacksPerSecond);
>   SetMaxPlayoutVolume();
>   EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
>   StartRecording();
>   StartPlayout();
>-  test_is_done_->Wait(std::max(kTestTimeOutInMilliseconds,
>-                               1000 * kFullDuplexTimeInSec));
>+  test_is_done_->Wait(
>+      std::max(kTestTimeOutInMilliseconds, 1000 * kFullDuplexTimeInSec));
>   StopPlayout();
>   StopRecording();
> 
>@@ -1006,16 +997,15 @@ TEST_F(AudioDeviceTest, DISABLED_MeasureLoopbackLatency) {
>   NiceMock<MockAudioTransportAndroid> mock(kPlayout | kRecording);
>   std::unique_ptr<LatencyMeasuringAudioStream> latency_audio_stream(
>       new LatencyMeasuringAudioStream(playout_frames_per_10ms_buffer()));
>-  mock.HandleCallbacks(test_is_done_.get(),
>-                       latency_audio_stream.get(),
>+  mock.HandleCallbacks(test_is_done_.get(), latency_audio_stream.get(),
>                        kMeasureLatencyTimeInSec * kNumCallbacksPerSecond);
>   EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
>   SetMaxPlayoutVolume();
>   DisableBuiltInAECIfAvailable();
>   StartRecording();
>   StartPlayout();
>-  test_is_done_->Wait(std::max(kTestTimeOutInMilliseconds,
>-                               1000 * kMeasureLatencyTimeInSec));
>+  test_is_done_->Wait(
>+      std::max(kTestTimeOutInMilliseconds, 1000 * kMeasureLatencyTimeInSec));
>   StopPlayout();
>   StopRecording();
>   // Verify that the correct number of transmitted impulses are detected.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/audio_manager.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/audio_manager.cc
>index fae4ecce0cd..98a2afd20c9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/audio_manager.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/audio_manager.cc
>@@ -12,21 +12,13 @@
> 
> #include <utility>
> 
>-#include <android/log.h>
>-
> #include "modules/audio_device/android/audio_common.h"
> #include "modules/utility/include/helpers_android.h"
> #include "rtc_base/arraysize.h"
> #include "rtc_base/checks.h"
>+#include "rtc_base/logging.h"
> #include "rtc_base/platform_thread.h"
> 
>-#define TAG "AudioManager"
>-#define ALOGV(...) __android_log_print(ANDROID_LOG_VERBOSE, TAG, __VA_ARGS__)
>-#define ALOGD(...) __android_log_print(ANDROID_LOG_DEBUG, TAG, __VA_ARGS__)
>-#define ALOGE(...) __android_log_print(ANDROID_LOG_ERROR, TAG, __VA_ARGS__)
>-#define ALOGW(...) __android_log_print(ANDROID_LOG_WARN, TAG, __VA_ARGS__)
>-#define ALOGI(...) __android_log_print(ANDROID_LOG_INFO, TAG, __VA_ARGS__)
>-
> namespace webrtc {
> 
> // AudioManager::JavaAudioManager implementation
>@@ -41,11 +33,11 @@ AudioManager::JavaAudioManager::JavaAudioManager(
>       is_device_blacklisted_for_open_sles_usage_(
>           native_reg->GetMethodId("isDeviceBlacklistedForOpenSLESUsage",
>                                   "()Z")) {
>-  ALOGD("JavaAudioManager::ctor @[tid=%d]", rtc::CurrentThreadId());
>+  RTC_LOG(INFO) << "JavaAudioManager::ctor";
> }
> 
> AudioManager::JavaAudioManager::~JavaAudioManager() {
>-  ALOGD("JavaAudioManager::dtor[tid=%d]", rtc::CurrentThreadId());
>+  RTC_LOG(INFO) << "JavaAudioManager::~dtor";
> }
> 
> bool AudioManager::JavaAudioManager::Init() {
>@@ -76,10 +68,10 @@ AudioManager::AudioManager()
>       low_latency_playout_(false),
>       low_latency_record_(false),
>       delay_estimate_in_milliseconds_(0) {
>-  ALOGD("ctor[tid=%d]", rtc::CurrentThreadId());
>+  RTC_LOG(INFO) << "ctor";
>   RTC_CHECK(j_environment_);
>   JNINativeMethod native_methods[] = {
>-      {"nativeCacheAudioParameters", "(IIIZZZZZZIIJ)V",
>+      {"nativeCacheAudioParameters", "(IIIZZZZZZZIIJ)V",
>        reinterpret_cast<void*>(&webrtc::AudioManager::CacheAudioParameters)}};
>   j_native_registration_ = j_environment_->RegisterNatives(
>       "org/webrtc/voiceengine/WebRtcAudioManager", native_methods,
>@@ -91,14 +83,14 @@ AudioManager::AudioManager()
> }
> 
> AudioManager::~AudioManager() {
>-  ALOGD("~dtor[tid=%d]", rtc::CurrentThreadId());
>+  RTC_LOG(INFO) << "dtor";
>   RTC_DCHECK(thread_checker_.CalledOnValidThread());
>   Close();
> }
> 
> void AudioManager::SetActiveAudioLayer(
>     AudioDeviceModule::AudioLayer audio_layer) {
>-  ALOGD("SetActiveAudioLayer(%d)[tid=%d]", audio_layer, rtc::CurrentThreadId());
>+  RTC_LOG(INFO) << "SetActiveAudioLayer: " << audio_layer;
>   RTC_DCHECK(thread_checker_.CalledOnValidThread());
>   RTC_DCHECK(!initialized_);
>   // Store the currently utilized audio layer.
>@@ -111,25 +103,27 @@ void AudioManager::SetActiveAudioLayer(
>       (audio_layer == AudioDeviceModule::kAndroidJavaAudio)
>           ? kHighLatencyModeDelayEstimateInMilliseconds
>           : kLowLatencyModeDelayEstimateInMilliseconds;
>-  ALOGD("delay_estimate_in_milliseconds: %d", delay_estimate_in_milliseconds_);
>+  RTC_LOG(INFO) << "delay_estimate_in_milliseconds: "
>+                << delay_estimate_in_milliseconds_;
> }
> 
> SLObjectItf AudioManager::GetOpenSLEngine() {
>-  ALOGD("GetOpenSLEngine[tid=%d]", rtc::CurrentThreadId());
>+  RTC_LOG(INFO) << "GetOpenSLEngine";
>   RTC_DCHECK(thread_checker_.CalledOnValidThread());
>   // Only allow usage of OpenSL ES if such an audio layer has been specified.
>   if (audio_layer_ != AudioDeviceModule::kAndroidOpenSLESAudio &&
>       audio_layer_ !=
>           AudioDeviceModule::kAndroidJavaInputAndOpenSLESOutputAudio) {
>-    ALOGI("Unable to create OpenSL engine for the current audio layer: %d",
>-          audio_layer_);
>+    RTC_LOG(INFO)
>+        << "Unable to create OpenSL engine for the current audio layer: "
>+        << audio_layer_;
>     return nullptr;
>   }
>   // OpenSL ES for Android only supports a single engine per application.
>   // If one already has been created, return existing object instead of
>   // creating a new.
>   if (engine_object_.Get() != nullptr) {
>-    ALOGI("The OpenSL ES engine object has already been created");
>+    RTC_LOG(WARNING) << "The OpenSL ES engine object has already been created";
>     return engine_object_.Get();
>   }
>   // Create the engine object in thread safe mode.
>@@ -138,14 +132,15 @@ SLObjectItf AudioManager::GetOpenSLEngine() {
>   SLresult result =
>       slCreateEngine(engine_object_.Receive(), 1, option, 0, NULL, NULL);
>   if (result != SL_RESULT_SUCCESS) {
>-    ALOGE("slCreateEngine() failed: %s", GetSLErrorString(result));
>+    RTC_LOG(LS_ERROR) << "slCreateEngine() failed: "
>+                      << GetSLErrorString(result);
>     engine_object_.Reset();
>     return nullptr;
>   }
>   // Realize the SL Engine in synchronous mode.
>   result = engine_object_->Realize(engine_object_.Get(), SL_BOOLEAN_FALSE);
>   if (result != SL_RESULT_SUCCESS) {
>-    ALOGE("Realize() failed: %s", GetSLErrorString(result));
>+    RTC_LOG(LS_ERROR) << "Realize() failed: " << GetSLErrorString(result);
>     engine_object_.Reset();
>     return nullptr;
>   }
>@@ -154,12 +149,12 @@ SLObjectItf AudioManager::GetOpenSLEngine() {
> }
> 
> bool AudioManager::Init() {
>-  ALOGD("Init[tid=%d]", rtc::CurrentThreadId());
>+  RTC_LOG(INFO) << "Init";
>   RTC_DCHECK(thread_checker_.CalledOnValidThread());
>   RTC_DCHECK(!initialized_);
>   RTC_DCHECK_NE(audio_layer_, AudioDeviceModule::kPlatformDefaultAudio);
>   if (!j_audio_manager_->Init()) {
>-    ALOGE("init failed!");
>+    RTC_LOG(LS_ERROR) << "Init() failed";
>     return false;
>   }
>   initialized_ = true;
>@@ -167,7 +162,7 @@ bool AudioManager::Init() {
> }
> 
> bool AudioManager::Close() {
>-  ALOGD("Close[tid=%d]", rtc::CurrentThreadId());
>+  RTC_LOG(INFO) << "Close";
>   RTC_DCHECK(thread_checker_.CalledOnValidThread());
>   if (!initialized_)
>     return true;
>@@ -177,7 +172,6 @@ bool AudioManager::Close() {
> }
> 
> bool AudioManager::IsCommunicationModeEnabled() const {
>-  ALOGD("IsCommunicationModeEnabled()");
>   RTC_DCHECK(thread_checker_.CalledOnValidThread());
>   return j_audio_manager_->IsCommunicationModeEnabled();
> }
>@@ -199,7 +193,6 @@ bool AudioManager::IsNoiseSuppressorSupported() const {
> 
> bool AudioManager::IsLowLatencyPlayoutSupported() const {
>   RTC_DCHECK(thread_checker_.CalledOnValidThread());
>-  ALOGD("IsLowLatencyPlayoutSupported()");
>   // Some devices are blacklisted for usage of OpenSL ES even if they report
>   // that low-latency playout is supported. See b/21485703 for details.
>   return j_audio_manager_->IsDeviceBlacklistedForOpenSLESUsage()
>@@ -209,28 +202,33 @@ bool AudioManager::IsLowLatencyPlayoutSupported() const {
> 
> bool AudioManager::IsLowLatencyRecordSupported() const {
>   RTC_DCHECK(thread_checker_.CalledOnValidThread());
>-  ALOGD("IsLowLatencyRecordSupported()");
>   return low_latency_record_;
> }
> 
> bool AudioManager::IsProAudioSupported() const {
>   RTC_DCHECK(thread_checker_.CalledOnValidThread());
>-  ALOGD("IsProAudioSupported()");
>   // TODO(henrika): return the state independently of if OpenSL ES is
>   // blacklisted or not for now. We could use the same approach as in
>   // IsLowLatencyPlayoutSupported() but I can't see the need for it yet.
>   return pro_audio_;
> }
> 
>+// TODO(henrika): improve comments...
>+bool AudioManager::IsAAudioSupported() const {
>+#if defined(AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO)
>+  return a_audio_;
>+#else
>+  return false;
>+#endif
>+}
>+
> bool AudioManager::IsStereoPlayoutSupported() const {
>   RTC_DCHECK(thread_checker_.CalledOnValidThread());
>-  ALOGD("IsStereoPlayoutSupported()");
>   return (playout_parameters_.channels() == 2);
> }
> 
> bool AudioManager::IsStereoRecordSupported() const {
>   RTC_DCHECK(thread_checker_.CalledOnValidThread());
>-  ALOGD("IsStereoRecordSupported()");
>   return (record_parameters_.channels() == 2);
> }
> 
>@@ -238,6 +236,7 @@ int AudioManager::GetDelayEstimateInMilliseconds() const {
>   return delay_estimate_in_milliseconds_;
> }
> 
>+JNI_FUNCTION_ALIGN
> void JNICALL AudioManager::CacheAudioParameters(JNIEnv* env,
>                                                 jobject obj,
>                                                 jint sample_rate,
>@@ -249,6 +248,7 @@ void JNICALL AudioManager::CacheAudioParameters(JNIEnv* env,
>                                                 jboolean low_latency_output,
>                                                 jboolean low_latency_input,
>                                                 jboolean pro_audio,
>+                                                jboolean a_audio,
>                                                 jint output_buffer_size,
>                                                 jint input_buffer_size,
>                                                 jlong native_audio_manager) {
>@@ -257,7 +257,7 @@ void JNICALL AudioManager::CacheAudioParameters(JNIEnv* env,
>   this_object->OnCacheAudioParameters(
>       env, sample_rate, output_channels, input_channels, hardware_aec,
>       hardware_agc, hardware_ns, low_latency_output, low_latency_input,
>-      pro_audio, output_buffer_size, input_buffer_size);
>+      pro_audio, a_audio, output_buffer_size, input_buffer_size);
> }
> 
> void AudioManager::OnCacheAudioParameters(JNIEnv* env,
>@@ -270,20 +270,23 @@ void AudioManager::OnCacheAudioParameters(JNIEnv* env,
>                                           jboolean low_latency_output,
>                                           jboolean low_latency_input,
>                                           jboolean pro_audio,
>+                                          jboolean a_audio,
>                                           jint output_buffer_size,
>                                           jint input_buffer_size) {
>-  ALOGD("OnCacheAudioParameters[tid=%d]", rtc::CurrentThreadId());
>-  ALOGD("hardware_aec: %d", hardware_aec);
>-  ALOGD("hardware_agc: %d", hardware_agc);
>-  ALOGD("hardware_ns: %d", hardware_ns);
>-  ALOGD("low_latency_output: %d", low_latency_output);
>-  ALOGD("low_latency_input: %d", low_latency_input);
>-  ALOGD("pro_audio: %d", pro_audio);
>-  ALOGD("sample_rate: %d", sample_rate);
>-  ALOGD("output_channels: %d", output_channels);
>-  ALOGD("input_channels: %d", input_channels);
>-  ALOGD("output_buffer_size: %d", output_buffer_size);
>-  ALOGD("input_buffer_size: %d", input_buffer_size);
>+  RTC_LOG(INFO)
>+      << "OnCacheAudioParameters: "
>+      << "hardware_aec: " << static_cast<bool>(hardware_aec)
>+      << ", hardware_agc: " << static_cast<bool>(hardware_agc)
>+      << ", hardware_ns: " << static_cast<bool>(hardware_ns)
>+      << ", low_latency_output: " << static_cast<bool>(low_latency_output)
>+      << ", low_latency_input: " << static_cast<bool>(low_latency_input)
>+      << ", pro_audio: " << static_cast<bool>(pro_audio)
>+      << ", a_audio: " << static_cast<bool>(a_audio)
>+      << ", sample_rate: " << static_cast<int>(sample_rate)
>+      << ", output_channels: " << static_cast<int>(output_channels)
>+      << ", input_channels: " << static_cast<int>(input_channels)
>+      << ", output_buffer_size: " << static_cast<int>(output_buffer_size)
>+      << ", input_buffer_size: " << static_cast<int>(input_buffer_size);
>   RTC_DCHECK(thread_checker_.CalledOnValidThread());
>   hardware_aec_ = hardware_aec;
>   hardware_agc_ = hardware_agc;
>@@ -291,6 +294,7 @@ void AudioManager::OnCacheAudioParameters(JNIEnv* env,
>   low_latency_playout_ = low_latency_output;
>   low_latency_record_ = low_latency_input;
>   pro_audio_ = pro_audio;
>+  a_audio_ = a_audio;
>   playout_parameters_.reset(sample_rate, static_cast<size_t>(output_channels),
>                             static_cast<size_t>(output_buffer_size));
>   record_parameters_.reset(sample_rate, static_cast<size_t>(input_channels),
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/audio_manager.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/audio_manager.h
>index dd6b51ff90d..d2d0eb623b4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/audio_manager.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/audio_manager.h
>@@ -11,10 +11,10 @@
> #ifndef MODULES_AUDIO_DEVICE_ANDROID_AUDIO_MANAGER_H_
> #define MODULES_AUDIO_DEVICE_ANDROID_AUDIO_MANAGER_H_
> 
>-#include <memory>
>-
>-#include <jni.h>
> #include <SLES/OpenSLES.h>
>+#include <jni.h>
>+
>+#include <memory>
> 
> #include "modules/audio_device/android/audio_common.h"
> #include "modules/audio_device/android/opensles_common.h"
>@@ -115,6 +115,9 @@ class AudioManager {
>   // OpenSL ES.
>   bool IsProAudioSupported() const;
> 
>+  // Returns true if the device supports AAudio.
>+  bool IsAAudioSupported() const;
>+
>   // Returns the estimated total delay of this device. Unit is in milliseconds.
>   // The vaule is set once at construction and never changes after that.
>   // Possible values are webrtc::kLowLatencyModeDelayEstimateInMilliseconds and
>@@ -136,6 +139,7 @@ class AudioManager {
>                                            jboolean low_latency_output,
>                                            jboolean low_latency_input,
>                                            jboolean pro_audio,
>+                                           jboolean a_audio,
>                                            jint output_buffer_size,
>                                            jint input_buffer_size,
>                                            jlong native_audio_manager);
>@@ -149,6 +153,7 @@ class AudioManager {
>                               jboolean low_latency_output,
>                               jboolean low_latency_input,
>                               jboolean pro_audio,
>+                              jboolean a_audio,
>                               jint output_buffer_size,
>                               jint input_buffer_size);
> 
>@@ -200,6 +205,9 @@ class AudioManager {
>   // True if device supports the low-latency OpenSL ES pro-audio path.
>   bool pro_audio_;
> 
>+  // True if device supports the low-latency AAudio audio path.
>+  bool a_audio_;
>+
>   // The delay estimate can take one of two fixed values depending on if the
>   // device supports low-latency output or not.
>   int delay_estimate_in_milliseconds_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/audio_manager_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/audio_manager_unittest.cc
>index 00314ebfebd..ed713cbede1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/audio_manager_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/audio_manager_unittest.cc
>@@ -73,8 +73,7 @@ class AudioManagerTest : public ::testing::Test {
>   AudioParameters record_parameters_;
> };
> 
>-TEST_F(AudioManagerTest, ConstructDestruct) {
>-}
>+TEST_F(AudioManagerTest, ConstructDestruct) {}
> 
> // It should not be possible to create an OpenSL engine object if Java based
> // audio is requested in both directions.
>@@ -187,8 +186,8 @@ TEST_F(AudioManagerTest, ShowDeviceInfo) {
>   BuildInfo build_info;
>   PRINT("%smodel: %s\n", kTag, build_info.GetDeviceModel().c_str());
>   PRINT("%sbrand: %s\n", kTag, build_info.GetBrand().c_str());
>-  PRINT("%smanufacturer: %s\n",
>-        kTag, build_info.GetDeviceManufacturer().c_str());
>+  PRINT("%smanufacturer: %s\n", kTag,
>+        build_info.GetDeviceManufacturer().c_str());
> }
> 
> // Add Android build information to the test for logging purposes.
>@@ -238,4 +237,3 @@ TEST_F(AudioManagerTest, AudioParametersWithNonDefaultConstruction) {
> }
> 
> }  // namespace webrtc
>-
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/audio_record_jni.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/audio_record_jni.cc
>index 0d5d32ed8ff..97ba6c668c2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/audio_record_jni.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/audio_record_jni.cc
>@@ -140,7 +140,7 @@ int32_t AudioRecordJni::InitRecording() {
>   RTC_DCHECK(!recording_);
>   ScopedHistogramTimer timer("WebRTC.Audio.InitRecordingDurationMs");
>   int frames_per_buffer = j_audio_record_->InitRecording(
>-        audio_parameters_.sample_rate(), audio_parameters_.channels());
>+      audio_parameters_.sample_rate(), audio_parameters_.channels());
>   if (frames_per_buffer < 0) {
>     direct_buffer_address_ = nullptr;
>     RTC_LOG(LS_ERROR) << "InitRecording failed";
>@@ -159,8 +159,12 @@ int32_t AudioRecordJni::InitRecording() {
> int32_t AudioRecordJni::StartRecording() {
>   RTC_LOG(INFO) << "StartRecording";
>   RTC_DCHECK(thread_checker_.CalledOnValidThread());
>-  RTC_DCHECK(initialized_);
>   RTC_DCHECK(!recording_);
>+  if (!initialized_) {
>+    RTC_DLOG(LS_WARNING)
>+        << "Recording can not start since InitRecording must succeed first";
>+    return 0;
>+  }
>   ScopedHistogramTimer timer("WebRTC.Audio.StartRecordingDurationMs");
>   if (!j_audio_record_->StartRecording()) {
>     RTC_LOG(LS_ERROR) << "StartRecording failed";
>@@ -225,6 +229,7 @@ int32_t AudioRecordJni::EnableBuiltInNS(bool enable) {
>   return j_audio_record_->EnableBuiltInNS(enable) ? 0 : -1;
> }
> 
>+JNI_FUNCTION_ALIGN
> void JNICALL AudioRecordJni::CacheDirectBufferAddress(JNIEnv* env,
>                                                       jobject obj,
>                                                       jobject byte_buffer,
>@@ -241,10 +246,11 @@ void AudioRecordJni::OnCacheDirectBufferAddress(JNIEnv* env,
>   RTC_DCHECK(!direct_buffer_address_);
>   direct_buffer_address_ = env->GetDirectBufferAddress(byte_buffer);
>   jlong capacity = env->GetDirectBufferCapacity(byte_buffer);
>-  RTC_LOG(INFO) <<  "direct buffer capacity: " << capacity;
>+  RTC_LOG(INFO) << "direct buffer capacity: " << capacity;
>   direct_buffer_capacity_in_bytes_ = static_cast<size_t>(capacity);
> }
> 
>+JNI_FUNCTION_ALIGN
> void JNICALL AudioRecordJni::DataIsRecorded(JNIEnv* env,
>                                             jobject obj,
>                                             jint length,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/audio_record_jni.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/audio_record_jni.h
>index f508280ad0b..a700a51b2ba 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/audio_record_jni.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/audio_record_jni.h
>@@ -48,7 +48,7 @@ class AudioRecordJni {
>   class JavaAudioRecord {
>    public:
>     JavaAudioRecord(NativeRegistration* native_registration,
>-                   std::unique_ptr<GlobalRef> audio_track);
>+                    std::unique_ptr<GlobalRef> audio_track);
>     ~JavaAudioRecord();
> 
>     int InitRecording(int sample_rate, size_t channels);
>@@ -91,8 +91,10 @@ class AudioRecordJni {
>   // is also stored in |direct_buffer_capacity_in_bytes_|.
>   // This method will be called by the WebRtcAudioRecord constructor, i.e.,
>   // on the same thread that this object is created on.
>-  static void JNICALL CacheDirectBufferAddress(
>-    JNIEnv* env, jobject obj, jobject byte_buffer, jlong nativeAudioRecord);
>+  static void JNICALL CacheDirectBufferAddress(JNIEnv* env,
>+                                               jobject obj,
>+                                               jobject byte_buffer,
>+                                               jlong nativeAudioRecord);
>   void OnCacheDirectBufferAddress(JNIEnv* env, jobject byte_buffer);
> 
>   // Called periodically by the Java based WebRtcAudioRecord object when
>@@ -101,8 +103,10 @@ class AudioRecordJni {
>   // now time to send these to the consumer.
>   // This method is called on a high-priority thread from Java. The name of
>   // the thread is 'AudioRecordThread'.
>-  static void JNICALL DataIsRecorded(
>-    JNIEnv* env, jobject obj, jint length, jlong nativeAudioRecord);
>+  static void JNICALL DataIsRecorded(JNIEnv* env,
>+                                     jobject obj,
>+                                     jint length,
>+                                     jlong nativeAudioRecord);
>   void OnDataIsRecorded(int length);
> 
>   // Stores thread ID in constructor.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/audio_track_jni.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/audio_track_jni.cc
>index 04ca35a7c02..b0e5ddf5f6e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/audio_track_jni.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/audio_track_jni.cc
>@@ -127,8 +127,12 @@ int32_t AudioTrackJni::InitPlayout() {
> int32_t AudioTrackJni::StartPlayout() {
>   RTC_LOG(INFO) << "StartPlayout";
>   RTC_DCHECK(thread_checker_.CalledOnValidThread());
>-  RTC_DCHECK(initialized_);
>   RTC_DCHECK(!playing_);
>+  if (!initialized_) {
>+    RTC_DLOG(LS_WARNING)
>+        << "Playout can not start since InitPlayout must succeed first";
>+    return 0;
>+  }
>   if (!j_audio_track_->StartPlayout()) {
>     RTC_LOG(LS_ERROR) << "StartPlayout failed";
>     return -1;
>@@ -200,6 +204,7 @@ void AudioTrackJni::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
>   audio_device_buffer_->SetPlayoutChannels(channels);
> }
> 
>+JNI_FUNCTION_ALIGN
> void JNICALL AudioTrackJni::CacheDirectBufferAddress(JNIEnv* env,
>                                                      jobject obj,
>                                                      jobject byte_buffer,
>@@ -216,13 +221,14 @@ void AudioTrackJni::OnCacheDirectBufferAddress(JNIEnv* env,
>   RTC_DCHECK(!direct_buffer_address_);
>   direct_buffer_address_ = env->GetDirectBufferAddress(byte_buffer);
>   jlong capacity = env->GetDirectBufferCapacity(byte_buffer);
>-  RTC_LOG(INFO) <<  "direct buffer capacity: " << capacity;
>+  RTC_LOG(INFO) << "direct buffer capacity: " << capacity;
>   direct_buffer_capacity_in_bytes_ = static_cast<size_t>(capacity);
>   const size_t bytes_per_frame = audio_parameters_.channels() * sizeof(int16_t);
>   frames_per_buffer_ = direct_buffer_capacity_in_bytes_ / bytes_per_frame;
>-  RTC_LOG(INFO) <<  "frames_per_buffer: " << frames_per_buffer_;
>+  RTC_LOG(INFO) << "frames_per_buffer: " << frames_per_buffer_;
> }
> 
>+JNI_FUNCTION_ALIGN
> void JNICALL AudioTrackJni::GetPlayoutData(JNIEnv* env,
>                                            jobject obj,
>                                            jint length,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/audio_track_jni.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/audio_track_jni.h
>index fd9c61e1ff8..469aa82bbdc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/audio_track_jni.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/audio_track_jni.h
>@@ -90,8 +90,10 @@ class AudioTrackJni {
>   // |byte_buffer| in |direct_buffer_address_|. The size of the buffer
>   // is also stored in |direct_buffer_capacity_in_bytes_|.
>   // Called on the same thread as the creating thread.
>-  static void JNICALL CacheDirectBufferAddress(
>-    JNIEnv* env, jobject obj, jobject byte_buffer, jlong nativeAudioTrack);
>+  static void JNICALL CacheDirectBufferAddress(JNIEnv* env,
>+                                               jobject obj,
>+                                               jobject byte_buffer,
>+                                               jlong nativeAudioTrack);
>   void OnCacheDirectBufferAddress(JNIEnv* env, jobject byte_buffer);
> 
>   // Called periodically by the Java based WebRtcAudioTrack object when
>@@ -99,8 +101,10 @@ class AudioTrackJni {
>   // be written to the memory area |direct_buffer_address_| for playout.
>   // This method is called on a high-priority thread from Java. The name of
>   // the thread is 'AudioTrackThread'.
>-  static void JNICALL GetPlayoutData(
>-    JNIEnv* env, jobject obj, jint length, jlong nativeAudioTrack);
>+  static void JNICALL GetPlayoutData(JNIEnv* env,
>+                                     jobject obj,
>+                                     jint length,
>+                                     jlong nativeAudioTrack);
>   void OnGetPlayoutData(size_t length);
> 
>   // Stores thread ID in constructor.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/build_info.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/build_info.cc
>index 9eab4b33efe..916be8244eb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/build_info.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/build_info.cc
>@@ -21,8 +21,8 @@ BuildInfo::BuildInfo()
> 
> std::string BuildInfo::GetStringFromJava(const char* name) {
>   jmethodID id = j_build_info_.GetStaticMethodId(name, "()Ljava/lang/String;");
>-  jstring j_string = static_cast<jstring>(
>-      j_build_info_.CallStaticObjectMethod(id));
>+  jstring j_string =
>+      static_cast<jstring>(j_build_info_.CallStaticObjectMethod(id));
>   return j_environment_->JavaToStdString(j_string);
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioEffects.java b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioEffects.java
>index 90a48d3bc27..24ba0a12b87 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioEffects.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioEffects.java
>@@ -19,6 +19,7 @@ import android.media.audiofx.NoiseSuppressor;
> import android.os.Build;
> import java.util.List;
> import java.util.UUID;
>+import javax.annotation.Nullable;
> import org.webrtc.Logging;
> 
> // This class wraps control of three different platform effects. Supported
>@@ -40,12 +41,12 @@ public class WebRtcAudioEffects {
>   // Contains the available effect descriptors returned from the
>   // AudioEffect.getEffects() call. This result is cached to avoid doing the
>   // slow OS call multiple times.
>-  private static Descriptor[] cachedEffects = null;
>+  private static @Nullable Descriptor[] cachedEffects = null;
> 
>   // Contains the audio effect objects. Created in enable() and destroyed
>   // in release().
>-  private AcousticEchoCanceler aec = null;
>-  private NoiseSuppressor ns = null;
>+  private @Nullable AcousticEchoCanceler aec = null;
>+  private @Nullable NoiseSuppressor ns = null;
> 
>   // Affects the final state given to the setEnabled() method on each effect.
>   // The default state is set to "disabled" but each effect can also be enabled
>@@ -291,7 +292,7 @@ public class WebRtcAudioEffects {
> 
>   // Returns the cached copy of the audio effects array, if available, or
>   // queries the operating system for the list of effects.
>-  private static Descriptor[] getAvailableEffects() {
>+  private static @Nullable Descriptor[] getAvailableEffects() {
>     if (cachedEffects != null) {
>       return cachedEffects;
>     }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioManager.java b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioManager.java
>index 08979aa6fa6..dea84ca2268 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioManager.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioManager.java
>@@ -20,6 +20,7 @@ import android.media.AudioTrack;
> import android.os.Build;
> import java.util.Timer;
> import java.util.TimerTask;
>+import javax.annotation.Nullable;
> import org.webrtc.ContextUtils;
> import org.webrtc.Logging;
> 
>@@ -37,6 +38,10 @@ public class WebRtcAudioManager {
> 
>   private static final String TAG = "WebRtcAudioManager";
> 
>+  // TODO(bugs.webrtc.org/8914): disabled by default until AAudio support has
>+  // been completed. Goal is to always return false on Android O MR1 and higher.
>+  private static final boolean blacklistDeviceForAAudioUsage = true;
>+
>   // Use mono as default for both audio directions.
>   private static boolean useStereoOutput = false;
>   private static boolean useStereoInput = false;
>@@ -98,7 +103,7 @@ public class WebRtcAudioManager {
>     private static final int TIMER_PERIOD_IN_SECONDS = 30;
> 
>     private final AudioManager audioManager;
>-    private Timer timer;
>+    private @Nullable Timer timer;
> 
>     public VolumeLogger(AudioManager audioManager) {
>       this.audioManager = audioManager;
>@@ -156,6 +161,7 @@ public class WebRtcAudioManager {
>   private boolean lowLatencyOutput;
>   private boolean lowLatencyInput;
>   private boolean proAudio;
>+  private boolean aAudio;
>   private int sampleRate;
>   private int outputChannels;
>   private int inputChannels;
>@@ -175,8 +181,9 @@ public class WebRtcAudioManager {
>     volumeLogger = new VolumeLogger(audioManager);
>     storeAudioParameters();
>     nativeCacheAudioParameters(sampleRate, outputChannels, inputChannels, hardwareAEC, hardwareAGC,
>-        hardwareNS, lowLatencyOutput, lowLatencyInput, proAudio, outputBufferSize, inputBufferSize,
>-        nativeAudioManager);
>+        hardwareNS, lowLatencyOutput, lowLatencyInput, proAudio, aAudio, outputBufferSize,
>+        inputBufferSize, nativeAudioManager);
>+    WebRtcAudioUtils.logAudioState(TAG);
>   }
> 
>   private boolean init() {
>@@ -225,6 +232,7 @@ public class WebRtcAudioManager {
>     lowLatencyOutput = isLowLatencyOutputSupported();
>     lowLatencyInput = isLowLatencyInputSupported();
>     proAudio = isProAudioSupported();
>+    aAudio = isAAudioSupported();
>     outputBufferSize = lowLatencyOutput ? getLowLatencyOutputFramesPerBuffer()
>                                         : getMinOutputFrameSize(sampleRate, outputChannels);
>     inputBufferSize = lowLatencyInput ? getLowLatencyInputFramesPerBuffer()
>@@ -263,6 +271,15 @@ public class WebRtcAudioManager {
>                PackageManager.FEATURE_AUDIO_PRO);
>   }
> 
>+  // AAudio is supported on Androio Oreo MR1 (API 27) and higher.
>+  // TODO(bugs.webrtc.org/8914): currently disabled by default.
>+  private boolean isAAudioSupported() {
>+    if (blacklistDeviceForAAudioUsage) {
>+      Logging.w(TAG, "AAudio support is currently disabled on all devices!");
>+    }
>+    return !blacklistDeviceForAAudioUsage && WebRtcAudioUtils.runningOnOreoMR1OrHigher();
>+  }
>+
>   // Returns the native output sample rate for this device's output stream.
>   private int getNativeOutputSampleRate() {
>     // Override this if we're running on an old emulator image which only
>@@ -361,6 +378,6 @@ public class WebRtcAudioManager {
> 
>   private native void nativeCacheAudioParameters(int sampleRate, int outputChannels,
>       int inputChannels, boolean hardwareAEC, boolean hardwareAGC, boolean hardwareNS,
>-      boolean lowLatencyOutput, boolean lowLatencyInput, boolean proAudio, int outputBufferSize,
>-      int inputBufferSize, long nativeAudioManager);
>+      boolean lowLatencyOutput, boolean lowLatencyInput, boolean proAudio, boolean aAudio,
>+      int outputBufferSize, int inputBufferSize, long nativeAudioManager);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioRecord.java b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioRecord.java
>index 2b4ef1a0030..e3946a6249a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioRecord.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioRecord.java
>@@ -17,7 +17,9 @@ import android.media.MediaRecorder.AudioSource;
> import android.os.Process;
> import java.lang.System;
> import java.nio.ByteBuffer;
>+import java.util.Arrays;
> import java.util.concurrent.TimeUnit;
>+import javax.annotation.Nullable;
> import org.webrtc.Logging;
> import org.webrtc.ThreadUtils;
> 
>@@ -50,12 +52,12 @@ public class WebRtcAudioRecord {
> 
>   private final long nativeAudioRecord;
> 
>-  private WebRtcAudioEffects effects = null;
>+  private @Nullable WebRtcAudioEffects effects = null;
> 
>   private ByteBuffer byteBuffer;
> 
>-  private AudioRecord audioRecord = null;
>-  private AudioRecordThread audioThread = null;
>+  private @Nullable AudioRecord audioRecord = null;
>+  private @Nullable AudioRecordThread audioThread = null;
> 
>   private static volatile boolean microphoneMute = false;
>   private byte[] emptyBytes;
>@@ -72,13 +74,62 @@ public class WebRtcAudioRecord {
>     void onWebRtcAudioRecordError(String errorMessage);
>   }
> 
>-  private static WebRtcAudioRecordErrorCallback errorCallback = null;
>+  private static @Nullable WebRtcAudioRecordErrorCallback errorCallback = null;
> 
>   public static void setErrorCallback(WebRtcAudioRecordErrorCallback errorCallback) {
>     Logging.d(TAG, "Set error callback");
>     WebRtcAudioRecord.errorCallback = errorCallback;
>   }
> 
>+  /**
>+   * Contains audio sample information. Object is passed using {@link
>+   * WebRtcAudioRecord.WebRtcAudioRecordSamplesReadyCallback}
>+   */
>+  public static class AudioSamples {
>+    /** See {@link AudioRecord#getAudioFormat()} */
>+    private final int audioFormat;
>+    /** See {@link AudioRecord#getChannelCount()} */
>+    private final int channelCount;
>+    /** See {@link AudioRecord#getSampleRate()} */
>+    private final int sampleRate;
>+
>+    private final byte[] data;
>+
>+    private AudioSamples(AudioRecord audioRecord, byte[] data) {
>+      this.audioFormat = audioRecord.getAudioFormat();
>+      this.channelCount = audioRecord.getChannelCount();
>+      this.sampleRate = audioRecord.getSampleRate();
>+      this.data = data;
>+    }
>+
>+    public int getAudioFormat() {
>+      return audioFormat;
>+    }
>+
>+    public int getChannelCount() {
>+      return channelCount;
>+    }
>+
>+    public int getSampleRate() {
>+      return sampleRate;
>+    }
>+
>+    public byte[] getData() {
>+      return data;
>+    }
>+  }
>+
>+  /** Called when new audio samples are ready. This should only be set for debug purposes */
>+  public static interface WebRtcAudioRecordSamplesReadyCallback {
>+    void onWebRtcAudioRecordSamplesReady(AudioSamples samples);
>+  }
>+
>+  private static @Nullable WebRtcAudioRecordSamplesReadyCallback audioSamplesReadyCallback = null;
>+
>+  public static void setOnAudioSamplesReady(WebRtcAudioRecordSamplesReadyCallback callback) {
>+    audioSamplesReadyCallback = callback;
>+  }
>+
>   /**
>    * Audio thread which keeps calling ByteBuffer.read() waiting for audio
>    * to be recorded. Feeds recorded data to the native counterpart as a
>@@ -92,6 +143,8 @@ public class WebRtcAudioRecord {
>       super(name);
>     }
> 
>+    // TODO(titovartem) make correct fix during webrtc:9175
>+    @SuppressWarnings("ByteBufferBackingArray")
>     @Override
>     public void run() {
>       Process.setThreadPriority(Process.THREAD_PRIORITY_URGENT_AUDIO);
>@@ -112,6 +165,13 @@ public class WebRtcAudioRecord {
>           if (keepAlive) {
>             nativeDataIsRecorded(bytesRead, nativeAudioRecord);
>           }
>+          if (audioSamplesReadyCallback != null) {
>+            // Copy the entire byte buffer array.  Assume that the start of the byteBuffer is
>+            // at index 0.
>+            byte[] data = Arrays.copyOf(byteBuffer.array(), byteBuffer.capacity());
>+            audioSamplesReadyCallback.onWebRtcAudioRecordSamplesReady(
>+                new AudioSamples(audioRecord, data));
>+          }
>         } else {
>           String errorMessage = "AudioRecord.read failed: " + bytesRead;
>           Logging.e(TAG, errorMessage);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioTrack.java b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioTrack.java
>index 3c8c9ebb291..ee2c76c130f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioTrack.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioTrack.java
>@@ -20,6 +20,7 @@ import android.media.AudioTrack;
> import android.os.Process;
> import java.lang.Thread;
> import java.nio.ByteBuffer;
>+import javax.annotation.Nullable;
> import org.webrtc.ContextUtils;
> import org.webrtc.Logging;
> import org.webrtc.ThreadUtils;
>@@ -79,8 +80,8 @@ public class WebRtcAudioTrack {
> 
>   private ByteBuffer byteBuffer;
> 
>-  private AudioTrack audioTrack = null;
>-  private AudioTrackThread audioThread = null;
>+  private @Nullable AudioTrack audioTrack = null;
>+  private @Nullable AudioTrackThread audioThread = null;
> 
>   // Samples to be played are replaced by zeros if |speakerMute| is set to true.
>   // Can be used to ensure that the speaker is fully muted.
>@@ -107,8 +108,8 @@ public class WebRtcAudioTrack {
>     void onWebRtcAudioTrackError(String errorMessage);
>   }
> 
>-  private static WebRtcAudioTrackErrorCallback errorCallbackOld = null;
>-  private static ErrorCallback errorCallback = null;
>+  private static @Nullable WebRtcAudioTrackErrorCallback errorCallbackOld = null;
>+  private static @Nullable ErrorCallback errorCallback = null;
> 
>   @Deprecated
>   public static void setErrorCallback(WebRtcAudioTrackErrorCallback errorCallback) {
>@@ -497,7 +498,7 @@ public class WebRtcAudioTrack {
>   private void reportWebRtcAudioTrackInitError(String errorMessage) {
>     Logging.e(TAG, "Init playout error: " + errorMessage);
>     WebRtcAudioUtils.logAudioState(TAG);
>-    if (errorCallback != null) {
>+    if (errorCallbackOld != null) {
>       errorCallbackOld.onWebRtcAudioTrackInitError(errorMessage);
>     }
>     if (errorCallback != null) {
>@@ -509,7 +510,7 @@ public class WebRtcAudioTrack {
>       AudioTrackStartErrorCode errorCode, String errorMessage) {
>     Logging.e(TAG, "Start playout error: "  + errorCode + ". " + errorMessage);
>     WebRtcAudioUtils.logAudioState(TAG);
>-    if (errorCallback != null) {
>+    if (errorCallbackOld != null) {
>       errorCallbackOld.onWebRtcAudioTrackStartError(errorMessage);
>     }
>     if (errorCallback != null) {
>@@ -520,7 +521,7 @@ public class WebRtcAudioTrack {
>   private void reportWebRtcAudioTrackError(String errorMessage) {
>     Logging.e(TAG, "Run-time playback error: " + errorMessage);
>     WebRtcAudioUtils.logAudioState(TAG);
>-    if (errorCallback != null) {
>+    if (errorCallbackOld != null) {
>       errorCallbackOld.onWebRtcAudioTrackError(errorMessage);
>     }
>     if (errorCallback != null) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioUtils.java b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioUtils.java
>index f31e497d386..097f3bee335 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioUtils.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/WebRtcAudioUtils.java
>@@ -191,6 +191,16 @@ public final class WebRtcAudioUtils {
>     return Build.VERSION.SDK_INT >= Build.VERSION_CODES.N;
>   }
> 
>+  public static boolean runningOnOreoOrHigher() {
>+    // API Level 26.
>+    return Build.VERSION.SDK_INT >= Build.VERSION_CODES.O;
>+  }
>+
>+  public static boolean runningOnOreoMR1OrHigher() {
>+    // API Level 27.
>+    return Build.VERSION.SDK_INT >= Build.VERSION_CODES.O_MR1;
>+  }
>+
>   // Helper method for building a string of thread information.
>   public static String getThreadInfo() {
>     return "@[name=" + Thread.currentThread().getName() + ", id=" + Thread.currentThread().getId()
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/opensles_common.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/opensles_common.h
>index 90988eef5ad..438c5220722 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/opensles_common.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/opensles_common.h
>@@ -11,8 +11,8 @@
> #ifndef MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_COMMON_H_
> #define MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_COMMON_H_
> 
>-#include <stddef.h>
> #include <SLES/OpenSLES.h>
>+#include <stddef.h>
> 
> #include "rtc_base/checks.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/opensles_player.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/opensles_player.cc
>index 2732f99b824..6f86ba5794e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/opensles_player.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/opensles_player.cc
>@@ -12,6 +12,7 @@
> 
> #include <android/log.h>
> 
>+#include "absl/memory/memory.h"
> #include "api/array_view.h"
> #include "modules/audio_device/android/audio_common.h"
> #include "modules/audio_device/android/audio_manager.h"
>@@ -81,9 +82,7 @@ int OpenSLESPlayer::Init() {
>   ALOGD("Init[tid=%d]", rtc::CurrentThreadId());
>   RTC_DCHECK(thread_checker_.CalledOnValidThread());
>   if (audio_parameters_.channels() == 2) {
>-    // TODO(henrika): FineAudioBuffer needs more work to support stereo.
>-    ALOGE("OpenSLESPlayer does not support stereo");
>-    return -1;
>+    ALOGW("Stereo mode is enabled");
>   }
>   return 0;
> }
>@@ -212,16 +211,15 @@ void OpenSLESPlayer::AllocateDataBuffers() {
>   // recommended to construct audio buffers so that they contain an exact
>   // multiple of this number. If so, callbacks will occur at regular intervals,
>   // which reduces jitter.
>-  const size_t buffer_size_in_bytes = audio_parameters_.GetBytesPerBuffer();
>-  ALOGD("native buffer size: %" PRIuS, buffer_size_in_bytes);
>+  const size_t buffer_size_in_samples =
>+      audio_parameters_.frames_per_buffer() * audio_parameters_.channels();
>+  ALOGD("native buffer size: %" PRIuS, buffer_size_in_samples);
>   ALOGD("native buffer size in ms: %.2f",
>         audio_parameters_.GetBufferSizeInMilliseconds());
>-  fine_audio_buffer_.reset(new FineAudioBuffer(audio_device_buffer_,
>-                                               audio_parameters_.sample_rate(),
>-                                               2 * buffer_size_in_bytes));
>+  fine_audio_buffer_ = absl::make_unique<FineAudioBuffer>(audio_device_buffer_);
>   // Allocated memory for audio buffers.
>   for (int i = 0; i < kNumOfOpenSLESBuffers; ++i) {
>-    audio_buffers_[i].reset(new SLint8[buffer_size_in_bytes]);
>+    audio_buffers_[i].reset(new SLint16[buffer_size_in_samples]);
>   }
> }
> 
>@@ -393,24 +391,29 @@ void OpenSLESPlayer::EnqueuePlayoutData(bool silence) {
>     ALOGW("Bad OpenSL ES playout timing, dT=%u [ms]", diff);
>   }
>   last_play_time_ = current_time;
>-  SLint8* audio_ptr = audio_buffers_[buffer_index_].get();
>+  SLint8* audio_ptr8 =
>+      reinterpret_cast<SLint8*>(audio_buffers_[buffer_index_].get());
>   if (silence) {
>     RTC_DCHECK(thread_checker_.CalledOnValidThread());
>-    // Avoid aquiring real audio data from WebRTC and fill the buffer with
>+    // Avoid acquiring real audio data from WebRTC and fill the buffer with
>     // zeros instead. Used to prime the buffer with silence and to avoid asking
>     // for audio data from two different threads.
>-    memset(audio_ptr, 0, audio_parameters_.GetBytesPerBuffer());
>+    memset(audio_ptr8, 0, audio_parameters_.GetBytesPerBuffer());
>   } else {
>     RTC_DCHECK(thread_checker_opensles_.CalledOnValidThread());
>     // Read audio data from the WebRTC source using the FineAudioBuffer object
>     // to adjust for differences in buffer size between WebRTC (10ms) and native
>-    // OpenSL ES.
>-    fine_audio_buffer_->GetPlayoutData(rtc::ArrayView<SLint8>(
>-        audio_ptr, audio_parameters_.GetBytesPerBuffer()));
>+    // OpenSL ES. Use hardcoded delay estimate since OpenSL ES does not support
>+    // delay estimation.
>+    fine_audio_buffer_->GetPlayoutData(
>+        rtc::ArrayView<int16_t>(audio_buffers_[buffer_index_].get(),
>+                                audio_parameters_.frames_per_buffer() *
>+                                    audio_parameters_.channels()),
>+        25);
>   }
>   // Enqueue the decoded audio buffer for playback.
>   SLresult err = (*simple_buffer_queue_)
>-                     ->Enqueue(simple_buffer_queue_, audio_ptr,
>+                     ->Enqueue(simple_buffer_queue_, audio_ptr8,
>                                audio_parameters_.GetBytesPerBuffer());
>   if (SL_RESULT_SUCCESS != err) {
>     ALOGE("Enqueue failed: %d", err);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/opensles_player.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/opensles_player.h
>index 167c3563301..20107585a69 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/opensles_player.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/opensles_player.h
>@@ -143,9 +143,8 @@ class OpenSLESPlayer {
>   SLDataFormat_PCM pcm_format_;
> 
>   // Queue of audio buffers to be used by the player object for rendering
>-  // audio. They will be used in a Round-robin way and the size of each buffer
>-  // is given by FineAudioBuffer::RequiredBufferSizeBytes().
>-  std::unique_ptr<SLint8[]> audio_buffers_[kNumOfOpenSLESBuffers];
>+  // audio.
>+  std::unique_ptr<SLint16[]> audio_buffers_[kNumOfOpenSLESBuffers];
> 
>   // FineAudioBuffer takes an AudioDeviceBuffer which delivers audio data
>   // in chunks of 10ms. It then allows for this data to be pulled in
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/opensles_recorder.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/opensles_recorder.cc
>index 919486ba708..6bc1bf57db7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/opensles_recorder.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/opensles_recorder.cc
>@@ -12,6 +12,7 @@
> 
> #include <android/log.h>
> 
>+#include "absl/memory/memory.h"
> #include "api/array_view.h"
> #include "modules/audio_device/android/audio_common.h"
> #include "modules/audio_device/android/audio_manager.h"
>@@ -78,9 +79,7 @@ int OpenSLESRecorder::Init() {
>   ALOGD("Init[tid=%d]", rtc::CurrentThreadId());
>   RTC_DCHECK(thread_checker_.CalledOnValidThread());
>   if (audio_parameters_.channels() == 2) {
>-    // TODO(henrika): FineAudioBuffer needs more work to support stereo.
>-    ALOGE("OpenSLESRecorder does not support stereo");
>-    return -1;
>+    ALOGD("Stereo mode is enabled");
>   }
>   return 0;
> }
>@@ -342,14 +341,13 @@ void OpenSLESRecorder::AllocateDataBuffers() {
>         audio_parameters_.GetBytesPerBuffer());
>   ALOGD("native sample rate: %d", audio_parameters_.sample_rate());
>   RTC_DCHECK(audio_device_buffer_);
>-  fine_audio_buffer_.reset(
>-      new FineAudioBuffer(audio_device_buffer_, audio_parameters_.sample_rate(),
>-                          2 * audio_parameters_.GetBytesPerBuffer()));
>+  fine_audio_buffer_ = absl::make_unique<FineAudioBuffer>(audio_device_buffer_);
>   // Allocate queue of audio buffers that stores recorded audio samples.
>-  const int data_size_bytes = audio_parameters_.GetBytesPerBuffer();
>-  audio_buffers_.reset(new std::unique_ptr<SLint8[]>[kNumOfOpenSLESBuffers]);
>+  const int buffer_size_samples =
>+      audio_parameters_.frames_per_buffer() * audio_parameters_.channels();
>+  audio_buffers_.reset(new std::unique_ptr<SLint16[]>[kNumOfOpenSLESBuffers]);
>   for (int i = 0; i < kNumOfOpenSLESBuffers; ++i) {
>-    audio_buffers_[i].reset(new SLint8[data_size_bytes]);
>+    audio_buffers_[i].reset(new SLint16[buffer_size_samples]);
>   }
> }
> 
>@@ -374,12 +372,11 @@ void OpenSLESRecorder::ReadBufferQueue() {
>   // since there is no support to turn off built-in EC in combination with
>   // OpenSL ES anyhow. Hence, as is, the WebRTC based AEC (which would use
>   // these estimates) will never be active.
>-  const size_t size_in_bytes =
>-      static_cast<size_t>(audio_parameters_.GetBytesPerBuffer());
>-  const int8_t* data =
>-      static_cast<const int8_t*>(audio_buffers_[buffer_index_].get());
>   fine_audio_buffer_->DeliverRecordedData(
>-      rtc::ArrayView<const int8_t>(data, size_in_bytes), 25, 25);
>+      rtc::ArrayView<const int16_t>(
>+          audio_buffers_[buffer_index_].get(),
>+          audio_parameters_.frames_per_buffer() * audio_parameters_.channels()),
>+      25);
>   // Enqueue the utilized audio buffer and use if for recording again.
>   EnqueueAudioBuffer();
> }
>@@ -387,8 +384,10 @@ void OpenSLESRecorder::ReadBufferQueue() {
> bool OpenSLESRecorder::EnqueueAudioBuffer() {
>   SLresult err =
>       (*simple_buffer_queue_)
>-          ->Enqueue(simple_buffer_queue_, audio_buffers_[buffer_index_].get(),
>-                    audio_parameters_.GetBytesPerBuffer());
>+          ->Enqueue(
>+              simple_buffer_queue_,
>+              reinterpret_cast<SLint8*>(audio_buffers_[buffer_index_].get()),
>+              audio_parameters_.GetBytesPerBuffer());
>   if (SL_RESULT_SUCCESS != err) {
>     ALOGE("Enqueue failed: %s", GetSLErrorString(err));
>     return false;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/opensles_recorder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/opensles_recorder.h
>index 478c0340af3..ee1ede51d58 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/opensles_recorder.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/opensles_recorder.h
>@@ -176,9 +176,9 @@ class OpenSLESRecorder {
> 
>   // Queue of audio buffers to be used by the recorder object for capturing
>   // audio. They will be used in a Round-robin way and the size of each buffer
>-  // is given by AudioParameters::GetBytesPerBuffer(), i.e., it corresponds to
>+  // is given by AudioParameters::frames_per_buffer(), i.e., it corresponds to
>   // the native OpenSL ES buffer size.
>-  std::unique_ptr<std::unique_ptr<SLint8[]>[]> audio_buffers_;
>+  std::unique_ptr<std::unique_ptr<SLint16[]>[]> audio_buffers_;
> 
>   // Keeps track of active audio buffer 'n' in the audio_buffers_[n] queue.
>   // Example (kNumOfOpenSLESBuffers = 2): counts 0, 1, 0, 1, ...
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_buffer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_buffer.cc
>index 892a3af0628..d872da5fc92 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_buffer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_buffer.cc
>@@ -234,14 +234,17 @@ int32_t AudioDeviceBuffer::SetTypingStatus(bool typing_status) {
>   return 0;
> }
> 
>-void AudioDeviceBuffer::NativeAudioInterrupted() {
>+void AudioDeviceBuffer::NativeAudioPlayoutInterrupted() {
>   RTC_DCHECK(main_thread_checker_.CalledOnValidThread());
>   playout_thread_checker_.DetachFromThread();
>+}
>+
>+void AudioDeviceBuffer::NativeAudioRecordingInterrupted() {
>+  RTC_DCHECK(main_thread_checker_.CalledOnValidThread());
>   recording_thread_checker_.DetachFromThread();
> }
> 
>-void AudioDeviceBuffer::SetVQEData(int play_delay_ms,
>-                                   int rec_delay_ms) {
>+void AudioDeviceBuffer::SetVQEData(int play_delay_ms, int rec_delay_ms) {
>   RTC_DCHECK_RUN_ON(&recording_thread_checker_);
>   play_delay_ms_ = play_delay_ms;
>   rec_delay_ms_ = rec_delay_ms;
>@@ -352,9 +355,17 @@ int32_t AudioDeviceBuffer::GetPlayoutData(void* audio_buffer) {
>   const double phase_increment =
>       k2Pi * 440.0 / static_cast<double>(play_sample_rate_);
>   int16_t* destination_r = reinterpret_cast<int16_t*>(audio_buffer);
>-  for (size_t i = 0; i < play_buffer_.size(); ++i) {
>-    destination_r[i] = static_cast<int16_t>((sin(phase_) * (1 << 14)));
>-    phase_ += phase_increment;
>+  if (play_channels_ == 1) {
>+    for (size_t i = 0; i < play_buffer_.size(); ++i) {
>+      destination_r[i] = static_cast<int16_t>((sin(phase_) * (1 << 14)));
>+      phase_ += phase_increment;
>+    }
>+  } else if (play_channels_ == 2) {
>+    for (size_t i = 0; i < play_buffer_.size() / 2; ++i) {
>+      destination_r[2 * i] = destination_r[2 * i + 1] =
>+          static_cast<int16_t>((sin(phase_) * (1 << 14)));
>+      phase_ += phase_increment;
>+    }
>   }
> #else
>   memcpy(audio_buffer, play_buffer_.data(),
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_buffer.h
>index c4ef95b3211..8dd95501947 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_buffer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_buffer.h
>@@ -17,8 +17,6 @@
> #include "rtc_base/task_queue.h"
> #include "rtc_base/thread_annotations.h"
> #include "rtc_base/thread_checker.h"
>-#include "system_wrappers/include/file_wrapper.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -104,11 +102,12 @@ class AudioDeviceBuffer {
> 
>   int32_t SetTypingStatus(bool typing_status);
> 
>-  // Called on iOS where the native audio layer can be interrupted by other
>-  // audio applications. This method can then be used to reset internal
>-  // states and detach thread checkers to allow for a new audio session where
>-  // native callbacks may come from a new set of I/O threads.
>-  void NativeAudioInterrupted();
>+  // Called on iOS and Android where the native audio layer can be interrupted
>+  // by other audio applications. These methods can then be used to reset
>+  // internal states and detach thread checkers to allow for new audio sessions
>+  // where native callbacks may come from a new set of I/O threads.
>+  void NativeAudioPlayoutInterrupted();
>+  void NativeAudioRecordingInterrupted();
> 
>  private:
>   // Starts/stops periodic logging of audio stats.
>@@ -182,46 +181,46 @@ class AudioDeviceBuffer {
>   // Keeps track of if playout/recording are active or not. A combination
>   // of these states are used to determine when to start and stop the timer.
>   // Only used on the creating thread and not used to control any media flow.
>-  bool playing_ RTC_ACCESS_ON(main_thread_checker_);
>-  bool recording_ RTC_ACCESS_ON(main_thread_checker_);
>+  bool playing_ RTC_GUARDED_BY(main_thread_checker_);
>+  bool recording_ RTC_GUARDED_BY(main_thread_checker_);
> 
>   // Buffer used for audio samples to be played out. Size can be changed
>   // dynamically. The 16-bit samples are interleaved, hence the size is
>   // proportional to the number of channels.
>-  rtc::BufferT<int16_t> play_buffer_ RTC_ACCESS_ON(playout_thread_checker_);
>+  rtc::BufferT<int16_t> play_buffer_ RTC_GUARDED_BY(playout_thread_checker_);
> 
>   // Byte buffer used for recorded audio samples. Size can be changed
>   // dynamically.
>-  rtc::BufferT<int16_t> rec_buffer_ RTC_ACCESS_ON(recording_thread_checker_);
>+  rtc::BufferT<int16_t> rec_buffer_ RTC_GUARDED_BY(recording_thread_checker_);
> 
>   // Contains true of a key-press has been detected.
>-  bool typing_status_ RTC_ACCESS_ON(recording_thread_checker_);
>+  bool typing_status_ RTC_GUARDED_BY(recording_thread_checker_);
> 
>   // Delay values used by the AEC.
>-  int play_delay_ms_ RTC_ACCESS_ON(recording_thread_checker_);
>-  int rec_delay_ms_ RTC_ACCESS_ON(recording_thread_checker_);
>+  int play_delay_ms_ RTC_GUARDED_BY(recording_thread_checker_);
>+  int rec_delay_ms_ RTC_GUARDED_BY(recording_thread_checker_);
> 
>   // Counts number of times LogStats() has been called.
>-  size_t num_stat_reports_ RTC_ACCESS_ON(task_queue_);
>+  size_t num_stat_reports_ RTC_GUARDED_BY(task_queue_);
> 
>   // Time stamp of last timer task (drives logging).
>-  int64_t last_timer_task_time_ RTC_ACCESS_ON(task_queue_);
>+  int64_t last_timer_task_time_ RTC_GUARDED_BY(task_queue_);
> 
>   // Counts number of audio callbacks modulo 50 to create a signal when
>   // a new storage of audio stats shall be done.
>-  int16_t rec_stat_count_ RTC_ACCESS_ON(recording_thread_checker_);
>-  int16_t play_stat_count_ RTC_ACCESS_ON(playout_thread_checker_);
>+  int16_t rec_stat_count_ RTC_GUARDED_BY(recording_thread_checker_);
>+  int16_t play_stat_count_ RTC_GUARDED_BY(playout_thread_checker_);
> 
>   // Time stamps of when playout and recording starts.
>-  int64_t play_start_time_ RTC_ACCESS_ON(main_thread_checker_);
>-  int64_t rec_start_time_ RTC_ACCESS_ON(main_thread_checker_);
>+  int64_t play_start_time_ RTC_GUARDED_BY(main_thread_checker_);
>+  int64_t rec_start_time_ RTC_GUARDED_BY(main_thread_checker_);
> 
>   // Contains counters for playout and recording statistics.
>   Stats stats_ RTC_GUARDED_BY(lock_);
> 
>   // Stores current stats at each timer task. Used to calculate differences
>   // between two successive timer events.
>-  Stats last_stats_ RTC_ACCESS_ON(task_queue_);
>+  Stats last_stats_ RTC_GUARDED_BY(task_queue_);
> 
>   // Set to true at construction and modified to false as soon as one audio-
>   // level estimate larger than zero is detected.
>@@ -231,12 +230,12 @@ class AudioDeviceBuffer {
>   // StartPeriodicLogging() and set to false by StopPeriodicLogging().
>   // Setting this member to false prevents (possiby invalid) log messages from
>   // being printed in the LogStats() task.
>-  bool log_stats_ RTC_ACCESS_ON(task_queue_);
>+  bool log_stats_ RTC_GUARDED_BY(task_queue_);
> 
> // Should *never* be defined in production builds. Only used for testing.
> // When defined, the output signal will be replaced by a sinus tone at 440Hz.
> #ifdef AUDIO_DEVICE_PLAYS_SINUS_TONE
>-  double phase_ RTC_ACCESS_ON(playout_thread_checker_);
>+  double phase_ RTC_GUARDED_BY(playout_thread_checker_);
> #endif
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_config.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_config.h
>index fd12c8bedd3..fa51747b67a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_config.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_config.h
>@@ -28,4 +28,3 @@ enum { GET_MIC_VOLUME_INTERVAL_MS = 1000 };
> #endif
> 
> #endif  // AUDIO_DEVICE_AUDIO_DEVICE_CONFIG_H_
>-
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_data_observer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_data_observer.cc
>index 5f7259d13f3..723d1c8bbdc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_data_observer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_data_observer.cc
>@@ -71,6 +71,11 @@ class ADMWrapper : public AudioDeviceModule, public AudioTransport {
>                            int64_t* elapsed_time_ms,
>                            int64_t* ntp_time_ms) override {
>     int32_t res = 0;
>+    // Set out parameters to safe values to be sure not to return corrupted
>+    // data.
>+    nSamplesOut = 0;
>+    *elapsed_time_ms = -1;
>+    *ntp_time_ms = -1;
>     // Request data from audio transport.
>     if (audio_transport_) {
>       res = audio_transport_->NeedMorePlayData(
>@@ -87,15 +92,6 @@ class ADMWrapper : public AudioDeviceModule, public AudioTransport {
>     return res;
>   }
> 
>-  void PushCaptureData(int voe_channel,
>-                       const void* audio_data,
>-                       int bits_per_sample,
>-                       int sample_rate,
>-                       size_t number_of_channels,
>-                       size_t number_of_frames) override {
>-    RTC_NOTREACHED();
>-  }
>-
>   void PullRenderData(int bits_per_sample,
>                       int sample_rate,
>                       size_t number_of_channels,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_generic.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_generic.h
>index e7abcb37ff1..be9c072c412 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_generic.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_generic.h
>@@ -122,8 +122,8 @@ class AudioDeviceGeneric {
>   virtual int32_t EnableBuiltInAGC(bool enable);
>   virtual int32_t EnableBuiltInNS(bool enable);
> 
>-  // iOS only.
>-  // TODO(henrika): add Android support.
>+// iOS only.
>+// TODO(henrika): add Android support.
> #if defined(WEBRTC_IOS)
>   virtual int GetPlayoutAudioParameters(AudioParameters* params) const;
>   virtual int GetRecordAudioParameters(AudioParameters* params) const;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_impl.cc
>index ff651d4c90d..834f4aa9176 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_impl.cc
>@@ -20,10 +20,14 @@
> 
> #if defined(_WIN32)
> #if defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
>-#include "audio_device_core_win.h"
>+#include "modules/audio_device/win/audio_device_core_win.h"
> #endif
> #elif defined(WEBRTC_ANDROID)
> #include <stdlib.h>
>+#if defined(AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO)
>+#include "modules/audio_device/android/aaudio_player.h"
>+#include "modules/audio_device/android/aaudio_recorder.h"
>+#endif
> #include "modules/audio_device/android/audio_device_template.h"
> #include "modules/audio_device/android/audio_manager.h"
> #include "modules/audio_device/android/audio_record_jni.h"
>@@ -32,15 +36,15 @@
> #include "modules/audio_device/android/opensles_recorder.h"
> #elif defined(WEBRTC_LINUX)
> #if defined(LINUX_ALSA)
>-#include "audio_device_alsa_linux.h"
>+#include "modules/audio_device/linux/audio_device_alsa_linux.h"
> #endif
> #if defined(LINUX_PULSE)
>-#include "audio_device_pulse_linux.h"
>+#include "modules/audio_device/linux/audio_device_pulse_linux.h"
> #endif
> #elif defined(WEBRTC_IOS)
>-#include "audio_device_ios.h"
>+#include "modules/audio_device/ios/audio_device_ios.h"
> #elif defined(WEBRTC_MAC)
>-#include "audio_device_mac.h"
>+#include "modules/audio_device/mac/audio_device_mac.h"
> #endif
> #if defined(WEBRTC_DUMMY_FILE_DEVICES)
> #include "modules/audio_device/dummy/file_audio_device_factory.h"
>@@ -52,14 +56,14 @@
>   {                         \
>     if (!initialized_) {    \
>       return -1;            \
>-    };                      \
>+    }                       \
>   }
> 
> #define CHECKinitialized__BOOL() \
>   {                              \
>     if (!initialized_) {         \
>       return false;              \
>-    };                           \
>+    }                            \
>   }
> 
> namespace webrtc {
>@@ -68,6 +72,15 @@ namespace webrtc {
> rtc::scoped_refptr<AudioDeviceModule> AudioDeviceModule::Create(
>     const AudioLayer audio_layer) {
>   RTC_LOG(INFO) << __FUNCTION__;
>+
>+  // The "AudioDeviceModule::kWindowsCoreAudio2" audio layer has its own
>+  // dedicated factory method which should be used instead.
>+  if (audio_layer == AudioDeviceModule::kWindowsCoreAudio2) {
>+    RTC_LOG(LS_ERROR) << "Use the CreateWindowsCoreAudioAudioDeviceModule() "
>+                         "factory method instead for this option.";
>+    return nullptr;
>+  }
>+
>   // Create the generic reference counted (platform independent) implementation.
>   rtc::scoped_refptr<AudioDeviceModuleImpl> audioDevice(
>       new rtc::RefCountedObject<AudioDeviceModuleImpl>(audio_layer));
>@@ -170,8 +183,11 @@ int32_t AudioDeviceModuleImpl::CreatePlatformSpecificObjects() {
>   audio_manager_android_.reset(new AudioManager());
>   // Select best possible combination of audio layers.
>   if (audio_layer == kPlatformDefaultAudio) {
>-    if (audio_manager_android_->IsLowLatencyPlayoutSupported() &&
>-        audio_manager_android_->IsLowLatencyRecordSupported()) {
>+    if (audio_manager_android_->IsAAudioSupported()) {
>+      // Use of AAudio for both playout and recording has highest priority.
>+      audio_layer = kAndroidAAudioAudio;
>+    } else if (audio_manager_android_->IsLowLatencyPlayoutSupported() &&
>+               audio_manager_android_->IsLowLatencyRecordSupported()) {
>       // Use OpenSL ES for both playout and recording.
>       audio_layer = kAndroidOpenSLESAudio;
>     } else if (audio_manager_android_->IsLowLatencyPlayoutSupported() &&
>@@ -201,31 +217,58 @@ int32_t AudioDeviceModuleImpl::CreatePlatformSpecificObjects() {
>     // time support for HW AEC using the AudioRecord Java API.
>     audio_device_.reset(new AudioDeviceTemplate<AudioRecordJni, OpenSLESPlayer>(
>         audio_layer, audio_manager));
>+  } else if (audio_layer == kAndroidAAudioAudio) {
>+#if defined(AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO)
>+    // AAudio based audio for both input and output.
>+    audio_device_.reset(new AudioDeviceTemplate<AAudioRecorder, AAudioPlayer>(
>+        audio_layer, audio_manager));
>+#endif
>+  } else if (audio_layer == kAndroidJavaInputAndAAudioOutputAudio) {
>+#if defined(AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO)
>+    // Java audio for input and AAudio for output audio (i.e. mixed APIs).
>+    audio_device_.reset(new AudioDeviceTemplate<AudioRecordJni, AAudioPlayer>(
>+        audio_layer, audio_manager));
>+#endif
>   } else {
>-    // Invalid audio layer.
>+    RTC_LOG(LS_ERROR) << "The requested audio layer is not supported";
>     audio_device_.reset(nullptr);
>   }
> // END #if defined(WEBRTC_ANDROID)
> 
> // Linux ADM implementation.
>+// Note that, LINUX_ALSA is always defined by default when WEBRTC_LINUX is
>+// defined. LINUX_PULSE depends on the 'rtc_include_pulse_audio' build flag.
>+// TODO(bugs.webrtc.org/9127): improve support and make it more clear that
>+// PulseAudio is the default selection.
> #elif defined(WEBRTC_LINUX)
>+#if !defined(LINUX_PULSE)
>+  // Build flag 'rtc_include_pulse_audio' is set to false. In this mode:
>+  // - kPlatformDefaultAudio => ALSA, and
>+  // - kLinuxAlsaAudio => ALSA, and
>+  // - kLinuxPulseAudio => Invalid selection.
>+  RTC_LOG(WARNING) << "PulseAudio is disabled using build flag.";
>+  if ((audio_layer == kLinuxAlsaAudio) ||
>+      (audio_layer == kPlatformDefaultAudio)) {
>+    audio_device_.reset(new AudioDeviceLinuxALSA());
>+    RTC_LOG(INFO) << "Linux ALSA APIs will be utilized.";
>+  }
>+#else
>+  // Build flag 'rtc_include_pulse_audio' is set to true (default). In this
>+  // mode:
>+  // - kPlatformDefaultAudio => PulseAudio, and
>+  // - kLinuxPulseAudio => PulseAudio, and
>+  // - kLinuxAlsaAudio => ALSA (supported but not default).
>+  RTC_LOG(INFO) << "PulseAudio support is enabled.";
>   if ((audio_layer == kLinuxPulseAudio) ||
>       (audio_layer == kPlatformDefaultAudio)) {
>-#if defined(LINUX_PULSE)
>-    RTC_LOG(INFO) << "Attempting to use Linux PulseAudio APIs...";
>-    // Linux PulseAudio implementation.
>+    // Linux PulseAudio implementation is default.
>     audio_device_.reset(new AudioDeviceLinuxPulse());
>     RTC_LOG(INFO) << "Linux PulseAudio APIs will be utilized";
>-#endif
>-#if defined(LINUX_PULSE)
>-#endif
>   } else if (audio_layer == kLinuxAlsaAudio) {
>-#if defined(LINUX_ALSA)
>-    // Linux ALSA implementation.
>     audio_device_.reset(new AudioDeviceLinuxALSA());
>-    RTC_LOG(INFO) << "Linux ALSA APIs will be utilized.";
>-#endif
>+    RTC_LOG(WARNING) << "Linux ALSA APIs will be utilized.";
>   }
>+#endif  // #if !defined(LINUX_PULSE)
> #endif  // #if defined(WEBRTC_LINUX)
> 
> // iOS ADM implementation.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_name.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_name.cc
>new file mode 100644
>index 00000000000..0a1ed0c9683
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_name.cc
>@@ -0,0 +1,28 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_device/audio_device_name.h"
>+
>+#include <utility>
>+
>+namespace webrtc {
>+
>+const char AudioDeviceName::kDefaultDeviceId[] = "default";
>+const char AudioDeviceName::kDefaultCommunicationsDeviceId[] = "communications";
>+
>+AudioDeviceName::AudioDeviceName(const std::string device_name,
>+                                 const std::string unique_id)
>+    : device_name(std::move(device_name)), unique_id(std::move(unique_id)) {}
>+
>+bool AudioDeviceName::IsValid() {
>+  return !device_name.empty() && !unique_id.empty();
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_name.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_name.h
>new file mode 100644
>index 00000000000..c03384e4091
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_name.h
>@@ -0,0 +1,47 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_DEVICE_AUDIO_DEVICE_NAME_H_
>+#define MODULES_AUDIO_DEVICE_AUDIO_DEVICE_NAME_H_
>+
>+#include <string>
>+#include <vector>
>+
>+namespace webrtc {
>+
>+struct AudioDeviceName {
>+  // Unique ID of the generic default device.
>+  static const char kDefaultDeviceId[];
>+
>+  // Unique ID of the generic default communications device.
>+  static const char kDefaultCommunicationsDeviceId[];
>+
>+  AudioDeviceName() = default;
>+  AudioDeviceName(const std::string device_name, const std::string unique_id);
>+
>+  ~AudioDeviceName() = default;
>+
>+  // Support copy and move.
>+  AudioDeviceName(const AudioDeviceName& other) = default;
>+  AudioDeviceName(AudioDeviceName&&) = default;
>+  AudioDeviceName& operator=(const AudioDeviceName&) = default;
>+  AudioDeviceName& operator=(AudioDeviceName&&) = default;
>+
>+  bool IsValid();
>+
>+  std::string device_name;  // Friendly name of the device.
>+  std::string unique_id;    // Unique identifier for the device.
>+};
>+
>+typedef std::vector<AudioDeviceName> AudioDeviceNames;
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_DEVICE_AUDIO_DEVICE_NAME_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_unittest.cc
>index c309ca6819a..16aef4d8b61 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/audio_device_unittest.cc
>@@ -10,10 +10,11 @@
> 
> #include <algorithm>
> #include <cstring>
>+#include <memory>
> #include <numeric>
> 
>+#include "absl/types/optional.h"
> #include "api/array_view.h"
>-#include "api/optional.h"
> #include "modules/audio_device/audio_device_impl.h"
> #include "modules/audio_device/include/audio_device.h"
> #include "modules/audio_device/include/mock_audio_transport.h"
>@@ -29,6 +30,10 @@
> #include "rtc_base/timeutils.h"
> #include "test/gmock.h"
> #include "test/gtest.h"
>+#ifdef WEBRTC_WIN
>+#include "modules/audio_device/include/audio_device_factory.h"
>+#include "modules/audio_device/win/core_audio_utility_win.h"
>+#endif
> 
> using ::testing::_;
> using ::testing::AtLeast;
>@@ -94,8 +99,8 @@ enum class TransportType {
> // measurements.
> class AudioStream {
>  public:
>-  virtual void Write(rtc::ArrayView<const int16_t> source, size_t channels) = 0;
>-  virtual void Read(rtc::ArrayView<int16_t> destination, size_t channels) = 0;
>+  virtual void Write(rtc::ArrayView<const int16_t> source) = 0;
>+  virtual void Read(rtc::ArrayView<int16_t> destination) = 0;
> 
>   virtual ~AudioStream() = default;
> };
>@@ -122,8 +127,7 @@ int IndexToMilliseconds(size_t index, size_t frames_per_10ms_buffer) {
> // change over time and that both sides will use the same size.
> class FifoAudioStream : public AudioStream {
>  public:
>-  void Write(rtc::ArrayView<const int16_t> source, size_t channels) override {
>-    EXPECT_EQ(channels, 1u);
>+  void Write(rtc::ArrayView<const int16_t> source) override {
>     RTC_DCHECK_RUNS_SERIALIZED(&race_checker_);
>     const size_t size = [&] {
>       rtc::CritScope lock(&lock_);
>@@ -140,8 +144,7 @@ class FifoAudioStream : public AudioStream {
>     written_elements_ += size;
>   }
> 
>-  void Read(rtc::ArrayView<int16_t> destination, size_t channels) override {
>-    EXPECT_EQ(channels, 1u);
>+  void Read(rtc::ArrayView<int16_t> destination) override {
>     rtc::CritScope lock(&lock_);
>     if (fifo_.empty()) {
>       std::fill(destination.begin(), destination.end(), 0);
>@@ -191,9 +194,8 @@ class LatencyAudioStream : public AudioStream {
>   }
> 
>   // Insert periodic impulses in first two samples of |destination|.
>-  void Read(rtc::ArrayView<int16_t> destination, size_t channels) override {
>+  void Read(rtc::ArrayView<int16_t> destination) override {
>     RTC_DCHECK_RUN_ON(&read_thread_checker_);
>-    EXPECT_EQ(channels, 1u);
>     if (read_count_ == 0) {
>       PRINT("[");
>     }
>@@ -214,8 +216,7 @@ class LatencyAudioStream : public AudioStream {
> 
>   // Detect received impulses in |source|, derive time between transmission and
>   // detection and add the calculated delay to list of latencies.
>-  void Write(rtc::ArrayView<const int16_t> source, size_t channels) override {
>-    EXPECT_EQ(channels, 1u);
>+  void Write(rtc::ArrayView<const int16_t> source) override {
>     RTC_DCHECK_RUN_ON(&write_thread_checker_);
>     RTC_DCHECK_RUNS_SERIALIZED(&race_checker_);
>     rtc::CritScope lock(&lock_);
>@@ -293,10 +294,10 @@ class LatencyAudioStream : public AudioStream {
>   rtc::ThreadChecker read_thread_checker_;
>   rtc::ThreadChecker write_thread_checker_;
> 
>-  rtc::Optional<int64_t> pulse_time_ RTC_GUARDED_BY(lock_);
>+  absl::optional<int64_t> pulse_time_ RTC_GUARDED_BY(lock_);
>   std::vector<int> latencies_ RTC_GUARDED_BY(race_checker_);
>-  size_t read_count_ RTC_ACCESS_ON(read_thread_checker_) = 0;
>-  size_t write_count_ RTC_ACCESS_ON(write_thread_checker_) = 0;
>+  size_t read_count_ RTC_GUARDED_BY(read_thread_checker_) = 0;
>+  size_t write_count_ RTC_GUARDED_BY(write_thread_checker_) = 0;
> };
> 
> // Mocks the AudioTransport object and proxies actions for the two callbacks
>@@ -360,8 +361,7 @@ class MockAudioTransport : public test::MockAudioTransport {
>     if (audio_stream_) {
>       audio_stream_->Write(
>           rtc::MakeArrayView(static_cast<const int16_t*>(audio_buffer),
>-                             samples_per_channel * channels),
>-          channels);
>+                             samples_per_channel * channels));
>     }
>     // Signal the event after given amount of callbacks.
>     if (ReceivedEnoughCallbacks()) {
>@@ -375,7 +375,7 @@ class MockAudioTransport : public test::MockAudioTransport {
>                                const size_t channels,
>                                const uint32_t sample_rate,
>                                void* audio_buffer,
>-                               size_t& samples_per_channel_out,
>+                               size_t& samples_out,
>                                int64_t* elapsed_time_ms,
>                                int64_t* ntp_time_ms) {
>     EXPECT_TRUE(play_mode()) << "No test is expecting these callbacks.";
>@@ -395,13 +395,11 @@ class MockAudioTransport : public test::MockAudioTransport {
>                 playout_parameters_.frames_per_10ms_buffer());
>     }
>     play_count_++;
>-    samples_per_channel_out = samples_per_channel;
>+    samples_out = samples_per_channel * channels;
>     // Read audio data from audio stream object if one has been injected.
>     if (audio_stream_) {
>-      audio_stream_->Read(
>-          rtc::MakeArrayView(static_cast<int16_t*>(audio_buffer),
>-                             samples_per_channel * channels),
>-          channels);
>+      audio_stream_->Read(rtc::MakeArrayView(
>+          static_cast<int16_t*>(audio_buffer), samples_per_channel * channels));
>     } else {
>       // Fill the audio buffer with zeros to avoid disturbing audio.
>       const size_t num_bytes = samples_per_channel * bytes_per_frame;
>@@ -452,17 +450,17 @@ class MockAudioTransport : public test::MockAudioTransport {
> };
> 
> // AudioDeviceTest test fixture.
>-class AudioDeviceTest : public ::testing::Test {
>+class AudioDeviceTest
>+    : public ::testing::TestWithParam<webrtc::AudioDeviceModule::AudioLayer> {
>  protected:
>-  AudioDeviceTest() : event_(false, false) {
>+  AudioDeviceTest() : audio_layer_(GetParam()), event_(false, false) {
> #if !defined(ADDRESS_SANITIZER) && !defined(MEMORY_SANITIZER) && \
>     !defined(WEBRTC_DUMMY_AUDIO_BUILD)
>     rtc::LogMessage::LogToDebug(rtc::LS_INFO);
>     // Add extra logging fields here if needed for debugging.
>-    // rtc::LogMessage::LogTimestamps();
>-    // rtc::LogMessage::LogThreads();
>-    audio_device_ =
>-        AudioDeviceModule::Create(AudioDeviceModule::kPlatformDefaultAudio);
>+    rtc::LogMessage::LogTimestamps();
>+    rtc::LogMessage::LogThreads();
>+    audio_device_ = CreateAudioDevice();
>     EXPECT_NE(audio_device_.get(), nullptr);
>     AudioDeviceModule::AudioLayer audio_layer;
>     int got_platform_audio_layer =
>@@ -513,6 +511,32 @@ class AudioDeviceTest : public ::testing::Test {
>     return audio_device_;
>   }
> 
>+  rtc::scoped_refptr<AudioDeviceModule> CreateAudioDevice() {
>+    // Use the default factory for kPlatformDefaultAudio and a special factory
>+    // CreateWindowsCoreAudioAudioDeviceModule() for kWindowsCoreAudio2.
>+    // The value of |audio_layer_| is set at construction by GetParam() and two
>+    // different layers are tested on Windows only.
>+    if (audio_layer_ == AudioDeviceModule::kPlatformDefaultAudio) {
>+      return AudioDeviceModule::Create(audio_layer_);
>+    } else if (audio_layer_ == AudioDeviceModule::kWindowsCoreAudio2) {
>+#ifdef WEBRTC_WIN
>+      // We must initialize the COM library on a thread before we calling any of
>+      // the library functions. All COM functions in the ADM will return
>+      // CO_E_NOTINITIALIZED otherwise.
>+      com_initializer_ = absl::make_unique<webrtc_win::ScopedCOMInitializer>(
>+          webrtc_win::ScopedCOMInitializer::kMTA);
>+      EXPECT_TRUE(com_initializer_->Succeeded());
>+      EXPECT_TRUE(webrtc_win::core_audio_utility::IsSupported());
>+      EXPECT_TRUE(webrtc_win::core_audio_utility::IsMMCSSSupported());
>+      return CreateWindowsCoreAudioAudioDeviceModule();
>+#else
>+      return nullptr;
>+#endif
>+    } else {
>+      return nullptr;
>+    }
>+  }
>+
>   void StartPlayout() {
>     EXPECT_FALSE(audio_device()->Playing());
>     EXPECT_EQ(0, audio_device()->InitPlayout());
>@@ -541,17 +565,70 @@ class AudioDeviceTest : public ::testing::Test {
>     EXPECT_FALSE(audio_device()->RecordingIsInitialized());
>   }
> 
>+  bool NewWindowsAudioDeviceModuleIsUsed() {
>+#ifdef WEBRTC_WIN
>+    AudioDeviceModule::AudioLayer audio_layer;
>+    EXPECT_EQ(0, audio_device()->ActiveAudioLayer(&audio_layer));
>+    if (audio_layer == AudioDeviceModule::kWindowsCoreAudio2) {
>+      // Default device is always added as first element in the list and the
>+      // default communication device as the second element. Hence, the list
>+      // contains two extra elements in this case.
>+      return true;
>+    }
>+#endif
>+    return false;
>+  }
>+
>  private:
>+#ifdef WEBRTC_WIN
>+  // Windows Core Audio based ADM needs to run on a COM initialized thread.
>+  std::unique_ptr<webrtc_win::ScopedCOMInitializer> com_initializer_;
>+#endif
>+  AudioDeviceModule::AudioLayer audio_layer_;
>   bool requirements_satisfied_ = true;
>   rtc::Event event_;
>   rtc::scoped_refptr<AudioDeviceModule> audio_device_;
>   bool stereo_playout_ = false;
> };
> 
>+// Instead of using the test fixture, verify that the different factory methods
>+// work as intended.
>+TEST(AudioDeviceTestWin, ConstructDestructWithFactory) {
>+  rtc::scoped_refptr<AudioDeviceModule> audio_device;
>+  // The default factory should work for all platforms when a default ADM is
>+  // requested.
>+  audio_device =
>+      AudioDeviceModule::Create(AudioDeviceModule::kPlatformDefaultAudio);
>+  EXPECT_TRUE(audio_device);
>+  audio_device = nullptr;
>+#ifdef WEBRTC_WIN
>+  // For Windows, the old factory method creates an ADM where the platform-
>+  // specific parts are implemented by an AudioDeviceGeneric object. Verify
>+  // that the old factory can't be used in combination with the latest audio
>+  // layer AudioDeviceModule::kWindowsCoreAudio2.
>+  audio_device =
>+      AudioDeviceModule::Create(AudioDeviceModule::kWindowsCoreAudio2);
>+  EXPECT_FALSE(audio_device);
>+  audio_device = nullptr;
>+  // Instead, ensure that the new dedicated factory method called
>+  // CreateWindowsCoreAudioAudioDeviceModule() can be used on Windows and that
>+  // it sets the audio layer to kWindowsCoreAudio2 implicitly. Note that, the
>+  // new ADM for Windows must be created on a COM thread.
>+  webrtc_win::ScopedCOMInitializer com_initializer(
>+      webrtc_win::ScopedCOMInitializer::kMTA);
>+  EXPECT_TRUE(com_initializer.Succeeded());
>+  audio_device = CreateWindowsCoreAudioAudioDeviceModule();
>+  EXPECT_TRUE(audio_device);
>+  AudioDeviceModule::AudioLayer audio_layer;
>+  EXPECT_EQ(0, audio_device->ActiveAudioLayer(&audio_layer));
>+  EXPECT_EQ(audio_layer, AudioDeviceModule::kWindowsCoreAudio2);
>+#endif
>+}
>+
> // Uses the test fixture to create, initialize and destruct the ADM.
>-TEST_F(AudioDeviceTest, ConstructDestruct) {}
>+TEST_P(AudioDeviceTest, ConstructDestructDefault) {}
> 
>-TEST_F(AudioDeviceTest, InitTerminate) {
>+TEST_P(AudioDeviceTest, InitTerminate) {
>   SKIP_TEST_IF_NOT(requirements_satisfied());
>   // Initialization is part of the test fixture.
>   EXPECT_TRUE(audio_device()->Initialized());
>@@ -559,28 +636,105 @@ TEST_F(AudioDeviceTest, InitTerminate) {
>   EXPECT_FALSE(audio_device()->Initialized());
> }
> 
>+// Enumerate all available and active output devices.
>+TEST_P(AudioDeviceTest, PlayoutDeviceNames) {
>+  SKIP_TEST_IF_NOT(requirements_satisfied());
>+  char device_name[kAdmMaxDeviceNameSize];
>+  char unique_id[kAdmMaxGuidSize];
>+  int num_devices = audio_device()->PlayoutDevices();
>+  if (NewWindowsAudioDeviceModuleIsUsed()) {
>+    num_devices += 2;
>+  }
>+  EXPECT_GT(num_devices, 0);
>+  for (int i = 0; i < num_devices; ++i) {
>+    EXPECT_EQ(0, audio_device()->PlayoutDeviceName(i, device_name, unique_id));
>+  }
>+  EXPECT_EQ(-1, audio_device()->PlayoutDeviceName(num_devices, device_name,
>+                                                  unique_id));
>+}
>+
>+// Enumerate all available and active input devices.
>+TEST_P(AudioDeviceTest, RecordingDeviceNames) {
>+  SKIP_TEST_IF_NOT(requirements_satisfied());
>+  char device_name[kAdmMaxDeviceNameSize];
>+  char unique_id[kAdmMaxGuidSize];
>+  int num_devices = audio_device()->RecordingDevices();
>+  if (NewWindowsAudioDeviceModuleIsUsed()) {
>+    num_devices += 2;
>+  }
>+  EXPECT_GT(num_devices, 0);
>+  for (int i = 0; i < num_devices; ++i) {
>+    EXPECT_EQ(0,
>+              audio_device()->RecordingDeviceName(i, device_name, unique_id));
>+  }
>+  EXPECT_EQ(-1, audio_device()->RecordingDeviceName(num_devices, device_name,
>+                                                    unique_id));
>+}
>+
>+// Counts number of active output devices and ensure that all can be selected.
>+TEST_P(AudioDeviceTest, SetPlayoutDevice) {
>+  SKIP_TEST_IF_NOT(requirements_satisfied());
>+  int num_devices = audio_device()->PlayoutDevices();
>+  if (NewWindowsAudioDeviceModuleIsUsed()) {
>+    num_devices += 2;
>+  }
>+  EXPECT_GT(num_devices, 0);
>+  // Verify that all available playout devices can be set (not enabled yet).
>+  for (int i = 0; i < num_devices; ++i) {
>+    EXPECT_EQ(0, audio_device()->SetPlayoutDevice(i));
>+  }
>+  EXPECT_EQ(-1, audio_device()->SetPlayoutDevice(num_devices));
>+#ifdef WEBRTC_WIN
>+  // On Windows, verify the alternative method where the user can select device
>+  // by role.
>+  EXPECT_EQ(
>+      0, audio_device()->SetPlayoutDevice(AudioDeviceModule::kDefaultDevice));
>+  EXPECT_EQ(0, audio_device()->SetPlayoutDevice(
>+                   AudioDeviceModule::kDefaultCommunicationDevice));
>+#endif
>+}
>+
>+// Counts number of active input devices and ensure that all can be selected.
>+TEST_P(AudioDeviceTest, SetRecordingDevice) {
>+  SKIP_TEST_IF_NOT(requirements_satisfied());
>+  int num_devices = audio_device()->RecordingDevices();
>+  if (NewWindowsAudioDeviceModuleIsUsed()) {
>+    num_devices += 2;
>+  }
>+  EXPECT_GT(num_devices, 0);
>+  // Verify that all available recording devices can be set (not enabled yet).
>+  for (int i = 0; i < num_devices; ++i) {
>+    EXPECT_EQ(0, audio_device()->SetRecordingDevice(i));
>+  }
>+  EXPECT_EQ(-1, audio_device()->SetRecordingDevice(num_devices));
>+#ifdef WEBRTC_WIN
>+  // On Windows, verify the alternative method where the user can select device
>+  // by role.
>+  EXPECT_EQ(
>+      0, audio_device()->SetRecordingDevice(AudioDeviceModule::kDefaultDevice));
>+  EXPECT_EQ(0, audio_device()->SetRecordingDevice(
>+                   AudioDeviceModule::kDefaultCommunicationDevice));
>+#endif
>+}
>+
> // Tests Start/Stop playout without any registered audio callback.
>-TEST_F(AudioDeviceTest, StartStopPlayout) {
>+TEST_P(AudioDeviceTest, StartStopPlayout) {
>   SKIP_TEST_IF_NOT(requirements_satisfied());
>   StartPlayout();
>   StopPlayout();
>-  StartPlayout();
>-  StopPlayout();
> }
> 
> // Tests Start/Stop recording without any registered audio callback.
>-TEST_F(AudioDeviceTest, StartStopRecording) {
>+TEST_P(AudioDeviceTest, StartStopRecording) {
>   SKIP_TEST_IF_NOT(requirements_satisfied());
>   StartRecording();
>   StopRecording();
>-  StartRecording();
>-  StopRecording();
> }
> 
> // Tests Init/Stop/Init recording without any registered audio callback.
> // See https://bugs.chromium.org/p/webrtc/issues/detail?id=8041 for details
> // on why this test is useful.
>-TEST_F(AudioDeviceTest, InitStopInitRecording) {
>+TEST_P(AudioDeviceTest, InitStopInitRecording) {
>   SKIP_TEST_IF_NOT(requirements_satisfied());
>   EXPECT_EQ(0, audio_device()->InitRecording());
>   EXPECT_TRUE(audio_device()->RecordingIsInitialized());
>@@ -590,7 +744,7 @@ TEST_F(AudioDeviceTest, InitStopInitRecording) {
> }
> 
> // Tests Init/Stop/Init recording while playout is active.
>-TEST_F(AudioDeviceTest, InitStopInitRecordingWhilePlaying) {
>+TEST_P(AudioDeviceTest, InitStopInitRecordingWhilePlaying) {
>   SKIP_TEST_IF_NOT(requirements_satisfied());
>   StartPlayout();
>   EXPECT_EQ(0, audio_device()->InitRecording());
>@@ -602,7 +756,7 @@ TEST_F(AudioDeviceTest, InitStopInitRecordingWhilePlaying) {
> }
> 
> // Tests Init/Stop/Init playout without any registered audio callback.
>-TEST_F(AudioDeviceTest, InitStopInitPlayout) {
>+TEST_P(AudioDeviceTest, InitStopInitPlayout) {
>   SKIP_TEST_IF_NOT(requirements_satisfied());
>   EXPECT_EQ(0, audio_device()->InitPlayout());
>   EXPECT_TRUE(audio_device()->PlayoutIsInitialized());
>@@ -612,7 +766,7 @@ TEST_F(AudioDeviceTest, InitStopInitPlayout) {
> }
> 
> // Tests Init/Stop/Init playout while recording is active.
>-TEST_F(AudioDeviceTest, InitStopInitPlayoutWhileRecording) {
>+TEST_P(AudioDeviceTest, InitStopInitPlayoutWhileRecording) {
>   SKIP_TEST_IF_NOT(requirements_satisfied());
>   StartRecording();
>   EXPECT_EQ(0, audio_device()->InitPlayout());
>@@ -628,7 +782,7 @@ TEST_F(AudioDeviceTest, InitStopInitPlayoutWhileRecording) {
> // Note that we can't add expectations on audio parameters in EXPECT_CALL
> // since parameter are not provided in the each callback. We therefore test and
> // verify the parameters in the fake audio transport implementation instead.
>-TEST_F(AudioDeviceTest, StartPlayoutVerifyCallbacks) {
>+TEST_P(AudioDeviceTest, StartPlayoutVerifyCallbacks) {
>   SKIP_TEST_IF_NOT(requirements_satisfied());
>   MockAudioTransport mock(TransportType::kPlay);
>   mock.HandleCallbacks(event(), nullptr, kNumCallbacks);
>@@ -642,7 +796,7 @@ TEST_F(AudioDeviceTest, StartPlayoutVerifyCallbacks) {
> 
> // Start recording and verify that the native audio layer starts providing real
> // audio samples using the RecordedDataIsAvailable() callback.
>-TEST_F(AudioDeviceTest, StartRecordingVerifyCallbacks) {
>+TEST_P(AudioDeviceTest, StartRecordingVerifyCallbacks) {
>   SKIP_TEST_IF_NOT(requirements_satisfied());
>   MockAudioTransport mock(TransportType::kRecord);
>   mock.HandleCallbacks(event(), nullptr, kNumCallbacks);
>@@ -657,7 +811,7 @@ TEST_F(AudioDeviceTest, StartRecordingVerifyCallbacks) {
> 
> // Start playout and recording (full-duplex audio) and verify that audio is
> // active in both directions.
>-TEST_F(AudioDeviceTest, StartPlayoutAndRecordingVerifyCallbacks) {
>+TEST_P(AudioDeviceTest, StartPlayoutAndRecordingVerifyCallbacks) {
>   SKIP_TEST_IF_NOT(requirements_satisfied());
>   MockAudioTransport mock(TransportType::kPlayAndRecord);
>   mock.HandleCallbacks(event(), nullptr, kNumCallbacks);
>@@ -686,18 +840,18 @@ TEST_F(AudioDeviceTest, StartPlayoutAndRecordingVerifyCallbacks) {
> // sequence by running in loopback for a few seconds while measuring the size
> // (max and average) of the FIFO. The size of the FIFO is increased by the
> // recording side and decreased by the playout side.
>-TEST_F(AudioDeviceTest, RunPlayoutAndRecordingInFullDuplex) {
>+TEST_P(AudioDeviceTest, RunPlayoutAndRecordingInFullDuplex) {
>   SKIP_TEST_IF_NOT(requirements_satisfied());
>   NiceMock<MockAudioTransport> mock(TransportType::kPlayAndRecord);
>   FifoAudioStream audio_stream;
>   mock.HandleCallbacks(event(), &audio_stream,
>                        kFullDuplexTimeInSec * kNumCallbacksPerSecond);
>   EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
>-  // Run both sides in mono to make the loopback packet handling less complex.
>-  // The test works for stereo as well; the only requirement is that both sides
>-  // use the same configuration.
>-  EXPECT_EQ(0, audio_device()->SetStereoPlayout(false));
>-  EXPECT_EQ(0, audio_device()->SetStereoRecording(false));
>+  // Run both sides using the same channel configuration to avoid conversions
>+  // between mono/stereo while running in full duplex mode. Also, some devices
>+  // (mainly on Windows) do not support mono.
>+  EXPECT_EQ(0, audio_device()->SetStereoPlayout(true));
>+  EXPECT_EQ(0, audio_device()->SetStereoRecording(true));
>   StartPlayout();
>   StartRecording();
>   event()->Wait(static_cast<int>(
>@@ -723,15 +877,15 @@ TEST_F(AudioDeviceTest, RunPlayoutAndRecordingInFullDuplex) {
> // some sort of audio feedback loop. E.g. a headset where the mic is placed
> // close to the speaker to ensure highest possible echo. It is also recommended
> // to run the test at highest possible output volume.
>-TEST_F(AudioDeviceTest, DISABLED_MeasureLoopbackLatency) {
>+TEST_P(AudioDeviceTest, DISABLED_MeasureLoopbackLatency) {
>   SKIP_TEST_IF_NOT(requirements_satisfied());
>   NiceMock<MockAudioTransport> mock(TransportType::kPlayAndRecord);
>   LatencyAudioStream audio_stream;
>   mock.HandleCallbacks(event(), &audio_stream,
>                        kMeasureLatencyTimeInSec * kNumCallbacksPerSecond);
>   EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
>-  EXPECT_EQ(0, audio_device()->SetStereoPlayout(false));
>-  EXPECT_EQ(0, audio_device()->SetStereoRecording(false));
>+  EXPECT_EQ(0, audio_device()->SetStereoPlayout(true));
>+  EXPECT_EQ(0, audio_device()->SetStereoRecording(true));
>   StartPlayout();
>   StartRecording();
>   event()->Wait(static_cast<int>(
>@@ -746,4 +900,20 @@ TEST_F(AudioDeviceTest, DISABLED_MeasureLoopbackLatency) {
>   audio_stream.PrintResults();
> }
> 
>+#ifdef WEBRTC_WIN
>+// Test two different audio layers (or rather two different Core Audio
>+// implementations) for Windows.
>+INSTANTIATE_TEST_CASE_P(
>+    AudioLayerWin,
>+    AudioDeviceTest,
>+    ::testing::Values(AudioDeviceModule::kPlatformDefaultAudio,
>+                      AudioDeviceModule::kWindowsCoreAudio2));
>+#else
>+// For all platforms but Windows, only test the default audio layer.
>+INSTANTIATE_TEST_CASE_P(
>+    AudioLayer,
>+    AudioDeviceTest,
>+    ::testing::Values(AudioDeviceModule::kPlatformDefaultAudio));
>+#endif
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/dummy/audio_device_dummy.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/dummy/audio_device_dummy.cc
>index b7b61a72abd..b8fd8370389 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/dummy/audio_device_dummy.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/dummy/audio_device_dummy.cc
>@@ -21,13 +21,21 @@ AudioDeviceGeneric::InitStatus AudioDeviceDummy::Init() {
>   return InitStatus::OK;
> }
> 
>-int32_t AudioDeviceDummy::Terminate() { return 0; }
>+int32_t AudioDeviceDummy::Terminate() {
>+  return 0;
>+}
> 
>-bool AudioDeviceDummy::Initialized() const { return true; }
>+bool AudioDeviceDummy::Initialized() const {
>+  return true;
>+}
> 
>-int16_t AudioDeviceDummy::PlayoutDevices() { return -1; }
>+int16_t AudioDeviceDummy::PlayoutDevices() {
>+  return -1;
>+}
> 
>-int16_t AudioDeviceDummy::RecordingDevices() { return -1; }
>+int16_t AudioDeviceDummy::RecordingDevices() {
>+  return -1;
>+}
> 
> int32_t AudioDeviceDummy::PlayoutDeviceName(uint16_t index,
>                                             char name[kAdmMaxDeviceNameSize],
>@@ -41,59 +49,99 @@ int32_t AudioDeviceDummy::RecordingDeviceName(uint16_t index,
>   return -1;
> }
> 
>-int32_t AudioDeviceDummy::SetPlayoutDevice(uint16_t index) { return -1; }
>+int32_t AudioDeviceDummy::SetPlayoutDevice(uint16_t index) {
>+  return -1;
>+}
> 
> int32_t AudioDeviceDummy::SetPlayoutDevice(
>     AudioDeviceModule::WindowsDeviceType device) {
>   return -1;
> }
> 
>-int32_t AudioDeviceDummy::SetRecordingDevice(uint16_t index) { return -1; }
>+int32_t AudioDeviceDummy::SetRecordingDevice(uint16_t index) {
>+  return -1;
>+}
> 
> int32_t AudioDeviceDummy::SetRecordingDevice(
>     AudioDeviceModule::WindowsDeviceType device) {
>   return -1;
> }
> 
>-int32_t AudioDeviceDummy::PlayoutIsAvailable(bool& available) { return -1; }
>+int32_t AudioDeviceDummy::PlayoutIsAvailable(bool& available) {
>+  return -1;
>+}
> 
>-int32_t AudioDeviceDummy::InitPlayout() { return -1; }
>+int32_t AudioDeviceDummy::InitPlayout() {
>+  return -1;
>+}
> 
>-bool AudioDeviceDummy::PlayoutIsInitialized() const { return false; }
>+bool AudioDeviceDummy::PlayoutIsInitialized() const {
>+  return false;
>+}
> 
>-int32_t AudioDeviceDummy::RecordingIsAvailable(bool& available) { return -1; }
>+int32_t AudioDeviceDummy::RecordingIsAvailable(bool& available) {
>+  return -1;
>+}
> 
>-int32_t AudioDeviceDummy::InitRecording() { return -1; }
>+int32_t AudioDeviceDummy::InitRecording() {
>+  return -1;
>+}
> 
>-bool AudioDeviceDummy::RecordingIsInitialized() const { return false; }
>+bool AudioDeviceDummy::RecordingIsInitialized() const {
>+  return false;
>+}
> 
>-int32_t AudioDeviceDummy::StartPlayout() { return -1; }
>+int32_t AudioDeviceDummy::StartPlayout() {
>+  return -1;
>+}
> 
>-int32_t AudioDeviceDummy::StopPlayout() { return 0; }
>+int32_t AudioDeviceDummy::StopPlayout() {
>+  return 0;
>+}
> 
>-bool AudioDeviceDummy::Playing() const { return false; }
>+bool AudioDeviceDummy::Playing() const {
>+  return false;
>+}
> 
>-int32_t AudioDeviceDummy::StartRecording() { return -1; }
>+int32_t AudioDeviceDummy::StartRecording() {
>+  return -1;
>+}
> 
>-int32_t AudioDeviceDummy::StopRecording() { return 0; }
>+int32_t AudioDeviceDummy::StopRecording() {
>+  return 0;
>+}
> 
>-bool AudioDeviceDummy::Recording() const { return false; }
>+bool AudioDeviceDummy::Recording() const {
>+  return false;
>+}
> 
>-int32_t AudioDeviceDummy::InitSpeaker() { return -1; }
>+int32_t AudioDeviceDummy::InitSpeaker() {
>+  return -1;
>+}
> 
>-bool AudioDeviceDummy::SpeakerIsInitialized() const { return false; }
>+bool AudioDeviceDummy::SpeakerIsInitialized() const {
>+  return false;
>+}
> 
>-int32_t AudioDeviceDummy::InitMicrophone() { return -1; }
>+int32_t AudioDeviceDummy::InitMicrophone() {
>+  return -1;
>+}
> 
>-bool AudioDeviceDummy::MicrophoneIsInitialized() const { return false; }
>+bool AudioDeviceDummy::MicrophoneIsInitialized() const {
>+  return false;
>+}
> 
> int32_t AudioDeviceDummy::SpeakerVolumeIsAvailable(bool& available) {
>   return -1;
> }
> 
>-int32_t AudioDeviceDummy::SetSpeakerVolume(uint32_t volume) { return -1; }
>+int32_t AudioDeviceDummy::SetSpeakerVolume(uint32_t volume) {
>+  return -1;
>+}
> 
>-int32_t AudioDeviceDummy::SpeakerVolume(uint32_t& volume) const { return -1; }
>+int32_t AudioDeviceDummy::SpeakerVolume(uint32_t& volume) const {
>+  return -1;
>+}
> 
> int32_t AudioDeviceDummy::MaxSpeakerVolume(uint32_t& maxVolume) const {
>   return -1;
>@@ -107,7 +155,9 @@ int32_t AudioDeviceDummy::MicrophoneVolumeIsAvailable(bool& available) {
>   return -1;
> }
> 
>-int32_t AudioDeviceDummy::SetMicrophoneVolume(uint32_t volume) { return -1; }
>+int32_t AudioDeviceDummy::SetMicrophoneVolume(uint32_t volume) {
>+  return -1;
>+}
> 
> int32_t AudioDeviceDummy::MicrophoneVolume(uint32_t& volume) const {
>   return -1;
>@@ -121,36 +171,56 @@ int32_t AudioDeviceDummy::MinMicrophoneVolume(uint32_t& minVolume) const {
>   return -1;
> }
> 
>-int32_t AudioDeviceDummy::SpeakerMuteIsAvailable(bool& available) { return -1; }
>+int32_t AudioDeviceDummy::SpeakerMuteIsAvailable(bool& available) {
>+  return -1;
>+}
> 
>-int32_t AudioDeviceDummy::SetSpeakerMute(bool enable) { return -1; }
>+int32_t AudioDeviceDummy::SetSpeakerMute(bool enable) {
>+  return -1;
>+}
> 
>-int32_t AudioDeviceDummy::SpeakerMute(bool& enabled) const { return -1; }
>+int32_t AudioDeviceDummy::SpeakerMute(bool& enabled) const {
>+  return -1;
>+}
> 
> int32_t AudioDeviceDummy::MicrophoneMuteIsAvailable(bool& available) {
>   return -1;
> }
> 
>-int32_t AudioDeviceDummy::SetMicrophoneMute(bool enable) { return -1; }
>+int32_t AudioDeviceDummy::SetMicrophoneMute(bool enable) {
>+  return -1;
>+}
> 
>-int32_t AudioDeviceDummy::MicrophoneMute(bool& enabled) const { return -1; }
>+int32_t AudioDeviceDummy::MicrophoneMute(bool& enabled) const {
>+  return -1;
>+}
> 
> int32_t AudioDeviceDummy::StereoPlayoutIsAvailable(bool& available) {
>   return -1;
> }
>-int32_t AudioDeviceDummy::SetStereoPlayout(bool enable) { return -1; }
>+int32_t AudioDeviceDummy::SetStereoPlayout(bool enable) {
>+  return -1;
>+}
> 
>-int32_t AudioDeviceDummy::StereoPlayout(bool& enabled) const { return -1; }
>+int32_t AudioDeviceDummy::StereoPlayout(bool& enabled) const {
>+  return -1;
>+}
> 
> int32_t AudioDeviceDummy::StereoRecordingIsAvailable(bool& available) {
>   return -1;
> }
> 
>-int32_t AudioDeviceDummy::SetStereoRecording(bool enable) { return -1; }
>+int32_t AudioDeviceDummy::SetStereoRecording(bool enable) {
>+  return -1;
>+}
> 
>-int32_t AudioDeviceDummy::StereoRecording(bool& enabled) const { return -1; }
>+int32_t AudioDeviceDummy::StereoRecording(bool& enabled) const {
>+  return -1;
>+}
> 
>-int32_t AudioDeviceDummy::PlayoutDelay(uint16_t& delayMS) const { return -1; }
>+int32_t AudioDeviceDummy::PlayoutDelay(uint16_t& delayMS) const {
>+  return -1;
>+}
> 
> void AudioDeviceDummy::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {}
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/dummy/file_audio_device.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/dummy/file_audio_device.h
>index 1a7c78acab7..f07690e7e03 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/dummy/file_audio_device.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/dummy/file_audio_device.h
>@@ -18,8 +18,8 @@
> 
> #include "modules/audio_device/audio_device_generic.h"
> #include "rtc_base/criticalsection.h"
>+#include "rtc_base/system/file_wrapper.h"
> #include "rtc_base/timeutils.h"
>-#include "system_wrappers/include/file_wrapper.h"
> 
> namespace rtc {
> class PlatformThread;
>@@ -38,8 +38,7 @@ class FileAudioDevice : public AudioDeviceGeneric {
>   // The input file should be a readable 48k stereo raw file, and the output
>   // file should point to a writable location. The output format will also be
>   // 48k stereo raw audio.
>-  FileAudioDevice(const char* inputFilename,
>-                  const char* outputFilename);
>+  FileAudioDevice(const char* inputFilename, const char* outputFilename);
>   virtual ~FileAudioDevice();
> 
>   // Retrieve the currently utilized audio layer
>@@ -138,7 +137,7 @@ class FileAudioDevice : public AudioDeviceGeneric {
>   int32_t _record_index;
>   AudioDeviceBuffer* _ptrAudioBuffer;
>   int8_t* _recordingBuffer;  // In bytes.
>-  int8_t* _playoutBuffer;  // In bytes.
>+  int8_t* _playoutBuffer;    // In bytes.
>   uint32_t _recordingFramesLeft;
>   uint32_t _playoutFramesLeft;
>   rtc::CriticalSection _critSect;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/fine_audio_buffer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/fine_audio_buffer.cc
>index 7683834f7cb..bd18359009d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/fine_audio_buffer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/fine_audio_buffer.cc
>@@ -10,26 +10,32 @@
> 
> #include "modules/audio_device/fine_audio_buffer.h"
> 
>-#include <memory.h>
>-#include <stdio.h>
>-#include <algorithm>
>-
> #include "modules/audio_device/audio_device_buffer.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>+#include "rtc_base/numerics/safe_conversions.h"
> 
> namespace webrtc {
> 
>-FineAudioBuffer::FineAudioBuffer(AudioDeviceBuffer* device_buffer,
>-                                 int sample_rate,
>-                                 size_t capacity)
>-    : device_buffer_(device_buffer),
>-      sample_rate_(sample_rate),
>-      samples_per_10_ms_(static_cast<size_t>(sample_rate_ * 10 / 1000)),
>-      bytes_per_10_ms_(samples_per_10_ms_ * sizeof(int16_t)),
>-      playout_buffer_(0, capacity),
>-      record_buffer_(0, capacity) {
>-  RTC_LOG(INFO) << "samples_per_10_ms_:" << samples_per_10_ms_;
>+FineAudioBuffer::FineAudioBuffer(AudioDeviceBuffer* audio_device_buffer)
>+    : audio_device_buffer_(audio_device_buffer),
>+      playout_samples_per_channel_10ms_(rtc::dchecked_cast<size_t>(
>+          audio_device_buffer->PlayoutSampleRate() * 10 / 1000)),
>+      record_samples_per_channel_10ms_(rtc::dchecked_cast<size_t>(
>+          audio_device_buffer->RecordingSampleRate() * 10 / 1000)),
>+      playout_channels_(audio_device_buffer->PlayoutChannels()),
>+      record_channels_(audio_device_buffer->RecordingChannels()) {
>+  RTC_DCHECK(audio_device_buffer_);
>+  if (IsReadyForPlayout()) {
>+    RTC_DLOG(INFO) << "playout_samples_per_channel_10ms: "
>+                   << playout_samples_per_channel_10ms_;
>+    RTC_DLOG(INFO) << "playout_channels: " << playout_channels_;
>+  }
>+  if (IsReadyForRecord()) {
>+    RTC_DLOG(INFO) << "record_samples_per_channel_10ms: "
>+                   << record_samples_per_channel_10ms_;
>+    RTC_DLOG(INFO) << "record_channels: " << record_channels_;
>+  }
> }
> 
> FineAudioBuffer::~FineAudioBuffer() {}
>@@ -42,50 +48,76 @@ void FineAudioBuffer::ResetRecord() {
>   record_buffer_.Clear();
> }
> 
>-void FineAudioBuffer::GetPlayoutData(rtc::ArrayView<int8_t> audio_buffer) {
>+bool FineAudioBuffer::IsReadyForPlayout() const {
>+  return playout_samples_per_channel_10ms_ > 0 && playout_channels_ > 0;
>+}
>+
>+bool FineAudioBuffer::IsReadyForRecord() const {
>+  return record_samples_per_channel_10ms_ > 0 && record_channels_ > 0;
>+}
>+
>+void FineAudioBuffer::GetPlayoutData(rtc::ArrayView<int16_t> audio_buffer,
>+                                     int playout_delay_ms) {
>+  RTC_DCHECK(IsReadyForPlayout());
>   // Ask WebRTC for new data in chunks of 10ms until we have enough to
>   // fulfill the request. It is possible that the buffer already contains
>   // enough samples from the last round.
>-  const size_t num_bytes = audio_buffer.size();
>-  while (playout_buffer_.size() < num_bytes) {
>-    // Get 10ms decoded audio from WebRTC.
>-    device_buffer_->RequestPlayoutData(samples_per_10_ms_);
>-    // Append |bytes_per_10_ms_| elements to the end of the buffer.
>-    const size_t bytes_written = playout_buffer_.AppendData(
>-        bytes_per_10_ms_, [&](rtc::ArrayView<int8_t> buf) {
>-          const size_t samples_per_channel =
>-              device_buffer_->GetPlayoutData(buf.data());
>-          // TODO(henrika): this class is only used on mobile devices and is
>-          // currently limited to mono. Modifications are needed for stereo.
>-          return sizeof(int16_t) * samples_per_channel;
>-        });
>-    RTC_DCHECK_EQ(bytes_per_10_ms_, bytes_written);
>+  while (playout_buffer_.size() < audio_buffer.size()) {
>+    // Get 10ms decoded audio from WebRTC. The ADB knows about number of
>+    // channels; hence we can ask for number of samples per channel here.
>+    if (audio_device_buffer_->RequestPlayoutData(
>+            playout_samples_per_channel_10ms_) ==
>+        static_cast<int32_t>(playout_samples_per_channel_10ms_)) {
>+      // Append 10ms to the end of the local buffer taking number of channels
>+      // into account.
>+      const size_t num_elements_10ms =
>+          playout_channels_ * playout_samples_per_channel_10ms_;
>+      const size_t written_elements = playout_buffer_.AppendData(
>+          num_elements_10ms, [&](rtc::ArrayView<int16_t> buf) {
>+            const size_t samples_per_channel_10ms =
>+                audio_device_buffer_->GetPlayoutData(buf.data());
>+            return playout_channels_ * samples_per_channel_10ms;
>+          });
>+      RTC_DCHECK_EQ(num_elements_10ms, written_elements);
>+    } else {
>+      // Provide silence if AudioDeviceBuffer::RequestPlayoutData() fails.
>+      // Can e.g. happen when an AudioTransport has not been registered.
>+      const size_t num_bytes = audio_buffer.size() * sizeof(int16_t);
>+      std::memset(audio_buffer.data(), 0, num_bytes);
>+      return;
>+    }
>   }
>+
>   // Provide the requested number of bytes to the consumer.
>+  const size_t num_bytes = audio_buffer.size() * sizeof(int16_t);
>   memcpy(audio_buffer.data(), playout_buffer_.data(), num_bytes);
>   // Move remaining samples to start of buffer to prepare for next round.
>-  memmove(playout_buffer_.data(), playout_buffer_.data() + num_bytes,
>-          playout_buffer_.size() - num_bytes);
>-  playout_buffer_.SetSize(playout_buffer_.size() - num_bytes);
>+  memmove(playout_buffer_.data(), playout_buffer_.data() + audio_buffer.size(),
>+          (playout_buffer_.size() - audio_buffer.size()) * sizeof(int16_t));
>+  playout_buffer_.SetSize(playout_buffer_.size() - audio_buffer.size());
>+  // Cache playout latency for usage in DeliverRecordedData();
>+  playout_delay_ms_ = playout_delay_ms;
> }
> 
> void FineAudioBuffer::DeliverRecordedData(
>-    rtc::ArrayView<const int8_t> audio_buffer,
>-    int playout_delay_ms,
>+    rtc::ArrayView<const int16_t> audio_buffer,
>     int record_delay_ms) {
>-  // Always append new data and grow the buffer if needed.
>+  RTC_DCHECK(IsReadyForRecord());
>+  // Always append new data and grow the buffer when needed.
>   record_buffer_.AppendData(audio_buffer.data(), audio_buffer.size());
>   // Consume samples from buffer in chunks of 10ms until there is not
>-  // enough data left. The number of remaining bytes in the cache is given by
>-  // the new size of the buffer.
>-  while (record_buffer_.size() >= bytes_per_10_ms_) {
>-    device_buffer_->SetRecordedBuffer(record_buffer_.data(),
>-                                      samples_per_10_ms_);
>-    device_buffer_->SetVQEData(playout_delay_ms, record_delay_ms);
>-    device_buffer_->DeliverRecordedData();
>-    memmove(record_buffer_.data(), record_buffer_.data() + bytes_per_10_ms_,
>-            record_buffer_.size() - bytes_per_10_ms_);
>-    record_buffer_.SetSize(record_buffer_.size() - bytes_per_10_ms_);
>+  // enough data left. The number of remaining samples in the cache is given by
>+  // the new size of the internal |record_buffer_|.
>+  const size_t num_elements_10ms =
>+      record_channels_ * record_samples_per_channel_10ms_;
>+  while (record_buffer_.size() >= num_elements_10ms) {
>+    audio_device_buffer_->SetRecordedBuffer(record_buffer_.data(),
>+                                            record_samples_per_channel_10ms_);
>+    audio_device_buffer_->SetVQEData(playout_delay_ms_, record_delay_ms);
>+    audio_device_buffer_->DeliverRecordedData();
>+    memmove(record_buffer_.data(), record_buffer_.data() + num_elements_10ms,
>+            (record_buffer_.size() - num_elements_10ms) * sizeof(int16_t));
>+    record_buffer_.SetSize(record_buffer_.size() - num_elements_10ms);
>   }
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/fine_audio_buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/fine_audio_buffer.h
>index 6bd047eca46..210eda834a8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/fine_audio_buffer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/fine_audio_buffer.h
>@@ -11,79 +11,82 @@
> #ifndef MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_
> #define MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_
> 
>-#include <memory>
>-
> #include "api/array_view.h"
> #include "rtc_base/buffer.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
> class AudioDeviceBuffer;
> 
>-// FineAudioBuffer takes an AudioDeviceBuffer (ADB) which deals with audio data
>-// corresponding to 10ms of data. It then allows for this data to be pulled in
>-// a finer or coarser granularity. I.e. interacting with this class instead of
>-// directly with the AudioDeviceBuffer one can ask for any number of audio data
>-// samples. This class also ensures that audio data can be delivered to the ADB
>-// in 10ms chunks when the size of the provided audio buffers differs from 10ms.
>+// FineAudioBuffer takes an AudioDeviceBuffer (ADB) which deals with 16-bit PCM
>+// audio samples corresponding to 10ms of data. It then allows for this data
>+// to be pulled in a finer or coarser granularity. I.e. interacting with this
>+// class instead of directly with the AudioDeviceBuffer one can ask for any
>+// number of audio data samples. This class also ensures that audio data can be
>+// delivered to the ADB in 10ms chunks when the size of the provided audio
>+// buffers differs from 10ms.
> // As an example: calling DeliverRecordedData() with 5ms buffers will deliver
> // accumulated 10ms worth of data to the ADB every second call.
>-// TODO(henrika): add support for stereo when mobile platforms need it.
> class FineAudioBuffer {
>  public:
>   // |device_buffer| is a buffer that provides 10ms of audio data.
>-  // |sample_rate| is the sample rate of the audio data. This is needed because
>-  // |device_buffer| delivers 10ms of data. Given the sample rate the number
>-  // of samples can be calculated. The |capacity| ensures that the buffer size
>-  // can be increased to at least capacity without further reallocation.
>-  FineAudioBuffer(AudioDeviceBuffer* device_buffer,
>-                  int sample_rate,
>-                  size_t capacity);
>+  FineAudioBuffer(AudioDeviceBuffer* audio_device_buffer);
>   ~FineAudioBuffer();
> 
>-  // Clears buffers and counters dealing with playour and/or recording.
>+  // Clears buffers and counters dealing with playout and/or recording.
>   void ResetPlayout();
>   void ResetRecord();
> 
>+  // Utility methods which returns true if valid parameters are acquired at
>+  // constructions.
>+  bool IsReadyForPlayout() const;
>+  bool IsReadyForRecord() const;
>+
>   // Copies audio samples into |audio_buffer|Â where number of requested
>   // elements is specified by |audio_buffer.size()|. The producer will always
>   // fill up the audio buffer and if no audio exists, the buffer will contain
>-  // silence instead.
>-  void GetPlayoutData(rtc::ArrayView<int8_t> audio_buffer);
>+  // silence instead. The provided delay estimate in |playout_delay_ms| should
>+  // contain an estimate of the latency between when an audio frame is read from
>+  // WebRTC and when it is played out on the speaker.
>+  void GetPlayoutData(rtc::ArrayView<int16_t> audio_buffer,
>+                      int playout_delay_ms);
> 
>   // Consumes the audio data in |audio_buffer| and sends it to the WebRTC layer
>-  // in chunks of 10ms. The provided delay estimates in |playout_delay_ms| and
>-  // |record_delay_ms| are given to the AEC in the audio processing module.
>+  // in chunks of 10ms. The sum of the provided delay estimate in
>+  // |record_delay_ms| and the latest |playout_delay_ms| in GetPlayoutData()
>+  // are given to the AEC in the audio processing module.
>   // They can be fixed values on most platforms and they are ignored if an
>   // external (hardware/built-in) AEC is used.
>   // Example: buffer size is 5ms => call #1 stores 5ms of data, call #2 stores
>-  // 5ms of data and sends a total of 10ms to WebRTC and clears the intenal
>+  // 5ms of data and sends a total of 10ms to WebRTC and clears the internal
>   // cache. Call #3 restarts the scheme above.
>-  void DeliverRecordedData(rtc::ArrayView<const int8_t> audio_buffer,
>-                           int playout_delay_ms,
>+  void DeliverRecordedData(rtc::ArrayView<const int16_t> audio_buffer,
>                            int record_delay_ms);
> 
>  private:
>   // Device buffer that works with 10ms chunks of data both for playout and
>   // for recording. I.e., the WebRTC side will always be asked for audio to be
>   // played out in 10ms chunks and recorded audio will be sent to WebRTC in
>-  // 10ms chunks as well. This pointer is owned by the constructor of this
>+  // 10ms chunks as well. This raw pointer is owned by the constructor of this
>   // class and the owner must ensure that the pointer is valid during the life-
>   // time of this object.
>-  AudioDeviceBuffer* const device_buffer_;
>-  // Sample rate in Hertz.
>-  const int sample_rate_;
>-  // Number of audio samples per 10ms.
>-  const size_t samples_per_10_ms_;
>-  // Number of audio bytes per 10ms.
>-  const size_t bytes_per_10_ms_;
>+  AudioDeviceBuffer* const audio_device_buffer_;
>+  // Number of audio samples per channel per 10ms. Set once at construction
>+  // based on parameters in |audio_device_buffer|.
>+  const size_t playout_samples_per_channel_10ms_;
>+  const size_t record_samples_per_channel_10ms_;
>+  // Number of audio channels. Set once at construction based on parameters in
>+  // |audio_device_buffer|.
>+  const size_t playout_channels_;
>+  const size_t record_channels_;
>   // Storage for output samples from which a consumer can read audio buffers
>   // in any size using GetPlayoutData().
>-  rtc::BufferT<int8_t> playout_buffer_;
>+  rtc::BufferT<int16_t> playout_buffer_;
>   // Storage for input samples that are about to be delivered to the WebRTC
>   // ADB or remains from the last successful delivery of a 10ms audio buffer.
>-  rtc::BufferT<int8_t> record_buffer_;
>+  rtc::BufferT<int16_t> record_buffer_;
>+  // Contains latest delay estimate given to GetPlayoutData().
>+  int playout_delay_ms_ = 0;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/fine_audio_buffer_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/fine_audio_buffer_unittest.cc
>index f493e51a268..aea899c94ad 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/fine_audio_buffer_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/fine_audio_buffer_unittest.cc
>@@ -26,6 +26,7 @@ using ::testing::Return;
> namespace webrtc {
> 
> const int kSampleRate = 44100;
>+const int kChannels = 2;
> const int kSamplesPer10Ms = kSampleRate * 10 / 1000;
> 
> // The fake audio data is 0,1,..SCHAR_MAX-1,0,1,... This is to make it easy
>@@ -34,7 +35,7 @@ const int kSamplesPer10Ms = kSampleRate * 10 / 1000;
> // buffer 2 would contain 3,4,5. Note that SCHAR_MAX is 127 so wrap-around
> // will happen.
> // |buffer| is the audio buffer to verify.
>-bool VerifyBuffer(const int8_t* buffer, int buffer_number, int size) {
>+bool VerifyBuffer(const int16_t* buffer, int buffer_number, int size) {
>   int start_value = (buffer_number * size) % SCHAR_MAX;
>   for (int i = 0; i < size; ++i) {
>     if (buffer[i] != (i + start_value) % SCHAR_MAX) {
>@@ -52,18 +53,18 @@ bool VerifyBuffer(const int8_t* buffer, int buffer_number, int size) {
> // |samples_per_10_ms| is the number of samples that should be written to the
> // buffer (|arg0|).
> ACTION_P2(UpdateBuffer, iteration, samples_per_10_ms) {
>-  int8_t* buffer = static_cast<int8_t*>(arg0);
>-  int bytes_per_10_ms = samples_per_10_ms * static_cast<int>(sizeof(int16_t));
>-  int start_value = (iteration * bytes_per_10_ms) % SCHAR_MAX;
>-  for (int i = 0; i < bytes_per_10_ms; ++i) {
>+  int16_t* buffer = static_cast<int16_t*>(arg0);
>+  int start_value = (iteration * samples_per_10_ms) % SCHAR_MAX;
>+  for (int i = 0; i < samples_per_10_ms; ++i) {
>     buffer[i] = (i + start_value) % SCHAR_MAX;
>   }
>-  return samples_per_10_ms;
>+  // Should return samples per channel.
>+  return samples_per_10_ms / kChannels;
> }
> 
> // Writes a periodic ramp pattern to the supplied |buffer|. See UpdateBuffer()
> // for details.
>-void UpdateInputBuffer(int8_t* buffer, int iteration, int size) {
>+void UpdateInputBuffer(int16_t* buffer, int iteration, int size) {
>   int start_value = (iteration * size) % SCHAR_MAX;
>   for (int i = 0; i < size; ++i) {
>     buffer[i] = (i + start_value) % SCHAR_MAX;
>@@ -75,31 +76,34 @@ void UpdateInputBuffer(int8_t* buffer, int iteration, int size) {
> // supplied using a buffer size that is smaller or larger than 10ms.
> // See VerifyBuffer() for details.
> ACTION_P2(VerifyInputBuffer, iteration, samples_per_10_ms) {
>-  const int8_t* buffer = static_cast<const int8_t*>(arg0);
>-  int bytes_per_10_ms = samples_per_10_ms * static_cast<int>(sizeof(int16_t));
>-  int start_value = (iteration * bytes_per_10_ms) % SCHAR_MAX;
>-  for (int i = 0; i < bytes_per_10_ms; ++i) {
>+  const int16_t* buffer = static_cast<const int16_t*>(arg0);
>+  int start_value = (iteration * samples_per_10_ms) % SCHAR_MAX;
>+  for (int i = 0; i < samples_per_10_ms; ++i) {
>     EXPECT_EQ(buffer[i], (i + start_value) % SCHAR_MAX);
>   }
>   return 0;
> }
> 
> void RunFineBufferTest(int frame_size_in_samples) {
>-  const int kFrameSizeBytes =
>-      frame_size_in_samples * static_cast<int>(sizeof(int16_t));
>+  const int kFrameSizeSamples = frame_size_in_samples;
>   const int kNumberOfFrames = 5;
>   // Ceiling of integer division: 1 + ((x - 1) / y)
>   const int kNumberOfUpdateBufferCalls =
>       1 + ((kNumberOfFrames * frame_size_in_samples - 1) / kSamplesPer10Ms);
> 
>   MockAudioDeviceBuffer audio_device_buffer;
>+  audio_device_buffer.SetPlayoutSampleRate(kSampleRate);
>+  audio_device_buffer.SetPlayoutChannels(kChannels);
>+  audio_device_buffer.SetRecordingSampleRate(kSampleRate);
>+  audio_device_buffer.SetRecordingChannels(kChannels);
>+
>   EXPECT_CALL(audio_device_buffer, RequestPlayoutData(_))
>       .WillRepeatedly(Return(kSamplesPer10Ms));
>   {
>     InSequence s;
>     for (int i = 0; i < kNumberOfUpdateBufferCalls; ++i) {
>       EXPECT_CALL(audio_device_buffer, GetPlayoutData(_))
>-          .WillOnce(UpdateBuffer(i, kSamplesPer10Ms))
>+          .WillOnce(UpdateBuffer(i, kChannels * kSamplesPer10Ms))
>           .RetiresOnSaturation();
>     }
>   }
>@@ -107,7 +111,7 @@ void RunFineBufferTest(int frame_size_in_samples) {
>     InSequence s;
>     for (int j = 0; j < kNumberOfUpdateBufferCalls - 1; ++j) {
>       EXPECT_CALL(audio_device_buffer, SetRecordedBuffer(_, kSamplesPer10Ms))
>-          .WillOnce(VerifyInputBuffer(j, kSamplesPer10Ms))
>+          .WillOnce(VerifyInputBuffer(j, kChannels * kSamplesPer10Ms))
>           .RetiresOnSaturation();
>     }
>   }
>@@ -115,20 +119,26 @@ void RunFineBufferTest(int frame_size_in_samples) {
>       .Times(kNumberOfUpdateBufferCalls - 1);
>   EXPECT_CALL(audio_device_buffer, DeliverRecordedData())
>       .Times(kNumberOfUpdateBufferCalls - 1)
>-      .WillRepeatedly(Return(kSamplesPer10Ms));
>+      .WillRepeatedly(Return(0));
> 
>-  FineAudioBuffer fine_buffer(&audio_device_buffer, kSampleRate,
>-                              kFrameSizeBytes);
>-  std::unique_ptr<int8_t[]> out_buffer(new int8_t[kFrameSizeBytes]);
>-  std::unique_ptr<int8_t[]> in_buffer(new int8_t[kFrameSizeBytes]);
>+  FineAudioBuffer fine_buffer(&audio_device_buffer);
>+  std::unique_ptr<int16_t[]> out_buffer(
>+      new int16_t[kChannels * kFrameSizeSamples]);
>+  std::unique_ptr<int16_t[]> in_buffer(
>+      new int16_t[kChannels * kFrameSizeSamples]);
> 
>   for (int i = 0; i < kNumberOfFrames; ++i) {
>     fine_buffer.GetPlayoutData(
>-        rtc::ArrayView<int8_t>(out_buffer.get(), kFrameSizeBytes));
>-    EXPECT_TRUE(VerifyBuffer(out_buffer.get(), i, kFrameSizeBytes));
>-    UpdateInputBuffer(in_buffer.get(), i, kFrameSizeBytes);
>+        rtc::ArrayView<int16_t>(out_buffer.get(),
>+                                kChannels * kFrameSizeSamples),
>+        0);
>+    EXPECT_TRUE(
>+        VerifyBuffer(out_buffer.get(), i, kChannels * kFrameSizeSamples));
>+    UpdateInputBuffer(in_buffer.get(), i, kChannels * kFrameSizeSamples);
>     fine_buffer.DeliverRecordedData(
>-        rtc::ArrayView<const int8_t>(in_buffer.get(), kFrameSizeBytes), 0, 0);
>+        rtc::ArrayView<const int16_t>(in_buffer.get(),
>+                                      kChannels * kFrameSizeSamples),
>+        0);
>   }
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/audio_device.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/audio_device.h
>index 5e88b88f177..a8613193740 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/audio_device.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/audio_device.h
>@@ -1,4 +1,4 @@
>-  /*
>+/*
>  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>  *
>  *  Use of this source code is governed by a BSD-style license
>@@ -12,8 +12,8 @@
> #define MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_H_
> 
> #include "modules/audio_device/include/audio_device_defines.h"
>-#include "rtc_base/scoped_ref_ptr.h"
> #include "rtc_base/refcount.h"
>+#include "rtc_base/scoped_ref_ptr.h"
> 
> namespace webrtc {
> 
>@@ -21,20 +21,20 @@ class AudioDeviceModule : public rtc::RefCountInterface {
>  public:
>   // Deprecated.
>   // TODO(henrika): to be removed.
>-  enum ErrorCode {
>-    kAdmErrNone = 0,
>-    kAdmErrArgument = 1
>-  };
>+  enum ErrorCode { kAdmErrNone = 0, kAdmErrArgument = 1 };
> 
>   enum AudioLayer {
>     kPlatformDefaultAudio = 0,
>-    kWindowsCoreAudio = 2,
>-    kLinuxAlsaAudio = 3,
>-    kLinuxPulseAudio = 4,
>-    kAndroidJavaAudio = 5,
>-    kAndroidOpenSLESAudio = 6,
>-    kAndroidJavaInputAndOpenSLESOutputAudio = 7,
>-    kDummyAudio = 8
>+    kWindowsCoreAudio,
>+    kWindowsCoreAudio2,  // experimental
>+    kLinuxAlsaAudio,
>+    kLinuxPulseAudio,
>+    kAndroidJavaAudio,
>+    kAndroidOpenSLESAudio,
>+    kAndroidJavaInputAndOpenSLESOutputAudio,
>+    kAndroidAAudioAudio,
>+    kAndroidJavaInputAndAAudioOutputAudio,
>+    kDummyAudio,
>   };
> 
>   enum WindowsDeviceType {
>@@ -43,11 +43,7 @@ class AudioDeviceModule : public rtc::RefCountInterface {
>   };
> 
>   // TODO(bugs.webrtc.org/7306): deprecated.
>-  enum ChannelType {
>-    kChannelLeft = 0,
>-    kChannelRight = 1,
>-    kChannelBoth = 2
>-  };
>+  enum ChannelType { kChannelLeft = 0, kChannelRight = 1, kChannelBoth = 2 };
> 
>  public:
>   // Creates an ADM.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/audio_device_default.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/audio_device_default.h
>new file mode 100644
>index 00000000000..8b052fb3405
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/audio_device_default.h
>@@ -0,0 +1,130 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_DEFAULT_H_
>+#define MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_DEFAULT_H_
>+
>+#include "modules/audio_device/include/audio_device.h"
>+
>+namespace webrtc {
>+namespace webrtc_impl {
>+
>+// AudioDeviceModuleDefault template adds default implementation for all
>+// AudioDeviceModule methods to the class, which inherits from
>+// AudioDeviceModuleDefault<T>.
>+template <typename T>
>+class AudioDeviceModuleDefault : public T {
>+ public:
>+  AudioDeviceModuleDefault() {}
>+  virtual ~AudioDeviceModuleDefault() {}
>+
>+  int32_t RegisterAudioCallback(AudioTransport* audioCallback) override {
>+    return 0;
>+  }
>+  int32_t Init() override { return 0; }
>+  int32_t InitSpeaker() override { return 0; }
>+  int32_t SetPlayoutDevice(uint16_t index) override { return 0; }
>+  int32_t SetPlayoutDevice(
>+      AudioDeviceModule::WindowsDeviceType device) override {
>+    return 0;
>+  }
>+  int32_t SetStereoPlayout(bool enable) override { return 0; }
>+  int32_t StopPlayout() override { return 0; }
>+  int32_t InitMicrophone() override { return 0; }
>+  int32_t SetRecordingDevice(uint16_t index) override { return 0; }
>+  int32_t SetRecordingDevice(
>+      AudioDeviceModule::WindowsDeviceType device) override {
>+    return 0;
>+  }
>+  int32_t SetStereoRecording(bool enable) override { return 0; }
>+  int32_t StopRecording() override { return 0; }
>+
>+  int32_t Terminate() override { return 0; }
>+
>+  int32_t ActiveAudioLayer(
>+      AudioDeviceModule::AudioLayer* audioLayer) const override {
>+    return 0;
>+  }
>+  bool Initialized() const override { return true; }
>+  int16_t PlayoutDevices() override { return 0; }
>+  int16_t RecordingDevices() override { return 0; }
>+  int32_t PlayoutDeviceName(uint16_t index,
>+                            char name[kAdmMaxDeviceNameSize],
>+                            char guid[kAdmMaxGuidSize]) override {
>+    return 0;
>+  }
>+  int32_t RecordingDeviceName(uint16_t index,
>+                              char name[kAdmMaxDeviceNameSize],
>+                              char guid[kAdmMaxGuidSize]) override {
>+    return 0;
>+  }
>+  int32_t PlayoutIsAvailable(bool* available) override { return 0; }
>+  int32_t InitPlayout() override { return 0; }
>+  bool PlayoutIsInitialized() const override { return true; }
>+  int32_t RecordingIsAvailable(bool* available) override { return 0; }
>+  int32_t InitRecording() override { return 0; }
>+  bool RecordingIsInitialized() const override { return true; }
>+  int32_t StartPlayout() override { return 0; }
>+  bool Playing() const override { return false; }
>+  int32_t StartRecording() override { return 0; }
>+  bool Recording() const override { return false; }
>+  bool SpeakerIsInitialized() const override { return true; }
>+  bool MicrophoneIsInitialized() const override { return true; }
>+  int32_t SpeakerVolumeIsAvailable(bool* available) override { return 0; }
>+  int32_t SetSpeakerVolume(uint32_t volume) override { return 0; }
>+  int32_t SpeakerVolume(uint32_t* volume) const override { return 0; }
>+  int32_t MaxSpeakerVolume(uint32_t* maxVolume) const override { return 0; }
>+  int32_t MinSpeakerVolume(uint32_t* minVolume) const override { return 0; }
>+  int32_t MicrophoneVolumeIsAvailable(bool* available) override { return 0; }
>+  int32_t SetMicrophoneVolume(uint32_t volume) override { return 0; }
>+  int32_t MicrophoneVolume(uint32_t* volume) const override { return 0; }
>+  int32_t MaxMicrophoneVolume(uint32_t* maxVolume) const override { return 0; }
>+  int32_t MinMicrophoneVolume(uint32_t* minVolume) const override { return 0; }
>+  int32_t SpeakerMuteIsAvailable(bool* available) override { return 0; }
>+  int32_t SetSpeakerMute(bool enable) override { return 0; }
>+  int32_t SpeakerMute(bool* enabled) const override { return 0; }
>+  int32_t MicrophoneMuteIsAvailable(bool* available) override { return 0; }
>+  int32_t SetMicrophoneMute(bool enable) override { return 0; }
>+  int32_t MicrophoneMute(bool* enabled) const override { return 0; }
>+  int32_t StereoPlayoutIsAvailable(bool* available) const override {
>+    *available = false;
>+    return 0;
>+  }
>+  int32_t StereoPlayout(bool* enabled) const override { return 0; }
>+  int32_t StereoRecordingIsAvailable(bool* available) const override {
>+    *available = false;
>+    return 0;
>+  }
>+  int32_t StereoRecording(bool* enabled) const override { return 0; }
>+  int32_t PlayoutDelay(uint16_t* delayMS) const override {
>+    *delayMS = 0;
>+    return 0;
>+  }
>+  bool BuiltInAECIsAvailable() const override { return false; }
>+  int32_t EnableBuiltInAEC(bool enable) override { return -1; }
>+  bool BuiltInAGCIsAvailable() const override { return false; }
>+  int32_t EnableBuiltInAGC(bool enable) override { return -1; }
>+  bool BuiltInNSIsAvailable() const override { return false; }
>+  int32_t EnableBuiltInNS(bool enable) override { return -1; }
>+
>+#if defined(WEBRTC_IOS)
>+  int GetPlayoutAudioParameters(AudioParameters* params) const override {
>+    return -1;
>+  }
>+  int GetRecordAudioParameters(AudioParameters* params) const override {
>+    return -1;
>+  }
>+#endif  // WEBRTC_IOS
>+};
>+
>+}  // namespace webrtc_impl
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_DEFAULT_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/audio_device_defines.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/audio_device_defines.h
>index 9bf5d1a7157..f488ab6e0ff 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/audio_device_defines.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/audio_device_defines.h
>@@ -12,10 +12,11 @@
> #define MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_DEFINES_H_
> 
> #include <stddef.h>
>+#include <string>
> 
> #include "rtc_base/checks.h"
> #include "rtc_base/deprecation.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include "rtc_base/strings/string_builder.h"
> 
> namespace webrtc {
> 
>@@ -41,31 +42,17 @@ class AudioTransport {
>                                           const int32_t clockDrift,
>                                           const uint32_t currentMicLevel,
>                                           const bool keyPressed,
>-                                          uint32_t& newMicLevel) = 0;
>+                                          uint32_t& newMicLevel) = 0;  // NOLINT
> 
>+  // Implementation has to setup safe values for all specified out parameters.
>   virtual int32_t NeedMorePlayData(const size_t nSamples,
>                                    const size_t nBytesPerSample,
>                                    const size_t nChannels,
>                                    const uint32_t samplesPerSec,
>                                    void* audioSamples,
>-                                   size_t& nSamplesOut,
>+                                   size_t& nSamplesOut,  // NOLINT
>                                    int64_t* elapsed_time_ms,
>-                                   int64_t* ntp_time_ms) = 0;
>-
>-  // Method to push the captured audio data to the specific VoE channel.
>-  // The data will not undergo audio processing.
>-  // |voe_channel| is the id of the VoE channel which is the sink to the
>-  // capture data.
>-  // TODO(bugs.webrtc.org/8659): Remove this method once clients updated.
>-  RTC_DEPRECATED virtual void PushCaptureData(
>-      int /* voe_channel */,
>-      const void* /* audio_data */,
>-      int /* bits_per_sample */,
>-      int /* sample_rate */,
>-      size_t /* number_of_channels */,
>-      size_t /* number_of_frames */) {
>-    RTC_NOTREACHED();
>-  }
>+                                   int64_t* ntp_time_ms) = 0;  // NOLINT
> 
>   // Method to pull mixed render audio data from all active VoE channels.
>   // The data will not be passed as reference for audio processing internally.
>@@ -143,6 +130,19 @@ class AudioParameters {
>       return 0.0;
>     return static_cast<double>(frames_per_buffer_) / (sample_rate_);
>   }
>+  std::string ToString() const {
>+    char ss_buf[1024];
>+    rtc::SimpleStringBuilder ss(ss_buf);
>+    ss << "AudioParameters: ";
>+    ss << "sample_rate=" << sample_rate() << ", channels=" << channels();
>+    ss << ", frames_per_buffer=" << frames_per_buffer();
>+    ss << ", frames_per_10ms_buffer=" << frames_per_10ms_buffer();
>+    ss << ", bytes_per_frame=" << GetBytesPerFrame();
>+    ss << ", bytes_per_buffer=" << GetBytesPerBuffer();
>+    ss << ", bytes_per_10ms_buffer=" << GetBytesPer10msBuffer();
>+    ss << ", size_in_ms=" << GetBufferSizeInMilliseconds();
>+    return ss.str();
>+  }
> 
>  private:
>   int sample_rate_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/audio_device_factory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/audio_device_factory.cc
>new file mode 100644
>index 00000000000..abcaa212e23
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/audio_device_factory.cc
>@@ -0,0 +1,40 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_device/include/audio_device_factory.h"
>+
>+#if defined(WEBRTC_WIN)
>+#include "modules/audio_device/win/audio_device_module_win.h"
>+#include "modules/audio_device/win/core_audio_input_win.h"
>+#include "modules/audio_device/win/core_audio_output_win.h"
>+#include "modules/audio_device/win/core_audio_utility_win.h"
>+#endif
>+
>+#include "absl/memory/memory.h"
>+#include "rtc_base/logging.h"
>+
>+namespace webrtc {
>+
>+rtc::scoped_refptr<AudioDeviceModule>
>+CreateWindowsCoreAudioAudioDeviceModule() {
>+  RTC_DLOG(INFO) << __FUNCTION__;
>+  // Returns NULL if Core Audio is not supported or if COM has not been
>+  // initialized correctly using webrtc_win::ScopedCOMInitializer.
>+  if (!webrtc_win::core_audio_utility::IsSupported()) {
>+    RTC_LOG(LS_ERROR)
>+        << "Unable to create ADM since Core Audio is not supported";
>+    return nullptr;
>+  }
>+  return CreateWindowsCoreAudioAudioDeviceModuleFromInputAndOutput(
>+      absl::make_unique<webrtc_win::CoreAudioInput>(),
>+      absl::make_unique<webrtc_win::CoreAudioOutput>());
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/audio_device_factory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/audio_device_factory.h
>new file mode 100644
>index 00000000000..4297988a130
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/audio_device_factory.h
>@@ -0,0 +1,40 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_FACTORY_H_
>+#define MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_FACTORY_H_
>+
>+#include "modules/audio_device/include/audio_device.h"
>+
>+namespace webrtc {
>+
>+// Creates an AudioDeviceModule (ADM) for Windows based on the Core Audio API.
>+// The creating thread must be a COM thread; otherwise nullptr will be returned.
>+// Example (assuming webrtc namespace):
>+//
>+//  public:
>+//   rtc::scoped_refptr<AudioDeviceModule> CreateAudioDevice() {
>+//     // Tell COM that this thread shall live in the MTA.
>+//     com_initializer_ = absl::make_unique<webrtc_win::ScopedCOMInitializer>(
>+//         webrtc_win::ScopedCOMInitializer::kMTA);
>+//     if (!com_initializer_->Succeeded()) {
>+//       return nullptr;
>+//     }
>+//     return CreateWindowsCoreAudioAudioDeviceModule();
>+//   }
>+//
>+//   private:
>+//    std::unique_ptr<webrtc_win::ScopedCOMInitializer> com_initializer_;
>+//
>+rtc::scoped_refptr<AudioDeviceModule> CreateWindowsCoreAudioAudioDeviceModule();
>+
>+}  // namespace webrtc
>+
>+#endif  //  MODULES_AUDIO_DEVICE_INCLUDE_AUDIO_DEVICE_FACTORY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/fake_audio_device.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/fake_audio_device.h
>index f66cbfdadfd..9949627a733 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/fake_audio_device.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/fake_audio_device.h
>@@ -12,110 +12,19 @@
> #define MODULES_AUDIO_DEVICE_INCLUDE_FAKE_AUDIO_DEVICE_H_
> 
> #include "modules/audio_device/include/audio_device.h"
>+#include "modules/audio_device/include/audio_device_default.h"
> 
> namespace webrtc {
> 
>-class FakeAudioDeviceModule : public AudioDeviceModule {
>+class FakeAudioDeviceModule
>+    : public webrtc_impl::AudioDeviceModuleDefault<AudioDeviceModule> {
>  public:
>-  FakeAudioDeviceModule() {}
>-  virtual ~FakeAudioDeviceModule() {}
>-
>   // TODO(nisse): Fix all users of this class to managed references using
>   // scoped_refptr. Current code doesn't always use refcounting for this class.
>   void AddRef() const override {}
>   rtc::RefCountReleaseStatus Release() const override {
>     return rtc::RefCountReleaseStatus::kDroppedLastRef;
>   }
>-
>- private:
>-  int32_t RegisterAudioCallback(AudioTransport* audioCallback) override {
>-    return 0;
>-  }
>-  int32_t Init() override { return 0; }
>-  int32_t InitSpeaker() override { return 0; }
>-  int32_t SetPlayoutDevice(uint16_t index) override { return 0; }
>-  int32_t SetPlayoutDevice(WindowsDeviceType device) override { return 0; }
>-  int32_t SetStereoPlayout(bool enable) override { return 0; }
>-  int32_t StopPlayout() override { return 0; }
>-  int32_t InitMicrophone() override { return 0; }
>-  int32_t SetRecordingDevice(uint16_t index) override { return 0; }
>-  int32_t SetRecordingDevice(WindowsDeviceType device) override { return 0; }
>-  int32_t SetStereoRecording(bool enable) override { return 0; }
>-  int32_t StopRecording() override { return 0; }
>-
>-  int32_t Terminate() override { return 0; }
>-
>-  int32_t ActiveAudioLayer(AudioLayer* audioLayer) const override { return 0; }
>-  bool Initialized() const override { return true; }
>-  int16_t PlayoutDevices() override { return 0; }
>-  int16_t RecordingDevices() override { return 0; }
>-  int32_t PlayoutDeviceName(uint16_t index,
>-                            char name[kAdmMaxDeviceNameSize],
>-                            char guid[kAdmMaxGuidSize]) override {
>-    return 0;
>-  }
>-  int32_t RecordingDeviceName(uint16_t index,
>-                              char name[kAdmMaxDeviceNameSize],
>-                              char guid[kAdmMaxGuidSize]) override {
>-    return 0;
>-  }
>-  int32_t PlayoutIsAvailable(bool* available) override { return 0; }
>-  int32_t InitPlayout() override { return 0; }
>-  bool PlayoutIsInitialized() const override { return true; }
>-  int32_t RecordingIsAvailable(bool* available) override { return 0; }
>-  int32_t InitRecording() override { return 0; }
>-  bool RecordingIsInitialized() const override { return true; }
>-  int32_t StartPlayout() override { return 0; }
>-  bool Playing() const override { return false; }
>-  int32_t StartRecording() override { return 0; }
>-  bool Recording() const override { return false; }
>-  bool SpeakerIsInitialized() const override { return true; }
>-  bool MicrophoneIsInitialized() const override { return true; }
>-  int32_t SpeakerVolumeIsAvailable(bool* available) override { return 0; }
>-  int32_t SetSpeakerVolume(uint32_t volume) override { return 0; }
>-  int32_t SpeakerVolume(uint32_t* volume) const override { return 0; }
>-  int32_t MaxSpeakerVolume(uint32_t* maxVolume) const override { return 0; }
>-  int32_t MinSpeakerVolume(uint32_t* minVolume) const override { return 0; }
>-  int32_t MicrophoneVolumeIsAvailable(bool* available) override { return 0; }
>-  int32_t SetMicrophoneVolume(uint32_t volume) override { return 0; }
>-  int32_t MicrophoneVolume(uint32_t* volume) const override { return 0; }
>-  int32_t MaxMicrophoneVolume(uint32_t* maxVolume) const override { return 0; }
>-  int32_t MinMicrophoneVolume(uint32_t* minVolume) const override { return 0; }
>-  int32_t SpeakerMuteIsAvailable(bool* available) override { return 0; }
>-  int32_t SetSpeakerMute(bool enable) override { return 0; }
>-  int32_t SpeakerMute(bool* enabled) const override { return 0; }
>-  int32_t MicrophoneMuteIsAvailable(bool* available) override { return 0; }
>-  int32_t SetMicrophoneMute(bool enable) override { return 0; }
>-  int32_t MicrophoneMute(bool* enabled) const override { return 0; }
>-  int32_t StereoPlayoutIsAvailable(bool* available) const override {
>-    *available = false;
>-    return 0;
>-  }
>-  int32_t StereoPlayout(bool* enabled) const override { return 0; }
>-  int32_t StereoRecordingIsAvailable(bool* available) const override {
>-    *available = false;
>-    return 0;
>-  }
>-  int32_t StereoRecording(bool* enabled) const override { return 0; }
>-  int32_t PlayoutDelay(uint16_t* delayMS) const override {
>-    *delayMS = 0;
>-    return 0;
>-  }
>-  bool BuiltInAECIsAvailable() const override { return false; }
>-  int32_t EnableBuiltInAEC(bool enable) override { return -1; }
>-  bool BuiltInAGCIsAvailable() const override { return false; }
>-  int32_t EnableBuiltInAGC(bool enable) override { return -1; }
>-  bool BuiltInNSIsAvailable() const override { return false; }
>-  int32_t EnableBuiltInNS(bool enable) override { return -1; }
>-
>-#if defined(WEBRTC_IOS)
>-  int GetPlayoutAudioParameters(AudioParameters* params) const override {
>-    return -1;
>-  }
>-  int GetRecordAudioParameters(AudioParameters* params) const override {
>-    return -1;
>-  }
>-#endif  // WEBRTC_IOS
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/mock_audio_device.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/mock_audio_device.h
>index 7984e0b5bd6..105442b8079 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/mock_audio_device.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/mock_audio_device.h
>@@ -33,12 +33,14 @@ class MockAudioDeviceModule : public AudioDeviceModule {
>   MOCK_CONST_METHOD0(Initialized, bool());
>   MOCK_METHOD0(PlayoutDevices, int16_t());
>   MOCK_METHOD0(RecordingDevices, int16_t());
>-  MOCK_METHOD3(PlayoutDeviceName, int32_t(uint16_t index,
>-                                          char name[kAdmMaxDeviceNameSize],
>-                                          char guid[kAdmMaxGuidSize]));
>-  MOCK_METHOD3(RecordingDeviceName, int32_t(uint16_t index,
>-                                            char name[kAdmMaxDeviceNameSize],
>-                                            char guid[kAdmMaxGuidSize]));
>+  MOCK_METHOD3(PlayoutDeviceName,
>+               int32_t(uint16_t index,
>+                       char name[kAdmMaxDeviceNameSize],
>+                       char guid[kAdmMaxGuidSize]));
>+  MOCK_METHOD3(RecordingDeviceName,
>+               int32_t(uint16_t index,
>+                       char name[kAdmMaxDeviceNameSize],
>+                       char guid[kAdmMaxGuidSize]));
>   MOCK_METHOD1(SetPlayoutDevice, int32_t(uint16_t index));
>   MOCK_METHOD1(SetPlayoutDevice, int32_t(WindowsDeviceType device));
>   MOCK_METHOD1(SetRecordingDevice, int32_t(uint16_t index));
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/mock_audio_transport.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/mock_audio_transport.h
>index baa0227b391..ebdfbc70154 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/mock_audio_transport.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/mock_audio_transport.h
>@@ -44,14 +44,6 @@ class MockAudioTransport : public AudioTransport {
>                        int64_t* elapsed_time_ms,
>                        int64_t* ntp_time_ms));
> 
>-  MOCK_METHOD6(PushCaptureData,
>-               void(int voe_channel,
>-                    const void* audio_data,
>-                    int bits_per_sample,
>-                    int sample_rate,
>-                    size_t number_of_channels,
>-                    size_t number_of_frames));
>-
>   MOCK_METHOD7(PullRenderData,
>                void(int bits_per_sample,
>                     int sample_rate,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/test_audio_device.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/test_audio_device.cc
>new file mode 100644
>index 00000000000..7d18d825a1d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/test_audio_device.cc
>@@ -0,0 +1,560 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#include <algorithm>
>+#include <memory>
>+#include <string>
>+#include <utility>
>+#include <vector>
>+
>+#include "absl/memory/memory.h"
>+#include "api/array_view.h"
>+#include "common_audio/wav_file.h"
>+#include "modules/audio_device/include/audio_device_default.h"
>+#include "modules/audio_device/include/test_audio_device.h"
>+#include "rtc_base/buffer.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/criticalsection.h"
>+#include "rtc_base/event.h"
>+#include "rtc_base/numerics/safe_conversions.h"
>+#include "rtc_base/platform_thread.h"
>+#include "rtc_base/random.h"
>+#include "rtc_base/refcountedobject.h"
>+#include "system_wrappers/include/event_wrapper.h"
>+
>+namespace webrtc {
>+
>+class EventTimerWrapper;
>+
>+namespace {
>+
>+constexpr int kFrameLengthMs = 10;
>+constexpr int kFramesPerSecond = 1000 / kFrameLengthMs;
>+
>+// TestAudioDeviceModule implements an AudioDevice module that can act both as a
>+// capturer and a renderer. It will use 10ms audio frames.
>+class TestAudioDeviceModuleImpl
>+    : public webrtc_impl::AudioDeviceModuleDefault<TestAudioDeviceModule> {
>+ public:
>+  // Creates a new TestAudioDeviceModule. When capturing or playing, 10 ms audio
>+  // frames will be processed every 10ms / |speed|.
>+  // |capturer| is an object that produces audio data. Can be nullptr if this
>+  // device is never used for recording.
>+  // |renderer| is an object that receives audio data that would have been
>+  // played out. Can be nullptr if this device is never used for playing.
>+  // Use one of the Create... functions to get these instances.
>+  TestAudioDeviceModuleImpl(std::unique_ptr<Capturer> capturer,
>+                            std::unique_ptr<Renderer> renderer,
>+                            float speed = 1)
>+      : capturer_(std::move(capturer)),
>+        renderer_(std::move(renderer)),
>+        speed_(speed),
>+        audio_callback_(nullptr),
>+        rendering_(false),
>+        capturing_(false),
>+        done_rendering_(true, true),
>+        done_capturing_(true, true),
>+        tick_(EventTimerWrapper::Create()) {
>+    auto good_sample_rate = [](int sr) {
>+      return sr == 8000 || sr == 16000 || sr == 32000 || sr == 44100 ||
>+             sr == 48000;
>+    };
>+
>+    if (renderer_) {
>+      const int sample_rate = renderer_->SamplingFrequency();
>+      playout_buffer_.resize(
>+          SamplesPerFrame(sample_rate) * renderer_->NumChannels(), 0);
>+      RTC_CHECK(good_sample_rate(sample_rate));
>+    }
>+    if (capturer_) {
>+      RTC_CHECK(good_sample_rate(capturer_->SamplingFrequency()));
>+    }
>+  }
>+
>+  ~TestAudioDeviceModuleImpl() {
>+    StopPlayout();
>+    StopRecording();
>+    if (thread_) {
>+      thread_->Stop();
>+    }
>+  }
>+
>+  int32_t Init() {
>+    RTC_CHECK(tick_->StartTimer(true, kFrameLengthMs / speed_));
>+    thread_ = absl::make_unique<rtc::PlatformThread>(
>+        TestAudioDeviceModuleImpl::Run, this, "TestAudioDeviceModuleImpl");
>+    thread_->Start();
>+    thread_->SetPriority(rtc::kHighPriority);
>+    return 0;
>+  }
>+
>+  int32_t RegisterAudioCallback(AudioTransport* callback) {
>+    rtc::CritScope cs(&lock_);
>+    RTC_DCHECK(callback || audio_callback_);
>+    audio_callback_ = callback;
>+    return 0;
>+  }
>+
>+  int32_t StartPlayout() {
>+    rtc::CritScope cs(&lock_);
>+    RTC_CHECK(renderer_);
>+    rendering_ = true;
>+    done_rendering_.Reset();
>+    return 0;
>+  }
>+
>+  int32_t StopPlayout() {
>+    rtc::CritScope cs(&lock_);
>+    rendering_ = false;
>+    done_rendering_.Set();
>+    return 0;
>+  }
>+
>+  int32_t StartRecording() {
>+    rtc::CritScope cs(&lock_);
>+    RTC_CHECK(capturer_);
>+    capturing_ = true;
>+    done_capturing_.Reset();
>+    return 0;
>+  }
>+
>+  int32_t StopRecording() {
>+    rtc::CritScope cs(&lock_);
>+    capturing_ = false;
>+    done_capturing_.Set();
>+    return 0;
>+  }
>+
>+  bool Playing() const {
>+    rtc::CritScope cs(&lock_);
>+    return rendering_;
>+  }
>+
>+  bool Recording() const {
>+    rtc::CritScope cs(&lock_);
>+    return capturing_;
>+  }
>+
>+  // Blocks until the Renderer refuses to receive data.
>+  // Returns false if |timeout_ms| passes before that happens.
>+  bool WaitForPlayoutEnd(int timeout_ms = rtc::Event::kForever) {
>+    return done_rendering_.Wait(timeout_ms);
>+  }
>+
>+  // Blocks until the Recorder stops producing data.
>+  // Returns false if |timeout_ms| passes before that happens.
>+  bool WaitForRecordingEnd(int timeout_ms = rtc::Event::kForever) {
>+    return done_capturing_.Wait(timeout_ms);
>+  }
>+
>+ private:
>+  void ProcessAudio() {
>+    {
>+      rtc::CritScope cs(&lock_);
>+      if (capturing_) {
>+        // Capture 10ms of audio. 2 bytes per sample.
>+        const bool keep_capturing = capturer_->Capture(&recording_buffer_);
>+        uint32_t new_mic_level = 0;
>+        if (recording_buffer_.size() > 0) {
>+          audio_callback_->RecordedDataIsAvailable(
>+              recording_buffer_.data(), recording_buffer_.size(), 2,
>+              capturer_->NumChannels(), capturer_->SamplingFrequency(), 0, 0, 0,
>+              false, new_mic_level);
>+        }
>+        if (!keep_capturing) {
>+          capturing_ = false;
>+          done_capturing_.Set();
>+        }
>+      }
>+      if (rendering_) {
>+        size_t samples_out = 0;
>+        int64_t elapsed_time_ms = -1;
>+        int64_t ntp_time_ms = -1;
>+        const int sampling_frequency = renderer_->SamplingFrequency();
>+        audio_callback_->NeedMorePlayData(
>+            SamplesPerFrame(sampling_frequency), 2, renderer_->NumChannels(),
>+            sampling_frequency, playout_buffer_.data(), samples_out,
>+            &elapsed_time_ms, &ntp_time_ms);
>+        const bool keep_rendering = renderer_->Render(
>+            rtc::ArrayView<const int16_t>(playout_buffer_.data(), samples_out));
>+        if (!keep_rendering) {
>+          rendering_ = false;
>+          done_rendering_.Set();
>+        }
>+      }
>+    }
>+    tick_->Wait(WEBRTC_EVENT_INFINITE);
>+  }
>+
>+  static bool Run(void* obj) {
>+    static_cast<TestAudioDeviceModuleImpl*>(obj)->ProcessAudio();
>+    return true;
>+  }
>+
>+  const std::unique_ptr<Capturer> capturer_ RTC_GUARDED_BY(lock_);
>+  const std::unique_ptr<Renderer> renderer_ RTC_GUARDED_BY(lock_);
>+  const float speed_;
>+
>+  rtc::CriticalSection lock_;
>+  AudioTransport* audio_callback_ RTC_GUARDED_BY(lock_);
>+  bool rendering_ RTC_GUARDED_BY(lock_);
>+  bool capturing_ RTC_GUARDED_BY(lock_);
>+  rtc::Event done_rendering_;
>+  rtc::Event done_capturing_;
>+
>+  std::vector<int16_t> playout_buffer_ RTC_GUARDED_BY(lock_);
>+  rtc::BufferT<int16_t> recording_buffer_ RTC_GUARDED_BY(lock_);
>+
>+  std::unique_ptr<EventTimerWrapper> tick_;
>+  std::unique_ptr<rtc::PlatformThread> thread_;
>+};
>+
>+// A fake capturer that generates pulses with random samples between
>+// -max_amplitude and +max_amplitude.
>+class PulsedNoiseCapturerImpl final
>+    : public TestAudioDeviceModule::PulsedNoiseCapturer {
>+ public:
>+  // Assuming 10ms audio packets.
>+  PulsedNoiseCapturerImpl(int16_t max_amplitude,
>+                          int sampling_frequency_in_hz,
>+                          int num_channels)
>+      : sampling_frequency_in_hz_(sampling_frequency_in_hz),
>+        fill_with_zero_(false),
>+        random_generator_(1),
>+        max_amplitude_(max_amplitude),
>+        num_channels_(num_channels) {
>+    RTC_DCHECK_GT(max_amplitude, 0);
>+  }
>+
>+  int SamplingFrequency() const override { return sampling_frequency_in_hz_; }
>+
>+  int NumChannels() const override { return num_channels_; }
>+
>+  bool Capture(rtc::BufferT<int16_t>* buffer) override {
>+    fill_with_zero_ = !fill_with_zero_;
>+    int16_t max_amplitude;
>+    {
>+      rtc::CritScope cs(&lock_);
>+      max_amplitude = max_amplitude_;
>+    }
>+    buffer->SetData(
>+        TestAudioDeviceModule::SamplesPerFrame(sampling_frequency_in_hz_) *
>+            num_channels_,
>+        [&](rtc::ArrayView<int16_t> data) {
>+          if (fill_with_zero_) {
>+            std::fill(data.begin(), data.end(), 0);
>+          } else {
>+            std::generate(data.begin(), data.end(), [&]() {
>+              return random_generator_.Rand(-max_amplitude, max_amplitude);
>+            });
>+          }
>+          return data.size();
>+        });
>+    return true;
>+  }
>+
>+  void SetMaxAmplitude(int16_t amplitude) override {
>+    rtc::CritScope cs(&lock_);
>+    max_amplitude_ = amplitude;
>+  }
>+
>+ private:
>+  int sampling_frequency_in_hz_;
>+  bool fill_with_zero_;
>+  Random random_generator_;
>+  rtc::CriticalSection lock_;
>+  int16_t max_amplitude_ RTC_GUARDED_BY(lock_);
>+  const int num_channels_;
>+};
>+
>+class WavFileReader final : public TestAudioDeviceModule::Capturer {
>+ public:
>+  WavFileReader(std::string filename,
>+                int sampling_frequency_in_hz,
>+                int num_channels)
>+      : WavFileReader(absl::make_unique<WavReader>(filename),
>+                      sampling_frequency_in_hz,
>+                      num_channels) {}
>+
>+  WavFileReader(rtc::PlatformFile file,
>+                int sampling_frequency_in_hz,
>+                int num_channels)
>+      : WavFileReader(absl::make_unique<WavReader>(file),
>+                      sampling_frequency_in_hz,
>+                      num_channels) {}
>+
>+  int SamplingFrequency() const override { return sampling_frequency_in_hz_; }
>+
>+  int NumChannels() const override { return num_channels_; }
>+
>+  bool Capture(rtc::BufferT<int16_t>* buffer) override {
>+    buffer->SetData(
>+        TestAudioDeviceModule::SamplesPerFrame(sampling_frequency_in_hz_) *
>+            num_channels_,
>+        [&](rtc::ArrayView<int16_t> data) {
>+          return wav_reader_->ReadSamples(data.size(), data.data());
>+        });
>+    return buffer->size() > 0;
>+  }
>+
>+ private:
>+  WavFileReader(std::unique_ptr<WavReader> wav_reader,
>+                int sampling_frequency_in_hz,
>+                int num_channels)
>+      : sampling_frequency_in_hz_(sampling_frequency_in_hz),
>+        num_channels_(num_channels),
>+        wav_reader_(std::move(wav_reader)) {
>+    RTC_CHECK_EQ(wav_reader_->sample_rate(), sampling_frequency_in_hz);
>+    RTC_CHECK_EQ(wav_reader_->num_channels(), num_channels);
>+  }
>+
>+  int sampling_frequency_in_hz_;
>+  const int num_channels_;
>+  std::unique_ptr<WavReader> wav_reader_;
>+};
>+
>+class WavFileWriter final : public TestAudioDeviceModule::Renderer {
>+ public:
>+  WavFileWriter(std::string filename,
>+                int sampling_frequency_in_hz,
>+                int num_channels)
>+      : WavFileWriter(absl::make_unique<WavWriter>(filename,
>+                                                   sampling_frequency_in_hz,
>+                                                   num_channels),
>+                      sampling_frequency_in_hz,
>+                      num_channels) {}
>+
>+  WavFileWriter(rtc::PlatformFile file,
>+                int sampling_frequency_in_hz,
>+                int num_channels)
>+      : WavFileWriter(absl::make_unique<WavWriter>(file,
>+                                                   sampling_frequency_in_hz,
>+                                                   num_channels),
>+                      sampling_frequency_in_hz,
>+                      num_channels) {}
>+
>+  int SamplingFrequency() const override { return sampling_frequency_in_hz_; }
>+
>+  int NumChannels() const override { return num_channels_; }
>+
>+  bool Render(rtc::ArrayView<const int16_t> data) override {
>+    wav_writer_->WriteSamples(data.data(), data.size());
>+    return true;
>+  }
>+
>+ private:
>+  WavFileWriter(std::unique_ptr<WavWriter> wav_writer,
>+                int sampling_frequency_in_hz,
>+                int num_channels)
>+      : sampling_frequency_in_hz_(sampling_frequency_in_hz),
>+        wav_writer_(std::move(wav_writer)),
>+        num_channels_(num_channels) {}
>+
>+  int sampling_frequency_in_hz_;
>+  std::unique_ptr<WavWriter> wav_writer_;
>+  const int num_channels_;
>+};
>+
>+class BoundedWavFileWriter : public TestAudioDeviceModule::Renderer {
>+ public:
>+  BoundedWavFileWriter(std::string filename,
>+                       int sampling_frequency_in_hz,
>+                       int num_channels)
>+      : sampling_frequency_in_hz_(sampling_frequency_in_hz),
>+        wav_writer_(filename, sampling_frequency_in_hz, num_channels),
>+        num_channels_(num_channels),
>+        silent_audio_(
>+            TestAudioDeviceModule::SamplesPerFrame(sampling_frequency_in_hz) *
>+                num_channels,
>+            0),
>+        started_writing_(false),
>+        trailing_zeros_(0) {}
>+
>+  BoundedWavFileWriter(rtc::PlatformFile file,
>+                       int sampling_frequency_in_hz,
>+                       int num_channels)
>+      : sampling_frequency_in_hz_(sampling_frequency_in_hz),
>+        wav_writer_(file, sampling_frequency_in_hz, num_channels),
>+        num_channels_(num_channels),
>+        silent_audio_(
>+            TestAudioDeviceModule::SamplesPerFrame(sampling_frequency_in_hz) *
>+                num_channels,
>+            0),
>+        started_writing_(false),
>+        trailing_zeros_(0) {}
>+
>+  int SamplingFrequency() const override { return sampling_frequency_in_hz_; }
>+
>+  int NumChannels() const override { return num_channels_; }
>+
>+  bool Render(rtc::ArrayView<const int16_t> data) override {
>+    const int16_t kAmplitudeThreshold = 5;
>+
>+    const int16_t* begin = data.begin();
>+    const int16_t* end = data.end();
>+    if (!started_writing_) {
>+      // Cut off silence at the beginning.
>+      while (begin < end) {
>+        if (std::abs(*begin) > kAmplitudeThreshold) {
>+          started_writing_ = true;
>+          break;
>+        }
>+        ++begin;
>+      }
>+    }
>+    if (started_writing_) {
>+      // Cut off silence at the end.
>+      while (begin < end) {
>+        if (*(end - 1) != 0) {
>+          break;
>+        }
>+        --end;
>+      }
>+      if (begin < end) {
>+        // If it turns out that the silence was not final, need to write all the
>+        // skipped zeros and continue writing audio.
>+        while (trailing_zeros_ > 0) {
>+          const size_t zeros_to_write =
>+              std::min(trailing_zeros_, silent_audio_.size());
>+          wav_writer_.WriteSamples(silent_audio_.data(), zeros_to_write);
>+          trailing_zeros_ -= zeros_to_write;
>+        }
>+        wav_writer_.WriteSamples(begin, end - begin);
>+      }
>+      // Save the number of zeros we skipped in case this needs to be restored.
>+      trailing_zeros_ += data.end() - end;
>+    }
>+    return true;
>+  }
>+
>+ private:
>+  int sampling_frequency_in_hz_;
>+  WavWriter wav_writer_;
>+  const int num_channels_;
>+  std::vector<int16_t> silent_audio_;
>+  bool started_writing_;
>+  size_t trailing_zeros_;
>+};
>+
>+class DiscardRenderer final : public TestAudioDeviceModule::Renderer {
>+ public:
>+  explicit DiscardRenderer(int sampling_frequency_in_hz, int num_channels)
>+      : sampling_frequency_in_hz_(sampling_frequency_in_hz),
>+        num_channels_(num_channels) {}
>+
>+  int SamplingFrequency() const override { return sampling_frequency_in_hz_; }
>+
>+  int NumChannels() const override { return num_channels_; }
>+
>+  bool Render(rtc::ArrayView<const int16_t> data) override { return true; }
>+
>+ private:
>+  int sampling_frequency_in_hz_;
>+  const int num_channels_;
>+};
>+
>+}  // namespace
>+
>+size_t TestAudioDeviceModule::SamplesPerFrame(int sampling_frequency_in_hz) {
>+  return rtc::CheckedDivExact(sampling_frequency_in_hz, kFramesPerSecond);
>+}
>+
>+rtc::scoped_refptr<TestAudioDeviceModule>
>+TestAudioDeviceModule::CreateTestAudioDeviceModule(
>+    std::unique_ptr<Capturer> capturer,
>+    std::unique_ptr<Renderer> renderer,
>+    float speed) {
>+  return new rtc::RefCountedObject<TestAudioDeviceModuleImpl>(
>+      std::move(capturer), std::move(renderer), speed);
>+}
>+
>+std::unique_ptr<TestAudioDeviceModule::PulsedNoiseCapturer>
>+TestAudioDeviceModule::CreatePulsedNoiseCapturer(int16_t max_amplitude,
>+                                                 int sampling_frequency_in_hz,
>+                                                 int num_channels) {
>+  return std::unique_ptr<TestAudioDeviceModule::PulsedNoiseCapturer>(
>+      new PulsedNoiseCapturerImpl(max_amplitude, sampling_frequency_in_hz,
>+                                  num_channels));
>+}
>+
>+std::unique_ptr<TestAudioDeviceModule::Renderer>
>+TestAudioDeviceModule::CreateDiscardRenderer(int sampling_frequency_in_hz,
>+                                             int num_channels) {
>+  return std::unique_ptr<TestAudioDeviceModule::Renderer>(
>+      new DiscardRenderer(sampling_frequency_in_hz, num_channels));
>+}
>+
>+std::unique_ptr<TestAudioDeviceModule::Capturer>
>+TestAudioDeviceModule::CreateWavFileReader(std::string filename,
>+                                           int sampling_frequency_in_hz,
>+                                           int num_channels) {
>+  return std::unique_ptr<TestAudioDeviceModule::Capturer>(
>+      new WavFileReader(filename, sampling_frequency_in_hz, num_channels));
>+}
>+
>+std::unique_ptr<TestAudioDeviceModule::Capturer>
>+TestAudioDeviceModule::CreateWavFileReader(std::string filename) {
>+  WavReader reader(filename);
>+  int sampling_frequency_in_hz = reader.sample_rate();
>+  int num_channels = rtc::checked_cast<int>(reader.num_channels());
>+  return std::unique_ptr<TestAudioDeviceModule::Capturer>(
>+      new WavFileReader(filename, sampling_frequency_in_hz, num_channels));
>+}
>+
>+std::unique_ptr<TestAudioDeviceModule::Renderer>
>+TestAudioDeviceModule::CreateWavFileWriter(std::string filename,
>+                                           int sampling_frequency_in_hz,
>+                                           int num_channels) {
>+  return std::unique_ptr<TestAudioDeviceModule::Renderer>(
>+      new WavFileWriter(filename, sampling_frequency_in_hz, num_channels));
>+}
>+
>+std::unique_ptr<TestAudioDeviceModule::Renderer>
>+TestAudioDeviceModule::CreateBoundedWavFileWriter(std::string filename,
>+                                                  int sampling_frequency_in_hz,
>+                                                  int num_channels) {
>+  return std::unique_ptr<TestAudioDeviceModule::Renderer>(
>+      new BoundedWavFileWriter(filename, sampling_frequency_in_hz,
>+                               num_channels));
>+}
>+
>+std::unique_ptr<TestAudioDeviceModule::Capturer>
>+TestAudioDeviceModule::CreateWavFileReader(rtc::PlatformFile file,
>+                                           int sampling_frequency_in_hz,
>+                                           int num_channels) {
>+  return std::unique_ptr<TestAudioDeviceModule::Capturer>(
>+      new WavFileReader(file, sampling_frequency_in_hz, num_channels));
>+}
>+
>+std::unique_ptr<TestAudioDeviceModule::Capturer>
>+TestAudioDeviceModule::CreateWavFileReader(rtc::PlatformFile file) {
>+  WavReader reader(file);
>+  int sampling_frequency_in_hz = reader.sample_rate();
>+  int num_channels = rtc::checked_cast<int>(reader.num_channels());
>+  return std::unique_ptr<TestAudioDeviceModule::Capturer>(
>+      new WavFileReader(file, sampling_frequency_in_hz, num_channels));
>+}
>+
>+std::unique_ptr<TestAudioDeviceModule::Renderer>
>+TestAudioDeviceModule::CreateWavFileWriter(rtc::PlatformFile file,
>+                                           int sampling_frequency_in_hz,
>+                                           int num_channels) {
>+  return std::unique_ptr<TestAudioDeviceModule::Renderer>(
>+      new WavFileWriter(file, sampling_frequency_in_hz, num_channels));
>+}
>+
>+std::unique_ptr<TestAudioDeviceModule::Renderer>
>+TestAudioDeviceModule::CreateBoundedWavFileWriter(rtc::PlatformFile file,
>+                                                  int sampling_frequency_in_hz,
>+                                                  int num_channels) {
>+  return std::unique_ptr<TestAudioDeviceModule::Renderer>(
>+      new BoundedWavFileWriter(file, sampling_frequency_in_hz, num_channels));
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/test_audio_device.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/test_audio_device.h
>new file mode 100644
>index 00000000000..7d034dce8c7
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/test_audio_device.h
>@@ -0,0 +1,170 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#ifndef MODULES_AUDIO_DEVICE_INCLUDE_TEST_AUDIO_DEVICE_H_
>+#define MODULES_AUDIO_DEVICE_INCLUDE_TEST_AUDIO_DEVICE_H_
>+
>+#include <memory>
>+#include <string>
>+#include <vector>
>+
>+#include "modules/audio_device/include/audio_device.h"
>+#include "rtc_base/buffer.h"
>+#include "rtc_base/event.h"
>+#include "rtc_base/platform_file.h"
>+
>+namespace webrtc {
>+
>+// TestAudioDeviceModule implements an AudioDevice module that can act both as a
>+// capturer and a renderer. It will use 10ms audio frames.
>+class TestAudioDeviceModule : public AudioDeviceModule {
>+ public:
>+  // Returns the number of samples that Capturers and Renderers with this
>+  // sampling frequency will work with every time Capture or Render is called.
>+  static size_t SamplesPerFrame(int sampling_frequency_in_hz);
>+
>+  class Capturer {
>+   public:
>+    virtual ~Capturer() {}
>+    // Returns the sampling frequency in Hz of the audio data that this
>+    // capturer produces.
>+    virtual int SamplingFrequency() const = 0;
>+    // Returns the number of channels of captured audio data.
>+    virtual int NumChannels() const = 0;
>+    // Replaces the contents of |buffer| with 10ms of captured audio data
>+    // (see TestAudioDeviceModule::SamplesPerFrame). Returns true if the
>+    // capturer can keep producing data, or false when the capture finishes.
>+    virtual bool Capture(rtc::BufferT<int16_t>* buffer) = 0;
>+  };
>+
>+  class Renderer {
>+   public:
>+    virtual ~Renderer() {}
>+    // Returns the sampling frequency in Hz of the audio data that this
>+    // renderer receives.
>+    virtual int SamplingFrequency() const = 0;
>+    // Returns the number of channels of audio data to be required.
>+    virtual int NumChannels() const = 0;
>+    // Renders the passed audio data and returns true if the renderer wants
>+    // to keep receiving data, or false otherwise.
>+    virtual bool Render(rtc::ArrayView<const int16_t> data) = 0;
>+  };
>+
>+  // A fake capturer that generates pulses with random samples between
>+  // -max_amplitude and +max_amplitude.
>+  class PulsedNoiseCapturer : public Capturer {
>+   public:
>+    virtual ~PulsedNoiseCapturer() {}
>+
>+    virtual void SetMaxAmplitude(int16_t amplitude) = 0;
>+  };
>+
>+  virtual ~TestAudioDeviceModule() {}
>+
>+  // Creates a new TestAudioDeviceModule. When capturing or playing, 10 ms audio
>+  // frames will be processed every 10ms / |speed|.
>+  // |capturer| is an object that produces audio data. Can be nullptr if this
>+  // device is never used for recording.
>+  // |renderer| is an object that receives audio data that would have been
>+  // played out. Can be nullptr if this device is never used for playing.
>+  // Use one of the Create... functions to get these instances.
>+  static rtc::scoped_refptr<TestAudioDeviceModule> CreateTestAudioDeviceModule(
>+      std::unique_ptr<Capturer> capturer,
>+      std::unique_ptr<Renderer> renderer,
>+      float speed = 1);
>+
>+  // Returns a Capturer instance that generates a signal of |num_channels|
>+  // channels where every second frame is zero and every second frame is evenly
>+  // distributed random noise with max amplitude |max_amplitude|.
>+  static std::unique_ptr<PulsedNoiseCapturer> CreatePulsedNoiseCapturer(
>+      int16_t max_amplitude,
>+      int sampling_frequency_in_hz,
>+      int num_channels = 1);
>+
>+  // Returns a Renderer instance that does nothing with the audio data.
>+  static std::unique_ptr<Renderer> CreateDiscardRenderer(
>+      int sampling_frequency_in_hz,
>+      int num_channels = 1);
>+
>+  // WavReader and WavWriter creation based on file name.
>+
>+  // Returns a Capturer instance that gets its data from a file. The sample rate
>+  // and channels will be checked against the Wav file.
>+  static std::unique_ptr<Capturer> CreateWavFileReader(
>+      std::string filename,
>+      int sampling_frequency_in_hz,
>+      int num_channels = 1);
>+
>+  // Returns a Capturer instance that gets its data from a file.
>+  // Automatically detects sample rate and num of channels.
>+  static std::unique_ptr<Capturer> CreateWavFileReader(std::string filename);
>+
>+  // Returns a Renderer instance that writes its data to a file.
>+  static std::unique_ptr<Renderer> CreateWavFileWriter(
>+      std::string filename,
>+      int sampling_frequency_in_hz,
>+      int num_channels = 1);
>+
>+  // Returns a Renderer instance that writes its data to a WAV file, cutting
>+  // off silence at the beginning (not necessarily perfect silence, see
>+  // kAmplitudeThreshold) and at the end (only actual 0 samples in this case).
>+  static std::unique_ptr<Renderer> CreateBoundedWavFileWriter(
>+      std::string filename,
>+      int sampling_frequency_in_hz,
>+      int num_channels = 1);
>+
>+  // WavReader and WavWriter creation based on rtc::PlatformFile.
>+
>+  // Returns a Capturer instance that gets its data from a file. The sample rate
>+  // and channels will be checked against the Wav file.
>+  static std::unique_ptr<Capturer> CreateWavFileReader(
>+      rtc::PlatformFile file,
>+      int sampling_frequency_in_hz,
>+      int num_channels = 1);
>+
>+  // Returns a Capturer instance that gets its data from a file.
>+  // Automatically detects sample rate and num of channels.
>+  static std::unique_ptr<Capturer> CreateWavFileReader(rtc::PlatformFile file);
>+
>+  // Returns a Renderer instance that writes its data to a file.
>+  static std::unique_ptr<Renderer> CreateWavFileWriter(
>+      rtc::PlatformFile file,
>+      int sampling_frequency_in_hz,
>+      int num_channels = 1);
>+
>+  // Returns a Renderer instance that writes its data to a WAV file, cutting
>+  // off silence at the beginning (not necessarily perfect silence, see
>+  // kAmplitudeThreshold) and at the end (only actual 0 samples in this case).
>+  static std::unique_ptr<Renderer> CreateBoundedWavFileWriter(
>+      rtc::PlatformFile file,
>+      int sampling_frequency_in_hz,
>+      int num_channels = 1);
>+
>+  virtual int32_t Init() = 0;
>+  virtual int32_t RegisterAudioCallback(AudioTransport* callback) = 0;
>+
>+  virtual int32_t StartPlayout() = 0;
>+  virtual int32_t StopPlayout() = 0;
>+  virtual int32_t StartRecording() = 0;
>+  virtual int32_t StopRecording() = 0;
>+
>+  virtual bool Playing() const = 0;
>+  virtual bool Recording() const = 0;
>+
>+  // Blocks until the Renderer refuses to receive data.
>+  // Returns false if |timeout_ms| passes before that happens.
>+  virtual bool WaitForPlayoutEnd(int timeout_ms = rtc::Event::kForever) = 0;
>+  // Blocks until the Recorder stops producing data.
>+  // Returns false if |timeout_ms| passes before that happens.
>+  virtual bool WaitForRecordingEnd(int timeout_ms = rtc::Event::kForever) = 0;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_DEVICE_INCLUDE_TEST_AUDIO_DEVICE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/test_audio_device_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/test_audio_device_unittest.cc
>new file mode 100644
>index 00000000000..bdb1285e85d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/test_audio_device_unittest.cc
>@@ -0,0 +1,202 @@
>+/*
>+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <algorithm>
>+#include <array>
>+
>+#include "api/array_view.h"
>+#include "common_audio/wav_file.h"
>+#include "common_audio/wav_header.h"
>+#include "modules/audio_device/include/test_audio_device.h"
>+#include "rtc_base/logging.h"
>+#include "test/gmock.h"
>+#include "test/gtest.h"
>+#include "test/testsupport/fileutils.h"
>+
>+namespace webrtc {
>+
>+namespace {
>+
>+void RunTestViaRtcPlatformFileAPI(
>+    rtc::ArrayView<const int16_t> input_samples,
>+    rtc::ArrayView<const int16_t> expected_samples,
>+    size_t samples_per_frame) {
>+  const ::testing::TestInfo* const test_info =
>+      ::testing::UnitTest::GetInstance()->current_test_info();
>+
>+  const std::string output_filename =
>+      test::OutputPath() + "BoundedWavFileWriterTest_" + test_info->name() +
>+      "_" + std::to_string(std::rand()) + ".wav";
>+
>+  static const size_t kSamplesPerFrame = 8;
>+  static const int kSampleRate = kSamplesPerFrame * 100;
>+  EXPECT_EQ(TestAudioDeviceModule::SamplesPerFrame(kSampleRate),
>+            kSamplesPerFrame);
>+
>+  // Test through rtc::PlatformFile API.
>+  {
>+    auto file = rtc::CreatePlatformFile(output_filename);
>+    std::unique_ptr<TestAudioDeviceModule::Renderer> writer =
>+        TestAudioDeviceModule::CreateBoundedWavFileWriter(file, 800);
>+
>+    for (size_t i = 0; i < input_samples.size(); i += kSamplesPerFrame) {
>+      EXPECT_TRUE(writer->Render(rtc::ArrayView<const int16_t>(
>+          &input_samples[i],
>+          std::min(kSamplesPerFrame, input_samples.size() - i))));
>+    }
>+  }
>+
>+  {
>+    auto file = rtc::OpenPlatformFile(output_filename);
>+    WavReader reader(file);
>+    std::vector<int16_t> read_samples(expected_samples.size());
>+    EXPECT_EQ(expected_samples.size(),
>+              reader.ReadSamples(read_samples.size(), read_samples.data()));
>+    EXPECT_THAT(expected_samples, testing::ElementsAreArray(read_samples));
>+
>+    EXPECT_EQ(0u, reader.ReadSamples(read_samples.size(), read_samples.data()));
>+  }
>+
>+  remove(output_filename.c_str());
>+}
>+
>+void RunTest(const std::vector<int16_t>& input_samples,
>+             const std::vector<int16_t>& expected_samples,
>+             size_t samples_per_frame) {
>+  const ::testing::TestInfo* const test_info =
>+      ::testing::UnitTest::GetInstance()->current_test_info();
>+
>+  const std::string output_filename =
>+      test::OutputPath() + "BoundedWavFileWriterTest_" + test_info->name() +
>+      "_" + std::to_string(std::rand()) + ".wav";
>+
>+  static const size_t kSamplesPerFrame = 8;
>+  static const int kSampleRate = kSamplesPerFrame * 100;
>+  EXPECT_EQ(TestAudioDeviceModule::SamplesPerFrame(kSampleRate),
>+            kSamplesPerFrame);
>+
>+  // Test through file name API.
>+  {
>+    std::unique_ptr<TestAudioDeviceModule::Renderer> writer =
>+        TestAudioDeviceModule::CreateBoundedWavFileWriter(output_filename, 800);
>+
>+    for (size_t i = 0; i < input_samples.size(); i += kSamplesPerFrame) {
>+      EXPECT_TRUE(writer->Render(rtc::ArrayView<const int16_t>(
>+          &input_samples[i],
>+          std::min(kSamplesPerFrame, input_samples.size() - i))));
>+    }
>+  }
>+
>+  {
>+    WavReader reader(output_filename);
>+    std::vector<int16_t> read_samples(expected_samples.size());
>+    EXPECT_EQ(expected_samples.size(),
>+              reader.ReadSamples(read_samples.size(), read_samples.data()));
>+    EXPECT_EQ(expected_samples, read_samples);
>+
>+    EXPECT_EQ(0u, reader.ReadSamples(read_samples.size(), read_samples.data()));
>+  }
>+
>+  remove(output_filename.c_str());
>+}
>+}  // namespace
>+
>+TEST(BoundedWavFileWriterTest, NoSilence) {
>+  static const std::vector<int16_t> kInputSamples = {
>+      75,   1234,  243,    -1231, -22222, 0,    3,      88,
>+      1222, -1213, -13222, -7,    -3525,  5787, -25247, 8};
>+  static const std::vector<int16_t> kExpectedSamples = kInputSamples;
>+  RunTest(kInputSamples, kExpectedSamples, 8);
>+}
>+
>+TEST(BoundedWavFileWriterTest, SomeStartSilence) {
>+  static const std::vector<int16_t> kInputSamples = {
>+      0, 0, 0, 0, 3, 0, 0, 0, 0, 3, -13222, -7, -3525, 5787, -25247, 8};
>+  static const std::vector<int16_t> kExpectedSamples(kInputSamples.begin() + 10,
>+                                                     kInputSamples.end());
>+  RunTest(kInputSamples, kExpectedSamples, 8);
>+}
>+
>+TEST(BoundedWavFileWriterTest, SomeStartSilenceRtcFile) {
>+  static const std::vector<int16_t> kInputSamples = {
>+      0, 0, 0, 0, 3, 0, 0, 0, 0, 3, -13222, -7, -3525, 5787, -25247, 8};
>+  static const std::vector<int16_t> kExpectedSamples(kInputSamples.begin() + 10,
>+                                                     kInputSamples.end());
>+  RunTestViaRtcPlatformFileAPI(kInputSamples, kExpectedSamples, 8);
>+}
>+
>+TEST(BoundedWavFileWriterTest, NegativeStartSilence) {
>+  static const std::vector<int16_t> kInputSamples = {
>+      0, -4, -6, 0, 3, 0, 0, 0, 0, 3, -13222, -7, -3525, 5787, -25247, 8};
>+  static const std::vector<int16_t> kExpectedSamples(kInputSamples.begin() + 2,
>+                                                     kInputSamples.end());
>+  RunTest(kInputSamples, kExpectedSamples, 8);
>+}
>+
>+TEST(BoundedWavFileWriterTest, SomeEndSilence) {
>+  static const std::vector<int16_t> kInputSamples = {
>+      75, 1234, 243, -1231, -22222, 0, 1, 0, 0, 0, 0, 0, 0, 0, 0, 0};
>+  static const std::vector<int16_t> kExpectedSamples(kInputSamples.begin(),
>+                                                     kInputSamples.end() - 9);
>+  RunTest(kInputSamples, kExpectedSamples, 8);
>+}
>+
>+TEST(BoundedWavFileWriterTest, DoubleEndSilence) {
>+  static const std::vector<int16_t> kInputSamples = {
>+      75, 1234,  243,    -1231, -22222, 0,    0, 0,
>+      0,  -1213, -13222, -7,    -3525,  5787, 0, 0};
>+  static const std::vector<int16_t> kExpectedSamples(kInputSamples.begin(),
>+                                                     kInputSamples.end() - 2);
>+  RunTest(kInputSamples, kExpectedSamples, 8);
>+}
>+
>+TEST(BoundedWavFileWriterTest, DoubleSilence) {
>+  static const std::vector<int16_t> kInputSamples = {0,     -1213, -13222, -7,
>+                                                     -3525, 5787,  0,      0};
>+  static const std::vector<int16_t> kExpectedSamples(kInputSamples.begin() + 1,
>+                                                     kInputSamples.end() - 2);
>+  RunTest(kInputSamples, kExpectedSamples, 8);
>+}
>+
>+TEST(BoundedWavFileWriterTest, EndSilenceCutoff) {
>+  static const std::vector<int16_t> kInputSamples = {
>+      75, 1234, 243, -1231, -22222, 0, 1, 0, 0, 0, 0};
>+  static const std::vector<int16_t> kExpectedSamples(kInputSamples.begin(),
>+                                                     kInputSamples.end() - 4);
>+  RunTest(kInputSamples, kExpectedSamples, 8);
>+}
>+
>+TEST(PulsedNoiseCapturerTest, SetMaxAmplitude) {
>+  const int16_t kAmplitude = 50;
>+  std::unique_ptr<TestAudioDeviceModule::PulsedNoiseCapturer> capturer =
>+      TestAudioDeviceModule::CreatePulsedNoiseCapturer(
>+          kAmplitude, /*sampling_frequency_in_hz=*/8000);
>+  rtc::BufferT<int16_t> recording_buffer;
>+
>+  // Verify that the capturer doesn't create entries louder than than
>+  // kAmplitude. Since the pulse generator alternates between writing
>+  // zeroes and actual entries, we need to do the capturing twice.
>+  capturer->Capture(&recording_buffer);
>+  capturer->Capture(&recording_buffer);
>+  int16_t max_sample =
>+      *std::max_element(recording_buffer.begin(), recording_buffer.end());
>+  EXPECT_LE(max_sample, kAmplitude);
>+
>+  // Increase the amplitude and verify that the samples can now be louder
>+  // than the previous max.
>+  capturer->SetMaxAmplitude(kAmplitude * 2);
>+  capturer->Capture(&recording_buffer);
>+  capturer->Capture(&recording_buffer);
>+  max_sample =
>+      *std::max_element(recording_buffer.begin(), recording_buffer.end());
>+  EXPECT_GT(max_sample, kAmplitude);
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/ios/audio_device_ios.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/ios/audio_device_ios.h
>index d97e124a56e..eeebbf382f7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/ios/audio_device_ios.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/ios/audio_device_ios.h
>@@ -13,7 +13,6 @@
> 
> #include <memory>
> 
>-#include "sdk/objc/Framework/Headers/WebRTC/RTCMacros.h"
> #include "modules/audio_device/audio_device_generic.h"
> #include "modules/audio_device/ios/audio_session_observer.h"
> #include "modules/audio_device/ios/voice_processing_audio_unit.h"
>@@ -22,6 +21,7 @@
> #include "rtc_base/thread.h"
> #include "rtc_base/thread_annotations.h"
> #include "rtc_base/thread_checker.h"
>+#include "sdk/objc/Framework/Headers/WebRTC/RTCMacros.h"
> 
> RTC_FWD_DECL_OBJC_CLASS(RTCAudioSessionDelegateAdapter);
> 
>@@ -154,7 +154,7 @@ class AudioDeviceIOS : public AudioDeviceGeneric,
>                             AudioBufferList* io_data) override;
> 
>   // Handles messages from posts.
>-  void OnMessage(rtc::Message *msg) override;
>+  void OnMessage(rtc::Message* msg) override;
> 
>  private:
>   // Called by the relevant AudioSessionObserver methods on |thread_|.
>@@ -195,6 +195,9 @@ class AudioDeviceIOS : public AudioDeviceGeneric,
>   // Closes and deletes the voice-processing I/O unit.
>   void ShutdownPlayOrRecord();
> 
>+  // Resets thread-checkers before a call is restarted.
>+  void PrepareForNewStart();
>+
>   // Ensures that methods are called from the same thread as this object is
>   // created on.
>   rtc::ThreadChecker thread_checker_;
>@@ -246,7 +249,7 @@ class AudioDeviceIOS : public AudioDeviceGeneric,
>   // On real iOS devices, the size will be fixed and set once. For iOS
>   // simulators, the size can vary from callback to callback and the size
>   // will be changed dynamically to account for this behavior.
>-  rtc::BufferT<int8_t> record_audio_buffer_;
>+  rtc::BufferT<int16_t> record_audio_buffer_;
> 
>   // Set to 1 when recording is active and 0 otherwise.
>   volatile int recording_;
>@@ -255,7 +258,7 @@ class AudioDeviceIOS : public AudioDeviceGeneric,
>   volatile int playing_;
> 
>   // Set to true after successful call to Init(), false otherwise.
>-  bool initialized_ RTC_ACCESS_ON(thread_checker_);
>+  bool initialized_ RTC_GUARDED_BY(thread_checker_);
> 
>   // Set to true after successful call to InitRecording() or InitPlayout(),
>   // false otherwise.
>@@ -266,14 +269,14 @@ class AudioDeviceIOS : public AudioDeviceGeneric,
> 
>   // Audio interruption observer instance.
>   RTCAudioSessionDelegateAdapter* audio_session_observer_
>-      RTC_ACCESS_ON(thread_checker_);
>+      RTC_GUARDED_BY(thread_checker_);
> 
>   // Set to true if we've activated the audio session.
>-  bool has_configured_session_ RTC_ACCESS_ON(thread_checker_);
>+  bool has_configured_session_ RTC_GUARDED_BY(thread_checker_);
> 
>   // Counts number of detected audio glitches on the playout side.
>-  int64_t num_detected_playout_glitches_ RTC_ACCESS_ON(thread_checker_);
>-  int64_t last_playout_time_ RTC_ACCESS_ON(io_thread_checker_);
>+  int64_t num_detected_playout_glitches_ RTC_GUARDED_BY(thread_checker_);
>+  int64_t last_playout_time_ RTC_GUARDED_BY(io_thread_checker_);
> 
>   // Counts number of playout callbacks per call.
>   // The value isupdated on the native I/O thread and later read on the
>@@ -282,7 +285,7 @@ class AudioDeviceIOS : public AudioDeviceGeneric,
>   int64_t num_playout_callbacks_;
> 
>   // Contains the time for when the last output volume change was detected.
>-  int64_t last_output_volume_change_time_ RTC_ACCESS_ON(thread_checker_);
>+  int64_t last_output_volume_change_time_ RTC_GUARDED_BY(thread_checker_);
> 
>   // Exposes private members for testing purposes only.
>   FRIEND_TEST_ALL_PREFIXES(AudioDeviceTest, testInterruptedAudioSession);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/ios/audio_device_ios.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/ios/audio_device_ios.mm
>index 5dd816a92f5..7e3d8f2181f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/ios/audio_device_ios.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/ios/audio_device_ios.mm
>@@ -360,12 +360,11 @@ OSStatus AudioDeviceIOS::OnDeliverRecordedData(AudioUnitRenderActionFlags* flags
>   // Simply return if recording is not enabled.
>   if (!rtc::AtomicOps::AcquireLoad(&recording_)) return result;
> 
>-  const size_t num_bytes = num_frames * VoiceProcessingAudioUnit::kBytesPerSample;
>   // Set the size of our own audio buffer and clear it first to avoid copying
>   // in combination with potential reallocations.
>   // On real iOS devices, the size will only be set once (at first callback).
>   record_audio_buffer_.Clear();
>-  record_audio_buffer_.SetSize(num_bytes);
>+  record_audio_buffer_.SetSize(num_frames);
> 
>   // Allocate AudioBuffers to be used as storage for the received audio.
>   // The AudioBufferList structure works as a placeholder for the
>@@ -376,8 +375,9 @@ OSStatus AudioDeviceIOS::OnDeliverRecordedData(AudioUnitRenderActionFlags* flags
>   audio_buffer_list.mNumberBuffers = 1;
>   AudioBuffer* audio_buffer = &audio_buffer_list.mBuffers[0];
>   audio_buffer->mNumberChannels = record_parameters_.channels();
>-  audio_buffer->mDataByteSize = record_audio_buffer_.size();
>-  audio_buffer->mData = record_audio_buffer_.data();
>+  audio_buffer->mDataByteSize =
>+      record_audio_buffer_.size() * VoiceProcessingAudioUnit::kBytesPerSample;
>+  audio_buffer->mData = reinterpret_cast<int8_t*>(record_audio_buffer_.data());
> 
>   // Obtain the recorded audio samples by initiating a rendering cycle.
>   // Since it happens on the input bus, the |io_data| parameter is a reference
>@@ -394,8 +394,7 @@ OSStatus AudioDeviceIOS::OnDeliverRecordedData(AudioUnitRenderActionFlags* flags
>   // Get a pointer to the recorded audio and send it to the WebRTC ADB.
>   // Use the FineAudioBuffer instance to convert between native buffer size
>   // and the 10ms buffer size used by WebRTC.
>-  fine_audio_buffer_->DeliverRecordedData(
>-      record_audio_buffer_, kFixedPlayoutDelayEstimate, kFixedRecordDelayEstimate);
>+  fine_audio_buffer_->DeliverRecordedData(record_audio_buffer_, kFixedRecordDelayEstimate);
>   return noErr;
> }
> 
>@@ -410,16 +409,13 @@ OSStatus AudioDeviceIOS::OnGetPlayoutData(AudioUnitRenderActionFlags* flags,
>   AudioBuffer* audio_buffer = &io_data->mBuffers[0];
>   RTC_DCHECK_EQ(1, audio_buffer->mNumberChannels);
> 
>-  // Get pointer to internal audio buffer to which new audio data shall be
>-  // written.
>-  const size_t size_in_bytes = audio_buffer->mDataByteSize;
>-  RTC_CHECK_EQ(size_in_bytes / VoiceProcessingAudioUnit::kBytesPerSample, num_frames);
>-  int8_t* destination = reinterpret_cast<int8_t*>(audio_buffer->mData);
>   // Produce silence and give audio unit a hint about it if playout is not
>   // activated.
>   if (!rtc::AtomicOps::AcquireLoad(&playing_)) {
>+    const size_t size_in_bytes = audio_buffer->mDataByteSize;
>+    RTC_CHECK_EQ(size_in_bytes / VoiceProcessingAudioUnit::kBytesPerSample, num_frames);
>     *flags |= kAudioUnitRenderAction_OutputIsSilence;
>-    memset(destination, 0, size_in_bytes);
>+    memset(static_cast<int8_t*>(audio_buffer->mData), 0, size_in_bytes);
>     return noErr;
>   }
> 
>@@ -455,7 +451,9 @@ OSStatus AudioDeviceIOS::OnGetPlayoutData(AudioUnitRenderActionFlags* flags,
>   // Read decoded 16-bit PCM samples from WebRTC (using a size that matches
>   // the native I/O audio unit) and copy the result to the audio buffer in the
>   // |io_data| destination.
>-  fine_audio_buffer_->GetPlayoutData(rtc::ArrayView<int8_t>(destination, size_in_bytes));
>+  fine_audio_buffer_->GetPlayoutData(
>+      rtc::ArrayView<int16_t>(static_cast<int16_t*>(audio_buffer->mData), num_frames),
>+      kFixedPlayoutDelayEstimate);
>   return noErr;
> }
> 
>@@ -493,14 +491,7 @@ void AudioDeviceIOS::HandleInterruptionBegin() {
>     if (!audio_unit_->Stop()) {
>       RTCLogError(@"Failed to stop the audio unit for interruption begin.");
>     } else {
>-      // The audio unit has been stopped but will be restarted when the
>-      // interruption ends in HandleInterruptionEnd(). It will result in audio
>-      // callbacks from a new native I/O thread which means that we must detach
>-      // thread checkers here to be prepared for an upcoming new audio stream.
>-      io_thread_checker_.DetachFromThread();
>-      // The audio device buffer must also be informed about the interrupted
>-      // state so it can detach its thread checkers as well.
>-      audio_device_buffer_->NativeAudioInterrupted();
>+      PrepareForNewStart();
>     }
>   }
>   is_interrupted_ = true;
>@@ -582,6 +573,7 @@ void AudioDeviceIOS::HandleSampleRateChange(float sample_rate) {
>   if (audio_unit_->GetState() == VoiceProcessingAudioUnit::kStarted) {
>     audio_unit_->Stop();
>     restart_audio_unit = true;
>+    PrepareForNewStart();
>   }
>   if (audio_unit_->GetState() == VoiceProcessingAudioUnit::kInitialized) {
>     audio_unit_->Uninitialize();
>@@ -700,12 +692,9 @@ void AudioDeviceIOS::SetupAudioBuffersForActiveAudioSession() {
> 
>   // Create a modified audio buffer class which allows us to ask for,
>   // or deliver, any number of samples (and not only multiple of 10ms) to match
>-  // the native audio unit buffer size. Use a reasonable capacity to avoid
>-  // reallocations while audio is played to reduce risk of glitches.
>+  // the native audio unit buffer size.
>   RTC_DCHECK(audio_device_buffer_);
>-  const size_t capacity_in_bytes = 2 * playout_parameters_.GetBytesPerBuffer();
>-  fine_audio_buffer_.reset(new FineAudioBuffer(
>-      audio_device_buffer_, playout_parameters_.sample_rate(), capacity_in_bytes));
>+  fine_audio_buffer_.reset(new FineAudioBuffer(audio_device_buffer_));
> }
> 
> bool AudioDeviceIOS::CreateAudioUnit() {
>@@ -857,6 +846,7 @@ bool AudioDeviceIOS::InitPlayOrRecord() {
>   if (![session beginWebRTCSession:&error]) {
>     [session unlockForConfiguration];
>     RTCLogError(@"Failed to begin WebRTC session: %@", error.localizedDescription);
>+    audio_unit_.reset();
>     return false;
>   }
> 
>@@ -868,6 +858,7 @@ bool AudioDeviceIOS::InitPlayOrRecord() {
>       // audio session during or after a Media Services failure.
>       // See AVAudioSessionErrorCodeMediaServicesFailed for details.
>       [session unlockForConfiguration];
>+      audio_unit_.reset();
>       return false;
>     }
>     SetupAudioBuffersForActiveAudioSession();
>@@ -905,4 +896,19 @@ void AudioDeviceIOS::ShutdownPlayOrRecord() {
>   [session unlockForConfiguration];
> }
> 
>+void AudioDeviceIOS::PrepareForNewStart() {
>+  LOGI() << "PrepareForNewStart";
>+  // The audio unit has been stopped and preparations are needed for an upcoming
>+  // restart. It will result in audio callbacks from a new native I/O thread
>+  // which means that we must detach thread checkers here to be prepared for an
>+  // upcoming new audio stream.
>+  io_thread_checker_.DetachFromThread();
>+  // The audio device buffer must also be informed about the interrupted
>+  // state so it can detach its thread checkers as well.
>+  if (audio_device_buffer_) {
>+    audio_device_buffer_->NativeAudioPlayoutInterrupted();
>+    audio_device_buffer_->NativeAudioRecordingInterrupted();
>+  }
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/ios/objc/RTCAudioSessionDelegateAdapter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/ios/objc/RTCAudioSessionDelegateAdapter.h
>index 7a0f9d2594e..f024bfb00f6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/ios/objc/RTCAudioSessionDelegateAdapter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/ios/objc/RTCAudioSessionDelegateAdapter.h
>@@ -24,7 +24,6 @@ class AudioSessionObserver;
> /** |observer| is a raw pointer and should be kept alive
>  *  for this object's lifetime.
>  */
>-- (instancetype)initWithObserver:(webrtc::AudioSessionObserver *)observer
>-    NS_DESIGNATED_INITIALIZER;
>+- (instancetype)initWithObserver:(webrtc::AudioSessionObserver *)observer NS_DESIGNATED_INITIALIZER;
> 
> @end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/ios/voice_processing_audio_unit.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/ios/voice_processing_audio_unit.mm
>index c350969335c..48bd7e117a2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/ios/voice_processing_audio_unit.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/ios/voice_processing_audio_unit.mm
>@@ -11,6 +11,7 @@
> #import "modules/audio_device/ios/voice_processing_audio_unit.h"
> 
> #include "rtc_base/checks.h"
>+#include "rtc_base/system/fallthrough.h"
> #include "system_wrappers/include/metrics.h"
> 
> #import "WebRTC/RTCLogging.h"
>@@ -444,12 +445,12 @@ void VoiceProcessingAudioUnit::DisposeAudioUnit() {
>       case kStarted:
>         Stop();
>         // Fall through.
>-        FALLTHROUGH();
>+        RTC_FALLTHROUGH();
>       case kInitialized:
>         Uninitialize();
>         break;
>       case kUninitialized:
>-        FALLTHROUGH();
>+        RTC_FALLTHROUGH();
>       case kInitRequired:
>         break;
>     }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/alsasymboltable_linux.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/alsasymboltable_linux.cc
>index d605dbaac71..5dfb91d6f42 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/alsasymboltable_linux.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/alsasymboltable_linux.cc
>@@ -31,8 +31,7 @@ namespace webrtc {
> namespace adm_linux_alsa {
> 
> LATE_BINDING_SYMBOL_TABLE_DEFINE_BEGIN(AlsaSymbolTable, "libasound.so.2")
>-#define X(sym) \
>-    LATE_BINDING_SYMBOL_TABLE_DEFINE_ENTRY(AlsaSymbolTable, sym)
>+#define X(sym) LATE_BINDING_SYMBOL_TABLE_DEFINE_ENTRY(AlsaSymbolTable, sym)
> ALSA_SYMBOLS_LIST
> #undef X
> LATE_BINDING_SYMBOL_TABLE_DEFINE_END(AlsaSymbolTable)
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/alsasymboltable_linux.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/alsasymboltable_linux.h
>index 9c1f9545220..c9970b02bc0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/alsasymboltable_linux.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/alsasymboltable_linux.h
>@@ -36,109 +36,108 @@ namespace adm_linux_alsa {
> // The ALSA symbols we need, as an X-Macro list.
> // This list must contain precisely every libasound function that is used in
> // alsasoundsystem.cc.
>-#define ALSA_SYMBOLS_LIST \
>-  X(snd_device_name_free_hint) \
>-  X(snd_device_name_get_hint) \
>-  X(snd_device_name_hint) \
>-  X(snd_pcm_avail_update) \
>-  X(snd_pcm_close) \
>-  X(snd_pcm_delay) \
>-  X(snd_pcm_drop) \
>-  X(snd_pcm_open) \
>-  X(snd_pcm_prepare) \
>-  X(snd_pcm_readi) \
>-  X(snd_pcm_recover) \
>-  X(snd_pcm_resume) \
>-  X(snd_pcm_reset) \
>-  X(snd_pcm_state) \
>-  X(snd_pcm_set_params) \
>-  X(snd_pcm_get_params) \
>-  X(snd_pcm_start) \
>-  X(snd_pcm_stream) \
>-  X(snd_pcm_frames_to_bytes) \
>-  X(snd_pcm_bytes_to_frames) \
>-  X(snd_pcm_wait) \
>-  X(snd_pcm_writei) \
>-  X(snd_pcm_info_get_class) \
>-  X(snd_pcm_info_get_subdevices_avail) \
>-  X(snd_pcm_info_get_subdevice_name) \
>-  X(snd_pcm_info_set_subdevice) \
>-  X(snd_pcm_info_get_id) \
>-  X(snd_pcm_info_set_device) \
>-  X(snd_pcm_info_set_stream) \
>-  X(snd_pcm_info_get_name) \
>-  X(snd_pcm_info_get_subdevices_count) \
>-  X(snd_pcm_info_sizeof) \
>-  X(snd_pcm_hw_params) \
>-  X(snd_pcm_hw_params_malloc) \
>-  X(snd_pcm_hw_params_free) \
>-  X(snd_pcm_hw_params_any) \
>-  X(snd_pcm_hw_params_set_access) \
>-  X(snd_pcm_hw_params_set_format) \
>-  X(snd_pcm_hw_params_set_channels) \
>-  X(snd_pcm_hw_params_set_rate_near) \
>-  X(snd_pcm_hw_params_set_buffer_size_near) \
>-  X(snd_card_next) \
>-  X(snd_card_get_name) \
>-  X(snd_config_update) \
>-  X(snd_config_copy) \
>-  X(snd_config_get_id) \
>-  X(snd_ctl_open) \
>-  X(snd_ctl_close) \
>-  X(snd_ctl_card_info) \
>-  X(snd_ctl_card_info_sizeof) \
>-  X(snd_ctl_card_info_get_id) \
>-  X(snd_ctl_card_info_get_name) \
>-  X(snd_ctl_pcm_next_device) \
>-  X(snd_ctl_pcm_info) \
>-  X(snd_mixer_load) \
>-  X(snd_mixer_free) \
>-  X(snd_mixer_detach) \
>-  X(snd_mixer_close) \
>-  X(snd_mixer_open) \
>-  X(snd_mixer_attach) \
>-  X(snd_mixer_first_elem) \
>-  X(snd_mixer_elem_next) \
>-  X(snd_mixer_selem_get_name) \
>-  X(snd_mixer_selem_is_active) \
>-  X(snd_mixer_selem_register) \
>-  X(snd_mixer_selem_set_playback_volume_all) \
>-  X(snd_mixer_selem_get_playback_volume) \
>-  X(snd_mixer_selem_has_playback_volume) \
>+#define ALSA_SYMBOLS_LIST                      \
>+  X(snd_device_name_free_hint)                 \
>+  X(snd_device_name_get_hint)                  \
>+  X(snd_device_name_hint)                      \
>+  X(snd_pcm_avail_update)                      \
>+  X(snd_pcm_close)                             \
>+  X(snd_pcm_delay)                             \
>+  X(snd_pcm_drop)                              \
>+  X(snd_pcm_open)                              \
>+  X(snd_pcm_prepare)                           \
>+  X(snd_pcm_readi)                             \
>+  X(snd_pcm_recover)                           \
>+  X(snd_pcm_resume)                            \
>+  X(snd_pcm_reset)                             \
>+  X(snd_pcm_state)                             \
>+  X(snd_pcm_set_params)                        \
>+  X(snd_pcm_get_params)                        \
>+  X(snd_pcm_start)                             \
>+  X(snd_pcm_stream)                            \
>+  X(snd_pcm_frames_to_bytes)                   \
>+  X(snd_pcm_bytes_to_frames)                   \
>+  X(snd_pcm_wait)                              \
>+  X(snd_pcm_writei)                            \
>+  X(snd_pcm_info_get_class)                    \
>+  X(snd_pcm_info_get_subdevices_avail)         \
>+  X(snd_pcm_info_get_subdevice_name)           \
>+  X(snd_pcm_info_set_subdevice)                \
>+  X(snd_pcm_info_get_id)                       \
>+  X(snd_pcm_info_set_device)                   \
>+  X(snd_pcm_info_set_stream)                   \
>+  X(snd_pcm_info_get_name)                     \
>+  X(snd_pcm_info_get_subdevices_count)         \
>+  X(snd_pcm_info_sizeof)                       \
>+  X(snd_pcm_hw_params)                         \
>+  X(snd_pcm_hw_params_malloc)                  \
>+  X(snd_pcm_hw_params_free)                    \
>+  X(snd_pcm_hw_params_any)                     \
>+  X(snd_pcm_hw_params_set_access)              \
>+  X(snd_pcm_hw_params_set_format)              \
>+  X(snd_pcm_hw_params_set_channels)            \
>+  X(snd_pcm_hw_params_set_rate_near)           \
>+  X(snd_pcm_hw_params_set_buffer_size_near)    \
>+  X(snd_card_next)                             \
>+  X(snd_card_get_name)                         \
>+  X(snd_config_update)                         \
>+  X(snd_config_copy)                           \
>+  X(snd_config_get_id)                         \
>+  X(snd_ctl_open)                              \
>+  X(snd_ctl_close)                             \
>+  X(snd_ctl_card_info)                         \
>+  X(snd_ctl_card_info_sizeof)                  \
>+  X(snd_ctl_card_info_get_id)                  \
>+  X(snd_ctl_card_info_get_name)                \
>+  X(snd_ctl_pcm_next_device)                   \
>+  X(snd_ctl_pcm_info)                          \
>+  X(snd_mixer_load)                            \
>+  X(snd_mixer_free)                            \
>+  X(snd_mixer_detach)                          \
>+  X(snd_mixer_close)                           \
>+  X(snd_mixer_open)                            \
>+  X(snd_mixer_attach)                          \
>+  X(snd_mixer_first_elem)                      \
>+  X(snd_mixer_elem_next)                       \
>+  X(snd_mixer_selem_get_name)                  \
>+  X(snd_mixer_selem_is_active)                 \
>+  X(snd_mixer_selem_register)                  \
>+  X(snd_mixer_selem_set_playback_volume_all)   \
>+  X(snd_mixer_selem_get_playback_volume)       \
>+  X(snd_mixer_selem_has_playback_volume)       \
>   X(snd_mixer_selem_get_playback_volume_range) \
>-  X(snd_mixer_selem_has_playback_switch) \
>-  X(snd_mixer_selem_get_playback_switch) \
>-  X(snd_mixer_selem_set_playback_switch_all) \
>-  X(snd_mixer_selem_has_capture_switch) \
>-  X(snd_mixer_selem_get_capture_switch) \
>-  X(snd_mixer_selem_set_capture_switch_all) \
>-  X(snd_mixer_selem_has_capture_volume) \
>-  X(snd_mixer_selem_set_capture_volume_all) \
>-  X(snd_mixer_selem_get_capture_volume) \
>-  X(snd_mixer_selem_get_capture_volume_range) \
>-  X(snd_dlopen) \
>-  X(snd_dlclose) \
>-  X(snd_config) \
>-  X(snd_config_search) \
>-  X(snd_config_get_string) \
>-  X(snd_config_search_definition) \
>-  X(snd_config_get_type) \
>-  X(snd_config_delete) \
>-  X(snd_config_iterator_entry) \
>-  X(snd_config_iterator_first) \
>-  X(snd_config_iterator_next) \
>-  X(snd_config_iterator_end) \
>-  X(snd_config_delete_compound_members) \
>-  X(snd_config_get_integer) \
>-  X(snd_config_get_bool) \
>-  X(snd_dlsym) \
>-  X(snd_strerror) \
>-  X(snd_lib_error) \
>+  X(snd_mixer_selem_has_playback_switch)       \
>+  X(snd_mixer_selem_get_playback_switch)       \
>+  X(snd_mixer_selem_set_playback_switch_all)   \
>+  X(snd_mixer_selem_has_capture_switch)        \
>+  X(snd_mixer_selem_get_capture_switch)        \
>+  X(snd_mixer_selem_set_capture_switch_all)    \
>+  X(snd_mixer_selem_has_capture_volume)        \
>+  X(snd_mixer_selem_set_capture_volume_all)    \
>+  X(snd_mixer_selem_get_capture_volume)        \
>+  X(snd_mixer_selem_get_capture_volume_range)  \
>+  X(snd_dlopen)                                \
>+  X(snd_dlclose)                               \
>+  X(snd_config)                                \
>+  X(snd_config_search)                         \
>+  X(snd_config_get_string)                     \
>+  X(snd_config_search_definition)              \
>+  X(snd_config_get_type)                       \
>+  X(snd_config_delete)                         \
>+  X(snd_config_iterator_entry)                 \
>+  X(snd_config_iterator_first)                 \
>+  X(snd_config_iterator_next)                  \
>+  X(snd_config_iterator_end)                   \
>+  X(snd_config_delete_compound_members)        \
>+  X(snd_config_get_integer)                    \
>+  X(snd_config_get_bool)                       \
>+  X(snd_dlsym)                                 \
>+  X(snd_strerror)                              \
>+  X(snd_lib_error)                             \
>   X(snd_lib_error_set_handler)
> 
> LATE_BINDING_SYMBOL_TABLE_DECLARE_BEGIN(AlsaSymbolTable)
>-#define X(sym) \
>-    LATE_BINDING_SYMBOL_TABLE_DECLARE_ENTRY(AlsaSymbolTable, sym)
>+#define X(sym) LATE_BINDING_SYMBOL_TABLE_DECLARE_ENTRY(AlsaSymbolTable, sym)
> ALSA_SYMBOLS_LIST
> #undef X
> LATE_BINDING_SYMBOL_TABLE_DECLARE_END(AlsaSymbolTable)
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/audio_device_alsa_linux.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/audio_device_alsa_linux.cc
>index 735474c518e..f8be24e7f0b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/audio_device_alsa_linux.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/audio_device_alsa_linux.cc
>@@ -13,7 +13,7 @@
> #include "modules/audio_device/audio_device_config.h"
> #include "modules/audio_device/linux/audio_device_alsa_linux.h"
> #include "rtc_base/logging.h"
>-
>+#include "rtc_base/system/arch.h"
> #include "system_wrappers/include/event_wrapper.h"
> #include "system_wrappers/include/sleep.h"
> webrtc::adm_linux_alsa::AlsaSymbolTable AlsaSymbolTable;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/audio_device_alsa_linux.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/audio_device_alsa_linux.h
>index 60e5743fa9f..f8961453b56 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/audio_device_alsa_linux.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/audio_device_alsa_linux.h
>@@ -25,178 +25,175 @@
> #include <sys/ioctl.h>
> #include <sys/soundcard.h>
> 
>-
>-namespace webrtc
>-{
>+namespace webrtc {
> class EventWrapper;
> 
>-class AudioDeviceLinuxALSA : public AudioDeviceGeneric
>-{
>-public:
>-    AudioDeviceLinuxALSA();
>-    virtual ~AudioDeviceLinuxALSA();
>-
>-    // Retrieve the currently utilized audio layer
>-    int32_t ActiveAudioLayer(
>-        AudioDeviceModule::AudioLayer& audioLayer) const override;
>-
>-    // Main initializaton and termination
>-    InitStatus Init() override;
>-    int32_t Terminate() override;
>-    bool Initialized() const override;
>-
>-    // Device enumeration
>-    int16_t PlayoutDevices() override;
>-    int16_t RecordingDevices() override;
>-    int32_t PlayoutDeviceName(uint16_t index,
>+class AudioDeviceLinuxALSA : public AudioDeviceGeneric {
>+ public:
>+  AudioDeviceLinuxALSA();
>+  virtual ~AudioDeviceLinuxALSA();
>+
>+  // Retrieve the currently utilized audio layer
>+  int32_t ActiveAudioLayer(
>+      AudioDeviceModule::AudioLayer& audioLayer) const override;
>+
>+  // Main initializaton and termination
>+  InitStatus Init() override;
>+  int32_t Terminate() override;
>+  bool Initialized() const override;
>+
>+  // Device enumeration
>+  int16_t PlayoutDevices() override;
>+  int16_t RecordingDevices() override;
>+  int32_t PlayoutDeviceName(uint16_t index,
>+                            char name[kAdmMaxDeviceNameSize],
>+                            char guid[kAdmMaxGuidSize]) override;
>+  int32_t RecordingDeviceName(uint16_t index,
>                               char name[kAdmMaxDeviceNameSize],
>                               char guid[kAdmMaxGuidSize]) override;
>-    int32_t RecordingDeviceName(uint16_t index,
>-                                char name[kAdmMaxDeviceNameSize],
>-                                char guid[kAdmMaxGuidSize]) override;
>-
>-    // Device selection
>-    int32_t SetPlayoutDevice(uint16_t index) override;
>-    int32_t SetPlayoutDevice(
>-        AudioDeviceModule::WindowsDeviceType device) override;
>-    int32_t SetRecordingDevice(uint16_t index) override;
>-    int32_t SetRecordingDevice(
>-        AudioDeviceModule::WindowsDeviceType device) override;
>-
>-    // Audio transport initialization
>-    int32_t PlayoutIsAvailable(bool& available) override;
>-    int32_t InitPlayout() override;
>-    bool PlayoutIsInitialized() const override;
>-    int32_t RecordingIsAvailable(bool& available) override;
>-    int32_t InitRecording() override;
>-    bool RecordingIsInitialized() const override;
>-
>-    // Audio transport control
>-    int32_t StartPlayout() override;
>-    int32_t StopPlayout() override;
>-    bool Playing() const override;
>-    int32_t StartRecording() override;
>-    int32_t StopRecording() override;
>-    bool Recording() const override;
>-
>-    // Audio mixer initialization
>-    int32_t InitSpeaker() override;
>-    bool SpeakerIsInitialized() const override;
>-    int32_t InitMicrophone() override;
>-    bool MicrophoneIsInitialized() const override;
>-
>-    // Speaker volume controls
>-    int32_t SpeakerVolumeIsAvailable(bool& available) override;
>-    int32_t SetSpeakerVolume(uint32_t volume) override;
>-    int32_t SpeakerVolume(uint32_t& volume) const override;
>-    int32_t MaxSpeakerVolume(uint32_t& maxVolume) const override;
>-    int32_t MinSpeakerVolume(uint32_t& minVolume) const override;
>-
>-    // Microphone volume controls
>-    int32_t MicrophoneVolumeIsAvailable(bool& available) override;
>-    int32_t SetMicrophoneVolume(uint32_t volume) override;
>-    int32_t MicrophoneVolume(uint32_t& volume) const override;
>-    int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const override;
>-    int32_t MinMicrophoneVolume(uint32_t& minVolume) const override;
>-
>-    // Speaker mute control
>-    int32_t SpeakerMuteIsAvailable(bool& available) override;
>-    int32_t SetSpeakerMute(bool enable) override;
>-    int32_t SpeakerMute(bool& enabled) const override;
>-
>-    // Microphone mute control
>-    int32_t MicrophoneMuteIsAvailable(bool& available) override;
>-    int32_t SetMicrophoneMute(bool enable) override;
>-    int32_t MicrophoneMute(bool& enabled) const override;
>-
>-    // Stereo support
>-    int32_t StereoPlayoutIsAvailable(bool& available) override;
>-    int32_t SetStereoPlayout(bool enable) override;
>-    int32_t StereoPlayout(bool& enabled) const override;
>-    int32_t StereoRecordingIsAvailable(bool& available) override;
>-    int32_t SetStereoRecording(bool enable) override;
>-    int32_t StereoRecording(bool& enabled) const override;
>-
>-    // Delay information and control
>-    int32_t PlayoutDelay(uint16_t& delayMS) const override;
>-
>-    void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override;
>-
>-private:
>-    int32_t GetDevicesInfo(const int32_t function,
>-                           const bool playback,
>-                           const int32_t enumDeviceNo = 0,
>-                           char* enumDeviceName = NULL,
>-                           const int32_t ednLen = 0) const;
>-    int32_t ErrorRecovery(int32_t error, snd_pcm_t* deviceHandle);
>-
>-    bool KeyPressed() const;
>-
>-    void Lock() RTC_EXCLUSIVE_LOCK_FUNCTION(_critSect) { _critSect.Enter(); };
>-    void UnLock() RTC_UNLOCK_FUNCTION(_critSect) { _critSect.Leave(); };
>-
>-    inline int32_t InputSanityCheckAfterUnlockedPeriod() const;
>-    inline int32_t OutputSanityCheckAfterUnlockedPeriod() const;
>-
>-    static bool RecThreadFunc(void*);
>-    static bool PlayThreadFunc(void*);
>-    bool RecThreadProcess();
>-    bool PlayThreadProcess();
>-
>-    AudioDeviceBuffer* _ptrAudioBuffer;
>-
>-    rtc::CriticalSection _critSect;
>-
>-    // TODO(pbos): Make plain members and start/stop instead of resetting these
>-    // pointers. A thread can be reused.
>-    std::unique_ptr<rtc::PlatformThread> _ptrThreadRec;
>-    std::unique_ptr<rtc::PlatformThread> _ptrThreadPlay;
>-
>-    AudioMixerManagerLinuxALSA _mixerManager;
>-
>-    uint16_t _inputDeviceIndex;
>-    uint16_t _outputDeviceIndex;
>-    bool _inputDeviceIsSpecified;
>-    bool _outputDeviceIsSpecified;
>-
>-    snd_pcm_t* _handleRecord;
>-    snd_pcm_t* _handlePlayout;
>-
>-    snd_pcm_uframes_t _recordingBuffersizeInFrame;
>-    snd_pcm_uframes_t _recordingPeriodSizeInFrame;
>-    snd_pcm_uframes_t _playoutBufferSizeInFrame;
>-    snd_pcm_uframes_t _playoutPeriodSizeInFrame;
>-
>-    ssize_t _recordingBufferSizeIn10MS;
>-    ssize_t _playoutBufferSizeIn10MS;
>-    uint32_t _recordingFramesIn10MS;
>-    uint32_t _playoutFramesIn10MS;
>-
>-    uint32_t _recordingFreq;
>-    uint32_t _playoutFreq;
>-    uint8_t _recChannels;
>-    uint8_t _playChannels;
>-
>-    int8_t* _recordingBuffer; // in byte
>-    int8_t* _playoutBuffer; // in byte
>-    uint32_t _recordingFramesLeft;
>-    uint32_t _playoutFramesLeft;
>-
>-    bool _initialized;
>-    bool _recording;
>-    bool _playing;
>-    bool _recIsInitialized;
>-    bool _playIsInitialized;
>-
>-    snd_pcm_sframes_t _recordingDelay;
>-    snd_pcm_sframes_t _playoutDelay;
>-
>-    char _oldKeyState[32];
>+
>+  // Device selection
>+  int32_t SetPlayoutDevice(uint16_t index) override;
>+  int32_t SetPlayoutDevice(
>+      AudioDeviceModule::WindowsDeviceType device) override;
>+  int32_t SetRecordingDevice(uint16_t index) override;
>+  int32_t SetRecordingDevice(
>+      AudioDeviceModule::WindowsDeviceType device) override;
>+
>+  // Audio transport initialization
>+  int32_t PlayoutIsAvailable(bool& available) override;
>+  int32_t InitPlayout() override;
>+  bool PlayoutIsInitialized() const override;
>+  int32_t RecordingIsAvailable(bool& available) override;
>+  int32_t InitRecording() override;
>+  bool RecordingIsInitialized() const override;
>+
>+  // Audio transport control
>+  int32_t StartPlayout() override;
>+  int32_t StopPlayout() override;
>+  bool Playing() const override;
>+  int32_t StartRecording() override;
>+  int32_t StopRecording() override;
>+  bool Recording() const override;
>+
>+  // Audio mixer initialization
>+  int32_t InitSpeaker() override;
>+  bool SpeakerIsInitialized() const override;
>+  int32_t InitMicrophone() override;
>+  bool MicrophoneIsInitialized() const override;
>+
>+  // Speaker volume controls
>+  int32_t SpeakerVolumeIsAvailable(bool& available) override;
>+  int32_t SetSpeakerVolume(uint32_t volume) override;
>+  int32_t SpeakerVolume(uint32_t& volume) const override;
>+  int32_t MaxSpeakerVolume(uint32_t& maxVolume) const override;
>+  int32_t MinSpeakerVolume(uint32_t& minVolume) const override;
>+
>+  // Microphone volume controls
>+  int32_t MicrophoneVolumeIsAvailable(bool& available) override;
>+  int32_t SetMicrophoneVolume(uint32_t volume) override;
>+  int32_t MicrophoneVolume(uint32_t& volume) const override;
>+  int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const override;
>+  int32_t MinMicrophoneVolume(uint32_t& minVolume) const override;
>+
>+  // Speaker mute control
>+  int32_t SpeakerMuteIsAvailable(bool& available) override;
>+  int32_t SetSpeakerMute(bool enable) override;
>+  int32_t SpeakerMute(bool& enabled) const override;
>+
>+  // Microphone mute control
>+  int32_t MicrophoneMuteIsAvailable(bool& available) override;
>+  int32_t SetMicrophoneMute(bool enable) override;
>+  int32_t MicrophoneMute(bool& enabled) const override;
>+
>+  // Stereo support
>+  int32_t StereoPlayoutIsAvailable(bool& available) override;
>+  int32_t SetStereoPlayout(bool enable) override;
>+  int32_t StereoPlayout(bool& enabled) const override;
>+  int32_t StereoRecordingIsAvailable(bool& available) override;
>+  int32_t SetStereoRecording(bool enable) override;
>+  int32_t StereoRecording(bool& enabled) const override;
>+
>+  // Delay information and control
>+  int32_t PlayoutDelay(uint16_t& delayMS) const override;
>+
>+  void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override;
>+
>+ private:
>+  int32_t GetDevicesInfo(const int32_t function,
>+                         const bool playback,
>+                         const int32_t enumDeviceNo = 0,
>+                         char* enumDeviceName = NULL,
>+                         const int32_t ednLen = 0) const;
>+  int32_t ErrorRecovery(int32_t error, snd_pcm_t* deviceHandle);
>+
>+  bool KeyPressed() const;
>+
>+  void Lock() RTC_EXCLUSIVE_LOCK_FUNCTION(_critSect) { _critSect.Enter(); };
>+  void UnLock() RTC_UNLOCK_FUNCTION(_critSect) { _critSect.Leave(); };
>+
>+  inline int32_t InputSanityCheckAfterUnlockedPeriod() const;
>+  inline int32_t OutputSanityCheckAfterUnlockedPeriod() const;
>+
>+  static bool RecThreadFunc(void*);
>+  static bool PlayThreadFunc(void*);
>+  bool RecThreadProcess();
>+  bool PlayThreadProcess();
>+
>+  AudioDeviceBuffer* _ptrAudioBuffer;
>+
>+  rtc::CriticalSection _critSect;
>+
>+  // TODO(pbos): Make plain members and start/stop instead of resetting these
>+  // pointers. A thread can be reused.
>+  std::unique_ptr<rtc::PlatformThread> _ptrThreadRec;
>+  std::unique_ptr<rtc::PlatformThread> _ptrThreadPlay;
>+
>+  AudioMixerManagerLinuxALSA _mixerManager;
>+
>+  uint16_t _inputDeviceIndex;
>+  uint16_t _outputDeviceIndex;
>+  bool _inputDeviceIsSpecified;
>+  bool _outputDeviceIsSpecified;
>+
>+  snd_pcm_t* _handleRecord;
>+  snd_pcm_t* _handlePlayout;
>+
>+  snd_pcm_uframes_t _recordingBuffersizeInFrame;
>+  snd_pcm_uframes_t _recordingPeriodSizeInFrame;
>+  snd_pcm_uframes_t _playoutBufferSizeInFrame;
>+  snd_pcm_uframes_t _playoutPeriodSizeInFrame;
>+
>+  ssize_t _recordingBufferSizeIn10MS;
>+  ssize_t _playoutBufferSizeIn10MS;
>+  uint32_t _recordingFramesIn10MS;
>+  uint32_t _playoutFramesIn10MS;
>+
>+  uint32_t _recordingFreq;
>+  uint32_t _playoutFreq;
>+  uint8_t _recChannels;
>+  uint8_t _playChannels;
>+
>+  int8_t* _recordingBuffer;  // in byte
>+  int8_t* _playoutBuffer;    // in byte
>+  uint32_t _recordingFramesLeft;
>+  uint32_t _playoutFramesLeft;
>+
>+  bool _initialized;
>+  bool _recording;
>+  bool _playing;
>+  bool _recIsInitialized;
>+  bool _playIsInitialized;
>+
>+  snd_pcm_sframes_t _recordingDelay;
>+  snd_pcm_sframes_t _playoutDelay;
>+
>+  char _oldKeyState[32];
> #if defined(WEBRTC_USE_X11)
>-    Display* _XDisplay;
>+  Display* _XDisplay;
> #endif
> };
> 
>-}
>+}  // namespace webrtc
> 
> #endif  // MODULES_AUDIO_DEVICE_MAIN_SOURCE_LINUX_AUDIO_DEVICE_ALSA_LINUX_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/audio_device_pulse_linux.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/audio_device_pulse_linux.cc
>index 33eeee929bd..2f343bae069 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/audio_device_pulse_linux.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/audio_device_pulse_linux.cc
>@@ -155,12 +155,14 @@ AudioDeviceGeneric::InitStatus AudioDeviceLinuxPulse::Init() {
>     return InitStatus::OTHER_ERROR;
>   }
> 
>+#if defined(WEBRTC_USE_X11)
>   // Get X display handle for typing detection
>   _XDisplay = XOpenDisplay(NULL);
>   if (!_XDisplay) {
>     RTC_LOG(LS_WARNING)
>         << "failed to open X display, typing detection will not work";
>   }
>+#endif
> 
>   // RECORDING
>   _ptrThreadRec.reset(new rtc::PlatformThread(
>@@ -212,10 +214,12 @@ int32_t AudioDeviceLinuxPulse::Terminate() {
>     return -1;
>   }
> 
>+#if defined(WEBRTC_USE_X11)
>   if (_XDisplay) {
>     XCloseDisplay(_XDisplay);
>     _XDisplay = NULL;
>   }
>+#endif
> 
>   _initialized = false;
>   _outputDeviceIsSpecified = false;
>@@ -1951,8 +1955,7 @@ int32_t AudioDeviceLinuxPulse::ReadRecordedData(const void* bufferData,
> int32_t AudioDeviceLinuxPulse::ProcessRecordedData(int8_t* bufferData,
>                                                    uint32_t bufferSizeInSamples,
>                                                    uint32_t recDelay)
>-  RTC_EXCLUSIVE_LOCKS_REQUIRED(_critSect) {
>-
>+    RTC_EXCLUSIVE_LOCKS_REQUIRED(_critSect) {
>   _ptrAudioBuffer->SetRecordedBuffer(bufferData, bufferSizeInSamples);
> 
>   // TODO(andrew): this is a temporary hack, to avoid non-causal far- and
>@@ -2281,6 +2284,7 @@ bool AudioDeviceLinuxPulse::RecThreadProcess() {
> }
> 
> bool AudioDeviceLinuxPulse::KeyPressed() const {
>+#if defined(WEBRTC_USE_X11)
>   char szKey[32];
>   unsigned int i = 0;
>   char state = 0;
>@@ -2298,5 +2302,8 @@ bool AudioDeviceLinuxPulse::KeyPressed() const {
>   // Save old state
>   memcpy((char*)_oldKeyState, (char*)szKey, sizeof(_oldKeyState));
>   return (state != 0);
>+#else
>+  return false;
>+#endif
> }
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/audio_device_pulse_linux.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/audio_device_pulse_linux.h
>index 9aadfbe0dff..854c7c6f2fd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/audio_device_pulse_linux.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/audio_device_pulse_linux.h
>@@ -19,7 +19,10 @@
> #include "rtc_base/platform_thread.h"
> #include "rtc_base/thread_checker.h"
> 
>+#if defined(WEBRTC_USE_X11)
> #include <X11/Xlib.h>
>+#endif
>+
> #include <pulse/pulseaudio.h>
> 
> // We define this flag if it's missing from our headers, because we want to be
>@@ -85,253 +88,254 @@ const uint32_t WEBRTC_PA_MSECS_PER_SEC = 1000;
> // Init _configuredLatencyRec/Play to this value to disable latency requirements
> const int32_t WEBRTC_PA_NO_LATENCY_REQUIREMENTS = -1;
> 
>-// Set this const to 1 to account for peeked and used data in latency calculation
>+// Set this const to 1 to account for peeked and used data in latency
>+// calculation
> const uint32_t WEBRTC_PA_CAPTURE_BUFFER_LATENCY_ADJUSTMENT = 0;
> 
>-namespace webrtc
>-{
>+namespace webrtc {
> class EventWrapper;
> 
>-class AudioDeviceLinuxPulse: public AudioDeviceGeneric
>-{
>-public:
>-    AudioDeviceLinuxPulse();
>-    virtual ~AudioDeviceLinuxPulse();
>-
>-    // Retrieve the currently utilized audio layer
>-    int32_t ActiveAudioLayer(
>-        AudioDeviceModule::AudioLayer& audioLayer) const override;
>-
>-    // Main initializaton and termination
>-    InitStatus Init() override;
>-    int32_t Terminate() override;
>-    bool Initialized() const override;
>-
>-    // Device enumeration
>-    int16_t PlayoutDevices() override;
>-    int16_t RecordingDevices() override;
>-    int32_t PlayoutDeviceName(uint16_t index,
>+class AudioDeviceLinuxPulse : public AudioDeviceGeneric {
>+ public:
>+  AudioDeviceLinuxPulse();
>+  virtual ~AudioDeviceLinuxPulse();
>+
>+  // Retrieve the currently utilized audio layer
>+  int32_t ActiveAudioLayer(
>+      AudioDeviceModule::AudioLayer& audioLayer) const override;
>+
>+  // Main initializaton and termination
>+  InitStatus Init() override;
>+  int32_t Terminate() override;
>+  bool Initialized() const override;
>+
>+  // Device enumeration
>+  int16_t PlayoutDevices() override;
>+  int16_t RecordingDevices() override;
>+  int32_t PlayoutDeviceName(uint16_t index,
>+                            char name[kAdmMaxDeviceNameSize],
>+                            char guid[kAdmMaxGuidSize]) override;
>+  int32_t RecordingDeviceName(uint16_t index,
>                               char name[kAdmMaxDeviceNameSize],
>                               char guid[kAdmMaxGuidSize]) override;
>-    int32_t RecordingDeviceName(uint16_t index,
>-                                char name[kAdmMaxDeviceNameSize],
>-                                char guid[kAdmMaxGuidSize]) override;
>-
>-    // Device selection
>-    int32_t SetPlayoutDevice(uint16_t index) override;
>-    int32_t SetPlayoutDevice(
>-        AudioDeviceModule::WindowsDeviceType device) override;
>-    int32_t SetRecordingDevice(uint16_t index) override;
>-    int32_t SetRecordingDevice(
>-        AudioDeviceModule::WindowsDeviceType device) override;
>-
>-    // Audio transport initialization
>-    int32_t PlayoutIsAvailable(bool& available) override;
>-    int32_t InitPlayout() override;
>-    bool PlayoutIsInitialized() const override;
>-    int32_t RecordingIsAvailable(bool& available) override;
>-    int32_t InitRecording() override;
>-    bool RecordingIsInitialized() const override;
>-
>-    // Audio transport control
>-    int32_t StartPlayout() override;
>-    int32_t StopPlayout() override;
>-    bool Playing() const override;
>-    int32_t StartRecording() override;
>-    int32_t StopRecording() override;
>-    bool Recording() const override;
>-
>-    // Audio mixer initialization
>-    int32_t InitSpeaker() override;
>-    bool SpeakerIsInitialized() const override;
>-    int32_t InitMicrophone() override;
>-    bool MicrophoneIsInitialized() const override;
>-
>-    // Speaker volume controls
>-    int32_t SpeakerVolumeIsAvailable(bool& available) override;
>-    int32_t SetSpeakerVolume(uint32_t volume) override;
>-    int32_t SpeakerVolume(uint32_t& volume) const override;
>-    int32_t MaxSpeakerVolume(uint32_t& maxVolume) const override;
>-    int32_t MinSpeakerVolume(uint32_t& minVolume) const override;
>-
>-    // Microphone volume controls
>-    int32_t MicrophoneVolumeIsAvailable(bool& available) override;
>-    int32_t SetMicrophoneVolume(uint32_t volume) override;
>-    int32_t MicrophoneVolume(uint32_t& volume) const override;
>-    int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const override;
>-    int32_t MinMicrophoneVolume(uint32_t& minVolume) const override;
>-
>-    // Speaker mute control
>-    int32_t SpeakerMuteIsAvailable(bool& available) override;
>-    int32_t SetSpeakerMute(bool enable) override;
>-    int32_t SpeakerMute(bool& enabled) const override;
>-
>-    // Microphone mute control
>-    int32_t MicrophoneMuteIsAvailable(bool& available) override;
>-    int32_t SetMicrophoneMute(bool enable) override;
>-    int32_t MicrophoneMute(bool& enabled) const override;
>-
>-    // Stereo support
>-    int32_t StereoPlayoutIsAvailable(bool& available) override;
>-    int32_t SetStereoPlayout(bool enable) override;
>-    int32_t StereoPlayout(bool& enabled) const override;
>-    int32_t StereoRecordingIsAvailable(bool& available) override;
>-    int32_t SetStereoRecording(bool enable) override;
>-    int32_t StereoRecording(bool& enabled) const override;
>-
>-    // Delay information and control
>-    int32_t PlayoutDelay(uint16_t& delayMS) const override;
>-
>-   void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override;
>-
>-private:
>- void Lock() RTC_EXCLUSIVE_LOCK_FUNCTION(_critSect) { _critSect.Enter(); }
>- void UnLock() RTC_UNLOCK_FUNCTION(_critSect) { _critSect.Leave(); }
>- void WaitForOperationCompletion(pa_operation* paOperation) const;
>- void WaitForSuccess(pa_operation* paOperation) const;
>-
>- bool KeyPressed() const;
>-
>- static void PaContextStateCallback(pa_context* c, void* pThis);
>- static void PaSinkInfoCallback(pa_context* c,
>-                                const pa_sink_info* i,
>-                                int eol,
>-                                void* pThis);
>- static void PaSourceInfoCallback(pa_context* c,
>-                                  const pa_source_info* i,
>-                                  int eol,
>-                                  void* pThis);
>- static void PaServerInfoCallback(pa_context* c,
>-                                  const pa_server_info* i,
>-                                  void* pThis);
>- static void PaStreamStateCallback(pa_stream* p, void* pThis);
>- void PaContextStateCallbackHandler(pa_context* c);
>- void PaSinkInfoCallbackHandler(const pa_sink_info* i, int eol);
>- void PaSourceInfoCallbackHandler(const pa_source_info* i, int eol);
>- void PaServerInfoCallbackHandler(const pa_server_info* i);
>- void PaStreamStateCallbackHandler(pa_stream* p);
>-
>- void EnableWriteCallback();
>- void DisableWriteCallback();
>- static void PaStreamWriteCallback(pa_stream* unused,
>-                                   size_t buffer_space,
>+
>+  // Device selection
>+  int32_t SetPlayoutDevice(uint16_t index) override;
>+  int32_t SetPlayoutDevice(
>+      AudioDeviceModule::WindowsDeviceType device) override;
>+  int32_t SetRecordingDevice(uint16_t index) override;
>+  int32_t SetRecordingDevice(
>+      AudioDeviceModule::WindowsDeviceType device) override;
>+
>+  // Audio transport initialization
>+  int32_t PlayoutIsAvailable(bool& available) override;
>+  int32_t InitPlayout() override;
>+  bool PlayoutIsInitialized() const override;
>+  int32_t RecordingIsAvailable(bool& available) override;
>+  int32_t InitRecording() override;
>+  bool RecordingIsInitialized() const override;
>+
>+  // Audio transport control
>+  int32_t StartPlayout() override;
>+  int32_t StopPlayout() override;
>+  bool Playing() const override;
>+  int32_t StartRecording() override;
>+  int32_t StopRecording() override;
>+  bool Recording() const override;
>+
>+  // Audio mixer initialization
>+  int32_t InitSpeaker() override;
>+  bool SpeakerIsInitialized() const override;
>+  int32_t InitMicrophone() override;
>+  bool MicrophoneIsInitialized() const override;
>+
>+  // Speaker volume controls
>+  int32_t SpeakerVolumeIsAvailable(bool& available) override;
>+  int32_t SetSpeakerVolume(uint32_t volume) override;
>+  int32_t SpeakerVolume(uint32_t& volume) const override;
>+  int32_t MaxSpeakerVolume(uint32_t& maxVolume) const override;
>+  int32_t MinSpeakerVolume(uint32_t& minVolume) const override;
>+
>+  // Microphone volume controls
>+  int32_t MicrophoneVolumeIsAvailable(bool& available) override;
>+  int32_t SetMicrophoneVolume(uint32_t volume) override;
>+  int32_t MicrophoneVolume(uint32_t& volume) const override;
>+  int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const override;
>+  int32_t MinMicrophoneVolume(uint32_t& minVolume) const override;
>+
>+  // Speaker mute control
>+  int32_t SpeakerMuteIsAvailable(bool& available) override;
>+  int32_t SetSpeakerMute(bool enable) override;
>+  int32_t SpeakerMute(bool& enabled) const override;
>+
>+  // Microphone mute control
>+  int32_t MicrophoneMuteIsAvailable(bool& available) override;
>+  int32_t SetMicrophoneMute(bool enable) override;
>+  int32_t MicrophoneMute(bool& enabled) const override;
>+
>+  // Stereo support
>+  int32_t StereoPlayoutIsAvailable(bool& available) override;
>+  int32_t SetStereoPlayout(bool enable) override;
>+  int32_t StereoPlayout(bool& enabled) const override;
>+  int32_t StereoRecordingIsAvailable(bool& available) override;
>+  int32_t SetStereoRecording(bool enable) override;
>+  int32_t StereoRecording(bool& enabled) const override;
>+
>+  // Delay information and control
>+  int32_t PlayoutDelay(uint16_t& delayMS) const override;
>+
>+  void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override;
>+
>+ private:
>+  void Lock() RTC_EXCLUSIVE_LOCK_FUNCTION(_critSect) { _critSect.Enter(); }
>+  void UnLock() RTC_UNLOCK_FUNCTION(_critSect) { _critSect.Leave(); }
>+  void WaitForOperationCompletion(pa_operation* paOperation) const;
>+  void WaitForSuccess(pa_operation* paOperation) const;
>+
>+  bool KeyPressed() const;
>+
>+  static void PaContextStateCallback(pa_context* c, void* pThis);
>+  static void PaSinkInfoCallback(pa_context* c,
>+                                 const pa_sink_info* i,
>+                                 int eol,
>+                                 void* pThis);
>+  static void PaSourceInfoCallback(pa_context* c,
>+                                   const pa_source_info* i,
>+                                   int eol,
>                                    void* pThis);
>- void PaStreamWriteCallbackHandler(size_t buffer_space);
>- static void PaStreamUnderflowCallback(pa_stream* unused, void* pThis);
>- void PaStreamUnderflowCallbackHandler();
>- void EnableReadCallback();
>- void DisableReadCallback();
>- static void PaStreamReadCallback(pa_stream* unused1,
>-                                  size_t unused2,
>-                                  void* pThis);
>- void PaStreamReadCallbackHandler();
>- static void PaStreamOverflowCallback(pa_stream* unused, void* pThis);
>- void PaStreamOverflowCallbackHandler();
>- int32_t LatencyUsecs(pa_stream* stream);
>- int32_t ReadRecordedData(const void* bufferData, size_t bufferSize);
>- int32_t ProcessRecordedData(int8_t* bufferData,
>-                             uint32_t bufferSizeInSamples,
>-                             uint32_t recDelay);
>-
>- int32_t CheckPulseAudioVersion();
>- int32_t InitSamplingFrequency();
>- int32_t GetDefaultDeviceInfo(bool recDevice, char* name, uint16_t& index);
>- int32_t InitPulseAudio();
>- int32_t TerminatePulseAudio();
>-
>- void PaLock();
>- void PaUnLock();
>-
>- static bool RecThreadFunc(void*);
>- static bool PlayThreadFunc(void*);
>- bool RecThreadProcess();
>- bool PlayThreadProcess();
>-
>- AudioDeviceBuffer* _ptrAudioBuffer;
>-
>- rtc::CriticalSection _critSect;
>- EventWrapper& _timeEventRec;
>- EventWrapper& _timeEventPlay;
>- EventWrapper& _recStartEvent;
>- EventWrapper& _playStartEvent;
>-
>- // TODO(pbos): Remove unique_ptr and use directly without resetting.
>- std::unique_ptr<rtc::PlatformThread> _ptrThreadPlay;
>- std::unique_ptr<rtc::PlatformThread> _ptrThreadRec;
>-
>- AudioMixerManagerLinuxPulse _mixerManager;
>-
>- uint16_t _inputDeviceIndex;
>- uint16_t _outputDeviceIndex;
>- bool _inputDeviceIsSpecified;
>- bool _outputDeviceIsSpecified;
>-
>- int sample_rate_hz_;
>- uint8_t _recChannels;
>- uint8_t _playChannels;
>-
>- // Stores thread ID in constructor.
>- // We can then use ThreadChecker::CalledOnValidThread() to ensure that
>- // other methods are called from the same thread.
>- // Currently only does RTC_DCHECK(thread_checker_.CalledOnValidThread()).
>- rtc::ThreadChecker thread_checker_;
>-
>- bool _initialized;
>- bool _recording;
>- bool _playing;
>- bool _recIsInitialized;
>- bool _playIsInitialized;
>- bool _startRec;
>- bool _stopRec;
>- bool _startPlay;
>- bool _stopPlay;
>- bool update_speaker_volume_at_startup_;
>-
>- uint32_t _sndCardPlayDelay;
>- uint32_t _sndCardRecDelay;
>-
>- int32_t _writeErrors;
>-
>- uint16_t _deviceIndex;
>- int16_t _numPlayDevices;
>- int16_t _numRecDevices;
>- char* _playDeviceName;
>- char* _recDeviceName;
>- char* _playDisplayDeviceName;
>- char* _recDisplayDeviceName;
>- char _paServerVersion[32];
>-
>- int8_t* _playBuffer;
>- size_t _playbackBufferSize;
>- size_t _playbackBufferUnused;
>- size_t _tempBufferSpace;
>- int8_t* _recBuffer;
>- size_t _recordBufferSize;
>- size_t _recordBufferUsed;
>- const void* _tempSampleData;
>- size_t _tempSampleDataSize;
>- int32_t _configuredLatencyPlay;
>- int32_t _configuredLatencyRec;
>-
>- // PulseAudio
>- uint16_t _paDeviceIndex;
>- bool _paStateChanged;
>-
>- pa_threaded_mainloop* _paMainloop;
>- pa_mainloop_api* _paMainloopApi;
>- pa_context* _paContext;
>-
>- pa_stream* _recStream;
>- pa_stream* _playStream;
>- uint32_t _recStreamFlags;
>- uint32_t _playStreamFlags;
>- pa_buffer_attr _playBufferAttr;
>- pa_buffer_attr _recBufferAttr;
>-
>- char _oldKeyState[32];
>- Display* _XDisplay;
>+  static void PaServerInfoCallback(pa_context* c,
>+                                   const pa_server_info* i,
>+                                   void* pThis);
>+  static void PaStreamStateCallback(pa_stream* p, void* pThis);
>+  void PaContextStateCallbackHandler(pa_context* c);
>+  void PaSinkInfoCallbackHandler(const pa_sink_info* i, int eol);
>+  void PaSourceInfoCallbackHandler(const pa_source_info* i, int eol);
>+  void PaServerInfoCallbackHandler(const pa_server_info* i);
>+  void PaStreamStateCallbackHandler(pa_stream* p);
>+
>+  void EnableWriteCallback();
>+  void DisableWriteCallback();
>+  static void PaStreamWriteCallback(pa_stream* unused,
>+                                    size_t buffer_space,
>+                                    void* pThis);
>+  void PaStreamWriteCallbackHandler(size_t buffer_space);
>+  static void PaStreamUnderflowCallback(pa_stream* unused, void* pThis);
>+  void PaStreamUnderflowCallbackHandler();
>+  void EnableReadCallback();
>+  void DisableReadCallback();
>+  static void PaStreamReadCallback(pa_stream* unused1,
>+                                   size_t unused2,
>+                                   void* pThis);
>+  void PaStreamReadCallbackHandler();
>+  static void PaStreamOverflowCallback(pa_stream* unused, void* pThis);
>+  void PaStreamOverflowCallbackHandler();
>+  int32_t LatencyUsecs(pa_stream* stream);
>+  int32_t ReadRecordedData(const void* bufferData, size_t bufferSize);
>+  int32_t ProcessRecordedData(int8_t* bufferData,
>+                              uint32_t bufferSizeInSamples,
>+                              uint32_t recDelay);
>+
>+  int32_t CheckPulseAudioVersion();
>+  int32_t InitSamplingFrequency();
>+  int32_t GetDefaultDeviceInfo(bool recDevice, char* name, uint16_t& index);
>+  int32_t InitPulseAudio();
>+  int32_t TerminatePulseAudio();
>+
>+  void PaLock();
>+  void PaUnLock();
>+
>+  static bool RecThreadFunc(void*);
>+  static bool PlayThreadFunc(void*);
>+  bool RecThreadProcess();
>+  bool PlayThreadProcess();
>+
>+  AudioDeviceBuffer* _ptrAudioBuffer;
>+
>+  rtc::CriticalSection _critSect;
>+  EventWrapper& _timeEventRec;
>+  EventWrapper& _timeEventPlay;
>+  EventWrapper& _recStartEvent;
>+  EventWrapper& _playStartEvent;
>+
>+  // TODO(pbos): Remove unique_ptr and use directly without resetting.
>+  std::unique_ptr<rtc::PlatformThread> _ptrThreadPlay;
>+  std::unique_ptr<rtc::PlatformThread> _ptrThreadRec;
>+
>+  AudioMixerManagerLinuxPulse _mixerManager;
>+
>+  uint16_t _inputDeviceIndex;
>+  uint16_t _outputDeviceIndex;
>+  bool _inputDeviceIsSpecified;
>+  bool _outputDeviceIsSpecified;
>+
>+  int sample_rate_hz_;
>+  uint8_t _recChannels;
>+  uint8_t _playChannels;
>+
>+  // Stores thread ID in constructor.
>+  // We can then use ThreadChecker::CalledOnValidThread() to ensure that
>+  // other methods are called from the same thread.
>+  // Currently only does RTC_DCHECK(thread_checker_.CalledOnValidThread()).
>+  rtc::ThreadChecker thread_checker_;
>+
>+  bool _initialized;
>+  bool _recording;
>+  bool _playing;
>+  bool _recIsInitialized;
>+  bool _playIsInitialized;
>+  bool _startRec;
>+  bool _stopRec;
>+  bool _startPlay;
>+  bool _stopPlay;
>+  bool update_speaker_volume_at_startup_;
>+
>+  uint32_t _sndCardPlayDelay;
>+  uint32_t _sndCardRecDelay;
>+
>+  int32_t _writeErrors;
>+
>+  uint16_t _deviceIndex;
>+  int16_t _numPlayDevices;
>+  int16_t _numRecDevices;
>+  char* _playDeviceName;
>+  char* _recDeviceName;
>+  char* _playDisplayDeviceName;
>+  char* _recDisplayDeviceName;
>+  char _paServerVersion[32];
>+
>+  int8_t* _playBuffer;
>+  size_t _playbackBufferSize;
>+  size_t _playbackBufferUnused;
>+  size_t _tempBufferSpace;
>+  int8_t* _recBuffer;
>+  size_t _recordBufferSize;
>+  size_t _recordBufferUsed;
>+  const void* _tempSampleData;
>+  size_t _tempSampleDataSize;
>+  int32_t _configuredLatencyPlay;
>+  int32_t _configuredLatencyRec;
>+
>+  // PulseAudio
>+  uint16_t _paDeviceIndex;
>+  bool _paStateChanged;
>+
>+  pa_threaded_mainloop* _paMainloop;
>+  pa_mainloop_api* _paMainloopApi;
>+  pa_context* _paContext;
>+
>+  pa_stream* _recStream;
>+  pa_stream* _playStream;
>+  uint32_t _recStreamFlags;
>+  uint32_t _playStreamFlags;
>+  pa_buffer_attr _playBufferAttr;
>+  pa_buffer_attr _recBufferAttr;
>+
>+  char _oldKeyState[32];
>+#if defined(WEBRTC_USE_X11)
>+  Display* _XDisplay;
>+#endif
> };
> 
>-}
>+}  // namespace webrtc
> 
> #endif  // MODULES_AUDIO_DEVICE_MAIN_SOURCE_LINUX_AUDIO_DEVICE_PULSE_LINUX_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/audio_mixer_manager_alsa_linux.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/audio_mixer_manager_alsa_linux.h
>index 61de607cf70..5d40e08bf56 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/audio_mixer_manager_alsa_linux.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/audio_mixer_manager_alsa_linux.h
>@@ -14,59 +14,56 @@
> #include "modules/audio_device/include/audio_device.h"
> #include "modules/audio_device/linux/alsasymboltable_linux.h"
> #include "rtc_base/criticalsection.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> #include <alsa/asoundlib.h>
> 
>-namespace webrtc
>-{
>+namespace webrtc {
> 
>-class AudioMixerManagerLinuxALSA
>-{
>-public:
>-    int32_t OpenSpeaker(char* deviceName);
>-    int32_t OpenMicrophone(char* deviceName);
>-    int32_t SetSpeakerVolume(uint32_t volume);
>-    int32_t SpeakerVolume(uint32_t& volume) const;
>-    int32_t MaxSpeakerVolume(uint32_t& maxVolume) const;
>-    int32_t MinSpeakerVolume(uint32_t& minVolume) const;
>-    int32_t SpeakerVolumeIsAvailable(bool& available);
>-    int32_t SpeakerMuteIsAvailable(bool& available);
>-    int32_t SetSpeakerMute(bool enable);
>-    int32_t SpeakerMute(bool& enabled) const;
>-    int32_t MicrophoneMuteIsAvailable(bool& available);
>-    int32_t SetMicrophoneMute(bool enable);
>-    int32_t MicrophoneMute(bool& enabled) const;
>-    int32_t MicrophoneVolumeIsAvailable(bool& available);
>-    int32_t SetMicrophoneVolume(uint32_t volume);
>-    int32_t MicrophoneVolume(uint32_t& volume) const;
>-    int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const;
>-    int32_t MinMicrophoneVolume(uint32_t& minVolume) const;
>-    int32_t Close();
>-    int32_t CloseSpeaker();
>-    int32_t CloseMicrophone();
>-    bool SpeakerIsInitialized() const;
>-    bool MicrophoneIsInitialized() const;
>+class AudioMixerManagerLinuxALSA {
>+ public:
>+  int32_t OpenSpeaker(char* deviceName);
>+  int32_t OpenMicrophone(char* deviceName);
>+  int32_t SetSpeakerVolume(uint32_t volume);
>+  int32_t SpeakerVolume(uint32_t& volume) const;
>+  int32_t MaxSpeakerVolume(uint32_t& maxVolume) const;
>+  int32_t MinSpeakerVolume(uint32_t& minVolume) const;
>+  int32_t SpeakerVolumeIsAvailable(bool& available);
>+  int32_t SpeakerMuteIsAvailable(bool& available);
>+  int32_t SetSpeakerMute(bool enable);
>+  int32_t SpeakerMute(bool& enabled) const;
>+  int32_t MicrophoneMuteIsAvailable(bool& available);
>+  int32_t SetMicrophoneMute(bool enable);
>+  int32_t MicrophoneMute(bool& enabled) const;
>+  int32_t MicrophoneVolumeIsAvailable(bool& available);
>+  int32_t SetMicrophoneVolume(uint32_t volume);
>+  int32_t MicrophoneVolume(uint32_t& volume) const;
>+  int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const;
>+  int32_t MinMicrophoneVolume(uint32_t& minVolume) const;
>+  int32_t Close();
>+  int32_t CloseSpeaker();
>+  int32_t CloseMicrophone();
>+  bool SpeakerIsInitialized() const;
>+  bool MicrophoneIsInitialized() const;
> 
>-public:
>-    AudioMixerManagerLinuxALSA();
>-    ~AudioMixerManagerLinuxALSA();
>+ public:
>+  AudioMixerManagerLinuxALSA();
>+  ~AudioMixerManagerLinuxALSA();
> 
>-private:
>-    int32_t LoadMicMixerElement() const;
>-    int32_t LoadSpeakerMixerElement() const;
>-    void GetControlName(char *controlName, char* deviceName) const;
>+ private:
>+  int32_t LoadMicMixerElement() const;
>+  int32_t LoadSpeakerMixerElement() const;
>+  void GetControlName(char* controlName, char* deviceName) const;
> 
>-private:
>-    rtc::CriticalSection _critSect;
>-    mutable snd_mixer_t* _outputMixerHandle;
>-    char _outputMixerStr[kAdmMaxDeviceNameSize];
>-    mutable snd_mixer_t* _inputMixerHandle;
>-    char _inputMixerStr[kAdmMaxDeviceNameSize];
>-    mutable snd_mixer_elem_t* _outputMixerElement;
>-    mutable snd_mixer_elem_t* _inputMixerElement;
>+ private:
>+  rtc::CriticalSection _critSect;
>+  mutable snd_mixer_t* _outputMixerHandle;
>+  char _outputMixerStr[kAdmMaxDeviceNameSize];
>+  mutable snd_mixer_t* _inputMixerHandle;
>+  char _inputMixerStr[kAdmMaxDeviceNameSize];
>+  mutable snd_mixer_elem_t* _outputMixerElement;
>+  mutable snd_mixer_elem_t* _inputMixerElement;
> };
> 
>-}
>+}  // namespace webrtc
> 
> #endif  // MODULES_AUDIO_DEVICE_MAIN_SOURCE_LINUX_AUDIO_MIXER_MANAGER_ALSA_LINUX_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/audio_mixer_manager_pulse_linux.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/audio_mixer_manager_pulse_linux.h
>index 7846ccbcc04..c5460b6237a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/audio_mixer_manager_pulse_linux.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/audio_mixer_manager_pulse_linux.h
>@@ -14,100 +14,103 @@
> #include "modules/audio_device/include/audio_device.h"
> #include "modules/audio_device/linux/pulseaudiosymboltable_linux.h"
> #include "rtc_base/thread_checker.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> #include <pulse/pulseaudio.h>
> #include <stdint.h>
> 
> #ifndef UINT32_MAX
>-#define UINT32_MAX  ((uint32_t)-1)
>+#define UINT32_MAX ((uint32_t)-1)
> #endif
> 
>-namespace webrtc
>-{
>-
>-class AudioMixerManagerLinuxPulse
>-{
>-public:
>-    int32_t SetPlayStream(pa_stream* playStream);
>-    int32_t SetRecStream(pa_stream* recStream);
>-    int32_t OpenSpeaker(uint16_t deviceIndex);
>-    int32_t OpenMicrophone(uint16_t deviceIndex);
>-    int32_t SetSpeakerVolume(uint32_t volume);
>-    int32_t SpeakerVolume(uint32_t& volume) const;
>-    int32_t MaxSpeakerVolume(uint32_t& maxVolume) const;
>-    int32_t MinSpeakerVolume(uint32_t& minVolume) const;
>-    int32_t SpeakerVolumeIsAvailable(bool& available);
>-    int32_t SpeakerMuteIsAvailable(bool& available);
>-    int32_t SetSpeakerMute(bool enable);
>-    int32_t StereoPlayoutIsAvailable(bool& available);
>-    int32_t StereoRecordingIsAvailable(bool& available);
>-    int32_t SpeakerMute(bool& enabled) const;
>-    int32_t MicrophoneMuteIsAvailable(bool& available);
>-    int32_t SetMicrophoneMute(bool enable);
>-    int32_t MicrophoneMute(bool& enabled) const;
>-    int32_t MicrophoneVolumeIsAvailable(bool& available);
>-    int32_t SetMicrophoneVolume(uint32_t volume);
>-    int32_t MicrophoneVolume(uint32_t& volume) const;
>-    int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const;
>-    int32_t MinMicrophoneVolume(uint32_t& minVolume) const;
>-    int32_t SetPulseAudioObjects(pa_threaded_mainloop* mainloop,
>-                                 pa_context* context);
>-    int32_t Close();
>-    int32_t CloseSpeaker();
>-    int32_t CloseMicrophone();
>-    bool SpeakerIsInitialized() const;
>-    bool MicrophoneIsInitialized() const;
>-
>-public:
>-    AudioMixerManagerLinuxPulse();
>-    ~AudioMixerManagerLinuxPulse();
>-
>-private:
>-    static void PaSinkInfoCallback(pa_context *c, const pa_sink_info *i,
>-                                   int eol, void *pThis);
>-    static void PaSinkInputInfoCallback(pa_context *c,
>-                                        const pa_sink_input_info *i, int eol,
>-                                        void *pThis);
>-    static void PaSourceInfoCallback(pa_context *c, const pa_source_info *i,
>-                                     int eol, void *pThis);
>-    static void
>-        PaSetVolumeCallback(pa_context* /*c*/, int success, void* /*pThis*/);
>-    void PaSinkInfoCallbackHandler(const pa_sink_info *i, int eol);
>-    void PaSinkInputInfoCallbackHandler(const pa_sink_input_info *i, int eol);
>-    void PaSourceInfoCallbackHandler(const pa_source_info *i, int eol);
>-
>-    void WaitForOperationCompletion(pa_operation* paOperation) const;
>-
>-    bool GetSinkInputInfo() const;
>-    bool GetSinkInfoByIndex(int device_index)const ;
>-    bool GetSourceInfoByIndex(int device_index) const;
>-
>-private:
>-    int16_t _paOutputDeviceIndex;
>-    int16_t _paInputDeviceIndex;
>-
>-    pa_stream* _paPlayStream;
>-    pa_stream* _paRecStream;
>-
>-    pa_threaded_mainloop* _paMainloop;
>-    pa_context* _paContext;
>-
>-    mutable uint32_t _paVolume;
>-    mutable uint32_t _paMute;
>-    mutable uint32_t _paVolSteps;
>-    bool _paSpeakerMute;
>-    mutable uint32_t _paSpeakerVolume;
>-    mutable uint8_t _paChannels;
>-    bool _paObjectsSet;
>-
>-    // Stores thread ID in constructor.
>-    // We can then use ThreadChecker::CalledOnValidThread() to ensure that
>-    // other methods are called from the same thread.
>-    // Currently only does RTC_DCHECK(thread_checker_.CalledOnValidThread()).
>-    rtc::ThreadChecker thread_checker_;
>+namespace webrtc {
>+
>+class AudioMixerManagerLinuxPulse {
>+ public:
>+  int32_t SetPlayStream(pa_stream* playStream);
>+  int32_t SetRecStream(pa_stream* recStream);
>+  int32_t OpenSpeaker(uint16_t deviceIndex);
>+  int32_t OpenMicrophone(uint16_t deviceIndex);
>+  int32_t SetSpeakerVolume(uint32_t volume);
>+  int32_t SpeakerVolume(uint32_t& volume) const;
>+  int32_t MaxSpeakerVolume(uint32_t& maxVolume) const;
>+  int32_t MinSpeakerVolume(uint32_t& minVolume) const;
>+  int32_t SpeakerVolumeIsAvailable(bool& available);
>+  int32_t SpeakerMuteIsAvailable(bool& available);
>+  int32_t SetSpeakerMute(bool enable);
>+  int32_t StereoPlayoutIsAvailable(bool& available);
>+  int32_t StereoRecordingIsAvailable(bool& available);
>+  int32_t SpeakerMute(bool& enabled) const;
>+  int32_t MicrophoneMuteIsAvailable(bool& available);
>+  int32_t SetMicrophoneMute(bool enable);
>+  int32_t MicrophoneMute(bool& enabled) const;
>+  int32_t MicrophoneVolumeIsAvailable(bool& available);
>+  int32_t SetMicrophoneVolume(uint32_t volume);
>+  int32_t MicrophoneVolume(uint32_t& volume) const;
>+  int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const;
>+  int32_t MinMicrophoneVolume(uint32_t& minVolume) const;
>+  int32_t SetPulseAudioObjects(pa_threaded_mainloop* mainloop,
>+                               pa_context* context);
>+  int32_t Close();
>+  int32_t CloseSpeaker();
>+  int32_t CloseMicrophone();
>+  bool SpeakerIsInitialized() const;
>+  bool MicrophoneIsInitialized() const;
>+
>+ public:
>+  AudioMixerManagerLinuxPulse();
>+  ~AudioMixerManagerLinuxPulse();
>+
>+ private:
>+  static void PaSinkInfoCallback(pa_context* c,
>+                                 const pa_sink_info* i,
>+                                 int eol,
>+                                 void* pThis);
>+  static void PaSinkInputInfoCallback(pa_context* c,
>+                                      const pa_sink_input_info* i,
>+                                      int eol,
>+                                      void* pThis);
>+  static void PaSourceInfoCallback(pa_context* c,
>+                                   const pa_source_info* i,
>+                                   int eol,
>+                                   void* pThis);
>+  static void PaSetVolumeCallback(pa_context* /*c*/,
>+                                  int success,
>+                                  void* /*pThis*/);
>+  void PaSinkInfoCallbackHandler(const pa_sink_info* i, int eol);
>+  void PaSinkInputInfoCallbackHandler(const pa_sink_input_info* i, int eol);
>+  void PaSourceInfoCallbackHandler(const pa_source_info* i, int eol);
>+
>+  void WaitForOperationCompletion(pa_operation* paOperation) const;
>+
>+  bool GetSinkInputInfo() const;
>+  bool GetSinkInfoByIndex(int device_index) const;
>+  bool GetSourceInfoByIndex(int device_index) const;
>+
>+ private:
>+  int16_t _paOutputDeviceIndex;
>+  int16_t _paInputDeviceIndex;
>+
>+  pa_stream* _paPlayStream;
>+  pa_stream* _paRecStream;
>+
>+  pa_threaded_mainloop* _paMainloop;
>+  pa_context* _paContext;
>+
>+  mutable uint32_t _paVolume;
>+  mutable uint32_t _paMute;
>+  mutable uint32_t _paVolSteps;
>+  bool _paSpeakerMute;
>+  mutable uint32_t _paSpeakerVolume;
>+  mutable uint8_t _paChannels;
>+  bool _paObjectsSet;
>+
>+  // Stores thread ID in constructor.
>+  // We can then use ThreadChecker::CalledOnValidThread() to ensure that
>+  // other methods are called from the same thread.
>+  // Currently only does RTC_DCHECK(thread_checker_.CalledOnValidThread()).
>+  rtc::ThreadChecker thread_checker_;
> };
> 
>-}
>+}  // namespace webrtc
> 
> #endif  // MODULES_AUDIO_DEVICE_MAIN_SOURCE_LINUX_AUDIO_MIXER_MANAGER_PULSE_LINUX_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/latebindingsymboltable_linux.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/latebindingsymboltable_linux.h
>index b834f5c76e4..1e28c223cf4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/latebindingsymboltable_linux.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/latebindingsymboltable_linux.h
>@@ -26,7 +26,7 @@ namespace webrtc {
> namespace adm_linux {
> 
> #ifdef WEBRTC_LINUX
>-typedef void *DllHandle;
>+typedef void* DllHandle;
> 
> const DllHandle kInvalidDllHandle = NULL;
> #else
>@@ -40,37 +40,30 @@ void InternalUnloadDll(DllHandle handle);
> 
> bool InternalLoadSymbols(DllHandle handle,
>                          int num_symbols,
>-                         const char *const symbol_names[],
>-                         void *symbols[]);
>+                         const char* const symbol_names[],
>+                         void* symbols[]);
> 
> template <int SYMBOL_TABLE_SIZE,
>           const char kDllName[],
>-          const char *const kSymbolNames[]>
>+          const char* const kSymbolNames[]>
> class LateBindingSymbolTable {
>  public:
>   LateBindingSymbolTable()
>-      : handle_(kInvalidDllHandle),
>-        undefined_symbols_(false) {
>+      : handle_(kInvalidDllHandle), undefined_symbols_(false) {
>     memset(symbols_, 0, sizeof(symbols_));
>   }
> 
>-  ~LateBindingSymbolTable() {
>-    Unload();
>-  }
>+  ~LateBindingSymbolTable() { Unload(); }
> 
>-  static int NumSymbols() {
>-    return SYMBOL_TABLE_SIZE;
>-  }
>+  static int NumSymbols() { return SYMBOL_TABLE_SIZE; }
> 
>   // We do not use this, but we offer it for theoretical convenience.
>-  static const char *GetSymbolName(int index) {
>+  static const char* GetSymbolName(int index) {
>     assert(index < NumSymbols());
>     return kSymbolNames[index];
>   }
> 
>-  bool IsLoaded() const {
>-    return handle_ != kInvalidDllHandle;
>-  }
>+  bool IsLoaded() const { return handle_ != kInvalidDllHandle; }
> 
>   // Loads the DLL and the symbol table. Returns true iff the DLL and symbol
>   // table loaded successfully.
>@@ -106,7 +99,7 @@ class LateBindingSymbolTable {
> 
>   // Retrieves the given symbol. NOTE: Recommended to use LATESYM_GET below
>   // instead of this.
>-  void *GetSymbol(int index) const {
>+  void* GetSymbol(int index) const {
>     assert(IsLoaded());
>     assert(index < NumSymbols());
>     return symbols_[index];
>@@ -115,15 +108,13 @@ class LateBindingSymbolTable {
>  private:
>   DllHandle handle_;
>   bool undefined_symbols_;
>-  void *symbols_[SYMBOL_TABLE_SIZE];
>+  void* symbols_[SYMBOL_TABLE_SIZE];
> 
>   RTC_DISALLOW_COPY_AND_ASSIGN(LateBindingSymbolTable);
> };
> 
> // This macro must be invoked in a header to declare a symbol table class.
>-#define LATE_BINDING_SYMBOL_TABLE_DECLARE_BEGIN(ClassName) \
>-enum {
>-
>+#define LATE_BINDING_SYMBOL_TABLE_DECLARE_BEGIN(ClassName) enum {
> // This macro must be invoked in the header declaration once for each symbol
> // (recommended to use an X-Macro to avoid duplication).
> // This macro defines an enum with names built from the symbols, which
>@@ -150,26 +141,24 @@ enum {
> // This macro must be invoked in a .cc file to define a previously-declared
> // symbol table class.
> #define LATE_BINDING_SYMBOL_TABLE_DEFINE_BEGIN(ClassName, dllName) \
>-const char ClassName##_kDllName[] = dllName; \
>-const char *const ClassName##_kSymbolNames[ClassName##_SYMBOL_TABLE_SIZE] = {
>-
>+  const char ClassName##_kDllName[] = dllName;                     \
>+  const char* const ClassName##_kSymbolNames[ClassName##_SYMBOL_TABLE_SIZE] = {
> // This macro must be invoked in the .cc definition once for each symbol
> // (recommended to use an X-Macro to avoid duplication).
> // This would have to use the mangled name if we were to ever support C++
> // symbols.
>-#define LATE_BINDING_SYMBOL_TABLE_DEFINE_ENTRY(ClassName, sym) \
>-  #sym,
>+#define LATE_BINDING_SYMBOL_TABLE_DEFINE_ENTRY(ClassName, sym) #sym,
> 
> #define LATE_BINDING_SYMBOL_TABLE_DEFINE_END(ClassName) \
>-};
>+  }                                                     \
>+  ;
> 
> // Index of a given symbol in the given symbol table class.
>-#define LATESYM_INDEXOF(ClassName, sym) \
>-  (ClassName##_SYMBOL_TABLE_INDEX_##sym)
>+#define LATESYM_INDEXOF(ClassName, sym) (ClassName##_SYMBOL_TABLE_INDEX_##sym)
> 
> // Returns a reference to the given late-binded symbol, with the correct type.
> #define LATESYM_GET(ClassName, inst, sym) \
>-  (*reinterpret_cast<typeof(&sym)>( \
>+  (*reinterpret_cast<typeof(&sym)>(       \
>       (inst)->GetSymbol(LATESYM_INDEXOF(ClassName, sym))))
> 
> }  // namespace adm_linux
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/pulseaudiosymboltable_linux.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/pulseaudiosymboltable_linux.cc
>index 3d012fe6199..e0759e6ca35 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/pulseaudiosymboltable_linux.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/pulseaudiosymboltable_linux.cc
>@@ -32,7 +32,7 @@ namespace adm_linux_pulse {
> 
> LATE_BINDING_SYMBOL_TABLE_DEFINE_BEGIN(PulseAudioSymbolTable, "libpulse.so.0")
> #define X(sym) \
>-    LATE_BINDING_SYMBOL_TABLE_DEFINE_ENTRY(PulseAudioSymbolTable, sym)
>+  LATE_BINDING_SYMBOL_TABLE_DEFINE_ENTRY(PulseAudioSymbolTable, sym)
> PULSE_AUDIO_SYMBOLS_LIST
> #undef X
> LATE_BINDING_SYMBOL_TABLE_DEFINE_END(PulseAudioSymbolTable)
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/pulseaudiosymboltable_linux.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/pulseaudiosymboltable_linux.h
>index d21446ce8af..2f6a9510d83 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/pulseaudiosymboltable_linux.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/linux/pulseaudiosymboltable_linux.h
>@@ -36,66 +36,66 @@ namespace adm_linux_pulse {
> // The PulseAudio symbols we need, as an X-Macro list.
> // This list must contain precisely every libpulse function that is used in
> // the ADM LINUX PULSE Device and Mixer classes
>-#define PULSE_AUDIO_SYMBOLS_LIST \
>-  X(pa_bytes_per_second) \
>-  X(pa_context_connect) \
>-  X(pa_context_disconnect) \
>-  X(pa_context_errno) \
>-  X(pa_context_get_protocol_version) \
>-  X(pa_context_get_server_info) \
>-  X(pa_context_get_sink_info_list) \
>-  X(pa_context_get_sink_info_by_index) \
>-  X(pa_context_get_sink_info_by_name) \
>-  X(pa_context_get_sink_input_info) \
>-  X(pa_context_get_source_info_by_index) \
>-  X(pa_context_get_source_info_by_name) \
>-  X(pa_context_get_source_info_list) \
>-  X(pa_context_get_state) \
>-  X(pa_context_new) \
>-  X(pa_context_set_sink_input_volume) \
>-  X(pa_context_set_sink_input_mute) \
>+#define PULSE_AUDIO_SYMBOLS_LIST           \
>+  X(pa_bytes_per_second)                   \
>+  X(pa_context_connect)                    \
>+  X(pa_context_disconnect)                 \
>+  X(pa_context_errno)                      \
>+  X(pa_context_get_protocol_version)       \
>+  X(pa_context_get_server_info)            \
>+  X(pa_context_get_sink_info_list)         \
>+  X(pa_context_get_sink_info_by_index)     \
>+  X(pa_context_get_sink_info_by_name)      \
>+  X(pa_context_get_sink_input_info)        \
>+  X(pa_context_get_source_info_by_index)   \
>+  X(pa_context_get_source_info_by_name)    \
>+  X(pa_context_get_source_info_list)       \
>+  X(pa_context_get_state)                  \
>+  X(pa_context_new)                        \
>+  X(pa_context_set_sink_input_volume)      \
>+  X(pa_context_set_sink_input_mute)        \
>   X(pa_context_set_source_volume_by_index) \
>-  X(pa_context_set_source_mute_by_index) \
>-  X(pa_context_set_state_callback) \
>-  X(pa_context_unref) \
>-  X(pa_cvolume_set) \
>-  X(pa_operation_get_state) \
>-  X(pa_operation_unref) \
>-  X(pa_stream_connect_playback) \
>-  X(pa_stream_connect_record) \
>-  X(pa_stream_disconnect) \
>-  X(pa_stream_drop) \
>-  X(pa_stream_get_device_index) \
>-  X(pa_stream_get_index) \
>-  X(pa_stream_get_latency) \
>-  X(pa_stream_get_sample_spec) \
>-  X(pa_stream_get_state) \
>-  X(pa_stream_new) \
>-  X(pa_stream_peek) \
>-  X(pa_stream_readable_size) \
>-  X(pa_stream_set_buffer_attr) \
>-  X(pa_stream_set_overflow_callback) \
>-  X(pa_stream_set_read_callback) \
>-  X(pa_stream_set_state_callback) \
>-  X(pa_stream_set_underflow_callback) \
>-  X(pa_stream_set_write_callback) \
>-  X(pa_stream_unref) \
>-  X(pa_stream_writable_size) \
>-  X(pa_stream_write) \
>-  X(pa_strerror) \
>-  X(pa_threaded_mainloop_free) \
>-  X(pa_threaded_mainloop_get_api) \
>-  X(pa_threaded_mainloop_lock) \
>-  X(pa_threaded_mainloop_new) \
>-  X(pa_threaded_mainloop_signal) \
>-  X(pa_threaded_mainloop_start) \
>-  X(pa_threaded_mainloop_stop) \
>-  X(pa_threaded_mainloop_unlock) \
>+  X(pa_context_set_source_mute_by_index)   \
>+  X(pa_context_set_state_callback)         \
>+  X(pa_context_unref)                      \
>+  X(pa_cvolume_set)                        \
>+  X(pa_operation_get_state)                \
>+  X(pa_operation_unref)                    \
>+  X(pa_stream_connect_playback)            \
>+  X(pa_stream_connect_record)              \
>+  X(pa_stream_disconnect)                  \
>+  X(pa_stream_drop)                        \
>+  X(pa_stream_get_device_index)            \
>+  X(pa_stream_get_index)                   \
>+  X(pa_stream_get_latency)                 \
>+  X(pa_stream_get_sample_spec)             \
>+  X(pa_stream_get_state)                   \
>+  X(pa_stream_new)                         \
>+  X(pa_stream_peek)                        \
>+  X(pa_stream_readable_size)               \
>+  X(pa_stream_set_buffer_attr)             \
>+  X(pa_stream_set_overflow_callback)       \
>+  X(pa_stream_set_read_callback)           \
>+  X(pa_stream_set_state_callback)          \
>+  X(pa_stream_set_underflow_callback)      \
>+  X(pa_stream_set_write_callback)          \
>+  X(pa_stream_unref)                       \
>+  X(pa_stream_writable_size)               \
>+  X(pa_stream_write)                       \
>+  X(pa_strerror)                           \
>+  X(pa_threaded_mainloop_free)             \
>+  X(pa_threaded_mainloop_get_api)          \
>+  X(pa_threaded_mainloop_lock)             \
>+  X(pa_threaded_mainloop_new)              \
>+  X(pa_threaded_mainloop_signal)           \
>+  X(pa_threaded_mainloop_start)            \
>+  X(pa_threaded_mainloop_stop)             \
>+  X(pa_threaded_mainloop_unlock)           \
>   X(pa_threaded_mainloop_wait)
> 
> LATE_BINDING_SYMBOL_TABLE_DECLARE_BEGIN(PulseAudioSymbolTable)
> #define X(sym) \
>-    LATE_BINDING_SYMBOL_TABLE_DECLARE_ENTRY(PulseAudioSymbolTable, sym)
>+  LATE_BINDING_SYMBOL_TABLE_DECLARE_ENTRY(PulseAudioSymbolTable, sym)
> PULSE_AUDIO_SYMBOLS_LIST
> #undef X
> LATE_BINDING_SYMBOL_TABLE_DECLARE_END(PulseAudioSymbolTable)
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/mac/audio_device_mac.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/mac/audio_device_mac.cc
>index 89ea9317c1f..063ac85736d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/mac/audio_device_mac.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/mac/audio_device_mac.cc
>@@ -9,17 +9,18 @@
>  */
> 
> #include "modules/audio_device/mac/audio_device_mac.h"
>+#include "absl/memory/memory.h"
> #include "modules/audio_device/audio_device_config.h"
>-#include "modules/audio_device/mac/portaudio/pa_ringbuffer.h"
>+#include "modules/third_party/portaudio/pa_ringbuffer.h"
> #include "rtc_base/arraysize.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/platform_thread.h"
>+#include "rtc_base/system/arch.h"
> #include "system_wrappers/include/event_wrapper.h"
> 
> #include <ApplicationServices/ApplicationServices.h>
> #include <libkern/OSAtomic.h>  // OSAtomicCompareAndSwap()
> #include <mach/mach.h>         // mach_task_self()
>-#include <memory>              // std::make_unique<>() and std::unique_ptr<>()
> #include <sys/sysctl.h>        // sysctlbyname()
> 
> namespace webrtc {
>@@ -1566,7 +1567,7 @@ int32_t AudioDeviceMac::GetNumberDevices(const AudioObjectPropertyScope scope,
>   }
> 
>   UInt32 numberDevices = size / sizeof(AudioDeviceID);
>-  auto deviceIds = std::make_unique<AudioDeviceID[]>(numberDevices);
>+  const auto deviceIds = absl::make_unique<AudioDeviceID[]>(numberDevices);
>   AudioBufferList* bufferList = NULL;
>   UInt32 numberScopedDevices = 0;
> 
>@@ -1597,8 +1598,9 @@ int32_t AudioDeviceMac::GetNumberDevices(const AudioObjectPropertyScope scope,
>   // Then list the rest of the devices
>   bool listOK = true;
> 
>-  WEBRTC_CA_LOG_ERR(AudioObjectGetPropertyData(
>-      kAudioObjectSystemObject, &propertyAddress, 0, NULL, &size, deviceIds.get()));
>+  WEBRTC_CA_LOG_ERR(AudioObjectGetPropertyData(kAudioObjectSystemObject,
>+                                               &propertyAddress, 0, NULL, &size,
>+                                               deviceIds.get()));
>   if (err != noErr) {
>     listOK = false;
>   } else {
>@@ -1646,7 +1648,6 @@ int32_t AudioDeviceMac::GetNumberDevices(const AudioObjectPropertyScope scope,
>       free(bufferList);
>       bufferList = NULL;
>     }
>-
>     return -1;
>   }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/mac/audio_mixer_manager_mac.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/mac/audio_mixer_manager_mac.cc
>index 0932e8b01b8..faedfb0ce33 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/mac/audio_mixer_manager_mac.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/mac/audio_mixer_manager_mac.cc
>@@ -9,6 +9,7 @@
>  */
> 
> #include "modules/audio_device/mac/audio_mixer_manager_mac.h"
>+#include "rtc_base/system/arch.h"
> 
> #include <unistd.h>  // getpid()
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/mac/audio_mixer_manager_mac.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/mac/audio_mixer_manager_mac.h
>index 6681da5913a..bb67f2058ca 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/mac/audio_mixer_manager_mac.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/mac/audio_mixer_manager_mac.h
>@@ -14,7 +14,6 @@
> #include "modules/audio_device/include/audio_device.h"
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/logging.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> #include <CoreAudio/CoreAudio.h>
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/mac/portaudio/pa_memorybarrier.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/mac/portaudio/pa_memorybarrier.h
>deleted file mode 100644
>index f6896222010..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/mac/portaudio/pa_memorybarrier.h
>+++ /dev/null
>@@ -1,127 +0,0 @@
>-/*
>- * $Id: pa_memorybarrier.h 1240 2007-07-17 13:05:07Z bjornroche $
>- * Portable Audio I/O Library
>- * Memory barrier utilities
>- *
>- * Author: Bjorn Roche, XO Audio, LLC
>- *
>- * This program uses the PortAudio Portable Audio Library.
>- * For more information see: http://www.portaudio.com
>- * Copyright (c) 1999-2000 Ross Bencina and Phil Burk
>- *
>- * Permission is hereby granted, free of charge, to any person obtaining
>- * a copy of this software and associated documentation files
>- * (the "Software"), to deal in the Software without restriction,
>- * including without limitation the rights to use, copy, modify, merge,
>- * publish, distribute, sublicense, and/or sell copies of the Software,
>- * and to permit persons to whom the Software is furnished to do so,
>- * subject to the following conditions:
>- *
>- * The above copyright notice and this permission notice shall be
>- * included in all copies or substantial portions of the Software.
>- *
>- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
>- * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
>- * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
>- * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
>- * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
>- * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
>- * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
>- */
>-
>-/*
>- * The text above constitutes the entire PortAudio license; however, 
>- * the PortAudio community also makes the following non-binding requests:
>- *
>- * Any person wishing to distribute modifications to the Software is
>- * requested to send the modifications to the original developer so that
>- * they can be incorporated into the canonical version. It is also 
>- * requested that these non-binding requests be included along with the 
>- * license above.
>- */
>-
>-/**
>- @file pa_memorybarrier.h
>- @ingroup common_src
>-*/
>-
>-/****************
>- * Some memory barrier primitives based on the system.
>- * right now only OS X, FreeBSD, and Linux are supported. In addition to providing
>- * memory barriers, these functions should ensure that data cached in registers
>- * is written out to cache where it can be snooped by other CPUs. (ie, the volatile
>- * keyword should not be required)
>- *
>- * the primitives that must be defined are:
>- *
>- * PaUtil_FullMemoryBarrier()
>- * PaUtil_ReadMemoryBarrier()
>- * PaUtil_WriteMemoryBarrier()
>- *
>- ****************/
>-
>-#if defined(__APPLE__)
>-#   include <libkern/OSAtomic.h>
>-    /* Here are the memory barrier functions. Mac OS X only provides
>-       full memory barriers, so the three types of barriers are the same,
>-       however, these barriers are superior to compiler-based ones. */
>-#   define PaUtil_FullMemoryBarrier()  OSMemoryBarrier()
>-#   define PaUtil_ReadMemoryBarrier()  OSMemoryBarrier()
>-#   define PaUtil_WriteMemoryBarrier() OSMemoryBarrier()
>-#elif defined(__GNUC__)
>-    /* GCC >= 4.1 has built-in intrinsics. We'll use those */
>-#   if (__GNUC__ > 4) || (__GNUC__ == 4 && __GNUC_MINOR__ >= 1)
>-#      define PaUtil_FullMemoryBarrier()  __sync_synchronize()
>-#      define PaUtil_ReadMemoryBarrier()  __sync_synchronize()
>-#      define PaUtil_WriteMemoryBarrier() __sync_synchronize()
>-    /* as a fallback, GCC understands volatile asm and "memory" to mean it
>-     * should not reorder memory read/writes */
>-    /* Note that it is not clear that any compiler actually defines __PPC__,
>-     * it can probably removed safely. */
>-#   elif defined( __ppc__ ) || defined( __powerpc__) || defined( __PPC__ )
>-#      define PaUtil_FullMemoryBarrier()  asm volatile("sync":::"memory")
>-#      define PaUtil_ReadMemoryBarrier()  asm volatile("sync":::"memory")
>-#      define PaUtil_WriteMemoryBarrier() asm volatile("sync":::"memory")
>-#   elif defined( __i386__ ) || defined( __i486__ ) || defined( __i586__ ) || \
>-         defined( __i686__ ) || defined( __x86_64__ )
>-#      define PaUtil_FullMemoryBarrier()  asm volatile("mfence":::"memory")
>-#      define PaUtil_ReadMemoryBarrier()  asm volatile("lfence":::"memory")
>-#      define PaUtil_WriteMemoryBarrier() asm volatile("sfence":::"memory")
>-#   else
>-#      ifdef ALLOW_SMP_DANGERS
>-#         warning Memory barriers not defined on this system or system unknown
>-#         warning For SMP safety, you should fix this.
>-#         define PaUtil_FullMemoryBarrier()
>-#         define PaUtil_ReadMemoryBarrier()
>-#         define PaUtil_WriteMemoryBarrier()
>-#      else
>-#         error Memory barriers are not defined on this system. You can still compile by defining ALLOW_SMP_DANGERS, but SMP safety will not be guaranteed.
>-#      endif
>-#   endif
>-#elif (_MSC_VER >= 1400) && !defined(_WIN32_WCE)
>-#   include <intrin.h>
>-#   pragma intrinsic(_ReadWriteBarrier)
>-#   pragma intrinsic(_ReadBarrier)
>-#   pragma intrinsic(_WriteBarrier)
>-#   define PaUtil_FullMemoryBarrier()  _ReadWriteBarrier()
>-#   define PaUtil_ReadMemoryBarrier()  _ReadBarrier()
>-#   define PaUtil_WriteMemoryBarrier() _WriteBarrier()
>-#elif defined(_WIN32_WCE)
>-#   define PaUtil_FullMemoryBarrier()
>-#   define PaUtil_ReadMemoryBarrier()
>-#   define PaUtil_WriteMemoryBarrier()
>-#elif defined(_MSC_VER) || defined(__BORLANDC__)
>-#   define PaUtil_FullMemoryBarrier()  _asm { lock add    [esp], 0 }
>-#   define PaUtil_ReadMemoryBarrier()  _asm { lock add    [esp], 0 }
>-#   define PaUtil_WriteMemoryBarrier() _asm { lock add    [esp], 0 }
>-#else
>-#   ifdef ALLOW_SMP_DANGERS
>-#      warning Memory barriers not defined on this system or system unknown
>-#      warning For SMP safety, you should fix this.
>-#      define PaUtil_FullMemoryBarrier()
>-#      define PaUtil_ReadMemoryBarrier()
>-#      define PaUtil_WriteMemoryBarrier()
>-#   else
>-#      error Memory barriers are not defined on this system. You can still compile by defining ALLOW_SMP_DANGERS, but SMP safety will not be guaranteed.
>-#   endif
>-#endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/win/audio_device_core_win.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/win/audio_device_core_win.cc
>index 834fa4ed259..fef8e50ffda 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/win/audio_device_core_win.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/win/audio_device_core_win.cc
>@@ -28,9 +28,9 @@
> #include <assert.h>
> #include <string.h>
> 
>-#include <Functiondiscoverykeys_devpkey.h>
> #include <comdef.h>
> #include <dmo.h>
>+#include <functiondiscoverykeys_devpkey.h>
> #include <mmsystem.h>
> #include <strsafe.h>
> #include <uuids.h>
>@@ -40,6 +40,8 @@
> 
> #include "rtc_base/logging.h"
> #include "rtc_base/platform_thread.h"
>+#include "rtc_base/stringutils.h"
>+#include "rtc_base/thread_annotations.h"
> #include "system_wrappers/include/sleep.h"
> 
> // Macro that calls a COM method returning HRESULT value.
>@@ -379,7 +381,9 @@ bool AudioDeviceWindowsCore::CoreAudioIsSupported() {
> // ----------------------------------------------------------------------------
> 
> AudioDeviceWindowsCore::AudioDeviceWindowsCore()
>-    : _comInit(ScopedCOMInitializer::kMTA),
>+    : _avrtLibrary(NULL),
>+      _winSupportAvrt(false),
>+      _comInit(ScopedCOMInitializer::kMTA),
>       _ptrAudioBuffer(NULL),
>       _ptrEnumerator(NULL),
>       _ptrRenderCollection(NULL),
>@@ -395,6 +399,15 @@ AudioDeviceWindowsCore::AudioDeviceWindowsCore()
>       _dmo(NULL),
>       _mediaBuffer(NULL),
>       _builtInAecEnabled(false),
>+      _hRenderSamplesReadyEvent(NULL),
>+      _hPlayThread(NULL),
>+      _hRenderStartedEvent(NULL),
>+      _hShutdownRenderEvent(NULL),
>+      _hCaptureSamplesReadyEvent(NULL),
>+      _hRecThread(NULL),
>+      _hCaptureStartedEvent(NULL),
>+      _hShutdownCaptureEvent(NULL),
>+      _hMmTask(NULL),
>       _playAudioFrameSize(0),
>       _playSampleRate(0),
>       _playBlockSize(0),
>@@ -407,17 +420,6 @@ AudioDeviceWindowsCore::AudioDeviceWindowsCore()
>       _recSampleRate(0),
>       _recBlockSize(0),
>       _recChannels(2),
>-      _avrtLibrary(NULL),
>-      _winSupportAvrt(false),
>-      _hRenderSamplesReadyEvent(NULL),
>-      _hPlayThread(NULL),
>-      _hCaptureSamplesReadyEvent(NULL),
>-      _hRecThread(NULL),
>-      _hShutdownRenderEvent(NULL),
>-      _hShutdownCaptureEvent(NULL),
>-      _hRenderStartedEvent(NULL),
>-      _hCaptureStartedEvent(NULL),
>-      _hMmTask(NULL),
>       _initialized(false),
>       _recording(false),
>       _playing(false),
>@@ -425,7 +427,6 @@ AudioDeviceWindowsCore::AudioDeviceWindowsCore()
>       _playIsInitialized(false),
>       _speakerIsInitialized(false),
>       _microphoneIsInitialized(false),
>-      _playBufDelay(80),
>       _usingInputDeviceIndex(false),
>       _usingOutputDeviceIndex(false),
>       _inputDevice(AudioDeviceModule::kDefaultCommunicationDevice),
>@@ -1872,8 +1873,8 @@ int32_t AudioDeviceWindowsCore::InitPlayout() {
>   if (SUCCEEDED(hr)) {
>     RTC_LOG(LS_VERBOSE) << "Audio Engine's current rendering mix format:";
>     // format type
>-    RTC_LOG(LS_VERBOSE) << "wFormatTag     : 0x" << std::hex
>-                        << pWfxOut->wFormatTag << std::dec << " ("
>+    RTC_LOG(LS_VERBOSE) << "wFormatTag     : 0x"
>+                        << rtc::ToHex(pWfxOut->wFormatTag) << " ("
>                         << pWfxOut->wFormatTag << ")";
>     // number of channels (i.e. mono, stereo...)
>     RTC_LOG(LS_VERBOSE) << "nChannels      : " << pWfxOut->nChannels;
>@@ -1947,8 +1948,8 @@ int32_t AudioDeviceWindowsCore::InitPlayout() {
>     _playChannels = Wfx.nChannels;
> 
>     RTC_LOG(LS_VERBOSE) << "VoE selected this rendering format:";
>-    RTC_LOG(LS_VERBOSE) << "wFormatTag         : 0x" << std::hex
>-                        << Wfx.wFormatTag << std::dec << " (" << Wfx.wFormatTag
>+    RTC_LOG(LS_VERBOSE) << "wFormatTag         : 0x"
>+                        << rtc::ToHex(Wfx.wFormatTag) << " (" << Wfx.wFormatTag
>                         << ")";
>     RTC_LOG(LS_VERBOSE) << "nChannels          : " << Wfx.nChannels;
>     RTC_LOG(LS_VERBOSE) << "nSamplesPerSec     : " << Wfx.nSamplesPerSec;
>@@ -1958,8 +1959,7 @@ int32_t AudioDeviceWindowsCore::InitPlayout() {
>     RTC_LOG(LS_VERBOSE) << "cbSize             : " << Wfx.cbSize;
>     RTC_LOG(LS_VERBOSE) << "Additional settings:";
>     RTC_LOG(LS_VERBOSE) << "_playAudioFrameSize: " << _playAudioFrameSize;
>-    RTC_LOG(LS_VERBOSE) << "_playBlockSize     : "
>-                        << _playBlockSize;
>+    RTC_LOG(LS_VERBOSE) << "_playBlockSize     : " << _playBlockSize;
>     RTC_LOG(LS_VERBOSE) << "_playChannels      : " << _playChannels;
>   }
> 
>@@ -2070,7 +2070,7 @@ int32_t AudioDeviceWindowsCore::InitRecordingDMO() {
>     return -1;
>   }
> 
>-  DMO_MEDIA_TYPE mt = {0};
>+  DMO_MEDIA_TYPE mt = {};
>   HRESULT hr = MoInitMediaType(&mt, sizeof(WAVEFORMATEX));
>   if (FAILED(hr)) {
>     MoFreeMediaType(&mt);
>@@ -2189,8 +2189,8 @@ int32_t AudioDeviceWindowsCore::InitRecording() {
>   if (SUCCEEDED(hr)) {
>     RTC_LOG(LS_VERBOSE) << "Audio Engine's current capturing mix format:";
>     // format type
>-    RTC_LOG(LS_VERBOSE) << "wFormatTag     : 0x" << std::hex
>-                        << pWfxIn->wFormatTag << std::dec << " ("
>+    RTC_LOG(LS_VERBOSE) << "wFormatTag     : 0x"
>+                        << rtc::ToHex(pWfxIn->wFormatTag) << " ("
>                         << pWfxIn->wFormatTag << ")";
>     // number of channels (i.e. mono, stereo...)
>     RTC_LOG(LS_VERBOSE) << "nChannels      : " << pWfxIn->nChannels;
>@@ -2262,8 +2262,8 @@ int32_t AudioDeviceWindowsCore::InitRecording() {
>     _recChannels = Wfx.Format.nChannels;
> 
>     RTC_LOG(LS_VERBOSE) << "VoE selected this capturing format:";
>-    RTC_LOG(LS_VERBOSE) << "wFormatTag        : 0x" << std::hex
>-                        << Wfx.Format.wFormatTag << std::dec << " ("
>+    RTC_LOG(LS_VERBOSE) << "wFormatTag        : 0x"
>+                        << rtc::ToHex(Wfx.Format.wFormatTag) << " ("
>                         << Wfx.Format.wFormatTag << ")";
>     RTC_LOG(LS_VERBOSE) << "nChannels         : " << Wfx.Format.nChannels;
>     RTC_LOG(LS_VERBOSE) << "nSamplesPerSec    : " << Wfx.Format.nSamplesPerSec;
>@@ -2634,6 +2634,10 @@ int32_t AudioDeviceWindowsCore::PlayoutDelay(uint16_t& delayMS) const {
>   return 0;
> }
> 
>+bool AudioDeviceWindowsCore::BuiltInAECIsAvailable() const {
>+  return _dmo != nullptr;
>+}
>+
> // ----------------------------------------------------------------------------
> //  Playing
> // ----------------------------------------------------------------------------
>@@ -3323,8 +3327,8 @@ DWORD AudioDeviceWindowsCore::DoCaptureThread() {
>         // IAudioClient::Stop, IAudioClient::Reset, and releasing the audio
>         // client.
>         RTC_LOG(LS_ERROR) << "IAudioCaptureClient::GetBuffer returned"
>-                          << " AUDCLNT_E_BUFFER_ERROR, hr = 0x" << std::hex
>-                          << hr << std::dec;
>+                          << " AUDCLNT_E_BUFFER_ERROR, hr = 0x"
>+                          << rtc::ToHex(hr);
>         goto Exit;
>       }
> 
>@@ -3396,6 +3400,14 @@ int32_t AudioDeviceWindowsCore::EnableBuiltInAEC(bool enable) {
>   return 0;
> }
> 
>+void AudioDeviceWindowsCore::_Lock() RTC_NO_THREAD_SAFETY_ANALYSIS {
>+  _critSect.Enter();
>+}
>+
>+void AudioDeviceWindowsCore::_UnLock() RTC_NO_THREAD_SAFETY_ANALYSIS {
>+  _critSect.Leave();
>+}
>+
> int AudioDeviceWindowsCore::SetDMOProperties() {
>   HRESULT hr = S_OK;
>   assert(_dmo != NULL);
>@@ -3829,7 +3841,7 @@ int32_t AudioDeviceWindowsCore::_GetDeviceName(IMMDevice* pDevice,
>     hr = pDevice->OpenPropertyStore(STGM_READ, &pProps);
>     if (FAILED(hr)) {
>       RTC_LOG(LS_ERROR) << "IMMDevice::OpenPropertyStore failed, hr = 0x"
>-                        << std::hex << hr << std::dec;
>+                        << rtc::ToHex(hr);
>     }
>   }
> 
>@@ -3841,21 +3853,21 @@ int32_t AudioDeviceWindowsCore::_GetDeviceName(IMMDevice* pDevice,
>     hr = pProps->GetValue(PKEY_Device_FriendlyName, &varName);
>     if (FAILED(hr)) {
>       RTC_LOG(LS_ERROR) << "IPropertyStore::GetValue failed, hr = 0x"
>-                        << std::hex << hr << std::dec;
>+                        << rtc::ToHex(hr);
>     }
>   }
> 
>   if ((SUCCEEDED(hr)) && (VT_EMPTY == varName.vt)) {
>     hr = E_FAIL;
>     RTC_LOG(LS_ERROR) << "IPropertyStore::GetValue returned no value,"
>-                      << " hr = 0x" << std::hex << hr << std::dec;
>+                      << " hr = 0x" << rtc::ToHex(hr);
>   }
> 
>   if ((SUCCEEDED(hr)) && (VT_LPWSTR != varName.vt)) {
>     // The returned value is not a wide null terminated string.
>     hr = E_UNEXPECTED;
>     RTC_LOG(LS_ERROR) << "IPropertyStore::GetValue returned unexpected"
>-                      << " type, hr = 0x" << std::hex << hr << std::dec;
>+                      << " type, hr = 0x" << rtc::ToHex(hr);
>   }
> 
>   if (SUCCEEDED(hr) && (varName.pwszVal != NULL)) {
>@@ -4044,16 +4056,16 @@ int32_t AudioDeviceWindowsCore::_EnumerateEndpointDevicesAll(
>     hr = pEndpoint->GetState(&dwState);
>     CONTINUE_ON_ERROR(hr);
>     if (dwState & DEVICE_STATE_ACTIVE)
>-      RTC_LOG(LS_VERBOSE) << "state (0x" << std::hex << dwState << std::dec
>+      RTC_LOG(LS_VERBOSE) << "state (0x" << rtc::ToHex(dwState)
>                           << ")  : *ACTIVE*";
>     if (dwState & DEVICE_STATE_DISABLED)
>-      RTC_LOG(LS_VERBOSE) << "state (0x" << std::hex << dwState << std::dec
>+      RTC_LOG(LS_VERBOSE) << "state (0x" << rtc::ToHex(dwState)
>                           << ")  : DISABLED";
>     if (dwState & DEVICE_STATE_NOTPRESENT)
>-      RTC_LOG(LS_VERBOSE) << "state (0x" << std::hex << dwState << std::dec
>+      RTC_LOG(LS_VERBOSE) << "state (0x" << rtc::ToHex(dwState)
>                           << ")  : NOTPRESENT";
>     if (dwState & DEVICE_STATE_UNPLUGGED)
>-      RTC_LOG(LS_VERBOSE) << "state (0x" << std::hex << dwState << std::dec
>+      RTC_LOG(LS_VERBOSE) << "state (0x" << rtc::ToHex(dwState)
>                           << ")  : UNPLUGGED";
> 
>     // Check the hardware volume capabilities.
>@@ -4065,16 +4077,16 @@ int32_t AudioDeviceWindowsCore::_EnumerateEndpointDevicesAll(
>     CONTINUE_ON_ERROR(hr);
>     if (dwHwSupportMask & ENDPOINT_HARDWARE_SUPPORT_VOLUME)
>       // The audio endpoint device supports a hardware volume control
>-      RTC_LOG(LS_VERBOSE) << "hwmask (0x" << std::hex << dwHwSupportMask
>-                          << std::dec << ") : HARDWARE_SUPPORT_VOLUME";
>+      RTC_LOG(LS_VERBOSE) << "hwmask (0x" << rtc::ToHex(dwHwSupportMask)
>+                          << ") : HARDWARE_SUPPORT_VOLUME";
>     if (dwHwSupportMask & ENDPOINT_HARDWARE_SUPPORT_MUTE)
>       // The audio endpoint device supports a hardware mute control
>-      RTC_LOG(LS_VERBOSE) << "hwmask (0x" << std::hex << dwHwSupportMask
>-                          << std::dec << ") : HARDWARE_SUPPORT_MUTE";
>+      RTC_LOG(LS_VERBOSE) << "hwmask (0x" << rtc::ToHex(dwHwSupportMask)
>+                          << ") : HARDWARE_SUPPORT_MUTE";
>     if (dwHwSupportMask & ENDPOINT_HARDWARE_SUPPORT_METER)
>       // The audio endpoint device supports a hardware peak meter
>-      RTC_LOG(LS_VERBOSE) << "hwmask (0x" << std::hex << dwHwSupportMask
>-                          << std::dec << ") : HARDWARE_SUPPORT_METER";
>+      RTC_LOG(LS_VERBOSE) << "hwmask (0x" << rtc::ToHex(dwHwSupportMask)
>+                          << ") : HARDWARE_SUPPORT_METER";
> 
>     // Check the channel count (#channels in the audio stream that enters or
>     // leaves the audio endpoint device)
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/win/audio_device_core_win.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/win/audio_device_core_win.h
>index b5c2b72d301..7d06c8c8906 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/win/audio_device_core_win.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/win/audio_device_core_win.h
>@@ -13,26 +13,28 @@
> 
> #if (_MSC_VER >= 1400)  // only include for VS 2005 and higher
> 
>+#include "rtc_base/win32.h"
>+
> #include "modules/audio_device/audio_device_generic.h"
> 
>-#include <wmcodecdsp.h>      // CLSID_CWMAudioAEC
>-                             // (must be before audioclient.h)
>-#include <Audioclient.h>     // WASAPI
>-#include <Audiopolicy.h>
>-#include <Mmdeviceapi.h>     // MMDevice
>-#include <avrt.h>            // Avrt
>+#include <wmcodecdsp.h>   // CLSID_CWMAudioAEC
>+                          // (must be before audioclient.h)
>+#include <audioclient.h>  // WASAPI
>+#include <audiopolicy.h>
>+#include <avrt.h>  // Avrt
> #include <endpointvolume.h>
>-#include <mediaobj.h>        // IMediaObject
>+#include <mediaobj.h>     // IMediaObject
>+#include <mmdeviceapi.h>  // MMDevice
> 
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/scoped_ref_ptr.h"
> 
> // Use Multimedia Class Scheduler Service (MMCSS) to boost the thread priority
>-#pragma comment( lib, "avrt.lib" )
>+#pragma comment(lib, "avrt.lib")
> // AVRT function pointers
>-typedef BOOL (WINAPI *PAvRevertMmThreadCharacteristics)(HANDLE);
>-typedef HANDLE (WINAPI *PAvSetMmThreadCharacteristicsA)(LPCSTR, LPDWORD);
>-typedef BOOL (WINAPI *PAvSetMmThreadPriority)(HANDLE, AVRT_PRIORITY);
>+typedef BOOL(WINAPI* PAvRevertMmThreadCharacteristics)(HANDLE);
>+typedef HANDLE(WINAPI* PAvSetMmThreadCharacteristicsA)(LPCSTR, LPDWORD);
>+typedef BOOL(WINAPI* PAvSetMmThreadPriority)(HANDLE, AVRT_PRIORITY);
> 
> namespace webrtc {
> 
>@@ -51,16 +53,14 @@ class ScopedCOMInitializer {
>   enum SelectMTA { kMTA };
> 
>   // Constructor for STA initialization.
>-  ScopedCOMInitializer() {
>-    Initialize(COINIT_APARTMENTTHREADED);
>-  }
>+  ScopedCOMInitializer() { Initialize(COINIT_APARTMENTTHREADED); }
> 
>   // Constructor for MTA initialization.
>   explicit ScopedCOMInitializer(SelectMTA mta) {
>     Initialize(COINIT_MULTITHREADED);
>   }
> 
>-  ScopedCOMInitializer::~ScopedCOMInitializer() {
>+  ~ScopedCOMInitializer() {
>     if (SUCCEEDED(hr_))
>       CoUninitialize();
>   }
>@@ -68,9 +68,7 @@ class ScopedCOMInitializer {
>   bool succeeded() const { return SUCCEEDED(hr_); }
> 
>  private:
>-  void Initialize(COINIT init) {
>-    hr_ = CoInitializeEx(NULL, init);
>-  }
>+  void Initialize(COINIT init) { hr_ = CoInitializeEx(NULL, init); }
> 
>   HRESULT hr_;
> 
>@@ -78,238 +76,246 @@ class ScopedCOMInitializer {
>   void operator=(const ScopedCOMInitializer&);
> };
> 
>+class AudioDeviceWindowsCore : public AudioDeviceGeneric {
>+ public:
>+  AudioDeviceWindowsCore();
>+  ~AudioDeviceWindowsCore();
>+
>+  static bool CoreAudioIsSupported();
>+
>+  // Retrieve the currently utilized audio layer
>+  virtual int32_t ActiveAudioLayer(
>+      AudioDeviceModule::AudioLayer& audioLayer) const;
>+
>+  // Main initializaton and termination
>+  virtual InitStatus Init();
>+  virtual int32_t Terminate();
>+  virtual bool Initialized() const;
>+
>+  // Device enumeration
>+  virtual int16_t PlayoutDevices();
>+  virtual int16_t RecordingDevices();
>+  virtual int32_t PlayoutDeviceName(uint16_t index,
>+                                    char name[kAdmMaxDeviceNameSize],
>+                                    char guid[kAdmMaxGuidSize]);
>+  virtual int32_t RecordingDeviceName(uint16_t index,
>+                                      char name[kAdmMaxDeviceNameSize],
>+                                      char guid[kAdmMaxGuidSize]);
>+
>+  // Device selection
>+  virtual int32_t SetPlayoutDevice(uint16_t index);
>+  virtual int32_t SetPlayoutDevice(AudioDeviceModule::WindowsDeviceType device);
>+  virtual int32_t SetRecordingDevice(uint16_t index);
>+  virtual int32_t SetRecordingDevice(
>+      AudioDeviceModule::WindowsDeviceType device);
>+
>+  // Audio transport initialization
>+  virtual int32_t PlayoutIsAvailable(bool& available);
>+  virtual int32_t InitPlayout();
>+  virtual bool PlayoutIsInitialized() const;
>+  virtual int32_t RecordingIsAvailable(bool& available);
>+  virtual int32_t InitRecording();
>+  virtual bool RecordingIsInitialized() const;
>+
>+  // Audio transport control
>+  virtual int32_t StartPlayout();
>+  virtual int32_t StopPlayout();
>+  virtual bool Playing() const;
>+  virtual int32_t StartRecording();
>+  virtual int32_t StopRecording();
>+  virtual bool Recording() const;
>+
>+  // Audio mixer initialization
>+  virtual int32_t InitSpeaker();
>+  virtual bool SpeakerIsInitialized() const;
>+  virtual int32_t InitMicrophone();
>+  virtual bool MicrophoneIsInitialized() const;
>+
>+  // Speaker volume controls
>+  virtual int32_t SpeakerVolumeIsAvailable(bool& available);
>+  virtual int32_t SetSpeakerVolume(uint32_t volume);
>+  virtual int32_t SpeakerVolume(uint32_t& volume) const;
>+  virtual int32_t MaxSpeakerVolume(uint32_t& maxVolume) const;
>+  virtual int32_t MinSpeakerVolume(uint32_t& minVolume) const;
>+
>+  // Microphone volume controls
>+  virtual int32_t MicrophoneVolumeIsAvailable(bool& available);
>+  virtual int32_t SetMicrophoneVolume(uint32_t volume);
>+  virtual int32_t MicrophoneVolume(uint32_t& volume) const;
>+  virtual int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const;
>+  virtual int32_t MinMicrophoneVolume(uint32_t& minVolume) const;
>+
>+  // Speaker mute control
>+  virtual int32_t SpeakerMuteIsAvailable(bool& available);
>+  virtual int32_t SetSpeakerMute(bool enable);
>+  virtual int32_t SpeakerMute(bool& enabled) const;
>+
>+  // Microphone mute control
>+  virtual int32_t MicrophoneMuteIsAvailable(bool& available);
>+  virtual int32_t SetMicrophoneMute(bool enable);
>+  virtual int32_t MicrophoneMute(bool& enabled) const;
>+
>+  // Stereo support
>+  virtual int32_t StereoPlayoutIsAvailable(bool& available);
>+  virtual int32_t SetStereoPlayout(bool enable);
>+  virtual int32_t StereoPlayout(bool& enabled) const;
>+  virtual int32_t StereoRecordingIsAvailable(bool& available);
>+  virtual int32_t SetStereoRecording(bool enable);
>+  virtual int32_t StereoRecording(bool& enabled) const;
>+
>+  // Delay information and control
>+  virtual int32_t PlayoutDelay(uint16_t& delayMS) const;
>+
>+  virtual bool BuiltInAECIsAvailable() const;
>+
>+  virtual int32_t EnableBuiltInAEC(bool enable);
> 
>-class AudioDeviceWindowsCore : public AudioDeviceGeneric
>-{
>-public:
>-    AudioDeviceWindowsCore();
>-    ~AudioDeviceWindowsCore();
>-
>-    static bool CoreAudioIsSupported();
>-
>-    // Retrieve the currently utilized audio layer
>-    virtual int32_t ActiveAudioLayer(AudioDeviceModule::AudioLayer& audioLayer) const;
>-
>-    // Main initializaton and termination
>-    virtual InitStatus Init();
>-    virtual int32_t Terminate();
>-    virtual bool Initialized() const;
>-
>-    // Device enumeration
>-    virtual int16_t PlayoutDevices();
>-    virtual int16_t RecordingDevices();
>-    virtual int32_t PlayoutDeviceName(
>-        uint16_t index,
>-        char name[kAdmMaxDeviceNameSize],
>-        char guid[kAdmMaxGuidSize]);
>-    virtual int32_t RecordingDeviceName(
>-        uint16_t index,
>-        char name[kAdmMaxDeviceNameSize],
>-        char guid[kAdmMaxGuidSize]);
>-
>-    // Device selection
>-    virtual int32_t SetPlayoutDevice(uint16_t index);
>-    virtual int32_t SetPlayoutDevice(AudioDeviceModule::WindowsDeviceType device);
>-    virtual int32_t SetRecordingDevice(uint16_t index);
>-    virtual int32_t SetRecordingDevice(AudioDeviceModule::WindowsDeviceType device);
>-
>-    // Audio transport initialization
>-    virtual int32_t PlayoutIsAvailable(bool& available);
>-    virtual int32_t InitPlayout();
>-    virtual bool PlayoutIsInitialized() const;
>-    virtual int32_t RecordingIsAvailable(bool& available);
>-    virtual int32_t InitRecording();
>-    virtual bool RecordingIsInitialized() const;
>-
>-    // Audio transport control
>-    virtual int32_t StartPlayout();
>-    virtual int32_t StopPlayout();
>-    virtual bool Playing() const;
>-    virtual int32_t StartRecording();
>-    virtual int32_t StopRecording();
>-    virtual bool Recording() const;
>-
>-    // Audio mixer initialization
>-    virtual int32_t InitSpeaker();
>-    virtual bool SpeakerIsInitialized() const;
>-    virtual int32_t InitMicrophone();
>-    virtual bool MicrophoneIsInitialized() const;
>-
>-    // Speaker volume controls
>-    virtual int32_t SpeakerVolumeIsAvailable(bool& available);
>-    virtual int32_t SetSpeakerVolume(uint32_t volume);
>-    virtual int32_t SpeakerVolume(uint32_t& volume) const;
>-    virtual int32_t MaxSpeakerVolume(uint32_t& maxVolume) const;
>-    virtual int32_t MinSpeakerVolume(uint32_t& minVolume) const;
>-
>-    // Microphone volume controls
>-    virtual int32_t MicrophoneVolumeIsAvailable(bool& available);
>-    virtual int32_t SetMicrophoneVolume(uint32_t volume);
>-    virtual int32_t MicrophoneVolume(uint32_t& volume) const;
>-    virtual int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const;
>-    virtual int32_t MinMicrophoneVolume(uint32_t& minVolume) const;
>-
>-    // Speaker mute control
>-    virtual int32_t SpeakerMuteIsAvailable(bool& available);
>-    virtual int32_t SetSpeakerMute(bool enable);
>-    virtual int32_t SpeakerMute(bool& enabled) const;
>-
>-    // Microphone mute control
>-    virtual int32_t MicrophoneMuteIsAvailable(bool& available);
>-    virtual int32_t SetMicrophoneMute(bool enable);
>-    virtual int32_t MicrophoneMute(bool& enabled) const;
>-
>-    // Stereo support
>-    virtual int32_t StereoPlayoutIsAvailable(bool& available);
>-    virtual int32_t SetStereoPlayout(bool enable);
>-    virtual int32_t StereoPlayout(bool& enabled) const;
>-    virtual int32_t StereoRecordingIsAvailable(bool& available);
>-    virtual int32_t SetStereoRecording(bool enable);
>-    virtual int32_t StereoRecording(bool& enabled) const;
>-
>-    // Delay information and control
>-    virtual int32_t PlayoutDelay(uint16_t& delayMS) const;
>-
>-    virtual int32_t EnableBuiltInAEC(bool enable);
>-
>-public:
>-    virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
>-
>-private:
>-    bool KeyPressed() const;
>-
>-private:    // avrt function pointers
>-    PAvRevertMmThreadCharacteristics    _PAvRevertMmThreadCharacteristics;
>-    PAvSetMmThreadCharacteristicsA      _PAvSetMmThreadCharacteristicsA;
>-    PAvSetMmThreadPriority              _PAvSetMmThreadPriority;
>-    HMODULE                             _avrtLibrary;
>-    bool                                _winSupportAvrt;
>-
>-private:    // thread functions
>-    DWORD InitCaptureThreadPriority();
>-    void RevertCaptureThreadPriority();
>-    static DWORD WINAPI WSAPICaptureThread(LPVOID context);
>-    DWORD DoCaptureThread();
>-
>-    static DWORD WINAPI WSAPICaptureThreadPollDMO(LPVOID context);
>-    DWORD DoCaptureThreadPollDMO();
>-
>-    static DWORD WINAPI WSAPIRenderThread(LPVOID context);
>-    DWORD DoRenderThread();
>-
>-    void _Lock() { _critSect.Enter(); };
>-    void _UnLock() { _critSect.Leave(); };
>-
>-    int SetDMOProperties();
>-
>-    int SetBoolProperty(IPropertyStore* ptrPS,
>-                        REFPROPERTYKEY key,
>-                        VARIANT_BOOL value);
>-
>-    int SetVtI4Property(IPropertyStore* ptrPS,
>-                        REFPROPERTYKEY key,
>-                        LONG value);
>-
>-    int32_t _EnumerateEndpointDevicesAll(EDataFlow dataFlow) const;
>-    void _TraceCOMError(HRESULT hr) const;
>-
>-    int32_t _RefreshDeviceList(EDataFlow dir);
>-    int16_t _DeviceListCount(EDataFlow dir);
>-    int32_t _GetDefaultDeviceName(EDataFlow dir, ERole role, LPWSTR szBuffer, int bufferLen);
>-    int32_t _GetListDeviceName(EDataFlow dir, int index, LPWSTR szBuffer, int bufferLen);
>-    int32_t _GetDeviceName(IMMDevice* pDevice, LPWSTR pszBuffer, int bufferLen);
>-    int32_t _GetListDeviceID(EDataFlow dir, int index, LPWSTR szBuffer, int bufferLen);
>-    int32_t _GetDefaultDeviceID(EDataFlow dir, ERole role, LPWSTR szBuffer, int bufferLen);
>-    int32_t _GetDefaultDeviceIndex(EDataFlow dir, ERole role, int* index);
>-    int32_t _GetDeviceID(IMMDevice* pDevice, LPWSTR pszBuffer, int bufferLen);
>-    int32_t _GetDefaultDevice(EDataFlow dir, ERole role, IMMDevice** ppDevice);
>-    int32_t _GetListDevice(EDataFlow dir, int index, IMMDevice** ppDevice);
>-
>-    // Converts from wide-char to UTF-8 if UNICODE is defined.
>-    // Does nothing if UNICODE is undefined.
>-    char* WideToUTF8(const TCHAR* src) const;
>-
>-    int32_t InitRecordingDMO();
>-
>-    ScopedCOMInitializer                    _comInit;
>-    AudioDeviceBuffer*                      _ptrAudioBuffer;
>-    rtc::CriticalSection                    _critSect;
>-    rtc::CriticalSection                    _volumeMutex;
>-
>-    IMMDeviceEnumerator*                    _ptrEnumerator;
>-    IMMDeviceCollection*                    _ptrRenderCollection;
>-    IMMDeviceCollection*                    _ptrCaptureCollection;
>-    IMMDevice*                              _ptrDeviceOut;
>-    IMMDevice*                              _ptrDeviceIn;
>-
>-    IAudioClient*                           _ptrClientOut;
>-    IAudioClient*                           _ptrClientIn;
>-    IAudioRenderClient*                     _ptrRenderClient;
>-    IAudioCaptureClient*                    _ptrCaptureClient;
>-    IAudioEndpointVolume*                   _ptrCaptureVolume;
>-    ISimpleAudioVolume*                     _ptrRenderSimpleVolume;
>-
>-    // DirectX Media Object (DMO) for the built-in AEC.
>-    rtc::scoped_refptr<IMediaObject> _dmo;
>-    rtc::scoped_refptr<IMediaBuffer> _mediaBuffer;
>-    bool                                    _builtInAecEnabled;
>-
>-    HANDLE                                  _hRenderSamplesReadyEvent;
>-    HANDLE                                  _hPlayThread;
>-    HANDLE                                  _hRenderStartedEvent;
>-    HANDLE                                  _hShutdownRenderEvent;
>-
>-    HANDLE                                  _hCaptureSamplesReadyEvent;
>-    HANDLE                                  _hRecThread;
>-    HANDLE                                  _hCaptureStartedEvent;
>-    HANDLE                                  _hShutdownCaptureEvent;
>-
>-    HANDLE                                  _hMmTask;
>-
>-    UINT                                    _playAudioFrameSize;
>-    uint32_t                          _playSampleRate;
>-    uint32_t                          _devicePlaySampleRate;
>-    uint32_t                          _playBlockSize;
>-    uint32_t                          _devicePlayBlockSize;
>-    uint32_t                          _playChannels;
>-    uint32_t                          _sndCardPlayDelay;
>-    UINT64                                  _writtenSamples;
>-
>-    UINT                                    _recAudioFrameSize;
>-    uint32_t                          _recSampleRate;
>-    uint32_t                          _recBlockSize;
>-    uint32_t                          _recChannels;
>-    UINT64                                  _readSamples;
>-    uint32_t                          _sndCardRecDelay;
>-
>-    uint16_t                          _recChannelsPrioList[3];
>-    uint16_t                          _playChannelsPrioList[2];
>-
>-    LARGE_INTEGER                           _perfCounterFreq;
>-    double                                  _perfCounterFactor;
>-
>-private:
>-    bool                                    _initialized;
>-    bool                                    _recording;
>-    bool                                    _playing;
>-    bool                                    _recIsInitialized;
>-    bool                                    _playIsInitialized;
>-    bool                                    _speakerIsInitialized;
>-    bool                                    _microphoneIsInitialized;
>-
>-    bool                                    _usingInputDeviceIndex;
>-    bool                                    _usingOutputDeviceIndex;
>-    AudioDeviceModule::WindowsDeviceType    _inputDevice;
>-    AudioDeviceModule::WindowsDeviceType    _outputDevice;
>-    uint16_t                          _inputDeviceIndex;
>-    uint16_t                          _outputDeviceIndex;
>-
>-    uint16_t                          _playBufDelay;
>-
>-    mutable char                            _str[512];
>+ public:
>+  virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
>+
>+ private:
>+  bool KeyPressed() const;
>+
>+ private:  // avrt function pointers
>+  PAvRevertMmThreadCharacteristics _PAvRevertMmThreadCharacteristics;
>+  PAvSetMmThreadCharacteristicsA _PAvSetMmThreadCharacteristicsA;
>+  PAvSetMmThreadPriority _PAvSetMmThreadPriority;
>+  HMODULE _avrtLibrary;
>+  bool _winSupportAvrt;
>+
>+ private:  // thread functions
>+  DWORD InitCaptureThreadPriority();
>+  void RevertCaptureThreadPriority();
>+  static DWORD WINAPI WSAPICaptureThread(LPVOID context);
>+  DWORD DoCaptureThread();
>+
>+  static DWORD WINAPI WSAPICaptureThreadPollDMO(LPVOID context);
>+  DWORD DoCaptureThreadPollDMO();
>+
>+  static DWORD WINAPI WSAPIRenderThread(LPVOID context);
>+  DWORD DoRenderThread();
>+
>+  void _Lock();
>+  void _UnLock();
>+
>+  int SetDMOProperties();
>+
>+  int SetBoolProperty(IPropertyStore* ptrPS,
>+                      REFPROPERTYKEY key,
>+                      VARIANT_BOOL value);
>+
>+  int SetVtI4Property(IPropertyStore* ptrPS, REFPROPERTYKEY key, LONG value);
>+
>+  int32_t _EnumerateEndpointDevicesAll(EDataFlow dataFlow) const;
>+  void _TraceCOMError(HRESULT hr) const;
>+
>+  int32_t _RefreshDeviceList(EDataFlow dir);
>+  int16_t _DeviceListCount(EDataFlow dir);
>+  int32_t _GetDefaultDeviceName(EDataFlow dir,
>+                                ERole role,
>+                                LPWSTR szBuffer,
>+                                int bufferLen);
>+  int32_t _GetListDeviceName(EDataFlow dir,
>+                             int index,
>+                             LPWSTR szBuffer,
>+                             int bufferLen);
>+  int32_t _GetDeviceName(IMMDevice* pDevice, LPWSTR pszBuffer, int bufferLen);
>+  int32_t _GetListDeviceID(EDataFlow dir,
>+                           int index,
>+                           LPWSTR szBuffer,
>+                           int bufferLen);
>+  int32_t _GetDefaultDeviceID(EDataFlow dir,
>+                              ERole role,
>+                              LPWSTR szBuffer,
>+                              int bufferLen);
>+  int32_t _GetDefaultDeviceIndex(EDataFlow dir, ERole role, int* index);
>+  int32_t _GetDeviceID(IMMDevice* pDevice, LPWSTR pszBuffer, int bufferLen);
>+  int32_t _GetDefaultDevice(EDataFlow dir, ERole role, IMMDevice** ppDevice);
>+  int32_t _GetListDevice(EDataFlow dir, int index, IMMDevice** ppDevice);
>+
>+  // Converts from wide-char to UTF-8 if UNICODE is defined.
>+  // Does nothing if UNICODE is undefined.
>+  char* WideToUTF8(const TCHAR* src) const;
>+
>+  int32_t InitRecordingDMO();
>+
>+  ScopedCOMInitializer _comInit;
>+  AudioDeviceBuffer* _ptrAudioBuffer;
>+  rtc::CriticalSection _critSect;
>+  rtc::CriticalSection _volumeMutex;
>+
>+  IMMDeviceEnumerator* _ptrEnumerator;
>+  IMMDeviceCollection* _ptrRenderCollection;
>+  IMMDeviceCollection* _ptrCaptureCollection;
>+  IMMDevice* _ptrDeviceOut;
>+  IMMDevice* _ptrDeviceIn;
>+
>+  IAudioClient* _ptrClientOut;
>+  IAudioClient* _ptrClientIn;
>+  IAudioRenderClient* _ptrRenderClient;
>+  IAudioCaptureClient* _ptrCaptureClient;
>+  IAudioEndpointVolume* _ptrCaptureVolume;
>+  ISimpleAudioVolume* _ptrRenderSimpleVolume;
>+
>+  // DirectX Media Object (DMO) for the built-in AEC.
>+  rtc::scoped_refptr<IMediaObject> _dmo;
>+  rtc::scoped_refptr<IMediaBuffer> _mediaBuffer;
>+  bool _builtInAecEnabled;
>+
>+  HANDLE _hRenderSamplesReadyEvent;
>+  HANDLE _hPlayThread;
>+  HANDLE _hRenderStartedEvent;
>+  HANDLE _hShutdownRenderEvent;
>+
>+  HANDLE _hCaptureSamplesReadyEvent;
>+  HANDLE _hRecThread;
>+  HANDLE _hCaptureStartedEvent;
>+  HANDLE _hShutdownCaptureEvent;
>+
>+  HANDLE _hMmTask;
>+
>+  UINT _playAudioFrameSize;
>+  uint32_t _playSampleRate;
>+  uint32_t _devicePlaySampleRate;
>+  uint32_t _playBlockSize;
>+  uint32_t _devicePlayBlockSize;
>+  uint32_t _playChannels;
>+  uint32_t _sndCardPlayDelay;
>+  uint32_t _sndCardRecDelay;
>+  UINT64 _writtenSamples;
>+  UINT64 _readSamples;
>+
>+  UINT _recAudioFrameSize;
>+  uint32_t _recSampleRate;
>+  uint32_t _recBlockSize;
>+  uint32_t _recChannels;
>+
>+  uint16_t _recChannelsPrioList[3];
>+  uint16_t _playChannelsPrioList[2];
>+
>+  LARGE_INTEGER _perfCounterFreq;
>+  double _perfCounterFactor;
>+
>+ private:
>+  bool _initialized;
>+  bool _recording;
>+  bool _playing;
>+  bool _recIsInitialized;
>+  bool _playIsInitialized;
>+  bool _speakerIsInitialized;
>+  bool _microphoneIsInitialized;
>+
>+  bool _usingInputDeviceIndex;
>+  bool _usingOutputDeviceIndex;
>+  AudioDeviceModule::WindowsDeviceType _inputDevice;
>+  AudioDeviceModule::WindowsDeviceType _outputDevice;
>+  uint16_t _inputDeviceIndex;
>+  uint16_t _outputDeviceIndex;
>+
>+  mutable char _str[512];
> };
> 
>-#endif    // #if (_MSC_VER >= 1400)
>+#endif  // #if (_MSC_VER >= 1400)
> 
> }  // namespace webrtc
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/win/audio_device_module_win.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/win/audio_device_module_win.cc
>new file mode 100644
>index 00000000000..b919334bc79
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/win/audio_device_module_win.cc
>@@ -0,0 +1,423 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_device/win/audio_device_module_win.h"
>+
>+#include <utility>
>+
>+#include "modules/audio_device/audio_device_buffer.h"
>+#include "modules/audio_device/include/audio_device.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/logging.h"
>+#include "rtc_base/refcountedobject.h"
>+#include "rtc_base/stringutils.h"
>+#include "rtc_base/thread_checker.h"
>+
>+namespace webrtc {
>+namespace webrtc_win {
>+namespace {
>+
>+// This class combines a generic instance of an AudioInput and a generic
>+// instance of an AudioOutput to create an AudioDeviceModule. This is mostly
>+// done by delegating to the audio input/output with some glue code. This class
>+// also directly implements some of the AudioDeviceModule methods with dummy
>+// implementations.
>+//
>+// An instance must be created, destroyed and used on one and the same thread,
>+// i.e., all public methods must also be called on the same thread. A thread
>+// checker will RTC_DCHECK if any method is called on an invalid thread.
>+// TODO(henrika): is thread checking needed in AudioInput and AudioOutput?
>+class WindowsAudioDeviceModule : public AudioDeviceModule {
>+ public:
>+  enum class InitStatus {
>+    OK = 0,
>+    PLAYOUT_ERROR = 1,
>+    RECORDING_ERROR = 2,
>+    OTHER_ERROR = 3,
>+    NUM_STATUSES = 4
>+  };
>+
>+  WindowsAudioDeviceModule(std::unique_ptr<AudioInput> audio_input,
>+                           std::unique_ptr<AudioOutput> audio_output)
>+      : input_(std::move(audio_input)), output_(std::move(audio_output)) {
>+    RTC_CHECK(input_);
>+    RTC_CHECK(output_);
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    RTC_DCHECK_RUN_ON(&thread_checker_);
>+  }
>+
>+  ~WindowsAudioDeviceModule() override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    RTC_DCHECK_RUN_ON(&thread_checker_);
>+    Terminate();
>+  }
>+
>+  WindowsAudioDeviceModule(const WindowsAudioDeviceModule&) = delete;
>+  WindowsAudioDeviceModule& operator=(const WindowsAudioDeviceModule&) = delete;
>+
>+  int32_t ActiveAudioLayer(
>+      AudioDeviceModule::AudioLayer* audioLayer) const override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    RTC_DCHECK_RUN_ON(&thread_checker_);
>+    // TODO(henrika): it might be possible to remove this unique signature.
>+    *audioLayer = AudioDeviceModule::kWindowsCoreAudio2;
>+    return 0;
>+  }
>+
>+  int32_t RegisterAudioCallback(AudioTransport* audioCallback) override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    RTC_DCHECK(audio_device_buffer_);
>+    RTC_DCHECK_RUN_ON(&thread_checker_);
>+    return audio_device_buffer_->RegisterAudioCallback(audioCallback);
>+  }
>+
>+  int32_t Init() override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    RTC_DCHECK_RUN_ON(&thread_checker_);
>+    if (initialized_) {
>+      return 0;
>+    }
>+    audio_device_buffer_ = absl::make_unique<AudioDeviceBuffer>();
>+    AttachAudioBuffer();
>+    InitStatus status;
>+    if (output_->Init() != 0) {
>+      status = InitStatus::PLAYOUT_ERROR;
>+    } else if (input_->Init() != 0) {
>+      output_->Terminate();
>+      status = InitStatus::RECORDING_ERROR;
>+    } else {
>+      initialized_ = true;
>+      status = InitStatus::OK;
>+    }
>+    if (status != InitStatus::OK) {
>+      RTC_LOG(LS_ERROR) << "Audio device initialization failed";
>+      return -1;
>+    }
>+    return 0;
>+  }
>+
>+  int32_t Terminate() override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    RTC_DCHECK_RUN_ON(&thread_checker_);
>+    if (!initialized_)
>+      return 0;
>+    int32_t err = input_->Terminate();
>+    err |= output_->Terminate();
>+    initialized_ = false;
>+    RTC_DCHECK_EQ(err, 0);
>+    return err;
>+  }
>+
>+  bool Initialized() const override {
>+    RTC_DCHECK_RUN_ON(&thread_checker_);
>+    return initialized_;
>+  }
>+
>+  int16_t PlayoutDevices() override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    RTC_DCHECK_RUN_ON(&thread_checker_);
>+    return output_->NumDevices();
>+  }
>+
>+  int16_t RecordingDevices() override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    RTC_DCHECK_RUN_ON(&thread_checker_);
>+    return input_->NumDevices();
>+  }
>+
>+  int32_t PlayoutDeviceName(uint16_t index,
>+                            char name[kAdmMaxDeviceNameSize],
>+                            char guid[kAdmMaxGuidSize]) override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    RTC_DCHECK_RUN_ON(&thread_checker_);
>+    std::string name_str, guid_str;
>+    int ret = -1;
>+    if (guid != nullptr) {
>+      ret = output_->DeviceName(index, &name_str, &guid_str);
>+      rtc::strcpyn(guid, kAdmMaxGuidSize, guid_str.c_str());
>+    } else {
>+      ret = output_->DeviceName(index, &name_str, nullptr);
>+    }
>+    rtc::strcpyn(name, kAdmMaxDeviceNameSize, name_str.c_str());
>+    return ret;
>+  }
>+  int32_t RecordingDeviceName(uint16_t index,
>+                              char name[kAdmMaxDeviceNameSize],
>+                              char guid[kAdmMaxGuidSize]) override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    RTC_DCHECK_RUN_ON(&thread_checker_);
>+    std::string name_str, guid_str;
>+    int ret = -1;
>+    if (guid != nullptr) {
>+      ret = input_->DeviceName(index, &name_str, &guid_str);
>+      rtc::strcpyn(guid, kAdmMaxGuidSize, guid_str.c_str());
>+    } else {
>+      ret = input_->DeviceName(index, &name_str, nullptr);
>+    }
>+    rtc::strcpyn(name, kAdmMaxDeviceNameSize, name_str.c_str());
>+    return ret;
>+  }
>+
>+  int32_t SetPlayoutDevice(uint16_t index) override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    RTC_DCHECK_RUN_ON(&thread_checker_);
>+    return output_->SetDevice(index);
>+  }
>+
>+  int32_t SetPlayoutDevice(
>+      AudioDeviceModule::WindowsDeviceType device) override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    RTC_DCHECK_RUN_ON(&thread_checker_);
>+    return output_->SetDevice(device);
>+  }
>+  int32_t SetRecordingDevice(uint16_t index) override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    RTC_DCHECK_RUN_ON(&thread_checker_);
>+    return input_->SetDevice(index);
>+  }
>+
>+  int32_t SetRecordingDevice(
>+      AudioDeviceModule::WindowsDeviceType device) override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    RTC_DCHECK_RUN_ON(&thread_checker_);
>+    return input_->SetDevice(device);
>+  }
>+
>+  int32_t PlayoutIsAvailable(bool* available) override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    RTC_DCHECK_RUN_ON(&thread_checker_);
>+    *available = true;
>+    return 0;
>+  }
>+
>+  int32_t InitPlayout() override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    RTC_DCHECK_RUN_ON(&thread_checker_);
>+    return output_->InitPlayout();
>+  }
>+
>+  bool PlayoutIsInitialized() const override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    RTC_DCHECK_RUN_ON(&thread_checker_);
>+    return output_->PlayoutIsInitialized();
>+  }
>+
>+  int32_t RecordingIsAvailable(bool* available) override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    RTC_DCHECK_RUN_ON(&thread_checker_);
>+    *available = true;
>+    return 0;
>+  }
>+
>+  int32_t InitRecording() override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    RTC_DCHECK_RUN_ON(&thread_checker_);
>+    return input_->InitRecording();
>+  }
>+
>+  bool RecordingIsInitialized() const override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    RTC_DCHECK_RUN_ON(&thread_checker_);
>+    return input_->RecordingIsInitialized();
>+  }
>+
>+  int32_t StartPlayout() override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    RTC_DCHECK_RUN_ON(&thread_checker_);
>+    return output_->StartPlayout();
>+  }
>+
>+  int32_t StopPlayout() override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    RTC_DCHECK_RUN_ON(&thread_checker_);
>+    return output_->StopPlayout();
>+  }
>+
>+  bool Playing() const override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    RTC_DCHECK_RUN_ON(&thread_checker_);
>+    return output_->Playing();
>+  }
>+
>+  int32_t StartRecording() override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    RTC_DCHECK_RUN_ON(&thread_checker_);
>+    return input_->StartRecording();
>+  }
>+
>+  int32_t StopRecording() override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    RTC_DCHECK_RUN_ON(&thread_checker_);
>+    return input_->StopRecording();
>+  }
>+
>+  bool Recording() const override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    return input_->Recording();
>+  }
>+
>+  int32_t InitSpeaker() override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    RTC_DCHECK_RUN_ON(&thread_checker_);
>+    RTC_DLOG(LS_WARNING) << "This method has no effect";
>+    return initialized_ ? 0 : -1;
>+  }
>+
>+  bool SpeakerIsInitialized() const override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    RTC_DCHECK_RUN_ON(&thread_checker_);
>+    RTC_DLOG(LS_WARNING) << "This method has no effect";
>+    return initialized_;
>+  }
>+
>+  int32_t InitMicrophone() override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    RTC_DCHECK_RUN_ON(&thread_checker_);
>+    RTC_DLOG(LS_WARNING) << "This method has no effect";
>+    return initialized_ ? 0 : -1;
>+  }
>+
>+  bool MicrophoneIsInitialized() const override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    RTC_DCHECK_RUN_ON(&thread_checker_);
>+    RTC_DLOG(LS_WARNING) << "This method has no effect";
>+    return initialized_;
>+  }
>+
>+  int32_t SpeakerVolumeIsAvailable(bool* available) override {
>+    // TODO(henrika): improve support.
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    RTC_DCHECK_RUN_ON(&thread_checker_);
>+    *available = false;
>+    return 0;
>+  }
>+
>+  int32_t SetSpeakerVolume(uint32_t volume) override { return 0; }
>+  int32_t SpeakerVolume(uint32_t* volume) const override { return 0; }
>+  int32_t MaxSpeakerVolume(uint32_t* maxVolume) const override { return 0; }
>+  int32_t MinSpeakerVolume(uint32_t* minVolume) const override { return 0; }
>+
>+  int32_t MicrophoneVolumeIsAvailable(bool* available) override {
>+    // TODO(henrika): improve support.
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    RTC_DCHECK_RUN_ON(&thread_checker_);
>+    *available = false;
>+    return 0;
>+  }
>+
>+  int32_t SetMicrophoneVolume(uint32_t volume) override { return 0; }
>+  int32_t MicrophoneVolume(uint32_t* volume) const override { return 0; }
>+  int32_t MaxMicrophoneVolume(uint32_t* maxVolume) const override { return 0; }
>+  int32_t MinMicrophoneVolume(uint32_t* minVolume) const override { return 0; }
>+
>+  int32_t SpeakerMuteIsAvailable(bool* available) override { return 0; }
>+  int32_t SetSpeakerMute(bool enable) override { return 0; }
>+  int32_t SpeakerMute(bool* enabled) const override { return 0; }
>+
>+  int32_t MicrophoneMuteIsAvailable(bool* available) override { return 0; }
>+  int32_t SetMicrophoneMute(bool enable) override { return 0; }
>+  int32_t MicrophoneMute(bool* enabled) const override { return 0; }
>+
>+  int32_t StereoPlayoutIsAvailable(bool* available) const override {
>+    // TODO(henrika): improve support.
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    RTC_DCHECK_RUN_ON(&thread_checker_);
>+    *available = true;
>+    return 0;
>+  }
>+
>+  int32_t SetStereoPlayout(bool enable) override {
>+    // TODO(henrika): improve support.
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    RTC_DCHECK_RUN_ON(&thread_checker_);
>+    return 0;
>+  }
>+
>+  int32_t StereoPlayout(bool* enabled) const override {
>+    // TODO(henrika): improve support.
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    RTC_DCHECK_RUN_ON(&thread_checker_);
>+    *enabled = true;
>+    return 0;
>+  }
>+
>+  int32_t StereoRecordingIsAvailable(bool* available) const override {
>+    // TODO(henrika): improve support.
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    RTC_DCHECK_RUN_ON(&thread_checker_);
>+    *available = true;
>+    return 0;
>+  }
>+
>+  int32_t SetStereoRecording(bool enable) override {
>+    // TODO(henrika): improve support.
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    RTC_DCHECK_RUN_ON(&thread_checker_);
>+    return 0;
>+  }
>+
>+  int32_t StereoRecording(bool* enabled) const override {
>+    // TODO(henrika): improve support.
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    RTC_DCHECK_RUN_ON(&thread_checker_);
>+    *enabled = true;
>+    return 0;
>+  }
>+
>+  int32_t PlayoutDelay(uint16_t* delayMS) const override { return 0; }
>+
>+  bool BuiltInAECIsAvailable() const override { return false; }
>+  bool BuiltInAGCIsAvailable() const override { return false; }
>+  bool BuiltInNSIsAvailable() const override { return false; }
>+
>+  int32_t EnableBuiltInAEC(bool enable) override { return 0; }
>+  int32_t EnableBuiltInAGC(bool enable) override { return 0; }
>+  int32_t EnableBuiltInNS(bool enable) override { return 0; }
>+
>+  int32_t AttachAudioBuffer() {
>+    RTC_DLOG(INFO) << __FUNCTION__;
>+    output_->AttachAudioBuffer(audio_device_buffer_.get());
>+    input_->AttachAudioBuffer(audio_device_buffer_.get());
>+    return 0;
>+  }
>+
>+ private:
>+  // Ensures that the class is used on the same thread as it is constructed
>+  // and destroyed on.
>+  rtc::ThreadChecker thread_checker_;
>+
>+  // Implements the AudioInput interface and deals with audio capturing parts.
>+  const std::unique_ptr<AudioInput> input_;
>+
>+  // Implements the AudioOutput interface and deals with audio rendering parts.
>+  const std::unique_ptr<AudioOutput> output_;
>+
>+  // The AudioDeviceBuffer (ADB) instance is needed for sending/receiving audio
>+  // to/from the WebRTC layer. Created and owned by this object. Used by
>+  // both |input_| and |output_| but they use orthogonal parts of the ADB.
>+  std::unique_ptr<AudioDeviceBuffer> audio_device_buffer_;
>+
>+  // Set to true after a successful call to Init(). Cleared by Terminate().
>+  bool initialized_ = false;
>+};
>+
>+}  // namespace
>+
>+rtc::scoped_refptr<AudioDeviceModule>
>+CreateWindowsCoreAudioAudioDeviceModuleFromInputAndOutput(
>+    std::unique_ptr<AudioInput> audio_input,
>+    std::unique_ptr<AudioOutput> audio_output) {
>+  RTC_LOG(INFO) << __FUNCTION__;
>+  return new rtc::RefCountedObject<WindowsAudioDeviceModule>(
>+      std::move(audio_input), std::move(audio_output));
>+}
>+
>+}  // namespace webrtc_win
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/win/audio_device_module_win.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/win/audio_device_module_win.h
>new file mode 100644
>index 00000000000..7484ef7f8dc
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/win/audio_device_module_win.h
>@@ -0,0 +1,79 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_DEVICE_WIN_AUDIO_DEVICE_MODULE_WIN_H_
>+#define MODULES_AUDIO_DEVICE_WIN_AUDIO_DEVICE_MODULE_WIN_H_
>+
>+#include <memory>
>+#include <string>
>+
>+#include "modules/audio_device/include/audio_device.h"
>+#include "rtc_base/scoped_ref_ptr.h"
>+
>+namespace webrtc {
>+
>+class AudioDeviceBuffer;
>+
>+namespace webrtc_win {
>+
>+// This interface represents the main input-related parts of the complete
>+// AudioDeviceModule interface.
>+class AudioInput {
>+ public:
>+  virtual ~AudioInput() {}
>+
>+  virtual int Init() = 0;
>+  virtual int Terminate() = 0;
>+  virtual int NumDevices() const = 0;
>+  virtual int SetDevice(int index) = 0;
>+  virtual int SetDevice(AudioDeviceModule::WindowsDeviceType device) = 0;
>+  virtual int DeviceName(int index, std::string* name, std::string* guid) = 0;
>+  virtual void AttachAudioBuffer(AudioDeviceBuffer* audio_buffer) = 0;
>+  virtual bool RecordingIsInitialized() const = 0;
>+  virtual int InitRecording() = 0;
>+  virtual int StartRecording() = 0;
>+  virtual int StopRecording() = 0;
>+  virtual bool Recording() = 0;
>+  virtual int VolumeIsAvailable(bool* available) = 0;
>+};
>+
>+// This interface represents the main output-related parts of the complete
>+// AudioDeviceModule interface.
>+class AudioOutput {
>+ public:
>+  virtual ~AudioOutput() {}
>+
>+  virtual int Init() = 0;
>+  virtual int Terminate() = 0;
>+  virtual int NumDevices() const = 0;
>+  virtual int SetDevice(int index) = 0;
>+  virtual int SetDevice(AudioDeviceModule::WindowsDeviceType device) = 0;
>+  virtual int DeviceName(int index, std::string* name, std::string* guid) = 0;
>+  virtual void AttachAudioBuffer(AudioDeviceBuffer* audio_buffer) = 0;
>+  virtual bool PlayoutIsInitialized() const = 0;
>+  virtual int InitPlayout() = 0;
>+  virtual int StartPlayout() = 0;
>+  virtual int StopPlayout() = 0;
>+  virtual bool Playing() = 0;
>+  virtual int VolumeIsAvailable(bool* available) = 0;
>+};
>+
>+// Combines an AudioInput and an AudioOutput implementation to build an
>+// AudioDeviceModule. Hides most parts of the full ADM interface.
>+rtc::scoped_refptr<AudioDeviceModule>
>+CreateWindowsCoreAudioAudioDeviceModuleFromInputAndOutput(
>+    std::unique_ptr<AudioInput> audio_input,
>+    std::unique_ptr<AudioOutput> audio_output);
>+
>+}  // namespace webrtc_win
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_DEVICE_WIN_AUDIO_DEVICE_MODULE_WIN_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/win/core_audio_base_win.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/win/core_audio_base_win.cc
>new file mode 100644
>index 00000000000..2f37e4bcb6a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/win/core_audio_base_win.cc
>@@ -0,0 +1,439 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_device/win/core_audio_base_win.h"
>+
>+#include <string>
>+
>+#include "absl/memory/memory.h"
>+#include "rtc_base/arraysize.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/logging.h"
>+#include "rtc_base/numerics/safe_conversions.h"
>+#include "rtc_base/win/windows_version.h"
>+
>+using Microsoft::WRL::ComPtr;
>+
>+namespace webrtc {
>+namespace webrtc_win {
>+namespace {
>+
>+enum DefaultDeviceType {
>+  kDefault,
>+  kDefaultCommunications,
>+  kDefaultDeviceTypeMaxCount,
>+};
>+
>+const char* DirectionToString(CoreAudioBase::Direction direction) {
>+  switch (direction) {
>+    case CoreAudioBase::Direction::kOutput:
>+      return "Output";
>+    case CoreAudioBase::Direction::kInput:
>+      return "Input";
>+    default:
>+      return "Unkown";
>+  }
>+}
>+
>+void Run(void* obj) {
>+  RTC_DCHECK(obj);
>+  reinterpret_cast<CoreAudioBase*>(obj)->ThreadRun();
>+}
>+
>+}  // namespace
>+
>+CoreAudioBase::CoreAudioBase(Direction direction, OnDataCallback callback)
>+    : direction_(direction), on_data_callback_(callback), format_() {
>+  RTC_DLOG(INFO) << __FUNCTION__ << "[" << DirectionToString(direction) << "]";
>+
>+  // Create the event which the audio engine will signal each time a buffer
>+  // becomes ready to be processed by the client.
>+  audio_samples_event_.Set(CreateEvent(nullptr, false, false, nullptr));
>+  RTC_DCHECK(audio_samples_event_.IsValid());
>+
>+  // Event to be be set in Stop() when rendering/capturing shall stop.
>+  stop_event_.Set(CreateEvent(nullptr, false, false, nullptr));
>+  RTC_DCHECK(stop_event_.IsValid());
>+}
>+
>+CoreAudioBase::~CoreAudioBase() {
>+  RTC_DLOG(INFO) << __FUNCTION__;
>+}
>+
>+EDataFlow CoreAudioBase::GetDataFlow() const {
>+  return direction_ == CoreAudioBase::Direction::kOutput ? eRender : eCapture;
>+}
>+
>+int CoreAudioBase::NumberOfActiveDevices() const {
>+  return core_audio_utility::NumberOfActiveDevices(GetDataFlow());
>+}
>+
>+int CoreAudioBase::NumberOfEnumeratedDevices() const {
>+  const int num_active = NumberOfActiveDevices();
>+  return num_active > 0 ? num_active + kDefaultDeviceTypeMaxCount : 0;
>+}
>+
>+bool CoreAudioBase::IsDefaultDevice(int index) const {
>+  return index == kDefault;
>+}
>+
>+bool CoreAudioBase::IsDefaultCommunicationsDevice(int index) const {
>+  return index == kDefaultCommunications;
>+}
>+
>+bool CoreAudioBase::IsDefaultDevice(const std::string& device_id) const {
>+  return (IsInput() &&
>+          (device_id == core_audio_utility::GetDefaultInputDeviceID())) ||
>+         (IsOutput() &&
>+          (device_id == core_audio_utility::GetDefaultOutputDeviceID()));
>+}
>+
>+bool CoreAudioBase::IsDefaultCommunicationsDevice(
>+    const std::string& device_id) const {
>+  return (IsInput() &&
>+          (device_id ==
>+           core_audio_utility::GetCommunicationsInputDeviceID())) ||
>+         (IsOutput() &&
>+          (device_id == core_audio_utility::GetCommunicationsOutputDeviceID()));
>+}
>+
>+bool CoreAudioBase::IsInput() const {
>+  return direction_ == CoreAudioBase::Direction::kInput;
>+}
>+
>+bool CoreAudioBase::IsOutput() const {
>+  return direction_ == CoreAudioBase::Direction::kOutput;
>+}
>+
>+std::string CoreAudioBase::GetDeviceID(int index) const {
>+  if (index >= NumberOfEnumeratedDevices()) {
>+    RTC_LOG(LS_ERROR) << "Invalid device index";
>+    return std::string();
>+  }
>+
>+  std::string device_id;
>+  if (IsDefaultDevice(index)) {
>+    device_id = IsInput() ? core_audio_utility::GetDefaultInputDeviceID()
>+                          : core_audio_utility::GetDefaultOutputDeviceID();
>+  } else if (IsDefaultCommunicationsDevice(index)) {
>+    device_id = IsInput()
>+                    ? core_audio_utility::GetCommunicationsInputDeviceID()
>+                    : core_audio_utility::GetCommunicationsOutputDeviceID();
>+  } else {
>+    AudioDeviceNames device_names;
>+    bool ok = IsInput()
>+                  ? core_audio_utility::GetInputDeviceNames(&device_names)
>+                  : core_audio_utility::GetOutputDeviceNames(&device_names);
>+    if (ok) {
>+      device_id = device_names[index].unique_id;
>+    }
>+  }
>+  return device_id;
>+}
>+
>+int CoreAudioBase::DeviceName(int index,
>+                              std::string* name,
>+                              std::string* guid) const {
>+  RTC_DLOG(INFO) << __FUNCTION__ << "[" << DirectionToString(direction())
>+                 << "]";
>+  if (index > NumberOfEnumeratedDevices() - 1) {
>+    RTC_LOG(LS_ERROR) << "Invalid device index";
>+    return -1;
>+  }
>+
>+  AudioDeviceNames device_names;
>+  bool ok = IsInput() ? core_audio_utility::GetInputDeviceNames(&device_names)
>+                      : core_audio_utility::GetOutputDeviceNames(&device_names);
>+  if (!ok) {
>+    RTC_LOG(LS_ERROR) << "Failed to get the device name";
>+    return -1;
>+  }
>+
>+  *name = device_names[index].device_name;
>+  RTC_DLOG(INFO) << "name: " << *name;
>+  if (guid != nullptr) {
>+    *guid = device_names[index].unique_id;
>+    RTC_DLOG(INFO) << "guid: " << guid;
>+  }
>+  return 0;
>+}
>+
>+bool CoreAudioBase::Init() {
>+  RTC_DLOG(INFO) << __FUNCTION__ << "[" << DirectionToString(direction())
>+                 << "]";
>+  RTC_DCHECK(!device_id_.empty());
>+  RTC_DCHECK(audio_device_buffer_);
>+  RTC_DCHECK(!audio_client_.Get());
>+
>+  // Use an existing |device_id_| and set parameters which are required to
>+  // create an audio client. It is up to the parent class to set |device_id_|.
>+  // TODO(henrika): improve device notification.
>+  std::string device_id = device_id_;
>+  ERole role = eConsole;
>+  if (IsDefaultDevice(device_id)) {
>+    device_id = AudioDeviceName::kDefaultDeviceId;
>+    role = eConsole;
>+  } else if (IsDefaultCommunicationsDevice(device_id)) {
>+    device_id = AudioDeviceName::kDefaultCommunicationsDeviceId;
>+    role = eCommunications;
>+  }
>+
>+  // Create an IAudioClient interface which enables us to create and initialize
>+  // an audio stream between an audio application and the audio engine.
>+  ComPtr<IAudioClient> audio_client =
>+      core_audio_utility::CreateClient(device_id, GetDataFlow(), role);
>+  if (!audio_client.Get()) {
>+    return false;
>+  }
>+
>+  // Retrieve preferred audio input or output parameters for the given client.
>+  AudioParameters params;
>+  if (FAILED(core_audio_utility::GetPreferredAudioParameters(audio_client.Get(),
>+                                                             &params))) {
>+    return false;
>+  }
>+
>+  // Define the output WAVEFORMATEXTENSIBLE format in |format_|.
>+  WAVEFORMATEX* format = &format_.Format;
>+  format->wFormatTag = WAVE_FORMAT_EXTENSIBLE;
>+  format->nChannels = rtc::dchecked_cast<WORD>(params.channels());
>+  format->nSamplesPerSec = params.sample_rate();
>+  format->wBitsPerSample = rtc::dchecked_cast<WORD>(params.bits_per_sample());
>+  format->nBlockAlign = (format->wBitsPerSample / 8) * format->nChannels;
>+  format->nAvgBytesPerSec = format->nSamplesPerSec * format->nBlockAlign;
>+  format->cbSize = sizeof(WAVEFORMATEXTENSIBLE) - sizeof(WAVEFORMATEX);
>+  // Add the parts which are unique for the WAVE_FORMAT_EXTENSIBLE structure.
>+  format_.Samples.wValidBitsPerSample =
>+      rtc::dchecked_cast<WORD>(params.bits_per_sample());
>+  // TODO(henrika): improve (common for input and output?)
>+  format_.dwChannelMask = params.channels() == 1
>+                              ? SPEAKER_FRONT_CENTER
>+                              : SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT;
>+  format_.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
>+  RTC_DLOG(INFO) << core_audio_utility::WaveFormatExToString(&format_);
>+
>+  // Verify that the format is supported.
>+  if (!core_audio_utility::IsFormatSupported(
>+          audio_client.Get(), AUDCLNT_SHAREMODE_SHARED, &format_)) {
>+    return false;
>+  }
>+
>+  // Initialize the audio stream between the client and the device in shared
>+  // mode using event-driven buffer handling.
>+  if (FAILED(core_audio_utility::SharedModeInitialize(
>+          audio_client.Get(), &format_, audio_samples_event_,
>+          &endpoint_buffer_size_frames_))) {
>+    return false;
>+  }
>+
>+  // Check device period and the preferred buffer size and log a warning if
>+  // WebRTC's buffer size is not an even divisor of the preferred buffer size
>+  // in Core Audio.
>+  // TODO(henrik): sort out if a non-perfect match really is an issue.
>+  REFERENCE_TIME device_period;
>+  if (FAILED(core_audio_utility::GetDevicePeriod(
>+          audio_client.Get(), AUDCLNT_SHAREMODE_SHARED, &device_period))) {
>+    return false;
>+  }
>+  const double device_period_in_seconds =
>+      static_cast<double>(
>+          core_audio_utility::ReferenceTimeToTimeDelta(device_period).ms()) /
>+      1000.0L;
>+  const int preferred_frames_per_buffer =
>+      static_cast<int>(params.sample_rate() * device_period_in_seconds + 0.5);
>+  RTC_DLOG(INFO) << "preferred_frames_per_buffer: "
>+                 << preferred_frames_per_buffer;
>+  if (preferred_frames_per_buffer % params.frames_per_buffer()) {
>+    RTC_LOG(WARNING) << "Buffer size of " << params.frames_per_buffer()
>+                     << " is not an even divisor of "
>+                     << preferred_frames_per_buffer;
>+  }
>+
>+  // Store valid COM interfaces.
>+  audio_client_ = audio_client;
>+
>+  return true;
>+}
>+
>+bool CoreAudioBase::Start() {
>+  RTC_DLOG(INFO) << __FUNCTION__ << "[" << DirectionToString(direction())
>+                 << "]";
>+
>+  audio_thread_ = absl::make_unique<rtc::PlatformThread>(
>+      Run, this, IsInput() ? "wasapi_capture_thread" : "wasapi_render_thread",
>+      rtc::kRealtimePriority);
>+  audio_thread_->Start();
>+  if (!audio_thread_->IsRunning()) {
>+    StopThread();
>+    RTC_LOG(LS_ERROR) << "Failed to start audio thread";
>+    return false;
>+  }
>+  RTC_DLOG(INFO) << "Started thread with name: " << audio_thread_->name();
>+
>+  // Start streaming data between the endpoint buffer and the audio engine.
>+  _com_error error = audio_client_->Start();
>+  if (error.Error() != S_OK) {
>+    StopThread();
>+    RTC_LOG(LS_ERROR) << "IAudioClient::Start failed: "
>+                      << core_audio_utility::ErrorToString(error);
>+    return false;
>+  }
>+
>+  return true;
>+}
>+
>+bool CoreAudioBase::Stop() {
>+  RTC_DLOG(INFO) << __FUNCTION__ << "[" << DirectionToString(direction())
>+                 << "]";
>+
>+  // Stop streaming and the internal audio thread.
>+  _com_error error = audio_client_->Stop();
>+  if (error.Error() != S_OK) {
>+    RTC_LOG(LS_ERROR) << "IAudioClient::Stop failed: "
>+                      << core_audio_utility::ErrorToString(error);
>+  }
>+  StopThread();
>+
>+  // Flush all pending data and reset the audio clock stream position to 0.
>+  error = audio_client_->Reset();
>+  if (error.Error() != S_OK) {
>+    RTC_LOG(LS_ERROR) << "IAudioClient::Reset failed: "
>+                      << core_audio_utility::ErrorToString(error);
>+  }
>+
>+  if (IsOutput()) {
>+    // Extra safety check to ensure that the buffers are cleared.
>+    // If the buffers are not cleared correctly, the next call to Start()
>+    // would fail with AUDCLNT_E_BUFFER_ERROR at
>+    // IAudioRenderClient::GetBuffer().
>+    UINT32 num_queued_frames = 0;
>+    audio_client_->GetCurrentPadding(&num_queued_frames);
>+    RTC_DCHECK_EQ(0u, num_queued_frames);
>+  }
>+
>+  return true;
>+}
>+
>+bool CoreAudioBase::IsVolumeControlAvailable(bool* available) const {
>+  // A valid IAudioClient is required to access the ISimpleAudioVolume interface
>+  // properly. It is possible to use IAudioSessionManager::GetSimpleAudioVolume
>+  // as well but we use the audio client here to ensure that the initialized
>+  // audio session is visible under group box labeled "Applications" in
>+  // Sndvol.exe.
>+  if (!audio_client_.Get()) {
>+    return false;
>+  }
>+
>+  // Try to create an ISimpleAudioVolume instance.
>+  ComPtr<ISimpleAudioVolume> audio_volume =
>+      core_audio_utility::CreateSimpleAudioVolume(audio_client_.Get());
>+  if (!audio_volume.Get()) {
>+    RTC_DLOG(LS_ERROR) << "Volume control is not supported";
>+    return false;
>+  }
>+
>+  // Try to use the valid volume control.
>+  float volume = 0.0;
>+  _com_error error = audio_volume->GetMasterVolume(&volume);
>+  if (error.Error() != S_OK) {
>+    RTC_LOG(LS_ERROR) << "ISimpleAudioVolume::GetMasterVolume failed: "
>+                      << core_audio_utility::ErrorToString(error);
>+    *available = false;
>+  }
>+  RTC_DLOG(INFO) << "master volume for output audio session: " << volume;
>+
>+  *available = true;
>+  return false;
>+}
>+
>+void CoreAudioBase::StopThread() {
>+  RTC_DLOG(INFO) << __FUNCTION__;
>+  if (audio_thread_) {
>+    if (audio_thread_->IsRunning()) {
>+      RTC_DLOG(INFO) << "Sets stop_event...";
>+      SetEvent(stop_event_.Get());
>+      RTC_DLOG(INFO) << "PlatformThread::Stop...";
>+      audio_thread_->Stop();
>+    }
>+    audio_thread_.reset();
>+
>+    // Ensure that we don't quit the main thread loop immediately next
>+    // time Start() is called.
>+    ResetEvent(stop_event_.Get());
>+  }
>+}
>+
>+void CoreAudioBase::ThreadRun() {
>+  if (!core_audio_utility::IsMMCSSSupported()) {
>+    RTC_LOG(LS_ERROR) << "MMCSS is not supported";
>+    return;
>+  }
>+  RTC_DLOG(INFO) << "ThreadRun starts...";
>+  // TODO(henrika): difference between "Pro Audio" and "Audio"?
>+  ScopedMMCSSRegistration mmcss_registration(L"Pro Audio");
>+  ScopedCOMInitializer com_initializer(ScopedCOMInitializer::kMTA);
>+  RTC_DCHECK(mmcss_registration.Succeeded());
>+  RTC_DCHECK(com_initializer.Succeeded());
>+  RTC_DCHECK(stop_event_.IsValid());
>+  RTC_DCHECK(audio_samples_event_.IsValid());
>+
>+  bool streaming = true;
>+  bool error = false;
>+  HANDLE wait_array[] = {stop_event_.Get(), audio_samples_event_.Get()};
>+
>+  // The device frequency is the frequency generated by the hardware clock in
>+  // the audio device. The GetFrequency() method reports a constant frequency.
>+  UINT64 device_frequency = 0;
>+  if (audio_clock_.Get()) {
>+    RTC_DCHECK(IsOutput());
>+    _com_error result = audio_clock_->GetFrequency(&device_frequency);
>+    if ((error = result.Error()) != S_OK) {
>+      RTC_LOG(LS_ERROR) << "IAudioClock::GetFrequency failed: "
>+                        << core_audio_utility::ErrorToString(error);
>+    }
>+  }
>+
>+  // Keep streaming audio until the stop event or the stream-switch event
>+  // is signaled. An error event can also break the main thread loop.
>+  while (streaming && !error) {
>+    // Wait for a close-down event, stream-switch event or a new render event.
>+    DWORD wait_result = WaitForMultipleObjects(arraysize(wait_array),
>+                                               wait_array, false, INFINITE);
>+    switch (wait_result) {
>+      case WAIT_OBJECT_0 + 0:
>+        // |stop_event_| has been set.
>+        streaming = false;
>+        break;
>+      case WAIT_OBJECT_0 + 1:
>+        // |audio_samples_event_| has been set.
>+        error = !on_data_callback_(device_frequency);
>+        break;
>+      default:
>+        error = true;
>+        break;
>+    }
>+  }
>+
>+  if (streaming && error) {
>+    RTC_LOG(LS_ERROR) << "WASAPI streaming failed.";
>+    // Stop audio streaming since something has gone wrong in our main thread
>+    // loop. Note that, we are still in a "started" state, hence a Stop() call
>+    // is required to join the thread properly.
>+    audio_client_->Stop();
>+
>+    // TODO(henrika): notify clients that something has gone wrong and that
>+    // this stream should be destroyed instead of reused in the future.
>+  }
>+
>+  RTC_DLOG(INFO) << "...ThreadRun stops";
>+}
>+
>+}  // namespace webrtc_win
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/win/core_audio_base_win.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/win/core_audio_base_win.h
>new file mode 100644
>index 00000000000..b0886f99d34
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/win/core_audio_base_win.h
>@@ -0,0 +1,102 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_DEVICE_WIN_CORE_AUDIO_BASE_WIN_H_
>+#define MODULES_AUDIO_DEVICE_WIN_CORE_AUDIO_BASE_WIN_H_
>+
>+#include <functional>
>+#include <memory>
>+#include <string>
>+
>+#include "modules/audio_device/win/core_audio_utility_win.h"
>+#include "rtc_base/platform_thread.h"
>+#include "rtc_base/thread_checker.h"
>+
>+namespace webrtc {
>+
>+class AudioDeviceBuffer;
>+class FineAudioBuffer;
>+
>+namespace webrtc_win {
>+
>+// Serves as base class for CoreAudioInput and CoreAudioOutput and supports
>+// device handling and audio streaming where the direction (input or output)
>+// is set at constructions by the parent.
>+class CoreAudioBase {
>+ public:
>+  enum class Direction {
>+    kInput,
>+    kOutput,
>+  };
>+
>+  // Callback definition for notifications of new audio data. For input clients,
>+  // it means that "new audio data has now been captured", and for output
>+  // clients, "the output layer now needs new audio data".
>+  typedef std::function<bool(uint64_t device_frequency)> OnDataCallback;
>+
>+  explicit CoreAudioBase(Direction direction, OnDataCallback callback);
>+  ~CoreAudioBase();
>+
>+  std::string GetDeviceID(int index) const;
>+  int DeviceName(int index, std::string* name, std::string* guid) const;
>+
>+  bool Init();
>+  bool Start();
>+  bool Stop();
>+  bool IsVolumeControlAvailable(bool* available) const;
>+
>+  Direction direction() const { return direction_; }
>+
>+  void ThreadRun();
>+
>+  CoreAudioBase(const CoreAudioBase&) = delete;
>+  CoreAudioBase& operator=(const CoreAudioBase&) = delete;
>+
>+ protected:
>+  // Returns number of active devices given the specified |direction_|.
>+  int NumberOfActiveDevices() const;
>+
>+  // Returns total number of enumerated audio devices which is the sum of all
>+  // active devices plus two extra (one default and one default
>+  // communications). The value in |direction_| determines if capture or
>+  // render devices are counted.
>+  int NumberOfEnumeratedDevices() const;
>+
>+  bool IsInput() const;
>+  bool IsOutput() const;
>+  bool IsDefaultDevice(int index) const;
>+  bool IsDefaultCommunicationsDevice(int index) const;
>+  bool IsDefaultDevice(const std::string& device_id) const;
>+  bool IsDefaultCommunicationsDevice(const std::string& device_id) const;
>+  EDataFlow GetDataFlow() const;
>+
>+  rtc::ThreadChecker thread_checker_;
>+  rtc::ThreadChecker thread_checker_audio_;
>+  const Direction direction_;
>+  const OnDataCallback on_data_callback_;
>+  AudioDeviceBuffer* audio_device_buffer_ = nullptr;
>+  bool initialized_ = false;
>+  std::string device_id_;
>+  WAVEFORMATEXTENSIBLE format_ = {};
>+  uint32_t endpoint_buffer_size_frames_ = 0;
>+  Microsoft::WRL::ComPtr<IAudioClient> audio_client_;
>+  Microsoft::WRL::ComPtr<IAudioClock> audio_clock_;
>+  ScopedHandle audio_samples_event_;
>+  ScopedHandle stop_event_;
>+  std::unique_ptr<rtc::PlatformThread> audio_thread_;
>+
>+ private:
>+  void StopThread();
>+};
>+
>+}  // namespace webrtc_win
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_DEVICE_WIN_CORE_AUDIO_BASE_WIN_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/win/core_audio_input_win.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/win/core_audio_input_win.cc
>new file mode 100644
>index 00000000000..a389393f04d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/win/core_audio_input_win.cc
>@@ -0,0 +1,342 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_device/win/core_audio_input_win.h"
>+
>+#include "absl/memory/memory.h"
>+#include "modules/audio_device/audio_device_buffer.h"
>+#include "modules/audio_device/fine_audio_buffer.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/logging.h"
>+#include "rtc_base/numerics/safe_conversions.h"
>+
>+using Microsoft::WRL::ComPtr;
>+
>+namespace webrtc {
>+namespace webrtc_win {
>+
>+CoreAudioInput::CoreAudioInput()
>+    : CoreAudioBase(CoreAudioBase::Direction::kInput,
>+                    [this](uint64_t freq) { return OnDataCallback(freq); }) {
>+  RTC_DLOG(INFO) << __FUNCTION__;
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  thread_checker_audio_.DetachFromThread();
>+}
>+
>+CoreAudioInput::~CoreAudioInput() {
>+  RTC_DLOG(INFO) << __FUNCTION__;
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+}
>+
>+int CoreAudioInput::Init() {
>+  RTC_DLOG(INFO) << __FUNCTION__;
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  StopRecording();
>+  return 0;
>+}
>+
>+int CoreAudioInput::Terminate() {
>+  RTC_DLOG(INFO) << __FUNCTION__;
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  return 0;
>+}
>+
>+int CoreAudioInput::NumDevices() const {
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  return core_audio_utility::NumberOfActiveDevices(eCapture);
>+}
>+
>+int CoreAudioInput::SetDevice(int index) {
>+  RTC_DLOG(INFO) << __FUNCTION__ << ": " << index;
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  if (initialized_) {
>+    return -1;
>+  }
>+
>+  std::string device_id = GetDeviceID(index);
>+  RTC_DLOG(INFO) << "index=" << index << " => device_id: " << device_id;
>+  device_id_ = device_id;
>+
>+  return device_id_.empty() ? -1 : 0;
>+}
>+
>+int CoreAudioInput::SetDevice(AudioDeviceModule::WindowsDeviceType device) {
>+  RTC_DLOG(INFO) << __FUNCTION__ << ": " << device;
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  return SetDevice((device == AudioDeviceModule::kDefaultDevice) ? 0 : 1);
>+}
>+
>+int CoreAudioInput::DeviceName(int index,
>+                               std::string* name,
>+                               std::string* guid) {
>+  RTC_DLOG(INFO) << __FUNCTION__ << ": " << index;
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  RTC_DCHECK(name);
>+  return CoreAudioBase::DeviceName(index, name, guid);
>+}
>+
>+void CoreAudioInput::AttachAudioBuffer(AudioDeviceBuffer* audio_buffer) {
>+  RTC_DLOG(INFO) << __FUNCTION__;
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  audio_device_buffer_ = audio_buffer;
>+}
>+
>+bool CoreAudioInput::RecordingIsInitialized() const {
>+  RTC_DLOG(INFO) << __FUNCTION__ << ": " << initialized_;
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  return initialized_;
>+}
>+
>+int CoreAudioInput::InitRecording() {
>+  RTC_DLOG(INFO) << __FUNCTION__;
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  RTC_DCHECK(!initialized_);
>+  RTC_DCHECK(!Recording());
>+  RTC_DCHECK(!audio_client_.Get());
>+  RTC_DCHECK(!audio_capture_client_.Get());
>+
>+  // Create an IAudioClient and store the valid interface pointer in
>+  // |audio_client_|. The base class will use optimal input parameters and do
>+  // an event driven shared mode initialization. The utilized format will be
>+  // stored in |format_| and can be used for configuration and allocation of
>+  // audio buffers.
>+  if (!CoreAudioBase::Init()) {
>+    return -1;
>+  }
>+  RTC_DCHECK(audio_client_.Get());
>+
>+  // Configure the recording side of the audio device buffer using |format_|
>+  // after a trivial sanity check of the format structure.
>+  RTC_DCHECK(audio_device_buffer_);
>+  WAVEFORMATEX* format = &format_.Format;
>+  RTC_DCHECK_EQ(format->wFormatTag, WAVE_FORMAT_EXTENSIBLE);
>+  audio_device_buffer_->SetRecordingSampleRate(format->nSamplesPerSec);
>+  audio_device_buffer_->SetRecordingChannels(format->nChannels);
>+
>+  // Create a modified audio buffer class which allows us to supply any number
>+  // of samples (and not only multiple of 10ms) to match the optimal buffer
>+  // size per callback used by Core Audio.
>+  // TODO(henrika): can we share one FineAudioBuffer?
>+  fine_audio_buffer_ = absl::make_unique<FineAudioBuffer>(audio_device_buffer_);
>+
>+  // Create an IAudioCaptureClient for an initialized IAudioClient.
>+  // The IAudioCaptureClient interface enables a client to read input data from
>+  // a capture endpoint buffer.
>+  ComPtr<IAudioCaptureClient> audio_capture_client =
>+      core_audio_utility::CreateCaptureClient(audio_client_.Get());
>+  if (!audio_capture_client.Get()) {
>+    return -1;
>+  }
>+
>+  // Query performance frequency.
>+  LARGE_INTEGER ticks_per_sec = {};
>+  qpc_to_100ns_.reset();
>+  if (::QueryPerformanceFrequency(&ticks_per_sec)) {
>+    double qpc_ticks_per_second =
>+        rtc::dchecked_cast<double>(ticks_per_sec.QuadPart);
>+    qpc_to_100ns_ = 10000000.0 / qpc_ticks_per_second;
>+  }
>+
>+  // Store valid COM interfaces. Note that, |audio_client_| has already been
>+  // set in CoreAudioBase::Init().
>+  audio_capture_client_ = audio_capture_client;
>+
>+  initialized_ = true;
>+  return 0;
>+}
>+
>+int CoreAudioInput::StartRecording() {
>+  RTC_DLOG(INFO) << __FUNCTION__;
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  RTC_DCHECK(!Recording());
>+  if (!initialized_) {
>+    RTC_DLOG(LS_WARNING)
>+        << "Recording can not start since InitRecording must succeed first";
>+    return 0;
>+  }
>+  if (fine_audio_buffer_) {
>+    fine_audio_buffer_->ResetRecord();
>+  }
>+
>+  if (!Start()) {
>+    return -1;
>+  }
>+
>+  return 0;
>+}
>+
>+int CoreAudioInput::StopRecording() {
>+  RTC_DLOG(INFO) << __FUNCTION__;
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  if (!initialized_) {
>+    return 0;
>+  }
>+
>+  // Release resources allocated in InitRecording() and then return if this
>+  // method is called without any active input audio.
>+  if (!Recording()) {
>+    RTC_DLOG(WARNING) << "No input stream is active";
>+    audio_client_.Reset();
>+    audio_capture_client_.Reset();
>+    initialized_ = false;
>+    return 0;
>+  }
>+
>+  if (!Stop()) {
>+    RTC_LOG(LS_ERROR) << "StopRecording failed";
>+    return -1;
>+  }
>+
>+  // TODO(henrika): if we want to support Init(), Start(), Stop(), Init(),
>+  // Start(), Stop() without close in between, these lines are needed.
>+  // Not supported on mobile ADMs, hence we can probably live without it.
>+  // audio_client_.Reset();
>+  // audio_capture_client_.Reset();
>+  // audio_device_buffer_->NativeAudioRecordingInterrupted();
>+  thread_checker_audio_.DetachFromThread();
>+  qpc_to_100ns_.reset();
>+  initialized_ = false;
>+  return 0;
>+}
>+
>+bool CoreAudioInput::Recording() {
>+  RTC_DLOG(INFO) << __FUNCTION__ << ": " << (audio_thread_ != nullptr);
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  return audio_thread_ != nullptr;
>+}
>+
>+// TODO(henrika): finalize support of audio session volume control. As is, we
>+// are not compatible with the old ADM implementation since it allows accessing
>+// the volume control with any active audio output stream.
>+int CoreAudioInput::VolumeIsAvailable(bool* available) {
>+  RTC_DLOG(INFO) << __FUNCTION__;
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  return IsVolumeControlAvailable(available) ? 0 : -1;
>+}
>+
>+bool CoreAudioInput::OnDataCallback(uint64_t device_frequency) {
>+  RTC_DCHECK_RUN_ON(&thread_checker_audio_);
>+  UINT32 num_frames_in_next_packet = 0;
>+  _com_error error =
>+      audio_capture_client_->GetNextPacketSize(&num_frames_in_next_packet);
>+  if (error.Error() != S_OK) {
>+    RTC_LOG(LS_ERROR) << "IAudioCaptureClient::GetNextPacketSize failed: "
>+                      << core_audio_utility::ErrorToString(error);
>+    return false;
>+  }
>+
>+  // Drain the WASAPI capture buffer fully if audio has been recorded.
>+  while (num_frames_in_next_packet > 0) {
>+    uint8_t* audio_data;
>+    UINT32 num_frames_to_read = 0;
>+    DWORD flags = 0;
>+    UINT64 device_position_frames = 0;
>+    UINT64 capture_time_100ns = 0;
>+    error = audio_capture_client_->GetBuffer(&audio_data, &num_frames_to_read,
>+                                             &flags, &device_position_frames,
>+                                             &capture_time_100ns);
>+    if (error.Error() == AUDCLNT_S_BUFFER_EMPTY) {
>+      // The call succeeded but no capture data is available to be read.
>+      // Return and start waiting for new capture event
>+      RTC_DCHECK_EQ(num_frames_to_read, 0u);
>+      return true;
>+    }
>+    if (error.Error() != S_OK) {
>+      RTC_LOG(LS_ERROR) << "IAudioCaptureClient::GetBuffer failed: "
>+                        << core_audio_utility::ErrorToString(error);
>+      return false;
>+    }
>+
>+    // TODO(henrika): only update the latency estimate N times per second to
>+    // save resources.
>+    // TODO(henrika): note that FineAudioBuffer adds latency as well.
>+    auto opt_record_delay_ms = EstimateLatencyMillis(capture_time_100ns);
>+
>+    // The data in the packet is not correlated with the previous packet's
>+    // device position; possibly due to a stream state transition or timing
>+    // glitch. The behavior of the AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY flag
>+    // is undefined on the application's first call to GetBuffer after Start.
>+    if (device_position_frames != 0 &&
>+        flags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY) {
>+      RTC_DLOG(LS_WARNING) << "AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY";
>+    }
>+    // The time at which the device's stream position was recorded is uncertain.
>+    // Thus, the client might be unable to accurately set a time stamp for the
>+    // current data packet.
>+    if (flags & AUDCLNT_BUFFERFLAGS_TIMESTAMP_ERROR) {
>+      RTC_DLOG(LS_WARNING) << "AUDCLNT_BUFFERFLAGS_TIMESTAMP_ERROR";
>+    }
>+
>+    // Treat all of the data in the packet as silence and ignore the actual
>+    // data values when AUDCLNT_BUFFERFLAGS_SILENT is set.
>+    if (flags & AUDCLNT_BUFFERFLAGS_SILENT) {
>+      rtc::ExplicitZeroMemory(audio_data,
>+                              format_.Format.nBlockAlign * num_frames_to_read);
>+      RTC_DLOG(LS_WARNING) << "Captured audio is replaced by silence";
>+    } else {
>+      // Copy recorded audio in |audio_data| to the WebRTC sink using the
>+      // FineAudioBuffer object.
>+      // TODO(henrika): fix delay estimation.
>+      int record_delay_ms = 0;
>+      if (opt_record_delay_ms) {
>+        record_delay_ms = *opt_record_delay_ms;
>+        // RTC_DLOG(INFO) << "record_delay_ms: " << record_delay_ms;
>+      }
>+      fine_audio_buffer_->DeliverRecordedData(
>+          rtc::MakeArrayView(reinterpret_cast<const int16_t*>(audio_data),
>+                             format_.Format.nChannels * num_frames_to_read),
>+
>+          record_delay_ms);
>+    }
>+
>+    error = audio_capture_client_->ReleaseBuffer(num_frames_to_read);
>+    if (error.Error() != S_OK) {
>+      RTC_LOG(LS_ERROR) << "IAudioCaptureClient::ReleaseBuffer failed: "
>+                        << core_audio_utility::ErrorToString(error);
>+      return false;
>+    }
>+
>+    error =
>+        audio_capture_client_->GetNextPacketSize(&num_frames_in_next_packet);
>+    if (error.Error() != S_OK) {
>+      RTC_LOG(LS_ERROR) << "IAudioCaptureClient::GetNextPacketSize failed: "
>+                        << core_audio_utility::ErrorToString(error);
>+      return false;
>+    }
>+  }
>+
>+  return true;
>+}
>+
>+absl::optional<int> CoreAudioInput::EstimateLatencyMillis(
>+    uint64_t capture_time_100ns) {
>+  if (!qpc_to_100ns_) {
>+    return absl::nullopt;
>+  }
>+  // Input parameter |capture_time_100ns| contains the performance counter at
>+  // the time that the audio endpoint device recorded the device position of
>+  // the first audio frame in the data packet converted into 100ns units.
>+  // We derive a delay estimate by:
>+  // - sampling the current performance counter (qpc_now_raw),
>+  // - converting it into 100ns time units (now_time_100ns), and
>+  // - subtracting |capture_time_100ns| from now_time_100ns.
>+  LARGE_INTEGER perf_counter_now = {};
>+  if (!::QueryPerformanceCounter(&perf_counter_now)) {
>+    return absl::nullopt;
>+  }
>+  uint64_t qpc_now_raw = perf_counter_now.QuadPart;
>+  uint64_t now_time_100ns = qpc_now_raw * (*qpc_to_100ns_);
>+  webrtc::TimeDelta delay_us =
>+      webrtc::TimeDelta::us(0.1 * (now_time_100ns - capture_time_100ns) + 0.5);
>+  return delay_us.ms();
>+}
>+
>+}  // namespace webrtc_win
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/win/core_audio_input_win.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/win/core_audio_input_win.h
>new file mode 100644
>index 00000000000..0dd2e3730a3
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/win/core_audio_input_win.h
>@@ -0,0 +1,67 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_DEVICE_WIN_CORE_AUDIO_INPUT_WIN_H_
>+#define MODULES_AUDIO_DEVICE_WIN_CORE_AUDIO_INPUT_WIN_H_
>+
>+#include <memory>
>+#include <string>
>+
>+#include "absl/types/optional.h"
>+#include "modules/audio_device/win/audio_device_module_win.h"
>+#include "modules/audio_device/win/core_audio_base_win.h"
>+
>+namespace webrtc {
>+
>+class AudioDeviceBuffer;
>+class FineAudioBuffer;
>+
>+namespace webrtc_win {
>+
>+// Windows specific AudioInput implementation using a CoreAudioBase class where
>+// an input direction is set at construction. Supports capture device handling
>+// and streaming of captured audio to a WebRTC client.
>+class CoreAudioInput final : public CoreAudioBase, public AudioInput {
>+ public:
>+  CoreAudioInput();
>+  ~CoreAudioInput() override;
>+
>+  // AudioInput implementation.
>+  int Init() override;
>+  int Terminate() override;
>+  int NumDevices() const override;
>+  int SetDevice(int index) override;
>+  int SetDevice(AudioDeviceModule::WindowsDeviceType device) override;
>+  int DeviceName(int index, std::string* name, std::string* guid) override;
>+  void AttachAudioBuffer(AudioDeviceBuffer* audio_buffer) override;
>+  bool RecordingIsInitialized() const override;
>+  int InitRecording() override;
>+  int StartRecording() override;
>+  int StopRecording() override;
>+  bool Recording() override;
>+  int VolumeIsAvailable(bool* available) override;
>+
>+  CoreAudioInput(const CoreAudioInput&) = delete;
>+  CoreAudioInput& operator=(const CoreAudioInput&) = delete;
>+
>+ private:
>+  bool OnDataCallback(uint64_t device_frequency);
>+  absl::optional<int> EstimateLatencyMillis(uint64_t capture_time_100ns);
>+
>+  std::unique_ptr<FineAudioBuffer> fine_audio_buffer_;
>+  Microsoft::WRL::ComPtr<IAudioCaptureClient> audio_capture_client_;
>+  absl::optional<double> qpc_to_100ns_;
>+};
>+
>+}  // namespace webrtc_win
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_DEVICE_WIN_CORE_AUDIO_INPUT_WIN_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/win/core_audio_output_win.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/win/core_audio_output_win.cc
>new file mode 100644
>index 00000000000..a391109dba5
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/win/core_audio_output_win.cc
>@@ -0,0 +1,306 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_device/win/core_audio_output_win.h"
>+
>+#include "absl/memory/memory.h"
>+#include "modules/audio_device/audio_device_buffer.h"
>+#include "modules/audio_device/fine_audio_buffer.h"
>+#include "rtc_base/bind.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/logging.h"
>+
>+using Microsoft::WRL::ComPtr;
>+
>+namespace webrtc {
>+namespace webrtc_win {
>+
>+CoreAudioOutput::CoreAudioOutput()
>+    : CoreAudioBase(CoreAudioBase::Direction::kOutput,
>+                    [this](uint64_t freq) { return OnDataCallback(freq); }) {
>+  RTC_DLOG(INFO) << __FUNCTION__;
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  thread_checker_audio_.DetachFromThread();
>+}
>+
>+CoreAudioOutput::~CoreAudioOutput() {
>+  RTC_DLOG(INFO) << __FUNCTION__;
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  Terminate();
>+}
>+
>+int CoreAudioOutput::Init() {
>+  RTC_DLOG(INFO) << __FUNCTION__;
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  return 0;
>+}
>+
>+int CoreAudioOutput::Terminate() {
>+  RTC_DLOG(INFO) << __FUNCTION__;
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  StopPlayout();
>+  return 0;
>+}
>+
>+int CoreAudioOutput::NumDevices() const {
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  return core_audio_utility::NumberOfActiveDevices(eRender);
>+}
>+
>+int CoreAudioOutput::SetDevice(int index) {
>+  RTC_DLOG(INFO) << __FUNCTION__ << ": " << index;
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  if (initialized_) {
>+    return -1;
>+  }
>+
>+  std::string device_id = GetDeviceID(index);
>+  RTC_DLOG(INFO) << "index=" << index << " => device_id: " << device_id;
>+  device_id_ = device_id;
>+
>+  return device_id_.empty() ? -1 : 0;
>+}
>+
>+int CoreAudioOutput::SetDevice(AudioDeviceModule::WindowsDeviceType device) {
>+  RTC_DLOG(INFO) << __FUNCTION__ << ": " << device;
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  return SetDevice((device == AudioDeviceModule::kDefaultDevice) ? 0 : 1);
>+}
>+
>+int CoreAudioOutput::DeviceName(int index,
>+                                std::string* name,
>+                                std::string* guid) {
>+  RTC_DLOG(INFO) << __FUNCTION__ << ": " << index;
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  RTC_DCHECK(name);
>+  return CoreAudioBase::DeviceName(index, name, guid);
>+}
>+
>+void CoreAudioOutput::AttachAudioBuffer(AudioDeviceBuffer* audio_buffer) {
>+  RTC_DLOG(INFO) << __FUNCTION__;
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  audio_device_buffer_ = audio_buffer;
>+}
>+
>+bool CoreAudioOutput::PlayoutIsInitialized() const {
>+  RTC_DLOG(INFO) << __FUNCTION__;
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  return initialized_;
>+}
>+
>+int CoreAudioOutput::InitPlayout() {
>+  RTC_DLOG(INFO) << __FUNCTION__;
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  RTC_DCHECK(!initialized_);
>+  RTC_DCHECK(!Playing());
>+  RTC_DCHECK(!audio_client_.Get());
>+  RTC_DCHECK(!audio_render_client_.Get());
>+
>+  // Create an IAudioClient client and store the valid interface pointer in
>+  // |audio_client_|. The base class will use optimal output parameters and do
>+  // an event driven shared mode initialization. The utilized format will be
>+  // stored in |format_| and can be used for configuration and allocation of
>+  // audio buffers.
>+  if (!CoreAudioBase::Init()) {
>+    return -1;
>+  }
>+  RTC_DCHECK(audio_client_.Get());
>+
>+  // Configure the playout side of the audio device buffer using |format_|
>+  // after a trivial sanity check of the format structure.
>+  RTC_DCHECK(audio_device_buffer_);
>+  WAVEFORMATEX* format = &format_.Format;
>+  RTC_DCHECK_EQ(format->wFormatTag, WAVE_FORMAT_EXTENSIBLE);
>+  audio_device_buffer_->SetPlayoutSampleRate(format->nSamplesPerSec);
>+  audio_device_buffer_->SetPlayoutChannels(format->nChannels);
>+
>+  // Create a modified audio buffer class which allows us to ask for any number
>+  // of samples (and not only multiple of 10ms) to match the optimal
>+  // buffer size per callback used by Core Audio.
>+  // TODO(henrika): can we use a shared buffer instead?
>+  fine_audio_buffer_ = absl::make_unique<FineAudioBuffer>(audio_device_buffer_);
>+
>+  // Create an IAudioRenderClient for an initialized IAudioClient.
>+  // The IAudioRenderClient interface enables us to write output data to
>+  // a rendering endpoint buffer.
>+  ComPtr<IAudioRenderClient> audio_render_client =
>+      core_audio_utility::CreateRenderClient(audio_client_.Get());
>+  if (!audio_render_client.Get())
>+    return -1;
>+
>+  ComPtr<IAudioClock> audio_clock =
>+      core_audio_utility::CreateAudioClock(audio_client_.Get());
>+  if (!audio_clock.Get())
>+    return -1;
>+
>+  // Store valid COM interfaces. Note that, |audio_client_| has already been
>+  // set in CoreAudioBase::Init().
>+  audio_render_client_ = audio_render_client;
>+  audio_clock_ = audio_clock;
>+
>+  initialized_ = true;
>+  return 0;
>+}
>+
>+int CoreAudioOutput::StartPlayout() {
>+  RTC_DLOG(INFO) << __FUNCTION__;
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  RTC_DCHECK(!Playing());
>+  if (!initialized_) {
>+    RTC_DLOG(LS_WARNING)
>+        << "Playout can not start since InitPlayout must succeed first";
>+    return 0;
>+  }
>+  if (fine_audio_buffer_) {
>+    fine_audio_buffer_->ResetPlayout();
>+  }
>+
>+  if (!core_audio_utility::FillRenderEndpointBufferWithSilence(
>+          audio_client_.Get(), audio_render_client_.Get())) {
>+    RTC_LOG(LS_WARNING) << "Failed to prepare output endpoint with silence";
>+  }
>+
>+  num_frames_written_ = endpoint_buffer_size_frames_;
>+
>+  if (!Start()) {
>+    return -1;
>+  }
>+
>+  return 0;
>+}
>+
>+int CoreAudioOutput::StopPlayout() {
>+  RTC_DLOG(INFO) << __FUNCTION__;
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  if (!initialized_) {
>+    return 0;
>+  }
>+
>+  // Release resources allocated in InitPlayout() and then return if this
>+  // method is called without any active output audio.
>+  if (!Playing()) {
>+    RTC_DLOG(WARNING) << "No output stream is active";
>+    audio_client_.Reset();
>+    audio_render_client_.Reset();
>+    initialized_ = false;
>+    return 0;
>+  }
>+
>+  if (!Stop()) {
>+    RTC_LOG(LS_ERROR) << "StopPlayout failed";
>+    return -1;
>+  }
>+
>+  thread_checker_audio_.DetachFromThread();
>+  initialized_ = false;
>+  return 0;
>+}
>+
>+bool CoreAudioOutput::Playing() {
>+  RTC_DLOG(INFO) << __FUNCTION__;
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  return audio_thread_ != nullptr;
>+}
>+
>+// TODO(henrika): finalize support of audio session volume control. As is, we
>+// are not compatible with the old ADM implementation since it allows accessing
>+// the volume control with any active audio output stream.
>+int CoreAudioOutput::VolumeIsAvailable(bool* available) {
>+  RTC_DLOG(INFO) << __FUNCTION__;
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  return IsVolumeControlAvailable(available) ? 0 : -1;
>+}
>+
>+bool CoreAudioOutput::OnDataCallback(uint64_t device_frequency) {
>+  RTC_DCHECK_RUN_ON(&thread_checker_audio_);
>+  // Get the padding value which indicates the amount of valid unread data that
>+  // the endpoint buffer currently contains.
>+  UINT32 num_unread_frames = 0;
>+  _com_error error = audio_client_->GetCurrentPadding(&num_unread_frames);
>+  if (error.Error() != S_OK) {
>+    RTC_LOG(LS_ERROR) << "IAudioClient::GetCurrentPadding failed: "
>+                      << core_audio_utility::ErrorToString(error);
>+    return false;
>+  }
>+
>+  // Contains how much new data we can write to the buffer without the risk of
>+  // overwriting previously written data that the audio engine has not yet read
>+  // from the buffer. I.e., it is the maximum buffer size we can request when
>+  // calling IAudioRenderClient::GetBuffer().
>+  UINT32 num_requested_frames =
>+      endpoint_buffer_size_frames_ - num_unread_frames;
>+
>+  // Request all available space in the rendering endpoint buffer into which the
>+  // client can later write an audio packet.
>+  uint8_t* audio_data;
>+  error = audio_render_client_->GetBuffer(num_requested_frames, &audio_data);
>+  if (error.Error() != S_OK) {
>+    RTC_LOG(LS_ERROR) << "IAudioRenderClient::GetBuffer failed: "
>+                      << core_audio_utility::ErrorToString(error);
>+    return false;
>+  }
>+
>+  // TODO(henrika): only update the latency estimate N times per second to
>+  // save resources.
>+  // TODO(henrika): note that FineAudioBuffer adds latency as well.
>+  int playout_delay_ms = EstimateOutputLatencyMillis(device_frequency);
>+  // RTC_DLOG(INFO) << "playout_delay_ms: " << playout_delay_ms;
>+
>+  // Get audio data from WebRTC and write it to the allocated buffer in
>+  // |audio_data|.
>+  fine_audio_buffer_->GetPlayoutData(
>+      rtc::MakeArrayView(reinterpret_cast<int16_t*>(audio_data),
>+                         num_requested_frames * format_.Format.nChannels),
>+      playout_delay_ms);
>+
>+  // Release the buffer space acquired in IAudioRenderClient::GetBuffer.
>+  error = audio_render_client_->ReleaseBuffer(num_requested_frames, 0);
>+  if (error.Error() != S_OK) {
>+    RTC_LOG(LS_ERROR) << "IAudioRenderClient::ReleaseBuffer failed: "
>+                      << core_audio_utility::ErrorToString(error);
>+    return false;
>+  }
>+
>+  num_frames_written_ += num_requested_frames;
>+
>+  return true;
>+}
>+
>+// TODO(henrika): IAudioClock2::GetDevicePosition could perhaps be used here
>+// instead. Tried it once, but it crashed for capture devices.
>+int CoreAudioOutput::EstimateOutputLatencyMillis(uint64_t device_frequency) {
>+  UINT64 position = 0;
>+  UINT64 qpc_position = 0;
>+  int delay_ms = 0;
>+  // Get the device position through output parameter |position|. This is the
>+  // stream position of the sample that is currently playing through the
>+  // speakers.
>+  _com_error error = audio_clock_->GetPosition(&position, &qpc_position);
>+  if (error.Error() == S_OK) {
>+    // Number of frames already played out through the speaker.
>+    const uint64_t num_played_out_frames =
>+        format_.Format.nSamplesPerSec * position / device_frequency;
>+
>+    // Number of frames that have been written to the buffer but not yet
>+    // played out corresponding to the estimated latency measured in number
>+    // of audio frames.
>+    const uint64_t delay_frames = num_frames_written_ - num_played_out_frames;
>+
>+    // Convert latency in number of frames into milliseconds.
>+    webrtc::TimeDelta delay = webrtc::TimeDelta::us(
>+        delay_frames * kNumMicrosecsPerSec / format_.Format.nSamplesPerSec);
>+    delay_ms = delay.ms();
>+  }
>+  return delay_ms;
>+}
>+
>+}  // namespace webrtc_win
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/win/core_audio_output_win.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/win/core_audio_output_win.h
>new file mode 100644
>index 00000000000..c947f3dd81a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/win/core_audio_output_win.h
>@@ -0,0 +1,66 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_DEVICE_WIN_CORE_AUDIO_OUTPUT_WIN_H_
>+#define MODULES_AUDIO_DEVICE_WIN_CORE_AUDIO_OUTPUT_WIN_H_
>+
>+#include <memory>
>+#include <string>
>+
>+#include "modules/audio_device/win/audio_device_module_win.h"
>+#include "modules/audio_device/win/core_audio_base_win.h"
>+
>+namespace webrtc {
>+
>+class AudioDeviceBuffer;
>+class FineAudioBuffer;
>+
>+namespace webrtc_win {
>+
>+// Windows specific AudioOutput implementation using a CoreAudioBase class where
>+// an output direction is set at construction. Supports render device handling
>+// and streaming of decoded audio from a WebRTC client to the native audio
>+// layer.
>+class CoreAudioOutput final : public CoreAudioBase, public AudioOutput {
>+ public:
>+  CoreAudioOutput();
>+  ~CoreAudioOutput() override;
>+
>+  // AudioOutput implementation.
>+  int Init() override;
>+  int Terminate() override;
>+  int NumDevices() const override;
>+  int SetDevice(int index) override;
>+  int SetDevice(AudioDeviceModule::WindowsDeviceType device) override;
>+  int DeviceName(int index, std::string* name, std::string* guid) override;
>+  void AttachAudioBuffer(AudioDeviceBuffer* audio_buffer) override;
>+  bool PlayoutIsInitialized() const override;
>+  int InitPlayout() override;
>+  int StartPlayout() override;
>+  int StopPlayout() override;
>+  bool Playing() override;
>+  int VolumeIsAvailable(bool* available) override;
>+
>+  CoreAudioOutput(const CoreAudioOutput&) = delete;
>+  CoreAudioOutput& operator=(const CoreAudioOutput&) = delete;
>+
>+ private:
>+  bool OnDataCallback(uint64_t device_frequency);
>+  int EstimateOutputLatencyMillis(uint64_t device_frequency);
>+
>+  std::unique_ptr<FineAudioBuffer> fine_audio_buffer_;
>+  Microsoft::WRL::ComPtr<IAudioRenderClient> audio_render_client_;
>+  uint64_t num_frames_written_ = 0;
>+};
>+
>+}  // namespace webrtc_win
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_DEVICE_WIN_CORE_AUDIO_OUTPUT_WIN_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/win/core_audio_utility_win.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/win/core_audio_utility_win.cc
>new file mode 100644
>index 00000000000..5dc45a8a989
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/win/core_audio_utility_win.cc
>@@ -0,0 +1,960 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_device/win/core_audio_utility_win.h"
>+
>+#include <Functiondiscoverykeys_devpkey.h>
>+#include <atlbase.h>
>+#include <stdio.h>
>+#include <tchar.h>
>+
>+#include <iomanip>
>+#include <string>
>+#include <utility>
>+
>+#include "rtc_base/arraysize.h"
>+#include "rtc_base/logging.h"
>+#include "rtc_base/platform_thread_types.h"
>+#include "rtc_base/strings/string_builder.h"
>+#include "rtc_base/stringutils.h"
>+
>+using ATL::CComHeapPtr;
>+using Microsoft::WRL::ComPtr;
>+using webrtc::AudioDeviceName;
>+using webrtc::AudioParameters;
>+
>+namespace webrtc {
>+namespace webrtc_win {
>+namespace {
>+
>+using core_audio_utility::ErrorToString;
>+
>+bool LoadAudiosesDll() {
>+  static const wchar_t* const kAudiosesDLL =
>+      L"%WINDIR%\\system32\\audioses.dll";
>+  wchar_t path[MAX_PATH] = {0};
>+  ExpandEnvironmentStringsW(kAudiosesDLL, path, arraysize(path));
>+  RTC_DLOG(INFO) << rtc::ToUtf8(path);
>+  return (LoadLibraryExW(path, nullptr, LOAD_WITH_ALTERED_SEARCH_PATH) !=
>+          nullptr);
>+}
>+
>+bool LoadAvrtDll() {
>+  static const wchar_t* const kAvrtDLL = L"%WINDIR%\\system32\\Avrt.dll";
>+  wchar_t path[MAX_PATH] = {0};
>+  ExpandEnvironmentStringsW(kAvrtDLL, path, arraysize(path));
>+  RTC_DLOG(INFO) << rtc::ToUtf8(path);
>+  return (LoadLibraryExW(path, nullptr, LOAD_WITH_ALTERED_SEARCH_PATH) !=
>+          nullptr);
>+}
>+
>+ComPtr<IMMDeviceEnumerator> CreateDeviceEnumeratorInternal(
>+    bool allow_reinitialize) {
>+  ComPtr<IMMDeviceEnumerator> device_enumerator;
>+  _com_error error =
>+      ::CoCreateInstance(__uuidof(MMDeviceEnumerator), nullptr, CLSCTX_ALL,
>+                         IID_PPV_ARGS(&device_enumerator));
>+  if (error.Error() != S_OK) {
>+    RTC_LOG(LS_ERROR) << "CoCreateInstance failed: " << ErrorToString(error);
>+  }
>+
>+  if (error.Error() == CO_E_NOTINITIALIZED && allow_reinitialize) {
>+    RTC_LOG(LS_ERROR) << "CoCreateInstance failed with CO_E_NOTINITIALIZED";
>+    // We have seen crashes which indicates that this method can in fact
>+    // fail with CO_E_NOTINITIALIZED in combination with certain 3rd party
>+    // modules. Calling CoInitializeEx() is an attempt to resolve the reported
>+    // issues. See http://crbug.com/378465 for details.
>+    error = CoInitializeEx(nullptr, COINIT_MULTITHREADED);
>+    if (error.Error() != S_OK) {
>+      error = ::CoCreateInstance(__uuidof(MMDeviceEnumerator), nullptr,
>+                                 CLSCTX_ALL, IID_PPV_ARGS(&device_enumerator));
>+      if (error.Error() != S_OK) {
>+        RTC_LOG(LS_ERROR) << "CoCreateInstance failed: "
>+                          << ErrorToString(error);
>+      }
>+    }
>+  }
>+  return device_enumerator;
>+}
>+
>+bool IsSupportedInternal() {
>+  // The Core Audio APIs are implemented in the user-mode system components
>+  // Audioses.dll and Mmdevapi.dll. Dependency Walker shows that it is
>+  // enough to verify possibility to load the Audioses DLL since it depends
>+  // on Mmdevapi.dll. See http://crbug.com/166397 why this extra step is
>+  // required to guarantee Core Audio support.
>+  if (!LoadAudiosesDll())
>+    return false;
>+
>+  // Being able to load the Audioses.dll does not seem to be sufficient for
>+  // all devices to guarantee Core Audio support. To be 100%, we also verify
>+  // that it is possible to a create the IMMDeviceEnumerator interface. If
>+  // this works as well we should be home free.
>+  ComPtr<IMMDeviceEnumerator> device_enumerator =
>+      CreateDeviceEnumeratorInternal(false);
>+  if (!device_enumerator) {
>+    RTC_LOG(LS_ERROR)
>+        << "Failed to create Core Audio device enumerator on thread with ID "
>+        << rtc::CurrentThreadId();
>+    return false;
>+  }
>+
>+  return true;
>+}
>+
>+bool IsDeviceActive(IMMDevice* device) {
>+  DWORD state = DEVICE_STATE_DISABLED;
>+  return SUCCEEDED(device->GetState(&state)) && (state & DEVICE_STATE_ACTIVE);
>+}
>+
>+// Retrieve an audio device specified by |device_id| or a default device
>+// specified by data-flow direction and role if |device_id| is default.
>+ComPtr<IMMDevice> CreateDeviceInternal(const std::string& device_id,
>+                                       EDataFlow data_flow,
>+                                       ERole role) {
>+  RTC_DLOG(INFO) << "CreateDeviceInternal: " << role;
>+  ComPtr<IMMDevice> audio_endpoint_device;
>+
>+  // Create the IMMDeviceEnumerator interface.
>+  ComPtr<IMMDeviceEnumerator> device_enum(CreateDeviceEnumeratorInternal(true));
>+  if (!device_enum.Get())
>+    return audio_endpoint_device;
>+
>+  _com_error error(S_FALSE);
>+  if (device_id == AudioDeviceName::kDefaultDeviceId) {
>+    error = device_enum->GetDefaultAudioEndpoint(
>+        data_flow, role, audio_endpoint_device.GetAddressOf());
>+    if (error.Error() != S_OK) {
>+      RTC_LOG(LS_ERROR)
>+          << "IMMDeviceEnumerator::GetDefaultAudioEndpoint failed: "
>+          << ErrorToString(error);
>+    }
>+  } else {
>+    error = device_enum->GetDevice(rtc::ToUtf16(device_id).c_str(),
>+                                   audio_endpoint_device.GetAddressOf());
>+    if (error.Error() != S_OK) {
>+      RTC_LOG(LS_ERROR) << "IMMDeviceEnumerator::GetDevice failed: "
>+                        << ErrorToString(error);
>+    }
>+  }
>+
>+  // Verify that the audio endpoint device is active, i.e., that the audio
>+  // adapter that connects to the endpoint device is present and enabled.
>+  if (SUCCEEDED(error.Error()) &&
>+      !IsDeviceActive(audio_endpoint_device.Get())) {
>+    RTC_LOG(LS_WARNING) << "Selected endpoint device is not active";
>+    audio_endpoint_device.Reset();
>+  }
>+
>+  return audio_endpoint_device;
>+}
>+
>+std::string GetDeviceIdInternal(IMMDevice* device) {
>+  // Retrieve unique name of endpoint device.
>+  // Example: "{0.0.1.00000000}.{8db6020f-18e3-4f25-b6f5-7726c9122574}".
>+  CComHeapPtr<WCHAR> device_id;
>+  if (SUCCEEDED(device->GetId(&device_id))) {
>+    return rtc::ToUtf8(device_id, wcslen(device_id));
>+  } else {
>+    return std::string();
>+  }
>+}
>+
>+std::string GetDeviceFriendlyNameInternal(IMMDevice* device) {
>+  // Retrieve user-friendly name of endpoint device.
>+  // Example: "Microphone (Realtek High Definition Audio)".
>+  ComPtr<IPropertyStore> properties;
>+  HRESULT hr = device->OpenPropertyStore(STGM_READ, properties.GetAddressOf());
>+  if (FAILED(hr))
>+    return std::string();
>+
>+  ScopedPropVariant friendly_name_pv;
>+  hr = properties->GetValue(PKEY_Device_FriendlyName,
>+                            friendly_name_pv.Receive());
>+  if (FAILED(hr))
>+    return std::string();
>+
>+  if (friendly_name_pv.get().vt == VT_LPWSTR &&
>+      friendly_name_pv.get().pwszVal) {
>+    return rtc::ToUtf8(friendly_name_pv.get().pwszVal,
>+                       wcslen(friendly_name_pv.get().pwszVal));
>+  } else {
>+    return std::string();
>+  }
>+}
>+
>+ComPtr<IAudioSessionManager2> CreateSessionManager2Internal(
>+    IMMDevice* audio_device) {
>+  if (!audio_device)
>+    return ComPtr<IAudioSessionManager2>();
>+
>+  ComPtr<IAudioSessionManager2> audio_session_manager;
>+  _com_error error =
>+      audio_device->Activate(__uuidof(IAudioSessionManager2), CLSCTX_ALL,
>+                             nullptr, &audio_session_manager);
>+  if (error.Error() != S_OK) {
>+    RTC_LOG(LS_ERROR) << "IMMDevice::Activate(IAudioSessionManager2) failed: "
>+                      << ErrorToString(error);
>+  }
>+  return audio_session_manager;
>+}
>+
>+ComPtr<IAudioSessionEnumerator> CreateSessionEnumeratorInternal(
>+    IMMDevice* audio_device) {
>+  if (!audio_device) {
>+    return ComPtr<IAudioSessionEnumerator>();
>+  }
>+
>+  ComPtr<IAudioSessionEnumerator> audio_session_enumerator;
>+  ComPtr<IAudioSessionManager2> audio_session_manager =
>+      CreateSessionManager2Internal(audio_device);
>+  if (!audio_session_manager.Get()) {
>+    return audio_session_enumerator;
>+  }
>+  _com_error error =
>+      audio_session_manager->GetSessionEnumerator(&audio_session_enumerator);
>+  if (error.Error() != S_OK) {
>+    RTC_LOG(LS_ERROR)
>+        << "IAudioSessionEnumerator::IAudioSessionEnumerator failed: "
>+        << ErrorToString(error);
>+    return ComPtr<IAudioSessionEnumerator>();
>+  }
>+  return audio_session_enumerator;
>+}
>+
>+// Creates and activates an IAudioClient COM object given the selected
>+// endpoint device.
>+ComPtr<IAudioClient> CreateClientInternal(IMMDevice* audio_device) {
>+  if (!audio_device)
>+    return ComPtr<IAudioClient>();
>+
>+  ComPtr<IAudioClient> audio_client;
>+  _com_error error = audio_device->Activate(__uuidof(IAudioClient), CLSCTX_ALL,
>+                                            nullptr, &audio_client);
>+  if (error.Error() != S_OK) {
>+    RTC_LOG(LS_ERROR) << "IMMDevice::Activate(IAudioClient) failed: "
>+                      << ErrorToString(error);
>+  }
>+  return audio_client;
>+}
>+
>+ComPtr<IAudioClient2> CreateClient2Internal(IMMDevice* audio_device) {
>+  if (!audio_device)
>+    return ComPtr<IAudioClient2>();
>+
>+  ComPtr<IAudioClient2> audio_client;
>+  _com_error error = audio_device->Activate(__uuidof(IAudioClient2), CLSCTX_ALL,
>+                                            nullptr, &audio_client);
>+  if (error.Error() != S_OK) {
>+    RTC_LOG(LS_ERROR) << "IMMDevice::Activate(IAudioClient2) failed: "
>+                      << ErrorToString(error);
>+  }
>+  return audio_client;
>+}
>+
>+ComPtr<IMMDeviceCollection> CreateCollectionInternal(EDataFlow data_flow) {
>+  ComPtr<IMMDeviceEnumerator> device_enumerator(
>+      CreateDeviceEnumeratorInternal(true));
>+  if (!device_enumerator) {
>+    return ComPtr<IMMDeviceCollection>();
>+  }
>+
>+  // Generate a collection of active (present and not disabled) audio endpoint
>+  // devices for the specified data-flow direction.
>+  // This method will succeed even if all devices are disabled.
>+  ComPtr<IMMDeviceCollection> collection;
>+  _com_error error = device_enumerator->EnumAudioEndpoints(
>+      data_flow, DEVICE_STATE_ACTIVE, collection.GetAddressOf());
>+  if (error.Error() != S_OK) {
>+    RTC_LOG(LS_ERROR) << "IMMDeviceCollection::EnumAudioEndpoints failed: "
>+                      << ErrorToString(error);
>+  }
>+  return collection;
>+}
>+
>+bool GetDeviceNamesInternal(EDataFlow data_flow,
>+                            webrtc::AudioDeviceNames* device_names) {
>+  // Always add the default device in index 0 and the default communication
>+  // device as index 1 in the vector. The name of the default device starts
>+  // with "Default - " and the default communication device starts with
>+  // "Communication - ".
>+  //  Example of friendly name: "Default - Headset (SB Arena Headset)"
>+  ERole role[] = {eConsole, eCommunications};
>+  ComPtr<IMMDevice> default_device;
>+  AudioDeviceName default_device_name;
>+  for (size_t i = 0; i < arraysize(role); ++i) {
>+    default_device = CreateDeviceInternal(AudioDeviceName::kDefaultDeviceId,
>+                                          data_flow, role[i]);
>+    std::string device_name;
>+    device_name += (role[i] == eConsole ? "Default - " : "Communication - ");
>+    device_name += GetDeviceFriendlyNameInternal(default_device.Get());
>+    std::string unique_id = GetDeviceIdInternal(default_device.Get());
>+
>+    default_device_name.device_name = std::move(device_name);
>+    default_device_name.unique_id = std::move(unique_id);
>+    RTC_DLOG(INFO) << "friendly name: " << default_device_name.device_name;
>+    RTC_DLOG(INFO) << "unique id    : " << default_device_name.unique_id;
>+    // Add combination of user-friendly and unique name to the output list.
>+    device_names->emplace_back(default_device_name);
>+  }
>+
>+  // Next, add all active input devices on index 2 and above. Note that,
>+  // one device can have more than one role. Hence, if only one input device
>+  // is present, the output vector will contain three elements all with the
>+  // same unique ID but with different names.
>+  // Example (one capture device but three elements in device_names):
>+  //   0: friendly name: Default - Headset (SB Arena Headset)
>+  //   0: unique id    : {0.0.1.00000000}.{822d99bb-d9b0-4f6f-b2a5-cd1be220d338}
>+  //   1: friendly name: Communication - Headset (SB Arena Headset)
>+  //   1: unique id    : {0.0.1.00000000}.{822d99bb-d9b0-4f6f-b2a5-cd1be220d338}
>+  //   2: friendly name: Headset (SB Arena Headset)
>+  //   2: unique id    : {0.0.1.00000000}.{822d99bb-d9b0-4f6f-b2a5-cd1be220d338}
>+
>+  // Generate a collection of active audio endpoint devices for the specified
>+  // direction.
>+  ComPtr<IMMDeviceCollection> collection = CreateCollectionInternal(data_flow);
>+  if (!collection.Get()) {
>+    return false;
>+  }
>+
>+  // Retrieve the number of active audio devices for the specified direction.
>+  UINT number_of_active_devices = 0;
>+  collection->GetCount(&number_of_active_devices);
>+  if (number_of_active_devices == 0) {
>+    return true;
>+  }
>+
>+  // Loop over all active devices and add friendly name and unique ID to the
>+  // |device_names| list which already contains two elements
>+  RTC_DCHECK_EQ(device_names->size(), 2);
>+  for (UINT i = 0; i < number_of_active_devices; ++i) {
>+    // Retrieve a pointer to the specified item in the device collection.
>+    ComPtr<IMMDevice> audio_device;
>+    _com_error error = collection->Item(i, audio_device.GetAddressOf());
>+    if (error.Error() != S_OK)
>+      continue;
>+    // Retrieve the complete device name for the given audio device endpoint.
>+    AudioDeviceName device_name(
>+        GetDeviceFriendlyNameInternal(audio_device.Get()),
>+        GetDeviceIdInternal(audio_device.Get()));
>+    RTC_DLOG(INFO) << "friendly name: " << device_name.device_name;
>+    RTC_DLOG(INFO) << "unique id    : " << device_name.unique_id;
>+    // Add combination of user-friendly and unique name to the output list.
>+    device_names->emplace_back(device_name);
>+  }
>+
>+  return true;
>+}
>+
>+HRESULT GetPreferredAudioParametersInternal(IAudioClient* client,
>+                                            AudioParameters* params) {
>+  WAVEFORMATPCMEX mix_format;
>+  HRESULT hr = core_audio_utility::GetSharedModeMixFormat(client, &mix_format);
>+  if (FAILED(hr))
>+    return hr;
>+
>+  REFERENCE_TIME default_period = 0;
>+  hr = core_audio_utility::GetDevicePeriod(client, AUDCLNT_SHAREMODE_SHARED,
>+                                           &default_period);
>+  if (FAILED(hr))
>+    return hr;
>+
>+  const int sample_rate = mix_format.Format.nSamplesPerSec;
>+  // TODO(henrika): utilize full mix_format.Format.wBitsPerSample.
>+  // const size_t bits_per_sample = AudioParameters::kBitsPerSample;
>+  // TODO(henrika): improve channel layout support.
>+  const size_t channels = mix_format.Format.nChannels;
>+  RTC_DCHECK_LE(channels, 2);
>+
>+  // Use the native device period to derive the smallest possible buffer size
>+  // in shared mode.
>+  double device_period_in_seconds =
>+      static_cast<double>(
>+          core_audio_utility::ReferenceTimeToTimeDelta(default_period).ms()) /
>+      1000.0L;
>+  const size_t frames_per_buffer =
>+      static_cast<size_t>(sample_rate * device_period_in_seconds + 0.5);
>+
>+  AudioParameters audio_params(sample_rate, channels, frames_per_buffer);
>+  *params = audio_params;
>+  RTC_DLOG(INFO) << audio_params.ToString();
>+
>+  return hr;
>+}
>+
>+}  // namespace
>+
>+namespace core_audio_utility {
>+
>+bool IsSupported() {
>+  RTC_DLOG(INFO) << "IsSupported";
>+  static bool g_is_supported = IsSupportedInternal();
>+  return g_is_supported;
>+}
>+
>+bool IsMMCSSSupported() {
>+  RTC_DLOG(INFO) << "IsMMCSSSupported";
>+  return LoadAvrtDll();
>+}
>+
>+int NumberOfActiveDevices(EDataFlow data_flow) {
>+  // Generate a collection of active audio endpoint devices for the specified
>+  // data-flow direction.
>+  ComPtr<IMMDeviceCollection> collection = CreateCollectionInternal(data_flow);
>+  if (!collection.Get()) {
>+    return 0;
>+  }
>+
>+  // Retrieve the number of active audio devices for the specified direction.
>+  UINT number_of_active_devices = 0;
>+  collection->GetCount(&number_of_active_devices);
>+  std::string str;
>+  if (data_flow == eCapture) {
>+    str = "Number of capture devices: ";
>+  } else if (data_flow == eRender) {
>+    str = "Number of render devices: ";
>+  } else if (data_flow == eAll) {
>+    str = "Total number of devices: ";
>+  }
>+  RTC_DLOG(INFO) << str << number_of_active_devices;
>+  return static_cast<int>(number_of_active_devices);
>+}
>+
>+ComPtr<IMMDeviceEnumerator> CreateDeviceEnumerator() {
>+  RTC_DLOG(INFO) << "CreateDeviceEnumerator";
>+  return CreateDeviceEnumeratorInternal(true);
>+}
>+
>+std::string GetDefaultInputDeviceID() {
>+  RTC_DLOG(INFO) << "GetDefaultInputDeviceID";
>+  ComPtr<IMMDevice> device(
>+      CreateDevice(AudioDeviceName::kDefaultDeviceId, eCapture, eConsole));
>+  return device.Get() ? GetDeviceIdInternal(device.Get()) : std::string();
>+}
>+
>+std::string GetDefaultOutputDeviceID() {
>+  RTC_DLOG(INFO) << "GetDefaultOutputDeviceID";
>+  ComPtr<IMMDevice> device(
>+      CreateDevice(AudioDeviceName::kDefaultDeviceId, eRender, eConsole));
>+  return device.Get() ? GetDeviceIdInternal(device.Get()) : std::string();
>+}
>+
>+std::string GetCommunicationsInputDeviceID() {
>+  RTC_DLOG(INFO) << "GetCommunicationsInputDeviceID";
>+  ComPtr<IMMDevice> device(CreateDevice(AudioDeviceName::kDefaultDeviceId,
>+                                        eCapture, eCommunications));
>+  return device.Get() ? GetDeviceIdInternal(device.Get()) : std::string();
>+}
>+
>+std::string GetCommunicationsOutputDeviceID() {
>+  RTC_DLOG(INFO) << "GetCommunicationsOutputDeviceID";
>+  ComPtr<IMMDevice> device(CreateDevice(AudioDeviceName::kDefaultDeviceId,
>+                                        eRender, eCommunications));
>+  return device.Get() ? GetDeviceIdInternal(device.Get()) : std::string();
>+}
>+
>+ComPtr<IMMDevice> CreateDevice(const std::string& device_id,
>+                               EDataFlow data_flow,
>+                               ERole role) {
>+  RTC_DLOG(INFO) << "CreateDevice";
>+  return CreateDeviceInternal(device_id, data_flow, role);
>+}
>+
>+AudioDeviceName GetDeviceName(IMMDevice* device) {
>+  RTC_DLOG(INFO) << "GetDeviceName";
>+  RTC_DCHECK(device);
>+  AudioDeviceName device_name(GetDeviceFriendlyNameInternal(device),
>+                              GetDeviceIdInternal(device));
>+  RTC_DLOG(INFO) << "friendly name: " << device_name.device_name;
>+  RTC_DLOG(INFO) << "unique id    : " << device_name.unique_id;
>+  return device_name;
>+}
>+
>+std::string GetFriendlyName(const std::string& device_id,
>+                            EDataFlow data_flow,
>+                            ERole role) {
>+  RTC_DLOG(INFO) << "GetFriendlyName";
>+  ComPtr<IMMDevice> audio_device = CreateDevice(device_id, data_flow, role);
>+  if (!audio_device.Get())
>+    return std::string();
>+
>+  AudioDeviceName device_name = GetDeviceName(audio_device.Get());
>+  return device_name.device_name;
>+}
>+
>+EDataFlow GetDataFlow(IMMDevice* device) {
>+  RTC_DLOG(INFO) << "GetDataFlow";
>+  RTC_DCHECK(device);
>+  ComPtr<IMMEndpoint> endpoint;
>+  _com_error error = device->QueryInterface(endpoint.GetAddressOf());
>+  if (error.Error() != S_OK) {
>+    RTC_LOG(LS_ERROR) << "IMMDevice::QueryInterface failed: "
>+                      << ErrorToString(error);
>+    return eAll;
>+  }
>+
>+  EDataFlow data_flow;
>+  error = endpoint->GetDataFlow(&data_flow);
>+  if (error.Error() != S_OK) {
>+    RTC_LOG(LS_ERROR) << "IMMEndpoint::GetDataFlow failed: "
>+                      << ErrorToString(error);
>+    return eAll;
>+  }
>+  return data_flow;
>+}
>+
>+bool GetInputDeviceNames(webrtc::AudioDeviceNames* device_names) {
>+  RTC_DLOG(INFO) << "GetInputDeviceNames";
>+  RTC_DCHECK(device_names);
>+  return GetDeviceNamesInternal(eCapture, device_names);
>+}
>+
>+bool GetOutputDeviceNames(webrtc::AudioDeviceNames* device_names) {
>+  RTC_DLOG(INFO) << "GetOutputDeviceNames";
>+  RTC_DCHECK(device_names);
>+  return GetDeviceNamesInternal(eRender, device_names);
>+}
>+
>+ComPtr<IAudioSessionManager2> CreateSessionManager2(IMMDevice* device) {
>+  RTC_DLOG(INFO) << "CreateSessionManager2";
>+  return CreateSessionManager2Internal(device);
>+}
>+
>+Microsoft::WRL::ComPtr<IAudioSessionEnumerator> CreateSessionEnumerator(
>+    IMMDevice* device) {
>+  RTC_DLOG(INFO) << "CreateSessionEnumerator";
>+  return CreateSessionEnumeratorInternal(device);
>+}
>+
>+int NumberOfActiveSessions(IMMDevice* device) {
>+  RTC_DLOG(INFO) << "NumberOfActiveSessions";
>+  ComPtr<IAudioSessionEnumerator> session_enumerator =
>+      CreateSessionEnumerator(device);
>+
>+  // Iterate over all audio sessions for the given device.
>+  int session_count = 0;
>+  _com_error error = session_enumerator->GetCount(&session_count);
>+  if (error.Error() != S_OK) {
>+    RTC_LOG(LS_ERROR) << "IAudioSessionEnumerator::GetCount failed: "
>+                      << ErrorToString(error);
>+    return 0;
>+  }
>+  RTC_DLOG(INFO) << "Total number of audio sessions: " << session_count;
>+
>+  int num_active = 0;
>+  for (int session = 0; session < session_count; session++) {
>+    // Acquire the session control interface.
>+    ComPtr<IAudioSessionControl> session_control;
>+    error = session_enumerator->GetSession(session, &session_control);
>+    if (error.Error() != S_OK) {
>+      RTC_LOG(LS_ERROR) << "IAudioSessionEnumerator::GetSession failed: "
>+                        << ErrorToString(error);
>+      return 0;
>+    }
>+
>+    // Log the display name of the audio session for debugging purposes.
>+    CComHeapPtr<WCHAR> display_name;
>+    if (SUCCEEDED(session_control->GetDisplayName(&display_name))) {
>+      RTC_DLOG(INFO) << "display name: "
>+                     << rtc::ToUtf8(display_name, wcslen(display_name));
>+    }
>+
>+    // Get the current state and check if the state is active or not.
>+    AudioSessionState state;
>+    error = session_control->GetState(&state);
>+    if (error.Error() != S_OK) {
>+      RTC_LOG(LS_ERROR) << "IAudioSessionControl::GetState failed: "
>+                        << ErrorToString(error);
>+      return 0;
>+    }
>+    if (state == AudioSessionStateActive) {
>+      ++num_active;
>+    }
>+  }
>+
>+  RTC_DLOG(INFO) << "Number of active audio sessions: " << num_active;
>+  return num_active;
>+}
>+
>+ComPtr<IAudioClient> CreateClient(const std::string& device_id,
>+                                  EDataFlow data_flow,
>+                                  ERole role) {
>+  RTC_DLOG(INFO) << "CreateClient";
>+  ComPtr<IMMDevice> device(CreateDevice(device_id, data_flow, role));
>+  return CreateClientInternal(device.Get());
>+}
>+
>+ComPtr<IAudioClient2> CreateClient2(const std::string& device_id,
>+                                    EDataFlow data_flow,
>+                                    ERole role) {
>+  RTC_DLOG(INFO) << "CreateClient2";
>+  ComPtr<IMMDevice> device(CreateDevice(device_id, data_flow, role));
>+  return CreateClient2Internal(device.Get());
>+}
>+
>+HRESULT SetClientProperties(IAudioClient2* client) {
>+  RTC_DLOG(INFO) << "SetClientProperties";
>+  RTC_DCHECK(client);
>+
>+  AudioClientProperties properties = {0};
>+  properties.cbSize = sizeof(AudioClientProperties);
>+  properties.bIsOffload = false;
>+  // Real-time VoIP communication.
>+  // TODO(henrika): other categories?
>+  properties.eCategory = AudioCategory_Communications;
>+  // TODO(henrika): can AUDCLNT_STREAMOPTIONS_RAW be used?
>+  properties.Options = AUDCLNT_STREAMOPTIONS_NONE;
>+  _com_error error = client->SetClientProperties(&properties);
>+  if (error.Error() != S_OK) {
>+    RTC_LOG(LS_ERROR) << "IAudioClient2::SetClientProperties failed: "
>+                      << ErrorToString(error);
>+  }
>+  return error.Error();
>+}
>+
>+HRESULT GetSharedModeMixFormat(IAudioClient* client,
>+                               WAVEFORMATEXTENSIBLE* format) {
>+  RTC_DLOG(INFO) << "GetSharedModeMixFormat";
>+  RTC_DCHECK(client);
>+  ScopedCoMem<WAVEFORMATEXTENSIBLE> format_ex;
>+  _com_error error =
>+      client->GetMixFormat(reinterpret_cast<WAVEFORMATEX**>(&format_ex));
>+  if (error.Error() != S_OK) {
>+    RTC_LOG(LS_ERROR) << "IAudioClient::GetMixFormat failed: "
>+                      << ErrorToString(error);
>+    return error.Error();
>+  }
>+
>+  size_t bytes = sizeof(WAVEFORMATEX) + format_ex->Format.cbSize;
>+  RTC_DCHECK_EQ(bytes, sizeof(WAVEFORMATEXTENSIBLE));
>+  memcpy(format, format_ex, bytes);
>+  RTC_DLOG(INFO) << WaveFormatExToString(format);
>+
>+  return error.Error();
>+}
>+
>+bool IsFormatSupported(IAudioClient* client,
>+                       AUDCLNT_SHAREMODE share_mode,
>+                       const WAVEFORMATEXTENSIBLE* format) {
>+  RTC_DLOG(INFO) << "IsFormatSupported";
>+  RTC_DCHECK(client);
>+  ScopedCoMem<WAVEFORMATEXTENSIBLE> closest_match;
>+  // This method provides a way for a client to determine, before calling
>+  // IAudioClient::Initialize, whether the audio engine supports a particular
>+  // stream format or not. In shared mode, the audio engine always supports
>+  // the mix format (see GetSharedModeMixFormat).
>+  // TODO(henrika): verify support for exclusive mode as well?
>+  _com_error error = client->IsFormatSupported(
>+      share_mode, reinterpret_cast<const WAVEFORMATEX*>(format),
>+      reinterpret_cast<WAVEFORMATEX**>(&closest_match));
>+  if ((error.Error() == S_OK) && (closest_match == nullptr)) {
>+    RTC_DLOG(INFO)
>+        << "The audio endpoint device supports the specified stream format";
>+  } else if ((error.Error() == S_FALSE) && (closest_match != nullptr)) {
>+    // Call succeeded with a closest match to the specified format. This log can
>+    // only be triggered for shared mode.
>+    RTC_LOG(LS_WARNING)
>+        << "Exact format is not supported, but a closest match exists";
>+    RTC_LOG(INFO) << WaveFormatExToString(closest_match);
>+  } else if ((error.Error() == AUDCLNT_E_UNSUPPORTED_FORMAT) &&
>+             (closest_match == nullptr)) {
>+    // The audio engine does not support the caller-specified format or any
>+    // similar format.
>+    RTC_DLOG(INFO) << "The audio endpoint device does not support the "
>+                      "specified stream format";
>+  } else {
>+    RTC_LOG(LS_ERROR) << "IAudioClient::IsFormatSupported failed: "
>+                      << ErrorToString(error);
>+  }
>+
>+  return (error.Error() == S_OK);
>+}
>+
>+HRESULT GetDevicePeriod(IAudioClient* client,
>+                        AUDCLNT_SHAREMODE share_mode,
>+                        REFERENCE_TIME* device_period) {
>+  RTC_DLOG(INFO) << "GetDevicePeriod";
>+  RTC_DCHECK(client);
>+  // The |default_period| parameter specifies the default scheduling period
>+  // for a shared-mode stream. The |minimum_period| parameter specifies the
>+  // minimum scheduling period for an exclusive-mode stream.
>+  // The time is expressed in 100-nanosecond units.
>+  REFERENCE_TIME default_period = 0;
>+  REFERENCE_TIME minimum_period = 0;
>+  _com_error error = client->GetDevicePeriod(&default_period, &minimum_period);
>+  if (error.Error() != S_OK) {
>+    RTC_LOG(LS_ERROR) << "IAudioClient::GetDevicePeriod failed: "
>+                      << ErrorToString(error);
>+    return error.Error();
>+  }
>+
>+  *device_period = (share_mode == AUDCLNT_SHAREMODE_SHARED) ? default_period
>+                                                            : minimum_period;
>+  RTC_LOG(INFO) << "device_period: "
>+                << ReferenceTimeToTimeDelta(*device_period).ms() << " [ms]";
>+  return error.Error();
>+}
>+
>+HRESULT GetPreferredAudioParameters(const std::string& device_id,
>+                                    bool is_output_device,
>+                                    AudioParameters* params) {
>+  RTC_DLOG(INFO) << "GetPreferredAudioParameters";
>+  EDataFlow data_flow = is_output_device ? eRender : eCapture;
>+  ComPtr<IMMDevice> device;
>+  if (device_id == AudioDeviceName::kDefaultCommunicationsDeviceId) {
>+    device = CreateDeviceInternal(AudioDeviceName::kDefaultDeviceId, data_flow,
>+                                  eCommunications);
>+  } else {
>+    // If |device_id| equals AudioDeviceName::kDefaultDeviceId, a default
>+    // device will be created.
>+    device = CreateDeviceInternal(device_id, data_flow, eConsole);
>+  }
>+  if (!device.Get()) {
>+    return E_FAIL;
>+  }
>+
>+  ComPtr<IAudioClient> client(CreateClientInternal(device.Get()));
>+  if (!client.Get())
>+    return E_FAIL;
>+
>+  return GetPreferredAudioParametersInternal(client.Get(), params);
>+}
>+
>+HRESULT GetPreferredAudioParameters(IAudioClient* client,
>+                                    AudioParameters* params) {
>+  RTC_DCHECK(client);
>+  return GetPreferredAudioParametersInternal(client, params);
>+}
>+
>+HRESULT SharedModeInitialize(IAudioClient* client,
>+                             const WAVEFORMATEXTENSIBLE* format,
>+                             HANDLE event_handle,
>+                             uint32_t* endpoint_buffer_size) {
>+  RTC_DLOG(INFO) << "SharedModeInitialize";
>+  RTC_DCHECK(client);
>+  // The AUDCLNT_STREAMFLAGS_NOPERSIST flag disables persistence of the volume
>+  // and mute settings for a session that contains rendering streams.
>+  // By default, the volume level and muting state for a rendering session are
>+  // persistent across system restarts. The volume level and muting state for a
>+  // capture session are never persistent.
>+  DWORD stream_flags = AUDCLNT_STREAMFLAGS_NOPERSIST;
>+
>+  // Enable event-driven streaming if a valid event handle is provided.
>+  // After the stream starts, the audio engine will signal the event handle
>+  // to notify the client each time a buffer becomes ready to process.
>+  // Event-driven buffering is supported for both rendering and capturing.
>+  // Both shared-mode and exclusive-mode streams can use event-driven buffering.
>+  bool use_event =
>+      (event_handle != nullptr && event_handle != INVALID_HANDLE_VALUE);
>+  if (use_event) {
>+    stream_flags |= AUDCLNT_STREAMFLAGS_EVENTCALLBACK;
>+    RTC_DLOG(INFO) << "The stream is initialized to be event driven";
>+  }
>+  RTC_DLOG(INFO) << "stream_flags: 0x" << rtc::ToHex(stream_flags);
>+
>+  // Initialize the shared mode client for minimal delay.
>+  _com_error error = client->Initialize(
>+      AUDCLNT_SHAREMODE_SHARED, stream_flags, 0, 0,
>+      reinterpret_cast<const WAVEFORMATEX*>(format), nullptr);
>+  if (error.Error() != S_OK) {
>+    RTC_LOG(LS_ERROR) << "IAudioClient::Initialize failed: "
>+                      << ErrorToString(error);
>+    return error.Error();
>+  }
>+
>+  // If a stream is initialized to be event driven and in shared mode, the
>+  // associated application must also obtain a handle by making a call to
>+  // IAudioClient::SetEventHandle.
>+  if (use_event) {
>+    error = client->SetEventHandle(event_handle);
>+    if (error.Error() != S_OK) {
>+      RTC_LOG(LS_ERROR) << "IAudioClient::SetEventHandle failed: "
>+                        << ErrorToString(error);
>+      return error.Error();
>+    }
>+  }
>+
>+  UINT32 buffer_size_in_frames = 0;
>+  // Retrieves the size (maximum capacity) of the endpoint buffer. The size is
>+  // expressed as the number of audio frames the buffer can hold.
>+  error = client->GetBufferSize(&buffer_size_in_frames);
>+  if (error.Error() != S_OK) {
>+    RTC_LOG(LS_ERROR) << "IAudioClient::GetBufferSize failed: "
>+                      << ErrorToString(error);
>+    return error.Error();
>+  }
>+
>+  *endpoint_buffer_size = buffer_size_in_frames;
>+  RTC_DLOG(INFO) << "endpoint buffer size: " << buffer_size_in_frames
>+                 << " [audio frames]";
>+  RTC_DLOG(INFO) << "bytes per audio frame: " << format->Format.nBlockAlign;
>+  RTC_DLOG(INFO) << "endpoint buffer size: "
>+                 << buffer_size_in_frames * format->Format.nChannels *
>+                        (format->Format.wBitsPerSample / 8)
>+                 << " [bytes]";
>+
>+  // TODO(henrika): utilize when delay measurements are added.
>+  REFERENCE_TIME latency = 0;
>+  error = client->GetStreamLatency(&latency);
>+  RTC_DLOG(INFO) << "stream latency: " << ReferenceTimeToTimeDelta(latency).ms()
>+                 << " [ms]";
>+  return error.Error();
>+}
>+
>+ComPtr<IAudioRenderClient> CreateRenderClient(IAudioClient* client) {
>+  RTC_DLOG(INFO) << "CreateRenderClient";
>+  RTC_DCHECK(client);
>+  // Get access to the IAudioRenderClient interface. This interface
>+  // enables us to write output data to a rendering endpoint buffer.
>+  ComPtr<IAudioRenderClient> audio_render_client;
>+  _com_error error = client->GetService(IID_PPV_ARGS(&audio_render_client));
>+  if (error.Error() != S_OK) {
>+    RTC_LOG(LS_ERROR)
>+        << "IAudioClient::GetService(IID_IAudioRenderClient) failed: "
>+        << ErrorToString(error);
>+    return ComPtr<IAudioRenderClient>();
>+  }
>+  return audio_render_client;
>+}
>+
>+ComPtr<IAudioCaptureClient> CreateCaptureClient(IAudioClient* client) {
>+  RTC_DLOG(INFO) << "CreateCaptureClient";
>+  RTC_DCHECK(client);
>+  // Get access to the IAudioCaptureClient interface. This interface
>+  // enables us to read input data from a capturing endpoint buffer.
>+  ComPtr<IAudioCaptureClient> audio_capture_client;
>+  _com_error error = client->GetService(IID_PPV_ARGS(&audio_capture_client));
>+  if (error.Error() != S_OK) {
>+    RTC_LOG(LS_ERROR)
>+        << "IAudioClient::GetService(IID_IAudioCaptureClient) failed: "
>+        << ErrorToString(error);
>+    return ComPtr<IAudioCaptureClient>();
>+  }
>+  return audio_capture_client;
>+}
>+
>+ComPtr<IAudioClock> CreateAudioClock(IAudioClient* client) {
>+  RTC_DLOG(INFO) << "CreateAudioClock";
>+  RTC_DCHECK(client);
>+  // Get access to the IAudioClock interface. This interface enables us to
>+  // monitor a stream's data rate and the current position in the stream.
>+  ComPtr<IAudioClock> audio_clock;
>+  _com_error error = client->GetService(IID_PPV_ARGS(&audio_clock));
>+  if (error.Error() != S_OK) {
>+    RTC_LOG(LS_ERROR) << "IAudioClient::GetService(IID_IAudioClock) failed: "
>+                      << ErrorToString(error);
>+    return ComPtr<IAudioClock>();
>+  }
>+  return audio_clock;
>+}
>+
>+ComPtr<ISimpleAudioVolume> CreateSimpleAudioVolume(IAudioClient* client) {
>+  RTC_DLOG(INFO) << "CreateSimpleAudioVolume";
>+  RTC_DCHECK(client);
>+  // Get access to the ISimpleAudioVolume interface. This interface enables a
>+  // client to control the master volume level of an audio session.
>+  ComPtr<ISimpleAudioVolume> simple_audio_volume;
>+  _com_error error = client->GetService(IID_PPV_ARGS(&simple_audio_volume));
>+  if (error.Error() != S_OK) {
>+    RTC_LOG(LS_ERROR)
>+        << "IAudioClient::GetService(IID_ISimpleAudioVolume) failed: "
>+        << ErrorToString(error);
>+    return ComPtr<ISimpleAudioVolume>();
>+  }
>+  return simple_audio_volume;
>+}
>+
>+bool FillRenderEndpointBufferWithSilence(IAudioClient* client,
>+                                         IAudioRenderClient* render_client) {
>+  RTC_DLOG(INFO) << "FillRenderEndpointBufferWithSilence";
>+  RTC_DCHECK(client);
>+  RTC_DCHECK(render_client);
>+  UINT32 endpoint_buffer_size = 0;
>+  _com_error error = client->GetBufferSize(&endpoint_buffer_size);
>+  if (error.Error() != S_OK) {
>+    RTC_LOG(LS_ERROR) << "IAudioClient::GetBufferSize failed: "
>+                      << ErrorToString(error);
>+    return false;
>+  }
>+
>+  UINT32 num_queued_frames = 0;
>+  // Get number of audio frames that are queued up to play in the endpoint
>+  // buffer.
>+  error = client->GetCurrentPadding(&num_queued_frames);
>+  if (error.Error() != S_OK) {
>+    RTC_LOG(LS_ERROR) << "IAudioClient::GetCurrentPadding failed: "
>+                      << ErrorToString(error);
>+    return false;
>+  }
>+  RTC_DLOG(INFO) << "num_queued_frames: " << num_queued_frames;
>+
>+  BYTE* data = nullptr;
>+  int num_frames_to_fill = endpoint_buffer_size - num_queued_frames;
>+  RTC_DLOG(INFO) << "num_frames_to_fill: " << num_frames_to_fill;
>+  error = render_client->GetBuffer(num_frames_to_fill, &data);
>+  if (error.Error() != S_OK) {
>+    RTC_LOG(LS_ERROR) << "IAudioRenderClient::GetBuffer failed: "
>+                      << ErrorToString(error);
>+    return false;
>+  }
>+
>+  // Using the AUDCLNT_BUFFERFLAGS_SILENT flag eliminates the need to
>+  // explicitly write silence data to the rendering buffer.
>+  error = render_client->ReleaseBuffer(num_frames_to_fill,
>+                                       AUDCLNT_BUFFERFLAGS_SILENT);
>+  if (error.Error() != S_OK) {
>+    RTC_LOG(LS_ERROR) << "IAudioRenderClient::ReleaseBuffer failed: "
>+                      << ErrorToString(error);
>+    return false;
>+  }
>+
>+  return true;
>+}
>+
>+std::string WaveFormatExToString(const WAVEFORMATEXTENSIBLE* format) {
>+  RTC_DCHECK_EQ(format->Format.wFormatTag, WAVE_FORMAT_EXTENSIBLE);
>+  char ss_buf[1024];
>+  rtc::SimpleStringBuilder ss(ss_buf);
>+  ss.AppendFormat("wFormatTag: WAVE_FORMAT_EXTENSIBLE");
>+  ss.AppendFormat(", nChannels: %d", format->Format.nChannels);
>+  ss.AppendFormat(", nSamplesPerSec: %d", format->Format.nSamplesPerSec);
>+  ss.AppendFormat(", nAvgBytesPerSec: %d", format->Format.nAvgBytesPerSec);
>+  ss.AppendFormat(", nBlockAlign: %d", format->Format.nBlockAlign);
>+  ss.AppendFormat(", wBitsPerSample: %d", format->Format.wBitsPerSample);
>+  ss.AppendFormat(", cbSize: %d", format->Format.cbSize);
>+  ss.AppendFormat(", wValidBitsPerSample: %d",
>+                  format->Samples.wValidBitsPerSample);
>+  ss.AppendFormat(", dwChannelMask: 0x%X", format->dwChannelMask);
>+  if (format->SubFormat == KSDATAFORMAT_SUBTYPE_PCM) {
>+    ss << ", SubFormat: KSDATAFORMAT_SUBTYPE_PCM";
>+  } else if (format->SubFormat == KSDATAFORMAT_SUBTYPE_IEEE_FLOAT) {
>+    ss << ", SubFormat: KSDATAFORMAT_SUBTYPE_IEEE_FLOAT";
>+  } else {
>+    ss << ", SubFormat: NOT_SUPPORTED";
>+  }
>+  return ss.str();
>+}
>+
>+webrtc::TimeDelta ReferenceTimeToTimeDelta(REFERENCE_TIME time) {
>+  // Each unit of reference time is 100 nanoseconds <=> 0.1 microsecond.
>+  return webrtc::TimeDelta::us(0.1 * time + 0.5);
>+}
>+
>+std::string ErrorToString(const _com_error& error) {
>+  char ss_buf[1024];
>+  rtc::SimpleStringBuilder ss(ss_buf);
>+  ss.AppendFormat("%s (0x%08X)", rtc::ToUtf8(error.ErrorMessage()).c_str(),
>+                  error.Error());
>+  return ss.str();
>+}
>+
>+}  // namespace core_audio_utility
>+}  // namespace webrtc_win
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/win/core_audio_utility_win.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/win/core_audio_utility_win.h
>new file mode 100644
>index 00000000000..87e49606cac
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/win/core_audio_utility_win.h
>@@ -0,0 +1,479 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_DEVICE_WIN_CORE_AUDIO_UTILITY_WIN_H_
>+#define MODULES_AUDIO_DEVICE_WIN_CORE_AUDIO_UTILITY_WIN_H_
>+
>+#include <Audioclient.h>
>+#include <Audiopolicy.h>
>+#include <Mmdeviceapi.h>
>+#include <avrt.h>
>+#include <comdef.h>
>+#include <objbase.h>
>+#include <propidl.h>
>+#include <wrl/client.h>
>+
>+#include <string>
>+
>+#include "api/units/time_delta.h"
>+#include "modules/audio_device/audio_device_name.h"
>+#include "modules/audio_device/include/audio_device_defines.h"
>+#include "rtc_base/logging.h"
>+
>+#pragma comment(lib, "Avrt.lib")
>+
>+namespace webrtc {
>+namespace webrtc_win {
>+
>+static const int64_t kNumMicrosecsPerSec = webrtc::TimeDelta::seconds(1).us();
>+
>+// Utility class which registers a thread with MMCSS in the constructor and
>+// deregisters MMCSS in the destructor. The task name is given by |task_name|.
>+// The Multimedia Class Scheduler service (MMCSS) enables multimedia
>+// applications to ensure that their time-sensitive processing receives
>+// prioritized access to CPU resources without denying CPU resources to
>+// lower-priority applications.
>+class ScopedMMCSSRegistration {
>+ public:
>+  explicit ScopedMMCSSRegistration(const TCHAR* task_name) {
>+    RTC_DLOG(INFO) << "ScopedMMCSSRegistration: " << rtc::ToUtf8(task_name);
>+    // Register the calling thread with MMCSS for the supplied |task_name|.
>+    DWORD mmcss_task_index = 0;
>+    mmcss_handle_ = AvSetMmThreadCharacteristics(task_name, &mmcss_task_index);
>+    if (mmcss_handle_ == nullptr) {
>+      RTC_LOG(LS_ERROR) << "Failed to enable MMCSS on this thread: "
>+                        << GetLastError();
>+    }
>+  }
>+
>+  ~ScopedMMCSSRegistration() {
>+    if (Succeeded()) {
>+      // Deregister with MMCSS.
>+      RTC_DLOG(INFO) << "~ScopedMMCSSRegistration";
>+      AvRevertMmThreadCharacteristics(mmcss_handle_);
>+    }
>+  }
>+
>+  ScopedMMCSSRegistration(const ScopedMMCSSRegistration&) = delete;
>+  ScopedMMCSSRegistration& operator=(const ScopedMMCSSRegistration&) = delete;
>+
>+  bool Succeeded() const { return mmcss_handle_ != nullptr; }
>+
>+ private:
>+  HANDLE mmcss_handle_ = nullptr;
>+};
>+
>+// Initializes COM in the constructor (STA or MTA), and uninitializes COM in the
>+// destructor. Taken from base::win::ScopedCOMInitializer.
>+//
>+// WARNING: This should only be used once per thread, ideally scoped to a
>+// similar lifetime as the thread itself.  You should not be using this in
>+// random utility functions that make COM calls; instead ensure that these
>+// functions are running on a COM-supporting thread!
>+// See https://msdn.microsoft.com/en-us/library/ms809971.aspx for details.
>+class ScopedCOMInitializer {
>+ public:
>+  // Enum value provided to initialize the thread as an MTA instead of STA.
>+  // There are two types of apartments, Single Threaded Apartments (STAs)
>+  // and Multi Threaded Apartments (MTAs). Within a given process there can
>+  // be multiple STAâs but there is only one MTA. STA is typically used by
>+  // "GUI applications" and MTA by "worker threads" with no UI message loop.
>+  enum SelectMTA { kMTA };
>+
>+  // Constructor for STA initialization.
>+  ScopedCOMInitializer() {
>+    RTC_DLOG(INFO) << "Single-Threaded Apartment (STA) COM thread";
>+    Initialize(COINIT_APARTMENTTHREADED);
>+  }
>+
>+  // Constructor for MTA initialization.
>+  explicit ScopedCOMInitializer(SelectMTA mta) {
>+    RTC_DLOG(INFO) << "Multi-Threaded Apartment (MTA) COM thread";
>+    Initialize(COINIT_MULTITHREADED);
>+  }
>+
>+  ~ScopedCOMInitializer() {
>+    if (Succeeded()) {
>+      CoUninitialize();
>+    }
>+  }
>+
>+  ScopedCOMInitializer(const ScopedCOMInitializer&) = delete;
>+  ScopedCOMInitializer& operator=(const ScopedCOMInitializer&) = delete;
>+
>+  bool Succeeded() { return SUCCEEDED(hr_); }
>+
>+ private:
>+  void Initialize(COINIT init) {
>+    // Initializes the COM library for use by the calling thread, sets the
>+    // thread's concurrency model, and creates a new apartment for the thread
>+    // if one is required. CoInitializeEx must be called at least once, and is
>+    // usually called only once, for each thread that uses the COM library.
>+    hr_ = CoInitializeEx(NULL, init);
>+    RTC_CHECK_NE(RPC_E_CHANGED_MODE, hr_)
>+        << "Invalid COM thread model change (MTA->STA)";
>+    // Multiple calls to CoInitializeEx by the same thread are allowed as long
>+    // as they pass the same concurrency flag, but subsequent valid calls
>+    // return S_FALSE. To close the COM library gracefully on a thread, each
>+    // successful call to CoInitializeEx, including any call that returns
>+    // S_FALSE, must be balanced by a corresponding call to CoUninitialize.
>+    if (hr_ == S_OK) {
>+      RTC_DLOG(INFO)
>+          << "The COM library was initialized successfully on this thread";
>+    } else if (hr_ == S_FALSE) {
>+      RTC_DLOG(WARNING)
>+          << "The COM library is already initialized on this thread";
>+    }
>+  }
>+  HRESULT hr_;
>+};
>+
>+// A PROPVARIANT that is automatically initialized and cleared upon respective
>+// construction and destruction of this class.
>+class ScopedPropVariant {
>+ public:
>+  ScopedPropVariant() { PropVariantInit(&pv_); }
>+
>+  ~ScopedPropVariant() { Reset(); }
>+
>+  ScopedPropVariant(const ScopedPropVariant&) = delete;
>+  ScopedPropVariant& operator=(const ScopedPropVariant&) = delete;
>+  bool operator==(const ScopedPropVariant&) const = delete;
>+  bool operator!=(const ScopedPropVariant&) const = delete;
>+
>+  // Returns a pointer to the underlying PROPVARIANT for use as an out param in
>+  // a function call.
>+  PROPVARIANT* Receive() {
>+    RTC_DCHECK_EQ(pv_.vt, VT_EMPTY);
>+    return &pv_;
>+  }
>+
>+  // Clears the instance to prepare it for re-use (e.g., via Receive).
>+  void Reset() {
>+    if (pv_.vt != VT_EMPTY) {
>+      HRESULT result = PropVariantClear(&pv_);
>+      RTC_DCHECK_EQ(result, S_OK);
>+    }
>+  }
>+
>+  const PROPVARIANT& get() const { return pv_; }
>+  const PROPVARIANT* ptr() const { return &pv_; }
>+
>+ private:
>+  PROPVARIANT pv_;
>+};
>+
>+// Simple scoped memory releaser class for COM allocated memory.
>+template <typename T>
>+class ScopedCoMem {
>+ public:
>+  ScopedCoMem() : mem_ptr_(nullptr) {}
>+
>+  ~ScopedCoMem() { Reset(nullptr); }
>+
>+  ScopedCoMem(const ScopedCoMem&) = delete;
>+  ScopedCoMem& operator=(const ScopedCoMem&) = delete;
>+
>+  T** operator&() {                   // NOLINT
>+    RTC_DCHECK(mem_ptr_ == nullptr);  // To catch memory leaks.
>+    return &mem_ptr_;
>+  }
>+
>+  operator T*() { return mem_ptr_; }
>+
>+  T* operator->() {
>+    RTC_DCHECK(mem_ptr_ != nullptr);
>+    return mem_ptr_;
>+  }
>+
>+  const T* operator->() const {
>+    RTC_DCHECK(mem_ptr_ != nullptr);
>+    return mem_ptr_;
>+  }
>+
>+  explicit operator bool() const { return mem_ptr_; }
>+
>+  friend bool operator==(const ScopedCoMem& lhs, std::nullptr_t) {
>+    return lhs.Get() == nullptr;
>+  }
>+
>+  friend bool operator==(std::nullptr_t, const ScopedCoMem& rhs) {
>+    return rhs.Get() == nullptr;
>+  }
>+
>+  friend bool operator!=(const ScopedCoMem& lhs, std::nullptr_t) {
>+    return lhs.Get() != nullptr;
>+  }
>+
>+  friend bool operator!=(std::nullptr_t, const ScopedCoMem& rhs) {
>+    return rhs.Get() != nullptr;
>+  }
>+
>+  void Reset(T* ptr) {
>+    if (mem_ptr_)
>+      CoTaskMemFree(mem_ptr_);
>+    mem_ptr_ = ptr;
>+  }
>+
>+  T* Get() const { return mem_ptr_; }
>+
>+ private:
>+  T* mem_ptr_;
>+};
>+
>+// A HANDLE that is automatically initialized and closed upon respective
>+// construction and destruction of this class.
>+class ScopedHandle {
>+ public:
>+  ScopedHandle() : handle_(nullptr) {}
>+  explicit ScopedHandle(HANDLE h) : handle_(nullptr) { Set(h); }
>+
>+  ~ScopedHandle() { Close(); }
>+
>+  ScopedHandle& operator=(const ScopedHandle&) = delete;
>+  bool operator==(const ScopedHandle&) const = delete;
>+  bool operator!=(const ScopedHandle&) const = delete;
>+
>+  // Use this instead of comparing to INVALID_HANDLE_VALUE.
>+  bool IsValid() const { return handle_ != nullptr; }
>+
>+  void Set(HANDLE new_handle) {
>+    Close();
>+    // Windows is inconsistent about invalid handles.
>+    // See https://blogs.msdn.microsoft.com/oldnewthing/20040302-00/?p=40443
>+    // for details.
>+    if (new_handle != INVALID_HANDLE_VALUE) {
>+      handle_ = new_handle;
>+    }
>+  }
>+
>+  HANDLE Get() const { return handle_; }
>+
>+  operator HANDLE() const { return handle_; }
>+
>+  void Close() {
>+    if (handle_) {
>+      if (!::CloseHandle(handle_)) {
>+        RTC_NOTREACHED();
>+      }
>+      handle_ = nullptr;
>+    }
>+  }
>+
>+ private:
>+  HANDLE handle_;
>+};
>+
>+// Utility methods for the Core Audio API on Windows.
>+// Always ensure that Core Audio is supported before using these methods.
>+// Use webrtc_win::core_audio_utility::IsSupported() for this purpose.
>+// Also, all methods must be called on a valid COM thread. This can be done
>+// by using the webrtc_win::ScopedCOMInitializer helper class.
>+// These methods are based on media::CoreAudioUtil in Chrome.
>+namespace core_audio_utility {
>+
>+// Returns true if Windows Core Audio is supported.
>+// Always verify that this method returns true before using any of the
>+// other methods in this class.
>+bool IsSupported();
>+
>+// Returns true if Multimedia Class Scheduler service (MMCSS) is supported.
>+// The MMCSS enables multimedia applications to ensure that their time-sensitive
>+// processing receives prioritized access to CPU resources without denying CPU
>+// resources to lower-priority applications.
>+bool IsMMCSSSupported();
>+
>+// The MMDevice API lets clients discover the audio endpoint devices in the
>+// system and determine which devices are suitable for the application to use.
>+// Header file Mmdeviceapi.h defines the interfaces in the MMDevice API.
>+
>+// Number of active audio devices in the specified data flow direction.
>+// Set |data_flow| to eAll to retrieve the total number of active audio
>+// devices.
>+int NumberOfActiveDevices(EDataFlow data_flow);
>+
>+// Creates an IMMDeviceEnumerator interface which provides methods for
>+// enumerating audio endpoint devices.
>+// TODO(henrika): IMMDeviceEnumerator::RegisterEndpointNotificationCallback.
>+Microsoft::WRL::ComPtr<IMMDeviceEnumerator> CreateDeviceEnumerator();
>+
>+// These functions return the unique device id of the default or
>+// communications input/output device, or an empty string if no such device
>+// exists or if the device has been disabled.
>+std::string GetDefaultInputDeviceID();
>+std::string GetDefaultOutputDeviceID();
>+std::string GetCommunicationsInputDeviceID();
>+std::string GetCommunicationsOutputDeviceID();
>+
>+// Creates an IMMDevice interface corresponding to the unique device id in
>+// |device_id|, or by data-flow direction and role if |device_id| is set to
>+// AudioDeviceName::kDefaultDeviceId.
>+Microsoft::WRL::ComPtr<IMMDevice> CreateDevice(const std::string& device_id,
>+                                               EDataFlow data_flow,
>+                                               ERole role);
>+
>+// Returns the unique ID and user-friendly name of a given endpoint device.
>+// Example: "{0.0.1.00000000}.{8db6020f-18e3-4f25-b6f5-7726c9122574}", and
>+//          "Microphone (Realtek High Definition Audio)".
>+webrtc::AudioDeviceName GetDeviceName(IMMDevice* device);
>+
>+// Gets the user-friendly name of the endpoint device which is represented
>+// by a unique id in |device_id|, or by data-flow direction and role if
>+// |device_id| is set to AudioDeviceName::kDefaultDeviceId.
>+std::string GetFriendlyName(const std::string& device_id,
>+                            EDataFlow data_flow,
>+                            ERole role);
>+
>+// Query if the audio device is a rendering device or a capture device.
>+EDataFlow GetDataFlow(IMMDevice* device);
>+
>+// Enumerates all input devices and adds the names (friendly name and unique
>+// device id) to the list in |device_names|.
>+bool GetInputDeviceNames(webrtc::AudioDeviceNames* device_names);
>+
>+// Enumerates all output devices and adds the names (friendly name and unique
>+// device id) to the list in |device_names|.
>+bool GetOutputDeviceNames(webrtc::AudioDeviceNames* device_names);
>+
>+// The Windows Audio Session API (WASAPI) enables client applications to
>+// manage the flow of audio data between the application and an audio endpoint
>+// device. Header files Audioclient.h and Audiopolicy.h define the WASAPI
>+// interfaces.
>+
>+// Creates an IAudioSessionManager2 interface for the specified |device|.
>+// This interface provides access to e.g. the IAudioSessionEnumerator
>+Microsoft::WRL::ComPtr<IAudioSessionManager2> CreateSessionManager2(
>+    IMMDevice* device);
>+
>+// Creates an IAudioSessionEnumerator interface for the specified |device|.
>+// The client can use the interface to enumerate audio sessions on the audio
>+// device
>+Microsoft::WRL::ComPtr<IAudioSessionEnumerator> CreateSessionEnumerator(
>+    IMMDevice* device);
>+
>+// Number of active audio sessions for the given |device|. Expired or inactive
>+// sessions are not included.
>+int NumberOfActiveSessions(IMMDevice* device);
>+
>+// Creates an IAudioClient instance for a specific device or the default
>+// device specified by data-flow direction and role.
>+Microsoft::WRL::ComPtr<IAudioClient> CreateClient(const std::string& device_id,
>+                                                  EDataFlow data_flow,
>+                                                  ERole role);
>+
>+Microsoft::WRL::ComPtr<IAudioClient2>
>+CreateClient2(const std::string& device_id, EDataFlow data_flow, ERole role);
>+
>+// Sets the AudioCategory_Communications category. Should be called before
>+// GetSharedModeMixFormat() and IsFormatSupported().
>+// Minimum supported client: Windows 8.
>+// TODO(henrika): evaluate effect (if any).
>+HRESULT SetClientProperties(IAudioClient2* client);
>+
>+// Get the mix format that the audio engine uses internally for processing
>+// of shared-mode streams. The client can call this method before calling
>+// IAudioClient::Initialize. When creating a shared-mode stream for an audio
>+// endpoint device, the Initialize method always accepts the stream format
>+// obtained by this method.
>+HRESULT GetSharedModeMixFormat(IAudioClient* client,
>+                               WAVEFORMATEXTENSIBLE* format);
>+
>+// Returns true if the specified |client| supports the format in |format|
>+// for the given |share_mode| (shared or exclusive). The client can call this
>+// method before calling IAudioClient::Initialize.
>+bool IsFormatSupported(IAudioClient* client,
>+                       AUDCLNT_SHAREMODE share_mode,
>+                       const WAVEFORMATEXTENSIBLE* format);
>+
>+// For a shared-mode stream, the audio engine periodically processes the
>+// data in the endpoint buffer at the period obtained in |device_period|.
>+// For an exclusive mode stream, |device_period| corresponds to the minimum
>+// time interval between successive processing by the endpoint device.
>+// This period plus the stream latency between the buffer and endpoint device
>+// represents the minimum possible latency that an audio application can
>+// achieve. The time in |device_period| is expressed in 100-nanosecond units.
>+HRESULT GetDevicePeriod(IAudioClient* client,
>+                        AUDCLNT_SHAREMODE share_mode,
>+                        REFERENCE_TIME* device_period);
>+
>+// Get the preferred audio parameters for the given |device_id|. The acquired
>+// values should only be utilized for shared mode streamed since there are no
>+// preferred settings for an exclusive mode stream.
>+HRESULT GetPreferredAudioParameters(const std::string& device_id,
>+                                    bool is_output_device,
>+                                    webrtc::AudioParameters* params);
>+
>+HRESULT GetPreferredAudioParameters(IAudioClient* client,
>+                                    webrtc::AudioParameters* params);
>+
>+// After activating an IAudioClient interface on an audio endpoint device,
>+// the client must initialize it once, and only once, to initialize the audio
>+// stream between the client and the device. In shared mode, the client
>+// connects indirectly through the audio engine which does the mixing.
>+// If a valid event is provided in |event_handle|, the client will be
>+// initialized for event-driven buffer handling. If |event_handle| is set to
>+// nullptr, event-driven buffer handling is not utilized.
>+// The output parameter |endpoint_buffer_size| contains the size of the
>+// endpoint buffer and it is expressed as the number of audio frames the
>+// buffer can hold.
>+// TODO(henrika):
>+// - use IAudioClient2::SetClientProperties before calling this method
>+// - IAudioClient::Initialize(your_format, AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM
>+//                            AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY)
>+HRESULT SharedModeInitialize(IAudioClient* client,
>+                             const WAVEFORMATEXTENSIBLE* format,
>+                             HANDLE event_handle,
>+                             uint32_t* endpoint_buffer_size);
>+
>+// Creates an IAudioRenderClient client for an existing IAudioClient given by
>+// |client|. The IAudioRenderClient interface enables a client to write
>+// output data to a rendering endpoint buffer. The methods in this interface
>+// manage the movement of data packets that contain audio-rendering data.
>+Microsoft::WRL::ComPtr<IAudioRenderClient> CreateRenderClient(
>+    IAudioClient* client);
>+
>+// Creates an IAudioCaptureClient client for an existing IAudioClient given by
>+// |client|. The IAudioCaptureClient interface enables a client to read
>+// input data from a capture endpoint buffer. The methods in this interface
>+// manage the movement of data packets that contain capture data.
>+Microsoft::WRL::ComPtr<IAudioCaptureClient> CreateCaptureClient(
>+    IAudioClient* client);
>+
>+// Creates an IAudioClock interface for an existing IAudioClient given by
>+// |client|. The IAudioClock interface enables a client to monitor a stream's
>+// data rate and the current position in the stream.
>+Microsoft::WRL::ComPtr<IAudioClock> CreateAudioClock(IAudioClient* client);
>+
>+// Creates an ISimpleAudioVolume interface for an existing IAudioClient given by
>+// |client|. This interface enables a client to control the master volume level
>+// of an active audio session.
>+Microsoft::WRL::ComPtr<ISimpleAudioVolume> CreateSimpleAudioVolume(
>+    IAudioClient* client);
>+
>+// Fills up the endpoint rendering buffer with silence for an existing
>+// IAudioClient given by |client| and a corresponding IAudioRenderClient
>+// given by |render_client|.
>+bool FillRenderEndpointBufferWithSilence(IAudioClient* client,
>+                                         IAudioRenderClient* render_client);
>+// Transforms a WAVEFORMATEXTENSIBLE struct to a human-readable string.
>+std::string WaveFormatExToString(const WAVEFORMATEXTENSIBLE* format);
>+
>+// Converts Windows internal REFERENCE_TIME (100 nanosecond units) into
>+// generic webrtc::TimeDelta which then can be converted to any time unit.
>+webrtc::TimeDelta ReferenceTimeToTimeDelta(REFERENCE_TIME time);
>+
>+// Converts a COM error into a human-readable string.
>+std::string ErrorToString(const _com_error& error);
>+
>+}  // namespace core_audio_utility
>+}  // namespace webrtc_win
>+}  // namespace webrtc
>+
>+#endif  //  MODULES_AUDIO_DEVICE_WIN_CORE_AUDIO_UTILITY_WIN_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/win/core_audio_utility_win_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/win/core_audio_utility_win_unittest.cc
>new file mode 100644
>index 00000000000..3944cb002de
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/win/core_audio_utility_win_unittest.cc
>@@ -0,0 +1,694 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_device/win/core_audio_utility_win.h"
>+#include "rtc_base/arraysize.h"
>+#include "rtc_base/logging.h"
>+#include "rtc_base/win/windows_version.h"
>+#include "test/gtest.h"
>+
>+#include "system_wrappers/include/sleep.h"
>+
>+using Microsoft::WRL::ComPtr;
>+using webrtc::AudioDeviceName;
>+
>+namespace webrtc {
>+namespace webrtc_win {
>+namespace {
>+
>+#define ABORT_TEST_IF_NOT(requirements_satisfied)                        \
>+  do {                                                                   \
>+    bool fail = false;                                                   \
>+    if (ShouldAbortTest(requirements_satisfied, #requirements_satisfied, \
>+                        &fail)) {                                        \
>+      if (fail)                                                          \
>+        FAIL();                                                          \
>+      else                                                               \
>+        return;                                                          \
>+    }                                                                    \
>+  } while (false)
>+
>+bool ShouldAbortTest(bool requirements_satisfied,
>+                     const char* requirements_expression,
>+                     bool* should_fail) {
>+  if (!requirements_satisfied) {
>+    RTC_LOG(LS_ERROR) << "Requirement(s) not satisfied ("
>+                      << requirements_expression << ")";
>+    // TODO(henrika): improve hard-coded condition to determine if test should
>+    // fail or be ignored. Could use e.g. a command-line argument here to
>+    // determine if the test should fail or not.
>+    *should_fail = false;
>+    return true;
>+  }
>+  *should_fail = false;
>+  return false;
>+}
>+
>+}  // namespace
>+
>+// CoreAudioUtilityWinTest test fixture.
>+class CoreAudioUtilityWinTest : public ::testing::Test {
>+ protected:
>+  CoreAudioUtilityWinTest()
>+      : com_init_(webrtc_win::ScopedCOMInitializer::kMTA) {
>+    // We must initialize the COM library on a thread before we calling any of
>+    // the library functions. All COM functions will return CO_E_NOTINITIALIZED
>+    // otherwise.
>+    EXPECT_TRUE(com_init_.Succeeded());
>+
>+    // Configure logging.
>+    rtc::LogMessage::LogToDebug(rtc::LS_INFO);
>+    rtc::LogMessage::LogTimestamps();
>+    rtc::LogMessage::LogThreads();
>+  }
>+
>+  virtual ~CoreAudioUtilityWinTest() {}
>+
>+  bool DevicesAvailable() {
>+    return core_audio_utility::IsSupported() &&
>+           core_audio_utility::NumberOfActiveDevices(eCapture) > 0 &&
>+           core_audio_utility::NumberOfActiveDevices(eRender) > 0;
>+  }
>+
>+ private:
>+  ScopedCOMInitializer com_init_;
>+};
>+
>+TEST_F(CoreAudioUtilityWinTest, NumberOfActiveDevices) {
>+  ABORT_TEST_IF_NOT(DevicesAvailable());
>+  int render_devices = core_audio_utility::NumberOfActiveDevices(eRender);
>+  EXPECT_GT(render_devices, 0);
>+  int capture_devices = core_audio_utility::NumberOfActiveDevices(eCapture);
>+  EXPECT_GT(capture_devices, 0);
>+  int total_devices = core_audio_utility::NumberOfActiveDevices(eAll);
>+  EXPECT_EQ(total_devices, render_devices + capture_devices);
>+}
>+
>+TEST_F(CoreAudioUtilityWinTest, CreateDeviceEnumerator) {
>+  ABORT_TEST_IF_NOT(DevicesAvailable());
>+  ComPtr<IMMDeviceEnumerator> enumerator =
>+      core_audio_utility::CreateDeviceEnumerator();
>+  EXPECT_TRUE(enumerator.Get());
>+}
>+
>+TEST_F(CoreAudioUtilityWinTest, GetDefaultInputDeviceID) {
>+  ABORT_TEST_IF_NOT(DevicesAvailable());
>+  std::string default_device_id = core_audio_utility::GetDefaultInputDeviceID();
>+  EXPECT_FALSE(default_device_id.empty());
>+}
>+
>+TEST_F(CoreAudioUtilityWinTest, GetDefaultOutputDeviceID) {
>+  ABORT_TEST_IF_NOT(DevicesAvailable());
>+  std::string default_device_id =
>+      core_audio_utility::GetDefaultOutputDeviceID();
>+  EXPECT_FALSE(default_device_id.empty());
>+}
>+
>+TEST_F(CoreAudioUtilityWinTest, GetCommunicationsInputDeviceID) {
>+  ABORT_TEST_IF_NOT(DevicesAvailable());
>+  std::string default_device_id =
>+      core_audio_utility::GetCommunicationsInputDeviceID();
>+  EXPECT_FALSE(default_device_id.empty());
>+}
>+
>+TEST_F(CoreAudioUtilityWinTest, GetCommunicationsOutputDeviceID) {
>+  ABORT_TEST_IF_NOT(DevicesAvailable());
>+  std::string default_device_id =
>+      core_audio_utility::GetCommunicationsOutputDeviceID();
>+  EXPECT_FALSE(default_device_id.empty());
>+}
>+
>+TEST_F(CoreAudioUtilityWinTest, CreateDefaultDevice) {
>+  ABORT_TEST_IF_NOT(DevicesAvailable());
>+
>+  struct {
>+    EDataFlow flow;
>+    ERole role;
>+  } data[] = {{eRender, eConsole},         {eRender, eCommunications},
>+              {eRender, eMultimedia},      {eCapture, eConsole},
>+              {eCapture, eCommunications}, {eCapture, eMultimedia}};
>+
>+  // Create default devices for all flow/role combinations above.
>+  ComPtr<IMMDevice> audio_device;
>+  for (size_t i = 0; i < arraysize(data); ++i) {
>+    audio_device = core_audio_utility::CreateDevice(
>+        AudioDeviceName::kDefaultDeviceId, data[i].flow, data[i].role);
>+    EXPECT_TRUE(audio_device.Get());
>+    EXPECT_EQ(data[i].flow,
>+              core_audio_utility::GetDataFlow(audio_device.Get()));
>+  }
>+
>+  // Only eRender and eCapture are allowed as flow parameter.
>+  audio_device = core_audio_utility::CreateDevice(
>+      AudioDeviceName::kDefaultDeviceId, eAll, eConsole);
>+  EXPECT_FALSE(audio_device.Get());
>+}
>+
>+TEST_F(CoreAudioUtilityWinTest, CreateDevice) {
>+  ABORT_TEST_IF_NOT(DevicesAvailable());
>+
>+  // Get name and ID of default device used for playback.
>+  ComPtr<IMMDevice> default_render_device = core_audio_utility::CreateDevice(
>+      AudioDeviceName::kDefaultDeviceId, eRender, eConsole);
>+  AudioDeviceName default_render_name =
>+      core_audio_utility::GetDeviceName(default_render_device.Get());
>+  EXPECT_TRUE(default_render_name.IsValid());
>+
>+  // Use the unique ID as input to CreateDevice() and create a corresponding
>+  // IMMDevice. The data-flow direction and role parameters are ignored for
>+  // this scenario.
>+  ComPtr<IMMDevice> audio_device = core_audio_utility::CreateDevice(
>+      default_render_name.unique_id, EDataFlow(), ERole());
>+  EXPECT_TRUE(audio_device.Get());
>+
>+  // Verify that the two IMMDevice interfaces represents the same endpoint
>+  // by comparing their unique IDs.
>+  AudioDeviceName device_name =
>+      core_audio_utility::GetDeviceName(audio_device.Get());
>+  EXPECT_EQ(default_render_name.unique_id, device_name.unique_id);
>+}
>+
>+TEST_F(CoreAudioUtilityWinTest, GetDefaultDeviceName) {
>+  ABORT_TEST_IF_NOT(DevicesAvailable());
>+
>+  struct {
>+    EDataFlow flow;
>+    ERole role;
>+  } data[] = {{eRender, eConsole},
>+              {eRender, eCommunications},
>+              {eCapture, eConsole},
>+              {eCapture, eCommunications}};
>+
>+  // Get name and ID of default devices for all flow/role combinations above.
>+  ComPtr<IMMDevice> audio_device;
>+  AudioDeviceName device_name;
>+  for (size_t i = 0; i < arraysize(data); ++i) {
>+    audio_device = core_audio_utility::CreateDevice(
>+        AudioDeviceName::kDefaultDeviceId, data[i].flow, data[i].role);
>+    device_name = core_audio_utility::GetDeviceName(audio_device.Get());
>+    EXPECT_TRUE(device_name.IsValid());
>+  }
>+}
>+
>+TEST_F(CoreAudioUtilityWinTest, GetFriendlyName) {
>+  ABORT_TEST_IF_NOT(DevicesAvailable());
>+
>+  // Get name and ID of default device used for recording.
>+  ComPtr<IMMDevice> audio_device = core_audio_utility::CreateDevice(
>+      AudioDeviceName::kDefaultDeviceId, eCapture, eConsole);
>+  AudioDeviceName device_name =
>+      core_audio_utility::GetDeviceName(audio_device.Get());
>+  EXPECT_TRUE(device_name.IsValid());
>+
>+  // Use unique ID as input to GetFriendlyName() and compare the result
>+  // with the already obtained friendly name for the default capture device.
>+  std::string friendly_name = core_audio_utility::GetFriendlyName(
>+      device_name.unique_id, eCapture, eConsole);
>+  EXPECT_EQ(friendly_name, device_name.device_name);
>+
>+  // Same test as above but for playback.
>+  audio_device = core_audio_utility::CreateDevice(
>+      AudioDeviceName::kDefaultDeviceId, eRender, eConsole);
>+  device_name = core_audio_utility::GetDeviceName(audio_device.Get());
>+  friendly_name = core_audio_utility::GetFriendlyName(device_name.unique_id,
>+                                                      eRender, eConsole);
>+  EXPECT_EQ(friendly_name, device_name.device_name);
>+}
>+
>+TEST_F(CoreAudioUtilityWinTest, GetInputDeviceNames) {
>+  ABORT_TEST_IF_NOT(DevicesAvailable());
>+
>+  webrtc::AudioDeviceNames device_names;
>+  EXPECT_TRUE(core_audio_utility::GetInputDeviceNames(&device_names));
>+  // Number of elements in the list should be two more than the number of
>+  // active devices since we always add default and default communication
>+  // devices on index 0 and 1.
>+  EXPECT_EQ(static_cast<int>(device_names.size()),
>+            2 + core_audio_utility::NumberOfActiveDevices(eCapture));
>+}
>+
>+TEST_F(CoreAudioUtilityWinTest, GetOutputDeviceNames) {
>+  ABORT_TEST_IF_NOT(DevicesAvailable());
>+
>+  webrtc::AudioDeviceNames device_names;
>+  EXPECT_TRUE(core_audio_utility::GetOutputDeviceNames(&device_names));
>+  // Number of elements in the list should be two more than the number of
>+  // active devices since we always add default and default communication
>+  // devices on index 0 and 1.
>+  EXPECT_EQ(static_cast<int>(device_names.size()),
>+            2 + core_audio_utility::NumberOfActiveDevices(eRender));
>+}
>+
>+TEST_F(CoreAudioUtilityWinTest, CreateSessionManager2) {
>+  ABORT_TEST_IF_NOT(DevicesAvailable() &&
>+                    rtc::rtc_win::GetVersion() >= rtc::rtc_win::VERSION_WIN7);
>+
>+  EDataFlow data_flow[] = {eRender, eCapture};
>+
>+  // Obtain reference to an IAudioSessionManager2 interface for a default audio
>+  // endpoint device specified by two different data flows and the |eConsole|
>+  // role.
>+  for (size_t i = 0; i < arraysize(data_flow); ++i) {
>+    ComPtr<IMMDevice> device(core_audio_utility::CreateDevice(
>+        AudioDeviceName::kDefaultDeviceId, data_flow[i], eConsole));
>+    EXPECT_TRUE(device.Get());
>+    ComPtr<IAudioSessionManager2> session_manager =
>+        core_audio_utility::CreateSessionManager2(device.Get());
>+    EXPECT_TRUE(session_manager.Get());
>+  }
>+}
>+
>+TEST_F(CoreAudioUtilityWinTest, CreateSessionEnumerator) {
>+  ABORT_TEST_IF_NOT(DevicesAvailable() &&
>+                    rtc::rtc_win::GetVersion() >= rtc::rtc_win::VERSION_WIN7);
>+
>+  EDataFlow data_flow[] = {eRender, eCapture};
>+
>+  // Obtain reference to an IAudioSessionEnumerator interface for a default
>+  // audio endpoint device specified by two different data flows and the
>+  // |eConsole| role.
>+  for (size_t i = 0; i < arraysize(data_flow); ++i) {
>+    ComPtr<IMMDevice> device(core_audio_utility::CreateDevice(
>+        AudioDeviceName::kDefaultDeviceId, data_flow[i], eConsole));
>+    EXPECT_TRUE(device.Get());
>+    ComPtr<IAudioSessionEnumerator> session_enumerator =
>+        core_audio_utility::CreateSessionEnumerator(device.Get());
>+    EXPECT_TRUE(session_enumerator.Get());
>+
>+    // Perform a sanity test of the interface by asking for the total number
>+    // of audio sessions that are open on the audio device. Note that, we do
>+    // not check if the session is active or not.
>+    int session_count = 0;
>+    EXPECT_TRUE(SUCCEEDED(session_enumerator->GetCount(&session_count)));
>+    EXPECT_GE(session_count, 0);
>+  }
>+}
>+
>+TEST_F(CoreAudioUtilityWinTest, NumberOfActiveSessions) {
>+  ABORT_TEST_IF_NOT(DevicesAvailable() &&
>+                    rtc::rtc_win::GetVersion() >= rtc::rtc_win::VERSION_WIN7);
>+
>+  EDataFlow data_flow[] = {eRender, eCapture};
>+
>+  // Count number of active audio session for a default audio endpoint device
>+  // specified by two different data flows and the |eConsole| role.
>+  // Ensure that the number of active audio sessions is less than or equal to
>+  // the total number of audio sessions on that same device.
>+  for (size_t i = 0; i < arraysize(data_flow); ++i) {
>+    // Create an audio endpoint device.
>+    ComPtr<IMMDevice> device(core_audio_utility::CreateDevice(
>+        AudioDeviceName::kDefaultDeviceId, data_flow[i], eConsole));
>+    EXPECT_TRUE(device.Get());
>+
>+    // Ask for total number of audio sessions on the created device.
>+    ComPtr<IAudioSessionEnumerator> session_enumerator =
>+        core_audio_utility::CreateSessionEnumerator(device.Get());
>+    EXPECT_TRUE(session_enumerator.Get());
>+    int total_session_count = 0;
>+    EXPECT_TRUE(SUCCEEDED(session_enumerator->GetCount(&total_session_count)));
>+    EXPECT_GE(total_session_count, 0);
>+
>+    // Use NumberOfActiveSessions and get number of active audio sessions.
>+    int active_session_count =
>+        core_audio_utility::NumberOfActiveSessions(device.Get());
>+    EXPECT_LE(active_session_count, total_session_count);
>+  }
>+}
>+
>+TEST_F(CoreAudioUtilityWinTest, CreateClient) {
>+  ABORT_TEST_IF_NOT(DevicesAvailable());
>+
>+  EDataFlow data_flow[] = {eRender, eCapture};
>+
>+  // Obtain reference to an IAudioClient interface for a default audio endpoint
>+  // device specified by two different data flows and the |eConsole| role.
>+  for (size_t i = 0; i < arraysize(data_flow); ++i) {
>+    ComPtr<IAudioClient> client = core_audio_utility::CreateClient(
>+        AudioDeviceName::kDefaultDeviceId, data_flow[i], eConsole);
>+    EXPECT_TRUE(client.Get());
>+  }
>+}
>+
>+TEST_F(CoreAudioUtilityWinTest, CreateClient2) {
>+  ABORT_TEST_IF_NOT(DevicesAvailable() &&
>+                    rtc::rtc_win::GetVersion() >= rtc::rtc_win::VERSION_WIN10);
>+
>+  EDataFlow data_flow[] = {eRender, eCapture};
>+
>+  // Obtain reference to an IAudioClient2 interface for a default audio endpoint
>+  // device specified by two different data flows and the |eConsole| role.
>+  for (size_t i = 0; i < arraysize(data_flow); ++i) {
>+    ComPtr<IAudioClient2> client = core_audio_utility::CreateClient2(
>+        AudioDeviceName::kDefaultDeviceId, data_flow[i], eConsole);
>+    EXPECT_TRUE(client.Get());
>+  }
>+}
>+
>+TEST_F(CoreAudioUtilityWinTest, SetClientProperties) {
>+  ABORT_TEST_IF_NOT(DevicesAvailable());
>+
>+  ComPtr<IAudioClient2> client = core_audio_utility::CreateClient2(
>+      AudioDeviceName::kDefaultDeviceId, eRender, eConsole);
>+  EXPECT_TRUE(client.Get());
>+
>+  EXPECT_TRUE(SUCCEEDED(core_audio_utility::SetClientProperties(client.Get())));
>+}
>+
>+TEST_F(CoreAudioUtilityWinTest, GetSharedModeMixFormat) {
>+  ABORT_TEST_IF_NOT(DevicesAvailable());
>+
>+  ComPtr<IAudioClient> client = core_audio_utility::CreateClient(
>+      AudioDeviceName::kDefaultDeviceId, eRender, eConsole);
>+  EXPECT_TRUE(client.Get());
>+
>+  // Perform a simple sanity test of the acquired format structure.
>+  WAVEFORMATPCMEX format;
>+  EXPECT_TRUE(SUCCEEDED(
>+      core_audio_utility::GetSharedModeMixFormat(client.Get(), &format)));
>+  EXPECT_GE(format.Format.nChannels, 1);
>+  EXPECT_GE(format.Format.nSamplesPerSec, 8000u);
>+  EXPECT_GE(format.Format.wBitsPerSample, 16);
>+  EXPECT_GE(format.Samples.wValidBitsPerSample, 16);
>+  EXPECT_EQ(format.Format.wFormatTag, WAVE_FORMAT_EXTENSIBLE);
>+}
>+
>+TEST_F(CoreAudioUtilityWinTest, IsFormatSupported) {
>+  ABORT_TEST_IF_NOT(DevicesAvailable());
>+
>+  // Create a default render client.
>+  ComPtr<IAudioClient> client = core_audio_utility::CreateClient(
>+      AudioDeviceName::kDefaultDeviceId, eRender, eConsole);
>+  EXPECT_TRUE(client.Get());
>+
>+  // Get the default, shared mode, mixing format.
>+  WAVEFORMATEXTENSIBLE format;
>+  EXPECT_TRUE(SUCCEEDED(
>+      core_audio_utility::GetSharedModeMixFormat(client.Get(), &format)));
>+
>+  // In shared mode, the audio engine always supports the mix format.
>+  EXPECT_TRUE(core_audio_utility::IsFormatSupported(
>+      client.Get(), AUDCLNT_SHAREMODE_SHARED, &format));
>+
>+  // Use an invalid format and verify that it is not supported.
>+  format.Format.nSamplesPerSec += 1;
>+  EXPECT_FALSE(core_audio_utility::IsFormatSupported(
>+      client.Get(), AUDCLNT_SHAREMODE_SHARED, &format));
>+}
>+
>+TEST_F(CoreAudioUtilityWinTest, GetDevicePeriod) {
>+  ABORT_TEST_IF_NOT(DevicesAvailable());
>+
>+  EDataFlow data_flow[] = {eRender, eCapture};
>+
>+  // Verify that the device periods are valid for the default render and
>+  // capture devices.
>+  for (size_t i = 0; i < arraysize(data_flow); ++i) {
>+    ComPtr<IAudioClient> client;
>+    REFERENCE_TIME shared_time_period = 0;
>+    REFERENCE_TIME exclusive_time_period = 0;
>+    client = core_audio_utility::CreateClient(AudioDeviceName::kDefaultDeviceId,
>+                                              data_flow[i], eConsole);
>+    EXPECT_TRUE(client.Get());
>+    EXPECT_TRUE(SUCCEEDED(core_audio_utility::GetDevicePeriod(
>+        client.Get(), AUDCLNT_SHAREMODE_SHARED, &shared_time_period)));
>+    EXPECT_GT(shared_time_period, 0);
>+    EXPECT_TRUE(SUCCEEDED(core_audio_utility::GetDevicePeriod(
>+        client.Get(), AUDCLNT_SHAREMODE_EXCLUSIVE, &exclusive_time_period)));
>+    EXPECT_GT(exclusive_time_period, 0);
>+    EXPECT_LE(exclusive_time_period, shared_time_period);
>+  }
>+}
>+
>+TEST_F(CoreAudioUtilityWinTest, GetPreferredAudioParameters) {
>+  ABORT_TEST_IF_NOT(DevicesAvailable());
>+
>+  EDataFlow data_flow[] = {eRender, eCapture};
>+
>+  // Verify that the preferred audio parameters are OK for the default render
>+  // and capture devices.
>+  for (size_t i = 0; i < arraysize(data_flow); ++i) {
>+    webrtc::AudioParameters params;
>+    EXPECT_TRUE(SUCCEEDED(core_audio_utility::GetPreferredAudioParameters(
>+        AudioDeviceName::kDefaultDeviceId, data_flow[i] == eRender, &params)));
>+    EXPECT_TRUE(params.is_valid());
>+    EXPECT_TRUE(params.is_complete());
>+  }
>+
>+  // Verify that the preferred audio parameters are OK for the default
>+  // communication devices.
>+  for (size_t i = 0; i < arraysize(data_flow); ++i) {
>+    webrtc::AudioParameters params;
>+    EXPECT_TRUE(SUCCEEDED(core_audio_utility::GetPreferredAudioParameters(
>+        AudioDeviceName::kDefaultCommunicationsDeviceId,
>+        data_flow[i] == eRender, &params)));
>+    EXPECT_TRUE(params.is_valid());
>+    EXPECT_TRUE(params.is_complete());
>+  }
>+}
>+
>+TEST_F(CoreAudioUtilityWinTest, SharedModeInitialize) {
>+  ABORT_TEST_IF_NOT(DevicesAvailable());
>+
>+  ComPtr<IAudioClient> client;
>+  client = core_audio_utility::CreateClient(AudioDeviceName::kDefaultDeviceId,
>+                                            eRender, eConsole);
>+  EXPECT_TRUE(client.Get());
>+
>+  WAVEFORMATPCMEX format;
>+  EXPECT_TRUE(SUCCEEDED(
>+      core_audio_utility::GetSharedModeMixFormat(client.Get(), &format)));
>+
>+  // Perform a shared-mode initialization without event-driven buffer handling.
>+  uint32_t endpoint_buffer_size = 0;
>+  HRESULT hr = core_audio_utility::SharedModeInitialize(
>+      client.Get(), &format, nullptr, &endpoint_buffer_size);
>+  EXPECT_TRUE(SUCCEEDED(hr));
>+  EXPECT_GT(endpoint_buffer_size, 0u);
>+
>+  // It is only possible to create a client once.
>+  hr = core_audio_utility::SharedModeInitialize(client.Get(), &format, nullptr,
>+                                                &endpoint_buffer_size);
>+  EXPECT_FALSE(SUCCEEDED(hr));
>+  EXPECT_EQ(hr, AUDCLNT_E_ALREADY_INITIALIZED);
>+
>+  // Verify that it is possible to reinitialize the client after releasing it
>+  // and then creating a new client.
>+  client = core_audio_utility::CreateClient(AudioDeviceName::kDefaultDeviceId,
>+                                            eRender, eConsole);
>+  EXPECT_TRUE(client.Get());
>+  hr = core_audio_utility::SharedModeInitialize(client.Get(), &format, nullptr,
>+                                                &endpoint_buffer_size);
>+  EXPECT_TRUE(SUCCEEDED(hr));
>+  EXPECT_GT(endpoint_buffer_size, 0u);
>+
>+  // Use a non-supported format and verify that initialization fails.
>+  // A simple way to emulate an invalid format is to use the shared-mode
>+  // mixing format and modify the preferred sample rate.
>+  client = core_audio_utility::CreateClient(AudioDeviceName::kDefaultDeviceId,
>+                                            eRender, eConsole);
>+  EXPECT_TRUE(client.Get());
>+  format.Format.nSamplesPerSec = format.Format.nSamplesPerSec + 1;
>+  EXPECT_FALSE(core_audio_utility::IsFormatSupported(
>+      client.Get(), AUDCLNT_SHAREMODE_SHARED, &format));
>+  hr = core_audio_utility::SharedModeInitialize(client.Get(), &format, nullptr,
>+                                                &endpoint_buffer_size);
>+  EXPECT_TRUE(FAILED(hr));
>+  EXPECT_EQ(hr, E_INVALIDARG);
>+
>+  // Finally, perform a shared-mode initialization using event-driven buffer
>+  // handling. The event handle will be signaled when an audio buffer is ready
>+  // to be processed by the client (not verified here). The event handle should
>+  // be in the non-signaled state.
>+  ScopedHandle event_handle(::CreateEvent(nullptr, TRUE, FALSE, nullptr));
>+  client = core_audio_utility::CreateClient(AudioDeviceName::kDefaultDeviceId,
>+                                            eRender, eConsole);
>+  EXPECT_TRUE(client.Get());
>+  EXPECT_TRUE(SUCCEEDED(
>+      core_audio_utility::GetSharedModeMixFormat(client.Get(), &format)));
>+  EXPECT_TRUE(core_audio_utility::IsFormatSupported(
>+      client.Get(), AUDCLNT_SHAREMODE_SHARED, &format));
>+  hr = core_audio_utility::SharedModeInitialize(
>+      client.Get(), &format, event_handle, &endpoint_buffer_size);
>+  EXPECT_TRUE(SUCCEEDED(hr));
>+  EXPECT_GT(endpoint_buffer_size, 0u);
>+}
>+
>+TEST_F(CoreAudioUtilityWinTest, CreateRenderAndCaptureClients) {
>+  ABORT_TEST_IF_NOT(DevicesAvailable());
>+
>+  EDataFlow data_flow[] = {eRender, eCapture};
>+
>+  WAVEFORMATPCMEX format;
>+  uint32_t endpoint_buffer_size = 0;
>+
>+  for (size_t i = 0; i < arraysize(data_flow); ++i) {
>+    ComPtr<IAudioClient> client;
>+    ComPtr<IAudioRenderClient> render_client;
>+    ComPtr<IAudioCaptureClient> capture_client;
>+
>+    // Create a default client for the given data-flow direction.
>+    client = core_audio_utility::CreateClient(AudioDeviceName::kDefaultDeviceId,
>+                                              data_flow[i], eConsole);
>+    EXPECT_TRUE(client.Get());
>+    EXPECT_TRUE(SUCCEEDED(
>+        core_audio_utility::GetSharedModeMixFormat(client.Get(), &format)));
>+    if (data_flow[i] == eRender) {
>+      // It is not possible to create a render client using an unitialized
>+      // client interface.
>+      render_client = core_audio_utility::CreateRenderClient(client.Get());
>+      EXPECT_FALSE(render_client.Get());
>+
>+      // Do a proper initialization and verify that it works this time.
>+      core_audio_utility::SharedModeInitialize(client.Get(), &format, nullptr,
>+                                               &endpoint_buffer_size);
>+      render_client = core_audio_utility::CreateRenderClient(client.Get());
>+      EXPECT_TRUE(render_client.Get());
>+      EXPECT_GT(endpoint_buffer_size, 0u);
>+    } else if (data_flow[i] == eCapture) {
>+      // It is not possible to create a capture client using an unitialized
>+      // client interface.
>+      capture_client = core_audio_utility::CreateCaptureClient(client.Get());
>+      EXPECT_FALSE(capture_client.Get());
>+
>+      // Do a proper initialization and verify that it works this time.
>+      core_audio_utility::SharedModeInitialize(client.Get(), &format, nullptr,
>+                                               &endpoint_buffer_size);
>+      capture_client = core_audio_utility::CreateCaptureClient(client.Get());
>+      EXPECT_TRUE(capture_client.Get());
>+      EXPECT_GT(endpoint_buffer_size, 0u);
>+    }
>+  }
>+}
>+
>+TEST_F(CoreAudioUtilityWinTest, CreateAudioClock) {
>+  ABORT_TEST_IF_NOT(DevicesAvailable());
>+
>+  EDataFlow data_flow[] = {eRender, eCapture};
>+
>+  WAVEFORMATPCMEX format;
>+  uint32_t endpoint_buffer_size = 0;
>+
>+  for (size_t i = 0; i < arraysize(data_flow); ++i) {
>+    ComPtr<IAudioClient> client;
>+    ComPtr<IAudioClock> audio_clock;
>+
>+    // Create a default client for the given data-flow direction.
>+    client = core_audio_utility::CreateClient(AudioDeviceName::kDefaultDeviceId,
>+                                              data_flow[i], eConsole);
>+    EXPECT_TRUE(client.Get());
>+    EXPECT_TRUE(SUCCEEDED(
>+        core_audio_utility::GetSharedModeMixFormat(client.Get(), &format)));
>+
>+    // It is not possible to create an audio clock using an unitialized client
>+    // interface.
>+    audio_clock = core_audio_utility::CreateAudioClock(client.Get());
>+    EXPECT_FALSE(audio_clock.Get());
>+
>+    // Do a proper initialization and verify that it works this time.
>+    core_audio_utility::SharedModeInitialize(client.Get(), &format, nullptr,
>+                                             &endpoint_buffer_size);
>+    audio_clock = core_audio_utility::CreateAudioClock(client.Get());
>+    EXPECT_TRUE(audio_clock.Get());
>+    EXPECT_GT(endpoint_buffer_size, 0u);
>+
>+    // Use the audio clock and verify that querying the device frequency works.
>+    UINT64 frequency = 0;
>+    EXPECT_TRUE(SUCCEEDED(audio_clock->GetFrequency(&frequency)));
>+    EXPECT_GT(frequency, 0u);
>+  }
>+}
>+
>+TEST_F(CoreAudioUtilityWinTest, CreateSimpleAudioVolume) {
>+  ABORT_TEST_IF_NOT(DevicesAvailable());
>+
>+  EDataFlow data_flow[] = {eRender, eCapture};
>+
>+  WAVEFORMATPCMEX format;
>+  uint32_t endpoint_buffer_size = 0;
>+
>+  for (size_t i = 0; i < arraysize(data_flow); ++i) {
>+    ComPtr<IAudioClient> client;
>+    ComPtr<ISimpleAudioVolume> simple_audio_volume;
>+
>+    // Create a default client for the given data-flow direction.
>+    client = core_audio_utility::CreateClient(AudioDeviceName::kDefaultDeviceId,
>+                                              data_flow[i], eConsole);
>+    EXPECT_TRUE(client.Get());
>+    EXPECT_TRUE(SUCCEEDED(
>+        core_audio_utility::GetSharedModeMixFormat(client.Get(), &format)));
>+
>+    // It is not possible to create an audio volume using an uninitialized
>+    // client interface.
>+    simple_audio_volume =
>+        core_audio_utility::CreateSimpleAudioVolume(client.Get());
>+    EXPECT_FALSE(simple_audio_volume.Get());
>+
>+    // Do a proper initialization and verify that it works this time.
>+    core_audio_utility::SharedModeInitialize(client.Get(), &format, nullptr,
>+                                             &endpoint_buffer_size);
>+    simple_audio_volume =
>+        core_audio_utility::CreateSimpleAudioVolume(client.Get());
>+    EXPECT_TRUE(simple_audio_volume.Get());
>+    EXPECT_GT(endpoint_buffer_size, 0u);
>+
>+    // Use the audio volume interface and validate that it works. The volume
>+    // level should be value in the range 0.0 to 1.0 at first call.
>+    float volume = 0.0;
>+    EXPECT_TRUE(SUCCEEDED(simple_audio_volume->GetMasterVolume(&volume)));
>+    EXPECT_GE(volume, 0.0);
>+    EXPECT_LE(volume, 1.0);
>+
>+    // Next, set a new volume and verify that the setter does its job.
>+    const float target_volume = 0.5;
>+    EXPECT_TRUE(SUCCEEDED(
>+        simple_audio_volume->SetMasterVolume(target_volume, nullptr)));
>+    EXPECT_TRUE(SUCCEEDED(simple_audio_volume->GetMasterVolume(&volume)));
>+    EXPECT_EQ(volume, target_volume);
>+  }
>+}
>+
>+TEST_F(CoreAudioUtilityWinTest, FillRenderEndpointBufferWithSilence) {
>+  ABORT_TEST_IF_NOT(DevicesAvailable());
>+
>+  // Create default clients using the default mixing format for shared mode.
>+  ComPtr<IAudioClient> client(core_audio_utility::CreateClient(
>+      AudioDeviceName::kDefaultDeviceId, eRender, eConsole));
>+  EXPECT_TRUE(client.Get());
>+
>+  WAVEFORMATPCMEX format;
>+  uint32_t endpoint_buffer_size = 0;
>+  EXPECT_TRUE(SUCCEEDED(
>+      core_audio_utility::GetSharedModeMixFormat(client.Get(), &format)));
>+  core_audio_utility::SharedModeInitialize(client.Get(), &format, nullptr,
>+                                           &endpoint_buffer_size);
>+  EXPECT_GT(endpoint_buffer_size, 0u);
>+
>+  ComPtr<IAudioRenderClient> render_client(
>+      core_audio_utility::CreateRenderClient(client.Get()));
>+  EXPECT_TRUE(render_client.Get());
>+
>+  // The endpoint audio buffer should not be filled up by default after being
>+  // created.
>+  UINT32 num_queued_frames = 0;
>+  client->GetCurrentPadding(&num_queued_frames);
>+  EXPECT_EQ(num_queued_frames, 0u);
>+
>+  // Fill it up with zeros and verify that the buffer is full.
>+  // It is not possible to verify that the actual data consists of zeros
>+  // since we can't access data that has already been sent to the endpoint
>+  // buffer.
>+  EXPECT_TRUE(core_audio_utility::FillRenderEndpointBufferWithSilence(
>+      client.Get(), render_client.Get()));
>+  client->GetCurrentPadding(&num_queued_frames);
>+  EXPECT_EQ(num_queued_frames, endpoint_buffer_size);
>+}
>+
>+}  // namespace webrtc_win
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/BUILD.gn
>index 13183984d7e..da54e7bc973 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/BUILD.gn
>@@ -32,18 +32,25 @@ rtc_static_library("audio_mixer_impl") {
>     "frame_combiner.h",
>   ]
> 
>+  configs += [ "../audio_processing:apm_debug_dump" ]
>+
>   deps = [
>     ":audio_frame_manipulator",
>-    "..:module_api",
>     "../..:webrtc_common",
>-    "../../:typedefs",
>     "../../api:array_view",
>-    "../../api:audio_mixer_api",
>+    "../../api/audio:audio_frame_api",
>+    "../../api/audio:audio_mixer_api",
>     "../../audio/utility:audio_frame_operations",
>+    "../../common_audio",
>     "../../rtc_base:checks",
>     "../../rtc_base:rtc_base_approved",
>     "../../system_wrappers",
>+    "../../system_wrappers:field_trial_api",
>+    "../../system_wrappers:metrics_api",
>     "../audio_processing",
>+    "../audio_processing:apm_logging",
>+    "../audio_processing:audio_frame_view",
>+    "../audio_processing/agc2:fixed_digital",
>   ]
> }
> 
>@@ -59,7 +66,7 @@ rtc_static_library("audio_frame_manipulator") {
>   ]
> 
>   deps = [
>-    "..:module_api",
>+    "../../api/audio:audio_frame_api",
>     "../../audio/utility:audio_frame_operations",
>     "../../rtc_base:checks",
>     "../../rtc_base:rtc_base_approved",
>@@ -79,18 +86,33 @@ if (rtc_include_tests) {
>       "sine_wave_generator.cc",
>       "sine_wave_generator.h",
>     ]
>+
>     deps = [
>       ":audio_frame_manipulator",
>       ":audio_mixer_impl",
>-      "..:module_api",
>       "../../api:array_view",
>-      "../../api:audio_mixer_api",
>+      "../../api/audio:audio_frame_api",
>+      "../../api/audio:audio_mixer_api",
>       "../../audio/utility:audio_frame_operations",
>       "../../rtc_base:checks",
>       "../../rtc_base:rtc_base_approved",
>-      "../../rtc_base:rtc_task_queue",
>+      "../../rtc_base:rtc_task_queue_for_test",
>       "../../test:test_support",
>-      "//testing/gmock",
>+    ]
>+  }
>+
>+  rtc_executable("audio_mixer_test") {
>+    testonly = true
>+    sources = [
>+      "audio_mixer_test.cc",
>+    ]
>+
>+    deps = [
>+      ":audio_mixer_impl",
>+      "../../api/audio:audio_mixer_api",
>+      "../../common_audio",
>+      "../../rtc_base:rtc_base_approved",
>+      "../../system_wrappers:system_wrappers_default",
>     ]
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/DEPS b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/DEPS
>index 51608ad47d9..dcee4a37d3a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/DEPS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/DEPS
>@@ -5,10 +5,8 @@ include_rules = [
>   "+modules/audio_coding",
>   "+modules/audio_device",
>   "+modules/audio_processing",
>-  "+modules/media_file",
>   "+modules/pacing",
>   "+modules/rtp_rtcp",
>   "+modules/utility",
>   "+system_wrappers",
>-  "+voice_engine",
> ]
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/audio_frame_manipulator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/audio_frame_manipulator.cc
>index 0f5a83fed71..e8a5f759b01 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/audio_frame_manipulator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/audio_frame_manipulator.cc
>@@ -10,7 +10,6 @@
> 
> #include "modules/audio_mixer/audio_frame_manipulator.h"
> #include "audio/utility/audio_frame_operations.h"
>-#include "modules/include/module_common_types.h"
> #include "rtc_base/checks.h"
> 
> namespace webrtc {
>@@ -22,7 +21,8 @@ uint32_t AudioMixerCalculateEnergy(const AudioFrame& audio_frame) {
> 
>   uint32_t energy = 0;
>   const int16_t* frame_data = audio_frame.data();
>-  for (size_t position = 0; position < audio_frame.samples_per_channel_;
>+  for (size_t position = 0;
>+       position < audio_frame.samples_per_channel_ * audio_frame.num_channels_;
>        position++) {
>     // TODO(aleloi): This can overflow. Convert to floats.
>     energy += frame_data[position] * frame_data[position];
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/audio_frame_manipulator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/audio_frame_manipulator.h
>index fe87169676c..dc24caea5a9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/audio_frame_manipulator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/audio_frame_manipulator.h
>@@ -11,7 +11,7 @@
> #ifndef MODULES_AUDIO_MIXER_AUDIO_FRAME_MANIPULATOR_H_
> #define MODULES_AUDIO_MIXER_AUDIO_FRAME_MANIPULATOR_H_
> 
>-#include "modules/include/module_common_types.h"
>+#include "api/audio/audio_frame.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/audio_frame_manipulator_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/audio_frame_manipulator_unittest.cc
>index b3be883ce02..28154a53c68 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/audio_frame_manipulator_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/audio_frame_manipulator_unittest.cc
>@@ -11,7 +11,6 @@
> #include <algorithm>
> 
> #include "modules/audio_mixer/audio_frame_manipulator.h"
>-#include "modules/include/module_common_types.h"
> #include "test/gtest.h"
> 
> namespace webrtc {
>@@ -24,8 +23,8 @@ void FillFrameWithConstants(size_t samples_per_channel,
>   frame->num_channels_ = number_of_channels;
>   frame->samples_per_channel_ = samples_per_channel;
>   int16_t* frame_data = frame->mutable_data();
>-  std::fill(frame_data,
>-            frame_data + samples_per_channel * number_of_channels, value);
>+  std::fill(frame_data, frame_data + samples_per_channel * number_of_channels,
>+            value);
> }
> }  // namespace
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/audio_mixer_impl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/audio_mixer_impl.h
>index 8d49e913e48..e8994a674bc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/audio_mixer_impl.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/audio_mixer_impl.h
>@@ -17,12 +17,9 @@
> #include "api/audio/audio_mixer.h"
> #include "modules/audio_mixer/frame_combiner.h"
> #include "modules/audio_mixer/output_rate_calculator.h"
>-#include "modules/audio_processing/include/audio_processing.h"
>-#include "modules/include/module_common_types.h"
> #include "rtc_base/race_checker.h"
> #include "rtc_base/scoped_ref_ptr.h"
> #include "rtc_base/thread_annotations.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/audio_mixer_impl_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/audio_mixer_impl_unittest.cc
>index 4461e6e038d..57e4526c926 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/audio_mixer_impl_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/audio_mixer_impl_unittest.cc
>@@ -21,8 +21,7 @@
> #include "modules/audio_mixer/default_output_rate_calculator.h"
> #include "rtc_base/bind.h"
> #include "rtc_base/checks.h"
>-#include "rtc_base/event.h"
>-#include "rtc_base/task_queue.h"
>+#include "rtc_base/task_queue_for_test.h"
> #include "test/gmock.h"
> 
> using testing::_;
>@@ -62,7 +61,7 @@ AudioFrame frame_for_mixing;
> 
> }  // namespace
> 
>-class MockMixerAudioSource : public AudioMixer::Source {
>+class MockMixerAudioSource : public testing::NiceMock<AudioMixer::Source> {
>  public:
>   MockMixerAudioSource()
>       : fake_audio_frame_info_(AudioMixer::Source::AudioFrameInfo::kNormal) {
>@@ -189,11 +188,11 @@ TEST(AudioMixer, LargestEnergyVadActiveMixed) {
>     if (i == kAudioSources - 1 ||
>         i < kAudioSources - 1 -
>                 AudioMixerImpl::kMaximumAmountOfMixedAudioSources) {
>-      EXPECT_FALSE(is_mixed) << "Mixing status of AudioSource #" << i
>-                             << " wrong.";
>+      EXPECT_FALSE(is_mixed)
>+          << "Mixing status of AudioSource #" << i << " wrong.";
>     } else {
>-      EXPECT_TRUE(is_mixed) << "Mixing status of AudioSource #" << i
>-                            << " wrong.";
>+      EXPECT_TRUE(is_mixed)
>+          << "Mixing status of AudioSource #" << i << " wrong.";
>     }
>   }
> }
>@@ -222,9 +221,8 @@ TEST(AudioMixer, FrameNotModifiedForSingleParticipant) {
>                &audio_frame);
>   }
> 
>-  EXPECT_EQ(
>-      0,
>-      memcmp(participant.fake_frame()->data(), audio_frame.data(), n_samples));
>+  EXPECT_EQ(0, memcmp(participant.fake_frame()->data(), audio_frame.data(),
>+                      n_samples));
> }
> 
> TEST(AudioMixer, SourceAtNativeRateShouldNeverResample) {
>@@ -372,14 +370,9 @@ TEST(AudioMixer, RampedOutSourcesShouldNotBeMarkedMixed) {
> // This test checks that the initialization and participant addition
> // can be done on a different thread.
> TEST(AudioMixer, ConstructFromOtherThread) {
>-  rtc::TaskQueue init_queue("init");
>+  rtc::test::TaskQueueForTest init_queue("init");
>   rtc::scoped_refptr<AudioMixer> mixer;
>-  rtc::Event event(false, false);
>-  init_queue.PostTask([&mixer, &event]() {
>-    mixer = AudioMixerImpl::Create();
>-    event.Set();
>-  });
>-  event.Wait(rtc::Event::kForever);
>+  init_queue.SendTask([&mixer]() { mixer = AudioMixerImpl::Create(); });
> 
>   MockMixerAudioSource participant;
>   EXPECT_CALL(participant, PreferredSampleRate())
>@@ -387,12 +380,9 @@ TEST(AudioMixer, ConstructFromOtherThread) {
> 
>   ResetFrame(participant.fake_frame());
> 
>-  rtc::TaskQueue participant_queue("participant");
>-  participant_queue.PostTask([&mixer, &event, &participant]() {
>-    mixer->AddSource(&participant);
>-    event.Set();
>-  });
>-  event.Wait(rtc::Event::kForever);
>+  rtc::test::TaskQueueForTest participant_queue("participant");
>+  participant_queue.SendTask(
>+      [&mixer, &participant]() { mixer->AddSource(&participant); });
> 
>   EXPECT_CALL(participant, GetAudioFrameWithInfo(kDefaultSampleRateHz, _))
>       .Times(Exactly(1));
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/audio_mixer_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/audio_mixer_test.cc
>new file mode 100644
>index 00000000000..a2b41ab733c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/audio_mixer_test.cc
>@@ -0,0 +1,174 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <cstring>
>+#include <iostream>
>+#include <vector>
>+
>+#include "api/audio/audio_mixer.h"
>+#include "common_audio/wav_file.h"
>+#include "modules/audio_mixer/audio_mixer_impl.h"
>+#include "modules/audio_mixer/default_output_rate_calculator.h"
>+#include "rtc_base/flags.h"
>+
>+DEFINE_bool(help, false, "Prints this message");
>+DEFINE_int(sampling_rate,
>+           16000,
>+           "Rate at which to mix (all input streams must have this rate)");
>+
>+DEFINE_bool(
>+    stereo,
>+    false,
>+    "Enable stereo (interleaved). Inputs need not be as this parameter.");
>+
>+DEFINE_bool(limiter, true, "Enable limiter.");
>+DEFINE_string(output_file,
>+              "mixed_file.wav",
>+              "File in which to store the mixed result.");
>+DEFINE_string(input_file_1, "", "First input. Default none.");
>+DEFINE_string(input_file_2, "", "Second input. Default none.");
>+DEFINE_string(input_file_3, "", "Third input. Default none.");
>+DEFINE_string(input_file_4, "", "Fourth input. Default none.");
>+
>+namespace webrtc {
>+namespace test {
>+
>+class FilePlayingSource : public AudioMixer::Source {
>+ public:
>+  explicit FilePlayingSource(std::string filename)
>+      : wav_reader_(new WavReader(filename)),
>+        sample_rate_hz_(wav_reader_->sample_rate()),
>+        samples_per_channel_(sample_rate_hz_ / 100),
>+        number_of_channels_(wav_reader_->num_channels()) {}
>+
>+  AudioFrameInfo GetAudioFrameWithInfo(int target_rate_hz,
>+                                       AudioFrame* frame) override {
>+    frame->samples_per_channel_ = samples_per_channel_;
>+    frame->num_channels_ = number_of_channels_;
>+    frame->sample_rate_hz_ = target_rate_hz;
>+
>+    RTC_CHECK_EQ(target_rate_hz, sample_rate_hz_);
>+
>+    const size_t num_to_read = number_of_channels_ * samples_per_channel_;
>+    const size_t num_read =
>+        wav_reader_->ReadSamples(num_to_read, frame->mutable_data());
>+
>+    file_has_ended_ = num_to_read != num_read;
>+    if (file_has_ended_) {
>+      frame->Mute();
>+    }
>+    return file_has_ended_ ? AudioFrameInfo::kMuted : AudioFrameInfo::kNormal;
>+  }
>+
>+  int Ssrc() const override { return 0; }
>+
>+  int PreferredSampleRate() const override { return sample_rate_hz_; }
>+
>+  bool FileHasEnded() const { return file_has_ended_; }
>+
>+  std::string ToString() const {
>+    std::stringstream ss;
>+    ss << "{rate: " << sample_rate_hz_ << ", channels: " << number_of_channels_
>+       << ", samples_tot: " << wav_reader_->num_samples() << "}";
>+    return ss.str();
>+  }
>+
>+ private:
>+  std::unique_ptr<WavReader> wav_reader_;
>+  int sample_rate_hz_;
>+  int samples_per_channel_;
>+  int number_of_channels_;
>+  bool file_has_ended_ = false;
>+};
>+}  // namespace test
>+}  // namespace webrtc
>+
>+namespace {
>+
>+const std::vector<std::string> parse_input_files() {
>+  std::vector<std::string> result;
>+  for (auto* x : {FLAG_input_file_1, FLAG_input_file_2, FLAG_input_file_3,
>+                  FLAG_input_file_4}) {
>+    if (strcmp(x, "") != 0) {
>+      result.push_back(x);
>+    }
>+  }
>+  return result;
>+}
>+}  // namespace
>+
>+int main(int argc, char* argv[]) {
>+  rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, false);
>+  if (FLAG_help) {
>+    rtc::FlagList::Print(nullptr, false);
>+    return 0;
>+  }
>+
>+  rtc::scoped_refptr<webrtc::AudioMixerImpl> mixer(
>+      webrtc::AudioMixerImpl::Create(
>+          std::unique_ptr<webrtc::OutputRateCalculator>(
>+              new webrtc::DefaultOutputRateCalculator()),
>+          FLAG_limiter));
>+
>+  const std::vector<std::string> input_files = parse_input_files();
>+  std::vector<webrtc::test::FilePlayingSource> sources;
>+  const int num_channels = FLAG_stereo ? 2 : 1;
>+  for (auto input_file : input_files) {
>+    sources.emplace_back(input_file);
>+  }
>+
>+  for (auto& source : sources) {
>+    auto error = mixer->AddSource(&source);
>+    RTC_CHECK(error);
>+  }
>+
>+  if (sources.empty()) {
>+    std::cout << "Need at least one source!\n";
>+    rtc::FlagList::Print(nullptr, false);
>+    return 1;
>+  }
>+
>+  const size_t sample_rate = sources[0].PreferredSampleRate();
>+  for (const auto& source : sources) {
>+    RTC_CHECK_EQ(sample_rate, source.PreferredSampleRate());
>+  }
>+
>+  // Print stats.
>+  std::cout << "Limiting is: " << (FLAG_limiter ? "on" : "off") << "\n"
>+            << "Channels: " << num_channels << "\n"
>+            << "Rate: " << sample_rate << "\n"
>+            << "Number of input streams: " << input_files.size() << "\n";
>+  for (const auto& source : sources) {
>+    std::cout << "\t" << source.ToString() << "\n";
>+  }
>+  std::cout << "Now mixing\n...\n";
>+
>+  webrtc::WavWriter wav_writer(FLAG_output_file, sample_rate, num_channels);
>+
>+  webrtc::AudioFrame frame;
>+
>+  bool all_streams_finished = false;
>+  while (!all_streams_finished) {
>+    mixer->Mix(num_channels, &frame);
>+    RTC_CHECK_EQ(sample_rate / 100, frame.samples_per_channel_);
>+    RTC_CHECK_EQ(sample_rate, frame.sample_rate_hz_);
>+    RTC_CHECK_EQ(num_channels, frame.num_channels_);
>+    wav_writer.WriteSamples(frame.data(),
>+                            num_channels * frame.samples_per_channel_);
>+
>+    all_streams_finished =
>+        std::all_of(sources.begin(), sources.end(),
>+                    [](const webrtc::test::FilePlayingSource& source) {
>+                      return source.FileHasEnded();
>+                    });
>+  }
>+
>+  std::cout << "Done!\n" << std::endl;
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/frame_combiner.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/frame_combiner.cc
>index 7c671ec6aa7..4759c3a40fb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/frame_combiner.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/frame_combiner.cc
>@@ -13,160 +13,113 @@
> #include <algorithm>
> #include <array>
> #include <functional>
>-#include <memory>
> 
> #include "api/array_view.h"
> #include "audio/utility/audio_frame_operations.h"
>+#include "common_audio/include/audio_util.h"
> #include "modules/audio_mixer/audio_frame_manipulator.h"
> #include "modules/audio_mixer/audio_mixer_impl.h"
>+#include "modules/audio_processing/include/audio_processing.h"
>+#include "modules/audio_processing/logging/apm_data_dumper.h"
>+#include "rtc_base/arraysize.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>+#include "system_wrappers/include/metrics.h"
> 
> namespace webrtc {
> namespace {
> 
> // Stereo, 48 kHz, 10 ms.
>-constexpr int kMaximalFrameSize = 2 * 48 * 10;
>-
>-void CombineZeroFrames(bool use_limiter,
>-                       AudioProcessing* limiter,
>-                       AudioFrame* audio_frame_for_mixing) {
>-  audio_frame_for_mixing->elapsed_time_ms_ = -1;
>-  AudioFrameOperations::Mute(audio_frame_for_mixing);
>-  // The limiter should still process a zero frame to avoid jumps in
>-  // its gain curve.
>-  if (use_limiter) {
>-    RTC_DCHECK(limiter);
>-    // The limiter smoothly increases frames with half gain to full
>-    // volume.  Here there's no need to apply half gain, since the frame
>-    // is zero anyway.
>-    limiter->ProcessStream(audio_frame_for_mixing);
>-  }
>-}
>-
>-void CombineOneFrame(const AudioFrame* input_frame,
>-                     bool use_limiter,
>-                     AudioProcessing* limiter,
>-                     AudioFrame* audio_frame_for_mixing) {
>-  audio_frame_for_mixing->timestamp_ = input_frame->timestamp_;
>-  audio_frame_for_mixing->elapsed_time_ms_ = input_frame->elapsed_time_ms_;
>-  // TODO(yujo): can we optimize muted frames?
>-  std::copy(input_frame->data(),
>-            input_frame->data() +
>-                input_frame->num_channels_ * input_frame->samples_per_channel_,
>-            audio_frame_for_mixing->mutable_data());
>-  if (use_limiter) {
>-    AudioFrameOperations::ApplyHalfGain(audio_frame_for_mixing);
>-    RTC_DCHECK(limiter);
>-    limiter->ProcessStream(audio_frame_for_mixing);
>-    AudioFrameOperations::Add(*audio_frame_for_mixing, audio_frame_for_mixing);
>-  }
>-}
>+constexpr int kMaximumAmountOfChannels = 2;
>+constexpr int kMaximumChannelSize = 48 * AudioMixerImpl::kFrameDurationInMs;
> 
>-// Lower-level helper function called from Combine(...) when there
>-// are several input frames.
>-//
>-// TODO(aleloi): change interface to ArrayView<int16_t> output_frame
>-// once we have gotten rid of the APM limiter.
>-//
>-// Only the 'data' field of output_frame should be modified. The
>-// rest are used for potentially sending the output to the APM
>-// limiter.
>-void CombineMultipleFrames(
>-    const std::vector<rtc::ArrayView<const int16_t>>& input_frames,
>-    bool use_limiter,
>-    AudioProcessing* limiter,
>-    AudioFrame* audio_frame_for_mixing) {
>-  RTC_DCHECK(!input_frames.empty());
>-  RTC_DCHECK(audio_frame_for_mixing);
>+using OneChannelBuffer = std::array<float, kMaximumChannelSize>;
> 
>-  const size_t frame_length = input_frames.front().size();
>-  for (const auto& frame : input_frames) {
>-    RTC_DCHECK_EQ(frame_length, frame.size());
>-  }
>+void SetAudioFrameFields(const std::vector<AudioFrame*>& mix_list,
>+                         size_t number_of_channels,
>+                         int sample_rate,
>+                         size_t number_of_streams,
>+                         AudioFrame* audio_frame_for_mixing) {
>+  const size_t samples_per_channel = static_cast<size_t>(
>+      (sample_rate * webrtc::AudioMixerImpl::kFrameDurationInMs) / 1000);
> 
>-  // Algorithm: int16 frames are added to a sufficiently large
>-  // statically allocated int32 buffer. For > 2 participants this is
>-  // more efficient than addition in place in the int16 audio
>-  // frame. The audio quality loss due to halving the samples is
>-  // smaller than 16-bit addition in place.
>-  RTC_DCHECK_GE(kMaximalFrameSize, frame_length);
>-  std::array<int32_t, kMaximalFrameSize> add_buffer;
>+  // TODO(minyue): Issue bugs.webrtc.org/3390.
>+  // Audio frame timestamp. The 'timestamp_' field is set to dummy
>+  // value '0', because it is only supported in the one channel case and
>+  // is then updated in the helper functions.
>+  audio_frame_for_mixing->UpdateFrame(
>+      0, nullptr, samples_per_channel, sample_rate, AudioFrame::kUndefined,
>+      AudioFrame::kVadUnknown, number_of_channels);
> 
>-  add_buffer.fill(0);
>+  if (mix_list.empty()) {
>+    audio_frame_for_mixing->elapsed_time_ms_ = -1;
>+  } else if (mix_list.size() == 1) {
>+    audio_frame_for_mixing->timestamp_ = mix_list[0]->timestamp_;
>+    audio_frame_for_mixing->elapsed_time_ms_ = mix_list[0]->elapsed_time_ms_;
>+  }
>+}
> 
>-  for (const auto& frame : input_frames) {
>-    // TODO(yujo): skip this for muted frames.
>-    std::transform(frame.begin(), frame.end(), add_buffer.begin(),
>-                   add_buffer.begin(), std::plus<int32_t>());
>+void MixFewFramesWithNoLimiter(const std::vector<AudioFrame*>& mix_list,
>+                               AudioFrame* audio_frame_for_mixing) {
>+  if (mix_list.empty()) {
>+    audio_frame_for_mixing->Mute();
>+    return;
>   }
>+  RTC_DCHECK_LE(mix_list.size(), 1);
>+  std::copy(mix_list[0]->data(),
>+            mix_list[0]->data() +
>+                mix_list[0]->num_channels_ * mix_list[0]->samples_per_channel_,
>+            audio_frame_for_mixing->mutable_data());
>+}
> 
>-  if (use_limiter) {
>-    // Halve all samples to avoid saturation before limiting.
>-    std::transform(add_buffer.begin(), add_buffer.begin() + frame_length,
>-                   audio_frame_for_mixing->mutable_data(), [](int32_t a) {
>-                     return rtc::saturated_cast<int16_t>(a / 2);
>-                   });
>-
>-    // Smoothly limit the audio.
>-    RTC_DCHECK(limiter);
>-    const int error = limiter->ProcessStream(audio_frame_for_mixing);
>-    if (error != limiter->kNoError) {
>-      RTC_LOG_F(LS_ERROR) << "Error from AudioProcessing: " << error;
>-      RTC_NOTREACHED();
>+std::array<OneChannelBuffer, kMaximumAmountOfChannels> MixToFloatFrame(
>+    const std::vector<AudioFrame*>& mix_list,
>+    size_t samples_per_channel,
>+    size_t number_of_channels) {
>+  // Convert to FloatS16 and mix.
>+  using OneChannelBuffer = std::array<float, kMaximumChannelSize>;
>+  std::array<OneChannelBuffer, kMaximumAmountOfChannels> mixing_buffer{};
>+
>+  for (size_t i = 0; i < mix_list.size(); ++i) {
>+    const AudioFrame* const frame = mix_list[i];
>+    for (size_t j = 0; j < number_of_channels; ++j) {
>+      for (size_t k = 0; k < samples_per_channel; ++k) {
>+        mixing_buffer[j][k] += frame->data()[number_of_channels * k + j];
>+      }
>     }
>-
>-    // And now we can safely restore the level. This procedure results in
>-    // some loss of resolution, deemed acceptable.
>-    //
>-    // It's possible to apply the gain in the AGC (with a target level of 0 dbFS
>-    // and compression gain of 6 dB). However, in the transition frame when this
>-    // is enabled (moving from one to two audio sources) it has the potential to
>-    // create discontinuities in the mixed frame.
>-    //
>-    // Instead we double the frame (with addition since left-shifting a
>-    // negative value is undefined).
>-    AudioFrameOperations::Add(*audio_frame_for_mixing, audio_frame_for_mixing);
>-  } else {
>-    std::transform(add_buffer.begin(), add_buffer.begin() + frame_length,
>-                   audio_frame_for_mixing->mutable_data(),
>-                   [](int32_t a) { return rtc::saturated_cast<int16_t>(a); });
>   }
>+  return mixing_buffer;
> }
> 
>-std::unique_ptr<AudioProcessing> CreateLimiter() {
>-  Config config;
>-  config.Set<ExperimentalAgc>(new ExperimentalAgc(false));
>-
>-  std::unique_ptr<AudioProcessing> limiter(
>-      AudioProcessingBuilder().Create(config));
>-  RTC_DCHECK(limiter);
>-
>-  webrtc::AudioProcessing::Config apm_config;
>-  apm_config.residual_echo_detector.enabled = false;
>-  limiter->ApplyConfig(apm_config);
>-
>-  const auto check_no_error = [](int x) {
>-    RTC_DCHECK_EQ(x, AudioProcessing::kNoError);
>-  };
>-  auto* const gain_control = limiter->gain_control();
>-  check_no_error(gain_control->set_mode(GainControl::kFixedDigital));
>-
>-  // We smoothly limit the mixed frame to -7 dbFS. -6 would correspond to the
>-  // divide-by-2 but -7 is used instead to give a bit of headroom since the
>-  // AGC is not a hard limiter.
>-  check_no_error(gain_control->set_target_level_dbfs(7));
>-
>-  check_no_error(gain_control->set_compression_gain_db(0));
>-  check_no_error(gain_control->enable_limiter(true));
>-  check_no_error(gain_control->Enable(true));
>-  return limiter;
>+void RunLimiter(AudioFrameView<float> mixing_buffer_view,
>+                FixedGainController* limiter) {
>+  const size_t sample_rate = mixing_buffer_view.samples_per_channel() * 1000 /
>+                             AudioMixerImpl::kFrameDurationInMs;
>+  limiter->SetSampleRate(sample_rate);
>+  limiter->Process(mixing_buffer_view);
>+}
>+
>+// Both interleaves and rounds.
>+void InterleaveToAudioFrame(AudioFrameView<const float> mixing_buffer_view,
>+                            AudioFrame* audio_frame_for_mixing) {
>+  const size_t number_of_channels = mixing_buffer_view.num_channels();
>+  const size_t samples_per_channel = mixing_buffer_view.samples_per_channel();
>+  // Put data in the result frame.
>+  for (size_t i = 0; i < number_of_channels; ++i) {
>+    for (size_t j = 0; j < samples_per_channel; ++j) {
>+      audio_frame_for_mixing->mutable_data()[number_of_channels * j + i] =
>+          FloatS16ToS16(mixing_buffer_view.channel(i)[j]);
>+    }
>+  }
> }
> }  // namespace
> 
>-FrameCombiner::FrameCombiner(bool use_apm_limiter)
>-    : use_apm_limiter_(use_apm_limiter),
>-      limiter_(use_apm_limiter ? CreateLimiter() : nullptr) {}
>+FrameCombiner::FrameCombiner(bool use_limiter)
>+    : data_dumper_(new ApmDataDumper(0)),
>+      limiter_(data_dumper_.get(), "AudioMixer"),
>+      use_limiter_(use_limiter) {
>+  limiter_.SetGain(0.f);
>+}
> 
> FrameCombiner::~FrameCombiner() = default;
> 
>@@ -174,8 +127,14 @@ void FrameCombiner::Combine(const std::vector<AudioFrame*>& mix_list,
>                             size_t number_of_channels,
>                             int sample_rate,
>                             size_t number_of_streams,
>-                            AudioFrame* audio_frame_for_mixing) const {
>+                            AudioFrame* audio_frame_for_mixing) {
>   RTC_DCHECK(audio_frame_for_mixing);
>+
>+  LogMixingStats(mix_list, sample_rate, number_of_streams);
>+
>+  SetAudioFrameFields(mix_list, number_of_channels, sample_rate,
>+                      number_of_streams, audio_frame_for_mixing);
>+
>   const size_t samples_per_channel = static_cast<size_t>(
>       (sample_rate * webrtc::AudioMixerImpl::kFrameDurationInMs) / 1000);
> 
>@@ -184,35 +143,60 @@ void FrameCombiner::Combine(const std::vector<AudioFrame*>& mix_list,
>     RTC_DCHECK_EQ(sample_rate, frame->sample_rate_hz_);
>   }
> 
>-  // Frames could be both stereo and mono.
>+  // The 'num_channels_' field of frames in 'mix_list' could be
>+  // different from 'number_of_channels'.
>   for (auto* frame : mix_list) {
>     RemixFrame(number_of_channels, frame);
>   }
> 
>-  // TODO(aleloi): Issue bugs.webrtc.org/3390.
>-  // Audio frame timestamp. The 'timestamp_' field is set to dummy
>-  // value '0', because it is only supported in the one channel case and
>-  // is then updated in the helper functions.
>-  audio_frame_for_mixing->UpdateFrame(
>-      0, nullptr, samples_per_channel, sample_rate, AudioFrame::kUndefined,
>-      AudioFrame::kVadUnknown, number_of_channels);
>+  if (number_of_streams <= 1) {
>+    MixFewFramesWithNoLimiter(mix_list, audio_frame_for_mixing);
>+    return;
>+  }
> 
>-  const bool use_limiter_this_round = use_apm_limiter_ && number_of_streams > 1;
>+  std::array<OneChannelBuffer, kMaximumAmountOfChannels> mixing_buffer =
>+      MixToFloatFrame(mix_list, samples_per_channel, number_of_channels);
> 
>-  if (mix_list.empty()) {
>-    CombineZeroFrames(use_limiter_this_round, limiter_.get(),
>-                      audio_frame_for_mixing);
>-  } else if (mix_list.size() == 1) {
>-    CombineOneFrame(mix_list.front(), use_limiter_this_round, limiter_.get(),
>-                    audio_frame_for_mixing);
>-  } else {
>-    std::vector<rtc::ArrayView<const int16_t>> input_frames;
>-    for (size_t i = 0; i < mix_list.size(); ++i) {
>-      input_frames.push_back(rtc::ArrayView<const int16_t>(
>-          mix_list[i]->data(), samples_per_channel * number_of_channels));
>-    }
>-    CombineMultipleFrames(input_frames, use_limiter_this_round, limiter_.get(),
>-                          audio_frame_for_mixing);
>+  // Put float data in an AudioFrameView.
>+  std::array<float*, kMaximumAmountOfChannels> channel_pointers{};
>+  for (size_t i = 0; i < number_of_channels; ++i) {
>+    channel_pointers[i] = &mixing_buffer[i][0];
>+  }
>+  AudioFrameView<float> mixing_buffer_view(
>+      &channel_pointers[0], number_of_channels, samples_per_channel);
>+
>+  if (use_limiter_) {
>+    RunLimiter(mixing_buffer_view, &limiter_);
>+  }
>+
>+  InterleaveToAudioFrame(mixing_buffer_view, audio_frame_for_mixing);
>+}
>+
>+void FrameCombiner::LogMixingStats(const std::vector<AudioFrame*>& mix_list,
>+                                   int sample_rate,
>+                                   size_t number_of_streams) const {
>+  // Log every second.
>+  uma_logging_counter_++;
>+  if (uma_logging_counter_ > 1000 / AudioMixerImpl::kFrameDurationInMs) {
>+    uma_logging_counter_ = 0;
>+    RTC_HISTOGRAM_COUNTS_100("WebRTC.Audio.AudioMixer.NumIncomingStreams",
>+                             static_cast<int>(number_of_streams));
>+    RTC_HISTOGRAM_ENUMERATION(
>+        "WebRTC.Audio.AudioMixer.NumIncomingActiveStreams",
>+        static_cast<int>(mix_list.size()),
>+        AudioMixerImpl::kMaximumAmountOfMixedAudioSources);
>+
>+    using NativeRate = AudioProcessing::NativeRate;
>+    static constexpr NativeRate native_rates[] = {
>+        NativeRate::kSampleRate8kHz, NativeRate::kSampleRate16kHz,
>+        NativeRate::kSampleRate32kHz, NativeRate::kSampleRate48kHz};
>+    const auto* rate_position = std::lower_bound(
>+        std::begin(native_rates), std::end(native_rates), sample_rate);
>+
>+    RTC_HISTOGRAM_ENUMERATION(
>+        "WebRTC.Audio.AudioMixer.MixingRate",
>+        std::distance(std::begin(native_rates), rate_position),
>+        arraysize(native_rates));
>   }
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/frame_combiner.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/frame_combiner.h
>index 88ab0d7f9ad..70ac0277589 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/frame_combiner.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/frame_combiner.h
>@@ -14,14 +14,17 @@
> #include <memory>
> #include <vector>
> 
>-#include "modules/audio_processing/include/audio_processing.h"
>-#include "modules/include/module_common_types.h"
>+#include "api/audio/audio_frame.h"
>+#include "modules/audio_processing/agc2/fixed_gain_controller.h"
> 
> namespace webrtc {
>+class ApmDataDumper;
>+class FixedGainController;
> 
> class FrameCombiner {
>  public:
>-  explicit FrameCombiner(bool use_apm_limiter);
>+  enum class LimiterType { kNoLimiter, kApmAgcLimiter, kApmAgc2Limiter };
>+  explicit FrameCombiner(bool use_limiter);
>   ~FrameCombiner();
> 
>   // Combine several frames into one. Assumes sample_rate,
>@@ -34,11 +37,17 @@ class FrameCombiner {
>                size_t number_of_channels,
>                int sample_rate,
>                size_t number_of_streams,
>-               AudioFrame* audio_frame_for_mixing) const;
>+               AudioFrame* audio_frame_for_mixing);
> 
>  private:
>-  const bool use_apm_limiter_;
>-  std::unique_ptr<AudioProcessing> limiter_;
>+  void LogMixingStats(const std::vector<AudioFrame*>& mix_list,
>+                      int sample_rate,
>+                      size_t number_of_streams) const;
>+
>+  std::unique_ptr<ApmDataDumper> data_dumper_;
>+  FixedGainController limiter_;
>+  const bool use_limiter_;
>+  mutable int uma_logging_counter_ = 0;
> };
> }  // namespace webrtc
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/frame_combiner_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/frame_combiner_unittest.cc
>index 490e99e4954..c7bf9c31359 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/frame_combiner_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/frame_combiner_unittest.cc
>@@ -23,6 +23,14 @@
> namespace webrtc {
> 
> namespace {
>+using LimiterType = FrameCombiner::LimiterType;
>+struct FrameCombinerConfig {
>+  bool use_limiter;
>+  int sample_rate_hz;
>+  int number_of_channels;
>+  float wave_frequency;
>+};
>+
> std::string ProduceDebugText(int sample_rate_hz,
>                              int number_of_channels,
>                              int number_of_sources) {
>@@ -33,17 +41,12 @@ std::string ProduceDebugText(int sample_rate_hz,
>   return ss.str();
> }
> 
>-std::string ProduceDebugText(int sample_rate_hz,
>-                             int number_of_channels,
>-                             int number_of_sources,
>-                             bool limiter_active,
>-                             float wave_frequency) {
>+std::string ProduceDebugText(const FrameCombinerConfig& config) {
>   std::ostringstream ss;
>-  ss << "Sample rate: " << sample_rate_hz << " ,";
>-  ss << "number of channels: " << number_of_channels << " ,";
>-  ss << "number of sources: " << number_of_sources << " ,";
>-  ss << "limiter active: " << (limiter_active ? "true" : "false") << " ,";
>-  ss << "wave frequency: " << wave_frequency << " ,";
>+  ss << "Sample rate: " << config.sample_rate_hz << " ,";
>+  ss << "number of channels: " << config.number_of_channels << " ,";
>+  ss << "limiter active: " << (config.use_limiter ? "on" : "off") << " ,";
>+  ss << "wave frequency: " << config.wave_frequency << " ,";
>   return ss.str();
> }
> 
>@@ -60,9 +63,10 @@ void SetUpFrames(int sample_rate_hz, int number_of_channels) {
> }
> }  // namespace
> 
>+// The limiter requires sample rate divisible by 2000.
> TEST(FrameCombiner, BasicApiCallsLimiter) {
>   FrameCombiner combiner(true);
>-  for (const int rate : {8000, 16000, 32000, 48000}) {
>+  for (const int rate : {8000, 18000, 34000, 48000}) {
>     for (const int number_of_channels : {1, 2}) {
>       const std::vector<AudioFrame*> all_frames = {&frame1, &frame2};
>       SetUpFrames(rate, number_of_channels);
>@@ -79,9 +83,8 @@ TEST(FrameCombiner, BasicApiCallsLimiter) {
>   }
> }
> 
>-// No APM limiter means no AudioProcessing::NativeRate restriction
>-// on rate. The rate has to be divisible by 100 since we use
>-// 10 ms frames, though.
>+// With no limiter, the rate has to be divisible by 100 since we use
>+// 10 ms frames.
> TEST(FrameCombiner, BasicApiCallsNoLimiter) {
>   FrameCombiner combiner(false);
>   for (const int rate : {8000, 10000, 11000, 32000, 44100}) {
>@@ -150,56 +153,60 @@ TEST(FrameCombiner, CombiningOneFrameShouldNotChangeFrame) {
> }
> 
> // Send a sine wave through the FrameCombiner, and check that the
>-// difference between input and output varies smoothly. This is to
>-// catch issues like chromium:695993.
>+// difference between input and output varies smoothly. Also check
>+// that it is inside reasonable bounds. This is to catch issues like
>+// chromium:695993 and chromium:816875.
> TEST(FrameCombiner, GainCurveIsSmoothForAlternatingNumberOfStreams) {
>-  // Test doesn't work with rates requiring a band split, because it
>-  // introduces a small delay measured in single samples, and this
>-  // test cannot handle it.
>-  //
>-  // TODO(aleloi): Add more rates when APM limiter doesn't use band
>-  // split.
>-  for (const bool use_limiter : {true, false}) {
>-    for (const int rate : {8000, 16000}) {
>-      constexpr int number_of_channels = 2;
>-      for (const float wave_frequency : {50, 400, 3200}) {
>-        SCOPED_TRACE(ProduceDebugText(rate, number_of_channels, 1, use_limiter,
>-                                      wave_frequency));
>-
>-        FrameCombiner combiner(use_limiter);
>-
>-        constexpr int16_t wave_amplitude = 30000;
>-        SineWaveGenerator wave_generator(wave_frequency, wave_amplitude);
>-
>-        GainChangeCalculator change_calculator;
>-        float cumulative_change = 0.f;
>-
>-        constexpr size_t iterations = 100;
>-
>-        for (size_t i = 0; i < iterations; ++i) {
>-          SetUpFrames(rate, number_of_channels);
>-          wave_generator.GenerateNextFrame(&frame1);
>-          AudioFrameOperations::Mute(&frame2);
>-
>-          std::vector<AudioFrame*> frames_to_combine = {&frame1};
>-          if (i % 2 == 0) {
>-            frames_to_combine.push_back(&frame2);
>-          }
>-          const size_t number_of_samples =
>-              frame1.samples_per_channel_ * number_of_channels;
>-
>-          // Ensures limiter is on if 'use_limiter'.
>-          constexpr size_t number_of_streams = 2;
>-          combiner.Combine(frames_to_combine, number_of_channels, rate,
>-                           number_of_streams, &audio_frame_for_mixing);
>-          cumulative_change += change_calculator.CalculateGainChange(
>-              rtc::ArrayView<const int16_t>(frame1.data(), number_of_samples),
>-              rtc::ArrayView<const int16_t>(audio_frame_for_mixing.data(),
>-                                            number_of_samples));
>-        }
>-        RTC_DCHECK_LT(cumulative_change, 10);
>+  // Rates are divisible by 2000 when limiter is active.
>+  std::vector<FrameCombinerConfig> configs = {
>+      {false, 30100, 2, 50.f},  {false, 16500, 1, 3200.f},
>+      {true, 8000, 1, 3200.f},  {true, 16000, 1, 50.f},
>+      {true, 18000, 2, 3200.f}, {true, 10000, 2, 50.f},
>+  };
>+
>+  for (const auto& config : configs) {
>+    SCOPED_TRACE(ProduceDebugText(config));
>+
>+    FrameCombiner combiner(config.use_limiter);
>+
>+    constexpr int16_t wave_amplitude = 30000;
>+    SineWaveGenerator wave_generator(config.wave_frequency, wave_amplitude);
>+
>+    GainChangeCalculator change_calculator;
>+    float cumulative_change = 0.f;
>+
>+    constexpr size_t iterations = 100;
>+
>+    for (size_t i = 0; i < iterations; ++i) {
>+      SetUpFrames(config.sample_rate_hz, config.number_of_channels);
>+      wave_generator.GenerateNextFrame(&frame1);
>+      AudioFrameOperations::Mute(&frame2);
>+
>+      std::vector<AudioFrame*> frames_to_combine = {&frame1};
>+      if (i % 2 == 0) {
>+        frames_to_combine.push_back(&frame2);
>       }
>+      const size_t number_of_samples =
>+          frame1.samples_per_channel_ * config.number_of_channels;
>+
>+      // Ensures limiter is on if 'use_limiter'.
>+      constexpr size_t number_of_streams = 2;
>+      combiner.Combine(frames_to_combine, config.number_of_channels,
>+                       config.sample_rate_hz, number_of_streams,
>+                       &audio_frame_for_mixing);
>+      cumulative_change += change_calculator.CalculateGainChange(
>+          rtc::ArrayView<const int16_t>(frame1.data(), number_of_samples),
>+          rtc::ArrayView<const int16_t>(audio_frame_for_mixing.data(),
>+                                        number_of_samples));
>     }
>+
>+    // Check that the gain doesn't vary too much.
>+    EXPECT_LT(cumulative_change, 10);
>+
>+    // Check that the latest gain is within reasonable bounds. It
>+    // should be slightly less that 1.
>+    EXPECT_LT(0.9f, change_calculator.LatestGain());
>+    EXPECT_LT(change_calculator.LatestGain(), 1.01f);
>   }
> }
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/gain_change_calculator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/gain_change_calculator.cc
>index ae7c6d59e70..64727938fab 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/gain_change_calculator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/gain_change_calculator.cc
>@@ -29,6 +29,10 @@ float GainChangeCalculator::CalculateGainChange(
>   return CalculateDifferences(gain);
> }
> 
>+float GainChangeCalculator::LatestGain() const {
>+  return last_reliable_gain_;
>+}
>+
> void GainChangeCalculator::CalculateGain(rtc::ArrayView<const int16_t> in,
>                                          rtc::ArrayView<const int16_t> out,
>                                          rtc::ArrayView<float> gain) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/gain_change_calculator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/gain_change_calculator.h
>index fe7957cdd57..1d7f151edb9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/gain_change_calculator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/gain_change_calculator.h
>@@ -23,6 +23,8 @@ class GainChangeCalculator {
>   float CalculateGainChange(rtc::ArrayView<const int16_t> in,
>                             rtc::ArrayView<const int16_t> out);
> 
>+  float LatestGain() const;
>+
>  private:
>   void CalculateGain(rtc::ArrayView<const int16_t> in,
>                      rtc::ArrayView<const int16_t> out,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/sine_wave_generator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/sine_wave_generator.h
>index b5c4674e201..4a50c0e88ea 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/sine_wave_generator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_mixer/sine_wave_generator.h
>@@ -11,7 +11,7 @@
> #ifndef MODULES_AUDIO_MIXER_SINE_WAVE_GENERATOR_H_
> #define MODULES_AUDIO_MIXER_SINE_WAVE_GENERATOR_H_
> 
>-#include "modules/include/module_common_types.h"
>+#include "api/audio/audio_frame.h"
> #include "rtc_base/checks.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/BUILD.gn
>index c7d36c6472c..9e714461e96 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/BUILD.gn
>@@ -6,7 +6,6 @@
> # in the file PATENTS.  All contributing project authors may
> # be found in the AUTHORS file in the root of the source tree.
> 
>-import("//build/config/arm.gni")
> import("../../webrtc.gni")
> if (!build_with_mozilla) {
>   import("//third_party/protobuf/proto_library.gni")
>@@ -31,114 +30,19 @@ rtc_static_library("audio_processing") {
>   visibility = [ "*" ]
>   configs += [ ":apm_debug_dump" ]
>   sources = [
>-    "aec/aec_resampler.cc",
>-    "aec/aec_resampler.h",
>-    "aec/echo_cancellation.cc",
>-    "aec/echo_cancellation.h",
>-    "aec3/adaptive_fir_filter.cc",
>-    "aec3/adaptive_fir_filter.h",
>-    "aec3/aec3_common.cc",
>-    "aec3/aec3_common.h",
>-    "aec3/aec3_fft.cc",
>-    "aec3/aec3_fft.h",
>-    "aec3/aec_state.cc",
>-    "aec3/aec_state.h",
>-    "aec3/block_framer.cc",
>-    "aec3/block_framer.h",
>-    "aec3/block_processor.cc",
>-    "aec3/block_processor.h",
>-    "aec3/block_processor_metrics.cc",
>-    "aec3/block_processor_metrics.h",
>-    "aec3/cascaded_biquad_filter.cc",
>-    "aec3/cascaded_biquad_filter.h",
>-    "aec3/comfort_noise_generator.cc",
>-    "aec3/comfort_noise_generator.h",
>-    "aec3/decimator.cc",
>-    "aec3/decimator.h",
>-    "aec3/delay_estimate.h",
>-    "aec3/downsampled_render_buffer.cc",
>-    "aec3/downsampled_render_buffer.h",
>-    "aec3/echo_canceller3.cc",
>-    "aec3/echo_canceller3.h",
>-    "aec3/echo_path_delay_estimator.cc",
>-    "aec3/echo_path_delay_estimator.h",
>-    "aec3/echo_path_variability.cc",
>-    "aec3/echo_path_variability.h",
>-    "aec3/echo_remover.cc",
>-    "aec3/echo_remover.h",
>-    "aec3/echo_remover_metrics.cc",
>-    "aec3/echo_remover_metrics.h",
>-    "aec3/erl_estimator.cc",
>-    "aec3/erl_estimator.h",
>-    "aec3/erle_estimator.cc",
>-    "aec3/erle_estimator.h",
>-    "aec3/fft_buffer.cc",
>-    "aec3/fft_buffer.h",
>-    "aec3/fft_data.h",
>-    "aec3/frame_blocker.cc",
>-    "aec3/frame_blocker.h",
>-    "aec3/main_filter_update_gain.cc",
>-    "aec3/main_filter_update_gain.h",
>-    "aec3/matched_filter.cc",
>-    "aec3/matched_filter.h",
>-    "aec3/matched_filter_lag_aggregator.cc",
>-    "aec3/matched_filter_lag_aggregator.h",
>-    "aec3/matrix_buffer.cc",
>-    "aec3/matrix_buffer.h",
>-    "aec3/output_selector.cc",
>-    "aec3/output_selector.h",
>-    "aec3/render_buffer.cc",
>-    "aec3/render_buffer.h",
>-    "aec3/render_delay_buffer.cc",
>-    "aec3/render_delay_buffer.h",
>-    "aec3/render_delay_controller.cc",
>-    "aec3/render_delay_controller.h",
>-    "aec3/render_delay_controller_metrics.cc",
>-    "aec3/render_delay_controller_metrics.h",
>-    "aec3/render_signal_analyzer.cc",
>-    "aec3/render_signal_analyzer.h",
>-    "aec3/residual_echo_estimator.cc",
>-    "aec3/residual_echo_estimator.h",
>-    "aec3/shadow_filter_update_gain.cc",
>-    "aec3/shadow_filter_update_gain.h",
>-    "aec3/subtractor.cc",
>-    "aec3/subtractor.h",
>-    "aec3/subtractor_output.h",
>-    "aec3/suppression_filter.cc",
>-    "aec3/suppression_filter.h",
>-    "aec3/suppression_gain.cc",
>-    "aec3/suppression_gain.h",
>-    "aec3/vector_buffer.cc",
>-    "aec3/vector_buffer.h",
>-    "aec3/vector_math.h",
>-    "agc/agc.cc",
>-    "agc/agc.h",
>-    "agc/agc_manager_direct.cc",
>-    "agc/agc_manager_direct.h",
>-    "agc/gain_map_internal.h",
>-    "agc/loudness_histogram.cc",
>-    "agc/loudness_histogram.h",
>-    "agc/utility.cc",
>-    "agc/utility.h",
>-    "agc2/gain_controller2.cc",
>-    "agc2/gain_controller2.h",
>     "audio_buffer.cc",
>     "audio_buffer.h",
>     "audio_processing_impl.cc",
>     "audio_processing_impl.h",
>-    "beamformer/array_util.cc",
>-    "beamformer/array_util.h",
>-    "beamformer/complex_matrix.h",
>-    "beamformer/covariance_matrix_generator.cc",
>-    "beamformer/covariance_matrix_generator.h",
>-    "beamformer/matrix.h",
>-    "beamformer/nonlinear_beamformer.cc",
>-    "beamformer/nonlinear_beamformer.h",
>     "common.h",
>     "echo_cancellation_impl.cc",
>     "echo_cancellation_impl.h",
>+    "echo_cancellation_proxy.cc",
>+    "echo_cancellation_proxy.h",
>     "echo_control_mobile_impl.cc",
>     "echo_control_mobile_impl.h",
>+    "echo_control_mobile_proxy.cc",
>+    "echo_control_mobile_proxy.h",
>     "echo_detector/circular_buffer.cc",
>     "echo_detector/circular_buffer.h",
>     "echo_detector/mean_variance_estimator.cc",
>@@ -151,31 +55,14 @@ rtc_static_library("audio_processing") {
>     "gain_control_for_experimental_agc.h",
>     "gain_control_impl.cc",
>     "gain_control_impl.h",
>+    "gain_controller2.cc",
>+    "gain_controller2.h",
>+    "include/aec_dump.cc",
>+    "include/aec_dump.h",
>     "include/audio_processing.cc",
>     "include/audio_processing.h",
>     "include/config.cc",
>     "include/config.h",
>-    "level_controller/biquad_filter.cc",
>-    "level_controller/biquad_filter.h",
>-    "level_controller/down_sampler.cc",
>-    "level_controller/down_sampler.h",
>-    "level_controller/gain_applier.cc",
>-    "level_controller/gain_applier.h",
>-    "level_controller/gain_selector.cc",
>-    "level_controller/gain_selector.h",
>-    "level_controller/level_controller.cc",
>-    "level_controller/level_controller.h",
>-    "level_controller/level_controller_constants.h",
>-    "level_controller/noise_level_estimator.cc",
>-    "level_controller/noise_level_estimator.h",
>-    "level_controller/noise_spectrum_estimator.cc",
>-    "level_controller/noise_spectrum_estimator.h",
>-    "level_controller/peak_level_estimator.cc",
>-    "level_controller/peak_level_estimator.h",
>-    "level_controller/saturating_gain_estimator.cc",
>-    "level_controller/saturating_gain_estimator.h",
>-    "level_controller/signal_classifier.cc",
>-    "level_controller/signal_classifier.h",
>     "level_estimator_impl.cc",
>     "level_estimator_impl.h",
>     "low_cut_filter.cc",
>@@ -206,63 +93,51 @@ rtc_static_library("audio_processing") {
>     "transient/wpd_tree.h",
>     "typing_detection.cc",
>     "typing_detection.h",
>-    "vad/common.h",
>-    "vad/gmm.cc",
>-    "vad/gmm.h",
>-    "vad/noise_gmm_tables.h",
>-    "vad/pitch_based_vad.cc",
>-    "vad/pitch_based_vad.h",
>-    "vad/pitch_internal.cc",
>-    "vad/pitch_internal.h",
>-    "vad/pole_zero_filter.cc",
>-    "vad/pole_zero_filter.h",
>-    "vad/standalone_vad.cc",
>-    "vad/standalone_vad.h",
>-    "vad/vad_audio_proc.cc",
>-    "vad/vad_audio_proc.h",
>-    "vad/vad_audio_proc_internal.h",
>-    "vad/vad_circular_buffer.cc",
>-    "vad/vad_circular_buffer.h",
>-    "vad/voice_activity_detector.cc",
>-    "vad/voice_activity_detector.h",
>-    "vad/voice_gmm_tables.h",
>     "voice_detection_impl.cc",
>     "voice_detection_impl.h",
>   ]
> 
>   defines = []
>   deps = [
>-    ":aec_core",
>-    ":aec_dump_interface",
>     ":apm_logging",
>+    ":audio_frame_view",
>+    ":audio_generator_interface",
>     ":audio_processing_c",
>     ":audio_processing_statistics",
>-    "..:module_api",
>+    ":gain_control_interface",
>     "../..:webrtc_common",
>-    "../../:typedefs",
>     "../../api:array_view",
>-    "../../api:optional",
>+    "../../api/audio:aec3_config",
>+    "../../api/audio:audio_frame_api",
>+    "../../api/audio:echo_control",
>     "../../audio/utility:audio_frame_operations",
>+    "../../common_audio:common_audio_c",
>+    "../../common_audio/third_party/fft4g:fft4g",
>     "../../rtc_base:checks",
>     "../../rtc_base:deprecation",
>     "../../rtc_base:gtest_prod",
>-    "../../rtc_base:protobuf_utils",
>+    "../../rtc_base:safe_minmax",
>     "../../rtc_base:sanitizer",
>+    "../../rtc_base/system:arch",
>     "../../system_wrappers:cpu_features_api",
>     "../../system_wrappers:field_trial_api",
>     "../../system_wrappers:metrics_api",
>-    "../audio_coding:isac",
>+    "aec:aec",
>+    "aec:aec_core",
>+    "aecm:aecm_core",
>+    "agc",
>+    "agc:agc_legacy_c",
>+    "agc2:adaptive_digital",
>+    "agc2:fixed_digital",
>+    "agc2:gain_applier",
>+    "vad",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> 
>   if (aec_untrusted_delay_for_testing) {
>     defines += [ "WEBRTC_UNTRUSTED_DELAY" ]
>   }
> 
>-  if (rtc_enable_protobuf) {
>-    defines += [ "WEBRTC_AUDIOPROC_DEBUG_DUMP" ]
>-    deps += [ ":audioproc_debug_proto" ]
>-  }
>-
>   if (rtc_enable_intelligibility_enhancer) {
>     defines += [ "WEBRTC_INTELLIGIBILITY_ENHANCER=1" ]
>     sources += [
>@@ -281,9 +156,6 @@ rtc_static_library("audio_processing") {
>     defines += [ "WEBRTC_NS_FLOAT" ]
>   }
> 
>-  # TODO(jschuh): Bug 1348: fix this warning.
>-  configs += [ "//build/config/compiler:no_size_t_to_int_warning" ]
>-
>   deps += [
>     "../../common_audio",
>     "../../common_audio:fir_filter",
>@@ -293,6 +165,12 @@ rtc_static_library("audio_processing") {
>   ]
> }
> 
>+rtc_source_set("gain_control_interface") {
>+  sources = [
>+    "include/gain_control.h",
>+  ]
>+}
>+
> rtc_source_set("audio_processing_statistics") {
>   visibility = [ "*" ]
>   sources = [
>@@ -300,31 +178,63 @@ rtc_source_set("audio_processing_statistics") {
>     "include/audio_processing_statistics.h",
>   ]
>   deps = [
>-    "../../api:optional",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
> 
>-rtc_source_set("aec_dump_interface") {
>+rtc_source_set("audio_frame_view") {
>   sources = [
>-    "include/aec_dump.cc",
>-    "include/aec_dump.h",
>+    "include/audio_frame_view.h",
>   ]
>-
>   deps = [
>     "../../api:array_view",
>+  ]
>+}
>+
>+rtc_source_set("audio_generator_interface") {
>+  visibility = [ "*" ]
>+  sources = [
>+    "include/audio_generator.h",
>+  ]
>+  deps = [
>+    ":audio_frame_view",
>     "../../rtc_base:rtc_base_approved",
>+    "../../system_wrappers:system_wrappers",
>   ]
> }
> 
>-rtc_source_set("audio_processing_c") {
>-  visibility = [ ":*" ]  # Only targets in this file can depend on this.
>+rtc_source_set("audio_generator_factory") {
>+  visibility = [ "*" ]
>   sources = [
>-    "agc/legacy/analog_agc.c",
>-    "agc/legacy/analog_agc.h",
>-    "agc/legacy/digital_agc.c",
>-    "agc/legacy/digital_agc.h",
>-    "agc/legacy/gain_control.h",
>+    "include/audio_generator_factory.cc",
>+    "include/audio_generator_factory.h",
>   ]
>+  deps = [
>+    ":audio_generator_interface",
>+    ":file_audio_generator",
>+    "../../common_audio:common_audio",
>+    "../../rtc_base:rtc_base_approved",
>+    "../../system_wrappers:system_wrappers",
>+    "//third_party/abseil-cpp/absl/memory",
>+  ]
>+}
>+
>+rtc_source_set("file_audio_generator") {
>+  sources = [
>+    "audio_generator/file_audio_generator.cc",
>+    "audio_generator/file_audio_generator.h",
>+  ]
>+  deps = [
>+    ":audio_generator_interface",
>+    "../../common_audio:common_audio",
>+    "../../rtc_base:rtc_base_approved",
>+    "../../system_wrappers:system_wrappers",
>+  ]
>+}
>+
>+rtc_source_set("audio_processing_c") {
>+  visibility = [ ":*" ]  # Only targets in this file can depend on this.
>+  sources = []
> 
>   if (rtc_prefer_fixed_point) {
>     sources += [
>@@ -352,20 +262,20 @@ rtc_source_set("audio_processing_c") {
> 
>   deps = [
>     "../..:webrtc_common",
>-    "../../:typedefs",
>     "../../common_audio",
>+    "../../common_audio:common_audio_c",
>+    "../../common_audio/third_party/fft4g:fft4g",
>     "../../rtc_base:checks",
>     "../../rtc_base:rtc_base_approved",
>     "../../system_wrappers:cpu_features_api",
>+    "agc:agc_legacy_c",
>   ]
> 
>   if (rtc_build_with_neon) {
>     sources += [ "ns/nsx_core_neon.c" ]
> 
>     if (current_cpu != "arm64") {
>-      # Enable compilation for the NEON instruction set. This is needed
>-      # since //build/config/arm.gni only enables NEON for iOS, not Android.
>-      # This provides the same functionality as webrtc/build/arm_neon.gypi.
>+      # Enable compilation for the NEON instruction set.
>       suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
>       cflags = [ "-mfpu=neon" ]
>     }
>@@ -406,99 +316,6 @@ rtc_source_set("apm_logging") {
>   defines = []
> }
> 
>-rtc_source_set("aec_core") {
>-  configs += [ ":apm_debug_dump" ]
>-  sources = [
>-    "aec/aec_common.h",
>-    "aec/aec_core.cc",
>-    "aec/aec_core.h",
>-    "aec/aec_core_optimized_methods.h",
>-    "aecm/aecm_core.cc",
>-    "aecm/aecm_core.h",
>-    "aecm/aecm_defines.h",
>-    "aecm/echo_control_mobile.cc",
>-    "aecm/echo_control_mobile.h",
>-    "utility/block_mean_calculator.cc",
>-    "utility/block_mean_calculator.h",
>-    "utility/delay_estimator.cc",
>-    "utility/delay_estimator.h",
>-    "utility/delay_estimator_internal.h",
>-    "utility/delay_estimator_wrapper.cc",
>-    "utility/delay_estimator_wrapper.h",
>-    "utility/ooura_fft.cc",
>-    "utility/ooura_fft.h",
>-    "utility/ooura_fft_tables_common.h",
>-  ]
>-  deps = [
>-    ":apm_logging",
>-    ":audio_processing_statistics",
>-    "../..:webrtc_common",
>-    "../../:typedefs",
>-    "../../common_audio:common_audio",
>-    "../../rtc_base:checks",
>-    "../../rtc_base:rtc_base_approved",
>-    "../../rtc_base:sanitizer",
>-    "../../system_wrappers:cpu_features_api",
>-    "../../system_wrappers:metrics_api",
>-  ]
>-  cflags = []
>-  defines = []
>-
>-  if (current_cpu == "x86" || current_cpu == "x64") {
>-    sources += [
>-      "aec/aec_core_sse2.cc",
>-      "utility/ooura_fft_sse2.cc",
>-      "utility/ooura_fft_tables_neon_sse2.h",
>-    ]
>-    if (is_posix) {
>-      cflags += [ "-msse2" ]
>-    }
>-  }
>-
>-  if (rtc_build_with_neon) {
>-    sources += [
>-      "aec/aec_core_neon.cc",
>-      "aecm/aecm_core_neon.cc",
>-      "utility/ooura_fft_neon.cc",
>-      "utility/ooura_fft_tables_neon_sse2.h",
>-    ]
>-
>-    if (current_cpu != "arm64") {
>-      # Enable compilation for the NEON instruction set. This is needed
>-      # since //build/config/arm.gni only enables NEON for iOS, not Android.
>-      # This provides the same functionality as webrtc/build/arm_neon.gypi.
>-      suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
>-      cflags += [ "-mfpu=neon" ]
>-    }
>-
>-    # Disable LTO on NEON targets due to compiler bug.
>-    # TODO(fdegans): Enable this. See crbug.com/408997.
>-    if (rtc_use_lto) {
>-      cflags -= [
>-        "-flto",
>-        "-ffat-lto-objects",
>-      ]
>-    }
>-
>-    deps += [ "../../common_audio" ]
>-  }
>-
>-  if (current_cpu == "mipsel") {
>-    sources += [ "aecm/aecm_core_mips.cc" ]
>-    if (mips_float_abi == "hard") {
>-      sources += [
>-        "aec/aec_core_mips.cc",
>-        "utility/ooura_fft_mips.cc",
>-      ]
>-    }
>-  } else {
>-    sources += [ "aecm/aecm_core_c.cc" ]
>-  }
>-
>-  # TODO(jschuh): Bug 1348: fix this warning.
>-  configs += [ "//build/config/compiler:no_size_t_to_int_warning" ]
>-}
>-
> if (rtc_include_tests) {
>   rtc_source_set("mocks") {
>     testonly = true
>@@ -506,7 +323,6 @@ if (rtc_include_tests) {
>       "include/mock_audio_processing.h",
>     ]
>     deps = [
>-      ":aec_dump_interface",
>       ":audio_processing",
>       ":audio_processing_statistics",
>       "../../test:test_support",
>@@ -518,7 +334,6 @@ if (rtc_include_tests) {
>     deps = [
>       ":audioproc_test_utils",
>       ":click_annotate",
>-      ":nonlinear_beamformer_test",
>       ":transient_suppression_test",
>     ]
> 
>@@ -542,24 +357,11 @@ if (rtc_include_tests) {
> 
>     configs += [ ":apm_debug_dump" ]
>     sources = [
>-      "aec/echo_cancellation_unittest.cc",
>-      "aec/system_delay_unittest.cc",
>-      "aec3/mock/mock_block_processor.h",
>-      "aec3/mock/mock_echo_remover.h",
>-      "aec3/mock/mock_render_delay_buffer.h",
>-      "aec3/mock/mock_render_delay_controller.h",
>-      "agc/agc_manager_direct_unittest.cc",
>-      "agc/loudness_histogram_unittest.cc",
>-      "agc/mock_agc.h",
>       "audio_buffer_unittest.cc",
>-      "beamformer/array_util_unittest.cc",
>-      "beamformer/complex_matrix_unittest.cc",
>-      "beamformer/covariance_matrix_generator_unittest.cc",
>-      "beamformer/matrix_test_helpers.h",
>-      "beamformer/matrix_unittest.cc",
>-      "beamformer/mock_nonlinear_beamformer.h",
>+      "audio_frame_view_unittest.cc",
>       "config_unittest.cc",
>       "echo_cancellation_impl_unittest.cc",
>+      "gain_controller2_unittest.cc",
>       "splitting_filter_unittest.cc",
>       "test/fake_recording_device_unittest.cc",
>       "transient/dyadic_decimator_unittest.cc",
>@@ -571,44 +373,51 @@ if (rtc_include_tests) {
>       "transient/transient_suppressor_unittest.cc",
>       "transient/wpd_node_unittest.cc",
>       "transient/wpd_tree_unittest.cc",
>-      "utility/block_mean_calculator_unittest.cc",
>-      "utility/delay_estimator_unittest.cc",
>-      "vad/gmm_unittest.cc",
>-      "vad/pitch_based_vad_unittest.cc",
>-      "vad/pitch_internal_unittest.cc",
>-      "vad/pole_zero_filter_unittest.cc",
>-      "vad/standalone_vad_unittest.cc",
>-      "vad/vad_audio_proc_unittest.cc",
>-      "vad/vad_circular_buffer_unittest.cc",
>-      "vad/voice_activity_detector_unittest.cc",
>     ]
> 
>     deps = [
>-      ":aec_core",
>       ":analog_mic_simulation",
>       ":apm_logging",
>+      ":audio_frame_view",
>       ":audio_processing",
>       ":audioproc_test_utils",
>+      ":file_audio_generator_unittests",
>       ":mocks",
>-      "..:module_api",
>       "../..:webrtc_common",
>-      "../../:typedefs",
>       "../../api:array_view",
>-      "../../api:optional",
>+      "../../api/audio:aec3_config",
>+      "../../api/audio:aec3_factory",
>       "../../common_audio:common_audio",
>+      "../../common_audio:common_audio_c",
>       "../../rtc_base:checks",
>       "../../rtc_base:gtest_prod",
>       "../../rtc_base:protobuf_utils",
>       "../../rtc_base:rtc_base",
>       "../../rtc_base:rtc_base_approved",
>+      "../../rtc_base:safe_minmax",
>+      "../../rtc_base/system:arch",
>+      "../../rtc_base/system:file_wrapper",
>       "../../system_wrappers",
>       "../../system_wrappers:cpu_features_api",
>+      "../../test:fileutils",
>       "../../test:test_support",
>       "../audio_coding:neteq_input_audio_tools",
>+      "aec:aec_core",
>+      "aec:aec_unittests",
>       "aec_dump:mock_aec_dump_unittests",
>+      "agc:agc_unittests",
>+      "agc2:adaptive_digital_unittests",
>+      "agc2:biquad_filter_unittests",
>+      "agc2:fixed_digital_unittests",
>+      "agc2:noise_estimator_unittests",
>+      "agc2/rnn_vad:unittests",
>       "test/conversational_speech:unittest",
>-      "//testing/gmock",
>+      "utility:block_mean_calculator_unittest",
>+      "utility:legacy_delay_estimator_unittest",
>+      "vad:vad_unittests",
>       "//testing/gtest",
>+      "//third_party/abseil-cpp/absl/memory",
>+      "//third_party/abseil-cpp/absl/types:optional",
>     ]
> 
>     defines = []
>@@ -636,49 +445,16 @@ if (rtc_include_tests) {
>         ":audioproc_protobuf_utils",
>         ":audioproc_test_utils",
>         ":audioproc_unittest_proto",
>+        "../../api/audio:audio_frame_api",
>+        "../../rtc_base:rtc_base_tests_utils",
>         "../../rtc_base:rtc_task_queue",
>         "aec_dump",
>         "aec_dump:aec_dump_unittests",
>       ]
>       sources += [
>-        "aec3/adaptive_fir_filter_unittest.cc",
>-        "aec3/aec3_fft_unittest.cc",
>-        "aec3/aec_state_unittest.cc",
>-        "aec3/block_framer_unittest.cc",
>-        "aec3/block_processor_metrics_unittest.cc",
>-        "aec3/block_processor_unittest.cc",
>-        "aec3/cascaded_biquad_filter_unittest.cc",
>-        "aec3/comfort_noise_generator_unittest.cc",
>-        "aec3/decimator_unittest.cc",
>-        "aec3/echo_canceller3_unittest.cc",
>-        "aec3/echo_path_delay_estimator_unittest.cc",
>-        "aec3/echo_path_variability_unittest.cc",
>-        "aec3/echo_remover_metrics_unittest.cc",
>-        "aec3/echo_remover_unittest.cc",
>-        "aec3/erl_estimator_unittest.cc",
>-        "aec3/erle_estimator_unittest.cc",
>-        "aec3/fft_data_unittest.cc",
>-        "aec3/frame_blocker_unittest.cc",
>-        "aec3/main_filter_update_gain_unittest.cc",
>-        "aec3/matched_filter_lag_aggregator_unittest.cc",
>-        "aec3/matched_filter_unittest.cc",
>-        "aec3/output_selector_unittest.cc",
>-        "aec3/render_buffer_unittest.cc",
>-        "aec3/render_delay_buffer_unittest.cc",
>-        "aec3/render_delay_controller_metrics_unittest.cc",
>-        "aec3/render_delay_controller_unittest.cc",
>-        "aec3/render_signal_analyzer_unittest.cc",
>-        "aec3/residual_echo_estimator_unittest.cc",
>-        "aec3/shadow_filter_update_gain_unittest.cc",
>-        "aec3/subtractor_unittest.cc",
>-        "aec3/suppression_filter_unittest.cc",
>-        "aec3/suppression_gain_unittest.cc",
>-        "aec3/vector_math_unittest.cc",
>-        "agc2/gain_controller2_unittest.cc",
>         "audio_processing_impl_locking_unittest.cc",
>         "audio_processing_impl_unittest.cc",
>         "audio_processing_unittest.cc",
>-        "beamformer/nonlinear_beamformer_unittest.cc",
>         "echo_cancellation_bit_exact_unittest.cc",
>         "echo_control_mobile_unittest.cc",
>         "echo_detector/circular_buffer_unittest.cc",
>@@ -686,7 +462,6 @@ if (rtc_include_tests) {
>         "echo_detector/moving_max_unittest.cc",
>         "echo_detector/normalized_covariance_estimator_unittest.cc",
>         "gain_control_unittest.cc",
>-        "level_controller/level_controller_unittest.cc",
>         "level_estimator_unittest.cc",
>         "low_cut_filter_unittest.cc",
>         "noise_suppression_unittest.cc",
>@@ -714,13 +489,11 @@ if (rtc_include_tests) {
> 
>     sources = [
>       "audio_processing_performance_unittest.cc",
>-      "level_controller/level_controller_complexity_unittest.cc",
>     ]
>     deps = [
>       ":audio_processing",
>       ":audioproc_test_utils",
>       "../../api:array_view",
>-      "../../modules:module_api",
>       "../../rtc_base:protobuf_utils",
>       "../../rtc_base:rtc_base_approved",
>       "../../system_wrappers",
>@@ -735,6 +508,23 @@ if (rtc_include_tests) {
>     }
>   }
> 
>+  rtc_source_set("file_audio_generator_unittests") {
>+    testonly = true
>+
>+    sources = [
>+      "audio_generator/file_audio_generator_unittest.cc",
>+    ]
>+
>+    deps = [
>+      ":audio_generator_factory",
>+      ":audio_processing",
>+      ":file_audio_generator",
>+      "../../rtc_base:rtc_base_approved",
>+      "../../test:fileutils",
>+      "../../test:test_support",
>+    ]
>+  }
>+
>   rtc_source_set("analog_mic_simulation") {
>     sources = [
>       "test/fake_recording_device.cc",
>@@ -742,22 +532,27 @@ if (rtc_include_tests) {
>     ]
>     deps = [
>       "../../api:array_view",
>+      "../../api/audio:audio_frame_api",
>       "../../common_audio:common_audio",
>-      "../../modules:module_api",
>       "../../rtc_base:checks",
>       "../../rtc_base:rtc_base_approved",
>+      "../../rtc_base:safe_minmax",
>+      "agc:gain_map",
>+      "//third_party/abseil-cpp/absl/memory",
>+      "//third_party/abseil-cpp/absl/types:optional",
>     ]
>   }
> 
>   if (rtc_enable_protobuf) {
>-    rtc_executable("audioproc_f") {
>+    rtc_source_set("audioproc_f_impl") {
>       testonly = true
>       sources = [
>         "test/aec_dump_based_simulator.cc",
>         "test/aec_dump_based_simulator.h",
>         "test/audio_processing_simulator.cc",
>         "test/audio_processing_simulator.h",
>-        "test/audioproc_float.cc",
>+        "test/audioproc_float_impl.cc",
>+        "test/audioproc_float_impl.h",
>         "test/wav_based_simulator.cc",
>         "test/wav_based_simulator.h",
>       ]
>@@ -768,11 +563,12 @@ if (rtc_include_tests) {
>         ":audioproc_debug_proto",
>         ":audioproc_protobuf_utils",
>         ":audioproc_test_utils",
>-        "../../api:optional",
>+        "../../api/audio:aec3_factory",
>         "../../common_audio:common_audio",
>         "../../rtc_base:checks",
>         "../../rtc_base:protobuf_utils",
>         "../../rtc_base:rtc_base_approved",
>+        "../../rtc_base:rtc_json",
>         "../../rtc_base:rtc_task_queue",
>         "../../rtc_base:stringutils",
>         "../../system_wrappers",
>@@ -781,6 +577,20 @@ if (rtc_include_tests) {
>         "aec_dump",
>         "aec_dump:aec_dump_impl",
>         "//testing/gtest",
>+        "//third_party/abseil-cpp/absl/memory",
>+        "//third_party/abseil-cpp/absl/types:optional",
>+      ]
>+    }  # audioproc_f_impl
>+    rtc_executable("audioproc_f") {
>+      testonly = true
>+      sources = [
>+        "test/audioproc_float_main.cc",
>+      ]
>+      deps = [
>+        ":audio_processing",
>+        "../../api:audioproc_f_api",
>+        "../../rtc_base:rtc_base_approved",
>+        "//third_party/abseil-cpp/absl/memory",
>       ]
>     }  # audioproc_f
>   }
>@@ -803,16 +613,18 @@ if (rtc_include_tests) {
> 
>     deps = [
>       ":audio_processing",
>-      "..:module_api",
>       "../../api:array_view",
>-      "../../api:optional",
>+      "../../api/audio:audio_frame_api",
>       "../../common_audio",
>       "../../rtc_base:checks",
>       "../../rtc_base:rtc_base_approved",
>+      "../../rtc_base/system:arch",
>       "../../system_wrappers",
>+      "../../test:fileutils",
>       "../../test:test_support",
>       "../audio_coding:neteq_input_audio_tools",
>       "//testing/gtest",
>+      "//third_party/abseil-cpp/absl/types:optional",
>     ]
>   }
> 
>@@ -825,14 +637,15 @@ if (rtc_include_tests) {
>     ]
>     deps = [
>       ":audio_processing",
>-      "..:module_api",
>       "../..:webrtc_common",
>-      "../../:typedefs",
>       "../../common_audio:common_audio",
>       "../../rtc_base:rtc_base_approved",
>+      "../../rtc_base/system:file_wrapper",
>       "../../system_wrappers",
>       "../../system_wrappers:metrics_default",
>+      "../../test:fileutils",
>       "../../test:test_support",
>+      "agc:level_estimation",
>       "//testing/gtest",
>     ]
>   }
>@@ -847,27 +660,12 @@ if (rtc_include_tests) {
>     deps = [
>       ":audio_processing",
>       "../..:webrtc_common",
>-      "../../:typedefs",
>+      "../../rtc_base/system:file_wrapper",
>       "../../system_wrappers",
>       "../../system_wrappers:metrics_default",
>     ]
>   }
> 
>-  rtc_executable("nonlinear_beamformer_test") {
>-    testonly = true
>-    sources = [
>-      "beamformer/nonlinear_beamformer_test.cc",
>-    ]
>-    deps = [
>-      ":audio_processing",
>-      ":audioproc_test_utils",
>-      "../../common_audio:common_audio",
>-      "../../rtc_base:checks",
>-      "../../rtc_base:rtc_base_approved",
>-      "../../system_wrappers:metrics_default",
>-    ]
>-  }
>-
>   if (rtc_enable_intelligibility_enhancer) {
>     rtc_executable("intelligibility_proc") {
>       testonly = true
>@@ -877,6 +675,7 @@ if (rtc_include_tests) {
>       deps = [
>         ":audio_processing",
>         ":audioproc_test_utils",
>+        "../../common_audio",
>         "../../rtc_base:rtc_base_approved",
>         "../../system_wrappers:metrics_default",
>         "../../test:test_support",
>@@ -902,9 +701,9 @@ if (rtc_include_tests) {
>       deps = [
>         ":audioproc_debug_proto",
>         "../..:webrtc_common",
>-        "../../:typedefs",
>         "../../rtc_base:protobuf_utils",
>         "../../rtc_base:rtc_base_approved",
>+        "../../rtc_base/system:arch",
>       ]
>     }
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/OWNERS
>index 9c521d230ae..9a82da61a3c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/OWNERS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/OWNERS
>@@ -1,7 +1,12 @@
> aleloi@webrtc.org
>+alessiob@webrtc.org
> aluebs@webrtc.org
>+gustaf@webrtc.org
> henrik.lundin@webrtc.org
>+ivoc@webrtc.org
>+minyue@webrtc.org
> peah@webrtc.org
>+saza@webrtc.org
> 
> # These are for the common case of adding or renaming files. If you're doing
> # structural changes, please get a review from a reviewer in this file.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/BUILD.gn
>new file mode 100644
>index 00000000000..52441d7fdad
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/BUILD.gn
>@@ -0,0 +1,100 @@
>+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+#
>+# Use of this source code is governed by a BSD-style license
>+# that can be found in the LICENSE file in the root of the source
>+# tree. An additional intellectual property rights grant can be found
>+# in the file PATENTS.  All contributing project authors may
>+# be found in the AUTHORS file in the root of the source tree.
>+
>+import("../../../webrtc.gni")
>+
>+rtc_source_set("aec") {
>+  configs += [ "..:apm_debug_dump" ]
>+  sources = [
>+    "aec_resampler.cc",
>+    "aec_resampler.h",
>+    "echo_cancellation.cc",
>+    "echo_cancellation.h",
>+  ]
>+  deps = [
>+    ":aec_core",
>+    "..:apm_logging",
>+    "../../../common_audio:common_audio_c",
>+    "../../../rtc_base:checks",
>+    "../../../rtc_base:rtc_base_approved",
>+  ]
>+}
>+
>+rtc_source_set("aec_core") {
>+  configs += [ "..:apm_debug_dump" ]
>+  sources = [
>+    "aec_common.h",
>+    "aec_core.cc",
>+    "aec_core.h",
>+    "aec_core_optimized_methods.h",
>+  ]
>+  deps = [
>+    "..:apm_logging",
>+    "../../../common_audio:common_audio_c",
>+    "../../../rtc_base:checks",
>+    "../../../rtc_base:rtc_base_approved",
>+    "../../../rtc_base/system:arch",
>+    "../../../system_wrappers:cpu_features_api",
>+    "../../../system_wrappers:metrics_api",
>+    "../utility:block_mean_calculator",
>+    "../utility:legacy_delay_estimator",
>+    "../utility:ooura_fft",
>+  ]
>+  cflags = []
>+
>+  if (current_cpu == "x86" || current_cpu == "x64") {
>+    sources += [ "aec_core_sse2.cc" ]
>+    if (is_posix || is_fuchsia) {
>+      cflags += [ "-msse2" ]
>+    }
>+  }
>+
>+  if (rtc_build_with_neon) {
>+    sources += [ "aec_core_neon.cc" ]
>+
>+    if (current_cpu != "arm64") {
>+      # Enable compilation for the NEON instruction set.
>+      suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
>+      cflags += [ "-mfpu=neon" ]
>+    }
>+
>+    # Disable LTO on NEON targets due to compiler bug.
>+    # TODO(fdegans): Enable this. See crbug.com/408997.
>+    if (rtc_use_lto) {
>+      cflags -= [
>+        "-flto",
>+        "-ffat-lto-objects",
>+      ]
>+    }
>+
>+    deps += [ "../../../common_audio" ]
>+  }
>+
>+  if (current_cpu == "mipsel" && mips_float_abi == "hard") {
>+    sources += [ "aec_core_mips.cc" ]
>+  }
>+}
>+
>+if (rtc_include_tests) {
>+  rtc_source_set("aec_unittests") {
>+    testonly = true
>+
>+    sources = [
>+      "echo_cancellation_unittest.cc",
>+      "system_delay_unittest.cc",
>+    ]
>+    deps = [
>+      ":aec",
>+      ":aec_core",
>+      "../../../rtc_base:checks",
>+      "../../../rtc_base:rtc_base_approved",
>+      "../../../test:test_support",
>+      "//testing/gtest",
>+    ]
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/aec_common.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/aec_common.h
>index 80c5c14813b..ac1f3394562 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/aec_common.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/aec_common.h
>@@ -11,8 +11,6 @@
> #ifndef MODULES_AUDIO_PROCESSING_AEC_AEC_COMMON_H_
> #define MODULES_AUDIO_PROCESSING_AEC_AEC_COMMON_H_
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>-
> #ifdef _MSC_VER /* visual c++ */
> #define ALIGN16_BEG __declspec(align(16))
> #define ALIGN16_END
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/aec_core.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/aec_core.cc
>index c9bd7d36856..5a8cf8f58b6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/aec_core.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/aec_core.cc
>@@ -14,11 +14,11 @@
> 
> #include "modules/audio_processing/aec/aec_core.h"
> 
>-#include <algorithm>
> #include <math.h>
> #include <stddef.h>  // size_t
> #include <stdlib.h>
> #include <string.h>
>+#include <algorithm>
> 
> #include "rtc_base/checks.h"
> extern "C" {
>@@ -30,9 +30,9 @@ extern "C" {
> #include "modules/audio_processing/logging/apm_data_dumper.h"
> #include "modules/audio_processing/utility/delay_estimator_wrapper.h"
> #include "rtc_base/checks.h"
>+#include "rtc_base/system/arch.h"
> #include "system_wrappers/include/cpu_features_wrapper.h"
> #include "system_wrappers/include/metrics.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> namespace {
>@@ -178,9 +178,7 @@ __inline static float MulIm(float aRe, float aIm, float bRe, float bIm) {
> // window, of which the length is 1 unit longer than indicated. Remove "+1" when
> // the code is refactored.
> PowerLevel::PowerLevel()
>-    : framelevel(kSubCountLen + 1),
>-      averagelevel(kCountLen + 1) {
>-}
>+    : framelevel(kSubCountLen + 1), averagelevel(kCountLen + 1) {}
> 
> BlockBuffer::BlockBuffer() {
>   buffer_ = WebRtc_CreateBuffer(kBufferSizeBlocks, sizeof(float) * PART_LEN);
>@@ -238,10 +236,7 @@ size_t BlockBuffer::AvaliableSpace() {
> }
> 
> DivergentFilterFraction::DivergentFilterFraction()
>-    : count_(0),
>-      occurrence_(0),
>-      fraction_(-1.0) {
>-}
>+    : count_(0), occurrence_(0), fraction_(-1.0) {}
> 
> void DivergentFilterFraction::Reset() {
>   Clear();
>@@ -254,19 +249,18 @@ void DivergentFilterFraction::AddObservation(const PowerLevel& nearlevel,
>   const float near_level = nearlevel.framelevel.GetLatestMean();
>   const float level_increase =
>       linoutlevel.framelevel.GetLatestMean() - near_level;
>-  const bool output_signal_active = nlpoutlevel.framelevel.GetLatestMean() >
>-          40.0 * nlpoutlevel.minlevel;
>+  const bool output_signal_active =
>+      nlpoutlevel.framelevel.GetLatestMean() > 40.0 * nlpoutlevel.minlevel;
>   // Level increase should be, in principle, negative, when the filter
>   // does not diverge. Here we allow some margin (0.01 * near end level) and
>   // numerical error (1.0). We count divergence only when the AEC output
>   // signal is active.
>-  if (output_signal_active &&
>-      level_increase > std::max(0.01 * near_level, 1.0))
>+  if (output_signal_active && level_increase > std::max(0.01 * near_level, 1.0))
>     occurrence_++;
>   ++count_;
>   if (count_ == kDivergentFilterFractionAggregationWindowSize) {
>     fraction_ = static_cast<float>(occurrence_) /
>-        kDivergentFilterFractionAggregationWindowSize;
>+                kDivergentFilterFractionAggregationWindowSize;
>     Clear();
>   }
> }
>@@ -416,9 +410,9 @@ static void Suppress(const float hNl[PART_LEN1], float efw[2][PART_LEN1]) {
>   }
> }
> 
>-static int PartitionDelay(int num_partitions,
>-                          float h_fft_buf[2]
>-                                         [kExtendedNumPartitions * PART_LEN1]) {
>+static int PartitionDelay(
>+    int num_partitions,
>+    float h_fft_buf[2][kExtendedNumPartitions * PART_LEN1]) {
>   // Measures the energy in each filter partition and returns the partition with
>   // highest energy.
>   // TODO(bjornv): Spread computational cost by computing one partition per
>@@ -445,7 +439,8 @@ static int PartitionDelay(int num_partitions,
> }
> 
> // Update metric with 10 * log10(numerator / denominator).
>-static void UpdateLogRatioMetric(Stats* metric, float numerator,
>+static void UpdateLogRatioMetric(Stats* metric,
>+                                 float numerator,
>                                  float denominator) {
>   RTC_DCHECK(metric);
>   RTC_CHECK(numerator >= 0);
>@@ -736,9 +731,8 @@ static void UpdateMetrics(AecCore* aec) {
>   }
> 
>   if (aec->linoutlevel.framelevel.EndOfBlock()) {
>-    aec->divergent_filter_fraction.AddObservation(aec->nearlevel,
>-                                                  aec->linoutlevel,
>-                                                  aec->nlpoutlevel);
>+    aec->divergent_filter_fraction.AddObservation(
>+        aec->nearlevel, aec->linoutlevel, aec->nlpoutlevel);
>   }
> 
>   if (aec->farlevel.averagelevel.EndOfBlock()) {
>@@ -755,7 +749,6 @@ static void UpdateMetrics(AecCore* aec) {
>     if ((aec->stateCounter > (0.5f * kCountLen * kSubCountLen)) &&
>         (aec->farlevel.framelevel.EndOfBlock()) &&
>         (far_average_level > (actThreshold * aec->farlevel.minlevel))) {
>-
>       // ERL: error return loss.
>       const float near_average_level =
>           aec->nearlevel.averagelevel.GetLatestMean();
>@@ -815,9 +808,9 @@ static void UpdateDelayMetrics(AecCore* self) {
>   for (i = 0; i < kHistorySizeBlocks; i++) {
>     l1_norm += abs(i - median) * self->delay_histogram[i];
>   }
>-  self->delay_std =
>-      static_cast<int>((l1_norm + self->num_delay_values / 2) /
>-                       self->num_delay_values) * kMsPerBlock;
>+  self->delay_std = static_cast<int>((l1_norm + self->num_delay_values / 2) /
>+                                     self->num_delay_values) *
>+                    kMsPerBlock;
> 
>   // Determine fraction of delays that are out of bounds, that is, either
>   // negative (anti-causal system) or larger than the AEC filter length.
>@@ -949,11 +942,11 @@ static int SignalBasedDelayCorrection(AecCore* self) {
>   return delay_correction;
> }
> 
>-static void RegressorPower(int num_partitions,
>-                           int latest_added_partition,
>-                           float x_fft_buf[2]
>-                                          [kExtendedNumPartitions * PART_LEN1],
>-                           float x_pow[PART_LEN1]) {
>+static void RegressorPower(
>+    int num_partitions,
>+    int latest_added_partition,
>+    float x_fft_buf[2][kExtendedNumPartitions * PART_LEN1],
>+    float x_pow[PART_LEN1]) {
>   RTC_DCHECK_LT(latest_added_partition, num_partitions);
>   memset(x_pow, 0, PART_LEN1 * sizeof(x_pow[0]));
> 
>@@ -976,21 +969,20 @@ static void RegressorPower(int num_partitions,
>   }
> }
> 
>-static void EchoSubtraction(const OouraFft& ooura_fft,
>-                            int num_partitions,
>-                            int extended_filter_enabled,
>-                            int* extreme_filter_divergence,
>-                            float filter_step_size,
>-                            float error_threshold,
>-                            float* x_fft,
>-                            int* x_fft_buf_block_pos,
>-                            float x_fft_buf[2]
>-                                           [kExtendedNumPartitions * PART_LEN1],
>-                            float* const y,
>-                            float x_pow[PART_LEN1],
>-                            float h_fft_buf[2]
>-                                           [kExtendedNumPartitions * PART_LEN1],
>-                            float echo_subtractor_output[PART_LEN]) {
>+static void EchoSubtraction(
>+    const OouraFft& ooura_fft,
>+    int num_partitions,
>+    int extended_filter_enabled,
>+    int* extreme_filter_divergence,
>+    float filter_step_size,
>+    float error_threshold,
>+    float* x_fft,
>+    int* x_fft_buf_block_pos,
>+    float x_fft_buf[2][kExtendedNumPartitions * PART_LEN1],
>+    float* const y,
>+    float x_pow[PART_LEN1],
>+    float h_fft_buf[2][kExtendedNumPartitions * PART_LEN1],
>+    float echo_subtractor_output[PART_LEN]) {
>   float s_fft[2][PART_LEN1];
>   float e_extended[PART_LEN2];
>   float s_extended[PART_LEN2];
>@@ -1095,6 +1087,7 @@ static void FormSuppressionGain(AecCore* aec,
>     } else {
>       for (int i = 0; i < PART_LEN1; ++i) {
>         hNl[i] = 1 - cohxd[i];
>+        hNl[i] = std::max(hNl[i], 0.f);
>       }
>       hNlFb = hNlXdAvg;
>       hNlFbLow = hNlXdAvg;
>@@ -1109,6 +1102,7 @@ static void FormSuppressionGain(AecCore* aec,
>       aec->echoState = 1;
>       for (int i = 0; i < PART_LEN1; ++i) {
>         hNl[i] = WEBRTC_SPL_MIN(cohde[i], 1 - cohxd[i]);
>+        hNl[i] = std::max(hNl[i], 0.f);
>       }
> 
>       // Select an order statistic from the preferred bands.
>@@ -1116,10 +1110,10 @@ static void FormSuppressionGain(AecCore* aec,
>       // preferred.
>       memcpy(hNlPref, &hNl[minPrefBand], sizeof(float) * prefBandSize);
>       qsort(hNlPref, prefBandSize, sizeof(float), CmpFloat);
>-      hNlFb = hNlPref[static_cast<int>(floor(prefBandQuant *
>-                                             (prefBandSize - 1)))];
>-      hNlFbLow = hNlPref[static_cast<int>(floor(prefBandQuantLow *
>-                                                (prefBandSize - 1)))];
>+      hNlFb =
>+          hNlPref[static_cast<int>(floor(prefBandQuant * (prefBandSize - 1)))];
>+      hNlFbLow = hNlPref[static_cast<int>(
>+          floor(prefBandQuantLow * (prefBandSize - 1)))];
>     }
>   }
> 
>@@ -1140,10 +1134,10 @@ static void FormSuppressionGain(AecCore* aec,
>   if (aec->hNlMinCtr == 2) {
>     aec->hNlNewMin = 0;
>     aec->hNlMinCtr = 0;
>-    aec->overDrive =
>-        WEBRTC_SPL_MAX(kTargetSupp[aec->nlp_mode] /
>-                       static_cast<float>(log(aec->hNlFbMin + 1e-10f) + 1e-10f),
>-                       min_overdrive[aec->nlp_mode]);
>+    aec->overDrive = WEBRTC_SPL_MAX(
>+        kTargetSupp[aec->nlp_mode] /
>+            static_cast<float>(log(aec->hNlFbMin + 1e-10f) + 1e-10f),
>+        min_overdrive[aec->nlp_mode]);
>   }
> 
>   // Smooth the overdrive.
>@@ -1771,20 +1765,18 @@ void BufferNearendFrame(
>     float nearend_buffer[NUM_HIGH_BANDS_MAX + 1]
>                         [PART_LEN - (FRAME_LEN - PART_LEN)]) {
>   for (size_t i = 0; i < num_bands; ++i) {
>-    memcpy(
>-        &nearend_buffer[i][0],
>-        &nearend_frame[i]
>-                      [nearend_start_index + FRAME_LEN - num_samples_to_buffer],
>-        num_samples_to_buffer * sizeof(float));
>+    memcpy(&nearend_buffer[i][0],
>+           &nearend_frame[i][nearend_start_index + FRAME_LEN -
>+                             num_samples_to_buffer],
>+           num_samples_to_buffer * sizeof(float));
>   }
> }
> 
>-void BufferOutputBlock(size_t num_bands,
>-                       const float output_block[NUM_HIGH_BANDS_MAX + 1]
>-                                               [PART_LEN],
>-                       size_t* output_buffer_size,
>-                       float output_buffer[NUM_HIGH_BANDS_MAX + 1]
>-                                          [2 * PART_LEN]) {
>+void BufferOutputBlock(
>+    size_t num_bands,
>+    const float output_block[NUM_HIGH_BANDS_MAX + 1][PART_LEN],
>+    size_t* output_buffer_size,
>+    float output_buffer[NUM_HIGH_BANDS_MAX + 1][2 * PART_LEN]) {
>   for (size_t i = 0; i < num_bands; ++i) {
>     memcpy(&output_buffer[i][*output_buffer_size], &output_block[i][0],
>            PART_LEN * sizeof(float));
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/aec_core.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/aec_core.h
>index 78596ec4b45..35a6c9b2164 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/aec_core.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/aec_core.h
>@@ -22,12 +22,10 @@
> extern "C" {
> #include "common_audio/ring_buffer.h"
> }
>-#include "common_audio/wav_file.h"
> #include "modules/audio_processing/aec/aec_common.h"
> #include "modules/audio_processing/utility/block_mean_calculator.h"
> #include "modules/audio_processing/utility/ooura_fft.h"
> #include "rtc_base/constructormagic.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/aec_core_mips.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/aec_core_mips.cc
>index ebe6349e935..bf89cfaf93b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/aec_core_mips.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/aec_core_mips.cc
>@@ -51,95 +51,94 @@ void WebRtcAec_FilterFar_mips(
>     int len = PART_LEN1 >> 1;
> 
>     __asm __volatile(
>-      ".set       push                                                \n\t"
>-      ".set       noreorder                                           \n\t"
>-      "1:                                                             \n\t"
>-      "lwc1       %[f0],      0(%[aRe])                               \n\t"
>-      "lwc1       %[f1],      0(%[bRe])                               \n\t"
>-      "lwc1       %[f2],      0(%[bIm])                               \n\t"
>-      "lwc1       %[f3],      0(%[aIm])                               \n\t"
>-      "lwc1       %[f4],      4(%[aRe])                               \n\t"
>-      "lwc1       %[f5],      4(%[bRe])                               \n\t"
>-      "lwc1       %[f6],      4(%[bIm])                               \n\t"
>-      "mul.s      %[f8],      %[f0],          %[f1]                   \n\t"
>-      "mul.s      %[f0],      %[f0],          %[f2]                   \n\t"
>-      "mul.s      %[f9],      %[f4],          %[f5]                   \n\t"
>-      "mul.s      %[f4],      %[f4],          %[f6]                   \n\t"
>-      "lwc1       %[f7],      4(%[aIm])                               \n\t"
>+        ".set       push                                                \n\t"
>+        ".set       noreorder                                           \n\t"
>+        "1:                                                             \n\t"
>+        "lwc1       %[f0],      0(%[aRe])                               \n\t"
>+        "lwc1       %[f1],      0(%[bRe])                               \n\t"
>+        "lwc1       %[f2],      0(%[bIm])                               \n\t"
>+        "lwc1       %[f3],      0(%[aIm])                               \n\t"
>+        "lwc1       %[f4],      4(%[aRe])                               \n\t"
>+        "lwc1       %[f5],      4(%[bRe])                               \n\t"
>+        "lwc1       %[f6],      4(%[bIm])                               \n\t"
>+        "mul.s      %[f8],      %[f0],          %[f1]                   \n\t"
>+        "mul.s      %[f0],      %[f0],          %[f2]                   \n\t"
>+        "mul.s      %[f9],      %[f4],          %[f5]                   \n\t"
>+        "mul.s      %[f4],      %[f4],          %[f6]                   \n\t"
>+        "lwc1       %[f7],      4(%[aIm])                               \n\t"
> #if !defined(MIPS32_R2_LE)
>-      "mul.s      %[f12],     %[f2],          %[f3]                   \n\t"
>-      "mul.s      %[f1],      %[f3],          %[f1]                   \n\t"
>-      "mul.s      %[f11],     %[f6],          %[f7]                   \n\t"
>-      "addiu      %[aRe],     %[aRe],         8                       \n\t"
>-      "addiu      %[aIm],     %[aIm],         8                       \n\t"
>-      "addiu      %[len],     %[len],         -1                      \n\t"
>-      "sub.s      %[f8],      %[f8],          %[f12]                  \n\t"
>-      "mul.s      %[f12],     %[f7],          %[f5]                   \n\t"
>-      "lwc1       %[f2],      0(%[yf0])                               \n\t"
>-      "add.s      %[f1],      %[f0],          %[f1]                   \n\t"
>-      "lwc1       %[f3],      0(%[yf1])                               \n\t"
>-      "sub.s      %[f9],      %[f9],          %[f11]                  \n\t"
>-      "lwc1       %[f6],      4(%[yf0])                               \n\t"
>-      "add.s      %[f4],      %[f4],          %[f12]                  \n\t"
>-#else  // #if !defined(MIPS32_R2_LE)
>-      "addiu      %[aRe],     %[aRe],         8                       \n\t"
>-      "addiu      %[aIm],     %[aIm],         8                       \n\t"
>-      "addiu      %[len],     %[len],         -1                      \n\t"
>-      "nmsub.s    %[f8],      %[f8],          %[f2],      %[f3]       \n\t"
>-      "lwc1       %[f2],      0(%[yf0])                               \n\t"
>-      "madd.s     %[f1],      %[f0],          %[f3],      %[f1]       \n\t"
>-      "lwc1       %[f3],      0(%[yf1])                               \n\t"
>-      "nmsub.s    %[f9],      %[f9],          %[f6],      %[f7]       \n\t"
>-      "lwc1       %[f6],      4(%[yf0])                               \n\t"
>-      "madd.s     %[f4],      %[f4],          %[f7],      %[f5]       \n\t"
>+        "mul.s      %[f12],     %[f2],          %[f3]                   \n\t"
>+        "mul.s      %[f1],      %[f3],          %[f1]                   \n\t"
>+        "mul.s      %[f11],     %[f6],          %[f7]                   \n\t"
>+        "addiu      %[aRe],     %[aRe],         8                       \n\t"
>+        "addiu      %[aIm],     %[aIm],         8                       \n\t"
>+        "addiu      %[len],     %[len],         -1                      \n\t"
>+        "sub.s      %[f8],      %[f8],          %[f12]                  \n\t"
>+        "mul.s      %[f12],     %[f7],          %[f5]                   \n\t"
>+        "lwc1       %[f2],      0(%[yf0])                               \n\t"
>+        "add.s      %[f1],      %[f0],          %[f1]                   \n\t"
>+        "lwc1       %[f3],      0(%[yf1])                               \n\t"
>+        "sub.s      %[f9],      %[f9],          %[f11]                  \n\t"
>+        "lwc1       %[f6],      4(%[yf0])                               \n\t"
>+        "add.s      %[f4],      %[f4],          %[f12]                  \n\t"
>+#else   // #if !defined(MIPS32_R2_LE)
>+        "addiu      %[aRe],     %[aRe],         8                       \n\t"
>+        "addiu      %[aIm],     %[aIm],         8                       \n\t"
>+        "addiu      %[len],     %[len],         -1                      \n\t"
>+        "nmsub.s    %[f8],      %[f8],          %[f2],      %[f3]       \n\t"
>+        "lwc1       %[f2],      0(%[yf0])                               \n\t"
>+        "madd.s     %[f1],      %[f0],          %[f3],      %[f1]       \n\t"
>+        "lwc1       %[f3],      0(%[yf1])                               \n\t"
>+        "nmsub.s    %[f9],      %[f9],          %[f6],      %[f7]       \n\t"
>+        "lwc1       %[f6],      4(%[yf0])                               \n\t"
>+        "madd.s     %[f4],      %[f4],          %[f7],      %[f5]       \n\t"
> #endif  // #if !defined(MIPS32_R2_LE)
>-      "lwc1       %[f5],      4(%[yf1])                               \n\t"
>-      "add.s      %[f2],      %[f2],          %[f8]                   \n\t"
>-      "addiu      %[bRe],     %[bRe],         8                       \n\t"
>-      "addiu      %[bIm],     %[bIm],         8                       \n\t"
>-      "add.s      %[f3],      %[f3],          %[f1]                   \n\t"
>-      "add.s      %[f6],      %[f6],          %[f9]                   \n\t"
>-      "add.s      %[f5],      %[f5],          %[f4]                   \n\t"
>-      "swc1       %[f2],      0(%[yf0])                               \n\t"
>-      "swc1       %[f3],      0(%[yf1])                               \n\t"
>-      "swc1       %[f6],      4(%[yf0])                               \n\t"
>-      "swc1       %[f5],      4(%[yf1])                               \n\t"
>-      "addiu      %[yf0],     %[yf0],         8                       \n\t"
>-      "bgtz       %[len],     1b                                      \n\t"
>-      " addiu     %[yf1],     %[yf1],         8                       \n\t"
>-      "lwc1       %[f0],      0(%[aRe])                               \n\t"
>-      "lwc1       %[f1],      0(%[bRe])                               \n\t"
>-      "lwc1       %[f2],      0(%[bIm])                               \n\t"
>-      "lwc1       %[f3],      0(%[aIm])                               \n\t"
>-      "mul.s      %[f8],      %[f0],          %[f1]                   \n\t"
>-      "mul.s      %[f0],      %[f0],          %[f2]                   \n\t"
>+        "lwc1       %[f5],      4(%[yf1])                               \n\t"
>+        "add.s      %[f2],      %[f2],          %[f8]                   \n\t"
>+        "addiu      %[bRe],     %[bRe],         8                       \n\t"
>+        "addiu      %[bIm],     %[bIm],         8                       \n\t"
>+        "add.s      %[f3],      %[f3],          %[f1]                   \n\t"
>+        "add.s      %[f6],      %[f6],          %[f9]                   \n\t"
>+        "add.s      %[f5],      %[f5],          %[f4]                   \n\t"
>+        "swc1       %[f2],      0(%[yf0])                               \n\t"
>+        "swc1       %[f3],      0(%[yf1])                               \n\t"
>+        "swc1       %[f6],      4(%[yf0])                               \n\t"
>+        "swc1       %[f5],      4(%[yf1])                               \n\t"
>+        "addiu      %[yf0],     %[yf0],         8                       \n\t"
>+        "bgtz       %[len],     1b                                      \n\t"
>+        " addiu     %[yf1],     %[yf1],         8                       \n\t"
>+        "lwc1       %[f0],      0(%[aRe])                               \n\t"
>+        "lwc1       %[f1],      0(%[bRe])                               \n\t"
>+        "lwc1       %[f2],      0(%[bIm])                               \n\t"
>+        "lwc1       %[f3],      0(%[aIm])                               \n\t"
>+        "mul.s      %[f8],      %[f0],          %[f1]                   \n\t"
>+        "mul.s      %[f0],      %[f0],          %[f2]                   \n\t"
> #if !defined(MIPS32_R2_LE)
>-      "mul.s      %[f12],     %[f2],          %[f3]                   \n\t"
>-      "mul.s      %[f1],      %[f3],          %[f1]                   \n\t"
>-      "sub.s      %[f8],      %[f8],          %[f12]                  \n\t"
>-      "lwc1       %[f2],      0(%[yf0])                               \n\t"
>-      "add.s      %[f1],      %[f0],          %[f1]                   \n\t"
>-      "lwc1       %[f3],      0(%[yf1])                               \n\t"
>-#else  // #if !defined(MIPS32_R2_LE)
>-      "nmsub.s    %[f8],      %[f8],          %[f2],      %[f3]       \n\t"
>-      "lwc1       %[f2],      0(%[yf0])                               \n\t"
>-      "madd.s     %[f1],      %[f0],          %[f3],      %[f1]       \n\t"
>-      "lwc1       %[f3],      0(%[yf1])                               \n\t"
>+        "mul.s      %[f12],     %[f2],          %[f3]                   \n\t"
>+        "mul.s      %[f1],      %[f3],          %[f1]                   \n\t"
>+        "sub.s      %[f8],      %[f8],          %[f12]                  \n\t"
>+        "lwc1       %[f2],      0(%[yf0])                               \n\t"
>+        "add.s      %[f1],      %[f0],          %[f1]                   \n\t"
>+        "lwc1       %[f3],      0(%[yf1])                               \n\t"
>+#else   // #if !defined(MIPS32_R2_LE)
>+        "nmsub.s    %[f8],      %[f8],          %[f2],      %[f3]       \n\t"
>+        "lwc1       %[f2],      0(%[yf0])                               \n\t"
>+        "madd.s     %[f1],      %[f0],          %[f3],      %[f1]       \n\t"
>+        "lwc1       %[f3],      0(%[yf1])                               \n\t"
> #endif  // #if !defined(MIPS32_R2_LE)
>-      "add.s      %[f2],      %[f2],          %[f8]                   \n\t"
>-      "add.s      %[f3],      %[f3],          %[f1]                   \n\t"
>-      "swc1       %[f2],      0(%[yf0])                               \n\t"
>-      "swc1       %[f3],      0(%[yf1])                               \n\t"
>-      ".set       pop                                                 \n\t"
>-      : [f0] "=&f" (f0), [f1] "=&f" (f1), [f2] "=&f" (f2),
>-        [f3] "=&f" (f3), [f4] "=&f" (f4), [f5] "=&f" (f5),
>-        [f6] "=&f" (f6), [f7] "=&f" (f7), [f8] "=&f" (f8),
>-        [f9] "=&f" (f9), [f10] "=&f" (f10), [f11] "=&f" (f11),
>-        [f12] "=&f" (f12), [f13] "=&f" (f13), [aRe] "+r" (aRe),
>-        [aIm] "+r" (aIm), [bRe] "+r" (bRe), [bIm] "+r" (bIm),
>-        [yf0] "+r" (yf0), [yf1] "+r" (yf1), [len] "+r" (len)
>-      :
>-      : "memory");
>+        "add.s      %[f2],      %[f2],          %[f8]                   \n\t"
>+        "add.s      %[f3],      %[f3],          %[f1]                   \n\t"
>+        "swc1       %[f2],      0(%[yf0])                               \n\t"
>+        "swc1       %[f3],      0(%[yf1])                               \n\t"
>+        ".set       pop                                                 \n\t"
>+        : [f0] "=&f"(f0), [f1] "=&f"(f1), [f2] "=&f"(f2), [f3] "=&f"(f3),
>+          [f4] "=&f"(f4), [f5] "=&f"(f5), [f6] "=&f"(f6), [f7] "=&f"(f7),
>+          [f8] "=&f"(f8), [f9] "=&f"(f9), [f10] "=&f"(f10), [f11] "=&f"(f11),
>+          [f12] "=&f"(f12), [f13] "=&f"(f13), [aRe] "+r"(aRe), [aIm] "+r"(aIm),
>+          [bRe] "+r"(bRe), [bIm] "+r"(bIm), [yf0] "+r"(yf0), [yf1] "+r"(yf1),
>+          [len] "+r"(len)
>+        :
>+        : "memory");
>   }
> }
> 
>@@ -171,73 +170,71 @@ void WebRtcAec_FilterAdaptation_mips(
>     int len = PART_LEN >> 1;
> 
>     __asm __volatile(
>-      ".set       push                                                \n\t"
>-      ".set       noreorder                                           \n\t"
>-      "addiu      %[fft_tmp], %[fft],         0                       \n\t"
>-      "1:                                                             \n\t"
>-      "lwc1       %[f0],      0(%[aRe])                               \n\t"
>-      "lwc1       %[f1],      0(%[bRe])                               \n\t"
>-      "lwc1       %[f2],      0(%[bIm])                               \n\t"
>-      "lwc1       %[f4],      4(%[aRe])                               \n\t"
>-      "lwc1       %[f5],      4(%[bRe])                               \n\t"
>-      "lwc1       %[f6],      4(%[bIm])                               \n\t"
>-      "addiu      %[aRe],     %[aRe],         8                       \n\t"
>-      "addiu      %[bRe],     %[bRe],         8                       \n\t"
>-      "mul.s      %[f8],      %[f0],          %[f1]                   \n\t"
>-      "mul.s      %[f0],      %[f0],          %[f2]                   \n\t"
>-      "lwc1       %[f3],      0(%[aIm])                               \n\t"
>-      "mul.s      %[f9],      %[f4],          %[f5]                   \n\t"
>-      "lwc1       %[f7],      4(%[aIm])                               \n\t"
>-      "mul.s      %[f4],      %[f4],          %[f6]                   \n\t"
>+        ".set       push                                                \n\t"
>+        ".set       noreorder                                           \n\t"
>+        "addiu      %[fft_tmp], %[fft],         0                       \n\t"
>+        "1:                                                             \n\t"
>+        "lwc1       %[f0],      0(%[aRe])                               \n\t"
>+        "lwc1       %[f1],      0(%[bRe])                               \n\t"
>+        "lwc1       %[f2],      0(%[bIm])                               \n\t"
>+        "lwc1       %[f4],      4(%[aRe])                               \n\t"
>+        "lwc1       %[f5],      4(%[bRe])                               \n\t"
>+        "lwc1       %[f6],      4(%[bIm])                               \n\t"
>+        "addiu      %[aRe],     %[aRe],         8                       \n\t"
>+        "addiu      %[bRe],     %[bRe],         8                       \n\t"
>+        "mul.s      %[f8],      %[f0],          %[f1]                   \n\t"
>+        "mul.s      %[f0],      %[f0],          %[f2]                   \n\t"
>+        "lwc1       %[f3],      0(%[aIm])                               \n\t"
>+        "mul.s      %[f9],      %[f4],          %[f5]                   \n\t"
>+        "lwc1       %[f7],      4(%[aIm])                               \n\t"
>+        "mul.s      %[f4],      %[f4],          %[f6]                   \n\t"
> #if !defined(MIPS32_R2_LE)
>-      "mul.s      %[f10],     %[f3],          %[f2]                   \n\t"
>-      "mul.s      %[f1],      %[f3],          %[f1]                   \n\t"
>-      "mul.s      %[f11],     %[f7],          %[f6]                   \n\t"
>-      "mul.s      %[f5],      %[f7],          %[f5]                   \n\t"
>-      "addiu      %[aIm],     %[aIm],         8                       \n\t"
>-      "addiu      %[bIm],     %[bIm],         8                       \n\t"
>-      "addiu      %[len],     %[len],         -1                      \n\t"
>-      "add.s      %[f8],      %[f8],          %[f10]                  \n\t"
>-      "sub.s      %[f1],      %[f0],          %[f1]                   \n\t"
>-      "add.s      %[f9],      %[f9],          %[f11]                  \n\t"
>-      "sub.s      %[f5],      %[f4],          %[f5]                   \n\t"
>-#else  // #if !defined(MIPS32_R2_LE)
>-      "addiu      %[aIm],     %[aIm],         8                       \n\t"
>-      "addiu      %[bIm],     %[bIm],         8                       \n\t"
>-      "addiu      %[len],     %[len],         -1                      \n\t"
>-      "madd.s     %[f8],      %[f8],          %[f3],      %[f2]       \n\t"
>-      "nmsub.s    %[f1],      %[f0],          %[f3],      %[f1]       \n\t"
>-      "madd.s     %[f9],      %[f9],          %[f7],      %[f6]       \n\t"
>-      "nmsub.s    %[f5],      %[f4],          %[f7],      %[f5]       \n\t"
>+        "mul.s      %[f10],     %[f3],          %[f2]                   \n\t"
>+        "mul.s      %[f1],      %[f3],          %[f1]                   \n\t"
>+        "mul.s      %[f11],     %[f7],          %[f6]                   \n\t"
>+        "mul.s      %[f5],      %[f7],          %[f5]                   \n\t"
>+        "addiu      %[aIm],     %[aIm],         8                       \n\t"
>+        "addiu      %[bIm],     %[bIm],         8                       \n\t"
>+        "addiu      %[len],     %[len],         -1                      \n\t"
>+        "add.s      %[f8],      %[f8],          %[f10]                  \n\t"
>+        "sub.s      %[f1],      %[f0],          %[f1]                   \n\t"
>+        "add.s      %[f9],      %[f9],          %[f11]                  \n\t"
>+        "sub.s      %[f5],      %[f4],          %[f5]                   \n\t"
>+#else   // #if !defined(MIPS32_R2_LE)
>+        "addiu      %[aIm],     %[aIm],         8                       \n\t"
>+        "addiu      %[bIm],     %[bIm],         8                       \n\t"
>+        "addiu      %[len],     %[len],         -1                      \n\t"
>+        "madd.s     %[f8],      %[f8],          %[f3],      %[f2]       \n\t"
>+        "nmsub.s    %[f1],      %[f0],          %[f3],      %[f1]       \n\t"
>+        "madd.s     %[f9],      %[f9],          %[f7],      %[f6]       \n\t"
>+        "nmsub.s    %[f5],      %[f4],          %[f7],      %[f5]       \n\t"
> #endif  // #if !defined(MIPS32_R2_LE)
>-      "swc1       %[f8],      0(%[fft_tmp])                           \n\t"
>-      "swc1       %[f1],      4(%[fft_tmp])                           \n\t"
>-      "swc1       %[f9],      8(%[fft_tmp])                           \n\t"
>-      "swc1       %[f5],      12(%[fft_tmp])                          \n\t"
>-      "bgtz       %[len],     1b                                      \n\t"
>-      " addiu     %[fft_tmp], %[fft_tmp],     16                      \n\t"
>-      "lwc1       %[f0],      0(%[aRe])                               \n\t"
>-      "lwc1       %[f1],      0(%[bRe])                               \n\t"
>-      "lwc1       %[f2],      0(%[bIm])                               \n\t"
>-      "lwc1       %[f3],      0(%[aIm])                               \n\t"
>-      "mul.s      %[f8],      %[f0],          %[f1]                   \n\t"
>+        "swc1       %[f8],      0(%[fft_tmp])                           \n\t"
>+        "swc1       %[f1],      4(%[fft_tmp])                           \n\t"
>+        "swc1       %[f9],      8(%[fft_tmp])                           \n\t"
>+        "swc1       %[f5],      12(%[fft_tmp])                          \n\t"
>+        "bgtz       %[len],     1b                                      \n\t"
>+        " addiu     %[fft_tmp], %[fft_tmp],     16                      \n\t"
>+        "lwc1       %[f0],      0(%[aRe])                               \n\t"
>+        "lwc1       %[f1],      0(%[bRe])                               \n\t"
>+        "lwc1       %[f2],      0(%[bIm])                               \n\t"
>+        "lwc1       %[f3],      0(%[aIm])                               \n\t"
>+        "mul.s      %[f8],      %[f0],          %[f1]                   \n\t"
> #if !defined(MIPS32_R2_LE)
>-      "mul.s      %[f10],     %[f3],          %[f2]                   \n\t"
>-      "add.s      %[f8],      %[f8],          %[f10]                  \n\t"
>-#else  // #if !defined(MIPS32_R2_LE)
>-      "madd.s     %[f8],      %[f8],          %[f3],      %[f2]       \n\t"
>+        "mul.s      %[f10],     %[f3],          %[f2]                   \n\t"
>+        "add.s      %[f8],      %[f8],          %[f10]                  \n\t"
>+#else   // #if !defined(MIPS32_R2_LE)
>+        "madd.s     %[f8],      %[f8],          %[f3],      %[f2]       \n\t"
> #endif  // #if !defined(MIPS32_R2_LE)
>-      "swc1       %[f8],      4(%[fft])                               \n\t"
>-      ".set       pop                                                 \n\t"
>-      : [f0] "=&f" (f0), [f1] "=&f" (f1), [f2] "=&f" (f2),
>-        [f3] "=&f" (f3), [f4] "=&f" (f4), [f5] "=&f" (f5),
>-        [f6] "=&f" (f6), [f7] "=&f" (f7), [f8] "=&f" (f8),
>-        [f9] "=&f" (f9), [f10] "=&f" (f10), [f11] "=&f" (f11),
>-        [f12] "=&f" (f12), [aRe] "+r" (aRe), [aIm] "+r" (aIm),
>-        [bRe] "+r" (bRe), [bIm] "+r" (bIm), [fft_tmp] "=&r" (fft_tmp),
>-        [len] "+r" (len)
>-      : [fft] "r" (fft)
>-      : "memory");
>+        "swc1       %[f8],      4(%[fft])                               \n\t"
>+        ".set       pop                                                 \n\t"
>+        : [f0] "=&f"(f0), [f1] "=&f"(f1), [f2] "=&f"(f2), [f3] "=&f"(f3),
>+          [f4] "=&f"(f4), [f5] "=&f"(f5), [f6] "=&f"(f6), [f7] "=&f"(f7),
>+          [f8] "=&f"(f8), [f9] "=&f"(f9), [f10] "=&f"(f10), [f11] "=&f"(f11),
>+          [f12] "=&f"(f12), [aRe] "+r"(aRe), [aIm] "+r"(aIm), [bRe] "+r"(bRe),
>+          [bIm] "+r"(bIm), [fft_tmp] "=&r"(fft_tmp), [len] "+r"(len)
>+        : [fft] "r"(fft)
>+        : "memory");
> 
>     ooura_fft.InverseFft(fft);
>     memset(fft + PART_LEN, 0, sizeof(float) * PART_LEN);
>@@ -246,102 +243,101 @@ void WebRtcAec_FilterAdaptation_mips(
>     {
>       float scale = 2.0f / PART_LEN2;
>       __asm __volatile(
>+          ".set     push                                    \n\t"
>+          ".set     noreorder                               \n\t"
>+          "addiu    %[fft_tmp], %[fft],        0            \n\t"
>+          "addiu    %[len],     $zero,         8            \n\t"
>+          "1:                                               \n\t"
>+          "addiu    %[len],     %[len],        -1           \n\t"
>+          "lwc1     %[f0],      0(%[fft_tmp])               \n\t"
>+          "lwc1     %[f1],      4(%[fft_tmp])               \n\t"
>+          "lwc1     %[f2],      8(%[fft_tmp])               \n\t"
>+          "lwc1     %[f3],      12(%[fft_tmp])              \n\t"
>+          "mul.s    %[f0],      %[f0],         %[scale]     \n\t"
>+          "mul.s    %[f1],      %[f1],         %[scale]     \n\t"
>+          "mul.s    %[f2],      %[f2],         %[scale]     \n\t"
>+          "mul.s    %[f3],      %[f3],         %[scale]     \n\t"
>+          "lwc1     %[f4],      16(%[fft_tmp])              \n\t"
>+          "lwc1     %[f5],      20(%[fft_tmp])              \n\t"
>+          "lwc1     %[f6],      24(%[fft_tmp])              \n\t"
>+          "lwc1     %[f7],      28(%[fft_tmp])              \n\t"
>+          "mul.s    %[f4],      %[f4],         %[scale]     \n\t"
>+          "mul.s    %[f5],      %[f5],         %[scale]     \n\t"
>+          "mul.s    %[f6],      %[f6],         %[scale]     \n\t"
>+          "mul.s    %[f7],      %[f7],         %[scale]     \n\t"
>+          "swc1     %[f0],      0(%[fft_tmp])               \n\t"
>+          "swc1     %[f1],      4(%[fft_tmp])               \n\t"
>+          "swc1     %[f2],      8(%[fft_tmp])               \n\t"
>+          "swc1     %[f3],      12(%[fft_tmp])              \n\t"
>+          "swc1     %[f4],      16(%[fft_tmp])              \n\t"
>+          "swc1     %[f5],      20(%[fft_tmp])              \n\t"
>+          "swc1     %[f6],      24(%[fft_tmp])              \n\t"
>+          "swc1     %[f7],      28(%[fft_tmp])              \n\t"
>+          "bgtz     %[len],     1b                          \n\t"
>+          " addiu   %[fft_tmp], %[fft_tmp],    32           \n\t"
>+          ".set     pop                                     \n\t"
>+          : [f0] "=&f"(f0), [f1] "=&f"(f1), [f2] "=&f"(f2), [f3] "=&f"(f3),
>+            [f4] "=&f"(f4), [f5] "=&f"(f5), [f6] "=&f"(f6), [f7] "=&f"(f7),
>+            [len] "=&r"(len), [fft_tmp] "=&r"(fft_tmp)
>+          : [scale] "f"(scale), [fft] "r"(fft)
>+          : "memory");
>+    }
>+    ooura_fft.Fft(fft);
>+    aRe = h_fft_buf[0] + pos;
>+    aIm = h_fft_buf[1] + pos;
>+    __asm __volatile(
>         ".set     push                                    \n\t"
>         ".set     noreorder                               \n\t"
>         "addiu    %[fft_tmp], %[fft],        0            \n\t"
>-        "addiu    %[len],     $zero,         8            \n\t"
>+        "addiu    %[len],     $zero,         31           \n\t"
>+        "lwc1     %[f0],      0(%[aRe])                   \n\t"
>+        "lwc1     %[f1],      0(%[fft_tmp])               \n\t"
>+        "lwc1     %[f2],      256(%[aRe])                 \n\t"
>+        "lwc1     %[f3],      4(%[fft_tmp])               \n\t"
>+        "lwc1     %[f4],      4(%[aRe])                   \n\t"
>+        "lwc1     %[f5],      8(%[fft_tmp])               \n\t"
>+        "lwc1     %[f6],      4(%[aIm])                   \n\t"
>+        "lwc1     %[f7],      12(%[fft_tmp])              \n\t"
>+        "add.s    %[f0],      %[f0],         %[f1]        \n\t"
>+        "add.s    %[f2],      %[f2],         %[f3]        \n\t"
>+        "add.s    %[f4],      %[f4],         %[f5]        \n\t"
>+        "add.s    %[f6],      %[f6],         %[f7]        \n\t"
>+        "addiu    %[fft_tmp], %[fft_tmp],    16           \n\t"
>+        "swc1     %[f0],      0(%[aRe])                   \n\t"
>+        "swc1     %[f2],      256(%[aRe])                 \n\t"
>+        "swc1     %[f4],      4(%[aRe])                   \n\t"
>+        "addiu    %[aRe],     %[aRe],        8            \n\t"
>+        "swc1     %[f6],      4(%[aIm])                   \n\t"
>+        "addiu    %[aIm],     %[aIm],        8            \n\t"
>         "1:                                               \n\t"
>+        "lwc1     %[f0],      0(%[aRe])                   \n\t"
>+        "lwc1     %[f1],      0(%[fft_tmp])               \n\t"
>+        "lwc1     %[f2],      0(%[aIm])                   \n\t"
>+        "lwc1     %[f3],      4(%[fft_tmp])               \n\t"
>+        "lwc1     %[f4],      4(%[aRe])                   \n\t"
>+        "lwc1     %[f5],      8(%[fft_tmp])               \n\t"
>+        "lwc1     %[f6],      4(%[aIm])                   \n\t"
>+        "lwc1     %[f7],      12(%[fft_tmp])              \n\t"
>+        "add.s    %[f0],      %[f0],         %[f1]        \n\t"
>+        "add.s    %[f2],      %[f2],         %[f3]        \n\t"
>+        "add.s    %[f4],      %[f4],         %[f5]        \n\t"
>+        "add.s    %[f6],      %[f6],         %[f7]        \n\t"
>         "addiu    %[len],     %[len],        -1           \n\t"
>-        "lwc1     %[f0],      0(%[fft_tmp])               \n\t"
>-        "lwc1     %[f1],      4(%[fft_tmp])               \n\t"
>-        "lwc1     %[f2],      8(%[fft_tmp])               \n\t"
>-        "lwc1     %[f3],      12(%[fft_tmp])              \n\t"
>-        "mul.s    %[f0],      %[f0],         %[scale]     \n\t"
>-        "mul.s    %[f1],      %[f1],         %[scale]     \n\t"
>-        "mul.s    %[f2],      %[f2],         %[scale]     \n\t"
>-        "mul.s    %[f3],      %[f3],         %[scale]     \n\t"
>-        "lwc1     %[f4],      16(%[fft_tmp])              \n\t"
>-        "lwc1     %[f5],      20(%[fft_tmp])              \n\t"
>-        "lwc1     %[f6],      24(%[fft_tmp])              \n\t"
>-        "lwc1     %[f7],      28(%[fft_tmp])              \n\t"
>-        "mul.s    %[f4],      %[f4],         %[scale]     \n\t"
>-        "mul.s    %[f5],      %[f5],         %[scale]     \n\t"
>-        "mul.s    %[f6],      %[f6],         %[scale]     \n\t"
>-        "mul.s    %[f7],      %[f7],         %[scale]     \n\t"
>-        "swc1     %[f0],      0(%[fft_tmp])               \n\t"
>-        "swc1     %[f1],      4(%[fft_tmp])               \n\t"
>-        "swc1     %[f2],      8(%[fft_tmp])               \n\t"
>-        "swc1     %[f3],      12(%[fft_tmp])              \n\t"
>-        "swc1     %[f4],      16(%[fft_tmp])              \n\t"
>-        "swc1     %[f5],      20(%[fft_tmp])              \n\t"
>-        "swc1     %[f6],      24(%[fft_tmp])              \n\t"
>-        "swc1     %[f7],      28(%[fft_tmp])              \n\t"
>+        "addiu    %[fft_tmp], %[fft_tmp],    16           \n\t"
>+        "swc1     %[f0],      0(%[aRe])                   \n\t"
>+        "swc1     %[f2],      0(%[aIm])                   \n\t"
>+        "swc1     %[f4],      4(%[aRe])                   \n\t"
>+        "addiu    %[aRe],     %[aRe],        8            \n\t"
>+        "swc1     %[f6],      4(%[aIm])                   \n\t"
>         "bgtz     %[len],     1b                          \n\t"
>-        " addiu   %[fft_tmp], %[fft_tmp],    32           \n\t"
>+        " addiu   %[aIm],     %[aIm],        8            \n\t"
>         ".set     pop                                     \n\t"
>-        : [f0] "=&f" (f0), [f1] "=&f" (f1), [f2] "=&f" (f2),
>-          [f3] "=&f" (f3), [f4] "=&f" (f4), [f5] "=&f" (f5),
>-          [f6] "=&f" (f6), [f7] "=&f" (f7), [len] "=&r" (len),
>-          [fft_tmp] "=&r" (fft_tmp)
>-        : [scale] "f" (scale), [fft] "r" (fft)
>+        : [f0] "=&f"(f0), [f1] "=&f"(f1), [f2] "=&f"(f2), [f3] "=&f"(f3),
>+          [f4] "=&f"(f4), [f5] "=&f"(f5), [f6] "=&f"(f6), [f7] "=&f"(f7),
>+          [len] "=&r"(len), [fft_tmp] "=&r"(fft_tmp), [aRe] "+r"(aRe),
>+          [aIm] "+r"(aIm)
>+        : [fft] "r"(fft)
>         : "memory");
>-    }
>-    ooura_fft.Fft(fft);
>-    aRe = h_fft_buf[0] + pos;
>-    aIm = h_fft_buf[1] + pos;
>-    __asm __volatile(
>-      ".set     push                                    \n\t"
>-      ".set     noreorder                               \n\t"
>-      "addiu    %[fft_tmp], %[fft],        0            \n\t"
>-      "addiu    %[len],     $zero,         31           \n\t"
>-      "lwc1     %[f0],      0(%[aRe])                   \n\t"
>-      "lwc1     %[f1],      0(%[fft_tmp])               \n\t"
>-      "lwc1     %[f2],      256(%[aRe])                 \n\t"
>-      "lwc1     %[f3],      4(%[fft_tmp])               \n\t"
>-      "lwc1     %[f4],      4(%[aRe])                   \n\t"
>-      "lwc1     %[f5],      8(%[fft_tmp])               \n\t"
>-      "lwc1     %[f6],      4(%[aIm])                   \n\t"
>-      "lwc1     %[f7],      12(%[fft_tmp])              \n\t"
>-      "add.s    %[f0],      %[f0],         %[f1]        \n\t"
>-      "add.s    %[f2],      %[f2],         %[f3]        \n\t"
>-      "add.s    %[f4],      %[f4],         %[f5]        \n\t"
>-      "add.s    %[f6],      %[f6],         %[f7]        \n\t"
>-      "addiu    %[fft_tmp], %[fft_tmp],    16           \n\t"
>-      "swc1     %[f0],      0(%[aRe])                   \n\t"
>-      "swc1     %[f2],      256(%[aRe])                 \n\t"
>-      "swc1     %[f4],      4(%[aRe])                   \n\t"
>-      "addiu    %[aRe],     %[aRe],        8            \n\t"
>-      "swc1     %[f6],      4(%[aIm])                   \n\t"
>-      "addiu    %[aIm],     %[aIm],        8            \n\t"
>-      "1:                                               \n\t"
>-      "lwc1     %[f0],      0(%[aRe])                   \n\t"
>-      "lwc1     %[f1],      0(%[fft_tmp])               \n\t"
>-      "lwc1     %[f2],      0(%[aIm])                   \n\t"
>-      "lwc1     %[f3],      4(%[fft_tmp])               \n\t"
>-      "lwc1     %[f4],      4(%[aRe])                   \n\t"
>-      "lwc1     %[f5],      8(%[fft_tmp])               \n\t"
>-      "lwc1     %[f6],      4(%[aIm])                   \n\t"
>-      "lwc1     %[f7],      12(%[fft_tmp])              \n\t"
>-      "add.s    %[f0],      %[f0],         %[f1]        \n\t"
>-      "add.s    %[f2],      %[f2],         %[f3]        \n\t"
>-      "add.s    %[f4],      %[f4],         %[f5]        \n\t"
>-      "add.s    %[f6],      %[f6],         %[f7]        \n\t"
>-      "addiu    %[len],     %[len],        -1           \n\t"
>-      "addiu    %[fft_tmp], %[fft_tmp],    16           \n\t"
>-      "swc1     %[f0],      0(%[aRe])                   \n\t"
>-      "swc1     %[f2],      0(%[aIm])                   \n\t"
>-      "swc1     %[f4],      4(%[aRe])                   \n\t"
>-      "addiu    %[aRe],     %[aRe],        8            \n\t"
>-      "swc1     %[f6],      4(%[aIm])                   \n\t"
>-      "bgtz     %[len],     1b                          \n\t"
>-      " addiu   %[aIm],     %[aIm],        8            \n\t"
>-      ".set     pop                                     \n\t"
>-      : [f0] "=&f" (f0), [f1] "=&f" (f1), [f2] "=&f" (f2),
>-        [f3] "=&f" (f3), [f4] "=&f" (f4), [f5] "=&f" (f5),
>-        [f6] "=&f" (f6), [f7] "=&f" (f7), [len] "=&r" (len),
>-        [fft_tmp] "=&r" (fft_tmp), [aRe] "+r" (aRe), [aIm] "+r" (aIm)
>-      : [fft] "r" (fft)
>-      : "memory");
>   }
> }
> 
>@@ -359,28 +355,28 @@ void WebRtcAec_Overdrive_mips(float overdrive_scaling,
>   for (int i = 0; i < PART_LEN1; ++i) {
>     // Weight subbands
>     __asm __volatile(
>-      ".set      push                                              \n\t"
>-      ".set      noreorder                                         \n\t"
>-      "lwc1      %[temp1],    0(%[p_hNl])                          \n\t"
>-      "lwc1      %[temp2],    0(%[p_wC])                           \n\t"
>-      "c.lt.s    %[hNlFb],    %[temp1]                             \n\t"
>-      "bc1f      1f                                                \n\t"
>-      " mul.s    %[temp3],    %[temp2],     %[hNlFb]               \n\t"
>-      "sub.s     %[temp4],    %[one],       %[temp2]               \n\t"
>+        ".set      push                                              \n\t"
>+        ".set      noreorder                                         \n\t"
>+        "lwc1      %[temp1],    0(%[p_hNl])                          \n\t"
>+        "lwc1      %[temp2],    0(%[p_wC])                           \n\t"
>+        "c.lt.s    %[hNlFb],    %[temp1]                             \n\t"
>+        "bc1f      1f                                                \n\t"
>+        " mul.s    %[temp3],    %[temp2],     %[hNlFb]               \n\t"
>+        "sub.s     %[temp4],    %[one],       %[temp2]               \n\t"
> #if !defined(MIPS32_R2_LE)
>-      "mul.s     %[temp1],    %[temp1],     %[temp4]               \n\t"
>-      "add.s     %[temp1],    %[temp3],     %[temp1]               \n\t"
>-#else  // #if !defined(MIPS32_R2_LE)
>-      "madd.s    %[temp1],    %[temp3],     %[temp1],   %[temp4]   \n\t"
>+        "mul.s     %[temp1],    %[temp1],     %[temp4]               \n\t"
>+        "add.s     %[temp1],    %[temp3],     %[temp1]               \n\t"
>+#else   // #if !defined(MIPS32_R2_LE)
>+        "madd.s    %[temp1],    %[temp3],     %[temp1],   %[temp4]   \n\t"
> #endif  // #if !defined(MIPS32_R2_LE)
>-      "swc1      %[temp1],    0(%[p_hNl])                          \n\t"
>-     "1:                                                           \n\t"
>-      "addiu     %[p_wC],     %[p_wC],      4                      \n\t"
>-      ".set      pop                                               \n\t"
>-      : [temp1] "=&f" (temp1), [temp2] "=&f" (temp2), [temp3] "=&f" (temp3),
>-        [temp4] "=&f" (temp4), [p_wC] "+r" (p_WebRtcAec_wC)
>-      : [hNlFb] "f" (hNlFb), [one] "f" (one), [p_hNl] "r" (p_hNl)
>-      : "memory");
>+        "swc1      %[temp1],    0(%[p_hNl])                          \n\t"
>+        "1:                                                           \n\t"
>+        "addiu     %[p_wC],     %[p_wC],      4                      \n\t"
>+        ".set      pop                                               \n\t"
>+        : [temp1] "=&f"(temp1), [temp2] "=&f"(temp2), [temp3] "=&f"(temp3),
>+          [temp4] "=&f"(temp4), [p_wC] "+r"(p_WebRtcAec_wC)
>+        : [hNlFb] "f"(hNlFb), [one] "f"(one), [p_hNl] "r"(p_hNl)
>+        : "memory");
> 
>     hNl[i] = powf(hNl[i], overdrive_scaling * WebRtcAec_overDriveCurve[i]);
>   }
>@@ -399,22 +395,22 @@ void WebRtcAec_Suppress_mips(const float hNl[PART_LEN1],
> 
>   for (int i = 0; i < PART_LEN1; ++i) {
>     __asm __volatile(
>-      "lwc1      %[temp1],    0(%[p_hNl])              \n\t"
>-      "lwc1      %[temp3],    0(%[p_efw1])             \n\t"
>-      "lwc1      %[temp2],    0(%[p_efw0])             \n\t"
>-      "addiu     %[p_hNl],    %[p_hNl],     4          \n\t"
>-      "mul.s     %[temp3],    %[temp3],     %[temp1]   \n\t"
>-      "mul.s     %[temp2],    %[temp2],     %[temp1]   \n\t"
>-      "addiu     %[p_efw0],   %[p_efw0],    4          \n\t"
>-      "addiu     %[p_efw1],   %[p_efw1],    4          \n\t"
>-      "neg.s     %[temp4],    %[temp3]                 \n\t"
>-      "swc1      %[temp2],    -4(%[p_efw0])            \n\t"
>-      "swc1      %[temp4],    -4(%[p_efw1])            \n\t"
>-      : [temp1] "=&f" (temp1), [temp2] "=&f" (temp2), [temp3] "=&f" (temp3),
>-        [temp4] "=&f" (temp4), [p_efw0] "+r" (p_efw0), [p_efw1] "+r" (p_efw1),
>-        [p_hNl] "+r" (p_hNl)
>-      :
>-      : "memory");
>+        "lwc1      %[temp1],    0(%[p_hNl])              \n\t"
>+        "lwc1      %[temp3],    0(%[p_efw1])             \n\t"
>+        "lwc1      %[temp2],    0(%[p_efw0])             \n\t"
>+        "addiu     %[p_hNl],    %[p_hNl],     4          \n\t"
>+        "mul.s     %[temp3],    %[temp3],     %[temp1]   \n\t"
>+        "mul.s     %[temp2],    %[temp2],     %[temp1]   \n\t"
>+        "addiu     %[p_efw0],   %[p_efw0],    4          \n\t"
>+        "addiu     %[p_efw1],   %[p_efw1],    4          \n\t"
>+        "neg.s     %[temp4],    %[temp3]                 \n\t"
>+        "swc1      %[temp2],    -4(%[p_efw0])            \n\t"
>+        "swc1      %[temp4],    -4(%[p_efw1])            \n\t"
>+        : [temp1] "=&f"(temp1), [temp2] "=&f"(temp2), [temp3] "=&f"(temp3),
>+          [temp4] "=&f"(temp4), [p_efw0] "+r"(p_efw0), [p_efw1] "+r"(p_efw1),
>+          [p_hNl] "+r"(p_hNl)
>+        :
>+        : "memory");
>   }
> }
> 
>@@ -433,51 +429,50 @@ void WebRtcAec_ScaleErrorSignal_mips(float mu,
> #endif
> 
>   __asm __volatile(
>-    ".set       push                                   \n\t"
>-    ".set       noreorder                              \n\t"
>-    "1:                                                \n\t"
>-    "lwc1       %[f0],     0(%[x_pow])                 \n\t"
>-    "lwc1       %[f1],     0(%[ef0])                   \n\t"
>-    "lwc1       %[f2],     0(%[ef1])                   \n\t"
>-    "add.s      %[f0],     %[f0],       %[fac1]        \n\t"
>-    "div.s      %[f1],     %[f1],       %[f0]          \n\t"
>-    "div.s      %[f2],     %[f2],       %[f0]          \n\t"
>-    "mul.s      %[f0],     %[f1],       %[f1]          \n\t"
>+      ".set       push                                   \n\t"
>+      ".set       noreorder                              \n\t"
>+      "1:                                                \n\t"
>+      "lwc1       %[f0],     0(%[x_pow])                 \n\t"
>+      "lwc1       %[f1],     0(%[ef0])                   \n\t"
>+      "lwc1       %[f2],     0(%[ef1])                   \n\t"
>+      "add.s      %[f0],     %[f0],       %[fac1]        \n\t"
>+      "div.s      %[f1],     %[f1],       %[f0]          \n\t"
>+      "div.s      %[f2],     %[f2],       %[f0]          \n\t"
>+      "mul.s      %[f0],     %[f1],       %[f1]          \n\t"
> #if defined(MIPS32_R2_LE)
>-    "madd.s     %[f0],     %[f0],       %[f2],   %[f2] \n\t"
>+      "madd.s     %[f0],     %[f0],       %[f2],   %[f2] \n\t"
> #else
>-    "mul.s      %[f3],     %[f2],       %[f2]          \n\t"
>-    "add.s      %[f0],     %[f0],       %[f3]          \n\t"
>+      "mul.s      %[f3],     %[f2],       %[f2]          \n\t"
>+      "add.s      %[f0],     %[f0],       %[f3]          \n\t"
> #endif
>-    "c.le.s     %[f0],     %[err_th2]                  \n\t"
>-    "nop                                               \n\t"
>-    "bc1t       2f                                     \n\t"
>-    " nop                                              \n\t"
>-    "sqrt.s     %[f0],     %[f0]                       \n\t"
>-    "add.s      %[f0],     %[f0],       %[fac1]        \n\t"
>-    "div.s      %[f0],     %[err_th],   %[f0]          \n\t"
>-    "mul.s      %[f1],     %[f1],       %[f0]          \n\t"
>-    "mul.s      %[f2],     %[f2],       %[f0]          \n\t"
>-    "2:                                                \n\t"
>-    "mul.s      %[f1],     %[f1],       %[mu]          \n\t"
>-    "mul.s      %[f2],     %[f2],       %[mu]          \n\t"
>-    "swc1       %[f1],     0(%[ef0])                   \n\t"
>-    "swc1       %[f2],     0(%[ef1])                   \n\t"
>-    "addiu      %[len],    %[len],      -1             \n\t"
>-    "addiu      %[x_pow],  %[x_pow],    4              \n\t"
>-    "addiu      %[ef0],    %[ef0],      4              \n\t"
>-    "bgtz       %[len],    1b                          \n\t"
>-    " addiu     %[ef1],    %[ef1],      4              \n\t"
>-    ".set       pop                                    \n\t"
>-    : [f0] "=&f" (f0), [f1] "=&f" (f1), [f2] "=&f" (f2),
>+      "c.le.s     %[f0],     %[err_th2]                  \n\t"
>+      "nop                                               \n\t"
>+      "bc1t       2f                                     \n\t"
>+      " nop                                              \n\t"
>+      "sqrt.s     %[f0],     %[f0]                       \n\t"
>+      "add.s      %[f0],     %[f0],       %[fac1]        \n\t"
>+      "div.s      %[f0],     %[err_th],   %[f0]          \n\t"
>+      "mul.s      %[f1],     %[f1],       %[f0]          \n\t"
>+      "mul.s      %[f2],     %[f2],       %[f0]          \n\t"
>+      "2:                                                \n\t"
>+      "mul.s      %[f1],     %[f1],       %[mu]          \n\t"
>+      "mul.s      %[f2],     %[f2],       %[mu]          \n\t"
>+      "swc1       %[f1],     0(%[ef0])                   \n\t"
>+      "swc1       %[f2],     0(%[ef1])                   \n\t"
>+      "addiu      %[len],    %[len],      -1             \n\t"
>+      "addiu      %[x_pow],  %[x_pow],    4              \n\t"
>+      "addiu      %[ef0],    %[ef0],      4              \n\t"
>+      "bgtz       %[len],    1b                          \n\t"
>+      " addiu     %[ef1],    %[ef1],      4              \n\t"
>+      ".set       pop                                    \n\t"
>+      : [f0] "=&f"(f0), [f1] "=&f"(f1), [f2] "=&f"(f2),
> #if !defined(MIPS32_R2_LE)
>-      [f3] "=&f" (f3),
>+        [f3] "=&f"(f3),
> #endif
>-      [x_pow] "+r" (x_pow), [ef0] "+r" (ef0), [ef1] "+r" (ef1),
>-      [len] "+r" (len)
>-    : [fac1] "f" (fac1), [err_th2] "f" (err_th2), [mu] "f" (mu),
>-      [err_th] "f" (error_threshold)
>-    : "memory");
>+        [x_pow] "+r"(x_pow), [ef0] "+r"(ef0), [ef1] "+r"(ef1), [len] "+r"(len)
>+      : [fac1] "f"(fac1), [err_th2] "f"(err_th2), [mu] "f"(mu),
>+        [err_th] "f"(error_threshold)
>+      : "memory");
> }
> 
> void WebRtcAec_InitAec_mips(void) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/aec_core_neon.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/aec_core_neon.cc
>index 1fbf56b8b67..072bd17dfe0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/aec_core_neon.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/aec_core_neon.cc
>@@ -38,13 +38,12 @@ __inline static float MulIm(float aRe, float aIm, float bRe, float bIm) {
>   return aRe * bIm + aIm * bRe;
> }
> 
>-static void FilterFarNEON(int num_partitions,
>-                          int x_fft_buf_block_pos,
>-                          float x_fft_buf[2]
>-                                         [kExtendedNumPartitions * PART_LEN1],
>-                          float h_fft_buf[2]
>-                                         [kExtendedNumPartitions * PART_LEN1],
>-                          float y_fft[2][PART_LEN1]) {
>+static void FilterFarNEON(
>+    int num_partitions,
>+    int x_fft_buf_block_pos,
>+    float x_fft_buf[2][kExtendedNumPartitions * PART_LEN1],
>+    float h_fft_buf[2][kExtendedNumPartitions * PART_LEN1],
>+    float y_fft[2][PART_LEN1]) {
>   int i;
>   for (i = 0; i < num_partitions; i++) {
>     int j;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/aec_core_optimized_methods.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/aec_core_optimized_methods.h
>index a8a20e94d89..03c027dabfb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/aec_core_optimized_methods.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/aec_core_optimized_methods.h
>@@ -14,7 +14,6 @@
> #include <memory>
> 
> #include "modules/audio_processing/aec/aec_core.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/aec_core_sse2.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/aec_core_sse2.cc
>index 0532662bed1..ede04ddfc3f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/aec_core_sse2.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/aec_core_sse2.cc
>@@ -33,13 +33,12 @@ __inline static float MulIm(float aRe, float aIm, float bRe, float bIm) {
>   return aRe * bIm + aIm * bRe;
> }
> 
>-static void FilterFarSSE2(int num_partitions,
>-                          int x_fft_buf_block_pos,
>-                          float x_fft_buf[2]
>-                                         [kExtendedNumPartitions * PART_LEN1],
>-                          float h_fft_buf[2]
>-                                         [kExtendedNumPartitions * PART_LEN1],
>-                          float y_fft[2][PART_LEN1]) {
>+static void FilterFarSSE2(
>+    int num_partitions,
>+    int x_fft_buf_block_pos,
>+    float x_fft_buf[2][kExtendedNumPartitions * PART_LEN1],
>+    float h_fft_buf[2][kExtendedNumPartitions * PART_LEN1],
>+    float y_fft[2][PART_LEN1]) {
>   int i;
>   for (i = 0; i < num_partitions; i++) {
>     int j;
>@@ -257,10 +256,10 @@ static __m128 mm_pow_ps(__m128 a, __m128 b) {
>         _mm_and_ps(a, *(reinterpret_cast<const __m128*>(float_exponent_mask)));
>     const __m128 n_1 = _mm_castsi128_ps(_mm_srli_epi32(
>         _mm_castps_si128(two_n), shift_exponent_into_top_mantissa));
>-    const __m128 n_0 =
>-      _mm_or_ps(n_1, *(reinterpret_cast<const __m128*>(eight_biased_exponent)));
>-    const __m128 n =
>-      _mm_sub_ps(n_0, *(reinterpret_cast<const __m128*>(implicit_leading_one)));
>+    const __m128 n_0 = _mm_or_ps(
>+        n_1, *(reinterpret_cast<const __m128*>(eight_biased_exponent)));
>+    const __m128 n = _mm_sub_ps(
>+        n_0, *(reinterpret_cast<const __m128*>(implicit_leading_one)));
> 
>     // Compute y.
>     static const ALIGN16_BEG int mantissa_mask[4] ALIGN16_END = {
>@@ -269,9 +268,9 @@ static __m128 mm_pow_ps(__m128 a, __m128 b) {
>         0x3F800000, 0x3F800000, 0x3F800000, 0x3F800000};
>     const __m128 mantissa =
>         _mm_and_ps(a, *(reinterpret_cast<const __m128*>(mantissa_mask)));
>-    const __m128 y =
>-        _mm_or_ps(mantissa,
>-               *(reinterpret_cast<const __m128*>(zero_biased_exponent_is_one)));
>+    const __m128 y = _mm_or_ps(
>+        mantissa,
>+        *(reinterpret_cast<const __m128*>(zero_biased_exponent_is_one)));
> 
>     // Approximate log2(y) ~= (y - 1) * pol5(y).
>     //    pol5(y) = C5 * y^5 + C4 * y^4 + C3 * y^3 + C2 * y^2 + C1 * y + C0
>@@ -303,9 +302,8 @@ static __m128 mm_pow_ps(__m128 a, __m128 b) {
>     const __m128 pol5_y_8 = _mm_mul_ps(pol5_y_7, y);
>     const __m128 pol5_y =
>         _mm_add_ps(pol5_y_8, *(reinterpret_cast<const __m128*>(C0)));
>-    const __m128 y_minus_one =
>-        _mm_sub_ps(y,
>-               *(reinterpret_cast<const __m128*>(zero_biased_exponent_is_one)));
>+    const __m128 y_minus_one = _mm_sub_ps(
>+        y, *(reinterpret_cast<const __m128*>(zero_biased_exponent_is_one)));
>     const __m128 log2_y = _mm_mul_ps(y_minus_one, pol5_y);
> 
>     // Combine parts.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/echo_cancellation.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/echo_cancellation.cc
>index 864db5308e6..fd1aec4058e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/echo_cancellation.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/echo_cancellation.cc
>@@ -24,7 +24,6 @@ extern "C" {
> #include "modules/audio_processing/aec/aec_core.h"
> #include "modules/audio_processing/aec/aec_resampler.h"
> #include "modules/audio_processing/logging/apm_data_dumper.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -291,7 +290,7 @@ int32_t WebRtcAec_BufferFarend(void* aecInst,
> 
>   aecpc->farend_started = 1;
>   WebRtcAec_SetSystemDelay(aecpc->aec, WebRtcAec_system_delay(aecpc->aec) +
>-                           static_cast<int>(newNrOfSamples));
>+                                           static_cast<int>(newNrOfSamples));
> 
>   // Write the time-domain data to |far_pre_buf|.
>   WebRtc_WriteBuffer(aecpc->far_pre_buf, farend_ptr, newNrOfSamples);
>@@ -302,8 +301,8 @@ int32_t WebRtcAec_BufferFarend(void* aecInst,
>     {
>       float* ptmp = NULL;
>       float tmp[PART_LEN2];
>-      WebRtc_ReadBuffer(aecpc->far_pre_buf,
>-                        reinterpret_cast<void**>(&ptmp), tmp, PART_LEN2);
>+      WebRtc_ReadBuffer(aecpc->far_pre_buf, reinterpret_cast<void**>(&ptmp),
>+                        tmp, PART_LEN2);
>       WebRtcAec_BufferFarendBlock(aecpc->aec, &ptmp[PART_LEN]);
>     }
> 
>@@ -545,9 +544,9 @@ static int ProcessNormal(Aec* aecInst,
>   const float minSkewEst = -0.5f;
>   const float maxSkewEst = 1.0f;
> 
>-  reported_delay_ms =
>-      reported_delay_ms > kMaxTrustedDelayMs ? kMaxTrustedDelayMs :
>-      reported_delay_ms;
>+  reported_delay_ms = reported_delay_ms > kMaxTrustedDelayMs
>+                          ? kMaxTrustedDelayMs
>+                          : reported_delay_ms;
>   // TODO(andrew): we need to investigate if this +10 is really wanted.
>   reported_delay_ms += 10;
>   aecInst->msInSndCardBuf = reported_delay_ms;
>@@ -642,9 +641,8 @@ static int ProcessNormal(Aec* aecInst,
>       // for too long). When the far-end buffer is filled with
>       // approximately the same amount of data as reported by the system
>       // we end the startup phase.
>-      int overhead_elements =
>-          WebRtcAec_system_delay(aecInst->aec) / PART_LEN -
>-          aecInst->bufSizeStart;
>+      int overhead_elements = WebRtcAec_system_delay(aecInst->aec) / PART_LEN -
>+                              aecInst->bufSizeStart;
>       if (overhead_elements == 0) {
>         // Enable the AEC
>         aecInst->startup_phase = 0;
>@@ -677,7 +675,7 @@ static int ProcessNormal(Aec* aecInst,
> }
> 
> static void ProcessExtended(Aec* self,
>-                            const float* const* near,
>+                            const float* const* nearend,
>                             size_t num_bands,
>                             float* const* out,
>                             size_t num_samples,
>@@ -709,8 +707,8 @@ static void ProcessExtended(Aec* self,
>   if (!self->farend_started) {
>     for (i = 0; i < num_bands; ++i) {
>       // Only needed if they don't already point to the same place.
>-      if (near[i] != out[i]) {
>-        memcpy(out[i], near[i], sizeof(near[i][0]) * num_samples);
>+      if (nearend[i] != out[i]) {
>+        memcpy(out[i], nearend[i], sizeof(nearend[i][0]) * num_samples);
>       }
>     }
>     return;
>@@ -746,7 +744,7 @@ static void ProcessExtended(Aec* self,
>     const int adjusted_known_delay =
>         WEBRTC_SPL_MAX(0, self->knownDelay + delay_diff_offset);
> 
>-    WebRtcAec_ProcessFrames(self->aec, near, num_bands, num_samples,
>+    WebRtcAec_ProcessFrames(self->aec, nearend, num_bands, num_samples,
>                             adjusted_known_delay, out);
>   }
> }
>@@ -780,10 +778,8 @@ static void EstBufDelayNormal(Aec* aecInst) {
>   // We use -1 to signal an initialized state in the "extended" implementation;
>   // compensate for that.
>   aecInst->filtDelay = aecInst->filtDelay < 0 ? 0 : aecInst->filtDelay;
>-  aecInst->filtDelay =
>-      WEBRTC_SPL_MAX(0, static_cast<int16_t>(0.8 *
>-                                             aecInst->filtDelay +
>-                                             0.2 * current_delay));
>+  aecInst->filtDelay = WEBRTC_SPL_MAX(
>+      0, static_cast<int16_t>(0.8 * aecInst->filtDelay + 0.2 * current_delay));
> 
>   delay_difference = aecInst->filtDelay - aecInst->knownDelay;
>   if (delay_difference > 224) {
>@@ -809,8 +805,8 @@ static void EstBufDelayNormal(Aec* aecInst) {
> }
> 
> static void EstBufDelayExtended(Aec* aecInst) {
>-  int reported_delay = aecInst->msInSndCardBuf * sampMsNb *
>-      aecInst->rate_factor;
>+  int reported_delay =
>+      aecInst->msInSndCardBuf * sampMsNb * aecInst->rate_factor;
>   int current_delay = reported_delay - WebRtcAec_system_delay(aecInst->aec);
>   int delay_difference = 0;
> 
>@@ -839,8 +835,8 @@ static void EstBufDelayExtended(Aec* aecInst) {
>     aecInst->filtDelay = WEBRTC_SPL_MAX(0, 0.5 * current_delay);
>   } else {
>     aecInst->filtDelay = WEBRTC_SPL_MAX(
>-        0, static_cast<int16_t>(0.95 * aecInst->filtDelay + 0.05 *
>-                                current_delay));
>+        0,
>+        static_cast<int16_t>(0.95 * aecInst->filtDelay + 0.05 * current_delay));
>   }
> 
>   delay_difference = aecInst->filtDelay - aecInst->knownDelay;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/echo_cancellation.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/echo_cancellation.h
>index d5c703e46b4..20393471f2a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/echo_cancellation.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/echo_cancellation.h
>@@ -19,7 +19,6 @@ extern "C" {
> #include "common_audio/ring_buffer.h"
> }
> #include "modules/audio_processing/aec/aec_core.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/system_delay_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/system_delay_unittest.cc
>index fc57af8b87a..9c57e8b84ef 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/system_delay_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec/system_delay_unittest.cc
>@@ -12,15 +12,14 @@
> #include "modules/audio_processing/aec/echo_cancellation.h"
> #include "rtc_base/numerics/safe_conversions.h"
> #include "test/gtest.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> namespace webrtc {
> namespace {
> 
> class SystemDelayTest : public ::testing::Test {
>  protected:
>   SystemDelayTest();
>-  virtual void SetUp();
>-  virtual void TearDown();
>+  void SetUp() override;
>+  void TearDown() override;
> 
>   // Initialization of AEC handle with respect to |sample_rate_hz|. Since the
>   // device sample rate is unimportant we set that value to 48000 Hz.
>@@ -104,14 +103,8 @@ void SystemDelayTest::Init(int sample_rate_hz) {
> 
> void SystemDelayTest::RenderAndCapture(int device_buffer_ms) {
>   EXPECT_EQ(0, WebRtcAec_BufferFarend(handle_, far_, samples_per_frame_));
>-  EXPECT_EQ(0,
>-            WebRtcAec_Process(handle_,
>-                              &near_ptr_,
>-                              1,
>-                              &out_ptr_,
>-                              samples_per_frame_,
>-                              device_buffer_ms,
>-                              0));
>+  EXPECT_EQ(0, WebRtcAec_Process(handle_, &near_ptr_, 1, &out_ptr_,
>+                                 samples_per_frame_, device_buffer_ms, 0));
> }
> 
> size_t SystemDelayTest::BufferFillUp() {
>@@ -157,16 +150,15 @@ void SystemDelayTest::RunStableStartup() {
>     EXPECT_GT(kStableConvergenceMs, process_time_ms);
>   }
>   // Verify that the buffer has been flushed.
>-  EXPECT_GE(static_cast<int>(buffer_size),
>-            WebRtcAec_system_delay(self_->aec));
>+  EXPECT_GE(static_cast<int>(buffer_size), WebRtcAec_system_delay(self_->aec));
> }
> 
>-  int SystemDelayTest::MapBufferSizeToSamples(int size_in_ms,
>-                                              bool extended_filter) {
>+int SystemDelayTest::MapBufferSizeToSamples(int size_in_ms,
>+                                            bool extended_filter) {
>   // If extended_filter is disabled we add an extra 10 ms for the unprocessed
>   // frame. That is simply how the algorithm is constructed.
>-  return static_cast<int>(
>-      (size_in_ms + (extended_filter ? 0 : 10)) * samples_per_frame_ / 10);
>+  return static_cast<int>((size_in_ms + (extended_filter ? 0 : 10)) *
>+                          samples_per_frame_ / 10);
> }
> 
> // The tests should meet basic requirements and not be adjusted to what is
>@@ -241,9 +233,9 @@ TEST_F(SystemDelayTest, CorrectDelayAfterStableStartup) {
>             static_cast<int>(kDeviceBufMs * samples_per_frame_ / 10);
>         EXPECT_GE(average_reported_delay, WebRtcAec_system_delay(self_->aec));
>         int lower_bound = WebRtcAec_extended_filter_enabled(self_->aec)
>-            ? (average_reported_delay / 2 -
>-               rtc::checked_cast<int>(samples_per_frame_))
>-            : average_reported_delay * 3 / 4;
>+                              ? (average_reported_delay / 2 -
>+                                 rtc::checked_cast<int>(samples_per_frame_))
>+                              : average_reported_delay * 3 / 4;
>         EXPECT_LE(lower_bound, WebRtcAec_system_delay(self_->aec));
>       }
>     }
>@@ -322,14 +314,8 @@ TEST_F(SystemDelayTest, CorrectDelayAfterStableBufferBuildUp) {
>     // can make that assumption since we have a separate stability test.
>     int process_time_ms = 0;
>     for (; process_time_ms < kStableConvergenceMs; process_time_ms += 10) {
>-      EXPECT_EQ(0,
>-                WebRtcAec_Process(handle_,
>-                                  &near_ptr_,
>-                                  1,
>-                                  &out_ptr_,
>-                                  samples_per_frame_,
>-                                  kDeviceBufMs,
>-                                  0));
>+      EXPECT_EQ(0, WebRtcAec_Process(handle_, &near_ptr_, 1, &out_ptr_,
>+                                     samples_per_frame_, kDeviceBufMs, 0));
>     }
>     // Verify that a buffer size has been established.
>     EXPECT_EQ(0, self_->checkBuffSize);
>@@ -414,8 +400,8 @@ TEST_F(SystemDelayTest, CorrectDelayDuringDrift) {
>         for (int j = 0; j < 1000; j++) {
>           // Drift = -1 ms per 100 ms of data.
>           int device_buf_ms = kDeviceBufMs - (j / 10) + jump;
>-          int device_buf = MapBufferSizeToSamples(device_buf_ms,
>-                                                  extended_filter == 1);
>+          int device_buf =
>+              MapBufferSizeToSamples(device_buf_ms, extended_filter == 1);
> 
>           if (device_buf_ms < 30) {
>             // Add 10 ms data, taking affect next frame.
>@@ -452,8 +438,8 @@ TEST_F(SystemDelayTest, ShouldRecoverAfterGlitch) {
>       for (size_t i = 0; i < kNumSampleRates; i++) {
>         Init(kSampleRateHz[i]);
>         RunStableStartup();
>-        int device_buf = MapBufferSizeToSamples(kDeviceBufMs,
>-                                                extended_filter == 1);
>+        int device_buf =
>+            MapBufferSizeToSamples(kDeviceBufMs, extended_filter == 1);
>         // Glitch state.
>         for (int j = 0; j < 20; j++) {
>           EXPECT_EQ(0,
>@@ -513,8 +499,8 @@ TEST_F(SystemDelayTest, UnaffectedWhenSpuriousDeviceBufferValues) {
>     for (size_t i = 0; i < kNumSampleRates; i++) {
>       Init(kSampleRateHz[i]);
>       RunStableStartup();
>-      int device_buf = MapBufferSizeToSamples(kDeviceBufMs,
>-                                              extended_filter == 1);
>+      int device_buf =
>+          MapBufferSizeToSamples(kDeviceBufMs, extended_filter == 1);
> 
>       // Normal state. We are currently not in a non-causal state.
>       bool non_causal = false;
>@@ -566,8 +552,8 @@ TEST_F(SystemDelayTest, CorrectImpactWhenTogglingDeviceBufferValues) {
>       for (size_t i = 0; i < kNumSampleRates; i++) {
>         Init(kSampleRateHz[i]);
>         RunStableStartup();
>-        const int device_buf = MapBufferSizeToSamples(kDeviceBufMs,
>-                                                      extended_filter == 1);
>+        const int device_buf =
>+            MapBufferSizeToSamples(kDeviceBufMs, extended_filter == 1);
> 
>         // Normal state. We are currently not in a non-causal state.
>         bool non_causal = false;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/BUILD.gn
>new file mode 100644
>index 00000000000..d0df5f7b36f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/BUILD.gn
>@@ -0,0 +1,217 @@
>+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+#
>+# Use of this source code is governed by a BSD-style license
>+# that can be found in the LICENSE file in the root of the source
>+# tree. An additional intellectual property rights grant can be found
>+# in the file PATENTS.  All contributing project authors may
>+# be found in the AUTHORS file in the root of the source tree.
>+
>+import("../../../webrtc.gni")
>+
>+rtc_static_library("aec3") {
>+  visibility = [ "*" ]
>+  configs += [ "..:apm_debug_dump" ]
>+  sources = [
>+    "adaptive_fir_filter.cc",
>+    "adaptive_fir_filter.h",
>+    "aec3_common.cc",
>+    "aec3_common.h",
>+    "aec3_fft.cc",
>+    "aec3_fft.h",
>+    "aec_state.cc",
>+    "aec_state.h",
>+    "block_framer.cc",
>+    "block_framer.h",
>+    "block_processor.cc",
>+    "block_processor.h",
>+    "block_processor_metrics.cc",
>+    "block_processor_metrics.h",
>+    "cascaded_biquad_filter.cc",
>+    "cascaded_biquad_filter.h",
>+    "coherence_gain.cc",
>+    "coherence_gain.h",
>+    "comfort_noise_generator.cc",
>+    "comfort_noise_generator.h",
>+    "decimator.cc",
>+    "decimator.h",
>+    "delay_estimate.h",
>+    "downsampled_render_buffer.cc",
>+    "downsampled_render_buffer.h",
>+    "echo_audibility.cc",
>+    "echo_audibility.h",
>+    "echo_canceller3.cc",
>+    "echo_canceller3.h",
>+    "echo_path_delay_estimator.cc",
>+    "echo_path_delay_estimator.h",
>+    "echo_path_variability.cc",
>+    "echo_path_variability.h",
>+    "echo_remover.cc",
>+    "echo_remover.h",
>+    "echo_remover_metrics.cc",
>+    "echo_remover_metrics.h",
>+    "erl_estimator.cc",
>+    "erl_estimator.h",
>+    "erle_estimator.cc",
>+    "erle_estimator.h",
>+    "fft_buffer.cc",
>+    "fft_buffer.h",
>+    "fft_data.h",
>+    "filter_analyzer.cc",
>+    "filter_analyzer.h",
>+    "frame_blocker.cc",
>+    "frame_blocker.h",
>+    "main_filter_update_gain.cc",
>+    "main_filter_update_gain.h",
>+    "matched_filter.cc",
>+    "matched_filter.h",
>+    "matched_filter_lag_aggregator.cc",
>+    "matched_filter_lag_aggregator.h",
>+    "matrix_buffer.cc",
>+    "matrix_buffer.h",
>+    "moving_average.cc",
>+    "moving_average.h",
>+    "render_buffer.cc",
>+    "render_buffer.h",
>+    "render_delay_buffer.cc",
>+    "render_delay_buffer.h",
>+    "render_delay_controller.cc",
>+    "render_delay_controller.h",
>+    "render_delay_controller_metrics.cc",
>+    "render_delay_controller_metrics.h",
>+    "render_signal_analyzer.cc",
>+    "render_signal_analyzer.h",
>+    "residual_echo_estimator.cc",
>+    "residual_echo_estimator.h",
>+    "reverb_decay_estimator.cc",
>+    "reverb_decay_estimator.h",
>+    "reverb_frequency_response.cc",
>+    "reverb_frequency_response.h",
>+    "reverb_model.cc",
>+    "reverb_model.h",
>+    "reverb_model_estimator.cc",
>+    "reverb_model_estimator.h",
>+    "reverb_model_fallback.cc",
>+    "reverb_model_fallback.h",
>+    "shadow_filter_update_gain.cc",
>+    "shadow_filter_update_gain.h",
>+    "skew_estimator.cc",
>+    "skew_estimator.h",
>+    "stationarity_estimator.cc",
>+    "stationarity_estimator.h",
>+    "subtractor.cc",
>+    "subtractor.h",
>+    "subtractor_output.cc",
>+    "subtractor_output.h",
>+    "subtractor_output_analyzer.cc",
>+    "subtractor_output_analyzer.h",
>+    "suppression_filter.cc",
>+    "suppression_filter.h",
>+    "suppression_gain.cc",
>+    "suppression_gain.h",
>+    "suppression_gain_limiter.cc",
>+    "suppression_gain_limiter.h",
>+    "vector_buffer.cc",
>+    "vector_buffer.h",
>+    "vector_math.h",
>+  ]
>+
>+  defines = []
>+  if (rtc_build_with_neon && current_cpu != "arm64") {
>+    suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
>+    cflags = [ "-mfpu=neon" ]
>+  }
>+
>+  deps = [
>+    "..:apm_logging",
>+    "..:audio_processing",
>+    "../../../api:array_view",
>+    "../../../api/audio:aec3_config",
>+    "../../../api/audio:echo_control",
>+    "../../../common_audio:common_audio_c",
>+    "../../../rtc_base:checks",
>+    "../../../rtc_base:rtc_base_approved",
>+    "../../../rtc_base:safe_minmax",
>+    "../../../rtc_base/system:arch",
>+    "../../../system_wrappers:cpu_features_api",
>+    "../../../system_wrappers:field_trial_api",
>+    "../../../system_wrappers:metrics_api",
>+    "../utility:ooura_fft",
>+    "//third_party/abseil-cpp/absl/types:optional",
>+  ]
>+}
>+
>+if (rtc_include_tests) {
>+  rtc_source_set("aec3_unittests") {
>+    testonly = true
>+
>+    configs += [ "..:apm_debug_dump" ]
>+    sources = [
>+      "mock/mock_block_processor.cc",
>+      "mock/mock_block_processor.h",
>+      "mock/mock_echo_remover.cc",
>+      "mock/mock_echo_remover.h",
>+      "mock/mock_render_delay_buffer.cc",
>+      "mock/mock_render_delay_buffer.h",
>+      "mock/mock_render_delay_controller.cc",
>+      "mock/mock_render_delay_controller.h",
>+    ]
>+
>+    deps = [
>+      ":aec3",
>+      "..:apm_logging",
>+      "..:audio_processing",
>+      "..:audio_processing_unittests",
>+      "../../../api:array_view",
>+      "../../../api/audio:aec3_config",
>+      "../../../rtc_base:checks",
>+      "../../../rtc_base:rtc_base_approved",
>+      "../../../rtc_base:safe_minmax",
>+      "../../../rtc_base/system:arch",
>+      "../../../system_wrappers:cpu_features_api",
>+      "../../../test:test_support",
>+      "//third_party/abseil-cpp/absl/types:optional",
>+    ]
>+
>+    defines = []
>+
>+    if (rtc_enable_protobuf) {
>+      sources += [
>+        "adaptive_fir_filter_unittest.cc",
>+        "aec3_fft_unittest.cc",
>+        "aec_state_unittest.cc",
>+        "block_framer_unittest.cc",
>+        "block_processor_metrics_unittest.cc",
>+        "block_processor_unittest.cc",
>+        "cascaded_biquad_filter_unittest.cc",
>+        "comfort_noise_generator_unittest.cc",
>+        "decimator_unittest.cc",
>+        "echo_canceller3_unittest.cc",
>+        "echo_path_delay_estimator_unittest.cc",
>+        "echo_path_variability_unittest.cc",
>+        "echo_remover_metrics_unittest.cc",
>+        "echo_remover_unittest.cc",
>+        "erl_estimator_unittest.cc",
>+        "erle_estimator_unittest.cc",
>+        "fft_data_unittest.cc",
>+        "frame_blocker_unittest.cc",
>+        "main_filter_update_gain_unittest.cc",
>+        "matched_filter_lag_aggregator_unittest.cc",
>+        "matched_filter_unittest.cc",
>+        "moving_average_unittest.cc",
>+        "render_buffer_unittest.cc",
>+        "render_delay_buffer_unittest.cc",
>+        "render_delay_controller_metrics_unittest.cc",
>+        "render_delay_controller_unittest.cc",
>+        "render_signal_analyzer_unittest.cc",
>+        "residual_echo_estimator_unittest.cc",
>+        "reverb_model_estimator_unittest.cc",
>+        "shadow_filter_update_gain_unittest.cc",
>+        "skew_estimator_unittest.cc",
>+        "subtractor_unittest.cc",
>+        "suppression_filter_unittest.cc",
>+        "suppression_gain_unittest.cc",
>+        "vector_math_unittest.cc",
>+      ]
>+    }
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/adaptive_fir_filter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/adaptive_fir_filter.cc
>index e080b4b590a..9a1e811fcfe 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/adaptive_fir_filter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/adaptive_fir_filter.cc
>@@ -10,10 +10,12 @@
> 
> #include "modules/audio_processing/aec3/adaptive_fir_filter.h"
> 
>+// Defines WEBRTC_ARCH_X86_FAMILY, used below.
>+#include "rtc_base/system/arch.h"
>+
> #if defined(WEBRTC_HAS_NEON)
> #include <arm_neon.h>
> #endif
>-#include "typedefs.h"  // NOLINT(build/include)
> #if defined(WEBRTC_ARCH_X86_FAMILY)
> #include <emmintrin.h>
> #endif
>@@ -416,16 +418,27 @@ void ApplyFilter_SSE2(const RenderBuffer& render_buffer,
> }  // namespace aec3
> 
> AdaptiveFirFilter::AdaptiveFirFilter(size_t max_size_partitions,
>+                                     size_t initial_size_partitions,
>+                                     size_t size_change_duration_blocks,
>                                      Aec3Optimization optimization,
>                                      ApmDataDumper* data_dumper)
>     : data_dumper_(data_dumper),
>       fft_(),
>       optimization_(optimization),
>       max_size_partitions_(max_size_partitions),
>+      size_change_duration_blocks_(
>+          static_cast<int>(size_change_duration_blocks)),
>+      current_size_partitions_(initial_size_partitions),
>+      target_size_partitions_(initial_size_partitions),
>+      old_target_size_partitions_(initial_size_partitions),
>       H_(max_size_partitions_),
>       H2_(max_size_partitions_, std::array<float, kFftLengthBy2Plus1>()),
>       h_(GetTimeDomainLength(max_size_partitions_), 0.f) {
>   RTC_DCHECK(data_dumper_);
>+  RTC_DCHECK_GE(max_size_partitions, initial_size_partitions);
>+
>+  RTC_DCHECK_LT(0, size_change_duration_blocks_);
>+  one_by_size_change_duration_blocks_ = 1.f / size_change_duration_blocks_;
> 
>   for (auto& H_j : H_) {
>     H_j.Clear();
>@@ -434,6 +447,7 @@ AdaptiveFirFilter::AdaptiveFirFilter(size_t max_size_partitions,
>     H2_k.fill(0.f);
>   }
>   erl_.fill(0.f);
>+  SetSizePartitions(current_size_partitions_, true);
> }
> 
> AdaptiveFirFilter::~AdaptiveFirFilter() = default;
>@@ -460,30 +474,64 @@ void AdaptiveFirFilter::HandleEchoPathChange() {
>   erl_.fill(0.f);
> }
> 
>-void AdaptiveFirFilter::SetSizePartitions(size_t size) {
>+void AdaptiveFirFilter::SetSizePartitions(size_t size, bool immediate_effect) {
>   RTC_DCHECK_EQ(max_size_partitions_, H_.capacity());
>   RTC_DCHECK_EQ(max_size_partitions_, H2_.capacity());
>   RTC_DCHECK_EQ(GetTimeDomainLength(max_size_partitions_), h_.capacity());
>   RTC_DCHECK_EQ(H_.size(), H2_.size());
>   RTC_DCHECK_EQ(h_.size(), GetTimeDomainLength(H_.size()));
>-
>-  if (size > max_size_partitions_) {
>-    RTC_LOG(LS_ERROR) << "Too large adaptive filter size specificed: " << size;
>-    size = max_size_partitions_;
>+  RTC_DCHECK_LE(size, max_size_partitions_);
>+
>+  target_size_partitions_ = std::min(max_size_partitions_, size);
>+  if (immediate_effect) {
>+    current_size_partitions_ = old_target_size_partitions_ =
>+        target_size_partitions_;
>+    ResetFilterBuffersToCurrentSize();
>+    size_change_counter_ = 0;
>+  } else {
>+    size_change_counter_ = size_change_duration_blocks_;
>   }
>+}
> 
>-  if (size < H_.size()) {
>-    for (size_t k = size; k < H_.size(); ++k) {
>+void AdaptiveFirFilter::ResetFilterBuffersToCurrentSize() {
>+  if (current_size_partitions_ < H_.size()) {
>+    for (size_t k = current_size_partitions_; k < H_.size(); ++k) {
>       H_[k].Clear();
>       H2_[k].fill(0.f);
>     }
>-
>-    std::fill(h_.begin() + GetTimeDomainLength(size), h_.end(), 0.f);
>+    std::fill(h_.begin() + GetTimeDomainLength(current_size_partitions_),
>+              h_.end(), 0.f);
>   }
> 
>-  H_.resize(size);
>-  H2_.resize(size);
>-  h_.resize(GetTimeDomainLength(size));
>+  H_.resize(current_size_partitions_);
>+  H2_.resize(current_size_partitions_);
>+  h_.resize(GetTimeDomainLength(current_size_partitions_));
>+  RTC_DCHECK_LT(0, current_size_partitions_);
>+  partition_to_constrain_ =
>+      std::min(partition_to_constrain_, current_size_partitions_ - 1);
>+}
>+
>+void AdaptiveFirFilter::UpdateSize() {
>+  RTC_DCHECK_GE(size_change_duration_blocks_, size_change_counter_);
>+  if (size_change_counter_ > 0) {
>+    --size_change_counter_;
>+
>+    auto average = [](float from, float to, float from_weight) {
>+      return from * from_weight + to * (1.f - from_weight);
>+    };
>+
>+    float change_factor =
>+        size_change_counter_ * one_by_size_change_duration_blocks_;
>+
>+    current_size_partitions_ = average(old_target_size_partitions_,
>+                                       target_size_partitions_, change_factor);
>+
>+    ResetFilterBuffersToCurrentSize();
>+  } else {
>+    current_size_partitions_ = old_target_size_partitions_ =
>+        target_size_partitions_;
>+  }
>+  RTC_DCHECK_LE(0, size_change_counter_);
> }
> 
> void AdaptiveFirFilter::Filter(const RenderBuffer& render_buffer,
>@@ -507,6 +555,9 @@ void AdaptiveFirFilter::Filter(const RenderBuffer& render_buffer,
> 
> void AdaptiveFirFilter::Adapt(const RenderBuffer& render_buffer,
>                               const FftData& G) {
>+  // Update the filter size if needed.
>+  UpdateSize();
>+
>   // Adapt the filter.
>   switch (optimization_) {
> #if defined(WEBRTC_ARCH_X86_FAMILY)
>@@ -567,4 +618,27 @@ void AdaptiveFirFilter::Constrain() {
>                                 : 0;
> }
> 
>+void AdaptiveFirFilter::ScaleFilter(float factor) {
>+  for (auto& H : H_) {
>+    for (auto& re : H.re) {
>+      re *= factor;
>+    }
>+    for (auto& im : H.im) {
>+      im *= factor;
>+    }
>+  }
>+  for (auto& h : h_) {
>+    h *= factor;
>+  }
>+}
>+
>+// Set the filter coefficients.
>+void AdaptiveFirFilter::SetFilter(const std::vector<FftData>& H) {
>+  const size_t num_partitions = std::min(H_.size(), H.size());
>+  for (size_t k = 0; k < num_partitions; ++k) {
>+    std::copy(H[k].re.begin(), H[k].re.end(), H_[k].re.begin());
>+    std::copy(H[k].im.begin(), H[k].im.end(), H_[k].im.begin());
>+  }
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/adaptive_fir_filter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/adaptive_fir_filter.h
>index e993c765871..5dfb4660fa2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/adaptive_fir_filter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/adaptive_fir_filter.h
>@@ -22,6 +22,7 @@
> #include "modules/audio_processing/aec3/render_buffer.h"
> #include "modules/audio_processing/logging/apm_data_dumper.h"
> #include "rtc_base/constructormagic.h"
>+#include "rtc_base/system/arch.h"
> 
> namespace webrtc {
> namespace aec3 {
>@@ -92,6 +93,8 @@ void ApplyFilter_SSE2(const RenderBuffer& render_buffer,
> class AdaptiveFirFilter {
>  public:
>   AdaptiveFirFilter(size_t max_size_partitions,
>+                    size_t initial_size_partitions,
>+                    size_t size_change_duration_blocks,
>                     Aec3Optimization optimization,
>                     ApmDataDumper* data_dumper);
> 
>@@ -111,7 +114,7 @@ class AdaptiveFirFilter {
>   size_t SizePartitions() const { return H_.size(); }
> 
>   // Sets the filter size.
>-  void SetSizePartitions(size_t size);
>+  void SetSizePartitions(size_t size, bool immediate_effect);
> 
>   // Returns the filter based echo return loss.
>   const std::array<float, kFftLengthBy2Plus1>& Erl() const { return erl_; }
>@@ -125,24 +128,51 @@ class AdaptiveFirFilter {
>   // Returns the estimate of the impulse response.
>   const std::vector<float>& FilterImpulseResponse() const { return h_; }
> 
>-  void DumpFilter(const char* name) {
>-    size_t current_size_partitions = H_.size();
>+  void DumpFilter(const char* name_frequency_domain,
>+                  const char* name_time_domain) {
>+    size_t current_size = H_.size();
>     H_.resize(max_size_partitions_);
>     for (auto& H : H_) {
>-      data_dumper_->DumpRaw(name, H.re);
>-      data_dumper_->DumpRaw(name, H.im);
>+      data_dumper_->DumpRaw(name_frequency_domain, H.re);
>+      data_dumper_->DumpRaw(name_frequency_domain, H.im);
>     }
>-    H_.resize(current_size_partitions);
>+    H_.resize(current_size);
>+
>+    current_size = h_.size();
>+    h_.resize(GetTimeDomainLength(max_size_partitions_));
>+    data_dumper_->DumpRaw(name_time_domain, h_);
>+    h_.resize(current_size);
>   }
> 
>+  // Scale the filter impulse response and spectrum by a factor.
>+  void ScaleFilter(float factor);
>+
>+  // Set the filter coefficients.
>+  void SetFilter(const std::vector<FftData>& H);
>+
>+  // Gets the filter coefficients.
>+  const std::vector<FftData>& GetFilter() const { return H_; }
>+
>  private:
>   // Constrain the filter partitions in a cyclic manner.
>   void Constrain();
> 
>+  // Resets the filter buffers to use the current size.
>+  void ResetFilterBuffersToCurrentSize();
>+
>+  // Gradually Updates the current filter size towards the target size.
>+  void UpdateSize();
>+
>   ApmDataDumper* const data_dumper_;
>   const Aec3Fft fft_;
>   const Aec3Optimization optimization_;
>   const size_t max_size_partitions_;
>+  const int size_change_duration_blocks_;
>+  float one_by_size_change_duration_blocks_;
>+  size_t current_size_partitions_;
>+  size_t target_size_partitions_;
>+  size_t old_target_size_partitions_;
>+  int size_change_counter_ = 0;
>   std::vector<FftData> H_;
>   std::vector<std::array<float, kFftLengthBy2Plus1>> H2_;
>   std::vector<float> h_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/adaptive_fir_filter_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/adaptive_fir_filter_unittest.cc
>index fe9da33d132..077586e51fa 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/adaptive_fir_filter_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/adaptive_fir_filter_unittest.cc
>@@ -10,11 +10,13 @@
> 
> #include "modules/audio_processing/aec3/adaptive_fir_filter.h"
> 
>+// Defines WEBRTC_ARCH_X86_FAMILY, used below.
>+#include "rtc_base/system/arch.h"
>+
> #include <math.h>
> #include <algorithm>
> #include <numeric>
> #include <string>
>-#include "typedefs.h"  // NOLINT(build/include)
> #if defined(WEBRTC_ARCH_X86_FAMILY)
> #include <emmintrin.h>
> #endif
>@@ -73,7 +75,7 @@ TEST(AdaptiveFirFilter, FilterAdaptationNeonOptimizations) {
>     }
>     render_delay_buffer->PrepareCaptureProcessing();
>   }
>-  const auto& render_buffer = render_delay_buffer->GetRenderBuffer();
>+  auto* const render_buffer = render_delay_buffer->GetRenderBuffer();
> 
>   for (size_t j = 0; j < G.re.size(); ++j) {
>     G.re[j] = j / 10001.f;
>@@ -183,7 +185,7 @@ TEST(AdaptiveFirFilter, FilterAdaptationSse2Optimizations) {
>         render_delay_buffer->Reset();
>       }
>       render_delay_buffer->PrepareCaptureProcessing();
>-      const auto& render_buffer = render_delay_buffer->GetRenderBuffer();
>+      auto* const render_buffer = render_delay_buffer->GetRenderBuffer();
> 
>       ApplyFilter_SSE2(*render_buffer, H_SSE2, &S_SSE2);
>       ApplyFilter(*render_buffer, H_C, &S_C);
>@@ -268,13 +270,13 @@ TEST(AdaptiveFirFilter, UpdateErlSse2Optimization) {
> #if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
> // Verifies that the check for non-null data dumper works.
> TEST(AdaptiveFirFilter, NullDataDumper) {
>-  EXPECT_DEATH(AdaptiveFirFilter(9, DetectOptimization(), nullptr), "");
>+  EXPECT_DEATH(AdaptiveFirFilter(9, 9, 250, DetectOptimization(), nullptr), "");
> }
> 
> // Verifies that the check for non-null filter output works.
> TEST(AdaptiveFirFilter, NullFilterOutput) {
>   ApmDataDumper data_dumper(42);
>-  AdaptiveFirFilter filter(9, DetectOptimization(), &data_dumper);
>+  AdaptiveFirFilter filter(9, 9, 250, DetectOptimization(), &data_dumper);
>   std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
>       RenderDelayBuffer::Create(EchoCanceller3Config(), 3));
>   EXPECT_DEATH(filter.Filter(*render_delay_buffer->GetRenderBuffer(), nullptr),
>@@ -287,7 +289,7 @@ TEST(AdaptiveFirFilter, NullFilterOutput) {
> // are turned on.
> TEST(AdaptiveFirFilter, FilterStatisticsAccess) {
>   ApmDataDumper data_dumper(42);
>-  AdaptiveFirFilter filter(9, DetectOptimization(), &data_dumper);
>+  AdaptiveFirFilter filter(9, 9, 250, DetectOptimization(), &data_dumper);
>   filter.Erl();
>   filter.FilterFrequencyResponse();
> }
>@@ -296,7 +298,8 @@ TEST(AdaptiveFirFilter, FilterStatisticsAccess) {
> TEST(AdaptiveFirFilter, FilterSize) {
>   ApmDataDumper data_dumper(42);
>   for (size_t filter_size = 1; filter_size < 5; ++filter_size) {
>-    AdaptiveFirFilter filter(filter_size, DetectOptimization(), &data_dumper);
>+    AdaptiveFirFilter filter(filter_size, filter_size, 250,
>+                             DetectOptimization(), &data_dumper);
>     EXPECT_EQ(filter_size, filter.SizePartitions());
>   }
> }
>@@ -308,22 +311,26 @@ TEST(AdaptiveFirFilter, FilterAndAdapt) {
>   ApmDataDumper data_dumper(42);
>   EchoCanceller3Config config;
>   AdaptiveFirFilter filter(config.filter.main.length_blocks,
>+                           config.filter.main.length_blocks,
>+                           config.filter.config_change_duration_blocks,
>                            DetectOptimization(), &data_dumper);
>   Aec3Fft fft;
>   config.delay.min_echo_path_delay_blocks = 0;
>   config.delay.default_delay = 1;
>   std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
>       RenderDelayBuffer::Create(config, 3));
>-  ShadowFilterUpdateGain gain(config.filter.shadow);
>+  ShadowFilterUpdateGain gain(config.filter.shadow,
>+                              config.filter.config_change_duration_blocks);
>   Random random_generator(42U);
>   std::vector<std::vector<float>> x(3, std::vector<float>(kBlockSize, 0.f));
>   std::vector<float> n(kBlockSize, 0.f);
>   std::vector<float> y(kBlockSize, 0.f);
>   AecState aec_state(EchoCanceller3Config{});
>-  RenderSignalAnalyzer render_signal_analyzer;
>+  RenderSignalAnalyzer render_signal_analyzer(config);
>+  absl::optional<DelayEstimate> delay_estimate;
>   std::vector<float> e(kBlockSize, 0.f);
>   std::array<float, kFftLength> s_scratch;
>-  std::array<float, kBlockSize> s;
>+  SubtractorOutput output;
>   FftData S;
>   FftData G;
>   FftData E;
>@@ -337,6 +344,7 @@ TEST(AdaptiveFirFilter, FilterAndAdapt) {
>   Y2.fill(0.f);
>   E2_main.fill(0.f);
>   E2_shadow.fill(0.f);
>+  output.Reset();
> 
>   constexpr float kScale = 1.0f / kFftLengthBy2;
> 
>@@ -346,27 +354,27 @@ TEST(AdaptiveFirFilter, FilterAndAdapt) {
>     CascadedBiQuadFilter y_hp_filter(kHighPassFilterCoefficients, 1);
> 
>     SCOPED_TRACE(ProduceDebugText(delay_samples));
>-    for (size_t k = 0; k < kNumBlocksToProcess; ++k) {
>+    for (size_t j = 0; j < kNumBlocksToProcess; ++j) {
>       RandomizeSampleVector(&random_generator, x[0]);
>       delay_buffer.Delay(x[0], y);
> 
>       RandomizeSampleVector(&random_generator, n);
>       static constexpr float kNoiseScaling = 1.f / 100.f;
>-      std::transform(
>-          y.begin(), y.end(), n.begin(), y.begin(),
>-          [](float a, float b) { return a + b * kNoiseScaling; });
>+      std::transform(y.begin(), y.end(), n.begin(), y.begin(),
>+                     [](float a, float b) { return a + b * kNoiseScaling; });
> 
>       x_hp_filter.Process(x[0]);
>       y_hp_filter.Process(y);
> 
>       render_delay_buffer->Insert(x);
>-      if (k == 0) {
>+      if (j == 0) {
>         render_delay_buffer->Reset();
>       }
>       render_delay_buffer->PrepareCaptureProcessing();
>-      const auto& render_buffer = render_delay_buffer->GetRenderBuffer();
>+      auto* const render_buffer = render_delay_buffer->GetRenderBuffer();
> 
>-      render_signal_analyzer.Update(*render_buffer, aec_state.FilterDelay());
>+      render_signal_analyzer.Update(*render_buffer,
>+                                    aec_state.FilterDelayBlocks());
> 
>       filter.Filter(*render_buffer, &S);
>       fft.Ifft(S, &s_scratch);
>@@ -377,7 +385,7 @@ TEST(AdaptiveFirFilter, FilterAndAdapt) {
>                     [](float& a) { a = rtc::SafeClamp(a, -32768.f, 32767.f); });
>       fft.ZeroPaddedFft(e, Aec3Fft::Window::kRectangular, &E);
>       for (size_t k = 0; k < kBlockSize; ++k) {
>-        s[k] = kScale * s_scratch[k + kFftLengthBy2];
>+        output.s_main[k] = kScale * s_scratch[k + kFftLengthBy2];
>       }
> 
>       std::array<float, kFftLengthBy2Plus1> render_power;
>@@ -387,15 +395,14 @@ TEST(AdaptiveFirFilter, FilterAndAdapt) {
>       filter.Adapt(*render_buffer, G);
>       aec_state.HandleEchoPathChange(EchoPathVariability(
>           false, EchoPathVariability::DelayAdjustment::kNone, false));
>-      aec_state.Update(filter.FilterFrequencyResponse(),
>-                       filter.FilterImpulseResponse(), true, *render_buffer,
>-                       E2_main, Y2, s, false);
>+
>+      aec_state.Update(delay_estimate, filter.FilterFrequencyResponse(),
>+                       filter.FilterImpulseResponse(), *render_buffer, E2_main,
>+                       Y2, output, y);
>     }
>     // Verify that the filter is able to perform well.
>     EXPECT_LT(1000 * std::inner_product(e.begin(), e.end(), e.begin(), 0.f),
>               std::inner_product(y.begin(), y.end(), y.begin(), 0.f));
>-    EXPECT_EQ(delay_samples / kBlockSize,
>-              static_cast<size_t>(aec_state.FilterDelay()));
>   }
> }
> }  // namespace aec3
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec3_common.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec3_common.cc
>index 7becce49f17..c374a3517d6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec3_common.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec3_common.cc
>@@ -10,9 +10,11 @@
> 
> #include "modules/audio_processing/aec3/aec3_common.h"
> 
>-#include "typedefs.h"  // NOLINT(build/include)
> #include "system_wrappers/include/cpu_features_wrapper.h"
> 
>+#include "rtc_base/checks.h"
>+#include "rtc_base/system/arch.h"
>+
> namespace webrtc {
> 
> Aec3Optimization DetectOptimization() {
>@@ -29,4 +31,25 @@ Aec3Optimization DetectOptimization() {
>   return Aec3Optimization::kNone;
> }
> 
>+float FastApproxLog2f(const float in) {
>+  RTC_DCHECK_GT(in, .0f);
>+  // Read and interpret float as uint32_t and then cast to float.
>+  // This is done to extract the exponent (bits 30 - 23).
>+  // "Right shift" of the exponent is then performed by multiplying
>+  // with the constant (1/2^23). Finally, we subtract a constant to
>+  // remove the bias (https://en.wikipedia.org/wiki/Exponent_bias).
>+  union {
>+    float dummy;
>+    uint32_t a;
>+  } x = {in};
>+  float out = x.a;
>+  out *= 1.1920929e-7f;  // 1/2^23
>+  out -= 126.942695f;    // Remove bias.
>+  return out;
>+}
>+
>+float Log2TodB(const float in_log2) {
>+  return 3.0102999566398121 * in_log2;
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec3_common.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec3_common.h
>index 47f078415aa..56c7a9024a3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec3_common.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec3_common.h
>@@ -12,7 +12,6 @@
> #define MODULES_AUDIO_PROCESSING_AEC3_AEC3_COMMON_H_
> 
> #include <stddef.h>
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -37,6 +36,7 @@ constexpr size_t kFftLengthBy2 = 64;
> constexpr size_t kFftLengthBy2Plus1 = kFftLengthBy2 + 1;
> constexpr size_t kFftLengthBy2Minus1 = kFftLengthBy2 - 1;
> constexpr size_t kFftLength = 2 * kFftLengthBy2;
>+constexpr size_t kFftLengthBy2Log2 = 6;
> 
> constexpr int kMaxAdaptiveFilterLength = 50;
> constexpr int kRenderTransferQueueSizeFrames = 100;
>@@ -45,7 +45,7 @@ constexpr size_t kMaxNumBands = 3;
> constexpr size_t kSubFrameLength = 80;
> 
> constexpr size_t kBlockSize = kFftLengthBy2;
>-constexpr size_t kBlockSizeLog2 = 6;
>+constexpr size_t kBlockSizeLog2 = kFftLengthBy2Log2;
> 
> constexpr size_t kExtendedBlockSize = 2 * kFftLengthBy2;
> constexpr size_t kMatchedFilterWindowSizeSubBlocks = 32;
>@@ -88,9 +88,18 @@ constexpr size_t GetRenderDelayBufferSize(size_t down_sampling_factor,
> // Detects what kind of optimizations to use for the code.
> Aec3Optimization DetectOptimization();
> 
>+// Computes the log2 of the input in a fast an approximate manner.
>+float FastApproxLog2f(const float in);
>+
>+// Returns dB from a power quantity expressed in log2.
>+float Log2TodB(const float in_log2);
>+
> static_assert(1 << kBlockSizeLog2 == kBlockSize,
>               "Proper number of shifts for blocksize");
> 
>+static_assert(1 << kFftLengthBy2Log2 == kFftLengthBy2,
>+              "Proper number of shifts for the fft length");
>+
> static_assert(1 == NumBandsForRate(8000), "Number of bands for 8 kHz");
> static_assert(1 == NumBandsForRate(16000), "Number of bands for 16 kHz");
> static_assert(2 == NumBandsForRate(32000), "Number of bands for 32 kHz");
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec3_fft.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec3_fft.cc
>index d6690360f86..b02f3425b71 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec3_fft.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec3_fft.cc
>@@ -33,6 +33,41 @@ const float kHanning64[kFftLengthBy2] = {
>     0.15088159f, 0.11697778f, 0.08688061f, 0.06088921f, 0.03926189f,
>     0.0222136f,  0.00991376f, 0.00248461f, 0.f};
> 
>+// Hanning window from Matlab command win = sqrt(hanning(128)).
>+const float kSqrtHanning128[kFftLength] = {
>+    0.00000000000000f, 0.02454122852291f, 0.04906767432742f, 0.07356456359967f,
>+    0.09801714032956f, 0.12241067519922f, 0.14673047445536f, 0.17096188876030f,
>+    0.19509032201613f, 0.21910124015687f, 0.24298017990326f, 0.26671275747490f,
>+    0.29028467725446f, 0.31368174039889f, 0.33688985339222f, 0.35989503653499f,
>+    0.38268343236509f, 0.40524131400499f, 0.42755509343028f, 0.44961132965461f,
>+    0.47139673682600f, 0.49289819222978f, 0.51410274419322f, 0.53499761988710f,
>+    0.55557023301960f, 0.57580819141785f, 0.59569930449243f, 0.61523159058063f,
>+    0.63439328416365f, 0.65317284295378f, 0.67155895484702f, 0.68954054473707f,
>+    0.70710678118655f, 0.72424708295147f, 0.74095112535496f, 0.75720884650648f,
>+    0.77301045336274f, 0.78834642762661f, 0.80320753148064f, 0.81758481315158f,
>+    0.83146961230255f, 0.84485356524971f, 0.85772861000027f, 0.87008699110871f,
>+    0.88192126434835f, 0.89322430119552f, 0.90398929312344f, 0.91420975570353f,
>+    0.92387953251129f, 0.93299279883474f, 0.94154406518302f, 0.94952818059304f,
>+    0.95694033573221f, 0.96377606579544f, 0.97003125319454f, 0.97570213003853f,
>+    0.98078528040323f, 0.98527764238894f, 0.98917650996478f, 0.99247953459871f,
>+    0.99518472667220f, 0.99729045667869f, 0.99879545620517f, 0.99969881869620f,
>+    1.00000000000000f, 0.99969881869620f, 0.99879545620517f, 0.99729045667869f,
>+    0.99518472667220f, 0.99247953459871f, 0.98917650996478f, 0.98527764238894f,
>+    0.98078528040323f, 0.97570213003853f, 0.97003125319454f, 0.96377606579544f,
>+    0.95694033573221f, 0.94952818059304f, 0.94154406518302f, 0.93299279883474f,
>+    0.92387953251129f, 0.91420975570353f, 0.90398929312344f, 0.89322430119552f,
>+    0.88192126434835f, 0.87008699110871f, 0.85772861000027f, 0.84485356524971f,
>+    0.83146961230255f, 0.81758481315158f, 0.80320753148064f, 0.78834642762661f,
>+    0.77301045336274f, 0.75720884650648f, 0.74095112535496f, 0.72424708295147f,
>+    0.70710678118655f, 0.68954054473707f, 0.67155895484702f, 0.65317284295378f,
>+    0.63439328416365f, 0.61523159058063f, 0.59569930449243f, 0.57580819141785f,
>+    0.55557023301960f, 0.53499761988710f, 0.51410274419322f, 0.49289819222978f,
>+    0.47139673682600f, 0.44961132965461f, 0.42755509343028f, 0.40524131400499f,
>+    0.38268343236509f, 0.35989503653499f, 0.33688985339222f, 0.31368174039889f,
>+    0.29028467725446f, 0.26671275747490f, 0.24298017990326f, 0.21910124015687f,
>+    0.19509032201613f, 0.17096188876030f, 0.14673047445536f, 0.12241067519922f,
>+    0.09801714032956f, 0.07356456359967f, 0.04906767432742f, 0.02454122852291f};
>+
> }  // namespace
> 
> // TODO(peah): Change x to be std::array once the rest of the code allows this.
>@@ -52,6 +87,9 @@ void Aec3Fft::ZeroPaddedFft(rtc::ArrayView<const float> x,
>                      fft.begin() + kFftLengthBy2,
>                      [](float a, float b) { return a * b; });
>       break;
>+    case Window::kSqrtHanning:
>+      RTC_NOTREACHED();
>+      break;
>     default:
>       RTC_NOTREACHED();
>   }
>@@ -60,15 +98,33 @@ void Aec3Fft::ZeroPaddedFft(rtc::ArrayView<const float> x,
> }
> 
> void Aec3Fft::PaddedFft(rtc::ArrayView<const float> x,
>-                        rtc::ArrayView<float> x_old,
>+                        rtc::ArrayView<const float> x_old,
>+                        Window window,
>                         FftData* X) const {
>   RTC_DCHECK(X);
>   RTC_DCHECK_EQ(kFftLengthBy2, x.size());
>   RTC_DCHECK_EQ(kFftLengthBy2, x_old.size());
>   std::array<float, kFftLength> fft;
>-  std::copy(x_old.begin(), x_old.end(), fft.begin());
>-  std::copy(x.begin(), x.end(), fft.begin() + x_old.size());
>-  std::copy(x.begin(), x.end(), x_old.begin());
>+
>+  switch (window) {
>+    case Window::kRectangular:
>+      std::copy(x_old.begin(), x_old.end(), fft.begin());
>+      std::copy(x.begin(), x.end(), fft.begin() + x_old.size());
>+      break;
>+    case Window::kHanning:
>+      RTC_NOTREACHED();
>+      break;
>+    case Window::kSqrtHanning:
>+      std::transform(x_old.begin(), x_old.end(), std::begin(kSqrtHanning128),
>+                     fft.begin(), std::multiplies<float>());
>+      std::transform(x.begin(), x.end(),
>+                     std::begin(kSqrtHanning128) + x_old.size(),
>+                     fft.begin() + x_old.size(), std::multiplies<float>());
>+      break;
>+    default:
>+      RTC_NOTREACHED();
>+  }
>+
>   Fft(&fft, X);
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec3_fft.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec3_fft.h
>index f3dddb3f1bb..b70022255d9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec3_fft.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec3_fft.h
>@@ -25,7 +25,7 @@ namespace webrtc {
> // FftData type.
> class Aec3Fft {
>  public:
>-  enum class Window { kRectangular, kHanning };
>+  enum class Window { kRectangular, kHanning, kSqrtHanning };
> 
>   Aec3Fft() = default;
>   // Computes the FFT. Note that both the input and output are modified.
>@@ -51,7 +51,15 @@ class Aec3Fft {
>   // Concatenates the kFftLengthBy2 values long x and x_old before computing the
>   // Fft. After that, x is copied to x_old.
>   void PaddedFft(rtc::ArrayView<const float> x,
>-                 rtc::ArrayView<float> x_old,
>+                 rtc::ArrayView<const float> x_old,
>+                 FftData* X) const {
>+    PaddedFft(x, x_old, Window::kRectangular, X);
>+  }
>+
>+  // Padded Fft using a time-domain window.
>+  void PaddedFft(rtc::ArrayView<const float> x,
>+                 rtc::ArrayView<const float> x_old,
>+                 Window window,
>                  FftData* X) const;
> 
>  private:
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec3_fft_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec3_fft_unittest.cc
>index 87fe7a8fbbe..82d6e766cc8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec3_fft_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec3_fft_unittest.cc
>@@ -199,6 +199,7 @@ TEST(Aec3Fft, PaddedFft) {
>     std::for_each(x_ref.begin(), x_ref.end(), [](float& a) { a *= 64.f; });
> 
>     fft.PaddedFft(x_in, x_old, &X);
>+    std::copy(x_in.begin(), x_in.end(), x_old.begin());
>     fft.Ifft(X, &x_out);
> 
>     for (size_t j = 0; j < x_out.size(); ++j) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec_state.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec_state.cc
>index e225e032347..0ca8618bee7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec_state.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec_state.cc
>@@ -15,38 +15,71 @@
> #include <numeric>
> #include <vector>
> 
>+#include "absl/types/optional.h"
> #include "api/array_view.h"
>+#include "modules/audio_processing/aec3/aec3_common.h"
> #include "modules/audio_processing/logging/apm_data_dumper.h"
> #include "rtc_base/atomicops.h"
> #include "rtc_base/checks.h"
>+#include "system_wrappers/include/field_trial.h"
> 
> namespace webrtc {
> namespace {
> 
>-// Computes delay of the adaptive filter.
>-int EstimateFilterDelay(
>-    const std::vector<std::array<float, kFftLengthBy2Plus1>>&
>-        adaptive_filter_frequency_response) {
>-  const auto& H2 = adaptive_filter_frequency_response;
>-  constexpr size_t kUpperBin = kFftLengthBy2 - 5;
>-  RTC_DCHECK_GE(kMaxAdaptiveFilterLength, H2.size());
>-  std::array<int, kMaxAdaptiveFilterLength> delays;
>-  delays.fill(0);
>-  for (size_t k = 1; k < kUpperBin; ++k) {
>-    // Find the maximum of H2[j].
>-    size_t peak = 0;
>-    for (size_t j = 0; j < H2.size(); ++j) {
>-      if (H2[j][k] > H2[peak][k]) {
>-        peak = j;
>-      }
>-    }
>-    ++delays[peak];
>-  }
>+bool EnableTransparentMode() {
>+  return !field_trial::IsEnabled("WebRTC-Aec3TransparentModeKillSwitch");
>+}
>+
>+bool EnableStationaryRenderImprovements() {
>+  return !field_trial::IsEnabled(
>+      "WebRTC-Aec3StationaryRenderImprovementsKillSwitch");
>+}
>+
>+bool EnableEnforcingDelayAfterRealignment() {
>+  return !field_trial::IsEnabled(
>+      "WebRTC-Aec3EnforceDelayAfterRealignmentKillSwitch");
>+}
>+
>+bool EnableLinearModeWithDivergedFilter() {
>+  return !field_trial::IsEnabled(
>+      "WebRTC-Aec3LinearModeWithDivergedFilterKillSwitch");
>+}
>+
>+bool EnableEarlyFilterUsage() {
>+  return !field_trial::IsEnabled("WebRTC-Aec3EarlyLinearFilterUsageKillSwitch");
>+}
>+
>+bool EnableShortInitialState() {
>+  return !field_trial::IsEnabled("WebRTC-Aec3ShortInitialStateKillSwitch");
>+}
>+
>+bool EnableNoWaitForAlignment() {
>+  return !field_trial::IsEnabled("WebRTC-Aec3NoAlignmentWaitKillSwitch");
>+}
>+
>+bool EnableConvergenceTriggeredLinearMode() {
>+  return !field_trial::IsEnabled(
>+      "WebRTC-Aec3ConvergenceTriggingLinearKillSwitch");
>+}
>+
>+bool EnableUncertaintyUntilSufficientAdapted() {
>+  return !field_trial::IsEnabled(
>+      "WebRTC-Aec3ErleUncertaintyUntilSufficientlyAdaptedKillSwitch");
>+}
>+
>+bool TreatTransparentModeAsNonlinear() {
>+  return !field_trial::IsEnabled(
>+      "WebRTC-Aec3TreatTransparentModeAsNonlinearKillSwitch");
>+}
> 
>-  return std::distance(delays.begin(),
>-                       std::max_element(delays.begin(), delays.end()));
>+float ComputeGainRampupIncrease(const EchoCanceller3Config& config) {
>+  const auto& c = config.echo_removal_control.gain_rampup;
>+  return powf(1.f / c.first_non_zero_gain, 1.f / c.non_zero_gain_blocks);
> }
> 
>+constexpr size_t kBlocksSinceConvergencedFilterInit = 10000;
>+constexpr size_t kBlocksSinceConsistentEstimateInit = 10000;
>+
> }  // namespace
> 
> int AecState::instance_count_ = 0;
>@@ -54,207 +87,297 @@ int AecState::instance_count_ = 0;
> AecState::AecState(const EchoCanceller3Config& config)
>     : data_dumper_(
>           new ApmDataDumper(rtc::AtomicOps::Increment(&instance_count_))),
>-      erle_estimator_(config.erle.min, config.erle.max_l, config.erle.max_h),
>       config_(config),
>+      allow_transparent_mode_(EnableTransparentMode()),
>+      use_stationary_properties_(
>+          EnableStationaryRenderImprovements() &&
>+          config_.echo_audibility.use_stationary_properties),
>+      enforce_delay_after_realignment_(EnableEnforcingDelayAfterRealignment()),
>+      allow_linear_mode_with_diverged_filter_(
>+          EnableLinearModeWithDivergedFilter()),
>+      early_filter_usage_activated_(EnableEarlyFilterUsage()),
>+      use_short_initial_state_(EnableShortInitialState()),
>+      convergence_trigger_linear_mode_(EnableConvergenceTriggeredLinearMode()),
>+      no_alignment_required_for_linear_mode_(EnableNoWaitForAlignment()),
>+      use_uncertainty_until_sufficiently_adapted_(
>+          EnableUncertaintyUntilSufficientAdapted()),
>+      transparent_mode_enforces_nonlinear_mode_(
>+          TreatTransparentModeAsNonlinear()),
>+      erle_estimator_(config.erle.min, config.erle.max_l, config.erle.max_h),
>       max_render_(config_.filter.main.length_blocks, 0.f),
>-      reverb_decay_(config_.ep_strength.default_len) {}
>+      gain_rampup_increase_(ComputeGainRampupIncrease(config_)),
>+      suppression_gain_limiter_(config_),
>+      filter_analyzer_(config_),
>+      blocks_since_converged_filter_(kBlocksSinceConvergencedFilterInit),
>+      active_blocks_since_consistent_filter_estimate_(
>+          kBlocksSinceConsistentEstimateInit),
>+      reverb_model_estimator_(config) {}
> 
> AecState::~AecState() = default;
> 
> void AecState::HandleEchoPathChange(
>     const EchoPathVariability& echo_path_variability) {
>   const auto full_reset = [&]() {
>+    filter_analyzer_.Reset();
>     blocks_since_last_saturation_ = 0;
>     usable_linear_estimate_ = false;
>-    echo_leakage_detected_ = false;
>+    diverged_linear_filter_ = false;
>     capture_signal_saturation_ = false;
>     echo_saturation_ = false;
>-    previous_max_sample_ = 0.f;
>     std::fill(max_render_.begin(), max_render_.end(), 0.f);
>-    force_zero_gain_counter_ = 0;
>     blocks_with_proper_filter_adaptation_ = 0;
>-    capture_block_counter_ = 0;
>+    blocks_since_reset_ = 0;
>     filter_has_had_time_to_converge_ = false;
>     render_received_ = false;
>-    force_zero_gain_ = true;
>     blocks_with_active_render_ = 0;
>     initial_state_ = true;
>+    suppression_gain_limiter_.Reset();
>+    blocks_since_converged_filter_ = kBlocksSinceConvergencedFilterInit;
>+    diverged_blocks_ = 0;
>   };
> 
>   // TODO(peah): Refine the reset scheme according to the type of gain and
>   // delay adjustment.
>-  if (echo_path_variability.gain_change) {
>-    full_reset();
>-  }
> 
>   if (echo_path_variability.delay_change !=
>-      EchoPathVariability::DelayAdjustment::kBufferReadjustment) {
>-    full_reset();
>-  } else if (echo_path_variability.delay_change !=
>-             EchoPathVariability::DelayAdjustment::kBufferFlush) {
>+      EchoPathVariability::DelayAdjustment::kNone) {
>     full_reset();
>-
>-  } else if (echo_path_variability.delay_change !=
>-             EchoPathVariability::DelayAdjustment::kDelayReset) {
>-    full_reset();
>-  } else if (echo_path_variability.delay_change !=
>-             EchoPathVariability::DelayAdjustment::kNewDetectedDelay) {
>-    full_reset();
>-  } else if (echo_path_variability.gain_change) {
>-    capture_block_counter_ = kNumBlocksPerSecond;
>   }
>+
>+  subtractor_output_analyzer_.HandleEchoPathChange();
> }
> 
> void AecState::Update(
>+    const absl::optional<DelayEstimate>& external_delay,
>     const std::vector<std::array<float, kFftLengthBy2Plus1>>&
>         adaptive_filter_frequency_response,
>     const std::vector<float>& adaptive_filter_impulse_response,
>-    bool converged_filter,
>     const RenderBuffer& render_buffer,
>     const std::array<float, kFftLengthBy2Plus1>& E2_main,
>     const std::array<float, kFftLengthBy2Plus1>& Y2,
>-    const std::array<float, kBlockSize>& s,
>-    bool echo_leakage_detected) {
>-  // Store input parameters.
>-  echo_leakage_detected_ = echo_leakage_detected;
>+    const SubtractorOutput& subtractor_output,
>+    rtc::ArrayView<const float> y) {
>+  // Analyze the filter output.
>+  subtractor_output_analyzer_.Update(subtractor_output);
>+
>+  const bool converged_filter = subtractor_output_analyzer_.ConvergedFilter();
>+  const bool diverged_filter = subtractor_output_analyzer_.DivergedFilter();
>+
>+  // Analyze the filter and compute the delays.
>+  filter_analyzer_.Update(adaptive_filter_impulse_response,
>+                          adaptive_filter_frequency_response, render_buffer);
>+  filter_delay_blocks_ = filter_analyzer_.DelayBlocks();
>+  if (enforce_delay_after_realignment_) {
>+    if (external_delay &&
>+        (!external_delay_ || external_delay_->delay != external_delay->delay)) {
>+      frames_since_external_delay_change_ = 0;
>+      external_delay_ = external_delay;
>+    }
>+    if (blocks_with_proper_filter_adaptation_ < 2 * kNumBlocksPerSecond &&
>+        external_delay_) {
>+      filter_delay_blocks_ = config_.delay.delay_headroom_blocks;
>+    }
>+  }
>+
>+  if (filter_analyzer_.Consistent()) {
>+    internal_delay_ = filter_analyzer_.DelayBlocks();
>+  } else {
>+    internal_delay_ = absl::nullopt;
>+  }
>+
>+  external_delay_seen_ = external_delay_seen_ || external_delay;
> 
>-  // Estimate the filter delay.
>-  filter_delay_ = EstimateFilterDelay(adaptive_filter_frequency_response);
>-  const std::vector<float>& x = render_buffer.Block(-filter_delay_)[0];
>+  const std::vector<float>& x = render_buffer.Block(-filter_delay_blocks_)[0];
> 
>   // Update counters.
>   ++capture_block_counter_;
>+  ++blocks_since_reset_;
>   const bool active_render_block = DetectActiveRender(x);
>   blocks_with_active_render_ += active_render_block ? 1 : 0;
>   blocks_with_proper_filter_adaptation_ +=
>       active_render_block && !SaturatedCapture() ? 1 : 0;
> 
>-  // Force zero echo suppression gain after an echo path change to allow at
>-  // least some render data to be collected in order to avoid an initial echo
>-  // burst.
>-  force_zero_gain_ = ++force_zero_gain_counter_ < kNumBlocksPerSecond / 5;
>+  // Update the limit on the echo suppression after an echo path change to avoid
>+  // an initial echo burst.
>+  suppression_gain_limiter_.Update(render_buffer.GetRenderActivity(),
>+                                   transparent_mode_);
> 
>+  if (UseStationaryProperties()) {
>+    // Update the echo audibility evaluator.
>+    echo_audibility_.Update(
>+        render_buffer, FilterDelayBlocks(), external_delay_seen_,
>+        config_.ep_strength.reverb_based_on_render ? ReverbDecay() : 0.f);
>+  }
> 
>   // Update the ERL and ERLE measures.
>-  if (converged_filter && capture_block_counter_ >= 2 * kNumBlocksPerSecond) {
>-    const auto& X2 = render_buffer.Spectrum(filter_delay_);
>-    erle_estimator_.Update(X2, Y2, E2_main);
>-    erl_estimator_.Update(X2, Y2);
>+  if (blocks_since_reset_ >= 2 * kNumBlocksPerSecond) {
>+    const auto& X2 = render_buffer.Spectrum(filter_delay_blocks_);
>+    erle_estimator_.Update(X2, Y2, E2_main, converged_filter);
>+    if (converged_filter) {
>+      erl_estimator_.Update(X2, Y2);
>+    }
>   }
> 
>-  // Update the echo audibility evaluator.
>-  echo_audibility_.Update(x, s, converged_filter);
>-
>   // Detect and flag echo saturation.
>-  // TODO(peah): Add the delay in this computation to ensure that the render and
>-  // capture signals are properly aligned.
>   if (config_.ep_strength.echo_can_saturate) {
>-    echo_saturation_ = DetectEchoSaturation(x);
>+    echo_saturation_ = DetectEchoSaturation(x, EchoPathGain());
>+  }
>+
>+  if (early_filter_usage_activated_) {
>+    filter_has_had_time_to_converge_ =
>+        blocks_with_proper_filter_adaptation_ >= 0.8f * kNumBlocksPerSecond;
>+  } else {
>+    filter_has_had_time_to_converge_ =
>+        blocks_with_proper_filter_adaptation_ >= 1.5f * kNumBlocksPerSecond;
>   }
> 
>-  // TODO(peah): Move?
>-  filter_has_had_time_to_converge_ =
>-      blocks_with_proper_filter_adaptation_ >= 1.5f * kNumBlocksPerSecond;
>+  if (!filter_should_have_converged_) {
>+    filter_should_have_converged_ =
>+        blocks_with_proper_filter_adaptation_ > 6 * kNumBlocksPerSecond;
>+  }
> 
>-  initial_state_ =
>-      blocks_with_proper_filter_adaptation_ < 5 * kNumBlocksPerSecond;
>+  // Flag whether the initial state is still active.
>+  if (use_short_initial_state_) {
>+    initial_state_ =
>+        blocks_with_proper_filter_adaptation_ < 2.5f * kNumBlocksPerSecond;
>+  } else {
>+    initial_state_ =
>+        blocks_with_proper_filter_adaptation_ < 5 * kNumBlocksPerSecond;
>+  }
> 
>-  // Flag whether the linear filter estimate is usable.
>-  usable_linear_estimate_ =
>-      !echo_saturation_ &&
>-      (converged_filter && filter_has_had_time_to_converge_) &&
>-      capture_block_counter_ >= 1.f * kNumBlocksPerSecond && !TransparentMode();
>+  // Update counters for the filter divergence and convergence.
>+  diverged_blocks_ = diverged_filter ? diverged_blocks_ + 1 : 0;
>+  if (diverged_blocks_ >= 60) {
>+    blocks_since_converged_filter_ = kBlocksSinceConvergencedFilterInit;
>+  } else {
>+    blocks_since_converged_filter_ =
>+        converged_filter ? 0 : blocks_since_converged_filter_ + 1;
>+  }
>+  if (converged_filter) {
>+    active_blocks_since_converged_filter_ = 0;
>+  } else if (active_render_block) {
>+    ++active_blocks_since_converged_filter_;
>+  }
> 
>-  // After an amount of active render samples for which an echo should have been
>-  // detected in the capture signal if the ERL was not infinite, flag that a
>-  // transparent mode should be entered.
>-  transparent_mode_ =
>-      !converged_filter &&
>-      (blocks_with_active_render_ == 0 ||
>-       blocks_with_proper_filter_adaptation_ >= 5 * kNumBlocksPerSecond);
>-}
>+  bool recently_converged_filter =
>+      blocks_since_converged_filter_ < 60 * kNumBlocksPerSecond;
> 
>-void AecState::UpdateReverb(const std::vector<float>& impulse_response) {
>-  if ((!(filter_delay_ && usable_linear_estimate_)) ||
>-      (filter_delay_ >
>-       static_cast<int>(config_.filter.main.length_blocks) - 4)) {
>-    return;
>+  if (blocks_since_converged_filter_ > 20 * kNumBlocksPerSecond) {
>+    converged_filter_count_ = 0;
>+  } else if (converged_filter) {
>+    ++converged_filter_count_;
>+  }
>+  if (converged_filter_count_ > 50) {
>+    finite_erl_ = true;
>   }
> 
>-  // Form the data to match against by squaring the impulse response
>-  // coefficients.
>-  std::array<float, GetTimeDomainLength(kMaxAdaptiveFilterLength)>
>-      matching_data_data;
>-  RTC_DCHECK_LE(GetTimeDomainLength(config_.filter.main.length_blocks),
>-                matching_data_data.size());
>-  rtc::ArrayView<float> matching_data(
>-      matching_data_data.data(),
>-      GetTimeDomainLength(config_.filter.main.length_blocks));
>-  std::transform(impulse_response.begin(), impulse_response.end(),
>-                 matching_data.begin(), [](float a) { return a * a; });
>-
>-  // Avoid matching against noise in the model by subtracting an estimate of the
>-  // model noise power.
>-  constexpr size_t kTailLength = 64;
>-  const size_t tail_index =
>-      GetTimeDomainLength(config_.filter.main.length_blocks) - kTailLength;
>-  const float tail_power = *std::max_element(matching_data.begin() + tail_index,
>-                                             matching_data.end());
>-  std::for_each(matching_data.begin(), matching_data.begin() + tail_index,
>-                [tail_power](float& a) { a = std::max(0.f, a - tail_power); });
>-
>-  // Identify the peak index of the impulse response.
>-  const size_t peak_index = *std::max_element(
>-      matching_data.begin(), matching_data.begin() + tail_index);
>-
>-  if (peak_index + 128 < tail_index) {
>-    size_t start_index = peak_index + 64;
>-    // Compute the matching residual error for the current candidate to match.
>-    float residual_sqr_sum = 0.f;
>-    float d_k = reverb_decay_to_test_;
>-    for (size_t k = start_index; k < tail_index; ++k) {
>-      if (matching_data[start_index + 1] == 0.f) {
>-        break;
>-      }
>-
>-      float residual = matching_data[k] - matching_data[peak_index] * d_k;
>-      residual_sqr_sum += residual * residual;
>-      d_k *= reverb_decay_to_test_;
>-    }
>+  if (filter_analyzer_.Consistent() && filter_delay_blocks_ < 5) {
>+    consistent_filter_seen_ = true;
>+    active_blocks_since_consistent_filter_estimate_ = 0;
>+  } else if (active_render_block) {
>+    ++active_blocks_since_consistent_filter_estimate_;
>+  }
> 
>-    // If needed, update the best candidate for the reverb decay.
>-    if (reverb_decay_candidate_residual_ < 0.f ||
>-        residual_sqr_sum < reverb_decay_candidate_residual_) {
>-      reverb_decay_candidate_residual_ = residual_sqr_sum;
>-      reverb_decay_candidate_ = reverb_decay_to_test_;
>-    }
>+  bool consistent_filter_estimate_not_seen;
>+  if (!consistent_filter_seen_) {
>+    consistent_filter_estimate_not_seen =
>+        capture_block_counter_ > 5 * kNumBlocksPerSecond;
>+  } else {
>+    consistent_filter_estimate_not_seen =
>+        active_blocks_since_consistent_filter_estimate_ >
>+        30 * kNumBlocksPerSecond;
>   }
> 
>-  // Compute the next reverb candidate to evaluate such that all candidates will
>-  // be evaluated within one second.
>-  reverb_decay_to_test_ += (0.9965f - 0.9f) / (5 * kNumBlocksPerSecond);
>+  converged_filter_seen_ = converged_filter_seen_ || converged_filter;
> 
>-  // If all reverb candidates have been evaluated, choose the best one as the
>-  // reverb decay.
>-  if (reverb_decay_to_test_ >= 0.9965f) {
>-    if (reverb_decay_candidate_residual_ < 0.f) {
>-      // Transform the decay to be in the unit of blocks.
>-      reverb_decay_ = powf(reverb_decay_candidate_, kFftLengthBy2);
>+  // If no filter convergence is seen for a long time, reset the estimated
>+  // properties of the echo path.
>+  if (active_blocks_since_converged_filter_ > 60 * kNumBlocksPerSecond) {
>+    converged_filter_seen_ = false;
>+    finite_erl_ = false;
>+  }
> 
>-      // Limit the estimated reverb_decay_ to the maximum one needed in practice
>-      // to minimize the impact of incorrect estimates.
>-      reverb_decay_ = std::min(config_.ep_strength.default_len, reverb_decay_);
>-    }
>-    reverb_decay_to_test_ = 0.9f;
>-    reverb_decay_candidate_residual_ = -1.f;
>+  // After an amount of active render samples for which an echo should have been
>+  // detected in the capture signal if the ERL was not infinite, flag that a
>+  // transparent mode should be entered.
>+  transparent_mode_ = !config_.ep_strength.bounded_erl && !finite_erl_;
>+  transparent_mode_ =
>+      transparent_mode_ &&
>+      (consistent_filter_estimate_not_seen || !converged_filter_seen_);
>+  transparent_mode_ = transparent_mode_ && filter_should_have_converged_;
>+  transparent_mode_ = transparent_mode_ && allow_transparent_mode_;
>+
>+  usable_linear_estimate_ = !echo_saturation_;
>+
>+  if (convergence_trigger_linear_mode_) {
>+    usable_linear_estimate_ =
>+        usable_linear_estimate_ &&
>+        ((filter_has_had_time_to_converge_ && external_delay) ||
>+         converged_filter_seen_);
>+  } else {
>+    usable_linear_estimate_ =
>+        usable_linear_estimate_ && filter_has_had_time_to_converge_;
>   }
> 
>-  // For noisy impulse responses, assume a fixed tail length.
>-  if (tail_power > 0.0005f) {
>-    reverb_decay_ = config_.ep_strength.default_len;
>+  if (!no_alignment_required_for_linear_mode_) {
>+    usable_linear_estimate_ = usable_linear_estimate_ && external_delay;
>+  }
>+
>+  if (!config_.echo_removal_control.linear_and_stable_echo_path) {
>+    usable_linear_estimate_ =
>+        usable_linear_estimate_ && recently_converged_filter;
>+    if (!allow_linear_mode_with_diverged_filter_) {
>+      usable_linear_estimate_ = usable_linear_estimate_ && !diverged_filter;
>+    }
>   }
>-  data_dumper_->DumpRaw("aec3_reverb_decay", reverb_decay_);
>-  data_dumper_->DumpRaw("aec3_tail_power", tail_power);
>+  if (transparent_mode_enforces_nonlinear_mode_) {
>+    usable_linear_estimate_ = usable_linear_estimate_ && !TransparentMode();
>+  }
>+
>+  use_linear_filter_output_ = usable_linear_estimate_ && !TransparentMode();
>+  diverged_linear_filter_ = diverged_filter;
>+
>+  const bool stationary_block =
>+      use_stationary_properties_ && echo_audibility_.IsBlockStationary();
>+
>+  reverb_model_estimator_.Update(
>+      filter_analyzer_.GetAdjustedFilter(), adaptive_filter_frequency_response,
>+      erle_estimator_.GetInstLinearQualityEstimate(), filter_delay_blocks_,
>+      usable_linear_estimate_, stationary_block);
>+
>+  erle_estimator_.Dump(data_dumper_);
>+  reverb_model_estimator_.Dump(data_dumper_.get());
>+  data_dumper_->DumpRaw("aec3_erl", Erl());
>+  data_dumper_->DumpRaw("aec3_erl_time_domain", ErlTimeDomain());
>+  data_dumper_->DumpRaw("aec3_usable_linear_estimate", UsableLinearEstimate());
>+  data_dumper_->DumpRaw("aec3_transparent_mode", transparent_mode_);
>+  data_dumper_->DumpRaw("aec3_state_internal_delay",
>+                        internal_delay_ ? *internal_delay_ : -1);
>+  data_dumper_->DumpRaw("aec3_filter_delay", filter_analyzer_.DelayBlocks());
>+
>+  data_dumper_->DumpRaw("aec3_consistent_filter",
>+                        filter_analyzer_.Consistent());
>+  data_dumper_->DumpRaw("aec3_suppression_gain_limit", SuppressionGainLimit());
>+  data_dumper_->DumpRaw("aec3_initial_state", InitialState());
>+  data_dumper_->DumpRaw("aec3_capture_saturation", SaturatedCapture());
>+  data_dumper_->DumpRaw("aec3_echo_saturation", echo_saturation_);
>+  data_dumper_->DumpRaw("aec3_converged_filter", converged_filter);
>+  data_dumper_->DumpRaw("aec3_diverged_filter", diverged_filter);
>+
>+  data_dumper_->DumpRaw("aec3_external_delay_avaliable",
>+                        external_delay ? 1 : 0);
>+  data_dumper_->DumpRaw("aec3_consistent_filter_estimate_not_seen",
>+                        consistent_filter_estimate_not_seen);
>+  data_dumper_->DumpRaw("aec3_filter_should_have_converged",
>+                        filter_should_have_converged_);
>+  data_dumper_->DumpRaw("aec3_filter_has_had_time_to_converge",
>+                        filter_has_had_time_to_converge_);
>+  data_dumper_->DumpRaw("aec3_recently_converged_filter",
>+                        recently_converged_filter);
>+  data_dumper_->DumpRaw("aec3_suppresion_gain_limiter_running",
>+                        IsSuppressionGainLimitActive());
>+  data_dumper_->DumpRaw("aec3_filter_tail_freq_resp_est",
>+                        GetReverbFrequencyResponse());
> }
> 
> bool AecState::DetectActiveRender(rtc::ArrayView<const float> x) const {
>@@ -264,69 +387,22 @@ bool AecState::DetectActiveRender(rtc::ArrayView<const float> x) const {
>                         kFftLengthBy2;
> }
> 
>-bool AecState::DetectEchoSaturation(rtc::ArrayView<const float> x) {
>+bool AecState::DetectEchoSaturation(rtc::ArrayView<const float> x,
>+                                    float echo_path_gain) {
>   RTC_DCHECK_LT(0, x.size());
>   const float max_sample = fabs(*std::max_element(
>       x.begin(), x.end(), [](float a, float b) { return a * a < b * b; }));
>-  previous_max_sample_ = max_sample;
> 
>   // Set flag for potential presence of saturated echo
>-  blocks_since_last_saturation_ =
>-      previous_max_sample_ > 200.f && SaturatedCapture()
>-          ? 0
>-          : blocks_since_last_saturation_ + 1;
>-
>-  return blocks_since_last_saturation_ < 20;
>-}
>-
>-void AecState::EchoAudibility::Update(rtc::ArrayView<const float> x,
>-                                      const std::array<float, kBlockSize>& s,
>-                                      bool converged_filter) {
>-  auto result_x = std::minmax_element(x.begin(), x.end());
>-  auto result_s = std::minmax_element(s.begin(), s.end());
>-  const float x_abs =
>-      std::max(fabsf(*result_x.first), fabsf(*result_x.second));
>-  const float s_abs =
>-      std::max(fabsf(*result_s.first), fabsf(*result_s.second));
>-
>-  if (converged_filter) {
>-    if (x_abs < 20.f) {
>-      ++low_farend_counter_;
>-    } else {
>-      low_farend_counter_ = 0;
>-    }
>+  const float kMargin = 10.f;
>+  float peak_echo_amplitude = max_sample * echo_path_gain * kMargin;
>+  if (SaturatedCapture() && peak_echo_amplitude > 32000) {
>+    blocks_since_last_saturation_ = 0;
>   } else {
>-    if (x_abs < 100.f) {
>-      ++low_farend_counter_;
>-    } else {
>-      low_farend_counter_ = 0;
>-    }
>+    ++blocks_since_last_saturation_;
>   }
> 
>-  // The echo is deemed as not audible if the echo estimate is on the level of
>-  // the quantization noise in the FFTs and the nearend level is sufficiently
>-  // strong to mask that by ensuring that the playout and AGC gains do not boost
>-  // any residual echo that is below the quantization noise level. Furthermore,
>-  // cases where the render signal is very close to zero are also identified as
>-  // not producing audible echo.
>-  inaudible_echo_ = (max_nearend_ > 500 && s_abs < 30.f) ||
>-                    (!converged_filter && x_abs < 500);
>-  inaudible_echo_ = inaudible_echo_ || low_farend_counter_ > 20;
>-}
>-
>-void AecState::EchoAudibility::UpdateWithOutput(rtc::ArrayView<const float> e) {
>-  const float e_max = *std::max_element(e.begin(), e.end());
>-  const float e_min = *std::min_element(e.begin(), e.end());
>-  const float e_abs = std::max(fabsf(e_max), fabsf(e_min));
>-
>-  if (max_nearend_ < e_abs) {
>-    max_nearend_ = e_abs;
>-    max_nearend_counter_ = 0;
>-  } else {
>-    if (++max_nearend_counter_ > 5 * kNumBlocksPerSecond) {
>-      max_nearend_ *= 0.995f;
>-    }
>-  }
>+  return blocks_since_last_saturation_ < 5;
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec_state.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec_state.h
>index e2039ad4dbf..99353b34a0d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec_state.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec_state.h
>@@ -11,18 +11,27 @@
> #ifndef MODULES_AUDIO_PROCESSING_AEC3_AEC_STATE_H_
> #define MODULES_AUDIO_PROCESSING_AEC3_AEC_STATE_H_
> 
>+#include <math.h>
>+
> #include <algorithm>
> #include <memory>
> #include <vector>
> 
>+#include "absl/types/optional.h"
> #include "api/array_view.h"
>-#include "api/optional.h"
>+#include "api/audio/echo_canceller3_config.h"
> #include "modules/audio_processing/aec3/aec3_common.h"
>+#include "modules/audio_processing/aec3/delay_estimate.h"
>+#include "modules/audio_processing/aec3/echo_audibility.h"
> #include "modules/audio_processing/aec3/echo_path_variability.h"
> #include "modules/audio_processing/aec3/erl_estimator.h"
> #include "modules/audio_processing/aec3/erle_estimator.h"
>+#include "modules/audio_processing/aec3/filter_analyzer.h"
> #include "modules/audio_processing/aec3/render_buffer.h"
>-#include "modules/audio_processing/include/audio_processing.h"
>+#include "modules/audio_processing/aec3/reverb_model_estimator.h"
>+#include "modules/audio_processing/aec3/subtractor_output.h"
>+#include "modules/audio_processing/aec3/subtractor_output_analyzer.h"
>+#include "modules/audio_processing/aec3/suppression_gain_limiter.h"
> #include "rtc_base/constructormagic.h"
> 
> namespace webrtc {
>@@ -39,19 +48,47 @@ class AecState {
>   // echo.
>   bool UsableLinearEstimate() const { return usable_linear_estimate_; }
> 
>-  // Returns whether there has been echo leakage detected.
>-  bool EchoLeakageDetected() const { return echo_leakage_detected_; }
>+  // Returns whether the echo subtractor output should be used as output.
>+  bool UseLinearFilterOutput() const { return use_linear_filter_output_; }
>+
>+  // Returns the estimated echo path gain.
>+  float EchoPathGain() const { return filter_analyzer_.Gain(); }
> 
>   // Returns whether the render signal is currently active.
>   bool ActiveRender() const { return blocks_with_active_render_ > 200; }
> 
>+  // Returns the appropriate scaling of the residual echo to match the
>+  // audibility.
>+  void GetResidualEchoScaling(rtc::ArrayView<float> residual_scaling) const {
>+    echo_audibility_.GetResidualEchoScaling(residual_scaling);
>+  }
>+
>+  // Returns whether the stationary properties of the signals are used in the
>+  // aec.
>+  bool UseStationaryProperties() const { return use_stationary_properties_; }
>+
>   // Returns the ERLE.
>   const std::array<float, kFftLengthBy2Plus1>& Erle() const {
>     return erle_estimator_.Erle();
>   }
> 
>-  // Returns the time-domain ERLE.
>-  float ErleTimeDomain() const { return erle_estimator_.ErleTimeDomain(); }
>+  // Returns any uncertainty in the ERLE estimate.
>+  absl::optional<float> ErleUncertainty() const {
>+    if (allow_linear_mode_with_diverged_filter_ && diverged_linear_filter_) {
>+      return 10.f;
>+    }
>+
>+    if (!filter_has_had_time_to_converge_ &&
>+        use_uncertainty_until_sufficiently_adapted_) {
>+      return 10.f;
>+    }
>+    return absl::nullopt;
>+  }
>+
>+  // Returns the time-domain ERLE in log2 units.
>+  float ErleTimeDomainLog2() const {
>+    return erle_estimator_.ErleTimeDomainLog2();
>+  }
> 
>   // Returns the ERL.
>   const std::array<float, kFftLengthBy2Plus1>& Erl() const {
>@@ -62,7 +99,10 @@ class AecState {
>   float ErlTimeDomain() const { return erl_estimator_.ErlTimeDomain(); }
> 
>   // Returns the delay estimate based on the linear filter.
>-  int FilterDelay() const { return filter_delay_; }
>+  int FilterDelayBlocks() const { return filter_delay_blocks_; }
>+
>+  // Returns the internal delay estimate based on the linear filter.
>+  absl::optional<int> InternalDelay() const { return internal_delay_; }
> 
>   // Returns whether the capture signal is saturated.
>   bool SaturatedCapture() const { return capture_signal_saturation_; }
>@@ -70,9 +110,6 @@ class AecState {
>   // Returns whether the echo signal is saturated.
>   bool SaturatedEcho() const { return echo_saturation_; }
> 
>-  // Returns whether the echo path can saturate.
>-  bool SaturatingEchoPath() const { return saturating_echo_path_; }
>-
>   // Updates the capture signal saturation.
>   void UpdateCaptureSaturation(bool capture_signal_saturation) {
>     capture_signal_saturation_ = capture_signal_saturation;
>@@ -85,17 +122,21 @@ class AecState {
>   void HandleEchoPathChange(const EchoPathVariability& echo_path_variability);
> 
>   // Returns the decay factor for the echo reverberation.
>-  float ReverbDecay() const { return reverb_decay_; }
>+  float ReverbDecay() const { return reverb_model_estimator_.ReverbDecay(); }
> 
>-  // Returns whether the echo suppression gain should be forced to zero.
>-  bool ForcedZeroGain() const { return force_zero_gain_; }
>+  // Return the frequency response of the reverberant echo.
>+  rtc::ArrayView<const float> GetReverbFrequencyResponse() const {
>+    return reverb_model_estimator_.GetReverbFrequencyResponse();
>+  }
> 
>-  // Returns whether the echo in the capture signal is audible.
>-  bool InaudibleEcho() const { return echo_audibility_.InaudibleEcho(); }
>+  // Returns the upper limit for the echo suppression gain.
>+  float SuppressionGainLimit() const {
>+    return suppression_gain_limiter_.Limit();
>+  }
> 
>-  // Updates the aec state with the AEC output signal.
>-  void UpdateWithOutput(rtc::ArrayView<const float> e) {
>-    echo_audibility_.UpdateWithOutput(e);
>+  // Returns whether the suppression gain limiter is active.
>+  bool IsSuppressionGainLimitActive() const {
>+    return suppression_gain_limiter_.IsActive();
>   }
> 
>   // Returns whether the linear filter should have been able to properly adapt.
>@@ -107,65 +148,78 @@ class AecState {
>   bool InitialState() const { return initial_state_; }
> 
>   // Updates the aec state.
>-  void Update(const std::vector<std::array<float, kFftLengthBy2Plus1>>&
>+  void Update(const absl::optional<DelayEstimate>& external_delay,
>+              const std::vector<std::array<float, kFftLengthBy2Plus1>>&
>                   adaptive_filter_frequency_response,
>               const std::vector<float>& adaptive_filter_impulse_response,
>-              bool converged_filter,
>               const RenderBuffer& render_buffer,
>               const std::array<float, kFftLengthBy2Plus1>& E2_main,
>               const std::array<float, kFftLengthBy2Plus1>& Y2,
>-              const std::array<float, kBlockSize>& s_main,
>-              bool echo_leakage_detected);
>+              const SubtractorOutput& subtractor_output,
>+              rtc::ArrayView<const float> y);
>+
>+  // Returns filter length in blocks.
>+  int FilterLengthBlocks() const {
>+    return filter_analyzer_.FilterLengthBlocks();
>+  }
> 
>  private:
>-  class EchoAudibility {
>-   public:
>-    void Update(rtc::ArrayView<const float> x,
>-                const std::array<float, kBlockSize>& s,
>-                bool converged_filter);
>-    void UpdateWithOutput(rtc::ArrayView<const float> e);
>-    bool InaudibleEcho() const { return inaudible_echo_; }
>-
>-   private:
>-    float max_nearend_ = 0.f;
>-    size_t max_nearend_counter_ = 0;
>-    size_t low_farend_counter_ = 0;
>-    bool inaudible_echo_ = false;
>-  };
>-
>-  void UpdateReverb(const std::vector<float>& impulse_response);
>   bool DetectActiveRender(rtc::ArrayView<const float> x) const;
>-  bool DetectEchoSaturation(rtc::ArrayView<const float> x);
>+  void UpdateSuppressorGainLimit(bool render_activity);
>+  bool DetectEchoSaturation(rtc::ArrayView<const float> x,
>+                            float echo_path_gain);
> 
>   static int instance_count_;
>   std::unique_ptr<ApmDataDumper> data_dumper_;
>+  const EchoCanceller3Config config_;
>+  const bool allow_transparent_mode_;
>+  const bool use_stationary_properties_;
>+  const bool enforce_delay_after_realignment_;
>+  const bool allow_linear_mode_with_diverged_filter_;
>+  const bool early_filter_usage_activated_;
>+  const bool use_short_initial_state_;
>+  const bool convergence_trigger_linear_mode_;
>+  const bool no_alignment_required_for_linear_mode_;
>+  const bool use_uncertainty_until_sufficiently_adapted_;
>+  const bool transparent_mode_enforces_nonlinear_mode_;
>   ErlEstimator erl_estimator_;
>   ErleEstimator erle_estimator_;
>   size_t capture_block_counter_ = 0;
>+  size_t blocks_since_reset_ = 0;
>   size_t blocks_with_proper_filter_adaptation_ = 0;
>   size_t blocks_with_active_render_ = 0;
>   bool usable_linear_estimate_ = false;
>-  bool echo_leakage_detected_ = false;
>+  bool diverged_linear_filter_ = false;
>   bool capture_signal_saturation_ = false;
>   bool echo_saturation_ = false;
>   bool transparent_mode_ = false;
>-  float previous_max_sample_ = 0.f;
>-  bool force_zero_gain_ = false;
>   bool render_received_ = false;
>-  size_t force_zero_gain_counter_ = 0;
>-  int filter_delay_ = 0;
>+  int filter_delay_blocks_ = 0;
>   size_t blocks_since_last_saturation_ = 1000;
>-  float reverb_decay_to_test_ = 0.9f;
>-  float reverb_decay_candidate_ = 0.f;
>-  float reverb_decay_candidate_residual_ = -1.f;
>-  EchoAudibility echo_audibility_;
>-  const EchoCanceller3Config config_;
>+
>   std::vector<float> max_render_;
>-  float reverb_decay_;
>-  bool saturating_echo_path_ = false;
>   bool filter_has_had_time_to_converge_ = false;
>   bool initial_state_ = true;
>-
>+  const float gain_rampup_increase_;
>+  SuppressionGainUpperLimiter suppression_gain_limiter_;
>+  FilterAnalyzer filter_analyzer_;
>+  bool use_linear_filter_output_ = false;
>+  absl::optional<int> internal_delay_;
>+  size_t diverged_blocks_ = 0;
>+  bool filter_should_have_converged_ = false;
>+  size_t blocks_since_converged_filter_;
>+  size_t active_blocks_since_consistent_filter_estimate_;
>+  bool converged_filter_seen_ = false;
>+  bool consistent_filter_seen_ = false;
>+  bool external_delay_seen_ = false;
>+  absl::optional<DelayEstimate> external_delay_;
>+  size_t frames_since_external_delay_change_ = 0;
>+  size_t converged_filter_count_ = 0;
>+  bool finite_erl_ = false;
>+  size_t active_blocks_since_converged_filter_ = 0;
>+  EchoAudibility echo_audibility_;
>+  ReverbModelEstimator reverb_model_estimator_;
>+  SubtractorOutputAnalyzer subtractor_output_analyzer_;
>   RTC_DISALLOW_COPY_AND_ASSIGN(AecState);
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec_state_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec_state_unittest.cc
>index 846a83c79d5..50b97f44370 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec_state_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/aec_state_unittest.cc
>@@ -22,6 +22,8 @@ TEST(AecState, NormalUsage) {
>   ApmDataDumper data_dumper(42);
>   EchoCanceller3Config config;
>   AecState state(config);
>+  absl::optional<DelayEstimate> delay_estimate =
>+      DelayEstimate(DelayEstimate::Quality::kRefined, 10);
>   std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
>       RenderDelayBuffer::Create(config, 3));
>   std::array<float, kFftLengthBy2Plus1> E2_main = {};
>@@ -29,9 +31,13 @@ TEST(AecState, NormalUsage) {
>   std::vector<std::vector<float>> x(3, std::vector<float>(kBlockSize, 0.f));
>   EchoPathVariability echo_path_variability(
>       false, EchoPathVariability::DelayAdjustment::kNone, false);
>-  std::array<float, kBlockSize> s;
>+  SubtractorOutput output;
>+  output.Reset();
>+  std::array<float, kBlockSize> y;
>   Aec3Fft fft;
>-  s.fill(100.f);
>+  output.s_main.fill(100.f);
>+  output.e_main.fill(100.f);
>+  y.fill(1000.f);
> 
>   std::vector<std::array<float, kFftLengthBy2Plus1>>
>       converged_filter_frequency_response(10);
>@@ -46,55 +52,47 @@ TEST(AecState, NormalUsage) {
>   std::vector<float> impulse_response(
>       GetTimeDomainLength(config.filter.main.length_blocks), 0.f);
> 
>-  // Verify that linear AEC usability is false when the filter is diverged.
>-  state.Update(diverged_filter_frequency_response, impulse_response, true,
>-               *render_delay_buffer->GetRenderBuffer(), E2_main, Y2, s, false);
>-  EXPECT_FALSE(state.UsableLinearEstimate());
>-
>   // Verify that linear AEC usability is true when the filter is converged
>   std::fill(x[0].begin(), x[0].end(), 101.f);
>   for (int k = 0; k < 3000; ++k) {
>     render_delay_buffer->Insert(x);
>-    state.Update(converged_filter_frequency_response, impulse_response, true,
>-                 *render_delay_buffer->GetRenderBuffer(), E2_main, Y2, s,
>-                 false);
>+    output.UpdatePowers(y);
>+    state.Update(delay_estimate, converged_filter_frequency_response,
>+                 impulse_response, *render_delay_buffer->GetRenderBuffer(),
>+                 E2_main, Y2, output, y);
>   }
>   EXPECT_TRUE(state.UsableLinearEstimate());
> 
>   // Verify that linear AEC usability becomes false after an echo path change is
>   // reported
>+  output.UpdatePowers(y);
>   state.HandleEchoPathChange(EchoPathVariability(
>-      true, EchoPathVariability::DelayAdjustment::kNone, false));
>-  state.Update(converged_filter_frequency_response, impulse_response, true,
>-               *render_delay_buffer->GetRenderBuffer(), E2_main, Y2, s, false);
>+      false, EchoPathVariability::DelayAdjustment::kBufferReadjustment, false));
>+  state.Update(delay_estimate, converged_filter_frequency_response,
>+               impulse_response, *render_delay_buffer->GetRenderBuffer(),
>+               E2_main, Y2, output, y);
>   EXPECT_FALSE(state.UsableLinearEstimate());
> 
>   // Verify that the active render detection works as intended.
>   std::fill(x[0].begin(), x[0].end(), 101.f);
>   render_delay_buffer->Insert(x);
>+  output.UpdatePowers(y);
>   state.HandleEchoPathChange(EchoPathVariability(
>       true, EchoPathVariability::DelayAdjustment::kNewDetectedDelay, false));
>-  state.Update(converged_filter_frequency_response, impulse_response, true,
>-               *render_delay_buffer->GetRenderBuffer(), E2_main, Y2, s, false);
>+  state.Update(delay_estimate, converged_filter_frequency_response,
>+               impulse_response, *render_delay_buffer->GetRenderBuffer(),
>+               E2_main, Y2, output, y);
>   EXPECT_FALSE(state.ActiveRender());
> 
>   for (int k = 0; k < 1000; ++k) {
>     render_delay_buffer->Insert(x);
>-    state.Update(converged_filter_frequency_response, impulse_response, true,
>-                 *render_delay_buffer->GetRenderBuffer(), E2_main, Y2, s,
>-                 false);
>+    output.UpdatePowers(y);
>+    state.Update(delay_estimate, converged_filter_frequency_response,
>+                 impulse_response, *render_delay_buffer->GetRenderBuffer(),
>+                 E2_main, Y2, output, y);
>   }
>   EXPECT_TRUE(state.ActiveRender());
> 
>-  // Verify that echo leakage is properly reported.
>-  state.Update(converged_filter_frequency_response, impulse_response, true,
>-               *render_delay_buffer->GetRenderBuffer(), E2_main, Y2, s, false);
>-  EXPECT_FALSE(state.EchoLeakageDetected());
>-
>-  state.Update(converged_filter_frequency_response, impulse_response, true,
>-               *render_delay_buffer->GetRenderBuffer(), E2_main, Y2, s, true);
>-  EXPECT_TRUE(state.EchoLeakageDetected());
>-
>   // Verify that the ERL is properly estimated
>   for (auto& x_k : x) {
>     x_k = std::vector<float>(kBlockSize, 0.f);
>@@ -112,9 +110,10 @@ TEST(AecState, NormalUsage) {
> 
>   Y2.fill(10.f * 10000.f * 10000.f);
>   for (size_t k = 0; k < 1000; ++k) {
>-    state.Update(converged_filter_frequency_response, impulse_response, true,
>-                 *render_delay_buffer->GetRenderBuffer(), E2_main, Y2, s,
>-                 false);
>+    output.UpdatePowers(y);
>+    state.Update(delay_estimate, converged_filter_frequency_response,
>+                 impulse_response, *render_delay_buffer->GetRenderBuffer(),
>+                 E2_main, Y2, output, y);
>   }
> 
>   ASSERT_TRUE(state.UsableLinearEstimate());
>@@ -129,9 +128,10 @@ TEST(AecState, NormalUsage) {
>   E2_main.fill(1.f * 10000.f * 10000.f);
>   Y2.fill(10.f * E2_main[0]);
>   for (size_t k = 0; k < 1000; ++k) {
>-    state.Update(converged_filter_frequency_response, impulse_response, true,
>-                 *render_delay_buffer->GetRenderBuffer(), E2_main, Y2, s,
>-                 false);
>+    output.UpdatePowers(y);
>+    state.Update(delay_estimate, converged_filter_frequency_response,
>+                 impulse_response, *render_delay_buffer->GetRenderBuffer(),
>+                 E2_main, Y2, output, y);
>   }
>   ASSERT_TRUE(state.UsableLinearEstimate());
>   {
>@@ -139,7 +139,7 @@ TEST(AecState, NormalUsage) {
>     EXPECT_EQ(erle[0], erle[1]);
>     constexpr size_t kLowFrequencyLimit = 32;
>     for (size_t k = 1; k < kLowFrequencyLimit; ++k) {
>-      EXPECT_NEAR(k % 2 == 0 ? 8.f : 1.f, erle[k], 0.1);
>+      EXPECT_NEAR(k % 2 == 0 ? 4.f : 1.f, erle[k], 0.1);
>     }
>     for (size_t k = kLowFrequencyLimit; k < erle.size() - 1; ++k) {
>       EXPECT_NEAR(k % 2 == 0 ? 1.5f : 1.f, erle[k], 0.1);
>@@ -150,9 +150,10 @@ TEST(AecState, NormalUsage) {
>   E2_main.fill(1.f * 10000.f * 10000.f);
>   Y2.fill(5.f * E2_main[0]);
>   for (size_t k = 0; k < 1000; ++k) {
>-    state.Update(converged_filter_frequency_response, impulse_response, true,
>-                 *render_delay_buffer->GetRenderBuffer(), E2_main, Y2, s,
>-                 false);
>+    output.UpdatePowers(y);
>+    state.Update(delay_estimate, converged_filter_frequency_response,
>+                 impulse_response, *render_delay_buffer->GetRenderBuffer(),
>+                 E2_main, Y2, output, y);
>   }
> 
>   ASSERT_TRUE(state.UsableLinearEstimate());
>@@ -161,7 +162,7 @@ TEST(AecState, NormalUsage) {
>     EXPECT_EQ(erle[0], erle[1]);
>     constexpr size_t kLowFrequencyLimit = 32;
>     for (size_t k = 1; k < kLowFrequencyLimit; ++k) {
>-      EXPECT_NEAR(k % 2 == 0 ? 5.f : 1.f, erle[k], 0.1);
>+      EXPECT_NEAR(k % 2 == 0 ? 4.f : 1.f, erle[k], 0.1);
>     }
>     for (size_t k = kLowFrequencyLimit; k < erle.size() - 1; ++k) {
>       EXPECT_NEAR(k % 2 == 0 ? 1.5f : 1.f, erle[k], 0.1);
>@@ -172,40 +173,43 @@ TEST(AecState, NormalUsage) {
> 
> // Verifies the delay for a converged filter is correctly identified.
> TEST(AecState, ConvergedFilterDelay) {
>-  constexpr int kFilterLength = 10;
>+  constexpr int kFilterLengthBlocks = 10;
>   EchoCanceller3Config config;
>   AecState state(config);
>   std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
>       RenderDelayBuffer::Create(config, 3));
>+  absl::optional<DelayEstimate> delay_estimate;
>   std::array<float, kFftLengthBy2Plus1> E2_main;
>   std::array<float, kFftLengthBy2Plus1> Y2;
>   std::array<float, kBlockSize> x;
>   EchoPathVariability echo_path_variability(
>       false, EchoPathVariability::DelayAdjustment::kNone, false);
>-  std::array<float, kBlockSize> s;
>-  s.fill(100.f);
>+  SubtractorOutput output;
>+  output.Reset();
>+  std::array<float, kBlockSize> y;
>+  output.s_main.fill(100.f);
>   x.fill(0.f);
>+  y.fill(0.f);
> 
>   std::vector<std::array<float, kFftLengthBy2Plus1>> frequency_response(
>-      kFilterLength);
>+      kFilterLengthBlocks);
>+  for (auto& v : frequency_response) {
>+    v.fill(0.01f);
>+  }
> 
>   std::vector<float> impulse_response(
>       GetTimeDomainLength(config.filter.main.length_blocks), 0.f);
> 
>   // Verify that the filter delay for a converged filter is properly identified.
>-  for (int k = 0; k < kFilterLength; ++k) {
>-    for (auto& v : frequency_response) {
>-      v.fill(0.01f);
>-    }
>-    frequency_response[k].fill(100.f);
>-    frequency_response[k][0] = 0.f;
>+  for (int k = 0; k < kFilterLengthBlocks; ++k) {
>+    std::fill(impulse_response.begin(), impulse_response.end(), 0.f);
>+    impulse_response[k * kBlockSize + 1] = 1.f;
>+
>     state.HandleEchoPathChange(echo_path_variability);
>-    state.Update(frequency_response, impulse_response, true,
>-                 *render_delay_buffer->GetRenderBuffer(), E2_main, Y2, s,
>-                 false);
>-    if (k != (kFilterLength - 1)) {
>-      EXPECT_EQ(k, state.FilterDelay());
>-    }
>+    output.UpdatePowers(y);
>+    state.Update(delay_estimate, frequency_response, impulse_response,
>+                 *render_delay_buffer->GetRenderBuffer(), E2_main, Y2, output,
>+                 y);
>   }
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/block_processor.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/block_processor.cc
>index dafe384bedc..1bb69584cc3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/block_processor.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/block_processor.cc
>@@ -9,7 +9,7 @@
>  */
> #include "modules/audio_processing/aec3/block_processor.h"
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> #include "modules/audio_processing/aec3/aec3_common.h"
> #include "modules/audio_processing/aec3/block_processor_metrics.h"
> #include "modules/audio_processing/aec3/echo_path_variability.h"
>@@ -43,6 +43,8 @@ class BlockProcessorImpl final : public BlockProcessor {
> 
>   void GetMetrics(EchoControl::Metrics* metrics) const override;
> 
>+  void SetAudioBufferDelay(size_t delay_ms) override;
>+
>  private:
>   static int instance_count_;
>   std::unique_ptr<ApmDataDumper> data_dumper_;
>@@ -56,7 +58,8 @@ class BlockProcessorImpl final : public BlockProcessor {
>   BlockProcessorMetrics metrics_;
>   RenderDelayBuffer::BufferingEvent render_event_;
>   size_t capture_call_counter_ = 0;
>-  rtc::Optional<DelayEstimate> estimated_delay_;
>+  absl::optional<DelayEstimate> estimated_delay_;
>+  absl::optional<int> echo_remover_delay_;
>   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(BlockProcessorImpl);
> };
> 
>@@ -158,7 +161,8 @@ void BlockProcessorImpl::ProcessCapture(
>   // Compute and and apply the render delay required to achieve proper signal
>   // alignment.
>   estimated_delay_ = delay_controller_->GetDelay(
>-      render_buffer_->GetDownsampledRenderBuffer(), (*capture_block)[0]);
>+      render_buffer_->GetDownsampledRenderBuffer(), render_buffer_->Delay(),
>+      echo_remover_delay_, (*capture_block)[0]);
> 
>   if (estimated_delay_) {
>     if (render_buffer_->CausalDelay(estimated_delay_->delay)) {
>@@ -188,9 +192,13 @@ void BlockProcessorImpl::ProcessCapture(
> 
>   // Remove the echo from the capture signal.
>   echo_remover_->ProcessCapture(
>-      echo_path_variability, capture_signal_saturation,
>+      echo_path_variability, capture_signal_saturation, estimated_delay_,
>       render_buffer_->GetRenderBuffer(), capture_block);
> 
>+  // Check to see if a refined delay estimate has been obtained from the echo
>+  // remover.
>+  echo_remover_delay_ = echo_remover_->Delay();
>+
>   // Update the metrics.
>   metrics_.UpdateCapture(false);
> 
>@@ -224,10 +232,14 @@ void BlockProcessorImpl::UpdateEchoLeakageStatus(bool leakage_detected) {
> void BlockProcessorImpl::GetMetrics(EchoControl::Metrics* metrics) const {
>   echo_remover_->GetMetrics(metrics);
>   const int block_size_ms = sample_rate_hz_ == 8000 ? 8 : 4;
>-  rtc::Optional<size_t> delay = render_buffer_->Delay();
>+  absl::optional<size_t> delay = render_buffer_->Delay();
>   metrics->delay_ms = delay ? static_cast<int>(*delay) * block_size_ms : 0;
> }
> 
>+void BlockProcessorImpl::SetAudioBufferDelay(size_t delay_ms) {
>+  render_buffer_->SetAudioBufferDelay(delay_ms);
>+}
>+
> }  // namespace
> 
> BlockProcessor* BlockProcessor::Create(const EchoCanceller3Config& config,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/block_processor.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/block_processor.h
>index 8687bc2282d..a3967ea7e4a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/block_processor.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/block_processor.h
>@@ -42,6 +42,9 @@ class BlockProcessor {
>   // Get current metrics.
>   virtual void GetMetrics(EchoControl::Metrics* metrics) const = 0;
> 
>+  // Provides an optional external estimate of the audio buffer delay.
>+  virtual void SetAudioBufferDelay(size_t delay_ms) = 0;
>+
>   // Processes a block of capture data.
>   virtual void ProcessCapture(
>       bool echo_path_gain_change,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/block_processor_metrics_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/block_processor_metrics_unittest.cc
>index 7ce8573efa6..73f76895a66 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/block_processor_metrics_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/block_processor_metrics_unittest.cc
>@@ -8,8 +8,8 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "modules/audio_processing/aec3/aec3_common.h"
> #include "modules/audio_processing/aec3/block_processor_metrics.h"
>+#include "modules/audio_processing/aec3/aec3_common.h"
> 
> #include "test/gtest.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/block_processor_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/block_processor_unittest.cc
>index 29b249b3b6c..59060181be3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/block_processor_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/block_processor_unittest.cc
>@@ -35,15 +35,17 @@ using testing::_;
> 
> // Verifies that the basic BlockProcessor functionality works and that the API
> // methods are callable.
>-void RunBasicSetupAndApiCallTest(int sample_rate_hz) {
>+void RunBasicSetupAndApiCallTest(int sample_rate_hz, int num_iterations) {
>   std::unique_ptr<BlockProcessor> block_processor(
>       BlockProcessor::Create(EchoCanceller3Config(), sample_rate_hz));
>   std::vector<std::vector<float>> block(NumBandsForRate(sample_rate_hz),
>-                                        std::vector<float>(kBlockSize, 0.f));
>+                                        std::vector<float>(kBlockSize, 1000.f));
> 
>-  block_processor->BufferRender(block);
>-  block_processor->ProcessCapture(false, false, &block);
>-  block_processor->UpdateEchoLeakageStatus(false);
>+  for (int k = 0; k < num_iterations; ++k) {
>+    block_processor->BufferRender(block);
>+    block_processor->ProcessCapture(false, false, &block);
>+    block_processor->UpdateEchoLeakageStatus(false);
>+  }
> }
> 
> #if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
>@@ -164,9 +166,9 @@ TEST(BlockProcessor, DISABLED_SubmoduleIntegration) {
>     EXPECT_CALL(*render_delay_buffer_mock, Delay())
>         .Times(kNumBlocks)
>         .WillRepeatedly(Return(0));
>-    EXPECT_CALL(*render_delay_controller_mock, GetDelay(_, _))
>+    EXPECT_CALL(*render_delay_controller_mock, GetDelay(_, _, _, _))
>         .Times(kNumBlocks);
>-    EXPECT_CALL(*echo_remover_mock, ProcessCapture(_, _, _, _))
>+    EXPECT_CALL(*echo_remover_mock, ProcessCapture(_, _, _, _, _))
>         .Times(kNumBlocks);
>     EXPECT_CALL(*echo_remover_mock, UpdateEchoLeakageStatus(_))
>         .Times(kNumBlocks);
>@@ -193,10 +195,14 @@ TEST(BlockProcessor, DISABLED_SubmoduleIntegration) {
> TEST(BlockProcessor, BasicSetupAndApiCalls) {
>   for (auto rate : {8000, 16000, 32000, 48000}) {
>     SCOPED_TRACE(ProduceDebugText(rate));
>-    RunBasicSetupAndApiCallTest(rate);
>+    RunBasicSetupAndApiCallTest(rate, 1);
>   }
> }
> 
>+TEST(BlockProcessor, TestLongerCall) {
>+  RunBasicSetupAndApiCallTest(16000, 20 * kNumBlocksPerSecond);
>+}
>+
> #if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
> // TODO(gustaf): Re-enable the test once the issue with memory leaks during
> // DEATH tests on test bots has been fixed.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/cascaded_biquad_filter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/cascaded_biquad_filter.cc
>index 9a472f516fc..333d2267a41 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/cascaded_biquad_filter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/cascaded_biquad_filter.cc
>@@ -13,37 +13,84 @@
> 
> namespace webrtc {
> 
>+CascadedBiQuadFilter::BiQuadParam::BiQuadParam(std::complex<float> zero,
>+                                               std::complex<float> pole,
>+                                               float gain,
>+                                               bool mirror_zero_along_i_axis)
>+    : zero(zero),
>+      pole(pole),
>+      gain(gain),
>+      mirror_zero_along_i_axis(mirror_zero_along_i_axis) {}
>+
>+CascadedBiQuadFilter::BiQuadParam::BiQuadParam(const BiQuadParam&) = default;
>+
>+CascadedBiQuadFilter::BiQuad::BiQuad(
>+    const CascadedBiQuadFilter::BiQuadParam& param)
>+    : x(), y() {
>+  float z_r = std::real(param.zero);
>+  float z_i = std::imag(param.zero);
>+  float p_r = std::real(param.pole);
>+  float p_i = std::imag(param.pole);
>+  float gain = param.gain;
>+
>+  if (param.mirror_zero_along_i_axis) {
>+    // Assuming zeroes at z_r and -z_r.
>+    RTC_DCHECK(z_i == 0.f);
>+    coefficients.b[0] = gain * 1.f;
>+    coefficients.b[1] = 0.f;
>+    coefficients.b[2] = gain * -(z_r * z_r);
>+  } else {
>+    // Assuming zeros at (z_r + z_i*i) and (z_r - z_i*i).
>+    coefficients.b[0] = gain * 1.f;
>+    coefficients.b[1] = gain * -2.f * z_r;
>+    coefficients.b[2] = gain * (z_r * z_r + z_i * z_i);
>+  }
>+
>+  // Assuming poles at (p_r + p_i*i) and (p_r - p_i*i).
>+  coefficients.a[0] = -2.f * p_r;
>+  coefficients.a[1] = p_r * p_r + p_i * p_i;
>+}
>+
> CascadedBiQuadFilter::CascadedBiQuadFilter(
>     const CascadedBiQuadFilter::BiQuadCoefficients& coefficients,
>     size_t num_biquads)
>-    : biquad_states_(num_biquads), coefficients_(coefficients) {}
>+    : biquads_(num_biquads, coefficients) {}
>+
>+CascadedBiQuadFilter::CascadedBiQuadFilter(
>+    const std::vector<CascadedBiQuadFilter::BiQuadParam>& biquad_params) {
>+  for (const auto& param : biquad_params) {
>+    biquads_.push_back(BiQuad(param));
>+  }
>+}
> 
> CascadedBiQuadFilter::~CascadedBiQuadFilter() = default;
> 
> void CascadedBiQuadFilter::Process(rtc::ArrayView<const float> x,
>                                    rtc::ArrayView<float> y) {
>-  ApplyBiQuad(x, y, &biquad_states_[0]);
>-  for (size_t k = 1; k < biquad_states_.size(); ++k) {
>-    ApplyBiQuad(y, y, &biquad_states_[k]);
>+  if (biquads_.size() > 0) {
>+    ApplyBiQuad(x, y, &biquads_[0]);
>+    for (size_t k = 1; k < biquads_.size(); ++k) {
>+      ApplyBiQuad(y, y, &biquads_[k]);
>+    }
>+  } else {
>+    std::copy(x.begin(), x.end(), y.begin());
>   }
> }
> 
> void CascadedBiQuadFilter::Process(rtc::ArrayView<float> y) {
>-  for (auto& biquad : biquad_states_) {
>+  for (auto& biquad : biquads_) {
>     ApplyBiQuad(y, y, &biquad);
>   }
> }
> 
>-void CascadedBiQuadFilter::ApplyBiQuad(
>-    rtc::ArrayView<const float> x,
>-    rtc::ArrayView<float> y,
>-    CascadedBiQuadFilter::BiQuadState* biquad_state) {
>+void CascadedBiQuadFilter::ApplyBiQuad(rtc::ArrayView<const float> x,
>+                                       rtc::ArrayView<float> y,
>+                                       CascadedBiQuadFilter::BiQuad* biquad) {
>   RTC_DCHECK_EQ(x.size(), y.size());
>-  RTC_DCHECK(biquad_state);
>-  const auto* c_b = coefficients_.b;
>-  const auto* c_a = coefficients_.a;
>-  auto* m_x = biquad_state->x;
>-  auto* m_y = biquad_state->y;
>+  const auto* c_b = biquad->coefficients.b;
>+  const auto* c_a = biquad->coefficients.a;
>+  auto* m_x = biquad->x;
>+  auto* m_y = biquad->y;
>   for (size_t k = 0; k < x.size(); ++k) {
>     const float tmp = x[k];
>     y[k] = c_b[0] * tmp + c_b[1] * m_x[0] + c_b[2] * m_x[1] - c_a[0] * m_y[0] -
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/cascaded_biquad_filter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/cascaded_biquad_filter.h
>index aea889ab537..1e09fa6e5a4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/cascaded_biquad_filter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/cascaded_biquad_filter.h
>@@ -11,6 +11,7 @@
> #ifndef MODULES_AUDIO_PROCESSING_AEC3_CASCADED_BIQUAD_FILTER_H_
> #define MODULES_AUDIO_PROCESSING_AEC3_CASCADED_BIQUAD_FILTER_H_
> 
>+#include <complex>
> #include <vector>
> 
> #include "api/array_view.h"
>@@ -18,14 +19,20 @@
> 
> namespace webrtc {
> 
>-// Applies a number of identical biquads in a cascaded manner. The filter
>-// implementation is direct form 1.
>+// Applies a number of biquads in a cascaded manner. The filter implementation
>+// is direct form 1.
> class CascadedBiQuadFilter {
>  public:
>-  struct BiQuadState {
>-    BiQuadState() : x(), y() {}
>-    float x[2];
>-    float y[2];
>+  struct BiQuadParam {
>+    BiQuadParam(std::complex<float> zero,
>+                std::complex<float> pole,
>+                float gain,
>+                bool mirror_zero_along_i_axis = false);
>+    BiQuadParam(const BiQuadParam&);
>+    std::complex<float> zero;
>+    std::complex<float> pole;
>+    float gain;
>+    bool mirror_zero_along_i_axis;
>   };
> 
>   struct BiQuadCoefficients {
>@@ -33,9 +40,20 @@ class CascadedBiQuadFilter {
>     float a[2];
>   };
> 
>+  struct BiQuad {
>+    BiQuad(const BiQuadCoefficients& coefficients)
>+        : coefficients(coefficients), x(), y() {}
>+    BiQuad(const CascadedBiQuadFilter::BiQuadParam& param);
>+    BiQuadCoefficients coefficients;
>+    float x[2];
>+    float y[2];
>+  };
>+
>   CascadedBiQuadFilter(
>       const CascadedBiQuadFilter::BiQuadCoefficients& coefficients,
>       size_t num_biquads);
>+  CascadedBiQuadFilter(
>+      const std::vector<CascadedBiQuadFilter::BiQuadParam>& biquad_params);
>   ~CascadedBiQuadFilter();
>   // Applies the biquads on the values in x in order to form the output in y.
>   void Process(rtc::ArrayView<const float> x, rtc::ArrayView<float> y);
>@@ -45,10 +63,9 @@ class CascadedBiQuadFilter {
>  private:
>   void ApplyBiQuad(rtc::ArrayView<const float> x,
>                    rtc::ArrayView<float> y,
>-                   CascadedBiQuadFilter::BiQuadState* biquad_state);
>+                   CascadedBiQuadFilter::BiQuad* biquad);
> 
>-  std::vector<BiQuadState> biquad_states_;
>-  const BiQuadCoefficients coefficients_;
>+  std::vector<BiQuad> biquads_;
> 
>   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(CascadedBiQuadFilter);
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/cascaded_biquad_filter_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/cascaded_biquad_filter_unittest.cc
>index fcb77e1f6e4..57f4b04f5f6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/cascaded_biquad_filter_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/cascaded_biquad_filter_unittest.cc
>@@ -95,4 +95,46 @@ TEST(CascadedBiquadFilter, InputSizeCheckVerification) {
> }
> #endif
> 
>+// Verifies the conversion from zero, pole, gain to filter coefficients for
>+// lowpass filter.
>+TEST(CascadedBiquadFilter, BiQuadParamLowPass) {
>+  CascadedBiQuadFilter::BiQuadParam param(
>+      {-1.0f, 0.0f}, {0.23146901f, 0.39514232f}, 0.1866943331163784f);
>+  CascadedBiQuadFilter::BiQuad filter(param);
>+  const float epsilon = 1e-6f;
>+  EXPECT_NEAR(filter.coefficients.b[0], 0.18669433f, epsilon);
>+  EXPECT_NEAR(filter.coefficients.b[1], 0.37338867f, epsilon);
>+  EXPECT_NEAR(filter.coefficients.b[2], 0.18669433f, epsilon);
>+  EXPECT_NEAR(filter.coefficients.a[0], -0.46293803f, epsilon);
>+  EXPECT_NEAR(filter.coefficients.a[1], 0.20971536f, epsilon);
>+}
>+
>+// Verifies the conversion from zero, pole, gain to filter coefficients for
>+// highpass filter.
>+TEST(CascadedBiquadFilter, BiQuadParamHighPass) {
>+  CascadedBiQuadFilter::BiQuadParam param(
>+      {1.0f, 0.0f}, {0.72712179f, 0.21296904f}, 0.75707637533388494f);
>+  CascadedBiQuadFilter::BiQuad filter(param);
>+  const float epsilon = 1e-6f;
>+  EXPECT_NEAR(filter.coefficients.b[0], 0.75707638f, epsilon);
>+  EXPECT_NEAR(filter.coefficients.b[1], -1.51415275f, epsilon);
>+  EXPECT_NEAR(filter.coefficients.b[2], 0.75707638f, epsilon);
>+  EXPECT_NEAR(filter.coefficients.a[0], -1.45424359f, epsilon);
>+  EXPECT_NEAR(filter.coefficients.a[1], 0.57406192f, epsilon);
>+}
>+
>+// Verifies the conversion from zero, pole, gain to filter coefficients for
>+// bandpass filter.
>+TEST(CascadedBiquadFilter, BiQuadParamBandPass) {
>+  CascadedBiQuadFilter::BiQuadParam param(
>+      {1.0f, 0.0f}, {1.11022302e-16f, 0.71381051f}, 0.2452372752527856f, true);
>+  CascadedBiQuadFilter::BiQuad filter(param);
>+  const float epsilon = 1e-6f;
>+  EXPECT_NEAR(filter.coefficients.b[0], 0.24523728f, epsilon);
>+  EXPECT_NEAR(filter.coefficients.b[1], 0.f, epsilon);
>+  EXPECT_NEAR(filter.coefficients.b[2], -0.24523728f, epsilon);
>+  EXPECT_NEAR(filter.coefficients.a[0], -2.22044605e-16f, epsilon);
>+  EXPECT_NEAR(filter.coefficients.a[1], 5.09525449e-01f, epsilon);
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/coherence_gain.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/coherence_gain.cc
>new file mode 100644
>index 00000000000..ad33382b3ae
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/coherence_gain.cc
>@@ -0,0 +1,257 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/aec3/coherence_gain.h"
>+
>+#include <math.h>
>+
>+#include <algorithm>
>+
>+#include "rtc_base/checks.h"
>+
>+namespace webrtc {
>+
>+namespace {
>+
>+// Matlab code to produce table:
>+// overDriveCurve = [sqrt(linspace(0,1,65))' + 1];
>+// fprintf(1, '\t%.4f, %.4f, %.4f, %.4f, %.4f, %.4f,\n', overDriveCurve);
>+const float kOverDriveCurve[kFftLengthBy2Plus1] = {
>+    1.0000f, 1.1250f, 1.1768f, 1.2165f, 1.2500f, 1.2795f, 1.3062f, 1.3307f,
>+    1.3536f, 1.3750f, 1.3953f, 1.4146f, 1.4330f, 1.4507f, 1.4677f, 1.4841f,
>+    1.5000f, 1.5154f, 1.5303f, 1.5449f, 1.5590f, 1.5728f, 1.5863f, 1.5995f,
>+    1.6124f, 1.6250f, 1.6374f, 1.6495f, 1.6614f, 1.6731f, 1.6847f, 1.6960f,
>+    1.7071f, 1.7181f, 1.7289f, 1.7395f, 1.7500f, 1.7603f, 1.7706f, 1.7806f,
>+    1.7906f, 1.8004f, 1.8101f, 1.8197f, 1.8292f, 1.8385f, 1.8478f, 1.8570f,
>+    1.8660f, 1.8750f, 1.8839f, 1.8927f, 1.9014f, 1.9100f, 1.9186f, 1.9270f,
>+    1.9354f, 1.9437f, 1.9520f, 1.9601f, 1.9682f, 1.9763f, 1.9843f, 1.9922f,
>+    2.0000f};
>+
>+// Matlab code to produce table:
>+// weightCurve = [0 ; 0.3 * sqrt(linspace(0,1,64))' + 0.1];
>+// fprintf(1, '\t%.4f, %.4f, %.4f, %.4f, %.4f, %.4f,\n', weightCurve);
>+const float kWeightCurve[kFftLengthBy2Plus1] = {
>+    0.0000f, 0.1000f, 0.1378f, 0.1535f, 0.1655f, 0.1756f, 0.1845f, 0.1926f,
>+    0.2000f, 0.2069f, 0.2134f, 0.2195f, 0.2254f, 0.2309f, 0.2363f, 0.2414f,
>+    0.2464f, 0.2512f, 0.2558f, 0.2604f, 0.2648f, 0.2690f, 0.2732f, 0.2773f,
>+    0.2813f, 0.2852f, 0.2890f, 0.2927f, 0.2964f, 0.3000f, 0.3035f, 0.3070f,
>+    0.3104f, 0.3138f, 0.3171f, 0.3204f, 0.3236f, 0.3268f, 0.3299f, 0.3330f,
>+    0.3360f, 0.3390f, 0.3420f, 0.3449f, 0.3478f, 0.3507f, 0.3535f, 0.3563f,
>+    0.3591f, 0.3619f, 0.3646f, 0.3673f, 0.3699f, 0.3726f, 0.3752f, 0.3777f,
>+    0.3803f, 0.3828f, 0.3854f, 0.3878f, 0.3903f, 0.3928f, 0.3952f, 0.3976f,
>+    0.4000f};
>+
>+int CmpFloat(const void* a, const void* b) {
>+  const float* da = static_cast<const float*>(a);
>+  const float* db = static_cast<const float*>(b);
>+  return (*da > *db) - (*da < *db);
>+}
>+
>+}  // namespace
>+
>+CoherenceGain::CoherenceGain(int sample_rate_hz, size_t num_bands_to_compute)
>+    : num_bands_to_compute_(num_bands_to_compute),
>+      sample_rate_scaler_(sample_rate_hz >= 16000 ? 2 : 1) {
>+  spectra_.Cye.Clear();
>+  spectra_.Cxy.Clear();
>+  spectra_.Pe.fill(0.f);
>+  // Initialize to 1 in order to prevent numerical instability in the first
>+  // block.
>+  spectra_.Py.fill(1.f);
>+  spectra_.Px.fill(1.f);
>+}
>+
>+CoherenceGain::~CoherenceGain() = default;
>+
>+void CoherenceGain::ComputeGain(const FftData& E,
>+                                const FftData& X,
>+                                const FftData& Y,
>+                                rtc::ArrayView<float> gain) {
>+  std::array<float, kFftLengthBy2Plus1> coherence_ye;
>+  std::array<float, kFftLengthBy2Plus1> coherence_xy;
>+
>+  UpdateCoherenceSpectra(E, X, Y);
>+  ComputeCoherence(coherence_ye, coherence_xy);
>+  FormSuppressionGain(coherence_ye, coherence_xy, gain);
>+}
>+
>+// Updates the following smoothed Power Spectral Densities (PSD):
>+//  - sd  : near-end
>+//  - se  : residual echo
>+//  - sx  : far-end
>+//  - sde : cross-PSD of near-end and residual echo
>+//  - sxd : cross-PSD of near-end and far-end
>+//
>+void CoherenceGain::UpdateCoherenceSpectra(const FftData& E,
>+                                           const FftData& X,
>+                                           const FftData& Y) {
>+  const float s = sample_rate_scaler_ == 1 ? 0.9f : 0.92f;
>+  const float one_minus_s = 1.f - s;
>+  auto& c = spectra_;
>+
>+  for (size_t i = 0; i < c.Py.size(); i++) {
>+    c.Py[i] =
>+        s * c.Py[i] + one_minus_s * (Y.re[i] * Y.re[i] + Y.im[i] * Y.im[i]);
>+    c.Pe[i] =
>+        s * c.Pe[i] + one_minus_s * (E.re[i] * E.re[i] + E.im[i] * E.im[i]);
>+    // We threshold here to protect against the ill-effects of a zero farend.
>+    // The threshold is not arbitrarily chosen, but balances protection and
>+    // adverse interaction with the algorithm's tuning.
>+
>+    // Threshold to protect against the ill-effects of a zero far-end.
>+    c.Px[i] =
>+        s * c.Px[i] +
>+        one_minus_s * std::max(X.re[i] * X.re[i] + X.im[i] * X.im[i], 15.f);
>+
>+    c.Cye.re[i] =
>+        s * c.Cye.re[i] + one_minus_s * (Y.re[i] * E.re[i] + Y.im[i] * E.im[i]);
>+    c.Cye.im[i] =
>+        s * c.Cye.im[i] + one_minus_s * (Y.re[i] * E.im[i] - Y.im[i] * E.re[i]);
>+
>+    c.Cxy.re[i] =
>+        s * c.Cxy.re[i] + one_minus_s * (Y.re[i] * X.re[i] + Y.im[i] * X.im[i]);
>+    c.Cxy.im[i] =
>+        s * c.Cxy.im[i] + one_minus_s * (Y.re[i] * X.im[i] - Y.im[i] * X.re[i]);
>+  }
>+}
>+
>+void CoherenceGain::FormSuppressionGain(
>+    rtc::ArrayView<const float> coherence_ye,
>+    rtc::ArrayView<const float> coherence_xy,
>+    rtc::ArrayView<float> gain) {
>+  RTC_DCHECK_EQ(kFftLengthBy2Plus1, coherence_ye.size());
>+  RTC_DCHECK_EQ(kFftLengthBy2Plus1, coherence_xy.size());
>+  RTC_DCHECK_EQ(kFftLengthBy2Plus1, gain.size());
>+  constexpr int kPrefBandSize = 24;
>+  auto& gs = gain_state_;
>+  std::array<float, kPrefBandSize> h_nl_pref;
>+  float h_nl_fb = 0;
>+  float h_nl_fb_low = 0;
>+  const int pref_band_size = kPrefBandSize / sample_rate_scaler_;
>+  const int min_pref_band = 4 / sample_rate_scaler_;
>+
>+  float h_nl_de_avg = 0.f;
>+  float h_nl_xd_avg = 0.f;
>+  for (int i = min_pref_band; i < pref_band_size + min_pref_band; ++i) {
>+    h_nl_xd_avg += coherence_xy[i];
>+    h_nl_de_avg += coherence_ye[i];
>+  }
>+  h_nl_xd_avg /= pref_band_size;
>+  h_nl_xd_avg = 1 - h_nl_xd_avg;
>+  h_nl_de_avg /= pref_band_size;
>+
>+  if (h_nl_xd_avg < 0.75f && h_nl_xd_avg < gs.h_nl_xd_avg_min) {
>+    gs.h_nl_xd_avg_min = h_nl_xd_avg;
>+  }
>+
>+  if (h_nl_de_avg > 0.98f && h_nl_xd_avg > 0.9f) {
>+    gs.near_state = true;
>+  } else if (h_nl_de_avg < 0.95f || h_nl_xd_avg < 0.8f) {
>+    gs.near_state = false;
>+  }
>+
>+  std::array<float, kFftLengthBy2Plus1> h_nl;
>+  if (gs.h_nl_xd_avg_min == 1) {
>+    gs.overdrive = 15.f;
>+
>+    if (gs.near_state) {
>+      std::copy(coherence_ye.begin(), coherence_ye.end(), h_nl.begin());
>+      h_nl_fb = h_nl_de_avg;
>+      h_nl_fb_low = h_nl_de_avg;
>+    } else {
>+      for (size_t i = 0; i < h_nl.size(); ++i) {
>+        h_nl[i] = 1 - coherence_xy[i];
>+        h_nl[i] = std::max(h_nl[i], 0.f);
>+      }
>+      h_nl_fb = h_nl_xd_avg;
>+      h_nl_fb_low = h_nl_xd_avg;
>+    }
>+  } else {
>+    if (gs.near_state) {
>+      std::copy(coherence_ye.begin(), coherence_ye.end(), h_nl.begin());
>+      h_nl_fb = h_nl_de_avg;
>+      h_nl_fb_low = h_nl_de_avg;
>+    } else {
>+      for (size_t i = 0; i < h_nl.size(); ++i) {
>+        h_nl[i] = std::min(coherence_ye[i], 1 - coherence_xy[i]);
>+        h_nl[i] = std::max(h_nl[i], 0.f);
>+      }
>+
>+      // Select an order statistic from the preferred bands.
>+      // TODO(peah): Using quicksort now, but a selection algorithm may be
>+      // preferred.
>+      std::copy(h_nl.begin() + min_pref_band,
>+                h_nl.begin() + min_pref_band + pref_band_size,
>+                h_nl_pref.begin());
>+      std::qsort(h_nl_pref.data(), pref_band_size, sizeof(float), CmpFloat);
>+
>+      constexpr float kPrefBandQuant = 0.75f;
>+      h_nl_fb = h_nl_pref[static_cast<int>(
>+          floor(kPrefBandQuant * (pref_band_size - 1)))];
>+      constexpr float kPrefBandQuantLow = 0.5f;
>+      h_nl_fb_low = h_nl_pref[static_cast<int>(
>+          floor(kPrefBandQuantLow * (pref_band_size - 1)))];
>+    }
>+  }
>+
>+  // Track the local filter minimum to determine suppression overdrive.
>+  if (h_nl_fb_low < 0.6f && h_nl_fb_low < gs.h_nl_fb_local_min) {
>+    gs.h_nl_fb_local_min = h_nl_fb_low;
>+    gs.h_nl_fb_min = h_nl_fb_low;
>+    gs.h_nl_new_min = 1;
>+    gs.h_nl_min_ctr = 0;
>+  }
>+  gs.h_nl_fb_local_min =
>+      std::min(gs.h_nl_fb_local_min + 0.0008f / sample_rate_scaler_, 1.f);
>+  gs.h_nl_xd_avg_min =
>+      std::min(gs.h_nl_xd_avg_min + 0.0006f / sample_rate_scaler_, 1.f);
>+
>+  if (gs.h_nl_new_min == 1) {
>+    ++gs.h_nl_min_ctr;
>+  }
>+  if (gs.h_nl_min_ctr == 2) {
>+    gs.h_nl_new_min = 0;
>+    gs.h_nl_min_ctr = 0;
>+    constexpr float epsilon = 1e-10f;
>+    gs.overdrive = std::max(
>+        -18.4f / static_cast<float>(log(gs.h_nl_fb_min + epsilon) + epsilon),
>+        15.f);
>+  }
>+
>+  // Smooth the overdrive.
>+  if (gs.overdrive < gs.overdrive_scaling) {
>+    gs.overdrive_scaling = 0.99f * gs.overdrive_scaling + 0.01f * gs.overdrive;
>+  } else {
>+    gs.overdrive_scaling = 0.9f * gs.overdrive_scaling + 0.1f * gs.overdrive;
>+  }
>+
>+  // Apply the overdrive.
>+  RTC_DCHECK_LE(num_bands_to_compute_, gain.size());
>+  for (size_t i = 0; i < num_bands_to_compute_; ++i) {
>+    if (h_nl[i] > h_nl_fb) {
>+      h_nl[i] = kWeightCurve[i] * h_nl_fb + (1 - kWeightCurve[i]) * h_nl[i];
>+    }
>+    gain[i] = powf(h_nl[i], gs.overdrive_scaling * kOverDriveCurve[i]);
>+  }
>+}
>+
>+void CoherenceGain::ComputeCoherence(rtc::ArrayView<float> coherence_ye,
>+                                     rtc::ArrayView<float> coherence_xy) const {
>+  const auto& c = spectra_;
>+  constexpr float epsilon = 1e-10f;
>+  for (size_t i = 0; i < coherence_ye.size(); ++i) {
>+    coherence_ye[i] = (c.Cye.re[i] * c.Cye.re[i] + c.Cye.im[i] * c.Cye.im[i]) /
>+                      (c.Py[i] * c.Pe[i] + epsilon);
>+    coherence_xy[i] = (c.Cxy.re[i] * c.Cxy.re[i] + c.Cxy.im[i] * c.Cxy.im[i]) /
>+                      (c.Px[i] * c.Py[i] + epsilon);
>+  }
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/coherence_gain.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/coherence_gain.h
>new file mode 100644
>index 00000000000..b6e22fdbbcf
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/coherence_gain.h
>@@ -0,0 +1,77 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_AEC3_COHERENCE_GAIN_H_
>+#define MODULES_AUDIO_PROCESSING_AEC3_COHERENCE_GAIN_H_
>+
>+#include <array>
>+
>+#include "modules/audio_processing/aec3/aec3_common.h"
>+#include "modules/audio_processing/aec3/aec3_fft.h"
>+#include "rtc_base/constructormagic.h"
>+
>+namespace webrtc {
>+
>+// Class for computing an echo suppression gain based on the coherence measure.
>+class CoherenceGain {
>+ public:
>+  CoherenceGain(int sample_rate_hz, size_t num_bands_to_compute);
>+  ~CoherenceGain();
>+
>+  // Computes the gain based on the FFTs of the filter error output signal, the
>+  // render signal and the capture signal.
>+  void ComputeGain(const FftData& E,
>+                   const FftData& X,
>+                   const FftData& Y,
>+                   rtc::ArrayView<float> gain);
>+
>+ private:
>+  struct {
>+    FftData Cye;
>+    FftData Cxy;
>+    std::array<float, kFftLengthBy2Plus1> Px;
>+    std::array<float, kFftLengthBy2Plus1> Py;
>+    std::array<float, kFftLengthBy2Plus1> Pe;
>+  } spectra_;
>+
>+  struct {
>+    float h_nl_fb_min = 1;
>+    float h_nl_fb_local_min = 1;
>+    float h_nl_xd_avg_min = 1.f;
>+    int h_nl_new_min = 0;
>+    float h_nl_min_ctr = 0;
>+    float overdrive = 2;
>+    float overdrive_scaling = 2;
>+    bool near_state = false;
>+  } gain_state_;
>+
>+  const Aec3Fft fft_;
>+  const size_t num_bands_to_compute_;
>+  const int sample_rate_scaler_;
>+
>+  // Updates the spectral estimates used for the coherence computation.
>+  void UpdateCoherenceSpectra(const FftData& E,
>+                              const FftData& X,
>+                              const FftData& Y);
>+
>+  // Compute the suppression gain based on the coherence.
>+  void FormSuppressionGain(rtc::ArrayView<const float> coherence_ye,
>+                           rtc::ArrayView<const float> coherence_xy,
>+                           rtc::ArrayView<float> h_nl);
>+
>+  // Compute the coherence.
>+  void ComputeCoherence(rtc::ArrayView<float> coherence_ye,
>+                        rtc::ArrayView<float> coherence_xy) const;
>+
>+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(CoherenceGain);
>+};
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_AEC3_COHERENCE_GAIN_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/comfort_noise_generator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/comfort_noise_generator.cc
>index dab40a9d7ed..55faf4b5291 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/comfort_noise_generator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/comfort_noise_generator.cc
>@@ -10,7 +10,9 @@
> 
> #include "modules/audio_processing/aec3/comfort_noise_generator.h"
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>+// Defines WEBRTC_ARCH_X86_FAMILY, used below.
>+#include "rtc_base/system/arch.h"
>+
> #if defined(WEBRTC_ARCH_X86_FAMILY)
> #include <emmintrin.h>
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/comfort_noise_generator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/comfort_noise_generator.h
>index 2d998beb94f..6a479896ecb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/comfort_noise_generator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/comfort_noise_generator.h
>@@ -18,6 +18,7 @@
> #include "modules/audio_processing/aec3/aec_state.h"
> #include "modules/audio_processing/aec3/fft_data.h"
> #include "rtc_base/constructormagic.h"
>+#include "rtc_base/system/arch.h"
> 
> namespace webrtc {
> namespace aec3 {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/comfort_noise_generator_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/comfort_noise_generator_unittest.cc
>index d7e9407dff3..77a09edcd6a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/comfort_noise_generator_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/comfort_noise_generator_unittest.cc
>@@ -14,9 +14,9 @@
> #include <numeric>
> 
> #include "rtc_base/random.h"
>+#include "rtc_base/system/arch.h"
> #include "system_wrappers/include/cpu_features_wrapper.h"
> #include "test/gtest.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> namespace aec3 {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/decimator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/decimator.cc
>index 135a771a7ce..d9faa62dc44 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/decimator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/decimator.cc
>@@ -14,39 +14,53 @@
> namespace webrtc {
> namespace {
> 
>-// b, a = signal.butter(2, 3400/8000.0, 'lowpass', analog=False) which are the
>-// same as b, a = signal.butter(2, 1700/4000.0, 'lowpass', analog=False).
>-const CascadedBiQuadFilter::BiQuadCoefficients kLowPassFilterCoefficients2 = {
>-    {0.22711796f, 0.45423593f, 0.22711796f},
>-    {-0.27666461f, 0.18513647f}};
>-constexpr int kNumFilters2 = 3;
>+// signal.butter(2, 3400/8000.0, 'lowpass', analog=False)
>+const std::vector<CascadedBiQuadFilter::BiQuadParam> GetLowPassFilterDS2() {
>+  return std::vector<CascadedBiQuadFilter::BiQuadParam>{
>+      {{-1.f, 0.f}, {0.13833231f, 0.40743176f}, 0.22711796393486466f},
>+      {{-1.f, 0.f}, {0.13833231f, 0.40743176f}, 0.22711796393486466f},
>+      {{-1.f, 0.f}, {0.13833231f, 0.40743176f}, 0.22711796393486466f}};
>+}
> 
>-// b, a = signal.butter(2, 1500/8000.0, 'lowpass', analog=False) which are the
>-// same as b, a = signal.butter(2, 75/4000.0, 'lowpass', analog=False).
>-const CascadedBiQuadFilter::BiQuadCoefficients kLowPassFilterCoefficients4 = {
>-    {0.0179f, 0.0357f, 0.0179f},
>-    {-1.5879f, 0.6594f}};
>-constexpr int kNumFilters4 = 3;
>+// signal.ellip(6, 1, 40, 1800/8000, btype='lowpass', analog=False)
>+const std::vector<CascadedBiQuadFilter::BiQuadParam> GetLowPassFilterDS4() {
>+  return std::vector<CascadedBiQuadFilter::BiQuadParam>{
>+      {{-0.08873842f, 0.99605496f}, {0.75916227f, 0.23841065f}, 0.26250696827f},
>+      {{0.62273832f, 0.78243018f}, {0.74892112f, 0.5410152f}, 0.26250696827f},
>+      {{0.71107693f, 0.70311421f}, {0.74895534f, 0.63924616f}, 0.26250696827f}};
>+}
> 
>-// b, a = signal.butter(2, 800/8000.0, 'lowpass', analog=False) which are the
>-// same as b, a = signal.butter(2, 400/4000.0, 'lowpass', analog=False).
>-const CascadedBiQuadFilter::BiQuadCoefficients kLowPassFilterCoefficients8 = {
>-    {0.02008337f, 0.04016673f, 0.02008337f},
>-    {-1.56101808f, 0.64135154f}};
>-constexpr int kNumFilters8 = 4;
>+// signal.cheby1(1, 6, [1000/8000, 2000/8000], btype='bandpass', analog=False)
>+const std::vector<CascadedBiQuadFilter::BiQuadParam> GetBandPassFilterDS8() {
>+  return std::vector<CascadedBiQuadFilter::BiQuadParam>{
>+      {{1.f, 0.f}, {0.7601815f, 0.46423542f}, 0.10330478266505948f, true},
>+      {{1.f, 0.f}, {0.7601815f, 0.46423542f}, 0.10330478266505948f, true},
>+      {{1.f, 0.f}, {0.7601815f, 0.46423542f}, 0.10330478266505948f, true},
>+      {{1.f, 0.f}, {0.7601815f, 0.46423542f}, 0.10330478266505948f, true},
>+      {{1.f, 0.f}, {0.7601815f, 0.46423542f}, 0.10330478266505948f, true}};
>+}
> 
>+// signal.butter(2, 1000/8000.0, 'highpass', analog=False)
>+const std::vector<CascadedBiQuadFilter::BiQuadParam> GetHighPassFilter() {
>+  return std::vector<CascadedBiQuadFilter::BiQuadParam>{
>+      {{1.f, 0.f}, {0.72712179f, 0.21296904f}, 0.7570763753338849f}};
>+}
>+
>+const std::vector<CascadedBiQuadFilter::BiQuadParam> GetPassThroughFilter() {
>+  return std::vector<CascadedBiQuadFilter::BiQuadParam>{};
>+}
> }  // namespace
> 
> Decimator::Decimator(size_t down_sampling_factor)
>     : down_sampling_factor_(down_sampling_factor),
>-      low_pass_filter_(
>-          down_sampling_factor_ == 4
>-              ? kLowPassFilterCoefficients4
>-              : (down_sampling_factor_ == 8 ? kLowPassFilterCoefficients8
>-                                            : kLowPassFilterCoefficients2),
>-          down_sampling_factor_ == 4
>-              ? kNumFilters4
>-              : (down_sampling_factor_ == 8 ? kNumFilters8 : kNumFilters2)) {
>+      anti_aliasing_filter_(down_sampling_factor_ == 4
>+                                ? GetLowPassFilterDS4()
>+                                : (down_sampling_factor_ == 8
>+                                       ? GetBandPassFilterDS8()
>+                                       : GetLowPassFilterDS2())),
>+      noise_reduction_filter_(down_sampling_factor_ == 8
>+                                  ? GetPassThroughFilter()
>+                                  : GetHighPassFilter()) {
>   RTC_DCHECK(down_sampling_factor_ == 2 || down_sampling_factor_ == 4 ||
>              down_sampling_factor_ == 8);
> }
>@@ -58,7 +72,10 @@ void Decimator::Decimate(rtc::ArrayView<const float> in,
>   std::array<float, kBlockSize> x;
> 
>   // Limit the frequency content of the signal to avoid aliasing.
>-  low_pass_filter_.Process(in, x);
>+  anti_aliasing_filter_.Process(in, x);
>+
>+  // Reduce the impact of near-end noise.
>+  noise_reduction_filter_.Process(x);
> 
>   // Downsample the signal.
>   for (size_t j = 0, k = 0; j < out.size(); ++j, k += down_sampling_factor_) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/decimator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/decimator.h
>index 7418a26fb01..2bb60a4d7b7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/decimator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/decimator.h
>@@ -30,7 +30,8 @@ class Decimator {
> 
>  private:
>   const size_t down_sampling_factor_;
>-  CascadedBiQuadFilter low_pass_filter_;
>+  CascadedBiQuadFilter anti_aliasing_filter_;
>+  CascadedBiQuadFilter noise_reduction_filter_;
> 
>   RTC_DISALLOW_COPY_AND_ASSIGN(Decimator);
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/decimator_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/decimator_unittest.cc
>index e77a990f9ca..e9ab21b70cf 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/decimator_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/decimator_unittest.cc
>@@ -99,21 +99,6 @@ TEST(Decimator, NoLeakageFromUpperFrequencies) {
>   }
> }
> 
>-// Verifies that the impact of low-frequency content is small during the
>-// downsampling.
>-TEST(Decimator, NoImpactOnLowerFrequencies) {
>-  float input_power;
>-  float output_power;
>-  for (auto rate : {8000, 16000, 32000, 48000}) {
>-    for (auto down_sampling_factor : kDownSamplingFactors) {
>-      ProduceDebugText(rate);
>-      ProduceDecimatedSinusoidalOutputPower(rate, down_sampling_factor, 200.f,
>-                                            &input_power, &output_power);
>-      EXPECT_LT(0.7f * input_power, output_power);
>-    }
>-  }
>-}
>-
> #if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
> // Verifies the check for the input size.
> TEST(Decimator, WrongInputSize) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/delay_estimate.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/delay_estimate.h
>index c3911f7f425..ea5dd271532 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/delay_estimate.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/delay_estimate.h
>@@ -22,6 +22,8 @@ struct DelayEstimate {
> 
>   Quality quality;
>   size_t delay;
>+  size_t blocks_since_last_change = 0;
>+  size_t blocks_since_last_update = 0;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_audibility.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_audibility.cc
>new file mode 100644
>index 00000000000..567873b181b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_audibility.cc
>@@ -0,0 +1,107 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/aec3/echo_audibility.h"
>+
>+#include <algorithm>
>+#include <cmath>
>+
>+#include "modules/audio_processing/aec3/aec3_common.h"
>+#include "modules/audio_processing/aec3/matrix_buffer.h"
>+#include "modules/audio_processing/aec3/stationarity_estimator.h"
>+#include "modules/audio_processing/aec3/vector_buffer.h"
>+
>+namespace webrtc {
>+
>+EchoAudibility::EchoAudibility() {
>+  Reset();
>+}
>+
>+EchoAudibility::~EchoAudibility() = default;
>+
>+void EchoAudibility::Update(const RenderBuffer& render_buffer,
>+                            int delay_blocks,
>+                            bool external_delay_seen,
>+                            float reverb_decay) {
>+  UpdateRenderNoiseEstimator(render_buffer.GetSpectrumBuffer(),
>+                             render_buffer.GetBlockBuffer(),
>+                             external_delay_seen);
>+
>+  if (external_delay_seen) {
>+    UpdateRenderStationarityFlags(render_buffer, delay_blocks, reverb_decay);
>+  }
>+}
>+
>+void EchoAudibility::Reset() {
>+  render_stationarity_.Reset();
>+  non_zero_render_seen_ = false;
>+  render_spectrum_write_prev_ = absl::nullopt;
>+}
>+
>+void EchoAudibility::UpdateRenderStationarityFlags(
>+    const RenderBuffer& render_buffer,
>+    int delay_blocks,
>+    float reverb_decay) {
>+  const VectorBuffer& spectrum_buffer = render_buffer.GetSpectrumBuffer();
>+  int idx_at_delay =
>+      spectrum_buffer.OffsetIndex(spectrum_buffer.read, delay_blocks);
>+
>+  int num_lookahead = render_buffer.Headroom() - delay_blocks + 1;
>+  num_lookahead = std::max(0, num_lookahead);
>+
>+  render_stationarity_.UpdateStationarityFlags(spectrum_buffer, idx_at_delay,
>+                                               num_lookahead, reverb_decay);
>+}
>+
>+void EchoAudibility::UpdateRenderNoiseEstimator(
>+    const VectorBuffer& spectrum_buffer,
>+    const MatrixBuffer& block_buffer,
>+    bool external_delay_seen) {
>+  if (!render_spectrum_write_prev_) {
>+    render_spectrum_write_prev_ = spectrum_buffer.write;
>+    render_block_write_prev_ = block_buffer.write;
>+    return;
>+  }
>+  int render_spectrum_write_current = spectrum_buffer.write;
>+  if (!non_zero_render_seen_ && !external_delay_seen) {
>+    non_zero_render_seen_ = !IsRenderTooLow(block_buffer);
>+  }
>+  if (non_zero_render_seen_) {
>+    for (int idx = render_spectrum_write_prev_.value();
>+         idx != render_spectrum_write_current;
>+         idx = spectrum_buffer.DecIndex(idx)) {
>+      render_stationarity_.UpdateNoiseEstimator(spectrum_buffer.buffer[idx]);
>+    }
>+  }
>+  render_spectrum_write_prev_ = render_spectrum_write_current;
>+}
>+
>+bool EchoAudibility::IsRenderTooLow(const MatrixBuffer& block_buffer) {
>+  bool too_low = false;
>+  const int render_block_write_current = block_buffer.write;
>+  if (render_block_write_current == render_block_write_prev_) {
>+    too_low = true;
>+  } else {
>+    for (int idx = render_block_write_prev_; idx != render_block_write_current;
>+         idx = block_buffer.IncIndex(idx)) {
>+      auto block = block_buffer.buffer[idx][0];
>+      auto r = std::minmax_element(block.cbegin(), block.cend());
>+      float max_abs = std::max(std::fabs(*r.first), std::fabs(*r.second));
>+      if (max_abs < 10) {
>+        too_low = true;  // Discards all blocks if one of them is too low.
>+        break;
>+      }
>+    }
>+  }
>+  render_block_write_prev_ = render_block_write_current;
>+  return too_low;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_audibility.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_audibility.h
>new file mode 100644
>index 00000000000..4650fa53862
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_audibility.h
>@@ -0,0 +1,87 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_AEC3_ECHO_AUDIBILITY_H_
>+#define MODULES_AUDIO_PROCESSING_AEC3_ECHO_AUDIBILITY_H_
>+
>+#include <algorithm>
>+#include <array>
>+#include <limits>
>+#include <memory>
>+#include <vector>
>+
>+#include "absl/types/optional.h"
>+#include "api/array_view.h"
>+#include "modules/audio_processing/aec3/matrix_buffer.h"
>+#include "modules/audio_processing/aec3/render_buffer.h"
>+#include "modules/audio_processing/aec3/stationarity_estimator.h"
>+#include "modules/audio_processing/aec3/vector_buffer.h"
>+#include "rtc_base/constructormagic.h"
>+
>+namespace webrtc {
>+
>+class ApmDataDumper;
>+
>+class EchoAudibility {
>+ public:
>+  EchoAudibility();
>+  ~EchoAudibility();
>+
>+  // Feed new render data to the echo audibility estimator.
>+  void Update(const RenderBuffer& render_buffer,
>+              int delay_blocks,
>+              bool external_delay_seen,
>+              float reverb_decay);
>+
>+  // Get the residual echo scaling.
>+  void GetResidualEchoScaling(rtc::ArrayView<float> residual_scaling) const {
>+    for (size_t band = 0; band < residual_scaling.size(); ++band) {
>+      if (render_stationarity_.IsBandStationary(band)) {
>+        residual_scaling[band] = 0.f;
>+      } else {
>+        residual_scaling[band] = 1.0f;
>+      }
>+    }
>+  }
>+
>+  // Returns true if the current render block is estimated as stationary.
>+  bool IsBlockStationary() const {
>+    return render_stationarity_.IsBlockStationary();
>+  }
>+
>+ private:
>+  // Reset the EchoAudibility class.
>+  void Reset();
>+
>+  // Updates the render stationarity flags for the current frame.
>+  void UpdateRenderStationarityFlags(const RenderBuffer& render_buffer,
>+                                     int delay_blocks,
>+                                     float reverb_decay);
>+
>+  // Updates the noise estimator with the new render data since the previous
>+  // call to this method.
>+  void UpdateRenderNoiseEstimator(const VectorBuffer& spectrum_buffer,
>+                                  const MatrixBuffer& block_buffer,
>+                                  bool external_delay_seen);
>+
>+  // Returns a bool being true if the render signal contains just close to zero
>+  // values.
>+  bool IsRenderTooLow(const MatrixBuffer& block_buffer);
>+
>+  absl::optional<int> render_spectrum_write_prev_;
>+  int render_block_write_prev_;
>+  bool non_zero_render_seen_;
>+  StationarityEstimator render_stationarity_;
>+  RTC_DISALLOW_COPY_AND_ASSIGN(EchoAudibility);
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_AEC3_ECHO_AUDIBILITY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_canceller3.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_canceller3.cc
>index 100ed27160b..07d8b254aa0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_canceller3.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_canceller3.cc
>@@ -9,10 +9,10 @@
>  */
> #include "modules/audio_processing/aec3/echo_canceller3.h"
> 
>-#include <sstream>
>-
> #include "modules/audio_processing/logging/apm_data_dumper.h"
> #include "rtc_base/atomicops.h"
>+#include "rtc_base/logging.h"
>+#include "system_wrappers/include/field_trial.h"
> 
> namespace webrtc {
> 
>@@ -29,6 +29,106 @@ bool DetectSaturation(rtc::ArrayView<const float> y) {
>   return false;
> }
> 
>+bool UseShortDelayEstimatorWindow() {
>+  return field_trial::IsEnabled("WebRTC-Aec3UseShortDelayEstimatorWindow");
>+}
>+
>+bool EnableReverbBasedOnRender() {
>+  return !field_trial::IsEnabled("WebRTC-Aec3ReverbBasedOnRenderKillSwitch");
>+}
>+
>+bool EnableReverbModelling() {
>+  return !field_trial::IsEnabled("WebRTC-Aec3ReverbModellingKillSwitch");
>+}
>+
>+bool EnableSuppressorNearendAveraging() {
>+  return !field_trial::IsEnabled(
>+      "WebRTC-Aec3SuppressorNearendAveragingKillSwitch");
>+}
>+
>+bool EnableSlowFilterAdaptation() {
>+  return !field_trial::IsEnabled("WebRTC-Aec3SlowFilterAdaptationKillSwitch");
>+}
>+
>+bool EnableShadowFilterJumpstart() {
>+  return !field_trial::IsEnabled("WebRTC-Aec3ShadowFilterJumpstartKillSwitch");
>+}
>+
>+// Method for adjusting config parameter dependencies..
>+EchoCanceller3Config AdjustConfig(const EchoCanceller3Config& config) {
>+  EchoCanceller3Config adjusted_cfg = config;
>+
>+  if (!EnableReverbModelling()) {
>+    adjusted_cfg.ep_strength.default_len = 0.f;
>+  }
>+
>+  // Use customized parameters when the system has clock-drift.
>+  if (config.echo_removal_control.has_clock_drift) {
>+    RTC_LOG(LS_WARNING)
>+        << "Customizing parameters to work well for the clock-drift case.";
>+    if (config.ep_strength.bounded_erl) {
>+      adjusted_cfg.ep_strength.default_len = 0.85f;
>+      adjusted_cfg.ep_strength.lf = 0.01f;
>+      adjusted_cfg.ep_strength.mf = 0.01f;
>+      adjusted_cfg.ep_strength.hf = 0.01f;
>+      adjusted_cfg.echo_model.render_pre_window_size = 1;
>+      adjusted_cfg.echo_model.render_post_window_size = 1;
>+      adjusted_cfg.echo_model.nonlinear_hold = 3;
>+      adjusted_cfg.echo_model.nonlinear_release = 0.001f;
>+    } else {
>+      adjusted_cfg.ep_strength.bounded_erl = true;
>+      adjusted_cfg.delay.down_sampling_factor = 2;
>+      adjusted_cfg.ep_strength.default_len = 0.8f;
>+      adjusted_cfg.ep_strength.lf = 0.01f;
>+      adjusted_cfg.ep_strength.mf = 0.01f;
>+      adjusted_cfg.ep_strength.hf = 0.01f;
>+      adjusted_cfg.filter.main = {30, 0.1f, 0.8f, 0.001f, 20075344.f};
>+      adjusted_cfg.filter.shadow = {30, 0.7f, 20075344.f};
>+      adjusted_cfg.filter.main_initial = {30, 0.1f, 1.5f, 0.001f, 20075344.f};
>+      adjusted_cfg.filter.shadow_initial = {30, 0.9f, 20075344.f};
>+      adjusted_cfg.echo_model.render_pre_window_size = 2;
>+      adjusted_cfg.echo_model.render_post_window_size = 2;
>+      adjusted_cfg.echo_model.nonlinear_hold = 3;
>+      adjusted_cfg.echo_model.nonlinear_release = 0.6f;
>+    }
>+  }
>+
>+  if (UseShortDelayEstimatorWindow()) {
>+    adjusted_cfg.delay.num_filters =
>+        std::min(adjusted_cfg.delay.num_filters, static_cast<size_t>(5));
>+  }
>+
>+  if (EnableReverbBasedOnRender() == false) {
>+    adjusted_cfg.ep_strength.reverb_based_on_render = false;
>+  }
>+
>+  if (!EnableSuppressorNearendAveraging()) {
>+    adjusted_cfg.suppressor.nearend_average_blocks = 1;
>+  }
>+
>+  if (!EnableSlowFilterAdaptation()) {
>+    if (!EnableShadowFilterJumpstart()) {
>+      adjusted_cfg.filter.main.leakage_converged = 0.005f;
>+      adjusted_cfg.filter.main.leakage_diverged = 0.1f;
>+    }
>+    adjusted_cfg.filter.main_initial.leakage_converged = 0.05f;
>+    adjusted_cfg.filter.main_initial.leakage_diverged = 5.f;
>+  }
>+
>+  if (!EnableShadowFilterJumpstart()) {
>+    if (EnableSlowFilterAdaptation()) {
>+      adjusted_cfg.filter.main.leakage_converged = 0.0005f;
>+      adjusted_cfg.filter.main.leakage_diverged = 0.01f;
>+    } else {
>+      adjusted_cfg.filter.main.leakage_converged = 0.005f;
>+      adjusted_cfg.filter.main.leakage_diverged = 0.1f;
>+    }
>+    adjusted_cfg.filter.main.error_floor = 0.001f;
>+  }
>+
>+  return adjusted_cfg;
>+}
>+
> void FillSubFrameView(AudioBuffer* frame,
>                       size_t sub_frame_index,
>                       std::vector<rtc::ArrayView<float>>* sub_frame_view) {
>@@ -184,6 +284,12 @@ EchoCanceller3::RenderWriter::~RenderWriter() = default;
> void EchoCanceller3::RenderWriter::Insert(AudioBuffer* input) {
>   RTC_DCHECK_EQ(1, input->num_channels());
>   RTC_DCHECK_EQ(frame_length_, input->num_frames_per_band());
>+  RTC_DCHECK_EQ(num_bands_, input->num_bands());
>+
>+  // TODO(bugs.webrtc.org/8759) Temporary work-around.
>+  if (num_bands_ != static_cast<int>(input->num_bands()))
>+    return;
>+
>   data_dumper_->DumpWav("aec3_render_input", frame_length_,
>                         &input->split_bands_f(0)[0][0],
>                         LowestBandRate(sample_rate_hz_), 1);
>@@ -203,11 +309,12 @@ int EchoCanceller3::instance_count_ = 0;
> EchoCanceller3::EchoCanceller3(const EchoCanceller3Config& config,
>                                int sample_rate_hz,
>                                bool use_highpass_filter)
>-    : EchoCanceller3(config,
>-                     sample_rate_hz,
>-                     use_highpass_filter,
>-                     std::unique_ptr<BlockProcessor>(
>-                         BlockProcessor::Create(config, sample_rate_hz))) {}
>+    : EchoCanceller3(
>+          AdjustConfig(config),
>+          sample_rate_hz,
>+          use_highpass_filter,
>+          std::unique_ptr<BlockProcessor>(
>+              BlockProcessor::Create(AdjustConfig(config), sample_rate_hz))) {}
> EchoCanceller3::EchoCanceller3(const EchoCanceller3Config& config,
>                                int sample_rate_hz,
>                                bool use_highpass_filter,
>@@ -332,8 +439,9 @@ EchoControl::Metrics EchoCanceller3::GetMetrics() const {
>   return metrics;
> }
> 
>-bool EchoCanceller3::Validate(const EchoCanceller3Config& config) {
>-  return true;
>+void EchoCanceller3::SetAudioBufferDelay(size_t delay_ms) {
>+  RTC_DCHECK_RUNS_SERIALIZED(&capture_race_checker_);
>+  block_processor_->SetAudioBufferDelay(delay_ms);
> }
> 
> void EchoCanceller3::EmptyRenderQueue() {
>@@ -357,20 +465,4 @@ void EchoCanceller3::EmptyRenderQueue() {
>         render_transfer_queue_.Remove(&render_queue_output_frame_);
>   }
> }
>-
>-EchoCanceller3Factory::EchoCanceller3Factory() {}
>-
>-EchoCanceller3Factory::EchoCanceller3Factory(const EchoCanceller3Config& config)
>-    : config_(config) {
>-  // Revert to default configuration if needed.
>-  if (!EchoCanceller3::Validate(config_)) {
>-    config_ = EchoCanceller3Config();
>-  }
>-}
>-
>-std::unique_ptr<EchoControl> EchoCanceller3Factory::Create(int sample_rate_hz) {
>-  return std::unique_ptr<EchoControl>(
>-      new EchoCanceller3(config_, sample_rate_hz, true));
>-}
>-
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_canceller3.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_canceller3.h
>index d136c1e0112..b66b8b1dd1e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_canceller3.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_canceller3.h
>@@ -11,12 +11,12 @@
> #ifndef MODULES_AUDIO_PROCESSING_AEC3_ECHO_CANCELLER3_H_
> #define MODULES_AUDIO_PROCESSING_AEC3_ECHO_CANCELLER3_H_
> 
>+#include "api/audio/echo_canceller3_config.h"
> #include "modules/audio_processing/aec3/block_framer.h"
> #include "modules/audio_processing/aec3/block_processor.h"
> #include "modules/audio_processing/aec3/cascaded_biquad_filter.h"
> #include "modules/audio_processing/aec3/frame_blocker.h"
> #include "modules/audio_processing/audio_buffer.h"
>-#include "modules/audio_processing/include/audio_processing.h"
> #include "modules/audio_processing/logging/apm_data_dumper.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/race_checker.h"
>@@ -82,6 +82,8 @@ class EchoCanceller3 : public EchoControl {
>   void ProcessCapture(AudioBuffer* capture, bool level_change) override;
>   // Collect current metrics from the echo canceller.
>   Metrics GetMetrics() const override;
>+  // Provides an optional external estimate of the audio buffer delay.
>+  void SetAudioBufferDelay(size_t delay_ms) override;
> 
>   // Signals whether an external detector has detected echo leakage from the
>   // echo canceller.
>@@ -92,9 +94,6 @@ class EchoCanceller3 : public EchoControl {
>     block_processor_->UpdateEchoLeakageStatus(leakage_detected);
>   }
> 
>-  // Validates a config.
>-  static bool Validate(const EchoCanceller3Config& config);
>-
>  private:
>   class RenderWriter;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_canceller3_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_canceller3_unittest.cc
>index d54295f90ab..f6526421304 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_canceller3_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_canceller3_unittest.cc
>@@ -104,6 +104,8 @@ class CaptureTransportVerificationProcessor : public BlockProcessor {
> 
>   void GetMetrics(EchoControl::Metrics* metrics) const override {}
> 
>+  void SetAudioBufferDelay(size_t delay_ms) override{};
>+
>  private:
>   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(CaptureTransportVerificationProcessor);
> };
>@@ -132,6 +134,8 @@ class RenderTransportVerificationProcessor : public BlockProcessor {
> 
>   void GetMetrics(EchoControl::Metrics* metrics) const override {}
> 
>+  void SetAudioBufferDelay(size_t delay_ms) override{};
>+
>  private:
>   std::deque<std::vector<std::vector<float>>> received_render_blocks_;
>   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RenderTransportVerificationProcessor);
>@@ -681,11 +685,6 @@ TEST(EchoCanceller3Messaging, EchoLeakage) {
>   }
> }
> 
>-TEST(EchoCanceller3, ConfigValidation) {
>-  EchoCanceller3Config config;
>-  EXPECT_TRUE(EchoCanceller3::Validate(config));
>-}
>-
> #if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
> 
> TEST(EchoCanceller3InputCheck, WrongCaptureNumBandsCheckVerification) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_path_delay_estimator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_path_delay_estimator.cc
>index d9d753cd432..4cae277c8ee 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_path_delay_estimator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_path_delay_estimator.cc
>@@ -12,29 +12,42 @@
> #include <algorithm>
> #include <array>
> 
>+#include "api/audio/echo_canceller3_config.h"
> #include "modules/audio_processing/aec3/aec3_common.h"
>-#include "modules/audio_processing/include/audio_processing.h"
> #include "modules/audio_processing/logging/apm_data_dumper.h"
> #include "rtc_base/checks.h"
>+#include "system_wrappers/include/field_trial.h"
> 
> namespace webrtc {
>+namespace {
>+size_t GetDownSamplingFactor(const EchoCanceller3Config& config) {
>+  // Do not use down sampling factor 8 if kill switch is triggered.
>+  return (config.delay.down_sampling_factor == 8 &&
>+          field_trial::IsEnabled("WebRTC-Aec3DownSamplingFactor8KillSwitch"))
>+             ? 4
>+             : config.delay.down_sampling_factor;
>+}
>+}  // namespace
> 
> EchoPathDelayEstimator::EchoPathDelayEstimator(
>     ApmDataDumper* data_dumper,
>     const EchoCanceller3Config& config)
>     : data_dumper_(data_dumper),
>-      down_sampling_factor_(config.delay.down_sampling_factor),
>+      down_sampling_factor_(GetDownSamplingFactor(config)),
>       sub_block_size_(down_sampling_factor_ != 0
>                           ? kBlockSize / down_sampling_factor_
>                           : kBlockSize),
>       capture_decimator_(down_sampling_factor_),
>-      matched_filter_(data_dumper_,
>-                      DetectOptimization(),
>-                      sub_block_size_,
>-                      kMatchedFilterWindowSizeSubBlocks,
>-                      config.delay.num_filters,
>-                      kMatchedFilterAlignmentShiftSizeSubBlocks,
>-                      config.render_levels.poor_excitation_render_limit),
>+      matched_filter_(
>+          data_dumper_,
>+          DetectOptimization(),
>+          sub_block_size_,
>+          kMatchedFilterWindowSizeSubBlocks,
>+          config.delay.num_filters,
>+          kMatchedFilterAlignmentShiftSizeSubBlocks,
>+          GetDownSamplingFactor(config) == 8
>+              ? config.render_levels.poor_excitation_render_limit_ds8
>+              : config.render_levels.poor_excitation_render_limit),
>       matched_filter_lag_aggregator_(data_dumper_,
>                                      matched_filter_.GetMaxFilterLag()) {
>   RTC_DCHECK(data_dumper);
>@@ -48,9 +61,11 @@ void EchoPathDelayEstimator::Reset(bool soft_reset) {
>     matched_filter_lag_aggregator_.Reset();
>   }
>   matched_filter_.Reset();
>+  old_aggregated_lag_ = absl::nullopt;
>+  consistent_estimate_counter_ = 0;
> }
> 
>-rtc::Optional<DelayEstimate> EchoPathDelayEstimator::EstimateDelay(
>+absl::optional<DelayEstimate> EchoPathDelayEstimator::EstimateDelay(
>     const DownsampledRenderBuffer& render_buffer,
>     rtc::ArrayView<const float> capture) {
>   RTC_DCHECK_EQ(kBlockSize, capture.size());
>@@ -66,7 +81,7 @@ rtc::Optional<DelayEstimate> EchoPathDelayEstimator::EstimateDelay(
>                         16000 / down_sampling_factor_, 1);
>   matched_filter_.Update(render_buffer, downsampled_capture);
> 
>-  rtc::Optional<DelayEstimate> aggregated_matched_filter_lag =
>+  absl::optional<DelayEstimate> aggregated_matched_filter_lag =
>       matched_filter_lag_aggregator_.Aggregate(
>           matched_filter_.GetLagEstimates());
> 
>@@ -84,6 +99,19 @@ rtc::Optional<DelayEstimate> EchoPathDelayEstimator::EstimateDelay(
>   if (aggregated_matched_filter_lag) {
>     aggregated_matched_filter_lag->delay *= down_sampling_factor_;
>   }
>+
>+  if (old_aggregated_lag_ && aggregated_matched_filter_lag &&
>+      old_aggregated_lag_->delay == aggregated_matched_filter_lag->delay) {
>+    ++consistent_estimate_counter_;
>+  } else {
>+    consistent_estimate_counter_ = 0;
>+  }
>+  old_aggregated_lag_ = aggregated_matched_filter_lag;
>+  constexpr size_t kNumBlocksPerSecondBy2 = kNumBlocksPerSecond / 2;
>+  if (consistent_estimate_counter_ > kNumBlocksPerSecondBy2) {
>+    Reset(true);
>+  }
>+
>   return aggregated_matched_filter_lag;
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_path_delay_estimator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_path_delay_estimator.h
>index 10fa862fdb0..cea9abcb515 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_path_delay_estimator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_path_delay_estimator.h
>@@ -13,13 +13,13 @@
> 
> #include <vector>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
>+#include "api/audio/echo_canceller3_config.h"
> #include "modules/audio_processing/aec3/decimator.h"
> #include "modules/audio_processing/aec3/delay_estimate.h"
> #include "modules/audio_processing/aec3/downsampled_render_buffer.h"
> #include "modules/audio_processing/aec3/matched_filter.h"
> #include "modules/audio_processing/aec3/matched_filter_lag_aggregator.h"
>-#include "modules/audio_processing/include/audio_processing.h"
> #include "rtc_base/constructormagic.h"
> 
> namespace webrtc {
>@@ -38,7 +38,7 @@ class EchoPathDelayEstimator {
>   void Reset(bool soft_reset);
> 
>   // Produce a delay estimate if such is avaliable.
>-  rtc::Optional<DelayEstimate> EstimateDelay(
>+  absl::optional<DelayEstimate> EstimateDelay(
>       const DownsampledRenderBuffer& render_buffer,
>       rtc::ArrayView<const float> capture);
> 
>@@ -55,6 +55,8 @@ class EchoPathDelayEstimator {
>   Decimator capture_decimator_;
>   MatchedFilter matched_filter_;
>   MatchedFilterLagAggregator matched_filter_lag_aggregator_;
>+  absl::optional<DelayEstimate> old_aggregated_lag_;
>+  size_t consistent_estimate_counter_ = 0;
> 
>   RTC_DISALLOW_COPY_AND_ASSIGN(EchoPathDelayEstimator);
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_path_delay_estimator_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_path_delay_estimator_unittest.cc
>index dbd7c0bbb50..0e237a57862 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_path_delay_estimator_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_path_delay_estimator_unittest.cc
>@@ -14,9 +14,9 @@
> #include <sstream>
> #include <string>
> 
>+#include "api/audio/echo_canceller3_config.h"
> #include "modules/audio_processing/aec3/aec3_common.h"
> #include "modules/audio_processing/aec3/render_delay_buffer.h"
>-#include "modules/audio_processing/include/audio_processing.h"
> #include "modules/audio_processing/logging/apm_data_dumper.h"
> #include "modules/audio_processing/test/echo_canceller_test_tools.h"
> #include "rtc_base/random.h"
>@@ -72,7 +72,7 @@ TEST(EchoPathDelayEstimator, DelayEstimation) {
>           delay_samples + 2 * config.delay.api_call_jitter_blocks * 64);
>       EchoPathDelayEstimator estimator(&data_dumper, config);
> 
>-      rtc::Optional<DelayEstimate> estimated_delay_samples;
>+      absl::optional<DelayEstimate> estimated_delay_samples;
>       for (size_t k = 0; k < (500 + (delay_samples) / kBlockSize); ++k) {
>         RandomizeSampleVector(&random_generator, render[0]);
>         signal_delay_buffer.Delay(render[0], capture);
>@@ -84,23 +84,29 @@ TEST(EchoPathDelayEstimator, DelayEstimation) {
> 
>         render_delay_buffer->PrepareCaptureProcessing();
> 
>-        estimated_delay_samples = estimator.EstimateDelay(
>+        auto estimate = estimator.EstimateDelay(
>             render_delay_buffer->GetDownsampledRenderBuffer(), capture);
>+
>+        if (estimate) {
>+          estimated_delay_samples = estimate;
>+        }
>       }
> 
>       if (estimated_delay_samples) {
>-        // Due to the internal down-sampling done inside the delay estimator
>-        // the estimated delay cannot be expected to be exact to the true delay.
>-        EXPECT_NEAR(delay_samples,
>-                    estimated_delay_samples->delay -
>-                        (config.delay.api_call_jitter_blocks + 1) * 64,
>-                    config.delay.down_sampling_factor);
>+        // Allow estimated delay to be off by one sample in the down-sampled
>+        // domain.
>+        size_t delay_ds = delay_samples / down_sampling_factor;
>+        size_t estimated_delay_ds =
>+            (estimated_delay_samples->delay -
>+             (config.delay.api_call_jitter_blocks + 1) * 64) /
>+            down_sampling_factor;
>+        EXPECT_NEAR(delay_ds, estimated_delay_ds, 1);
>       } else {
>         ADD_FAILURE();
>       }
>+    }
>   }
> }
>-}
> 
> // Verifies that the delay estimator does not produce delay estimates for render
> // signals of low level.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_remover.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_remover.cc
>index a153deb863d..5394eaf69ec 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_remover.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_remover.cc
>@@ -22,7 +22,6 @@
> #include "modules/audio_processing/aec3/echo_path_variability.h"
> #include "modules/audio_processing/aec3/echo_remover_metrics.h"
> #include "modules/audio_processing/aec3/fft_data.h"
>-#include "modules/audio_processing/aec3/output_selector.h"
> #include "modules/audio_processing/aec3/render_buffer.h"
> #include "modules/audio_processing/aec3/render_delay_buffer.h"
> #include "modules/audio_processing/aec3/residual_echo_estimator.h"
>@@ -32,11 +31,23 @@
> #include "modules/audio_processing/logging/apm_data_dumper.h"
> #include "rtc_base/atomicops.h"
> #include "rtc_base/constructormagic.h"
>+#include "rtc_base/logging.h"
>+#include "system_wrappers/include/field_trial.h"
> 
> namespace webrtc {
> 
> namespace {
> 
>+bool UseShadowFilterOutput() {
>+  return !field_trial::IsEnabled(
>+      "WebRTC-Aec3UtilizeShadowFilterOutputKillSwitch");
>+}
>+
>+bool UseSmoothSignalTransitions() {
>+  return !field_trial::IsEnabled(
>+      "WebRTC-Aec3SmoothSignalTransitionsKillSwitch");
>+}
>+
> void LinearEchoPower(const FftData& E,
>                      const FftData& Y,
>                      std::array<float, kFftLengthBy2Plus1>* S2) {
>@@ -46,11 +57,40 @@ void LinearEchoPower(const FftData& E,
>   }
> }
> 
>+// Fades between two input signals using a fix-sized transition.
>+void SignalTransition(rtc::ArrayView<const float> from,
>+                      rtc::ArrayView<const float> to,
>+                      rtc::ArrayView<float> out) {
>+  constexpr size_t kTransitionSize = 30;
>+  constexpr float kOneByTransitionSize = 1.f / kTransitionSize;
>+
>+  RTC_DCHECK_EQ(from.size(), to.size());
>+  RTC_DCHECK_EQ(from.size(), out.size());
>+  RTC_DCHECK_LE(kTransitionSize, out.size());
>+
>+  for (size_t k = 0; k < kTransitionSize; ++k) {
>+    out[k] = k * kOneByTransitionSize * to[k];
>+    out[k] += (kTransitionSize - k) * kOneByTransitionSize * to[k];
>+  }
>+
>+  std::copy(to.begin() + kTransitionSize, to.end(),
>+            out.begin() + kTransitionSize);
>+}
>+
>+// Computes a windowed (square root Hanning) padded FFT and updates the related
>+// memory.
>+void WindowedPaddedFft(const Aec3Fft& fft,
>+                       rtc::ArrayView<const float> v,
>+                       rtc::ArrayView<float> v_old,
>+                       FftData* V) {
>+  fft.PaddedFft(v, v_old, Aec3Fft::Window::kSqrtHanning, V);
>+  std::copy(v.begin(), v.end(), v_old.begin());
>+}
>+
> // Class for removing the echo from the capture signal.
> class EchoRemoverImpl final : public EchoRemover {
>  public:
>-  explicit EchoRemoverImpl(const EchoCanceller3Config& config,
>-                           int sample_rate_hz);
>+  EchoRemoverImpl(const EchoCanceller3Config& config, int sample_rate_hz);
>   ~EchoRemoverImpl() override;
> 
>   void GetMetrics(EchoControl::Metrics* metrics) const override;
>@@ -58,11 +98,18 @@ class EchoRemoverImpl final : public EchoRemover {
>   // Removes the echo from a block of samples from the capture signal. The
>   // supplied render signal is assumed to be pre-aligned with the capture
>   // signal.
>-  void ProcessCapture(const EchoPathVariability& echo_path_variability,
>+  void ProcessCapture(EchoPathVariability echo_path_variability,
>                       bool capture_signal_saturation,
>+                      const absl::optional<DelayEstimate>& external_delay,
>                       RenderBuffer* render_buffer,
>                       std::vector<std::vector<float>>* capture) override;
> 
>+  // Returns the internal delay estimate in blocks.
>+  absl::optional<int> Delay() const override {
>+    // TODO(peah): Remove or reactivate this functionality.
>+    return absl::nullopt;
>+  }
>+
>   // Updates the status on whether echo leakage is detected in the output of the
>   // echo remover.
>   void UpdateEchoLeakageStatus(bool leakage_detected) override {
>@@ -70,23 +117,38 @@ class EchoRemoverImpl final : public EchoRemover {
>   }
> 
>  private:
>+  // Selects which of the shadow and main linear filter outputs that is most
>+  // appropriate to pass to the suppressor and forms the linear filter output by
>+  // smoothly transition between those.
>+  void FormLinearFilterOutput(bool smooth_transition,
>+                              const SubtractorOutput& subtractor_output,
>+                              rtc::ArrayView<float> output);
>+
>   static int instance_count_;
>   const EchoCanceller3Config config_;
>   const Aec3Fft fft_;
>   std::unique_ptr<ApmDataDumper> data_dumper_;
>   const Aec3Optimization optimization_;
>   const int sample_rate_hz_;
>+  const bool use_shadow_filter_output_;
>+  const bool use_smooth_signal_transitions_;
>   Subtractor subtractor_;
>   SuppressionGain suppression_gain_;
>   ComfortNoiseGenerator cng_;
>   SuppressionFilter suppression_filter_;
>   RenderSignalAnalyzer render_signal_analyzer_;
>-  OutputSelector output_selector_;
>   ResidualEchoEstimator residual_echo_estimator_;
>   bool echo_leakage_detected_ = false;
>   AecState aec_state_;
>   EchoRemoverMetrics metrics_;
>   bool initial_state_ = true;
>+  std::array<float, kFftLengthBy2> e_old_;
>+  std::array<float, kFftLengthBy2> x_old_;
>+  std::array<float, kFftLengthBy2> y_old_;
>+  size_t block_counter_ = 0;
>+  int gain_change_hangover_ = 0;
>+  bool main_filter_output_last_selected_ = true;
>+  bool linear_filter_output_last_selected_ = true;
> 
>   RTC_DISALLOW_COPY_AND_ASSIGN(EchoRemoverImpl);
> };
>@@ -101,13 +163,19 @@ EchoRemoverImpl::EchoRemoverImpl(const EchoCanceller3Config& config,
>           new ApmDataDumper(rtc::AtomicOps::Increment(&instance_count_))),
>       optimization_(DetectOptimization()),
>       sample_rate_hz_(sample_rate_hz),
>+      use_shadow_filter_output_(UseShadowFilterOutput()),
>+      use_smooth_signal_transitions_(UseSmoothSignalTransitions()),
>       subtractor_(config, data_dumper_.get(), optimization_),
>-      suppression_gain_(config_, optimization_),
>+      suppression_gain_(config_, optimization_, sample_rate_hz),
>       cng_(optimization_),
>       suppression_filter_(sample_rate_hz_),
>+      render_signal_analyzer_(config_),
>       residual_echo_estimator_(config_),
>       aec_state_(config_) {
>   RTC_DCHECK(ValidFullBandRate(sample_rate_hz));
>+  x_old_.fill(0.f);
>+  y_old_.fill(0.f);
>+  e_old_.fill(0.f);
> }
> 
> EchoRemoverImpl::~EchoRemoverImpl() = default;
>@@ -116,14 +184,16 @@ void EchoRemoverImpl::GetMetrics(EchoControl::Metrics* metrics) const {
>   // Echo return loss (ERL) is inverted to go from gain to attenuation.
>   metrics->echo_return_loss = -10.0 * log10(aec_state_.ErlTimeDomain());
>   metrics->echo_return_loss_enhancement =
>-      10.0 * log10(aec_state_.ErleTimeDomain());
>+      Log2TodB(aec_state_.ErleTimeDomainLog2());
> }
> 
> void EchoRemoverImpl::ProcessCapture(
>-    const EchoPathVariability& echo_path_variability,
>+    EchoPathVariability echo_path_variability,
>     bool capture_signal_saturation,
>+    const absl::optional<DelayEstimate>& external_delay,
>     RenderBuffer* render_buffer,
>     std::vector<std::vector<float>>* capture) {
>+  ++block_counter_;
>   const std::vector<std::vector<float>>& x = render_buffer->Block(0);
>   std::vector<std::vector<float>>* y = capture;
>   RTC_DCHECK(render_buffer);
>@@ -145,54 +215,103 @@ void EchoRemoverImpl::ProcessCapture(
>   aec_state_.UpdateCaptureSaturation(capture_signal_saturation);
> 
>   if (echo_path_variability.AudioPathChanged()) {
>+    // Ensure that the gain change is only acted on once per frame.
>+    if (echo_path_variability.gain_change) {
>+      if (gain_change_hangover_ == 0) {
>+        constexpr int kMaxBlocksPerFrame = 3;
>+        gain_change_hangover_ = kMaxBlocksPerFrame;
>+        RTC_LOG(LS_WARNING)
>+            << "Gain change detected at block " << block_counter_;
>+      } else {
>+        echo_path_variability.gain_change = false;
>+      }
>+    }
>+
>     subtractor_.HandleEchoPathChange(echo_path_variability);
>     aec_state_.HandleEchoPathChange(echo_path_variability);
>-    initial_state_ = true;
>+
>+    if (echo_path_variability.delay_change !=
>+        EchoPathVariability::DelayAdjustment::kNone) {
>+      suppression_gain_.SetInitialState(true);
>+      initial_state_ = true;
>+    }
>+  }
>+  if (gain_change_hangover_ > 0) {
>+    --gain_change_hangover_;
>   }
> 
>   std::array<float, kFftLengthBy2Plus1> Y2;
>+  std::array<float, kFftLengthBy2Plus1> E2;
>   std::array<float, kFftLengthBy2Plus1> R2;
>   std::array<float, kFftLengthBy2Plus1> S2_linear;
>   std::array<float, kFftLengthBy2Plus1> G;
>   float high_bands_gain;
>   FftData Y;
>+  FftData E;
>   FftData comfort_noise;
>   FftData high_band_comfort_noise;
>   SubtractorOutput subtractor_output;
>-  FftData& E_main_nonwindowed = subtractor_output.E_main_nonwindowed;
>-  auto& E2_main = subtractor_output.E2_main_nonwindowed;
>-  auto& E2_shadow = subtractor_output.E2_shadow;
>-  auto& e_main = subtractor_output.e_main;
> 
>   // Analyze the render signal.
>-  render_signal_analyzer_.Update(*render_buffer, aec_state_.FilterDelay());
>+  render_signal_analyzer_.Update(*render_buffer,
>+                                 aec_state_.FilterDelayBlocks());
> 
>   // Perform linear echo cancellation.
>   if (initial_state_ && !aec_state_.InitialState()) {
>     subtractor_.ExitInitialState();
>+    suppression_gain_.SetInitialState(false);
>     initial_state_ = false;
>   }
>+
>+  // If the delay is known, use the echo subtractor.
>   subtractor_.Process(*render_buffer, y0, render_signal_analyzer_, aec_state_,
>                       &subtractor_output);
>+  std::array<float, kBlockSize> e;
>+  FormLinearFilterOutput(use_smooth_signal_transitions_, subtractor_output, e);
> 
>   // Compute spectra.
>-  // fft_.ZeroPaddedFft(y0, Aec3Fft::Window::kHanning, &Y);
>-  fft_.ZeroPaddedFft(y0, Aec3Fft::Window::kRectangular, &Y);
>-  LinearEchoPower(E_main_nonwindowed, Y, &S2_linear);
>+  WindowedPaddedFft(fft_, y0, y_old_, &Y);
>+  WindowedPaddedFft(fft_, e, e_old_, &E);
>+  LinearEchoPower(E, Y, &S2_linear);
>   Y.Spectrum(optimization_, Y2);
>+  E.Spectrum(optimization_, E2);
> 
>   // Update the AEC state information.
>-  aec_state_.Update(subtractor_.FilterFrequencyResponse(),
>-                    subtractor_.FilterImpulseResponse(),
>-                    subtractor_.ConvergedFilter(), *render_buffer, E2_main, Y2,
>-                    subtractor_output.s_main, echo_leakage_detected_);
>+  aec_state_.Update(external_delay, subtractor_.FilterFrequencyResponse(),
>+                    subtractor_.FilterImpulseResponse(), *render_buffer, E2, Y2,
>+                    subtractor_output, y0);
>+
>+  // Compute spectra.
>+  const bool suppression_gain_uses_ffts =
>+      config_.suppressor.bands_with_reliable_coherence > 0;
>+  FftData X;
>+  if (suppression_gain_uses_ffts) {
>+    auto& x_aligned = render_buffer->Block(-aec_state_.FilterDelayBlocks())[0];
>+    WindowedPaddedFft(fft_, x_aligned, x_old_, &X);
>+  } else {
>+    X.Clear();
>+  }
> 
>   // Choose the linear output.
>-  output_selector_.FormLinearOutput(!aec_state_.TransparentMode(), e_main, y0);
>+  data_dumper_->DumpWav("aec3_output_linear2", kBlockSize, &e[0],
>+                        LowestBandRate(sample_rate_hz_), 1);
>+  if (aec_state_.UseLinearFilterOutput()) {
>+    if (!linear_filter_output_last_selected_ &&
>+        use_smooth_signal_transitions_) {
>+      SignalTransition(y0, e, y0);
>+    } else {
>+      std::copy(e.begin(), e.end(), y0.begin());
>+    }
>+  } else {
>+    if (linear_filter_output_last_selected_ && use_smooth_signal_transitions_) {
>+      SignalTransition(e, y0, y0);
>+    }
>+  }
>+  linear_filter_output_last_selected_ = aec_state_.UseLinearFilterOutput();
>+  const auto& Y_fft = aec_state_.UseLinearFilterOutput() ? E : Y;
>+
>   data_dumper_->DumpWav("aec3_output_linear", kBlockSize, &y0[0],
>                         LowestBandRate(sample_rate_hz_), 1);
>-  data_dumper_->DumpRaw("aec3_output_linear", y0);
>-  const auto& E2 = output_selector_.UseSubtractorOutput() ? E2_main : Y2;
> 
>   // Estimate the residual echo power.
>   residual_echo_estimator_.Estimate(aec_state_, *render_buffer, S2_linear, Y2,
>@@ -201,19 +320,17 @@ void EchoRemoverImpl::ProcessCapture(
>   // Estimate the comfort noise.
>   cng_.Compute(aec_state_, Y2, &comfort_noise, &high_band_comfort_noise);
> 
>-  // A choose and apply echo suppression gain.
>-  suppression_gain_.GetGain(E2, R2, cng_.NoiseSpectrum(),
>+  // Compute and apply the suppression gain.
>+  suppression_gain_.GetGain(E2, R2, cng_.NoiseSpectrum(), E, X, Y,
>                             render_signal_analyzer_, aec_state_, x,
>                             &high_bands_gain, &G);
>+
>   suppression_filter_.ApplyGain(comfort_noise, high_band_comfort_noise, G,
>-                                high_bands_gain, y);
>+                                high_bands_gain, Y_fft, y);
> 
>   // Update the metrics.
>   metrics_.Update(aec_state_, cng_.NoiseSpectrum(), G);
> 
>-  // Update the aec state with the aec output characteristics.
>-  aec_state_.UpdateWithOutput(y0);
>-
>   // Debug outputs for the purpose of development and analysis.
>   data_dumper_->DumpWav("aec3_echo_estimate", kBlockSize,
>                         &subtractor_output.s_main[0],
>@@ -227,23 +344,66 @@ void EchoRemoverImpl::ProcessCapture(
>                         rtc::ArrayView<const float>(&y0[0], kBlockSize),
>                         LowestBandRate(sample_rate_hz_), 1);
>   data_dumper_->DumpRaw("aec3_using_subtractor_output",
>-                        output_selector_.UseSubtractorOutput() ? 1 : 0);
>+                        aec_state_.UseLinearFilterOutput() ? 1 : 0);
>   data_dumper_->DumpRaw("aec3_E2", E2);
>-  data_dumper_->DumpRaw("aec3_E2_main", E2_main);
>-  data_dumper_->DumpRaw("aec3_E2_shadow", E2_shadow);
>   data_dumper_->DumpRaw("aec3_S2_linear", S2_linear);
>   data_dumper_->DumpRaw("aec3_Y2", Y2);
>-  data_dumper_->DumpRaw("aec3_X2", render_buffer->Spectrum(0));
>+  data_dumper_->DumpRaw(
>+      "aec3_X2", render_buffer->Spectrum(aec_state_.FilterDelayBlocks()));
>   data_dumper_->DumpRaw("aec3_R2", R2);
>-  data_dumper_->DumpRaw("aec3_erle", aec_state_.Erle());
>-  data_dumper_->DumpRaw("aec3_erl", aec_state_.Erl());
>-  data_dumper_->DumpRaw("aec3_usable_linear_estimate",
>-                        aec_state_.UsableLinearEstimate());
>-  data_dumper_->DumpRaw("aec3_filter_delay", aec_state_.FilterDelay());
>+  data_dumper_->DumpRaw("aec3_R2_reverb",
>+                        residual_echo_estimator_.GetReverbPowerSpectrum());
>+  data_dumper_->DumpRaw("aec3_filter_delay", aec_state_.FilterDelayBlocks());
>   data_dumper_->DumpRaw("aec3_capture_saturation",
>                         aec_state_.SaturatedCapture() ? 1 : 0);
> }
> 
>+void EchoRemoverImpl::FormLinearFilterOutput(
>+    bool smooth_transition,
>+    const SubtractorOutput& subtractor_output,
>+    rtc::ArrayView<float> output) {
>+  RTC_DCHECK_EQ(subtractor_output.e_main.size(), output.size());
>+  RTC_DCHECK_EQ(subtractor_output.e_shadow.size(), output.size());
>+  bool use_main_output = true;
>+  if (use_shadow_filter_output_) {
>+    // As the output of the main adaptive filter generally should be better than
>+    // the shadow filter output, add a margin and threshold for when choosing
>+    // the shadow filter output.
>+    if (subtractor_output.e2_shadow < 0.9f * subtractor_output.e2_main &&
>+        subtractor_output.y2 > 30.f * 30.f * kBlockSize &&
>+        (subtractor_output.s2_main > 60.f * 60.f * kBlockSize ||
>+         subtractor_output.s2_shadow > 60.f * 60.f * kBlockSize)) {
>+      use_main_output = false;
>+    } else {
>+      // If the main filter is diverged, choose the filter output that has the
>+      // lowest power.
>+      if (subtractor_output.e2_shadow < subtractor_output.e2_main &&
>+          subtractor_output.y2 < subtractor_output.e2_main) {
>+        use_main_output = false;
>+      }
>+    }
>+  }
>+
>+  if (use_main_output) {
>+    if (!main_filter_output_last_selected_ && smooth_transition) {
>+      SignalTransition(subtractor_output.e_shadow, subtractor_output.e_main,
>+                       output);
>+    } else {
>+      std::copy(subtractor_output.e_main.begin(),
>+                subtractor_output.e_main.end(), output.begin());
>+    }
>+  } else {
>+    if (main_filter_output_last_selected_ && smooth_transition) {
>+      SignalTransition(subtractor_output.e_main, subtractor_output.e_shadow,
>+                       output);
>+    } else {
>+      std::copy(subtractor_output.e_shadow.begin(),
>+                subtractor_output.e_shadow.end(), output.begin());
>+    }
>+  }
>+  main_filter_output_last_selected_ = use_main_output;
>+}
>+
> }  // namespace
> 
> EchoRemover* EchoRemover::Create(const EchoCanceller3Config& config,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_remover.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_remover.h
>index 795cefa9433..cc8dae9c179 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_remover.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_remover.h
>@@ -13,10 +13,12 @@
> 
> #include <vector>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
>+#include "api/audio/echo_canceller3_config.h"
>+#include "api/audio/echo_control.h"
>+#include "modules/audio_processing/aec3/delay_estimate.h"
> #include "modules/audio_processing/aec3/echo_path_variability.h"
> #include "modules/audio_processing/aec3/render_buffer.h"
>-#include "modules/audio_processing/include/audio_processing.h"
> 
> namespace webrtc {
> 
>@@ -34,11 +36,15 @@ class EchoRemover {
>   // supplied render signal is assumed to be pre-aligned with the capture
>   // signal.
>   virtual void ProcessCapture(
>-      const EchoPathVariability& echo_path_variability,
>+      EchoPathVariability echo_path_variability,
>       bool capture_signal_saturation,
>+      const absl::optional<DelayEstimate>& external_delay,
>       RenderBuffer* render_buffer,
>       std::vector<std::vector<float>>* capture) = 0;
> 
>+  // Returns the internal delay estimate in blocks.
>+  virtual absl::optional<int> Delay() const = 0;
>+
>   // Updates the status on whether echo leakage is detected in the output of the
>   // echo remover.
>   virtual void UpdateEchoLeakageStatus(bool leakage_detected) = 0;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_remover_metrics.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_remover_metrics.cc
>index bc815eb9624..8592a93b657 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_remover_metrics.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_remover_metrics.cc
>@@ -67,7 +67,7 @@ void EchoRemoverMetrics::Update(
>     aec3::UpdateDbMetric(aec_state.Erl(), &erl_);
>     erl_time_domain_.UpdateInstant(aec_state.ErlTimeDomain());
>     aec3::UpdateDbMetric(aec_state.Erle(), &erle_);
>-    erle_time_domain_.UpdateInstant(aec_state.ErleTimeDomain());
>+    erle_time_domain_.UpdateInstant(aec_state.ErleTimeDomainLog2());
>     aec3::UpdateDbMetric(comfort_noise_spectrum, &comfort_noise_);
>     aec3::UpdateDbMetric(suppressor_gain, &suppressor_gain_);
>     active_render_count_ += (aec_state.ActiveRender() ? 1 : 0);
>@@ -237,7 +237,7 @@ void EchoRemoverMetrics::Update(
>             static_cast<int>(
>                 active_render_count_ > kMetricsCollectionBlocksBy2 ? 1 : 0));
>         RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.EchoCanceller.FilterDelay",
>-                                    aec_state.FilterDelay(), 0, 30, 31);
>+                                    aec_state.FilterDelayBlocks(), 0, 30, 31);
>         RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.EchoCanceller.CaptureSaturation",
>                               static_cast<int>(saturated_capture_ ? 1 : 0));
>         break;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_remover_metrics_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_remover_metrics_unittest.cc
>index fbd30d179b9..fb9f151b42b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_remover_metrics_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_remover_metrics_unittest.cc
>@@ -12,8 +12,8 @@
> 
> #include <math.h>
> 
>-#include "modules/audio_processing/aec3/aec_state.h"
> #include "modules/audio_processing/aec3/aec3_fft.h"
>+#include "modules/audio_processing/aec3/aec_state.h"
> #include "test/gtest.h"
> 
> namespace webrtc {
>@@ -78,23 +78,19 @@ TEST(TransformDbMetricForReporting, DbFsScaling) {
> // Verifies that the TransformDbMetricForReporting method is able to properly
> // limit the output.
> TEST(TransformDbMetricForReporting, Limits) {
>-  EXPECT_EQ(
>-      0,
>-      aec3::TransformDbMetricForReporting(false, 0.f, 10.f, 0.f, 1.f, 0.001f));
>-  EXPECT_EQ(
>-      10,
>-      aec3::TransformDbMetricForReporting(false, 0.f, 10.f, 0.f, 1.f, 100.f));
>+  EXPECT_EQ(0, aec3::TransformDbMetricForReporting(false, 0.f, 10.f, 0.f, 1.f,
>+                                                   0.001f));
>+  EXPECT_EQ(10, aec3::TransformDbMetricForReporting(false, 0.f, 10.f, 0.f, 1.f,
>+                                                    100.f));
> }
> 
> // Verifies that the TransformDbMetricForReporting method is able to properly
> // negate output.
> TEST(TransformDbMetricForReporting, Negate) {
>-  EXPECT_EQ(
>-      10,
>-      aec3::TransformDbMetricForReporting(true, -20.f, 20.f, 0.f, 1.f, 0.1f));
>-  EXPECT_EQ(
>-      -10,
>-      aec3::TransformDbMetricForReporting(true, -20.f, 20.f, 0.f, 1.f, 10.f));
>+  EXPECT_EQ(10, aec3::TransformDbMetricForReporting(true, -20.f, 20.f, 0.f, 1.f,
>+                                                    0.1f));
>+  EXPECT_EQ(-10, aec3::TransformDbMetricForReporting(true, -20.f, 20.f, 0.f,
>+                                                     1.f, 10.f));
> }
> 
> // Verify the Update functionality of DbMetric.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_remover_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_remover_unittest.cc
>index c9fad802b13..04a27273d1b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_remover_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/echo_remover_unittest.cc
>@@ -43,6 +43,7 @@ std::string ProduceDebugText(int sample_rate_hz, int delay) {
> 
> // Verifies the basic API call sequence
> TEST(EchoRemover, BasicApiCalls) {
>+  absl::optional<DelayEstimate> delay_estimate;
>   for (auto rate : {8000, 16000, 32000, 48000}) {
>     SCOPED_TRACE(ProduceDebugText(rate));
>     std::unique_ptr<EchoRemover> remover(
>@@ -64,7 +65,8 @@ TEST(EchoRemover, BasicApiCalls) {
>       render_buffer->PrepareCaptureProcessing();
> 
>       remover->ProcessCapture(echo_path_variability, k % 2 == 0 ? true : false,
>-                              render_buffer->GetRenderBuffer(), &capture);
>+                              delay_estimate, render_buffer->GetRenderBuffer(),
>+                              &capture);
>     }
>   }
> }
>@@ -82,6 +84,7 @@ TEST(EchoRemover, DISABLED_WrongSampleRate) {
> 
> // Verifies the check for the capture block size.
> TEST(EchoRemover, WrongCaptureBlockSize) {
>+  absl::optional<DelayEstimate> delay_estimate;
>   for (auto rate : {8000, 16000, 32000, 48000}) {
>     SCOPED_TRACE(ProduceDebugText(rate));
>     std::unique_ptr<EchoRemover> remover(
>@@ -93,7 +96,7 @@ TEST(EchoRemover, WrongCaptureBlockSize) {
>     EchoPathVariability echo_path_variability(
>         false, EchoPathVariability::DelayAdjustment::kNone, false);
>     EXPECT_DEATH(
>-        remover->ProcessCapture(echo_path_variability, false,
>+        remover->ProcessCapture(echo_path_variability, false, delay_estimate,
>                                 render_buffer->GetRenderBuffer(), &capture),
>         "");
>   }
>@@ -103,6 +106,7 @@ TEST(EchoRemover, WrongCaptureBlockSize) {
> // TODO(peah): Re-enable the test once the issue with memory leaks during DEATH
> // tests on test bots has been fixed.c
> TEST(EchoRemover, DISABLED_WrongCaptureNumBands) {
>+  absl::optional<DelayEstimate> delay_estimate;
>   for (auto rate : {16000, 32000, 48000}) {
>     SCOPED_TRACE(ProduceDebugText(rate));
>     std::unique_ptr<EchoRemover> remover(
>@@ -115,7 +119,7 @@ TEST(EchoRemover, DISABLED_WrongCaptureNumBands) {
>     EchoPathVariability echo_path_variability(
>         false, EchoPathVariability::DelayAdjustment::kNone, false);
>     EXPECT_DEATH(
>-        remover->ProcessCapture(echo_path_variability, false,
>+        remover->ProcessCapture(echo_path_variability, false, delay_estimate,
>                                 render_buffer->GetRenderBuffer(), &capture),
>         "");
>   }
>@@ -123,6 +127,7 @@ TEST(EchoRemover, DISABLED_WrongCaptureNumBands) {
> 
> // Verifies the check for non-null capture block.
> TEST(EchoRemover, NullCapture) {
>+  absl::optional<DelayEstimate> delay_estimate;
>   std::unique_ptr<EchoRemover> remover(
>       EchoRemover::Create(EchoCanceller3Config(), 8000));
>   std::unique_ptr<RenderDelayBuffer> render_buffer(
>@@ -130,7 +135,7 @@ TEST(EchoRemover, NullCapture) {
>   EchoPathVariability echo_path_variability(
>       false, EchoPathVariability::DelayAdjustment::kNone, false);
>   EXPECT_DEATH(
>-      remover->ProcessCapture(echo_path_variability, false,
>+      remover->ProcessCapture(echo_path_variability, false, delay_estimate,
>                               render_buffer->GetRenderBuffer(), nullptr),
>       "");
> }
>@@ -142,6 +147,7 @@ TEST(EchoRemover, NullCapture) {
> TEST(EchoRemover, BasicEchoRemoval) {
>   constexpr int kNumBlocksToProcess = 500;
>   Random random_generator(42U);
>+  absl::optional<DelayEstimate> delay_estimate;
>   for (auto rate : {8000, 16000, 32000, 48000}) {
>     std::vector<std::vector<float>> x(NumBandsForRate(rate),
>                                       std::vector<float>(kBlockSize, 0.f));
>@@ -187,7 +193,7 @@ TEST(EchoRemover, BasicEchoRemoval) {
>         render_buffer->Insert(x);
>         render_buffer->PrepareCaptureProcessing();
> 
>-        remover->ProcessCapture(echo_path_variability, false,
>+        remover->ProcessCapture(echo_path_variability, false, delay_estimate,
>                                 render_buffer->GetRenderBuffer(), &y);
> 
>         if (k > kNumBlocksToProcess / 2) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/erle_estimator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/erle_estimator.cc
>index 0e4cbe14691..52ef8edf2fe 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/erle_estimator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/erle_estimator.cc
>@@ -13,27 +13,161 @@
> #include <algorithm>
> #include <numeric>
> 
>+#include "absl/types/optional.h"
>+#include "modules/audio_processing/aec3/aec3_common.h"
>+#include "modules/audio_processing/logging/apm_data_dumper.h"
> #include "rtc_base/numerics/safe_minmax.h"
> 
> namespace webrtc {
> 
>+namespace {
>+constexpr int kPointsToAccumulate = 6;
>+constexpr float kEpsilon = 1e-3f;
>+}  // namespace
>+
> ErleEstimator::ErleEstimator(float min_erle,
>                              float max_erle_lf,
>                              float max_erle_hf)
>     : min_erle_(min_erle),
>+      min_erle_log2_(FastApproxLog2f(min_erle_ + kEpsilon)),
>       max_erle_lf_(max_erle_lf),
>-      max_erle_hf_(max_erle_hf) {
>+      max_erle_lf_log2(FastApproxLog2f(max_erle_lf_ + kEpsilon)),
>+      max_erle_hf_(max_erle_hf),
>+      erle_freq_inst_(kPointsToAccumulate),
>+      erle_time_inst_(kPointsToAccumulate) {
>   erle_.fill(min_erle_);
>+  erle_onsets_.fill(min_erle_);
>   hold_counters_.fill(0);
>-  erle_time_domain_ = min_erle_;
>+  coming_onset_.fill(true);
>+  erle_time_domain_log2_ = min_erle_log2_;
>   hold_counter_time_domain_ = 0;
> }
> 
> ErleEstimator::~ErleEstimator() = default;
> 
>+ErleEstimator::ErleTimeInstantaneous::ErleTimeInstantaneous(
>+    int points_to_accumulate)
>+    : points_to_accumulate_(points_to_accumulate) {
>+  Reset();
>+}
>+ErleEstimator::ErleTimeInstantaneous::~ErleTimeInstantaneous() = default;
>+
>+bool ErleEstimator::ErleTimeInstantaneous::Update(const float Y2_sum,
>+                                                  const float E2_sum) {
>+  bool ret = false;
>+  E2_acum_ += E2_sum;
>+  Y2_acum_ += Y2_sum;
>+  num_points_++;
>+  if (num_points_ == points_to_accumulate_) {
>+    if (E2_acum_ > 0.f) {
>+      ret = true;
>+      erle_log2_ = FastApproxLog2f(Y2_acum_ / E2_acum_ + kEpsilon);
>+    }
>+    num_points_ = 0;
>+    E2_acum_ = 0.f;
>+    Y2_acum_ = 0.f;
>+  }
>+
>+  if (ret) {
>+    UpdateMaxMin();
>+    UpdateQualityEstimate();
>+  }
>+  return ret;
>+}
>+
>+void ErleEstimator::ErleTimeInstantaneous::Reset() {
>+  ResetAccumulators();
>+  max_erle_log2_ = -10.f;  // -30 dB.
>+  min_erle_log2_ = 33.f;   // 100 dB.
>+  inst_quality_estimate_ = 0.f;
>+}
>+
>+void ErleEstimator::ErleTimeInstantaneous::ResetAccumulators() {
>+  erle_log2_ = absl::nullopt;
>+  inst_quality_estimate_ = 0.f;
>+  num_points_ = 0;
>+  E2_acum_ = 0.f;
>+  Y2_acum_ = 0.f;
>+}
>+
>+void ErleEstimator::ErleTimeInstantaneous::Dump(
>+    const std::unique_ptr<ApmDataDumper>& data_dumper) {
>+  data_dumper->DumpRaw("aec3_erle_time_inst_log2",
>+                       erle_log2_ ? *erle_log2_ : -10.f);
>+  data_dumper->DumpRaw(
>+      "aec3_erle_time_quality",
>+      GetInstQualityEstimate() ? GetInstQualityEstimate().value() : 0.f);
>+  data_dumper->DumpRaw("aec3_erle_time_max_log2", max_erle_log2_);
>+  data_dumper->DumpRaw("aec3_erle_time_min_log2", min_erle_log2_);
>+}
>+
>+void ErleEstimator::ErleTimeInstantaneous::UpdateMaxMin() {
>+  RTC_DCHECK(erle_log2_);
>+  if (erle_log2_.value() > max_erle_log2_) {
>+    max_erle_log2_ = erle_log2_.value();
>+  } else {
>+    max_erle_log2_ -= 0.0004;  // Forget factor, approx 1dB every 3 sec.
>+  }
>+
>+  if (erle_log2_.value() < min_erle_log2_) {
>+    min_erle_log2_ = erle_log2_.value();
>+  } else {
>+    min_erle_log2_ += 0.0004;  // Forget factor, approx 1dB every 3 sec.
>+  }
>+}
>+
>+void ErleEstimator::ErleTimeInstantaneous::UpdateQualityEstimate() {
>+  const float alpha = 0.07f;
>+  float quality_estimate = 0.f;
>+  RTC_DCHECK(erle_log2_);
>+  if (max_erle_log2_ > min_erle_log2_) {
>+    quality_estimate = (erle_log2_.value() - min_erle_log2_) /
>+                       (max_erle_log2_ - min_erle_log2_);
>+  }
>+  if (quality_estimate > inst_quality_estimate_) {
>+    inst_quality_estimate_ = quality_estimate;
>+  } else {
>+    inst_quality_estimate_ +=
>+        alpha * (quality_estimate - inst_quality_estimate_);
>+  }
>+}
>+
>+ErleEstimator::ErleFreqInstantaneous::ErleFreqInstantaneous(
>+    int points_to_accumulate)
>+    : points_to_accumulate_(points_to_accumulate) {
>+  Reset();
>+}
>+
>+ErleEstimator::ErleFreqInstantaneous::~ErleFreqInstantaneous() = default;
>+
>+absl::optional<float>
>+ErleEstimator::ErleFreqInstantaneous::Update(float Y2, float E2, size_t band) {
>+  absl::optional<float> ret = absl::nullopt;
>+  RTC_DCHECK_LT(band, kFftLengthBy2Plus1);
>+  Y2_acum_[band] += Y2;
>+  E2_acum_[band] += E2;
>+  if (++num_points_[band] == points_to_accumulate_) {
>+    if (E2_acum_[band]) {
>+      ret = Y2_acum_[band] / E2_acum_[band];
>+    }
>+    num_points_[band] = 0;
>+    Y2_acum_[band] = 0.f;
>+    E2_acum_[band] = 0.f;
>+  }
>+
>+  return ret;
>+}
>+
>+void ErleEstimator::ErleFreqInstantaneous::Reset() {
>+  Y2_acum_.fill(0.f);
>+  E2_acum_.fill(0.f);
>+  num_points_.fill(0);
>+}
>+
> void ErleEstimator::Update(rtc::ArrayView<const float> render_spectrum,
>                            rtc::ArrayView<const float> capture_spectrum,
>-                           rtc::ArrayView<const float> subtractor_spectrum) {
>+                           rtc::ArrayView<const float> subtractor_spectrum,
>+                           bool converged_filter) {
>   RTC_DCHECK_EQ(kFftLengthBy2Plus1, render_spectrum.size());
>   RTC_DCHECK_EQ(kFftLengthBy2Plus1, capture_spectrum.size());
>   RTC_DCHECK_EQ(kFftLengthBy2Plus1, subtractor_spectrum.size());
>@@ -44,49 +178,96 @@ void ErleEstimator::Update(rtc::ArrayView<const float> render_spectrum,
>   // Corresponds of WGN of power -46 dBFS.
>   constexpr float kX2Min = 44015068.0f;
> 
>+  constexpr int kErleHold = 100;
>+  constexpr int kBlocksForOnsetDetection = kErleHold + 150;
>+
>+  auto erle_band_update = [](float erle_band, float new_erle, float alpha_inc,
>+                             float alpha_dec, float min_erle, float max_erle) {
>+    float alpha = new_erle > erle_band ? alpha_inc : alpha_dec;
>+    float erle_band_out = erle_band;
>+    erle_band_out = erle_band + alpha * (new_erle - erle_band);
>+    erle_band_out = rtc::SafeClamp(erle_band_out, min_erle, max_erle);
>+    return erle_band_out;
>+  };
>+
>   // Update the estimates in a clamped minimum statistics manner.
>   auto erle_update = [&](size_t start, size_t stop, float max_erle) {
>     for (size_t k = start; k < stop; ++k) {
>-      if (X2[k] > kX2Min && E2[k] > 0.f) {
>-        const float new_erle = Y2[k] / E2[k];
>-        if (new_erle > erle_[k]) {
>-          hold_counters_[k - 1] = 100;
>-          erle_[k] += 0.1f * (new_erle - erle_[k]);
>-          erle_[k] = rtc::SafeClamp(erle_[k], min_erle_, max_erle);
>+      if (X2[k] > kX2Min) {
>+        absl::optional<float> new_erle =
>+            erle_freq_inst_.Update(Y2[k], E2[k], k);
>+        if (new_erle) {
>+          if (coming_onset_[k]) {
>+            coming_onset_[k] = false;
>+            erle_onsets_[k] =
>+                erle_band_update(erle_onsets_[k], new_erle.value(), 0.15f, 0.3f,
>+                                 min_erle_, max_erle);
>+          }
>+          hold_counters_[k] = kBlocksForOnsetDetection;
>+          erle_[k] = erle_band_update(erle_[k], new_erle.value(), 0.05f, 0.1f,
>+                                      min_erle_, max_erle);
>         }
>       }
>     }
>   };
>-  erle_update(1, kFftLengthBy2 / 2, max_erle_lf_);
>-  erle_update(kFftLengthBy2 / 2, kFftLengthBy2, max_erle_hf_);
> 
>-  std::for_each(hold_counters_.begin(), hold_counters_.end(),
>-                [](int& a) { --a; });
>-  std::transform(hold_counters_.begin(), hold_counters_.end(),
>-                 erle_.begin() + 1, erle_.begin() + 1, [&](int a, float b) {
>-                   return a > 0 ? b : std::max(min_erle_, 0.97f * b);
>-                 });
>+  if (converged_filter) {
>+    // Note that the use of the converged_filter flag already imposed
>+    // a minimum of the erle that can be estimated as that flag would
>+    // be false if the filter is performing poorly.
>+    constexpr size_t kFftLengthBy4 = kFftLengthBy2 / 2;
>+    erle_update(1, kFftLengthBy4, max_erle_lf_);
>+    erle_update(kFftLengthBy4, kFftLengthBy2, max_erle_hf_);
>+  }
>+
>+  for (size_t k = 1; k < kFftLengthBy2; ++k) {
>+    hold_counters_[k]--;
>+    if (hold_counters_[k] <= (kBlocksForOnsetDetection - kErleHold)) {
>+      if (erle_[k] > erle_onsets_[k]) {
>+        erle_[k] = std::max(erle_onsets_[k], 0.97f * erle_[k]);
>+        RTC_DCHECK_LE(min_erle_, erle_[k]);
>+      }
>+      if (hold_counters_[k] <= 0) {
>+        coming_onset_[k] = true;
>+        hold_counters_[k] = 0;
>+      }
>+    }
>+  }
> 
>   erle_[0] = erle_[1];
>   erle_[kFftLengthBy2] = erle_[kFftLengthBy2 - 1];
> 
>-  // Compute ERLE over all frequency bins.
>-  const float X2_sum = std::accumulate(X2.begin(), X2.end(), 0.0f);
>-  const float E2_sum = std::accumulate(E2.begin(), E2.end(), 0.0f);
>-  if (X2_sum > kX2Min * X2.size() && E2_sum > 0.f) {
>-    const float Y2_sum = std::accumulate(Y2.begin(), Y2.end(), 0.0f);
>-    const float new_erle = Y2_sum / E2_sum;
>-    if (new_erle > erle_time_domain_) {
>-      hold_counter_time_domain_ = 100;
>-      erle_time_domain_ += 0.1f * (new_erle - erle_time_domain_);
>-      erle_time_domain_ =
>-          rtc::SafeClamp(erle_time_domain_, min_erle_, max_erle_lf_);
>+  if (converged_filter) {
>+    // Compute ERLE over all frequency bins.
>+    const float X2_sum = std::accumulate(X2.begin(), X2.end(), 0.0f);
>+    if (X2_sum > kX2Min * X2.size()) {
>+      const float Y2_sum = std::accumulate(Y2.begin(), Y2.end(), 0.0f);
>+      const float E2_sum = std::accumulate(E2.begin(), E2.end(), 0.0f);
>+      if (erle_time_inst_.Update(Y2_sum, E2_sum)) {
>+        hold_counter_time_domain_ = kErleHold;
>+        erle_time_domain_log2_ +=
>+            0.1f * ((erle_time_inst_.GetInstErle_log2().value()) -
>+                    erle_time_domain_log2_);
>+        erle_time_domain_log2_ = rtc::SafeClamp(
>+            erle_time_domain_log2_, min_erle_log2_, max_erle_lf_log2);
>+      }
>     }
>   }
>   --hold_counter_time_domain_;
>-  erle_time_domain_ = (hold_counter_time_domain_ > 0)
>-                        ? erle_time_domain_
>-                        : std::max(min_erle_, 0.97f * erle_time_domain_);
>+  if (hold_counter_time_domain_ <= 0) {
>+    erle_time_domain_log2_ =
>+        std::max(min_erle_log2_, erle_time_domain_log2_ - 0.044f);
>+  }
>+  if (hold_counter_time_domain_ == 0) {
>+    erle_time_inst_.ResetAccumulators();
>+  }
>+}
>+
>+void ErleEstimator::Dump(const std::unique_ptr<ApmDataDumper>& data_dumper) {
>+  data_dumper->DumpRaw("aec3_erle", Erle());
>+  data_dumper->DumpRaw("aec3_erle_onset", ErleOnsets());
>+  data_dumper->DumpRaw("aec3_erle_time_domain_log2", ErleTimeDomainLog2());
>+  erle_time_inst_.Dump(data_dumper);
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/erle_estimator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/erle_estimator.h
>index cb9fce6e147..19bc7b407d7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/erle_estimator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/erle_estimator.h
>@@ -13,8 +13,10 @@
> 
> #include <array>
> 
>+#include "absl/types/optional.h"
> #include "api/array_view.h"
> #include "modules/audio_processing/aec3/aec3_common.h"
>+#include "modules/audio_processing/logging/apm_data_dumper.h"
> #include "rtc_base/constructormagic.h"
> 
> namespace webrtc {
>@@ -28,21 +30,89 @@ class ErleEstimator {
>   // Updates the ERLE estimate.
>   void Update(rtc::ArrayView<const float> render_spectrum,
>               rtc::ArrayView<const float> capture_spectrum,
>-              rtc::ArrayView<const float> subtractor_spectrum);
>+              rtc::ArrayView<const float> subtractor_spectrum,
>+              bool converged_filter);
> 
>   // Returns the most recent ERLE estimate.
>   const std::array<float, kFftLengthBy2Plus1>& Erle() const { return erle_; }
>-  float ErleTimeDomain() const { return erle_time_domain_; }
>+  // Returns the ERLE that is estimated during onsets. Use for logging/testing.
>+  const std::array<float, kFftLengthBy2Plus1>& ErleOnsets() const {
>+    return erle_onsets_;
>+  }
>+  float ErleTimeDomainLog2() const { return erle_time_domain_log2_; }
>+
>+  absl::optional<float> GetInstLinearQualityEstimate() const {
>+    return erle_time_inst_.GetInstQualityEstimate();
>+  }
>+
>+  void Dump(const std::unique_ptr<ApmDataDumper>& data_dumper);
>+
>+  class ErleTimeInstantaneous {
>+   public:
>+    ErleTimeInstantaneous(int points_to_accumulate);
>+    ~ErleTimeInstantaneous();
>+    // Update the estimator with a new point, returns true
>+    // if the instantaneous erle was updated due to having enough
>+    // points for performing the estimate.
>+    bool Update(const float Y2_sum, const float E2_sum);
>+    // Reset all the members of the class.
>+    void Reset();
>+    // Reset the members realated with an instantaneous estimate.
>+    void ResetAccumulators();
>+    // Returns the instantaneous ERLE in log2 units.
>+    absl::optional<float> GetInstErle_log2() const { return erle_log2_; }
>+    // Get an indication between 0 and 1 of the performance of the linear filter
>+    // for the current time instant.
>+    absl::optional<float> GetInstQualityEstimate() const {
>+      return erle_log2_ ? absl::optional<float>(inst_quality_estimate_)
>+                        : absl::nullopt;
>+    }
>+    void Dump(const std::unique_ptr<ApmDataDumper>& data_dumper);
>+
>+   private:
>+    void UpdateMaxMin();
>+    void UpdateQualityEstimate();
>+    absl::optional<float> erle_log2_;
>+    float inst_quality_estimate_;
>+    float max_erle_log2_;
>+    float min_erle_log2_;
>+    float Y2_acum_;
>+    float E2_acum_;
>+    int num_points_;
>+    const int points_to_accumulate_;
>+  };
>+
>+  class ErleFreqInstantaneous {
>+   public:
>+    ErleFreqInstantaneous(int points_to_accumulate);
>+    ~ErleFreqInstantaneous();
>+    // Updates the ERLE for a band with a new block. Returns absl::nullopt
>+    // if not enough points were accuulated for doing the estimation.
>+    absl::optional<float> Update(float Y2, float E2, size_t band);
>+    // Reset all the member of the class.
>+    void Reset();
>+
>+   private:
>+    std::array<float, kFftLengthBy2Plus1> Y2_acum_;
>+    std::array<float, kFftLengthBy2Plus1> E2_acum_;
>+    std::array<int, kFftLengthBy2Plus1> num_points_;
>+    const int points_to_accumulate_;
>+  };
> 
>  private:
>   std::array<float, kFftLengthBy2Plus1> erle_;
>-  std::array<int, kFftLengthBy2Minus1> hold_counters_;
>-  float erle_time_domain_;
>+  std::array<float, kFftLengthBy2Plus1> erle_onsets_;
>+  std::array<bool, kFftLengthBy2Plus1> coming_onset_;
>+  std::array<int, kFftLengthBy2Plus1> hold_counters_;
>   int hold_counter_time_domain_;
>+  float erle_time_domain_log2_;
>   const float min_erle_;
>+  const float min_erle_log2_;
>   const float max_erle_lf_;
>+  const float max_erle_lf_log2;
>   const float max_erle_hf_;
>-
>+  ErleFreqInstantaneous erle_freq_inst_;
>+  ErleTimeInstantaneous erle_time_inst_;
>   RTC_DISALLOW_COPY_AND_ASSIGN(ErleEstimator);
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/erle_estimator_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/erle_estimator_unittest.cc
>index f3dd7d9bbbe..86ac5dfd1f6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/erle_estimator_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/erle_estimator_unittest.cc
>@@ -8,7 +8,10 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>+#include <cmath>
>+
> #include "modules/audio_processing/aec3/erle_estimator.h"
>+#include "api/array_view.h"
> #include "test/gtest.h"
> 
> namespace webrtc {
>@@ -16,60 +19,122 @@ namespace webrtc {
> namespace {
> 
> constexpr int kLowFrequencyLimit = kFftLengthBy2 / 2;
>+constexpr float kMaxErleLf = 8.f;
>+constexpr float kMaxErleHf = 1.5f;
>+constexpr float kMinErle = 1.0f;
>+constexpr float kTrueErle = 10.f;
>+constexpr float kTrueErleOnsets = 1.0f;
> 
>-void VerifyErle(const std::array<float, kFftLengthBy2Plus1>& erle,
>-                float erle_time_domain,
>-                float reference_lf,
>-                float reference_hf) {
>+void VerifyErleBands(rtc::ArrayView<const float> erle,
>+                     float reference_lf,
>+                     float reference_hf) {
>   std::for_each(
>       erle.begin(), erle.begin() + kLowFrequencyLimit,
>       [reference_lf](float a) { EXPECT_NEAR(reference_lf, a, 0.001); });
>   std::for_each(
>       erle.begin() + kLowFrequencyLimit, erle.end(),
>       [reference_hf](float a) { EXPECT_NEAR(reference_hf, a, 0.001); });
>-  EXPECT_NEAR(reference_lf, erle_time_domain, 0.001);
>+}
>+
>+void VerifyErle(rtc::ArrayView<const float> erle,
>+                float erle_time_domain,
>+                float reference_lf,
>+                float reference_hf) {
>+  VerifyErleBands(erle, reference_lf, reference_hf);
>+  EXPECT_NEAR(reference_lf, erle_time_domain, 0.5);
>+}
>+
>+void FormFarendFrame(std::array<float, kFftLengthBy2Plus1>* X2,
>+                     std::array<float, kFftLengthBy2Plus1>* E2,
>+                     std::array<float, kFftLengthBy2Plus1>* Y2,
>+                     float erle) {
>+  X2->fill(500 * 1000.f * 1000.f);
>+  E2->fill(1000.f * 1000.f);
>+  Y2->fill(erle * (*E2)[0]);
>+}
>+
>+void FormNearendFrame(std::array<float, kFftLengthBy2Plus1>* X2,
>+                      std::array<float, kFftLengthBy2Plus1>* E2,
>+                      std::array<float, kFftLengthBy2Plus1>* Y2) {
>+  X2->fill(0.f);
>+  Y2->fill(500.f * 1000.f * 1000.f);
>+  E2->fill((*Y2)[0]);
> }
> 
> }  // namespace
> 
>-// Verifies that the correct ERLE estimates are achieved.
>-TEST(ErleEstimator, Estimates) {
>+TEST(ErleEstimator, VerifyErleIncreaseAndHold) {
>   std::array<float, kFftLengthBy2Plus1> X2;
>   std::array<float, kFftLengthBy2Plus1> E2;
>   std::array<float, kFftLengthBy2Plus1> Y2;
> 
>-  ErleEstimator estimator(1.f, 8.f, 1.5f);
>+  ErleEstimator estimator(kMinErle, kMaxErleLf, kMaxErleHf);
>+
>+  // Verifies that the ERLE estimate is properly increased to higher values.
>+  FormFarendFrame(&X2, &E2, &Y2, kTrueErle);
> 
>-  // Verifies that the ERLE estimate is properley increased to higher values.
>-  X2.fill(500 * 1000.f * 1000.f);
>-  E2.fill(1000.f * 1000.f);
>-  Y2.fill(10 * E2[0]);
>   for (size_t k = 0; k < 200; ++k) {
>-    estimator.Update(X2, Y2, E2);
>+    estimator.Update(X2, Y2, E2, true);
>   }
>-  VerifyErle(estimator.Erle(), estimator.ErleTimeDomain(), 8.f, 1.5f);
>+  VerifyErle(estimator.Erle(), std::pow(2.f, estimator.ErleTimeDomainLog2()),
>+             kMaxErleLf, kMaxErleHf);
> 
>-  // Verifies that the ERLE is not immediately decreased when the ERLE in the
>-  // data decreases.
>-  Y2.fill(0.1f * E2[0]);
>-  for (size_t k = 0; k < 98; ++k) {
>-    estimator.Update(X2, Y2, E2);
>+  FormNearendFrame(&X2, &E2, &Y2);
>+  // Verifies that the ERLE is not immediately decreased during nearend
>+  // activity.
>+  for (size_t k = 0; k < 50; ++k) {
>+    estimator.Update(X2, Y2, E2, true);
>   }
>-  VerifyErle(estimator.Erle(), estimator.ErleTimeDomain(), 8.f, 1.5f);
>+  VerifyErle(estimator.Erle(), std::pow(2.f, estimator.ErleTimeDomainLog2()),
>+             kMaxErleLf, kMaxErleHf);
>+}
>+
>+TEST(ErleEstimator, VerifyErleTrackingOnOnsets) {
>+  std::array<float, kFftLengthBy2Plus1> X2;
>+  std::array<float, kFftLengthBy2Plus1> E2;
>+  std::array<float, kFftLengthBy2Plus1> Y2;
>+
>+  ErleEstimator estimator(kMinErle, kMaxErleLf, kMaxErleHf);
> 
>-  // Verifies that the minimum ERLE is eventually achieved.
>-  for (size_t k = 0; k < 1000; ++k) {
>-    estimator.Update(X2, Y2, E2);
>+  for (size_t burst = 0; burst < 20; ++burst) {
>+    FormFarendFrame(&X2, &E2, &Y2, kTrueErleOnsets);
>+    for (size_t k = 0; k < 10; ++k) {
>+      estimator.Update(X2, Y2, E2, true);
>+    }
>+    FormFarendFrame(&X2, &E2, &Y2, kTrueErle);
>+    for (size_t k = 0; k < 200; ++k) {
>+      estimator.Update(X2, Y2, E2, true);
>+    }
>+    FormNearendFrame(&X2, &E2, &Y2);
>+    for (size_t k = 0; k < 300; ++k) {
>+      estimator.Update(X2, Y2, E2, true);
>+    }
>   }
>-  VerifyErle(estimator.Erle(), estimator.ErleTimeDomain(), 1.f, 1.f);
>+  VerifyErleBands(estimator.ErleOnsets(), kMinErle, kMinErle);
>+  FormNearendFrame(&X2, &E2, &Y2);
>+  for (size_t k = 0; k < 1000; k++) {
>+    estimator.Update(X2, Y2, E2, true);
>+  }
>+  // Verifies that during ne activity, Erle converges to the Erle for onsets.
>+  VerifyErle(estimator.Erle(), std::pow(2.f, estimator.ErleTimeDomainLog2()),
>+             kMinErle, kMinErle);
>+}
>+
>+TEST(ErleEstimator, VerifyNoErleUpdateDuringLowActivity) {
>+  std::array<float, kFftLengthBy2Plus1> X2;
>+  std::array<float, kFftLengthBy2Plus1> E2;
>+  std::array<float, kFftLengthBy2Plus1> Y2;
>+  ErleEstimator estimator(kMinErle, kMaxErleLf, kMaxErleHf);
> 
>   // Verifies that the ERLE estimate is is not updated for low-level render
>   // signals.
>   X2.fill(1000.f * 1000.f);
>   Y2.fill(10 * E2[0]);
>   for (size_t k = 0; k < 200; ++k) {
>-    estimator.Update(X2, Y2, E2);
>+    estimator.Update(X2, Y2, E2, true);
>   }
>-  VerifyErle(estimator.Erle(), estimator.ErleTimeDomain(), 1.f, 1.f);
>+  VerifyErle(estimator.Erle(), std::pow(2.f, estimator.ErleTimeDomainLog2()),
>+             kMinErle, kMinErle);
> }
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/fft_data.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/fft_data.h
>index 59511b56fe9..5e5adb62de9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/fft_data.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/fft_data.h
>@@ -11,7 +11,9 @@
> #ifndef MODULES_AUDIO_PROCESSING_AEC3_FFT_DATA_H_
> #define MODULES_AUDIO_PROCESSING_AEC3_FFT_DATA_H_
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>+// Defines WEBRTC_ARCH_X86_FAMILY, used below.
>+#include "rtc_base/system/arch.h"
>+
> #if defined(WEBRTC_ARCH_X86_FAMILY)
> #include <emmintrin.h>
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/fft_data_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/fft_data_unittest.cc
>index 8fc5ca7c764..0812fd64208 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/fft_data_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/fft_data_unittest.cc
>@@ -10,9 +10,9 @@
> 
> #include "modules/audio_processing/aec3/fft_data.h"
> 
>+#include "rtc_base/system/arch.h"
> #include "system_wrappers/include/cpu_features_wrapper.h"
> #include "test/gtest.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/filter_analyzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/filter_analyzer.cc
>new file mode 100644
>index 00000000000..790cb1e8b52
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/filter_analyzer.cc
>@@ -0,0 +1,171 @@
>+/*
>+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/aec3/filter_analyzer.h"
>+#include <math.h>
>+
>+#include <algorithm>
>+#include <array>
>+#include <numeric>
>+
>+#include "modules/audio_processing/aec3/aec3_common.h"
>+#include "modules/audio_processing/logging/apm_data_dumper.h"
>+#include "rtc_base/atomicops.h"
>+#include "rtc_base/checks.h"
>+#include "system_wrappers/include/field_trial.h"
>+
>+namespace webrtc {
>+namespace {
>+
>+size_t FindPeakIndex(rtc::ArrayView<const float> filter_time_domain) {
>+  size_t peak_index = 0;
>+  float max_h2 = filter_time_domain[0] * filter_time_domain[0];
>+  for (size_t k = 1; k < filter_time_domain.size(); ++k) {
>+    float tmp = filter_time_domain[k] * filter_time_domain[k];
>+    if (tmp > max_h2) {
>+      peak_index = k;
>+      max_h2 = tmp;
>+    }
>+  }
>+
>+  return peak_index;
>+}
>+
>+bool EnableFilterPreprocessing() {
>+  return !field_trial::IsEnabled(
>+      "WebRTC-Aec3FilterAnalyzerPreprocessorKillSwitch");
>+}
>+
>+}  // namespace
>+
>+int FilterAnalyzer::instance_count_ = 0;
>+
>+FilterAnalyzer::FilterAnalyzer(const EchoCanceller3Config& config)
>+    : data_dumper_(
>+          new ApmDataDumper(rtc::AtomicOps::Increment(&instance_count_))),
>+      use_preprocessed_filter_(EnableFilterPreprocessing()),
>+      bounded_erl_(config.ep_strength.bounded_erl),
>+      default_gain_(config.ep_strength.lf),
>+      active_render_threshold_(config.render_levels.active_render_limit *
>+                               config.render_levels.active_render_limit *
>+                               kFftLengthBy2),
>+      h_highpass_(GetTimeDomainLength(config.filter.main.length_blocks), 0.f),
>+      filter_length_blocks_(config.filter.main_initial.length_blocks) {
>+  Reset();
>+}
>+
>+void FilterAnalyzer::PreProcessFilter(
>+    rtc::ArrayView<const float> filter_time_domain) {
>+  RTC_DCHECK_GE(h_highpass_.capacity(), filter_time_domain.size());
>+  h_highpass_.resize(filter_time_domain.size());
>+  // Minimum phase high-pass filter with cutoff frequency at about 600 Hz.
>+  constexpr std::array<float, 3> h = {{0.7929742f, -0.36072128f, -0.47047766f}};
>+
>+  std::fill(h_highpass_.begin(), h_highpass_.end(), 0.f);
>+  for (size_t k = h.size() - 1; k < filter_time_domain.size(); ++k) {
>+    for (size_t j = 0; j < h.size(); ++j) {
>+      h_highpass_[k] += filter_time_domain[k - j] * h[j];
>+    }
>+  }
>+}
>+
>+FilterAnalyzer::~FilterAnalyzer() = default;
>+
>+void FilterAnalyzer::Reset() {
>+  delay_blocks_ = 0;
>+  consistent_estimate_ = false;
>+  blocks_since_reset_ = 0;
>+  consistent_estimate_ = false;
>+  consistent_estimate_counter_ = 0;
>+  consistent_delay_reference_ = -10;
>+  gain_ = default_gain_;
>+}
>+
>+void FilterAnalyzer::Update(
>+    rtc::ArrayView<const float> filter_time_domain,
>+    const std::vector<std::array<float, kFftLengthBy2Plus1>>&
>+        filter_freq_response,
>+    const RenderBuffer& render_buffer) {
>+  // Preprocess the filter to avoid issues with low-frequency components in the
>+  // filter.
>+  PreProcessFilter(filter_time_domain);
>+  data_dumper_->DumpRaw("aec3_linear_filter_processed_td", h_highpass_);
>+
>+  const auto& filter_to_analyze =
>+      use_preprocessed_filter_ ? h_highpass_ : filter_time_domain;
>+  RTC_DCHECK_EQ(filter_to_analyze.size(), filter_time_domain.size());
>+
>+  size_t peak_index = FindPeakIndex(filter_to_analyze);
>+  delay_blocks_ = peak_index >> kBlockSizeLog2;
>+  UpdateFilterGain(filter_to_analyze, peak_index);
>+
>+  float filter_floor = 0;
>+  float filter_secondary_peak = 0;
>+  size_t limit1 = peak_index < 64 ? 0 : peak_index - 64;
>+  size_t limit2 =
>+      peak_index > filter_to_analyze.size() - 129 ? 0 : peak_index + 128;
>+
>+  for (size_t k = 0; k < limit1; ++k) {
>+    float abs_h = fabsf(filter_to_analyze[k]);
>+    filter_floor += abs_h;
>+    filter_secondary_peak = std::max(filter_secondary_peak, abs_h);
>+  }
>+  for (size_t k = limit2; k < filter_to_analyze.size(); ++k) {
>+    float abs_h = fabsf(filter_to_analyze[k]);
>+    filter_floor += abs_h;
>+    filter_secondary_peak = std::max(filter_secondary_peak, abs_h);
>+  }
>+
>+  filter_floor /= (limit1 + filter_to_analyze.size() - limit2);
>+
>+  float abs_peak = fabsf(filter_to_analyze[peak_index]);
>+  bool significant_peak_index =
>+      abs_peak > 10.f * filter_floor && abs_peak > 2.f * filter_secondary_peak;
>+
>+  if (consistent_delay_reference_ != delay_blocks_ || !significant_peak_index) {
>+    consistent_estimate_counter_ = 0;
>+    consistent_delay_reference_ = delay_blocks_;
>+  } else {
>+    const auto& x = render_buffer.Block(-delay_blocks_)[0];
>+    const float x_energy =
>+        std::inner_product(x.begin(), x.end(), x.begin(), 0.f);
>+    const bool active_render_block = x_energy > active_render_threshold_;
>+
>+    if (active_render_block) {
>+      ++consistent_estimate_counter_;
>+    }
>+  }
>+
>+  consistent_estimate_ =
>+      consistent_estimate_counter_ > 1.5f * kNumBlocksPerSecond;
>+
>+  filter_length_blocks_ = filter_time_domain.size() * (1.f / kBlockSize);
>+}
>+
>+void FilterAnalyzer::UpdateFilterGain(
>+    rtc::ArrayView<const float> filter_time_domain,
>+    size_t peak_index) {
>+  bool sufficient_time_to_converge =
>+      ++blocks_since_reset_ > 5 * kNumBlocksPerSecond;
>+
>+  if (sufficient_time_to_converge && consistent_estimate_) {
>+    gain_ = fabsf(filter_time_domain[peak_index]);
>+  } else {
>+    if (gain_) {
>+      gain_ = std::max(gain_, fabsf(filter_time_domain[peak_index]));
>+    }
>+  }
>+
>+  if (bounded_erl_ && gain_) {
>+    gain_ = std::max(gain_, 0.01f);
>+  }
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/filter_analyzer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/filter_analyzer.h
>new file mode 100644
>index 00000000000..47e95336662
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/filter_analyzer.h
>@@ -0,0 +1,84 @@
>+/*
>+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_AEC3_FILTER_ANALYZER_H_
>+#define MODULES_AUDIO_PROCESSING_AEC3_FILTER_ANALYZER_H_
>+
>+#include <array>
>+#include <vector>
>+
>+#include "absl/types/optional.h"
>+#include "api/array_view.h"
>+#include "api/audio/echo_canceller3_config.h"
>+#include "modules/audio_processing/aec3/aec3_common.h"
>+#include "modules/audio_processing/aec3/cascaded_biquad_filter.h"
>+#include "modules/audio_processing/aec3/render_buffer.h"
>+#include "rtc_base/constructormagic.h"
>+
>+namespace webrtc {
>+
>+class ApmDataDumper;
>+
>+// Class for analyzing the properties of an adaptive filter.
>+class FilterAnalyzer {
>+ public:
>+  explicit FilterAnalyzer(const EchoCanceller3Config& config);
>+  ~FilterAnalyzer();
>+
>+  // Resets the analysis.
>+  void Reset();
>+
>+  // Updates the estimates with new input data.
>+  void Update(rtc::ArrayView<const float> filter_time_domain,
>+              const std::vector<std::array<float, kFftLengthBy2Plus1>>&
>+                  filter_freq_response,
>+              const RenderBuffer& render_buffer);
>+
>+  // Returns the delay of the filter in terms of blocks.
>+  int DelayBlocks() const { return delay_blocks_; }
>+
>+  // Returns whether the filter is consistent in the sense that it does not
>+  // change much over time.
>+  bool Consistent() const { return consistent_estimate_; }
>+
>+  // Returns the estimated filter gain.
>+  float Gain() const { return gain_; }
>+
>+  // Returns the number of blocks for the current used filter.
>+  float FilterLengthBlocks() const { return filter_length_blocks_; }
>+
>+  // Returns the preprocessed filter.
>+  rtc::ArrayView<const float> GetAdjustedFilter() const { return h_highpass_; }
>+
>+ private:
>+  void UpdateFilterGain(rtc::ArrayView<const float> filter_time_domain,
>+                        size_t max_index);
>+  void PreProcessFilter(rtc::ArrayView<const float> filter_time_domain);
>+
>+  static int instance_count_;
>+  std::unique_ptr<ApmDataDumper> data_dumper_;
>+  const bool use_preprocessed_filter_;
>+  const bool bounded_erl_;
>+  const float default_gain_;
>+  const float active_render_threshold_;
>+  std::vector<float> h_highpass_;
>+  int delay_blocks_ = 0;
>+  size_t blocks_since_reset_ = 0;
>+  bool consistent_estimate_ = false;
>+  size_t consistent_estimate_counter_ = 0;
>+  int consistent_delay_reference_ = -10;
>+  float gain_;
>+  int filter_length_blocks_;
>+  RTC_DISALLOW_COPY_AND_ASSIGN(FilterAnalyzer);
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_AEC3_FILTER_ANALYZER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/main_filter_update_gain.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/main_filter_update_gain.cc
>index 7f7e1b389c1..6d31ed00c27 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/main_filter_update_gain.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/main_filter_update_gain.cc
>@@ -22,6 +22,7 @@ namespace webrtc {
> namespace {
> 
> constexpr float kHErrorInitial = 10000.f;
>+constexpr float kHErrorGainChange = 10000.f;
> constexpr int kPoorExcitationCounterInitial = 1000;
> 
> }  // namespace
>@@ -29,22 +30,36 @@ constexpr int kPoorExcitationCounterInitial = 1000;
> int MainFilterUpdateGain::instance_count_ = 0;
> 
> MainFilterUpdateGain::MainFilterUpdateGain(
>-    const EchoCanceller3Config::Filter::MainConfiguration& config)
>+    const EchoCanceller3Config::Filter::MainConfiguration& config,
>+    size_t config_change_duration_blocks)
>     : data_dumper_(
>           new ApmDataDumper(rtc::AtomicOps::Increment(&instance_count_))),
>-      config_(config),
>+      config_change_duration_blocks_(
>+          static_cast<int>(config_change_duration_blocks)),
>       poor_excitation_counter_(kPoorExcitationCounterInitial) {
>+  SetConfig(config, true);
>   H_error_.fill(kHErrorInitial);
>+  RTC_DCHECK_LT(0, config_change_duration_blocks_);
>+  one_by_config_change_duration_blocks_ = 1.f / config_change_duration_blocks_;
> }
> 
> MainFilterUpdateGain::~MainFilterUpdateGain() {}
> 
> void MainFilterUpdateGain::HandleEchoPathChange(
>     const EchoPathVariability& echo_path_variability) {
>-  // TODO(peah): Add even-specific behavior.
>-  H_error_.fill(kHErrorInitial);
>-  poor_excitation_counter_ = kPoorExcitationCounterInitial;
>-  call_counter_ = 0;
>+  if (echo_path_variability.gain_change) {
>+    H_error_.fill(kHErrorGainChange);
>+  }
>+
>+  if (echo_path_variability.delay_change !=
>+      EchoPathVariability::DelayAdjustment::kNone) {
>+    H_error_.fill(kHErrorInitial);
>+  }
>+
>+  if (!echo_path_variability.gain_change) {
>+    poor_excitation_counter_ = kPoorExcitationCounterInitial;
>+    call_counter_ = 0;
>+  }
> }
> 
> void MainFilterUpdateGain::Compute(
>@@ -63,9 +78,10 @@ void MainFilterUpdateGain::Compute(
>   const size_t size_partitions = filter.SizePartitions();
>   auto X2 = render_power;
>   const auto& erl = filter.Erl();
>-
>   ++call_counter_;
> 
>+  UpdateCurrentConfig();
>+
>   if (render_signal_analyzer.PoorSignalExcitation()) {
>     poor_excitation_counter_ = 0;
>   }
>@@ -80,7 +96,7 @@ void MainFilterUpdateGain::Compute(
>     std::array<float, kFftLengthBy2Plus1> mu;
>     // mu = H_error / (0.5* H_error* X2 + n * E2).
>     for (size_t k = 0; k < kFftLengthBy2Plus1; ++k) {
>-      mu[k] = X2[k] > config_.noise_gate
>+      mu[k] = X2[k] > current_config_.noise_gate
>                   ? H_error_[k] / (0.5f * H_error_[k] * X2[k] +
>                                    size_partitions * E2_main[k])
>                   : 0.f;
>@@ -105,17 +121,47 @@ void MainFilterUpdateGain::Compute(
>   std::array<float, kFftLengthBy2Plus1> H_error_increase;
>   std::transform(E2_shadow.begin(), E2_shadow.end(), E2_main.begin(),
>                  H_error_increase.begin(), [&](float a, float b) {
>-                   return a >= b ? config_.leakage_converged
>-                                 : config_.leakage_diverged;
>+                   return a >= b ? current_config_.leakage_converged
>+                                 : current_config_.leakage_diverged;
>                  });
>   std::transform(erl.begin(), erl.end(), H_error_increase.begin(),
>                  H_error_increase.begin(), std::multiplies<float>());
>   std::transform(H_error_.begin(), H_error_.end(), H_error_increase.begin(),
>                  H_error_.begin(), [&](float a, float b) {
>-                   return std::max(a + b, config_.error_floor);
>+                   return std::max(a + b, current_config_.error_floor);
>                  });
> 
>   data_dumper_->DumpRaw("aec3_main_gain_H_error", H_error_);
> }
> 
>+void MainFilterUpdateGain::UpdateCurrentConfig() {
>+  RTC_DCHECK_GE(config_change_duration_blocks_, config_change_counter_);
>+  if (config_change_counter_ > 0) {
>+    if (--config_change_counter_ > 0) {
>+      auto average = [](float from, float to, float from_weight) {
>+        return from * from_weight + to * (1.f - from_weight);
>+      };
>+
>+      float change_factor =
>+          config_change_counter_ * one_by_config_change_duration_blocks_;
>+
>+      current_config_.leakage_converged =
>+          average(old_target_config_.leakage_converged,
>+                  target_config_.leakage_converged, change_factor);
>+      current_config_.leakage_diverged =
>+          average(old_target_config_.leakage_diverged,
>+                  target_config_.leakage_diverged, change_factor);
>+      current_config_.error_floor =
>+          average(old_target_config_.error_floor, target_config_.error_floor,
>+                  change_factor);
>+      current_config_.noise_gate =
>+          average(old_target_config_.noise_gate, target_config_.noise_gate,
>+                  change_factor);
>+    } else {
>+      current_config_ = old_target_config_ = target_config_;
>+    }
>+  }
>+  RTC_DCHECK_LE(0, config_change_counter_);
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/main_filter_update_gain.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/main_filter_update_gain.h
>index 802676868e1..525b52279c4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/main_filter_update_gain.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/main_filter_update_gain.h
>@@ -1,6 +1,6 @@
> /*
>  *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>-spect *
>+ *
>  *  Use of this source code is governed by a BSD-style license
>  *  that can be found in the LICENSE file in the root of the source
>  *  tree. An additional intellectual property rights grant can be found
>@@ -14,23 +14,24 @@ spect *
> #include <memory>
> #include <vector>
> 
>+#include "api/audio/echo_canceller3_config.h"
> #include "modules/audio_processing/aec3/adaptive_fir_filter.h"
> #include "modules/audio_processing/aec3/aec3_common.h"
> #include "modules/audio_processing/aec3/echo_path_variability.h"
> #include "modules/audio_processing/aec3/render_signal_analyzer.h"
> #include "modules/audio_processing/aec3/subtractor_output.h"
>-#include "modules/audio_processing/include/audio_processing.h"
> #include "rtc_base/constructormagic.h"
> 
> namespace webrtc {
> 
> class ApmDataDumper;
> 
>-// Provides functionality for computing the adaptive gain for the main filter.
>+// Provides functionality for  computing the adaptive gain for the main filter.
> class MainFilterUpdateGain {
>  public:
>   explicit MainFilterUpdateGain(
>-      const EchoCanceller3Config::Filter::MainConfiguration& config);
>+      const EchoCanceller3Config::Filter::MainConfiguration& config,
>+      size_t config_change_duration_blocks);
>   ~MainFilterUpdateGain();
> 
>   // Takes action in the case of a known echo path change.
>@@ -45,18 +46,34 @@ class MainFilterUpdateGain {
>                FftData* gain_fft);
> 
>   // Sets a new config.
>-  void SetConfig(
>-      const EchoCanceller3Config::Filter::MainConfiguration& config) {
>-    config_ = config;
>+  void SetConfig(const EchoCanceller3Config::Filter::MainConfiguration& config,
>+                 bool immediate_effect) {
>+    if (immediate_effect) {
>+      old_target_config_ = current_config_ = target_config_ = config;
>+      config_change_counter_ = 0;
>+    } else {
>+      old_target_config_ = current_config_;
>+      target_config_ = config;
>+      config_change_counter_ = config_change_duration_blocks_;
>+    }
>   }
> 
>  private:
>   static int instance_count_;
>   std::unique_ptr<ApmDataDumper> data_dumper_;
>-  EchoCanceller3Config::Filter::MainConfiguration config_;
>+  const int config_change_duration_blocks_;
>+  float one_by_config_change_duration_blocks_;
>+  EchoCanceller3Config::Filter::MainConfiguration current_config_;
>+  EchoCanceller3Config::Filter::MainConfiguration target_config_;
>+  EchoCanceller3Config::Filter::MainConfiguration old_target_config_;
>   std::array<float, kFftLengthBy2Plus1> H_error_;
>   size_t poor_excitation_counter_;
>   size_t call_counter_ = 0;
>+  int config_change_counter_ = 0;
>+
>+  // Updates the current config towards the target config.
>+  void UpdateCurrentConfig();
>+
>   RTC_DISALLOW_COPY_AND_ASSIGN(MainFilterUpdateGain);
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/main_filter_update_gain_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/main_filter_update_gain_unittest.cc
>index fbf444975ad..bed148a57db 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/main_filter_update_gain_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/main_filter_update_gain_unittest.cc
>@@ -45,14 +45,20 @@ void RunFilterUpdateTest(int num_blocks_to_process,
>   config.filter.main.length_blocks = filter_length_blocks;
>   config.filter.shadow.length_blocks = filter_length_blocks;
>   AdaptiveFirFilter main_filter(config.filter.main.length_blocks,
>+                                config.filter.main.length_blocks,
>+                                config.filter.config_change_duration_blocks,
>                                 DetectOptimization(), &data_dumper);
>   AdaptiveFirFilter shadow_filter(config.filter.shadow.length_blocks,
>+                                  config.filter.shadow.length_blocks,
>+                                  config.filter.config_change_duration_blocks,
>                                   DetectOptimization(), &data_dumper);
>   Aec3Fft fft;
>   std::array<float, kBlockSize> x_old;
>   x_old.fill(0.f);
>-  ShadowFilterUpdateGain shadow_gain(config.filter.shadow);
>-  MainFilterUpdateGain main_gain(config.filter.main);
>+  ShadowFilterUpdateGain shadow_gain(
>+      config.filter.shadow, config.filter.config_change_duration_blocks);
>+  MainFilterUpdateGain main_gain(config.filter.main,
>+                                 config.filter.config_change_duration_blocks);
>   Random random_generator(42U);
>   std::vector<std::vector<float>> x(3, std::vector<float>(kBlockSize, 0.f));
>   std::vector<float> y(kBlockSize, 0.f);
>@@ -61,7 +67,8 @@ void RunFilterUpdateTest(int num_blocks_to_process,
>   std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
>       RenderDelayBuffer::Create(config, 3));
>   AecState aec_state(config);
>-  RenderSignalAnalyzer render_signal_analyzer;
>+  RenderSignalAnalyzer render_signal_analyzer(config);
>+  absl::optional<DelayEstimate> delay_estimate;
>   std::array<float, kFftLength> s_scratch;
>   std::array<float, kBlockSize> s;
>   FftData S;
>@@ -107,7 +114,7 @@ void RunFilterUpdateTest(int num_blocks_to_process,
>     render_delay_buffer->PrepareCaptureProcessing();
> 
>     render_signal_analyzer.Update(*render_delay_buffer->GetRenderBuffer(),
>-                                  aec_state.FilterDelay());
>+                                  aec_state.FilterDelayBlocks());
> 
>     // Apply the main filter.
>     main_filter.Filter(*render_delay_buffer->GetRenderBuffer(), &S);
>@@ -154,10 +161,10 @@ void RunFilterUpdateTest(int num_blocks_to_process,
>     // Update the delay.
>     aec_state.HandleEchoPathChange(EchoPathVariability(
>         false, EchoPathVariability::DelayAdjustment::kNone, false));
>-    aec_state.Update(main_filter.FilterFrequencyResponse(),
>-                     main_filter.FilterImpulseResponse(), true,
>-                     *render_delay_buffer->GetRenderBuffer(), E2_main, Y2, s,
>-                     false);
>+    aec_state.Update(delay_estimate, main_filter.FilterFrequencyResponse(),
>+                     main_filter.FilterImpulseResponse(),
>+                     *render_delay_buffer->GetRenderBuffer(), E2_main, Y2,
>+                     output, y);
>   }
> 
>   std::copy(e_main.begin(), e_main.end(), e_last_block->begin());
>@@ -188,10 +195,13 @@ TEST(MainFilterUpdateGain, NullDataOutputGain) {
>   ApmDataDumper data_dumper(42);
>   EchoCanceller3Config config;
>   AdaptiveFirFilter filter(config.filter.main.length_blocks,
>+                           config.filter.main.length_blocks,
>+                           config.filter.config_change_duration_blocks,
>                            DetectOptimization(), &data_dumper);
>-  RenderSignalAnalyzer analyzer;
>+  RenderSignalAnalyzer analyzer(EchoCanceller3Config{});
>   SubtractorOutput output;
>-  MainFilterUpdateGain gain(config.filter.main);
>+  MainFilterUpdateGain gain(config.filter.main,
>+                            config.filter.config_change_duration_blocks);
>   std::array<float, kFftLengthBy2Plus1> render_power;
>   render_power.fill(0.f);
>   EXPECT_DEATH(
>@@ -212,7 +222,7 @@ TEST(MainFilterUpdateGain, GainCausesFilterToConverge) {
>       std::array<float, kBlockSize> y;
>       FftData G;
> 
>-      RunFilterUpdateTest(500, delay_samples, filter_length_blocks,
>+      RunFilterUpdateTest(600, delay_samples, filter_length_blocks,
>                           blocks_with_echo_path_changes, blocks_with_saturation,
>                           false, &e, &y, &G);
> 
>@@ -232,34 +242,34 @@ TEST(MainFilterUpdateGain, GainCausesFilterToConverge) {
> // Verifies that the magnitude of the gain on average decreases for a
> // persistently exciting signal.
> TEST(MainFilterUpdateGain, DecreasingGain) {
>-    std::vector<int> blocks_with_echo_path_changes;
>-    std::vector<int> blocks_with_saturation;
>-
>-    std::array<float, kBlockSize> e;
>-    std::array<float, kBlockSize> y;
>-    FftData G_a;
>-    FftData G_b;
>-    FftData G_c;
>-    std::array<float, kFftLengthBy2Plus1> G_a_power;
>-    std::array<float, kFftLengthBy2Plus1> G_b_power;
>-    std::array<float, kFftLengthBy2Plus1> G_c_power;
>-
>-    RunFilterUpdateTest(100, 65, 12, blocks_with_echo_path_changes,
>-                        blocks_with_saturation, false, &e, &y, &G_a);
>-    RunFilterUpdateTest(300, 65, 12, blocks_with_echo_path_changes,
>-                        blocks_with_saturation, false, &e, &y, &G_b);
>-    RunFilterUpdateTest(600, 65, 12, blocks_with_echo_path_changes,
>-                        blocks_with_saturation, false, &e, &y, &G_c);
>-
>-    G_a.Spectrum(Aec3Optimization::kNone, G_a_power);
>-    G_b.Spectrum(Aec3Optimization::kNone, G_b_power);
>-    G_c.Spectrum(Aec3Optimization::kNone, G_c_power);
>-
>-    EXPECT_GT(std::accumulate(G_a_power.begin(), G_a_power.end(), 0.),
>-              std::accumulate(G_b_power.begin(), G_b_power.end(), 0.));
>+  std::vector<int> blocks_with_echo_path_changes;
>+  std::vector<int> blocks_with_saturation;
> 
>-    EXPECT_GT(std::accumulate(G_b_power.begin(), G_b_power.end(), 0.),
>-              std::accumulate(G_c_power.begin(), G_c_power.end(), 0.));
>+  std::array<float, kBlockSize> e;
>+  std::array<float, kBlockSize> y;
>+  FftData G_a;
>+  FftData G_b;
>+  FftData G_c;
>+  std::array<float, kFftLengthBy2Plus1> G_a_power;
>+  std::array<float, kFftLengthBy2Plus1> G_b_power;
>+  std::array<float, kFftLengthBy2Plus1> G_c_power;
>+
>+  RunFilterUpdateTest(250, 65, 12, blocks_with_echo_path_changes,
>+                      blocks_with_saturation, false, &e, &y, &G_a);
>+  RunFilterUpdateTest(500, 65, 12, blocks_with_echo_path_changes,
>+                      blocks_with_saturation, false, &e, &y, &G_b);
>+  RunFilterUpdateTest(750, 65, 12, blocks_with_echo_path_changes,
>+                      blocks_with_saturation, false, &e, &y, &G_c);
>+
>+  G_a.Spectrum(Aec3Optimization::kNone, G_a_power);
>+  G_b.Spectrum(Aec3Optimization::kNone, G_b_power);
>+  G_c.Spectrum(Aec3Optimization::kNone, G_c_power);
>+
>+  EXPECT_GT(std::accumulate(G_a_power.begin(), G_a_power.end(), 0.),
>+            std::accumulate(G_b_power.begin(), G_b_power.end(), 0.));
>+
>+  EXPECT_GT(std::accumulate(G_b_power.begin(), G_b_power.end(), 0.),
>+            std::accumulate(G_c_power.begin(), G_c_power.end(), 0.));
> }
> 
> // Verifies that the gain is zero when there is saturation and that the internal
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/matched_filter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/matched_filter.cc
>index 2b2e9d6bfab..7486ddfa080 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/matched_filter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/matched_filter.cc
>@@ -9,17 +9,19 @@
>  */
> #include "modules/audio_processing/aec3/matched_filter.h"
> 
>+// Defines WEBRTC_ARCH_X86_FAMILY, used below.
>+#include "rtc_base/system/arch.h"
>+
> #if defined(WEBRTC_HAS_NEON)
> #include <arm_neon.h>
> #endif
>-#include "typedefs.h"  // NOLINT(build/include)
> #if defined(WEBRTC_ARCH_X86_FAMILY)
> #include <emmintrin.h>
> #endif
> #include <algorithm>
> #include <numeric>
> 
>-#include "modules/audio_processing/include/audio_processing.h"
>+#include "api/audio/echo_canceller3_config.h"
> #include "modules/audio_processing/logging/apm_data_dumper.h"
> #include "rtc_base/logging.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/matched_filter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/matched_filter.h
>index c9bdc462e88..1c06b5eb2ef 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/matched_filter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/matched_filter.h
>@@ -15,10 +15,12 @@
> #include <memory>
> #include <vector>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
>+#include "api/array_view.h"
> #include "modules/audio_processing/aec3/aec3_common.h"
> #include "modules/audio_processing/aec3/downsampled_render_buffer.h"
> #include "rtc_base/constructormagic.h"
>+#include "rtc_base/system/arch.h"
> 
> namespace webrtc {
> namespace aec3 {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/matched_filter_lag_aggregator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/matched_filter_lag_aggregator.cc
>index 9041924b34c..9cd2eb3d7b2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/matched_filter_lag_aggregator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/matched_filter_lag_aggregator.cc
>@@ -30,7 +30,7 @@ void MatchedFilterLagAggregator::Reset() {
>   significant_candidate_found_ = false;
> }
> 
>-rtc::Optional<DelayEstimate> MatchedFilterLagAggregator::Aggregate(
>+absl::optional<DelayEstimate> MatchedFilterLagAggregator::Aggregate(
>     rtc::ArrayView<const MatchedFilter::LagEstimate> lag_estimates) {
>   // Choose the strongest lag estimate as the best one.
>   float best_accuracy = 0.f;
>@@ -70,11 +70,9 @@ rtc::Optional<DelayEstimate> MatchedFilterLagAggregator::Aggregate(
>     if (histogram_[candidate] > 25) {
>       significant_candidate_found_ = true;
>       return DelayEstimate(DelayEstimate::Quality::kRefined, candidate);
>-    } else if (!significant_candidate_found_) {
>-      return DelayEstimate(DelayEstimate::Quality::kCoarse, candidate);
>     }
>   }
>-  return rtc::nullopt;
>+  return absl::nullopt;
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/matched_filter_lag_aggregator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/matched_filter_lag_aggregator.h
>index 86968bd24f8..fddcfbff798 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/matched_filter_lag_aggregator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/matched_filter_lag_aggregator.h
>@@ -13,7 +13,7 @@
> 
> #include <vector>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> #include "modules/audio_processing/aec3/delay_estimate.h"
> #include "modules/audio_processing/aec3/matched_filter.h"
> #include "rtc_base/constructormagic.h"
>@@ -33,7 +33,7 @@ class MatchedFilterLagAggregator {
>   void Reset();
> 
>   // Aggregates the provided lag estimates.
>-  rtc::Optional<DelayEstimate> Aggregate(
>+  absl::optional<DelayEstimate> Aggregate(
>       rtc::ArrayView<const MatchedFilter::LagEstimate> lag_estimates);
> 
>  private:
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/matched_filter_lag_aggregator_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/matched_filter_lag_aggregator_unittest.cc
>index ce303d4e382..cea5f13ce74 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/matched_filter_lag_aggregator_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/matched_filter_lag_aggregator_unittest.cc
>@@ -6,7 +6,7 @@
>  *  tree. An additional intellectual property rights grant can be found
>  *  in the file PATENTS.  All contributing project authors may
>  *  be found in the AUTHORS file in the root of the source tree.
>-  */
>+ */
> 
> #include "modules/audio_processing/aec3/matched_filter_lag_aggregator.h"
> 
>@@ -22,7 +22,7 @@
> namespace webrtc {
> namespace {
> 
>-constexpr size_t kNumLagsBeforeDetection = 25;
>+constexpr size_t kNumLagsBeforeDetection = 26;
> 
> }  // namespace
> 
>@@ -37,10 +37,10 @@ TEST(MatchedFilterLagAggregator, MostAccurateLagChosen) {
>   lag_estimates[1] = MatchedFilter::LagEstimate(0.5f, true, kLag2, true);
> 
>   for (size_t k = 0; k < kNumLagsBeforeDetection; ++k) {
>-    EXPECT_TRUE(aggregator.Aggregate(lag_estimates));
>+    aggregator.Aggregate(lag_estimates);
>   }
> 
>-  rtc::Optional<DelayEstimate> aggregated_lag =
>+  absl::optional<DelayEstimate> aggregated_lag =
>       aggregator.Aggregate(lag_estimates);
>   EXPECT_TRUE(aggregated_lag);
>   EXPECT_EQ(kLag1, aggregated_lag->delay);
>@@ -68,7 +68,7 @@ TEST(MatchedFilterLagAggregator,
>   std::vector<MatchedFilter::LagEstimate> lag_estimates(1);
>   MatchedFilterLagAggregator aggregator(&data_dumper, 100);
> 
>-  rtc::Optional<DelayEstimate> aggregated_lag;
>+  absl::optional<DelayEstimate> aggregated_lag;
>   for (size_t k = 0; k < kNumLagsBeforeDetection; ++k) {
>     lag_estimates[0] = MatchedFilter::LagEstimate(1.f, true, 10, true);
>     aggregated_lag = aggregator.Aggregate(lag_estimates);
>@@ -98,7 +98,7 @@ TEST(MatchedFilterLagAggregator,
>   MatchedFilterLagAggregator aggregator(&data_dumper, kLag);
>   for (size_t k = 0; k < kNumLagsBeforeDetection * 10; ++k) {
>     lag_estimates[0] = MatchedFilter::LagEstimate(1.f, true, kLag, false);
>-    rtc::Optional<DelayEstimate> aggregated_lag =
>+    absl::optional<DelayEstimate> aggregated_lag =
>         aggregator.Aggregate(lag_estimates);
>     EXPECT_FALSE(aggregated_lag);
>     EXPECT_EQ(kLag, aggregated_lag->delay);
>@@ -114,7 +114,7 @@ TEST(MatchedFilterLagAggregator, DISABLED_PersistentAggregatedLag) {
>   ApmDataDumper data_dumper(0);
>   std::vector<MatchedFilter::LagEstimate> lag_estimates(1);
>   MatchedFilterLagAggregator aggregator(&data_dumper, std::max(kLag1, kLag2));
>-  rtc::Optional<DelayEstimate> aggregated_lag;
>+  absl::optional<DelayEstimate> aggregated_lag;
>   for (size_t k = 0; k < kNumLagsBeforeDetection; ++k) {
>     lag_estimates[0] = MatchedFilter::LagEstimate(1.f, true, kLag1, true);
>     aggregated_lag = aggregator.Aggregate(lag_estimates);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/matched_filter_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/matched_filter_unittest.cc
>index e30c78caa64..c7dc211baa2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/matched_filter_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/matched_filter_unittest.cc
>@@ -6,11 +6,13 @@
>  *  tree. An additional intellectual property rights grant can be found
>  *  in the file PATENTS.  All contributing project authors may
>  *  be found in the AUTHORS file in the root of the source tree.
>-  */
>+ */
> 
> #include "modules/audio_processing/aec3/matched_filter.h"
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>+// Defines WEBRTC_ARCH_X86_FAMILY, used below.
>+#include "rtc_base/system/arch.h"
>+
> #if defined(WEBRTC_ARCH_X86_FAMILY)
> #include <emmintrin.h>
> #endif
>@@ -183,7 +185,7 @@ TEST(MatchedFilter, LagEstimation) {
>       auto lag_estimates = filter.GetLagEstimates();
> 
>       // Find which lag estimate should be the most accurate.
>-      rtc::Optional<size_t> expected_most_accurate_lag_estimate;
>+      absl::optional<size_t> expected_most_accurate_lag_estimate;
>       size_t alignment_shift_sub_blocks = 0;
>       for (size_t k = 0; k < config.delay.num_filters; ++k) {
>         if ((alignment_shift_sub_blocks + 3 * kWindowSizeSubBlocks / 4) *
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/mock/mock_block_processor.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/mock/mock_block_processor.cc
>new file mode 100644
>index 00000000000..c5c33dbd687
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/mock/mock_block_processor.cc
>@@ -0,0 +1,20 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/aec3/mock/mock_block_processor.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+MockBlockProcessor::MockBlockProcessor() = default;
>+MockBlockProcessor::~MockBlockProcessor() = default;
>+
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/mock/mock_block_processor.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/mock/mock_block_processor.h
>index 5fff4561858..85b88f7fc79 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/mock/mock_block_processor.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/mock/mock_block_processor.h
>@@ -21,7 +21,8 @@ namespace test {
> 
> class MockBlockProcessor : public BlockProcessor {
>  public:
>-  virtual ~MockBlockProcessor() {}
>+  MockBlockProcessor();
>+  virtual ~MockBlockProcessor();
> 
>   MOCK_METHOD3(ProcessCapture,
>                void(bool level_change,
>@@ -31,6 +32,7 @@ class MockBlockProcessor : public BlockProcessor {
>                void(const std::vector<std::vector<float>>& block));
>   MOCK_METHOD1(UpdateEchoLeakageStatus, void(bool leakage_detected));
>   MOCK_CONST_METHOD1(GetMetrics, void(EchoControl::Metrics* metrics));
>+  MOCK_METHOD1(SetAudioBufferDelay, void(size_t delay_ms));
> };
> 
> }  // namespace test
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/mock/mock_echo_remover.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/mock/mock_echo_remover.cc
>new file mode 100644
>index 00000000000..b903bf07855
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/mock/mock_echo_remover.cc
>@@ -0,0 +1,20 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/aec3/mock/mock_echo_remover.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+MockEchoRemover::MockEchoRemover() = default;
>+MockEchoRemover::~MockEchoRemover() = default;
>+
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/mock/mock_echo_remover.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/mock/mock_echo_remover.h
>index 11ced154e02..5faea265133 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/mock/mock_echo_remover.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/mock/mock_echo_remover.h
>@@ -13,7 +13,7 @@
> 
> #include <vector>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> #include "modules/audio_processing/aec3/echo_path_variability.h"
> #include "modules/audio_processing/aec3/echo_remover.h"
> #include "modules/audio_processing/aec3/render_buffer.h"
>@@ -24,14 +24,16 @@ namespace test {
> 
> class MockEchoRemover : public EchoRemover {
>  public:
>-  virtual ~MockEchoRemover() = default;
>+  MockEchoRemover();
>+  virtual ~MockEchoRemover();
> 
>-  MOCK_METHOD4(ProcessCapture,
>-               void(const EchoPathVariability& echo_path_variability,
>+  MOCK_METHOD5(ProcessCapture,
>+               void(EchoPathVariability echo_path_variability,
>                     bool capture_signal_saturation,
>+                    const absl::optional<DelayEstimate>& delay_estimate,
>                     RenderBuffer* render_buffer,
>                     std::vector<std::vector<float>>* capture));
>-
>+  MOCK_CONST_METHOD0(Delay, absl::optional<int>());
>   MOCK_METHOD1(UpdateEchoLeakageStatus, void(bool leakage_detected));
>   MOCK_CONST_METHOD1(GetMetrics, void(EchoControl::Metrics* metrics));
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/mock/mock_render_delay_buffer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/mock/mock_render_delay_buffer.cc
>new file mode 100644
>index 00000000000..20214cbc2d2
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/mock/mock_render_delay_buffer.cc
>@@ -0,0 +1,35 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/aec3/mock/mock_render_delay_buffer.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+MockRenderDelayBuffer::MockRenderDelayBuffer(int sample_rate_hz)
>+    : block_buffer_(GetRenderDelayBufferSize(4, 4, 12),
>+                    NumBandsForRate(sample_rate_hz),
>+                    kBlockSize),
>+      spectrum_buffer_(block_buffer_.buffer.size(), kFftLengthBy2Plus1),
>+      fft_buffer_(block_buffer_.buffer.size()),
>+      render_buffer_(&block_buffer_, &spectrum_buffer_, &fft_buffer_),
>+      downsampled_render_buffer_(GetDownSampledBufferSize(4, 4)) {
>+  ON_CALL(*this, GetRenderBuffer())
>+      .WillByDefault(
>+          testing::Invoke(this, &MockRenderDelayBuffer::FakeGetRenderBuffer));
>+  ON_CALL(*this, GetDownsampledRenderBuffer())
>+      .WillByDefault(testing::Invoke(
>+          this, &MockRenderDelayBuffer::FakeGetDownsampledRenderBuffer));
>+}
>+
>+MockRenderDelayBuffer::~MockRenderDelayBuffer() = default;
>+
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/mock/mock_render_delay_buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/mock/mock_render_delay_buffer.h
>index 1ed2b40e0c8..ba5e98cb9a9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/mock/mock_render_delay_buffer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/mock/mock_render_delay_buffer.h
>@@ -24,22 +24,8 @@ namespace test {
> 
> class MockRenderDelayBuffer : public RenderDelayBuffer {
>  public:
>-  explicit MockRenderDelayBuffer(int sample_rate_hz)
>-      : block_buffer_(GetRenderDelayBufferSize(4, 4, 12),
>-                      NumBandsForRate(sample_rate_hz),
>-                      kBlockSize),
>-        spectrum_buffer_(block_buffer_.buffer.size(), kFftLengthBy2Plus1),
>-        fft_buffer_(block_buffer_.buffer.size()),
>-        render_buffer_(&block_buffer_, &spectrum_buffer_, &fft_buffer_),
>-        downsampled_render_buffer_(GetDownSampledBufferSize(4, 4)) {
>-    ON_CALL(*this, GetRenderBuffer())
>-        .WillByDefault(
>-            testing::Invoke(this, &MockRenderDelayBuffer::FakeGetRenderBuffer));
>-    ON_CALL(*this, GetDownsampledRenderBuffer())
>-        .WillByDefault(testing::Invoke(
>-            this, &MockRenderDelayBuffer::FakeGetDownsampledRenderBuffer));
>-  }
>-  virtual ~MockRenderDelayBuffer() = default;
>+  explicit MockRenderDelayBuffer(int sample_rate_hz);
>+  virtual ~MockRenderDelayBuffer();
> 
>   MOCK_METHOD0(Reset, void());
>   MOCK_METHOD1(Insert,
>@@ -47,12 +33,13 @@ class MockRenderDelayBuffer : public RenderDelayBuffer {
>                    const std::vector<std::vector<float>>& block));
>   MOCK_METHOD0(PrepareCaptureProcessing, RenderDelayBuffer::BufferingEvent());
>   MOCK_METHOD1(SetDelay, bool(size_t delay));
>-  MOCK_CONST_METHOD0(Delay, rtc::Optional<size_t>());
>+  MOCK_CONST_METHOD0(Delay, size_t());
>   MOCK_CONST_METHOD0(MaxDelay, size_t());
>   MOCK_METHOD0(GetRenderBuffer, RenderBuffer*());
>   MOCK_CONST_METHOD0(GetDownsampledRenderBuffer,
>                      const DownsampledRenderBuffer&());
>   MOCK_CONST_METHOD1(CausalDelay, bool(size_t delay));
>+  MOCK_METHOD1(SetAudioBufferDelay, void(size_t delay_ms));
> 
>  private:
>   RenderBuffer* FakeGetRenderBuffer() { return &render_buffer_; }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/mock/mock_render_delay_controller.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/mock/mock_render_delay_controller.cc
>new file mode 100644
>index 00000000000..4ae2af96bf1
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/mock/mock_render_delay_controller.cc
>@@ -0,0 +1,20 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/aec3/mock/mock_render_delay_controller.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+MockRenderDelayController::MockRenderDelayController() = default;
>+MockRenderDelayController::~MockRenderDelayController() = default;
>+
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/mock/mock_render_delay_controller.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/mock/mock_render_delay_controller.h
>index 8fb7a8e900c..ed5971cda0d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/mock/mock_render_delay_controller.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/mock/mock_render_delay_controller.h
>@@ -11,8 +11,8 @@
> #ifndef MODULES_AUDIO_PROCESSING_AEC3_MOCK_MOCK_RENDER_DELAY_CONTROLLER_H_
> #define MODULES_AUDIO_PROCESSING_AEC3_MOCK_MOCK_RENDER_DELAY_CONTROLLER_H_
> 
>+#include "absl/types/optional.h"
> #include "api/array_view.h"
>-#include "api/optional.h"
> #include "modules/audio_processing/aec3/downsampled_render_buffer.h"
> #include "modules/audio_processing/aec3/render_delay_controller.h"
> #include "test/gmock.h"
>@@ -22,14 +22,17 @@ namespace test {
> 
> class MockRenderDelayController : public RenderDelayController {
>  public:
>-  virtual ~MockRenderDelayController() = default;
>+  MockRenderDelayController();
>+  virtual ~MockRenderDelayController();
> 
>   MOCK_METHOD0(Reset, void());
>   MOCK_METHOD0(LogRenderCall, void());
>-  MOCK_METHOD2(
>-      GetDelay,
>-      rtc::Optional<DelayEstimate>(const DownsampledRenderBuffer& render_buffer,
>-                                   rtc::ArrayView<const float> capture));
>+  MOCK_METHOD4(GetDelay,
>+               absl::optional<DelayEstimate>(
>+                   const DownsampledRenderBuffer& render_buffer,
>+                   size_t render_delay_buffer_delay,
>+                   const absl::optional<int>& echo_remover_delay,
>+                   rtc::ArrayView<const float> capture));
> };
> 
> }  // namespace test
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/moving_average.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/moving_average.cc
>new file mode 100644
>index 00000000000..e9d64e6b326
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/moving_average.cc
>@@ -0,0 +1,58 @@
>+
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/aec3/moving_average.h"
>+
>+#include <algorithm>
>+#include <functional>
>+
>+namespace webrtc {
>+namespace aec3 {
>+
>+MovingAverage::MovingAverage(size_t num_elem, size_t mem_len)
>+    : num_elem_(num_elem),
>+      mem_len_(mem_len - 1),
>+      scaling_(1.0f / static_cast<float>(mem_len)),
>+      memory_(num_elem * mem_len_, 0.f),
>+      mem_index_(0) {
>+  RTC_DCHECK(num_elem_ > 0);
>+  RTC_DCHECK(mem_len > 0);
>+}
>+
>+MovingAverage::~MovingAverage() = default;
>+
>+void MovingAverage::Average(rtc::ArrayView<const float> input,
>+                            rtc::ArrayView<float> output) {
>+  RTC_DCHECK(input.size() == num_elem_);
>+  RTC_DCHECK(output.size() == num_elem_);
>+
>+  // Sum all contributions.
>+  std::copy(input.begin(), input.end(), output.begin());
>+  for (auto i = memory_.begin(); i < memory_.end(); i += num_elem_) {
>+    std::transform(i, i + num_elem_, output.begin(), output.begin(),
>+                   std::plus<float>());
>+  }
>+
>+  // Divide by mem_len_.
>+  for (float& o : output) {
>+    o *= scaling_;
>+  }
>+
>+  // Update memory.
>+  if (mem_len_ > 0) {
>+    std::copy(input.begin(), input.end(),
>+              memory_.begin() + mem_index_ * num_elem_);
>+    mem_index_ = (mem_index_ + 1) % mem_len_;
>+  }
>+}
>+
>+}  // namespace aec3
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/moving_average.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/moving_average.h
>new file mode 100644
>index 00000000000..94497d782c8
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/moving_average.h
>@@ -0,0 +1,43 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_AEC3_MOVING_AVERAGE_H_
>+#define MODULES_AUDIO_PROCESSING_AEC3_MOVING_AVERAGE_H_
>+
>+#include <vector>
>+
>+#include "api/array_view.h"
>+
>+namespace webrtc {
>+namespace aec3 {
>+
>+class MovingAverage {
>+ public:
>+  // Creates an instance of MovingAverage that accepts inputs of length num_elem
>+  // and averages over mem_len inputs.
>+  MovingAverage(size_t num_elem, size_t mem_len);
>+  ~MovingAverage();
>+
>+  // Computes the average of input and mem_len-1 previous inputs and stores the
>+  // result in output.
>+  void Average(rtc::ArrayView<const float> input, rtc::ArrayView<float> output);
>+
>+ private:
>+  const size_t num_elem_;
>+  const size_t mem_len_;
>+  const float scaling_;
>+  std::vector<float> memory_;
>+  size_t mem_index_;
>+};
>+
>+}  // namespace aec3
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_AEC3_MOVING_AVERAGE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/moving_average_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/moving_average_unittest.cc
>new file mode 100644
>index 00000000000..05542d1eef9
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/moving_average_unittest.cc
>@@ -0,0 +1,88 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/aec3/moving_average.h"
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+
>+TEST(MovingAverage, Average) {
>+  constexpr size_t num_elem = 4;
>+  constexpr size_t mem_len = 3;
>+  constexpr float e = 1e-6f;
>+  aec3::MovingAverage ma(num_elem, mem_len);
>+  std::array<float, num_elem> data1 = {1, 2, 3, 4};
>+  std::array<float, num_elem> data2 = {5, 1, 9, 7};
>+  std::array<float, num_elem> data3 = {3, 3, 5, 6};
>+  std::array<float, num_elem> data4 = {8, 4, 2, 1};
>+  std::array<float, num_elem> output;
>+
>+  ma.Average(data1, output);
>+  EXPECT_NEAR(output[0], data1[0] / 3.0f, e);
>+  EXPECT_NEAR(output[1], data1[1] / 3.0f, e);
>+  EXPECT_NEAR(output[2], data1[2] / 3.0f, e);
>+  EXPECT_NEAR(output[3], data1[3] / 3.0f, e);
>+
>+  ma.Average(data2, output);
>+  EXPECT_NEAR(output[0], (data1[0] + data2[0]) / 3.0f, e);
>+  EXPECT_NEAR(output[1], (data1[1] + data2[1]) / 3.0f, e);
>+  EXPECT_NEAR(output[2], (data1[2] + data2[2]) / 3.0f, e);
>+  EXPECT_NEAR(output[3], (data1[3] + data2[3]) / 3.0f, e);
>+
>+  ma.Average(data3, output);
>+  EXPECT_NEAR(output[0], (data1[0] + data2[0] + data3[0]) / 3.0f, e);
>+  EXPECT_NEAR(output[1], (data1[1] + data2[1] + data3[1]) / 3.0f, e);
>+  EXPECT_NEAR(output[2], (data1[2] + data2[2] + data3[2]) / 3.0f, e);
>+  EXPECT_NEAR(output[3], (data1[3] + data2[3] + data3[3]) / 3.0f, e);
>+
>+  ma.Average(data4, output);
>+  EXPECT_NEAR(output[0], (data2[0] + data3[0] + data4[0]) / 3.0f, e);
>+  EXPECT_NEAR(output[1], (data2[1] + data3[1] + data4[1]) / 3.0f, e);
>+  EXPECT_NEAR(output[2], (data2[2] + data3[2] + data4[2]) / 3.0f, e);
>+  EXPECT_NEAR(output[3], (data2[3] + data3[3] + data4[3]) / 3.0f, e);
>+}
>+
>+TEST(MovingAverage, PassThrough) {
>+  constexpr size_t num_elem = 4;
>+  constexpr size_t mem_len = 1;
>+  constexpr float e = 1e-6f;
>+  aec3::MovingAverage ma(num_elem, mem_len);
>+  std::array<float, num_elem> data1 = {1, 2, 3, 4};
>+  std::array<float, num_elem> data2 = {5, 1, 9, 7};
>+  std::array<float, num_elem> data3 = {3, 3, 5, 6};
>+  std::array<float, num_elem> data4 = {8, 4, 2, 1};
>+  std::array<float, num_elem> output;
>+
>+  ma.Average(data1, output);
>+  EXPECT_NEAR(output[0], data1[0], e);
>+  EXPECT_NEAR(output[1], data1[1], e);
>+  EXPECT_NEAR(output[2], data1[2], e);
>+  EXPECT_NEAR(output[3], data1[3], e);
>+
>+  ma.Average(data2, output);
>+  EXPECT_NEAR(output[0], data2[0], e);
>+  EXPECT_NEAR(output[1], data2[1], e);
>+  EXPECT_NEAR(output[2], data2[2], e);
>+  EXPECT_NEAR(output[3], data2[3], e);
>+
>+  ma.Average(data3, output);
>+  EXPECT_NEAR(output[0], data3[0], e);
>+  EXPECT_NEAR(output[1], data3[1], e);
>+  EXPECT_NEAR(output[2], data3[2], e);
>+  EXPECT_NEAR(output[3], data3[3], e);
>+
>+  ma.Average(data4, output);
>+  EXPECT_NEAR(output[0], data4[0], e);
>+  EXPECT_NEAR(output[1], data4[1], e);
>+  EXPECT_NEAR(output[2], data4[2], e);
>+  EXPECT_NEAR(output[3], data4[3], e);
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/output_selector.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/output_selector.cc
>deleted file mode 100644
>index 4f547d98d91..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/output_selector.cc
>+++ /dev/null
>@@ -1,59 +0,0 @@
>-/*
>- *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "modules/audio_processing/aec3/output_selector.h"
>-
>-#include <algorithm>
>-#include <numeric>
>-
>-#include "rtc_base/checks.h"
>-
>-namespace webrtc {
>-namespace {
>-
>-// Performs the transition between the signals in a smooth manner.
>-void SmoothFrameTransition(bool from_y_to_e,
>-                           rtc::ArrayView<const float> e,
>-                           rtc::ArrayView<float> y) {
>-  RTC_DCHECK_LT(0u, e.size());
>-  RTC_DCHECK_EQ(y.size(), e.size());
>-
>-  const float change_factor = (from_y_to_e ? 1.f : -1.f) / e.size();
>-  float averaging = from_y_to_e ? 0.f : 1.f;
>-  for (size_t k = 0; k < e.size(); ++k) {
>-    y[k] += averaging * (e[k] - y[k]);
>-    averaging += change_factor;
>-  }
>-  RTC_DCHECK_EQ(from_y_to_e ? 1.f : 0.f, averaging);
>-}
>-
>-}  // namespace
>-
>-OutputSelector::OutputSelector() = default;
>-
>-OutputSelector::~OutputSelector() = default;
>-
>-void OutputSelector::FormLinearOutput(
>-    bool use_subtractor_output,
>-    rtc::ArrayView<const float> subtractor_output,
>-    rtc::ArrayView<float> capture) {
>-  RTC_DCHECK_EQ(subtractor_output.size(), capture.size());
>-  rtc::ArrayView<const float>& e_main = subtractor_output;
>-  rtc::ArrayView<float> y = capture;
>-
>-  if (use_subtractor_output != use_subtractor_output_) {
>-    use_subtractor_output_ = use_subtractor_output;
>-    SmoothFrameTransition(use_subtractor_output_, e_main, y);
>-  } else if (use_subtractor_output_) {
>-    std::copy(e_main.begin(), e_main.end(), y.begin());
>-  }
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/output_selector.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/output_selector.h
>deleted file mode 100644
>index a406c61745a..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/output_selector.h
>+++ /dev/null
>@@ -1,41 +0,0 @@
>-/*
>- *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MODULES_AUDIO_PROCESSING_AEC3_OUTPUT_SELECTOR_H_
>-#define MODULES_AUDIO_PROCESSING_AEC3_OUTPUT_SELECTOR_H_
>-
>-#include "api/array_view.h"
>-#include "rtc_base/constructormagic.h"
>-
>-namespace webrtc {
>-
>-// Performs the selection between which of the linear aec output and the
>-// microphone signal should be used as the echo suppressor output.
>-class OutputSelector {
>- public:
>-  OutputSelector();
>-  ~OutputSelector();
>-
>-  // Forms the most appropriate output signal.
>-  void FormLinearOutput(bool use_subtractor_output,
>-                        rtc::ArrayView<const float> subtractor_output,
>-                        rtc::ArrayView<float> capture);
>-
>-  // Returns true if the linear aec output is the one used.
>-  bool UseSubtractorOutput() const { return use_subtractor_output_; }
>-
>- private:
>-  bool use_subtractor_output_ = false;
>-  RTC_DISALLOW_COPY_AND_ASSIGN(OutputSelector);
>-};
>-
>-}  // namespace webrtc
>-
>-#endif  // MODULES_AUDIO_PROCESSING_AEC3_OUTPUT_SELECTOR_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/output_selector_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/output_selector_unittest.cc
>deleted file mode 100644
>index c7add1c838d..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/output_selector_unittest.cc
>+++ /dev/null
>@@ -1,69 +0,0 @@
>-/*
>- *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "modules/audio_processing/aec3/output_selector.h"
>-
>-#include <algorithm>
>-#include <array>
>-
>-#include "modules/audio_processing/aec3/aec3_common.h"
>-#include "test/gtest.h"
>-
>-namespace webrtc {
>-
>-// Verifies that the switching between the signals in the output works as
>-// intended.
>-TEST(OutputSelector, ProperSwitching) {
>-  OutputSelector selector;
>-
>-  std::array<float, kBlockSize> y;
>-  std::array<float, kBlockSize> e;
>-  std::array<float, kBlockSize> e_ref;
>-  std::array<float, kBlockSize> y_ref;
>-  auto init_blocks = [](std::array<float, kBlockSize>* e,
>-                        std::array<float, kBlockSize>* y) {
>-    e->fill(10.f);
>-    y->fill(20.f);
>-  };
>-
>-  init_blocks(&e_ref, &y_ref);
>-
>-  init_blocks(&e, &y);
>-  selector.FormLinearOutput(false, e, y);
>-  EXPECT_EQ(y_ref, y);
>-
>-  init_blocks(&e, &y);
>-  selector.FormLinearOutput(true, e, y);
>-  EXPECT_NE(e_ref, y);
>-  EXPECT_NE(y_ref, y);
>-
>-  init_blocks(&e, &y);
>-  selector.FormLinearOutput(true, e, y);
>-  EXPECT_EQ(e_ref, y);
>-
>-  init_blocks(&e, &y);
>-  selector.FormLinearOutput(true, e, y);
>-  EXPECT_EQ(e_ref, y);
>-
>-  init_blocks(&e, &y);
>-  selector.FormLinearOutput(false, e, y);
>-  EXPECT_NE(e_ref, y);
>-  EXPECT_NE(y_ref, y);
>-
>-  init_blocks(&e, &y);
>-  selector.FormLinearOutput(false, e, y);
>-  EXPECT_EQ(y_ref, y);
>-
>-  init_blocks(&e, &y);
>-  selector.FormLinearOutput(false, e, y);
>-  EXPECT_EQ(y_ref, y);
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_buffer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_buffer.cc
>index 6e224be7039..235e3e335af 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_buffer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_buffer.cc
>@@ -47,4 +47,28 @@ void RenderBuffer::SpectralSum(
>   }
> }
> 
>+void RenderBuffer::SpectralSums(
>+    size_t num_spectra_shorter,
>+    size_t num_spectra_longer,
>+    std::array<float, kFftLengthBy2Plus1>* X2_shorter,
>+    std::array<float, kFftLengthBy2Plus1>* X2_longer) const {
>+  RTC_DCHECK_LE(num_spectra_shorter, num_spectra_longer);
>+  X2_shorter->fill(0.f);
>+  int position = spectrum_buffer_->read;
>+  size_t j = 0;
>+  for (; j < num_spectra_shorter; ++j) {
>+    std::transform(X2_shorter->begin(), X2_shorter->end(),
>+                   spectrum_buffer_->buffer[position].begin(),
>+                   X2_shorter->begin(), std::plus<float>());
>+    position = spectrum_buffer_->IncIndex(position);
>+  }
>+  std::copy(X2_shorter->begin(), X2_shorter->end(), X2_longer->begin());
>+  for (; j < num_spectra_longer; ++j) {
>+    std::transform(X2_longer->begin(), X2_longer->end(),
>+                   spectrum_buffer_->buffer[position].begin(),
>+                   X2_longer->begin(), std::plus<float>());
>+    position = spectrum_buffer_->IncIndex(position);
>+  }
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_buffer.h
>index db94e74cc73..dd67268efd7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_buffer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_buffer.h
>@@ -61,10 +61,44 @@ class RenderBuffer {
>   void SpectralSum(size_t num_spectra,
>                    std::array<float, kFftLengthBy2Plus1>* X2) const;
> 
>+  // Returns the sums of the spectrums for two numbers of FFTs.
>+  void SpectralSums(size_t num_spectra_shorter,
>+                    size_t num_spectra_longer,
>+                    std::array<float, kFftLengthBy2Plus1>* X2_shorter,
>+                    std::array<float, kFftLengthBy2Plus1>* X2_longer) const;
>+
>+  // Gets the recent activity seen in the render signal.
>+  bool GetRenderActivity() const { return render_activity_; }
>+
>+  // Specifies the recent activity seen in the render signal.
>+  void SetRenderActivity(bool activity) { render_activity_ = activity; }
>+
>+  // Returns the headroom between the write and the read positions in the
>+  // buffer.
>+  int Headroom() const {
>+    // The write and read indices are decreased over time.
>+    int headroom =
>+        fft_buffer_->write < fft_buffer_->read
>+            ? fft_buffer_->read - fft_buffer_->write
>+            : fft_buffer_->size - fft_buffer_->write + fft_buffer_->read;
>+
>+    RTC_DCHECK_LE(0, headroom);
>+    RTC_DCHECK_GE(fft_buffer_->size, headroom);
>+
>+    return headroom;
>+  }
>+
>+  // Returns a reference to the spectrum buffer.
>+  const VectorBuffer& GetSpectrumBuffer() const { return *spectrum_buffer_; }
>+
>+  // Returns a reference to the block buffer.
>+  const MatrixBuffer& GetBlockBuffer() const { return *block_buffer_; }
>+
>  private:
>   const MatrixBuffer* const block_buffer_;
>   const VectorBuffer* const spectrum_buffer_;
>   const FftBuffer* const fft_buffer_;
>+  bool render_activity_ = false;
>   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RenderBuffer);
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_buffer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_buffer.cc
>index 13737296df0..37a2378738c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_buffer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_buffer.cc
>@@ -12,6 +12,7 @@
> 
> #include <string.h>
> #include <algorithm>
>+#include <numeric>
> 
> #include "modules/audio_processing/aec3/aec3_common.h"
> #include "modules/audio_processing/aec3/aec3_fft.h"
>@@ -24,10 +25,24 @@
> #include "rtc_base/checks.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/logging.h"
>+#include "system_wrappers/include/field_trial.h"
> 
> namespace webrtc {
> namespace {
> 
>+bool EnableZeroExternalDelayHeadroom() {
>+  return !field_trial::IsEnabled(
>+      "WebRTC-Aec3ZeroExternalDelayHeadroomKillSwitch");
>+}
>+
>+size_t GetDownSamplingFactor(const EchoCanceller3Config& config) {
>+  // Do not use down sampling factor 8 if kill switch is triggered.
>+  return (config.delay.down_sampling_factor == 8 &&
>+          field_trial::IsEnabled("WebRTC-Aec3DownSamplingFactor8KillSwitch"))
>+             ? 4
>+             : config.delay.down_sampling_factor;
>+}
>+
> class RenderDelayBufferImpl final : public RenderDelayBuffer {
>  public:
>   RenderDelayBufferImpl(const EchoCanceller3Config& config, size_t num_bands);
>@@ -37,7 +52,7 @@ class RenderDelayBufferImpl final : public RenderDelayBuffer {
>   BufferingEvent Insert(const std::vector<std::vector<float>>& block) override;
>   BufferingEvent PrepareCaptureProcessing() override;
>   bool SetDelay(size_t delay) override;
>-  rtc::Optional<size_t> Delay() const override { return delay_; }
>+  size_t Delay() const override { return MapInternalDelayToExternalDelay(); }
>   size_t MaxDelay() const override {
>     return blocks_.buffer.size() - 1 - buffer_headroom_;
>   }
>@@ -49,17 +64,21 @@ class RenderDelayBufferImpl final : public RenderDelayBuffer {
> 
>   bool CausalDelay(size_t delay) const override;
> 
>+  void SetAudioBufferDelay(size_t delay_ms) override;
>+
>  private:
>   static int instance_count_;
>   std::unique_ptr<ApmDataDumper> data_dumper_;
>   const Aec3Optimization optimization_;
>   const EchoCanceller3Config config_;
>+  size_t down_sampling_factor_;
>+  const bool use_zero_external_delay_headroom_;
>   const int sub_block_size_;
>   MatrixBuffer blocks_;
>   VectorBuffer spectra_;
>   FftBuffer ffts_;
>-  rtc::Optional<size_t> delay_;
>-  rtc::Optional<int> internal_delay_;
>+  absl::optional<size_t> delay_;
>+  absl::optional<int> internal_delay_;
>   RenderBuffer echo_remover_buffer_;
>   DownsampledRenderBuffer low_rate_;
>   Decimator render_decimator_;
>@@ -72,12 +91,18 @@ class RenderDelayBufferImpl final : public RenderDelayBuffer {
>   int max_observed_jitter_ = 1;
>   size_t capture_call_counter_ = 0;
>   size_t render_call_counter_ = 0;
>+  bool render_activity_ = false;
>+  size_t render_activity_counter_ = 0;
>+  absl::optional<size_t> external_audio_buffer_delay_;
>+  bool external_delay_verified_after_reset_ = false;
> 
>   int LowRateBufferOffset() const { return DelayEstimatorOffset(config_) >> 1; }
>-  int MaxExternalDelayToInternalDelay(size_t delay) const;
>+  int MapExternalDelayToInternalDelay(size_t external_delay_blocks) const;
>+  int MapInternalDelayToExternalDelay() const;
>   void ApplyDelay(int delay);
>   void InsertBlock(const std::vector<std::vector<float>>& block,
>                    int previous_write);
>+  bool DetectActiveRender(rtc::ArrayView<const float> x) const;
> 
>   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RenderDelayBufferImpl);
> };
>@@ -95,7 +120,7 @@ void IncreaseWriteIndices(int sub_block_size,
> }
> 
> // Increases the read indices for the render buffers.
>-void IncreaseReadIndices(const rtc::Optional<int>& delay,
>+void IncreaseReadIndices(const absl::optional<int>& delay,
>                          int sub_block_size,
>                          MatrixBuffer* blocks,
>                          VectorBuffer* spectra,
>@@ -122,7 +147,7 @@ bool RenderOverrun(const MatrixBuffer& b, const DownsampledRenderBuffer& l) {
> // Checks for a render buffer underrun. If the delay is not specified, only the
> // low rate buffer underrun is counted as the delay offset for the other buffers
> // is unknown.
>-bool RenderUnderrun(const rtc::Optional<int>& delay,
>+bool RenderUnderrun(const absl::optional<int>& delay,
>                     const MatrixBuffer& b,
>                     const DownsampledRenderBuffer& l) {
>   return l.read == l.write || (delay && b.read == b.write);
>@@ -152,22 +177,23 @@ RenderDelayBufferImpl::RenderDelayBufferImpl(const EchoCanceller3Config& config,
>           new ApmDataDumper(rtc::AtomicOps::Increment(&instance_count_))),
>       optimization_(DetectOptimization()),
>       config_(config),
>-      sub_block_size_(
>-          static_cast<int>(config.delay.down_sampling_factor > 0
>-                               ? kBlockSize / config.delay.down_sampling_factor
>-                               : kBlockSize)),
>-      blocks_(GetRenderDelayBufferSize(config.delay.down_sampling_factor,
>+      down_sampling_factor_(GetDownSamplingFactor(config)),
>+      use_zero_external_delay_headroom_(EnableZeroExternalDelayHeadroom()),
>+      sub_block_size_(static_cast<int>(down_sampling_factor_ > 0
>+                                           ? kBlockSize / down_sampling_factor_
>+                                           : kBlockSize)),
>+      blocks_(GetRenderDelayBufferSize(down_sampling_factor_,
>                                        config.delay.num_filters,
>                                        config.filter.main.length_blocks),
>               num_bands,
>               kBlockSize),
>       spectra_(blocks_.buffer.size(), kFftLengthBy2Plus1),
>       ffts_(blocks_.buffer.size()),
>-      delay_(config_.delay.min_echo_path_delay_blocks),
>+      delay_(config_.delay.default_delay),
>       echo_remover_buffer_(&blocks_, &spectra_, &ffts_),
>-      low_rate_(GetDownSampledBufferSize(config.delay.down_sampling_factor,
>+      low_rate_(GetDownSampledBufferSize(down_sampling_factor_,
>                                          config.delay.num_filters)),
>-      render_decimator_(config.delay.down_sampling_factor),
>+      render_decimator_(down_sampling_factor_),
>       zero_block_(num_bands, std::vector<float>(kBlockSize, 0.f)),
>       fft_(),
>       render_ds_(sub_block_size_, 0.f),
>@@ -192,12 +218,34 @@ void RenderDelayBufferImpl::Reset() {
>   low_rate_.read = low_rate_.OffsetIndex(
>       low_rate_.write, LowRateBufferOffset() * sub_block_size_);
> 
>-  // Set the render buffer delays to the default delay.
>-  ApplyDelay(config_.delay.default_delay);
>+  // Check for any external audio buffer delay and whether it is feasible.
>+  if (external_audio_buffer_delay_) {
>+    const size_t headroom = use_zero_external_delay_headroom_ ? 0 : 2;
>+    size_t external_delay_to_set = 0;
>+    if (*external_audio_buffer_delay_ < headroom) {
>+      external_delay_to_set = 0;
>+    } else {
>+      external_delay_to_set = *external_audio_buffer_delay_ - headroom;
>+    }
>+
>+    external_delay_to_set = std::min(external_delay_to_set, MaxDelay());
>+
>+    // When an external delay estimate is available, use that delay as the
>+    // initial render buffer delay.
>+    internal_delay_ = external_delay_to_set;
>+    ApplyDelay(*internal_delay_);
>+    delay_ = MapInternalDelayToExternalDelay();
>+
>+    external_delay_verified_after_reset_ = false;
>+  } else {
>+    // If an external delay estimate is not available, use that delay as the
>+    // initial delay. Set the render buffer delays to the default delay.
>+    ApplyDelay(config_.delay.default_delay);
> 
>-  // Unset the delays which are set by ApplyConfig.
>-  delay_ = rtc::nullopt;
>-  internal_delay_ = rtc::nullopt;
>+    // Unset the delays which are set by SetDelay.
>+    delay_ = absl::nullopt;
>+    internal_delay_ = absl::nullopt;
>+  }
> }
> 
> // Inserts a new block into the render buffers.
>@@ -211,7 +259,7 @@ RenderDelayBuffer::BufferingEvent RenderDelayBufferImpl::Insert(
>     } else {
>       if (++num_api_calls_in_a_row_ > max_observed_jitter_) {
>         max_observed_jitter_ = num_api_calls_in_a_row_;
>-        RTC_LOG(LS_INFO)
>+        RTC_LOG(LS_WARNING)
>             << "New max number api jitter observed at render block "
>             << render_call_counter_ << ":  " << num_api_calls_in_a_row_
>             << " blocks";
>@@ -230,6 +278,12 @@ RenderDelayBuffer::BufferingEvent RenderDelayBufferImpl::Insert(
>                              ? BufferingEvent::kRenderOverrun
>                              : BufferingEvent::kNone;
> 
>+  // Detect and update render activity.
>+  if (!render_activity_) {
>+    render_activity_counter_ += DetectActiveRender(block[0]) ? 1 : 0;
>+    render_activity_ = render_activity_counter_ >= 20;
>+  }
>+
>   // Insert the new render block into the specified position.
>   InsertBlock(block, previous_write);
> 
>@@ -253,7 +307,7 @@ RenderDelayBufferImpl::PrepareCaptureProcessing() {
>     } else {
>       if (++num_api_calls_in_a_row_ > max_observed_jitter_) {
>         max_observed_jitter_ = num_api_calls_in_a_row_;
>-        RTC_LOG(LS_INFO)
>+        RTC_LOG(LS_WARNING)
>             << "New max number api jitter observed at capture block "
>             << capture_call_counter_ << ":  " << num_api_calls_in_a_row_
>             << " blocks";
>@@ -283,18 +337,32 @@ RenderDelayBufferImpl::PrepareCaptureProcessing() {
>     Reset();
>   }
> 
>+  echo_remover_buffer_.SetRenderActivity(render_activity_);
>+  if (render_activity_) {
>+    render_activity_counter_ = 0;
>+    render_activity_ = false;
>+  }
>+
>   return event;
> }
> 
> // Sets the delay and returns a bool indicating whether the delay was changed.
> bool RenderDelayBufferImpl::SetDelay(size_t delay) {
>+  if (!external_delay_verified_after_reset_ && external_audio_buffer_delay_ &&
>+      delay_) {
>+    int difference = static_cast<int>(delay) - static_cast<int>(*delay_);
>+    RTC_LOG(LS_WARNING) << "Mismatch between first estimated delay after reset "
>+                           "and external delay: "
>+                        << difference << " blocks";
>+    external_delay_verified_after_reset_ = true;
>+  }
>   if (delay_ && *delay_ == delay) {
>     return false;
>   }
>   delay_ = delay;
> 
>   // Compute the internal delay and limit the delay to the allowed range.
>-  int internal_delay = MaxExternalDelayToInternalDelay(*delay_);
>+  int internal_delay = MapExternalDelayToInternalDelay(*delay_);
>   internal_delay_ =
>       std::min(MaxDelay(), static_cast<size_t>(std::max(internal_delay, 0)));
> 
>@@ -306,7 +374,7 @@ bool RenderDelayBufferImpl::SetDelay(size_t delay) {
> // Returns whether the specified delay is causal.
> bool RenderDelayBufferImpl::CausalDelay(size_t delay) const {
>   // Compute the internal delay and limit the delay to the allowed range.
>-  int internal_delay = MaxExternalDelayToInternalDelay(delay);
>+  int internal_delay = MapExternalDelayToInternalDelay(delay);
>   internal_delay =
>       std::min(MaxDelay(), static_cast<size_t>(std::max(internal_delay, 0)));
> 
>@@ -314,8 +382,19 @@ bool RenderDelayBufferImpl::CausalDelay(size_t delay) const {
>          static_cast<int>(config_.delay.min_echo_path_delay_blocks);
> }
> 
>+void RenderDelayBufferImpl::SetAudioBufferDelay(size_t delay_ms) {
>+  if (!external_audio_buffer_delay_) {
>+    RTC_LOG(LS_WARNING)
>+        << "Receiving a first reported externally buffer delay of " << delay_ms
>+        << " ms.";
>+  }
>+
>+  // Convert delay from milliseconds to blocks (rounded down).
>+  external_audio_buffer_delay_ = delay_ms / 4;
>+}
>+
> // Maps the externally computed delay to the delay used internally.
>-int RenderDelayBufferImpl::MaxExternalDelayToInternalDelay(
>+int RenderDelayBufferImpl::MapExternalDelayToInternalDelay(
>     size_t external_delay_blocks) const {
>   const int latency = BufferLatency(low_rate_);
>   RTC_DCHECK_LT(0, sub_block_size_);
>@@ -325,8 +404,20 @@ int RenderDelayBufferImpl::MaxExternalDelayToInternalDelay(
>          DelayEstimatorOffset(config_);
> }
> 
>+// Maps the internally used delay to the delay used externally.
>+int RenderDelayBufferImpl::MapInternalDelayToExternalDelay() const {
>+  const int latency = BufferLatency(low_rate_);
>+  int latency_blocks = latency / sub_block_size_;
>+  int internal_delay = spectra_.read >= spectra_.write
>+                           ? spectra_.read - spectra_.write
>+                           : spectra_.size + spectra_.read - spectra_.write;
>+
>+  return internal_delay - latency_blocks + DelayEstimatorOffset(config_);
>+}
>+
> // Set the read indices according to the delay.
> void RenderDelayBufferImpl::ApplyDelay(int delay) {
>+  RTC_LOG(LS_WARNING) << "Applying internal delay of " << delay << " blocks.";
>   blocks_.read = blocks_.OffsetIndex(blocks_.write, -delay);
>   spectra_.read = spectra_.OffsetIndex(spectra_.write, delay);
>   ffts_.read = ffts_.OffsetIndex(ffts_.write, delay);
>@@ -347,12 +438,24 @@ void RenderDelayBufferImpl::InsertBlock(
>     std::copy(block[k].begin(), block[k].end(), b.buffer[b.write][k].begin());
>   }
> 
>+  data_dumper_->DumpWav("aec3_render_decimator_input", block[0].size(),
>+                        block[0].data(), 16000, 1);
>   render_decimator_.Decimate(block[0], ds);
>+  data_dumper_->DumpWav("aec3_render_decimator_output", ds.size(), ds.data(),
>+                        16000 / down_sampling_factor_, 1);
>   std::copy(ds.rbegin(), ds.rend(), lr.buffer.begin() + lr.write);
>   fft_.PaddedFft(block[0], b.buffer[previous_write][0], &f.buffer[f.write]);
>   f.buffer[f.write].Spectrum(optimization_, s.buffer[s.write]);
> }
> 
>+bool RenderDelayBufferImpl::DetectActiveRender(
>+    rtc::ArrayView<const float> x) const {
>+  const float x_energy = std::inner_product(x.begin(), x.end(), x.begin(), 0.f);
>+  return x_energy > (config_.render_levels.active_render_limit *
>+                     config_.render_levels.active_render_limit) *
>+                        kFftLengthBy2;
>+}
>+
> }  // namespace
> 
> int RenderDelayBuffer::RenderDelayBuffer::DelayEstimatorOffset(
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_buffer.h
>index 28ae32ba06b..a6d6874e3e4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_buffer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_buffer.h
>@@ -15,12 +15,13 @@
> #include <array>
> #include <vector>
> 
>+#include "absl/types/optional.h"
> #include "api/array_view.h"
>+#include "api/audio/echo_canceller3_config.h"
> #include "modules/audio_processing/aec3/aec3_common.h"
> #include "modules/audio_processing/aec3/downsampled_render_buffer.h"
> #include "modules/audio_processing/aec3/fft_data.h"
> #include "modules/audio_processing/aec3/render_buffer.h"
>-#include "modules/audio_processing/include/audio_processing.h"
> 
> namespace webrtc {
> 
>@@ -56,7 +57,7 @@ class RenderDelayBuffer {
>   virtual bool SetDelay(size_t delay) = 0;
> 
>   // Gets the buffer delay.
>-  virtual rtc::Optional<size_t> Delay() const = 0;
>+  virtual size_t Delay() const = 0;
> 
>   // Gets the buffer delay.
>   virtual size_t MaxDelay() const = 0;
>@@ -72,6 +73,9 @@ class RenderDelayBuffer {
> 
>   // Returns the maximum non calusal offset that can occur in the delay buffer.
>   static int DelayEstimatorOffset(const EchoCanceller3Config& config);
>+
>+  // Provides an optional external estimate of the audio buffer delay.
>+  virtual void SetAudioBufferDelay(size_t delay_ms) = 0;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_buffer_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_buffer_unittest.cc
>index fb9c48d0e84..78f0b5a4f7e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_buffer_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_buffer_unittest.cc
>@@ -75,12 +75,14 @@ TEST(RenderDelayBuffer, SetDelay) {
>   EchoCanceller3Config config;
>   std::unique_ptr<RenderDelayBuffer> delay_buffer(
>       RenderDelayBuffer::Create(config, 1));
>-  ASSERT_FALSE(delay_buffer->Delay());
>-  for (size_t delay = config.delay.min_echo_path_delay_blocks + 1; delay < 20;
>-       ++delay) {
>-    delay_buffer->SetDelay(delay);
>-    ASSERT_TRUE(delay_buffer->Delay());
>-    EXPECT_EQ(delay, *delay_buffer->Delay());
>+  ASSERT_TRUE(delay_buffer->Delay());
>+  delay_buffer->Reset();
>+  size_t initial_internal_delay = config.delay.min_echo_path_delay_blocks +
>+                                  config.delay.api_call_jitter_blocks;
>+  for (size_t delay = initial_internal_delay;
>+       delay < initial_internal_delay + 20; ++delay) {
>+    ASSERT_TRUE(delay_buffer->SetDelay(delay));
>+    EXPECT_EQ(delay, delay_buffer->Delay());
>   }
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_controller.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_controller.cc
>index 6120c3ecae2..8adf5f51029 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_controller.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_controller.cc
>@@ -15,57 +15,32 @@
> #include <string>
> #include <vector>
> 
>+#include "api/audio/echo_canceller3_config.h"
> #include "modules/audio_processing/aec3/aec3_common.h"
> #include "modules/audio_processing/aec3/echo_path_delay_estimator.h"
> #include "modules/audio_processing/aec3/render_delay_controller_metrics.h"
>-#include "modules/audio_processing/include/audio_processing.h"
>+#include "modules/audio_processing/aec3/skew_estimator.h"
> #include "rtc_base/atomicops.h"
> #include "rtc_base/constructormagic.h"
>+#include "rtc_base/logging.h"
>+#include "system_wrappers/include/field_trial.h"
> 
> namespace webrtc {
> 
> namespace {
> 
>-constexpr int kSkewHistorySizeLog2 = 8;
>-
>-// Estimator of API call skew between render and capture.
>-class SkewEstimator {
>- public:
>-  // Resets the estimation.
>-  void Reset() {
>-    skew_ = 0;
>-    next_index_ = 0;
>-    sufficient_skew_stored_ = false;
>+int GetSkewHysteresis(const EchoCanceller3Config& config) {
>+  if (field_trial::IsEnabled("WebRTC-Aec3EnforceSkewHysteresis1")) {
>+    return 1;
>   }
>-
>-  // Updates the skew data for a render call.
>-  void LogRenderCall() { ++skew_; }
>-
>-  // Updates and computes the skew at a capture call. Returns an optional which
>-  // is non-null if a reliable skew has been found.
>-  rtc::Optional<int> GetSkewFromCapture() {
>-    --skew_;
>-
>-    skew_history_[next_index_] = skew_;
>-    if (++next_index_ == skew_history_.size()) {
>-      next_index_ = 0;
>-      sufficient_skew_stored_ = true;
>-    }
>-
>-    if (!sufficient_skew_stored_) {
>-      return rtc::nullopt;
>-    }
>-
>-    return std::accumulate(skew_history_.begin(), skew_history_.end(), 0) >>
>-           kSkewHistorySizeLog2;
>+  if (field_trial::IsEnabled("WebRTC-Aec3EnforceSkewHysteresis2")) {
>+    return 2;
>   }
> 
>- private:
>-  int skew_ = 0;
>-  std::array<int, 1 << kSkewHistorySizeLog2> skew_history_;
>-  size_t next_index_ = 0;
>-  bool sufficient_skew_stored_ = false;
>-};
>+  return static_cast<int>(config.delay.skew_hysteresis_blocks);
>+}
>+
>+constexpr int kSkewHistorySizeLog2 = 8;
> 
> class RenderDelayControllerImpl final : public RenderDelayController {
>  public:
>@@ -75,8 +50,10 @@ class RenderDelayControllerImpl final : public RenderDelayController {
>   ~RenderDelayControllerImpl() override;
>   void Reset() override;
>   void LogRenderCall() override;
>-  rtc::Optional<DelayEstimate> GetDelay(
>+  absl::optional<DelayEstimate> GetDelay(
>       const DownsampledRenderBuffer& render_buffer,
>+      size_t render_delay_buffer_delay,
>+      const absl::optional<int>& echo_remover_delay,
>       rtc::ArrayView<const float> capture) override;
> 
>  private:
>@@ -85,21 +62,25 @@ class RenderDelayControllerImpl final : public RenderDelayController {
>   const int delay_headroom_blocks_;
>   const int hysteresis_limit_1_blocks_;
>   const int hysteresis_limit_2_blocks_;
>-  rtc::Optional<DelayEstimate> delay_;
>+  const int skew_hysteresis_blocks_;
>+  absl::optional<DelayEstimate> delay_;
>   EchoPathDelayEstimator delay_estimator_;
>   std::vector<float> delay_buf_;
>   int delay_buf_index_ = 0;
>   RenderDelayControllerMetrics metrics_;
>   SkewEstimator skew_estimator_;
>-  rtc::Optional<DelayEstimate> delay_samples_;
>-  rtc::Optional<int> skew_;
>+  absl::optional<DelayEstimate> delay_samples_;
>+  absl::optional<int> skew_;
>+  int previous_offset_blocks_ = 0;
>+  int skew_shift_reporting_counter_ = 0;
>+  size_t capture_call_counter_ = 0;
>   int delay_change_counter_ = 0;
>   size_t soft_reset_counter_ = 0;
>   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RenderDelayControllerImpl);
> };
> 
> DelayEstimate ComputeBufferDelay(
>-    const rtc::Optional<DelayEstimate>& current_delay,
>+    const absl::optional<DelayEstimate>& current_delay,
>     int delay_headroom_blocks,
>     int hysteresis_limit_1_blocks,
>     int hysteresis_limit_2_blocks,
>@@ -130,7 +111,9 @@ DelayEstimate ComputeBufferDelay(
>     }
>   }
> 
>-  return DelayEstimate(estimated_delay.quality, new_delay_blocks);
>+  DelayEstimate new_delay = estimated_delay;
>+  new_delay.delay = new_delay_blocks;
>+  return new_delay;
> }
> 
> int RenderDelayControllerImpl::instance_count_ = 0;
>@@ -147,8 +130,10 @@ RenderDelayControllerImpl::RenderDelayControllerImpl(
>           static_cast<int>(config.delay.hysteresis_limit_1_blocks)),
>       hysteresis_limit_2_blocks_(
>           static_cast<int>(config.delay.hysteresis_limit_2_blocks)),
>+      skew_hysteresis_blocks_(GetSkewHysteresis(config)),
>       delay_estimator_(data_dumper_.get(), config),
>-      delay_buf_(kBlockSize * non_causal_offset, 0.f) {
>+      delay_buf_(kBlockSize * non_causal_offset, 0.f),
>+      skew_estimator_(kSkewHistorySizeLog2) {
>   RTC_DCHECK(ValidFullBandRate(sample_rate_hz));
>   delay_estimator_.LogDelayEstimationProperties(sample_rate_hz,
>                                                 delay_buf_.size());
>@@ -157,9 +142,10 @@ RenderDelayControllerImpl::RenderDelayControllerImpl(
> RenderDelayControllerImpl::~RenderDelayControllerImpl() = default;
> 
> void RenderDelayControllerImpl::Reset() {
>-  delay_ = rtc::nullopt;
>-  delay_samples_ = rtc::nullopt;
>-  skew_ = rtc::nullopt;
>+  delay_ = absl::nullopt;
>+  delay_samples_ = absl::nullopt;
>+  skew_ = absl::nullopt;
>+  previous_offset_blocks_ = 0;
>   std::fill(delay_buf_.begin(), delay_buf_.end(), 0.f);
>   delay_estimator_.Reset(false);
>   skew_estimator_.Reset();
>@@ -171,10 +157,13 @@ void RenderDelayControllerImpl::LogRenderCall() {
>   skew_estimator_.LogRenderCall();
> }
> 
>-rtc::Optional<DelayEstimate> RenderDelayControllerImpl::GetDelay(
>+absl::optional<DelayEstimate> RenderDelayControllerImpl::GetDelay(
>     const DownsampledRenderBuffer& render_buffer,
>+    size_t render_delay_buffer_delay,
>+    const absl::optional<int>& echo_remover_delay,
>     rtc::ArrayView<const float> capture) {
>   RTC_DCHECK_EQ(kBlockSize, capture.size());
>+  ++capture_call_counter_;
> 
>   // Estimate the delay with a delayed capture.
>   RTC_DCHECK_LT(delay_buf_index_ + kBlockSize - 1, delay_buf_.size());
>@@ -183,18 +172,44 @@ rtc::Optional<DelayEstimate> RenderDelayControllerImpl::GetDelay(
>   auto delay_samples =
>       delay_estimator_.EstimateDelay(render_buffer, capture_delayed);
> 
>+  // Overrule the delay estimator delay if the echo remover reports a delay.
>+  if (echo_remover_delay) {
>+    int total_echo_remover_delay_samples =
>+        (render_delay_buffer_delay + *echo_remover_delay) * kBlockSize;
>+    delay_samples = DelayEstimate(DelayEstimate::Quality::kRefined,
>+                                  total_echo_remover_delay_samples);
>+  }
>+
>   std::copy(capture.begin(), capture.end(),
>             delay_buf_.begin() + delay_buf_index_);
>   delay_buf_index_ = (delay_buf_index_ + kBlockSize) % delay_buf_.size();
> 
>   // Compute the latest skew update.
>-  rtc::Optional<int> skew = skew_estimator_.GetSkewFromCapture();
>+  absl::optional<int> skew = skew_estimator_.GetSkewFromCapture();
> 
>   if (delay_samples) {
>+    // TODO(peah): Refactor the rest of the code to assume a kRefined estimate
>+    // quality.
>+    RTC_DCHECK(DelayEstimate::Quality::kRefined == delay_samples->quality);
>     if (!delay_samples_ || delay_samples->delay != delay_samples_->delay) {
>       delay_change_counter_ = 0;
>     }
>-    delay_samples_ = delay_samples;
>+    if (delay_samples_) {
>+      delay_samples_->blocks_since_last_change =
>+          delay_samples_->delay == delay_samples->delay
>+              ? delay_samples_->blocks_since_last_change + 1
>+              : 0;
>+      delay_samples_->blocks_since_last_update = 0;
>+      delay_samples_->delay = delay_samples->delay;
>+      delay_samples_->quality = delay_samples->quality;
>+    } else {
>+      delay_samples_ = delay_samples;
>+    }
>+  } else {
>+    if (delay_samples_) {
>+      ++delay_samples_->blocks_since_last_change;
>+      ++delay_samples_->blocks_since_last_update;
>+    }
>   }
> 
>   if (delay_change_counter_ < 2 * kNumBlocksPerSecond) {
>@@ -213,7 +228,9 @@ rtc::Optional<DelayEstimate> RenderDelayControllerImpl::GetDelay(
>       delay_samples_->quality == DelayEstimate::Quality::kRefined) {
>     // Compute the skew offset and add a margin.
>     offset_blocks = *skew_ - *skew;
>-    if (offset_blocks != 0 && soft_reset_counter_ > 10 * kNumBlocksPerSecond) {
>+    if (abs(offset_blocks) <= skew_hysteresis_blocks_) {
>+      offset_blocks = 0;
>+    } else if (soft_reset_counter_ > 10 * kNumBlocksPerSecond) {
>       // Soft reset the delay estimator if there is a significant offset
>       // detected.
>       delay_estimator_.Reset(true);
>@@ -221,6 +238,22 @@ rtc::Optional<DelayEstimate> RenderDelayControllerImpl::GetDelay(
>     }
>   }
> 
>+  // Log any changes in the skew.
>+  skew_shift_reporting_counter_ =
>+      std::max(0, skew_shift_reporting_counter_ - 1);
>+  absl::optional<int> skew_shift =
>+      skew_shift_reporting_counter_ == 0 &&
>+              previous_offset_blocks_ != offset_blocks
>+          ? absl::optional<int>(offset_blocks - previous_offset_blocks_)
>+          : absl::nullopt;
>+  previous_offset_blocks_ = offset_blocks;
>+  if (skew_shift) {
>+    RTC_LOG(LS_WARNING) << "API call skew shift of " << *skew_shift
>+                        << " blocks detected at capture block "
>+                        << capture_call_counter_;
>+    skew_shift_reporting_counter_ = 3 * kNumBlocksPerSecond;
>+  }
>+
>   if (delay_samples_) {
>     // Compute the render delay buffer delay.
>     delay_ = ComputeBufferDelay(
>@@ -228,9 +261,9 @@ rtc::Optional<DelayEstimate> RenderDelayControllerImpl::GetDelay(
>         hysteresis_limit_2_blocks_, offset_blocks, *delay_samples_);
>   }
> 
>-  metrics_.Update(delay_samples_ ? rtc::Optional<size_t>(delay_samples_->delay)
>-                                 : rtc::nullopt,
>-                  delay_ ? delay_->delay : 0);
>+  metrics_.Update(delay_samples_ ? absl::optional<size_t>(delay_samples_->delay)
>+                                 : absl::nullopt,
>+                  delay_ ? delay_->delay : 0, skew_shift);
> 
>   data_dumper_->DumpRaw("aec3_render_delay_controller_delay",
>                         delay_samples ? delay_samples->delay : 0);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_controller.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_controller.h
>index 842b8fffe51..ddd95482fcc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_controller.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_controller.h
>@@ -11,12 +11,12 @@
> #ifndef MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_
> #define MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_H_
> 
>+#include "absl/types/optional.h"
> #include "api/array_view.h"
>-#include "api/optional.h"
>+#include "api/audio/echo_canceller3_config.h"
> #include "modules/audio_processing/aec3/delay_estimate.h"
> #include "modules/audio_processing/aec3/downsampled_render_buffer.h"
> #include "modules/audio_processing/aec3/render_delay_buffer.h"
>-#include "modules/audio_processing/include/audio_processing.h"
> #include "modules/audio_processing/logging/apm_data_dumper.h"
> 
> namespace webrtc {
>@@ -36,8 +36,10 @@ class RenderDelayController {
>   virtual void LogRenderCall() = 0;
> 
>   // Aligns the render buffer content with the capture signal.
>-  virtual rtc::Optional<DelayEstimate> GetDelay(
>+  virtual absl::optional<DelayEstimate> GetDelay(
>       const DownsampledRenderBuffer& render_buffer,
>+      size_t render_delay_buffer_delay,
>+      const absl::optional<int>& echo_remover_delay,
>       rtc::ArrayView<const float> capture) = 0;
> };
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_controller_metrics.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_controller_metrics.cc
>index 696ac29a733..09db3396834 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_controller_metrics.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_controller_metrics.cc
>@@ -36,21 +36,34 @@ enum class DelayChangesCategory {
>   kNumCategories
> };
> 
>+constexpr int kMaxSkewShiftCount = 20;
>+
> }  // namespace
> 
>-void RenderDelayControllerMetrics::Update(rtc::Optional<size_t> delay_samples,
>-                                          size_t buffer_delay_blocks) {
>+RenderDelayControllerMetrics::RenderDelayControllerMetrics() = default;
>+
>+void RenderDelayControllerMetrics::Update(
>+    absl::optional<size_t> delay_samples,
>+    size_t buffer_delay_blocks,
>+    absl::optional<int> skew_shift_blocks) {
>   ++call_counter_;
> 
>   if (!initial_update) {
>+    size_t delay_blocks;
>     if (delay_samples) {
>       ++reliable_delay_estimate_counter_;
>-      size_t delay_blocks = (*delay_samples) / kBlockSize;
>+      delay_blocks = (*delay_samples) / kBlockSize + 2;
>+    } else {
>+      delay_blocks = 0;
>+    }
> 
>-      if (delay_blocks != delay_blocks_) {
>-        ++delay_change_counter_;
>-        delay_blocks_ = delay_blocks;
>-      }
>+    if (delay_blocks != delay_blocks_) {
>+      ++delay_change_counter_;
>+      delay_blocks_ = delay_blocks;
>+    }
>+
>+    if (skew_shift_blocks) {
>+      skew_shift_count_ = std::min(kMaxSkewShiftCount, skew_shift_count_);
>     }
>   } else if (++initial_call_counter_ == 5 * kNumBlocksPerSecond) {
>     initial_update = false;
>@@ -58,12 +71,12 @@ void RenderDelayControllerMetrics::Update(rtc::Optional<size_t> delay_samples,
> 
>   if (call_counter_ == kMetricsReportingIntervalBlocks) {
>     int value_to_report = static_cast<int>(delay_blocks_);
>-    value_to_report = std::min(124, value_to_report);
>+    value_to_report = std::min(124, value_to_report >> 1);
>     RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.EchoCanceller.EchoPathDelay",
>                                 value_to_report, 0, 124, 125);
> 
>-    value_to_report = static_cast<int>(buffer_delay_blocks);
>-    value_to_report = std::min(124, value_to_report);
>+    value_to_report = static_cast<int>(buffer_delay_blocks + 2);
>+    value_to_report = std::min(124, value_to_report >> 1);
>     RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.EchoCanceller.BufferDelay",
>                                 value_to_report, 0, 124, 125);
> 
>@@ -107,6 +120,15 @@ void RenderDelayControllerMetrics::Update(rtc::Optional<size_t> delay_samples,
>   } else {
>     metrics_reported_ = false;
>   }
>+
>+  if (!initial_update && ++skew_report_timer_ == 60 * kNumBlocksPerSecond) {
>+    RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.EchoCanceller.MaxSkewShiftCount",
>+                                skew_shift_count_, 0, kMaxSkewShiftCount,
>+                                kMaxSkewShiftCount + 1);
>+
>+    skew_shift_count_ = 0;
>+    skew_report_timer_ = 0;
>+  }
> }
> 
> void RenderDelayControllerMetrics::ResetMetrics() {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_controller_metrics.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_controller_metrics.h
>index a93f0764edc..1cfe899b488 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_controller_metrics.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_controller_metrics.h
>@@ -11,7 +11,7 @@
> #ifndef MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_METRICS_H_
> #define MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_CONTROLLER_METRICS_H_
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> #include "rtc_base/constructormagic.h"
> 
> namespace webrtc {
>@@ -19,10 +19,12 @@ namespace webrtc {
> // Handles the reporting of metrics for the render delay controller.
> class RenderDelayControllerMetrics {
>  public:
>-  RenderDelayControllerMetrics() = default;
>+  RenderDelayControllerMetrics();
> 
>   // Updates the metric with new data.
>-  void Update(rtc::Optional<size_t> delay_samples, size_t buffer_delay_blocks);
>+  void Update(absl::optional<size_t> delay_samples,
>+              size_t buffer_delay_blocks,
>+              absl::optional<int> skew_shift_blocks);
> 
>   // Returns true if the metrics have just been reported, otherwise false.
>   bool MetricsReported() { return metrics_reported_; }
>@@ -35,9 +37,11 @@ class RenderDelayControllerMetrics {
>   int reliable_delay_estimate_counter_ = 0;
>   int delay_change_counter_ = 0;
>   int call_counter_ = 0;
>+  int skew_report_timer_ = 0;
>   int initial_call_counter_ = 0;
>   bool metrics_reported_ = false;
>   bool initial_update = true;
>+  int skew_shift_count_ = 0;
> 
>   RTC_DISALLOW_COPY_AND_ASSIGN(RenderDelayControllerMetrics);
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_controller_metrics_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_controller_metrics_unittest.cc
>index 433c2495540..e867de4df42 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_controller_metrics_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_controller_metrics_unittest.cc
>@@ -9,7 +9,7 @@
>  */
> 
> #include "modules/audio_processing/aec3/render_delay_controller_metrics.h"
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> #include "modules/audio_processing/aec3/aec3_common.h"
> 
> #include "test/gtest.h"
>@@ -22,10 +22,10 @@ TEST(RenderDelayControllerMetrics, NormalUsage) {
> 
>   for (int j = 0; j < 3; ++j) {
>     for (int k = 0; k < kMetricsReportingIntervalBlocks - 1; ++k) {
>-      metrics.Update(rtc::nullopt, 0);
>+      metrics.Update(absl::nullopt, 0, absl::nullopt);
>       EXPECT_FALSE(metrics.MetricsReported());
>     }
>-    metrics.Update(rtc::nullopt, 0);
>+    metrics.Update(absl::nullopt, 0, absl::nullopt);
>     EXPECT_TRUE(metrics.MetricsReported());
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_controller_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_controller_unittest.cc
>index 656c5e88174..98d5b259dce 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_controller_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_delay_controller_unittest.cc
>@@ -48,6 +48,7 @@ constexpr size_t kDownSamplingFactors[] = {2, 4, 8};
> TEST(RenderDelayController, NoRenderSignal) {
>   std::vector<float> block(kBlockSize, 0.f);
>   EchoCanceller3Config config;
>+  absl::optional<int> echo_remover_delay_;
>   for (size_t num_matched_filters = 4; num_matched_filters == 10;
>        num_matched_filters++) {
>     for (auto down_sampling_factor : kDownSamplingFactors) {
>@@ -62,7 +63,8 @@ TEST(RenderDelayController, NoRenderSignal) {
>                 config, RenderDelayBuffer::DelayEstimatorOffset(config), rate));
>         for (size_t k = 0; k < 100; ++k) {
>           auto delay = delay_controller->GetDelay(
>-              delay_buffer->GetDownsampledRenderBuffer(), block);
>+              delay_buffer->GetDownsampledRenderBuffer(), delay_buffer->Delay(),
>+              echo_remover_delay_, block);
>           EXPECT_EQ(config.delay.min_echo_path_delay_blocks, delay->delay);
>         }
>       }
>@@ -73,7 +75,8 @@ TEST(RenderDelayController, NoRenderSignal) {
> // Verifies the basic API call sequence.
> TEST(RenderDelayController, BasicApiCalls) {
>   std::vector<float> capture_block(kBlockSize, 0.f);
>-  rtc::Optional<DelayEstimate> delay_blocks;
>+  absl::optional<DelayEstimate> delay_blocks;
>+  absl::optional<int> echo_remover_delay;
>   for (size_t num_matched_filters = 4; num_matched_filters == 10;
>        num_matched_filters++) {
>     for (auto down_sampling_factor : kDownSamplingFactors) {
>@@ -94,7 +97,8 @@ TEST(RenderDelayController, BasicApiCalls) {
>           render_delay_buffer->PrepareCaptureProcessing();
> 
>           delay_blocks = delay_controller->GetDelay(
>-              render_delay_buffer->GetDownsampledRenderBuffer(), capture_block);
>+              render_delay_buffer->GetDownsampledRenderBuffer(),
>+              render_delay_buffer->Delay(), echo_remover_delay, capture_block);
>         }
>         EXPECT_TRUE(delay_blocks);
>         EXPECT_EQ(config.delay.min_echo_path_delay_blocks, delay_blocks->delay);
>@@ -107,6 +111,7 @@ TEST(RenderDelayController, BasicApiCalls) {
> // simple timeshifts between the signals.
> TEST(RenderDelayController, Alignment) {
>   Random random_generator(42U);
>+  absl::optional<int> echo_remover_delay;
>   std::vector<float> capture_block(kBlockSize, 0.f);
>   for (size_t num_matched_filters = 4; num_matched_filters == 10;
>        num_matched_filters++) {
>@@ -120,7 +125,7 @@ TEST(RenderDelayController, Alignment) {
>             NumBandsForRate(rate), std::vector<float>(kBlockSize, 0.f));
> 
>         for (size_t delay_samples : {15, 50, 150, 200, 800, 4000}) {
>-          rtc::Optional<DelayEstimate> delay_blocks;
>+          absl::optional<DelayEstimate> delay_blocks;
>           SCOPED_TRACE(ProduceDebugText(rate, delay_samples));
>           std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
>               RenderDelayBuffer::Create(config, NumBandsForRate(rate)));
>@@ -136,6 +141,7 @@ TEST(RenderDelayController, Alignment) {
>             render_delay_buffer->PrepareCaptureProcessing();
>             delay_blocks = delay_controller->GetDelay(
>                 render_delay_buffer->GetDownsampledRenderBuffer(),
>+                render_delay_buffer->Delay(), echo_remover_delay,
>                 capture_block);
>           }
>           ASSERT_TRUE(!!delay_blocks);
>@@ -156,6 +162,7 @@ TEST(RenderDelayController, Alignment) {
> // delays.
> TEST(RenderDelayController, NonCausalAlignment) {
>   Random random_generator(42U);
>+  absl::optional<int> echo_remover_delay;
>   for (size_t num_matched_filters = 4; num_matched_filters == 10;
>        num_matched_filters++) {
>     for (auto down_sampling_factor : kDownSamplingFactors) {
>@@ -169,7 +176,7 @@ TEST(RenderDelayController, NonCausalAlignment) {
>             NumBandsForRate(rate), std::vector<float>(kBlockSize, 0.f));
> 
>         for (int delay_samples : {-15, -50, -150, -200}) {
>-          rtc::Optional<DelayEstimate> delay_blocks;
>+          absl::optional<DelayEstimate> delay_blocks;
>           SCOPED_TRACE(ProduceDebugText(rate, -delay_samples));
>           std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
>               RenderDelayBuffer::Create(config, NumBandsForRate(rate)));
>@@ -186,6 +193,7 @@ TEST(RenderDelayController, NonCausalAlignment) {
>             render_delay_buffer->PrepareCaptureProcessing();
>             delay_blocks = delay_controller->GetDelay(
>                 render_delay_buffer->GetDownsampledRenderBuffer(),
>+                render_delay_buffer->Delay(), echo_remover_delay,
>                 capture_block[0]);
>           }
> 
>@@ -200,6 +208,7 @@ TEST(RenderDelayController, NonCausalAlignment) {
> // simple timeshifts between the signals when there is jitter in the API calls.
> TEST(RenderDelayController, AlignmentWithJitter) {
>   Random random_generator(42U);
>+  absl::optional<int> echo_remover_delay;
>   std::vector<float> capture_block(kBlockSize, 0.f);
>   for (size_t num_matched_filters = 4; num_matched_filters == 10;
>        num_matched_filters++) {
>@@ -211,7 +220,7 @@ TEST(RenderDelayController, AlignmentWithJitter) {
>         std::vector<std::vector<float>> render_block(
>             NumBandsForRate(rate), std::vector<float>(kBlockSize, 0.f));
>         for (size_t delay_samples : {15, 50, 300, 800}) {
>-          rtc::Optional<DelayEstimate> delay_blocks;
>+          absl::optional<DelayEstimate> delay_blocks;
>           SCOPED_TRACE(ProduceDebugText(rate, delay_samples));
>           std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
>               RenderDelayBuffer::Create(config, NumBandsForRate(rate)));
>@@ -237,6 +246,7 @@ TEST(RenderDelayController, AlignmentWithJitter) {
>               render_delay_buffer->PrepareCaptureProcessing();
>               delay_blocks = delay_controller->GetDelay(
>                   render_delay_buffer->GetDownsampledRenderBuffer(),
>+                  render_delay_buffer->Delay(), echo_remover_delay,
>                   capture_block_buffer[k]);
>             }
>           }
>@@ -286,6 +296,7 @@ TEST(RenderDelayController, InitialHeadroom) {
> TEST(RenderDelayController, WrongCaptureSize) {
>   std::vector<float> block(kBlockSize - 1, 0.f);
>   EchoCanceller3Config config;
>+  absl::optional<int> echo_remover_delay;
>   for (auto rate : {8000, 16000, 32000, 48000}) {
>     SCOPED_TRACE(ProduceDebugText(rate));
>     std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
>@@ -296,7 +307,7 @@ TEST(RenderDelayController, WrongCaptureSize) {
>                 EchoCanceller3Config(),
>                 RenderDelayBuffer::DelayEstimatorOffset(config), rate))
>             ->GetDelay(render_delay_buffer->GetDownsampledRenderBuffer(),
>-                       block),
>+                       render_delay_buffer->Delay(), echo_remover_delay, block),
>         "");
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_signal_analyzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_signal_analyzer.cc
>index 84546c4c2fb..50c34ce43f6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_signal_analyzer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_signal_analyzer.cc
>@@ -23,7 +23,7 @@ constexpr size_t kCounterThreshold = 5;
> // Identifies local bands with narrow characteristics.
> void IdentifySmallNarrowBandRegions(
>     const RenderBuffer& render_buffer,
>-    const rtc::Optional<size_t>& delay_partitions,
>+    const absl::optional<size_t>& delay_partitions,
>     std::array<size_t, kFftLengthBy2 - 1>* narrow_band_counters) {
>   if (!delay_partitions) {
>     narrow_band_counters->fill(0);
>@@ -42,7 +42,8 @@ void IdentifySmallNarrowBandRegions(
> 
> // Identifies whether the signal has a single strong narrow-band component.
> void IdentifyStrongNarrowBandComponent(const RenderBuffer& render_buffer,
>-                                       rtc::Optional<int>* narrow_peak_band,
>+                                       int strong_peak_freeze_duration,
>+                                       absl::optional<int>* narrow_peak_band,
>                                        size_t* narrow_peak_counter) {
>   const auto X2_latest = render_buffer.Spectrum(0);
> 
>@@ -52,7 +53,7 @@ void IdentifyStrongNarrowBandComponent(const RenderBuffer& render_buffer,
> 
>   // Compute the level around the peak.
>   float non_peak_power = 0.f;
>-  for (int k = std::max(5, peak_bin - 14); k < peak_bin - 4; ++k) {
>+  for (int k = std::max(0, peak_bin - 14); k < peak_bin - 4; ++k) {
>     non_peak_power = std::max(X2_latest[k], non_peak_power);
>   }
>   for (int k = peak_bin + 5;
>@@ -60,7 +61,7 @@ void IdentifyStrongNarrowBandComponent(const RenderBuffer& render_buffer,
>     non_peak_power = std::max(X2_latest[k], non_peak_power);
>   }
> 
>-  // Assess the render signal strength
>+  // Assess the render signal strength.
>   const std::vector<std::vector<float>>& x_latest = render_buffer.Block(0);
>   auto result0 = std::minmax_element(x_latest[0].begin(), x_latest[0].end());
>   float max_abs = std::max(fabs(*result0.first), fabs(*result0.second));
>@@ -74,34 +75,37 @@ void IdentifyStrongNarrowBandComponent(const RenderBuffer& render_buffer,
>   }
> 
>   // Detect whether the spectal peak has as strong narrowband nature.
>-  if (peak_bin > 6 && max_abs > 100 &&
>+  if (peak_bin > 0 && max_abs > 100 &&
>       X2_latest[peak_bin] > 100 * non_peak_power) {
>     *narrow_peak_band = peak_bin;
>     *narrow_peak_counter = 0;
>   } else {
>-    if (*narrow_peak_band && ++(*narrow_peak_counter) > 7) {
>-      *narrow_peak_band = rtc::nullopt;
>+    if (*narrow_peak_band &&
>+        ++(*narrow_peak_counter) >
>+            static_cast<size_t>(strong_peak_freeze_duration)) {
>+      *narrow_peak_band = absl::nullopt;
>     }
>   }
> }
> 
> }  // namespace
> 
>-RenderSignalAnalyzer::RenderSignalAnalyzer() {
>+RenderSignalAnalyzer::RenderSignalAnalyzer(const EchoCanceller3Config& config)
>+    : strong_peak_freeze_duration_(config.filter.main.length_blocks) {
>   narrow_band_counters_.fill(0);
> }
> RenderSignalAnalyzer::~RenderSignalAnalyzer() = default;
> 
> void RenderSignalAnalyzer::Update(
>     const RenderBuffer& render_buffer,
>-    const rtc::Optional<size_t>& delay_partitions) {
>+    const absl::optional<size_t>& delay_partitions) {
>   // Identify bands of narrow nature.
>   IdentifySmallNarrowBandRegions(render_buffer, delay_partitions,
>                                  &narrow_band_counters_);
> 
>   // Identify the presence of a strong narrow band.
>-  IdentifyStrongNarrowBandComponent(render_buffer, &narrow_peak_band_,
>-                                    &narrow_peak_counter_);
>+  IdentifyStrongNarrowBandComponent(render_buffer, strong_peak_freeze_duration_,
>+                                    &narrow_peak_band_, &narrow_peak_counter_);
> }
> 
> void RenderSignalAnalyzer::MaskRegionsAroundNarrowBands(
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_signal_analyzer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_signal_analyzer.h
>index 64d74f4e3f5..c603c921447 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_signal_analyzer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_signal_analyzer.h
>@@ -14,7 +14,8 @@
> #include <array>
> #include <memory>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
>+#include "api/audio/echo_canceller3_config.h"
> #include "modules/audio_processing/aec3/aec3_common.h"
> #include "modules/audio_processing/aec3/render_buffer.h"
> #include "rtc_base/constructormagic.h"
>@@ -24,12 +25,12 @@ namespace webrtc {
> // Provides functionality for analyzing the properties of the render signal.
> class RenderSignalAnalyzer {
>  public:
>-  RenderSignalAnalyzer();
>+  explicit RenderSignalAnalyzer(const EchoCanceller3Config& config);
>   ~RenderSignalAnalyzer();
> 
>   // Updates the render signal analysis with the most recent render signal.
>   void Update(const RenderBuffer& render_buffer,
>-              const rtc::Optional<size_t>& delay_partitions);
>+              const absl::optional<size_t>& delay_partitions);
> 
>   // Returns true if the render signal is poorly exciting.
>   bool PoorSignalExcitation() const {
>@@ -43,11 +44,12 @@ class RenderSignalAnalyzer {
>   void MaskRegionsAroundNarrowBands(
>       std::array<float, kFftLengthBy2Plus1>* v) const;
> 
>-  rtc::Optional<int> NarrowPeakBand() const { return narrow_peak_band_; }
>+  absl::optional<int> NarrowPeakBand() const { return narrow_peak_band_; }
> 
>  private:
>+  const int strong_peak_freeze_duration_;
>   std::array<size_t, kFftLengthBy2 - 1> narrow_band_counters_;
>-  rtc::Optional<int> narrow_peak_band_;
>+  absl::optional<int> narrow_peak_band_;
>   size_t narrow_peak_counter_;
> 
>   RTC_DISALLOW_COPY_AND_ASSIGN(RenderSignalAnalyzer);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_signal_analyzer_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_signal_analyzer_unittest.cc
>index 8b68960beea..f9b19558387 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_signal_analyzer_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/render_signal_analyzer_unittest.cc
>@@ -46,7 +46,7 @@ void ProduceSinusoid(int sample_rate_hz,
> #if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
> // Verifies that the check for non-null output parameter works.
> TEST(RenderSignalAnalyzer, NullMaskOutput) {
>-  RenderSignalAnalyzer analyzer;
>+  RenderSignalAnalyzer analyzer(EchoCanceller3Config{});
>   EXPECT_DEATH(analyzer.MaskRegionsAroundNarrowBands(nullptr), "");
> }
> 
>@@ -54,7 +54,7 @@ TEST(RenderSignalAnalyzer, NullMaskOutput) {
> 
> // Verify that no narrow bands are detected in a Gaussian noise signal.
> TEST(RenderSignalAnalyzer, NoFalseDetectionOfNarrowBands) {
>-  RenderSignalAnalyzer analyzer;
>+  RenderSignalAnalyzer analyzer(EchoCanceller3Config{});
>   Random random_generator(42U);
>   std::vector<std::vector<float>> x(3, std::vector<float>(kBlockSize, 0.f));
>   std::array<float, kBlockSize> x_old;
>@@ -73,7 +73,7 @@ TEST(RenderSignalAnalyzer, NoFalseDetectionOfNarrowBands) {
>     render_delay_buffer->PrepareCaptureProcessing();
> 
>     analyzer.Update(*render_delay_buffer->GetRenderBuffer(),
>-                    rtc::Optional<size_t>(0));
>+                    absl::optional<size_t>(0));
>   }
> 
>   mask.fill(1.f);
>@@ -85,7 +85,7 @@ TEST(RenderSignalAnalyzer, NoFalseDetectionOfNarrowBands) {
> 
> // Verify that a sinusiod signal is detected as narrow bands.
> TEST(RenderSignalAnalyzer, NarrowBandDetection) {
>-  RenderSignalAnalyzer analyzer;
>+  RenderSignalAnalyzer analyzer(EchoCanceller3Config{});
>   Random random_generator(42U);
>   std::vector<std::vector<float>> x(3, std::vector<float>(kBlockSize, 0.f));
>   std::array<float, kBlockSize> x_old;
>@@ -112,7 +112,7 @@ TEST(RenderSignalAnalyzer, NarrowBandDetection) {
>       render_delay_buffer->PrepareCaptureProcessing();
> 
>       analyzer.Update(*render_delay_buffer->GetRenderBuffer(),
>-                      known_delay ? rtc::Optional<size_t>(0) : rtc::nullopt);
>+                      known_delay ? absl::optional<size_t>(0) : absl::nullopt);
>     }
>   };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/residual_echo_estimator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/residual_echo_estimator.cc
>index f0c971dd5ab..2e3ad9fea72 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/residual_echo_estimator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/residual_echo_estimator.cc
>@@ -1,3 +1,4 @@
>+
> /*
>  *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>  *
>@@ -13,71 +14,79 @@
> #include <numeric>
> #include <vector>
> 
>+#include "modules/audio_processing/aec3/reverb_model.h"
>+#include "modules/audio_processing/aec3/reverb_model_fallback.h"
> #include "rtc_base/checks.h"
>+#include "system_wrappers/include/field_trial.h"
> 
> namespace webrtc {
> namespace {
> 
>-// Estimates the echo generating signal power as gated maximal power over a time
>-// window.
>-void EchoGeneratingPower(const RenderBuffer& render_buffer,
>-                         size_t min_delay,
>-                         size_t max_delay,
>-                         std::array<float, kFftLengthBy2Plus1>* X2) {
>-  X2->fill(0.f);
>-  for (size_t k = min_delay; k <= max_delay; ++k) {
>-    std::transform(X2->begin(), X2->end(), render_buffer.Spectrum(k).begin(),
>-                   X2->begin(),
>-                   [](float a, float b) { return std::max(a, b); });
>-  }
>-
>-  // Apply soft noise gate of -78 dBFS.
>-  static constexpr float kNoiseGatePower = 27509.42f;
>-  std::for_each(X2->begin(), X2->end(), [](float& a) {
>-    if (kNoiseGatePower > a) {
>-      a = std::max(0.f, a - 0.3f * (kNoiseGatePower - a));
>-    }
>-  });
>+bool EnableSoftTransparentMode() {
>+  return !field_trial::IsEnabled("WebRTC-Aec3SoftTransparentModeKillSwitch");
> }
> 
>-constexpr int kNoiseFloorCounterMax = 50;
>-constexpr float kNoiseFloorMin = 10.f * 10.f * 128.f * 128.f;
>-
>-// Updates estimate for the power of the stationary noise component in the
>-// render signal.
>-void RenderNoisePower(
>-    const RenderBuffer& render_buffer,
>-    std::array<float, kFftLengthBy2Plus1>* X2_noise_floor,
>-    std::array<int, kFftLengthBy2Plus1>* X2_noise_floor_counter) {
>-  RTC_DCHECK(X2_noise_floor);
>-  RTC_DCHECK(X2_noise_floor_counter);
>+bool OverrideEstimatedEchoPathGain() {
>+  return !field_trial::IsEnabled("WebRTC-Aec3OverrideEchoPathGainKillSwitch");
>+}
> 
>-  const auto render_power = render_buffer.Spectrum(0);
>-  RTC_DCHECK_EQ(X2_noise_floor->size(), render_power.size());
>-  RTC_DCHECK_EQ(X2_noise_floor_counter->size(), render_power.size());
>+bool UseFixedNonLinearReverbModel() {
>+  return field_trial::IsEnabled(
>+      "WebRTC-Aec3StandardNonlinearReverbModelKillSwitch");
>+}
> 
>-  // Estimate the stationary noise power in a minimum statistics manner.
>-  for (size_t k = 0; k < render_power.size(); ++k) {
>-    // Decrease rapidly.
>-    if (render_power[k] < (*X2_noise_floor)[k]) {
>-      (*X2_noise_floor)[k] = render_power[k];
>-      (*X2_noise_floor_counter)[k] = 0;
>+// Computes the indexes that will be used for computing spectral power over
>+// the blocks surrounding the delay.
>+void GetRenderIndexesToAnalyze(
>+    const VectorBuffer& spectrum_buffer,
>+    const EchoCanceller3Config::EchoModel& echo_model,
>+    int filter_delay_blocks,
>+    bool gain_limiter_running,
>+    int headroom,
>+    int* idx_start,
>+    int* idx_stop) {
>+  RTC_DCHECK(idx_start);
>+  RTC_DCHECK(idx_stop);
>+  if (gain_limiter_running) {
>+    if (static_cast<size_t>(headroom) >
>+        echo_model.render_post_window_size_init) {
>+      *idx_start = spectrum_buffer.OffsetIndex(
>+          spectrum_buffer.read,
>+          -static_cast<int>(echo_model.render_post_window_size_init));
>     } else {
>-      // Increase in a delayed, leaky manner.
>-      if ((*X2_noise_floor_counter)[k] >= kNoiseFloorCounterMax) {
>-        (*X2_noise_floor)[k] =
>-            std::max((*X2_noise_floor)[k] * 1.1f, kNoiseFloorMin);
>-      } else {
>-        ++(*X2_noise_floor_counter)[k];
>-      }
>+      *idx_start = spectrum_buffer.IncIndex(spectrum_buffer.write);
>     }
>+
>+    *idx_stop = spectrum_buffer.OffsetIndex(
>+        spectrum_buffer.read, echo_model.render_pre_window_size_init);
>+  } else {
>+    size_t window_start;
>+    size_t window_end;
>+    window_start =
>+        std::max(0, filter_delay_blocks -
>+                        static_cast<int>(echo_model.render_pre_window_size));
>+    window_end = filter_delay_blocks +
>+                 static_cast<int>(echo_model.render_post_window_size);
>+    *idx_start =
>+        spectrum_buffer.OffsetIndex(spectrum_buffer.read, window_start);
>+    *idx_stop =
>+        spectrum_buffer.OffsetIndex(spectrum_buffer.read, window_end + 1);
>   }
> }
> 
> }  // namespace
> 
> ResidualEchoEstimator::ResidualEchoEstimator(const EchoCanceller3Config& config)
>-    : config_(config), S2_old_(config_.filter.main.length_blocks) {
>+    : config_(config),
>+      soft_transparent_mode_(EnableSoftTransparentMode()),
>+      override_estimated_echo_path_gain_(OverrideEstimatedEchoPathGain()),
>+      use_fixed_nonlinear_reverb_model_(UseFixedNonLinearReverbModel()) {
>+  if (config_.ep_strength.reverb_based_on_render) {
>+    echo_reverb_.reset(new ReverbModel());
>+  } else {
>+    echo_reverb_fallback.reset(
>+        new ReverbModelFallback(config_.filter.main.length_blocks));
>+  }
>   Reset();
> }
> 
>@@ -96,169 +105,216 @@ void ResidualEchoEstimator::Estimate(
> 
>   // Estimate the residual echo power.
>   if (aec_state.UsableLinearEstimate()) {
>-    LinearEstimate(S2_linear, aec_state.Erle(), aec_state.FilterDelay(), R2);
>-    AddEchoReverb(S2_linear, aec_state.SaturatedEcho(), aec_state.FilterDelay(),
>-                  aec_state.ReverbDecay(), R2);
>+    RTC_DCHECK(!aec_state.SaturatedEcho());
>+    LinearEstimate(S2_linear, aec_state.Erle(), aec_state.ErleUncertainty(),
>+                   R2);
>+    // Adds the estimated unmodelled echo power to the residual echo power
>+    // estimate.
>+    if (echo_reverb_) {
>+      echo_reverb_->AddReverb(
>+          render_buffer.Spectrum(aec_state.FilterLengthBlocks() + 1),
>+          aec_state.GetReverbFrequencyResponse(), aec_state.ReverbDecay(), *R2);
> 
>-    // If the echo is saturated, estimate the echo power as the maximum echo
>-    // power with a leakage factor.
>-    if (aec_state.SaturatedEcho()) {
>-      R2->fill((*std::max_element(R2->begin(), R2->end())) * 100.f);
>+    } else {
>+      RTC_DCHECK(echo_reverb_fallback);
>+      echo_reverb_fallback->AddEchoReverb(S2_linear,
>+                                          aec_state.FilterDelayBlocks(),
>+                                          aec_state.ReverbDecay(), R2);
>     }
>+
>   } else {
>     // Estimate the echo generating signal power.
>     std::array<float, kFftLengthBy2Plus1> X2;
> 
>-    // Computes the spectral power over the blocks surrounding the delay.
>-    EchoGeneratingPower(render_buffer, std::max(0, aec_state.FilterDelay() - 1),
>-                        aec_state.FilterDelay() + 10, &X2);
>+    EchoGeneratingPower(render_buffer.GetSpectrumBuffer(), config_.echo_model,
>+                        render_buffer.Headroom(), aec_state.FilterDelayBlocks(),
>+                        aec_state.IsSuppressionGainLimitActive(),
>+                        !aec_state.UseStationaryProperties(), &X2);
> 
>     // Subtract the stationary noise power to avoid stationary noise causing
>     // excessive echo suppression.
>-    std::transform(
>-        X2.begin(), X2.end(), X2_noise_floor_.begin(), X2.begin(),
>-        [](float a, float b) { return std::max(0.f, a - 10.f * b); });
>-
>-    NonLinearEstimate(aec_state.FilterHasHadTimeToConverge(),
>-                      aec_state.SaturatedEcho(),
>-                      config_.ep_strength.bounded_erl,
>-                      aec_state.TransparentMode(), X2, Y2, R2);
>+    std::transform(X2.begin(), X2.end(), X2_noise_floor_.begin(), X2.begin(),
>+                   [&](float a, float b) {
>+                     return std::max(
>+                         0.f, a - config_.echo_model.stationary_gate_slope * b);
>+                   });
>+
>+    float echo_path_gain;
>+    if (override_estimated_echo_path_gain_) {
>+      echo_path_gain = aec_state.TransparentMode() && soft_transparent_mode_
>+                           ? 0.01f
>+                           : config_.ep_strength.lf;
>+    } else {
>+      echo_path_gain = aec_state.TransparentMode() && soft_transparent_mode_
>+                           ? 0.01f
>+                           : aec_state.EchoPathGain();
>+    }
>+    NonLinearEstimate(echo_path_gain, X2, Y2, R2);
> 
>+    // If the echo is saturated, estimate the echo power as the maximum echo
>+    // power with a leakage factor.
>     if (aec_state.SaturatedEcho()) {
>-      // TODO(peah): Modify to make sense theoretically.
>-      AddEchoReverb(*R2, aec_state.SaturatedEcho(),
>-                    config_.filter.main.length_blocks, aec_state.ReverbDecay(),
>-                    R2);
>+      R2->fill((*std::max_element(R2->begin(), R2->end())) * 100.f);
>+    }
>+
>+    if (!(aec_state.TransparentMode() && soft_transparent_mode_)) {
>+      if (echo_reverb_) {
>+        echo_reverb_->AddReverbNoFreqShaping(
>+            render_buffer.Spectrum(aec_state.FilterDelayBlocks() + 1),
>+            echo_path_gain * echo_path_gain, aec_state.ReverbDecay(), *R2);
>+      } else {
>+        RTC_DCHECK(echo_reverb_fallback);
>+        echo_reverb_fallback->AddEchoReverb(*R2,
>+                                            config_.filter.main.length_blocks,
>+                                            aec_state.ReverbDecay(), R2);
>+      }
>     }
>   }
> 
>-  // If the echo is deemed inaudible, set the residual echo to zero.
>-  if (aec_state.InaudibleEcho()) {
>-    R2->fill(0.f);
>-    R2_old_.fill(0.f);
>-    R2_hold_counter_.fill(0.f);
>+  if (aec_state.UseStationaryProperties()) {
>+    // Scale the echo according to echo audibility.
>+    std::array<float, kFftLengthBy2Plus1> residual_scaling;
>+    aec_state.GetResidualEchoScaling(residual_scaling);
>+    for (size_t k = 0; k < R2->size(); ++k) {
>+      (*R2)[k] *= residual_scaling[k];
>+      if (residual_scaling[k] == 0.f) {
>+        R2_hold_counter_[k] = 0;
>+      }
>+    }
>+  }
>+  if (!soft_transparent_mode_) {
>+    // If the echo is deemed inaudible, set the residual echo to zero.
>+    if (aec_state.TransparentMode()) {
>+      R2->fill(0.f);
>+      R2_old_.fill(0.f);
>+      R2_hold_counter_.fill(0.f);
>+    }
>   }
> 
>   std::copy(R2->begin(), R2->end(), R2_old_.begin());
> }
> 
> void ResidualEchoEstimator::Reset() {
>-  X2_noise_floor_counter_.fill(kNoiseFloorCounterMax);
>-  X2_noise_floor_.fill(kNoiseFloorMin);
>-  R2_reverb_.fill(0.f);
>+  if (echo_reverb_) {
>+    echo_reverb_->Reset();
>+  } else {
>+    RTC_DCHECK(echo_reverb_fallback);
>+    echo_reverb_fallback->Reset();
>+  }
>+  X2_noise_floor_counter_.fill(config_.echo_model.noise_floor_hold);
>+  X2_noise_floor_.fill(config_.echo_model.min_noise_floor_power);
>   R2_old_.fill(0.f);
>   R2_hold_counter_.fill(0.f);
>-  for (auto& S2_k : S2_old_) {
>-    S2_k.fill(0.f);
>-  }
> }
> 
> void ResidualEchoEstimator::LinearEstimate(
>     const std::array<float, kFftLengthBy2Plus1>& S2_linear,
>     const std::array<float, kFftLengthBy2Plus1>& erle,
>-    size_t delay,
>+    absl::optional<float> erle_uncertainty,
>     std::array<float, kFftLengthBy2Plus1>* R2) {
>   std::fill(R2_hold_counter_.begin(), R2_hold_counter_.end(), 10.f);
>-  std::transform(erle.begin(), erle.end(), S2_linear.begin(), R2->begin(),
>-                 [](float a, float b) {
>-                   RTC_DCHECK_LT(0.f, a);
>-                   return b / a;
>-                 });
>+  if (erle_uncertainty) {
>+    for (size_t k = 0; k < R2->size(); ++k) {
>+      (*R2)[k] = S2_linear[k] * *erle_uncertainty;
>+    }
>+  } else {
>+    std::transform(erle.begin(), erle.end(), S2_linear.begin(), R2->begin(),
>+                   [](float a, float b) {
>+                     RTC_DCHECK_LT(0.f, a);
>+                     return b / a;
>+                   });
>+  }
> }
> 
> void ResidualEchoEstimator::NonLinearEstimate(
>-    bool sufficient_filter_updates,
>-    bool saturated_echo,
>-    bool bounded_erl,
>-    bool transparent_mode,
>+    float echo_path_gain,
>     const std::array<float, kFftLengthBy2Plus1>& X2,
>     const std::array<float, kFftLengthBy2Plus1>& Y2,
>     std::array<float, kFftLengthBy2Plus1>* R2) {
>-  float echo_path_gain_lf;
>-  float echo_path_gain_mf;
>-  float echo_path_gain_hf;
>-
>-  // Set echo path gains.
>-  if (saturated_echo) {
>-    // If the echo could be saturated, use a very conservative gain.
>-    echo_path_gain_lf = echo_path_gain_mf = echo_path_gain_hf = 10000.f;
>-  } else if (sufficient_filter_updates && !bounded_erl) {
>-    // If the filter should have been able to converge, and no assumption is
>-    // possible on the ERL, use a low gain.
>-    echo_path_gain_lf = echo_path_gain_mf = echo_path_gain_hf = 0.01f;
>-  } else if ((sufficient_filter_updates && bounded_erl) || transparent_mode) {
>-    // If the filter should have been able to converge, and and it is known that
>-    // the ERL is bounded, use a very low gain.
>-    echo_path_gain_lf = echo_path_gain_mf = echo_path_gain_hf = 0.001f;
>-  } else {
>-    // In the initial state, use conservative gains.
>-    echo_path_gain_lf = config_.ep_strength.lf;
>-    echo_path_gain_mf = config_.ep_strength.mf;
>-    echo_path_gain_hf = config_.ep_strength.hf;
>+  // Compute preliminary residual echo.
>+  std::transform(X2.begin(), X2.end(), R2->begin(), [echo_path_gain](float a) {
>+    return a * echo_path_gain * echo_path_gain;
>+  });
>+
>+  if (use_fixed_nonlinear_reverb_model_) {
>+    for (size_t k = 0; k < R2->size(); ++k) {
>+      // Update hold counter.
>+      R2_hold_counter_[k] = R2_old_[k] < (*R2)[k] ? 0 : R2_hold_counter_[k] + 1;
>+
>+      // Compute the residual echo by holding a maximum echo powers and an echo
>+      // fading corresponding to a room with an RT60 value of about 50 ms.
>+      (*R2)[k] =
>+          R2_hold_counter_[k] < config_.echo_model.nonlinear_hold
>+              ? std::max((*R2)[k], R2_old_[k])
>+              : std::min((*R2)[k] +
>+                             R2_old_[k] * config_.echo_model.nonlinear_release,
>+                         Y2[k]);
>+    }
>   }
>+}
> 
>-  // Compute preliminary residual echo.
>-  std::transform(
>-      X2.begin(), X2.begin() + 12, R2->begin(),
>-      [echo_path_gain_lf](float a) { return a * echo_path_gain_lf; });
>-  std::transform(
>-      X2.begin() + 12, X2.begin() + 25, R2->begin() + 12,
>-      [echo_path_gain_mf](float a) { return a * echo_path_gain_mf; });
>-  std::transform(
>-      X2.begin() + 25, X2.end(), R2->begin() + 25,
>-      [echo_path_gain_hf](float a) { return a * echo_path_gain_hf; });
>-
>-  for (size_t k = 0; k < R2->size(); ++k) {
>-    // Update hold counter.
>-    R2_hold_counter_[k] = R2_old_[k] < (*R2)[k] ? 0 : R2_hold_counter_[k] + 1;
>-
>-    // Compute the residual echo by holding a maximum echo powers and an echo
>-    // fading corresponding to a room with an RT60 value of about 50 ms.
>-    (*R2)[k] = R2_hold_counter_[k] < 2
>-                   ? std::max((*R2)[k], R2_old_[k])
>-                   : std::min((*R2)[k] + R2_old_[k] * 0.1f, Y2[k]);
>+void ResidualEchoEstimator::EchoGeneratingPower(
>+    const VectorBuffer& spectrum_buffer,
>+    const EchoCanceller3Config::EchoModel& echo_model,
>+    int headroom_spectrum_buffer,
>+    int filter_delay_blocks,
>+    bool gain_limiter_running,
>+    bool apply_noise_gating,
>+    std::array<float, kFftLengthBy2Plus1>* X2) const {
>+  int idx_stop, idx_start;
>+
>+  RTC_DCHECK(X2);
>+  GetRenderIndexesToAnalyze(spectrum_buffer, config_.echo_model,
>+                            filter_delay_blocks, gain_limiter_running,
>+                            headroom_spectrum_buffer, &idx_start, &idx_stop);
>+
>+  X2->fill(0.f);
>+  for (int k = idx_start; k != idx_stop; k = spectrum_buffer.IncIndex(k)) {
>+    std::transform(X2->begin(), X2->end(), spectrum_buffer.buffer[k].begin(),
>+                   X2->begin(),
>+                   [](float a, float b) { return std::max(a, b); });
>+  }
>+
>+  if (apply_noise_gating) {
>+    // Apply soft noise gate.
>+    std::for_each(X2->begin(), X2->end(), [&](float& a) {
>+      if (config_.echo_model.noise_gate_power > a) {
>+        a = std::max(0.f, a - config_.echo_model.noise_gate_slope *
>+                                  (config_.echo_model.noise_gate_power - a));
>+      }
>+    });
>   }
> }
> 
>-void ResidualEchoEstimator::AddEchoReverb(
>-    const std::array<float, kFftLengthBy2Plus1>& S2,
>-    bool saturated_echo,
>-    size_t delay,
>-    float reverb_decay_factor,
>-    std::array<float, kFftLengthBy2Plus1>* R2) {
>-  // Compute the decay factor for how much the echo has decayed before leaving
>-  // the region covered by the linear model.
>-  auto integer_power = [](float base, int exp) {
>-    float result = 1.f;
>-    for (int k = 0; k < exp; ++k) {
>-      result *= base;
>+void ResidualEchoEstimator::RenderNoisePower(
>+    const RenderBuffer& render_buffer,
>+    std::array<float, kFftLengthBy2Plus1>* X2_noise_floor,
>+    std::array<int, kFftLengthBy2Plus1>* X2_noise_floor_counter) const {
>+  RTC_DCHECK(X2_noise_floor);
>+  RTC_DCHECK(X2_noise_floor_counter);
>+
>+  const auto render_power = render_buffer.Spectrum(0);
>+  RTC_DCHECK_EQ(X2_noise_floor->size(), render_power.size());
>+  RTC_DCHECK_EQ(X2_noise_floor_counter->size(), render_power.size());
>+
>+  // Estimate the stationary noise power in a minimum statistics manner.
>+  for (size_t k = 0; k < render_power.size(); ++k) {
>+    // Decrease rapidly.
>+    if (render_power[k] < (*X2_noise_floor)[k]) {
>+      (*X2_noise_floor)[k] = render_power[k];
>+      (*X2_noise_floor_counter)[k] = 0;
>+    } else {
>+      // Increase in a delayed, leaky manner.
>+      if ((*X2_noise_floor_counter)[k] >=
>+          static_cast<int>(config_.echo_model.noise_floor_hold)) {
>+        (*X2_noise_floor)[k] =
>+            std::max((*X2_noise_floor)[k] * 1.1f,
>+                     config_.echo_model.min_noise_floor_power);
>+      } else {
>+        ++(*X2_noise_floor_counter)[k];
>+      }
>     }
>-    return result;
>-  };
>-  RTC_DCHECK_LE(delay, S2_old_.size());
>-  const float reverb_decay_for_delay =
>-      integer_power(reverb_decay_factor, S2_old_.size() - delay);
>-
>-  // Update the estimate of the reverberant residual echo power.
>-  S2_old_index_ = S2_old_index_ > 0 ? S2_old_index_ - 1 : S2_old_.size() - 1;
>-  const auto& S2_end = S2_old_[S2_old_index_];
>-  std::transform(
>-      S2_end.begin(), S2_end.end(), R2_reverb_.begin(), R2_reverb_.begin(),
>-      [reverb_decay_for_delay, reverb_decay_factor](float a, float b) {
>-        return (b + a * reverb_decay_for_delay) * reverb_decay_factor;
>-      });
>-
>-  // Update the buffer of old echo powers.
>-  if (saturated_echo) {
>-    S2_old_[S2_old_index_].fill((*std::max_element(S2.begin(), S2.end())) *
>-                                100.f);
>-  } else {
>-    std::copy(S2.begin(), S2.end(), S2_old_[S2_old_index_].begin());
>   }
>-
>-  // Add the power of the echo reverb to the residual echo power.
>-  std::transform(R2->begin(), R2->end(), R2_reverb_.begin(), R2->begin(),
>-                 std::plus<float>());
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/residual_echo_estimator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/residual_echo_estimator.h
>index f2ecea19ccc..6dcf24f0907 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/residual_echo_estimator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/residual_echo_estimator.h
>@@ -13,13 +13,16 @@
> 
> #include <algorithm>
> #include <array>
>+#include <memory>
> #include <vector>
> 
> #include "api/array_view.h"
>+#include "api/audio/echo_canceller3_config.h"
> #include "modules/audio_processing/aec3/aec3_common.h"
> #include "modules/audio_processing/aec3/aec_state.h"
> #include "modules/audio_processing/aec3/render_buffer.h"
>-#include "modules/audio_processing/include/audio_processing.h"
>+#include "modules/audio_processing/aec3/reverb_model.h"
>+#include "modules/audio_processing/aec3/reverb_model_fallback.h"
> #include "rtc_base/constructormagic.h"
> 
> namespace webrtc {
>@@ -35,6 +38,16 @@ class ResidualEchoEstimator {
>                 const std::array<float, kFftLengthBy2Plus1>& Y2,
>                 std::array<float, kFftLengthBy2Plus1>* R2);
> 
>+  // Returns the reverberant power spectrum contributions to the echo residual.
>+  const std::array<float, kFftLengthBy2Plus1>& GetReverbPowerSpectrum() const {
>+    if (echo_reverb_) {
>+      return echo_reverb_->GetPowerSpectrum();
>+    } else {
>+      RTC_DCHECK(echo_reverb_fallback);
>+      return echo_reverb_fallback->GetPowerSpectrum();
>+    }
>+  }
>+
>  private:
>   // Resets the state.
>   void Reset();
>@@ -43,35 +56,44 @@ class ResidualEchoEstimator {
>   // (ERLE) and the linear power estimate.
>   void LinearEstimate(const std::array<float, kFftLengthBy2Plus1>& S2_linear,
>                       const std::array<float, kFftLengthBy2Plus1>& erle,
>-                      size_t delay,
>+                      absl::optional<float> erle_uncertainty,
>                       std::array<float, kFftLengthBy2Plus1>* R2);
> 
>   // Estimates the residual echo power based on the estimate of the echo path
>   // gain.
>-  void NonLinearEstimate(bool sufficient_filter_updates,
>-                         bool saturated_echo,
>-                         bool bounded_erl,
>-                         bool transparent_mode,
>+  void NonLinearEstimate(float echo_path_gain,
>                          const std::array<float, kFftLengthBy2Plus1>& X2,
>                          const std::array<float, kFftLengthBy2Plus1>& Y2,
>                          std::array<float, kFftLengthBy2Plus1>* R2);
> 
>-  // Adds the estimated unmodelled echo power to the residual echo power
>-  // estimate.
>-  void AddEchoReverb(const std::array<float, kFftLengthBy2Plus1>& S2,
>-                     bool saturated_echo,
>-                     size_t delay,
>-                     float reverb_decay_factor,
>-                     std::array<float, kFftLengthBy2Plus1>* R2);
>+
>+  // Estimates the echo generating signal power as gated maximal power over a
>+  // time window.
>+  void EchoGeneratingPower(const VectorBuffer& spectrum_buffer,
>+                           const EchoCanceller3Config::EchoModel& echo_model,
>+                           int headroom_spectrum_buffer,
>+                           int filter_delay_blocks,
>+                           bool gain_limiter_running,
>+                           bool apply_noise_gating,
>+                           std::array<float, kFftLengthBy2Plus1>* X2) const;
>+
>+  // Updates estimate for the power of the stationary noise component in the
>+  // render signal.
>+  void RenderNoisePower(
>+      const RenderBuffer& render_buffer,
>+      std::array<float, kFftLengthBy2Plus1>* X2_noise_floor,
>+      std::array<int, kFftLengthBy2Plus1>* X2_noise_floor_counter) const;
>+
>   const EchoCanceller3Config config_;
>   std::array<float, kFftLengthBy2Plus1> R2_old_;
>   std::array<int, kFftLengthBy2Plus1> R2_hold_counter_;
>-  std::array<float, kFftLengthBy2Plus1> R2_reverb_;
>-  int S2_old_index_ = 0;
>-  std::vector<std::array<float, kFftLengthBy2Plus1>> S2_old_;
>   std::array<float, kFftLengthBy2Plus1> X2_noise_floor_;
>   std::array<int, kFftLengthBy2Plus1> X2_noise_floor_counter_;
>-
>+  const bool soft_transparent_mode_;
>+  const bool override_estimated_echo_path_gain_;
>+  const bool use_fixed_nonlinear_reverb_model_;
>+  std::unique_ptr<ReverbModel> echo_reverb_;
>+  std::unique_ptr<ReverbModelFallback> echo_reverb_fallback;
>   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(ResidualEchoEstimator);
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/residual_echo_estimator_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/residual_echo_estimator_unittest.cc
>index e617a168eaf..832d8ca1d33 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/residual_echo_estimator_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/residual_echo_estimator_unittest.cc
>@@ -10,10 +10,10 @@
> 
> #include "modules/audio_processing/aec3/residual_echo_estimator.h"
> 
>+#include "api/audio/echo_canceller3_config.h"
> #include "modules/audio_processing/aec3/aec3_fft.h"
> #include "modules/audio_processing/aec3/aec_state.h"
> #include "modules/audio_processing/aec3/render_delay_buffer.h"
>-#include "modules/audio_processing/include/audio_processing.h"
> #include "modules/audio_processing/test/echo_canceller_test_tools.h"
> #include "rtc_base/random.h"
> #include "test/gtest.h"
>@@ -61,8 +61,10 @@ TEST(ResidualEchoEstimator, DISABLED_BasicTest) {
>   std::vector<std::vector<float>> x(3, std::vector<float>(kBlockSize, 0.f));
>   std::vector<std::array<float, kFftLengthBy2Plus1>> H2(10);
>   Random random_generator(42U);
>-  std::array<float, kBlockSize> s;
>+  SubtractorOutput output;
>+  std::array<float, kBlockSize> y;
>   Aec3Fft fft;
>+  absl::optional<DelayEstimate> delay_estimate;
> 
>   for (auto& H2_k : H2) {
>     H2_k.fill(0.01f);
>@@ -73,7 +75,9 @@ TEST(ResidualEchoEstimator, DISABLED_BasicTest) {
>   std::vector<float> h(GetTimeDomainLength(config.filter.main.length_blocks),
>                        0.f);
> 
>-  s.fill(100.f);
>+  output.Reset();
>+  output.s_main.fill(100.f);
>+  y.fill(0.f);
> 
>   constexpr float kLevel = 10.f;
>   E2_shadow.fill(kLevel);
>@@ -92,8 +96,9 @@ TEST(ResidualEchoEstimator, DISABLED_BasicTest) {
>     render_delay_buffer->PrepareCaptureProcessing();
> 
>     aec_state.HandleEchoPathChange(echo_path_variability);
>-    aec_state.Update(H2, h, true, *render_delay_buffer->GetRenderBuffer(),
>-                     E2_main, Y2, s, false);
>+    aec_state.Update(delay_estimate, H2, h,
>+                     *render_delay_buffer->GetRenderBuffer(), E2_main, Y2,
>+                     output, y);
> 
>     estimator.Estimate(aec_state, *render_delay_buffer->GetRenderBuffer(),
>                        S2_linear, Y2, &R2);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_decay_estimator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_decay_estimator.cc
>new file mode 100644
>index 00000000000..427a54fce51
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_decay_estimator.cc
>@@ -0,0 +1,361 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/aec3/reverb_decay_estimator.h"
>+
>+#include <algorithm>
>+#include <cmath>
>+#include <numeric>
>+
>+#include "api/array_view.h"
>+#include "modules/audio_processing/logging/apm_data_dumper.h"
>+#include "rtc_base/checks.h"
>+#include "system_wrappers/include/field_trial.h"
>+
>+namespace webrtc {
>+
>+namespace {
>+
>+bool EnforceAdaptiveEchoReverbEstimation() {
>+  return field_trial::IsEnabled(
>+      "WebRTC-Aec3EnableAdaptiveEchoReverbEstimation");
>+}
>+
>+constexpr int kEarlyReverbMinSizeBlocks = 3;
>+constexpr int kBlocksPerSection = 3;
>+
>+// Averages the values in a block of size kFftLengthBy2;
>+float BlockAverage(rtc::ArrayView<const float> v, size_t block_index) {
>+  constexpr float kOneByFftLengthBy2 = 1.f / kFftLengthBy2;
>+  const int i = block_index * kFftLengthBy2;
>+  RTC_DCHECK_GE(v.size(), i + kFftLengthBy2);
>+  const float sum =
>+      std::accumulate(v.begin() + i, v.begin() + i + kFftLengthBy2, 0.f);
>+  return sum * kOneByFftLengthBy2;
>+}
>+
>+// Analyzes the gain in a block.
>+void AnalyzeBlockGain(const std::array<float, kFftLengthBy2>& h2,
>+                      float floor_gain,
>+                      float* previous_gain,
>+                      bool* block_adapting,
>+                      bool* decaying_gain) {
>+  float gain = std::max(BlockAverage(h2, 0), 1e-32f);
>+  *block_adapting =
>+      *previous_gain > 1.1f * gain || *previous_gain < 0.9f * gain;
>+  *decaying_gain = gain > floor_gain;
>+  *previous_gain = gain;
>+}
>+
>+// Arithmetic sum of $2 \sum_{i=0.5}^{(N-1)/2}i^2$ calculated directly.
>+constexpr float SymmetricArithmetricSum(int N) {
>+  return N * (N * N - 1.0f) * (1.f / 12.f);
>+}
>+
>+}  // namespace
>+
>+ReverbDecayEstimator::ReverbDecayEstimator(const EchoCanceller3Config& config)
>+    : filter_length_blocks_(config.filter.main.length_blocks),
>+      filter_length_coefficients_(GetTimeDomainLength(filter_length_blocks_)),
>+      use_adaptive_echo_decay_(config.ep_strength.default_len < 0.f ||
>+                               EnforceAdaptiveEchoReverbEstimation()),
>+      early_reverb_estimator_(config.filter.main.length_blocks -
>+                              kEarlyReverbMinSizeBlocks),
>+      late_reverb_start_(kEarlyReverbMinSizeBlocks),
>+      late_reverb_end_(kEarlyReverbMinSizeBlocks),
>+      decay_(std::fabs(config.ep_strength.default_len)) {
>+  previous_gains_.fill(0.f);
>+  RTC_DCHECK_GT(config.filter.main.length_blocks,
>+                static_cast<size_t>(kEarlyReverbMinSizeBlocks));
>+}
>+
>+ReverbDecayEstimator::~ReverbDecayEstimator() = default;
>+
>+void ReverbDecayEstimator::Update(rtc::ArrayView<const float> filter,
>+                                  const absl::optional<float>& filter_quality,
>+                                  int filter_delay_blocks,
>+                                  bool usable_linear_filter,
>+                                  bool stationary_signal) {
>+  const int filter_size = static_cast<int>(filter.size());
>+
>+  if (stationary_signal) {
>+    return;
>+  }
>+
>+  // TODO(devicentepena): Verify that the below is correct.
>+  bool estimation_feasible =
>+      filter_delay_blocks <=
>+      filter_length_blocks_ - kEarlyReverbMinSizeBlocks - 1;
>+  estimation_feasible =
>+      estimation_feasible && filter_size == filter_length_coefficients_;
>+  estimation_feasible = estimation_feasible && filter_delay_blocks > 0;
>+  estimation_feasible = estimation_feasible && usable_linear_filter;
>+
>+  if (!estimation_feasible) {
>+    ResetDecayEstimation();
>+    return;
>+  }
>+
>+  if (!use_adaptive_echo_decay_) {
>+    return;
>+  }
>+
>+  const float new_smoothing = filter_quality ? *filter_quality * 0.2f : 0.f;
>+  smoothing_constant_ = std::max(new_smoothing, smoothing_constant_);
>+  if (smoothing_constant_ == 0.f) {
>+    return;
>+  }
>+
>+  if (block_to_analyze_ < filter_length_blocks_) {
>+    // Analyze the filter and accumulate data for reverb estimation.
>+    AnalyzeFilter(filter);
>+    ++block_to_analyze_;
>+  } else {
>+    // When the filter is fully analyzed, estimate the reverb decay and reset
>+    // the block_to_analyze_ counter.
>+    EstimateDecay(filter);
>+  }
>+}
>+
>+void ReverbDecayEstimator::ResetDecayEstimation() {
>+  early_reverb_estimator_.Reset();
>+  late_reverb_decay_estimator_.Reset(0);
>+  block_to_analyze_ = 0;
>+  estimation_region_candidate_size_ = 0;
>+  estimation_region_identified_ = false;
>+  smoothing_constant_ = 0.f;
>+  late_reverb_start_ = 0;
>+  late_reverb_end_ = 0;
>+}
>+
>+void ReverbDecayEstimator::EstimateDecay(rtc::ArrayView<const float> filter) {
>+  auto& h = filter;
>+
>+  RTC_DCHECK_EQ(0, h.size() % kFftLengthBy2);
>+
>+  // Compute the squared filter coefficients.
>+  std::array<float, GetTimeDomainLength(kMaxAdaptiveFilterLength)> h2_data;
>+  RTC_DCHECK_GE(h2_data.size(), filter_length_coefficients_);
>+  rtc::ArrayView<float> h2(h2_data.data(), filter_length_coefficients_);
>+  std::transform(h.begin(), h.end(), h2.begin(), [](float a) { return a * a; });
>+
>+  // Identify the peak index of the filter.
>+  const int peak_coefficient =
>+      std::distance(h2.begin(), std::max_element(h2.begin(), h2.end()));
>+  int peak_block = peak_coefficient >> kFftLengthBy2Log2;
>+
>+  // Reset the block analysis counter.
>+  block_to_analyze_ =
>+      std::min(peak_block + kEarlyReverbMinSizeBlocks, filter_length_blocks_);
>+
>+  // To estimate the reverb decay, the energy of the first filter section must
>+  // be substantially larger than the last. Also, the first filter section
>+  // energy must not deviate too much from the max peak.
>+  const float first_reverb_gain = BlockAverage(h2, block_to_analyze_);
>+  tail_gain_ = BlockAverage(h2, (h2.size() >> kFftLengthBy2Log2) - 1);
>+  const bool sufficient_reverb_decay = first_reverb_gain > 4.f * tail_gain_;
>+  const bool valid_filter =
>+      first_reverb_gain > 2.f * tail_gain_ && h2[peak_coefficient] < 100.f;
>+
>+  // Estimate the size of the regions with early and late reflections.
>+  const int size_early_reverb = early_reverb_estimator_.Estimate();
>+  const int size_late_reverb =
>+      std::max(estimation_region_candidate_size_ - size_early_reverb, 0);
>+
>+  // Only update the reverb decay estimate if the size of the identified late
>+  // reverb is sufficiently large.
>+  if (size_late_reverb >= 5) {
>+    if (valid_filter && late_reverb_decay_estimator_.EstimateAvailable()) {
>+      float decay = std::pow(
>+          2.0f, late_reverb_decay_estimator_.Estimate() * kFftLengthBy2);
>+      constexpr float kMaxDecay = 0.95f;  // ~1 sec min RT60.
>+      constexpr float kMinDecay = 0.02f;  // ~15 ms max RT60.
>+      decay = std::max(.97f * decay_, decay);
>+      decay = std::min(decay, kMaxDecay);
>+      decay = std::max(decay, kMinDecay);
>+      decay_ += smoothing_constant_ * (decay - decay_);
>+    }
>+
>+    // Update length of decay. Must have enough data (number of sections) in
>+    // order to estimate decay rate.
>+    late_reverb_decay_estimator_.Reset(size_late_reverb * kFftLengthBy2);
>+    late_reverb_start_ =
>+        peak_block + kEarlyReverbMinSizeBlocks + size_early_reverb;
>+    late_reverb_end_ =
>+        block_to_analyze_ + estimation_region_candidate_size_ - 1;
>+  } else {
>+    late_reverb_decay_estimator_.Reset(0);
>+    late_reverb_start_ = 0;
>+    late_reverb_end_ = 0;
>+  }
>+
>+  // Reset variables for the identification of the region for reverb decay
>+  // estimation.
>+  estimation_region_identified_ = !(valid_filter && sufficient_reverb_decay);
>+  estimation_region_candidate_size_ = 0;
>+
>+  // Stop estimation of the decay until another good filter is received.
>+  smoothing_constant_ = 0.f;
>+
>+  // Reset early reflections detector.
>+  early_reverb_estimator_.Reset();
>+}
>+
>+void ReverbDecayEstimator::AnalyzeFilter(rtc::ArrayView<const float> filter) {
>+  auto h = rtc::ArrayView<const float>(
>+      filter.begin() + block_to_analyze_ * kFftLengthBy2, kFftLengthBy2);
>+
>+  // Compute squared filter coeffiecients for the block to analyze_;
>+  std::array<float, kFftLengthBy2> h2;
>+  std::transform(h.begin(), h.end(), h2.begin(), [](float a) { return a * a; });
>+
>+  // Map out the region for estimating the reverb decay.
>+  bool adapting;
>+  bool above_noise_floor;
>+  AnalyzeBlockGain(h2, tail_gain_, &previous_gains_[block_to_analyze_],
>+                   &adapting, &above_noise_floor);
>+
>+  // Count consecutive number of "good" filter sections, where "good" means:
>+  // 1) energy is above noise floor.
>+  // 2) energy of current section has not changed too much from last check.
>+  estimation_region_identified_ =
>+      estimation_region_identified_ || adapting || !above_noise_floor;
>+  if (!estimation_region_identified_) {
>+    ++estimation_region_candidate_size_;
>+  }
>+
>+  // Accumulate data for reverb decay estimation and for the estimation of early
>+  // reflections.
>+  if (block_to_analyze_ <= late_reverb_end_) {
>+    if (block_to_analyze_ >= late_reverb_start_) {
>+      for (float h2_k : h2) {
>+        float h2_log2 = FastApproxLog2f(h2_k + 1e-10);
>+        late_reverb_decay_estimator_.Accumulate(h2_log2);
>+        early_reverb_estimator_.Accumulate(h2_log2, smoothing_constant_);
>+      }
>+    } else {
>+      for (float h2_k : h2) {
>+        float h2_log2 = FastApproxLog2f(h2_k + 1e-10);
>+        early_reverb_estimator_.Accumulate(h2_log2, smoothing_constant_);
>+      }
>+    }
>+  }
>+}
>+
>+void ReverbDecayEstimator::Dump(ApmDataDumper* data_dumper) const {
>+  data_dumper->DumpRaw("aec3_reverb_decay", decay_);
>+  data_dumper->DumpRaw("aec3_reverb_tail_energy", tail_gain_);
>+  data_dumper->DumpRaw("aec3_reverb_alpha", smoothing_constant_);
>+  data_dumper->DumpRaw("aec3_num_reverb_decay_blocks",
>+                       late_reverb_end_ - late_reverb_start_);
>+  data_dumper->DumpRaw("aec3_blocks_after_early_reflections",
>+                       late_reverb_start_);
>+  early_reverb_estimator_.Dump(data_dumper);
>+}
>+
>+void ReverbDecayEstimator::LateReverbLinearRegressor::Reset(
>+    int num_data_points) {
>+  RTC_DCHECK_LE(0, num_data_points);
>+  RTC_DCHECK_EQ(0, num_data_points % 2);
>+  const int N = num_data_points;
>+  nz_ = 0.f;
>+  // Arithmetic sum of $2 \sum_{i=0.5}^{(N-1)/2}i^2$ calculated directly.
>+  nn_ = SymmetricArithmetricSum(N);
>+  // The linear regression approach assumes symmetric index around 0.
>+  count_ = N > 0 ? count_ = -N * 0.5f + 0.5f : 0.f;
>+  N_ = N;
>+  n_ = 0;
>+}
>+
>+void ReverbDecayEstimator::LateReverbLinearRegressor::Accumulate(float z) {
>+  nz_ += count_ * z;
>+  ++count_;
>+  ++n_;
>+}
>+
>+float ReverbDecayEstimator::LateReverbLinearRegressor::Estimate() {
>+  RTC_DCHECK(EstimateAvailable());
>+  if (nn_ == 0.f) {
>+    RTC_NOTREACHED();
>+    return 0.f;
>+  }
>+  return nz_ / nn_;
>+}
>+
>+ReverbDecayEstimator::EarlyReverbLengthEstimator::EarlyReverbLengthEstimator(
>+    int max_blocks)
>+    : numerators_(1 + max_blocks / kBlocksPerSection, 0.f),
>+      nz_(numerators_.size(), 0.f),
>+      count_(numerators_.size(), 0.f) {
>+  RTC_DCHECK_LE(0, max_blocks);
>+}
>+
>+ReverbDecayEstimator::EarlyReverbLengthEstimator::
>+    ~EarlyReverbLengthEstimator() = default;
>+
>+void ReverbDecayEstimator::EarlyReverbLengthEstimator::Reset() {
>+  // Linear regression approach assumes symmetric index around 0.
>+  constexpr float kCount = -0.5f * kBlocksPerSection * kFftLengthBy2 + 0.5f;
>+  std::fill(count_.begin(), count_.end(), kCount);
>+  std::fill(nz_.begin(), nz_.end(), 0.f);
>+  section_ = 0;
>+  section_update_counter_ = 0;
>+}
>+
>+void ReverbDecayEstimator::EarlyReverbLengthEstimator::Accumulate(
>+    float value,
>+    float smoothing) {
>+  nz_[section_] += count_[section_] * value;
>+  ++count_[section_];
>+
>+  if (++section_update_counter_ == kBlocksPerSection * kFftLengthBy2) {
>+    RTC_DCHECK_GT(nz_.size(), section_);
>+    RTC_DCHECK_GT(numerators_.size(), section_);
>+    numerators_[section_] +=
>+        smoothing * (nz_[section_] - numerators_[section_]);
>+    section_update_counter_ = 0;
>+    ++section_;
>+  }
>+}
>+
>+int ReverbDecayEstimator::EarlyReverbLengthEstimator::Estimate() {
>+  constexpr float N = kBlocksPerSection * kFftLengthBy2;
>+  constexpr float nn = SymmetricArithmetricSum(N);
>+  // numerator_11 refers to the quantity that the linear regressor needs in the
>+  // numerator for getting a decay equal to 1.1 (which is not a decay).
>+  // log2(1.1) * nn / kFftLengthBy2.
>+  constexpr float numerator_11 = 0.13750352374993502f * nn / kFftLengthBy2;
>+  // log2(0.8) *  nn / kFftLengthBy2.
>+  constexpr float numerator_08 = -0.32192809488736229f * nn / kFftLengthBy2;
>+  constexpr int kNumSectionsToAnalyze = 3;
>+
>+  // Analyze the first kNumSectionsToAnalyze regions.
>+  // TODO(devicentepena): Add a more thorough comment for explaining the logic
>+  // below.
>+  const float min_stable_region = *std::min_element(
>+      numerators_.begin() + kNumSectionsToAnalyze, numerators_.end());
>+  int early_reverb_size = 0;
>+  for (int k = 0; k < kNumSectionsToAnalyze; ++k) {
>+    if ((numerators_[k] > numerator_11) ||
>+        (numerators_[k] < numerator_08 &&
>+         numerators_[k] < 0.9f * min_stable_region)) {
>+      early_reverb_size = (k + 1) * kBlocksPerSection;
>+    }
>+  }
>+
>+  return early_reverb_size;
>+}
>+
>+void ReverbDecayEstimator::EarlyReverbLengthEstimator::Dump(
>+    ApmDataDumper* data_dumper) const {
>+  data_dumper->DumpRaw("aec3_er_acum_numerator", numerators_);
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_decay_estimator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_decay_estimator.h
>new file mode 100644
>index 00000000000..9ce519db78f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_decay_estimator.h
>@@ -0,0 +1,110 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_AEC3_REVERB_DECAY_ESTIMATOR_H_
>+#define MODULES_AUDIO_PROCESSING_AEC3_REVERB_DECAY_ESTIMATOR_H_
>+
>+#include <array>
>+#include <vector>
>+
>+#include "absl/types/optional.h"
>+#include "api/array_view.h"
>+#include "api/audio/echo_canceller3_config.h"
>+#include "modules/audio_processing/aec3/aec3_common.h"
>+
>+namespace webrtc {
>+
>+class ApmDataDumper;
>+
>+// Class for estimating the decay of the late reverb.
>+class ReverbDecayEstimator {
>+ public:
>+  explicit ReverbDecayEstimator(const EchoCanceller3Config& config);
>+  ~ReverbDecayEstimator();
>+  // Updates the decay estimate.
>+  void Update(rtc::ArrayView<const float> filter,
>+              const absl::optional<float>& filter_quality,
>+              int filter_delay_blocks,
>+              bool usable_linear_filter,
>+              bool stationary_signal);
>+  // Returns the decay for the exponential model.
>+  float Decay() const { return decay_; }
>+  // Dumps debug data.
>+  void Dump(ApmDataDumper* data_dumper) const;
>+
>+ private:
>+  void EstimateDecay(rtc::ArrayView<const float> filter);
>+  void AnalyzeFilter(rtc::ArrayView<const float> filter);
>+
>+  void ResetDecayEstimation();
>+
>+  // Class for estimating the decay of the late reverb from the linear filter.
>+  class LateReverbLinearRegressor {
>+   public:
>+    // Resets the estimator to receive a specified number of data points.
>+    void Reset(int num_data_points);
>+    // Accumulates estimation data.
>+    void Accumulate(float z);
>+    // Estimates the decay.
>+    float Estimate();
>+    // Returns whether an estimate is available.
>+    bool EstimateAvailable() const { return n_ == N_ && N_ != 0; }
>+
>+   public:
>+    float nz_ = 0.f;
>+    float nn_ = 0.f;
>+    float count_ = 0.f;
>+    int N_ = 0;
>+    int n_ = 0;
>+  };
>+
>+  // Class for identifying the length of the early reverb from the linear
>+  // filter.
>+  class EarlyReverbLengthEstimator {
>+   public:
>+    explicit EarlyReverbLengthEstimator(int max_blocks);
>+    ~EarlyReverbLengthEstimator();
>+
>+    // Resets the estimator.
>+    void Reset();
>+    // Accumulates estimation data.
>+    void Accumulate(float value, float smoothing);
>+    // Estimates the size in blocks of the early reverb.
>+    int Estimate();
>+    // Dumps debug data.
>+    void Dump(ApmDataDumper* data_dumper) const;
>+
>+   private:
>+    std::vector<float> numerators_;
>+    std::vector<float> nz_;
>+    std::vector<float> count_;
>+    int section_ = 0;
>+    int section_update_counter_ = 0;
>+  };
>+
>+  const int filter_length_blocks_;
>+  const int filter_length_coefficients_;
>+  const bool use_adaptive_echo_decay_;
>+  LateReverbLinearRegressor late_reverb_decay_estimator_;
>+  EarlyReverbLengthEstimator early_reverb_estimator_;
>+  int late_reverb_start_;
>+  int late_reverb_end_;
>+  int block_to_analyze_ = 0;
>+  int estimation_region_candidate_size_ = 0;
>+  bool estimation_region_identified_ = false;
>+  std::array<float, kMaxAdaptiveFilterLength> previous_gains_;
>+  float decay_;
>+  float tail_gain_ = 0.f;
>+  float smoothing_constant_ = 0.f;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_AEC3_REVERB_DECAY_ESTIMATOR_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_frequency_response.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_frequency_response.cc
>new file mode 100644
>index 00000000000..a20e776ad9c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_frequency_response.cc
>@@ -0,0 +1,113 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/aec3/reverb_frequency_response.h"
>+
>+#include <algorithm>
>+#include <array>
>+#include <cmath>
>+#include <memory>
>+#include <numeric>
>+
>+#include "api/array_view.h"
>+#include "api/audio/echo_canceller3_config.h"
>+#include "modules/audio_processing/aec3/aec3_common.h"
>+#include "rtc_base/checks.h"
>+#include "system_wrappers/include/field_trial.h"
>+
>+namespace webrtc {
>+
>+namespace {
>+
>+bool EnableSmoothUpdatesTailFreqResp() {
>+  return !field_trial::IsEnabled(
>+      "WebRTC-Aec3SmoothUpdatesTailFreqRespKillSwitch");
>+}
>+
>+// Computes the ratio of the energies between the direct path and the tail. The
>+// energy is computed in the power spectrum domain discarding the DC
>+// contributions.
>+float AverageDecayWithinFilter(
>+    rtc::ArrayView<const float> freq_resp_direct_path,
>+    rtc::ArrayView<const float> freq_resp_tail) {
>+  // Skipping the DC for the ratio computation
>+  constexpr size_t kSkipBins = 1;
>+  RTC_CHECK_EQ(freq_resp_direct_path.size(), freq_resp_tail.size());
>+
>+  float direct_path_energy =
>+      std::accumulate(freq_resp_direct_path.begin() + kSkipBins,
>+                      freq_resp_direct_path.end(), 0.f);
>+
>+  if (direct_path_energy == 0.f) {
>+    return 0.f;
>+  }
>+
>+  float tail_energy = std::accumulate(freq_resp_tail.begin() + kSkipBins,
>+                                      freq_resp_tail.end(), 0.f);
>+  return tail_energy / direct_path_energy;
>+}
>+
>+}  // namespace
>+
>+ReverbFrequencyResponse::ReverbFrequencyResponse()
>+    : enable_smooth_tail_response_updates_(EnableSmoothUpdatesTailFreqResp()) {
>+  tail_response_.fill(0.f);
>+}
>+ReverbFrequencyResponse::~ReverbFrequencyResponse() = default;
>+
>+void ReverbFrequencyResponse::Update(
>+    const std::vector<std::array<float, kFftLengthBy2Plus1>>&
>+        frequency_response,
>+    int filter_delay_blocks,
>+    const absl::optional<float>& linear_filter_quality,
>+    bool stationary_block) {
>+  if (!enable_smooth_tail_response_updates_) {
>+    Update(frequency_response, filter_delay_blocks, 0.1f);
>+    return;
>+  }
>+
>+  if (stationary_block || !linear_filter_quality) {
>+    return;
>+  }
>+
>+  Update(frequency_response, filter_delay_blocks, *linear_filter_quality);
>+}
>+
>+void ReverbFrequencyResponse::Update(
>+    const std::vector<std::array<float, kFftLengthBy2Plus1>>&
>+        frequency_response,
>+    int filter_delay_blocks,
>+    float linear_filter_quality) {
>+  rtc::ArrayView<const float> freq_resp_tail(
>+      frequency_response[frequency_response.size() - 1]);
>+
>+  rtc::ArrayView<const float> freq_resp_direct_path(
>+      frequency_response[filter_delay_blocks]);
>+
>+  float average_decay =
>+      AverageDecayWithinFilter(freq_resp_direct_path, freq_resp_tail);
>+
>+  const float smoothing = 0.2f * linear_filter_quality;
>+  average_decay_ += smoothing * (average_decay - average_decay_);
>+
>+  for (size_t k = 0; k < kFftLengthBy2Plus1; ++k) {
>+    tail_response_[k] = freq_resp_direct_path[k] * average_decay_;
>+  }
>+
>+  // TODO(devicentepena): Check if this should be done using a max that weights
>+  // both the lower and upper bands equally.
>+  for (size_t k = 1; k < kFftLengthBy2; ++k) {
>+    const float avg_neighbour =
>+        0.5f * (tail_response_[k - 1] + tail_response_[k + 1]);
>+    tail_response_[k] = std::max(tail_response_[k], avg_neighbour);
>+  }
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_frequency_response.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_frequency_response.h
>new file mode 100644
>index 00000000000..23485e009fb
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_frequency_response.h
>@@ -0,0 +1,55 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_AEC3_REVERB_FREQUENCY_RESPONSE_H_
>+#define MODULES_AUDIO_PROCESSING_AEC3_REVERB_FREQUENCY_RESPONSE_H_
>+
>+#include <memory>
>+#include <vector>
>+
>+#include "absl/types/optional.h"
>+#include "api/array_view.h"
>+#include "modules/audio_processing/aec3/aec3_common.h"
>+#include "modules/audio_processing/logging/apm_data_dumper.h"
>+
>+namespace webrtc {
>+
>+// Class for updating the frequency response for the reverb.
>+class ReverbFrequencyResponse {
>+ public:
>+  ReverbFrequencyResponse();
>+  ~ReverbFrequencyResponse();
>+
>+  // Updates the frequency response estimate of the reverb.
>+  void Update(const std::vector<std::array<float, kFftLengthBy2Plus1>>&
>+                  frequency_response,
>+              int filter_delay_blocks,
>+              const absl::optional<float>& linear_filter_quality,
>+              bool stationary_block);
>+
>+  // Returns the estimated frequency response for the reverb.
>+  rtc::ArrayView<const float> FrequencyResponse() const {
>+    return tail_response_;
>+  }
>+
>+ private:
>+  void Update(const std::vector<std::array<float, kFftLengthBy2Plus1>>&
>+                  frequency_response,
>+              int filter_delay_blocks,
>+              float linear_filter_quality);
>+
>+  const bool enable_smooth_tail_response_updates_;
>+  float average_decay_ = 0.f;
>+  std::array<float, kFftLengthBy2Plus1> tail_response_;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_AEC3_REVERB_FREQUENCY_RESPONSE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_model.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_model.cc
>new file mode 100644
>index 00000000000..0ca248fc754
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_model.cc
>@@ -0,0 +1,87 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/aec3/reverb_model.h"
>+
>+#include <math.h>
>+
>+#include <algorithm>
>+#include <functional>
>+
>+#include "api/array_view.h"
>+#include "modules/audio_processing/aec3/aec3_common.h"
>+
>+namespace webrtc {
>+
>+ReverbModel::ReverbModel() {
>+  Reset();
>+}
>+
>+ReverbModel::~ReverbModel() = default;
>+
>+void ReverbModel::Reset() {
>+  reverb_.fill(0.);
>+}
>+
>+void ReverbModel::AddReverbNoFreqShaping(
>+    rtc::ArrayView<const float> power_spectrum,
>+    float power_spectrum_scaling,
>+    float reverb_decay,
>+    rtc::ArrayView<float> reverb_power_spectrum) {
>+  UpdateReverbContributionsNoFreqShaping(power_spectrum, power_spectrum_scaling,
>+                                         reverb_decay);
>+
>+  // Add the power of the echo reverb to the residual echo power.
>+  std::transform(reverb_power_spectrum.begin(), reverb_power_spectrum.end(),
>+                 reverb_.begin(), reverb_power_spectrum.begin(),
>+                 std::plus<float>());
>+}
>+
>+void ReverbModel::AddReverb(rtc::ArrayView<const float> power_spectrum,
>+                            rtc::ArrayView<const float> power_spectrum_scaling,
>+                            float reverb_decay,
>+                            rtc::ArrayView<float> reverb_power_spectrum) {
>+  UpdateReverbContributions(power_spectrum, power_spectrum_scaling,
>+                            reverb_decay);
>+
>+  // Add the power of the echo reverb to the residual echo power.
>+  std::transform(reverb_power_spectrum.begin(), reverb_power_spectrum.end(),
>+                 reverb_.begin(), reverb_power_spectrum.begin(),
>+                 std::plus<float>());
>+}
>+
>+void ReverbModel::UpdateReverbContributionsNoFreqShaping(
>+    rtc::ArrayView<const float> power_spectrum,
>+    float power_spectrum_scaling,
>+    float reverb_decay) {
>+  if (reverb_decay > 0) {
>+    // Update the estimate of the reverberant power.
>+    for (size_t k = 0; k < power_spectrum.size(); ++k) {
>+      reverb_[k] = (reverb_[k] + power_spectrum[k] * power_spectrum_scaling) *
>+                   reverb_decay;
>+    }
>+  }
>+}
>+
>+void ReverbModel::UpdateReverbContributions(
>+    rtc::ArrayView<const float>& power_spectrum,
>+    rtc::ArrayView<const float>& power_spectrum_scaling,
>+    float reverb_decay) {
>+  if (reverb_decay > 0) {
>+    // Update the estimate of the reverberant power.
>+    for (size_t k = 0; k < power_spectrum.size(); ++k) {
>+      reverb_[k] =
>+          (reverb_[k] + power_spectrum[k] * power_spectrum_scaling[k]) *
>+          reverb_decay;
>+    }
>+  }
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_model.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_model.h
>new file mode 100644
>index 00000000000..d3087a72d91
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_model.h
>@@ -0,0 +1,68 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_AEC3_REVERB_MODEL_H_
>+#define MODULES_AUDIO_PROCESSING_AEC3_REVERB_MODEL_H_
>+
>+#include "api/array_view.h"
>+#include "modules/audio_processing/aec3/aec3_common.h"
>+
>+namespace webrtc {
>+
>+// The ReverbModel class describes an exponential reverberant model
>+// that can be applied over power spectrums.
>+class ReverbModel {
>+ public:
>+  ReverbModel();
>+  ~ReverbModel();
>+
>+  // Resets the state.
>+  void Reset();
>+
>+  // The methods AddReverbNoFreqShaping and AddReverb add the reverberation
>+  // contribution to an input/output power spectrum
>+  // Before applying the exponential reverberant model, the input power spectrum
>+  // is pre-scaled. Use the method AddReverb when a different scaling should be
>+  // applied per frequency and AddReverb_no_freq_shape if the same scaling
>+  // should be used for all the frequencies.
>+  void AddReverbNoFreqShaping(rtc::ArrayView<const float> power_spectrum,
>+                              float power_spectrum_scaling,
>+                              float reverb_decay,
>+                              rtc::ArrayView<float> reverb_power_spectrum);
>+
>+  void AddReverb(rtc::ArrayView<const float> power_spectrum,
>+                 rtc::ArrayView<const float> freq_response_tail,
>+                 float reverb_decay,
>+                 rtc::ArrayView<float> reverb_power_spectrum);
>+
>+  // Updates the reverberation contributions without applying any shaping of the
>+  // spectrum.
>+  void UpdateReverbContributionsNoFreqShaping(
>+      rtc::ArrayView<const float> power_spectrum,
>+      float power_spectrum_scaling,
>+      float reverb_decay);
>+
>+  // Returns the current power spectrum reverberation contributions.
>+  const std::array<float, kFftLengthBy2Plus1>& GetPowerSpectrum() const {
>+    return reverb_;
>+  }
>+
>+ private:
>+  // Updates the reverberation contributions.
>+  void UpdateReverbContributions(rtc::ArrayView<const float>& power_spectrum,
>+                                 rtc::ArrayView<const float>& freq_resp_tail,
>+                                 float reverb_decay);
>+
>+  std::array<float, kFftLengthBy2Plus1> reverb_;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_AEC3_REVERB_MODEL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_model_estimator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_model_estimator.cc
>new file mode 100644
>index 00000000000..ce3e2be3351
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_model_estimator.cc
>@@ -0,0 +1,37 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/aec3/reverb_model_estimator.h"
>+
>+namespace webrtc {
>+
>+ReverbModelEstimator::ReverbModelEstimator(const EchoCanceller3Config& config)
>+    : reverb_decay_estimator_(config) {}
>+
>+ReverbModelEstimator::~ReverbModelEstimator() = default;
>+
>+void ReverbModelEstimator::Update(
>+    rtc::ArrayView<const float> impulse_response,
>+    const std::vector<std::array<float, kFftLengthBy2Plus1>>&
>+        frequency_response,
>+    const absl::optional<float>& linear_filter_quality,
>+    int filter_delay_blocks,
>+    bool usable_linear_estimate,
>+    bool stationary_block) {
>+  // Estimate the frequency response for the reverb.
>+  reverb_frequency_response_.Update(frequency_response, filter_delay_blocks,
>+                                    linear_filter_quality, stationary_block);
>+
>+  // Estimate the reverb decay,
>+  reverb_decay_estimator_.Update(impulse_response, linear_filter_quality,
>+                                 filter_delay_blocks, usable_linear_estimate,
>+                                 stationary_block);
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_model_estimator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_model_estimator.h
>new file mode 100644
>index 00000000000..b6a3591e091
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_model_estimator.h
>@@ -0,0 +1,61 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_AEC3_REVERB_MODEL_ESTIMATOR_H_
>+#define MODULES_AUDIO_PROCESSING_AEC3_REVERB_MODEL_ESTIMATOR_H_
>+
>+#include <vector>
>+
>+#include "absl/types/optional.h"
>+#include "api/array_view.h"
>+#include "api/audio/echo_canceller3_config.h"
>+#include "modules/audio_processing/aec3/reverb_decay_estimator.h"
>+#include "modules/audio_processing/aec3/reverb_frequency_response.h"
>+
>+namespace webrtc {
>+
>+class ApmDataDumper;
>+
>+// Class for estimating the model parameters for the reverberant echo.
>+class ReverbModelEstimator {
>+ public:
>+  explicit ReverbModelEstimator(const EchoCanceller3Config& config);
>+  ~ReverbModelEstimator();
>+
>+  // Updates the estimates based on new data.
>+  void Update(rtc::ArrayView<const float> impulse_response,
>+              const std::vector<std::array<float, kFftLengthBy2Plus1>>&
>+                  frequency_response,
>+              const absl::optional<float>& linear_filter_quality,
>+              int filter_delay_blocks,
>+              bool usable_linear_estimate,
>+              bool stationary_block);
>+
>+  // Returns the exponential decay of the reverberant echo.
>+  float ReverbDecay() const { return reverb_decay_estimator_.Decay(); }
>+
>+  // Return the frequency response of the reverberant echo.
>+  rtc::ArrayView<const float> GetReverbFrequencyResponse() const {
>+    return reverb_frequency_response_.FrequencyResponse();
>+  }
>+
>+  // Dumps debug data.
>+  void Dump(ApmDataDumper* data_dumper) const {
>+    reverb_decay_estimator_.Dump(data_dumper);
>+  }
>+
>+ private:
>+  ReverbDecayEstimator reverb_decay_estimator_;
>+  ReverbFrequencyResponse reverb_frequency_response_;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_AEC3_REVERB_MODEL_ESTIMATOR_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_model_estimator_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_model_estimator_unittest.cc
>new file mode 100644
>index 00000000000..aa5e96f1041
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_model_estimator_unittest.cc
>@@ -0,0 +1,115 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/aec3/reverb_model_estimator.h"
>+
>+#include <algorithm>
>+#include <cmath>
>+#include <numeric>
>+
>+#include "absl/types/optional.h"
>+#include "api/array_view.h"
>+#include "api/audio/echo_canceller3_config.h"
>+#include "modules/audio_processing/aec3/aec3_common.h"
>+#include "modules/audio_processing/aec3/aec3_fft.h"
>+#include "modules/audio_processing/aec3/fft_data.h"
>+#include "rtc_base/checks.h"
>+
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+
>+class ReverbModelEstimatorTest {
>+ public:
>+  explicit ReverbModelEstimatorTest(float default_decay)
>+      : default_decay_(default_decay), estimated_decay_(default_decay) {
>+    aec3_config_.ep_strength.default_len = default_decay_;
>+    aec3_config_.filter.main.length_blocks = 40;
>+    h_.resize(aec3_config_.filter.main.length_blocks * kBlockSize);
>+    H2_.resize(aec3_config_.filter.main.length_blocks);
>+    CreateImpulseResponseWithDecay();
>+  }
>+  void RunEstimator();
>+  float GetDecay() { return estimated_decay_; }
>+  float GetTrueDecay() { return kTruePowerDecay; }
>+  float GetPowerTailDb() { return 10.f * log10(estimated_power_tail_); }
>+  float GetTruePowerTailDb() { return 10.f * log10(true_power_tail_); }
>+
>+ private:
>+  void CreateImpulseResponseWithDecay();
>+
>+  absl::optional<float> quality_linear_ = 1.0f;
>+  static constexpr int kFilterDelayBlocks = 2;
>+  static constexpr bool kUsableLinearEstimate = true;
>+  static constexpr bool kStationaryBlock = false;
>+  static constexpr float kTruePowerDecay = 0.5f;
>+  EchoCanceller3Config aec3_config_;
>+  float default_decay_;
>+  float estimated_decay_;
>+  float estimated_power_tail_ = 0.f;
>+  float true_power_tail_ = 0.f;
>+  std::vector<float> h_;
>+  std::vector<std::array<float, kFftLengthBy2Plus1>> H2_;
>+};
>+
>+void ReverbModelEstimatorTest::CreateImpulseResponseWithDecay() {
>+  const Aec3Fft fft;
>+  RTC_DCHECK_EQ(h_.size(), aec3_config_.filter.main.length_blocks * kBlockSize);
>+  RTC_DCHECK_EQ(H2_.size(), aec3_config_.filter.main.length_blocks);
>+  RTC_DCHECK_EQ(kFilterDelayBlocks, 2);
>+
>+  float decay_sample = std::sqrt(powf(kTruePowerDecay, 1.f / kBlockSize));
>+  const size_t filter_delay_coefficients = kFilterDelayBlocks * kBlockSize;
>+  std::fill(h_.begin(), h_.end(), 0.f);
>+  h_[filter_delay_coefficients] = 1.f;
>+  for (size_t k = filter_delay_coefficients + 1; k < h_.size(); ++k) {
>+    h_[k] = h_[k - 1] * decay_sample;
>+  }
>+
>+  std::array<float, kFftLength> fft_data;
>+  FftData H_j;
>+  for (size_t j = 0, k = 0; j < H2_.size(); ++j, k += kBlockSize) {
>+    fft_data.fill(0.f);
>+    std::copy(h_.begin() + k, h_.begin() + k + kBlockSize, fft_data.begin());
>+    fft.Fft(&fft_data, &H_j);
>+    H_j.Spectrum(Aec3Optimization::kNone, H2_[j]);
>+  }
>+  rtc::ArrayView<float> H2_tail(H2_[H2_.size() - 1]);
>+  true_power_tail_ = std::accumulate(H2_tail.begin(), H2_tail.end(), 0.f);
>+}
>+void ReverbModelEstimatorTest::RunEstimator() {
>+  ReverbModelEstimator estimator(aec3_config_);
>+  for (size_t k = 0; k < 3000; ++k) {
>+    estimator.Update(h_, H2_, quality_linear_, kFilterDelayBlocks,
>+                     kUsableLinearEstimate, kStationaryBlock);
>+  }
>+  estimated_decay_ = estimator.ReverbDecay();
>+  auto freq_resp_tail = estimator.GetReverbFrequencyResponse();
>+  estimated_power_tail_ =
>+      std::accumulate(freq_resp_tail.begin(), freq_resp_tail.end(), 0.f);
>+}
>+
>+TEST(ReverbModelEstimatorTests, NotChangingDecay) {
>+  constexpr float default_decay = 0.9f;
>+  ReverbModelEstimatorTest test(default_decay);
>+  test.RunEstimator();
>+  EXPECT_EQ(test.GetDecay(), default_decay);
>+  EXPECT_NEAR(test.GetPowerTailDb(), test.GetTruePowerTailDb(), 5.f);
>+}
>+
>+TEST(ReverbModelEstimatorTests, ChangingDecay) {
>+  constexpr float default_decay = -0.9f;
>+  ReverbModelEstimatorTest test(default_decay);
>+  test.RunEstimator();
>+  EXPECT_NEAR(test.GetDecay(), test.GetTrueDecay(), 0.1);
>+  EXPECT_NEAR(test.GetPowerTailDb(), test.GetTruePowerTailDb(), 5.f);
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_model_fallback.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_model_fallback.cc
>new file mode 100644
>index 00000000000..67df37b3cf8
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_model_fallback.cc
>@@ -0,0 +1,70 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/aec3/reverb_model_fallback.h"
>+
>+#include <algorithm>
>+#include <functional>
>+
>+#include "modules/audio_processing/aec3/aec3_common.h"
>+#include "rtc_base/checks.h"
>+
>+namespace webrtc {
>+
>+ReverbModelFallback::ReverbModelFallback(size_t length_blocks)
>+    : S2_old_(length_blocks) {
>+  Reset();
>+}
>+
>+ReverbModelFallback::~ReverbModelFallback() = default;
>+
>+void ReverbModelFallback::Reset() {
>+  R2_reverb_.fill(0.f);
>+  for (auto& S2_k : S2_old_) {
>+    S2_k.fill(0.f);
>+  }
>+}
>+
>+void ReverbModelFallback::AddEchoReverb(
>+    const std::array<float, kFftLengthBy2Plus1>& S2,
>+    size_t delay,
>+    float reverb_decay_factor,
>+    std::array<float, kFftLengthBy2Plus1>* R2) {
>+  // Compute the decay factor for how much the echo has decayed before leaving
>+  // the region covered by the linear model.
>+  auto integer_power = [](float base, int exp) {
>+    float result = 1.f;
>+    for (int k = 0; k < exp; ++k) {
>+      result *= base;
>+    }
>+    return result;
>+  };
>+  RTC_DCHECK_LE(delay, S2_old_.size());
>+  const float reverb_decay_for_delay =
>+      integer_power(reverb_decay_factor, S2_old_.size() - delay);
>+
>+  // Update the estimate of the reverberant residual echo power.
>+  S2_old_index_ = S2_old_index_ > 0 ? S2_old_index_ - 1 : S2_old_.size() - 1;
>+  const auto& S2_end = S2_old_[S2_old_index_];
>+  std::transform(
>+      S2_end.begin(), S2_end.end(), R2_reverb_.begin(), R2_reverb_.begin(),
>+      [reverb_decay_for_delay, reverb_decay_factor](float a, float b) {
>+        return (b + a * reverb_decay_for_delay) * reverb_decay_factor;
>+      });
>+
>+  // Update the buffer of old echo powers.
>+  std::copy(S2.begin(), S2.end(), S2_old_[S2_old_index_].begin());
>+
>+  // Add the power of the echo reverb to the residual echo power.
>+  std::transform(R2->begin(), R2->end(), R2_reverb_.begin(), R2->begin(),
>+                 std::plus<float>());
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_model_fallback.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_model_fallback.h
>new file mode 100644
>index 00000000000..1b2a953ea64
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/reverb_model_fallback.h
>@@ -0,0 +1,53 @@
>+/*
>+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_AEC3_REVERB_MODEL_FALLBACK_H_
>+#define MODULES_AUDIO_PROCESSING_AEC3_REVERB_MODEL_FALLBACK_H_
>+
>+#include <array>
>+#include <vector>
>+
>+#include "modules/audio_processing/aec3/aec3_common.h"
>+
>+namespace webrtc {
>+
>+// The ReverbModelFallback class describes an exponential reverberant model.
>+// This model is expected to be applied over the echo power spectrum that
>+// is estimated by the linear filter.
>+
>+class ReverbModelFallback {
>+ public:
>+  explicit ReverbModelFallback(size_t length_blocks);
>+  ~ReverbModelFallback();
>+
>+  // Resets the state
>+  void Reset();
>+
>+  // Adds the estimated unmodelled echo power to the residual echo power
>+  // estimate.
>+  void AddEchoReverb(const std::array<float, kFftLengthBy2Plus1>& S2,
>+                     size_t delay,
>+                     float reverb_decay_factor,
>+                     std::array<float, kFftLengthBy2Plus1>* R2);
>+
>+  // Returns the current power spectrum reverberation contributions.
>+  const std::array<float, kFftLengthBy2Plus1>& GetPowerSpectrum() const {
>+    return R2_reverb_;
>+  }
>+
>+ private:
>+  std::array<float, kFftLengthBy2Plus1> R2_reverb_;
>+  int S2_old_index_ = 0;
>+  std::vector<std::array<float, kFftLengthBy2Plus1>> S2_old_;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_AEC3_REVERB_MODEL_FALLBACK_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/shadow_filter_update_gain.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/shadow_filter_update_gain.cc
>index 0fc940a555b..e27437aff28 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/shadow_filter_update_gain.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/shadow_filter_update_gain.cc
>@@ -18,8 +18,14 @@
> namespace webrtc {
> 
> ShadowFilterUpdateGain::ShadowFilterUpdateGain(
>-    const EchoCanceller3Config::Filter::ShadowConfiguration& config)
>-    : config_(config) {}
>+    const EchoCanceller3Config::Filter::ShadowConfiguration& config,
>+    size_t config_change_duration_blocks)
>+    : config_change_duration_blocks_(
>+          static_cast<int>(config_change_duration_blocks)) {
>+  SetConfig(config, true);
>+  RTC_DCHECK_LT(0, config_change_duration_blocks_);
>+  one_by_config_change_duration_blocks_ = 1.f / config_change_duration_blocks_;
>+}
> 
> void ShadowFilterUpdateGain::HandleEchoPathChange() {
>   // TODO(peah): Check whether this counter should instead be initialized to a
>@@ -38,6 +44,8 @@ void ShadowFilterUpdateGain::Compute(
>   RTC_DCHECK(G);
>   ++call_counter_;
> 
>+  UpdateCurrentConfig();
>+
>   if (render_signal_analyzer.PoorSignalExcitation()) {
>     poor_signal_excitation_counter_ = 0;
>   }
>@@ -54,7 +62,7 @@ void ShadowFilterUpdateGain::Compute(
>   std::array<float, kFftLengthBy2Plus1> mu;
>   auto X2 = render_power;
>   std::transform(X2.begin(), X2.end(), mu.begin(), [&](float a) {
>-    return a > config_.noise_gate ? config_.rate / a : 0.f;
>+    return a > current_config_.noise_gate ? current_config_.rate / a : 0.f;
>   });
> 
>   // Avoid updating the filter close to narrow bands in the render signals.
>@@ -67,4 +75,27 @@ void ShadowFilterUpdateGain::Compute(
>                  std::multiplies<float>());
> }
> 
>+void ShadowFilterUpdateGain::UpdateCurrentConfig() {
>+  RTC_DCHECK_GE(config_change_duration_blocks_, config_change_counter_);
>+  if (config_change_counter_ > 0) {
>+    if (--config_change_counter_ > 0) {
>+      auto average = [](float from, float to, float from_weight) {
>+        return from * from_weight + to * (1.f - from_weight);
>+      };
>+
>+      float change_factor =
>+          config_change_counter_ * one_by_config_change_duration_blocks_;
>+
>+      current_config_.rate =
>+          average(old_target_config_.rate, target_config_.rate, change_factor);
>+      current_config_.noise_gate =
>+          average(old_target_config_.noise_gate, target_config_.noise_gate,
>+                  change_factor);
>+    } else {
>+      current_config_ = old_target_config_ = target_config_;
>+    }
>+  }
>+  RTC_DCHECK_LE(0, config_change_counter_);
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/shadow_filter_update_gain.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/shadow_filter_update_gain.h
>index 5f7f5a1f47b..a92bc3b8b7f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/shadow_filter_update_gain.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/shadow_filter_update_gain.h
>@@ -11,10 +11,10 @@
> #ifndef MODULES_AUDIO_PROCESSING_AEC3_SHADOW_FILTER_UPDATE_GAIN_H_
> #define MODULES_AUDIO_PROCESSING_AEC3_SHADOW_FILTER_UPDATE_GAIN_H_
> 
>+#include "api/audio/echo_canceller3_config.h"
> #include "modules/audio_processing/aec3/aec3_common.h"
> #include "modules/audio_processing/aec3/render_buffer.h"
> #include "modules/audio_processing/aec3/render_signal_analyzer.h"
>-#include "modules/audio_processing/include/audio_processing.h"
> #include "rtc_base/constructormagic.h"
> 
> namespace webrtc {
>@@ -23,7 +23,8 @@ namespace webrtc {
> class ShadowFilterUpdateGain {
>  public:
>   explicit ShadowFilterUpdateGain(
>-      const EchoCanceller3Config::Filter::ShadowConfiguration& config);
>+      const EchoCanceller3Config::Filter::ShadowConfiguration& config,
>+      size_t config_change_duration_blocks);
> 
>   // Takes action in the case of a known echo path change.
>   void HandleEchoPathChange();
>@@ -38,16 +39,31 @@ class ShadowFilterUpdateGain {
> 
>   // Sets a new config.
>   void SetConfig(
>-      const EchoCanceller3Config::Filter::ShadowConfiguration& config) {
>-    config_ = config;
>+      const EchoCanceller3Config::Filter::ShadowConfiguration& config,
>+      bool immediate_effect) {
>+    if (immediate_effect) {
>+      old_target_config_ = current_config_ = target_config_ = config;
>+      config_change_counter_ = 0;
>+    } else {
>+      old_target_config_ = current_config_;
>+      target_config_ = config;
>+      config_change_counter_ = config_change_duration_blocks_;
>+    }
>   }
> 
>  private:
>-  EchoCanceller3Config::Filter::ShadowConfiguration config_;
>+  EchoCanceller3Config::Filter::ShadowConfiguration current_config_;
>+  EchoCanceller3Config::Filter::ShadowConfiguration target_config_;
>+  EchoCanceller3Config::Filter::ShadowConfiguration old_target_config_;
>+  const int config_change_duration_blocks_;
>+  float one_by_config_change_duration_blocks_;
>   // TODO(peah): Check whether this counter should instead be initialized to a
>   // large value.
>   size_t poor_signal_excitation_counter_ = 0;
>   size_t call_counter_ = 0;
>+  int config_change_counter_ = 0;
>+
>+  void UpdateCurrentConfig();
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/shadow_filter_update_gain_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/shadow_filter_update_gain_unittest.cc
>index 77750fd7c8d..d77da33f102 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/shadow_filter_update_gain_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/shadow_filter_update_gain_unittest.cc
>@@ -40,8 +40,12 @@ void RunFilterUpdateTest(int num_blocks_to_process,
>   EchoCanceller3Config config;
>   config.filter.main.length_blocks = filter_length_blocks;
>   AdaptiveFirFilter main_filter(config.filter.main.length_blocks,
>+                                config.filter.main.length_blocks,
>+                                config.filter.config_change_duration_blocks,
>                                 DetectOptimization(), &data_dumper);
>   AdaptiveFirFilter shadow_filter(config.filter.shadow.length_blocks,
>+                                  config.filter.shadow.length_blocks,
>+                                  config.filter.config_change_duration_blocks,
>                                   DetectOptimization(), &data_dumper);
>   Aec3Fft fft;
> 
>@@ -52,12 +56,13 @@ void RunFilterUpdateTest(int num_blocks_to_process,
> 
>   std::array<float, kBlockSize> x_old;
>   x_old.fill(0.f);
>-  ShadowFilterUpdateGain shadow_gain(config.filter.shadow);
>+  ShadowFilterUpdateGain shadow_gain(
>+      config.filter.shadow, config.filter.config_change_duration_blocks);
>   Random random_generator(42U);
>   std::vector<std::vector<float>> x(3, std::vector<float>(kBlockSize, 0.f));
>   std::vector<float> y(kBlockSize, 0.f);
>   AecState aec_state(config);
>-  RenderSignalAnalyzer render_signal_analyzer;
>+  RenderSignalAnalyzer render_signal_analyzer(config);
>   std::array<float, kFftLength> s;
>   FftData S;
>   FftData G;
>@@ -130,11 +135,11 @@ std::string ProduceDebugText(size_t delay, int filter_length_blocks) {
> TEST(ShadowFilterUpdateGain, NullDataOutputGain) {
>   ApmDataDumper data_dumper(42);
>   FftBuffer fft_buffer(1);
>-  RenderSignalAnalyzer analyzer;
>+  RenderSignalAnalyzer analyzer(EchoCanceller3Config{});
>   FftData E;
>   const EchoCanceller3Config::Filter::ShadowConfiguration& config = {
>       12, 0.5f, 220075344.f};
>-  ShadowFilterUpdateGain gain(config);
>+  ShadowFilterUpdateGain gain(config, 250);
>   std::array<float, kFftLengthBy2Plus1> render_power;
>   render_power.fill(0.f);
>   EXPECT_DEATH(gain.Compute(render_power, analyzer, E, 1, false, nullptr), "");
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/skew_estimator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/skew_estimator.cc
>new file mode 100644
>index 00000000000..310e4e9dd1d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/skew_estimator.cc
>@@ -0,0 +1,46 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#include "modules/audio_processing/aec3/skew_estimator.h"
>+
>+#include <algorithm>
>+#include <numeric>
>+
>+namespace webrtc {
>+
>+SkewEstimator::SkewEstimator(size_t skew_history_size_log2)
>+    : skew_history_size_log2_(static_cast<int>(skew_history_size_log2)),
>+      skew_history_(1ULL << skew_history_size_log2_, 0) {}
>+
>+SkewEstimator::~SkewEstimator() = default;
>+
>+void SkewEstimator::Reset() {
>+  skew_ = 0;
>+  skew_sum_ = 0;
>+  next_index_ = 0;
>+  sufficient_skew_stored_ = false;
>+  std::fill(skew_history_.begin(), skew_history_.end(), 0);
>+}
>+
>+absl::optional<int> SkewEstimator::GetSkewFromCapture() {
>+  --skew_;
>+
>+  skew_sum_ += skew_ - skew_history_[next_index_];
>+  skew_history_[next_index_] = skew_;
>+  if (++next_index_ == skew_history_.size()) {
>+    next_index_ = 0;
>+    sufficient_skew_stored_ = true;
>+  }
>+
>+  const int bias = static_cast<int>(skew_history_.size()) >> 1;
>+  const int average = (skew_sum_ + bias) >> skew_history_size_log2_;
>+  return sufficient_skew_stored_ ? absl::optional<int>(average) : absl::nullopt;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/skew_estimator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/skew_estimator.h
>new file mode 100644
>index 00000000000..ff2526025a8
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/skew_estimator.h
>@@ -0,0 +1,50 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_AEC3_SKEW_ESTIMATOR_H_
>+#define MODULES_AUDIO_PROCESSING_AEC3_SKEW_ESTIMATOR_H_
>+
>+#include <vector>
>+
>+#include "absl/types/optional.h"
>+#include "rtc_base/constructormagic.h"
>+
>+namespace webrtc {
>+
>+// Estimator of API call skew between render and capture.
>+class SkewEstimator {
>+ public:
>+  explicit SkewEstimator(size_t skew_history_size_log2);
>+  ~SkewEstimator();
>+
>+  // Resets the estimation.
>+  void Reset();
>+
>+  // Updates the skew data for a render call.
>+  void LogRenderCall() { ++skew_; }
>+
>+  // Updates and computes the skew at a capture call. Returns an optional which
>+  // is non-null if a reliable skew has been found.
>+  absl::optional<int> GetSkewFromCapture();
>+
>+ private:
>+  const int skew_history_size_log2_;
>+  std::vector<float> skew_history_;
>+  int skew_ = 0;
>+  int skew_sum_ = 0;
>+  size_t next_index_ = 0;
>+  bool sufficient_skew_stored_ = false;
>+
>+  RTC_DISALLOW_COPY_AND_ASSIGN(SkewEstimator);
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_AEC3_SKEW_ESTIMATOR_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/skew_estimator_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/skew_estimator_unittest.cc
>new file mode 100644
>index 00000000000..3e8c13e23c1
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/skew_estimator_unittest.cc
>@@ -0,0 +1,157 @@
>+/*
>+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/aec3/skew_estimator.h"
>+
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+namespace aec3 {
>+
>+// Tests that the skew ends up as it should after a skew change.
>+TEST(SkewEstimator, SkewChangeAdaptation) {
>+  constexpr int kNumSkewsLog2 = 7;
>+  constexpr int kNumSkews = 1 << kNumSkewsLog2;
>+
>+  SkewEstimator estimator(kNumSkewsLog2);
>+
>+  for (int k = 0; k < kNumSkews - 1; ++k) {
>+    estimator.LogRenderCall();
>+    auto skew = estimator.GetSkewFromCapture();
>+    EXPECT_FALSE(skew);
>+  }
>+
>+  estimator.LogRenderCall();
>+
>+  absl::optional<int> skew;
>+  for (int k = 0; k < kNumSkews; ++k) {
>+    estimator.LogRenderCall();
>+    skew = estimator.GetSkewFromCapture();
>+    EXPECT_TRUE(skew);
>+  }
>+  EXPECT_EQ(1, *skew);
>+
>+  estimator.LogRenderCall();
>+
>+  for (int k = 0; k < kNumSkews; ++k) {
>+    estimator.LogRenderCall();
>+    skew = estimator.GetSkewFromCapture();
>+    EXPECT_TRUE(skew);
>+  }
>+  EXPECT_EQ(2, *skew);
>+}
>+
>+// Tests that the skew ends up as it should for a surplus of render calls.
>+TEST(SkewEstimator, SkewForSurplusRender) {
>+  constexpr int kNumSkewsLog2 = 7;
>+  constexpr int kNumSkews = 1 << kNumSkewsLog2;
>+
>+  SkewEstimator estimator(kNumSkewsLog2);
>+
>+  for (int k = 0; k < kNumSkews - 1; ++k) {
>+    estimator.LogRenderCall();
>+    auto skew = estimator.GetSkewFromCapture();
>+    EXPECT_FALSE(skew);
>+  }
>+
>+  estimator.LogRenderCall();
>+
>+  absl::optional<int> skew;
>+  for (int k = 0; k < kNumSkews; ++k) {
>+    estimator.LogRenderCall();
>+    skew = estimator.GetSkewFromCapture();
>+    EXPECT_TRUE(skew);
>+  }
>+  EXPECT_EQ(1, *skew);
>+}
>+
>+// Tests that the skew ends up as it should for a surplus of capture calls.
>+TEST(SkewEstimator, SkewForSurplusCapture) {
>+  constexpr int kNumSkewsLog2 = 7;
>+  constexpr int kNumSkews = 1 << kNumSkewsLog2;
>+
>+  SkewEstimator estimator(kNumSkewsLog2);
>+
>+  for (int k = 0; k < kNumSkews - 1; ++k) {
>+    estimator.LogRenderCall();
>+    auto skew = estimator.GetSkewFromCapture();
>+    EXPECT_FALSE(skew);
>+  }
>+
>+  absl::optional<int> skew;
>+  skew = estimator.GetSkewFromCapture();
>+
>+  for (int k = 0; k < kNumSkews; ++k) {
>+    estimator.LogRenderCall();
>+    skew = estimator.GetSkewFromCapture();
>+    EXPECT_TRUE(skew);
>+  }
>+  EXPECT_EQ(-1, *skew);
>+}
>+
>+// Tests that the skew estimator returns a null optional when it should.
>+TEST(SkewEstimator, NullEstimate) {
>+  constexpr int kNumSkewsLog2 = 4;
>+  constexpr int kNumSkews = 1 << kNumSkewsLog2;
>+
>+  SkewEstimator estimator(kNumSkewsLog2);
>+
>+  for (int k = 0; k < kNumSkews - 1; ++k) {
>+    estimator.LogRenderCall();
>+    auto skew = estimator.GetSkewFromCapture();
>+    EXPECT_FALSE(skew);
>+  }
>+
>+  estimator.LogRenderCall();
>+  auto skew = estimator.GetSkewFromCapture();
>+  EXPECT_TRUE(skew);
>+
>+  estimator.Reset();
>+  for (int k = 0; k < kNumSkews - 1; ++k) {
>+    estimator.LogRenderCall();
>+    auto skew = estimator.GetSkewFromCapture();
>+    EXPECT_FALSE(skew);
>+  }
>+}
>+
>+// Tests that the skew estimator properly rounds the average skew.
>+TEST(SkewEstimator, SkewRounding) {
>+  constexpr int kNumSkewsLog2 = 4;
>+  constexpr int kNumSkews = 1 << kNumSkewsLog2;
>+
>+  SkewEstimator estimator(kNumSkewsLog2);
>+
>+  absl::optional<int> skew;
>+  for (int k = 0; k < kNumSkews; ++k) {
>+    if (k == kNumSkews - 1) {
>+      // Reverse call order once.
>+      skew = estimator.GetSkewFromCapture();
>+      estimator.LogRenderCall();
>+    } else {
>+      // Normal call order.
>+      estimator.LogRenderCall();
>+      skew = estimator.GetSkewFromCapture();
>+    }
>+  }
>+  EXPECT_EQ(*skew, 0);
>+
>+  estimator.Reset();
>+  for (int k = 0; k < kNumSkews; ++k) {
>+    estimator.LogRenderCall();
>+    estimator.LogRenderCall();
>+    estimator.LogRenderCall();
>+    estimator.GetSkewFromCapture();
>+    estimator.GetSkewFromCapture();
>+    skew = estimator.GetSkewFromCapture();
>+  }
>+  EXPECT_EQ(*skew, 1);
>+}
>+}  // namespace aec3
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/stationarity_estimator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/stationarity_estimator.cc
>new file mode 100644
>index 00000000000..efeabf1117f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/stationarity_estimator.cc
>@@ -0,0 +1,218 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/aec3/stationarity_estimator.h"
>+
>+#include <algorithm>
>+#include <array>
>+#include <vector>
>+
>+#include "modules/audio_processing/aec3/aec3_common.h"
>+#include "modules/audio_processing/aec3/vector_buffer.h"
>+#include "modules/audio_processing/logging/apm_data_dumper.h"
>+#include "rtc_base/atomicops.h"
>+
>+namespace webrtc {
>+
>+namespace {
>+constexpr float kMinNoisePower = 10.f;
>+constexpr int kHangoverBlocks = kNumBlocksPerSecond / 20;
>+constexpr int kNBlocksAverageInitPhase = 20;
>+constexpr int kNBlocksInitialPhase = kNumBlocksPerSecond * 2.;
>+}  // namespace
>+
>+StationarityEstimator::StationarityEstimator()
>+    : data_dumper_(
>+          new ApmDataDumper(rtc::AtomicOps::Increment(&instance_count_))) {
>+  Reset();
>+}
>+
>+StationarityEstimator::~StationarityEstimator() = default;
>+
>+void StationarityEstimator::Reset() {
>+  noise_.Reset();
>+  hangovers_.fill(0);
>+  stationarity_flags_.fill(false);
>+  render_reverb_.Reset();
>+}
>+
>+// Update just the noise estimator. Usefull until the delay is known
>+void StationarityEstimator::UpdateNoiseEstimator(
>+    rtc::ArrayView<const float> spectrum) {
>+  noise_.Update(spectrum);
>+  data_dumper_->DumpRaw("aec3_stationarity_noise_spectrum", noise_.Spectrum());
>+  data_dumper_->DumpRaw("aec3_stationarity_is_block_stationary",
>+                        IsBlockStationary());
>+}
>+
>+void StationarityEstimator::UpdateStationarityFlags(
>+    const VectorBuffer& spectrum_buffer,
>+    int idx_current,
>+    int num_lookahead,
>+    float reverb_decay) {
>+  std::array<int, kWindowLength> indexes;
>+  int num_lookahead_bounded = std::min(num_lookahead, kWindowLength - 1);
>+  int idx = idx_current;
>+
>+  if (num_lookahead_bounded < kWindowLength - 1) {
>+    int num_lookback = (kWindowLength - 1) - num_lookahead_bounded;
>+    idx = spectrum_buffer.OffsetIndex(idx_current, num_lookback);
>+  }
>+  // For estimating the stationarity properties of the current frame, the
>+  // power for each band is accumulated for several consecutive spectra in the
>+  // method EstimateBandStationarity.
>+  // In order to avoid getting the indexes of the spectra for every band with
>+  // its associated overhead, those indexes are stored in an array and then use
>+  // when the estimation is done.
>+  indexes[0] = idx;
>+  for (size_t k = 1; k < indexes.size(); ++k) {
>+    indexes[k] = spectrum_buffer.DecIndex(indexes[k - 1]);
>+  }
>+  RTC_DCHECK_EQ(
>+      spectrum_buffer.DecIndex(indexes[kWindowLength - 1]),
>+      spectrum_buffer.OffsetIndex(idx_current, -(num_lookahead_bounded + 1)));
>+
>+  int idx_past = spectrum_buffer.IncIndex(idx_current);
>+  render_reverb_.UpdateReverbContributionsNoFreqShaping(
>+      spectrum_buffer.buffer[idx_past], 1.0f, reverb_decay);
>+  for (size_t k = 0; k < stationarity_flags_.size(); ++k) {
>+    stationarity_flags_[k] = EstimateBandStationarity(
>+        spectrum_buffer, render_reverb_.GetPowerSpectrum(), indexes, k);
>+  }
>+  UpdateHangover();
>+  SmoothStationaryPerFreq();
>+}
>+
>+bool StationarityEstimator::IsBlockStationary() const {
>+  float acum_stationarity = 0.f;
>+  RTC_DCHECK_EQ(stationarity_flags_.size(), kFftLengthBy2Plus1);
>+  for (size_t band = 0; band < stationarity_flags_.size(); ++band) {
>+    bool st = IsBandStationary(band);
>+    acum_stationarity += static_cast<float>(st);
>+  }
>+  return ((acum_stationarity * (1.f / kFftLengthBy2Plus1)) > 0.75f);
>+}
>+
>+bool StationarityEstimator::EstimateBandStationarity(
>+    const VectorBuffer& spectrum_buffer,
>+    const std::array<float, kFftLengthBy2Plus1>& reverb,
>+    const std::array<int, kWindowLength>& indexes,
>+    size_t band) const {
>+  constexpr float kThrStationarity = 10.f;
>+  float acum_power = 0.f;
>+  for (auto idx : indexes) {
>+    acum_power += spectrum_buffer.buffer[idx][band];
>+  }
>+  acum_power += reverb[band];
>+  float noise = kWindowLength * GetStationarityPowerBand(band);
>+  RTC_CHECK_LT(0.f, noise);
>+  bool stationary = acum_power < kThrStationarity * noise;
>+  data_dumper_->DumpRaw("aec3_stationarity_long_ratio", acum_power / noise);
>+  return stationary;
>+}
>+
>+bool StationarityEstimator::AreAllBandsStationary() {
>+  for (auto b : stationarity_flags_) {
>+    if (!b)
>+      return false;
>+  }
>+  return true;
>+}
>+
>+void StationarityEstimator::UpdateHangover() {
>+  bool reduce_hangover = AreAllBandsStationary();
>+  for (size_t k = 0; k < stationarity_flags_.size(); ++k) {
>+    if (!stationarity_flags_[k]) {
>+      hangovers_[k] = kHangoverBlocks;
>+    } else if (reduce_hangover) {
>+      hangovers_[k] = std::max(hangovers_[k] - 1, 0);
>+    }
>+  }
>+}
>+
>+void StationarityEstimator::SmoothStationaryPerFreq() {
>+  std::array<bool, kFftLengthBy2Plus1> all_ahead_stationary_smooth;
>+  for (size_t k = 1; k < kFftLengthBy2Plus1 - 1; ++k) {
>+    all_ahead_stationary_smooth[k] = stationarity_flags_[k - 1] &&
>+                                     stationarity_flags_[k] &&
>+                                     stationarity_flags_[k + 1];
>+  }
>+
>+  all_ahead_stationary_smooth[0] = all_ahead_stationary_smooth[1];
>+  all_ahead_stationary_smooth[kFftLengthBy2Plus1 - 1] =
>+      all_ahead_stationary_smooth[kFftLengthBy2Plus1 - 2];
>+
>+  stationarity_flags_ = all_ahead_stationary_smooth;
>+}
>+
>+int StationarityEstimator::instance_count_ = 0;
>+
>+StationarityEstimator::NoiseSpectrum::NoiseSpectrum() {
>+  Reset();
>+}
>+
>+StationarityEstimator::NoiseSpectrum::~NoiseSpectrum() = default;
>+
>+void StationarityEstimator::NoiseSpectrum::Reset() {
>+  block_counter_ = 0;
>+  noise_spectrum_.fill(kMinNoisePower);
>+}
>+
>+void StationarityEstimator::NoiseSpectrum::Update(
>+    rtc::ArrayView<const float> spectrum) {
>+  RTC_DCHECK_EQ(kFftLengthBy2Plus1, spectrum.size());
>+  ++block_counter_;
>+  float alpha = GetAlpha();
>+  for (size_t k = 0; k < spectrum.size(); ++k) {
>+    if (block_counter_ <= kNBlocksAverageInitPhase) {
>+      noise_spectrum_[k] += (1.f / kNBlocksAverageInitPhase) * spectrum[k];
>+    } else {
>+      noise_spectrum_[k] =
>+          UpdateBandBySmoothing(spectrum[k], noise_spectrum_[k], alpha);
>+    }
>+  }
>+}
>+
>+float StationarityEstimator::NoiseSpectrum::GetAlpha() const {
>+  constexpr float kAlpha = 0.004f;
>+  constexpr float kAlphaInit = 0.04f;
>+  constexpr float kTiltAlpha = (kAlphaInit - kAlpha) / kNBlocksInitialPhase;
>+
>+  if (block_counter_ > (kNBlocksInitialPhase + kNBlocksAverageInitPhase)) {
>+    return kAlpha;
>+  } else {
>+    return kAlphaInit -
>+           kTiltAlpha * (block_counter_ - kNBlocksAverageInitPhase);
>+  }
>+}
>+
>+float StationarityEstimator::NoiseSpectrum::UpdateBandBySmoothing(
>+    float power_band,
>+    float power_band_noise,
>+    float alpha) const {
>+  float power_band_noise_updated = power_band_noise;
>+  if (power_band_noise < power_band) {
>+    RTC_DCHECK_GT(power_band, 0.f);
>+    float alpha_inc = alpha * (power_band_noise / power_band);
>+    if (block_counter_ > kNBlocksInitialPhase) {
>+      if (10.f * power_band_noise < power_band) {
>+        alpha_inc *= 0.1f;
>+      }
>+    }
>+    power_band_noise_updated += alpha_inc * (power_band - power_band_noise);
>+  } else {
>+    power_band_noise_updated += alpha * (power_band - power_band_noise);
>+    power_band_noise_updated =
>+        std::max(power_band_noise_updated, kMinNoisePower);
>+  }
>+  return power_band_noise_updated;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/stationarity_estimator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/stationarity_estimator.h
>new file mode 100644
>index 00000000000..e2c5a625347
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/stationarity_estimator.h
>@@ -0,0 +1,119 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_AEC3_STATIONARITY_ESTIMATOR_H_
>+#define MODULES_AUDIO_PROCESSING_AEC3_STATIONARITY_ESTIMATOR_H_
>+
>+#include <array>
>+#include <memory>
>+#include <vector>
>+
>+#include "api/array_view.h"
>+#include "modules/audio_processing/aec3/aec3_common.h"
>+#include "modules/audio_processing/aec3/reverb_model.h"
>+#include "modules/audio_processing/aec3/vector_buffer.h"
>+
>+namespace webrtc {
>+
>+class ApmDataDumper;
>+
>+class StationarityEstimator {
>+ public:
>+  StationarityEstimator();
>+  ~StationarityEstimator();
>+
>+  // Reset the stationarity estimator.
>+  void Reset();
>+
>+  // Update just the noise estimator. Usefull until the delay is known
>+  void UpdateNoiseEstimator(rtc::ArrayView<const float> spectrum);
>+
>+  // Update the flag indicating whether this current frame is stationary. For
>+  // getting a more robust estimation, it looks at future and/or past frames.
>+  void UpdateStationarityFlags(const VectorBuffer& spectrum_buffer,
>+                               int idx_current,
>+                               int num_lookahead,
>+                               float reverb_decay);
>+
>+  // Returns true if the current band is stationary.
>+  bool IsBandStationary(size_t band) const {
>+    return stationarity_flags_[band] && (hangovers_[band] == 0);
>+  }
>+
>+  // Returns true if the current block is estimated as stationary.
>+  bool IsBlockStationary() const;
>+
>+ private:
>+  static constexpr int kWindowLength = 13;
>+  // Returns the power of the stationary noise spectrum at a band.
>+  float GetStationarityPowerBand(size_t k) const { return noise_.Power(k); }
>+
>+  // Get an estimation of the stationarity for the current band by looking
>+  // at the past/present/future available data.
>+  bool EstimateBandStationarity(
>+      const VectorBuffer& spectrum_buffer,
>+      const std::array<float, kFftLengthBy2Plus1>& reverb,
>+      const std::array<int, kWindowLength>& indexes,
>+      size_t band) const;
>+
>+  // True if all bands at the current point are stationary.
>+  bool AreAllBandsStationary();
>+
>+  // Update the hangover depending on the stationary status of the current
>+  // frame.
>+  void UpdateHangover();
>+
>+  // Smooth the stationarity detection by looking at neighbouring frequency
>+  // bands.
>+  void SmoothStationaryPerFreq();
>+
>+  class NoiseSpectrum {
>+   public:
>+    NoiseSpectrum();
>+    ~NoiseSpectrum();
>+
>+    // Reset the noise power spectrum estimate state.
>+    void Reset();
>+
>+    // Update the noise power spectrum with a new frame.
>+    void Update(rtc::ArrayView<const float> spectrum);
>+
>+    // Get the noise estimation power spectrum.
>+    rtc::ArrayView<const float> Spectrum() const { return noise_spectrum_; }
>+
>+    // Get the noise power spectrum at a certain band.
>+    float Power(size_t band) const {
>+      RTC_DCHECK_LT(band, noise_spectrum_.size());
>+      return noise_spectrum_[band];
>+    }
>+
>+   private:
>+    // Get the update coefficient to be used for the current frame.
>+    float GetAlpha() const;
>+
>+    // Update the noise power spectrum at a certain band with a new frame.
>+    float UpdateBandBySmoothing(float power_band,
>+                                float power_band_noise,
>+                                float alpha) const;
>+    std::array<float, kFftLengthBy2Plus1> noise_spectrum_;
>+    size_t block_counter_;
>+  };
>+
>+  static int instance_count_;
>+  std::unique_ptr<ApmDataDumper> data_dumper_;
>+  NoiseSpectrum noise_;
>+  std::array<int, kFftLengthBy2Plus1> hangovers_;
>+  std::array<bool, kFftLengthBy2Plus1> stationarity_flags_;
>+  ReverbModel render_reverb_;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_AEC3_STATIONARITY_ESTIMATOR_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subtractor.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subtractor.cc
>index e3b4f87c0fe..b2fc993f48b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subtractor.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subtractor.cc
>@@ -16,17 +16,46 @@
> #include "api/array_view.h"
> #include "modules/audio_processing/logging/apm_data_dumper.h"
> #include "rtc_base/checks.h"
>+#include "rtc_base/logging.h"
> #include "rtc_base/numerics/safe_minmax.h"
>+#include "system_wrappers/include/field_trial.h"
> 
> namespace webrtc {
> 
> namespace {
> 
>+bool EnableAgcGainChangeResponse() {
>+  return !field_trial::IsEnabled("WebRTC-Aec3AgcGainChangeResponseKillSwitch");
>+}
>+
>+bool EnableAdaptationDuringSaturation() {
>+  return !field_trial::IsEnabled("WebRTC-Aec3RapidAgcGainRecoveryKillSwitch");
>+}
>+
>+bool EnableMisadjustmentEstimator() {
>+  return !field_trial::IsEnabled("WebRTC-Aec3MisadjustmentEstimatorKillSwitch");
>+}
>+
>+bool EnableShadowFilterJumpstart() {
>+  return !field_trial::IsEnabled("WebRTC-Aec3ShadowFilterJumpstartKillSwitch");
>+}
>+
>+bool EnableShadowFilterBoostedJumpstart() {
>+  return !field_trial::IsEnabled(
>+      "WebRTC-Aec3ShadowFilterBoostedJumpstartKillSwitch");
>+}
>+
>+bool EnableEarlyShadowFilterJumpstart() {
>+  return !field_trial::IsEnabled(
>+      "WebRTC-Aec3EarlyShadowFilterJumpstartKillSwitch");
>+}
>+
> void PredictionError(const Aec3Fft& fft,
>                      const FftData& S,
>                      rtc::ArrayView<const float> y,
>                      std::array<float, kBlockSize>* e,
>                      std::array<float, kBlockSize>* s,
>+                     bool adaptation_during_saturation,
>                      bool* saturation) {
>   std::array<float, kFftLength> tmp;
>   fft.Ifft(S, &tmp);
>@@ -48,8 +77,24 @@ void PredictionError(const Aec3Fft& fft,
>     *saturation = *result.first <= -32768 || *result.first >= 32767;
>   }
> 
>-  std::for_each(e->begin(), e->end(),
>-                [](float& a) { a = rtc::SafeClamp(a, -32768.f, 32767.f); });
>+  if (!adaptation_during_saturation) {
>+    std::for_each(e->begin(), e->end(),
>+                  [](float& a) { a = rtc::SafeClamp(a, -32768.f, 32767.f); });
>+  } else {
>+    *saturation = false;
>+  }
>+}
>+
>+void ScaleFilterOutput(rtc::ArrayView<const float> y,
>+                       float factor,
>+                       rtc::ArrayView<float> e,
>+                       rtc::ArrayView<float> s) {
>+  RTC_DCHECK_EQ(y.size(), e.size());
>+  RTC_DCHECK_EQ(y.size(), s.size());
>+  for (size_t k = 0; k < y.size(); ++k) {
>+    s[k] *= factor;
>+    e[k] = y[k] - s[k];
>+  }
> }
> 
> }  // namespace
>@@ -61,29 +106,28 @@ Subtractor::Subtractor(const EchoCanceller3Config& config,
>       data_dumper_(data_dumper),
>       optimization_(optimization),
>       config_(config),
>+      adaptation_during_saturation_(EnableAdaptationDuringSaturation()),
>+      enable_misadjustment_estimator_(EnableMisadjustmentEstimator()),
>+      enable_agc_gain_change_response_(EnableAgcGainChangeResponse()),
>+      enable_shadow_filter_jumpstart_(EnableShadowFilterJumpstart()),
>+      enable_shadow_filter_boosted_jumpstart_(
>+          EnableShadowFilterBoostedJumpstart()),
>+      enable_early_shadow_filter_jumpstart_(EnableEarlyShadowFilterJumpstart()),
>       main_filter_(config_.filter.main.length_blocks,
>+                   config_.filter.main_initial.length_blocks,
>+                   config.filter.config_change_duration_blocks,
>                    optimization,
>                    data_dumper_),
>       shadow_filter_(config_.filter.shadow.length_blocks,
>+                     config_.filter.shadow_initial.length_blocks,
>+                     config.filter.config_change_duration_blocks,
>                      optimization,
>                      data_dumper_),
>-      G_main_(config_.filter.main_initial),
>-      G_shadow_(config_.filter.shadow_initial) {
>+      G_main_(config_.filter.main_initial,
>+              config_.filter.config_change_duration_blocks),
>+      G_shadow_(config_.filter.shadow_initial,
>+                config.filter.config_change_duration_blocks) {
>   RTC_DCHECK(data_dumper_);
>-  // Currently, the rest of AEC3 requires the main and shadow filter lengths to
>-  // be identical.
>-  RTC_DCHECK_EQ(config_.filter.main.length_blocks,
>-                config_.filter.shadow.length_blocks);
>-  RTC_DCHECK_EQ(config_.filter.main_initial.length_blocks,
>-                config_.filter.shadow_initial.length_blocks);
>-
>-  RTC_DCHECK_GE(config_.filter.main.length_blocks,
>-                config_.filter.main_initial.length_blocks);
>-  RTC_DCHECK_GE(config_.filter.shadow.length_blocks,
>-                config_.filter.shadow_initial.length_blocks);
>-
>-  main_filter_.SetSizePartitions(config_.filter.main_initial.length_blocks);
>-  shadow_filter_.SetSizePartitions(config_.filter.shadow_initial.length_blocks);
> }
> 
> Subtractor::~Subtractor() = default;
>@@ -95,35 +139,31 @@ void Subtractor::HandleEchoPathChange(
>     shadow_filter_.HandleEchoPathChange();
>     G_main_.HandleEchoPathChange(echo_path_variability);
>     G_shadow_.HandleEchoPathChange();
>-    G_main_.SetConfig(config_.filter.main_initial);
>-    G_shadow_.SetConfig(config_.filter.shadow_initial);
>-    main_filter_converged_ = false;
>-    shadow_filter_converged_ = false;
>-    main_filter_.SetSizePartitions(config_.filter.main_initial.length_blocks);
>+    G_main_.SetConfig(config_.filter.main_initial, true);
>+    G_shadow_.SetConfig(config_.filter.shadow_initial, true);
>+    main_filter_.SetSizePartitions(config_.filter.main_initial.length_blocks,
>+                                   true);
>     shadow_filter_.SetSizePartitions(
>-        config_.filter.shadow_initial.length_blocks);
>+        config_.filter.shadow_initial.length_blocks, true);
>   };
> 
>-  // TODO(peah): Add delay-change specific reset behavior.
>-  if ((echo_path_variability.delay_change ==
>-       EchoPathVariability::DelayAdjustment::kBufferFlush) ||
>-      (echo_path_variability.delay_change ==
>-       EchoPathVariability::DelayAdjustment::kDelayReset)) {
>-    full_reset();
>-  } else if (echo_path_variability.delay_change ==
>-             EchoPathVariability::DelayAdjustment::kNewDetectedDelay) {
>-    full_reset();
>-  } else if (echo_path_variability.delay_change ==
>-             EchoPathVariability::DelayAdjustment::kBufferReadjustment) {
>+  if (echo_path_variability.delay_change !=
>+      EchoPathVariability::DelayAdjustment::kNone) {
>     full_reset();
>   }
>+
>+  if (echo_path_variability.gain_change && enable_agc_gain_change_response_) {
>+    RTC_LOG(LS_WARNING) << "Resetting main filter adaptation speed due to "
>+                           "microphone gain change";
>+    G_main_.HandleEchoPathChange(echo_path_variability);
>+  }
> }
> 
> void Subtractor::ExitInitialState() {
>-  G_main_.SetConfig(config_.filter.main);
>-  G_shadow_.SetConfig(config_.filter.shadow);
>-  main_filter_.SetSizePartitions(config_.filter.main.length_blocks);
>-  shadow_filter_.SetSizePartitions(config_.filter.shadow.length_blocks);
>+  G_main_.SetConfig(config_.filter.main, false);
>+  G_shadow_.SetConfig(config_.filter.shadow, false);
>+  main_filter_.SetSizePartitions(config_.filter.main.length_blocks, false);
>+  shadow_filter_.SetSizePartitions(config_.filter.shadow.length_blocks, false);
> }
> 
> void Subtractor::Process(const RenderBuffer& render_buffer,
>@@ -134,7 +174,6 @@ void Subtractor::Process(const RenderBuffer& render_buffer,
>   RTC_DCHECK_EQ(kBlockSize, capture.size());
>   rtc::ArrayView<const float> y = capture;
>   FftData& E_main = output->E_main;
>-  FftData& E_main_nonwindowed = output->E_main_nonwindowed;
>   FftData E_shadow;
>   std::array<float, kBlockSize>& e_main = output->e_main;
>   std::array<float, kBlockSize>& e_shadow = output->e_shadow;
>@@ -142,72 +181,151 @@ void Subtractor::Process(const RenderBuffer& render_buffer,
>   FftData S;
>   FftData& G = S;
> 
>-  // Form the output of the main filter.
>+  // Form the outputs of the main and shadow filters.
>   main_filter_.Filter(render_buffer, &S);
>   bool main_saturation = false;
>-  PredictionError(fft_, S, y, &e_main, &output->s_main, &main_saturation);
>-  fft_.ZeroPaddedFft(e_main, Aec3Fft::Window::kHanning, &E_main);
>+  PredictionError(fft_, S, y, &e_main, &output->s_main,
>+                  adaptation_during_saturation_, &main_saturation);
> 
>-  // Form the output of the shadow filter.
>   shadow_filter_.Filter(render_buffer, &S);
>   bool shadow_saturation = false;
>-  PredictionError(fft_, S, y, &e_shadow, nullptr, &shadow_saturation);
>-  fft_.ZeroPaddedFft(e_shadow, Aec3Fft::Window::kHanning, &E_shadow);
>-
>-  if (!(main_filter_converged_ || shadow_filter_converged_)) {
>-    const auto sum_of_squares = [](float a, float b) { return a + b * b; };
>-    const float y2 = std::accumulate(y.begin(), y.end(), 0.f, sum_of_squares);
>+  PredictionError(fft_, S, y, &e_shadow, &output->s_shadow,
>+                  adaptation_during_saturation_, &shadow_saturation);
> 
>-    if (!main_filter_converged_) {
>-      const float e2_main =
>-          std::accumulate(e_main.begin(), e_main.end(), 0.f, sum_of_squares);
>-      main_filter_converged_ = e2_main > 0.1 * y2;
>-    }
>+  // Compute the signal powers in the subtractor output.
>+  output->UpdatePowers(y);
> 
>-    if (!shadow_filter_converged_) {
>-      const float e2_shadow = std::accumulate(e_shadow.begin(), e_shadow.end(),
>-                                              0.f, sum_of_squares);
>-      shadow_filter_converged_ = e2_shadow > 0.1 * y2;
>+  // Adjust the filter if needed.
>+  bool main_filter_adjusted = false;
>+  if (enable_misadjustment_estimator_) {
>+    filter_misadjustment_estimator_.Update(*output);
>+    if (filter_misadjustment_estimator_.IsAdjustmentNeeded()) {
>+      float scale = filter_misadjustment_estimator_.GetMisadjustment();
>+      main_filter_.ScaleFilter(scale);
>+      ScaleFilterOutput(y, scale, e_main, output->s_main);
>+      filter_misadjustment_estimator_.Reset();
>+      main_filter_adjusted = true;
>     }
>   }
> 
>+  // Compute the FFts of the main and shadow filter outputs.
>+  fft_.ZeroPaddedFft(e_main, Aec3Fft::Window::kHanning, &E_main);
>+  fft_.ZeroPaddedFft(e_shadow, Aec3Fft::Window::kHanning, &E_shadow);
>+
>   // Compute spectra for future use.
>   E_shadow.Spectrum(optimization_, output->E2_shadow);
>   E_main.Spectrum(optimization_, output->E2_main);
> 
>-  if (main_filter_converged_ || !shadow_filter_converged_) {
>-    fft_.ZeroPaddedFft(e_main, Aec3Fft::Window::kRectangular,
>-                       &E_main_nonwindowed);
>-    E_main_nonwindowed.Spectrum(optimization_, output->E2_main_nonwindowed);
>+  // Compute the render powers.
>+  std::array<float, kFftLengthBy2Plus1> X2_main;
>+  std::array<float, kFftLengthBy2Plus1> X2_shadow_data;
>+  std::array<float, kFftLengthBy2Plus1>& X2_shadow =
>+      main_filter_.SizePartitions() == shadow_filter_.SizePartitions()
>+          ? X2_main
>+          : X2_shadow_data;
>+  if (main_filter_.SizePartitions() == shadow_filter_.SizePartitions()) {
>+    render_buffer.SpectralSum(main_filter_.SizePartitions(), &X2_main);
>+  } else if (main_filter_.SizePartitions() > shadow_filter_.SizePartitions()) {
>+    render_buffer.SpectralSums(shadow_filter_.SizePartitions(),
>+                               main_filter_.SizePartitions(), &X2_shadow,
>+                               &X2_main);
>   } else {
>-    fft_.ZeroPaddedFft(e_shadow, Aec3Fft::Window::kRectangular,
>-                       &E_main_nonwindowed);
>-    E_main_nonwindowed.Spectrum(optimization_, output->E2_main_nonwindowed);
>+    render_buffer.SpectralSums(main_filter_.SizePartitions(),
>+                               shadow_filter_.SizePartitions(), &X2_main,
>+                               &X2_shadow);
>   }
> 
>   // Update the main filter.
>-  std::array<float, kFftLengthBy2Plus1> X2;
>-  render_buffer.SpectralSum(main_filter_.SizePartitions(), &X2);
>-  G_main_.Compute(X2, render_signal_analyzer, *output, main_filter_,
>-                  aec_state.SaturatedCapture() || main_saturation, &G);
>+  if (!main_filter_adjusted) {
>+    G_main_.Compute(X2_main, render_signal_analyzer, *output, main_filter_,
>+                    aec_state.SaturatedCapture() || main_saturation, &G);
>+  } else {
>+    G.re.fill(0.f);
>+    G.im.fill(0.f);
>+  }
>   main_filter_.Adapt(render_buffer, G);
>   data_dumper_->DumpRaw("aec3_subtractor_G_main", G.re);
>   data_dumper_->DumpRaw("aec3_subtractor_G_main", G.im);
> 
>   // Update the shadow filter.
>-  if (shadow_filter_.SizePartitions() != main_filter_.SizePartitions()) {
>-    render_buffer.SpectralSum(shadow_filter_.SizePartitions(), &X2);
>+  poor_shadow_filter_counter_ =
>+      output->e2_main < output->e2_shadow ? poor_shadow_filter_counter_ + 1 : 0;
>+  if (((poor_shadow_filter_counter_ < 5 &&
>+        enable_early_shadow_filter_jumpstart_) ||
>+       (poor_shadow_filter_counter_ < 10 &&
>+        !enable_early_shadow_filter_jumpstart_)) ||
>+      !enable_shadow_filter_jumpstart_) {
>+    G_shadow_.Compute(X2_shadow, render_signal_analyzer, E_shadow,
>+                      shadow_filter_.SizePartitions(),
>+                      aec_state.SaturatedCapture() || shadow_saturation, &G);
>+    shadow_filter_.Adapt(render_buffer, G);
>+  } else {
>+    poor_shadow_filter_counter_ = 0;
>+    if (enable_shadow_filter_boosted_jumpstart_) {
>+      shadow_filter_.SetFilter(main_filter_.GetFilter());
>+      G_shadow_.Compute(X2_shadow, render_signal_analyzer, E_main,
>+                        shadow_filter_.SizePartitions(),
>+                        aec_state.SaturatedCapture() || main_saturation, &G);
>+      shadow_filter_.Adapt(render_buffer, G);
>+    } else {
>+      G.re.fill(0.f);
>+      G.im.fill(0.f);
>+      shadow_filter_.Adapt(render_buffer, G);
>+      shadow_filter_.SetFilter(main_filter_.GetFilter());
>+    }
>   }
>-  G_shadow_.Compute(X2, render_signal_analyzer, E_shadow,
>-                    shadow_filter_.SizePartitions(),
>-                    aec_state.SaturatedCapture() || shadow_saturation, &G);
>-  shadow_filter_.Adapt(render_buffer, G);
> 
>   data_dumper_->DumpRaw("aec3_subtractor_G_shadow", G.re);
>   data_dumper_->DumpRaw("aec3_subtractor_G_shadow", G.im);
>+  filter_misadjustment_estimator_.Dump(data_dumper_);
>+  DumpFilters();
>+
>+  if (adaptation_during_saturation_) {
>+    std::for_each(e_main.begin(), e_main.end(),
>+                  [](float& a) { a = rtc::SafeClamp(a, -32768.f, 32767.f); });
>+  }
>+
>+  data_dumper_->DumpWav("aec3_main_filter_output", kBlockSize, &e_main[0],
>+                        16000, 1);
>+  data_dumper_->DumpWav("aec3_shadow_filter_output", kBlockSize, &e_shadow[0],
>+                        16000, 1);
>+}
>+
>+void Subtractor::FilterMisadjustmentEstimator::Update(
>+    const SubtractorOutput& output) {
>+  e2_acum_ += output.e2_main;
>+  y2_acum_ += output.y2;
>+  if (++n_blocks_acum_ == n_blocks_) {
>+    if (y2_acum_ > n_blocks_ * 200.f * 200.f * kBlockSize) {
>+      float update = (e2_acum_ / y2_acum_);
>+      if (e2_acum_ > n_blocks_ * 7500.f * 7500.f * kBlockSize) {
>+        // Duration equal to blockSizeMs * n_blocks_ * 4.
>+        overhang_ = 4;
>+      } else {
>+        overhang_ = std::max(overhang_ - 1, 0);
>+      }
>+
>+      if ((update < inv_misadjustment_) || (overhang_ > 0)) {
>+        inv_misadjustment_ += 0.1f * (update - inv_misadjustment_);
>+      }
>+    }
>+    e2_acum_ = 0.f;
>+    y2_acum_ = 0.f;
>+    n_blocks_acum_ = 0;
>+  }
>+}
>+
>+void Subtractor::FilterMisadjustmentEstimator::Reset() {
>+  e2_acum_ = 0.f;
>+  y2_acum_ = 0.f;
>+  n_blocks_acum_ = 0;
>+  inv_misadjustment_ = 0.f;
>+  overhang_ = 0.f;
>+}
> 
>-  main_filter_.DumpFilter("aec3_subtractor_H_main");
>-  shadow_filter_.DumpFilter("aec3_subtractor_H_shadow");
>+void Subtractor::FilterMisadjustmentEstimator::Dump(
>+    ApmDataDumper* data_dumper) const {
>+  data_dumper->DumpRaw("aec3_inv_misadjustment_factor", inv_misadjustment_);
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subtractor.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subtractor.h
>index b3c8506c813..c92a971f117 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subtractor.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subtractor.h
>@@ -11,9 +11,10 @@
> #ifndef MODULES_AUDIO_PROCESSING_AEC3_SUBTRACTOR_H_
> #define MODULES_AUDIO_PROCESSING_AEC3_SUBTRACTOR_H_
> 
>-#include <array>
> #include <algorithm>
>+#include <array>
> #include <vector>
>+#include "math.h"
> 
> #include "modules/audio_processing/aec3/adaptive_fir_filter.h"
> #include "modules/audio_processing/aec3/aec3_common.h"
>@@ -53,33 +54,69 @@ class Subtractor {
>   // Returns the block-wise frequency response for the main adaptive filter.
>   const std::vector<std::array<float, kFftLengthBy2Plus1>>&
>   FilterFrequencyResponse() const {
>-    return main_filter_converged_ || (!shadow_filter_converged_)
>-               ? main_filter_.FilterFrequencyResponse()
>-               : shadow_filter_.FilterFrequencyResponse();
>+    return main_filter_.FilterFrequencyResponse();
>   }
> 
>   // Returns the estimate of the impulse response for the main adaptive filter.
>   const std::vector<float>& FilterImpulseResponse() const {
>-    return main_filter_converged_ || (!shadow_filter_converged_)
>-               ? main_filter_.FilterImpulseResponse()
>-               : shadow_filter_.FilterImpulseResponse();
>+    return main_filter_.FilterImpulseResponse();
>   }
> 
>-  bool ConvergedFilter() const {
>-    return main_filter_converged_ || shadow_filter_converged_;
>+  void DumpFilters() {
>+    main_filter_.DumpFilter("aec3_subtractor_H_main", "aec3_subtractor_h_main");
>+    shadow_filter_.DumpFilter("aec3_subtractor_H_shadow",
>+                              "aec3_subtractor_h_shadow");
>   }
> 
>  private:
>+  class FilterMisadjustmentEstimator {
>+   public:
>+    FilterMisadjustmentEstimator() = default;
>+    ~FilterMisadjustmentEstimator() = default;
>+    // Update the misadjustment estimator.
>+    void Update(const SubtractorOutput& output);
>+    // GetMisadjustment() Returns a recommended scale for the filter so the
>+    // prediction error energy gets closer to the energy that is seen at the
>+    // microphone input.
>+    float GetMisadjustment() const {
>+      RTC_DCHECK_GT(inv_misadjustment_, 0.0f);
>+      // It is not aiming to adjust all the estimated mismatch. Instead,
>+      // it adjusts half of that estimated mismatch.
>+      return 2.f / sqrtf(inv_misadjustment_);
>+    }
>+    // Returns true if the prediciton error energy is significantly larger
>+    // than the microphone signal energy and, therefore, an adjustment is
>+    // recommended.
>+    bool IsAdjustmentNeeded() const { return inv_misadjustment_ > 10.f; }
>+    void Reset();
>+    void Dump(ApmDataDumper* data_dumper) const;
>+
>+   private:
>+    const int n_blocks_ = 4;
>+    int n_blocks_acum_ = 0;
>+    float e2_acum_ = 0.f;
>+    float y2_acum_ = 0.f;
>+    float inv_misadjustment_ = 0.f;
>+    int overhang_ = 0.f;
>+  };
>+
>   const Aec3Fft fft_;
>   ApmDataDumper* data_dumper_;
>   const Aec3Optimization optimization_;
>   const EchoCanceller3Config config_;
>+  const bool adaptation_during_saturation_;
>+  const bool enable_misadjustment_estimator_;
>+  const bool enable_agc_gain_change_response_;
>+  const bool enable_shadow_filter_jumpstart_;
>+  const bool enable_shadow_filter_boosted_jumpstart_;
>+  const bool enable_early_shadow_filter_jumpstart_;
>+
>   AdaptiveFirFilter main_filter_;
>   AdaptiveFirFilter shadow_filter_;
>   MainFilterUpdateGain G_main_;
>   ShadowFilterUpdateGain G_shadow_;
>-  bool main_filter_converged_ = false;
>-  bool shadow_filter_converged_ = false;
>+  FilterMisadjustmentEstimator filter_misadjustment_estimator_;
>+  size_t poor_shadow_filter_counter_ = 0;
>   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(Subtractor);
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subtractor_output.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subtractor_output.cc
>new file mode 100644
>index 00000000000..affa4a3a06f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subtractor_output.cc
>@@ -0,0 +1,47 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/aec3/subtractor_output.h"
>+
>+#include <numeric>
>+
>+namespace webrtc {
>+
>+SubtractorOutput::SubtractorOutput() = default;
>+SubtractorOutput::~SubtractorOutput() = default;
>+
>+void SubtractorOutput::Reset() {
>+  s_main.fill(0.f);
>+  s_shadow.fill(0.f);
>+  e_main.fill(0.f);
>+  e_shadow.fill(0.f);
>+  E_main.re.fill(0.f);
>+  E_main.im.fill(0.f);
>+  E2_main.fill(0.f);
>+  E2_shadow.fill(0.f);
>+  e2_main = 0.f;
>+  e2_shadow = 0.f;
>+  s2_main = 0.f;
>+  s2_shadow = 0.f;
>+  y2 = 0.f;
>+}
>+
>+void SubtractorOutput::UpdatePowers(rtc::ArrayView<const float> y) {
>+  const auto sum_of_squares = [](float a, float b) { return a + b * b; };
>+  y2 = std::accumulate(y.begin(), y.end(), 0.f, sum_of_squares);
>+  e2_main = std::accumulate(e_main.begin(), e_main.end(), 0.f, sum_of_squares);
>+  e2_shadow =
>+      std::accumulate(e_shadow.begin(), e_shadow.end(), 0.f, sum_of_squares);
>+  s2_main = std::accumulate(s_main.begin(), s_main.end(), 0.f, sum_of_squares);
>+  s2_shadow =
>+      std::accumulate(s_shadow.begin(), s_shadow.end(), 0.f, sum_of_squares);
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subtractor_output.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subtractor_output.h
>index 83f6cf58b75..89727e9396f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subtractor_output.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subtractor_output.h
>@@ -13,6 +13,7 @@
> 
> #include <array>
> 
>+#include "api/array_view.h"
> #include "modules/audio_processing/aec3/aec3_common.h"
> #include "modules/audio_processing/aec3/fft_data.h"
> 
>@@ -20,24 +21,27 @@ namespace webrtc {
> 
> // Stores the values being returned from the echo subtractor.
> struct SubtractorOutput {
>+  SubtractorOutput();
>+  ~SubtractorOutput();
>+
>   std::array<float, kBlockSize> s_main;
>+  std::array<float, kBlockSize> s_shadow;
>   std::array<float, kBlockSize> e_main;
>   std::array<float, kBlockSize> e_shadow;
>   FftData E_main;
>-  FftData E_main_nonwindowed;
>   std::array<float, kFftLengthBy2Plus1> E2_main;
>-  std::array<float, kFftLengthBy2Plus1> E2_main_nonwindowed;
>   std::array<float, kFftLengthBy2Plus1> E2_shadow;
>+  float s2_main = 0.f;
>+  float s2_shadow = 0.f;
>+  float e2_main = 0.f;
>+  float e2_shadow = 0.f;
>+  float y2 = 0.f;
>+
>+  // Reset the struct content.
>+  void Reset();
> 
>-  void Reset() {
>-    s_main.fill(0.f);
>-    e_main.fill(0.f);
>-    e_shadow.fill(0.f);
>-    E_main.re.fill(0.f);
>-    E_main.im.fill(0.f);
>-    E2_main.fill(0.f);
>-    E2_shadow.fill(0.f);
>-  }
>+  // Updates the powers of the signals.
>+  void UpdatePowers(rtc::ArrayView<const float> y);
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subtractor_output_analyzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subtractor_output_analyzer.cc
>new file mode 100644
>index 00000000000..ff495857598
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subtractor_output_analyzer.cc
>@@ -0,0 +1,51 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/aec3/subtractor_output_analyzer.h"
>+
>+#include <array>
>+#include <numeric>
>+
>+#include "system_wrappers/include/field_trial.h"
>+
>+namespace webrtc {
>+namespace {
>+
>+bool EnableStrictDivergenceCheck() {
>+  return !field_trial::IsEnabled("WebRTC-Aec3StrictDivergenceCheckKillSwitch");
>+}
>+
>+}  // namespace
>+
>+SubtractorOutputAnalyzer::SubtractorOutputAnalyzer()
>+    : strict_divergence_check_(EnableStrictDivergenceCheck()) {}
>+
>+void SubtractorOutputAnalyzer::Update(
>+    const SubtractorOutput& subtractor_output) {
>+  const float y2 = subtractor_output.y2;
>+  const float e2_main = subtractor_output.e2_main;
>+  const float e2_shadow = subtractor_output.e2_shadow;
>+
>+  constexpr float kConvergenceThreshold = 50 * 50 * kBlockSize;
>+  main_filter_converged_ = e2_main < 0.5f * y2 && y2 > kConvergenceThreshold;
>+  shadow_filter_converged_ =
>+      e2_shadow < 0.05 * y2 && y2 > kConvergenceThreshold;
>+  float min_e2 =
>+      strict_divergence_check_ ? std::min(e2_main, e2_shadow) : e2_main;
>+  filter_diverged_ = min_e2 > 1.5f * y2 && y2 > 30.f * 30.f * kBlockSize;
>+}
>+
>+void SubtractorOutputAnalyzer::HandleEchoPathChange() {
>+  shadow_filter_converged_ = false;
>+  main_filter_converged_ = false;
>+  filter_diverged_ = false;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subtractor_output_analyzer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subtractor_output_analyzer.h
>new file mode 100644
>index 00000000000..b59a68e5e60
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subtractor_output_analyzer.h
>@@ -0,0 +1,46 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_AEC3_SUBTRACTOR_OUTPUT_ANALYZER_H_
>+#define MODULES_AUDIO_PROCESSING_AEC3_SUBTRACTOR_OUTPUT_ANALYZER_H_
>+
>+#include "api/array_view.h"
>+#include "modules/audio_processing/aec3/subtractor_output.h"
>+
>+namespace webrtc {
>+
>+// Class for analyzing the properties subtractor output
>+class SubtractorOutputAnalyzer {
>+ public:
>+  SubtractorOutputAnalyzer();
>+  ~SubtractorOutputAnalyzer() = default;
>+
>+  // Analyses the subtractor output.
>+  void Update(const SubtractorOutput& subtractor_output);
>+
>+  bool ConvergedFilter() const {
>+    return main_filter_converged_ || shadow_filter_converged_;
>+  }
>+
>+  bool DivergedFilter() const { return filter_diverged_; }
>+
>+  // Handle echo path change.
>+  void HandleEchoPathChange();
>+
>+ private:
>+  const bool strict_divergence_check_;
>+  bool shadow_filter_converged_ = false;
>+  bool main_filter_converged_ = false;
>+  bool filter_diverged_ = false;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_AEC3_SUBTRACTOR_OUTPUT_ANALYZER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subtractor_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subtractor_unittest.cc
>index 32221563a92..77918056b4b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subtractor_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/subtractor_unittest.cc
>@@ -25,14 +25,17 @@ namespace {
> 
> float RunSubtractorTest(int num_blocks_to_process,
>                         int delay_samples,
>-                        int filter_length_blocks,
>+                        int main_filter_length_blocks,
>+                        int shadow_filter_length_blocks,
>                         bool uncorrelated_inputs,
>                         const std::vector<int>& blocks_with_echo_path_changes) {
>   ApmDataDumper data_dumper(42);
>   EchoCanceller3Config config;
>-  config.filter.main.length_blocks = config.filter.shadow.length_blocks =
>-      filter_length_blocks;
>+  config.filter.main.length_blocks = main_filter_length_blocks;
>+  config.filter.shadow.length_blocks = shadow_filter_length_blocks;
>+
>   Subtractor subtractor(config, &data_dumper, DetectOptimization());
>+  absl::optional<DelayEstimate> delay_estimate;
>   std::vector<std::vector<float>> x(3, std::vector<float>(kBlockSize, 0.f));
>   std::vector<float> y(kBlockSize, 0.f);
>   std::array<float, kBlockSize> x_old;
>@@ -41,7 +44,7 @@ float RunSubtractorTest(int num_blocks_to_process,
>   config.delay.default_delay = 1;
>   std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
>       RenderDelayBuffer::Create(config, 3));
>-  RenderSignalAnalyzer render_signal_analyzer;
>+  RenderSignalAnalyzer render_signal_analyzer(config);
>   Random random_generator(42U);
>   Aec3Fft fft;
>   std::array<float, kFftLengthBy2Plus1> Y2;
>@@ -67,7 +70,7 @@ float RunSubtractorTest(int num_blocks_to_process,
>     }
>     render_delay_buffer->PrepareCaptureProcessing();
>     render_signal_analyzer.Update(*render_delay_buffer->GetRenderBuffer(),
>-                                  aec_state.FilterDelay());
>+                                  aec_state.FilterDelayBlocks());
> 
>     // Handle echo path changes.
>     if (std::find(blocks_with_echo_path_changes.begin(),
>@@ -82,11 +85,10 @@ float RunSubtractorTest(int num_blocks_to_process,
> 
>     aec_state.HandleEchoPathChange(EchoPathVariability(
>         false, EchoPathVariability::DelayAdjustment::kNone, false));
>-    aec_state.Update(subtractor.FilterFrequencyResponse(),
>+    aec_state.Update(delay_estimate, subtractor.FilterFrequencyResponse(),
>                      subtractor.FilterImpulseResponse(),
>-                     subtractor.ConvergedFilter(),
>                      *render_delay_buffer->GetRenderBuffer(), E2_main, Y2,
>-                     output.s_main, false);
>+                     output, y);
>   }
> 
>   const float output_power = std::inner_product(
>@@ -125,7 +127,7 @@ TEST(Subtractor, DISABLED_NullOutput) {
>   Subtractor subtractor(config, &data_dumper, DetectOptimization());
>   std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
>       RenderDelayBuffer::Create(config, 3));
>-  RenderSignalAnalyzer render_signal_analyzer;
>+  RenderSignalAnalyzer render_signal_analyzer(config);
>   std::vector<float> y(kBlockSize, 0.f);
> 
>   EXPECT_DEATH(
>@@ -141,7 +143,7 @@ TEST(Subtractor, WrongCaptureSize) {
>   Subtractor subtractor(config, &data_dumper, DetectOptimization());
>   std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
>       RenderDelayBuffer::Create(config, 3));
>-  RenderSignalAnalyzer render_signal_analyzer;
>+  RenderSignalAnalyzer render_signal_analyzer(config);
>   std::vector<float> y(kBlockSize - 1, 0.f);
>   SubtractorOutput output;
> 
>@@ -160,9 +162,9 @@ TEST(Subtractor, Convergence) {
>     for (size_t delay_samples : {0, 64, 150, 200, 301}) {
>       SCOPED_TRACE(ProduceDebugText(delay_samples, filter_length_blocks));
> 
>-      float echo_to_nearend_power =
>-          RunSubtractorTest(400, delay_samples, filter_length_blocks, false,
>-                            blocks_with_echo_path_changes);
>+      float echo_to_nearend_power = RunSubtractorTest(
>+          400, delay_samples, filter_length_blocks, filter_length_blocks, false,
>+          blocks_with_echo_path_changes);
> 
>       // Use different criteria to take overmodelling into account.
>       if (filter_length_blocks == 12) {
>@@ -174,6 +176,24 @@ TEST(Subtractor, Convergence) {
>   }
> }
> 
>+// Verifies that the subtractor is able to handle the case when the main filter
>+// is longer than the shadow filter.
>+TEST(Subtractor, MainFilterLongerThanShadowFilter) {
>+  std::vector<int> blocks_with_echo_path_changes;
>+  float echo_to_nearend_power =
>+      RunSubtractorTest(400, 64, 20, 15, false, blocks_with_echo_path_changes);
>+  EXPECT_GT(0.5f, echo_to_nearend_power);
>+}
>+
>+// Verifies that the subtractor is able to handle the case when the shadow
>+// filter is longer than the main filter.
>+TEST(Subtractor, ShadowFilterLongerThanMainFilter) {
>+  std::vector<int> blocks_with_echo_path_changes;
>+  float echo_to_nearend_power =
>+      RunSubtractorTest(400, 64, 15, 20, false, blocks_with_echo_path_changes);
>+  EXPECT_GT(0.5f, echo_to_nearend_power);
>+}
>+
> // Verifies that the subtractor does not converge on uncorrelated signals.
> TEST(Subtractor, NonConvergenceOnUncorrelatedSignals) {
>   std::vector<int> blocks_with_echo_path_changes;
>@@ -181,9 +201,9 @@ TEST(Subtractor, NonConvergenceOnUncorrelatedSignals) {
>     for (size_t delay_samples : {0, 64, 150, 200, 301}) {
>       SCOPED_TRACE(ProduceDebugText(delay_samples, filter_length_blocks));
> 
>-      float echo_to_nearend_power =
>-          RunSubtractorTest(300, delay_samples, filter_length_blocks, true,
>-                            blocks_with_echo_path_changes);
>+      float echo_to_nearend_power = RunSubtractorTest(
>+          300, delay_samples, filter_length_blocks, filter_length_blocks, true,
>+          blocks_with_echo_path_changes);
>       EXPECT_NEAR(1.f, echo_to_nearend_power, 0.1);
>     }
>   }
>@@ -198,9 +218,9 @@ TEST(Subtractor, EchoPathChangeReset) {
>     for (size_t delay_samples : {0, 64, 150, 200, 301}) {
>       SCOPED_TRACE(ProduceDebugText(delay_samples, filter_length_blocks));
> 
>-      float echo_to_nearend_power =
>-          RunSubtractorTest(100, delay_samples, filter_length_blocks, false,
>-                            blocks_with_echo_path_changes);
>+      float echo_to_nearend_power = RunSubtractorTest(
>+          100, delay_samples, filter_length_blocks, filter_length_blocks, false,
>+          blocks_with_echo_path_changes);
>       EXPECT_NEAR(1.f, echo_to_nearend_power, 0.0000001f);
>     }
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_filter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_filter.cc
>index 8c92bf5762b..87e3008dc76 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_filter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_filter.cc
>@@ -64,7 +64,6 @@ SuppressionFilter::SuppressionFilter(int sample_rate_hz)
>       fft_(),
>       e_output_old_(NumBandsForRate(sample_rate_hz_)) {
>   RTC_DCHECK(ValidFullBandRate(sample_rate_hz_));
>-  e_input_old_.fill(0.f);
>   std::for_each(e_output_old_.begin(), e_output_old_.end(),
>                 [](std::array<float, kFftLengthBy2>& a) { a.fill(0.f); });
> }
>@@ -76,22 +75,14 @@ void SuppressionFilter::ApplyGain(
>     const FftData& comfort_noise_high_band,
>     const std::array<float, kFftLengthBy2Plus1>& suppression_gain,
>     float high_bands_gain,
>+    const FftData& E_lowest_band,
>     std::vector<std::vector<float>>* e) {
>   RTC_DCHECK(e);
>   RTC_DCHECK_EQ(e->size(), NumBandsForRate(sample_rate_hz_));
>   FftData E;
>-  std::array<float, kFftLength> e_extended;
>-  constexpr float kIfftNormalization = 2.f / kFftLength;
> 
>   // Analysis filterbank.
>-  std::transform(e_input_old_.begin(), e_input_old_.end(),
>-                 std::begin(kSqrtHanning), e_extended.begin(),
>-                 std::multiplies<float>());
>-  std::transform((*e)[0].begin(), (*e)[0].end(),
>-                 std::begin(kSqrtHanning) + kFftLengthBy2,
>-                 e_extended.begin() + kFftLengthBy2, std::multiplies<float>());
>-  std::copy((*e)[0].begin(), (*e)[0].end(), e_input_old_.begin());
>-  fft_.Fft(&e_extended, &E);
>+  E.Assign(E_lowest_band);
> 
>   // Apply gain.
>   std::transform(suppression_gain.begin(), suppression_gain.end(), E.re.begin(),
>@@ -113,6 +104,9 @@ void SuppressionFilter::ApplyGain(
>                  E.im.begin(), E.im.begin(), std::plus<float>());
> 
>   // Synthesis filterbank.
>+  std::array<float, kFftLength> e_extended;
>+  constexpr float kIfftNormalization = 2.f / kFftLength;
>+
>   fft_.Ifft(E, &e_extended);
>   std::transform(e_output_old_[0].begin(), e_output_old_[0].end(),
>                  std::begin(kSqrtHanning) + kFftLengthBy2, (*e)[0].begin(),
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_filter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_filter.h
>index 5f91dea28fc..237408d7f9d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_filter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_filter.h
>@@ -28,13 +28,13 @@ class SuppressionFilter {
>                  const FftData& comfort_noise_high_bands,
>                  const std::array<float, kFftLengthBy2Plus1>& suppression_gain,
>                  float high_bands_gain,
>+                 const FftData& E_lowest_band,
>                  std::vector<std::vector<float>>* e);
> 
>  private:
>   const int sample_rate_hz_;
>   const OouraFft ooura_fft_;
>   const Aec3Fft fft_;
>-  std::array<float, kFftLengthBy2> e_input_old_;
>   std::vector<std::array<float, kFftLengthBy2>> e_output_old_;
>   RTC_DISALLOW_COPY_AND_ASSIGN(SuppressionFilter);
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_filter_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_filter_unittest.cc
>index 51b3f91f2a5..eaa608eed5d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_filter_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_filter_unittest.cc
>@@ -42,10 +42,11 @@ void ProduceSinusoid(int sample_rate_hz,
> TEST(SuppressionFilter, NullOutput) {
>   FftData cn;
>   FftData cn_high_bands;
>+  FftData E;
>   std::array<float, kFftLengthBy2Plus1> gain;
> 
>   EXPECT_DEATH(SuppressionFilter(16000).ApplyGain(cn, cn_high_bands, gain, 1.0f,
>-                                                  nullptr),
>+                                                  E, nullptr),
>                "");
> }
> 
>@@ -62,7 +63,10 @@ TEST(SuppressionFilter, ComfortNoiseInUnityGain) {
>   FftData cn;
>   FftData cn_high_bands;
>   std::array<float, kFftLengthBy2Plus1> gain;
>+  std::array<float, kFftLengthBy2> e_old_;
>+  Aec3Fft fft;
> 
>+  e_old_.fill(0.f);
>   gain.fill(1.f);
>   cn.re.fill(1.f);
>   cn.im.fill(1.f);
>@@ -71,7 +75,12 @@ TEST(SuppressionFilter, ComfortNoiseInUnityGain) {
> 
>   std::vector<std::vector<float>> e(3, std::vector<float>(kBlockSize, 0.f));
>   std::vector<std::vector<float>> e_ref = e;
>-  filter.ApplyGain(cn, cn_high_bands, gain, 1.f, &e);
>+
>+  FftData E;
>+  fft.PaddedFft(e[0], e_old_, Aec3Fft::Window::kSqrtHanning, &E);
>+  std::copy(e[0].begin(), e[0].end(), e_old_.begin());
>+
>+  filter.ApplyGain(cn, cn_high_bands, gain, 1.f, E, &e);
> 
>   for (size_t k = 0; k < e.size(); ++k) {
>     EXPECT_EQ(e_ref[k], e[k]);
>@@ -83,8 +92,11 @@ TEST(SuppressionFilter, SignalSuppression) {
>   SuppressionFilter filter(48000);
>   FftData cn;
>   FftData cn_high_bands;
>+  std::array<float, kFftLengthBy2> e_old_;
>+  Aec3Fft fft;
>   std::array<float, kFftLengthBy2Plus1> gain;
>   std::vector<std::vector<float>> e(3, std::vector<float>(kBlockSize, 0.f));
>+  e_old_.fill(0.f);
> 
>   gain.fill(1.f);
>   std::for_each(gain.begin() + 10, gain.end(), [](float& a) { a = 0.f; });
>@@ -103,7 +115,12 @@ TEST(SuppressionFilter, SignalSuppression) {
>                     e[0]);
>     e0_input =
>         std::inner_product(e[0].begin(), e[0].end(), e[0].begin(), e0_input);
>-    filter.ApplyGain(cn, cn_high_bands, gain, 1.f, &e);
>+
>+    FftData E;
>+    fft.PaddedFft(e[0], e_old_, Aec3Fft::Window::kSqrtHanning, &E);
>+    std::copy(e[0].begin(), e[0].end(), e_old_.begin());
>+
>+    filter.ApplyGain(cn, cn_high_bands, gain, 1.f, E, &e);
>     e0_output =
>         std::inner_product(e[0].begin(), e[0].end(), e[0].begin(), e0_output);
>   }
>@@ -116,10 +133,12 @@ TEST(SuppressionFilter, SignalSuppression) {
> TEST(SuppressionFilter, SignalTransparency) {
>   SuppressionFilter filter(48000);
>   FftData cn;
>+  std::array<float, kFftLengthBy2> e_old_;
>+  Aec3Fft fft;
>   FftData cn_high_bands;
>   std::array<float, kFftLengthBy2Plus1> gain;
>   std::vector<std::vector<float>> e(3, std::vector<float>(kBlockSize, 0.f));
>-
>+  e_old_.fill(0.f);
>   gain.fill(1.f);
>   std::for_each(gain.begin() + 30, gain.end(), [](float& a) { a = 0.f; });
> 
>@@ -137,7 +156,12 @@ TEST(SuppressionFilter, SignalTransparency) {
>                     e[0]);
>     e0_input =
>         std::inner_product(e[0].begin(), e[0].end(), e[0].begin(), e0_input);
>-    filter.ApplyGain(cn, cn_high_bands, gain, 1.f, &e);
>+
>+    FftData E;
>+    fft.PaddedFft(e[0], e_old_, Aec3Fft::Window::kSqrtHanning, &E);
>+    std::copy(e[0].begin(), e[0].end(), e_old_.begin());
>+
>+    filter.ApplyGain(cn, cn_high_bands, gain, 1.f, E, &e);
>     e0_output =
>         std::inner_product(e[0].begin(), e[0].end(), e[0].begin(), e0_output);
>   }
>@@ -150,6 +174,8 @@ TEST(SuppressionFilter, Delay) {
>   SuppressionFilter filter(48000);
>   FftData cn;
>   FftData cn_high_bands;
>+  std::array<float, kFftLengthBy2> e_old_;
>+  Aec3Fft fft;
>   std::array<float, kFftLengthBy2Plus1> gain;
>   std::vector<std::vector<float>> e(3, std::vector<float>(kBlockSize, 0.f));
> 
>@@ -167,7 +193,11 @@ TEST(SuppressionFilter, Delay) {
>       }
>     }
> 
>-    filter.ApplyGain(cn, cn_high_bands, gain, 1.f, &e);
>+    FftData E;
>+    fft.PaddedFft(e[0], e_old_, Aec3Fft::Window::kSqrtHanning, &E);
>+    std::copy(e[0].begin(), e[0].end(), e_old_.begin());
>+
>+    filter.ApplyGain(cn, cn_high_bands, gain, 1.f, E, &e);
>     if (k > 2) {
>       for (size_t j = 0; j < 2; ++j) {
>         for (size_t i = 0; i < kBlockSize; ++i) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_gain.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_gain.cc
>index d1543c69835..a042c5fd9af 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_gain.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_gain.cc
>@@ -1,3 +1,4 @@
>+
> /*
>  *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>  *
>@@ -10,7 +11,9 @@
> 
> #include "modules/audio_processing/aec3/suppression_gain.h"
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>+// Defines WEBRTC_ARCH_X86_FAMILY, used below.
>+#include "rtc_base/system/arch.h"
>+
> #if defined(WEBRTC_ARCH_X86_FAMILY)
> #include <emmintrin.h>
> #endif
>@@ -19,20 +22,23 @@
> #include <functional>
> #include <numeric>
> 
>+#include "modules/audio_processing/aec3/moving_average.h"
> #include "modules/audio_processing/aec3/vector_math.h"
>+#include "modules/audio_processing/logging/apm_data_dumper.h"
>+#include "rtc_base/atomicops.h"
> #include "rtc_base/checks.h"
>+#include "system_wrappers/include/field_trial.h"
> 
> namespace webrtc {
> namespace {
> 
>-// Reduce gain to avoid narrow band echo leakage.
>-void NarrowBandAttenuation(int narrow_bin,
>-                           std::array<float, kFftLengthBy2Plus1>* gain) {
>-  const int upper_bin =
>-      std::min(narrow_bin + 6, static_cast<int>(kFftLengthBy2Plus1 - 1));
>-  for (int k = std::max(0, narrow_bin - 6); k <= upper_bin; ++k) {
>-    (*gain)[k] = std::min((*gain)[k], 0.001f);
>-  }
>+bool EnableTransparencyImprovements() {
>+  return !field_trial::IsEnabled(
>+      "WebRTC-Aec3TransparencyImprovementsKillSwitch");
>+}
>+
>+bool EnableNewSuppression() {
>+  return !field_trial::IsEnabled("WebRTC-Aec3NewSuppressionKillSwitch");
> }
> 
> // Adjust the gains according to the presence of known external filters.
>@@ -55,7 +61,7 @@ void AdjustForExternalFilters(std::array<float, kFftLengthBy2Plus1>* gain) {
> 
> // Computes the gain to apply for the bands beyond the first band.
> float UpperBandsGain(
>-    const rtc::Optional<int>& narrow_peak_band,
>+    const absl::optional<int>& narrow_peak_band,
>     bool saturated_echo,
>     const std::vector<std::vector<float>>& render,
>     const std::array<float, kFftLengthBy2Plus1>& low_band_gain) {
>@@ -107,91 +113,61 @@ float UpperBandsGain(
>   return std::min(gain_below_8_khz, anti_howling_gain);
> }
> 
>-// Limits the gain increase.
>-void UpdateMaxGainIncrease(
>-    const EchoCanceller3Config& config,
>-    size_t no_saturation_counter,
>-    bool low_noise_render,
>-    bool initial_state,
>-    bool linear_echo_estimate,
>-    const std::array<float, kFftLengthBy2Plus1>& last_echo,
>-    const std::array<float, kFftLengthBy2Plus1>& echo,
>-    const std::array<float, kFftLengthBy2Plus1>& last_gain,
>-    const std::array<float, kFftLengthBy2Plus1>& new_gain,
>-    std::array<float, kFftLengthBy2Plus1>* gain_increase) {
>-  float max_increasing;
>-  float max_decreasing;
>-  float rate_increasing;
>-  float rate_decreasing;
>-  float min_increasing;
>-  float min_decreasing;
>-
>-  auto& param = config.gain_updates;
>-  if (!linear_echo_estimate) {
>-    max_increasing = param.nonlinear.max_inc;
>-    max_decreasing = param.nonlinear.max_dec;
>-    rate_increasing = param.nonlinear.rate_inc;
>-    rate_decreasing = param.nonlinear.rate_dec;
>-    min_increasing = param.nonlinear.min_inc;
>-    min_decreasing = param.nonlinear.min_dec;
>-  } else if (initial_state && no_saturation_counter > 10) {
>-    max_increasing = param.initial.max_inc;
>-    max_decreasing = param.initial.max_dec;
>-    rate_increasing = param.initial.rate_inc;
>-    rate_decreasing = param.initial.rate_dec;
>-    min_increasing = param.initial.min_inc;
>-    min_decreasing = param.initial.min_dec;
>-  } else if (low_noise_render) {
>-    max_increasing = param.low_noise.max_inc;
>-    max_decreasing = param.low_noise.max_dec;
>-    rate_increasing = param.low_noise.rate_inc;
>-    rate_decreasing = param.low_noise.rate_dec;
>-    min_increasing = param.low_noise.min_inc;
>-    min_decreasing = param.low_noise.min_dec;
>-  } else if (no_saturation_counter > 10) {
>-    max_increasing = param.normal.max_inc;
>-    max_decreasing = param.normal.max_dec;
>-    rate_increasing = param.normal.rate_inc;
>-    rate_decreasing = param.normal.rate_dec;
>-    min_increasing = param.normal.min_inc;
>-    min_decreasing = param.normal.min_dec;
>-  } else {
>-    max_increasing = param.saturation.max_inc;
>-    max_decreasing = param.saturation.max_dec;
>-    rate_increasing = param.saturation.rate_inc;
>-    rate_decreasing = param.saturation.rate_dec;
>-    min_increasing = param.saturation.min_inc;
>-    min_decreasing = param.saturation.min_dec;
>-  }
>-
>-  for (size_t k = 0; k < new_gain.size(); ++k) {
>-    if (echo[k] > last_echo[k]) {
>-      (*gain_increase)[k] =
>-          new_gain[k] > last_gain[k]
>-              ? std::min(max_increasing, (*gain_increase)[k] * rate_increasing)
>-              : min_increasing;
>-    } else {
>-      (*gain_increase)[k] =
>-          new_gain[k] > last_gain[k]
>-              ? std::min(max_decreasing, (*gain_increase)[k] * rate_decreasing)
>-              : min_decreasing;
>+// Scales the echo according to assessed audibility at the other end.
>+void WeightEchoForAudibility(const EchoCanceller3Config& config,
>+                             rtc::ArrayView<const float> echo,
>+                             rtc::ArrayView<float> weighted_echo,
>+                             rtc::ArrayView<float> one_by_weighted_echo) {
>+  RTC_DCHECK_EQ(kFftLengthBy2Plus1, echo.size());
>+  RTC_DCHECK_EQ(kFftLengthBy2Plus1, weighted_echo.size());
>+  RTC_DCHECK_EQ(kFftLengthBy2Plus1, one_by_weighted_echo.size());
>+
>+  auto weigh = [](float threshold, float normalizer, size_t begin, size_t end,
>+                  rtc::ArrayView<const float> echo,
>+                  rtc::ArrayView<float> weighted_echo,
>+                  rtc::ArrayView<float> one_by_weighted_echo) {
>+    for (size_t k = begin; k < end; ++k) {
>+      if (echo[k] < threshold) {
>+        float tmp = (threshold - echo[k]) * normalizer;
>+        weighted_echo[k] = echo[k] * std::max(0.f, 1.f - tmp * tmp);
>+      } else {
>+        weighted_echo[k] = echo[k];
>+      }
>+      one_by_weighted_echo[k] =
>+          weighted_echo[k] > 0.f ? 1.f / weighted_echo[k] : 1.f;
>     }
>-  }
>+  };
>+
>+  float threshold = config.echo_audibility.floor_power *
>+                    config.echo_audibility.audibility_threshold_lf;
>+  float normalizer = 1.f / (threshold - config.echo_audibility.floor_power);
>+  weigh(threshold, normalizer, 0, 3, echo, weighted_echo, one_by_weighted_echo);
>+
>+  threshold = config.echo_audibility.floor_power *
>+              config.echo_audibility.audibility_threshold_mf;
>+  normalizer = 1.f / (threshold - config.echo_audibility.floor_power);
>+  weigh(threshold, normalizer, 3, 7, echo, weighted_echo, one_by_weighted_echo);
>+
>+  threshold = config.echo_audibility.floor_power *
>+              config.echo_audibility.audibility_threshold_hf;
>+  normalizer = 1.f / (threshold - config.echo_audibility.floor_power);
>+  weigh(threshold, normalizer, 7, kFftLengthBy2Plus1, echo, weighted_echo,
>+        one_by_weighted_echo);
> }
> 
> // Computes the gain to reduce the echo to a non audible level.
>-void GainToNoAudibleEcho(
>+void GainToNoAudibleEchoFallback(
>     const EchoCanceller3Config& config,
>     bool low_noise_render,
>     bool saturated_echo,
>-    bool saturating_echo_path,
>     bool linear_echo_estimate,
>+    bool enable_transparency_improvements,
>     const std::array<float, kFftLengthBy2Plus1>& nearend,
>-    const std::array<float, kFftLengthBy2Plus1>& echo,
>+    const std::array<float, kFftLengthBy2Plus1>& weighted_echo,
>     const std::array<float, kFftLengthBy2Plus1>& masker,
>     const std::array<float, kFftLengthBy2Plus1>& min_gain,
>     const std::array<float, kFftLengthBy2Plus1>& max_gain,
>-    const std::array<float, kFftLengthBy2Plus1>& one_by_echo,
>+    const std::array<float, kFftLengthBy2Plus1>& one_by_weighted_echo,
>     std::array<float, kFftLengthBy2Plus1>* gain) {
>   float nearend_masking_margin = 0.f;
>   if (linear_echo_estimate) {
>@@ -205,24 +181,29 @@ void GainToNoAudibleEcho(
> 
>   RTC_DCHECK_LE(0.f, nearend_masking_margin);
>   RTC_DCHECK_GT(1.f, nearend_masking_margin);
>-  const float one_by_one_minus_nearend_masking_margin =
>-      1.f / (1.0f - nearend_masking_margin);
> 
>   const float masker_margin =
>-      linear_echo_estimate ? config.gain_mask.m1 : config.gain_mask.m8;
>+      linear_echo_estimate
>+          ? (enable_transparency_improvements ? config.gain_mask.m0
>+                                              : config.gain_mask.m1)
>+          : config.gain_mask.m8;
> 
>   for (size_t k = 0; k < gain->size(); ++k) {
>+    // TODO(devicentepena): Experiment by removing the reverberation estimation
>+    // from the nearend signal before computing the gains.
>     const float unity_gain_masker = std::max(nearend[k], masker[k]);
>     RTC_DCHECK_LE(0.f, nearend_masking_margin * unity_gain_masker);
>-    if (echo[k] <= nearend_masking_margin * unity_gain_masker ||
>+    if (weighted_echo[k] <= nearend_masking_margin * unity_gain_masker ||
>         unity_gain_masker <= 0.f) {
>       (*gain)[k] = 1.f;
>     } else {
>       RTC_DCHECK_LT(0.f, unity_gain_masker);
>-      (*gain)[k] = std::max(0.f, (1.f - 5.f * echo[k] / unity_gain_masker) *
>-                                     one_by_one_minus_nearend_masking_margin);
>       (*gain)[k] =
>-          std::max(masker_margin * masker[k] * one_by_echo[k], (*gain)[k]);
>+          std::max(0.f, (1.f - config.gain_mask.gain_curve_slope *
>+                                   weighted_echo[k] / unity_gain_masker) *
>+                            config.gain_mask.gain_curve_offset);
>+      (*gain)[k] = std::max(masker_margin * masker[k] * one_by_weighted_echo[k],
>+                            (*gain)[k]);
>     }
> 
>     (*gain)[k] = std::min(std::max((*gain)[k], min_gain[k]), max_gain[k]);
>@@ -234,11 +215,31 @@ constexpr size_t kUpperAccurateBandPlus1 = 29;
> 
> // Computes the signal output power that masks the echo signal.
> void MaskingPower(const EchoCanceller3Config& config,
>+                  bool enable_transparency_improvements,
>                   const std::array<float, kFftLengthBy2Plus1>& nearend,
>                   const std::array<float, kFftLengthBy2Plus1>& comfort_noise,
>                   const std::array<float, kFftLengthBy2Plus1>& last_masker,
>                   const std::array<float, kFftLengthBy2Plus1>& gain,
>                   std::array<float, kFftLengthBy2Plus1>* masker) {
>+  if (enable_transparency_improvements) {
>+    std::copy(comfort_noise.begin(), comfort_noise.end(), masker->begin());
>+    return;
>+  }
>+
>+  // Apply masking over time.
>+  float masking_factor = config.gain_mask.temporal_masking_lf;
>+  auto limit = config.gain_mask.temporal_masking_lf_bands;
>+  std::transform(
>+      comfort_noise.begin(), comfort_noise.begin() + limit, last_masker.begin(),
>+      masker->begin(),
>+      [masking_factor](float a, float b) { return a + masking_factor * b; });
>+  masking_factor = config.gain_mask.temporal_masking_hf;
>+  std::transform(
>+      comfort_noise.begin() + limit, comfort_noise.end(),
>+      last_masker.begin() + limit, masker->begin() + limit,
>+      [masking_factor](float a, float b) { return a + masking_factor * b; });
>+
>+  // Apply masking only between lower frequency bands.
>   std::array<float, kFftLengthBy2Plus1> side_band_masker;
>   float max_nearend_after_gain = 0.f;
>   for (size_t k = 0; k < gain.size(); ++k) {
>@@ -246,10 +247,8 @@ void MaskingPower(const EchoCanceller3Config& config,
>     max_nearend_after_gain =
>         std::max(max_nearend_after_gain, nearend_after_gain);
>     side_band_masker[k] = nearend_after_gain + comfort_noise[k];
>-    (*masker)[k] = comfort_noise[k] + config.gain_mask.m4 * last_masker[k];
>   }
> 
>-  // Apply masking only between lower frequency bands.
>   RTC_DCHECK_LT(kUpperAccurateBandPlus1, gain.size());
>   for (size_t k = 1; k < kUpperAccurateBandPlus1; ++k) {
>     (*masker)[k] += config.gain_mask.m5 *
>@@ -281,26 +280,44 @@ void AdjustNonConvergedFrequencies(
> 
> }  // namespace
> 
>+int SuppressionGain::instance_count_ = 0;
>+
>+// Computes the gain to reduce the echo to a non audible level.
>+void SuppressionGain::GainToNoAudibleEcho(
>+    const std::array<float, kFftLengthBy2Plus1>& nearend,
>+    const std::array<float, kFftLengthBy2Plus1>& echo,
>+    const std::array<float, kFftLengthBy2Plus1>& masker,
>+    const std::array<float, kFftLengthBy2Plus1>& min_gain,
>+    const std::array<float, kFftLengthBy2Plus1>& max_gain,
>+    std::array<float, kFftLengthBy2Plus1>* gain) const {
>+  for (size_t k = 0; k < gain->size(); ++k) {
>+    float enr = echo[k] / (nearend[k] + 1.f);  // Echo-to-nearend ratio.
>+    float emr = echo[k] / (masker[k] + 1.f);   // Echo-to-masker (noise) ratio.
>+    float g = 1.0f;
>+    if (enr > enr_transparent_[k] && emr > emr_transparent_[k]) {
>+      g = (enr_suppress_[k] - enr) / (enr_suppress_[k] - enr_transparent_[k]);
>+      g = std::max(g, emr_transparent_[k] / emr);
>+    }
>+    (*gain)[k] = std::max(std::min(g, max_gain[k]), min_gain[k]);
>+  }
>+}
>+
> // TODO(peah): Add further optimizations, in particular for the divisions.
> void SuppressionGain::LowerBandGain(
>     bool low_noise_render,
>-    const rtc::Optional<int>& narrow_peak_band,
>-    bool saturated_echo,
>-    bool saturating_echo_path,
>-    bool initial_state,
>-    bool linear_echo_estimate,
>+    const AecState& aec_state,
>     const std::array<float, kFftLengthBy2Plus1>& nearend,
>     const std::array<float, kFftLengthBy2Plus1>& echo,
>     const std::array<float, kFftLengthBy2Plus1>& comfort_noise,
>     std::array<float, kFftLengthBy2Plus1>* gain) {
>-  // Count the number of blocks since saturation.
>-  no_saturation_counter_ = saturated_echo ? 0 : no_saturation_counter_ + 1;
>+  const bool saturated_echo = aec_state.SaturatedEcho();
>+  const bool linear_echo_estimate = aec_state.UsableLinearEstimate();
> 
>-  // Precompute 1/echo (note that when the echo is zero, the precomputed value
>-  // is never used).
>-  std::array<float, kFftLengthBy2Plus1> one_by_echo;
>-  std::transform(echo.begin(), echo.end(), one_by_echo.begin(),
>-                 [](float a) { return a > 0.f ? 1.f / a : 1.f; });
>+  // Weight echo power in terms of audibility. // Precompute 1/weighted echo
>+  // (note that when the echo is zero, the precomputed value is never used).
>+  std::array<float, kFftLengthBy2Plus1> weighted_echo;
>+  std::array<float, kFftLengthBy2Plus1> one_by_weighted_echo;
>+  WeightEchoForAudibility(config_, echo, weighted_echo, one_by_weighted_echo);
> 
>   // Compute the minimum gain as the attenuating gain to put the signal just
>   // above the zero sample values.
>@@ -308,12 +325,24 @@ void SuppressionGain::LowerBandGain(
>   const float min_echo_power =
>       low_noise_render ? config_.echo_audibility.low_render_limit
>                        : config_.echo_audibility.normal_render_limit;
>-  if (no_saturation_counter_ > 10) {
>+  if (!saturated_echo) {
>     for (size_t k = 0; k < nearend.size(); ++k) {
>-      const float denom = std::min(nearend[k], echo[k]);
>+      const float denom = std::min(nearend[k], weighted_echo[k]);
>       min_gain[k] = denom > 0.f ? min_echo_power / denom : 1.f;
>       min_gain[k] = std::min(min_gain[k], 1.f);
>     }
>+    if (enable_transparency_improvements_) {
>+      for (size_t k = 0; k < 6; ++k) {
>+        // Make sure the gains of the low frequencies do not decrease too
>+        // quickly after strong nearend.
>+        if (last_nearend_[k] > last_echo_[k]) {
>+          min_gain[k] =
>+              std::max(min_gain[k],
>+                       last_gain_[k] * config_.gain_updates.max_dec_factor_lf);
>+          min_gain[k] = std::min(min_gain[k], 1.f);
>+        }
>+      }
>+    }
>   } else {
>     min_gain.fill(0.f);
>   }
>@@ -321,24 +350,39 @@ void SuppressionGain::LowerBandGain(
>   // Compute the maximum gain by limiting the gain increase from the previous
>   // gain.
>   std::array<float, kFftLengthBy2Plus1> max_gain;
>-  for (size_t k = 0; k < gain->size(); ++k) {
>-    max_gain[k] = std::min(std::max(last_gain_[k] * gain_increase_[k],
>-                                    config_.gain_updates.floor_first_increase),
>-                           1.f);
>+  if (enable_transparency_improvements_) {
>+    for (size_t k = 0; k < gain->size(); ++k) {
>+      max_gain[k] =
>+          std::min(std::max(last_gain_[k] * config_.gain_updates.max_inc_factor,
>+                            config_.gain_updates.floor_first_increase),
>+                   1.f);
>+    }
>+  } else {
>+    for (size_t k = 0; k < gain->size(); ++k) {
>+      max_gain[k] =
>+          std::min(std::max(last_gain_[k] * gain_increase_[k],
>+                            config_.gain_updates.floor_first_increase),
>+                   1.f);
>+    }
>   }
> 
>   // Iteratively compute the gain required to attenuate the echo to a non
>   // noticeable level.
>-  gain->fill(0.f);
>-  for (int k = 0; k < 2; ++k) {
>-    std::array<float, kFftLengthBy2Plus1> masker;
>-    MaskingPower(config_, nearend, comfort_noise, last_masker_, *gain, &masker);
>-    GainToNoAudibleEcho(config_, low_noise_render, saturated_echo,
>-                        saturating_echo_path, linear_echo_estimate, nearend,
>-                        echo, masker, min_gain, max_gain, one_by_echo, gain);
>+  std::array<float, kFftLengthBy2Plus1> masker;
>+  if (enable_new_suppression_) {
>+    GainToNoAudibleEcho(nearend, weighted_echo, comfort_noise, min_gain,
>+                        max_gain, gain);
>     AdjustForExternalFilters(gain);
>-    if (narrow_peak_band) {
>-      NarrowBandAttenuation(*narrow_peak_band, gain);
>+  } else {
>+    gain->fill(0.f);
>+    for (int k = 0; k < 2; ++k) {
>+      MaskingPower(config_, enable_transparency_improvements_, nearend,
>+                   comfort_noise, last_masker_, *gain, &masker);
>+      GainToNoAudibleEchoFallback(
>+          config_, low_noise_render, saturated_echo, linear_echo_estimate,
>+          enable_transparency_improvements_, nearend, weighted_echo, masker,
>+          min_gain, max_gain, one_by_weighted_echo, gain);
>+      AdjustForExternalFilters(gain);
>     }
>   }
> 
>@@ -346,37 +390,79 @@ void SuppressionGain::LowerBandGain(
>   AdjustNonConvergedFrequencies(gain);
> 
>   // Update the allowed maximum gain increase.
>-  UpdateMaxGainIncrease(config_, no_saturation_counter_, low_noise_render,
>-                        initial_state, linear_echo_estimate, last_echo_, echo,
>-                        last_gain_, *gain, &gain_increase_);
>-
>-  // Adjust gain dynamics.
>-  const float gain_bound =
>-      std::max(0.001f, *std::min_element(gain->begin(), gain->end()) * 10000.f);
>-  std::for_each(gain->begin(), gain->end(),
>-                [gain_bound](float& a) { a = std::min(a, gain_bound); });
>+  UpdateGainIncrease(low_noise_render, linear_echo_estimate, saturated_echo,
>+                     weighted_echo, *gain);
> 
>   // Store data required for the gain computation of the next block.
>-  std::copy(echo.begin(), echo.end(), last_echo_.begin());
>+  std::copy(nearend.begin(), nearend.end(), last_nearend_.begin());
>+  std::copy(weighted_echo.begin(), weighted_echo.end(), last_echo_.begin());
>   std::copy(gain->begin(), gain->end(), last_gain_.begin());
>-  MaskingPower(config_, nearend, comfort_noise, last_masker_, *gain,
>-               &last_masker_);
>+  MaskingPower(config_, enable_transparency_improvements_, nearend,
>+               comfort_noise, last_masker_, *gain, &last_masker_);
>   aec3::VectorMath(optimization_).Sqrt(*gain);
>+
>+  // Debug outputs for the purpose of development and analysis.
>+  data_dumper_->DumpRaw("aec3_suppressor_min_gain", min_gain);
>+  data_dumper_->DumpRaw("aec3_suppressor_max_gain", max_gain);
>+  data_dumper_->DumpRaw("aec3_suppressor_masker", masker);
>+  data_dumper_->DumpRaw("aec3_suppressor_last_masker", last_masker_);
> }
> 
> SuppressionGain::SuppressionGain(const EchoCanceller3Config& config,
>-                                 Aec3Optimization optimization)
>-    : optimization_(optimization), config_(config) {
>+                                 Aec3Optimization optimization,
>+                                 int sample_rate_hz)
>+    : data_dumper_(
>+          new ApmDataDumper(rtc::AtomicOps::Increment(&instance_count_))),
>+      optimization_(optimization),
>+      config_(config),
>+      state_change_duration_blocks_(
>+          static_cast<int>(config_.filter.config_change_duration_blocks)),
>+      coherence_gain_(sample_rate_hz,
>+                      config_.suppressor.bands_with_reliable_coherence),
>+      enable_transparency_improvements_(EnableTransparencyImprovements()),
>+      enable_new_suppression_(EnableNewSuppression()),
>+      moving_average_(kFftLengthBy2Plus1,
>+                      config.suppressor.nearend_average_blocks) {
>+  RTC_DCHECK_LT(0, state_change_duration_blocks_);
>+  one_by_state_change_duration_blocks_ = 1.f / state_change_duration_blocks_;
>   last_gain_.fill(1.f);
>   last_masker_.fill(0.f);
>   gain_increase_.fill(1.f);
>+  last_nearend_.fill(0.f);
>   last_echo_.fill(0.f);
>+
>+  // Compute per-band masking thresholds.
>+  constexpr size_t kLastLfBand = 5;
>+  constexpr size_t kFirstHfBand = 8;
>+  RTC_DCHECK_LT(kLastLfBand, kFirstHfBand);
>+  auto& lf = config.suppressor.mask_lf;
>+  auto& hf = config.suppressor.mask_hf;
>+  RTC_DCHECK_LT(lf.enr_transparent, lf.enr_suppress);
>+  RTC_DCHECK_LT(hf.enr_transparent, hf.enr_suppress);
>+  for (size_t k = 0; k < kFftLengthBy2Plus1; k++) {
>+    float a;
>+    if (k <= kLastLfBand) {
>+      a = 0.f;
>+    } else if (k < kFirstHfBand) {
>+      a = (k - kLastLfBand) / static_cast<float>(kFirstHfBand - kLastLfBand);
>+    } else {
>+      a = 1.f;
>+    }
>+    enr_transparent_[k] = (1 - a) * lf.enr_transparent + a * hf.enr_transparent;
>+    enr_suppress_[k] = (1 - a) * lf.enr_suppress + a * hf.enr_suppress;
>+    emr_transparent_[k] = (1 - a) * lf.emr_transparent + a * hf.emr_transparent;
>+  }
> }
> 
>+SuppressionGain::~SuppressionGain() = default;
>+
> void SuppressionGain::GetGain(
>-    const std::array<float, kFftLengthBy2Plus1>& nearend,
>-    const std::array<float, kFftLengthBy2Plus1>& echo,
>-    const std::array<float, kFftLengthBy2Plus1>& comfort_noise,
>+    const std::array<float, kFftLengthBy2Plus1>& nearend_spectrum,
>+    const std::array<float, kFftLengthBy2Plus1>& echo_spectrum,
>+    const std::array<float, kFftLengthBy2Plus1>& comfort_noise_spectrum,
>+    const FftData& linear_aec_fft,
>+    const FftData& render_fft,
>+    const FftData& capture_fft,
>     const RenderSignalAnalyzer& render_signal_analyzer,
>     const AecState& aec_state,
>     const std::vector<std::vector<float>>& render,
>@@ -385,32 +471,144 @@ void SuppressionGain::GetGain(
>   RTC_DCHECK(high_bands_gain);
>   RTC_DCHECK(low_band_gain);
> 
>-  const bool saturated_echo = aec_state.SaturatedEcho();
>-  const bool saturating_echo_path = aec_state.SaturatingEchoPath();
>-  const bool force_zero_gain = aec_state.ForcedZeroGain();
>-  const bool linear_echo_estimate = aec_state.UsableLinearEstimate();
>-  const bool initial_state = aec_state.InitialState();
>-  if (force_zero_gain) {
>-    last_gain_.fill(0.f);
>-    std::copy(comfort_noise.begin(), comfort_noise.end(), last_masker_.begin());
>-    low_band_gain->fill(0.f);
>-    gain_increase_.fill(1.f);
>-    *high_bands_gain = 0.f;
>-    return;
>-  }
>-
>-  bool low_noise_render = low_render_detector_.Detect(render);
>+  std::array<float, kFftLengthBy2Plus1> nearend_average;
>+  moving_average_.Average(nearend_spectrum, nearend_average);
> 
>   // Compute gain for the lower band.
>-  const rtc::Optional<int> narrow_peak_band =
>+  bool low_noise_render = low_render_detector_.Detect(render);
>+  const absl::optional<int> narrow_peak_band =
>       render_signal_analyzer.NarrowPeakBand();
>-  LowerBandGain(low_noise_render, narrow_peak_band, saturated_echo,
>-                saturating_echo_path, initial_state, linear_echo_estimate,
>-                nearend, echo, comfort_noise, low_band_gain);
>+  LowerBandGain(low_noise_render, aec_state, nearend_average, echo_spectrum,
>+                comfort_noise_spectrum, low_band_gain);
>+
>+  // Adjust the gain for bands where the coherence indicates not echo.
>+  if (config_.suppressor.bands_with_reliable_coherence > 0 &&
>+      !enable_transparency_improvements_) {
>+    std::array<float, kFftLengthBy2Plus1> G_coherence;
>+    coherence_gain_.ComputeGain(linear_aec_fft, render_fft, capture_fft,
>+                                G_coherence);
>+    for (size_t k = 0; k < config_.suppressor.bands_with_reliable_coherence;
>+         ++k) {
>+      (*low_band_gain)[k] = std::max((*low_band_gain)[k], G_coherence[k]);
>+    }
>+  }
>+
>+  // Limit the gain of the lower bands during start up and after resets.
>+  const float gain_upper_bound = aec_state.SuppressionGainLimit();
>+  if (gain_upper_bound < 1.f) {
>+    for (size_t k = 0; k < low_band_gain->size(); ++k) {
>+      (*low_band_gain)[k] = std::min((*low_band_gain)[k], gain_upper_bound);
>+    }
>+  }
> 
>   // Compute the gain for the upper bands.
>-  *high_bands_gain =
>-      UpperBandsGain(narrow_peak_band, saturated_echo, render, *low_band_gain);
>+  *high_bands_gain = UpperBandsGain(narrow_peak_band, aec_state.SaturatedEcho(),
>+                                    render, *low_band_gain);
>+}
>+
>+void SuppressionGain::SetInitialState(bool state) {
>+  initial_state_ = state;
>+  if (state) {
>+    initial_state_change_counter_ = state_change_duration_blocks_;
>+  } else {
>+    initial_state_change_counter_ = 0;
>+  }
>+}
>+
>+void SuppressionGain::UpdateGainIncrease(
>+    bool low_noise_render,
>+    bool linear_echo_estimate,
>+    bool saturated_echo,
>+    const std::array<float, kFftLengthBy2Plus1>& echo,
>+    const std::array<float, kFftLengthBy2Plus1>& new_gain) {
>+  float max_inc;
>+  float max_dec;
>+  float rate_inc;
>+  float rate_dec;
>+  float min_inc;
>+  float min_dec;
>+
>+  RTC_DCHECK_GE(state_change_duration_blocks_, initial_state_change_counter_);
>+  if (initial_state_change_counter_ > 0) {
>+    if (--initial_state_change_counter_ == 0) {
>+      initial_state_ = false;
>+    }
>+  }
>+  RTC_DCHECK_LE(0, initial_state_change_counter_);
>+
>+  // EchoCanceller3Config::GainUpdates
>+  auto& p = config_.gain_updates;
>+  if (!linear_echo_estimate) {
>+    max_inc = p.nonlinear.max_inc;
>+    max_dec = p.nonlinear.max_dec;
>+    rate_inc = p.nonlinear.rate_inc;
>+    rate_dec = p.nonlinear.rate_dec;
>+    min_inc = p.nonlinear.min_inc;
>+    min_dec = p.nonlinear.min_dec;
>+  } else if (initial_state_ && !saturated_echo) {
>+    if (initial_state_change_counter_ > 0) {
>+      float change_factor =
>+          initial_state_change_counter_ * one_by_state_change_duration_blocks_;
>+
>+      auto average = [](float from, float to, float from_weight) {
>+        return from * from_weight + to * (1.f - from_weight);
>+      };
>+
>+      max_inc = average(p.initial.max_inc, p.normal.max_inc, change_factor);
>+      max_dec = average(p.initial.max_dec, p.normal.max_dec, change_factor);
>+      rate_inc = average(p.initial.rate_inc, p.normal.rate_inc, change_factor);
>+      rate_dec = average(p.initial.rate_dec, p.normal.rate_dec, change_factor);
>+      min_inc = average(p.initial.min_inc, p.normal.min_inc, change_factor);
>+      min_dec = average(p.initial.min_dec, p.normal.min_dec, change_factor);
>+    } else {
>+      max_inc = p.initial.max_inc;
>+      max_dec = p.initial.max_dec;
>+      rate_inc = p.initial.rate_inc;
>+      rate_dec = p.initial.rate_dec;
>+      min_inc = p.initial.min_inc;
>+      min_dec = p.initial.min_dec;
>+    }
>+  } else if (low_noise_render) {
>+    max_inc = p.low_noise.max_inc;
>+    max_dec = p.low_noise.max_dec;
>+    rate_inc = p.low_noise.rate_inc;
>+    rate_dec = p.low_noise.rate_dec;
>+    min_inc = p.low_noise.min_inc;
>+    min_dec = p.low_noise.min_dec;
>+  } else if (!saturated_echo) {
>+    max_inc = p.normal.max_inc;
>+    max_dec = p.normal.max_dec;
>+    rate_inc = p.normal.rate_inc;
>+    rate_dec = p.normal.rate_dec;
>+    min_inc = p.normal.min_inc;
>+    min_dec = p.normal.min_dec;
>+  } else {
>+    max_inc = p.saturation.max_inc;
>+    max_dec = p.saturation.max_dec;
>+    rate_inc = p.saturation.rate_inc;
>+    rate_dec = p.saturation.rate_dec;
>+    min_inc = p.saturation.min_inc;
>+    min_dec = p.saturation.min_dec;
>+  }
>+
>+  for (size_t k = 0; k < new_gain.size(); ++k) {
>+    auto increase_update = [](float new_gain, float last_gain,
>+                              float current_inc, float max_inc, float min_inc,
>+                              float change_rate) {
>+      return new_gain > last_gain ? std::min(max_inc, current_inc * change_rate)
>+                                  : min_inc;
>+    };
>+
>+    if (echo[k] > last_echo_[k]) {
>+      gain_increase_[k] =
>+          increase_update(new_gain[k], last_gain_[k], gain_increase_[k],
>+                          max_inc, min_inc, rate_inc);
>+    } else {
>+      gain_increase_[k] =
>+          increase_update(new_gain[k], last_gain_[k], gain_increase_[k],
>+                          max_dec, min_dec, rate_dec);
>+    }
>+  }
> }
> 
> // Detects when the render signal can be considered to have low power and
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_gain.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_gain.h
>index 36e4e343e8c..4c0c85dcf73 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_gain.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_gain.h
>@@ -14,10 +14,12 @@
> #include <array>
> #include <vector>
> 
>+#include "api/audio/echo_canceller3_config.h"
> #include "modules/audio_processing/aec3/aec3_common.h"
> #include "modules/audio_processing/aec3/aec_state.h"
>+#include "modules/audio_processing/aec3/coherence_gain.h"
>+#include "modules/audio_processing/aec3/moving_average.h"
> #include "modules/audio_processing/aec3/render_signal_analyzer.h"
>-#include "modules/audio_processing/include/audio_processing.h"
> #include "rtc_base/constructormagic.h"
> 
> namespace webrtc {
>@@ -25,28 +27,49 @@ namespace webrtc {
> class SuppressionGain {
>  public:
>   SuppressionGain(const EchoCanceller3Config& config,
>-                  Aec3Optimization optimization);
>-  void GetGain(const std::array<float, kFftLengthBy2Plus1>& nearend,
>-               const std::array<float, kFftLengthBy2Plus1>& echo,
>-               const std::array<float, kFftLengthBy2Plus1>& comfort_noise,
>-               const RenderSignalAnalyzer& render_signal_analyzer,
>-               const AecState& aec_state,
>-               const std::vector<std::vector<float>>& render,
>-               float* high_bands_gain,
>-               std::array<float, kFftLengthBy2Plus1>* low_band_gain);
>+                  Aec3Optimization optimization,
>+                  int sample_rate_hz);
>+  ~SuppressionGain();
>+  void GetGain(
>+      const std::array<float, kFftLengthBy2Plus1>& nearend_spectrum,
>+      const std::array<float, kFftLengthBy2Plus1>& echo_spectrum,
>+      const std::array<float, kFftLengthBy2Plus1>& comfort_noise_spectrum,
>+      const FftData& linear_aec_fft,
>+      const FftData& render_fft,
>+      const FftData& capture_fft,
>+      const RenderSignalAnalyzer& render_signal_analyzer,
>+      const AecState& aec_state,
>+      const std::vector<std::vector<float>>& render,
>+      float* high_bands_gain,
>+      std::array<float, kFftLengthBy2Plus1>* low_band_gain);
>+
>+  // Toggles the usage of the initial state.
>+  void SetInitialState(bool state);
> 
>  private:
>+  void GainToNoAudibleEcho(
>+      const std::array<float, kFftLengthBy2Plus1>& nearend,
>+      const std::array<float, kFftLengthBy2Plus1>& echo,
>+      const std::array<float, kFftLengthBy2Plus1>& masker,
>+      const std::array<float, kFftLengthBy2Plus1>& min_gain,
>+      const std::array<float, kFftLengthBy2Plus1>& max_gain,
>+      std::array<float, kFftLengthBy2Plus1>* gain) const;
>+
>   void LowerBandGain(bool stationary_with_low_power,
>-                     const rtc::Optional<int>& narrow_peak_band,
>-                     bool saturated_echo,
>-                     bool saturating_echo_path,
>-                     bool initial_state,
>-                     bool linear_echo_estimate,
>+                     const AecState& aec_state,
>                      const std::array<float, kFftLengthBy2Plus1>& nearend,
>                      const std::array<float, kFftLengthBy2Plus1>& echo,
>                      const std::array<float, kFftLengthBy2Plus1>& comfort_noise,
>                      std::array<float, kFftLengthBy2Plus1>* gain);
> 
>+  // Limits the gain increase.
>+  void UpdateGainIncrease(
>+      bool low_noise_render,
>+      bool linear_echo_estimate,
>+      bool saturated_echo,
>+      const std::array<float, kFftLengthBy2Plus1>& echo,
>+      const std::array<float, kFftLengthBy2Plus1>& new_gain);
>+
>   class LowNoiseRenderDetector {
>    public:
>     bool Detect(const std::vector<std::vector<float>>& render);
>@@ -55,15 +78,28 @@ class SuppressionGain {
>     float average_power_ = 32768.f * 32768.f;
>   };
> 
>+  static int instance_count_;
>+  std::unique_ptr<ApmDataDumper> data_dumper_;
>   const Aec3Optimization optimization_;
>+  const EchoCanceller3Config config_;
>+  const int state_change_duration_blocks_;
>+  float one_by_state_change_duration_blocks_;
>   std::array<float, kFftLengthBy2Plus1> last_gain_;
>   std::array<float, kFftLengthBy2Plus1> last_masker_;
>   std::array<float, kFftLengthBy2Plus1> gain_increase_;
>+  std::array<float, kFftLengthBy2Plus1> last_nearend_;
>   std::array<float, kFftLengthBy2Plus1> last_echo_;
>-
>+  std::array<float, kFftLengthBy2Plus1> enr_transparent_;
>+  std::array<float, kFftLengthBy2Plus1> enr_suppress_;
>+  std::array<float, kFftLengthBy2Plus1> emr_transparent_;
>   LowNoiseRenderDetector low_render_detector_;
>-  size_t no_saturation_counter_ = 0;
>-  const EchoCanceller3Config config_;
>+  bool initial_state_ = true;
>+  int initial_state_change_counter_ = 0;
>+  CoherenceGain coherence_gain_;
>+  const bool enable_transparency_improvements_;
>+  const bool enable_new_suppression_;
>+  aec3::MovingAverage moving_average_;
>+
>   RTC_DISALLOW_COPY_AND_ASSIGN(SuppressionGain);
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_gain_limiter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_gain_limiter.cc
>new file mode 100644
>index 00000000000..e3d7a660bf0
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_gain_limiter.cc
>@@ -0,0 +1,92 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/aec3/suppression_gain_limiter.h"
>+
>+#include <math.h>
>+#include <algorithm>
>+
>+#include "modules/audio_processing/aec3/aec3_common.h"
>+
>+namespace webrtc {
>+namespace {
>+
>+// Computes the gain rampup factor to use.
>+float ComputeGainRampupIncrease(
>+    const EchoCanceller3Config::EchoRemovalControl::GainRampup& rampup_config) {
>+  return powf(1.f / rampup_config.first_non_zero_gain,
>+              1.f / rampup_config.non_zero_gain_blocks);
>+}
>+
>+}  // namespace
>+
>+SuppressionGainUpperLimiter::SuppressionGainUpperLimiter(
>+    const EchoCanceller3Config& config)
>+    : rampup_config_(config.echo_removal_control.gain_rampup),
>+      gain_rampup_increase_(ComputeGainRampupIncrease(rampup_config_)) {
>+  Reset();
>+}
>+
>+void SuppressionGainUpperLimiter::Reset() {
>+  recent_reset_ = true;
>+}
>+
>+void SuppressionGainUpperLimiter::Update(bool render_activity,
>+                                         bool transparent_mode) {
>+  if (transparent_mode) {
>+    active_render_seen_ = true;
>+    call_startup_phase_ = false;
>+    recent_reset_ = false;
>+    suppressor_gain_limit_ = 1.f;
>+    return;
>+  }
>+
>+  if (recent_reset_ && !call_startup_phase_) {
>+    // Only enforce 250 ms full suppression after in-call resets,
>+    constexpr int kMuteFramesAfterReset = kNumBlocksPerSecond / 4;
>+    realignment_counter_ = kMuteFramesAfterReset;
>+  } else if (!active_render_seen_ && render_activity) {
>+    // Enforce a tailormade suppression limiting during call startup.
>+    active_render_seen_ = true;
>+    realignment_counter_ = rampup_config_.full_gain_blocks;
>+  } else if (realignment_counter_ > 0) {
>+    if (--realignment_counter_ == 0) {
>+      call_startup_phase_ = false;
>+    }
>+  }
>+  recent_reset_ = false;
>+
>+  // Do not enforce any gain limit on the suppressor.
>+  if (!IsActive()) {
>+    suppressor_gain_limit_ = 1.f;
>+    return;
>+  }
>+
>+  // Enforce full suppression.
>+  if (realignment_counter_ > rampup_config_.non_zero_gain_blocks ||
>+      (!call_startup_phase_ && realignment_counter_ > 0)) {
>+    suppressor_gain_limit_ = rampup_config_.initial_gain;
>+    return;
>+  }
>+
>+  // Start increasing the gain limit.
>+  if (realignment_counter_ == rampup_config_.non_zero_gain_blocks) {
>+    suppressor_gain_limit_ = rampup_config_.first_non_zero_gain;
>+    return;
>+  }
>+
>+  // Increase the gain limit until it reaches 1.f.
>+  RTC_DCHECK_LT(0.f, suppressor_gain_limit_);
>+  suppressor_gain_limit_ =
>+      std::min(1.f, suppressor_gain_limit_ * gain_rampup_increase_);
>+  RTC_DCHECK_GE(1.f, suppressor_gain_limit_);
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_gain_limiter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_gain_limiter.h
>new file mode 100644
>index 00000000000..717868ac0a6
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_gain_limiter.h
>@@ -0,0 +1,52 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_AEC3_SUPPRESSION_GAIN_LIMITER_H_
>+#define MODULES_AUDIO_PROCESSING_AEC3_SUPPRESSION_GAIN_LIMITER_H_
>+
>+#include "api/array_view.h"
>+#include "api/audio/echo_canceller3_config.h"
>+#include "rtc_base/constructormagic.h"
>+
>+namespace webrtc {
>+
>+// Class for applying a smoothly increasing limit for the suppression gain
>+// during call startup and after in-call resets.
>+class SuppressionGainUpperLimiter {
>+ public:
>+  explicit SuppressionGainUpperLimiter(const EchoCanceller3Config& config);
>+
>+  // Reset the limiting behavior.
>+  void Reset();
>+
>+  // Updates the limiting behavior for the current capture bloc.
>+  void Update(bool render_activity, bool transparent_mode);
>+
>+  // Returns the current suppressor gain limit.
>+  float Limit() const { return suppressor_gain_limit_; }
>+
>+  // Return whether the suppressor gain limit is active.
>+  bool IsActive() const { return (realignment_counter_ > 0); }
>+
>+ private:
>+  const EchoCanceller3Config::EchoRemovalControl::GainRampup rampup_config_;
>+  const float gain_rampup_increase_;
>+  bool call_startup_phase_ = true;
>+  int realignment_counter_ = 0;
>+  bool active_render_seen_ = false;
>+  float suppressor_gain_limit_ = 1.f;
>+  bool recent_reset_ = false;
>+
>+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(SuppressionGainUpperLimiter);
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_AEC3_SUPPRESSION_GAIN_LIMITER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_gain_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_gain_unittest.cc
>index bcdcd23c328..630596ea79e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_gain_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/suppression_gain_unittest.cc
>@@ -13,11 +13,11 @@
> #include "modules/audio_processing/aec3/aec_state.h"
> #include "modules/audio_processing/aec3/render_delay_buffer.h"
> #include "modules/audio_processing/aec3/subtractor.h"
>+#include "modules/audio_processing/aec3/subtractor_output.h"
> #include "modules/audio_processing/logging/apm_data_dumper.h"
> #include "rtc_base/checks.h"
> #include "system_wrappers/include/cpu_features_wrapper.h"
> #include "test/gtest.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> namespace aec3 {
>@@ -29,33 +29,49 @@ TEST(SuppressionGain, NullOutputGains) {
>   std::array<float, kFftLengthBy2Plus1> E2;
>   std::array<float, kFftLengthBy2Plus1> R2;
>   std::array<float, kFftLengthBy2Plus1> N2;
>+  FftData E;
>+  FftData X;
>+  FftData Y;
>   E2.fill(0.f);
>   R2.fill(0.f);
>   N2.fill(0.f);
>+  E.re.fill(0.f);
>+  E.im.fill(0.f);
>+  X.re.fill(0.f);
>+  X.im.fill(0.f);
>+  Y.re.fill(0.f);
>+  Y.im.fill(0.f);
>+
>   float high_bands_gain;
>   AecState aec_state(EchoCanceller3Config{});
>-  EXPECT_DEATH(SuppressionGain(EchoCanceller3Config{}, DetectOptimization())
>-                   .GetGain(E2, R2, N2, RenderSignalAnalyzer(), aec_state,
>-                            std::vector<std::vector<float>>(
>-                                3, std::vector<float>(kBlockSize, 0.f)),
>-                            &high_bands_gain, nullptr),
>-               "");
>+  EXPECT_DEATH(
>+      SuppressionGain(EchoCanceller3Config{}, DetectOptimization(), 16000)
>+          .GetGain(E2, R2, N2, E, X, Y,
>+                   RenderSignalAnalyzer((EchoCanceller3Config{})), aec_state,
>+                   std::vector<std::vector<float>>(
>+                       3, std::vector<float>(kBlockSize, 0.f)),
>+                   &high_bands_gain, nullptr),
>+      "");
> }
> 
> #endif
> 
> // Does a sanity check that the gains are correctly computed.
> TEST(SuppressionGain, BasicGainComputation) {
>-  SuppressionGain suppression_gain(EchoCanceller3Config(),
>-                                   DetectOptimization());
>-  RenderSignalAnalyzer analyzer;
>+  SuppressionGain suppression_gain(EchoCanceller3Config(), DetectOptimization(),
>+                                   16000);
>+  RenderSignalAnalyzer analyzer(EchoCanceller3Config{});
>   float high_bands_gain;
>   std::array<float, kFftLengthBy2Plus1> E2;
>   std::array<float, kFftLengthBy2Plus1> Y2;
>   std::array<float, kFftLengthBy2Plus1> R2;
>   std::array<float, kFftLengthBy2Plus1> N2;
>   std::array<float, kFftLengthBy2Plus1> g;
>-  std::array<float, kBlockSize> s;
>+  SubtractorOutput output;
>+  std::array<float, kBlockSize> y;
>+  FftData E;
>+  FftData X;
>+  FftData Y;
>   std::vector<std::vector<float>> x(1, std::vector<float>(kBlockSize, 0.f));
>   EchoCanceller3Config config;
>   AecState aec_state(config);
>@@ -63,40 +79,36 @@ TEST(SuppressionGain, BasicGainComputation) {
>   Subtractor subtractor(config, &data_dumper, DetectOptimization());
>   std::unique_ptr<RenderDelayBuffer> render_delay_buffer(
>       RenderDelayBuffer::Create(config, 3));
>-
>-  // Verify the functionality for forcing a zero gain.
>-  E2.fill(1000000000.f);
>-  R2.fill(10000000000000.f);
>-  N2.fill(0.f);
>-  s.fill(10.f);
>-  aec_state.Update(subtractor.FilterFrequencyResponse(),
>-                   subtractor.FilterImpulseResponse(),
>-                   subtractor.ConvergedFilter(),
>-                   *render_delay_buffer->GetRenderBuffer(), E2, Y2, s, false);
>-  suppression_gain.GetGain(E2, R2, N2, analyzer, aec_state, x, &high_bands_gain,
>-                           &g);
>-  std::for_each(g.begin(), g.end(), [](float a) { EXPECT_FLOAT_EQ(0.f, a); });
>-  EXPECT_FLOAT_EQ(0.f, high_bands_gain);
>+  absl::optional<DelayEstimate> delay_estimate;
> 
>   // Ensure that a strong noise is detected to mask any echoes.
>   E2.fill(10.f);
>   Y2.fill(10.f);
>   R2.fill(0.1f);
>   N2.fill(100.f);
>+  output.Reset();
>+  y.fill(0.f);
>+  E.re.fill(sqrtf(E2[0]));
>+  E.im.fill(0.f);
>+  X.re.fill(sqrtf(R2[0]));
>+  X.im.fill(0.f);
>+  Y.re.fill(sqrtf(Y2[0]));
>+  Y.im.fill(0.f);
>+
>   // Ensure that the gain is no longer forced to zero.
>   for (int k = 0; k <= kNumBlocksPerSecond / 5 + 1; ++k) {
>-    aec_state.Update(subtractor.FilterFrequencyResponse(),
>+    aec_state.Update(delay_estimate, subtractor.FilterFrequencyResponse(),
>                      subtractor.FilterImpulseResponse(),
>-                     subtractor.ConvergedFilter(),
>-                     *render_delay_buffer->GetRenderBuffer(), E2, Y2, s, false);
>+                     *render_delay_buffer->GetRenderBuffer(), E2, Y2, output,
>+                     y);
>   }
> 
>   for (int k = 0; k < 100; ++k) {
>-    aec_state.Update(subtractor.FilterFrequencyResponse(),
>+    aec_state.Update(delay_estimate, subtractor.FilterFrequencyResponse(),
>                      subtractor.FilterImpulseResponse(),
>-                     subtractor.ConvergedFilter(),
>-                     *render_delay_buffer->GetRenderBuffer(), E2, Y2, s, false);
>-    suppression_gain.GetGain(E2, R2, N2, analyzer, aec_state, x,
>+                     *render_delay_buffer->GetRenderBuffer(), E2, Y2, output,
>+                     y);
>+    suppression_gain.GetGain(E2, R2, N2, E, X, Y, analyzer, aec_state, x,
>                              &high_bands_gain, &g);
>   }
>   std::for_each(g.begin(), g.end(),
>@@ -107,12 +119,16 @@ TEST(SuppressionGain, BasicGainComputation) {
>   Y2.fill(100.f);
>   R2.fill(0.1f);
>   N2.fill(0.f);
>+  E.re.fill(sqrtf(E2[0]));
>+  X.re.fill(sqrtf(R2[0]));
>+  Y.re.fill(sqrtf(Y2[0]));
>+
>   for (int k = 0; k < 100; ++k) {
>-    aec_state.Update(subtractor.FilterFrequencyResponse(),
>+    aec_state.Update(delay_estimate, subtractor.FilterFrequencyResponse(),
>                      subtractor.FilterImpulseResponse(),
>-                     subtractor.ConvergedFilter(),
>-                     *render_delay_buffer->GetRenderBuffer(), E2, Y2, s, false);
>-    suppression_gain.GetGain(E2, R2, N2, analyzer, aec_state, x,
>+                     *render_delay_buffer->GetRenderBuffer(), E2, Y2, output,
>+                     y);
>+    suppression_gain.GetGain(E2, R2, N2, E, X, Y, analyzer, aec_state, x,
>                              &high_bands_gain, &g);
>   }
>   std::for_each(g.begin(), g.end(),
>@@ -121,14 +137,15 @@ TEST(SuppressionGain, BasicGainComputation) {
>   // Ensure that a strong echo is suppressed.
>   E2.fill(1000000000.f);
>   R2.fill(10000000000000.f);
>-  N2.fill(0.f);
>+  E.re.fill(sqrtf(E2[0]));
>+  X.re.fill(sqrtf(R2[0]));
>+
>   for (int k = 0; k < 10; ++k) {
>-    suppression_gain.GetGain(E2, R2, N2, analyzer, aec_state, x,
>+    suppression_gain.GetGain(E2, R2, N2, E, X, Y, analyzer, aec_state, x,
>                              &high_bands_gain, &g);
>   }
>   std::for_each(g.begin(), g.end(),
>                 [](float a) { EXPECT_NEAR(0.f, a, 0.001); });
>-
> }
> 
> }  // namespace aec3
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/vector_buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/vector_buffer.h
>index a7f99325457..c7d4f68b27c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/vector_buffer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/vector_buffer.h
>@@ -36,6 +36,7 @@ struct VectorBuffer {
>   int OffsetIndex(int index, int offset) const {
>     RTC_DCHECK_GE(size, offset);
>     RTC_DCHECK_EQ(buffer.size(), static_cast<size_t>(size));
>+    RTC_DCHECK_GE(size + index + offset, 0);
>     return (size + index + offset) % size;
>   }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/vector_math.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/vector_math.h
>index 0672b513f3c..255331b672e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/vector_math.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/vector_math.h
>@@ -11,7 +11,9 @@
> #ifndef MODULES_AUDIO_PROCESSING_AEC3_VECTOR_MATH_H_
> #define MODULES_AUDIO_PROCESSING_AEC3_VECTOR_MATH_H_
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>+// Defines WEBRTC_ARCH_X86_FAMILY, used below.
>+#include "rtc_base/system/arch.h"
>+
> #if defined(WEBRTC_HAS_NEON)
> #include <arm_neon.h>
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/vector_math_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/vector_math_unittest.cc
>index 6bf60ecb64d..fdab2e52cae 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/vector_math_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec3/vector_math_unittest.cc
>@@ -12,9 +12,9 @@
> 
> #include <math.h>
> 
>+#include "rtc_base/system/arch.h"
> #include "system_wrappers/include/cpu_features_wrapper.h"
> #include "test/gtest.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec_dump/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec_dump/BUILD.gn
>index aac4b0a49d5..e5fee3e95f7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec_dump/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec_dump/BUILD.gn
>@@ -14,11 +14,8 @@ rtc_source_set("aec_dump") {
>     "aec_dump_factory.h",
>   ]
> 
>-  public_deps = [
>-    "..:aec_dump_interface",
>-  ]
>-
>   deps = [
>+    "../",
>     "../../../rtc_base:rtc_base_approved",
>   ]
> }
>@@ -31,12 +28,8 @@ rtc_source_set("mock_aec_dump") {
>   ]
> 
>   deps = [
>-    "..:aec_dump_interface",
>-  ]
>-  public_deps = [
>-    "../..:module_api",
>+    "../",
>     "../../../test:test_support",
>-    "//testing/gmock",
>   ]
> }
> 
>@@ -49,9 +42,10 @@ rtc_source_set("mock_aec_dump_unittests") {
> 
>   deps = [
>     ":mock_aec_dump",
>-    "..:audio_processing",
>+    "../",
>     "../../../rtc_base:rtc_base_approved",
>     "//testing/gtest",
>+    "//third_party/abseil-cpp/absl/memory",
>   ]
> }
> 
>@@ -66,20 +60,17 @@ if (rtc_enable_protobuf) {
>       "write_to_file_task.h",
>     ]
> 
>-    public = []
>-
>-    public_deps = [
>-      ":aec_dump",
>-      "..:aec_dump_interface",
>-    ]
>-
>     deps = [
>-      "../../../modules:module_api",
>+      ":aec_dump",
>+      "../",
>+      "../../../api/audio:audio_frame_api",
>       "../../../rtc_base:checks",
>       "../../../rtc_base:protobuf_utils",
>       "../../../rtc_base:rtc_base_approved",
>       "../../../rtc_base:rtc_task_queue",
>+      "../../../rtc_base/system:file_wrapper",
>       "../../../system_wrappers",
>+      "//third_party/abseil-cpp/absl/memory",
>     ]
> 
>     deps += [ "../:audioproc_debug_proto" ]
>@@ -89,11 +80,12 @@ if (rtc_enable_protobuf) {
>     testonly = true
>     defines = []
>     deps = [
>+      ":aec_dump",
>       ":aec_dump_impl",
>-      "..:aec_dump_interface",
>       "..:audioproc_debug_proto",
>-      "../../../modules:module_api",
>+      "../",
>       "../../../rtc_base:rtc_task_queue",
>+      "../../../test:fileutils",
>       "../../../test:test_support",
>       "//testing/gtest",
>     ]
>@@ -109,8 +101,8 @@ rtc_source_set("null_aec_dump_factory") {
>     "null_aec_dump_factory.cc",
>   ]
> 
>-  public_deps = [
>+  deps = [
>     ":aec_dump",
>-    "..:aec_dump_interface",
>+    "../",
>   ]
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec_dump/aec_dump_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec_dump/aec_dump_impl.cc
>index 594bf85cd9d..d94822dbfce 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec_dump/aec_dump_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec_dump/aec_dump_impl.cc
>@@ -12,10 +12,10 @@
> 
> #include "modules/audio_processing/aec_dump/aec_dump_impl.h"
> 
>+#include "absl/memory/memory.h"
> #include "modules/audio_processing/aec_dump/aec_dump_factory.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/event.h"
>-#include "rtc_base/ptr_util.h"
> 
> namespace webrtc {
> 
>@@ -48,6 +48,10 @@ void CopyFromConfigToEvent(const webrtc::InternalAPMConfig& config,
>   pb_cfg->set_intelligibility_enhancer_enabled(
>       config.intelligibility_enhancer_enabled);
> 
>+  pb_cfg->set_pre_amplifier_enabled(config.pre_amplifier_enabled);
>+  pb_cfg->set_pre_amplifier_fixed_gain_factor(
>+      config.pre_amplifier_fixed_gain_factor);
>+
>   pb_cfg->set_experiments_description(config.experiments_description);
> }
> 
>@@ -70,35 +74,40 @@ AecDumpImpl::~AecDumpImpl() {
>   thread_sync_event.Wait(rtc::Event::kForever);
> }
> 
>-void AecDumpImpl::WriteInitMessage(
>-    const InternalAPMStreamsConfig& streams_config) {
>+void AecDumpImpl::WriteInitMessage(const ProcessingConfig& api_format,
>+                                   int64_t time_now_ms) {
>   auto task = CreateWriteToFileTask();
>   auto* event = task->GetEvent();
>   event->set_type(audioproc::Event::INIT);
>   audioproc::Init* msg = event->mutable_init();
> 
>-  msg->set_sample_rate(streams_config.input_sample_rate);
>-  msg->set_output_sample_rate(streams_config.output_sample_rate);
>-  msg->set_reverse_sample_rate(streams_config.render_input_sample_rate);
>-  msg->set_reverse_output_sample_rate(streams_config.render_output_sample_rate);
>+  msg->set_sample_rate(api_format.input_stream().sample_rate_hz());
>+  msg->set_output_sample_rate(api_format.output_stream().sample_rate_hz());
>+  msg->set_reverse_sample_rate(
>+      api_format.reverse_input_stream().sample_rate_hz());
>+  msg->set_reverse_output_sample_rate(
>+      api_format.reverse_output_stream().sample_rate_hz());
> 
>   msg->set_num_input_channels(
>-      static_cast<int32_t>(streams_config.input_num_channels));
>+      static_cast<int32_t>(api_format.input_stream().num_channels()));
>   msg->set_num_output_channels(
>-      static_cast<int32_t>(streams_config.output_num_channels));
>+      static_cast<int32_t>(api_format.output_stream().num_channels()));
>   msg->set_num_reverse_channels(
>-      static_cast<int32_t>(streams_config.render_input_num_channels));
>+      static_cast<int32_t>(api_format.reverse_input_stream().num_channels()));
>   msg->set_num_reverse_output_channels(
>-      streams_config.render_output_num_channels);
>+      api_format.reverse_output_stream().num_channels());
>+  msg->set_timestamp_ms(time_now_ms);
> 
>   worker_queue_->PostTask(std::unique_ptr<rtc::QueuedTask>(std::move(task)));
> }
> 
>-void AecDumpImpl::AddCaptureStreamInput(const FloatAudioFrame& src) {
>+void AecDumpImpl::AddCaptureStreamInput(
>+    const AudioFrameView<const float>& src) {
>   capture_stream_info_.AddInput(src);
> }
> 
>-void AecDumpImpl::AddCaptureStreamOutput(const FloatAudioFrame& src) {
>+void AecDumpImpl::AddCaptureStreamOutput(
>+    const AudioFrameView<const float>& src) {
>   capture_stream_info_.AddOutput(src);
> }
> 
>@@ -134,7 +143,8 @@ void AecDumpImpl::WriteRenderStreamMessage(const AudioFrame& frame) {
>   worker_queue_->PostTask(std::unique_ptr<rtc::QueuedTask>(std::move(task)));
> }
> 
>-void AecDumpImpl::WriteRenderStreamMessage(const FloatAudioFrame& src) {
>+void AecDumpImpl::WriteRenderStreamMessage(
>+    const AudioFrameView<const float>& src) {
>   auto task = CreateWriteToFileTask();
>   auto* event = task->GetEvent();
> 
>@@ -160,8 +170,8 @@ void AecDumpImpl::WriteConfig(const InternalAPMConfig& config) {
> }
> 
> std::unique_ptr<WriteToFileTask> AecDumpImpl::CreateWriteToFileTask() {
>-  return rtc::MakeUnique<WriteToFileTask>(debug_file_.get(),
>-                                          &num_bytes_left_for_log_);
>+  return absl::make_unique<WriteToFileTask>(debug_file_.get(),
>+                                            &num_bytes_left_for_log_);
> }
> 
> std::unique_ptr<AecDump> AecDumpFactory::Create(rtc::PlatformFile file,
>@@ -176,8 +186,8 @@ std::unique_ptr<AecDump> AecDumpFactory::Create(rtc::PlatformFile file,
>   if (!debug_file->OpenFromFileHandle(handle)) {
>     return nullptr;
>   }
>-  return rtc::MakeUnique<AecDumpImpl>(std::move(debug_file), max_log_size_bytes,
>-                                      worker_queue);
>+  return absl::make_unique<AecDumpImpl>(std::move(debug_file),
>+                                        max_log_size_bytes, worker_queue);
> }
> 
> std::unique_ptr<AecDump> AecDumpFactory::Create(std::string file_name,
>@@ -188,8 +198,8 @@ std::unique_ptr<AecDump> AecDumpFactory::Create(std::string file_name,
>   if (!debug_file->OpenFile(file_name.c_str(), false)) {
>     return nullptr;
>   }
>-  return rtc::MakeUnique<AecDumpImpl>(std::move(debug_file), max_log_size_bytes,
>-                                      worker_queue);
>+  return absl::make_unique<AecDumpImpl>(std::move(debug_file),
>+                                        max_log_size_bytes, worker_queue);
> }
> 
> std::unique_ptr<AecDump> AecDumpFactory::Create(FILE* handle,
>@@ -201,7 +211,7 @@ std::unique_ptr<AecDump> AecDumpFactory::Create(FILE* handle,
>   if (!debug_file->OpenFromFileHandle(handle)) {
>     return nullptr;
>   }
>-  return rtc::MakeUnique<AecDumpImpl>(std::move(debug_file), max_log_size_bytes,
>-                                      worker_queue);
>+  return absl::make_unique<AecDumpImpl>(std::move(debug_file),
>+                                        max_log_size_bytes, worker_queue);
> }
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec_dump/aec_dump_impl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec_dump/aec_dump_impl.h
>index 5be876b6b73..a5416a07da8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec_dump/aec_dump_impl.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec_dump/aec_dump_impl.h
>@@ -15,16 +15,16 @@
> #include <string>
> #include <vector>
> 
>+#include "api/audio/audio_frame.h"
> #include "modules/audio_processing/aec_dump/capture_stream_info.h"
> #include "modules/audio_processing/aec_dump/write_to_file_task.h"
> #include "modules/audio_processing/include/aec_dump.h"
>-#include "modules/include/module_common_types.h"
> #include "rtc_base/ignore_wundef.h"
> #include "rtc_base/platform_file.h"
> #include "rtc_base/race_checker.h"
>+#include "rtc_base/system/file_wrapper.h"
> #include "rtc_base/task_queue.h"
> #include "rtc_base/thread_annotations.h"
>-#include "system_wrappers/include/file_wrapper.h"
> 
> // Files generated at build-time by the protobuf compiler.
> RTC_PUSH_IGNORING_WUNDEF()
>@@ -52,17 +52,18 @@ class AecDumpImpl : public AecDump {
> 
>   ~AecDumpImpl() override;
> 
>-  void WriteInitMessage(const InternalAPMStreamsConfig& api_format) override;
>-
>-  void AddCaptureStreamInput(const FloatAudioFrame& src) override;
>-  void AddCaptureStreamOutput(const FloatAudioFrame& src) override;
>+  void WriteInitMessage(const ProcessingConfig& api_format,
>+                        int64_t time_now_ms) override;
>+  void AddCaptureStreamInput(const AudioFrameView<const float>& src) override;
>+  void AddCaptureStreamOutput(const AudioFrameView<const float>& src) override;
>   void AddCaptureStreamInput(const AudioFrame& frame) override;
>   void AddCaptureStreamOutput(const AudioFrame& frame) override;
>   void AddAudioProcessingState(const AudioProcessingState& state) override;
>   void WriteCaptureStreamMessage() override;
> 
>   void WriteRenderStreamMessage(const AudioFrame& frame) override;
>-  void WriteRenderStreamMessage(const FloatAudioFrame& src) override;
>+  void WriteRenderStreamMessage(
>+      const AudioFrameView<const float>& src) override;
> 
>   void WriteConfig(const InternalAPMConfig& config) override;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec_dump/aec_dump_integration_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec_dump/aec_dump_integration_test.cc
>index a7d53b522f6..b6d8756d274 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec_dump/aec_dump_integration_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec_dump/aec_dump_integration_test.cc
>@@ -10,9 +10,9 @@
> 
> #include <utility>
> 
>+#include "absl/memory/memory.h"
> #include "modules/audio_processing/aec_dump/mock_aec_dump.h"
> #include "modules/audio_processing/include/audio_processing.h"
>-#include "rtc_base/ptr_util.h"
> 
> using testing::_;
> using testing::AtLeast;
>@@ -31,14 +31,14 @@ std::unique_ptr<webrtc::AudioProcessing> CreateAudioProcessing() {
> 
> std::unique_ptr<webrtc::test::MockAecDump> CreateMockAecDump() {
>   auto mock_aec_dump =
>-      rtc::MakeUnique<testing::StrictMock<webrtc::test::MockAecDump>>();
>+      absl::make_unique<testing::StrictMock<webrtc::test::MockAecDump>>();
>   EXPECT_CALL(*mock_aec_dump.get(), WriteConfig(_)).Times(AtLeast(1));
>-  EXPECT_CALL(*mock_aec_dump.get(), WriteInitMessage(_)).Times(AtLeast(1));
>+  EXPECT_CALL(*mock_aec_dump.get(), WriteInitMessage(_, _)).Times(AtLeast(1));
>   return std::unique_ptr<webrtc::test::MockAecDump>(std::move(mock_aec_dump));
> }
> 
> std::unique_ptr<webrtc::AudioFrame> CreateFakeFrame() {
>-  auto fake_frame = rtc::MakeUnique<webrtc::AudioFrame>();
>+  auto fake_frame = absl::make_unique<webrtc::AudioFrame>();
>   fake_frame->num_channels_ = 1;
>   fake_frame->sample_rate_hz_ = 48000;
>   fake_frame->samples_per_channel_ = 480;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec_dump/aec_dump_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec_dump/aec_dump_unittest.cc
>index 965ac03bd7d..75ed5291fea 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec_dump/aec_dump_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec_dump/aec_dump_unittest.cc
>@@ -12,7 +12,6 @@
> 
> #include "modules/audio_processing/aec_dump/aec_dump_factory.h"
> 
>-#include "modules/include/module_common_types.h"
> #include "rtc_base/task_queue.h"
> #include "test/gtest.h"
> #include "test/testsupport/fileutils.h"
>@@ -40,8 +39,9 @@ TEST(AecDumper, APICallsDoNotCrash) {
>     webrtc::InternalAPMConfig apm_config;
>     aec_dump->WriteConfig(apm_config);
> 
>-    webrtc::InternalAPMStreamsConfig streams_config;
>-    aec_dump->WriteInitMessage(streams_config);
>+    webrtc::ProcessingConfig api_format;
>+    constexpr int64_t kTimeNowMs = 123456789ll;
>+    aec_dump->WriteInitMessage(api_format, kTimeNowMs);
>   }
>   // Remove file after the AecDump d-tor has finished.
>   ASSERT_EQ(0, remove(filename.c_str()));
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec_dump/capture_stream_info.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec_dump/capture_stream_info.cc
>index e3284d88222..dd48fd42108 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec_dump/capture_stream_info.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec_dump/capture_stream_info.cc
>@@ -19,7 +19,7 @@ CaptureStreamInfo::CaptureStreamInfo(std::unique_ptr<WriteToFileTask> task)
> 
> CaptureStreamInfo::~CaptureStreamInfo() = default;
> 
>-void CaptureStreamInfo::AddInput(const FloatAudioFrame& src) {
>+void CaptureStreamInfo::AddInput(const AudioFrameView<const float>& src) {
>   RTC_DCHECK(task_);
>   auto* stream = task_->GetEvent()->mutable_stream();
> 
>@@ -30,7 +30,7 @@ void CaptureStreamInfo::AddInput(const FloatAudioFrame& src) {
>   }
> }
> 
>-void CaptureStreamInfo::AddOutput(const FloatAudioFrame& src) {
>+void CaptureStreamInfo::AddOutput(const AudioFrameView<const float>& src) {
>   RTC_DCHECK(task_);
>   auto* stream = task_->GetEvent()->mutable_stream();
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec_dump/capture_stream_info.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec_dump/capture_stream_info.h
>index 9999c3fbd04..da8fb58895c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec_dump/capture_stream_info.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec_dump/capture_stream_info.h
>@@ -15,9 +15,9 @@
> #include <utility>
> #include <vector>
> 
>+#include "api/audio/audio_frame.h"
> #include "modules/audio_processing/aec_dump/write_to_file_task.h"
> #include "modules/audio_processing/include/aec_dump.h"
>-#include "modules/include/module_common_types.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/ignore_wundef.h"
> #include "rtc_base/logging.h"
>@@ -37,8 +37,8 @@ class CaptureStreamInfo {
>  public:
>   explicit CaptureStreamInfo(std::unique_ptr<WriteToFileTask> task);
>   ~CaptureStreamInfo();
>-  void AddInput(const FloatAudioFrame& src);
>-  void AddOutput(const FloatAudioFrame& src);
>+  void AddInput(const AudioFrameView<const float>& src);
>+  void AddOutput(const AudioFrameView<const float>& src);
> 
>   void AddInput(const AudioFrame& frame);
>   void AddOutput(const AudioFrame& frame);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec_dump/mock_aec_dump.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec_dump/mock_aec_dump.cc
>index aa89e45fe86..fe35d81db9a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec_dump/mock_aec_dump.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec_dump/mock_aec_dump.cc
>@@ -15,5 +15,5 @@ namespace test {
> 
> MockAecDump::MockAecDump() = default;
> MockAecDump::~MockAecDump() = default;
>-}
>-}
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec_dump/mock_aec_dump.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec_dump/mock_aec_dump.h
>index 6df6f2849cb..c01de51e376 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec_dump/mock_aec_dump.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec_dump/mock_aec_dump.h
>@@ -14,7 +14,6 @@
> #include <memory>
> 
> #include "modules/audio_processing/include/aec_dump.h"
>-#include "modules/include/module_common_types.h"
> #include "test/gmock.h"
> 
> namespace webrtc {
>@@ -26,11 +25,13 @@ class MockAecDump : public AecDump {
>   MockAecDump();
>   virtual ~MockAecDump();
> 
>-  MOCK_METHOD1(WriteInitMessage,
>-               void(const InternalAPMStreamsConfig& streams_config));
>+  MOCK_METHOD2(WriteInitMessage,
>+               void(const ProcessingConfig& api_format, int64_t time_now_ms));
> 
>-  MOCK_METHOD1(AddCaptureStreamInput, void(const FloatAudioFrame& src));
>-  MOCK_METHOD1(AddCaptureStreamOutput, void(const FloatAudioFrame& src));
>+  MOCK_METHOD1(AddCaptureStreamInput,
>+               void(const AudioFrameView<const float>& src));
>+  MOCK_METHOD1(AddCaptureStreamOutput,
>+               void(const AudioFrameView<const float>& src));
>   MOCK_METHOD1(AddCaptureStreamInput, void(const AudioFrame& frame));
>   MOCK_METHOD1(AddCaptureStreamOutput, void(const AudioFrame& frame));
>   MOCK_METHOD1(AddAudioProcessingState,
>@@ -38,7 +39,8 @@ class MockAecDump : public AecDump {
>   MOCK_METHOD0(WriteCaptureStreamMessage, void());
> 
>   MOCK_METHOD1(WriteRenderStreamMessage, void(const AudioFrame& frame));
>-  MOCK_METHOD1(WriteRenderStreamMessage, void(const FloatAudioFrame& src));
>+  MOCK_METHOD1(WriteRenderStreamMessage,
>+               void(const AudioFrameView<const float>& src));
> 
>   MOCK_METHOD1(WriteConfig, void(const InternalAPMConfig& config));
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec_dump/write_to_file_task.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec_dump/write_to_file_task.h
>index 73014732472..711afb218ad 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec_dump/write_to_file_task.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aec_dump/write_to_file_task.h
>@@ -19,8 +19,8 @@
> #include "rtc_base/event.h"
> #include "rtc_base/ignore_wundef.h"
> #include "rtc_base/platform_file.h"
>+#include "rtc_base/system/file_wrapper.h"
> #include "rtc_base/task_queue.h"
>-#include "system_wrappers/include/file_wrapper.h"
> 
> // Files generated at build-time by the protobuf compiler.
> RTC_PUSH_IGNORING_WUNDEF()
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aecm/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aecm/BUILD.gn
>new file mode 100644
>index 00000000000..8250173afea
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aecm/BUILD.gn
>@@ -0,0 +1,53 @@
>+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+#
>+# Use of this source code is governed by a BSD-style license
>+# that can be found in the LICENSE file in the root of the source
>+# tree. An additional intellectual property rights grant can be found
>+# in the file PATENTS.  All contributing project authors may
>+# be found in the AUTHORS file in the root of the source tree.
>+
>+import("../../../webrtc.gni")
>+
>+rtc_source_set("aecm_core") {
>+  sources = [
>+    "aecm_core.cc",
>+    "aecm_core.h",
>+    "aecm_defines.h",
>+    "echo_control_mobile.cc",
>+    "echo_control_mobile.h",
>+  ]
>+  deps = [
>+    "../../../common_audio:common_audio_c",
>+    "../../../rtc_base:checks",
>+    "../../../rtc_base:rtc_base_approved",
>+    "../../../rtc_base:sanitizer",
>+    "../../../system_wrappers:cpu_features_api",
>+    "../utility:legacy_delay_estimator",
>+  ]
>+  cflags = []
>+
>+  if (rtc_build_with_neon) {
>+    sources += [ "aecm_core_neon.cc" ]
>+
>+    if (current_cpu != "arm64") {
>+      # Enable compilation for the NEON instruction set.
>+      suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
>+      cflags += [ "-mfpu=neon" ]
>+    }
>+
>+    # Disable LTO on NEON targets due to compiler bug.
>+    # TODO(fdegans): Enable this. See crbug.com/408997.
>+    if (rtc_use_lto) {
>+      cflags -= [
>+        "-flto",
>+        "-ffat-lto-objects",
>+      ]
>+    }
>+  }
>+
>+  if (current_cpu == "mipsel") {
>+    sources += [ "aecm_core_mips.cc" ]
>+  } else {
>+    sources += [ "aecm_core_c.cc" ]
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aecm/aecm_core.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aecm/aecm_core.cc
>index d69dc1ce768..0e56b50c5ef 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aecm/aecm_core.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aecm/aecm_core.cc
>@@ -25,129 +25,99 @@ extern "C" {
> 
> #include "rtc_base/checks.h"
> #include "rtc_base/numerics/safe_conversions.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> #ifdef AEC_DEBUG
>-FILE *dfile;
>-FILE *testfile;
>+FILE* dfile;
>+FILE* testfile;
> #endif
> 
> const int16_t WebRtcAecm_kCosTable[] = {
>-    8192,  8190,  8187,  8180,  8172,  8160,  8147,  8130,  8112,
>-    8091,  8067,  8041,  8012,  7982,  7948,  7912,  7874,  7834,
>-    7791,  7745,  7697,  7647,  7595,  7540,  7483,  7424,  7362,
>-    7299,  7233,  7164,  7094,  7021,  6947,  6870,  6791,  6710,
>-    6627,  6542,  6455,  6366,  6275,  6182,  6087,  5991,  5892,
>-    5792,  5690,  5586,  5481,  5374,  5265,  5155,  5043,  4930,
>-    4815,  4698,  4580,  4461,  4341,  4219,  4096,  3971,  3845,
>-    3719,  3591,  3462,  3331,  3200,  3068,  2935,  2801,  2667,
>-    2531,  2395,  2258,  2120,  1981,  1842,  1703,  1563,  1422,
>-    1281,  1140,   998,   856,   713,   571,   428,   285,   142,
>-       0,  -142,  -285,  -428,  -571,  -713,  -856,  -998, -1140,
>-   -1281, -1422, -1563, -1703, -1842, -1981, -2120, -2258, -2395,
>-   -2531, -2667, -2801, -2935, -3068, -3200, -3331, -3462, -3591,
>-   -3719, -3845, -3971, -4095, -4219, -4341, -4461, -4580, -4698,
>-   -4815, -4930, -5043, -5155, -5265, -5374, -5481, -5586, -5690,
>-   -5792, -5892, -5991, -6087, -6182, -6275, -6366, -6455, -6542,
>-   -6627, -6710, -6791, -6870, -6947, -7021, -7094, -7164, -7233,
>-   -7299, -7362, -7424, -7483, -7540, -7595, -7647, -7697, -7745,
>-   -7791, -7834, -7874, -7912, -7948, -7982, -8012, -8041, -8067,
>-   -8091, -8112, -8130, -8147, -8160, -8172, -8180, -8187, -8190,
>-   -8191, -8190, -8187, -8180, -8172, -8160, -8147, -8130, -8112,
>-   -8091, -8067, -8041, -8012, -7982, -7948, -7912, -7874, -7834,
>-   -7791, -7745, -7697, -7647, -7595, -7540, -7483, -7424, -7362,
>-   -7299, -7233, -7164, -7094, -7021, -6947, -6870, -6791, -6710,
>-   -6627, -6542, -6455, -6366, -6275, -6182, -6087, -5991, -5892,
>-   -5792, -5690, -5586, -5481, -5374, -5265, -5155, -5043, -4930,
>-   -4815, -4698, -4580, -4461, -4341, -4219, -4096, -3971, -3845,
>-   -3719, -3591, -3462, -3331, -3200, -3068, -2935, -2801, -2667,
>-   -2531, -2395, -2258, -2120, -1981, -1842, -1703, -1563, -1422,
>-   -1281, -1140,  -998,  -856,  -713,  -571,  -428,  -285,  -142,
>-       0,   142,   285,   428,   571,   713,   856,   998,  1140,
>-    1281,  1422,  1563,  1703,  1842,  1981,  2120,  2258,  2395,
>-    2531,  2667,  2801,  2935,  3068,  3200,  3331,  3462,  3591,
>-    3719,  3845,  3971,  4095,  4219,  4341,  4461,  4580,  4698,
>-    4815,  4930,  5043,  5155,  5265,  5374,  5481,  5586,  5690,
>-    5792,  5892,  5991,  6087,  6182,  6275,  6366,  6455,  6542,
>-    6627,  6710,  6791,  6870,  6947,  7021,  7094,  7164,  7233,
>-    7299,  7362,  7424,  7483,  7540,  7595,  7647,  7697,  7745,
>-    7791,  7834,  7874,  7912,  7948,  7982,  8012,  8041,  8067,
>-    8091,  8112,  8130,  8147,  8160,  8172,  8180,  8187,  8190
>-};
>+    8192,  8190,  8187,  8180,  8172,  8160,  8147,  8130,  8112,  8091,  8067,
>+    8041,  8012,  7982,  7948,  7912,  7874,  7834,  7791,  7745,  7697,  7647,
>+    7595,  7540,  7483,  7424,  7362,  7299,  7233,  7164,  7094,  7021,  6947,
>+    6870,  6791,  6710,  6627,  6542,  6455,  6366,  6275,  6182,  6087,  5991,
>+    5892,  5792,  5690,  5586,  5481,  5374,  5265,  5155,  5043,  4930,  4815,
>+    4698,  4580,  4461,  4341,  4219,  4096,  3971,  3845,  3719,  3591,  3462,
>+    3331,  3200,  3068,  2935,  2801,  2667,  2531,  2395,  2258,  2120,  1981,
>+    1842,  1703,  1563,  1422,  1281,  1140,  998,   856,   713,   571,   428,
>+    285,   142,   0,     -142,  -285,  -428,  -571,  -713,  -856,  -998,  -1140,
>+    -1281, -1422, -1563, -1703, -1842, -1981, -2120, -2258, -2395, -2531, -2667,
>+    -2801, -2935, -3068, -3200, -3331, -3462, -3591, -3719, -3845, -3971, -4095,
>+    -4219, -4341, -4461, -4580, -4698, -4815, -4930, -5043, -5155, -5265, -5374,
>+    -5481, -5586, -5690, -5792, -5892, -5991, -6087, -6182, -6275, -6366, -6455,
>+    -6542, -6627, -6710, -6791, -6870, -6947, -7021, -7094, -7164, -7233, -7299,
>+    -7362, -7424, -7483, -7540, -7595, -7647, -7697, -7745, -7791, -7834, -7874,
>+    -7912, -7948, -7982, -8012, -8041, -8067, -8091, -8112, -8130, -8147, -8160,
>+    -8172, -8180, -8187, -8190, -8191, -8190, -8187, -8180, -8172, -8160, -8147,
>+    -8130, -8112, -8091, -8067, -8041, -8012, -7982, -7948, -7912, -7874, -7834,
>+    -7791, -7745, -7697, -7647, -7595, -7540, -7483, -7424, -7362, -7299, -7233,
>+    -7164, -7094, -7021, -6947, -6870, -6791, -6710, -6627, -6542, -6455, -6366,
>+    -6275, -6182, -6087, -5991, -5892, -5792, -5690, -5586, -5481, -5374, -5265,
>+    -5155, -5043, -4930, -4815, -4698, -4580, -4461, -4341, -4219, -4096, -3971,
>+    -3845, -3719, -3591, -3462, -3331, -3200, -3068, -2935, -2801, -2667, -2531,
>+    -2395, -2258, -2120, -1981, -1842, -1703, -1563, -1422, -1281, -1140, -998,
>+    -856,  -713,  -571,  -428,  -285,  -142,  0,     142,   285,   428,   571,
>+    713,   856,   998,   1140,  1281,  1422,  1563,  1703,  1842,  1981,  2120,
>+    2258,  2395,  2531,  2667,  2801,  2935,  3068,  3200,  3331,  3462,  3591,
>+    3719,  3845,  3971,  4095,  4219,  4341,  4461,  4580,  4698,  4815,  4930,
>+    5043,  5155,  5265,  5374,  5481,  5586,  5690,  5792,  5892,  5991,  6087,
>+    6182,  6275,  6366,  6455,  6542,  6627,  6710,  6791,  6870,  6947,  7021,
>+    7094,  7164,  7233,  7299,  7362,  7424,  7483,  7540,  7595,  7647,  7697,
>+    7745,  7791,  7834,  7874,  7912,  7948,  7982,  8012,  8041,  8067,  8091,
>+    8112,  8130,  8147,  8160,  8172,  8180,  8187,  8190};
> 
> const int16_t WebRtcAecm_kSinTable[] = {
>-       0,    142,    285,    428,    571,    713,    856,    998,
>-    1140,   1281,   1422,   1563,   1703,   1842,   1981,   2120,
>-    2258,   2395,   2531,   2667,   2801,   2935,   3068,   3200,
>-    3331,   3462,   3591,   3719,   3845,   3971,   4095,   4219,
>-    4341,   4461,   4580,   4698,   4815,   4930,   5043,   5155,
>-    5265,   5374,   5481,   5586,   5690,   5792,   5892,   5991,
>-    6087,   6182,   6275,   6366,   6455,   6542,   6627,   6710,
>-    6791,   6870,   6947,   7021,   7094,   7164,   7233,   7299,
>-    7362,   7424,   7483,   7540,   7595,   7647,   7697,   7745,
>-    7791,   7834,   7874,   7912,   7948,   7982,   8012,   8041,
>-    8067,   8091,   8112,   8130,   8147,   8160,   8172,   8180,
>-    8187,   8190,   8191,   8190,   8187,   8180,   8172,   8160,
>-    8147,   8130,   8112,   8091,   8067,   8041,   8012,   7982,
>-    7948,   7912,   7874,   7834,   7791,   7745,   7697,   7647,
>-    7595,   7540,   7483,   7424,   7362,   7299,   7233,   7164,
>-    7094,   7021,   6947,   6870,   6791,   6710,   6627,   6542,
>-    6455,   6366,   6275,   6182,   6087,   5991,   5892,   5792,
>-    5690,   5586,   5481,   5374,   5265,   5155,   5043,   4930,
>-    4815,   4698,   4580,   4461,   4341,   4219,   4096,   3971,
>-    3845,   3719,   3591,   3462,   3331,   3200,   3068,   2935,
>-    2801,   2667,   2531,   2395,   2258,   2120,   1981,   1842,
>-    1703,   1563,   1422,   1281,   1140,    998,    856,    713,
>-     571,    428,    285,    142,      0,   -142,   -285,   -428,
>-    -571,   -713,   -856,   -998,  -1140,  -1281,  -1422,  -1563,
>-   -1703,  -1842,  -1981,  -2120,  -2258,  -2395,  -2531,  -2667,
>-   -2801,  -2935,  -3068,  -3200,  -3331,  -3462,  -3591,  -3719,
>-   -3845,  -3971,  -4095,  -4219,  -4341,  -4461,  -4580,  -4698,
>-   -4815,  -4930,  -5043,  -5155,  -5265,  -5374,  -5481,  -5586,
>-   -5690,  -5792,  -5892,  -5991,  -6087,  -6182,  -6275,  -6366,
>-   -6455,  -6542,  -6627,  -6710,  -6791,  -6870,  -6947,  -7021,
>-   -7094,  -7164,  -7233,  -7299,  -7362,  -7424,  -7483,  -7540,
>-   -7595,  -7647,  -7697,  -7745,  -7791,  -7834,  -7874,  -7912,
>-   -7948,  -7982,  -8012,  -8041,  -8067,  -8091,  -8112,  -8130,
>-   -8147,  -8160,  -8172,  -8180,  -8187,  -8190,  -8191,  -8190,
>-   -8187,  -8180,  -8172,  -8160,  -8147,  -8130,  -8112,  -8091,
>-   -8067,  -8041,  -8012,  -7982,  -7948,  -7912,  -7874,  -7834,
>-   -7791,  -7745,  -7697,  -7647,  -7595,  -7540,  -7483,  -7424,
>-   -7362,  -7299,  -7233,  -7164,  -7094,  -7021,  -6947,  -6870,
>-   -6791,  -6710,  -6627,  -6542,  -6455,  -6366,  -6275,  -6182,
>-   -6087,  -5991,  -5892,  -5792,  -5690,  -5586,  -5481,  -5374,
>-   -5265,  -5155,  -5043,  -4930,  -4815,  -4698,  -4580,  -4461,
>-   -4341,  -4219,  -4096,  -3971,  -3845,  -3719,  -3591,  -3462,
>-   -3331,  -3200,  -3068,  -2935,  -2801,  -2667,  -2531,  -2395,
>-   -2258,  -2120,  -1981,  -1842,  -1703,  -1563,  -1422,  -1281,
>-   -1140,   -998,   -856,   -713,   -571,   -428,   -285,   -142
>-};
>+    0,     142,   285,   428,   571,   713,   856,   998,   1140,  1281,  1422,
>+    1563,  1703,  1842,  1981,  2120,  2258,  2395,  2531,  2667,  2801,  2935,
>+    3068,  3200,  3331,  3462,  3591,  3719,  3845,  3971,  4095,  4219,  4341,
>+    4461,  4580,  4698,  4815,  4930,  5043,  5155,  5265,  5374,  5481,  5586,
>+    5690,  5792,  5892,  5991,  6087,  6182,  6275,  6366,  6455,  6542,  6627,
>+    6710,  6791,  6870,  6947,  7021,  7094,  7164,  7233,  7299,  7362,  7424,
>+    7483,  7540,  7595,  7647,  7697,  7745,  7791,  7834,  7874,  7912,  7948,
>+    7982,  8012,  8041,  8067,  8091,  8112,  8130,  8147,  8160,  8172,  8180,
>+    8187,  8190,  8191,  8190,  8187,  8180,  8172,  8160,  8147,  8130,  8112,
>+    8091,  8067,  8041,  8012,  7982,  7948,  7912,  7874,  7834,  7791,  7745,
>+    7697,  7647,  7595,  7540,  7483,  7424,  7362,  7299,  7233,  7164,  7094,
>+    7021,  6947,  6870,  6791,  6710,  6627,  6542,  6455,  6366,  6275,  6182,
>+    6087,  5991,  5892,  5792,  5690,  5586,  5481,  5374,  5265,  5155,  5043,
>+    4930,  4815,  4698,  4580,  4461,  4341,  4219,  4096,  3971,  3845,  3719,
>+    3591,  3462,  3331,  3200,  3068,  2935,  2801,  2667,  2531,  2395,  2258,
>+    2120,  1981,  1842,  1703,  1563,  1422,  1281,  1140,  998,   856,   713,
>+    571,   428,   285,   142,   0,     -142,  -285,  -428,  -571,  -713,  -856,
>+    -998,  -1140, -1281, -1422, -1563, -1703, -1842, -1981, -2120, -2258, -2395,
>+    -2531, -2667, -2801, -2935, -3068, -3200, -3331, -3462, -3591, -3719, -3845,
>+    -3971, -4095, -4219, -4341, -4461, -4580, -4698, -4815, -4930, -5043, -5155,
>+    -5265, -5374, -5481, -5586, -5690, -5792, -5892, -5991, -6087, -6182, -6275,
>+    -6366, -6455, -6542, -6627, -6710, -6791, -6870, -6947, -7021, -7094, -7164,
>+    -7233, -7299, -7362, -7424, -7483, -7540, -7595, -7647, -7697, -7745, -7791,
>+    -7834, -7874, -7912, -7948, -7982, -8012, -8041, -8067, -8091, -8112, -8130,
>+    -8147, -8160, -8172, -8180, -8187, -8190, -8191, -8190, -8187, -8180, -8172,
>+    -8160, -8147, -8130, -8112, -8091, -8067, -8041, -8012, -7982, -7948, -7912,
>+    -7874, -7834, -7791, -7745, -7697, -7647, -7595, -7540, -7483, -7424, -7362,
>+    -7299, -7233, -7164, -7094, -7021, -6947, -6870, -6791, -6710, -6627, -6542,
>+    -6455, -6366, -6275, -6182, -6087, -5991, -5892, -5792, -5690, -5586, -5481,
>+    -5374, -5265, -5155, -5043, -4930, -4815, -4698, -4580, -4461, -4341, -4219,
>+    -4096, -3971, -3845, -3719, -3591, -3462, -3331, -3200, -3068, -2935, -2801,
>+    -2667, -2531, -2395, -2258, -2120, -1981, -1842, -1703, -1563, -1422, -1281,
>+    -1140, -998,  -856,  -713,  -571,  -428,  -285,  -142};
> 
> // Initialization table for echo channel in 8 kHz
> static const int16_t kChannelStored8kHz[PART_LEN1] = {
>-    2040,   1815,   1590,   1498,   1405,   1395,   1385,   1418,
>-    1451,   1506,   1562,   1644,   1726,   1804,   1882,   1918,
>-    1953,   1982,   2010,   2025,   2040,   2034,   2027,   2021,
>-    2014,   1997,   1980,   1925,   1869,   1800,   1732,   1683,
>-    1635,   1604,   1572,   1545,   1517,   1481,   1444,   1405,
>-    1367,   1331,   1294,   1270,   1245,   1239,   1233,   1247,
>-    1260,   1282,   1303,   1338,   1373,   1407,   1441,   1470,
>-    1499,   1524,   1549,   1565,   1582,   1601,   1621,   1649,
>-    1676
>-};
>+    2040, 1815, 1590, 1498, 1405, 1395, 1385, 1418, 1451, 1506, 1562,
>+    1644, 1726, 1804, 1882, 1918, 1953, 1982, 2010, 2025, 2040, 2034,
>+    2027, 2021, 2014, 1997, 1980, 1925, 1869, 1800, 1732, 1683, 1635,
>+    1604, 1572, 1545, 1517, 1481, 1444, 1405, 1367, 1331, 1294, 1270,
>+    1245, 1239, 1233, 1247, 1260, 1282, 1303, 1338, 1373, 1407, 1441,
>+    1470, 1499, 1524, 1549, 1565, 1582, 1601, 1621, 1649, 1676};
> 
> // Initialization table for echo channel in 16 kHz
> static const int16_t kChannelStored16kHz[PART_LEN1] = {
>-    2040,   1590,   1405,   1385,   1451,   1562,   1726,   1882,
>-    1953,   2010,   2040,   2027,   2014,   1980,   1869,   1732,
>-    1635,   1572,   1517,   1444,   1367,   1294,   1245,   1233,
>-    1260,   1303,   1373,   1441,   1499,   1549,   1582,   1621,
>-    1676,   1741,   1802,   1861,   1921,   1983,   2040,   2102,
>-    2170,   2265,   2375,   2515,   2651,   2781,   2922,   3075,
>-    3253,   3471,   3738,   3976,   4151,   4258,   4308,   4288,
>-    4270,   4253,   4237,   4179,   4086,   3947,   3757,   3484,
>-    3153
>-};
>+    2040, 1590, 1405, 1385, 1451, 1562, 1726, 1882, 1953, 2010, 2040,
>+    2027, 2014, 1980, 1869, 1732, 1635, 1572, 1517, 1444, 1367, 1294,
>+    1245, 1233, 1260, 1303, 1373, 1441, 1499, 1549, 1582, 1621, 1676,
>+    1741, 1802, 1861, 1921, 1983, 2040, 2102, 2170, 2265, 2375, 2515,
>+    2651, 2781, 2922, 3075, 3253, 3471, 3738, 3976, 4151, 4258, 4308,
>+    4288, 4270, 4253, 4237, 4179, 4086, 3947, 3757, 3484, 3153};
> 
> // Moves the pointer to the next entry and inserts |far_spectrum| and
> // corresponding Q-domain in its buffer.
>@@ -168,8 +138,7 @@ void WebRtcAecm_UpdateFarHistory(AecmCore* self,
>   // Update Q-domain buffer
>   self->far_q_domains[self->far_history_pos] = far_q;
>   // Update far end spectrum buffer
>-  memcpy(&(self->far_history[self->far_history_pos * PART_LEN1]),
>-         far_spectrum,
>+  memcpy(&(self->far_history[self->far_history_pos * PART_LEN1]), far_spectrum,
>          sizeof(uint16_t) * PART_LEN1);
> }
> 
>@@ -213,95 +182,90 @@ StoreAdaptiveChannel WebRtcAecm_StoreAdaptiveChannel;
> ResetAdaptiveChannel WebRtcAecm_ResetAdaptiveChannel;
> 
> AecmCore* WebRtcAecm_CreateCore() {
>-    AecmCore* aecm = static_cast<AecmCore*>(malloc(sizeof(AecmCore)));
>-
>-    aecm->farFrameBuf = WebRtc_CreateBuffer(FRAME_LEN + PART_LEN,
>-                                            sizeof(int16_t));
>-    if (!aecm->farFrameBuf)
>-    {
>-        WebRtcAecm_FreeCore(aecm);
>-        return NULL;
>-    }
>+  AecmCore* aecm = static_cast<AecmCore*>(malloc(sizeof(AecmCore)));
> 
>-    aecm->nearNoisyFrameBuf = WebRtc_CreateBuffer(FRAME_LEN + PART_LEN,
>-                                                  sizeof(int16_t));
>-    if (!aecm->nearNoisyFrameBuf)
>-    {
>-        WebRtcAecm_FreeCore(aecm);
>-        return NULL;
>-    }
>+  aecm->farFrameBuf =
>+      WebRtc_CreateBuffer(FRAME_LEN + PART_LEN, sizeof(int16_t));
>+  if (!aecm->farFrameBuf) {
>+    WebRtcAecm_FreeCore(aecm);
>+    return NULL;
>+  }
> 
>-    aecm->nearCleanFrameBuf = WebRtc_CreateBuffer(FRAME_LEN + PART_LEN,
>-                                                  sizeof(int16_t));
>-    if (!aecm->nearCleanFrameBuf)
>-    {
>-        WebRtcAecm_FreeCore(aecm);
>-        return NULL;
>-    }
>+  aecm->nearNoisyFrameBuf =
>+      WebRtc_CreateBuffer(FRAME_LEN + PART_LEN, sizeof(int16_t));
>+  if (!aecm->nearNoisyFrameBuf) {
>+    WebRtcAecm_FreeCore(aecm);
>+    return NULL;
>+  }
> 
>-    aecm->outFrameBuf = WebRtc_CreateBuffer(FRAME_LEN + PART_LEN,
>-                                            sizeof(int16_t));
>-    if (!aecm->outFrameBuf)
>-    {
>-        WebRtcAecm_FreeCore(aecm);
>-        return NULL;
>-    }
>+  aecm->nearCleanFrameBuf =
>+      WebRtc_CreateBuffer(FRAME_LEN + PART_LEN, sizeof(int16_t));
>+  if (!aecm->nearCleanFrameBuf) {
>+    WebRtcAecm_FreeCore(aecm);
>+    return NULL;
>+  }
> 
>-    aecm->delay_estimator_farend = WebRtc_CreateDelayEstimatorFarend(PART_LEN1,
>-                                                                     MAX_DELAY);
>-    if (aecm->delay_estimator_farend == NULL) {
>-      WebRtcAecm_FreeCore(aecm);
>-      return NULL;
>-    }
>-    aecm->delay_estimator =
>-        WebRtc_CreateDelayEstimator(aecm->delay_estimator_farend, 0);
>-    if (aecm->delay_estimator == NULL) {
>-      WebRtcAecm_FreeCore(aecm);
>-      return NULL;
>-    }
>-    // TODO(bjornv): Explicitly disable robust delay validation until no
>-    // performance regression has been established.  Then remove the line.
>-    WebRtc_enable_robust_validation(aecm->delay_estimator, 0);
>-
>-    aecm->real_fft = WebRtcSpl_CreateRealFFT(PART_LEN_SHIFT);
>-    if (aecm->real_fft == NULL) {
>-      WebRtcAecm_FreeCore(aecm);
>-      return NULL;
>-    }
>+  aecm->outFrameBuf =
>+      WebRtc_CreateBuffer(FRAME_LEN + PART_LEN, sizeof(int16_t));
>+  if (!aecm->outFrameBuf) {
>+    WebRtcAecm_FreeCore(aecm);
>+    return NULL;
>+  }
> 
>-    // Init some aecm pointers. 16 and 32 byte alignment is only necessary
>-    // for Neon code currently.
>-    aecm->xBuf = (int16_t*) (((uintptr_t)aecm->xBuf_buf + 31) & ~ 31);
>-    aecm->dBufClean = (int16_t*) (((uintptr_t)aecm->dBufClean_buf + 31) & ~ 31);
>-    aecm->dBufNoisy = (int16_t*) (((uintptr_t)aecm->dBufNoisy_buf + 31) & ~ 31);
>-    aecm->outBuf = (int16_t*) (((uintptr_t)aecm->outBuf_buf + 15) & ~ 15);
>-    aecm->channelStored = (int16_t*) (((uintptr_t)
>-                                             aecm->channelStored_buf + 15) & ~ 15);
>-    aecm->channelAdapt16 = (int16_t*) (((uintptr_t)
>-                                              aecm->channelAdapt16_buf + 15) & ~ 15);
>-    aecm->channelAdapt32 = (int32_t*) (((uintptr_t)
>-                                              aecm->channelAdapt32_buf + 31) & ~ 31);
>-
>-    return aecm;
>+  aecm->delay_estimator_farend =
>+      WebRtc_CreateDelayEstimatorFarend(PART_LEN1, MAX_DELAY);
>+  if (aecm->delay_estimator_farend == NULL) {
>+    WebRtcAecm_FreeCore(aecm);
>+    return NULL;
>+  }
>+  aecm->delay_estimator =
>+      WebRtc_CreateDelayEstimator(aecm->delay_estimator_farend, 0);
>+  if (aecm->delay_estimator == NULL) {
>+    WebRtcAecm_FreeCore(aecm);
>+    return NULL;
>+  }
>+  // TODO(bjornv): Explicitly disable robust delay validation until no
>+  // performance regression has been established.  Then remove the line.
>+  WebRtc_enable_robust_validation(aecm->delay_estimator, 0);
>+
>+  aecm->real_fft = WebRtcSpl_CreateRealFFT(PART_LEN_SHIFT);
>+  if (aecm->real_fft == NULL) {
>+    WebRtcAecm_FreeCore(aecm);
>+    return NULL;
>+  }
>+
>+  // Init some aecm pointers. 16 and 32 byte alignment is only necessary
>+  // for Neon code currently.
>+  aecm->xBuf = (int16_t*)(((uintptr_t)aecm->xBuf_buf + 31) & ~31);
>+  aecm->dBufClean = (int16_t*)(((uintptr_t)aecm->dBufClean_buf + 31) & ~31);
>+  aecm->dBufNoisy = (int16_t*)(((uintptr_t)aecm->dBufNoisy_buf + 31) & ~31);
>+  aecm->outBuf = (int16_t*)(((uintptr_t)aecm->outBuf_buf + 15) & ~15);
>+  aecm->channelStored =
>+      (int16_t*)(((uintptr_t)aecm->channelStored_buf + 15) & ~15);
>+  aecm->channelAdapt16 =
>+      (int16_t*)(((uintptr_t)aecm->channelAdapt16_buf + 15) & ~15);
>+  aecm->channelAdapt32 =
>+      (int32_t*)(((uintptr_t)aecm->channelAdapt32_buf + 31) & ~31);
>+
>+  return aecm;
> }
> 
> void WebRtcAecm_InitEchoPathCore(AecmCore* aecm, const int16_t* echo_path) {
>-    int i = 0;
>-
>-    // Reset the stored channel
>-    memcpy(aecm->channelStored, echo_path, sizeof(int16_t) * PART_LEN1);
>-    // Reset the adapted channels
>-    memcpy(aecm->channelAdapt16, echo_path, sizeof(int16_t) * PART_LEN1);
>-    for (i = 0; i < PART_LEN1; i++)
>-    {
>-        aecm->channelAdapt32[i] = (int32_t)aecm->channelAdapt16[i] << 16;
>-    }
>+  int i = 0;
>+
>+  // Reset the stored channel
>+  memcpy(aecm->channelStored, echo_path, sizeof(int16_t) * PART_LEN1);
>+  // Reset the adapted channels
>+  memcpy(aecm->channelAdapt16, echo_path, sizeof(int16_t) * PART_LEN1);
>+  for (i = 0; i < PART_LEN1; i++) {
>+    aecm->channelAdapt32[i] = (int32_t)aecm->channelAdapt16[i] << 16;
>+  }
> 
>-    // Reset channel storing variables
>-    aecm->mseAdaptOld = 1000;
>-    aecm->mseStoredOld = 1000;
>-    aecm->mseThreshold = WEBRTC_SPL_WORD32_MAX;
>-    aecm->mseChannelCount = 0;
>+  // Reset channel storing variables
>+  aecm->mseAdaptOld = 1000;
>+  aecm->mseStoredOld = 1000;
>+  aecm->mseThreshold = WEBRTC_SPL_WORD32_MAX;
>+  aecm->mseChannelCount = 0;
> }
> 
> static void CalcLinearEnergiesC(AecmCore* aecm,
>@@ -310,65 +274,61 @@ static void CalcLinearEnergiesC(AecmCore* aecm,
>                                 uint32_t* far_energy,
>                                 uint32_t* echo_energy_adapt,
>                                 uint32_t* echo_energy_stored) {
>-    int i;
>-
>-    // Get energy for the delayed far end signal and estimated
>-    // echo using both stored and adapted channels.
>-    for (i = 0; i < PART_LEN1; i++)
>-    {
>-        echo_est[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i],
>-                                           far_spectrum[i]);
>-        (*far_energy) += (uint32_t)(far_spectrum[i]);
>-        *echo_energy_adapt += aecm->channelAdapt16[i] * far_spectrum[i];
>-        (*echo_energy_stored) += (uint32_t)echo_est[i];
>-    }
>+  int i;
>+
>+  // Get energy for the delayed far end signal and estimated
>+  // echo using both stored and adapted channels.
>+  for (i = 0; i < PART_LEN1; i++) {
>+    echo_est[i] =
>+        WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i], far_spectrum[i]);
>+    (*far_energy) += (uint32_t)(far_spectrum[i]);
>+    *echo_energy_adapt += aecm->channelAdapt16[i] * far_spectrum[i];
>+    (*echo_energy_stored) += (uint32_t)echo_est[i];
>+  }
> }
> 
> static void StoreAdaptiveChannelC(AecmCore* aecm,
>                                   const uint16_t* far_spectrum,
>                                   int32_t* echo_est) {
>-    int i;
>-
>-    // During startup we store the channel every block.
>-    memcpy(aecm->channelStored, aecm->channelAdapt16, sizeof(int16_t) * PART_LEN1);
>-    // Recalculate echo estimate
>-    for (i = 0; i < PART_LEN; i += 4)
>-    {
>-        echo_est[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i],
>-                                           far_spectrum[i]);
>-        echo_est[i + 1] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i + 1],
>-                                           far_spectrum[i + 1]);
>-        echo_est[i + 2] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i + 2],
>-                                           far_spectrum[i + 2]);
>-        echo_est[i + 3] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i + 3],
>-                                           far_spectrum[i + 3]);
>-    }
>-    echo_est[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i],
>-                                       far_spectrum[i]);
>+  int i;
>+
>+  // During startup we store the channel every block.
>+  memcpy(aecm->channelStored, aecm->channelAdapt16,
>+         sizeof(int16_t) * PART_LEN1);
>+  // Recalculate echo estimate
>+  for (i = 0; i < PART_LEN; i += 4) {
>+    echo_est[i] =
>+        WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i], far_spectrum[i]);
>+    echo_est[i + 1] =
>+        WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i + 1], far_spectrum[i + 1]);
>+    echo_est[i + 2] =
>+        WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i + 2], far_spectrum[i + 2]);
>+    echo_est[i + 3] =
>+        WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i + 3], far_spectrum[i + 3]);
>+  }
>+  echo_est[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i], far_spectrum[i]);
> }
> 
> static void ResetAdaptiveChannelC(AecmCore* aecm) {
>-    int i;
>-
>-    // The stored channel has a significantly lower MSE than the adaptive one for
>-    // two consecutive calculations. Reset the adaptive channel.
>-    memcpy(aecm->channelAdapt16, aecm->channelStored,
>-           sizeof(int16_t) * PART_LEN1);
>-    // Restore the W32 channel
>-    for (i = 0; i < PART_LEN; i += 4)
>-    {
>-        aecm->channelAdapt32[i] = (int32_t)aecm->channelStored[i] << 16;
>-        aecm->channelAdapt32[i + 1] = (int32_t)aecm->channelStored[i + 1] << 16;
>-        aecm->channelAdapt32[i + 2] = (int32_t)aecm->channelStored[i + 2] << 16;
>-        aecm->channelAdapt32[i + 3] = (int32_t)aecm->channelStored[i + 3] << 16;
>-    }
>+  int i;
>+
>+  // The stored channel has a significantly lower MSE than the adaptive one for
>+  // two consecutive calculations. Reset the adaptive channel.
>+  memcpy(aecm->channelAdapt16, aecm->channelStored,
>+         sizeof(int16_t) * PART_LEN1);
>+  // Restore the W32 channel
>+  for (i = 0; i < PART_LEN; i += 4) {
>     aecm->channelAdapt32[i] = (int32_t)aecm->channelStored[i] << 16;
>+    aecm->channelAdapt32[i + 1] = (int32_t)aecm->channelStored[i + 1] << 16;
>+    aecm->channelAdapt32[i + 2] = (int32_t)aecm->channelStored[i + 2] << 16;
>+    aecm->channelAdapt32[i + 3] = (int32_t)aecm->channelStored[i + 3] << 16;
>+  }
>+  aecm->channelAdapt32[i] = (int32_t)aecm->channelStored[i] << 16;
> }
> 
> // Initialize function pointers for ARM Neon platform.
> #if defined(WEBRTC_HAS_NEON)
>-static void WebRtcAecm_InitNeon(void)
>-{
>+static void WebRtcAecm_InitNeon(void) {
>   WebRtcAecm_StoreAdaptiveChannel = WebRtcAecm_StoreAdaptiveChannelNeon;
>   WebRtcAecm_ResetAdaptiveChannel = WebRtcAecm_ResetAdaptiveChannelNeon;
>   WebRtcAecm_CalcLinearEnergies = WebRtcAecm_CalcLinearEnergiesNeon;
>@@ -377,8 +337,7 @@ static void WebRtcAecm_InitNeon(void)
> 
> // Initialize function pointers for MIPS platform.
> #if defined(MIPS32_LE)
>-static void WebRtcAecm_InitMips(void)
>-{
>+static void WebRtcAecm_InitMips(void) {
> #if defined(MIPS_DSP_R1_LE)
>   WebRtcAecm_StoreAdaptiveChannel = WebRtcAecm_StoreAdaptiveChannel_mips;
>   WebRtcAecm_ResetAdaptiveChannel = WebRtcAecm_ResetAdaptiveChannel_mips;
>@@ -389,8 +348,8 @@ static void WebRtcAecm_InitMips(void)
> 
> // WebRtcAecm_InitCore(...)
> //
>-// This function initializes the AECM instant created with WebRtcAecm_CreateCore(...)
>-// Input:
>+// This function initializes the AECM instant created with
>+// WebRtcAecm_CreateCore(...) Input:
> //      - aecm            : Pointer to the Echo Suppression instance
> //      - samplingFreq   : Sampling Frequency
> //
>@@ -401,152 +360,147 @@ static void WebRtcAecm_InitMips(void)
> //                        -1 - Error
> //
> int WebRtcAecm_InitCore(AecmCore* const aecm, int samplingFreq) {
>-    int i = 0;
>-    int32_t tmp32 = PART_LEN1 * PART_LEN1;
>-    int16_t tmp16 = PART_LEN1;
>-
>-    if (samplingFreq != 8000 && samplingFreq != 16000)
>-    {
>-        samplingFreq = 8000;
>-        return -1;
>-    }
>-    // sanity check of sampling frequency
>-    aecm->mult = (int16_t)samplingFreq / 8000;
>+  int i = 0;
>+  int32_t tmp32 = PART_LEN1 * PART_LEN1;
>+  int16_t tmp16 = PART_LEN1;
> 
>-    aecm->farBufWritePos = 0;
>-    aecm->farBufReadPos = 0;
>-    aecm->knownDelay = 0;
>-    aecm->lastKnownDelay = 0;
>-
>-    WebRtc_InitBuffer(aecm->farFrameBuf);
>-    WebRtc_InitBuffer(aecm->nearNoisyFrameBuf);
>-    WebRtc_InitBuffer(aecm->nearCleanFrameBuf);
>-    WebRtc_InitBuffer(aecm->outFrameBuf);
>+  if (samplingFreq != 8000 && samplingFreq != 16000) {
>+    samplingFreq = 8000;
>+    return -1;
>+  }
>+  // sanity check of sampling frequency
>+  aecm->mult = (int16_t)samplingFreq / 8000;
> 
>-    memset(aecm->xBuf_buf, 0, sizeof(aecm->xBuf_buf));
>-    memset(aecm->dBufClean_buf, 0, sizeof(aecm->dBufClean_buf));
>-    memset(aecm->dBufNoisy_buf, 0, sizeof(aecm->dBufNoisy_buf));
>-    memset(aecm->outBuf_buf, 0, sizeof(aecm->outBuf_buf));
>+  aecm->farBufWritePos = 0;
>+  aecm->farBufReadPos = 0;
>+  aecm->knownDelay = 0;
>+  aecm->lastKnownDelay = 0;
> 
>-    aecm->seed = 666;
>-    aecm->totCount = 0;
>+  WebRtc_InitBuffer(aecm->farFrameBuf);
>+  WebRtc_InitBuffer(aecm->nearNoisyFrameBuf);
>+  WebRtc_InitBuffer(aecm->nearCleanFrameBuf);
>+  WebRtc_InitBuffer(aecm->outFrameBuf);
> 
>-    if (WebRtc_InitDelayEstimatorFarend(aecm->delay_estimator_farend) != 0) {
>-      return -1;
>-    }
>-    if (WebRtc_InitDelayEstimator(aecm->delay_estimator) != 0) {
>-      return -1;
>-    }
>-    // Set far end histories to zero
>-    memset(aecm->far_history, 0, sizeof(uint16_t) * PART_LEN1 * MAX_DELAY);
>-    memset(aecm->far_q_domains, 0, sizeof(int) * MAX_DELAY);
>-    aecm->far_history_pos = MAX_DELAY;
>-
>-    aecm->nlpFlag = 1;
>-    aecm->fixedDelay = -1;
>-
>-    aecm->dfaCleanQDomain = 0;
>-    aecm->dfaCleanQDomainOld = 0;
>-    aecm->dfaNoisyQDomain = 0;
>-    aecm->dfaNoisyQDomainOld = 0;
>-
>-    memset(aecm->nearLogEnergy, 0, sizeof(aecm->nearLogEnergy));
>-    aecm->farLogEnergy = 0;
>-    memset(aecm->echoAdaptLogEnergy, 0, sizeof(aecm->echoAdaptLogEnergy));
>-    memset(aecm->echoStoredLogEnergy, 0, sizeof(aecm->echoStoredLogEnergy));
>-
>-    // Initialize the echo channels with a stored shape.
>-    if (samplingFreq == 8000)
>-    {
>-        WebRtcAecm_InitEchoPathCore(aecm, kChannelStored8kHz);
>-    }
>-    else
>-    {
>-        WebRtcAecm_InitEchoPathCore(aecm, kChannelStored16kHz);
>-    }
>+  memset(aecm->xBuf_buf, 0, sizeof(aecm->xBuf_buf));
>+  memset(aecm->dBufClean_buf, 0, sizeof(aecm->dBufClean_buf));
>+  memset(aecm->dBufNoisy_buf, 0, sizeof(aecm->dBufNoisy_buf));
>+  memset(aecm->outBuf_buf, 0, sizeof(aecm->outBuf_buf));
> 
>-    memset(aecm->echoFilt, 0, sizeof(aecm->echoFilt));
>-    memset(aecm->nearFilt, 0, sizeof(aecm->nearFilt));
>-    aecm->noiseEstCtr = 0;
>+  aecm->seed = 666;
>+  aecm->totCount = 0;
> 
>-    aecm->cngMode = AecmTrue;
>-
>-    memset(aecm->noiseEstTooLowCtr, 0, sizeof(aecm->noiseEstTooLowCtr));
>-    memset(aecm->noiseEstTooHighCtr, 0, sizeof(aecm->noiseEstTooHighCtr));
>-    // Shape the initial noise level to an approximate pink noise.
>-    for (i = 0; i < (PART_LEN1 >> 1) - 1; i++)
>-    {
>-        aecm->noiseEst[i] = (tmp32 << 8);
>-        tmp16--;
>-        tmp32 -= (int32_t)((tmp16 << 1) + 1);
>-    }
>-    for (; i < PART_LEN1; i++)
>-    {
>-        aecm->noiseEst[i] = (tmp32 << 8);
>-    }
>-
>-    aecm->farEnergyMin = WEBRTC_SPL_WORD16_MAX;
>-    aecm->farEnergyMax = WEBRTC_SPL_WORD16_MIN;
>-    aecm->farEnergyMaxMin = 0;
>-    aecm->farEnergyVAD = FAR_ENERGY_MIN; // This prevents false speech detection at the
>-                                         // beginning.
>-    aecm->farEnergyMSE = 0;
>-    aecm->currentVADValue = 0;
>-    aecm->vadUpdateCount = 0;
>-    aecm->firstVAD = 1;
>+  if (WebRtc_InitDelayEstimatorFarend(aecm->delay_estimator_farend) != 0) {
>+    return -1;
>+  }
>+  if (WebRtc_InitDelayEstimator(aecm->delay_estimator) != 0) {
>+    return -1;
>+  }
>+  // Set far end histories to zero
>+  memset(aecm->far_history, 0, sizeof(uint16_t) * PART_LEN1 * MAX_DELAY);
>+  memset(aecm->far_q_domains, 0, sizeof(int) * MAX_DELAY);
>+  aecm->far_history_pos = MAX_DELAY;
>+
>+  aecm->nlpFlag = 1;
>+  aecm->fixedDelay = -1;
>+
>+  aecm->dfaCleanQDomain = 0;
>+  aecm->dfaCleanQDomainOld = 0;
>+  aecm->dfaNoisyQDomain = 0;
>+  aecm->dfaNoisyQDomainOld = 0;
>+
>+  memset(aecm->nearLogEnergy, 0, sizeof(aecm->nearLogEnergy));
>+  aecm->farLogEnergy = 0;
>+  memset(aecm->echoAdaptLogEnergy, 0, sizeof(aecm->echoAdaptLogEnergy));
>+  memset(aecm->echoStoredLogEnergy, 0, sizeof(aecm->echoStoredLogEnergy));
>+
>+  // Initialize the echo channels with a stored shape.
>+  if (samplingFreq == 8000) {
>+    WebRtcAecm_InitEchoPathCore(aecm, kChannelStored8kHz);
>+  } else {
>+    WebRtcAecm_InitEchoPathCore(aecm, kChannelStored16kHz);
>+  }
> 
>-    aecm->startupState = 0;
>-    aecm->supGain = SUPGAIN_DEFAULT;
>-    aecm->supGainOld = SUPGAIN_DEFAULT;
>+  memset(aecm->echoFilt, 0, sizeof(aecm->echoFilt));
>+  memset(aecm->nearFilt, 0, sizeof(aecm->nearFilt));
>+  aecm->noiseEstCtr = 0;
> 
>-    aecm->supGainErrParamA = SUPGAIN_ERROR_PARAM_A;
>-    aecm->supGainErrParamD = SUPGAIN_ERROR_PARAM_D;
>-    aecm->supGainErrParamDiffAB = SUPGAIN_ERROR_PARAM_A - SUPGAIN_ERROR_PARAM_B;
>-    aecm->supGainErrParamDiffBD = SUPGAIN_ERROR_PARAM_B - SUPGAIN_ERROR_PARAM_D;
>+  aecm->cngMode = AecmTrue;
> 
>-    // Assert a preprocessor definition at compile-time. It's an assumption
>-    // used in assembly code, so check the assembly files before any change.
>-    static_assert(PART_LEN % 16 == 0, "PART_LEN is not a multiple of 16");
>+  memset(aecm->noiseEstTooLowCtr, 0, sizeof(aecm->noiseEstTooLowCtr));
>+  memset(aecm->noiseEstTooHighCtr, 0, sizeof(aecm->noiseEstTooHighCtr));
>+  // Shape the initial noise level to an approximate pink noise.
>+  for (i = 0; i < (PART_LEN1 >> 1) - 1; i++) {
>+    aecm->noiseEst[i] = (tmp32 << 8);
>+    tmp16--;
>+    tmp32 -= (int32_t)((tmp16 << 1) + 1);
>+  }
>+  for (; i < PART_LEN1; i++) {
>+    aecm->noiseEst[i] = (tmp32 << 8);
>+  }
> 
>-    // Initialize function pointers.
>-    WebRtcAecm_CalcLinearEnergies = CalcLinearEnergiesC;
>-    WebRtcAecm_StoreAdaptiveChannel = StoreAdaptiveChannelC;
>-    WebRtcAecm_ResetAdaptiveChannel = ResetAdaptiveChannelC;
>+  aecm->farEnergyMin = WEBRTC_SPL_WORD16_MAX;
>+  aecm->farEnergyMax = WEBRTC_SPL_WORD16_MIN;
>+  aecm->farEnergyMaxMin = 0;
>+  aecm->farEnergyVAD =
>+      FAR_ENERGY_MIN;  // This prevents false speech detection at the
>+                       // beginning.
>+  aecm->farEnergyMSE = 0;
>+  aecm->currentVADValue = 0;
>+  aecm->vadUpdateCount = 0;
>+  aecm->firstVAD = 1;
>+
>+  aecm->startupState = 0;
>+  aecm->supGain = SUPGAIN_DEFAULT;
>+  aecm->supGainOld = SUPGAIN_DEFAULT;
>+
>+  aecm->supGainErrParamA = SUPGAIN_ERROR_PARAM_A;
>+  aecm->supGainErrParamD = SUPGAIN_ERROR_PARAM_D;
>+  aecm->supGainErrParamDiffAB = SUPGAIN_ERROR_PARAM_A - SUPGAIN_ERROR_PARAM_B;
>+  aecm->supGainErrParamDiffBD = SUPGAIN_ERROR_PARAM_B - SUPGAIN_ERROR_PARAM_D;
>+
>+  // Assert a preprocessor definition at compile-time. It's an assumption
>+  // used in assembly code, so check the assembly files before any change.
>+  static_assert(PART_LEN % 16 == 0, "PART_LEN is not a multiple of 16");
>+
>+  // Initialize function pointers.
>+  WebRtcAecm_CalcLinearEnergies = CalcLinearEnergiesC;
>+  WebRtcAecm_StoreAdaptiveChannel = StoreAdaptiveChannelC;
>+  WebRtcAecm_ResetAdaptiveChannel = ResetAdaptiveChannelC;
> 
> #if defined(WEBRTC_HAS_NEON)
>-    WebRtcAecm_InitNeon();
>+  WebRtcAecm_InitNeon();
> #endif
> 
> #if defined(MIPS32_LE)
>-    WebRtcAecm_InitMips();
>+  WebRtcAecm_InitMips();
> #endif
>-    return 0;
>+  return 0;
> }
> 
> // TODO(bjornv): This function is currently not used. Add support for these
> // parameters from a higher level
> int WebRtcAecm_Control(AecmCore* aecm, int delay, int nlpFlag) {
>-    aecm->nlpFlag = nlpFlag;
>-    aecm->fixedDelay = delay;
>+  aecm->nlpFlag = nlpFlag;
>+  aecm->fixedDelay = delay;
> 
>-    return 0;
>+  return 0;
> }
> 
> void WebRtcAecm_FreeCore(AecmCore* aecm) {
>-    if (aecm == NULL) {
>-      return;
>-    }
>+  if (aecm == NULL) {
>+    return;
>+  }
> 
>-    WebRtc_FreeBuffer(aecm->farFrameBuf);
>-    WebRtc_FreeBuffer(aecm->nearNoisyFrameBuf);
>-    WebRtc_FreeBuffer(aecm->nearCleanFrameBuf);
>-    WebRtc_FreeBuffer(aecm->outFrameBuf);
>+  WebRtc_FreeBuffer(aecm->farFrameBuf);
>+  WebRtc_FreeBuffer(aecm->nearNoisyFrameBuf);
>+  WebRtc_FreeBuffer(aecm->nearCleanFrameBuf);
>+  WebRtc_FreeBuffer(aecm->outFrameBuf);
> 
>-    WebRtc_FreeDelayEstimator(aecm->delay_estimator);
>-    WebRtc_FreeDelayEstimatorFarend(aecm->delay_estimator_farend);
>-    WebRtcSpl_FreeRealFFT(aecm->real_fft);
>+  WebRtc_FreeDelayEstimator(aecm->delay_estimator);
>+  WebRtc_FreeDelayEstimatorFarend(aecm->delay_estimator_farend);
>+  WebRtcSpl_FreeRealFFT(aecm->real_fft);
> 
>-    free(aecm);
>+  free(aecm);
> }
> 
> int WebRtcAecm_ProcessFrame(AecmCore* aecm,
>@@ -554,89 +508,72 @@ int WebRtcAecm_ProcessFrame(AecmCore* aecm,
>                             const int16_t* nearendNoisy,
>                             const int16_t* nearendClean,
>                             int16_t* out) {
>-    int16_t outBlock_buf[PART_LEN + 8]; // Align buffer to 8-byte boundary.
>-    int16_t* outBlock = (int16_t*) (((uintptr_t) outBlock_buf + 15) & ~ 15);
>-
>-    int16_t farFrame[FRAME_LEN];
>-    const int16_t* out_ptr = NULL;
>-    int size = 0;
>-
>-    // Buffer the current frame.
>-    // Fetch an older one corresponding to the delay.
>-    WebRtcAecm_BufferFarFrame(aecm, farend, FRAME_LEN);
>-    WebRtcAecm_FetchFarFrame(aecm, farFrame, FRAME_LEN, aecm->knownDelay);
>-
>-    // Buffer the synchronized far and near frames,
>-    // to pass the smaller blocks individually.
>-    WebRtc_WriteBuffer(aecm->farFrameBuf, farFrame, FRAME_LEN);
>-    WebRtc_WriteBuffer(aecm->nearNoisyFrameBuf, nearendNoisy, FRAME_LEN);
>-    if (nearendClean != NULL)
>-    {
>-        WebRtc_WriteBuffer(aecm->nearCleanFrameBuf, nearendClean, FRAME_LEN);
>-    }
>-
>-    // Process as many blocks as possible.
>-    while (WebRtc_available_read(aecm->farFrameBuf) >= PART_LEN)
>-    {
>-        int16_t far_block[PART_LEN];
>-        const int16_t* far_block_ptr = NULL;
>-        int16_t near_noisy_block[PART_LEN];
>-        const int16_t* near_noisy_block_ptr = NULL;
>-
>-        WebRtc_ReadBuffer(aecm->farFrameBuf, (void**) &far_block_ptr, far_block,
>-                          PART_LEN);
>-        WebRtc_ReadBuffer(aecm->nearNoisyFrameBuf,
>-                          (void**) &near_noisy_block_ptr,
>-                          near_noisy_block,
>-                          PART_LEN);
>-        if (nearendClean != NULL)
>-        {
>-            int16_t near_clean_block[PART_LEN];
>-            const int16_t* near_clean_block_ptr = NULL;
>-
>-            WebRtc_ReadBuffer(aecm->nearCleanFrameBuf,
>-                              (void**) &near_clean_block_ptr,
>-                              near_clean_block,
>-                              PART_LEN);
>-            if (WebRtcAecm_ProcessBlock(aecm,
>-                                        far_block_ptr,
>-                                        near_noisy_block_ptr,
>-                                        near_clean_block_ptr,
>-                                        outBlock) == -1)
>-            {
>-                return -1;
>-            }
>-        } else
>-        {
>-            if (WebRtcAecm_ProcessBlock(aecm,
>-                                        far_block_ptr,
>-                                        near_noisy_block_ptr,
>-                                        NULL,
>-                                        outBlock) == -1)
>-            {
>-                return -1;
>-            }
>-        }
>+  int16_t outBlock_buf[PART_LEN + 8];  // Align buffer to 8-byte boundary.
>+  int16_t* outBlock = (int16_t*)(((uintptr_t)outBlock_buf + 15) & ~15);
>+
>+  int16_t farFrame[FRAME_LEN];
>+  const int16_t* out_ptr = NULL;
>+  int size = 0;
>+
>+  // Buffer the current frame.
>+  // Fetch an older one corresponding to the delay.
>+  WebRtcAecm_BufferFarFrame(aecm, farend, FRAME_LEN);
>+  WebRtcAecm_FetchFarFrame(aecm, farFrame, FRAME_LEN, aecm->knownDelay);
>+
>+  // Buffer the synchronized far and near frames,
>+  // to pass the smaller blocks individually.
>+  WebRtc_WriteBuffer(aecm->farFrameBuf, farFrame, FRAME_LEN);
>+  WebRtc_WriteBuffer(aecm->nearNoisyFrameBuf, nearendNoisy, FRAME_LEN);
>+  if (nearendClean != NULL) {
>+    WebRtc_WriteBuffer(aecm->nearCleanFrameBuf, nearendClean, FRAME_LEN);
>+  }
> 
>-        WebRtc_WriteBuffer(aecm->outFrameBuf, outBlock, PART_LEN);
>+  // Process as many blocks as possible.
>+  while (WebRtc_available_read(aecm->farFrameBuf) >= PART_LEN) {
>+    int16_t far_block[PART_LEN];
>+    const int16_t* far_block_ptr = NULL;
>+    int16_t near_noisy_block[PART_LEN];
>+    const int16_t* near_noisy_block_ptr = NULL;
>+
>+    WebRtc_ReadBuffer(aecm->farFrameBuf, (void**)&far_block_ptr, far_block,
>+                      PART_LEN);
>+    WebRtc_ReadBuffer(aecm->nearNoisyFrameBuf, (void**)&near_noisy_block_ptr,
>+                      near_noisy_block, PART_LEN);
>+    if (nearendClean != NULL) {
>+      int16_t near_clean_block[PART_LEN];
>+      const int16_t* near_clean_block_ptr = NULL;
>+
>+      WebRtc_ReadBuffer(aecm->nearCleanFrameBuf, (void**)&near_clean_block_ptr,
>+                        near_clean_block, PART_LEN);
>+      if (WebRtcAecm_ProcessBlock(aecm, far_block_ptr, near_noisy_block_ptr,
>+                                  near_clean_block_ptr, outBlock) == -1) {
>+        return -1;
>+      }
>+    } else {
>+      if (WebRtcAecm_ProcessBlock(aecm, far_block_ptr, near_noisy_block_ptr,
>+                                  NULL, outBlock) == -1) {
>+        return -1;
>+      }
>     }
> 
>-    // Stuff the out buffer if we have less than a frame to output.
>-    // This should only happen for the first frame.
>-    size = (int) WebRtc_available_read(aecm->outFrameBuf);
>-    if (size < FRAME_LEN)
>-    {
>-        WebRtc_MoveReadPtr(aecm->outFrameBuf, size - FRAME_LEN);
>-    }
>+    WebRtc_WriteBuffer(aecm->outFrameBuf, outBlock, PART_LEN);
>+  }
> 
>-    // Obtain an output frame.
>-    WebRtc_ReadBuffer(aecm->outFrameBuf, (void**) &out_ptr, out, FRAME_LEN);
>-    if (out_ptr != out) {
>-      // ReadBuffer() hasn't copied to |out| in this case.
>-      memcpy(out, out_ptr, FRAME_LEN * sizeof(int16_t));
>-    }
>+  // Stuff the out buffer if we have less than a frame to output.
>+  // This should only happen for the first frame.
>+  size = (int)WebRtc_available_read(aecm->outFrameBuf);
>+  if (size < FRAME_LEN) {
>+    WebRtc_MoveReadPtr(aecm->outFrameBuf, size - FRAME_LEN);
>+  }
> 
>-    return 0;
>+  // Obtain an output frame.
>+  WebRtc_ReadBuffer(aecm->outFrameBuf, (void**)&out_ptr, out, FRAME_LEN);
>+  if (out_ptr != out) {
>+    // ReadBuffer() hasn't copied to |out| in this case.
>+    memcpy(out, out_ptr, FRAME_LEN * sizeof(int16_t));
>+  }
>+
>+  return 0;
> }
> 
> // WebRtcAecm_AsymFilt(...)
>@@ -653,26 +590,23 @@ int WebRtcAecm_ProcessFrame(AecmCore* aecm,
> //
> // Return: - Filtered value.
> //
>-int16_t WebRtcAecm_AsymFilt(const int16_t filtOld, const int16_t inVal,
>+int16_t WebRtcAecm_AsymFilt(const int16_t filtOld,
>+                            const int16_t inVal,
>                             const int16_t stepSizePos,
>-                            const int16_t stepSizeNeg)
>-{
>-    int16_t retVal;
>+                            const int16_t stepSizeNeg) {
>+  int16_t retVal;
> 
>-    if ((filtOld == WEBRTC_SPL_WORD16_MAX) | (filtOld == WEBRTC_SPL_WORD16_MIN))
>-    {
>-        return inVal;
>-    }
>-    retVal = filtOld;
>-    if (filtOld > inVal)
>-    {
>-        retVal -= (filtOld - inVal) >> stepSizeNeg;
>-    } else
>-    {
>-        retVal += (inVal - filtOld) >> stepSizePos;
>-    }
>+  if ((filtOld == WEBRTC_SPL_WORD16_MAX) | (filtOld == WEBRTC_SPL_WORD16_MIN)) {
>+    return inVal;
>+  }
>+  retVal = filtOld;
>+  if (filtOld > inVal) {
>+    retVal -= (filtOld - inVal) >> stepSizeNeg;
>+  } else {
>+    retVal += (inVal - filtOld) >> stepSizePos;
>+  }
> 
>-    return retVal;
>+  return retVal;
> }
> 
> // ExtractFractionPart(a, zeros)
>@@ -700,8 +634,9 @@ static int16_t LogOfEnergyInQ8(uint32_t energy, int q_domain) {
> 
> // WebRtcAecm_CalcEnergies(...)
> //
>-// This function calculates the log of energies for nearend, farend and estimated
>-// echoes. There is also an update of energy decision levels, i.e. internal VAD.
>+// This function calculates the log of energies for nearend, farend and
>+// estimated echoes. There is also an update of energy decision levels, i.e.
>+// internal VAD.
> //
> //
> // @param  aecm         [i/o]   Handle of the AECM instance.
>@@ -716,123 +651,112 @@ void WebRtcAecm_CalcEnergies(AecmCore* aecm,
>                              const int16_t far_q,
>                              const uint32_t nearEner,
>                              int32_t* echoEst) {
>-    // Local variables
>-    uint32_t tmpAdapt = 0;
>-    uint32_t tmpStored = 0;
>-    uint32_t tmpFar = 0;
>-
>-    int i;
>-
>-    int16_t tmp16;
>-    int16_t increase_max_shifts = 4;
>-    int16_t decrease_max_shifts = 11;
>-    int16_t increase_min_shifts = 11;
>-    int16_t decrease_min_shifts = 3;
>-
>-    // Get log of near end energy and store in buffer
>-
>-    // Shift buffer
>-    memmove(aecm->nearLogEnergy + 1, aecm->nearLogEnergy,
>-            sizeof(int16_t) * (MAX_BUF_LEN - 1));
>-
>-    // Logarithm of integrated magnitude spectrum (nearEner)
>-    aecm->nearLogEnergy[0] = LogOfEnergyInQ8(nearEner, aecm->dfaNoisyQDomain);
>-
>-    WebRtcAecm_CalcLinearEnergies(aecm, far_spectrum, echoEst, &tmpFar, &tmpAdapt, &tmpStored);
>-
>-    // Shift buffers
>-    memmove(aecm->echoAdaptLogEnergy + 1, aecm->echoAdaptLogEnergy,
>-            sizeof(int16_t) * (MAX_BUF_LEN - 1));
>-    memmove(aecm->echoStoredLogEnergy + 1, aecm->echoStoredLogEnergy,
>-            sizeof(int16_t) * (MAX_BUF_LEN - 1));
>-
>-    // Logarithm of delayed far end energy
>-    aecm->farLogEnergy = LogOfEnergyInQ8(tmpFar, far_q);
>-
>-    // Logarithm of estimated echo energy through adapted channel
>-    aecm->echoAdaptLogEnergy[0] = LogOfEnergyInQ8(tmpAdapt,
>-                                                  RESOLUTION_CHANNEL16 + far_q);
>-
>-    // Logarithm of estimated echo energy through stored channel
>-    aecm->echoStoredLogEnergy[0] =
>-        LogOfEnergyInQ8(tmpStored, RESOLUTION_CHANNEL16 + far_q);
>-
>-    // Update farend energy levels (min, max, vad, mse)
>-    if (aecm->farLogEnergy > FAR_ENERGY_MIN)
>-    {
>-        if (aecm->startupState == 0)
>-        {
>-            increase_max_shifts = 2;
>-            decrease_min_shifts = 2;
>-            increase_min_shifts = 8;
>-        }
>+  // Local variables
>+  uint32_t tmpAdapt = 0;
>+  uint32_t tmpStored = 0;
>+  uint32_t tmpFar = 0;
>+
>+  int i;
>+
>+  int16_t tmp16;
>+  int16_t increase_max_shifts = 4;
>+  int16_t decrease_max_shifts = 11;
>+  int16_t increase_min_shifts = 11;
>+  int16_t decrease_min_shifts = 3;
>+
>+  // Get log of near end energy and store in buffer
>+
>+  // Shift buffer
>+  memmove(aecm->nearLogEnergy + 1, aecm->nearLogEnergy,
>+          sizeof(int16_t) * (MAX_BUF_LEN - 1));
>+
>+  // Logarithm of integrated magnitude spectrum (nearEner)
>+  aecm->nearLogEnergy[0] = LogOfEnergyInQ8(nearEner, aecm->dfaNoisyQDomain);
>+
>+  WebRtcAecm_CalcLinearEnergies(aecm, far_spectrum, echoEst, &tmpFar, &tmpAdapt,
>+                                &tmpStored);
>+
>+  // Shift buffers
>+  memmove(aecm->echoAdaptLogEnergy + 1, aecm->echoAdaptLogEnergy,
>+          sizeof(int16_t) * (MAX_BUF_LEN - 1));
>+  memmove(aecm->echoStoredLogEnergy + 1, aecm->echoStoredLogEnergy,
>+          sizeof(int16_t) * (MAX_BUF_LEN - 1));
>+
>+  // Logarithm of delayed far end energy
>+  aecm->farLogEnergy = LogOfEnergyInQ8(tmpFar, far_q);
>+
>+  // Logarithm of estimated echo energy through adapted channel
>+  aecm->echoAdaptLogEnergy[0] =
>+      LogOfEnergyInQ8(tmpAdapt, RESOLUTION_CHANNEL16 + far_q);
>+
>+  // Logarithm of estimated echo energy through stored channel
>+  aecm->echoStoredLogEnergy[0] =
>+      LogOfEnergyInQ8(tmpStored, RESOLUTION_CHANNEL16 + far_q);
>+
>+  // Update farend energy levels (min, max, vad, mse)
>+  if (aecm->farLogEnergy > FAR_ENERGY_MIN) {
>+    if (aecm->startupState == 0) {
>+      increase_max_shifts = 2;
>+      decrease_min_shifts = 2;
>+      increase_min_shifts = 8;
>+    }
> 
>-        aecm->farEnergyMin = WebRtcAecm_AsymFilt(aecm->farEnergyMin, aecm->farLogEnergy,
>-                                                 increase_min_shifts, decrease_min_shifts);
>-        aecm->farEnergyMax = WebRtcAecm_AsymFilt(aecm->farEnergyMax, aecm->farLogEnergy,
>-                                                 increase_max_shifts, decrease_max_shifts);
>-        aecm->farEnergyMaxMin = (aecm->farEnergyMax - aecm->farEnergyMin);
>-
>-        // Dynamic VAD region size
>-        tmp16 = 2560 - aecm->farEnergyMin;
>-        if (tmp16 > 0)
>-        {
>-          tmp16 = (int16_t)((tmp16 * FAR_ENERGY_VAD_REGION) >> 9);
>-        } else
>-        {
>-            tmp16 = 0;
>-        }
>-        tmp16 += FAR_ENERGY_VAD_REGION;
>-
>-        if ((aecm->startupState == 0) | (aecm->vadUpdateCount > 1024))
>-        {
>-            // In startup phase or VAD update halted
>-            aecm->farEnergyVAD = aecm->farEnergyMin + tmp16;
>-        } else
>-        {
>-            if (aecm->farEnergyVAD > aecm->farLogEnergy)
>-            {
>-                aecm->farEnergyVAD +=
>-                    (aecm->farLogEnergy + tmp16 - aecm->farEnergyVAD) >> 6;
>-                aecm->vadUpdateCount = 0;
>-            } else
>-            {
>-                aecm->vadUpdateCount++;
>-            }
>-        }
>-        // Put MSE threshold higher than VAD
>-        aecm->farEnergyMSE = aecm->farEnergyVAD + (1 << 8);
>+    aecm->farEnergyMin =
>+        WebRtcAecm_AsymFilt(aecm->farEnergyMin, aecm->farLogEnergy,
>+                            increase_min_shifts, decrease_min_shifts);
>+    aecm->farEnergyMax =
>+        WebRtcAecm_AsymFilt(aecm->farEnergyMax, aecm->farLogEnergy,
>+                            increase_max_shifts, decrease_max_shifts);
>+    aecm->farEnergyMaxMin = (aecm->farEnergyMax - aecm->farEnergyMin);
>+
>+    // Dynamic VAD region size
>+    tmp16 = 2560 - aecm->farEnergyMin;
>+    if (tmp16 > 0) {
>+      tmp16 = (int16_t)((tmp16 * FAR_ENERGY_VAD_REGION) >> 9);
>+    } else {
>+      tmp16 = 0;
>     }
>+    tmp16 += FAR_ENERGY_VAD_REGION;
>+
>+    if ((aecm->startupState == 0) | (aecm->vadUpdateCount > 1024)) {
>+      // In startup phase or VAD update halted
>+      aecm->farEnergyVAD = aecm->farEnergyMin + tmp16;
>+    } else {
>+      if (aecm->farEnergyVAD > aecm->farLogEnergy) {
>+        aecm->farEnergyVAD +=
>+            (aecm->farLogEnergy + tmp16 - aecm->farEnergyVAD) >> 6;
>+        aecm->vadUpdateCount = 0;
>+      } else {
>+        aecm->vadUpdateCount++;
>+      }
>+    }
>+    // Put MSE threshold higher than VAD
>+    aecm->farEnergyMSE = aecm->farEnergyVAD + (1 << 8);
>+  }
> 
>-    // Update VAD variables
>-    if (aecm->farLogEnergy > aecm->farEnergyVAD)
>-    {
>-        if ((aecm->startupState == 0) | (aecm->farEnergyMaxMin > FAR_ENERGY_DIFF))
>-        {
>-            // We are in startup or have significant dynamics in input speech level
>-            aecm->currentVADValue = 1;
>-        }
>-    } else
>-    {
>-        aecm->currentVADValue = 0;
>+  // Update VAD variables
>+  if (aecm->farLogEnergy > aecm->farEnergyVAD) {
>+    if ((aecm->startupState == 0) | (aecm->farEnergyMaxMin > FAR_ENERGY_DIFF)) {
>+      // We are in startup or have significant dynamics in input speech level
>+      aecm->currentVADValue = 1;
>     }
>-    if ((aecm->currentVADValue) && (aecm->firstVAD))
>-    {
>-        aecm->firstVAD = 0;
>-        if (aecm->echoAdaptLogEnergy[0] > aecm->nearLogEnergy[0])
>-        {
>-            // The estimated echo has higher energy than the near end signal.
>-            // This means that the initialization was too aggressive. Scale
>-            // down by a factor 8
>-            for (i = 0; i < PART_LEN1; i++)
>-            {
>-                aecm->channelAdapt16[i] >>= 3;
>-            }
>-            // Compensate the adapted echo energy level accordingly.
>-            aecm->echoAdaptLogEnergy[0] -= (3 << 8);
>-            aecm->firstVAD = 1;
>-        }
>+  } else {
>+    aecm->currentVADValue = 0;
>+  }
>+  if ((aecm->currentVADValue) && (aecm->firstVAD)) {
>+    aecm->firstVAD = 0;
>+    if (aecm->echoAdaptLogEnergy[0] > aecm->nearLogEnergy[0]) {
>+      // The estimated echo has higher energy than the near end signal.
>+      // This means that the initialization was too aggressive. Scale
>+      // down by a factor 8
>+      for (i = 0; i < PART_LEN1; i++) {
>+        aecm->channelAdapt16[i] >>= 3;
>+      }
>+      // Compensate the adapted echo energy level accordingly.
>+      aecm->echoAdaptLogEnergy[0] -= (3 << 8);
>+      aecm->firstVAD = 1;
>     }
>+  }
> }
> 
> // WebRtcAecm_CalcStepSize(...)
>@@ -841,52 +765,50 @@ void WebRtcAecm_CalcEnergies(AecmCore* aecm,
> //
> //
> // @param  aecm  [in]    Handle of the AECM instance.
>-// @param  mu    [out]   (Return value) Stepsize in log2(), i.e. number of shifts.
>+// @param  mu    [out]   (Return value) Stepsize in log2(), i.e. number of
>+// shifts.
> //
> //
> int16_t WebRtcAecm_CalcStepSize(AecmCore* const aecm) {
>-    int32_t tmp32;
>-    int16_t tmp16;
>-    int16_t mu = MU_MAX;
>-
>-    // Here we calculate the step size mu used in the
>-    // following NLMS based Channel estimation algorithm
>-    if (!aecm->currentVADValue)
>-    {
>-        // Far end energy level too low, no channel update
>-        mu = 0;
>-    } else if (aecm->startupState > 0)
>-    {
>-        if (aecm->farEnergyMin >= aecm->farEnergyMax)
>-        {
>-            mu = MU_MIN;
>-        } else
>-        {
>-            tmp16 = (aecm->farLogEnergy - aecm->farEnergyMin);
>-            tmp32 = tmp16 * MU_DIFF;
>-            tmp32 = WebRtcSpl_DivW32W16(tmp32, aecm->farEnergyMaxMin);
>-            mu = MU_MIN - 1 - (int16_t)(tmp32);
>-            // The -1 is an alternative to rounding. This way we get a larger
>-            // stepsize, so we in some sense compensate for truncation in NLMS
>-        }
>-        if (mu < MU_MAX)
>-        {
>-            mu = MU_MAX; // Equivalent with maximum step size of 2^-MU_MAX
>-        }
>+  int32_t tmp32;
>+  int16_t tmp16;
>+  int16_t mu = MU_MAX;
>+
>+  // Here we calculate the step size mu used in the
>+  // following NLMS based Channel estimation algorithm
>+  if (!aecm->currentVADValue) {
>+    // Far end energy level too low, no channel update
>+    mu = 0;
>+  } else if (aecm->startupState > 0) {
>+    if (aecm->farEnergyMin >= aecm->farEnergyMax) {
>+      mu = MU_MIN;
>+    } else {
>+      tmp16 = (aecm->farLogEnergy - aecm->farEnergyMin);
>+      tmp32 = tmp16 * MU_DIFF;
>+      tmp32 = WebRtcSpl_DivW32W16(tmp32, aecm->farEnergyMaxMin);
>+      mu = MU_MIN - 1 - (int16_t)(tmp32);
>+      // The -1 is an alternative to rounding. This way we get a larger
>+      // stepsize, so we in some sense compensate for truncation in NLMS
>+    }
>+    if (mu < MU_MAX) {
>+      mu = MU_MAX;  // Equivalent with maximum step size of 2^-MU_MAX
>     }
>+  }
> 
>-    return mu;
>+  return mu;
> }
> 
> // WebRtcAecm_UpdateChannel(...)
> //
>-// This function performs channel estimation. NLMS and decision on channel storage.
>+// This function performs channel estimation. NLMS and decision on channel
>+// storage.
> //
> //
> // @param  aecm         [i/o]   Handle of the AECM instance.
> // @param  far_spectrum [in]    Absolute value of the farend signal in Q(far_q)
> // @param  far_q        [in]    Q-domain of the farend signal
>-// @param  dfa          [in]    Absolute value of the nearend signal (Q[aecm->dfaQDomain])
>+// @param  dfa          [in]    Absolute value of the nearend signal
>+// (Q[aecm->dfaQDomain])
> // @param  mu           [in]    NLMS step size.
> // @param  echoEst      [i/o]   Estimated echo in Q(far_q+RESOLUTION_CHANNEL16).
> //
>@@ -896,337 +818,301 @@ void WebRtcAecm_UpdateChannel(AecmCore* aecm,
>                               const uint16_t* const dfa,
>                               const int16_t mu,
>                               int32_t* echoEst) {
>-    uint32_t tmpU32no1, tmpU32no2;
>-    int32_t tmp32no1, tmp32no2;
>-    int32_t mseStored;
>-    int32_t mseAdapt;
>-
>-    int i;
>-
>-    int16_t zerosFar, zerosNum, zerosCh, zerosDfa;
>-    int16_t shiftChFar, shiftNum, shift2ResChan;
>-    int16_t tmp16no1;
>-    int16_t xfaQ, dfaQ;
>-
>-    // This is the channel estimation algorithm. It is base on NLMS but has a variable step
>-    // length, which was calculated above.
>-    if (mu)
>-    {
>-        for (i = 0; i < PART_LEN1; i++)
>-        {
>-            // Determine norm of channel and farend to make sure we don't get overflow in
>-            // multiplication
>-            zerosCh = WebRtcSpl_NormU32(aecm->channelAdapt32[i]);
>-            zerosFar = WebRtcSpl_NormU32((uint32_t)far_spectrum[i]);
>-            if (zerosCh + zerosFar > 31)
>-            {
>-                // Multiplication is safe
>-                tmpU32no1 = WEBRTC_SPL_UMUL_32_16(aecm->channelAdapt32[i],
>-                        far_spectrum[i]);
>-                shiftChFar = 0;
>-            } else
>-            {
>-                // We need to shift down before multiplication
>-                shiftChFar = 32 - zerosCh - zerosFar;
>-                tmpU32no1 = rtc::dchecked_cast<uint32_t>(
>-                    aecm->channelAdapt32[i] >> shiftChFar) * far_spectrum[i];
>-            }
>-            // Determine Q-domain of numerator
>-            zerosNum = WebRtcSpl_NormU32(tmpU32no1);
>-            if (dfa[i])
>-            {
>-                zerosDfa = WebRtcSpl_NormU32((uint32_t)dfa[i]);
>-            } else
>-            {
>-                zerosDfa = 32;
>-            }
>-            tmp16no1 = zerosDfa - 2 + aecm->dfaNoisyQDomain -
>-                RESOLUTION_CHANNEL32 - far_q + shiftChFar;
>-            if (zerosNum > tmp16no1 + 1)
>-            {
>-                xfaQ = tmp16no1;
>-                dfaQ = zerosDfa - 2;
>-            } else
>-            {
>-                xfaQ = zerosNum - 2;
>-                dfaQ = RESOLUTION_CHANNEL32 + far_q - aecm->dfaNoisyQDomain -
>-                    shiftChFar + xfaQ;
>-            }
>-            // Add in the same Q-domain
>-            tmpU32no1 = WEBRTC_SPL_SHIFT_W32(tmpU32no1, xfaQ);
>-            tmpU32no2 = WEBRTC_SPL_SHIFT_W32((uint32_t)dfa[i], dfaQ);
>-            tmp32no1 = (int32_t)tmpU32no2 - (int32_t)tmpU32no1;
>-            zerosNum = WebRtcSpl_NormW32(tmp32no1);
>-            if ((tmp32no1) && (far_spectrum[i] > (CHANNEL_VAD << far_q)))
>-            {
>-                //
>-                // Update is needed
>-                //
>-                // This is what we would like to compute
>-                //
>-                // tmp32no1 = dfa[i] - (aecm->channelAdapt[i] * far_spectrum[i])
>-                // tmp32norm = (i + 1)
>-                // aecm->channelAdapt[i] += (2^mu) * tmp32no1
>-                //                        / (tmp32norm * far_spectrum[i])
>-                //
>-
>-                // Make sure we don't get overflow in multiplication.
>-                if (zerosNum + zerosFar > 31)
>-                {
>-                    if (tmp32no1 > 0)
>-                    {
>-                        tmp32no2 = (int32_t)WEBRTC_SPL_UMUL_32_16(tmp32no1,
>-                                                                        far_spectrum[i]);
>-                    } else
>-                    {
>-                        tmp32no2 = -(int32_t)WEBRTC_SPL_UMUL_32_16(-tmp32no1,
>-                                                                         far_spectrum[i]);
>-                    }
>-                    shiftNum = 0;
>-                } else
>-                {
>-                    shiftNum = 32 - (zerosNum + zerosFar);
>-                    if (tmp32no1 > 0)
>-                    {
>-                        tmp32no2 = (tmp32no1 >> shiftNum) * far_spectrum[i];
>-                    } else
>-                    {
>-                        tmp32no2 = -((-tmp32no1 >> shiftNum) * far_spectrum[i]);
>-                    }
>-                }
>-                // Normalize with respect to frequency bin
>-                tmp32no2 = WebRtcSpl_DivW32W16(tmp32no2, i + 1);
>-                // Make sure we are in the right Q-domain
>-                shift2ResChan = shiftNum + shiftChFar - xfaQ - mu - ((30 - zerosFar) << 1);
>-                if (WebRtcSpl_NormW32(tmp32no2) < shift2ResChan)
>-                {
>-                    tmp32no2 = WEBRTC_SPL_WORD32_MAX;
>-                } else
>-                {
>-                    tmp32no2 = WEBRTC_SPL_SHIFT_W32(tmp32no2, shift2ResChan);
>-                }
>-                aecm->channelAdapt32[i] =
>-                    WebRtcSpl_AddSatW32(aecm->channelAdapt32[i], tmp32no2);
>-                if (aecm->channelAdapt32[i] < 0)
>-                {
>-                    // We can never have negative channel gain
>-                    aecm->channelAdapt32[i] = 0;
>-                }
>-                aecm->channelAdapt16[i] =
>-                    (int16_t)(aecm->channelAdapt32[i] >> 16);
>-            }
>+  uint32_t tmpU32no1, tmpU32no2;
>+  int32_t tmp32no1, tmp32no2;
>+  int32_t mseStored;
>+  int32_t mseAdapt;
>+
>+  int i;
>+
>+  int16_t zerosFar, zerosNum, zerosCh, zerosDfa;
>+  int16_t shiftChFar, shiftNum, shift2ResChan;
>+  int16_t tmp16no1;
>+  int16_t xfaQ, dfaQ;
>+
>+  // This is the channel estimation algorithm. It is base on NLMS but has a
>+  // variable step length, which was calculated above.
>+  if (mu) {
>+    for (i = 0; i < PART_LEN1; i++) {
>+      // Determine norm of channel and farend to make sure we don't get overflow
>+      // in multiplication
>+      zerosCh = WebRtcSpl_NormU32(aecm->channelAdapt32[i]);
>+      zerosFar = WebRtcSpl_NormU32((uint32_t)far_spectrum[i]);
>+      if (zerosCh + zerosFar > 31) {
>+        // Multiplication is safe
>+        tmpU32no1 =
>+            WEBRTC_SPL_UMUL_32_16(aecm->channelAdapt32[i], far_spectrum[i]);
>+        shiftChFar = 0;
>+      } else {
>+        // We need to shift down before multiplication
>+        shiftChFar = 32 - zerosCh - zerosFar;
>+        // If zerosCh == zerosFar == 0, shiftChFar is 32. A
>+        // right shift of 32 is undefined. To avoid that, we
>+        // do this check.
>+        tmpU32no1 =
>+            rtc::dchecked_cast<uint32_t>(
>+                shiftChFar >= 32 ? 0 : aecm->channelAdapt32[i] >> shiftChFar) *
>+            far_spectrum[i];
>+      }
>+      // Determine Q-domain of numerator
>+      zerosNum = WebRtcSpl_NormU32(tmpU32no1);
>+      if (dfa[i]) {
>+        zerosDfa = WebRtcSpl_NormU32((uint32_t)dfa[i]);
>+      } else {
>+        zerosDfa = 32;
>+      }
>+      tmp16no1 = zerosDfa - 2 + aecm->dfaNoisyQDomain - RESOLUTION_CHANNEL32 -
>+                 far_q + shiftChFar;
>+      if (zerosNum > tmp16no1 + 1) {
>+        xfaQ = tmp16no1;
>+        dfaQ = zerosDfa - 2;
>+      } else {
>+        xfaQ = zerosNum - 2;
>+        dfaQ = RESOLUTION_CHANNEL32 + far_q - aecm->dfaNoisyQDomain -
>+               shiftChFar + xfaQ;
>+      }
>+      // Add in the same Q-domain
>+      tmpU32no1 = WEBRTC_SPL_SHIFT_W32(tmpU32no1, xfaQ);
>+      tmpU32no2 = WEBRTC_SPL_SHIFT_W32((uint32_t)dfa[i], dfaQ);
>+      tmp32no1 = (int32_t)tmpU32no2 - (int32_t)tmpU32no1;
>+      zerosNum = WebRtcSpl_NormW32(tmp32no1);
>+      if ((tmp32no1) && (far_spectrum[i] > (CHANNEL_VAD << far_q))) {
>+        //
>+        // Update is needed
>+        //
>+        // This is what we would like to compute
>+        //
>+        // tmp32no1 = dfa[i] - (aecm->channelAdapt[i] * far_spectrum[i])
>+        // tmp32norm = (i + 1)
>+        // aecm->channelAdapt[i] += (2^mu) * tmp32no1
>+        //                        / (tmp32norm * far_spectrum[i])
>+        //
>+
>+        // Make sure we don't get overflow in multiplication.
>+        if (zerosNum + zerosFar > 31) {
>+          if (tmp32no1 > 0) {
>+            tmp32no2 =
>+                (int32_t)WEBRTC_SPL_UMUL_32_16(tmp32no1, far_spectrum[i]);
>+          } else {
>+            tmp32no2 =
>+                -(int32_t)WEBRTC_SPL_UMUL_32_16(-tmp32no1, far_spectrum[i]);
>+          }
>+          shiftNum = 0;
>+        } else {
>+          shiftNum = 32 - (zerosNum + zerosFar);
>+          if (tmp32no1 > 0) {
>+            tmp32no2 = (tmp32no1 >> shiftNum) * far_spectrum[i];
>+          } else {
>+            tmp32no2 = -((-tmp32no1 >> shiftNum) * far_spectrum[i]);
>+          }
>+        }
>+        // Normalize with respect to frequency bin
>+        tmp32no2 = WebRtcSpl_DivW32W16(tmp32no2, i + 1);
>+        // Make sure we are in the right Q-domain
>+        shift2ResChan =
>+            shiftNum + shiftChFar - xfaQ - mu - ((30 - zerosFar) << 1);
>+        if (WebRtcSpl_NormW32(tmp32no2) < shift2ResChan) {
>+          tmp32no2 = WEBRTC_SPL_WORD32_MAX;
>+        } else {
>+          tmp32no2 = WEBRTC_SPL_SHIFT_W32(tmp32no2, shift2ResChan);
>+        }
>+        aecm->channelAdapt32[i] =
>+            WebRtcSpl_AddSatW32(aecm->channelAdapt32[i], tmp32no2);
>+        if (aecm->channelAdapt32[i] < 0) {
>+          // We can never have negative channel gain
>+          aecm->channelAdapt32[i] = 0;
>         }
>+        aecm->channelAdapt16[i] = (int16_t)(aecm->channelAdapt32[i] >> 16);
>+      }
>     }
>-    // END: Adaptive channel update
>-
>-    // Determine if we should store or restore the channel
>-    if ((aecm->startupState == 0) & (aecm->currentVADValue))
>-    {
>-        // During startup we store the channel every block,
>-        // and we recalculate echo estimate
>+  }
>+  // END: Adaptive channel update
>+
>+  // Determine if we should store or restore the channel
>+  if ((aecm->startupState == 0) & (aecm->currentVADValue)) {
>+    // During startup we store the channel every block,
>+    // and we recalculate echo estimate
>+    WebRtcAecm_StoreAdaptiveChannel(aecm, far_spectrum, echoEst);
>+  } else {
>+    if (aecm->farLogEnergy < aecm->farEnergyMSE) {
>+      aecm->mseChannelCount = 0;
>+    } else {
>+      aecm->mseChannelCount++;
>+    }
>+    // Enough data for validation. Store channel if we can.
>+    if (aecm->mseChannelCount >= (MIN_MSE_COUNT + 10)) {
>+      // We have enough data.
>+      // Calculate MSE of "Adapt" and "Stored" versions.
>+      // It is actually not MSE, but average absolute error.
>+      mseStored = 0;
>+      mseAdapt = 0;
>+      for (i = 0; i < MIN_MSE_COUNT; i++) {
>+        tmp32no1 = ((int32_t)aecm->echoStoredLogEnergy[i] -
>+                    (int32_t)aecm->nearLogEnergy[i]);
>+        tmp32no2 = WEBRTC_SPL_ABS_W32(tmp32no1);
>+        mseStored += tmp32no2;
>+
>+        tmp32no1 = ((int32_t)aecm->echoAdaptLogEnergy[i] -
>+                    (int32_t)aecm->nearLogEnergy[i]);
>+        tmp32no2 = WEBRTC_SPL_ABS_W32(tmp32no1);
>+        mseAdapt += tmp32no2;
>+      }
>+      if (((mseStored << MSE_RESOLUTION) < (MIN_MSE_DIFF * mseAdapt)) &
>+          ((aecm->mseStoredOld << MSE_RESOLUTION) <
>+           (MIN_MSE_DIFF * aecm->mseAdaptOld))) {
>+        // The stored channel has a significantly lower MSE than the adaptive
>+        // one for two consecutive calculations. Reset the adaptive channel.
>+        WebRtcAecm_ResetAdaptiveChannel(aecm);
>+      } else if (((MIN_MSE_DIFF * mseStored) > (mseAdapt << MSE_RESOLUTION)) &
>+                 (mseAdapt < aecm->mseThreshold) &
>+                 (aecm->mseAdaptOld < aecm->mseThreshold)) {
>+        // The adaptive channel has a significantly lower MSE than the stored
>+        // one. The MSE for the adaptive channel has also been low for two
>+        // consecutive calculations. Store the adaptive channel.
>         WebRtcAecm_StoreAdaptiveChannel(aecm, far_spectrum, echoEst);
>-    } else
>-    {
>-        if (aecm->farLogEnergy < aecm->farEnergyMSE)
>-        {
>-            aecm->mseChannelCount = 0;
>-        } else
>-        {
>-            aecm->mseChannelCount++;
>-        }
>-        // Enough data for validation. Store channel if we can.
>-        if (aecm->mseChannelCount >= (MIN_MSE_COUNT + 10))
>-        {
>-            // We have enough data.
>-            // Calculate MSE of "Adapt" and "Stored" versions.
>-            // It is actually not MSE, but average absolute error.
>-            mseStored = 0;
>-            mseAdapt = 0;
>-            for (i = 0; i < MIN_MSE_COUNT; i++)
>-            {
>-                tmp32no1 = ((int32_t)aecm->echoStoredLogEnergy[i]
>-                        - (int32_t)aecm->nearLogEnergy[i]);
>-                tmp32no2 = WEBRTC_SPL_ABS_W32(tmp32no1);
>-                mseStored += tmp32no2;
>-
>-                tmp32no1 = ((int32_t)aecm->echoAdaptLogEnergy[i]
>-                        - (int32_t)aecm->nearLogEnergy[i]);
>-                tmp32no2 = WEBRTC_SPL_ABS_W32(tmp32no1);
>-                mseAdapt += tmp32no2;
>-            }
>-            if (((mseStored << MSE_RESOLUTION) < (MIN_MSE_DIFF * mseAdapt))
>-                    & ((aecm->mseStoredOld << MSE_RESOLUTION) < (MIN_MSE_DIFF
>-                            * aecm->mseAdaptOld)))
>-            {
>-                // The stored channel has a significantly lower MSE than the adaptive one for
>-                // two consecutive calculations. Reset the adaptive channel.
>-                WebRtcAecm_ResetAdaptiveChannel(aecm);
>-            } else if (((MIN_MSE_DIFF * mseStored) > (mseAdapt << MSE_RESOLUTION)) & (mseAdapt
>-                    < aecm->mseThreshold) & (aecm->mseAdaptOld < aecm->mseThreshold))
>-            {
>-                // The adaptive channel has a significantly lower MSE than the stored one.
>-                // The MSE for the adaptive channel has also been low for two consecutive
>-                // calculations. Store the adaptive channel.
>-                WebRtcAecm_StoreAdaptiveChannel(aecm, far_spectrum, echoEst);
>-
>-                // Update threshold
>-                if (aecm->mseThreshold == WEBRTC_SPL_WORD32_MAX)
>-                {
>-                    aecm->mseThreshold = (mseAdapt + aecm->mseAdaptOld);
>-                } else
>-                {
>-                  int scaled_threshold = aecm->mseThreshold * 5 / 8;
>-                  aecm->mseThreshold +=
>-                      ((mseAdapt - scaled_threshold) * 205) >> 8;
>-                }
>-
>-            }
>-
>-            // Reset counter
>-            aecm->mseChannelCount = 0;
>-
>-            // Store the MSE values.
>-            aecm->mseStoredOld = mseStored;
>-            aecm->mseAdaptOld = mseAdapt;
>+
>+        // Update threshold
>+        if (aecm->mseThreshold == WEBRTC_SPL_WORD32_MAX) {
>+          aecm->mseThreshold = (mseAdapt + aecm->mseAdaptOld);
>+        } else {
>+          int scaled_threshold = aecm->mseThreshold * 5 / 8;
>+          aecm->mseThreshold += ((mseAdapt - scaled_threshold) * 205) >> 8;
>         }
>+      }
>+
>+      // Reset counter
>+      aecm->mseChannelCount = 0;
>+
>+      // Store the MSE values.
>+      aecm->mseStoredOld = mseStored;
>+      aecm->mseAdaptOld = mseAdapt;
>     }
>-    // END: Determine if we should store or reset channel estimate.
>+  }
>+  // END: Determine if we should store or reset channel estimate.
> }
> 
> // CalcSuppressionGain(...)
> //
>-// This function calculates the suppression gain that is used in the Wiener filter.
>+// This function calculates the suppression gain that is used in the Wiener
>+// filter.
> //
> //
> // @param  aecm     [i/n]   Handle of the AECM instance.
>-// @param  supGain  [out]   (Return value) Suppression gain with which to scale the noise
>+// @param  supGain  [out]   (Return value) Suppression gain with which to scale
>+// the noise
> //                          level (Q14).
> //
> //
> int16_t WebRtcAecm_CalcSuppressionGain(AecmCore* const aecm) {
>-    int32_t tmp32no1;
>-
>-    int16_t supGain = SUPGAIN_DEFAULT;
>-    int16_t tmp16no1;
>-    int16_t dE = 0;
>-
>-    // Determine suppression gain used in the Wiener filter. The gain is based on a mix of far
>-    // end energy and echo estimation error.
>-    // Adjust for the far end signal level. A low signal level indicates no far end signal,
>-    // hence we set the suppression gain to 0
>-    if (!aecm->currentVADValue)
>-    {
>-        supGain = 0;
>-    } else
>-    {
>-        // Adjust for possible double talk. If we have large variations in estimation error we
>-        // likely have double talk (or poor channel).
>-        tmp16no1 = (aecm->nearLogEnergy[0] - aecm->echoStoredLogEnergy[0] - ENERGY_DEV_OFFSET);
>-        dE = WEBRTC_SPL_ABS_W16(tmp16no1);
>-
>-        if (dE < ENERGY_DEV_TOL)
>-        {
>-            // Likely no double talk. The better estimation, the more we can suppress signal.
>-            // Update counters
>-            if (dE < SUPGAIN_EPC_DT)
>-            {
>-                tmp32no1 = aecm->supGainErrParamDiffAB * dE;
>-                tmp32no1 += (SUPGAIN_EPC_DT >> 1);
>-                tmp16no1 = (int16_t)WebRtcSpl_DivW32W16(tmp32no1, SUPGAIN_EPC_DT);
>-                supGain = aecm->supGainErrParamA - tmp16no1;
>-            } else
>-            {
>-                tmp32no1 = aecm->supGainErrParamDiffBD * (ENERGY_DEV_TOL - dE);
>-                tmp32no1 += ((ENERGY_DEV_TOL - SUPGAIN_EPC_DT) >> 1);
>-                tmp16no1 = (int16_t)WebRtcSpl_DivW32W16(tmp32no1, (ENERGY_DEV_TOL
>-                        - SUPGAIN_EPC_DT));
>-                supGain = aecm->supGainErrParamD + tmp16no1;
>-            }
>-        } else
>-        {
>-            // Likely in double talk. Use default value
>-            supGain = aecm->supGainErrParamD;
>-        }
>+  int32_t tmp32no1;
>+
>+  int16_t supGain = SUPGAIN_DEFAULT;
>+  int16_t tmp16no1;
>+  int16_t dE = 0;
>+
>+  // Determine suppression gain used in the Wiener filter. The gain is based on
>+  // a mix of far end energy and echo estimation error. Adjust for the far end
>+  // signal level. A low signal level indicates no far end signal, hence we set
>+  // the suppression gain to 0
>+  if (!aecm->currentVADValue) {
>+    supGain = 0;
>+  } else {
>+    // Adjust for possible double talk. If we have large variations in
>+    // estimation error we likely have double talk (or poor channel).
>+    tmp16no1 = (aecm->nearLogEnergy[0] - aecm->echoStoredLogEnergy[0] -
>+                ENERGY_DEV_OFFSET);
>+    dE = WEBRTC_SPL_ABS_W16(tmp16no1);
>+
>+    if (dE < ENERGY_DEV_TOL) {
>+      // Likely no double talk. The better estimation, the more we can suppress
>+      // signal. Update counters
>+      if (dE < SUPGAIN_EPC_DT) {
>+        tmp32no1 = aecm->supGainErrParamDiffAB * dE;
>+        tmp32no1 += (SUPGAIN_EPC_DT >> 1);
>+        tmp16no1 = (int16_t)WebRtcSpl_DivW32W16(tmp32no1, SUPGAIN_EPC_DT);
>+        supGain = aecm->supGainErrParamA - tmp16no1;
>+      } else {
>+        tmp32no1 = aecm->supGainErrParamDiffBD * (ENERGY_DEV_TOL - dE);
>+        tmp32no1 += ((ENERGY_DEV_TOL - SUPGAIN_EPC_DT) >> 1);
>+        tmp16no1 = (int16_t)WebRtcSpl_DivW32W16(
>+            tmp32no1, (ENERGY_DEV_TOL - SUPGAIN_EPC_DT));
>+        supGain = aecm->supGainErrParamD + tmp16no1;
>+      }
>+    } else {
>+      // Likely in double talk. Use default value
>+      supGain = aecm->supGainErrParamD;
>     }
>+  }
> 
>-    if (supGain > aecm->supGainOld)
>-    {
>-        tmp16no1 = supGain;
>-    } else
>-    {
>-        tmp16no1 = aecm->supGainOld;
>-    }
>-    aecm->supGainOld = supGain;
>-    if (tmp16no1 < aecm->supGain)
>-    {
>-        aecm->supGain += (int16_t)((tmp16no1 - aecm->supGain) >> 4);
>-    } else
>-    {
>-        aecm->supGain += (int16_t)((tmp16no1 - aecm->supGain) >> 4);
>-    }
>+  if (supGain > aecm->supGainOld) {
>+    tmp16no1 = supGain;
>+  } else {
>+    tmp16no1 = aecm->supGainOld;
>+  }
>+  aecm->supGainOld = supGain;
>+  if (tmp16no1 < aecm->supGain) {
>+    aecm->supGain += (int16_t)((tmp16no1 - aecm->supGain) >> 4);
>+  } else {
>+    aecm->supGain += (int16_t)((tmp16no1 - aecm->supGain) >> 4);
>+  }
> 
>-    // END: Update suppression gain
>+  // END: Update suppression gain
> 
>-    return aecm->supGain;
>+  return aecm->supGain;
> }
> 
> void WebRtcAecm_BufferFarFrame(AecmCore* const aecm,
>                                const int16_t* const farend,
>                                const int farLen) {
>-    int writeLen = farLen, writePos = 0;
>-
>-    // Check if the write position must be wrapped
>-    while (aecm->farBufWritePos + writeLen > FAR_BUF_LEN)
>-    {
>-        // Write to remaining buffer space before wrapping
>-        writeLen = FAR_BUF_LEN - aecm->farBufWritePos;
>-        memcpy(aecm->farBuf + aecm->farBufWritePos, farend + writePos,
>-               sizeof(int16_t) * writeLen);
>-        aecm->farBufWritePos = 0;
>-        writePos = writeLen;
>-        writeLen = farLen - writeLen;
>-    }
>+  int writeLen = farLen, writePos = 0;
> 
>+  // Check if the write position must be wrapped
>+  while (aecm->farBufWritePos + writeLen > FAR_BUF_LEN) {
>+    // Write to remaining buffer space before wrapping
>+    writeLen = FAR_BUF_LEN - aecm->farBufWritePos;
>     memcpy(aecm->farBuf + aecm->farBufWritePos, farend + writePos,
>            sizeof(int16_t) * writeLen);
>-    aecm->farBufWritePos += writeLen;
>+    aecm->farBufWritePos = 0;
>+    writePos = writeLen;
>+    writeLen = farLen - writeLen;
>+  }
>+
>+  memcpy(aecm->farBuf + aecm->farBufWritePos, farend + writePos,
>+         sizeof(int16_t) * writeLen);
>+  aecm->farBufWritePos += writeLen;
> }
> 
> void WebRtcAecm_FetchFarFrame(AecmCore* const aecm,
>                               int16_t* const farend,
>                               const int farLen,
>                               const int knownDelay) {
>-    int readLen = farLen;
>-    int readPos = 0;
>-    int delayChange = knownDelay - aecm->lastKnownDelay;
>+  int readLen = farLen;
>+  int readPos = 0;
>+  int delayChange = knownDelay - aecm->lastKnownDelay;
> 
>-    aecm->farBufReadPos -= delayChange;
>+  aecm->farBufReadPos -= delayChange;
> 
>-    // Check if delay forces a read position wrap
>-    while (aecm->farBufReadPos < 0)
>-    {
>-        aecm->farBufReadPos += FAR_BUF_LEN;
>-    }
>-    while (aecm->farBufReadPos > FAR_BUF_LEN - 1)
>-    {
>-        aecm->farBufReadPos -= FAR_BUF_LEN;
>-    }
>-
>-    aecm->lastKnownDelay = knownDelay;
>+  // Check if delay forces a read position wrap
>+  while (aecm->farBufReadPos < 0) {
>+    aecm->farBufReadPos += FAR_BUF_LEN;
>+  }
>+  while (aecm->farBufReadPos > FAR_BUF_LEN - 1) {
>+    aecm->farBufReadPos -= FAR_BUF_LEN;
>+  }
> 
>-    // Check if read position must be wrapped
>-    while (aecm->farBufReadPos + readLen > FAR_BUF_LEN)
>-    {
>+  aecm->lastKnownDelay = knownDelay;
> 
>-        // Read from remaining buffer space before wrapping
>-        readLen = FAR_BUF_LEN - aecm->farBufReadPos;
>-        memcpy(farend + readPos, aecm->farBuf + aecm->farBufReadPos,
>-               sizeof(int16_t) * readLen);
>-        aecm->farBufReadPos = 0;
>-        readPos = readLen;
>-        readLen = farLen - readLen;
>-    }
>+  // Check if read position must be wrapped
>+  while (aecm->farBufReadPos + readLen > FAR_BUF_LEN) {
>+    // Read from remaining buffer space before wrapping
>+    readLen = FAR_BUF_LEN - aecm->farBufReadPos;
>     memcpy(farend + readPos, aecm->farBuf + aecm->farBufReadPos,
>            sizeof(int16_t) * readLen);
>-    aecm->farBufReadPos += readLen;
>+    aecm->farBufReadPos = 0;
>+    readPos = readLen;
>+    readLen = farLen - readLen;
>+  }
>+  memcpy(farend + readPos, aecm->farBuf + aecm->farBufReadPos,
>+         sizeof(int16_t) * readLen);
>+  aecm->farBufReadPos += readLen;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aecm/aecm_core.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aecm/aecm_core.h
>index feb997e9fa1..1681d2a92f6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aecm/aecm_core.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aecm/aecm_core.h
>@@ -18,7 +18,6 @@ extern "C" {
> #include "common_audio/signal_processing/include/signal_processing_library.h"
> }
> #include "modules/audio_processing/aecm/aecm_defines.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> #ifdef _MSC_VER  // visual c++
> #define ALIGN8_BEG __declspec(align(8))
>@@ -29,109 +28,109 @@ extern "C" {
> #endif
> 
> typedef struct {
>-    int16_t real;
>-    int16_t imag;
>+  int16_t real;
>+  int16_t imag;
> } ComplexInt16;
> 
> typedef struct {
>-    int farBufWritePos;
>-    int farBufReadPos;
>-    int knownDelay;
>-    int lastKnownDelay;
>-    int firstVAD;  // Parameter to control poorly initialized channels
>-
>-    RingBuffer* farFrameBuf;
>-    RingBuffer* nearNoisyFrameBuf;
>-    RingBuffer* nearCleanFrameBuf;
>-    RingBuffer* outFrameBuf;
>-
>-    int16_t farBuf[FAR_BUF_LEN];
>-
>-    int16_t mult;
>-    uint32_t seed;
>-
>-    // Delay estimation variables
>-    void* delay_estimator_farend;
>-    void* delay_estimator;
>-    uint16_t currentDelay;
>-    // Far end history variables
>-    // TODO(bjornv): Replace |far_history| with ring_buffer.
>-    uint16_t far_history[PART_LEN1 * MAX_DELAY];
>-    int far_history_pos;
>-    int far_q_domains[MAX_DELAY];
>-
>-    int16_t nlpFlag;
>-    int16_t fixedDelay;
>-
>-    uint32_t totCount;
>-
>-    int16_t dfaCleanQDomain;
>-    int16_t dfaCleanQDomainOld;
>-    int16_t dfaNoisyQDomain;
>-    int16_t dfaNoisyQDomainOld;
>-
>-    int16_t nearLogEnergy[MAX_BUF_LEN];
>-    int16_t farLogEnergy;
>-    int16_t echoAdaptLogEnergy[MAX_BUF_LEN];
>-    int16_t echoStoredLogEnergy[MAX_BUF_LEN];
>-
>-    // The extra 16 or 32 bytes in the following buffers are for alignment based
>-    // Neon code.
>-    // It's designed this way since the current GCC compiler can't align a
>-    // buffer in 16 or 32 byte boundaries properly.
>-    int16_t channelStored_buf[PART_LEN1 + 8];
>-    int16_t channelAdapt16_buf[PART_LEN1 + 8];
>-    int32_t channelAdapt32_buf[PART_LEN1 + 8];
>-    int16_t xBuf_buf[PART_LEN2 + 16];  // farend
>-    int16_t dBufClean_buf[PART_LEN2 + 16];  // nearend
>-    int16_t dBufNoisy_buf[PART_LEN2 + 16];  // nearend
>-    int16_t outBuf_buf[PART_LEN + 8];
>-
>-    // Pointers to the above buffers
>-    int16_t *channelStored;
>-    int16_t *channelAdapt16;
>-    int32_t *channelAdapt32;
>-    int16_t *xBuf;
>-    int16_t *dBufClean;
>-    int16_t *dBufNoisy;
>-    int16_t *outBuf;
>-
>-    int32_t echoFilt[PART_LEN1];
>-    int16_t nearFilt[PART_LEN1];
>-    int32_t noiseEst[PART_LEN1];
>-    int           noiseEstTooLowCtr[PART_LEN1];
>-    int           noiseEstTooHighCtr[PART_LEN1];
>-    int16_t noiseEstCtr;
>-    int16_t cngMode;
>-
>-    int32_t mseAdaptOld;
>-    int32_t mseStoredOld;
>-    int32_t mseThreshold;
>-
>-    int16_t farEnergyMin;
>-    int16_t farEnergyMax;
>-    int16_t farEnergyMaxMin;
>-    int16_t farEnergyVAD;
>-    int16_t farEnergyMSE;
>-    int currentVADValue;
>-    int16_t vadUpdateCount;
>-
>-    int16_t startupState;
>-    int16_t mseChannelCount;
>-    int16_t supGain;
>-    int16_t supGainOld;
>-
>-    int16_t supGainErrParamA;
>-    int16_t supGainErrParamD;
>-    int16_t supGainErrParamDiffAB;
>-    int16_t supGainErrParamDiffBD;
>-
>-    struct RealFFT* real_fft;
>+  int farBufWritePos;
>+  int farBufReadPos;
>+  int knownDelay;
>+  int lastKnownDelay;
>+  int firstVAD;  // Parameter to control poorly initialized channels
>+
>+  RingBuffer* farFrameBuf;
>+  RingBuffer* nearNoisyFrameBuf;
>+  RingBuffer* nearCleanFrameBuf;
>+  RingBuffer* outFrameBuf;
>+
>+  int16_t farBuf[FAR_BUF_LEN];
>+
>+  int16_t mult;
>+  uint32_t seed;
>+
>+  // Delay estimation variables
>+  void* delay_estimator_farend;
>+  void* delay_estimator;
>+  uint16_t currentDelay;
>+  // Far end history variables
>+  // TODO(bjornv): Replace |far_history| with ring_buffer.
>+  uint16_t far_history[PART_LEN1 * MAX_DELAY];
>+  int far_history_pos;
>+  int far_q_domains[MAX_DELAY];
>+
>+  int16_t nlpFlag;
>+  int16_t fixedDelay;
>+
>+  uint32_t totCount;
>+
>+  int16_t dfaCleanQDomain;
>+  int16_t dfaCleanQDomainOld;
>+  int16_t dfaNoisyQDomain;
>+  int16_t dfaNoisyQDomainOld;
>+
>+  int16_t nearLogEnergy[MAX_BUF_LEN];
>+  int16_t farLogEnergy;
>+  int16_t echoAdaptLogEnergy[MAX_BUF_LEN];
>+  int16_t echoStoredLogEnergy[MAX_BUF_LEN];
>+
>+  // The extra 16 or 32 bytes in the following buffers are for alignment based
>+  // Neon code.
>+  // It's designed this way since the current GCC compiler can't align a
>+  // buffer in 16 or 32 byte boundaries properly.
>+  int16_t channelStored_buf[PART_LEN1 + 8];
>+  int16_t channelAdapt16_buf[PART_LEN1 + 8];
>+  int32_t channelAdapt32_buf[PART_LEN1 + 8];
>+  int16_t xBuf_buf[PART_LEN2 + 16];       // farend
>+  int16_t dBufClean_buf[PART_LEN2 + 16];  // nearend
>+  int16_t dBufNoisy_buf[PART_LEN2 + 16];  // nearend
>+  int16_t outBuf_buf[PART_LEN + 8];
>+
>+  // Pointers to the above buffers
>+  int16_t* channelStored;
>+  int16_t* channelAdapt16;
>+  int32_t* channelAdapt32;
>+  int16_t* xBuf;
>+  int16_t* dBufClean;
>+  int16_t* dBufNoisy;
>+  int16_t* outBuf;
>+
>+  int32_t echoFilt[PART_LEN1];
>+  int16_t nearFilt[PART_LEN1];
>+  int32_t noiseEst[PART_LEN1];
>+  int noiseEstTooLowCtr[PART_LEN1];
>+  int noiseEstTooHighCtr[PART_LEN1];
>+  int16_t noiseEstCtr;
>+  int16_t cngMode;
>+
>+  int32_t mseAdaptOld;
>+  int32_t mseStoredOld;
>+  int32_t mseThreshold;
>+
>+  int16_t farEnergyMin;
>+  int16_t farEnergyMax;
>+  int16_t farEnergyMaxMin;
>+  int16_t farEnergyVAD;
>+  int16_t farEnergyMSE;
>+  int currentVADValue;
>+  int16_t vadUpdateCount;
>+
>+  int16_t startupState;
>+  int16_t mseChannelCount;
>+  int16_t supGain;
>+  int16_t supGainOld;
>+
>+  int16_t supGainErrParamA;
>+  int16_t supGainErrParamD;
>+  int16_t supGainErrParamDiffAB;
>+  int16_t supGainErrParamDiffBD;
>+
>+  struct RealFFT* real_fft;
> 
> #ifdef AEC_DEBUG
>-    FILE *farFile;
>-    FILE *nearFile;
>-    FILE *outFile;
>+  FILE* farFile;
>+  FILE* nearFile;
>+  FILE* outFile;
> #endif
> } AecmCore;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aecm/aecm_core_c.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aecm/aecm_core_c.cc
>index fa5f15cf105..905274fa97b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aecm/aecm_core_c.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aecm/aecm_core_c.cc
>@@ -26,32 +26,28 @@ extern "C" {
> #include "rtc_base/checks.h"
> #include "rtc_base/numerics/safe_conversions.h"
> #include "rtc_base/sanitizer.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> // Square root of Hanning window in Q14.
> static const ALIGN8_BEG int16_t WebRtcAecm_kSqrtHanning[] ALIGN8_END = {
>-  0, 399, 798, 1196, 1594, 1990, 2386, 2780, 3172,
>-  3562, 3951, 4337, 4720, 5101, 5478, 5853, 6224,
>-  6591, 6954, 7313, 7668, 8019, 8364, 8705, 9040,
>-  9370, 9695, 10013, 10326, 10633, 10933, 11227, 11514,
>-  11795, 12068, 12335, 12594, 12845, 13089, 13325, 13553,
>-  13773, 13985, 14189, 14384, 14571, 14749, 14918, 15079,
>-  15231, 15373, 15506, 15631, 15746, 15851, 15947, 16034,
>-  16111, 16179, 16237, 16286, 16325, 16354, 16373, 16384
>-};
>+    0,     399,   798,   1196,  1594,  1990,  2386,  2780,  3172,  3562,  3951,
>+    4337,  4720,  5101,  5478,  5853,  6224,  6591,  6954,  7313,  7668,  8019,
>+    8364,  8705,  9040,  9370,  9695,  10013, 10326, 10633, 10933, 11227, 11514,
>+    11795, 12068, 12335, 12594, 12845, 13089, 13325, 13553, 13773, 13985, 14189,
>+    14384, 14571, 14749, 14918, 15079, 15231, 15373, 15506, 15631, 15746, 15851,
>+    15947, 16034, 16111, 16179, 16237, 16286, 16325, 16354, 16373, 16384};
> 
> #ifdef AECM_WITH_ABS_APPROX
>-//Q15 alpha = 0.99439986968132  const Factor for magnitude approximation
>+// Q15 alpha = 0.99439986968132  const Factor for magnitude approximation
> static const uint16_t kAlpha1 = 32584;
>-//Q15 beta = 0.12967166976970   const Factor for magnitude approximation
>+// Q15 beta = 0.12967166976970   const Factor for magnitude approximation
> static const uint16_t kBeta1 = 4249;
>-//Q15 alpha = 0.94234827210087  const Factor for magnitude approximation
>+// Q15 alpha = 0.94234827210087  const Factor for magnitude approximation
> static const uint16_t kAlpha2 = 30879;
>-//Q15 beta = 0.33787806009150   const Factor for magnitude approximation
>+// Q15 beta = 0.33787806009150   const Factor for magnitude approximation
> static const uint16_t kBeta2 = 11072;
>-//Q15 alpha = 0.82247698684306  const Factor for magnitude approximation
>+// Q15 alpha = 0.82247698684306  const Factor for magnitude approximation
> static const uint16_t kAlpha3 = 26951;
>-//Q15 beta = 0.57762063060713   const Factor for magnitude approximation
>+// Q15 beta = 0.57762063060713   const Factor for magnitude approximation
> static const uint16_t kBeta3 = 18927;
> #endif
> 
>@@ -77,8 +73,8 @@ static void WindowAndFFT(AecmCore* aecm,
>     int16_t scaled_time_signal = time_signal[i] * (1 << time_signal_scaling);
>     fft[i] = (int16_t)((scaled_time_signal * WebRtcAecm_kSqrtHanning[i]) >> 14);
>     scaled_time_signal = time_signal[i + PART_LEN] * (1 << time_signal_scaling);
>-    fft[PART_LEN + i] = (int16_t)((
>-        scaled_time_signal * WebRtcAecm_kSqrtHanning[PART_LEN - i]) >> 14);
>+    fft[PART_LEN + i] = (int16_t)(
>+        (scaled_time_signal * WebRtcAecm_kSqrtHanning[PART_LEN - i]) >> 14);
>   }
> 
>   // Do forward FFT, then take only the first PART_LEN complex samples,
>@@ -115,32 +111,27 @@ static void InverseFFTAndWindow(AecmCore* aecm,
>   outCFFT = WebRtcSpl_RealInverseFFT(aecm->real_fft, fft, ifft_out);
>   for (i = 0; i < PART_LEN; i++) {
>     ifft_out[i] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
>-                    ifft_out[i], WebRtcAecm_kSqrtHanning[i], 14);
>+        ifft_out[i], WebRtcAecm_kSqrtHanning[i], 14);
>     tmp32no1 = WEBRTC_SPL_SHIFT_W32((int32_t)ifft_out[i],
>-                                     outCFFT - aecm->dfaCleanQDomain);
>+                                    outCFFT - aecm->dfaCleanQDomain);
>     output[i] = (int16_t)WEBRTC_SPL_SAT(WEBRTC_SPL_WORD16_MAX,
>                                         tmp32no1 + aecm->outBuf[i],
>                                         WEBRTC_SPL_WORD16_MIN);
> 
>-    tmp32no1 = (ifft_out[PART_LEN + i] *
>-        WebRtcAecm_kSqrtHanning[PART_LEN - i]) >> 14;
>-    tmp32no1 = WEBRTC_SPL_SHIFT_W32(tmp32no1,
>-                                    outCFFT - aecm->dfaCleanQDomain);
>-    aecm->outBuf[i] = (int16_t)WEBRTC_SPL_SAT(WEBRTC_SPL_WORD16_MAX,
>-                                                tmp32no1,
>-                                                WEBRTC_SPL_WORD16_MIN);
>+    tmp32no1 =
>+        (ifft_out[PART_LEN + i] * WebRtcAecm_kSqrtHanning[PART_LEN - i]) >> 14;
>+    tmp32no1 = WEBRTC_SPL_SHIFT_W32(tmp32no1, outCFFT - aecm->dfaCleanQDomain);
>+    aecm->outBuf[i] = (int16_t)WEBRTC_SPL_SAT(WEBRTC_SPL_WORD16_MAX, tmp32no1,
>+                                              WEBRTC_SPL_WORD16_MIN);
>   }
> 
>   // Copy the current block to the old position
>   // (aecm->outBuf is shifted elsewhere)
>   memcpy(aecm->xBuf, aecm->xBuf + PART_LEN, sizeof(int16_t) * PART_LEN);
>-  memcpy(aecm->dBufNoisy,
>-         aecm->dBufNoisy + PART_LEN,
>+  memcpy(aecm->dBufNoisy, aecm->dBufNoisy + PART_LEN,
>          sizeof(int16_t) * PART_LEN);
>-  if (nearendClean != NULL)
>-  {
>-    memcpy(aecm->dBufClean,
>-           aecm->dBufClean + PART_LEN,
>+  if (nearendClean != NULL) {
>+    memcpy(aecm->dBufClean, aecm->dBufClean + PART_LEN,
>            sizeof(int16_t) * PART_LEN);
>   }
> }
>@@ -171,7 +162,7 @@ static int TimeToFrequencyDomain(AecmCore* aecm,
> 
>   // In fft_buf, +16 for 32-byte alignment.
>   int16_t fft_buf[PART_LEN4 + 16];
>-  int16_t *fft = (int16_t *) (((uintptr_t) fft_buf + 31) & ~31);
>+  int16_t* fft = (int16_t*)(((uintptr_t)fft_buf + 31) & ~31);
> 
>   int16_t tmp16no1;
> #ifndef WEBRTC_ARCH_ARM_V7
>@@ -196,54 +187,43 @@ static int TimeToFrequencyDomain(AecmCore* aecm,
>   freq_signal[0].imag = 0;
>   freq_signal[PART_LEN].imag = 0;
>   freq_signal_abs[0] = (uint16_t)WEBRTC_SPL_ABS_W16(freq_signal[0].real);
>-  freq_signal_abs[PART_LEN] = (uint16_t)WEBRTC_SPL_ABS_W16(
>-                                freq_signal[PART_LEN].real);
>-  (*freq_signal_sum_abs) = (uint32_t)(freq_signal_abs[0]) +
>-                           (uint32_t)(freq_signal_abs[PART_LEN]);
>-
>-  for (i = 1; i < PART_LEN; i++)
>-  {
>-    if (freq_signal[i].real == 0)
>-    {
>+  freq_signal_abs[PART_LEN] =
>+      (uint16_t)WEBRTC_SPL_ABS_W16(freq_signal[PART_LEN].real);
>+  (*freq_signal_sum_abs) =
>+      (uint32_t)(freq_signal_abs[0]) + (uint32_t)(freq_signal_abs[PART_LEN]);
>+
>+  for (i = 1; i < PART_LEN; i++) {
>+    if (freq_signal[i].real == 0) {
>       freq_signal_abs[i] = (uint16_t)WEBRTC_SPL_ABS_W16(freq_signal[i].imag);
>-    }
>-    else if (freq_signal[i].imag == 0)
>-    {
>+    } else if (freq_signal[i].imag == 0) {
>       freq_signal_abs[i] = (uint16_t)WEBRTC_SPL_ABS_W16(freq_signal[i].real);
>-    }
>-    else
>-    {
>-      // Approximation for magnitude of complex fft output
>-      // magn = sqrt(real^2 + imag^2)
>-      // magn ~= alpha * max(|imag|,|real|) + beta * min(|imag|,|real|)
>-      //
>-      // The parameters alpha and beta are stored in Q15
>+    } else {
>+// Approximation for magnitude of complex fft output
>+// magn = sqrt(real^2 + imag^2)
>+// magn ~= alpha * max(|imag|,|real|) + beta * min(|imag|,|real|)
>+//
>+// The parameters alpha and beta are stored in Q15
> 
> #ifdef AECM_WITH_ABS_APPROX
>       tmp16no1 = WEBRTC_SPL_ABS_W16(freq_signal[i].real);
>       tmp16no2 = WEBRTC_SPL_ABS_W16(freq_signal[i].imag);
> 
>-      if(tmp16no1 > tmp16no2)
>-      {
>+      if (tmp16no1 > tmp16no2) {
>         max_value = tmp16no1;
>         min_value = tmp16no2;
>-      } else
>-      {
>+      } else {
>         max_value = tmp16no2;
>         min_value = tmp16no1;
>       }
> 
>       // Magnitude in Q(-6)
>-      if ((max_value >> 2) > min_value)
>-      {
>+      if ((max_value >> 2) > min_value) {
>         alpha = kAlpha1;
>         beta = kBeta1;
>-      } else if ((max_value >> 1) > min_value)
>-      {
>+      } else if ((max_value >> 1) > min_value) {
>         alpha = kAlpha2;
>         beta = kBeta2;
>-      } else
>-      {
>+      } else {
>         alpha = kAlpha3;
>         beta = kBeta3;
>       }
>@@ -253,24 +233,21 @@ static int TimeToFrequencyDomain(AecmCore* aecm,
> #else
> #ifdef WEBRTC_ARCH_ARM_V7
>       __asm __volatile(
>-        "smulbb %[tmp32no1], %[real], %[real]\n\t"
>-        "smlabb %[tmp32no2], %[imag], %[imag], %[tmp32no1]\n\t"
>-        :[tmp32no1]"+&r"(tmp32no1),
>-         [tmp32no2]"=r"(tmp32no2)
>-        :[real]"r"(freq_signal[i].real),
>-         [imag]"r"(freq_signal[i].imag)
>-      );
>+          "smulbb %[tmp32no1], %[real], %[real]\n\t"
>+          "smlabb %[tmp32no2], %[imag], %[imag], %[tmp32no1]\n\t"
>+          : [tmp32no1] "+&r"(tmp32no1), [tmp32no2] "=r"(tmp32no2)
>+          : [real] "r"(freq_signal[i].real), [imag] "r"(freq_signal[i].imag));
> #else
>       tmp16no1 = WEBRTC_SPL_ABS_W16(freq_signal[i].real);
>       tmp16no2 = WEBRTC_SPL_ABS_W16(freq_signal[i].imag);
>       tmp32no1 = tmp16no1 * tmp16no1;
>       tmp32no2 = tmp16no2 * tmp16no2;
>       tmp32no2 = WebRtcSpl_AddSatW32(tmp32no1, tmp32no2);
>-#endif // WEBRTC_ARCH_ARM_V7
>+#endif  // WEBRTC_ARCH_ARM_V7
>       tmp32no1 = WebRtcSpl_SqrtFloor(tmp32no2);
> 
>       freq_signal_abs[i] = (uint16_t)tmp32no1;
>-#endif // AECM_WITH_ABS_APPROX
>+#endif  // AECM_WITH_ABS_APPROX
>     }
>     (*freq_signal_sum_abs) += (uint32_t)freq_signal_abs[i];
>   }
>@@ -279,11 +256,11 @@ static int TimeToFrequencyDomain(AecmCore* aecm,
> }
> 
> int RTC_NO_SANITIZE("signed-integer-overflow")  // bugs.webrtc.org/8200
>-WebRtcAecm_ProcessBlock(AecmCore* aecm,
>-                        const int16_t* farend,
>-                        const int16_t* nearendNoisy,
>-                        const int16_t* nearendClean,
>-                        int16_t* output) {
>+    WebRtcAecm_ProcessBlock(AecmCore* aecm,
>+                            const int16_t* farend,
>+                            const int16_t* nearendNoisy,
>+                            const int16_t* nearendClean,
>+                            int16_t* output) {
>   int i;
> 
>   uint32_t xfaSum;
>@@ -302,13 +279,13 @@ WebRtcAecm_ProcessBlock(AecmCore* aecm,
> 
>   // 32 byte aligned buffers (with +8 or +16).
>   // TODO(kma): define fft with ComplexInt16.
>-  int16_t fft_buf[PART_LEN4 + 2 + 16]; // +2 to make a loop safe.
>+  int16_t fft_buf[PART_LEN4 + 2 + 16];  // +2 to make a loop safe.
>   int32_t echoEst32_buf[PART_LEN1 + 8];
>   int32_t dfw_buf[PART_LEN2 + 8];
>   int32_t efw_buf[PART_LEN2 + 8];
> 
>-  int16_t* fft = (int16_t*) (((uintptr_t) fft_buf + 31) & ~ 31);
>-  int32_t* echoEst32 = (int32_t*) (((uintptr_t) echoEst32_buf + 31) & ~ 31);
>+  int16_t* fft = (int16_t*)(((uintptr_t)fft_buf + 31) & ~31);
>+  int32_t* echoEst32 = (int32_t*)(((uintptr_t)echoEst32_buf + 31) & ~31);
>   ComplexInt16* dfw = (ComplexInt16*)(((uintptr_t)dfw_buf + 31) & ~31);
>   ComplexInt16* efw = (ComplexInt16*)(((uintptr_t)efw_buf + 31) & ~31);
> 
>@@ -334,53 +311,37 @@ WebRtcAecm_ProcessBlock(AecmCore* aecm,
>   // (1) another CONV_LEN blocks
>   // (2) the rest
> 
>-  if (aecm->startupState < 2)
>-  {
>-    aecm->startupState = (aecm->totCount >= CONV_LEN) +
>-                         (aecm->totCount >= CONV_LEN2);
>+  if (aecm->startupState < 2) {
>+    aecm->startupState =
>+        (aecm->totCount >= CONV_LEN) + (aecm->totCount >= CONV_LEN2);
>   }
>   // END: Determine startup state
> 
>   // Buffer near and far end signals
>   memcpy(aecm->xBuf + PART_LEN, farend, sizeof(int16_t) * PART_LEN);
>   memcpy(aecm->dBufNoisy + PART_LEN, nearendNoisy, sizeof(int16_t) * PART_LEN);
>-  if (nearendClean != NULL)
>-  {
>-    memcpy(aecm->dBufClean + PART_LEN,
>-           nearendClean,
>+  if (nearendClean != NULL) {
>+    memcpy(aecm->dBufClean + PART_LEN, nearendClean,
>            sizeof(int16_t) * PART_LEN);
>   }
> 
>   // Transform far end signal from time domain to frequency domain.
>-  far_q = TimeToFrequencyDomain(aecm,
>-                                aecm->xBuf,
>-                                dfw,
>-                                xfa,
>-                                &xfaSum);
>+  far_q = TimeToFrequencyDomain(aecm, aecm->xBuf, dfw, xfa, &xfaSum);
> 
>   // Transform noisy near end signal from time domain to frequency domain.
>-  zerosDBufNoisy = TimeToFrequencyDomain(aecm,
>-                                         aecm->dBufNoisy,
>-                                         dfw,
>-                                         dfaNoisy,
>-                                         &dfaNoisySum);
>+  zerosDBufNoisy =
>+      TimeToFrequencyDomain(aecm, aecm->dBufNoisy, dfw, dfaNoisy, &dfaNoisySum);
>   aecm->dfaNoisyQDomainOld = aecm->dfaNoisyQDomain;
>   aecm->dfaNoisyQDomain = (int16_t)zerosDBufNoisy;
> 
>-
>-  if (nearendClean == NULL)
>-  {
>+  if (nearendClean == NULL) {
>     ptrDfaClean = dfaNoisy;
>     aecm->dfaCleanQDomainOld = aecm->dfaNoisyQDomainOld;
>     aecm->dfaCleanQDomain = aecm->dfaNoisyQDomain;
>     dfaCleanSum = dfaNoisySum;
>-  } else
>-  {
>+  } else {
>     // Transform clean near end signal from time domain to frequency domain.
>-    zerosDBufClean = TimeToFrequencyDomain(aecm,
>-                                           aecm->dBufClean,
>-                                           dfw,
>-                                           dfaClean,
>+    zerosDBufClean = TimeToFrequencyDomain(aecm, aecm->dBufClean, dfw, dfaClean,
>                                            &dfaCleanSum);
>     aecm->dfaCleanQDomainOld = aecm->dfaCleanQDomain;
>     aecm->dfaCleanQDomain = (int16_t)zerosDBufClean;
>@@ -389,46 +350,34 @@ WebRtcAecm_ProcessBlock(AecmCore* aecm,
>   // Get the delay
>   // Save far-end history and estimate delay
>   WebRtcAecm_UpdateFarHistory(aecm, xfa, far_q);
>-  if (WebRtc_AddFarSpectrumFix(aecm->delay_estimator_farend,
>-                               xfa,
>-                               PART_LEN1,
>+  if (WebRtc_AddFarSpectrumFix(aecm->delay_estimator_farend, xfa, PART_LEN1,
>                                far_q) == -1) {
>     return -1;
>   }
>-  delay = WebRtc_DelayEstimatorProcessFix(aecm->delay_estimator,
>-                                          dfaNoisy,
>-                                          PART_LEN1,
>-                                          zerosDBufNoisy);
>-  if (delay == -1)
>-  {
>+  delay = WebRtc_DelayEstimatorProcessFix(aecm->delay_estimator, dfaNoisy,
>+                                          PART_LEN1, zerosDBufNoisy);
>+  if (delay == -1) {
>     return -1;
>-  }
>-  else if (delay == -2)
>-  {
>+  } else if (delay == -2) {
>     // If the delay is unknown, we assume zero.
>     // NOTE: this will have to be adjusted if we ever add lookahead.
>     delay = 0;
>   }
> 
>-  if (aecm->fixedDelay >= 0)
>-  {
>+  if (aecm->fixedDelay >= 0) {
>     // Use fixed delay
>     delay = aecm->fixedDelay;
>   }
> 
>   // Get aligned far end spectrum
>   far_spectrum_ptr = WebRtcAecm_AlignedFarend(aecm, &far_q, delay);
>-  zerosXBuf = (int16_t) far_q;
>-  if (far_spectrum_ptr == NULL)
>-  {
>+  zerosXBuf = (int16_t)far_q;
>+  if (far_spectrum_ptr == NULL) {
>     return -1;
>   }
> 
>   // Calculate log(energy) and update energy threshold levels
>-  WebRtcAecm_CalcEnergies(aecm,
>-                          far_spectrum_ptr,
>-                          zerosXBuf,
>-                          dfaNoisySum,
>+  WebRtcAecm_CalcEnergies(aecm, far_spectrum_ptr, zerosXBuf, dfaNoisySum,
>                           echoEst32);
> 
>   // Calculate stepsize
>@@ -440,18 +389,12 @@ WebRtcAecm_ProcessBlock(AecmCore* aecm,
>   // This is the channel estimation algorithm.
>   // It is base on NLMS but has a variable step length,
>   // which was calculated above.
>-  WebRtcAecm_UpdateChannel(aecm,
>-                           far_spectrum_ptr,
>-                           zerosXBuf,
>-                           dfaNoisy,
>-                           mu,
>+  WebRtcAecm_UpdateChannel(aecm, far_spectrum_ptr, zerosXBuf, dfaNoisy, mu,
>                            echoEst32);
>   supGain = WebRtcAecm_CalcSuppressionGain(aecm);
> 
>-
>   // Calculate Wiener filter hnl[]
>-  for (i = 0; i < PART_LEN1; i++)
>-  {
>+  for (i = 0; i < PART_LEN1; i++) {
>     // Far end signal through channel estimate in Q8
>     // How much can we shift right to preserve resolution
>     tmp32no1 = echoEst32[i] - aecm->echoFilt[i];
>@@ -460,28 +403,24 @@ WebRtcAecm_ProcessBlock(AecmCore* aecm,
> 
>     zeros32 = WebRtcSpl_NormW32(aecm->echoFilt[i]) + 1;
>     zeros16 = WebRtcSpl_NormW16(supGain) + 1;
>-    if (zeros32 + zeros16 > 16)
>-    {
>+    if (zeros32 + zeros16 > 16) {
>       // Multiplication is safe
>       // Result in
>       // Q(RESOLUTION_CHANNEL+RESOLUTION_SUPGAIN+
>       //   aecm->xfaQDomainBuf[diff])
>-      echoEst32Gained = WEBRTC_SPL_UMUL_32_16((uint32_t)aecm->echoFilt[i],
>-                                              (uint16_t)supGain);
>+      echoEst32Gained =
>+          WEBRTC_SPL_UMUL_32_16((uint32_t)aecm->echoFilt[i], (uint16_t)supGain);
>       resolutionDiff = 14 - RESOLUTION_CHANNEL16 - RESOLUTION_SUPGAIN;
>       resolutionDiff += (aecm->dfaCleanQDomain - zerosXBuf);
>-    } else
>-    {
>+    } else {
>       tmp16no1 = 17 - zeros32 - zeros16;
>-      resolutionDiff = 14 + tmp16no1 - RESOLUTION_CHANNEL16 -
>-                       RESOLUTION_SUPGAIN;
>+      resolutionDiff =
>+          14 + tmp16no1 - RESOLUTION_CHANNEL16 - RESOLUTION_SUPGAIN;
>       resolutionDiff += (aecm->dfaCleanQDomain - zerosXBuf);
>-      if (zeros32 > tmp16no1)
>-      {
>+      if (zeros32 > tmp16no1) {
>         echoEst32Gained = WEBRTC_SPL_UMUL_32_16((uint32_t)aecm->echoFilt[i],
>                                                 supGain >> tmp16no1);
>-      } else
>-      {
>+      } else {
>         // Result in Q-(RESOLUTION_CHANNEL+RESOLUTION_SUPGAIN-16)
>         echoEst32Gained = (aecm->echoFilt[i] >> tmp16no1) * supGain;
>       }
>@@ -491,13 +430,13 @@ WebRtcAecm_ProcessBlock(AecmCore* aecm,
>     RTC_DCHECK_GE(zeros16, 0);  // |zeros16| is a norm, hence non-negative.
>     dfa_clean_q_domain_diff = aecm->dfaCleanQDomain - aecm->dfaCleanQDomainOld;
>     if (zeros16 < dfa_clean_q_domain_diff && aecm->nearFilt[i]) {
>-      tmp16no1 = aecm->nearFilt[i] << zeros16;
>+      tmp16no1 = aecm->nearFilt[i] * (1 << zeros16);
>       qDomainDiff = zeros16 - dfa_clean_q_domain_diff;
>       tmp16no2 = ptrDfaClean[i] >> -qDomainDiff;
>     } else {
>       tmp16no1 = dfa_clean_q_domain_diff < 0
>-          ? aecm->nearFilt[i] >> -dfa_clean_q_domain_diff
>-          : aecm->nearFilt[i] << dfa_clean_q_domain_diff;
>+                     ? aecm->nearFilt[i] >> -dfa_clean_q_domain_diff
>+                     : aecm->nearFilt[i] * (1 << dfa_clean_q_domain_diff);
>       qDomainDiff = 0;
>       tmp16no2 = ptrDfaClean[i];
>     }
>@@ -508,130 +447,105 @@ WebRtcAecm_ProcessBlock(AecmCore* aecm,
>     if ((tmp16no2) & (-qDomainDiff > zeros16)) {
>       aecm->nearFilt[i] = WEBRTC_SPL_WORD16_MAX;
>     } else {
>-      aecm->nearFilt[i] = qDomainDiff < 0 ? tmp16no2 << -qDomainDiff
>+      aecm->nearFilt[i] = qDomainDiff < 0 ? tmp16no2 * (1 << -qDomainDiff)
>                                           : tmp16no2 >> qDomainDiff;
>     }
> 
>     // Wiener filter coefficients, resulting hnl in Q14
>-    if (echoEst32Gained == 0)
>-    {
>+    if (echoEst32Gained == 0) {
>       hnl[i] = ONE_Q14;
>-    } else if (aecm->nearFilt[i] == 0)
>-    {
>+    } else if (aecm->nearFilt[i] == 0) {
>       hnl[i] = 0;
>-    } else
>-    {
>+    } else {
>       // Multiply the suppression gain
>       // Rounding
>       echoEst32Gained += (uint32_t)(aecm->nearFilt[i] >> 1);
>-      tmpU32 = WebRtcSpl_DivU32U16(echoEst32Gained,
>-                                   (uint16_t)aecm->nearFilt[i]);
>+      tmpU32 =
>+          WebRtcSpl_DivU32U16(echoEst32Gained, (uint16_t)aecm->nearFilt[i]);
> 
>       // Current resolution is
>       // Q-(RESOLUTION_CHANNEL+RESOLUTION_SUPGAIN- max(0,17-zeros16- zeros32))
>       // Make sure we are in Q14
>       tmp32no1 = (int32_t)WEBRTC_SPL_SHIFT_W32(tmpU32, resolutionDiff);
>-      if (tmp32no1 > ONE_Q14)
>-      {
>+      if (tmp32no1 > ONE_Q14) {
>         hnl[i] = 0;
>-      } else if (tmp32no1 < 0)
>-      {
>+      } else if (tmp32no1 < 0) {
>         hnl[i] = ONE_Q14;
>-      } else
>-      {
>+      } else {
>         // 1-echoEst/dfa
>         hnl[i] = ONE_Q14 - (int16_t)tmp32no1;
>-        if (hnl[i] < 0)
>-        {
>+        if (hnl[i] < 0) {
>           hnl[i] = 0;
>         }
>       }
>     }
>-    if (hnl[i])
>-    {
>+    if (hnl[i]) {
>       numPosCoef++;
>     }
>   }
>   // Only in wideband. Prevent the gain in upper band from being larger than
>   // in lower band.
>-  if (aecm->mult == 2)
>-  {
>+  if (aecm->mult == 2) {
>     // TODO(bjornv): Investigate if the scaling of hnl[i] below can cause
>     //               speech distortion in double-talk.
>-    for (i = 0; i < PART_LEN1; i++)
>-    {
>+    for (i = 0; i < PART_LEN1; i++) {
>       hnl[i] = (int16_t)((hnl[i] * hnl[i]) >> 14);
>     }
> 
>-    for (i = kMinPrefBand; i <= kMaxPrefBand; i++)
>-    {
>+    for (i = kMinPrefBand; i <= kMaxPrefBand; i++) {
>       avgHnl32 += (int32_t)hnl[i];
>     }
>     RTC_DCHECK_GT(kMaxPrefBand - kMinPrefBand + 1, 0);
>     avgHnl32 /= (kMaxPrefBand - kMinPrefBand + 1);
> 
>-    for (i = kMaxPrefBand; i < PART_LEN1; i++)
>-    {
>-      if (hnl[i] > (int16_t)avgHnl32)
>-      {
>+    for (i = kMaxPrefBand; i < PART_LEN1; i++) {
>+      if (hnl[i] > (int16_t)avgHnl32) {
>         hnl[i] = (int16_t)avgHnl32;
>       }
>     }
>   }
> 
>   // Calculate NLP gain, result is in Q14
>-  if (aecm->nlpFlag)
>-  {
>-    for (i = 0; i < PART_LEN1; i++)
>-    {
>+  if (aecm->nlpFlag) {
>+    for (i = 0; i < PART_LEN1; i++) {
>       // Truncate values close to zero and one.
>-      if (hnl[i] > NLP_COMP_HIGH)
>-      {
>+      if (hnl[i] > NLP_COMP_HIGH) {
>         hnl[i] = ONE_Q14;
>-      } else if (hnl[i] < NLP_COMP_LOW)
>-      {
>+      } else if (hnl[i] < NLP_COMP_LOW) {
>         hnl[i] = 0;
>       }
> 
>       // Remove outliers
>-      if (numPosCoef < 3)
>-      {
>+      if (numPosCoef < 3) {
>         nlpGain = 0;
>-      } else
>-      {
>+      } else {
>         nlpGain = ONE_Q14;
>       }
> 
>       // NLP
>-      if ((hnl[i] == ONE_Q14) && (nlpGain == ONE_Q14))
>-      {
>+      if ((hnl[i] == ONE_Q14) && (nlpGain == ONE_Q14)) {
>         hnl[i] = ONE_Q14;
>-      } else
>-      {
>+      } else {
>         hnl[i] = (int16_t)((hnl[i] * nlpGain) >> 14);
>       }
> 
>       // multiply with Wiener coefficients
>-      efw[i].real = (int16_t)(WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].real,
>-                                                                   hnl[i], 14));
>-      efw[i].imag = (int16_t)(WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].imag,
>-                                                                   hnl[i], 14));
>+      efw[i].real = (int16_t)(
>+          WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].real, hnl[i], 14));
>+      efw[i].imag = (int16_t)(
>+          WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].imag, hnl[i], 14));
>     }
>-  }
>-  else
>-  {
>+  } else {
>     // multiply with Wiener coefficients
>-    for (i = 0; i < PART_LEN1; i++)
>-    {
>-      efw[i].real = (int16_t)(WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].real,
>-                                                                   hnl[i], 14));
>-      efw[i].imag = (int16_t)(WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].imag,
>-                                                                   hnl[i], 14));
>+    for (i = 0; i < PART_LEN1; i++) {
>+      efw[i].real = (int16_t)(
>+          WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].real, hnl[i], 14));
>+      efw[i].imag = (int16_t)(
>+          WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].imag, hnl[i], 14));
>     }
>   }
> 
>-  if (aecm->cngMode == AecmTrue)
>-  {
>+  if (aecm->cngMode == AecmTrue) {
>     ComfortNoise(aecm, ptrDfaClean, efw, hnl);
>   }
> 
>@@ -660,83 +574,66 @@ static void ComfortNoise(AecmCore* aecm,
>   RTC_DCHECK_GE(shiftFromNearToNoise, 0);
>   RTC_DCHECK_LT(shiftFromNearToNoise, 16);
> 
>-  if (aecm->noiseEstCtr < 100)
>-  {
>+  if (aecm->noiseEstCtr < 100) {
>     // Track the minimum more quickly initially.
>     aecm->noiseEstCtr++;
>     minTrackShift = 6;
>-  } else
>-  {
>+  } else {
>     minTrackShift = 9;
>   }
> 
>   // Estimate noise power.
>-  for (i = 0; i < PART_LEN1; i++)
>-  {
>+  for (i = 0; i < PART_LEN1; i++) {
>     // Shift to the noise domain.
>     tmp32 = (int32_t)dfa[i];
>     outLShift32 = tmp32 << shiftFromNearToNoise;
> 
>-    if (outLShift32 < aecm->noiseEst[i])
>-    {
>+    if (outLShift32 < aecm->noiseEst[i]) {
>       // Reset "too low" counter
>       aecm->noiseEstTooLowCtr[i] = 0;
>       // Track the minimum.
>-      if (aecm->noiseEst[i] < (1 << minTrackShift))
>-      {
>+      if (aecm->noiseEst[i] < (1 << minTrackShift)) {
>         // For small values, decrease noiseEst[i] every
>         // |kNoiseEstIncCount| block. The regular approach below can not
>         // go further down due to truncation.
>         aecm->noiseEstTooHighCtr[i]++;
>-        if (aecm->noiseEstTooHighCtr[i] >= kNoiseEstIncCount)
>-        {
>+        if (aecm->noiseEstTooHighCtr[i] >= kNoiseEstIncCount) {
>           aecm->noiseEst[i]--;
>-          aecm->noiseEstTooHighCtr[i] = 0; // Reset the counter
>+          aecm->noiseEstTooHighCtr[i] = 0;  // Reset the counter
>         }
>+      } else {
>+        aecm->noiseEst[i] -=
>+            ((aecm->noiseEst[i] - outLShift32) >> minTrackShift);
>       }
>-      else
>-      {
>-        aecm->noiseEst[i] -= ((aecm->noiseEst[i] - outLShift32)
>-                              >> minTrackShift);
>-      }
>-    } else
>-    {
>+    } else {
>       // Reset "too high" counter
>       aecm->noiseEstTooHighCtr[i] = 0;
>       // Ramp slowly upwards until we hit the minimum again.
>-      if ((aecm->noiseEst[i] >> 19) > 0)
>-      {
>+      if ((aecm->noiseEst[i] >> 19) > 0) {
>         // Avoid overflow.
>         // Multiplication with 2049 will cause wrap around. Scale
>         // down first and then multiply
>         aecm->noiseEst[i] >>= 11;
>         aecm->noiseEst[i] *= 2049;
>-      }
>-      else if ((aecm->noiseEst[i] >> 11) > 0)
>-      {
>+      } else if ((aecm->noiseEst[i] >> 11) > 0) {
>         // Large enough for relative increase
>         aecm->noiseEst[i] *= 2049;
>         aecm->noiseEst[i] >>= 11;
>-      }
>-      else
>-      {
>+      } else {
>         // Make incremental increases based on size every
>         // |kNoiseEstIncCount| block
>         aecm->noiseEstTooLowCtr[i]++;
>-        if (aecm->noiseEstTooLowCtr[i] >= kNoiseEstIncCount)
>-        {
>+        if (aecm->noiseEstTooLowCtr[i] >= kNoiseEstIncCount) {
>           aecm->noiseEst[i] += (aecm->noiseEst[i] >> 9) + 1;
>-          aecm->noiseEstTooLowCtr[i] = 0; // Reset counter
>+          aecm->noiseEstTooLowCtr[i] = 0;  // Reset counter
>         }
>       }
>     }
>   }
> 
>-  for (i = 0; i < PART_LEN1; i++)
>-  {
>+  for (i = 0; i < PART_LEN1; i++) {
>     tmp32 = aecm->noiseEst[i] >> shiftFromNearToNoise;
>-    if (tmp32 > 32767)
>-    {
>+    if (tmp32 > 32767) {
>       tmp32 = 32767;
>       aecm->noiseEst[i] = tmp32 << shiftFromNearToNoise;
>     }
>@@ -750,23 +647,21 @@ static void ComfortNoise(AecmCore* aecm,
>   WebRtcSpl_RandUArray(randW16, PART_LEN, &aecm->seed);
> 
>   // Generate noise according to estimated energy.
>-  uReal[0] = 0; // Reject LF noise.
>+  uReal[0] = 0;  // Reject LF noise.
>   uImag[0] = 0;
>-  for (i = 1; i < PART_LEN1; i++)
>-  {
>+  for (i = 1; i < PART_LEN1; i++) {
>     // Get a random index for the cos and sin tables over [0 359].
>     tmp16 = (int16_t)((359 * randW16[i - 1]) >> 15);
> 
>     // Tables are in Q13.
>-    uReal[i] = (int16_t)((noiseRShift16[i] * WebRtcAecm_kCosTable[tmp16]) >>
>-        13);
>-    uImag[i] = (int16_t)((-noiseRShift16[i] * WebRtcAecm_kSinTable[tmp16]) >>
>-        13);
>+    uReal[i] =
>+        (int16_t)((noiseRShift16[i] * WebRtcAecm_kCosTable[tmp16]) >> 13);
>+    uImag[i] =
>+        (int16_t)((-noiseRShift16[i] * WebRtcAecm_kSinTable[tmp16]) >> 13);
>   }
>   uImag[PART_LEN] = 0;
> 
>-  for (i = 0; i < PART_LEN1; i++)
>-  {
>+  for (i = 0; i < PART_LEN1; i++) {
>     out[i].real = WebRtcSpl_AddSatW16(out[i].real, uReal[i]);
>     out[i].imag = WebRtcSpl_AddSatW16(out[i].imag, uImag[i]);
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aecm/aecm_core_mips.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aecm/aecm_core_mips.cc
>index 58e5ec5e358..11e4095ef58 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aecm/aecm_core_mips.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aecm/aecm_core_mips.cc
>@@ -16,56 +16,37 @@
> #include "rtc_base/numerics/safe_conversions.h"
> 
> static const ALIGN8_BEG int16_t WebRtcAecm_kSqrtHanning[] ALIGN8_END = {
>-  0, 399, 798, 1196, 1594, 1990, 2386, 2780, 3172,
>-  3562, 3951, 4337, 4720, 5101, 5478, 5853, 6224,
>-  6591, 6954, 7313, 7668, 8019, 8364, 8705, 9040,
>-  9370, 9695, 10013, 10326, 10633, 10933, 11227, 11514,
>-  11795, 12068, 12335, 12594, 12845, 13089, 13325, 13553,
>-  13773, 13985, 14189, 14384, 14571, 14749, 14918, 15079,
>-  15231, 15373, 15506, 15631, 15746, 15851, 15947, 16034,
>-  16111, 16179, 16237, 16286, 16325, 16354, 16373, 16384
>-};
>+    0,     399,   798,   1196,  1594,  1990,  2386,  2780,  3172,  3562,  3951,
>+    4337,  4720,  5101,  5478,  5853,  6224,  6591,  6954,  7313,  7668,  8019,
>+    8364,  8705,  9040,  9370,  9695,  10013, 10326, 10633, 10933, 11227, 11514,
>+    11795, 12068, 12335, 12594, 12845, 13089, 13325, 13553, 13773, 13985, 14189,
>+    14384, 14571, 14749, 14918, 15079, 15231, 15373, 15506, 15631, 15746, 15851,
>+    15947, 16034, 16111, 16179, 16237, 16286, 16325, 16354, 16373, 16384};
> 
> static const int16_t kNoiseEstQDomain = 15;
> static const int16_t kNoiseEstIncCount = 5;
> 
> static int16_t coefTable[] = {
>-   0,   4, 256, 260, 128, 132, 384, 388,
>-  64,  68, 320, 324, 192, 196, 448, 452,
>-  32,  36, 288, 292, 160, 164, 416, 420,
>-  96, 100, 352, 356, 224, 228, 480, 484,
>-  16,  20, 272, 276, 144, 148, 400, 404,
>-  80,  84, 336, 340, 208, 212, 464, 468,
>-  48,  52, 304, 308, 176, 180, 432, 436,
>- 112, 116, 368, 372, 240, 244, 496, 500,
>-   8,  12, 264, 268, 136, 140, 392, 396,
>-  72,  76, 328, 332, 200, 204, 456, 460,
>-  40,  44, 296, 300, 168, 172, 424, 428,
>- 104, 108, 360, 364, 232, 236, 488, 492,
>-  24,  28, 280, 284, 152, 156, 408, 412,
>-  88,  92, 344, 348, 216, 220, 472, 476,
>-  56,  60, 312, 316, 184, 188, 440, 444,
>- 120, 124, 376, 380, 248, 252, 504, 508
>-};
>+    0,   4,   256, 260, 128, 132, 384, 388, 64,  68,  320, 324, 192, 196, 448,
>+    452, 32,  36,  288, 292, 160, 164, 416, 420, 96,  100, 352, 356, 224, 228,
>+    480, 484, 16,  20,  272, 276, 144, 148, 400, 404, 80,  84,  336, 340, 208,
>+    212, 464, 468, 48,  52,  304, 308, 176, 180, 432, 436, 112, 116, 368, 372,
>+    240, 244, 496, 500, 8,   12,  264, 268, 136, 140, 392, 396, 72,  76,  328,
>+    332, 200, 204, 456, 460, 40,  44,  296, 300, 168, 172, 424, 428, 104, 108,
>+    360, 364, 232, 236, 488, 492, 24,  28,  280, 284, 152, 156, 408, 412, 88,
>+    92,  344, 348, 216, 220, 472, 476, 56,  60,  312, 316, 184, 188, 440, 444,
>+    120, 124, 376, 380, 248, 252, 504, 508};
> 
> static int16_t coefTable_ifft[] = {
>-    0, 512, 256, 508, 128, 252, 384, 380,
>-   64, 124, 320, 444, 192, 188, 448, 316,
>-   32,  60, 288, 476, 160, 220, 416, 348,
>-   96,  92, 352, 412, 224, 156, 480, 284,
>-   16,  28, 272, 492, 144, 236, 400, 364,
>-   80, 108, 336, 428, 208, 172, 464, 300,
>-   48,  44, 304, 460, 176, 204, 432, 332,
>-  112,  76, 368, 396, 240, 140, 496, 268,
>-    8,  12, 264, 500, 136, 244, 392, 372,
>-   72, 116, 328, 436, 200, 180, 456, 308,
>-   40,  52, 296, 468, 168, 212, 424, 340,
>-  104,  84, 360, 404, 232, 148, 488, 276,
>-   24,  20, 280, 484, 152, 228, 408, 356,
>-   88, 100, 344, 420, 216, 164, 472, 292,
>-   56,  36, 312, 452, 184, 196, 440, 324,
>-  120,  68, 376, 388, 248, 132, 504, 260
>-};
>+    0,   512, 256, 508, 128, 252, 384, 380, 64,  124, 320, 444, 192, 188, 448,
>+    316, 32,  60,  288, 476, 160, 220, 416, 348, 96,  92,  352, 412, 224, 156,
>+    480, 284, 16,  28,  272, 492, 144, 236, 400, 364, 80,  108, 336, 428, 208,
>+    172, 464, 300, 48,  44,  304, 460, 176, 204, 432, 332, 112, 76,  368, 396,
>+    240, 140, 496, 268, 8,   12,  264, 500, 136, 244, 392, 372, 72,  116, 328,
>+    436, 200, 180, 456, 308, 40,  52,  296, 468, 168, 212, 424, 340, 104, 84,
>+    360, 404, 232, 148, 488, 276, 24,  20,  280, 484, 152, 228, 408, 356, 88,
>+    100, 344, 420, 216, 164, 472, 292, 56,  36,  312, 452, 184, 196, 440, 324,
>+    120, 68,  376, 388, 248, 132, 504, 260};
> 
> static void ComfortNoise(AecmCore* aecm,
>                          const uint16_t* dfa,
>@@ -81,122 +62,152 @@ static void WindowAndFFT(AecmCore* aecm,
>   int32_t tmp1, tmp2, tmp3, tmp4;
>   int16_t* pfrfi;
>   ComplexInt16* pfreq_signal;
>-  int16_t  f_coef, s_coef;
>+  int16_t f_coef, s_coef;
>   int32_t load_ptr, store_ptr1, store_ptr2, shift, shift1;
>   int32_t hann, hann1, coefs;
> 
>   memset(fft, 0, sizeof(int16_t) * PART_LEN4);
> 
>   // FFT of signal
>-  __asm __volatile (
>-    ".set        push                                                    \n\t"
>-    ".set        noreorder                                               \n\t"
>-    "addiu       %[shift],          %[time_signal_scaling], -14          \n\t"
>-    "addiu       %[i],              $zero,                  64           \n\t"
>-    "addiu       %[load_ptr],       %[time_signal],         0            \n\t"
>-    "addiu       %[hann],           %[hanning],             0            \n\t"
>-    "addiu       %[hann1],          %[hanning],             128          \n\t"
>-    "addiu       %[coefs],          %[coefTable],           0            \n\t"
>-    "bltz        %[shift],          2f                                   \n\t"
>-    " negu       %[shift1],         %[shift]                             \n\t"
>-   "1:                                                                   \n\t"
>-    "lh          %[tmp1],           0(%[load_ptr])                       \n\t"
>-    "lh          %[tmp2],           0(%[hann])                           \n\t"
>-    "lh          %[tmp3],           128(%[load_ptr])                     \n\t"
>-    "lh          %[tmp4],           0(%[hann1])                          \n\t"
>-    "addiu       %[i],              %[i],                   -1           \n\t"
>-    "mul         %[tmp1],           %[tmp1],                %[tmp2]      \n\t"
>-    "mul         %[tmp3],           %[tmp3],                %[tmp4]      \n\t"
>-    "lh          %[f_coef],         0(%[coefs])                          \n\t"
>-    "lh          %[s_coef],         2(%[coefs])                          \n\t"
>-    "addiu       %[load_ptr],       %[load_ptr],            2            \n\t"
>-    "addiu       %[hann],           %[hann],                2            \n\t"
>-    "addiu       %[hann1],          %[hann1],               -2           \n\t"
>-    "addu        %[store_ptr1],     %[fft],                 %[f_coef]    \n\t"
>-    "addu        %[store_ptr2],     %[fft],                 %[s_coef]    \n\t"
>-    "sllv        %[tmp1],           %[tmp1],                %[shift]     \n\t"
>-    "sllv        %[tmp3],           %[tmp3],                %[shift]     \n\t"
>-    "sh          %[tmp1],           0(%[store_ptr1])                     \n\t"
>-    "sh          %[tmp3],           0(%[store_ptr2])                     \n\t"
>-    "bgtz        %[i],              1b                                   \n\t"
>-    " addiu      %[coefs],          %[coefs],               4            \n\t"
>-    "b           3f                                                      \n\t"
>-    " nop                                                                \n\t"
>-   "2:                                                                   \n\t"
>-    "lh          %[tmp1],           0(%[load_ptr])                       \n\t"
>-    "lh          %[tmp2],           0(%[hann])                           \n\t"
>-    "lh          %[tmp3],           128(%[load_ptr])                     \n\t"
>-    "lh          %[tmp4],           0(%[hann1])                          \n\t"
>-    "addiu       %[i],              %[i],                   -1           \n\t"
>-    "mul         %[tmp1],           %[tmp1],                %[tmp2]      \n\t"
>-    "mul         %[tmp3],           %[tmp3],                %[tmp4]      \n\t"
>-    "lh          %[f_coef],         0(%[coefs])                          \n\t"
>-    "lh          %[s_coef],         2(%[coefs])                          \n\t"
>-    "addiu       %[load_ptr],       %[load_ptr],            2            \n\t"
>-    "addiu       %[hann],           %[hann],                2            \n\t"
>-    "addiu       %[hann1],          %[hann1],               -2           \n\t"
>-    "addu        %[store_ptr1],     %[fft],                 %[f_coef]    \n\t"
>-    "addu        %[store_ptr2],     %[fft],                 %[s_coef]    \n\t"
>-    "srav        %[tmp1],           %[tmp1],                %[shift1]    \n\t"
>-    "srav        %[tmp3],           %[tmp3],                %[shift1]    \n\t"
>-    "sh          %[tmp1],           0(%[store_ptr1])                     \n\t"
>-    "sh          %[tmp3],           0(%[store_ptr2])                     \n\t"
>-    "bgtz        %[i],              2b                                   \n\t"
>-    " addiu      %[coefs],          %[coefs],               4            \n\t"
>-   "3:                                                                   \n\t"
>-    ".set        pop                                                     \n\t"
>-    : [load_ptr] "=&r" (load_ptr), [shift] "=&r" (shift), [hann] "=&r" (hann),
>-      [hann1] "=&r" (hann1), [shift1] "=&r" (shift1), [coefs] "=&r" (coefs),
>-      [tmp1] "=&r" (tmp1), [tmp2] "=&r" (tmp2), [tmp3] "=&r" (tmp3),
>-      [tmp4] "=&r" (tmp4), [i] "=&r" (i), [f_coef] "=&r" (f_coef),
>-      [s_coef] "=&r" (s_coef), [store_ptr1] "=&r" (store_ptr1),
>-      [store_ptr2] "=&r" (store_ptr2)
>-    : [time_signal] "r" (time_signal), [coefTable] "r" (coefTable),
>-      [time_signal_scaling] "r" (time_signal_scaling),
>-      [hanning] "r" (WebRtcAecm_kSqrtHanning), [fft] "r" (fft)
>-    : "memory", "hi", "lo"
>-  );
>+  __asm __volatile(
>+      ".set        push                                                    \n\t"
>+      ".set        noreorder                                               \n\t"
>+      "addiu       %[shift],          %[time_signal_scaling], -14          \n\t"
>+      "addiu       %[i],              $zero,                  64           \n\t"
>+      "addiu       %[load_ptr],       %[time_signal],         0            \n\t"
>+      "addiu       %[hann],           %[hanning],             0            \n\t"
>+      "addiu       %[hann1],          %[hanning],             128          \n\t"
>+      "addiu       %[coefs],          %[coefTable],           0            \n\t"
>+      "bltz        %[shift],          2f                                   \n\t"
>+      " negu       %[shift1],         %[shift]                             \n\t"
>+      "1:                                                                   "
>+      "\n\t"
>+      "lh          %[tmp1],           0(%[load_ptr])                       \n\t"
>+      "lh          %[tmp2],           0(%[hann])                           \n\t"
>+      "lh          %[tmp3],           128(%[load_ptr])                     \n\t"
>+      "lh          %[tmp4],           0(%[hann1])                          \n\t"
>+      "addiu       %[i],              %[i],                   -1           \n\t"
>+      "mul         %[tmp1],           %[tmp1],                %[tmp2]      \n\t"
>+      "mul         %[tmp3],           %[tmp3],                %[tmp4]      \n\t"
>+      "lh          %[f_coef],         0(%[coefs])                          \n\t"
>+      "lh          %[s_coef],         2(%[coefs])                          \n\t"
>+      "addiu       %[load_ptr],       %[load_ptr],            2            \n\t"
>+      "addiu       %[hann],           %[hann],                2            \n\t"
>+      "addiu       %[hann1],          %[hann1],               -2           \n\t"
>+      "addu        %[store_ptr1],     %[fft],                 %[f_coef]    \n\t"
>+      "addu        %[store_ptr2],     %[fft],                 %[s_coef]    \n\t"
>+      "sllv        %[tmp1],           %[tmp1],                %[shift]     \n\t"
>+      "sllv        %[tmp3],           %[tmp3],                %[shift]     \n\t"
>+      "sh          %[tmp1],           0(%[store_ptr1])                     \n\t"
>+      "sh          %[tmp3],           0(%[store_ptr2])                     \n\t"
>+      "bgtz        %[i],              1b                                   \n\t"
>+      " addiu      %[coefs],          %[coefs],               4            \n\t"
>+      "b           3f                                                      \n\t"
>+      " nop                                                                \n\t"
>+      "2:                                                                   "
>+      "\n\t"
>+      "lh          %[tmp1],           0(%[load_ptr])                       \n\t"
>+      "lh          %[tmp2],           0(%[hann])                           \n\t"
>+      "lh          %[tmp3],           128(%[load_ptr])                     \n\t"
>+      "lh          %[tmp4],           0(%[hann1])                          \n\t"
>+      "addiu       %[i],              %[i],                   -1           \n\t"
>+      "mul         %[tmp1],           %[tmp1],                %[tmp2]      \n\t"
>+      "mul         %[tmp3],           %[tmp3],                %[tmp4]      \n\t"
>+      "lh          %[f_coef],         0(%[coefs])                          \n\t"
>+      "lh          %[s_coef],         2(%[coefs])                          \n\t"
>+      "addiu       %[load_ptr],       %[load_ptr],            2            \n\t"
>+      "addiu       %[hann],           %[hann],                2            \n\t"
>+      "addiu       %[hann1],          %[hann1],               -2           \n\t"
>+      "addu        %[store_ptr1],     %[fft],                 %[f_coef]    \n\t"
>+      "addu        %[store_ptr2],     %[fft],                 %[s_coef]    \n\t"
>+      "srav        %[tmp1],           %[tmp1],                %[shift1]    \n\t"
>+      "srav        %[tmp3],           %[tmp3],                %[shift1]    \n\t"
>+      "sh          %[tmp1],           0(%[store_ptr1])                     \n\t"
>+      "sh          %[tmp3],           0(%[store_ptr2])                     \n\t"
>+      "bgtz        %[i],              2b                                   \n\t"
>+      " addiu      %[coefs],          %[coefs],               4            \n\t"
>+      "3:                                                                   "
>+      "\n\t"
>+      ".set        pop                                                     \n\t"
>+      : [load_ptr] "=&r"(load_ptr), [shift] "=&r"(shift), [hann] "=&r"(hann),
>+        [hann1] "=&r"(hann1), [shift1] "=&r"(shift1), [coefs] "=&r"(coefs),
>+        [tmp1] "=&r"(tmp1), [tmp2] "=&r"(tmp2), [tmp3] "=&r"(tmp3),
>+        [tmp4] "=&r"(tmp4), [i] "=&r"(i), [f_coef] "=&r"(f_coef),
>+        [s_coef] "=&r"(s_coef), [store_ptr1] "=&r"(store_ptr1),
>+        [store_ptr2] "=&r"(store_ptr2)
>+      : [time_signal] "r"(time_signal), [coefTable] "r"(coefTable),
>+        [time_signal_scaling] "r"(time_signal_scaling),
>+        [hanning] "r"(WebRtcAecm_kSqrtHanning), [fft] "r"(fft)
>+      : "memory", "hi", "lo");
> 
>   WebRtcSpl_ComplexFFT(fft, PART_LEN_SHIFT, 1);
>   pfrfi = fft;
>   pfreq_signal = freq_signal;
> 
>-  __asm __volatile (
>-    ".set        push                                                     \n\t"
>-    ".set        noreorder                                                \n\t"
>-    "addiu       %[j],              $zero,                 128            \n\t"
>-   "1:                                                                    \n\t"
>-    "lh          %[tmp1],           0(%[pfrfi])                           \n\t"
>-    "lh          %[tmp2],           2(%[pfrfi])                           \n\t"
>-    "lh          %[tmp3],           4(%[pfrfi])                           \n\t"
>-    "lh          %[tmp4],           6(%[pfrfi])                           \n\t"
>-    "subu        %[tmp2],           $zero,                 %[tmp2]        \n\t"
>-    "sh          %[tmp1],           0(%[pfreq_signal])                    \n\t"
>-    "sh          %[tmp2],           2(%[pfreq_signal])                    \n\t"
>-    "subu        %[tmp4],           $zero,                 %[tmp4]        \n\t"
>-    "sh          %[tmp3],           4(%[pfreq_signal])                    \n\t"
>-    "sh          %[tmp4],           6(%[pfreq_signal])                    \n\t"
>-    "lh          %[tmp1],           8(%[pfrfi])                           \n\t"
>-    "lh          %[tmp2],           10(%[pfrfi])                          \n\t"
>-    "lh          %[tmp3],           12(%[pfrfi])                          \n\t"
>-    "lh          %[tmp4],           14(%[pfrfi])                          \n\t"
>-    "addiu       %[j],              %[j],                  -8             \n\t"
>-    "subu        %[tmp2],           $zero,                 %[tmp2]        \n\t"
>-    "sh          %[tmp1],           8(%[pfreq_signal])                    \n\t"
>-    "sh          %[tmp2],           10(%[pfreq_signal])                   \n\t"
>-    "subu        %[tmp4],           $zero,                 %[tmp4]        \n\t"
>-    "sh          %[tmp3],           12(%[pfreq_signal])                   \n\t"
>-    "sh          %[tmp4],           14(%[pfreq_signal])                   \n\t"
>-    "addiu       %[pfreq_signal],   %[pfreq_signal],       16             \n\t"
>-    "bgtz        %[j],              1b                                    \n\t"
>-    " addiu      %[pfrfi],          %[pfrfi],              16             \n\t"
>-    ".set        pop                                                      \n\t"
>-    : [tmp1] "=&r" (tmp1), [tmp2] "=&r" (tmp2), [tmp3] "=&r" (tmp3),
>-      [j] "=&r" (j), [pfrfi] "+r" (pfrfi), [pfreq_signal] "+r" (pfreq_signal),
>-      [tmp4] "=&r" (tmp4)
>-    :
>-    : "memory"
>-  );
>+  __asm __volatile(
>+      ".set        push                                                     "
>+      "\n\t"
>+      ".set        noreorder                                                "
>+      "\n\t"
>+      "addiu       %[j],              $zero,                 128            "
>+      "\n\t"
>+      "1:                                                                    "
>+      "\n\t"
>+      "lh          %[tmp1],           0(%[pfrfi])                           "
>+      "\n\t"
>+      "lh          %[tmp2],           2(%[pfrfi])                           "
>+      "\n\t"
>+      "lh          %[tmp3],           4(%[pfrfi])                           "
>+      "\n\t"
>+      "lh          %[tmp4],           6(%[pfrfi])                           "
>+      "\n\t"
>+      "subu        %[tmp2],           $zero,                 %[tmp2]        "
>+      "\n\t"
>+      "sh          %[tmp1],           0(%[pfreq_signal])                    "
>+      "\n\t"
>+      "sh          %[tmp2],           2(%[pfreq_signal])                    "
>+      "\n\t"
>+      "subu        %[tmp4],           $zero,                 %[tmp4]        "
>+      "\n\t"
>+      "sh          %[tmp3],           4(%[pfreq_signal])                    "
>+      "\n\t"
>+      "sh          %[tmp4],           6(%[pfreq_signal])                    "
>+      "\n\t"
>+      "lh          %[tmp1],           8(%[pfrfi])                           "
>+      "\n\t"
>+      "lh          %[tmp2],           10(%[pfrfi])                          "
>+      "\n\t"
>+      "lh          %[tmp3],           12(%[pfrfi])                          "
>+      "\n\t"
>+      "lh          %[tmp4],           14(%[pfrfi])                          "
>+      "\n\t"
>+      "addiu       %[j],              %[j],                  -8             "
>+      "\n\t"
>+      "subu        %[tmp2],           $zero,                 %[tmp2]        "
>+      "\n\t"
>+      "sh          %[tmp1],           8(%[pfreq_signal])                    "
>+      "\n\t"
>+      "sh          %[tmp2],           10(%[pfreq_signal])                   "
>+      "\n\t"
>+      "subu        %[tmp4],           $zero,                 %[tmp4]        "
>+      "\n\t"
>+      "sh          %[tmp3],           12(%[pfreq_signal])                   "
>+      "\n\t"
>+      "sh          %[tmp4],           14(%[pfreq_signal])                   "
>+      "\n\t"
>+      "addiu       %[pfreq_signal],   %[pfreq_signal],       16             "
>+      "\n\t"
>+      "bgtz        %[j],              1b                                    "
>+      "\n\t"
>+      " addiu      %[pfrfi],          %[pfrfi],              16             "
>+      "\n\t"
>+      ".set        pop                                                      "
>+      "\n\t"
>+      : [tmp1] "=&r"(tmp1), [tmp2] "=&r"(tmp2), [tmp3] "=&r"(tmp3),
>+        [j] "=&r"(j), [pfrfi] "+r"(pfrfi), [pfreq_signal] "+r"(pfreq_signal),
>+        [tmp4] "=&r"(tmp4)
>+      :
>+      : "memory");
> }
> 
> static void InverseFFTAndWindow(AecmCore* aecm,
>@@ -216,67 +227,118 @@ static void InverseFFTAndWindow(AecmCore* aecm,
>   const int16_t* pp_kSqrtHanning = &WebRtcAecm_kSqrtHanning[PART_LEN];
>   int16_t* output1 = output;
> 
>-  __asm __volatile (
>-    ".set      push                                                        \n\t"
>-    ".set      noreorder                                                   \n\t"
>-    "addiu     %[i],                $zero,                   64            \n\t"
>-   "1:                                                                     \n\t"
>-    "lh        %[tmp1],             0(%[pcoefTable_ifft])                  \n\t"
>-    "lh        %[tmp2],             2(%[pcoefTable_ifft])                  \n\t"
>-    "lh        %[tmp_re],           0(%[pefw])                             \n\t"
>-    "lh        %[tmp_im],           2(%[pefw])                             \n\t"
>-    "addu      %[pfft],             %[fft],                  %[tmp2]       \n\t"
>-    "sh        %[tmp_re],           0(%[pfft])                             \n\t"
>-    "sh        %[tmp_im],           2(%[pfft])                             \n\t"
>-    "addu      %[pfft],             %[fft],                  %[tmp1]       \n\t"
>-    "sh        %[tmp_re],           0(%[pfft])                             \n\t"
>-    "subu      %[tmp_im],           $zero,                   %[tmp_im]     \n\t"
>-    "sh        %[tmp_im],           2(%[pfft])                             \n\t"
>-    "lh        %[tmp1],             4(%[pcoefTable_ifft])                  \n\t"
>-    "lh        %[tmp2],             6(%[pcoefTable_ifft])                  \n\t"
>-    "lh        %[tmp_re],           4(%[pefw])                             \n\t"
>-    "lh        %[tmp_im],           6(%[pefw])                             \n\t"
>-    "addu      %[pfft],             %[fft],                  %[tmp2]       \n\t"
>-    "sh        %[tmp_re],           0(%[pfft])                             \n\t"
>-    "sh        %[tmp_im],           2(%[pfft])                             \n\t"
>-    "addu      %[pfft],             %[fft],                  %[tmp1]       \n\t"
>-    "sh        %[tmp_re],           0(%[pfft])                             \n\t"
>-    "subu      %[tmp_im],           $zero,                   %[tmp_im]     \n\t"
>-    "sh        %[tmp_im],           2(%[pfft])                             \n\t"
>-    "lh        %[tmp1],             8(%[pcoefTable_ifft])                  \n\t"
>-    "lh        %[tmp2],             10(%[pcoefTable_ifft])                 \n\t"
>-    "lh        %[tmp_re],           8(%[pefw])                             \n\t"
>-    "lh        %[tmp_im],           10(%[pefw])                            \n\t"
>-    "addu      %[pfft],             %[fft],                  %[tmp2]       \n\t"
>-    "sh        %[tmp_re],           0(%[pfft])                             \n\t"
>-    "sh        %[tmp_im],           2(%[pfft])                             \n\t"
>-    "addu      %[pfft],             %[fft],                  %[tmp1]       \n\t"
>-    "sh        %[tmp_re],           0(%[pfft])                             \n\t"
>-    "subu      %[tmp_im],           $zero,                   %[tmp_im]     \n\t"
>-    "sh        %[tmp_im],           2(%[pfft])                             \n\t"
>-    "lh        %[tmp1],             12(%[pcoefTable_ifft])                 \n\t"
>-    "lh        %[tmp2],             14(%[pcoefTable_ifft])                 \n\t"
>-    "lh        %[tmp_re],           12(%[pefw])                            \n\t"
>-    "lh        %[tmp_im],           14(%[pefw])                            \n\t"
>-    "addu      %[pfft],             %[fft],                  %[tmp2]       \n\t"
>-    "sh        %[tmp_re],           0(%[pfft])                             \n\t"
>-    "sh        %[tmp_im],           2(%[pfft])                             \n\t"
>-    "addu      %[pfft],             %[fft],                  %[tmp1]       \n\t"
>-    "sh        %[tmp_re],           0(%[pfft])                             \n\t"
>-    "subu      %[tmp_im],           $zero,                   %[tmp_im]     \n\t"
>-    "sh        %[tmp_im],           2(%[pfft])                             \n\t"
>-    "addiu     %[pcoefTable_ifft],  %[pcoefTable_ifft],      16            \n\t"
>-    "addiu     %[i],                %[i],                    -4            \n\t"
>-    "bgtz      %[i],                1b                                     \n\t"
>-    " addiu    %[pefw],             %[pefw],                 16            \n\t"
>-    ".set      pop                                                         \n\t"
>-    : [tmp1] "=&r" (tmp1), [tmp2] "=&r" (tmp2), [pfft] "+r" (pfft),
>-      [i] "=&r" (i), [tmp_re] "=&r" (tmp_re), [tmp_im] "=&r" (tmp_im),
>-      [pefw] "+r" (pefw), [pcoefTable_ifft] "+r" (pcoefTable_ifft),
>-      [fft] "+r" (fft)
>-    :
>-    : "memory"
>-  );
>+  __asm __volatile(
>+      ".set      push                                                        "
>+      "\n\t"
>+      ".set      noreorder                                                   "
>+      "\n\t"
>+      "addiu     %[i],                $zero,                   64            "
>+      "\n\t"
>+      "1:                                                                     "
>+      "\n\t"
>+      "lh        %[tmp1],             0(%[pcoefTable_ifft])                  "
>+      "\n\t"
>+      "lh        %[tmp2],             2(%[pcoefTable_ifft])                  "
>+      "\n\t"
>+      "lh        %[tmp_re],           0(%[pefw])                             "
>+      "\n\t"
>+      "lh        %[tmp_im],           2(%[pefw])                             "
>+      "\n\t"
>+      "addu      %[pfft],             %[fft],                  %[tmp2]       "
>+      "\n\t"
>+      "sh        %[tmp_re],           0(%[pfft])                             "
>+      "\n\t"
>+      "sh        %[tmp_im],           2(%[pfft])                             "
>+      "\n\t"
>+      "addu      %[pfft],             %[fft],                  %[tmp1]       "
>+      "\n\t"
>+      "sh        %[tmp_re],           0(%[pfft])                             "
>+      "\n\t"
>+      "subu      %[tmp_im],           $zero,                   %[tmp_im]     "
>+      "\n\t"
>+      "sh        %[tmp_im],           2(%[pfft])                             "
>+      "\n\t"
>+      "lh        %[tmp1],             4(%[pcoefTable_ifft])                  "
>+      "\n\t"
>+      "lh        %[tmp2],             6(%[pcoefTable_ifft])                  "
>+      "\n\t"
>+      "lh        %[tmp_re],           4(%[pefw])                             "
>+      "\n\t"
>+      "lh        %[tmp_im],           6(%[pefw])                             "
>+      "\n\t"
>+      "addu      %[pfft],             %[fft],                  %[tmp2]       "
>+      "\n\t"
>+      "sh        %[tmp_re],           0(%[pfft])                             "
>+      "\n\t"
>+      "sh        %[tmp_im],           2(%[pfft])                             "
>+      "\n\t"
>+      "addu      %[pfft],             %[fft],                  %[tmp1]       "
>+      "\n\t"
>+      "sh        %[tmp_re],           0(%[pfft])                             "
>+      "\n\t"
>+      "subu      %[tmp_im],           $zero,                   %[tmp_im]     "
>+      "\n\t"
>+      "sh        %[tmp_im],           2(%[pfft])                             "
>+      "\n\t"
>+      "lh        %[tmp1],             8(%[pcoefTable_ifft])                  "
>+      "\n\t"
>+      "lh        %[tmp2],             10(%[pcoefTable_ifft])                 "
>+      "\n\t"
>+      "lh        %[tmp_re],           8(%[pefw])                             "
>+      "\n\t"
>+      "lh        %[tmp_im],           10(%[pefw])                            "
>+      "\n\t"
>+      "addu      %[pfft],             %[fft],                  %[tmp2]       "
>+      "\n\t"
>+      "sh        %[tmp_re],           0(%[pfft])                             "
>+      "\n\t"
>+      "sh        %[tmp_im],           2(%[pfft])                             "
>+      "\n\t"
>+      "addu      %[pfft],             %[fft],                  %[tmp1]       "
>+      "\n\t"
>+      "sh        %[tmp_re],           0(%[pfft])                             "
>+      "\n\t"
>+      "subu      %[tmp_im],           $zero,                   %[tmp_im]     "
>+      "\n\t"
>+      "sh        %[tmp_im],           2(%[pfft])                             "
>+      "\n\t"
>+      "lh        %[tmp1],             12(%[pcoefTable_ifft])                 "
>+      "\n\t"
>+      "lh        %[tmp2],             14(%[pcoefTable_ifft])                 "
>+      "\n\t"
>+      "lh        %[tmp_re],           12(%[pefw])                            "
>+      "\n\t"
>+      "lh        %[tmp_im],           14(%[pefw])                            "
>+      "\n\t"
>+      "addu      %[pfft],             %[fft],                  %[tmp2]       "
>+      "\n\t"
>+      "sh        %[tmp_re],           0(%[pfft])                             "
>+      "\n\t"
>+      "sh        %[tmp_im],           2(%[pfft])                             "
>+      "\n\t"
>+      "addu      %[pfft],             %[fft],                  %[tmp1]       "
>+      "\n\t"
>+      "sh        %[tmp_re],           0(%[pfft])                             "
>+      "\n\t"
>+      "subu      %[tmp_im],           $zero,                   %[tmp_im]     "
>+      "\n\t"
>+      "sh        %[tmp_im],           2(%[pfft])                             "
>+      "\n\t"
>+      "addiu     %[pcoefTable_ifft],  %[pcoefTable_ifft],      16            "
>+      "\n\t"
>+      "addiu     %[i],                %[i],                    -4            "
>+      "\n\t"
>+      "bgtz      %[i],                1b                                     "
>+      "\n\t"
>+      " addiu    %[pefw],             %[pefw],                 16            "
>+      "\n\t"
>+      ".set      pop                                                         "
>+      "\n\t"
>+      : [tmp1] "=&r"(tmp1), [tmp2] "=&r"(tmp2), [pfft] "+r"(pfft), [i] "=&r"(i),
>+        [tmp_re] "=&r"(tmp_re), [tmp_im] "=&r"(tmp_im), [pefw] "+r"(pefw),
>+        [pcoefTable_ifft] "+r"(pcoefTable_ifft), [fft] "+r"(fft)
>+      :
>+      : "memory");
> 
>   fft[2] = efw[PART_LEN].real;
>   fft[3] = -efw[PART_LEN].imag;
>@@ -284,150 +346,235 @@ static void InverseFFTAndWindow(AecmCore* aecm,
>   outCFFT = WebRtcSpl_ComplexIFFT(fft, PART_LEN_SHIFT, 1);
>   pfft = fft;
> 
>-  __asm __volatile (
>-    ".set       push                                               \n\t"
>-    ".set       noreorder                                          \n\t"
>-    "addiu      %[i],            $zero,               128          \n\t"
>-   "1:                                                             \n\t"
>-    "lh         %[tmp1],         0(%[ppfft])                       \n\t"
>-    "lh         %[tmp2],         4(%[ppfft])                       \n\t"
>-    "lh         %[tmp3],         8(%[ppfft])                       \n\t"
>-    "lh         %[tmp4],         12(%[ppfft])                      \n\t"
>-    "addiu      %[i],            %[i],                -4           \n\t"
>-    "sh         %[tmp1],         0(%[pfft])                        \n\t"
>-    "sh         %[tmp2],         2(%[pfft])                        \n\t"
>-    "sh         %[tmp3],         4(%[pfft])                        \n\t"
>-    "sh         %[tmp4],         6(%[pfft])                        \n\t"
>-    "addiu      %[ppfft],        %[ppfft],            16           \n\t"
>-    "bgtz       %[i],            1b                                \n\t"
>-    " addiu     %[pfft],         %[pfft],             8            \n\t"
>-    ".set       pop                                                \n\t"
>-    : [tmp1] "=&r" (tmp1), [tmp2] "=&r" (tmp2), [pfft] "+r" (pfft),
>-      [i] "=&r" (i), [tmp3] "=&r" (tmp3), [tmp4] "=&r" (tmp4),
>-      [ppfft] "+r" (ppfft)
>-    :
>-    : "memory"
>-  );
>+  __asm __volatile(
>+      ".set       push                                               \n\t"
>+      ".set       noreorder                                          \n\t"
>+      "addiu      %[i],            $zero,               128          \n\t"
>+      "1:                                                             \n\t"
>+      "lh         %[tmp1],         0(%[ppfft])                       \n\t"
>+      "lh         %[tmp2],         4(%[ppfft])                       \n\t"
>+      "lh         %[tmp3],         8(%[ppfft])                       \n\t"
>+      "lh         %[tmp4],         12(%[ppfft])                      \n\t"
>+      "addiu      %[i],            %[i],                -4           \n\t"
>+      "sh         %[tmp1],         0(%[pfft])                        \n\t"
>+      "sh         %[tmp2],         2(%[pfft])                        \n\t"
>+      "sh         %[tmp3],         4(%[pfft])                        \n\t"
>+      "sh         %[tmp4],         6(%[pfft])                        \n\t"
>+      "addiu      %[ppfft],        %[ppfft],            16           \n\t"
>+      "bgtz       %[i],            1b                                \n\t"
>+      " addiu     %[pfft],         %[pfft],             8            \n\t"
>+      ".set       pop                                                \n\t"
>+      : [tmp1] "=&r"(tmp1), [tmp2] "=&r"(tmp2), [pfft] "+r"(pfft), [i] "=&r"(i),
>+        [tmp3] "=&r"(tmp3), [tmp4] "=&r"(tmp4), [ppfft] "+r"(ppfft)
>+      :
>+      : "memory");
> 
>   pfft = fft;
>   out_aecm = (int32_t)(outCFFT - aecm->dfaCleanQDomain);
> 
>-  __asm __volatile (
>-    ".set       push                                                       \n\t"
>-    ".set       noreorder                                                  \n\t"
>-    "addiu      %[i],                $zero,                  64            \n\t"
>-   "11:                                                                    \n\t"
>-    "lh         %[tmp1],             0(%[pfft])                            \n\t"
>-    "lh         %[tmp2],             0(%[p_kSqrtHanning])                  \n\t"
>-    "addiu      %[i],                %[i],                   -2            \n\t"
>-    "mul        %[tmp1],             %[tmp1],                %[tmp2]       \n\t"
>-    "lh         %[tmp3],             2(%[pfft])                            \n\t"
>-    "lh         %[tmp4],             2(%[p_kSqrtHanning])                  \n\t"
>-    "mul        %[tmp3],             %[tmp3],                %[tmp4]       \n\t"
>-    "addiu      %[tmp1],             %[tmp1],                8192          \n\t"
>-    "sra        %[tmp1],             %[tmp1],                14            \n\t"
>-    "addiu      %[tmp3],             %[tmp3],                8192          \n\t"
>-    "sra        %[tmp3],             %[tmp3],                14            \n\t"
>-    "bgez       %[out_aecm],         1f                                    \n\t"
>-    " negu      %[tmp2],             %[out_aecm]                           \n\t"
>-    "srav       %[tmp1],             %[tmp1],                %[tmp2]       \n\t"
>-    "b          2f                                                         \n\t"
>-    " srav      %[tmp3],             %[tmp3],                %[tmp2]       \n\t"
>-   "1:                                                                     \n\t"
>-    "sllv       %[tmp1],             %[tmp1],                %[out_aecm]   \n\t"
>-    "sllv       %[tmp3],             %[tmp3],                %[out_aecm]   \n\t"
>-   "2:                                                                     \n\t"
>-    "lh         %[tmp4],             0(%[paecm_buf])                       \n\t"
>-    "lh         %[tmp2],             2(%[paecm_buf])                       \n\t"
>-    "addu       %[tmp3],             %[tmp3],                %[tmp2]       \n\t"
>-    "addu       %[tmp1],             %[tmp1],                %[tmp4]       \n\t"
>+  __asm __volatile(
>+      ".set       push                                                       "
>+      "\n\t"
>+      ".set       noreorder                                                  "
>+      "\n\t"
>+      "addiu      %[i],                $zero,                  64            "
>+      "\n\t"
>+      "11:                                                                    "
>+      "\n\t"
>+      "lh         %[tmp1],             0(%[pfft])                            "
>+      "\n\t"
>+      "lh         %[tmp2],             0(%[p_kSqrtHanning])                  "
>+      "\n\t"
>+      "addiu      %[i],                %[i],                   -2            "
>+      "\n\t"
>+      "mul        %[tmp1],             %[tmp1],                %[tmp2]       "
>+      "\n\t"
>+      "lh         %[tmp3],             2(%[pfft])                            "
>+      "\n\t"
>+      "lh         %[tmp4],             2(%[p_kSqrtHanning])                  "
>+      "\n\t"
>+      "mul        %[tmp3],             %[tmp3],                %[tmp4]       "
>+      "\n\t"
>+      "addiu      %[tmp1],             %[tmp1],                8192          "
>+      "\n\t"
>+      "sra        %[tmp1],             %[tmp1],                14            "
>+      "\n\t"
>+      "addiu      %[tmp3],             %[tmp3],                8192          "
>+      "\n\t"
>+      "sra        %[tmp3],             %[tmp3],                14            "
>+      "\n\t"
>+      "bgez       %[out_aecm],         1f                                    "
>+      "\n\t"
>+      " negu      %[tmp2],             %[out_aecm]                           "
>+      "\n\t"
>+      "srav       %[tmp1],             %[tmp1],                %[tmp2]       "
>+      "\n\t"
>+      "b          2f                                                         "
>+      "\n\t"
>+      " srav      %[tmp3],             %[tmp3],                %[tmp2]       "
>+      "\n\t"
>+      "1:                                                                     "
>+      "\n\t"
>+      "sllv       %[tmp1],             %[tmp1],                %[out_aecm]   "
>+      "\n\t"
>+      "sllv       %[tmp3],             %[tmp3],                %[out_aecm]   "
>+      "\n\t"
>+      "2:                                                                     "
>+      "\n\t"
>+      "lh         %[tmp4],             0(%[paecm_buf])                       "
>+      "\n\t"
>+      "lh         %[tmp2],             2(%[paecm_buf])                       "
>+      "\n\t"
>+      "addu       %[tmp3],             %[tmp3],                %[tmp2]       "
>+      "\n\t"
>+      "addu       %[tmp1],             %[tmp1],                %[tmp4]       "
>+      "\n\t"
> #if defined(MIPS_DSP_R1_LE)
>-    "shll_s.w   %[tmp1],             %[tmp1],                16            \n\t"
>-    "sra        %[tmp1],             %[tmp1],                16            \n\t"
>-    "shll_s.w   %[tmp3],             %[tmp3],                16            \n\t"
>-    "sra        %[tmp3],             %[tmp3],                16            \n\t"
>-#else  // #if defined(MIPS_DSP_R1_LE)
>-    "sra        %[tmp4],             %[tmp1],                31            \n\t"
>-    "sra        %[tmp2],             %[tmp1],                15            \n\t"
>-    "beq        %[tmp4],             %[tmp2],                3f            \n\t"
>-    " ori       %[tmp2],             $zero,                  0x7fff        \n\t"
>-    "xor        %[tmp1],             %[tmp2],                %[tmp4]       \n\t"
>-   "3:                                                                     \n\t"
>-    "sra        %[tmp2],             %[tmp3],                31            \n\t"
>-    "sra        %[tmp4],             %[tmp3],                15            \n\t"
>-    "beq        %[tmp2],             %[tmp4],                4f            \n\t"
>-    " ori       %[tmp4],             $zero,                  0x7fff        \n\t"
>-    "xor        %[tmp3],             %[tmp4],                %[tmp2]       \n\t"
>-   "4:                                                                     \n\t"
>+      "shll_s.w   %[tmp1],             %[tmp1],                16            "
>+      "\n\t"
>+      "sra        %[tmp1],             %[tmp1],                16            "
>+      "\n\t"
>+      "shll_s.w   %[tmp3],             %[tmp3],                16            "
>+      "\n\t"
>+      "sra        %[tmp3],             %[tmp3],                16            "
>+      "\n\t"
>+#else   // #if defined(MIPS_DSP_R1_LE)
>+      "sra        %[tmp4],             %[tmp1],                31            "
>+      "\n\t"
>+      "sra        %[tmp2],             %[tmp1],                15            "
>+      "\n\t"
>+      "beq        %[tmp4],             %[tmp2],                3f            "
>+      "\n\t"
>+      " ori       %[tmp2],             $zero,                  0x7fff        "
>+      "\n\t"
>+      "xor        %[tmp1],             %[tmp2],                %[tmp4]       "
>+      "\n\t"
>+      "3:                                                                     "
>+      "\n\t"
>+      "sra        %[tmp2],             %[tmp3],                31            "
>+      "\n\t"
>+      "sra        %[tmp4],             %[tmp3],                15            "
>+      "\n\t"
>+      "beq        %[tmp2],             %[tmp4],                4f            "
>+      "\n\t"
>+      " ori       %[tmp4],             $zero,                  0x7fff        "
>+      "\n\t"
>+      "xor        %[tmp3],             %[tmp4],                %[tmp2]       "
>+      "\n\t"
>+      "4:                                                                     "
>+      "\n\t"
> #endif  // #if defined(MIPS_DSP_R1_LE)
>-    "sh         %[tmp1],             0(%[pfft])                            \n\t"
>-    "sh         %[tmp1],             0(%[output1])                         \n\t"
>-    "sh         %[tmp3],             2(%[pfft])                            \n\t"
>-    "sh         %[tmp3],             2(%[output1])                         \n\t"
>-    "lh         %[tmp1],             128(%[pfft])                          \n\t"
>-    "lh         %[tmp2],             0(%[pp_kSqrtHanning])                 \n\t"
>-    "mul        %[tmp1],             %[tmp1],                %[tmp2]       \n\t"
>-    "lh         %[tmp3],             130(%[pfft])                          \n\t"
>-    "lh         %[tmp4],             -2(%[pp_kSqrtHanning])                \n\t"
>-    "mul        %[tmp3],             %[tmp3],                %[tmp4]       \n\t"
>-    "sra        %[tmp1],             %[tmp1],                14            \n\t"
>-    "sra        %[tmp3],             %[tmp3],                14            \n\t"
>-    "bgez       %[out_aecm],         5f                                    \n\t"
>-    " negu      %[tmp2],             %[out_aecm]                           \n\t"
>-    "srav       %[tmp3],             %[tmp3],                %[tmp2]       \n\t"
>-    "b          6f                                                         \n\t"
>-    " srav      %[tmp1],             %[tmp1],                %[tmp2]       \n\t"
>-   "5:                                                                     \n\t"
>-    "sllv       %[tmp1],             %[tmp1],                %[out_aecm]   \n\t"
>-    "sllv       %[tmp3],             %[tmp3],                %[out_aecm]   \n\t"
>-   "6:                                                                     \n\t"
>+      "sh         %[tmp1],             0(%[pfft])                            "
>+      "\n\t"
>+      "sh         %[tmp1],             0(%[output1])                         "
>+      "\n\t"
>+      "sh         %[tmp3],             2(%[pfft])                            "
>+      "\n\t"
>+      "sh         %[tmp3],             2(%[output1])                         "
>+      "\n\t"
>+      "lh         %[tmp1],             128(%[pfft])                          "
>+      "\n\t"
>+      "lh         %[tmp2],             0(%[pp_kSqrtHanning])                 "
>+      "\n\t"
>+      "mul        %[tmp1],             %[tmp1],                %[tmp2]       "
>+      "\n\t"
>+      "lh         %[tmp3],             130(%[pfft])                          "
>+      "\n\t"
>+      "lh         %[tmp4],             -2(%[pp_kSqrtHanning])                "
>+      "\n\t"
>+      "mul        %[tmp3],             %[tmp3],                %[tmp4]       "
>+      "\n\t"
>+      "sra        %[tmp1],             %[tmp1],                14            "
>+      "\n\t"
>+      "sra        %[tmp3],             %[tmp3],                14            "
>+      "\n\t"
>+      "bgez       %[out_aecm],         5f                                    "
>+      "\n\t"
>+      " negu      %[tmp2],             %[out_aecm]                           "
>+      "\n\t"
>+      "srav       %[tmp3],             %[tmp3],                %[tmp2]       "
>+      "\n\t"
>+      "b          6f                                                         "
>+      "\n\t"
>+      " srav      %[tmp1],             %[tmp1],                %[tmp2]       "
>+      "\n\t"
>+      "5:                                                                     "
>+      "\n\t"
>+      "sllv       %[tmp1],             %[tmp1],                %[out_aecm]   "
>+      "\n\t"
>+      "sllv       %[tmp3],             %[tmp3],                %[out_aecm]   "
>+      "\n\t"
>+      "6:                                                                     "
>+      "\n\t"
> #if defined(MIPS_DSP_R1_LE)
>-    "shll_s.w   %[tmp1],             %[tmp1],                16            \n\t"
>-    "sra        %[tmp1],             %[tmp1],                16            \n\t"
>-    "shll_s.w   %[tmp3],             %[tmp3],                16            \n\t"
>-    "sra        %[tmp3],             %[tmp3],                16            \n\t"
>-#else  // #if defined(MIPS_DSP_R1_LE)
>-    "sra        %[tmp4],             %[tmp1],                31            \n\t"
>-    "sra        %[tmp2],             %[tmp1],                15            \n\t"
>-    "beq        %[tmp4],             %[tmp2],                7f            \n\t"
>-    " ori       %[tmp2],             $zero,                  0x7fff        \n\t"
>-    "xor        %[tmp1],             %[tmp2],                %[tmp4]       \n\t"
>-   "7:                                                                     \n\t"
>-    "sra        %[tmp2],             %[tmp3],                31            \n\t"
>-    "sra        %[tmp4],             %[tmp3],                15            \n\t"
>-    "beq        %[tmp2],             %[tmp4],                8f            \n\t"
>-    " ori       %[tmp4],             $zero,                  0x7fff        \n\t"
>-    "xor        %[tmp3],             %[tmp4],                %[tmp2]       \n\t"
>-   "8:                                                                     \n\t"
>+      "shll_s.w   %[tmp1],             %[tmp1],                16            "
>+      "\n\t"
>+      "sra        %[tmp1],             %[tmp1],                16            "
>+      "\n\t"
>+      "shll_s.w   %[tmp3],             %[tmp3],                16            "
>+      "\n\t"
>+      "sra        %[tmp3],             %[tmp3],                16            "
>+      "\n\t"
>+#else   // #if defined(MIPS_DSP_R1_LE)
>+      "sra        %[tmp4],             %[tmp1],                31            "
>+      "\n\t"
>+      "sra        %[tmp2],             %[tmp1],                15            "
>+      "\n\t"
>+      "beq        %[tmp4],             %[tmp2],                7f            "
>+      "\n\t"
>+      " ori       %[tmp2],             $zero,                  0x7fff        "
>+      "\n\t"
>+      "xor        %[tmp1],             %[tmp2],                %[tmp4]       "
>+      "\n\t"
>+      "7:                                                                     "
>+      "\n\t"
>+      "sra        %[tmp2],             %[tmp3],                31            "
>+      "\n\t"
>+      "sra        %[tmp4],             %[tmp3],                15            "
>+      "\n\t"
>+      "beq        %[tmp2],             %[tmp4],                8f            "
>+      "\n\t"
>+      " ori       %[tmp4],             $zero,                  0x7fff        "
>+      "\n\t"
>+      "xor        %[tmp3],             %[tmp4],                %[tmp2]       "
>+      "\n\t"
>+      "8:                                                                     "
>+      "\n\t"
> #endif  // #if defined(MIPS_DSP_R1_LE)
>-    "sh         %[tmp1],             0(%[paecm_buf])                       \n\t"
>-    "sh         %[tmp3],             2(%[paecm_buf])                       \n\t"
>-    "addiu      %[output1],          %[output1],             4             \n\t"
>-    "addiu      %[paecm_buf],        %[paecm_buf],           4             \n\t"
>-    "addiu      %[pfft],             %[pfft],                4             \n\t"
>-    "addiu      %[p_kSqrtHanning],   %[p_kSqrtHanning],      4             \n\t"
>-    "bgtz       %[i],                11b                                   \n\t"
>-    " addiu     %[pp_kSqrtHanning],  %[pp_kSqrtHanning],     -4            \n\t"
>-    ".set       pop                                                        \n\t"
>-    : [tmp1] "=&r" (tmp1), [tmp2] "=&r" (tmp2), [pfft] "+r" (pfft),
>-      [output1] "+r" (output1), [tmp3] "=&r" (tmp3), [tmp4] "=&r" (tmp4),
>-      [paecm_buf] "+r" (paecm_buf), [i] "=&r" (i),
>-      [pp_kSqrtHanning] "+r" (pp_kSqrtHanning),
>-      [p_kSqrtHanning] "+r" (p_kSqrtHanning)
>-    : [out_aecm] "r" (out_aecm),
>-      [WebRtcAecm_kSqrtHanning] "r" (WebRtcAecm_kSqrtHanning)
>-    : "hi", "lo","memory"
>-  );
>+      "sh         %[tmp1],             0(%[paecm_buf])                       "
>+      "\n\t"
>+      "sh         %[tmp3],             2(%[paecm_buf])                       "
>+      "\n\t"
>+      "addiu      %[output1],          %[output1],             4             "
>+      "\n\t"
>+      "addiu      %[paecm_buf],        %[paecm_buf],           4             "
>+      "\n\t"
>+      "addiu      %[pfft],             %[pfft],                4             "
>+      "\n\t"
>+      "addiu      %[p_kSqrtHanning],   %[p_kSqrtHanning],      4             "
>+      "\n\t"
>+      "bgtz       %[i],                11b                                   "
>+      "\n\t"
>+      " addiu     %[pp_kSqrtHanning],  %[pp_kSqrtHanning],     -4            "
>+      "\n\t"
>+      ".set       pop                                                        "
>+      "\n\t"
>+      : [tmp1] "=&r"(tmp1), [tmp2] "=&r"(tmp2), [pfft] "+r"(pfft),
>+        [output1] "+r"(output1), [tmp3] "=&r"(tmp3), [tmp4] "=&r"(tmp4),
>+        [paecm_buf] "+r"(paecm_buf), [i] "=&r"(i),
>+        [pp_kSqrtHanning] "+r"(pp_kSqrtHanning),
>+        [p_kSqrtHanning] "+r"(p_kSqrtHanning)
>+      : [out_aecm] "r"(out_aecm),
>+        [WebRtcAecm_kSqrtHanning] "r"(WebRtcAecm_kSqrtHanning)
>+      : "hi", "lo", "memory");
> 
>   // Copy the current block to the old position
>   // (aecm->outBuf is shifted elsewhere)
>   memcpy(aecm->xBuf, aecm->xBuf + PART_LEN, sizeof(int16_t) * PART_LEN);
>-  memcpy(aecm->dBufNoisy,
>-         aecm->dBufNoisy + PART_LEN,
>+  memcpy(aecm->dBufNoisy, aecm->dBufNoisy + PART_LEN,
>          sizeof(int16_t) * PART_LEN);
>   if (nearendClean != NULL) {
>-    memcpy(aecm->dBufClean,
>-           aecm->dBufClean + PART_LEN,
>+    memcpy(aecm->dBufClean, aecm->dBufClean + PART_LEN,
>            sizeof(int16_t) * PART_LEN);
>   }
> }
>@@ -451,62 +598,61 @@ void WebRtcAecm_CalcLinearEnergies_mips(AecmCore* aecm,
> 
>   // Get energy for the delayed far end signal and estimated
>   // echo using both stored and adapted channels.
>-  for (i = 0; i < PART_LEN; i+= 4) {
>-    __asm __volatile (
>-      ".set           push                                            \n\t"
>-      ".set           noreorder                                       \n\t"
>-      "lh             %[stored0],     0(%[ch_stored_p])               \n\t"
>-      "lhu            %[adept0],      0(%[ch_adapt_p])                \n\t"
>-      "lhu            %[spectrum0],   0(%[spectrum_p])                \n\t"
>-      "lh             %[stored1],     2(%[ch_stored_p])               \n\t"
>-      "lhu            %[adept1],      2(%[ch_adapt_p])                \n\t"
>-      "lhu            %[spectrum1],   2(%[spectrum_p])                \n\t"
>-      "mul            %[echo0],       %[stored0],     %[spectrum0]    \n\t"
>-      "mul            %[temp0],       %[adept0],      %[spectrum0]    \n\t"
>-      "mul            %[echo1],       %[stored1],     %[spectrum1]    \n\t"
>-      "mul            %[temp1],       %[adept1],      %[spectrum1]    \n\t"
>-      "addu           %[par1],        %[par1],        %[spectrum0]    \n\t"
>-      "addu           %[par1],        %[par1],        %[spectrum1]    \n\t"
>-      "addiu          %[echo_p],      %[echo_p],      16              \n\t"
>-      "addu           %[par3],        %[par3],        %[echo0]        \n\t"
>-      "addu           %[par2],        %[par2],        %[temp0]        \n\t"
>-      "addu           %[par3],        %[par3],        %[echo1]        \n\t"
>-      "addu           %[par2],        %[par2],        %[temp1]        \n\t"
>-      "usw            %[echo0],       -16(%[echo_p])                  \n\t"
>-      "usw            %[echo1],       -12(%[echo_p])                  \n\t"
>-      "lh             %[stored0],     4(%[ch_stored_p])               \n\t"
>-      "lhu            %[adept0],      4(%[ch_adapt_p])                \n\t"
>-      "lhu            %[spectrum0],   4(%[spectrum_p])                \n\t"
>-      "lh             %[stored1],     6(%[ch_stored_p])               \n\t"
>-      "lhu            %[adept1],      6(%[ch_adapt_p])                \n\t"
>-      "lhu            %[spectrum1],   6(%[spectrum_p])                \n\t"
>-      "mul            %[echo0],       %[stored0],     %[spectrum0]    \n\t"
>-      "mul            %[temp0],       %[adept0],      %[spectrum0]    \n\t"
>-      "mul            %[echo1],       %[stored1],     %[spectrum1]    \n\t"
>-      "mul            %[temp1],       %[adept1],      %[spectrum1]    \n\t"
>-      "addu           %[par1],        %[par1],        %[spectrum0]    \n\t"
>-      "addu           %[par1],        %[par1],        %[spectrum1]    \n\t"
>-      "addiu          %[ch_stored_p], %[ch_stored_p], 8               \n\t"
>-      "addiu          %[ch_adapt_p],  %[ch_adapt_p],  8               \n\t"
>-      "addiu          %[spectrum_p],  %[spectrum_p],  8               \n\t"
>-      "addu           %[par3],        %[par3],        %[echo0]        \n\t"
>-      "addu           %[par2],        %[par2],        %[temp0]        \n\t"
>-      "addu           %[par3],        %[par3],        %[echo1]        \n\t"
>-      "addu           %[par2],        %[par2],        %[temp1]        \n\t"
>-      "usw            %[echo0],       -8(%[echo_p])                   \n\t"
>-      "usw            %[echo1],       -4(%[echo_p])                   \n\t"
>-      ".set           pop                                             \n\t"
>-      : [temp0] "=&r" (temp0), [stored0] "=&r" (stored0),
>-        [adept0] "=&r" (adept0), [spectrum0] "=&r" (spectrum0),
>-        [echo0] "=&r" (echo0), [echo_p] "+r" (echo_p), [par3] "+r" (par3),
>-        [par1] "+r" (par1), [par2] "+r" (par2), [stored1] "=&r" (stored1),
>-        [adept1] "=&r" (adept1), [echo1] "=&r" (echo1),
>-        [spectrum1] "=&r" (spectrum1), [temp1] "=&r" (temp1),
>-        [ch_stored_p] "+r" (ch_stored_p), [ch_adapt_p] "+r" (ch_adapt_p),
>-        [spectrum_p] "+r" (spectrum_p)
>-      :
>-      : "hi", "lo", "memory"
>-    );
>+  for (i = 0; i < PART_LEN; i += 4) {
>+    __asm __volatile(
>+        ".set           push                                            \n\t"
>+        ".set           noreorder                                       \n\t"
>+        "lh             %[stored0],     0(%[ch_stored_p])               \n\t"
>+        "lhu            %[adept0],      0(%[ch_adapt_p])                \n\t"
>+        "lhu            %[spectrum0],   0(%[spectrum_p])                \n\t"
>+        "lh             %[stored1],     2(%[ch_stored_p])               \n\t"
>+        "lhu            %[adept1],      2(%[ch_adapt_p])                \n\t"
>+        "lhu            %[spectrum1],   2(%[spectrum_p])                \n\t"
>+        "mul            %[echo0],       %[stored0],     %[spectrum0]    \n\t"
>+        "mul            %[temp0],       %[adept0],      %[spectrum0]    \n\t"
>+        "mul            %[echo1],       %[stored1],     %[spectrum1]    \n\t"
>+        "mul            %[temp1],       %[adept1],      %[spectrum1]    \n\t"
>+        "addu           %[par1],        %[par1],        %[spectrum0]    \n\t"
>+        "addu           %[par1],        %[par1],        %[spectrum1]    \n\t"
>+        "addiu          %[echo_p],      %[echo_p],      16              \n\t"
>+        "addu           %[par3],        %[par3],        %[echo0]        \n\t"
>+        "addu           %[par2],        %[par2],        %[temp0]        \n\t"
>+        "addu           %[par3],        %[par3],        %[echo1]        \n\t"
>+        "addu           %[par2],        %[par2],        %[temp1]        \n\t"
>+        "usw            %[echo0],       -16(%[echo_p])                  \n\t"
>+        "usw            %[echo1],       -12(%[echo_p])                  \n\t"
>+        "lh             %[stored0],     4(%[ch_stored_p])               \n\t"
>+        "lhu            %[adept0],      4(%[ch_adapt_p])                \n\t"
>+        "lhu            %[spectrum0],   4(%[spectrum_p])                \n\t"
>+        "lh             %[stored1],     6(%[ch_stored_p])               \n\t"
>+        "lhu            %[adept1],      6(%[ch_adapt_p])                \n\t"
>+        "lhu            %[spectrum1],   6(%[spectrum_p])                \n\t"
>+        "mul            %[echo0],       %[stored0],     %[spectrum0]    \n\t"
>+        "mul            %[temp0],       %[adept0],      %[spectrum0]    \n\t"
>+        "mul            %[echo1],       %[stored1],     %[spectrum1]    \n\t"
>+        "mul            %[temp1],       %[adept1],      %[spectrum1]    \n\t"
>+        "addu           %[par1],        %[par1],        %[spectrum0]    \n\t"
>+        "addu           %[par1],        %[par1],        %[spectrum1]    \n\t"
>+        "addiu          %[ch_stored_p], %[ch_stored_p], 8               \n\t"
>+        "addiu          %[ch_adapt_p],  %[ch_adapt_p],  8               \n\t"
>+        "addiu          %[spectrum_p],  %[spectrum_p],  8               \n\t"
>+        "addu           %[par3],        %[par3],        %[echo0]        \n\t"
>+        "addu           %[par2],        %[par2],        %[temp0]        \n\t"
>+        "addu           %[par3],        %[par3],        %[echo1]        \n\t"
>+        "addu           %[par2],        %[par2],        %[temp1]        \n\t"
>+        "usw            %[echo0],       -8(%[echo_p])                   \n\t"
>+        "usw            %[echo1],       -4(%[echo_p])                   \n\t"
>+        ".set           pop                                             \n\t"
>+        : [temp0] "=&r"(temp0), [stored0] "=&r"(stored0),
>+          [adept0] "=&r"(adept0), [spectrum0] "=&r"(spectrum0),
>+          [echo0] "=&r"(echo0), [echo_p] "+r"(echo_p), [par3] "+r"(par3),
>+          [par1] "+r"(par1), [par2] "+r"(par2), [stored1] "=&r"(stored1),
>+          [adept1] "=&r"(adept1), [echo1] "=&r"(echo1),
>+          [spectrum1] "=&r"(spectrum1), [temp1] "=&r"(temp1),
>+          [ch_stored_p] "+r"(ch_stored_p), [ch_adapt_p] "+r"(ch_adapt_p),
>+          [spectrum_p] "+r"(spectrum_p)
>+        :
>+        : "hi", "lo", "memory");
>   }
> 
>   echo_est[PART_LEN] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[PART_LEN],
>@@ -537,35 +683,33 @@ void WebRtcAecm_StoreAdaptiveChannel_mips(AecmCore* aecm,
>          sizeof(int16_t) * PART_LEN1);
>   // Recalculate echo estimate
>   for (i = 0; i < PART_LEN; i += 4) {
>-    __asm __volatile (
>-      "ulw            %[temp0],   0(%[temp8])               \n\t"
>-      "ulw            %[temp2],   0(%[temp1])               \n\t"
>-      "ulw            %[temp4],   4(%[temp8])               \n\t"
>-      "ulw            %[temp5],   4(%[temp1])               \n\t"
>-      "muleq_s.w.phl  %[temp3],   %[temp2],     %[temp0]    \n\t"
>-      "muleq_s.w.phr  %[temp0],   %[temp2],     %[temp0]    \n\t"
>-      "muleq_s.w.phl  %[temp6],   %[temp5],     %[temp4]    \n\t"
>-      "muleq_s.w.phr  %[temp4],   %[temp5],     %[temp4]    \n\t"
>-      "addiu          %[temp7],   %[temp7],     16          \n\t"
>-      "addiu          %[temp1],   %[temp1],     8           \n\t"
>-      "addiu          %[temp8],   %[temp8],     8           \n\t"
>-      "sra            %[temp3],   %[temp3],     1           \n\t"
>-      "sra            %[temp0],   %[temp0],     1           \n\t"
>-      "sra            %[temp6],   %[temp6],     1           \n\t"
>-      "sra            %[temp4],   %[temp4],     1           \n\t"
>-      "usw            %[temp3],   -12(%[temp7])             \n\t"
>-      "usw            %[temp0],   -16(%[temp7])             \n\t"
>-      "usw            %[temp6],   -4(%[temp7])              \n\t"
>-      "usw            %[temp4],   -8(%[temp7])              \n\t"
>-      : [temp0] "=&r" (temp0), [temp2] "=&r" (temp2), [temp3] "=&r" (temp3),
>-        [temp4] "=&r" (temp4), [temp5] "=&r" (temp5), [temp6] "=&r" (temp6),
>-        [temp1] "+r" (temp1), [temp8] "+r" (temp8), [temp7] "+r" (temp7)
>-      :
>-      : "hi", "lo", "memory"
>-    );
>+    __asm __volatile(
>+        "ulw            %[temp0],   0(%[temp8])               \n\t"
>+        "ulw            %[temp2],   0(%[temp1])               \n\t"
>+        "ulw            %[temp4],   4(%[temp8])               \n\t"
>+        "ulw            %[temp5],   4(%[temp1])               \n\t"
>+        "muleq_s.w.phl  %[temp3],   %[temp2],     %[temp0]    \n\t"
>+        "muleq_s.w.phr  %[temp0],   %[temp2],     %[temp0]    \n\t"
>+        "muleq_s.w.phl  %[temp6],   %[temp5],     %[temp4]    \n\t"
>+        "muleq_s.w.phr  %[temp4],   %[temp5],     %[temp4]    \n\t"
>+        "addiu          %[temp7],   %[temp7],     16          \n\t"
>+        "addiu          %[temp1],   %[temp1],     8           \n\t"
>+        "addiu          %[temp8],   %[temp8],     8           \n\t"
>+        "sra            %[temp3],   %[temp3],     1           \n\t"
>+        "sra            %[temp0],   %[temp0],     1           \n\t"
>+        "sra            %[temp6],   %[temp6],     1           \n\t"
>+        "sra            %[temp4],   %[temp4],     1           \n\t"
>+        "usw            %[temp3],   -12(%[temp7])             \n\t"
>+        "usw            %[temp0],   -16(%[temp7])             \n\t"
>+        "usw            %[temp6],   -4(%[temp7])              \n\t"
>+        "usw            %[temp4],   -8(%[temp7])              \n\t"
>+        : [temp0] "=&r"(temp0), [temp2] "=&r"(temp2), [temp3] "=&r"(temp3),
>+          [temp4] "=&r"(temp4), [temp5] "=&r"(temp5), [temp6] "=&r"(temp6),
>+          [temp1] "+r"(temp1), [temp8] "+r"(temp8), [temp7] "+r"(temp7)
>+        :
>+        : "hi", "lo", "memory");
>   }
>-  echo_est[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i],
>-                                      far_spectrum[i]);
>+  echo_est[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i], far_spectrum[i]);
> }
> 
> void WebRtcAecm_ResetAdaptiveChannel_mips(AecmCore* aecm) {
>@@ -579,31 +723,28 @@ void WebRtcAecm_ResetAdaptiveChannel_mips(AecmCore* aecm) {
> 
>   // The stored channel has a significantly lower MSE than the adaptive one for
>   // two consecutive calculations. Reset the adaptive channel.
>-  memcpy(aecm->channelAdapt16,
>-         aecm->channelStored,
>+  memcpy(aecm->channelAdapt16, aecm->channelStored,
>          sizeof(int16_t) * PART_LEN1);
> 
>   // Restore the W32 channel
>   for (i = 0; i < PART_LEN; i += 4) {
>-    __asm __volatile (
>-      "ulw            %[temp1], 0(%[temp0])           \n\t"
>-      "ulw            %[temp4], 4(%[temp0])           \n\t"
>-      "preceq.w.phl   %[temp2], %[temp1]              \n\t"
>-      "preceq.w.phr   %[temp1], %[temp1]              \n\t"
>-      "preceq.w.phl   %[temp5], %[temp4]              \n\t"
>-      "preceq.w.phr   %[temp4], %[temp4]              \n\t"
>-      "addiu          %[temp0], %[temp0], 8           \n\t"
>-      "usw            %[temp2], 4(%[temp3])           \n\t"
>-      "usw            %[temp1], 0(%[temp3])           \n\t"
>-      "usw            %[temp5], 12(%[temp3])          \n\t"
>-      "usw            %[temp4], 8(%[temp3])           \n\t"
>-      "addiu          %[temp3], %[temp3], 16          \n\t"
>-      : [temp1] "=&r" (temp1), [temp2] "=&r" (temp2),
>-        [temp4] "=&r" (temp4), [temp5] "=&r" (temp5),
>-        [temp3] "+r" (temp3), [temp0] "+r" (temp0)
>-      :
>-      : "memory"
>-    );
>+    __asm __volatile(
>+        "ulw            %[temp1], 0(%[temp0])           \n\t"
>+        "ulw            %[temp4], 4(%[temp0])           \n\t"
>+        "preceq.w.phl   %[temp2], %[temp1]              \n\t"
>+        "preceq.w.phr   %[temp1], %[temp1]              \n\t"
>+        "preceq.w.phl   %[temp5], %[temp4]              \n\t"
>+        "preceq.w.phr   %[temp4], %[temp4]              \n\t"
>+        "addiu          %[temp0], %[temp0], 8           \n\t"
>+        "usw            %[temp2], 4(%[temp3])           \n\t"
>+        "usw            %[temp1], 0(%[temp3])           \n\t"
>+        "usw            %[temp5], 12(%[temp3])          \n\t"
>+        "usw            %[temp4], 8(%[temp3])           \n\t"
>+        "addiu          %[temp3], %[temp3], 16          \n\t"
>+        : [temp1] "=&r"(temp1), [temp2] "=&r"(temp2), [temp4] "=&r"(temp4),
>+          [temp5] "=&r"(temp5), [temp3] "+r"(temp3), [temp0] "+r"(temp0)
>+        :
>+        : "memory");
>   }
> 
>   aecm->channelAdapt32[i] = (int32_t)aecm->channelStored[i] << 16;
>@@ -633,7 +774,7 @@ static int TimeToFrequencyDomain(AecmCore* aecm,
> 
>   // In fft_buf, +16 for 32-byte alignment.
>   int16_t fft_buf[PART_LEN4 + 16];
>-  int16_t *fft = (int16_t *) (((uintptr_t) fft_buf + 31) & ~31);
>+  int16_t* fft = (int16_t*)(((uintptr_t)fft_buf + 31) & ~31);
> 
>   int16_t tmp16no1;
> #if !defined(MIPS_DSP_R2_LE)
>@@ -662,25 +803,18 @@ static int TimeToFrequencyDomain(AecmCore* aecm,
>   freq_signal[PART_LEN].imag = 0;
>   freq_signal[PART_LEN].real = fft[PART_LEN2];
>   freq_signal_abs[0] = (uint16_t)WEBRTC_SPL_ABS_W16(freq_signal[0].real);
>-  freq_signal_abs[PART_LEN] = (uint16_t)WEBRTC_SPL_ABS_W16(
>-    freq_signal[PART_LEN].real);
>-  (*freq_signal_sum_abs) = (uint32_t)(freq_signal_abs[0]) +
>-    (uint32_t)(freq_signal_abs[PART_LEN]);
>+  freq_signal_abs[PART_LEN] =
>+      (uint16_t)WEBRTC_SPL_ABS_W16(freq_signal[PART_LEN].real);
>+  (*freq_signal_sum_abs) =
>+      (uint32_t)(freq_signal_abs[0]) + (uint32_t)(freq_signal_abs[PART_LEN]);
> 
> #if !defined(MIPS_DSP_R2_LE)
>   for (i = 1; i < PART_LEN; i++) {
>-    if (freq_signal[i].real == 0)
>-    {
>-      freq_signal_abs[i] = (uint16_t)WEBRTC_SPL_ABS_W16(
>-        freq_signal[i].imag);
>-    }
>-    else if (freq_signal[i].imag == 0)
>-    {
>-      freq_signal_abs[i] = (uint16_t)WEBRTC_SPL_ABS_W16(
>-        freq_signal[i].real);
>-    }
>-    else
>-    {
>+    if (freq_signal[i].real == 0) {
>+      freq_signal_abs[i] = (uint16_t)WEBRTC_SPL_ABS_W16(freq_signal[i].imag);
>+    } else if (freq_signal[i].imag == 0) {
>+      freq_signal_abs[i] = (uint16_t)WEBRTC_SPL_ABS_W16(freq_signal[i].real);
>+    } else {
>       // Approximation for magnitude of complex fft output
>       // magn = sqrt(real^2 + imag^2)
>       // magn ~= alpha * max(|imag|,|real|) + beta * min(|imag|,|real|)
>@@ -697,32 +831,30 @@ static int TimeToFrequencyDomain(AecmCore* aecm,
>     }
>     (*freq_signal_sum_abs) += (uint32_t)freq_signal_abs[i];
>   }
>-#else // #if !defined(MIPS_DSP_R2_LE)
>-  freqs = (uint32_t)(freq_signal_abs[0]) +
>-          (uint32_t)(freq_signal_abs[PART_LEN]);
>+#else  // #if !defined(MIPS_DSP_R2_LE)
>+  freqs =
>+      (uint32_t)(freq_signal_abs[0]) + (uint32_t)(freq_signal_abs[PART_LEN]);
>   freqp = &(freq_signal[1].real);
> 
>-  __asm __volatile (
>-    "lw             %[freqt0],      0(%[freqp])             \n\t"
>-    "lw             %[freqt1],      4(%[freqp])             \n\t"
>-    "lw             %[freqt2],      8(%[freqp])             \n\t"
>-    "mult           $ac0,           $zero,      $zero       \n\t"
>-    "mult           $ac1,           $zero,      $zero       \n\t"
>-    "mult           $ac2,           $zero,      $zero       \n\t"
>-    "dpaq_s.w.ph    $ac0,           %[freqt0],  %[freqt0]   \n\t"
>-    "dpaq_s.w.ph    $ac1,           %[freqt1],  %[freqt1]   \n\t"
>-    "dpaq_s.w.ph    $ac2,           %[freqt2],  %[freqt2]   \n\t"
>-    "addiu          %[freqp],       %[freqp],   12          \n\t"
>-    "extr.w         %[tmp32no20],   $ac0,       1           \n\t"
>-    "extr.w         %[tmp32no21],   $ac1,       1           \n\t"
>-    "extr.w         %[tmp32no22],   $ac2,       1           \n\t"
>-    : [freqt0] "=&r" (freqt0), [freqt1] "=&r" (freqt1),
>-      [freqt2] "=&r" (freqt2), [freqp] "+r" (freqp),
>-      [tmp32no20] "=r" (tmp32no20), [tmp32no21] "=r" (tmp32no21),
>-      [tmp32no22] "=r" (tmp32no22)
>-    :
>-    : "memory", "hi", "lo", "$ac1hi", "$ac1lo", "$ac2hi", "$ac2lo"
>-  );
>+  __asm __volatile(
>+      "lw             %[freqt0],      0(%[freqp])             \n\t"
>+      "lw             %[freqt1],      4(%[freqp])             \n\t"
>+      "lw             %[freqt2],      8(%[freqp])             \n\t"
>+      "mult           $ac0,           $zero,      $zero       \n\t"
>+      "mult           $ac1,           $zero,      $zero       \n\t"
>+      "mult           $ac2,           $zero,      $zero       \n\t"
>+      "dpaq_s.w.ph    $ac0,           %[freqt0],  %[freqt0]   \n\t"
>+      "dpaq_s.w.ph    $ac1,           %[freqt1],  %[freqt1]   \n\t"
>+      "dpaq_s.w.ph    $ac2,           %[freqt2],  %[freqt2]   \n\t"
>+      "addiu          %[freqp],       %[freqp],   12          \n\t"
>+      "extr.w         %[tmp32no20],   $ac0,       1           \n\t"
>+      "extr.w         %[tmp32no21],   $ac1,       1           \n\t"
>+      "extr.w         %[tmp32no22],   $ac2,       1           \n\t"
>+      : [freqt0] "=&r"(freqt0), [freqt1] "=&r"(freqt1), [freqt2] "=&r"(freqt2),
>+        [freqp] "+r"(freqp), [tmp32no20] "=r"(tmp32no20),
>+        [tmp32no21] "=r"(tmp32no21), [tmp32no22] "=r"(tmp32no22)
>+      :
>+      : "memory", "hi", "lo", "$ac1hi", "$ac1lo", "$ac2hi", "$ac2lo");
> 
>   tmp32no10 = WebRtcSpl_SqrtFloor(tmp32no20);
>   tmp32no11 = WebRtcSpl_SqrtFloor(tmp32no21);
>@@ -734,57 +866,54 @@ static int TimeToFrequencyDomain(AecmCore* aecm,
>   freqs += (uint32_t)tmp32no11;
>   freqs += (uint32_t)tmp32no12;
>   freqabsp = &(freq_signal_abs[4]);
>-  for (i = 4; i < PART_LEN; i+=4)
>-  {
>-    __asm __volatile (
>-      "ulw            %[freqt0],      0(%[freqp])                 \n\t"
>-      "ulw            %[freqt1],      4(%[freqp])                 \n\t"
>-      "ulw            %[freqt2],      8(%[freqp])                 \n\t"
>-      "ulw            %[freqt3],      12(%[freqp])                \n\t"
>-      "mult           $ac0,           $zero,          $zero       \n\t"
>-      "mult           $ac1,           $zero,          $zero       \n\t"
>-      "mult           $ac2,           $zero,          $zero       \n\t"
>-      "mult           $ac3,           $zero,          $zero       \n\t"
>-      "dpaq_s.w.ph    $ac0,           %[freqt0],      %[freqt0]   \n\t"
>-      "dpaq_s.w.ph    $ac1,           %[freqt1],      %[freqt1]   \n\t"
>-      "dpaq_s.w.ph    $ac2,           %[freqt2],      %[freqt2]   \n\t"
>-      "dpaq_s.w.ph    $ac3,           %[freqt3],      %[freqt3]   \n\t"
>-      "addiu          %[freqp],       %[freqp],       16          \n\t"
>-      "addiu          %[freqabsp],    %[freqabsp],    8           \n\t"
>-      "extr.w         %[tmp32no20],   $ac0,           1           \n\t"
>-      "extr.w         %[tmp32no21],   $ac1,           1           \n\t"
>-      "extr.w         %[tmp32no22],   $ac2,           1           \n\t"
>-      "extr.w         %[tmp32no23],   $ac3,           1           \n\t"
>-      : [freqt0] "=&r" (freqt0), [freqt1] "=&r" (freqt1),
>-        [freqt2] "=&r" (freqt2), [freqt3] "=&r" (freqt3),
>-        [tmp32no20] "=r" (tmp32no20), [tmp32no21] "=r" (tmp32no21),
>-        [tmp32no22] "=r" (tmp32no22), [tmp32no23] "=r" (tmp32no23),
>-        [freqabsp] "+r" (freqabsp), [freqp] "+r" (freqp)
>-      :
>-      : "memory", "hi", "lo", "$ac1hi", "$ac1lo",
>-        "$ac2hi", "$ac2lo", "$ac3hi", "$ac3lo"
>-    );
>+  for (i = 4; i < PART_LEN; i += 4) {
>+    __asm __volatile(
>+        "ulw            %[freqt0],      0(%[freqp])                 \n\t"
>+        "ulw            %[freqt1],      4(%[freqp])                 \n\t"
>+        "ulw            %[freqt2],      8(%[freqp])                 \n\t"
>+        "ulw            %[freqt3],      12(%[freqp])                \n\t"
>+        "mult           $ac0,           $zero,          $zero       \n\t"
>+        "mult           $ac1,           $zero,          $zero       \n\t"
>+        "mult           $ac2,           $zero,          $zero       \n\t"
>+        "mult           $ac3,           $zero,          $zero       \n\t"
>+        "dpaq_s.w.ph    $ac0,           %[freqt0],      %[freqt0]   \n\t"
>+        "dpaq_s.w.ph    $ac1,           %[freqt1],      %[freqt1]   \n\t"
>+        "dpaq_s.w.ph    $ac2,           %[freqt2],      %[freqt2]   \n\t"
>+        "dpaq_s.w.ph    $ac3,           %[freqt3],      %[freqt3]   \n\t"
>+        "addiu          %[freqp],       %[freqp],       16          \n\t"
>+        "addiu          %[freqabsp],    %[freqabsp],    8           \n\t"
>+        "extr.w         %[tmp32no20],   $ac0,           1           \n\t"
>+        "extr.w         %[tmp32no21],   $ac1,           1           \n\t"
>+        "extr.w         %[tmp32no22],   $ac2,           1           \n\t"
>+        "extr.w         %[tmp32no23],   $ac3,           1           \n\t"
>+        : [freqt0] "=&r"(freqt0), [freqt1] "=&r"(freqt1),
>+          [freqt2] "=&r"(freqt2), [freqt3] "=&r"(freqt3),
>+          [tmp32no20] "=r"(tmp32no20), [tmp32no21] "=r"(tmp32no21),
>+          [tmp32no22] "=r"(tmp32no22), [tmp32no23] "=r"(tmp32no23),
>+          [freqabsp] "+r"(freqabsp), [freqp] "+r"(freqp)
>+        :
>+        : "memory", "hi", "lo", "$ac1hi", "$ac1lo", "$ac2hi", "$ac2lo",
>+          "$ac3hi", "$ac3lo");
> 
>     tmp32no10 = WebRtcSpl_SqrtFloor(tmp32no20);
>     tmp32no11 = WebRtcSpl_SqrtFloor(tmp32no21);
>     tmp32no12 = WebRtcSpl_SqrtFloor(tmp32no22);
>     tmp32no13 = WebRtcSpl_SqrtFloor(tmp32no23);
> 
>-    __asm __volatile (
>-      "sh             %[tmp32no10],   -8(%[freqabsp])                 \n\t"
>-      "sh             %[tmp32no11],   -6(%[freqabsp])                 \n\t"
>-      "sh             %[tmp32no12],   -4(%[freqabsp])                 \n\t"
>-      "sh             %[tmp32no13],   -2(%[freqabsp])                 \n\t"
>-      "addu           %[freqs],       %[freqs],       %[tmp32no10]    \n\t"
>-      "addu           %[freqs],       %[freqs],       %[tmp32no11]    \n\t"
>-      "addu           %[freqs],       %[freqs],       %[tmp32no12]    \n\t"
>-      "addu           %[freqs],       %[freqs],       %[tmp32no13]    \n\t"
>-      : [freqs] "+r" (freqs)
>-      : [tmp32no10] "r" (tmp32no10), [tmp32no11] "r" (tmp32no11),
>-        [tmp32no12] "r" (tmp32no12), [tmp32no13] "r" (tmp32no13),
>-        [freqabsp] "r" (freqabsp)
>-      : "memory"
>-    );
>+    __asm __volatile(
>+        "sh             %[tmp32no10],   -8(%[freqabsp])                 \n\t"
>+        "sh             %[tmp32no11],   -6(%[freqabsp])                 \n\t"
>+        "sh             %[tmp32no12],   -4(%[freqabsp])                 \n\t"
>+        "sh             %[tmp32no13],   -2(%[freqabsp])                 \n\t"
>+        "addu           %[freqs],       %[freqs],       %[tmp32no10]    \n\t"
>+        "addu           %[freqs],       %[freqs],       %[tmp32no11]    \n\t"
>+        "addu           %[freqs],       %[freqs],       %[tmp32no12]    \n\t"
>+        "addu           %[freqs],       %[freqs],       %[tmp32no13]    \n\t"
>+        : [freqs] "+r"(freqs)
>+        : [tmp32no10] "r"(tmp32no10), [tmp32no11] "r"(tmp32no11),
>+          [tmp32no12] "r"(tmp32no12), [tmp32no13] "r"(tmp32no13),
>+          [freqabsp] "r"(freqabsp)
>+        : "memory");
>   }
> 
>   (*freq_signal_sum_abs) = freqs;
>@@ -813,13 +942,13 @@ int WebRtcAecm_ProcessBlock(AecmCore* aecm,
>   const uint16_t* far_spectrum_ptr = NULL;
> 
>   // 32 byte aligned buffers (with +8 or +16).
>-  int16_t fft_buf[PART_LEN4 + 2 + 16]; // +2 to make a loop safe.
>+  int16_t fft_buf[PART_LEN4 + 2 + 16];  // +2 to make a loop safe.
>   int32_t echoEst32_buf[PART_LEN1 + 8];
>   int32_t dfw_buf[PART_LEN2 + 8];
>   int32_t efw_buf[PART_LEN2 + 8];
> 
>-  int16_t* fft = (int16_t*)(((uint32_t)fft_buf + 31) & ~ 31);
>-  int32_t* echoEst32 = (int32_t*)(((uint32_t)echoEst32_buf + 31) & ~ 31);
>+  int16_t* fft = (int16_t*)(((uint32_t)fft_buf + 31) & ~31);
>+  int32_t* echoEst32 = (int32_t*)(((uint32_t)echoEst32_buf + 31) & ~31);
>   ComplexInt16* dfw = (ComplexInt16*)(((uint32_t)dfw_buf + 31) & ~31);
>   ComplexInt16* efw = (ComplexInt16*)(((uint32_t)efw_buf + 31) & ~31);
> 
>@@ -856,35 +985,25 @@ int WebRtcAecm_ProcessBlock(AecmCore* aecm,
>   // (2) the rest
> 
>   if (aecm->startupState < 2) {
>-    aecm->startupState = (aecm->totCount >= CONV_LEN) +
>-                         (aecm->totCount >= CONV_LEN2);
>+    aecm->startupState =
>+        (aecm->totCount >= CONV_LEN) + (aecm->totCount >= CONV_LEN2);
>   }
>   // END: Determine startup state
> 
>   // Buffer near and far end signals
>   memcpy(aecm->xBuf + PART_LEN, farend, sizeof(int16_t) * PART_LEN);
>-  memcpy(aecm->dBufNoisy + PART_LEN,
>-         nearendNoisy,
>-         sizeof(int16_t) * PART_LEN);
>+  memcpy(aecm->dBufNoisy + PART_LEN, nearendNoisy, sizeof(int16_t) * PART_LEN);
>   if (nearendClean != NULL) {
>-    memcpy(aecm->dBufClean + PART_LEN,
>-           nearendClean,
>+    memcpy(aecm->dBufClean + PART_LEN, nearendClean,
>            sizeof(int16_t) * PART_LEN);
>   }
> 
>   // Transform far end signal from time domain to frequency domain.
>-  far_q = TimeToFrequencyDomain(aecm,
>-                                aecm->xBuf,
>-                                dfw,
>-                                xfa,
>-                                &xfaSum);
>+  far_q = TimeToFrequencyDomain(aecm, aecm->xBuf, dfw, xfa, &xfaSum);
> 
>   // Transform noisy near end signal from time domain to frequency domain.
>-  zerosDBufNoisy = TimeToFrequencyDomain(aecm,
>-                                         aecm->dBufNoisy,
>-                                         dfw,
>-                                         dfaNoisy,
>-                                         &dfaNoisySum);
>+  zerosDBufNoisy =
>+      TimeToFrequencyDomain(aecm, aecm->dBufNoisy, dfw, dfaNoisy, &dfaNoisySum);
>   aecm->dfaNoisyQDomainOld = aecm->dfaNoisyQDomain;
>   aecm->dfaNoisyQDomain = (int16_t)zerosDBufNoisy;
> 
>@@ -895,10 +1014,7 @@ int WebRtcAecm_ProcessBlock(AecmCore* aecm,
>     dfaCleanSum = dfaNoisySum;
>   } else {
>     // Transform clean near end signal from time domain to frequency domain.
>-    zerosDBufClean = TimeToFrequencyDomain(aecm,
>-                                           aecm->dBufClean,
>-                                           dfw,
>-                                           dfaClean,
>+    zerosDBufClean = TimeToFrequencyDomain(aecm, aecm->dBufClean, dfw, dfaClean,
>                                            &dfaCleanSum);
>     aecm->dfaCleanQDomainOld = aecm->dfaCleanQDomain;
>     aecm->dfaCleanQDomain = (int16_t)zerosDBufClean;
>@@ -912,14 +1028,11 @@ int WebRtcAecm_ProcessBlock(AecmCore* aecm,
>                                far_q) == -1) {
>     return -1;
>   }
>-  delay = WebRtc_DelayEstimatorProcessFix(aecm->delay_estimator,
>-                                          dfaNoisy,
>-                                          PART_LEN1,
>-                                          zerosDBufNoisy);
>+  delay = WebRtc_DelayEstimatorProcessFix(aecm->delay_estimator, dfaNoisy,
>+                                          PART_LEN1, zerosDBufNoisy);
>   if (delay == -1) {
>     return -1;
>-  }
>-  else if (delay == -2) {
>+  } else if (delay == -2) {
>     // If the delay is unknown, we assume zero.
>     // NOTE: this will have to be adjusted if we ever add lookahead.
>     delay = 0;
>@@ -932,17 +1045,14 @@ int WebRtcAecm_ProcessBlock(AecmCore* aecm,
> 
>   // Get aligned far end spectrum
>   far_spectrum_ptr = WebRtcAecm_AlignedFarend(aecm, &far_q, delay);
>-  zerosXBuf = (int16_t) far_q;
>+  zerosXBuf = (int16_t)far_q;
> 
>   if (far_spectrum_ptr == NULL) {
>     return -1;
>   }
> 
>   // Calculate log(energy) and update energy threshold levels
>-  WebRtcAecm_CalcEnergies(aecm,
>-                          far_spectrum_ptr,
>-                          zerosXBuf,
>-                          dfaNoisySum,
>+  WebRtcAecm_CalcEnergies(aecm, far_spectrum_ptr, zerosXBuf, dfaNoisySum,
>                           echoEst32);
>   // Calculate stepsize
>   mu = WebRtcAecm_CalcStepSize(aecm);
>@@ -953,11 +1063,7 @@ int WebRtcAecm_ProcessBlock(AecmCore* aecm,
>   // This is the channel estimation algorithm.
>   // It is base on NLMS but has a variable step length,
>   // which was calculated above.
>-  WebRtcAecm_UpdateChannel(aecm,
>-                           far_spectrum_ptr,
>-                           zerosXBuf,
>-                           dfaNoisy,
>-                           mu,
>+  WebRtcAecm_UpdateChannel(aecm, far_spectrum_ptr, zerosXBuf, dfaNoisy, mu,
>                            echoEst32);
> 
>   supGain = WebRtcAecm_CalcSuppressionGain(aecm);
>@@ -976,19 +1082,18 @@ int WebRtcAecm_ProcessBlock(AecmCore* aecm,
>       // Multiplication is safe
>       // Result in
>       // Q(RESOLUTION_CHANNEL+RESOLUTION_SUPGAIN+aecm->xfaQDomainBuf[diff])
>-      echoEst32Gained = WEBRTC_SPL_UMUL_32_16((uint32_t)aecm->echoFilt[i],
>-                                              (uint16_t)supGain);
>+      echoEst32Gained =
>+          WEBRTC_SPL_UMUL_32_16((uint32_t)aecm->echoFilt[i], (uint16_t)supGain);
>       resolutionDiff = 14 - RESOLUTION_CHANNEL16 - RESOLUTION_SUPGAIN;
>       resolutionDiff += (aecm->dfaCleanQDomain - zerosXBuf);
>     } else {
>       tmp16no1 = 17 - zeros32 - zeros16;
>-      resolutionDiff = 14 + tmp16no1 - RESOLUTION_CHANNEL16 -
>-                       RESOLUTION_SUPGAIN;
>+      resolutionDiff =
>+          14 + tmp16no1 - RESOLUTION_CHANNEL16 - RESOLUTION_SUPGAIN;
>       resolutionDiff += (aecm->dfaCleanQDomain - zerosXBuf);
>       if (zeros32 > tmp16no1) {
>-        echoEst32Gained = WEBRTC_SPL_UMUL_32_16(
>-                            (uint32_t)aecm->echoFilt[i],
>-                            supGain >> tmp16no1);
>+        echoEst32Gained = WEBRTC_SPL_UMUL_32_16((uint32_t)aecm->echoFilt[i],
>+                                                supGain >> tmp16no1);
>       } else {
>         // Result in Q-(RESOLUTION_CHANNEL+RESOLUTION_SUPGAIN-16)
>         echoEst32Gained = (aecm->echoFilt[i] >> tmp16no1) * supGain;
>@@ -1004,8 +1109,8 @@ int WebRtcAecm_ProcessBlock(AecmCore* aecm,
>       tmp16no2 = ptrDfaClean[i] >> -qDomainDiff;
>     } else {
>       tmp16no1 = dfa_clean_q_domain_diff < 0
>-          ? aecm->nearFilt[i] >> -dfa_clean_q_domain_diff
>-          : aecm->nearFilt[i] << dfa_clean_q_domain_diff;
>+                     ? aecm->nearFilt[i] >> -dfa_clean_q_domain_diff
>+                     : aecm->nearFilt[i] << dfa_clean_q_domain_diff;
>       qDomainDiff = 0;
>       tmp16no2 = ptrDfaClean[i];
>     }
>@@ -1017,8 +1122,8 @@ int WebRtcAecm_ProcessBlock(AecmCore* aecm,
>     if ((tmp16no2) & (-qDomainDiff > zeros16)) {
>       aecm->nearFilt[i] = WEBRTC_SPL_WORD16_MAX;
>     } else {
>-      aecm->nearFilt[i] = qDomainDiff < 0 ? tmp16no2 << -qDomainDiff
>-                                          : tmp16no2 >> qDomainDiff;
>+      aecm->nearFilt[i] =
>+          qDomainDiff < 0 ? tmp16no2 << -qDomainDiff : tmp16no2 >> qDomainDiff;
>     }
> 
>     // Wiener filter coefficients, resulting hnl in Q14
>@@ -1031,8 +1136,8 @@ int WebRtcAecm_ProcessBlock(AecmCore* aecm,
>       // Multiply the suppression gain
>       // Rounding
>       echoEst32Gained += (uint32_t)(aecm->nearFilt[i] >> 1);
>-      tmpU32 = WebRtcSpl_DivU32U16(echoEst32Gained,
>-                                   (uint16_t)aecm->nearFilt[i]);
>+      tmpU32 =
>+          WebRtcSpl_DivU32U16(echoEst32Gained, (uint16_t)aecm->nearFilt[i]);
> 
>       // Current resolution is
>       // Q-(RESOLUTION_CHANNEL + RESOLUTION_SUPGAIN
>@@ -1062,58 +1167,56 @@ int WebRtcAecm_ProcessBlock(AecmCore* aecm,
>     // TODO(bjornv): Investigate if the scaling of hnl[i] below can cause
>     //               speech distortion in double-talk.
>     for (i = 0; i < (PART_LEN1 >> 3); i++) {
>-      __asm __volatile (
>-        "lh         %[temp1],       0(%[ptr1])                  \n\t"
>-        "lh         %[temp2],       2(%[ptr1])                  \n\t"
>-        "lh         %[temp3],       4(%[ptr1])                  \n\t"
>-        "lh         %[temp4],       6(%[ptr1])                  \n\t"
>-        "lh         %[temp5],       8(%[ptr1])                  \n\t"
>-        "lh         %[temp6],       10(%[ptr1])                 \n\t"
>-        "lh         %[temp7],       12(%[ptr1])                 \n\t"
>-        "lh         %[temp8],       14(%[ptr1])                 \n\t"
>-        "mul        %[temp1],       %[temp1],       %[temp1]    \n\t"
>-        "mul        %[temp2],       %[temp2],       %[temp2]    \n\t"
>-        "mul        %[temp3],       %[temp3],       %[temp3]    \n\t"
>-        "mul        %[temp4],       %[temp4],       %[temp4]    \n\t"
>-        "mul        %[temp5],       %[temp5],       %[temp5]    \n\t"
>-        "mul        %[temp6],       %[temp6],       %[temp6]    \n\t"
>-        "mul        %[temp7],       %[temp7],       %[temp7]    \n\t"
>-        "mul        %[temp8],       %[temp8],       %[temp8]    \n\t"
>-        "sra        %[temp1],       %[temp1],       14          \n\t"
>-        "sra        %[temp2],       %[temp2],       14          \n\t"
>-        "sra        %[temp3],       %[temp3],       14          \n\t"
>-        "sra        %[temp4],       %[temp4],       14          \n\t"
>-        "sra        %[temp5],       %[temp5],       14          \n\t"
>-        "sra        %[temp6],       %[temp6],       14          \n\t"
>-        "sra        %[temp7],       %[temp7],       14          \n\t"
>-        "sra        %[temp8],       %[temp8],       14          \n\t"
>-        "sh         %[temp1],       0(%[ptr1])                  \n\t"
>-        "sh         %[temp2],       2(%[ptr1])                  \n\t"
>-        "sh         %[temp3],       4(%[ptr1])                  \n\t"
>-        "sh         %[temp4],       6(%[ptr1])                  \n\t"
>-        "sh         %[temp5],       8(%[ptr1])                  \n\t"
>-        "sh         %[temp6],       10(%[ptr1])                 \n\t"
>-        "sh         %[temp7],       12(%[ptr1])                 \n\t"
>-        "sh         %[temp8],       14(%[ptr1])                 \n\t"
>-        "addiu      %[ptr1],        %[ptr1],        16          \n\t"
>-        : [temp1] "=&r" (temp1), [temp2] "=&r" (temp2), [temp3] "=&r" (temp3),
>-          [temp4] "=&r" (temp4), [temp5] "=&r" (temp5), [temp6] "=&r" (temp6),
>-          [temp7] "=&r" (temp7), [temp8] "=&r" (temp8), [ptr1] "+r" (ptr1)
>-        :
>-        : "memory", "hi", "lo"
>-      );
>+      __asm __volatile(
>+          "lh         %[temp1],       0(%[ptr1])                  \n\t"
>+          "lh         %[temp2],       2(%[ptr1])                  \n\t"
>+          "lh         %[temp3],       4(%[ptr1])                  \n\t"
>+          "lh         %[temp4],       6(%[ptr1])                  \n\t"
>+          "lh         %[temp5],       8(%[ptr1])                  \n\t"
>+          "lh         %[temp6],       10(%[ptr1])                 \n\t"
>+          "lh         %[temp7],       12(%[ptr1])                 \n\t"
>+          "lh         %[temp8],       14(%[ptr1])                 \n\t"
>+          "mul        %[temp1],       %[temp1],       %[temp1]    \n\t"
>+          "mul        %[temp2],       %[temp2],       %[temp2]    \n\t"
>+          "mul        %[temp3],       %[temp3],       %[temp3]    \n\t"
>+          "mul        %[temp4],       %[temp4],       %[temp4]    \n\t"
>+          "mul        %[temp5],       %[temp5],       %[temp5]    \n\t"
>+          "mul        %[temp6],       %[temp6],       %[temp6]    \n\t"
>+          "mul        %[temp7],       %[temp7],       %[temp7]    \n\t"
>+          "mul        %[temp8],       %[temp8],       %[temp8]    \n\t"
>+          "sra        %[temp1],       %[temp1],       14          \n\t"
>+          "sra        %[temp2],       %[temp2],       14          \n\t"
>+          "sra        %[temp3],       %[temp3],       14          \n\t"
>+          "sra        %[temp4],       %[temp4],       14          \n\t"
>+          "sra        %[temp5],       %[temp5],       14          \n\t"
>+          "sra        %[temp6],       %[temp6],       14          \n\t"
>+          "sra        %[temp7],       %[temp7],       14          \n\t"
>+          "sra        %[temp8],       %[temp8],       14          \n\t"
>+          "sh         %[temp1],       0(%[ptr1])                  \n\t"
>+          "sh         %[temp2],       2(%[ptr1])                  \n\t"
>+          "sh         %[temp3],       4(%[ptr1])                  \n\t"
>+          "sh         %[temp4],       6(%[ptr1])                  \n\t"
>+          "sh         %[temp5],       8(%[ptr1])                  \n\t"
>+          "sh         %[temp6],       10(%[ptr1])                 \n\t"
>+          "sh         %[temp7],       12(%[ptr1])                 \n\t"
>+          "sh         %[temp8],       14(%[ptr1])                 \n\t"
>+          "addiu      %[ptr1],        %[ptr1],        16          \n\t"
>+          : [temp1] "=&r"(temp1), [temp2] "=&r"(temp2), [temp3] "=&r"(temp3),
>+            [temp4] "=&r"(temp4), [temp5] "=&r"(temp5), [temp6] "=&r"(temp6),
>+            [temp7] "=&r"(temp7), [temp8] "=&r"(temp8), [ptr1] "+r"(ptr1)
>+          :
>+          : "memory", "hi", "lo");
>     }
>-    for(i = 0; i < (PART_LEN1 & 7); i++) {
>-      __asm __volatile (
>-        "lh         %[temp1],       0(%[ptr1])                  \n\t"
>-        "mul        %[temp1],       %[temp1],       %[temp1]    \n\t"
>-        "sra        %[temp1],       %[temp1],       14          \n\t"
>-        "sh         %[temp1],       0(%[ptr1])                  \n\t"
>-        "addiu      %[ptr1],        %[ptr1],        2           \n\t"
>-        : [temp1] "=&r" (temp1), [ptr1] "+r" (ptr1)
>-        :
>-        : "memory", "hi", "lo"
>-      );
>+    for (i = 0; i < (PART_LEN1 & 7); i++) {
>+      __asm __volatile(
>+          "lh         %[temp1],       0(%[ptr1])                  \n\t"
>+          "mul        %[temp1],       %[temp1],       %[temp1]    \n\t"
>+          "sra        %[temp1],       %[temp1],       14          \n\t"
>+          "sh         %[temp1],       0(%[ptr1])                  \n\t"
>+          "addiu      %[ptr1],        %[ptr1],        2           \n\t"
>+          : [temp1] "=&r"(temp1), [ptr1] "+r"(ptr1)
>+          :
>+          : "memory", "hi", "lo");
>     }
> 
>     for (i = kMinPrefBand; i <= kMaxPrefBand; i++) {
>@@ -1141,102 +1244,95 @@ int WebRtcAecm_ProcessBlock(AecmCore* aecm,
>     } else {
>       for (i = 0; i < PART_LEN1; i++) {
> #if defined(MIPS_DSP_R1_LE)
>-        __asm __volatile (
>-          ".set       push                                        \n\t"
>-          ".set       noreorder                                   \n\t"
>-          "lh         %[temp1],       0(%[ptr])                   \n\t"
>-          "lh         %[temp2],       0(%[dr_ptr])                \n\t"
>-          "slti       %[temp4],       %[temp1],       0x4001      \n\t"
>-          "beqz       %[temp4],       3f                          \n\t"
>-          " lh        %[temp3],       2(%[dr_ptr])                \n\t"
>-          "slti       %[temp5],       %[temp1],       3277        \n\t"
>-          "bnez       %[temp5],       2f                          \n\t"
>-          " addiu     %[dr_ptr],      %[dr_ptr],      4           \n\t"
>-          "mul        %[temp2],       %[temp2],       %[temp1]    \n\t"
>-          "mul        %[temp3],       %[temp3],       %[temp1]    \n\t"
>-          "shra_r.w   %[temp2],       %[temp2],       14          \n\t"
>-          "shra_r.w   %[temp3],       %[temp3],       14          \n\t"
>-          "b          4f                                          \n\t"
>-          " nop                                                   \n\t"
>-         "2:                                                      \n\t"
>-          "addu       %[temp1],       $zero,          $zero       \n\t"
>-          "addu       %[temp2],       $zero,          $zero       \n\t"
>-          "addu       %[temp3],       $zero,          $zero       \n\t"
>-          "b          1f                                          \n\t"
>-          " nop                                                   \n\t"
>-         "3:                                                      \n\t"
>-          "addiu      %[temp1],       $0,             0x4000      \n\t"
>-         "1:                                                      \n\t"
>-          "sh         %[temp1],       0(%[ptr])                   \n\t"
>-         "4:                                                      \n\t"
>-          "sh         %[temp2],       0(%[er_ptr])                \n\t"
>-          "sh         %[temp3],       2(%[er_ptr])                \n\t"
>-          "addiu      %[ptr],         %[ptr],         2           \n\t"
>-          "addiu      %[er_ptr],      %[er_ptr],      4           \n\t"
>-          ".set       pop                                         \n\t"
>-          : [temp1] "=&r" (temp1), [temp2] "=&r" (temp2), [temp3] "=&r" (temp3),
>-            [temp4] "=&r" (temp4), [temp5] "=&r" (temp5), [ptr] "+r" (ptr),
>-            [er_ptr] "+r" (er_ptr), [dr_ptr] "+r" (dr_ptr)
>-          :
>-          : "memory", "hi", "lo"
>-        );
>+        __asm __volatile(
>+            ".set       push                                        \n\t"
>+            ".set       noreorder                                   \n\t"
>+            "lh         %[temp1],       0(%[ptr])                   \n\t"
>+            "lh         %[temp2],       0(%[dr_ptr])                \n\t"
>+            "slti       %[temp4],       %[temp1],       0x4001      \n\t"
>+            "beqz       %[temp4],       3f                          \n\t"
>+            " lh        %[temp3],       2(%[dr_ptr])                \n\t"
>+            "slti       %[temp5],       %[temp1],       3277        \n\t"
>+            "bnez       %[temp5],       2f                          \n\t"
>+            " addiu     %[dr_ptr],      %[dr_ptr],      4           \n\t"
>+            "mul        %[temp2],       %[temp2],       %[temp1]    \n\t"
>+            "mul        %[temp3],       %[temp3],       %[temp1]    \n\t"
>+            "shra_r.w   %[temp2],       %[temp2],       14          \n\t"
>+            "shra_r.w   %[temp3],       %[temp3],       14          \n\t"
>+            "b          4f                                          \n\t"
>+            " nop                                                   \n\t"
>+            "2:                                                      \n\t"
>+            "addu       %[temp1],       $zero,          $zero       \n\t"
>+            "addu       %[temp2],       $zero,          $zero       \n\t"
>+            "addu       %[temp3],       $zero,          $zero       \n\t"
>+            "b          1f                                          \n\t"
>+            " nop                                                   \n\t"
>+            "3:                                                      \n\t"
>+            "addiu      %[temp1],       $0,             0x4000      \n\t"
>+            "1:                                                      \n\t"
>+            "sh         %[temp1],       0(%[ptr])                   \n\t"
>+            "4:                                                      \n\t"
>+            "sh         %[temp2],       0(%[er_ptr])                \n\t"
>+            "sh         %[temp3],       2(%[er_ptr])                \n\t"
>+            "addiu      %[ptr],         %[ptr],         2           \n\t"
>+            "addiu      %[er_ptr],      %[er_ptr],      4           \n\t"
>+            ".set       pop                                         \n\t"
>+            : [temp1] "=&r"(temp1), [temp2] "=&r"(temp2), [temp3] "=&r"(temp3),
>+              [temp4] "=&r"(temp4), [temp5] "=&r"(temp5), [ptr] "+r"(ptr),
>+              [er_ptr] "+r"(er_ptr), [dr_ptr] "+r"(dr_ptr)
>+            :
>+            : "memory", "hi", "lo");
> #else
>-        __asm __volatile (
>-          ".set       push                                        \n\t"
>-          ".set       noreorder                                   \n\t"
>-          "lh         %[temp1],       0(%[ptr])                   \n\t"
>-          "lh         %[temp2],       0(%[dr_ptr])                \n\t"
>-          "slti       %[temp4],       %[temp1],       0x4001      \n\t"
>-          "beqz       %[temp4],       3f                          \n\t"
>-          " lh        %[temp3],       2(%[dr_ptr])                \n\t"
>-          "slti       %[temp5],       %[temp1],       3277        \n\t"
>-          "bnez       %[temp5],       2f                          \n\t"
>-          " addiu     %[dr_ptr],      %[dr_ptr],      4           \n\t"
>-          "mul        %[temp2],       %[temp2],       %[temp1]    \n\t"
>-          "mul        %[temp3],       %[temp3],       %[temp1]    \n\t"
>-          "addiu      %[temp2],       %[temp2],       0x2000      \n\t"
>-          "addiu      %[temp3],       %[temp3],       0x2000      \n\t"
>-          "sra        %[temp2],       %[temp2],       14          \n\t"
>-          "sra        %[temp3],       %[temp3],       14          \n\t"
>-          "b          4f                                          \n\t"
>-          " nop                                                   \n\t"
>-         "2:                                                      \n\t"
>-          "addu       %[temp1],       $zero,          $zero       \n\t"
>-          "addu       %[temp2],       $zero,          $zero       \n\t"
>-          "addu       %[temp3],       $zero,          $zero       \n\t"
>-          "b          1f                                          \n\t"
>-          " nop                                                   \n\t"
>-         "3:                                                      \n\t"
>-          "addiu      %[temp1],       $0,             0x4000      \n\t"
>-         "1:                                                      \n\t"
>-          "sh         %[temp1],       0(%[ptr])                   \n\t"
>-         "4:                                                      \n\t"
>-          "sh         %[temp2],       0(%[er_ptr])                \n\t"
>-          "sh         %[temp3],       2(%[er_ptr])                \n\t"
>-          "addiu      %[ptr],         %[ptr],         2           \n\t"
>-          "addiu      %[er_ptr],      %[er_ptr],      4           \n\t"
>-          ".set       pop                                         \n\t"
>-          : [temp1] "=&r" (temp1), [temp2] "=&r" (temp2), [temp3] "=&r" (temp3),
>-            [temp4] "=&r" (temp4), [temp5] "=&r" (temp5), [ptr] "+r" (ptr),
>-            [er_ptr] "+r" (er_ptr), [dr_ptr] "+r" (dr_ptr)
>-          :
>-          : "memory", "hi", "lo"
>-        );
>+        __asm __volatile(
>+            ".set       push                                        \n\t"
>+            ".set       noreorder                                   \n\t"
>+            "lh         %[temp1],       0(%[ptr])                   \n\t"
>+            "lh         %[temp2],       0(%[dr_ptr])                \n\t"
>+            "slti       %[temp4],       %[temp1],       0x4001      \n\t"
>+            "beqz       %[temp4],       3f                          \n\t"
>+            " lh        %[temp3],       2(%[dr_ptr])                \n\t"
>+            "slti       %[temp5],       %[temp1],       3277        \n\t"
>+            "bnez       %[temp5],       2f                          \n\t"
>+            " addiu     %[dr_ptr],      %[dr_ptr],      4           \n\t"
>+            "mul        %[temp2],       %[temp2],       %[temp1]    \n\t"
>+            "mul        %[temp3],       %[temp3],       %[temp1]    \n\t"
>+            "addiu      %[temp2],       %[temp2],       0x2000      \n\t"
>+            "addiu      %[temp3],       %[temp3],       0x2000      \n\t"
>+            "sra        %[temp2],       %[temp2],       14          \n\t"
>+            "sra        %[temp3],       %[temp3],       14          \n\t"
>+            "b          4f                                          \n\t"
>+            " nop                                                   \n\t"
>+            "2:                                                      \n\t"
>+            "addu       %[temp1],       $zero,          $zero       \n\t"
>+            "addu       %[temp2],       $zero,          $zero       \n\t"
>+            "addu       %[temp3],       $zero,          $zero       \n\t"
>+            "b          1f                                          \n\t"
>+            " nop                                                   \n\t"
>+            "3:                                                      \n\t"
>+            "addiu      %[temp1],       $0,             0x4000      \n\t"
>+            "1:                                                      \n\t"
>+            "sh         %[temp1],       0(%[ptr])                   \n\t"
>+            "4:                                                      \n\t"
>+            "sh         %[temp2],       0(%[er_ptr])                \n\t"
>+            "sh         %[temp3],       2(%[er_ptr])                \n\t"
>+            "addiu      %[ptr],         %[ptr],         2           \n\t"
>+            "addiu      %[er_ptr],      %[er_ptr],      4           \n\t"
>+            ".set       pop                                         \n\t"
>+            : [temp1] "=&r"(temp1), [temp2] "=&r"(temp2), [temp3] "=&r"(temp3),
>+              [temp4] "=&r"(temp4), [temp5] "=&r"(temp5), [ptr] "+r"(ptr),
>+              [er_ptr] "+r"(er_ptr), [dr_ptr] "+r"(dr_ptr)
>+            :
>+            : "memory", "hi", "lo");
> #endif
>       }
>     }
>-  }
>-  else {
>+  } else {
>     // multiply with Wiener coefficients
>     for (i = 0; i < PART_LEN1; i++) {
>-      efw[i].real = (int16_t)
>-                      (WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].real,
>-                                                            hnl[i],
>-                                                            14));
>-      efw[i].imag = (int16_t)
>-                      (WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].imag,
>-                                                            hnl[i],
>-                                                            14));
>+      efw[i].real = (int16_t)(
>+          WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].real, hnl[i], 14));
>+      efw[i].imag = (int16_t)(
>+          WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].imag, hnl[i], 14));
>     }
>   }
> 
>@@ -1284,26 +1380,25 @@ static void ComfortNoise(AecmCore* aecm,
>   // Generate a uniform random array on [0 2^15-1].
>   WebRtcSpl_RandUArray(randW16, PART_LEN, &aecm->seed);
>   int16_t* randW16p = (int16_t*)randW16;
>-#if defined (MIPS_DSP_R1_LE)
>+#if defined(MIPS_DSP_R1_LE)
>   int16_t* kCosTablep = (int16_t*)WebRtcAecm_kCosTable;
>   int16_t* kSinTablep = (int16_t*)WebRtcAecm_kSinTable;
>-#endif   // #if defined(MIPS_DSP_R1_LE)
>+#endif  // #if defined(MIPS_DSP_R1_LE)
>   tmp1 = (int32_t*)aecm->noiseEst + 1;
>   dfap = (int16_t*)dfa + 1;
>   lambdap = (int16_t*)lambda + 1;
>   // Estimate noise power.
>-  for (i = 1; i < PART_LEN1; i+=2) {
>-  // Shift to the noise domain.
>-    __asm __volatile (
>-      "lh     %[tmp32],       0(%[dfap])                              \n\t"
>-      "lw     %[tnoise],      0(%[tmp1])                              \n\t"
>-      "sllv   %[outLShift32], %[tmp32],   %[shiftFromNearToNoise]     \n\t"
>-      : [tmp32] "=&r" (tmp32), [outLShift32] "=r" (outLShift32),
>-        [tnoise] "=&r" (tnoise)
>-      : [tmp1] "r" (tmp1), [dfap] "r" (dfap),
>-        [shiftFromNearToNoise] "r" (shiftFromNearToNoise)
>-      : "memory"
>-    );
>+  for (i = 1; i < PART_LEN1; i += 2) {
>+    // Shift to the noise domain.
>+    __asm __volatile(
>+        "lh     %[tmp32],       0(%[dfap])                              \n\t"
>+        "lw     %[tnoise],      0(%[tmp1])                              \n\t"
>+        "sllv   %[outLShift32], %[tmp32],   %[shiftFromNearToNoise]     \n\t"
>+        : [tmp32] "=&r"(tmp32), [outLShift32] "=r"(outLShift32),
>+          [tnoise] "=&r"(tnoise)
>+        : [tmp1] "r"(tmp1), [dfap] "r"(dfap),
>+          [shiftFromNearToNoise] "r"(shiftFromNearToNoise)
>+        : "memory");
> 
>     if (outLShift32 < tnoise) {
>       // Reset "too low" counter
>@@ -1319,13 +1414,13 @@ static void ComfortNoise(AecmCore* aecm,
>           aecm->noiseEstTooHighCtr[i] = 0;  // Reset the counter
>         }
>       } else {
>-        __asm __volatile (
>-          "subu   %[tmp32],       %[tnoise],      %[outLShift32]      \n\t"
>-          "srav   %[tmp32],       %[tmp32],       %[minTrackShift]    \n\t"
>-          "subu   %[tnoise],      %[tnoise],      %[tmp32]            \n\t"
>-          : [tmp32] "=&r" (tmp32), [tnoise] "+r" (tnoise)
>-          : [outLShift32] "r" (outLShift32), [minTrackShift] "r" (minTrackShift)
>-        );
>+        __asm __volatile(
>+            "subu   %[tmp32],       %[tnoise],      %[outLShift32]      \n\t"
>+            "srav   %[tmp32],       %[tmp32],       %[minTrackShift]    \n\t"
>+            "subu   %[tnoise],      %[tnoise],      %[tmp32]            \n\t"
>+            : [tmp32] "=&r"(tmp32), [tnoise] "+r"(tnoise)
>+            :
>+            [outLShift32] "r"(outLShift32), [minTrackShift] "r"(minTrackShift));
>       }
>     } else {
>       // Reset "too high" counter
>@@ -1334,53 +1429,49 @@ static void ComfortNoise(AecmCore* aecm,
>       if ((tnoise >> 19) <= 0) {
>         if ((tnoise >> 11) > 0) {
>           // Large enough for relative increase
>-          __asm __volatile (
>-            "mul    %[tnoise],  %[tnoise],  %[c2049]    \n\t"
>-            "sra    %[tnoise],  %[tnoise],  11          \n\t"
>-            : [tnoise] "+r" (tnoise)
>-            : [c2049] "r" (c2049)
>-            : "hi", "lo"
>-          );
>+          __asm __volatile(
>+              "mul    %[tnoise],  %[tnoise],  %[c2049]    \n\t"
>+              "sra    %[tnoise],  %[tnoise],  11          \n\t"
>+              : [tnoise] "+r"(tnoise)
>+              : [c2049] "r"(c2049)
>+              : "hi", "lo");
>         } else {
>           // Make incremental increases based on size every
>           // |kNoiseEstIncCount| block
>           aecm->noiseEstTooLowCtr[i]++;
>           if (aecm->noiseEstTooLowCtr[i] >= kNoiseEstIncCount) {
>-            __asm __volatile (
>-              "sra    %[tmp32],   %[tnoise],  9           \n\t"
>-              "addi   %[tnoise],  %[tnoise],  1           \n\t"
>-              "addu   %[tnoise],  %[tnoise],  %[tmp32]    \n\t"
>-              : [tnoise] "+r" (tnoise), [tmp32] "=&r" (tmp32)
>-              :
>-            );
>-            aecm->noiseEstTooLowCtr[i] = 0; // Reset counter
>+            __asm __volatile(
>+                "sra    %[tmp32],   %[tnoise],  9           \n\t"
>+                "addi   %[tnoise],  %[tnoise],  1           \n\t"
>+                "addu   %[tnoise],  %[tnoise],  %[tmp32]    \n\t"
>+                : [tnoise] "+r"(tnoise), [tmp32] "=&r"(tmp32)
>+                :);
>+            aecm->noiseEstTooLowCtr[i] = 0;  // Reset counter
>           }
>         }
>       } else {
>         // Avoid overflow.
>         // Multiplication with 2049 will cause wrap around. Scale
>         // down first and then multiply
>-        __asm __volatile (
>-          "sra    %[tnoise],  %[tnoise],  11          \n\t"
>-          "mul    %[tnoise],  %[tnoise],  %[c2049]    \n\t"
>-          : [tnoise] "+r" (tnoise)
>-          : [c2049] "r" (c2049)
>-          : "hi", "lo"
>-        );
>+        __asm __volatile(
>+            "sra    %[tnoise],  %[tnoise],  11          \n\t"
>+            "mul    %[tnoise],  %[tnoise],  %[c2049]    \n\t"
>+            : [tnoise] "+r"(tnoise)
>+            : [c2049] "r"(c2049)
>+            : "hi", "lo");
>       }
>     }
> 
>     // Shift to the noise domain.
>-    __asm __volatile (
>-      "lh     %[tmp32],       2(%[dfap])                              \n\t"
>-      "lw     %[tnoise1],     4(%[tmp1])                              \n\t"
>-      "addiu  %[dfap],        %[dfap],    4                           \n\t"
>-      "sllv   %[outLShift32], %[tmp32],   %[shiftFromNearToNoise]     \n\t"
>-      : [tmp32] "=&r" (tmp32), [dfap] "+r" (dfap),
>-        [outLShift32] "=r" (outLShift32), [tnoise1] "=&r" (tnoise1)
>-      : [tmp1] "r" (tmp1), [shiftFromNearToNoise] "r" (shiftFromNearToNoise)
>-      : "memory"
>-    );
>+    __asm __volatile(
>+        "lh     %[tmp32],       2(%[dfap])                              \n\t"
>+        "lw     %[tnoise1],     4(%[tmp1])                              \n\t"
>+        "addiu  %[dfap],        %[dfap],    4                           \n\t"
>+        "sllv   %[outLShift32], %[tmp32],   %[shiftFromNearToNoise]     \n\t"
>+        : [tmp32] "=&r"(tmp32), [dfap] "+r"(dfap),
>+          [outLShift32] "=r"(outLShift32), [tnoise1] "=&r"(tnoise1)
>+        : [tmp1] "r"(tmp1), [shiftFromNearToNoise] "r"(shiftFromNearToNoise)
>+        : "memory");
> 
>     if (outLShift32 < tnoise1) {
>       // Reset "too low" counter
>@@ -1393,16 +1484,16 @@ static void ComfortNoise(AecmCore* aecm,
>         aecm->noiseEstTooHighCtr[i + 1]++;
>         if (aecm->noiseEstTooHighCtr[i + 1] >= kNoiseEstIncCount) {
>           tnoise1--;
>-          aecm->noiseEstTooHighCtr[i + 1] = 0; // Reset the counter
>+          aecm->noiseEstTooHighCtr[i + 1] = 0;  // Reset the counter
>         }
>       } else {
>-        __asm __volatile (
>-          "subu   %[tmp32],       %[tnoise1],     %[outLShift32]      \n\t"
>-          "srav   %[tmp32],       %[tmp32],       %[minTrackShift]    \n\t"
>-          "subu   %[tnoise1],     %[tnoise1],     %[tmp32]            \n\t"
>-          : [tmp32] "=&r" (tmp32), [tnoise1] "+r" (tnoise1)
>-          : [outLShift32] "r" (outLShift32), [minTrackShift] "r" (minTrackShift)
>-        );
>+        __asm __volatile(
>+            "subu   %[tmp32],       %[tnoise1],     %[outLShift32]      \n\t"
>+            "srav   %[tmp32],       %[tmp32],       %[minTrackShift]    \n\t"
>+            "subu   %[tnoise1],     %[tnoise1],     %[tmp32]            \n\t"
>+            : [tmp32] "=&r"(tmp32), [tnoise1] "+r"(tnoise1)
>+            :
>+            [outLShift32] "r"(outLShift32), [minTrackShift] "r"(minTrackShift));
>       }
>     } else {
>       // Reset "too high" counter
>@@ -1411,59 +1502,55 @@ static void ComfortNoise(AecmCore* aecm,
>       if ((tnoise1 >> 19) <= 0) {
>         if ((tnoise1 >> 11) > 0) {
>           // Large enough for relative increase
>-          __asm __volatile (
>-            "mul    %[tnoise1], %[tnoise1], %[c2049]   \n\t"
>-            "sra    %[tnoise1], %[tnoise1], 11         \n\t"
>-            : [tnoise1] "+r" (tnoise1)
>-            : [c2049] "r" (c2049)
>-            : "hi", "lo"
>-          );
>+          __asm __volatile(
>+              "mul    %[tnoise1], %[tnoise1], %[c2049]   \n\t"
>+              "sra    %[tnoise1], %[tnoise1], 11         \n\t"
>+              : [tnoise1] "+r"(tnoise1)
>+              : [c2049] "r"(c2049)
>+              : "hi", "lo");
>         } else {
>           // Make incremental increases based on size every
>           // |kNoiseEstIncCount| block
>           aecm->noiseEstTooLowCtr[i + 1]++;
>           if (aecm->noiseEstTooLowCtr[i + 1] >= kNoiseEstIncCount) {
>-            __asm __volatile (
>-              "sra    %[tmp32],   %[tnoise1], 9           \n\t"
>-              "addi   %[tnoise1], %[tnoise1], 1           \n\t"
>-              "addu   %[tnoise1], %[tnoise1], %[tmp32]    \n\t"
>-              : [tnoise1] "+r" (tnoise1), [tmp32] "=&r" (tmp32)
>-              :
>-            );
>-            aecm->noiseEstTooLowCtr[i + 1] = 0; // Reset counter
>+            __asm __volatile(
>+                "sra    %[tmp32],   %[tnoise1], 9           \n\t"
>+                "addi   %[tnoise1], %[tnoise1], 1           \n\t"
>+                "addu   %[tnoise1], %[tnoise1], %[tmp32]    \n\t"
>+                : [tnoise1] "+r"(tnoise1), [tmp32] "=&r"(tmp32)
>+                :);
>+            aecm->noiseEstTooLowCtr[i + 1] = 0;  // Reset counter
>           }
>         }
>       } else {
>         // Avoid overflow.
>         // Multiplication with 2049 will cause wrap around. Scale
>         // down first and then multiply
>-        __asm __volatile (
>-          "sra    %[tnoise1], %[tnoise1], 11          \n\t"
>-          "mul    %[tnoise1], %[tnoise1], %[c2049]    \n\t"
>-          : [tnoise1] "+r" (tnoise1)
>-          : [c2049] "r" (c2049)
>-          : "hi", "lo"
>-        );
>+        __asm __volatile(
>+            "sra    %[tnoise1], %[tnoise1], 11          \n\t"
>+            "mul    %[tnoise1], %[tnoise1], %[c2049]    \n\t"
>+            : [tnoise1] "+r"(tnoise1)
>+            : [c2049] "r"(c2049)
>+            : "hi", "lo");
>       }
>     }
> 
>-    __asm __volatile (
>-      "lh     %[tmp16],   0(%[lambdap])                           \n\t"
>-      "lh     %[tmp161],  2(%[lambdap])                           \n\t"
>-      "sw     %[tnoise],  0(%[tmp1])                              \n\t"
>-      "sw     %[tnoise1], 4(%[tmp1])                              \n\t"
>-      "subu   %[tmp16],   %[c114],        %[tmp16]                \n\t"
>-      "subu   %[tmp161],  %[c114],        %[tmp161]               \n\t"
>-      "srav   %[tmp32],   %[tnoise],      %[shiftFromNearToNoise] \n\t"
>-      "srav   %[tmp321],  %[tnoise1],     %[shiftFromNearToNoise] \n\t"
>-      "addiu  %[lambdap], %[lambdap],     4                       \n\t"
>-      "addiu  %[tmp1],    %[tmp1],        8                       \n\t"
>-      : [tmp16] "=&r" (tmp16), [tmp161] "=&r" (tmp161), [tmp1] "+r" (tmp1),
>-        [tmp32] "=&r" (tmp32), [tmp321] "=&r" (tmp321), [lambdap] "+r" (lambdap)
>-      : [tnoise] "r" (tnoise), [tnoise1] "r" (tnoise1), [c114] "r" (c114),
>-        [shiftFromNearToNoise] "r" (shiftFromNearToNoise)
>-      : "memory"
>-    );
>+    __asm __volatile(
>+        "lh     %[tmp16],   0(%[lambdap])                           \n\t"
>+        "lh     %[tmp161],  2(%[lambdap])                           \n\t"
>+        "sw     %[tnoise],  0(%[tmp1])                              \n\t"
>+        "sw     %[tnoise1], 4(%[tmp1])                              \n\t"
>+        "subu   %[tmp16],   %[c114],        %[tmp16]                \n\t"
>+        "subu   %[tmp161],  %[c114],        %[tmp161]               \n\t"
>+        "srav   %[tmp32],   %[tnoise],      %[shiftFromNearToNoise] \n\t"
>+        "srav   %[tmp321],  %[tnoise1],     %[shiftFromNearToNoise] \n\t"
>+        "addiu  %[lambdap], %[lambdap],     4                       \n\t"
>+        "addiu  %[tmp1],    %[tmp1],        8                       \n\t"
>+        : [tmp16] "=&r"(tmp16), [tmp161] "=&r"(tmp161), [tmp1] "+r"(tmp1),
>+          [tmp32] "=&r"(tmp32), [tmp321] "=&r"(tmp321), [lambdap] "+r"(lambdap)
>+        : [tnoise] "r"(tnoise), [tnoise1] "r"(tnoise1), [c114] "r"(c114),
>+          [shiftFromNearToNoise] "r"(shiftFromNearToNoise)
>+        : "memory");
> 
>     if (tmp32 > 32767) {
>       tmp32 = 32767;
>@@ -1471,33 +1558,31 @@ static void ComfortNoise(AecmCore* aecm,
>     }
>     if (tmp321 > 32767) {
>       tmp321 = 32767;
>-      aecm->noiseEst[i+1] = tmp321 << shiftFromNearToNoise;
>+      aecm->noiseEst[i + 1] = tmp321 << shiftFromNearToNoise;
>     }
> 
>-    __asm __volatile (
>-      "mul    %[tmp32],   %[tmp32],       %[tmp16]                \n\t"
>-      "mul    %[tmp321],  %[tmp321],      %[tmp161]               \n\t"
>-      "sra    %[nrsh1],   %[tmp32],       14                      \n\t"
>-      "sra    %[nrsh2],   %[tmp321],      14                      \n\t"
>-      : [nrsh1] "=&r" (nrsh1), [nrsh2] "=r" (nrsh2)
>-      : [tmp16] "r" (tmp16), [tmp161] "r" (tmp161), [tmp32] "r" (tmp32),
>-        [tmp321] "r" (tmp321)
>-      : "memory", "hi", "lo"
>-    );
>-
>-    __asm __volatile (
>-      "lh     %[tmp32],       0(%[randW16p])              \n\t"
>-      "lh     %[tmp321],      2(%[randW16p])              \n\t"
>-      "addiu  %[randW16p],    %[randW16p],    4           \n\t"
>-      "mul    %[tmp32],       %[tmp32],       %[c359]     \n\t"
>-      "mul    %[tmp321],      %[tmp321],      %[c359]     \n\t"
>-      "sra    %[tmp16],       %[tmp32],       15          \n\t"
>-      "sra    %[tmp161],      %[tmp321],      15          \n\t"
>-      : [randW16p] "+r" (randW16p), [tmp32] "=&r" (tmp32),
>-        [tmp16] "=r" (tmp16), [tmp161] "=r" (tmp161), [tmp321] "=&r" (tmp321)
>-      : [c359] "r" (c359)
>-      : "memory", "hi", "lo"
>-    );
>+    __asm __volatile(
>+        "mul    %[tmp32],   %[tmp32],       %[tmp16]                \n\t"
>+        "mul    %[tmp321],  %[tmp321],      %[tmp161]               \n\t"
>+        "sra    %[nrsh1],   %[tmp32],       14                      \n\t"
>+        "sra    %[nrsh2],   %[tmp321],      14                      \n\t"
>+        : [nrsh1] "=&r"(nrsh1), [nrsh2] "=r"(nrsh2)
>+        : [tmp16] "r"(tmp16), [tmp161] "r"(tmp161), [tmp32] "r"(tmp32),
>+          [tmp321] "r"(tmp321)
>+        : "memory", "hi", "lo");
>+
>+    __asm __volatile(
>+        "lh     %[tmp32],       0(%[randW16p])              \n\t"
>+        "lh     %[tmp321],      2(%[randW16p])              \n\t"
>+        "addiu  %[randW16p],    %[randW16p],    4           \n\t"
>+        "mul    %[tmp32],       %[tmp32],       %[c359]     \n\t"
>+        "mul    %[tmp321],      %[tmp321],      %[c359]     \n\t"
>+        "sra    %[tmp16],       %[tmp32],       15          \n\t"
>+        "sra    %[tmp161],      %[tmp321],      15          \n\t"
>+        : [randW16p] "+r"(randW16p), [tmp32] "=&r"(tmp32), [tmp16] "=r"(tmp16),
>+          [tmp161] "=r"(tmp161), [tmp321] "=&r"(tmp321)
>+        : [c359] "r"(c359)
>+        : "memory", "hi", "lo");
> 
> #if !defined(MIPS_DSP_R1_LE)
>     tmp32 = WebRtcAecm_kCosTable[tmp16];
>@@ -1505,36 +1590,34 @@ static void ComfortNoise(AecmCore* aecm,
>     tmp322 = WebRtcAecm_kCosTable[tmp161];
>     tmp323 = WebRtcAecm_kSinTable[tmp161];
> #else
>-    __asm __volatile (
>-      "sll    %[tmp16],       %[tmp16],                   1           \n\t"
>-      "sll    %[tmp161],      %[tmp161],                  1           \n\t"
>-      "lhx    %[tmp32],       %[tmp16](%[kCosTablep])                 \n\t"
>-      "lhx    %[tmp321],      %[tmp16](%[kSinTablep])                 \n\t"
>-      "lhx    %[tmp322],      %[tmp161](%[kCosTablep])                \n\t"
>-      "lhx    %[tmp323],      %[tmp161](%[kSinTablep])                \n\t"
>-      : [tmp32] "=&r" (tmp32), [tmp321] "=&r" (tmp321),
>-        [tmp322] "=&r" (tmp322), [tmp323] "=&r" (tmp323)
>-      : [kCosTablep] "r" (kCosTablep), [tmp16] "r" (tmp16),
>-        [tmp161] "r" (tmp161), [kSinTablep] "r" (kSinTablep)
>-      : "memory"
>-    );
>+    __asm __volatile(
>+        "sll    %[tmp16],       %[tmp16],                   1           \n\t"
>+        "sll    %[tmp161],      %[tmp161],                  1           \n\t"
>+        "lhx    %[tmp32],       %[tmp16](%[kCosTablep])                 \n\t"
>+        "lhx    %[tmp321],      %[tmp16](%[kSinTablep])                 \n\t"
>+        "lhx    %[tmp322],      %[tmp161](%[kCosTablep])                \n\t"
>+        "lhx    %[tmp323],      %[tmp161](%[kSinTablep])                \n\t"
>+        : [tmp32] "=&r"(tmp32), [tmp321] "=&r"(tmp321), [tmp322] "=&r"(tmp322),
>+          [tmp323] "=&r"(tmp323)
>+        : [kCosTablep] "r"(kCosTablep), [tmp16] "r"(tmp16),
>+          [tmp161] "r"(tmp161), [kSinTablep] "r"(kSinTablep)
>+        : "memory");
> #endif
>-    __asm __volatile (
>-      "mul    %[tmp32],       %[tmp32],                   %[nrsh1]    \n\t"
>-      "negu   %[tmp162],      %[nrsh1]                                \n\t"
>-      "mul    %[tmp322],      %[tmp322],                  %[nrsh2]    \n\t"
>-      "negu   %[tmp163],      %[nrsh2]                                \n\t"
>-      "sra    %[tmp32],       %[tmp32],                   13          \n\t"
>-      "mul    %[tmp321],      %[tmp321],                  %[tmp162]   \n\t"
>-      "sra    %[tmp322],      %[tmp322],                  13          \n\t"
>-      "mul    %[tmp323],      %[tmp323],                  %[tmp163]   \n\t"
>-      "sra    %[tmp321],      %[tmp321],                  13          \n\t"
>-      "sra    %[tmp323],      %[tmp323],                  13          \n\t"
>-      : [tmp32] "+r" (tmp32), [tmp321] "+r" (tmp321), [tmp162] "=&r" (tmp162),
>-        [tmp322] "+r" (tmp322), [tmp323] "+r" (tmp323), [tmp163] "=&r" (tmp163)
>-      : [nrsh1] "r" (nrsh1), [nrsh2] "r" (nrsh2)
>-      : "hi", "lo"
>-    );
>+    __asm __volatile(
>+        "mul    %[tmp32],       %[tmp32],                   %[nrsh1]    \n\t"
>+        "negu   %[tmp162],      %[nrsh1]                                \n\t"
>+        "mul    %[tmp322],      %[tmp322],                  %[nrsh2]    \n\t"
>+        "negu   %[tmp163],      %[nrsh2]                                \n\t"
>+        "sra    %[tmp32],       %[tmp32],                   13          \n\t"
>+        "mul    %[tmp321],      %[tmp321],                  %[tmp162]   \n\t"
>+        "sra    %[tmp322],      %[tmp322],                  13          \n\t"
>+        "mul    %[tmp323],      %[tmp323],                  %[tmp163]   \n\t"
>+        "sra    %[tmp321],      %[tmp321],                  13          \n\t"
>+        "sra    %[tmp323],      %[tmp323],                  13          \n\t"
>+        : [tmp32] "+r"(tmp32), [tmp321] "+r"(tmp321), [tmp162] "=&r"(tmp162),
>+          [tmp322] "+r"(tmp322), [tmp323] "+r"(tmp323), [tmp163] "=&r"(tmp163)
>+        : [nrsh1] "r"(nrsh1), [nrsh2] "r"(nrsh2)
>+        : "hi", "lo");
>     // Tables are in Q13.
>     uReal[i] = (int16_t)tmp32;
>     uImag[i] = (int16_t)tmp321;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aecm/aecm_core_neon.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aecm/aecm_core_neon.cc
>index ca7211f7385..a2153a21a7d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aecm/aecm_core_neon.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aecm/aecm_core_neon.cc
>@@ -77,12 +77,12 @@ void WebRtcAecm_CalcLinearEnergiesNeon(AecmCore* aecm,
>     echo_stored_v = vaddq_u32(echo_est_v_low, echo_stored_v);
>     echo_stored_v = vaddq_u32(echo_est_v_high, echo_stored_v);
> 
>-    echo_adapt_v = vmlal_u16(echo_adapt_v,
>-                             vreinterpret_u16_s16(vget_low_s16(adapt_v)),
>-                             vget_low_u16(spectrum_v));
>-    echo_adapt_v = vmlal_u16(echo_adapt_v,
>-                             vreinterpret_u16_s16(vget_high_s16(adapt_v)),
>-                             vget_high_u16(spectrum_v));
>+    echo_adapt_v =
>+        vmlal_u16(echo_adapt_v, vreinterpret_u16_s16(vget_low_s16(adapt_v)),
>+                  vget_low_u16(spectrum_v));
>+    echo_adapt_v =
>+        vmlal_u16(echo_adapt_v, vreinterpret_u16_s16(vget_high_s16(adapt_v)),
>+                  vget_high_u16(spectrum_v));
> 
>     start_stored_p += 8;
>     start_adapt_p += 8;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aecm/aecm_defines.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aecm/aecm_defines.h
>index ae2d2bc03a1..5805549e2c4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aecm/aecm_defines.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aecm/aecm_defines.h
>@@ -11,77 +11,77 @@
> #ifndef MODULES_AUDIO_PROCESSING_AECM_AECM_DEFINES_H_
> #define MODULES_AUDIO_PROCESSING_AECM_AECM_DEFINES_H_
> 
>-#define AECM_DYNAMIC_Q                 /* Turn on/off dynamic Q-domain. */
>+#define AECM_DYNAMIC_Q /* Turn on/off dynamic Q-domain. */
> 
> /* Algorithm parameters */
>-#define FRAME_LEN       80             /* Total frame length, 10 ms. */
>+#define FRAME_LEN 80 /* Total frame length, 10 ms. */
> 
>-#define PART_LEN        64             /* Length of partition. */
>-#define PART_LEN_SHIFT  7              /* Length of (PART_LEN * 2) in base 2. */
>+#define PART_LEN 64      /* Length of partition. */
>+#define PART_LEN_SHIFT 7 /* Length of (PART_LEN * 2) in base 2. */
> 
>-#define PART_LEN1       (PART_LEN + 1)  /* Unique fft coefficients. */
>-#define PART_LEN2       (PART_LEN << 1) /* Length of partition * 2. */
>-#define PART_LEN4       (PART_LEN << 2) /* Length of partition * 4. */
>-#define FAR_BUF_LEN     PART_LEN4       /* Length of buffers. */
>-#define MAX_DELAY       100
>+#define PART_LEN1 (PART_LEN + 1)  /* Unique fft coefficients. */
>+#define PART_LEN2 (PART_LEN << 1) /* Length of partition * 2. */
>+#define PART_LEN4 (PART_LEN << 2) /* Length of partition * 4. */
>+#define FAR_BUF_LEN PART_LEN4     /* Length of buffers. */
>+#define MAX_DELAY 100
> 
> /* Counter parameters */
>-#define CONV_LEN        512          /* Convergence length used at startup. */
>-#define CONV_LEN2       (CONV_LEN << 1) /* Used at startup. */
>+#define CONV_LEN 512              /* Convergence length used at startup. */
>+#define CONV_LEN2 (CONV_LEN << 1) /* Used at startup. */
> 
> /* Energy parameters */
>-#define MAX_BUF_LEN     64           /* History length of energy signals. */
>-#define FAR_ENERGY_MIN  1025         /* Lowest Far energy level: At least 2 */
>-                                     /* in energy. */
>-#define FAR_ENERGY_DIFF 929          /* Allowed difference between max */
>-                                     /* and min. */
>-#define ENERGY_DEV_OFFSET       0    /* The energy error offset in Q8. */
>-#define ENERGY_DEV_TOL  400          /* The energy estimation tolerance (Q8). */
>-#define FAR_ENERGY_VAD_REGION   230  /* Far VAD tolerance region. */
>+#define MAX_BUF_LEN 64            /* History length of energy signals. */
>+#define FAR_ENERGY_MIN 1025       /* Lowest Far energy level: At least 2 */
>+                                  /* in energy. */
>+#define FAR_ENERGY_DIFF 929       /* Allowed difference between max */
>+                                  /* and min. */
>+#define ENERGY_DEV_OFFSET 0       /* The energy error offset in Q8. */
>+#define ENERGY_DEV_TOL 400        /* The energy estimation tolerance (Q8). */
>+#define FAR_ENERGY_VAD_REGION 230 /* Far VAD tolerance region. */
> 
> /* Stepsize parameters */
>-#define MU_MIN          10          /* Min stepsize 2^-MU_MIN (far end energy */
>-                                    /* dependent). */
>-#define MU_MAX          1           /* Max stepsize 2^-MU_MAX (far end energy */
>-                                    /* dependent). */
>-#define MU_DIFF         9           /* MU_MIN - MU_MAX */
>+#define MU_MIN 10 /* Min stepsize 2^-MU_MIN (far end energy */
>+                  /* dependent). */
>+#define MU_MAX 1  /* Max stepsize 2^-MU_MAX (far end energy */
>+                  /* dependent). */
>+#define MU_DIFF 9 /* MU_MIN - MU_MAX */
> 
> /* Channel parameters */
>-#define MIN_MSE_COUNT   20 /* Min number of consecutive blocks with enough */
>-                           /* far end energy to compare channel estimates. */
>-#define MIN_MSE_DIFF    29 /* The ratio between adapted and stored channel to */
>-                           /* accept a new storage (0.8 in Q-MSE_RESOLUTION). */
>-#define MSE_RESOLUTION  5           /* MSE parameter resolution. */
>-#define RESOLUTION_CHANNEL16    12  /* W16 Channel in Q-RESOLUTION_CHANNEL16. */
>-#define RESOLUTION_CHANNEL32    28  /* W32 Channel in Q-RESOLUTION_CHANNEL. */
>-#define CHANNEL_VAD     16          /* Minimum energy in frequency band */
>-                                    /* to update channel. */
>+#define MIN_MSE_COUNT 20 /* Min number of consecutive blocks with enough */
>+                         /* far end energy to compare channel estimates. */
>+#define MIN_MSE_DIFF 29  /* The ratio between adapted and stored channel to */
>+                         /* accept a new storage (0.8 in Q-MSE_RESOLUTION). */
>+#define MSE_RESOLUTION 5 /* MSE parameter resolution. */
>+#define RESOLUTION_CHANNEL16 12 /* W16 Channel in Q-RESOLUTION_CHANNEL16. */
>+#define RESOLUTION_CHANNEL32 28 /* W32 Channel in Q-RESOLUTION_CHANNEL. */
>+#define CHANNEL_VAD 16          /* Minimum energy in frequency band */
>+                                /* to update channel. */
> 
> /* Suppression gain parameters: SUPGAIN parameters in Q-(RESOLUTION_SUPGAIN). */
>-#define RESOLUTION_SUPGAIN      8     /* Channel in Q-(RESOLUTION_SUPGAIN). */
>-#define SUPGAIN_DEFAULT (1 << RESOLUTION_SUPGAIN)  /* Default. */
>-#define SUPGAIN_ERROR_PARAM_A   3072  /* Estimation error parameter */
>-                                      /* (Maximum gain) (8 in Q8). */
>-#define SUPGAIN_ERROR_PARAM_B   1536  /* Estimation error parameter */
>-                                      /* (Gain before going down). */
>-#define SUPGAIN_ERROR_PARAM_D   SUPGAIN_DEFAULT /* Estimation error parameter */
>-                                /* (Should be the same as Default) (1 in Q8). */
>-#define SUPGAIN_EPC_DT  200     /* SUPGAIN_ERROR_PARAM_C * ENERGY_DEV_TOL */
>+#define RESOLUTION_SUPGAIN 8 /* Channel in Q-(RESOLUTION_SUPGAIN). */
>+#define SUPGAIN_DEFAULT (1 << RESOLUTION_SUPGAIN) /* Default. */
>+#define SUPGAIN_ERROR_PARAM_A 3072            /* Estimation error parameter */
>+                                              /* (Maximum gain) (8 in Q8). */
>+#define SUPGAIN_ERROR_PARAM_B 1536            /* Estimation error parameter */
>+                                              /* (Gain before going down). */
>+#define SUPGAIN_ERROR_PARAM_D SUPGAIN_DEFAULT /* Estimation error parameter */
>+/* (Should be the same as Default) (1 in Q8). */
>+#define SUPGAIN_EPC_DT 200 /* SUPGAIN_ERROR_PARAM_C * ENERGY_DEV_TOL */
> 
> /* Defines for "check delay estimation" */
>-#define CORR_WIDTH      31      /* Number of samples to correlate over. */
>-#define CORR_MAX        16      /* Maximum correlation offset. */
>-#define CORR_MAX_BUF    63
>-#define CORR_DEV        4
>-#define CORR_MAX_LEVEL  20
>-#define CORR_MAX_LOW    4
>-#define CORR_BUF_LEN    (CORR_MAX << 1) + 1
>+#define CORR_WIDTH 31 /* Number of samples to correlate over. */
>+#define CORR_MAX 16   /* Maximum correlation offset. */
>+#define CORR_MAX_BUF 63
>+#define CORR_DEV 4
>+#define CORR_MAX_LEVEL 20
>+#define CORR_MAX_LOW 4
>+#define CORR_BUF_LEN (CORR_MAX << 1) + 1
> /* Note that CORR_WIDTH + 2*CORR_MAX <= MAX_BUF_LEN. */
> 
>-#define ONE_Q14         (1 << 14)
>+#define ONE_Q14 (1 << 14)
> 
> /* NLP defines */
>-#define NLP_COMP_LOW    3277    /* 0.2 in Q14 */
>-#define NLP_COMP_HIGH   ONE_Q14 /* 1 in Q14 */
>+#define NLP_COMP_LOW 3277     /* 0.2 in Q14 */
>+#define NLP_COMP_HIGH ONE_Q14 /* 1 in Q14 */
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aecm/echo_control_mobile.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aecm/echo_control_mobile.cc
>index 36e227166f1..c947563b280 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aecm/echo_control_mobile.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aecm/echo_control_mobile.cc
>@@ -21,56 +21,56 @@ extern "C" {
> }
> #include "modules/audio_processing/aecm/aecm_core.h"
> 
>-#define BUF_SIZE_FRAMES 50 // buffer size (frames)
>+#define BUF_SIZE_FRAMES 50  // buffer size (frames)
> // Maximum length of resampled signal. Must be an integer multiple of frames
> // (ceil(1/(1 + MIN_SKEW)*2) + 1)*FRAME_LEN
> // The factor of 2 handles wb, and the + 1 is as a safety margin
> #define MAX_RESAMP_LEN (5 * FRAME_LEN)
> 
>-static const size_t kBufSizeSamp = BUF_SIZE_FRAMES * FRAME_LEN; // buffer size (samples)
>-static const int kSampMsNb = 8; // samples per ms in nb
>+static const size_t kBufSizeSamp =
>+    BUF_SIZE_FRAMES * FRAME_LEN;  // buffer size (samples)
>+static const int kSampMsNb = 8;   // samples per ms in nb
> // Target suppression levels for nlp modes
> // log{0.001, 0.00001, 0.00000001}
> static const int kInitCheck = 42;
> 
>-typedef struct
>-{
>-    int sampFreq;
>-    int scSampFreq;
>-    short bufSizeStart;
>-    int knownDelay;
>-
>-    // Stores the last frame added to the farend buffer
>-    short farendOld[2][FRAME_LEN];
>-    short initFlag; // indicates if AEC has been initialized
>-
>-    // Variables used for averaging far end buffer size
>-    short counter;
>-    short sum;
>-    short firstVal;
>-    short checkBufSizeCtr;
>-
>-    // Variables used for delay shifts
>-    short msInSndCardBuf;
>-    short filtDelay;
>-    int timeForDelayChange;
>-    int ECstartup;
>-    int checkBuffSize;
>-    int delayChange;
>-    short lastDelayDiff;
>-
>-    int16_t echoMode;
>+typedef struct {
>+  int sampFreq;
>+  int scSampFreq;
>+  short bufSizeStart;
>+  int knownDelay;
>+
>+  // Stores the last frame added to the farend buffer
>+  short farendOld[2][FRAME_LEN];
>+  short initFlag;  // indicates if AEC has been initialized
>+
>+  // Variables used for averaging far end buffer size
>+  short counter;
>+  short sum;
>+  short firstVal;
>+  short checkBufSizeCtr;
>+
>+  // Variables used for delay shifts
>+  short msInSndCardBuf;
>+  short filtDelay;
>+  int timeForDelayChange;
>+  int ECstartup;
>+  int checkBuffSize;
>+  int delayChange;
>+  short lastDelayDiff;
>+
>+  int16_t echoMode;
> 
> #ifdef AEC_DEBUG
>-    FILE *bufFile;
>-    FILE *delayFile;
>-    FILE *preCompFile;
>-    FILE *postCompFile;
>-#endif // AEC_DEBUG
>-    // Structures
>-    RingBuffer *farendBuf;
>-
>-    AecmCore* aecmCore;
>+  FILE* bufFile;
>+  FILE* delayFile;
>+  FILE* preCompFile;
>+  FILE* postCompFile;
>+#endif  // AEC_DEBUG
>+  // Structures
>+  RingBuffer* farendBuf;
>+
>+  AecmCore* aecmCore;
> } AecMobile;
> 
> // Estimates delay to set the position of the farend buffer read pointer
>@@ -81,123 +81,117 @@ static int WebRtcAecm_EstBufDelay(AecMobile* aecm, short msInSndCardBuf);
> static int WebRtcAecm_DelayComp(AecMobile* aecm);
> 
> void* WebRtcAecm_Create() {
>-    AecMobile* aecm = static_cast<AecMobile*>(malloc(sizeof(AecMobile)));
>+  AecMobile* aecm = static_cast<AecMobile*>(malloc(sizeof(AecMobile)));
> 
>-    WebRtcSpl_Init();
>+  WebRtcSpl_Init();
> 
>-    aecm->aecmCore = WebRtcAecm_CreateCore();
>-    if (!aecm->aecmCore) {
>-        WebRtcAecm_Free(aecm);
>-        return NULL;
>-    }
>+  aecm->aecmCore = WebRtcAecm_CreateCore();
>+  if (!aecm->aecmCore) {
>+    WebRtcAecm_Free(aecm);
>+    return NULL;
>+  }
> 
>-    aecm->farendBuf = WebRtc_CreateBuffer(kBufSizeSamp,
>-                                          sizeof(int16_t));
>-    if (!aecm->farendBuf)
>-    {
>-        WebRtcAecm_Free(aecm);
>-        return NULL;
>-    }
>+  aecm->farendBuf = WebRtc_CreateBuffer(kBufSizeSamp, sizeof(int16_t));
>+  if (!aecm->farendBuf) {
>+    WebRtcAecm_Free(aecm);
>+    return NULL;
>+  }
> 
>-    aecm->initFlag = 0;
>+  aecm->initFlag = 0;
> 
> #ifdef AEC_DEBUG
>-    aecm->aecmCore->farFile = fopen("aecFar.pcm","wb");
>-    aecm->aecmCore->nearFile = fopen("aecNear.pcm","wb");
>-    aecm->aecmCore->outFile = fopen("aecOut.pcm","wb");
>-    //aecm->aecmCore->outLpFile = fopen("aecOutLp.pcm","wb");
>-
>-    aecm->bufFile = fopen("aecBuf.dat", "wb");
>-    aecm->delayFile = fopen("aecDelay.dat", "wb");
>-    aecm->preCompFile = fopen("preComp.pcm", "wb");
>-    aecm->postCompFile = fopen("postComp.pcm", "wb");
>-#endif // AEC_DEBUG
>-    return aecm;
>+  aecm->aecmCore->farFile = fopen("aecFar.pcm", "wb");
>+  aecm->aecmCore->nearFile = fopen("aecNear.pcm", "wb");
>+  aecm->aecmCore->outFile = fopen("aecOut.pcm", "wb");
>+  // aecm->aecmCore->outLpFile = fopen("aecOutLp.pcm","wb");
>+
>+  aecm->bufFile = fopen("aecBuf.dat", "wb");
>+  aecm->delayFile = fopen("aecDelay.dat", "wb");
>+  aecm->preCompFile = fopen("preComp.pcm", "wb");
>+  aecm->postCompFile = fopen("postComp.pcm", "wb");
>+#endif  // AEC_DEBUG
>+  return aecm;
> }
> 
> void WebRtcAecm_Free(void* aecmInst) {
>   AecMobile* aecm = static_cast<AecMobile*>(aecmInst);
> 
>-    if (aecm == NULL) {
>-      return;
>-    }
>+  if (aecm == NULL) {
>+    return;
>+  }
> 
> #ifdef AEC_DEBUG
>-    fclose(aecm->aecmCore->farFile);
>-    fclose(aecm->aecmCore->nearFile);
>-    fclose(aecm->aecmCore->outFile);
>-    //fclose(aecm->aecmCore->outLpFile);
>-
>-    fclose(aecm->bufFile);
>-    fclose(aecm->delayFile);
>-    fclose(aecm->preCompFile);
>-    fclose(aecm->postCompFile);
>-#endif // AEC_DEBUG
>-    WebRtcAecm_FreeCore(aecm->aecmCore);
>-    WebRtc_FreeBuffer(aecm->farendBuf);
>-    free(aecm);
>+  fclose(aecm->aecmCore->farFile);
>+  fclose(aecm->aecmCore->nearFile);
>+  fclose(aecm->aecmCore->outFile);
>+  // fclose(aecm->aecmCore->outLpFile);
>+
>+  fclose(aecm->bufFile);
>+  fclose(aecm->delayFile);
>+  fclose(aecm->preCompFile);
>+  fclose(aecm->postCompFile);
>+#endif  // AEC_DEBUG
>+  WebRtcAecm_FreeCore(aecm->aecmCore);
>+  WebRtc_FreeBuffer(aecm->farendBuf);
>+  free(aecm);
> }
> 
>-int32_t WebRtcAecm_Init(void *aecmInst, int32_t sampFreq)
>-{
>-    AecMobile* aecm = static_cast<AecMobile*>(aecmInst);
>-    AecmConfig aecConfig;
>+int32_t WebRtcAecm_Init(void* aecmInst, int32_t sampFreq) {
>+  AecMobile* aecm = static_cast<AecMobile*>(aecmInst);
>+  AecmConfig aecConfig;
> 
>-    if (aecm == NULL)
>-    {
>-        return -1;
>-    }
>+  if (aecm == NULL) {
>+    return -1;
>+  }
> 
>-    if (sampFreq != 8000 && sampFreq != 16000)
>-    {
>-        return AECM_BAD_PARAMETER_ERROR;
>-    }
>-    aecm->sampFreq = sampFreq;
>+  if (sampFreq != 8000 && sampFreq != 16000) {
>+    return AECM_BAD_PARAMETER_ERROR;
>+  }
>+  aecm->sampFreq = sampFreq;
> 
>-    // Initialize AECM core
>-    if (WebRtcAecm_InitCore(aecm->aecmCore, aecm->sampFreq) == -1)
>-    {
>-        return AECM_UNSPECIFIED_ERROR;
>-    }
>+  // Initialize AECM core
>+  if (WebRtcAecm_InitCore(aecm->aecmCore, aecm->sampFreq) == -1) {
>+    return AECM_UNSPECIFIED_ERROR;
>+  }
> 
>-    // Initialize farend buffer
>-    WebRtc_InitBuffer(aecm->farendBuf);
>+  // Initialize farend buffer
>+  WebRtc_InitBuffer(aecm->farendBuf);
> 
>-    aecm->initFlag = kInitCheck; // indicates that initialization has been done
>+  aecm->initFlag = kInitCheck;  // indicates that initialization has been done
> 
>-    aecm->delayChange = 1;
>+  aecm->delayChange = 1;
> 
>-    aecm->sum = 0;
>-    aecm->counter = 0;
>-    aecm->checkBuffSize = 1;
>-    aecm->firstVal = 0;
>+  aecm->sum = 0;
>+  aecm->counter = 0;
>+  aecm->checkBuffSize = 1;
>+  aecm->firstVal = 0;
> 
>-    aecm->ECstartup = 1;
>-    aecm->bufSizeStart = 0;
>-    aecm->checkBufSizeCtr = 0;
>-    aecm->filtDelay = 0;
>-    aecm->timeForDelayChange = 0;
>-    aecm->knownDelay = 0;
>-    aecm->lastDelayDiff = 0;
>+  aecm->ECstartup = 1;
>+  aecm->bufSizeStart = 0;
>+  aecm->checkBufSizeCtr = 0;
>+  aecm->filtDelay = 0;
>+  aecm->timeForDelayChange = 0;
>+  aecm->knownDelay = 0;
>+  aecm->lastDelayDiff = 0;
> 
>-    memset(&aecm->farendOld, 0, sizeof(aecm->farendOld));
>+  memset(&aecm->farendOld, 0, sizeof(aecm->farendOld));
> 
>-    // Default settings.
>-    aecConfig.cngMode = AecmTrue;
>-    aecConfig.echoMode = 3;
>+  // Default settings.
>+  aecConfig.cngMode = AecmTrue;
>+  aecConfig.echoMode = 3;
> 
>-    if (WebRtcAecm_set_config(aecm, aecConfig) == -1)
>-    {
>-        return AECM_UNSPECIFIED_ERROR;
>-    }
>+  if (WebRtcAecm_set_config(aecm, aecConfig) == -1) {
>+    return AECM_UNSPECIFIED_ERROR;
>+  }
> 
>-    return 0;
>+  return 0;
> }
> 
> // Returns any error that is caused when buffering the
> // farend signal.
>-int32_t WebRtcAecm_GetBufferFarendError(void *aecmInst, const int16_t *farend,
>-                                size_t nrOfSamples) {
>+int32_t WebRtcAecm_GetBufferFarendError(void* aecmInst,
>+                                        const int16_t* farend,
>+                                        size_t nrOfSamples) {
>   AecMobile* aecm = static_cast<AecMobile*>(aecmInst);
> 
>   if (aecm == NULL)
>@@ -215,8 +209,8 @@ int32_t WebRtcAecm_GetBufferFarendError(void *aecmInst, const int16_t *farend,
>   return 0;
> }
> 
>-
>-int32_t WebRtcAecm_BufferFarend(void *aecmInst, const int16_t *farend,
>+int32_t WebRtcAecm_BufferFarend(void* aecmInst,
>+                                const int16_t* farend,
>                                 size_t nrOfSamples) {
>   AecMobile* aecm = static_cast<AecMobile*>(aecmInst);
> 
>@@ -227,8 +221,7 @@ int32_t WebRtcAecm_BufferFarend(void *aecmInst, const int16_t *farend,
>     return err;
> 
>   // TODO(unknown): Is this really a good idea?
>-  if (!aecm->ECstartup)
>-  {
>+  if (!aecm->ECstartup) {
>     WebRtcAecm_DelayComp(aecm);
>   }
> 
>@@ -237,412 +230,363 @@ int32_t WebRtcAecm_BufferFarend(void *aecmInst, const int16_t *farend,
>   return 0;
> }
> 
>-int32_t WebRtcAecm_Process(void *aecmInst, const int16_t *nearendNoisy,
>-                           const int16_t *nearendClean, int16_t *out,
>-                           size_t nrOfSamples, int16_t msInSndCardBuf)
>-{
>-    AecMobile* aecm = static_cast<AecMobile*>(aecmInst);
>-    int32_t retVal = 0;
>-    size_t i;
>-    short nmbrOfFilledBuffers;
>-    size_t nBlocks10ms;
>-    size_t nFrames;
>+int32_t WebRtcAecm_Process(void* aecmInst,
>+                           const int16_t* nearendNoisy,
>+                           const int16_t* nearendClean,
>+                           int16_t* out,
>+                           size_t nrOfSamples,
>+                           int16_t msInSndCardBuf) {
>+  AecMobile* aecm = static_cast<AecMobile*>(aecmInst);
>+  int32_t retVal = 0;
>+  size_t i;
>+  short nmbrOfFilledBuffers;
>+  size_t nBlocks10ms;
>+  size_t nFrames;
> #ifdef AEC_DEBUG
>-    short msInAECBuf;
>+  short msInAECBuf;
> #endif
> 
>-    if (aecm == NULL)
>-    {
>-        return -1;
>-    }
>+  if (aecm == NULL) {
>+    return -1;
>+  }
> 
>-    if (nearendNoisy == NULL)
>-    {
>-        return AECM_NULL_POINTER_ERROR;
>-    }
>+  if (nearendNoisy == NULL) {
>+    return AECM_NULL_POINTER_ERROR;
>+  }
> 
>-    if (out == NULL)
>-    {
>-        return AECM_NULL_POINTER_ERROR;
>-    }
>+  if (out == NULL) {
>+    return AECM_NULL_POINTER_ERROR;
>+  }
>+
>+  if (aecm->initFlag != kInitCheck) {
>+    return AECM_UNINITIALIZED_ERROR;
>+  }
>+
>+  if (nrOfSamples != 80 && nrOfSamples != 160) {
>+    return AECM_BAD_PARAMETER_ERROR;
>+  }
> 
>-    if (aecm->initFlag != kInitCheck)
>-    {
>-        return AECM_UNINITIALIZED_ERROR;
>+  if (msInSndCardBuf < 0) {
>+    msInSndCardBuf = 0;
>+    retVal = AECM_BAD_PARAMETER_WARNING;
>+  } else if (msInSndCardBuf > 500) {
>+    msInSndCardBuf = 500;
>+    retVal = AECM_BAD_PARAMETER_WARNING;
>+  }
>+  msInSndCardBuf += 10;
>+  aecm->msInSndCardBuf = msInSndCardBuf;
>+
>+  nFrames = nrOfSamples / FRAME_LEN;
>+  nBlocks10ms = nFrames / aecm->aecmCore->mult;
>+
>+  if (aecm->ECstartup) {
>+    if (nearendClean == NULL) {
>+      if (out != nearendNoisy) {
>+        memcpy(out, nearendNoisy, sizeof(short) * nrOfSamples);
>+      }
>+    } else if (out != nearendClean) {
>+      memcpy(out, nearendClean, sizeof(short) * nrOfSamples);
>     }
> 
>-    if (nrOfSamples != 80 && nrOfSamples != 160)
>-    {
>-        return AECM_BAD_PARAMETER_ERROR;
>+    nmbrOfFilledBuffers =
>+        (short)WebRtc_available_read(aecm->farendBuf) / FRAME_LEN;
>+    // The AECM is in the start up mode
>+    // AECM is disabled until the soundcard buffer and farend buffers are OK
>+
>+    // Mechanism to ensure that the soundcard buffer is reasonably stable.
>+    if (aecm->checkBuffSize) {
>+      aecm->checkBufSizeCtr++;
>+      // Before we fill up the far end buffer we require the amount of data on
>+      // the sound card to be stable (+/-8 ms) compared to the first value. This
>+      // comparison is made during the following 4 consecutive frames. If it
>+      // seems to be stable then we start to fill up the far end buffer.
>+
>+      if (aecm->counter == 0) {
>+        aecm->firstVal = aecm->msInSndCardBuf;
>+        aecm->sum = 0;
>+      }
>+
>+      if (abs(aecm->firstVal - aecm->msInSndCardBuf) <
>+          WEBRTC_SPL_MAX(0.2 * aecm->msInSndCardBuf, kSampMsNb)) {
>+        aecm->sum += aecm->msInSndCardBuf;
>+        aecm->counter++;
>+      } else {
>+        aecm->counter = 0;
>+      }
>+
>+      if (aecm->counter * nBlocks10ms >= 6) {
>+        // The farend buffer size is determined in blocks of 80 samples
>+        // Use 75% of the average value of the soundcard buffer
>+        aecm->bufSizeStart = WEBRTC_SPL_MIN(
>+            (3 * aecm->sum * aecm->aecmCore->mult) / (aecm->counter * 40),
>+            BUF_SIZE_FRAMES);
>+        // buffersize has now been determined
>+        aecm->checkBuffSize = 0;
>+      }
>+
>+      if (aecm->checkBufSizeCtr * nBlocks10ms > 50) {
>+        // for really bad sound cards, don't disable echocanceller for more than
>+        // 0.5 sec
>+        aecm->bufSizeStart = WEBRTC_SPL_MIN(
>+            (3 * aecm->msInSndCardBuf * aecm->aecmCore->mult) / 40,
>+            BUF_SIZE_FRAMES);
>+        aecm->checkBuffSize = 0;
>+      }
>     }
> 
>-    if (msInSndCardBuf < 0)
>-    {
>-        msInSndCardBuf = 0;
>-        retVal = AECM_BAD_PARAMETER_WARNING;
>-    } else if (msInSndCardBuf > 500)
>-    {
>-        msInSndCardBuf = 500;
>-        retVal = AECM_BAD_PARAMETER_WARNING;
>+    // if checkBuffSize changed in the if-statement above
>+    if (!aecm->checkBuffSize) {
>+      // soundcard buffer is now reasonably stable
>+      // When the far end buffer is filled with approximately the same amount of
>+      // data as the amount on the sound card we end the start up phase and
>+      // start to cancel echoes.
>+
>+      if (nmbrOfFilledBuffers == aecm->bufSizeStart) {
>+        aecm->ECstartup = 0;  // Enable the AECM
>+      } else if (nmbrOfFilledBuffers > aecm->bufSizeStart) {
>+        WebRtc_MoveReadPtr(aecm->farendBuf,
>+                           (int)WebRtc_available_read(aecm->farendBuf) -
>+                               (int)aecm->bufSizeStart * FRAME_LEN);
>+        aecm->ECstartup = 0;
>+      }
>     }
>-    msInSndCardBuf += 10;
>-    aecm->msInSndCardBuf = msInSndCardBuf;
>-
>-    nFrames = nrOfSamples / FRAME_LEN;
>-    nBlocks10ms = nFrames / aecm->aecmCore->mult;
>-
>-    if (aecm->ECstartup)
>-    {
>-        if (nearendClean == NULL)
>-        {
>-            if (out != nearendNoisy)
>-            {
>-                memcpy(out, nearendNoisy, sizeof(short) * nrOfSamples);
>-            }
>-        } else if (out != nearendClean)
>-        {
>-            memcpy(out, nearendClean, sizeof(short) * nrOfSamples);
>-        }
>-
>-        nmbrOfFilledBuffers =
>-            (short) WebRtc_available_read(aecm->farendBuf) / FRAME_LEN;
>-        // The AECM is in the start up mode
>-        // AECM is disabled until the soundcard buffer and farend buffers are OK
>-
>-        // Mechanism to ensure that the soundcard buffer is reasonably stable.
>-        if (aecm->checkBuffSize)
>-        {
>-            aecm->checkBufSizeCtr++;
>-            // Before we fill up the far end buffer we require the amount of data on the
>-            // sound card to be stable (+/-8 ms) compared to the first value. This
>-            // comparison is made during the following 4 consecutive frames. If it seems
>-            // to be stable then we start to fill up the far end buffer.
>-
>-            if (aecm->counter == 0)
>-            {
>-                aecm->firstVal = aecm->msInSndCardBuf;
>-                aecm->sum = 0;
>-            }
>-
>-            if (abs(aecm->firstVal - aecm->msInSndCardBuf)
>-                    < WEBRTC_SPL_MAX(0.2 * aecm->msInSndCardBuf, kSampMsNb))
>-            {
>-                aecm->sum += aecm->msInSndCardBuf;
>-                aecm->counter++;
>-            } else
>-            {
>-                aecm->counter = 0;
>-            }
>-
>-            if (aecm->counter * nBlocks10ms >= 6)
>-            {
>-                // The farend buffer size is determined in blocks of 80 samples
>-                // Use 75% of the average value of the soundcard buffer
>-                aecm->bufSizeStart
>-                        = WEBRTC_SPL_MIN((3 * aecm->sum
>-                                        * aecm->aecmCore->mult) / (aecm->counter * 40), BUF_SIZE_FRAMES);
>-                // buffersize has now been determined
>-                aecm->checkBuffSize = 0;
>-            }
>-
>-            if (aecm->checkBufSizeCtr * nBlocks10ms > 50)
>-            {
>-                // for really bad sound cards, don't disable echocanceller for more than 0.5 sec
>-                aecm->bufSizeStart = WEBRTC_SPL_MIN((3 * aecm->msInSndCardBuf
>-                                * aecm->aecmCore->mult) / 40, BUF_SIZE_FRAMES);
>-                aecm->checkBuffSize = 0;
>-            }
>-        }
>-
>-        // if checkBuffSize changed in the if-statement above
>-        if (!aecm->checkBuffSize)
>-        {
>-            // soundcard buffer is now reasonably stable
>-            // When the far end buffer is filled with approximately the same amount of
>-            // data as the amount on the sound card we end the start up phase and start
>-            // to cancel echoes.
>-
>-            if (nmbrOfFilledBuffers == aecm->bufSizeStart)
>-            {
>-                aecm->ECstartup = 0; // Enable the AECM
>-            } else if (nmbrOfFilledBuffers > aecm->bufSizeStart)
>-            {
>-                WebRtc_MoveReadPtr(aecm->farendBuf,
>-                                   (int) WebRtc_available_read(aecm->farendBuf)
>-                                   - (int) aecm->bufSizeStart * FRAME_LEN);
>-                aecm->ECstartup = 0;
>-            }
>-        }
>-
>-    } else
>-    {
>-        // AECM is enabled
>-
>-        // Note only 1 block supported for nb and 2 blocks for wb
>-        for (i = 0; i < nFrames; i++)
>-        {
>-            int16_t farend[FRAME_LEN];
>-            const int16_t* farend_ptr = NULL;
>-
>-            nmbrOfFilledBuffers =
>-                (short) WebRtc_available_read(aecm->farendBuf) / FRAME_LEN;
>-
>-            // Check that there is data in the far end buffer
>-            if (nmbrOfFilledBuffers > 0)
>-            {
>-                // Get the next 80 samples from the farend buffer
>-                WebRtc_ReadBuffer(aecm->farendBuf, (void**) &farend_ptr, farend,
>-                                  FRAME_LEN);
>-
>-                // Always store the last frame for use when we run out of data
>-                memcpy(&(aecm->farendOld[i][0]), farend_ptr,
>-                       FRAME_LEN * sizeof(short));
>-            } else
>-            {
>-                // We have no data so we use the last played frame
>-                memcpy(farend, &(aecm->farendOld[i][0]), FRAME_LEN * sizeof(short));
>-                farend_ptr = farend;
>-            }
>-
>-            // Call buffer delay estimator when all data is extracted,
>-            // i,e. i = 0 for NB and i = 1 for WB
>-            if ((i == 0 && aecm->sampFreq == 8000) || (i == 1 && aecm->sampFreq == 16000))
>-            {
>-                WebRtcAecm_EstBufDelay(aecm, aecm->msInSndCardBuf);
>-            }
>-
>-            // Call the AECM
>-            /*WebRtcAecm_ProcessFrame(aecm->aecmCore, farend, &nearend[FRAME_LEN * i],
>-             &out[FRAME_LEN * i], aecm->knownDelay);*/
>-            if (WebRtcAecm_ProcessFrame(aecm->aecmCore,
>-                                        farend_ptr,
>-                                        &nearendNoisy[FRAME_LEN * i],
>-                                        (nearendClean
>-                                         ? &nearendClean[FRAME_LEN * i]
>-                                         : NULL),
>-                                        &out[FRAME_LEN * i]) == -1)
>-                return -1;
>-        }
>+
>+  } else {
>+    // AECM is enabled
>+
>+    // Note only 1 block supported for nb and 2 blocks for wb
>+    for (i = 0; i < nFrames; i++) {
>+      int16_t farend[FRAME_LEN];
>+      const int16_t* farend_ptr = NULL;
>+
>+      nmbrOfFilledBuffers =
>+          (short)WebRtc_available_read(aecm->farendBuf) / FRAME_LEN;
>+
>+      // Check that there is data in the far end buffer
>+      if (nmbrOfFilledBuffers > 0) {
>+        // Get the next 80 samples from the farend buffer
>+        WebRtc_ReadBuffer(aecm->farendBuf, (void**)&farend_ptr, farend,
>+                          FRAME_LEN);
>+
>+        // Always store the last frame for use when we run out of data
>+        memcpy(&(aecm->farendOld[i][0]), farend_ptr, FRAME_LEN * sizeof(short));
>+      } else {
>+        // We have no data so we use the last played frame
>+        memcpy(farend, &(aecm->farendOld[i][0]), FRAME_LEN * sizeof(short));
>+        farend_ptr = farend;
>+      }
>+
>+      // Call buffer delay estimator when all data is extracted,
>+      // i,e. i = 0 for NB and i = 1 for WB
>+      if ((i == 0 && aecm->sampFreq == 8000) ||
>+          (i == 1 && aecm->sampFreq == 16000)) {
>+        WebRtcAecm_EstBufDelay(aecm, aecm->msInSndCardBuf);
>+      }
>+
>+      // Call the AECM
>+      /*WebRtcAecm_ProcessFrame(aecm->aecmCore, farend, &nearend[FRAME_LEN * i],
>+       &out[FRAME_LEN * i], aecm->knownDelay);*/
>+      if (WebRtcAecm_ProcessFrame(
>+              aecm->aecmCore, farend_ptr, &nearendNoisy[FRAME_LEN * i],
>+              (nearendClean ? &nearendClean[FRAME_LEN * i] : NULL),
>+              &out[FRAME_LEN * i]) == -1)
>+        return -1;
>     }
>+  }
> 
> #ifdef AEC_DEBUG
>-    msInAECBuf = (short) WebRtc_available_read(aecm->farendBuf) /
>-        (kSampMsNb * aecm->aecmCore->mult);
>-    fwrite(&msInAECBuf, 2, 1, aecm->bufFile);
>-    fwrite(&(aecm->knownDelay), sizeof(aecm->knownDelay), 1, aecm->delayFile);
>+  msInAECBuf = (short)WebRtc_available_read(aecm->farendBuf) /
>+               (kSampMsNb * aecm->aecmCore->mult);
>+  fwrite(&msInAECBuf, 2, 1, aecm->bufFile);
>+  fwrite(&(aecm->knownDelay), sizeof(aecm->knownDelay), 1, aecm->delayFile);
> #endif
> 
>-    return retVal;
>+  return retVal;
> }
> 
>-int32_t WebRtcAecm_set_config(void *aecmInst, AecmConfig config)
>-{
>-    AecMobile* aecm = static_cast<AecMobile*>(aecmInst);
>+int32_t WebRtcAecm_set_config(void* aecmInst, AecmConfig config) {
>+  AecMobile* aecm = static_cast<AecMobile*>(aecmInst);
> 
>-    if (aecm == NULL)
>-    {
>-        return -1;
>-    }
>+  if (aecm == NULL) {
>+    return -1;
>+  }
> 
>-    if (aecm->initFlag != kInitCheck)
>-    {
>-        return AECM_UNINITIALIZED_ERROR;
>-    }
>+  if (aecm->initFlag != kInitCheck) {
>+    return AECM_UNINITIALIZED_ERROR;
>+  }
> 
>-    if (config.cngMode != AecmFalse && config.cngMode != AecmTrue)
>-    {
>-        return AECM_BAD_PARAMETER_ERROR;
>-    }
>-    aecm->aecmCore->cngMode = config.cngMode;
>+  if (config.cngMode != AecmFalse && config.cngMode != AecmTrue) {
>+    return AECM_BAD_PARAMETER_ERROR;
>+  }
>+  aecm->aecmCore->cngMode = config.cngMode;
> 
>-    if (config.echoMode < 0 || config.echoMode > 4)
>-    {
>-        return AECM_BAD_PARAMETER_ERROR;
>-    }
>-    aecm->echoMode = config.echoMode;
>-
>-    if (aecm->echoMode == 0)
>-    {
>-        aecm->aecmCore->supGain = SUPGAIN_DEFAULT >> 3;
>-        aecm->aecmCore->supGainOld = SUPGAIN_DEFAULT >> 3;
>-        aecm->aecmCore->supGainErrParamA = SUPGAIN_ERROR_PARAM_A >> 3;
>-        aecm->aecmCore->supGainErrParamD = SUPGAIN_ERROR_PARAM_D >> 3;
>-        aecm->aecmCore->supGainErrParamDiffAB = (SUPGAIN_ERROR_PARAM_A >> 3)
>-                - (SUPGAIN_ERROR_PARAM_B >> 3);
>-        aecm->aecmCore->supGainErrParamDiffBD = (SUPGAIN_ERROR_PARAM_B >> 3)
>-                - (SUPGAIN_ERROR_PARAM_D >> 3);
>-    } else if (aecm->echoMode == 1)
>-    {
>-        aecm->aecmCore->supGain = SUPGAIN_DEFAULT >> 2;
>-        aecm->aecmCore->supGainOld = SUPGAIN_DEFAULT >> 2;
>-        aecm->aecmCore->supGainErrParamA = SUPGAIN_ERROR_PARAM_A >> 2;
>-        aecm->aecmCore->supGainErrParamD = SUPGAIN_ERROR_PARAM_D >> 2;
>-        aecm->aecmCore->supGainErrParamDiffAB = (SUPGAIN_ERROR_PARAM_A >> 2)
>-                - (SUPGAIN_ERROR_PARAM_B >> 2);
>-        aecm->aecmCore->supGainErrParamDiffBD = (SUPGAIN_ERROR_PARAM_B >> 2)
>-                - (SUPGAIN_ERROR_PARAM_D >> 2);
>-    } else if (aecm->echoMode == 2)
>-    {
>-        aecm->aecmCore->supGain = SUPGAIN_DEFAULT >> 1;
>-        aecm->aecmCore->supGainOld = SUPGAIN_DEFAULT >> 1;
>-        aecm->aecmCore->supGainErrParamA = SUPGAIN_ERROR_PARAM_A >> 1;
>-        aecm->aecmCore->supGainErrParamD = SUPGAIN_ERROR_PARAM_D >> 1;
>-        aecm->aecmCore->supGainErrParamDiffAB = (SUPGAIN_ERROR_PARAM_A >> 1)
>-                - (SUPGAIN_ERROR_PARAM_B >> 1);
>-        aecm->aecmCore->supGainErrParamDiffBD = (SUPGAIN_ERROR_PARAM_B >> 1)
>-                - (SUPGAIN_ERROR_PARAM_D >> 1);
>-    } else if (aecm->echoMode == 3)
>-    {
>-        aecm->aecmCore->supGain = SUPGAIN_DEFAULT;
>-        aecm->aecmCore->supGainOld = SUPGAIN_DEFAULT;
>-        aecm->aecmCore->supGainErrParamA = SUPGAIN_ERROR_PARAM_A;
>-        aecm->aecmCore->supGainErrParamD = SUPGAIN_ERROR_PARAM_D;
>-        aecm->aecmCore->supGainErrParamDiffAB = SUPGAIN_ERROR_PARAM_A - SUPGAIN_ERROR_PARAM_B;
>-        aecm->aecmCore->supGainErrParamDiffBD = SUPGAIN_ERROR_PARAM_B - SUPGAIN_ERROR_PARAM_D;
>-    } else if (aecm->echoMode == 4)
>-    {
>-        aecm->aecmCore->supGain = SUPGAIN_DEFAULT << 1;
>-        aecm->aecmCore->supGainOld = SUPGAIN_DEFAULT << 1;
>-        aecm->aecmCore->supGainErrParamA = SUPGAIN_ERROR_PARAM_A << 1;
>-        aecm->aecmCore->supGainErrParamD = SUPGAIN_ERROR_PARAM_D << 1;
>-        aecm->aecmCore->supGainErrParamDiffAB = (SUPGAIN_ERROR_PARAM_A << 1)
>-                - (SUPGAIN_ERROR_PARAM_B << 1);
>-        aecm->aecmCore->supGainErrParamDiffBD = (SUPGAIN_ERROR_PARAM_B << 1)
>-                - (SUPGAIN_ERROR_PARAM_D << 1);
>-    }
>+  if (config.echoMode < 0 || config.echoMode > 4) {
>+    return AECM_BAD_PARAMETER_ERROR;
>+  }
>+  aecm->echoMode = config.echoMode;
>+
>+  if (aecm->echoMode == 0) {
>+    aecm->aecmCore->supGain = SUPGAIN_DEFAULT >> 3;
>+    aecm->aecmCore->supGainOld = SUPGAIN_DEFAULT >> 3;
>+    aecm->aecmCore->supGainErrParamA = SUPGAIN_ERROR_PARAM_A >> 3;
>+    aecm->aecmCore->supGainErrParamD = SUPGAIN_ERROR_PARAM_D >> 3;
>+    aecm->aecmCore->supGainErrParamDiffAB =
>+        (SUPGAIN_ERROR_PARAM_A >> 3) - (SUPGAIN_ERROR_PARAM_B >> 3);
>+    aecm->aecmCore->supGainErrParamDiffBD =
>+        (SUPGAIN_ERROR_PARAM_B >> 3) - (SUPGAIN_ERROR_PARAM_D >> 3);
>+  } else if (aecm->echoMode == 1) {
>+    aecm->aecmCore->supGain = SUPGAIN_DEFAULT >> 2;
>+    aecm->aecmCore->supGainOld = SUPGAIN_DEFAULT >> 2;
>+    aecm->aecmCore->supGainErrParamA = SUPGAIN_ERROR_PARAM_A >> 2;
>+    aecm->aecmCore->supGainErrParamD = SUPGAIN_ERROR_PARAM_D >> 2;
>+    aecm->aecmCore->supGainErrParamDiffAB =
>+        (SUPGAIN_ERROR_PARAM_A >> 2) - (SUPGAIN_ERROR_PARAM_B >> 2);
>+    aecm->aecmCore->supGainErrParamDiffBD =
>+        (SUPGAIN_ERROR_PARAM_B >> 2) - (SUPGAIN_ERROR_PARAM_D >> 2);
>+  } else if (aecm->echoMode == 2) {
>+    aecm->aecmCore->supGain = SUPGAIN_DEFAULT >> 1;
>+    aecm->aecmCore->supGainOld = SUPGAIN_DEFAULT >> 1;
>+    aecm->aecmCore->supGainErrParamA = SUPGAIN_ERROR_PARAM_A >> 1;
>+    aecm->aecmCore->supGainErrParamD = SUPGAIN_ERROR_PARAM_D >> 1;
>+    aecm->aecmCore->supGainErrParamDiffAB =
>+        (SUPGAIN_ERROR_PARAM_A >> 1) - (SUPGAIN_ERROR_PARAM_B >> 1);
>+    aecm->aecmCore->supGainErrParamDiffBD =
>+        (SUPGAIN_ERROR_PARAM_B >> 1) - (SUPGAIN_ERROR_PARAM_D >> 1);
>+  } else if (aecm->echoMode == 3) {
>+    aecm->aecmCore->supGain = SUPGAIN_DEFAULT;
>+    aecm->aecmCore->supGainOld = SUPGAIN_DEFAULT;
>+    aecm->aecmCore->supGainErrParamA = SUPGAIN_ERROR_PARAM_A;
>+    aecm->aecmCore->supGainErrParamD = SUPGAIN_ERROR_PARAM_D;
>+    aecm->aecmCore->supGainErrParamDiffAB =
>+        SUPGAIN_ERROR_PARAM_A - SUPGAIN_ERROR_PARAM_B;
>+    aecm->aecmCore->supGainErrParamDiffBD =
>+        SUPGAIN_ERROR_PARAM_B - SUPGAIN_ERROR_PARAM_D;
>+  } else if (aecm->echoMode == 4) {
>+    aecm->aecmCore->supGain = SUPGAIN_DEFAULT << 1;
>+    aecm->aecmCore->supGainOld = SUPGAIN_DEFAULT << 1;
>+    aecm->aecmCore->supGainErrParamA = SUPGAIN_ERROR_PARAM_A << 1;
>+    aecm->aecmCore->supGainErrParamD = SUPGAIN_ERROR_PARAM_D << 1;
>+    aecm->aecmCore->supGainErrParamDiffAB =
>+        (SUPGAIN_ERROR_PARAM_A << 1) - (SUPGAIN_ERROR_PARAM_B << 1);
>+    aecm->aecmCore->supGainErrParamDiffBD =
>+        (SUPGAIN_ERROR_PARAM_B << 1) - (SUPGAIN_ERROR_PARAM_D << 1);
>+  }
> 
>-    return 0;
>+  return 0;
> }
> 
> int32_t WebRtcAecm_InitEchoPath(void* aecmInst,
>                                 const void* echo_path,
>-                                size_t size_bytes)
>-{
>-    AecMobile* aecm = static_cast<AecMobile*>(aecmInst);
>-    const int16_t* echo_path_ptr = static_cast<const int16_t*>(echo_path);
>+                                size_t size_bytes) {
>+  AecMobile* aecm = static_cast<AecMobile*>(aecmInst);
>+  const int16_t* echo_path_ptr = static_cast<const int16_t*>(echo_path);
> 
>-    if (aecmInst == NULL) {
>-      return -1;
>-    }
>-    if (echo_path == NULL) {
>-      return AECM_NULL_POINTER_ERROR;
>-    }
>-    if (size_bytes != WebRtcAecm_echo_path_size_bytes())
>-    {
>-        // Input channel size does not match the size of AECM
>-        return AECM_BAD_PARAMETER_ERROR;
>-    }
>-    if (aecm->initFlag != kInitCheck)
>-    {
>-        return AECM_UNINITIALIZED_ERROR;
>-    }
>+  if (aecmInst == NULL) {
>+    return -1;
>+  }
>+  if (echo_path == NULL) {
>+    return AECM_NULL_POINTER_ERROR;
>+  }
>+  if (size_bytes != WebRtcAecm_echo_path_size_bytes()) {
>+    // Input channel size does not match the size of AECM
>+    return AECM_BAD_PARAMETER_ERROR;
>+  }
>+  if (aecm->initFlag != kInitCheck) {
>+    return AECM_UNINITIALIZED_ERROR;
>+  }
> 
>-    WebRtcAecm_InitEchoPathCore(aecm->aecmCore, echo_path_ptr);
>+  WebRtcAecm_InitEchoPathCore(aecm->aecmCore, echo_path_ptr);
> 
>-    return 0;
>+  return 0;
> }
> 
> int32_t WebRtcAecm_GetEchoPath(void* aecmInst,
>                                void* echo_path,
>-                               size_t size_bytes)
>-{
>-    AecMobile* aecm = static_cast<AecMobile*>(aecmInst);
>-    int16_t* echo_path_ptr = static_cast<int16_t*>(echo_path);
>+                               size_t size_bytes) {
>+  AecMobile* aecm = static_cast<AecMobile*>(aecmInst);
>+  int16_t* echo_path_ptr = static_cast<int16_t*>(echo_path);
> 
>-    if (aecmInst == NULL) {
>-      return -1;
>-    }
>-    if (echo_path == NULL) {
>-      return AECM_NULL_POINTER_ERROR;
>-    }
>-    if (size_bytes != WebRtcAecm_echo_path_size_bytes())
>-    {
>-        // Input channel size does not match the size of AECM
>-        return AECM_BAD_PARAMETER_ERROR;
>-    }
>-    if (aecm->initFlag != kInitCheck)
>-    {
>-        return AECM_UNINITIALIZED_ERROR;
>-    }
>+  if (aecmInst == NULL) {
>+    return -1;
>+  }
>+  if (echo_path == NULL) {
>+    return AECM_NULL_POINTER_ERROR;
>+  }
>+  if (size_bytes != WebRtcAecm_echo_path_size_bytes()) {
>+    // Input channel size does not match the size of AECM
>+    return AECM_BAD_PARAMETER_ERROR;
>+  }
>+  if (aecm->initFlag != kInitCheck) {
>+    return AECM_UNINITIALIZED_ERROR;
>+  }
> 
>-    memcpy(echo_path_ptr, aecm->aecmCore->channelStored, size_bytes);
>-    return 0;
>+  memcpy(echo_path_ptr, aecm->aecmCore->channelStored, size_bytes);
>+  return 0;
> }
> 
>-size_t WebRtcAecm_echo_path_size_bytes()
>-{
>-    return (PART_LEN1 * sizeof(int16_t));
>+size_t WebRtcAecm_echo_path_size_bytes() {
>+  return (PART_LEN1 * sizeof(int16_t));
> }
> 
>-
> static int WebRtcAecm_EstBufDelay(AecMobile* aecm, short msInSndCardBuf) {
>-    short delayNew, nSampSndCard;
>-    short nSampFar = (short) WebRtc_available_read(aecm->farendBuf);
>-    short diff;
>+  short delayNew, nSampSndCard;
>+  short nSampFar = (short)WebRtc_available_read(aecm->farendBuf);
>+  short diff;
> 
>-    nSampSndCard = msInSndCardBuf * kSampMsNb * aecm->aecmCore->mult;
>+  nSampSndCard = msInSndCardBuf * kSampMsNb * aecm->aecmCore->mult;
> 
>-    delayNew = nSampSndCard - nSampFar;
>+  delayNew = nSampSndCard - nSampFar;
> 
>-    if (delayNew < FRAME_LEN)
>-    {
>-        WebRtc_MoveReadPtr(aecm->farendBuf, FRAME_LEN);
>-        delayNew += FRAME_LEN;
>-    }
>+  if (delayNew < FRAME_LEN) {
>+    WebRtc_MoveReadPtr(aecm->farendBuf, FRAME_LEN);
>+    delayNew += FRAME_LEN;
>+  }
> 
>-    aecm->filtDelay = WEBRTC_SPL_MAX(0, (8 * aecm->filtDelay + 2 * delayNew) / 10);
>-
>-    diff = aecm->filtDelay - aecm->knownDelay;
>-    if (diff > 224)
>-    {
>-        if (aecm->lastDelayDiff < 96)
>-        {
>-            aecm->timeForDelayChange = 0;
>-        } else
>-        {
>-            aecm->timeForDelayChange++;
>-        }
>-    } else if (diff < 96 && aecm->knownDelay > 0)
>-    {
>-        if (aecm->lastDelayDiff > 224)
>-        {
>-            aecm->timeForDelayChange = 0;
>-        } else
>-        {
>-            aecm->timeForDelayChange++;
>-        }
>-    } else
>-    {
>-        aecm->timeForDelayChange = 0;
>-    }
>-    aecm->lastDelayDiff = diff;
>+  aecm->filtDelay =
>+      WEBRTC_SPL_MAX(0, (8 * aecm->filtDelay + 2 * delayNew) / 10);
> 
>-    if (aecm->timeForDelayChange > 25)
>-    {
>-        aecm->knownDelay = WEBRTC_SPL_MAX((int)aecm->filtDelay - 160, 0);
>+  diff = aecm->filtDelay - aecm->knownDelay;
>+  if (diff > 224) {
>+    if (aecm->lastDelayDiff < 96) {
>+      aecm->timeForDelayChange = 0;
>+    } else {
>+      aecm->timeForDelayChange++;
>     }
>-    return 0;
>+  } else if (diff < 96 && aecm->knownDelay > 0) {
>+    if (aecm->lastDelayDiff > 224) {
>+      aecm->timeForDelayChange = 0;
>+    } else {
>+      aecm->timeForDelayChange++;
>+    }
>+  } else {
>+    aecm->timeForDelayChange = 0;
>+  }
>+  aecm->lastDelayDiff = diff;
>+
>+  if (aecm->timeForDelayChange > 25) {
>+    aecm->knownDelay = WEBRTC_SPL_MAX((int)aecm->filtDelay - 160, 0);
>+  }
>+  return 0;
> }
> 
> static int WebRtcAecm_DelayComp(AecMobile* aecm) {
>-    int nSampFar = (int) WebRtc_available_read(aecm->farendBuf);
>-    int nSampSndCard, delayNew, nSampAdd;
>-    const int maxStuffSamp = 10 * FRAME_LEN;
>-
>-    nSampSndCard = aecm->msInSndCardBuf * kSampMsNb * aecm->aecmCore->mult;
>-    delayNew = nSampSndCard - nSampFar;
>-
>-    if (delayNew > FAR_BUF_LEN - FRAME_LEN * aecm->aecmCore->mult)
>-    {
>-        // The difference of the buffer sizes is larger than the maximum
>-        // allowed known delay. Compensate by stuffing the buffer.
>-        nSampAdd = (int)(WEBRTC_SPL_MAX(((nSampSndCard >> 1) - nSampFar),
>-                FRAME_LEN));
>-        nSampAdd = WEBRTC_SPL_MIN(nSampAdd, maxStuffSamp);
>-
>-        WebRtc_MoveReadPtr(aecm->farendBuf, -nSampAdd);
>-        aecm->delayChange = 1; // the delay needs to be updated
>-    }
>+  int nSampFar = (int)WebRtc_available_read(aecm->farendBuf);
>+  int nSampSndCard, delayNew, nSampAdd;
>+  const int maxStuffSamp = 10 * FRAME_LEN;
>+
>+  nSampSndCard = aecm->msInSndCardBuf * kSampMsNb * aecm->aecmCore->mult;
>+  delayNew = nSampSndCard - nSampFar;
>+
>+  if (delayNew > FAR_BUF_LEN - FRAME_LEN * aecm->aecmCore->mult) {
>+    // The difference of the buffer sizes is larger than the maximum
>+    // allowed known delay. Compensate by stuffing the buffer.
>+    nSampAdd =
>+        (int)(WEBRTC_SPL_MAX(((nSampSndCard >> 1) - nSampFar), FRAME_LEN));
>+    nSampAdd = WEBRTC_SPL_MIN(nSampAdd, maxStuffSamp);
>+
>+    WebRtc_MoveReadPtr(aecm->farendBuf, -nSampAdd);
>+    aecm->delayChange = 1;  // the delay needs to be updated
>+  }
> 
>-    return 0;
>+  return 0;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aecm/echo_control_mobile.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aecm/echo_control_mobile.h
>index e0091c3e575..cea13b8b5e9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aecm/echo_control_mobile.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/aecm/echo_control_mobile.h
>@@ -11,28 +11,24 @@
> #ifndef MODULES_AUDIO_PROCESSING_AECM_ECHO_CONTROL_MOBILE_H_
> #define MODULES_AUDIO_PROCESSING_AECM_ECHO_CONTROL_MOBILE_H_
> 
>-#include <stdlib.h>
>+#include <stddef.h>
>+#include <stdint.h>
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>-
>-enum {
>-    AecmFalse = 0,
>-    AecmTrue
>-};
>+enum { AecmFalse = 0, AecmTrue };
> 
> // Errors
>-#define AECM_UNSPECIFIED_ERROR           12000
>-#define AECM_UNSUPPORTED_FUNCTION_ERROR  12001
>-#define AECM_UNINITIALIZED_ERROR         12002
>-#define AECM_NULL_POINTER_ERROR          12003
>-#define AECM_BAD_PARAMETER_ERROR         12004
>+#define AECM_UNSPECIFIED_ERROR 12000
>+#define AECM_UNSUPPORTED_FUNCTION_ERROR 12001
>+#define AECM_UNINITIALIZED_ERROR 12002
>+#define AECM_NULL_POINTER_ERROR 12003
>+#define AECM_BAD_PARAMETER_ERROR 12004
> 
> // Warnings
>-#define AECM_BAD_PARAMETER_WARNING       12100
>+#define AECM_BAD_PARAMETER_WARNING 12100
> 
> typedef struct {
>-    int16_t cngMode;            // AECM_FALSE, AECM_TRUE (default)
>-    int16_t echoMode;           // 0, 1, 2, 3 (default), 4
>+  int16_t cngMode;   // AECM_FALSE, AECM_TRUE (default)
>+  int16_t echoMode;  // 0, 1, 2, 3 (default), 4
> } AecmConfig;
> 
> #ifdef __cplusplus
>@@ -202,7 +198,6 @@ int32_t WebRtcAecm_GetEchoPath(void* aecmInst,
>  */
> size_t WebRtcAecm_echo_path_size_bytes();
> 
>-
> #ifdef __cplusplus
> }
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/BUILD.gn
>new file mode 100644
>index 00000000000..65653a0d3e6
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/BUILD.gn
>@@ -0,0 +1,126 @@
>+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+#
>+# Use of this source code is governed by a BSD-style license
>+# that can be found in the LICENSE file in the root of the source
>+# tree. An additional intellectual property rights grant can be found
>+# in the file PATENTS.  All contributing project authors may
>+# be found in the AUTHORS file in the root of the source tree.
>+
>+import("../../../webrtc.gni")
>+
>+rtc_source_set("agc") {
>+  sources = [
>+    "agc_manager_direct.cc",
>+    "agc_manager_direct.h",
>+  ]
>+  configs += [ "..:apm_debug_dump" ]
>+  deps = [
>+    ":gain_map",
>+    ":level_estimation",
>+    "..:apm_logging",
>+    "..:gain_control_interface",
>+    "../../..:webrtc_common",
>+    "../../../rtc_base:checks",
>+    "../../../rtc_base:gtest_prod",
>+    "../../../rtc_base:logging",
>+    "../../../rtc_base:macromagic",
>+    "../../../rtc_base:safe_minmax",
>+    "../../../system_wrappers:metrics_api",
>+    "../agc2:level_estimation_agc",
>+    "../vad",
>+  ]
>+}
>+
>+rtc_source_set("level_estimation") {
>+  sources = [
>+    "agc.cc",
>+    "agc.h",
>+    "loudness_histogram.cc",
>+    "loudness_histogram.h",
>+    "utility.cc",
>+    "utility.h",
>+  ]
>+  deps = [
>+    "../../..:webrtc_common",
>+    "../../../rtc_base:checks",
>+    "../../../rtc_base:macromagic",
>+    "../vad",
>+  ]
>+}
>+
>+rtc_source_set("agc_legacy_c") {
>+  visibility = [
>+    ":*",
>+    "..:*",
>+  ]  # Only targets in this file and in
>+     # audio_processing can depend on
>+     # this.
>+
>+  sources = [
>+    "legacy/analog_agc.c",
>+    "legacy/analog_agc.h",
>+    "legacy/digital_agc.c",
>+    "legacy/digital_agc.h",
>+    "legacy/gain_control.h",
>+  ]
>+
>+  deps = [
>+    "../../..:webrtc_common",
>+    "../../../common_audio",
>+    "../../../common_audio:common_audio_c",
>+    "../../../common_audio/third_party/fft4g",
>+    "../../../rtc_base:checks",
>+    "../../../rtc_base:rtc_base_approved",
>+    "../../../system_wrappers:cpu_features_api",
>+  ]
>+
>+  if (rtc_build_with_neon) {
>+    if (current_cpu != "arm64") {
>+      # Enable compilation for the NEON instruction set.
>+      suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
>+      cflags = [ "-mfpu=neon" ]
>+    }
>+
>+    # Disable LTO on NEON targets due to compiler bug.
>+    # TODO(fdegans): Enable this. See crbug.com/408997.
>+    if (rtc_use_lto) {
>+      cflags -= [
>+        "-flto",
>+        "-ffat-lto-objects",
>+      ]
>+    }
>+  }
>+}
>+
>+rtc_source_set("gain_map") {
>+  sources = [
>+    "gain_map_internal.h",
>+  ]
>+}
>+
>+if (rtc_include_tests) {
>+  rtc_source_set("agc_unittests") {
>+    testonly = true
>+    sources = [
>+      "agc_manager_direct_unittest.cc",
>+      "loudness_histogram_unittest.cc",
>+      "mock_agc.h",
>+    ]
>+    configs += [ "..:apm_debug_dump" ]
>+
>+    if ((!build_with_chromium || is_win) && is_clang) {
>+      # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>+      suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>+    }
>+
>+    deps = [
>+      ":agc",
>+      ":level_estimation",
>+      "..:mocks",
>+      "../../..:webrtc_common",
>+      "../../../test:fileutils",
>+      "../../../test:test_support",
>+      "//testing/gtest",
>+    ]
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/agc.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/agc.cc
>index 0c6478e803e..12c8cfb7fbd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/agc.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/agc.cc
>@@ -18,7 +18,6 @@
> 
> #include "modules/audio_processing/agc/loudness_histogram.h"
> #include "modules/audio_processing/agc/utility.h"
>-#include "modules/include/module_common_types.h"
> #include "rtc_base/checks.h"
> 
> namespace webrtc {
>@@ -48,7 +47,7 @@ float Agc::AnalyzePreproc(const int16_t* audio, size_t length) {
>   return 1.0f * num_clipped / length;
> }
> 
>-int Agc::Process(const int16_t* audio, size_t length, int sample_rate_hz) {
>+void Agc::Process(const int16_t* audio, size_t length, int sample_rate_hz) {
>   vad_.ProcessChunk(audio, length, sample_rate_hz);
>   const std::vector<double>& rms = vad_.chunkwise_rms();
>   const std::vector<double>& probabilities =
>@@ -57,7 +56,6 @@ int Agc::Process(const int16_t* audio, size_t length, int sample_rate_hz) {
>   for (size_t i = 0; i < rms.size(); ++i) {
>     histogram_->Update(rms[i], probabilities[i]);
>   }
>-  return 0;
> }
> 
> bool Agc::GetRmsErrorDb(int* error) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/agc.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/agc.h
>index 3a8d5c8122c..5d34c212f3a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/agc.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/agc.h
>@@ -14,7 +14,6 @@
> #include <memory>
> 
> #include "modules/audio_processing/vad/voice_activity_detector.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -31,7 +30,7 @@ class Agc {
>   virtual float AnalyzePreproc(const int16_t* audio, size_t length);
>   // |audio| must be mono; in a multi-channel stream, provide the first (usually
>   // left) channel.
>-  virtual int Process(const int16_t* audio, size_t length, int sample_rate_hz);
>+  virtual void Process(const int16_t* audio, size_t length, int sample_rate_hz);
> 
>   // Retrieves the difference between the target RMS level and the current
>   // signal RMS level in dB. Returns true if an update is available and false
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/agc_manager_direct.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/agc_manager_direct.cc
>index 706d4ab037c..dc6d45121cd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/agc_manager_direct.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/agc_manager_direct.cc
>@@ -17,8 +17,8 @@
> #endif
> 
> #include "modules/audio_processing/agc/gain_map_internal.h"
>-#include "modules/audio_processing/gain_control_impl.h"
>-#include "modules/include/module_common_types.h"
>+#include "modules/audio_processing/agc2/adaptive_mode_level_estimator_agc.h"
>+#include "modules/audio_processing/include/gain_control.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/numerics/safe_minmax.h"
>@@ -26,6 +26,8 @@
> 
> namespace webrtc {
> 
>+int AgcManagerDirect::instance_counter_ = 0;
>+
> namespace {
> 
> // Amount the microphone level is lowered with every clipping event.
>@@ -70,18 +72,43 @@ int LevelFromGainError(int gain_error, int level) {
>   int new_level = level;
>   if (gain_error > 0) {
>     while (kGainMap[new_level] - kGainMap[level] < gain_error &&
>-          new_level < kMaxMicLevel) {
>+           new_level < kMaxMicLevel) {
>       ++new_level;
>     }
>   } else {
>     while (kGainMap[new_level] - kGainMap[level] > gain_error &&
>-          new_level > kMinMicLevel) {
>+           new_level > kMinMicLevel) {
>       --new_level;
>     }
>   }
>   return new_level;
> }
> 
>+int InitializeGainControl(GainControl* gain_control,
>+                          bool disable_digital_adaptive) {
>+  if (gain_control->set_mode(GainControl::kFixedDigital) != 0) {
>+    RTC_LOG(LS_ERROR) << "set_mode(GainControl::kFixedDigital) failed.";
>+    return -1;
>+  }
>+  const int target_level_dbfs = disable_digital_adaptive ? 0 : 2;
>+  if (gain_control->set_target_level_dbfs(target_level_dbfs) != 0) {
>+    RTC_LOG(LS_ERROR) << "set_target_level_dbfs() failed.";
>+    return -1;
>+  }
>+  const int compression_gain_db =
>+      disable_digital_adaptive ? 0 : kDefaultCompressionGain;
>+  if (gain_control->set_compression_gain_db(compression_gain_db) != 0) {
>+    RTC_LOG(LS_ERROR) << "set_compression_gain_db() failed.";
>+    return -1;
>+  }
>+  const bool enable_limiter = !disable_digital_adaptive;
>+  if (gain_control->enable_limiter(enable_limiter) != 0) {
>+    RTC_LOG(LS_ERROR) << "enable_limiter() failed.";
>+    return -1;
>+  }
>+  return 0;
>+}
>+
> }  // namespace
> 
> // Facility for dumping debug audio files. All methods are no-ops in the
>@@ -89,57 +116,64 @@ int LevelFromGainError(int gain_error, int level) {
> class DebugFile {
> #ifdef WEBRTC_AGC_DEBUG_DUMP
>  public:
>-  explicit DebugFile(const char* filename)
>-      : file_(fopen(filename, "wb")) {
>+  explicit DebugFile(const char* filename) : file_(fopen(filename, "wb")) {
>     RTC_DCHECK(file_);
>   }
>-  ~DebugFile() {
>-    fclose(file_);
>-  }
>+  ~DebugFile() { fclose(file_); }
>   void Write(const int16_t* data, size_t length_samples) {
>     fwrite(data, 1, length_samples * sizeof(int16_t), file_);
>   }
>+
>  private:
>   FILE* file_;
> #else
>  public:
>-  explicit DebugFile(const char* filename) {
>-  }
>-  ~DebugFile() {
>-  }
>-  void Write(const int16_t* data, size_t length_samples) {
>-  }
>+  explicit DebugFile(const char* filename) {}
>+  ~DebugFile() {}
>+  void Write(const int16_t* data, size_t length_samples) {}
> #endif  // WEBRTC_AGC_DEBUG_DUMP
> };
> 
> AgcManagerDirect::AgcManagerDirect(GainControl* gctrl,
>                                    VolumeCallbacks* volume_callbacks,
>                                    int startup_min_level,
>-                                   int clipped_level_min)
>-    : agc_(new Agc()),
>-      gctrl_(gctrl),
>-      volume_callbacks_(volume_callbacks),
>-      frames_since_clipped_(kClippedWaitFrames),
>-      level_(0),
>-      max_level_(kMaxMicLevel),
>-      max_compression_gain_(kMaxCompressionGain),
>-      target_compression_(kDefaultCompressionGain),
>-      compression_(target_compression_),
>-      compression_accumulator_(compression_),
>-      capture_muted_(false),
>-      check_volume_on_next_process_(true),  // Check at startup.
>-      startup_(true),
>-      startup_min_level_(ClampLevel(startup_min_level)),
>-      clipped_level_min_(clipped_level_min),
>-      file_preproc_(new DebugFile("agc_preproc.pcm")),
>-      file_postproc_(new DebugFile("agc_postproc.pcm")) {}
>+                                   int clipped_level_min,
>+                                   bool use_agc2_level_estimation,
>+                                   bool disable_digital_adaptive)
>+    : AgcManagerDirect(use_agc2_level_estimation ? nullptr : new Agc(),
>+                       gctrl,
>+                       volume_callbacks,
>+                       startup_min_level,
>+                       clipped_level_min,
>+                       use_agc2_level_estimation,
>+                       disable_digital_adaptive) {
>+  RTC_DCHECK(agc_);
>+}
> 
> AgcManagerDirect::AgcManagerDirect(Agc* agc,
>                                    GainControl* gctrl,
>                                    VolumeCallbacks* volume_callbacks,
>                                    int startup_min_level,
>                                    int clipped_level_min)
>-    : agc_(agc),
>+    : AgcManagerDirect(agc,
>+                       gctrl,
>+                       volume_callbacks,
>+                       startup_min_level,
>+                       clipped_level_min,
>+                       false,
>+                       false) {
>+  RTC_DCHECK(agc_);
>+}
>+
>+AgcManagerDirect::AgcManagerDirect(Agc* agc,
>+                                   GainControl* gctrl,
>+                                   VolumeCallbacks* volume_callbacks,
>+                                   int startup_min_level,
>+                                   int clipped_level_min,
>+                                   bool use_agc2_level_estimation,
>+                                   bool disable_digital_adaptive)
>+    : data_dumper_(new ApmDataDumper(instance_counter_)),
>+      agc_(agc),
>       gctrl_(gctrl),
>       volume_callbacks_(volume_callbacks),
>       frames_since_clipped_(kClippedWaitFrames),
>@@ -152,42 +186,37 @@ AgcManagerDirect::AgcManagerDirect(Agc* agc,
>       capture_muted_(false),
>       check_volume_on_next_process_(true),  // Check at startup.
>       startup_(true),
>+      use_agc2_level_estimation_(use_agc2_level_estimation),
>+      disable_digital_adaptive_(disable_digital_adaptive),
>       startup_min_level_(ClampLevel(startup_min_level)),
>       clipped_level_min_(clipped_level_min),
>       file_preproc_(new DebugFile("agc_preproc.pcm")),
>-      file_postproc_(new DebugFile("agc_postproc.pcm")) {}
>+      file_postproc_(new DebugFile("agc_postproc.pcm")) {
>+  instance_counter_++;
>+  if (use_agc2_level_estimation_) {
>+    RTC_DCHECK(!agc);
>+    agc_.reset(new AdaptiveModeLevelEstimatorAgc(data_dumper_.get()));
>+  } else {
>+    RTC_DCHECK(agc);
>+  }
>+}
> 
> AgcManagerDirect::~AgcManagerDirect() {}
> 
> int AgcManagerDirect::Initialize() {
>   max_level_ = kMaxMicLevel;
>   max_compression_gain_ = kMaxCompressionGain;
>-  target_compression_ = kDefaultCompressionGain;
>-  compression_ = target_compression_;
>+  target_compression_ = disable_digital_adaptive_ ? 0 : kDefaultCompressionGain;
>+  compression_ = disable_digital_adaptive_ ? 0 : target_compression_;
>   compression_accumulator_ = compression_;
>   capture_muted_ = false;
>   check_volume_on_next_process_ = true;
>   // TODO(bjornv): Investigate if we need to reset |startup_| as well. For
>   // example, what happens when we change devices.
> 
>-  if (gctrl_->set_mode(GainControl::kFixedDigital) != 0) {
>-    RTC_LOG(LS_ERROR) << "set_mode(GainControl::kFixedDigital) failed.";
>-    return -1;
>-  }
>-  if (gctrl_->set_target_level_dbfs(2) != 0) {
>-    RTC_LOG(LS_ERROR) << "set_target_level_dbfs(2) failed.";
>-    return -1;
>-  }
>-  if (gctrl_->set_compression_gain_db(kDefaultCompressionGain) != 0) {
>-    RTC_LOG(LS_ERROR)
>-        << "set_compression_gain_db(kDefaultCompressionGain) failed.";
>-    return -1;
>-  }
>-  if (gctrl_->enable_limiter(true) != 0) {
>-    RTC_LOG(LS_ERROR) << "enable_limiter(true) failed.";
>-    return -1;
>-  }
>-  return 0;
>+  data_dumper_->InitiateNewSetOfRecordings();
>+
>+  return InitializeGainControl(gctrl_, disable_digital_adaptive_);
> }
> 
> void AgcManagerDirect::AnalyzePreProcess(int16_t* audio,
>@@ -216,8 +245,8 @@ void AgcManagerDirect::AnalyzePreProcess(int16_t* audio,
>   // gain is increased, through SetMaxLevel().
>   float clipped_ratio = agc_->AnalyzePreproc(audio, length);
>   if (clipped_ratio > kClippedRatioThreshold) {
>-    RTC_LOG(LS_INFO) << "[agc] Clipping detected. clipped_ratio="
>-                     << clipped_ratio;
>+    RTC_DLOG(LS_INFO) << "[agc] Clipping detected. clipped_ratio="
>+                      << clipped_ratio;
>     // Always decrease the maximum level, even if the current level is below
>     // threshold.
>     SetMaxLevel(std::max(clipped_level_min_, max_level_ - kClippedLevelStep));
>@@ -249,28 +278,27 @@ void AgcManagerDirect::Process(const int16_t* audio,
>     CheckVolumeAndReset();
>   }
> 
>-  if (agc_->Process(audio, length, sample_rate_hz) != 0) {
>-    RTC_LOG(LS_ERROR) << "Agc::Process failed";
>-    RTC_NOTREACHED();
>-  }
>+  agc_->Process(audio, length, sample_rate_hz);
> 
>   UpdateGain();
>-  UpdateCompressor();
>+  if (!disable_digital_adaptive_) {
>+    UpdateCompressor();
>+  }
> 
>   file_postproc_->Write(audio, length);
>+
>+  data_dumper_->DumpRaw("experimental_gain_control_compression_gain_db", 1,
>+                        &compression_);
> }
> 
> void AgcManagerDirect::SetLevel(int new_level) {
>   int voe_level = volume_callbacks_->GetMicVolume();
>-  if (voe_level < 0) {
>-    return;
>-  }
>   if (voe_level == 0) {
>-    RTC_LOG(LS_INFO)
>+    RTC_DLOG(LS_INFO)
>         << "[agc] VolumeCallbacks returned level=0, taking no action.";
>     return;
>   }
>-  if (voe_level > kMaxMicLevel) {
>+  if (voe_level < 0 || voe_level > kMaxMicLevel) {
>     RTC_LOG(LS_ERROR) << "VolumeCallbacks returned an invalid level="
>                       << voe_level;
>     return;
>@@ -278,8 +306,9 @@ void AgcManagerDirect::SetLevel(int new_level) {
> 
>   if (voe_level > level_ + kLevelQuantizationSlack ||
>       voe_level < level_ - kLevelQuantizationSlack) {
>-    RTC_LOG(LS_INFO) << "[agc] Mic volume was manually adjusted. Updating "
>-                     << "stored level from " << level_ << " to " << voe_level;
>+    RTC_DLOG(LS_INFO) << "[agc] Mic volume was manually adjusted. Updating "
>+                         "stored level from "
>+                      << level_ << " to " << voe_level;
>     level_ = voe_level;
>     // Always allow the user to increase the volume.
>     if (level_ > max_level_) {
>@@ -298,9 +327,9 @@ void AgcManagerDirect::SetLevel(int new_level) {
>   }
> 
>   volume_callbacks_->SetMicVolume(new_level);
>-  RTC_LOG(LS_INFO) << "[agc] voe_level=" << voe_level << ", "
>-                   << "level_=" << level_ << ", "
>-                   << "new_level=" << new_level;
>+  RTC_DLOG(LS_INFO) << "[agc] voe_level=" << voe_level << ", "
>+                    << "level_=" << level_ << ", "
>+                    << "new_level=" << new_level;
>   level_ = new_level;
> }
> 
>@@ -314,8 +343,8 @@ void AgcManagerDirect::SetMaxLevel(int level) {
>                                            (kMaxMicLevel - clipped_level_min_) *
>                                            kSurplusCompressionGain +
>                                        0.5f);
>-  RTC_LOG(LS_INFO) << "[agc] max_level_=" << max_level_
>-                   << ", max_compression_gain_=" << max_compression_gain_;
>+  RTC_DLOG(LS_INFO) << "[agc] max_level_=" << max_level_
>+                    << ", max_compression_gain_=" << max_compression_gain_;
> }
> 
> void AgcManagerDirect::SetCaptureMuted(bool muted) {
>@@ -336,28 +365,26 @@ float AgcManagerDirect::voice_probability() {
> 
> int AgcManagerDirect::CheckVolumeAndReset() {
>   int level = volume_callbacks_->GetMicVolume();
>-  if (level < 0) {
>-    return -1;
>-  }
>   // Reasons for taking action at startup:
>   // 1) A person starting a call is expected to be heard.
>   // 2) Independent of interpretation of |level| == 0 we should raise it so the
>   // AGC can do its job properly.
>   if (level == 0 && !startup_) {
>-    RTC_LOG(LS_INFO)
>+    RTC_DLOG(LS_INFO)
>         << "[agc] VolumeCallbacks returned level=0, taking no action.";
>     return 0;
>   }
>-  if (level > kMaxMicLevel) {
>-    RTC_LOG(LS_ERROR) << "VolumeCallbacks returned an invalid level=" << level;
>+  if (level < 0 || level > kMaxMicLevel) {
>+    RTC_LOG(LS_ERROR) << "[agc] VolumeCallbacks returned an invalid level="
>+                      << level;
>     return -1;
>   }
>-  RTC_LOG(LS_INFO) << "[agc] Initial GetMicVolume()=" << level;
>+  RTC_DLOG(LS_INFO) << "[agc] Initial GetMicVolume()=" << level;
> 
>   int minLevel = startup_ ? startup_min_level_ : kMinMicLevel;
>   if (level < minLevel) {
>     level = minLevel;
>-    RTC_LOG(LS_INFO) << "[agc] Initial volume too low, raising to " << level;
>+    RTC_DLOG(LS_INFO) << "[agc] Initial volume too low, raising to " << level;
>     volume_callbacks_->SetMicVolume(level);
>   }
>   agc_->Reset();
>@@ -392,15 +419,15 @@ void AgcManagerDirect::UpdateGain() {
>   // target and the newly received target. This serves to soften perceptible
>   // intra-talkspurt adjustments, at the cost of some adaptation speed.
>   if ((raw_compression == max_compression_gain_ &&
>-      target_compression_ == max_compression_gain_ - 1) ||
>+       target_compression_ == max_compression_gain_ - 1) ||
>       (raw_compression == kMinCompressionGain &&
>-      target_compression_ == kMinCompressionGain + 1)) {
>+       target_compression_ == kMinCompressionGain + 1)) {
>     // Special case to allow the target to reach the endpoints of the
>     // compression range. The deemphasis would otherwise halt it at 1 dB shy.
>     target_compression_ = raw_compression;
>   } else {
>-    target_compression_ = (raw_compression - target_compression_) / 2
>-        + target_compression_;
>+    target_compression_ =
>+        (raw_compression - target_compression_) / 2 + target_compression_;
>   }
> 
>   // Residual error will be handled by adjusting the volume slider. Use the
>@@ -409,9 +436,9 @@ void AgcManagerDirect::UpdateGain() {
>   const int residual_gain =
>       rtc::SafeClamp(rms_error - raw_compression, -kMaxResidualGainChange,
>                      kMaxResidualGainChange);
>-  RTC_LOG(LS_INFO) << "[agc] rms_error=" << rms_error << ", "
>-                   << "target_compression=" << target_compression_ << ", "
>-                   << "residual_gain=" << residual_gain;
>+  RTC_DLOG(LS_INFO) << "[agc] rms_error=" << rms_error
>+                    << ", target_compression=" << target_compression_
>+                    << ", residual_gain=" << residual_gain;
>   if (residual_gain == 0)
>     return;
> 
>@@ -421,10 +448,19 @@ void AgcManagerDirect::UpdateGain() {
>     // level_ was updated by SetLevel; log the new value.
>     RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.AgcSetLevel", level_, 1,
>                                 kMaxMicLevel, 50);
>+    // Reset the AGC since the level has changed.
>+    agc_->Reset();
>   }
> }
> 
> void AgcManagerDirect::UpdateCompressor() {
>+  calls_since_last_gain_log_++;
>+  if (calls_since_last_gain_log_ == 100) {
>+    calls_since_last_gain_log_ = 0;
>+    RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.Agc.DigitalGainApplied",
>+                                compression_, 0, kMaxCompressionGain,
>+                                kMaxCompressionGain + 1);
>+  }
>   if (compression_ == target_compression_) {
>     return;
>   }
>@@ -449,6 +485,9 @@ void AgcManagerDirect::UpdateCompressor() {
> 
>   // Set the new compression gain.
>   if (new_compression != compression_) {
>+    RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.Agc.DigitalGainUpdated",
>+                                new_compression, 0, kMaxCompressionGain,
>+                                kMaxCompressionGain + 1);
>     compression_ = new_compression;
>     compression_accumulator_ = new_compression;
>     if (gctrl_->set_compression_gain_db(compression_) != 0) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/agc_manager_direct.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/agc_manager_direct.h
>index 03d2607f85f..cbfd6a1018f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/agc_manager_direct.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/agc_manager_direct.h
>@@ -14,7 +14,9 @@
> #include <memory>
> 
> #include "modules/audio_processing/agc/agc.h"
>+#include "modules/audio_processing/logging/apm_data_dumper.h"
> #include "rtc_base/constructormagic.h"
>+#include "rtc_base/gtest_prod_util.h"
> 
> namespace webrtc {
> 
>@@ -47,14 +49,10 @@ class AgcManagerDirect final {
>   AgcManagerDirect(GainControl* gctrl,
>                    VolumeCallbacks* volume_callbacks,
>                    int startup_min_level,
>-                   int clipped_level_min);
>-  // Dependency injection for testing. Don't delete |agc| as the memory is owned
>-  // by the manager.
>-  AgcManagerDirect(Agc* agc,
>-                   GainControl* gctrl,
>-                   VolumeCallbacks* volume_callbacks,
>-                   int startup_min_level,
>-                   int clipped_level_min);
>+                   int clipped_level_min,
>+                   bool use_agc2_level_estimation,
>+                   bool disable_digital_adaptive);
>+
>   ~AgcManagerDirect();
> 
>   int Initialize();
>@@ -72,6 +70,28 @@ class AgcManagerDirect final {
>   float voice_probability();
> 
>  private:
>+  friend class AgcManagerDirectTest;
>+
>+  FRIEND_TEST_ALL_PREFIXES(AgcManagerDirectStandaloneTest,
>+                           DisableDigitalDisablesDigital);
>+
>+  // Dependency injection for testing. Don't delete |agc| as the memory is owned
>+  // by the manager.
>+  AgcManagerDirect(Agc* agc,
>+                   GainControl* gctrl,
>+                   VolumeCallbacks* volume_callbacks,
>+                   int startup_min_level,
>+                   int clipped_level_min);
>+
>+  // Most general c-tor.
>+  AgcManagerDirect(Agc* agc,
>+                   GainControl* gctrl,
>+                   VolumeCallbacks* volume_callbacks,
>+                   int startup_min_level,
>+                   int clipped_level_min,
>+                   bool use_agc2_level_estimation,
>+                   bool disable_digital_adaptive);
>+
>   // Sets a new microphone level, after first checking that it hasn't been
>   // updated by the user, in which case no action is taken.
>   void SetLevel(int new_level);
>@@ -85,6 +105,9 @@ class AgcManagerDirect final {
>   void UpdateGain();
>   void UpdateCompressor();
> 
>+  std::unique_ptr<ApmDataDumper> data_dumper_;
>+  static int instance_counter_;
>+
>   std::unique_ptr<Agc> agc_;
>   GainControl* gctrl_;
>   VolumeCallbacks* volume_callbacks_;
>@@ -99,8 +122,11 @@ class AgcManagerDirect final {
>   bool capture_muted_;
>   bool check_volume_on_next_process_;
>   bool startup_;
>+  const bool use_agc2_level_estimation_;
>+  const bool disable_digital_adaptive_;
>   int startup_min_level_;
>   const int clipped_level_min_;
>+  int calls_since_last_gain_log_ = 0;
> 
>   std::unique_ptr<DebugFile> file_preproc_;
>   std::unique_ptr<DebugFile> file_postproc_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/agc_manager_direct_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/agc_manager_direct_unittest.cc
>index 6a8af6aa71e..73ea55d06c7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/agc_manager_direct_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/agc_manager_direct_unittest.cc
>@@ -17,6 +17,7 @@
> #include "test/gtest.h"
> 
> using ::testing::_;
>+using ::testing::AtLeast;
> using ::testing::DoAll;
> using ::testing::Return;
> using ::testing::SetArgPointee;
>@@ -53,7 +54,7 @@ class AgcManagerDirectTest : public ::testing::Test {
>   }
> 
>   void FirstProcess() {
>-    EXPECT_CALL(*agc_, Reset());
>+    EXPECT_CALL(*agc_, Reset()).Times(AtLeast(1));
>     EXPECT_CALL(*agc_, GetRmsErrorDb(_)).WillOnce(Return(false));
>     CallProcess(1);
>   }
>@@ -77,7 +78,7 @@ class AgcManagerDirectTest : public ::testing::Test {
> 
>   void CallProcess(int num_calls) {
>     for (int i = 0; i < num_calls; ++i) {
>-      EXPECT_CALL(*agc_, Process(_, _, _)).WillOnce(Return(0));
>+      EXPECT_CALL(*agc_, Process(_, _, _)).WillOnce(Return());
>       manager_.Process(nullptr, kSamplesPerChannel, kSampleRateHz);
>     }
>   }
>@@ -366,11 +367,14 @@ TEST_F(AgcManagerDirectTest, ManualLevelChangeResultsInNoSetMicCall) {
>   // to SetMicVolume.
>   EXPECT_CALL(*agc_, GetRmsErrorDb(_))
>       .WillOnce(DoAll(SetArgPointee<0>(11), Return(true)));
>+
>+  // When the analog volume changes, the gain controller is reset.
>+  EXPECT_CALL(*agc_, Reset()).Times(AtLeast(1));
>+
>   // GetMicVolume returns a value outside of the quantization slack, indicating
>   // a manual volume change.
>+  ASSERT_NE(volume_.GetMicVolume(), 154);
>   volume_.SetMicVolume(154);
>-  // SetMicVolume should not be called.
>-  EXPECT_CALL(*agc_, Reset()).Times(1);
>   CallProcess(1);
>   EXPECT_EQ(154, volume_.GetMicVolume());
> 
>@@ -378,7 +382,7 @@ TEST_F(AgcManagerDirectTest, ManualLevelChangeResultsInNoSetMicCall) {
>   EXPECT_CALL(*agc_, GetRmsErrorDb(_))
>       .WillOnce(DoAll(SetArgPointee<0>(-1), Return(true)));
>   volume_.SetMicVolume(100);
>-  EXPECT_CALL(*agc_, Reset()).Times(1);
>+  EXPECT_CALL(*agc_, Reset()).Times(AtLeast(1));
>   CallProcess(1);
>   EXPECT_EQ(100, volume_.GetMicVolume());
> 
>@@ -407,7 +411,7 @@ TEST_F(AgcManagerDirectTest, RecoveryAfterManualLevelChangeFromMax) {
>   EXPECT_CALL(*agc_, GetRmsErrorDb(_))
>       .WillOnce(DoAll(SetArgPointee<0>(-1), Return(true)));
>   volume_.SetMicVolume(50);
>-  EXPECT_CALL(*agc_, Reset()).Times(1);
>+  EXPECT_CALL(*agc_, Reset()).Times(AtLeast(1));
>   CallProcess(1);
>   EXPECT_EQ(50, volume_.GetMicVolume());
> 
>@@ -426,7 +430,7 @@ TEST_F(AgcManagerDirectTest, RecoveryAfterManualLevelChangeBelowMin) {
>       .WillOnce(DoAll(SetArgPointee<0>(-1), Return(true)));
>   // Don't set to zero, which will cause AGC to take no action.
>   volume_.SetMicVolume(1);
>-  EXPECT_CALL(*agc_, Reset()).Times(1);
>+  EXPECT_CALL(*agc_, Reset()).Times(AtLeast(1));
>   CallProcess(1);
>   EXPECT_EQ(1, volume_.GetMicVolume());
> 
>@@ -467,7 +471,7 @@ TEST_F(AgcManagerDirectTest, ClippingLowersVolume) {
>   SetVolumeAndProcess(255);
> 
>   EXPECT_CALL(*agc_, AnalyzePreproc(_, _)).WillOnce(Return(0.101));
>-  EXPECT_CALL(*agc_, Reset()).Times(1);
>+  EXPECT_CALL(*agc_, Reset()).Times(AtLeast(1));
>   CallPreProc(1);
>   EXPECT_EQ(240, volume_.GetMicVolume());
> }
>@@ -477,7 +481,7 @@ TEST_F(AgcManagerDirectTest, WaitingPeriodBetweenClippingChecks) {
> 
>   EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
>       .WillOnce(Return(kAboveClippedThreshold));
>-  EXPECT_CALL(*agc_, Reset()).Times(1);
>+  EXPECT_CALL(*agc_, Reset()).Times(AtLeast(1));
>   CallPreProc(1);
>   EXPECT_EQ(240, volume_.GetMicVolume());
> 
>@@ -489,7 +493,7 @@ TEST_F(AgcManagerDirectTest, WaitingPeriodBetweenClippingChecks) {
> 
>   EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
>       .WillOnce(Return(kAboveClippedThreshold));
>-  EXPECT_CALL(*agc_, Reset()).Times(1);
>+  EXPECT_CALL(*agc_, Reset()).Times(AtLeast(1));
>   CallPreProc(1);
>   EXPECT_EQ(225, volume_.GetMicVolume());
> }
>@@ -499,7 +503,7 @@ TEST_F(AgcManagerDirectTest, ClippingLoweringIsLimited) {
> 
>   EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
>       .WillOnce(Return(kAboveClippedThreshold));
>-  EXPECT_CALL(*agc_, Reset()).Times(1);
>+  EXPECT_CALL(*agc_, Reset()).Times(AtLeast(1));
>   CallPreProc(1);
>   EXPECT_EQ(kClippedMin, volume_.GetMicVolume());
> 
>@@ -515,7 +519,7 @@ TEST_F(AgcManagerDirectTest, ClippingMaxIsRespectedWhenEqualToLevel) {
> 
>   EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
>       .WillOnce(Return(kAboveClippedThreshold));
>-  EXPECT_CALL(*agc_, Reset()).Times(1);
>+  EXPECT_CALL(*agc_, Reset()).Times(AtLeast(1));
>   CallPreProc(1);
>   EXPECT_EQ(240, volume_.GetMicVolume());
> 
>@@ -530,7 +534,7 @@ TEST_F(AgcManagerDirectTest, ClippingMaxIsRespectedWhenHigherThanLevel) {
> 
>   EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
>       .WillOnce(Return(kAboveClippedThreshold));
>-  EXPECT_CALL(*agc_, Reset()).Times(1);
>+  EXPECT_CALL(*agc_, Reset()).Times(AtLeast(1));
>   CallPreProc(1);
>   EXPECT_EQ(185, volume_.GetMicVolume());
> 
>@@ -547,7 +551,7 @@ TEST_F(AgcManagerDirectTest, MaxCompressionIsIncreasedAfterClipping) {
> 
>   EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
>       .WillOnce(Return(kAboveClippedThreshold));
>-  EXPECT_CALL(*agc_, Reset()).Times(1);
>+  EXPECT_CALL(*agc_, Reset()).Times(AtLeast(1));
>   CallPreProc(1);
>   EXPECT_EQ(195, volume_.GetMicVolume());
> 
>@@ -576,14 +580,14 @@ TEST_F(AgcManagerDirectTest, MaxCompressionIsIncreasedAfterClipping) {
>   CallPreProc(300);
>   EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
>       .WillOnce(Return(kAboveClippedThreshold));
>-  EXPECT_CALL(*agc_, Reset()).Times(1);
>+  EXPECT_CALL(*agc_, Reset()).Times(AtLeast(1));
>   CallPreProc(1);
>   EXPECT_EQ(180, volume_.GetMicVolume());
> 
>   CallPreProc(300);
>   EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
>       .WillOnce(Return(kAboveClippedThreshold));
>-  EXPECT_CALL(*agc_, Reset()).Times(1);
>+  EXPECT_CALL(*agc_, Reset()).Times(AtLeast(1));
>   CallPreProc(1);
>   EXPECT_EQ(kClippedMin, volume_.GetMicVolume());
> 
>@@ -628,7 +632,7 @@ TEST_F(AgcManagerDirectTest, UserCanRaiseVolumeAfterClipping) {
> 
>   EXPECT_CALL(*agc_, AnalyzePreproc(_, _))
>       .WillOnce(Return(kAboveClippedThreshold));
>-  EXPECT_CALL(*agc_, Reset()).Times(1);
>+  EXPECT_CALL(*agc_, Reset()).Times(AtLeast(1));
>   CallPreProc(1);
>   EXPECT_EQ(210, volume_.GetMicVolume());
> 
>@@ -637,7 +641,7 @@ TEST_F(AgcManagerDirectTest, UserCanRaiseVolumeAfterClipping) {
>       .WillOnce(DoAll(SetArgPointee<0>(14), Return(true)));
>   // User changed the volume.
>   volume_.SetMicVolume(250);
>-  EXPECT_CALL(*agc_, Reset()).Times(1);
>+  EXPECT_CALL(*agc_, Reset()).Times(AtLeast(1));
>   CallProcess(1);
>   EXPECT_EQ(250, volume_.GetMicVolume());
> 
>@@ -679,4 +683,22 @@ TEST_F(AgcManagerDirectTest, TakesNoActionOnZeroMicVolume) {
>   EXPECT_EQ(0, volume_.GetMicVolume());
> }
> 
>+TEST(AgcManagerDirectStandaloneTest, DisableDigitalDisablesDigital) {
>+  auto agc = std::unique_ptr<Agc>(new testing::NiceMock<MockAgc>());
>+  test::MockGainControl gctrl;
>+  TestVolumeCallbacks volume;
>+
>+  AgcManagerDirect manager(agc.release(), &gctrl, &volume, kInitialVolume,
>+                           kClippedMin,
>+                           /* use agc2 level estimation */ false,
>+                           /* disable digital adaptive */ true);
>+
>+  EXPECT_CALL(gctrl, set_mode(GainControl::kFixedDigital));
>+  EXPECT_CALL(gctrl, set_target_level_dbfs(0));
>+  EXPECT_CALL(gctrl, set_compression_gain_db(0));
>+  EXPECT_CALL(gctrl, enable_limiter(false));
>+
>+  manager.Initialize();
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/gain_map_internal.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/gain_map_internal.h
>index f09c748b59b..63d9b510ca5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/gain_map_internal.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/gain_map_internal.h
>@@ -14,262 +14,23 @@
> static const int kGainMapSize = 256;
> // Uses parameters: si = 2, sf = 0.25, D = 8/256
> static const int kGainMap[kGainMapSize] = {
>-  -56,
>-  -54,
>-  -52,
>-  -50,
>-  -48,
>-  -47,
>-  -45,
>-  -43,
>-  -42,
>-  -40,
>-  -38,
>-  -37,
>-  -35,
>-  -34,
>-  -33,
>-  -31,
>-  -30,
>-  -29,
>-  -27,
>-  -26,
>-  -25,
>-  -24,
>-  -23,
>-  -22,
>-  -20,
>-  -19,
>-  -18,
>-  -17,
>-  -16,
>-  -15,
>-  -14,
>-  -14,
>-  -13,
>-  -12,
>-  -11,
>-  -10,
>-  -9,
>-  -8,
>-  -8,
>-  -7,
>-  -6,
>-  -5,
>-  -5,
>-  -4,
>-  -3,
>-  -2,
>-  -2,
>-  -1,
>-  0,
>-  0,
>-  1,
>-  1,
>-  2,
>-  3,
>-  3,
>-  4,
>-  4,
>-  5,
>-  5,
>-  6,
>-  6,
>-  7,
>-  7,
>-  8,
>-  8,
>-  9,
>-  9,
>-  10,
>-  10,
>-  11,
>-  11,
>-  12,
>-  12,
>-  13,
>-  13,
>-  13,
>-  14,
>-  14,
>-  15,
>-  15,
>-  15,
>-  16,
>-  16,
>-  17,
>-  17,
>-  17,
>-  18,
>-  18,
>-  18,
>-  19,
>-  19,
>-  19,
>-  20,
>-  20,
>-  21,
>-  21,
>-  21,
>-  22,
>-  22,
>-  22,
>-  23,
>-  23,
>-  23,
>-  24,
>-  24,
>-  24,
>-  24,
>-  25,
>-  25,
>-  25,
>-  26,
>-  26,
>-  26,
>-  27,
>-  27,
>-  27,
>-  28,
>-  28,
>-  28,
>-  28,
>-  29,
>-  29,
>-  29,
>-  30,
>-  30,
>-  30,
>-  30,
>-  31,
>-  31,
>-  31,
>-  32,
>-  32,
>-  32,
>-  32,
>-  33,
>-  33,
>-  33,
>-  33,
>-  34,
>-  34,
>-  34,
>-  35,
>-  35,
>-  35,
>-  35,
>-  36,
>-  36,
>-  36,
>-  36,
>-  37,
>-  37,
>-  37,
>-  38,
>-  38,
>-  38,
>-  38,
>-  39,
>-  39,
>-  39,
>-  39,
>-  40,
>-  40,
>-  40,
>-  40,
>-  41,
>-  41,
>-  41,
>-  41,
>-  42,
>-  42,
>-  42,
>-  42,
>-  43,
>-  43,
>-  43,
>-  44,
>-  44,
>-  44,
>-  44,
>-  45,
>-  45,
>-  45,
>-  45,
>-  46,
>-  46,
>-  46,
>-  46,
>-  47,
>-  47,
>-  47,
>-  47,
>-  48,
>-  48,
>-  48,
>-  48,
>-  49,
>-  49,
>-  49,
>-  49,
>-  50,
>-  50,
>-  50,
>-  50,
>-  51,
>-  51,
>-  51,
>-  51,
>-  52,
>-  52,
>-  52,
>-  52,
>-  53,
>-  53,
>-  53,
>-  53,
>-  54,
>-  54,
>-  54,
>-  54,
>-  55,
>-  55,
>-  55,
>-  55,
>-  56,
>-  56,
>-  56,
>-  56,
>-  57,
>-  57,
>-  57,
>-  57,
>-  58,
>-  58,
>-  58,
>-  58,
>-  59,
>-  59,
>-  59,
>-  59,
>-  60,
>-  60,
>-  60,
>-  60,
>-  61,
>-  61,
>-  61,
>-  61,
>-  62,
>-  62,
>-  62,
>-  62,
>-  63,
>-  63,
>-  63,
>-  63,
>-  64
>-};
>+    -56, -54, -52, -50, -48, -47, -45, -43, -42, -40, -38, -37, -35, -34, -33,
>+    -31, -30, -29, -27, -26, -25, -24, -23, -22, -20, -19, -18, -17, -16, -15,
>+    -14, -14, -13, -12, -11, -10, -9,  -8,  -8,  -7,  -6,  -5,  -5,  -4,  -3,
>+    -2,  -2,  -1,  0,   0,   1,   1,   2,   3,   3,   4,   4,   5,   5,   6,
>+    6,   7,   7,   8,   8,   9,   9,   10,  10,  11,  11,  12,  12,  13,  13,
>+    13,  14,  14,  15,  15,  15,  16,  16,  17,  17,  17,  18,  18,  18,  19,
>+    19,  19,  20,  20,  21,  21,  21,  22,  22,  22,  23,  23,  23,  24,  24,
>+    24,  24,  25,  25,  25,  26,  26,  26,  27,  27,  27,  28,  28,  28,  28,
>+    29,  29,  29,  30,  30,  30,  30,  31,  31,  31,  32,  32,  32,  32,  33,
>+    33,  33,  33,  34,  34,  34,  35,  35,  35,  35,  36,  36,  36,  36,  37,
>+    37,  37,  38,  38,  38,  38,  39,  39,  39,  39,  40,  40,  40,  40,  41,
>+    41,  41,  41,  42,  42,  42,  42,  43,  43,  43,  44,  44,  44,  44,  45,
>+    45,  45,  45,  46,  46,  46,  46,  47,  47,  47,  47,  48,  48,  48,  48,
>+    49,  49,  49,  49,  50,  50,  50,  50,  51,  51,  51,  51,  52,  52,  52,
>+    52,  53,  53,  53,  53,  54,  54,  54,  54,  55,  55,  55,  55,  56,  56,
>+    56,  56,  57,  57,  57,  57,  58,  58,  58,  58,  59,  59,  59,  59,  60,
>+    60,  60,  60,  61,  61,  61,  61,  62,  62,  62,  62,  63,  63,  63,  63,
>+    64};
> 
> #endif  // MODULES_AUDIO_PROCESSING_AGC_GAIN_MAP_INTERNAL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/legacy/analog_agc.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/legacy/analog_agc.h
>index 1fed3779ae4..619d95ad0ad 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/legacy/analog_agc.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/legacy/analog_agc.h
>@@ -18,7 +18,6 @@
> 
> #include "modules/audio_processing/agc/legacy/digital_agc.h"
> #include "modules/audio_processing/agc/legacy/gain_control.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> /* Analog Automatic Gain Control variables:
>  * Constant declarations (inner limits inside which no changes are done)
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/legacy/digital_agc.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/legacy/digital_agc.c
>index 3269a17ce4d..d1c30bd2833 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/legacy/digital_agc.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/legacy/digital_agc.c
>@@ -412,8 +412,10 @@ int32_t WebRtcAgc_ProcessDigital(DigitalAgc* stt,
>     }
>     tmp32 = ((uint32_t)cur_level << zeros) & 0x7FFFFFFF;
>     frac = (int16_t)(tmp32 >> 19);  // Q12.
>-    tmp32 = (stt->gainTable[zeros - 1] - stt->gainTable[zeros]) * frac;
>-    gains[k + 1] = stt->gainTable[zeros] + (tmp32 >> 12);
>+    // Interpolate between gainTable[zeros] and gainTable[zeros-1].
>+    tmp32 = ((stt->gainTable[zeros - 1] - stt->gainTable[zeros]) *
>+             (int64_t)frac) >> 12;
>+    gains[k + 1] = stt->gainTable[zeros] + tmp32;
> #ifdef WEBRTC_AGC_DEBUG_DUMP
>     if (k == 0) {
>       fprintf(stt->logFile, "%d\t%d\t%d\t%d\t%d\n", env[0], cur_level,
>@@ -465,9 +467,10 @@ int32_t WebRtcAgc_ProcessDigital(DigitalAgc* stt,
> 
>   // Limit gain to avoid overload distortion
>   for (k = 0; k < 10; k++) {
>-    // To prevent wrap around
>+    // Find a shift of gains[k + 1] such that it can be squared without
>+    // overflow, but at least by 10 bits.
>     zeros = 10;
>-    if (gains[k + 1] > 47453132) {
>+    if (gains[k + 1] > 47452159) {
>       zeros = 16 - WebRtcSpl_NormW32(gains[k + 1]);
>     }
>     gain32 = (gains[k + 1] >> zeros) + 1;
>@@ -502,18 +505,16 @@ int32_t WebRtcAgc_ProcessDigital(DigitalAgc* stt,
>   // iterate over samples
>   for (n = 0; n < L; n++) {
>     for (i = 0; i < num_bands; ++i) {
>-      tmp32 = out[i][n] * ((gain32 + 127) >> 7);
>-      out_tmp = tmp32 >> 16;
>+      out_tmp = (int64_t)out[i][n] * ((gain32 + 127) >> 7) >> 16;
>       if (out_tmp > 4095) {
>         out[i][n] = (int16_t)32767;
>       } else if (out_tmp < -4096) {
>         out[i][n] = (int16_t)-32768;
>       } else {
>-        tmp32 = out[i][n] * (gain32 >> 4);
>-        out[i][n] = (int16_t)(tmp32 >> 16);
>+        tmp32 = ((int64_t)out[i][n] * (gain32 >> 4)) >> 16;
>+        out[i][n] = (int16_t)tmp32;
>       }
>     }
>-    //
> 
>     gain32 += delta;
>   }
>@@ -582,6 +583,7 @@ int16_t WebRtcAgc_ProcessVad(AgcVad* state,      // (i) VAD state
>   int16_t buf2[4];
>   int16_t HPstate;
>   int16_t zeros, dB;
>+  int64_t tmp64;
> 
>   // process in 10 sub frames of 1 ms (to save on memory)
>   nrg = 0;
>@@ -687,17 +689,17 @@ int16_t WebRtcAgc_ProcessVad(AgcVad* state,      // (i) VAD state
>   tmp32 = WebRtcSpl_DivW32W16(tmp32, state->stdLongTerm);
>   tmpU16 = (13 << 12);
>   tmp32b = WEBRTC_SPL_MUL_16_U16(state->logRatio, tmpU16);
>-  tmp32 += tmp32b >> 10;
>-
>-  state->logRatio = (int16_t)(tmp32 >> 6);
>+  tmp64 = tmp32;
>+  tmp64 += tmp32b >> 10;
>+  tmp64 >>= 6;
> 
>   // limit
>-  if (state->logRatio > 2048) {
>-    state->logRatio = 2048;
>-  }
>-  if (state->logRatio < -2048) {
>-    state->logRatio = -2048;
>+  if (tmp64 > 2048) {
>+    tmp64 = 2048;
>+  } else if (tmp64 < -2048) {
>+    tmp64 = -2048;
>   }
>+  state->logRatio = (int16_t)tmp64;
> 
>   return state->logRatio;  // Q10
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/legacy/digital_agc.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/legacy/digital_agc.h
>index af6cf488371..f086294cf25 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/legacy/digital_agc.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/legacy/digital_agc.h
>@@ -15,7 +15,6 @@
> #include <stdio.h>
> #endif
> #include "common_audio/signal_processing/include/signal_processing_library.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> // the 32 most significant bits of A(19) * B(26) >> 13
> #define AGC_MUL32(A, B) (((B) >> 13) * (A) + (((0x00001FFF & (B)) * (A)) >> 13))
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/legacy/gain_control.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/legacy/gain_control.h
>index 05b7ff67976..a0ac96dfa82 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/legacy/gain_control.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/legacy/gain_control.h
>@@ -11,8 +11,6 @@
> #ifndef MODULES_AUDIO_PROCESSING_AGC_LEGACY_GAIN_CONTROL_H_
> #define MODULES_AUDIO_PROCESSING_AGC_LEGACY_GAIN_CONTROL_H_
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>-
> // Errors
> #define AGC_UNSPECIFIED_ERROR 18000
> #define AGC_UNSUPPORTED_FUNCTION_ERROR 18001
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/loudness_histogram.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/loudness_histogram.cc
>index 63d5f7cad04..0ed5850e72d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/loudness_histogram.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/loudness_histogram.cc
>@@ -13,7 +13,6 @@
> #include <cmath>
> #include <cstring>
> 
>-#include "modules/include/module_common_types.h"
> #include "rtc_base/checks.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/loudness_histogram.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/loudness_histogram.h
>index ab45276f060..d8ad7511764 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/loudness_histogram.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/loudness_histogram.h
>@@ -15,8 +15,6 @@
> 
> #include <memory>
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>-
> namespace webrtc {
> 
> // This class implements the histogram of loudness with circular buffers so that
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/mock_agc.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/mock_agc.h
>index 36901af50c2..4297e2a08bd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/mock_agc.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc/mock_agc.h
>@@ -13,16 +13,16 @@
> 
> #include "modules/audio_processing/agc/agc.h"
> 
>-#include "modules/include/module_common_types.h"
> #include "test/gmock.h"
> 
> namespace webrtc {
> 
> class MockAgc : public Agc {
>  public:
>+  virtual ~MockAgc() {}
>   MOCK_METHOD2(AnalyzePreproc, float(const int16_t* audio, size_t length));
>-  MOCK_METHOD3(Process, int(const int16_t* audio, size_t length,
>-                            int sample_rate_hz));
>+  MOCK_METHOD3(Process,
>+               void(const int16_t* audio, size_t length, int sample_rate_hz));
>   MOCK_METHOD1(GetRmsErrorDb, bool(int* error));
>   MOCK_METHOD0(Reset, void());
>   MOCK_METHOD1(set_target_level_dbfs, int(int level));
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/BUILD.gn
>new file mode 100644
>index 00000000000..45ef9685066
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/BUILD.gn
>@@ -0,0 +1,274 @@
>+# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>+#
>+# Use of this source code is governed by a BSD-style license
>+# that can be found in the LICENSE file in the root of the source
>+# tree. An additional intellectual property rights grant can be found
>+# in the file PATENTS.  All contributing project authors may
>+# be found in the AUTHORS file in the root of the source tree.
>+
>+import("../../../webrtc.gni")
>+
>+group("agc2") {
>+  deps = [
>+    ":adaptive_digital",
>+    ":fixed_digital",
>+  ]
>+}
>+
>+rtc_source_set("level_estimation_agc") {
>+  sources = [
>+    "adaptive_mode_level_estimator_agc.cc",
>+    "adaptive_mode_level_estimator_agc.h",
>+  ]
>+  configs += [ "..:apm_debug_dump" ]
>+  deps = [
>+    ":adaptive_digital",
>+    ":common",
>+    ":gain_applier",
>+    ":noise_level_estimator",
>+    ":rnn_vad_with_level",
>+    "..:apm_logging",
>+    "..:audio_frame_view",
>+    "../../../api:array_view",
>+    "../../../common_audio",
>+    "../../../rtc_base:checks",
>+    "../../../rtc_base:rtc_base_approved",
>+    "../../../rtc_base:safe_minmax",
>+    "../agc:level_estimation",
>+    "../vad",
>+  ]
>+}
>+
>+rtc_source_set("adaptive_digital") {
>+  sources = [
>+    "adaptive_agc.cc",
>+    "adaptive_agc.h",
>+    "adaptive_digital_gain_applier.cc",
>+    "adaptive_digital_gain_applier.h",
>+    "adaptive_mode_level_estimator.cc",
>+    "adaptive_mode_level_estimator.h",
>+    "saturation_protector.cc",
>+    "saturation_protector.h",
>+  ]
>+
>+  configs += [ "..:apm_debug_dump" ]
>+
>+  deps = [
>+    ":common",
>+    ":gain_applier",
>+    ":noise_level_estimator",
>+    ":rnn_vad_with_level",
>+    "..:apm_logging",
>+    "..:audio_frame_view",
>+    "../../../api:array_view",
>+    "../../../common_audio",
>+    "../../../rtc_base:checks",
>+    "../../../rtc_base:rtc_base_approved",
>+    "../../../rtc_base:safe_minmax",
>+    "../../../system_wrappers:metrics_api",
>+  ]
>+}
>+
>+rtc_source_set("biquad_filter") {
>+  visibility = [ "./*" ]
>+  sources = [
>+    "biquad_filter.cc",
>+    "biquad_filter.h",
>+  ]
>+  deps = [
>+    "../../../api:array_view",
>+    "../../../rtc_base:rtc_base_approved",
>+  ]
>+}
>+
>+rtc_source_set("common") {
>+  sources = [
>+    "agc2_common.h",
>+  ]
>+  deps = [
>+    "../../../rtc_base:rtc_base_approved",
>+  ]
>+}
>+
>+rtc_source_set("fixed_digital") {
>+  sources = [
>+    "fixed_digital_level_estimator.cc",
>+    "fixed_digital_level_estimator.h",
>+    "fixed_gain_controller.cc",
>+    "fixed_gain_controller.h",
>+    "gain_curve_applier.cc",
>+    "gain_curve_applier.h",
>+    "interpolated_gain_curve.cc",
>+    "interpolated_gain_curve.h",
>+  ]
>+
>+  configs += [ "..:apm_debug_dump" ]
>+
>+  deps = [
>+    ":common",
>+    "..:apm_logging",
>+    "..:audio_frame_view",
>+    "../../../api:array_view",
>+    "../../../common_audio",
>+    "../../../rtc_base:checks",
>+    "../../../rtc_base:gtest_prod",
>+    "../../../rtc_base:rtc_base_approved",
>+    "../../../rtc_base:safe_minmax",
>+    "../../../system_wrappers:metrics_api",
>+  ]
>+}
>+
>+rtc_source_set("gain_applier") {
>+  sources = [
>+    "gain_applier.cc",
>+    "gain_applier.h",
>+  ]
>+  deps = [
>+    ":common",
>+    "..:audio_frame_view",
>+    "../../../rtc_base:safe_minmax",
>+  ]
>+}
>+
>+rtc_source_set("noise_level_estimator") {
>+  sources = [
>+    "down_sampler.cc",
>+    "down_sampler.h",
>+    "noise_level_estimator.cc",
>+    "noise_level_estimator.h",
>+    "noise_spectrum_estimator.cc",
>+    "noise_spectrum_estimator.h",
>+    "signal_classifier.cc",
>+    "signal_classifier.h",
>+  ]
>+  deps = [
>+    ":biquad_filter",
>+    "..:apm_logging",
>+    "..:audio_frame_view",
>+    "../../../api:array_view",
>+    "../../../common_audio",
>+    "../../../rtc_base:checks",
>+    "../../../rtc_base:macromagic",
>+    "../utility:ooura_fft",
>+  ]
>+
>+  configs += [ "..:apm_debug_dump" ]
>+}
>+
>+rtc_source_set("rnn_vad_with_level") {
>+  sources = [
>+    "vad_with_level.cc",
>+    "vad_with_level.h",
>+  ]
>+  deps = [
>+    "..:audio_frame_view",
>+    "../../../api:array_view",
>+    "../../../common_audio",
>+    "../../../rtc_base:checks",
>+    "rnn_vad",
>+  ]
>+}
>+
>+rtc_source_set("adaptive_digital_unittests") {
>+  testonly = true
>+  configs += [ "..:apm_debug_dump" ]
>+
>+  sources = [
>+    "adaptive_digital_gain_applier_unittest.cc",
>+    "adaptive_mode_level_estimator_unittest.cc",
>+    "gain_applier_unittest.cc",
>+    "saturation_protector_unittest.cc",
>+  ]
>+  deps = [
>+    ":adaptive_digital",
>+    ":common",
>+    ":gain_applier",
>+    ":test_utils",
>+    "..:apm_logging",
>+    "..:audio_frame_view",
>+    "../../../api:array_view",
>+    "../../../common_audio",
>+    "../../../rtc_base:checks",
>+    "../../../rtc_base:rtc_base_approved",
>+    "../../../rtc_base:rtc_base_tests_utils",
>+  ]
>+}
>+
>+rtc_source_set("biquad_filter_unittests") {
>+  testonly = true
>+  sources = [
>+    "biquad_filter_unittest.cc",
>+  ]
>+  deps = [
>+    ":biquad_filter",
>+    "../../../rtc_base:rtc_base_tests_utils",
>+  ]
>+}
>+
>+rtc_source_set("fixed_digital_unittests") {
>+  testonly = true
>+  configs += [ "..:apm_debug_dump" ]
>+
>+  sources = [
>+    "agc2_testing_common_unittest.cc",
>+    "compute_interpolated_gain_curve.cc",
>+    "compute_interpolated_gain_curve.h",
>+    "fixed_digital_level_estimator_unittest.cc",
>+    "fixed_gain_controller_unittest.cc",
>+    "gain_curve_applier_unittest.cc",
>+    "interpolated_gain_curve_unittest.cc",
>+    "limiter.cc",
>+    "limiter.h",
>+    "limiter_unittest.cc",
>+  ]
>+  deps = [
>+    ":common",
>+    ":fixed_digital",
>+    ":test_utils",
>+    "..:apm_logging",
>+    "..:audio_frame_view",
>+    "../../../api:array_view",
>+    "../../../common_audio",
>+    "../../../rtc_base:checks",
>+    "../../../rtc_base:rtc_base_approved",
>+    "../../../rtc_base:rtc_base_tests_utils",
>+    "../../../system_wrappers:metrics_default",
>+    "//third_party/abseil-cpp/absl/memory",
>+  ]
>+}
>+
>+rtc_source_set("noise_estimator_unittests") {
>+  testonly = true
>+  configs += [ "..:apm_debug_dump" ]
>+
>+  sources = [
>+    "noise_level_estimator_unittest.cc",
>+    "signal_classifier_unittest.cc",
>+  ]
>+  deps = [
>+    ":noise_level_estimator",
>+    ":test_utils",
>+    "..:apm_logging",
>+    "..:audio_frame_view",
>+    "../../../api:array_view",
>+    "../../../rtc_base:checks",
>+    "../../../rtc_base:rtc_base_approved",
>+    "../../../rtc_base:rtc_base_tests_utils",
>+  ]
>+}
>+
>+rtc_source_set("test_utils") {
>+  testonly = true
>+  visibility = [ ":*" ]
>+  sources = [
>+    "agc2_testing_common.cc",
>+    "agc2_testing_common.h",
>+    "vector_float_frame.cc",
>+    "vector_float_frame.h",
>+  ]
>+  deps = [
>+    "..:audio_frame_view",
>+    "../../../rtc_base:checks",
>+    "../../../rtc_base:rtc_base_approved",
>+  ]
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_agc.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_agc.cc
>new file mode 100644
>index 00000000000..805be0cf7d1
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_agc.cc
>@@ -0,0 +1,57 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/agc2/adaptive_agc.h"
>+
>+#include <algorithm>
>+#include <numeric>
>+
>+#include "common_audio/include/audio_util.h"
>+#include "modules/audio_processing/agc2/vad_with_level.h"
>+#include "modules/audio_processing/logging/apm_data_dumper.h"
>+
>+namespace webrtc {
>+
>+AdaptiveAgc::AdaptiveAgc(ApmDataDumper* apm_data_dumper)
>+    : speech_level_estimator_(apm_data_dumper),
>+      gain_applier_(apm_data_dumper),
>+      apm_data_dumper_(apm_data_dumper),
>+      noise_level_estimator_(apm_data_dumper) {
>+  RTC_DCHECK(apm_data_dumper);
>+}
>+
>+AdaptiveAgc::~AdaptiveAgc() = default;
>+
>+void AdaptiveAgc::Process(AudioFrameView<float> float_frame) {
>+  const VadWithLevel::LevelAndProbability vad_result =
>+      vad_.AnalyzeFrame(float_frame);
>+  apm_data_dumper_->DumpRaw("agc2_vad_probability",
>+                            vad_result.speech_probability);
>+  apm_data_dumper_->DumpRaw("agc2_vad_rms_dbfs", vad_result.speech_rms_dbfs);
>+
>+  apm_data_dumper_->DumpRaw("agc2_vad_peak_dbfs", vad_result.speech_peak_dbfs);
>+  speech_level_estimator_.UpdateEstimation(vad_result);
>+
>+  const float speech_level_dbfs = speech_level_estimator_.LatestLevelEstimate();
>+
>+  const float noise_level_dbfs = noise_level_estimator_.Analyze(float_frame);
>+
>+  apm_data_dumper_->DumpRaw("agc2_noise_estimate_dbfs", noise_level_dbfs);
>+
>+  // The gain applier applies the gain.
>+  gain_applier_.Process(speech_level_dbfs, noise_level_dbfs, vad_result,
>+                        float_frame);
>+}
>+
>+void AdaptiveAgc::Reset() {
>+  speech_level_estimator_.Reset();
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_agc.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_agc.h
>new file mode 100644
>index 00000000000..8f5efec79b4
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_agc.h
>@@ -0,0 +1,43 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_AGC_H_
>+#define MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_AGC_H_
>+
>+#include <memory>
>+
>+#include "modules/audio_processing/agc2/adaptive_digital_gain_applier.h"
>+#include "modules/audio_processing/agc2/adaptive_mode_level_estimator.h"
>+#include "modules/audio_processing/agc2/noise_level_estimator.h"
>+#include "modules/audio_processing/agc2/vad_with_level.h"
>+#include "modules/audio_processing/include/audio_frame_view.h"
>+
>+namespace webrtc {
>+class ApmDataDumper;
>+
>+class AdaptiveAgc {
>+ public:
>+  explicit AdaptiveAgc(ApmDataDumper* apm_data_dumper);
>+  ~AdaptiveAgc();
>+
>+  void Process(AudioFrameView<float> float_frame);
>+  void Reset();
>+
>+ private:
>+  AdaptiveModeLevelEstimator speech_level_estimator_;
>+  VadWithLevel vad_;
>+  AdaptiveDigitalGainApplier gain_applier_;
>+  ApmDataDumper* const apm_data_dumper_;
>+  NoiseLevelEstimator noise_level_estimator_;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_AGC_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_digital_gain_applier.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_digital_gain_applier.cc
>new file mode 100644
>index 00000000000..f5342df8297
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_digital_gain_applier.cc
>@@ -0,0 +1,123 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/agc2/adaptive_digital_gain_applier.h"
>+
>+#include <algorithm>
>+
>+#include "common_audio/include/audio_util.h"
>+#include "modules/audio_processing/agc2/agc2_common.h"
>+#include "modules/audio_processing/logging/apm_data_dumper.h"
>+#include "rtc_base/numerics/safe_minmax.h"
>+#include "system_wrappers/include/metrics.h"
>+
>+namespace webrtc {
>+namespace {
>+
>+// This function maps input level to desired applied gain. We want to
>+// boost the signal so that peaks are at -kHeadroomDbfs. We can't
>+// apply more than kMaxGainDb gain.
>+float ComputeGainDb(float input_level_dbfs) {
>+  // If the level is very low, boost it as much as we can.
>+  if (input_level_dbfs < -(kHeadroomDbfs + kMaxGainDb)) {
>+    return kMaxGainDb;
>+  }
>+
>+  // We expect to end up here most of the time: the level is below
>+  // -headroom, but we can boost it to -headroom.
>+  if (input_level_dbfs < -kHeadroomDbfs) {
>+    return -kHeadroomDbfs - input_level_dbfs;
>+  }
>+
>+  // Otherwise, the level is too high and we can't boost. The
>+  // LevelEstimator is responsible for not reporting bogus gain
>+  // values.
>+  RTC_DCHECK_LE(input_level_dbfs, 0.f);
>+  return 0.f;
>+}
>+
>+// We require 'gain + noise_level <= kMaxNoiseLevelDbfs'.
>+float LimitGainByNoise(float target_gain,
>+                       float input_noise_level_dbfs,
>+                       ApmDataDumper* apm_data_dumper) {
>+  const float noise_headroom_db = kMaxNoiseLevelDbfs - input_noise_level_dbfs;
>+  apm_data_dumper->DumpRaw("agc2_noise_headroom_db", noise_headroom_db);
>+  return std::min(target_gain, std::max(noise_headroom_db, 0.f));
>+}
>+
>+// Computes how the gain should change during this frame.
>+// Return the gain difference in db to 'last_gain_db'.
>+float ComputeGainChangeThisFrameDb(float target_gain_db,
>+                                   float last_gain_db,
>+                                   bool gain_increase_allowed) {
>+  float target_gain_difference_db = target_gain_db - last_gain_db;
>+  if (!gain_increase_allowed) {
>+    target_gain_difference_db = std::min(target_gain_difference_db, 0.f);
>+  }
>+
>+  return rtc::SafeClamp(target_gain_difference_db, -kMaxGainChangePerFrameDb,
>+                        kMaxGainChangePerFrameDb);
>+}
>+}  // namespace
>+
>+AdaptiveDigitalGainApplier::AdaptiveDigitalGainApplier(
>+    ApmDataDumper* apm_data_dumper)
>+    : gain_applier_(false, DbToRatio(last_gain_db_)),
>+      apm_data_dumper_(apm_data_dumper) {}
>+
>+void AdaptiveDigitalGainApplier::Process(
>+    float input_level_dbfs,
>+    float input_noise_level_dbfs,
>+    const VadWithLevel::LevelAndProbability vad_result,
>+    AudioFrameView<float> float_frame) {
>+  calls_since_last_gain_log_++;
>+  if (calls_since_last_gain_log_ == 100) {
>+    calls_since_last_gain_log_ = 0;
>+    RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.Agc2.DigitalGainApplied",
>+                                last_gain_db_, 0, kMaxGainDb, kMaxGainDb + 1);
>+    RTC_HISTOGRAM_COUNTS_LINEAR("WebRTC.Audio.Agc2.EstimatedNoiseLevel",
>+                                input_noise_level_dbfs, 0, 100, 101);
>+  }
>+
>+  input_level_dbfs = std::min(input_level_dbfs, 0.f);
>+
>+  RTC_DCHECK_GE(input_level_dbfs, -150.f);
>+  RTC_DCHECK_GE(float_frame.num_channels(), 1);
>+  RTC_DCHECK_GE(float_frame.samples_per_channel(), 1);
>+
>+  const float target_gain_db =
>+      LimitGainByNoise(ComputeGainDb(input_level_dbfs), input_noise_level_dbfs,
>+                       apm_data_dumper_);
>+
>+  // Forbid increasing the gain when there is no speech.
>+  gain_increase_allowed_ =
>+      vad_result.speech_probability > kVadConfidenceThreshold;
>+
>+  const float gain_change_this_frame_db = ComputeGainChangeThisFrameDb(
>+      target_gain_db, last_gain_db_, gain_increase_allowed_);
>+
>+  apm_data_dumper_->DumpRaw("agc2_want_to_change_by_db",
>+                            target_gain_db - last_gain_db_);
>+  apm_data_dumper_->DumpRaw("agc2_will_change_by_db",
>+                            gain_change_this_frame_db);
>+
>+  // Optimization: avoid calling math functions if gain does not
>+  // change.
>+  if (gain_change_this_frame_db != 0.f) {
>+    gain_applier_.SetGainFactor(
>+        DbToRatio(last_gain_db_ + gain_change_this_frame_db));
>+  }
>+  gain_applier_.ApplyGain(float_frame);
>+
>+  // Remember that the gain has changed for the next iteration.
>+  last_gain_db_ = last_gain_db_ + gain_change_this_frame_db;
>+  apm_data_dumper_->DumpRaw("agc2_applied_gain_db", last_gain_db_);
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_digital_gain_applier.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_digital_gain_applier.h
>new file mode 100644
>index 00000000000..a3a1ff57f5a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_digital_gain_applier.h
>@@ -0,0 +1,46 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_DIGITAL_GAIN_APPLIER_H_
>+#define MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_DIGITAL_GAIN_APPLIER_H_
>+
>+#include "modules/audio_processing/agc2/agc2_common.h"
>+#include "modules/audio_processing/agc2/gain_applier.h"
>+#include "modules/audio_processing/agc2/vad_with_level.h"
>+#include "modules/audio_processing/include/audio_frame_view.h"
>+
>+namespace webrtc {
>+
>+class ApmDataDumper;
>+
>+class AdaptiveDigitalGainApplier {
>+ public:
>+  explicit AdaptiveDigitalGainApplier(ApmDataDumper* apm_data_dumper);
>+  // Decide what gain to apply.
>+  void Process(float input_level_dbfs,
>+               float input_noise_level_dbfs,
>+               const VadWithLevel::LevelAndProbability vad_result,
>+               AudioFrameView<float> float_frame);
>+
>+ private:
>+  float last_gain_db_ = kInitialAdaptiveDigitalGainDb;
>+  GainApplier gain_applier_;
>+  int calls_since_last_gain_log_ = 0;
>+
>+  // For some combinations of noise and speech probability, increasing
>+  // the level is not allowed. Since we may get VAD results in bursts,
>+  // we keep track of this variable until the next VAD results come
>+  // in.
>+  bool gain_increase_allowed_ = true;
>+  ApmDataDumper* apm_data_dumper_ = nullptr;
>+};
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_DIGITAL_GAIN_APPLIER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_digital_gain_applier_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_digital_gain_applier_unittest.cc
>new file mode 100644
>index 00000000000..ea9e5c74dd3
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_digital_gain_applier_unittest.cc
>@@ -0,0 +1,188 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/agc2/adaptive_digital_gain_applier.h"
>+
>+#include <algorithm>
>+
>+#include "common_audio/include/audio_util.h"
>+#include "modules/audio_processing/agc2/agc2_common.h"
>+#include "modules/audio_processing/agc2/vector_float_frame.h"
>+#include "modules/audio_processing/logging/apm_data_dumper.h"
>+#include "rtc_base/gunit.h"
>+
>+namespace webrtc {
>+namespace {
>+// Constants used in place of estimated noise levels.
>+constexpr float kNoNoiseDbfs = -90.f;
>+constexpr float kWithNoiseDbfs = -20.f;
>+
>+// Runs gain applier and returns the applied gain in linear scale.
>+float RunOnConstantLevel(int num_iterations,
>+                         VadWithLevel::LevelAndProbability vad_data,
>+                         float input_level_dbfs,
>+                         AdaptiveDigitalGainApplier* gain_applier) {
>+  float gain_linear = 0.f;
>+
>+  for (int i = 0; i < num_iterations; ++i) {
>+    VectorFloatFrame fake_audio(1, 1, 1.f);
>+    gain_applier->Process(input_level_dbfs, kNoNoiseDbfs, vad_data,
>+                          fake_audio.float_frame_view());
>+    gain_linear = fake_audio.float_frame_view().channel(0)[0];
>+  }
>+  return gain_linear;
>+}
>+
>+constexpr VadWithLevel::LevelAndProbability kVadSpeech(1.f, -20.f, 0.f);
>+}  // namespace
>+
>+TEST(AutomaticGainController2AdaptiveGainApplier, GainApplierShouldNotCrash) {
>+  static_assert(
>+      std::is_trivially_destructible<VadWithLevel::LevelAndProbability>::value,
>+      "");
>+  ApmDataDumper apm_data_dumper(0);
>+  AdaptiveDigitalGainApplier gain_applier(&apm_data_dumper);
>+
>+  // Make one call with reasonable audio level values and settings.
>+  VectorFloatFrame fake_audio(2, 480, 10000.f);
>+  gain_applier.Process(-5.0, kNoNoiseDbfs, kVadSpeech,
>+                       fake_audio.float_frame_view());
>+}
>+
>+// Check that the output is -kHeadroom dBFS.
>+TEST(AutomaticGainController2AdaptiveGainApplier, TargetLevelIsReached) {
>+  ApmDataDumper apm_data_dumper(0);
>+  AdaptiveDigitalGainApplier gain_applier(&apm_data_dumper);
>+
>+  constexpr float initial_level_dbfs = -5.f;
>+
>+  const float applied_gain =
>+      RunOnConstantLevel(200, kVadSpeech, initial_level_dbfs, &gain_applier);
>+
>+  EXPECT_NEAR(applied_gain, DbToRatio(-kHeadroomDbfs - initial_level_dbfs),
>+              0.1f);
>+}
>+
>+// Check that the output is -kHeadroom dBFS
>+TEST(AutomaticGainController2AdaptiveGainApplier, GainApproachesMaxGain) {
>+  ApmDataDumper apm_data_dumper(0);
>+  AdaptiveDigitalGainApplier gain_applier(&apm_data_dumper);
>+
>+  constexpr float initial_level_dbfs = -kHeadroomDbfs - kMaxGainDb - 10.f;
>+  // A few extra frames for safety.
>+  constexpr int kNumFramesToAdapt =
>+      static_cast<int>(kMaxGainDb / kMaxGainChangePerFrameDb) + 10;
>+
>+  const float applied_gain = RunOnConstantLevel(
>+      kNumFramesToAdapt, kVadSpeech, initial_level_dbfs, &gain_applier);
>+  EXPECT_NEAR(applied_gain, DbToRatio(kMaxGainDb), 0.1f);
>+
>+  const float applied_gain_db = 20.f * std::log10(applied_gain);
>+  EXPECT_NEAR(applied_gain_db, kMaxGainDb, 0.1f);
>+}
>+
>+TEST(AutomaticGainController2AdaptiveGainApplier, GainDoesNotChangeFast) {
>+  ApmDataDumper apm_data_dumper(0);
>+  AdaptiveDigitalGainApplier gain_applier(&apm_data_dumper);
>+
>+  constexpr float initial_level_dbfs = -25.f;
>+  // A few extra frames for safety.
>+  constexpr int kNumFramesToAdapt =
>+      static_cast<int>(initial_level_dbfs / kMaxGainChangePerFrameDb) + 10;
>+
>+  const float kMaxChangePerFrameLinear = DbToRatio(kMaxGainChangePerFrameDb);
>+
>+  float last_gain_linear = 1.f;
>+  for (int i = 0; i < kNumFramesToAdapt; ++i) {
>+    SCOPED_TRACE(i);
>+    VectorFloatFrame fake_audio(1, 1, 1.f);
>+    gain_applier.Process(initial_level_dbfs, kNoNoiseDbfs, kVadSpeech,
>+                         fake_audio.float_frame_view());
>+    float current_gain_linear = fake_audio.float_frame_view().channel(0)[0];
>+    EXPECT_LE(std::abs(current_gain_linear - last_gain_linear),
>+              kMaxChangePerFrameLinear);
>+    last_gain_linear = current_gain_linear;
>+  }
>+
>+  // Check that the same is true when gain decreases as well.
>+  for (int i = 0; i < kNumFramesToAdapt; ++i) {
>+    SCOPED_TRACE(i);
>+    VectorFloatFrame fake_audio(1, 1, 1.f);
>+    gain_applier.Process(0.f, kNoNoiseDbfs, kVadSpeech,
>+                         fake_audio.float_frame_view());
>+    float current_gain_linear = fake_audio.float_frame_view().channel(0)[0];
>+    EXPECT_LE(std::abs(current_gain_linear - last_gain_linear),
>+              kMaxChangePerFrameLinear);
>+    last_gain_linear = current_gain_linear;
>+  }
>+}
>+
>+TEST(AutomaticGainController2AdaptiveGainApplier, GainIsRampedInAFrame) {
>+  ApmDataDumper apm_data_dumper(0);
>+  AdaptiveDigitalGainApplier gain_applier(&apm_data_dumper);
>+
>+  constexpr float initial_level_dbfs = -25.f;
>+  constexpr int num_samples = 480;
>+
>+  VectorFloatFrame fake_audio(1, num_samples, 1.f);
>+  gain_applier.Process(initial_level_dbfs, kNoNoiseDbfs, kVadSpeech,
>+                       fake_audio.float_frame_view());
>+  float maximal_difference = 0.f;
>+  float current_value = 1.f * DbToRatio(kInitialAdaptiveDigitalGainDb);
>+  for (const auto& x : fake_audio.float_frame_view().channel(0)) {
>+    const float difference = std::abs(x - current_value);
>+    maximal_difference = std::max(maximal_difference, difference);
>+    current_value = x;
>+  }
>+
>+  const float kMaxChangePerFrameLinear = DbToRatio(kMaxGainChangePerFrameDb);
>+  const float kMaxChangePerSample = kMaxChangePerFrameLinear / num_samples;
>+
>+  EXPECT_LE(maximal_difference, kMaxChangePerSample);
>+}
>+
>+TEST(AutomaticGainController2AdaptiveGainApplier, NoiseLimitsGain) {
>+  ApmDataDumper apm_data_dumper(0);
>+  AdaptiveDigitalGainApplier gain_applier(&apm_data_dumper);
>+
>+  constexpr float initial_level_dbfs = -25.f;
>+  constexpr int num_samples = 480;
>+  constexpr int num_initial_frames =
>+      kInitialAdaptiveDigitalGainDb / kMaxGainChangePerFrameDb;
>+  constexpr int num_frames = 50;
>+
>+  ASSERT_GT(kWithNoiseDbfs, kMaxNoiseLevelDbfs) << "kWithNoiseDbfs is too low";
>+
>+  for (int i = 0; i < num_initial_frames + num_frames; ++i) {
>+    VectorFloatFrame fake_audio(1, num_samples, 1.f);
>+    gain_applier.Process(initial_level_dbfs, kWithNoiseDbfs, kVadSpeech,
>+                         fake_audio.float_frame_view());
>+
>+    // Wait so that the adaptive gain applier has time to lower the gain.
>+    if (i > num_initial_frames) {
>+      const float maximal_ratio =
>+          *std::max_element(fake_audio.float_frame_view().channel(0).begin(),
>+                            fake_audio.float_frame_view().channel(0).end());
>+
>+      EXPECT_NEAR(maximal_ratio, 1.f, 0.001f);
>+    }
>+  }
>+}
>+
>+TEST(AutomaticGainController2GainApplier, CanHandlePositiveSpeechLevels) {
>+  ApmDataDumper apm_data_dumper(0);
>+  AdaptiveDigitalGainApplier gain_applier(&apm_data_dumper);
>+
>+  // Make one call with positive audio level values and settings.
>+  VectorFloatFrame fake_audio(2, 480, 10000.f);
>+  gain_applier.Process(5.0f, kNoNoiseDbfs, kVadSpeech,
>+                       fake_audio.float_frame_view());
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_mode_level_estimator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_mode_level_estimator.cc
>new file mode 100644
>index 00000000000..b670f4b4692
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_mode_level_estimator.cc
>@@ -0,0 +1,77 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/agc2/adaptive_mode_level_estimator.h"
>+
>+#include "modules/audio_processing/agc2/agc2_common.h"
>+#include "modules/audio_processing/logging/apm_data_dumper.h"
>+#include "rtc_base/numerics/safe_minmax.h"
>+
>+namespace webrtc {
>+
>+AdaptiveModeLevelEstimator::AdaptiveModeLevelEstimator(
>+    ApmDataDumper* apm_data_dumper)
>+    : saturation_protector_(apm_data_dumper),
>+      apm_data_dumper_(apm_data_dumper) {}
>+
>+void AdaptiveModeLevelEstimator::UpdateEstimation(
>+    const VadWithLevel::LevelAndProbability& vad_data) {
>+  RTC_DCHECK_GT(vad_data.speech_rms_dbfs, -150.f);
>+  RTC_DCHECK_LT(vad_data.speech_rms_dbfs, 50.f);
>+  RTC_DCHECK_GT(vad_data.speech_peak_dbfs, -150.f);
>+  RTC_DCHECK_LT(vad_data.speech_peak_dbfs, 50.f);
>+  RTC_DCHECK_GE(vad_data.speech_probability, 0.f);
>+  RTC_DCHECK_LE(vad_data.speech_probability, 1.f);
>+
>+  if (vad_data.speech_probability < kVadConfidenceThreshold) {
>+    DebugDumpEstimate();
>+    return;
>+  }
>+
>+  const bool buffer_is_full = buffer_size_ms_ >= kFullBufferSizeMs;
>+  if (!buffer_is_full) {
>+    buffer_size_ms_ += kFrameDurationMs;
>+  }
>+
>+  const float leak_factor = buffer_is_full ? kFullBufferLeakFactor : 1.f;
>+
>+  estimate_numerator_ = estimate_numerator_ * leak_factor +
>+                        vad_data.speech_rms_dbfs * vad_data.speech_probability;
>+  estimate_denominator_ =
>+      estimate_denominator_ * leak_factor + vad_data.speech_probability;
>+
>+  last_estimate_with_offset_dbfs_ = estimate_numerator_ / estimate_denominator_;
>+
>+  saturation_protector_.UpdateMargin(vad_data, last_estimate_with_offset_dbfs_);
>+  DebugDumpEstimate();
>+}
>+
>+float AdaptiveModeLevelEstimator::LatestLevelEstimate() const {
>+  return rtc::SafeClamp<float>(
>+      last_estimate_with_offset_dbfs_ + saturation_protector_.LastMargin(),
>+      -90.f, 30.f);
>+}
>+
>+void AdaptiveModeLevelEstimator::Reset() {
>+  buffer_size_ms_ = 0;
>+  last_estimate_with_offset_dbfs_ = kInitialSpeechLevelEstimateDbfs;
>+  estimate_numerator_ = 0.f;
>+  estimate_denominator_ = 0.f;
>+  saturation_protector_.Reset();
>+}
>+
>+void AdaptiveModeLevelEstimator::DebugDumpEstimate() {
>+  apm_data_dumper_->DumpRaw("agc2_adaptive_level_estimate_with_offset_dbfs",
>+                            last_estimate_with_offset_dbfs_);
>+  apm_data_dumper_->DumpRaw("agc2_adaptive_level_estimate_dbfs",
>+                            LatestLevelEstimate());
>+  saturation_protector_.DebugDumpEstimate();
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_mode_level_estimator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_mode_level_estimator.h
>new file mode 100644
>index 00000000000..5ca7c550be5
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_mode_level_estimator.h
>@@ -0,0 +1,40 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_MODE_LEVEL_ESTIMATOR_H_
>+#define MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_MODE_LEVEL_ESTIMATOR_H_
>+
>+#include "modules/audio_processing/agc2/saturation_protector.h"
>+#include "modules/audio_processing/agc2/vad_with_level.h"
>+
>+namespace webrtc {
>+class ApmDataDumper;
>+
>+class AdaptiveModeLevelEstimator {
>+ public:
>+  explicit AdaptiveModeLevelEstimator(ApmDataDumper* apm_data_dumper);
>+  void UpdateEstimation(const VadWithLevel::LevelAndProbability& vad_data);
>+  float LatestLevelEstimate() const;
>+  void Reset();
>+
>+ private:
>+  void DebugDumpEstimate();
>+
>+  size_t buffer_size_ms_ = 0;
>+  float last_estimate_with_offset_dbfs_ = kInitialSpeechLevelEstimateDbfs;
>+  float estimate_numerator_ = 0.f;
>+  float estimate_denominator_ = 0.f;
>+  SaturationProtector saturation_protector_;
>+  ApmDataDumper* const apm_data_dumper_;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_MODE_LEVEL_ESTIMATOR_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_mode_level_estimator_agc.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_mode_level_estimator_agc.cc
>new file mode 100644
>index 00000000000..b0922be677c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_mode_level_estimator_agc.cc
>@@ -0,0 +1,61 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/agc2/adaptive_mode_level_estimator_agc.h"
>+
>+#include "modules/audio_processing/include/audio_frame_view.h"
>+
>+namespace webrtc {
>+
>+AdaptiveModeLevelEstimatorAgc::AdaptiveModeLevelEstimatorAgc(
>+    ApmDataDumper* apm_data_dumper)
>+    : level_estimator_(apm_data_dumper) {
>+  set_target_level_dbfs(kDefaultLevelDbfs);
>+}
>+
>+// |audio| must be mono; in a multi-channel stream, provide the first (usually
>+// left) channel.
>+void AdaptiveModeLevelEstimatorAgc::Process(const int16_t* audio,
>+                                            size_t length,
>+                                            int sample_rate_hz) {
>+  std::vector<float> float_audio_frame(audio, audio + length);
>+  const float* const first_channel = &float_audio_frame[0];
>+  AudioFrameView<const float> frame_view(&first_channel, 1 /* num channels */,
>+                                         length);
>+  const auto vad_prob = agc2_vad_.AnalyzeFrame(frame_view);
>+  latest_voice_probability_ = vad_prob.speech_probability;
>+  if (latest_voice_probability_ > kVadConfidenceThreshold) {
>+    time_in_ms_since_last_estimate_ += kFrameDurationMs;
>+  }
>+  level_estimator_.UpdateEstimation(vad_prob);
>+}
>+
>+// Retrieves the difference between the target RMS level and the current
>+// signal RMS level in dB. Returns true if an update is available and false
>+// otherwise, in which case |error| should be ignored and no action taken.
>+bool AdaptiveModeLevelEstimatorAgc::GetRmsErrorDb(int* error) {
>+  if (time_in_ms_since_last_estimate_ <= kTimeUntilConfidentMs) {
>+    return false;
>+  }
>+  *error = std::floor(target_level_dbfs() -
>+                      level_estimator_.LatestLevelEstimate() + 0.5f);
>+  time_in_ms_since_last_estimate_ = 0;
>+  return true;
>+}
>+
>+void AdaptiveModeLevelEstimatorAgc::Reset() {
>+  level_estimator_.Reset();
>+}
>+
>+float AdaptiveModeLevelEstimatorAgc::voice_probability() const {
>+  return latest_voice_probability_;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_mode_level_estimator_agc.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_mode_level_estimator_agc.h
>new file mode 100644
>index 00000000000..df01dd9d27d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_mode_level_estimator_agc.h
>@@ -0,0 +1,49 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_MODE_LEVEL_ESTIMATOR_AGC_H_
>+#define MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_MODE_LEVEL_ESTIMATOR_AGC_H_
>+
>+#include <vector>
>+
>+#include "modules/audio_processing/agc/agc.h"
>+#include "modules/audio_processing/agc2/adaptive_mode_level_estimator.h"
>+#include "modules/audio_processing/agc2/vad_with_level.h"
>+
>+namespace webrtc {
>+class AdaptiveModeLevelEstimatorAgc : public Agc {
>+ public:
>+  explicit AdaptiveModeLevelEstimatorAgc(ApmDataDumper* apm_data_dumper);
>+
>+  // |audio| must be mono; in a multi-channel stream, provide the first (usually
>+  // left) channel.
>+  void Process(const int16_t* audio,
>+               size_t length,
>+               int sample_rate_hz) override;
>+
>+  // Retrieves the difference between the target RMS level and the current
>+  // signal RMS level in dB. Returns true if an update is available and false
>+  // otherwise, in which case |error| should be ignored and no action taken.
>+  bool GetRmsErrorDb(int* error) override;
>+  void Reset() override;
>+
>+  float voice_probability() const override;
>+
>+ private:
>+  static constexpr int kTimeUntilConfidentMs = 700;
>+  static constexpr int kDefaultLevelDbfs = 0;
>+  int32_t time_in_ms_since_last_estimate_ = 0;
>+  AdaptiveModeLevelEstimator level_estimator_;
>+  VadWithLevel agc2_vad_;
>+  float latest_voice_probability_ = 0.f;
>+};
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_MODE_LEVEL_ESTIMATOR_AGC_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_mode_level_estimator_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_mode_level_estimator_unittest.cc
>new file mode 100644
>index 00000000000..1915ce23a4f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/adaptive_mode_level_estimator_unittest.cc
>@@ -0,0 +1,149 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/agc2/adaptive_mode_level_estimator.h"
>+
>+#include "modules/audio_processing/agc2/agc2_common.h"
>+#include "modules/audio_processing/logging/apm_data_dumper.h"
>+#include "rtc_base/gunit.h"
>+
>+namespace webrtc {
>+namespace {
>+void RunOnConstantLevel(int num_iterations,
>+                        VadWithLevel::LevelAndProbability vad_data,
>+                        AdaptiveModeLevelEstimator* level_estimator) {
>+  for (int i = 0; i < num_iterations; ++i) {
>+    level_estimator->UpdateEstimation(vad_data);  // By copy
>+  }
>+}
>+}  // namespace
>+
>+TEST(AutomaticGainController2AdaptiveModeLevelEstimator,
>+     EstimatorShouldNotCrash) {
>+  ApmDataDumper apm_data_dumper(0);
>+  AdaptiveModeLevelEstimator level_estimator(&apm_data_dumper);
>+
>+  VadWithLevel::LevelAndProbability vad_data(1.f, -20.f, -10.f);
>+  level_estimator.UpdateEstimation(vad_data);
>+  static_cast<void>(level_estimator.LatestLevelEstimate());
>+}
>+
>+TEST(AutomaticGainController2AdaptiveModeLevelEstimator, LevelShouldStabilize) {
>+  ApmDataDumper apm_data_dumper(0);
>+  AdaptiveModeLevelEstimator level_estimator(&apm_data_dumper);
>+
>+  constexpr float kSpeechRmsDbfs = -15.f;
>+  RunOnConstantLevel(
>+      100,
>+      VadWithLevel::LevelAndProbability(
>+          1.f, kSpeechRmsDbfs - kInitialSaturationMarginDb, kSpeechRmsDbfs),
>+      &level_estimator);
>+
>+  EXPECT_NEAR(level_estimator.LatestLevelEstimate(), kSpeechRmsDbfs, 0.1f);
>+}
>+
>+TEST(AutomaticGainController2AdaptiveModeLevelEstimator,
>+     EstimatorIgnoresZeroProbabilityFrames) {
>+  ApmDataDumper apm_data_dumper(0);
>+  AdaptiveModeLevelEstimator level_estimator(&apm_data_dumper);
>+
>+  // Run for one second of fake audio.
>+  constexpr float kSpeechRmsDbfs = -25.f;
>+  RunOnConstantLevel(
>+      100,
>+      VadWithLevel::LevelAndProbability(
>+          1.f, kSpeechRmsDbfs - kInitialSaturationMarginDb, kSpeechRmsDbfs),
>+      &level_estimator);
>+
>+  // Run for one more second, but mark as not speech.
>+  constexpr float kNoiseRmsDbfs = 0.f;
>+  RunOnConstantLevel(
>+      100, VadWithLevel::LevelAndProbability(0.f, kNoiseRmsDbfs, kNoiseRmsDbfs),
>+      &level_estimator);
>+
>+  // Level should not have changed.
>+  EXPECT_NEAR(level_estimator.LatestLevelEstimate(), kSpeechRmsDbfs, 0.1f);
>+}
>+
>+TEST(AutomaticGainController2AdaptiveModeLevelEstimator, TimeToAdapt) {
>+  ApmDataDumper apm_data_dumper(0);
>+  AdaptiveModeLevelEstimator level_estimator(&apm_data_dumper);
>+
>+  // Run for one 'window size' interval.
>+  constexpr float kInitialSpeechRmsDbfs = -30.f;
>+  RunOnConstantLevel(
>+      kFullBufferSizeMs / kFrameDurationMs,
>+      VadWithLevel::LevelAndProbability(
>+          1.f, kInitialSpeechRmsDbfs - kInitialSaturationMarginDb,
>+          kInitialSpeechRmsDbfs),
>+      &level_estimator);
>+
>+  // Run for one half 'window size' interval. This should not be enough to
>+  // adapt.
>+  constexpr float kDifferentSpeechRmsDbfs = -10.f;
>+  // It should at most differ by 25% after one half 'window size' interval.
>+  const float kMaxDifferenceDb =
>+      0.25 * std::abs(kDifferentSpeechRmsDbfs - kInitialSpeechRmsDbfs);
>+  RunOnConstantLevel(
>+      static_cast<int>(kFullBufferSizeMs / kFrameDurationMs / 2),
>+      VadWithLevel::LevelAndProbability(
>+          1.f, kDifferentSpeechRmsDbfs - kInitialSaturationMarginDb,
>+          kDifferentSpeechRmsDbfs),
>+      &level_estimator);
>+  EXPECT_GT(
>+      std::abs(kDifferentSpeechRmsDbfs - level_estimator.LatestLevelEstimate()),
>+      kMaxDifferenceDb);
>+
>+  // Run for some more time. Afterwards, we should have adapted.
>+  RunOnConstantLevel(
>+      static_cast<int>(3 * kFullBufferSizeMs / kFrameDurationMs),
>+      VadWithLevel::LevelAndProbability(
>+          1.f, kDifferentSpeechRmsDbfs - kInitialSaturationMarginDb,
>+          kDifferentSpeechRmsDbfs),
>+      &level_estimator);
>+  EXPECT_NEAR(level_estimator.LatestLevelEstimate(), kDifferentSpeechRmsDbfs,
>+              kMaxDifferenceDb * 0.5f);
>+}
>+
>+TEST(AutomaticGainController2AdaptiveModeLevelEstimator,
>+     ResetGivesFastAdaptation) {
>+  ApmDataDumper apm_data_dumper(0);
>+  AdaptiveModeLevelEstimator level_estimator(&apm_data_dumper);
>+
>+  // Run the level estimator for one window size interval. This gives time to
>+  // adapt.
>+  constexpr float kInitialSpeechRmsDbfs = -30.f;
>+  RunOnConstantLevel(
>+      kFullBufferSizeMs / kFrameDurationMs,
>+      VadWithLevel::LevelAndProbability(
>+          1.f, kInitialSpeechRmsDbfs - kInitialSaturationMarginDb,
>+          kInitialSpeechRmsDbfs),
>+      &level_estimator);
>+
>+  constexpr float kDifferentSpeechRmsDbfs = -10.f;
>+  // Reset and run one half window size interval.
>+  level_estimator.Reset();
>+
>+  RunOnConstantLevel(
>+      kFullBufferSizeMs / kFrameDurationMs / 2,
>+      VadWithLevel::LevelAndProbability(
>+          1.f, kDifferentSpeechRmsDbfs - kInitialSaturationMarginDb,
>+          kDifferentSpeechRmsDbfs),
>+      &level_estimator);
>+
>+  // The level should be close to 'kDifferentSpeechRmsDbfs'.
>+  const float kMaxDifferenceDb =
>+      0.1f * std::abs(kDifferentSpeechRmsDbfs - kInitialSpeechRmsDbfs);
>+  EXPECT_LT(
>+      std::abs(kDifferentSpeechRmsDbfs - level_estimator.LatestLevelEstimate()),
>+      kMaxDifferenceDb);
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/agc2_common.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/agc2_common.h
>new file mode 100644
>index 00000000000..35e8f58587f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/agc2_common.h
>@@ -0,0 +1,87 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_AGC2_AGC2_COMMON_H_
>+#define MODULES_AUDIO_PROCESSING_AGC2_AGC2_COMMON_H_
>+
>+#include <stddef.h>
>+
>+#include <cmath>
>+
>+namespace webrtc {
>+
>+constexpr float kMinFloatS16Value = -32768.f;
>+constexpr float kMaxFloatS16Value = 32767.f;
>+constexpr float kMaxAbsFloatS16Value = 32768.0f;
>+
>+constexpr size_t kFrameDurationMs = 10;
>+constexpr size_t kSubFramesInFrame = 20;
>+constexpr size_t kMaximalNumberOfSamplesPerChannel = 480;
>+
>+constexpr float kAttackFilterConstant = 0.f;
>+
>+// Adaptive digital gain applier settings below.
>+constexpr float kMaxGainChangePerSecondDb = 3.f;
>+constexpr float kMaxGainChangePerFrameDb =
>+    kMaxGainChangePerSecondDb * kFrameDurationMs / 1000.f;
>+constexpr float kHeadroomDbfs = 1.f;
>+constexpr float kMaxGainDb = 30.f;
>+constexpr float kInitialAdaptiveDigitalGainDb = 8.f;
>+
>+// This parameter must be tuned together with the noise estimator.
>+constexpr float kMaxNoiseLevelDbfs = -50.f;
>+
>+// This is the threshold for speech. Speech frames are used for updating the
>+// speech level, measuring the amount of speech, and decide when to allow target
>+// gain reduction.
>+constexpr float kVadConfidenceThreshold = 0.4f;
>+
>+// The amount of 'memory' of the Level Estimator. Decides leak factors.
>+constexpr size_t kFullBufferSizeMs = 1600;
>+constexpr float kFullBufferLeakFactor = 1.f - 1.f / kFullBufferSizeMs;
>+
>+constexpr float kInitialSpeechLevelEstimateDbfs = -30.f;
>+
>+// Saturation Protector settings.
>+constexpr float kInitialSaturationMarginDb = 17.f;
>+
>+constexpr size_t kPeakEnveloperSuperFrameLengthMs = 400;
>+static_assert(kFullBufferSizeMs % kPeakEnveloperSuperFrameLengthMs == 0,
>+              "Full buffer size should be a multiple of super frame length for "
>+              "optimal Saturation Protector performance.");
>+
>+constexpr size_t kPeakEnveloperBufferSize =
>+    kFullBufferSizeMs / kPeakEnveloperSuperFrameLengthMs + 1;
>+
>+// This value is 10 ** (-1/20 * frame_size_ms / satproc_attack_ms),
>+// where satproc_attack_ms is 5000.
>+constexpr float kSaturationProtectorAttackConstant = 0.9988493699365052f;
>+
>+// This value is 10 ** (-1/20 * frame_size_ms / satproc_decay_ms),
>+// where satproc_decay_ms is 1000.
>+constexpr float kSaturationProtectorDecayConstant = 0.9997697679981565f;
>+
>+// This is computed from kDecayMs by
>+// 10 ** (-1/20 * subframe_duration / kDecayMs).
>+// |subframe_duration| is |kFrameDurationMs / kSubFramesInFrame|.
>+// kDecayMs is defined in agc2_testing_common.h
>+constexpr float kDecayFilterConstant = 0.9998848773724686f;
>+
>+// Number of interpolation points for each region of the limiter.
>+// These values have been tuned to limit the interpolated gain curve error given
>+// the limiter parameters and allowing a maximum error of +/- 32768^-1.
>+constexpr size_t kInterpolatedGainCurveKneePoints = 22;
>+constexpr size_t kInterpolatedGainCurveBeyondKneePoints = 10;
>+constexpr size_t kInterpolatedGainCurveTotalPoints =
>+    kInterpolatedGainCurveKneePoints + kInterpolatedGainCurveBeyondKneePoints;
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_AGC2_AGC2_COMMON_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/agc2_testing_common.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/agc2_testing_common.cc
>new file mode 100644
>index 00000000000..6c22492e882
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/agc2_testing_common.cc
>@@ -0,0 +1,33 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/agc2/agc2_testing_common.h"
>+
>+#include "rtc_base/checks.h"
>+
>+namespace webrtc {
>+
>+namespace test {
>+
>+std::vector<double> LinSpace(const double l,
>+                             const double r,
>+                             size_t num_points) {
>+  RTC_CHECK(num_points >= 2);
>+  std::vector<double> points(num_points);
>+  const double step = (r - l) / (num_points - 1.0);
>+  points[0] = l;
>+  for (size_t i = 1; i < num_points - 1; i++) {
>+    points[i] = static_cast<double>(l) + i * step;
>+  }
>+  points[num_points - 1] = r;
>+  return points;
>+}
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/agc2_testing_common.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/agc2_testing_common.h
>new file mode 100644
>index 00000000000..7bfadbb3fce
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/agc2_testing_common.h
>@@ -0,0 +1,78 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_AGC2_AGC2_TESTING_COMMON_H_
>+#define MODULES_AUDIO_PROCESSING_AGC2_AGC2_TESTING_COMMON_H_
>+
>+#include <math.h>
>+
>+#include <limits>
>+#include <vector>
>+
>+#include "rtc_base/checks.h"
>+
>+namespace webrtc {
>+
>+namespace test {
>+
>+// Level Estimator test parameters.
>+constexpr float kDecayMs = 500.f;
>+
>+// Limiter parameters.
>+constexpr float kLimiterMaxInputLevelDbFs = 1.f;
>+constexpr float kLimiterKneeSmoothnessDb = 1.f;
>+constexpr float kLimiterCompressionRatio = 5.f;
>+constexpr float kPi = 3.1415926536f;
>+
>+std::vector<double> LinSpace(const double l, const double r, size_t num_points);
>+
>+class SineGenerator {
>+ public:
>+  SineGenerator(float frequency, int rate)
>+      : frequency_(frequency), rate_(rate) {}
>+  float operator()() {
>+    x_radians_ += frequency_ / rate_ * 2 * kPi;
>+    if (x_radians_ > 2 * kPi) {
>+      x_radians_ -= 2 * kPi;
>+    }
>+    return 1000.f * sinf(x_radians_);
>+  }
>+
>+ private:
>+  float frequency_;
>+  int rate_;
>+  float x_radians_ = 0.f;
>+};
>+
>+class PulseGenerator {
>+ public:
>+  PulseGenerator(float frequency, int rate)
>+      : samples_period_(
>+            static_cast<int>(static_cast<float>(rate) / frequency)) {
>+    RTC_DCHECK_GT(rate, frequency);
>+  }
>+  float operator()() {
>+    sample_counter_++;
>+    if (sample_counter_ >= samples_period_) {
>+      sample_counter_ -= samples_period_;
>+    }
>+    return static_cast<float>(
>+        sample_counter_ == 0 ? std::numeric_limits<int16_t>::max() : 10.f);
>+  }
>+
>+ private:
>+  int samples_period_;
>+  int sample_counter_ = 0;
>+};
>+
>+}  // namespace test
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_AGC2_AGC2_TESTING_COMMON_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/agc2_testing_common_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/agc2_testing_common_unittest.cc
>new file mode 100644
>index 00000000000..b9f712617d3
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/agc2_testing_common_unittest.cc
>@@ -0,0 +1,26 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/agc2/agc2_testing_common.h"
>+#include "rtc_base/gunit.h"
>+
>+namespace webrtc {
>+
>+TEST(AutomaticGainController2Common, TestLinSpace) {
>+  std::vector<double> points1 = test::LinSpace(-1.0, 2.0, 4);
>+  const std::vector<double> expected_points1{{-1.0, 0.0, 1.0, 2.0}};
>+  EXPECT_EQ(expected_points1, points1);
>+
>+  std::vector<double> points2 = test::LinSpace(0.0, 1.0, 4);
>+  const std::vector<double> expected_points2{{0.0, 1.0 / 3.0, 2.0 / 3.0, 1.0}};
>+  EXPECT_EQ(points2, expected_points2);
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/biquad_filter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/biquad_filter.cc
>similarity index 82%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/biquad_filter.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/biquad_filter.cc
>index 5a4ddc891ec..9858d505846 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/biquad_filter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/biquad_filter.cc
>@@ -8,13 +8,12 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "modules/audio_processing/level_controller/biquad_filter.h"
>+#include "modules/audio_processing/agc2/biquad_filter.h"
> 
> namespace webrtc {
> 
>-// This method applies a biquad filter to an input signal x to produce an
>-// output signal y. The biquad coefficients are specified at the construction
>-// of the object.
>+// Transposed direct form I implementation of a bi-quad filter applied to an
>+// input signal |x| to produce an output signal |y|.
> void BiQuadFilter::Process(rtc::ArrayView<const float> x,
>                            rtc::ArrayView<float> y) {
>   for (size_t k = 0; k < x.size(); ++k) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/biquad_filter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/biquad_filter.h
>similarity index 70%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/biquad_filter.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/biquad_filter.h
>index dad104d43fd..284930c5952 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/biquad_filter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/biquad_filter.h
>@@ -8,10 +8,8 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#ifndef MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_BIQUAD_FILTER_H_
>-#define MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_BIQUAD_FILTER_H_
>-
>-#include <vector>
>+#ifndef MODULES_AUDIO_PROCESSING_AGC2_BIQUAD_FILTER_H_
>+#define MODULES_AUDIO_PROCESSING_AGC2_BIQUAD_FILTER_H_
> 
> #include "api/array_view.h"
> #include "rtc_base/arraysize.h"
>@@ -21,6 +19,10 @@ namespace webrtc {
> 
> class BiQuadFilter {
>  public:
>+  // Normalized filter coefficients.
>+  //        b_0 + b_1 â¢ z^(-1) + b_2 â¢ z^(-2)
>+  // H(z) = ---------------------------------
>+  //         1 + a_1 â¢ z^(-1) + a_2 â¢ z^(-2)
>   struct BiQuadCoefficients {
>     float b[3];
>     float a[2];
>@@ -32,13 +34,17 @@ class BiQuadFilter {
>     coefficients_ = coefficients;
>   }
> 
>+  void Reset() { biquad_state_.Reset(); }
>+
>   // Produces a filtered output y of the input x. Both x and y need to
>-  // have the same length.
>+  // have the same length. In-place modification is allowed.
>   void Process(rtc::ArrayView<const float> x, rtc::ArrayView<float> y);
> 
>  private:
>   struct BiQuadState {
>-    BiQuadState() {
>+    BiQuadState() { Reset(); }
>+
>+    void Reset() {
>       std::fill(b, b + arraysize(b), 0.f);
>       std::fill(a, a + arraysize(a), 0.f);
>     }
>@@ -55,4 +61,4 @@ class BiQuadFilter {
> 
> }  // namespace webrtc
> 
>-#endif  // MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_BIQUAD_FILTER_H_
>+#endif  // MODULES_AUDIO_PROCESSING_AGC2_BIQUAD_FILTER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/biquad_filter_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/biquad_filter_unittest.cc
>new file mode 100644
>index 00000000000..cd9a272787c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/biquad_filter_unittest.cc
>@@ -0,0 +1,136 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/agc2/biquad_filter.h"
>+
>+#include <algorithm>
>+#include <array>
>+#include <cmath>
>+
>+// TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
>+// #include "test/fpe_observer.h"
>+#include "rtc_base/gunit.h"
>+
>+namespace webrtc {
>+namespace test {
>+namespace {
>+
>+constexpr size_t kFrameSize = 8;
>+constexpr size_t kNumFrames = 4;
>+using FloatArraySequence =
>+    std::array<std::array<float, kFrameSize>, kNumFrames>;
>+
>+constexpr FloatArraySequence kBiQuadInputSeq = {
>+    {{{-87.166290f, -8.029022f, 101.619583f, -0.294296f, -5.825764f, -8.890625f,
>+       10.310432f, 54.845333f}},
>+     {{-64.647644f, -6.883945f, 11.059189f, -95.242538f, -108.870834f,
>+       11.024944f, 63.044102f, -52.709583f}},
>+     {{-32.350529f, -18.108028f, -74.022339f, -8.986874f, -1.525581f,
>+       103.705513f, 6.346226f, -14.319557f}},
>+     {{22.645832f, -64.597153f, 55.462521f, -109.393188f, 10.117825f,
>+       -40.019642f, -98.612228f, -8.330326f}}}};
>+
>+// Generated via "B, A = scipy.signal.butter(2, 30/12000, btype='highpass')"
>+const BiQuadFilter::BiQuadCoefficients kBiQuadConfig = {
>+    {0.99446179f, -1.98892358f, 0.99446179f},
>+    {-1.98889291f, 0.98895425f}};
>+
>+// Comparing to scipy. The expected output is generated as follows:
>+// zi = np.float32([0, 0])
>+// for i in range(4):
>+//   yn, zi = scipy.signal.lfilter(B, A, x[i], zi=zi)
>+//   print(yn)
>+constexpr FloatArraySequence kBiQuadOutputSeq = {
>+    {{{-86.68354497f, -7.02175351f, 102.10290352f, -0.37487333f, -5.87205847f,
>+       -8.85521608f, 10.33772563f, 54.51157181f}},
>+     {{-64.92531604f, -6.76395978f, 11.15534507f, -94.68073341f, -107.18177856f,
>+       13.24642474f, 64.84288941f, -50.97822629f}},
>+     {{-30.1579652f, -15.64850899f, -71.06662821f, -5.5883229f, 1.91175353f,
>+       106.5572003f, 8.57183046f, -12.06298473f}},
>+     {{24.84286614f, -62.18094158f, 57.91488056f, -106.65685933f, 13.38760103f,
>+       -36.60367134f, -94.44880104f, -3.59920354f}}}};
>+
>+// Fail for every pair from two equally sized rtc::ArrayView<float> views such
>+// that their relative error is above a given threshold. If the expected value
>+// of a pair is 0, the tolerance is used to check the absolute error.
>+void ExpectNearRelative(rtc::ArrayView<const float> expected,
>+                        rtc::ArrayView<const float> computed,
>+                        const float tolerance) {
>+  // The relative error is undefined when the expected value is 0.
>+  // When that happens, check the absolute error instead. |safe_den| is used
>+  // below to implement such logic.
>+  auto safe_den = [](float x) { return (x == 0.f) ? 1.f : std::fabs(x); };
>+  ASSERT_EQ(expected.size(), computed.size());
>+  for (size_t i = 0; i < expected.size(); ++i) {
>+    const float abs_diff = std::fabs(expected[i] - computed[i]);
>+    // No failure when the values are equal.
>+    if (abs_diff == 0.f)
>+      continue;
>+    SCOPED_TRACE(i);
>+    SCOPED_TRACE(expected[i]);
>+    SCOPED_TRACE(computed[i]);
>+    EXPECT_LE(abs_diff / safe_den(expected[i]), tolerance);
>+  }
>+}
>+
>+}  // namespace
>+
>+TEST(BiQuadFilterTest, FilterNotInPlace) {
>+  BiQuadFilter filter;
>+  filter.Initialize(kBiQuadConfig);
>+  std::array<float, kFrameSize> samples;
>+
>+  // TODO(https://bugs.webrtc.org/8948): Add when the issue is fixed.
>+  // FloatingPointExceptionObserver fpe_observer;
>+
>+  for (size_t i = 0; i < kNumFrames; ++i) {
>+    SCOPED_TRACE(i);
>+    filter.Process(kBiQuadInputSeq[i], samples);
>+    ExpectNearRelative(kBiQuadOutputSeq[i], samples, 2e-4f);
>+  }
>+}
>+
>+TEST(BiQuadFilterTest, FilterInPlace) {
>+  BiQuadFilter filter;
>+  filter.Initialize(kBiQuadConfig);
>+  std::array<float, kFrameSize> samples;
>+
>+  // TODO(https://bugs.webrtc.org/8948): Add when the issue is fixed.
>+  // FloatingPointExceptionObserver fpe_observer;
>+
>+  for (size_t i = 0; i < kNumFrames; ++i) {
>+    SCOPED_TRACE(i);
>+    std::copy(kBiQuadInputSeq[i].begin(), kBiQuadInputSeq[i].end(),
>+              samples.begin());
>+    filter.Process({samples}, {samples});
>+    ExpectNearRelative(kBiQuadOutputSeq[i], samples, 2e-4f);
>+  }
>+}
>+
>+TEST(BiQuadFilterTest, Reset) {
>+  BiQuadFilter filter;
>+  filter.Initialize(kBiQuadConfig);
>+
>+  std::array<float, kFrameSize> samples1;
>+  for (size_t i = 0; i < kNumFrames; ++i) {
>+    filter.Process(kBiQuadInputSeq[i], samples1);
>+  }
>+
>+  filter.Reset();
>+  std::array<float, kFrameSize> samples2;
>+  for (size_t i = 0; i < kNumFrames; ++i) {
>+    filter.Process(kBiQuadInputSeq[i], samples2);
>+  }
>+
>+  EXPECT_EQ(samples1, samples2);
>+}
>+
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/compute_interpolated_gain_curve.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/compute_interpolated_gain_curve.cc
>new file mode 100644
>index 00000000000..f395bceffa1
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/compute_interpolated_gain_curve.cc
>@@ -0,0 +1,228 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/agc2/compute_interpolated_gain_curve.h"
>+
>+#include <algorithm>
>+#include <cmath>
>+#include <queue>
>+#include <tuple>
>+#include <utility>
>+#include <vector>
>+
>+#include "modules/audio_processing/agc2/agc2_common.h"
>+#include "modules/audio_processing/agc2/agc2_testing_common.h"
>+#include "modules/audio_processing/agc2/limiter.h"
>+#include "rtc_base/checks.h"
>+
>+namespace webrtc {
>+namespace {
>+
>+std::pair<double, double> ComputeLinearApproximationParams(
>+    const Limiter* limiter,
>+    const double x) {
>+  const double m = limiter->GetGainFirstDerivativeLinear(x);
>+  const double q = limiter->GetGainLinear(x) - m * x;
>+  return {m, q};
>+}
>+
>+double ComputeAreaUnderPiecewiseLinearApproximation(const Limiter* limiter,
>+                                                    const double x0,
>+                                                    const double x1) {
>+  RTC_CHECK_LT(x0, x1);
>+
>+  // Linear approximation in x0 and x1.
>+  double m0, q0, m1, q1;
>+  std::tie(m0, q0) = ComputeLinearApproximationParams(limiter, x0);
>+  std::tie(m1, q1) = ComputeLinearApproximationParams(limiter, x1);
>+
>+  // Intersection point between two adjacent linear pieces.
>+  RTC_CHECK_NE(m1, m0);
>+  const double x_split = (q0 - q1) / (m1 - m0);
>+  RTC_CHECK_LT(x0, x_split);
>+  RTC_CHECK_LT(x_split, x1);
>+
>+  auto area_under_linear_piece = [](double x_l, double x_r, double m,
>+                                    double q) {
>+    return x_r * (m * x_r / 2.0 + q) - x_l * (m * x_l / 2.0 + q);
>+  };
>+  return area_under_linear_piece(x0, x_split, m0, q0) +
>+         area_under_linear_piece(x_split, x1, m1, q1);
>+}
>+
>+// Computes the approximation error in the limiter region for a given interval.
>+// The error is computed as the difference between the areas beneath the limiter
>+// curve to approximate and its linear under-approximation.
>+double LimiterUnderApproximationNegativeError(const Limiter* limiter,
>+                                              const double x0,
>+                                              const double x1) {
>+  const double area_limiter = limiter->GetGainIntegralLinear(x0, x1);
>+  const double area_interpolated_curve =
>+      ComputeAreaUnderPiecewiseLinearApproximation(limiter, x0, x1);
>+  RTC_CHECK_GE(area_limiter, area_interpolated_curve);
>+  return area_limiter - area_interpolated_curve;
>+}
>+
>+// Automatically finds where to sample the beyond-knee region of a limiter using
>+// a greedy optimization algorithm that iteratively decreases the approximation
>+// error.
>+// The solution is sub-optimal because the algorithm is greedy and the points
>+// are assigned by halving intervals (starting with the whole beyond-knee region
>+// as a single interval). However, even if sub-optimal, this algorithm works
>+// well in practice and it is efficiently implemented using priority queues.
>+std::vector<double> SampleLimiterRegion(const Limiter* limiter) {
>+  static_assert(kInterpolatedGainCurveBeyondKneePoints > 2, "");
>+
>+  struct Interval {
>+    Interval() = default;  // Ctor required by std::priority_queue.
>+    Interval(double l, double r, double e) : x0(l), x1(r), error(e) {
>+      RTC_CHECK(x0 < x1);
>+    }
>+    bool operator<(const Interval& other) const { return error < other.error; }
>+
>+    double x0;
>+    double x1;
>+    double error;
>+  };
>+
>+  std::priority_queue<Interval, std::vector<Interval>> q;
>+  q.emplace(limiter->limiter_start_linear(), limiter->max_input_level_linear(),
>+            LimiterUnderApproximationNegativeError(
>+                limiter, limiter->limiter_start_linear(),
>+                limiter->max_input_level_linear()));
>+
>+  // Iteratively find points by halving the interval with greatest error.
>+  while (q.size() < kInterpolatedGainCurveBeyondKneePoints) {
>+    // Get the interval with highest error.
>+    const auto interval = q.top();
>+    q.pop();
>+
>+    // Split |interval| and enqueue.
>+    double x_split = (interval.x0 + interval.x1) / 2.0;
>+    q.emplace(interval.x0, x_split,
>+              LimiterUnderApproximationNegativeError(limiter, interval.x0,
>+                                                     x_split));  // Left.
>+    q.emplace(x_split, interval.x1,
>+              LimiterUnderApproximationNegativeError(limiter, x_split,
>+                                                     interval.x1));  // Right.
>+  }
>+
>+  // Copy x1 values and sort them.
>+  RTC_CHECK_EQ(q.size(), kInterpolatedGainCurveBeyondKneePoints);
>+  std::vector<double> samples(kInterpolatedGainCurveBeyondKneePoints);
>+  for (size_t i = 0; i < kInterpolatedGainCurveBeyondKneePoints; ++i) {
>+    const auto interval = q.top();
>+    q.pop();
>+    samples[i] = interval.x1;
>+  }
>+  RTC_CHECK(q.empty());
>+  std::sort(samples.begin(), samples.end());
>+
>+  return samples;
>+}
>+
>+// Compute the parameters to over-approximate the knee region via linear
>+// interpolation. Over-approximating is saturation-safe since the knee region is
>+// convex.
>+void PrecomputeKneeApproxParams(const Limiter* limiter,
>+                                test::InterpolatedParameters* parameters) {
>+  static_assert(kInterpolatedGainCurveKneePoints > 2, "");
>+  // Get |kInterpolatedGainCurveKneePoints| - 1 equally spaced points.
>+  const std::vector<double> points = test::LinSpace(
>+      limiter->knee_start_linear(), limiter->limiter_start_linear(),
>+      kInterpolatedGainCurveKneePoints - 1);
>+
>+  // Set the first two points. The second is computed to help with the beginning
>+  // of the knee region, which has high curvature.
>+  parameters->computed_approximation_params_x[0] = points[0];
>+  parameters->computed_approximation_params_x[1] =
>+      (points[0] + points[1]) / 2.0;
>+  // Copy the remaining points.
>+  std::copy(std::begin(points) + 1, std::end(points),
>+            std::begin(parameters->computed_approximation_params_x) + 2);
>+
>+  // Compute (m, q) pairs for each linear piece y = mx + q.
>+  for (size_t i = 0; i < kInterpolatedGainCurveKneePoints - 1; ++i) {
>+    const double x0 = parameters->computed_approximation_params_x[i];
>+    const double x1 = parameters->computed_approximation_params_x[i + 1];
>+    const double y0 = limiter->GetGainLinear(x0);
>+    const double y1 = limiter->GetGainLinear(x1);
>+    RTC_CHECK_NE(x1, x0);
>+    parameters->computed_approximation_params_m[i] = (y1 - y0) / (x1 - x0);
>+    parameters->computed_approximation_params_q[i] =
>+        y0 - parameters->computed_approximation_params_m[i] * x0;
>+  }
>+}
>+
>+// Compute the parameters to under-approximate the beyond-knee region via linear
>+// interpolation and greedy sampling. Under-approximating is saturation-safe
>+// since the beyond-knee region is concave.
>+void PrecomputeBeyondKneeApproxParams(
>+    const Limiter* limiter,
>+    test::InterpolatedParameters* parameters) {
>+  // Find points on which the linear pieces are tangent to the gain curve.
>+  const auto samples = SampleLimiterRegion(limiter);
>+
>+  // Parametrize each linear piece.
>+  double m, q;
>+  std::tie(m, q) = ComputeLinearApproximationParams(
>+      limiter,
>+      parameters
>+          ->computed_approximation_params_x[kInterpolatedGainCurveKneePoints -
>+                                            1]);
>+  parameters
>+      ->computed_approximation_params_m[kInterpolatedGainCurveKneePoints - 1] =
>+      m;
>+  parameters
>+      ->computed_approximation_params_q[kInterpolatedGainCurveKneePoints - 1] =
>+      q;
>+  for (size_t i = 0; i < samples.size(); ++i) {
>+    std::tie(m, q) = ComputeLinearApproximationParams(limiter, samples[i]);
>+    parameters
>+        ->computed_approximation_params_m[i +
>+                                          kInterpolatedGainCurveKneePoints] = m;
>+    parameters
>+        ->computed_approximation_params_q[i +
>+                                          kInterpolatedGainCurveKneePoints] = q;
>+  }
>+
>+  // Find the point of intersection between adjacent linear pieces. They will be
>+  // used as boundaries between adjacent linear pieces.
>+  for (size_t i = kInterpolatedGainCurveKneePoints;
>+       i < kInterpolatedGainCurveKneePoints +
>+               kInterpolatedGainCurveBeyondKneePoints;
>+       ++i) {
>+    RTC_CHECK_NE(parameters->computed_approximation_params_m[i],
>+                 parameters->computed_approximation_params_m[i - 1]);
>+    parameters->computed_approximation_params_x[i] =
>+        (  // Formula: (q0 - q1) / (m1 - m0).
>+            parameters->computed_approximation_params_q[i - 1] -
>+            parameters->computed_approximation_params_q[i]) /
>+        (parameters->computed_approximation_params_m[i] -
>+         parameters->computed_approximation_params_m[i - 1]);
>+  }
>+}
>+
>+}  // namespace
>+
>+namespace test {
>+
>+InterpolatedParameters ComputeInterpolatedGainCurveApproximationParams() {
>+  InterpolatedParameters parameters;
>+  Limiter limiter;
>+  parameters.computed_approximation_params_x.fill(0.0f);
>+  parameters.computed_approximation_params_m.fill(0.0f);
>+  parameters.computed_approximation_params_q.fill(0.0f);
>+  PrecomputeKneeApproxParams(&limiter, &parameters);
>+  PrecomputeBeyondKneeApproxParams(&limiter, &parameters);
>+  return parameters;
>+}
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/compute_interpolated_gain_curve.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/compute_interpolated_gain_curve.h
>new file mode 100644
>index 00000000000..5f52441ec36
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/compute_interpolated_gain_curve.h
>@@ -0,0 +1,48 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_AGC2_COMPUTE_INTERPOLATED_GAIN_CURVE_H_
>+#define MODULES_AUDIO_PROCESSING_AGC2_COMPUTE_INTERPOLATED_GAIN_CURVE_H_
>+
>+#include <array>
>+
>+#include "modules/audio_processing/agc2/agc2_common.h"
>+
>+namespace webrtc {
>+
>+namespace test {
>+
>+// Parameters for interpolated gain curve using under-approximation to
>+// avoid saturation.
>+//
>+// The saturation gain is defined in order to let hard-clipping occur for
>+// those samples having a level that falls in the saturation region. It is an
>+// upper bound of the actual gain to apply - i.e., that returned by the
>+// limiter.
>+
>+// Knee and beyond-knee regions approximation parameters.
>+// The gain curve is approximated as a piece-wise linear function.
>+// |approx_params_x_| are the boundaries between adjacent linear pieces,
>+// |approx_params_m_| and |approx_params_q_| are the slope and the y-intercept
>+// values of each piece.
>+struct InterpolatedParameters {
>+  std::array<float, kInterpolatedGainCurveTotalPoints>
>+      computed_approximation_params_x;
>+  std::array<float, kInterpolatedGainCurveTotalPoints>
>+      computed_approximation_params_m;
>+  std::array<float, kInterpolatedGainCurveTotalPoints>
>+      computed_approximation_params_q;
>+};
>+
>+InterpolatedParameters ComputeInterpolatedGainCurveApproximationParams();
>+}  // namespace test
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_AGC2_COMPUTE_INTERPOLATED_GAIN_CURVE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/down_sampler.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/down_sampler.cc
>similarity index 67%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/down_sampler.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/down_sampler.cc
>index a1702f432c5..50486e0a367 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/down_sampler.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/down_sampler.cc
>@@ -8,20 +8,24 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "modules/audio_processing/level_controller/down_sampler.h"
>+#include "modules/audio_processing/agc2/down_sampler.h"
> 
> #include <string.h>
> #include <algorithm>
>-#include <vector>
> 
>-#include "modules/audio_processing/include/audio_processing.h"
>-#include "modules/audio_processing/level_controller/biquad_filter.h"
>+#include "modules/audio_processing/agc2/biquad_filter.h"
> #include "modules/audio_processing/logging/apm_data_dumper.h"
> #include "rtc_base/checks.h"
> 
> namespace webrtc {
> namespace {
> 
>+constexpr int kChunkSizeMs = 10;
>+constexpr int kSampleRate8kHz = 8000;
>+constexpr int kSampleRate16kHz = 16000;
>+constexpr int kSampleRate32kHz = 32000;
>+constexpr int kSampleRate48kHz = 48000;
>+
> // Bandlimiter coefficients computed based on that only
> // the first 40 bins of the spectrum for the downsampled
> // signal are used.
>@@ -47,21 +51,20 @@ DownSampler::DownSampler(ApmDataDumper* data_dumper)
>   Initialize(48000);
> }
> void DownSampler::Initialize(int sample_rate_hz) {
>-  RTC_DCHECK(sample_rate_hz == AudioProcessing::kSampleRate8kHz ||
>-             sample_rate_hz == AudioProcessing::kSampleRate16kHz ||
>-             sample_rate_hz == AudioProcessing::kSampleRate32kHz ||
>-             sample_rate_hz == AudioProcessing::kSampleRate48kHz);
>+  RTC_DCHECK(
>+      sample_rate_hz == kSampleRate8kHz || sample_rate_hz == kSampleRate16kHz ||
>+      sample_rate_hz == kSampleRate32kHz || sample_rate_hz == kSampleRate48kHz);
> 
>   sample_rate_hz_ = sample_rate_hz;
>   down_sampling_factor_ = rtc::CheckedDivExact(sample_rate_hz_, 8000);
> 
>   /// Note that the down sampling filter is not used if the sample rate is 8
>   /// kHz.
>-  if (sample_rate_hz_ == AudioProcessing::kSampleRate16kHz) {
>+  if (sample_rate_hz_ == kSampleRate16kHz) {
>     low_pass_filter_.Initialize(kLowPassFilterCoefficients_16kHz);
>-  } else if (sample_rate_hz_ == AudioProcessing::kSampleRate32kHz) {
>+  } else if (sample_rate_hz_ == kSampleRate32kHz) {
>     low_pass_filter_.Initialize(kLowPassFilterCoefficients_32kHz);
>-  } else if (sample_rate_hz_ == AudioProcessing::kSampleRate48kHz) {
>+  } else if (sample_rate_hz_ == kSampleRate48kHz) {
>     low_pass_filter_.Initialize(kLowPassFilterCoefficients_48kHz);
>   }
> }
>@@ -69,17 +72,13 @@ void DownSampler::Initialize(int sample_rate_hz) {
> void DownSampler::DownSample(rtc::ArrayView<const float> in,
>                              rtc::ArrayView<float> out) {
>   data_dumper_->DumpWav("lc_down_sampler_input", in, sample_rate_hz_, 1);
>-  RTC_DCHECK_EQ(sample_rate_hz_ * AudioProcessing::kChunkSizeMs / 1000,
>-                in.size());
>-  RTC_DCHECK_EQ(
>-      AudioProcessing::kSampleRate8kHz * AudioProcessing::kChunkSizeMs / 1000,
>-      out.size());
>-  const size_t kMaxNumFrames =
>-      AudioProcessing::kSampleRate48kHz * AudioProcessing::kChunkSizeMs / 1000;
>+  RTC_DCHECK_EQ(sample_rate_hz_ * kChunkSizeMs / 1000, in.size());
>+  RTC_DCHECK_EQ(kSampleRate8kHz * kChunkSizeMs / 1000, out.size());
>+  const size_t kMaxNumFrames = kSampleRate48kHz * kChunkSizeMs / 1000;
>   float x[kMaxNumFrames];
> 
>   // Band-limit the signal to 4 kHz.
>-  if (sample_rate_hz_ != AudioProcessing::kSampleRate8kHz) {
>+  if (sample_rate_hz_ != kSampleRate8kHz) {
>     low_pass_filter_.Process(in, rtc::ArrayView<float>(x, in.size()));
> 
>     // Downsample the signal.
>@@ -93,8 +92,7 @@ void DownSampler::DownSample(rtc::ArrayView<const float> in,
>     std::copy(in.data(), in.data() + in.size(), out.data());
>   }
> 
>-  data_dumper_->DumpWav("lc_down_sampler_output", out,
>-                        AudioProcessing::kSampleRate8kHz, 1);
>+  data_dumper_->DumpWav("lc_down_sampler_output", out, kSampleRate8kHz, 1);
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/down_sampler.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/down_sampler.h
>similarity index 77%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/down_sampler.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/down_sampler.h
>index d6502425a1f..a609ea8e0c5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/down_sampler.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/down_sampler.h
>@@ -8,11 +8,11 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#ifndef MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_DOWN_SAMPLER_H_
>-#define MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_DOWN_SAMPLER_H_
>+#ifndef MODULES_AUDIO_PROCESSING_AGC2_DOWN_SAMPLER_H_
>+#define MODULES_AUDIO_PROCESSING_AGC2_DOWN_SAMPLER_H_
> 
> #include "api/array_view.h"
>-#include "modules/audio_processing/level_controller/biquad_filter.h"
>+#include "modules/audio_processing/agc2/biquad_filter.h"
> #include "rtc_base/constructormagic.h"
> 
> namespace webrtc {
>@@ -37,4 +37,4 @@ class DownSampler {
> 
> }  // namespace webrtc
> 
>-#endif  // MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_DOWN_SAMPLER_H_
>+#endif  // MODULES_AUDIO_PROCESSING_AGC2_DOWN_SAMPLER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/fixed_digital_level_estimator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/fixed_digital_level_estimator.cc
>new file mode 100644
>index 00000000000..9a1fd281ca6
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/fixed_digital_level_estimator.cc
>@@ -0,0 +1,100 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/agc2/fixed_digital_level_estimator.h"
>+
>+#include <algorithm>
>+#include <cmath>
>+
>+#include "modules/audio_processing/logging/apm_data_dumper.h"
>+#include "rtc_base/checks.h"
>+
>+namespace webrtc {
>+
>+FixedDigitalLevelEstimator::FixedDigitalLevelEstimator(
>+    size_t sample_rate_hz,
>+    ApmDataDumper* apm_data_dumper)
>+    : apm_data_dumper_(apm_data_dumper) {
>+  SetSampleRate(sample_rate_hz);
>+  CheckParameterCombination();
>+  RTC_DCHECK(apm_data_dumper_);
>+  apm_data_dumper_->DumpRaw("agc2_level_estimator_samplerate", sample_rate_hz);
>+}
>+
>+void FixedDigitalLevelEstimator::CheckParameterCombination() {
>+  RTC_DCHECK_GT(samples_in_frame_, 0);
>+  RTC_DCHECK_LE(kSubFramesInFrame, samples_in_frame_);
>+  RTC_DCHECK_EQ(samples_in_frame_ % kSubFramesInFrame, 0);
>+  RTC_DCHECK_GT(samples_in_sub_frame_, 1);
>+}
>+
>+std::array<float, kSubFramesInFrame> FixedDigitalLevelEstimator::ComputeLevel(
>+    const AudioFrameView<const float>& float_frame) {
>+  RTC_DCHECK_GT(float_frame.num_channels(), 0);
>+  RTC_DCHECK_EQ(float_frame.samples_per_channel(), samples_in_frame_);
>+
>+  // Compute max envelope without smoothing.
>+  std::array<float, kSubFramesInFrame> envelope{};
>+  for (size_t channel_idx = 0; channel_idx < float_frame.num_channels();
>+       ++channel_idx) {
>+    const auto channel = float_frame.channel(channel_idx);
>+    for (size_t sub_frame = 0; sub_frame < kSubFramesInFrame; ++sub_frame) {
>+      for (size_t sample_in_sub_frame = 0;
>+           sample_in_sub_frame < samples_in_sub_frame_; ++sample_in_sub_frame) {
>+        envelope[sub_frame] =
>+            std::max(envelope[sub_frame],
>+                     std::abs(channel[sub_frame * samples_in_sub_frame_ +
>+                                      sample_in_sub_frame]));
>+      }
>+    }
>+  }
>+
>+  // Make sure envelope increases happen one step earlier so that the
>+  // corresponding *gain decrease* doesn't miss a sudden signal
>+  // increase due to interpolation.
>+  for (size_t sub_frame = 0; sub_frame < kSubFramesInFrame - 1; ++sub_frame) {
>+    if (envelope[sub_frame] < envelope[sub_frame + 1]) {
>+      envelope[sub_frame] = envelope[sub_frame + 1];
>+    }
>+  }
>+
>+  // Add attack / decay smoothing.
>+  for (size_t sub_frame = 0; sub_frame < kSubFramesInFrame; ++sub_frame) {
>+    const float envelope_value = envelope[sub_frame];
>+    if (envelope_value > filter_state_level_) {
>+      envelope[sub_frame] = envelope_value * (1 - kAttackFilterConstant) +
>+                            filter_state_level_ * kAttackFilterConstant;
>+    } else {
>+      envelope[sub_frame] = envelope_value * (1 - kDecayFilterConstant) +
>+                            filter_state_level_ * kDecayFilterConstant;
>+    }
>+    filter_state_level_ = envelope[sub_frame];
>+
>+    // Dump data for debug.
>+    RTC_DCHECK(apm_data_dumper_);
>+    const auto channel = float_frame.channel(0);
>+    apm_data_dumper_->DumpRaw("agc2_level_estimator_samples",
>+                              samples_in_sub_frame_,
>+                              &channel[sub_frame * samples_in_sub_frame_]);
>+    apm_data_dumper_->DumpRaw("agc2_level_estimator_level",
>+                              envelope[sub_frame]);
>+  }
>+
>+  return envelope;
>+}
>+
>+void FixedDigitalLevelEstimator::SetSampleRate(size_t sample_rate_hz) {
>+  samples_in_frame_ = rtc::CheckedDivExact(sample_rate_hz * kFrameDurationMs,
>+                                           static_cast<size_t>(1000));
>+  samples_in_sub_frame_ =
>+      rtc::CheckedDivExact(samples_in_frame_, kSubFramesInFrame);
>+  CheckParameterCombination();
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/fixed_digital_level_estimator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/fixed_digital_level_estimator.h
>new file mode 100644
>index 00000000000..b0e7a6d0887
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/fixed_digital_level_estimator.h
>@@ -0,0 +1,60 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_AGC2_FIXED_DIGITAL_LEVEL_ESTIMATOR_H_
>+#define MODULES_AUDIO_PROCESSING_AGC2_FIXED_DIGITAL_LEVEL_ESTIMATOR_H_
>+
>+#include <array>
>+#include <vector>
>+
>+#include "modules/audio_processing/agc2/agc2_common.h"
>+#include "modules/audio_processing/include/audio_frame_view.h"
>+#include "rtc_base/constructormagic.h"
>+
>+namespace webrtc {
>+
>+class ApmDataDumper;
>+// Produces a smooth signal level estimate from an input audio
>+// stream. The estimate smoothing is done through exponential
>+// filtering.
>+class FixedDigitalLevelEstimator {
>+ public:
>+  // Sample rates are allowed if the number of samples in a frame
>+  // (sample_rate_hz * kFrameDurationMs / 1000) is divisible by
>+  // kSubFramesInSample. For kFrameDurationMs=10 and
>+  // kSubFramesInSample=20, this means that sample_rate_hz has to be
>+  // divisible by 2000.
>+  FixedDigitalLevelEstimator(size_t sample_rate_hz,
>+                             ApmDataDumper* apm_data_dumper);
>+
>+  // The input is assumed to be in FloatS16 format. Scaled input will
>+  // produce similarly scaled output. A frame of with kFrameDurationMs
>+  // ms of audio produces a level estimates in the same scale. The
>+  // level estimate contains kSubFramesInFrame values.
>+  std::array<float, kSubFramesInFrame> ComputeLevel(
>+      const AudioFrameView<const float>& float_frame);
>+
>+  // Rate may be changed at any time (but not concurrently) from the
>+  // value passed to the constructor. The class is not thread safe.
>+  void SetSampleRate(size_t sample_rate_hz);
>+
>+ private:
>+  void CheckParameterCombination();
>+
>+  ApmDataDumper* const apm_data_dumper_ = nullptr;
>+  float filter_state_level_ = 0.f;
>+  size_t samples_in_frame_;
>+  size_t samples_in_sub_frame_;
>+
>+  RTC_DISALLOW_COPY_AND_ASSIGN(FixedDigitalLevelEstimator);
>+};
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_AGC2_FIXED_DIGITAL_LEVEL_ESTIMATOR_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/fixed_digital_level_estimator_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/fixed_digital_level_estimator_unittest.cc
>new file mode 100644
>index 00000000000..7547f8e2ed1
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/fixed_digital_level_estimator_unittest.cc
>@@ -0,0 +1,158 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/agc2/fixed_digital_level_estimator.h"
>+
>+#include <limits>
>+
>+#include "common_audio/include/audio_util.h"
>+#include "modules/audio_processing/agc2/agc2_common.h"
>+#include "modules/audio_processing/agc2/agc2_testing_common.h"
>+#include "modules/audio_processing/agc2/vector_float_frame.h"
>+#include "modules/audio_processing/logging/apm_data_dumper.h"
>+#include "rtc_base/gunit.h"
>+
>+namespace webrtc {
>+namespace {
>+
>+constexpr float kInputLevel = 10000.f;
>+
>+// Run audio at specified settings through the level estimator, and
>+// verify that the output level falls within the bounds.
>+void TestLevelEstimator(int sample_rate_hz,
>+                        int num_channels,
>+                        float input_level_linear_scale,
>+                        float expected_min,
>+                        float expected_max) {
>+  ApmDataDumper apm_data_dumper(0);
>+  FixedDigitalLevelEstimator level_estimator(sample_rate_hz, &apm_data_dumper);
>+
>+  const VectorFloatFrame vectors_with_float_frame(
>+      num_channels, rtc::CheckedDivExact(sample_rate_hz, 100),
>+      input_level_linear_scale);
>+
>+  for (int i = 0; i < 500; ++i) {
>+    const auto level = level_estimator.ComputeLevel(
>+        vectors_with_float_frame.float_frame_view());
>+
>+    // Give the estimator some time to ramp up.
>+    if (i < 50) {
>+      continue;
>+    }
>+
>+    for (const auto& x : level) {
>+      EXPECT_LE(expected_min, x);
>+      EXPECT_LE(x, expected_max);
>+    }
>+  }
>+}
>+
>+// Returns time it takes for the level estimator to decrease its level
>+// estimate by 'level_reduction_db'.
>+float TimeMsToDecreaseLevel(int sample_rate_hz,
>+                            int num_channels,
>+                            float input_level_db,
>+                            float level_reduction_db) {
>+  const float input_level = DbfsToFloatS16(input_level_db);
>+  RTC_DCHECK_GT(level_reduction_db, 0);
>+
>+  const VectorFloatFrame vectors_with_float_frame(
>+      num_channels, rtc::CheckedDivExact(sample_rate_hz, 100), input_level);
>+
>+  ApmDataDumper apm_data_dumper(0);
>+  FixedDigitalLevelEstimator level_estimator(sample_rate_hz, &apm_data_dumper);
>+
>+  // Give the LevelEstimator plenty of time to ramp up and stabilize
>+  float last_level = 0.f;
>+  for (int i = 0; i < 500; ++i) {
>+    const auto level_envelope = level_estimator.ComputeLevel(
>+        vectors_with_float_frame.float_frame_view());
>+    last_level = *level_envelope.rbegin();
>+  }
>+
>+  // Set input to 0.
>+  VectorFloatFrame vectors_with_zero_float_frame(
>+      num_channels, rtc::CheckedDivExact(sample_rate_hz, 100), 0);
>+
>+  const float reduced_level_linear =
>+      DbfsToFloatS16(input_level_db - level_reduction_db);
>+  int sub_frames_until_level_reduction = 0;
>+  while (last_level > reduced_level_linear) {
>+    const auto level_envelope = level_estimator.ComputeLevel(
>+        vectors_with_zero_float_frame.float_frame_view());
>+    for (const auto& v : level_envelope) {
>+      EXPECT_LT(v, last_level);
>+      sub_frames_until_level_reduction++;
>+      last_level = v;
>+      if (last_level <= reduced_level_linear) {
>+        break;
>+      }
>+    }
>+  }
>+  return static_cast<float>(sub_frames_until_level_reduction) *
>+         kFrameDurationMs / kSubFramesInFrame;
>+}
>+}  // namespace
>+
>+TEST(AutomaticGainController2LevelEstimator, EstimatorShouldNotCrash) {
>+  TestLevelEstimator(8000, 1, 0, std::numeric_limits<float>::lowest(),
>+                     std::numeric_limits<float>::max());
>+}
>+
>+TEST(AutomaticGainController2LevelEstimator,
>+     EstimatorShouldEstimateConstantLevel) {
>+  TestLevelEstimator(10000, 1, kInputLevel, kInputLevel * 0.99,
>+                     kInputLevel * 1.01);
>+}
>+
>+TEST(AutomaticGainController2LevelEstimator,
>+     EstimatorShouldEstimateConstantLevelForManyChannels) {
>+  constexpr size_t num_channels = 10;
>+  TestLevelEstimator(20000, num_channels, kInputLevel, kInputLevel * 0.99,
>+                     kInputLevel * 1.01);
>+}
>+
>+TEST(AutomaticGainController2LevelEstimator, TimeToDecreaseForLowLevel) {
>+  constexpr float kLevelReductionDb = 25;
>+  constexpr float kInitialLowLevel = -40;
>+  constexpr float kExpectedTime = kLevelReductionDb * test::kDecayMs;
>+
>+  const float time_to_decrease =
>+      TimeMsToDecreaseLevel(22000, 1, kInitialLowLevel, kLevelReductionDb);
>+
>+  EXPECT_LE(kExpectedTime * 0.9, time_to_decrease);
>+  EXPECT_LE(time_to_decrease, kExpectedTime * 1.1);
>+}
>+
>+TEST(AutomaticGainController2LevelEstimator, TimeToDecreaseForFullScaleLevel) {
>+  constexpr float kLevelReductionDb = 25;
>+  constexpr float kExpectedTime = kLevelReductionDb * test::kDecayMs;
>+
>+  const float time_to_decrease =
>+      TimeMsToDecreaseLevel(26000, 1, 0, kLevelReductionDb);
>+
>+  EXPECT_LE(kExpectedTime * 0.9, time_to_decrease);
>+  EXPECT_LE(time_to_decrease, kExpectedTime * 1.1);
>+}
>+
>+TEST(AutomaticGainController2LevelEstimator,
>+     TimeToDecreaseForMultipleChannels) {
>+  constexpr float kLevelReductionDb = 25;
>+  constexpr float kExpectedTime = kLevelReductionDb * test::kDecayMs;
>+  constexpr size_t kNumChannels = 10;
>+
>+  const float time_to_decrease =
>+      TimeMsToDecreaseLevel(28000, kNumChannels, 0, kLevelReductionDb);
>+
>+  EXPECT_LE(kExpectedTime * 0.9, time_to_decrease);
>+  EXPECT_LE(time_to_decrease, kExpectedTime * 1.1);
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/fixed_gain_controller.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/fixed_gain_controller.cc
>new file mode 100644
>index 00000000000..6aa390ea366
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/fixed_gain_controller.cc
>@@ -0,0 +1,95 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/agc2/fixed_gain_controller.h"
>+
>+#include <algorithm>
>+#include <cmath>
>+
>+#include "api/array_view.h"
>+#include "common_audio/include/audio_util.h"
>+#include "modules/audio_processing/agc2/agc2_common.h"
>+#include "modules/audio_processing/agc2/interpolated_gain_curve.h"
>+#include "modules/audio_processing/logging/apm_data_dumper.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/logging.h"
>+#include "rtc_base/numerics/safe_minmax.h"
>+
>+namespace webrtc {
>+namespace {
>+
>+// Returns true when the gain factor is so close to 1 that it would
>+// not affect int16 samples.
>+bool CloseToOne(float gain_factor) {
>+  return 1.f - 1.f / kMaxFloatS16Value <= gain_factor &&
>+         gain_factor <= 1.f + 1.f / kMaxFloatS16Value;
>+}
>+}  // namespace
>+
>+FixedGainController::FixedGainController(ApmDataDumper* apm_data_dumper)
>+    : FixedGainController(apm_data_dumper, "Agc2") {}
>+
>+FixedGainController::FixedGainController(ApmDataDumper* apm_data_dumper,
>+                                         std::string histogram_name_prefix)
>+    : apm_data_dumper_(apm_data_dumper),
>+      gain_curve_applier_(48000, apm_data_dumper_, histogram_name_prefix) {
>+  // Do update histograms.xml when adding name prefixes.
>+  RTC_DCHECK(histogram_name_prefix == "" || histogram_name_prefix == "Test" ||
>+             histogram_name_prefix == "AudioMixer" ||
>+             histogram_name_prefix == "Agc2");
>+}
>+
>+void FixedGainController::SetGain(float gain_to_apply_db) {
>+  // Changes in gain_to_apply_ cause discontinuities. We assume
>+  // gain_to_apply_ is set in the beginning of the call. If it is
>+  // frequently changed, we should add interpolation between the
>+  // values.
>+  // The gain
>+  RTC_DCHECK_LE(-50.f, gain_to_apply_db);
>+  RTC_DCHECK_LE(gain_to_apply_db, 50.f);
>+  gain_to_apply_ = DbToRatio(gain_to_apply_db);
>+  RTC_DCHECK_LT(0.f, gain_to_apply_);
>+  RTC_DLOG(LS_INFO) << "Gain to apply: " << gain_to_apply_db << " db.";
>+}
>+
>+void FixedGainController::SetSampleRate(size_t sample_rate_hz) {
>+  gain_curve_applier_.SetSampleRate(sample_rate_hz);
>+}
>+
>+void FixedGainController::Process(AudioFrameView<float> signal) {
>+  // Apply fixed digital gain. One of the
>+  // planned usages of the FGC is to only use the limiter. In that
>+  // case, the gain would be 1.0. Not doing the multiplications speeds
>+  // it up considerably. Hence the check.
>+  if (!CloseToOne(gain_to_apply_)) {
>+    for (size_t k = 0; k < signal.num_channels(); ++k) {
>+      rtc::ArrayView<float> channel_view = signal.channel(k);
>+      for (auto& sample : channel_view) {
>+        sample *= gain_to_apply_;
>+      }
>+    }
>+  }
>+
>+  // Use the limiter.
>+  gain_curve_applier_.Process(signal);
>+
>+  // Dump data for debug.
>+  const auto channel_view = signal.channel(0);
>+  apm_data_dumper_->DumpRaw("agc2_fixed_digital_gain_curve_applier",
>+                            channel_view.size(), channel_view.data());
>+  // Hard-clipping.
>+  for (size_t k = 0; k < signal.num_channels(); ++k) {
>+    rtc::ArrayView<float> channel_view = signal.channel(k);
>+    for (auto& sample : channel_view) {
>+      sample = rtc::SafeClamp(sample, kMinFloatS16Value, kMaxFloatS16Value);
>+    }
>+  }
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/fixed_gain_controller.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/fixed_gain_controller.h
>new file mode 100644
>index 00000000000..a41a13f516f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/fixed_gain_controller.h
>@@ -0,0 +1,41 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_AGC2_FIXED_GAIN_CONTROLLER_H_
>+#define MODULES_AUDIO_PROCESSING_AGC2_FIXED_GAIN_CONTROLLER_H_
>+
>+#include "modules/audio_processing/agc2/gain_curve_applier.h"
>+#include "modules/audio_processing/include/audio_frame_view.h"
>+
>+namespace webrtc {
>+class ApmDataDumper;
>+
>+class FixedGainController {
>+ public:
>+  explicit FixedGainController(ApmDataDumper* apm_data_dumper);
>+  FixedGainController(ApmDataDumper* apm_data_dumper,
>+                      std::string histogram_name_prefix);
>+
>+  void Process(AudioFrameView<float> signal);
>+
>+  // Rate and gain may be changed at any time (but not concurrently
>+  // with any other method call).
>+  void SetGain(float gain_to_apply_db);
>+  void SetSampleRate(size_t sample_rate_hz);
>+
>+ private:
>+  float gain_to_apply_ = 1.f;
>+  ApmDataDumper* apm_data_dumper_ = nullptr;
>+  GainCurveApplier gain_curve_applier_;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_AGC2_FIXED_GAIN_CONTROLLER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/fixed_gain_controller_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/fixed_gain_controller_unittest.cc
>new file mode 100644
>index 00000000000..fa68a01a4b9
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/fixed_gain_controller_unittest.cc
>@@ -0,0 +1,213 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/agc2/fixed_gain_controller.h"
>+
>+#include "absl/memory/memory.h"
>+#include "api/array_view.h"
>+#include "modules/audio_processing/agc2/agc2_testing_common.h"
>+#include "modules/audio_processing/agc2/vector_float_frame.h"
>+#include "modules/audio_processing/logging/apm_data_dumper.h"
>+#include "rtc_base/gunit.h"
>+#include "system_wrappers/include/metrics_default.h"
>+
>+namespace webrtc {
>+namespace {
>+
>+constexpr float kInputLevelLinear = 15000.f;
>+
>+constexpr float kGainToApplyDb = 15.f;
>+
>+float RunFixedGainControllerWithConstantInput(FixedGainController* fixed_gc,
>+                                              const float input_level,
>+                                              const size_t num_frames,
>+                                              const int sample_rate) {
>+  // Give time to the level etimator to converge.
>+  for (size_t i = 0; i < num_frames; ++i) {
>+    VectorFloatFrame vectors_with_float_frame(
>+        1, rtc::CheckedDivExact(sample_rate, 100), input_level);
>+    fixed_gc->Process(vectors_with_float_frame.float_frame_view());
>+  }
>+
>+  // Process the last frame with constant input level.
>+  VectorFloatFrame vectors_with_float_frame_last(
>+      1, rtc::CheckedDivExact(sample_rate, 100), input_level);
>+  fixed_gc->Process(vectors_with_float_frame_last.float_frame_view());
>+
>+  // Return the last sample from the last processed frame.
>+  const auto channel =
>+      vectors_with_float_frame_last.float_frame_view().channel(0);
>+  return channel[channel.size() - 1];
>+}
>+ApmDataDumper test_data_dumper(0);
>+
>+std::unique_ptr<FixedGainController> CreateFixedGainController(
>+    float gain_to_apply,
>+    size_t rate,
>+    std::string histogram_name_prefix) {
>+  std::unique_ptr<FixedGainController> fgc =
>+      absl::make_unique<FixedGainController>(&test_data_dumper,
>+                                             histogram_name_prefix);
>+  fgc->SetGain(gain_to_apply);
>+  fgc->SetSampleRate(rate);
>+  return fgc;
>+}
>+
>+std::unique_ptr<FixedGainController> CreateFixedGainController(
>+    float gain_to_apply,
>+    size_t rate) {
>+  return CreateFixedGainController(gain_to_apply, rate, "");
>+}
>+
>+}  // namespace
>+
>+TEST(AutomaticGainController2FixedDigital, CreateUse) {
>+  const int kSampleRate = 44000;
>+  std::unique_ptr<FixedGainController> fixed_gc =
>+      CreateFixedGainController(kGainToApplyDb, kSampleRate);
>+  VectorFloatFrame vectors_with_float_frame(
>+      1, rtc::CheckedDivExact(kSampleRate, 100), kInputLevelLinear);
>+  auto float_frame = vectors_with_float_frame.float_frame_view();
>+  fixed_gc->Process(float_frame);
>+  const auto channel = float_frame.channel(0);
>+  EXPECT_LT(kInputLevelLinear, channel[0]);
>+}
>+
>+TEST(AutomaticGainController2FixedDigital, CheckSaturationBehaviorWithLimiter) {
>+  const float kInputLevel = 32767.f;
>+  const size_t kNumFrames = 5;
>+  const size_t kSampleRate = 42000;
>+
>+  const auto gains_no_saturation =
>+      test::LinSpace(0.1, test::kLimiterMaxInputLevelDbFs - 0.01, 10);
>+  for (const auto gain_db : gains_no_saturation) {
>+    // Since |test::kLimiterMaxInputLevelDbFs| > |gain_db|, the
>+    // limiter will not saturate the signal.
>+    std::unique_ptr<FixedGainController> fixed_gc_no_saturation =
>+        CreateFixedGainController(gain_db, kSampleRate);
>+
>+    // Saturation not expected.
>+    SCOPED_TRACE(std::to_string(gain_db));
>+    EXPECT_LT(
>+        RunFixedGainControllerWithConstantInput(
>+            fixed_gc_no_saturation.get(), kInputLevel, kNumFrames, kSampleRate),
>+        32767.f);
>+  }
>+
>+  const auto gains_saturation =
>+      test::LinSpace(test::kLimiterMaxInputLevelDbFs + 0.01, 10, 10);
>+  for (const auto gain_db : gains_saturation) {
>+    // Since |test::kLimiterMaxInputLevelDbFs| < |gain|, the limiter
>+    // will saturate the signal.
>+    std::unique_ptr<FixedGainController> fixed_gc_saturation =
>+        CreateFixedGainController(gain_db, kSampleRate);
>+
>+    // Saturation expected.
>+    SCOPED_TRACE(std::to_string(gain_db));
>+    EXPECT_FLOAT_EQ(
>+        RunFixedGainControllerWithConstantInput(
>+            fixed_gc_saturation.get(), kInputLevel, kNumFrames, kSampleRate),
>+        32767.f);
>+  }
>+}
>+
>+TEST(AutomaticGainController2FixedDigital,
>+     CheckSaturationBehaviorWithLimiterSingleSample) {
>+  const float kInputLevel = 32767.f;
>+  const size_t kNumFrames = 5;
>+  const size_t kSampleRate = 8000;
>+
>+  const auto gains_no_saturation =
>+      test::LinSpace(0.1, test::kLimiterMaxInputLevelDbFs - 0.01, 10);
>+  for (const auto gain_db : gains_no_saturation) {
>+    // Since |gain| > |test::kLimiterMaxInputLevelDbFs|, the limiter will
>+    // not saturate the signal.
>+    std::unique_ptr<FixedGainController> fixed_gc_no_saturation =
>+        CreateFixedGainController(gain_db, kSampleRate);
>+
>+    // Saturation not expected.
>+    SCOPED_TRACE(std::to_string(gain_db));
>+    EXPECT_LT(
>+        RunFixedGainControllerWithConstantInput(
>+            fixed_gc_no_saturation.get(), kInputLevel, kNumFrames, kSampleRate),
>+        32767.f);
>+  }
>+
>+  const auto gains_saturation =
>+      test::LinSpace(test::kLimiterMaxInputLevelDbFs + 0.01, 10, 10);
>+  for (const auto gain_db : gains_saturation) {
>+    // Singe |gain| < |test::kLimiterMaxInputLevelDbFs|, the limiter will
>+    // saturate the signal.
>+    std::unique_ptr<FixedGainController> fixed_gc_saturation =
>+        CreateFixedGainController(gain_db, kSampleRate);
>+
>+    // Saturation expected.
>+    SCOPED_TRACE(std::to_string(gain_db));
>+    EXPECT_FLOAT_EQ(
>+        RunFixedGainControllerWithConstantInput(
>+            fixed_gc_saturation.get(), kInputLevel, kNumFrames, kSampleRate),
>+        32767.f);
>+  }
>+}
>+
>+TEST(AutomaticGainController2FixedDigital, GainShouldChangeOnSetGain) {
>+  constexpr float kInputLevel = 1000.f;
>+  constexpr size_t kNumFrames = 5;
>+  constexpr size_t kSampleRate = 8000;
>+  constexpr float kGainDbNoChange = 0.f;
>+  constexpr float kGainDbFactor10 = 20.f;
>+
>+  std::unique_ptr<FixedGainController> fixed_gc_no_saturation =
>+      CreateFixedGainController(kGainDbNoChange, kSampleRate);
>+
>+  // Signal level is unchanged with 0 db gain.
>+  EXPECT_FLOAT_EQ(
>+      RunFixedGainControllerWithConstantInput(
>+          fixed_gc_no_saturation.get(), kInputLevel, kNumFrames, kSampleRate),
>+      kInputLevel);
>+
>+  fixed_gc_no_saturation->SetGain(kGainDbFactor10);
>+
>+  // +20db should increase signal by a factor of 10.
>+  EXPECT_FLOAT_EQ(
>+      RunFixedGainControllerWithConstantInput(
>+          fixed_gc_no_saturation.get(), kInputLevel, kNumFrames, kSampleRate),
>+      kInputLevel * 10);
>+}
>+
>+TEST(AutomaticGainController2FixedDigital, RegionHistogramIsUpdated) {
>+  constexpr size_t kSampleRate = 8000;
>+  constexpr float kGainDb = 0.f;
>+  constexpr float kInputLevel = 1000.f;
>+  constexpr size_t kNumFrames = 5;
>+
>+  metrics::Reset();
>+
>+  std::unique_ptr<FixedGainController> fixed_gc_no_saturation =
>+      CreateFixedGainController(kGainDb, kSampleRate, "Test");
>+
>+  static_cast<void>(RunFixedGainControllerWithConstantInput(
>+      fixed_gc_no_saturation.get(), kInputLevel, kNumFrames, kSampleRate));
>+
>+  // Destroying FixedGainController should cause the last limiter region to be
>+  // logged.
>+  fixed_gc_no_saturation.reset();
>+
>+  EXPECT_EQ(1, metrics::NumSamples(
>+                   "WebRTC.Audio.Test.FixedDigitalGainCurveRegion.Identity"));
>+  EXPECT_EQ(0, metrics::NumSamples(
>+                   "WebRTC.Audio.Test.FixedDigitalGainCurveRegion.Knee"));
>+  EXPECT_EQ(0, metrics::NumSamples(
>+                   "WebRTC.Audio.Test.FixedDigitalGainCurveRegion.Limiter"));
>+  EXPECT_EQ(0, metrics::NumSamples(
>+                   "WebRTC.Audio.Test.FixedDigitalGainCurveRegion.Saturation"));
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/gain_applier.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/gain_applier.cc
>new file mode 100644
>index 00000000000..38eb1de03f9
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/gain_applier.cc
>@@ -0,0 +1,101 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/agc2/gain_applier.h"
>+
>+#include "modules/audio_processing/agc2/agc2_common.h"
>+#include "rtc_base/numerics/safe_minmax.h"
>+
>+namespace webrtc {
>+namespace {
>+
>+// Returns true when the gain factor is so close to 1 that it would
>+// not affect int16 samples.
>+bool GainCloseToOne(float gain_factor) {
>+  return 1.f - 1.f / kMaxFloatS16Value <= gain_factor &&
>+         gain_factor <= 1.f + 1.f / kMaxFloatS16Value;
>+}
>+
>+void ClipSignal(AudioFrameView<float> signal) {
>+  for (size_t k = 0; k < signal.num_channels(); ++k) {
>+    rtc::ArrayView<float> channel_view = signal.channel(k);
>+    for (auto& sample : channel_view) {
>+      sample = rtc::SafeClamp(sample, kMinFloatS16Value, kMaxFloatS16Value);
>+    }
>+  }
>+}
>+
>+void ApplyGainWithRamping(float last_gain_linear,
>+                          float gain_at_end_of_frame_linear,
>+                          float inverse_samples_per_channel,
>+                          AudioFrameView<float> float_frame) {
>+  // Do not modify the signal.
>+  if (last_gain_linear == gain_at_end_of_frame_linear &&
>+      GainCloseToOne(gain_at_end_of_frame_linear)) {
>+    return;
>+  }
>+
>+  // Gain is constant and different from 1.
>+  if (last_gain_linear == gain_at_end_of_frame_linear) {
>+    for (size_t k = 0; k < float_frame.num_channels(); ++k) {
>+      rtc::ArrayView<float> channel_view = float_frame.channel(k);
>+      for (auto& sample : channel_view) {
>+        sample *= gain_at_end_of_frame_linear;
>+      }
>+    }
>+    return;
>+  }
>+
>+  // The gain changes. We have to change slowly to avoid discontinuities.
>+  const float increment = (gain_at_end_of_frame_linear - last_gain_linear) *
>+                          inverse_samples_per_channel;
>+  float gain = last_gain_linear;
>+  for (size_t i = 0; i < float_frame.samples_per_channel(); ++i) {
>+    for (size_t ch = 0; ch < float_frame.num_channels(); ++ch) {
>+      float_frame.channel(ch)[i] *= gain;
>+    }
>+    gain += increment;
>+  }
>+}
>+
>+}  // namespace
>+
>+GainApplier::GainApplier(bool hard_clip_samples, float initial_gain_factor)
>+    : hard_clip_samples_(hard_clip_samples),
>+      last_gain_factor_(initial_gain_factor),
>+      current_gain_factor_(initial_gain_factor) {}
>+
>+void GainApplier::ApplyGain(AudioFrameView<float> signal) {
>+  if (static_cast<int>(signal.samples_per_channel()) != samples_per_channel_) {
>+    Initialize(signal.samples_per_channel());
>+  }
>+
>+  ApplyGainWithRamping(last_gain_factor_, current_gain_factor_,
>+                       inverse_samples_per_channel_, signal);
>+
>+  last_gain_factor_ = current_gain_factor_;
>+
>+  if (hard_clip_samples_) {
>+    ClipSignal(signal);
>+  }
>+}
>+
>+void GainApplier::SetGainFactor(float gain_factor) {
>+  RTC_DCHECK_GT(gain_factor, 0.f);
>+  current_gain_factor_ = gain_factor;
>+}
>+
>+void GainApplier::Initialize(size_t samples_per_channel) {
>+  RTC_DCHECK_GT(samples_per_channel, 0);
>+  samples_per_channel_ = static_cast<int>(samples_per_channel);
>+  inverse_samples_per_channel_ = 1.f / samples_per_channel_;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/gain_applier.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/gain_applier.h
>new file mode 100644
>index 00000000000..a4f56cee4be
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/gain_applier.h
>@@ -0,0 +1,41 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_AGC2_GAIN_APPLIER_H_
>+#define MODULES_AUDIO_PROCESSING_AGC2_GAIN_APPLIER_H_
>+
>+#include "modules/audio_processing/include/audio_frame_view.h"
>+
>+namespace webrtc {
>+class GainApplier {
>+ public:
>+  GainApplier(bool hard_clip_samples, float initial_gain_factor);
>+
>+  void ApplyGain(AudioFrameView<float> signal);
>+  void SetGainFactor(float gain_factor);
>+
>+ private:
>+  void Initialize(size_t samples_per_channel);
>+
>+  // Whether to clip samples after gain is applied. If 'true', result
>+  // will fit in FloatS16 range.
>+  const bool hard_clip_samples_;
>+  float last_gain_factor_;
>+
>+  // If this value is not equal to 'last_gain_factor', gain will be
>+  // ramped from 'last_gain_factor_' to this value during the next
>+  // 'ApplyGain'.
>+  float current_gain_factor_;
>+  int samples_per_channel_ = -1;
>+  float inverse_samples_per_channel_ = -1.f;
>+};
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_AGC2_GAIN_APPLIER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/gain_applier_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/gain_applier_unittest.cc
>new file mode 100644
>index 00000000000..3296345e62e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/gain_applier_unittest.cc
>@@ -0,0 +1,93 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/agc2/gain_applier.h"
>+
>+#include <math.h>
>+
>+#include <algorithm>
>+#include <limits>
>+
>+#include "modules/audio_processing/agc2/vector_float_frame.h"
>+#include "rtc_base/gunit.h"
>+
>+namespace webrtc {
>+TEST(AutomaticGainController2GainApplier, InitialGainIsRespected) {
>+  constexpr float initial_signal_level = 123.f;
>+  constexpr float gain_factor = 10.f;
>+  VectorFloatFrame fake_audio(1, 1, initial_signal_level);
>+  GainApplier gain_applier(true, gain_factor);
>+
>+  gain_applier.ApplyGain(fake_audio.float_frame_view());
>+  EXPECT_NEAR(fake_audio.float_frame_view().channel(0)[0],
>+              initial_signal_level * gain_factor, 0.1f);
>+}
>+
>+TEST(AutomaticGainController2GainApplier, ClippingIsDone) {
>+  constexpr float initial_signal_level = 30000.f;
>+  constexpr float gain_factor = 10.f;
>+  VectorFloatFrame fake_audio(1, 1, initial_signal_level);
>+  GainApplier gain_applier(true, gain_factor);
>+
>+  gain_applier.ApplyGain(fake_audio.float_frame_view());
>+  EXPECT_NEAR(fake_audio.float_frame_view().channel(0)[0],
>+              std::numeric_limits<int16_t>::max(), 0.1f);
>+}
>+
>+TEST(AutomaticGainController2GainApplier, ClippingIsNotDone) {
>+  constexpr float initial_signal_level = 30000.f;
>+  constexpr float gain_factor = 10.f;
>+  VectorFloatFrame fake_audio(1, 1, initial_signal_level);
>+  GainApplier gain_applier(false, gain_factor);
>+
>+  gain_applier.ApplyGain(fake_audio.float_frame_view());
>+
>+  EXPECT_NEAR(fake_audio.float_frame_view().channel(0)[0],
>+              initial_signal_level * gain_factor, 0.1f);
>+}
>+
>+TEST(AutomaticGainController2GainApplier, RampingIsDone) {
>+  constexpr float initial_signal_level = 30000.f;
>+  constexpr float initial_gain_factor = 1.f;
>+  constexpr float target_gain_factor = 0.5f;
>+  constexpr int num_channels = 3;
>+  constexpr int samples_per_channel = 4;
>+  VectorFloatFrame fake_audio(num_channels, samples_per_channel,
>+                              initial_signal_level);
>+  GainApplier gain_applier(false, initial_gain_factor);
>+
>+  gain_applier.SetGainFactor(target_gain_factor);
>+  gain_applier.ApplyGain(fake_audio.float_frame_view());
>+
>+  // The maximal gain change should be close to that in linear interpolation.
>+  for (size_t channel = 0; channel < num_channels; ++channel) {
>+    float max_signal_change = 0.f;
>+    float last_signal_level = initial_signal_level;
>+    for (const auto sample : fake_audio.float_frame_view().channel(channel)) {
>+      const float current_change = fabs(last_signal_level - sample);
>+      max_signal_change = std::max(max_signal_change, current_change);
>+      last_signal_level = sample;
>+    }
>+    const float total_gain_change =
>+        fabs((initial_gain_factor - target_gain_factor) * initial_signal_level);
>+    EXPECT_NEAR(max_signal_change, total_gain_change / samples_per_channel,
>+                0.1f);
>+  }
>+
>+  // Next frame should have the desired level.
>+  VectorFloatFrame next_fake_audio_frame(num_channels, samples_per_channel,
>+                                         initial_signal_level);
>+  gain_applier.ApplyGain(next_fake_audio_frame.float_frame_view());
>+
>+  // The last sample should have the new gain.
>+  EXPECT_NEAR(next_fake_audio_frame.float_frame_view().channel(0)[0],
>+              initial_signal_level * target_gain_factor, 0.1f);
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/gain_curve_applier.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/gain_curve_applier.cc
>new file mode 100644
>index 00000000000..7c1410b232b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/gain_curve_applier.cc
>@@ -0,0 +1,133 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/agc2/gain_curve_applier.h"
>+
>+#include <algorithm>
>+#include <array>
>+#include <cmath>
>+
>+#include "api/array_view.h"
>+#include "modules/audio_processing/logging/apm_data_dumper.h"
>+#include "rtc_base/checks.h"
>+
>+namespace webrtc {
>+namespace {
>+
>+// This constant affects the way scaling factors are interpolated for the first
>+// sub-frame of a frame. Only in the case in which the first sub-frame has an
>+// estimated level which is greater than the that of the previous analyzed
>+// sub-frame, linear interpolation is replaced with a power function which
>+// reduces the chances of over-shooting (and hence saturation), however reducing
>+// the fixed gain effectiveness.
>+constexpr float kAttackFirstSubframeInterpolationPower = 8.f;
>+
>+void InterpolateFirstSubframe(float last_factor,
>+                              float current_factor,
>+                              rtc::ArrayView<float> subframe) {
>+  const auto n = subframe.size();
>+  constexpr auto p = kAttackFirstSubframeInterpolationPower;
>+  for (size_t i = 0; i < n; ++i) {
>+    subframe[i] = std::pow(1.f - i / n, p) * (last_factor - current_factor) +
>+                  current_factor;
>+  }
>+}
>+
>+void ComputePerSampleSubframeFactors(
>+    const std::array<float, kSubFramesInFrame + 1>& scaling_factors,
>+    size_t samples_per_channel,
>+    rtc::ArrayView<float> per_sample_scaling_factors) {
>+  const size_t num_subframes = scaling_factors.size() - 1;
>+  const size_t subframe_size =
>+      rtc::CheckedDivExact(samples_per_channel, num_subframes);
>+
>+  // Handle first sub-frame differently in case of attack.
>+  const bool is_attack = scaling_factors[0] > scaling_factors[1];
>+  if (is_attack) {
>+    InterpolateFirstSubframe(
>+        scaling_factors[0], scaling_factors[1],
>+        rtc::ArrayView<float>(
>+            per_sample_scaling_factors.subview(0, subframe_size)));
>+  }
>+
>+  for (size_t i = is_attack ? 1 : 0; i < num_subframes; ++i) {
>+    const size_t subframe_start = i * subframe_size;
>+    const float scaling_start = scaling_factors[i];
>+    const float scaling_end = scaling_factors[i + 1];
>+    const float scaling_diff = (scaling_end - scaling_start) / subframe_size;
>+    for (size_t j = 0; j < subframe_size; ++j) {
>+      per_sample_scaling_factors[subframe_start + j] =
>+          scaling_start + scaling_diff * j;
>+    }
>+  }
>+}
>+
>+void ScaleSamples(rtc::ArrayView<const float> per_sample_scaling_factors,
>+                  AudioFrameView<float> signal) {
>+  const size_t samples_per_channel = signal.samples_per_channel();
>+  RTC_DCHECK_EQ(samples_per_channel, per_sample_scaling_factors.size());
>+  for (size_t i = 0; i < signal.num_channels(); ++i) {
>+    auto channel = signal.channel(i);
>+    for (size_t j = 0; j < samples_per_channel; ++j) {
>+      channel[j] *= per_sample_scaling_factors[j];
>+    }
>+  }
>+}
>+
>+}  // namespace
>+
>+GainCurveApplier::GainCurveApplier(size_t sample_rate_hz,
>+                                   ApmDataDumper* apm_data_dumper,
>+                                   std::string histogram_name)
>+    : interp_gain_curve_(apm_data_dumper, histogram_name),
>+      level_estimator_(sample_rate_hz, apm_data_dumper),
>+      apm_data_dumper_(apm_data_dumper) {}
>+
>+GainCurveApplier::~GainCurveApplier() = default;
>+
>+void GainCurveApplier::Process(AudioFrameView<float> signal) {
>+  const auto level_estimate = level_estimator_.ComputeLevel(signal);
>+
>+  RTC_DCHECK_EQ(level_estimate.size() + 1, scaling_factors_.size());
>+  scaling_factors_[0] = last_scaling_factor_;
>+  std::transform(level_estimate.begin(), level_estimate.end(),
>+                 scaling_factors_.begin() + 1, [this](float x) {
>+                   return interp_gain_curve_.LookUpGainToApply(x);
>+                 });
>+
>+  const size_t samples_per_channel = signal.samples_per_channel();
>+  RTC_DCHECK_LE(samples_per_channel, kMaximalNumberOfSamplesPerChannel);
>+
>+  auto per_sample_scaling_factors = rtc::ArrayView<float>(
>+      &per_sample_scaling_factors_[0], samples_per_channel);
>+  ComputePerSampleSubframeFactors(scaling_factors_, samples_per_channel,
>+                                  per_sample_scaling_factors);
>+  ScaleSamples(per_sample_scaling_factors, signal);
>+
>+  last_scaling_factor_ = scaling_factors_.back();
>+
>+  // Dump data for debug.
>+  apm_data_dumper_->DumpRaw("agc2_gain_curve_applier_scaling_factors",
>+                            samples_per_channel,
>+                            per_sample_scaling_factors_.data());
>+}
>+
>+InterpolatedGainCurve::Stats GainCurveApplier::GetGainCurveStats() const {
>+  return interp_gain_curve_.get_stats();
>+}
>+
>+void GainCurveApplier::SetSampleRate(size_t sample_rate_hz) {
>+  level_estimator_.SetSampleRate(sample_rate_hz);
>+  // Check that per_sample_scaling_factors_ is large enough.
>+  RTC_DCHECK_LE(sample_rate_hz,
>+                kMaximalNumberOfSamplesPerChannel * 1000 / kFrameDurationMs);
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/gain_curve_applier.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/gain_curve_applier.h
>new file mode 100644
>index 00000000000..f546f6325d3
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/gain_curve_applier.h
>@@ -0,0 +1,58 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_AGC2_GAIN_CURVE_APPLIER_H_
>+#define MODULES_AUDIO_PROCESSING_AGC2_GAIN_CURVE_APPLIER_H_
>+
>+#include <vector>
>+
>+#include "modules/audio_processing/agc2/fixed_digital_level_estimator.h"
>+#include "modules/audio_processing/agc2/interpolated_gain_curve.h"
>+#include "modules/audio_processing/include/audio_frame_view.h"
>+#include "rtc_base/constructormagic.h"
>+
>+namespace webrtc {
>+class ApmDataDumper;
>+
>+class GainCurveApplier {
>+ public:
>+  GainCurveApplier(size_t sample_rate_hz,
>+                   ApmDataDumper* apm_data_dumper,
>+                   std::string histogram_name_prefix);
>+
>+  ~GainCurveApplier();
>+
>+  void Process(AudioFrameView<float> signal);
>+  InterpolatedGainCurve::Stats GetGainCurveStats() const;
>+
>+  // Supported rates must be
>+  // * supported by FixedDigitalLevelEstimator
>+  // * below kMaximalNumberOfSamplesPerChannel*1000/kFrameDurationMs
>+  //   so that samples_per_channel fit in the
>+  //   per_sample_scaling_factors_ array.
>+  void SetSampleRate(size_t sample_rate_hz);
>+
>+ private:
>+  const InterpolatedGainCurve interp_gain_curve_;
>+  FixedDigitalLevelEstimator level_estimator_;
>+  ApmDataDumper* const apm_data_dumper_ = nullptr;
>+
>+  // Work array containing the sub-frame scaling factors to be interpolated.
>+  std::array<float, kSubFramesInFrame + 1> scaling_factors_ = {};
>+  std::array<float, kMaximalNumberOfSamplesPerChannel>
>+      per_sample_scaling_factors_ = {};
>+  float last_scaling_factor_ = 1.f;
>+
>+  RTC_DISALLOW_COPY_AND_ASSIGN(GainCurveApplier);
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_AGC2_GAIN_CURVE_APPLIER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/gain_curve_applier_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/gain_curve_applier_unittest.cc
>new file mode 100644
>index 00000000000..0f75f62c407
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/gain_curve_applier_unittest.cc
>@@ -0,0 +1,60 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/agc2/gain_curve_applier.h"
>+
>+#include "common_audio/include/audio_util.h"
>+#include "modules/audio_processing/agc2/agc2_common.h"
>+#include "modules/audio_processing/agc2/agc2_testing_common.h"
>+#include "modules/audio_processing/agc2/vector_float_frame.h"
>+#include "modules/audio_processing/logging/apm_data_dumper.h"
>+#include "rtc_base/gunit.h"
>+
>+namespace webrtc {
>+
>+TEST(GainCurveApplier, GainCurveApplierShouldConstructAndRun) {
>+  const int sample_rate_hz = 48000;
>+  ApmDataDumper apm_data_dumper(0);
>+
>+  GainCurveApplier gain_curve_applier(sample_rate_hz, &apm_data_dumper, "");
>+
>+  VectorFloatFrame vectors_with_float_frame(1, sample_rate_hz / 100,
>+                                            kMaxAbsFloatS16Value);
>+  gain_curve_applier.Process(vectors_with_float_frame.float_frame_view());
>+}
>+
>+TEST(GainCurveApplier, OutputVolumeAboveThreshold) {
>+  const int sample_rate_hz = 48000;
>+  const float input_level =
>+      (kMaxAbsFloatS16Value + DbfsToFloatS16(test::kLimiterMaxInputLevelDbFs)) /
>+      2.f;
>+  ApmDataDumper apm_data_dumper(0);
>+
>+  GainCurveApplier gain_curve_applier(sample_rate_hz, &apm_data_dumper, "");
>+
>+  // Give the level estimator time to adapt.
>+  for (int i = 0; i < 5; ++i) {
>+    VectorFloatFrame vectors_with_float_frame(1, sample_rate_hz / 100,
>+                                              input_level);
>+    gain_curve_applier.Process(vectors_with_float_frame.float_frame_view());
>+  }
>+
>+  VectorFloatFrame vectors_with_float_frame(1, sample_rate_hz / 100,
>+                                            input_level);
>+  gain_curve_applier.Process(vectors_with_float_frame.float_frame_view());
>+  rtc::ArrayView<const float> channel =
>+      vectors_with_float_frame.float_frame_view().channel(0);
>+
>+  for (const auto& sample : channel) {
>+    EXPECT_LT(0.9f * kMaxAbsFloatS16Value, sample);
>+  }
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/interpolated_gain_curve.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/interpolated_gain_curve.cc
>new file mode 100644
>index 00000000000..73e6a8ecc36
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/interpolated_gain_curve.cc
>@@ -0,0 +1,191 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/agc2/interpolated_gain_curve.h"
>+
>+#include "modules/audio_processing/agc2/agc2_common.h"
>+#include "modules/audio_processing/logging/apm_data_dumper.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/logging.h"
>+
>+namespace webrtc {
>+
>+constexpr std::array<float, kInterpolatedGainCurveTotalPoints>
>+    InterpolatedGainCurve::approximation_params_x_;
>+
>+constexpr std::array<float, kInterpolatedGainCurveTotalPoints>
>+    InterpolatedGainCurve::approximation_params_m_;
>+
>+constexpr std::array<float, kInterpolatedGainCurveTotalPoints>
>+    InterpolatedGainCurve::approximation_params_q_;
>+
>+InterpolatedGainCurve::InterpolatedGainCurve(ApmDataDumper* apm_data_dumper,
>+                                             std::string histogram_name_prefix)
>+    : region_logger_("WebRTC.Audio." + histogram_name_prefix +
>+                         ".FixedDigitalGainCurveRegion.Identity",
>+                     "WebRTC.Audio." + histogram_name_prefix +
>+                         ".FixedDigitalGainCurveRegion.Knee",
>+                     "WebRTC.Audio." + histogram_name_prefix +
>+                         ".FixedDigitalGainCurveRegion.Limiter",
>+                     "WebRTC.Audio." + histogram_name_prefix +
>+                         ".FixedDigitalGainCurveRegion.Saturation"),
>+      apm_data_dumper_(apm_data_dumper) {}
>+
>+InterpolatedGainCurve::~InterpolatedGainCurve() {
>+  if (stats_.available) {
>+    RTC_DCHECK(apm_data_dumper_);
>+    apm_data_dumper_->DumpRaw("agc2_interp_gain_curve_lookups_identity",
>+                              stats_.look_ups_identity_region);
>+    apm_data_dumper_->DumpRaw("agc2_interp_gain_curve_lookups_knee",
>+                              stats_.look_ups_knee_region);
>+    apm_data_dumper_->DumpRaw("agc2_interp_gain_curve_lookups_limiter",
>+                              stats_.look_ups_limiter_region);
>+    apm_data_dumper_->DumpRaw("agc2_interp_gain_curve_lookups_saturation",
>+                              stats_.look_ups_saturation_region);
>+    region_logger_.LogRegionStats(stats_);
>+  }
>+}
>+
>+InterpolatedGainCurve::RegionLogger::RegionLogger(
>+    std::string identity_histogram_name,
>+    std::string knee_histogram_name,
>+    std::string limiter_histogram_name,
>+    std::string saturation_histogram_name)
>+    : identity_histogram(
>+          metrics::HistogramFactoryGetCounts(identity_histogram_name,
>+                                             1,
>+                                             10000,
>+                                             50)),
>+      knee_histogram(metrics::HistogramFactoryGetCounts(knee_histogram_name,
>+                                                        1,
>+                                                        10000,
>+                                                        50)),
>+      limiter_histogram(
>+          metrics::HistogramFactoryGetCounts(limiter_histogram_name,
>+                                             1,
>+                                             10000,
>+                                             50)),
>+      saturation_histogram(
>+          metrics::HistogramFactoryGetCounts(saturation_histogram_name,
>+                                             1,
>+                                             10000,
>+                                             50)) {}
>+
>+InterpolatedGainCurve::RegionLogger::~RegionLogger() = default;
>+
>+void InterpolatedGainCurve::RegionLogger::LogRegionStats(
>+    const InterpolatedGainCurve::Stats& stats) const {
>+  using Region = InterpolatedGainCurve::GainCurveRegion;
>+  const int duration_s =
>+      stats.region_duration_frames / (1000 / kFrameDurationMs);
>+
>+  switch (stats.region) {
>+    case Region::kIdentity: {
>+      if (identity_histogram) {
>+        metrics::HistogramAdd(identity_histogram, duration_s);
>+      }
>+      break;
>+    }
>+    case Region::kKnee: {
>+      if (knee_histogram) {
>+        metrics::HistogramAdd(knee_histogram, duration_s);
>+      }
>+      break;
>+    }
>+    case Region::kLimiter: {
>+      if (limiter_histogram) {
>+        metrics::HistogramAdd(limiter_histogram, duration_s);
>+      }
>+      break;
>+    }
>+    case Region::kSaturation: {
>+      if (saturation_histogram) {
>+        metrics::HistogramAdd(saturation_histogram, duration_s);
>+      }
>+      break;
>+    }
>+    default: { RTC_NOTREACHED(); }
>+  }
>+}
>+
>+void InterpolatedGainCurve::UpdateStats(float input_level) const {
>+  stats_.available = true;
>+
>+  GainCurveRegion region;
>+
>+  if (input_level < approximation_params_x_[0]) {
>+    stats_.look_ups_identity_region++;
>+    region = GainCurveRegion::kIdentity;
>+  } else if (input_level <
>+             approximation_params_x_[kInterpolatedGainCurveKneePoints - 1]) {
>+    stats_.look_ups_knee_region++;
>+    region = GainCurveRegion::kKnee;
>+  } else if (input_level < kMaxInputLevelLinear) {
>+    stats_.look_ups_limiter_region++;
>+    region = GainCurveRegion::kLimiter;
>+  } else {
>+    stats_.look_ups_saturation_region++;
>+    region = GainCurveRegion::kSaturation;
>+  }
>+
>+  if (region == stats_.region) {
>+    ++stats_.region_duration_frames;
>+  } else {
>+    region_logger_.LogRegionStats(stats_);
>+
>+    stats_.region_duration_frames = 0;
>+    stats_.region = region;
>+  }
>+}
>+
>+// Looks up a gain to apply given a non-negative input level.
>+// The cost of this operation depends on the region in which |input_level|
>+// falls.
>+// For the identity and the saturation regions the cost is O(1).
>+// For the other regions, namely knee and limiter, the cost is
>+// O(2 + log2(|LightkInterpolatedGainCurveTotalPoints|), plus O(1) for the
>+// linear interpolation (one product and one sum).
>+float InterpolatedGainCurve::LookUpGainToApply(float input_level) const {
>+  UpdateStats(input_level);
>+
>+  if (input_level <= approximation_params_x_[0]) {
>+    // Identity region.
>+    return 1.0f;
>+  }
>+
>+  if (input_level >= kMaxInputLevelLinear) {
>+    // Saturating lower bound. The saturing samples exactly hit the clipping
>+    // level. This method achieves has the lowest harmonic distorsion, but it
>+    // may reduce the amplitude of the non-saturating samples too much.
>+    return 32768.f / input_level;
>+  }
>+
>+  // Knee and limiter regions; find the linear piece index. Spelling
>+  // out the complete type was the only way to silence both the clang
>+  // plugin and the windows compilers.
>+  std::array<float, kInterpolatedGainCurveTotalPoints>::const_iterator it =
>+      std::lower_bound(approximation_params_x_.begin(),
>+                       approximation_params_x_.end(), input_level);
>+  const size_t index = std::distance(approximation_params_x_.begin(), it) - 1;
>+  RTC_DCHECK_LE(0, index);
>+  RTC_DCHECK_LT(index, approximation_params_m_.size());
>+  RTC_DCHECK_LE(approximation_params_x_[index], input_level);
>+  if (index < approximation_params_m_.size() - 1) {
>+    RTC_DCHECK_LE(input_level, approximation_params_x_[index + 1]);
>+  }
>+
>+  // Piece-wise linear interploation.
>+  const float gain = approximation_params_m_[index] * input_level +
>+                     approximation_params_q_[index];
>+  RTC_DCHECK_LE(0.f, gain);
>+  return gain;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/interpolated_gain_curve.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/interpolated_gain_curve.h
>new file mode 100644
>index 00000000000..68d45322387
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/interpolated_gain_curve.h
>@@ -0,0 +1,153 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_AGC2_INTERPOLATED_GAIN_CURVE_H_
>+#define MODULES_AUDIO_PROCESSING_AGC2_INTERPOLATED_GAIN_CURVE_H_
>+
>+#include <array>
>+#include <string>
>+
>+#include "modules/audio_processing/agc2/agc2_common.h"
>+
>+#include "rtc_base/constructormagic.h"
>+#include "rtc_base/gtest_prod_util.h"
>+#include "system_wrappers/include/metrics.h"
>+
>+namespace webrtc {
>+
>+class ApmDataDumper;
>+
>+constexpr float kInputLevelScalingFactor = 32768.0f;
>+
>+// Defined as DbfsToLinear(kLimiterMaxInputLevelDbFs)
>+constexpr float kMaxInputLevelLinear = static_cast<float>(36766.300710566735);
>+
>+// Interpolated gain curve using under-approximation to avoid saturation.
>+//
>+// The goal of this class is allowing fast look ups to get an accurate
>+// estimates of the gain to apply given an estimated input level.
>+class InterpolatedGainCurve {
>+ public:
>+  enum class GainCurveRegion {
>+    kIdentity = 0,
>+    kKnee = 1,
>+    kLimiter = 2,
>+    kSaturation = 3
>+  };
>+
>+  struct Stats {
>+    // Region in which the output level equals the input one.
>+    size_t look_ups_identity_region = 0;
>+    // Smoothing between the identity and the limiter regions.
>+    size_t look_ups_knee_region = 0;
>+    // Limiter region in which the output and input levels are linearly related.
>+    size_t look_ups_limiter_region = 0;
>+    // Region in which saturation may occur since the input level is beyond the
>+    // maximum expected by the limiter.
>+    size_t look_ups_saturation_region = 0;
>+    // True if stats have been populated.
>+    bool available = false;
>+
>+    // The current region, and for how many frames the level has been
>+    // in that region.
>+    GainCurveRegion region = GainCurveRegion::kIdentity;
>+    int64_t region_duration_frames = 0;
>+  };
>+
>+  InterpolatedGainCurve(ApmDataDumper* apm_data_dumper,
>+                        std::string histogram_name_prefix);
>+  ~InterpolatedGainCurve();
>+
>+  Stats get_stats() const { return stats_; }
>+
>+  // Given a non-negative input level (linear scale), a scalar factor to apply
>+  // to a sub-frame is returned.
>+  // Levels above kLimiterMaxInputLevelDbFs will be reduced to 0 dBFS
>+  // after applying this gain
>+  float LookUpGainToApply(float input_level) const;
>+
>+ private:
>+  // For comparing 'approximation_params_*_' with ones computed by
>+  // ComputeInterpolatedGainCurve.
>+  FRIEND_TEST_ALL_PREFIXES(AutomaticGainController2InterpolatedGainCurve,
>+                           CheckApproximationParams);
>+
>+  struct RegionLogger {
>+    metrics::Histogram* identity_histogram;
>+    metrics::Histogram* knee_histogram;
>+    metrics::Histogram* limiter_histogram;
>+    metrics::Histogram* saturation_histogram;
>+
>+    RegionLogger(std::string identity_histogram_name,
>+                 std::string knee_histogram_name,
>+                 std::string limiter_histogram_name,
>+                 std::string saturation_histogram_name);
>+
>+    ~RegionLogger();
>+
>+    void LogRegionStats(const InterpolatedGainCurve::Stats& stats) const;
>+  } region_logger_;
>+
>+  void UpdateStats(float input_level) const;
>+
>+  ApmDataDumper* const apm_data_dumper_;
>+
>+  static constexpr std::array<float, kInterpolatedGainCurveTotalPoints>
>+      approximation_params_x_ = {
>+          {30057.296875,    30148.986328125, 30240.67578125,  30424.052734375,
>+           30607.4296875,   30790.806640625, 30974.18359375,  31157.560546875,
>+           31340.939453125, 31524.31640625,  31707.693359375, 31891.0703125,
>+           32074.447265625, 32257.82421875,  32441.201171875, 32624.580078125,
>+           32807.95703125,  32991.33203125,  33174.7109375,   33358.08984375,
>+           33541.46484375,  33724.84375,     33819.53515625,  34009.5390625,
>+           34200.05859375,  34389.81640625,  34674.48828125,  35054.375,
>+           35434.86328125,  35814.81640625,  36195.16796875,  36575.03125}};
>+  static constexpr std::array<float, kInterpolatedGainCurveTotalPoints>
>+      approximation_params_m_ = {
>+          {-3.515235675877192989e-07, -1.050251626111275982e-06,
>+           -2.085213736791047268e-06, -3.443004743530764244e-06,
>+           -4.773849468620028347e-06, -6.077375928725814447e-06,
>+           -7.353257842623861507e-06, -8.601219633419532329e-06,
>+           -9.821013009059242904e-06, -1.101243378798244521e-05,
>+           -1.217532644659513608e-05, -1.330956911260727793e-05,
>+           -1.441507538402220234e-05, -1.549179251014720649e-05,
>+           -1.653970684856176376e-05, -1.755882840370759368e-05,
>+           -1.854918446042574942e-05, -1.951086778717581183e-05,
>+           -2.044398024736437947e-05, -2.1348627342376858e-05,
>+           -2.222496914328075945e-05, -2.265374678245279938e-05,
>+           -2.242570917587727308e-05, -2.220122041762806475e-05,
>+           -2.19802095671184361e-05,  -2.176260204578284174e-05,
>+           -2.133731686626560986e-05, -2.092481918225530535e-05,
>+           -2.052459603874012828e-05, -2.013615448959171772e-05,
>+           -1.975903069251216948e-05, -1.939277899509761482e-05}};
>+
>+  static constexpr std::array<float, kInterpolatedGainCurveTotalPoints>
>+      approximation_params_q_ = {
>+          {1.010565876960754395, 1.031631827354431152, 1.062929749488830566,
>+           1.104239225387573242, 1.144973039627075195, 1.185109615325927734,
>+           1.224629044532775879, 1.263512492179870605, 1.301741957664489746,
>+           1.339300632476806641, 1.376173257827758789, 1.412345528602600098,
>+           1.447803974151611328, 1.482536554336547852, 1.516532182693481445,
>+           1.549780607223510742, 1.582272171974182129, 1.613999366760253906,
>+           1.644955039024353027, 1.675132393836975098, 1.704526185989379883,
>+           1.718986630439758301, 1.711274504661560059, 1.703639745712280273,
>+           1.696081161499023438, 1.688597679138183594, 1.673851132392883301,
>+           1.659391283988952637, 1.645209431648254395, 1.631297469139099121,
>+           1.617647409439086914, 1.604251742362976074}};
>+
>+  // Stats.
>+  mutable Stats stats_;
>+
>+  RTC_DISALLOW_COPY_AND_ASSIGN(InterpolatedGainCurve);
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_AGC2_INTERPOLATED_GAIN_CURVE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/interpolated_gain_curve_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/interpolated_gain_curve_unittest.cc
>new file mode 100644
>index 00000000000..dd696313ae2
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/interpolated_gain_curve_unittest.cc
>@@ -0,0 +1,202 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <array>
>+#include <vector>
>+
>+#include "api/array_view.h"
>+#include "common_audio/include/audio_util.h"
>+#include "modules/audio_processing/agc2/agc2_common.h"
>+#include "modules/audio_processing/agc2/compute_interpolated_gain_curve.h"
>+#include "modules/audio_processing/agc2/interpolated_gain_curve.h"
>+#include "modules/audio_processing/agc2/limiter.h"
>+#include "modules/audio_processing/logging/apm_data_dumper.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/gunit.h"
>+
>+namespace webrtc {
>+namespace {
>+
>+constexpr double kLevelEpsilon = 1e-2 * kMaxAbsFloatS16Value;
>+constexpr float kInterpolatedGainCurveTolerance = 1.f / 32768.f;
>+ApmDataDumper apm_data_dumper(0);
>+const Limiter limiter;
>+
>+}  // namespace
>+
>+TEST(AutomaticGainController2InterpolatedGainCurve, CreateUse) {
>+  InterpolatedGainCurve igc(&apm_data_dumper, "");
>+
>+  const auto levels = test::LinSpace(
>+      kLevelEpsilon, DbfsToFloatS16(limiter.max_input_level_db() + 1), 500);
>+  for (const auto level : levels) {
>+    EXPECT_GE(igc.LookUpGainToApply(level), 0.0f);
>+  }
>+}
>+
>+TEST(AutomaticGainController2InterpolatedGainCurve, CheckValidOutput) {
>+  InterpolatedGainCurve igc(&apm_data_dumper, "");
>+
>+  const auto levels = test::LinSpace(
>+      kLevelEpsilon, limiter.max_input_level_linear() * 2.0, 500);
>+  for (const auto level : levels) {
>+    SCOPED_TRACE(std::to_string(level));
>+    const float gain = igc.LookUpGainToApply(level);
>+    EXPECT_LE(0.0f, gain);
>+    EXPECT_LE(gain, 1.0f);
>+  }
>+}
>+
>+TEST(AutomaticGainController2InterpolatedGainCurve, CheckMonotonicity) {
>+  InterpolatedGainCurve igc(&apm_data_dumper, "");
>+
>+  const auto levels = test::LinSpace(
>+      kLevelEpsilon, limiter.max_input_level_linear() + kLevelEpsilon + 0.5,
>+      500);
>+  float prev_gain = igc.LookUpGainToApply(0.0f);
>+  for (const auto level : levels) {
>+    const float gain = igc.LookUpGainToApply(level);
>+    EXPECT_GE(prev_gain, gain);
>+    prev_gain = gain;
>+  }
>+}
>+
>+TEST(AutomaticGainController2InterpolatedGainCurve, CheckApproximation) {
>+  InterpolatedGainCurve igc(&apm_data_dumper, "");
>+
>+  const auto levels = test::LinSpace(
>+      kLevelEpsilon, limiter.max_input_level_linear() - kLevelEpsilon, 500);
>+  for (const auto level : levels) {
>+    SCOPED_TRACE(std::to_string(level));
>+    EXPECT_LT(
>+        std::fabs(limiter.GetGainLinear(level) - igc.LookUpGainToApply(level)),
>+        kInterpolatedGainCurveTolerance);
>+  }
>+}
>+
>+TEST(AutomaticGainController2InterpolatedGainCurve, CheckRegionBoundaries) {
>+  InterpolatedGainCurve igc(&apm_data_dumper, "");
>+
>+  const std::vector<double> levels{
>+      {kLevelEpsilon, limiter.knee_start_linear() + kLevelEpsilon,
>+       limiter.limiter_start_linear() + kLevelEpsilon,
>+       limiter.max_input_level_linear() + kLevelEpsilon}};
>+  for (const auto level : levels) {
>+    igc.LookUpGainToApply(level);
>+  }
>+
>+  const auto stats = igc.get_stats();
>+  EXPECT_EQ(1ul, stats.look_ups_identity_region);
>+  EXPECT_EQ(1ul, stats.look_ups_knee_region);
>+  EXPECT_EQ(1ul, stats.look_ups_limiter_region);
>+  EXPECT_EQ(1ul, stats.look_ups_saturation_region);
>+}
>+
>+TEST(AutomaticGainController2InterpolatedGainCurve, CheckIdentityRegion) {
>+  constexpr size_t kNumSteps = 10;
>+  InterpolatedGainCurve igc(&apm_data_dumper, "");
>+
>+  const auto levels =
>+      test::LinSpace(kLevelEpsilon, limiter.knee_start_linear(), kNumSteps);
>+  for (const auto level : levels) {
>+    SCOPED_TRACE(std::to_string(level));
>+    EXPECT_EQ(1.0f, igc.LookUpGainToApply(level));
>+  }
>+
>+  const auto stats = igc.get_stats();
>+  EXPECT_EQ(kNumSteps - 1, stats.look_ups_identity_region);
>+  EXPECT_EQ(1ul, stats.look_ups_knee_region);
>+  EXPECT_EQ(0ul, stats.look_ups_limiter_region);
>+  EXPECT_EQ(0ul, stats.look_ups_saturation_region);
>+}
>+
>+TEST(AutomaticGainController2InterpolatedGainCurve,
>+     CheckNoOverApproximationKnee) {
>+  constexpr size_t kNumSteps = 10;
>+  InterpolatedGainCurve igc(&apm_data_dumper, "");
>+
>+  const auto levels =
>+      test::LinSpace(limiter.knee_start_linear() + kLevelEpsilon,
>+                     limiter.limiter_start_linear(), kNumSteps);
>+  for (const auto level : levels) {
>+    SCOPED_TRACE(std::to_string(level));
>+    // Small tolerance added (needed because comparing a float with a double).
>+    EXPECT_LE(igc.LookUpGainToApply(level),
>+              limiter.GetGainLinear(level) + 1e-7);
>+  }
>+
>+  const auto stats = igc.get_stats();
>+  EXPECT_EQ(0ul, stats.look_ups_identity_region);
>+  EXPECT_EQ(kNumSteps - 1, stats.look_ups_knee_region);
>+  EXPECT_EQ(1ul, stats.look_ups_limiter_region);
>+  EXPECT_EQ(0ul, stats.look_ups_saturation_region);
>+}
>+
>+TEST(AutomaticGainController2InterpolatedGainCurve,
>+     CheckNoOverApproximationBeyondKnee) {
>+  constexpr size_t kNumSteps = 10;
>+  InterpolatedGainCurve igc(&apm_data_dumper, "");
>+
>+  const auto levels = test::LinSpace(
>+      limiter.limiter_start_linear() + kLevelEpsilon,
>+      limiter.max_input_level_linear() - kLevelEpsilon, kNumSteps);
>+  for (const auto level : levels) {
>+    SCOPED_TRACE(std::to_string(level));
>+    // Small tolerance added (needed because comparing a float with a double).
>+    EXPECT_LE(igc.LookUpGainToApply(level),
>+              limiter.GetGainLinear(level) + 1e-7);
>+  }
>+
>+  const auto stats = igc.get_stats();
>+  EXPECT_EQ(0ul, stats.look_ups_identity_region);
>+  EXPECT_EQ(0ul, stats.look_ups_knee_region);
>+  EXPECT_EQ(kNumSteps, stats.look_ups_limiter_region);
>+  EXPECT_EQ(0ul, stats.look_ups_saturation_region);
>+}
>+
>+TEST(AutomaticGainController2InterpolatedGainCurve,
>+     CheckNoOverApproximationWithSaturation) {
>+  constexpr size_t kNumSteps = 3;
>+  InterpolatedGainCurve igc(&apm_data_dumper, "");
>+
>+  const auto levels = test::LinSpace(
>+      limiter.max_input_level_linear() + kLevelEpsilon,
>+      limiter.max_input_level_linear() + kLevelEpsilon + 0.5, kNumSteps);
>+  for (const auto level : levels) {
>+    SCOPED_TRACE(std::to_string(level));
>+    EXPECT_LE(igc.LookUpGainToApply(level), limiter.GetGainLinear(level));
>+  }
>+
>+  const auto stats = igc.get_stats();
>+  EXPECT_EQ(0ul, stats.look_ups_identity_region);
>+  EXPECT_EQ(0ul, stats.look_ups_knee_region);
>+  EXPECT_EQ(0ul, stats.look_ups_limiter_region);
>+  EXPECT_EQ(kNumSteps, stats.look_ups_saturation_region);
>+}
>+
>+TEST(AutomaticGainController2InterpolatedGainCurve, CheckApproximationParams) {
>+  test::InterpolatedParameters parameters =
>+      test::ComputeInterpolatedGainCurveApproximationParams();
>+
>+  InterpolatedGainCurve igc(&apm_data_dumper, "");
>+
>+  for (size_t i = 0; i < kInterpolatedGainCurveTotalPoints; ++i) {
>+    // The tolerance levels are chosen to account for deviations due
>+    // to computing with single precision floating point numbers.
>+    EXPECT_NEAR(igc.approximation_params_x_[i],
>+                parameters.computed_approximation_params_x[i], 0.9f);
>+    EXPECT_NEAR(igc.approximation_params_m_[i],
>+                parameters.computed_approximation_params_m[i], 0.00001f);
>+    EXPECT_NEAR(igc.approximation_params_q_[i],
>+                parameters.computed_approximation_params_q[i], 0.001f);
>+  }
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/limiter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/limiter.cc
>new file mode 100644
>index 00000000000..d2b9877be11
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/limiter.cc
>@@ -0,0 +1,137 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/agc2/limiter.h"
>+
>+#include <cmath>
>+
>+#include "common_audio/include/audio_util.h"
>+#include "modules/audio_processing/agc2/agc2_common.h"
>+#include "rtc_base/checks.h"
>+
>+namespace webrtc {
>+namespace {
>+
>+double ComputeKneeStart(double max_input_level_db,
>+                        double knee_smoothness_db,
>+                        double compression_ratio) {
>+  RTC_CHECK_LT((compression_ratio - 1.0) * knee_smoothness_db /
>+                   (2.0 * compression_ratio),
>+               max_input_level_db);
>+  return -knee_smoothness_db / 2.0 -
>+         max_input_level_db / (compression_ratio - 1.0);
>+}
>+
>+std::array<double, 3> ComputeKneeRegionPolynomial(double knee_start_dbfs,
>+                                                  double knee_smoothness_db,
>+                                                  double compression_ratio) {
>+  const double a = (1.0 - compression_ratio) /
>+                   (2.0 * knee_smoothness_db * compression_ratio);
>+  const double b = 1.0 - 2.0 * a * knee_start_dbfs;
>+  const double c = a * knee_start_dbfs * knee_start_dbfs;
>+  return {{a, b, c}};
>+}
>+
>+double ComputeLimiterD1(double max_input_level_db, double compression_ratio) {
>+  return (std::pow(10.0, -max_input_level_db / (20.0 * compression_ratio)) *
>+          (1.0 - compression_ratio) / compression_ratio) /
>+         kMaxAbsFloatS16Value;
>+}
>+
>+constexpr double ComputeLimiterD2(double compression_ratio) {
>+  return (1.0 - 2.0 * compression_ratio) / compression_ratio;
>+}
>+
>+double ComputeLimiterI2(double max_input_level_db,
>+                        double compression_ratio,
>+                        double gain_curve_limiter_i1) {
>+  RTC_CHECK_NE(gain_curve_limiter_i1, 0.f);
>+  return std::pow(10.0, -max_input_level_db / (20.0 * compression_ratio)) /
>+         gain_curve_limiter_i1 /
>+         std::pow(kMaxAbsFloatS16Value, gain_curve_limiter_i1 - 1);
>+}
>+
>+}  // namespace
>+
>+Limiter::Limiter()
>+    : max_input_level_linear_(DbfsToFloatS16(max_input_level_db_)),
>+      knee_start_dbfs_(ComputeKneeStart(max_input_level_db_,
>+                                        knee_smoothness_db_,
>+                                        compression_ratio_)),
>+      knee_start_linear_(DbfsToFloatS16(knee_start_dbfs_)),
>+      limiter_start_dbfs_(knee_start_dbfs_ + knee_smoothness_db_),
>+      limiter_start_linear_(DbfsToFloatS16(limiter_start_dbfs_)),
>+      knee_region_polynomial_(ComputeKneeRegionPolynomial(knee_start_dbfs_,
>+                                                          knee_smoothness_db_,
>+                                                          compression_ratio_)),
>+      gain_curve_limiter_d1_(
>+          ComputeLimiterD1(max_input_level_db_, compression_ratio_)),
>+      gain_curve_limiter_d2_(ComputeLimiterD2(compression_ratio_)),
>+      gain_curve_limiter_i1_(1.0 / compression_ratio_),
>+      gain_curve_limiter_i2_(ComputeLimiterI2(max_input_level_db_,
>+                                              compression_ratio_,
>+                                              gain_curve_limiter_i1_)) {
>+  static_assert(knee_smoothness_db_ > 0.0f, "");
>+  static_assert(compression_ratio_ > 1.0f, "");
>+  RTC_CHECK_GE(max_input_level_db_, knee_start_dbfs_ + knee_smoothness_db_);
>+}
>+
>+constexpr double Limiter::max_input_level_db_;
>+constexpr double Limiter::knee_smoothness_db_;
>+constexpr double Limiter::compression_ratio_;
>+
>+double Limiter::GetOutputLevelDbfs(double input_level_dbfs) const {
>+  if (input_level_dbfs < knee_start_dbfs_) {
>+    return input_level_dbfs;
>+  } else if (input_level_dbfs < limiter_start_dbfs_) {
>+    return GetKneeRegionOutputLevelDbfs(input_level_dbfs);
>+  }
>+  return GetCompressorRegionOutputLevelDbfs(input_level_dbfs);
>+}
>+
>+double Limiter::GetGainLinear(double input_level_linear) const {
>+  if (input_level_linear < knee_start_linear_) {
>+    return 1.0;
>+  }
>+  return DbfsToFloatS16(
>+             GetOutputLevelDbfs(FloatS16ToDbfs(input_level_linear))) /
>+         input_level_linear;
>+}
>+
>+// Computes the first derivative of GetGainLinear() in |x|.
>+double Limiter::GetGainFirstDerivativeLinear(double x) const {
>+  // Beyond-knee region only.
>+  RTC_CHECK_GE(x, limiter_start_linear_ - 1e-7 * kMaxAbsFloatS16Value);
>+  return gain_curve_limiter_d1_ *
>+         std::pow(x / kMaxAbsFloatS16Value, gain_curve_limiter_d2_);
>+}
>+
>+// Computes the integral of GetGainLinear() in the range [x0, x1].
>+double Limiter::GetGainIntegralLinear(double x0, double x1) const {
>+  RTC_CHECK_LE(x0, x1);                     // Valid interval.
>+  RTC_CHECK_GE(x0, limiter_start_linear_);  // Beyond-knee region only.
>+  auto limiter_integral = [this](const double& x) {
>+    return gain_curve_limiter_i2_ * std::pow(x, gain_curve_limiter_i1_);
>+  };
>+  return limiter_integral(x1) - limiter_integral(x0);
>+}
>+
>+double Limiter::GetKneeRegionOutputLevelDbfs(double input_level_dbfs) const {
>+  return knee_region_polynomial_[0] * input_level_dbfs * input_level_dbfs +
>+         knee_region_polynomial_[1] * input_level_dbfs +
>+         knee_region_polynomial_[2];
>+}
>+
>+double Limiter::GetCompressorRegionOutputLevelDbfs(
>+    double input_level_dbfs) const {
>+  return (input_level_dbfs - max_input_level_db_) / compression_ratio_;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/limiter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/limiter.h
>new file mode 100644
>index 00000000000..f350baeef10
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/limiter.h
>@@ -0,0 +1,74 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_AGC2_LIMITER_H_
>+#define MODULES_AUDIO_PROCESSING_AGC2_LIMITER_H_
>+
>+#include <array>
>+
>+#include "modules/audio_processing/agc2/agc2_testing_common.h"
>+
>+namespace webrtc {
>+
>+// A class for computing gain curve parameters. The gain curve is
>+// defined by constants kLimiterMaxInputLevelDbFs, kLimiterKneeSmoothnessDb,
>+// kLimiterCompressionRatio. The curve consints of one linear part,
>+// one quadratic polynomial part and another linear part. The
>+// constants define the parameters of the parts.
>+class Limiter {
>+ public:
>+  Limiter();
>+
>+  double max_input_level_db() const { return max_input_level_db_; }
>+  double max_input_level_linear() const { return max_input_level_linear_; }
>+  double knee_start_linear() const { return knee_start_linear_; }
>+  double limiter_start_linear() const { return limiter_start_linear_; }
>+
>+  // These methods can be marked 'constexpr' in C++ 14.
>+  double GetOutputLevelDbfs(double input_level_dbfs) const;
>+  double GetGainLinear(double input_level_linear) const;
>+  double GetGainFirstDerivativeLinear(double x) const;
>+  double GetGainIntegralLinear(double x0, double x1) const;
>+
>+ private:
>+  double GetKneeRegionOutputLevelDbfs(double input_level_dbfs) const;
>+  double GetCompressorRegionOutputLevelDbfs(double input_level_dbfs) const;
>+
>+  static constexpr double max_input_level_db_ = test::kLimiterMaxInputLevelDbFs;
>+  static constexpr double knee_smoothness_db_ = test::kLimiterKneeSmoothnessDb;
>+  static constexpr double compression_ratio_ = test::kLimiterCompressionRatio;
>+
>+  const double max_input_level_linear_;
>+
>+  // Do not modify signal with level <= knee_start_dbfs_.
>+  const double knee_start_dbfs_;
>+  const double knee_start_linear_;
>+
>+  // The upper end of the knee region, which is between knee_start_dbfs_ and
>+  // limiter_start_dbfs_.
>+  const double limiter_start_dbfs_;
>+  const double limiter_start_linear_;
>+
>+  // Coefficients {a, b, c} of the knee region polynomial
>+  // ax^2 + bx + c in the DB scale.
>+  const std::array<double, 3> knee_region_polynomial_;
>+
>+  // Parameters for the computation of the first derivative of GetGainLinear().
>+  const double gain_curve_limiter_d1_;
>+  const double gain_curve_limiter_d2_;
>+
>+  // Parameters for the computation of the integral of GetGainLinear().
>+  const double gain_curve_limiter_i1_;
>+  const double gain_curve_limiter_i2_;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_AGC2_LIMITER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/limiter_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/limiter_unittest.cc
>new file mode 100644
>index 00000000000..70798126347
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/limiter_unittest.cc
>@@ -0,0 +1,60 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/agc2/limiter.h"
>+
>+#include "rtc_base/gunit.h"
>+
>+namespace webrtc {
>+
>+TEST(FixedDigitalGainController2Limiter, ConstructDestruct) {
>+  Limiter l;
>+}
>+
>+TEST(FixedDigitalGainController2Limiter, GainCurveShouldBeMonotone) {
>+  Limiter l;
>+  float last_output_level = 0.f;
>+  bool has_last_output_level = false;
>+  for (float level = -90.f; level <= l.max_input_level_db(); level += 0.5f) {
>+    const float current_output_level = l.GetOutputLevelDbfs(level);
>+    if (!has_last_output_level) {
>+      last_output_level = current_output_level;
>+      has_last_output_level = true;
>+    }
>+    EXPECT_LE(last_output_level, current_output_level);
>+    last_output_level = current_output_level;
>+  }
>+}
>+
>+TEST(FixedDigitalGainController2Limiter, GainCurveShouldBeContinuous) {
>+  Limiter l;
>+  float last_output_level = 0.f;
>+  bool has_last_output_level = false;
>+  constexpr float kMaxDelta = 0.5f;
>+  for (float level = -90.f; level <= l.max_input_level_db(); level += 0.5f) {
>+    const float current_output_level = l.GetOutputLevelDbfs(level);
>+    if (!has_last_output_level) {
>+      last_output_level = current_output_level;
>+      has_last_output_level = true;
>+    }
>+    EXPECT_LE(current_output_level, last_output_level + kMaxDelta);
>+    last_output_level = current_output_level;
>+  }
>+}
>+
>+TEST(FixedDigitalGainController2Limiter, OutputGainShouldBeLessThanFullScale) {
>+  Limiter l;
>+  for (float level = -90.f; level <= l.max_input_level_db(); level += 0.5f) {
>+    const float current_output_level = l.GetOutputLevelDbfs(level);
>+    EXPECT_LE(current_output_level, 0.f);
>+  }
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/noise_level_estimator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/noise_level_estimator.cc
>similarity index 51%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/noise_level_estimator.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/noise_level_estimator.cc
>index abf4ea2cb19..d9aaf1f9bd3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/noise_level_estimator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/noise_level_estimator.cc
>@@ -8,40 +8,77 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "modules/audio_processing/level_controller/noise_level_estimator.h"
>+#include "modules/audio_processing/agc2/noise_level_estimator.h"
>+
>+#include <math.h>
> 
> #include <algorithm>
>+#include <numeric>
> 
>-#include "modules/audio_processing/audio_buffer.h"
>+#include "common_audio/include/audio_util.h"
> #include "modules/audio_processing/logging/apm_data_dumper.h"
> 
> namespace webrtc {
> 
>-NoiseLevelEstimator::NoiseLevelEstimator() {
>-  Initialize(AudioProcessing::kSampleRate48kHz);
>+namespace {
>+constexpr int kFramesPerSecond = 100;
>+
>+float FrameEnergy(const AudioFrameView<const float>& audio) {
>+  float energy = 0.f;
>+  for (size_t k = 0; k < audio.num_channels(); ++k) {
>+    float channel_energy =
>+        std::accumulate(audio.channel(k).begin(), audio.channel(k).end(), 0.f,
>+                        [](float a, float b) -> float { return a + b * b; });
>+    energy = std::max(channel_energy, energy);
>+  }
>+  return energy;
>+}
>+
>+float EnergyToDbfs(float signal_energy, size_t num_samples) {
>+  const float rms = std::sqrt(signal_energy / num_samples);
>+  return FloatS16ToDbfs(rms);
>+}
>+}  // namespace
>+
>+NoiseLevelEstimator::NoiseLevelEstimator(ApmDataDumper* data_dumper)
>+    : signal_classifier_(data_dumper) {
>+  Initialize(48000);
> }
> 
> NoiseLevelEstimator::~NoiseLevelEstimator() {}
> 
> void NoiseLevelEstimator::Initialize(int sample_rate_hz) {
>+  sample_rate_hz_ = sample_rate_hz;
>   noise_energy_ = 1.f;
>   first_update_ = true;
>-  min_noise_energy_ = sample_rate_hz * 2.f * 2.f / 100.f;
>+  min_noise_energy_ = sample_rate_hz * 2.f * 2.f / kFramesPerSecond;
>   noise_energy_hold_counter_ = 0;
>+  signal_classifier_.Initialize(sample_rate_hz);
> }
> 
>-float NoiseLevelEstimator::Analyze(SignalClassifier::SignalType signal_type,
>-                                   float frame_energy) {
>+float NoiseLevelEstimator::Analyze(const AudioFrameView<const float>& frame) {
>+  const int rate =
>+      static_cast<int>(frame.samples_per_channel() * kFramesPerSecond);
>+  if (rate != sample_rate_hz_) {
>+    Initialize(rate);
>+  }
>+  const float frame_energy = FrameEnergy(frame);
>   if (frame_energy <= 0.f) {
>-    return noise_energy_;
>+    RTC_DCHECK_GE(frame_energy, 0.f);
>+    return EnergyToDbfs(noise_energy_, frame.samples_per_channel());
>   }
> 
>   if (first_update_) {
>     // Initialize the noise energy to the frame energy.
>     first_update_ = false;
>-    return noise_energy_ = std::max(frame_energy, min_noise_energy_);
>+    return EnergyToDbfs(
>+        noise_energy_ = std::max(frame_energy, min_noise_energy_),
>+        frame.samples_per_channel());
>   }
> 
>+  const SignalClassifier::SignalType signal_type =
>+      signal_classifier_.Analyze(frame.channel(0));
>+
>   // Update the noise estimate in a minimum statistics-type manner.
>   if (signal_type == SignalClassifier::SignalType::kStationary) {
>     if (frame_energy > noise_energy_) {
>@@ -66,7 +103,9 @@ float NoiseLevelEstimator::Analyze(SignalClassifier::SignalType signal_type,
>   }
> 
>   // Ensure a minimum of the estimate.
>-  return noise_energy_ = std::max(noise_energy_, min_noise_energy_);
>+  return EnergyToDbfs(
>+      noise_energy_ = std::max(noise_energy_, min_noise_energy_),
>+      frame.samples_per_channel());
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/noise_level_estimator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/noise_level_estimator.h
>similarity index 56%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/noise_level_estimator.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/noise_level_estimator.h
>index 94ef6737e77..24067a16655 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/noise_level_estimator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/noise_level_estimator.h
>@@ -8,30 +8,36 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#ifndef MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_NOISE_LEVEL_ESTIMATOR_H_
>-#define MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_NOISE_LEVEL_ESTIMATOR_H_
>+#ifndef MODULES_AUDIO_PROCESSING_AGC2_NOISE_LEVEL_ESTIMATOR_H_
>+#define MODULES_AUDIO_PROCESSING_AGC2_NOISE_LEVEL_ESTIMATOR_H_
> 
>-#include "modules/audio_processing/level_controller/signal_classifier.h"
>+#include "modules/audio_processing/agc2/signal_classifier.h"
>+#include "modules/audio_processing/include/audio_frame_view.h"
> #include "rtc_base/constructormagic.h"
> 
> namespace webrtc {
>+class ApmDataDumper;
> 
> class NoiseLevelEstimator {
>  public:
>-  NoiseLevelEstimator();
>+  NoiseLevelEstimator(ApmDataDumper* data_dumper);
>   ~NoiseLevelEstimator();
>-  void Initialize(int sample_rate_hz);
>-  float Analyze(SignalClassifier::SignalType signal_type, float frame_energy);
>+  // Returns the estimated noise level in dBFS.
>+  float Analyze(const AudioFrameView<const float>& frame);
> 
>  private:
>-  float min_noise_energy_ = 0.f;
>+  void Initialize(int sample_rate_hz);
>+
>+  int sample_rate_hz_;
>+  float min_noise_energy_;
>   bool first_update_;
>   float noise_energy_;
>   int noise_energy_hold_counter_;
>+  SignalClassifier signal_classifier_;
> 
>   RTC_DISALLOW_COPY_AND_ASSIGN(NoiseLevelEstimator);
> };
> 
> }  // namespace webrtc
> 
>-#endif  // MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_NOISE_LEVEL_ESTIMATOR_H_
>+#endif  // MODULES_AUDIO_PROCESSING_AGC2_NOISE_LEVEL_ESTIMATOR_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/noise_level_estimator_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/noise_level_estimator_unittest.cc
>new file mode 100644
>index 00000000000..c4fd33b0a08
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/noise_level_estimator_unittest.cc
>@@ -0,0 +1,83 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/agc2/noise_level_estimator.h"
>+
>+#include <array>
>+#include <functional>
>+#include <limits>
>+
>+#include "modules/audio_processing/agc2/agc2_testing_common.h"
>+#include "modules/audio_processing/agc2/vector_float_frame.h"
>+#include "modules/audio_processing/logging/apm_data_dumper.h"
>+#include "rtc_base/gunit.h"
>+#include "rtc_base/random.h"
>+
>+namespace webrtc {
>+namespace {
>+Random rand_gen(42);
>+ApmDataDumper data_dumper(0);
>+constexpr int kNumIterations = 200;
>+constexpr int kFramesPerSecond = 100;
>+
>+// Runs the noise estimator on audio generated by 'sample_generator'
>+// for kNumIterations. Returns the last noise level estimate.
>+float RunEstimator(std::function<float()> sample_generator, int rate) {
>+  NoiseLevelEstimator estimator(&data_dumper);
>+  const size_t samples_per_channel =
>+      rtc::CheckedDivExact(rate, kFramesPerSecond);
>+  VectorFloatFrame signal(1, static_cast<int>(samples_per_channel), 0.f);
>+
>+  for (int i = 0; i < kNumIterations; ++i) {
>+    AudioFrameView<float> frame_view = signal.float_frame_view();
>+    for (size_t j = 0; j < samples_per_channel; ++j) {
>+      frame_view.channel(0)[j] = sample_generator();
>+    }
>+    estimator.Analyze(frame_view);
>+  }
>+  return estimator.Analyze(signal.float_frame_view());
>+}
>+
>+float WhiteNoiseGenerator() {
>+  return static_cast<float>(rand_gen.Rand(std::numeric_limits<int16_t>::min(),
>+                                          std::numeric_limits<int16_t>::max()));
>+}
>+}  // namespace
>+
>+// White random noise is stationary, but does not trigger the detector
>+// every frame due to the randomness.
>+TEST(AutomaticGainController2NoiseEstimator, RandomNoise) {
>+  for (const auto rate : {8000, 16000, 32000, 48000}) {
>+    const float noise_level = RunEstimator(WhiteNoiseGenerator, rate);
>+    EXPECT_NEAR(noise_level, -5.f, 1.f);
>+  }
>+}
>+
>+// Sine curves are (very) stationary. They trigger the detector all
>+// the time. Except for a few initial frames.
>+TEST(AutomaticGainController2NoiseEstimator, SineTone) {
>+  for (const auto rate : {8000, 16000, 32000, 48000}) {
>+    test::SineGenerator gen(600.f, rate);
>+    const float noise_level = RunEstimator(gen, rate);
>+    EXPECT_NEAR(noise_level, -33.f, 1.f);
>+  }
>+}
>+
>+// Pulses are transient if they are far enough apart. They shouldn't
>+// trigger the noise detector.
>+TEST(AutomaticGainController2NoiseEstimator, PulseTone) {
>+  for (const auto rate : {8000, 16000, 32000, 48000}) {
>+    test::PulseGenerator gen(20.f, rate);
>+    const int noise_level = RunEstimator(gen, rate);
>+    EXPECT_NEAR(noise_level, -79.f, 1.f);
>+  }
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/noise_spectrum_estimator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/noise_spectrum_estimator.cc
>similarity index 96%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/noise_spectrum_estimator.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/noise_spectrum_estimator.cc
>index 6e921c24d11..9e08126e892 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/noise_spectrum_estimator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/noise_spectrum_estimator.cc
>@@ -8,7 +8,7 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "modules/audio_processing/level_controller/noise_spectrum_estimator.h"
>+#include "modules/audio_processing/agc2/noise_spectrum_estimator.h"
> 
> #include <string.h>
> #include <algorithm>
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/noise_spectrum_estimator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/noise_spectrum_estimator.h
>similarity index 80%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/noise_spectrum_estimator.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/noise_spectrum_estimator.h
>index f10933ec968..fd1cc13a3f5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/noise_spectrum_estimator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/noise_spectrum_estimator.h
>@@ -8,8 +8,8 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#ifndef MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_NOISE_SPECTRUM_ESTIMATOR_H_
>-#define MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_NOISE_SPECTRUM_ESTIMATOR_H_
>+#ifndef MODULES_AUDIO_PROCESSING_AGC2_NOISE_SPECTRUM_ESTIMATOR_H_
>+#define MODULES_AUDIO_PROCESSING_AGC2_NOISE_SPECTRUM_ESTIMATOR_H_
> 
> #include "api/array_view.h"
> #include "rtc_base/constructormagic.h"
>@@ -37,4 +37,4 @@ class NoiseSpectrumEstimator {
> 
> }  // namespace webrtc
> 
>-#endif  // MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_NOISE_SPECTRUM_ESTIMATOR_H_
>+#endif  // MODULES_AUDIO_PROCESSING_AGC2_NOISE_SPECTRUM_ESTIMATOR_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/BUILD.gn
>new file mode 100644
>index 00000000000..d1edecea33b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/BUILD.gn
>@@ -0,0 +1,130 @@
>+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+#
>+# Use of this source code is governed by a BSD-style license
>+# that can be found in the LICENSE file in the root of the source
>+# tree. An additional intellectual property rights grant can be found
>+# in the file PATENTS.  All contributing project authors may
>+# be found in the AUTHORS file in the root of the source tree.
>+
>+import("../../../../webrtc.gni")
>+
>+rtc_source_set("rnn_vad") {
>+  sources = [
>+    "common.h",
>+    "features_extraction.cc",
>+    "features_extraction.h",
>+    "fft_util.cc",
>+    "fft_util.h",
>+    "lp_residual.cc",
>+    "lp_residual.h",
>+    "pitch_info.h",
>+    "pitch_search.cc",
>+    "pitch_search.h",
>+    "pitch_search_internal.cc",
>+    "pitch_search_internal.h",
>+    "ring_buffer.h",
>+    "rnn.cc",
>+    "rnn.h",
>+    "sequence_buffer.h",
>+    "spectral_features.cc",
>+    "spectral_features.h",
>+    "spectral_features_internal.cc",
>+    "spectral_features_internal.h",
>+    "symmetric_matrix_buffer.h",
>+  ]
>+  deps = [
>+    "..:biquad_filter",
>+    "../../../../api:array_view",
>+    "../../../../common_audio/",
>+    "../../../../rtc_base:checks",
>+    "../../../../rtc_base:rtc_base_approved",
>+    "//third_party/rnnoise:kiss_fft",
>+    "//third_party/rnnoise:rnn_vad",
>+  ]
>+}
>+
>+if (rtc_include_tests) {
>+  rtc_source_set("test_utils") {
>+    testonly = true
>+    sources = [
>+      "test_utils.cc",
>+      "test_utils.h",
>+    ]
>+    deps = [
>+      "../../../../api:array_view",
>+      "../../../../rtc_base:checks",
>+      "../../../../rtc_base:ptr_util",
>+      "../../../../test:fileutils",
>+      "../../../../test:test_support",
>+      "//third_party/abseil-cpp/absl/memory",
>+    ]
>+  }
>+
>+  unittest_resources = [
>+    "../../../../resources/audio_processing/agc2/rnn_vad/band_energies.dat",
>+    "../../../../resources/audio_processing/agc2/rnn_vad/fft.dat",
>+    "../../../../resources/audio_processing/agc2/rnn_vad/pitch_buf_24k.dat",
>+    "../../../../resources/audio_processing/agc2/rnn_vad/pitch_search_int.dat",
>+    "../../../../resources/audio_processing/agc2/rnn_vad/pitch_lp_res.dat",
>+    "../../../../resources/audio_processing/agc2/rnn_vad/samples.pcm",
>+    "../../../../resources/audio_processing/agc2/rnn_vad/sil_features.dat",
>+    "../../../../resources/audio_processing/agc2/rnn_vad/vad_prob.dat",
>+  ]
>+
>+  if (is_ios) {
>+    bundle_data("unittests_bundle_data") {
>+      testonly = true
>+      sources = unittest_resources
>+      outputs = [
>+        "{{bundle_resources_dir}}/{{source_file_part}}",
>+      ]
>+    }
>+  }
>+
>+  rtc_source_set("unittests") {
>+    testonly = true
>+    sources = [
>+      "features_extraction_unittest.cc",
>+      "fft_util_unittest.cc",
>+      "lp_residual_unittest.cc",
>+      "pitch_search_internal_unittest.cc",
>+      "pitch_search_unittest.cc",
>+      "ring_buffer_unittest.cc",
>+      "rnn_unittest.cc",
>+      "rnn_vad_unittest.cc",
>+      "sequence_buffer_unittest.cc",
>+      "spectral_features_internal_unittest.cc",
>+      "spectral_features_unittest.cc",
>+      "symmetric_matrix_buffer_unittest.cc",
>+    ]
>+    deps = [
>+      ":rnn_vad",
>+      ":test_utils",
>+      "../..:audioproc_test_utils",
>+      "../../../../api:array_view",
>+      "../../../../common_audio/",
>+      "../../../../rtc_base:checks",
>+      "../../../../rtc_base:logging",
>+      "../../../../test:test_support",
>+      "//third_party/rnnoise:rnn_vad",
>+    ]
>+    data = unittest_resources
>+    if (is_ios) {
>+      deps += [ ":unittests_bundle_data" ]
>+    }
>+  }
>+
>+  rtc_executable("rnn_vad_tool") {
>+    testonly = true
>+    sources = [
>+      "rnn_vad_tool.cc",
>+    ]
>+    deps = [
>+      ":rnn_vad",
>+      "../../../../api:array_view",
>+      "../../../../common_audio:common_audio",
>+      "../../../../rtc_base:rtc_base_approved",
>+      "../../../../test:test_support",
>+    ]
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/DEPS b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/DEPS
>new file mode 100644
>index 00000000000..773c2d7edd5
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/DEPS
>@@ -0,0 +1,3 @@
>+include_rules = [
>+  "+third_party/rnnoise",
>+]
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/common.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/common.h
>new file mode 100644
>index 00000000000..b98438db68a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/common.h
>@@ -0,0 +1,67 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_COMMON_H_
>+#define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_COMMON_H_
>+
>+namespace webrtc {
>+namespace rnn_vad {
>+
>+constexpr double kPi = 3.14159265358979323846;
>+
>+constexpr size_t kSampleRate24kHz = 24000;
>+constexpr size_t kFrameSize10ms24kHz = kSampleRate24kHz / 100;
>+constexpr size_t kFrameSize20ms24kHz = kFrameSize10ms24kHz * 2;
>+
>+// Pitch analysis params.
>+constexpr size_t kMinPitch24kHz = kSampleRate24kHz / 800;   // 0.00125 s.
>+constexpr size_t kMaxPitch24kHz = kSampleRate24kHz / 62.5;  // 0.016 s.
>+constexpr size_t kBufSize24kHz = kMaxPitch24kHz + kFrameSize20ms24kHz;
>+static_assert((kBufSize24kHz & 1) == 0, "The buffer size must be even.");
>+
>+// Define a higher minimum pitch period for the initial search. This is used to
>+// avoid searching for very short periods, for which a refinement step is
>+// responsible.
>+constexpr size_t kInitialMinPitch24kHz = 3 * kMinPitch24kHz;
>+static_assert(kMinPitch24kHz < kInitialMinPitch24kHz, "");
>+static_assert(kInitialMinPitch24kHz < kMaxPitch24kHz, "");
>+
>+// 12 kHz analysis.
>+constexpr size_t kSampleRate12kHz = 12000;
>+constexpr size_t kFrameSize10ms12kHz = kSampleRate12kHz / 100;
>+constexpr size_t kFrameSize20ms12kHz = kFrameSize10ms12kHz * 2;
>+constexpr size_t kBufSize12kHz = kBufSize24kHz / 2;
>+constexpr size_t kInitialMinPitch12kHz = kInitialMinPitch24kHz / 2;
>+constexpr size_t kMaxPitch12kHz = kMaxPitch24kHz / 2;
>+
>+// 48 kHz constants.
>+constexpr size_t kMinPitch48kHz = kMinPitch24kHz * 2;
>+constexpr size_t kMaxPitch48kHz = kMaxPitch24kHz * 2;
>+
>+// Sub-band frequency boundaries.
>+constexpr size_t kNumBands = 22;
>+constexpr int kBandFrequencyBoundaries[kNumBands] = {
>+    0,    200,  400,  600,  800,  1000, 1200, 1400, 1600,  2000,  2400,
>+    2800, 3200, 4000, 4800, 5600, 6800, 8000, 9600, 12000, 15600, 20000};
>+
>+// Feature extraction parameters.
>+constexpr size_t kNumLowerBands = 6;
>+static_assert((0 < kNumLowerBands) && (kNumLowerBands < kNumBands), "");
>+constexpr size_t kSpectralCoeffsHistorySize = 8;
>+static_assert(kSpectralCoeffsHistorySize > 2,
>+              "The history size must at least be 3 to compute first and second "
>+              "derivatives.");
>+
>+constexpr size_t kFeatureVectorSize = 42;
>+
>+}  // namespace rnn_vad
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_COMMON_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/features_extraction.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/features_extraction.cc
>new file mode 100644
>index 00000000000..8ab56736a2f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/features_extraction.cc
>@@ -0,0 +1,88 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/agc2/rnn_vad/features_extraction.h"
>+
>+#include "modules/audio_processing/agc2/rnn_vad/lp_residual.h"
>+#include "rtc_base/checks.h"
>+
>+namespace webrtc {
>+namespace rnn_vad {
>+namespace {
>+
>+// Generated via "B, A = scipy.signal.butter(2, 30/12000, btype='highpass')"
>+const BiQuadFilter::BiQuadCoefficients kHpfConfig24k = {
>+    {0.99446179f, -1.98892358f, 0.99446179f},
>+    {-1.98889291f, 0.98895425f}};
>+
>+}  // namespace
>+
>+FeaturesExtractor::FeaturesExtractor()
>+    : use_high_pass_filter_(false),
>+      pitch_buf_24kHz_(),
>+      pitch_buf_24kHz_view_(pitch_buf_24kHz_.GetBufferView()),
>+      lp_residual_(kBufSize24kHz),
>+      lp_residual_view_(lp_residual_.data(), kBufSize24kHz),
>+      pitch_estimator_(),
>+      reference_frame_view_(pitch_buf_24kHz_.GetMostRecentValuesView()) {
>+  RTC_DCHECK_EQ(kBufSize24kHz, lp_residual_.size());
>+  hpf_.Initialize(kHpfConfig24k);
>+  Reset();
>+}
>+
>+FeaturesExtractor::~FeaturesExtractor() = default;
>+
>+void FeaturesExtractor::Reset() {
>+  pitch_buf_24kHz_.Reset();
>+  spectral_features_extractor_.Reset();
>+  if (use_high_pass_filter_)
>+    hpf_.Reset();
>+}
>+
>+bool FeaturesExtractor::CheckSilenceComputeFeatures(
>+    rtc::ArrayView<const float, kFrameSize10ms24kHz> samples,
>+    rtc::ArrayView<float, kFeatureVectorSize> feature_vector) {
>+  // Pre-processing.
>+  if (use_high_pass_filter_) {
>+    std::array<float, kFrameSize10ms24kHz> samples_filtered;
>+    hpf_.Process(samples, samples_filtered);
>+    // Feed buffer with the pre-processed version of |samples|.
>+    pitch_buf_24kHz_.Push(samples_filtered);
>+  } else {
>+    // Feed buffer with |samples|.
>+    pitch_buf_24kHz_.Push(samples);
>+  }
>+  // Extract the LP residual.
>+  float lpc_coeffs[kNumLpcCoefficients];
>+  ComputeAndPostProcessLpcCoefficients(pitch_buf_24kHz_view_, lpc_coeffs);
>+  ComputeLpResidual(lpc_coeffs, pitch_buf_24kHz_view_, lp_residual_view_);
>+  // Estimate pitch on the LP-residual and write the normalized pitch period
>+  // into the output vector (normalization based on training data stats).
>+  pitch_info_48kHz_ = pitch_estimator_.Estimate(lp_residual_view_);
>+  feature_vector[kFeatureVectorSize - 2] =
>+      0.01f * (static_cast<int>(pitch_info_48kHz_.period) - 300);
>+  // Extract lagged frames (according to the estimated pitch period).
>+  RTC_DCHECK_LE(pitch_info_48kHz_.period / 2, kMaxPitch24kHz);
>+  auto lagged_frame = pitch_buf_24kHz_view_.subview(
>+      kMaxPitch24kHz - pitch_info_48kHz_.period / 2, kFrameSize20ms24kHz);
>+  // Analyze reference and lagged frames checking if silence has been detected
>+  // and write the feature vector.
>+  return spectral_features_extractor_.CheckSilenceComputeFeatures(
>+      reference_frame_view_, {lagged_frame.data(), kFrameSize20ms24kHz},
>+      {{feature_vector.data() + kNumLowerBands, kNumBands - kNumLowerBands},
>+       {feature_vector.data(), kNumLowerBands},
>+       {feature_vector.data() + kNumBands, kNumLowerBands},
>+       {feature_vector.data() + kNumBands + kNumLowerBands, kNumLowerBands},
>+       {feature_vector.data() + kNumBands + 2 * kNumLowerBands, kNumLowerBands},
>+       &feature_vector[kFeatureVectorSize - 1]});
>+}
>+
>+}  // namespace rnn_vad
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/features_extraction.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/features_extraction.h
>new file mode 100644
>index 00000000000..1f63885c4e6
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/features_extraction.h
>@@ -0,0 +1,62 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_FEATURES_EXTRACTION_H_
>+#define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_FEATURES_EXTRACTION_H_
>+
>+#include <memory>
>+#include <vector>
>+
>+#include "api/array_view.h"
>+#include "modules/audio_processing/agc2/biquad_filter.h"
>+#include "modules/audio_processing/agc2/rnn_vad/common.h"
>+#include "modules/audio_processing/agc2/rnn_vad/pitch_search.h"
>+#include "modules/audio_processing/agc2/rnn_vad/sequence_buffer.h"
>+#include "modules/audio_processing/agc2/rnn_vad/spectral_features.h"
>+
>+namespace webrtc {
>+namespace rnn_vad {
>+
>+// Feature extractor to feed the VAD RNN.
>+class FeaturesExtractor {
>+ public:
>+  FeaturesExtractor();
>+  FeaturesExtractor(const FeaturesExtractor&) = delete;
>+  FeaturesExtractor& operator=(const FeaturesExtractor&) = delete;
>+  ~FeaturesExtractor();
>+  void Reset();
>+  // Analyzes the samples, computes the feature vector and returns true if
>+  // silence is detected (false if not). When silence is detected,
>+  // |feature_vector| is partially written and therefore must not be used to
>+  // feed the VAD RNN.
>+  bool CheckSilenceComputeFeatures(
>+      rtc::ArrayView<const float, kFrameSize10ms24kHz> samples,
>+      rtc::ArrayView<float, kFeatureVectorSize> feature_vector);
>+
>+ private:
>+  const bool use_high_pass_filter_;
>+  // TODO(bugs.webrtc.org/7494): Remove HPF depending on how AGC2 is used in APM
>+  // and on whether an HPF is already used as pre-processing step in APM.
>+  BiQuadFilter hpf_;
>+  SequenceBuffer<float, kBufSize24kHz, kFrameSize10ms24kHz, kFrameSize20ms24kHz>
>+      pitch_buf_24kHz_;
>+  rtc::ArrayView<const float, kBufSize24kHz> pitch_buf_24kHz_view_;
>+  std::vector<float> lp_residual_;
>+  rtc::ArrayView<float, kBufSize24kHz> lp_residual_view_;
>+  PitchEstimator pitch_estimator_;
>+  rtc::ArrayView<const float, kFrameSize20ms24kHz> reference_frame_view_;
>+  SpectralFeaturesExtractor spectral_features_extractor_;
>+  PitchInfo pitch_info_48kHz_;
>+};
>+
>+}  // namespace rnn_vad
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_FEATURES_EXTRACTION_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/features_extraction_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/features_extraction_unittest.cc
>new file mode 100644
>index 00000000000..c00fc232ebb
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/features_extraction_unittest.cc
>@@ -0,0 +1,104 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/agc2/rnn_vad/features_extraction.h"
>+
>+#include <cmath>
>+#include <vector>
>+
>+#include "modules/audio_processing/agc2/rnn_vad/test_utils.h"
>+// TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
>+// #include "test/fpe_observer.h"
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+namespace rnn_vad {
>+namespace test {
>+namespace {
>+
>+constexpr size_t ceil(size_t n, size_t m) {
>+  return (n + m - 1) / m;
>+}
>+
>+// Number of 10 ms frames required to fill a pitch buffer having size
>+// |kBufSize24kHz|.
>+constexpr size_t kNumTestDataFrames = ceil(kBufSize24kHz, kFrameSize10ms24kHz);
>+// Number of samples for the test data.
>+constexpr size_t kNumTestDataSize = kNumTestDataFrames * kFrameSize10ms24kHz;
>+
>+// Verifies that the pitch in Hz is in the detectable range.
>+bool PitchIsValid(float pitch_hz) {
>+  const size_t pitch_period =
>+      static_cast<size_t>(static_cast<float>(kSampleRate24kHz) / pitch_hz);
>+  return kInitialMinPitch24kHz <= pitch_period &&
>+         pitch_period <= kMaxPitch24kHz;
>+}
>+
>+void CreatePureTone(float amplitude, float freq_hz, rtc::ArrayView<float> dst) {
>+  for (size_t i = 0; i < dst.size(); ++i) {
>+    dst[i] = amplitude * std::sin(2.f * kPi * freq_hz * i / kSampleRate24kHz);
>+  }
>+}
>+
>+// Feeds |features_extractor| with |samples| splitting it in 10 ms frames.
>+// For every frame, the output is written into |feature_vector|. Returns true
>+// if silence is detected in the last frame.
>+bool FeedTestData(FeaturesExtractor* features_extractor,
>+                  rtc::ArrayView<const float> samples,
>+                  rtc::ArrayView<float, kFeatureVectorSize> feature_vector) {
>+  // TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
>+  // FloatingPointExceptionObserver fpe_observer;
>+  bool is_silence = true;
>+  const size_t num_frames = samples.size() / kFrameSize10ms24kHz;
>+  for (size_t i = 0; i < num_frames; ++i) {
>+    is_silence = features_extractor->CheckSilenceComputeFeatures(
>+        {samples.data() + i * kFrameSize10ms24kHz, kFrameSize10ms24kHz},
>+        feature_vector);
>+  }
>+  return is_silence;
>+}
>+
>+}  // namespace
>+
>+// Extracts the features for two pure tones and verifies that the pitch field
>+// values reflect the known tone frequencies.
>+TEST(RnnVadTest, FeatureExtractionLowHighPitch) {
>+  constexpr float amplitude = 1000.f;
>+  constexpr float low_pitch_hz = 150.f;
>+  constexpr float high_pitch_hz = 250.f;
>+  ASSERT_TRUE(PitchIsValid(low_pitch_hz));
>+  ASSERT_TRUE(PitchIsValid(high_pitch_hz));
>+
>+  FeaturesExtractor features_extractor;
>+  std::vector<float> samples(kNumTestDataSize);
>+  std::vector<float> feature_vector(kFeatureVectorSize);
>+  ASSERT_EQ(kFeatureVectorSize, feature_vector.size());
>+  rtc::ArrayView<float, kFeatureVectorSize> feature_vector_view(
>+      feature_vector.data(), kFeatureVectorSize);
>+
>+  // Extract the normalized scalar feature that is proportional to the estimated
>+  // pitch period.
>+  constexpr size_t pitch_feature_index = kFeatureVectorSize - 2;
>+  // Low frequency tone - i.e., high period.
>+  CreatePureTone(amplitude, low_pitch_hz, samples);
>+  ASSERT_FALSE(FeedTestData(&features_extractor, samples, feature_vector_view));
>+  float high_pitch_period = feature_vector_view[pitch_feature_index];
>+  // High frequency tone - i.e., low period.
>+  features_extractor.Reset();
>+  CreatePureTone(amplitude, high_pitch_hz, samples);
>+  ASSERT_FALSE(FeedTestData(&features_extractor, samples, feature_vector_view));
>+  float low_pitch_period = feature_vector_view[pitch_feature_index];
>+  // Check.
>+  EXPECT_LT(low_pitch_period, high_pitch_period);
>+}
>+
>+}  // namespace test
>+}  // namespace rnn_vad
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/fft_util.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/fft_util.cc
>new file mode 100644
>index 00000000000..1017400a282
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/fft_util.cc
>@@ -0,0 +1,58 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/agc2/rnn_vad/fft_util.h"
>+
>+#include <cmath>
>+
>+#include "rtc_base/checks.h"
>+
>+namespace webrtc {
>+namespace rnn_vad {
>+namespace {
>+
>+constexpr size_t kHalfFrameSize = kFrameSize20ms24kHz / 2;
>+
>+// Computes the first half of the Vorbis window.
>+std::array<float, kHalfFrameSize> ComputeHalfVorbisWindow() {
>+  std::array<float, kHalfFrameSize> half_window{};
>+  for (size_t i = 0; i < kHalfFrameSize; ++i) {
>+    half_window[i] =
>+        std::sin(0.5 * kPi * std::sin(0.5 * kPi * (i + 0.5) / kHalfFrameSize) *
>+                 std::sin(0.5 * kPi * (i + 0.5) / kHalfFrameSize));
>+  }
>+  return half_window;
>+}
>+
>+}  // namespace
>+
>+BandAnalysisFft::BandAnalysisFft()
>+    : half_window_(ComputeHalfVorbisWindow()),
>+      fft_(static_cast<int>(input_buf_.size())) {}
>+
>+BandAnalysisFft::~BandAnalysisFft() = default;
>+
>+void BandAnalysisFft::ForwardFft(rtc::ArrayView<const float> samples,
>+                                 rtc::ArrayView<std::complex<float>> dst) {
>+  RTC_DCHECK_EQ(input_buf_.size(), samples.size());
>+  RTC_DCHECK_EQ(samples.size(), dst.size());
>+  // Apply windowing.
>+  RTC_DCHECK_EQ(input_buf_.size(), 2 * half_window_.size());
>+  for (size_t i = 0; i < input_buf_.size() / 2; ++i) {
>+    input_buf_[i].real(samples[i] * half_window_[i]);
>+    size_t j = kFrameSize20ms24kHz - i - 1;
>+    input_buf_[j].real(samples[j] * half_window_[i]);
>+  }
>+  fft_.ForwardFft(kFrameSize20ms24kHz, input_buf_.data(), kFrameSize20ms24kHz,
>+                  dst.data());
>+}
>+
>+}  // namespace rnn_vad
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/fft_util.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/fft_util.h
>new file mode 100644
>index 00000000000..f4265f4e327
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/fft_util.h
>@@ -0,0 +1,51 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_FFT_UTIL_H_
>+#define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_FFT_UTIL_H_
>+
>+#include <array>
>+#include <complex>
>+
>+#include "api/array_view.h"
>+#include "modules/audio_processing/agc2/rnn_vad/common.h"
>+#include "third_party/rnnoise/src/kiss_fft.h"
>+
>+namespace webrtc {
>+namespace rnn_vad {
>+
>+// FFT implementation wrapper for the band-wise analysis step in which 20 ms
>+// frames at 24 kHz are analyzed in the frequency domain. The goal of this class
>+// are (i) making easy to switch to another FFT implementation, (ii) own the
>+// input buffer for the FFT and (iii) apply a windowing function before
>+// computing the FFT.
>+class BandAnalysisFft {
>+ public:
>+  BandAnalysisFft();
>+  BandAnalysisFft(const BandAnalysisFft&) = delete;
>+  BandAnalysisFft& operator=(const BandAnalysisFft&) = delete;
>+  ~BandAnalysisFft();
>+  // Applies a windowing function to |samples|, computes the real forward FFT
>+  // and writes the result in |dst|.
>+  void ForwardFft(rtc::ArrayView<const float> samples,
>+                  rtc::ArrayView<std::complex<float>> dst);
>+
>+ private:
>+  static_assert((kFrameSize20ms24kHz & 1) == 0,
>+                "kFrameSize20ms24kHz must be even.");
>+  const std::array<float, kFrameSize20ms24kHz / 2> half_window_;
>+  std::array<std::complex<float>, kFrameSize20ms24kHz> input_buf_{};
>+  rnnoise::KissFft fft_;
>+};
>+
>+}  // namespace rnn_vad
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_FFT_UTIL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/fft_util_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/fft_util_unittest.cc
>new file mode 100644
>index 00000000000..985460051d9
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/fft_util_unittest.cc
>@@ -0,0 +1,49 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <array>
>+
>+#include "modules/audio_processing/agc2/rnn_vad/common.h"
>+#include "modules/audio_processing/agc2/rnn_vad/fft_util.h"
>+#include "rtc_base/checks.h"
>+// TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
>+// #include "test/fpe_observer.h"
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+namespace rnn_vad {
>+namespace test {
>+
>+TEST(RnnVadTest, CheckBandAnalysisFftOutput) {
>+  // Input data.
>+  std::array<float, kFrameSize20ms24kHz> samples{};
>+  for (int i = 0; i < static_cast<int>(kFrameSize20ms24kHz); ++i) {
>+    samples[i] = i - static_cast<int>(kFrameSize20ms24kHz / 2);
>+  }
>+  // TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
>+  // FloatingPointExceptionObserver fpe_observer;
>+  BandAnalysisFft fft;
>+  std::array<std::complex<float>, kFrameSize20ms24kHz> fft_coeffs;
>+  fft.ForwardFft(samples, fft_coeffs);
>+  // First coefficient is DC - i.e., real number.
>+  EXPECT_EQ(0.f, fft_coeffs[0].imag());
>+  // Check conjugated symmetry of the FFT output.
>+  for (size_t i = 1; i < fft_coeffs.size() / 2; ++i) {
>+    SCOPED_TRACE(i);
>+    const auto& a = fft_coeffs[i];
>+    const auto& b = fft_coeffs[fft_coeffs.size() - i];
>+    EXPECT_NEAR(a.real(), b.real(), 2e-6f);
>+    EXPECT_NEAR(a.imag(), -b.imag(), 2e-6f);
>+  }
>+}
>+
>+}  // namespace test
>+}  // namespace rnn_vad
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/lp_residual.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/lp_residual.cc
>new file mode 100644
>index 00000000000..1a124a34938
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/lp_residual.cc
>@@ -0,0 +1,138 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/agc2/rnn_vad/lp_residual.h"
>+
>+#include <algorithm>
>+#include <array>
>+#include <cmath>
>+#include <numeric>
>+
>+#include "rtc_base/checks.h"
>+
>+namespace webrtc {
>+namespace rnn_vad {
>+namespace {
>+
>+// Computes cross-correlation coefficients between |x| and |y| and writes them
>+// in |x_corr|. The lag values are in {0, ..., max_lag - 1}, where max_lag
>+// equals the size of |x_corr|.
>+// The |x| and |y| sub-arrays used to compute a cross-correlation coefficients
>+// for a lag l have both size "size of |x| - l" - i.e., the longest sub-array is
>+// used. |x| and |y| must have the same size.
>+void ComputeCrossCorrelation(
>+    rtc::ArrayView<const float> x,
>+    rtc::ArrayView<const float> y,
>+    rtc::ArrayView<float, kNumLpcCoefficients> x_corr) {
>+  constexpr size_t max_lag = x_corr.size();
>+  RTC_DCHECK_EQ(x.size(), y.size());
>+  RTC_DCHECK_LT(max_lag, x.size());
>+  for (size_t lag = 0; lag < max_lag; ++lag) {
>+    x_corr[lag] =
>+        std::inner_product(x.begin(), x.end() - lag, y.begin() + lag, 0.f);
>+  }
>+}
>+
>+// Applies denoising to the auto-correlation coefficients.
>+void DenoiseAutoCorrelation(
>+    rtc::ArrayView<float, kNumLpcCoefficients> auto_corr) {
>+  // Assume -40 dB white noise floor.
>+  auto_corr[0] *= 1.0001f;
>+  for (size_t i = 1; i < kNumLpcCoefficients; ++i) {
>+    auto_corr[i] -= auto_corr[i] * (0.008f * i) * (0.008f * i);
>+  }
>+}
>+
>+// Computes the initial inverse filter coefficients given the auto-correlation
>+// coefficients of an input frame.
>+void ComputeInitialInverseFilterCoefficients(
>+    rtc::ArrayView<const float, kNumLpcCoefficients> auto_corr,
>+    rtc::ArrayView<float, kNumLpcCoefficients - 1> lpc_coeffs) {
>+  float error = auto_corr[0];
>+  for (size_t i = 0; i < kNumLpcCoefficients - 1; ++i) {
>+    float reflection_coeff = 0.f;
>+    for (size_t j = 0; j < i; ++j) {
>+      reflection_coeff += lpc_coeffs[j] * auto_corr[i - j];
>+    }
>+    reflection_coeff += auto_corr[i + 1];
>+
>+    // Avoid division by numbers close to zero.
>+    constexpr float kMinErrorMagnitude = 1e-6f;
>+    if (std::fabs(error) < kMinErrorMagnitude) {
>+      error = std::copysign(kMinErrorMagnitude, error);
>+    }
>+
>+    reflection_coeff /= -error;
>+    // Update LPC coefficients and total error.
>+    lpc_coeffs[i] = reflection_coeff;
>+    for (size_t j = 0; j<(i + 1)>> 1; ++j) {
>+      const float tmp1 = lpc_coeffs[j];
>+      const float tmp2 = lpc_coeffs[i - 1 - j];
>+      lpc_coeffs[j] = tmp1 + reflection_coeff * tmp2;
>+      lpc_coeffs[i - 1 - j] = tmp2 + reflection_coeff * tmp1;
>+    }
>+    error -= reflection_coeff * reflection_coeff * error;
>+    if (error < 0.001f * auto_corr[0]) {
>+      break;
>+    }
>+  }
>+}
>+
>+}  // namespace
>+
>+void ComputeAndPostProcessLpcCoefficients(
>+    rtc::ArrayView<const float> x,
>+    rtc::ArrayView<float, kNumLpcCoefficients> lpc_coeffs) {
>+  std::array<float, kNumLpcCoefficients> auto_corr;
>+  ComputeCrossCorrelation(x, x, {auto_corr.data(), auto_corr.size()});
>+  if (auto_corr[0] == 0.f) {  // Empty frame.
>+    std::fill(lpc_coeffs.begin(), lpc_coeffs.end(), 0);
>+    return;
>+  }
>+  DenoiseAutoCorrelation({auto_corr.data(), auto_corr.size()});
>+  std::array<float, kNumLpcCoefficients - 1> lpc_coeffs_pre{};
>+  ComputeInitialInverseFilterCoefficients(auto_corr, lpc_coeffs_pre);
>+  // LPC coefficients post-processing.
>+  // TODO(bugs.webrtc.org/9076): Consider removing these steps.
>+  float c1 = 1.f;
>+  for (size_t i = 0; i < kNumLpcCoefficients - 1; ++i) {
>+    c1 *= 0.9f;
>+    lpc_coeffs_pre[i] *= c1;
>+  }
>+  const float c2 = 0.8f;
>+  lpc_coeffs[0] = lpc_coeffs_pre[0] + c2;
>+  lpc_coeffs[1] = lpc_coeffs_pre[1] + c2 * lpc_coeffs_pre[0];
>+  lpc_coeffs[2] = lpc_coeffs_pre[2] + c2 * lpc_coeffs_pre[1];
>+  lpc_coeffs[3] = lpc_coeffs_pre[3] + c2 * lpc_coeffs_pre[2];
>+  lpc_coeffs[4] = c2 * lpc_coeffs_pre[3];
>+}
>+
>+void ComputeLpResidual(
>+    rtc::ArrayView<const float, kNumLpcCoefficients> lpc_coeffs,
>+    rtc::ArrayView<const float> x,
>+    rtc::ArrayView<float> y) {
>+  RTC_DCHECK_LT(kNumLpcCoefficients, x.size());
>+  RTC_DCHECK_EQ(x.size(), y.size());
>+  std::array<float, kNumLpcCoefficients> input_chunk;
>+  input_chunk.fill(0.f);
>+  for (size_t i = 0; i < y.size(); ++i) {
>+    const float sum = std::inner_product(input_chunk.begin(), input_chunk.end(),
>+                                         lpc_coeffs.begin(), x[i]);
>+    // Circular shift and add a new sample.
>+    for (size_t j = kNumLpcCoefficients - 1; j > 0; --j)
>+      input_chunk[j] = input_chunk[j - 1];
>+    input_chunk[0] = x[i];
>+    // Copy result.
>+    y[i] = sum;
>+  }
>+}
>+
>+}  // namespace rnn_vad
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/lp_residual.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/lp_residual.h
>new file mode 100644
>index 00000000000..bffafd291f9
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/lp_residual.h
>@@ -0,0 +1,39 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_LP_RESIDUAL_H_
>+#define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_LP_RESIDUAL_H_
>+
>+#include "api/array_view.h"
>+
>+namespace webrtc {
>+namespace rnn_vad {
>+
>+// LPC inverse filter length.
>+constexpr size_t kNumLpcCoefficients = 5;
>+
>+// Given a frame |x|, computes a post-processed version of LPC coefficients
>+// tailored for pitch estimation.
>+void ComputeAndPostProcessLpcCoefficients(
>+    rtc::ArrayView<const float> x,
>+    rtc::ArrayView<float, kNumLpcCoefficients> lpc_coeffs);
>+
>+// Computes the LP residual for the input frame |x| and the LPC coefficients
>+// |lpc_coeffs|. |y| and |x| can point to the same array for in-place
>+// computation.
>+void ComputeLpResidual(
>+    rtc::ArrayView<const float, kNumLpcCoefficients> lpc_coeffs,
>+    rtc::ArrayView<const float> x,
>+    rtc::ArrayView<float> y);
>+
>+}  // namespace rnn_vad
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_LP_RESIDUAL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/lp_residual_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/lp_residual_unittest.cc
>new file mode 100644
>index 00000000000..47d8bf531fb
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/lp_residual_unittest.cc
>@@ -0,0 +1,84 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/agc2/rnn_vad/lp_residual.h"
>+
>+#include <array>
>+
>+#include "modules/audio_processing/agc2/rnn_vad/common.h"
>+#include "modules/audio_processing/agc2/rnn_vad/test_utils.h"
>+// TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
>+// #include "test/fpe_observer.h"
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+namespace rnn_vad {
>+namespace test {
>+
>+TEST(RnnVadTest, LpResidualOfEmptyFrame) {
>+  // TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
>+  // FloatingPointExceptionObserver fpe_observer;
>+
>+  // Input frame (empty, i.e., all samples set to 0).
>+  std::array<float, kFrameSize10ms24kHz> empty_frame;
>+  empty_frame.fill(0.f);
>+  // Compute inverse filter coefficients.
>+  std::array<float, kNumLpcCoefficients> lpc_coeffs;
>+  ComputeAndPostProcessLpcCoefficients(empty_frame, lpc_coeffs);
>+  // Compute LP residual.
>+  std::array<float, kFrameSize10ms24kHz> lp_residual;
>+  ComputeLpResidual(lpc_coeffs, empty_frame, lp_residual);
>+}
>+
>+// TODO(bugs.webrtc.org/9076): Remove when the issue is fixed.
>+TEST(RnnVadTest, LpResidualPipelineBitExactness) {
>+  // Pitch buffer 24 kHz data reader.
>+  auto pitch_buf_24kHz_reader = CreatePitchBuffer24kHzReader();
>+  const size_t num_frames = pitch_buf_24kHz_reader.second;
>+  std::array<float, kBufSize24kHz> pitch_buf_data;
>+  // Read ground-truth.
>+  auto lp_residual_reader = CreateLpResidualAndPitchPeriodGainReader();
>+  ASSERT_EQ(num_frames, lp_residual_reader.second);
>+  std::array<float, kBufSize24kHz> expected_lp_residual;
>+  rtc::ArrayView<float, kBufSize24kHz> expected_lp_residual_view(
>+      expected_lp_residual.data(), expected_lp_residual.size());
>+  // Init pipeline.
>+  std::array<float, kNumLpcCoefficients> lpc_coeffs;
>+  rtc::ArrayView<float, kNumLpcCoefficients> lpc_coeffs_view(
>+      lpc_coeffs.data(), kNumLpcCoefficients);
>+  std::array<float, kBufSize24kHz> computed_lp_residual;
>+  rtc::ArrayView<float, kBufSize24kHz> computed_lp_residual_view(
>+      computed_lp_residual.data(), computed_lp_residual.size());
>+  {
>+    // TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
>+    // FloatingPointExceptionObserver fpe_observer;
>+    for (size_t i = 0; i < num_frames; ++i) {
>+      SCOPED_TRACE(i);
>+      // Read input and expected output.
>+      pitch_buf_24kHz_reader.first->ReadChunk(pitch_buf_data);
>+      lp_residual_reader.first->ReadChunk(expected_lp_residual_view);
>+      // Skip pitch gain and period.
>+      float unused;
>+      lp_residual_reader.first->ReadValue(&unused);
>+      lp_residual_reader.first->ReadValue(&unused);
>+      // Run pipeline.
>+      ComputeAndPostProcessLpcCoefficients(pitch_buf_data, lpc_coeffs_view);
>+      ComputeLpResidual(lpc_coeffs_view, pitch_buf_data,
>+                        computed_lp_residual_view);
>+      // Compare.
>+      ExpectNearAbsolute(expected_lp_residual_view, computed_lp_residual_view,
>+                         kFloatMin);
>+    }
>+  }
>+}
>+
>+}  // namespace test
>+}  // namespace rnn_vad
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/pitch_info.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/pitch_info.h
>new file mode 100644
>index 00000000000..f0998d1fadd
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/pitch_info.h
>@@ -0,0 +1,29 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_PITCH_INFO_H_
>+#define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_PITCH_INFO_H_
>+
>+namespace webrtc {
>+namespace rnn_vad {
>+
>+// Stores pitch period and gain information. The pitch gain measures the
>+// strength of the pitch (the higher, the stronger).
>+struct PitchInfo {
>+  PitchInfo() : period(0), gain(0.f) {}
>+  PitchInfo(size_t p, float g) : period(p), gain(g) {}
>+  size_t period;
>+  float gain;
>+};
>+
>+}  // namespace rnn_vad
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_PITCH_INFO_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/pitch_search.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/pitch_search.cc
>new file mode 100644
>index 00000000000..1f8859d5b04
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/pitch_search.cc
>@@ -0,0 +1,56 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/agc2/rnn_vad/pitch_search.h"
>+
>+namespace webrtc {
>+namespace rnn_vad {
>+
>+PitchEstimator::PitchEstimator()
>+    : fft_(RealFourier::Create(kAutoCorrelationFftOrder)),
>+      pitch_buf_decimated_(kBufSize12kHz),
>+      pitch_buf_decimated_view_(pitch_buf_decimated_.data(), kBufSize12kHz),
>+      auto_corr_(kNumInvertedLags12kHz),
>+      auto_corr_view_(auto_corr_.data(), kNumInvertedLags12kHz) {
>+  RTC_DCHECK_EQ(kBufSize12kHz, pitch_buf_decimated_.size());
>+  RTC_DCHECK_EQ(kNumInvertedLags12kHz, auto_corr_view_.size());
>+}
>+
>+PitchEstimator::~PitchEstimator() = default;
>+
>+PitchInfo PitchEstimator::Estimate(
>+    rtc::ArrayView<const float, kBufSize24kHz> pitch_buf) {
>+  // Perform the initial pitch search at 12 kHz.
>+  Decimate2x(pitch_buf, pitch_buf_decimated_view_);
>+  // Compute auto-correlation terms.
>+  ComputePitchAutoCorrelation(pitch_buf_decimated_view_, kMaxPitch12kHz,
>+                              auto_corr_view_, fft_.get());
>+
>+  // Search for pitch at 12 kHz.
>+  std::array<size_t, 2> pitch_candidates_inv_lags = FindBestPitchPeriods(
>+      auto_corr_view_, pitch_buf_decimated_view_, kMaxPitch12kHz);
>+
>+  // Refine the pitch period estimation.
>+  // The refinement is done using the pitch buffer that contains 24 kHz samples.
>+  // Therefore, adapt the inverted lags in |pitch_candidates_inv_lags| from 12
>+  // to 24 kHz.
>+  for (size_t i = 0; i < pitch_candidates_inv_lags.size(); ++i)
>+    pitch_candidates_inv_lags[i] *= 2;
>+  size_t pitch_inv_lag_48kHz =
>+      RefinePitchPeriod48kHz(pitch_buf, pitch_candidates_inv_lags);
>+  // Look for stronger harmonics to find the final pitch period and its gain.
>+  RTC_DCHECK_LT(pitch_inv_lag_48kHz, kMaxPitch48kHz);
>+  last_pitch_48kHz_ = CheckLowerPitchPeriodsAndComputePitchGain(
>+      pitch_buf, kMaxPitch48kHz - pitch_inv_lag_48kHz, last_pitch_48kHz_);
>+  return last_pitch_48kHz_;
>+}
>+
>+}  // namespace rnn_vad
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/pitch_search.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/pitch_search.h
>new file mode 100644
>index 00000000000..59145353c16
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/pitch_search.h
>@@ -0,0 +1,49 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_PITCH_SEARCH_H_
>+#define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_PITCH_SEARCH_H_
>+
>+#include <memory>
>+#include <vector>
>+
>+#include "api/array_view.h"
>+#include "common_audio/real_fourier.h"
>+#include "modules/audio_processing/agc2/rnn_vad/common.h"
>+#include "modules/audio_processing/agc2/rnn_vad/pitch_info.h"
>+#include "modules/audio_processing/agc2/rnn_vad/pitch_search_internal.h"
>+
>+namespace webrtc {
>+namespace rnn_vad {
>+
>+// Pitch estimator.
>+class PitchEstimator {
>+ public:
>+  PitchEstimator();
>+  PitchEstimator(const PitchEstimator&) = delete;
>+  PitchEstimator& operator=(const PitchEstimator&) = delete;
>+  ~PitchEstimator();
>+  // Estimates the pitch period and gain. Returns the pitch estimation data for
>+  // 48 kHz.
>+  PitchInfo Estimate(rtc::ArrayView<const float, kBufSize24kHz> pitch_buf);
>+
>+ private:
>+  PitchInfo last_pitch_48kHz_;
>+  std::unique_ptr<RealFourier> fft_;
>+  std::vector<float> pitch_buf_decimated_;
>+  rtc::ArrayView<float, kBufSize12kHz> pitch_buf_decimated_view_;
>+  std::vector<float> auto_corr_;
>+  rtc::ArrayView<float, kNumInvertedLags12kHz> auto_corr_view_;
>+};
>+
>+}  // namespace rnn_vad
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_PITCH_SEARCH_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.cc
>new file mode 100644
>index 00000000000..b7b44d29376
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.cc
>@@ -0,0 +1,451 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/agc2/rnn_vad/pitch_search_internal.h"
>+
>+#include <algorithm>
>+#include <cmath>
>+#include <numeric>
>+#include <utility>
>+
>+#include "modules/audio_processing/agc2/rnn_vad/common.h"
>+#include "rtc_base/checks.h"
>+
>+namespace webrtc {
>+namespace rnn_vad {
>+namespace {
>+
>+// Converts a lag to an inverted lag (only for 24kHz).
>+size_t GetInvertedLag(size_t lag) {
>+  RTC_DCHECK_LE(lag, kMaxPitch24kHz);
>+  return kMaxPitch24kHz - lag;
>+}
>+
>+float ComputeAutoCorrelationCoeff(rtc::ArrayView<const float> pitch_buf,
>+                                  size_t inv_lag,
>+                                  size_t max_pitch_period) {
>+  RTC_DCHECK_LT(inv_lag, pitch_buf.size());
>+  RTC_DCHECK_LT(max_pitch_period, pitch_buf.size());
>+  RTC_DCHECK_LE(inv_lag, max_pitch_period);
>+  // TODO(bugs.webrtc.org/9076): Maybe optimize using vectorization.
>+  return std::inner_product(pitch_buf.begin() + max_pitch_period,
>+                            pitch_buf.end(), pitch_buf.begin() + inv_lag, 0.f);
>+}
>+
>+// Computes a pseudo-interpolation offset for an estimated pitch period |lag| by
>+// looking at the auto-correlation coefficients in the neighborhood of |lag|.
>+// (namely, |prev_auto_corr|, |lag_auto_corr| and |next_auto_corr|). The output
>+// is a lag in {-1, 0, +1}.
>+// TODO(bugs.webrtc.org/9076): Consider removing pseudo-i since it
>+// is relevant only if the spectral analysis works at a sample rate that is
>+// twice as that of the pitch buffer (not so important instead for the estimated
>+// pitch period feature fed into the RNN).
>+int GetPitchPseudoInterpolationOffset(size_t lag,
>+                                      float prev_auto_corr,
>+                                      float lag_auto_corr,
>+                                      float next_auto_corr) {
>+  const float& a = prev_auto_corr;
>+  const float& b = lag_auto_corr;
>+  const float& c = next_auto_corr;
>+
>+  int offset = 0;
>+  if ((c - a) > 0.7f * (b - a)) {
>+    offset = 1;  // |c| is the largest auto-correlation coefficient.
>+  } else if ((a - c) > 0.7f * (b - c)) {
>+    offset = -1;  // |a| is the largest auto-correlation coefficient.
>+  }
>+  return offset;
>+}
>+
>+// Refines a pitch period |lag| encoded as lag with pseudo-interpolation. The
>+// output sample rate is twice as that of |lag|.
>+size_t PitchPseudoInterpolationLagPitchBuf(
>+    size_t lag,
>+    rtc::ArrayView<const float, kBufSize24kHz> pitch_buf) {
>+  int offset = 0;
>+  // Cannot apply pseudo-interpolation at the boundaries.
>+  if (lag > 0 && lag < kMaxPitch24kHz) {
>+    offset = GetPitchPseudoInterpolationOffset(
>+        lag,
>+        ComputeAutoCorrelationCoeff(pitch_buf, GetInvertedLag(lag - 1),
>+                                    kMaxPitch24kHz),
>+        ComputeAutoCorrelationCoeff(pitch_buf, GetInvertedLag(lag),
>+                                    kMaxPitch24kHz),
>+        ComputeAutoCorrelationCoeff(pitch_buf, GetInvertedLag(lag + 1),
>+                                    kMaxPitch24kHz));
>+  }
>+  return 2 * lag + offset;
>+}
>+
>+// Refines a pitch period |inv_lag| encoded as inverted lag with
>+// pseudo-interpolation. The output sample rate is twice as that of
>+// |inv_lag|.
>+size_t PitchPseudoInterpolationInvLagAutoCorr(
>+    size_t inv_lag,
>+    rtc::ArrayView<const float> auto_corr) {
>+  int offset = 0;
>+  // Cannot apply pseudo-interpolation at the boundaries.
>+  if (inv_lag > 0 && inv_lag < auto_corr.size() - 1) {
>+    offset = GetPitchPseudoInterpolationOffset(inv_lag, auto_corr[inv_lag + 1],
>+                                               auto_corr[inv_lag],
>+                                               auto_corr[inv_lag - 1]);
>+  }
>+  // TODO(bugs.webrtc.org/9076): When retraining, check if |offset| below should
>+  // be subtracted since |inv_lag| is an inverted lag but offset is a lag.
>+  return 2 * inv_lag + offset;
>+}
>+
>+// Integer multipliers used in CheckLowerPitchPeriodsAndComputePitchGain() when
>+// looking for sub-harmonics.
>+// The values have been chosen to serve the following algorithm. Given the
>+// initial pitch period T, we examine whether one of its harmonics is the true
>+// fundamental frequency. We consider T/k with k in {2, ..., 15}. For each of
>+// these harmonics, in addition to the pitch gain of itself, we choose one
>+// multiple of its pitch period, n*T/k, to validate it (by averaging their pitch
>+// gains). The multiplier n is chosen so that n*T/k is used only one time over
>+// all k. When for example k = 4, we should also expect a peak at 3*T/4. When
>+// k = 8 instead we don't want to look at 2*T/8, since we have already checked
>+// T/4 before. Instead, we look at T*3/8.
>+// The array can be generate in Python as follows:
>+//   from fractions import Fraction
>+//   # Smallest positive integer not in X.
>+//   def mex(X):
>+//     for i in range(1, int(max(X)+2)):
>+//       if i not in X:
>+//         return i
>+//   # Visited multiples of the period.
>+//   S = {1}
>+//   for n in range(2, 16):
>+//     sn = mex({n * i for i in S} | {1})
>+//     S = S | {Fraction(1, n), Fraction(sn, n)}
>+//     print(sn, end=', ')
>+constexpr std::array<size_t, 14> kSubHarmonicMultipliers = {
>+    {3, 2, 3, 2, 5, 2, 3, 2, 3, 2, 5, 2, 3, 2}};
>+
>+// Initial pitch period candidate thresholds for ComputePitchGainThreshold() for
>+// a sample rate of 24 kHz. Computed as [5*k*k for k in range(16)].
>+constexpr std::array<size_t, 14> kInitialPitchPeriodThresholds = {
>+    {20, 45, 80, 125, 180, 245, 320, 405, 500, 605, 720, 845, 980, 1125}};
>+
>+}  // namespace
>+
>+void Decimate2x(rtc::ArrayView<const float, kBufSize24kHz> src,
>+                rtc::ArrayView<float, kBufSize12kHz> dst) {
>+  // TODO(bugs.webrtc.org/9076): Consider adding anti-aliasing filter.
>+  static_assert(2 * dst.size() == src.size(), "");
>+  for (size_t i = 0; i < dst.size(); ++i) {
>+    dst[i] = src[2 * i];
>+  }
>+}
>+
>+float ComputePitchGainThreshold(size_t candidate_pitch_period,
>+                                size_t pitch_period_ratio,
>+                                size_t initial_pitch_period,
>+                                float initial_pitch_gain,
>+                                size_t prev_pitch_period,
>+                                size_t prev_pitch_gain) {
>+  // Map arguments to more compact aliases.
>+  const size_t& t1 = candidate_pitch_period;
>+  const size_t& k = pitch_period_ratio;
>+  const size_t& t0 = initial_pitch_period;
>+  const float& g0 = initial_pitch_gain;
>+  const size_t& t_prev = prev_pitch_period;
>+  const size_t& g_prev = prev_pitch_gain;
>+
>+  // Validate input.
>+  RTC_DCHECK_GE(k, 2);
>+
>+  // Compute a term that lowers the threshold when |t1| is close to the last
>+  // estimated period |t_prev| - i.e., pitch tracking.
>+  float lower_threshold_term = 0;
>+  if (abs(static_cast<int>(t1) - static_cast<int>(t_prev)) <= 1) {
>+    // The candidate pitch period is within 1 sample from the previous one.
>+    // Make the candidate at |t1| very easy to be accepted.
>+    lower_threshold_term = g_prev;
>+  } else if (abs(static_cast<int>(t1) - static_cast<int>(t_prev)) == 2 &&
>+             t0 > kInitialPitchPeriodThresholds[k - 2]) {
>+    // The candidate pitch period is 2 samples far from the previous one and the
>+    // period |t0| (from which |t1| has been derived) is greater than a
>+    // threshold. Make |t1| easy to be accepted.
>+    lower_threshold_term = 0.5f * g_prev;
>+  }
>+  // Set the threshold based on the gain of the initial estimate |t0|. Also
>+  // reduce the chance of false positives caused by a bias towards high
>+  // frequencies (originating from short-term correlations).
>+  float threshold = std::max(0.3f, 0.7f * g0 - lower_threshold_term);
>+  if (t1 < 3 * kMinPitch24kHz) {  // High frequency.
>+    threshold = std::max(0.4f, 0.85f * g0 - lower_threshold_term);
>+  } else if (t1 < 2 * kMinPitch24kHz) {  // Even higher frequency.
>+    threshold = std::max(0.5f, 0.9f * g0 - lower_threshold_term);
>+  }
>+  return threshold;
>+}
>+
>+void ComputeSlidingFrameSquareEnergies(
>+    rtc::ArrayView<const float, kBufSize24kHz> pitch_buf,
>+    rtc::ArrayView<float, kMaxPitch24kHz + 1> yy_values) {
>+  float yy =
>+      ComputeAutoCorrelationCoeff(pitch_buf, kMaxPitch24kHz, kMaxPitch24kHz);
>+  yy_values[0] = yy;
>+  for (size_t i = 1; i < yy_values.size(); ++i) {
>+    RTC_DCHECK_LE(i, kMaxPitch24kHz + kFrameSize20ms24kHz);
>+    RTC_DCHECK_LE(i, kMaxPitch24kHz);
>+    const float old_coeff = pitch_buf[kMaxPitch24kHz + kFrameSize20ms24kHz - i];
>+    const float new_coeff = pitch_buf[kMaxPitch24kHz - i];
>+    yy -= old_coeff * old_coeff;
>+    yy += new_coeff * new_coeff;
>+    yy = std::max(0.f, yy);
>+    yy_values[i] = yy;
>+  }
>+}
>+
>+void ComputePitchAutoCorrelation(
>+    rtc::ArrayView<const float, kBufSize12kHz> pitch_buf,
>+    size_t max_pitch_period,
>+    rtc::ArrayView<float, kNumInvertedLags12kHz> auto_corr,
>+    webrtc::RealFourier* fft) {
>+  RTC_DCHECK_GT(max_pitch_period, auto_corr.size());
>+  RTC_DCHECK_LT(max_pitch_period, pitch_buf.size());
>+  RTC_DCHECK(fft);
>+
>+  constexpr size_t time_domain_fft_length = 1 << kAutoCorrelationFftOrder;
>+  constexpr size_t freq_domain_fft_length = time_domain_fft_length / 2 + 1;
>+
>+  RTC_DCHECK_EQ(RealFourier::FftLength(fft->order()), time_domain_fft_length);
>+  RTC_DCHECK_EQ(RealFourier::ComplexLength(fft->order()),
>+                freq_domain_fft_length);
>+
>+  // Cross-correlation of y_i=pitch_buf[i:i+convolution_length] and
>+  // x=pitch_buf[-convolution_length:] is equivalent to convolution of
>+  // y_i and reversed(x). New notation: h=reversed(x), x=y.
>+  std::array<float, time_domain_fft_length> h{};
>+  std::array<float, time_domain_fft_length> x{};
>+
>+  const size_t convolution_length = kBufSize12kHz - max_pitch_period;
>+  // Check that the FFT-length is big enough to avoid cyclic
>+  // convolution errors.
>+  RTC_DCHECK_GT(time_domain_fft_length,
>+                kNumInvertedLags12kHz + convolution_length);
>+
>+  // h[0:convolution_length] is reversed pitch_buf[-convolution_length:].
>+  std::reverse_copy(pitch_buf.end() - convolution_length, pitch_buf.end(),
>+                    h.begin());
>+
>+  // x is pitch_buf[:kNumInvertedLags12kHz + convolution_length].
>+  std::copy(pitch_buf.begin(),
>+            pitch_buf.begin() + kNumInvertedLags12kHz + convolution_length,
>+            x.begin());
>+
>+  // Shift to frequency domain.
>+  std::array<std::complex<float>, freq_domain_fft_length> X{};
>+  std::array<std::complex<float>, freq_domain_fft_length> H{};
>+  fft->Forward(&x[0], &X[0]);
>+  fft->Forward(&h[0], &H[0]);
>+
>+  // Convolve in frequency domain.
>+  for (size_t i = 0; i < X.size(); ++i) {
>+    X[i] *= H[i];
>+  }
>+
>+  // Shift back to time domain.
>+  std::array<float, time_domain_fft_length> x_conv_h;
>+  fft->Inverse(&X[0], &x_conv_h[0]);
>+
>+  // Collect the result.
>+  std::copy(x_conv_h.begin() + convolution_length - 1,
>+            x_conv_h.begin() + convolution_length + kNumInvertedLags12kHz - 1,
>+            auto_corr.begin());
>+}
>+
>+std::array<size_t, 2> FindBestPitchPeriods(
>+    rtc::ArrayView<const float> auto_corr,
>+    rtc::ArrayView<const float> pitch_buf,
>+    size_t max_pitch_period) {
>+  // Stores a pitch candidate period and strength information.
>+  struct PitchCandidate {
>+    // Pitch period encoded as inverted lag.
>+    size_t period_inverted_lag = 0;
>+    // Pitch strength encoded as a ratio.
>+    float strength_numerator = -1.f;
>+    float strength_denominator = 0.f;
>+    // Compare the strength of two pitch candidates.
>+    bool HasStrongerPitchThan(const PitchCandidate& b) const {
>+      // Comparing the numerator/denominator ratios without using divisions.
>+      return strength_numerator * b.strength_denominator >
>+             b.strength_numerator * strength_denominator;
>+    }
>+  };
>+
>+  RTC_DCHECK_GT(max_pitch_period, auto_corr.size());
>+  RTC_DCHECK_LT(max_pitch_period, pitch_buf.size());
>+  const size_t frame_size = pitch_buf.size() - max_pitch_period;
>+  // TODO(bugs.webrtc.org/9076): Maybe optimize using vectorization.
>+  float yy =
>+      std::inner_product(pitch_buf.begin(), pitch_buf.begin() + frame_size + 1,
>+                         pitch_buf.begin(), 1.f);
>+  // Search best and second best pitches by looking at the scaled
>+  // auto-correlation.
>+  PitchCandidate candidate;
>+  PitchCandidate best;
>+  PitchCandidate second_best;
>+  second_best.period_inverted_lag = 1;
>+  for (size_t inv_lag = 0; inv_lag < auto_corr.size(); ++inv_lag) {
>+    // A pitch candidate must have positive correlation.
>+    if (auto_corr[inv_lag] > 0) {
>+      candidate.period_inverted_lag = inv_lag;
>+      candidate.strength_numerator = auto_corr[inv_lag] * auto_corr[inv_lag];
>+      candidate.strength_denominator = yy;
>+      if (candidate.HasStrongerPitchThan(second_best)) {
>+        if (candidate.HasStrongerPitchThan(best)) {
>+          second_best = best;
>+          best = candidate;
>+        } else {
>+          second_best = candidate;
>+        }
>+      }
>+    }
>+    // Update |squared_energy_y| for the next inverted lag.
>+    const float old_coeff = pitch_buf[inv_lag];
>+    const float new_coeff = pitch_buf[inv_lag + frame_size];
>+    yy -= old_coeff * old_coeff;
>+    yy += new_coeff * new_coeff;
>+    yy = std::max(0.f, yy);
>+  }
>+  return {{best.period_inverted_lag, second_best.period_inverted_lag}};
>+}
>+
>+size_t RefinePitchPeriod48kHz(
>+    rtc::ArrayView<const float, kBufSize24kHz> pitch_buf,
>+    rtc::ArrayView<const size_t, 2> inv_lags) {
>+  // Compute the auto-correlation terms only for neighbors of the given pitch
>+  // candidates (similar to what is done in ComputePitchAutoCorrelation(), but
>+  // for a few lag values).
>+  std::array<float, kNumInvertedLags24kHz> auto_corr;
>+  auto_corr.fill(0.f);  // Zeros become ignored lags in FindBestPitchPeriods().
>+  auto is_neighbor = [](size_t i, size_t j) {
>+    return ((i > j) ? (i - j) : (j - i)) <= 2;
>+  };
>+  for (size_t inv_lag = 0; inv_lag < auto_corr.size(); ++inv_lag) {
>+    if (is_neighbor(inv_lag, inv_lags[0]) || is_neighbor(inv_lag, inv_lags[1]))
>+      auto_corr[inv_lag] =
>+          ComputeAutoCorrelationCoeff(pitch_buf, inv_lag, kMaxPitch24kHz);
>+  }
>+  // Find best pitch at 24 kHz.
>+  const auto pitch_candidates_inv_lags = FindBestPitchPeriods(
>+      {auto_corr.data(), auto_corr.size()},
>+      {pitch_buf.data(), pitch_buf.size()}, kMaxPitch24kHz);
>+  const auto inv_lag = pitch_candidates_inv_lags[0];  // Refine the best.
>+  // Pseudo-interpolation.
>+  return PitchPseudoInterpolationInvLagAutoCorr(inv_lag, auto_corr);
>+}
>+
>+PitchInfo CheckLowerPitchPeriodsAndComputePitchGain(
>+    rtc::ArrayView<const float, kBufSize24kHz> pitch_buf,
>+    size_t initial_pitch_period_48kHz,
>+    PitchInfo prev_pitch_48kHz) {
>+  RTC_DCHECK_LE(kMinPitch48kHz, initial_pitch_period_48kHz);
>+  RTC_DCHECK_LE(initial_pitch_period_48kHz, kMaxPitch48kHz);
>+  // Stores information for a refined pitch candidate.
>+  struct RefinedPitchCandidate {
>+    RefinedPitchCandidate() {}
>+    RefinedPitchCandidate(size_t period_24kHz, float gain, float xy, float yy)
>+        : period_24kHz(period_24kHz), gain(gain), xy(xy), yy(yy) {}
>+    size_t period_24kHz;
>+    // Pitch strength information.
>+    float gain;
>+    // Additional pitch strength information used for the final estimation of
>+    // pitch gain.
>+    float xy;  // Cross-correlation.
>+    float yy;  // Auto-correlation.
>+  };
>+
>+  // Initialize.
>+  std::array<float, kMaxPitch24kHz + 1> yy_values;
>+  ComputeSlidingFrameSquareEnergies(pitch_buf,
>+                                    {yy_values.data(), yy_values.size()});
>+  const float xx = yy_values[0];
>+  // Helper lambdas.
>+  const auto pitch_gain = [](float xy, float yy, float xx) {
>+    RTC_DCHECK_LE(0.f, xx * yy);
>+    return xy / std::sqrt(1.f + xx * yy);
>+  };
>+  // Initial pitch candidate gain.
>+  RefinedPitchCandidate best_pitch;
>+  best_pitch.period_24kHz =
>+      std::min(initial_pitch_period_48kHz / 2, kMaxPitch24kHz - 1);
>+  best_pitch.xy = ComputeAutoCorrelationCoeff(
>+      pitch_buf, GetInvertedLag(best_pitch.period_24kHz), kMaxPitch24kHz);
>+  best_pitch.yy = yy_values[best_pitch.period_24kHz];
>+  best_pitch.gain = pitch_gain(best_pitch.xy, best_pitch.yy, xx);
>+
>+  // Store the initial pitch period information.
>+  const size_t initial_pitch_period = best_pitch.period_24kHz;
>+  const float initial_pitch_gain = best_pitch.gain;
>+
>+  // Given the initial pitch estimation, check lower periods (i.e., harmonics).
>+  const auto alternative_period = [](size_t period, size_t k,
>+                                     size_t n) -> size_t {
>+    RTC_DCHECK_LT(0, k);
>+    return (2 * n * period + k) / (2 * k);  // Same as round(n*period/k).
>+  };
>+  for (size_t k = 2; k < kSubHarmonicMultipliers.size() + 2; ++k) {
>+    size_t candidate_pitch_period =
>+        alternative_period(initial_pitch_period, k, 1);
>+    if (candidate_pitch_period < kMinPitch24kHz)
>+      break;
>+    // When looking at |candidate_pitch_period|, we also look at one of its
>+    // sub-harmonics. |kSubHarmonicMultipliers| is used to know where to look.
>+    // |k| == 2 is a special case since |candidate_pitch_secondary_period| might
>+    // be greater than the maximum pitch period.
>+    size_t candidate_pitch_secondary_period = alternative_period(
>+        initial_pitch_period, k, kSubHarmonicMultipliers[k - 2]);
>+    if (k == 2 && candidate_pitch_secondary_period > kMaxPitch24kHz)
>+      candidate_pitch_secondary_period = initial_pitch_period;
>+    RTC_DCHECK_NE(candidate_pitch_period, candidate_pitch_secondary_period)
>+        << "The lower pitch period and the additional sub-harmonic must not "
>+        << "coincide.";
>+    // Compute an auto-correlation score for the primary pitch candidate
>+    // |candidate_pitch_period| by also looking at its possible sub-harmonic
>+    // |candidate_pitch_secondary_period|.
>+    float xy_primary_period = ComputeAutoCorrelationCoeff(
>+        pitch_buf, GetInvertedLag(candidate_pitch_period), kMaxPitch24kHz);
>+    float xy_secondary_period = ComputeAutoCorrelationCoeff(
>+        pitch_buf, GetInvertedLag(candidate_pitch_secondary_period),
>+        kMaxPitch24kHz);
>+    float xy = 0.5f * (xy_primary_period + xy_secondary_period);
>+    float yy = 0.5f * (yy_values[candidate_pitch_period] +
>+                       yy_values[candidate_pitch_secondary_period]);
>+    float candidate_pitch_gain = pitch_gain(xy, yy, xx);
>+
>+    // Maybe update best period.
>+    float threshold = ComputePitchGainThreshold(
>+        candidate_pitch_period, k, initial_pitch_period, initial_pitch_gain,
>+        prev_pitch_48kHz.period / 2, prev_pitch_48kHz.gain);
>+    if (candidate_pitch_gain > threshold) {
>+      best_pitch = {candidate_pitch_period, candidate_pitch_gain, xy, yy};
>+    }
>+  }
>+
>+  // Final pitch gain and period.
>+  best_pitch.xy = std::max(0.f, best_pitch.xy);
>+  RTC_DCHECK_LE(0.f, best_pitch.yy);
>+  float final_pitch_gain = (best_pitch.yy <= best_pitch.xy)
>+                               ? 1.f
>+                               : best_pitch.xy / (best_pitch.yy + 1.f);
>+  final_pitch_gain = std::min(best_pitch.gain, final_pitch_gain);
>+  size_t final_pitch_period_48kHz = std::max(
>+      kMinPitch48kHz,
>+      PitchPseudoInterpolationLagPitchBuf(best_pitch.period_24kHz, pitch_buf));
>+
>+  return {final_pitch_period_48kHz, final_pitch_gain};
>+}
>+
>+}  // namespace rnn_vad
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.h
>new file mode 100644
>index 00000000000..75f7f17a42f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/pitch_search_internal.h
>@@ -0,0 +1,107 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_PITCH_SEARCH_INTERNAL_H_
>+#define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_PITCH_SEARCH_INTERNAL_H_
>+
>+#include <array>
>+
>+#include "api/array_view.h"
>+#include "common_audio/real_fourier.h"
>+#include "modules/audio_processing/agc2/rnn_vad/common.h"
>+#include "modules/audio_processing/agc2/rnn_vad/pitch_info.h"
>+
>+namespace webrtc {
>+namespace rnn_vad {
>+
>+// The inverted lags for the pitch interval [|kInitialMinPitch12kHz|,
>+// |kMaxPitch12kHz|] are in the range [0, |kNumInvertedLags|].
>+static_assert(kMaxPitch12kHz > kInitialMinPitch12kHz, "");
>+static_assert(kMaxPitch24kHz > kInitialMinPitch24kHz, "");
>+constexpr size_t kNumInvertedLags12kHz = kMaxPitch12kHz - kInitialMinPitch12kHz;
>+constexpr size_t kNumInvertedLags24kHz = kMaxPitch24kHz - kInitialMinPitch24kHz;
>+constexpr int kAutoCorrelationFftOrder = 9;  // Length-512 FFT.
>+
>+static_assert(1 << kAutoCorrelationFftOrder >
>+                  kNumInvertedLags12kHz + kBufSize12kHz - kMaxPitch12kHz,
>+              "");
>+
>+// Performs 2x decimation without any anti-aliasing filter.
>+void Decimate2x(rtc::ArrayView<const float, kBufSize24kHz> src,
>+                rtc::ArrayView<float, kBufSize12kHz> dst);
>+
>+// Computes a gain threshold for a candidate pitch period given the initial and
>+// the previous pitch period and gain estimates and the pitch period ratio used
>+// to derive the candidate pitch period from the initial period.
>+float ComputePitchGainThreshold(size_t candidate_pitch_period,
>+                                size_t pitch_period_ratio,
>+                                size_t initial_pitch_period,
>+                                float initial_pitch_gain,
>+                                size_t prev_pitch_period,
>+                                size_t prev_pitch_gain);
>+
>+// Computes the sum of squared samples for every sliding frame in the pitch
>+// buffer. |yy_values| indexes are lags.
>+//
>+// The pitch buffer is structured as depicted below:
>+// |.........|...........|
>+//      a          b
>+// The part on the left, named "a" contains the oldest samples, whereas "b" the
>+// most recent ones. The size of "a" corresponds to the maximum pitch period,
>+// that of "b" to the frame size (e.g., 16 ms and 20 ms respectively).
>+void ComputeSlidingFrameSquareEnergies(
>+    rtc::ArrayView<const float, kBufSize24kHz> pitch_buf,
>+    rtc::ArrayView<float, kMaxPitch24kHz + 1> yy_values);
>+
>+// Computes the auto-correlation coefficients for a given pitch interval.
>+// |auto_corr| indexes are inverted lags.
>+//
>+// The auto-correlations coefficients are computed as follows:
>+// |.........|...........|  <- pitch buffer
>+//           [ x (fixed) ]
>+// [   y_0   ]
>+//         [ y_{m-1} ]
>+// x and y are sub-array of equal length; x is never moved, whereas y slides.
>+// The cross-correlation between y_0 and x corresponds to the auto-correlation
>+// for the maximum pitch period. Hence, the first value in |auto_corr| has an
>+// inverted lag equal to 0 that corresponds to a lag equal to the maximum pitch
>+// period.
>+void ComputePitchAutoCorrelation(
>+    rtc::ArrayView<const float, kBufSize12kHz> pitch_buf,
>+    size_t max_pitch_period,
>+    rtc::ArrayView<float, kNumInvertedLags12kHz> auto_corr,
>+    webrtc::RealFourier* fft);
>+
>+// Given the auto-correlation coefficients stored according to
>+// ComputePitchAutoCorrelation() (i.e., using inverted lags), returns the best
>+// and the second best pitch periods.
>+std::array<size_t, 2> FindBestPitchPeriods(
>+    rtc::ArrayView<const float> auto_corr,
>+    rtc::ArrayView<const float> pitch_buf,
>+    size_t max_pitch_period);
>+
>+// Refines the pitch period estimation given the pitch buffer |pitch_buf| and
>+// the initial pitch period estimation |inv_lags|. Returns an inverted lag at
>+// 48 kHz.
>+size_t RefinePitchPeriod48kHz(
>+    rtc::ArrayView<const float, kBufSize24kHz> pitch_buf,
>+    rtc::ArrayView<const size_t, 2> inv_lags);
>+
>+// Refines the pitch period estimation and compute the pitch gain. Returns the
>+// refined pitch estimation data at 48 kHz.
>+PitchInfo CheckLowerPitchPeriodsAndComputePitchGain(
>+    rtc::ArrayView<const float, kBufSize24kHz> pitch_buf,
>+    size_t initial_pitch_period_48kHz,
>+    PitchInfo prev_pitch_48kHz);
>+
>+}  // namespace rnn_vad
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_PITCH_SEARCH_INTERNAL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/pitch_search_internal_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/pitch_search_internal_unittest.cc
>new file mode 100644
>index 00000000000..82b48101ff8
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/pitch_search_internal_unittest.cc
>@@ -0,0 +1,266 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/agc2/rnn_vad/pitch_search_internal.h"
>+#include "common_audio/real_fourier.h"
>+
>+#include <array>
>+#include <tuple>
>+
>+#include "modules/audio_processing/agc2/rnn_vad/test_utils.h"
>+// TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
>+// #include "test/fpe_observer.h"
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+namespace rnn_vad {
>+namespace test {
>+namespace {
>+
>+constexpr std::array<size_t, 2> kTestPitchPeriods = {
>+    3 * kMinPitch48kHz / 2, (3 * kMinPitch48kHz + kMaxPitch48kHz) / 2,
>+};
>+constexpr std::array<float, 2> kTestPitchGains = {0.35f, 0.75f};
>+
>+constexpr size_t kNumPitchBufSquareEnergies = 385;
>+constexpr size_t kNumPitchBufAutoCorrCoeffs = 147;
>+constexpr size_t kTestDataSize =
>+    kBufSize24kHz + kNumPitchBufSquareEnergies + kNumPitchBufAutoCorrCoeffs;
>+
>+class TestData {
>+ public:
>+  TestData() {
>+    auto test_data_reader = CreatePitchSearchTestDataReader();
>+    test_data_reader->ReadChunk(test_data_);
>+  }
>+  rtc::ArrayView<const float, kBufSize24kHz> GetPitchBufView() {
>+    return {test_data_.data(), kBufSize24kHz};
>+  }
>+  rtc::ArrayView<const float, kNumPitchBufSquareEnergies>
>+  GetPitchBufSquareEnergiesView() {
>+    return {test_data_.data() + kBufSize24kHz, kNumPitchBufSquareEnergies};
>+  }
>+  rtc::ArrayView<const float, kNumPitchBufAutoCorrCoeffs>
>+  GetPitchBufAutoCorrCoeffsView() {
>+    return {test_data_.data() + kBufSize24kHz + kNumPitchBufSquareEnergies,
>+            kNumPitchBufAutoCorrCoeffs};
>+  }
>+
>+ private:
>+  std::array<float, kTestDataSize> test_data_;
>+};
>+
>+}  // namespace
>+
>+class ComputePitchGainThresholdTest
>+    : public testing::Test,
>+      public ::testing::WithParamInterface<
>+          std::tuple<size_t, size_t, size_t, float, size_t, float, float>> {};
>+
>+TEST_P(ComputePitchGainThresholdTest, BitExactness) {
>+  const auto params = GetParam();
>+  const size_t candidate_pitch_period = std::get<0>(params);
>+  const size_t pitch_period_ratio = std::get<1>(params);
>+  const size_t initial_pitch_period = std::get<2>(params);
>+  const float initial_pitch_gain = std::get<3>(params);
>+  const size_t prev_pitch_period = std::get<4>(params);
>+  const size_t prev_pitch_gain = std::get<5>(params);
>+  const float threshold = std::get<6>(params);
>+
>+  {
>+    // TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
>+    // FloatingPointExceptionObserver fpe_observer;
>+
>+    EXPECT_NEAR(
>+        threshold,
>+        ComputePitchGainThreshold(candidate_pitch_period, pitch_period_ratio,
>+                                  initial_pitch_period, initial_pitch_gain,
>+                                  prev_pitch_period, prev_pitch_gain),
>+        3e-6f);
>+  }
>+}
>+
>+INSTANTIATE_TEST_CASE_P(
>+    RnnVadTest,
>+    ComputePitchGainThresholdTest,
>+    ::testing::Values(
>+        std::make_tuple(31, 7, 219, 0.45649201f, 199, 0.604747f, 0.40000001f),
>+        std::make_tuple(113,
>+                        2,
>+                        226,
>+                        0.20967799f,
>+                        219,
>+                        0.40392199f,
>+                        0.30000001f),
>+        std::make_tuple(63, 2, 126, 0.210788f, 364, 0.098519f, 0.40000001f),
>+        std::make_tuple(30, 5, 152, 0.82356697f, 149, 0.55535901f, 0.700032f),
>+        std::make_tuple(76, 2, 151, 0.79522997f, 151, 0.82356697f, 0.675946f),
>+        std::make_tuple(31, 5, 153, 0.85069299f, 150, 0.79073799f, 0.72308898f),
>+        std::make_tuple(78, 2, 156, 0.72750503f, 153, 0.85069299f, 0.618379f)));
>+
>+TEST(RnnVadTest, ComputeSlidingFrameSquareEnergiesBitExactness) {
>+  TestData test_data;
>+  std::array<float, kNumPitchBufSquareEnergies> computed_output;
>+  {
>+    // TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
>+    // FloatingPointExceptionObserver fpe_observer;
>+    ComputeSlidingFrameSquareEnergies(test_data.GetPitchBufView(),
>+                                      computed_output);
>+  }
>+  auto square_energies_view = test_data.GetPitchBufSquareEnergiesView();
>+  ExpectNearAbsolute({square_energies_view.data(), square_energies_view.size()},
>+                     computed_output, 3e-2f);
>+}
>+
>+TEST(RnnVadTest, ComputePitchAutoCorrelationBitExactness) {
>+  TestData test_data;
>+  std::array<float, kBufSize12kHz> pitch_buf_decimated;
>+  Decimate2x(test_data.GetPitchBufView(), pitch_buf_decimated);
>+  std::array<float, kNumPitchBufAutoCorrCoeffs> computed_output;
>+  {
>+    // TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
>+    // FloatingPointExceptionObserver fpe_observer;
>+    std::unique_ptr<RealFourier> fft =
>+        RealFourier::Create(kAutoCorrelationFftOrder);
>+    ComputePitchAutoCorrelation(pitch_buf_decimated, kMaxPitch12kHz,
>+                                computed_output, fft.get());
>+  }
>+  auto auto_corr_view = test_data.GetPitchBufAutoCorrCoeffsView();
>+  ExpectNearAbsolute({auto_corr_view.data(), auto_corr_view.size()},
>+                     computed_output, 3e-3f);
>+}
>+
>+// Check that the auto correlation function computes the right thing for a
>+// simple use case.
>+TEST(RnnVadTest, ComputePitchAutoCorrelationConstantBuffer) {
>+  // Create constant signal with no pitch.
>+  std::array<float, kBufSize12kHz> pitch_buf_decimated;
>+  std::fill(pitch_buf_decimated.begin(), pitch_buf_decimated.end(), 1.f);
>+
>+  std::array<float, kNumPitchBufAutoCorrCoeffs> computed_output;
>+  {
>+    // TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
>+    // FloatingPointExceptionObserver fpe_observer;
>+    std::unique_ptr<RealFourier> fft =
>+        RealFourier::Create(kAutoCorrelationFftOrder);
>+    ComputePitchAutoCorrelation(pitch_buf_decimated, kMaxPitch12kHz,
>+                                computed_output, fft.get());
>+  }
>+
>+  // The expected output is constantly the length of the fixed 'x'
>+  // array in ComputePitchAutoCorrelation.
>+  std::array<float, kNumPitchBufAutoCorrCoeffs> expected_output;
>+  std::fill(expected_output.begin(), expected_output.end(),
>+            kBufSize12kHz - kMaxPitch12kHz);
>+  ExpectNearAbsolute(expected_output, computed_output, 4e-5f);
>+}
>+
>+TEST(RnnVadTest, FindBestPitchPeriodsBitExactness) {
>+  TestData test_data;
>+  std::array<float, kBufSize12kHz> pitch_buf_decimated;
>+  Decimate2x(test_data.GetPitchBufView(), pitch_buf_decimated);
>+  std::array<size_t, 2> pitch_candidates_inv_lags;
>+  {
>+    // TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
>+    // FloatingPointExceptionObserver fpe_observer;
>+    auto auto_corr_view = test_data.GetPitchBufAutoCorrCoeffsView();
>+    pitch_candidates_inv_lags =
>+        FindBestPitchPeriods({auto_corr_view.data(), auto_corr_view.size()},
>+                             pitch_buf_decimated, kMaxPitch12kHz);
>+  }
>+  const std::array<size_t, 2> expected_output = {140, 142};
>+  EXPECT_EQ(expected_output, pitch_candidates_inv_lags);
>+}
>+
>+TEST(RnnVadTest, RefinePitchPeriod48kHzBitExactness) {
>+  TestData test_data;
>+  std::array<float, kBufSize12kHz> pitch_buf_decimated;
>+  Decimate2x(test_data.GetPitchBufView(), pitch_buf_decimated);
>+  size_t pitch_inv_lag;
>+  {
>+    // TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
>+    // FloatingPointExceptionObserver fpe_observer;
>+    const std::array<size_t, 2> pitch_candidates_inv_lags = {280, 284};
>+    pitch_inv_lag = RefinePitchPeriod48kHz(test_data.GetPitchBufView(),
>+                                           pitch_candidates_inv_lags);
>+  }
>+  EXPECT_EQ(560u, pitch_inv_lag);
>+}
>+
>+class CheckLowerPitchPeriodsAndComputePitchGainTest
>+    : public testing::Test,
>+      public ::testing::WithParamInterface<
>+          std::tuple<size_t, size_t, float, size_t, float>> {};
>+
>+TEST_P(CheckLowerPitchPeriodsAndComputePitchGainTest, BitExactness) {
>+  const auto params = GetParam();
>+  const size_t initial_pitch_period = std::get<0>(params);
>+  const size_t prev_pitch_period = std::get<1>(params);
>+  const float prev_pitch_gain = std::get<2>(params);
>+  const size_t expected_pitch_period = std::get<3>(params);
>+  const float expected_pitch_gain = std::get<4>(params);
>+  TestData test_data;
>+  {
>+    // TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
>+    // FloatingPointExceptionObserver fpe_observer;
>+    const auto computed_output = CheckLowerPitchPeriodsAndComputePitchGain(
>+        test_data.GetPitchBufView(), initial_pitch_period,
>+        {prev_pitch_period, prev_pitch_gain});
>+    EXPECT_EQ(expected_pitch_period, computed_output.period);
>+    EXPECT_NEAR(expected_pitch_gain, computed_output.gain, 1e-6f);
>+  }
>+}
>+
>+INSTANTIATE_TEST_CASE_P(RnnVadTest,
>+                        CheckLowerPitchPeriodsAndComputePitchGainTest,
>+                        ::testing::Values(std::make_tuple(kTestPitchPeriods[0],
>+                                                          kTestPitchPeriods[0],
>+                                                          kTestPitchGains[0],
>+                                                          91,
>+                                                          -0.0188608f),
>+                                          std::make_tuple(kTestPitchPeriods[0],
>+                                                          kTestPitchPeriods[0],
>+                                                          kTestPitchGains[1],
>+                                                          91,
>+                                                          -0.0188608f),
>+                                          std::make_tuple(kTestPitchPeriods[0],
>+                                                          kTestPitchPeriods[1],
>+                                                          kTestPitchGains[0],
>+                                                          91,
>+                                                          -0.0188608f),
>+                                          std::make_tuple(kTestPitchPeriods[0],
>+                                                          kTestPitchPeriods[1],
>+                                                          kTestPitchGains[1],
>+                                                          91,
>+                                                          -0.0188608f),
>+                                          std::make_tuple(kTestPitchPeriods[1],
>+                                                          kTestPitchPeriods[0],
>+                                                          kTestPitchGains[0],
>+                                                          475,
>+                                                          -0.0904344f),
>+                                          std::make_tuple(kTestPitchPeriods[1],
>+                                                          kTestPitchPeriods[0],
>+                                                          kTestPitchGains[1],
>+                                                          475,
>+                                                          -0.0904344f),
>+                                          std::make_tuple(kTestPitchPeriods[1],
>+                                                          kTestPitchPeriods[1],
>+                                                          kTestPitchGains[0],
>+                                                          475,
>+                                                          -0.0904344f),
>+                                          std::make_tuple(kTestPitchPeriods[1],
>+                                                          kTestPitchPeriods[1],
>+                                                          kTestPitchGains[1],
>+                                                          475,
>+                                                          -0.0904344f)));
>+
>+}  // namespace test
>+}  // namespace rnn_vad
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/pitch_search_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/pitch_search_unittest.cc
>new file mode 100644
>index 00000000000..4c56238dd92
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/pitch_search_unittest.cc
>@@ -0,0 +1,50 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/agc2/rnn_vad/pitch_search.h"
>+#include "modules/audio_processing/agc2/rnn_vad/pitch_info.h"
>+#include "modules/audio_processing/agc2/rnn_vad/pitch_search_internal.h"
>+
>+#include <array>
>+
>+#include "modules/audio_processing/agc2/rnn_vad/test_utils.h"
>+// TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
>+// #include "test/fpe_observer.h"
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+namespace rnn_vad {
>+namespace test {
>+
>+// TODO(bugs.webrtc.org/9076): Remove when the issue is fixed.
>+TEST(RnnVadTest, PitchSearchBitExactness) {
>+  auto lp_residual_reader = CreateLpResidualAndPitchPeriodGainReader();
>+  const size_t num_frames = lp_residual_reader.second;
>+  std::array<float, 864> lp_residual;
>+  float expected_pitch_period, expected_pitch_gain;
>+  PitchEstimator pitch_estimator;
>+  {
>+    // TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
>+    // FloatingPointExceptionObserver fpe_observer;
>+    for (size_t i = 0; i < num_frames; ++i) {
>+      SCOPED_TRACE(i);
>+      lp_residual_reader.first->ReadChunk(lp_residual);
>+      lp_residual_reader.first->ReadValue(&expected_pitch_period);
>+      lp_residual_reader.first->ReadValue(&expected_pitch_gain);
>+      PitchInfo pitch_info = pitch_estimator.Estimate(lp_residual);
>+      EXPECT_EQ(static_cast<size_t>(expected_pitch_period), pitch_info.period);
>+      EXPECT_NEAR(expected_pitch_gain, pitch_info.gain, 1e-5f);
>+    }
>+  }
>+}
>+
>+}  // namespace test
>+}  // namespace rnn_vad
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/ring_buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/ring_buffer.h
>new file mode 100644
>index 00000000000..294b0c0ba87
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/ring_buffer.h
>@@ -0,0 +1,66 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_RING_BUFFER_H_
>+#define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_RING_BUFFER_H_
>+
>+#include <array>
>+#include <cstring>
>+#include <type_traits>
>+
>+#include "api/array_view.h"
>+
>+namespace webrtc {
>+namespace rnn_vad {
>+
>+// Ring buffer for N arrays of type T each one with size S.
>+template <typename T, size_t S, size_t N>
>+class RingBuffer {
>+  static_assert(S > 0, "");
>+  static_assert(N > 0, "");
>+  static_assert(std::is_arithmetic<T>::value,
>+                "Integral or floating point required.");
>+
>+ public:
>+  RingBuffer() : tail_(0) {}
>+  RingBuffer(const RingBuffer&) = delete;
>+  RingBuffer& operator=(const RingBuffer&) = delete;
>+  ~RingBuffer() = default;
>+  // Set the ring buffer values to zero.
>+  void Reset() { buffer_.fill(0); }
>+  // Replace the least recently pushed array in the buffer with |new_values|.
>+  void Push(rtc::ArrayView<const T, S> new_values) {
>+    std::memcpy(buffer_.data() + S * tail_, new_values.data(), S * sizeof(T));
>+    tail_ += 1;
>+    if (tail_ == N)
>+      tail_ = 0;
>+  }
>+  // Return an array view onto the array with a given delay. A view on the last
>+  // and least recently push array is returned when |delay| is 0 and N - 1
>+  // respectively.
>+  rtc::ArrayView<const T, S> GetArrayView(size_t delay) const {
>+    const int delay_int = static_cast<int>(delay);
>+    RTC_DCHECK_LE(0, delay_int);
>+    RTC_DCHECK_LT(delay_int, N);
>+    int offset = tail_ - 1 - delay_int;
>+    if (offset < 0)
>+      offset += N;
>+    return {buffer_.data() + S * offset, S};
>+  }
>+
>+ private:
>+  int tail_;  // Index of the least recently pushed sub-array.
>+  std::array<T, S * N> buffer_{};
>+};
>+
>+}  // namespace rnn_vad
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_RING_BUFFER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/ring_buffer_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/ring_buffer_unittest.cc
>new file mode 100644
>index 00000000000..f0646517985
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/ring_buffer_unittest.cc
>@@ -0,0 +1,115 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/agc2/rnn_vad/ring_buffer.h"
>+
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+namespace rnn_vad {
>+namespace test {
>+namespace {
>+
>+// Compare the elements of two given array views.
>+template <typename T, std::ptrdiff_t S>
>+void ExpectEq(rtc::ArrayView<const T, S> a, rtc::ArrayView<const T, S> b) {
>+  for (size_t i = 0; i < S; ++i) {
>+    SCOPED_TRACE(i);
>+    EXPECT_EQ(a[i], b[i]);
>+  }
>+}
>+
>+// Test push/read sequences.
>+template <typename T, size_t S, size_t N>
>+void TestRingBuffer() {
>+  SCOPED_TRACE(N);
>+  SCOPED_TRACE(S);
>+  std::array<T, S> prev_pushed_array;
>+  std::array<T, S> pushed_array;
>+  rtc::ArrayView<const T, S> pushed_array_view(pushed_array.data(), S);
>+
>+  // Init.
>+  RingBuffer<T, S, N> ring_buf;
>+  ring_buf.GetArrayView(0);
>+  pushed_array.fill(0);
>+  ring_buf.Push(pushed_array_view);
>+  ExpectEq(pushed_array_view, ring_buf.GetArrayView(0));
>+
>+  // Push N times and check most recent and second most recent.
>+  for (T v = 1; v <= static_cast<T>(N); ++v) {
>+    SCOPED_TRACE(v);
>+    prev_pushed_array = pushed_array;
>+    pushed_array.fill(v);
>+    ring_buf.Push(pushed_array_view);
>+    ExpectEq(pushed_array_view, ring_buf.GetArrayView(0));
>+    if (N > 1) {
>+      pushed_array.fill(v - 1);
>+      ExpectEq(pushed_array_view, ring_buf.GetArrayView(1));
>+    }
>+  }
>+
>+  // Check buffer.
>+  for (size_t delay = 2; delay < N; ++delay) {
>+    SCOPED_TRACE(delay);
>+    T expected_value = N - static_cast<T>(delay);
>+    pushed_array.fill(expected_value);
>+    ExpectEq(pushed_array_view, ring_buf.GetArrayView(delay));
>+  }
>+}
>+
>+}  // namespace
>+
>+// Check that for different delays, different views are returned.
>+TEST(RnnVadTest, RingBufferArrayViews) {
>+  constexpr size_t s = 3;
>+  constexpr size_t n = 4;
>+  RingBuffer<int, s, n> ring_buf;
>+  std::array<int, s> pushed_array;
>+  pushed_array.fill(1);
>+  for (size_t k = 0; k <= n; ++k) {  // Push data n + 1 times.
>+    SCOPED_TRACE(k);
>+    // Check array views.
>+    for (size_t i = 0; i < n; ++i) {
>+      SCOPED_TRACE(i);
>+      auto view_i = ring_buf.GetArrayView(i);
>+      for (size_t j = i + 1; j < n; ++j) {
>+        SCOPED_TRACE(j);
>+        auto view_j = ring_buf.GetArrayView(j);
>+        EXPECT_NE(view_i, view_j);
>+      }
>+    }
>+    ring_buf.Push(pushed_array);
>+  }
>+}
>+
>+TEST(RnnVadTest, RingBufferUnsigned) {
>+  TestRingBuffer<uint8_t, 1, 1>();
>+  TestRingBuffer<uint8_t, 2, 5>();
>+  TestRingBuffer<uint8_t, 5, 2>();
>+  TestRingBuffer<uint8_t, 5, 5>();
>+}
>+
>+TEST(RnnVadTest, RingBufferSigned) {
>+  TestRingBuffer<int, 1, 1>();
>+  TestRingBuffer<int, 2, 5>();
>+  TestRingBuffer<int, 5, 2>();
>+  TestRingBuffer<int, 5, 5>();
>+}
>+
>+TEST(RnnVadTest, RingBufferFloating) {
>+  TestRingBuffer<float, 1, 1>();
>+  TestRingBuffer<float, 2, 5>();
>+  TestRingBuffer<float, 5, 2>();
>+  TestRingBuffer<float, 5, 5>();
>+}
>+
>+}  // namespace test
>+}  // namespace rnn_vad
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/rnn.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/rnn.cc
>new file mode 100644
>index 00000000000..2b360347342
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/rnn.cc
>@@ -0,0 +1,232 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/agc2/rnn_vad/rnn.h"
>+
>+#include <algorithm>
>+#include <array>
>+#include <cmath>
>+
>+#include "rtc_base/checks.h"
>+#include "third_party/rnnoise/src/rnn_activations.h"
>+#include "third_party/rnnoise/src/rnn_vad_weights.h"
>+
>+namespace webrtc {
>+namespace rnn_vad {
>+
>+using rnnoise::kWeightsScale;
>+
>+using rnnoise::kInputLayerInputSize;
>+static_assert(kFeatureVectorSize == kInputLayerInputSize, "");
>+using rnnoise::kInputDenseWeights;
>+using rnnoise::kInputDenseBias;
>+using rnnoise::kInputLayerOutputSize;
>+static_assert(kInputLayerOutputSize <= kFullyConnectedLayersMaxUnits,
>+              "Increase kFullyConnectedLayersMaxUnits.");
>+
>+using rnnoise::kHiddenGruRecurrentWeights;
>+using rnnoise::kHiddenGruWeights;
>+using rnnoise::kHiddenGruBias;
>+using rnnoise::kHiddenLayerOutputSize;
>+static_assert(kHiddenLayerOutputSize <= kRecurrentLayersMaxUnits,
>+              "Increase kRecurrentLayersMaxUnits.");
>+
>+using rnnoise::kOutputDenseWeights;
>+using rnnoise::kOutputDenseBias;
>+using rnnoise::kOutputLayerOutputSize;
>+static_assert(kOutputLayerOutputSize <= kFullyConnectedLayersMaxUnits,
>+              "Increase kFullyConnectedLayersMaxUnits.");
>+
>+using rnnoise::RectifiedLinearUnit;
>+using rnnoise::SigmoidApproximated;
>+using rnnoise::TansigApproximated;
>+
>+FullyConnectedLayer::FullyConnectedLayer(
>+    const size_t input_size,
>+    const size_t output_size,
>+    const rtc::ArrayView<const int8_t> bias,
>+    const rtc::ArrayView<const int8_t> weights,
>+    float (*const activation_function)(float))
>+    : input_size_(input_size),
>+      output_size_(output_size),
>+      bias_(bias),
>+      weights_(weights),
>+      activation_function_(activation_function) {
>+  RTC_DCHECK_LE(output_size_, kFullyConnectedLayersMaxUnits)
>+      << "Static over-allocation of fully-connected layers output vectors is "
>+         "not sufficient.";
>+  RTC_DCHECK_EQ(output_size_, bias_.size())
>+      << "Mismatching output size and bias terms array size.";
>+  RTC_DCHECK_EQ(input_size_ * output_size_, weights_.size())
>+      << "Mismatching input-output size and weight coefficients array size.";
>+}
>+
>+FullyConnectedLayer::~FullyConnectedLayer() = default;
>+
>+rtc::ArrayView<const float> FullyConnectedLayer::GetOutput() const {
>+  return rtc::ArrayView<const float>(output_.data(), output_size_);
>+}
>+
>+void FullyConnectedLayer::ComputeOutput(rtc::ArrayView<const float> input) {
>+  // TODO(bugs.chromium.org/9076): Optimize using SSE/AVX fused multiply-add
>+  // operations.
>+  for (size_t o = 0; o < output_size_; ++o) {
>+    output_[o] = bias_[o];
>+    // TODO(bugs.chromium.org/9076): Benchmark how different layouts for
>+    // |weights_| change the performance across different platforms.
>+    for (size_t i = 0; i < input_size_; ++i) {
>+      output_[o] += input[i] * weights_[i * output_size_ + o];
>+    }
>+    output_[o] = (*activation_function_)(kWeightsScale * output_[o]);
>+  }
>+}
>+
>+GatedRecurrentLayer::GatedRecurrentLayer(
>+    const size_t input_size,
>+    const size_t output_size,
>+    const rtc::ArrayView<const int8_t> bias,
>+    const rtc::ArrayView<const int8_t> weights,
>+    const rtc::ArrayView<const int8_t> recurrent_weights,
>+    float (*const activation_function)(float))
>+    : input_size_(input_size),
>+      output_size_(output_size),
>+      bias_(bias),
>+      weights_(weights),
>+      recurrent_weights_(recurrent_weights),
>+      activation_function_(activation_function) {
>+  RTC_DCHECK_LE(output_size_, kRecurrentLayersMaxUnits)
>+      << "Static over-allocation of recurrent layers state vectors is not "
>+      << "sufficient.";
>+  RTC_DCHECK_EQ(3 * output_size_, bias_.size())
>+      << "Mismatching output size and bias terms array size.";
>+  RTC_DCHECK_EQ(3 * input_size_ * output_size_, weights_.size())
>+      << "Mismatching input-output size and weight coefficients array size.";
>+  RTC_DCHECK_EQ(3 * input_size_ * output_size_, recurrent_weights_.size())
>+      << "Mismatching input-output size and recurrent weight coefficients array"
>+      << " size.";
>+  Reset();
>+}
>+
>+GatedRecurrentLayer::~GatedRecurrentLayer() = default;
>+
>+rtc::ArrayView<const float> GatedRecurrentLayer::GetOutput() const {
>+  return rtc::ArrayView<const float>(state_.data(), output_size_);
>+}
>+
>+void GatedRecurrentLayer::Reset() {
>+  state_.fill(0.f);
>+}
>+
>+void GatedRecurrentLayer::ComputeOutput(rtc::ArrayView<const float> input) {
>+  // TODO(bugs.chromium.org/9076): Optimize using SSE/AVX fused multiply-add
>+  // operations.
>+  // Stride and offset used to read parameter arrays.
>+  const size_t stride = 3 * output_size_;
>+  size_t offset = 0;
>+
>+  // Compute update gates.
>+  std::array<float, kRecurrentLayersMaxUnits> update;
>+  for (size_t o = 0; o < output_size_; ++o) {
>+    update[o] = bias_[o];
>+    // TODO(bugs.chromium.org/9076): Benchmark how different layouts for
>+    // |weights_| and |recurrent_weights_| change the performance across
>+    // different platforms.
>+    for (size_t i = 0; i < input_size_; ++i) {  // Add input.
>+      update[o] += input[i] * weights_[i * stride + o];
>+    }
>+    for (size_t s = 0; s < output_size_; ++s) {
>+      update[o] += state_[s] * recurrent_weights_[s * stride + o];
>+    }  // Add state.
>+    update[o] = SigmoidApproximated(kWeightsScale * update[o]);
>+  }
>+
>+  // Compute reset gates.
>+  offset += output_size_;
>+  std::array<float, kRecurrentLayersMaxUnits> reset;
>+  for (size_t o = 0; o < output_size_; ++o) {
>+    reset[o] = bias_[offset + o];
>+    for (size_t i = 0; i < input_size_; ++i) {  // Add input.
>+      reset[o] += input[i] * weights_[offset + i * stride + o];
>+    }
>+    for (size_t s = 0; s < output_size_; ++s) {  // Add state.
>+      reset[o] += state_[s] * recurrent_weights_[offset + s * stride + o];
>+    }
>+    reset[o] = SigmoidApproximated(kWeightsScale * reset[o]);
>+  }
>+
>+  // Compute output.
>+  offset += output_size_;
>+  std::array<float, kRecurrentLayersMaxUnits> output;
>+  for (size_t o = 0; o < output_size_; ++o) {
>+    output[o] = bias_[offset + o];
>+    for (size_t i = 0; i < input_size_; ++i) {  // Add input.
>+      output[o] += input[i] * weights_[offset + i * stride + o];
>+    }
>+    for (size_t s = 0; s < output_size_;
>+         ++s) {  // Add state through reset gates.
>+      output[o] +=
>+          state_[s] * recurrent_weights_[offset + s * stride + o] * reset[s];
>+    }
>+    output[o] = (*activation_function_)(kWeightsScale * output[o]);
>+    // Update output through the update gates.
>+    output[o] = update[o] * state_[o] + (1.f - update[o]) * output[o];
>+  }
>+
>+  // Update the state. Not done in the previous loop since that would pollute
>+  // the current state and lead to incorrect output values.
>+  std::copy(output.begin(), output.end(), state_.begin());
>+}
>+
>+RnnBasedVad::RnnBasedVad()
>+    : input_layer_(kInputLayerInputSize,
>+                   kInputLayerOutputSize,
>+                   kInputDenseBias,
>+                   kInputDenseWeights,
>+                   TansigApproximated),
>+      hidden_layer_(kInputLayerOutputSize,
>+                    kHiddenLayerOutputSize,
>+                    kHiddenGruBias,
>+                    kHiddenGruWeights,
>+                    kHiddenGruRecurrentWeights,
>+                    RectifiedLinearUnit),
>+      output_layer_(kHiddenLayerOutputSize,
>+                    kOutputLayerOutputSize,
>+                    kOutputDenseBias,
>+                    kOutputDenseWeights,
>+                    SigmoidApproximated) {
>+  // Input-output chaining size checks.
>+  RTC_DCHECK_EQ(input_layer_.output_size(), hidden_layer_.input_size())
>+      << "The input and the hidden layers sizes do not match.";
>+  RTC_DCHECK_EQ(hidden_layer_.output_size(), output_layer_.input_size())
>+      << "The hidden and the output layers sizes do not match.";
>+}
>+
>+RnnBasedVad::~RnnBasedVad() = default;
>+
>+void RnnBasedVad::Reset() {
>+  hidden_layer_.Reset();
>+}
>+
>+float RnnBasedVad::ComputeVadProbability(
>+    rtc::ArrayView<const float, kFeatureVectorSize> feature_vector,
>+    bool is_silence) {
>+  if (is_silence) {
>+    Reset();
>+    return 0.f;
>+  }
>+  input_layer_.ComputeOutput(feature_vector);
>+  hidden_layer_.ComputeOutput(input_layer_.GetOutput());
>+  output_layer_.ComputeOutput(hidden_layer_.GetOutput());
>+  const auto vad_output = output_layer_.GetOutput();
>+  return vad_output[0];
>+}
>+
>+}  // namespace rnn_vad
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/rnn.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/rnn.h
>new file mode 100644
>index 00000000000..b3a3b9c485c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/rnn.h
>@@ -0,0 +1,115 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_RNN_H_
>+#define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_RNN_H_
>+
>+#include <array>
>+
>+#include "api/array_view.h"
>+#include "modules/audio_processing/agc2/rnn_vad/common.h"
>+
>+namespace webrtc {
>+namespace rnn_vad {
>+
>+// Maximum number of units for a fully-connected layer. This value is used to
>+// over-allocate space for fully-connected layers output vectors (implemented as
>+// std::array). The value should equal the number of units of the largest
>+// fully-connected layer.
>+constexpr size_t kFullyConnectedLayersMaxUnits = 24;
>+
>+// Maximum number of units for a recurrent layer. This value is used to
>+// over-allocate space for recurrent layers state vectors (implemented as
>+// std::array). The value should equal the number of units of the largest
>+// recurrent layer.
>+constexpr size_t kRecurrentLayersMaxUnits = 24;
>+
>+// Fully-connected layer.
>+class FullyConnectedLayer {
>+ public:
>+  FullyConnectedLayer(const size_t input_size,
>+                      const size_t output_size,
>+                      const rtc::ArrayView<const int8_t> bias,
>+                      const rtc::ArrayView<const int8_t> weights,
>+                      float (*const activation_function)(float));
>+  FullyConnectedLayer(const FullyConnectedLayer&) = delete;
>+  FullyConnectedLayer& operator=(const FullyConnectedLayer&) = delete;
>+  ~FullyConnectedLayer();
>+  size_t input_size() const { return input_size_; }
>+  size_t output_size() const { return output_size_; }
>+  rtc::ArrayView<const float> GetOutput() const;
>+  // Computes the fully-connected layer output.
>+  void ComputeOutput(rtc::ArrayView<const float> input);
>+
>+ private:
>+  const size_t input_size_;
>+  const size_t output_size_;
>+  const rtc::ArrayView<const int8_t> bias_;
>+  const rtc::ArrayView<const int8_t> weights_;
>+  float (*const activation_function_)(float);
>+  // The output vector of a recurrent layer has length equal to |output_size_|.
>+  // However, for efficiency, over-allocation is used.
>+  std::array<float, kFullyConnectedLayersMaxUnits> output_;
>+};
>+
>+// Recurrent layer with gated recurrent units (GRUs).
>+class GatedRecurrentLayer {
>+ public:
>+  GatedRecurrentLayer(const size_t input_size,
>+                      const size_t output_size,
>+                      const rtc::ArrayView<const int8_t> bias,
>+                      const rtc::ArrayView<const int8_t> weights,
>+                      const rtc::ArrayView<const int8_t> recurrent_weights,
>+                      float (*const activation_function)(float));
>+  GatedRecurrentLayer(const GatedRecurrentLayer&) = delete;
>+  GatedRecurrentLayer& operator=(const GatedRecurrentLayer&) = delete;
>+  ~GatedRecurrentLayer();
>+  size_t input_size() const { return input_size_; }
>+  size_t output_size() const { return output_size_; }
>+  rtc::ArrayView<const float> GetOutput() const;
>+  void Reset();
>+  // Computes the recurrent layer output and updates the status.
>+  void ComputeOutput(rtc::ArrayView<const float> input);
>+
>+ private:
>+  const size_t input_size_;
>+  const size_t output_size_;
>+  const rtc::ArrayView<const int8_t> bias_;
>+  const rtc::ArrayView<const int8_t> weights_;
>+  const rtc::ArrayView<const int8_t> recurrent_weights_;
>+  float (*const activation_function_)(float);
>+  // The state vector of a recurrent layer has length equal to |output_size_|.
>+  // However, to avoid dynamic allocation, over-allocation is used.
>+  std::array<float, kRecurrentLayersMaxUnits> state_;
>+};
>+
>+// Recurrent network based VAD.
>+class RnnBasedVad {
>+ public:
>+  RnnBasedVad();
>+  RnnBasedVad(const RnnBasedVad&) = delete;
>+  RnnBasedVad& operator=(const RnnBasedVad&) = delete;
>+  ~RnnBasedVad();
>+  void Reset();
>+  // Compute and returns the probability of voice (range: [0.0, 1.0]).
>+  float ComputeVadProbability(
>+      rtc::ArrayView<const float, kFeatureVectorSize> feature_vector,
>+      bool is_silence);
>+
>+ private:
>+  FullyConnectedLayer input_layer_;
>+  GatedRecurrentLayer hidden_layer_;
>+  FullyConnectedLayer output_layer_;
>+};
>+
>+}  // namespace rnn_vad
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_RNN_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/rnn_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/rnn_unittest.cc
>new file mode 100644
>index 00000000000..289ce8d7596
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/rnn_unittest.cc
>@@ -0,0 +1,179 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <algorithm>
>+#include <array>
>+#include <vector>
>+
>+#include "modules/audio_processing/agc2/rnn_vad/rnn.h"
>+#include "modules/audio_processing/agc2/rnn_vad/test_utils.h"
>+#include "rtc_base/checks.h"
>+#include "test/gtest.h"
>+#include "third_party/rnnoise/src/rnn_activations.h"
>+#include "third_party/rnnoise/src/rnn_vad_weights.h"
>+
>+namespace webrtc {
>+namespace rnn_vad {
>+namespace test {
>+
>+using rnnoise::RectifiedLinearUnit;
>+using rnnoise::SigmoidApproximated;
>+
>+namespace {
>+
>+void TestFullyConnectedLayer(FullyConnectedLayer* fc,
>+                             rtc::ArrayView<const float> input_vector,
>+                             const float expected_output) {
>+  RTC_CHECK(fc);
>+  fc->ComputeOutput(input_vector);
>+  const auto output = fc->GetOutput();
>+  EXPECT_NEAR(expected_output, output[0], 3e-6f);
>+}
>+
>+void TestGatedRecurrentLayer(
>+    GatedRecurrentLayer* gru,
>+    rtc::ArrayView<const float> input_sequence,
>+    rtc::ArrayView<const float> expected_output_sequence) {
>+  RTC_CHECK(gru);
>+  auto gru_output_view = gru->GetOutput();
>+  const size_t input_sequence_length =
>+      rtc::CheckedDivExact(input_sequence.size(), gru->input_size());
>+  const size_t output_sequence_length =
>+      rtc::CheckedDivExact(expected_output_sequence.size(), gru->output_size());
>+  ASSERT_EQ(input_sequence_length, output_sequence_length)
>+      << "The test data length is invalid.";
>+  // Feed the GRU layer and check the output at every step.
>+  gru->Reset();
>+  for (size_t i = 0; i < input_sequence_length; ++i) {
>+    SCOPED_TRACE(i);
>+    gru->ComputeOutput(
>+        input_sequence.subview(i * gru->input_size(), gru->input_size()));
>+    const auto expected_output = expected_output_sequence.subview(
>+        i * gru->output_size(), gru->output_size());
>+    ExpectNearAbsolute(expected_output, gru_output_view, 3e-6f);
>+  }
>+}
>+
>+}  // namespace
>+
>+// Bit-exactness check for fully connected layers.
>+TEST(RnnVadTest, CheckFullyConnectedLayerOutput) {
>+  const std::array<int8_t, 1> bias = {-50};
>+  const std::array<int8_t, 24> weights = {
>+      127,  127,  127, 127,  127,  20,  127,  -126, -126, -54, 14,  125,
>+      -126, -126, 127, -125, -126, 127, -127, -127, -57,  -30, 127, 80};
>+  FullyConnectedLayer fc(24, 1, bias, weights, SigmoidApproximated);
>+  // Test on different inputs.
>+  {
>+    const std::array<float, 24> input_vector = {
>+        0.f,           0.f,           0.f,          0.f,          0.f,
>+        0.f,           0.215833917f,  0.290601075f, 0.238759011f, 0.244751841f,
>+        0.f,           0.0461241305f, 0.106401242f, 0.223070428f, 0.630603909f,
>+        0.690453172f,  0.f,           0.387645692f, 0.166913897f, 0.f,
>+        0.0327451192f, 0.f,           0.136149868f, 0.446351469f};
>+    TestFullyConnectedLayer(&fc, input_vector, 0.436567038f);
>+  }
>+  {
>+    const std::array<float, 24> input_vector = {
>+        0.592162728f,  0.529089332f,  1.18205106f,
>+        1.21736848f,   0.f,           0.470851123f,
>+        0.130675942f,  0.320903003f,  0.305496395f,
>+        0.0571633279f, 1.57001138f,   0.0182026215f,
>+        0.0977443159f, 0.347477973f,  0.493206412f,
>+        0.9688586f,    0.0320267938f, 0.244722098f,
>+        0.312745273f,  0.f,           0.00650715502f,
>+        0.312553257f,  1.62619662f,   0.782880902f};
>+    TestFullyConnectedLayer(&fc, input_vector, 0.874741316f);
>+  }
>+  {
>+    const std::array<float, 24> input_vector = {
>+        0.395022154f,  0.333681047f,  0.76302278f,
>+        0.965480626f,  0.f,           0.941198349f,
>+        0.0892967582f, 0.745046318f,  0.635769248f,
>+        0.238564298f,  0.970656633f,  0.014159563f,
>+        0.094203949f,  0.446816623f,  0.640755892f,
>+        1.20532358f,   0.0254284926f, 0.283327013f,
>+        0.726210058f,  0.0550272502f, 0.000344108557f,
>+        0.369803518f,  1.56680179f,   0.997883797f};
>+    TestFullyConnectedLayer(&fc, input_vector, 0.672785878f);
>+  }
>+}
>+
>+TEST(RnnVadTest, CheckGatedRecurrentLayer) {
>+  const std::array<int8_t, 12> bias = {96,   -99, -81, -114, 49,  119,
>+                                       -118, 68,  -76, 91,   121, 125};
>+  const std::array<int8_t, 60> weights = {
>+      124, 9,    1,    116, -66, -21, -118, -110, 104,  75,  -23,  -51,
>+      -72, -111, 47,   93,  77,  -98, 41,   -8,   40,   -23, -43,  -107,
>+      9,   -73,  30,   -32, -2,  64,  -26,  91,   -48,  -24, -28,  -104,
>+      74,  -46,  116,  15,  32,  52,  -126, -38,  -121, 12,  -16,  110,
>+      -95, 66,   -103, -35, -38, 3,   -126, -61,  28,   98,  -117, -43};
>+  const std::array<int8_t, 60> recurrent_weights = {
>+      -3,  87,  50,  51,  -22,  27,  -39, 62,   31,  -83, -52,  -48,
>+      -6,  83,  -19, 104, 105,  48,  23,  68,   23,  40,  7,    -120,
>+      64,  -62, 117, 85,  -51,  -43, 54,  -105, 120, 56,  -128, -107,
>+      39,  50,  -17, -47, -117, 14,  108, 12,   -7,  -72, 103,  -87,
>+      -66, 82,  84,  100, -98,  102, -49, 44,   122, 106, -20,  -69};
>+  GatedRecurrentLayer gru(5, 4, bias, weights, recurrent_weights,
>+                          RectifiedLinearUnit);
>+  // Test on different inputs.
>+  {
>+    const std::array<float, 20> input_sequence = {
>+        0.89395463f, 0.93224651f, 0.55788344f, 0.32341808f, 0.93355054f,
>+        0.13475326f, 0.97370994f, 0.14253306f, 0.93710381f, 0.76093364f,
>+        0.65780413f, 0.41657975f, 0.49403164f, 0.46843281f, 0.75138855f,
>+        0.24517593f, 0.47657707f, 0.57064998f, 0.435184f,   0.19319285f};
>+    const std::array<float, 16> expected_output_sequence = {
>+        0.0239123f,  0.5773077f,  0.f,         0.f,
>+        0.01282811f, 0.64330572f, 0.f,         0.04863098f,
>+        0.00781069f, 0.75267816f, 0.f,         0.02579715f,
>+        0.00471378f, 0.59162533f, 0.11087593f, 0.01334511f};
>+    TestGatedRecurrentLayer(&gru, input_sequence, expected_output_sequence);
>+  }
>+}
>+
>+// TODO(bugs.webrtc.org/9076): Remove when the issue is fixed.
>+// Bit-exactness test checking that precomputed frame-wise features lead to the
>+// expected VAD probabilities.
>+TEST(RnnVadTest, RnnBitExactness) {
>+  // Init.
>+  auto features_reader = CreateSilenceFlagsFeatureMatrixReader();
>+  auto vad_probs_reader = CreateVadProbsReader();
>+  ASSERT_EQ(features_reader.second, vad_probs_reader.second);
>+  const size_t num_frames = features_reader.second;
>+  // Frame-wise buffers.
>+  float expected_vad_probability;
>+  float is_silence;
>+  std::array<float, kFeatureVectorSize> features;
>+
>+  // Compute VAD probability using the precomputed features.
>+  RnnBasedVad vad;
>+  for (size_t i = 0; i < num_frames; ++i) {
>+    SCOPED_TRACE(i);
>+    // Read frame data.
>+    RTC_CHECK(vad_probs_reader.first->ReadValue(&expected_vad_probability));
>+    // The features file also includes a silence flag for each frame.
>+    RTC_CHECK(features_reader.first->ReadValue(&is_silence));
>+    RTC_CHECK(features_reader.first->ReadChunk(features));
>+    // Compute and check VAD probability.
>+    float vad_probability = vad.ComputeVadProbability(features, is_silence);
>+    ASSERT_TRUE(is_silence == 0.f || is_silence == 1.f);
>+    if (is_silence == 1.f) {
>+      ASSERT_EQ(0.f, expected_vad_probability);
>+      EXPECT_EQ(0.f, vad_probability);
>+    } else {
>+      EXPECT_NEAR(expected_vad_probability, vad_probability, 3e-6f);
>+    }
>+  }
>+}
>+
>+}  // namespace test
>+}  // namespace rnn_vad
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_tool.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_tool.cc
>new file mode 100644
>index 00000000000..5fba0bfac87
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_tool.cc
>@@ -0,0 +1,133 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <array>
>+#include <string>
>+#include <vector>
>+
>+#include "common_audio/resampler/push_sinc_resampler.h"
>+#include "common_audio/wav_file.h"
>+#include "modules/audio_processing/agc2/rnn_vad/common.h"
>+#include "modules/audio_processing/agc2/rnn_vad/features_extraction.h"
>+#include "modules/audio_processing/agc2/rnn_vad/rnn.h"
>+#include "rtc_base/flags.h"
>+#include "rtc_base/logging.h"
>+
>+namespace webrtc {
>+namespace rnn_vad {
>+namespace test {
>+namespace {
>+
>+DEFINE_string(i, "", "Path to the input wav file");
>+std::string InputWavFile() {
>+  return static_cast<std::string>(FLAG_i);
>+}
>+
>+DEFINE_string(f, "", "Path to the output features file");
>+std::string OutputFeaturesFile() {
>+  return static_cast<std::string>(FLAG_f);
>+}
>+
>+DEFINE_string(o, "", "Path to the output VAD probabilities file");
>+std::string OutputVadProbsFile() {
>+  return static_cast<std::string>(FLAG_o);
>+}
>+
>+DEFINE_bool(help, false, "Prints this message");
>+
>+}  // namespace
>+
>+int main(int argc, char* argv[]) {
>+  rtc::LogMessage::LogToDebug(rtc::LS_INFO);
>+  rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true);
>+  if (FLAG_help) {
>+    rtc::FlagList::Print(nullptr, false);
>+    return 0;
>+  }
>+
>+  // Open wav input file and check properties.
>+  WavReader wav_reader(InputWavFile());
>+  if (wav_reader.num_channels() != 1) {
>+    RTC_LOG(LS_ERROR) << "Only mono wav files are supported";
>+    return 1;
>+  }
>+  if (wav_reader.sample_rate() % 100 != 0) {
>+    RTC_LOG(LS_ERROR) << "The sample rate rate must allow 10 ms frames.";
>+    return 1;
>+  }
>+  RTC_LOG(LS_INFO) << "Input sample rate: " << wav_reader.sample_rate();
>+
>+  // Init output files.
>+  FILE* vad_probs_file = fopen(OutputVadProbsFile().c_str(), "wb");
>+  FILE* features_file = nullptr;
>+  const std::string output_feature_file = OutputFeaturesFile();
>+  if (!output_feature_file.empty()) {
>+    features_file = fopen(output_feature_file.c_str(), "wb");
>+  }
>+
>+  // Initialize.
>+  const size_t frame_size_10ms =
>+      rtc::CheckedDivExact(wav_reader.sample_rate(), 100);
>+  std::vector<float> samples_10ms;
>+  samples_10ms.resize(frame_size_10ms);
>+  std::array<float, kFrameSize10ms24kHz> samples_10ms_24kHz;
>+  PushSincResampler resampler(frame_size_10ms, kFrameSize10ms24kHz);
>+  FeaturesExtractor features_extractor;
>+  std::array<float, kFeatureVectorSize> feature_vector;
>+  RnnBasedVad rnn_vad;
>+
>+  // Compute VAD probabilities.
>+  while (true) {
>+    // Read frame at the input sample rate.
>+    const auto read_samples =
>+        wav_reader.ReadSamples(frame_size_10ms, samples_10ms.data());
>+    if (read_samples < frame_size_10ms) {
>+      break;  // EOF.
>+    }
>+    // Resample input.
>+    resampler.Resample(samples_10ms.data(), samples_10ms.size(),
>+                       samples_10ms_24kHz.data(), samples_10ms_24kHz.size());
>+
>+    // Extract features and feed the RNN.
>+    bool is_silence = features_extractor.CheckSilenceComputeFeatures(
>+        samples_10ms_24kHz, feature_vector);
>+    float vad_probability =
>+        rnn_vad.ComputeVadProbability(feature_vector, is_silence);
>+    // Write voice probability.
>+    RTC_DCHECK_GE(vad_probability, 0.f);
>+    RTC_DCHECK_GE(1.f, vad_probability);
>+    fwrite(&vad_probability, sizeof(float), 1, vad_probs_file);
>+    // Write features.
>+    if (features_file) {
>+      const float float_is_silence = is_silence ? 1.f : 0.f;
>+      fwrite(&float_is_silence, sizeof(float), 1, features_file);
>+      fwrite(feature_vector.data(), sizeof(float), kFeatureVectorSize,
>+             features_file);
>+    }
>+  }
>+
>+  // Close output file(s).
>+  fclose(vad_probs_file);
>+  RTC_LOG(LS_INFO) << "VAD probabilities written to " << FLAG_o;
>+  if (features_file) {
>+    fclose(features_file);
>+    RTC_LOG(LS_INFO) << "features written to " << FLAG_f;
>+  }
>+
>+  return 0;
>+}
>+
>+}  // namespace test
>+}  // namespace rnn_vad
>+}  // namespace webrtc
>+
>+int main(int argc, char* argv[]) {
>+  return webrtc::rnn_vad::test::main(argc, argv);
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_unittest.cc
>new file mode 100644
>index 00000000000..4afe24b9f10
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/rnn_vad_unittest.cc
>@@ -0,0 +1,96 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <array>
>+#include <vector>
>+
>+#include "common_audio/resampler/push_sinc_resampler.h"
>+#include "modules/audio_processing/agc2/rnn_vad/features_extraction.h"
>+#include "modules/audio_processing/agc2/rnn_vad/rnn.h"
>+#include "modules/audio_processing/agc2/rnn_vad/test_utils.h"
>+#include "modules/audio_processing/test/performance_timer.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/logging.h"
>+#include "test/gtest.h"
>+#include "third_party/rnnoise/src/rnn_activations.h"
>+#include "third_party/rnnoise/src/rnn_vad_weights.h"
>+
>+namespace webrtc {
>+namespace rnn_vad {
>+namespace test {
>+namespace {
>+
>+constexpr size_t kFrameSize10ms48kHz = 480;
>+
>+void DumpPerfStats(size_t num_samples,
>+                   size_t sample_rate,
>+                   double average_us,
>+                   double standard_deviation) {
>+  float audio_track_length_ms =
>+      1e3f * static_cast<float>(num_samples) / static_cast<float>(sample_rate);
>+  float average_ms = static_cast<float>(average_us) / 1e3f;
>+  float speed = audio_track_length_ms / average_ms;
>+  RTC_LOG(LS_INFO) << "track duration (ms): " << audio_track_length_ms;
>+  RTC_LOG(LS_INFO) << "average processing time (ms): " << average_ms << " +/- "
>+                   << (standard_deviation / 1e3);
>+  RTC_LOG(LS_INFO) << "speed: " << speed << "x";
>+}
>+
>+}  // namespace
>+
>+// Performance test for the RNN VAD (pre-fetching and downsampling are
>+// excluded). Keep disabled and only enable locally to measure performance as
>+// follows:
>+// - on desktop: run the this unit test adding "--logs";
>+// - on android: run the this unit test adding "--logcat-output-file".
>+TEST(RnnVadTest, DISABLED_RnnVadPerformance) {
>+  // PCM samples reader and buffers.
>+  auto samples_reader = CreatePcmSamplesReader(kFrameSize10ms48kHz);
>+  const size_t num_frames = samples_reader.second;
>+  std::array<float, kFrameSize10ms48kHz> samples;
>+  // Pre-fetch and decimate samples.
>+  PushSincResampler decimator(kFrameSize10ms48kHz, kFrameSize10ms24kHz);
>+  std::vector<float> prefetched_decimated_samples;
>+  prefetched_decimated_samples.resize(num_frames * kFrameSize10ms24kHz);
>+  for (size_t i = 0; i < num_frames; ++i) {
>+    samples_reader.first->ReadChunk(samples);
>+    decimator.Resample(samples.data(), samples.size(),
>+                       &prefetched_decimated_samples[i * kFrameSize10ms24kHz],
>+                       kFrameSize10ms24kHz);
>+  }
>+  // Initialize.
>+  FeaturesExtractor features_extractor;
>+  std::array<float, kFeatureVectorSize> feature_vector;
>+  RnnBasedVad rnn_vad;
>+  constexpr size_t number_of_tests = 100;
>+  ::webrtc::test::PerformanceTimer perf_timer(number_of_tests);
>+  for (size_t k = 0; k < number_of_tests; ++k) {
>+    features_extractor.Reset();
>+    rnn_vad.Reset();
>+    // Process frames.
>+    perf_timer.StartTimer();
>+    for (size_t i = 0; i < num_frames; ++i) {
>+      bool is_silence = features_extractor.CheckSilenceComputeFeatures(
>+          {&prefetched_decimated_samples[i * kFrameSize10ms24kHz],
>+           kFrameSize10ms24kHz},
>+          feature_vector);
>+      rnn_vad.ComputeVadProbability(feature_vector, is_silence);
>+    }
>+    perf_timer.StopTimer();
>+    samples_reader.first->SeekBeginning();
>+  }
>+  DumpPerfStats(num_frames * kFrameSize10ms24kHz, kSampleRate24kHz,
>+                perf_timer.GetDurationAverage(),
>+                perf_timer.GetDurationStandardDeviation());
>+}
>+
>+}  // namespace test
>+}  // namespace rnn_vad
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/sequence_buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/sequence_buffer.h
>new file mode 100644
>index 00000000000..75d3d9bc092
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/sequence_buffer.h
>@@ -0,0 +1,79 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_SEQUENCE_BUFFER_H_
>+#define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_SEQUENCE_BUFFER_H_
>+
>+#include <algorithm>
>+#include <cstring>
>+#include <type_traits>
>+#include <vector>
>+
>+#include "api/array_view.h"
>+#include "rtc_base/checks.h"
>+
>+namespace webrtc {
>+namespace rnn_vad {
>+
>+// Linear buffer implementation to (i) push fixed size chunks of sequential data
>+// and (ii) view contiguous parts of the buffer. The buffer and the pushed
>+// chunks have size S and N respectively. For instance, when S = 2N the first
>+// half of the sequence buffer is replaced with its second half, and the new N
>+// values are written at the end of the buffer.
>+// The class also provides a view on the most recent M values, where 0 < M <= S
>+// and by default M = N.
>+template <typename T, size_t S, size_t N, size_t M = N>
>+class SequenceBuffer {
>+  static_assert(N <= S,
>+                "The new chunk size cannot be larger than the sequence buffer "
>+                "size.");
>+  static_assert(std::is_arithmetic<T>::value,
>+                "Integral or floating point required.");
>+
>+ public:
>+  SequenceBuffer() : buffer_(S) {
>+    RTC_DCHECK_EQ(S, buffer_.size());
>+    Reset();
>+  }
>+  SequenceBuffer(const SequenceBuffer&) = delete;
>+  SequenceBuffer& operator=(const SequenceBuffer&) = delete;
>+  ~SequenceBuffer() = default;
>+  size_t size() const { return S; }
>+  size_t chunks_size() const { return N; }
>+  // Sets the sequence buffer values to zero.
>+  void Reset() { std::fill(buffer_.begin(), buffer_.end(), 0); }
>+  // Returns a view on the whole buffer.
>+  rtc::ArrayView<const T, S> GetBufferView() const {
>+    return {buffer_.data(), S};
>+  }
>+  // Returns a view on the M most recent values of the buffer.
>+  rtc::ArrayView<const T, M> GetMostRecentValuesView() const {
>+    static_assert(M <= S,
>+                  "The number of most recent values cannot be larger than the "
>+                  "sequence buffer size.");
>+    return {buffer_.data() + S - M, M};
>+  }
>+  // Shifts left the buffer by N items and add new N items at the end.
>+  void Push(rtc::ArrayView<const T, N> new_values) {
>+    // Make space for the new values.
>+    if (S > N)
>+      std::memmove(buffer_.data(), buffer_.data() + N, (S - N) * sizeof(T));
>+    // Copy the new values at the end of the buffer.
>+    std::memcpy(buffer_.data() + S - N, new_values.data(), N * sizeof(T));
>+  }
>+
>+ private:
>+  std::vector<T> buffer_;
>+};
>+
>+}  // namespace rnn_vad
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_SEQUENCE_BUFFER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/sequence_buffer_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/sequence_buffer_unittest.cc
>new file mode 100644
>index 00000000000..9b66dcf701d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/sequence_buffer_unittest.cc
>@@ -0,0 +1,105 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/agc2/rnn_vad/sequence_buffer.h"
>+
>+#include <algorithm>
>+#include <array>
>+
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+namespace rnn_vad {
>+namespace test {
>+namespace {
>+
>+template <typename T, size_t S, size_t N>
>+void TestSequenceBufferPushOp() {
>+  SCOPED_TRACE(S);
>+  SCOPED_TRACE(N);
>+  SequenceBuffer<T, S, N> seq_buf;
>+  auto seq_buf_view = seq_buf.GetBufferView();
>+  std::array<T, N> chunk;
>+
>+  // Check that a chunk is fully gone after ceil(S / N) push ops.
>+  chunk.fill(1);
>+  seq_buf.Push(chunk);
>+  chunk.fill(0);
>+  constexpr size_t required_push_ops = (S % N) ? S / N + 1 : S / N;
>+  for (size_t i = 0; i < required_push_ops - 1; ++i) {
>+    SCOPED_TRACE(i);
>+    seq_buf.Push(chunk);
>+    // Still in the buffer.
>+    const auto* m = std::max_element(seq_buf_view.begin(), seq_buf_view.end());
>+    EXPECT_EQ(1, *m);
>+  }
>+  // Gone after another push.
>+  seq_buf.Push(chunk);
>+  const auto* m = std::max_element(seq_buf_view.begin(), seq_buf_view.end());
>+  EXPECT_EQ(0, *m);
>+
>+  // Check that the last item moves left by N positions after a push op.
>+  if (S > N) {
>+    // Fill in with non-zero values.
>+    for (size_t i = 0; i < N; ++i)
>+      chunk[i] = static_cast<T>(i + 1);
>+    seq_buf.Push(chunk);
>+    // With the next Push(), |last| will be moved left by N positions.
>+    const T last = chunk[N - 1];
>+    for (size_t i = 0; i < N; ++i)
>+      chunk[i] = static_cast<T>(last + i + 1);
>+    seq_buf.Push(chunk);
>+    EXPECT_EQ(last, seq_buf_view[S - N - 1]);
>+  }
>+}
>+
>+}  // namespace
>+
>+TEST(RnnVadTest, SequenceBufferGetters) {
>+  constexpr size_t buffer_size = 8;
>+  constexpr size_t chunk_size = 8;
>+  SequenceBuffer<int, buffer_size, chunk_size> seq_buf;
>+  EXPECT_EQ(buffer_size, seq_buf.size());
>+  EXPECT_EQ(chunk_size, seq_buf.chunks_size());
>+  // Test view.
>+  auto seq_buf_view = seq_buf.GetBufferView();
>+  EXPECT_EQ(0, seq_buf_view[0]);
>+  EXPECT_EQ(0, seq_buf_view[seq_buf_view.size() - 1]);
>+  constexpr std::array<int, chunk_size> chunk = {10, 20, 30, 40,
>+                                                 50, 60, 70, 80};
>+  seq_buf.Push(chunk);
>+  EXPECT_EQ(10, *seq_buf_view.begin());
>+  EXPECT_EQ(80, *(seq_buf_view.end() - 1));
>+}
>+
>+TEST(RnnVadTest, SequenceBufferPushOpsUnsigned) {
>+  TestSequenceBufferPushOp<uint8_t, 32, 8>();   // Chunk size: 25%.
>+  TestSequenceBufferPushOp<uint8_t, 32, 16>();  // Chunk size: 50%.
>+  TestSequenceBufferPushOp<uint8_t, 32, 32>();  // Chunk size: 100%.
>+  TestSequenceBufferPushOp<uint8_t, 23, 7>();   // Non-integer ratio.
>+}
>+
>+TEST(RnnVadTest, SequenceBufferPushOpsSigned) {
>+  TestSequenceBufferPushOp<int, 32, 8>();   // Chunk size: 25%.
>+  TestSequenceBufferPushOp<int, 32, 16>();  // Chunk size: 50%.
>+  TestSequenceBufferPushOp<int, 32, 32>();  // Chunk size: 100%.
>+  TestSequenceBufferPushOp<int, 23, 7>();   // Non-integer ratio.
>+}
>+
>+TEST(RnnVadTest, SequenceBufferPushOpsFloating) {
>+  TestSequenceBufferPushOp<float, 32, 8>();   // Chunk size: 25%.
>+  TestSequenceBufferPushOp<float, 32, 16>();  // Chunk size: 50%.
>+  TestSequenceBufferPushOp<float, 32, 32>();  // Chunk size: 100%.
>+  TestSequenceBufferPushOp<float, 23, 7>();   // Non-integer ratio.
>+}
>+
>+}  // namespace test
>+}  // namespace rnn_vad
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/spectral_features.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/spectral_features.cc
>new file mode 100644
>index 00000000000..695eed57f09
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/spectral_features.cc
>@@ -0,0 +1,198 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/agc2/rnn_vad/spectral_features.h"
>+
>+#include <algorithm>
>+#include <cmath>
>+#include <limits>
>+#include <numeric>
>+
>+#include "modules/audio_processing/agc2/rnn_vad/spectral_features_internal.h"
>+#include "rtc_base/checks.h"
>+
>+namespace webrtc {
>+namespace rnn_vad {
>+namespace {
>+
>+constexpr float kSilenceThreshold = 0.04f;
>+
>+// Computes the new spectral difference stats and pushes them into the passed
>+// symmetric matrix buffer.
>+void UpdateSpectralDifferenceStats(
>+    rtc::ArrayView<const float, kNumBands> new_spectral_coeffs,
>+    const RingBuffer<float, kNumBands, kSpectralCoeffsHistorySize>& ring_buf,
>+    SymmetricMatrixBuffer<float, kSpectralCoeffsHistorySize>* sym_matrix_buf) {
>+  RTC_DCHECK(sym_matrix_buf);
>+  // Compute the new spectral distance stats.
>+  std::array<float, kSpectralCoeffsHistorySize - 1> distances;
>+  for (size_t i = 0; i < kSpectralCoeffsHistorySize - 1; ++i) {
>+    const size_t delay = i + 1;
>+    auto old_spectral_coeffs = ring_buf.GetArrayView(delay);
>+    distances[i] = 0.f;
>+    for (size_t k = 0; k < kNumBands; ++k) {
>+      const float c = new_spectral_coeffs[k] - old_spectral_coeffs[k];
>+      distances[i] += c * c;
>+    }
>+  }
>+  // Push the new spectral distance stats into the symmetric matrix buffer.
>+  sym_matrix_buf->Push(distances);
>+}
>+
>+}  // namespace
>+
>+SpectralFeaturesView::SpectralFeaturesView(
>+    rtc::ArrayView<float, kNumBands - kNumLowerBands> coeffs,
>+    rtc::ArrayView<float, kNumLowerBands> average,
>+    rtc::ArrayView<float, kNumLowerBands> first_derivative,
>+    rtc::ArrayView<float, kNumLowerBands> second_derivative,
>+    rtc::ArrayView<float, kNumLowerBands> cross_correlations,
>+    float* variability)
>+    : coeffs(coeffs),
>+      average(average),
>+      first_derivative(first_derivative),
>+      second_derivative(second_derivative),
>+      cross_correlations(cross_correlations),
>+      variability(variability) {}
>+
>+SpectralFeaturesView::SpectralFeaturesView(const SpectralFeaturesView&) =
>+    default;
>+SpectralFeaturesView::~SpectralFeaturesView() = default;
>+
>+SpectralFeaturesExtractor::SpectralFeaturesExtractor()
>+    : fft_(),
>+      reference_frame_fft_(kFrameSize20ms24kHz),
>+      lagged_frame_fft_(kFrameSize20ms24kHz),
>+      band_boundaries_(
>+          ComputeBandBoundaryIndexes(kSampleRate24kHz, kFrameSize20ms24kHz)),
>+      dct_table_(ComputeDctTable()) {}
>+
>+SpectralFeaturesExtractor::~SpectralFeaturesExtractor() = default;
>+
>+void SpectralFeaturesExtractor::Reset() {
>+  spectral_coeffs_ring_buf_.Reset();
>+  spectral_diffs_buf_.Reset();
>+}
>+
>+bool SpectralFeaturesExtractor::CheckSilenceComputeFeatures(
>+    rtc::ArrayView<const float, kFrameSize20ms24kHz> reference_frame,
>+    rtc::ArrayView<const float, kFrameSize20ms24kHz> lagged_frame,
>+    SpectralFeaturesView spectral_features) {
>+  // Analyze reference frame.
>+  fft_.ForwardFft(reference_frame, reference_frame_fft_);
>+  ComputeBandEnergies(reference_frame_fft_, band_boundaries_,
>+                      reference_frame_energy_coeffs_);
>+  // Check if the reference frame has silence.
>+  const float tot_energy =
>+      std::accumulate(reference_frame_energy_coeffs_.begin(),
>+                      reference_frame_energy_coeffs_.end(), 0.f);
>+  if (tot_energy < kSilenceThreshold)
>+    return true;
>+  // Analyze lagged frame.
>+  fft_.ForwardFft(lagged_frame, lagged_frame_fft_);
>+  ComputeBandEnergies(lagged_frame_fft_, band_boundaries_,
>+                      lagged_frame_energy_coeffs_);
>+  // Log of the band energies for the reference frame.
>+  std::array<float, kNumBands> log_band_energy_coeffs;
>+  ComputeLogBandEnergiesCoefficients(reference_frame_energy_coeffs_,
>+                                     log_band_energy_coeffs);
>+  // Decorrelate band-wise log energy coefficients via DCT.
>+  std::array<float, kNumBands> log_band_energy_coeffs_decorrelated;
>+  ComputeDct(log_band_energy_coeffs, dct_table_,
>+             log_band_energy_coeffs_decorrelated);
>+  // Normalize (based on training set stats).
>+  log_band_energy_coeffs_decorrelated[0] -= 12;
>+  log_band_energy_coeffs_decorrelated[1] -= 4;
>+  // Update the ring buffer and the spectral difference stats.
>+  spectral_coeffs_ring_buf_.Push(log_band_energy_coeffs_decorrelated);
>+  UpdateSpectralDifferenceStats(log_band_energy_coeffs_decorrelated,
>+                                spectral_coeffs_ring_buf_,
>+                                &spectral_diffs_buf_);
>+  // Write the higher bands spectral coefficients.
>+  auto coeffs_src = spectral_coeffs_ring_buf_.GetArrayView(0);
>+  RTC_DCHECK_EQ(coeffs_src.size() - kNumLowerBands,
>+                spectral_features.coeffs.size());
>+  std::copy(coeffs_src.begin() + kNumLowerBands, coeffs_src.end(),
>+            spectral_features.coeffs.begin());
>+  // Compute and write remaining features.
>+  ComputeAvgAndDerivatives(spectral_features.average,
>+                           spectral_features.first_derivative,
>+                           spectral_features.second_derivative);
>+  ComputeCrossCorrelation(spectral_features.cross_correlations);
>+  RTC_DCHECK(spectral_features.variability);
>+  *(spectral_features.variability) = ComputeVariability();
>+  return false;
>+}
>+
>+void SpectralFeaturesExtractor::ComputeAvgAndDerivatives(
>+    rtc::ArrayView<float, kNumLowerBands> average,
>+    rtc::ArrayView<float, kNumLowerBands> first_derivative,
>+    rtc::ArrayView<float, kNumLowerBands> second_derivative) {
>+  auto curr = spectral_coeffs_ring_buf_.GetArrayView(0);
>+  auto prev1 = spectral_coeffs_ring_buf_.GetArrayView(1);
>+  auto prev2 = spectral_coeffs_ring_buf_.GetArrayView(2);
>+  RTC_DCHECK_EQ(average.size(), first_derivative.size());
>+  RTC_DCHECK_EQ(first_derivative.size(), second_derivative.size());
>+  RTC_DCHECK_LE(average.size(), curr.size());
>+  for (size_t i = 0; i < average.size(); ++i) {
>+    // Average, kernel: [1, 1, 1].
>+    average[i] = curr[i] + prev1[i] + prev2[i];
>+    // First derivative, kernel: [1, 0, - 1].
>+    first_derivative[i] = curr[i] - prev2[i];
>+    // Second derivative, Laplacian kernel: [1, -2, 1].
>+    second_derivative[i] = curr[i] - 2 * prev1[i] + prev2[i];
>+  }
>+}
>+
>+void SpectralFeaturesExtractor::ComputeCrossCorrelation(
>+    rtc::ArrayView<float, kNumLowerBands> cross_correlations) {
>+  const auto& x = reference_frame_fft_;
>+  const auto& y = lagged_frame_fft_;
>+  auto cross_corr = [x, y](const size_t freq_bin_index) -> float {
>+    return (x[freq_bin_index].real() * y[freq_bin_index].real() +
>+            x[freq_bin_index].imag() * y[freq_bin_index].imag());
>+  };
>+  std::array<float, kNumBands> cross_corr_coeffs;
>+  constexpr size_t kNumFftPoints = kFrameSize20ms24kHz / 2 + 1;
>+  ComputeBandCoefficients(cross_corr, band_boundaries_, kNumFftPoints - 1,
>+                          cross_corr_coeffs);
>+  // Normalize.
>+  for (size_t i = 0; i < cross_corr_coeffs.size(); ++i) {
>+    cross_corr_coeffs[i] =
>+        cross_corr_coeffs[i] /
>+        std::sqrt(0.001f + reference_frame_energy_coeffs_[i] *
>+                               lagged_frame_energy_coeffs_[i]);
>+  }
>+  // Decorrelate.
>+  ComputeDct(cross_corr_coeffs, dct_table_, cross_correlations);
>+  // Normalize (based on training set stats).
>+  cross_correlations[0] -= 1.3f;
>+  cross_correlations[1] -= 0.9f;
>+}
>+
>+float SpectralFeaturesExtractor::ComputeVariability() {
>+  // Compute spectral variability score.
>+  float spec_variability = 0.f;
>+  for (size_t delay1 = 0; delay1 < kSpectralCoeffsHistorySize; ++delay1) {
>+    float min_dist = std::numeric_limits<float>::max();
>+    for (size_t delay2 = 0; delay2 < kSpectralCoeffsHistorySize; ++delay2) {
>+      if (delay1 == delay2)  // The distance would be 0.
>+        continue;
>+      min_dist =
>+          std::min(min_dist, spectral_diffs_buf_.GetValue(delay1, delay2));
>+    }
>+    spec_variability += min_dist;
>+  }
>+  // Normalize (based on training set stats).
>+  return spec_variability / kSpectralCoeffsHistorySize - 2.1f;
>+}
>+
>+}  // namespace rnn_vad
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/spectral_features.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/spectral_features.h
>new file mode 100644
>index 00000000000..bedd7ab3d7d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/spectral_features.h
>@@ -0,0 +1,92 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_SPECTRAL_FEATURES_H_
>+#define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_SPECTRAL_FEATURES_H_
>+
>+#include <array>
>+#include <complex>
>+#include <vector>
>+
>+#include "api/array_view.h"
>+#include "modules/audio_processing/agc2/rnn_vad/common.h"
>+#include "modules/audio_processing/agc2/rnn_vad/fft_util.h"
>+#include "modules/audio_processing/agc2/rnn_vad/ring_buffer.h"
>+#include "modules/audio_processing/agc2/rnn_vad/symmetric_matrix_buffer.h"
>+
>+namespace webrtc {
>+namespace rnn_vad {
>+
>+// View on spectral features.
>+class SpectralFeaturesView {
>+ public:
>+  SpectralFeaturesView(rtc::ArrayView<float, kNumBands - kNumLowerBands> coeffs,
>+                       rtc::ArrayView<float, kNumLowerBands> average,
>+                       rtc::ArrayView<float, kNumLowerBands> first_derivative,
>+                       rtc::ArrayView<float, kNumLowerBands> second_derivative,
>+                       rtc::ArrayView<float, kNumLowerBands> cross_correlations,
>+                       float* variability);
>+  SpectralFeaturesView(const SpectralFeaturesView&);
>+  ~SpectralFeaturesView();
>+  // Higher bands spectral coefficients.
>+  const rtc::ArrayView<float, kNumBands - kNumLowerBands> coeffs;
>+  // Average and first and second derivative over time for the lower bands.
>+  const rtc::ArrayView<float, kNumLowerBands> average;
>+  const rtc::ArrayView<float, kNumLowerBands> first_derivative;
>+  const rtc::ArrayView<float, kNumLowerBands> second_derivative;
>+  // Spectral cross-correlation for the lower bands.
>+  const rtc::ArrayView<float, kNumLowerBands> cross_correlations;
>+  // Spectral variability score.
>+  float* const variability;
>+};
>+
>+// Class to compute spectral features.
>+class SpectralFeaturesExtractor {
>+ public:
>+  SpectralFeaturesExtractor();
>+  SpectralFeaturesExtractor(const SpectralFeaturesExtractor&) = delete;
>+  SpectralFeaturesExtractor& operator=(const SpectralFeaturesExtractor&) =
>+      delete;
>+  ~SpectralFeaturesExtractor();
>+  // Resets the internal state of the feature extractor.
>+  void Reset();
>+  // Analyzes a pair of reference and lagged frames from the pitch buffer,
>+  // detects silence and computes features. If silence is detected, the output
>+  // is neither computed nor written.
>+  bool CheckSilenceComputeFeatures(
>+      rtc::ArrayView<const float, kFrameSize20ms24kHz> reference_frame,
>+      rtc::ArrayView<const float, kFrameSize20ms24kHz> lagged_frame,
>+      SpectralFeaturesView spectral_features);
>+
>+ private:
>+  void ComputeAvgAndDerivatives(
>+      rtc::ArrayView<float, kNumLowerBands> average,
>+      rtc::ArrayView<float, kNumLowerBands> first_derivative,
>+      rtc::ArrayView<float, kNumLowerBands> second_derivative);
>+  void ComputeCrossCorrelation(
>+      rtc::ArrayView<float, kNumLowerBands> cross_correlations);
>+  float ComputeVariability();
>+
>+  BandAnalysisFft fft_;
>+  std::vector<std::complex<float>> reference_frame_fft_;
>+  std::vector<std::complex<float>> lagged_frame_fft_;
>+  std::array<float, kNumBands> reference_frame_energy_coeffs_{};
>+  std::array<float, kNumBands> lagged_frame_energy_coeffs_{};
>+  const std::array<size_t, kNumBands> band_boundaries_;
>+  const std::array<float, kNumBands * kNumBands> dct_table_;
>+  RingBuffer<float, kNumBands, kSpectralCoeffsHistorySize>
>+      spectral_coeffs_ring_buf_;
>+  SymmetricMatrixBuffer<float, kSpectralCoeffsHistorySize> spectral_diffs_buf_;
>+};
>+
>+}  // namespace rnn_vad
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_SPECTRAL_FEATURES_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.cc
>new file mode 100644
>index 00000000000..4cbe48a29e3
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.cc
>@@ -0,0 +1,128 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/agc2/rnn_vad/spectral_features_internal.h"
>+
>+#include <algorithm>
>+#include <cmath>
>+
>+#include "rtc_base/checks.h"
>+
>+namespace webrtc {
>+namespace rnn_vad {
>+namespace {
>+
>+// DCT scaling factor.
>+const float kDctScalingFactor = std::sqrt(2.f / kNumBands);
>+
>+}  // namespace
>+
>+std::array<size_t, kNumBands> ComputeBandBoundaryIndexes(
>+    size_t sample_rate_hz,
>+    size_t frame_size_samples) {
>+  std::array<size_t, kNumBands> indexes;
>+  for (size_t i = 0; i < kNumBands; ++i) {
>+    indexes[i] =
>+        kBandFrequencyBoundaries[i] * frame_size_samples / sample_rate_hz;
>+  }
>+  return indexes;
>+}
>+
>+void ComputeBandCoefficients(
>+    rtc::FunctionView<float(size_t)> functor,
>+    rtc::ArrayView<const size_t, kNumBands> band_boundaries,
>+    size_t max_freq_bin_index,
>+    rtc::ArrayView<float, kNumBands> coefficients) {
>+  std::fill(coefficients.begin(), coefficients.end(), 0.f);
>+  for (size_t i = 0; i < coefficients.size() - 1; ++i) {
>+    RTC_DCHECK_EQ(0.f, coefficients[i + 1]);
>+    RTC_DCHECK_GT(band_boundaries[i + 1], band_boundaries[i]);
>+    const size_t first_freq_bin = band_boundaries[i];
>+    const size_t last_freq_bin =
>+        std::min(max_freq_bin_index, first_freq_bin + band_boundaries[i + 1] -
>+                                         band_boundaries[i] - 1);
>+    // Depending on the sample rate, the highest bands can have no FFT
>+    // coefficients. Stop the iteration when coming across the first empty band.
>+    if (first_freq_bin >= last_freq_bin)
>+      break;
>+    const size_t band_size = last_freq_bin - first_freq_bin + 1;
>+    // Compute the band coefficient using a triangular band with peak response
>+    // at the band boundary.
>+    for (size_t j = first_freq_bin; j <= last_freq_bin; ++j) {
>+      const float w = static_cast<float>(j - first_freq_bin) / band_size;
>+      const float coefficient = functor(j);
>+      coefficients[i] += (1.f - w) * coefficient;
>+      coefficients[i + 1] += w * coefficient;
>+    }
>+  }
>+  // The first and the last bands in the loop above only got half contribution.
>+  coefficients[0] *= 2.f;
>+  coefficients[coefficients.size() - 1] *= 2.f;
>+  // TODO(bugs.webrtc.org/9076): Replace the line above with
>+  // "coefficients[i] *= 2.f" (*) since we now assume that the last band is
>+  // always |kNumBands| - 1.
>+  // (*): "size_t i" must be declared before the main loop.
>+}
>+
>+void ComputeBandEnergies(
>+    rtc::ArrayView<const std::complex<float>> fft_coeffs,
>+    rtc::ArrayView<const size_t, kNumBands> band_boundaries,
>+    rtc::ArrayView<float, kNumBands> band_energies) {
>+  RTC_DCHECK_EQ(band_boundaries.size(), band_energies.size());
>+  auto functor = [fft_coeffs](const size_t freq_bin_index) -> float {
>+    return std::norm(fft_coeffs[freq_bin_index]);
>+  };
>+  ComputeBandCoefficients(functor, band_boundaries, fft_coeffs.size() - 1,
>+                          band_energies);
>+}
>+
>+void ComputeLogBandEnergiesCoefficients(
>+    rtc::ArrayView<const float, kNumBands> band_energy_coeffs,
>+    rtc::ArrayView<float, kNumBands> log_band_energy_coeffs) {
>+  float log_max = -2.f;
>+  float follow = -2.f;
>+  for (size_t i = 0; i < band_energy_coeffs.size(); ++i) {
>+    log_band_energy_coeffs[i] = std::log10(1e-2f + band_energy_coeffs[i]);
>+    // Smoothing across frequency bands.
>+    log_band_energy_coeffs[i] = std::max(
>+        log_max - 7.f, std::max(follow - 1.5f, log_band_energy_coeffs[i]));
>+    log_max = std::max(log_max, log_band_energy_coeffs[i]);
>+    follow = std::max(follow - 1.5f, log_band_energy_coeffs[i]);
>+  }
>+}
>+
>+std::array<float, kNumBands * kNumBands> ComputeDctTable() {
>+  std::array<float, kNumBands * kNumBands> dct_table;
>+  const double k = std::sqrt(0.5);
>+  for (size_t i = 0; i < kNumBands; ++i) {
>+    for (size_t j = 0; j < kNumBands; ++j)
>+      dct_table[i * kNumBands + j] = std::cos((i + 0.5) * j * kPi / kNumBands);
>+    dct_table[i * kNumBands] *= k;
>+  }
>+  return dct_table;
>+}
>+
>+void ComputeDct(rtc::ArrayView<const float, kNumBands> in,
>+                rtc::ArrayView<const float, kNumBands * kNumBands> dct_table,
>+                rtc::ArrayView<float> out) {
>+  RTC_DCHECK_NE(in.data(), out.data()) << "In-place DCT is not supported.";
>+  RTC_DCHECK_LE(1, out.size());
>+  RTC_DCHECK_LE(out.size(), in.size());
>+  std::fill(out.begin(), out.end(), 0.f);
>+  for (size_t i = 0; i < out.size(); ++i) {
>+    for (size_t j = 0; j < in.size(); ++j) {
>+      out[i] += in[j] * dct_table[j * in.size() + i];
>+    }
>+    out[i] *= kDctScalingFactor;
>+  }
>+}
>+
>+}  // namespace rnn_vad
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.h
>new file mode 100644
>index 00000000000..45bb382078e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/spectral_features_internal.h
>@@ -0,0 +1,65 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_SPECTRAL_FEATURES_INTERNAL_H_
>+#define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_SPECTRAL_FEATURES_INTERNAL_H_
>+
>+#include <array>
>+#include <complex>
>+
>+#include "api/array_view.h"
>+#include "modules/audio_processing/agc2/rnn_vad/common.h"
>+#include "rtc_base/function_view.h"
>+
>+namespace webrtc {
>+namespace rnn_vad {
>+
>+// Computes FFT boundary indexes corresponding to sub-bands.
>+std::array<size_t, kNumBands> ComputeBandBoundaryIndexes(
>+    size_t sample_rate_hz,
>+    size_t frame_size_samples);
>+
>+// Iterates through frequency bands and computes coefficients via |functor| for
>+// triangular bands with peak response at each band boundary. |functor| returns
>+// a floating point value for the FFT coefficient having index equal to the
>+// argument passed to |functor|; that argument is in the range {0, ...
>+// |max_freq_bin_index| - 1}.
>+void ComputeBandCoefficients(
>+    rtc::FunctionView<float(size_t)> functor,
>+    rtc::ArrayView<const size_t, kNumBands> band_boundaries,
>+    const size_t max_freq_bin_index,
>+    rtc::ArrayView<float, kNumBands> coefficients);
>+
>+// Given an array of FFT coefficients and a vector of band boundary indexes,
>+// computes band energy coefficients.
>+void ComputeBandEnergies(
>+    rtc::ArrayView<const std::complex<float>> fft_coeffs,
>+    rtc::ArrayView<const size_t, kNumBands> band_boundaries,
>+    rtc::ArrayView<float, kNumBands> band_energies);
>+
>+// Computes log band energy coefficients.
>+void ComputeLogBandEnergiesCoefficients(
>+    rtc::ArrayView<const float, kNumBands> band_energy_coeffs,
>+    rtc::ArrayView<float, kNumBands> log_band_energy_coeffs);
>+
>+// Creates a DCT table for arrays having size equal to |kNumBands|.
>+std::array<float, kNumBands * kNumBands> ComputeDctTable();
>+
>+// Computes DCT for |in| given a pre-computed DCT table. In-place computation is
>+// not allowed and |out| can be smaller than |in| in order to only compute the
>+// first DCT coefficients.
>+void ComputeDct(rtc::ArrayView<const float, kNumBands> in,
>+                rtc::ArrayView<const float, kNumBands * kNumBands> dct_table,
>+                rtc::ArrayView<float> out);
>+
>+}  // namespace rnn_vad
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_SPECTRAL_FEATURES_INTERNAL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/spectral_features_internal_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/spectral_features_internal_unittest.cc
>new file mode 100644
>index 00000000000..5e769bf6746
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/spectral_features_internal_unittest.cc
>@@ -0,0 +1,125 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/agc2/rnn_vad/spectral_features_internal.h"
>+
>+#include "modules/audio_processing/agc2/rnn_vad/test_utils.h"
>+// TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
>+// #include "test/fpe_observer.h"
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+namespace rnn_vad {
>+namespace test {
>+namespace {
>+
>+constexpr size_t kSampleRate48kHz = 48000;
>+constexpr size_t kFrameSize20ms48kHz = 2 * kSampleRate48kHz / 100;
>+constexpr size_t kFftNumCoeffs20ms48kHz = kFrameSize20ms48kHz / 2 + 1;
>+
>+}  // namespace
>+
>+// TODO(bugs.webrtc.org/9076): Remove this test before closing the issue.
>+// Check that when using precomputed FFT coefficients for frames at 48 kHz, the
>+// output of ComputeBandEnergies() is bit exact.
>+TEST(RnnVadTest, ComputeBandEnergies48kHzBitExactness) {
>+  // Initialize input data reader and buffers.
>+  auto fft_coeffs_reader = CreateFftCoeffsReader();
>+  const size_t num_frames = fft_coeffs_reader.second;
>+  ASSERT_EQ(
>+      kFftNumCoeffs20ms48kHz,
>+      rtc::CheckedDivExact(fft_coeffs_reader.first->data_length(), num_frames) /
>+          2);
>+  std::array<float, kFftNumCoeffs20ms48kHz> fft_coeffs_real;
>+  std::array<float, kFftNumCoeffs20ms48kHz> fft_coeffs_imag;
>+  std::array<std::complex<float>, kFftNumCoeffs20ms48kHz> fft_coeffs;
>+  // Init expected output reader and buffer.
>+  auto band_energies_reader = CreateBandEnergyCoeffsReader();
>+  ASSERT_EQ(num_frames, band_energies_reader.second);
>+  std::array<float, kNumBands> expected_band_energies;
>+  // Init band energies coefficients computation.
>+  const auto band_boundary_indexes =
>+      ComputeBandBoundaryIndexes(kSampleRate48kHz, kFrameSize20ms48kHz);
>+  std::array<float, kNumBands> computed_band_energies;
>+
>+  // Check output for every frame.
>+  {
>+    // TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
>+    // FloatingPointExceptionObserver fpe_observer;
>+    for (size_t i = 0; i < num_frames; ++i) {
>+      SCOPED_TRACE(i);
>+      // Read input.
>+      fft_coeffs_reader.first->ReadChunk(fft_coeffs_real);
>+      fft_coeffs_reader.first->ReadChunk(fft_coeffs_imag);
>+      for (size_t i = 0; i < kFftNumCoeffs20ms48kHz; ++i) {
>+        fft_coeffs[i].real(fft_coeffs_real[i]);
>+        fft_coeffs[i].imag(fft_coeffs_imag[i]);
>+      }
>+      band_energies_reader.first->ReadChunk(expected_band_energies);
>+      // Compute band energy coefficients and check output.
>+      ComputeBandEnergies(fft_coeffs, band_boundary_indexes,
>+                          computed_band_energies);
>+      ExpectEqualFloatArray(expected_band_energies, computed_band_energies);
>+    }
>+  }
>+}
>+
>+TEST(RnnVadTest, ComputeLogBandEnergiesCoefficientsBitExactness) {
>+  constexpr std::array<float, kNumBands> input = {
>+      {86.060539245605f, 275.668334960938f, 43.406528472900f, 6.541896820068f,
>+       17.964015960693f, 8.090919494629f,   1.261920094490f,  1.212702631950f,
>+       1.619154453278f,  0.508935272694f,   0.346316039562f,  0.237035423517f,
>+       0.172424271703f,  0.271657168865f,   0.126088857651f,  0.139967113733f,
>+       0.207200810313f,  0.155893072486f,   0.091090843081f,  0.033391401172f,
>+       0.013879744336f,  0.011973354965f}};
>+  constexpr std::array<float, kNumBands> expected_output = {
>+      {1.934854507446f,  2.440402746201f,  1.637655138969f,  0.816367030144f,
>+       1.254645109177f,  0.908534288406f,  0.104459829628f,  0.087320849299f,
>+       0.211962252855f,  -0.284886807203f, -0.448164641857f, -0.607240796089f,
>+       -0.738917350769f, -0.550279200077f, -0.866177439690f, -0.824003994465f,
>+       -0.663138568401f, -0.780171751976f, -0.995288193226f, -1.362596273422f,
>+       -1.621970295906f, -1.658103585243f}};
>+  std::array<float, kNumBands> computed_output;
>+  {
>+    // TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
>+    // FloatingPointExceptionObserver fpe_observer;
>+    ComputeLogBandEnergiesCoefficients(input, computed_output);
>+    ExpectNearAbsolute(expected_output, computed_output, 1e-5f);
>+  }
>+}
>+
>+TEST(RnnVadTest, ComputeDctBitExactness) {
>+  constexpr std::array<float, kNumBands> input = {
>+      {0.232155621052f,  0.678957760334f, 0.220818966627f,  -0.077363930643f,
>+       -0.559227049351f, 0.432545185089f, 0.353900641203f,  0.398993015289f,
>+       0.409774333239f,  0.454977899790f, 0.300520688295f,  -0.010286616161f,
>+       0.272525429726f,  0.098067551851f, 0.083649002016f,  0.046226885170f,
>+       -0.033228103071f, 0.144773483276f, -0.117661058903f, -0.005628800020f,
>+       -0.009547689930f, -0.045382082462f}};
>+  constexpr std::array<float, kNumBands> expected_output = {
>+      {0.697072803974f,  0.442710995674f,  -0.293156713247f, -0.060711503029f,
>+       0.292050391436f,  0.489301353693f,  0.402255415916f,  0.134404733777f,
>+       -0.086305990815f, -0.199605688453f, -0.234511867166f, -0.413774639368f,
>+       -0.388507157564f, -0.032798115164f, 0.044605545700f,  0.112466648221f,
>+       -0.050096966326f, 0.045971218497f,  -0.029815061018f, -0.410366982222f,
>+       -0.209233760834f, -0.128037497401f}};
>+  auto dct_table = ComputeDctTable();
>+  std::array<float, kNumBands> computed_output;
>+  {
>+    // TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
>+    // FloatingPointExceptionObserver fpe_observer;
>+    ComputeDct(input, dct_table, computed_output);
>+    ExpectNearAbsolute(expected_output, computed_output, 1e-5f);
>+  }
>+}
>+
>+}  // namespace test
>+}  // namespace rnn_vad
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/spectral_features_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/spectral_features_unittest.cc
>new file mode 100644
>index 00000000000..557e41e97cc
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/spectral_features_unittest.cc
>@@ -0,0 +1,126 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/agc2/rnn_vad/spectral_features.h"
>+
>+#include <algorithm>
>+
>+#include "modules/audio_processing/agc2/rnn_vad/test_utils.h"
>+#include "rtc_base/checks.h"
>+// TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
>+// #include "test/fpe_observer.h"
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+namespace rnn_vad {
>+namespace test {
>+namespace {
>+
>+constexpr size_t kTestFeatureVectorSize = kNumBands + 3 * kNumLowerBands + 1;
>+
>+// Writes non-zero sample values.
>+void WriteTestData(rtc::ArrayView<float> samples) {
>+  for (size_t i = 0; i < samples.size(); ++i) {
>+    samples[i] = i % 100;
>+  }
>+}
>+
>+SpectralFeaturesView GetSpectralFeaturesView(
>+    std::array<float, kTestFeatureVectorSize>* feature_vector) {
>+  return {
>+      {feature_vector->data() + kNumLowerBands, kNumBands - kNumLowerBands},
>+      {feature_vector->data(), kNumLowerBands},
>+      {feature_vector->data() + kNumBands, kNumLowerBands},
>+      {feature_vector->data() + kNumBands + kNumLowerBands, kNumLowerBands},
>+      {feature_vector->data() + kNumBands + 2 * kNumLowerBands, kNumLowerBands},
>+      &(*feature_vector)[kNumBands + 3 * kNumLowerBands]};
>+}
>+
>+constexpr float kInitialFeatureVal = -9999.f;
>+
>+}  // namespace
>+
>+TEST(RnnVadTest, SpectralFeaturesWithAndWithoutSilence) {
>+  // Initialize.
>+  SpectralFeaturesExtractor sfe;
>+  std::array<float, kFrameSize20ms24kHz> samples;
>+  rtc::ArrayView<float, kFrameSize20ms24kHz> samples_view(samples);
>+  bool is_silence;
>+  std::array<float, kTestFeatureVectorSize> feature_vector;
>+  auto feature_vector_view = GetSpectralFeaturesView(&feature_vector);
>+
>+  // Write an initial value in the feature vector to detect changes.
>+  std::fill(feature_vector.begin(), feature_vector.end(), kInitialFeatureVal);
>+
>+  // TODO(bugs.webrtc.org/8948): Add when the issue is fixed.
>+  // FloatingPointExceptionObserver fpe_observer;
>+
>+  // With silence.
>+  std::fill(samples.begin(), samples.end(), 0.f);
>+  is_silence = sfe.CheckSilenceComputeFeatures(samples_view, samples_view,
>+                                               feature_vector_view);
>+  // Silence is expected, the output won't be overwritten.
>+  EXPECT_TRUE(is_silence);
>+  EXPECT_TRUE(std::all_of(feature_vector.begin(), feature_vector.end(),
>+                          [](float x) { return x == kInitialFeatureVal; }));
>+
>+  // With no silence.
>+  WriteTestData(samples);
>+  is_silence = sfe.CheckSilenceComputeFeatures(samples_view, samples_view,
>+                                               feature_vector_view);
>+  // Silence is not expected, the output will be overwritten.
>+  EXPECT_FALSE(is_silence);
>+  EXPECT_FALSE(std::all_of(feature_vector.begin(), feature_vector.end(),
>+                           [](float x) { return x == kInitialFeatureVal; }));
>+}
>+
>+// When the input signal does not change, the spectral coefficients average does
>+// not change and the derivatives are zero. Similarly, the spectral variability
>+// score does not change either.
>+TEST(RnnVadTest, SpectralFeaturesConstantAverageZeroDerivative) {
>+  // Initialize.
>+  SpectralFeaturesExtractor sfe;
>+  std::array<float, kFrameSize20ms24kHz> samples;
>+  rtc::ArrayView<float, kFrameSize20ms24kHz> samples_view(samples);
>+  WriteTestData(samples);
>+  bool is_silence;
>+
>+  // Fill the spectral features with test data.
>+  std::array<float, kTestFeatureVectorSize> feature_vector;
>+  auto feature_vector_view = GetSpectralFeaturesView(&feature_vector);
>+  for (size_t i = 0; i < kSpectralCoeffsHistorySize; ++i) {
>+    is_silence = sfe.CheckSilenceComputeFeatures(samples_view, samples_view,
>+                                                 feature_vector_view);
>+  }
>+
>+  // Feed the test data one last time but using a different output vector.
>+  std::array<float, kTestFeatureVectorSize> feature_vector_last;
>+  auto feature_vector_last_view = GetSpectralFeaturesView(&feature_vector_last);
>+  is_silence = sfe.CheckSilenceComputeFeatures(samples_view, samples_view,
>+                                               feature_vector_last_view);
>+
>+  // Average is unchanged.
>+  ExpectEqualFloatArray({feature_vector.data(), kNumLowerBands},
>+                        {feature_vector_last.data(), kNumLowerBands});
>+  // First and second derivatives are zero.
>+  constexpr std::array<float, kNumLowerBands> zeros{};
>+  ExpectEqualFloatArray(
>+      {feature_vector_last.data() + kNumBands, kNumLowerBands}, zeros);
>+  ExpectEqualFloatArray(
>+      {feature_vector_last.data() + kNumBands + kNumLowerBands, kNumLowerBands},
>+      zeros);
>+  // Spectral variability is unchanged.
>+  EXPECT_FLOAT_EQ(feature_vector[kNumBands + 3 * kNumLowerBands],
>+                  feature_vector_last[kNumBands + 3 * kNumLowerBands]);
>+}
>+
>+}  // namespace test
>+}  // namespace rnn_vad
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/symmetric_matrix_buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/symmetric_matrix_buffer.h
>new file mode 100644
>index 00000000000..f0282aaed5e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/symmetric_matrix_buffer.h
>@@ -0,0 +1,94 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_SYMMETRIC_MATRIX_BUFFER_H_
>+#define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_SYMMETRIC_MATRIX_BUFFER_H_
>+
>+#include <algorithm>
>+#include <array>
>+#include <cstring>
>+#include <utility>
>+
>+#include "api/array_view.h"
>+#include "rtc_base/checks.h"
>+
>+namespace webrtc {
>+namespace rnn_vad {
>+
>+// Data structure to buffer the results of pair-wise comparisons between items
>+// stored in a ring buffer. Every time that the oldest item is replaced in the
>+// ring buffer, the new one is compared to the remaining items in the ring
>+// buffer. The results of such comparisons need to be buffered and automatically
>+// removed when one of the two corresponding items that have been compared is
>+// removed from the ring buffer. It is assumed that the comparison is symmetric
>+// and that comparing an item with itself is not needed.
>+template <typename T, size_t S>
>+class SymmetricMatrixBuffer {
>+  static_assert(S > 2, "");
>+
>+ public:
>+  SymmetricMatrixBuffer() = default;
>+  SymmetricMatrixBuffer(const SymmetricMatrixBuffer&) = delete;
>+  SymmetricMatrixBuffer& operator=(const SymmetricMatrixBuffer&) = delete;
>+  ~SymmetricMatrixBuffer() = default;
>+  // Sets the buffer values to zero.
>+  void Reset() {
>+    static_assert(std::is_arithmetic<T>::value,
>+                  "Integral or floating point required.");
>+    buf_.fill(0);
>+  }
>+  // Pushes the results from the comparison between the most recent item and
>+  // those that are still in the ring buffer. The first element in |values| must
>+  // correspond to the comparison between the most recent item and the second
>+  // most recent one in the ring buffer, whereas the last element in |values|
>+  // must correspond to the comparison between the most recent item and the
>+  // oldest one in the ring buffer.
>+  void Push(rtc::ArrayView<T, S - 1> values) {
>+    // Move the lower-right sub-matrix of size (S-2) x (S-2) one row up and one
>+    // column left.
>+    std::memmove(buf_.data(), buf_.data() + S, (buf_.size() - S) * sizeof(T));
>+    // Copy new values in the last column in the right order.
>+    for (size_t i = 0; i < values.size(); ++i) {
>+      const size_t index = (S - 1 - i) * (S - 1) - 1;
>+      RTC_DCHECK_LE(static_cast<size_t>(0), index);
>+      RTC_DCHECK_LT(index, buf_.size());
>+      buf_[index] = values[i];
>+    }
>+  }
>+  // Reads the value that corresponds to comparison of two items in the ring
>+  // buffer having delay |delay1| and |delay2|. The two arguments must not be
>+  // equal and both must be in {0, ..., S - 1}.
>+  T GetValue(size_t delay1, size_t delay2) const {
>+    int row = S - 1 - static_cast<int>(delay1);
>+    int col = S - 1 - static_cast<int>(delay2);
>+    RTC_DCHECK_NE(row, col) << "The diagonal cannot be accessed.";
>+    if (row > col)
>+      std::swap(row, col);  // Swap to access the upper-right triangular part.
>+    RTC_DCHECK_LE(0, row);
>+    RTC_DCHECK_LT(row, S - 1) << "Not enforcing row < col and row != col.";
>+    RTC_DCHECK_LE(1, col) << "Not enforcing row < col and row != col.";
>+    RTC_DCHECK_LT(col, S);
>+    const int index = row * (S - 1) + (col - 1);
>+    RTC_DCHECK_LE(0, index);
>+    RTC_DCHECK_LT(index, buf_.size());
>+    return buf_[index];
>+  }
>+
>+ private:
>+  // Encode an upper-right triangular matrix (excluding its diagonal) using a
>+  // square matrix. This allows to move the data in Push() with one single
>+  // operation.
>+  std::array<T, (S - 1) * (S - 1)> buf_{};
>+};
>+
>+}  // namespace rnn_vad
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_SYMMETRIC_MATRIX_BUFFER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/symmetric_matrix_buffer_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/symmetric_matrix_buffer_unittest.cc
>new file mode 100644
>index 00000000000..a1b80076964
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/symmetric_matrix_buffer_unittest.cc
>@@ -0,0 +1,110 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/agc2/rnn_vad/symmetric_matrix_buffer.h"
>+
>+#include "modules/audio_processing/agc2/rnn_vad/ring_buffer.h"
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+namespace rnn_vad {
>+namespace test {
>+namespace {
>+
>+template <typename T, size_t S>
>+void CheckSymmetry(const SymmetricMatrixBuffer<T, S>* sym_matrix_buf) {
>+  for (size_t row = 0; row < S - 1; ++row)
>+    for (size_t col = row + 1; col < S; ++col)
>+      EXPECT_EQ(sym_matrix_buf->GetValue(row, col),
>+                sym_matrix_buf->GetValue(col, row));
>+}
>+
>+using PairType = std::pair<int, int>;
>+
>+// Checks that the symmetric matrix buffer contains any pair with a value equal
>+// to the given one.
>+template <size_t S>
>+bool CheckPairsWithValueExist(
>+    const SymmetricMatrixBuffer<PairType, S>* sym_matrix_buf,
>+    const int value) {
>+  for (size_t row = 0; row < S - 1; ++row) {
>+    for (size_t col = row + 1; col < S; ++col) {
>+      auto p = sym_matrix_buf->GetValue(row, col);
>+      if (p.first == value || p.second == value)
>+        return true;
>+    }
>+  }
>+  return false;
>+}
>+
>+}  // namespace
>+
>+// Test that shows how to combine RingBuffer and SymmetricMatrixBuffer to
>+// efficiently compute pair-wise scores. This test verifies that the evolution
>+// of a SymmetricMatrixBuffer instance follows that of RingBuffer.
>+TEST(RnnVadTest, SymmetricMatrixBufferUseCase) {
>+  // Instance a ring buffer which will be fed with a series of integer values.
>+  constexpr int kRingBufSize = 10;
>+  RingBuffer<int, 1, static_cast<size_t>(kRingBufSize)> ring_buf;
>+  // Instance a symmetric matrix buffer for the ring buffer above. It stores
>+  // pairs of integers with which this test can easily check that the evolution
>+  // of RingBuffer and SymmetricMatrixBuffer match.
>+  SymmetricMatrixBuffer<PairType, kRingBufSize> sym_matrix_buf;
>+  for (int t = 1; t <= 100; ++t) {  // Evolution steps.
>+    SCOPED_TRACE(t);
>+    const int t_removed = ring_buf.GetArrayView(kRingBufSize - 1)[0];
>+    ring_buf.Push({&t, 1});
>+    // The head of the ring buffer is |t|.
>+    ASSERT_EQ(t, ring_buf.GetArrayView(0)[0]);
>+    // Create the comparisons between |t| and the older elements in the ring
>+    // buffer.
>+    std::array<PairType, kRingBufSize - 1> new_comparions;
>+    for (int i = 0; i < kRingBufSize - 1; ++i) {
>+      // Start comparing |t| to the second newest element in the ring buffer.
>+      const int delay = i + 1;
>+      const auto t_prev = ring_buf.GetArrayView(delay)[0];
>+      ASSERT_EQ(std::max(0, t - delay), t_prev);
>+      // Compare the last element |t| with |t_prev|.
>+      new_comparions[i].first = t_prev;
>+      new_comparions[i].second = t;
>+    }
>+    // Push the new comparisons in the symmetric matrix buffer.
>+    sym_matrix_buf.Push({new_comparions.data(), new_comparions.size()});
>+    // Tests.
>+    CheckSymmetry(&sym_matrix_buf);
>+    // Check that the pairs resulting from the content in the ring buffer are
>+    // in the right position.
>+    for (size_t delay1 = 0; delay1 < kRingBufSize - 1; ++delay1) {
>+      for (size_t delay2 = delay1 + 1; delay2 < kRingBufSize; ++delay2) {
>+        const auto t1 = ring_buf.GetArrayView(delay1)[0];
>+        const auto t2 = ring_buf.GetArrayView(delay2)[0];
>+        ASSERT_LE(t2, t1);
>+        const auto p = sym_matrix_buf.GetValue(delay1, delay2);
>+        EXPECT_EQ(p.first, t2);
>+        EXPECT_EQ(p.second, t1);
>+      }
>+    }
>+    // Check that every older element in the ring buffer still has a
>+    // corresponding pair in the symmetric matrix buffer.
>+    for (size_t delay = 1; delay < kRingBufSize; ++delay) {
>+      const auto t_prev = ring_buf.GetArrayView(delay)[0];
>+      EXPECT_TRUE(CheckPairsWithValueExist(&sym_matrix_buf, t_prev));
>+    }
>+    // Check that the element removed from the ring buffer has no corresponding
>+    // pairs in the symmetric matrix buffer.
>+    if (t > kRingBufSize - 1) {
>+      EXPECT_FALSE(CheckPairsWithValueExist(&sym_matrix_buf, t_removed));
>+    }
>+  }
>+}
>+
>+}  // namespace test
>+}  // namespace rnn_vad
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/test_utils.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/test_utils.cc
>new file mode 100644
>index 00000000000..db38cab74c0
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/test_utils.cc
>@@ -0,0 +1,116 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/agc2/rnn_vad/test_utils.h"
>+
>+#include "absl/memory/memory.h"
>+#include "rtc_base/checks.h"
>+#include "test/gtest.h"
>+#include "test/testsupport/fileutils.h"
>+
>+namespace webrtc {
>+namespace rnn_vad {
>+namespace test {
>+namespace {
>+
>+using ReaderPairType =
>+    std::pair<std::unique_ptr<BinaryFileReader<float>>, const size_t>;
>+
>+}  // namespace
>+
>+using webrtc::test::ResourcePath;
>+
>+void ExpectEqualFloatArray(rtc::ArrayView<const float> expected,
>+                           rtc::ArrayView<const float> computed) {
>+  ASSERT_EQ(expected.size(), computed.size());
>+  for (size_t i = 0; i < expected.size(); ++i) {
>+    SCOPED_TRACE(i);
>+    EXPECT_FLOAT_EQ(expected[i], computed[i]);
>+  }
>+}
>+
>+void ExpectNearAbsolute(rtc::ArrayView<const float> expected,
>+                        rtc::ArrayView<const float> computed,
>+                        float tolerance) {
>+  ASSERT_EQ(expected.size(), computed.size());
>+  for (size_t i = 0; i < expected.size(); ++i) {
>+    SCOPED_TRACE(i);
>+    EXPECT_NEAR(expected[i], computed[i], tolerance);
>+  }
>+}
>+
>+std::unique_ptr<BinaryFileReader<float>> CreatePitchSearchTestDataReader() {
>+  constexpr size_t cols = 1396;
>+  return absl::make_unique<BinaryFileReader<float>>(
>+      ResourcePath("audio_processing/agc2/rnn_vad/pitch_search_int", "dat"),
>+      cols);
>+}
>+
>+std::pair<std::unique_ptr<BinaryFileReader<int16_t, float>>, const size_t>
>+CreatePcmSamplesReader(const size_t frame_length) {
>+  auto ptr = absl::make_unique<BinaryFileReader<int16_t, float>>(
>+      test::ResourcePath("audio_processing/agc2/rnn_vad/samples", "pcm"),
>+      frame_length);
>+  // The last incomplete frame is ignored.
>+  return {std::move(ptr), ptr->data_length() / frame_length};
>+}
>+
>+ReaderPairType CreatePitchBuffer24kHzReader() {
>+  constexpr size_t cols = 864;
>+  auto ptr = absl::make_unique<BinaryFileReader<float>>(
>+      ResourcePath("audio_processing/agc2/rnn_vad/pitch_buf_24k", "dat"), cols);
>+  return {std::move(ptr), rtc::CheckedDivExact(ptr->data_length(), cols)};
>+}
>+
>+ReaderPairType CreateLpResidualAndPitchPeriodGainReader() {
>+  constexpr size_t num_lp_residual_coeffs = 864;
>+  auto ptr = absl::make_unique<BinaryFileReader<float>>(
>+      ResourcePath("audio_processing/agc2/rnn_vad/pitch_lp_res", "dat"),
>+      num_lp_residual_coeffs);
>+  return {std::move(ptr),
>+          rtc::CheckedDivExact(ptr->data_length(), 2 + num_lp_residual_coeffs)};
>+}
>+
>+ReaderPairType CreateFftCoeffsReader() {
>+  constexpr size_t num_fft_points = 481;
>+  constexpr size_t row_size = 2 * num_fft_points;  // Real and imaginary values.
>+  auto ptr = absl::make_unique<BinaryFileReader<float>>(
>+      test::ResourcePath("audio_processing/agc2/rnn_vad/fft", "dat"),
>+      num_fft_points);
>+  return {std::move(ptr), rtc::CheckedDivExact(ptr->data_length(), row_size)};
>+}
>+
>+ReaderPairType CreateBandEnergyCoeffsReader() {
>+  constexpr size_t num_bands = 22;
>+  auto ptr = absl::make_unique<BinaryFileReader<float>>(
>+      test::ResourcePath("audio_processing/agc2/rnn_vad/band_energies", "dat"),
>+      num_bands);
>+  return {std::move(ptr), rtc::CheckedDivExact(ptr->data_length(), num_bands)};
>+}
>+
>+ReaderPairType CreateSilenceFlagsFeatureMatrixReader() {
>+  constexpr size_t feature_vector_size = 42;
>+  auto ptr = absl::make_unique<BinaryFileReader<float>>(
>+      test::ResourcePath("audio_processing/agc2/rnn_vad/sil_features", "dat"),
>+      feature_vector_size);
>+  // Features and silence flag.
>+  return {std::move(ptr),
>+          rtc::CheckedDivExact(ptr->data_length(), feature_vector_size + 1)};
>+}
>+
>+ReaderPairType CreateVadProbsReader() {
>+  auto ptr = absl::make_unique<BinaryFileReader<float>>(
>+      test::ResourcePath("audio_processing/agc2/rnn_vad/vad_prob", "dat"));
>+  return {std::move(ptr), ptr->data_length()};
>+}
>+
>+}  // namespace test
>+}  // namespace rnn_vad
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/test_utils.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/test_utils.h
>new file mode 100644
>index 00000000000..15be85ac000
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/rnn_vad/test_utils.h
>@@ -0,0 +1,125 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_TEST_UTILS_H_
>+#define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_TEST_UTILS_H_
>+
>+#include <algorithm>
>+#include <fstream>
>+#include <limits>
>+#include <memory>
>+#include <string>
>+#include <utility>
>+#include <vector>
>+
>+#include "api/array_view.h"
>+#include "rtc_base/checks.h"
>+
>+namespace webrtc {
>+namespace rnn_vad {
>+namespace test {
>+
>+constexpr float kFloatMin = std::numeric_limits<float>::min();
>+
>+// Fails for every pair from two equally sized rtc::ArrayView<float> views such
>+// that the values in the pair do not match.
>+void ExpectEqualFloatArray(rtc::ArrayView<const float> expected,
>+                           rtc::ArrayView<const float> computed);
>+
>+// Fails for every pair from two equally sized rtc::ArrayView<float> views such
>+// that their absolute error is above a given threshold.
>+void ExpectNearAbsolute(rtc::ArrayView<const float> expected,
>+                        rtc::ArrayView<const float> computed,
>+                        float tolerance);
>+
>+// Reader for binary files consisting of an arbitrary long sequence of elements
>+// having type T. It is possible to read and cast to another type D at once.
>+template <typename T, typename D = T>
>+class BinaryFileReader {
>+ public:
>+  explicit BinaryFileReader(const std::string& file_path, size_t chunk_size = 1)
>+      : is_(file_path, std::ios::binary | std::ios::ate),
>+        data_length_(is_.tellg() / sizeof(T)),
>+        chunk_size_(chunk_size) {
>+    RTC_CHECK_LT(0, chunk_size_);
>+    RTC_CHECK(is_);
>+    SeekBeginning();
>+    buf_.resize(chunk_size_);
>+  }
>+  BinaryFileReader(const BinaryFileReader&) = delete;
>+  BinaryFileReader& operator=(const BinaryFileReader&) = delete;
>+  ~BinaryFileReader() = default;
>+  size_t data_length() const { return data_length_; }
>+  bool ReadValue(D* dst) {
>+    if (std::is_same<T, D>::value) {
>+      is_.read(reinterpret_cast<char*>(dst), sizeof(T));
>+    } else {
>+      T v;
>+      is_.read(reinterpret_cast<char*>(&v), sizeof(T));
>+      *dst = static_cast<D>(v);
>+    }
>+    return is_.gcount() == sizeof(T);
>+  }
>+  bool ReadChunk(rtc::ArrayView<D> dst) {
>+    RTC_DCHECK_EQ(chunk_size_, dst.size());
>+    const std::streamsize bytes_to_read = chunk_size_ * sizeof(T);
>+    if (std::is_same<T, D>::value) {
>+      is_.read(reinterpret_cast<char*>(dst.data()), bytes_to_read);
>+    } else {
>+      is_.read(reinterpret_cast<char*>(buf_.data()), bytes_to_read);
>+      std::transform(buf_.begin(), buf_.end(), dst.begin(),
>+                     [](const T& v) -> D { return static_cast<D>(v); });
>+    }
>+    return is_.gcount() == bytes_to_read;
>+  }
>+  void SeekForward(size_t items) { is_.seekg(items * sizeof(T), is_.cur); }
>+  void SeekBeginning() { is_.seekg(0, is_.beg); }
>+
>+ private:
>+  std::ifstream is_;
>+  const size_t data_length_;
>+  const size_t chunk_size_;
>+  std::vector<T> buf_;
>+};
>+
>+// Factories for resource file readers.
>+// Creates a reader for the pitch search test data.
>+std::unique_ptr<BinaryFileReader<float>> CreatePitchSearchTestDataReader();
>+// The functions below return a pair where the first item is a reader unique
>+// pointer and the second the number of chunks that can be read from the file.
>+// Creates a reader for the PCM samples that casts from S16 to float and reads
>+// chunks with length |frame_length|.
>+std::pair<std::unique_ptr<BinaryFileReader<int16_t, float>>, const size_t>
>+CreatePcmSamplesReader(const size_t frame_length);
>+// Creates a reader for the pitch buffer content at 24 kHz.
>+std::pair<std::unique_ptr<BinaryFileReader<float>>, const size_t>
>+CreatePitchBuffer24kHzReader();
>+// Creates a reader for the the LP residual coefficients and the pitch period
>+// and gain values.
>+std::pair<std::unique_ptr<BinaryFileReader<float>>, const size_t>
>+CreateLpResidualAndPitchPeriodGainReader();
>+// Creates a reader for the FFT coefficients.
>+std::pair<std::unique_ptr<BinaryFileReader<float>>, const size_t>
>+CreateFftCoeffsReader();
>+// Instance a reader for the band energy coefficients.
>+std::pair<std::unique_ptr<BinaryFileReader<float>>, const size_t>
>+CreateBandEnergyCoeffsReader();
>+// Creates a reader for the silence flags and the feature matrix.
>+std::pair<std::unique_ptr<BinaryFileReader<float>>, const size_t>
>+CreateSilenceFlagsFeatureMatrixReader();
>+// Creates a reader for the VAD probabilities.
>+std::pair<std::unique_ptr<BinaryFileReader<float>>, const size_t>
>+CreateVadProbsReader();
>+
>+}  // namespace test
>+}  // namespace rnn_vad
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_TEST_UTILS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/saturation_protector.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/saturation_protector.cc
>new file mode 100644
>index 00000000000..dc9be474042
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/saturation_protector.cc
>@@ -0,0 +1,94 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/agc2/saturation_protector.h"
>+
>+#include <algorithm>
>+
>+#include "modules/audio_processing/logging/apm_data_dumper.h"
>+#include "rtc_base/numerics/safe_minmax.h"
>+
>+namespace webrtc {
>+
>+namespace {
>+void ShiftBuffer(std::array<float, kPeakEnveloperBufferSize>* buffer_) {
>+  // Move everything one element back.
>+  std::copy(buffer_->begin() + 1, buffer_->end(), buffer_->begin());
>+}
>+}  // namespace
>+
>+SaturationProtector::PeakEnveloper::PeakEnveloper() = default;
>+
>+void SaturationProtector::PeakEnveloper::Process(float frame_peak_dbfs) {
>+  // Update the delayed buffer and the current superframe peak.
>+  current_superframe_peak_dbfs_ =
>+      std::max(current_superframe_peak_dbfs_, frame_peak_dbfs);
>+  speech_time_in_estimate_ms_ += kFrameDurationMs;
>+  if (speech_time_in_estimate_ms_ > kPeakEnveloperSuperFrameLengthMs) {
>+    speech_time_in_estimate_ms_ = 0;
>+    const bool buffer_full = elements_in_buffer_ == kPeakEnveloperBufferSize;
>+    if (buffer_full) {
>+      ShiftBuffer(&peak_delay_buffer_);
>+      *peak_delay_buffer_.rbegin() = current_superframe_peak_dbfs_;
>+    } else {
>+      peak_delay_buffer_[elements_in_buffer_] = current_superframe_peak_dbfs_;
>+      elements_in_buffer_++;
>+    }
>+    current_superframe_peak_dbfs_ = -90.f;
>+  }
>+}
>+
>+float SaturationProtector::PeakEnveloper::Query() const {
>+  float result;
>+  if (elements_in_buffer_ > 0) {
>+    result = peak_delay_buffer_[0];
>+  } else {
>+    result = current_superframe_peak_dbfs_;
>+  }
>+  return result;
>+}
>+
>+SaturationProtector::SaturationProtector(ApmDataDumper* apm_data_dumper)
>+    : apm_data_dumper_(apm_data_dumper) {}
>+
>+void SaturationProtector::UpdateMargin(
>+    const VadWithLevel::LevelAndProbability& vad_data,
>+    float last_speech_level_estimate) {
>+  peak_enveloper_.Process(vad_data.speech_peak_dbfs);
>+  const float delayed_peak_dbfs = peak_enveloper_.Query();
>+  const float difference_db = delayed_peak_dbfs - last_speech_level_estimate;
>+
>+  if (last_margin_ < difference_db) {
>+    last_margin_ = last_margin_ * kSaturationProtectorAttackConstant +
>+                   difference_db * (1.f - kSaturationProtectorAttackConstant);
>+  } else {
>+    last_margin_ = last_margin_ * kSaturationProtectorDecayConstant +
>+                   difference_db * (1.f - kSaturationProtectorDecayConstant);
>+  }
>+
>+  last_margin_ = rtc::SafeClamp<float>(last_margin_, 12.f, 25.f);
>+}
>+
>+float SaturationProtector::LastMargin() const {
>+  return last_margin_;
>+}
>+
>+void SaturationProtector::Reset() {
>+  peak_enveloper_ = PeakEnveloper();
>+}
>+
>+void SaturationProtector::DebugDumpEstimate() const {
>+  apm_data_dumper_->DumpRaw(
>+      "agc2_adaptive_saturation_protector_delayed_peak_dbfs",
>+      peak_enveloper_.Query());
>+  apm_data_dumper_->DumpRaw("agc2_adaptive_saturation_margin_db", last_margin_);
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/saturation_protector.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/saturation_protector.h
>new file mode 100644
>index 00000000000..3f207da8301
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/saturation_protector.h
>@@ -0,0 +1,67 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_AGC2_SATURATION_PROTECTOR_H_
>+#define MODULES_AUDIO_PROCESSING_AGC2_SATURATION_PROTECTOR_H_
>+
>+#include <array>
>+
>+#include "modules/audio_processing/agc2/agc2_common.h"
>+#include "modules/audio_processing/agc2/vad_with_level.h"
>+
>+namespace webrtc {
>+
>+class ApmDataDumper;
>+
>+class SaturationProtector {
>+ public:
>+  explicit SaturationProtector(ApmDataDumper* apm_data_dumper);
>+
>+  // Update and return margin estimate. This method should be called
>+  // whenever a frame is reliably classified as 'speech'.
>+  //
>+  // Returned value is in DB scale.
>+  void UpdateMargin(const VadWithLevel::LevelAndProbability& vad_data,
>+                    float last_speech_level_estimate_dbfs);
>+
>+  // Returns latest computed margin. Used in cases when speech is not
>+  // detected.
>+  float LastMargin() const;
>+
>+  // Resets the internal memory.
>+  void Reset();
>+
>+  void DebugDumpEstimate() const;
>+
>+ private:
>+  // Computes a delayed envelope of peaks.
>+  class PeakEnveloper {
>+   public:
>+    PeakEnveloper();
>+    void Process(float frame_peak_dbfs);
>+
>+    float Query() const;
>+
>+   private:
>+    size_t speech_time_in_estimate_ms_ = 0;
>+    float current_superframe_peak_dbfs_ = -90.f;
>+    size_t elements_in_buffer_ = 0;
>+    std::array<float, kPeakEnveloperBufferSize> peak_delay_buffer_ = {};
>+  };
>+
>+  ApmDataDumper* apm_data_dumper_;
>+
>+  float last_margin_ = kInitialSaturationMarginDb;
>+  PeakEnveloper peak_enveloper_;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_AGC2_SATURATION_PROTECTOR_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/saturation_protector_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/saturation_protector_unittest.cc
>new file mode 100644
>index 00000000000..6013e13827d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/saturation_protector_unittest.cc
>@@ -0,0 +1,144 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/agc2/saturation_protector.h"
>+
>+#include <algorithm>
>+
>+#include "modules/audio_processing/agc2/agc2_common.h"
>+#include "modules/audio_processing/logging/apm_data_dumper.h"
>+#include "rtc_base/gunit.h"
>+
>+namespace webrtc {
>+namespace {
>+float RunOnConstantLevel(int num_iterations,
>+                         VadWithLevel::LevelAndProbability vad_data,
>+                         float estimated_level_dbfs,
>+                         SaturationProtector* saturation_protector) {
>+  float last_margin = saturation_protector->LastMargin();
>+  float max_difference = 0.f;
>+  for (int i = 0; i < num_iterations; ++i) {
>+    saturation_protector->UpdateMargin(vad_data, estimated_level_dbfs);
>+    const float new_margin = saturation_protector->LastMargin();
>+    max_difference =
>+        std::max(max_difference, std::abs(new_margin - last_margin));
>+    last_margin = new_margin;
>+    saturation_protector->DebugDumpEstimate();
>+  }
>+  return max_difference;
>+}
>+}  // namespace
>+
>+TEST(AutomaticGainController2SaturationProtector, ProtectorShouldNotCrash) {
>+  ApmDataDumper apm_data_dumper(0);
>+  SaturationProtector saturation_protector(&apm_data_dumper);
>+  VadWithLevel::LevelAndProbability vad_data(1.f, -20.f, -10.f);
>+
>+  saturation_protector.UpdateMargin(vad_data, -20.f);
>+  static_cast<void>(saturation_protector.LastMargin());
>+  saturation_protector.DebugDumpEstimate();
>+}
>+
>+// Check that the estimate converges to the ratio between peaks and
>+// level estimator values after a while.
>+TEST(AutomaticGainController2SaturationProtector,
>+     ProtectorEstimatesCrestRatio) {
>+  ApmDataDumper apm_data_dumper(0);
>+  SaturationProtector saturation_protector(&apm_data_dumper);
>+
>+  constexpr float kPeakLevel = -20.f;
>+  constexpr float kCrestFactor = kInitialSaturationMarginDb + 1.f;
>+  constexpr float kSpeechLevel = kPeakLevel - kCrestFactor;
>+  const float kMaxDifference =
>+      0.5 * std::abs(kInitialSaturationMarginDb - kCrestFactor);
>+
>+  static_cast<void>(RunOnConstantLevel(
>+      2000, VadWithLevel::LevelAndProbability(1.f, -90.f, kPeakLevel),
>+      kSpeechLevel, &saturation_protector));
>+
>+  EXPECT_NEAR(saturation_protector.LastMargin(), kCrestFactor, kMaxDifference);
>+}
>+
>+TEST(AutomaticGainController2SaturationProtector, ProtectorChangesSlowly) {
>+  ApmDataDumper apm_data_dumper(0);
>+  SaturationProtector saturation_protector(&apm_data_dumper);
>+
>+  constexpr float kPeakLevel = -20.f;
>+  constexpr float kCrestFactor = kInitialSaturationMarginDb - 5.f;
>+  constexpr float kOtherCrestFactor = kInitialSaturationMarginDb;
>+  constexpr float kSpeechLevel = kPeakLevel - kCrestFactor;
>+  constexpr float kOtherSpeechLevel = kPeakLevel - kOtherCrestFactor;
>+
>+  constexpr int kNumIterations = 1000;
>+  float max_difference = RunOnConstantLevel(
>+      kNumIterations, VadWithLevel::LevelAndProbability(1.f, -90.f, kPeakLevel),
>+      kSpeechLevel, &saturation_protector);
>+
>+  max_difference =
>+      std::max(RunOnConstantLevel(
>+                   kNumIterations,
>+                   VadWithLevel::LevelAndProbability(1.f, -90.f, kPeakLevel),
>+                   kOtherSpeechLevel, &saturation_protector),
>+               max_difference);
>+
>+  constexpr float kMaxChangeSpeedDbPerSecond = 0.5;  // 1 db / 2 seconds.
>+
>+  EXPECT_LE(max_difference,
>+            kMaxChangeSpeedDbPerSecond / 1000 * kFrameDurationMs);
>+}
>+
>+TEST(AutomaticGainController2SaturationProtector,
>+     ProtectorAdaptsToDelayedChanges) {
>+  ApmDataDumper apm_data_dumper(0);
>+  SaturationProtector saturation_protector(&apm_data_dumper);
>+
>+  constexpr int kDelayIterations = kFullBufferSizeMs / kFrameDurationMs;
>+  constexpr float kInitialSpeechLevelDbfs = -30;
>+  constexpr float kLaterSpeechLevelDbfs = -15;
>+
>+  // First run on initial level.
>+  float max_difference = RunOnConstantLevel(
>+      kDelayIterations,
>+      VadWithLevel::LevelAndProbability(
>+          1.f, -90.f, kInitialSpeechLevelDbfs + kInitialSaturationMarginDb),
>+      kInitialSpeechLevelDbfs, &saturation_protector);
>+
>+  // Then peak changes, but not RMS.
>+  max_difference = std::max(
>+      RunOnConstantLevel(
>+          kDelayIterations,
>+          VadWithLevel::LevelAndProbability(
>+              1.f, -90.f, kLaterSpeechLevelDbfs + kInitialSaturationMarginDb),
>+          kInitialSpeechLevelDbfs, &saturation_protector),
>+      max_difference);
>+
>+  // Then both change.
>+  max_difference = std::max(
>+      RunOnConstantLevel(
>+          kDelayIterations,
>+          VadWithLevel::LevelAndProbability(
>+              1.f, -90.f, kLaterSpeechLevelDbfs + kInitialSaturationMarginDb),
>+          kLaterSpeechLevelDbfs, &saturation_protector),
>+      max_difference);
>+
>+  // The saturation protector expects that the RMS changes roughly
>+  // 'kFullBufferSizeMs' after peaks change. This is to account for
>+  // delay introduces by the level estimator. Therefore, the input
>+  // above is 'normal' and 'expected', and shouldn't influence the
>+  // margin by much.
>+
>+  const float total_difference =
>+      std::abs(saturation_protector.LastMargin() - kInitialSaturationMarginDb);
>+
>+  EXPECT_LE(total_difference, 0.05f);
>+  EXPECT_LE(max_difference, 0.01f);
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/signal_classifier.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/signal_classifier.cc
>similarity index 81%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/signal_classifier.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/signal_classifier.cc
>index d2d5917387a..0ec34148b93 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/signal_classifier.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/signal_classifier.cc
>@@ -8,16 +8,15 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "modules/audio_processing/level_controller/signal_classifier.h"
>+#include "modules/audio_processing/agc2/signal_classifier.h"
> 
> #include <algorithm>
> #include <numeric>
> #include <vector>
> 
> #include "api/array_view.h"
>-#include "modules/audio_processing/audio_buffer.h"
>-#include "modules/audio_processing/level_controller/down_sampler.h"
>-#include "modules/audio_processing/level_controller/noise_spectrum_estimator.h"
>+#include "modules/audio_processing/agc2/down_sampler.h"
>+#include "modules/audio_processing/agc2/noise_spectrum_estimator.h"
> #include "modules/audio_processing/logging/apm_data_dumper.h"
> #include "rtc_base/constructormagic.h"
> 
>@@ -84,8 +83,6 @@ webrtc::SignalClassifier::SignalType ClassifySignal(
>   // stationarity.
>   if (num_stationary_bands > 15) {
>     return SignalClassifier::SignalType::kStationary;
>-  } else if (num_highly_nonstationary_bands > 15) {
>-    return SignalClassifier::SignalType::kHighlyNonStationary;
>   } else {
>     return SignalClassifier::SignalType::kNonStationary;
>   }
>@@ -113,7 +110,7 @@ SignalClassifier::SignalClassifier(ApmDataDumper* data_dumper)
>     : data_dumper_(data_dumper),
>       down_sampler_(data_dumper_),
>       noise_spectrum_estimator_(data_dumper_) {
>-  Initialize(AudioProcessing::kSampleRate48kHz);
>+  Initialize(48000);
> }
> SignalClassifier::~SignalClassifier() {}
> 
>@@ -127,15 +124,13 @@ void SignalClassifier::Initialize(int sample_rate_hz) {
>   last_signal_type_ = SignalClassifier::SignalType::kNonStationary;
> }
> 
>-void SignalClassifier::Analyze(const AudioBuffer& audio,
>-                               SignalType* signal_type) {
>-  RTC_DCHECK_EQ(audio.num_frames(), sample_rate_hz_ / 100);
>+SignalClassifier::SignalType SignalClassifier::Analyze(
>+    rtc::ArrayView<const float> signal) {
>+  RTC_DCHECK_EQ(signal.size(), sample_rate_hz_ / 100);
> 
>   // Compute the signal power spectrum.
>   float downsampled_frame[80];
>-  down_sampler_.DownSample(rtc::ArrayView<const float>(
>-                               audio.channels_const_f()[0], audio.num_frames()),
>-                           downsampled_frame);
>+  down_sampler_.DownSample(signal, downsampled_frame);
>   float extended_frame[128];
>   frame_extender_->ExtendFrame(downsampled_frame, extended_frame);
>   RemoveDcLevel(extended_frame);
>@@ -144,9 +139,9 @@ void SignalClassifier::Analyze(const AudioBuffer& audio,
> 
>   // Classify the signal based on the estimate of the noise spectrum and the
>   // signal spectrum estimate.
>-  *signal_type = ClassifySignal(signal_spectrum,
>-                                noise_spectrum_estimator_.GetNoiseSpectrum(),
>-                                data_dumper_);
>+  const SignalType signal_type = ClassifySignal(
>+      signal_spectrum, noise_spectrum_estimator_.GetNoiseSpectrum(),
>+      data_dumper_);
> 
>   // Update the noise spectrum based on the signal spectrum.
>   noise_spectrum_estimator_.Update(signal_spectrum,
>@@ -155,17 +150,18 @@ void SignalClassifier::Analyze(const AudioBuffer& audio,
>   // Update the number of frames until a reliable signal spectrum is achieved.
>   initialization_frames_left_ = std::max(0, initialization_frames_left_ - 1);
> 
>-  if (last_signal_type_ == *signal_type) {
>+  if (last_signal_type_ == signal_type) {
>     consistent_classification_counter_ =
>         std::max(0, consistent_classification_counter_ - 1);
>   } else {
>-    last_signal_type_ = *signal_type;
>+    last_signal_type_ = signal_type;
>     consistent_classification_counter_ = 3;
>   }
> 
>   if (consistent_classification_counter_ > 0) {
>-    *signal_type = SignalClassifier::SignalType::kNonStationary;
>+    return SignalClassifier::SignalType::kNonStationary;
>   }
>+  return signal_type;
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/signal_classifier.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/signal_classifier.h
>similarity index 75%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/signal_classifier.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/signal_classifier.h
>index 2be13fef7ad..23fe3159774 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/signal_classifier.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/signal_classifier.h
>@@ -8,15 +8,15 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#ifndef MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_SIGNAL_CLASSIFIER_H_
>-#define MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_SIGNAL_CLASSIFIER_H_
>+#ifndef MODULES_AUDIO_PROCESSING_AGC2_SIGNAL_CLASSIFIER_H_
>+#define MODULES_AUDIO_PROCESSING_AGC2_SIGNAL_CLASSIFIER_H_
> 
> #include <memory>
> #include <vector>
> 
> #include "api/array_view.h"
>-#include "modules/audio_processing/level_controller/down_sampler.h"
>-#include "modules/audio_processing/level_controller/noise_spectrum_estimator.h"
>+#include "modules/audio_processing/agc2/down_sampler.h"
>+#include "modules/audio_processing/agc2/noise_spectrum_estimator.h"
> #include "modules/audio_processing/utility/ooura_fft.h"
> #include "rtc_base/constructormagic.h"
> 
>@@ -27,13 +27,13 @@ class AudioBuffer;
> 
> class SignalClassifier {
>  public:
>-  enum class SignalType { kHighlyNonStationary, kNonStationary, kStationary };
>+  enum class SignalType { kNonStationary, kStationary };
> 
>   explicit SignalClassifier(ApmDataDumper* data_dumper);
>   ~SignalClassifier();
> 
>   void Initialize(int sample_rate_hz);
>-  void Analyze(const AudioBuffer& audio, SignalType* signal_type);
>+  SignalType Analyze(rtc::ArrayView<const float> signal);
> 
>  private:
>   class FrameExtender {
>@@ -64,4 +64,4 @@ class SignalClassifier {
> 
> }  // namespace webrtc
> 
>-#endif  // MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_SIGNAL_CLASSIFIER_H_
>+#endif  // MODULES_AUDIO_PROCESSING_AGC2_SIGNAL_CLASSIFIER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/signal_classifier_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/signal_classifier_unittest.cc
>new file mode 100644
>index 00000000000..62171b32e60
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/signal_classifier_unittest.cc
>@@ -0,0 +1,82 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/agc2/signal_classifier.h"
>+
>+#include <array>
>+#include <functional>
>+#include <limits>
>+
>+#include "modules/audio_processing/agc2/agc2_testing_common.h"
>+#include "modules/audio_processing/logging/apm_data_dumper.h"
>+#include "rtc_base/gunit.h"
>+#include "rtc_base/random.h"
>+
>+namespace webrtc {
>+
>+namespace {
>+Random rand_gen(42);
>+ApmDataDumper data_dumper(0);
>+constexpr int kNumIterations = 100;
>+
>+// Runs the signal classifier on audio generated by 'sample_generator'
>+// for kNumIterations. Returns the number of frames classified as noise.
>+int RunClassifier(std::function<float()> sample_generator, int rate) {
>+  SignalClassifier classifier(&data_dumper);
>+  std::array<float, 480> signal;
>+  classifier.Initialize(rate);
>+  const size_t samples_per_channel = rtc::CheckedDivExact(rate, 100);
>+  int number_of_noise_frames = 0;
>+  for (int i = 0; i < kNumIterations; ++i) {
>+    for (size_t j = 0; j < samples_per_channel; ++j) {
>+      signal[j] = sample_generator();
>+    }
>+    number_of_noise_frames +=
>+        classifier.Analyze({&signal[0], samples_per_channel}) ==
>+        SignalClassifier::SignalType::kStationary;
>+  }
>+  return number_of_noise_frames;
>+}
>+
>+float WhiteNoiseGenerator() {
>+  return static_cast<float>(rand_gen.Rand(std::numeric_limits<int16_t>::min(),
>+                                          std::numeric_limits<int16_t>::max()));
>+}
>+}  // namespace
>+
>+// White random noise is stationary, but does not trigger the detector
>+// every frame due to the randomness.
>+TEST(AutomaticGainController2SignalClassifier, WhiteNoise) {
>+  for (const auto rate : {8000, 16000, 32000, 48000}) {
>+    const int number_of_noise_frames = RunClassifier(WhiteNoiseGenerator, rate);
>+    EXPECT_GT(number_of_noise_frames, kNumIterations / 2);
>+  }
>+}
>+
>+// Sine curves are (very) stationary. They trigger the detector all
>+// the time. Except for a few initial frames.
>+TEST(AutomaticGainController2SignalClassifier, SineTone) {
>+  for (const auto rate : {8000, 16000, 32000, 48000}) {
>+    test::SineGenerator gen(600.f, rate);
>+    const int number_of_noise_frames = RunClassifier(gen, rate);
>+    EXPECT_GE(number_of_noise_frames, kNumIterations - 5);
>+  }
>+}
>+
>+// Pulses are transient if they are far enough apart. They shouldn't
>+// trigger the noise detector.
>+TEST(AutomaticGainController2SignalClassifier, PulseTone) {
>+  for (const auto rate : {8000, 16000, 32000, 48000}) {
>+    test::PulseGenerator gen(30.f, rate);
>+    const int number_of_noise_frames = RunClassifier(gen, rate);
>+    EXPECT_EQ(number_of_noise_frames, 0);
>+  }
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/vad_with_level.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/vad_with_level.cc
>new file mode 100644
>index 00000000000..decfacd0c18
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/vad_with_level.cc
>@@ -0,0 +1,68 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/agc2/vad_with_level.h"
>+
>+#include <algorithm>
>+
>+#include "common_audio/include/audio_util.h"
>+#include "modules/audio_processing/agc2/rnn_vad/common.h"
>+#include "rtc_base/checks.h"
>+
>+namespace webrtc {
>+
>+namespace {
>+float ProcessForPeak(AudioFrameView<const float> frame) {
>+  float current_max = 0;
>+  for (const auto& x : frame.channel(0)) {
>+    current_max = std::max(std::fabs(x), current_max);
>+  }
>+  return current_max;
>+}
>+
>+float ProcessForRms(AudioFrameView<const float> frame) {
>+  float rms = 0;
>+  for (const auto& x : frame.channel(0)) {
>+    rms += x * x;
>+  }
>+  return sqrt(rms / frame.samples_per_channel());
>+}
>+}  // namespace
>+
>+VadWithLevel::VadWithLevel() = default;
>+VadWithLevel::~VadWithLevel() = default;
>+
>+VadWithLevel::LevelAndProbability VadWithLevel::AnalyzeFrame(
>+    AudioFrameView<const float> frame) {
>+  SetSampleRate(static_cast<int>(frame.samples_per_channel() * 100));
>+  std::array<float, rnn_vad::kFrameSize10ms24kHz> work_frame;
>+  // Feed the 1st channel to the resampler.
>+  resampler_.Resample(frame.channel(0).data(), frame.samples_per_channel(),
>+                      work_frame.data(), rnn_vad::kFrameSize10ms24kHz);
>+
>+  std::array<float, rnn_vad::kFeatureVectorSize> feature_vector;
>+
>+  const bool is_silence = features_extractor_.CheckSilenceComputeFeatures(
>+      work_frame, feature_vector);
>+  const float vad_probability =
>+      rnn_vad_.ComputeVadProbability(feature_vector, is_silence);
>+  return LevelAndProbability(vad_probability,
>+                             FloatS16ToDbfs(ProcessForRms(frame)),
>+                             FloatS16ToDbfs(ProcessForPeak(frame)));
>+}
>+
>+void VadWithLevel::SetSampleRate(int sample_rate_hz) {
>+  // The source number of channels in 1, because we always use the 1st
>+  // channel.
>+  resampler_.InitializeIfNeeded(sample_rate_hz, rnn_vad::kSampleRate24kHz,
>+                                1 /* num_channels */);
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/vad_with_level.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/vad_with_level.h
>new file mode 100644
>index 00000000000..67a00ced6c1
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/vad_with_level.h
>@@ -0,0 +1,49 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_AGC2_VAD_WITH_LEVEL_H_
>+#define MODULES_AUDIO_PROCESSING_AGC2_VAD_WITH_LEVEL_H_
>+
>+#include "api/array_view.h"
>+#include "common_audio/resampler/include/push_resampler.h"
>+#include "modules/audio_processing/agc2/rnn_vad/features_extraction.h"
>+#include "modules/audio_processing/agc2/rnn_vad/rnn.h"
>+#include "modules/audio_processing/include/audio_frame_view.h"
>+
>+namespace webrtc {
>+class VadWithLevel {
>+ public:
>+  struct LevelAndProbability {
>+    constexpr LevelAndProbability(float prob, float rms, float peak)
>+        : speech_probability(prob),
>+          speech_rms_dbfs(rms),
>+          speech_peak_dbfs(peak) {}
>+    LevelAndProbability() = default;
>+    float speech_probability = 0;
>+    float speech_rms_dbfs = 0;  // Root mean square in decibels to full-scale.
>+    float speech_peak_dbfs = 0;
>+  };
>+
>+  VadWithLevel();
>+  ~VadWithLevel();
>+
>+  LevelAndProbability AnalyzeFrame(AudioFrameView<const float> frame);
>+
>+ private:
>+  void SetSampleRate(int sample_rate_hz);
>+
>+  rnn_vad::RnnBasedVad rnn_vad_;
>+  rnn_vad::FeaturesExtractor features_extractor_;
>+  PushResampler<float> resampler_;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_AGC2_VAD_WITH_LEVEL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/vector_float_frame.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/vector_float_frame.cc
>new file mode 100644
>index 00000000000..a70d8151960
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/vector_float_frame.cc
>@@ -0,0 +1,39 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/agc2/vector_float_frame.h"
>+
>+namespace webrtc {
>+
>+namespace {
>+
>+std::vector<float*> ConstructChannelPointers(
>+    std::vector<std::vector<float>>* x) {
>+  std::vector<float*> channel_ptrs;
>+  for (auto& v : *x) {
>+    channel_ptrs.push_back(v.data());
>+  }
>+  return channel_ptrs;
>+}
>+}  // namespace
>+
>+VectorFloatFrame::VectorFloatFrame(int num_channels,
>+                                   int samples_per_channel,
>+                                   float start_value)
>+    : channels_(num_channels,
>+                std::vector<float>(samples_per_channel, start_value)),
>+      channel_ptrs_(ConstructChannelPointers(&channels_)),
>+      float_frame_view_(channel_ptrs_.data(),
>+                        channels_.size(),
>+                        samples_per_channel) {}
>+
>+VectorFloatFrame::~VectorFloatFrame() = default;
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/vector_float_frame.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/vector_float_frame.h
>new file mode 100644
>index 00000000000..b521f346f95
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/vector_float_frame.h
>@@ -0,0 +1,42 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_AGC2_VECTOR_FLOAT_FRAME_H_
>+#define MODULES_AUDIO_PROCESSING_AGC2_VECTOR_FLOAT_FRAME_H_
>+
>+#include <vector>
>+
>+#include "modules/audio_processing/include/audio_frame_view.h"
>+
>+namespace webrtc {
>+
>+// A construct consisting of a multi-channel audio frame, and a FloatFrame view
>+// of it.
>+class VectorFloatFrame {
>+ public:
>+  VectorFloatFrame(int num_channels,
>+                   int samples_per_channel,
>+                   float start_value);
>+  const AudioFrameView<float>& float_frame_view() { return float_frame_view_; }
>+  AudioFrameView<const float> float_frame_view() const {
>+    return float_frame_view_;
>+  }
>+
>+  ~VectorFloatFrame();
>+
>+ private:
>+  std::vector<std::vector<float>> channels_;
>+  std::vector<float*> channel_ptrs_;
>+  AudioFrameView<float> float_frame_view_;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_AGC2_VECTOR_FLOAT_FRAME_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_buffer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_buffer.cc
>index 16f11742e3f..f163f5a07de 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_buffer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_buffer.cc
>@@ -49,20 +49,20 @@ AudioBuffer::AudioBuffer(size_t input_num_frames,
>                          size_t process_num_frames,
>                          size_t num_process_channels,
>                          size_t output_num_frames)
>-  : input_num_frames_(input_num_frames),
>-    num_input_channels_(num_input_channels),
>-    proc_num_frames_(process_num_frames),
>-    num_proc_channels_(num_process_channels),
>-    output_num_frames_(output_num_frames),
>-    num_channels_(num_process_channels),
>-    num_bands_(NumBandsFromSamplesPerChannel(proc_num_frames_)),
>-    num_split_frames_(rtc::CheckedDivExact(proc_num_frames_, num_bands_)),
>-    mixed_low_pass_valid_(false),
>-    reference_copied_(false),
>-    activity_(AudioFrame::kVadUnknown),
>-    keyboard_data_(NULL),
>-    data_(new IFChannelBuffer(proc_num_frames_, num_proc_channels_)),
>-    output_buffer_(new IFChannelBuffer(output_num_frames_, num_channels_)) {
>+    : input_num_frames_(input_num_frames),
>+      num_input_channels_(num_input_channels),
>+      proc_num_frames_(process_num_frames),
>+      num_proc_channels_(num_process_channels),
>+      output_num_frames_(output_num_frames),
>+      num_channels_(num_process_channels),
>+      num_bands_(NumBandsFromSamplesPerChannel(proc_num_frames_)),
>+      num_split_frames_(rtc::CheckedDivExact(proc_num_frames_, num_bands_)),
>+      mixed_low_pass_valid_(false),
>+      reference_copied_(false),
>+      activity_(AudioFrame::kVadUnknown),
>+      keyboard_data_(NULL),
>+      data_(new IFChannelBuffer(proc_num_frames_, num_proc_channels_)),
>+      output_buffer_(new IFChannelBuffer(output_num_frames_, num_channels_)) {
>   RTC_DCHECK_GT(input_num_frames_, 0);
>   RTC_DCHECK_GT(proc_num_frames_, 0);
>   RTC_DCHECK_GT(output_num_frames_, 0);
>@@ -73,8 +73,8 @@ AudioBuffer::AudioBuffer(size_t input_num_frames,
>   if (input_num_frames_ != proc_num_frames_ ||
>       output_num_frames_ != proc_num_frames_) {
>     // Create an intermediate buffer for resampling.
>-    process_buffer_.reset(new ChannelBuffer<float>(proc_num_frames_,
>-                                                   num_proc_channels_));
>+    process_buffer_.reset(
>+        new ChannelBuffer<float>(proc_num_frames_, num_proc_channels_));
> 
>     if (input_num_frames_ != proc_num_frames_) {
>       for (size_t i = 0; i < num_proc_channels_; ++i) {
>@@ -92,12 +92,10 @@ AudioBuffer::AudioBuffer(size_t input_num_frames,
>   }
> 
>   if (num_bands_ > 1) {
>-    split_data_.reset(new IFChannelBuffer(proc_num_frames_,
>-                                          num_proc_channels_,
>-                                          num_bands_));
>-    splitting_filter_.reset(new SplittingFilter(num_proc_channels_,
>-                                                num_bands_,
>-                                                proc_num_frames_));
>+    split_data_.reset(
>+        new IFChannelBuffer(proc_num_frames_, num_proc_channels_, num_bands_));
>+    splitting_filter_.reset(
>+        new SplittingFilter(num_proc_channels_, num_bands_, proc_num_frames_));
>   }
> }
> 
>@@ -132,8 +130,7 @@ void AudioBuffer::CopyFrom(const float* const* data,
>   // Resample.
>   if (input_num_frames_ != proc_num_frames_) {
>     for (size_t i = 0; i < num_proc_channels_; ++i) {
>-      input_resamplers_[i]->Resample(data_ptr[i],
>-                                     input_num_frames_,
>+      input_resamplers_[i]->Resample(data_ptr[i], input_num_frames_,
>                                      process_buffer_->channels()[i],
>                                      proc_num_frames_);
>     }
>@@ -142,8 +139,7 @@ void AudioBuffer::CopyFrom(const float* const* data,
> 
>   // Convert to the S16 range.
>   for (size_t i = 0; i < num_proc_channels_; ++i) {
>-    FloatToFloatS16(data_ptr[i],
>-                    proc_num_frames_,
>+    FloatToFloatS16(data_ptr[i], proc_num_frames_,
>                     data_->fbuf()->channels()[i]);
>   }
> }
>@@ -161,17 +157,14 @@ void AudioBuffer::CopyTo(const StreamConfig& stream_config,
>     data_ptr = process_buffer_->channels();
>   }
>   for (size_t i = 0; i < num_channels_; ++i) {
>-    FloatS16ToFloat(data_->fbuf()->channels()[i],
>-                    proc_num_frames_,
>+    FloatS16ToFloat(data_->fbuf()->channels()[i], proc_num_frames_,
>                     data_ptr[i]);
>   }
> 
>   // Resample.
>   if (output_num_frames_ != proc_num_frames_) {
>     for (size_t i = 0; i < num_channels_; ++i) {
>-      output_resamplers_[i]->Resample(data_ptr[i],
>-                                      proc_num_frames_,
>-                                      data[i],
>+      output_resamplers_[i]->Resample(data_ptr[i], proc_num_frames_, data[i],
>                                       output_num_frames_);
>     }
>   }
>@@ -204,16 +197,14 @@ int16_t* const* AudioBuffer::channels() {
> }
> 
> const int16_t* const* AudioBuffer::split_bands_const(size_t channel) const {
>-  return split_data_.get() ?
>-         split_data_->ibuf_const()->bands(channel) :
>-         data_->ibuf_const()->bands(channel);
>+  return split_data_.get() ? split_data_->ibuf_const()->bands(channel)
>+                           : data_->ibuf_const()->bands(channel);
> }
> 
> int16_t* const* AudioBuffer::split_bands(size_t channel) {
>   mixed_low_pass_valid_ = false;
>-  return split_data_.get() ?
>-         split_data_->ibuf()->bands(channel) :
>-         data_->ibuf()->bands(channel);
>+  return split_data_.get() ? split_data_->ibuf()->bands(channel)
>+                           : data_->ibuf()->bands(channel);
> }
> 
> const int16_t* const* AudioBuffer::split_channels_const(Band band) const {
>@@ -261,16 +252,14 @@ float* const* AudioBuffer::channels_f() {
> }
> 
> const float* const* AudioBuffer::split_bands_const_f(size_t channel) const {
>-  return split_data_.get() ?
>-         split_data_->fbuf_const()->bands(channel) :
>-         data_->fbuf_const()->bands(channel);
>+  return split_data_.get() ? split_data_->fbuf_const()->bands(channel)
>+                           : data_->fbuf_const()->bands(channel);
> }
> 
> float* const* AudioBuffer::split_bands_f(size_t channel) {
>   mixed_low_pass_valid_ = false;
>-  return split_data_.get() ?
>-         split_data_->fbuf()->bands(channel) :
>-         data_->fbuf()->bands(channel);
>+  return split_data_.get() ? split_data_->fbuf()->bands(channel)
>+                           : data_->fbuf()->bands(channel);
> }
> 
> const float* const* AudioBuffer::split_channels_const_f(Band band) const {
>@@ -401,19 +390,16 @@ void AudioBuffer::DeinterleaveFrom(AudioFrame* frame) {
>                              num_input_channels_, deinterleaved[0]);
>   } else {
>     RTC_DCHECK_EQ(num_proc_channels_, num_input_channels_);
>-    Deinterleave(frame->data(),
>-                 input_num_frames_,
>-                 num_proc_channels_,
>+    Deinterleave(frame->data(), input_num_frames_, num_proc_channels_,
>                  deinterleaved);
>   }
> 
>   // Resample.
>   if (input_num_frames_ != proc_num_frames_) {
>     for (size_t i = 0; i < num_proc_channels_; ++i) {
>-      input_resamplers_[i]->Resample(input_buffer_->fbuf_const()->channels()[i],
>-                                     input_num_frames_,
>-                                     data_->fbuf()->channels()[i],
>-                                     proc_num_frames_);
>+      input_resamplers_[i]->Resample(
>+          input_buffer_->fbuf_const()->channels()[i], input_num_frames_,
>+          data_->fbuf()->channels()[i], proc_num_frames_);
>     }
>   }
> }
>@@ -453,8 +439,7 @@ void AudioBuffer::CopyLowPassToReference() {
>   if (!low_pass_reference_channels_.get() ||
>       low_pass_reference_channels_->num_channels() != num_channels_) {
>     low_pass_reference_channels_.reset(
>-        new ChannelBuffer<int16_t>(num_split_frames_,
>-                                   num_proc_channels_));
>+        new ChannelBuffer<int16_t>(num_split_frames_, num_proc_channels_));
>   }
>   for (size_t i = 0; i < num_proc_channels_; i++) {
>     memcpy(low_pass_reference_channels_->channels()[i],
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_buffer.h
>index 8451bdeeaa7..ade3eec13db 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_buffer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_buffer.h
>@@ -14,22 +14,17 @@
> #include <memory>
> #include <vector>
> 
>+#include "api/audio/audio_frame.h"
> #include "common_audio/channel_buffer.h"
> #include "modules/audio_processing/include/audio_processing.h"
> #include "modules/audio_processing/splitting_filter.h"
>-#include "modules/include/module_common_types.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
> class PushSincResampler;
> class IFChannelBuffer;
> 
>-enum Band {
>-  kBand0To8kHz = 0,
>-  kBand8To16kHz = 1,
>-  kBand16To24kHz = 2
>-};
>+enum Band { kBand0To8kHz = 0, kBand8To16kHz = 1, kBand16To24kHz = 2 };
> 
> class AudioBuffer {
>  public:
>@@ -152,11 +147,11 @@ class AudioBuffer {
>   std::unique_ptr<IFChannelBuffer> data_;
>   std::unique_ptr<IFChannelBuffer> split_data_;
>   std::unique_ptr<SplittingFilter> splitting_filter_;
>-  std::unique_ptr<ChannelBuffer<int16_t> > mixed_low_pass_channels_;
>-  std::unique_ptr<ChannelBuffer<int16_t> > low_pass_reference_channels_;
>+  std::unique_ptr<ChannelBuffer<int16_t>> mixed_low_pass_channels_;
>+  std::unique_ptr<ChannelBuffer<int16_t>> low_pass_reference_channels_;
>   std::unique_ptr<IFChannelBuffer> input_buffer_;
>   std::unique_ptr<IFChannelBuffer> output_buffer_;
>-  std::unique_ptr<ChannelBuffer<float> > process_buffer_;
>+  std::unique_ptr<ChannelBuffer<float>> process_buffer_;
>   std::vector<std::unique_ptr<PushSincResampler>> input_resamplers_;
>   std::vector<std::unique_ptr<PushSincResampler>> output_resamplers_;
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_frame_view_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_frame_view_unittest.cc
>new file mode 100644
>index 00000000000..7a9d1261030
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_frame_view_unittest.cc
>@@ -0,0 +1,51 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/include/audio_frame_view.h"
>+
>+#include "modules/audio_processing/audio_buffer.h"
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+TEST(AudioFrameTest, ConstructFromAudioBuffer) {
>+  constexpr int kSampleRateHz = 48000;
>+  constexpr int kNumChannels = 2;
>+  constexpr float kFloatConstant = 1272.f;
>+  constexpr float kIntConstant = 17252;
>+  const webrtc::StreamConfig stream_config(kSampleRateHz, kNumChannels, false);
>+  webrtc::AudioBuffer buffer(
>+      stream_config.num_frames(), stream_config.num_channels(),
>+      stream_config.num_frames(), stream_config.num_channels(),
>+      stream_config.num_frames());
>+
>+  AudioFrameView<float> non_const_view(
>+      buffer.channels_f(), buffer.num_channels(), buffer.num_frames());
>+  // Modification is allowed.
>+  non_const_view.channel(0)[0] = kFloatConstant;
>+  EXPECT_EQ(buffer.channels_f()[0][0], kFloatConstant);
>+
>+  AudioFrameView<const float> const_view(
>+      buffer.channels_f(), buffer.num_channels(), buffer.num_frames());
>+  // Modification is not allowed.
>+  // const_view.channel(0)[0] = kFloatConstant;
>+
>+  // Assignment is allowed.
>+  AudioFrameView<const float> other_const_view = non_const_view;
>+  static_cast<void>(other_const_view);
>+
>+  // But not the other way. The following will fail:
>+  // non_const_view = other_const_view;
>+
>+  AudioFrameView<int16_t> non_const_int16_view(
>+      buffer.channels(), buffer.num_channels(), buffer.num_frames());
>+  non_const_int16_view.channel(0)[0] = kIntConstant;
>+  EXPECT_EQ(buffer.channels()[0][0], kIntConstant);
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_generator/file_audio_generator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_generator/file_audio_generator.cc
>new file mode 100644
>index 00000000000..0d691b8bbf9
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_generator/file_audio_generator.cc
>@@ -0,0 +1,36 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/audio_generator/file_audio_generator.h"
>+
>+namespace webrtc {
>+
>+FileAudioGenerator::FileAudioGenerator(
>+    std::unique_ptr<WavReader> input_audio_file) {
>+  // TODO(bugs.webrtc.org/8882) Stub.
>+  // Read audio from file into internal buffer.
>+}
>+
>+FileAudioGenerator::~FileAudioGenerator() = default;
>+
>+void FileAudioGenerator::FillFrame(AudioFrameView<float> audio) {
>+  // TODO(bugs.webrtc.org/8882) Stub.
>+  // Fill |audio| from internal buffer.
>+}
>+
>+size_t FileAudioGenerator::NumChannels() {
>+  return num_channels_;
>+}
>+
>+size_t FileAudioGenerator::SampleRateHz() {
>+  return sample_rate_hz_;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_generator/file_audio_generator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_generator/file_audio_generator.h
>new file mode 100644
>index 00000000000..01979a42b30
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_generator/file_audio_generator.h
>@@ -0,0 +1,48 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_AUDIO_GENERATOR_FILE_AUDIO_GENERATOR_H_
>+#define MODULES_AUDIO_PROCESSING_AUDIO_GENERATOR_FILE_AUDIO_GENERATOR_H_
>+
>+#include <memory>
>+
>+#include "common_audio/wav_file.h"
>+#include "modules/audio_processing/include/audio_generator.h"
>+#include "rtc_base/constructormagic.h"
>+
>+namespace webrtc {
>+
>+// Provides looping audio from a file. The file is read in its entirety on
>+// construction and then closed. This class wraps a webrtc::WavReader, and is
>+// hence unsuitable for non-diagnostic code.
>+class FileAudioGenerator : public AudioGenerator {
>+ public:
>+  // Reads the playout audio from a given WAV file.
>+  explicit FileAudioGenerator(std::unique_ptr<WavReader> input_audio_file);
>+
>+  ~FileAudioGenerator() override;
>+
>+  // Fill |audio| with audio from a file.
>+  void FillFrame(AudioFrameView<float> audio) override;
>+
>+  size_t NumChannels() override;
>+
>+  size_t SampleRateHz() override;
>+
>+ private:
>+  size_t num_channels_;
>+  size_t sample_rate_hz_;
>+
>+  RTC_DISALLOW_COPY_AND_ASSIGN(FileAudioGenerator);
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_AUDIO_GENERATOR_FILE_AUDIO_GENERATOR_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_generator/file_audio_generator_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_generator/file_audio_generator_unittest.cc
>new file mode 100644
>index 00000000000..6ed3b94667e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_generator/file_audio_generator_unittest.cc
>@@ -0,0 +1,31 @@
>+/*
>+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <string>
>+
>+#include "modules/audio_processing/include/audio_generator_factory.h"
>+#include "modules/audio_processing/include/audio_processing.h"
>+#include "test/gtest.h"
>+#include "test/testsupport/fileutils.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+// TODO(bugs.webrtc.org/8882) Stub.
>+// Add unit tests for both file audio and generated audio.
>+
>+TEST(FileAudioGeneratorTest, CreationDeletion) {
>+  const std::string audio_filename =
>+      test::ResourcePath("voice_engine/audio_tiny48", "wav");
>+  auto audio_generator = AudioGeneratorFactory::Create(audio_filename);
>+}
>+
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_processing_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_processing_impl.cc
>index accd03b255d..52693859aeb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_processing_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_processing_impl.cc
>@@ -19,33 +19,35 @@
> #include "common_audio/include/audio_util.h"
> #include "common_audio/signal_processing/include/signal_processing_library.h"
> #include "modules/audio_processing/aec/aec_core.h"
>-#include "modules/audio_processing/aec3/echo_canceller3.h"
> #include "modules/audio_processing/agc/agc_manager_direct.h"
>-#include "modules/audio_processing/agc2/gain_controller2.h"
>+#include "modules/audio_processing/agc2/gain_applier.h"
> #include "modules/audio_processing/audio_buffer.h"
>-#include "modules/audio_processing/beamformer/nonlinear_beamformer.h"
> #include "modules/audio_processing/common.h"
> #include "modules/audio_processing/echo_cancellation_impl.h"
>+#include "modules/audio_processing/echo_cancellation_proxy.h"
> #include "modules/audio_processing/echo_control_mobile_impl.h"
>+#include "modules/audio_processing/echo_control_mobile_proxy.h"
> #include "modules/audio_processing/gain_control_for_experimental_agc.h"
> #include "modules/audio_processing/gain_control_impl.h"
>+#include "modules/audio_processing/gain_controller2.h"
>+#include "modules/audio_processing/logging/apm_data_dumper.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/platform_file.h"
> #include "rtc_base/refcountedobject.h"
>+#include "rtc_base/system/arch.h"
>+#include "rtc_base/timeutils.h"
> #include "rtc_base/trace_event.h"
> #if WEBRTC_INTELLIGIBILITY_ENHANCER
> #include "modules/audio_processing/intelligibility/intelligibility_enhancer.h"
> #endif
>-#include "modules/audio_processing/level_controller/level_controller.h"
> #include "modules/audio_processing/level_estimator_impl.h"
> #include "modules/audio_processing/low_cut_filter.h"
> #include "modules/audio_processing/noise_suppression_impl.h"
> #include "modules/audio_processing/residual_echo_detector.h"
> #include "modules/audio_processing/transient/transient_suppressor.h"
> #include "modules/audio_processing/voice_detection_impl.h"
>-#include "modules/include/module_common_types.h"
>-#include "system_wrappers/include/file_wrapper.h"
>+#include "rtc_base/atomicops.h"
> #include "system_wrappers/include/metrics.h"
> 
> // Check to verify that the define for the intelligibility enhancer is properly
>@@ -67,6 +69,7 @@
> namespace webrtc {
> 
> constexpr int AudioProcessing::kNativeSampleRatesHz[];
>+constexpr int kRuntimeSettingQueueSize = 100;
> 
> namespace {
> 
>@@ -148,24 +151,6 @@ class HighPassFilterImpl : public HighPassFilter {
>   AudioProcessingImpl* apm_;
>   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(HighPassFilterImpl);
> };
>-
>-webrtc::InternalAPMStreamsConfig ToStreamsConfig(
>-    const ProcessingConfig& api_format) {
>-  webrtc::InternalAPMStreamsConfig result;
>-  result.input_sample_rate = api_format.input_stream().sample_rate_hz();
>-  result.input_num_channels = api_format.input_stream().num_channels();
>-  result.output_num_channels = api_format.output_stream().num_channels();
>-  result.render_input_num_channels =
>-      api_format.reverse_input_stream().num_channels();
>-  result.render_input_sample_rate =
>-      api_format.reverse_input_stream().sample_rate_hz();
>-  result.output_sample_rate = api_format.output_stream().sample_rate_hz();
>-  result.render_output_sample_rate =
>-      api_format.reverse_output_stream().sample_rate_hz();
>-  result.render_output_num_channels =
>-      api_format.reverse_output_stream().num_channels();
>-  return result;
>-}
> }  // namespace
> 
> // Throughout webrtc, it's assumed that success is represented by zero.
>@@ -184,10 +169,9 @@ bool AudioProcessingImpl::ApmSubmoduleStates::Update(
>     bool residual_echo_detector_enabled,
>     bool noise_suppressor_enabled,
>     bool intelligibility_enhancer_enabled,
>-    bool beamformer_enabled,
>     bool adaptive_gain_controller_enabled,
>     bool gain_controller2_enabled,
>-    bool level_controller_enabled,
>+    bool pre_amplifier_enabled,
>     bool echo_controller_enabled,
>     bool voice_activity_detector_enabled,
>     bool level_estimator_enabled,
>@@ -202,12 +186,11 @@ bool AudioProcessingImpl::ApmSubmoduleStates::Update(
>   changed |= (noise_suppressor_enabled != noise_suppressor_enabled_);
>   changed |=
>       (intelligibility_enhancer_enabled != intelligibility_enhancer_enabled_);
>-  changed |= (beamformer_enabled != beamformer_enabled_);
>   changed |=
>       (adaptive_gain_controller_enabled != adaptive_gain_controller_enabled_);
>   changed |=
>       (gain_controller2_enabled != gain_controller2_enabled_);
>-  changed |= (level_controller_enabled != level_controller_enabled_);
>+  changed |= (pre_amplifier_enabled_ != pre_amplifier_enabled);
>   changed |= (echo_controller_enabled != echo_controller_enabled_);
>   changed |= (level_estimator_enabled != level_estimator_enabled_);
>   changed |=
>@@ -220,10 +203,9 @@ bool AudioProcessingImpl::ApmSubmoduleStates::Update(
>     residual_echo_detector_enabled_ = residual_echo_detector_enabled;
>     noise_suppressor_enabled_ = noise_suppressor_enabled;
>     intelligibility_enhancer_enabled_ = intelligibility_enhancer_enabled;
>-    beamformer_enabled_ = beamformer_enabled;
>     adaptive_gain_controller_enabled_ = adaptive_gain_controller_enabled;
>     gain_controller2_enabled_ = gain_controller2_enabled;
>-    level_controller_enabled_ = level_controller_enabled;
>+    pre_amplifier_enabled_ = pre_amplifier_enabled;
>     echo_controller_enabled_ = echo_controller_enabled;
>     level_estimator_enabled_ = level_estimator_enabled;
>     voice_activity_detector_enabled_ = voice_activity_detector_enabled;
>@@ -249,14 +231,13 @@ bool AudioProcessingImpl::ApmSubmoduleStates::CaptureMultiBandProcessingActive()
>     const {
>   return low_cut_filter_enabled_ || echo_canceller_enabled_ ||
>          mobile_echo_controller_enabled_ || noise_suppressor_enabled_ ||
>-         beamformer_enabled_ || adaptive_gain_controller_enabled_ ||
>-         echo_controller_enabled_;
>+         adaptive_gain_controller_enabled_ || echo_controller_enabled_;
> }
> 
> bool AudioProcessingImpl::ApmSubmoduleStates::CaptureFullBandProcessingActive()
>     const {
>-  return level_controller_enabled_ || gain_controller2_enabled_ ||
>-         capture_post_processor_enabled_;
>+  return gain_controller2_enabled_ || capture_post_processor_enabled_ ||
>+         pre_amplifier_enabled_;
> }
> 
> bool AudioProcessingImpl::ApmSubmoduleStates::RenderMultiBandSubModulesActive()
>@@ -285,6 +266,8 @@ struct AudioProcessingImpl::ApmPublicSubmodules {
>   // Accessed externally of APM without any lock acquired.
>   std::unique_ptr<EchoCancellationImpl> echo_cancellation;
>   std::unique_ptr<EchoControlMobileImpl> echo_control_mobile;
>+  std::unique_ptr<EchoCancellationProxy> echo_cancellation_proxy;
>+  std::unique_ptr<EchoControlMobileProxy> echo_control_mobile_proxy;
>   std::unique_ptr<GainControlImpl> gain_control;
>   std::unique_ptr<LevelEstimatorImpl> level_estimator;
>   std::unique_ptr<NoiseSuppressionImpl> noise_suppression;
>@@ -300,24 +283,21 @@ struct AudioProcessingImpl::ApmPublicSubmodules {
> };
> 
> struct AudioProcessingImpl::ApmPrivateSubmodules {
>-  ApmPrivateSubmodules(NonlinearBeamformer* beamformer,
>-                       std::unique_ptr<CustomProcessing> capture_post_processor,
>+  ApmPrivateSubmodules(std::unique_ptr<CustomProcessing> capture_post_processor,
>                        std::unique_ptr<CustomProcessing> render_pre_processor,
>-                       std::unique_ptr<EchoDetector> echo_detector)
>-      : beamformer(beamformer),
>-        echo_detector(std::move(echo_detector)),
>+                       rtc::scoped_refptr<EchoDetector> echo_detector)
>+      : echo_detector(std::move(echo_detector)),
>         capture_post_processor(std::move(capture_post_processor)),
>         render_pre_processor(std::move(render_pre_processor)) {}
>   // Accessed internally from capture or during initialization
>-  std::unique_ptr<NonlinearBeamformer> beamformer;
>   std::unique_ptr<AgcManagerDirect> agc_manager;
>   std::unique_ptr<GainController2> gain_controller2;
>   std::unique_ptr<LowCutFilter> low_cut_filter;
>-  std::unique_ptr<LevelController> level_controller;
>-  std::unique_ptr<EchoDetector> echo_detector;
>+  rtc::scoped_refptr<EchoDetector> echo_detector;
>   std::unique_ptr<EchoControl> echo_controller;
>   std::unique_ptr<CustomProcessing> capture_post_processor;
>   std::unique_ptr<CustomProcessing> render_pre_processor;
>+  std::unique_ptr<GainApplier> pre_amplifier;
> };
> 
> AudioProcessingBuilder::AudioProcessingBuilder() = default;
>@@ -341,14 +321,8 @@ AudioProcessingBuilder& AudioProcessingBuilder::SetEchoControlFactory(
>   return *this;
> }
> 
>-AudioProcessingBuilder& AudioProcessingBuilder::SetNonlinearBeamformer(
>-    std::unique_ptr<NonlinearBeamformer> nonlinear_beamformer) {
>-  nonlinear_beamformer_ = std::move(nonlinear_beamformer);
>-  return *this;
>-}
>-
> AudioProcessingBuilder& AudioProcessingBuilder::SetEchoDetector(
>-    std::unique_ptr<EchoDetector> echo_detector) {
>+    rtc::scoped_refptr<EchoDetector> echo_detector) {
>   echo_detector_ = std::move(echo_detector);
>   return *this;
> }
>@@ -362,7 +336,7 @@ AudioProcessing* AudioProcessingBuilder::Create(const webrtc::Config& config) {
>   AudioProcessingImpl* apm = new rtc::RefCountedObject<AudioProcessingImpl>(
>       config, std::move(capture_post_processing_),
>       std::move(render_pre_processing_), std::move(echo_control_factory_),
>-      std::move(echo_detector_), nonlinear_beamformer_.release());
>+      std::move(echo_detector_));
>   if (apm->Initialize() != AudioProcessing::kNoError) {
>     delete apm;
>     apm = nullptr;
>@@ -370,83 +344,48 @@ AudioProcessing* AudioProcessingBuilder::Create(const webrtc::Config& config) {
>   return apm;
> }
> 
>-AudioProcessing* AudioProcessing::Create() {
>-  webrtc::Config config;
>-  return Create(config, nullptr, nullptr, nullptr, nullptr);
>-}
>-
>-AudioProcessing* AudioProcessing::Create(const webrtc::Config& config) {
>-  return Create(config, nullptr, nullptr, nullptr, nullptr);
>-}
>-
>-AudioProcessing* AudioProcessing::Create(const webrtc::Config& config,
>-                                         NonlinearBeamformer* beamformer) {
>-  return Create(config, nullptr, nullptr, nullptr, beamformer);
>-}
>-
>-AudioProcessing* AudioProcessing::Create(
>-    const webrtc::Config& config,
>-    std::unique_ptr<CustomProcessing> capture_post_processor,
>-    std::unique_ptr<EchoControlFactory> echo_control_factory,
>-    NonlinearBeamformer* beamformer) {
>-  return Create(config, std::move(capture_post_processor), nullptr,
>-                std::move(echo_control_factory), beamformer);
>-}
>-
>-AudioProcessing* AudioProcessing::Create(
>-    const webrtc::Config& config,
>-    std::unique_ptr<CustomProcessing> capture_post_processor,
>-    std::unique_ptr<CustomProcessing> render_pre_processor,
>-    std::unique_ptr<EchoControlFactory> echo_control_factory,
>-    NonlinearBeamformer* beamformer) {
>-  AudioProcessingImpl* apm = new rtc::RefCountedObject<AudioProcessingImpl>(
>-      config, std::move(capture_post_processor),
>-      std::move(render_pre_processor), std::move(echo_control_factory), nullptr,
>-      beamformer);
>-  if (apm->Initialize() != kNoError) {
>-    delete apm;
>-    apm = nullptr;
>-  }
>-
>-  return apm;
>-}
>-
> AudioProcessingImpl::AudioProcessingImpl(const webrtc::Config& config)
>-    : AudioProcessingImpl(config, nullptr, nullptr, nullptr, nullptr, nullptr) {
>-}
>+    : AudioProcessingImpl(config, nullptr, nullptr, nullptr, nullptr) {}
>+
>+int AudioProcessingImpl::instance_count_ = 0;
> 
> AudioProcessingImpl::AudioProcessingImpl(
>     const webrtc::Config& config,
>     std::unique_ptr<CustomProcessing> capture_post_processor,
>     std::unique_ptr<CustomProcessing> render_pre_processor,
>     std::unique_ptr<EchoControlFactory> echo_control_factory,
>-    std::unique_ptr<EchoDetector> echo_detector,
>-    NonlinearBeamformer* beamformer)
>-    : high_pass_filter_impl_(new HighPassFilterImpl(this)),
>+    rtc::scoped_refptr<EchoDetector> echo_detector)
>+    : data_dumper_(
>+          new ApmDataDumper(rtc::AtomicOps::Increment(&instance_count_))),
>+      capture_runtime_settings_(kRuntimeSettingQueueSize),
>+      render_runtime_settings_(kRuntimeSettingQueueSize),
>+      capture_runtime_settings_enqueuer_(&capture_runtime_settings_),
>+      render_runtime_settings_enqueuer_(&render_runtime_settings_),
>+      high_pass_filter_impl_(new HighPassFilterImpl(this)),
>       echo_control_factory_(std::move(echo_control_factory)),
>       submodule_states_(!!capture_post_processor, !!render_pre_processor),
>       public_submodules_(new ApmPublicSubmodules()),
>       private_submodules_(
>-          new ApmPrivateSubmodules(beamformer,
>-                                   std::move(capture_post_processor),
>+          new ApmPrivateSubmodules(std::move(capture_post_processor),
>                                    std::move(render_pre_processor),
>                                    std::move(echo_detector))),
>       constants_(config.Get<ExperimentalAgc>().startup_min_volume,
>                  config.Get<ExperimentalAgc>().clipped_level_min,
> #if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
>+                 false,
>+                 false,
>                  false),
> #else
>-                 config.Get<ExperimentalAgc>().enabled),
>+                 config.Get<ExperimentalAgc>().enabled,
>+                 config.Get<ExperimentalAgc>().enabled_agc2_level_estimator,
>+                 config.Get<ExperimentalAgc>().digital_adaptive_disabled),
> #endif
> #if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
>-      capture_(false,
>+      capture_(false),
> #else
>-      capture_(config.Get<ExperimentalNs>().enabled,
>+      capture_(config.Get<ExperimentalNs>().enabled),
> #endif
>-               config.Get<Beamforming>().array_geometry,
>-               config.Get<Beamforming>().target_direction),
>-      capture_nonlocked_(config.Get<Beamforming>().enabled,
>-                         config.Get<Intelligibility>().enabled) {
>+      capture_nonlocked_(config.Get<Intelligibility>().enabled) {
>   {
>     rtc::CritScope cs_render(&crit_render_);
>     rtc::CritScope cs_capture(&crit_capture_);
>@@ -459,8 +398,13 @@ AudioProcessingImpl::AudioProcessingImpl(
>         new EchoCancellationImpl(&crit_render_, &crit_capture_));
>     public_submodules_->echo_control_mobile.reset(
>         new EchoControlMobileImpl(&crit_render_, &crit_capture_));
>+    public_submodules_->echo_cancellation_proxy.reset(new EchoCancellationProxy(
>+        this, public_submodules_->echo_cancellation.get()));
>+    public_submodules_->echo_control_mobile_proxy.reset(
>+        new EchoControlMobileProxy(
>+            this, public_submodules_->echo_control_mobile.get()));
>     public_submodules_->gain_control.reset(
>-        new GainControlImpl(&crit_capture_, &crit_capture_));
>+        new GainControlImpl(&crit_render_, &crit_capture_));
>     public_submodules_->level_estimator.reset(
>         new LevelEstimatorImpl(&crit_capture_));
>     public_submodules_->noise_suppression.reset(
>@@ -473,21 +417,17 @@ AudioProcessingImpl::AudioProcessingImpl(
> 
>     // If no echo detector is injected, use the ResidualEchoDetector.
>     if (!private_submodules_->echo_detector) {
>-      private_submodules_->echo_detector.reset(new ResidualEchoDetector());
>+      private_submodules_->echo_detector =
>+          new rtc::RefCountedObject<ResidualEchoDetector>();
>     }
> 
>-    // TODO(peah): Move this creation to happen only when the level controller
>-    // is enabled.
>-    private_submodules_->level_controller.reset(new LevelController());
>-
>     // TODO(alessiob): Move the injected gain controller once injection is
>     // implemented.
>     private_submodules_->gain_controller2.reset(new GainController2());
> 
>     RTC_LOG(LS_INFO) << "Capture post processor activated: "
>-                     << !!private_submodules_->capture_post_processor;
>-
>-    RTC_LOG(LS_INFO) << "Render pre processor activated: "
>+                     << !!private_submodules_->capture_post_processor
>+                     << "\nRender pre processor activated: "
>                      << !!private_submodules_->render_pre_processor;
>   }
> 
>@@ -564,11 +504,6 @@ int AudioProcessingImpl::MaybeInitialize(
> int AudioProcessingImpl::InitializeLocked() {
>   UpdateActiveSubmoduleStates();
> 
>-  const int capture_audiobuffer_num_channels =
>-      capture_nonlocked_.beamformer_enabled
>-          ? formats_.api_format.input_stream().num_channels()
>-          : formats_.api_format.output_stream().num_channels();
>-
>   const int render_audiobuffer_num_output_frames =
>       formats_.api_format.reverse_output_stream().num_frames() == 0
>           ? formats_.render_processing_format.num_frames()
>@@ -599,7 +534,7 @@ int AudioProcessingImpl::InitializeLocked() {
>       new AudioBuffer(formats_.api_format.input_stream().num_frames(),
>                       formats_.api_format.input_stream().num_channels(),
>                       capture_nonlocked_.capture_processing_format.num_frames(),
>-                      capture_audiobuffer_num_channels,
>+                      formats_.api_format.output_stream().num_channels(),
>                       formats_.api_format.output_stream().num_frames()));
> 
>   public_submodules_->echo_cancellation->Initialize(
>@@ -622,7 +557,9 @@ int AudioProcessingImpl::InitializeLocked() {
>       private_submodules_->agc_manager.reset(new AgcManagerDirect(
>           public_submodules_->gain_control.get(),
>           public_submodules_->gain_control_for_experimental_agc.get(),
>-          constants_.agc_startup_min_volume, constants_.agc_clipped_level_min));
>+          constants_.agc_startup_min_volume, constants_.agc_clipped_level_min,
>+          constants_.use_experimental_agc_agc2_level_estimation,
>+          constants_.use_experimental_agc_agc2_digital_adaptive));
>     }
>     private_submodules_->agc_manager->Initialize();
>     private_submodules_->agc_manager->SetCaptureMuted(
>@@ -630,7 +567,6 @@ int AudioProcessingImpl::InitializeLocked() {
>     public_submodules_->gain_control_for_experimental_agc->Initialize();
>   }
>   InitializeTransient();
>-  InitializeBeamformer();
> #if WEBRTC_INTELLIGIBILITY_ENHANCER
>   InitializeIntelligibility();
> #endif
>@@ -639,7 +575,6 @@ int AudioProcessingImpl::InitializeLocked() {
>                                                     proc_sample_rate_hz());
>   public_submodules_->voice_detection->Initialize(proc_split_sample_rate_hz());
>   public_submodules_->level_estimator->Initialize();
>-  InitializeLevelController();
>   InitializeResidualEchoDetector();
>   InitializeEchoController();
>   InitializeGainController2();
>@@ -647,7 +582,7 @@ int AudioProcessingImpl::InitializeLocked() {
>   InitializePreProcessor();
> 
>   if (aec_dump_) {
>-    aec_dump_->WriteInitMessage(ToStreamsConfig(formats_.api_format));
>+    aec_dump_->WriteInitMessage(formats_.api_format, rtc::TimeUTCMillis());
>   }
>   return kNoError;
> }
>@@ -671,11 +606,6 @@ int AudioProcessingImpl::InitializeLocked(const ProcessingConfig& config) {
>     return kBadNumberChannelsError;
>   }
> 
>-  if (capture_nonlocked_.beamformer_enabled &&
>-      num_in_channels != capture_.array_geometry.size()) {
>-    return kBadNumberChannelsError;
>-  }
>-
>   formats_.api_format = config;
> 
>   int capture_processing_rate = FindNativeProcessRateToUse(
>@@ -743,65 +673,30 @@ int AudioProcessingImpl::InitializeLocked(const ProcessingConfig& config) {
> void AudioProcessingImpl::ApplyConfig(const AudioProcessing::Config& config) {
>   config_ = config;
> 
>-  bool config_ok = LevelController::Validate(config_.level_controller);
>-  if (!config_ok) {
>-    RTC_LOG(LS_ERROR) << "AudioProcessing module config error" << std::endl
>-                      << "level_controller: "
>-                      << LevelController::ToString(config_.level_controller)
>-                      << std::endl
>-                      << "Reverting to default parameter set";
>-    config_.level_controller = AudioProcessing::Config::LevelController();
>-  }
>-
>   // Run in a single-threaded manner when applying the settings.
>   rtc::CritScope cs_render(&crit_render_);
>   rtc::CritScope cs_capture(&crit_capture_);
> 
>-  // TODO(peah): Replace the use of capture_nonlocked_.level_controller_enabled
>-  // with the value in config_ everywhere in the code.
>-  if (capture_nonlocked_.level_controller_enabled !=
>-      config_.level_controller.enabled) {
>-    capture_nonlocked_.level_controller_enabled =
>-        config_.level_controller.enabled;
>-    // TODO(peah): Remove the conditional initialization to always initialize
>-    // the level controller regardless of whether it is enabled or not.
>-    InitializeLevelController();
>-  }
>-  RTC_LOG(LS_INFO) << "Level controller activated: "
>-                   << capture_nonlocked_.level_controller_enabled;
>-
>-  private_submodules_->level_controller->ApplyConfig(config_.level_controller);
>-
>   InitializeLowCutFilter();
> 
>   RTC_LOG(LS_INFO) << "Highpass filter activated: "
>                    << config_.high_pass_filter.enabled;
> 
>-  // Deprecated way of activating AEC3.
>-  // TODO(gustaf): Remove when possible.
>-  if (config.echo_canceller3.enabled && !echo_control_factory_) {
>-    capture_nonlocked_.echo_controller_enabled =
>-        config_.echo_canceller3.enabled;
>-    echo_control_factory_ =
>-        std::unique_ptr<EchoControlFactory>(new EchoCanceller3Factory());
>-    InitializeEchoController();
>-    RTC_LOG(LS_INFO) << "Echo canceller 3 activated: "
>-                     << capture_nonlocked_.echo_controller_enabled;
>-  }
>-
>-  config_ok = GainController2::Validate(config_.gain_controller2);
>+  const bool config_ok = GainController2::Validate(config_.gain_controller2);
>   if (!config_ok) {
>-    RTC_LOG(LS_ERROR) << "AudioProcessing module config error" << std::endl
>-                      << "Gain Controller 2: "
>+    RTC_LOG(LS_ERROR) << "AudioProcessing module config error\n"
>+                         "Gain Controller 2: "
>                       << GainController2::ToString(config_.gain_controller2)
>-                      << std::endl
>-                      << "Reverting to default parameter set";
>+                      << "\nReverting to default parameter set";
>     config_.gain_controller2 = AudioProcessing::Config::GainController2();
>   }
>   InitializeGainController2();
>+  InitializePreAmplifier();
>   private_submodules_->gain_controller2->ApplyConfig(config_.gain_controller2);
>   RTC_LOG(LS_INFO) << "Gain Controller 2 activated: "
>                    << config_.gain_controller2.enabled;
>+  RTC_LOG(LS_INFO) << "Pre-amplifier activated: "
>+                   << config_.pre_amplifier.enabled;
> }
> 
> void AudioProcessingImpl::SetExtraOptions(const webrtc::Config& config) {
>@@ -826,18 +721,6 @@ void AudioProcessingImpl::SetExtraOptions(const webrtc::Config& config) {
>     InitializeIntelligibility();
>   }
> #endif
>-
>-#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
>-  if (capture_nonlocked_.beamformer_enabled !=
>-          config.Get<Beamforming>().enabled) {
>-    capture_nonlocked_.beamformer_enabled = config.Get<Beamforming>().enabled;
>-    if (config.Get<Beamforming>().array_geometry.size() > 1) {
>-      capture_.array_geometry = config.Get<Beamforming>().array_geometry;
>-    }
>-    capture_.target_direction = config.Get<Beamforming>().target_direction;
>-    InitializeBeamformer();
>-  }
>-#endif  // WEBRTC_ANDROID_PLATFORM_BUILD
> }
> 
> int AudioProcessingImpl::proc_sample_rate_hz() const {
>@@ -862,10 +745,7 @@ size_t AudioProcessingImpl::num_input_channels() const {
> 
> size_t AudioProcessingImpl::num_proc_channels() const {
>   // Used as callback from submodules, hence locking is not allowed.
>-  return (capture_nonlocked_.beamformer_enabled ||
>-          capture_nonlocked_.echo_controller_enabled)
>-             ? 1
>-             : num_output_channels();
>+  return capture_nonlocked_.echo_controller_enabled ? 1 : num_output_channels();
> }
> 
> size_t AudioProcessingImpl::num_output_channels() const {
>@@ -882,6 +762,44 @@ void AudioProcessingImpl::set_output_will_be_muted(bool muted) {
>   }
> }
> 
>+void AudioProcessingImpl::SetRuntimeSetting(RuntimeSetting setting) {
>+  switch (setting.type()) {
>+    case RuntimeSetting::Type::kCustomRenderProcessingRuntimeSetting:
>+      render_runtime_settings_enqueuer_.Enqueue(setting);
>+      return;
>+    case RuntimeSetting::Type::kNotSpecified:
>+      RTC_NOTREACHED();
>+      return;
>+    case RuntimeSetting::Type::kCapturePreGain:
>+      capture_runtime_settings_enqueuer_.Enqueue(setting);
>+      return;
>+  }
>+  // The language allows the enum to have a non-enumerator
>+  // value. Check that this doesn't happen.
>+  RTC_NOTREACHED();
>+}
>+
>+AudioProcessingImpl::RuntimeSettingEnqueuer::RuntimeSettingEnqueuer(
>+    SwapQueue<RuntimeSetting>* runtime_settings)
>+    : runtime_settings_(*runtime_settings) {
>+  RTC_DCHECK(runtime_settings);
>+}
>+
>+AudioProcessingImpl::RuntimeSettingEnqueuer::~RuntimeSettingEnqueuer() =
>+    default;
>+
>+void AudioProcessingImpl::RuntimeSettingEnqueuer::Enqueue(
>+    RuntimeSetting setting) {
>+  size_t remaining_attempts = 10;
>+  while (!runtime_settings_.Insert(&setting) && remaining_attempts-- > 0) {
>+    RuntimeSetting setting_to_discard;
>+    if (runtime_settings_.Remove(&setting_to_discard))
>+      RTC_LOG(LS_ERROR)
>+          << "The runtime settings queue is full. Oldest setting discarded.";
>+  }
>+  if (remaining_attempts == 0)
>+    RTC_LOG(LS_ERROR) << "Cannot enqueue a new runtime setting.";
>+}
> 
> int AudioProcessingImpl::ProcessStream(const float* const* src,
>                                        size_t samples_per_channel,
>@@ -964,6 +882,47 @@ int AudioProcessingImpl::ProcessStream(const float* const* src,
>   return kNoError;
> }
> 
>+void AudioProcessingImpl::HandleCaptureRuntimeSettings() {
>+  RuntimeSetting setting;
>+  while (capture_runtime_settings_.Remove(&setting)) {
>+    switch (setting.type()) {
>+      case RuntimeSetting::Type::kCapturePreGain:
>+        if (config_.pre_amplifier.enabled) {
>+          float value;
>+          setting.GetFloat(&value);
>+          private_submodules_->pre_amplifier->SetGainFactor(value);
>+        }
>+        // TODO(bugs.chromium.org/9138): Log setting handling by Aec Dump.
>+        break;
>+      case RuntimeSetting::Type::kCustomRenderProcessingRuntimeSetting:
>+        RTC_NOTREACHED();
>+        break;
>+      case RuntimeSetting::Type::kNotSpecified:
>+        RTC_NOTREACHED();
>+        break;
>+    }
>+  }
>+}
>+
>+void AudioProcessingImpl::HandleRenderRuntimeSettings() {
>+  RuntimeSetting setting;
>+  while (render_runtime_settings_.Remove(&setting)) {
>+    switch (setting.type()) {
>+      case RuntimeSetting::Type::kCustomRenderProcessingRuntimeSetting:
>+        if (private_submodules_->render_pre_processor) {
>+          private_submodules_->render_pre_processor->SetRuntimeSetting(setting);
>+        }
>+        break;
>+      case RuntimeSetting::Type::kCapturePreGain:
>+        RTC_NOTREACHED();
>+        break;
>+      case RuntimeSetting::Type::kNotSpecified:
>+        RTC_NOTREACHED();
>+        break;
>+    }
>+  }
>+}
>+
> void AudioProcessingImpl::QueueBandedRenderAudio(AudioBuffer* audio) {
>   EchoCancellationImpl::PackRenderAudioBuffer(audio, num_output_channels(),
>                                               num_reverse_channels(),
>@@ -1150,10 +1109,10 @@ int AudioProcessingImpl::ProcessStream(AudioFrame* frame) {
>   TRACE_EVENT0("webrtc", "AudioProcessing::ProcessStream_AudioFrame");
>   {
>     // Acquire the capture lock in order to safely call the function
>-    // that retrieves the render side data. This function accesses apm
>+    // that retrieves the render side data. This function accesses APM
>     // getters that need the capture lock held when being called.
>     // The lock needs to be released as
>-    // public_submodules_->echo_control_mobile->is_enabled() aquires this lock
>+    // public_submodules_->echo_control_mobile->is_enabled() acquires this lock
>     // as well.
>     rtc::CritScope cs_capture(&crit_capture_);
>     EmptyQueuedRenderAudio();
>@@ -1218,6 +1177,8 @@ int AudioProcessingImpl::ProcessStream(AudioFrame* frame) {
> }
> 
> int AudioProcessingImpl::ProcessCaptureStreamLocked() {
>+  HandleCaptureRuntimeSettings();
>+
>   // Ensure that not both the AEC and AECM are active at the same time.
>   // TODO(peah): Simplify once the public API Enable functions for these
>   // are moved to APM.
>@@ -1228,6 +1189,12 @@ int AudioProcessingImpl::ProcessCaptureStreamLocked() {
> 
>   AudioBuffer* capture_buffer = capture_.capture_audio.get();  // For brevity.
> 
>+  if (private_submodules_->pre_amplifier) {
>+    private_submodules_->pre_amplifier->ApplyGain(AudioFrameView<float>(
>+        capture_buffer->channels_f(), capture_buffer->num_channels(),
>+        capture_buffer->num_frames()));
>+  }
>+
>   capture_input_rms_.Analyze(rtc::ArrayView<const int16_t>(
>       capture_buffer->channels_const()[0],
>       capture_nonlocked_.capture_processing_format.num_frames()));
>@@ -1242,9 +1209,13 @@ int AudioProcessingImpl::ProcessCaptureStreamLocked() {
>   }
> 
>   if (private_submodules_->echo_controller) {
>-    // TODO(peah): Reactivate analogue AGC gain detection once the analogue AGC
>-    // issues have been addressed.
>-    capture_.echo_path_gain_change = false;
>+    // Detect and flag any change in the analog gain.
>+    int analog_mic_level = gain_control()->stream_analog_level();
>+    capture_.echo_path_gain_change =
>+        capture_.prev_analog_mic_level != analog_mic_level &&
>+        capture_.prev_analog_mic_level != -1;
>+    capture_.prev_analog_mic_level = analog_mic_level;
>+
>     private_submodules_->echo_controller->AnalyzeCapture(capture_buffer);
>   }
> 
>@@ -1269,13 +1240,6 @@ int AudioProcessingImpl::ProcessCaptureStreamLocked() {
>     capture_buffer->set_num_channels(1);
>   }
> 
>-  if (capture_nonlocked_.beamformer_enabled) {
>-    private_submodules_->beamformer->AnalyzeChunk(
>-        *capture_buffer->split_data_f());
>-    // Discards all channels by the leftmost one.
>-    capture_buffer->set_num_channels(1);
>-  }
>-
>   // TODO(peah): Move the AEC3 low-cut filter to this place.
>   if (private_submodules_->low_cut_filter &&
>       !private_submodules_->echo_controller) {
>@@ -1288,11 +1252,18 @@ int AudioProcessingImpl::ProcessCaptureStreamLocked() {
>   // Ensure that the stream delay was set before the call to the
>   // AEC ProcessCaptureAudio function.
>   if (public_submodules_->echo_cancellation->is_enabled() &&
>-      !was_stream_delay_set()) {
>+      !private_submodules_->echo_controller && !was_stream_delay_set()) {
>     return AudioProcessing::kStreamParameterNotSetError;
>   }
> 
>   if (private_submodules_->echo_controller) {
>+    data_dumper_->DumpRaw("stream_delay", stream_delay_ms());
>+
>+    if (was_stream_delay_set()) {
>+      private_submodules_->echo_controller->SetAudioBufferDelay(
>+          stream_delay_ms());
>+    }
>+
>     private_submodules_->echo_controller->ProcessCapture(
>         capture_buffer, capture_.echo_path_gain_change);
>   } else {
>@@ -1308,13 +1279,11 @@ int AudioProcessingImpl::ProcessCaptureStreamLocked() {
> #if WEBRTC_INTELLIGIBILITY_ENHANCER
>   if (capture_nonlocked_.intelligibility_enabled) {
>     RTC_DCHECK(public_submodules_->noise_suppression->is_enabled());
>-    int gain_db = public_submodules_->gain_control->is_enabled() ?
>-                  public_submodules_->gain_control->compression_gain_db() :
>-                  0;
>-    float gain = std::pow(10.f, gain_db / 20.f);
>-    gain *= capture_nonlocked_.level_controller_enabled ?
>-            private_submodules_->level_controller->GetLastGain() :
>-            1.f;
>+    const int gain_db =
>+        public_submodules_->gain_control->is_enabled()
>+            ? public_submodules_->gain_control->compression_gain_db()
>+            : 0;
>+    const float gain = DbToRatio(gain_db);
>     public_submodules_->intelligibility_enhancer->SetCaptureNoiseEstimate(
>         public_submodules_->noise_suppression->NoiseEstimate(), gain);
>   }
>@@ -1333,16 +1302,10 @@ int AudioProcessingImpl::ProcessCaptureStreamLocked() {
>         capture_buffer, stream_delay_ms()));
>   }
> 
>-  if (capture_nonlocked_.beamformer_enabled) {
>-    private_submodules_->beamformer->PostFilter(capture_buffer->split_data_f());
>-  }
>-
>   public_submodules_->voice_detection->ProcessCaptureAudio(capture_buffer);
> 
>   if (constants_.use_experimental_agc &&
>-      public_submodules_->gain_control->is_enabled() &&
>-      (!capture_nonlocked_.beamformer_enabled ||
>-       private_submodules_->beamformer->is_target_present())) {
>+      public_submodules_->gain_control->is_enabled()) {
>     private_submodules_->agc_manager->Process(
>         capture_buffer->split_bands_const(0)[kBand0To8kHz],
>         capture_buffer->num_frames_per_band(), capture_nonlocked_.split_rate);
>@@ -1381,13 +1344,11 @@ int AudioProcessingImpl::ProcessCaptureStreamLocked() {
>   }
> 
>   if (config_.gain_controller2.enabled) {
>+    private_submodules_->gain_controller2->NotifyAnalogLevel(
>+        gain_control()->stream_analog_level());
>     private_submodules_->gain_controller2->Process(capture_buffer);
>   }
> 
>-  if (capture_nonlocked_.level_controller_enabled) {
>-    private_submodules_->level_controller->Process(capture_buffer);
>-  }
>-
>   if (private_submodules_->capture_post_processor) {
>     private_submodules_->capture_post_processor->Process(capture_buffer);
>   }
>@@ -1465,8 +1426,8 @@ int AudioProcessingImpl::AnalyzeReverseStreamLocked(
>   processing_config.reverse_output_stream() = output_config;
> 
>   RETURN_ON_ERR(MaybeInitializeRender(processing_config));
>-  assert(input_config.num_frames() ==
>-         formats_.api_format.reverse_input_stream().num_frames());
>+  RTC_DCHECK_EQ(input_config.num_frames(),
>+                formats_.api_format.reverse_input_stream().num_frames());
> 
>   if (aec_dump_) {
>     const size_t channel_size =
>@@ -1474,7 +1435,7 @@ int AudioProcessingImpl::AnalyzeReverseStreamLocked(
>     const size_t num_channels =
>         formats_.api_format.reverse_input_stream().num_channels();
>     aec_dump_->WriteRenderStreamMessage(
>-        FloatAudioFrame(src, num_channels, channel_size));
>+        AudioFrameView<const float>(src, num_channels, channel_size));
>   }
>   render_.render_audio->CopyFrom(src,
>                                  formats_.api_format.reverse_input_stream());
>@@ -1530,12 +1491,14 @@ int AudioProcessingImpl::ProcessReverseStream(AudioFrame* frame) {
> int AudioProcessingImpl::ProcessRenderStreamLocked() {
>   AudioBuffer* render_buffer = render_.render_audio.get();  // For brevity.
> 
>-  QueueNonbandedRenderAudio(render_buffer);
>+  HandleRenderRuntimeSettings();
> 
>   if (private_submodules_->render_pre_processor) {
>     private_submodules_->render_pre_processor->Process(render_buffer);
>   }
> 
>+  QueueNonbandedRenderAudio(render_buffer);
>+
>   if (submodule_states_.RenderMultiBandSubModulesActive() &&
>       SampleRateSupportsMultiBand(
>           formats_.render_processing_format.sample_rate_hz())) {
>@@ -1553,7 +1516,7 @@ int AudioProcessingImpl::ProcessRenderStreamLocked() {
>     QueueBandedRenderAudio(render_buffer);
>   }
> 
>-  // TODO(peah): Perform the queueing Ãnside QueueRenderAudiuo().
>+  // TODO(peah): Perform the queuing inside QueueRenderAudiuo().
>   if (private_submodules_->echo_controller) {
>     private_submodules_->echo_controller->AnalyzeRender(render_buffer);
>   }
>@@ -1622,7 +1585,7 @@ void AudioProcessingImpl::AttachAecDump(std::unique_ptr<AecDump> aec_dump) {
>   // 'aec_dump' parameter, which is after locks are released.
>   aec_dump_.swap(aec_dump);
>   WriteAecDumpConfigMessage(true);
>-  aec_dump_->WriteInitMessage(ToStreamsConfig(formats_.api_format));
>+  aec_dump_->WriteInitMessage(formats_.api_format, rtc::TimeUTCMillis());
> }
> 
> void AudioProcessingImpl::DetachAecDump() {
>@@ -1637,6 +1600,17 @@ void AudioProcessingImpl::DetachAecDump() {
>   }
> }
> 
>+void AudioProcessingImpl::AttachPlayoutAudioGenerator(
>+    std::unique_ptr<AudioGenerator> audio_generator) {
>+  // TODO(bugs.webrtc.org/8882) Stub.
>+  // Reset internal audio generator with audio_generator.
>+}
>+
>+void AudioProcessingImpl::DetachPlayoutAudioGenerator() {
>+  // TODO(bugs.webrtc.org/8882) Stub.
>+  // Delete audio generator, if one is attached.
>+}
>+
> AudioProcessing::AudioProcessingStatistics::AudioProcessingStatistics() {
>   residual_echo_return_loss.Set(-100.0f, -100.0f, -100.0f, -100.0f);
>   echo_return_loss.Set(-100.0f, -100.0f, -100.0f, -100.0f);
>@@ -1713,15 +1687,15 @@ AudioProcessingStats AudioProcessingImpl::GetStatistics(
>                Error::kNoError) {
>       if (metrics.divergent_filter_fraction != -1.0f) {
>         stats.divergent_filter_fraction =
>-            rtc::Optional<double>(metrics.divergent_filter_fraction);
>+            absl::optional<double>(metrics.divergent_filter_fraction);
>       }
>       if (metrics.echo_return_loss.instant != -100) {
>         stats.echo_return_loss =
>-            rtc::Optional<double>(metrics.echo_return_loss.instant);
>+            absl::optional<double>(metrics.echo_return_loss.instant);
>       }
>       if (metrics.echo_return_loss_enhancement.instant != -100) {
>-        stats.echo_return_loss_enhancement =
>-            rtc::Optional<double>(metrics.echo_return_loss_enhancement.instant);
>+        stats.echo_return_loss_enhancement = absl::optional<double>(
>+            metrics.echo_return_loss_enhancement.instant);
>       }
>     }
>     if (config_.residual_echo_detector.enabled) {
>@@ -1738,10 +1712,10 @@ AudioProcessingStats AudioProcessingImpl::GetStatistics(
>             &delay_median, &delay_std, &fraction_poor_delays) ==
>         Error::kNoError) {
>       if (delay_median >= 0) {
>-        stats.delay_median_ms = rtc::Optional<int32_t>(delay_median);
>+        stats.delay_median_ms = absl::optional<int32_t>(delay_median);
>       }
>       if (delay_std >= 0) {
>-        stats.delay_standard_deviation_ms = rtc::Optional<int32_t>(delay_std);
>+        stats.delay_standard_deviation_ms = absl::optional<int32_t>(delay_std);
>       }
>     }
>   }
>@@ -1749,11 +1723,11 @@ AudioProcessingStats AudioProcessingImpl::GetStatistics(
> }
> 
> EchoCancellation* AudioProcessingImpl::echo_cancellation() const {
>-  return public_submodules_->echo_cancellation.get();
>+  return public_submodules_->echo_cancellation_proxy.get();
> }
> 
> EchoControlMobile* AudioProcessingImpl::echo_control_mobile() const {
>-  return public_submodules_->echo_control_mobile.get();
>+  return public_submodules_->echo_control_mobile_proxy.get();
> }
> 
> GainControl* AudioProcessingImpl::gain_control() const {
>@@ -1801,10 +1775,8 @@ bool AudioProcessingImpl::UpdateActiveSubmoduleStates() {
>       config_.residual_echo_detector.enabled,
>       public_submodules_->noise_suppression->is_enabled(),
>       capture_nonlocked_.intelligibility_enabled,
>-      capture_nonlocked_.beamformer_enabled,
>       public_submodules_->gain_control->is_enabled(),
>-      config_.gain_controller2.enabled,
>-      capture_nonlocked_.level_controller_enabled,
>+      config_.gain_controller2.enabled, config_.pre_amplifier.enabled,
>       capture_nonlocked_.echo_controller_enabled,
>       public_submodules_->voice_detection->is_enabled(),
>       public_submodules_->level_estimator->is_enabled(),
>@@ -1823,17 +1795,6 @@ void AudioProcessingImpl::InitializeTransient() {
>   }
> }
> 
>-void AudioProcessingImpl::InitializeBeamformer() {
>-  if (capture_nonlocked_.beamformer_enabled) {
>-    if (!private_submodules_->beamformer) {
>-      private_submodules_->beamformer.reset(new NonlinearBeamformer(
>-          capture_.array_geometry, 1u, capture_.target_direction));
>-    }
>-    private_submodules_->beamformer->Initialize(kChunkSizeMs,
>-                                                capture_nonlocked_.split_rate);
>-  }
>-}
>-
> void AudioProcessingImpl::InitializeIntelligibility() {
> #if WEBRTC_INTELLIGIBILITY_ENHANCER
>   if (capture_nonlocked_.intelligibility_enabled) {
>@@ -1870,14 +1831,20 @@ void AudioProcessingImpl::InitializeGainController2() {
>   }
> }
> 
>-void AudioProcessingImpl::InitializeLevelController() {
>-  private_submodules_->level_controller->Initialize(proc_sample_rate_hz());
>+void AudioProcessingImpl::InitializePreAmplifier() {
>+  if (config_.pre_amplifier.enabled) {
>+    private_submodules_->pre_amplifier.reset(
>+        new GainApplier(true, config_.pre_amplifier.fixed_gain_factor));
>+  } else {
>+    private_submodules_->pre_amplifier.reset();
>+  }
> }
> 
> void AudioProcessingImpl::InitializeResidualEchoDetector() {
>   RTC_DCHECK(private_submodules_->echo_detector);
>-  private_submodules_->echo_detector->Initialize(proc_sample_rate_hz(),
>-                                                 num_proc_channels());
>+  private_submodules_->echo_detector->Initialize(
>+      proc_sample_rate_hz(), 1,
>+      formats_.render_processing_format.sample_rate_hz(), 1);
> }
> 
> void AudioProcessingImpl::InitializePostProcessor() {
>@@ -1976,9 +1943,6 @@ void AudioProcessingImpl::WriteAecDumpConfigMessage(bool forced) {
>       public_submodules_->echo_cancellation->GetExperimentsDescription();
>   // TODO(peah): Add semicolon-separated concatenations of experiment
>   // descriptions for other submodules.
>-  if (capture_nonlocked_.level_controller_enabled) {
>-    experiments_description += "LevelController;";
>-  }
>   if (constants_.agc_clipped_level_min != kClippedLevelMin) {
>     experiments_description += "AgcClippingLevelExperiment;";
>   }
>@@ -2026,6 +1990,9 @@ void AudioProcessingImpl::WriteAecDumpConfigMessage(bool forced) {
>   apm_config.intelligibility_enhancer_enabled =
>       capture_nonlocked_.intelligibility_enabled;
>   apm_config.experiments_description = experiments_description;
>+  apm_config.pre_amplifier_enabled = config_.pre_amplifier.enabled;
>+  apm_config.pre_amplifier_fixed_gain_factor =
>+      config_.pre_amplifier.fixed_gain_factor;
> 
>   if (!forced && apm_config == apm_config_for_aec_dump_) {
>     return;
>@@ -2042,7 +2009,7 @@ void AudioProcessingImpl::RecordUnprocessedCaptureStream(
>   const size_t channel_size = formats_.api_format.input_stream().num_frames();
>   const size_t num_channels = formats_.api_format.input_stream().num_channels();
>   aec_dump_->AddCaptureStreamInput(
>-      FloatAudioFrame(src, num_channels, channel_size));
>+      AudioFrameView<const float>(src, num_channels, channel_size));
>   RecordAudioProcessingState();
> }
> 
>@@ -2062,8 +2029,8 @@ void AudioProcessingImpl::RecordProcessedCaptureStream(
>   const size_t channel_size = formats_.api_format.output_stream().num_frames();
>   const size_t num_channels =
>       formats_.api_format.output_stream().num_channels();
>-  aec_dump_->AddCaptureStreamOutput(
>-      FloatAudioFrame(processed_capture_stream, num_channels, channel_size));
>+  aec_dump_->AddCaptureStreamOutput(AudioFrameView<const float>(
>+      processed_capture_stream, num_channels, channel_size));
>   aec_dump_->WriteCaptureStreamMessage();
> }
> 
>@@ -2087,9 +2054,7 @@ void AudioProcessingImpl::RecordAudioProcessingState() {
> }
> 
> AudioProcessingImpl::ApmCaptureState::ApmCaptureState(
>-    bool transient_suppressor_enabled,
>-    const std::vector<Point>& array_geometry,
>-    SphericalPointf target_direction)
>+    bool transient_suppressor_enabled)
>     : aec_system_delay_jumps(-1),
>       delay_offset_ms(0),
>       was_stream_delay_set(false),
>@@ -2099,11 +2064,10 @@ AudioProcessingImpl::ApmCaptureState::ApmCaptureState(
>       output_will_be_muted(false),
>       key_pressed(false),
>       transient_suppressor_enabled(transient_suppressor_enabled),
>-      array_geometry(array_geometry),
>-      target_direction(target_direction),
>       capture_processing_format(kSampleRate16kHz),
>       split_rate(kSampleRate16kHz),
>-      echo_path_gain_change(false) {}
>+      echo_path_gain_change(false),
>+      prev_analog_mic_level(-1) {}
> 
> AudioProcessingImpl::ApmCaptureState::~ApmCaptureState() = default;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_processing_impl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_processing_impl.h
>index 8ece0297237..44d0d087a5b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_processing_impl.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_processing_impl.h
>@@ -24,29 +24,25 @@
> #include "rtc_base/function_view.h"
> #include "rtc_base/gtest_prod_util.h"
> #include "rtc_base/ignore_wundef.h"
>-#include "rtc_base/protobuf_utils.h"
> #include "rtc_base/swap_queue.h"
> #include "rtc_base/thread_annotations.h"
>-#include "system_wrappers/include/file_wrapper.h"
> 
> namespace webrtc {
> 
>+class ApmDataDumper;
> class AudioConverter;
>-class NonlinearBeamformer;
> 
> class AudioProcessingImpl : public AudioProcessing {
>  public:
>   // Methods forcing APM to run in a single-threaded manner.
>   // Acquires both the render and capture locks.
>   explicit AudioProcessingImpl(const webrtc::Config& config);
>-  // AudioProcessingImpl takes ownership of capture post processor and
>-  // beamformer.
>+  // AudioProcessingImpl takes ownership of capture post processor.
>   AudioProcessingImpl(const webrtc::Config& config,
>                       std::unique_ptr<CustomProcessing> capture_post_processor,
>                       std::unique_ptr<CustomProcessing> render_pre_processor,
>                       std::unique_ptr<EchoControlFactory> echo_control_factory,
>-                      std::unique_ptr<EchoDetector> echo_detector,
>-                      NonlinearBeamformer* beamformer);
>+                      rtc::scoped_refptr<EchoDetector> echo_detector);
>   ~AudioProcessingImpl() override;
>   int Initialize() override;
>   int Initialize(int capture_input_sample_rate_hz,
>@@ -61,6 +57,11 @@ class AudioProcessingImpl : public AudioProcessing {
>   void UpdateHistogramsOnCallEnd() override;
>   void AttachAecDump(std::unique_ptr<AecDump> aec_dump) override;
>   void DetachAecDump() override;
>+  void AttachPlayoutAudioGenerator(
>+      std::unique_ptr<AudioGenerator> audio_generator) override;
>+  void DetachPlayoutAudioGenerator() override;
>+
>+  void SetRuntimeSetting(RuntimeSetting setting) override;
> 
>   // Capture-side exclusive methods possibly running APM in a
>   // multi-threaded manner. Acquire the capture lock.
>@@ -139,9 +140,31 @@ class AudioProcessingImpl : public AudioProcessing {
>   FRIEND_TEST_ALL_PREFIXES(ApmConfiguration, DefaultBehavior);
>   FRIEND_TEST_ALL_PREFIXES(ApmConfiguration, ValidConfigBehavior);
>   FRIEND_TEST_ALL_PREFIXES(ApmConfiguration, InValidConfigBehavior);
>+
>+  // Class providing thread-safe message pipe functionality for
>+  // |runtime_settings_|.
>+  class RuntimeSettingEnqueuer {
>+   public:
>+    explicit RuntimeSettingEnqueuer(
>+        SwapQueue<RuntimeSetting>* runtime_settings);
>+    ~RuntimeSettingEnqueuer();
>+    void Enqueue(RuntimeSetting setting);
>+
>+   private:
>+    SwapQueue<RuntimeSetting>& runtime_settings_;
>+  };
>   struct ApmPublicSubmodules;
>   struct ApmPrivateSubmodules;
> 
>+  std::unique_ptr<ApmDataDumper> data_dumper_;
>+  static int instance_count_;
>+
>+  SwapQueue<RuntimeSetting> capture_runtime_settings_;
>+  SwapQueue<RuntimeSetting> render_runtime_settings_;
>+
>+  RuntimeSettingEnqueuer capture_runtime_settings_enqueuer_;
>+  RuntimeSettingEnqueuer render_runtime_settings_enqueuer_;
>+
>   // Submodule interface implementations.
>   std::unique_ptr<HighPassFilter> high_pass_filter_impl_;
> 
>@@ -159,10 +182,9 @@ class AudioProcessingImpl : public AudioProcessing {
>                 bool residual_echo_detector_enabled,
>                 bool noise_suppressor_enabled,
>                 bool intelligibility_enhancer_enabled,
>-                bool beamformer_enabled,
>                 bool adaptive_gain_controller_enabled,
>                 bool gain_controller2_enabled,
>-                bool level_controller_enabled,
>+                bool pre_amplifier_enabled,
>                 bool echo_controller_enabled,
>                 bool voice_activity_detector_enabled,
>                 bool level_estimator_enabled,
>@@ -183,10 +205,9 @@ class AudioProcessingImpl : public AudioProcessing {
>     bool residual_echo_detector_enabled_ = false;
>     bool noise_suppressor_enabled_ = false;
>     bool intelligibility_enhancer_enabled_ = false;
>-    bool beamformer_enabled_ = false;
>     bool adaptive_gain_controller_enabled_ = false;
>     bool gain_controller2_enabled_ = false;
>-    bool level_controller_enabled_ = false;
>+    bool pre_amplifier_enabled_ = false;
>     bool echo_controller_enabled_ = false;
>     bool level_estimator_enabled_ = false;
>     bool voice_activity_detector_enabled_ = false;
>@@ -220,21 +241,24 @@ class AudioProcessingImpl : public AudioProcessing {
>   // acquired.
>   void InitializeTransient()
>       RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
>-  void InitializeBeamformer()
>-      RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
>   void InitializeIntelligibility()
>       RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
>   int InitializeLocked(const ProcessingConfig& config)
>       RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
>-  void InitializeLevelController() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
>   void InitializeResidualEchoDetector()
>       RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
>   void InitializeLowCutFilter() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
>   void InitializeEchoController() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
>   void InitializeGainController2() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
>+  void InitializePreAmplifier() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
>   void InitializePostProcessor() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
>   void InitializePreProcessor() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
> 
>+  // Empties and handles the respective RuntimeSetting queues.
>+  void HandleCaptureRuntimeSettings()
>+      RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_capture_);
>+  void HandleRenderRuntimeSettings() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_render_);
>+
>   void EmptyQueuedRenderAudio();
>   void AllocateRenderQueue()
>       RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_render_, crit_capture_);
>@@ -328,20 +352,27 @@ class AudioProcessingImpl : public AudioProcessing {
>   const struct ApmConstants {
>     ApmConstants(int agc_startup_min_volume,
>                  int agc_clipped_level_min,
>-                 bool use_experimental_agc)
>+                 bool use_experimental_agc,
>+                 bool use_experimental_agc_agc2_level_estimation,
>+                 bool use_experimental_agc_agc2_digital_adaptive)
>         :  // Format of processing streams at input/output call sites.
>           agc_startup_min_volume(agc_startup_min_volume),
>           agc_clipped_level_min(agc_clipped_level_min),
>-          use_experimental_agc(use_experimental_agc) {}
>+          use_experimental_agc(use_experimental_agc),
>+          use_experimental_agc_agc2_level_estimation(
>+              use_experimental_agc_agc2_level_estimation),
>+          use_experimental_agc_agc2_digital_adaptive(
>+              use_experimental_agc_agc2_digital_adaptive) {}
>     int agc_startup_min_volume;
>     int agc_clipped_level_min;
>     bool use_experimental_agc;
>+    bool use_experimental_agc_agc2_level_estimation;
>+    bool use_experimental_agc_agc2_digital_adaptive;
>+
>   } constants_;
> 
>   struct ApmCaptureState {
>-    ApmCaptureState(bool transient_suppressor_enabled,
>-                    const std::vector<Point>& array_geometry,
>-                    SphericalPointf target_direction);
>+    ApmCaptureState(bool transient_suppressor_enabled);
>     ~ApmCaptureState();
>     int aec_system_delay_jumps;
>     int delay_offset_ms;
>@@ -352,8 +383,6 @@ class AudioProcessingImpl : public AudioProcessing {
>     bool output_will_be_muted;
>     bool key_pressed;
>     bool transient_suppressor_enabled;
>-    std::vector<Point> array_geometry;
>-    SphericalPointf target_direction;
>     std::unique_ptr<AudioBuffer> capture_audio;
>     // Only the rate and samples fields of capture_processing_format_ are used
>     // because the capture processing number of channels is mutable and is
>@@ -361,15 +390,14 @@ class AudioProcessingImpl : public AudioProcessing {
>     StreamConfig capture_processing_format;
>     int split_rate;
>     bool echo_path_gain_change;
>+    int prev_analog_mic_level;
>   } capture_ RTC_GUARDED_BY(crit_capture_);
> 
>   struct ApmCaptureNonLockedState {
>-    ApmCaptureNonLockedState(bool beamformer_enabled,
>-                             bool intelligibility_enabled)
>+    ApmCaptureNonLockedState(bool intelligibility_enabled)
>         : capture_processing_format(kSampleRate16kHz),
>           split_rate(kSampleRate16kHz),
>           stream_delay_ms(0),
>-          beamformer_enabled(beamformer_enabled),
>           intelligibility_enabled(intelligibility_enabled) {}
>     // Only the rate and samples fields of capture_processing_format_ are used
>     // because the forward processing number of channels is mutable and is
>@@ -377,9 +405,7 @@ class AudioProcessingImpl : public AudioProcessing {
>     StreamConfig capture_processing_format;
>     int split_rate;
>     int stream_delay_ms;
>-    bool beamformer_enabled;
>     bool intelligibility_enabled;
>-    bool level_controller_enabled = false;
>     bool echo_controller_enabled = false;
>   } capture_nonlocked_;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_processing_impl_locking_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_processing_impl_locking_unittest.cc
>index d4cff4582bf..b770fef18a1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_processing_impl_locking_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_processing_impl_locking_unittest.cc
>@@ -16,7 +16,6 @@
> 
> #include "api/array_view.h"
> #include "modules/audio_processing/test/test_utils.h"
>-#include "modules/include/module_common_types.h"
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/event.h"
> #include "rtc_base/platform_thread.h"
>@@ -147,8 +146,7 @@ struct TestConfig {
> 
>       // Create test config for the first processing API function set.
>       test_configs.push_back(test_config);
>-      test_config.render_api_function =
>-          RenderApiImpl::AnalyzeReverseStreamImpl;
>+      test_config.render_api_function = RenderApiImpl::AnalyzeReverseStreamImpl;
>       test_config.capture_api_function = CaptureApiImpl::ProcessStreamImpl3;
>       test_configs.push_back(test_config);
>     }
>@@ -483,8 +481,7 @@ void PopulateAudioFrame(AudioFrame* frame,
>     for (size_t k = 0; k < frame->samples_per_channel_; k++) {
>       // Store random 16 bit number between -(amplitude+1) and
>       // amplitude.
>-      frame_data[k * ch] =
>-          rand_gen->RandInt(2 * amplitude + 1) - amplitude - 1;
>+      frame_data[k * ch] = rand_gen->RandInt(2 * amplitude + 1) - amplitude - 1;
>     }
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_processing_impl_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_processing_impl_unittest.cc
>index e152befc5cf..8926f0223f2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_processing_impl_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_processing_impl_unittest.cc
>@@ -10,8 +10,10 @@
> 
> #include "modules/audio_processing/audio_processing_impl.h"
> 
>+#include "modules/audio_processing/include/audio_processing.h"
> #include "modules/audio_processing/test/test_utils.h"
>-#include "modules/include/module_common_types.h"
>+#include "rtc_base/refcountedobject.h"
>+#include "rtc_base/scoped_ref_ptr.h"
> #include "test/gmock.h"
> #include "test/gtest.h"
> 
>@@ -34,6 +36,83 @@ class MockInitialize : public AudioProcessingImpl {
>   MOCK_CONST_METHOD0(Release, rtc::RefCountReleaseStatus());
> };
> 
>+void InitializeAudioFrame(size_t input_rate,
>+                          size_t num_channels,
>+                          AudioFrame* frame) {
>+  const size_t samples_per_input_channel = rtc::CheckedDivExact(
>+      input_rate, static_cast<size_t>(rtc::CheckedDivExact(
>+                      1000, AudioProcessing::kChunkSizeMs)));
>+  RTC_DCHECK_LE(samples_per_input_channel * num_channels,
>+                AudioFrame::kMaxDataSizeSamples);
>+  frame->samples_per_channel_ = samples_per_input_channel;
>+  frame->sample_rate_hz_ = input_rate;
>+  frame->num_channels_ = num_channels;
>+}
>+
>+void FillFixedFrame(int16_t audio_level, AudioFrame* frame) {
>+  const size_t num_samples = frame->samples_per_channel_ * frame->num_channels_;
>+  for (size_t i = 0; i < num_samples; ++i) {
>+    frame->mutable_data()[i] = audio_level;
>+  }
>+}
>+
>+// Mocks EchoDetector and records the first samples of the last analyzed render
>+// stream frame. Used to check what data is read by an EchoDetector
>+// implementation injected into an APM.
>+class TestEchoDetector : public EchoDetector {
>+ public:
>+  TestEchoDetector()
>+      : analyze_render_audio_called_(false),
>+        last_render_audio_first_sample_(0.f) {}
>+  ~TestEchoDetector() override = default;
>+  void AnalyzeRenderAudio(rtc::ArrayView<const float> render_audio) override {
>+    last_render_audio_first_sample_ = render_audio[0];
>+    analyze_render_audio_called_ = true;
>+  }
>+  void AnalyzeCaptureAudio(rtc::ArrayView<const float> capture_audio) override {
>+  }
>+  void Initialize(int capture_sample_rate_hz,
>+                  int num_capture_channels,
>+                  int render_sample_rate_hz,
>+                  int num_render_channels) override {}
>+  EchoDetector::Metrics GetMetrics() const override { return {}; }
>+  // Returns true if AnalyzeRenderAudio() has been called at least once.
>+  bool analyze_render_audio_called() const {
>+    return analyze_render_audio_called_;
>+  }
>+  // Returns the first sample of the last analyzed render frame.
>+  float last_render_audio_first_sample() const {
>+    return last_render_audio_first_sample_;
>+  }
>+
>+ private:
>+  bool analyze_render_audio_called_;
>+  float last_render_audio_first_sample_;
>+};
>+
>+// Mocks CustomProcessing and applies ProcessSample() to all the samples.
>+// Meant to be injected into an APM to modify samples in a known and detectable
>+// way.
>+class TestRenderPreProcessor : public CustomProcessing {
>+ public:
>+  TestRenderPreProcessor() = default;
>+  ~TestRenderPreProcessor() = default;
>+  void Initialize(int sample_rate_hz, int num_channels) override {}
>+  void Process(AudioBuffer* audio) override {
>+    for (size_t k = 0; k < audio->num_channels(); ++k) {
>+      rtc::ArrayView<float> channel_view(audio->channels_f()[k],
>+                                         audio->num_frames());
>+      std::transform(channel_view.begin(), channel_view.end(),
>+                     channel_view.begin(), ProcessSample);
>+    }
>+  };
>+  std::string ToString() const override { return "TestRenderPreProcessor"; }
>+  void SetRuntimeSetting(AudioProcessing::RuntimeSetting setting) override {}
>+  // Modifies a sample. This member is used in Process() to modify a frame and
>+  // it is publicly visible to enable tests.
>+  static constexpr float ProcessSample(float x) { return 2.f * x; }
>+};
>+
> }  // namespace
> 
> TEST(AudioProcessingImplTest, AudioParameterChangeTriggersInit) {
>@@ -55,15 +134,13 @@ TEST(AudioProcessingImplTest, AudioParameterChangeTriggersInit) {
> 
>   // New sample rate. (Only impacts ProcessStream).
>   SetFrameSampleRate(&frame, 32000);
>-  EXPECT_CALL(mock, InitializeLocked())
>-      .Times(1);
>+  EXPECT_CALL(mock, InitializeLocked()).Times(1);
>   EXPECT_NOERR(mock.ProcessStream(&frame));
> 
>   // New number of channels.
>   // TODO(peah): Investigate why this causes 2 inits.
>   frame.num_channels_ = 2;
>-  EXPECT_CALL(mock, InitializeLocked())
>-      .Times(2);
>+  EXPECT_CALL(mock, InitializeLocked()).Times(2);
>   EXPECT_NOERR(mock.ProcessStream(&frame));
>   // ProcessStream sets num_channels_ == num_output_channels.
>   frame.num_channels_ = 2;
>@@ -75,4 +152,82 @@ TEST(AudioProcessingImplTest, AudioParameterChangeTriggersInit) {
>   EXPECT_NOERR(mock.ProcessReverseStream(&frame));
> }
> 
>+TEST(AudioProcessingImplTest, UpdateCapturePreGainRuntimeSetting) {
>+  std::unique_ptr<AudioProcessing> apm(AudioProcessingBuilder().Create());
>+  webrtc::AudioProcessing::Config apm_config;
>+  apm_config.pre_amplifier.enabled = true;
>+  apm_config.pre_amplifier.fixed_gain_factor = 1.f;
>+  apm->ApplyConfig(apm_config);
>+
>+  AudioFrame frame;
>+  constexpr int16_t kAudioLevel = 10000;
>+  constexpr size_t kSampleRateHz = 48000;
>+  constexpr size_t kNumChannels = 2;
>+  InitializeAudioFrame(kSampleRateHz, kNumChannels, &frame);
>+
>+  FillFixedFrame(kAudioLevel, &frame);
>+  apm->ProcessStream(&frame);
>+  EXPECT_EQ(frame.data()[100], kAudioLevel)
>+      << "With factor 1, frame shouldn't be modified.";
>+
>+  constexpr float kGainFactor = 2.f;
>+  apm->SetRuntimeSetting(
>+      AudioProcessing::RuntimeSetting::CreateCapturePreGain(kGainFactor));
>+
>+  // Process for two frames to have time to ramp up gain.
>+  for (int i = 0; i < 2; ++i) {
>+    FillFixedFrame(kAudioLevel, &frame);
>+    apm->ProcessStream(&frame);
>+  }
>+  EXPECT_EQ(frame.data()[100], kGainFactor * kAudioLevel)
>+      << "Frame should be amplified.";
>+}
>+
>+TEST(AudioProcessingImplTest, RenderPreProcessorBeforeEchoDetector) {
>+  // Make sure that signal changes caused by a render pre-processing sub-module
>+  // take place before any echo detector analysis.
>+  rtc::scoped_refptr<TestEchoDetector> test_echo_detector(
>+      new rtc::RefCountedObject<TestEchoDetector>());
>+  std::unique_ptr<CustomProcessing> test_render_pre_processor(
>+      new TestRenderPreProcessor());
>+  // Create APM injecting the test echo detector and render pre-processor.
>+  std::unique_ptr<AudioProcessing> apm(
>+      AudioProcessingBuilder()
>+          .SetEchoDetector(test_echo_detector)
>+          .SetRenderPreProcessing(std::move(test_render_pre_processor))
>+          .Create());
>+  webrtc::AudioProcessing::Config apm_config;
>+  apm_config.pre_amplifier.enabled = true;
>+  apm_config.residual_echo_detector.enabled = true;
>+  apm->ApplyConfig(apm_config);
>+
>+  constexpr int16_t kAudioLevel = 1000;
>+  constexpr int kSampleRateHz = 16000;
>+  constexpr size_t kNumChannels = 1;
>+  AudioFrame frame;
>+  InitializeAudioFrame(kSampleRateHz, kNumChannels, &frame);
>+
>+  constexpr float kAudioLevelFloat = static_cast<float>(kAudioLevel);
>+  constexpr float kExpectedPreprocessedAudioLevel =
>+      TestRenderPreProcessor::ProcessSample(kAudioLevelFloat);
>+  ASSERT_NE(kAudioLevelFloat, kExpectedPreprocessedAudioLevel);
>+
>+  // Analyze a render stream frame.
>+  FillFixedFrame(kAudioLevel, &frame);
>+  ASSERT_EQ(AudioProcessing::Error::kNoError,
>+            apm->ProcessReverseStream(&frame));
>+  // Trigger a call to in EchoDetector::AnalyzeRenderAudio() via
>+  // ProcessStream().
>+  FillFixedFrame(kAudioLevel, &frame);
>+  ASSERT_EQ(AudioProcessing::Error::kNoError, apm->ProcessStream(&frame));
>+  // Regardless of how the call to in EchoDetector::AnalyzeRenderAudio() is
>+  // triggered, the line below checks that the call has occurred. If not, the
>+  // APM implementation may have changed and this test might need to be adapted.
>+  ASSERT_TRUE(test_echo_detector->analyze_render_audio_called());
>+  // Check that the data read in EchoDetector::AnalyzeRenderAudio() is that
>+  // produced by the render pre-processor.
>+  EXPECT_EQ(kExpectedPreprocessedAudioLevel,
>+            test_echo_detector->last_render_audio_first_sample());
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_processing_performance_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_processing_performance_unittest.cc
>index 8dd81b2a581..56615cb532c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_processing_performance_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_processing_performance_unittest.cc
>@@ -17,7 +17,6 @@
> 
> #include "api/array_view.h"
> #include "modules/audio_processing/test/test_utils.h"
>-#include "modules/include/module_common_types.h"
> #include "rtc_base/atomicops.h"
> #include "rtc_base/numerics/safe_conversions.h"
> #include "rtc_base/platform_thread.h"
>@@ -50,7 +49,6 @@ enum class ProcessorType { kRender, kCapture };
> enum class SettingsType {
>   kDefaultApmDesktop,
>   kDefaultApmMobile,
>-  kDefaultApmDesktopAndBeamformer,
>   kDefaultApmDesktopAndIntelligibilityEnhancer,
>   kAllSubmodulesTurnedOff,
>   kDefaultApmDesktopWithoutDelayAgnostic,
>@@ -115,17 +113,6 @@ struct SimulationConfig {
>       }
>     }
> #endif
>-
>-    const SettingsType beamformer_settings[] = {
>-        SettingsType::kDefaultApmDesktopAndBeamformer};
>-
>-    const int beamformer_sample_rates[] = {8000, 16000, 32000, 48000};
>-
>-    for (auto sample_rate : beamformer_sample_rates) {
>-      for (auto settings : beamformer_settings) {
>-        simulation_configs.push_back(SimulationConfig(sample_rate, settings));
>-      }
>-    }
> #endif
> 
>     const SettingsType mobile_settings[] = {SettingsType::kDefaultApmMobile};
>@@ -150,9 +137,6 @@ struct SimulationConfig {
>       case SettingsType::kDefaultApmDesktop:
>         description = "DefaultApmDesktop";
>         break;
>-      case SettingsType::kDefaultApmDesktopAndBeamformer:
>-        description = "DefaultApmDesktopAndBeamformer";
>-        break;
>       case SettingsType::kDefaultApmDesktopAndIntelligibilityEnhancer:
>         description = "DefaultApmDesktopAndIntelligibilityEnhancer";
>         break;
>@@ -176,13 +160,9 @@ struct SimulationConfig {
> // Handler for the frame counters.
> class FrameCounters {
>  public:
>-  void IncreaseRenderCounter() {
>-    rtc::AtomicOps::Increment(&render_count_);
>-  }
>+  void IncreaseRenderCounter() { rtc::AtomicOps::Increment(&render_count_); }
> 
>-  void IncreaseCaptureCounter() {
>-    rtc::AtomicOps::Increment(&capture_count_);
>-  }
>+  void IncreaseCaptureCounter() { rtc::AtomicOps::Increment(&capture_count_); }
> 
>   int CaptureMinusRenderCounters() const {
>     // The return value will be approximate, but that's good enough since
>@@ -212,9 +192,7 @@ class FrameCounters {
> // Class that represents a flag that can only be raised.
> class LockedFlag {
>  public:
>-  bool get_flag() const {
>-    return rtc::AtomicOps::AcquireLoad(&flag_);
>-  }
>+  bool get_flag() const { return rtc::AtomicOps::AcquireLoad(&flag_); }
> 
>   void set_flag() {
>     if (!get_flag())  // read-only operation to avoid affecting the cache-line.
>@@ -264,9 +242,8 @@ class TimedThreadApiProcessor {
>         "_" + std::to_string(simulation_config_->sample_rate_hz) + "Hz";
> 
>     webrtc::test::PrintResultMeanAndError(
>-        "apm_timing", sample_rate_name, processor_name,
>-        GetDurationAverage(), GetDurationStandardDeviation(),
>-        "us", false);
>+        "apm_timing", sample_rate_name, processor_name, GetDurationAverage(),
>+        GetDurationStandardDeviation(), "us", false);
> 
>     if (kPrintAllDurations) {
>       webrtc::test::PrintResultList("apm_call_durations", sample_rate_name,
>@@ -544,18 +521,6 @@ class CallSimulator : public ::testing::TestWithParam<SimulationConfig> {
>       config->Set<DelayAgnostic>(new DelayAgnostic(true));
>     };
> 
>-    // Lambda function for adding beamformer settings to a config.
>-    auto add_beamformer_config = [](Config* config) {
>-      const size_t num_mics = 2;
>-      const std::vector<Point> array_geometry =
>-          ParseArrayGeometry("0 0 0 0.05 0 0", num_mics);
>-      RTC_CHECK_EQ(array_geometry.size(), num_mics);
>-
>-      config->Set<Beamforming>(
>-          new Beamforming(true, array_geometry,
>-                          SphericalPointf(DegreesToRadians(90), 0.f, 1.f)));
>-    };
>-
>     int num_capture_channels = 1;
>     switch (simulation_config_.simulation_settings) {
>       case SettingsType::kDefaultApmMobile: {
>@@ -573,17 +538,6 @@ class CallSimulator : public ::testing::TestWithParam<SimulationConfig> {
>         apm_->SetExtraOptions(config);
>         break;
>       }
>-      case SettingsType::kDefaultApmDesktopAndBeamformer: {
>-        Config config;
>-        add_beamformer_config(&config);
>-        add_default_desktop_config(&config);
>-        apm_.reset(AudioProcessingBuilder().Create(config));
>-        ASSERT_TRUE(!!apm_);
>-        set_default_desktop_apm_runtime_settings(apm_.get());
>-        apm_->SetExtraOptions(config);
>-        num_capture_channels = 2;
>-        break;
>-      }
>       case SettingsType::kDefaultApmDesktopAndIntelligibilityEnhancer: {
>         Config config;
>         config.Set<Intelligibility>(new Intelligibility(true));
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_processing_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_processing_unittest.cc
>index a75037b2478..346c2a2c40f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_processing_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/audio_processing_unittest.cc
>@@ -21,22 +21,22 @@
> #include "common_audio/signal_processing/include/signal_processing_library.h"
> #include "modules/audio_processing/aec_dump/aec_dump_factory.h"
> #include "modules/audio_processing/audio_processing_impl.h"
>-#include "modules/audio_processing/beamformer/mock_nonlinear_beamformer.h"
> #include "modules/audio_processing/common.h"
> #include "modules/audio_processing/include/audio_processing.h"
> #include "modules/audio_processing/include/mock_audio_processing.h"
>-#include "modules/audio_processing/level_controller/level_controller_constants.h"
> #include "modules/audio_processing/test/protobuf_utils.h"
> #include "modules/audio_processing/test/test_utils.h"
>-#include "modules/include/module_common_types.h"
> #include "rtc_base/arraysize.h"
> #include "rtc_base/checks.h"
>+#include "rtc_base/fakeclock.h"
> #include "rtc_base/gtest_prod_util.h"
> #include "rtc_base/ignore_wundef.h"
> #include "rtc_base/numerics/safe_conversions.h"
> #include "rtc_base/numerics/safe_minmax.h"
> #include "rtc_base/protobuf_utils.h"
> #include "rtc_base/refcountedobject.h"
>+#include "rtc_base/swap_queue.h"
>+#include "rtc_base/system/arch.h"
> #include "rtc_base/task_queue.h"
> #include "rtc_base/thread.h"
> #include "system_wrappers/include/event_wrapper.h"
>@@ -187,6 +187,8 @@ void EnableAllAPComponents(AudioProcessing* ap) {
>   EXPECT_NOERR(ap->echo_cancellation()->enable_drift_compensation(true));
>   EXPECT_NOERR(ap->echo_cancellation()->enable_metrics(true));
>   EXPECT_NOERR(ap->echo_cancellation()->enable_delay_logging(true));
>+  EXPECT_NOERR(ap->echo_cancellation()->set_suppression_level(
>+      EchoCancellation::SuppressionLevel::kModerateSuppression));
>   EXPECT_NOERR(ap->echo_cancellation()->Enable(true));
> 
>   EXPECT_NOERR(ap->gain_control()->set_mode(GainControl::kAdaptiveAnalog));
>@@ -1301,95 +1303,6 @@ TEST_F(ApmTest, ManualVolumeChangeIsPossible) {
>   }
> }
> 
>-#if !defined(WEBRTC_ANDROID) && !defined(WEBRTC_IOS)
>-TEST_F(ApmTest, AgcOnlyAdaptsWhenTargetSignalIsPresent) {
>-  const int kSampleRateHz = 16000;
>-  const size_t kSamplesPerChannel =
>-      static_cast<size_t>(AudioProcessing::kChunkSizeMs * kSampleRateHz / 1000);
>-  const size_t kNumInputChannels = 2;
>-  const size_t kNumOutputChannels = 1;
>-  const size_t kNumChunks = 700;
>-  const float kScaleFactor = 0.25f;
>-  Config config;
>-  std::vector<webrtc::Point> geometry;
>-  geometry.push_back(webrtc::Point(0.f, 0.f, 0.f));
>-  geometry.push_back(webrtc::Point(0.05f, 0.f, 0.f));
>-  config.Set<Beamforming>(new Beamforming(true, geometry));
>-  testing::NiceMock<MockNonlinearBeamformer>* beamformer =
>-      new testing::NiceMock<MockNonlinearBeamformer>(geometry, 1u);
>-  std::unique_ptr<AudioProcessing> apm(
>-      AudioProcessingBuilder()
>-          .SetNonlinearBeamformer(
>-              std::unique_ptr<webrtc::NonlinearBeamformer>(beamformer))
>-          .Create(config));
>-  EXPECT_EQ(kNoErr, apm->gain_control()->Enable(true));
>-  ChannelBuffer<float> src_buf(kSamplesPerChannel, kNumInputChannels);
>-  ChannelBuffer<float> dest_buf(kSamplesPerChannel, kNumOutputChannels);
>-  const size_t max_length = kSamplesPerChannel * std::max(kNumInputChannels,
>-                                                          kNumOutputChannels);
>-  std::unique_ptr<int16_t[]> int_data(new int16_t[max_length]);
>-  std::unique_ptr<float[]> float_data(new float[max_length]);
>-  std::string filename = ResourceFilePath("far", kSampleRateHz);
>-  FILE* far_file = fopen(filename.c_str(), "rb");
>-  ASSERT_TRUE(far_file != NULL) << "Could not open file " << filename << "\n";
>-  const int kDefaultVolume = apm->gain_control()->stream_analog_level();
>-  const int kDefaultCompressionGain =
>-      apm->gain_control()->compression_gain_db();
>-  bool is_target = false;
>-  EXPECT_CALL(*beamformer, is_target_present())
>-      .WillRepeatedly(testing::ReturnPointee(&is_target));
>-  for (size_t i = 0; i < kNumChunks; ++i) {
>-    ASSERT_TRUE(ReadChunk(far_file,
>-                          int_data.get(),
>-                          float_data.get(),
>-                          &src_buf));
>-    for (size_t j = 0; j < kNumInputChannels; ++j) {
>-      for (size_t k = 0; k < kSamplesPerChannel; ++k) {
>-        src_buf.channels()[j][k] *= kScaleFactor;
>-      }
>-    }
>-    EXPECT_EQ(kNoErr,
>-              apm->ProcessStream(src_buf.channels(),
>-                                 src_buf.num_frames(),
>-                                 kSampleRateHz,
>-                                 LayoutFromChannels(src_buf.num_channels()),
>-                                 kSampleRateHz,
>-                                 LayoutFromChannels(dest_buf.num_channels()),
>-                                 dest_buf.channels()));
>-  }
>-  EXPECT_EQ(kDefaultVolume,
>-            apm->gain_control()->stream_analog_level());
>-  EXPECT_EQ(kDefaultCompressionGain,
>-            apm->gain_control()->compression_gain_db());
>-  rewind(far_file);
>-  is_target = true;
>-  for (size_t i = 0; i < kNumChunks; ++i) {
>-    ASSERT_TRUE(ReadChunk(far_file,
>-                          int_data.get(),
>-                          float_data.get(),
>-                          &src_buf));
>-    for (size_t j = 0; j < kNumInputChannels; ++j) {
>-      for (size_t k = 0; k < kSamplesPerChannel; ++k) {
>-        src_buf.channels()[j][k] *= kScaleFactor;
>-      }
>-    }
>-    EXPECT_EQ(kNoErr,
>-              apm->ProcessStream(src_buf.channels(),
>-                                 src_buf.num_frames(),
>-                                 kSampleRateHz,
>-                                 LayoutFromChannels(src_buf.num_channels()),
>-                                 kSampleRateHz,
>-                                 LayoutFromChannels(dest_buf.num_channels()),
>-                                 dest_buf.channels()));
>-  }
>-  EXPECT_LT(kDefaultVolume,
>-            apm->gain_control()->stream_analog_level());
>-  EXPECT_LT(kDefaultCompressionGain,
>-            apm->gain_control()->compression_gain_db());
>-  ASSERT_EQ(0, fclose(far_file));
>-}
>-#endif
>-
> TEST_F(ApmTest, NoiseSuppression) {
>   // Test valid suppression levels.
>   NoiseSuppression::Level level[] = {
>@@ -1820,6 +1733,7 @@ void ApmTest::ProcessDebugDump(const std::string& in_filename,
> }
> 
> void ApmTest::VerifyDebugDumpTest(Format format) {
>+  rtc::ScopedFakeClock fake_clock;
>   const std::string in_filename = test::ResourcePath("ref03", "aecdump");
>   std::string format_string;
>   switch (format) {
>@@ -2273,9 +2187,9 @@ TEST_F(ApmTest, Process) {
>       // or generate a separate android reference.
> #if defined(WEBRTC_ANDROID)
>       const int kHasVoiceCountOffset = 3;
>-      const int kHasVoiceCountNear = 4;
>+      const int kHasVoiceCountNear = 8;
>       const int kMaxOutputAverageOffset = 9;
>-      const int kMaxOutputAverageNear = 9;
>+      const int kMaxOutputAverageNear = 26;
> #else
>       const int kHasVoiceCountOffset = 0;
>       const int kHasVoiceCountNear = kIntNear;
>@@ -2821,101 +2735,36 @@ INSTANTIATE_TEST_CASE_P(
> 
> }  // namespace
> 
>-TEST(ApmConfiguration, DefaultBehavior) {
>-  // Verify that the level controller is default off, it can be activated using
>-  // the config, and that the default initial level is maintained after the
>-  // config has been applied.
>-  std::unique_ptr<AudioProcessingImpl> apm(
>-      new rtc::RefCountedObject<AudioProcessingImpl>(webrtc::Config()));
>-  AudioProcessing::Config config;
>-  EXPECT_FALSE(apm->config_.level_controller.enabled);
>-  // TODO(peah): Add test for the existence of the level controller object once
>-  // that is created only when that is specified in the config.
>-  // TODO(peah): Remove the testing for
>-  // apm->capture_nonlocked_.level_controller_enabled once the value in config_
>-  // is instead used to activate the level controller.
>-  EXPECT_FALSE(apm->capture_nonlocked_.level_controller_enabled);
>-  EXPECT_NEAR(kTargetLcPeakLeveldBFS,
>-              apm->config_.level_controller.initial_peak_level_dbfs,
>-              std::numeric_limits<float>::epsilon());
>-  config.level_controller.enabled = true;
>-  apm->ApplyConfig(config);
>-  EXPECT_TRUE(apm->config_.level_controller.enabled);
>-  // TODO(peah): Add test for the existence of the level controller object once
>-  // that is created only when the that is specified in the config.
>-  // TODO(peah): Remove the testing for
>-  // apm->capture_nonlocked_.level_controller_enabled once the value in config_
>-  // is instead used to activate the level controller.
>-  EXPECT_TRUE(apm->capture_nonlocked_.level_controller_enabled);
>-  EXPECT_NEAR(kTargetLcPeakLeveldBFS,
>-              apm->config_.level_controller.initial_peak_level_dbfs,
>-              std::numeric_limits<float>::epsilon());
>-}
>-
>-TEST(ApmConfiguration, ValidConfigBehavior) {
>-  // Verify that the initial level can be specified and is retained after the
>-  // config has been applied.
>-  std::unique_ptr<AudioProcessingImpl> apm(
>-      new rtc::RefCountedObject<AudioProcessingImpl>(webrtc::Config()));
>-  AudioProcessing::Config config;
>-  config.level_controller.initial_peak_level_dbfs = -50.f;
>-  apm->ApplyConfig(config);
>-  EXPECT_FALSE(apm->config_.level_controller.enabled);
>-  // TODO(peah): Add test for the existence of the level controller object once
>-  // that is created only when the that is specified in the config.
>-  // TODO(peah): Remove the testing for
>-  // apm->capture_nonlocked_.level_controller_enabled once the value in config_
>-  // is instead used to activate the level controller.
>-  EXPECT_FALSE(apm->capture_nonlocked_.level_controller_enabled);
>-  EXPECT_NEAR(-50.f, apm->config_.level_controller.initial_peak_level_dbfs,
>-              std::numeric_limits<float>::epsilon());
>-}
>-
>-TEST(ApmConfiguration, InValidConfigBehavior) {
>-  // Verify that the config is properly reset when nonproper values are applied
>-  // for the initial level.
>-
>-  // Verify that the config is properly reset when the specified initial peak
>-  // level is too low.
>-  std::unique_ptr<AudioProcessingImpl> apm(
>-      new rtc::RefCountedObject<AudioProcessingImpl>(webrtc::Config()));
>-  AudioProcessing::Config config;
>-  config.level_controller.enabled = true;
>-  config.level_controller.initial_peak_level_dbfs = -101.f;
>-  apm->ApplyConfig(config);
>-  EXPECT_FALSE(apm->config_.level_controller.enabled);
>-  // TODO(peah): Add test for the existence of the level controller object once
>-  // that is created only when the that is specified in the config.
>-  // TODO(peah): Remove the testing for
>-  // apm->capture_nonlocked_.level_controller_enabled once the value in config_
>-  // is instead used to activate the level controller.
>-  EXPECT_FALSE(apm->capture_nonlocked_.level_controller_enabled);
>-  EXPECT_NEAR(kTargetLcPeakLeveldBFS,
>-              apm->config_.level_controller.initial_peak_level_dbfs,
>-              std::numeric_limits<float>::epsilon());
>-
>-  // Verify that the config is properly reset when the specified initial peak
>-  // level is too high.
>-  apm.reset(new rtc::RefCountedObject<AudioProcessingImpl>(webrtc::Config()));
>-  config = AudioProcessing::Config();
>-  config.level_controller.enabled = true;
>-  config.level_controller.initial_peak_level_dbfs = 1.f;
>-  apm->ApplyConfig(config);
>-  EXPECT_FALSE(apm->config_.level_controller.enabled);
>-  // TODO(peah): Add test for the existence of the level controller object once
>-  // that is created only when that is specified in the config.
>-  // TODO(peah): Remove the testing for
>-  // apm->capture_nonlocked_.level_controller_enabled once the value in config_
>-  // is instead used to activate the level controller.
>-  EXPECT_FALSE(apm->capture_nonlocked_.level_controller_enabled);
>-  EXPECT_NEAR(kTargetLcPeakLeveldBFS,
>-              apm->config_.level_controller.initial_peak_level_dbfs,
>-              std::numeric_limits<float>::epsilon());
>+TEST(RuntimeSettingTest, TestDefaultCtor) {
>+  auto s = AudioProcessing::RuntimeSetting();
>+  EXPECT_EQ(AudioProcessing::RuntimeSetting::Type::kNotSpecified, s.type());
>+}
>+
>+TEST(RuntimeSettingTest, TestCapturePreGain) {
>+  using Type = AudioProcessing::RuntimeSetting::Type;
>+  {
>+    auto s = AudioProcessing::RuntimeSetting::CreateCapturePreGain(1.25f);
>+    EXPECT_EQ(Type::kCapturePreGain, s.type());
>+    float v;
>+    s.GetFloat(&v);
>+    EXPECT_EQ(1.25f, v);
>+  }
>+
>+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
>+  EXPECT_DEATH(AudioProcessing::RuntimeSetting::CreateCapturePreGain(0.1f), "");
>+#endif
>+}
>+
>+TEST(RuntimeSettingTest, TestUsageWithSwapQueue) {
>+  SwapQueue<AudioProcessing::RuntimeSetting> q(1);
>+  auto s = AudioProcessing::RuntimeSetting();
>+  ASSERT_TRUE(q.Insert(&s));
>+  ASSERT_TRUE(q.Remove(&s));
>+  EXPECT_EQ(AudioProcessing::RuntimeSetting::Type::kNotSpecified, s.type());
> }
> 
> TEST(ApmConfiguration, EnablePostProcessing) {
>   // Verify that apm uses a capture post processing module if one is provided.
>-  webrtc::Config webrtc_config;
>   auto mock_post_processor_ptr =
>       new testing::NiceMock<test::MockCustomProcessing>();
>   auto mock_post_processor =
>@@ -2923,7 +2772,7 @@ TEST(ApmConfiguration, EnablePostProcessing) {
>   rtc::scoped_refptr<AudioProcessing> apm =
>       AudioProcessingBuilder()
>           .SetCapturePostProcessing(std::move(mock_post_processor))
>-          .Create(webrtc_config);
>+          .Create();
> 
>   AudioFrame audio;
>   audio.num_channels_ = 1;
>@@ -2935,7 +2784,6 @@ TEST(ApmConfiguration, EnablePostProcessing) {
> 
> TEST(ApmConfiguration, EnablePreProcessing) {
>   // Verify that apm uses a capture post processing module if one is provided.
>-  webrtc::Config webrtc_config;
>   auto mock_pre_processor_ptr =
>       new testing::NiceMock<test::MockCustomProcessing>();
>   auto mock_pre_processor =
>@@ -2943,7 +2791,7 @@ TEST(ApmConfiguration, EnablePreProcessing) {
>   rtc::scoped_refptr<AudioProcessing> apm =
>       AudioProcessingBuilder()
>           .SetRenderPreProcessing(std::move(mock_pre_processor))
>-          .Create(webrtc_config);
>+          .Create();
> 
>   AudioFrame audio;
>   audio.num_channels_ = 1;
>@@ -2953,6 +2801,28 @@ TEST(ApmConfiguration, EnablePreProcessing) {
>   apm->ProcessReverseStream(&audio);
> }
> 
>+TEST(ApmConfiguration, PreProcessingReceivesRuntimeSettings) {
>+  auto mock_pre_processor_ptr =
>+      new testing::NiceMock<test::MockCustomProcessing>();
>+  auto mock_pre_processor =
>+      std::unique_ptr<CustomProcessing>(mock_pre_processor_ptr);
>+  rtc::scoped_refptr<AudioProcessing> apm =
>+      AudioProcessingBuilder()
>+          .SetRenderPreProcessing(std::move(mock_pre_processor))
>+          .Create();
>+  apm->SetRuntimeSetting(
>+      AudioProcessing::RuntimeSetting::CreateCustomRenderSetting(0));
>+
>+  // RuntimeSettings forwarded during 'Process*Stream' calls.
>+  // Therefore we have to make one such call.
>+  AudioFrame audio;
>+  audio.num_channels_ = 1;
>+  SetFrameSampleRate(&audio, AudioProcessing::NativeRate::kSampleRate16kHz);
>+
>+  EXPECT_CALL(*mock_pre_processor_ptr, SetRuntimeSetting(testing::_)).Times(1);
>+  apm->ProcessReverseStream(&audio);
>+}
>+
> class MyEchoControlFactory : public EchoControlFactory {
>  public:
>   std::unique_ptr<EchoControl> Create(int sample_rate_hz) {
>@@ -3005,10 +2875,8 @@ std::unique_ptr<AudioProcessing> CreateApm(bool use_AEC2) {
>   // Disable all components except for an AEC and the residual echo detector.
>   AudioProcessing::Config config;
>   config.residual_echo_detector.enabled = true;
>-  config.echo_canceller3.enabled = false;
>   config.high_pass_filter.enabled = false;
>   config.gain_controller2.enabled = false;
>-  config.level_controller.enabled = false;
>   apm->ApplyConfig(config);
>   EXPECT_EQ(apm->gain_control()->Enable(false), 0);
>   EXPECT_EQ(apm->level_estimator()->Enable(false), 0);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/array_util.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/array_util.cc
>deleted file mode 100644
>index e8535591403..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/array_util.cc
>+++ /dev/null
>@@ -1,119 +0,0 @@
>-/*
>- *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "modules/audio_processing/beamformer/array_util.h"
>-
>-#include <algorithm>
>-#include <limits>
>-
>-#include "rtc_base/checks.h"
>-
>-namespace webrtc {
>-namespace {
>-
>-const float kMaxDotProduct = 1e-6f;
>-
>-}  // namespace
>-
>-float GetMinimumSpacing(const std::vector<Point>& array_geometry) {
>-  RTC_CHECK_GT(array_geometry.size(), 1);
>-  float mic_spacing = std::numeric_limits<float>::max();
>-  for (size_t i = 0; i < (array_geometry.size() - 1); ++i) {
>-    for (size_t j = i + 1; j < array_geometry.size(); ++j) {
>-      mic_spacing =
>-          std::min(mic_spacing, Distance(array_geometry[i], array_geometry[j]));
>-    }
>-  }
>-  return mic_spacing;
>-}
>-
>-Point PairDirection(const Point& a, const Point& b) {
>-  return {b.x() - a.x(), b.y() - a.y(), b.z() - a.z()};
>-}
>-
>-float DotProduct(const Point& a, const Point& b) {
>-  return a.x() * b.x() + a.y() * b.y() + a.z() * b.z();
>-}
>-
>-Point CrossProduct(const Point& a, const Point& b) {
>-  return {a.y() * b.z() - a.z() * b.y(), a.z() * b.x() - a.x() * b.z(),
>-          a.x() * b.y() - a.y() * b.x()};
>-}
>-
>-bool AreParallel(const Point& a, const Point& b) {
>-  Point cross_product = CrossProduct(a, b);
>-  return DotProduct(cross_product, cross_product) < kMaxDotProduct;
>-}
>-
>-bool ArePerpendicular(const Point& a, const Point& b) {
>-  return std::abs(DotProduct(a, b)) < kMaxDotProduct;
>-}
>-
>-rtc::Optional<Point> GetDirectionIfLinear(
>-    const std::vector<Point>& array_geometry) {
>-  RTC_DCHECK_GT(array_geometry.size(), 1);
>-  const Point first_pair_direction =
>-      PairDirection(array_geometry[0], array_geometry[1]);
>-  for (size_t i = 2u; i < array_geometry.size(); ++i) {
>-    const Point pair_direction =
>-        PairDirection(array_geometry[i - 1], array_geometry[i]);
>-    if (!AreParallel(first_pair_direction, pair_direction)) {
>-      return rtc::nullopt;
>-    }
>-  }
>-  return first_pair_direction;
>-}
>-
>-rtc::Optional<Point> GetNormalIfPlanar(
>-    const std::vector<Point>& array_geometry) {
>-  RTC_DCHECK_GT(array_geometry.size(), 1);
>-  const Point first_pair_direction =
>-      PairDirection(array_geometry[0], array_geometry[1]);
>-  Point pair_direction(0.f, 0.f, 0.f);
>-  size_t i = 2u;
>-  bool is_linear = true;
>-  for (; i < array_geometry.size() && is_linear; ++i) {
>-    pair_direction = PairDirection(array_geometry[i - 1], array_geometry[i]);
>-    if (!AreParallel(first_pair_direction, pair_direction)) {
>-      is_linear = false;
>-    }
>-  }
>-  if (is_linear) {
>-    return rtc::nullopt;
>-  }
>-  const Point normal_direction =
>-      CrossProduct(first_pair_direction, pair_direction);
>-  for (; i < array_geometry.size(); ++i) {
>-    pair_direction = PairDirection(array_geometry[i - 1], array_geometry[i]);
>-    if (!ArePerpendicular(normal_direction, pair_direction)) {
>-      return rtc::nullopt;
>-    }
>-  }
>-  return normal_direction;
>-}
>-
>-rtc::Optional<Point> GetArrayNormalIfExists(
>-    const std::vector<Point>& array_geometry) {
>-  const rtc::Optional<Point> direction = GetDirectionIfLinear(array_geometry);
>-  if (direction) {
>-    return Point(direction->y(), -direction->x(), 0.f);
>-  }
>-  const rtc::Optional<Point> normal = GetNormalIfPlanar(array_geometry);
>-  if (normal && normal->z() < kMaxDotProduct) {
>-    return normal;
>-  }
>-  return rtc::nullopt;
>-}
>-
>-Point AzimuthToPoint(float azimuth) {
>-  return Point(std::cos(azimuth), std::sin(azimuth), 0.f);
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/array_util.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/array_util.h
>deleted file mode 100644
>index a7271fcbacc..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/array_util.h
>+++ /dev/null
>@@ -1,117 +0,0 @@
>-/*
>- *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MODULES_AUDIO_PROCESSING_BEAMFORMER_ARRAY_UTIL_H_
>-#define MODULES_AUDIO_PROCESSING_BEAMFORMER_ARRAY_UTIL_H_
>-
>-#include <cmath>
>-#include <vector>
>-
>-#include "api/optional.h"
>-
>-namespace webrtc {
>-
>-// Coordinates in meters. The convention used is:
>-// x: the horizontal dimension, with positive to the right from the camera's
>-//    perspective.
>-// y: the depth dimension, with positive forward from the camera's
>-//    perspective.
>-// z: the vertical dimension, with positive upwards.
>-template<typename T>
>-struct CartesianPoint {
>-  CartesianPoint() {
>-    c[0] = 0;
>-    c[1] = 0;
>-    c[2] = 0;
>-  }
>-  CartesianPoint(T x, T y, T z) {
>-    c[0] = x;
>-    c[1] = y;
>-    c[2] = z;
>-  }
>-  T x() const { return c[0]; }
>-  T y() const { return c[1]; }
>-  T z() const { return c[2]; }
>-  T c[3];
>-};
>-
>-using Point = CartesianPoint<float>;
>-
>-// Calculates the direction from a to b.
>-Point PairDirection(const Point& a, const Point& b);
>-
>-float DotProduct(const Point& a, const Point& b);
>-Point CrossProduct(const Point& a, const Point& b);
>-
>-bool AreParallel(const Point& a, const Point& b);
>-bool ArePerpendicular(const Point& a, const Point& b);
>-
>-// Returns the minimum distance between any two Points in the given
>-// |array_geometry|.
>-float GetMinimumSpacing(const std::vector<Point>& array_geometry);
>-
>-// If the given array geometry is linear it returns the direction without
>-// normalizing.
>-rtc::Optional<Point> GetDirectionIfLinear(
>-    const std::vector<Point>& array_geometry);
>-
>-// If the given array geometry is planar it returns the normal without
>-// normalizing.
>-rtc::Optional<Point> GetNormalIfPlanar(
>-    const std::vector<Point>& array_geometry);
>-
>-// Returns the normal of an array if it has one and it is in the xy-plane.
>-rtc::Optional<Point> GetArrayNormalIfExists(
>-    const std::vector<Point>& array_geometry);
>-
>-// The resulting Point will be in the xy-plane.
>-Point AzimuthToPoint(float azimuth);
>-
>-template<typename T>
>-float Distance(CartesianPoint<T> a, CartesianPoint<T> b) {
>-  return std::sqrt((a.x() - b.x()) * (a.x() - b.x()) +
>-                   (a.y() - b.y()) * (a.y() - b.y()) +
>-                   (a.z() - b.z()) * (a.z() - b.z()));
>-}
>-
>-// The convention used:
>-// azimuth: zero is to the right from the camera's perspective, with positive
>-//          angles in radians counter-clockwise.
>-// elevation: zero is horizontal, with positive angles in radians upwards.
>-// radius: distance from the camera in meters.
>-template <typename T>
>-struct SphericalPoint {
>-  SphericalPoint(T azimuth, T elevation, T radius) {
>-    s[0] = azimuth;
>-    s[1] = elevation;
>-    s[2] = radius;
>-  }
>-  T azimuth() const { return s[0]; }
>-  T elevation() const { return s[1]; }
>-  T distance() const { return s[2]; }
>-  T s[3];
>-};
>-
>-using SphericalPointf = SphericalPoint<float>;
>-
>-// Helper functions to transform degrees to radians and the inverse.
>-template <typename T>
>-constexpr T DegreesToRadians(T angle_degrees) {
>-  return M_PI * angle_degrees / 180;
>-}
>-
>-template <typename T>
>-T RadiansToDegrees(T angle_radians) {
>-  return 180 * angle_radians / M_PI;
>-}
>-
>-}  // namespace webrtc
>-
>-#endif  // MODULES_AUDIO_PROCESSING_BEAMFORMER_ARRAY_UTIL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/array_util_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/array_util_unittest.cc
>deleted file mode 100644
>index a5c075ab05f..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/array_util_unittest.cc
>+++ /dev/null
>@@ -1,185 +0,0 @@
>-/*
>- *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-// MSVC++ requires this to be set before any other includes to get M_PI.
>-#define _USE_MATH_DEFINES
>-
>-#include "modules/audio_processing/beamformer/array_util.h"
>-
>-#include <math.h>
>-#include <vector>
>-
>-#include "test/gtest.h"
>-
>-namespace webrtc {
>-
>-bool operator==(const Point& lhs, const Point& rhs) {
>-  return lhs.x() == rhs.x() && lhs.y() == rhs.y() && lhs.z() == rhs.z();
>-}
>-
>-TEST(ArrayUtilTest, PairDirection) {
>-  EXPECT_EQ(Point(1.f, 2.f, 3.f),
>-            PairDirection(Point(0.f, 0.f, 0.f), Point(1.f, 2.f, 3.f)));
>-  EXPECT_EQ(Point(-1.f, -2.f, -3.f),
>-            PairDirection(Point(1.f, 2.f, 3.f), Point(0.f, 0.f, 0.f)));
>-  EXPECT_EQ(Point(0.f, 0.f, 0.f),
>-            PairDirection(Point(1.f, 0.f, 0.f), Point(1.f, 0.f, 0.f)));
>-  EXPECT_EQ(Point(-1.f, 2.f, 0.f),
>-            PairDirection(Point(1.f, 0.f, 0.f), Point(0.f, 2.f, 0.f)));
>-  EXPECT_EQ(Point(-4.f, 4.f, -4.f),
>-            PairDirection(Point(1.f, -2.f, 3.f), Point(-3.f, 2.f, -1.f)));
>-}
>-
>-TEST(ArrayUtilTest, DotProduct) {
>-  EXPECT_FLOAT_EQ(0.f, DotProduct(Point(0.f, 0.f, 0.f), Point(1.f, 2.f, 3.f)));
>-  EXPECT_FLOAT_EQ(0.f, DotProduct(Point(1.f, 0.f, 2.f), Point(0.f, 3.f, 0.f)));
>-  EXPECT_FLOAT_EQ(0.f, DotProduct(Point(1.f, 1.f, 0.f), Point(1.f, -1.f, 0.f)));
>-  EXPECT_FLOAT_EQ(2.f, DotProduct(Point(1.f, 0.f, 0.f), Point(2.f, 0.f, 0.f)));
>-  EXPECT_FLOAT_EQ(-6.f,
>-                  DotProduct(Point(-2.f, 0.f, 0.f), Point(3.f, 0.f, 0.f)));
>-  EXPECT_FLOAT_EQ(-10.f,
>-                  DotProduct(Point(1.f, -2.f, 3.f), Point(-3.f, 2.f, -1.f)));
>-}
>-
>-TEST(ArrayUtilTest, CrossProduct) {
>-  EXPECT_EQ(Point(0.f, 0.f, 0.f),
>-            CrossProduct(Point(0.f, 0.f, 0.f), Point(1.f, 2.f, 3.f)));
>-  EXPECT_EQ(Point(0.f, 0.f, 1.f),
>-            CrossProduct(Point(1.f, 0.f, 0.f), Point(0.f, 1.f, 0.f)));
>-  EXPECT_EQ(Point(1.f, 0.f, 0.f),
>-            CrossProduct(Point(0.f, 1.f, 0.f), Point(0.f, 0.f, 1.f)));
>-  EXPECT_EQ(Point(0.f, -1.f, 0.f),
>-            CrossProduct(Point(1.f, 0.f, 0.f), Point(0.f, 0.f, 1.f)));
>-  EXPECT_EQ(Point(-4.f, -8.f, -4.f),
>-            CrossProduct(Point(1.f, -2.f, 3.f), Point(-3.f, 2.f, -1.f)));
>-}
>-
>-TEST(ArrayUtilTest, AreParallel) {
>-  EXPECT_TRUE(AreParallel(Point(0.f, 0.f, 0.f), Point(1.f, 2.f, 3.f)));
>-  EXPECT_FALSE(AreParallel(Point(1.f, 0.f, 2.f), Point(0.f, 3.f, 0.f)));
>-  EXPECT_FALSE(AreParallel(Point(1.f, 2.f, 0.f), Point(1.f, -0.5f, 0.f)));
>-  EXPECT_FALSE(AreParallel(Point(1.f, -2.f, 3.f), Point(-3.f, 2.f, -1.f)));
>-  EXPECT_TRUE(AreParallel(Point(1.f, 0.f, 0.f), Point(2.f, 0.f, 0.f)));
>-  EXPECT_TRUE(AreParallel(Point(1.f, 2.f, 3.f), Point(-2.f, -4.f, -6.f)));
>-}
>-
>-TEST(ArrayUtilTest, ArePerpendicular) {
>-  EXPECT_TRUE(ArePerpendicular(Point(0.f, 0.f, 0.f), Point(1.f, 2.f, 3.f)));
>-  EXPECT_TRUE(ArePerpendicular(Point(1.f, 0.f, 2.f), Point(0.f, 3.f, 0.f)));
>-  EXPECT_TRUE(ArePerpendicular(Point(1.f, 2.f, 0.f), Point(1.f, -0.5f, 0.f)));
>-  EXPECT_FALSE(ArePerpendicular(Point(1.f, -2.f, 3.f), Point(-3.f, 2.f, -1.f)));
>-  EXPECT_FALSE(ArePerpendicular(Point(1.f, 0.f, 0.f), Point(2.f, 0.f, 0.f)));
>-  EXPECT_FALSE(ArePerpendicular(Point(1.f, 2.f, 3.f), Point(-2.f, -4.f, -6.f)));
>-}
>-
>-TEST(ArrayUtilTest, GetMinimumSpacing) {
>-  std::vector<Point> geometry;
>-  geometry.push_back(Point(0.f, 0.f, 0.f));
>-  geometry.push_back(Point(0.1f, 0.f, 0.f));
>-  EXPECT_FLOAT_EQ(0.1f, GetMinimumSpacing(geometry));
>-  geometry.push_back(Point(0.f, 0.05f, 0.f));
>-  EXPECT_FLOAT_EQ(0.05f, GetMinimumSpacing(geometry));
>-  geometry.push_back(Point(0.f, 0.f, 0.02f));
>-  EXPECT_FLOAT_EQ(0.02f, GetMinimumSpacing(geometry));
>-  geometry.push_back(Point(-0.003f, -0.004f, 0.02f));
>-  EXPECT_FLOAT_EQ(0.005f, GetMinimumSpacing(geometry));
>-}
>-
>-TEST(ArrayUtilTest, GetDirectionIfLinear) {
>-  std::vector<Point> geometry;
>-  geometry.push_back(Point(0.f, 0.f, 0.f));
>-  geometry.push_back(Point(0.1f, 0.f, 0.f));
>-  EXPECT_TRUE(
>-      AreParallel(Point(1.f, 0.f, 0.f), *GetDirectionIfLinear(geometry)));
>-  geometry.push_back(Point(0.15f, 0.f, 0.f));
>-  EXPECT_TRUE(
>-      AreParallel(Point(1.f, 0.f, 0.f), *GetDirectionIfLinear(geometry)));
>-  geometry.push_back(Point(-0.2f, 0.f, 0.f));
>-  EXPECT_TRUE(
>-      AreParallel(Point(1.f, 0.f, 0.f), *GetDirectionIfLinear(geometry)));
>-  geometry.push_back(Point(0.05f, 0.f, 0.f));
>-  EXPECT_TRUE(
>-      AreParallel(Point(1.f, 0.f, 0.f), *GetDirectionIfLinear(geometry)));
>-  geometry.push_back(Point(0.1f, 0.1f, 0.f));
>-  EXPECT_FALSE(GetDirectionIfLinear(geometry));
>-  geometry.push_back(Point(0.f, 0.f, -0.2f));
>-  EXPECT_FALSE(GetDirectionIfLinear(geometry));
>-}
>-
>-TEST(ArrayUtilTest, GetNormalIfPlanar) {
>-  std::vector<Point> geometry;
>-  geometry.push_back(Point(0.f, 0.f, 0.f));
>-  geometry.push_back(Point(0.1f, 0.f, 0.f));
>-  EXPECT_FALSE(GetNormalIfPlanar(geometry));
>-  geometry.push_back(Point(0.15f, 0.f, 0.f));
>-  EXPECT_FALSE(GetNormalIfPlanar(geometry));
>-  geometry.push_back(Point(0.1f, 0.2f, 0.f));
>-  EXPECT_TRUE(AreParallel(Point(0.f, 0.f, 1.f), *GetNormalIfPlanar(geometry)));
>-  geometry.push_back(Point(0.f, -0.15f, 0.f));
>-  EXPECT_TRUE(AreParallel(Point(0.f, 0.f, 1.f), *GetNormalIfPlanar(geometry)));
>-  geometry.push_back(Point(0.f, 0.1f, 0.2f));
>-  EXPECT_FALSE(GetNormalIfPlanar(geometry));
>-  geometry.push_back(Point(0.f, 0.f, -0.15f));
>-  EXPECT_FALSE(GetNormalIfPlanar(geometry));
>-  geometry.push_back(Point(0.1f, 0.2f, 0.f));
>-  EXPECT_FALSE(GetNormalIfPlanar(geometry));
>-}
>-
>-TEST(ArrayUtilTest, GetArrayNormalIfExists) {
>-  std::vector<Point> geometry;
>-  geometry.push_back(Point(0.f, 0.f, 0.f));
>-  geometry.push_back(Point(0.1f, 0.f, 0.f));
>-  EXPECT_TRUE(
>-      AreParallel(Point(0.f, 1.f, 0.f), *GetArrayNormalIfExists(geometry)));
>-  geometry.push_back(Point(0.15f, 0.f, 0.f));
>-  EXPECT_TRUE(
>-      AreParallel(Point(0.f, 1.f, 0.f), *GetArrayNormalIfExists(geometry)));
>-  geometry.push_back(Point(0.1f, 0.f, 0.2f));
>-  EXPECT_TRUE(
>-      AreParallel(Point(0.f, 1.f, 0.f), *GetArrayNormalIfExists(geometry)));
>-  geometry.push_back(Point(0.f, 0.f, -0.1f));
>-  EXPECT_TRUE(
>-      AreParallel(Point(0.f, 1.f, 0.f), *GetArrayNormalIfExists(geometry)));
>-  geometry.push_back(Point(0.1f, 0.2f, 0.3f));
>-  EXPECT_FALSE(GetArrayNormalIfExists(geometry));
>-  geometry.push_back(Point(0.f, -0.1f, 0.f));
>-  EXPECT_FALSE(GetArrayNormalIfExists(geometry));
>-  geometry.push_back(Point(1.f, 0.f, -0.2f));
>-  EXPECT_FALSE(GetArrayNormalIfExists(geometry));
>-}
>-
>-TEST(ArrayUtilTest, DegreesToRadians) {
>-  EXPECT_FLOAT_EQ(0.f, DegreesToRadians(0.f));
>-  EXPECT_FLOAT_EQ(static_cast<float>(M_PI) / 6.f, DegreesToRadians(30.f));
>-  EXPECT_FLOAT_EQ(-static_cast<float>(M_PI) / 4.f, DegreesToRadians(-45.f));
>-  EXPECT_FLOAT_EQ(static_cast<float>(M_PI) / 3.f, DegreesToRadians(60.f));
>-  EXPECT_FLOAT_EQ(-static_cast<float>(M_PI) / 2.f, DegreesToRadians(-90.f));
>-  EXPECT_FLOAT_EQ(2.f * static_cast<float>(M_PI) / 3.f,
>-                  DegreesToRadians(120.f));
>-  EXPECT_FLOAT_EQ(-3.f * static_cast<float>(M_PI) / 4.f,
>-                  DegreesToRadians(-135.f));
>-  EXPECT_FLOAT_EQ(5.f * static_cast<float>(M_PI) / 6.f,
>-                  DegreesToRadians(150.f));
>-  EXPECT_FLOAT_EQ(-static_cast<float>(M_PI), DegreesToRadians(-180.f));
>-}
>-
>-TEST(ArrayUtilTest, RadiansToDegrees) {
>-  EXPECT_FLOAT_EQ(0.f, RadiansToDegrees(0.f));
>-  EXPECT_FLOAT_EQ(30.f, RadiansToDegrees(M_PI / 6.f));
>-  EXPECT_FLOAT_EQ(-45.f, RadiansToDegrees(-M_PI / 4.f));
>-  EXPECT_FLOAT_EQ(60.f, RadiansToDegrees(M_PI / 3.f));
>-  EXPECT_FLOAT_EQ(-90.f, RadiansToDegrees(-M_PI / 2.f));
>-  EXPECT_FLOAT_EQ(120.f, RadiansToDegrees(2.f * M_PI / 3.f));
>-  EXPECT_FLOAT_EQ(-135.f, RadiansToDegrees(-3.f * M_PI / 4.f));
>-  EXPECT_FLOAT_EQ(150.f, RadiansToDegrees(5.f * M_PI / 6.f));
>-  EXPECT_FLOAT_EQ(-180.f, RadiansToDegrees(-M_PI));
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/complex_matrix.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/complex_matrix.h
>deleted file mode 100644
>index 9960e1dd870..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/complex_matrix.h
>+++ /dev/null
>@@ -1,96 +0,0 @@
>-/*
>- *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MODULES_AUDIO_PROCESSING_BEAMFORMER_COMPLEX_MATRIX_H_
>-#define MODULES_AUDIO_PROCESSING_BEAMFORMER_COMPLEX_MATRIX_H_
>-
>-#include <complex>
>-
>-#include "modules/audio_processing/beamformer/matrix.h"
>-#include "rtc_base/checks.h"
>-
>-namespace webrtc {
>-
>-using std::complex;
>-
>-// An extension of Matrix for operations that only work on a complex type.
>-template <typename T>
>-class ComplexMatrix : public Matrix<complex<T> > {
>- public:
>-  ComplexMatrix() : Matrix<complex<T> >() {}
>-
>-  ComplexMatrix(size_t num_rows, size_t num_columns)
>-      : Matrix<complex<T> >(num_rows, num_columns) {}
>-
>-  ComplexMatrix(const complex<T>* data, size_t num_rows, size_t num_columns)
>-      : Matrix<complex<T> >(data, num_rows, num_columns) {}
>-
>-  // Complex Matrix operations.
>-  ComplexMatrix& PointwiseConjugate() {
>-    complex<T>* const data = this->data();
>-    size_t size = this->num_rows() * this->num_columns();
>-    for (size_t i = 0; i < size; ++i) {
>-      data[i] = conj(data[i]);
>-    }
>-
>-    return *this;
>-  }
>-
>-  ComplexMatrix& PointwiseConjugate(const ComplexMatrix& operand) {
>-    this->CopyFrom(operand);
>-    return PointwiseConjugate();
>-  }
>-
>-  ComplexMatrix& ConjugateTranspose() {
>-    this->CopyDataToScratch();
>-    size_t num_rows = this->num_rows();
>-    this->SetNumRows(this->num_columns());
>-    this->SetNumColumns(num_rows);
>-    this->Resize();
>-    return ConjugateTranspose(this->scratch_elements());
>-  }
>-
>-  ComplexMatrix& ConjugateTranspose(const ComplexMatrix& operand) {
>-    RTC_CHECK_EQ(operand.num_rows(), this->num_columns());
>-    RTC_CHECK_EQ(operand.num_columns(), this->num_rows());
>-    return ConjugateTranspose(operand.elements());
>-  }
>-
>-  ComplexMatrix& ZeroImag() {
>-    complex<T>* const data = this->data();
>-    size_t size = this->num_rows() * this->num_columns();
>-    for (size_t i = 0; i < size; ++i) {
>-      data[i] = complex<T>(data[i].real(), 0);
>-    }
>-
>-    return *this;
>-  }
>-
>-  ComplexMatrix& ZeroImag(const ComplexMatrix& operand) {
>-    this->CopyFrom(operand);
>-    return ZeroImag();
>-  }
>-
>- private:
>-  ComplexMatrix& ConjugateTranspose(const complex<T>* const* src) {
>-    complex<T>* const* elements = this->elements();
>-    for (size_t i = 0; i < this->num_rows(); ++i) {
>-      for (size_t j = 0; j < this->num_columns(); ++j) {
>-        elements[i][j] = conj(src[j][i]);
>-      }
>-    }
>-
>-    return *this;
>-  }
>-};
>-
>-}  // namespace webrtc
>-
>-#endif  // MODULES_AUDIO_PROCESSING_BEAMFORMER_COMPLEX_MATRIX_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/complex_matrix_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/complex_matrix_unittest.cc
>deleted file mode 100644
>index e11dfd2b5c3..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/complex_matrix_unittest.cc
>+++ /dev/null
>@@ -1,102 +0,0 @@
>-/*
>- *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "modules/audio_processing/beamformer/complex_matrix.h"
>-#include "modules/audio_processing/beamformer/matrix_test_helpers.h"
>-#include "test/gtest.h"
>-
>-namespace webrtc {
>-
>-TEST(ComplexMatrixTest, TestPointwiseConjugate) {
>-  const int kNumRows = 2;
>-  const int kNumCols = 4;
>-
>-  const complex<float> kValuesInitial[kNumRows][kNumCols] = {
>-      {complex<float>(1.1f, 1.1f), complex<float>(2.2f, -2.2f),
>-       complex<float>(3.3f, 3.3f), complex<float>(4.4f, -4.4f)},
>-      {complex<float>(5.5f, 5.5f), complex<float>(6.6f, -6.6f),
>-       complex<float>(7.7f, 7.7f), complex<float>(8.8f, -8.8f)}};
>-
>-  const complex<float> kValuesExpected[kNumRows][kNumCols] = {
>-      {complex<float>(1.1f, -1.1f), complex<float>(2.2f, 2.2f),
>-       complex<float>(3.3f, -3.3f), complex<float>(4.4f, 4.4f)},
>-      {complex<float>(5.5f, -5.5f), complex<float>(6.6f, 6.6f),
>-       complex<float>(7.7f, -7.7f), complex<float>(8.8f, 8.8f)}};
>-
>-  ComplexMatrix<float> initial_mat(*kValuesInitial, kNumRows, kNumCols);
>-  ComplexMatrix<float> expected_result(*kValuesExpected, kNumRows, kNumCols);
>-  ComplexMatrix<float> actual_result(kNumRows, kNumCols);
>-
>-  actual_result.PointwiseConjugate(initial_mat);
>-  MatrixTestHelpers::ValidateMatrixEqualityComplexFloat(expected_result,
>-                                                        actual_result);
>-
>-  initial_mat.PointwiseConjugate();
>-  MatrixTestHelpers::ValidateMatrixEqualityComplexFloat(initial_mat,
>-                                                        actual_result);
>-}
>-
>-TEST(ComplexMatrixTest, TestConjugateTranspose) {
>-  const int kNumInitialRows = 2;
>-  const int kNumInitialCols = 4;
>-  const int kNumResultRows = 4;
>-  const int kNumResultCols = 2;
>-
>-  const complex<float> kValuesInitial[kNumInitialRows][kNumInitialCols] = {
>-      {complex<float>(1.1f, 1.1f), complex<float>(2.2f, 2.2f),
>-       complex<float>(3.3f, 3.3f), complex<float>(4.4f, 4.4f)},
>-      {complex<float>(5.5f, 5.5f), complex<float>(6.6f, 6.6f),
>-       complex<float>(7.7f, 7.7f), complex<float>(8.8f, 8.8f)}};
>-
>-  const complex<float> kValuesExpected[kNumResultRows][kNumResultCols] = {
>-      {complex<float>(1.1f, -1.1f), complex<float>(5.5f, -5.5f)},
>-      {complex<float>(2.2f, -2.2f), complex<float>(6.6f, -6.6f)},
>-      {complex<float>(3.3f, -3.3f), complex<float>(7.7f, -7.7f)},
>-      {complex<float>(4.4f, -4.4f), complex<float>(8.8f, -8.8f)}};
>-
>-  ComplexMatrix<float> initial_mat(
>-      *kValuesInitial, kNumInitialRows, kNumInitialCols);
>-  ComplexMatrix<float> expected_result(
>-      *kValuesExpected, kNumResultRows, kNumResultCols);
>-  ComplexMatrix<float> actual_result(kNumResultRows, kNumResultCols);
>-
>-  actual_result.ConjugateTranspose(initial_mat);
>-  MatrixTestHelpers::ValidateMatrixEqualityComplexFloat(expected_result,
>-                                                        actual_result);
>-
>-  initial_mat.ConjugateTranspose();
>-  MatrixTestHelpers::ValidateMatrixEqualityComplexFloat(initial_mat,
>-                                                        actual_result);
>-}
>-
>-TEST(ComplexMatrixTest, TestZeroImag) {
>-  const int kNumRows = 2;
>-  const int kNumCols = 2;
>-  const complex<float> kValuesInitial[kNumRows][kNumCols] = {
>-      {complex<float>(1.1f, 1.1f), complex<float>(2.2f, 2.2f)},
>-      {complex<float>(3.3f, 3.3f), complex<float>(4.4f, 4.4f)}};
>-  const complex<float> kValuesExpected[kNumRows][kNumCols] = {
>-      {complex<float>(1.1f, 0.f), complex<float>(2.2f, 0.f)},
>-      {complex<float>(3.3f, 0.f), complex<float>(4.4f, 0.f)}};
>-
>-  ComplexMatrix<float> initial_mat(*kValuesInitial, kNumRows, kNumCols);
>-  ComplexMatrix<float> expected_result(*kValuesExpected, kNumRows, kNumCols);
>-  ComplexMatrix<float> actual_result;
>-
>-  actual_result.ZeroImag(initial_mat);
>-  MatrixTestHelpers::ValidateMatrixEqualityComplexFloat(expected_result,
>-                                                        actual_result);
>-
>-  initial_mat.ZeroImag();
>-  MatrixTestHelpers::ValidateMatrixEqualityComplexFloat(initial_mat,
>-                                                        actual_result);
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/covariance_matrix_generator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/covariance_matrix_generator.cc
>deleted file mode 100644
>index df36d5974fd..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/covariance_matrix_generator.cc
>+++ /dev/null
>@@ -1,114 +0,0 @@
>-/*
>- *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#define _USE_MATH_DEFINES
>-
>-#include "modules/audio_processing/beamformer/covariance_matrix_generator.h"
>-
>-#include <cmath>
>-
>-namespace webrtc {
>-namespace {
>-
>-float BesselJ0(float x) {
>-#ifdef WEBRTC_WIN
>-  return _j0(x);
>-#else
>-  return j0(x);
>-#endif
>-}
>-
>-// Calculates the Euclidean norm for a row vector.
>-float Norm(const ComplexMatrix<float>& x) {
>-  RTC_CHECK_EQ(1, x.num_rows());
>-  const size_t length = x.num_columns();
>-  const complex<float>* elems = x.elements()[0];
>-  float result = 0.f;
>-  for (size_t i = 0u; i < length; ++i) {
>-    result += std::norm(elems[i]);
>-  }
>-  return std::sqrt(result);
>-}
>-
>-}  // namespace
>-
>-void CovarianceMatrixGenerator::UniformCovarianceMatrix(
>-    float wave_number,
>-    const std::vector<Point>& geometry,
>-    ComplexMatrix<float>* mat) {
>-  RTC_CHECK_EQ(geometry.size(), mat->num_rows());
>-  RTC_CHECK_EQ(geometry.size(), mat->num_columns());
>-
>-  complex<float>* const* mat_els = mat->elements();
>-  for (size_t i = 0; i < geometry.size(); ++i) {
>-    for (size_t j = 0; j < geometry.size(); ++j) {
>-      if (wave_number > 0.f) {
>-        mat_els[i][j] =
>-            BesselJ0(wave_number * Distance(geometry[i], geometry[j]));
>-      } else {
>-        mat_els[i][j] = i == j ? 1.f : 0.f;
>-      }
>-    }
>-  }
>-}
>-
>-void CovarianceMatrixGenerator::AngledCovarianceMatrix(
>-    float sound_speed,
>-    float angle,
>-    size_t frequency_bin,
>-    size_t fft_size,
>-    size_t num_freq_bins,
>-    int sample_rate,
>-    const std::vector<Point>& geometry,
>-    ComplexMatrix<float>* mat) {
>-  RTC_CHECK_EQ(geometry.size(), mat->num_rows());
>-  RTC_CHECK_EQ(geometry.size(), mat->num_columns());
>-
>-  ComplexMatrix<float> interf_cov_vector(1, geometry.size());
>-  ComplexMatrix<float> interf_cov_vector_transposed(geometry.size(), 1);
>-  PhaseAlignmentMasks(frequency_bin,
>-                      fft_size,
>-                      sample_rate,
>-                      sound_speed,
>-                      geometry,
>-                      angle,
>-                      &interf_cov_vector);
>-  interf_cov_vector.Scale(1.f / Norm(interf_cov_vector));
>-  interf_cov_vector_transposed.Transpose(interf_cov_vector);
>-  interf_cov_vector.PointwiseConjugate();
>-  mat->Multiply(interf_cov_vector_transposed, interf_cov_vector);
>-}
>-
>-void CovarianceMatrixGenerator::PhaseAlignmentMasks(
>-    size_t frequency_bin,
>-    size_t fft_size,
>-    int sample_rate,
>-    float sound_speed,
>-    const std::vector<Point>& geometry,
>-    float angle,
>-    ComplexMatrix<float>* mat) {
>-  RTC_CHECK_EQ(1, mat->num_rows());
>-  RTC_CHECK_EQ(geometry.size(), mat->num_columns());
>-
>-  float freq_in_hertz =
>-      (static_cast<float>(frequency_bin) / fft_size) * sample_rate;
>-
>-  complex<float>* const* mat_els = mat->elements();
>-  for (size_t c_ix = 0; c_ix < geometry.size(); ++c_ix) {
>-    float distance = std::cos(angle) * geometry[c_ix].x() +
>-                     std::sin(angle) * geometry[c_ix].y();
>-    float phase_shift = -2.f * M_PI * distance * freq_in_hertz / sound_speed;
>-
>-    // Euler's formula for mat[0][c_ix] = e^(j * phase_shift).
>-    mat_els[0][c_ix] = complex<float>(cos(phase_shift), sin(phase_shift));
>-  }
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/covariance_matrix_generator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/covariance_matrix_generator.h
>deleted file mode 100644
>index 6a5841f631c..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/covariance_matrix_generator.h
>+++ /dev/null
>@@ -1,54 +0,0 @@
>-/*
>- *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MODULES_AUDIO_PROCESSING_BEAMFORMER_COVARIANCE_MATRIX_GENERATOR_H_
>-#define MODULES_AUDIO_PROCESSING_BEAMFORMER_COVARIANCE_MATRIX_GENERATOR_H_
>-
>-#include "modules/audio_processing/beamformer/complex_matrix.h"
>-#include "modules/audio_processing/beamformer/array_util.h"
>-
>-namespace webrtc {
>-
>-// Helper class for Beamformer in charge of generating covariance matrices. For
>-// each function, the passed-in ComplexMatrix is expected to be of size
>-// |num_input_channels| x |num_input_channels|.
>-class CovarianceMatrixGenerator {
>- public:
>-  // A uniform covariance matrix with a gap at the target location. WARNING:
>-  // The target angle is assumed to be 0.
>-  static void UniformCovarianceMatrix(float wave_number,
>-                                      const std::vector<Point>& geometry,
>-                                      ComplexMatrix<float>* mat);
>-
>-  // The covariance matrix of a source at the given angle.
>-  static void AngledCovarianceMatrix(float sound_speed,
>-                                     float angle,
>-                                     size_t frequency_bin,
>-                                     size_t fft_size,
>-                                     size_t num_freq_bins,
>-                                     int sample_rate,
>-                                     const std::vector<Point>& geometry,
>-                                     ComplexMatrix<float>* mat);
>-
>-  // Calculates phase shifts that, when applied to a multichannel signal and
>-  // added together, cause constructive interferernce for sources located at
>-  // the given angle.
>-  static void PhaseAlignmentMasks(size_t frequency_bin,
>-                                  size_t fft_size,
>-                                  int sample_rate,
>-                                  float sound_speed,
>-                                  const std::vector<Point>& geometry,
>-                                  float angle,
>-                                  ComplexMatrix<float>* mat);
>-};
>-
>-}  // namespace webrtc
>-
>-#endif  // MODULES_AUDIO_PROCESSING_BEAMFORMER_BF_HELPERS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/covariance_matrix_generator_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/covariance_matrix_generator_unittest.cc
>deleted file mode 100644
>index a6518e5d95f..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/covariance_matrix_generator_unittest.cc
>+++ /dev/null
>@@ -1,230 +0,0 @@
>-/*
>- *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#define _USE_MATH_DEFINES
>-
>-#include "modules/audio_processing/beamformer/covariance_matrix_generator.h"
>-
>-#include <cmath>
>-
>-#include "modules/audio_processing/beamformer/matrix_test_helpers.h"
>-#include "test/gtest.h"
>-
>-namespace webrtc {
>-
>-using std::complex;
>-
>-TEST(CovarianceMatrixGeneratorTest, TestUniformCovarianceMatrix2Mics) {
>-  const float kWaveNumber = 0.5775f;
>-  const int kNumberMics = 2;
>-  const float kMicSpacing = 0.05f;
>-  const float kTolerance = 0.0001f;
>-  std::vector<Point> geometry;
>-  float first_mic = (kNumberMics - 1) * kMicSpacing / 2.f;
>-  for (int i = 0; i < kNumberMics; ++i) {
>-    geometry.push_back(Point(i * kMicSpacing - first_mic, 0.f, 0.f));
>-  }
>-  ComplexMatrix<float> actual_covariance_matrix(kNumberMics, kNumberMics);
>-  CovarianceMatrixGenerator::UniformCovarianceMatrix(kWaveNumber,
>-                                                     geometry,
>-                                                     &actual_covariance_matrix);
>-
>-  complex<float>* const* actual_els = actual_covariance_matrix.elements();
>-
>-  EXPECT_NEAR(actual_els[0][0].real(), 1.f, kTolerance);
>-  EXPECT_NEAR(actual_els[0][1].real(), 0.9998f, kTolerance);
>-  EXPECT_NEAR(actual_els[1][0].real(), 0.9998f, kTolerance);
>-  EXPECT_NEAR(actual_els[1][1].real(), 1.f, kTolerance);
>-
>-  EXPECT_NEAR(actual_els[0][0].imag(), 0.f, kTolerance);
>-  EXPECT_NEAR(actual_els[0][1].imag(), 0.f, kTolerance);
>-  EXPECT_NEAR(actual_els[1][0].imag(), 0.f, kTolerance);
>-  EXPECT_NEAR(actual_els[1][1].imag(), 0.f, kTolerance);
>-}
>-
>-TEST(CovarianceMatrixGeneratorTest, TestUniformCovarianceMatrix3Mics) {
>-  const float kWaveNumber = 10.3861f;
>-  const int kNumberMics = 3;
>-  const float kMicSpacing = 0.04f;
>-  const float kTolerance = 0.0001f;
>-  std::vector<Point> geometry;
>-  float first_mic = (kNumberMics - 1) * kMicSpacing / 2.f;
>-  for (int i = 0; i < kNumberMics; ++i) {
>-    geometry.push_back(Point(i * kMicSpacing - first_mic, 0.f, 0.f));
>-  }
>-  ComplexMatrix<float> actual_covariance_matrix(kNumberMics, kNumberMics);
>-  CovarianceMatrixGenerator::UniformCovarianceMatrix(kWaveNumber,
>-                                                     geometry,
>-                                                     &actual_covariance_matrix);
>-
>-  complex<float>* const* actual_els = actual_covariance_matrix.elements();
>-
>-  EXPECT_NEAR(actual_els[0][0].real(), 1.f, kTolerance);
>-  EXPECT_NEAR(actual_els[0][1].real(), 0.9573f, kTolerance);
>-  EXPECT_NEAR(actual_els[0][2].real(), 0.8347f, kTolerance);
>-  EXPECT_NEAR(actual_els[1][0].real(), 0.9573f, kTolerance);
>-  EXPECT_NEAR(actual_els[1][1].real(), 1.f, kTolerance);
>-  EXPECT_NEAR(actual_els[1][2].real(), 0.9573f, kTolerance);
>-  EXPECT_NEAR(actual_els[2][0].real(), 0.8347f, kTolerance);
>-  EXPECT_NEAR(actual_els[2][1].real(), 0.9573f, kTolerance);
>-  EXPECT_NEAR(actual_els[2][2].real(), 1.f, kTolerance);
>-
>-  EXPECT_NEAR(actual_els[0][0].imag(), 0.f, kTolerance);
>-  EXPECT_NEAR(actual_els[0][1].imag(), 0.f, kTolerance);
>-  EXPECT_NEAR(actual_els[0][2].imag(), 0.f, kTolerance);
>-  EXPECT_NEAR(actual_els[1][0].imag(), 0.f, kTolerance);
>-  EXPECT_NEAR(actual_els[1][1].imag(), 0.f, kTolerance);
>-  EXPECT_NEAR(actual_els[1][2].imag(), 0.f, kTolerance);
>-  EXPECT_NEAR(actual_els[2][0].imag(), 0.f, kTolerance);
>-  EXPECT_NEAR(actual_els[2][1].imag(), 0.f, kTolerance);
>-  EXPECT_NEAR(actual_els[2][2].imag(), 0.f, kTolerance);
>-}
>-
>-TEST(CovarianceMatrixGeneratorTest, TestUniformCovarianceMatrix3DArray) {
>-  const float kWaveNumber = 1.2345f;
>-  const int kNumberMics = 4;
>-  const float kTolerance = 0.0001f;
>-  std::vector<Point> geometry;
>-  geometry.push_back(Point(-0.025f, -0.05f, -0.075f));
>-  geometry.push_back(Point(0.075f, -0.05f, -0.075f));
>-  geometry.push_back(Point(-0.025f, 0.15f, -0.075f));
>-  geometry.push_back(Point(-0.025f, -0.05f, 0.225f));
>-  ComplexMatrix<float> actual_covariance_matrix(kNumberMics, kNumberMics);
>-  CovarianceMatrixGenerator::UniformCovarianceMatrix(kWaveNumber,
>-                                                     geometry,
>-                                                     &actual_covariance_matrix);
>-
>-  complex<float>* const* actual_els = actual_covariance_matrix.elements();
>-
>-  EXPECT_NEAR(actual_els[0][0].real(), 1.f, kTolerance);
>-  EXPECT_NEAR(actual_els[0][1].real(), 0.9962f, kTolerance);
>-  EXPECT_NEAR(actual_els[0][2].real(), 0.9848f, kTolerance);
>-  EXPECT_NEAR(actual_els[0][3].real(), 0.9660f, kTolerance);
>-  EXPECT_NEAR(actual_els[1][0].real(), 0.9962f, kTolerance);
>-  EXPECT_NEAR(actual_els[1][1].real(), 1.f, kTolerance);
>-  EXPECT_NEAR(actual_els[1][2].real(), 0.9810f, kTolerance);
>-  EXPECT_NEAR(actual_els[1][3].real(), 0.9623f, kTolerance);
>-  EXPECT_NEAR(actual_els[2][0].real(), 0.9848f, kTolerance);
>-  EXPECT_NEAR(actual_els[2][1].real(), 0.9810f, kTolerance);
>-  EXPECT_NEAR(actual_els[2][2].real(), 1.f, kTolerance);
>-  EXPECT_NEAR(actual_els[2][3].real(), 0.9511f, kTolerance);
>-  EXPECT_NEAR(actual_els[3][0].real(), 0.9660f, kTolerance);
>-  EXPECT_NEAR(actual_els[3][1].real(), 0.9623f, kTolerance);
>-  EXPECT_NEAR(actual_els[3][2].real(), 0.9511f, kTolerance);
>-  EXPECT_NEAR(actual_els[3][3].real(), 1.f, kTolerance);
>-
>-  EXPECT_NEAR(actual_els[0][0].imag(), 0.f, kTolerance);
>-  EXPECT_NEAR(actual_els[0][1].imag(), 0.f, kTolerance);
>-  EXPECT_NEAR(actual_els[0][2].imag(), 0.f, kTolerance);
>-  EXPECT_NEAR(actual_els[0][3].imag(), 0.f, kTolerance);
>-  EXPECT_NEAR(actual_els[1][0].imag(), 0.f, kTolerance);
>-  EXPECT_NEAR(actual_els[1][1].imag(), 0.f, kTolerance);
>-  EXPECT_NEAR(actual_els[1][2].imag(), 0.f, kTolerance);
>-  EXPECT_NEAR(actual_els[1][3].imag(), 0.f, kTolerance);
>-  EXPECT_NEAR(actual_els[2][0].imag(), 0.f, kTolerance);
>-  EXPECT_NEAR(actual_els[2][1].imag(), 0.f, kTolerance);
>-  EXPECT_NEAR(actual_els[2][2].imag(), 0.f, kTolerance);
>-  EXPECT_NEAR(actual_els[2][3].imag(), 0.f, kTolerance);
>-  EXPECT_NEAR(actual_els[3][0].imag(), 0.f, kTolerance);
>-  EXPECT_NEAR(actual_els[3][1].imag(), 0.f, kTolerance);
>-  EXPECT_NEAR(actual_els[3][2].imag(), 0.f, kTolerance);
>-  EXPECT_NEAR(actual_els[3][3].imag(), 0.f, kTolerance);
>-}
>-
>-TEST(CovarianceMatrixGeneratorTest, TestAngledCovarianceMatrix2Mics) {
>-  const float kSpeedOfSound = 340;
>-  const float kAngle = static_cast<float>(M_PI) / 4.f;
>-  const float kFrequencyBin = 6;
>-  const float kFftSize = 512;
>-  const int kNumberFrequencyBins = 257;
>-  const int kSampleRate = 16000;
>-  const int kNumberMics = 2;
>-  const float kMicSpacing = 0.04f;
>-  const float kTolerance = 0.0001f;
>-  std::vector<Point> geometry;
>-  float first_mic = (kNumberMics - 1) * kMicSpacing / 2.f;
>-  for (int i = 0; i < kNumberMics; ++i) {
>-    geometry.push_back(Point(i * kMicSpacing - first_mic, 0.f, 0.f));
>-  }
>-  ComplexMatrix<float> actual_covariance_matrix(kNumberMics, kNumberMics);
>-  CovarianceMatrixGenerator::AngledCovarianceMatrix(kSpeedOfSound,
>-                                                    kAngle,
>-                                                    kFrequencyBin,
>-                                                    kFftSize,
>-                                                    kNumberFrequencyBins,
>-                                                    kSampleRate,
>-                                                    geometry,
>-                                                    &actual_covariance_matrix);
>-
>-  complex<float>* const* actual_els = actual_covariance_matrix.elements();
>-
>-  EXPECT_NEAR(actual_els[0][0].real(), 0.5f, kTolerance);
>-  EXPECT_NEAR(actual_els[0][1].real(), 0.4976f, kTolerance);
>-  EXPECT_NEAR(actual_els[1][0].real(), 0.4976f, kTolerance);
>-  EXPECT_NEAR(actual_els[1][1].real(), 0.5f, kTolerance);
>-
>-  EXPECT_NEAR(actual_els[0][0].imag(), 0.f, kTolerance);
>-  EXPECT_NEAR(actual_els[0][1].imag(), 0.0489f, kTolerance);
>-  EXPECT_NEAR(actual_els[1][0].imag(), -0.0489f, kTolerance);
>-  EXPECT_NEAR(actual_els[1][1].imag(), 0.f, kTolerance);
>-}
>-
>-TEST(CovarianceMatrixGeneratorTest, TestAngledCovarianceMatrix3Mics) {
>-  const float kSpeedOfSound = 340;
>-  const float kAngle = static_cast<float>(M_PI) / 4.f;
>-  const float kFrequencyBin = 9;
>-  const float kFftSize = 512;
>-  const int kNumberFrequencyBins = 257;
>-  const int kSampleRate = 42000;
>-  const int kNumberMics = 3;
>-  const float kMicSpacing = 0.05f;
>-  const float kTolerance = 0.0001f;
>-  std::vector<Point> geometry;
>-  float first_mic = (kNumberMics - 1) * kMicSpacing / 2.f;
>-  for (int i = 0; i < kNumberMics; ++i) {
>-    geometry.push_back(Point(i * kMicSpacing - first_mic, 0.f, 0.f));
>-  }
>-  ComplexMatrix<float> actual_covariance_matrix(kNumberMics, kNumberMics);
>-  CovarianceMatrixGenerator::AngledCovarianceMatrix(kSpeedOfSound,
>-                                                    kAngle,
>-                                                    kFrequencyBin,
>-                                                    kFftSize,
>-                                                    kNumberFrequencyBins,
>-                                                    kSampleRate,
>-                                                    geometry,
>-                                                    &actual_covariance_matrix);
>-
>-  complex<float>* const* actual_els = actual_covariance_matrix.elements();
>-
>-  EXPECT_NEAR(actual_els[0][0].real(), 0.3333f, kTolerance);
>-  EXPECT_NEAR(actual_els[0][1].real(), 0.2953f, kTolerance);
>-  EXPECT_NEAR(actual_els[0][2].real(), 0.1899f, kTolerance);
>-  EXPECT_NEAR(actual_els[1][0].real(), 0.2953f, kTolerance);
>-  EXPECT_NEAR(actual_els[1][1].real(), 0.3333f, kTolerance);
>-  EXPECT_NEAR(actual_els[1][2].real(), 0.2953f, kTolerance);
>-  EXPECT_NEAR(actual_els[2][0].real(), 0.1899f, kTolerance);
>-  EXPECT_NEAR(actual_els[2][1].real(), 0.2953f, kTolerance);
>-  EXPECT_NEAR(actual_els[2][2].real(), 0.3333f, kTolerance);
>-
>-  EXPECT_NEAR(actual_els[0][0].imag(), 0.f, kTolerance);
>-  EXPECT_NEAR(actual_els[0][1].imag(), 0.1546f, kTolerance);
>-  EXPECT_NEAR(actual_els[0][2].imag(), 0.274f, kTolerance);
>-  EXPECT_NEAR(actual_els[1][0].imag(), -0.1546f, kTolerance);
>-  EXPECT_NEAR(actual_els[1][1].imag(), 0.f, kTolerance);
>-  EXPECT_NEAR(actual_els[1][2].imag(), 0.1546f, kTolerance);
>-  EXPECT_NEAR(actual_els[2][0].imag(), -0.274f, kTolerance);
>-  EXPECT_NEAR(actual_els[2][1].imag(), -0.1546f, kTolerance);
>-  EXPECT_NEAR(actual_els[2][2].imag(), 0.f, kTolerance);
>-}
>-
>-// PhaseAlignmentMasks is tested by AngledCovarianceMatrix and by
>-// InitBeamformerWeights in BeamformerUnittest.
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/matrix.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/matrix.h
>deleted file mode 100644
>index bf94c25df01..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/matrix.h
>+++ /dev/null
>@@ -1,369 +0,0 @@
>-/*
>- *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MODULES_AUDIO_PROCESSING_BEAMFORMER_MATRIX_H_
>-#define MODULES_AUDIO_PROCESSING_BEAMFORMER_MATRIX_H_
>-
>-#include <algorithm>
>-#include <cstring>
>-#include <string>
>-#include <vector>
>-
>-#include "rtc_base/checks.h"
>-#include "rtc_base/constructormagic.h"
>-
>-namespace {
>-
>-// Wrappers to get around the compiler warning resulting from the fact that
>-// there's no std::sqrt overload for ints. We cast all non-complex types to
>-// a double for the sqrt method.
>-template <typename T>
>-T sqrt_wrapper(T x) {
>-  return sqrt(static_cast<double>(x));
>-}
>-
>-template <typename S>
>-std::complex<S> sqrt_wrapper(std::complex<S> x) {
>-  return sqrt(x);
>-}
>-} // namespace
>-
>-namespace webrtc {
>-
>-// Matrix is a class for doing standard matrix operations on 2 dimensional
>-// matrices of any size. Results of matrix operations are stored in the
>-// calling object. Function overloads exist for both in-place (the calling
>-// object is used as both an operand and the result) and out-of-place (all
>-// operands are passed in as parameters) operations. If operand dimensions
>-// mismatch, the program crashes. Out-of-place operations change the size of
>-// the calling object, if necessary, before operating.
>-//
>-// 'In-place' operations that inherently change the size of the matrix (eg.
>-// Transpose, Multiply on different-sized matrices) must make temporary copies
>-// (|scratch_elements_| and |scratch_data_|) of existing data to complete the
>-// operations.
>-//
>-// The data is stored contiguously. Data can be accessed internally as a flat
>-// array, |data_|, or as an array of row pointers, |elements_|, but is
>-// available to users only as an array of row pointers through |elements()|.
>-// Memory for storage is allocated when a matrix is resized only if the new
>-// size overflows capacity. Memory needed temporarily for any operations is
>-// similarly resized only if the new size overflows capacity.
>-//
>-// If you pass in storage through the ctor, that storage is copied into the
>-// matrix. TODO(claguna): albeit tricky, allow for data to be referenced
>-// instead of copied, and owned by the user.
>-template <typename T>
>-class Matrix {
>- public:
>-  Matrix() : num_rows_(0), num_columns_(0) {}
>-
>-  // Allocates space for the elements and initializes all values to zero.
>-  Matrix(size_t num_rows, size_t num_columns)
>-      : num_rows_(num_rows), num_columns_(num_columns) {
>-    Resize();
>-    scratch_data_.resize(num_rows_ * num_columns_);
>-    scratch_elements_.resize(num_rows_);
>-  }
>-
>-  // Copies |data| into the new Matrix.
>-  Matrix(const T* data, size_t num_rows, size_t num_columns)
>-      : num_rows_(0), num_columns_(0) {
>-    CopyFrom(data, num_rows, num_columns);
>-    scratch_data_.resize(num_rows_ * num_columns_);
>-    scratch_elements_.resize(num_rows_);
>-  }
>-
>-  virtual ~Matrix() {}
>-
>-  // Deep copy an existing matrix.
>-  void CopyFrom(const Matrix& other) {
>-    CopyFrom(&other.data_[0], other.num_rows_, other.num_columns_);
>-  }
>-
>-  // Copy |data| into the Matrix. The current data is lost.
>-  void CopyFrom(const T* const data, size_t num_rows, size_t num_columns) {
>-    Resize(num_rows, num_columns);
>-    memcpy(&data_[0], data, num_rows_ * num_columns_ * sizeof(data_[0]));
>-  }
>-
>-  Matrix& CopyFromColumn(const T* const* src,
>-                         size_t column_index,
>-                         size_t num_rows) {
>-    Resize(1, num_rows);
>-    for (size_t i = 0; i < num_columns_; ++i) {
>-      data_[i] = src[i][column_index];
>-    }
>-
>-    return *this;
>-  }
>-
>-  void Resize(size_t num_rows, size_t num_columns) {
>-    if (num_rows != num_rows_ || num_columns != num_columns_) {
>-      num_rows_ = num_rows;
>-      num_columns_ = num_columns;
>-      Resize();
>-    }
>-  }
>-
>-  // Accessors and mutators.
>-  size_t num_rows() const { return num_rows_; }
>-  size_t num_columns() const { return num_columns_; }
>-  T* const* elements() { return &elements_[0]; }
>-  const T* const* elements() const { return &elements_[0]; }
>-
>-  T Trace() {
>-    RTC_CHECK_EQ(num_rows_, num_columns_);
>-
>-    T trace = 0;
>-    for (size_t i = 0; i < num_rows_; ++i) {
>-      trace += elements_[i][i];
>-    }
>-    return trace;
>-  }
>-
>-  // Matrix Operations. Returns *this to support method chaining.
>-  Matrix& Transpose() {
>-    CopyDataToScratch();
>-    Resize(num_columns_, num_rows_);
>-    return Transpose(scratch_elements());
>-  }
>-
>-  Matrix& Transpose(const Matrix& operand) {
>-    RTC_CHECK_EQ(operand.num_rows_, num_columns_);
>-    RTC_CHECK_EQ(operand.num_columns_, num_rows_);
>-
>-    return Transpose(operand.elements());
>-  }
>-
>-  template <typename S>
>-  Matrix& Scale(const S& scalar) {
>-    for (size_t i = 0; i < data_.size(); ++i) {
>-      data_[i] *= scalar;
>-    }
>-
>-    return *this;
>-  }
>-
>-  template <typename S>
>-  Matrix& Scale(const Matrix& operand, const S& scalar) {
>-    CopyFrom(operand);
>-    return Scale(scalar);
>-  }
>-
>-  Matrix& Add(const Matrix& operand) {
>-    RTC_CHECK_EQ(num_rows_, operand.num_rows_);
>-    RTC_CHECK_EQ(num_columns_, operand.num_columns_);
>-
>-    for (size_t i = 0; i < data_.size(); ++i) {
>-      data_[i] += operand.data_[i];
>-    }
>-
>-    return *this;
>-  }
>-
>-  Matrix& Add(const Matrix& lhs, const Matrix& rhs) {
>-    CopyFrom(lhs);
>-    return Add(rhs);
>-  }
>-
>-  Matrix& Subtract(const Matrix& operand) {
>-    RTC_CHECK_EQ(num_rows_, operand.num_rows_);
>-    RTC_CHECK_EQ(num_columns_, operand.num_columns_);
>-
>-    for (size_t i = 0; i < data_.size(); ++i) {
>-      data_[i] -= operand.data_[i];
>-    }
>-
>-    return *this;
>-  }
>-
>-  Matrix& Subtract(const Matrix& lhs, const Matrix& rhs) {
>-    CopyFrom(lhs);
>-    return Subtract(rhs);
>-  }
>-
>-  Matrix& PointwiseMultiply(const Matrix& operand) {
>-    RTC_CHECK_EQ(num_rows_, operand.num_rows_);
>-    RTC_CHECK_EQ(num_columns_, operand.num_columns_);
>-
>-    for (size_t i = 0; i < data_.size(); ++i) {
>-      data_[i] *= operand.data_[i];
>-    }
>-
>-    return *this;
>-  }
>-
>-  Matrix& PointwiseMultiply(const Matrix& lhs, const Matrix& rhs) {
>-    CopyFrom(lhs);
>-    return PointwiseMultiply(rhs);
>-  }
>-
>-  Matrix& PointwiseDivide(const Matrix& operand) {
>-    RTC_CHECK_EQ(num_rows_, operand.num_rows_);
>-    RTC_CHECK_EQ(num_columns_, operand.num_columns_);
>-
>-    for (size_t i = 0; i < data_.size(); ++i) {
>-      data_[i] /= operand.data_[i];
>-    }
>-
>-    return *this;
>-  }
>-
>-  Matrix& PointwiseDivide(const Matrix& lhs, const Matrix& rhs) {
>-    CopyFrom(lhs);
>-    return PointwiseDivide(rhs);
>-  }
>-
>-  Matrix& PointwiseSquareRoot() {
>-    for (size_t i = 0; i < data_.size(); ++i) {
>-      data_[i] = sqrt_wrapper(data_[i]);
>-    }
>-
>-    return *this;
>-  }
>-
>-  Matrix& PointwiseSquareRoot(const Matrix& operand) {
>-    CopyFrom(operand);
>-    return PointwiseSquareRoot();
>-  }
>-
>-  Matrix& PointwiseAbsoluteValue() {
>-    for (size_t i = 0; i < data_.size(); ++i) {
>-      data_[i] = abs(data_[i]);
>-    }
>-
>-    return *this;
>-  }
>-
>-  Matrix& PointwiseAbsoluteValue(const Matrix& operand) {
>-    CopyFrom(operand);
>-    return PointwiseAbsoluteValue();
>-  }
>-
>-  Matrix& PointwiseSquare() {
>-    for (size_t i = 0; i < data_.size(); ++i) {
>-      data_[i] *= data_[i];
>-    }
>-
>-    return *this;
>-  }
>-
>-  Matrix& PointwiseSquare(const Matrix& operand) {
>-    CopyFrom(operand);
>-    return PointwiseSquare();
>-  }
>-
>-  Matrix& Multiply(const Matrix& lhs, const Matrix& rhs) {
>-    RTC_CHECK_EQ(lhs.num_columns_, rhs.num_rows_);
>-    RTC_CHECK_EQ(num_rows_, lhs.num_rows_);
>-    RTC_CHECK_EQ(num_columns_, rhs.num_columns_);
>-
>-    return Multiply(lhs.elements(), rhs.num_rows_, rhs.elements());
>-  }
>-
>-  Matrix& Multiply(const Matrix& rhs) {
>-    RTC_CHECK_EQ(num_columns_, rhs.num_rows_);
>-
>-    CopyDataToScratch();
>-    Resize(num_rows_, rhs.num_columns_);
>-    return Multiply(scratch_elements(), rhs.num_rows_, rhs.elements());
>-  }
>-
>-  std::string ToString() const {
>-    std::ostringstream ss;
>-    ss << std::endl << "Matrix" << std::endl;
>-
>-    for (size_t i = 0; i < num_rows_; ++i) {
>-      for (size_t j = 0; j < num_columns_; ++j) {
>-        ss << elements_[i][j] << " ";
>-      }
>-      ss << std::endl;
>-    }
>-    ss << std::endl;
>-
>-    return ss.str();
>-  }
>-
>- protected:
>-  void SetNumRows(const size_t num_rows) { num_rows_ = num_rows; }
>-  void SetNumColumns(const size_t num_columns) { num_columns_ = num_columns; }
>-  T* data() { return &data_[0]; }
>-  const T* data() const { return &data_[0]; }
>-  const T* const* scratch_elements() const { return &scratch_elements_[0]; }
>-
>-  // Resize the matrix. If an increase in capacity is required, the current
>-  // data is lost.
>-  void Resize() {
>-    size_t size = num_rows_ * num_columns_;
>-    data_.resize(size);
>-    elements_.resize(num_rows_);
>-
>-    for (size_t i = 0; i < num_rows_; ++i) {
>-      elements_[i] = &data_[i * num_columns_];
>-    }
>-  }
>-
>-  // Copies data_ into scratch_data_ and updates scratch_elements_ accordingly.
>-  void CopyDataToScratch() {
>-    scratch_data_ = data_;
>-    scratch_elements_.resize(num_rows_);
>-
>-    for (size_t i = 0; i < num_rows_; ++i) {
>-      scratch_elements_[i] = &scratch_data_[i * num_columns_];
>-    }
>-  }
>-
>- private:
>-  size_t num_rows_;
>-  size_t num_columns_;
>-  std::vector<T> data_;
>-  std::vector<T*> elements_;
>-
>-  // Stores temporary copies of |data_| and |elements_| for in-place operations
>-  // where referring to original data is necessary.
>-  std::vector<T> scratch_data_;
>-  std::vector<T*> scratch_elements_;
>-
>-  // Helpers for Transpose and Multiply operations that unify in-place and
>-  // out-of-place solutions.
>-  Matrix& Transpose(const T* const* src) {
>-    for (size_t i = 0; i < num_rows_; ++i) {
>-      for (size_t j = 0; j < num_columns_; ++j) {
>-        elements_[i][j] = src[j][i];
>-      }
>-    }
>-
>-    return *this;
>-  }
>-
>-  Matrix& Multiply(const T* const* lhs,
>-                   size_t num_rows_rhs,
>-                   const T* const* rhs) {
>-    for (size_t row = 0; row < num_rows_; ++row) {
>-      for (size_t col = 0; col < num_columns_; ++col) {
>-        T cur_element = 0;
>-        for (size_t i = 0; i < num_rows_rhs; ++i) {
>-          cur_element += lhs[row][i] * rhs[i][col];
>-        }
>-
>-        elements_[row][col] = cur_element;
>-      }
>-    }
>-
>-    return *this;
>-  }
>-
>-  RTC_DISALLOW_COPY_AND_ASSIGN(Matrix);
>-};
>-
>-}  // namespace webrtc
>-
>-#endif  // MODULES_AUDIO_PROCESSING_BEAMFORMER_MATRIX_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/matrix_test_helpers.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/matrix_test_helpers.h
>deleted file mode 100644
>index 62b47082e36..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/matrix_test_helpers.h
>+++ /dev/null
>@@ -1,102 +0,0 @@
>-/*
>- *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MODULES_AUDIO_PROCESSING_BEAMFORMER_MATRIX_TEST_HELPERS_H_
>-#define MODULES_AUDIO_PROCESSING_BEAMFORMER_MATRIX_TEST_HELPERS_H_
>-
>-#include "modules/audio_processing/beamformer/complex_matrix.h"
>-#include "modules/audio_processing/beamformer/matrix.h"
>-#include "test/gtest.h"
>-
>-namespace {
>-const float kTolerance = 0.001f;
>-}
>-
>-namespace webrtc {
>-
>-using std::complex;
>-
>-// Functions used in both matrix_unittest and complex_matrix_unittest.
>-class MatrixTestHelpers {
>- public:
>-  template <typename T>
>-  static void ValidateMatrixEquality(const Matrix<T>& expected,
>-                                     const Matrix<T>& actual) {
>-    EXPECT_EQ(expected.num_rows(), actual.num_rows());
>-    EXPECT_EQ(expected.num_columns(), actual.num_columns());
>-
>-    const T* const* expected_elements = expected.elements();
>-    const T* const* actual_elements = actual.elements();
>-    for (size_t i = 0; i < expected.num_rows(); ++i) {
>-      for (size_t j = 0; j < expected.num_columns(); ++j) {
>-        EXPECT_EQ(expected_elements[i][j], actual_elements[i][j]);
>-      }
>-    }
>-  }
>-
>-  static void ValidateMatrixEqualityFloat(const Matrix<float>& expected,
>-                                          const Matrix<float>& actual) {
>-    EXPECT_EQ(expected.num_rows(), actual.num_rows());
>-    EXPECT_EQ(expected.num_columns(), actual.num_columns());
>-
>-    const float* const* expected_elements = expected.elements();
>-    const float* const* actual_elements = actual.elements();
>-    for (size_t i = 0; i < expected.num_rows(); ++i) {
>-      for (size_t j = 0; j < expected.num_columns(); ++j) {
>-        EXPECT_NEAR(expected_elements[i][j], actual_elements[i][j], kTolerance);
>-      }
>-    }
>-  }
>-
>-  static void ValidateMatrixEqualityComplexFloat(
>-      const Matrix<complex<float> >& expected,
>-      const Matrix<complex<float> >& actual) {
>-    EXPECT_EQ(expected.num_rows(), actual.num_rows());
>-    EXPECT_EQ(expected.num_columns(), actual.num_columns());
>-
>-    const complex<float>* const* expected_elements = expected.elements();
>-    const complex<float>* const* actual_elements = actual.elements();
>-    for (size_t i = 0; i < expected.num_rows(); ++i) {
>-      for (size_t j = 0; j < expected.num_columns(); ++j) {
>-        EXPECT_NEAR(expected_elements[i][j].real(),
>-                    actual_elements[i][j].real(),
>-                    kTolerance);
>-        EXPECT_NEAR(expected_elements[i][j].imag(),
>-                    actual_elements[i][j].imag(),
>-                    kTolerance);
>-      }
>-    }
>-  }
>-
>-  static void ValidateMatrixNearEqualityComplexFloat(
>-      const Matrix<complex<float> >& expected,
>-      const Matrix<complex<float> >& actual,
>-      float tolerance) {
>-    EXPECT_EQ(expected.num_rows(), actual.num_rows());
>-    EXPECT_EQ(expected.num_columns(), actual.num_columns());
>-
>-    const complex<float>* const* expected_elements = expected.elements();
>-    const complex<float>* const* actual_elements = actual.elements();
>-    for (size_t i = 0; i < expected.num_rows(); ++i) {
>-      for (size_t j = 0; j < expected.num_columns(); ++j) {
>-        EXPECT_NEAR(expected_elements[i][j].real(),
>-                    actual_elements[i][j].real(),
>-                    tolerance);
>-        EXPECT_NEAR(expected_elements[i][j].imag(),
>-                    actual_elements[i][j].imag(),
>-                    tolerance);
>-      }
>-    }
>-  }
>-};
>-
>-}  // namespace webrtc
>-
>-#endif  // MODULES_AUDIO_PROCESSING_BEAMFORMER_MATRIX_TEST_HELPERS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/matrix_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/matrix_unittest.cc
>deleted file mode 100644
>index 4badfd07ed6..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/matrix_unittest.cc
>+++ /dev/null
>@@ -1,326 +0,0 @@
>-/*
>- *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include <complex>
>-
>-#include "modules/audio_processing/beamformer/matrix.h"
>-#include "modules/audio_processing/beamformer/matrix_test_helpers.h"
>-#include "test/gtest.h"
>-
>-namespace webrtc {
>-
>-using std::complex;
>-
>-TEST(MatrixTest, TestMultiplySameSize) {
>-  const int kNumRows = 2;
>-  const int kNumCols = 2;
>-  const float kValuesLeft[kNumRows][kNumCols] = {{1.1f, 2.2f}, {3.3f, 4.4f}};
>-  const float kValuesRight[kNumRows][kNumCols] = {{5.4f, 127.f},
>-                                                  {4600.f, -555.f}};
>-  const float kValuesExpected[kNumRows][kNumCols] = {{10125.94f, -1081.3f},
>-                                                     {20257.82f, -2022.9f}};
>-
>-  Matrix<float> lh_mat(*kValuesLeft, kNumRows, kNumCols);
>-  Matrix<float> rh_mat(*kValuesRight, kNumRows, kNumCols);
>-  Matrix<float> expected_result(*kValuesExpected, kNumRows, kNumCols);
>-  Matrix<float> actual_result(kNumRows, kNumCols);
>-
>-  actual_result.Multiply(lh_mat, rh_mat);
>-  MatrixTestHelpers::ValidateMatrixEquality(expected_result, actual_result);
>-
>-  lh_mat.Multiply(rh_mat);
>-  MatrixTestHelpers::ValidateMatrixEquality(lh_mat, actual_result);
>-}
>-
>-TEST(MatrixTest, TestMultiplyDifferentSize) {
>-  const int kNumRowsLeft = 2;
>-  const int kNumColsLeft = 3;
>-  const int kNumRowsRight = 3;
>-  const int kNumColsRight = 2;
>-  const int kValuesLeft[kNumRowsLeft][kNumColsLeft] = {{35, 466, -15},
>-                                                       {-3, 3422, 9}};
>-  const int kValuesRight[kNumRowsRight][kNumColsRight] = {
>-      {765, -42}, {0, 194}, {625, 66321}};
>-  const int kValuesExpected[kNumRowsLeft][kNumColsRight] = {{17400, -905881},
>-                                                            {3330, 1260883}};
>-
>-  Matrix<int> lh_mat(*kValuesLeft, kNumRowsLeft, kNumColsLeft);
>-  Matrix<int> rh_mat(*kValuesRight, kNumRowsRight, kNumColsRight);
>-  Matrix<int> expected_result(*kValuesExpected, kNumRowsLeft, kNumColsRight);
>-  Matrix<int> actual_result(kNumRowsLeft, kNumColsRight);
>-
>-  actual_result.Multiply(lh_mat, rh_mat);
>-  MatrixTestHelpers::ValidateMatrixEquality(expected_result, actual_result);
>-
>-  lh_mat.Multiply(rh_mat);
>-  MatrixTestHelpers::ValidateMatrixEquality(lh_mat, actual_result);
>-}
>-
>-TEST(MatrixTest, TestTranspose) {
>-  const int kNumInitialRows = 2;
>-  const int kNumInitialCols = 4;
>-  const int kNumResultRows = 4;
>-  const int kNumResultCols = 2;
>-  const float kValuesInitial[kNumInitialRows][kNumInitialCols] = {
>-      {1.1f, 2.2f, 3.3f, 4.4f}, {5.5f, 6.6f, 7.7f, 8.8f}};
>-  const float kValuesExpected[kNumResultRows][kNumResultCols] = {
>-      {1.1f, 5.5f}, {2.2f, 6.6f}, {3.3f, 7.7f}, {4.4f, 8.8f}};
>-
>-  Matrix<float> initial_mat(*kValuesInitial, kNumInitialRows, kNumInitialCols);
>-  Matrix<float> expected_result(
>-      *kValuesExpected, kNumResultRows, kNumResultCols);
>-  Matrix<float> actual_result(kNumResultRows, kNumResultCols);
>-
>-  actual_result.Transpose(initial_mat);
>-  MatrixTestHelpers::ValidateMatrixEqualityFloat(expected_result,
>-                                                 actual_result);
>-  initial_mat.Transpose();
>-  MatrixTestHelpers::ValidateMatrixEqualityFloat(initial_mat, actual_result);
>-}
>-
>-TEST(MatrixTest, TestScale) {
>-  const int kNumRows = 3;
>-  const int kNumCols = 3;
>-  const int kScaleFactor = -9;
>-  const int kValuesInitial[kNumRows][kNumCols] = {
>-      {1, 20, 5000}, {-3, -29, 66}, {7654, 0, -23455}};
>-  const int kValuesExpected[kNumRows][kNumCols] = {
>-      {-9, -180, -45000}, {27, 261, -594}, {-68886, 0, 211095}};
>-
>-  Matrix<int> initial_mat(*kValuesInitial, kNumRows, kNumCols);
>-  Matrix<int> expected_result(*kValuesExpected, kNumRows, kNumCols);
>-  Matrix<int> actual_result;
>-
>-  actual_result.Scale(initial_mat, kScaleFactor);
>-  MatrixTestHelpers::ValidateMatrixEquality(expected_result, actual_result);
>-
>-  initial_mat.Scale(kScaleFactor);
>-  MatrixTestHelpers::ValidateMatrixEquality(initial_mat, actual_result);
>-}
>-
>-TEST(MatrixTest, TestPointwiseAdd) {
>-  const int kNumRows = 2;
>-  const int kNumCols = 3;
>-  const float kValuesLeft[kNumRows][kNumCols] = {{1.1f, 210.45f, -549.2f},
>-                                                 {11.876f, 586.7f, -64.35f}};
>-  const float kValuesRight[kNumRows][kNumCols] = {{-50.4f, 1.f, 0.5f},
>-                                                  {460.f, -554.2f, 4566.f}};
>-  const float kValuesExpected[kNumRows][kNumCols] = {
>-      {-49.3f, 211.45f, -548.7f}, {471.876f, 32.5f, 4501.65f}};
>-
>-  Matrix<float> lh_mat(*kValuesLeft, kNumRows, kNumCols);
>-  Matrix<float> rh_mat(*kValuesRight, kNumRows, kNumCols);
>-  Matrix<float> expected_result(*kValuesExpected, kNumRows, kNumCols);
>-  Matrix<float> actual_result;
>-
>-  actual_result.Add(lh_mat, rh_mat);
>-  MatrixTestHelpers::ValidateMatrixEqualityFloat(expected_result,
>-                                                 actual_result);
>-  lh_mat.Add(rh_mat);
>-  MatrixTestHelpers::ValidateMatrixEqualityFloat(lh_mat, actual_result);
>-}
>-
>-TEST(MatrixTest, TestPointwiseSubtract) {
>-  const int kNumRows = 3;
>-  const int kNumCols = 2;
>-  const float kValuesLeft[kNumRows][kNumCols] = {
>-      {1.1f, 210.45f}, {-549.2f, 11.876f}, {586.7f, -64.35f}};
>-  const float kValuesRight[kNumRows][kNumCols] = {
>-      {-50.4f, 1.f}, {0.5f, 460.f}, {-554.2f, 4566.f}};
>-  const float kValuesExpected[kNumRows][kNumCols] = {
>-      {51.5f, 209.45f}, {-549.7f, -448.124f}, {1140.9f, -4630.35f}};
>-
>-  Matrix<float> lh_mat(*kValuesLeft, kNumRows, kNumCols);
>-  Matrix<float> rh_mat(*kValuesRight, kNumRows, kNumCols);
>-  Matrix<float> expected_result(*kValuesExpected, kNumRows, kNumCols);
>-  Matrix<float> actual_result;
>-
>-  actual_result.Subtract(lh_mat, rh_mat);
>-  MatrixTestHelpers::ValidateMatrixEqualityFloat(expected_result,
>-                                                 actual_result);
>-
>-  lh_mat.Subtract(rh_mat);
>-  MatrixTestHelpers::ValidateMatrixEqualityFloat(lh_mat, actual_result);
>-}
>-
>-TEST(MatrixTest, TestPointwiseMultiply) {
>-  const int kNumRows = 1;
>-  const int kNumCols = 5;
>-  const float kValuesLeft[kNumRows][kNumCols] = {
>-      {1.1f, 6.4f, 0.f, -1.f, -88.3f}};
>-  const float kValuesRight[kNumRows][kNumCols] = {
>-      {53.2f, -210.45f, -549.2f, 99.99f, -45.2f}};
>-  const float kValuesExpected[kNumRows][kNumCols] = {
>-      {58.52f, -1346.88f, 0.f, -99.99f, 3991.16f}};
>-
>-  Matrix<float> lh_mat(*kValuesLeft, kNumRows, kNumCols);
>-  Matrix<float> rh_mat(*kValuesRight, kNumRows, kNumCols);
>-  Matrix<float> expected_result(*kValuesExpected, kNumRows, kNumCols);
>-  Matrix<float> actual_result;
>-
>-  actual_result.PointwiseMultiply(lh_mat, rh_mat);
>-  MatrixTestHelpers::ValidateMatrixEqualityFloat(expected_result,
>-                                                 actual_result);
>-
>-  lh_mat.PointwiseMultiply(rh_mat);
>-  MatrixTestHelpers::ValidateMatrixEqualityFloat(lh_mat, actual_result);
>-}
>-
>-TEST(MatrixTest, TestPointwiseDivide) {
>-  const int kNumRows = 5;
>-  const int kNumCols = 1;
>-  const float kValuesLeft[kNumRows][kNumCols] = {
>-      {1.1f}, {6.4f}, {0.f}, {-1.f}, {-88.3f}};
>-  const float kValuesRight[kNumRows][kNumCols] = {
>-      {53.2f}, {-210.45f}, {-549.2f}, {99.99f}, {-45.2f}};
>-  const float kValuesExpected[kNumRows][kNumCols] = {
>-      {0.020676691f}, {-0.03041102399f}, {0.f}, {-0.010001f}, {1.9535398f}};
>-
>-  Matrix<float> lh_mat(*kValuesLeft, kNumRows, kNumCols);
>-  Matrix<float> rh_mat(*kValuesRight, kNumRows, kNumCols);
>-  Matrix<float> expected_result(*kValuesExpected, kNumRows, kNumCols);
>-  Matrix<float> actual_result;
>-
>-  actual_result.PointwiseDivide(lh_mat, rh_mat);
>-  MatrixTestHelpers::ValidateMatrixEqualityFloat(expected_result,
>-                                                 actual_result);
>-
>-  lh_mat.PointwiseDivide(rh_mat);
>-  MatrixTestHelpers::ValidateMatrixEqualityFloat(lh_mat, actual_result);
>-}
>-
>-TEST(MatrixTest, TestPointwiseSquareRoot) {
>-  const int kNumRows = 2;
>-  const int kNumCols = 2;
>-  const int kValues[kNumRows][kNumCols] = {{4, 9}, {16, 0}};
>-  const int kValuesExpected[kNumRows][kNumCols] = {{2, 3}, {4, 0}};
>-
>-  Matrix<int> operand_mat(*kValues, kNumRows, kNumCols);
>-  Matrix<int> expected_result(*kValuesExpected, kNumRows, kNumCols);
>-  Matrix<int> actual_result;
>-
>-  actual_result.PointwiseSquareRoot(operand_mat);
>-  MatrixTestHelpers::ValidateMatrixEquality(expected_result, actual_result);
>-
>-  operand_mat.PointwiseSquareRoot();
>-  MatrixTestHelpers::ValidateMatrixEquality(operand_mat, actual_result);
>-}
>-
>-TEST(MatrixTest, TestPointwiseSquareRootComplex) {
>-  const int kNumRows = 1;
>-  const int kNumCols = 3;
>-  const complex<float> kValues[kNumRows][kNumCols] = {
>-      {complex<float>(-4.f, 0), complex<float>(0, 9), complex<float>(3, -4)}};
>-  const complex<float> kValuesExpected[kNumRows][kNumCols] = {
>-      {complex<float>(0.f, 2.f), complex<float>(2.1213202f, 2.1213202f),
>-       complex<float>(2.f, -1.f)}};
>-
>-  Matrix<complex<float> > operand_mat(*kValues, kNumRows, kNumCols);
>-  Matrix<complex<float> > expected_result(*kValuesExpected, kNumRows, kNumCols);
>-  Matrix<complex<float> > actual_result;
>-
>-  actual_result.PointwiseSquareRoot(operand_mat);
>-  MatrixTestHelpers::ValidateMatrixEqualityComplexFloat(expected_result,
>-                                                        actual_result);
>-
>-  operand_mat.PointwiseSquareRoot();
>-  MatrixTestHelpers::ValidateMatrixEqualityComplexFloat(operand_mat,
>-                                                        actual_result);
>-}
>-
>-TEST(MatrixTest, TestPointwiseAbsoluteValue) {
>-  const int kNumRows = 1;
>-  const int kNumCols = 3;
>-  const complex<float> kValues[kNumRows][kNumCols] = {
>-      {complex<float>(-4.f, 0), complex<float>(0, 9), complex<float>(3, -4)}};
>-  const complex<float> kValuesExpected[kNumRows][kNumCols] = {
>-      {complex<float>(4.f, 0), complex<float>(9.f, 0), complex<float>(5.f, 0)}};
>-
>-  Matrix<complex<float> > operand_mat(*kValues, kNumRows, kNumCols);
>-  Matrix<complex<float> > expected_result(*kValuesExpected, kNumRows, kNumCols);
>-  Matrix<complex<float> > actual_result;
>-
>-  actual_result.PointwiseAbsoluteValue(operand_mat);
>-  MatrixTestHelpers::ValidateMatrixEqualityComplexFloat(expected_result,
>-                                                        actual_result);
>-
>-  operand_mat.PointwiseAbsoluteValue();
>-  MatrixTestHelpers::ValidateMatrixEqualityComplexFloat(operand_mat,
>-                                                        actual_result);
>-}
>-
>-TEST(MatrixTest, TestPointwiseSquare) {
>-  const int kNumRows = 1;
>-  const int kNumCols = 3;
>-  const float kValues[kNumRows][kNumCols] = {{2.4f, -4.f, 3.3f}};
>-  const float kValuesExpected[kNumRows][kNumCols] = {{5.76f, 16.f, 10.89f}};
>-
>-  Matrix<float> operand_mat(*kValues, kNumRows, kNumCols);
>-  Matrix<float> expected_result(*kValuesExpected, kNumRows, kNumCols);
>-  Matrix<float> actual_result;
>-
>-  actual_result.PointwiseSquare(operand_mat);
>-  MatrixTestHelpers::ValidateMatrixEqualityFloat(expected_result,
>-                                                 actual_result);
>-
>-  operand_mat.PointwiseSquare();
>-  MatrixTestHelpers::ValidateMatrixEqualityFloat(operand_mat, actual_result);
>-}
>-
>-TEST(MatrixTest, TestComplexOperations) {
>-  const int kNumRows = 2;
>-  const int kNumCols = 2;
>-
>-  const complex<float> kValuesLeft[kNumRows][kNumCols] = {
>-      {complex<float>(1.f, 1.f), complex<float>(2.f, 2.f)},
>-      {complex<float>(3.f, 3.f), complex<float>(4.f, 4.f)}};
>-
>-  const complex<float> kValuesRight[kNumRows][kNumCols] = {
>-      {complex<float>(5.f, 5.f), complex<float>(6.f, 6.f)},
>-      {complex<float>(7.f, 7.f), complex<float>(8.f, 8.f)}};
>-
>-  const complex<float> kValuesExpectedAdd[kNumRows][kNumCols] = {
>-      {complex<float>(6.f, 6.f), complex<float>(8.f, 8.f)},
>-      {complex<float>(10.f, 10.f), complex<float>(12.f, 12.f)}};
>-
>-  const complex<float> kValuesExpectedMultiply[kNumRows][kNumCols] = {
>-      {complex<float>(0.f, 38.f), complex<float>(0.f, 44.f)},
>-      {complex<float>(0.f, 86.f), complex<float>(0.f, 100.f)}};
>-
>-  const complex<float> kValuesExpectedPointwiseDivide[kNumRows][kNumCols] = {
>-      {complex<float>(0.2f, 0.f), complex<float>(0.33333333f, 0.f)},
>-      {complex<float>(0.42857143f, 0.f), complex<float>(0.5f, 0.f)}};
>-
>-  Matrix<complex<float> > lh_mat(*kValuesLeft, kNumRows, kNumCols);
>-  Matrix<complex<float> > rh_mat(*kValuesRight, kNumRows, kNumCols);
>-  Matrix<complex<float> > expected_result_add(
>-      *kValuesExpectedAdd, kNumRows, kNumCols);
>-  Matrix<complex<float> > expected_result_multiply(
>-      *kValuesExpectedMultiply, kNumRows, kNumCols);
>-  Matrix<complex<float> > expected_result_pointwise_divide(
>-      *kValuesExpectedPointwiseDivide, kNumRows, kNumCols);
>-  Matrix<complex<float> > actual_result_add;
>-  Matrix<complex<float> > actual_result_multiply(kNumRows, kNumCols);
>-  Matrix<complex<float> > actual_result_pointwise_divide;
>-
>-  actual_result_add.Add(lh_mat, rh_mat);
>-  MatrixTestHelpers::ValidateMatrixEqualityComplexFloat(expected_result_add,
>-                                                        actual_result_add);
>-
>-  actual_result_multiply.Multiply(lh_mat, rh_mat);
>-  MatrixTestHelpers::ValidateMatrixEqualityComplexFloat(
>-      expected_result_multiply, actual_result_multiply);
>-
>-  actual_result_pointwise_divide.PointwiseDivide(lh_mat, rh_mat);
>-  MatrixTestHelpers::ValidateMatrixEqualityComplexFloat(
>-      expected_result_pointwise_divide, actual_result_pointwise_divide);
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/mock_nonlinear_beamformer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/mock_nonlinear_beamformer.h
>deleted file mode 100644
>index c4c7358b631..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/mock_nonlinear_beamformer.h
>+++ /dev/null
>@@ -1,39 +0,0 @@
>-/*
>- *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MODULES_AUDIO_PROCESSING_BEAMFORMER_MOCK_BEAMFORMER_H_
>-#define MODULES_AUDIO_PROCESSING_BEAMFORMER_MOCK_BEAMFORMER_H_
>-
>-#include <vector>
>-
>-#include "modules/audio_processing/beamformer/nonlinear_beamformer.h"
>-#include "test/gmock.h"
>-
>-namespace webrtc {
>-
>-class MockNonlinearBeamformer : public NonlinearBeamformer {
>- public:
>-  MockNonlinearBeamformer(const std::vector<Point>& array_geometry,
>-                          size_t num_postfilter_channels)
>-      : NonlinearBeamformer(array_geometry, num_postfilter_channels) {}
>-
>-  MockNonlinearBeamformer(const std::vector<Point>& array_geometry)
>-      : NonlinearBeamformer(array_geometry, 1u) {}
>-
>-  MOCK_METHOD2(Initialize, void(int chunk_size_ms, int sample_rate_hz));
>-  MOCK_METHOD1(AnalyzeChunk, void(const ChannelBuffer<float>& data));
>-  MOCK_METHOD1(PostFilter, void(ChannelBuffer<float>* data));
>-  MOCK_METHOD1(IsInBeam, bool(const SphericalPointf& spherical_point));
>-  MOCK_METHOD0(is_target_present, bool());
>-};
>-
>-}  // namespace webrtc
>-
>-#endif  // MODULES_AUDIO_PROCESSING_BEAMFORMER_MOCK_BEAMFORMER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.cc
>deleted file mode 100644
>index 12f6d2f4652..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.cc
>+++ /dev/null
>@@ -1,599 +0,0 @@
>-/*
>- *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#define _USE_MATH_DEFINES
>-
>-#include "modules/audio_processing/beamformer/nonlinear_beamformer.h"
>-
>-#include <algorithm>
>-#include <cmath>
>-#include <numeric>
>-#include <vector>
>-
>-#include "common_audio/window_generator.h"
>-#include "modules/audio_processing/beamformer/covariance_matrix_generator.h"
>-#include "rtc_base/arraysize.h"
>-
>-namespace webrtc {
>-namespace {
>-
>-// Alpha for the Kaiser Bessel Derived window.
>-const float kKbdAlpha = 1.5f;
>-
>-const float kSpeedOfSoundMeterSeconds = 343;
>-
>-// The minimum separation in radians between the target direction and an
>-// interferer scenario.
>-const float kMinAwayRadians = 0.2f;
>-
>-// The separation between the target direction and the closest interferer
>-// scenario is proportional to this constant.
>-const float kAwaySlope = 0.008f;
>-
>-// When calculating the interference covariance matrix, this is the weight for
>-// the weighted average between the uniform covariance matrix and the angled
>-// covariance matrix.
>-// Rpsi = Rpsi_angled * kBalance + Rpsi_uniform * (1 - kBalance)
>-const float kBalance = 0.95f;
>-
>-// Alpha coefficients for mask smoothing.
>-const float kMaskTimeSmoothAlpha = 0.2f;
>-const float kMaskFrequencySmoothAlpha = 0.6f;
>-
>-// The average mask is computed from masks in this mid-frequency range. If these
>-// ranges are changed |kMaskQuantile| might need to be adjusted.
>-const int kLowMeanStartHz = 200;
>-const int kLowMeanEndHz = 400;
>-
>-// Range limiter for subtractive terms in the nominator and denominator of the
>-// postfilter expression. It handles the scenario mismatch between the true and
>-// model sources (target and interference).
>-const float kCutOffConstant = 0.9999f;
>-
>-// Quantile of mask values which is used to estimate target presence.
>-const float kMaskQuantile = 0.7f;
>-// Mask threshold over which the data is considered signal and not interference.
>-// It has to be updated every time the postfilter calculation is changed
>-// significantly.
>-// TODO(aluebs): Write a tool to tune the target threshold automatically based
>-// on files annotated with target and interference ground truth.
>-const float kMaskTargetThreshold = 0.01f;
>-// Time in seconds after which the data is considered interference if the mask
>-// does not pass |kMaskTargetThreshold|.
>-const float kHoldTargetSeconds = 0.25f;
>-
>-// To compensate for the attenuation this algorithm introduces to the target
>-// signal. It was estimated empirically from a low-noise low-reverberation
>-// recording from broadside.
>-const float kCompensationGain = 2.f;
>-
>-// Does conjugate(|norm_mat|) * |mat| * transpose(|norm_mat|). No extra space is
>-// used; to accomplish this, we compute both multiplications in the same loop.
>-// The returned norm is clamped to be non-negative.
>-float Norm(const ComplexMatrix<float>& mat,
>-           const ComplexMatrix<float>& norm_mat) {
>-  RTC_CHECK_EQ(1, norm_mat.num_rows());
>-  RTC_CHECK_EQ(norm_mat.num_columns(), mat.num_rows());
>-  RTC_CHECK_EQ(norm_mat.num_columns(), mat.num_columns());
>-
>-  complex<float> first_product = complex<float>(0.f, 0.f);
>-  complex<float> second_product = complex<float>(0.f, 0.f);
>-
>-  const complex<float>* const* mat_els = mat.elements();
>-  const complex<float>* const* norm_mat_els = norm_mat.elements();
>-
>-  for (size_t i = 0; i < norm_mat.num_columns(); ++i) {
>-    for (size_t j = 0; j < norm_mat.num_columns(); ++j) {
>-      first_product += conj(norm_mat_els[0][j]) * mat_els[j][i];
>-    }
>-    second_product += first_product * norm_mat_els[0][i];
>-    first_product = 0.f;
>-  }
>-  return std::max(second_product.real(), 0.f);
>-}
>-
>-// Does conjugate(|lhs|) * |rhs| for row vectors |lhs| and |rhs|.
>-complex<float> ConjugateDotProduct(const ComplexMatrix<float>& lhs,
>-                                   const ComplexMatrix<float>& rhs) {
>-  RTC_CHECK_EQ(1, lhs.num_rows());
>-  RTC_CHECK_EQ(1, rhs.num_rows());
>-  RTC_CHECK_EQ(lhs.num_columns(), rhs.num_columns());
>-
>-  const complex<float>* const* lhs_elements = lhs.elements();
>-  const complex<float>* const* rhs_elements = rhs.elements();
>-
>-  complex<float> result = complex<float>(0.f, 0.f);
>-  for (size_t i = 0; i < lhs.num_columns(); ++i) {
>-    result += conj(lhs_elements[0][i]) * rhs_elements[0][i];
>-  }
>-
>-  return result;
>-}
>-
>-// Works for positive numbers only.
>-size_t Round(float x) {
>-  return static_cast<size_t>(std::floor(x + 0.5f));
>-}
>-
>-// Calculates the sum of squares of a complex matrix.
>-float SumSquares(const ComplexMatrix<float>& mat) {
>-  float sum_squares = 0.f;
>-  const complex<float>* const* mat_els = mat.elements();
>-  for (size_t i = 0; i < mat.num_rows(); ++i) {
>-    for (size_t j = 0; j < mat.num_columns(); ++j) {
>-      float abs_value = std::abs(mat_els[i][j]);
>-      sum_squares += abs_value * abs_value;
>-    }
>-  }
>-  return sum_squares;
>-}
>-
>-// Does |out| = |in|.' * conj(|in|) for row vector |in|.
>-void TransposedConjugatedProduct(const ComplexMatrix<float>& in,
>-                                 ComplexMatrix<float>* out) {
>-  RTC_CHECK_EQ(1, in.num_rows());
>-  RTC_CHECK_EQ(out->num_rows(), in.num_columns());
>-  RTC_CHECK_EQ(out->num_columns(), in.num_columns());
>-  const complex<float>* in_elements = in.elements()[0];
>-  complex<float>* const* out_elements = out->elements();
>-  for (size_t i = 0; i < out->num_rows(); ++i) {
>-    for (size_t j = 0; j < out->num_columns(); ++j) {
>-      out_elements[i][j] = in_elements[i] * conj(in_elements[j]);
>-    }
>-  }
>-}
>-
>-std::vector<Point> GetCenteredArray(std::vector<Point> array_geometry) {
>-  for (size_t dim = 0; dim < 3; ++dim) {
>-    float center = 0.f;
>-    for (size_t i = 0; i < array_geometry.size(); ++i) {
>-      center += array_geometry[i].c[dim];
>-    }
>-    center /= array_geometry.size();
>-    for (size_t i = 0; i < array_geometry.size(); ++i) {
>-      array_geometry[i].c[dim] -= center;
>-    }
>-  }
>-  return array_geometry;
>-}
>-
>-}  // namespace
>-
>-const float NonlinearBeamformer::kHalfBeamWidthRadians = DegreesToRadians(20.f);
>-
>-// static
>-const size_t NonlinearBeamformer::kNumFreqBins;
>-
>-PostFilterTransform::PostFilterTransform(size_t num_channels,
>-                                         size_t chunk_length,
>-                                         float* window,
>-                                         size_t fft_size)
>-    : transform_(num_channels,
>-                 num_channels,
>-                 chunk_length,
>-                 window,
>-                 fft_size,
>-                 fft_size / 2,
>-                 this),
>-      num_freq_bins_(fft_size / 2 + 1) {}
>-
>-void PostFilterTransform::ProcessChunk(float* const* data, float* final_mask) {
>-  final_mask_ = final_mask;
>-  transform_.ProcessChunk(data, data);
>-}
>-
>-void PostFilterTransform::ProcessAudioBlock(const complex<float>* const* input,
>-                                            size_t num_input_channels,
>-                                            size_t num_freq_bins,
>-                                            size_t num_output_channels,
>-                                            complex<float>* const* output) {
>-  RTC_DCHECK_EQ(num_freq_bins_, num_freq_bins);
>-  RTC_DCHECK_EQ(num_input_channels, num_output_channels);
>-
>-  for (size_t ch = 0; ch < num_input_channels; ++ch) {
>-    for (size_t f_ix = 0; f_ix < num_freq_bins_; ++f_ix) {
>-      output[ch][f_ix] =
>-          kCompensationGain * final_mask_[f_ix] * input[ch][f_ix];
>-    }
>-  }
>-}
>-
>-NonlinearBeamformer::NonlinearBeamformer(
>-    const std::vector<Point>& array_geometry,
>-    size_t num_postfilter_channels,
>-    SphericalPointf target_direction)
>-    : num_input_channels_(array_geometry.size()),
>-      num_postfilter_channels_(num_postfilter_channels),
>-      array_geometry_(GetCenteredArray(array_geometry)),
>-      array_normal_(GetArrayNormalIfExists(array_geometry)),
>-      min_mic_spacing_(GetMinimumSpacing(array_geometry)),
>-      target_angle_radians_(target_direction.azimuth()),
>-      away_radians_(std::min(
>-          static_cast<float>(M_PI),
>-          std::max(kMinAwayRadians,
>-                   kAwaySlope * static_cast<float>(M_PI) / min_mic_spacing_))) {
>-  WindowGenerator::KaiserBesselDerived(kKbdAlpha, kFftSize, window_);
>-}
>-
>-NonlinearBeamformer::~NonlinearBeamformer() = default;
>-
>-void NonlinearBeamformer::Initialize(int chunk_size_ms, int sample_rate_hz) {
>-  chunk_length_ =
>-      static_cast<size_t>(sample_rate_hz / (1000.f / chunk_size_ms));
>-  sample_rate_hz_ = sample_rate_hz;
>-
>-  high_pass_postfilter_mask_ = 1.f;
>-  is_target_present_ = false;
>-  hold_target_blocks_ = kHoldTargetSeconds * 2 * sample_rate_hz / kFftSize;
>-  interference_blocks_count_ = hold_target_blocks_;
>-
>-  process_transform_.reset(new LappedTransform(num_input_channels_,
>-                                               0u,
>-                                               chunk_length_,
>-                                               window_,
>-                                               kFftSize,
>-                                               kFftSize / 2,
>-                                               this));
>-  postfilter_transform_.reset(new PostFilterTransform(
>-      num_postfilter_channels_, chunk_length_, window_, kFftSize));
>-  const float wave_number_step =
>-      (2.f * M_PI * sample_rate_hz_) / (kFftSize * kSpeedOfSoundMeterSeconds);
>-  for (size_t i = 0; i < kNumFreqBins; ++i) {
>-    time_smooth_mask_[i] = 1.f;
>-    final_mask_[i] = 1.f;
>-    wave_numbers_[i] = i * wave_number_step;
>-  }
>-
>-  InitLowFrequencyCorrectionRanges();
>-  InitDiffuseCovMats();
>-  AimAt(SphericalPointf(target_angle_radians_, 0.f, 1.f));
>-}
>-
>-// These bin indexes determine the regions over which a mean is taken. This is
>-// applied as a constant value over the adjacent end "frequency correction"
>-// regions.
>-//
>-//             low_mean_start_bin_     high_mean_start_bin_
>-//                   v                         v              constant
>-// |----------------|--------|----------------|-------|----------------|
>-//   constant               ^                        ^
>-//             low_mean_end_bin_       high_mean_end_bin_
>-//
>-void NonlinearBeamformer::InitLowFrequencyCorrectionRanges() {
>-  low_mean_start_bin_ = Round(static_cast<float>(kLowMeanStartHz) *
>-                                  kFftSize / sample_rate_hz_);
>-  low_mean_end_bin_ = Round(static_cast<float>(kLowMeanEndHz) *
>-                                  kFftSize / sample_rate_hz_);
>-
>-  RTC_DCHECK_GT(low_mean_start_bin_, 0U);
>-  RTC_DCHECK_LT(low_mean_start_bin_, low_mean_end_bin_);
>-}
>-
>-void NonlinearBeamformer::InitHighFrequencyCorrectionRanges() {
>-  const float kAliasingFreqHz =
>-      kSpeedOfSoundMeterSeconds /
>-      (min_mic_spacing_ * (1.f + std::abs(std::cos(target_angle_radians_))));
>-  const float kHighMeanStartHz = std::min(0.5f *  kAliasingFreqHz,
>-                                          sample_rate_hz_ / 2.f);
>-  const float kHighMeanEndHz = std::min(0.75f *  kAliasingFreqHz,
>-                                        sample_rate_hz_ / 2.f);
>-  high_mean_start_bin_ = Round(kHighMeanStartHz * kFftSize / sample_rate_hz_);
>-  high_mean_end_bin_ = Round(kHighMeanEndHz * kFftSize / sample_rate_hz_);
>-
>-  RTC_DCHECK_LT(low_mean_end_bin_, high_mean_end_bin_);
>-  RTC_DCHECK_LT(high_mean_start_bin_, high_mean_end_bin_);
>-  RTC_DCHECK_LT(high_mean_end_bin_, kNumFreqBins - 1);
>-}
>-
>-void NonlinearBeamformer::InitInterfAngles() {
>-  interf_angles_radians_.clear();
>-  const Point target_direction = AzimuthToPoint(target_angle_radians_);
>-  const Point clockwise_interf_direction =
>-      AzimuthToPoint(target_angle_radians_ - away_radians_);
>-  if (!array_normal_ ||
>-      DotProduct(*array_normal_, target_direction) *
>-              DotProduct(*array_normal_, clockwise_interf_direction) >=
>-          0.f) {
>-    // The target and clockwise interferer are in the same half-plane defined
>-    // by the array.
>-    interf_angles_radians_.push_back(target_angle_radians_ - away_radians_);
>-  } else {
>-    // Otherwise, the interferer will begin reflecting back at the target.
>-    // Instead rotate it away 180 degrees.
>-    interf_angles_radians_.push_back(target_angle_radians_ - away_radians_ +
>-                                     M_PI);
>-  }
>-  const Point counterclock_interf_direction =
>-      AzimuthToPoint(target_angle_radians_ + away_radians_);
>-  if (!array_normal_ ||
>-      DotProduct(*array_normal_, target_direction) *
>-              DotProduct(*array_normal_, counterclock_interf_direction) >=
>-          0.f) {
>-    // The target and counter-clockwise interferer are in the same half-plane
>-    // defined by the array.
>-    interf_angles_radians_.push_back(target_angle_radians_ + away_radians_);
>-  } else {
>-    // Otherwise, the interferer will begin reflecting back at the target.
>-    // Instead rotate it away 180 degrees.
>-    interf_angles_radians_.push_back(target_angle_radians_ + away_radians_ -
>-                                     M_PI);
>-  }
>-}
>-
>-void NonlinearBeamformer::InitDelaySumMasks() {
>-  for (size_t f_ix = 0; f_ix < kNumFreqBins; ++f_ix) {
>-    delay_sum_masks_[f_ix].Resize(1, num_input_channels_);
>-    CovarianceMatrixGenerator::PhaseAlignmentMasks(
>-        f_ix, kFftSize, sample_rate_hz_, kSpeedOfSoundMeterSeconds,
>-        array_geometry_, target_angle_radians_, &delay_sum_masks_[f_ix]);
>-
>-    complex_f norm_factor = sqrt(
>-        ConjugateDotProduct(delay_sum_masks_[f_ix], delay_sum_masks_[f_ix]));
>-    delay_sum_masks_[f_ix].Scale(1.f / norm_factor);
>-  }
>-}
>-
>-void NonlinearBeamformer::InitTargetCovMats() {
>-  for (size_t i = 0; i < kNumFreqBins; ++i) {
>-    target_cov_mats_[i].Resize(num_input_channels_, num_input_channels_);
>-    TransposedConjugatedProduct(delay_sum_masks_[i], &target_cov_mats_[i]);
>-  }
>-}
>-
>-void NonlinearBeamformer::InitDiffuseCovMats() {
>-  for (size_t i = 0; i < kNumFreqBins; ++i) {
>-    uniform_cov_mat_[i].Resize(num_input_channels_, num_input_channels_);
>-    CovarianceMatrixGenerator::UniformCovarianceMatrix(
>-        wave_numbers_[i], array_geometry_, &uniform_cov_mat_[i]);
>-    complex_f normalization_factor = uniform_cov_mat_[i].elements()[0][0];
>-    uniform_cov_mat_[i].Scale(1.f / normalization_factor);
>-    uniform_cov_mat_[i].Scale(1 - kBalance);
>-  }
>-}
>-
>-void NonlinearBeamformer::InitInterfCovMats() {
>-  for (size_t i = 0; i < kNumFreqBins; ++i) {
>-    interf_cov_mats_[i].clear();
>-    for (size_t j = 0; j < interf_angles_radians_.size(); ++j) {
>-      interf_cov_mats_[i].push_back(std::unique_ptr<ComplexMatrixF>(
>-          new ComplexMatrixF(num_input_channels_, num_input_channels_)));
>-      ComplexMatrixF angled_cov_mat(num_input_channels_, num_input_channels_);
>-      CovarianceMatrixGenerator::AngledCovarianceMatrix(
>-          kSpeedOfSoundMeterSeconds,
>-          interf_angles_radians_[j],
>-          i,
>-          kFftSize,
>-          kNumFreqBins,
>-          sample_rate_hz_,
>-          array_geometry_,
>-          &angled_cov_mat);
>-      // Normalize matrices before averaging them.
>-      complex_f normalization_factor = angled_cov_mat.elements()[0][0];
>-      angled_cov_mat.Scale(1.f / normalization_factor);
>-      // Weighted average of matrices.
>-      angled_cov_mat.Scale(kBalance);
>-      interf_cov_mats_[i][j]->Add(uniform_cov_mat_[i], angled_cov_mat);
>-    }
>-  }
>-}
>-
>-void NonlinearBeamformer::NormalizeCovMats() {
>-  for (size_t i = 0; i < kNumFreqBins; ++i) {
>-    rxiws_[i] = Norm(target_cov_mats_[i], delay_sum_masks_[i]);
>-    rpsiws_[i].clear();
>-    for (size_t j = 0; j < interf_angles_radians_.size(); ++j) {
>-      rpsiws_[i].push_back(Norm(*interf_cov_mats_[i][j], delay_sum_masks_[i]));
>-    }
>-  }
>-}
>-
>-void NonlinearBeamformer::AnalyzeChunk(const ChannelBuffer<float>& data) {
>-  RTC_DCHECK_EQ(data.num_channels(), num_input_channels_);
>-  RTC_DCHECK_EQ(data.num_frames_per_band(), chunk_length_);
>-
>-  old_high_pass_mask_ = high_pass_postfilter_mask_;
>-  process_transform_->ProcessChunk(data.channels(0), nullptr);
>-}
>-
>-void NonlinearBeamformer::PostFilter(ChannelBuffer<float>* data) {
>-  RTC_DCHECK_EQ(data->num_frames_per_band(), chunk_length_);
>-  // TODO(aluebs): Change to RTC_CHECK_EQ once the ChannelBuffer is updated.
>-  RTC_DCHECK_GE(data->num_channels(), num_postfilter_channels_);
>-
>-  postfilter_transform_->ProcessChunk(data->channels(0), final_mask_);
>-
>-  // Ramp up/down for smoothing is needed in order to avoid discontinuities in
>-  // the transitions between 10 ms frames.
>-  const float ramp_increment =
>-      (high_pass_postfilter_mask_ - old_high_pass_mask_) /
>-      data->num_frames_per_band();
>-  for (size_t i = 1; i < data->num_bands(); ++i) {
>-    float smoothed_mask = old_high_pass_mask_;
>-    for (size_t j = 0; j < data->num_frames_per_band(); ++j) {
>-      smoothed_mask += ramp_increment;
>-      for (size_t k = 0; k < num_postfilter_channels_; ++k) {
>-        data->channels(i)[k][j] *= smoothed_mask;
>-      }
>-    }
>-  }
>-}
>-
>-void NonlinearBeamformer::AimAt(const SphericalPointf& target_direction) {
>-  target_angle_radians_ = target_direction.azimuth();
>-  InitHighFrequencyCorrectionRanges();
>-  InitInterfAngles();
>-  InitDelaySumMasks();
>-  InitTargetCovMats();
>-  InitInterfCovMats();
>-  NormalizeCovMats();
>-}
>-
>-bool NonlinearBeamformer::IsInBeam(const SphericalPointf& spherical_point) {
>-  // If more than half-beamwidth degrees away from the beam's center,
>-  // you are out of the beam.
>-  return fabs(spherical_point.azimuth() - target_angle_radians_) <
>-         kHalfBeamWidthRadians;
>-}
>-
>-bool NonlinearBeamformer::is_target_present() { return is_target_present_; }
>-
>-void NonlinearBeamformer::ProcessAudioBlock(const complex_f* const* input,
>-                                            size_t num_input_channels,
>-                                            size_t num_freq_bins,
>-                                            size_t num_output_channels,
>-                                            complex_f* const* output) {
>-  RTC_CHECK_EQ(kNumFreqBins, num_freq_bins);
>-  RTC_CHECK_EQ(num_input_channels_, num_input_channels);
>-  RTC_CHECK_EQ(0, num_output_channels);
>-
>-  // Calculating the post-filter masks. Note that we need two for each
>-  // frequency bin to account for the positive and negative interferer
>-  // angle.
>-  for (size_t i = low_mean_start_bin_; i <= high_mean_end_bin_; ++i) {
>-    eig_m_.CopyFromColumn(input, i, num_input_channels_);
>-    float eig_m_norm_factor = std::sqrt(SumSquares(eig_m_));
>-    if (eig_m_norm_factor != 0.f) {
>-      eig_m_.Scale(1.f / eig_m_norm_factor);
>-    }
>-
>-    float rxim = Norm(target_cov_mats_[i], eig_m_);
>-    float ratio_rxiw_rxim = 0.f;
>-    if (rxim > 0.f) {
>-      ratio_rxiw_rxim = rxiws_[i] / rxim;
>-    }
>-
>-    complex_f rmw = abs(ConjugateDotProduct(delay_sum_masks_[i], eig_m_));
>-    rmw *= rmw;
>-    float rmw_r = rmw.real();
>-
>-    new_mask_[i] = CalculatePostfilterMask(*interf_cov_mats_[i][0],
>-                                           rpsiws_[i][0],
>-                                           ratio_rxiw_rxim,
>-                                           rmw_r);
>-    for (size_t j = 1; j < interf_angles_radians_.size(); ++j) {
>-      float tmp_mask = CalculatePostfilterMask(*interf_cov_mats_[i][j],
>-                                               rpsiws_[i][j],
>-                                               ratio_rxiw_rxim,
>-                                               rmw_r);
>-      if (tmp_mask < new_mask_[i]) {
>-        new_mask_[i] = tmp_mask;
>-      }
>-    }
>-  }
>-
>-  ApplyMaskTimeSmoothing();
>-  EstimateTargetPresence();
>-  ApplyLowFrequencyCorrection();
>-  ApplyHighFrequencyCorrection();
>-  ApplyMaskFrequencySmoothing();
>-}
>-
>-float NonlinearBeamformer::CalculatePostfilterMask(
>-    const ComplexMatrixF& interf_cov_mat,
>-    float rpsiw,
>-    float ratio_rxiw_rxim,
>-    float rmw_r) {
>-  float rpsim = Norm(interf_cov_mat, eig_m_);
>-
>-  float ratio = 0.f;
>-  if (rpsim > 0.f) {
>-    ratio = rpsiw / rpsim;
>-  }
>-
>-  float numerator = 1.f - kCutOffConstant;
>-  if (rmw_r > 0.f) {
>-    numerator = 1.f - std::min(kCutOffConstant, ratio / rmw_r);
>-  }
>-
>-  float denominator = 1.f - kCutOffConstant;
>-  if (ratio_rxiw_rxim > 0.f) {
>-    denominator = 1.f - std::min(kCutOffConstant, ratio / ratio_rxiw_rxim);
>-  }
>-
>-  return numerator / denominator;
>-}
>-
>-// Smooth new_mask_ into time_smooth_mask_.
>-void NonlinearBeamformer::ApplyMaskTimeSmoothing() {
>-  for (size_t i = low_mean_start_bin_; i <= high_mean_end_bin_; ++i) {
>-    time_smooth_mask_[i] = kMaskTimeSmoothAlpha * new_mask_[i] +
>-                           (1 - kMaskTimeSmoothAlpha) * time_smooth_mask_[i];
>-  }
>-}
>-
>-// Copy time_smooth_mask_ to final_mask_ and smooth over frequency.
>-void NonlinearBeamformer::ApplyMaskFrequencySmoothing() {
>-  // Smooth over frequency in both directions. The "frequency correction"
>-  // regions have constant value, but we enter them to smooth over the jump
>-  // that exists at the boundary. However, this does mean when smoothing "away"
>-  // from the region that we only need to use the last element.
>-  //
>-  // Upward smoothing:
>-  //   low_mean_start_bin_
>-  //         v
>-  // |------|------------|------|
>-  //       ^------------------>^
>-  //
>-  // Downward smoothing:
>-  //         high_mean_end_bin_
>-  //                    v
>-  // |------|------------|------|
>-  //  ^<------------------^
>-  std::copy(time_smooth_mask_, time_smooth_mask_ + kNumFreqBins, final_mask_);
>-  for (size_t i = low_mean_start_bin_; i < kNumFreqBins; ++i) {
>-    final_mask_[i] = kMaskFrequencySmoothAlpha * final_mask_[i] +
>-                     (1 - kMaskFrequencySmoothAlpha) * final_mask_[i - 1];
>-  }
>-  for (size_t i = high_mean_end_bin_ + 1; i > 0; --i) {
>-    final_mask_[i - 1] = kMaskFrequencySmoothAlpha * final_mask_[i - 1] +
>-                         (1 - kMaskFrequencySmoothAlpha) * final_mask_[i];
>-  }
>-}
>-
>-// Apply low frequency correction to time_smooth_mask_.
>-void NonlinearBeamformer::ApplyLowFrequencyCorrection() {
>-  const float low_frequency_mask =
>-      MaskRangeMean(low_mean_start_bin_, low_mean_end_bin_ + 1);
>-  std::fill(time_smooth_mask_, time_smooth_mask_ + low_mean_start_bin_,
>-            low_frequency_mask);
>-}
>-
>-// Apply high frequency correction to time_smooth_mask_. Update
>-// high_pass_postfilter_mask_ to use for the high frequency time-domain bands.
>-void NonlinearBeamformer::ApplyHighFrequencyCorrection() {
>-  high_pass_postfilter_mask_ =
>-      MaskRangeMean(high_mean_start_bin_, high_mean_end_bin_ + 1);
>-  std::fill(time_smooth_mask_ + high_mean_end_bin_ + 1,
>-            time_smooth_mask_ + kNumFreqBins, high_pass_postfilter_mask_);
>-}
>-
>-// Compute mean over the given range of time_smooth_mask_, [first, last).
>-float NonlinearBeamformer::MaskRangeMean(size_t first, size_t last) {
>-  RTC_DCHECK_GT(last, first);
>-  const float sum = std::accumulate(time_smooth_mask_ + first,
>-                                    time_smooth_mask_ + last, 0.f);
>-  return sum / (last - first);
>-}
>-
>-void NonlinearBeamformer::EstimateTargetPresence() {
>-  const size_t quantile = static_cast<size_t>(
>-      (high_mean_end_bin_ - low_mean_start_bin_) * kMaskQuantile +
>-      low_mean_start_bin_);
>-  std::nth_element(new_mask_ + low_mean_start_bin_, new_mask_ + quantile,
>-                   new_mask_ + high_mean_end_bin_ + 1);
>-  if (new_mask_[quantile] > kMaskTargetThreshold) {
>-    is_target_present_ = true;
>-    interference_blocks_count_ = 0;
>-  } else {
>-    is_target_present_ = interference_blocks_count_++ < hold_target_blocks_;
>-  }
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.h
>deleted file mode 100644
>index 5b79dc467e2..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer.h
>+++ /dev/null
>@@ -1,230 +0,0 @@
>-/*
>- *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MODULES_AUDIO_PROCESSING_BEAMFORMER_NONLINEAR_BEAMFORMER_H_
>-#define MODULES_AUDIO_PROCESSING_BEAMFORMER_NONLINEAR_BEAMFORMER_H_
>-
>-// MSVC++ requires this to be set before any other includes to get M_PI.
>-#ifndef _USE_MATH_DEFINES
>-#define _USE_MATH_DEFINES
>-#endif
>-
>-#include <math.h>
>-
>-#include <memory>
>-#include <vector>
>-
>-#include "common_audio/lapped_transform.h"
>-#include "common_audio/channel_buffer.h"
>-#include "modules/audio_processing/beamformer/array_util.h"
>-#include "modules/audio_processing/beamformer/complex_matrix.h"
>-
>-namespace webrtc {
>-
>-class PostFilterTransform : public LappedTransform::Callback {
>- public:
>-  PostFilterTransform(size_t num_channels,
>-                      size_t chunk_length,
>-                      float* window,
>-                      size_t fft_size);
>-
>-  void ProcessChunk(float* const* data, float* final_mask);
>-
>- protected:
>-  void ProcessAudioBlock(const complex<float>* const* input,
>-                         size_t num_input_channels,
>-                         size_t num_freq_bins,
>-                         size_t num_output_channels,
>-                         complex<float>* const* output) override;
>-
>- private:
>-  LappedTransform transform_;
>-  const size_t num_freq_bins_;
>-  float* final_mask_;
>-};
>-
>-// Enhances sound sources coming directly in front of a uniform linear array
>-// and suppresses sound sources coming from all other directions. Operates on
>-// multichannel signals and produces single-channel output.
>-//
>-// The implemented nonlinear postfilter algorithm taken from "A Robust Nonlinear
>-// Beamforming Postprocessor" by Bastiaan Kleijn.
>-class NonlinearBeamformer : public LappedTransform::Callback {
>- public:
>-  static const float kHalfBeamWidthRadians;
>-
>-  explicit NonlinearBeamformer(
>-      const std::vector<Point>& array_geometry,
>-      size_t num_postfilter_channels = 1u,
>-      SphericalPointf target_direction =
>-          SphericalPointf(static_cast<float>(M_PI) / 2.f, 0.f, 1.f));
>-  ~NonlinearBeamformer() override;
>-
>-  // Sample rate corresponds to the lower band.
>-  // Needs to be called before the NonlinearBeamformer can be used.
>-  virtual void Initialize(int chunk_size_ms, int sample_rate_hz);
>-
>-  // Analyzes one time-domain chunk of audio. The audio is expected to be split
>-  // into frequency bands inside the ChannelBuffer. The number of frames and
>-  // channels must correspond to the constructor parameters.
>-  virtual void AnalyzeChunk(const ChannelBuffer<float>& data);
>-
>-  // Applies the postfilter mask to one chunk of audio. The audio is expected to
>-  // be split into frequency bands inside the ChannelBuffer. The number of
>-  // frames and channels must correspond to the constructor parameters.
>-  virtual void PostFilter(ChannelBuffer<float>* data);
>-
>-  virtual void AimAt(const SphericalPointf& target_direction);
>-
>-  virtual bool IsInBeam(const SphericalPointf& spherical_point);
>-
>-  // After processing each block |is_target_present_| is set to true if the
>-  // target signal es present and to false otherwise. This methods can be called
>-  // to know if the data is target signal or interference and process it
>-  // accordingly.
>-  virtual bool is_target_present();
>-
>- protected:
>-  // Process one frequency-domain block of audio. This is where the fun
>-  // happens. Implements LappedTransform::Callback.
>-  void ProcessAudioBlock(const complex<float>* const* input,
>-                         size_t num_input_channels,
>-                         size_t num_freq_bins,
>-                         size_t num_output_channels,
>-                         complex<float>* const* output) override;
>-
>- private:
>-  FRIEND_TEST_ALL_PREFIXES(NonlinearBeamformerTest,
>-                           InterfAnglesTakeAmbiguityIntoAccount);
>-
>-  typedef Matrix<float> MatrixF;
>-  typedef ComplexMatrix<float> ComplexMatrixF;
>-  typedef complex<float> complex_f;
>-
>-  void InitLowFrequencyCorrectionRanges();
>-  void InitHighFrequencyCorrectionRanges();
>-  void InitInterfAngles();
>-  void InitDelaySumMasks();
>-  void InitTargetCovMats();
>-  void InitDiffuseCovMats();
>-  void InitInterfCovMats();
>-  void NormalizeCovMats();
>-
>-  // Calculates postfilter masks that minimize the mean squared error of our
>-  // estimation of the desired signal.
>-  float CalculatePostfilterMask(const ComplexMatrixF& interf_cov_mat,
>-                                float rpsiw,
>-                                float ratio_rxiw_rxim,
>-                                float rmxi_r);
>-
>-  // Prevents the postfilter masks from degenerating too quickly (a cause of
>-  // musical noise).
>-  void ApplyMaskTimeSmoothing();
>-  void ApplyMaskFrequencySmoothing();
>-
>-  // The postfilter masks are unreliable at low frequencies. Calculates a better
>-  // mask by averaging mid-low frequency values.
>-  void ApplyLowFrequencyCorrection();
>-
>-  // Postfilter masks are also unreliable at high frequencies. Average mid-high
>-  // frequency masks to calculate a single mask per block which can be applied
>-  // in the time-domain. Further, we average these block-masks over a chunk,
>-  // resulting in one postfilter mask per audio chunk. This allows us to skip
>-  // both transforming and blocking the high-frequency signal.
>-  void ApplyHighFrequencyCorrection();
>-
>-  // Compute the means needed for the above frequency correction.
>-  float MaskRangeMean(size_t start_bin, size_t end_bin);
>-
>-  // Applies post-filter mask to |input| and store in |output|.
>-  void ApplyPostFilter(const complex_f* input, complex_f* output);
>-
>-  void EstimateTargetPresence();
>-
>-  static const size_t kFftSize = 256;
>-  static const size_t kNumFreqBins = kFftSize / 2 + 1;
>-
>-  // Deals with the fft transform and blocking.
>-  size_t chunk_length_;
>-  std::unique_ptr<LappedTransform> process_transform_;
>-  std::unique_ptr<PostFilterTransform> postfilter_transform_;
>-  float window_[kFftSize];
>-
>-  // Parameters exposed to the user.
>-  const size_t num_input_channels_;
>-  const size_t num_postfilter_channels_;
>-  int sample_rate_hz_;
>-
>-  const std::vector<Point> array_geometry_;
>-  // The normal direction of the array if it has one and it is in the xy-plane.
>-  const rtc::Optional<Point> array_normal_;
>-
>-  // Minimum spacing between microphone pairs.
>-  const float min_mic_spacing_;
>-
>-  // Calculated based on user-input and constants in the .cc file.
>-  size_t low_mean_start_bin_;
>-  size_t low_mean_end_bin_;
>-  size_t high_mean_start_bin_;
>-  size_t high_mean_end_bin_;
>-
>-  // Quickly varying mask updated every block.
>-  float new_mask_[kNumFreqBins];
>-  // Time smoothed mask.
>-  float time_smooth_mask_[kNumFreqBins];
>-  // Time and frequency smoothed mask.
>-  float final_mask_[kNumFreqBins];
>-
>-  float target_angle_radians_;
>-  // Angles of the interferer scenarios.
>-  std::vector<float> interf_angles_radians_;
>-  // The angle between the target and the interferer scenarios.
>-  const float away_radians_;
>-
>-  // Array of length |kNumFreqBins|, Matrix of size |1| x |num_channels_|.
>-  ComplexMatrixF delay_sum_masks_[kNumFreqBins];
>-
>-  // Arrays of length |kNumFreqBins|, Matrix of size |num_input_channels_| x
>-  // |num_input_channels_|.
>-  ComplexMatrixF target_cov_mats_[kNumFreqBins];
>-  ComplexMatrixF uniform_cov_mat_[kNumFreqBins];
>-  // Array of length |kNumFreqBins|, Matrix of size |num_input_channels_| x
>-  // |num_input_channels_|. The vector has a size equal to the number of
>-  // interferer scenarios.
>-  std::vector<std::unique_ptr<ComplexMatrixF>> interf_cov_mats_[kNumFreqBins];
>-
>-  // Of length |kNumFreqBins|.
>-  float wave_numbers_[kNumFreqBins];
>-
>-  // Preallocated for ProcessAudioBlock()
>-  // Of length |kNumFreqBins|.
>-  float rxiws_[kNumFreqBins];
>-  // The vector has a size equal to the number of interferer scenarios.
>-  std::vector<float> rpsiws_[kNumFreqBins];
>-
>-  // The microphone normalization factor.
>-  ComplexMatrixF eig_m_;
>-
>-  // For processing the high-frequency input signal.
>-  float high_pass_postfilter_mask_;
>-  float old_high_pass_mask_;
>-
>-  // True when the target signal is present.
>-  bool is_target_present_;
>-  // Number of blocks after which the data is considered interference if the
>-  // mask does not pass |kMaskSignalThreshold|.
>-  size_t hold_target_blocks_;
>-  // Number of blocks since the last mask that passed |kMaskSignalThreshold|.
>-  size_t interference_blocks_count_;
>-};
>-
>-}  // namespace webrtc
>-
>-#endif  // MODULES_AUDIO_PROCESSING_BEAMFORMER_NONLINEAR_BEAMFORMER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer_test.cc
>deleted file mode 100644
>index 296cd6d4ae0..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer_test.cc
>+++ /dev/null
>@@ -1,96 +0,0 @@
>-/*
>- *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include <vector>
>-
>-#include "common_audio/channel_buffer.h"
>-#include "common_audio/wav_file.h"
>-#include "modules/audio_processing/beamformer/nonlinear_beamformer.h"
>-#include "modules/audio_processing/test/test_utils.h"
>-#include "rtc_base/checks.h"
>-#include "rtc_base/flags.h"
>-#include "rtc_base/format_macros.h"
>-
>-DEFINE_string(i, "", "The name of the input file to read from.");
>-DEFINE_string(o, "out.wav", "Name of the output file to write to.");
>-DEFINE_string(mic_positions, "",
>-    "Space delimited cartesian coordinates of microphones in meters. "
>-    "The coordinates of each point are contiguous. "
>-    "For a two element array: \"x1 y1 z1 x2 y2 z2\"");
>-DEFINE_bool(help, false, "Prints this message.");
>-
>-namespace webrtc {
>-namespace {
>-
>-const int kChunksPerSecond = 100;
>-const int kChunkSizeMs = 1000 / kChunksPerSecond;
>-
>-const char kUsage[] =
>-    "Command-line tool to run beamforming on WAV files. The signal is passed\n"
>-    "in as a single band, unlike the audio processing interface which splits\n"
>-    "signals into multiple bands.\n";
>-
>-}  // namespace
>-
>-int main(int argc, char* argv[]) {
>-  if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) ||
>-      FLAG_help || argc != 1) {
>-    printf("%s", kUsage);
>-    if (FLAG_help) {
>-      rtc::FlagList::Print(nullptr, false);
>-      return 0;
>-    }
>-    return 1;
>-  }
>-
>-  WavReader in_file(FLAG_i);
>-  WavWriter out_file(FLAG_o, in_file.sample_rate(), in_file.num_channels());
>-
>-  const size_t num_mics = in_file.num_channels();
>-  const std::vector<Point> array_geometry =
>-      ParseArrayGeometry(FLAG_mic_positions, num_mics);
>-  RTC_CHECK_EQ(array_geometry.size(), num_mics);
>-
>-  NonlinearBeamformer bf(array_geometry, array_geometry.size());
>-  bf.Initialize(kChunkSizeMs, in_file.sample_rate());
>-
>-  printf("Input file: %s\nChannels: %" PRIuS ", Sample rate: %d Hz\n\n",
>-         FLAG_i, in_file.num_channels(), in_file.sample_rate());
>-  printf("Output file: %s\nChannels: %" PRIuS ", Sample rate: %d Hz\n\n",
>-         FLAG_o, out_file.num_channels(), out_file.sample_rate());
>-
>-  ChannelBuffer<float> buf(
>-      rtc::CheckedDivExact(in_file.sample_rate(), kChunksPerSecond),
>-      in_file.num_channels());
>-
>-  std::vector<float> interleaved(buf.size());
>-  while (in_file.ReadSamples(interleaved.size(),
>-                             &interleaved[0]) == interleaved.size()) {
>-    FloatS16ToFloat(&interleaved[0], interleaved.size(), &interleaved[0]);
>-    Deinterleave(&interleaved[0], buf.num_frames(),
>-                 buf.num_channels(), buf.channels());
>-
>-    bf.AnalyzeChunk(buf);
>-    bf.PostFilter(&buf);
>-
>-    Interleave(buf.channels(), buf.num_frames(),
>-               buf.num_channels(), &interleaved[0]);
>-    FloatToFloatS16(&interleaved[0], interleaved.size(), &interleaved[0]);
>-    out_file.WriteSamples(&interleaved[0], interleaved.size());
>-  }
>-
>-  return 0;
>-}
>-
>-}  // namespace webrtc
>-
>-int main(int argc, char* argv[]) {
>-  return webrtc::main(argc, argv);
>-}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer_unittest.cc
>deleted file mode 100644
>index 78b2f0a68e7..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/beamformer/nonlinear_beamformer_unittest.cc
>+++ /dev/null
>@@ -1,375 +0,0 @@
>-/*
>- *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-// MSVC++ requires this to be set before any other includes to get M_PI.
>-#define _USE_MATH_DEFINES
>-
>-#include "modules/audio_processing/beamformer/nonlinear_beamformer.h"
>-
>-#include <math.h>
>-
>-#include "api/array_view.h"
>-#include "modules/audio_processing/audio_buffer.h"
>-#include "modules/audio_processing/test/audio_buffer_tools.h"
>-#include "modules/audio_processing/test/bitexactness_tools.h"
>-#include "test/gtest.h"
>-
>-namespace webrtc {
>-namespace {
>-
>-const int kChunkSizeMs = 10;
>-const int kSampleRateHz = 16000;
>-
>-SphericalPointf AzimuthToSphericalPoint(float azimuth_radians) {
>-  return SphericalPointf(azimuth_radians, 0.f, 1.f);
>-}
>-
>-void Verify(NonlinearBeamformer* bf, float target_azimuth_radians) {
>-  EXPECT_TRUE(bf->IsInBeam(AzimuthToSphericalPoint(target_azimuth_radians)));
>-  EXPECT_TRUE(bf->IsInBeam(AzimuthToSphericalPoint(
>-      target_azimuth_radians - NonlinearBeamformer::kHalfBeamWidthRadians +
>-      0.001f)));
>-  EXPECT_TRUE(bf->IsInBeam(AzimuthToSphericalPoint(
>-      target_azimuth_radians + NonlinearBeamformer::kHalfBeamWidthRadians -
>-      0.001f)));
>-  EXPECT_FALSE(bf->IsInBeam(AzimuthToSphericalPoint(
>-      target_azimuth_radians - NonlinearBeamformer::kHalfBeamWidthRadians -
>-      0.001f)));
>-  EXPECT_FALSE(bf->IsInBeam(AzimuthToSphericalPoint(
>-      target_azimuth_radians + NonlinearBeamformer::kHalfBeamWidthRadians +
>-      0.001f)));
>-}
>-
>-void AimAndVerify(NonlinearBeamformer* bf, float target_azimuth_radians) {
>-  bf->AimAt(AzimuthToSphericalPoint(target_azimuth_radians));
>-  Verify(bf, target_azimuth_radians);
>-}
>-
>-// Bitexactness test code.
>-const size_t kNumFramesToProcess = 1000;
>-
>-void ProcessOneFrame(int sample_rate_hz,
>-                     AudioBuffer* capture_audio_buffer,
>-                     NonlinearBeamformer* beamformer) {
>-  if (sample_rate_hz > AudioProcessing::kSampleRate16kHz) {
>-    capture_audio_buffer->SplitIntoFrequencyBands();
>-  }
>-
>-  beamformer->AnalyzeChunk(*capture_audio_buffer->split_data_f());
>-  capture_audio_buffer->set_num_channels(1);
>-  beamformer->PostFilter(capture_audio_buffer->split_data_f());
>-
>-  if (sample_rate_hz > AudioProcessing::kSampleRate16kHz) {
>-    capture_audio_buffer->MergeFrequencyBands();
>-  }
>-}
>-
>-int BeamformerSampleRate(int sample_rate_hz) {
>-  return (sample_rate_hz > AudioProcessing::kSampleRate16kHz
>-              ? AudioProcessing::kSampleRate16kHz
>-              : sample_rate_hz);
>-}
>-
>-void RunBitExactnessTest(int sample_rate_hz,
>-                         const std::vector<Point>& array_geometry,
>-                         const SphericalPointf& target_direction,
>-                         rtc::ArrayView<const float> output_reference) {
>-  NonlinearBeamformer beamformer(array_geometry, 1u, target_direction);
>-  beamformer.Initialize(AudioProcessing::kChunkSizeMs,
>-                        BeamformerSampleRate(sample_rate_hz));
>-
>-  const StreamConfig capture_config(sample_rate_hz, array_geometry.size(),
>-                                    false);
>-  AudioBuffer capture_buffer(
>-      capture_config.num_frames(), capture_config.num_channels(),
>-      capture_config.num_frames(), capture_config.num_channels(),
>-      capture_config.num_frames());
>-  test::InputAudioFile capture_file(
>-      test::GetApmCaptureTestVectorFileName(sample_rate_hz));
>-  std::vector<float> capture_input(capture_config.num_frames() *
>-                                   capture_config.num_channels());
>-  for (size_t frame_no = 0u; frame_no < kNumFramesToProcess; ++frame_no) {
>-    ReadFloatSamplesFromStereoFile(capture_config.num_frames(),
>-                                   capture_config.num_channels(), &capture_file,
>-                                   capture_input);
>-
>-    test::CopyVectorToAudioBuffer(capture_config, capture_input,
>-                                  &capture_buffer);
>-
>-    ProcessOneFrame(sample_rate_hz, &capture_buffer, &beamformer);
>-  }
>-
>-  // Extract and verify the test results.
>-  std::vector<float> capture_output;
>-  test::ExtractVectorFromAudioBuffer(capture_config, &capture_buffer,
>-                                     &capture_output);
>-
>-  const float kElementErrorBound = 1.f / static_cast<float>(1 << 15);
>-
>-  // Compare the output with the reference. Only the first values of the output
>-  // from last frame processed are compared in order not having to specify all
>-  // preceeding frames as testvectors. As the algorithm being tested has a
>-  // memory, testing only the last frame implicitly also tests the preceeding
>-  // frames.
>-  EXPECT_TRUE(test::VerifyDeinterleavedArray(
>-      capture_config.num_frames(), capture_config.num_channels(),
>-      output_reference, capture_output, kElementErrorBound));
>-}
>-
>-// TODO(peah): Add bitexactness tests for scenarios with more than 2 input
>-// channels.
>-std::vector<Point> CreateArrayGeometry(int variant) {
>-  std::vector<Point> array_geometry;
>-  switch (variant) {
>-    case 1:
>-      array_geometry.push_back(Point(-0.025f, 0.f, 0.f));
>-      array_geometry.push_back(Point(0.025f, 0.f, 0.f));
>-      break;
>-    case 2:
>-      array_geometry.push_back(Point(-0.035f, 0.f, 0.f));
>-      array_geometry.push_back(Point(0.035f, 0.f, 0.f));
>-      break;
>-    case 3:
>-      array_geometry.push_back(Point(-0.5f, 0.f, 0.f));
>-      array_geometry.push_back(Point(0.5f, 0.f, 0.f));
>-      break;
>-    default:
>-      RTC_CHECK(false);
>-  }
>-  return array_geometry;
>-}
>-
>-const SphericalPointf TargetDirection1(0.4f * static_cast<float>(M_PI) / 2.f,
>-                                       0.f,
>-                                       1.f);
>-const SphericalPointf TargetDirection2(static_cast<float>(M_PI) / 2.f,
>-                                       1.f,
>-                                       2.f);
>-
>-}  // namespace
>-
>-TEST(NonlinearBeamformerTest, AimingModifiesBeam) {
>-  std::vector<Point> array_geometry;
>-  array_geometry.push_back(Point(-0.025f, 0.f, 0.f));
>-  array_geometry.push_back(Point(0.025f, 0.f, 0.f));
>-  NonlinearBeamformer bf(array_geometry, 1u);
>-  bf.Initialize(kChunkSizeMs, kSampleRateHz);
>-  // The default constructor parameter sets the target angle to PI / 2.
>-  Verify(&bf, static_cast<float>(M_PI) / 2.f);
>-  AimAndVerify(&bf, static_cast<float>(M_PI) / 3.f);
>-  AimAndVerify(&bf, 3.f * static_cast<float>(M_PI) / 4.f);
>-  AimAndVerify(&bf, static_cast<float>(M_PI) / 6.f);
>-  AimAndVerify(&bf, static_cast<float>(M_PI));
>-}
>-
>-TEST(NonlinearBeamformerTest, InterfAnglesTakeAmbiguityIntoAccount) {
>-  {
>-    // For linear arrays there is ambiguity.
>-    std::vector<Point> array_geometry;
>-    array_geometry.push_back(Point(-0.1f, 0.f, 0.f));
>-    array_geometry.push_back(Point(0.f, 0.f, 0.f));
>-    array_geometry.push_back(Point(0.2f, 0.f, 0.f));
>-    NonlinearBeamformer bf(array_geometry, 1u);
>-    bf.Initialize(kChunkSizeMs, kSampleRateHz);
>-    EXPECT_EQ(2u, bf.interf_angles_radians_.size());
>-    EXPECT_FLOAT_EQ(M_PI / 2.f - bf.away_radians_,
>-                    bf.interf_angles_radians_[0]);
>-    EXPECT_FLOAT_EQ(M_PI / 2.f + bf.away_radians_,
>-                    bf.interf_angles_radians_[1]);
>-    bf.AimAt(AzimuthToSphericalPoint(bf.away_radians_ / 2.f));
>-    EXPECT_EQ(2u, bf.interf_angles_radians_.size());
>-    EXPECT_FLOAT_EQ(M_PI - bf.away_radians_ / 2.f,
>-                    bf.interf_angles_radians_[0]);
>-    EXPECT_FLOAT_EQ(3.f * bf.away_radians_ / 2.f, bf.interf_angles_radians_[1]);
>-  }
>-  {
>-    // For planar arrays with normal in the xy-plane there is ambiguity.
>-    std::vector<Point> array_geometry;
>-    array_geometry.push_back(Point(-0.1f, 0.f, 0.f));
>-    array_geometry.push_back(Point(0.f, 0.f, 0.f));
>-    array_geometry.push_back(Point(0.2f, 0.f, 0.f));
>-    array_geometry.push_back(Point(0.1f, 0.f, 0.2f));
>-    array_geometry.push_back(Point(0.f, 0.f, -0.1f));
>-    NonlinearBeamformer bf(array_geometry, 1u);
>-    bf.Initialize(kChunkSizeMs, kSampleRateHz);
>-    EXPECT_EQ(2u, bf.interf_angles_radians_.size());
>-    EXPECT_FLOAT_EQ(M_PI / 2.f - bf.away_radians_,
>-                    bf.interf_angles_radians_[0]);
>-    EXPECT_FLOAT_EQ(M_PI / 2.f + bf.away_radians_,
>-                    bf.interf_angles_radians_[1]);
>-    bf.AimAt(AzimuthToSphericalPoint(bf.away_radians_ / 2.f));
>-    EXPECT_EQ(2u, bf.interf_angles_radians_.size());
>-    EXPECT_FLOAT_EQ(M_PI - bf.away_radians_ / 2.f,
>-                    bf.interf_angles_radians_[0]);
>-    EXPECT_FLOAT_EQ(3.f * bf.away_radians_ / 2.f, bf.interf_angles_radians_[1]);
>-  }
>-  {
>-    // For planar arrays with normal not in the xy-plane there is no ambiguity.
>-    std::vector<Point> array_geometry;
>-    array_geometry.push_back(Point(0.f, 0.f, 0.f));
>-    array_geometry.push_back(Point(0.2f, 0.f, 0.f));
>-    array_geometry.push_back(Point(0.f, 0.1f, -0.2f));
>-    NonlinearBeamformer bf(array_geometry, 1u);
>-    bf.Initialize(kChunkSizeMs, kSampleRateHz);
>-    EXPECT_EQ(2u, bf.interf_angles_radians_.size());
>-    EXPECT_FLOAT_EQ(M_PI / 2.f - bf.away_radians_,
>-                    bf.interf_angles_radians_[0]);
>-    EXPECT_FLOAT_EQ(M_PI / 2.f + bf.away_radians_,
>-                    bf.interf_angles_radians_[1]);
>-    bf.AimAt(AzimuthToSphericalPoint(bf.away_radians_ / 2.f));
>-    EXPECT_EQ(2u, bf.interf_angles_radians_.size());
>-    EXPECT_FLOAT_EQ(-bf.away_radians_ / 2.f, bf.interf_angles_radians_[0]);
>-    EXPECT_FLOAT_EQ(3.f * bf.away_radians_ / 2.f, bf.interf_angles_radians_[1]);
>-  }
>-  {
>-    // For arrays which are not linear or planar there is no ambiguity.
>-    std::vector<Point> array_geometry;
>-    array_geometry.push_back(Point(0.f, 0.f, 0.f));
>-    array_geometry.push_back(Point(0.1f, 0.f, 0.f));
>-    array_geometry.push_back(Point(0.f, 0.2f, 0.f));
>-    array_geometry.push_back(Point(0.f, 0.f, 0.3f));
>-    NonlinearBeamformer bf(array_geometry, 1u);
>-    bf.Initialize(kChunkSizeMs, kSampleRateHz);
>-    EXPECT_EQ(2u, bf.interf_angles_radians_.size());
>-    EXPECT_FLOAT_EQ(M_PI / 2.f - bf.away_radians_,
>-                    bf.interf_angles_radians_[0]);
>-    EXPECT_FLOAT_EQ(M_PI / 2.f + bf.away_radians_,
>-                    bf.interf_angles_radians_[1]);
>-    bf.AimAt(AzimuthToSphericalPoint(bf.away_radians_ / 2.f));
>-    EXPECT_EQ(2u, bf.interf_angles_radians_.size());
>-    EXPECT_FLOAT_EQ(-bf.away_radians_ / 2.f, bf.interf_angles_radians_[0]);
>-    EXPECT_FLOAT_EQ(3.f * bf.away_radians_ / 2.f, bf.interf_angles_radians_[1]);
>-  }
>-}
>-
>-// TODO(peah): Investigate why the nonlinear_beamformer.cc causes a DCHECK in
>-// this setup.
>-TEST(BeamformerBitExactnessTest,
>-     DISABLED_Stereo8kHz_ArrayGeometry1_TargetDirection1) {
>-  const float kOutputReference[] = {0.001318f, -0.001091f, 0.000990f,
>-                                    0.001318f, -0.001091f, 0.000990f};
>-
>-  RunBitExactnessTest(AudioProcessing::kSampleRate8kHz, CreateArrayGeometry(1),
>-                      TargetDirection1, kOutputReference);
>-}
>-
>-TEST(BeamformerBitExactnessTest,
>-     Stereo16kHz_ArrayGeometry1_TargetDirection1) {
>-  const float kOutputReference[] = {-0.000077f, -0.000147f, -0.000138f,
>-                                    -0.000077f, -0.000147f, -0.000138f};
>-
>-  RunBitExactnessTest(AudioProcessing::kSampleRate16kHz, CreateArrayGeometry(1),
>-                      TargetDirection1, kOutputReference);
>-}
>-
>-TEST(BeamformerBitExactnessTest,
>-     Stereo32kHz_ArrayGeometry1_TargetDirection1) {
>-  const float kOutputReference[] = {-0.000061f, -0.000061f, -0.000061f,
>-                                    -0.000061f, -0.000061f, -0.000061f};
>-
>-  RunBitExactnessTest(AudioProcessing::kSampleRate32kHz, CreateArrayGeometry(1),
>-                      TargetDirection1, kOutputReference);
>-}
>-
>-TEST(BeamformerBitExactnessTest,
>-     Stereo48kHz_ArrayGeometry1_TargetDirection1) {
>-  const float kOutputReference[] = {0.000450f, 0.000436f, 0.000433f,
>-                                    0.000450f, 0.000436f, 0.000433f};
>-
>-  RunBitExactnessTest(AudioProcessing::kSampleRate48kHz, CreateArrayGeometry(1),
>-                      TargetDirection1, kOutputReference);
>-}
>-
>-// TODO(peah): Investigate why the nonlinear_beamformer.cc causes a DCHECK in
>-// this setup.
>-TEST(BeamformerBitExactnessTest,
>-     DISABLED_Stereo8kHz_ArrayGeometry1_TargetDirection2) {
>-  const float kOutputReference[] = {0.001144f,  -0.001026f, 0.001074f,
>-                                    -0.016205f, -0.007324f, -0.015656f};
>-
>-  RunBitExactnessTest(AudioProcessing::kSampleRate8kHz, CreateArrayGeometry(1),
>-                      TargetDirection2, kOutputReference);
>-}
>-
>-TEST(BeamformerBitExactnessTest,
>-     Stereo16kHz_ArrayGeometry1_TargetDirection2) {
>-  const float kOutputReference[] = {0.000221f, -0.000249f, 0.000140f,
>-                                    0.000221f, -0.000249f, 0.000140f};
>-
>-  RunBitExactnessTest(AudioProcessing::kSampleRate16kHz, CreateArrayGeometry(1),
>-                      TargetDirection2, kOutputReference);
>-}
>-
>-TEST(BeamformerBitExactnessTest,
>-     Stereo32kHz_ArrayGeometry1_TargetDirection2) {
>-  const float kOutputReference[] = {0.000763f, -0.000336f, 0.000549f,
>-                                    0.000763f, -0.000336f, 0.000549f};
>-
>-  RunBitExactnessTest(AudioProcessing::kSampleRate32kHz, CreateArrayGeometry(1),
>-                      TargetDirection2, kOutputReference);
>-}
>-
>-TEST(BeamformerBitExactnessTest,
>-     Stereo48kHz_ArrayGeometry1_TargetDirection2) {
>-  const float kOutputReference[] = {-0.000004f, -0.000494f, 0.000255f,
>-                                    -0.000004f, -0.000494f, 0.000255f};
>-
>-  RunBitExactnessTest(AudioProcessing::kSampleRate48kHz, CreateArrayGeometry(1),
>-                      TargetDirection2, kOutputReference);
>-}
>-
>-TEST(BeamformerBitExactnessTest,
>-     Stereo8kHz_ArrayGeometry2_TargetDirection2) {
>-  const float kOutputReference[] = {-0.000914f, 0.002170f, -0.002382f,
>-                                    -0.000914f, 0.002170f, -0.002382f};
>-
>-  RunBitExactnessTest(AudioProcessing::kSampleRate8kHz, CreateArrayGeometry(2),
>-                      TargetDirection2, kOutputReference);
>-}
>-
>-TEST(BeamformerBitExactnessTest,
>-     Stereo16kHz_ArrayGeometry2_TargetDirection2) {
>-  const float kOutputReference[] = {0.000179f, -0.000179f, 0.000081f,
>-                                    0.000179f, -0.000179f, 0.000081f};
>-
>-  RunBitExactnessTest(AudioProcessing::kSampleRate16kHz, CreateArrayGeometry(2),
>-                      TargetDirection2, kOutputReference);
>-}
>-
>-TEST(BeamformerBitExactnessTest,
>-     Stereo32kHz_ArrayGeometry2_TargetDirection2) {
>-  const float kOutputReference[] = {0.000549f, -0.000214f, 0.000366f,
>-                                    0.000549f, -0.000214f, 0.000366f};
>-
>-  RunBitExactnessTest(AudioProcessing::kSampleRate32kHz, CreateArrayGeometry(2),
>-                      TargetDirection2, kOutputReference);
>-}
>-
>-TEST(BeamformerBitExactnessTest,
>-     Stereo48kHz_ArrayGeometry2_TargetDirection2) {
>-  const float kOutputReference[] = {0.000019f, -0.000310f, 0.000182f,
>-                                    0.000019f, -0.000310f, 0.000182f};
>-
>-  RunBitExactnessTest(AudioProcessing::kSampleRate48kHz, CreateArrayGeometry(2),
>-                      TargetDirection2, kOutputReference);
>-}
>-
>-// TODO(peah): Investigate why the nonlinear_beamformer.cc causes a DCHECK in
>-// this setup.
>-TEST(BeamformerBitExactnessTest,
>-     DISABLED_Stereo16kHz_ArrayGeometry3_TargetDirection1) {
>-  const float kOutputReference[] = {-0.000161f, 0.000171f, -0.000096f,
>-                                    0.001007f,  0.000427f, 0.000977f};
>-
>-  RunBitExactnessTest(AudioProcessing::kSampleRate16kHz, CreateArrayGeometry(3),
>-                      TargetDirection1, kOutputReference);
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/config_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/config_unittest.cc
>index 8776ee3391c..19e9ab37d22 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/config_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/config_unittest.cc
>@@ -19,11 +19,9 @@ struct MyExperiment {
>   static const int kDefaultFactor;
>   static const int kDefaultOffset;
> 
>-  MyExperiment()
>-    : factor(kDefaultFactor), offset(kDefaultOffset) {}
>+  MyExperiment() : factor(kDefaultFactor), offset(kDefaultOffset) {}
> 
>-  MyExperiment(int factor, int offset)
>-    : factor(factor), offset(offset) {}
>+  MyExperiment(int factor, int offset) : factor(factor), offset(offset) {}
> 
>   int factor;
>   int offset;
>@@ -61,17 +59,13 @@ struct Algo1_CostFunction {
>       ConfigOptionID::kAlgo1CostFunctionForTest;
>   Algo1_CostFunction() {}
> 
>-  virtual int cost(int x) const {
>-    return x;
>-  }
>+  virtual int cost(int x) const { return x; }
> 
>   virtual ~Algo1_CostFunction() {}
> };
> 
> struct SqrCost : Algo1_CostFunction {
>-  virtual int cost(int x) const {
>-    return x*x;
>-  }
>+  virtual int cost(int x) const { return x * x; }
> };
> 
> TEST(Config, SupportsPolymorphism) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/debug.proto b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/debug.proto
>index 44177735e8c..ebfb84d9cd0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/debug.proto
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/debug.proto
>@@ -14,6 +14,7 @@ message Init {
>   optional int32 output_sample_rate = 7;
>   optional int32 reverse_output_sample_rate = 8;
>   optional int32 num_reverse_output_channels = 9;
>+  optional int64 timestamp_ms = 10;
> }
> 
> // May contain interleaved or deinterleaved data, but don't store both formats.
>@@ -72,8 +73,11 @@ message Config {
>   // Semicolon-separated string containing experimental feature
>   // descriptions.
>   optional string experiments_description = 17;
>-  // Intelligibility Enhancer
>+  // Intelligibility Enhancer.
>   optional bool intelligibility_enhancer_enabled = 18;
>+  // Pre amplifier.
>+  optional bool pre_amplifier_enabled = 19;
>+  optional float pre_amplifier_fixed_gain_factor = 20;
> }
> 
> message Event {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_cancellation_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_cancellation_impl.cc
>index 99f676c5628..9feaea7c3ff 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_cancellation_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_cancellation_impl.cc
>@@ -109,7 +109,7 @@ EchoCancellationImpl::EchoCancellationImpl(rtc::CriticalSection* crit_render,
>       crit_capture_(crit_capture),
>       drift_compensation_enabled_(false),
>       metrics_enabled_(false),
>-      suppression_level_(kModerateSuppression),
>+      suppression_level_(kHighSuppression),
>       stream_drift_samples_(0),
>       was_stream_drift_set_(false),
>       stream_has_echo_(false),
>@@ -147,7 +147,6 @@ void EchoCancellationImpl::ProcessRenderAudio(
>   }
> }
> 
>-
> int EchoCancellationImpl::ProcessCaptureAudio(AudioBuffer* audio,
>                                               int stream_delay_ms) {
>   rtc::CritScope cs_capture(crit_capture_);
>@@ -376,7 +375,8 @@ int EchoCancellationImpl::GetDelayMetrics(int* median, int* std) {
>   return GetDelayMetrics(median, std, &fraction_poor_delays);
> }
> 
>-int EchoCancellationImpl::GetDelayMetrics(int* median, int* std,
>+int EchoCancellationImpl::GetDelayMetrics(int* median,
>+                                          int* std,
>                                           float* fraction_poor_delays) {
>   rtc::CritScope cs(crit_capture_);
>   if (median == NULL) {
>@@ -445,8 +445,7 @@ int EchoCancellationImpl::GetSystemDelayInSamples() const {
>   rtc::CritScope cs(crit_capture_);
>   RTC_DCHECK(enabled_);
>   // Report the delay for the first AEC component.
>-  return WebRtcAec_system_delay(
>-      WebRtcAec_aec_core(cancellers_[0]->state()));
>+  return WebRtcAec_system_delay(WebRtcAec_aec_core(cancellers_[0]->state()));
> }
> 
> void EchoCancellationImpl::PackRenderAudioBuffer(
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_cancellation_proxy.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_cancellation_proxy.cc
>new file mode 100644
>index 00000000000..e8b1fbdd235
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_cancellation_proxy.cc
>@@ -0,0 +1,100 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/echo_cancellation_proxy.h"
>+
>+namespace webrtc {
>+
>+EchoCancellationProxy::EchoCancellationProxy(
>+    AudioProcessing* audio_processing,
>+    EchoCancellation* echo_cancellation)
>+    : audio_processing_(audio_processing),
>+      echo_cancellation_(echo_cancellation) {}
>+
>+EchoCancellationProxy::~EchoCancellationProxy() = default;
>+
>+int EchoCancellationProxy::Enable(bool enable) {
>+  // Change the config in APM to mirror the applied settings.
>+  // TODO(bugs.webrtc.org/9535): Remove the call to EchoCancellation::Enable
>+  // when APM starts taking the config into account.
>+  AudioProcessing::Config apm_config = audio_processing_->GetConfig();
>+  bool aec2_enabled = apm_config.echo_canceller.enabled &&
>+                      !apm_config.echo_canceller.mobile_mode;
>+  if ((aec2_enabled && !enable) || (!aec2_enabled && enable)) {
>+    apm_config.echo_canceller.enabled = enable;
>+    apm_config.echo_canceller.mobile_mode = false;
>+    audio_processing_->ApplyConfig(apm_config);
>+  }
>+  echo_cancellation_->Enable(enable);
>+  return AudioProcessing::kNoError;
>+}
>+
>+bool EchoCancellationProxy::is_enabled() const {
>+  return echo_cancellation_->is_enabled();
>+}
>+
>+int EchoCancellationProxy::enable_drift_compensation(bool enable) {
>+  return echo_cancellation_->enable_drift_compensation(enable);
>+}
>+
>+bool EchoCancellationProxy::is_drift_compensation_enabled() const {
>+  return echo_cancellation_->is_drift_compensation_enabled();
>+}
>+
>+void EchoCancellationProxy::set_stream_drift_samples(int drift) {
>+  echo_cancellation_->set_stream_drift_samples(drift);
>+}
>+
>+int EchoCancellationProxy::stream_drift_samples() const {
>+  return echo_cancellation_->stream_drift_samples();
>+}
>+
>+int EchoCancellationProxy::set_suppression_level(
>+    EchoCancellation::SuppressionLevel level) {
>+  return echo_cancellation_->set_suppression_level(level);
>+}
>+
>+EchoCancellation::SuppressionLevel EchoCancellationProxy::suppression_level()
>+    const {
>+  return echo_cancellation_->suppression_level();
>+}
>+
>+bool EchoCancellationProxy::stream_has_echo() const {
>+  return echo_cancellation_->stream_has_echo();
>+}
>+int EchoCancellationProxy::enable_metrics(bool enable) {
>+  return echo_cancellation_->enable_metrics(enable);
>+}
>+bool EchoCancellationProxy::are_metrics_enabled() const {
>+  return echo_cancellation_->are_metrics_enabled();
>+}
>+int EchoCancellationProxy::GetMetrics(Metrics* metrics) {
>+  return echo_cancellation_->GetMetrics(metrics);
>+}
>+int EchoCancellationProxy::enable_delay_logging(bool enable) {
>+  return echo_cancellation_->enable_delay_logging(enable);
>+}
>+bool EchoCancellationProxy::is_delay_logging_enabled() const {
>+  return echo_cancellation_->is_delay_logging_enabled();
>+}
>+int EchoCancellationProxy::GetDelayMetrics(int* median, int* std) {
>+  return echo_cancellation_->GetDelayMetrics(median, std);
>+}
>+int EchoCancellationProxy::GetDelayMetrics(int* median,
>+                                           int* std,
>+                                           float* fraction_poor_delays) {
>+  return echo_cancellation_->GetDelayMetrics(median, std, fraction_poor_delays);
>+}
>+
>+struct AecCore* EchoCancellationProxy::aec_core() const {
>+  return echo_cancellation_->aec_core();
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_cancellation_proxy.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_cancellation_proxy.h
>new file mode 100644
>index 00000000000..f2d30178d5e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_cancellation_proxy.h
>@@ -0,0 +1,55 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_ECHO_CANCELLATION_PROXY_H_
>+#define MODULES_AUDIO_PROCESSING_ECHO_CANCELLATION_PROXY_H_
>+
>+#include "modules/audio_processing/include/audio_processing.h"
>+#include "rtc_base/constructormagic.h"
>+#include "rtc_base/scoped_ref_ptr.h"
>+
>+namespace webrtc {
>+
>+// Class for temporarily redirecting AEC2 configuration to a new API.
>+class EchoCancellationProxy : public EchoCancellation {
>+ public:
>+  EchoCancellationProxy(AudioProcessing* audio_processing,
>+                        EchoCancellation* echo_cancellation);
>+  ~EchoCancellationProxy() override;
>+
>+  int Enable(bool enable) override;
>+  bool is_enabled() const override;
>+  int enable_drift_compensation(bool enable) override;
>+  bool is_drift_compensation_enabled() const override;
>+  void set_stream_drift_samples(int drift) override;
>+  int stream_drift_samples() const override;
>+  int set_suppression_level(SuppressionLevel level) override;
>+  SuppressionLevel suppression_level() const override;
>+  bool stream_has_echo() const override;
>+  int enable_metrics(bool enable) override;
>+  bool are_metrics_enabled() const override;
>+  int GetMetrics(Metrics* metrics) override;
>+  int enable_delay_logging(bool enable) override;
>+  bool is_delay_logging_enabled() const override;
>+  int GetDelayMetrics(int* median, int* std) override;
>+  int GetDelayMetrics(int* median,
>+                      int* std,
>+                      float* fraction_poor_delays) override;
>+  struct AecCore* aec_core() const override;
>+
>+ private:
>+  AudioProcessing* audio_processing_;
>+  EchoCancellation* echo_cancellation_;
>+
>+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(EchoCancellationProxy);
>+};
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_ECHO_CANCELLATION_PROXY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_control_mobile_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_control_mobile_impl.cc
>index ecb18106667..272da7a71ed 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_control_mobile_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_control_mobile_impl.cc
>@@ -113,17 +113,17 @@ EchoControlMobileImpl::EchoControlMobileImpl(rtc::CriticalSection* crit_render,
>     : crit_render_(crit_render),
>       crit_capture_(crit_capture),
>       routing_mode_(kSpeakerphone),
>-      comfort_noise_enabled_(true),
>+      comfort_noise_enabled_(false),
>       external_echo_path_(NULL) {
>   RTC_DCHECK(crit_render);
>   RTC_DCHECK(crit_capture);
> }
> 
> EchoControlMobileImpl::~EchoControlMobileImpl() {
>-    if (external_echo_path_ != NULL) {
>-      delete [] external_echo_path_;
>-      external_echo_path_ = NULL;
>-    }
>+  if (external_echo_path_ != NULL) {
>+    delete[] external_echo_path_;
>+    external_echo_path_ = NULL;
>+  }
> }
> 
> void EchoControlMobileImpl::ProcessRenderAudio(
>@@ -218,8 +218,7 @@ int EchoControlMobileImpl::ProcessCaptureAudio(AudioBuffer* audio,
>       ++handle_index;
>     }
>     for (size_t band = 1u; band < audio->num_bands(); ++band) {
>-      memset(audio->split_bands(capture)[band],
>-             0,
>+      memset(audio->split_bands(capture)[band], 0,
>              audio->num_frames_per_band() *
>                  sizeof(audio->split_bands(capture)[band][0]));
>     }
>@@ -269,8 +268,7 @@ int EchoControlMobileImpl::set_routing_mode(RoutingMode mode) {
>   return Configure();
> }
> 
>-EchoControlMobile::RoutingMode EchoControlMobileImpl::routing_mode()
>-    const {
>+EchoControlMobile::RoutingMode EchoControlMobileImpl::routing_mode() const {
>   rtc::CritScope cs(crit_capture_);
>   return routing_mode_;
> }
>@@ -353,9 +351,9 @@ void EchoControlMobileImpl::Initialize(int sample_rate_hz,
>     return;
>   }
> 
>-  if (stream_properties_->sample_rate_hz > AudioProcessing::kSampleRate16kHz) {
>-    RTC_LOG(LS_ERROR) << "AECM only supports 16 kHz or lower sample rates";
>-  }
>+  // AECM only supports 16 kHz or lower sample rates.
>+  RTC_DCHECK_LE(stream_properties_->sample_rate_hz,
>+                AudioProcessing::kSampleRate16kHz);
> 
>   cancellers_.resize(
>       NumCancellersRequired(stream_properties_->num_output_channels,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_control_mobile_proxy.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_control_mobile_proxy.cc
>new file mode 100644
>index 00000000000..fd298348f47
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_control_mobile_proxy.cc
>@@ -0,0 +1,69 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/echo_control_mobile_proxy.h"
>+
>+namespace webrtc {
>+
>+EchoControlMobileProxy::EchoControlMobileProxy(
>+    AudioProcessing* audio_processing,
>+    EchoControlMobile* echo_control_mobile)
>+    : audio_processing_(audio_processing),
>+      echo_control_mobile_(echo_control_mobile) {}
>+
>+EchoControlMobileProxy::~EchoControlMobileProxy() = default;
>+
>+int EchoControlMobileProxy::Enable(bool enable) {
>+  // Change the config in APM to mirror the applied settings.
>+  // TODO(bugs.webrtc.org/9535): Remove the call to EchoControlMobile::Enable
>+  // when APM starts taking the config into account.
>+  AudioProcessing::Config apm_config = audio_processing_->GetConfig();
>+  bool aecm_enabled = apm_config.echo_canceller.enabled &&
>+                      apm_config.echo_canceller.mobile_mode;
>+  if ((aecm_enabled && !enable) || (!aecm_enabled && enable)) {
>+    apm_config.echo_canceller.enabled = enable;
>+    apm_config.echo_canceller.mobile_mode = true;
>+    audio_processing_->ApplyConfig(apm_config);
>+  }
>+  echo_control_mobile_->Enable(enable);
>+  return AudioProcessing::kNoError;
>+}
>+
>+bool EchoControlMobileProxy::is_enabled() const {
>+  return echo_control_mobile_->is_enabled();
>+}
>+
>+int EchoControlMobileProxy::set_routing_mode(RoutingMode mode) {
>+  return echo_control_mobile_->set_routing_mode(mode);
>+}
>+
>+EchoControlMobile::RoutingMode EchoControlMobileProxy::routing_mode() const {
>+  return echo_control_mobile_->routing_mode();
>+}
>+
>+int EchoControlMobileProxy::enable_comfort_noise(bool enable) {
>+  return echo_control_mobile_->enable_comfort_noise(enable);
>+}
>+
>+bool EchoControlMobileProxy::is_comfort_noise_enabled() const {
>+  return echo_control_mobile_->is_comfort_noise_enabled();
>+}
>+
>+int EchoControlMobileProxy::SetEchoPath(const void* echo_path,
>+                                        size_t size_bytes) {
>+  return echo_control_mobile_->SetEchoPath(echo_path, size_bytes);
>+}
>+
>+int EchoControlMobileProxy::GetEchoPath(void* echo_path,
>+                                        size_t size_bytes) const {
>+  return echo_control_mobile_->GetEchoPath(echo_path, size_bytes);
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_control_mobile_proxy.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_control_mobile_proxy.h
>new file mode 100644
>index 00000000000..eb5908a4e0b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_control_mobile_proxy.h
>@@ -0,0 +1,44 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_ECHO_CONTROL_MOBILE_PROXY_H_
>+#define MODULES_AUDIO_PROCESSING_ECHO_CONTROL_MOBILE_PROXY_H_
>+
>+#include "modules/audio_processing/include/audio_processing.h"
>+#include "rtc_base/constructormagic.h"
>+#include "rtc_base/scoped_ref_ptr.h"
>+
>+namespace webrtc {
>+
>+// Class for temporarily redirecting AECM configuration to a new API.
>+class EchoControlMobileProxy : public EchoControlMobile {
>+ public:
>+  EchoControlMobileProxy(AudioProcessing* audio_processing,
>+                         EchoControlMobile* echo_control_mobile);
>+  ~EchoControlMobileProxy() override;
>+
>+  bool is_enabled() const override;
>+  RoutingMode routing_mode() const override;
>+  bool is_comfort_noise_enabled() const override;
>+  int Enable(bool enable) override;
>+  int set_routing_mode(RoutingMode mode) override;
>+  int enable_comfort_noise(bool enable) override;
>+  int SetEchoPath(const void* echo_path, size_t size_bytes) override;
>+  int GetEchoPath(void* echo_path, size_t size_bytes) const override;
>+
>+ private:
>+  AudioProcessing* audio_processing_;
>+  EchoControlMobile* echo_control_mobile_;
>+
>+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(EchoControlMobileProxy);
>+};
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_ECHO_CONTROL_MOBILE_PROXY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_detector/circular_buffer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_detector/circular_buffer.cc
>index 0c6cc8a9333..a6d10edfe2b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_detector/circular_buffer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_detector/circular_buffer.cc
>@@ -28,9 +28,9 @@ void CircularBuffer::Push(float value) {
>   RTC_DCHECK_LE(nr_elements_in_buffer_, buffer_.size());
> }
> 
>-rtc::Optional<float> CircularBuffer::Pop() {
>+absl::optional<float> CircularBuffer::Pop() {
>   if (nr_elements_in_buffer_ == 0) {
>-    return rtc::nullopt;
>+    return absl::nullopt;
>   }
>   const size_t index =
>       (buffer_.size() + next_insertion_index_ - nr_elements_in_buffer_) %
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_detector/circular_buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_detector/circular_buffer.h
>index 53d4afb6d50..9f5cdfa9536 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_detector/circular_buffer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_detector/circular_buffer.h
>@@ -13,7 +13,7 @@
> 
> #include <vector>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> 
> namespace webrtc {
> 
>@@ -24,7 +24,7 @@ struct CircularBuffer {
>   ~CircularBuffer();
> 
>   void Push(float value);
>-  rtc::Optional<float> Pop();
>+  absl::optional<float> Pop();
>   size_t Size() const { return nr_elements_in_buffer_; }
>   // This function fills the buffer with zeros, but does not change its size.
>   void Clear();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_detector/circular_buffer_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_detector/circular_buffer_unittest.cc
>index 657bd05888a..0fa2a2b2f7a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_detector/circular_buffer_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/echo_detector/circular_buffer_unittest.cc
>@@ -46,7 +46,7 @@ TEST(CircularBufferTests, OverflowTest) {
> 
> TEST(CircularBufferTests, ReadFromEmpty) {
>   CircularBuffer test_buffer(3);
>-  EXPECT_EQ(rtc::nullopt, test_buffer.Pop());
>+  EXPECT_EQ(absl::nullopt, test_buffer.Pop());
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/gain_control_for_experimental_agc.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/gain_control_for_experimental_agc.cc
>index d5d978c2e0b..4ab856c2c33 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/gain_control_for_experimental_agc.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/gain_control_for_experimental_agc.cc
>@@ -12,6 +12,7 @@
> 
> #include "modules/audio_processing/include/audio_processing.h"
> #include "modules/audio_processing/logging/apm_data_dumper.h"
>+#include "rtc_base/atomicops.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/criticalsection.h"
> 
>@@ -22,12 +23,11 @@ int GainControlForExperimentalAgc::instance_counter_ = 0;
> GainControlForExperimentalAgc::GainControlForExperimentalAgc(
>     GainControl* gain_control,
>     rtc::CriticalSection* crit_capture)
>-    : data_dumper_(new ApmDataDumper(instance_counter_)),
>+    : data_dumper_(
>+          new ApmDataDumper(rtc::AtomicOps::Increment(&instance_counter_))),
>       real_gain_control_(gain_control),
>       volume_(0),
>-      crit_capture_(crit_capture) {
>-  instance_counter_++;
>-}
>+      crit_capture_(crit_capture) {}
> 
> GainControlForExperimentalAgc::~GainControlForExperimentalAgc() = default;
> 
>@@ -43,14 +43,18 @@ int GainControlForExperimentalAgc::set_stream_analog_level(int level) {
>   rtc::CritScope cs_capture(crit_capture_);
>   data_dumper_->DumpRaw("experimental_gain_control_set_stream_analog_level", 1,
>                         &level);
>+  do_log_level_ = true;
>   volume_ = level;
>   return AudioProcessing::kNoError;
> }
> 
> int GainControlForExperimentalAgc::stream_analog_level() {
>   rtc::CritScope cs_capture(crit_capture_);
>-  data_dumper_->DumpRaw("experimental_gain_control_stream_analog_level", 1,
>-                        &volume_);
>+  if (do_log_level_) {
>+    data_dumper_->DumpRaw("experimental_gain_control_stream_analog_level", 1,
>+                          &volume_);
>+    do_log_level_ = false;
>+  }
>   return volume_;
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/gain_control_for_experimental_agc.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/gain_control_for_experimental_agc.h
>index 0894a0e30e5..b20fbc96f97 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/gain_control_for_experimental_agc.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/gain_control_for_experimental_agc.h
>@@ -68,6 +68,7 @@ class GainControlForExperimentalAgc : public GainControl,
>   GainControl* real_gain_control_;
>   int volume_;
>   rtc::CriticalSection* crit_capture_;
>+  bool do_log_level_ = true;
>   static int instance_counter_;
>   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(GainControlForExperimentalAgc);
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/gain_control_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/gain_control_impl.cc
>index e550ebbfad0..685a27ff668 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/gain_control_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/gain_control_impl.cc
>@@ -10,7 +10,7 @@
> 
> #include "modules/audio_processing/gain_control_impl.h"
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> #include "modules/audio_processing/agc/legacy/gain_control.h"
> #include "modules/audio_processing/audio_buffer.h"
> #include "modules/audio_processing/logging/apm_data_dumper.h"
>@@ -67,9 +67,7 @@ class GainControlImpl::GainController {
>     set_capture_level(capture_level);
>   }
> 
>-  void set_capture_level(int capture_level) {
>-    capture_level_ = capture_level;
>-  }
>+  void set_capture_level(int capture_level) { capture_level_ = capture_level; }
> 
>   int get_capture_level() {
>     RTC_DCHECK(capture_level_);
>@@ -80,7 +78,7 @@ class GainControlImpl::GainController {
>   Handle* state_;
>   // TODO(peah): Remove the optional once the initialization is moved into the
>   // ctor.
>-  rtc::Optional<int> capture_level_;
>+  absl::optional<int> capture_level_;
> 
>   RTC_DISALLOW_COPY_AND_ASSIGN(GainController);
> };
>@@ -259,7 +257,7 @@ int GainControlImpl::stream_analog_level() {
>   data_dumper_->DumpRaw("gain_control_stream_analog_level", 1,
>                         &analog_capture_level_);
>   // TODO(ajm): enable this assertion?
>-  //RTC_DCHECK_EQ(kAdaptiveAnalog, mode_);
>+  // RTC_DCHECK_EQ(kAdaptiveAnalog, mode_);
> 
>   return analog_capture_level_;
> }
>@@ -303,8 +301,7 @@ GainControl::Mode GainControlImpl::mode() const {
>   return mode_;
> }
> 
>-int GainControlImpl::set_analog_level_limits(int minimum,
>-                                             int maximum) {
>+int GainControlImpl::set_analog_level_limits(int minimum, int maximum) {
>   if (minimum < 0) {
>     return AudioProcessing::kBadParameterError;
>   }
>@@ -419,11 +416,10 @@ int GainControlImpl::Configure() {
>   WebRtcAgcConfig config;
>   // TODO(ajm): Flip the sign here (since AGC expects a positive value) if we
>   //            change the interface.
>-  //RTC_DCHECK_LE(target_level_dbfs_, 0);
>-  //config.targetLevelDbfs = static_cast<int16_t>(-target_level_dbfs_);
>+  // RTC_DCHECK_LE(target_level_dbfs_, 0);
>+  // config.targetLevelDbfs = static_cast<int16_t>(-target_level_dbfs_);
>   config.targetLevelDbfs = static_cast<int16_t>(target_level_dbfs_);
>-  config.compressionGaindB =
>-      static_cast<int16_t>(compression_gain_db_);
>+  config.compressionGaindB = static_cast<int16_t>(compression_gain_db_);
>   config.limiterEnable = limiter_enabled_;
> 
>   int error = AudioProcessing::kNoError;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/gain_control_impl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/gain_control_impl.h
>index 26745065cfa..959422fb44c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/gain_control_impl.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/gain_control_impl.h
>@@ -86,8 +86,8 @@ class GainControlImpl : public GainControl {
> 
>   std::vector<std::unique_ptr<GainController>> gain_controllers_;
> 
>-  rtc::Optional<size_t> num_proc_channels_ RTC_GUARDED_BY(crit_capture_);
>-  rtc::Optional<int> sample_rate_hz_ RTC_GUARDED_BY(crit_capture_);
>+  absl::optional<size_t> num_proc_channels_ RTC_GUARDED_BY(crit_capture_);
>+  absl::optional<int> sample_rate_hz_ RTC_GUARDED_BY(crit_capture_);
> 
>   static int instance_counter_;
>   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(GainControlImpl);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/gain_controller2.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/gain_controller2.cc
>similarity index 70%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/gain_controller2.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/gain_controller2.cc
>index ca564e8b1d6..cdfcffa9d59 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/gain_controller2.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/gain_controller2.cc
>@@ -8,15 +8,13 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "modules/audio_processing/agc2/gain_controller2.h"
>-
>-#include <cmath>
>+#include "modules/audio_processing/gain_controller2.h"
> 
> #include "modules/audio_processing/audio_buffer.h"
>+#include "modules/audio_processing/include/audio_frame_view.h"
> #include "modules/audio_processing/logging/apm_data_dumper.h"
> #include "rtc_base/atomicops.h"
> #include "rtc_base/checks.h"
>-#include "rtc_base/numerics/safe_minmax.h"
> 
> namespace webrtc {
> 
>@@ -25,8 +23,8 @@ int GainController2::instance_count_ = 0;
> GainController2::GainController2()
>     : data_dumper_(
>           new ApmDataDumper(rtc::AtomicOps::Increment(&instance_count_))),
>-      sample_rate_hz_(AudioProcessing::kSampleRate48kHz),
>-      fixed_gain_(1.f) {}
>+      fixed_gain_controller_(data_dumper_.get()),
>+      adaptive_agc_(data_dumper_.get()) {}
> 
> GainController2::~GainController2() = default;
> 
>@@ -35,28 +33,30 @@ void GainController2::Initialize(int sample_rate_hz) {
>              sample_rate_hz == AudioProcessing::kSampleRate16kHz ||
>              sample_rate_hz == AudioProcessing::kSampleRate32kHz ||
>              sample_rate_hz == AudioProcessing::kSampleRate48kHz);
>-  sample_rate_hz_ = sample_rate_hz;
>+  fixed_gain_controller_.SetSampleRate(sample_rate_hz);
>   data_dumper_->InitiateNewSetOfRecordings();
>-  data_dumper_->DumpRaw("sample_rate_hz", sample_rate_hz_);
>-  data_dumper_->DumpRaw("fixed_gain_linear", fixed_gain_);
>+  data_dumper_->DumpRaw("sample_rate_hz", sample_rate_hz);
> }
> 
> void GainController2::Process(AudioBuffer* audio) {
>-  if (fixed_gain_ == 1.f)
>-    return;
>+  AudioFrameView<float> float_frame(audio->channels_f(), audio->num_channels(),
>+                                    audio->num_frames());
>+  adaptive_agc_.Process(float_frame);
>+  fixed_gain_controller_.Process(float_frame);
>+}
> 
>-  for (size_t k = 0; k < audio->num_channels(); ++k) {
>-    for (size_t j = 0; j < audio->num_frames(); ++j) {
>-      audio->channels_f()[k][j] = rtc::SafeClamp(
>-          fixed_gain_ * audio->channels_f()[k][j], -32768.f, 32767.f);
>-    }
>+void GainController2::NotifyAnalogLevel(int level) {
>+  if (analog_level_ != level) {
>+    adaptive_agc_.Reset();
>   }
>+  analog_level_ = level;
> }
> 
> void GainController2::ApplyConfig(
>     const AudioProcessing::Config::GainController2& config) {
>   RTC_DCHECK(Validate(config));
>-  fixed_gain_ = std::pow(10.f, config.fixed_gain_db / 20.f);
>+  config_ = config;
>+  fixed_gain_controller_.SetGain(config_.fixed_gain_db);
> }
> 
> bool GainController2::Validate(
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/gain_controller2.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/gain_controller2.h
>similarity index 73%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/gain_controller2.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/gain_controller2.h
>index 11706870f43..41b7b6919ea 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/gain_controller2.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/gain_controller2.h
>@@ -8,12 +8,14 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#ifndef MODULES_AUDIO_PROCESSING_AGC2_GAIN_CONTROLLER2_H_
>-#define MODULES_AUDIO_PROCESSING_AGC2_GAIN_CONTROLLER2_H_
>+#ifndef MODULES_AUDIO_PROCESSING_GAIN_CONTROLLER2_H_
>+#define MODULES_AUDIO_PROCESSING_GAIN_CONTROLLER2_H_
> 
> #include <memory>
> #include <string>
> 
>+#include "modules/audio_processing/agc2/adaptive_agc.h"
>+#include "modules/audio_processing/agc2/fixed_gain_controller.h"
> #include "modules/audio_processing/include/audio_processing.h"
> #include "rtc_base/constructormagic.h"
> 
>@@ -24,8 +26,6 @@ class AudioBuffer;
> 
> // Gain Controller 2 aims to automatically adjust levels by acting on the
> // microphone gain and/or applying digital gain.
>-//
>-// Temporarily implements a fixed gain mode with hard-clipping.
> class GainController2 {
>  public:
>   GainController2();
>@@ -33,6 +33,7 @@ class GainController2 {
> 
>   void Initialize(int sample_rate_hz);
>   void Process(AudioBuffer* audio);
>+  void NotifyAnalogLevel(int level);
> 
>   void ApplyConfig(const AudioProcessing::Config::GainController2& config);
>   static bool Validate(const AudioProcessing::Config::GainController2& config);
>@@ -42,12 +43,14 @@ class GainController2 {
>  private:
>   static int instance_count_;
>   std::unique_ptr<ApmDataDumper> data_dumper_;
>-  int sample_rate_hz_;
>-  float fixed_gain_;
>+  FixedGainController fixed_gain_controller_;
>+  AudioProcessing::Config::GainController2 config_;
>+  AdaptiveAgc adaptive_agc_;
>+  int analog_level_ = -1;
> 
>   RTC_DISALLOW_COPY_AND_ASSIGN(GainController2);
> };
> 
> }  // namespace webrtc
> 
>-#endif  // MODULES_AUDIO_PROCESSING_AGC2_GAIN_CONTROLLER2_H_
>+#endif  // MODULES_AUDIO_PROCESSING_GAIN_CONTROLLER2_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/gain_controller2_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/gain_controller2_unittest.cc
>similarity index 90%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/gain_controller2_unittest.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/gain_controller2_unittest.cc
>index 46f654db62a..5bedccd64a8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/agc2/gain_controller2_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/gain_controller2_unittest.cc
>@@ -11,8 +11,8 @@
> #include <algorithm>
> 
> #include "api/array_view.h"
>-#include "modules/audio_processing/agc2/gain_controller2.h"
> #include "modules/audio_processing/audio_buffer.h"
>+#include "modules/audio_processing/gain_controller2.h"
> #include "rtc_base/checks.h"
> #include "test/gtest.h"
> 
>@@ -80,12 +80,6 @@ TEST(GainController2, Usage) {
>   SetAudioBufferSamples(sample_value, &ab);
>   AudioProcessing::Config::GainController2 config;
> 
>-  // Check that samples are not modified when the fixed gain is 0 dB.
>-  ASSERT_EQ(config.fixed_gain_db, 0.f);
>-  gain_controller2->ApplyConfig(config);
>-  gain_controller2->Process(&ab);
>-  EXPECT_EQ(ab.channels_f()[0][0], sample_value);
>-
>   // Check that samples are amplified when the fixed gain is greater than 0 dB.
>   config.fixed_gain_db = 5.f;
>   gain_controller2->ApplyConfig(config);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/aec_dump.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/aec_dump.cc
>index 365d01510ec..c243b523a37 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/aec_dump.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/aec_dump.cc
>@@ -35,6 +35,9 @@ bool InternalAPMConfig::operator==(const InternalAPMConfig& other) {
>          intelligibility_enhancer_enabled ==
>              other.intelligibility_enhancer_enabled &&
>          noise_robust_agc_enabled == other.noise_robust_agc_enabled &&
>+         pre_amplifier_enabled == other.pre_amplifier_enabled &&
>+         pre_amplifier_fixed_gain_factor ==
>+             other.pre_amplifier_fixed_gain_factor &&
>          experiments_description == other.experiments_description;
> }
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/aec_dump.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/aec_dump.h
>index 0c8d2271ccb..95c010b0a3e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/aec_dump.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/aec_dump.h
>@@ -16,11 +16,12 @@
> #include <vector>
> 
> #include "api/array_view.h"
>+#include "api/audio/audio_frame.h"
>+#include "modules/audio_processing/include/audio_frame_view.h"
>+#include "modules/audio_processing/include/audio_processing.h"
> 
> namespace webrtc {
> 
>-class AudioFrame;
>-
> // Struct for passing current config from APM without having to
> // include protobuf headers.
> struct InternalAPMConfig {
>@@ -50,52 +51,11 @@ struct InternalAPMConfig {
>   bool transient_suppression_enabled = false;
>   bool intelligibility_enhancer_enabled = false;
>   bool noise_robust_agc_enabled = false;
>+  bool pre_amplifier_enabled = false;
>+  float pre_amplifier_fixed_gain_factor = 1.f;
>   std::string experiments_description = "";
> };
> 
>-struct InternalAPMStreamsConfig {
>-  int input_sample_rate = 0;
>-  int output_sample_rate = 0;
>-  int render_input_sample_rate = 0;
>-  int render_output_sample_rate = 0;
>-
>-  size_t input_num_channels = 0;
>-  size_t output_num_channels = 0;
>-  size_t render_input_num_channels = 0;
>-  size_t render_output_num_channels = 0;
>-};
>-
>-// Class to pass audio data in float** format. This is to avoid
>-// dependence on AudioBuffer, and avoid problems associated with
>-// rtc::ArrayView<rtc::ArrayView>.
>-class FloatAudioFrame {
>- public:
>-  // |num_channels| and |channel_size| describe the float**
>-  // |audio_samples|. |audio_samples| is assumed to point to a
>-  // two-dimensional |num_channels * channel_size| array of floats.
>-  FloatAudioFrame(const float* const* audio_samples,
>-                  size_t num_channels,
>-                  size_t channel_size)
>-      : audio_samples_(audio_samples),
>-        num_channels_(num_channels),
>-        channel_size_(channel_size) {}
>-
>-  FloatAudioFrame() = delete;
>-
>-  size_t num_channels() const { return num_channels_; }
>-
>-  rtc::ArrayView<const float> channel(size_t idx) const {
>-    RTC_DCHECK_LE(0, idx);
>-    RTC_DCHECK_LE(idx, num_channels_);
>-    return rtc::ArrayView<const float>(audio_samples_[idx], channel_size_);
>-  }
>-
>- private:
>-  const float* const* audio_samples_;
>-  size_t num_channels_;
>-  size_t channel_size_;
>-};
>-
> // An interface for recording configuration and input/output streams
> // of the Audio Processing Module. The recordings are called
> // 'aec-dumps' and are stored in a protobuf format defined in
>@@ -116,14 +76,19 @@ class AecDump {
>   virtual ~AecDump() = default;
> 
>   // Logs Event::Type INIT message.
>-  virtual void WriteInitMessage(
>-      const InternalAPMStreamsConfig& streams_config) = 0;
>+  virtual void WriteInitMessage(const ProcessingConfig& api_format,
>+                                int64_t time_now_ms) = 0;
>+  RTC_DEPRECATED void WriteInitMessage(const ProcessingConfig& api_format) {
>+    WriteInitMessage(api_format, 0);
>+  }
> 
>   // Logs Event::Type STREAM message. To log an input/output pair,
>   // call the AddCapture* and AddAudioProcessingState methods followed
>   // by a WriteCaptureStreamMessage call.
>-  virtual void AddCaptureStreamInput(const FloatAudioFrame& src) = 0;
>-  virtual void AddCaptureStreamOutput(const FloatAudioFrame& src) = 0;
>+  virtual void AddCaptureStreamInput(
>+      const AudioFrameView<const float>& src) = 0;
>+  virtual void AddCaptureStreamOutput(
>+      const AudioFrameView<const float>& src) = 0;
>   virtual void AddCaptureStreamInput(const AudioFrame& frame) = 0;
>   virtual void AddCaptureStreamOutput(const AudioFrame& frame) = 0;
>   virtual void AddAudioProcessingState(const AudioProcessingState& state) = 0;
>@@ -131,7 +96,8 @@ class AecDump {
> 
>   // Logs Event::Type REVERSE_STREAM message.
>   virtual void WriteRenderStreamMessage(const AudioFrame& frame) = 0;
>-  virtual void WriteRenderStreamMessage(const FloatAudioFrame& src) = 0;
>+  virtual void WriteRenderStreamMessage(
>+      const AudioFrameView<const float>& src) = 0;
> 
>   // Logs Event::Type CONFIG message.
>   virtual void WriteConfig(const InternalAPMConfig& config) = 0;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/audio_frame_view.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/audio_frame_view.h
>new file mode 100644
>index 00000000000..ab5779ad6bb
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/audio_frame_view.h
>@@ -0,0 +1,67 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_FRAME_VIEW_H_
>+#define MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_FRAME_VIEW_H_
>+
>+#include "api/array_view.h"
>+
>+namespace webrtc {
>+
>+// Class to pass audio data in T** format, where T is a numeric type.
>+template <class T>
>+class AudioFrameView {
>+ public:
>+  // |num_channels| and |channel_size| describe the T**
>+  // |audio_samples|. |audio_samples| is assumed to point to a
>+  // two-dimensional |num_channels * channel_size| array of floats.
>+  AudioFrameView(T* const* audio_samples,
>+                 size_t num_channels,
>+                 size_t channel_size)
>+      : audio_samples_(audio_samples),
>+        num_channels_(num_channels),
>+        channel_size_(channel_size) {}
>+
>+  // Implicit cast to allow converting Frame<float> to
>+  // Frame<const float>.
>+  template <class U>
>+  AudioFrameView(AudioFrameView<U> other)
>+      : audio_samples_(other.data()),
>+        num_channels_(other.num_channels()),
>+        channel_size_(other.samples_per_channel()) {}
>+
>+  AudioFrameView() = delete;
>+
>+  size_t num_channels() const { return num_channels_; }
>+
>+  size_t samples_per_channel() const { return channel_size_; }
>+
>+  rtc::ArrayView<T> channel(size_t idx) {
>+    RTC_DCHECK_LE(0, idx);
>+    RTC_DCHECK_LE(idx, num_channels_);
>+    return rtc::ArrayView<T>(audio_samples_[idx], channel_size_);
>+  }
>+
>+  rtc::ArrayView<const T> channel(size_t idx) const {
>+    RTC_DCHECK_LE(0, idx);
>+    RTC_DCHECK_LE(idx, num_channels_);
>+    return rtc::ArrayView<const T>(audio_samples_[idx], channel_size_);
>+  }
>+
>+  T* const* data() { return audio_samples_; }
>+
>+ private:
>+  T* const* audio_samples_;
>+  size_t num_channels_;
>+  size_t channel_size_;
>+};
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_FRAME_VIEW_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/audio_generator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/audio_generator.h
>new file mode 100644
>index 00000000000..77e62849691
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/audio_generator.h
>@@ -0,0 +1,36 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_GENERATOR_H_
>+#define MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_GENERATOR_H_
>+
>+#include "modules/audio_processing/include/audio_frame_view.h"
>+
>+namespace webrtc {
>+// This class is used as input sink for the APM, for diagnostic purposes.
>+// Generates an infinite audio signal, [-1, 1] floating point values, in frames
>+// of fixed channel count and sample rate.
>+class AudioGenerator {
>+ public:
>+  virtual ~AudioGenerator() {}
>+
>+  // Fill |audio| with the next samples of the audio signal.
>+  virtual void FillFrame(AudioFrameView<float> audio) = 0;
>+
>+  // Return the number of channels output by the AudioGenerator.
>+  virtual size_t NumChannels() = 0;
>+
>+  // Return the sample rate output by the AudioGenerator.
>+  virtual size_t SampleRateHz() = 0;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_GENERATOR_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/audio_generator_factory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/audio_generator_factory.cc
>new file mode 100644
>index 00000000000..69a54015145
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/audio_generator_factory.cc
>@@ -0,0 +1,25 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/audio_processing/include/audio_generator_factory.h"
>+
>+#include "absl/memory/memory.h"
>+#include "common_audio/wav_file.h"
>+#include "modules/audio_processing/audio_generator/file_audio_generator.h"
>+
>+namespace webrtc {
>+
>+std::unique_ptr<AudioGenerator> AudioGeneratorFactory::Create(
>+    const std::string& file_name) {
>+  std::unique_ptr<WavReader> input_audio_file(new WavReader(file_name));
>+  return absl::make_unique<FileAudioGenerator>(std::move(input_audio_file));
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/audio_generator_factory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/audio_generator_factory.h
>new file mode 100644
>index 00000000000..a9513ef88d0
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/audio_generator_factory.h
>@@ -0,0 +1,31 @@
>+/*
>+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_GENERATOR_FACTORY_H_
>+#define MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_GENERATOR_FACTORY_H_
>+
>+#include <memory>
>+#include <string>
>+#include <utility>
>+
>+#include "modules/audio_processing/include/audio_generator.h"
>+
>+namespace webrtc {
>+
>+class AudioGeneratorFactory {
>+ public:
>+  // Creates an AudioGenerator that reads the playout audio from a given 16-bit
>+  // int-encoded WAV file.
>+  static std::unique_ptr<AudioGenerator> Create(const std::string& file_name);
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_GENERATOR_FACTORY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/audio_processing.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/audio_processing.cc
>index 8410a3dfbbf..75eedafc292 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/audio_processing.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/audio_processing.cc
>@@ -14,22 +14,7 @@
> 
> namespace webrtc {
> 
>-Beamforming::Beamforming()
>-    : enabled(false),
>-      array_geometry(),
>-      target_direction(
>-          SphericalPointf(static_cast<float>(M_PI) / 2.f, 0.f, 1.f)) {}
>-Beamforming::Beamforming(bool enabled, const std::vector<Point>& array_geometry)
>-    : Beamforming(enabled,
>-                  array_geometry,
>-                  SphericalPointf(static_cast<float>(M_PI) / 2.f, 0.f, 1.f)) {}
>+void CustomProcessing::SetRuntimeSetting(
>+    AudioProcessing::RuntimeSetting setting) {}
> 
>-Beamforming::Beamforming(bool enabled,
>-                         const std::vector<Point>& array_geometry,
>-                         SphericalPointf target_direction)
>-    : enabled(enabled),
>-      array_geometry(array_geometry),
>-      target_direction(target_direction) {}
>-
>-Beamforming::~Beamforming() {}
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/audio_processing.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/audio_processing.h
>index 07218264f4f..9c750f06925 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/audio_processing.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/audio_processing.h
>@@ -18,20 +18,22 @@
> 
> #include <math.h>
> #include <stddef.h>  // size_t
>-#include <stdio.h>  // FILE
>+#include <stdio.h>   // FILE
> #include <string.h>
> #include <vector>
> 
>-#include "api/optional.h"
>-#include "modules/audio_processing/beamformer/array_util.h"
>+#include "absl/types/optional.h"
>+#include "api/audio/echo_canceller3_config.h"
>+#include "api/audio/echo_control.h"
>+#include "modules/audio_processing/include/audio_generator.h"
> #include "modules/audio_processing/include/audio_processing_statistics.h"
> #include "modules/audio_processing/include/config.h"
>+#include "modules/audio_processing/include/gain_control.h"
> #include "rtc_base/arraysize.h"
> #include "rtc_base/deprecation.h"
> #include "rtc_base/platform_file.h"
> #include "rtc_base/refcount.h"
> #include "rtc_base/scoped_ref_ptr.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -41,14 +43,11 @@ class AecDump;
> class AudioBuffer;
> class AudioFrame;
> 
>-class NonlinearBeamformer;
>-
> class StreamConfig;
> class ProcessingConfig;
> 
> class EchoCancellation;
> class EchoControlMobile;
>-class EchoControlFactory;
> class EchoDetector;
> class GainControl;
> class HighPassFilter;
>@@ -57,9 +56,6 @@ class NoiseSuppression;
> class CustomProcessing;
> class VoiceDetection;
> 
>-// webrtc:8665, addedd temporarily to avoid breaking dependencies.
>-typedef CustomProcessing PostProcessing;
>-
> // Use to enable the extended filter mode in the AEC, along with robustness
> // measures around the reported system delays. It comes with a significant
> // increase in AEC complexity, but is much more robust to unreliable reported
>@@ -123,6 +119,13 @@ static constexpr int kClippedLevelMin = 70;
> struct ExperimentalAgc {
>   ExperimentalAgc() = default;
>   explicit ExperimentalAgc(bool enabled) : enabled(enabled) {}
>+  ExperimentalAgc(bool enabled,
>+                  bool enabled_agc2_level_estimator,
>+                  bool digital_adaptive_disabled)
>+      : enabled(enabled),
>+        enabled_agc2_level_estimator(enabled_agc2_level_estimator),
>+        digital_adaptive_disabled(digital_adaptive_disabled) {}
>+
>   ExperimentalAgc(bool enabled, int startup_min_volume)
>       : enabled(enabled), startup_min_volume(startup_min_volume) {}
>   ExperimentalAgc(bool enabled, int startup_min_volume, int clipped_level_min)
>@@ -134,6 +137,8 @@ struct ExperimentalAgc {
>   int startup_min_volume = kAgcStartupMinVolume;
>   // Lowest microphone level that will be applied in response to clipping.
>   int clipped_level_min = kClippedLevelMin;
>+  bool enabled_agc2_level_estimator = false;
>+  bool digital_adaptive_disabled = false;
> };
> 
> // Use to enable experimental noise suppression. It can be set in the
>@@ -145,22 +150,6 @@ struct ExperimentalNs {
>   bool enabled;
> };
> 
>-// Use to enable beamforming. Must be provided through the constructor. It will
>-// have no impact if used with AudioProcessing::SetExtraOptions().
>-struct Beamforming {
>-  Beamforming();
>-  Beamforming(bool enabled, const std::vector<Point>& array_geometry);
>-  Beamforming(bool enabled,
>-              const std::vector<Point>& array_geometry,
>-              SphericalPointf target_direction);
>-  ~Beamforming();
>-
>-  static const ConfigOptionID identifier = ConfigOptionID::kBeamforming;
>-  const bool enabled;
>-  const std::vector<Point> array_geometry;
>-  const SphericalPointf target_direction;
>-};
>-
> // Use to enable intelligibility enhancer in audio processing.
> //
> // Note: If enabled and the reverse stream has more than one output channel,
>@@ -209,8 +198,8 @@ struct Intelligibility {
> // AudioProcessing* apm = AudioProcessingBuilder().Create();
> //
> // AudioProcessing::Config config;
>-// config.level_controller.enabled = true;
> // config.high_pass_filter.enabled = true;
>+// config.gain_controller2.enabled = true;
> // apm->ApplyConfig(config)
> //
> // apm->echo_cancellation()->enable_drift_compensation(false);
>@@ -260,14 +249,15 @@ class AudioProcessing : public rtc::RefCountInterface {
>   // by changing the default values in the AudioProcessing::Config struct.
>   // The config is applied by passing the struct to the ApplyConfig method.
>   struct Config {
>-    struct LevelController {
>+    // TODO(bugs.webrtc.org/9535): Currently unused. Use this to determine AEC.
>+    struct EchoCanceller {
>       bool enabled = false;
>+      bool mobile_mode = false;
>+      // Recommended not to use. Will be removed in the future.
>+      // APM components are not fine-tuned for legacy suppression levels.
>+      bool legacy_moderate_suppression_level = false;
>+    } echo_canceller;
> 
>-      // Sets the initial peak level to use inside the level controller in order
>-      // to compute the signal gain. The unit for the peak level is dBFS and
>-      // the allowed range is [-100, 0].
>-      float initial_peak_level_dbfs = -6.0206f;
>-    } level_controller;
>     struct ResidualEchoDetector {
>       bool enabled = true;
>     } residual_echo_detector;
>@@ -276,16 +266,17 @@ class AudioProcessing : public rtc::RefCountInterface {
>       bool enabled = false;
>     } high_pass_filter;
> 
>-    // Deprecated way of activating AEC3.
>-    // TODO(gustaf): Remove when possible.
>-    struct EchoCanceller3 {
>+    // Enabled the pre-amplifier. It amplifies the capture signal
>+    // before any other processing is done.
>+    struct PreAmplifier {
>       bool enabled = false;
>-    } echo_canceller3;
>+      float fixed_gain_factor = 1.f;
>+    } pre_amplifier;
> 
>-    // Enables the next generation AGC functionality. This feature replaces the
>-    // standard methods of gain control in the previous AGC.
>-    // The functionality is not yet activated in the code and turning this on
>-    // does not yet have the desired behavior.
>+    // Enables the next generation AGC functionality. This feature
>+    // replaces the standard methods of gain control in the previous
>+    // AGC. This functionality is currently only partially
>+    // implemented.
>     struct GainController2 {
>       bool enabled = false;
>       float fixed_gain_db = 0.f;
>@@ -314,38 +305,40 @@ class AudioProcessing : public rtc::RefCountInterface {
>     kStereoAndKeyboard
>   };
> 
>-  // Creates an APM instance. Use one instance for every primary audio stream
>-  // requiring processing. On the client-side, this would typically be one
>-  // instance for the near-end stream, and additional instances for each far-end
>-  // stream which requires processing. On the server-side, this would typically
>-  // be one instance for every incoming stream.
>-  // The Create functions are deprecated, please use AudioProcessingBuilder
>-  // instead.
>-  // TODO(bugs.webrtc.org/8668): Remove these Create functions when all callers
>-  // have moved to AudioProcessingBuilder.
>-  static AudioProcessing* Create();
>-  // Allows passing in an optional configuration at create-time.
>-  static AudioProcessing* Create(const webrtc::Config& config);
>-  // Deprecated. Use the Create below, with nullptr CustomProcessing.
>-  RTC_DEPRECATED
>-  static AudioProcessing* Create(const webrtc::Config& config,
>-                                 NonlinearBeamformer* beamformer);
>-
>-  // Will be deprecated and removed as part of webrtc:8665. Use the
>-  // Create below, with nullptr CustomProcessing.
>-  static AudioProcessing* Create(
>-      const webrtc::Config& config,
>-      std::unique_ptr<CustomProcessing> capture_post_processor,
>-      std::unique_ptr<EchoControlFactory> echo_control_factory,
>-      NonlinearBeamformer* beamformer);
>-
>-  // Allows passing in optional user-defined processing modules.
>-  static AudioProcessing* Create(
>-      const webrtc::Config& config,
>-      std::unique_ptr<CustomProcessing> capture_post_processor,
>-      std::unique_ptr<CustomProcessing> render_pre_processor,
>-      std::unique_ptr<EchoControlFactory> echo_control_factory,
>-      NonlinearBeamformer* beamformer);
>+  // Specifies the properties of a setting to be passed to AudioProcessing at
>+  // runtime.
>+  class RuntimeSetting {
>+   public:
>+    enum class Type {
>+      kNotSpecified,
>+      kCapturePreGain,
>+      kCustomRenderProcessingRuntimeSetting
>+    };
>+
>+    RuntimeSetting() : type_(Type::kNotSpecified), value_(0.f) {}
>+    ~RuntimeSetting() = default;
>+
>+    static RuntimeSetting CreateCapturePreGain(float gain) {
>+      RTC_DCHECK_GE(gain, 1.f) << "Attenuation is not allowed.";
>+      return {Type::kCapturePreGain, gain};
>+    }
>+
>+    static RuntimeSetting CreateCustomRenderSetting(float payload) {
>+      return {Type::kCustomRenderProcessingRuntimeSetting, payload};
>+    }
>+
>+    Type type() const { return type_; }
>+    void GetFloat(float* value) const {
>+      RTC_DCHECK(value);
>+      *value = value_;
>+    }
>+
>+   private:
>+    RuntimeSetting(Type id, float value) : type_(id), value_(value) {}
>+    Type type_;
>+    float value_;
>+  };
>+
>   ~AudioProcessing() override {}
> 
>   // Initializes internal states, while retaining all user settings. This
>@@ -403,6 +396,9 @@ class AudioProcessing : public rtc::RefCountInterface {
>   // Default false.
>   virtual void set_output_will_be_muted(bool muted) = 0;
> 
>+  // Enqueue a runtime setting.
>+  virtual void SetRuntimeSetting(RuntimeSetting setting) = 0;
>+
>   // Processes a 10 ms |frame| of the primary audio stream. On the client-side,
>   // this is the near-end (or captured) audio.
>   //
>@@ -516,6 +512,16 @@ class AudioProcessing : public rtc::RefCountInterface {
>   // all pending logging tasks are completed.
>   virtual void DetachAecDump() = 0;
> 
>+  // Attaches provided webrtc::AudioGenerator for modifying playout audio.
>+  // Calling this method when another AudioGenerator is attached replaces the
>+  // active AudioGenerator with a new one.
>+  virtual void AttachPlayoutAudioGenerator(
>+      std::unique_ptr<AudioGenerator> audio_generator) = 0;
>+
>+  // If no AudioGenerator is attached, this has no effect. If an AecDump is
>+  // attached, its destructor is called.
>+  virtual void DetachPlayoutAudioGenerator() = 0;
>+
>   // Use to send UMA histograms at end of a call. Note that all histogram
>   // specific member variables are reset.
>   virtual void UpdateHistogramsOnCallEnd() = 0;
>@@ -663,12 +669,9 @@ class AudioProcessingBuilder {
>   // The AudioProcessingBuilder takes ownership of the render_pre_processing.
>   AudioProcessingBuilder& SetRenderPreProcessing(
>       std::unique_ptr<CustomProcessing> render_pre_processing);
>-  // The AudioProcessingBuilder takes ownership of the nonlinear beamformer.
>-  AudioProcessingBuilder& SetNonlinearBeamformer(
>-      std::unique_ptr<NonlinearBeamformer> nonlinear_beamformer);
>   // The AudioProcessingBuilder takes ownership of the echo_detector.
>   AudioProcessingBuilder& SetEchoDetector(
>-      std::unique_ptr<EchoDetector> echo_detector);
>+      rtc::scoped_refptr<EchoDetector> echo_detector);
>   // This creates an APM instance using the previously set components. Calling
>   // the Create function resets the AudioProcessingBuilder to its initial state.
>   AudioProcessing* Create();
>@@ -678,8 +681,7 @@ class AudioProcessingBuilder {
>   std::unique_ptr<EchoControlFactory> echo_control_factory_;
>   std::unique_ptr<CustomProcessing> capture_post_processing_;
>   std::unique_ptr<CustomProcessing> render_pre_processing_;
>-  std::unique_ptr<NonlinearBeamformer> nonlinear_beamformer_;
>-  std::unique_ptr<EchoDetector> echo_detector_;
>+  rtc::scoped_refptr<EchoDetector> echo_detector_;
>   RTC_DISALLOW_COPY_AND_ASSIGN(AudioProcessingBuilder);
> };
> 
>@@ -732,8 +734,8 @@ class StreamConfig {
> 
>  private:
>   static size_t calculate_frames(int sample_rate_hz) {
>-    return static_cast<size_t>(
>-        AudioProcessing::kChunkSizeMs * sample_rate_hz / 1000);
>+    return static_cast<size_t>(AudioProcessing::kChunkSizeMs * sample_rate_hz /
>+                               1000);
>   }
> 
>   int sample_rate_hz_;
>@@ -882,7 +884,8 @@ class EchoCancellation {
>   // Deprecated. Use GetStatistics on the AudioProcessing interface instead.
>   virtual int GetDelayMetrics(int* median, int* std) = 0;
>   // Deprecated. Use GetStatistics on the AudioProcessing interface instead.
>-  virtual int GetDelayMetrics(int* median, int* std,
>+  virtual int GetDelayMetrics(int* median,
>+                              int* std,
>                               float* fraction_poor_delays) = 0;
> 
>   // Returns a pointer to the low level AEC component.  In case of multiple
>@@ -951,128 +954,6 @@ class EchoControlMobile {
>   virtual ~EchoControlMobile() {}
> };
> 
>-// Interface for an acoustic echo cancellation (AEC) submodule.
>-class EchoControl {
>- public:
>-  // Analysis (not changing) of the render signal.
>-  virtual void AnalyzeRender(AudioBuffer* render) = 0;
>-
>-  // Analysis (not changing) of the capture signal.
>-  virtual void AnalyzeCapture(AudioBuffer* capture) = 0;
>-
>-  // Processes the capture signal in order to remove the echo.
>-  virtual void ProcessCapture(AudioBuffer* capture, bool echo_path_change) = 0;
>-
>-  struct Metrics {
>-    double echo_return_loss;
>-    double echo_return_loss_enhancement;
>-    int delay_ms;
>-  };
>-
>-  // Collect current metrics from the echo controller.
>-  virtual Metrics GetMetrics() const = 0;
>-
>-  virtual ~EchoControl() {}
>-};
>-
>-// Interface for a factory that creates EchoControllers.
>-class EchoControlFactory {
>- public:
>-  virtual std::unique_ptr<EchoControl> Create(int sample_rate_hz) = 0;
>-  virtual ~EchoControlFactory() = default;
>-};
>-
>-// The automatic gain control (AGC) component brings the signal to an
>-// appropriate range. This is done by applying a digital gain directly and, in
>-// the analog mode, prescribing an analog gain to be applied at the audio HAL.
>-//
>-// Recommended to be enabled on the client-side.
>-class GainControl {
>- public:
>-  virtual int Enable(bool enable) = 0;
>-  virtual bool is_enabled() const = 0;
>-
>-  // When an analog mode is set, this must be called prior to |ProcessStream()|
>-  // to pass the current analog level from the audio HAL. Must be within the
>-  // range provided to |set_analog_level_limits()|.
>-  virtual int set_stream_analog_level(int level) = 0;
>-
>-  // When an analog mode is set, this should be called after |ProcessStream()|
>-  // to obtain the recommended new analog level for the audio HAL. It is the
>-  // users responsibility to apply this level.
>-  virtual int stream_analog_level() = 0;
>-
>-  enum Mode {
>-    // Adaptive mode intended for use if an analog volume control is available
>-    // on the capture device. It will require the user to provide coupling
>-    // between the OS mixer controls and AGC through the |stream_analog_level()|
>-    // functions.
>-    //
>-    // It consists of an analog gain prescription for the audio device and a
>-    // digital compression stage.
>-    kAdaptiveAnalog,
>-
>-    // Adaptive mode intended for situations in which an analog volume control
>-    // is unavailable. It operates in a similar fashion to the adaptive analog
>-    // mode, but with scaling instead applied in the digital domain. As with
>-    // the analog mode, it additionally uses a digital compression stage.
>-    kAdaptiveDigital,
>-
>-    // Fixed mode which enables only the digital compression stage also used by
>-    // the two adaptive modes.
>-    //
>-    // It is distinguished from the adaptive modes by considering only a
>-    // short time-window of the input signal. It applies a fixed gain through
>-    // most of the input level range, and compresses (gradually reduces gain
>-    // with increasing level) the input signal at higher levels. This mode is
>-    // preferred on embedded devices where the capture signal level is
>-    // predictable, so that a known gain can be applied.
>-    kFixedDigital
>-  };
>-
>-  virtual int set_mode(Mode mode) = 0;
>-  virtual Mode mode() const = 0;
>-
>-  // Sets the target peak |level| (or envelope) of the AGC in dBFs (decibels
>-  // from digital full-scale). The convention is to use positive values. For
>-  // instance, passing in a value of 3 corresponds to -3 dBFs, or a target
>-  // level 3 dB below full-scale. Limited to [0, 31].
>-  //
>-  // TODO(ajm): use a negative value here instead, if/when VoE will similarly
>-  //            update its interface.
>-  virtual int set_target_level_dbfs(int level) = 0;
>-  virtual int target_level_dbfs() const = 0;
>-
>-  // Sets the maximum |gain| the digital compression stage may apply, in dB. A
>-  // higher number corresponds to greater compression, while a value of 0 will
>-  // leave the signal uncompressed. Limited to [0, 90].
>-  virtual int set_compression_gain_db(int gain) = 0;
>-  virtual int compression_gain_db() const = 0;
>-
>-  // When enabled, the compression stage will hard limit the signal to the
>-  // target level. Otherwise, the signal will be compressed but not limited
>-  // above the target level.
>-  virtual int enable_limiter(bool enable) = 0;
>-  virtual bool is_limiter_enabled() const = 0;
>-
>-  // Sets the |minimum| and |maximum| analog levels of the audio capture device.
>-  // Must be set if and only if an analog mode is used. Limited to [0, 65535].
>-  virtual int set_analog_level_limits(int minimum,
>-                                      int maximum) = 0;
>-  virtual int analog_level_minimum() const = 0;
>-  virtual int analog_level_maximum() const = 0;
>-
>-  // Returns true if the AGC has detected a saturation event (period where the
>-  // signal reaches digital full-scale) in the current frame and the analog
>-  // level cannot be reduced.
>-  //
>-  // This could be used as an indicator to reduce or disable analog mic gain at
>-  // the audio HAL.
>-  virtual bool stream_is_saturated() const = 0;
>-
>- protected:
>-  virtual ~GainControl() {}
>-};
> // TODO(peah): Remove this interface.
> // A filtering component which removes DC offset and low-frequency noise.
> // Recommended to be enabled on the client-side.
>@@ -1117,12 +998,7 @@ class NoiseSuppression {
> 
>   // Determines the aggressiveness of the suppression. Increasing the level
>   // will reduce the noise level at the expense of a higher speech distortion.
>-  enum Level {
>-    kLow,
>-    kModerate,
>-    kHigh,
>-    kVeryHigh
>-  };
>+  enum Level { kLow, kModerate, kHigh, kVeryHigh };
> 
>   virtual int set_level(Level level) = 0;
>   virtual Level level() const = 0;
>@@ -1148,15 +1024,21 @@ class CustomProcessing {
>   virtual void Process(AudioBuffer* audio) = 0;
>   // Returns a string representation of the module state.
>   virtual std::string ToString() const = 0;
>+  // Handles RuntimeSettings. TODO(webrtc:9262): make pure virtual
>+  // after updating dependencies.
>+  virtual void SetRuntimeSetting(AudioProcessing::RuntimeSetting setting);
> 
>   virtual ~CustomProcessing() {}
> };
> 
> // Interface for an echo detector submodule.
>-class EchoDetector {
>+class EchoDetector : public rtc::RefCountInterface {
>  public:
>   // (Re-)Initializes the submodule.
>-  virtual void Initialize(int sample_rate_hz, int num_channels) = 0;
>+  virtual void Initialize(int capture_sample_rate_hz,
>+                          int num_capture_channels,
>+                          int render_sample_rate_hz,
>+                          int num_render_channels) = 0;
> 
>   // Analysis (not changing) of the render signal.
>   virtual void AnalyzeRenderAudio(rtc::ArrayView<const float> render_audio) = 0;
>@@ -1176,8 +1058,6 @@ class EchoDetector {
> 
>   // Collect current metrics from the echo detector.
>   virtual Metrics GetMetrics() const = 0;
>-
>-  virtual ~EchoDetector() {}
> };
> 
> // The voice activity detection (VAD) component analyzes the stream to
>@@ -1230,107 +1110,6 @@ class VoiceDetection {
>   virtual ~VoiceDetection() {}
> };
> 
>-// Configuration struct for EchoCanceller3
>-struct EchoCanceller3Config {
>-  struct Delay {
>-    size_t default_delay = 5;
>-    size_t down_sampling_factor = 4;
>-    size_t num_filters = 5;
>-    size_t api_call_jitter_blocks = 26;
>-    size_t min_echo_path_delay_blocks = 0;
>-    size_t delay_headroom_blocks = 2;
>-    size_t hysteresis_limit_1_blocks = 1;
>-    size_t hysteresis_limit_2_blocks = 1;
>-  } delay;
>-
>-  struct Filter {
>-    struct MainConfiguration {
>-      size_t length_blocks;
>-      float leakage_converged;
>-      float leakage_diverged;
>-      float error_floor;
>-      float noise_gate;
>-    };
>-
>-    struct ShadowConfiguration {
>-      size_t length_blocks;
>-      float rate;
>-      float noise_gate;
>-    };
>-
>-    MainConfiguration main = {13, 0.005f, 0.1f, 0.001f, 20075344.f};
>-    ShadowConfiguration shadow = {13, 0.7f, 20075344.f};
>-
>-    MainConfiguration main_initial = {12, 0.05f, 5.f, 0.001f, 20075344.f};
>-    ShadowConfiguration shadow_initial = {12, 0.9f, 20075344.f};
>-  } filter;
>-
>-  struct Erle {
>-    float min = 1.f;
>-    float max_l = 8.f;
>-    float max_h = 1.5f;
>-  } erle;
>-
>-  struct EpStrength {
>-    float lf = 10.f;
>-    float mf = 10.f;
>-    float hf = 10.f;
>-    float default_len = 0.f;
>-    bool echo_can_saturate = true;
>-    bool bounded_erl = false;
>-  } ep_strength;
>-
>-  struct Mask {
>-    float m1 = 0.01f;
>-    float m2 = 0.0001f;
>-    float m3 = 0.01f;
>-    float m4 = 0.1f;
>-    float m5 = 0.1f;
>-    float m6 = 0.0001f;
>-    float m7 = 0.01f;
>-    float m8 = 0.0001f;
>-    float m9 = 0.1f;
>-  } gain_mask;
>-
>-  struct EchoAudibility {
>-    float low_render_limit = 4 * 64.f;
>-    float normal_render_limit = 64.f;
>-  } echo_audibility;
>-
>-  struct RenderLevels {
>-    float active_render_limit = 100.f;
>-    float poor_excitation_render_limit = 150.f;
>-  } render_levels;
>-
>-  struct GainUpdates {
>-    struct GainChanges {
>-      float max_inc;
>-      float max_dec;
>-      float rate_inc;
>-      float rate_dec;
>-      float min_inc;
>-      float min_dec;
>-    };
>-
>-    GainChanges low_noise = {2.f, 2.f, 1.4f, 1.4f, 1.1f, 1.1f};
>-    GainChanges initial = {2.f, 2.f, 1.5f, 1.5f, 1.2f, 1.2f};
>-    GainChanges normal = {2.f, 2.f, 1.5f, 1.5f, 1.2f, 1.2f};
>-    GainChanges saturation = {1.2f, 1.2f, 1.5f, 1.5f, 1.f, 1.f};
>-    GainChanges nonlinear = {1.5f, 1.5f, 1.2f, 1.2f, 1.1f, 1.1f};
>-
>-    float floor_first_increase = 0.00001f;
>-  } gain_updates;
>-};
>-
>-class EchoCanceller3Factory : public EchoControlFactory {
>- public:
>-  EchoCanceller3Factory();
>-  EchoCanceller3Factory(const EchoCanceller3Config& config);
>-  std::unique_ptr<EchoControl> Create(int sample_rate_hz) override;
>-
>- private:
>-  EchoCanceller3Config config_;
>-};
> }  // namespace webrtc
> 
> #endif  // MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/audio_processing_statistics.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/audio_processing_statistics.h
>index 05c59054b36..237d23c5bce 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/audio_processing_statistics.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/audio_processing_statistics.h
>@@ -11,7 +11,7 @@
> #ifndef MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_STATISTICS_H_
> #define MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_PROCESSING_STATISTICS_H_
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> 
> namespace webrtc {
> // This version of the stats uses Optionals, it will replace the regular
>@@ -23,12 +23,12 @@ struct AudioProcessingStats {
> 
>   // AEC Statistics.
>   // ERL = 10log_10(P_far / P_echo)
>-  rtc::Optional<double> echo_return_loss;
>+  absl::optional<double> echo_return_loss;
>   // ERLE = 10log_10(P_echo / P_out)
>-  rtc::Optional<double> echo_return_loss_enhancement;
>+  absl::optional<double> echo_return_loss_enhancement;
>   // Fraction of time that the AEC linear filter is divergent, in a 1-second
>   // non-overlapped aggregation window.
>-  rtc::Optional<double> divergent_filter_fraction;
>+  absl::optional<double> divergent_filter_fraction;
> 
>   // The delay metrics consists of the delay median and standard deviation. It
>   // also consists of the fraction of delay estimates that can make the echo
>@@ -37,18 +37,18 @@ struct AudioProcessingStats {
>   // second. Note that if there are several clients pulling metrics from
>   // |GetStatistics()| during a session the first call from any of them will
>   // change to one second aggregation window for all.
>-  rtc::Optional<int32_t> delay_median_ms;
>-  rtc::Optional<int32_t> delay_standard_deviation_ms;
>+  absl::optional<int32_t> delay_median_ms;
>+  absl::optional<int32_t> delay_standard_deviation_ms;
> 
>   // Residual echo detector likelihood.
>-  rtc::Optional<double> residual_echo_likelihood;
>+  absl::optional<double> residual_echo_likelihood;
>   // Maximum residual echo likelihood from the last time period.
>-  rtc::Optional<double> residual_echo_likelihood_recent_max;
>+  absl::optional<double> residual_echo_likelihood_recent_max;
> 
>   // The instantaneous delay estimate produced in the AEC. The unit is in
>   // milliseconds and the value is the instantaneous value at the time of the
>   // call to |GetStatistics()|.
>-  rtc::Optional<int32_t> delay_ms;
>+  absl::optional<int32_t> delay_ms;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/config.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/config.h
>index 7c34de8cccd..9232b2e0a83 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/config.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/config.h
>@@ -13,7 +13,6 @@
> 
> #include <map>
> 
>-#include "rtc_base/basictypes.h"
> #include "rtc_base/constructormagic.h"
> 
> namespace webrtc {
>@@ -31,11 +30,11 @@ enum class ConfigOptionID {
>   kDelayAgnostic,
>   kExperimentalAgc,
>   kExperimentalNs,
>-  kBeamforming,
>+  kBeamforming,  // Deprecated
>   kIntelligibility,
>   kEchoCanceller3,  // Deprecated
>   kAecRefinedAdaptiveFilter,
>-  kLevelControl
>+  kLevelControl  // Deprecated
> };
> 
> // Class Config is designed to ease passing a set of options across webrtc code.
>@@ -65,11 +64,13 @@ class Config {
>   // Returned references are owned by this.
>   //
>   // Requires std::is_default_constructible<T>
>-  template<typename T> const T& Get() const;
>+  template <typename T>
>+  const T& Get() const;
> 
>   // Set the option, deleting any previous instance of the same.
>   // This instance gets ownership of the newly set value.
>-  template<typename T> void Set(T* value);
>+  template <typename T>
>+  void Set(T* value);
> 
>   Config();
>   ~Config();
>@@ -79,16 +80,14 @@ class Config {
>     virtual ~BaseOption() {}
>   };
> 
>-  template<typename T>
>+  template <typename T>
>   struct Option : BaseOption {
>-    explicit Option(T* v): value(v) {}
>-    ~Option() {
>-      delete value;
>-    }
>+    explicit Option(T* v) : value(v) {}
>+    ~Option() { delete value; }
>     T* value;
>   };
> 
>-  template<typename T>
>+  template <typename T>
>   static ConfigOptionID identifier() {
>     return T::identifier;
>   }
>@@ -96,10 +95,10 @@ class Config {
>   // Used to instantiate a default constructed object that doesn't needs to be
>   // owned. This allows Get<T> to be implemented without requiring explicitly
>   // locks.
>-  template<typename T>
>+  template <typename T>
>   static const T& default_value() {
>-    RTC_DEFINE_STATIC_LOCAL(const T, def, ());
>-    return def;
>+    static const T* const def = new T();
>+    return *def;
>   }
> 
>   typedef std::map<ConfigOptionID, BaseOption*> OptionMap;
>@@ -110,7 +109,7 @@ class Config {
>   void operator=(const Config&);
> };
> 
>-template<typename T>
>+template <typename T>
> const T& Config::Get() const {
>   OptionMap::const_iterator it = options_.find(identifier<T>());
>   if (it != options_.end()) {
>@@ -122,7 +121,7 @@ const T& Config::Get() const {
>   return default_value<T>();
> }
> 
>-template<typename T>
>+template <typename T>
> void Config::Set(T* value) {
>   BaseOption*& it = options_[identifier<T>()];
>   delete it;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/gain_control.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/gain_control.h
>new file mode 100644
>index 00000000000..420b1c6e385
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/gain_control.h
>@@ -0,0 +1,108 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_INCLUDE_GAIN_CONTROL_H_
>+#define MODULES_AUDIO_PROCESSING_INCLUDE_GAIN_CONTROL_H_
>+
>+namespace webrtc {
>+
>+// The automatic gain control (AGC) component brings the signal to an
>+// appropriate range. This is done by applying a digital gain directly and, in
>+// the analog mode, prescribing an analog gain to be applied at the audio HAL.
>+//
>+// Recommended to be enabled on the client-side.
>+class GainControl {
>+ public:
>+  virtual int Enable(bool enable) = 0;
>+  virtual bool is_enabled() const = 0;
>+
>+  // When an analog mode is set, this must be called prior to |ProcessStream()|
>+  // to pass the current analog level from the audio HAL. Must be within the
>+  // range provided to |set_analog_level_limits()|.
>+  virtual int set_stream_analog_level(int level) = 0;
>+
>+  // When an analog mode is set, this should be called after |ProcessStream()|
>+  // to obtain the recommended new analog level for the audio HAL. It is the
>+  // users responsibility to apply this level.
>+  virtual int stream_analog_level() = 0;
>+
>+  enum Mode {
>+    // Adaptive mode intended for use if an analog volume control is available
>+    // on the capture device. It will require the user to provide coupling
>+    // between the OS mixer controls and AGC through the |stream_analog_level()|
>+    // functions.
>+    //
>+    // It consists of an analog gain prescription for the audio device and a
>+    // digital compression stage.
>+    kAdaptiveAnalog,
>+
>+    // Adaptive mode intended for situations in which an analog volume control
>+    // is unavailable. It operates in a similar fashion to the adaptive analog
>+    // mode, but with scaling instead applied in the digital domain. As with
>+    // the analog mode, it additionally uses a digital compression stage.
>+    kAdaptiveDigital,
>+
>+    // Fixed mode which enables only the digital compression stage also used by
>+    // the two adaptive modes.
>+    //
>+    // It is distinguished from the adaptive modes by considering only a
>+    // short time-window of the input signal. It applies a fixed gain through
>+    // most of the input level range, and compresses (gradually reduces gain
>+    // with increasing level) the input signal at higher levels. This mode is
>+    // preferred on embedded devices where the capture signal level is
>+    // predictable, so that a known gain can be applied.
>+    kFixedDigital
>+  };
>+
>+  virtual int set_mode(Mode mode) = 0;
>+  virtual Mode mode() const = 0;
>+
>+  // Sets the target peak |level| (or envelope) of the AGC in dBFs (decibels
>+  // from digital full-scale). The convention is to use positive values. For
>+  // instance, passing in a value of 3 corresponds to -3 dBFs, or a target
>+  // level 3 dB below full-scale. Limited to [0, 31].
>+  //
>+  // TODO(ajm): use a negative value here instead, if/when VoE will similarly
>+  //            update its interface.
>+  virtual int set_target_level_dbfs(int level) = 0;
>+  virtual int target_level_dbfs() const = 0;
>+
>+  // Sets the maximum |gain| the digital compression stage may apply, in dB. A
>+  // higher number corresponds to greater compression, while a value of 0 will
>+  // leave the signal uncompressed. Limited to [0, 90].
>+  virtual int set_compression_gain_db(int gain) = 0;
>+  virtual int compression_gain_db() const = 0;
>+
>+  // When enabled, the compression stage will hard limit the signal to the
>+  // target level. Otherwise, the signal will be compressed but not limited
>+  // above the target level.
>+  virtual int enable_limiter(bool enable) = 0;
>+  virtual bool is_limiter_enabled() const = 0;
>+
>+  // Sets the |minimum| and |maximum| analog levels of the audio capture device.
>+  // Must be set if and only if an analog mode is used. Limited to [0, 65535].
>+  virtual int set_analog_level_limits(int minimum, int maximum) = 0;
>+  virtual int analog_level_minimum() const = 0;
>+  virtual int analog_level_maximum() const = 0;
>+
>+  // Returns true if the AGC has detected a saturation event (period where the
>+  // signal reaches digital full-scale) in the current frame and the analog
>+  // level cannot be reduced.
>+  //
>+  // This could be used as an indicator to reduce or disable analog mic gain at
>+  // the audio HAL.
>+  virtual bool stream_is_saturated() const = 0;
>+
>+ protected:
>+  virtual ~GainControl() {}
>+};
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_INCLUDE_GAIN_CONTROL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/mock_audio_processing.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/mock_audio_processing.h
>index 8a1e463b237..9ceee106c08 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/mock_audio_processing.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/include/mock_audio_processing.h
>@@ -40,8 +40,8 @@ class MockEchoCancellation : public EchoCancellation {
>   MOCK_METHOD1(enable_delay_logging, int(bool enable));
>   MOCK_CONST_METHOD0(is_delay_logging_enabled, bool());
>   MOCK_METHOD2(GetDelayMetrics, int(int* median, int* std));
>-  MOCK_METHOD3(GetDelayMetrics, int(int* median, int* std,
>-                                    float* fraction_poor_delays));
>+  MOCK_METHOD3(GetDelayMetrics,
>+               int(int* median, int* std, float* fraction_poor_delays));
>   MOCK_CONST_METHOD0(aec_core, struct AecCore*());
> };
> 
>@@ -110,6 +110,8 @@ class MockCustomProcessing : public CustomProcessing {
>   virtual ~MockCustomProcessing() {}
>   MOCK_METHOD2(Initialize, void(int sample_rate_hz, int num_channels));
>   MOCK_METHOD1(Process, void(AudioBuffer* audio));
>+  MOCK_METHOD1(SetRuntimeSetting,
>+               void(AudioProcessing::RuntimeSetting setting));
>   MOCK_CONST_METHOD0(ToString, std::string());
> };
> 
>@@ -121,6 +123,7 @@ class MockEchoControl : public EchoControl {
>   MOCK_METHOD2(ProcessCapture,
>                void(AudioBuffer* capture, bool echo_path_change));
>   MOCK_CONST_METHOD0(GetMetrics, Metrics());
>+  MOCK_METHOD1(SetAudioBufferDelay, void(size_t delay_ms));
> };
> 
> class MockVoiceDetection : public VoiceDetection {
>@@ -136,7 +139,7 @@ class MockVoiceDetection : public VoiceDetection {
>   MOCK_CONST_METHOD0(frame_size_ms, int());
> };
> 
>-class MockAudioProcessing : public AudioProcessing {
>+class MockAudioProcessing : public testing::NiceMock<AudioProcessing> {
>  public:
>   MockAudioProcessing()
>       : echo_cancellation_(new testing::NiceMock<MockEchoCancellation>()),
>@@ -145,18 +148,18 @@ class MockAudioProcessing : public AudioProcessing {
>         high_pass_filter_(new testing::NiceMock<MockHighPassFilter>()),
>         level_estimator_(new testing::NiceMock<MockLevelEstimator>()),
>         noise_suppression_(new testing::NiceMock<MockNoiseSuppression>()),
>-        voice_detection_(new testing::NiceMock<MockVoiceDetection>()) {
>-  }
>+        voice_detection_(new testing::NiceMock<MockVoiceDetection>()) {}
> 
>   virtual ~MockAudioProcessing() {}
> 
>   MOCK_METHOD0(Initialize, int());
>-  MOCK_METHOD6(Initialize, int(int capture_input_sample_rate_hz,
>-                               int capture_output_sample_rate_hz,
>-                               int render_sample_rate_hz,
>-                               ChannelLayout capture_input_layout,
>-                               ChannelLayout capture_output_layout,
>-                               ChannelLayout render_input_layout));
>+  MOCK_METHOD6(Initialize,
>+               int(int capture_input_sample_rate_hz,
>+                   int capture_output_sample_rate_hz,
>+                   int render_sample_rate_hz,
>+                   ChannelLayout capture_input_layout,
>+                   ChannelLayout capture_output_layout,
>+                   ChannelLayout render_input_layout));
>   MOCK_METHOD1(Initialize, int(const ProcessingConfig& processing_config));
>   MOCK_METHOD1(ApplyConfig, void(const Config& config));
>   MOCK_METHOD1(SetExtraOptions, void(const webrtc::Config& config));
>@@ -167,27 +170,32 @@ class MockAudioProcessing : public AudioProcessing {
>   MOCK_CONST_METHOD0(num_output_channels, size_t());
>   MOCK_CONST_METHOD0(num_reverse_channels, size_t());
>   MOCK_METHOD1(set_output_will_be_muted, void(bool muted));
>+  MOCK_METHOD1(SetRuntimeSetting, void(RuntimeSetting setting));
>   MOCK_METHOD1(ProcessStream, int(AudioFrame* frame));
>-  MOCK_METHOD7(ProcessStream, int(const float* const* src,
>-                                  size_t samples_per_channel,
>-                                  int input_sample_rate_hz,
>-                                  ChannelLayout input_layout,
>-                                  int output_sample_rate_hz,
>-                                  ChannelLayout output_layout,
>-                                  float* const* dest));
>-  MOCK_METHOD4(ProcessStream, int(const float* const* src,
>-                                 const StreamConfig& input_config,
>-                                 const StreamConfig& output_config,
>-                                 float* const* dest));
>+  MOCK_METHOD7(ProcessStream,
>+               int(const float* const* src,
>+                   size_t samples_per_channel,
>+                   int input_sample_rate_hz,
>+                   ChannelLayout input_layout,
>+                   int output_sample_rate_hz,
>+                   ChannelLayout output_layout,
>+                   float* const* dest));
>+  MOCK_METHOD4(ProcessStream,
>+               int(const float* const* src,
>+                   const StreamConfig& input_config,
>+                   const StreamConfig& output_config,
>+                   float* const* dest));
>   MOCK_METHOD1(ProcessReverseStream, int(AudioFrame* frame));
>-  MOCK_METHOD4(AnalyzeReverseStream, int(const float* const* data,
>-                                         size_t samples_per_channel,
>-                                         int sample_rate_hz,
>-                                         ChannelLayout layout));
>-  MOCK_METHOD4(ProcessReverseStream, int(const float* const* src,
>-                                         const StreamConfig& input_config,
>-                                         const StreamConfig& output_config,
>-                                         float* const* dest));
>+  MOCK_METHOD4(AnalyzeReverseStream,
>+               int(const float* const* data,
>+                   size_t samples_per_channel,
>+                   int sample_rate_hz,
>+                   ChannelLayout layout));
>+  MOCK_METHOD4(ProcessReverseStream,
>+               int(const float* const* src,
>+                   const StreamConfig& input_config,
>+                   const StreamConfig& output_config,
>+                   float* const* dest));
>   MOCK_METHOD1(set_stream_delay_ms, int(int delay));
>   MOCK_CONST_METHOD0(stream_delay_ms, int());
>   MOCK_CONST_METHOD0(was_stream_delay_set, bool());
>@@ -198,6 +206,10 @@ class MockAudioProcessing : public AudioProcessing {
>   virtual void AttachAecDump(std::unique_ptr<AecDump> aec_dump) {}
>   MOCK_METHOD0(DetachAecDump, void());
> 
>+  virtual void AttachPlayoutAudioGenerator(
>+      std::unique_ptr<AudioGenerator> audio_generator) {}
>+  MOCK_METHOD0(DetachPlayoutAudioGenerator, void());
>+
>   MOCK_METHOD0(UpdateHistogramsOnCallEnd, void());
>   MOCK_CONST_METHOD0(GetStatistics, AudioProcessingStatistics());
>   MOCK_CONST_METHOD1(GetStatistics, AudioProcessingStats(bool));
>@@ -207,9 +219,7 @@ class MockAudioProcessing : public AudioProcessing {
>   virtual MockEchoControlMobile* echo_control_mobile() const {
>     return echo_control_mobile_.get();
>   }
>-  virtual MockGainControl* gain_control() const {
>-    return gain_control_.get();
>-  }
>+  virtual MockGainControl* gain_control() const { return gain_control_.get(); }
>   virtual MockHighPassFilter* high_pass_filter() const {
>     return high_pass_filter_.get();
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.cc
>index 0e696d9fff1..0f7b118ea15 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.cc
>@@ -31,16 +31,16 @@ const int kWindowSizeMs = 16;
> const int kChunkSizeMs = 10;  // Size provided by APM.
> const float kClipFreqKhz = 0.2f;
> const float kKbdAlpha = 1.5f;
>-const float kLambdaBot = -1.f;      // Extreme values in bisection
>-const float kLambdaTop = -1e-5f;      // search for lamda.
>+const float kLambdaBot = -1.f;    // Extreme values in bisection
>+const float kLambdaTop = -1e-5f;  // search for lamda.
> const float kVoiceProbabilityThreshold = 0.5f;
> // Number of chunks after voice activity which is still considered speech.
> const size_t kSpeechOffsetDelay = 10;
>-const float kDecayRate = 0.995f;              // Power estimation decay rate.
>+const float kDecayRate = 0.995f;  // Power estimation decay rate.
> const float kMaxRelativeGainChange = 0.005f;
> const float kRho = 0.0004f;  // Default production and interpretation SNR.
> const float kPowerNormalizationFactor = 1.f / (1 << 30);
>-const float kMaxActiveSNR = 128.f;  // 21dB
>+const float kMaxActiveSNR = 128.f;   // 21dB
> const float kMinInactiveSNR = 32.f;  // 15dB
> const size_t kGainUpdatePeriod = 10u;
> 
>@@ -133,8 +133,8 @@ IntelligibilityEnhancer::~IntelligibilityEnhancer() {
>   }
> }
> 
>-void IntelligibilityEnhancer::SetCaptureNoiseEstimate(
>-    std::vector<float> noise, float gain) {
>+void IntelligibilityEnhancer::SetCaptureNoiseEstimate(std::vector<float> noise,
>+                                                      float gain) {
>   RTC_DCHECK_EQ(noise.size(), num_noise_bins_);
>   for (auto& bin : noise) {
>     bin *= gain;
>@@ -176,10 +176,9 @@ void IntelligibilityEnhancer::ProcessAudioBlock(
>       MapToErbBands(noise_power_estimator_.power().data(), capture_filter_bank_,
>                     filtered_noise_pow_.data());
>       SolveForGainsGivenLambda(kLambdaTop, start_freq_, gains_eq_.data());
>-      const float power_target = std::accumulate(
>-          filtered_clear_pow_.data(),
>-          filtered_clear_pow_.data() + bank_size_,
>-          0.f);
>+      const float power_target =
>+          std::accumulate(filtered_clear_pow_.data(),
>+                          filtered_clear_pow_.data() + bank_size_, 0.f);
>       const float power_top =
>           DotProduct(gains_eq_.data(), filtered_clear_pow_.data(), bank_size_);
>       SolveForGainsGivenLambda(kLambdaBot, start_freq_, gains_eq_.data());
>@@ -199,12 +198,11 @@ void IntelligibilityEnhancer::ProcessAudioBlock(
> void IntelligibilityEnhancer::SnrBasedEffectActivation() {
>   const float* clear_psd = clear_power_estimator_.power().data();
>   const float* noise_psd = noise_power_estimator_.power().data();
>-  const float clear_power =
>-      std::accumulate(clear_psd, clear_psd + freqs_, 0.f);
>-  const float noise_power =
>-      std::accumulate(noise_psd, noise_psd + freqs_, 0.f);
>-  snr_ = kDecayRate * snr_ + (1.f - kDecayRate) * clear_power /
>-      (noise_power + std::numeric_limits<float>::epsilon());
>+  const float clear_power = std::accumulate(clear_psd, clear_psd + freqs_, 0.f);
>+  const float noise_power = std::accumulate(noise_psd, noise_psd + freqs_, 0.f);
>+  snr_ = kDecayRate * snr_ +
>+         (1.f - kDecayRate) * clear_power /
>+             (noise_power + std::numeric_limits<float>::epsilon());
>   if (is_active_) {
>     if (snr_ > kMaxActiveSNR) {
>       RTC_LOG(LS_INFO) << "Intelligibility Enhancer was deactivated at chunk "
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h
>index 3e0e269c583..35130921bb2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/intelligibility/intelligibility_enhancer.h
>@@ -93,7 +93,7 @@ class IntelligibilityEnhancer : public LappedTransform::Callback {
> 
>   static const size_t kMaxNumNoiseEstimatesToBuffer = 5;
> 
>-  const size_t freqs_;         // Num frequencies in frequency domain.
>+  const size_t freqs_;  // Num frequencies in frequency domain.
>   const size_t num_noise_bins_;
>   const size_t chunk_length_;  // Chunk size in samples.
>   const size_t bank_size_;     // Num ERB filters.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/intelligibility/intelligibility_utils.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/intelligibility/intelligibility_utils.cc
>index b6917f44077..b606d951327 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/intelligibility/intelligibility_utils.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/intelligibility/intelligibility_utils.cc
>@@ -37,11 +37,11 @@ float UpdateFactor(float target, float current, float limit) {
> 
> }  // namespace
> 
>-template<typename T>
>+template <typename T>
> PowerEstimator<T>::PowerEstimator(size_t num_freqs, float decay)
>     : power_(num_freqs, 0.f), decay_(decay) {}
> 
>-template<typename T>
>+template <typename T>
> void PowerEstimator<T>::Step(const T* data) {
>   for (size_t i = 0; i < power_.size(); ++i) {
>     power_[i] = decay_ * power_[i] +
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/gain_applier.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/gain_applier.cc
>deleted file mode 100644
>index 018f809e013..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/gain_applier.cc
>+++ /dev/null
>@@ -1,160 +0,0 @@
>-/*
>- *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "modules/audio_processing/level_controller/gain_applier.h"
>-
>-#include <algorithm>
>-
>-#include "api/array_view.h"
>-#include "rtc_base/checks.h"
>-
>-#include "modules/audio_processing/audio_buffer.h"
>-#include "modules/audio_processing/logging/apm_data_dumper.h"
>-
>-namespace webrtc {
>-namespace {
>-
>-const float kMaxSampleValue = 32767.f;
>-const float kMinSampleValue = -32767.f;
>-
>-int CountSaturations(rtc::ArrayView<const float> in) {
>-  return std::count_if(in.begin(), in.end(), [](const float& v) {
>-    return v >= kMaxSampleValue || v <= kMinSampleValue;
>-  });
>-}
>-
>-int CountSaturations(const AudioBuffer& audio) {
>-  int num_saturations = 0;
>-  for (size_t k = 0; k < audio.num_channels(); ++k) {
>-    num_saturations += CountSaturations(rtc::ArrayView<const float>(
>-        audio.channels_const_f()[k], audio.num_frames()));
>-  }
>-  return num_saturations;
>-}
>-
>-void LimitToAllowedRange(rtc::ArrayView<float> x) {
>-  for (auto& v : x) {
>-    v = std::max(kMinSampleValue, v);
>-    v = std::min(kMaxSampleValue, v);
>-  }
>-}
>-
>-void LimitToAllowedRange(AudioBuffer* audio) {
>-  for (size_t k = 0; k < audio->num_channels(); ++k) {
>-    LimitToAllowedRange(
>-        rtc::ArrayView<float>(audio->channels_f()[k], audio->num_frames()));
>-  }
>-}
>-
>-float ApplyIncreasingGain(float new_gain,
>-                          float old_gain,
>-                          float step_size,
>-                          rtc::ArrayView<float> x) {
>-  RTC_DCHECK_LT(0.f, step_size);
>-  float gain = old_gain;
>-  for (auto& v : x) {
>-    gain = std::min(new_gain, gain + step_size);
>-    v *= gain;
>-  }
>-  return gain;
>-}
>-
>-float ApplyDecreasingGain(float new_gain,
>-                          float old_gain,
>-                          float step_size,
>-                          rtc::ArrayView<float> x) {
>-  RTC_DCHECK_GT(0.f, step_size);
>-  float gain = old_gain;
>-  for (auto& v : x) {
>-    gain = std::max(new_gain, gain + step_size);
>-    v *= gain;
>-  }
>-  return gain;
>-}
>-
>-float ApplyConstantGain(float gain, rtc::ArrayView<float> x) {
>-  for (auto& v : x) {
>-    v *= gain;
>-  }
>-
>-  return gain;
>-}
>-
>-float ApplyGain(float new_gain,
>-                float old_gain,
>-                float increase_step_size,
>-                float decrease_step_size,
>-                rtc::ArrayView<float> x) {
>-  RTC_DCHECK_LT(0.f, increase_step_size);
>-  RTC_DCHECK_GT(0.f, decrease_step_size);
>-  if (new_gain == old_gain) {
>-    return ApplyConstantGain(new_gain, x);
>-  } else if (new_gain > old_gain) {
>-    return ApplyIncreasingGain(new_gain, old_gain, increase_step_size, x);
>-  } else {
>-    return ApplyDecreasingGain(new_gain, old_gain, decrease_step_size, x);
>-  }
>-}
>-
>-}  // namespace
>-
>-GainApplier::GainApplier(ApmDataDumper* data_dumper)
>-    : data_dumper_(data_dumper) {}
>-
>-void GainApplier::Initialize(int sample_rate_hz) {
>-  RTC_DCHECK(sample_rate_hz == AudioProcessing::kSampleRate8kHz ||
>-             sample_rate_hz == AudioProcessing::kSampleRate16kHz ||
>-             sample_rate_hz == AudioProcessing::kSampleRate32kHz ||
>-             sample_rate_hz == AudioProcessing::kSampleRate48kHz);
>-  const float kGainIncreaseStepSize48kHz = 0.0001f;
>-  const float kGainDecreaseStepSize48kHz = -0.01f;
>-  const float kGainSaturatedDecreaseStepSize48kHz = -0.05f;
>-
>-  last_frame_was_saturated_ = false;
>-  old_gain_ = 1.f;
>-  gain_increase_step_size_ =
>-      kGainIncreaseStepSize48kHz *
>-      (static_cast<float>(AudioProcessing::kSampleRate48kHz) / sample_rate_hz);
>-  gain_normal_decrease_step_size_ =
>-      kGainDecreaseStepSize48kHz *
>-      (static_cast<float>(AudioProcessing::kSampleRate48kHz) / sample_rate_hz);
>-  gain_saturated_decrease_step_size_ =
>-      kGainSaturatedDecreaseStepSize48kHz *
>-      (static_cast<float>(AudioProcessing::kSampleRate48kHz) / sample_rate_hz);
>-}
>-
>-int GainApplier::Process(float new_gain, AudioBuffer* audio) {
>-  RTC_CHECK_NE(0.f, gain_increase_step_size_);
>-  RTC_CHECK_NE(0.f, gain_normal_decrease_step_size_);
>-  RTC_CHECK_NE(0.f, gain_saturated_decrease_step_size_);
>-  int num_saturations = 0;
>-  if (new_gain != 1.f) {
>-    float last_applied_gain = 1.f;
>-    float gain_decrease_step_size = last_frame_was_saturated_
>-                                        ? gain_saturated_decrease_step_size_
>-                                        : gain_normal_decrease_step_size_;
>-    for (size_t k = 0; k < audio->num_channels(); ++k) {
>-      last_applied_gain = ApplyGain(
>-          new_gain, old_gain_, gain_increase_step_size_,
>-          gain_decrease_step_size,
>-          rtc::ArrayView<float>(audio->channels_f()[k], audio->num_frames()));
>-    }
>-
>-    num_saturations = CountSaturations(*audio);
>-    LimitToAllowedRange(audio);
>-    old_gain_ = last_applied_gain;
>-  }
>-
>-  data_dumper_->DumpRaw("lc_last_applied_gain", 1, &old_gain_);
>-
>-  return num_saturations;
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/gain_applier.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/gain_applier.h
>deleted file mode 100644
>index 5669f45bf7a..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/gain_applier.h
>+++ /dev/null
>@@ -1,42 +0,0 @@
>-/*
>- *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_GAIN_APPLIER_H_
>-#define MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_GAIN_APPLIER_H_
>-
>-#include "rtc_base/constructormagic.h"
>-
>-namespace webrtc {
>-
>-class ApmDataDumper;
>-class AudioBuffer;
>-
>-class GainApplier {
>- public:
>-  explicit GainApplier(ApmDataDumper* data_dumper);
>-  void Initialize(int sample_rate_hz);
>-
>-  // Applies the specified gain to the audio frame and returns the resulting
>-  // number of saturated sample values.
>-  int Process(float new_gain, AudioBuffer* audio);
>-
>- private:
>-  ApmDataDumper* const data_dumper_;
>-  float old_gain_ = 1.f;
>-  float gain_increase_step_size_ = 0.f;
>-  float gain_normal_decrease_step_size_ = 0.f;
>-  float gain_saturated_decrease_step_size_ = 0.f;
>-  bool last_frame_was_saturated_;
>-  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(GainApplier);
>-};
>-
>-}  // namespace webrtc
>-
>-#endif  // MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_GAIN_APPLIER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/gain_selector.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/gain_selector.cc
>deleted file mode 100644
>index 3ab75b1ce67..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/gain_selector.cc
>+++ /dev/null
>@@ -1,87 +0,0 @@
>-/*
>- *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "modules/audio_processing/level_controller/gain_selector.h"
>-
>-#include <math.h>
>-#include <algorithm>
>-
>-#include "modules/audio_processing/include/audio_processing.h"
>-#include "modules/audio_processing/level_controller/level_controller_constants.h"
>-#include "rtc_base/checks.h"
>-
>-namespace webrtc {
>-
>-GainSelector::GainSelector() {
>-  Initialize(AudioProcessing::kSampleRate48kHz);
>-}
>-
>-void GainSelector::Initialize(int sample_rate_hz) {
>-  gain_ = 1.f;
>-  frame_length_ = rtc::CheckedDivExact(sample_rate_hz, 100);
>-  highly_nonstationary_signal_hold_counter_ = 0;
>-}
>-
>-// Chooses the gain to apply by the level controller such that
>-// 1) The level of the stationary noise does not exceed
>-//    a predefined threshold.
>-// 2) The gain does not exceed the gain that has been found
>-//    to saturate the signal.
>-// 3) The peak level achieves the target peak level.
>-// 4) The gain is not below 1.
>-// 4) The gain is 1 if the signal has been classified as stationary
>-//    for a long time.
>-// 5) The gain is not above the maximum gain.
>-float GainSelector::GetNewGain(float peak_level,
>-                               float noise_energy,
>-                               float saturating_gain,
>-                               bool gain_jumpstart,
>-                               SignalClassifier::SignalType signal_type) {
>-  RTC_DCHECK_LT(0.f, peak_level);
>-
>-  if (signal_type == SignalClassifier::SignalType::kHighlyNonStationary ||
>-      gain_jumpstart) {
>-    highly_nonstationary_signal_hold_counter_ = 100;
>-  } else {
>-    highly_nonstationary_signal_hold_counter_ =
>-        std::max(0, highly_nonstationary_signal_hold_counter_ - 1);
>-  }
>-
>-  float desired_gain;
>-  if (highly_nonstationary_signal_hold_counter_ > 0) {
>-    // Compute a desired gain that ensures that the peak level is amplified to
>-    // the target level.
>-    desired_gain = kTargetLcPeakLevel / peak_level;
>-
>-    // Limit the desired gain so that it does not amplify the noise too much.
>-    float max_noise_energy = kMaxLcNoisePower * frame_length_;
>-    if (noise_energy * desired_gain * desired_gain > max_noise_energy) {
>-      RTC_DCHECK_LE(0.f, noise_energy);
>-      desired_gain = sqrtf(max_noise_energy / noise_energy);
>-    }
>-  } else {
>-    // If the signal has been stationary for a long while, apply a gain of 1 to
>-    // avoid amplifying pure noise.
>-    desired_gain = 1.0f;
>-  }
>-
>-  // Smootly update the gain towards the desired gain.
>-  gain_ += 0.2f * (desired_gain - gain_);
>-
>-  // Limit the gain to not exceed the maximum and the saturating gains, and to
>-  // ensure that the lowest possible gain is 1.
>-  gain_ = std::min(gain_, saturating_gain);
>-  gain_ = std::min(gain_, kMaxLcGain);
>-  gain_ = std::max(gain_, 1.f);
>-
>-  return gain_;
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/gain_selector.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/gain_selector.h
>deleted file mode 100644
>index 7966c438d7d..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/gain_selector.h
>+++ /dev/null
>@@ -1,40 +0,0 @@
>-/*
>- *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_GAIN_SELECTOR_H_
>-#define MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_GAIN_SELECTOR_H_
>-
>-#include "rtc_base/constructormagic.h"
>-
>-#include "modules/audio_processing/level_controller/signal_classifier.h"
>-
>-namespace webrtc {
>-
>-class GainSelector {
>- public:
>-  GainSelector();
>-  void Initialize(int sample_rate_hz);
>-  float GetNewGain(float peak_level,
>-                   float noise_energy,
>-                   float saturating_gain,
>-                   bool gain_jumpstart,
>-                   SignalClassifier::SignalType signal_type);
>-
>- private:
>-  float gain_;
>-  size_t frame_length_;
>-  int highly_nonstationary_signal_hold_counter_;
>-
>-  RTC_DISALLOW_COPY_AND_ASSIGN(GainSelector);
>-};
>-
>-}  // namespace webrtc
>-
>-#endif  // MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_GAIN_SELECTOR_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/level_controller.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/level_controller.cc
>deleted file mode 100644
>index 521f82e842b..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/level_controller.cc
>+++ /dev/null
>@@ -1,293 +0,0 @@
>-/*
>- *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "modules/audio_processing/level_controller/level_controller.h"
>-
>-#include <math.h>
>-#include <algorithm>
>-#include <numeric>
>-
>-#include "api/array_view.h"
>-#include "modules/audio_processing/audio_buffer.h"
>-#include "modules/audio_processing/level_controller/gain_applier.h"
>-#include "modules/audio_processing/level_controller/gain_selector.h"
>-#include "modules/audio_processing/level_controller/noise_level_estimator.h"
>-#include "modules/audio_processing/level_controller/peak_level_estimator.h"
>-#include "modules/audio_processing/level_controller/saturating_gain_estimator.h"
>-#include "modules/audio_processing/level_controller/signal_classifier.h"
>-#include "modules/audio_processing/logging/apm_data_dumper.h"
>-#include "rtc_base/arraysize.h"
>-#include "rtc_base/checks.h"
>-#include "rtc_base/logging.h"
>-#include "system_wrappers/include/metrics.h"
>-
>-namespace webrtc {
>-namespace {
>-
>-void UpdateAndRemoveDcLevel(float forgetting_factor,
>-                            float* dc_level,
>-                            rtc::ArrayView<float> x) {
>-  RTC_DCHECK(!x.empty());
>-  float mean =
>-      std::accumulate(x.begin(), x.end(), 0.0f) / static_cast<float>(x.size());
>-  *dc_level += forgetting_factor * (mean - *dc_level);
>-
>-  for (float& v : x) {
>-    v -= *dc_level;
>-  }
>-}
>-
>-float FrameEnergy(const AudioBuffer& audio) {
>-  float energy = 0.f;
>-  for (size_t k = 0; k < audio.num_channels(); ++k) {
>-    float channel_energy =
>-        std::accumulate(audio.channels_const_f()[k],
>-                        audio.channels_const_f()[k] + audio.num_frames(), 0.f,
>-                        [](float a, float b) -> float { return a + b * b; });
>-    energy = std::max(channel_energy, energy);
>-  }
>-  return energy;
>-}
>-
>-float PeakLevel(const AudioBuffer& audio) {
>-  float peak_level = 0.f;
>-  for (size_t k = 0; k < audio.num_channels(); ++k) {
>-    auto* channel_peak_level = std::max_element(
>-        audio.channels_const_f()[k],
>-        audio.channels_const_f()[k] + audio.num_frames(),
>-        [](float a, float b) { return std::abs(a) < std::abs(b); });
>-    peak_level = std::max(*channel_peak_level, peak_level);
>-  }
>-  return peak_level;
>-}
>-
>-const int kMetricsFrameInterval = 1000;
>-
>-}  // namespace
>-
>-int LevelController::instance_count_ = 0;
>-
>-void LevelController::Metrics::Initialize(int sample_rate_hz) {
>-  RTC_DCHECK(sample_rate_hz == AudioProcessing::kSampleRate8kHz ||
>-             sample_rate_hz == AudioProcessing::kSampleRate16kHz ||
>-             sample_rate_hz == AudioProcessing::kSampleRate32kHz ||
>-             sample_rate_hz == AudioProcessing::kSampleRate48kHz);
>-
>-  Reset();
>-  frame_length_ = rtc::CheckedDivExact(sample_rate_hz, 100);
>-}
>-
>-void LevelController::Metrics::Reset() {
>-  metrics_frame_counter_ = 0;
>-  gain_sum_ = 0.f;
>-  peak_level_sum_ = 0.f;
>-  noise_energy_sum_ = 0.f;
>-  max_gain_ = 0.f;
>-  max_peak_level_ = 0.f;
>-  max_noise_energy_ = 0.f;
>-}
>-
>-void LevelController::Metrics::Update(float long_term_peak_level,
>-                                      float noise_energy,
>-                                      float gain,
>-                                      float frame_peak_level) {
>-  const float kdBFSOffset = 90.3090f;
>-  gain_sum_ += gain;
>-  peak_level_sum_ += long_term_peak_level;
>-  noise_energy_sum_ += noise_energy;
>-  max_gain_ = std::max(max_gain_, gain);
>-  max_peak_level_ = std::max(max_peak_level_, long_term_peak_level);
>-  max_noise_energy_ = std::max(max_noise_energy_, noise_energy);
>-
>-  ++metrics_frame_counter_;
>-  if (metrics_frame_counter_ == kMetricsFrameInterval) {
>-    RTC_DCHECK_LT(0, frame_length_);
>-    RTC_DCHECK_LT(0, kMetricsFrameInterval);
>-
>-    const int max_noise_power_dbfs = static_cast<int>(
>-        10 * log10(max_noise_energy_ / frame_length_ + 1e-10f) - kdBFSOffset);
>-    RTC_HISTOGRAM_COUNTS("WebRTC.Audio.LevelControl.MaxNoisePower",
>-                         max_noise_power_dbfs, -90, 0, 50);
>-
>-    const int average_noise_power_dbfs = static_cast<int>(
>-        10 * log10(noise_energy_sum_ / (frame_length_ * kMetricsFrameInterval) +
>-                   1e-10f) -
>-        kdBFSOffset);
>-    RTC_HISTOGRAM_COUNTS("WebRTC.Audio.LevelControl.AverageNoisePower",
>-                         average_noise_power_dbfs, -90, 0, 50);
>-
>-    const int max_peak_level_dbfs = static_cast<int>(
>-        10 * log10(max_peak_level_ * max_peak_level_ + 1e-10f) - kdBFSOffset);
>-    RTC_HISTOGRAM_COUNTS("WebRTC.Audio.LevelControl.MaxPeakLevel",
>-                         max_peak_level_dbfs, -90, 0, 50);
>-
>-    const int average_peak_level_dbfs = static_cast<int>(
>-        10 * log10(peak_level_sum_ * peak_level_sum_ /
>-                       (kMetricsFrameInterval * kMetricsFrameInterval) +
>-                   1e-10f) -
>-        kdBFSOffset);
>-    RTC_HISTOGRAM_COUNTS("WebRTC.Audio.LevelControl.AveragePeakLevel",
>-                         average_peak_level_dbfs, -90, 0, 50);
>-
>-    RTC_DCHECK_LE(1.f, max_gain_);
>-    RTC_DCHECK_LE(1.f, gain_sum_ / kMetricsFrameInterval);
>-
>-    const int max_gain_db = static_cast<int>(10 * log10(max_gain_ * max_gain_));
>-    RTC_HISTOGRAM_COUNTS("WebRTC.Audio.LevelControl.MaxGain", max_gain_db, 0,
>-                         33, 30);
>-
>-    const int average_gain_db = static_cast<int>(
>-        10 * log10(gain_sum_ * gain_sum_ /
>-                   (kMetricsFrameInterval * kMetricsFrameInterval)));
>-    RTC_HISTOGRAM_COUNTS("WebRTC.Audio.LevelControl.AverageGain",
>-                         average_gain_db, 0, 33, 30);
>-
>-    const int long_term_peak_level_dbfs = static_cast<int>(
>-        10 * log10(long_term_peak_level * long_term_peak_level + 1e-10f) -
>-        kdBFSOffset);
>-
>-    const int frame_peak_level_dbfs = static_cast<int>(
>-        10 * log10(frame_peak_level * frame_peak_level + 1e-10f) - kdBFSOffset);
>-
>-    RTC_LOG(LS_INFO) << "Level Controller metrics: {"
>-                     << "Max noise power: " << max_noise_power_dbfs << " dBFS, "
>-                     << "Average noise power: " << average_noise_power_dbfs
>-                     << " dBFS, "
>-                     << "Max long term peak level: " << max_peak_level_dbfs
>-                     << " dBFS, "
>-                     << "Average long term peak level: "
>-                     << average_peak_level_dbfs << " dBFS, "
>-                     << "Max gain: " << max_gain_db << " dB, "
>-                     << "Average gain: " << average_gain_db << " dB, "
>-                     << "Long term peak level: " << long_term_peak_level_dbfs
>-                     << " dBFS, "
>-                     << "Last frame peak level: " << frame_peak_level_dbfs
>-                     << " dBFS"
>-                     << "}";
>-
>-    Reset();
>-  }
>-}
>-
>-LevelController::LevelController()
>-    : data_dumper_(new ApmDataDumper(instance_count_)),
>-      gain_applier_(data_dumper_.get()),
>-      signal_classifier_(data_dumper_.get()),
>-      peak_level_estimator_(kTargetLcPeakLeveldBFS) {
>-  Initialize(AudioProcessing::kSampleRate48kHz);
>-  ++instance_count_;
>-}
>-
>-LevelController::~LevelController() {}
>-
>-void LevelController::Initialize(int sample_rate_hz) {
>-  RTC_DCHECK(sample_rate_hz == AudioProcessing::kSampleRate8kHz ||
>-             sample_rate_hz == AudioProcessing::kSampleRate16kHz ||
>-             sample_rate_hz == AudioProcessing::kSampleRate32kHz ||
>-             sample_rate_hz == AudioProcessing::kSampleRate48kHz);
>-  data_dumper_->InitiateNewSetOfRecordings();
>-  gain_selector_.Initialize(sample_rate_hz);
>-  gain_applier_.Initialize(sample_rate_hz);
>-  signal_classifier_.Initialize(sample_rate_hz);
>-  noise_level_estimator_.Initialize(sample_rate_hz);
>-  peak_level_estimator_.Initialize(config_.initial_peak_level_dbfs);
>-  saturating_gain_estimator_.Initialize();
>-  metrics_.Initialize(sample_rate_hz);
>-
>-  last_gain_ = 1.0f;
>-  sample_rate_hz_ = sample_rate_hz;
>-  dc_forgetting_factor_ = 0.01f * sample_rate_hz / 48000.f;
>-  std::fill(dc_level_, dc_level_ + arraysize(dc_level_), 0.f);
>-}
>-
>-void LevelController::Process(AudioBuffer* audio) {
>-  RTC_DCHECK_LT(0, audio->num_channels());
>-  RTC_DCHECK_GE(2, audio->num_channels());
>-  RTC_DCHECK_NE(0.f, dc_forgetting_factor_);
>-  RTC_DCHECK(sample_rate_hz_);
>-  data_dumper_->DumpWav("lc_input", audio->num_frames(),
>-                        audio->channels_const_f()[0], *sample_rate_hz_, 1);
>-
>-  // Remove DC level.
>-  for (size_t k = 0; k < audio->num_channels(); ++k) {
>-    UpdateAndRemoveDcLevel(
>-        dc_forgetting_factor_, &dc_level_[k],
>-        rtc::ArrayView<float>(audio->channels_f()[k], audio->num_frames()));
>-  }
>-
>-  SignalClassifier::SignalType signal_type;
>-  signal_classifier_.Analyze(*audio, &signal_type);
>-  int tmp = static_cast<int>(signal_type);
>-  data_dumper_->DumpRaw("lc_signal_type", 1, &tmp);
>-
>-  // Estimate the noise energy.
>-  float noise_energy =
>-      noise_level_estimator_.Analyze(signal_type, FrameEnergy(*audio));
>-
>-  // Estimate the overall signal peak level.
>-  const float frame_peak_level = PeakLevel(*audio);
>-  const float long_term_peak_level =
>-      peak_level_estimator_.Analyze(signal_type, frame_peak_level);
>-
>-  float saturating_gain = saturating_gain_estimator_.GetGain();
>-
>-  // Compute the new gain to apply.
>-  last_gain_ =
>-      gain_selector_.GetNewGain(long_term_peak_level, noise_energy,
>-                                saturating_gain, gain_jumpstart_, signal_type);
>-
>-  // Unflag the jumpstart of the gain as it should only happen once.
>-  gain_jumpstart_ = false;
>-
>-  // Apply the gain to the signal.
>-  int num_saturations = gain_applier_.Process(last_gain_, audio);
>-
>-  // Estimate the gain that saturates the overall signal.
>-  saturating_gain_estimator_.Update(last_gain_, num_saturations);
>-
>-  // Update the metrics.
>-  metrics_.Update(long_term_peak_level, noise_energy, last_gain_,
>-                  frame_peak_level);
>-
>-  data_dumper_->DumpRaw("lc_selected_gain", 1, &last_gain_);
>-  data_dumper_->DumpRaw("lc_noise_energy", 1, &noise_energy);
>-  data_dumper_->DumpRaw("lc_peak_level", 1, &long_term_peak_level);
>-  data_dumper_->DumpRaw("lc_saturating_gain", 1, &saturating_gain);
>-
>-  data_dumper_->DumpWav("lc_output", audio->num_frames(),
>-                        audio->channels_f()[0], *sample_rate_hz_, 1);
>-}
>-
>-void LevelController::ApplyConfig(
>-    const AudioProcessing::Config::LevelController& config) {
>-  RTC_DCHECK(Validate(config));
>-  config_ = config;
>-  peak_level_estimator_.Initialize(config_.initial_peak_level_dbfs);
>-  gain_jumpstart_ = true;
>-}
>-
>-std::string LevelController::ToString(
>-    const AudioProcessing::Config::LevelController& config) {
>-  std::stringstream ss;
>-  ss << "{"
>-     << "enabled: " << (config.enabled ? "true" : "false") << ", "
>-     << "initial_peak_level_dbfs: " << config.initial_peak_level_dbfs << "}";
>-  return ss.str();
>-}
>-
>-bool LevelController::Validate(
>-    const AudioProcessing::Config::LevelController& config) {
>-  return (config.initial_peak_level_dbfs <
>-              std::numeric_limits<float>::epsilon() &&
>-          config.initial_peak_level_dbfs >
>-              -(100.f + std::numeric_limits<float>::epsilon()));
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/level_controller.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/level_controller.h
>deleted file mode 100644
>index 224b886abd1..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/level_controller.h
>+++ /dev/null
>@@ -1,95 +0,0 @@
>-/*
>- *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_LEVEL_CONTROLLER_H_
>-#define MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_LEVEL_CONTROLLER_H_
>-
>-#include <memory>
>-#include <vector>
>-
>-#include "api/optional.h"
>-#include "modules/audio_processing/include/audio_processing.h"
>-#include "modules/audio_processing/level_controller/gain_applier.h"
>-#include "modules/audio_processing/level_controller/gain_selector.h"
>-#include "modules/audio_processing/level_controller/noise_level_estimator.h"
>-#include "modules/audio_processing/level_controller/peak_level_estimator.h"
>-#include "modules/audio_processing/level_controller/saturating_gain_estimator.h"
>-#include "modules/audio_processing/level_controller/signal_classifier.h"
>-#include "rtc_base/constructormagic.h"
>-
>-namespace webrtc {
>-
>-class ApmDataDumper;
>-class AudioBuffer;
>-
>-class LevelController {
>- public:
>-  LevelController();
>-  ~LevelController();
>-
>-  void Initialize(int sample_rate_hz);
>-  void Process(AudioBuffer* audio);
>-  float GetLastGain() { return last_gain_; }
>-
>-  // TODO(peah): This method is a temporary solution as the the aim is to
>-  // instead apply the config inside the constructor. Therefore this is likely
>-  // to change.
>-  void ApplyConfig(const AudioProcessing::Config::LevelController& config);
>-  // Validates a config.
>-  static bool Validate(const AudioProcessing::Config::LevelController& config);
>-  // Dumps a config to a string.
>-  static std::string ToString(
>-      const AudioProcessing::Config::LevelController& config);
>-
>- private:
>-  class Metrics {
>-   public:
>-    Metrics() { Initialize(AudioProcessing::kSampleRate48kHz); }
>-    void Initialize(int sample_rate_hz);
>-    void Update(float long_term_peak_level,
>-                float noise_level,
>-                float gain,
>-                float frame_peak_level);
>-
>-   private:
>-    void Reset();
>-
>-    size_t metrics_frame_counter_;
>-    float gain_sum_;
>-    float peak_level_sum_;
>-    float noise_energy_sum_;
>-    float max_gain_;
>-    float max_peak_level_;
>-    float max_noise_energy_;
>-    float frame_length_;
>-  };
>-
>-  std::unique_ptr<ApmDataDumper> data_dumper_;
>-  GainSelector gain_selector_;
>-  GainApplier gain_applier_;
>-  SignalClassifier signal_classifier_;
>-  NoiseLevelEstimator noise_level_estimator_;
>-  PeakLevelEstimator peak_level_estimator_;
>-  SaturatingGainEstimator saturating_gain_estimator_;
>-  Metrics metrics_;
>-  rtc::Optional<int> sample_rate_hz_;
>-  static int instance_count_;
>-  float dc_level_[2];
>-  float dc_forgetting_factor_;
>-  float last_gain_;
>-  bool gain_jumpstart_ = false;
>-  AudioProcessing::Config::LevelController config_;
>-
>-  RTC_DISALLOW_COPY_AND_ASSIGN(LevelController);
>-};
>-
>-}  // namespace webrtc
>-
>-#endif  // MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_LEVEL_CONTROLLER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/level_controller_complexity_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/level_controller_complexity_unittest.cc
>deleted file mode 100644
>index 83f6725a0f1..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/level_controller_complexity_unittest.cc
>+++ /dev/null
>@@ -1,240 +0,0 @@
>-/*
>- *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include <numeric>
>-#include <vector>
>-
>-#include "api/array_view.h"
>-#include "modules/audio_processing/audio_buffer.h"
>-#include "modules/audio_processing/include/audio_processing.h"
>-#include "modules/audio_processing/level_controller/level_controller.h"
>-#include "modules/audio_processing/test/audio_buffer_tools.h"
>-#include "modules/audio_processing/test/bitexactness_tools.h"
>-#include "modules/audio_processing/test/performance_timer.h"
>-#include "modules/audio_processing/test/simulator_buffers.h"
>-#include "rtc_base/random.h"
>-#include "system_wrappers/include/clock.h"
>-#include "test/gtest.h"
>-#include "test/testsupport/perf_test.h"
>-
>-namespace webrtc {
>-namespace {
>-
>-const size_t kNumFramesToProcess = 300;
>-const size_t kNumFramesToProcessAtWarmup = 300;
>-const size_t kToTalNumFrames =
>-    kNumFramesToProcess + kNumFramesToProcessAtWarmup;
>-
>-void RunStandaloneSubmodule(int sample_rate_hz, size_t num_channels) {
>-  test::SimulatorBuffers buffers(sample_rate_hz, sample_rate_hz, sample_rate_hz,
>-                                 sample_rate_hz, num_channels, num_channels,
>-                                 num_channels, num_channels);
>-  test::PerformanceTimer timer(kNumFramesToProcess);
>-
>-  LevelController level_controller;
>-  level_controller.Initialize(sample_rate_hz);
>-
>-  for (size_t frame_no = 0; frame_no < kToTalNumFrames; ++frame_no) {
>-    buffers.UpdateInputBuffers();
>-
>-    if (frame_no >= kNumFramesToProcessAtWarmup) {
>-      timer.StartTimer();
>-    }
>-    level_controller.Process(buffers.capture_input_buffer.get());
>-    if (frame_no >= kNumFramesToProcessAtWarmup) {
>-      timer.StopTimer();
>-    }
>-  }
>-  webrtc::test::PrintResultMeanAndError(
>-      "level_controller_call_durations",
>-      "_" + std::to_string(sample_rate_hz) + "Hz_" +
>-          std::to_string(num_channels) + "_channels",
>-      "StandaloneLevelControl", timer.GetDurationAverage(),
>-      timer.GetDurationStandardDeviation(), "us", false);
>-}
>-
>-void RunTogetherWithApm(const std::string& test_description,
>-                        int render_input_sample_rate_hz,
>-                        int render_output_sample_rate_hz,
>-                        int capture_input_sample_rate_hz,
>-                        int capture_output_sample_rate_hz,
>-                        size_t num_channels,
>-                        bool use_mobile_aec,
>-                        bool include_default_apm_processing) {
>-  test::SimulatorBuffers buffers(
>-      render_input_sample_rate_hz, capture_input_sample_rate_hz,
>-      render_output_sample_rate_hz, capture_output_sample_rate_hz, num_channels,
>-      num_channels, num_channels, num_channels);
>-  test::PerformanceTimer render_timer(kNumFramesToProcess);
>-  test::PerformanceTimer capture_timer(kNumFramesToProcess);
>-  test::PerformanceTimer total_timer(kNumFramesToProcess);
>-
>-  webrtc::Config config;
>-  AudioProcessing::Config apm_config;
>-  if (include_default_apm_processing) {
>-    config.Set<DelayAgnostic>(new DelayAgnostic(true));
>-    config.Set<ExtendedFilter>(new ExtendedFilter(true));
>-  }
>-  apm_config.level_controller.enabled = true;
>-  apm_config.residual_echo_detector.enabled = include_default_apm_processing;
>-
>-  std::unique_ptr<AudioProcessing> apm;
>-  apm.reset(AudioProcessingBuilder().Create(config));
>-  ASSERT_TRUE(apm.get());
>-  apm->ApplyConfig(apm_config);
>-
>-  ASSERT_EQ(AudioProcessing::kNoError,
>-            apm->gain_control()->Enable(include_default_apm_processing));
>-  if (use_mobile_aec) {
>-    ASSERT_EQ(AudioProcessing::kNoError,
>-              apm->echo_cancellation()->Enable(false));
>-    ASSERT_EQ(AudioProcessing::kNoError, apm->echo_control_mobile()->Enable(
>-                                             include_default_apm_processing));
>-  } else {
>-    ASSERT_EQ(AudioProcessing::kNoError,
>-              apm->echo_cancellation()->Enable(include_default_apm_processing));
>-    ASSERT_EQ(AudioProcessing::kNoError,
>-              apm->echo_control_mobile()->Enable(false));
>-  }
>-  apm_config.high_pass_filter.enabled = include_default_apm_processing;
>-  ASSERT_EQ(AudioProcessing::kNoError,
>-            apm->noise_suppression()->Enable(include_default_apm_processing));
>-  ASSERT_EQ(AudioProcessing::kNoError,
>-            apm->voice_detection()->Enable(include_default_apm_processing));
>-  ASSERT_EQ(AudioProcessing::kNoError,
>-            apm->level_estimator()->Enable(include_default_apm_processing));
>-
>-  StreamConfig render_input_config(render_input_sample_rate_hz, num_channels,
>-                                   false);
>-  StreamConfig render_output_config(render_output_sample_rate_hz, num_channels,
>-                                    false);
>-  StreamConfig capture_input_config(capture_input_sample_rate_hz, num_channels,
>-                                    false);
>-  StreamConfig capture_output_config(capture_output_sample_rate_hz,
>-                                     num_channels, false);
>-
>-  for (size_t frame_no = 0; frame_no < kToTalNumFrames; ++frame_no) {
>-    buffers.UpdateInputBuffers();
>-
>-    if (frame_no >= kNumFramesToProcessAtWarmup) {
>-      total_timer.StartTimer();
>-      render_timer.StartTimer();
>-    }
>-    ASSERT_EQ(AudioProcessing::kNoError,
>-              apm->ProcessReverseStream(
>-                  &buffers.render_input[0], render_input_config,
>-                  render_output_config, &buffers.render_output[0]));
>-
>-    if (frame_no >= kNumFramesToProcessAtWarmup) {
>-      render_timer.StopTimer();
>-
>-      capture_timer.StartTimer();
>-    }
>-
>-    ASSERT_EQ(AudioProcessing::kNoError, apm->set_stream_delay_ms(0));
>-    ASSERT_EQ(
>-        AudioProcessing::kNoError,
>-        apm->ProcessStream(&buffers.capture_input[0], capture_input_config,
>-                           capture_output_config, &buffers.capture_output[0]));
>-
>-    if (frame_no >= kNumFramesToProcessAtWarmup) {
>-      capture_timer.StopTimer();
>-      total_timer.StopTimer();
>-    }
>-  }
>-
>-  webrtc::test::PrintResultMeanAndError(
>-      "level_controller_call_durations",
>-      "_" + std::to_string(render_input_sample_rate_hz) + "_" +
>-          std::to_string(render_output_sample_rate_hz) + "_" +
>-          std::to_string(capture_input_sample_rate_hz) + "_" +
>-          std::to_string(capture_output_sample_rate_hz) + "Hz_" +
>-          std::to_string(num_channels) + "_channels" + "_render",
>-      test_description, render_timer.GetDurationAverage(),
>-      render_timer.GetDurationStandardDeviation(), "us", false);
>-  webrtc::test::PrintResultMeanAndError(
>-      "level_controller_call_durations",
>-      "_" + std::to_string(render_input_sample_rate_hz) + "_" +
>-          std::to_string(render_output_sample_rate_hz) + "_" +
>-          std::to_string(capture_input_sample_rate_hz) + "_" +
>-          std::to_string(capture_output_sample_rate_hz) + "Hz_" +
>-          std::to_string(num_channels) + "_channels" + "_capture",
>-      test_description, capture_timer.GetDurationAverage(),
>-      capture_timer.GetDurationStandardDeviation(), "us", false);
>-  webrtc::test::PrintResultMeanAndError(
>-      "level_controller_call_durations",
>-      "_" + std::to_string(render_input_sample_rate_hz) + "_" +
>-          std::to_string(render_output_sample_rate_hz) + "_" +
>-          std::to_string(capture_input_sample_rate_hz) + "_" +
>-          std::to_string(capture_output_sample_rate_hz) + "Hz_" +
>-          std::to_string(num_channels) + "_channels" + "_total",
>-      test_description, total_timer.GetDurationAverage(),
>-      total_timer.GetDurationStandardDeviation(), "us", false);
>-}
>-
>-}  // namespace
>-
>-// TODO(peah): Reactivate once issue 7712 has been resolved.
>-TEST(LevelControllerPerformanceTest, DISABLED_StandaloneProcessing) {
>-  int sample_rates_to_test[] = {
>-      AudioProcessing::kSampleRate8kHz, AudioProcessing::kSampleRate16kHz,
>-      AudioProcessing::kSampleRate32kHz, AudioProcessing::kSampleRate48kHz};
>-  for (auto sample_rate : sample_rates_to_test) {
>-    for (size_t num_channels = 1; num_channels <= 2; ++num_channels) {
>-      RunStandaloneSubmodule(sample_rate, num_channels);
>-    }
>-  }
>-}
>-
>-void TestSomeSampleRatesWithApm(const std::string& test_name,
>-                                bool use_mobile_agc,
>-                                bool include_default_apm_processing) {
>-  // Test some stereo combinations first.
>-  size_t num_channels = 2;
>-  RunTogetherWithApm(test_name, 48000, 48000, AudioProcessing::kSampleRate16kHz,
>-                     AudioProcessing::kSampleRate32kHz, num_channels,
>-                     use_mobile_agc, include_default_apm_processing);
>-  RunTogetherWithApm(test_name, 48000, 48000, AudioProcessing::kSampleRate48kHz,
>-                     AudioProcessing::kSampleRate8kHz, num_channels,
>-                     use_mobile_agc, include_default_apm_processing);
>-  RunTogetherWithApm(test_name, 48000, 48000, 44100, 44100, num_channels,
>-                     use_mobile_agc, include_default_apm_processing);
>-
>-  // Then test mono combinations.
>-  num_channels = 1;
>-  RunTogetherWithApm(test_name, 48000, 48000, AudioProcessing::kSampleRate48kHz,
>-                     AudioProcessing::kSampleRate48kHz, num_channels,
>-                     use_mobile_agc, include_default_apm_processing);
>-}
>-
>-// TODO(peah): Reactivate once issue 7712 has been resolved.
>-#if !defined(WEBRTC_ANDROID)
>-TEST(LevelControllerPerformanceTest, DISABLED_ProcessingViaApm) {
>-#else
>-TEST(LevelControllerPerformanceTest, DISABLED_ProcessingViaApm) {
>-#endif
>-  // Run without default APM processing and desktop AGC.
>-  TestSomeSampleRatesWithApm("SimpleLevelControlViaApm", false, false);
>-}
>-
>-// TODO(peah): Reactivate once issue 7712 has been resolved.
>-#if !defined(WEBRTC_ANDROID)
>-TEST(LevelControllerPerformanceTest, DISABLED_InteractionWithDefaultApm) {
>-#else
>-TEST(LevelControllerPerformanceTest, DISABLED_InteractionWithDefaultApm) {
>-#endif
>-  bool include_default_apm_processing = true;
>-  TestSomeSampleRatesWithApm("LevelControlAndDefaultDesktopApm", false,
>-                             include_default_apm_processing);
>-  TestSomeSampleRatesWithApm("LevelControlAndDefaultMobileApm", true,
>-                             include_default_apm_processing);
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/level_controller_constants.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/level_controller_constants.h
>deleted file mode 100644
>index 6cf2cd4c7ef..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/level_controller_constants.h
>+++ /dev/null
>@@ -1,23 +0,0 @@
>-/*
>- *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_LEVEL_CONTROLLER_CONSTANTS_H_
>-#define MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_LEVEL_CONTROLLER_CONSTANTS_H_
>-
>-namespace webrtc {
>-
>-const float kMaxLcGain = 10;
>-const float kMaxLcNoisePower = 100.f * 100.f;
>-const float kTargetLcPeakLevel = 16384.f;
>-const float kTargetLcPeakLeveldBFS = -6.0206f;
>-
>-}  // namespace webrtc
>-
>-#endif  // MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_LEVEL_CONTROLLER_CONSTANTS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/level_controller_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/level_controller_unittest.cc
>deleted file mode 100644
>index cb36ae08f35..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/level_controller_unittest.cc
>+++ /dev/null
>@@ -1,156 +0,0 @@
>-/*
>- *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include <vector>
>-
>-#include "api/array_view.h"
>-#include "api/optional.h"
>-#include "modules/audio_processing/audio_buffer.h"
>-#include "modules/audio_processing/include/audio_processing.h"
>-#include "modules/audio_processing/level_controller/level_controller.h"
>-#include "modules/audio_processing/test/audio_buffer_tools.h"
>-#include "modules/audio_processing/test/bitexactness_tools.h"
>-#include "test/gtest.h"
>-
>-namespace webrtc {
>-namespace {
>-
>-const int kNumFramesToProcess = 1000;
>-
>-// Processes a specified amount of frames, verifies the results and reports
>-// any errors.
>-void RunBitexactnessTest(int sample_rate_hz,
>-                         size_t num_channels,
>-                         rtc::Optional<float> initial_peak_level_dbfs,
>-                         rtc::ArrayView<const float> output_reference) {
>-  LevelController level_controller;
>-  level_controller.Initialize(sample_rate_hz);
>-  if (initial_peak_level_dbfs) {
>-    AudioProcessing::Config::LevelController config;
>-    config.initial_peak_level_dbfs = *initial_peak_level_dbfs;
>-    level_controller.ApplyConfig(config);
>-  }
>-
>-  int samples_per_channel = rtc::CheckedDivExact(sample_rate_hz, 100);
>-  const StreamConfig capture_config(sample_rate_hz, num_channels, false);
>-  AudioBuffer capture_buffer(
>-      capture_config.num_frames(), capture_config.num_channels(),
>-      capture_config.num_frames(), capture_config.num_channels(),
>-      capture_config.num_frames());
>-  test::InputAudioFile capture_file(
>-      test::GetApmCaptureTestVectorFileName(sample_rate_hz));
>-  std::vector<float> capture_input(samples_per_channel * num_channels);
>-  for (size_t frame_no = 0; frame_no < kNumFramesToProcess; ++frame_no) {
>-    ReadFloatSamplesFromStereoFile(samples_per_channel, num_channels,
>-                                   &capture_file, capture_input);
>-
>-    test::CopyVectorToAudioBuffer(capture_config, capture_input,
>-                                  &capture_buffer);
>-
>-    level_controller.Process(&capture_buffer);
>-  }
>-
>-  // Extract test results.
>-  std::vector<float> capture_output;
>-  test::ExtractVectorFromAudioBuffer(capture_config, &capture_buffer,
>-                                     &capture_output);
>-
>-  // Compare the output with the reference. Only the first values of the output
>-  // from last frame processed are compared in order not having to specify all
>-  // preceding frames as testvectors. As the algorithm being tested has a
>-  // memory, testing only the last frame implicitly also tests the preceeding
>-  // frames.
>-  const float kVectorElementErrorBound = 1.0f / 32768.0f;
>-  EXPECT_TRUE(test::VerifyDeinterleavedArray(
>-      capture_config.num_frames(), capture_config.num_channels(),
>-      output_reference, capture_output, kVectorElementErrorBound));
>-}
>-
>-}  // namespace
>-
>-TEST(LevelControllerConfig, ToString) {
>-  AudioProcessing::Config config;
>-  config.level_controller.enabled = true;
>-  config.level_controller.initial_peak_level_dbfs = -6.0206f;
>-  EXPECT_EQ("{enabled: true, initial_peak_level_dbfs: -6.0206}",
>-            LevelController::ToString(config.level_controller));
>-
>-  config.level_controller.enabled = false;
>-  config.level_controller.initial_peak_level_dbfs = -50.f;
>-  EXPECT_EQ("{enabled: false, initial_peak_level_dbfs: -50}",
>-            LevelController::ToString(config.level_controller));
>-}
>-
>-TEST(LevelControlBitExactnessTest, Mono8kHz) {
>-  const float kOutputReference[] = {-0.013939f, -0.012154f, -0.009054f};
>-  RunBitexactnessTest(AudioProcessing::kSampleRate8kHz, 1, rtc::nullopt,
>-                      kOutputReference);
>-}
>-
>-TEST(LevelControlBitExactnessTest, Mono16kHz) {
>-  const float kOutputReference[] = {-0.013706f, -0.013215f, -0.013018f};
>-  RunBitexactnessTest(AudioProcessing::kSampleRate16kHz, 1, rtc::nullopt,
>-                      kOutputReference);
>-}
>-
>-TEST(LevelControlBitExactnessTest, Mono32kHz) {
>-  const float kOutputReference[] = {-0.014495f, -0.016425f, -0.016085f};
>-  RunBitexactnessTest(AudioProcessing::kSampleRate32kHz, 1, rtc::nullopt,
>-                      kOutputReference);
>-}
>-
>-// TODO(peah): Investigate why this particular testcase differ between Android
>-// and the rest of the platforms.
>-TEST(LevelControlBitExactnessTest, Mono48kHz) {
>-#if !(defined(WEBRTC_ARCH_ARM64) || defined(WEBRTC_ARCH_ARM) || \
>-      defined(WEBRTC_ANDROID))
>-  const float kOutputReference[] = {-0.014277f, -0.015180f, -0.017437f};
>-#else
>-  const float kOutputReference[] = {-0.014306f, -0.015209f, -0.017466f};
>-#endif
>-  RunBitexactnessTest(AudioProcessing::kSampleRate48kHz, 1, rtc::nullopt,
>-                      kOutputReference);
>-}
>-
>-TEST(LevelControlBitExactnessTest, Stereo8kHz) {
>-  const float kOutputReference[] = {-0.014063f, -0.008450f, -0.012159f,
>-                                    -0.051967f, -0.023202f, -0.047858f};
>-  RunBitexactnessTest(AudioProcessing::kSampleRate8kHz, 2, rtc::nullopt,
>-                      kOutputReference);
>-}
>-
>-TEST(LevelControlBitExactnessTest, Stereo16kHz) {
>-  const float kOutputReference[] = {-0.012714f, -0.005896f, -0.012220f,
>-                                    -0.053306f, -0.024549f, -0.051527f};
>-  RunBitexactnessTest(AudioProcessing::kSampleRate16kHz, 2, rtc::nullopt,
>-                      kOutputReference);
>-}
>-
>-TEST(LevelControlBitExactnessTest, Stereo32kHz) {
>-  const float kOutputReference[] = {-0.011764f, -0.007044f, -0.013472f,
>-                                    -0.053537f, -0.026322f, -0.056253f};
>-  RunBitexactnessTest(AudioProcessing::kSampleRate32kHz, 2, rtc::nullopt,
>-                      kOutputReference);
>-}
>-
>-TEST(LevelControlBitExactnessTest, Stereo48kHz) {
>-  const float kOutputReference[] = {-0.010643f, -0.006334f, -0.011377f,
>-                                    -0.049088f, -0.023600f, -0.050465f};
>-  RunBitexactnessTest(AudioProcessing::kSampleRate48kHz, 2, rtc::nullopt,
>-                      kOutputReference);
>-}
>-
>-TEST(LevelControlBitExactnessTest, MonoInitial48kHz) {
>-  const float kOutputReference[] = {-0.013884f, -0.014761f, -0.016951f};
>-  RunBitexactnessTest(AudioProcessing::kSampleRate48kHz, 1, -50,
>-                      kOutputReference);
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/peak_level_estimator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/peak_level_estimator.cc
>deleted file mode 100644
>index e78d74f514f..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/peak_level_estimator.cc
>+++ /dev/null
>@@ -1,74 +0,0 @@
>-/*
>- *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "modules/audio_processing/level_controller/peak_level_estimator.h"
>-
>-#include <algorithm>
>-
>-#include "modules/audio_processing/audio_buffer.h"
>-#include "modules/audio_processing/logging/apm_data_dumper.h"
>-
>-namespace webrtc {
>-namespace {
>-
>-constexpr float kMinLevel = 30.f;
>-
>-}  // namespace
>-
>-PeakLevelEstimator::PeakLevelEstimator(float initial_peak_level_dbfs) {
>-  Initialize(initial_peak_level_dbfs);
>-}
>-
>-PeakLevelEstimator::~PeakLevelEstimator() {}
>-
>-void PeakLevelEstimator::Initialize(float initial_peak_level_dbfs) {
>-  RTC_DCHECK_LE(-100.f, initial_peak_level_dbfs);
>-  RTC_DCHECK_GE(0.f, initial_peak_level_dbfs);
>-
>-  peak_level_ = std::pow(10.f, initial_peak_level_dbfs / 20.f) * 32768.f;
>-  peak_level_ = std::max(peak_level_, kMinLevel);
>-
>-  hold_counter_ = 0;
>-  initialization_phase_ = true;
>-}
>-
>-float PeakLevelEstimator::Analyze(SignalClassifier::SignalType signal_type,
>-                                  float frame_peak_level) {
>-  if (frame_peak_level == 0) {
>-    RTC_DCHECK_LE(kMinLevel, peak_level_);
>-    return peak_level_;
>-  }
>-
>-  if (peak_level_ < frame_peak_level) {
>-    // Smoothly update the estimate upwards when the frame peak level is
>-    // higher than the estimate.
>-    peak_level_ += 0.1f * (frame_peak_level - peak_level_);
>-    hold_counter_ = 100;
>-    initialization_phase_ = false;
>-  } else {
>-    hold_counter_ = std::max(0, hold_counter_ - 1);
>-
>-    // When the signal is highly non-stationary, update the estimate slowly
>-    // downwards if the estimate is lower than the frame peak level.
>-    if ((signal_type == SignalClassifier::SignalType::kHighlyNonStationary &&
>-         hold_counter_ == 0) ||
>-        initialization_phase_) {
>-      peak_level_ =
>-          std::max(peak_level_ + 0.01f * (frame_peak_level - peak_level_),
>-                   peak_level_ * 0.995f);
>-    }
>-  }
>-
>-  peak_level_ = std::max(peak_level_, kMinLevel);
>-
>-  return peak_level_;
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/peak_level_estimator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/peak_level_estimator.h
>deleted file mode 100644
>index 0aa55d2d558..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/peak_level_estimator.h
>+++ /dev/null
>@@ -1,37 +0,0 @@
>-/*
>- *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_PEAK_LEVEL_ESTIMATOR_H_
>-#define MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_PEAK_LEVEL_ESTIMATOR_H_
>-
>-#include "modules/audio_processing/level_controller/level_controller_constants.h"
>-#include "modules/audio_processing/level_controller/signal_classifier.h"
>-#include "rtc_base/constructormagic.h"
>-
>-namespace webrtc {
>-
>-class PeakLevelEstimator {
>- public:
>-  explicit PeakLevelEstimator(float initial_peak_level_dbfs);
>-  ~PeakLevelEstimator();
>-  void Initialize(float initial_peak_level_dbfs);
>-  float Analyze(SignalClassifier::SignalType signal_type,
>-                float frame_peak_level);
>- private:
>-  float peak_level_;
>-  int hold_counter_;
>-  bool initialization_phase_;
>-
>-  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(PeakLevelEstimator);
>-};
>-
>-}  // namespace webrtc
>-
>-#endif  // MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_PEAK_LEVEL_ESTIMATOR_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/saturating_gain_estimator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/saturating_gain_estimator.cc
>deleted file mode 100644
>index 60110c684ba..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/saturating_gain_estimator.cc
>+++ /dev/null
>@@ -1,48 +0,0 @@
>-/*
>- *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "modules/audio_processing/level_controller/saturating_gain_estimator.h"
>-
>-#include <math.h>
>-#include <algorithm>
>-
>-#include "modules/audio_processing/level_controller/level_controller_constants.h"
>-#include "modules/audio_processing/logging/apm_data_dumper.h"
>-
>-namespace webrtc {
>-
>-SaturatingGainEstimator::SaturatingGainEstimator() {
>-  Initialize();
>-}
>-
>-SaturatingGainEstimator::~SaturatingGainEstimator() {}
>-
>-void SaturatingGainEstimator::Initialize() {
>-  saturating_gain_ = kMaxLcGain;
>-  saturating_gain_hold_counter_ = 0;
>-}
>-
>-void SaturatingGainEstimator::Update(float gain, int num_saturations) {
>-  bool too_many_saturations = (num_saturations > 2);
>-
>-  if (too_many_saturations) {
>-    saturating_gain_ = 0.95f * gain;
>-    saturating_gain_hold_counter_ = 1000;
>-  } else {
>-    saturating_gain_hold_counter_ =
>-        std::max(0, saturating_gain_hold_counter_ - 1);
>-    if (saturating_gain_hold_counter_ == 0) {
>-      saturating_gain_ *= 1.001f;
>-      saturating_gain_ = std::min(kMaxLcGain, saturating_gain_);
>-    }
>-  }
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/saturating_gain_estimator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/saturating_gain_estimator.h
>deleted file mode 100644
>index 8980f4ef972..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/level_controller/saturating_gain_estimator.h
>+++ /dev/null
>@@ -1,37 +0,0 @@
>-/*
>- *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_SATURATING_GAIN_ESTIMATOR_H_
>-#define MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_SATURATING_GAIN_ESTIMATOR_H_
>-
>-#include "rtc_base/constructormagic.h"
>-
>-namespace webrtc {
>-
>-class ApmDataDumper;
>-
>-class SaturatingGainEstimator {
>- public:
>-  SaturatingGainEstimator();
>-  ~SaturatingGainEstimator();
>-  void Initialize();
>-  void Update(float gain, int num_saturations);
>-  float GetGain() const { return saturating_gain_; }
>-
>- private:
>-  float saturating_gain_;
>-  int saturating_gain_hold_counter_;
>-
>-  RTC_DISALLOW_COPY_AND_ASSIGN(SaturatingGainEstimator);
>-};
>-
>-}  // namespace webrtc
>-
>-#endif  // MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_SATURATING_GAIN_ESTIMATOR_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/logging/apm_data_dumper.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/logging/apm_data_dumper.cc
>index 2f6a6d6a809..e2e8602b989 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/logging/apm_data_dumper.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/logging/apm_data_dumper.cc
>@@ -10,8 +10,7 @@
> 
> #include "modules/audio_processing/logging/apm_data_dumper.h"
> 
>-#include <sstream>
>-
>+#include "rtc_base/strings/string_builder.h"
> #include "rtc_base/stringutils.h"
> 
> // Check to verify that the define is properly set.
>@@ -29,7 +28,8 @@ std::string FormFileName(const char* name,
>                          int instance_index,
>                          int reinit_index,
>                          const std::string& suffix) {
>-  std::stringstream ss;
>+  char buf[1024];
>+  rtc::SimpleStringBuilder ss(buf);
>   ss << name << "_" << instance_index << "-" << reinit_index << suffix;
>   return ss.str();
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/low_cut_filter_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/low_cut_filter_unittest.cc
>index d98d665d379..d7b3cb963d3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/low_cut_filter_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/low_cut_filter_unittest.cc
>@@ -51,12 +51,10 @@ void RunBitexactnessTest(int sample_rate,
>       (stream_config.num_frames() * stream_config.num_channels());
>   for (size_t frame_no = 0; frame_no < num_frames_to_process; ++frame_no) {
>     std::vector<float> frame_input(
>-        input.begin() +
>-            stream_config.num_frames() * stream_config.num_channels() *
>-                frame_no,
>-        input.begin() +
>-            stream_config.num_frames() * stream_config.num_channels() *
>-                (frame_no + 1));
>+        input.begin() + stream_config.num_frames() *
>+                            stream_config.num_channels() * frame_no,
>+        input.begin() + stream_config.num_frames() *
>+                            stream_config.num_channels() * (frame_no + 1));
> 
>     output = ProcessOneFrame(frame_input, stream_config, &low_cut_filter);
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/noise_suppression_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/noise_suppression_impl.cc
>index 8dd713f4e50..15d404323c3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/noise_suppression_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/noise_suppression_impl.cc
>@@ -37,10 +37,9 @@ class NoiseSuppressionImpl::Suppressor {
>     int error = NS_INIT(state_, sample_rate_hz);
>     RTC_DCHECK_EQ(0, error);
>   }
>-  ~Suppressor() {
>-    NS_FREE(state_);
>-  }
>+  ~Suppressor() { NS_FREE(state_); }
>   NsState* state() { return state_; }
>+
>  private:
>   NsState* state_ = nullptr;
>   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(Suppressor);
>@@ -96,15 +95,11 @@ void NoiseSuppressionImpl::ProcessCaptureAudio(AudioBuffer* audio) {
>   RTC_DCHECK_EQ(suppressors_.size(), audio->num_channels());
>   for (size_t i = 0; i < suppressors_.size(); i++) {
> #if defined(WEBRTC_NS_FLOAT)
>-    WebRtcNs_Process(suppressors_[i]->state(),
>-                     audio->split_bands_const_f(i),
>-                     audio->num_bands(),
>-                     audio->split_bands_f(i));
>+    WebRtcNs_Process(suppressors_[i]->state(), audio->split_bands_const_f(i),
>+                     audio->num_bands(), audio->split_bands_f(i));
> #elif defined(WEBRTC_NS_FIXED)
>-    WebRtcNsx_Process(suppressors_[i]->state(),
>-                      audio->split_bands_const(i),
>-                      audio->num_bands(),
>-                      audio->split_bands(i));
>+    WebRtcNsx_Process(suppressors_[i]->state(), audio->split_bands_const(i),
>+                      audio->num_bands(), audio->split_bands(i));
> #endif
>   }
> }
>@@ -190,8 +185,8 @@ std::vector<float> NoiseSuppressionImpl::NoiseEstimate() {
>   noise_estimate.assign(WebRtcNsx_num_freq(), 0.f);
>   for (auto& suppressor : suppressors_) {
>     int q_noise;
>-    const uint32_t* noise = WebRtcNsx_noise_estimate(suppressor->state(),
>-                                                     &q_noise);
>+    const uint32_t* noise =
>+        WebRtcNsx_noise_estimate(suppressor->state(), &q_noise);
>     const float kNormalizationFactor =
>         1.f / ((1 << q_noise) * suppressors_.size());
>     for (size_t i = 0; i < noise_estimate.size(); ++i) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/noise_suppression_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/noise_suppression_unittest.cc
>index 0b734fdd3f2..b2074859b04 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/noise_suppression_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/noise_suppression_unittest.cc
>@@ -97,18 +97,18 @@ void RunBitexactnessTest(int sample_rate_hz,
> TEST(NoiseSuppresionBitExactnessTest, Mono8kHzLow) {
> #if defined(WEBRTC_ARCH_ARM64)
>   const float kSpeechProbabilityReference = -4.0f;
>-  const float kNoiseEstimateReference[] =
>-      {1432.341431f, 3321.919922f, 7677.521973f};
>+  const float kNoiseEstimateReference[] = {1432.341431f, 3321.919922f,
>+                                           7677.521973f};
>   const float kOutputReference[] = {0.003510f, 0.004517f, 0.004669f};
> #elif defined(WEBRTC_ARCH_ARM)
>   const float kSpeechProbabilityReference = -4.0f;
>-  const float kNoiseEstimateReference[] =
>-      {1432.341431f, 3321.919922f, 7677.521973f};
>+  const float kNoiseEstimateReference[] = {1432.341431f, 3321.919922f,
>+                                           7677.521973f};
>   const float kOutputReference[] = {0.003510f, 0.004517f, 0.004669f};
> #else
>   const float kSpeechProbabilityReference = 0.73421317f;
>-  const float kNoiseEstimateReference[] =
>-      {1175.266113f, 3289.305908f, 7532.991211f};
>+  const float kNoiseEstimateReference[] = {1175.266113f, 3289.305908f,
>+                                           7532.991211f};
>   const float kOutputReference[] = {0.003263f, 0.004402f, 0.004537f};
> #endif
> 
>@@ -120,18 +120,18 @@ TEST(NoiseSuppresionBitExactnessTest, Mono8kHzLow) {
> TEST(NoiseSuppresionBitExactnessTest, Mono16kHzLow) {
> #if defined(WEBRTC_ARCH_ARM64)
>   const float kSpeechProbabilityReference = -4.0f;
>-  const float kNoiseEstimateReference[] =
>-      {2534.461914f, 6277.638672f, 14367.499023f};
>+  const float kNoiseEstimateReference[] = {2534.461914f, 6277.638672f,
>+                                           14367.499023f};
>   const float kOutputReference[] = {0.003449f, 0.004334f, 0.004303f};
> #elif defined(WEBRTC_ARCH_ARM)
>   const float kSpeechProbabilityReference = -4.0f;
>-  const float kNoiseEstimateReference[] =
>-      {2534.461914f, 6277.638672f, 14367.499023f};
>+  const float kNoiseEstimateReference[] = {2534.461914f, 6277.638672f,
>+                                           14367.499023f};
>   const float kOutputReference[] = {0.003449f, 0.004334f, 0.004303f};
> #else
>   const float kSpeechProbabilityReference = 0.71672988f;
>-  const float kNoiseEstimateReference[] =
>-      {2151.313965f, 6509.765137f, 15658.848633f};
>+  const float kNoiseEstimateReference[] = {2151.313965f, 6509.765137f,
>+                                           15658.848633f};
>   const float kOutputReference[] = {0.003574f, 0.004494f, 0.004499f};
> #endif
> 
>@@ -143,18 +143,18 @@ TEST(NoiseSuppresionBitExactnessTest, Mono16kHzLow) {
> TEST(NoiseSuppresionBitExactnessTest, Mono32kHzLow) {
> #if defined(WEBRTC_ARCH_ARM64)
>   const float kSpeechProbabilityReference = -4.0f;
>-  const float kNoiseEstimateReference[] =
>-      {2540.059082f, 6317.822754f, 14440.845703f};
>+  const float kNoiseEstimateReference[] = {2540.059082f, 6317.822754f,
>+                                           14440.845703f};
>   const float kOutputReference[] = {0.001679f, 0.002411f, 0.002594f};
> #elif defined(WEBRTC_ARCH_ARM)
>   const float kSpeechProbabilityReference = -4.0f;
>-  const float kNoiseEstimateReference[] =
>-      {2540.059082f, 6317.822754f, 14440.845703f};
>+  const float kNoiseEstimateReference[] = {2540.059082f, 6317.822754f,
>+                                           14440.845703f};
>   const float kOutputReference[] = {0.001679f, 0.002411f, 0.002594f};
> #else
>   const float kSpeechProbabilityReference = 0.67999554f;
>-  const float kNoiseEstimateReference[] =
>-      {2149.780518f, 7076.936035f, 14939.945312f};
>+  const float kNoiseEstimateReference[] = {2149.780518f, 7076.936035f,
>+                                           14939.945312f};
>   const float kOutputReference[] = {0.001221f, 0.001984f, 0.002228f};
> #endif
> 
>@@ -166,18 +166,18 @@ TEST(NoiseSuppresionBitExactnessTest, Mono32kHzLow) {
> TEST(NoiseSuppresionBitExactnessTest, Mono48kHzLow) {
> #if defined(WEBRTC_ARCH_ARM64)
>   const float kSpeechProbabilityReference = -4.0f;
>-  const float kNoiseEstimateReference[] =
>-      {2564.605713f, 6213.656250f, 13372.284180f};
>+  const float kNoiseEstimateReference[] = {2564.605713f, 6213.656250f,
>+                                           13372.284180f};
>   const float kOutputReference[] = {-0.013185f, -0.012769f, -0.012023f};
> #elif defined(WEBRTC_ARCH_ARM)
>   const float kSpeechProbabilityReference = -4.0f;
>-  const float kNoiseEstimateReference[] =
>-      {2564.605713f, 6213.656250f, 13372.284180f};
>+  const float kNoiseEstimateReference[] = {2564.605713f, 6213.656250f,
>+                                           13372.284180f};
>   const float kOutputReference[] = {-0.013185f, -0.012769f, -0.012023f};
> #else
>   const float kSpeechProbabilityReference = 0.70645678f;
>-  const float kNoiseEstimateReference[] =
>-      {2168.783203f, 6902.895508f, 13190.677734f};
>+  const float kNoiseEstimateReference[] = {2168.783203f, 6902.895508f,
>+                                           13190.677734f};
>   const float kOutputReference[] = {-0.013062f, -0.012657f, -0.011934f};
> #endif
> 
>@@ -189,20 +189,20 @@ TEST(NoiseSuppresionBitExactnessTest, Mono48kHzLow) {
> TEST(NoiseSuppresionBitExactnessTest, Stereo16kHzLow) {
> #if defined(WEBRTC_ARCH_ARM64)
>   const float kSpeechProbabilityReference = -4.0f;
>-  const float kNoiseEstimateReference[] =
>-      {9992.127930f, 12689.569336f, 11589.296875f};
>+  const float kNoiseEstimateReference[] = {9992.127930f, 12689.569336f,
>+                                           11589.296875f};
>   const float kOutputReference[] = {-0.011108f, -0.007904f, -0.012390f,
>                                     -0.002441f, 0.000855f,  -0.003204f};
> #elif defined(WEBRTC_ARCH_ARM)
>   const float kSpeechProbabilityReference = -4.0f;
>-  const float kNoiseEstimateReference[] =
>-      {10321.353516f, 12133.852539f, 10923.060547f};
>+  const float kNoiseEstimateReference[] = {10321.353516f, 12133.852539f,
>+                                           10923.060547f};
>   const float kOutputReference[] = {-0.011108f, -0.007904f, -0.012390f,
>                                     -0.002472f, 0.000916f,  -0.003235f};
> #else
>   const float kSpeechProbabilityReference = 0.67230678f;
>-  const float kNoiseEstimateReference[] =
>-      {9771.250000f, 11329.377930f, 10503.052734f};
>+  const float kNoiseEstimateReference[] = {9771.250000f, 11329.377930f,
>+                                           10503.052734f};
>   const float kOutputReference[] = {-0.011459f, -0.008110f, -0.012728f,
>                                     -0.002399f, 0.001018f,  -0.003189f};
> #endif
>@@ -215,18 +215,18 @@ TEST(NoiseSuppresionBitExactnessTest, Stereo16kHzLow) {
> TEST(NoiseSuppresionBitExactnessTest, Mono16kHzModerate) {
> #if defined(WEBRTC_ARCH_ARM64)
>   const float kSpeechProbabilityReference = -4.0f;
>-  const float kNoiseEstimateReference[] =
>-      {2057.085938f, 7601.055176f, 19666.187500f};
>+  const float kNoiseEstimateReference[] = {2057.085938f, 7601.055176f,
>+                                           19666.187500f};
>   const float kOutputReference[] = {0.004669f, 0.005524f, 0.005432f};
> #elif defined(WEBRTC_ARCH_ARM)
>   const float kSpeechProbabilityReference = -4.0f;
>-  const float kNoiseEstimateReference[] =
>-      {2244.497803f, 6864.164062f, 16726.523438f};
>+  const float kNoiseEstimateReference[] = {2244.497803f, 6864.164062f,
>+                                           16726.523438f};
>   const float kOutputReference[] = {0.004669f, 0.005615f, 0.005585f};
> #else
>   const float kSpeechProbabilityReference = 0.70897013f;
>-  const float kNoiseEstimateReference[] =
>-      {2171.490723f, 6553.567871f, 15626.562500f};
>+  const float kNoiseEstimateReference[] = {2171.490723f, 6553.567871f,
>+                                           15626.562500f};
>   const float kOutputReference[] = {0.004513f, 0.005590f, 0.005614f};
> #endif
> 
>@@ -238,18 +238,18 @@ TEST(NoiseSuppresionBitExactnessTest, Mono16kHzModerate) {
> TEST(NoiseSuppresionBitExactnessTest, Mono16kHzHigh) {
> #if defined(WEBRTC_ARCH_ARM64)
>   const float kSpeechProbabilityReference = -4.0f;
>-  const float kNoiseEstimateReference[] =
>-      {2095.148193f, 7698.553711f, 19689.533203f};
>+  const float kNoiseEstimateReference[] = {2095.148193f, 7698.553711f,
>+                                           19689.533203f};
>   const float kOutputReference[] = {0.004639f, 0.005402f, 0.005310f};
> #elif defined(WEBRTC_ARCH_ARM)
>   const float kSpeechProbabilityReference = -4.0f;
>-  const float kNoiseEstimateReference[] =
>-      {2282.515625f, 6984.408203f, 16920.960938f};
>+  const float kNoiseEstimateReference[] = {2282.515625f, 6984.408203f,
>+                                           16920.960938f};
>   const float kOutputReference[] = {0.004547f, 0.005432f, 0.005402f};
> #else
>   const float kSpeechProbabilityReference = 0.70106733f;
>-  const float kNoiseEstimateReference[] =
>-      {2224.968506f, 6712.025879f, 15785.087891f};
>+  const float kNoiseEstimateReference[] = {2224.968506f, 6712.025879f,
>+                                           15785.087891f};
>   const float kOutputReference[] = {0.004394f, 0.005406f, 0.005416f};
> #endif
> 
>@@ -261,18 +261,18 @@ TEST(NoiseSuppresionBitExactnessTest, Mono16kHzHigh) {
> TEST(NoiseSuppresionBitExactnessTest, Mono16kHzVeryHigh) {
> #if defined(WEBRTC_ARCH_ARM64)
>   const float kSpeechProbabilityReference = -4.0f;
>-  const float kNoiseEstimateReference[] =
>-      {2677.733398f, 6186.987305f, 14365.744141f};
>+  const float kNoiseEstimateReference[] = {2677.733398f, 6186.987305f,
>+                                           14365.744141f};
>   const float kOutputReference[] = {0.004273f, 0.005127f, 0.005188f};
> #elif defined(WEBRTC_ARCH_ARM)
>   const float kSpeechProbabilityReference = -4.0f;
>-  const float kNoiseEstimateReference[] =
>-      {2677.733398f, 6186.987305f, 14365.744141f};
>+  const float kNoiseEstimateReference[] = {2677.733398f, 6186.987305f,
>+                                           14365.744141f};
>   const float kOutputReference[] = {0.004273f, 0.005127f, 0.005188f};
> #else
>   const float kSpeechProbabilityReference = 0.70281971f;
>-  const float kNoiseEstimateReference[] =
>-      {2254.347900f, 6723.699707f, 15771.625977f};
>+  const float kNoiseEstimateReference[] = {2254.347900f, 6723.699707f,
>+                                           15771.625977f};
>   const float kOutputReference[] = {0.004321f, 0.005247f, 0.005263f};
> #endif
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/ns/defines.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/ns/defines.h
>index 66b45a9565e..d6abfea7e79 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/ns/defines.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/ns/defines.h
>@@ -11,39 +11,45 @@
> #ifndef MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_DEFINES_H_
> #define MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_DEFINES_H_
> 
>-#define BLOCKL_MAX          160 // max processing block length: 160
>-#define ANAL_BLOCKL_MAX     256 // max analysis block length: 256
>-#define HALF_ANAL_BLOCKL    129 // half max analysis block length + 1
>-#define NUM_HIGH_BANDS_MAX  2   // max number of high bands: 2
>+#define BLOCKL_MAX 160        // max processing block length: 160
>+#define ANAL_BLOCKL_MAX 256   // max analysis block length: 256
>+#define HALF_ANAL_BLOCKL 129  // half max analysis block length + 1
>+#define NUM_HIGH_BANDS_MAX 2  // max number of high bands: 2
> 
>-#define QUANTILE            (float)0.25
>+#define QUANTILE (float)0.25
> 
>-#define SIMULT              3
>-#define END_STARTUP_LONG    200
>-#define END_STARTUP_SHORT   50
>-#define FACTOR              (float)40.0
>-#define WIDTH               (float)0.01
>+#define SIMULT 3
>+#define END_STARTUP_LONG 200
>+#define END_STARTUP_SHORT 50
>+#define FACTOR (float)40.0
>+#define WIDTH (float)0.01
> 
> // Length of fft work arrays.
>-#define IP_LENGTH (ANAL_BLOCKL_MAX >> 1) // must be at least ceil(2 + sqrt(ANAL_BLOCKL_MAX/2))
>+#define IP_LENGTH \
>+  (ANAL_BLOCKL_MAX >> 1)  // must be at least ceil(2 + sqrt(ANAL_BLOCKL_MAX/2))
> #define W_LENGTH (ANAL_BLOCKL_MAX >> 1)
> 
>-//PARAMETERS FOR NEW METHOD
>-#define DD_PR_SNR           (float)0.98 // DD update of prior SNR
>-#define LRT_TAVG            (float)0.50 // tavg parameter for LRT (previously 0.90)
>-#define SPECT_FL_TAVG       (float)0.30 // tavg parameter for spectral flatness measure
>-#define SPECT_DIFF_TAVG     (float)0.30 // tavg parameter for spectral difference measure
>-#define PRIOR_UPDATE        (float)0.10 // update parameter of prior model
>-#define NOISE_UPDATE        (float)0.90 // update parameter for noise
>-#define SPEECH_UPDATE       (float)0.99 // update parameter when likely speech
>-#define WIDTH_PR_MAP        (float)4.0  // width parameter in sigmoid map for prior model
>-#define LRT_FEATURE_THR     (float)0.5  // default threshold for LRT feature
>-#define SF_FEATURE_THR      (float)0.5  // default threshold for Spectral Flatness feature
>-#define SD_FEATURE_THR      (float)0.5  // default threshold for Spectral Difference feature
>-#define PROB_RANGE          (float)0.20 // probability threshold for noise state in
>-                                        // speech/noise likelihood
>-#define HIST_PAR_EST         1000       // histogram size for estimation of parameters
>-#define GAMMA_PAUSE         (float)0.05 // update for conservative noise estimate
>+// PARAMETERS FOR NEW METHOD
>+#define DD_PR_SNR (float)0.98  // DD update of prior SNR
>+#define LRT_TAVG (float)0.50   // tavg parameter for LRT (previously 0.90)
>+#define SPECT_FL_TAVG \
>+  (float)0.30  // tavg parameter for spectral flatness measure
>+#define SPECT_DIFF_TAVG \
>+  (float)0.30  // tavg parameter for spectral difference measure
>+#define PRIOR_UPDATE (float)0.10   // update parameter of prior model
>+#define NOISE_UPDATE (float)0.90   // update parameter for noise
>+#define SPEECH_UPDATE (float)0.99  // update parameter when likely speech
>+#define WIDTH_PR_MAP \
>+  (float)4.0  // width parameter in sigmoid map for prior model
>+#define LRT_FEATURE_THR (float)0.5  // default threshold for LRT feature
>+#define SF_FEATURE_THR \
>+  (float)0.5  // default threshold for Spectral Flatness feature
>+#define SD_FEATURE_THR \
>+  (float)0.5  // default threshold for Spectral Difference feature
>+#define PROB_RANGE (float)0.20   // probability threshold for noise state in
>+                                 // speech/noise likelihood
>+#define HIST_PAR_EST 1000        // histogram size for estimation of parameters
>+#define GAMMA_PAUSE (float)0.05  // update for conservative noise estimate
> //
>-#define B_LIM               (float)0.5  // threshold in final energy gain factor calculation
>-#endif // MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_DEFINES_H_
>+#define B_LIM (float)0.5  // threshold in final energy gain factor calculation
>+#endif                    // MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_DEFINES_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/ns/noise_suppression.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/ns/noise_suppression.h
>index fa5da701899..0775ffa243c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/ns/noise_suppression.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/ns/noise_suppression.h
>@@ -12,8 +12,7 @@
> #define MODULES_AUDIO_PROCESSING_NS_NOISE_SUPPRESSION_H_
> 
> #include <stddef.h>
>-
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include <stdint.h>
> 
> typedef struct NsHandleT NsHandle;
> 
>@@ -93,9 +92,9 @@ void WebRtcNs_Analyze(NsHandle* NS_inst, const float* spframe);
>  *      - outframe      : Pointer to output frame for each band
>  */
> void WebRtcNs_Process(NsHandle* NS_inst,
>-                     const float* const* spframe,
>-                     size_t num_bands,
>-                     float* const* outframe);
>+                      const float* const* spframe,
>+                      size_t num_bands,
>+                      float* const* outframe);
> 
> /* Returns the internally used prior speech probability of the current frame.
>  * There is a frequency bin based one as well, with which this should not be
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/ns/noise_suppression_x.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/ns/noise_suppression_x.h
>index f25fb7aba1a..972784e5817 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/ns/noise_suppression_x.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/ns/noise_suppression_x.h
>@@ -12,8 +12,7 @@
> #define MODULES_AUDIO_PROCESSING_NS_NOISE_SUPPRESSION_X_H_
> 
> #include <stddef.h>
>-
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include <stdint.h>
> 
> typedef struct NsxHandleT NsxHandle;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/ns/ns_core.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/ns/ns_core.c
>index 3345b2451e3..bc5dd6d4511 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/ns/ns_core.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/ns/ns_core.c
>@@ -13,8 +13,8 @@
> #include <stdlib.h>
> 
> #include "rtc_base/checks.h"
>-#include "common_audio/fft4g.h"
> #include "common_audio/signal_processing/include/signal_processing_library.h"
>+#include "common_audio/third_party/fft4g/fft4g.h"
> #include "modules/audio_processing/ns/noise_suppression.h"
> #include "modules/audio_processing/ns/ns_core.h"
> #include "modules/audio_processing/ns/windows_private.h"
>@@ -1081,6 +1081,7 @@ void WebRtcNs_AnalyzeCore(NoiseSuppressionC* self, const float* speechFrame) {
>     // Depending on the duration of the inactive signal it takes a
>     // considerable amount of time for the system to learn what is noise and
>     // what is speech.
>+    self->signalEnergy = 0;
>     return;
>   }
> 
>@@ -1239,7 +1240,7 @@ void WebRtcNs_ProcessCore(NoiseSuppressionC* self,
> 
>   Windowing(self->window, self->dataBuf, self->anaLen, winData);
>   energy1 = Energy(winData, self->anaLen);
>-  if (energy1 == 0.0) {
>+  if (energy1 == 0.0 || self->signalEnergy == 0) {
>     // Synthesize the special case of zero input.
>     // Read out fully processed segment.
>     for (i = self->windShift; i < self->blockLen + self->windShift; i++) {
>@@ -1379,6 +1380,7 @@ void WebRtcNs_ProcessCore(NoiseSuppressionC* self,
>       sumMagnAnalyze += self->magnPrevAnalyze[i];
>       sumMagnProcess += self->magnPrevProcess[i];
>     }
>+    RTC_DCHECK_GT(sumMagnAnalyze, 0);
>     avgProbSpeechHB *= sumMagnProcess / sumMagnAnalyze;
>     // Average filter gain from low band.
>     // Average over second half (i.e., 4->8kHz) of frequencies spectrum.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/ns/ns_core.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/ns/ns_core.h
>index 97f76baf0d6..e90b8ce39b3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/ns/ns_core.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/ns/ns_core.h
>@@ -78,26 +78,26 @@ typedef struct NoiseSuppressionC_ {
>   float wfft[W_LENGTH];
> 
>   // Parameters for new method: some not needed, will reduce/cleanup later.
>-  int32_t blockInd;  // Frame index counter.
>+  int32_t blockInd;        // Frame index counter.
>   int modelUpdatePars[4];  // Parameters for updating or estimating.
>   // Thresholds/weights for prior model.
>-  float priorModelPars[7];  // Parameters for prior model.
>-  float noise[HALF_ANAL_BLOCKL];  // Noise spectrum from current frame.
>+  float priorModelPars[7];            // Parameters for prior model.
>+  float noise[HALF_ANAL_BLOCKL];      // Noise spectrum from current frame.
>   float noisePrev[HALF_ANAL_BLOCKL];  // Noise spectrum from previous frame.
>   // Magnitude spectrum of previous analyze frame.
>   float magnPrevAnalyze[HALF_ANAL_BLOCKL];
>   // Magnitude spectrum of previous process frame.
>   float magnPrevProcess[HALF_ANAL_BLOCKL];
>   float logLrtTimeAvg[HALF_ANAL_BLOCKL];  // Log LRT factor with time-smoothing.
>-  float priorSpeechProb;  // Prior speech/noise probability.
>+  float priorSpeechProb;                  // Prior speech/noise probability.
>   float featureData[7];
>   // Conservative noise spectrum estimate.
>   float magnAvgPause[HALF_ANAL_BLOCKL];
>   float signalEnergy;  // Energy of |magn|.
>   float sumMagn;
>-  float whiteNoiseLevel;  // Initial noise estimate.
>+  float whiteNoiseLevel;                // Initial noise estimate.
>   float initMagnEst[HALF_ANAL_BLOCKL];  // Initial magnitude spectrum estimate.
>-  float pinkNoiseNumerator;  // Pink noise parameter: numerator.
>+  float pinkNoiseNumerator;             // Pink noise parameter: numerator.
>   float pinkNoiseExp;  // Pink noise parameter: power of frequencies.
>   float parametricNoise[HALF_ANAL_BLOCKL];
>   // Parameters for feature extraction.
>@@ -110,7 +110,6 @@ typedef struct NoiseSuppressionC_ {
>   float speechProb[HALF_ANAL_BLOCKL];  // Final speech/noise prob: prior + LRT.
>   // Buffering data for HB.
>   float dataBufHB[NUM_HIGH_BANDS_MAX][ANAL_BLOCKL_MAX];
>-
> } NoiseSuppressionC;
> 
> #ifdef __cplusplus
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/ns/nsx_core.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/ns/nsx_core.h
>index c8097f79464..9e9d142818b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/ns/nsx_core.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/ns/nsx_core.h
>@@ -17,101 +17,99 @@
> 
> #include "common_audio/signal_processing/include/signal_processing_library.h"
> #include "modules/audio_processing/ns/nsx_defines.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> typedef struct NoiseSuppressionFixedC_ {
>-  uint32_t                fs;
>-
>-  const int16_t*          window;
>-  int16_t                 analysisBuffer[ANAL_BLOCKL_MAX];
>-  int16_t                 synthesisBuffer[ANAL_BLOCKL_MAX];
>-  uint16_t                noiseSupFilter[HALF_ANAL_BLOCKL];
>-  uint16_t                overdrive; /* Q8 */
>-  uint16_t                denoiseBound; /* Q14 */
>-  const int16_t*          factor2Table;
>-  int16_t                 noiseEstLogQuantile[SIMULT* HALF_ANAL_BLOCKL];
>-  int16_t                 noiseEstDensity[SIMULT* HALF_ANAL_BLOCKL];
>-  int16_t                 noiseEstCounter[SIMULT];
>-  int16_t                 noiseEstQuantile[HALF_ANAL_BLOCKL];
>-
>-  size_t                  anaLen;
>-  size_t                  anaLen2;
>-  size_t                  magnLen;
>-  int                     aggrMode;
>-  int                     stages;
>-  int                     initFlag;
>-  int                     gainMap;
>-
>-  int32_t                 maxLrt;
>-  int32_t                 minLrt;
>+  uint32_t fs;
>+
>+  const int16_t* window;
>+  int16_t analysisBuffer[ANAL_BLOCKL_MAX];
>+  int16_t synthesisBuffer[ANAL_BLOCKL_MAX];
>+  uint16_t noiseSupFilter[HALF_ANAL_BLOCKL];
>+  uint16_t overdrive;    /* Q8 */
>+  uint16_t denoiseBound; /* Q14 */
>+  const int16_t* factor2Table;
>+  int16_t noiseEstLogQuantile[SIMULT * HALF_ANAL_BLOCKL];
>+  int16_t noiseEstDensity[SIMULT * HALF_ANAL_BLOCKL];
>+  int16_t noiseEstCounter[SIMULT];
>+  int16_t noiseEstQuantile[HALF_ANAL_BLOCKL];
>+
>+  size_t anaLen;
>+  size_t anaLen2;
>+  size_t magnLen;
>+  int aggrMode;
>+  int stages;
>+  int initFlag;
>+  int gainMap;
>+
>+  int32_t maxLrt;
>+  int32_t minLrt;
>   // Log LRT factor with time-smoothing in Q8.
>-  int32_t                 logLrtTimeAvgW32[HALF_ANAL_BLOCKL];
>-  int32_t                 featureLogLrt;
>-  int32_t                 thresholdLogLrt;
>-  int16_t                 weightLogLrt;
>+  int32_t logLrtTimeAvgW32[HALF_ANAL_BLOCKL];
>+  int32_t featureLogLrt;
>+  int32_t thresholdLogLrt;
>+  int16_t weightLogLrt;
> 
>-  uint32_t                featureSpecDiff;
>-  uint32_t                thresholdSpecDiff;
>-  int16_t                 weightSpecDiff;
>+  uint32_t featureSpecDiff;
>+  uint32_t thresholdSpecDiff;
>+  int16_t weightSpecDiff;
> 
>-  uint32_t                featureSpecFlat;
>-  uint32_t                thresholdSpecFlat;
>-  int16_t                 weightSpecFlat;
>+  uint32_t featureSpecFlat;
>+  uint32_t thresholdSpecFlat;
>+  int16_t weightSpecFlat;
> 
>   // Conservative estimate of noise spectrum.
>-  int32_t                 avgMagnPause[HALF_ANAL_BLOCKL];
>-  uint32_t                magnEnergy;
>-  uint32_t                sumMagn;
>-  uint32_t                curAvgMagnEnergy;
>-  uint32_t                timeAvgMagnEnergy;
>-  uint32_t                timeAvgMagnEnergyTmp;
>-
>-  uint32_t                whiteNoiseLevel;  // Initial noise estimate.
>+  int32_t avgMagnPause[HALF_ANAL_BLOCKL];
>+  uint32_t magnEnergy;
>+  uint32_t sumMagn;
>+  uint32_t curAvgMagnEnergy;
>+  uint32_t timeAvgMagnEnergy;
>+  uint32_t timeAvgMagnEnergyTmp;
>+
>+  uint32_t whiteNoiseLevel;  // Initial noise estimate.
>   // Initial magnitude spectrum estimate.
>-  uint32_t                initMagnEst[HALF_ANAL_BLOCKL];
>+  uint32_t initMagnEst[HALF_ANAL_BLOCKL];
>   // Pink noise parameters:
>-  int32_t                 pinkNoiseNumerator;  // Numerator.
>-  int32_t                 pinkNoiseExp;  // Power of freq.
>-  int                     minNorm;  // Smallest normalization factor.
>-  int                     zeroInputSignal;  // Zero input signal flag.
>+  int32_t pinkNoiseNumerator;  // Numerator.
>+  int32_t pinkNoiseExp;        // Power of freq.
>+  int minNorm;                 // Smallest normalization factor.
>+  int zeroInputSignal;         // Zero input signal flag.
> 
>   // Noise spectrum from previous frame.
>-  uint32_t                prevNoiseU32[HALF_ANAL_BLOCKL];
>+  uint32_t prevNoiseU32[HALF_ANAL_BLOCKL];
>   // Magnitude spectrum from previous frame.
>-  uint16_t                prevMagnU16[HALF_ANAL_BLOCKL];
>+  uint16_t prevMagnU16[HALF_ANAL_BLOCKL];
>   // Prior speech/noise probability in Q14.
>-  int16_t                 priorNonSpeechProb;
>+  int16_t priorNonSpeechProb;
> 
>-  int                     blockIndex;  // Frame index counter.
>+  int blockIndex;  // Frame index counter.
>   // Parameter for updating or estimating thresholds/weights for prior model.
>-  int                     modelUpdate;
>-  int                     cntThresUpdate;
>+  int modelUpdate;
>+  int cntThresUpdate;
> 
>   // Histograms for parameter estimation.
>-  int16_t                 histLrt[HIST_PAR_EST];
>-  int16_t                 histSpecFlat[HIST_PAR_EST];
>-  int16_t                 histSpecDiff[HIST_PAR_EST];
>+  int16_t histLrt[HIST_PAR_EST];
>+  int16_t histSpecFlat[HIST_PAR_EST];
>+  int16_t histSpecDiff[HIST_PAR_EST];
> 
>   // Quantities for high band estimate.
>-  int16_t                 dataBufHBFX[NUM_HIGH_BANDS_MAX][ANAL_BLOCKL_MAX];
>+  int16_t dataBufHBFX[NUM_HIGH_BANDS_MAX][ANAL_BLOCKL_MAX];
> 
>-  int                     qNoise;
>-  int                     prevQNoise;
>-  int                     prevQMagn;
>-  size_t                  blockLen10ms;
>+  int qNoise;
>+  int prevQNoise;
>+  int prevQMagn;
>+  size_t blockLen10ms;
> 
>-  int16_t                 real[ANAL_BLOCKL_MAX];
>-  int16_t                 imag[ANAL_BLOCKL_MAX];
>-  int32_t                 energyIn;
>-  int                     scaleEnergyIn;
>-  int                     normData;
>+  int16_t real[ANAL_BLOCKL_MAX];
>+  int16_t imag[ANAL_BLOCKL_MAX];
>+  int32_t energyIn;
>+  int scaleEnergyIn;
>+  int normData;
> 
>   struct RealFFT* real_fft;
> } NoiseSuppressionFixedC;
> 
> #ifdef __cplusplus
>-extern "C"
>-{
>+extern "C" {
> #endif
> 
> /****************************************************************************
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/ns/nsx_defines.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/ns/nsx_defines.h
>index 12869b3cf50..dc14dbc7933 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/ns/nsx_defines.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/ns/nsx_defines.h
>@@ -11,54 +11,64 @@
> #ifndef MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_NSX_DEFINES_H_
> #define MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_NSX_DEFINES_H_
> 
>-#define ANAL_BLOCKL_MAX         256 /* Max analysis block length */
>-#define HALF_ANAL_BLOCKL        129 /* Half max analysis block length + 1 */
>-#define NUM_HIGH_BANDS_MAX      2   /* Max number of high bands */
>-#define SIMULT                  3
>-#define END_STARTUP_LONG        200
>-#define END_STARTUP_SHORT       50
>-#define FACTOR_Q16              2621440 /* 40 in Q16 */
>-#define FACTOR_Q7               5120 /* 40 in Q7 */
>-#define FACTOR_Q7_STARTUP       1024 /* 8 in Q7 */
>-#define WIDTH_Q8                3 /* 0.01 in Q8 (or 25 ) */
>+#define ANAL_BLOCKL_MAX 256  /* Max analysis block length */
>+#define HALF_ANAL_BLOCKL 129 /* Half max analysis block length + 1 */
>+#define NUM_HIGH_BANDS_MAX 2 /* Max number of high bands */
>+#define SIMULT 3
>+#define END_STARTUP_LONG 200
>+#define END_STARTUP_SHORT 50
>+#define FACTOR_Q16 2621440     /* 40 in Q16 */
>+#define FACTOR_Q7 5120         /* 40 in Q7 */
>+#define FACTOR_Q7_STARTUP 1024 /* 8 in Q7 */
>+#define WIDTH_Q8 3             /* 0.01 in Q8 (or 25 ) */
> 
> /* PARAMETERS FOR NEW METHOD */
>-#define DD_PR_SNR_Q11           2007 /* ~= Q11(0.98) DD update of prior SNR */
>+#define DD_PR_SNR_Q11 2007         /* ~= Q11(0.98) DD update of prior SNR */
> #define ONE_MINUS_DD_PR_SNR_Q11 41 /* DD update of prior SNR */
>-#define SPECT_FLAT_TAVG_Q14     4915 /* (0.30) tavg parameter for spectral flatness measure */
>-#define SPECT_DIFF_TAVG_Q8      77 /* (0.30) tavg parameter for spectral flatness measure */
>-#define PRIOR_UPDATE_Q14        1638 /* Q14(0.1) Update parameter of prior model */
>-#define NOISE_UPDATE_Q8         26 /* 26 ~= Q8(0.1) Update parameter for noise */
>+#define SPECT_FLAT_TAVG_Q14 \
>+  4915 /* (0.30) tavg parameter for spectral flatness measure */
>+#define SPECT_DIFF_TAVG_Q8 \
>+  77 /* (0.30) tavg parameter for spectral flatness measure */
>+#define PRIOR_UPDATE_Q14 1638 /* Q14(0.1) Update parameter of prior model */
>+#define NOISE_UPDATE_Q8 26    /* 26 ~= Q8(0.1) Update parameter for noise */
> 
> /* Probability threshold for noise state in speech/noise likelihood. */
> #define ONE_MINUS_PROB_RANGE_Q8 205 /* 205 ~= Q8(0.8) */
>-#define HIST_PAR_EST            1000 /* Histogram size for estimation of parameters */
>+#define HIST_PAR_EST 1000 /* Histogram size for estimation of parameters */
> 
> /* FEATURE EXTRACTION CONFIG  */
> /* Bin size of histogram */
>-#define BIN_SIZE_LRT            10
>-/* Scale parameters: multiply dominant peaks of the histograms by scale factor to obtain. */
>+#define BIN_SIZE_LRT 10
>+/* Scale parameters: multiply dominant peaks of the histograms by scale factor
>+ * to obtain. */
> /* Thresholds for prior model */
>-#define FACTOR_1_LRT_DIFF       6 /* For LRT and spectral difference (5 times bigger) */
>-/* For spectral_flatness: used when noise is flatter than speech (10 times bigger). */
>-#define FACTOR_2_FLAT_Q10       922
>+#define FACTOR_1_LRT_DIFF \
>+  6 /* For LRT and spectral difference (5 times bigger) */
>+/* For spectral_flatness: used when noise is flatter than speech (10 times
>+ * bigger). */
>+#define FACTOR_2_FLAT_Q10 922
> /* Peak limit for spectral flatness (varies between 0 and 1) */
>-#define THRES_PEAK_FLAT         24 /* * 2 * BIN_SIZE_FLAT_FX */
>-/* Limit on spacing of two highest peaks in histogram: spacing determined by bin size. */
>-#define LIM_PEAK_SPACE_FLAT_DIFF    4 /* * 2 * BIN_SIZE_DIFF_FX */
>+#define THRES_PEAK_FLAT 24 /* * 2 * BIN_SIZE_FLAT_FX */
>+/* Limit on spacing of two highest peaks in histogram: spacing determined by bin
>+ * size. */
>+#define LIM_PEAK_SPACE_FLAT_DIFF 4 /* * 2 * BIN_SIZE_DIFF_FX */
> /* Limit on relevance of second peak */
>-#define LIM_PEAK_WEIGHT_FLAT_DIFF   2
>-#define THRES_FLUCT_LRT         10240 /* = 20 * inst->modelUpdate; fluctuation limit of LRT feat. */
>+#define LIM_PEAK_WEIGHT_FLAT_DIFF 2
>+#define THRES_FLUCT_LRT \
>+  10240 /* = 20 * inst->modelUpdate; fluctuation limit of LRT feat. */
> /* Limit on the max and min values for the feature thresholds */
>-#define MAX_FLAT_Q10            38912 /*  * 2 * BIN_SIZE_FLAT_FX */
>-#define MIN_FLAT_Q10            4096 /*  * 2 * BIN_SIZE_FLAT_FX */
>-#define MAX_DIFF                100 /* * 2 * BIN_SIZE_DIFF_FX */
>-#define MIN_DIFF                16 /* * 2 * BIN_SIZE_DIFF_FX */
>+#define MAX_FLAT_Q10 38912 /*  * 2 * BIN_SIZE_FLAT_FX */
>+#define MIN_FLAT_Q10 4096  /*  * 2 * BIN_SIZE_FLAT_FX */
>+#define MAX_DIFF 100       /* * 2 * BIN_SIZE_DIFF_FX */
>+#define MIN_DIFF 16        /* * 2 * BIN_SIZE_DIFF_FX */
> /* Criteria of weight of histogram peak  to accept/reject feature */
>-#define THRES_WEIGHT_FLAT_DIFF  154 /*(int)(0.3*(inst->modelUpdate)) for flatness and difference */
>+#define THRES_WEIGHT_FLAT_DIFF \
>+  154 /*(int)(0.3*(inst->modelUpdate)) for flatness and difference */
> 
>-#define STAT_UPDATES            9 /* Update every 512 = 1 << 9 block */
>-#define ONE_MINUS_GAMMA_PAUSE_Q8    13 /* ~= Q8(0.05) Update for conservative noise estimate */
>-#define GAMMA_NOISE_TRANS_AND_SPEECH_Q8 3 /* ~= Q8(0.01) Update for transition and noise region */
>+#define STAT_UPDATES 9 /* Update every 512 = 1 << 9 block */
>+#define ONE_MINUS_GAMMA_PAUSE_Q8 \
>+  13 /* ~= Q8(0.05) Update for conservative noise estimate */
>+#define GAMMA_NOISE_TRANS_AND_SPEECH_Q8 \
>+  3 /* ~= Q8(0.01) Update for transition and noise region */
> 
> #endif /* MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_NSX_DEFINES_H_ */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/ns/windows_private.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/ns/windows_private.h
>index 2ffd693d8d6..17792ecdfef 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/ns/windows_private.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/ns/windows_private.h
>@@ -13,562 +13,660 @@
> 
> // Hanning window for 4ms 16kHz
> static const float kHanning64w128[128] = {
>-  0.00000000000000f, 0.02454122852291f, 0.04906767432742f,
>-  0.07356456359967f, 0.09801714032956f, 0.12241067519922f,
>-  0.14673047445536f, 0.17096188876030f, 0.19509032201613f,
>-  0.21910124015687f, 0.24298017990326f, 0.26671275747490f,
>-  0.29028467725446f, 0.31368174039889f, 0.33688985339222f,
>-  0.35989503653499f, 0.38268343236509f, 0.40524131400499f,
>-  0.42755509343028f, 0.44961132965461f, 0.47139673682600f,
>-  0.49289819222978f, 0.51410274419322f, 0.53499761988710f,
>-  0.55557023301960f, 0.57580819141785f, 0.59569930449243f,
>-  0.61523159058063f, 0.63439328416365f, 0.65317284295378f,
>-  0.67155895484702f, 0.68954054473707f, 0.70710678118655f,
>-  0.72424708295147f, 0.74095112535496f, 0.75720884650648f,
>-  0.77301045336274f, 0.78834642762661f, 0.80320753148064f,
>-  0.81758481315158f, 0.83146961230255f, 0.84485356524971f,
>-  0.85772861000027f, 0.87008699110871f, 0.88192126434835f,
>-  0.89322430119552f, 0.90398929312344f, 0.91420975570353f,
>-  0.92387953251129f, 0.93299279883474f, 0.94154406518302f,
>-  0.94952818059304f, 0.95694033573221f, 0.96377606579544f,
>-  0.97003125319454f, 0.97570213003853f, 0.98078528040323f,
>-  0.98527764238894f, 0.98917650996478f, 0.99247953459871f,
>-  0.99518472667220f, 0.99729045667869f, 0.99879545620517f,
>-  0.99969881869620f, 1.00000000000000f,
>-  0.99969881869620f, 0.99879545620517f, 0.99729045667869f,
>-  0.99518472667220f, 0.99247953459871f, 0.98917650996478f,
>-  0.98527764238894f, 0.98078528040323f, 0.97570213003853f,
>-  0.97003125319454f, 0.96377606579544f, 0.95694033573221f,
>-  0.94952818059304f, 0.94154406518302f, 0.93299279883474f,
>-  0.92387953251129f, 0.91420975570353f, 0.90398929312344f,
>-  0.89322430119552f, 0.88192126434835f, 0.87008699110871f,
>-  0.85772861000027f, 0.84485356524971f, 0.83146961230255f,
>-  0.81758481315158f, 0.80320753148064f, 0.78834642762661f,
>-  0.77301045336274f, 0.75720884650648f, 0.74095112535496f,
>-  0.72424708295147f, 0.70710678118655f, 0.68954054473707f,
>-  0.67155895484702f, 0.65317284295378f, 0.63439328416365f,
>-  0.61523159058063f, 0.59569930449243f, 0.57580819141785f,
>-  0.55557023301960f, 0.53499761988710f, 0.51410274419322f,
>-  0.49289819222978f, 0.47139673682600f, 0.44961132965461f,
>-  0.42755509343028f, 0.40524131400499f, 0.38268343236509f,
>-  0.35989503653499f, 0.33688985339222f, 0.31368174039889f,
>-  0.29028467725446f, 0.26671275747490f, 0.24298017990326f,
>-  0.21910124015687f, 0.19509032201613f, 0.17096188876030f,
>-  0.14673047445536f, 0.12241067519922f, 0.09801714032956f,
>-  0.07356456359967f, 0.04906767432742f, 0.02454122852291f
>-};
>-
>-
>+    0.00000000000000f, 0.02454122852291f, 0.04906767432742f, 0.07356456359967f,
>+    0.09801714032956f, 0.12241067519922f, 0.14673047445536f, 0.17096188876030f,
>+    0.19509032201613f, 0.21910124015687f, 0.24298017990326f, 0.26671275747490f,
>+    0.29028467725446f, 0.31368174039889f, 0.33688985339222f, 0.35989503653499f,
>+    0.38268343236509f, 0.40524131400499f, 0.42755509343028f, 0.44961132965461f,
>+    0.47139673682600f, 0.49289819222978f, 0.51410274419322f, 0.53499761988710f,
>+    0.55557023301960f, 0.57580819141785f, 0.59569930449243f, 0.61523159058063f,
>+    0.63439328416365f, 0.65317284295378f, 0.67155895484702f, 0.68954054473707f,
>+    0.70710678118655f, 0.72424708295147f, 0.74095112535496f, 0.75720884650648f,
>+    0.77301045336274f, 0.78834642762661f, 0.80320753148064f, 0.81758481315158f,
>+    0.83146961230255f, 0.84485356524971f, 0.85772861000027f, 0.87008699110871f,
>+    0.88192126434835f, 0.89322430119552f, 0.90398929312344f, 0.91420975570353f,
>+    0.92387953251129f, 0.93299279883474f, 0.94154406518302f, 0.94952818059304f,
>+    0.95694033573221f, 0.96377606579544f, 0.97003125319454f, 0.97570213003853f,
>+    0.98078528040323f, 0.98527764238894f, 0.98917650996478f, 0.99247953459871f,
>+    0.99518472667220f, 0.99729045667869f, 0.99879545620517f, 0.99969881869620f,
>+    1.00000000000000f, 0.99969881869620f, 0.99879545620517f, 0.99729045667869f,
>+    0.99518472667220f, 0.99247953459871f, 0.98917650996478f, 0.98527764238894f,
>+    0.98078528040323f, 0.97570213003853f, 0.97003125319454f, 0.96377606579544f,
>+    0.95694033573221f, 0.94952818059304f, 0.94154406518302f, 0.93299279883474f,
>+    0.92387953251129f, 0.91420975570353f, 0.90398929312344f, 0.89322430119552f,
>+    0.88192126434835f, 0.87008699110871f, 0.85772861000027f, 0.84485356524971f,
>+    0.83146961230255f, 0.81758481315158f, 0.80320753148064f, 0.78834642762661f,
>+    0.77301045336274f, 0.75720884650648f, 0.74095112535496f, 0.72424708295147f,
>+    0.70710678118655f, 0.68954054473707f, 0.67155895484702f, 0.65317284295378f,
>+    0.63439328416365f, 0.61523159058063f, 0.59569930449243f, 0.57580819141785f,
>+    0.55557023301960f, 0.53499761988710f, 0.51410274419322f, 0.49289819222978f,
>+    0.47139673682600f, 0.44961132965461f, 0.42755509343028f, 0.40524131400499f,
>+    0.38268343236509f, 0.35989503653499f, 0.33688985339222f, 0.31368174039889f,
>+    0.29028467725446f, 0.26671275747490f, 0.24298017990326f, 0.21910124015687f,
>+    0.19509032201613f, 0.17096188876030f, 0.14673047445536f, 0.12241067519922f,
>+    0.09801714032956f, 0.07356456359967f, 0.04906767432742f, 0.02454122852291f};
> 
> // hybrib Hanning & flat window
> static const float kBlocks80w128[128] = {
>-  (float)0.00000000, (float)0.03271908, (float)0.06540313, (float)0.09801714, (float)0.13052619,
>-  (float)0.16289547, (float)0.19509032, (float)0.22707626, (float)0.25881905, (float)0.29028468,
>-  (float)0.32143947, (float)0.35225005, (float)0.38268343, (float)0.41270703, (float)0.44228869,
>-  (float)0.47139674, (float)0.50000000, (float)0.52806785, (float)0.55557023, (float)0.58247770,
>-  (float)0.60876143, (float)0.63439328, (float)0.65934582, (float)0.68359230, (float)0.70710678,
>-  (float)0.72986407, (float)0.75183981, (float)0.77301045, (float)0.79335334, (float)0.81284668,
>-  (float)0.83146961, (float)0.84920218, (float)0.86602540, (float)0.88192126, (float)0.89687274,
>-  (float)0.91086382, (float)0.92387953, (float)0.93590593, (float)0.94693013, (float)0.95694034,
>-  (float)0.96592583, (float)0.97387698, (float)0.98078528, (float)0.98664333, (float)0.99144486,
>-  (float)0.99518473, (float)0.99785892, (float)0.99946459, (float)1.00000000, (float)1.00000000,
>-  (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>-  (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>-  (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>-  (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>-  (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>-  (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>-  (float)1.00000000, (float)0.99946459, (float)0.99785892, (float)0.99518473, (float)0.99144486,
>-  (float)0.98664333, (float)0.98078528, (float)0.97387698, (float)0.96592583, (float)0.95694034,
>-  (float)0.94693013, (float)0.93590593, (float)0.92387953, (float)0.91086382, (float)0.89687274,
>-  (float)0.88192126, (float)0.86602540, (float)0.84920218, (float)0.83146961, (float)0.81284668,
>-  (float)0.79335334, (float)0.77301045, (float)0.75183981, (float)0.72986407, (float)0.70710678,
>-  (float)0.68359230, (float)0.65934582, (float)0.63439328, (float)0.60876143, (float)0.58247770,
>-  (float)0.55557023, (float)0.52806785, (float)0.50000000, (float)0.47139674, (float)0.44228869,
>-  (float)0.41270703, (float)0.38268343, (float)0.35225005, (float)0.32143947, (float)0.29028468,
>-  (float)0.25881905, (float)0.22707626, (float)0.19509032, (float)0.16289547, (float)0.13052619,
>-  (float)0.09801714, (float)0.06540313, (float)0.03271908
>-};
>+    (float)0.00000000, (float)0.03271908, (float)0.06540313, (float)0.09801714,
>+    (float)0.13052619, (float)0.16289547, (float)0.19509032, (float)0.22707626,
>+    (float)0.25881905, (float)0.29028468, (float)0.32143947, (float)0.35225005,
>+    (float)0.38268343, (float)0.41270703, (float)0.44228869, (float)0.47139674,
>+    (float)0.50000000, (float)0.52806785, (float)0.55557023, (float)0.58247770,
>+    (float)0.60876143, (float)0.63439328, (float)0.65934582, (float)0.68359230,
>+    (float)0.70710678, (float)0.72986407, (float)0.75183981, (float)0.77301045,
>+    (float)0.79335334, (float)0.81284668, (float)0.83146961, (float)0.84920218,
>+    (float)0.86602540, (float)0.88192126, (float)0.89687274, (float)0.91086382,
>+    (float)0.92387953, (float)0.93590593, (float)0.94693013, (float)0.95694034,
>+    (float)0.96592583, (float)0.97387698, (float)0.98078528, (float)0.98664333,
>+    (float)0.99144486, (float)0.99518473, (float)0.99785892, (float)0.99946459,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)0.99946459, (float)0.99785892, (float)0.99518473,
>+    (float)0.99144486, (float)0.98664333, (float)0.98078528, (float)0.97387698,
>+    (float)0.96592583, (float)0.95694034, (float)0.94693013, (float)0.93590593,
>+    (float)0.92387953, (float)0.91086382, (float)0.89687274, (float)0.88192126,
>+    (float)0.86602540, (float)0.84920218, (float)0.83146961, (float)0.81284668,
>+    (float)0.79335334, (float)0.77301045, (float)0.75183981, (float)0.72986407,
>+    (float)0.70710678, (float)0.68359230, (float)0.65934582, (float)0.63439328,
>+    (float)0.60876143, (float)0.58247770, (float)0.55557023, (float)0.52806785,
>+    (float)0.50000000, (float)0.47139674, (float)0.44228869, (float)0.41270703,
>+    (float)0.38268343, (float)0.35225005, (float)0.32143947, (float)0.29028468,
>+    (float)0.25881905, (float)0.22707626, (float)0.19509032, (float)0.16289547,
>+    (float)0.13052619, (float)0.09801714, (float)0.06540313, (float)0.03271908};
> 
> // hybrib Hanning & flat window
> static const float kBlocks160w256[256] = {
>-  (float)0.00000000, (float)0.01636173, (float)0.03271908, (float)0.04906767, (float)0.06540313,
>-  (float)0.08172107, (float)0.09801714, (float)0.11428696, (float)0.13052619, (float)0.14673047,
>-  (float)0.16289547, (float)0.17901686, (float)0.19509032, (float)0.21111155, (float)0.22707626,
>-  (float)0.24298018, (float)0.25881905, (float)0.27458862, (float)0.29028468, (float)0.30590302,
>-  (float)0.32143947, (float)0.33688985, (float)0.35225005, (float)0.36751594, (float)0.38268343,
>-  (float)0.39774847, (float)0.41270703, (float)0.42755509, (float)0.44228869, (float)0.45690388,
>-  (float)0.47139674, (float)0.48576339, (float)0.50000000, (float)0.51410274, (float)0.52806785,
>-  (float)0.54189158, (float)0.55557023, (float)0.56910015, (float)0.58247770, (float)0.59569930,
>-  (float)0.60876143, (float)0.62166057, (float)0.63439328, (float)0.64695615, (float)0.65934582,
>-  (float)0.67155895, (float)0.68359230, (float)0.69544264, (float)0.70710678, (float)0.71858162,
>-  (float)0.72986407, (float)0.74095113, (float)0.75183981, (float)0.76252720, (float)0.77301045,
>-  (float)0.78328675, (float)0.79335334, (float)0.80320753, (float)0.81284668, (float)0.82226822,
>-  (float)0.83146961, (float)0.84044840, (float)0.84920218, (float)0.85772861, (float)0.86602540,
>-  (float)0.87409034, (float)0.88192126, (float)0.88951608, (float)0.89687274, (float)0.90398929,
>-  (float)0.91086382, (float)0.91749450, (float)0.92387953, (float)0.93001722, (float)0.93590593,
>-  (float)0.94154407, (float)0.94693013, (float)0.95206268, (float)0.95694034, (float)0.96156180,
>-  (float)0.96592583, (float)0.97003125, (float)0.97387698, (float)0.97746197, (float)0.98078528,
>-  (float)0.98384601, (float)0.98664333, (float)0.98917651, (float)0.99144486, (float)0.99344778,
>-  (float)0.99518473, (float)0.99665524, (float)0.99785892, (float)0.99879546, (float)0.99946459,
>-  (float)0.99986614, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>-  (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>-  (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>-  (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>-  (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>-  (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>-  (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>-  (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>-  (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>-  (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>-  (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>-  (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>-  (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>-  (float)1.00000000, (float)0.99986614, (float)0.99946459, (float)0.99879546, (float)0.99785892,
>-  (float)0.99665524, (float)0.99518473, (float)0.99344778, (float)0.99144486, (float)0.98917651,
>-  (float)0.98664333, (float)0.98384601, (float)0.98078528, (float)0.97746197, (float)0.97387698,
>-  (float)0.97003125, (float)0.96592583, (float)0.96156180, (float)0.95694034, (float)0.95206268,
>-  (float)0.94693013, (float)0.94154407, (float)0.93590593, (float)0.93001722, (float)0.92387953,
>-  (float)0.91749450, (float)0.91086382, (float)0.90398929, (float)0.89687274, (float)0.88951608,
>-  (float)0.88192126, (float)0.87409034, (float)0.86602540, (float)0.85772861, (float)0.84920218,
>-  (float)0.84044840, (float)0.83146961, (float)0.82226822, (float)0.81284668, (float)0.80320753,
>-  (float)0.79335334, (float)0.78328675, (float)0.77301045, (float)0.76252720, (float)0.75183981,
>-  (float)0.74095113, (float)0.72986407, (float)0.71858162, (float)0.70710678, (float)0.69544264,
>-  (float)0.68359230, (float)0.67155895, (float)0.65934582, (float)0.64695615, (float)0.63439328,
>-  (float)0.62166057, (float)0.60876143, (float)0.59569930, (float)0.58247770, (float)0.56910015,
>-  (float)0.55557023, (float)0.54189158, (float)0.52806785, (float)0.51410274, (float)0.50000000,
>-  (float)0.48576339, (float)0.47139674, (float)0.45690388, (float)0.44228869, (float)0.42755509,
>-  (float)0.41270703, (float)0.39774847, (float)0.38268343, (float)0.36751594, (float)0.35225005,
>-  (float)0.33688985, (float)0.32143947, (float)0.30590302, (float)0.29028468, (float)0.27458862,
>-  (float)0.25881905, (float)0.24298018, (float)0.22707626, (float)0.21111155, (float)0.19509032,
>-  (float)0.17901686, (float)0.16289547, (float)0.14673047, (float)0.13052619, (float)0.11428696,
>-  (float)0.09801714, (float)0.08172107, (float)0.06540313, (float)0.04906767, (float)0.03271908,
>-  (float)0.01636173
>-};
>+    (float)0.00000000, (float)0.01636173, (float)0.03271908, (float)0.04906767,
>+    (float)0.06540313, (float)0.08172107, (float)0.09801714, (float)0.11428696,
>+    (float)0.13052619, (float)0.14673047, (float)0.16289547, (float)0.17901686,
>+    (float)0.19509032, (float)0.21111155, (float)0.22707626, (float)0.24298018,
>+    (float)0.25881905, (float)0.27458862, (float)0.29028468, (float)0.30590302,
>+    (float)0.32143947, (float)0.33688985, (float)0.35225005, (float)0.36751594,
>+    (float)0.38268343, (float)0.39774847, (float)0.41270703, (float)0.42755509,
>+    (float)0.44228869, (float)0.45690388, (float)0.47139674, (float)0.48576339,
>+    (float)0.50000000, (float)0.51410274, (float)0.52806785, (float)0.54189158,
>+    (float)0.55557023, (float)0.56910015, (float)0.58247770, (float)0.59569930,
>+    (float)0.60876143, (float)0.62166057, (float)0.63439328, (float)0.64695615,
>+    (float)0.65934582, (float)0.67155895, (float)0.68359230, (float)0.69544264,
>+    (float)0.70710678, (float)0.71858162, (float)0.72986407, (float)0.74095113,
>+    (float)0.75183981, (float)0.76252720, (float)0.77301045, (float)0.78328675,
>+    (float)0.79335334, (float)0.80320753, (float)0.81284668, (float)0.82226822,
>+    (float)0.83146961, (float)0.84044840, (float)0.84920218, (float)0.85772861,
>+    (float)0.86602540, (float)0.87409034, (float)0.88192126, (float)0.88951608,
>+    (float)0.89687274, (float)0.90398929, (float)0.91086382, (float)0.91749450,
>+    (float)0.92387953, (float)0.93001722, (float)0.93590593, (float)0.94154407,
>+    (float)0.94693013, (float)0.95206268, (float)0.95694034, (float)0.96156180,
>+    (float)0.96592583, (float)0.97003125, (float)0.97387698, (float)0.97746197,
>+    (float)0.98078528, (float)0.98384601, (float)0.98664333, (float)0.98917651,
>+    (float)0.99144486, (float)0.99344778, (float)0.99518473, (float)0.99665524,
>+    (float)0.99785892, (float)0.99879546, (float)0.99946459, (float)0.99986614,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)0.99986614, (float)0.99946459, (float)0.99879546,
>+    (float)0.99785892, (float)0.99665524, (float)0.99518473, (float)0.99344778,
>+    (float)0.99144486, (float)0.98917651, (float)0.98664333, (float)0.98384601,
>+    (float)0.98078528, (float)0.97746197, (float)0.97387698, (float)0.97003125,
>+    (float)0.96592583, (float)0.96156180, (float)0.95694034, (float)0.95206268,
>+    (float)0.94693013, (float)0.94154407, (float)0.93590593, (float)0.93001722,
>+    (float)0.92387953, (float)0.91749450, (float)0.91086382, (float)0.90398929,
>+    (float)0.89687274, (float)0.88951608, (float)0.88192126, (float)0.87409034,
>+    (float)0.86602540, (float)0.85772861, (float)0.84920218, (float)0.84044840,
>+    (float)0.83146961, (float)0.82226822, (float)0.81284668, (float)0.80320753,
>+    (float)0.79335334, (float)0.78328675, (float)0.77301045, (float)0.76252720,
>+    (float)0.75183981, (float)0.74095113, (float)0.72986407, (float)0.71858162,
>+    (float)0.70710678, (float)0.69544264, (float)0.68359230, (float)0.67155895,
>+    (float)0.65934582, (float)0.64695615, (float)0.63439328, (float)0.62166057,
>+    (float)0.60876143, (float)0.59569930, (float)0.58247770, (float)0.56910015,
>+    (float)0.55557023, (float)0.54189158, (float)0.52806785, (float)0.51410274,
>+    (float)0.50000000, (float)0.48576339, (float)0.47139674, (float)0.45690388,
>+    (float)0.44228869, (float)0.42755509, (float)0.41270703, (float)0.39774847,
>+    (float)0.38268343, (float)0.36751594, (float)0.35225005, (float)0.33688985,
>+    (float)0.32143947, (float)0.30590302, (float)0.29028468, (float)0.27458862,
>+    (float)0.25881905, (float)0.24298018, (float)0.22707626, (float)0.21111155,
>+    (float)0.19509032, (float)0.17901686, (float)0.16289547, (float)0.14673047,
>+    (float)0.13052619, (float)0.11428696, (float)0.09801714, (float)0.08172107,
>+    (float)0.06540313, (float)0.04906767, (float)0.03271908, (float)0.01636173};
> 
> // hybrib Hanning & flat window: for 20ms
> static const float kBlocks320w512[512] = {
>-  (float)0.00000000, (float)0.00818114, (float)0.01636173, (float)0.02454123, (float)0.03271908,
>-  (float)0.04089475, (float)0.04906767, (float)0.05723732, (float)0.06540313, (float)0.07356456,
>-  (float)0.08172107, (float)0.08987211, (float)0.09801714, (float)0.10615561, (float)0.11428696,
>-  (float)0.12241068, (float)0.13052619, (float)0.13863297, (float)0.14673047, (float)0.15481816,
>-  (float)0.16289547, (float)0.17096189, (float)0.17901686, (float)0.18705985, (float)0.19509032,
>-  (float)0.20310773, (float)0.21111155, (float)0.21910124, (float)0.22707626, (float)0.23503609,
>-  (float)0.24298018, (float)0.25090801, (float)0.25881905, (float)0.26671276, (float)0.27458862,
>-  (float)0.28244610, (float)0.29028468, (float)0.29810383, (float)0.30590302, (float)0.31368174,
>-  (float)0.32143947, (float)0.32917568, (float)0.33688985, (float)0.34458148, (float)0.35225005,
>-  (float)0.35989504, (float)0.36751594, (float)0.37511224, (float)0.38268343, (float)0.39022901,
>-  (float)0.39774847, (float)0.40524131, (float)0.41270703, (float)0.42014512, (float)0.42755509,
>-  (float)0.43493645, (float)0.44228869, (float)0.44961133, (float)0.45690388, (float)0.46416584,
>-  (float)0.47139674, (float)0.47859608, (float)0.48576339, (float)0.49289819, (float)0.50000000,
>-  (float)0.50706834, (float)0.51410274, (float)0.52110274, (float)0.52806785, (float)0.53499762,
>-  (float)0.54189158, (float)0.54874927, (float)0.55557023, (float)0.56235401, (float)0.56910015,
>-  (float)0.57580819, (float)0.58247770, (float)0.58910822, (float)0.59569930, (float)0.60225052,
>-  (float)0.60876143, (float)0.61523159, (float)0.62166057, (float)0.62804795, (float)0.63439328,
>-  (float)0.64069616, (float)0.64695615, (float)0.65317284, (float)0.65934582, (float)0.66547466,
>-  (float)0.67155895, (float)0.67759830, (float)0.68359230, (float)0.68954054, (float)0.69544264,
>-  (float)0.70129818, (float)0.70710678, (float)0.71286806, (float)0.71858162, (float)0.72424708,
>-  (float)0.72986407, (float)0.73543221, (float)0.74095113, (float)0.74642045, (float)0.75183981,
>-  (float)0.75720885, (float)0.76252720, (float)0.76779452, (float)0.77301045, (float)0.77817464,
>-  (float)0.78328675, (float)0.78834643, (float)0.79335334, (float)0.79830715, (float)0.80320753,
>-  (float)0.80805415, (float)0.81284668, (float)0.81758481, (float)0.82226822, (float)0.82689659,
>-  (float)0.83146961, (float)0.83598698, (float)0.84044840, (float)0.84485357, (float)0.84920218,
>-  (float)0.85349396, (float)0.85772861, (float)0.86190585, (float)0.86602540, (float)0.87008699,
>-  (float)0.87409034, (float)0.87803519, (float)0.88192126, (float)0.88574831, (float)0.88951608,
>-  (float)0.89322430, (float)0.89687274, (float)0.90046115, (float)0.90398929, (float)0.90745693,
>-  (float)0.91086382, (float)0.91420976, (float)0.91749450, (float)0.92071783, (float)0.92387953,
>-  (float)0.92697940, (float)0.93001722, (float)0.93299280, (float)0.93590593, (float)0.93875641,
>-  (float)0.94154407, (float)0.94426870, (float)0.94693013, (float)0.94952818, (float)0.95206268,
>-  (float)0.95453345, (float)0.95694034, (float)0.95928317, (float)0.96156180, (float)0.96377607,
>-  (float)0.96592583, (float)0.96801094, (float)0.97003125, (float)0.97198664, (float)0.97387698,
>-  (float)0.97570213, (float)0.97746197, (float)0.97915640, (float)0.98078528, (float)0.98234852,
>-  (float)0.98384601, (float)0.98527764, (float)0.98664333, (float)0.98794298, (float)0.98917651,
>-  (float)0.99034383, (float)0.99144486, (float)0.99247953, (float)0.99344778, (float)0.99434953,
>-  (float)0.99518473, (float)0.99595331, (float)0.99665524, (float)0.99729046, (float)0.99785892,
>-  (float)0.99836060, (float)0.99879546, (float)0.99916346, (float)0.99946459, (float)0.99969882,
>-  (float)0.99986614, (float)0.99996653, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>-  (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>-  (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>-  (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>-  (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>-  (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>-  (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>-  (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>-  (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>-  (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>-  (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>-  (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>-  (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>-  (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>-  (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>-  (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>-  (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>-  (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>-  (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>-  (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>-  (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>-  (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>-  (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>-  (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>-  (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>-  (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>-  (float)1.00000000, (float)0.99996653, (float)0.99986614, (float)0.99969882, (float)0.99946459,
>-  (float)0.99916346, (float)0.99879546, (float)0.99836060, (float)0.99785892, (float)0.99729046,
>-  (float)0.99665524, (float)0.99595331, (float)0.99518473, (float)0.99434953, (float)0.99344778,
>-  (float)0.99247953, (float)0.99144486, (float)0.99034383, (float)0.98917651, (float)0.98794298,
>-  (float)0.98664333, (float)0.98527764, (float)0.98384601, (float)0.98234852, (float)0.98078528,
>-  (float)0.97915640, (float)0.97746197, (float)0.97570213, (float)0.97387698, (float)0.97198664,
>-  (float)0.97003125, (float)0.96801094, (float)0.96592583, (float)0.96377607, (float)0.96156180,
>-  (float)0.95928317, (float)0.95694034, (float)0.95453345, (float)0.95206268, (float)0.94952818,
>-  (float)0.94693013, (float)0.94426870, (float)0.94154407, (float)0.93875641, (float)0.93590593,
>-  (float)0.93299280, (float)0.93001722, (float)0.92697940, (float)0.92387953, (float)0.92071783,
>-  (float)0.91749450, (float)0.91420976, (float)0.91086382, (float)0.90745693, (float)0.90398929,
>-  (float)0.90046115, (float)0.89687274, (float)0.89322430, (float)0.88951608, (float)0.88574831,
>-  (float)0.88192126, (float)0.87803519, (float)0.87409034, (float)0.87008699, (float)0.86602540,
>-  (float)0.86190585, (float)0.85772861, (float)0.85349396, (float)0.84920218, (float)0.84485357,
>-  (float)0.84044840, (float)0.83598698, (float)0.83146961, (float)0.82689659, (float)0.82226822,
>-  (float)0.81758481, (float)0.81284668, (float)0.80805415, (float)0.80320753, (float)0.79830715,
>-  (float)0.79335334, (float)0.78834643, (float)0.78328675, (float)0.77817464, (float)0.77301045,
>-  (float)0.76779452, (float)0.76252720, (float)0.75720885, (float)0.75183981, (float)0.74642045,
>-  (float)0.74095113, (float)0.73543221, (float)0.72986407, (float)0.72424708, (float)0.71858162,
>-  (float)0.71286806, (float)0.70710678, (float)0.70129818, (float)0.69544264, (float)0.68954054,
>-  (float)0.68359230, (float)0.67759830, (float)0.67155895, (float)0.66547466, (float)0.65934582,
>-  (float)0.65317284, (float)0.64695615, (float)0.64069616, (float)0.63439328, (float)0.62804795,
>-  (float)0.62166057, (float)0.61523159, (float)0.60876143, (float)0.60225052, (float)0.59569930,
>-  (float)0.58910822, (float)0.58247770, (float)0.57580819, (float)0.56910015, (float)0.56235401,
>-  (float)0.55557023, (float)0.54874927, (float)0.54189158, (float)0.53499762, (float)0.52806785,
>-  (float)0.52110274, (float)0.51410274, (float)0.50706834, (float)0.50000000, (float)0.49289819,
>-  (float)0.48576339, (float)0.47859608, (float)0.47139674, (float)0.46416584, (float)0.45690388,
>-  (float)0.44961133, (float)0.44228869, (float)0.43493645, (float)0.42755509, (float)0.42014512,
>-  (float)0.41270703, (float)0.40524131, (float)0.39774847, (float)0.39022901, (float)0.38268343,
>-  (float)0.37511224, (float)0.36751594, (float)0.35989504, (float)0.35225005, (float)0.34458148,
>-  (float)0.33688985, (float)0.32917568, (float)0.32143947, (float)0.31368174, (float)0.30590302,
>-  (float)0.29810383, (float)0.29028468, (float)0.28244610, (float)0.27458862, (float)0.26671276,
>-  (float)0.25881905, (float)0.25090801, (float)0.24298018, (float)0.23503609, (float)0.22707626,
>-  (float)0.21910124, (float)0.21111155, (float)0.20310773, (float)0.19509032, (float)0.18705985,
>-  (float)0.17901686, (float)0.17096189, (float)0.16289547, (float)0.15481816, (float)0.14673047,
>-  (float)0.13863297, (float)0.13052619, (float)0.12241068, (float)0.11428696, (float)0.10615561,
>-  (float)0.09801714, (float)0.08987211, (float)0.08172107, (float)0.07356456, (float)0.06540313,
>-  (float)0.05723732, (float)0.04906767, (float)0.04089475, (float)0.03271908, (float)0.02454123,
>-  (float)0.01636173, (float)0.00818114
>-};
>-
>+    (float)0.00000000, (float)0.00818114, (float)0.01636173, (float)0.02454123,
>+    (float)0.03271908, (float)0.04089475, (float)0.04906767, (float)0.05723732,
>+    (float)0.06540313, (float)0.07356456, (float)0.08172107, (float)0.08987211,
>+    (float)0.09801714, (float)0.10615561, (float)0.11428696, (float)0.12241068,
>+    (float)0.13052619, (float)0.13863297, (float)0.14673047, (float)0.15481816,
>+    (float)0.16289547, (float)0.17096189, (float)0.17901686, (float)0.18705985,
>+    (float)0.19509032, (float)0.20310773, (float)0.21111155, (float)0.21910124,
>+    (float)0.22707626, (float)0.23503609, (float)0.24298018, (float)0.25090801,
>+    (float)0.25881905, (float)0.26671276, (float)0.27458862, (float)0.28244610,
>+    (float)0.29028468, (float)0.29810383, (float)0.30590302, (float)0.31368174,
>+    (float)0.32143947, (float)0.32917568, (float)0.33688985, (float)0.34458148,
>+    (float)0.35225005, (float)0.35989504, (float)0.36751594, (float)0.37511224,
>+    (float)0.38268343, (float)0.39022901, (float)0.39774847, (float)0.40524131,
>+    (float)0.41270703, (float)0.42014512, (float)0.42755509, (float)0.43493645,
>+    (float)0.44228869, (float)0.44961133, (float)0.45690388, (float)0.46416584,
>+    (float)0.47139674, (float)0.47859608, (float)0.48576339, (float)0.49289819,
>+    (float)0.50000000, (float)0.50706834, (float)0.51410274, (float)0.52110274,
>+    (float)0.52806785, (float)0.53499762, (float)0.54189158, (float)0.54874927,
>+    (float)0.55557023, (float)0.56235401, (float)0.56910015, (float)0.57580819,
>+    (float)0.58247770, (float)0.58910822, (float)0.59569930, (float)0.60225052,
>+    (float)0.60876143, (float)0.61523159, (float)0.62166057, (float)0.62804795,
>+    (float)0.63439328, (float)0.64069616, (float)0.64695615, (float)0.65317284,
>+    (float)0.65934582, (float)0.66547466, (float)0.67155895, (float)0.67759830,
>+    (float)0.68359230, (float)0.68954054, (float)0.69544264, (float)0.70129818,
>+    (float)0.70710678, (float)0.71286806, (float)0.71858162, (float)0.72424708,
>+    (float)0.72986407, (float)0.73543221, (float)0.74095113, (float)0.74642045,
>+    (float)0.75183981, (float)0.75720885, (float)0.76252720, (float)0.76779452,
>+    (float)0.77301045, (float)0.77817464, (float)0.78328675, (float)0.78834643,
>+    (float)0.79335334, (float)0.79830715, (float)0.80320753, (float)0.80805415,
>+    (float)0.81284668, (float)0.81758481, (float)0.82226822, (float)0.82689659,
>+    (float)0.83146961, (float)0.83598698, (float)0.84044840, (float)0.84485357,
>+    (float)0.84920218, (float)0.85349396, (float)0.85772861, (float)0.86190585,
>+    (float)0.86602540, (float)0.87008699, (float)0.87409034, (float)0.87803519,
>+    (float)0.88192126, (float)0.88574831, (float)0.88951608, (float)0.89322430,
>+    (float)0.89687274, (float)0.90046115, (float)0.90398929, (float)0.90745693,
>+    (float)0.91086382, (float)0.91420976, (float)0.91749450, (float)0.92071783,
>+    (float)0.92387953, (float)0.92697940, (float)0.93001722, (float)0.93299280,
>+    (float)0.93590593, (float)0.93875641, (float)0.94154407, (float)0.94426870,
>+    (float)0.94693013, (float)0.94952818, (float)0.95206268, (float)0.95453345,
>+    (float)0.95694034, (float)0.95928317, (float)0.96156180, (float)0.96377607,
>+    (float)0.96592583, (float)0.96801094, (float)0.97003125, (float)0.97198664,
>+    (float)0.97387698, (float)0.97570213, (float)0.97746197, (float)0.97915640,
>+    (float)0.98078528, (float)0.98234852, (float)0.98384601, (float)0.98527764,
>+    (float)0.98664333, (float)0.98794298, (float)0.98917651, (float)0.99034383,
>+    (float)0.99144486, (float)0.99247953, (float)0.99344778, (float)0.99434953,
>+    (float)0.99518473, (float)0.99595331, (float)0.99665524, (float)0.99729046,
>+    (float)0.99785892, (float)0.99836060, (float)0.99879546, (float)0.99916346,
>+    (float)0.99946459, (float)0.99969882, (float)0.99986614, (float)0.99996653,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
>+    (float)1.00000000, (float)0.99996653, (float)0.99986614, (float)0.99969882,
>+    (float)0.99946459, (float)0.99916346, (float)0.99879546, (float)0.99836060,
>+    (float)0.99785892, (float)0.99729046, (float)0.99665524, (float)0.99595331,
>+    (float)0.99518473, (float)0.99434953, (float)0.99344778, (float)0.99247953,
>+    (float)0.99144486, (float)0.99034383, (float)0.98917651, (float)0.98794298,
>+    (float)0.98664333, (float)0.98527764, (float)0.98384601, (float)0.98234852,
>+    (float)0.98078528, (float)0.97915640, (float)0.97746197, (float)0.97570213,
>+    (float)0.97387698, (float)0.97198664, (float)0.97003125, (float)0.96801094,
>+    (float)0.96592583, (float)0.96377607, (float)0.96156180, (float)0.95928317,
>+    (float)0.95694034, (float)0.95453345, (float)0.95206268, (float)0.94952818,
>+    (float)0.94693013, (float)0.94426870, (float)0.94154407, (float)0.93875641,
>+    (float)0.93590593, (float)0.93299280, (float)0.93001722, (float)0.92697940,
>+    (float)0.92387953, (float)0.92071783, (float)0.91749450, (float)0.91420976,
>+    (float)0.91086382, (float)0.90745693, (float)0.90398929, (float)0.90046115,
>+    (float)0.89687274, (float)0.89322430, (float)0.88951608, (float)0.88574831,
>+    (float)0.88192126, (float)0.87803519, (float)0.87409034, (float)0.87008699,
>+    (float)0.86602540, (float)0.86190585, (float)0.85772861, (float)0.85349396,
>+    (float)0.84920218, (float)0.84485357, (float)0.84044840, (float)0.83598698,
>+    (float)0.83146961, (float)0.82689659, (float)0.82226822, (float)0.81758481,
>+    (float)0.81284668, (float)0.80805415, (float)0.80320753, (float)0.79830715,
>+    (float)0.79335334, (float)0.78834643, (float)0.78328675, (float)0.77817464,
>+    (float)0.77301045, (float)0.76779452, (float)0.76252720, (float)0.75720885,
>+    (float)0.75183981, (float)0.74642045, (float)0.74095113, (float)0.73543221,
>+    (float)0.72986407, (float)0.72424708, (float)0.71858162, (float)0.71286806,
>+    (float)0.70710678, (float)0.70129818, (float)0.69544264, (float)0.68954054,
>+    (float)0.68359230, (float)0.67759830, (float)0.67155895, (float)0.66547466,
>+    (float)0.65934582, (float)0.65317284, (float)0.64695615, (float)0.64069616,
>+    (float)0.63439328, (float)0.62804795, (float)0.62166057, (float)0.61523159,
>+    (float)0.60876143, (float)0.60225052, (float)0.59569930, (float)0.58910822,
>+    (float)0.58247770, (float)0.57580819, (float)0.56910015, (float)0.56235401,
>+    (float)0.55557023, (float)0.54874927, (float)0.54189158, (float)0.53499762,
>+    (float)0.52806785, (float)0.52110274, (float)0.51410274, (float)0.50706834,
>+    (float)0.50000000, (float)0.49289819, (float)0.48576339, (float)0.47859608,
>+    (float)0.47139674, (float)0.46416584, (float)0.45690388, (float)0.44961133,
>+    (float)0.44228869, (float)0.43493645, (float)0.42755509, (float)0.42014512,
>+    (float)0.41270703, (float)0.40524131, (float)0.39774847, (float)0.39022901,
>+    (float)0.38268343, (float)0.37511224, (float)0.36751594, (float)0.35989504,
>+    (float)0.35225005, (float)0.34458148, (float)0.33688985, (float)0.32917568,
>+    (float)0.32143947, (float)0.31368174, (float)0.30590302, (float)0.29810383,
>+    (float)0.29028468, (float)0.28244610, (float)0.27458862, (float)0.26671276,
>+    (float)0.25881905, (float)0.25090801, (float)0.24298018, (float)0.23503609,
>+    (float)0.22707626, (float)0.21910124, (float)0.21111155, (float)0.20310773,
>+    (float)0.19509032, (float)0.18705985, (float)0.17901686, (float)0.17096189,
>+    (float)0.16289547, (float)0.15481816, (float)0.14673047, (float)0.13863297,
>+    (float)0.13052619, (float)0.12241068, (float)0.11428696, (float)0.10615561,
>+    (float)0.09801714, (float)0.08987211, (float)0.08172107, (float)0.07356456,
>+    (float)0.06540313, (float)0.05723732, (float)0.04906767, (float)0.04089475,
>+    (float)0.03271908, (float)0.02454123, (float)0.01636173, (float)0.00818114};
> 
> // Hanning window: for 15ms at 16kHz with symmetric zeros
> static const float kBlocks240w512[512] = {
>-  (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
>-  (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
>-  (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
>-  (float)0.00000000, (float)0.00000000, (float)0.00654494, (float)0.01308960, (float)0.01963369,
>-  (float)0.02617695, (float)0.03271908, (float)0.03925982, (float)0.04579887, (float)0.05233596,
>-  (float)0.05887080, (float)0.06540313, (float)0.07193266, (float)0.07845910, (float)0.08498218,
>-  (float)0.09150162, (float)0.09801714, (float)0.10452846, (float)0.11103531, (float)0.11753740,
>-  (float)0.12403446, (float)0.13052620, (float)0.13701233, (float)0.14349262, (float)0.14996676,
>-  (float)0.15643448, (float)0.16289547, (float)0.16934951, (float)0.17579629, (float)0.18223552,
>-  (float)0.18866697, (float)0.19509032, (float)0.20150533, (float)0.20791170, (float)0.21430916,
>-  (float)0.22069745, (float)0.22707628, (float)0.23344538, (float)0.23980446, (float)0.24615330,
>-  (float)0.25249159, (float)0.25881904, (float)0.26513544, (float)0.27144045, (float)0.27773386,
>-  (float)0.28401536, (float)0.29028466, (float)0.29654160, (float)0.30278578, (float)0.30901700,
>-  (float)0.31523499, (float)0.32143945, (float)0.32763019, (float)0.33380687, (float)0.33996925,
>-  (float)0.34611708, (float)0.35225007, (float)0.35836795, (float)0.36447051, (float)0.37055743,
>-  (float)0.37662852, (float)0.38268346, (float)0.38872197, (float)0.39474389, (float)0.40074885,
>-  (float)0.40673664, (float)0.41270703, (float)0.41865975, (float)0.42459452, (float)0.43051112,
>-  (float)0.43640924, (float)0.44228873, (float)0.44814920, (float)0.45399052, (float)0.45981237,
>-  (float)0.46561453, (float)0.47139674, (float)0.47715878, (float)0.48290035, (float)0.48862126,
>-  (float)0.49432120, (float)0.50000000, (float)0.50565743, (float)0.51129311, (float)0.51690692,
>-  (float)0.52249855, (float)0.52806789, (float)0.53361452, (float)0.53913832, (float)0.54463905,
>-  (float)0.55011642, (float)0.55557024, (float)0.56100029, (float)0.56640625, (float)0.57178795,
>-  (float)0.57714522, (float)0.58247769, (float)0.58778524, (float)0.59306765, (float)0.59832460,
>-  (float)0.60355598, (float)0.60876143, (float)0.61394083, (float)0.61909395, (float)0.62422055,
>-  (float)0.62932038, (float)0.63439333, (float)0.63943899, (float)0.64445734, (float)0.64944810,
>-  (float)0.65441096, (float)0.65934587, (float)0.66425246, (float)0.66913062, (float)0.67398012,
>-  (float)0.67880076, (float)0.68359232, (float)0.68835455, (float)0.69308740, (float)0.69779050,
>-  (float)0.70246369, (float)0.70710677, (float)0.71171963, (float)0.71630198, (float)0.72085363,
>-  (float)0.72537440, (float)0.72986406, (float)0.73432255, (float)0.73874950, (float)0.74314487,
>-  (float)0.74750835, (float)0.75183982, (float)0.75613910, (float)0.76040596, (float)0.76464027,
>-  (float)0.76884186, (float)0.77301043, (float)0.77714598, (float)0.78124821, (float)0.78531694,
>-  (float)0.78935206, (float)0.79335338, (float)0.79732066, (float)0.80125386, (float)0.80515265,
>-  (float)0.80901700, (float)0.81284672, (float)0.81664157, (float)0.82040149, (float)0.82412618,
>-  (float)0.82781565, (float)0.83146966, (float)0.83508795, (float)0.83867061, (float)0.84221727,
>-  (float)0.84572780, (float)0.84920216, (float)0.85264021, (float)0.85604161, (float)0.85940641,
>-  (float)0.86273444, (float)0.86602545, (float)0.86927933, (float)0.87249607, (float)0.87567532,
>-  (float)0.87881714, (float)0.88192129, (float)0.88498765, (float)0.88801610, (float)0.89100653,
>-  (float)0.89395881, (float)0.89687276, (float)0.89974827, (float)0.90258533, (float)0.90538365,
>-  (float)0.90814316, (float)0.91086388, (float)0.91354549, (float)0.91618794, (float)0.91879123,
>-  (float)0.92135513, (float)0.92387950, (float)0.92636442, (float)0.92880958, (float)0.93121493,
>-  (float)0.93358046, (float)0.93590593, (float)0.93819135, (float)0.94043654, (float)0.94264150,
>-  (float)0.94480604, (float)0.94693011, (float)0.94901365, (float)0.95105654, (float)0.95305866,
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>+    (float)0.13052608, (float)0.12403426, (float)0.11753736, (float)0.11103519,
>+    (float)0.10452849, (float)0.09801710, (float)0.09150149, (float)0.08498220,
>+    (float)0.07845904, (float)0.07193252, (float)0.06540315, (float)0.05887074,
>+    (float)0.05233581, (float)0.04579888, (float)0.03925974, (float)0.03271893,
>+    (float)0.02617695, (float)0.01963361, (float)0.01308943, (float)0.00654493,
>+    (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
>+    (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
>+    (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
>+    (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000};
> 
> // Hanning window: for 30ms with 1024 fft with symmetric zeros at 16kHz
> static const float kBlocks480w1024[1024] = {
>-  (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
>-  (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
>-  (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
>-  (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
>-  (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
>-  (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
>-  (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00327249, (float)0.00654494,
>-  (float)0.00981732, (float)0.01308960, (float)0.01636173, (float)0.01963369, (float)0.02290544,
>-  (float)0.02617695, (float)0.02944817, (float)0.03271908, (float)0.03598964, (float)0.03925982,
>-  (float)0.04252957, (float)0.04579887, (float)0.04906768, (float)0.05233596, (float)0.05560368,
>-  (float)0.05887080, (float)0.06213730, (float)0.06540313, (float)0.06866825, (float)0.07193266,
>-  (float)0.07519628, (float)0.07845910, (float)0.08172107, (float)0.08498218, (float)0.08824237,
>-  (float)0.09150162, (float)0.09475989, (float)0.09801714, (float)0.10127335, (float)0.10452846,
>-  (float)0.10778246, (float)0.11103531, (float)0.11428697, (float)0.11753740, (float)0.12078657,
>-  (float)0.12403446, (float)0.12728101, (float)0.13052620, (float)0.13376999, (float)0.13701233,
>-  (float)0.14025325, (float)0.14349262, (float)0.14673047, (float)0.14996676, (float)0.15320145,
>-  (float)0.15643448, (float)0.15966582, (float)0.16289547, (float)0.16612339, (float)0.16934951,
>-  (float)0.17257382, (float)0.17579629, (float)0.17901687, (float)0.18223552, (float)0.18545224,
>-  (float)0.18866697, (float)0.19187967, (float)0.19509032, (float)0.19829889, (float)0.20150533,
>-  (float)0.20470962, (float)0.20791170, (float)0.21111156, (float)0.21430916, (float)0.21750447,
>-  (float)0.22069745, (float)0.22388805, (float)0.22707628, (float)0.23026206, (float)0.23344538,
>-  (float)0.23662618, (float)0.23980446, (float)0.24298020, (float)0.24615330, (float)0.24932377,
>-  (float)0.25249159, (float)0.25565669, (float)0.25881904, (float)0.26197866, (float)0.26513544,
>-  (float)0.26828939, (float)0.27144045, (float)0.27458861, (float)0.27773386, (float)0.28087610,
>-  (float)0.28401536, (float)0.28715158, (float)0.29028466, (float)0.29341471, (float)0.29654160,
>-  (float)0.29966527, (float)0.30278578, (float)0.30590302, (float)0.30901700, (float)0.31212768,
>-  (float)0.31523499, (float)0.31833893, (float)0.32143945, (float)0.32453656, (float)0.32763019,
>-  (float)0.33072028, (float)0.33380687, (float)0.33688986, (float)0.33996925, (float)0.34304500,
>-  (float)0.34611708, (float)0.34918544, (float)0.35225007, (float)0.35531089, (float)0.35836795,
>-  (float)0.36142117, (float)0.36447051, (float)0.36751595, (float)0.37055743, (float)0.37359497,
>-  (float)0.37662852, (float)0.37965801, (float)0.38268346, (float)0.38570479, (float)0.38872197,
>-  (float)0.39173502, (float)0.39474389, (float)0.39774847, (float)0.40074885, (float)0.40374491,
>-  (float)0.40673664, (float)0.40972406, (float)0.41270703, (float)0.41568562, (float)0.41865975,
>-  (float)0.42162940, (float)0.42459452, (float)0.42755508, (float)0.43051112, (float)0.43346250,
>-  (float)0.43640924, (float)0.43935132, (float)0.44228873, (float)0.44522133, (float)0.44814920,
>-  (float)0.45107228, (float)0.45399052, (float)0.45690390, (float)0.45981237, (float)0.46271592,
>-  (float)0.46561453, (float)0.46850815, (float)0.47139674, (float)0.47428030, (float)0.47715878,
>-  (float)0.48003215, (float)0.48290035, (float)0.48576337, (float)0.48862126, (float)0.49147385,
>-  (float)0.49432120, (float)0.49716330, (float)0.50000000, (float)0.50283140, (float)0.50565743,
>-  (float)0.50847799, (float)0.51129311, (float)0.51410276, (float)0.51690692, (float)0.51970553,
>-  (float)0.52249855, (float)0.52528602, (float)0.52806789, (float)0.53084403, (float)0.53361452,
>-  (float)0.53637928, (float)0.53913832, (float)0.54189163, (float)0.54463905, (float)0.54738063,
>-  (float)0.55011642, (float)0.55284631, (float)0.55557024, (float)0.55828828, (float)0.56100029,
>-  (float)0.56370628, (float)0.56640625, (float)0.56910014, (float)0.57178795, (float)0.57446963,
>-  (float)0.57714522, (float)0.57981455, (float)0.58247769, (float)0.58513463, (float)0.58778524,
>-  (float)0.59042960, (float)0.59306765, (float)0.59569931, (float)0.59832460, (float)0.60094351,
>-  (float)0.60355598, (float)0.60616195, (float)0.60876143, (float)0.61135441, (float)0.61394083,
>-  (float)0.61652070, (float)0.61909395, (float)0.62166059, (float)0.62422055, (float)0.62677383,
>-  (float)0.62932038, (float)0.63186020, (float)0.63439333, (float)0.63691956, (float)0.63943899,
>-  (float)0.64195162, (float)0.64445734, (float)0.64695615, (float)0.64944810, (float)0.65193301,
>-  (float)0.65441096, (float)0.65688187, (float)0.65934587, (float)0.66180271, (float)0.66425246,
>-  (float)0.66669512, (float)0.66913062, (float)0.67155898, (float)0.67398012, (float)0.67639405,
>-  (float)0.67880076, (float)0.68120021, (float)0.68359232, (float)0.68597710, (float)0.68835455,
>-  (float)0.69072467, (float)0.69308740, (float)0.69544262, (float)0.69779050, (float)0.70013082,
>-  (float)0.70246369, (float)0.70478904, (float)0.70710677, (float)0.70941699, (float)0.71171963,
>-  (float)0.71401459, (float)0.71630198, (float)0.71858168, (float)0.72085363, (float)0.72311789,
>-  (float)0.72537440, (float)0.72762316, (float)0.72986406, (float)0.73209721, (float)0.73432255,
>-  (float)0.73653996, (float)0.73874950, (float)0.74095118, (float)0.74314487, (float)0.74533057,
>-  (float)0.74750835, (float)0.74967808, (float)0.75183982, (float)0.75399351, (float)0.75613910,
>-  (float)0.75827658, (float)0.76040596, (float)0.76252723, (float)0.76464027, (float)0.76674515,
>-  (float)0.76884186, (float)0.77093029, (float)0.77301043, (float)0.77508241, (float)0.77714598,
>-  (float)0.77920127, (float)0.78124821, (float)0.78328675, (float)0.78531694, (float)0.78733873,
>-  (float)0.78935206, (float)0.79135692, (float)0.79335338, (float)0.79534125, (float)0.79732066,
>-  (float)0.79929149, (float)0.80125386, (float)0.80320752, (float)0.80515265, (float)0.80708915,
>-  (float)0.80901700, (float)0.81093621, (float)0.81284672, (float)0.81474853, (float)0.81664157,
>-  (float)0.81852591, (float)0.82040149, (float)0.82226825, (float)0.82412618, (float)0.82597536,
>-  (float)0.82781565, (float)0.82964706, (float)0.83146966, (float)0.83328325, (float)0.83508795,
>-  (float)0.83688378, (float)0.83867061, (float)0.84044838, (float)0.84221727, (float)0.84397703,
>-  (float)0.84572780, (float)0.84746957, (float)0.84920216, (float)0.85092574, (float)0.85264021,
>-  (float)0.85434544, (float)0.85604161, (float)0.85772866, (float)0.85940641, (float)0.86107504,
>-  (float)0.86273444, (float)0.86438453, (float)0.86602545, (float)0.86765707, (float)0.86927933,
>-  (float)0.87089235, (float)0.87249607, (float)0.87409031, (float)0.87567532, (float)0.87725097,
>-  (float)0.87881714, (float)0.88037390, (float)0.88192129, (float)0.88345921, (float)0.88498765,
>-  (float)0.88650668, (float)0.88801610, (float)0.88951612, (float)0.89100653, (float)0.89248741,
>-  (float)0.89395881, (float)0.89542055, (float)0.89687276, (float)0.89831537, (float)0.89974827,
>-  (float)0.90117162, (float)0.90258533, (float)0.90398932, (float)0.90538365, (float)0.90676826,
>-  (float)0.90814316, (float)0.90950841, (float)0.91086388, (float)0.91220951, (float)0.91354549,
>-  (float)0.91487163, (float)0.91618794, (float)0.91749454, (float)0.91879123, (float)0.92007810,
>-  (float)0.92135513, (float)0.92262226, (float)0.92387950, (float)0.92512691, (float)0.92636442,
>-  (float)0.92759192, (float)0.92880958, (float)0.93001723, (float)0.93121493, (float)0.93240267,
>-  (float)0.93358046, (float)0.93474817, (float)0.93590593, (float)0.93705362, (float)0.93819135,
>-  (float)0.93931901, (float)0.94043654, (float)0.94154406, (float)0.94264150, (float)0.94372880,
>-  (float)0.94480604, (float)0.94587320, (float)0.94693011, (float)0.94797695, (float)0.94901365,
>-  (float)0.95004016, (float)0.95105654, (float)0.95206273, (float)0.95305866, (float)0.95404440,
>-  (float)0.95501995, (float)0.95598525, (float)0.95694035, (float)0.95788521, (float)0.95881975,
>-  (float)0.95974404, (float)0.96065807, (float)0.96156180, (float)0.96245527, (float)0.96333838,
>-  (float)0.96421117, (float)0.96507370, (float)0.96592581, (float)0.96676767, (float)0.96759909,
>-  (float)0.96842021, (float)0.96923089, (float)0.97003126, (float)0.97082120, (float)0.97160077,
>-  (float)0.97236991, (float)0.97312868, (float)0.97387701, (float)0.97461486, (float)0.97534233,
>-  (float)0.97605932, (float)0.97676587, (float)0.97746199, (float)0.97814763, (float)0.97882277,
>-  (float)0.97948742, (float)0.98014158, (float)0.98078531, (float)0.98141843, (float)0.98204112,
>-  (float)0.98265332, (float)0.98325491, (float)0.98384601, (float)0.98442656, (float)0.98499662,
>-  (float)0.98555607, (float)0.98610497, (float)0.98664331, (float)0.98717111, (float)0.98768836,
>-  (float)0.98819500, (float)0.98869103, (float)0.98917651, (float)0.98965138, (float)0.99011570,
>-  (float)0.99056935, (float)0.99101239, (float)0.99144489, (float)0.99186671, (float)0.99227792,
>-  (float)0.99267852, (float)0.99306846, (float)0.99344778, (float)0.99381649, (float)0.99417448,
>-  (float)0.99452192, (float)0.99485862, (float)0.99518472, (float)0.99550015, (float)0.99580491,
>-  (float)0.99609905, (float)0.99638247, (float)0.99665523, (float)0.99691731, (float)0.99716878,
>-  (float)0.99740952, (float)0.99763954, (float)0.99785894, (float)0.99806762, (float)0.99826562,
>-  (float)0.99845290, (float)0.99862951, (float)0.99879545, (float)0.99895066, (float)0.99909520,
>-  (float)0.99922901, (float)0.99935216, (float)0.99946457, (float)0.99956632, (float)0.99965733,
>-  (float)0.99973762, (float)0.99980724, (float)0.99986613, (float)0.99991435, (float)0.99995178,
>-  (float)0.99997860, (float)0.99999464, (float)1.00000000, (float)0.99999464, (float)0.99997860,
>-  (float)0.99995178, (float)0.99991435, (float)0.99986613, (float)0.99980724, (float)0.99973762,
>-  (float)0.99965733, (float)0.99956632, (float)0.99946457, (float)0.99935216, (float)0.99922901,
>-  (float)0.99909520, (float)0.99895066, (float)0.99879545, (float)0.99862951, (float)0.99845290,
>-  (float)0.99826562, (float)0.99806762, (float)0.99785894, (float)0.99763954, (float)0.99740946,
>-  (float)0.99716872, (float)0.99691731, (float)0.99665523, (float)0.99638247, (float)0.99609905,
>-  (float)0.99580491, (float)0.99550015, (float)0.99518472, (float)0.99485862, (float)0.99452192,
>-  (float)0.99417448, (float)0.99381644, (float)0.99344778, (float)0.99306846, (float)0.99267852,
>-  (float)0.99227792, (float)0.99186671, (float)0.99144489, (float)0.99101239, (float)0.99056935,
>-  (float)0.99011564, (float)0.98965138, (float)0.98917651, (float)0.98869103, (float)0.98819494,
>-  (float)0.98768836, (float)0.98717111, (float)0.98664331, (float)0.98610497, (float)0.98555607,
>-  (float)0.98499656, (float)0.98442656, (float)0.98384601, (float)0.98325491, (float)0.98265326,
>-  (float)0.98204112, (float)0.98141843, (float)0.98078525, (float)0.98014158, (float)0.97948742,
>-  (float)0.97882277, (float)0.97814757, (float)0.97746193, (float)0.97676587, (float)0.97605932,
>-  (float)0.97534227, (float)0.97461486, (float)0.97387695, (float)0.97312862, (float)0.97236991,
>-  (float)0.97160077, (float)0.97082120, (float)0.97003126, (float)0.96923089, (float)0.96842015,
>-  (float)0.96759909, (float)0.96676761, (float)0.96592581, (float)0.96507365, (float)0.96421117,
>-  (float)0.96333838, (float)0.96245521, (float)0.96156180, (float)0.96065807, (float)0.95974404,
>-  (float)0.95881969, (float)0.95788515, (float)0.95694029, (float)0.95598525, (float)0.95501995,
>-  (float)0.95404440, (float)0.95305860, (float)0.95206267, (float)0.95105648, (float)0.95004016,
>-  (float)0.94901365, (float)0.94797695, (float)0.94693011, (float)0.94587314, (float)0.94480604,
>-  (float)0.94372880, (float)0.94264150, (float)0.94154406, (float)0.94043654, (float)0.93931895,
>-  (float)0.93819129, (float)0.93705362, (float)0.93590593, (float)0.93474817, (float)0.93358046,
>-  (float)0.93240267, (float)0.93121493, (float)0.93001723, (float)0.92880952, (float)0.92759192,
>-  (float)0.92636436, (float)0.92512691, (float)0.92387950, (float)0.92262226, (float)0.92135507,
>-  (float)0.92007804, (float)0.91879123, (float)0.91749448, (float)0.91618794, (float)0.91487157,
>-  (float)0.91354543, (float)0.91220951, (float)0.91086382, (float)0.90950835, (float)0.90814310,
>-  (float)0.90676820, (float)0.90538365, (float)0.90398932, (float)0.90258527, (float)0.90117157,
>-  (float)0.89974827, (float)0.89831525, (float)0.89687276, (float)0.89542055, (float)0.89395875,
>-  (float)0.89248741, (float)0.89100647, (float)0.88951600, (float)0.88801610, (float)0.88650662,
>-  (float)0.88498759, (float)0.88345915, (float)0.88192123, (float)0.88037384, (float)0.87881714,
>-  (float)0.87725091, (float)0.87567532, (float)0.87409031, (float)0.87249595, (float)0.87089223,
>-  (float)0.86927933, (float)0.86765701, (float)0.86602539, (float)0.86438447, (float)0.86273432,
>-  (float)0.86107504, (float)0.85940641, (float)0.85772860, (float)0.85604161, (float)0.85434544,
>-  (float)0.85264009, (float)0.85092574, (float)0.84920216, (float)0.84746951, (float)0.84572780,
>-  (float)0.84397697, (float)0.84221715, (float)0.84044844, (float)0.83867055, (float)0.83688372,
>-  (float)0.83508795, (float)0.83328319, (float)0.83146954, (float)0.82964706, (float)0.82781565,
>-  (float)0.82597530, (float)0.82412612, (float)0.82226813, (float)0.82040137, (float)0.81852591,
>-  (float)0.81664157, (float)0.81474847, (float)0.81284660, (float)0.81093609, (float)0.80901700,
>-  (float)0.80708915, (float)0.80515265, (float)0.80320752, (float)0.80125374, (float)0.79929143,
>-  (float)0.79732066, (float)0.79534125, (float)0.79335332, (float)0.79135686, (float)0.78935200,
>-  (float)0.78733861, (float)0.78531694, (float)0.78328675, (float)0.78124815, (float)0.77920121,
>-  (float)0.77714586, (float)0.77508223, (float)0.77301049, (float)0.77093029, (float)0.76884180,
>-  (float)0.76674509, (float)0.76464021, (float)0.76252711, (float)0.76040596, (float)0.75827658,
>-  (float)0.75613904, (float)0.75399339, (float)0.75183970, (float)0.74967796, (float)0.74750835,
>-  (float)0.74533057, (float)0.74314481, (float)0.74095106, (float)0.73874938, (float)0.73653996,
>-  (float)0.73432249, (float)0.73209721, (float)0.72986400, (float)0.72762305, (float)0.72537428,
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>+    (float)1.00000000, (float)0.99999464, (float)0.99997860, (float)0.99995178,
>+    (float)0.99991435, (float)0.99986613, (float)0.99980724, (float)0.99973762,
>+    (float)0.99965733, (float)0.99956632, (float)0.99946457, (float)0.99935216,
>+    (float)0.99922901, (float)0.99909520, (float)0.99895066, (float)0.99879545,
>+    (float)0.99862951, (float)0.99845290, (float)0.99826562, (float)0.99806762,
>+    (float)0.99785894, (float)0.99763954, (float)0.99740946, (float)0.99716872,
>+    (float)0.99691731, (float)0.99665523, (float)0.99638247, (float)0.99609905,
>+    (float)0.99580491, (float)0.99550015, (float)0.99518472, (float)0.99485862,
>+    (float)0.99452192, (float)0.99417448, (float)0.99381644, (float)0.99344778,
>+    (float)0.99306846, (float)0.99267852, (float)0.99227792, (float)0.99186671,
>+    (float)0.99144489, (float)0.99101239, (float)0.99056935, (float)0.99011564,
>+    (float)0.98965138, (float)0.98917651, (float)0.98869103, (float)0.98819494,
>+    (float)0.98768836, (float)0.98717111, (float)0.98664331, (float)0.98610497,
>+    (float)0.98555607, (float)0.98499656, (float)0.98442656, (float)0.98384601,
>+    (float)0.98325491, (float)0.98265326, (float)0.98204112, (float)0.98141843,
>+    (float)0.98078525, (float)0.98014158, (float)0.97948742, (float)0.97882277,
>+    (float)0.97814757, (float)0.97746193, (float)0.97676587, (float)0.97605932,
>+    (float)0.97534227, (float)0.97461486, (float)0.97387695, (float)0.97312862,
>+    (float)0.97236991, (float)0.97160077, (float)0.97082120, (float)0.97003126,
>+    (float)0.96923089, (float)0.96842015, (float)0.96759909, (float)0.96676761,
>+    (float)0.96592581, (float)0.96507365, (float)0.96421117, (float)0.96333838,
>+    (float)0.96245521, (float)0.96156180, (float)0.96065807, (float)0.95974404,
>+    (float)0.95881969, (float)0.95788515, (float)0.95694029, (float)0.95598525,
>+    (float)0.95501995, (float)0.95404440, (float)0.95305860, (float)0.95206267,
>+    (float)0.95105648, (float)0.95004016, (float)0.94901365, (float)0.94797695,
>+    (float)0.94693011, (float)0.94587314, (float)0.94480604, (float)0.94372880,
>+    (float)0.94264150, (float)0.94154406, (float)0.94043654, (float)0.93931895,
>+    (float)0.93819129, (float)0.93705362, (float)0.93590593, (float)0.93474817,
>+    (float)0.93358046, (float)0.93240267, (float)0.93121493, (float)0.93001723,
>+    (float)0.92880952, (float)0.92759192, (float)0.92636436, (float)0.92512691,
>+    (float)0.92387950, (float)0.92262226, (float)0.92135507, (float)0.92007804,
>+    (float)0.91879123, (float)0.91749448, (float)0.91618794, (float)0.91487157,
>+    (float)0.91354543, (float)0.91220951, (float)0.91086382, (float)0.90950835,
>+    (float)0.90814310, (float)0.90676820, (float)0.90538365, (float)0.90398932,
>+    (float)0.90258527, (float)0.90117157, (float)0.89974827, (float)0.89831525,
>+    (float)0.89687276, (float)0.89542055, (float)0.89395875, (float)0.89248741,
>+    (float)0.89100647, (float)0.88951600, (float)0.88801610, (float)0.88650662,
>+    (float)0.88498759, (float)0.88345915, (float)0.88192123, (float)0.88037384,
>+    (float)0.87881714, (float)0.87725091, (float)0.87567532, (float)0.87409031,
>+    (float)0.87249595, (float)0.87089223, (float)0.86927933, (float)0.86765701,
>+    (float)0.86602539, (float)0.86438447, (float)0.86273432, (float)0.86107504,
>+    (float)0.85940641, (float)0.85772860, (float)0.85604161, (float)0.85434544,
>+    (float)0.85264009, (float)0.85092574, (float)0.84920216, (float)0.84746951,
>+    (float)0.84572780, (float)0.84397697, (float)0.84221715, (float)0.84044844,
>+    (float)0.83867055, (float)0.83688372, (float)0.83508795, (float)0.83328319,
>+    (float)0.83146954, (float)0.82964706, (float)0.82781565, (float)0.82597530,
>+    (float)0.82412612, (float)0.82226813, (float)0.82040137, (float)0.81852591,
>+    (float)0.81664157, (float)0.81474847, (float)0.81284660, (float)0.81093609,
>+    (float)0.80901700, (float)0.80708915, (float)0.80515265, (float)0.80320752,
>+    (float)0.80125374, (float)0.79929143, (float)0.79732066, (float)0.79534125,
>+    (float)0.79335332, (float)0.79135686, (float)0.78935200, (float)0.78733861,
>+    (float)0.78531694, (float)0.78328675, (float)0.78124815, (float)0.77920121,
>+    (float)0.77714586, (float)0.77508223, (float)0.77301049, (float)0.77093029,
>+    (float)0.76884180, (float)0.76674509, (float)0.76464021, (float)0.76252711,
>+    (float)0.76040596, (float)0.75827658, (float)0.75613904, (float)0.75399339,
>+    (float)0.75183970, (float)0.74967796, (float)0.74750835, (float)0.74533057,
>+    (float)0.74314481, (float)0.74095106, (float)0.73874938, (float)0.73653996,
>+    (float)0.73432249, (float)0.73209721, (float)0.72986400, (float)0.72762305,
>+    (float)0.72537428, (float)0.72311789, (float)0.72085363, (float)0.71858162,
>+    (float)0.71630186, (float)0.71401453, (float)0.71171951, (float)0.70941705,
>+    (float)0.70710677, (float)0.70478898, (float)0.70246363, (float)0.70013070,
>+    (float)0.69779032, (float)0.69544268, (float)0.69308734, (float)0.69072461,
>+    (float)0.68835449, (float)0.68597704, (float)0.68359220, (float)0.68120021,
>+    (float)0.67880070, (float)0.67639399, (float)0.67398006, (float)0.67155886,
>+    (float)0.66913044, (float)0.66669512, (float)0.66425240, (float)0.66180259,
>+    (float)0.65934575, (float)0.65688181, (float)0.65441096, (float)0.65193301,
>+    (float)0.64944804, (float)0.64695609, (float)0.64445722, (float)0.64195150,
>+    (float)0.63943905, (float)0.63691956, (float)0.63439327, (float)0.63186014,
>+    (float)0.62932026, (float)0.62677372, (float)0.62422055, (float)0.62166059,
>+    (float)0.61909389, (float)0.61652064, (float)0.61394072, (float)0.61135429,
>+    (float)0.60876143, (float)0.60616189, (float)0.60355592, (float)0.60094339,
>+    (float)0.59832448, (float)0.59569913, (float)0.59306765, (float)0.59042960,
>+    (float)0.58778518, (float)0.58513451, (float)0.58247757, (float)0.57981461,
>+    (float)0.57714522, (float)0.57446963, (float)0.57178789, (float)0.56910002,
>+    (float)0.56640613, (float)0.56370628, (float)0.56100023, (float)0.55828822,
>+    (float)0.55557019, (float)0.55284619, (float)0.55011630, (float)0.54738069,
>+    (float)0.54463905, (float)0.54189152, (float)0.53913826, (float)0.53637916,
>+    (float)0.53361434, (float)0.53084403, (float)0.52806783, (float)0.52528596,
>+    (float)0.52249849, (float)0.51970541, (float)0.51690674, (float)0.51410276,
>+    (float)0.51129305, (float)0.50847787, (float)0.50565726, (float)0.50283122,
>+    (float)0.50000006, (float)0.49716327, (float)0.49432117, (float)0.49147379,
>+    (float)0.48862115, (float)0.48576325, (float)0.48290038, (float)0.48003212,
>+    (float)0.47715873, (float)0.47428021, (float)0.47139663, (float)0.46850798,
>+    (float)0.46561456, (float)0.46271589, (float)0.45981231, (float)0.45690379,
>+    (float)0.45399037, (float)0.45107210, (float)0.44814920, (float)0.44522130,
>+    (float)0.44228864, (float)0.43935123, (float)0.43640912, (float)0.43346232,
>+    (float)0.43051112, (float)0.42755505, (float)0.42459446, (float)0.42162928,
>+    (float)0.41865960, (float)0.41568545, (float)0.41270703, (float)0.40972400,
>+    (float)0.40673658, (float)0.40374479, (float)0.40074870, (float)0.39774850,
>+    (float)0.39474386, (float)0.39173496, (float)0.38872188, (float)0.38570464,
>+    (float)0.38268328, (float)0.37965804, (float)0.37662849, (float)0.37359491,
>+    (float)0.37055734, (float)0.36751580, (float)0.36447033, (float)0.36142117,
>+    (float)0.35836792, (float)0.35531086, (float)0.35224995, (float)0.34918529,
>+    (float)0.34611690, (float)0.34304500, (float)0.33996922, (float)0.33688980,
>+    (float)0.33380675, (float)0.33072016, (float)0.32763001, (float)0.32453656,
>+    (float)0.32143945, (float)0.31833887, (float)0.31523487, (float)0.31212750,
>+    (float)0.30901679, (float)0.30590302, (float)0.30278572, (float)0.29966521,
>+    (float)0.29654145, (float)0.29341453, (float)0.29028472, (float)0.28715155,
>+    (float)0.28401530, (float)0.28087601, (float)0.27773371, (float)0.27458847,
>+    (float)0.27144048, (float)0.26828936, (float)0.26513538, (float)0.26197854,
>+    (float)0.25881892, (float)0.25565651, (float)0.25249159, (float)0.24932374,
>+    (float)0.24615324, (float)0.24298008, (float)0.23980433, (float)0.23662600,
>+    (float)0.23344538, (float)0.23026201, (float)0.22707619, (float)0.22388794,
>+    (float)0.22069728, (float)0.21750426, (float)0.21430916, (float)0.21111152,
>+    (float)0.20791161, (float)0.20470949, (float)0.20150517, (float)0.19829892,
>+    (float)0.19509031, (float)0.19187963, (float)0.18866688, (float)0.18545210,
>+    (float)0.18223536, (float)0.17901689, (float)0.17579627, (float)0.17257376,
>+    (float)0.16934940, (float)0.16612324, (float)0.16289529, (float)0.15966584,
>+    (float)0.15643445, (float)0.15320137, (float)0.14996666, (float)0.14673033,
>+    (float)0.14349243, (float)0.14025325, (float)0.13701232, (float)0.13376991,
>+    (float)0.13052608, (float)0.12728085, (float)0.12403426, (float)0.12078657,
>+    (float)0.11753736, (float)0.11428688, (float)0.11103519, (float)0.10778230,
>+    (float)0.10452849, (float)0.10127334, (float)0.09801710, (float)0.09475980,
>+    (float)0.09150149, (float)0.08824220, (float)0.08498220, (float)0.08172106,
>+    (float)0.07845904, (float)0.07519618, (float)0.07193252, (float)0.06866808,
>+    (float)0.06540315, (float)0.06213728, (float)0.05887074, (float)0.05560357,
>+    (float)0.05233581, (float)0.04906749, (float)0.04579888, (float)0.04252954,
>+    (float)0.03925974, (float)0.03598953, (float)0.03271893, (float)0.02944798,
>+    (float)0.02617695, (float)0.02290541, (float)0.01963361, (float)0.01636161,
>+    (float)0.01308943, (float)0.00981712, (float)0.00654493, (float)0.00327244,
>+    (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
>+    (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
>+    (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
>+    (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
>+    (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
>+    (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
>+    (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
>+    (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000};
> 
> #endif  // MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_WINDOWS_PRIVATE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/residual_echo_detector.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/residual_echo_detector.cc
>index ef325a032ba..e80501358ea 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/residual_echo_detector.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/residual_echo_detector.cc
>@@ -90,7 +90,7 @@ void ResidualEchoDetector::AnalyzeCaptureAudio(
>   }
> 
>   // Get the next render value.
>-  const rtc::Optional<float> buffered_render_power = render_buffer_.Pop();
>+  const absl::optional<float> buffered_render_power = render_buffer_.Pop();
>   if (!buffered_render_power) {
>     // This can happen in a few cases: at the start of a call, due to a glitch
>     // or due to clock drift. The excess capture value will be ignored.
>@@ -142,8 +142,9 @@ void ResidualEchoDetector::AnalyzeCaptureAudio(
>       }
>       RTC_DCHECK_LT(read_index, render_power_.size());
>       RTC_LOG_F(LS_ERROR) << "Echo detector internal state: {"
>-                          << "Echo likelihood: " << echo_likelihood_
>-                          << ", Best Delay: " << best_delay << ", Covariance: "
>+                             "Echo likelihood: "
>+                          << echo_likelihood_ << ", Best Delay: " << best_delay
>+                          << ", Covariance: "
>                           << covariances_[best_delay].covariance()
>                           << ", Last capture power: " << capture_power
>                           << ", Capture mean: " << capture_mean
>@@ -177,8 +178,10 @@ void ResidualEchoDetector::AnalyzeCaptureAudio(
>                               : 0;
> }
> 
>-void ResidualEchoDetector::Initialize(int /*sample_rate_hz*/,
>-                                      int /*num_channels*/) {
>+void ResidualEchoDetector::Initialize(int /*capture_sample_rate_hz*/,
>+                                      int /*num_capture_channels*/,
>+                                      int /*render_sample_rate_hz*/,
>+                                      int /*num_render_channels*/) {
>   render_buffer_.Clear();
>   std::fill(render_power_.begin(), render_power_.end(), 0.f);
>   std::fill(render_power_mean_.begin(), render_power_mean_.end(), 0.f);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/residual_echo_detector.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/residual_echo_detector.h
>index e8ae552d6c0..5d18ecbb221 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/residual_echo_detector.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/residual_echo_detector.h
>@@ -37,7 +37,10 @@ class ResidualEchoDetector : public EchoDetector {
>   void AnalyzeCaptureAudio(rtc::ArrayView<const float> capture_audio) override;
> 
>   // This function should be called while holding the capture lock.
>-  void Initialize(int sample_rate_hz, int num_channels) override;
>+  void Initialize(int capture_sample_rate_hz,
>+                  int num_capture_channels,
>+                  int render_sample_rate_hz,
>+                  int num_render_channels) override;
> 
>   // This function is for testing purposes only.
>   void SetReliabilityForTest(float value) { reliability_ = value; }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/residual_echo_detector_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/residual_echo_detector_unittest.cc
>index 7bfa0d2eecb..c6fb8e45c68 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/residual_echo_detector_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/residual_echo_detector_unittest.cc
>@@ -11,13 +11,15 @@
> #include <vector>
> 
> #include "modules/audio_processing/residual_echo_detector.h"
>+#include "rtc_base/refcountedobject.h"
> #include "test/gtest.h"
> 
> namespace webrtc {
> 
> TEST(ResidualEchoDetectorTests, Echo) {
>-  ResidualEchoDetector echo_detector;
>-  echo_detector.SetReliabilityForTest(1.0f);
>+  rtc::scoped_refptr<ResidualEchoDetector> echo_detector =
>+      new rtc::RefCountedObject<ResidualEchoDetector>();
>+  echo_detector->SetReliabilityForTest(1.0f);
>   std::vector<float> ones(160, 1.f);
>   std::vector<float> zeros(160, 0.f);
> 
>@@ -26,24 +28,25 @@ TEST(ResidualEchoDetectorTests, Echo) {
>   // frame interval.
>   for (int i = 0; i < 1000; i++) {
>     if (i % 20 == 0) {
>-      echo_detector.AnalyzeRenderAudio(ones);
>-      echo_detector.AnalyzeCaptureAudio(zeros);
>+      echo_detector->AnalyzeRenderAudio(ones);
>+      echo_detector->AnalyzeCaptureAudio(zeros);
>     } else if (i % 20 == 10) {
>-      echo_detector.AnalyzeRenderAudio(zeros);
>-      echo_detector.AnalyzeCaptureAudio(ones);
>+      echo_detector->AnalyzeRenderAudio(zeros);
>+      echo_detector->AnalyzeCaptureAudio(ones);
>     } else {
>-      echo_detector.AnalyzeRenderAudio(zeros);
>-      echo_detector.AnalyzeCaptureAudio(zeros);
>+      echo_detector->AnalyzeRenderAudio(zeros);
>+      echo_detector->AnalyzeCaptureAudio(zeros);
>     }
>   }
>   // We expect to detect echo with near certain likelihood.
>-  auto ed_metrics = echo_detector.GetMetrics();
>+  auto ed_metrics = echo_detector->GetMetrics();
>   EXPECT_NEAR(1.f, ed_metrics.echo_likelihood, 0.01f);
> }
> 
> TEST(ResidualEchoDetectorTests, NoEcho) {
>-  ResidualEchoDetector echo_detector;
>-  echo_detector.SetReliabilityForTest(1.0f);
>+  rtc::scoped_refptr<ResidualEchoDetector> echo_detector =
>+      new rtc::RefCountedObject<ResidualEchoDetector>();
>+  echo_detector->SetReliabilityForTest(1.0f);
>   std::vector<float> ones(160, 1.f);
>   std::vector<float> zeros(160, 0.f);
> 
>@@ -51,20 +54,21 @@ TEST(ResidualEchoDetectorTests, NoEcho) {
>   // detected.
>   for (int i = 0; i < 1000; i++) {
>     if (i % 20 == 0) {
>-      echo_detector.AnalyzeRenderAudio(ones);
>+      echo_detector->AnalyzeRenderAudio(ones);
>     } else {
>-      echo_detector.AnalyzeRenderAudio(zeros);
>+      echo_detector->AnalyzeRenderAudio(zeros);
>     }
>-    echo_detector.AnalyzeCaptureAudio(zeros);
>+    echo_detector->AnalyzeCaptureAudio(zeros);
>   }
>   // We expect to not detect any echo.
>-  auto ed_metrics = echo_detector.GetMetrics();
>+  auto ed_metrics = echo_detector->GetMetrics();
>   EXPECT_NEAR(0.f, ed_metrics.echo_likelihood, 0.01f);
> }
> 
> TEST(ResidualEchoDetectorTests, EchoWithRenderClockDrift) {
>-  ResidualEchoDetector echo_detector;
>-  echo_detector.SetReliabilityForTest(1.0f);
>+  rtc::scoped_refptr<ResidualEchoDetector> echo_detector =
>+      new rtc::RefCountedObject<ResidualEchoDetector>();
>+  echo_detector->SetReliabilityForTest(1.0f);
>   std::vector<float> ones(160, 1.f);
>   std::vector<float> zeros(160, 0.f);
> 
>@@ -74,18 +78,18 @@ TEST(ResidualEchoDetectorTests, EchoWithRenderClockDrift) {
>   // the render side producing data slightly faster.
>   for (int i = 0; i < 1000; i++) {
>     if (i % 20 == 0) {
>-      echo_detector.AnalyzeRenderAudio(ones);
>-      echo_detector.AnalyzeCaptureAudio(zeros);
>+      echo_detector->AnalyzeRenderAudio(ones);
>+      echo_detector->AnalyzeCaptureAudio(zeros);
>     } else if (i % 20 == 10) {
>-      echo_detector.AnalyzeRenderAudio(zeros);
>-      echo_detector.AnalyzeCaptureAudio(ones);
>+      echo_detector->AnalyzeRenderAudio(zeros);
>+      echo_detector->AnalyzeCaptureAudio(ones);
>     } else {
>-      echo_detector.AnalyzeRenderAudio(zeros);
>-      echo_detector.AnalyzeCaptureAudio(zeros);
>+      echo_detector->AnalyzeRenderAudio(zeros);
>+      echo_detector->AnalyzeCaptureAudio(zeros);
>     }
>     if (i % 100 == 0) {
>       // This is causing the simulated clock drift.
>-      echo_detector.AnalyzeRenderAudio(zeros);
>+      echo_detector->AnalyzeRenderAudio(zeros);
>     }
>   }
>   // We expect to detect echo with high likelihood. Clock drift is harder to
>@@ -94,13 +98,14 @@ TEST(ResidualEchoDetectorTests, EchoWithRenderClockDrift) {
>   // A growing buffer can be caused by jitter or clock drift and it's not
>   // possible to make this decision right away. For this reason we only expect
>   // an echo likelihood of 75% in this test.
>-  auto ed_metrics = echo_detector.GetMetrics();
>+  auto ed_metrics = echo_detector->GetMetrics();
>   EXPECT_GT(ed_metrics.echo_likelihood, 0.75f);
> }
> 
> TEST(ResidualEchoDetectorTests, EchoWithCaptureClockDrift) {
>-  ResidualEchoDetector echo_detector;
>-  echo_detector.SetReliabilityForTest(1.0f);
>+  rtc::scoped_refptr<ResidualEchoDetector> echo_detector =
>+      new rtc::RefCountedObject<ResidualEchoDetector>();
>+  echo_detector->SetReliabilityForTest(1.0f);
>   std::vector<float> ones(160, 1.f);
>   std::vector<float> zeros(160, 0.f);
> 
>@@ -110,22 +115,22 @@ TEST(ResidualEchoDetectorTests, EchoWithCaptureClockDrift) {
>   // the capture side producing data slightly faster.
>   for (int i = 0; i < 1000; i++) {
>     if (i % 20 == 0) {
>-      echo_detector.AnalyzeRenderAudio(ones);
>-      echo_detector.AnalyzeCaptureAudio(zeros);
>+      echo_detector->AnalyzeRenderAudio(ones);
>+      echo_detector->AnalyzeCaptureAudio(zeros);
>     } else if (i % 20 == 10) {
>-      echo_detector.AnalyzeRenderAudio(zeros);
>-      echo_detector.AnalyzeCaptureAudio(ones);
>+      echo_detector->AnalyzeRenderAudio(zeros);
>+      echo_detector->AnalyzeCaptureAudio(ones);
>     } else {
>-      echo_detector.AnalyzeRenderAudio(zeros);
>-      echo_detector.AnalyzeCaptureAudio(zeros);
>+      echo_detector->AnalyzeRenderAudio(zeros);
>+      echo_detector->AnalyzeCaptureAudio(zeros);
>     }
>     if (i % 100 == 0) {
>       // This is causing the simulated clock drift.
>-      echo_detector.AnalyzeCaptureAudio(zeros);
>+      echo_detector->AnalyzeCaptureAudio(zeros);
>     }
>   }
>   // We expect to detect echo with near certain likelihood.
>-  auto ed_metrics = echo_detector.GetMetrics();
>+  auto ed_metrics = echo_detector->GetMetrics();
>   EXPECT_NEAR(1.f, ed_metrics.echo_likelihood, 0.01f);
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/rms_level.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/rms_level.cc
>index 55db226af25..727ecfc9feb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/rms_level.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/rms_level.cc
>@@ -54,7 +54,7 @@ void RmsLevel::Reset() {
>   sum_square_ = 0.f;
>   sample_count_ = 0;
>   max_sum_square_ = 0.f;
>-  block_size_ = rtc::nullopt;
>+  block_size_ = absl::nullopt;
> }
> 
> void RmsLevel::Analyze(rtc::ArrayView<const int16_t> data) {
>@@ -88,7 +88,7 @@ int RmsLevel::Average() {
> 
> RmsLevel::Levels RmsLevel::AverageAndPeak() {
>   // Note that block_size_ should by design always be non-empty when
>-  // sample_count_ != 0. Also, the * operator of rtc::Optional enforces this
>+  // sample_count_ != 0. Also, the * operator of absl::optional enforces this
>   // with a DCHECK.
>   Levels levels = (sample_count_ == 0)
>                       ? Levels{RmsLevel::kMinLevelDb, RmsLevel::kMinLevelDb}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/rms_level.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/rms_level.h
>index 6fe22fd1204..9aa549a57c1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/rms_level.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/rms_level.h
>@@ -11,9 +11,8 @@
> #ifndef MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
> #define MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
> 
>+#include "absl/types/optional.h"
> #include "api/array_view.h"
>-#include "api/optional.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -66,10 +65,9 @@ class RmsLevel {
>   float sum_square_;
>   size_t sample_count_;
>   float max_sum_square_;
>-  rtc::Optional<size_t> block_size_;
>+  absl::optional<size_t> block_size_;
> };
> 
> }  // namespace webrtc
> 
> #endif  // MODULES_AUDIO_PROCESSING_RMS_LEVEL_H_
>-
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/splitting_filter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/splitting_filter.cc
>index e2b8f82c36e..122bc9cc7f7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/splitting_filter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/splitting_filter.cc
>@@ -63,8 +63,7 @@ void SplittingFilter::TwoBandsAnalysis(const IFChannelBuffer* data,
>                                        IFChannelBuffer* bands) {
>   RTC_DCHECK_EQ(two_bands_states_.size(), data->num_channels());
>   for (size_t i = 0; i < two_bands_states_.size(); ++i) {
>-    WebRtcSpl_AnalysisQMF(data->ibuf_const()->channels()[i],
>-                          data->num_frames(),
>+    WebRtcSpl_AnalysisQMF(data->ibuf_const()->channels()[i], data->num_frames(),
>                           bands->ibuf()->channels(0)[i],
>                           bands->ibuf()->channels(1)[i],
>                           two_bands_states_[i].analysis_state1,
>@@ -76,12 +75,11 @@ void SplittingFilter::TwoBandsSynthesis(const IFChannelBuffer* bands,
>                                         IFChannelBuffer* data) {
>   RTC_DCHECK_LE(data->num_channels(), two_bands_states_.size());
>   for (size_t i = 0; i < data->num_channels(); ++i) {
>-    WebRtcSpl_SynthesisQMF(bands->ibuf_const()->channels(0)[i],
>-                           bands->ibuf_const()->channels(1)[i],
>-                           bands->num_frames_per_band(),
>-                           data->ibuf()->channels()[i],
>-                           two_bands_states_[i].synthesis_state1,
>-                           two_bands_states_[i].synthesis_state2);
>+    WebRtcSpl_SynthesisQMF(
>+        bands->ibuf_const()->channels(0)[i],
>+        bands->ibuf_const()->channels(1)[i], bands->num_frames_per_band(),
>+        data->ibuf()->channels()[i], two_bands_states_[i].synthesis_state1,
>+        two_bands_states_[i].synthesis_state2);
>   }
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/splitting_filter_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/splitting_filter_unittest.cc
>index 3e0dbb9b9c8..1caee64a7de 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/splitting_filter_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/splitting_filter_unittest.cc
>@@ -39,8 +39,7 @@ TEST(SplittingFilterTest, SplitsIntoThreeBandsAndReconstructs) {
>   static const int kFrequenciesHz[kNumBands] = {1000, 12000, 18000};
>   static const float kAmplitude = 8192.f;
>   static const size_t kChunks = 8;
>-  SplittingFilter splitting_filter(kChannels,
>-                                   kNumBands,
>+  SplittingFilter splitting_filter(kChannels, kNumBands,
>                                    kSamplesPer48kHzChannel);
>   IFChannelBuffer in_data(kSamplesPer48kHzChannel, kChannels, kNumBands);
>   IFChannelBuffer bands(kSamplesPer48kHzChannel, kChannels, kNumBands);
>@@ -48,8 +47,7 @@ TEST(SplittingFilterTest, SplitsIntoThreeBandsAndReconstructs) {
>   for (size_t i = 0; i < kChunks; ++i) {
>     // Input signal generation.
>     bool is_present[kNumBands];
>-    memset(in_data.fbuf()->channels()[0],
>-           0,
>+    memset(in_data.fbuf()->channels()[0], 0,
>            kSamplesPer48kHzChannel * sizeof(in_data.fbuf()->channels()[0][0]));
>     for (size_t j = 0; j < kNumBands; ++j) {
>       is_present[j] = i & (static_cast<size_t>(1) << j);
>@@ -57,7 +55,7 @@ TEST(SplittingFilterTest, SplitsIntoThreeBandsAndReconstructs) {
>       for (size_t k = 0; k < kSamplesPer48kHzChannel; ++k) {
>         in_data.fbuf()->channels()[0][k] +=
>             amplitude * sin(2.f * M_PI * kFrequenciesHz[j] *
>-                (i * kSamplesPer48kHzChannel + k) / kSampleRateHz);
>+                            (i * kSamplesPer48kHzChannel + k) / kSampleRateHz);
>       }
>     }
>     // Three band splitting filter.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/aec_dump_based_simulator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/aec_dump_based_simulator.cc
>index 6d0b07c7ed0..495318a73cb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/aec_dump_based_simulator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/aec_dump_based_simulator.cc
>@@ -14,6 +14,7 @@
> 
> #include "modules/audio_processing/test/protobuf_utils.h"
> #include "rtc_base/checks.h"
>+#include "rtc_base/numerics/safe_conversions.h"
> 
> namespace webrtc {
> namespace test {
>@@ -62,8 +63,10 @@ bool VerifyFloatBitExactness(const webrtc::audioproc::Stream& msg,
> 
> }  // namespace
> 
>-AecDumpBasedSimulator::AecDumpBasedSimulator(const SimulationSettings& settings)
>-    : AudioProcessingSimulator(settings) {}
>+AecDumpBasedSimulator::AecDumpBasedSimulator(
>+    const SimulationSettings& settings,
>+    std::unique_ptr<AudioProcessingBuilder> ap_builder)
>+    : AudioProcessingSimulator(settings, std::move(ap_builder)) {}
> 
> AecDumpBasedSimulator::~AecDumpBasedSimulator() = default;
> 
>@@ -130,14 +133,16 @@ void AecDumpBasedSimulator::PrepareProcessStreamCall(
>     }
>   }
> 
>-  if (!settings_.stream_delay) {
>-    if (msg.has_delay()) {
>+  if (!settings_.use_stream_delay || *settings_.use_stream_delay) {
>+    if (!settings_.stream_delay) {
>+      if (msg.has_delay()) {
>+        RTC_CHECK_EQ(AudioProcessing::kNoError,
>+                     ap_->set_stream_delay_ms(msg.delay()));
>+      }
>+    } else {
>       RTC_CHECK_EQ(AudioProcessing::kNoError,
>-                   ap_->set_stream_delay_ms(msg.delay()));
>+                   ap_->set_stream_delay_ms(*settings_.stream_delay));
>     }
>-  } else {
>-    RTC_CHECK_EQ(AudioProcessing::kNoError,
>-                 ap_->set_stream_delay_ms(*settings_.stream_delay));
>   }
> 
>   if (!settings_.stream_drift_samples) {
>@@ -462,6 +467,15 @@ void AecDumpBasedSimulator::HandleMessage(
>       }
>     }
> 
>+    if (msg.has_pre_amplifier_enabled() || settings_.use_pre_amplifier) {
>+      const bool enable = settings_.use_pre_amplifier
>+                              ? *settings_.use_pre_amplifier
>+                              : msg.pre_amplifier_enabled();
>+      apm_config.pre_amplifier.enabled = enable;
>+      apm_config.pre_amplifier.fixed_gain_factor =
>+          settings_.pre_amplifier_gain_factor;
>+    }
>+
>     if (settings_.use_verbose_logging && msg.has_experiments_description() &&
>         !msg.experiments_description().empty()) {
>       std::cout << " experiments not included by default in the simulation: "
>@@ -473,10 +487,6 @@ void AecDumpBasedSimulator::HandleMessage(
>           new RefinedAdaptiveFilter(*settings_.use_refined_adaptive_filter));
>     }
> 
>-    if (settings_.use_lc) {
>-      apm_config.level_controller.enabled = *settings_.use_lc;
>-    }
>-
>     if (settings_.use_ed) {
>       apm_config.residual_echo_detector.enabled = *settings_.use_ed;
>     }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/aec_dump_based_simulator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/aec_dump_based_simulator.h
>index 4c29bf7315b..9899c6c29f8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/aec_dump_based_simulator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/aec_dump_based_simulator.h
>@@ -30,7 +30,8 @@ namespace test {
> // Used to perform an audio processing simulation from an aec dump.
> class AecDumpBasedSimulator final : public AudioProcessingSimulator {
>  public:
>-  explicit AecDumpBasedSimulator(const SimulationSettings& settings);
>+  AecDumpBasedSimulator(const SimulationSettings& settings,
>+                        std::unique_ptr<AudioProcessingBuilder> ap_builder);
>   ~AecDumpBasedSimulator() override;
> 
>   // Processes the messages in the aecdump file.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/audio_processing_simulator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/audio_processing_simulator.cc
>index a72ac1c9ac6..f6af179bf0d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/audio_processing_simulator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/audio_processing_simulator.cc
>@@ -11,17 +11,21 @@
> #include "modules/audio_processing/test/audio_processing_simulator.h"
> 
> #include <algorithm>
>+#include <fstream>
> #include <iostream>
> #include <sstream>
> #include <string>
> #include <utility>
> #include <vector>
> 
>+#include "absl/memory/memory.h"
>+#include "api/audio/echo_canceller3_factory.h"
> #include "common_audio/include/audio_util.h"
> #include "modules/audio_processing/aec_dump/aec_dump_factory.h"
> #include "modules/audio_processing/include/audio_processing.h"
> #include "modules/audio_processing/test/fake_recording_device.h"
> #include "rtc_base/checks.h"
>+#include "rtc_base/json.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/stringutils.h"
> 
>@@ -29,6 +33,304 @@ namespace webrtc {
> namespace test {
> namespace {
> 
>+void ReadParam(const Json::Value& root, std::string param_name, bool* param) {
>+  RTC_CHECK(param);
>+  bool v;
>+  if (rtc::GetBoolFromJsonObject(root, param_name, &v)) {
>+    *param = v;
>+    std::cout << param_name << ":" << (*param ? "true" : "false") << std::endl;
>+  }
>+}
>+
>+void ReadParam(const Json::Value& root, std::string param_name, size_t* param) {
>+  RTC_CHECK(param);
>+  int v;
>+  if (rtc::GetIntFromJsonObject(root, param_name, &v)) {
>+    *param = v;
>+    std::cout << param_name << ":" << *param << std::endl;
>+  }
>+}
>+
>+void ReadParam(const Json::Value& root, std::string param_name, int* param) {
>+  RTC_CHECK(param);
>+  int v;
>+  if (rtc::GetIntFromJsonObject(root, param_name, &v)) {
>+    *param = v;
>+    std::cout << param_name << ":" << *param << std::endl;
>+  }
>+}
>+
>+void ReadParam(const Json::Value& root, std::string param_name, float* param) {
>+  RTC_CHECK(param);
>+  double v;
>+  if (rtc::GetDoubleFromJsonObject(root, param_name, &v)) {
>+    *param = static_cast<float>(v);
>+    std::cout << param_name << ":" << *param << std::endl;
>+  }
>+}
>+
>+void ReadParam(const Json::Value& root,
>+               std::string param_name,
>+               EchoCanceller3Config::Filter::MainConfiguration* param) {
>+  RTC_CHECK(param);
>+  Json::Value json_array;
>+  if (rtc::GetValueFromJsonObject(root, param_name, &json_array)) {
>+    std::vector<double> v;
>+    rtc::JsonArrayToDoubleVector(json_array, &v);
>+    if (v.size() != 5) {
>+      std::cout << "Incorrect array size for " << param_name << std::endl;
>+      RTC_CHECK(false);
>+    }
>+    param->length_blocks = static_cast<size_t>(v[0]);
>+    param->leakage_converged = static_cast<float>(v[1]);
>+    param->leakage_diverged = static_cast<float>(v[2]);
>+    param->error_floor = static_cast<float>(v[3]);
>+    param->noise_gate = static_cast<float>(v[4]);
>+
>+    std::cout << param_name << ":"
>+              << "[" << param->length_blocks << "," << param->leakage_converged
>+              << "," << param->leakage_diverged << "," << param->error_floor
>+              << "," << param->noise_gate << "]" << std::endl;
>+  }
>+}
>+
>+void ReadParam(const Json::Value& root,
>+               std::string param_name,
>+               EchoCanceller3Config::Filter::ShadowConfiguration* param) {
>+  RTC_CHECK(param);
>+  Json::Value json_array;
>+  if (rtc::GetValueFromJsonObject(root, param_name, &json_array)) {
>+    std::vector<double> v;
>+    rtc::JsonArrayToDoubleVector(json_array, &v);
>+    if (v.size() != 3) {
>+      std::cout << "Incorrect array size for " << param_name << std::endl;
>+      RTC_CHECK(false);
>+    }
>+    param->length_blocks = static_cast<size_t>(v[0]);
>+    param->rate = static_cast<float>(v[1]);
>+    param->noise_gate = static_cast<float>(v[2]);
>+    std::cout << param_name << ":"
>+              << "[" << param->length_blocks << "," << param->rate << ","
>+              << param->noise_gate << "]" << std::endl;
>+  }
>+}
>+
>+void ReadParam(const Json::Value& root,
>+               std::string param_name,
>+               EchoCanceller3Config::GainUpdates::GainChanges* param) {
>+  RTC_CHECK(param);
>+  Json::Value json_array;
>+  if (rtc::GetValueFromJsonObject(root, param_name, &json_array)) {
>+    std::vector<double> v;
>+    rtc::JsonArrayToDoubleVector(json_array, &v);
>+    if (v.size() != 6) {
>+      std::cout << "Incorrect array size for " << param_name << std::endl;
>+      RTC_CHECK(false);
>+    }
>+    param->max_inc = static_cast<float>(v[0]);
>+    param->max_dec = static_cast<float>(v[1]);
>+    param->rate_inc = static_cast<float>(v[2]);
>+    param->rate_dec = static_cast<float>(v[3]);
>+    param->min_inc = static_cast<float>(v[4]);
>+    param->min_dec = static_cast<float>(v[5]);
>+
>+    std::cout << param_name << ":"
>+              << "[" << param->max_inc << "," << param->max_dec << ","
>+              << param->rate_inc << "," << param->rate_dec << ","
>+              << param->min_inc << "," << param->min_dec << "]" << std::endl;
>+  }
>+}
>+
>+void ReadParam(const Json::Value& root,
>+               std::string param_name,
>+               EchoCanceller3Config::Suppressor::MaskingThresholds* param) {
>+  RTC_CHECK(param);
>+  Json::Value json_array;
>+  if (rtc::GetValueFromJsonObject(root, param_name, &json_array)) {
>+    std::vector<double> v;
>+    rtc::JsonArrayToDoubleVector(json_array, &v);
>+    if (v.size() != 3) {
>+      std::cout << "Incorrect array size for " << param_name << std::endl;
>+      RTC_CHECK(false);
>+    }
>+    param->enr_transparent = static_cast<float>(v[0]);
>+    param->enr_suppress = static_cast<float>(v[1]);
>+    param->emr_transparent = static_cast<float>(v[2]);
>+
>+    std::cout << param_name << ":"
>+              << "[" << param->enr_transparent << "," << param->enr_suppress
>+              << "," << param->emr_transparent << "]" << std::endl;
>+  }
>+}
>+
>+EchoCanceller3Config ParseAec3Parameters(const std::string& filename) {
>+  EchoCanceller3Config cfg;
>+  Json::Value root;
>+  std::string s;
>+  std::string json_string;
>+  std::ifstream f(filename.c_str());
>+
>+  if (f.fail()) {
>+    std::cout << "Failed to open the file " << filename << std::endl;
>+    RTC_CHECK(false);
>+  }
>+
>+  while (std::getline(f, s)) {
>+    json_string += s;
>+  }
>+  bool success = Json::Reader().parse(json_string, root);
>+  if (!success) {
>+    std::cout << "Incorrect JSON format:" << std::endl;
>+    std::cout << json_string << std::endl;
>+    RTC_CHECK(false);
>+  }
>+
>+  std::cout << "AEC3 Parameters from JSON input:" << std::endl;
>+  Json::Value section;
>+  if (rtc::GetValueFromJsonObject(root, "delay", &section)) {
>+    ReadParam(section, "default_delay", &cfg.delay.default_delay);
>+    ReadParam(section, "down_sampling_factor", &cfg.delay.down_sampling_factor);
>+    ReadParam(section, "num_filters", &cfg.delay.num_filters);
>+    ReadParam(section, "api_call_jitter_blocks",
>+              &cfg.delay.api_call_jitter_blocks);
>+    ReadParam(section, "min_echo_path_delay_blocks",
>+              &cfg.delay.min_echo_path_delay_blocks);
>+    ReadParam(section, "delay_headroom_blocks",
>+              &cfg.delay.delay_headroom_blocks);
>+    ReadParam(section, "hysteresis_limit_1_blocks",
>+              &cfg.delay.hysteresis_limit_1_blocks);
>+    ReadParam(section, "hysteresis_limit_2_blocks",
>+              &cfg.delay.hysteresis_limit_2_blocks);
>+    ReadParam(section, "skew_hysteresis_blocks",
>+              &cfg.delay.skew_hysteresis_blocks);
>+  }
>+
>+  if (rtc::GetValueFromJsonObject(root, "filter", &section)) {
>+    ReadParam(section, "main", &cfg.filter.main);
>+    ReadParam(section, "shadow", &cfg.filter.shadow);
>+    ReadParam(section, "main_initial", &cfg.filter.main_initial);
>+    ReadParam(section, "shadow_initial", &cfg.filter.shadow_initial);
>+  }
>+
>+  if (rtc::GetValueFromJsonObject(root, "erle", &section)) {
>+    ReadParam(section, "min", &cfg.erle.min);
>+    ReadParam(section, "max_l", &cfg.erle.max_l);
>+    ReadParam(section, "max_h", &cfg.erle.max_h);
>+  }
>+
>+  if (rtc::GetValueFromJsonObject(root, "ep_strength", &section)) {
>+    ReadParam(section, "lf", &cfg.ep_strength.lf);
>+    ReadParam(section, "mf", &cfg.ep_strength.mf);
>+    ReadParam(section, "hf", &cfg.ep_strength.hf);
>+    ReadParam(section, "default_len", &cfg.ep_strength.default_len);
>+    ReadParam(section, "reverb_based_on_render",
>+              &cfg.ep_strength.reverb_based_on_render);
>+    ReadParam(section, "echo_can_saturate", &cfg.ep_strength.echo_can_saturate);
>+    ReadParam(section, "bounded_erl", &cfg.ep_strength.bounded_erl);
>+  }
>+
>+  if (rtc::GetValueFromJsonObject(root, "gain_mask", &section)) {
>+    ReadParam(section, "m1", &cfg.gain_mask.m1);
>+    ReadParam(section, "m2", &cfg.gain_mask.m2);
>+    ReadParam(section, "m3", &cfg.gain_mask.m3);
>+    ReadParam(section, "m5", &cfg.gain_mask.m5);
>+    ReadParam(section, "m6", &cfg.gain_mask.m6);
>+    ReadParam(section, "m7", &cfg.gain_mask.m7);
>+    ReadParam(section, "m8", &cfg.gain_mask.m8);
>+    ReadParam(section, "m9", &cfg.gain_mask.m9);
>+
>+    ReadParam(section, "gain_curve_offset", &cfg.gain_mask.gain_curve_offset);
>+    ReadParam(section, "gain_curve_slope", &cfg.gain_mask.gain_curve_slope);
>+    ReadParam(section, "temporal_masking_lf",
>+              &cfg.gain_mask.temporal_masking_lf);
>+    ReadParam(section, "temporal_masking_hf",
>+              &cfg.gain_mask.temporal_masking_hf);
>+    ReadParam(section, "temporal_masking_lf_bands",
>+              &cfg.gain_mask.temporal_masking_lf_bands);
>+  }
>+
>+  if (rtc::GetValueFromJsonObject(root, "echo_audibility", &section)) {
>+    ReadParam(section, "low_render_limit",
>+              &cfg.echo_audibility.low_render_limit);
>+    ReadParam(section, "normal_render_limit",
>+              &cfg.echo_audibility.normal_render_limit);
>+
>+    ReadParam(section, "floor_power", &cfg.echo_audibility.floor_power);
>+    ReadParam(section, "audibility_threshold_lf",
>+              &cfg.echo_audibility.audibility_threshold_lf);
>+    ReadParam(section, "audibility_threshold_mf",
>+              &cfg.echo_audibility.audibility_threshold_mf);
>+    ReadParam(section, "audibility_threshold_hf",
>+              &cfg.echo_audibility.audibility_threshold_hf);
>+    ReadParam(section, "use_stationary_properties",
>+              &cfg.echo_audibility.use_stationary_properties);
>+  }
>+
>+  if (rtc::GetValueFromJsonObject(root, "gain_updates", &section)) {
>+    ReadParam(section, "low_noise", &cfg.gain_updates.low_noise);
>+    ReadParam(section, "initial", &cfg.gain_updates.initial);
>+    ReadParam(section, "normal", &cfg.gain_updates.normal);
>+    ReadParam(section, "saturation", &cfg.gain_updates.saturation);
>+    ReadParam(section, "nonlinear", &cfg.gain_updates.nonlinear);
>+    ReadParam(section, "max_inc_factor", &cfg.gain_updates.max_inc_factor);
>+    ReadParam(section, "max_dec_factor_lf",
>+              &cfg.gain_updates.max_dec_factor_lf);
>+    ReadParam(section, "floor_first_increase",
>+              &cfg.gain_updates.floor_first_increase);
>+  }
>+
>+  if (rtc::GetValueFromJsonObject(root, "echo_removal_control", &section)) {
>+    Json::Value subsection;
>+    if (rtc::GetValueFromJsonObject(section, "gain_rampup", &subsection)) {
>+      ReadParam(subsection, "initial_gain",
>+                &cfg.echo_removal_control.gain_rampup.initial_gain);
>+      ReadParam(subsection, "first_non_zero_gain",
>+                &cfg.echo_removal_control.gain_rampup.first_non_zero_gain);
>+      ReadParam(subsection, "non_zero_gain_blocks",
>+                &cfg.echo_removal_control.gain_rampup.non_zero_gain_blocks);
>+      ReadParam(subsection, "full_gain_blocks",
>+                &cfg.echo_removal_control.gain_rampup.full_gain_blocks);
>+    }
>+    ReadParam(section, "has_clock_drift",
>+              &cfg.echo_removal_control.has_clock_drift);
>+    ReadParam(section, "linear_and_stable_echo_path",
>+              &cfg.echo_removal_control.linear_and_stable_echo_path);
>+  }
>+
>+  if (rtc::GetValueFromJsonObject(root, "echo_model", &section)) {
>+    Json::Value subsection;
>+    ReadParam(section, "noise_floor_hold", &cfg.echo_model.noise_floor_hold);
>+    ReadParam(section, "min_noise_floor_power",
>+              &cfg.echo_model.min_noise_floor_power);
>+    ReadParam(section, "stationary_gate_slope",
>+              &cfg.echo_model.stationary_gate_slope);
>+    ReadParam(section, "noise_gate_power", &cfg.echo_model.noise_gate_power);
>+    ReadParam(section, "noise_gate_slope", &cfg.echo_model.noise_gate_slope);
>+    ReadParam(section, "render_pre_window_size",
>+              &cfg.echo_model.render_pre_window_size);
>+    ReadParam(section, "render_post_window_size",
>+              &cfg.echo_model.render_post_window_size);
>+    ReadParam(section, "render_pre_window_size_init",
>+              &cfg.echo_model.render_pre_window_size_init);
>+    ReadParam(section, "render_post_window_size_init",
>+              &cfg.echo_model.render_post_window_size_init);
>+    ReadParam(section, "nonlinear_hold", &cfg.echo_model.nonlinear_hold);
>+    ReadParam(section, "nonlinear_release", &cfg.echo_model.nonlinear_release);
>+  }
>+
>+  if (rtc::GetValueFromJsonObject(root, "suppressor", &section)) {
>+    ReadParam(section, "bands_with_reliable_coherence",
>+              &cfg.suppressor.bands_with_reliable_coherence);
>+    ReadParam(section, "nearend_average_blocks",
>+              &cfg.suppressor.nearend_average_blocks);
>+    ReadParam(section, "mask_lf", &cfg.suppressor.mask_lf);
>+    ReadParam(section, "mask_hf", &cfg.suppressor.mask_hf);
>+  }
>+
>+  std::cout << std::endl;
>+  return cfg;
>+}
>+
> void CopyFromAudioFrame(const AudioFrame& src, ChannelBuffer<float>* dest) {
>   RTC_CHECK_EQ(src.num_channels_, dest->num_channels());
>   RTC_CHECK_EQ(src.samples_per_channel_, dest->num_frames());
>@@ -83,8 +385,11 @@ void CopyToAudioFrame(const ChannelBuffer<float>& src, AudioFrame* dest) {
> }
> 
> AudioProcessingSimulator::AudioProcessingSimulator(
>-    const SimulationSettings& settings)
>+    const SimulationSettings& settings,
>+    std::unique_ptr<AudioProcessingBuilder> ap_builder)
>     : settings_(settings),
>+      ap_builder_(ap_builder ? std::move(ap_builder)
>+                             : absl::make_unique<AudioProcessingBuilder>()),
>       analog_mic_level_(settings.initial_mic_level),
>       fake_recording_device_(
>           settings.initial_mic_level,
>@@ -308,12 +613,6 @@ void AudioProcessingSimulator::CreateAudioProcessor() {
>   Config config;
>   AudioProcessing::Config apm_config;
>   std::unique_ptr<EchoControlFactory> echo_control_factory;
>-  if (settings_.use_bf && *settings_.use_bf) {
>-    config.Set<Beamforming>(new Beamforming(
>-        true, ParseArrayGeometry(*settings_.microphone_positions),
>-        SphericalPointf(DegreesToRadians(settings_.target_angle_degrees), 0.f,
>-                        1.f)));
>-  }
>   if (settings_.use_ts) {
>     config.Set<ExperimentalNs>(new ExperimentalNs(*settings_.use_ts));
>   }
>@@ -324,11 +623,18 @@ void AudioProcessingSimulator::CreateAudioProcessor() {
>     apm_config.gain_controller2.enabled = *settings_.use_agc2;
>     apm_config.gain_controller2.fixed_gain_db = settings_.agc2_fixed_gain_db;
>   }
>-  if (settings_.use_aec3 && *settings_.use_aec3) {
>-    echo_control_factory.reset(new EchoCanceller3Factory());
>+  if (settings_.use_pre_amplifier) {
>+    apm_config.pre_amplifier.enabled = *settings_.use_pre_amplifier;
>+    apm_config.pre_amplifier.fixed_gain_factor =
>+        settings_.pre_amplifier_gain_factor;
>   }
>-  if (settings_.use_lc) {
>-    apm_config.level_controller.enabled = *settings_.use_lc;
>+
>+  if (settings_.use_aec3 && *settings_.use_aec3) {
>+    EchoCanceller3Config cfg;
>+    if (settings_.aec3_settings_filename) {
>+      cfg = ParseAec3Parameters(*settings_.aec3_settings_filename);
>+    }
>+    echo_control_factory.reset(new EchoCanceller3Factory(cfg));
>   }
>   if (settings_.use_hpf) {
>     apm_config.high_pass_filter.enabled = *settings_.use_hpf;
>@@ -343,12 +649,17 @@ void AudioProcessingSimulator::CreateAudioProcessor() {
>   config.Set<DelayAgnostic>(new DelayAgnostic(!settings_.use_delay_agnostic ||
>                                               *settings_.use_delay_agnostic));
>   config.Set<ExperimentalAgc>(new ExperimentalAgc(
>-      !settings_.use_experimental_agc || *settings_.use_experimental_agc));
>+      !settings_.use_experimental_agc || *settings_.use_experimental_agc,
>+      !!settings_.use_experimental_agc_agc2_level_estimator &&
>+          *settings_.use_experimental_agc_agc2_level_estimator,
>+      !!settings_.experimental_agc_disable_digital_adaptive &&
>+          *settings_.experimental_agc_disable_digital_adaptive));
>   if (settings_.use_ed) {
>     apm_config.residual_echo_detector.enabled = *settings_.use_ed;
>   }
> 
>-  ap_.reset(AudioProcessingBuilder()
>+  RTC_CHECK(ap_builder_);
>+  ap_.reset((*ap_builder_)
>                 .SetEchoControlFactory(std::move(echo_control_factory))
>                 .Create(config));
>   RTC_CHECK(ap_);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/audio_processing_simulator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/audio_processing_simulator.h
>index 41a3f451065..e63cc468dc1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/audio_processing_simulator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/audio_processing_simulator.h
>@@ -17,7 +17,7 @@
> #include <memory>
> #include <string>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> #include "common_audio/channel_buffer.h"
> #include "modules/audio_processing/include/audio_processing.h"
> #include "modules/audio_processing/test/fake_recording_device.h"
>@@ -34,62 +34,64 @@ struct SimulationSettings {
>   SimulationSettings();
>   SimulationSettings(const SimulationSettings&);
>   ~SimulationSettings();
>-  rtc::Optional<int> stream_delay;
>-  rtc::Optional<int> stream_drift_samples;
>-  rtc::Optional<int> output_sample_rate_hz;
>-  rtc::Optional<int> output_num_channels;
>-  rtc::Optional<int> reverse_output_sample_rate_hz;
>-  rtc::Optional<int> reverse_output_num_channels;
>-  rtc::Optional<std::string> microphone_positions;
>-  int target_angle_degrees = 90;
>-  rtc::Optional<std::string> output_filename;
>-  rtc::Optional<std::string> reverse_output_filename;
>-  rtc::Optional<std::string> input_filename;
>-  rtc::Optional<std::string> reverse_input_filename;
>-  rtc::Optional<std::string> artificial_nearend_filename;
>-  rtc::Optional<bool> use_aec;
>-  rtc::Optional<bool> use_aecm;
>-  rtc::Optional<bool> use_ed;  // Residual Echo Detector.
>-  rtc::Optional<std::string> ed_graph_output_filename;
>-  rtc::Optional<bool> use_agc;
>-  rtc::Optional<bool> use_agc2;
>-  rtc::Optional<bool> use_hpf;
>-  rtc::Optional<bool> use_ns;
>-  rtc::Optional<bool> use_ts;
>-  rtc::Optional<bool> use_bf;
>-  rtc::Optional<bool> use_ie;
>-  rtc::Optional<bool> use_vad;
>-  rtc::Optional<bool> use_le;
>-  rtc::Optional<bool> use_all;
>-  rtc::Optional<int> aec_suppression_level;
>-  rtc::Optional<bool> use_delay_agnostic;
>-  rtc::Optional<bool> use_extended_filter;
>-  rtc::Optional<bool> use_drift_compensation;
>-  rtc::Optional<bool> use_aec3;
>-  rtc::Optional<bool> use_lc;
>-  rtc::Optional<bool> use_experimental_agc;
>-  rtc::Optional<int> aecm_routing_mode;
>-  rtc::Optional<bool> use_aecm_comfort_noise;
>-  rtc::Optional<int> agc_mode;
>-  rtc::Optional<int> agc_target_level;
>-  rtc::Optional<bool> use_agc_limiter;
>-  rtc::Optional<int> agc_compression_gain;
>+  absl::optional<int> stream_delay;
>+  absl::optional<bool> use_stream_delay;
>+  absl::optional<int> stream_drift_samples;
>+  absl::optional<int> output_sample_rate_hz;
>+  absl::optional<int> output_num_channels;
>+  absl::optional<int> reverse_output_sample_rate_hz;
>+  absl::optional<int> reverse_output_num_channels;
>+  absl::optional<std::string> output_filename;
>+  absl::optional<std::string> reverse_output_filename;
>+  absl::optional<std::string> input_filename;
>+  absl::optional<std::string> reverse_input_filename;
>+  absl::optional<std::string> artificial_nearend_filename;
>+  absl::optional<bool> use_aec;
>+  absl::optional<bool> use_aecm;
>+  absl::optional<bool> use_ed;  // Residual Echo Detector.
>+  absl::optional<std::string> ed_graph_output_filename;
>+  absl::optional<bool> use_agc;
>+  absl::optional<bool> use_agc2;
>+  absl::optional<bool> use_pre_amplifier;
>+  absl::optional<bool> use_hpf;
>+  absl::optional<bool> use_ns;
>+  absl::optional<bool> use_ts;
>+  absl::optional<bool> use_ie;
>+  absl::optional<bool> use_vad;
>+  absl::optional<bool> use_le;
>+  absl::optional<bool> use_all;
>+  absl::optional<int> aec_suppression_level;
>+  absl::optional<bool> use_delay_agnostic;
>+  absl::optional<bool> use_extended_filter;
>+  absl::optional<bool> use_drift_compensation;
>+  absl::optional<bool> use_aec3;
>+  absl::optional<bool> use_experimental_agc;
>+  absl::optional<bool> use_experimental_agc_agc2_level_estimator;
>+  absl::optional<bool> experimental_agc_disable_digital_adaptive;
>+  absl::optional<int> aecm_routing_mode;
>+  absl::optional<bool> use_aecm_comfort_noise;
>+  absl::optional<int> agc_mode;
>+  absl::optional<int> agc_target_level;
>+  absl::optional<bool> use_agc_limiter;
>+  absl::optional<int> agc_compression_gain;
>   float agc2_fixed_gain_db;
>-  rtc::Optional<int> vad_likelihood;
>-  rtc::Optional<int> ns_level;
>-  rtc::Optional<bool> use_refined_adaptive_filter;
>+  float pre_amplifier_gain_factor;
>+  absl::optional<int> vad_likelihood;
>+  absl::optional<int> ns_level;
>+  absl::optional<bool> use_refined_adaptive_filter;
>   int initial_mic_level;
>   bool simulate_mic_gain = false;
>-  rtc::Optional<int> simulated_mic_kind;
>+  absl::optional<int> simulated_mic_kind;
>   bool report_performance = false;
>   bool report_bitexactness = false;
>   bool use_verbose_logging = false;
>   bool discard_all_settings_in_aecdump = true;
>-  rtc::Optional<std::string> aec_dump_input_filename;
>-  rtc::Optional<std::string> aec_dump_output_filename;
>+  absl::optional<std::string> aec_dump_input_filename;
>+  absl::optional<std::string> aec_dump_output_filename;
>   bool fixed_interface = false;
>   bool store_intermediate_output = false;
>-  rtc::Optional<std::string> custom_call_order_filename;
>+  absl::optional<std::string> custom_call_order_filename;
>+  absl::optional<std::string> aec3_settings_filename;
> };
> 
> // Holds a few statistics about a series of TickIntervals.
>@@ -108,7 +110,8 @@ class AudioProcessingSimulator {
>  public:
>   static const int kChunksPerSecond = 1000 / AudioProcessing::kChunkSizeMs;
> 
>-  explicit AudioProcessingSimulator(const SimulationSettings& settings);
>+  AudioProcessingSimulator(const SimulationSettings& settings,
>+                           std::unique_ptr<AudioProcessingBuilder> ap_builder);
>   virtual ~AudioProcessingSimulator();
> 
>   // Processes the data in the input.
>@@ -157,6 +160,7 @@ class AudioProcessingSimulator {
> 
>   const SimulationSettings settings_;
>   std::unique_ptr<AudioProcessing> ap_;
>+  std::unique_ptr<AudioProcessingBuilder> ap_builder_;
> 
>   std::unique_ptr<ChannelBuffer<float>> in_buf_;
>   std::unique_ptr<ChannelBuffer<float>> out_buf_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/audioproc_float.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/audioproc_float_impl.cc
>similarity index 90%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/audioproc_float.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/audioproc_float_impl.cc
>index c5229a4e10b..47576692bd7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/audioproc_float.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/audioproc_float_impl.cc
>@@ -8,14 +8,17 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>+#include <string.h>
>+
> #include <iostream>
> #include <memory>
>-
>-#include <string.h>
>+#include <string>
>+#include <utility>
> 
> #include "modules/audio_processing/include/audio_processing.h"
> #include "modules/audio_processing/test/aec_dump_based_simulator.h"
> #include "modules/audio_processing/test/audio_processing_simulator.h"
>+#include "modules/audio_processing/test/audioproc_float_impl.h"
> #include "modules/audio_processing/test/wav_based_simulator.h"
> #include "rtc_base/flags.h"
> 
>@@ -53,15 +56,6 @@ DEFINE_int(output_sample_rate_hz,
> DEFINE_int(reverse_output_sample_rate_hz,
>            kParameterNotSpecifiedValue,
>            "Reverse stream output sample rate in Hz");
>-DEFINE_string(mic_positions,
>-              "",
>-              "Space delimited cartesian coordinates of microphones in "
>-              "meters. The coordinates of each point are contiguous. For a "
>-              "two element array: \"x1 y1 z1 x2 y2 z2\"");
>-DEFINE_int(target_angle_degrees,
>-           90,
>-           "The azimuth of the target in degrees (0-359). Only applies to "
>-           "beamforming.");
> DEFINE_bool(fixed_interface,
>             false,
>             "Use the fixed interface when operating on wav files");
>@@ -81,6 +75,9 @@ DEFINE_int(agc,
> DEFINE_int(agc2,
>            kParameterNotSpecifiedValue,
>            "Activate (1) or deactivate(0) the AGC2");
>+DEFINE_int(pre_amplifier,
>+           kParameterNotSpecifiedValue,
>+           "Activate (1) or deactivate(0) the pre amplifier");
> DEFINE_int(hpf,
>            kParameterNotSpecifiedValue,
>            "Activate (1) or deactivate(0) the high-pass filter");
>@@ -90,9 +87,6 @@ DEFINE_int(ns,
> DEFINE_int(ts,
>            kParameterNotSpecifiedValue,
>            "Activate (1) or deactivate(0) the transient suppressor");
>-DEFINE_int(bf,
>-           kParameterNotSpecifiedValue,
>-           "Activate (1) or deactivate(0) the beamformer");
> DEFINE_int(ie,
>            kParameterNotSpecifiedValue,
>            "Activate (1) or deactivate(0) the intelligibility enhancer");
>@@ -121,12 +115,13 @@ DEFINE_int(drift_compensation,
> DEFINE_int(aec3,
>            kParameterNotSpecifiedValue,
>            "Activate (1) or deactivate(0) the experimental AEC mode AEC3");
>-DEFINE_int(lc,
>-           kParameterNotSpecifiedValue,
>-           "Activate (1) or deactivate(0) the level control");
> DEFINE_int(experimental_agc,
>            kParameterNotSpecifiedValue,
>            "Activate (1) or deactivate(0) the experimental AGC");
>+DEFINE_int(experimental_agc_disable_digital_adaptive,
>+           kParameterNotSpecifiedValue,
>+           "Force-deactivate (1) digital adaptation in "
>+           "experimental AGC. Digital adaptation is active by default (0).");
> DEFINE_int(
>     refined_adaptive_filter,
>     kParameterNotSpecifiedValue,
>@@ -137,9 +132,7 @@ DEFINE_int(aecm_routing_mode,
> DEFINE_int(aecm_comfort_noise,
>            kParameterNotSpecifiedValue,
>            "Activate (1) or deactivate(0) the AECM comfort noise");
>-DEFINE_int(agc_mode,
>-           kParameterNotSpecifiedValue,
>-           "Specify the AGC mode (0-2)");
>+DEFINE_int(agc_mode, kParameterNotSpecifiedValue, "Specify the AGC mode (0-2)");
> DEFINE_int(agc_target_level,
>            kParameterNotSpecifiedValue,
>            "Specify the AGC target level (0-31)");
>@@ -150,15 +143,19 @@ DEFINE_int(agc_compression_gain,
>            kParameterNotSpecifiedValue,
>            "Specify the AGC compression gain (0-90)");
> DEFINE_float(agc2_fixed_gain_db, 0.f, "AGC2 fixed gain (dB) to apply");
>+DEFINE_float(pre_amplifier_gain_factor,
>+             1.f,
>+             "Pre-amplifier gain factor (linear) to apply");
> DEFINE_int(vad_likelihood,
>            kParameterNotSpecifiedValue,
>            "Specify the VAD likelihood (0-3)");
>-DEFINE_int(ns_level,
>-           kParameterNotSpecifiedValue,
>-           "Specify the NS level (0-3)");
>+DEFINE_int(ns_level, kParameterNotSpecifiedValue, "Specify the NS level (0-3)");
> DEFINE_int(stream_delay,
>            kParameterNotSpecifiedValue,
>            "Specify the stream delay in ms to use");
>+DEFINE_int(use_stream_delay,
>+           kParameterNotSpecifiedValue,
>+           "Activate (1) or deactivate(0) reporting the stream delay");
> DEFINE_int(stream_drift_samples,
>            kParameterNotSpecifiedValue,
>            "Specify the number of stream drift samples to use");
>@@ -181,22 +178,25 @@ DEFINE_bool(store_intermediate_output,
>             false,
>             "Creates new output files after each init");
> DEFINE_string(custom_call_order_file, "", "Custom process API call order file");
>+DEFINE_string(aec3_settings,
>+              "",
>+              "File in JSON-format with custom AEC3 settings");
> DEFINE_bool(help, false, "Print this message");
> 
> void SetSettingIfSpecified(const std::string& value,
>-                           rtc::Optional<std::string>* parameter) {
>+                           absl::optional<std::string>* parameter) {
>   if (value.compare("") != 0) {
>     *parameter = value;
>   }
> }
> 
>-void SetSettingIfSpecified(int value, rtc::Optional<int>* parameter) {
>+void SetSettingIfSpecified(int value, absl::optional<int>* parameter) {
>   if (value != kParameterNotSpecifiedValue) {
>     *parameter = value;
>   }
> }
> 
>-void SetSettingIfFlagSet(int32_t flag, rtc::Optional<bool>* parameter) {
>+void SetSettingIfFlagSet(int32_t flag, absl::optional<bool>* parameter) {
>   if (flag == 0) {
>     *parameter = false;
>   } else if (flag == 1) {
>@@ -210,12 +210,12 @@ SimulationSettings CreateSettings() {
>     settings.use_le = true;
>     settings.use_vad = true;
>     settings.use_ie = false;
>-    settings.use_bf = false;
>     settings.use_ts = true;
>     settings.use_ns = true;
>     settings.use_hpf = true;
>     settings.use_agc = true;
>     settings.use_agc2 = false;
>+    settings.use_pre_amplifier = false;
>     settings.use_aec = true;
>     settings.use_aecm = false;
>     settings.use_ed = false;
>@@ -236,18 +236,16 @@ SimulationSettings CreateSettings() {
>                         &settings.output_sample_rate_hz);
>   SetSettingIfSpecified(FLAG_reverse_output_sample_rate_hz,
>                         &settings.reverse_output_sample_rate_hz);
>-  SetSettingIfSpecified(FLAG_mic_positions, &settings.microphone_positions);
>-  settings.target_angle_degrees = FLAG_target_angle_degrees;
>   SetSettingIfFlagSet(FLAG_aec, &settings.use_aec);
>   SetSettingIfFlagSet(FLAG_aecm, &settings.use_aecm);
>   SetSettingIfFlagSet(FLAG_ed, &settings.use_ed);
>   SetSettingIfSpecified(FLAG_ed_graph, &settings.ed_graph_output_filename);
>   SetSettingIfFlagSet(FLAG_agc, &settings.use_agc);
>   SetSettingIfFlagSet(FLAG_agc2, &settings.use_agc2);
>+  SetSettingIfFlagSet(FLAG_pre_amplifier, &settings.use_pre_amplifier);
>   SetSettingIfFlagSet(FLAG_hpf, &settings.use_hpf);
>   SetSettingIfFlagSet(FLAG_ns, &settings.use_ns);
>   SetSettingIfFlagSet(FLAG_ts, &settings.use_ts);
>-  SetSettingIfFlagSet(FLAG_bf, &settings.use_bf);
>   SetSettingIfFlagSet(FLAG_ie, &settings.use_ie);
>   SetSettingIfFlagSet(FLAG_vad, &settings.use_vad);
>   SetSettingIfFlagSet(FLAG_le, &settings.use_le);
>@@ -261,8 +259,10 @@ SimulationSettings CreateSettings() {
>                       &settings.use_refined_adaptive_filter);
> 
>   SetSettingIfFlagSet(FLAG_aec3, &settings.use_aec3);
>-  SetSettingIfFlagSet(FLAG_lc, &settings.use_lc);
>   SetSettingIfFlagSet(FLAG_experimental_agc, &settings.use_experimental_agc);
>+  SetSettingIfFlagSet(FLAG_experimental_agc_disable_digital_adaptive,
>+                      &settings.experimental_agc_disable_digital_adaptive);
>+
>   SetSettingIfSpecified(FLAG_aecm_routing_mode, &settings.aecm_routing_mode);
>   SetSettingIfFlagSet(FLAG_aecm_comfort_noise,
>                       &settings.use_aecm_comfort_noise);
>@@ -272,13 +272,16 @@ SimulationSettings CreateSettings() {
>   SetSettingIfSpecified(FLAG_agc_compression_gain,
>                         &settings.agc_compression_gain);
>   settings.agc2_fixed_gain_db = FLAG_agc2_fixed_gain_db;
>+  settings.pre_amplifier_gain_factor = FLAG_pre_amplifier_gain_factor;
>   SetSettingIfSpecified(FLAG_vad_likelihood, &settings.vad_likelihood);
>   SetSettingIfSpecified(FLAG_ns_level, &settings.ns_level);
>   SetSettingIfSpecified(FLAG_stream_delay, &settings.stream_delay);
>+  SetSettingIfFlagSet(FLAG_use_stream_delay, &settings.use_stream_delay);
>   SetSettingIfSpecified(FLAG_stream_drift_samples,
>                         &settings.stream_drift_samples);
>   SetSettingIfSpecified(FLAG_custom_call_order_file,
>                         &settings.custom_call_order_filename);
>+  SetSettingIfSpecified(FLAG_aec3_settings, &settings.aec3_settings_filename);
>   settings.initial_mic_level = FLAG_initial_mic_level;
>   settings.simulate_mic_gain = FLAG_simulate_mic_gain;
>   SetSettingIfSpecified(FLAG_simulated_mic_kind, &settings.simulated_mic_kind);
>@@ -348,15 +351,6 @@ void PerformBasicParameterSanityChecks(const SimulationSettings& settings) {
>           *settings.reverse_output_num_channels <= 0,
>       "Error: --reverse_output_num_channels must be positive!\n");
> 
>-  ReportConditionalErrorAndExit(
>-      settings.use_bf && *settings.use_bf && !settings.microphone_positions,
>-      "Error: --mic_positions must be specified when the beamformer is "
>-      "activated.\n");
>-
>-  ReportConditionalErrorAndExit(
>-      settings.target_angle_degrees < 0 || settings.target_angle_degrees > 359,
>-      "Error: -target_angle_degrees must be specified between 0 and 359.\n");
>-
>   ReportConditionalErrorAndExit(
>       settings.aec_suppression_level &&
>           ((*settings.aec_suppression_level) < 0 ||
>@@ -378,11 +372,6 @@ void PerformBasicParameterSanityChecks(const SimulationSettings& settings) {
>                                         (*settings.agc_compression_gain) > 90),
>       "Error: --agc_compression_gain must be specified between 0 and 90.\n");
> 
>-  ReportConditionalErrorAndExit(
>-      settings.use_agc && *settings.use_agc && settings.use_agc2 &&
>-          *settings.use_agc2,
>-      "Error: --agc and --agc2 cannot be both active.\n");
>-
>   ReportConditionalErrorAndExit(
>       settings.use_agc2 && *settings.use_agc2 &&
>           ((settings.agc2_fixed_gain_db) < 0 ||
>@@ -460,9 +449,11 @@ void PerformBasicParameterSanityChecks(const SimulationSettings& settings) {
> 
> }  // namespace
> 
>-int main(int argc, char* argv[]) {
>-  if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) ||
>-      FLAG_help || argc != 1) {
>+int AudioprocFloatImpl(std::unique_ptr<AudioProcessingBuilder> ap_builder,
>+                       int argc,
>+                       char* argv[]) {
>+  if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) || FLAG_help ||
>+      argc != 1) {
>     printf("%s", kUsageDescription);
>     if (FLAG_help) {
>       rtc::FlagList::Print(nullptr, false);
>@@ -476,9 +467,9 @@ int main(int argc, char* argv[]) {
>   std::unique_ptr<AudioProcessingSimulator> processor;
> 
>   if (settings.aec_dump_input_filename) {
>-    processor.reset(new AecDumpBasedSimulator(settings));
>+    processor.reset(new AecDumpBasedSimulator(settings, std::move(ap_builder)));
>   } else {
>-    processor.reset(new WavBasedSimulator(settings));
>+    processor.reset(new WavBasedSimulator(settings, std::move(ap_builder)));
>   }
> 
>   processor->Process();
>@@ -511,7 +502,3 @@ int main(int argc, char* argv[]) {
> 
> }  // namespace test
> }  // namespace webrtc
>-
>-int main(int argc, char* argv[]) {
>-  return webrtc::test::main(argc, argv);
>-}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/audioproc_float_impl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/audioproc_float_impl.h
>new file mode 100644
>index 00000000000..063ecb45b8f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/audioproc_float_impl.h
>@@ -0,0 +1,29 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_PROCESSING_TEST_AUDIOPROC_FLOAT_IMPL_H_
>+#define MODULES_AUDIO_PROCESSING_TEST_AUDIOPROC_FLOAT_IMPL_H_
>+
>+#include <memory>
>+
>+#include "modules/audio_processing/include/audio_processing.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+// This function implements the audio processing simulation utility.
>+int AudioprocFloatImpl(std::unique_ptr<AudioProcessingBuilder> ap_builder,
>+                       int argc,
>+                       char* argv[]);
>+
>+}  // namespace test
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_PROCESSING_TEST_AUDIOPROC_FLOAT_IMPL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/audioproc_float_main.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/audioproc_float_main.cc
>new file mode 100644
>index 00000000000..900b9272117
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/audioproc_float_main.cc
>@@ -0,0 +1,18 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "absl/memory/memory.h"
>+#include "api/test/audioproc_float.h"
>+#include "modules/audio_processing/include/audio_processing.h"
>+
>+int main(int argc, char* argv[]) {
>+  return webrtc::test::AudioprocFloat(
>+      absl::make_unique<webrtc::AudioProcessingBuilder>(), argc, argv);
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn
>index 00e4730abf6..721ebc7ee56 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn
>@@ -23,7 +23,9 @@ rtc_executable("conversational_speech_generator") {
>   deps = [
>     ":lib",
>     "../../../../rtc_base:rtc_base_approved",
>+    "../../../../test:fileutils",
>     "../../../../test:test_support",
>+    "//third_party/abseil-cpp/absl/memory",
>   ]
> }
> 
>@@ -45,11 +47,11 @@ rtc_static_library("lib") {
>   ]
>   deps = [
>     "../../../..:webrtc_common",
>-    "../../../../:typedefs",
>     "../../../../api:array_view",
>     "../../../../common_audio",
>     "../../../../rtc_base:checks",
>     "../../../../rtc_base:rtc_base_approved",
>+    "//third_party/abseil-cpp/absl/memory",
>   ]
>   visibility = [ ":*" ]  # Only targets in this file can depend on this.
> }
>@@ -66,13 +68,12 @@ rtc_source_set("unittest") {
>   deps = [
>     ":lib",
>     "../../../..:webrtc_common",
>-    "../../../../:typedefs",
>     "../../../../api:array_view",
>-    "../../../../api:optional",
>     "../../../../common_audio",
>     "../../../../rtc_base:rtc_base_approved",
>+    "../../../../test:fileutils",
>     "../../../../test:test_support",
>-    "//testing/gmock",
>     "//testing/gtest",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/README.md b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/README.md
>index bbb4112fc01..0fa66669e65 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/README.md
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/README.md
>@@ -36,7 +36,9 @@ A a4 0
> 
> The first column indicates the speaker name, the second contains the audio track
> file names, and the third the offsets (in milliseconds) used to concatenate the
>-chunks.
>+chunks. An optional fourth column contains positive or negative integral gains
>+in dB that will be applied to the tracks. It's possible to specify the gain for
>+some turns but not for others. If the gain is left out, no gain is applied.
> 
> Assume that all the audio tracks in the example above are 1000 ms long.
> The tool will then generate two tracks (A and B) that look like this:
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/generator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/generator.cc
>index ee3f81a9a20..741a1ca86ba 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/generator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/generator.cc
>@@ -10,13 +10,13 @@
> 
> #include <iostream>
> 
>+#include "absl/memory/memory.h"
> #include "modules/audio_processing/test/conversational_speech/config.h"
> #include "modules/audio_processing/test/conversational_speech/multiend_call.h"
> #include "modules/audio_processing/test/conversational_speech/simulator.h"
> #include "modules/audio_processing/test/conversational_speech/timing.h"
> #include "modules/audio_processing/test/conversational_speech/wavreader_factory.h"
> #include "rtc_base/flags.h"
>-#include "rtc_base/ptr_util.h"
> #include "test/testsupport/fileutils.h"
> 
> namespace webrtc {
>@@ -40,8 +40,8 @@ DEFINE_bool(help, false, "Prints this message");
> }  // namespace
> 
> int main(int argc, char* argv[]) {
>-  if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) ||
>-      FLAG_help || argc != 1) {
>+  if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) || FLAG_help ||
>+      argc != 1) {
>     printf("%s", kUsageDescription);
>     if (FLAG_help) {
>       rtc::FlagList::Print(nullptr, false);
>@@ -60,23 +60,23 @@ int main(int argc, char* argv[]) {
>       conversational_speech::LoadTiming(config.timing_filepath());
> 
>   // Parse timing and audio tracks.
>-  auto wavreader_factory = rtc::MakeUnique<
>-      conversational_speech::WavReaderFactory>();
>+  auto wavreader_factory =
>+      absl::make_unique<conversational_speech::WavReaderFactory>();
>   conversational_speech::MultiEndCall multiend_call(
>       timing, config.audiotracks_path(), std::move(wavreader_factory));
> 
>   // Generate output audio tracks.
>-  auto generated_audiotrack_pairs = conversational_speech::Simulate(
>-      multiend_call, config.output_path());
>+  auto generated_audiotrack_pairs =
>+      conversational_speech::Simulate(multiend_call, config.output_path());
> 
>   // Show paths to created audio tracks.
>   std::cout << "Output files:" << std::endl;
>   for (const auto& output_paths_entry : *generated_audiotrack_pairs) {
>     std::cout << "  speaker: " << output_paths_entry.first << std::endl;
>     std::cout << "    near end: " << output_paths_entry.second.near_end
>-        << std::endl;
>+              << std::endl;
>     std::cout << "    far end: " << output_paths_entry.second.far_end
>-        << std::endl;
>+              << std::endl;
>   }
> 
>   return 0;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/generator_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/generator_unittest.cc
>index 064e8c82d76..5ef11db5dba 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/generator_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/generator_unittest.cc
>@@ -42,7 +42,7 @@
> #include <memory>
> #include <vector>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> #include "common_audio/wav_file.h"
> #include "modules/audio_processing/test/conversational_speech/config.h"
> #include "modules/audio_processing/test/conversational_speech/mock_wavreader_factory.h"
>@@ -51,7 +51,6 @@
> #include "modules/audio_processing/test/conversational_speech/timing.h"
> #include "modules/audio_processing/test/conversational_speech/wavreader_factory.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/pathutils.h"
> #include "test/gmock.h"
> #include "test/gtest.h"
> #include "test/testsupport/fileutils.h"
>@@ -72,12 +71,8 @@ const char* const timing_filepath = "/path/to/timing_file.txt";
> const char* const output_path = "/path/to/output_dir";
> 
> const std::vector<Turn> expected_timing = {
>-    {"A", "a1", 0},
>-    {"B", "b1", 0},
>-    {"A", "a2", 100},
>-    {"B", "b2", -200},
>-    {"A", "a3", 0},
>-    {"A", "a3", 0},
>+    {"A", "a1", 0, 0},    {"B", "b1", 0, 0}, {"A", "a2", 100, 0},
>+    {"B", "b2", -200, 0}, {"A", "a3", 0, 0}, {"A", "a3", 0, 0},
> };
> const std::size_t kNumberOfTurns = expected_timing.size();
> 
>@@ -129,18 +124,18 @@ struct SineAudioTrackParams {
> std::string CreateTemporarySineAudioTracks(
>     const std::map<std::string, SineAudioTrackParams>& sine_tracks_params) {
>   // Create temporary directory.
>-  rtc::Pathname temp_directory(OutputPath());
>-  temp_directory.AppendFolder("TempConversationalSpeechAudioTracks");
>-  CreateDir(temp_directory.pathname());
>+  std::string temp_directory =
>+      OutputPath() + "TempConversationalSpeechAudioTracks";
>+  CreateDir(temp_directory);
> 
>   // Create sine tracks.
>   for (const auto& it : sine_tracks_params) {
>-    const rtc::Pathname temp_filepath(temp_directory.pathname(), it.first);
>+    const std::string temp_filepath = JoinFilename(temp_directory, it.first);
>     CreateSineWavFile(
>-        temp_filepath.pathname(), it.second.params, it.second.frequency);
>+        temp_filepath, it.second.params, it.second.frequency);
>   }
> 
>-  return temp_directory.pathname();
>+  return temp_directory;
> }
> 
> void CheckAudioTrackParams(const WavReaderFactory& wav_reader_factory,
>@@ -154,7 +149,7 @@ void CheckAudioTrackParams(const WavReaderFactory& wav_reader_factory,
> 
> void DeleteFolderAndContents(const std::string& dir) {
>   if (!DirExists(dir)) { return; }
>-  rtc::Optional<std::vector<std::string>> dir_content = ReadDirectory(dir);
>+  absl::optional<std::vector<std::string>> dir_content = ReadDirectory(dir);
>   EXPECT_TRUE(dir_content);
>   for (const auto& path : *dir_content) {
>     if (DirExists(path)) {
>@@ -192,7 +187,7 @@ TEST(ConversationalSpeechTest, TimingSaveLoad) {
> 
>   // Create a std::vector<Turn> instance by loading from file.
>   std::vector<Turn> actual_timing = LoadTiming(temporary_filepath);
>-  std::remove(temporary_filepath.c_str());
>+  RemoveFile(temporary_filepath);
> 
>   // Check size.
>   EXPECT_EQ(expected_timing.size(), actual_timing.size());
>@@ -223,8 +218,7 @@ TEST(ConversationalSpeechTest, MultiEndCallCreate) {
> 
> TEST(ConversationalSpeechTest, MultiEndCallSetupDifferentSampleRates) {
>   const std::vector<Turn> timing = {
>-      {"A", "sr8000", 0},
>-      {"B", "sr16000", 0},
>+      {"A", "sr8000", 0, 0}, {"B", "sr16000", 0, 0},
>   };
>   auto mock_wavreader_factory = CreateMockWavReaderFactory();
> 
>@@ -238,8 +232,7 @@ TEST(ConversationalSpeechTest, MultiEndCallSetupDifferentSampleRates) {
> 
> TEST(ConversationalSpeechTest, MultiEndCallSetupMultipleChannels) {
>   const std::vector<Turn> timing = {
>-      {"A", "sr16000_stereo", 0},
>-      {"B", "sr16000_stereo", 0},
>+      {"A", "sr16000_stereo", 0, 0}, {"B", "sr16000_stereo", 0, 0},
>   };
>   auto mock_wavreader_factory = CreateMockWavReaderFactory();
> 
>@@ -254,8 +247,7 @@ TEST(ConversationalSpeechTest, MultiEndCallSetupMultipleChannels) {
> TEST(ConversationalSpeechTest,
>        MultiEndCallSetupDifferentSampleRatesAndMultipleNumChannels) {
>   const std::vector<Turn> timing = {
>-      {"A", "sr8000", 0},
>-      {"B", "sr16000_stereo", 0},
>+      {"A", "sr8000", 0, 0}, {"B", "sr16000_stereo", 0, 0},
>   };
>   auto mock_wavreader_factory = CreateMockWavReaderFactory();
> 
>@@ -269,8 +261,7 @@ TEST(ConversationalSpeechTest,
> 
> TEST(ConversationalSpeechTest, MultiEndCallSetupFirstOffsetNegative) {
>   const std::vector<Turn> timing = {
>-      {"A", "t500", -100},
>-      {"B", "t500", 0},
>+      {"A", "t500", -100, 0}, {"B", "t500", 0, 0},
>   };
>   auto mock_wavreader_factory = CreateMockWavReaderFactory();
> 
>@@ -288,8 +279,7 @@ TEST(ConversationalSpeechTest, MultiEndCallSetupSimple) {
>   // B .....1****
>   constexpr std::size_t expected_duration = kDefaultSampleRate;
>   const std::vector<Turn> timing = {
>-      {"A", "t500", 0},
>-      {"B", "t500", 0},
>+      {"A", "t500", 0, 0}, {"B", "t500", 0, 0},
>   };
>   auto mock_wavreader_factory = CreateMockWavReaderFactory();
> 
>@@ -313,8 +303,7 @@ TEST(ConversationalSpeechTest, MultiEndCallSetupPause) {
>   // B .......1****
>   constexpr std::size_t expected_duration = kDefaultSampleRate * 1.2;
>   const std::vector<Turn> timing = {
>-      {"A", "t500", 0},
>-      {"B", "t500", 200},
>+      {"A", "t500", 0, 0}, {"B", "t500", 200, 0},
>   };
>   auto mock_wavreader_factory = CreateMockWavReaderFactory();
> 
>@@ -338,8 +327,7 @@ TEST(ConversationalSpeechTest, MultiEndCallSetupCrossTalk) {
>   // B ....1****
>   constexpr std::size_t expected_duration = kDefaultSampleRate * 0.9;
>   const std::vector<Turn> timing = {
>-      {"A", "t500", 0},
>-      {"B", "t500", -100},
>+      {"A", "t500", 0, 0}, {"B", "t500", -100, 0},
>   };
>   auto mock_wavreader_factory = CreateMockWavReaderFactory();
> 
>@@ -362,8 +350,7 @@ TEST(ConversationalSpeechTest, MultiEndCallSetupInvalidOrder) {
>   // A ..0****
>   // B .1****.  The n-th turn cannot start before the (n-1)-th one.
>   const std::vector<Turn> timing = {
>-      {"A", "t500", 200},
>-      {"B", "t500", -600},
>+      {"A", "t500", 200, 0}, {"B", "t500", -600, 0},
>   };
>   auto mock_wavreader_factory = CreateMockWavReaderFactory();
> 
>@@ -381,9 +368,7 @@ TEST(ConversationalSpeechTest, MultiEndCallSetupCrossTalkThree) {
>   // B ...1*********
>   constexpr std::size_t expected_duration = kDefaultSampleRate * 1.3;
>   const std::vector<Turn> timing = {
>-      {"A", "t500", 0},
>-      {"B", "t1000", -200},
>-      {"A", "t500", -800},
>+      {"A", "t500", 0, 0}, {"B", "t1000", -200, 0}, {"A", "t500", -800, 0},
>   };
>   auto mock_wavreader_factory = CreateMockWavReaderFactory();
> 
>@@ -408,9 +393,7 @@ TEST(ConversationalSpeechTest, MultiEndCallSetupSelfCrossTalkNearInvalid) {
>   // B ......2****
>   //      ^  Turn #1 overlaps with #0 which is from the same speaker.
>   const std::vector<Turn> timing = {
>-      {"A", "t500", 0},
>-      {"A", "t500", -200},
>-      {"B", "t500", -200},
>+      {"A", "t500", 0, 0}, {"A", "t500", -200, 0}, {"B", "t500", -200, 0},
>   };
>   auto mock_wavreader_factory = CreateMockWavReaderFactory();
> 
>@@ -430,10 +413,10 @@ TEST(ConversationalSpeechTest, MultiEndCallSetupSelfCrossTalkFarInvalid) {
>   // A ......3**.
>   //         ^  Turn #3 overlaps with #0 which is from the same speaker.
>   const std::vector<Turn> timing = {
>-      {"A", "t1000", 0},
>-      {"B", "t300", -1000},
>-      {"C", "t300", 0},
>-      {"A", "t300", 0},
>+      {"A", "t1000", 0, 0},
>+      {"B", "t300", -1000, 0},
>+      {"C", "t300", 0, 0},
>+      {"A", "t300", 0, 0},
>   };
>   auto mock_wavreader_factory = CreateMockWavReaderFactory();
> 
>@@ -452,9 +435,7 @@ TEST(ConversationalSpeechTest, MultiEndCallSetupCrossTalkMiddleValid) {
>   // C .......2****
>   constexpr std::size_t expected_duration = kDefaultSampleRate * 1.2;
>   const std::vector<Turn> timing = {
>-      {"A", "t1000", 0},
>-      {"B", "t500", -800},
>-      {"C", "t500", 0},
>+      {"A", "t1000", 0, 0}, {"B", "t500", -800, 0}, {"C", "t500", 0, 0},
>   };
>   auto mock_wavreader_factory = CreateMockWavReaderFactory();
> 
>@@ -480,9 +461,7 @@ TEST(ConversationalSpeechTest, MultiEndCallSetupCrossTalkMiddleInvalid) {
>   //       ^  Turn #2 overlaps both with #0 and #1 (cross-talk with 3+ speakers
>   //          not permitted).
>   const std::vector<Turn> timing = {
>-      {"A", "t1000", 0},
>-      {"B", "t500", -800},
>-      {"C", "t500", -300},
>+      {"A", "t1000", 0, 0}, {"B", "t500", -800, 0}, {"C", "t500", -300, 0},
>   };
>   auto mock_wavreader_factory = CreateMockWavReaderFactory();
> 
>@@ -501,9 +480,7 @@ TEST(ConversationalSpeechTest, MultiEndCallSetupCrossTalkMiddleAndPause) {
>   // C .......3****
>   constexpr std::size_t expected_duration = kDefaultSampleRate * 1.2;
>   const std::vector<Turn> timing = {
>-      {"A", "t1000", 0},
>-      {"B", "t500", -900},
>-      {"C", "t500", 100},
>+      {"A", "t1000", 0, 0}, {"B", "t500", -900, 0}, {"C", "t500", 100, 0},
>   };
>   auto mock_wavreader_factory = CreateMockWavReaderFactory();
> 
>@@ -526,8 +503,7 @@ TEST(ConversationalSpeechTest, MultiEndCallSetupCrossTalkFullOverlapValid) {
>   // A 0****
>   // B 1****
>   const std::vector<Turn> timing = {
>-      {"A", "t500", 0},
>-      {"B", "t500", -500},
>+      {"A", "t500", 0, 0}, {"B", "t500", -500, 0},
>   };
>   auto mock_wavreader_factory = CreateMockWavReaderFactory();
> 
>@@ -551,13 +527,9 @@ TEST(ConversationalSpeechTest, MultiEndCallSetupLongSequence) {
>   // C ......2**.......6**..
>   constexpr std::size_t expected_duration = kDefaultSampleRate * 1.9;
>   const std::vector<Turn> timing = {
>-      {"A", "t500", 0},
>-      {"B", "t500", 0},
>-      {"C", "t300", -400},
>-      {"A", "t500", 0},
>-      {"B", "t300", -100},
>-      {"A", "t300", -100},
>-      {"C", "t300", -200},
>+      {"A", "t500", 0, 0},    {"B", "t500", 0, 0},    {"C", "t300", -400, 0},
>+      {"A", "t500", 0, 0},    {"B", "t300", -100, 0}, {"A", "t300", -100, 0},
>+      {"C", "t300", -200, 0},
>   };
>   auto mock_wavreader_factory = std::unique_ptr<MockWavReaderFactory>(
>       new MockWavReaderFactory(kDefaultMockWavReaderFactoryParams,
>@@ -585,13 +557,9 @@ TEST(ConversationalSpeechTest, MultiEndCallSetupLongSequenceInvalid) {
>   //                 ^ Turns #4, #5 and #6 overlapping (cross-talk with 3+
>   //                   speakers not permitted).
>   const std::vector<Turn> timing = {
>-      {"A", "t500", 0},
>-      {"B", "t500", 0},
>-      {"C", "t300", -400},
>-      {"A", "t500", 0},
>-      {"B", "t300", -100},
>-      {"A", "t300", -200},
>-      {"C", "t300", -200},
>+      {"A", "t500", 0, 0},    {"B", "t500", 0, 0},    {"C", "t300", -400, 0},
>+      {"A", "t500", 0, 0},    {"B", "t300", -100, 0}, {"A", "t300", -200, 0},
>+      {"C", "t300", -200, 0},
>   };
>   auto mock_wavreader_factory = std::unique_ptr<MockWavReaderFactory>(
>       new MockWavReaderFactory(kDefaultMockWavReaderFactoryParams,
>@@ -611,24 +579,24 @@ TEST(ConversationalSpeechTest, MultiEndCallWavReaderAdaptorSine) {
>   const int sample_rates[] = {8000, 11025, 16000, 22050, 32000, 44100, 48000};
> 
>   for (int sample_rate : sample_rates) {
>-    const rtc::Pathname temp_filename(
>-        OutputPath(), "TempSineWavFile_" + std::to_string(sample_rate)
>-            + ".wav");
>+    const std::string temp_filename =
>+        OutputPath() + "TempSineWavFile_" +
>+        std::to_string(sample_rate) + ".wav";
> 
>     // Write wav file.
>     const std::size_t num_samples = duration_seconds * sample_rate;
>     MockWavReaderFactory::Params params = {sample_rate, 1u, num_samples};
>-    CreateSineWavFile(temp_filename.pathname(), params);
>+    CreateSineWavFile(temp_filename, params);
> 
>     // Load wav file and check if params match.
>     WavReaderFactory wav_reader_factory;
>     MockWavReaderFactory::Params expeted_params = {
>         sample_rate, 1u, num_samples};
>     CheckAudioTrackParams(
>-        wav_reader_factory, temp_filename.pathname(), expeted_params);
>+        wav_reader_factory, temp_filename, expeted_params);
> 
>     // Clean up.
>-    remove(temp_filename.pathname().c_str());
>+    RemoveFile(temp_filename);
>   }
> }
> 
>@@ -637,10 +605,10 @@ TEST(ConversationalSpeechTest, DISABLED_MultiEndCallSimulator) {
>   // A 0*********...........2*********.....
>   // B ...........1*********.....3*********
>   const std::vector<Turn> expected_timing = {
>-      {"A", "t5000_440.wav", 0},
>-      {"B", "t5000_880.wav", 500},
>-      {"A", "t5000_440.wav", 0},
>-      {"B", "t5000_880.wav", -2500},
>+      {"A", "t5000_440.wav", 0, 0},
>+      {"B", "t5000_880.wav", 500, 0},
>+      {"A", "t5000_440.wav", 0, 0},
>+      {"B", "t5000_880.wav", -2500, 0},
>   };
>   const std::size_t expected_duration_seconds = 18;
> 
>@@ -660,12 +628,11 @@ TEST(ConversationalSpeechTest, DISABLED_MultiEndCallSimulator) {
>       expected_timing, audiotracks_path, std::move(wavreader_factory));
> 
>   // Simulate the call.
>-  rtc::Pathname output_path(audiotracks_path);
>-  output_path.AppendFolder("output");
>-  CreateDir(output_path.pathname());
>-  RTC_LOG(LS_VERBOSE) << "simulator output path: " << output_path.pathname();
>+  std::string output_path = JoinFilename(audiotracks_path, "output");
>+  CreateDir(output_path);
>+  RTC_LOG(LS_VERBOSE) << "simulator output path: " << output_path;
>   auto generated_audiotrak_pairs = conversational_speech::Simulate(
>-      multiend_call, output_path.pathname());
>+      multiend_call, output_path);
>   EXPECT_EQ(2u, generated_audiotrak_pairs->size());
> 
>   // Check the output.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader.cc
>index 103483d83e1..58d68e68b21 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader.cc
>@@ -16,10 +16,12 @@ namespace conversational_speech {
> 
> using testing::Return;
> 
>-MockWavReader::MockWavReader(
>-    int sample_rate, size_t num_channels, size_t num_samples)
>-      : sample_rate_(sample_rate), num_channels_(num_channels),
>-          num_samples_(num_samples) {
>+MockWavReader::MockWavReader(int sample_rate,
>+                             size_t num_channels,
>+                             size_t num_samples)
>+    : sample_rate_(sample_rate),
>+      num_channels_(num_channels),
>+      num_samples_(num_samples) {
>   ON_CALL(*this, SampleRate()).WillByDefault(Return(sample_rate_));
>   ON_CALL(*this, NumChannels()).WillByDefault(Return(num_channels_));
>   ON_CALL(*this, NumSamples()).WillByDefault(Return(num_samples_));
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader.h
>index 55d00b31aed..591299eefe5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader.h
>@@ -17,7 +17,6 @@
> #include "api/array_view.h"
> #include "modules/audio_processing/test/conversational_speech/wavreader_interface.h"
> #include "test/gmock.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> namespace test {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader_factory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader_factory.cc
>index 76f5c807187..eb8e3bed3c5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader_factory.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader_factory.cc
>@@ -25,10 +25,9 @@ using testing::Invoke;
> MockWavReaderFactory::MockWavReaderFactory(
>     const Params& default_params,
>     const std::map<std::string, const Params>& params)
>-        : default_params_(default_params),
>-          audiotrack_names_params_(params) {
>-  ON_CALL(*this, Create(_)).WillByDefault(Invoke(
>-      this, &MockWavReaderFactory::CreateMock));
>+    : default_params_(default_params), audiotrack_names_params_(params) {
>+  ON_CALL(*this, Create(_))
>+      .WillByDefault(Invoke(this, &MockWavReaderFactory::CreateMock));
> }
> 
> MockWavReaderFactory::MockWavReaderFactory(const Params& default_params)
>@@ -41,16 +40,15 @@ std::unique_ptr<WavReaderInterface> MockWavReaderFactory::CreateMock(
>     const std::string& filepath) {
>   // Search the parameters corresponding to filepath.
>   const rtc::Pathname audiotrack_file_path(filepath);
>-  const auto it = audiotrack_names_params_.find(
>-      audiotrack_file_path.filename());
>+  const auto it =
>+      audiotrack_names_params_.find(audiotrack_file_path.filename());
> 
>   // If not found, use default parameters.
>   if (it == audiotrack_names_params_.end()) {
>     RTC_LOG(LS_VERBOSE) << "using default parameters for " << filepath;
>-    return std::unique_ptr<WavReaderInterface>(
>-        new MockWavReader(default_params_.sample_rate,
>-                          default_params_.num_channels,
>-                          default_params_.num_samples));
>+    return std::unique_ptr<WavReaderInterface>(new MockWavReader(
>+        default_params_.sample_rate, default_params_.num_channels,
>+        default_params_.num_samples));
>   }
> 
>   // Found, use the audiotrack-specific parameters.
>@@ -58,10 +56,8 @@ std::unique_ptr<WavReaderInterface> MockWavReaderFactory::CreateMock(
>   RTC_LOG(LS_VERBOSE) << "sample_rate " << it->second.sample_rate;
>   RTC_LOG(LS_VERBOSE) << "num_channels " << it->second.num_channels;
>   RTC_LOG(LS_VERBOSE) << "num_samples " << it->second.num_samples;
>-  return std::unique_ptr<WavReaderInterface>(
>-      new MockWavReader(it->second.sample_rate,
>-                        it->second.num_channels,
>-                        it->second.num_samples));
>+  return std::unique_ptr<WavReaderInterface>(new MockWavReader(
>+      it->second.sample_rate, it->second.num_channels, it->second.num_samples));
> }
> 
> }  // namespace conversational_speech
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader_factory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader_factory.h
>index 4c055cf3aed..e84f5f35ccb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader_factory.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader_factory.h
>@@ -25,7 +25,7 @@ namespace conversational_speech {
> 
> class MockWavReaderFactory : public WavReaderAbstractFactory {
>  public:
>-  struct Params{
>+  struct Params {
>     int sample_rate;
>     size_t num_channels;
>     size_t num_samples;
>@@ -36,8 +36,8 @@ class MockWavReaderFactory : public WavReaderAbstractFactory {
>   explicit MockWavReaderFactory(const Params& default_params);
>   ~MockWavReaderFactory();
> 
>-  MOCK_CONST_METHOD1(Create, std::unique_ptr<WavReaderInterface>(
>-      const std::string&));
>+  MOCK_CONST_METHOD1(Create,
>+                     std::unique_ptr<WavReaderInterface>(const std::string&));
> 
>  private:
>   // Creates a MockWavReader instance using the parameters in
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/multiend_call.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/multiend_call.cc
>index 76cf774afc8..d633d90e9c4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/multiend_call.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/multiend_call.cc
>@@ -21,11 +21,13 @@ namespace test {
> namespace conversational_speech {
> 
> MultiEndCall::MultiEndCall(
>-    rtc::ArrayView<const Turn> timing, const std::string& audiotracks_path,
>+    rtc::ArrayView<const Turn> timing,
>+    const std::string& audiotracks_path,
>     std::unique_ptr<WavReaderAbstractFactory> wavreader_abstract_factory)
>-        : timing_(timing), audiotracks_path_(audiotracks_path),
>-          wavreader_abstract_factory_(std::move(wavreader_abstract_factory)),
>-          valid_(false) {
>+    : timing_(timing),
>+      audiotracks_path_(audiotracks_path),
>+      wavreader_abstract_factory_(std::move(wavreader_abstract_factory)),
>+      valid_(false) {
>   FindSpeakerNames();
>   if (CreateAudioTrackReaders())
>     valid_ = CheckTiming();
>@@ -49,8 +51,8 @@ bool MultiEndCall::CreateAudioTrackReaders() {
>       continue;
> 
>     // Instance Pathname to retrieve the full path to the audiotrack file.
>-    const rtc::Pathname audiotrack_file_path(
>-        audiotracks_path_, turn.audiotrack_file_name);
>+    const rtc::Pathname audiotrack_file_path(audiotracks_path_,
>+                                             turn.audiotrack_file_name);
> 
>     // Map the audiotrack file name to a new instance of WavReaderInterface.
>     std::unique_ptr<WavReaderInterface> wavreader =
>@@ -69,8 +71,8 @@ bool MultiEndCall::CreateAudioTrackReaders() {
>       return false;
>     }
> 
>-    audiotrack_readers_.emplace(
>-        turn.audiotrack_file_name, std::move(wavreader));
>+    audiotrack_readers_.emplace(turn.audiotrack_file_name,
>+                                std::move(wavreader));
>   }
> 
>   return true;
>@@ -101,10 +103,9 @@ bool MultiEndCall::CheckTiming() {
>   // detect self cross-talk).
>   std::map<std::string, std::vector<size_t>> speaking_turn_indices;
>   for (const std::string& speaker_name : speaker_names_) {
>-    speaking_turn_indices.emplace(
>-        std::piecewise_construct,
>-        std::forward_as_tuple(speaker_name),
>-        std::forward_as_tuple());
>+    speaking_turn_indices.emplace(std::piecewise_construct,
>+                                  std::forward_as_tuple(speaker_name),
>+                                  std::forward_as_tuple());
>   }
> 
>   // Parse turns.
>@@ -115,8 +116,8 @@ bool MultiEndCall::CheckTiming() {
>         << "Audio track reader not created";
> 
>     // Begin and end timestamps for the current turn.
>-    int offset_samples = millisecond_to_samples(
>-        turn.offset, it->second->SampleRate());
>+    int offset_samples =
>+        millisecond_to_samples(turn.offset, it->second->SampleRate());
>     std::size_t begin_timestamp = last_turn.end + offset_samples;
>     std::size_t end_timestamp = begin_timestamp + it->second->NumSamples();
>     RTC_LOG(LS_INFO) << "turn #" << turn_index << " " << begin_timestamp << "-"
>@@ -124,24 +125,23 @@ bool MultiEndCall::CheckTiming() {
> 
>     // The order is invalid if the offset is negative and its absolute value is
>     // larger then the duration of the previous turn.
>-    if (offset_samples < 0 && -offset_samples > static_cast<int>(
>-        last_turn.end - last_turn.begin)) {
>+    if (offset_samples < 0 &&
>+        -offset_samples > static_cast<int>(last_turn.end - last_turn.begin)) {
>       RTC_LOG(LS_ERROR) << "invalid order";
>       return false;
>     }
> 
>     // Cross-talk with 3 or more speakers occurs when the beginning of the
>     // current interval falls in the last two turns.
>-    if (turn_index > 1 && in_interval(begin_timestamp, last_turn)
>-        && in_interval(begin_timestamp, second_last_turn)) {
>+    if (turn_index > 1 && in_interval(begin_timestamp, last_turn) &&
>+        in_interval(begin_timestamp, second_last_turn)) {
>       RTC_LOG(LS_ERROR) << "cross-talk with 3+ speakers";
>       return false;
>     }
> 
>     // Append turn.
>-    speaking_turns_.emplace_back(
>-        turn.speaker_name, turn.audiotrack_file_name,
>-        begin_timestamp, end_timestamp);
>+    speaking_turns_.emplace_back(turn.speaker_name, turn.audiotrack_file_name,
>+                                 begin_timestamp, end_timestamp, turn.gain);
> 
>     // Save speaking turn index for self cross-talk detection.
>     RTC_DCHECK_EQ(speaking_turns_.size(), turn_index + 1);
>@@ -165,8 +165,9 @@ bool MultiEndCall::CheckTiming() {
>     std::vector<SpeakingTurn> speaking_turns_for_name;
>     std::copy_if(speaking_turns_.begin(), speaking_turns_.end(),
>                  std::back_inserter(speaking_turns_for_name),
>-                 [&speaker_name](const SpeakingTurn& st){
>-                   return st.speaker_name == speaker_name; });
>+                 [&speaker_name](const SpeakingTurn& st) {
>+                   return st.speaker_name == speaker_name;
>+                 });
> 
>     // Check for overlap between adjacent elements.
>     // This is a sufficient condition for self cross-talk since the intervals
>@@ -174,7 +175,8 @@ bool MultiEndCall::CheckTiming() {
>     auto overlap = std::adjacent_find(
>         speaking_turns_for_name.begin(), speaking_turns_for_name.end(),
>         [](const SpeakingTurn& a, const SpeakingTurn& b) {
>-            return a.end > b.begin; });
>+          return a.end > b.begin;
>+        });
> 
>     if (overlap != speaking_turns_for_name.end()) {
>       RTC_LOG(LS_ERROR) << "Self cross-talk detected";
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/multiend_call.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/multiend_call.h
>index e4cee7ac965..a4eff382bd3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/multiend_call.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/multiend_call.h
>@@ -35,29 +35,38 @@ class MultiEndCall {
>     // Constructor required in order to use std::vector::emplace_back().
>     SpeakingTurn(std::string new_speaker_name,
>                  std::string new_audiotrack_file_name,
>-                 size_t new_begin, size_t new_end)
>+                 size_t new_begin,
>+                 size_t new_end,
>+                 int gain)
>         : speaker_name(std::move(new_speaker_name)),
>           audiotrack_file_name(std::move(new_audiotrack_file_name)),
>-          begin(new_begin), end(new_end) {}
>+          begin(new_begin),
>+          end(new_end),
>+          gain(gain) {}
>     std::string speaker_name;
>     std::string audiotrack_file_name;
>     size_t begin;
>     size_t end;
>+    int gain;
>   };
> 
>   MultiEndCall(
>-      rtc::ArrayView<const Turn> timing, const std::string& audiotracks_path,
>+      rtc::ArrayView<const Turn> timing,
>+      const std::string& audiotracks_path,
>       std::unique_ptr<WavReaderAbstractFactory> wavreader_abstract_factory);
>   ~MultiEndCall();
> 
>   const std::set<std::string>& speaker_names() const { return speaker_names_; }
>   const std::map<std::string, std::unique_ptr<WavReaderInterface>>&
>-      audiotrack_readers() const { return audiotrack_readers_; }
>+  audiotrack_readers() const {
>+    return audiotrack_readers_;
>+  }
>   bool valid() const { return valid_; }
>   int sample_rate() const { return sample_rate_hz_; }
>   size_t total_duration_samples() const { return total_duration_samples_; }
>   const std::vector<SpeakingTurn>& speaking_turns() const {
>-      return speaking_turns_; }
>+    return speaking_turns_;
>+  }
> 
>  private:
>   // Finds unique speaker names.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/simulator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/simulator.cc
>index 84a9ef55c8e..c2fb7808e08 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/simulator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/simulator.cc
>@@ -10,17 +10,22 @@
> 
> #include "modules/audio_processing/test/conversational_speech/simulator.h"
> 
>+#include <math.h>
>+
>+#include <algorithm>
> #include <set>
> #include <utility>
> #include <vector>
> 
>+#include "absl/memory/memory.h"
> #include "api/array_view.h"
>+#include "common_audio/include/audio_util.h"
> #include "common_audio/wav_file.h"
> #include "modules/audio_processing/test/conversational_speech/wavreader_interface.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/logging.h"
>+#include "rtc_base/numerics/safe_conversions.h"
> #include "rtc_base/pathutils.h"
>-#include "rtc_base/ptr_util.h"
> 
> namespace webrtc {
> namespace test {
>@@ -33,28 +38,27 @@ using conversational_speech::WavReaderInterface;
> // Combines output path and speaker names to define the output file paths for
> // the near-end and far=end audio tracks.
> std::unique_ptr<std::map<std::string, SpeakerOutputFilePaths>>
>-    InitSpeakerOutputFilePaths(const std::set<std::string>& speaker_names,
>-                               const std::string& output_path) {
>+InitSpeakerOutputFilePaths(const std::set<std::string>& speaker_names,
>+                           const std::string& output_path) {
>   // Create map.
>-  auto speaker_output_file_paths_map = rtc::MakeUnique<
>-      std::map<std::string, SpeakerOutputFilePaths>>();
>+  auto speaker_output_file_paths_map =
>+      absl::make_unique<std::map<std::string, SpeakerOutputFilePaths>>();
> 
>   // Add near-end and far-end output paths into the map.
>   for (const auto& speaker_name : speaker_names) {
>-    const rtc::Pathname near_end_path(
>-        output_path, "s_" + speaker_name + "-near_end.wav");
>+    const rtc::Pathname near_end_path(output_path,
>+                                      "s_" + speaker_name + "-near_end.wav");
>     RTC_LOG(LS_VERBOSE) << "The near-end audio track will be created in "
>                         << near_end_path.pathname() << ".";
> 
>-    const rtc::Pathname far_end_path(
>-        output_path, "s_" + speaker_name + "-far_end.wav");
>+    const rtc::Pathname far_end_path(output_path,
>+                                     "s_" + speaker_name + "-far_end.wav");
>     RTC_LOG(LS_VERBOSE) << "The far-end audio track will be created in "
>                         << far_end_path.pathname() << ".";
> 
>     // Add to map.
>     speaker_output_file_paths_map->emplace(
>-        std::piecewise_construct,
>-        std::forward_as_tuple(speaker_name),
>+        std::piecewise_construct, std::forward_as_tuple(speaker_name),
>         std::forward_as_tuple(near_end_path.pathname(),
>                               far_end_path.pathname()));
>   }
>@@ -66,16 +70,13 @@ std::unique_ptr<std::map<std::string, SpeakerOutputFilePaths>>
> // output track of a speaker.
> class SpeakerWavWriters {
>  public:
>-  SpeakerWavWriters(
>-      const SpeakerOutputFilePaths& output_file_paths, int sample_rate)
>-          : near_end_wav_writer_(output_file_paths.near_end, sample_rate, 1u),
>-            far_end_wav_writer_(output_file_paths.far_end, sample_rate, 1u) {}
>-  WavWriter* near_end_wav_writer() {
>-    return &near_end_wav_writer_;
>-  }
>-  WavWriter* far_end_wav_writer() {
>-    return &far_end_wav_writer_;
>-  }
>+  SpeakerWavWriters(const SpeakerOutputFilePaths& output_file_paths,
>+                    int sample_rate)
>+      : near_end_wav_writer_(output_file_paths.near_end, sample_rate, 1u),
>+        far_end_wav_writer_(output_file_paths.far_end, sample_rate, 1u) {}
>+  WavWriter* near_end_wav_writer() { return &near_end_wav_writer_; }
>+  WavWriter* far_end_wav_writer() { return &far_end_wav_writer_; }
>+
>  private:
>   WavWriter near_end_wav_writer_;
>   WavWriter far_end_wav_writer_;
>@@ -84,18 +85,18 @@ class SpeakerWavWriters {
> // Initializes one WavWriter instance for each speaker and both the near-end and
> // far-end output tracks.
> std::unique_ptr<std::map<std::string, SpeakerWavWriters>>
>-    InitSpeakersWavWriters(const std::map<std::string, SpeakerOutputFilePaths>&
>-                           speaker_output_file_paths, int sample_rate) {
>+InitSpeakersWavWriters(const std::map<std::string, SpeakerOutputFilePaths>&
>+                           speaker_output_file_paths,
>+                       int sample_rate) {
>   // Create map.
>-  auto speaker_wav_writers_map = rtc::MakeUnique<
>-      std::map<std::string, SpeakerWavWriters>>();
>+  auto speaker_wav_writers_map =
>+      absl::make_unique<std::map<std::string, SpeakerWavWriters>>();
> 
>   // Add SpeakerWavWriters instance into the map.
>   for (auto it = speaker_output_file_paths.begin();
>-      it != speaker_output_file_paths.end(); ++it) {
>+       it != speaker_output_file_paths.end(); ++it) {
>     speaker_wav_writers_map->emplace(
>-        std::piecewise_construct,
>-        std::forward_as_tuple(it->first),
>+        std::piecewise_construct, std::forward_as_tuple(it->first),
>         std::forward_as_tuple(it->second, sample_rate));
>   }
> 
>@@ -107,17 +108,16 @@ std::unique_ptr<std::map<std::string, std::vector<int16_t>>> PreloadAudioTracks(
>     const std::map<std::string, std::unique_ptr<WavReaderInterface>>&
>         audiotrack_readers) {
>   // Create map.
>-  auto audiotracks_map = rtc::MakeUnique<
>-      std::map<std::string, std::vector<int16_t>>>();
>+  auto audiotracks_map =
>+      absl::make_unique<std::map<std::string, std::vector<int16_t>>>();
> 
>   // Add audio track vectors.
>   for (auto it = audiotrack_readers.begin(); it != audiotrack_readers.end();
>-      ++it) {
>+       ++it) {
>     // Add map entry.
>-    audiotracks_map->emplace(
>-        std::piecewise_construct,
>-        std::forward_as_tuple(it->first),
>-        std::forward_as_tuple(it->second->NumSamples()));
>+    audiotracks_map->emplace(std::piecewise_construct,
>+                             std::forward_as_tuple(it->first),
>+                             std::forward_as_tuple(it->second->NumSamples()));
> 
>     // Read samples.
>     it->second->ReadInt16Samples(audiotracks_map->at(it->first));
>@@ -131,7 +131,8 @@ std::unique_ptr<std::map<std::string, std::vector<int16_t>>> PreloadAudioTracks(
> // adds zeros as left padding. The padding corresponds to intervals during which
> // a speaker is not active.
> void PadLeftWriteChunk(rtc::ArrayView<const int16_t> source_samples,
>-                       size_t interval_begin, WavWriter* wav_writer) {
>+                       size_t interval_begin,
>+                       WavWriter* wav_writer) {
>   // Add left padding.
>   RTC_CHECK(wav_writer);
>   RTC_CHECK_GE(interval_begin, wav_writer->num_samples());
>@@ -158,16 +159,28 @@ void PadRightWrite(WavWriter* wav_writer, size_t pad_samples) {
>   }
> }
> 
>+void ScaleSignal(rtc::ArrayView<const int16_t> source_samples,
>+                 int gain,
>+                 rtc::ArrayView<int16_t> output_samples) {
>+  const float gain_linear = DbToRatio(gain);
>+  RTC_DCHECK_EQ(source_samples.size(), output_samples.size());
>+  std::transform(source_samples.begin(), source_samples.end(),
>+                 output_samples.begin(), [gain_linear](int16_t x) -> int16_t {
>+                   return rtc::saturated_cast<int16_t>(x * gain_linear);
>+                 });
>+}
>+
> }  // namespace
> 
> namespace conversational_speech {
> 
> std::unique_ptr<std::map<std::string, SpeakerOutputFilePaths>> Simulate(
>-    const MultiEndCall& multiend_call, const std::string& output_path) {
>+    const MultiEndCall& multiend_call,
>+    const std::string& output_path) {
>   // Set output file paths and initialize wav writers.
>   const auto& speaker_names = multiend_call.speaker_names();
>-  auto speaker_output_file_paths = InitSpeakerOutputFilePaths(
>-      speaker_names, output_path);
>+  auto speaker_output_file_paths =
>+      InitSpeakerOutputFilePaths(speaker_names, output_path);
>   auto speakers_wav_writers = InitSpeakersWavWriters(
>       *speaker_output_file_paths, multiend_call.sample_rate());
> 
>@@ -185,21 +198,23 @@ std::unique_ptr<std::map<std::string, SpeakerOutputFilePaths>> Simulate(
>   // Write near-end and far-end output tracks.
>   for (const auto& speaking_turn : multiend_call.speaking_turns()) {
>     const std::string& active_speaker_name = speaking_turn.speaker_name;
>-    auto source_audiotrack = audiotracks->at(
>-        speaking_turn.audiotrack_file_name);
>+    const auto source_audiotrack =
>+        audiotracks->at(speaking_turn.audiotrack_file_name);
>+    std::vector<int16_t> scaled_audiotrack(source_audiotrack.size());
>+    ScaleSignal(source_audiotrack, speaking_turn.gain, scaled_audiotrack);
> 
>     // Write active speaker's chunk to active speaker's near-end.
>-    PadLeftWriteChunk(source_audiotrack, speaking_turn.begin,
>-                      speakers_wav_writers->at(
>-                          active_speaker_name).near_end_wav_writer());
>+    PadLeftWriteChunk(
>+        scaled_audiotrack, speaking_turn.begin,
>+        speakers_wav_writers->at(active_speaker_name).near_end_wav_writer());
> 
>     // Write active speaker's chunk to other participants' far-ends.
>     for (const std::string& speaker_name : speaker_names) {
>       if (speaker_name == active_speaker_name)
>         continue;
>-      PadLeftWriteChunk(source_audiotrack, speaking_turn.begin,
>-                        speakers_wav_writers->at(
>-                            speaker_name).far_end_wav_writer());
>+      PadLeftWriteChunk(
>+          scaled_audiotrack, speaking_turn.begin,
>+          speakers_wav_writers->at(speaker_name).far_end_wav_writer());
>     }
>   }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/simulator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/simulator.h
>index 671c3b9c735..743db420f5f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/simulator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/simulator.h
>@@ -26,16 +26,16 @@ namespace conversational_speech {
> struct SpeakerOutputFilePaths {
>   SpeakerOutputFilePaths(const std::string& new_near_end,
>                          const std::string& new_far_end)
>-      : near_end(new_near_end),
>-        far_end(new_far_end) {}
>+      : near_end(new_near_end), far_end(new_far_end) {}
>   // Paths to the near-end and far-end audio track files.
>   const std::string near_end;
>   const std::string far_end;
> };
> 
> // Generates the near-end and far-end audio track pairs for each speaker.
>-std::unique_ptr<std::map<std::string, SpeakerOutputFilePaths>>
>-    Simulate(const MultiEndCall& multiend_call, const std::string& output_path);
>+std::unique_ptr<std::map<std::string, SpeakerOutputFilePaths>> Simulate(
>+    const MultiEndCall& multiend_call,
>+    const std::string& output_path);
> 
> }  // namespace conversational_speech
> }  // namespace test
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/timing.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/timing.cc
>index 773a42ebd75..6c19f328075 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/timing.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/timing.cc
>@@ -19,10 +19,10 @@ namespace webrtc {
> namespace test {
> namespace conversational_speech {
> 
>-bool Turn::operator==(const Turn &b) const {
>+bool Turn::operator==(const Turn& b) const {
>   return b.speaker_name == speaker_name &&
>-         b.audiotrack_file_name == audiotrack_file_name &&
>-         b.offset == offset;
>+         b.audiotrack_file_name == audiotrack_file_name && b.offset == offset &&
>+         b.gain == gain;
> }
> 
> std::vector<Turn> LoadTiming(const std::string& timing_filepath) {
>@@ -30,8 +30,13 @@ std::vector<Turn> LoadTiming(const std::string& timing_filepath) {
>   auto parse_line = [](const std::string& line) {
>     std::vector<std::string> fields;
>     rtc::split(line, ' ', &fields);
>-    RTC_CHECK_EQ(fields.size(), 3);
>-    return Turn(fields[0], fields[1], std::atol(fields[2].c_str()));
>+    RTC_CHECK_GE(fields.size(), 3);
>+    RTC_CHECK_LE(fields.size(), 4);
>+    int gain = 0;
>+    if (fields.size() == 4) {
>+      gain = std::atof(fields[3].c_str());
>+    }
>+    return Turn(fields[0], fields[1], std::atol(fields[2].c_str()), gain);
>   };
> 
>   // Init.
>@@ -55,8 +60,8 @@ void SaveTiming(const std::string& timing_filepath,
>   std::ofstream outfile(timing_filepath);
>   RTC_CHECK(outfile.is_open());
>   for (const Turn& turn : timing) {
>-    outfile << turn.speaker_name << " " << turn.audiotrack_file_name
>-        << " " << turn.offset << std::endl;
>+    outfile << turn.speaker_name << " " << turn.audiotrack_file_name << " "
>+            << turn.offset << " " << turn.gain << std::endl;
>   }
>   outfile.close();
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/timing.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/timing.h
>index dc433618150..812f3d55cc6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/timing.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/timing.h
>@@ -20,16 +20,20 @@ namespace webrtc {
> namespace test {
> namespace conversational_speech {
> 
>-struct Turn{
>-  Turn(std::string new_speaker_name, std::string new_audiotrack_file_name,
>-       int new_offset)
>+struct Turn {
>+  Turn(std::string new_speaker_name,
>+       std::string new_audiotrack_file_name,
>+       int new_offset,
>+       int gain)
>       : speaker_name(new_speaker_name),
>         audiotrack_file_name(new_audiotrack_file_name),
>-        offset(new_offset) {}
>-  bool operator==(const Turn &b) const;
>+        offset(new_offset),
>+        gain(gain) {}
>+  bool operator==(const Turn& b) const;
>   std::string speaker_name;
>   std::string audiotrack_file_name;
>   int offset;
>+  int gain;
> };
> 
> // Loads a list of turns from a file.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/wavreader_factory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/wavreader_factory.cc
>index 8342d3fed83..01e3c9412f5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/wavreader_factory.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/wavreader_factory.cc
>@@ -15,7 +15,6 @@
> #include "api/array_view.h"
> #include "common_audio/wav_file.h"
> #include "rtc_base/checks.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> namespace test {
>@@ -37,17 +36,11 @@ class WavReaderAdaptor final : public WavReaderInterface {
>     return wav_reader_.ReadSamples(samples.size(), samples.begin());
>   }
> 
>-  int SampleRate() const override {
>-    return wav_reader_.sample_rate();
>-  }
>+  int SampleRate() const override { return wav_reader_.sample_rate(); }
> 
>-  size_t NumChannels() const override {
>-    return wav_reader_.num_channels();
>-  }
>+  size_t NumChannels() const override { return wav_reader_.num_channels(); }
> 
>-  size_t NumSamples() const override {
>-    return wav_reader_.num_samples();
>-  }
>+  size_t NumSamples() const override { return wav_reader_.num_samples(); }
> 
>  private:
>   WavReader wav_reader_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/wavreader_factory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/wavreader_factory.h
>index e168d0d71fa..2f86bf5ae07 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/wavreader_factory.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/wavreader_factory.h
>@@ -25,8 +25,8 @@ class WavReaderFactory : public WavReaderAbstractFactory {
>  public:
>   WavReaderFactory();
>   ~WavReaderFactory() override;
>-  std::unique_ptr<WavReaderInterface> Create(const std::string& filepath) const
>-      override;
>+  std::unique_ptr<WavReaderInterface> Create(
>+      const std::string& filepath) const override;
> };
> 
> }  // namespace conversational_speech
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/wavreader_interface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/wavreader_interface.h
>index 3aea273ad26..c74f6394610 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/wavreader_interface.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/conversational_speech/wavreader_interface.h
>@@ -14,7 +14,6 @@
> #include <stddef.h>
> 
> #include "api/array_view.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> namespace test {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/debug_dump_replayer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/debug_dump_replayer.cc
>index 262f347de3b..a06c76e32aa 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/debug_dump_replayer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/debug_dump_replayer.cc
>@@ -23,8 +23,8 @@ void MaybeResetBuffer(std::unique_ptr<ChannelBuffer<float>>* buffer,
>   auto& buffer_ref = *buffer;
>   if (!buffer_ref.get() || buffer_ref->num_frames() != config.num_frames() ||
>       buffer_ref->num_channels() != config.num_channels()) {
>-    buffer_ref.reset(new ChannelBuffer<float>(config.num_frames(),
>-                                             config.num_channels()));
>+    buffer_ref.reset(
>+        new ChannelBuffer<float>(config.num_frames(), config.num_channels()));
>   }
> }
> 
>@@ -49,9 +49,9 @@ bool DebugDumpReplayer::SetDumpFile(const std::string& filename) {
> }
> 
> // Get next event that has not run.
>-rtc::Optional<audioproc::Event> DebugDumpReplayer::GetNextEvent() const {
>+absl::optional<audioproc::Event> DebugDumpReplayer::GetNextEvent() const {
>   if (!has_next_event_)
>-    return rtc::nullopt;
>+    return absl::nullopt;
>   else
>     return next_event_;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/debug_dump_replayer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/debug_dump_replayer.h
>index 4cd961dd7f8..aa5d72735a3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/debug_dump_replayer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/debug_dump_replayer.h
>@@ -34,7 +34,7 @@ class DebugDumpReplayer {
>   bool SetDumpFile(const std::string& filename);
> 
>   // Return next event.
>-  rtc::Optional<audioproc::Event> GetNextEvent() const;
>+  absl::optional<audioproc::Event> GetNextEvent() const;
> 
>   // Run the next event. Returns true if succeeded.
>   bool RunNextEvent();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/debug_dump_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/debug_dump_test.cc
>index 69231b2b66f..4e2943395e4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/debug_dump_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/debug_dump_test.cc
>@@ -14,6 +14,7 @@
> #include <string>
> #include <vector>
> 
>+#include "api/audio/echo_canceller3_factory.h"
> #include "modules/audio_coding/neteq/tools/resample_input_audio_file.h"
> #include "modules/audio_processing/aec_dump/aec_dump_factory.h"
> #include "modules/audio_processing/test/debug_dump_replayer.h"
>@@ -32,8 +33,8 @@ void MaybeResetBuffer(std::unique_ptr<ChannelBuffer<float>>* buffer,
>   auto& buffer_ref = *buffer;
>   if (!buffer_ref.get() || buffer_ref->num_frames() != config.num_frames() ||
>       buffer_ref->num_channels() != config.num_channels()) {
>-    buffer_ref.reset(new ChannelBuffer<float>(config.num_frames(),
>-                                             config.num_channels()));
>+    buffer_ref.reset(
>+        new ChannelBuffer<float>(config.num_frames(), config.num_channels()));
>   }
> }
> 
>@@ -86,7 +87,8 @@ class DebugDumpGenerator {
>   AudioProcessing* apm() const { return apm_.get(); }
> 
>  private:
>-  static void ReadAndDeinterleave(ResampleInputAudioFile* audio, int channels,
>+  static void ReadAndDeinterleave(ResampleInputAudioFile* audio,
>+                                  int channels,
>                                   const StreamConfig& config,
>                                   float* const* buffer);
> 
>@@ -226,11 +228,10 @@ void DebugDumpGenerator::Process(size_t num_blocks) {
>                  apm_->ProcessStream(input_->channels(), input_config_,
>                                      output_config_, output_->channels()));
> 
>-    RTC_CHECK_EQ(AudioProcessing::kNoError,
>-                 apm_->ProcessReverseStream(reverse_->channels(),
>-                                            reverse_config_,
>-                                            reverse_config_,
>-                                            reverse_->channels()));
>+    RTC_CHECK_EQ(
>+        AudioProcessing::kNoError,
>+        apm_->ProcessReverseStream(reverse_->channels(), reverse_config_,
>+                                   reverse_config_, reverse_->channels()));
>   }
> }
> 
>@@ -274,8 +275,8 @@ class DebugDumpTest : public ::testing::Test {
> void DebugDumpTest::VerifyDebugDump(const std::string& in_filename) {
>   ASSERT_TRUE(debug_dump_replayer_.SetDumpFile(in_filename));
> 
>-  while (const rtc::Optional<audioproc::Event> event =
>-      debug_dump_replayer_.GetNextEvent()) {
>+  while (const absl::optional<audioproc::Event> event =
>+             debug_dump_replayer_.GetNextEvent()) {
>     debug_dump_replayer_.RunNextEvent();
>     if (event->type() == audioproc::Event::STREAM) {
>       const audioproc::Stream* msg = &event->stream();
>@@ -287,9 +288,9 @@ void DebugDumpTest::VerifyDebugDump(const std::string& in_filename) {
>       ASSERT_EQ(output_config.num_frames() * sizeof(float),
>                 msg->output_channel(0).size());
>       for (int i = 0; i < msg->output_channel_size(); ++i) {
>-        ASSERT_EQ(0, memcmp(output->channels()[i],
>-                            msg->output_channel(i).data(),
>-                            msg->output_channel(i).size()));
>+        ASSERT_EQ(0,
>+                  memcmp(output->channels()[i], msg->output_channel(i).data(),
>+                         msg->output_channel(i).size()));
>       }
>     }
>   }
>@@ -387,7 +388,7 @@ TEST_F(DebugDumpTest, VerifyRefinedAdaptiveFilterExperimentalString) {
> 
>   ASSERT_TRUE(debug_dump_replayer_.SetDumpFile(generator.dump_file_name()));
> 
>-  while (const rtc::Optional<audioproc::Event> event =
>+  while (const absl::optional<audioproc::Event> event =
>              debug_dump_replayer_.GetNextEvent()) {
>     debug_dump_replayer_.RunNextEvent();
>     if (event->type() == audioproc::Event::CONFIG) {
>@@ -415,7 +416,7 @@ TEST_F(DebugDumpTest, VerifyCombinedExperimentalStringInclusive) {
> 
>   ASSERT_TRUE(debug_dump_replayer_.SetDumpFile(generator.dump_file_name()));
> 
>-  while (const rtc::Optional<audioproc::Event> event =
>+  while (const absl::optional<audioproc::Event> event =
>              debug_dump_replayer_.GetNextEvent()) {
>     debug_dump_replayer_.RunNextEvent();
>     if (event->type() == audioproc::Event::CONFIG) {
>@@ -443,7 +444,7 @@ TEST_F(DebugDumpTest, VerifyCombinedExperimentalStringExclusive) {
> 
>   ASSERT_TRUE(debug_dump_replayer_.SetDumpFile(generator.dump_file_name()));
> 
>-  while (const rtc::Optional<audioproc::Event> event =
>+  while (const absl::optional<audioproc::Event> event =
>              debug_dump_replayer_.GetNextEvent()) {
>     debug_dump_replayer_.RunNextEvent();
>     if (event->type() == audioproc::Event::CONFIG) {
>@@ -471,7 +472,7 @@ TEST_F(DebugDumpTest, VerifyAec3ExperimentalString) {
> 
>   ASSERT_TRUE(debug_dump_replayer_.SetDumpFile(generator.dump_file_name()));
> 
>-  while (const rtc::Optional<audioproc::Event> event =
>+  while (const absl::optional<audioproc::Event> event =
>              debug_dump_replayer_.GetNextEvent()) {
>     debug_dump_replayer_.RunNextEvent();
>     if (event->type() == audioproc::Event::CONFIG) {
>@@ -483,31 +484,6 @@ TEST_F(DebugDumpTest, VerifyAec3ExperimentalString) {
>   }
> }
> 
>-TEST_F(DebugDumpTest, VerifyLevelControllerExperimentalString) {
>-  Config config;
>-  AudioProcessing::Config apm_config;
>-  apm_config.level_controller.enabled = true;
>-  DebugDumpGenerator generator(config, apm_config);
>-  generator.StartRecording();
>-  generator.Process(100);
>-  generator.StopRecording();
>-
>-  DebugDumpReplayer debug_dump_replayer_;
>-
>-  ASSERT_TRUE(debug_dump_replayer_.SetDumpFile(generator.dump_file_name()));
>-
>-  while (const rtc::Optional<audioproc::Event> event =
>-             debug_dump_replayer_.GetNextEvent()) {
>-    debug_dump_replayer_.RunNextEvent();
>-    if (event->type() == audioproc::Event::CONFIG) {
>-      const audioproc::Config* msg = &event->config();
>-      ASSERT_TRUE(msg->has_experiments_description());
>-      EXPECT_PRED_FORMAT2(testing::IsSubstring, "LevelController",
>-                          msg->experiments_description().c_str());
>-    }
>-  }
>-}
>-
> TEST_F(DebugDumpTest, VerifyAgcClippingLevelExperimentalString) {
>   Config config;
>   // Arbitrarily set clipping gain to 17, which will never be the default.
>@@ -521,7 +497,7 @@ TEST_F(DebugDumpTest, VerifyAgcClippingLevelExperimentalString) {
> 
>   ASSERT_TRUE(debug_dump_replayer_.SetDumpFile(generator.dump_file_name()));
> 
>-  while (const rtc::Optional<audioproc::Event> event =
>+  while (const absl::optional<audioproc::Event> event =
>              debug_dump_replayer_.GetNextEvent()) {
>     debug_dump_replayer_.RunNextEvent();
>     if (event->type() == audioproc::Event::CONFIG) {
>@@ -544,7 +520,7 @@ TEST_F(DebugDumpTest, VerifyEmptyExperimentalString) {
> 
>   ASSERT_TRUE(debug_dump_replayer_.SetDumpFile(generator.dump_file_name()));
> 
>-  while (const rtc::Optional<audioproc::Event> event =
>+  while (const absl::optional<audioproc::Event> event =
>              debug_dump_replayer_.GetNextEvent()) {
>     debug_dump_replayer_.RunNextEvent();
>     if (event->type() == audioproc::Event::CONFIG) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/fake_recording_device.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/fake_recording_device.cc
>index aee3dcef61d..ee134ea0874 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/fake_recording_device.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/fake_recording_device.cc
>@@ -12,16 +12,18 @@
> 
> #include <algorithm>
> 
>+#include "absl/memory/memory.h"
>+#include "absl/types/optional.h"
>+#include "modules/audio_processing/agc/gain_map_internal.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/ptr_util.h"
>+#include "rtc_base/numerics/safe_conversions.h"
>+#include "rtc_base/numerics/safe_minmax.h"
> 
> namespace webrtc {
> namespace test {
> 
> namespace {
> 
>-constexpr int16_t kInt16SampleMin = -32768;
>-constexpr int16_t kInt16SampleMax = 32767;
> constexpr float kFloatSampleMin = -32768.f;
> constexpr float kFloatSampleMax = 32767.0f;
> 
>@@ -43,7 +45,7 @@ class FakeRecordingDeviceWorker {
>   // Mic level to simulate.
>   int mic_level_;
>   // Optional mic level to undo.
>-  rtc::Optional<int> undo_mic_level_;
>+  absl::optional<int> undo_mic_level_;
> };
> 
> namespace {
>@@ -75,11 +77,7 @@ class FakeRecordingDeviceLinear final : public FakeRecordingDeviceWorker {
>     const float divisor =
>         (undo_mic_level_ && *undo_mic_level_ > 0) ? *undo_mic_level_ : 255.f;
>     for (size_t i = 0; i < number_of_samples; ++i) {
>-      data[i] =
>-          std::max(kInt16SampleMin,
>-                   std::min(kInt16SampleMax,
>-                            static_cast<int16_t>(static_cast<float>(data[i]) *
>-                                                 mic_level_ / divisor)));
>+      data[i] = rtc::saturated_cast<int16_t>(data[i] * mic_level_ / divisor);
>     }
>   }
>   void ModifyBufferFloat(ChannelBuffer<float>* buffer) override {
>@@ -90,9 +88,47 @@ class FakeRecordingDeviceLinear final : public FakeRecordingDeviceWorker {
>     for (size_t c = 0; c < buffer->num_channels(); ++c) {
>       for (size_t i = 0; i < buffer->num_frames(); ++i) {
>         buffer->channels()[c][i] =
>-            std::max(kFloatSampleMin,
>-                     std::min(kFloatSampleMax,
>-                              buffer->channels()[c][i] * mic_level_ / divisor));
>+            rtc::SafeClamp(buffer->channels()[c][i] * mic_level_ / divisor,
>+                           kFloatSampleMin, kFloatSampleMax);
>+      }
>+    }
>+  }
>+};
>+
>+float ComputeAgc1LinearFactor(const absl::optional<int>& undo_mic_level,
>+                              int mic_level) {
>+  // If an undo level is specified, virtually restore the unmodified
>+  // microphone level; otherwise simulate the mic gain only.
>+  const int undo_level =
>+      (undo_mic_level && *undo_mic_level > 0) ? *undo_mic_level : 100;
>+  return DbToRatio(kGainMap[mic_level] - kGainMap[undo_level]);
>+}
>+
>+// Roughly dB-scale fake recording device. Valid levels are [0, 255]. The mic
>+// applies a gain from kGainMap in agc/gain_map_internal.h.
>+class FakeRecordingDeviceAgc1 final : public FakeRecordingDeviceWorker {
>+ public:
>+  explicit FakeRecordingDeviceAgc1(const int initial_mic_level)
>+      : FakeRecordingDeviceWorker(initial_mic_level) {}
>+  ~FakeRecordingDeviceAgc1() override = default;
>+  void ModifyBufferInt16(AudioFrame* buffer) override {
>+    const float scaling_factor =
>+        ComputeAgc1LinearFactor(undo_mic_level_, mic_level_);
>+    const size_t number_of_samples =
>+        buffer->samples_per_channel_ * buffer->num_channels_;
>+    int16_t* data = buffer->mutable_data();
>+    for (size_t i = 0; i < number_of_samples; ++i) {
>+      data[i] = rtc::saturated_cast<int16_t>(data[i] * scaling_factor);
>+    }
>+  }
>+  void ModifyBufferFloat(ChannelBuffer<float>* buffer) override {
>+    const float scaling_factor =
>+        ComputeAgc1LinearFactor(undo_mic_level_, mic_level_);
>+    for (size_t c = 0; c < buffer->num_channels(); ++c) {
>+      for (size_t i = 0; i < buffer->num_frames(); ++i) {
>+        buffer->channels()[c][i] =
>+            rtc::SafeClamp(buffer->channels()[c][i] * scaling_factor,
>+                           kFloatSampleMin, kFloatSampleMax);
>       }
>     }
>   }
>@@ -104,10 +140,14 @@ FakeRecordingDevice::FakeRecordingDevice(int initial_mic_level,
>                                          int device_kind) {
>   switch (device_kind) {
>     case 0:
>-      worker_ = rtc::MakeUnique<FakeRecordingDeviceIdentity>(initial_mic_level);
>+      worker_ =
>+          absl::make_unique<FakeRecordingDeviceIdentity>(initial_mic_level);
>       break;
>     case 1:
>-      worker_ = rtc::MakeUnique<FakeRecordingDeviceLinear>(initial_mic_level);
>+      worker_ = absl::make_unique<FakeRecordingDeviceLinear>(initial_mic_level);
>+      break;
>+    case 2:
>+      worker_ = absl::make_unique<FakeRecordingDeviceAgc1>(initial_mic_level);
>       break;
>     default:
>       RTC_NOTREACHED();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/fake_recording_device.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/fake_recording_device.h
>index b1e37a331d9..13f94a7e224 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/fake_recording_device.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/fake_recording_device.h
>@@ -16,8 +16,8 @@
> #include <vector>
> 
> #include "api/array_view.h"
>+#include "api/audio/audio_frame.h"
> #include "common_audio/channel_buffer.h"
>-#include "modules/include/module_common_types.h"
> #include "rtc_base/checks.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/fake_recording_device_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/fake_recording_device_unittest.cc
>index 504459a3c64..65940f24f95 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/fake_recording_device_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/fake_recording_device_unittest.cc
>@@ -14,9 +14,9 @@
> #include <string>
> #include <vector>
> 
>+#include "absl/memory/memory.h"
> #include "api/array_view.h"
> #include "modules/audio_processing/test/fake_recording_device.h"
>-#include "rtc_base/ptr_util.h"
> #include "test/gtest.h"
> 
> namespace webrtc {
>@@ -27,7 +27,7 @@ constexpr int kInitialMicLevel = 100;
> 
> // TODO(alessiob): Add new fake recording device kind values here as they are
> // added in FakeRecordingDevice::FakeRecordingDevice.
>-const std::vector<int> kFakeRecDeviceKinds = {0, 1};
>+const std::vector<int> kFakeRecDeviceKinds = {0, 1, 2};
> 
> const std::vector<std::vector<float>> kTestMultiChannelSamples{
>     std::vector<float>{-10.f, -1.f, -0.1f, 0.f, 0.1f, 1.f, 10.f}};
>@@ -47,7 +47,7 @@ void WritesDataIntoChannelBuffer(const std::vector<std::vector<float>>& data,
> std::unique_ptr<ChannelBuffer<float>> CreateChannelBufferWithData(
>     const std::vector<std::vector<float>>& data) {
>   auto buff =
>-      rtc::MakeUnique<ChannelBuffer<float>>(data[0].size(), data.size());
>+      absl::make_unique<ChannelBuffer<float>>(data[0].size(), data.size());
>   WritesDataIntoChannelBuffer(data, buff.get());
>   return buff;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/performance_timer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/performance_timer.h
>index 1c862dc78ef..b6e0da70d85 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/performance_timer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/performance_timer.h
>@@ -13,7 +13,7 @@
> 
> #include <vector>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> #include "system_wrappers/include/clock.h"
> 
> namespace webrtc {
>@@ -37,7 +37,7 @@ class PerformanceTimer {
> 
>  private:
>   webrtc::Clock* clock_;
>-  rtc::Optional<int64_t> start_timestamp_us_;
>+  absl::optional<int64_t> start_timestamp_us_;
>   std::vector<int64_t> timestamps_us_;
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/protobuf_utils.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/protobuf_utils.cc
>index 3b623b915c8..c47f8ea40f5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/protobuf_utils.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/protobuf_utils.cc
>@@ -9,13 +9,13 @@
>  */
> 
> #include "modules/audio_processing/test/protobuf_utils.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include "rtc_base/system/arch.h"
> 
> namespace webrtc {
> 
> size_t ReadMessageBytesFromFile(FILE* file, std::unique_ptr<uint8_t[]>* bytes) {
>-  // The "wire format" for the size is little-endian. Assume we're running on
>-  // a little-endian machine.
>+// The "wire format" for the size is little-endian. Assume we're running on
>+// a little-endian machine.
> #ifndef WEBRTC_ARCH_LITTLE_ENDIAN
> #error "Need to convert messsage from little-endian."
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/py_quality_assessment/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/py_quality_assessment/BUILD.gn
>index a23151c010c..6a4e31f4a4c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/py_quality_assessment/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/py_quality_assessment/BUILD.gn
>@@ -137,6 +137,20 @@ rtc_executable("apm_vad") {
>   sources = [
>     "quality_assessment/apm_vad.cc",
>   ]
>+  deps = [
>+    "../..",
>+    "../../../..:webrtc_common",
>+    "../../../../common_audio",
>+    "../../../../rtc_base:rtc_base_approved",
>+    "../../../../system_wrappers:metrics_default",
>+    "../../vad",
>+  ]
>+}
>+
>+rtc_executable("sound_level") {
>+  sources = [
>+    "quality_assessment/sound_level.cc",
>+  ]
>   deps = [
>     "../..",
>     "../../../..:webrtc_common",
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment_optimize.py b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment_optimize.py
>index 7946fe2aeb4..3daf48248bf 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment_optimize.py
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/py_quality_assessment/apm_quality_assessment_optimize.py
>@@ -105,7 +105,7 @@ def _ConfigurationAndScores(data_frame, params,
>       else:
>         result['params'][param] = config_json['-' + param]
> 
>-    current_param_combination = param_combination( # pylint: disable=star-args
>+    current_param_combination = param_combination(
>         **config_optimize_params)
>     results[current_param_combination].append(result)
>   return results
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/annotations.py b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/annotations.py
>index 5c8c8bacbc4..ca5bd1f72cf 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/annotations.py
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/annotations.py
>@@ -181,7 +181,6 @@ class AudioAnnotationsExtractor(object):
>     ext_kwargs = {'extvad_conf-' + ext_vad:
>                   self._external_vads[ext_vad].GetVadOutput()
>                   for ext_vad in self._external_vads}
>-    # pylint: disable=star-args
>     np.savez_compressed(
>         file=os.path.join(
>             output_path,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/fake_external_vad.py b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/fake_external_vad.py
>old mode 100644
>new mode 100755
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/input_signal_creator.py b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/input_signal_creator.py
>index 5d97c3b2fc1..1feec47b4cd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/input_signal_creator.py
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/input_signal_creator.py
>@@ -61,7 +61,7 @@ class InputSignalCreator(object):
>       AudioSegment instance.
>     """
>     assert 0 < frequency <= 24000
>-    assert 0 < duration
>+    assert duration > 0
>     template = signal_processing.SignalProcessingUtils.GenerateSilence(duration)
>     return signal_processing.SignalProcessingUtils.GeneratePureTone(
>         template, frequency)
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/sound_level.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/sound_level.cc
>new file mode 100644
>index 00000000000..98cf84c888f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/sound_level.cc
>@@ -0,0 +1,125 @@
>+// Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+//
>+// Use of this source code is governed by a BSD-style license
>+// that can be found in the LICENSE file in the root of the source
>+// tree. An additional intellectual property rights grant can be found
>+// in the file PATENTS.  All contributing project authors may
>+// be found in the AUTHORS file in the root of the source tree.
>+
>+#include <algorithm>
>+#include <array>
>+#include <cmath>
>+#include <fstream>
>+
>+#include "common_audio/include/audio_util.h"
>+#include "common_audio/wav_file.h"
>+#include "rtc_base/flags.h"
>+#include "rtc_base/logging.h"
>+
>+namespace webrtc {
>+namespace test {
>+namespace {
>+
>+constexpr int kMaxSampleRate = 48000;
>+constexpr uint8_t kMaxFrameLenMs = 30;
>+constexpr size_t kMaxFrameLen = kMaxFrameLenMs * kMaxSampleRate / 1000;
>+
>+const double kOneDbReduction = DbToRatio(-1.0);
>+
>+DEFINE_string(i, "", "Input wav file");
>+DEFINE_string(oc, "", "Config output file");
>+DEFINE_string(ol, "", "Levels output file");
>+DEFINE_float(a, 5.f, "Attack (ms)");
>+DEFINE_float(d, 20.f, "Decay (ms)");
>+DEFINE_int(f, 10, "Frame length (ms)");
>+DEFINE_bool(help, false, "prints this message");
>+
>+int main(int argc, char* argv[]) {
>+  if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true)) {
>+    rtc::FlagList::Print(nullptr, false);
>+    return 1;
>+  }
>+  if (FLAG_help) {
>+    rtc::FlagList::Print(nullptr, false);
>+    return 0;
>+  }
>+
>+  // Check parameters.
>+  if (FLAG_f < 1 || FLAG_f > kMaxFrameLenMs) {
>+    RTC_LOG(LS_ERROR) << "Invalid frame length (min: 1, max: " << kMaxFrameLenMs
>+                      << ")";
>+    return 1;
>+  }
>+  if (FLAG_a < 0 || FLAG_d < 0) {
>+    RTC_LOG(LS_ERROR) << "Attack and decay must be non-negative";
>+    return 1;
>+  }
>+
>+  // Open wav input file and check properties.
>+  WavReader wav_reader(FLAG_i);
>+  if (wav_reader.num_channels() != 1) {
>+    RTC_LOG(LS_ERROR) << "Only mono wav files supported";
>+    return 1;
>+  }
>+  if (wav_reader.sample_rate() > kMaxSampleRate) {
>+    RTC_LOG(LS_ERROR) << "Beyond maximum sample rate (" << kMaxSampleRate
>+                      << ")";
>+    return 1;
>+  }
>+
>+  // Map from milliseconds to samples.
>+  const size_t audio_frame_length =
>+      rtc::CheckedDivExact(FLAG_f * wav_reader.sample_rate(), 1000);
>+  auto time_const = [](double c) {
>+    return std::pow(kOneDbReduction, FLAG_f / c);
>+  };
>+  const float attack = FLAG_a == 0.0 ? 0.0 : time_const(FLAG_a);
>+  const float decay = FLAG_d == 0.0 ? 0.0 : time_const(FLAG_d);
>+
>+  // Write config to file.
>+  std::ofstream out_config(FLAG_oc);
>+  out_config << "{"
>+             << "'frame_len_ms': " << FLAG_f << ", "
>+             << "'attack_ms': " << FLAG_a << ", "
>+             << "'decay_ms': " << FLAG_d << "}\n";
>+  out_config.close();
>+
>+  // Measure level frame-by-frame.
>+  std::ofstream out_levels(FLAG_ol, std::ofstream::binary);
>+  std::array<int16_t, kMaxFrameLen> samples;
>+  float level_prev = 0.f;
>+  while (true) {
>+    // Process frame.
>+    const auto read_samples =
>+        wav_reader.ReadSamples(audio_frame_length, samples.data());
>+    if (read_samples < audio_frame_length)
>+      break;  // EOF.
>+
>+    // Frame peak level.
>+    std::transform(samples.begin(), samples.begin() + audio_frame_length,
>+                   samples.begin(), [](int16_t s) { return std::abs(s); });
>+    const auto* peak_level =
>+        std::max_element(samples.begin(), samples.begin() + audio_frame_length);
>+    const float level_curr = static_cast<float>(*peak_level) / 32768.f;
>+
>+    // Temporal smoothing.
>+    auto smooth = [&level_prev, &level_curr](float c) {
>+      return (1.0 - c) * level_curr + c * level_prev;
>+    };
>+    level_prev = smooth(level_curr > level_prev ? attack : decay);
>+
>+    // Write output.
>+    out_levels.write(reinterpret_cast<const char*>(&level_prev), sizeof(float));
>+  }
>+  out_levels.close();
>+
>+  return 0;
>+}
>+
>+}  // namespace
>+}  // namespace test
>+}  // namespace webrtc
>+
>+int main(int argc, char* argv[]) {
>+  return webrtc::test::main(argc, argv);
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/test_utils.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/test_utils.cc
>index 42ba7ebb5af..9f1a4691d24 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/test_utils.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/test_utils.cc
>@@ -12,6 +12,7 @@
> 
> #include "modules/audio_processing/test/test_utils.h"
> #include "rtc_base/checks.h"
>+#include "rtc_base/system/arch.h"
> 
> namespace webrtc {
> 
>@@ -91,9 +92,9 @@ void WriteFloatData(const float* const* data,
>   }
>   // TODO(aluebs): Use ScaleToInt16Range() from audio_util
>   for (size_t i = 0; i < length; ++i) {
>-    buffer[i] = buffer[i] > 0 ?
>-                buffer[i] * std::numeric_limits<int16_t>::max() :
>-                -buffer[i] * std::numeric_limits<int16_t>::min();
>+    buffer[i] = buffer[i] > 0
>+                    ? buffer[i] * std::numeric_limits<int16_t>::max()
>+                    : -buffer[i] * std::numeric_limits<int16_t>::min();
>   }
>   if (wav_file) {
>     wav_file->WriteSamples(buffer.get(), length);
>@@ -113,11 +114,10 @@ size_t SamplesFromRate(int rate) {
>   return static_cast<size_t>(AudioProcessing::kChunkSizeMs * rate / 1000);
> }
> 
>-void SetFrameSampleRate(AudioFrame* frame,
>-                        int sample_rate_hz) {
>+void SetFrameSampleRate(AudioFrame* frame, int sample_rate_hz) {
>   frame->sample_rate_hz_ = sample_rate_hz;
>-  frame->samples_per_channel_ = AudioProcessing::kChunkSizeMs *
>-      sample_rate_hz / 1000;
>+  frame->samples_per_channel_ =
>+      AudioProcessing::kChunkSizeMs * sample_rate_hz / 1000;
> }
> 
> AudioProcessing::ChannelLayout LayoutFromChannels(size_t num_channels) {
>@@ -132,27 +132,4 @@ AudioProcessing::ChannelLayout LayoutFromChannels(size_t num_channels) {
>   }
> }
> 
>-std::vector<Point> ParseArrayGeometry(const std::string& mic_positions) {
>-  const std::vector<float> values = ParseList<float>(mic_positions);
>-  const size_t num_mics =
>-      rtc::CheckedDivExact(values.size(), static_cast<size_t>(3));
>-  RTC_CHECK_GT(num_mics, 0) << "mic_positions is not large enough.";
>-
>-  std::vector<Point> result;
>-  result.reserve(num_mics);
>-  for (size_t i = 0; i < values.size(); i += 3) {
>-    result.push_back(Point(values[i + 0], values[i + 1], values[i + 2]));
>-  }
>-
>-  return result;
>-}
>-
>-std::vector<Point> ParseArrayGeometry(const std::string& mic_positions,
>-                                      size_t num_mics) {
>-  std::vector<Point> result = ParseArrayGeometry(mic_positions);
>-  RTC_CHECK_EQ(result.size(), num_mics)
>-      << "Could not parse mic_positions or incorrect number of points.";
>-  return result;
>-}
>-
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/test_utils.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/test_utils.h
>index 57dc7b37972..6e4154ad6d4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/test_utils.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/test_utils.h
>@@ -18,10 +18,10 @@
> #include <string>
> #include <vector>
> 
>+#include "api/audio/audio_frame.h"
> #include "common_audio/channel_buffer.h"
> #include "common_audio/wav_file.h"
> #include "modules/audio_processing/include/audio_processing.h"
>-#include "modules/include/module_common_types.h"
> #include "rtc_base/constructormagic.h"
> 
> namespace webrtc {
>@@ -91,8 +91,7 @@ FILE* OpenFile(const std::string& filename, const char* mode);
> 
> size_t SamplesFromRate(int rate);
> 
>-void SetFrameSampleRate(AudioFrame* frame,
>-                        int sample_rate_hz);
>+void SetFrameSampleRate(AudioFrame* frame, int sample_rate_hz);
> 
> template <typename T>
> void SetContainerFormat(int sample_rate_hz,
>@@ -130,29 +129,19 @@ float ComputeSNR(const T* ref, const T* test, size_t length, float* variance) {
> 
> // Returns a vector<T> parsed from whitespace delimited values in to_parse,
> // or an empty vector if the string could not be parsed.
>-template<typename T>
>+template <typename T>
> std::vector<T> ParseList(const std::string& to_parse) {
>   std::vector<T> values;
> 
>   std::istringstream str(to_parse);
>   std::copy(
>-      std::istream_iterator<T>(str),
>-      std::istream_iterator<T>(),
>+      std::istream_iterator<T>(str),  // no-presubmit-check TODO(webrtc:8982)
>+      std::istream_iterator<T>(),     // no-presubmit-check TODO(webrtc:8982)
>       std::back_inserter(values));
> 
>   return values;
> }
> 
>-// Parses the array geometry from the command line.
>-//
>-// If a vector with size != num_mics is returned, an error has occurred and an
>-// appropriate error message has been printed to stdout.
>-std::vector<Point> ParseArrayGeometry(const std::string& mic_positions,
>-                                      size_t num_mics);
>-
>-// Same as above, but without the num_mics check for when it isn't available.
>-std::vector<Point> ParseArrayGeometry(const std::string& mic_positions);
>-
> }  // namespace webrtc
> 
> #endif  // MODULES_AUDIO_PROCESSING_TEST_TEST_UTILS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/wav_based_simulator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/wav_based_simulator.cc
>index f53d1e5d8d9..908a2e6f2a5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/wav_based_simulator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/wav_based_simulator.cc
>@@ -53,8 +53,10 @@ WavBasedSimulator::GetCustomEventChain(const std::string& filename) {
>   return call_chain;
> }
> 
>-WavBasedSimulator::WavBasedSimulator(const SimulationSettings& settings)
>-      : AudioProcessingSimulator(settings) {}
>+WavBasedSimulator::WavBasedSimulator(
>+    const SimulationSettings& settings,
>+    std::unique_ptr<AudioProcessingBuilder> ap_builder)
>+    : AudioProcessingSimulator(settings, std::move(ap_builder)) {}
> 
> WavBasedSimulator::~WavBasedSimulator() = default;
> 
>@@ -72,9 +74,11 @@ void WavBasedSimulator::PrepareProcessStreamCall() {
>   }
>   ap_->set_stream_key_pressed(settings_.use_ts && (*settings_.use_ts));
> 
>-  RTC_CHECK_EQ(AudioProcessing::kNoError,
>-               ap_->set_stream_delay_ms(
>-                   settings_.stream_delay ? *settings_.stream_delay : 0));
>+  if (!settings_.use_stream_delay || *settings_.use_stream_delay) {
>+    RTC_CHECK_EQ(AudioProcessing::kNoError,
>+                 ap_->set_stream_delay_ms(
>+                     settings_.stream_delay ? *settings_.stream_delay : 0));
>+  }
> 
>   ap_->echo_cancellation()->set_stream_drift_samples(
>       settings_.stream_drift_samples ? *settings_.stream_drift_samples : 0);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/wav_based_simulator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/wav_based_simulator.h
>index febcffb62c2..0294e89ee6a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/wav_based_simulator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/wav_based_simulator.h
>@@ -23,7 +23,8 @@ namespace test {
> // Used to perform an audio processing simulation from wav files.
> class WavBasedSimulator final : public AudioProcessingSimulator {
>  public:
>-  explicit WavBasedSimulator(const SimulationSettings& settings);
>+  WavBasedSimulator(const SimulationSettings& settings,
>+                    std::unique_ptr<AudioProcessingBuilder> ap_builder);
>   ~WavBasedSimulator() override;
> 
>   // Processes the WAV input.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/three_band_filter_bank.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/three_band_filter_bank.cc
>index f5a319bc542..dbbfc283c58 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/three_band_filter_bank.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/three_band_filter_bank.cc
>@@ -68,19 +68,19 @@ const size_t kNumCoeffs = 4;
> // A Kaiser window is used because of its flexibility and the alpha is set to
> // 3.5, since that sets a stop band attenuation of 40dB ensuring a fast
> // transition.
>-const float kLowpassCoeffs[kNumBands * kSparsity][kNumCoeffs] =
>-    {{-0.00047749f, -0.00496888f, +0.16547118f, +0.00425496f},
>-     {-0.00173287f, -0.01585778f, +0.14989004f, +0.00994113f},
>-     {-0.00304815f, -0.02536082f, +0.12154542f, +0.01157993f},
>-     {-0.00383509f, -0.02982767f, +0.08543175f, +0.00983212f},
>-     {-0.00346946f, -0.02587886f, +0.04760441f, +0.00607594f},
>-     {-0.00154717f, -0.01136076f, +0.01387458f, +0.00186353f},
>-     {+0.00186353f, +0.01387458f, -0.01136076f, -0.00154717f},
>-     {+0.00607594f, +0.04760441f, -0.02587886f, -0.00346946f},
>-     {+0.00983212f, +0.08543175f, -0.02982767f, -0.00383509f},
>-     {+0.01157993f, +0.12154542f, -0.02536082f, -0.00304815f},
>-     {+0.00994113f, +0.14989004f, -0.01585778f, -0.00173287f},
>-     {+0.00425496f, +0.16547118f, -0.00496888f, -0.00047749f}};
>+const float kLowpassCoeffs[kNumBands * kSparsity][kNumCoeffs] = {
>+    {-0.00047749f, -0.00496888f, +0.16547118f, +0.00425496f},
>+    {-0.00173287f, -0.01585778f, +0.14989004f, +0.00994113f},
>+    {-0.00304815f, -0.02536082f, +0.12154542f, +0.01157993f},
>+    {-0.00383509f, -0.02982767f, +0.08543175f, +0.00983212f},
>+    {-0.00346946f, -0.02587886f, +0.04760441f, +0.00607594f},
>+    {-0.00154717f, -0.01136076f, +0.01387458f, +0.00186353f},
>+    {+0.00186353f, +0.01387458f, -0.01136076f, -0.00154717f},
>+    {+0.00607594f, +0.04760441f, -0.02587886f, -0.00346946f},
>+    {+0.00983212f, +0.08543175f, -0.02982767f, -0.00383509f},
>+    {+0.01157993f, +0.12154542f, -0.02536082f, -0.00304815f},
>+    {+0.00994113f, +0.14989004f, -0.01585778f, -0.00173287f},
>+    {+0.00425496f, +0.16547118f, -0.00496888f, -0.00047749f}};
> 
> // Downsamples |in| into |out|, taking one every |kNumbands| starting from
> // |offset|. |split_length| is the |out| length. |in| has to be at least
>@@ -150,8 +150,7 @@ void ThreeBandFilterBank::Analysis(const float* in,
>     Downsample(in, in_buffer_.size(), kNumBands - i - 1, &in_buffer_[0]);
>     for (size_t j = 0; j < kSparsity; ++j) {
>       const size_t offset = i + j * kNumBands;
>-      analysis_filters_[offset]->Filter(&in_buffer_[0],
>-                                        in_buffer_.size(),
>+      analysis_filters_[offset]->Filter(&in_buffer_[0], in_buffer_.size(),
>                                         &out_buffer_[0]);
>       DownModulate(&out_buffer_[0], out_buffer_.size(), offset, out);
>     }
>@@ -173,15 +172,13 @@ void ThreeBandFilterBank::Synthesis(const float* const* in,
>     for (size_t j = 0; j < kSparsity; ++j) {
>       const size_t offset = i + j * kNumBands;
>       UpModulate(in, in_buffer_.size(), offset, &in_buffer_[0]);
>-      synthesis_filters_[offset]->Filter(&in_buffer_[0],
>-                                         in_buffer_.size(),
>+      synthesis_filters_[offset]->Filter(&in_buffer_[0], in_buffer_.size(),
>                                          &out_buffer_[0]);
>       Upsample(&out_buffer_[0], out_buffer_.size(), i, out);
>     }
>   }
> }
> 
>-
> // Modulates |in| by |dct_modulation_| and accumulates it in each of the
> // |kNumBands| bands of |out|. |offset| is the index in the period of the
> // cosines used for modulation. |split_length| is the length of |in| and each
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/click_annotate.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/click_annotate.cc
>index a8b4a307e14..ce646b51beb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/click_annotate.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/click_annotate.cc
>@@ -14,9 +14,9 @@
> #include <memory>
> #include <vector>
> 
>-#include "modules/audio_processing/transient/transient_detector.h"
> #include "modules/audio_processing/transient/file_utils.h"
>-#include "system_wrappers/include/file_wrapper.h"
>+#include "modules/audio_processing/transient/transient_detector.h"
>+#include "rtc_base/system/file_wrapper.h"
> 
> using webrtc::FileWrapper;
> using webrtc::TransientDetector;
>@@ -73,9 +73,7 @@ int main(int argc, char* argv[]) {
> 
>   // Read first buffer from the PCM test file.
>   size_t file_samples_read = ReadInt16FromFileToFloatBuffer(
>-      pcm_file.get(),
>-      audio_buffer_length,
>-      audio_buffer.get());
>+      pcm_file.get(), audio_buffer_length, audio_buffer.get());
>   for (int time = 0; file_samples_read > 0; time += chunk_size_ms) {
>     // Pad the rest of the buffer with zeros.
>     for (size_t i = file_samples_read; i < audio_buffer_length; ++i) {
>@@ -92,14 +90,12 @@ int main(int argc, char* argv[]) {
>     send_times.push_back(value);
> 
>     // Read next buffer from the PCM test file.
>-    file_samples_read = ReadInt16FromFileToFloatBuffer(pcm_file.get(),
>-                                                       audio_buffer_length,
>-                                                       audio_buffer.get());
>+    file_samples_read = ReadInt16FromFileToFloatBuffer(
>+        pcm_file.get(), audio_buffer_length, audio_buffer.get());
>   }
> 
>-  size_t floats_written = WriteFloatBufferToFile(dat_file.get(),
>-                                                 send_times.size(),
>-                                                 &send_times[0]);
>+  size_t floats_written =
>+      WriteFloatBufferToFile(dat_file.get(), send_times.size(), &send_times[0]);
> 
>   if (floats_written == 0) {
>     printf("\nThe send times could not be written to DAT file\n\n");
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/common.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/common.h
>index 69546fc57b0..63c9a7b3153 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/common.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/common.h
>@@ -22,6 +22,6 @@ enum {
>   kSampleRate48kHz = 48000
> };
> 
>-} // namespace ts
>-} // namespace webrtc
>+}  // namespace ts
>+}  // namespace webrtc
> #endif  // MODULES_AUDIO_PROCESSING_TRANSIENT_COMMON_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/daubechies_8_wavelet_coeffs.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/daubechies_8_wavelet_coeffs.h
>index 4de24e0fcbb..92233bfd745 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/daubechies_8_wavelet_coeffs.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/daubechies_8_wavelet_coeffs.h
>@@ -19,44 +19,25 @@ namespace webrtc {
> 
> const int kDaubechies8CoefficientsLength = 16;
> 
>-const float kDaubechies8HighPassCoefficients[kDaubechies8CoefficientsLength]
>-    = {
>-  -5.44158422430816093862e-02f,
>-  3.12871590914465924627e-01f,
>-  -6.75630736298012846142e-01f,
>-  5.85354683654869090148e-01f,
>-  1.58291052560238926228e-02f,
>-  -2.84015542962428091389e-01f,
>-  -4.72484573997972536787e-04f,
>-  1.28747426620186011803e-01f,
>-  1.73693010020221083600e-02f,
>-  -4.40882539310647192377e-02f,
>-  -1.39810279170155156436e-02f,
>-  8.74609404701565465445e-03f,
>-  4.87035299301066034600e-03f,
>-  -3.91740372995977108837e-04f,
>-  -6.75449405998556772109e-04f,
>-  -1.17476784002281916305e-04f
>-};
>+const float kDaubechies8HighPassCoefficients[kDaubechies8CoefficientsLength] = {
>+    -5.44158422430816093862e-02f, 3.12871590914465924627e-01f,
>+    -6.75630736298012846142e-01f, 5.85354683654869090148e-01f,
>+    1.58291052560238926228e-02f,  -2.84015542962428091389e-01f,
>+    -4.72484573997972536787e-04f, 1.28747426620186011803e-01f,
>+    1.73693010020221083600e-02f,  -4.40882539310647192377e-02f,
>+    -1.39810279170155156436e-02f, 8.74609404701565465445e-03f,
>+    4.87035299301066034600e-03f,  -3.91740372995977108837e-04f,
>+    -6.75449405998556772109e-04f, -1.17476784002281916305e-04f};
> 
> const float kDaubechies8LowPassCoefficients[kDaubechies8CoefficientsLength] = {
>-  -1.17476784002281916305e-04f,
>-  6.75449405998556772109e-04f,
>-  -3.91740372995977108837e-04f,
>-  -4.87035299301066034600e-03f,
>-  8.74609404701565465445e-03f,
>-  1.39810279170155156436e-02f,
>-  -4.40882539310647192377e-02f,
>-  -1.73693010020221083600e-02f,
>-  1.28747426620186011803e-01f,
>-  4.72484573997972536787e-04f,
>-  -2.84015542962428091389e-01f,
>-  -1.58291052560238926228e-02f,
>-  5.85354683654869090148e-01f,
>-  6.75630736298012846142e-01f,
>-  3.12871590914465924627e-01f,
>-  5.44158422430816093862e-02f
>-};
>+    -1.17476784002281916305e-04f, 6.75449405998556772109e-04f,
>+    -3.91740372995977108837e-04f, -4.87035299301066034600e-03f,
>+    8.74609404701565465445e-03f,  1.39810279170155156436e-02f,
>+    -4.40882539310647192377e-02f, -1.73693010020221083600e-02f,
>+    1.28747426620186011803e-01f,  4.72484573997972536787e-04f,
>+    -2.84015542962428091389e-01f, -1.58291052560238926228e-02f,
>+    5.85354683654869090148e-01f,  6.75630736298012846142e-01f,
>+    3.12871590914465924627e-01f,  5.44158422430816093862e-02f};
> 
> }  // namespace webrtc
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/dyadic_decimator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/dyadic_decimator.h
>index 104f95d3afe..fcb56b7dc9c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/dyadic_decimator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/dyadic_decimator.h
>@@ -13,8 +13,6 @@
> 
> #include <cstdlib>
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>-
> // Provides a set of static methods to perform dyadic decimations.
> 
> namespace webrtc {
>@@ -44,7 +42,7 @@ inline size_t GetOutLengthToDyadicDecimate(size_t in_length,
> //         GetOutLengthToDyadicDecimate().
> //         Must be previously allocated.
> // Returns the number of output samples, -1 on error.
>-template<typename T>
>+template <typename T>
> static size_t DyadicDecimate(const T* in,
>                              size_t in_length,
>                              bool odd_sequence,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/dyadic_decimator_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/dyadic_decimator_unittest.cc
>index c407f47ffb8..3e65a7b9bfb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/dyadic_decimator_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/dyadic_decimator_unittest.cc
>@@ -19,7 +19,7 @@ static const size_t kOddBufferLength = 5;
> static const size_t kOutBufferLength = 3;
> 
> int16_t const test_buffer_even_len[] = {0, 1, 2, 3, 4, 5};
>-int16_t const test_buffer_odd_len[]  = {0, 1, 2, 3, 4};
>+int16_t const test_buffer_odd_len[] = {0, 1, 2, 3, 4};
> int16_t test_buffer_out[kOutBufferLength];
> 
> TEST(DyadicDecimatorTest, GetOutLengthToDyadicDecimate) {
>@@ -29,30 +29,23 @@ TEST(DyadicDecimatorTest, GetOutLengthToDyadicDecimate) {
>   EXPECT_EQ(2u, GetOutLengthToDyadicDecimate(5, true));
> }
> 
>-
> TEST(DyadicDecimatorTest, DyadicDecimateErrorValues) {
>   size_t out_samples = 0;
> 
>-  out_samples = DyadicDecimate(static_cast<int16_t*>(NULL),
>-                               kEvenBufferLength,
>+  out_samples = DyadicDecimate(static_cast<int16_t*>(NULL), kEvenBufferLength,
>                                false,  // Even sequence.
>-                               test_buffer_out,
>-                               kOutBufferLength);
>+                               test_buffer_out, kOutBufferLength);
>   EXPECT_EQ(0u, out_samples);
> 
>-  out_samples = DyadicDecimate(test_buffer_even_len,
>-                               kEvenBufferLength,
>+  out_samples = DyadicDecimate(test_buffer_even_len, kEvenBufferLength,
>                                false,  // Even sequence.
>-                               static_cast<int16_t*>(NULL),
>-                               kOutBufferLength);
>+                               static_cast<int16_t*>(NULL), kOutBufferLength);
>   EXPECT_EQ(0u, out_samples);
> 
>   // Less than required |out_length|.
>-  out_samples = DyadicDecimate(test_buffer_even_len,
>-                               kEvenBufferLength,
>+  out_samples = DyadicDecimate(test_buffer_even_len, kEvenBufferLength,
>                                false,  // Even sequence.
>-                               test_buffer_out,
>-                               2);
>+                               test_buffer_out, 2);
>   EXPECT_EQ(0u, out_samples);
> }
> 
>@@ -60,11 +53,9 @@ TEST(DyadicDecimatorTest, DyadicDecimateEvenLengthEvenSequence) {
>   size_t expected_out_samples =
>       GetOutLengthToDyadicDecimate(kEvenBufferLength, false);
> 
>-  size_t out_samples = DyadicDecimate(test_buffer_even_len,
>-                                      kEvenBufferLength,
>+  size_t out_samples = DyadicDecimate(test_buffer_even_len, kEvenBufferLength,
>                                       false,  // Even sequence.
>-                                      test_buffer_out,
>-                                      kOutBufferLength);
>+                                      test_buffer_out, kOutBufferLength);
> 
>   EXPECT_EQ(expected_out_samples, out_samples);
> 
>@@ -77,11 +68,9 @@ TEST(DyadicDecimatorTest, DyadicDecimateEvenLengthOddSequence) {
>   size_t expected_out_samples =
>       GetOutLengthToDyadicDecimate(kEvenBufferLength, true);
> 
>-  size_t out_samples = DyadicDecimate(test_buffer_even_len,
>-                                      kEvenBufferLength,
>+  size_t out_samples = DyadicDecimate(test_buffer_even_len, kEvenBufferLength,
>                                       true,  // Odd sequence.
>-                                      test_buffer_out,
>-                                      kOutBufferLength);
>+                                      test_buffer_out, kOutBufferLength);
> 
>   EXPECT_EQ(expected_out_samples, out_samples);
> 
>@@ -94,11 +83,9 @@ TEST(DyadicDecimatorTest, DyadicDecimateOddLengthEvenSequence) {
>   size_t expected_out_samples =
>       GetOutLengthToDyadicDecimate(kOddBufferLength, false);
> 
>-  size_t out_samples = DyadicDecimate(test_buffer_odd_len,
>-                                      kOddBufferLength,
>+  size_t out_samples = DyadicDecimate(test_buffer_odd_len, kOddBufferLength,
>                                       false,  // Even sequence.
>-                                      test_buffer_out,
>-                                      kOutBufferLength);
>+                                      test_buffer_out, kOutBufferLength);
> 
>   EXPECT_EQ(expected_out_samples, out_samples);
> 
>@@ -111,11 +98,9 @@ TEST(DyadicDecimatorTest, DyadicDecimateOddLengthOddSequence) {
>   size_t expected_out_samples =
>       GetOutLengthToDyadicDecimate(kOddBufferLength, true);
> 
>-  size_t out_samples = DyadicDecimate(test_buffer_odd_len,
>-                                      kOddBufferLength,
>+  size_t out_samples = DyadicDecimate(test_buffer_odd_len, kOddBufferLength,
>                                       true,  // Odd sequence.
>-                                      test_buffer_out,
>-                                      kOutBufferLength);
>+                                      test_buffer_out, kOutBufferLength);
> 
>   EXPECT_EQ(expected_out_samples, out_samples);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/file_utils.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/file_utils.cc
>index 7bf2e084f80..58f99325d1c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/file_utils.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/file_utils.cc
>@@ -12,8 +12,7 @@
> 
> #include <memory>
> 
>-#include "system_wrappers/include/file_wrapper.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include "rtc_base/system/file_wrapper.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/file_utils.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/file_utils.h
>index 3f05c1dd523..618401761b4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/file_utils.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/file_utils.h
>@@ -13,8 +13,7 @@
> 
> #include <string.h>
> 
>-#include "system_wrappers/include/file_wrapper.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include "rtc_base/system/file_wrapper.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/file_utils_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/file_utils_unittest.cc
>index c5e03990b5f..89902eccb11 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/file_utils_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/file_utils_unittest.cc
>@@ -11,14 +11,13 @@
> #include "modules/audio_processing/transient/file_utils.h"
> 
> #include <string.h>
>-#include <string>
> #include <memory>
>+#include <string>
> #include <vector>
> 
>-#include "system_wrappers/include/file_wrapper.h"
>+#include "rtc_base/system/file_wrapper.h"
> #include "test/gtest.h"
> #include "test/testsupport/fileutils.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -26,18 +25,18 @@ static const uint8_t kPiBytesf[4] = {0xDB, 0x0F, 0x49, 0x40};
> static const uint8_t kEBytesf[4] = {0x54, 0xF8, 0x2D, 0x40};
> static const uint8_t kAvogadroBytesf[4] = {0x2F, 0x0C, 0xFF, 0x66};
> 
>-static const uint8_t kPiBytes[8] =
>-    {0x18, 0x2D, 0x44, 0x54, 0xFB, 0x21, 0x09, 0x40};
>-static const uint8_t kEBytes[8] =
>-    {0x69, 0x57, 0x14, 0x8B, 0x0A, 0xBF, 0x05, 0x40};
>-static const uint8_t kAvogadroBytes[8] =
>-    {0xF4, 0xBC, 0xA8, 0xDF, 0x85, 0xE1, 0xDF, 0x44};
>+static const uint8_t kPiBytes[8] = {0x18, 0x2D, 0x44, 0x54,
>+                                    0xFB, 0x21, 0x09, 0x40};
>+static const uint8_t kEBytes[8] = {0x69, 0x57, 0x14, 0x8B,
>+                                   0x0A, 0xBF, 0x05, 0x40};
>+static const uint8_t kAvogadroBytes[8] = {0xF4, 0xBC, 0xA8, 0xDF,
>+                                          0x85, 0xE1, 0xDF, 0x44};
> 
> static const double kPi = 3.14159265358979323846;
> static const double kE = 2.71828182845904523536;
> static const double kAvogadro = 602214100000000000000000.0;
> 
>-class TransientFileUtilsTest: public ::testing::Test {
>+class TransientFileUtilsTest : public ::testing::Test {
>  protected:
>   TransientFileUtilsTest()
>       : kTestFileName(
>@@ -47,12 +46,10 @@ class TransientFileUtilsTest: public ::testing::Test {
>             test::ResourcePath("audio_processing/transient/float-utils",
>                                "dat")) {}
> 
>-  ~TransientFileUtilsTest() override {
>-    CleanupTempFiles();
>-  }
>+  ~TransientFileUtilsTest() override { CleanupTempFiles(); }
> 
>   std::string CreateTempFilename(const std::string& dir,
>-      const std::string& prefix) {
>+                                 const std::string& prefix) {
>     std::string filename = test::TempFilename(dir, prefix);
>     temp_filenames_.push_back(filename);
>     return filename;
>@@ -170,9 +167,8 @@ TEST_F(TransientFileUtilsTest, MAYBE_ReadInt16BufferFromFile) {
>   const size_t kBufferLength = 12;
>   std::unique_ptr<int16_t[]> buffer(new int16_t[kBufferLength]);
> 
>-  EXPECT_EQ(kBufferLength, ReadInt16BufferFromFile(file.get(),
>-                                                   kBufferLength,
>-                                                   buffer.get()));
>+  EXPECT_EQ(kBufferLength,
>+            ReadInt16BufferFromFile(file.get(), kBufferLength, buffer.get()));
>   EXPECT_EQ(22377, buffer[4]);
>   EXPECT_EQ(16389, buffer[7]);
>   EXPECT_EQ(17631, buffer[kBufferLength - 1]);
>@@ -184,9 +180,9 @@ TEST_F(TransientFileUtilsTest, MAYBE_ReadInt16BufferFromFile) {
>   // int16s read.
>   const size_t kBufferLenghtLargerThanFile = kBufferLength * 2;
>   buffer.reset(new int16_t[kBufferLenghtLargerThanFile]);
>-  EXPECT_EQ(kBufferLength, ReadInt16BufferFromFile(file.get(),
>-                                                   kBufferLenghtLargerThanFile,
>-                                                   buffer.get()));
>+  EXPECT_EQ(kBufferLength,
>+            ReadInt16BufferFromFile(file.get(), kBufferLenghtLargerThanFile,
>+                                    buffer.get()));
>   EXPECT_EQ(11544, buffer[0]);
>   EXPECT_EQ(22377, buffer[4]);
>   EXPECT_EQ(16389, buffer[7]);
>@@ -211,9 +207,8 @@ TEST_F(TransientFileUtilsTest, MAYBE_ReadInt16FromFileToFloatBuffer) {
>   const size_t kBufferLength = 12;
>   std::unique_ptr<float[]> buffer(new float[kBufferLength]);
> 
>-  EXPECT_EQ(kBufferLength, ReadInt16FromFileToFloatBuffer(file.get(),
>-                                                          kBufferLength,
>-                                                          buffer.get()));
>+  EXPECT_EQ(kBufferLength, ReadInt16FromFileToFloatBuffer(
>+                               file.get(), kBufferLength, buffer.get()));
> 
>   EXPECT_DOUBLE_EQ(11544, buffer[0]);
>   EXPECT_DOUBLE_EQ(22377, buffer[4]);
>@@ -228,9 +223,8 @@ TEST_F(TransientFileUtilsTest, MAYBE_ReadInt16FromFileToFloatBuffer) {
>   const size_t kBufferLenghtLargerThanFile = kBufferLength * 2;
>   buffer.reset(new float[kBufferLenghtLargerThanFile]);
>   EXPECT_EQ(kBufferLength,
>-            ReadInt16FromFileToFloatBuffer(file.get(),
>-                                           kBufferLenghtLargerThanFile,
>-                                           buffer.get()));
>+            ReadInt16FromFileToFloatBuffer(
>+                file.get(), kBufferLenghtLargerThanFile, buffer.get()));
>   EXPECT_DOUBLE_EQ(11544, buffer[0]);
>   EXPECT_DOUBLE_EQ(22377, buffer[4]);
>   EXPECT_DOUBLE_EQ(16389, buffer[7]);
>@@ -255,9 +249,8 @@ TEST_F(TransientFileUtilsTest, MAYBE_ReadInt16FromFileToDoubleBuffer) {
>   const size_t kBufferLength = 12;
>   std::unique_ptr<double[]> buffer(new double[kBufferLength]);
> 
>-  EXPECT_EQ(kBufferLength, ReadInt16FromFileToDoubleBuffer(file.get(),
>-                                                           kBufferLength,
>-                                                           buffer.get()));
>+  EXPECT_EQ(kBufferLength, ReadInt16FromFileToDoubleBuffer(
>+                               file.get(), kBufferLength, buffer.get()));
>   EXPECT_DOUBLE_EQ(11544, buffer[0]);
>   EXPECT_DOUBLE_EQ(22377, buffer[4]);
>   EXPECT_DOUBLE_EQ(16389, buffer[7]);
>@@ -271,9 +264,8 @@ TEST_F(TransientFileUtilsTest, MAYBE_ReadInt16FromFileToDoubleBuffer) {
>   const size_t kBufferLenghtLargerThanFile = kBufferLength * 2;
>   buffer.reset(new double[kBufferLenghtLargerThanFile]);
>   EXPECT_EQ(kBufferLength,
>-            ReadInt16FromFileToDoubleBuffer(file.get(),
>-                                            kBufferLenghtLargerThanFile,
>-                                            buffer.get()));
>+            ReadInt16FromFileToDoubleBuffer(
>+                file.get(), kBufferLenghtLargerThanFile, buffer.get()));
>   EXPECT_DOUBLE_EQ(11544, buffer[0]);
>   EXPECT_DOUBLE_EQ(22377, buffer[4]);
>   EXPECT_DOUBLE_EQ(16389, buffer[7]);
>@@ -297,9 +289,8 @@ TEST_F(TransientFileUtilsTest, MAYBE_ReadFloatBufferFromFile) {
>   const size_t kBufferLength = 3;
>   std::unique_ptr<float[]> buffer(new float[kBufferLength]);
> 
>-  EXPECT_EQ(kBufferLength, ReadFloatBufferFromFile(file.get(),
>-                                                   kBufferLength,
>-                                                   buffer.get()));
>+  EXPECT_EQ(kBufferLength,
>+            ReadFloatBufferFromFile(file.get(), kBufferLength, buffer.get()));
>   EXPECT_FLOAT_EQ(kPi, buffer[0]);
>   EXPECT_FLOAT_EQ(kE, buffer[1]);
>   EXPECT_FLOAT_EQ(kAvogadro, buffer[2]);
>@@ -311,9 +302,9 @@ TEST_F(TransientFileUtilsTest, MAYBE_ReadFloatBufferFromFile) {
>   // doubles read.
>   const size_t kBufferLenghtLargerThanFile = kBufferLength * 2;
>   buffer.reset(new float[kBufferLenghtLargerThanFile]);
>-  EXPECT_EQ(kBufferLength, ReadFloatBufferFromFile(file.get(),
>-                                                   kBufferLenghtLargerThanFile,
>-                                                   buffer.get()));
>+  EXPECT_EQ(kBufferLength,
>+            ReadFloatBufferFromFile(file.get(), kBufferLenghtLargerThanFile,
>+                                    buffer.get()));
>   EXPECT_FLOAT_EQ(kPi, buffer[0]);
>   EXPECT_FLOAT_EQ(kE, buffer[1]);
>   EXPECT_FLOAT_EQ(kAvogadro, buffer[2]);
>@@ -336,9 +327,8 @@ TEST_F(TransientFileUtilsTest, MAYBE_ReadDoubleBufferFromFile) {
>   const size_t kBufferLength = 3;
>   std::unique_ptr<double[]> buffer(new double[kBufferLength]);
> 
>-  EXPECT_EQ(kBufferLength, ReadDoubleBufferFromFile(file.get(),
>-                                                    kBufferLength,
>-                                                    buffer.get()));
>+  EXPECT_EQ(kBufferLength,
>+            ReadDoubleBufferFromFile(file.get(), kBufferLength, buffer.get()));
>   EXPECT_DOUBLE_EQ(kPi, buffer[0]);
>   EXPECT_DOUBLE_EQ(kE, buffer[1]);
>   EXPECT_DOUBLE_EQ(kAvogadro, buffer[2]);
>@@ -350,9 +340,9 @@ TEST_F(TransientFileUtilsTest, MAYBE_ReadDoubleBufferFromFile) {
>   // doubles read.
>   const size_t kBufferLenghtLargerThanFile = kBufferLength * 2;
>   buffer.reset(new double[kBufferLenghtLargerThanFile]);
>-  EXPECT_EQ(kBufferLength, ReadDoubleBufferFromFile(file.get(),
>-                                                    kBufferLenghtLargerThanFile,
>-                                                    buffer.get()));
>+  EXPECT_EQ(kBufferLength,
>+            ReadDoubleBufferFromFile(file.get(), kBufferLenghtLargerThanFile,
>+                                     buffer.get()));
>   EXPECT_DOUBLE_EQ(kPi, buffer[0]);
>   EXPECT_DOUBLE_EQ(kE, buffer[1]);
>   EXPECT_DOUBLE_EQ(kAvogadro, buffer[2]);
>@@ -366,8 +356,8 @@ TEST_F(TransientFileUtilsTest, MAYBE_ReadDoubleBufferFromFile) {
> TEST_F(TransientFileUtilsTest, MAYBE_WriteInt16BufferToFile) {
>   std::unique_ptr<FileWrapper> file(FileWrapper::Create());
> 
>-  std::string kOutFileName = CreateTempFilename(test::OutputPath(),
>-                                                "utils_test");
>+  std::string kOutFileName =
>+      CreateTempFilename(test::OutputPath(), "utils_test");
> 
>   file->OpenFile(kOutFileName.c_str(), false);  // Write mode.
>   ASSERT_TRUE(file->is_open()) << "File could not be opened:\n"
>@@ -381,8 +371,7 @@ TEST_F(TransientFileUtilsTest, MAYBE_WriteInt16BufferToFile) {
>   written_buffer[1] = 2;
>   written_buffer[2] = 3;
> 
>-  EXPECT_EQ(kBufferLength, WriteInt16BufferToFile(file.get(),
>-                                                  kBufferLength,
>+  EXPECT_EQ(kBufferLength, WriteInt16BufferToFile(file.get(), kBufferLength,
>                                                   written_buffer.get()));
> 
>   file->CloseFile();
>@@ -391,11 +380,9 @@ TEST_F(TransientFileUtilsTest, MAYBE_WriteInt16BufferToFile) {
>   ASSERT_TRUE(file->is_open()) << "File could not be opened:\n"
>                                << kOutFileName.c_str();
> 
>-  EXPECT_EQ(kBufferLength, ReadInt16BufferFromFile(file.get(),
>-                                                   kBufferLength,
>+  EXPECT_EQ(kBufferLength, ReadInt16BufferFromFile(file.get(), kBufferLength,
>                                                    read_buffer.get()));
>-  EXPECT_EQ(0, memcmp(written_buffer.get(),
>-                      read_buffer.get(),
>+  EXPECT_EQ(0, memcmp(written_buffer.get(), read_buffer.get(),
>                       kBufferLength * sizeof(written_buffer[0])));
> }
> 
>@@ -407,8 +394,8 @@ TEST_F(TransientFileUtilsTest, MAYBE_WriteInt16BufferToFile) {
> TEST_F(TransientFileUtilsTest, MAYBE_WriteFloatBufferToFile) {
>   std::unique_ptr<FileWrapper> file(FileWrapper::Create());
> 
>-  std::string kOutFileName = CreateTempFilename(test::OutputPath(),
>-                                                "utils_test");
>+  std::string kOutFileName =
>+      CreateTempFilename(test::OutputPath(), "utils_test");
> 
>   file->OpenFile(kOutFileName.c_str(), false);  // Write mode.
>   ASSERT_TRUE(file->is_open()) << "File could not be opened:\n"
>@@ -422,8 +409,7 @@ TEST_F(TransientFileUtilsTest, MAYBE_WriteFloatBufferToFile) {
>   written_buffer[1] = static_cast<float>(kE);
>   written_buffer[2] = static_cast<float>(kAvogadro);
> 
>-  EXPECT_EQ(kBufferLength, WriteFloatBufferToFile(file.get(),
>-                                                  kBufferLength,
>+  EXPECT_EQ(kBufferLength, WriteFloatBufferToFile(file.get(), kBufferLength,
>                                                   written_buffer.get()));
> 
>   file->CloseFile();
>@@ -432,11 +418,9 @@ TEST_F(TransientFileUtilsTest, MAYBE_WriteFloatBufferToFile) {
>   ASSERT_TRUE(file->is_open()) << "File could not be opened:\n"
>                                << kOutFileName.c_str();
> 
>-  EXPECT_EQ(kBufferLength, ReadFloatBufferFromFile(file.get(),
>-                                                   kBufferLength,
>+  EXPECT_EQ(kBufferLength, ReadFloatBufferFromFile(file.get(), kBufferLength,
>                                                    read_buffer.get()));
>-  EXPECT_EQ(0, memcmp(written_buffer.get(),
>-                      read_buffer.get(),
>+  EXPECT_EQ(0, memcmp(written_buffer.get(), read_buffer.get(),
>                       kBufferLength * sizeof(written_buffer[0])));
> }
> 
>@@ -448,8 +432,8 @@ TEST_F(TransientFileUtilsTest, MAYBE_WriteFloatBufferToFile) {
> TEST_F(TransientFileUtilsTest, MAYBE_WriteDoubleBufferToFile) {
>   std::unique_ptr<FileWrapper> file(FileWrapper::Create());
> 
>-  std::string kOutFileName = CreateTempFilename(test::OutputPath(),
>-                                                "utils_test");
>+  std::string kOutFileName =
>+      CreateTempFilename(test::OutputPath(), "utils_test");
> 
>   file->OpenFile(kOutFileName.c_str(), false);  // Write mode.
>   ASSERT_TRUE(file->is_open()) << "File could not be opened:\n"
>@@ -463,8 +447,7 @@ TEST_F(TransientFileUtilsTest, MAYBE_WriteDoubleBufferToFile) {
>   written_buffer[1] = kE;
>   written_buffer[2] = kAvogadro;
> 
>-  EXPECT_EQ(kBufferLength, WriteDoubleBufferToFile(file.get(),
>-                                                   kBufferLength,
>+  EXPECT_EQ(kBufferLength, WriteDoubleBufferToFile(file.get(), kBufferLength,
>                                                    written_buffer.get()));
> 
>   file->CloseFile();
>@@ -473,11 +456,9 @@ TEST_F(TransientFileUtilsTest, MAYBE_WriteDoubleBufferToFile) {
>   ASSERT_TRUE(file->is_open()) << "File could not be opened:\n"
>                                << kOutFileName.c_str();
> 
>-  EXPECT_EQ(kBufferLength, ReadDoubleBufferFromFile(file.get(),
>-                                                    kBufferLength,
>+  EXPECT_EQ(kBufferLength, ReadDoubleBufferFromFile(file.get(), kBufferLength,
>                                                     read_buffer.get()));
>-  EXPECT_EQ(0, memcmp(written_buffer.get(),
>-                      read_buffer.get(),
>+  EXPECT_EQ(0, memcmp(written_buffer.get(), read_buffer.get(),
>                       kBufferLength * sizeof(written_buffer[0])));
> }
> 
>@@ -501,9 +482,8 @@ TEST_F(TransientFileUtilsTest, MAYBE_ExpectedErrorReturnValues) {
> 
>   // Tests with file not opened.
>   EXPECT_EQ(0u, ReadInt16BufferFromFile(file.get(), 1, int16_buffer.get()));
>-  EXPECT_EQ(0u, ReadInt16FromFileToDoubleBuffer(file.get(),
>-                                                1,
>-                                                double_buffer.get()));
>+  EXPECT_EQ(
>+      0u, ReadInt16FromFileToDoubleBuffer(file.get(), 1, double_buffer.get()));
>   EXPECT_EQ(0u, ReadDoubleBufferFromFile(file.get(), 1, double_buffer.get()));
>   EXPECT_EQ(0u, WriteInt16BufferToFile(file.get(), 1, int16_buffer.get()));
>   EXPECT_EQ(0u, WriteDoubleBufferToFile(file.get(), 1, double_buffer.get()));
>@@ -518,9 +498,8 @@ TEST_F(TransientFileUtilsTest, MAYBE_ExpectedErrorReturnValues) {
> 
>   EXPECT_EQ(0u, ReadInt16FromFileToDoubleBuffer(NULL, 1, double_buffer.get()));
>   EXPECT_EQ(0u, ReadInt16FromFileToDoubleBuffer(file.get(), 1, NULL));
>-  EXPECT_EQ(0u, ReadInt16FromFileToDoubleBuffer(file.get(),
>-                                                0,
>-                                                double_buffer.get()));
>+  EXPECT_EQ(
>+      0u, ReadInt16FromFileToDoubleBuffer(file.get(), 0, double_buffer.get()));
> 
>   EXPECT_EQ(0u, ReadDoubleBufferFromFile(NULL, 1, double_buffer.get()));
>   EXPECT_EQ(0u, ReadDoubleBufferFromFile(file.get(), 1, NULL));
>@@ -536,4 +515,3 @@ TEST_F(TransientFileUtilsTest, MAYBE_ExpectedErrorReturnValues) {
> }
> 
> }  // namespace webrtc
>-
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/moving_moments.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/moving_moments.cc
>index 46b16b82bc4..a199bb0c1b9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/moving_moments.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/moving_moments.cc
>@@ -10,18 +10,14 @@
> 
> #include "modules/audio_processing/transient/moving_moments.h"
> 
>-#include <math.h>
>-#include <string.h>
>+#include <cmath>
> 
> #include "rtc_base/checks.h"
> 
> namespace webrtc {
> 
> MovingMoments::MovingMoments(size_t length)
>-    : length_(length),
>-      queue_(),
>-      sum_(0.0),
>-      sum_of_squares_(0.0) {
>+    : length_(length), queue_(), sum_(0.0), sum_of_squares_(0.0) {
>   RTC_DCHECK_GT(length, 0);
>   for (size_t i = 0; i < length; ++i) {
>     queue_.push(0.0);
>@@ -30,8 +26,10 @@ MovingMoments::MovingMoments(size_t length)
> 
> MovingMoments::~MovingMoments() {}
> 
>-void MovingMoments::CalculateMoments(const float* in, size_t in_length,
>-                                     float* first, float* second) {
>+void MovingMoments::CalculateMoments(const float* in,
>+                                     size_t in_length,
>+                                     float* first,
>+                                     float* second) {
>   RTC_DCHECK(in);
>   RTC_DCHECK_GT(in_length, 0);
>   RTC_DCHECK(first);
>@@ -45,7 +43,7 @@ void MovingMoments::CalculateMoments(const float* in, size_t in_length,
>     sum_ += in[i] - old_value;
>     sum_of_squares_ += in[i] * in[i] - old_value * old_value;
>     first[i] = sum_ / length_;
>-    second[i] = sum_of_squares_ / length_;
>+    second[i] = std::max(0.f, sum_of_squares_ / length_);
>   }
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/moving_moments.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/moving_moments.h
>index f1b3e38cc0a..6dc05204dad 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/moving_moments.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/moving_moments.h
>@@ -33,8 +33,10 @@ class MovingMoments {
> 
>   // Calculates the new values using |in|. Results will be in the out buffers.
>   // |first| and |second| must be allocated with at least |in_length|.
>-  void CalculateMoments(const float* in, size_t in_length,
>-                        float* first, float* second);
>+  void CalculateMoments(const float* in,
>+                        size_t in_length,
>+                        float* first,
>+                        float* second);
> 
>  private:
>   size_t length_;
>@@ -48,5 +50,4 @@ class MovingMoments {
> 
> }  // namespace webrtc
> 
>-
> #endif  // MODULES_AUDIO_PROCESSING_TRANSIENT_MOVING_MOMENTS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/moving_moments_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/moving_moments_unittest.cc
>index 057bc454e07..b0e613e7ab1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/moving_moments_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/moving_moments_unittest.cc
>@@ -26,7 +26,8 @@ class MovingMomentsTest : public ::testing::Test {
>   virtual void SetUp();
>   // Calls CalculateMoments and verifies that it produces the expected
>   // outputs.
>-  void CalculateMomentsAndVerify(const float* input, size_t input_length,
>+  void CalculateMomentsAndVerify(const float* input,
>+                                 size_t input_length,
>                                  const float* expected_mean,
>                                  const float* expected_mean_squares);
> 
>@@ -42,14 +43,13 @@ void MovingMomentsTest::SetUp() {
> }
> 
> void MovingMomentsTest::CalculateMomentsAndVerify(
>-    const float* input, size_t input_length,
>+    const float* input,
>+    size_t input_length,
>     const float* expected_mean,
>     const float* expected_mean_squares) {
>   ASSERT_LE(input_length, kMaxOutputLength);
> 
>-  moving_moments_->CalculateMoments(input,
>-                                    input_length,
>-                                    output_mean_,
>+  moving_moments_->CalculateMoments(input, input_length, output_mean_,
>                                     output_mean_squares_);
> 
>   for (size_t i = 1; i < input_length; ++i) {
>@@ -73,10 +73,10 @@ TEST_F(MovingMomentsTest, CorrectMomentsOfAConstantBuffer) {
>   const float kInput[] = {5.f, 5.f, 5.f, 5.f, 5.f, 5.f, 5.f, 5.f, 5.f, 5.f};
>   const size_t kInputLength = sizeof(kInput) / sizeof(kInput[0]);
> 
>-  const float expected_mean[kInputLength] =
>-      {1.f, 2.f, 3.f, 4.f, 5.f, 5.f, 5.f, 5.f, 5.f, 5.f};
>-  const float expected_mean_squares[kInputLength] =
>-      {5.f, 10.f, 15.f, 20.f, 25.f, 25.f, 25.f, 25.f, 25.f, 25.f};
>+  const float expected_mean[kInputLength] = {1.f, 2.f, 3.f, 4.f, 5.f,
>+                                             5.f, 5.f, 5.f, 5.f, 5.f};
>+  const float expected_mean_squares[kInputLength] = {
>+      5.f, 10.f, 15.f, 20.f, 25.f, 25.f, 25.f, 25.f, 25.f, 25.f};
> 
>   CalculateMomentsAndVerify(kInput, kInputLength, expected_mean,
>                             expected_mean_squares);
>@@ -86,24 +86,23 @@ TEST_F(MovingMomentsTest, CorrectMomentsOfAnIncreasingBuffer) {
>   const float kInput[] = {1.f, 2.f, 3.f, 4.f, 5.f, 6.f, 7.f, 8.f, 9.f};
>   const size_t kInputLength = sizeof(kInput) / sizeof(kInput[0]);
> 
>-  const float expected_mean[kInputLength] =
>-      {0.2f, 0.6f, 1.2f, 2.f, 3.f, 4.f, 5.f, 6.f, 7.f};
>-  const float expected_mean_squares[kInputLength] =
>-      {0.2f, 1.f, 2.8f, 6.f, 11.f, 18.f, 27.f, 38.f, 51.f};
>+  const float expected_mean[kInputLength] = {0.2f, 0.6f, 1.2f, 2.f, 3.f,
>+                                             4.f,  5.f,  6.f,  7.f};
>+  const float expected_mean_squares[kInputLength] = {
>+      0.2f, 1.f, 2.8f, 6.f, 11.f, 18.f, 27.f, 38.f, 51.f};
> 
>   CalculateMomentsAndVerify(kInput, kInputLength, expected_mean,
>                             expected_mean_squares);
> }
> 
> TEST_F(MovingMomentsTest, CorrectMomentsOfADecreasingBuffer) {
>-  const float kInput[] =
>-      {-1.f, -2.f, -3.f, -4.f, -5.f, -6.f, -7.f, -8.f, -9.f};
>+  const float kInput[] = {-1.f, -2.f, -3.f, -4.f, -5.f, -6.f, -7.f, -8.f, -9.f};
>   const size_t kInputLength = sizeof(kInput) / sizeof(kInput[0]);
> 
>-  const float expected_mean[kInputLength] =
>-      {-0.2f, -0.6f, -1.2f, -2.f, -3.f, -4.f, -5.f, -6.f, -7.f};
>-  const float expected_mean_squares[kInputLength] =
>-      {0.2f, 1.f, 2.8f, 6.f, 11.f, 18.f, 27.f, 38.f, 51.f};
>+  const float expected_mean[kInputLength] = {-0.2f, -0.6f, -1.2f, -2.f, -3.f,
>+                                             -4.f,  -5.f,  -6.f,  -7.f};
>+  const float expected_mean_squares[kInputLength] = {
>+      0.2f, 1.f, 2.8f, 6.f, 11.f, 18.f, 27.f, 38.f, 51.f};
> 
>   CalculateMomentsAndVerify(kInput, kInputLength, expected_mean,
>                             expected_mean_squares);
>@@ -112,76 +111,78 @@ TEST_F(MovingMomentsTest, CorrectMomentsOfADecreasingBuffer) {
> TEST_F(MovingMomentsTest, CorrectMomentsOfAZeroMeanSequence) {
>   const size_t kMovingMomentsBufferLength = 4;
>   moving_moments_.reset(new MovingMoments(kMovingMomentsBufferLength));
>-  const float kInput[] =
>-      {1.f, -1.f, 1.f, -1.f, 1.f, -1.f, 1.f, -1.f, 1.f, -1.f};
>+  const float kInput[] = {1.f,  -1.f, 1.f,  -1.f, 1.f,
>+                          -1.f, 1.f,  -1.f, 1.f,  -1.f};
>   const size_t kInputLength = sizeof(kInput) / sizeof(kInput[0]);
> 
>-  const float expected_mean[kInputLength] =
>-      {0.25f, 0.f, 0.25f, 0.f, 0.f, 0.f, 0.f, 0.f, 0.f, 0.f};
>-  const float expected_mean_squares[kInputLength] =
>-      {0.25f, 0.5f, 0.75f, 1.f, 1.f, 1.f, 1.f, 1.f, 1.f, 1.f};
>+  const float expected_mean[kInputLength] = {0.25f, 0.f, 0.25f, 0.f, 0.f,
>+                                             0.f,   0.f, 0.f,   0.f, 0.f};
>+  const float expected_mean_squares[kInputLength] = {
>+      0.25f, 0.5f, 0.75f, 1.f, 1.f, 1.f, 1.f, 1.f, 1.f, 1.f};
> 
>   CalculateMomentsAndVerify(kInput, kInputLength, expected_mean,
>                             expected_mean_squares);
> }
> 
> TEST_F(MovingMomentsTest, CorrectMomentsOfAnArbitraryBuffer) {
>-  const float kInput[] =
>-      {0.2f, 0.3f, 0.5f, 0.7f, 0.11f, 0.13f, 0.17f, 0.19f, 0.23f};
>+  const float kInput[] = {0.2f,  0.3f,  0.5f,  0.7f, 0.11f,
>+                          0.13f, 0.17f, 0.19f, 0.23f};
>   const size_t kInputLength = sizeof(kInput) / sizeof(kInput[0]);
> 
>-  const float expected_mean[kInputLength] =
>-      {0.04f, 0.1f, 0.2f, 0.34f, 0.362f, 0.348f, 0.322f, 0.26f, 0.166f};
>-  const float expected_mean_squares[kInputLength] =
>-      {0.008f, 0.026f, 0.076f, 0.174f, 0.1764f, 0.1718f, 0.1596f, 0.1168f,
>-      0.0294f};
>+  const float expected_mean[kInputLength] = {
>+      0.04f, 0.1f, 0.2f, 0.34f, 0.362f, 0.348f, 0.322f, 0.26f, 0.166f};
>+  const float expected_mean_squares[kInputLength] = {0.008f,  0.026f,  0.076f,
>+                                                     0.174f,  0.1764f, 0.1718f,
>+                                                     0.1596f, 0.1168f, 0.0294f};
> 
>   CalculateMomentsAndVerify(kInput, kInputLength, expected_mean,
>                             expected_mean_squares);
> }
> 
> TEST_F(MovingMomentsTest, MutipleCalculateMomentsCalls) {
>-  const float kInputFirstCall[] =
>-      {0.2f, 0.3f, 0.5f, 0.7f, 0.11f, 0.13f, 0.17f, 0.19f, 0.23f};
>-  const size_t kInputFirstCallLength = sizeof(kInputFirstCall) /
>-                                    sizeof(kInputFirstCall[0]);
>+  const float kInputFirstCall[] = {0.2f,  0.3f,  0.5f,  0.7f, 0.11f,
>+                                   0.13f, 0.17f, 0.19f, 0.23f};
>+  const size_t kInputFirstCallLength =
>+      sizeof(kInputFirstCall) / sizeof(kInputFirstCall[0]);
>   const float kInputSecondCall[] = {0.29f, 0.31f};
>-  const size_t kInputSecondCallLength = sizeof(kInputSecondCall) /
>-                                     sizeof(kInputSecondCall[0]);
>+  const size_t kInputSecondCallLength =
>+      sizeof(kInputSecondCall) / sizeof(kInputSecondCall[0]);
>   const float kInputThirdCall[] = {0.37f, 0.41f, 0.43f, 0.47f};
>-  const size_t kInputThirdCallLength = sizeof(kInputThirdCall) /
>-                                    sizeof(kInputThirdCall[0]);
>+  const size_t kInputThirdCallLength =
>+      sizeof(kInputThirdCall) / sizeof(kInputThirdCall[0]);
> 
>-  const float expected_mean_first_call[kInputFirstCallLength] =
>-      {0.04f, 0.1f, 0.2f, 0.34f, 0.362f, 0.348f, 0.322f, 0.26f, 0.166f};
>-  const float expected_mean_squares_first_call[kInputFirstCallLength] =
>-      {0.008f, 0.026f, 0.076f, 0.174f, 0.1764f, 0.1718f, 0.1596f, 0.1168f,
>-      0.0294f};
>+  const float expected_mean_first_call[kInputFirstCallLength] = {
>+      0.04f, 0.1f, 0.2f, 0.34f, 0.362f, 0.348f, 0.322f, 0.26f, 0.166f};
>+  const float expected_mean_squares_first_call[kInputFirstCallLength] = {
>+      0.008f,  0.026f,  0.076f,  0.174f, 0.1764f,
>+      0.1718f, 0.1596f, 0.1168f, 0.0294f};
> 
>-  const float expected_mean_second_call[kInputSecondCallLength] =
>-      {0.202f, 0.238f};
>-  const float expected_mean_squares_second_call[kInputSecondCallLength] =
>-      {0.0438f, 0.0596f};
>+  const float expected_mean_second_call[kInputSecondCallLength] = {0.202f,
>+                                                                   0.238f};
>+  const float expected_mean_squares_second_call[kInputSecondCallLength] = {
>+      0.0438f, 0.0596f};
> 
>-  const float expected_mean_third_call[kInputThirdCallLength] =
>-      {0.278f, 0.322f, 0.362f, 0.398f};
>-  const float expected_mean_squares_third_call[kInputThirdCallLength] =
>-      {0.0812f, 0.1076f, 0.134f, 0.1614f};
>+  const float expected_mean_third_call[kInputThirdCallLength] = {
>+      0.278f, 0.322f, 0.362f, 0.398f};
>+  const float expected_mean_squares_third_call[kInputThirdCallLength] = {
>+      0.0812f, 0.1076f, 0.134f, 0.1614f};
> 
>   CalculateMomentsAndVerify(kInputFirstCall, kInputFirstCallLength,
>-      expected_mean_first_call, expected_mean_squares_first_call);
>+                            expected_mean_first_call,
>+                            expected_mean_squares_first_call);
> 
>   CalculateMomentsAndVerify(kInputSecondCall, kInputSecondCallLength,
>-      expected_mean_second_call, expected_mean_squares_second_call);
>+                            expected_mean_second_call,
>+                            expected_mean_squares_second_call);
> 
>   CalculateMomentsAndVerify(kInputThirdCall, kInputThirdCallLength,
>-      expected_mean_third_call, expected_mean_squares_third_call);
>+                            expected_mean_third_call,
>+                            expected_mean_squares_third_call);
> }
> 
>-TEST_F(MovingMomentsTest,
>-       VerifySampleBasedVsBlockBasedCalculation) {
>-  const float kInput[] =
>-      {0.2f, 0.3f, 0.5f, 0.7f, 0.11f, 0.13f, 0.17f, 0.19f, 0.23f};
>+TEST_F(MovingMomentsTest, VerifySampleBasedVsBlockBasedCalculation) {
>+  const float kInput[] = {0.2f,  0.3f,  0.5f,  0.7f, 0.11f,
>+                          0.13f, 0.17f, 0.19f, 0.23f};
>   const size_t kInputLength = sizeof(kInput) / sizeof(kInput[0]);
> 
>   float output_mean_block_based[kInputLength];
>@@ -190,17 +191,16 @@ TEST_F(MovingMomentsTest,
>   float output_mean_sample_based;
>   float output_mean_squares_sample_based;
> 
>-  moving_moments_->CalculateMoments(
>-      kInput, kInputLength, output_mean_block_based,
>-      output_mean_squares_block_based);
>+  moving_moments_->CalculateMoments(kInput, kInputLength,
>+                                    output_mean_block_based,
>+                                    output_mean_squares_block_based);
>   moving_moments_.reset(new MovingMoments(kMovingMomentsBufferLength));
>   for (size_t i = 0; i < kInputLength; ++i) {
>-    moving_moments_->CalculateMoments(
>-        &kInput[i], 1, &output_mean_sample_based,
>-        &output_mean_squares_sample_based);
>+    moving_moments_->CalculateMoments(&kInput[i], 1, &output_mean_sample_based,
>+                                      &output_mean_squares_sample_based);
>     EXPECT_FLOAT_EQ(output_mean_block_based[i], output_mean_sample_based);
>     EXPECT_FLOAT_EQ(output_mean_squares_block_based[i],
>-                     output_mean_squares_sample_based);
>+                    output_mean_squares_sample_based);
>   }
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/transient_detector.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/transient_detector.cc
>index 1bb6f9f10a3..c3bf282b0e1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/transient_detector.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/transient_detector.cc
>@@ -51,8 +51,7 @@ TransientDetector::TransientDetector(int sample_rate_hz)
>   wpd_tree_.reset(new WPDTree(samples_per_chunk_,
>                               kDaubechies8HighPassCoefficients,
>                               kDaubechies8LowPassCoefficients,
>-                              kDaubechies8CoefficientsLength,
>-                              kLevels));
>+                              kDaubechies8CoefficientsLength, kLevels));
>   for (size_t i = 0; i < kLeaves; ++i) {
>     moving_moments_[i].reset(
>         new MovingMoments(samples_per_transient / kLeaves));
>@@ -86,8 +85,7 @@ float TransientDetector::Detect(const float* data,
>   for (size_t i = 0; i < kLeaves; ++i) {
>     WPDNode* leaf = wpd_tree_->NodeAt(kLevels, i);
> 
>-    moving_moments_[i]->CalculateMoments(leaf->data(),
>-                                         tree_leaves_data_length_,
>+    moving_moments_[i]->CalculateMoments(leaf->data(), tree_leaves_data_length_,
>                                          first_moments_.get(),
>                                          second_moments_.get());
> 
>@@ -127,8 +125,9 @@ float TransientDetector::Detect(const float* data,
>     const float kVerticalScaling = 0.5f;
>     const float kVerticalShift = 1.f;
> 
>-    result = (cos(result * horizontal_scaling + kHorizontalShift)
>-        + kVerticalShift) * kVerticalScaling;
>+    result =
>+        (cos(result * horizontal_scaling + kHorizontalShift) + kVerticalShift) *
>+        kVerticalScaling;
>     result *= result;
>   }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/transient_detector_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/transient_detector_unittest.cc
>index 96af179c428..d1eb7af8571 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/transient_detector_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/transient_detector_unittest.cc
>@@ -16,15 +16,13 @@
> 
> #include "modules/audio_processing/transient/common.h"
> #include "modules/audio_processing/transient/file_utils.h"
>-#include "system_wrappers/include/file_wrapper.h"
>+#include "rtc_base/system/file_wrapper.h"
> #include "test/gtest.h"
> #include "test/testsupport/fileutils.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>-static const int kSampleRatesHz[] = {ts::kSampleRate8kHz,
>-                                     ts::kSampleRate16kHz,
>+static const int kSampleRatesHz[] = {ts::kSampleRate8kHz, ts::kSampleRate16kHz,
>                                      ts::kSampleRate32kHz,
>                                      ts::kSampleRate48kHz};
> static const size_t kNumberOfSampleRates =
>@@ -57,7 +55,7 @@ TEST(TransientDetectorTest, CorrectnessBasedOnFiles) {
> 
>     bool file_opened = detect_file->is_open();
>     ASSERT_TRUE(file_opened) << "File could not be opened.\n"
>-          << detect_file_name.str().c_str();
>+                             << detect_file_name.str().c_str();
> 
>     // Prepare audio file.
>     std::stringstream audio_file_name;
>@@ -80,8 +78,7 @@ TEST(TransientDetectorTest, CorrectnessBasedOnFiles) {
> 
>     size_t frames_read = 0;
> 
>-    while (ReadInt16FromFileToFloatBuffer(audio_file.get(),
>-                                          buffer_length,
>+    while (ReadInt16FromFileToFloatBuffer(audio_file.get(), buffer_length,
>                                           buffer.get()) == buffer_length) {
>       ++frames_read;
> 
>@@ -92,8 +89,8 @@ TEST(TransientDetectorTest, CorrectnessBasedOnFiles) {
>           << "Detect test file is malformed.\n";
> 
>       // Compare results with data from the matlab test file.
>-      EXPECT_NEAR(file_value, detector_value, kTolerance) << "Frame: "
>-          << frames_read;
>+      EXPECT_NEAR(file_value, detector_value, kTolerance)
>+          << "Frame: " << frames_read;
>     }
> 
>     detect_file->CloseFile();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/transient_suppression_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/transient_suppression_test.cc
>index 3442ee0af6f..9e7ecd57865 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/transient_suppression_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/transient_suppression_test.cc
>@@ -10,8 +10,8 @@
> 
> #include "modules/audio_processing/transient/transient_suppressor.h"
> 
>-#include <stdlib.h>
> #include <stdio.h>
>+#include <stdlib.h>
> #include <string.h>
> 
> #include <memory>
>@@ -19,11 +19,9 @@
> 
> #include "common_audio/include/audio_util.h"
> #include "modules/audio_processing/agc/agc.h"
>-#include "modules/include/module_common_types.h"
> #include "rtc_base/flags.h"
> #include "test/gtest.h"
> #include "test/testsupport/fileutils.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> DEFINE_string(in_file_name, "", "PCM file that contains the signal.");
> DEFINE_string(detection_file_name,
>@@ -37,9 +35,7 @@ DEFINE_int(chunk_size_ms,
>            10,
>            "Time between each chunk of samples in milliseconds.");
> 
>-DEFINE_int(sample_rate_hz,
>-           16000,
>-           "Sampling frequency of the signal in Hertz.");
>+DEFINE_int(sample_rate_hz, 16000, "Sampling frequency of the signal in Hertz.");
> DEFINE_int(detection_rate_hz,
>            0,
>            "Sampling frequency of the detection signal in Hertz.");
>@@ -80,9 +76,7 @@ bool ReadBuffers(FILE* in_file,
>     tmpbuf.reset(new int16_t[num_channels * audio_buffer_size]);
>     read_ptr = tmpbuf.get();
>   }
>-  if (fread(read_ptr,
>-            sizeof(*read_ptr),
>-            num_channels * audio_buffer_size,
>+  if (fread(read_ptr, sizeof(*read_ptr), num_channels * audio_buffer_size,
>             in_file) != num_channels * audio_buffer_size) {
>     return false;
>   }
>@@ -105,8 +99,8 @@ bool ReadBuffers(FILE* in_file,
>   }
>   if (reference_file) {
>     std::unique_ptr<int16_t[]> ibuf(new int16_t[audio_buffer_size]);
>-    if (fread(ibuf.get(), sizeof(ibuf[0]), audio_buffer_size, reference_file)
>-        != audio_buffer_size)
>+    if (fread(ibuf.get(), sizeof(ibuf[0]), audio_buffer_size, reference_file) !=
>+        audio_buffer_size)
>       return false;
>     S16ToFloat(ibuf.get(), audio_buffer_size, reference_buffer);
>   }
>@@ -164,8 +158,8 @@ void void_main() {
>   Agc agc;
> 
>   TransientSuppressor suppressor;
>-  suppressor.Initialize(
>-      FLAG_sample_rate_hz, detection_rate_hz, FLAG_num_channels);
>+  suppressor.Initialize(FLAG_sample_rate_hz, detection_rate_hz,
>+                        FLAG_num_channels);
> 
>   const size_t audio_buffer_size =
>       FLAG_chunk_size_ms * FLAG_sample_rate_hz / 1000;
>@@ -185,40 +179,27 @@ void void_main() {
>   if (reference_file)
>     reference_buffer.reset(new float[audio_buffer_size]);
> 
>-  while (ReadBuffers(in_file,
>-                     audio_buffer_size,
>-                     FLAG_num_channels,
>-                     audio_buffer_i.get(),
>-                     detection_file,
>-                     detection_buffer_size,
>-                     detection_buffer.get(),
>-                     reference_file,
>-                     reference_buffer.get())) {
>-    ASSERT_EQ(0,
>-              agc.Process(audio_buffer_i.get(),
>-                          static_cast<int>(audio_buffer_size),
>-                          FLAG_sample_rate_hz))
>-        << "The AGC could not process the frame";
>+  while (ReadBuffers(in_file, audio_buffer_size, FLAG_num_channels,
>+                     audio_buffer_i.get(), detection_file,
>+                     detection_buffer_size, detection_buffer.get(),
>+                     reference_file, reference_buffer.get())) {
>+    agc.Process(audio_buffer_i.get(), static_cast<int>(audio_buffer_size),
>+                FLAG_sample_rate_hz);
> 
>     for (size_t i = 0; i < FLAG_num_channels * audio_buffer_size; ++i) {
>       audio_buffer_f[i] = audio_buffer_i[i];
>     }
> 
>-    ASSERT_EQ(0,
>-              suppressor.Suppress(audio_buffer_f.get(),
>-                                  audio_buffer_size,
>-                                  FLAG_num_channels,
>-                                  detection_buffer.get(),
>-                                  detection_buffer_size,
>-                                  reference_buffer.get(),
>-                                  audio_buffer_size,
>-                                  agc.voice_probability(),
>-                                  true))
>+    ASSERT_EQ(0, suppressor.Suppress(audio_buffer_f.get(), audio_buffer_size,
>+                                     FLAG_num_channels, detection_buffer.get(),
>+                                     detection_buffer_size,
>+                                     reference_buffer.get(), audio_buffer_size,
>+                                     agc.voice_probability(), true))
>         << "The transient suppressor could not suppress the frame";
> 
>     // Write result to out file.
>-    WritePCM(
>-        out_file, audio_buffer_size, FLAG_num_channels, audio_buffer_f.get());
>+    WritePCM(out_file, audio_buffer_size, FLAG_num_channels,
>+             audio_buffer_f.get());
>   }
> 
>   fclose(in_file);
>@@ -234,8 +215,8 @@ void void_main() {
> }  // namespace webrtc
> 
> int main(int argc, char* argv[]) {
>-  if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) ||
>-      FLAG_help || argc != 1) {
>+  if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) || FLAG_help ||
>+      argc != 1) {
>     printf("%s", webrtc::kUsage);
>     if (FLAG_help) {
>       rtc::FlagList::Print(nullptr, false);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/transient_suppressor.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/transient_suppressor.cc
>index 9bbd7d95005..1a5ed5669fe 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/transient_suppressor.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/transient_suppressor.cc
>@@ -17,15 +17,14 @@
> #include <deque>
> #include <set>
> 
>-#include "common_audio/fft4g.h"
> #include "common_audio/include/audio_util.h"
> #include "common_audio/signal_processing/include/signal_processing_library.h"
>+#include "common_audio/third_party/fft4g/fft4g.h"
> #include "modules/audio_processing/ns/windows_private.h"
> #include "modules/audio_processing/transient/common.h"
> #include "modules/audio_processing/transient/transient_detector.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -60,8 +59,7 @@ TransientSuppressor::TransientSuppressor()
>       use_hard_restoration_(false),
>       chunks_since_voice_change_(0),
>       seed_(182),
>-      using_reference_(false) {
>-}
>+      using_reference_(false) {}
> 
> TransientSuppressor::~TransientSuppressor() {}
> 
>@@ -110,17 +108,14 @@ int TransientSuppressor::Initialize(int sample_rate_hz,
>   RTC_DCHECK_GE(complex_analysis_length_, kMaxVoiceBin);
>   num_channels_ = num_channels;
>   in_buffer_.reset(new float[analysis_length_ * num_channels_]);
>-  memset(in_buffer_.get(),
>-         0,
>+  memset(in_buffer_.get(), 0,
>          analysis_length_ * num_channels_ * sizeof(in_buffer_[0]));
>   detection_length_ = detection_rate_hz * ts::kChunkSizeMs / 1000;
>   detection_buffer_.reset(new float[detection_length_]);
>-  memset(detection_buffer_.get(),
>-         0,
>+  memset(detection_buffer_.get(), 0,
>          detection_length_ * sizeof(detection_buffer_[0]));
>   out_buffer_.reset(new float[analysis_length_ * num_channels_]);
>-  memset(out_buffer_.get(),
>-         0,
>+  memset(out_buffer_.get(), 0,
>          analysis_length_ * num_channels_ * sizeof(out_buffer_[0]));
>   // ip[0] must be zero to trigger initialization using rdft().
>   size_t ip_length = 2 + sqrtf(analysis_length_);
>@@ -129,14 +124,12 @@ int TransientSuppressor::Initialize(int sample_rate_hz,
>   wfft_.reset(new float[complex_analysis_length_ - 1]);
>   memset(wfft_.get(), 0, (complex_analysis_length_ - 1) * sizeof(wfft_[0]));
>   spectral_mean_.reset(new float[complex_analysis_length_ * num_channels_]);
>-  memset(spectral_mean_.get(),
>-         0,
>+  memset(spectral_mean_.get(), 0,
>          complex_analysis_length_ * num_channels_ * sizeof(spectral_mean_[0]));
>   fft_buffer_.reset(new float[analysis_length_ + 2]);
>   memset(fft_buffer_.get(), 0, (analysis_length_ + 2) * sizeof(fft_buffer_[0]));
>   magnitudes_.reset(new float[complex_analysis_length_]);
>-  memset(magnitudes_.get(),
>-         0,
>+  memset(magnitudes_.get(), 0,
>          complex_analysis_length_ * sizeof(magnitudes_[0]));
>   mean_factor_.reset(new float[complex_analysis_length_]);
> 
>@@ -190,8 +183,8 @@ int TransientSuppressor::Suppress(float* data,
>       detection_data = &in_buffer_[buffer_delay_];
>     }
> 
>-    float detector_result = detector_->Detect(
>-        detection_data, detection_length, reference_data, reference_length);
>+    float detector_result = detector_->Detect(detection_data, detection_length,
>+                                              reference_data, reference_length);
>     if (detector_result < 0) {
>       return -1;
>     }
>@@ -247,8 +240,8 @@ void TransientSuppressor::Suppress(float* in_ptr,
>   fft_buffer_[1] = 0.f;
> 
>   for (size_t i = 0; i < complex_analysis_length_; ++i) {
>-    magnitudes_[i] = ComplexMagnitude(fft_buffer_[i * 2],
>-                                      fft_buffer_[i * 2 + 1]);
>+    magnitudes_[i] =
>+        ComplexMagnitude(fft_buffer_[i * 2], fft_buffer_[i * 2 + 1]);
>   }
>   // Restore audio if necessary.
>   if (suppression_enabled_) {
>@@ -269,11 +262,7 @@ void TransientSuppressor::Suppress(float* in_ptr,
>   // Put R[n/2] back in fft_buffer_[1].
>   fft_buffer_[1] = fft_buffer_[analysis_length_];
> 
>-  WebRtc_rdft(analysis_length_,
>-              -1,
>-              fft_buffer_.get(),
>-              ip_.get(),
>-              wfft_.get());
>+  WebRtc_rdft(analysis_length_, -1, fft_buffer_.get(), ip_.get(), wfft_.get());
>   const float fft_scaling = 2.f / analysis_length_;
> 
>   for (size_t i = 0; i < analysis_length_; ++i) {
>@@ -301,8 +290,7 @@ void TransientSuppressor::UpdateKeypress(bool key_pressed) {
>     keypress_counter_ = 0;
>   }
> 
>-  if (detection_enabled_ &&
>-      ++chunks_since_keypress_ > kChunksUntilNotTyping) {
>+  if (detection_enabled_ && ++chunks_since_keypress_ > kChunksUntilNotTyping) {
>     if (suppression_enabled_) {
>       RTC_LOG(LS_INFO) << "[ts] Transient suppression is now disabled.";
>     }
>@@ -337,26 +325,22 @@ void TransientSuppressor::UpdateRestoration(float voice_probability) {
> // |detection_enabled_| is updated by UpdateKeypress().
> void TransientSuppressor::UpdateBuffers(float* data) {
>   // TODO(aluebs): Change to ring buffer.
>-  memmove(in_buffer_.get(),
>-          &in_buffer_[data_length_],
>+  memmove(in_buffer_.get(), &in_buffer_[data_length_],
>           (buffer_delay_ + (num_channels_ - 1) * analysis_length_) *
>               sizeof(in_buffer_[0]));
>   // Copy new chunk to buffer.
>   for (int i = 0; i < num_channels_; ++i) {
>     memcpy(&in_buffer_[buffer_delay_ + i * analysis_length_],
>-           &data[i * data_length_],
>-           data_length_ * sizeof(*data));
>+           &data[i * data_length_], data_length_ * sizeof(*data));
>   }
>   if (detection_enabled_) {
>     // Shift previous chunk in out buffer.
>-    memmove(out_buffer_.get(),
>-            &out_buffer_[data_length_],
>+    memmove(out_buffer_.get(), &out_buffer_[data_length_],
>             (buffer_delay_ + (num_channels_ - 1) * analysis_length_) *
>                 sizeof(out_buffer_[0]));
>     // Initialize new chunk in out buffer.
>     for (int i = 0; i < num_channels_; ++i) {
>-      memset(&out_buffer_[buffer_delay_ + i * analysis_length_],
>-             0,
>+      memset(&out_buffer_[buffer_delay_ + i * analysis_length_], 0,
>              data_length_ * sizeof(out_buffer_[0]));
>     }
>   }
>@@ -375,7 +359,7 @@ void TransientSuppressor::HardRestoration(float* spectral_mean) {
>     if (magnitudes_[i] > spectral_mean[i] && magnitudes_[i] > 0) {
>       // RandU() generates values on [0, int16::max()]
>       const float phase = 2 * ts::kPi * WebRtcSpl_RandU(&seed_) /
>-          std::numeric_limits<int16_t>::max();
>+                          std::numeric_limits<int16_t>::max();
>       const float scaled_mean = detector_result * spectral_mean[i];
> 
>       fft_buffer_[i * 2] = (1 - detector_result) * fft_buffer_[i * 2] +
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/transient_suppressor.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/transient_suppressor.h
>index 27b096c795c..9ae3fc660a8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/transient_suppressor.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/transient_suppressor.h
>@@ -16,7 +16,6 @@
> #include <set>
> 
> #include "rtc_base/gtest_prod_util.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/wpd_node.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/wpd_node.cc
>index 20d6a9013e4..2e0ee7e5b71 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/wpd_node.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/wpd_node.cc
>@@ -23,13 +23,12 @@ namespace webrtc {
> WPDNode::WPDNode(size_t length,
>                  const float* coefficients,
>                  size_t coefficients_length)
>-    : // The data buffer has parent data length to be able to contain and filter
>-      // it.
>+    :  // The data buffer has parent data length to be able to contain and
>+       // filter it.
>       data_(new float[2 * length + 1]),
>       length_(length),
>-      filter_(CreateFirFilter(coefficients,
>-                              coefficients_length,
>-                              2 * length + 1)) {
>+      filter_(
>+          CreateFirFilter(coefficients, coefficients_length, 2 * length + 1)) {
>   RTC_DCHECK_GT(length, 0);
>   RTC_DCHECK(coefficients);
>   RTC_DCHECK_GT(coefficients_length, 0);
>@@ -48,8 +47,8 @@ int WPDNode::Update(const float* parent_data, size_t parent_data_length) {
> 
>   // Decimate data.
>   const bool kOddSequence = true;
>-  size_t output_samples = DyadicDecimate(
>-      data_.get(), parent_data_length, kOddSequence, data_.get(), length_);
>+  size_t output_samples = DyadicDecimate(data_.get(), parent_data_length,
>+                                         kOddSequence, data_.get(), length_);
>   if (output_samples != length_) {
>     return -1;
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/wpd_node.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/wpd_node.h
>index 117a6338a02..6a52fb7284d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/wpd_node.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/wpd_node.h
>@@ -13,8 +13,6 @@
> 
> #include <memory>
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>-
> namespace webrtc {
> 
> class FIRFilter;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/wpd_node_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/wpd_node_unittest.cc
>index 19293610555..5f9238255c2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/wpd_node_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/wpd_node_unittest.cc
>@@ -20,18 +20,17 @@ static const size_t kDataLength = 5;
> static const float kTolerance = 0.0001f;
> 
> static const size_t kParentDataLength = kDataLength * 2;
>-static const float kParentData[kParentDataLength] =
>-    {1.f, 2.f, 3.f, 4.f, 5.f, 6.f, 7.f, 8.f, 9.f, 10.f};
>+static const float kParentData[kParentDataLength] = {1.f, 2.f, 3.f, 4.f, 5.f,
>+                                                     6.f, 7.f, 8.f, 9.f, 10.f};
> 
> static const float kCoefficients[] = {0.2f, -0.3f, 0.5f, -0.7f, 0.11f};
>-static const size_t kCoefficientsLength = sizeof(kCoefficients) /
>-                                       sizeof(kCoefficients[0]);
>+static const size_t kCoefficientsLength =
>+    sizeof(kCoefficients) / sizeof(kCoefficients[0]);
> 
> TEST(WPDNodeTest, Accessors) {
>   WPDNode node(kDataLength, kCoefficients, kCoefficientsLength);
>   EXPECT_EQ(0, node.set_data(kParentData, kDataLength));
>-  EXPECT_EQ(0, memcmp(node.data(),
>-                      kParentData,
>+  EXPECT_EQ(0, memcmp(node.data(), kParentData,
>                       kDataLength * sizeof(node.data()[0])));
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/wpd_tree.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/wpd_tree.cc
>index a01b816daf5..72f4d76891f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/wpd_tree.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/wpd_tree.cc
>@@ -19,8 +19,10 @@
> 
> namespace webrtc {
> 
>-WPDTree::WPDTree(size_t data_length, const float* high_pass_coefficients,
>-                 const float* low_pass_coefficients, size_t coefficients_length,
>+WPDTree::WPDTree(size_t data_length,
>+                 const float* high_pass_coefficients,
>+                 const float* low_pass_coefficients,
>+                 size_t coefficients_length,
>                  int levels)
>     : data_length_(data_length),
>       levels_(levels),
>@@ -98,8 +100,8 @@ int WPDTree::Update(const float* data, size_t data_length) {
>       index_left_child = index * 2;
>       index_right_child = index_left_child + 1;
> 
>-      update_result = nodes_[index_left_child]->Update(
>-          nodes_[index]->data(), nodes_[index]->length());
>+      update_result = nodes_[index_left_child]->Update(nodes_[index]->data(),
>+                                                       nodes_[index]->length());
>       if (update_result != 0) {
>         return -1;
>       }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/wpd_tree.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/wpd_tree.h
>index 53fc06b294e..707a89d44ac 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/wpd_tree.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/wpd_tree.h
>@@ -47,9 +47,7 @@ class WPDTree {
>   ~WPDTree();
> 
>   // Returns the number of nodes at any given level.
>-  static int NumberOfNodesAtLevel(int level) {
>-    return 1 << level;
>-  }
>+  static int NumberOfNodesAtLevel(int level) { return 1 << level; }
> 
>   // Returns a pointer to the node at the given level and index(of that level).
>   // Level goes from 0 to levels().
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/wpd_tree_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/wpd_tree_unittest.cc
>index a90af7766bd..88f073963b8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/wpd_tree_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/transient/wpd_tree_unittest.cc
>@@ -16,7 +16,7 @@
> 
> #include "modules/audio_processing/transient/daubechies_8_wavelet_coeffs.h"
> #include "modules/audio_processing/transient/file_utils.h"
>-#include "system_wrappers/include/file_wrapper.h"
>+#include "rtc_base/system/file_wrapper.h"
> #include "test/gtest.h"
> #include "test/testsupport/fileutils.h"
> 
>@@ -30,13 +30,10 @@ TEST(WPDTreeTest, Construction) {
>   float test_buffer[kTestBufferSize];
>   memset(test_buffer, 0.f, kTestBufferSize * sizeof(*test_buffer));
>   float test_coefficients[] = {1.f, 2.f, 3.f, 4.f, 5.f};
>-  const size_t kTestCoefficientsLength = sizeof(test_coefficients) /
>-      sizeof(test_coefficients[0]);
>-  WPDTree tree(kTestBufferSize,
>-               test_coefficients,
>-               test_coefficients,
>-               kTestCoefficientsLength,
>-               kLevels);
>+  const size_t kTestCoefficientsLength =
>+      sizeof(test_coefficients) / sizeof(test_coefficients[0]);
>+  WPDTree tree(kTestBufferSize, test_coefficients, test_coefficients,
>+               kTestCoefficientsLength, kLevels);
>   ASSERT_EQ(kExpectedNumberOfNodes, tree.num_nodes());
>   // Checks for NodeAt(level, index).
>   int nodes_at_level = 0;
>@@ -79,10 +76,8 @@ TEST(WPDTreeTest, CorrectnessBasedOnMatlabFiles) {
>   const int kLeaves = 1 << kLevels;
>   const size_t kLeavesSamples = kTestBufferSize >> kLevels;
>   // Create tree with Discrete Meyer Wavelet Coefficients.
>-  WPDTree tree(kTestBufferSize,
>-               kDaubechies8HighPassCoefficients,
>-               kDaubechies8LowPassCoefficients,
>-               kDaubechies8CoefficientsLength,
>+  WPDTree tree(kTestBufferSize, kDaubechies8HighPassCoefficients,
>+               kDaubechies8LowPassCoefficients, kDaubechies8CoefficientsLength,
>                kLevels);
>   // Allocate and open all matlab and out files.
>   std::unique_ptr<FileWrapper> matlab_files_data[kLeaves];
>@@ -134,9 +129,8 @@ TEST(WPDTreeTest, CorrectnessBasedOnMatlabFiles) {
>   size_t frames_read = 0;
> 
>   // Read first buffer from the PCM test file.
>-  size_t file_samples_read = ReadInt16FromFileToFloatBuffer(test_file.get(),
>-                                                            kTestBufferSize,
>-                                                            test_buffer);
>+  size_t file_samples_read = ReadInt16FromFileToFloatBuffer(
>+      test_file.get(), kTestBufferSize, test_buffer);
>   while (file_samples_read > 0 && frames_read < kMaxFramesToTest) {
>     ++frames_read;
> 
>@@ -152,10 +146,8 @@ TEST(WPDTreeTest, CorrectnessBasedOnMatlabFiles) {
>     // Compare results with data from the matlab test files.
>     for (int i = 0; i < kLeaves; ++i) {
>       // Compare data values
>-      size_t matlab_samples_read =
>-          ReadDoubleBufferFromFile(matlab_files_data[i].get(),
>-                                   kLeavesSamples,
>-                                   matlab_buffer);
>+      size_t matlab_samples_read = ReadDoubleBufferFromFile(
>+          matlab_files_data[i].get(), kLeavesSamples, matlab_buffer);
> 
>       ASSERT_EQ(kLeavesSamples, matlab_samples_read)
>           << "Matlab test files are malformed.\n"
>@@ -170,15 +162,13 @@ TEST(WPDTreeTest, CorrectnessBasedOnMatlabFiles) {
>       }
> 
>       // Write results to out files.
>-      WriteFloatBufferToFile(out_files_data[i].get(),
>-                             kLeavesSamples,
>+      WriteFloatBufferToFile(out_files_data[i].get(), kLeavesSamples,
>                              node_data);
>     }
> 
>     // Read next buffer from the PCM test file.
>-    file_samples_read = ReadInt16FromFileToFloatBuffer(test_file.get(),
>-                                                       kTestBufferSize,
>-                                                       test_buffer);
>+    file_samples_read = ReadInt16FromFileToFloatBuffer(
>+        test_file.get(), kTestBufferSize, test_buffer);
>   }
> 
>   // Close all matlab and out files.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/typing_detection.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/typing_detection.cc
>index 6e18124aadf..e725b264ee0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/typing_detection.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/typing_detection.cc
>@@ -24,8 +24,7 @@ TypingDetection::TypingDetection()
>       reporting_threshold_(300),
>       penalty_decay_(1),
>       type_event_delay_(2),
>-      report_detection_update_period_(1) {
>-}
>+      report_detection_update_period_(1) {}
> 
> TypingDetection::~TypingDetection() {}
> 
>@@ -41,8 +40,7 @@ bool TypingDetection::Process(bool key_pressed, bool vad_activity) {
>   else
>     ++time_since_last_typing_;
> 
>-  if (time_since_last_typing_ < type_event_delay_ &&
>-      vad_activity &&
>+  if (time_since_last_typing_ < type_event_delay_ && vad_activity &&
>       time_active_ < time_window_) {
>     penalty_counter_ += cost_per_typing_;
>     if (penalty_counter_ > reporting_threshold_)
>@@ -73,15 +71,20 @@ void TypingDetection::SetParameters(int time_window,
>                                     int penalty_decay,
>                                     int type_event_delay,
>                                     int report_detection_update_period) {
>-  if (time_window) time_window_ = time_window;
>+  if (time_window)
>+    time_window_ = time_window;
> 
>-  if (cost_per_typing) cost_per_typing_ = cost_per_typing;
>+  if (cost_per_typing)
>+    cost_per_typing_ = cost_per_typing;
> 
>-  if (reporting_threshold) reporting_threshold_ = reporting_threshold;
>+  if (reporting_threshold)
>+    reporting_threshold_ = reporting_threshold;
> 
>-  if (penalty_decay) penalty_decay_ = penalty_decay;
>+  if (penalty_decay)
>+    penalty_decay_ = penalty_decay;
> 
>-  if (type_event_delay) type_event_delay_ = type_event_delay;
>+  if (type_event_delay)
>+    type_event_delay_ = type_event_delay;
> 
>   if (report_detection_update_period)
>     report_detection_update_period_ = report_detection_update_period;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/typing_detection.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/typing_detection.h
>index fe74a5956e5..70fd9033661 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/typing_detection.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/typing_detection.h
>@@ -11,9 +11,6 @@
> #ifndef MODULES_AUDIO_PROCESSING_TYPING_DETECTION_H_
> #define MODULES_AUDIO_PROCESSING_TYPING_DETECTION_H_
> 
>-#include "modules/include/module_common_types.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>-
> namespace webrtc {
> 
> class TypingDetection {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/BUILD.gn
>new file mode 100644
>index 00000000000..6a4304d1921
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/BUILD.gn
>@@ -0,0 +1,114 @@
>+# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
>+#
>+# Use of this source code is governed by a BSD-style license
>+# that can be found in the LICENSE file in the root of the source
>+# tree. An additional intellectual property rights grant can be found
>+# in the file PATENTS.  All contributing project authors may
>+# be found in the AUTHORS file in the root of the source tree.
>+
>+import("../../../webrtc.gni")
>+
>+rtc_source_set("block_mean_calculator") {
>+  sources = [
>+    "block_mean_calculator.cc",
>+    "block_mean_calculator.h",
>+  ]
>+  deps = [
>+    "../../../rtc_base:checks",
>+    "../../../rtc_base:rtc_base_approved",
>+  ]
>+}
>+
>+rtc_source_set("legacy_delay_estimator") {
>+  sources = [
>+    "delay_estimator.cc",
>+    "delay_estimator.h",
>+    "delay_estimator_internal.h",
>+    "delay_estimator_wrapper.cc",
>+    "delay_estimator_wrapper.h",
>+  ]
>+  deps = [
>+    "../../../rtc_base:checks",
>+  ]
>+}
>+
>+rtc_source_set("ooura_fft") {
>+  sources = [
>+    "ooura_fft.cc",
>+    "ooura_fft.h",
>+    "ooura_fft_tables_common.h",
>+  ]
>+  deps = [
>+    "../../../rtc_base/system:arch",
>+    "../../../system_wrappers:cpu_features_api",
>+  ]
>+  cflags = []
>+
>+  if (current_cpu == "x86" || current_cpu == "x64") {
>+    sources += [
>+      "ooura_fft_sse2.cc",
>+      "ooura_fft_tables_neon_sse2.h",
>+    ]
>+    if (is_posix || is_fuchsia) {
>+      cflags += [ "-msse2" ]
>+    }
>+  }
>+
>+  if (rtc_build_with_neon) {
>+    sources += [
>+      "ooura_fft_neon.cc",
>+      "ooura_fft_tables_neon_sse2.h",
>+    ]
>+
>+    deps += [ "../../../common_audio" ]
>+
>+    if (current_cpu != "arm64") {
>+      # Enable compilation for the NEON instruction set.
>+      suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
>+      cflags += [ "-mfpu=neon" ]
>+    }
>+
>+    # Disable LTO on NEON targets due to compiler bug.
>+    # TODO(fdegans): Enable this. See crbug.com/408997.
>+    if (rtc_use_lto) {
>+      cflags -= [
>+        "-flto",
>+        "-ffat-lto-objects",
>+      ]
>+    }
>+  }
>+
>+  if (current_cpu == "mipsel" && mips_float_abi == "hard") {
>+    sources += [ "ooura_fft_mips.cc" ]
>+  }
>+}
>+
>+if (rtc_include_tests) {
>+  rtc_source_set("block_mean_calculator_unittest") {
>+    testonly = true
>+
>+    sources = [
>+      "block_mean_calculator_unittest.cc",
>+    ]
>+    deps = [
>+      ":block_mean_calculator",
>+      "../../../rtc_base:rtc_base_approved",
>+      "../../../test:test_support",
>+      "//testing/gtest",
>+    ]
>+  }
>+
>+  rtc_source_set("legacy_delay_estimator_unittest") {
>+    testonly = true
>+
>+    sources = [
>+      "delay_estimator_unittest.cc",
>+    ]
>+    deps = [
>+      ":legacy_delay_estimator",
>+      "../../../rtc_base:rtc_base_approved",
>+      "../../../test:test_support",
>+      "//testing/gtest",
>+    ]
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/block_mean_calculator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/block_mean_calculator.cc
>index 3d766929c6a..82c1c0face9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/block_mean_calculator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/block_mean_calculator.cc
>@@ -15,10 +15,7 @@
> namespace webrtc {
> 
> BlockMeanCalculator::BlockMeanCalculator(size_t block_length)
>-    : block_length_(block_length),
>-      count_(0),
>-      sum_(0.0),
>-      mean_(0.0) {
>+    : block_length_(block_length), count_(0), sum_(0.0), mean_(0.0) {
>   RTC_DCHECK(block_length_ != 0);
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/delay_estimator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/delay_estimator.cc
>index 871b5416513..a15b91406c5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/delay_estimator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/delay_estimator.cc
>@@ -21,9 +21,9 @@
> static const int kShiftsAtZero = 13;  // Right shifts at zero binary spectrum.
> static const int kShiftsLinearSlope = 3;
> 
>-static const int32_t kProbabilityOffset = 1024;  // 2 in Q9.
>+static const int32_t kProbabilityOffset = 1024;      // 2 in Q9.
> static const int32_t kProbabilityLowerLimit = 8704;  // 17 in Q9.
>-static const int32_t kProbabilityMinSpread = 2816;  // 5.5 in Q9.
>+static const int32_t kProbabilityMinSpread = 2816;   // 5.5 in Q9.
> 
> // Robust validation settings
> static const float kHistogramMax = 3000.f;
>@@ -39,13 +39,13 @@ static const float kMinFractionWhenPossiblyNonCausal = 0.25f;
> 
> // Counts and returns number of bits of a 32-bit word.
> static int BitCount(uint32_t u32) {
>-  uint32_t tmp = u32 - ((u32 >> 1) & 033333333333) -
>-      ((u32 >> 2) & 011111111111);
>+  uint32_t tmp =
>+      u32 - ((u32 >> 1) & 033333333333) - ((u32 >> 2) & 011111111111);
>   tmp = ((tmp + (tmp >> 3)) & 030707070707);
>   tmp = (tmp + (tmp >> 6));
>   tmp = (tmp + (tmp >> 12) + (tmp >> 24)) & 077;
> 
>-  return ((int) tmp);
>+  return ((int)tmp);
> }
> 
> // Compares the |binary_vector| with all rows of the |binary_matrix| and counts
>@@ -69,7 +69,7 @@ static void BitCountComparison(uint32_t binary_vector,
> 
>   // Compare |binary_vector| with all rows of the |binary_matrix|
>   for (; n < matrix_size; n++) {
>-    bit_counts[n] = (int32_t) BitCount(binary_vector ^ binary_matrix[n]);
>+    bit_counts[n] = (int32_t)BitCount(binary_vector ^ binary_matrix[n]);
>   }
> }
> 
>@@ -96,8 +96,9 @@ static void UpdateRobustValidationStatistics(BinaryDelayEstimator* self,
>                                              int32_t valley_level_q14) {
>   const float valley_depth = valley_depth_q14 * kQ14Scaling;
>   float decrease_in_last_set = valley_depth;
>-  const int max_hits_for_slow_change = (candidate_delay < self->last_delay) ?
>-      kMaxHitsWhenPossiblyNonCausal : kMaxHitsWhenPossiblyCausal;
>+  const int max_hits_for_slow_change = (candidate_delay < self->last_delay)
>+                                           ? kMaxHitsWhenPossiblyNonCausal
>+                                           : kMaxHitsWhenPossiblyCausal;
>   int i = 0;
> 
>   RTC_DCHECK_EQ(self->history_size, self->farend->history_size);
>@@ -127,18 +128,20 @@ static void UpdateRobustValidationStatistics(BinaryDelayEstimator* self,
>   //    |candidate_delay| is a "potential" candidate and we start decreasing
>   //    these histogram bins more rapidly with |valley_depth|.
>   if (self->candidate_hits < max_hits_for_slow_change) {
>-    decrease_in_last_set = (self->mean_bit_counts[self->compare_delay] -
>-        valley_level_q14) * kQ14Scaling;
>+    decrease_in_last_set =
>+        (self->mean_bit_counts[self->compare_delay] - valley_level_q14) *
>+        kQ14Scaling;
>   }
>   // 4. All other bins are decreased with |valley_depth|.
>   // TODO(bjornv): Investigate how to make this loop more efficient.  Split up
>   // the loop?  Remove parts that doesn't add too much.
>   for (i = 0; i < self->history_size; ++i) {
>     int is_in_last_set = (i >= self->last_delay - 2) &&
>-        (i <= self->last_delay + 1) && (i != candidate_delay);
>-    int is_in_candidate_set = (i >= candidate_delay - 2) &&
>-        (i <= candidate_delay + 1);
>-    self->histogram[i] -= decrease_in_last_set * is_in_last_set +
>+                         (i <= self->last_delay + 1) && (i != candidate_delay);
>+    int is_in_candidate_set =
>+        (i >= candidate_delay - 2) && (i <= candidate_delay + 1);
>+    self->histogram[i] -=
>+        decrease_in_last_set * is_in_last_set +
>         valley_depth * (!is_in_last_set && !is_in_candidate_set);
>     // 5. No histogram bin can go below 0.
>     if (self->histogram[i] < 0) {
>@@ -196,16 +199,18 @@ static int HistogramBasedValidation(const BinaryDelayEstimator* self,
>   // into tables?
>   if (delay_difference > self->allowed_offset) {
>     fraction = 1.f - kFractionSlope * (delay_difference - self->allowed_offset);
>-    fraction = (fraction > kMinFractionWhenPossiblyCausal ? fraction :
>-        kMinFractionWhenPossiblyCausal);
>+    fraction = (fraction > kMinFractionWhenPossiblyCausal
>+                    ? fraction
>+                    : kMinFractionWhenPossiblyCausal);
>   } else if (delay_difference < 0) {
>-    fraction = kMinFractionWhenPossiblyNonCausal -
>-        kFractionSlope * delay_difference;
>+    fraction =
>+        kMinFractionWhenPossiblyNonCausal - kFractionSlope * delay_difference;
>     fraction = (fraction > 1.f ? 1.f : fraction);
>   }
>   histogram_threshold *= fraction;
>-  histogram_threshold = (histogram_threshold > kMinHistogramThreshold ?
>-      histogram_threshold : kMinHistogramThreshold);
>+  histogram_threshold =
>+      (histogram_threshold > kMinHistogramThreshold ? histogram_threshold
>+                                                    : kMinHistogramThreshold);
> 
>   is_histogram_valid =
>       (self->histogram[candidate_delay] >= histogram_threshold) &&
>@@ -243,8 +248,8 @@ static int RobustValidation(const BinaryDelayEstimator* self,
>   //   i) Before we actually have a valid estimate (|last_delay| == -2), we say
>   //      a candidate is valid if either algorithm states so
>   //      (|is_instantaneous_valid| OR |is_histogram_valid|).
>-  is_robust = (self->last_delay < 0) &&
>-      (is_instantaneous_valid || is_histogram_valid);
>+  is_robust =
>+      (self->last_delay < 0) && (is_instantaneous_valid || is_histogram_valid);
>   //  ii) Otherwise, we need both algorithms to be certain
>   //      (|is_instantaneous_valid| AND |is_histogram_valid|)
>   is_robust |= is_instantaneous_valid && is_histogram_valid;
>@@ -252,13 +257,12 @@ static int RobustValidation(const BinaryDelayEstimator* self,
>   //      the instantaneous one if |is_histogram_valid| = 1 and the histogram
>   //      is significantly strong.
>   is_robust |= is_histogram_valid &&
>-      (self->histogram[candidate_delay] > self->last_delay_histogram);
>+               (self->histogram[candidate_delay] > self->last_delay_histogram);
> 
>   return is_robust;
> }
> 
> void WebRtc_FreeBinaryDelayEstimatorFarend(BinaryDelayEstimatorFarend* self) {
>-
>   if (self == NULL) {
>     return;
>   }
>@@ -302,20 +306,17 @@ int WebRtc_AllocateFarendBufferMemory(BinaryDelayEstimatorFarend* self,
>   self->binary_far_history = static_cast<uint32_t*>(
>       realloc(self->binary_far_history,
>               history_size * sizeof(*self->binary_far_history)));
>-  self->far_bit_counts = static_cast<int*>(
>-      realloc(self->far_bit_counts,
>-              history_size * sizeof(*self->far_bit_counts)));
>+  self->far_bit_counts = static_cast<int*>(realloc(
>+      self->far_bit_counts, history_size * sizeof(*self->far_bit_counts)));
>   if ((self->binary_far_history == NULL) || (self->far_bit_counts == NULL)) {
>     history_size = 0;
>   }
>   // Fill with zeros if we have expanded the buffers.
>   if (history_size > self->history_size) {
>     int size_diff = history_size - self->history_size;
>-    memset(&self->binary_far_history[self->history_size],
>-           0,
>+    memset(&self->binary_far_history[self->history_size], 0,
>            sizeof(*self->binary_far_history) * size_diff);
>-    memset(&self->far_bit_counts[self->history_size],
>-           0,
>+    memset(&self->far_bit_counts[self->history_size], 0,
>            sizeof(*self->far_bit_counts) * size_diff);
>   }
>   self->history_size = history_size;
>@@ -330,7 +331,8 @@ void WebRtc_InitBinaryDelayEstimatorFarend(BinaryDelayEstimatorFarend* self) {
> }
> 
> void WebRtc_SoftResetBinaryDelayEstimatorFarend(
>-    BinaryDelayEstimatorFarend* self, int delay_shift) {
>+    BinaryDelayEstimatorFarend* self,
>+    int delay_shift) {
>   int abs_shift = abs(delay_shift);
>   int shift_size = 0;
>   int dest_index = 0;
>@@ -355,8 +357,7 @@ void WebRtc_SoftResetBinaryDelayEstimatorFarend(
>           sizeof(*self->binary_far_history) * shift_size);
>   memset(&self->binary_far_history[padding_index], 0,
>          sizeof(*self->binary_far_history) * abs_shift);
>-  memmove(&self->far_bit_counts[dest_index],
>-          &self->far_bit_counts[src_index],
>+  memmove(&self->far_bit_counts[dest_index], &self->far_bit_counts[src_index],
>           sizeof(*self->far_bit_counts) * shift_size);
>   memset(&self->far_bit_counts[padding_index], 0,
>          sizeof(*self->far_bit_counts) * abs_shift);
>@@ -378,7 +379,6 @@ void WebRtc_AddBinaryFarSpectrum(BinaryDelayEstimatorFarend* handle,
> }
> 
> void WebRtc_FreeBinaryDelayEstimator(BinaryDelayEstimator* self) {
>-
>   if (self == NULL) {
>     return;
>   }
>@@ -403,7 +403,8 @@ void WebRtc_FreeBinaryDelayEstimator(BinaryDelayEstimator* self) {
> }
> 
> BinaryDelayEstimator* WebRtc_CreateBinaryDelayEstimator(
>-    BinaryDelayEstimatorFarend* farend, int max_lookahead) {
>+    BinaryDelayEstimatorFarend* farend,
>+    int max_lookahead) {
>   BinaryDelayEstimator* self = NULL;
> 
>   if ((farend != NULL) && (max_lookahead >= 0)) {
>@@ -457,22 +458,18 @@ int WebRtc_AllocateHistoryBufferMemory(BinaryDelayEstimator* self,
>   self->histogram = static_cast<float*>(
>       realloc(self->histogram, (history_size + 1) * sizeof(*self->histogram)));
> 
>-  if ((self->mean_bit_counts == NULL) ||
>-      (self->bit_counts == NULL) ||
>+  if ((self->mean_bit_counts == NULL) || (self->bit_counts == NULL) ||
>       (self->histogram == NULL)) {
>     history_size = 0;
>   }
>   // Fill with zeros if we have expanded the buffers.
>   if (history_size > self->history_size) {
>     int size_diff = history_size - self->history_size;
>-    memset(&self->mean_bit_counts[self->history_size],
>-           0,
>+    memset(&self->mean_bit_counts[self->history_size], 0,
>            sizeof(*self->mean_bit_counts) * size_diff);
>-    memset(&self->bit_counts[self->history_size],
>-           0,
>+    memset(&self->bit_counts[self->history_size], 0,
>            sizeof(*self->bit_counts) * size_diff);
>-    memset(&self->histogram[self->history_size],
>-           0,
>+    memset(&self->histogram[self->history_size], 0,
>            sizeof(*self->histogram) * size_diff);
>   }
>   self->history_size = history_size;
>@@ -485,15 +482,14 @@ void WebRtc_InitBinaryDelayEstimator(BinaryDelayEstimator* self) {
>   RTC_DCHECK(self);
> 
>   memset(self->bit_counts, 0, sizeof(int32_t) * self->history_size);
>-  memset(self->binary_near_history,
>-         0,
>+  memset(self->binary_near_history, 0,
>          sizeof(uint32_t) * self->near_history_size);
>   for (i = 0; i <= self->history_size; ++i) {
>     self->mean_bit_counts[i] = (20 << 9);  // 20 in Q9.
>     self->histogram[i] = 0.f;
>   }
>-  self->minimum_probability = kMaxBitCountsQ9;  // 32 in Q9.
>-  self->last_delay_probability = (int) kMaxBitCountsQ9;  // 32 in Q9.
>+  self->minimum_probability = kMaxBitCountsQ9;          // 32 in Q9.
>+  self->last_delay_probability = (int)kMaxBitCountsQ9;  // 32 in Q9.
> 
>   // Default return value if we're unable to estimate. -1 is used for errors.
>   self->last_delay = -2;
>@@ -617,8 +613,8 @@ int WebRtc_ProcessBinarySpectrum(BinaryDelayEstimator* self,
>   //     and deeper than the best estimate so far
>   //      (|value_best_candidate| < |last_delay_probability|)
>   valid_candidate = ((valley_depth > kProbabilityOffset) &&
>-      ((value_best_candidate < self->minimum_probability) ||
>-          (value_best_candidate < self->last_delay_probability)));
>+                     ((value_best_candidate < self->minimum_probability) ||
>+                      (value_best_candidate < self->last_delay_probability)));
> 
>   // Check for nonstationary farend signal.
>   const bool non_stationary_farend =
>@@ -637,7 +633,6 @@ int WebRtc_ProcessBinarySpectrum(BinaryDelayEstimator* self,
>     int is_histogram_valid = HistogramBasedValidation(self, candidate_delay);
>     valid_candidate = RobustValidation(self, candidate_delay, valid_candidate,
>                                        is_histogram_valid);
>-
>   }
> 
>   // Only update the delay estimate when the farend is nonstationary and when
>@@ -645,8 +640,9 @@ int WebRtc_ProcessBinarySpectrum(BinaryDelayEstimator* self,
>   if (non_stationary_farend && valid_candidate) {
>     if (candidate_delay != self->last_delay) {
>       self->last_delay_histogram =
>-          (self->histogram[candidate_delay] > kLastHistogramMax ?
>-              kLastHistogramMax : self->histogram[candidate_delay]);
>+          (self->histogram[candidate_delay] > kLastHistogramMax
>+               ? kLastHistogramMax
>+               : self->histogram[candidate_delay]);
>       // Adjust the histogram if we made a change to |last_delay|, though it was
>       // not the most likely one according to the histogram.
>       if (self->histogram[candidate_delay] <
>@@ -679,8 +675,8 @@ float WebRtc_binary_last_delay_quality(BinaryDelayEstimator* self) {
>   } else {
>     // Note that |last_delay_probability| states how deep the minimum of the
>     // cost function is, so it is rather an error probability.
>-    quality = (float) (kMaxBitCountsQ9 - self->last_delay_probability) /
>-        kMaxBitCountsQ9;
>+    quality = (float)(kMaxBitCountsQ9 - self->last_delay_probability) /
>+              kMaxBitCountsQ9;
>     if (quality < 0) {
>       quality = 0;
>     }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/delay_estimator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/delay_estimator.h
>index cce6113a539..2f47e26f36b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/delay_estimator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/delay_estimator.h
>@@ -14,7 +14,7 @@
> #ifndef MODULES_AUDIO_PROCESSING_UTILITY_DELAY_ESTIMATOR_H_
> #define MODULES_AUDIO_PROCESSING_UTILITY_DELAY_ESTIMATOR_H_
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include <stdint.h>
> 
> static const int32_t kMaxBitCountsQ9 = (32 << 9);  // 32 matching bits in Q9.
> 
>@@ -117,7 +117,8 @@ void WebRtc_InitBinaryDelayEstimatorFarend(BinaryDelayEstimatorFarend* self);
> //    - delay_shift   : The amount of blocks to shift history buffers.
> //
> void WebRtc_SoftResetBinaryDelayEstimatorFarend(
>-    BinaryDelayEstimatorFarend* self, int delay_shift);
>+    BinaryDelayEstimatorFarend* self,
>+    int delay_shift);
> 
> // Adds the binary far-end spectrum to the internal far-end history buffer. This
> // spectrum is used as reference when calculating the delay using
>@@ -153,7 +154,8 @@ void WebRtc_FreeBinaryDelayEstimator(BinaryDelayEstimator* self);
> // See WebRtc_CreateDelayEstimator(..) in delay_estimator_wrapper.c for detailed
> // description.
> BinaryDelayEstimator* WebRtc_CreateBinaryDelayEstimator(
>-    BinaryDelayEstimatorFarend* farend, int max_lookahead);
>+    BinaryDelayEstimatorFarend* farend,
>+    int max_lookahead);
> 
> // Re-allocates |history_size| dependent buffers. The far-end buffers will be
> // updated at the same time if needed.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/delay_estimator_internal.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/delay_estimator_internal.h
>index 46eea3ec185..e99fe21a854 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/delay_estimator_internal.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/delay_estimator_internal.h
>@@ -14,7 +14,6 @@
> #define MODULES_AUDIO_PROCESSING_UTILITY_DELAY_ESTIMATOR_INTERNAL_H_
> 
> #include "modules/audio_processing/utility/delay_estimator.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> typedef union {
>   float float_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/delay_estimator_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/delay_estimator_unittest.cc
>index 36700e57066..324bc37a9e9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/delay_estimator_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/delay_estimator_unittest.cc
>@@ -12,7 +12,6 @@
> #include "modules/audio_processing/utility/delay_estimator_internal.h"
> #include "modules/audio_processing/utility/delay_estimator_wrapper.h"
> #include "test/gtest.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace {
> 
>@@ -27,23 +26,27 @@ enum { kSequenceLength = 400 };
> const int kDifferentHistorySize = 3;
> const int kDifferentLookahead = 1;
> 
>-const int kEnable[] = { 0, 1 };
>+const int kEnable[] = {0, 1};
> const size_t kSizeEnable = sizeof(kEnable) / sizeof(*kEnable);
> 
> class DelayEstimatorTest : public ::testing::Test {
>  protected:
>   DelayEstimatorTest();
>-  virtual void SetUp();
>-  virtual void TearDown();
>+  void SetUp() override;
>+  void TearDown() override;
> 
>   void Init();
>   void InitBinary();
>   void VerifyDelay(BinaryDelayEstimator* binary_handle, int offset, int delay);
>   void RunBinarySpectra(BinaryDelayEstimator* binary1,
>                         BinaryDelayEstimator* binary2,
>-                        int near_offset, int lookahead_offset, int far_offset);
>-  void RunBinarySpectraTest(int near_offset, int lookahead_offset,
>-                            int ref_robust_validation, int robust_validation);
>+                        int near_offset,
>+                        int lookahead_offset,
>+                        int far_offset);
>+  void RunBinarySpectraTest(int near_offset,
>+                            int lookahead_offset,
>+                            int ref_robust_validation,
>+                            int robust_validation);
> 
>   void* handle_;
>   DelayEstimator* self_;
>@@ -83,8 +86,8 @@ DelayEstimatorTest::DelayEstimatorTest()
> }
> 
> void DelayEstimatorTest::SetUp() {
>-  farend_handle_ = WebRtc_CreateDelayEstimatorFarend(kSpectrumSize,
>-                                                     kHistorySize);
>+  farend_handle_ =
>+      WebRtc_CreateDelayEstimatorFarend(kSpectrumSize, kHistorySize);
>   ASSERT_TRUE(farend_handle_ != NULL);
>   farend_self_ = reinterpret_cast<DelayEstimatorFarend*>(farend_handle_);
>   handle_ = WebRtc_CreateDelayEstimator(farend_handle_, kLookahead);
>@@ -131,7 +134,8 @@ void DelayEstimatorTest::InitBinary() {
> }
> 
> void DelayEstimatorTest::VerifyDelay(BinaryDelayEstimator* binary_handle,
>-                                     int offset, int delay) {
>+                                     int offset,
>+                                     int delay) {
>   // Verify that we WebRtc_binary_last_delay() returns correct delay.
>   EXPECT_EQ(delay, WebRtc_binary_last_delay(binary_handle));
> 
>@@ -147,8 +151,8 @@ void DelayEstimatorTest::RunBinarySpectra(BinaryDelayEstimator* binary1,
>                                           int near_offset,
>                                           int lookahead_offset,
>                                           int far_offset) {
>-  int different_validations = binary1->robust_validation_enabled ^
>-      binary2->robust_validation_enabled;
>+  int different_validations =
>+      binary1->robust_validation_enabled ^ binary2->robust_validation_enabled;
>   WebRtc_InitBinaryDelayEstimatorFarend(binary_farend_);
>   WebRtc_InitBinaryDelayEstimator(binary1);
>   WebRtc_InitBinaryDelayEstimator(binary2);
>@@ -160,9 +164,8 @@ void DelayEstimatorTest::RunBinarySpectra(BinaryDelayEstimator* binary1,
>     WebRtc_AddBinaryFarSpectrum(binary_farend_,
>                                 binary_spectrum_[i + far_offset]);
>     int delay_1 = WebRtc_ProcessBinarySpectrum(binary1, binary_spectrum_[i]);
>-    int delay_2 =
>-        WebRtc_ProcessBinarySpectrum(binary2,
>-                                     binary_spectrum_[i - near_offset]);
>+    int delay_2 = WebRtc_ProcessBinarySpectrum(
>+        binary2, binary_spectrum_[i - near_offset]);
> 
>     VerifyDelay(binary1, far_offset + kLookahead, delay_1);
>     VerifyDelay(binary2,
>@@ -177,7 +180,7 @@ void DelayEstimatorTest::RunBinarySpectra(BinaryDelayEstimator* binary1,
>     // all the time, unless one of them has robust validation turned on.  In
>     // that case the robust validation leaves the initial state faster.
>     if ((near_offset == 0) && (lookahead_offset == 0)) {
>-      if  (!different_validations) {
>+      if (!different_validations) {
>         EXPECT_EQ(delay_1, delay_2);
>       } else {
>         if (binary1->robust_validation_enabled) {
>@@ -199,9 +202,8 @@ void DelayEstimatorTest::RunBinarySpectraTest(int near_offset,
>                                               int lookahead_offset,
>                                               int ref_robust_validation,
>                                               int robust_validation) {
>-  BinaryDelayEstimator* binary2 =
>-      WebRtc_CreateBinaryDelayEstimator(binary_farend_,
>-                                        kLookahead + lookahead_offset);
>+  BinaryDelayEstimator* binary2 = WebRtc_CreateBinaryDelayEstimator(
>+      binary_farend_, kLookahead + lookahead_offset);
>   // Verify the delay for both causal and non-causal systems. For causal systems
>   // the delay is equivalent with a positive |offset| of the far-end sequence.
>   // For non-causal systems the delay is equivalent with a negative |offset| of
>@@ -209,8 +211,7 @@ void DelayEstimatorTest::RunBinarySpectraTest(int near_offset,
>   binary_->robust_validation_enabled = ref_robust_validation;
>   binary2->robust_validation_enabled = robust_validation;
>   for (int offset = -kLookahead;
>-      offset < kMaxDelay - lookahead_offset - near_offset;
>-      offset++) {
>+       offset < kMaxDelay - lookahead_offset - near_offset; offset++) {
>     RunBinarySpectra(binary_, binary2, near_offset, lookahead_offset, offset);
>   }
>   WebRtc_FreeBinaryDelayEstimator(binary2);
>@@ -248,8 +249,8 @@ TEST_F(DelayEstimatorTest, CorrectErrorReturnsOfWrapper) {
>   // 3) Incorrect spectrum size.
>   EXPECT_EQ(-1, WebRtc_AddFarSpectrumFloat(NULL, far_f_, spectrum_size_));
>   // Use |farend_handle_| which is properly created at SetUp().
>-  EXPECT_EQ(-1, WebRtc_AddFarSpectrumFloat(farend_handle_, NULL,
>-                                           spectrum_size_));
>+  EXPECT_EQ(-1,
>+            WebRtc_AddFarSpectrumFloat(farend_handle_, NULL, spectrum_size_));
>   EXPECT_EQ(-1, WebRtc_AddFarSpectrumFloat(farend_handle_, far_f_,
>                                            spectrum_size_ + 1));
> 
>@@ -259,8 +260,8 @@ TEST_F(DelayEstimatorTest, CorrectErrorReturnsOfWrapper) {
>   // 3) Incorrect spectrum size.
>   // 4) Too high precision in far-end spectrum (Q-domain > 15).
>   EXPECT_EQ(-1, WebRtc_AddFarSpectrumFix(NULL, far_u16_, spectrum_size_, 0));
>-  EXPECT_EQ(-1, WebRtc_AddFarSpectrumFix(farend_handle_, NULL, spectrum_size_,
>-                                         0));
>+  EXPECT_EQ(-1,
>+            WebRtc_AddFarSpectrumFix(farend_handle_, NULL, spectrum_size_, 0));
>   EXPECT_EQ(-1, WebRtc_AddFarSpectrumFix(farend_handle_, far_u16_,
>                                          spectrum_size_ + 1, 0));
>   EXPECT_EQ(-1, WebRtc_AddFarSpectrumFix(farend_handle_, far_u16_,
>@@ -313,16 +314,15 @@ TEST_F(DelayEstimatorTest, CorrectErrorReturnsOfWrapper) {
>   // 3) Incorrect spectrum size.
>   // 4) Non matching history sizes if multiple delay estimators using the same
>   //    far-end reference.
>-  EXPECT_EQ(-1, WebRtc_DelayEstimatorProcessFloat(NULL, near_f_,
>-                                                  spectrum_size_));
>+  EXPECT_EQ(-1,
>+            WebRtc_DelayEstimatorProcessFloat(NULL, near_f_, spectrum_size_));
>   // Use |handle_| which is properly created at SetUp().
>-  EXPECT_EQ(-1, WebRtc_DelayEstimatorProcessFloat(handle_, NULL,
>-                                                  spectrum_size_));
>+  EXPECT_EQ(-1,
>+            WebRtc_DelayEstimatorProcessFloat(handle_, NULL, spectrum_size_));
>   EXPECT_EQ(-1, WebRtc_DelayEstimatorProcessFloat(handle_, near_f_,
>                                                   spectrum_size_ + 1));
>   // |tmp_handle| is already in a non-matching state.
>-  EXPECT_EQ(-1, WebRtc_DelayEstimatorProcessFloat(tmp_handle,
>-                                                  near_f_,
>+  EXPECT_EQ(-1, WebRtc_DelayEstimatorProcessFloat(tmp_handle, near_f_,
>                                                   spectrum_size_));
> 
>   // WebRtc_DelayEstimatorProcessFix() should return -1 if we have:
>@@ -332,19 +332,17 @@ TEST_F(DelayEstimatorTest, CorrectErrorReturnsOfWrapper) {
>   // 4) Too high precision in near-end spectrum (Q-domain > 15).
>   // 5) Non matching history sizes if multiple delay estimators using the same
>   //    far-end reference.
>-  EXPECT_EQ(-1, WebRtc_DelayEstimatorProcessFix(NULL, near_u16_, spectrum_size_,
>-                                                0));
>-  EXPECT_EQ(-1, WebRtc_DelayEstimatorProcessFix(handle_, NULL, spectrum_size_,
>-                                                0));
>+  EXPECT_EQ(
>+      -1, WebRtc_DelayEstimatorProcessFix(NULL, near_u16_, spectrum_size_, 0));
>+  EXPECT_EQ(-1,
>+            WebRtc_DelayEstimatorProcessFix(handle_, NULL, spectrum_size_, 0));
>   EXPECT_EQ(-1, WebRtc_DelayEstimatorProcessFix(handle_, near_u16_,
>                                                 spectrum_size_ + 1, 0));
>   EXPECT_EQ(-1, WebRtc_DelayEstimatorProcessFix(handle_, near_u16_,
>                                                 spectrum_size_, 16));
>   // |tmp_handle| is already in a non-matching state.
>-  EXPECT_EQ(-1, WebRtc_DelayEstimatorProcessFix(tmp_handle,
>-                                                near_u16_,
>-                                                spectrum_size_,
>-                                                0));
>+  EXPECT_EQ(-1, WebRtc_DelayEstimatorProcessFix(tmp_handle, near_u16_,
>+                                                spectrum_size_, 0));
>   WebRtc_FreeDelayEstimator(tmp_handle);
> 
>   // WebRtc_last_delay() should return -1 if we have a NULL pointer as |handle|.
>@@ -382,8 +380,8 @@ TEST_F(DelayEstimatorTest, InitializedSpectrumAfterProcess) {
>   // In this test we verify that the mean spectra are initialized after first
>   // time we call WebRtc_AddFarSpectrum() and Process() respectively. The test
>   // also verifies the state is not left for zero spectra.
>-  const float kZerosFloat[kSpectrumSize] = { 0.0 };
>-  const uint16_t kZerosU16[kSpectrumSize] = { 0 };
>+  const float kZerosFloat[kSpectrumSize] = {0.0};
>+  const uint16_t kZerosU16[kSpectrumSize] = {0};
> 
>   // For floating point operations, process one frame and verify initialization
>   // flag.
>@@ -391,14 +389,14 @@ TEST_F(DelayEstimatorTest, InitializedSpectrumAfterProcess) {
>   EXPECT_EQ(0, WebRtc_AddFarSpectrumFloat(farend_handle_, kZerosFloat,
>                                           spectrum_size_));
>   EXPECT_EQ(0, farend_self_->far_spectrum_initialized);
>-  EXPECT_EQ(0, WebRtc_AddFarSpectrumFloat(farend_handle_, far_f_,
>-                                           spectrum_size_));
>+  EXPECT_EQ(0,
>+            WebRtc_AddFarSpectrumFloat(farend_handle_, far_f_, spectrum_size_));
>   EXPECT_EQ(1, farend_self_->far_spectrum_initialized);
>   EXPECT_EQ(-2, WebRtc_DelayEstimatorProcessFloat(handle_, kZerosFloat,
>                                                   spectrum_size_));
>   EXPECT_EQ(0, self_->near_spectrum_initialized);
>-  EXPECT_EQ(-2, WebRtc_DelayEstimatorProcessFloat(handle_, near_f_,
>-                                                  spectrum_size_));
>+  EXPECT_EQ(
>+      -2, WebRtc_DelayEstimatorProcessFloat(handle_, near_f_, spectrum_size_));
>   EXPECT_EQ(1, self_->near_spectrum_initialized);
> 
>   // For fixed point operations, process one frame and verify initialization
>@@ -407,8 +405,8 @@ TEST_F(DelayEstimatorTest, InitializedSpectrumAfterProcess) {
>   EXPECT_EQ(0, WebRtc_AddFarSpectrumFix(farend_handle_, kZerosU16,
>                                         spectrum_size_, 0));
>   EXPECT_EQ(0, farend_self_->far_spectrum_initialized);
>-  EXPECT_EQ(0, WebRtc_AddFarSpectrumFix(farend_handle_, far_u16_,
>-                                         spectrum_size_, 0));
>+  EXPECT_EQ(
>+      0, WebRtc_AddFarSpectrumFix(farend_handle_, far_u16_, spectrum_size_, 0));
>   EXPECT_EQ(1, farend_self_->far_spectrum_initialized);
>   EXPECT_EQ(-2, WebRtc_DelayEstimatorProcessFix(handle_, kZerosU16,
>                                                 spectrum_size_, 0));
>@@ -429,10 +427,10 @@ TEST_F(DelayEstimatorTest, CorrectLastDelay) {
>   // Floating point operations.
>   Init();
>   for (int i = 0; i < 200; i++) {
>-    EXPECT_EQ(0, WebRtc_AddFarSpectrumFloat(farend_handle_, far_f_,
>-                                            spectrum_size_));
>-    last_delay = WebRtc_DelayEstimatorProcessFloat(handle_, near_f_,
>-                                                   spectrum_size_);
>+    EXPECT_EQ(
>+        0, WebRtc_AddFarSpectrumFloat(farend_handle_, far_f_, spectrum_size_));
>+    last_delay =
>+        WebRtc_DelayEstimatorProcessFloat(handle_, near_f_, spectrum_size_);
>     if (last_delay != -2) {
>       EXPECT_EQ(last_delay, WebRtc_last_delay(handle_));
>       if (!WebRtc_is_robust_validation_enabled(handle_)) {
>@@ -451,8 +449,8 @@ TEST_F(DelayEstimatorTest, CorrectLastDelay) {
>   for (int i = 0; i < 200; i++) {
>     EXPECT_EQ(0, WebRtc_AddFarSpectrumFix(farend_handle_, far_u16_,
>                                           spectrum_size_, 0));
>-    last_delay = WebRtc_DelayEstimatorProcessFix(handle_, near_u16_,
>-                                                 spectrum_size_, 0);
>+    last_delay =
>+        WebRtc_DelayEstimatorProcessFix(handle_, near_u16_, spectrum_size_, 0);
>     if (last_delay != -2) {
>       EXPECT_EQ(last_delay, WebRtc_last_delay(handle_));
>       if (!WebRtc_is_robust_validation_enabled(handle_)) {
>@@ -577,8 +575,8 @@ TEST_F(DelayEstimatorTest, AllowedOffsetNoImpactWhenRobustValidationDisabled) {
> }
> 
> TEST_F(DelayEstimatorTest, VerifyLookaheadAtCreate) {
>-  void* farend_handle = WebRtc_CreateDelayEstimatorFarend(kSpectrumSize,
>-                                                          kMaxDelay);
>+  void* farend_handle =
>+      WebRtc_CreateDelayEstimatorFarend(kSpectrumSize, kMaxDelay);
>   ASSERT_TRUE(farend_handle != NULL);
>   void* handle = WebRtc_CreateDelayEstimator(farend_handle, kLookahead);
>   ASSERT_TRUE(handle != NULL);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/delay_estimator_wrapper.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/delay_estimator_wrapper.cc
>index f907c80a355..27c2a3a1a29 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/delay_estimator_wrapper.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/delay_estimator_wrapper.cc
>@@ -72,7 +72,7 @@ static uint32_t BinarySpectrumFix(const uint16_t* spectrum,
>     for (i = kBandFirst; i <= kBandLast; i++) {
>       if (spectrum[i] > 0) {
>         // Convert input spectrum from Q(|q_domain|) to Q15.
>-        int32_t spectrum_q15 = ((int32_t) spectrum[i]) << (15 - q_domain);
>+        int32_t spectrum_q15 = ((int32_t)spectrum[i]) << (15 - q_domain);
>         threshold_spectrum[i].int32_ = (spectrum_q15 >> 1);
>         *threshold_initialized = 1;
>       }
>@@ -80,7 +80,7 @@ static uint32_t BinarySpectrumFix(const uint16_t* spectrum,
>   }
>   for (i = kBandFirst; i <= kBandLast; i++) {
>     // Convert input spectrum from Q(|q_domain|) to Q15.
>-    int32_t spectrum_q15 = ((int32_t) spectrum[i]) << (15 - q_domain);
>+    int32_t spectrum_q15 = ((int32_t)spectrum[i]) << (15 - q_domain);
>     // Update the |threshold_spectrum|.
>     WebRtc_MeanEstimatorFix(spectrum_q15, 6, &(threshold_spectrum[i].int32_));
>     // Convert |spectrum| at current frequency bin to a binary value.
>@@ -123,7 +123,7 @@ static uint32_t BinarySpectrumFloat(const float* spectrum,
> }
> 
> void WebRtc_FreeDelayEstimatorFarend(void* handle) {
>-  DelayEstimatorFarend* self = (DelayEstimatorFarend*) handle;
>+  DelayEstimatorFarend* self = (DelayEstimatorFarend*)handle;
> 
>   if (handle == NULL) {
>     return;
>@@ -158,8 +158,8 @@ void* WebRtc_CreateDelayEstimatorFarend(int spectrum_size, int history_size) {
>     memory_fail |= (self->binary_farend == NULL);
> 
>     // Allocate memory for spectrum buffers.
>-    self->mean_far_spectrum =
>-        static_cast<SpectrumType*>(malloc(spectrum_size * sizeof(SpectrumType)));
>+    self->mean_far_spectrum = static_cast<SpectrumType*>(
>+        malloc(spectrum_size * sizeof(SpectrumType)));
>     memory_fail |= (self->mean_far_spectrum == NULL);
> 
>     self->spectrum_size = spectrum_size;
>@@ -174,7 +174,7 @@ void* WebRtc_CreateDelayEstimatorFarend(int spectrum_size, int history_size) {
> }
> 
> int WebRtc_InitDelayEstimatorFarend(void* handle) {
>-  DelayEstimatorFarend* self = (DelayEstimatorFarend*) handle;
>+  DelayEstimatorFarend* self = (DelayEstimatorFarend*)handle;
> 
>   if (self == NULL) {
>     return -1;
>@@ -193,7 +193,7 @@ int WebRtc_InitDelayEstimatorFarend(void* handle) {
> }
> 
> void WebRtc_SoftResetDelayEstimatorFarend(void* handle, int delay_shift) {
>-  DelayEstimatorFarend* self = (DelayEstimatorFarend*) handle;
>+  DelayEstimatorFarend* self = (DelayEstimatorFarend*)handle;
>   RTC_DCHECK(self);
>   WebRtc_SoftResetBinaryDelayEstimatorFarend(self->binary_farend, delay_shift);
> }
>@@ -202,7 +202,7 @@ int WebRtc_AddFarSpectrumFix(void* handle,
>                              const uint16_t* far_spectrum,
>                              int spectrum_size,
>                              int far_q) {
>-  DelayEstimatorFarend* self = (DelayEstimatorFarend*) handle;
>+  DelayEstimatorFarend* self = (DelayEstimatorFarend*)handle;
>   uint32_t binary_spectrum = 0;
> 
>   if (self == NULL) {
>@@ -232,7 +232,7 @@ int WebRtc_AddFarSpectrumFix(void* handle,
> int WebRtc_AddFarSpectrumFloat(void* handle,
>                                const float* far_spectrum,
>                                int spectrum_size) {
>-  DelayEstimatorFarend* self = (DelayEstimatorFarend*) handle;
>+  DelayEstimatorFarend* self = (DelayEstimatorFarend*)handle;
>   uint32_t binary_spectrum = 0;
> 
>   if (self == NULL) {
>@@ -256,7 +256,7 @@ int WebRtc_AddFarSpectrumFloat(void* handle,
> }
> 
> void WebRtc_FreeDelayEstimator(void* handle) {
>-  DelayEstimator* self = (DelayEstimator*) handle;
>+  DelayEstimator* self = (DelayEstimator*)handle;
> 
>   if (handle == NULL) {
>     return;
>@@ -273,7 +273,7 @@ void WebRtc_FreeDelayEstimator(void* handle) {
> 
> void* WebRtc_CreateDelayEstimator(void* farend_handle, int max_lookahead) {
>   DelayEstimator* self = NULL;
>-  DelayEstimatorFarend* farend = (DelayEstimatorFarend*) farend_handle;
>+  DelayEstimatorFarend* farend = (DelayEstimatorFarend*)farend_handle;
> 
>   if (farend_handle != NULL) {
>     self = static_cast<DelayEstimator*>(malloc(sizeof(DelayEstimator)));
>@@ -304,7 +304,7 @@ void* WebRtc_CreateDelayEstimator(void* farend_handle, int max_lookahead) {
> }
> 
> int WebRtc_InitDelayEstimator(void* handle) {
>-  DelayEstimator* self = (DelayEstimator*) handle;
>+  DelayEstimator* self = (DelayEstimator*)handle;
> 
>   if (self == NULL) {
>     return -1;
>@@ -323,7 +323,7 @@ int WebRtc_InitDelayEstimator(void* handle) {
> }
> 
> int WebRtc_SoftResetDelayEstimator(void* handle, int delay_shift) {
>-  DelayEstimator* self = (DelayEstimator*) handle;
>+  DelayEstimator* self = (DelayEstimator*)handle;
>   RTC_DCHECK(self);
>   return WebRtc_SoftResetBinaryDelayEstimator(self->binary_handle, delay_shift);
> }
>@@ -352,7 +352,7 @@ int WebRtc_history_size(const void* handle) {
> }
> 
> int WebRtc_set_lookahead(void* handle, int lookahead) {
>-  DelayEstimator* self = (DelayEstimator*) handle;
>+  DelayEstimator* self = (DelayEstimator*)handle;
>   RTC_DCHECK(self);
>   RTC_DCHECK(self->binary_handle);
>   if ((lookahead > self->binary_handle->near_history_size - 1) ||
>@@ -364,14 +364,14 @@ int WebRtc_set_lookahead(void* handle, int lookahead) {
> }
> 
> int WebRtc_lookahead(void* handle) {
>-  DelayEstimator* self = (DelayEstimator*) handle;
>+  DelayEstimator* self = (DelayEstimator*)handle;
>   RTC_DCHECK(self);
>   RTC_DCHECK(self->binary_handle);
>   return self->binary_handle->lookahead;
> }
> 
> int WebRtc_set_allowed_offset(void* handle, int allowed_offset) {
>-  DelayEstimator* self = (DelayEstimator*) handle;
>+  DelayEstimator* self = (DelayEstimator*)handle;
> 
>   if ((self == NULL) || (allowed_offset < 0)) {
>     return -1;
>@@ -381,7 +381,7 @@ int WebRtc_set_allowed_offset(void* handle, int allowed_offset) {
> }
> 
> int WebRtc_get_allowed_offset(const void* handle) {
>-  const DelayEstimator* self = (const DelayEstimator*) handle;
>+  const DelayEstimator* self = (const DelayEstimator*)handle;
> 
>   if (self == NULL) {
>     return -1;
>@@ -390,7 +390,7 @@ int WebRtc_get_allowed_offset(const void* handle) {
> }
> 
> int WebRtc_enable_robust_validation(void* handle, int enable) {
>-  DelayEstimator* self = (DelayEstimator*) handle;
>+  DelayEstimator* self = (DelayEstimator*)handle;
> 
>   if (self == NULL) {
>     return -1;
>@@ -404,7 +404,7 @@ int WebRtc_enable_robust_validation(void* handle, int enable) {
> }
> 
> int WebRtc_is_robust_validation_enabled(const void* handle) {
>-  const DelayEstimator* self = (const DelayEstimator*) handle;
>+  const DelayEstimator* self = (const DelayEstimator*)handle;
> 
>   if (self == NULL) {
>     return -1;
>@@ -416,7 +416,7 @@ int WebRtc_DelayEstimatorProcessFix(void* handle,
>                                     const uint16_t* near_spectrum,
>                                     int spectrum_size,
>                                     int near_q) {
>-  DelayEstimator* self = (DelayEstimator*) handle;
>+  DelayEstimator* self = (DelayEstimator*)handle;
>   uint32_t binary_spectrum = 0;
> 
>   if (self == NULL) {
>@@ -436,10 +436,9 @@ int WebRtc_DelayEstimatorProcessFix(void* handle,
>   }
> 
>   // Get binary spectra.
>-  binary_spectrum = BinarySpectrumFix(near_spectrum,
>-                                      self->mean_near_spectrum,
>-                                      near_q,
>-                                      &(self->near_spectrum_initialized));
>+  binary_spectrum =
>+      BinarySpectrumFix(near_spectrum, self->mean_near_spectrum, near_q,
>+                        &(self->near_spectrum_initialized));
> 
>   return WebRtc_ProcessBinarySpectrum(self->binary_handle, binary_spectrum);
> }
>@@ -447,7 +446,7 @@ int WebRtc_DelayEstimatorProcessFix(void* handle,
> int WebRtc_DelayEstimatorProcessFloat(void* handle,
>                                       const float* near_spectrum,
>                                       int spectrum_size) {
>-  DelayEstimator* self = (DelayEstimator*) handle;
>+  DelayEstimator* self = (DelayEstimator*)handle;
>   uint32_t binary_spectrum = 0;
> 
>   if (self == NULL) {
>@@ -470,7 +469,7 @@ int WebRtc_DelayEstimatorProcessFloat(void* handle,
> }
> 
> int WebRtc_last_delay(void* handle) {
>-  DelayEstimator* self = (DelayEstimator*) handle;
>+  DelayEstimator* self = (DelayEstimator*)handle;
> 
>   if (self == NULL) {
>     return -1;
>@@ -480,7 +479,7 @@ int WebRtc_last_delay(void* handle) {
> }
> 
> float WebRtc_last_delay_quality(void* handle) {
>-  DelayEstimator* self = (DelayEstimator*) handle;
>+  DelayEstimator* self = (DelayEstimator*)handle;
>   RTC_DCHECK(self);
>   return WebRtc_binary_last_delay_quality(self->binary_handle);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/delay_estimator_wrapper.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/delay_estimator_wrapper.h
>index 6b6e51f82c6..995470f99de 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/delay_estimator_wrapper.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/delay_estimator_wrapper.h
>@@ -14,7 +14,7 @@
> #ifndef MODULES_AUDIO_PROCESSING_UTILITY_DELAY_ESTIMATOR_WRAPPER_H_
> #define MODULES_AUDIO_PROCESSING_UTILITY_DELAY_ESTIMATOR_WRAPPER_H_
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include <stdint.h>
> 
> // Releases the memory allocated by WebRtc_CreateDelayEstimatorFarend(...)
> void WebRtc_FreeDelayEstimatorFarend(void* handle);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/ooura_fft.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/ooura_fft.cc
>index d753a813a65..c3333cea5be 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/ooura_fft.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/ooura_fft.cc
>@@ -26,8 +26,8 @@
> #include <math.h>
> 
> #include "modules/audio_processing/utility/ooura_fft_tables_common.h"
>+#include "rtc_base/system/arch.h"
> #include "system_wrappers/include/cpu_features_wrapper.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -313,7 +313,6 @@ static void rftbsub_128_C(float* a) {
> }
> #endif
> 
>-
> }  // namespace
> 
> OouraFft::OouraFft() {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/ooura_fft.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/ooura_fft.h
>index 96d57dc908d..0cdd6aa66f4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/ooura_fft.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/ooura_fft.h
>@@ -11,7 +11,7 @@
> #ifndef MODULES_AUDIO_PROCESSING_UTILITY_OOURA_FFT_H_
> #define MODULES_AUDIO_PROCESSING_UTILITY_OOURA_FFT_H_
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include "rtc_base/system/arch.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/ooura_fft_mips.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/ooura_fft_mips.cc
>index 569e1d7e820..9fe577db10a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/ooura_fft_mips.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/ooura_fft_mips.cc
>@@ -11,7 +11,6 @@
> #include "modules/audio_processing/utility/ooura_fft.h"
> 
> #include "modules/audio_processing/utility/ooura_fft_tables_common.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -279,533 +278,600 @@ void cft1st_128_mips(float* a) {
>   const float* first = rdft_wk3ri_first;
>   const float* second = rdft_wk3ri_second;
> 
>-  __asm __volatile (
>-    ".set       push                                                    \n\t"
>-    ".set       noreorder                                               \n\t"
>-    // first 8
>-    "lwc1       %[f0],        0(%[a])                                   \n\t"
>-    "lwc1       %[f1],        4(%[a])                                   \n\t"
>-    "lwc1       %[f2],        8(%[a])                                   \n\t"
>-    "lwc1       %[f3],        12(%[a])                                  \n\t"
>-    "lwc1       %[f4],        16(%[a])                                  \n\t"
>-    "lwc1       %[f5],        20(%[a])                                  \n\t"
>-    "lwc1       %[f6],        24(%[a])                                  \n\t"
>-    "lwc1       %[f7],        28(%[a])                                  \n\t"
>-    "add.s      %[f8],        %[f0],        %[f2]                       \n\t"
>-    "sub.s      %[f0],        %[f0],        %[f2]                       \n\t"
>-    "add.s      %[f2],        %[f4],        %[f6]                       \n\t"
>-    "sub.s      %[f4],        %[f4],        %[f6]                       \n\t"
>-    "add.s      %[f6],        %[f1],        %[f3]                       \n\t"
>-    "sub.s      %[f1],        %[f1],        %[f3]                       \n\t"
>-    "add.s      %[f3],        %[f5],        %[f7]                       \n\t"
>-    "sub.s      %[f5],        %[f5],        %[f7]                       \n\t"
>-    "add.s      %[f7],        %[f8],        %[f2]                       \n\t"
>-    "sub.s      %[f8],        %[f8],        %[f2]                       \n\t"
>-    "sub.s      %[f2],        %[f1],        %[f4]                       \n\t"
>-    "add.s      %[f1],        %[f1],        %[f4]                       \n\t"
>-    "add.s      %[f4],        %[f6],        %[f3]                       \n\t"
>-    "sub.s      %[f6],        %[f6],        %[f3]                       \n\t"
>-    "sub.s      %[f3],        %[f0],        %[f5]                       \n\t"
>-    "add.s      %[f0],        %[f0],        %[f5]                       \n\t"
>-    "swc1       %[f7],        0(%[a])                                   \n\t"
>-    "swc1       %[f8],        16(%[a])                                  \n\t"
>-    "swc1       %[f2],        28(%[a])                                  \n\t"
>-    "swc1       %[f1],        12(%[a])                                  \n\t"
>-    "swc1       %[f4],        4(%[a])                                   \n\t"
>-    "swc1       %[f6],        20(%[a])                                  \n\t"
>-    "swc1       %[f3],        8(%[a])                                   \n\t"
>-    "swc1       %[f0],        24(%[a])                                  \n\t"
>-    // second 8
>-    "lwc1       %[f0],        32(%[a])                                  \n\t"
>-    "lwc1       %[f1],        36(%[a])                                  \n\t"
>-    "lwc1       %[f2],        40(%[a])                                  \n\t"
>-    "lwc1       %[f3],        44(%[a])                                  \n\t"
>-    "lwc1       %[f4],        48(%[a])                                  \n\t"
>-    "lwc1       %[f5],        52(%[a])                                  \n\t"
>-    "lwc1       %[f6],        56(%[a])                                  \n\t"
>-    "lwc1       %[f7],        60(%[a])                                  \n\t"
>-    "add.s      %[f8],        %[f4],        %[f6]                       \n\t"
>-    "sub.s      %[f4],        %[f4],        %[f6]                       \n\t"
>-    "add.s      %[f6],        %[f1],        %[f3]                       \n\t"
>-    "sub.s      %[f1],        %[f1],        %[f3]                       \n\t"
>-    "add.s      %[f3],        %[f0],        %[f2]                       \n\t"
>-    "sub.s      %[f0],        %[f0],        %[f2]                       \n\t"
>-    "add.s      %[f2],        %[f5],        %[f7]                       \n\t"
>-    "sub.s      %[f5],        %[f5],        %[f7]                       \n\t"
>-    "add.s      %[f7],        %[f4],        %[f1]                       \n\t"
>-    "sub.s      %[f4],        %[f4],        %[f1]                       \n\t"
>-    "add.s      %[f1],        %[f3],        %[f8]                       \n\t"
>-    "sub.s      %[f3],        %[f3],        %[f8]                       \n\t"
>-    "sub.s      %[f8],        %[f0],        %[f5]                       \n\t"
>-    "add.s      %[f0],        %[f0],        %[f5]                       \n\t"
>-    "add.s      %[f5],        %[f6],        %[f2]                       \n\t"
>-    "sub.s      %[f6],        %[f2],        %[f6]                       \n\t"
>-    "lwc1       %[f9],        8(%[rdft_w])                              \n\t"
>-    "sub.s      %[f2],        %[f8],        %[f7]                       \n\t"
>-    "add.s      %[f8],        %[f8],        %[f7]                       \n\t"
>-    "sub.s      %[f7],        %[f4],        %[f0]                       \n\t"
>-    "add.s      %[f4],        %[f4],        %[f0]                       \n\t"
>-    // prepare for loop
>-    "addiu      %[a_ptr],     %[a],         64                          \n\t"
>-    "addiu      %[p1_rdft],   %[rdft_w],    8                           \n\t"
>-    "addiu      %[p2_rdft],   %[rdft_w],    16                          \n\t"
>-    "addiu      %[count],     $zero,        7                           \n\t"
>-    // finish second 8
>-    "mul.s      %[f2],        %[f9],        %[f2]                       \n\t"
>-    "mul.s      %[f8],        %[f9],        %[f8]                       \n\t"
>-    "mul.s      %[f7],        %[f9],        %[f7]                       \n\t"
>-    "mul.s      %[f4],        %[f9],        %[f4]                       \n\t"
>-    "swc1       %[f1],        32(%[a])                                  \n\t"
>-    "swc1       %[f3],        52(%[a])                                  \n\t"
>-    "swc1       %[f5],        36(%[a])                                  \n\t"
>-    "swc1       %[f6],        48(%[a])                                  \n\t"
>-    "swc1       %[f2],        40(%[a])                                  \n\t"
>-    "swc1       %[f8],        44(%[a])                                  \n\t"
>-    "swc1       %[f7],        56(%[a])                                  \n\t"
>-    "swc1       %[f4],        60(%[a])                                  \n\t"
>-    // loop
>-   "1:                                                                  \n\t"
>-    "lwc1       %[f0],        0(%[a_ptr])                               \n\t"
>-    "lwc1       %[f1],        4(%[a_ptr])                               \n\t"
>-    "lwc1       %[f2],        8(%[a_ptr])                               \n\t"
>-    "lwc1       %[f3],        12(%[a_ptr])                              \n\t"
>-    "lwc1       %[f4],        16(%[a_ptr])                              \n\t"
>-    "lwc1       %[f5],        20(%[a_ptr])                              \n\t"
>-    "lwc1       %[f6],        24(%[a_ptr])                              \n\t"
>-    "lwc1       %[f7],        28(%[a_ptr])                              \n\t"
>-    "add.s      %[f8],        %[f0],        %[f2]                       \n\t"
>-    "sub.s      %[f0],        %[f0],        %[f2]                       \n\t"
>-    "add.s      %[f2],        %[f4],        %[f6]                       \n\t"
>-    "sub.s      %[f4],        %[f4],        %[f6]                       \n\t"
>-    "add.s      %[f6],        %[f1],        %[f3]                       \n\t"
>-    "sub.s      %[f1],        %[f1],        %[f3]                       \n\t"
>-    "add.s      %[f3],        %[f5],        %[f7]                       \n\t"
>-    "sub.s      %[f5],        %[f5],        %[f7]                       \n\t"
>-    "lwc1       %[f10],       4(%[p1_rdft])                             \n\t"
>-    "lwc1       %[f11],       0(%[p2_rdft])                             \n\t"
>-    "lwc1       %[f12],       4(%[p2_rdft])                             \n\t"
>-    "lwc1       %[f13],       8(%[first])                               \n\t"
>-    "lwc1       %[f14],       12(%[first])                              \n\t"
>-    "add.s      %[f7],        %[f8],        %[f2]                       \n\t"
>-    "sub.s      %[f8],        %[f8],        %[f2]                       \n\t"
>-    "add.s      %[f2],        %[f6],        %[f3]                       \n\t"
>-    "sub.s      %[f6],        %[f6],        %[f3]                       \n\t"
>-    "add.s      %[f3],        %[f0],        %[f5]                       \n\t"
>-    "sub.s      %[f0],        %[f0],        %[f5]                       \n\t"
>-    "add.s      %[f5],        %[f1],        %[f4]                       \n\t"
>-    "sub.s      %[f1],        %[f1],        %[f4]                       \n\t"
>-    "swc1       %[f7],        0(%[a_ptr])                               \n\t"
>-    "swc1       %[f2],        4(%[a_ptr])                               \n\t"
>-    "mul.s      %[f4],        %[f9],        %[f8]                       \n\t"
>+  __asm __volatile(
>+      ".set       push                                                    \n\t"
>+      ".set       noreorder                                               \n\t"
>+      // first 8
>+      "lwc1       %[f0],        0(%[a])                                   \n\t"
>+      "lwc1       %[f1],        4(%[a])                                   \n\t"
>+      "lwc1       %[f2],        8(%[a])                                   \n\t"
>+      "lwc1       %[f3],        12(%[a])                                  \n\t"
>+      "lwc1       %[f4],        16(%[a])                                  \n\t"
>+      "lwc1       %[f5],        20(%[a])                                  \n\t"
>+      "lwc1       %[f6],        24(%[a])                                  \n\t"
>+      "lwc1       %[f7],        28(%[a])                                  \n\t"
>+      "add.s      %[f8],        %[f0],        %[f2]                       \n\t"
>+      "sub.s      %[f0],        %[f0],        %[f2]                       \n\t"
>+      "add.s      %[f2],        %[f4],        %[f6]                       \n\t"
>+      "sub.s      %[f4],        %[f4],        %[f6]                       \n\t"
>+      "add.s      %[f6],        %[f1],        %[f3]                       \n\t"
>+      "sub.s      %[f1],        %[f1],        %[f3]                       \n\t"
>+      "add.s      %[f3],        %[f5],        %[f7]                       \n\t"
>+      "sub.s      %[f5],        %[f5],        %[f7]                       \n\t"
>+      "add.s      %[f7],        %[f8],        %[f2]                       \n\t"
>+      "sub.s      %[f8],        %[f8],        %[f2]                       \n\t"
>+      "sub.s      %[f2],        %[f1],        %[f4]                       \n\t"
>+      "add.s      %[f1],        %[f1],        %[f4]                       \n\t"
>+      "add.s      %[f4],        %[f6],        %[f3]                       \n\t"
>+      "sub.s      %[f6],        %[f6],        %[f3]                       \n\t"
>+      "sub.s      %[f3],        %[f0],        %[f5]                       \n\t"
>+      "add.s      %[f0],        %[f0],        %[f5]                       \n\t"
>+      "swc1       %[f7],        0(%[a])                                   \n\t"
>+      "swc1       %[f8],        16(%[a])                                  \n\t"
>+      "swc1       %[f2],        28(%[a])                                  \n\t"
>+      "swc1       %[f1],        12(%[a])                                  \n\t"
>+      "swc1       %[f4],        4(%[a])                                   \n\t"
>+      "swc1       %[f6],        20(%[a])                                  \n\t"
>+      "swc1       %[f3],        8(%[a])                                   \n\t"
>+      "swc1       %[f0],        24(%[a])                                  \n\t"
>+      // second 8
>+      "lwc1       %[f0],        32(%[a])                                  \n\t"
>+      "lwc1       %[f1],        36(%[a])                                  \n\t"
>+      "lwc1       %[f2],        40(%[a])                                  \n\t"
>+      "lwc1       %[f3],        44(%[a])                                  \n\t"
>+      "lwc1       %[f4],        48(%[a])                                  \n\t"
>+      "lwc1       %[f5],        52(%[a])                                  \n\t"
>+      "lwc1       %[f6],        56(%[a])                                  \n\t"
>+      "lwc1       %[f7],        60(%[a])                                  \n\t"
>+      "add.s      %[f8],        %[f4],        %[f6]                       \n\t"
>+      "sub.s      %[f4],        %[f4],        %[f6]                       \n\t"
>+      "add.s      %[f6],        %[f1],        %[f3]                       \n\t"
>+      "sub.s      %[f1],        %[f1],        %[f3]                       \n\t"
>+      "add.s      %[f3],        %[f0],        %[f2]                       \n\t"
>+      "sub.s      %[f0],        %[f0],        %[f2]                       \n\t"
>+      "add.s      %[f2],        %[f5],        %[f7]                       \n\t"
>+      "sub.s      %[f5],        %[f5],        %[f7]                       \n\t"
>+      "add.s      %[f7],        %[f4],        %[f1]                       \n\t"
>+      "sub.s      %[f4],        %[f4],        %[f1]                       \n\t"
>+      "add.s      %[f1],        %[f3],        %[f8]                       \n\t"
>+      "sub.s      %[f3],        %[f3],        %[f8]                       \n\t"
>+      "sub.s      %[f8],        %[f0],        %[f5]                       \n\t"
>+      "add.s      %[f0],        %[f0],        %[f5]                       \n\t"
>+      "add.s      %[f5],        %[f6],        %[f2]                       \n\t"
>+      "sub.s      %[f6],        %[f2],        %[f6]                       \n\t"
>+      "lwc1       %[f9],        8(%[rdft_w])                              \n\t"
>+      "sub.s      %[f2],        %[f8],        %[f7]                       \n\t"
>+      "add.s      %[f8],        %[f8],        %[f7]                       \n\t"
>+      "sub.s      %[f7],        %[f4],        %[f0]                       \n\t"
>+      "add.s      %[f4],        %[f4],        %[f0]                       \n\t"
>+      // prepare for loop
>+      "addiu      %[a_ptr],     %[a],         64                          \n\t"
>+      "addiu      %[p1_rdft],   %[rdft_w],    8                           \n\t"
>+      "addiu      %[p2_rdft],   %[rdft_w],    16                          \n\t"
>+      "addiu      %[count],     $zero,        7                           \n\t"
>+      // finish second 8
>+      "mul.s      %[f2],        %[f9],        %[f2]                       \n\t"
>+      "mul.s      %[f8],        %[f9],        %[f8]                       \n\t"
>+      "mul.s      %[f7],        %[f9],        %[f7]                       \n\t"
>+      "mul.s      %[f4],        %[f9],        %[f4]                       \n\t"
>+      "swc1       %[f1],        32(%[a])                                  \n\t"
>+      "swc1       %[f3],        52(%[a])                                  \n\t"
>+      "swc1       %[f5],        36(%[a])                                  \n\t"
>+      "swc1       %[f6],        48(%[a])                                  \n\t"
>+      "swc1       %[f2],        40(%[a])                                  \n\t"
>+      "swc1       %[f8],        44(%[a])                                  \n\t"
>+      "swc1       %[f7],        56(%[a])                                  \n\t"
>+      "swc1       %[f4],        60(%[a])                                  \n\t"
>+      // loop
>+      "1:                                                                  \n\t"
>+      "lwc1       %[f0],        0(%[a_ptr])                               \n\t"
>+      "lwc1       %[f1],        4(%[a_ptr])                               \n\t"
>+      "lwc1       %[f2],        8(%[a_ptr])                               \n\t"
>+      "lwc1       %[f3],        12(%[a_ptr])                              \n\t"
>+      "lwc1       %[f4],        16(%[a_ptr])                              \n\t"
>+      "lwc1       %[f5],        20(%[a_ptr])                              \n\t"
>+      "lwc1       %[f6],        24(%[a_ptr])                              \n\t"
>+      "lwc1       %[f7],        28(%[a_ptr])                              \n\t"
>+      "add.s      %[f8],        %[f0],        %[f2]                       \n\t"
>+      "sub.s      %[f0],        %[f0],        %[f2]                       \n\t"
>+      "add.s      %[f2],        %[f4],        %[f6]                       \n\t"
>+      "sub.s      %[f4],        %[f4],        %[f6]                       \n\t"
>+      "add.s      %[f6],        %[f1],        %[f3]                       \n\t"
>+      "sub.s      %[f1],        %[f1],        %[f3]                       \n\t"
>+      "add.s      %[f3],        %[f5],        %[f7]                       \n\t"
>+      "sub.s      %[f5],        %[f5],        %[f7]                       \n\t"
>+      "lwc1       %[f10],       4(%[p1_rdft])                             \n\t"
>+      "lwc1       %[f11],       0(%[p2_rdft])                             \n\t"
>+      "lwc1       %[f12],       4(%[p2_rdft])                             \n\t"
>+      "lwc1       %[f13],       8(%[first])                               \n\t"
>+      "lwc1       %[f14],       12(%[first])                              \n\t"
>+      "add.s      %[f7],        %[f8],        %[f2]                       \n\t"
>+      "sub.s      %[f8],        %[f8],        %[f2]                       \n\t"
>+      "add.s      %[f2],        %[f6],        %[f3]                       \n\t"
>+      "sub.s      %[f6],        %[f6],        %[f3]                       \n\t"
>+      "add.s      %[f3],        %[f0],        %[f5]                       \n\t"
>+      "sub.s      %[f0],        %[f0],        %[f5]                       \n\t"
>+      "add.s      %[f5],        %[f1],        %[f4]                       \n\t"
>+      "sub.s      %[f1],        %[f1],        %[f4]                       \n\t"
>+      "swc1       %[f7],        0(%[a_ptr])                               \n\t"
>+      "swc1       %[f2],        4(%[a_ptr])                               \n\t"
>+      "mul.s      %[f4],        %[f9],        %[f8]                       \n\t"
> #if defined(MIPS32_R2_LE)
>-    "mul.s      %[f8],        %[f10],       %[f8]                       \n\t"
>-    "mul.s      %[f7],        %[f11],       %[f0]                       \n\t"
>-    "mul.s      %[f0],        %[f12],       %[f0]                       \n\t"
>-    "mul.s      %[f2],        %[f13],       %[f3]                       \n\t"
>-    "mul.s      %[f3],        %[f14],       %[f3]                       \n\t"
>-    "nmsub.s    %[f4],        %[f4],        %[f10],       %[f6]         \n\t"
>-    "madd.s     %[f8],        %[f8],        %[f9],        %[f6]         \n\t"
>-    "nmsub.s    %[f7],        %[f7],        %[f12],       %[f5]         \n\t"
>-    "madd.s     %[f0],        %[f0],        %[f11],       %[f5]         \n\t"
>-    "nmsub.s    %[f2],        %[f2],        %[f14],       %[f1]         \n\t"
>-    "madd.s     %[f3],        %[f3],        %[f13],       %[f1]         \n\t"
>+      "mul.s      %[f8],        %[f10],       %[f8]                       \n\t"
>+      "mul.s      %[f7],        %[f11],       %[f0]                       \n\t"
>+      "mul.s      %[f0],        %[f12],       %[f0]                       \n\t"
>+      "mul.s      %[f2],        %[f13],       %[f3]                       \n\t"
>+      "mul.s      %[f3],        %[f14],       %[f3]                       \n\t"
>+      "nmsub.s    %[f4],        %[f4],        %[f10],       %[f6]         \n\t"
>+      "madd.s     %[f8],        %[f8],        %[f9],        %[f6]         \n\t"
>+      "nmsub.s    %[f7],        %[f7],        %[f12],       %[f5]         \n\t"
>+      "madd.s     %[f0],        %[f0],        %[f11],       %[f5]         \n\t"
>+      "nmsub.s    %[f2],        %[f2],        %[f14],       %[f1]         \n\t"
>+      "madd.s     %[f3],        %[f3],        %[f13],       %[f1]         \n\t"
> #else
>-    "mul.s      %[f7],        %[f10],       %[f6]                       \n\t"
>-    "mul.s      %[f6],        %[f9],        %[f6]                       \n\t"
>-    "mul.s      %[f8],        %[f10],       %[f8]                       \n\t"
>-    "mul.s      %[f2],        %[f11],       %[f0]                       \n\t"
>-    "mul.s      %[f11],       %[f11],       %[f5]                       \n\t"
>-    "mul.s      %[f5],        %[f12],       %[f5]                       \n\t"
>-    "mul.s      %[f0],        %[f12],       %[f0]                       \n\t"
>-    "mul.s      %[f12],       %[f13],       %[f3]                       \n\t"
>-    "mul.s      %[f13],       %[f13],       %[f1]                       \n\t"
>-    "mul.s      %[f1],        %[f14],       %[f1]                       \n\t"
>-    "mul.s      %[f3],        %[f14],       %[f3]                       \n\t"
>-    "sub.s      %[f4],        %[f4],        %[f7]                       \n\t"
>-    "add.s      %[f8],        %[f6],        %[f8]                       \n\t"
>-    "sub.s      %[f7],        %[f2],        %[f5]                       \n\t"
>-    "add.s      %[f0],        %[f11],       %[f0]                       \n\t"
>-    "sub.s      %[f2],        %[f12],       %[f1]                       \n\t"
>-    "add.s      %[f3],        %[f13],       %[f3]                       \n\t"
>+      "mul.s      %[f7],        %[f10],       %[f6]                       \n\t"
>+      "mul.s      %[f6],        %[f9],        %[f6]                       \n\t"
>+      "mul.s      %[f8],        %[f10],       %[f8]                       \n\t"
>+      "mul.s      %[f2],        %[f11],       %[f0]                       \n\t"
>+      "mul.s      %[f11],       %[f11],       %[f5]                       \n\t"
>+      "mul.s      %[f5],        %[f12],       %[f5]                       \n\t"
>+      "mul.s      %[f0],        %[f12],       %[f0]                       \n\t"
>+      "mul.s      %[f12],       %[f13],       %[f3]                       \n\t"
>+      "mul.s      %[f13],       %[f13],       %[f1]                       \n\t"
>+      "mul.s      %[f1],        %[f14],       %[f1]                       \n\t"
>+      "mul.s      %[f3],        %[f14],       %[f3]                       \n\t"
>+      "sub.s      %[f4],        %[f4],        %[f7]                       \n\t"
>+      "add.s      %[f8],        %[f6],        %[f8]                       \n\t"
>+      "sub.s      %[f7],        %[f2],        %[f5]                       \n\t"
>+      "add.s      %[f0],        %[f11],       %[f0]                       \n\t"
>+      "sub.s      %[f2],        %[f12],       %[f1]                       \n\t"
>+      "add.s      %[f3],        %[f13],       %[f3]                       \n\t"
> #endif
>-    "swc1       %[f4],        16(%[a_ptr])                              \n\t"
>-    "swc1       %[f8],        20(%[a_ptr])                              \n\t"
>-    "swc1       %[f7],        8(%[a_ptr])                               \n\t"
>-    "swc1       %[f0],        12(%[a_ptr])                              \n\t"
>-    "swc1       %[f2],        24(%[a_ptr])                              \n\t"
>-    "swc1       %[f3],        28(%[a_ptr])                              \n\t"
>-    "lwc1       %[f0],        32(%[a_ptr])                              \n\t"
>-    "lwc1       %[f1],        36(%[a_ptr])                              \n\t"
>-    "lwc1       %[f2],        40(%[a_ptr])                              \n\t"
>-    "lwc1       %[f3],        44(%[a_ptr])                              \n\t"
>-    "lwc1       %[f4],        48(%[a_ptr])                              \n\t"
>-    "lwc1       %[f5],        52(%[a_ptr])                              \n\t"
>-    "lwc1       %[f6],        56(%[a_ptr])                              \n\t"
>-    "lwc1       %[f7],        60(%[a_ptr])                              \n\t"
>-    "add.s      %[f8],        %[f0],        %[f2]                       \n\t"
>-    "sub.s      %[f0],        %[f0],        %[f2]                       \n\t"
>-    "add.s      %[f2],        %[f4],        %[f6]                       \n\t"
>-    "sub.s      %[f4],        %[f4],        %[f6]                       \n\t"
>-    "add.s      %[f6],        %[f1],        %[f3]                       \n\t"
>-    "sub.s      %[f1],        %[f1],        %[f3]                       \n\t"
>-    "add.s      %[f3],        %[f5],        %[f7]                       \n\t"
>-    "sub.s      %[f5],        %[f5],        %[f7]                       \n\t"
>-    "lwc1       %[f11],       8(%[p2_rdft])                             \n\t"
>-    "lwc1       %[f12],       12(%[p2_rdft])                            \n\t"
>-    "lwc1       %[f13],       8(%[second])                              \n\t"
>-    "lwc1       %[f14],       12(%[second])                             \n\t"
>-    "add.s      %[f7],        %[f8],        %[f2]                       \n\t"
>-    "sub.s      %[f8],        %[f2],        %[f8]                       \n\t"
>-    "add.s      %[f2],        %[f6],        %[f3]                       \n\t"
>-    "sub.s      %[f6],        %[f3],        %[f6]                       \n\t"
>-    "add.s      %[f3],        %[f0],        %[f5]                       \n\t"
>-    "sub.s      %[f0],        %[f0],        %[f5]                       \n\t"
>-    "add.s      %[f5],        %[f1],        %[f4]                       \n\t"
>-    "sub.s      %[f1],        %[f1],        %[f4]                       \n\t"
>-    "swc1       %[f7],        32(%[a_ptr])                              \n\t"
>-    "swc1       %[f2],        36(%[a_ptr])                              \n\t"
>-    "mul.s      %[f4],        %[f10],       %[f8]                       \n\t"
>+      "swc1       %[f4],        16(%[a_ptr])                              \n\t"
>+      "swc1       %[f8],        20(%[a_ptr])                              \n\t"
>+      "swc1       %[f7],        8(%[a_ptr])                               \n\t"
>+      "swc1       %[f0],        12(%[a_ptr])                              \n\t"
>+      "swc1       %[f2],        24(%[a_ptr])                              \n\t"
>+      "swc1       %[f3],        28(%[a_ptr])                              \n\t"
>+      "lwc1       %[f0],        32(%[a_ptr])                              \n\t"
>+      "lwc1       %[f1],        36(%[a_ptr])                              \n\t"
>+      "lwc1       %[f2],        40(%[a_ptr])                              \n\t"
>+      "lwc1       %[f3],        44(%[a_ptr])                              \n\t"
>+      "lwc1       %[f4],        48(%[a_ptr])                              \n\t"
>+      "lwc1       %[f5],        52(%[a_ptr])                              \n\t"
>+      "lwc1       %[f6],        56(%[a_ptr])                              \n\t"
>+      "lwc1       %[f7],        60(%[a_ptr])                              \n\t"
>+      "add.s      %[f8],        %[f0],        %[f2]                       \n\t"
>+      "sub.s      %[f0],        %[f0],        %[f2]                       \n\t"
>+      "add.s      %[f2],        %[f4],        %[f6]                       \n\t"
>+      "sub.s      %[f4],        %[f4],        %[f6]                       \n\t"
>+      "add.s      %[f6],        %[f1],        %[f3]                       \n\t"
>+      "sub.s      %[f1],        %[f1],        %[f3]                       \n\t"
>+      "add.s      %[f3],        %[f5],        %[f7]                       \n\t"
>+      "sub.s      %[f5],        %[f5],        %[f7]                       \n\t"
>+      "lwc1       %[f11],       8(%[p2_rdft])                             \n\t"
>+      "lwc1       %[f12],       12(%[p2_rdft])                            \n\t"
>+      "lwc1       %[f13],       8(%[second])                              \n\t"
>+      "lwc1       %[f14],       12(%[second])                             \n\t"
>+      "add.s      %[f7],        %[f8],        %[f2]                       \n\t"
>+      "sub.s      %[f8],        %[f2],        %[f8]                       \n\t"
>+      "add.s      %[f2],        %[f6],        %[f3]                       \n\t"
>+      "sub.s      %[f6],        %[f3],        %[f6]                       \n\t"
>+      "add.s      %[f3],        %[f0],        %[f5]                       \n\t"
>+      "sub.s      %[f0],        %[f0],        %[f5]                       \n\t"
>+      "add.s      %[f5],        %[f1],        %[f4]                       \n\t"
>+      "sub.s      %[f1],        %[f1],        %[f4]                       \n\t"
>+      "swc1       %[f7],        32(%[a_ptr])                              \n\t"
>+      "swc1       %[f2],        36(%[a_ptr])                              \n\t"
>+      "mul.s      %[f4],        %[f10],       %[f8]                       \n\t"
> #if defined(MIPS32_R2_LE)
>-    "mul.s      %[f10],       %[f10],       %[f6]                       \n\t"
>-    "mul.s      %[f7],        %[f11],       %[f0]                       \n\t"
>-    "mul.s      %[f11],       %[f11],       %[f5]                       \n\t"
>-    "mul.s      %[f2],        %[f13],       %[f3]                       \n\t"
>-    "mul.s      %[f13],       %[f13],       %[f1]                       \n\t"
>-    "madd.s     %[f4],        %[f4],        %[f9],        %[f6]         \n\t"
>-    "nmsub.s    %[f10],       %[f10],       %[f9],        %[f8]         \n\t"
>-    "nmsub.s    %[f7],        %[f7],        %[f12],       %[f5]         \n\t"
>-    "madd.s     %[f11],       %[f11],       %[f12],       %[f0]         \n\t"
>-    "nmsub.s    %[f2],        %[f2],        %[f14],       %[f1]         \n\t"
>-    "madd.s     %[f13],       %[f13],       %[f14],       %[f3]         \n\t"
>+      "mul.s      %[f10],       %[f10],       %[f6]                       \n\t"
>+      "mul.s      %[f7],        %[f11],       %[f0]                       \n\t"
>+      "mul.s      %[f11],       %[f11],       %[f5]                       \n\t"
>+      "mul.s      %[f2],        %[f13],       %[f3]                       \n\t"
>+      "mul.s      %[f13],       %[f13],       %[f1]                       \n\t"
>+      "madd.s     %[f4],        %[f4],        %[f9],        %[f6]         \n\t"
>+      "nmsub.s    %[f10],       %[f10],       %[f9],        %[f8]         \n\t"
>+      "nmsub.s    %[f7],        %[f7],        %[f12],       %[f5]         \n\t"
>+      "madd.s     %[f11],       %[f11],       %[f12],       %[f0]         \n\t"
>+      "nmsub.s    %[f2],        %[f2],        %[f14],       %[f1]         \n\t"
>+      "madd.s     %[f13],       %[f13],       %[f14],       %[f3]         \n\t"
> #else
>-    "mul.s      %[f2],        %[f9],        %[f6]                       \n\t"
>-    "mul.s      %[f10],       %[f10],       %[f6]                       \n\t"
>-    "mul.s      %[f9],        %[f9],        %[f8]                       \n\t"
>-    "mul.s      %[f7],        %[f11],       %[f0]                       \n\t"
>-    "mul.s      %[f8],        %[f12],       %[f5]                       \n\t"
>-    "mul.s      %[f11],       %[f11],       %[f5]                       \n\t"
>-    "mul.s      %[f12],       %[f12],       %[f0]                       \n\t"
>-    "mul.s      %[f5],        %[f13],       %[f3]                       \n\t"
>-    "mul.s      %[f0],        %[f14],       %[f1]                       \n\t"
>-    "mul.s      %[f13],       %[f13],       %[f1]                       \n\t"
>-    "mul.s      %[f14],       %[f14],       %[f3]                       \n\t"
>-    "add.s      %[f4],        %[f4],        %[f2]                       \n\t"
>-    "sub.s      %[f10],       %[f10],       %[f9]                       \n\t"
>-    "sub.s      %[f7],        %[f7],        %[f8]                       \n\t"
>-    "add.s      %[f11],       %[f11],       %[f12]                      \n\t"
>-    "sub.s      %[f2],        %[f5],        %[f0]                       \n\t"
>-    "add.s      %[f13],       %[f13],       %[f14]                      \n\t"
>+      "mul.s      %[f2],        %[f9],        %[f6]                       \n\t"
>+      "mul.s      %[f10],       %[f10],       %[f6]                       \n\t"
>+      "mul.s      %[f9],        %[f9],        %[f8]                       \n\t"
>+      "mul.s      %[f7],        %[f11],       %[f0]                       \n\t"
>+      "mul.s      %[f8],        %[f12],       %[f5]                       \n\t"
>+      "mul.s      %[f11],       %[f11],       %[f5]                       \n\t"
>+      "mul.s      %[f12],       %[f12],       %[f0]                       \n\t"
>+      "mul.s      %[f5],        %[f13],       %[f3]                       \n\t"
>+      "mul.s      %[f0],        %[f14],       %[f1]                       \n\t"
>+      "mul.s      %[f13],       %[f13],       %[f1]                       \n\t"
>+      "mul.s      %[f14],       %[f14],       %[f3]                       \n\t"
>+      "add.s      %[f4],        %[f4],        %[f2]                       \n\t"
>+      "sub.s      %[f10],       %[f10],       %[f9]                       \n\t"
>+      "sub.s      %[f7],        %[f7],        %[f8]                       \n\t"
>+      "add.s      %[f11],       %[f11],       %[f12]                      \n\t"
>+      "sub.s      %[f2],        %[f5],        %[f0]                       \n\t"
>+      "add.s      %[f13],       %[f13],       %[f14]                      \n\t"
> #endif
>-    "swc1       %[f4],        48(%[a_ptr])                              \n\t"
>-    "swc1       %[f10],       52(%[a_ptr])                              \n\t"
>-    "swc1       %[f7],        40(%[a_ptr])                              \n\t"
>-    "swc1       %[f11],       44(%[a_ptr])                              \n\t"
>-    "swc1       %[f2],        56(%[a_ptr])                              \n\t"
>-    "swc1       %[f13],       60(%[a_ptr])                              \n\t"
>-    "addiu      %[count],     %[count],     -1                          \n\t"
>-    "lwc1       %[f9],        8(%[p1_rdft])                             \n\t"
>-    "addiu      %[a_ptr],     %[a_ptr],     64                          \n\t"
>-    "addiu      %[p1_rdft],   %[p1_rdft],   8                           \n\t"
>-    "addiu      %[p2_rdft],   %[p2_rdft],   16                          \n\t"
>-    "addiu      %[first],     %[first],     8                           \n\t"
>-    "bgtz       %[count],     1b                                        \n\t"
>-    " addiu     %[second],    %[second],    8                           \n\t"
>-    ".set       pop                                                     \n\t"
>-    : [f0] "=&f" (f0), [f1] "=&f" (f1), [f2] "=&f" (f2), [f3] "=&f" (f3),
>-      [f4] "=&f" (f4), [f5] "=&f" (f5), [f6] "=&f" (f6), [f7] "=&f" (f7),
>-      [f8] "=&f" (f8), [f9] "=&f" (f9), [f10] "=&f" (f10), [f11] "=&f" (f11),
>-      [f12] "=&f" (f12), [f13] "=&f" (f13), [f14] "=&f" (f14),
>-      [a_ptr] "=&r" (a_ptr), [p1_rdft] "=&r" (p1_rdft), [first] "+r" (first),
>-      [p2_rdft] "=&r" (p2_rdft), [count] "=&r" (count), [second] "+r" (second)
>-    : [a] "r" (a), [rdft_w] "r" (rdft_w)
>-    : "memory"
>-  );
>+      "swc1       %[f4],        48(%[a_ptr])                              \n\t"
>+      "swc1       %[f10],       52(%[a_ptr])                              \n\t"
>+      "swc1       %[f7],        40(%[a_ptr])                              \n\t"
>+      "swc1       %[f11],       44(%[a_ptr])                              \n\t"
>+      "swc1       %[f2],        56(%[a_ptr])                              \n\t"
>+      "swc1       %[f13],       60(%[a_ptr])                              \n\t"
>+      "addiu      %[count],     %[count],     -1                          \n\t"
>+      "lwc1       %[f9],        8(%[p1_rdft])                             \n\t"
>+      "addiu      %[a_ptr],     %[a_ptr],     64                          \n\t"
>+      "addiu      %[p1_rdft],   %[p1_rdft],   8                           \n\t"
>+      "addiu      %[p2_rdft],   %[p2_rdft],   16                          \n\t"
>+      "addiu      %[first],     %[first],     8                           \n\t"
>+      "bgtz       %[count],     1b                                        \n\t"
>+      " addiu     %[second],    %[second],    8                           \n\t"
>+      ".set       pop                                                     \n\t"
>+      : [f0] "=&f"(f0), [f1] "=&f"(f1), [f2] "=&f"(f2), [f3] "=&f"(f3),
>+        [f4] "=&f"(f4), [f5] "=&f"(f5), [f6] "=&f"(f6), [f7] "=&f"(f7),
>+        [f8] "=&f"(f8), [f9] "=&f"(f9), [f10] "=&f"(f10), [f11] "=&f"(f11),
>+        [f12] "=&f"(f12), [f13] "=&f"(f13), [f14] "=&f"(f14),
>+        [a_ptr] "=&r"(a_ptr), [p1_rdft] "=&r"(p1_rdft), [first] "+r"(first),
>+        [p2_rdft] "=&r"(p2_rdft), [count] "=&r"(count), [second] "+r"(second)
>+      : [a] "r"(a), [rdft_w] "r"(rdft_w)
>+      : "memory");
> }
> 
> void cftmdl_128_mips(float* a) {
>   float f0, f1, f2, f3, f4, f5, f6, f7, f8, f9, f10, f11, f12, f13, f14;
>   int tmp_a, count;
>-  __asm __volatile (
>-    ".set       push                                      \n\t"
>-    ".set       noreorder                                 \n\t"
>-    "addiu      %[tmp_a],   %[a],         0               \n\t"
>-    "addiu      %[count],   $zero,        4               \n\t"
>-   "1:                                                    \n\t"
>-    "addiu      %[count],   %[count],     -1              \n\t"
>-    "lwc1       %[f0],      0(%[tmp_a])                   \n\t"
>-    "lwc1       %[f2],      32(%[tmp_a])                  \n\t"
>-    "lwc1       %[f4],      64(%[tmp_a])                  \n\t"
>-    "lwc1       %[f6],      96(%[tmp_a])                  \n\t"
>-    "lwc1       %[f1],      4(%[tmp_a])                   \n\t"
>-    "lwc1       %[f3],      36(%[tmp_a])                  \n\t"
>-    "lwc1       %[f5],      68(%[tmp_a])                  \n\t"
>-    "lwc1       %[f7],      100(%[tmp_a])                 \n\t"
>-    "add.s      %[f8],      %[f0],        %[f2]           \n\t"
>-    "sub.s      %[f0],      %[f0],        %[f2]           \n\t"
>-    "add.s      %[f2],      %[f4],        %[f6]           \n\t"
>-    "sub.s      %[f4],      %[f4],        %[f6]           \n\t"
>-    "add.s      %[f6],      %[f1],        %[f3]           \n\t"
>-    "sub.s      %[f1],      %[f1],        %[f3]           \n\t"
>-    "add.s      %[f3],      %[f5],        %[f7]           \n\t"
>-    "sub.s      %[f5],      %[f5],        %[f7]           \n\t"
>-    "add.s      %[f7],      %[f8],        %[f2]           \n\t"
>-    "sub.s      %[f8],      %[f8],        %[f2]           \n\t"
>-    "add.s      %[f2],      %[f1],        %[f4]           \n\t"
>-    "sub.s      %[f1],      %[f1],        %[f4]           \n\t"
>-    "add.s      %[f4],      %[f6],        %[f3]           \n\t"
>-    "sub.s      %[f6],      %[f6],        %[f3]           \n\t"
>-    "sub.s      %[f3],      %[f0],        %[f5]           \n\t"
>-    "add.s      %[f0],      %[f0],        %[f5]           \n\t"
>-    "swc1       %[f7],      0(%[tmp_a])                   \n\t"
>-    "swc1       %[f8],      64(%[tmp_a])                  \n\t"
>-    "swc1       %[f2],      36(%[tmp_a])                  \n\t"
>-    "swc1       %[f1],      100(%[tmp_a])                 \n\t"
>-    "swc1       %[f4],      4(%[tmp_a])                   \n\t"
>-    "swc1       %[f6],      68(%[tmp_a])                  \n\t"
>-    "swc1       %[f3],      32(%[tmp_a])                  \n\t"
>-    "swc1       %[f0],      96(%[tmp_a])                  \n\t"
>-    "bgtz       %[count],   1b                            \n\t"
>-    " addiu     %[tmp_a],   %[tmp_a],     8               \n\t"
>-    ".set       pop                                       \n\t"
>-    : [f0] "=&f" (f0), [f1] "=&f" (f1), [f2] "=&f" (f2), [f3] "=&f" (f3),
>-      [f4] "=&f" (f4), [f5] "=&f" (f5), [f6] "=&f" (f6), [f7] "=&f" (f7),
>-      [f8] "=&f" (f8), [tmp_a] "=&r" (tmp_a), [count] "=&r" (count)
>-    : [a] "r" (a)
>-    : "memory"
>-  );
>+  __asm __volatile(
>+      ".set       push                                      \n\t"
>+      ".set       noreorder                                 \n\t"
>+      "addiu      %[tmp_a],   %[a],         0               \n\t"
>+      "addiu      %[count],   $zero,        4               \n\t"
>+      "1:                                                    \n\t"
>+      "addiu      %[count],   %[count],     -1              \n\t"
>+      "lwc1       %[f0],      0(%[tmp_a])                   \n\t"
>+      "lwc1       %[f2],      32(%[tmp_a])                  \n\t"
>+      "lwc1       %[f4],      64(%[tmp_a])                  \n\t"
>+      "lwc1       %[f6],      96(%[tmp_a])                  \n\t"
>+      "lwc1       %[f1],      4(%[tmp_a])                   \n\t"
>+      "lwc1       %[f3],      36(%[tmp_a])                  \n\t"
>+      "lwc1       %[f5],      68(%[tmp_a])                  \n\t"
>+      "lwc1       %[f7],      100(%[tmp_a])                 \n\t"
>+      "add.s      %[f8],      %[f0],        %[f2]           \n\t"
>+      "sub.s      %[f0],      %[f0],        %[f2]           \n\t"
>+      "add.s      %[f2],      %[f4],        %[f6]           \n\t"
>+      "sub.s      %[f4],      %[f4],        %[f6]           \n\t"
>+      "add.s      %[f6],      %[f1],        %[f3]           \n\t"
>+      "sub.s      %[f1],      %[f1],        %[f3]           \n\t"
>+      "add.s      %[f3],      %[f5],        %[f7]           \n\t"
>+      "sub.s      %[f5],      %[f5],        %[f7]           \n\t"
>+      "add.s      %[f7],      %[f8],        %[f2]           \n\t"
>+      "sub.s      %[f8],      %[f8],        %[f2]           \n\t"
>+      "add.s      %[f2],      %[f1],        %[f4]           \n\t"
>+      "sub.s      %[f1],      %[f1],        %[f4]           \n\t"
>+      "add.s      %[f4],      %[f6],        %[f3]           \n\t"
>+      "sub.s      %[f6],      %[f6],        %[f3]           \n\t"
>+      "sub.s      %[f3],      %[f0],        %[f5]           \n\t"
>+      "add.s      %[f0],      %[f0],        %[f5]           \n\t"
>+      "swc1       %[f7],      0(%[tmp_a])                   \n\t"
>+      "swc1       %[f8],      64(%[tmp_a])                  \n\t"
>+      "swc1       %[f2],      36(%[tmp_a])                  \n\t"
>+      "swc1       %[f1],      100(%[tmp_a])                 \n\t"
>+      "swc1       %[f4],      4(%[tmp_a])                   \n\t"
>+      "swc1       %[f6],      68(%[tmp_a])                  \n\t"
>+      "swc1       %[f3],      32(%[tmp_a])                  \n\t"
>+      "swc1       %[f0],      96(%[tmp_a])                  \n\t"
>+      "bgtz       %[count],   1b                            \n\t"
>+      " addiu     %[tmp_a],   %[tmp_a],     8               \n\t"
>+      ".set       pop                                       \n\t"
>+      : [f0] "=&f"(f0), [f1] "=&f"(f1), [f2] "=&f"(f2), [f3] "=&f"(f3),
>+        [f4] "=&f"(f4), [f5] "=&f"(f5), [f6] "=&f"(f6), [f7] "=&f"(f7),
>+        [f8] "=&f"(f8), [tmp_a] "=&r"(tmp_a), [count] "=&r"(count)
>+      : [a] "r"(a)
>+      : "memory");
>   f9 = rdft_w[2];
>-  __asm __volatile (
>-    ".set       push                                      \n\t"
>-    ".set       noreorder                                 \n\t"
>-    "addiu      %[tmp_a],   %[a],         128             \n\t"
>-    "addiu      %[count],   $zero,        4               \n\t"
>-   "1:                                                    \n\t"
>-    "addiu      %[count],   %[count],     -1              \n\t"
>-    "lwc1       %[f0],      0(%[tmp_a])                   \n\t"
>-    "lwc1       %[f2],      32(%[tmp_a])                  \n\t"
>-    "lwc1       %[f5],      68(%[tmp_a])                  \n\t"
>-    "lwc1       %[f7],      100(%[tmp_a])                 \n\t"
>-    "lwc1       %[f1],      4(%[tmp_a])                   \n\t"
>-    "lwc1       %[f3],      36(%[tmp_a])                  \n\t"
>-    "lwc1       %[f4],      64(%[tmp_a])                  \n\t"
>-    "lwc1       %[f6],      96(%[tmp_a])                  \n\t"
>-    "sub.s      %[f8],      %[f0],        %[f2]           \n\t"
>-    "add.s      %[f0],      %[f0],        %[f2]           \n\t"
>-    "sub.s      %[f2],      %[f5],        %[f7]           \n\t"
>-    "add.s      %[f5],      %[f5],        %[f7]           \n\t"
>-    "sub.s      %[f7],      %[f1],        %[f3]           \n\t"
>-    "add.s      %[f1],      %[f1],        %[f3]           \n\t"
>-    "sub.s      %[f3],      %[f4],        %[f6]           \n\t"
>-    "add.s      %[f4],      %[f4],        %[f6]           \n\t"
>-    "sub.s      %[f6],      %[f8],        %[f2]           \n\t"
>-    "add.s      %[f8],      %[f8],        %[f2]           \n\t"
>-    "add.s      %[f2],      %[f5],        %[f1]           \n\t"
>-    "sub.s      %[f5],      %[f5],        %[f1]           \n\t"
>-    "add.s      %[f1],      %[f3],        %[f7]           \n\t"
>-    "sub.s      %[f3],      %[f3],        %[f7]           \n\t"
>-    "add.s      %[f7],      %[f0],        %[f4]           \n\t"
>-    "sub.s      %[f0],      %[f0],        %[f4]           \n\t"
>-    "sub.s      %[f4],      %[f6],        %[f1]           \n\t"
>-    "add.s      %[f6],      %[f6],        %[f1]           \n\t"
>-    "sub.s      %[f1],      %[f3],        %[f8]           \n\t"
>-    "add.s      %[f3],      %[f3],        %[f8]           \n\t"
>-    "mul.s      %[f4],      %[f4],        %[f9]           \n\t"
>-    "mul.s      %[f6],      %[f6],        %[f9]           \n\t"
>-    "mul.s      %[f1],      %[f1],        %[f9]           \n\t"
>-    "mul.s      %[f3],      %[f3],        %[f9]           \n\t"
>-    "swc1       %[f7],      0(%[tmp_a])                   \n\t"
>-    "swc1       %[f2],      4(%[tmp_a])                   \n\t"
>-    "swc1       %[f5],      64(%[tmp_a])                  \n\t"
>-    "swc1       %[f0],      68(%[tmp_a])                  \n\t"
>-    "swc1       %[f4],      32(%[tmp_a])                  \n\t"
>-    "swc1       %[f6],      36(%[tmp_a])                  \n\t"
>-    "swc1       %[f1],      96(%[tmp_a])                  \n\t"
>-    "swc1       %[f3],      100(%[tmp_a])                 \n\t"
>-    "bgtz       %[count],   1b                            \n\t"
>-    " addiu     %[tmp_a],   %[tmp_a],     8               \n\t"
>-    ".set       pop                                       \n\t"
>-    : [f0] "=&f" (f0), [f1] "=&f" (f1), [f2] "=&f" (f2), [f3] "=&f" (f3),
>-      [f4] "=&f" (f4), [f5] "=&f" (f5), [f6] "=&f" (f6), [f7] "=&f" (f7),
>-      [f8] "=&f" (f8), [tmp_a] "=&r" (tmp_a), [count] "=&r" (count)
>-    : [a] "r" (a), [f9] "f" (f9)
>-    : "memory"
>-  );
>+  __asm __volatile(
>+      ".set       push                                      \n\t"
>+      ".set       noreorder                                 \n\t"
>+      "addiu      %[tmp_a],   %[a],         128             \n\t"
>+      "addiu      %[count],   $zero,        4               \n\t"
>+      "1:                                                    \n\t"
>+      "addiu      %[count],   %[count],     -1              \n\t"
>+      "lwc1       %[f0],      0(%[tmp_a])                   \n\t"
>+      "lwc1       %[f2],      32(%[tmp_a])                  \n\t"
>+      "lwc1       %[f5],      68(%[tmp_a])                  \n\t"
>+      "lwc1       %[f7],      100(%[tmp_a])                 \n\t"
>+      "lwc1       %[f1],      4(%[tmp_a])                   \n\t"
>+      "lwc1       %[f3],      36(%[tmp_a])                  \n\t"
>+      "lwc1       %[f4],      64(%[tmp_a])                  \n\t"
>+      "lwc1       %[f6],      96(%[tmp_a])                  \n\t"
>+      "sub.s      %[f8],      %[f0],        %[f2]           \n\t"
>+      "add.s      %[f0],      %[f0],        %[f2]           \n\t"
>+      "sub.s      %[f2],      %[f5],        %[f7]           \n\t"
>+      "add.s      %[f5],      %[f5],        %[f7]           \n\t"
>+      "sub.s      %[f7],      %[f1],        %[f3]           \n\t"
>+      "add.s      %[f1],      %[f1],        %[f3]           \n\t"
>+      "sub.s      %[f3],      %[f4],        %[f6]           \n\t"
>+      "add.s      %[f4],      %[f4],        %[f6]           \n\t"
>+      "sub.s      %[f6],      %[f8],        %[f2]           \n\t"
>+      "add.s      %[f8],      %[f8],        %[f2]           \n\t"
>+      "add.s      %[f2],      %[f5],        %[f1]           \n\t"
>+      "sub.s      %[f5],      %[f5],        %[f1]           \n\t"
>+      "add.s      %[f1],      %[f3],        %[f7]           \n\t"
>+      "sub.s      %[f3],      %[f3],        %[f7]           \n\t"
>+      "add.s      %[f7],      %[f0],        %[f4]           \n\t"
>+      "sub.s      %[f0],      %[f0],        %[f4]           \n\t"
>+      "sub.s      %[f4],      %[f6],        %[f1]           \n\t"
>+      "add.s      %[f6],      %[f6],        %[f1]           \n\t"
>+      "sub.s      %[f1],      %[f3],        %[f8]           \n\t"
>+      "add.s      %[f3],      %[f3],        %[f8]           \n\t"
>+      "mul.s      %[f4],      %[f4],        %[f9]           \n\t"
>+      "mul.s      %[f6],      %[f6],        %[f9]           \n\t"
>+      "mul.s      %[f1],      %[f1],        %[f9]           \n\t"
>+      "mul.s      %[f3],      %[f3],        %[f9]           \n\t"
>+      "swc1       %[f7],      0(%[tmp_a])                   \n\t"
>+      "swc1       %[f2],      4(%[tmp_a])                   \n\t"
>+      "swc1       %[f5],      64(%[tmp_a])                  \n\t"
>+      "swc1       %[f0],      68(%[tmp_a])                  \n\t"
>+      "swc1       %[f4],      32(%[tmp_a])                  \n\t"
>+      "swc1       %[f6],      36(%[tmp_a])                  \n\t"
>+      "swc1       %[f1],      96(%[tmp_a])                  \n\t"
>+      "swc1       %[f3],      100(%[tmp_a])                 \n\t"
>+      "bgtz       %[count],   1b                            \n\t"
>+      " addiu     %[tmp_a],   %[tmp_a],     8               \n\t"
>+      ".set       pop                                       \n\t"
>+      : [f0] "=&f"(f0), [f1] "=&f"(f1), [f2] "=&f"(f2), [f3] "=&f"(f3),
>+        [f4] "=&f"(f4), [f5] "=&f"(f5), [f6] "=&f"(f6), [f7] "=&f"(f7),
>+        [f8] "=&f"(f8), [tmp_a] "=&r"(tmp_a), [count] "=&r"(count)
>+      : [a] "r"(a), [f9] "f"(f9)
>+      : "memory");
>   f10 = rdft_w[3];
>   f11 = rdft_w[4];
>   f12 = rdft_w[5];
>   f13 = rdft_wk3ri_first[2];
>   f14 = rdft_wk3ri_first[3];
> 
>-  __asm __volatile (
>-    ".set       push                                                    \n\t"
>-    ".set       noreorder                                               \n\t"
>-    "addiu      %[tmp_a],     %[a],         256                         \n\t"
>-    "addiu      %[count],     $zero,        4                           \n\t"
>-   "1:                                                                  \n\t"
>-    "addiu      %[count],     %[count],     -1                          \n\t"
>-    "lwc1       %[f0],        0(%[tmp_a])                               \n\t"
>-    "lwc1       %[f2],        32(%[tmp_a])                              \n\t"
>-    "lwc1       %[f4],        64(%[tmp_a])                              \n\t"
>-    "lwc1       %[f6],        96(%[tmp_a])                              \n\t"
>-    "lwc1       %[f1],        4(%[tmp_a])                               \n\t"
>-    "lwc1       %[f3],        36(%[tmp_a])                              \n\t"
>-    "lwc1       %[f5],        68(%[tmp_a])                              \n\t"
>-    "lwc1       %[f7],        100(%[tmp_a])                             \n\t"
>-    "add.s      %[f8],        %[f0],        %[f2]                       \n\t"
>-    "sub.s      %[f0],        %[f0],        %[f2]                       \n\t"
>-    "add.s      %[f2],        %[f4],        %[f6]                       \n\t"
>-    "sub.s      %[f4],        %[f4],        %[f6]                       \n\t"
>-    "add.s      %[f6],        %[f1],        %[f3]                       \n\t"
>-    "sub.s      %[f1],        %[f1],        %[f3]                       \n\t"
>-    "add.s      %[f3],        %[f5],        %[f7]                       \n\t"
>-    "sub.s      %[f5],        %[f5],        %[f7]                       \n\t"
>-    "sub.s      %[f7],        %[f8],        %[f2]                       \n\t"
>-    "add.s      %[f8],        %[f8],        %[f2]                       \n\t"
>-    "add.s      %[f2],        %[f1],        %[f4]                       \n\t"
>-    "sub.s      %[f1],        %[f1],        %[f4]                       \n\t"
>-    "sub.s      %[f4],        %[f6],        %[f3]                       \n\t"
>-    "add.s      %[f6],        %[f6],        %[f3]                       \n\t"
>-    "sub.s      %[f3],        %[f0],        %[f5]                       \n\t"
>-    "add.s      %[f0],        %[f0],        %[f5]                       \n\t"
>-    "swc1       %[f8],        0(%[tmp_a])                               \n\t"
>-    "swc1       %[f6],        4(%[tmp_a])                               \n\t"
>-    "mul.s      %[f5],        %[f9],        %[f7]                       \n\t"
>+  __asm __volatile(
>+      ".set       push                                                    \n\t"
>+      ".set       noreorder                                               \n\t"
>+      "addiu      %[tmp_a],     %[a],         256                         \n\t"
>+      "addiu      %[count],     $zero,        4                           \n\t"
>+      "1:                                                                  \n\t"
>+      "addiu      %[count],     %[count],     -1                          \n\t"
>+      "lwc1       %[f0],        0(%[tmp_a])                               \n\t"
>+      "lwc1       %[f2],        32(%[tmp_a])                              \n\t"
>+      "lwc1       %[f4],        64(%[tmp_a])                              \n\t"
>+      "lwc1       %[f6],        96(%[tmp_a])                              \n\t"
>+      "lwc1       %[f1],        4(%[tmp_a])                               \n\t"
>+      "lwc1       %[f3],        36(%[tmp_a])                              \n\t"
>+      "lwc1       %[f5],        68(%[tmp_a])                              \n\t"
>+      "lwc1       %[f7],        100(%[tmp_a])                             \n\t"
>+      "add.s      %[f8],        %[f0],        %[f2]                       \n\t"
>+      "sub.s      %[f0],        %[f0],        %[f2]                       \n\t"
>+      "add.s      %[f2],        %[f4],        %[f6]                       \n\t"
>+      "sub.s      %[f4],        %[f4],        %[f6]                       \n\t"
>+      "add.s      %[f6],        %[f1],        %[f3]                       \n\t"
>+      "sub.s      %[f1],        %[f1],        %[f3]                       \n\t"
>+      "add.s      %[f3],        %[f5],        %[f7]                       \n\t"
>+      "sub.s      %[f5],        %[f5],        %[f7]                       \n\t"
>+      "sub.s      %[f7],        %[f8],        %[f2]                       \n\t"
>+      "add.s      %[f8],        %[f8],        %[f2]                       \n\t"
>+      "add.s      %[f2],        %[f1],        %[f4]                       \n\t"
>+      "sub.s      %[f1],        %[f1],        %[f4]                       \n\t"
>+      "sub.s      %[f4],        %[f6],        %[f3]                       \n\t"
>+      "add.s      %[f6],        %[f6],        %[f3]                       \n\t"
>+      "sub.s      %[f3],        %[f0],        %[f5]                       \n\t"
>+      "add.s      %[f0],        %[f0],        %[f5]                       \n\t"
>+      "swc1       %[f8],        0(%[tmp_a])                               \n\t"
>+      "swc1       %[f6],        4(%[tmp_a])                               \n\t"
>+      "mul.s      %[f5],        %[f9],        %[f7]                       \n\t"
> #if defined(MIPS32_R2_LE)
>-    "mul.s      %[f7],        %[f10],       %[f7]                       \n\t"
>-    "mul.s      %[f8],        %[f11],       %[f3]                       \n\t"
>-    "mul.s      %[f3],        %[f12],       %[f3]                       \n\t"
>-    "mul.s      %[f6],        %[f13],       %[f0]                       \n\t"
>-    "mul.s      %[f0],        %[f14],       %[f0]                       \n\t"
>-    "nmsub.s    %[f5],        %[f5],        %[f10],       %[f4]         \n\t"
>-    "madd.s     %[f7],        %[f7],        %[f9],        %[f4]         \n\t"
>-    "nmsub.s    %[f8],        %[f8],        %[f12],       %[f2]         \n\t"
>-    "madd.s     %[f3],        %[f3],        %[f11],       %[f2]         \n\t"
>-    "nmsub.s    %[f6],        %[f6],        %[f14],       %[f1]         \n\t"
>-    "madd.s     %[f0],        %[f0],        %[f13],       %[f1]         \n\t"
>-    "swc1       %[f5],        64(%[tmp_a])                              \n\t"
>-    "swc1       %[f7],        68(%[tmp_a])                              \n\t"
>+      "mul.s      %[f7],        %[f10],       %[f7]                       \n\t"
>+      "mul.s      %[f8],        %[f11],       %[f3]                       \n\t"
>+      "mul.s      %[f3],        %[f12],       %[f3]                       \n\t"
>+      "mul.s      %[f6],        %[f13],       %[f0]                       \n\t"
>+      "mul.s      %[f0],        %[f14],       %[f0]                       \n\t"
>+      "nmsub.s    %[f5],        %[f5],        %[f10],       %[f4]         \n\t"
>+      "madd.s     %[f7],        %[f7],        %[f9],        %[f4]         \n\t"
>+      "nmsub.s    %[f8],        %[f8],        %[f12],       %[f2]         \n\t"
>+      "madd.s     %[f3],        %[f3],        %[f11],       %[f2]         \n\t"
>+      "nmsub.s    %[f6],        %[f6],        %[f14],       %[f1]         \n\t"
>+      "madd.s     %[f0],        %[f0],        %[f13],       %[f1]         \n\t"
>+      "swc1       %[f5],        64(%[tmp_a])                              \n\t"
>+      "swc1       %[f7],        68(%[tmp_a])                              \n\t"
> #else
>-    "mul.s      %[f8],        %[f10],       %[f4]                       \n\t"
>-    "mul.s      %[f4],        %[f9],        %[f4]                       \n\t"
>-    "mul.s      %[f7],        %[f10],       %[f7]                       \n\t"
>-    "mul.s      %[f6],        %[f11],       %[f3]                       \n\t"
>-    "mul.s      %[f3],        %[f12],       %[f3]                       \n\t"
>-    "sub.s      %[f5],        %[f5],        %[f8]                       \n\t"
>-    "mul.s      %[f8],        %[f12],       %[f2]                       \n\t"
>-    "mul.s      %[f2],        %[f11],       %[f2]                       \n\t"
>-    "add.s      %[f7],        %[f4],        %[f7]                       \n\t"
>-    "mul.s      %[f4],        %[f13],       %[f0]                       \n\t"
>-    "mul.s      %[f0],        %[f14],       %[f0]                       \n\t"
>-    "sub.s      %[f8],        %[f6],        %[f8]                       \n\t"
>-    "mul.s      %[f6],        %[f14],       %[f1]                       \n\t"
>-    "mul.s      %[f1],        %[f13],       %[f1]                       \n\t"
>-    "add.s      %[f3],        %[f2],        %[f3]                       \n\t"
>-    "swc1       %[f5],        64(%[tmp_a])                              \n\t"
>-    "swc1       %[f7],        68(%[tmp_a])                              \n\t"
>-    "sub.s      %[f6],        %[f4],        %[f6]                       \n\t"
>-    "add.s      %[f0],        %[f1],        %[f0]                       \n\t"
>+      "mul.s      %[f8],        %[f10],       %[f4]                       \n\t"
>+      "mul.s      %[f4],        %[f9],        %[f4]                       \n\t"
>+      "mul.s      %[f7],        %[f10],       %[f7]                       \n\t"
>+      "mul.s      %[f6],        %[f11],       %[f3]                       \n\t"
>+      "mul.s      %[f3],        %[f12],       %[f3]                       \n\t"
>+      "sub.s      %[f5],        %[f5],        %[f8]                       \n\t"
>+      "mul.s      %[f8],        %[f12],       %[f2]                       \n\t"
>+      "mul.s      %[f2],        %[f11],       %[f2]                       \n\t"
>+      "add.s      %[f7],        %[f4],        %[f7]                       \n\t"
>+      "mul.s      %[f4],        %[f13],       %[f0]                       \n\t"
>+      "mul.s      %[f0],        %[f14],       %[f0]                       \n\t"
>+      "sub.s      %[f8],        %[f6],        %[f8]                       \n\t"
>+      "mul.s      %[f6],        %[f14],       %[f1]                       \n\t"
>+      "mul.s      %[f1],        %[f13],       %[f1]                       \n\t"
>+      "add.s      %[f3],        %[f2],        %[f3]                       \n\t"
>+      "swc1       %[f5],        64(%[tmp_a])                              \n\t"
>+      "swc1       %[f7],        68(%[tmp_a])                              \n\t"
>+      "sub.s      %[f6],        %[f4],        %[f6]                       \n\t"
>+      "add.s      %[f0],        %[f1],        %[f0]                       \n\t"
> #endif
>-    "swc1       %[f8],        32(%[tmp_a])                              \n\t"
>-    "swc1       %[f3],        36(%[tmp_a])                              \n\t"
>-    "swc1       %[f6],        96(%[tmp_a])                              \n\t"
>-    "swc1       %[f0],        100(%[tmp_a])                             \n\t"
>-    "bgtz       %[count],     1b                                        \n\t"
>-    " addiu     %[tmp_a],     %[tmp_a],     8                           \n\t"
>-    ".set       pop                                                     \n\t"
>-    : [f0] "=&f" (f0), [f1] "=&f" (f1), [f2] "=&f" (f2), [f3] "=&f" (f3),
>-      [f4] "=&f" (f4), [f5] "=&f" (f5), [f6] "=&f" (f6), [f7] "=&f" (f7),
>-      [f8] "=&f" (f8), [tmp_a] "=&r" (tmp_a), [count] "=&r" (count)
>-    : [a] "r" (a),  [f9] "f" (f9), [f10] "f" (f10), [f11] "f" (f11),
>-      [f12] "f" (f12), [f13] "f" (f13), [f14] "f" (f14)
>-    : "memory"
>-  );
>+      "swc1       %[f8],        32(%[tmp_a])                              \n\t"
>+      "swc1       %[f3],        36(%[tmp_a])                              \n\t"
>+      "swc1       %[f6],        96(%[tmp_a])                              \n\t"
>+      "swc1       %[f0],        100(%[tmp_a])                             \n\t"
>+      "bgtz       %[count],     1b                                        \n\t"
>+      " addiu     %[tmp_a],     %[tmp_a],     8                           \n\t"
>+      ".set       pop                                                     \n\t"
>+      : [f0] "=&f"(f0), [f1] "=&f"(f1), [f2] "=&f"(f2), [f3] "=&f"(f3),
>+        [f4] "=&f"(f4), [f5] "=&f"(f5), [f6] "=&f"(f6), [f7] "=&f"(f7),
>+        [f8] "=&f"(f8), [tmp_a] "=&r"(tmp_a), [count] "=&r"(count)
>+      : [a] "r"(a), [f9] "f"(f9), [f10] "f"(f10), [f11] "f"(f11),
>+        [f12] "f"(f12), [f13] "f"(f13), [f14] "f"(f14)
>+      : "memory");
>   f11 = rdft_w[6];
>   f12 = rdft_w[7];
>   f13 = rdft_wk3ri_second[2];
>   f14 = rdft_wk3ri_second[3];
>-  __asm __volatile (
>-    ".set       push                                                       \n\t"
>-    ".set       noreorder                                                  \n\t"
>-    "addiu      %[tmp_a],       %[a],           384                        \n\t"
>-    "addiu      %[count],       $zero,          4                          \n\t"
>-   "1:                                                                     \n\t"
>-    "addiu      %[count],       %[count],       -1                         \n\t"
>-    "lwc1       %[f0],          0(%[tmp_a])                                \n\t"
>-    "lwc1       %[f1],          4(%[tmp_a])                                \n\t"
>-    "lwc1       %[f2],          32(%[tmp_a])                               \n\t"
>-    "lwc1       %[f3],          36(%[tmp_a])                               \n\t"
>-    "lwc1       %[f4],          64(%[tmp_a])                               \n\t"
>-    "lwc1       %[f5],          68(%[tmp_a])                               \n\t"
>-    "lwc1       %[f6],          96(%[tmp_a])                               \n\t"
>-    "lwc1       %[f7],          100(%[tmp_a])                              \n\t"
>-    "add.s      %[f8],          %[f0],          %[f2]                      \n\t"
>-    "sub.s      %[f0],          %[f0],          %[f2]                      \n\t"
>-    "add.s      %[f2],          %[f4],          %[f6]                      \n\t"
>-    "sub.s      %[f4],          %[f4],          %[f6]                      \n\t"
>-    "add.s      %[f6],          %[f1],          %[f3]                      \n\t"
>-    "sub.s      %[f1],          %[f1],          %[f3]                      \n\t"
>-    "add.s      %[f3],          %[f5],          %[f7]                      \n\t"
>-    "sub.s      %[f5],          %[f5],          %[f7]                      \n\t"
>-    "sub.s      %[f7],          %[f2],          %[f8]                      \n\t"
>-    "add.s      %[f2],          %[f2],          %[f8]                      \n\t"
>-    "add.s      %[f8],          %[f1],          %[f4]                      \n\t"
>-    "sub.s      %[f1],          %[f1],          %[f4]                      \n\t"
>-    "sub.s      %[f4],          %[f3],          %[f6]                      \n\t"
>-    "add.s      %[f3],          %[f3],          %[f6]                      \n\t"
>-    "sub.s      %[f6],          %[f0],          %[f5]                      \n\t"
>-    "add.s      %[f0],          %[f0],          %[f5]                      \n\t"
>-    "swc1       %[f2],          0(%[tmp_a])                                \n\t"
>-    "swc1       %[f3],          4(%[tmp_a])                                \n\t"
>-    "mul.s      %[f5],          %[f10],         %[f7]                      \n\t"
>+  __asm __volatile(
>+      ".set       push                                                       "
>+      "\n\t"
>+      ".set       noreorder                                                  "
>+      "\n\t"
>+      "addiu      %[tmp_a],       %[a],           384                        "
>+      "\n\t"
>+      "addiu      %[count],       $zero,          4                          "
>+      "\n\t"
>+      "1:                                                                     "
>+      "\n\t"
>+      "addiu      %[count],       %[count],       -1                         "
>+      "\n\t"
>+      "lwc1       %[f0],          0(%[tmp_a])                                "
>+      "\n\t"
>+      "lwc1       %[f1],          4(%[tmp_a])                                "
>+      "\n\t"
>+      "lwc1       %[f2],          32(%[tmp_a])                               "
>+      "\n\t"
>+      "lwc1       %[f3],          36(%[tmp_a])                               "
>+      "\n\t"
>+      "lwc1       %[f4],          64(%[tmp_a])                               "
>+      "\n\t"
>+      "lwc1       %[f5],          68(%[tmp_a])                               "
>+      "\n\t"
>+      "lwc1       %[f6],          96(%[tmp_a])                               "
>+      "\n\t"
>+      "lwc1       %[f7],          100(%[tmp_a])                              "
>+      "\n\t"
>+      "add.s      %[f8],          %[f0],          %[f2]                      "
>+      "\n\t"
>+      "sub.s      %[f0],          %[f0],          %[f2]                      "
>+      "\n\t"
>+      "add.s      %[f2],          %[f4],          %[f6]                      "
>+      "\n\t"
>+      "sub.s      %[f4],          %[f4],          %[f6]                      "
>+      "\n\t"
>+      "add.s      %[f6],          %[f1],          %[f3]                      "
>+      "\n\t"
>+      "sub.s      %[f1],          %[f1],          %[f3]                      "
>+      "\n\t"
>+      "add.s      %[f3],          %[f5],          %[f7]                      "
>+      "\n\t"
>+      "sub.s      %[f5],          %[f5],          %[f7]                      "
>+      "\n\t"
>+      "sub.s      %[f7],          %[f2],          %[f8]                      "
>+      "\n\t"
>+      "add.s      %[f2],          %[f2],          %[f8]                      "
>+      "\n\t"
>+      "add.s      %[f8],          %[f1],          %[f4]                      "
>+      "\n\t"
>+      "sub.s      %[f1],          %[f1],          %[f4]                      "
>+      "\n\t"
>+      "sub.s      %[f4],          %[f3],          %[f6]                      "
>+      "\n\t"
>+      "add.s      %[f3],          %[f3],          %[f6]                      "
>+      "\n\t"
>+      "sub.s      %[f6],          %[f0],          %[f5]                      "
>+      "\n\t"
>+      "add.s      %[f0],          %[f0],          %[f5]                      "
>+      "\n\t"
>+      "swc1       %[f2],          0(%[tmp_a])                                "
>+      "\n\t"
>+      "swc1       %[f3],          4(%[tmp_a])                                "
>+      "\n\t"
>+      "mul.s      %[f5],          %[f10],         %[f7]                      "
>+      "\n\t"
> #if defined(MIPS32_R2_LE)
>-    "mul.s      %[f7],          %[f9],          %[f7]                      \n\t"
>-    "mul.s      %[f2],          %[f12],         %[f8]                      \n\t"
>-    "mul.s      %[f8],          %[f11],         %[f8]                      \n\t"
>-    "mul.s      %[f3],          %[f14],         %[f1]                      \n\t"
>-    "mul.s      %[f1],          %[f13],         %[f1]                      \n\t"
>-    "madd.s     %[f5],          %[f5],          %[f9],       %[f4]         \n\t"
>-    "msub.s     %[f7],          %[f7],          %[f10],      %[f4]         \n\t"
>-    "msub.s     %[f2],          %[f2],          %[f11],      %[f6]         \n\t"
>-    "madd.s     %[f8],          %[f8],          %[f12],      %[f6]         \n\t"
>-    "msub.s     %[f3],          %[f3],          %[f13],      %[f0]         \n\t"
>-    "madd.s     %[f1],          %[f1],          %[f14],      %[f0]         \n\t"
>-    "swc1       %[f5],          64(%[tmp_a])                               \n\t"
>-    "swc1       %[f7],          68(%[tmp_a])                               \n\t"
>+      "mul.s      %[f7],          %[f9],          %[f7]                      "
>+      "\n\t"
>+      "mul.s      %[f2],          %[f12],         %[f8]                      "
>+      "\n\t"
>+      "mul.s      %[f8],          %[f11],         %[f8]                      "
>+      "\n\t"
>+      "mul.s      %[f3],          %[f14],         %[f1]                      "
>+      "\n\t"
>+      "mul.s      %[f1],          %[f13],         %[f1]                      "
>+      "\n\t"
>+      "madd.s     %[f5],          %[f5],          %[f9],       %[f4]         "
>+      "\n\t"
>+      "msub.s     %[f7],          %[f7],          %[f10],      %[f4]         "
>+      "\n\t"
>+      "msub.s     %[f2],          %[f2],          %[f11],      %[f6]         "
>+      "\n\t"
>+      "madd.s     %[f8],          %[f8],          %[f12],      %[f6]         "
>+      "\n\t"
>+      "msub.s     %[f3],          %[f3],          %[f13],      %[f0]         "
>+      "\n\t"
>+      "madd.s     %[f1],          %[f1],          %[f14],      %[f0]         "
>+      "\n\t"
>+      "swc1       %[f5],          64(%[tmp_a])                               "
>+      "\n\t"
>+      "swc1       %[f7],          68(%[tmp_a])                               "
>+      "\n\t"
> #else
>-    "mul.s      %[f2],          %[f9],          %[f4]                      \n\t"
>-    "mul.s      %[f4],          %[f10],         %[f4]                      \n\t"
>-    "mul.s      %[f7],          %[f9],          %[f7]                      \n\t"
>-    "mul.s      %[f3],          %[f11],         %[f6]                      \n\t"
>-    "mul.s      %[f6],          %[f12],         %[f6]                      \n\t"
>-    "add.s      %[f5],          %[f5],          %[f2]                      \n\t"
>-    "sub.s      %[f7],          %[f4],          %[f7]                      \n\t"
>-    "mul.s      %[f2],          %[f12],         %[f8]                      \n\t"
>-    "mul.s      %[f8],          %[f11],         %[f8]                      \n\t"
>-    "mul.s      %[f4],          %[f14],         %[f1]                      \n\t"
>-    "mul.s      %[f1],          %[f13],         %[f1]                      \n\t"
>-    "sub.s      %[f2],          %[f3],          %[f2]                      \n\t"
>-    "mul.s      %[f3],          %[f13],         %[f0]                      \n\t"
>-    "mul.s      %[f0],          %[f14],         %[f0]                      \n\t"
>-    "add.s      %[f8],          %[f8],          %[f6]                      \n\t"
>-    "swc1       %[f5],          64(%[tmp_a])                               \n\t"
>-    "swc1       %[f7],          68(%[tmp_a])                               \n\t"
>-    "sub.s      %[f3],          %[f3],          %[f4]                      \n\t"
>-    "add.s      %[f1],          %[f1],          %[f0]                      \n\t"
>+      "mul.s      %[f2],          %[f9],          %[f4]                      "
>+      "\n\t"
>+      "mul.s      %[f4],          %[f10],         %[f4]                      "
>+      "\n\t"
>+      "mul.s      %[f7],          %[f9],          %[f7]                      "
>+      "\n\t"
>+      "mul.s      %[f3],          %[f11],         %[f6]                      "
>+      "\n\t"
>+      "mul.s      %[f6],          %[f12],         %[f6]                      "
>+      "\n\t"
>+      "add.s      %[f5],          %[f5],          %[f2]                      "
>+      "\n\t"
>+      "sub.s      %[f7],          %[f4],          %[f7]                      "
>+      "\n\t"
>+      "mul.s      %[f2],          %[f12],         %[f8]                      "
>+      "\n\t"
>+      "mul.s      %[f8],          %[f11],         %[f8]                      "
>+      "\n\t"
>+      "mul.s      %[f4],          %[f14],         %[f1]                      "
>+      "\n\t"
>+      "mul.s      %[f1],          %[f13],         %[f1]                      "
>+      "\n\t"
>+      "sub.s      %[f2],          %[f3],          %[f2]                      "
>+      "\n\t"
>+      "mul.s      %[f3],          %[f13],         %[f0]                      "
>+      "\n\t"
>+      "mul.s      %[f0],          %[f14],         %[f0]                      "
>+      "\n\t"
>+      "add.s      %[f8],          %[f8],          %[f6]                      "
>+      "\n\t"
>+      "swc1       %[f5],          64(%[tmp_a])                               "
>+      "\n\t"
>+      "swc1       %[f7],          68(%[tmp_a])                               "
>+      "\n\t"
>+      "sub.s      %[f3],          %[f3],          %[f4]                      "
>+      "\n\t"
>+      "add.s      %[f1],          %[f1],          %[f0]                      "
>+      "\n\t"
> #endif
>-    "swc1       %[f2],          32(%[tmp_a])                               \n\t"
>-    "swc1       %[f8],          36(%[tmp_a])                               \n\t"
>-    "swc1       %[f3],          96(%[tmp_a])                               \n\t"
>-    "swc1       %[f1],          100(%[tmp_a])                              \n\t"
>-    "bgtz       %[count],       1b                                         \n\t"
>-    " addiu     %[tmp_a],       %[tmp_a],       8                          \n\t"
>-    ".set       pop                                                        \n\t"
>-    : [f0] "=&f" (f0), [f1] "=&f" (f1), [f2] "=&f" (f2), [f3] "=&f" (f3),
>-      [f4] "=&f" (f4), [f5] "=&f" (f5), [f6] "=&f" (f6), [f7] "=&f" (f7),
>-      [f8] "=&f" (f8), [tmp_a] "=&r" (tmp_a), [count] "=&r" (count)
>-    : [a] "r" (a), [f9] "f" (f9), [f10] "f" (f10), [f11] "f" (f11),
>-      [f12] "f" (f12), [f13] "f" (f13), [f14] "f" (f14)
>-    : "memory"
>-  );
>+      "swc1       %[f2],          32(%[tmp_a])                               "
>+      "\n\t"
>+      "swc1       %[f8],          36(%[tmp_a])                               "
>+      "\n\t"
>+      "swc1       %[f3],          96(%[tmp_a])                               "
>+      "\n\t"
>+      "swc1       %[f1],          100(%[tmp_a])                              "
>+      "\n\t"
>+      "bgtz       %[count],       1b                                         "
>+      "\n\t"
>+      " addiu     %[tmp_a],       %[tmp_a],       8                          "
>+      "\n\t"
>+      ".set       pop                                                        "
>+      "\n\t"
>+      : [f0] "=&f"(f0), [f1] "=&f"(f1), [f2] "=&f"(f2), [f3] "=&f"(f3),
>+        [f4] "=&f"(f4), [f5] "=&f"(f5), [f6] "=&f"(f6), [f7] "=&f"(f7),
>+        [f8] "=&f"(f8), [tmp_a] "=&r"(tmp_a), [count] "=&r"(count)
>+      : [a] "r"(a), [f9] "f"(f9), [f10] "f"(f10), [f11] "f"(f11),
>+        [f12] "f"(f12), [f13] "f"(f13), [f14] "f"(f14)
>+      : "memory");
> }
> 
> void cftfsub_128_mips(float* a) {
>@@ -815,55 +881,53 @@ void cftfsub_128_mips(float* a) {
>   cft1st_128_mips(a);
>   cftmdl_128_mips(a);
> 
>-  __asm __volatile (
>-    ".set       push                                      \n\t"
>-    ".set       noreorder                                 \n\t"
>-    "addiu      %[tmp_a],       %[a],         0           \n\t"
>-    "addiu      %[count],       $zero,        16          \n\t"
>-   "1:                                                    \n\t"
>-    "addiu      %[count],       %[count],     -1          \n\t"
>-    "lwc1       %[f0],          0(%[tmp_a])               \n\t"
>-    "lwc1       %[f2],          128(%[tmp_a])             \n\t"
>-    "lwc1       %[f4],          256(%[tmp_a])             \n\t"
>-    "lwc1       %[f6],          384(%[tmp_a])             \n\t"
>-    "lwc1       %[f1],          4(%[tmp_a])               \n\t"
>-    "lwc1       %[f3],          132(%[tmp_a])             \n\t"
>-    "lwc1       %[f5],          260(%[tmp_a])             \n\t"
>-    "lwc1       %[f7],          388(%[tmp_a])             \n\t"
>-    "add.s      %[f8],          %[f0],        %[f2]       \n\t"
>-    "sub.s      %[f0],          %[f0],        %[f2]       \n\t"
>-    "add.s      %[f2],          %[f4],        %[f6]       \n\t"
>-    "sub.s      %[f4],          %[f4],        %[f6]       \n\t"
>-    "add.s      %[f6],          %[f1],        %[f3]       \n\t"
>-    "sub.s      %[f1],          %[f1],        %[f3]       \n\t"
>-    "add.s      %[f3],          %[f5],        %[f7]       \n\t"
>-    "sub.s      %[f5],          %[f5],        %[f7]       \n\t"
>-    "add.s      %[f7],          %[f8],        %[f2]       \n\t"
>-    "sub.s      %[f8],          %[f8],        %[f2]       \n\t"
>-    "add.s      %[f2],          %[f1],        %[f4]       \n\t"
>-    "sub.s      %[f1],          %[f1],        %[f4]       \n\t"
>-    "add.s      %[f4],          %[f6],        %[f3]       \n\t"
>-    "sub.s      %[f6],          %[f6],        %[f3]       \n\t"
>-    "sub.s      %[f3],          %[f0],        %[f5]       \n\t"
>-    "add.s      %[f0],          %[f0],        %[f5]       \n\t"
>-    "swc1       %[f7],          0(%[tmp_a])               \n\t"
>-    "swc1       %[f8],          256(%[tmp_a])             \n\t"
>-    "swc1       %[f2],          132(%[tmp_a])             \n\t"
>-    "swc1       %[f1],          388(%[tmp_a])             \n\t"
>-    "swc1       %[f4],          4(%[tmp_a])               \n\t"
>-    "swc1       %[f6],          260(%[tmp_a])             \n\t"
>-    "swc1       %[f3],          128(%[tmp_a])             \n\t"
>-    "swc1       %[f0],          384(%[tmp_a])             \n\t"
>-    "bgtz       %[count],       1b                        \n\t"
>-    " addiu     %[tmp_a],       %[tmp_a],   8             \n\t"
>-    ".set       pop                                       \n\t"
>-    : [f0] "=&f" (f0), [f1] "=&f" (f1), [f2] "=&f" (f2), [f3] "=&f" (f3),
>-      [f4] "=&f" (f4), [f5] "=&f" (f5), [f6] "=&f" (f6), [f7] "=&f" (f7),
>-      [f8] "=&f" (f8), [tmp_a] "=&r" (tmp_a),
>-      [count] "=&r" (count)
>-    : [a] "r" (a)
>-    : "memory"
>-  );
>+  __asm __volatile(
>+      ".set       push                                      \n\t"
>+      ".set       noreorder                                 \n\t"
>+      "addiu      %[tmp_a],       %[a],         0           \n\t"
>+      "addiu      %[count],       $zero,        16          \n\t"
>+      "1:                                                    \n\t"
>+      "addiu      %[count],       %[count],     -1          \n\t"
>+      "lwc1       %[f0],          0(%[tmp_a])               \n\t"
>+      "lwc1       %[f2],          128(%[tmp_a])             \n\t"
>+      "lwc1       %[f4],          256(%[tmp_a])             \n\t"
>+      "lwc1       %[f6],          384(%[tmp_a])             \n\t"
>+      "lwc1       %[f1],          4(%[tmp_a])               \n\t"
>+      "lwc1       %[f3],          132(%[tmp_a])             \n\t"
>+      "lwc1       %[f5],          260(%[tmp_a])             \n\t"
>+      "lwc1       %[f7],          388(%[tmp_a])             \n\t"
>+      "add.s      %[f8],          %[f0],        %[f2]       \n\t"
>+      "sub.s      %[f0],          %[f0],        %[f2]       \n\t"
>+      "add.s      %[f2],          %[f4],        %[f6]       \n\t"
>+      "sub.s      %[f4],          %[f4],        %[f6]       \n\t"
>+      "add.s      %[f6],          %[f1],        %[f3]       \n\t"
>+      "sub.s      %[f1],          %[f1],        %[f3]       \n\t"
>+      "add.s      %[f3],          %[f5],        %[f7]       \n\t"
>+      "sub.s      %[f5],          %[f5],        %[f7]       \n\t"
>+      "add.s      %[f7],          %[f8],        %[f2]       \n\t"
>+      "sub.s      %[f8],          %[f8],        %[f2]       \n\t"
>+      "add.s      %[f2],          %[f1],        %[f4]       \n\t"
>+      "sub.s      %[f1],          %[f1],        %[f4]       \n\t"
>+      "add.s      %[f4],          %[f6],        %[f3]       \n\t"
>+      "sub.s      %[f6],          %[f6],        %[f3]       \n\t"
>+      "sub.s      %[f3],          %[f0],        %[f5]       \n\t"
>+      "add.s      %[f0],          %[f0],        %[f5]       \n\t"
>+      "swc1       %[f7],          0(%[tmp_a])               \n\t"
>+      "swc1       %[f8],          256(%[tmp_a])             \n\t"
>+      "swc1       %[f2],          132(%[tmp_a])             \n\t"
>+      "swc1       %[f1],          388(%[tmp_a])             \n\t"
>+      "swc1       %[f4],          4(%[tmp_a])               \n\t"
>+      "swc1       %[f6],          260(%[tmp_a])             \n\t"
>+      "swc1       %[f3],          128(%[tmp_a])             \n\t"
>+      "swc1       %[f0],          384(%[tmp_a])             \n\t"
>+      "bgtz       %[count],       1b                        \n\t"
>+      " addiu     %[tmp_a],       %[tmp_a],   8             \n\t"
>+      ".set       pop                                       \n\t"
>+      : [f0] "=&f"(f0), [f1] "=&f"(f1), [f2] "=&f"(f2), [f3] "=&f"(f3),
>+        [f4] "=&f"(f4), [f5] "=&f"(f5), [f6] "=&f"(f6), [f7] "=&f"(f7),
>+        [f8] "=&f"(f8), [tmp_a] "=&r"(tmp_a), [count] "=&r"(count)
>+      : [a] "r"(a)
>+      : "memory");
> }
> 
> void cftbsub_128_mips(float* a) {
>@@ -873,55 +937,54 @@ void cftbsub_128_mips(float* a) {
>   cft1st_128_mips(a);
>   cftmdl_128_mips(a);
> 
>-  __asm __volatile (
>-    ".set       push                                        \n\t"
>-    ".set       noreorder                                   \n\t"
>-    "addiu      %[tmp_a],   %[a],           0               \n\t"
>-    "addiu      %[count],   $zero,          16              \n\t"
>-   "1:                                                      \n\t"
>-    "addiu      %[count],   %[count],       -1              \n\t"
>-    "lwc1       %[f0],      0(%[tmp_a])                     \n\t"
>-    "lwc1       %[f2],      128(%[tmp_a])                   \n\t"
>-    "lwc1       %[f4],      256(%[tmp_a])                   \n\t"
>-    "lwc1       %[f6],      384(%[tmp_a])                   \n\t"
>-    "lwc1       %[f1],      4(%[tmp_a])                     \n\t"
>-    "lwc1       %[f3],      132(%[tmp_a])                   \n\t"
>-    "lwc1       %[f5],      260(%[tmp_a])                   \n\t"
>-    "lwc1       %[f7],      388(%[tmp_a])                   \n\t"
>-    "add.s      %[f8],      %[f0],          %[f2]           \n\t"
>-    "sub.s      %[f0],      %[f0],          %[f2]           \n\t"
>-    "add.s      %[f2],      %[f4],          %[f6]           \n\t"
>-    "sub.s      %[f4],      %[f4],          %[f6]           \n\t"
>-    "add.s      %[f6],      %[f1],          %[f3]           \n\t"
>-    "sub.s      %[f1],      %[f3],          %[f1]           \n\t"
>-    "add.s      %[f3],      %[f5],          %[f7]           \n\t"
>-    "sub.s      %[f5],      %[f5],          %[f7]           \n\t"
>-    "add.s      %[f7],      %[f8],          %[f2]           \n\t"
>-    "sub.s      %[f8],      %[f8],          %[f2]           \n\t"
>-    "sub.s      %[f2],      %[f1],          %[f4]           \n\t"
>-    "add.s      %[f1],      %[f1],          %[f4]           \n\t"
>-    "add.s      %[f4],      %[f3],          %[f6]           \n\t"
>-    "sub.s      %[f6],      %[f3],          %[f6]           \n\t"
>-    "sub.s      %[f3],      %[f0],          %[f5]           \n\t"
>-    "add.s      %[f0],      %[f0],          %[f5]           \n\t"
>-    "neg.s      %[f4],      %[f4]                           \n\t"
>-    "swc1       %[f7],      0(%[tmp_a])                     \n\t"
>-    "swc1       %[f8],      256(%[tmp_a])                   \n\t"
>-    "swc1       %[f2],      132(%[tmp_a])                   \n\t"
>-    "swc1       %[f1],      388(%[tmp_a])                   \n\t"
>-    "swc1       %[f6],      260(%[tmp_a])                   \n\t"
>-    "swc1       %[f3],      128(%[tmp_a])                   \n\t"
>-    "swc1       %[f0],      384(%[tmp_a])                   \n\t"
>-    "swc1       %[f4],       4(%[tmp_a])                     \n\t"
>-    "bgtz       %[count],   1b                              \n\t"
>-    " addiu     %[tmp_a],   %[tmp_a],       8               \n\t"
>-    ".set       pop                                         \n\t"
>-    : [f0] "=&f" (f0), [f1] "=&f" (f1), [f2] "=&f" (f2), [f3] "=&f" (f3),
>-      [f4] "=&f" (f4), [f5] "=&f" (f5), [f6] "=&f" (f6), [f7] "=&f" (f7),
>-      [f8] "=&f" (f8), [tmp_a] "=&r" (tmp_a), [count] "=&r" (count)
>-    : [a] "r" (a)
>-    : "memory"
>-  );
>+  __asm __volatile(
>+      ".set       push                                        \n\t"
>+      ".set       noreorder                                   \n\t"
>+      "addiu      %[tmp_a],   %[a],           0               \n\t"
>+      "addiu      %[count],   $zero,          16              \n\t"
>+      "1:                                                      \n\t"
>+      "addiu      %[count],   %[count],       -1              \n\t"
>+      "lwc1       %[f0],      0(%[tmp_a])                     \n\t"
>+      "lwc1       %[f2],      128(%[tmp_a])                   \n\t"
>+      "lwc1       %[f4],      256(%[tmp_a])                   \n\t"
>+      "lwc1       %[f6],      384(%[tmp_a])                   \n\t"
>+      "lwc1       %[f1],      4(%[tmp_a])                     \n\t"
>+      "lwc1       %[f3],      132(%[tmp_a])                   \n\t"
>+      "lwc1       %[f5],      260(%[tmp_a])                   \n\t"
>+      "lwc1       %[f7],      388(%[tmp_a])                   \n\t"
>+      "add.s      %[f8],      %[f0],          %[f2]           \n\t"
>+      "sub.s      %[f0],      %[f0],          %[f2]           \n\t"
>+      "add.s      %[f2],      %[f4],          %[f6]           \n\t"
>+      "sub.s      %[f4],      %[f4],          %[f6]           \n\t"
>+      "add.s      %[f6],      %[f1],          %[f3]           \n\t"
>+      "sub.s      %[f1],      %[f3],          %[f1]           \n\t"
>+      "add.s      %[f3],      %[f5],          %[f7]           \n\t"
>+      "sub.s      %[f5],      %[f5],          %[f7]           \n\t"
>+      "add.s      %[f7],      %[f8],          %[f2]           \n\t"
>+      "sub.s      %[f8],      %[f8],          %[f2]           \n\t"
>+      "sub.s      %[f2],      %[f1],          %[f4]           \n\t"
>+      "add.s      %[f1],      %[f1],          %[f4]           \n\t"
>+      "add.s      %[f4],      %[f3],          %[f6]           \n\t"
>+      "sub.s      %[f6],      %[f3],          %[f6]           \n\t"
>+      "sub.s      %[f3],      %[f0],          %[f5]           \n\t"
>+      "add.s      %[f0],      %[f0],          %[f5]           \n\t"
>+      "neg.s      %[f4],      %[f4]                           \n\t"
>+      "swc1       %[f7],      0(%[tmp_a])                     \n\t"
>+      "swc1       %[f8],      256(%[tmp_a])                   \n\t"
>+      "swc1       %[f2],      132(%[tmp_a])                   \n\t"
>+      "swc1       %[f1],      388(%[tmp_a])                   \n\t"
>+      "swc1       %[f6],      260(%[tmp_a])                   \n\t"
>+      "swc1       %[f3],      128(%[tmp_a])                   \n\t"
>+      "swc1       %[f0],      384(%[tmp_a])                   \n\t"
>+      "swc1       %[f4],       4(%[tmp_a])                     \n\t"
>+      "bgtz       %[count],   1b                              \n\t"
>+      " addiu     %[tmp_a],   %[tmp_a],       8               \n\t"
>+      ".set       pop                                         \n\t"
>+      : [f0] "=&f"(f0), [f1] "=&f"(f1), [f2] "=&f"(f2), [f3] "=&f"(f3),
>+        [f4] "=&f"(f4), [f5] "=&f"(f5), [f6] "=&f"(f6), [f7] "=&f"(f7),
>+        [f8] "=&f"(f8), [tmp_a] "=&r"(tmp_a), [count] "=&r"(count)
>+      : [a] "r"(a)
>+      : "memory");
> }
> 
> void rftfsub_128_mips(float* a) {
>@@ -931,254 +994,252 @@ void rftfsub_128_mips(float* a) {
>   float* a2 = &a[126];
>   const float* c1 = &c[1];
>   const float* c2 = &c[31];
>-  float f1, f2, f3 ,f4, f5, f6, f7, f8, f9, f10, f11, f12, f13, f14, f15;
>+  float f1, f2, f3, f4, f5, f6, f7, f8, f9, f10, f11, f12, f13, f14, f15;
>   int count;
> 
>-  __asm __volatile (
>-    ".set      push                                             \n\t"
>-    ".set      noreorder                                        \n\t"
>-    "lwc1      %[f6],       0(%[c2])                            \n\t"
>-    "lwc1      %[f1],       0(%[a1])                            \n\t"
>-    "lwc1      %[f2],       0(%[a2])                            \n\t"
>-    "lwc1      %[f3],       4(%[a1])                            \n\t"
>-    "lwc1      %[f4],       4(%[a2])                            \n\t"
>-    "lwc1      %[f5],       0(%[c1])                            \n\t"
>-    "sub.s     %[f6],       %[f0],        %[f6]                 \n\t"
>-    "sub.s     %[f7],       %[f1],        %[f2]                 \n\t"
>-    "add.s     %[f8],       %[f3],        %[f4]                 \n\t"
>-    "addiu     %[count],    $zero,        15                    \n\t"
>-    "mul.s     %[f9],       %[f6],        %[f7]                 \n\t"
>-    "mul.s     %[f6],       %[f6],        %[f8]                 \n\t"
>+  __asm __volatile(
>+      ".set      push                                             \n\t"
>+      ".set      noreorder                                        \n\t"
>+      "lwc1      %[f6],       0(%[c2])                            \n\t"
>+      "lwc1      %[f1],       0(%[a1])                            \n\t"
>+      "lwc1      %[f2],       0(%[a2])                            \n\t"
>+      "lwc1      %[f3],       4(%[a1])                            \n\t"
>+      "lwc1      %[f4],       4(%[a2])                            \n\t"
>+      "lwc1      %[f5],       0(%[c1])                            \n\t"
>+      "sub.s     %[f6],       %[f0],        %[f6]                 \n\t"
>+      "sub.s     %[f7],       %[f1],        %[f2]                 \n\t"
>+      "add.s     %[f8],       %[f3],        %[f4]                 \n\t"
>+      "addiu     %[count],    $zero,        15                    \n\t"
>+      "mul.s     %[f9],       %[f6],        %[f7]                 \n\t"
>+      "mul.s     %[f6],       %[f6],        %[f8]                 \n\t"
> #if !defined(MIPS32_R2_LE)
>-    "mul.s     %[f8],       %[f5],        %[f8]                 \n\t"
>-    "mul.s     %[f5],       %[f5],        %[f7]                 \n\t"
>-    "sub.s     %[f9],       %[f9],        %[f8]                 \n\t"
>-    "add.s     %[f6],       %[f6],        %[f5]                 \n\t"
>+      "mul.s     %[f8],       %[f5],        %[f8]                 \n\t"
>+      "mul.s     %[f5],       %[f5],        %[f7]                 \n\t"
>+      "sub.s     %[f9],       %[f9],        %[f8]                 \n\t"
>+      "add.s     %[f6],       %[f6],        %[f5]                 \n\t"
> #else
>-    "nmsub.s   %[f9],       %[f9],        %[f5],      %[f8]     \n\t"
>-    "madd.s    %[f6],       %[f6],        %[f5],      %[f7]     \n\t"
>+      "nmsub.s   %[f9],       %[f9],        %[f5],      %[f8]     \n\t"
>+      "madd.s    %[f6],       %[f6],        %[f5],      %[f7]     \n\t"
> #endif
>-    "sub.s     %[f1],       %[f1],        %[f9]                 \n\t"
>-    "add.s     %[f2],       %[f2],        %[f9]                 \n\t"
>-    "sub.s     %[f3],       %[f3],        %[f6]                 \n\t"
>-    "sub.s     %[f4],       %[f4],        %[f6]                 \n\t"
>-    "swc1      %[f1],       0(%[a1])                            \n\t"
>-    "swc1      %[f2],       0(%[a2])                            \n\t"
>-    "swc1      %[f3],       4(%[a1])                            \n\t"
>-    "swc1      %[f4],       4(%[a2])                            \n\t"
>-    "addiu     %[a1],       %[a1],        8                     \n\t"
>-    "addiu     %[a2],       %[a2],        -8                    \n\t"
>-    "addiu     %[c1],       %[c1],        4                     \n\t"
>-    "addiu     %[c2],       %[c2],        -4                    \n\t"
>-   "1:                                                          \n\t"
>-    "lwc1      %[f6],       0(%[c2])                            \n\t"
>-    "lwc1      %[f1],       0(%[a1])                            \n\t"
>-    "lwc1      %[f2],       0(%[a2])                            \n\t"
>-    "lwc1      %[f3],       4(%[a1])                            \n\t"
>-    "lwc1      %[f4],       4(%[a2])                            \n\t"
>-    "lwc1      %[f5],       0(%[c1])                            \n\t"
>-    "sub.s     %[f6],       %[f0],        %[f6]                 \n\t"
>-    "sub.s     %[f7],       %[f1],        %[f2]                 \n\t"
>-    "add.s     %[f8],       %[f3],        %[f4]                 \n\t"
>-    "lwc1      %[f10],      -4(%[c2])                           \n\t"
>-    "lwc1      %[f11],      8(%[a1])                            \n\t"
>-    "lwc1      %[f12],      -8(%[a2])                           \n\t"
>-    "mul.s     %[f9],       %[f6],        %[f7]                 \n\t"
>-    "mul.s     %[f6],       %[f6],        %[f8]                 \n\t"
>+      "sub.s     %[f1],       %[f1],        %[f9]                 \n\t"
>+      "add.s     %[f2],       %[f2],        %[f9]                 \n\t"
>+      "sub.s     %[f3],       %[f3],        %[f6]                 \n\t"
>+      "sub.s     %[f4],       %[f4],        %[f6]                 \n\t"
>+      "swc1      %[f1],       0(%[a1])                            \n\t"
>+      "swc1      %[f2],       0(%[a2])                            \n\t"
>+      "swc1      %[f3],       4(%[a1])                            \n\t"
>+      "swc1      %[f4],       4(%[a2])                            \n\t"
>+      "addiu     %[a1],       %[a1],        8                     \n\t"
>+      "addiu     %[a2],       %[a2],        -8                    \n\t"
>+      "addiu     %[c1],       %[c1],        4                     \n\t"
>+      "addiu     %[c2],       %[c2],        -4                    \n\t"
>+      "1:                                                          \n\t"
>+      "lwc1      %[f6],       0(%[c2])                            \n\t"
>+      "lwc1      %[f1],       0(%[a1])                            \n\t"
>+      "lwc1      %[f2],       0(%[a2])                            \n\t"
>+      "lwc1      %[f3],       4(%[a1])                            \n\t"
>+      "lwc1      %[f4],       4(%[a2])                            \n\t"
>+      "lwc1      %[f5],       0(%[c1])                            \n\t"
>+      "sub.s     %[f6],       %[f0],        %[f6]                 \n\t"
>+      "sub.s     %[f7],       %[f1],        %[f2]                 \n\t"
>+      "add.s     %[f8],       %[f3],        %[f4]                 \n\t"
>+      "lwc1      %[f10],      -4(%[c2])                           \n\t"
>+      "lwc1      %[f11],      8(%[a1])                            \n\t"
>+      "lwc1      %[f12],      -8(%[a2])                           \n\t"
>+      "mul.s     %[f9],       %[f6],        %[f7]                 \n\t"
>+      "mul.s     %[f6],       %[f6],        %[f8]                 \n\t"
> #if !defined(MIPS32_R2_LE)
>-    "mul.s     %[f8],       %[f5],        %[f8]                 \n\t"
>-    "mul.s     %[f5],       %[f5],        %[f7]                 \n\t"
>-    "lwc1      %[f13],      12(%[a1])                           \n\t"
>-    "lwc1      %[f14],      -4(%[a2])                           \n\t"
>-    "lwc1      %[f15],      4(%[c1])                            \n\t"
>-    "sub.s     %[f9],       %[f9],        %[f8]                 \n\t"
>-    "add.s     %[f6],       %[f6],        %[f5]                 \n\t"
>+      "mul.s     %[f8],       %[f5],        %[f8]                 \n\t"
>+      "mul.s     %[f5],       %[f5],        %[f7]                 \n\t"
>+      "lwc1      %[f13],      12(%[a1])                           \n\t"
>+      "lwc1      %[f14],      -4(%[a2])                           \n\t"
>+      "lwc1      %[f15],      4(%[c1])                            \n\t"
>+      "sub.s     %[f9],       %[f9],        %[f8]                 \n\t"
>+      "add.s     %[f6],       %[f6],        %[f5]                 \n\t"
> #else
>-    "lwc1      %[f13],      12(%[a1])                           \n\t"
>-    "lwc1      %[f14],      -4(%[a2])                           \n\t"
>-    "lwc1      %[f15],      4(%[c1])                            \n\t"
>-    "nmsub.s   %[f9],       %[f9],        %[f5],      %[f8]     \n\t"
>-    "madd.s    %[f6],       %[f6],        %[f5],      %[f7]     \n\t"
>+      "lwc1      %[f13],      12(%[a1])                           \n\t"
>+      "lwc1      %[f14],      -4(%[a2])                           \n\t"
>+      "lwc1      %[f15],      4(%[c1])                            \n\t"
>+      "nmsub.s   %[f9],       %[f9],        %[f5],      %[f8]     \n\t"
>+      "madd.s    %[f6],       %[f6],        %[f5],      %[f7]     \n\t"
> #endif
>-    "sub.s     %[f10],      %[f0],        %[f10]                \n\t"
>-    "sub.s     %[f5],       %[f11],       %[f12]                \n\t"
>-    "add.s     %[f7],       %[f13],       %[f14]                \n\t"
>-    "sub.s     %[f1],       %[f1],        %[f9]                 \n\t"
>-    "add.s     %[f2],       %[f2],        %[f9]                 \n\t"
>-    "sub.s     %[f3],       %[f3],        %[f6]                 \n\t"
>-    "mul.s     %[f8],       %[f10],       %[f5]                 \n\t"
>-    "mul.s     %[f10],      %[f10],       %[f7]                 \n\t"
>+      "sub.s     %[f10],      %[f0],        %[f10]                \n\t"
>+      "sub.s     %[f5],       %[f11],       %[f12]                \n\t"
>+      "add.s     %[f7],       %[f13],       %[f14]                \n\t"
>+      "sub.s     %[f1],       %[f1],        %[f9]                 \n\t"
>+      "add.s     %[f2],       %[f2],        %[f9]                 \n\t"
>+      "sub.s     %[f3],       %[f3],        %[f6]                 \n\t"
>+      "mul.s     %[f8],       %[f10],       %[f5]                 \n\t"
>+      "mul.s     %[f10],      %[f10],       %[f7]                 \n\t"
> #if !defined(MIPS32_R2_LE)
>-    "mul.s     %[f9],       %[f15],       %[f7]                 \n\t"
>-    "mul.s     %[f15],      %[f15],       %[f5]                 \n\t"
>-    "sub.s     %[f4],       %[f4],        %[f6]                 \n\t"
>-    "swc1      %[f1],       0(%[a1])                            \n\t"
>-    "swc1      %[f2],       0(%[a2])                            \n\t"
>-    "sub.s     %[f8],       %[f8],        %[f9]                 \n\t"
>-    "add.s     %[f10],      %[f10],       %[f15]                \n\t"
>+      "mul.s     %[f9],       %[f15],       %[f7]                 \n\t"
>+      "mul.s     %[f15],      %[f15],       %[f5]                 \n\t"
>+      "sub.s     %[f4],       %[f4],        %[f6]                 \n\t"
>+      "swc1      %[f1],       0(%[a1])                            \n\t"
>+      "swc1      %[f2],       0(%[a2])                            \n\t"
>+      "sub.s     %[f8],       %[f8],        %[f9]                 \n\t"
>+      "add.s     %[f10],      %[f10],       %[f15]                \n\t"
> #else
>-    "swc1      %[f1],       0(%[a1])                            \n\t"
>-    "swc1      %[f2],       0(%[a2])                            \n\t"
>-    "sub.s     %[f4],       %[f4],        %[f6]                 \n\t"
>-    "nmsub.s   %[f8],       %[f8],        %[f15],     %[f7]     \n\t"
>-    "madd.s    %[f10],      %[f10],       %[f15],     %[f5]     \n\t"
>+      "swc1      %[f1],       0(%[a1])                            \n\t"
>+      "swc1      %[f2],       0(%[a2])                            \n\t"
>+      "sub.s     %[f4],       %[f4],        %[f6]                 \n\t"
>+      "nmsub.s   %[f8],       %[f8],        %[f15],     %[f7]     \n\t"
>+      "madd.s    %[f10],      %[f10],       %[f15],     %[f5]     \n\t"
> #endif
>-    "swc1      %[f3],       4(%[a1])                            \n\t"
>-    "swc1      %[f4],       4(%[a2])                            \n\t"
>-    "sub.s     %[f11],      %[f11],       %[f8]                 \n\t"
>-    "add.s     %[f12],      %[f12],       %[f8]                 \n\t"
>-    "sub.s     %[f13],      %[f13],       %[f10]                \n\t"
>-    "sub.s     %[f14],      %[f14],       %[f10]                \n\t"
>-    "addiu     %[c2],       %[c2],        -8                    \n\t"
>-    "addiu     %[c1],       %[c1],        8                     \n\t"
>-    "swc1      %[f11],      8(%[a1])                            \n\t"
>-    "swc1      %[f12],      -8(%[a2])                           \n\t"
>-    "swc1      %[f13],      12(%[a1])                           \n\t"
>-    "swc1      %[f14],      -4(%[a2])                           \n\t"
>-    "addiu     %[a1],       %[a1],        16                    \n\t"
>-    "addiu     %[count],    %[count],     -1                    \n\t"
>-    "bgtz      %[count],    1b                                  \n\t"
>-    " addiu    %[a2],       %[a2],        -16                   \n\t"
>-    ".set      pop                                              \n\t"
>-    : [a1] "+r" (a1), [a2] "+r" (a2), [c1] "+r" (c1), [c2] "+r" (c2),
>-      [f1] "=&f" (f1), [f2] "=&f" (f2), [f3] "=&f" (f3), [f4] "=&f" (f4),
>-      [f5] "=&f" (f5), [f6] "=&f" (f6), [f7] "=&f" (f7), [f8] "=&f" (f8),
>-      [f9] "=&f" (f9), [f10] "=&f" (f10), [f11] "=&f" (f11), [f12] "=&f" (f12),
>-      [f13] "=&f" (f13), [f14] "=&f" (f14), [f15] "=&f" (f15),
>-      [count] "=&r" (count)
>-    : [f0] "f" (f0)
>-    : "memory"
>-  );
>+      "swc1      %[f3],       4(%[a1])                            \n\t"
>+      "swc1      %[f4],       4(%[a2])                            \n\t"
>+      "sub.s     %[f11],      %[f11],       %[f8]                 \n\t"
>+      "add.s     %[f12],      %[f12],       %[f8]                 \n\t"
>+      "sub.s     %[f13],      %[f13],       %[f10]                \n\t"
>+      "sub.s     %[f14],      %[f14],       %[f10]                \n\t"
>+      "addiu     %[c2],       %[c2],        -8                    \n\t"
>+      "addiu     %[c1],       %[c1],        8                     \n\t"
>+      "swc1      %[f11],      8(%[a1])                            \n\t"
>+      "swc1      %[f12],      -8(%[a2])                           \n\t"
>+      "swc1      %[f13],      12(%[a1])                           \n\t"
>+      "swc1      %[f14],      -4(%[a2])                           \n\t"
>+      "addiu     %[a1],       %[a1],        16                    \n\t"
>+      "addiu     %[count],    %[count],     -1                    \n\t"
>+      "bgtz      %[count],    1b                                  \n\t"
>+      " addiu    %[a2],       %[a2],        -16                   \n\t"
>+      ".set      pop                                              \n\t"
>+      : [a1] "+r"(a1), [a2] "+r"(a2), [c1] "+r"(c1), [c2] "+r"(c2),
>+        [f1] "=&f"(f1), [f2] "=&f"(f2), [f3] "=&f"(f3), [f4] "=&f"(f4),
>+        [f5] "=&f"(f5), [f6] "=&f"(f6), [f7] "=&f"(f7), [f8] "=&f"(f8),
>+        [f9] "=&f"(f9), [f10] "=&f"(f10), [f11] "=&f"(f11), [f12] "=&f"(f12),
>+        [f13] "=&f"(f13), [f14] "=&f"(f14), [f15] "=&f"(f15),
>+        [count] "=&r"(count)
>+      : [f0] "f"(f0)
>+      : "memory");
> }
> 
> void rftbsub_128_mips(float* a) {
>-  const float *c = rdft_w + 32;
>+  const float* c = rdft_w + 32;
>   const float f0 = 0.5f;
>   float* a1 = &a[2];
>   float* a2 = &a[126];
>   const float* c1 = &c[1];
>   const float* c2 = &c[31];
>-  float f1, f2, f3 ,f4, f5, f6, f7, f8, f9, f10, f11, f12, f13, f14, f15;
>+  float f1, f2, f3, f4, f5, f6, f7, f8, f9, f10, f11, f12, f13, f14, f15;
>   int count;
> 
>   a[1] = -a[1];
>   a[65] = -a[65];
> 
>-  __asm __volatile (
>-    ".set      push                                             \n\t"
>-    ".set      noreorder                                        \n\t"
>-    "lwc1      %[f6],       0(%[c2])                            \n\t"
>-    "lwc1      %[f1],       0(%[a1])                            \n\t"
>-    "lwc1      %[f2],       0(%[a2])                            \n\t"
>-    "lwc1      %[f3],       4(%[a1])                            \n\t"
>-    "lwc1      %[f4],       4(%[a2])                            \n\t"
>-    "lwc1      %[f5],       0(%[c1])                            \n\t"
>-    "sub.s     %[f6],       %[f0],        %[f6]                 \n\t"
>-    "sub.s     %[f7],       %[f1],        %[f2]                 \n\t"
>-    "add.s     %[f8],       %[f3],        %[f4]                 \n\t"
>-    "addiu     %[count],    $zero,        15                    \n\t"
>-    "mul.s     %[f9],       %[f6],        %[f7]                 \n\t"
>-    "mul.s     %[f6],       %[f6],        %[f8]                 \n\t"
>+  __asm __volatile(
>+      ".set      push                                             \n\t"
>+      ".set      noreorder                                        \n\t"
>+      "lwc1      %[f6],       0(%[c2])                            \n\t"
>+      "lwc1      %[f1],       0(%[a1])                            \n\t"
>+      "lwc1      %[f2],       0(%[a2])                            \n\t"
>+      "lwc1      %[f3],       4(%[a1])                            \n\t"
>+      "lwc1      %[f4],       4(%[a2])                            \n\t"
>+      "lwc1      %[f5],       0(%[c1])                            \n\t"
>+      "sub.s     %[f6],       %[f0],        %[f6]                 \n\t"
>+      "sub.s     %[f7],       %[f1],        %[f2]                 \n\t"
>+      "add.s     %[f8],       %[f3],        %[f4]                 \n\t"
>+      "addiu     %[count],    $zero,        15                    \n\t"
>+      "mul.s     %[f9],       %[f6],        %[f7]                 \n\t"
>+      "mul.s     %[f6],       %[f6],        %[f8]                 \n\t"
> #if !defined(MIPS32_R2_LE)
>-    "mul.s     %[f8],       %[f5],        %[f8]                 \n\t"
>-    "mul.s     %[f5],       %[f5],        %[f7]                 \n\t"
>-    "add.s     %[f9],       %[f9],        %[f8]                 \n\t"
>-    "sub.s     %[f6],       %[f6],        %[f5]                 \n\t"
>+      "mul.s     %[f8],       %[f5],        %[f8]                 \n\t"
>+      "mul.s     %[f5],       %[f5],        %[f7]                 \n\t"
>+      "add.s     %[f9],       %[f9],        %[f8]                 \n\t"
>+      "sub.s     %[f6],       %[f6],        %[f5]                 \n\t"
> #else
>-    "madd.s    %[f9],       %[f9],        %[f5],      %[f8]     \n\t"
>-    "nmsub.s   %[f6],       %[f6],        %[f5],      %[f7]     \n\t"
>+      "madd.s    %[f9],       %[f9],        %[f5],      %[f8]     \n\t"
>+      "nmsub.s   %[f6],       %[f6],        %[f5],      %[f7]     \n\t"
> #endif
>-    "sub.s     %[f1],       %[f1],        %[f9]                 \n\t"
>-    "add.s     %[f2],       %[f2],        %[f9]                 \n\t"
>-    "sub.s     %[f3],       %[f6],        %[f3]                 \n\t"
>-    "sub.s     %[f4],       %[f6],        %[f4]                 \n\t"
>-    "swc1      %[f1],       0(%[a1])                            \n\t"
>-    "swc1      %[f2],       0(%[a2])                            \n\t"
>-    "swc1      %[f3],       4(%[a1])                            \n\t"
>-    "swc1      %[f4],       4(%[a2])                            \n\t"
>-    "addiu     %[a1],       %[a1],        8                     \n\t"
>-    "addiu     %[a2],       %[a2],        -8                    \n\t"
>-    "addiu     %[c1],       %[c1],        4                     \n\t"
>-    "addiu     %[c2],       %[c2],        -4                    \n\t"
>-   "1:                                                          \n\t"
>-    "lwc1      %[f6],       0(%[c2])                            \n\t"
>-    "lwc1      %[f1],       0(%[a1])                            \n\t"
>-    "lwc1      %[f2],       0(%[a2])                            \n\t"
>-    "lwc1      %[f3],       4(%[a1])                            \n\t"
>-    "lwc1      %[f4],       4(%[a2])                            \n\t"
>-    "lwc1      %[f5],       0(%[c1])                            \n\t"
>-    "sub.s     %[f6],       %[f0],        %[f6]                 \n\t"
>-    "sub.s     %[f7],       %[f1],        %[f2]                 \n\t"
>-    "add.s     %[f8],       %[f3],        %[f4]                 \n\t"
>-    "lwc1      %[f10],      -4(%[c2])                           \n\t"
>-    "lwc1      %[f11],      8(%[a1])                            \n\t"
>-    "lwc1      %[f12],      -8(%[a2])                           \n\t"
>-    "mul.s     %[f9],       %[f6],        %[f7]                 \n\t"
>-    "mul.s     %[f6],       %[f6],        %[f8]                 \n\t"
>+      "sub.s     %[f1],       %[f1],        %[f9]                 \n\t"
>+      "add.s     %[f2],       %[f2],        %[f9]                 \n\t"
>+      "sub.s     %[f3],       %[f6],        %[f3]                 \n\t"
>+      "sub.s     %[f4],       %[f6],        %[f4]                 \n\t"
>+      "swc1      %[f1],       0(%[a1])                            \n\t"
>+      "swc1      %[f2],       0(%[a2])                            \n\t"
>+      "swc1      %[f3],       4(%[a1])                            \n\t"
>+      "swc1      %[f4],       4(%[a2])                            \n\t"
>+      "addiu     %[a1],       %[a1],        8                     \n\t"
>+      "addiu     %[a2],       %[a2],        -8                    \n\t"
>+      "addiu     %[c1],       %[c1],        4                     \n\t"
>+      "addiu     %[c2],       %[c2],        -4                    \n\t"
>+      "1:                                                          \n\t"
>+      "lwc1      %[f6],       0(%[c2])                            \n\t"
>+      "lwc1      %[f1],       0(%[a1])                            \n\t"
>+      "lwc1      %[f2],       0(%[a2])                            \n\t"
>+      "lwc1      %[f3],       4(%[a1])                            \n\t"
>+      "lwc1      %[f4],       4(%[a2])                            \n\t"
>+      "lwc1      %[f5],       0(%[c1])                            \n\t"
>+      "sub.s     %[f6],       %[f0],        %[f6]                 \n\t"
>+      "sub.s     %[f7],       %[f1],        %[f2]                 \n\t"
>+      "add.s     %[f8],       %[f3],        %[f4]                 \n\t"
>+      "lwc1      %[f10],      -4(%[c2])                           \n\t"
>+      "lwc1      %[f11],      8(%[a1])                            \n\t"
>+      "lwc1      %[f12],      -8(%[a2])                           \n\t"
>+      "mul.s     %[f9],       %[f6],        %[f7]                 \n\t"
>+      "mul.s     %[f6],       %[f6],        %[f8]                 \n\t"
> #if !defined(MIPS32_R2_LE)
>-    "mul.s     %[f8],       %[f5],        %[f8]                 \n\t"
>-    "mul.s     %[f5],       %[f5],        %[f7]                 \n\t"
>-    "lwc1      %[f13],      12(%[a1])                           \n\t"
>-    "lwc1      %[f14],      -4(%[a2])                           \n\t"
>-    "lwc1      %[f15],      4(%[c1])                            \n\t"
>-    "add.s     %[f9],       %[f9],        %[f8]                 \n\t"
>-    "sub.s     %[f6],       %[f6],        %[f5]                 \n\t"
>+      "mul.s     %[f8],       %[f5],        %[f8]                 \n\t"
>+      "mul.s     %[f5],       %[f5],        %[f7]                 \n\t"
>+      "lwc1      %[f13],      12(%[a1])                           \n\t"
>+      "lwc1      %[f14],      -4(%[a2])                           \n\t"
>+      "lwc1      %[f15],      4(%[c1])                            \n\t"
>+      "add.s     %[f9],       %[f9],        %[f8]                 \n\t"
>+      "sub.s     %[f6],       %[f6],        %[f5]                 \n\t"
> #else
>-    "lwc1      %[f13],      12(%[a1])                           \n\t"
>-    "lwc1      %[f14],      -4(%[a2])                           \n\t"
>-    "lwc1      %[f15],      4(%[c1])                            \n\t"
>-    "madd.s    %[f9],       %[f9],        %[f5],      %[f8]     \n\t"
>-    "nmsub.s   %[f6],       %[f6],        %[f5],      %[f7]     \n\t"
>+      "lwc1      %[f13],      12(%[a1])                           \n\t"
>+      "lwc1      %[f14],      -4(%[a2])                           \n\t"
>+      "lwc1      %[f15],      4(%[c1])                            \n\t"
>+      "madd.s    %[f9],       %[f9],        %[f5],      %[f8]     \n\t"
>+      "nmsub.s   %[f6],       %[f6],        %[f5],      %[f7]     \n\t"
> #endif
>-    "sub.s     %[f10],      %[f0],        %[f10]                \n\t"
>-    "sub.s     %[f5],       %[f11],       %[f12]                \n\t"
>-    "add.s     %[f7],       %[f13],       %[f14]                \n\t"
>-    "sub.s     %[f1],       %[f1],        %[f9]                 \n\t"
>-    "add.s     %[f2],       %[f2],        %[f9]                 \n\t"
>-    "sub.s     %[f3],       %[f6],        %[f3]                 \n\t"
>-    "mul.s     %[f8],       %[f10],       %[f5]                 \n\t"
>-    "mul.s     %[f10],      %[f10],       %[f7]                 \n\t"
>+      "sub.s     %[f10],      %[f0],        %[f10]                \n\t"
>+      "sub.s     %[f5],       %[f11],       %[f12]                \n\t"
>+      "add.s     %[f7],       %[f13],       %[f14]                \n\t"
>+      "sub.s     %[f1],       %[f1],        %[f9]                 \n\t"
>+      "add.s     %[f2],       %[f2],        %[f9]                 \n\t"
>+      "sub.s     %[f3],       %[f6],        %[f3]                 \n\t"
>+      "mul.s     %[f8],       %[f10],       %[f5]                 \n\t"
>+      "mul.s     %[f10],      %[f10],       %[f7]                 \n\t"
> #if !defined(MIPS32_R2_LE)
>-    "mul.s     %[f9],       %[f15],       %[f7]                 \n\t"
>-    "mul.s     %[f15],      %[f15],       %[f5]                 \n\t"
>-    "sub.s     %[f4],       %[f6],        %[f4]                 \n\t"
>-    "swc1      %[f1],       0(%[a1])                            \n\t"
>-    "swc1      %[f2],       0(%[a2])                            \n\t"
>-    "add.s     %[f8],       %[f8],        %[f9]                 \n\t"
>-    "sub.s     %[f10],      %[f10],       %[f15]                \n\t"
>+      "mul.s     %[f9],       %[f15],       %[f7]                 \n\t"
>+      "mul.s     %[f15],      %[f15],       %[f5]                 \n\t"
>+      "sub.s     %[f4],       %[f6],        %[f4]                 \n\t"
>+      "swc1      %[f1],       0(%[a1])                            \n\t"
>+      "swc1      %[f2],       0(%[a2])                            \n\t"
>+      "add.s     %[f8],       %[f8],        %[f9]                 \n\t"
>+      "sub.s     %[f10],      %[f10],       %[f15]                \n\t"
> #else
>-    "swc1      %[f1],       0(%[a1])                            \n\t"
>-    "swc1      %[f2],       0(%[a2])                            \n\t"
>-    "sub.s     %[f4],       %[f6],        %[f4]                 \n\t"
>-    "madd.s    %[f8],       %[f8],        %[f15],     %[f7]     \n\t"
>-    "nmsub.s   %[f10],      %[f10],       %[f15],     %[f5]     \n\t"
>+      "swc1      %[f1],       0(%[a1])                            \n\t"
>+      "swc1      %[f2],       0(%[a2])                            \n\t"
>+      "sub.s     %[f4],       %[f6],        %[f4]                 \n\t"
>+      "madd.s    %[f8],       %[f8],        %[f15],     %[f7]     \n\t"
>+      "nmsub.s   %[f10],      %[f10],       %[f15],     %[f5]     \n\t"
> #endif
>-    "swc1      %[f3],       4(%[a1])                            \n\t"
>-    "swc1      %[f4],       4(%[a2])                            \n\t"
>-    "sub.s     %[f11],      %[f11],       %[f8]                 \n\t"
>-    "add.s     %[f12],      %[f12],       %[f8]                 \n\t"
>-    "sub.s     %[f13],      %[f10],       %[f13]                \n\t"
>-    "sub.s     %[f14],      %[f10],       %[f14]                \n\t"
>-    "addiu     %[c2],       %[c2],        -8                    \n\t"
>-    "addiu     %[c1],       %[c1],        8                     \n\t"
>-    "swc1      %[f11],      8(%[a1])                            \n\t"
>-    "swc1      %[f12],      -8(%[a2])                           \n\t"
>-    "swc1      %[f13],      12(%[a1])                           \n\t"
>-    "swc1      %[f14],      -4(%[a2])                           \n\t"
>-    "addiu     %[a1],       %[a1],        16                    \n\t"
>-    "addiu     %[count],    %[count],     -1                    \n\t"
>-    "bgtz      %[count],    1b                                  \n\t"
>-    " addiu    %[a2],       %[a2],        -16                   \n\t"
>-    ".set      pop                                              \n\t"
>-    : [a1] "+r" (a1), [a2] "+r" (a2), [c1] "+r" (c1), [c2] "+r" (c2),
>-      [f1] "=&f" (f1), [f2] "=&f" (f2), [f3] "=&f" (f3), [f4] "=&f" (f4),
>-      [f5] "=&f" (f5), [f6] "=&f" (f6), [f7] "=&f" (f7), [f8] "=&f" (f8),
>-      [f9] "=&f" (f9), [f10] "=&f" (f10), [f11] "=&f" (f11), [f12] "=&f" (f12),
>-      [f13] "=&f" (f13), [f14] "=&f" (f14), [f15] "=&f" (f15),
>-      [count] "=&r" (count)
>-    : [f0] "f" (f0)
>-    : "memory"
>-  );
>+      "swc1      %[f3],       4(%[a1])                            \n\t"
>+      "swc1      %[f4],       4(%[a2])                            \n\t"
>+      "sub.s     %[f11],      %[f11],       %[f8]                 \n\t"
>+      "add.s     %[f12],      %[f12],       %[f8]                 \n\t"
>+      "sub.s     %[f13],      %[f10],       %[f13]                \n\t"
>+      "sub.s     %[f14],      %[f10],       %[f14]                \n\t"
>+      "addiu     %[c2],       %[c2],        -8                    \n\t"
>+      "addiu     %[c1],       %[c1],        8                     \n\t"
>+      "swc1      %[f11],      8(%[a1])                            \n\t"
>+      "swc1      %[f12],      -8(%[a2])                           \n\t"
>+      "swc1      %[f13],      12(%[a1])                           \n\t"
>+      "swc1      %[f14],      -4(%[a2])                           \n\t"
>+      "addiu     %[a1],       %[a1],        16                    \n\t"
>+      "addiu     %[count],    %[count],     -1                    \n\t"
>+      "bgtz      %[count],    1b                                  \n\t"
>+      " addiu    %[a2],       %[a2],        -16                   \n\t"
>+      ".set      pop                                              \n\t"
>+      : [a1] "+r"(a1), [a2] "+r"(a2), [c1] "+r"(c1), [c2] "+r"(c2),
>+        [f1] "=&f"(f1), [f2] "=&f"(f2), [f3] "=&f"(f3), [f4] "=&f"(f4),
>+        [f5] "=&f"(f5), [f6] "=&f"(f6), [f7] "=&f"(f7), [f8] "=&f"(f8),
>+        [f9] "=&f"(f9), [f10] "=&f"(f10), [f11] "=&f"(f11), [f12] "=&f"(f12),
>+        [f13] "=&f"(f13), [f14] "=&f"(f14), [f15] "=&f"(f15),
>+        [count] "=&r"(count)
>+      : [f0] "f"(f0)
>+      : "memory");
> }
> #endif
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/ooura_fft_sse2.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/ooura_fft_sse2.cc
>index b44458eea70..9b5d0f37a44 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/ooura_fft_sse2.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/ooura_fft_sse2.cc
>@@ -14,6 +14,7 @@
> 
> #include "modules/audio_processing/utility/ooura_fft_tables_common.h"
> #include "modules/audio_processing/utility/ooura_fft_tables_neon_sse2.h"
>+#include "rtc_base/system/arch.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/ooura_fft_tables_neon_sse2.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/ooura_fft_tables_neon_sse2.h
>index 1c44ae71973..b6e4a07fce3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/ooura_fft_tables_neon_sse2.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/utility/ooura_fft_tables_neon_sse2.h
>@@ -12,6 +12,7 @@
> #define MODULES_AUDIO_PROCESSING_UTILITY_OOURA_FFT_TABLES_NEON_SSE2_H_
> 
> #include "modules/audio_processing/utility/ooura_fft.h"
>+#include "rtc_base/system/arch.h"
> 
> #ifdef _MSC_VER /* visual c++ */
> #define ALIGN16_BEG __declspec(align(16))
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/BUILD.gn
>new file mode 100644
>index 00000000000..6c64e947e53
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/BUILD.gn
>@@ -0,0 +1,69 @@
>+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+#
>+# Use of this source code is governed by a BSD-style license
>+# that can be found in the LICENSE file in the root of the source
>+# tree. An additional intellectual property rights grant can be found
>+# in the file PATENTS.  All contributing project authors may
>+# be found in the AUTHORS file in the root of the source tree.
>+
>+import("../../../webrtc.gni")
>+rtc_static_library("vad") {
>+  visibility = [
>+    "../*",
>+    "../../../rtc_tools:*",
>+  ]
>+  sources = [
>+    "common.h",
>+    "gmm.cc",
>+    "gmm.h",
>+    "noise_gmm_tables.h",
>+    "pitch_based_vad.cc",
>+    "pitch_based_vad.h",
>+    "pitch_internal.cc",
>+    "pitch_internal.h",
>+    "pole_zero_filter.cc",
>+    "pole_zero_filter.h",
>+    "standalone_vad.cc",
>+    "standalone_vad.h",
>+    "vad_audio_proc.cc",
>+    "vad_audio_proc.h",
>+    "vad_audio_proc_internal.h",
>+    "vad_circular_buffer.cc",
>+    "vad_circular_buffer.h",
>+    "voice_activity_detector.cc",
>+    "voice_activity_detector.h",
>+    "voice_gmm_tables.h",
>+  ]
>+  deps = [
>+    "../../../audio/utility:audio_frame_operations",
>+    "../../../common_audio",
>+    "../../../common_audio:common_audio_c",
>+    "../../../common_audio/third_party/fft4g:fft4g",
>+    "../../../rtc_base:checks",
>+    "../../audio_coding:isac_vad",
>+  ]
>+}
>+
>+if (rtc_include_tests) {
>+  rtc_static_library("vad_unittests") {
>+    testonly = true
>+    sources = [
>+      "gmm_unittest.cc",
>+      "pitch_based_vad_unittest.cc",
>+      "pitch_internal_unittest.cc",
>+      "pole_zero_filter_unittest.cc",
>+      "standalone_vad_unittest.cc",
>+      "vad_audio_proc_unittest.cc",
>+      "vad_circular_buffer_unittest.cc",
>+      "voice_activity_detector_unittest.cc",
>+    ]
>+    deps = [
>+      ":vad",
>+      "../../../common_audio",
>+      "../../../test:fileutils",
>+      "../../../test:test_support",
>+      "//testing/gmock",
>+      "//testing/gtest",
>+    ]
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/gmm.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/gmm.cc
>index 266ca44cc99..cd8a1a8a375 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/gmm.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/gmm.cc
>@@ -13,8 +13,6 @@
> #include <math.h>
> #include <stdlib.h>
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>-
> namespace webrtc {
> 
> static const int kMaxDimension = 10;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/gmm_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/gmm_unittest.cc
>index dfc8855a143..d895afab7be 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/gmm_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/gmm_unittest.cc
>@@ -37,10 +37,10 @@ TEST(GmmTest, EvaluateGmm) {
>   voice_gmm.covar_inverse = &kVoiceGmmCovarInverse[0][0][0];
> 
>   // Test vectors. These are the mean of the GMM means.
>-  const double kXVoice[kVoiceGmmDim] = {
>-      -1.35893162459863, 602.862491970368, 178.022069191324};
>-  const double kXNoise[kNoiseGmmDim] = {
>-      -2.33443722724409, 2827.97828765184, 141.114178166812};
>+  const double kXVoice[kVoiceGmmDim] = {-1.35893162459863, 602.862491970368,
>+                                        178.022069191324};
>+  const double kXNoise[kNoiseGmmDim] = {-2.33443722724409, 2827.97828765184,
>+                                        141.114178166812};
> 
>   // Expected pdf values. These values are computed in MATLAB using EvalGmm.m
>   const double kPdfNoise = 1.88904409403101e-07;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/noise_gmm_tables.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/noise_gmm_tables.h
>index c07dadebd52..15562776e93 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/noise_gmm_tables.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/noise_gmm_tables.h
>@@ -70,16 +70,8 @@ static const double kNoiseGmmMean[kNoiseGmmNumMixtures][kNoiseGmmDim] = {
>     {-2.30193040814533e+00, 1.43953696546439e+03, 7.04085275122649e+01}};
> 
> static const double kNoiseGmmWeights[kNoiseGmmNumMixtures] = {
>-    -1.09422832086193e+01,
>-    -1.10847897513425e+01,
>-    -1.36767587732187e+01,
>-    -1.79789356118641e+01,
>-    -1.42830169160894e+01,
>-    -1.56500228061379e+01,
>-    -1.83124990950113e+01,
>-    -1.69979436177477e+01,
>-    -1.12329424387828e+01,
>-    -1.41311785780639e+01,
>-    -1.47171861448585e+01,
>-    -1.35963362781839e+01};
>+    -1.09422832086193e+01, -1.10847897513425e+01, -1.36767587732187e+01,
>+    -1.79789356118641e+01, -1.42830169160894e+01, -1.56500228061379e+01,
>+    -1.83124990950113e+01, -1.69979436177477e+01, -1.12329424387828e+01,
>+    -1.41311785780639e+01, -1.47171861448585e+01, -1.35963362781839e+01};
> #endif  // MODULES_AUDIO_PROCESSING_VAD_NOISE_GMM_TABLES_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/pitch_based_vad.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/pitch_based_vad.cc
>index bca2552c356..025ef2059d9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/pitch_based_vad.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/pitch_based_vad.cc
>@@ -13,11 +13,10 @@
> #include <math.h>
> #include <string.h>
> 
>-#include "modules/audio_processing/vad/vad_circular_buffer.h"
> #include "modules/audio_processing/vad/common.h"
> #include "modules/audio_processing/vad/noise_gmm_tables.h"
>+#include "modules/audio_processing/vad/vad_circular_buffer.h"
> #include "modules/audio_processing/vad/voice_gmm_tables.h"
>-#include "modules/include/module_common_types.h"
> 
> namespace webrtc {
> 
>@@ -59,8 +58,7 @@ PitchBasedVad::PitchBasedVad()
>   voice_gmm_.covar_inverse = &kVoiceGmmCovarInverse[0][0][0];
> }
> 
>-PitchBasedVad::~PitchBasedVad() {
>-}
>+PitchBasedVad::~PitchBasedVad() {}
> 
> int PitchBasedVad::VoicingProbability(const AudioFeatures& features,
>                                       double* p_combined) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/pitch_based_vad.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/pitch_based_vad.h
>index 584dcc73ce7..4d327652c68 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/pitch_based_vad.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/pitch_based_vad.h
>@@ -15,7 +15,6 @@
> 
> #include "modules/audio_processing/vad/common.h"
> #include "modules/audio_processing/vad/gmm.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/pitch_internal_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/pitch_internal_unittest.cc
>index c1fde100088..19c2e1a2711 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/pitch_internal_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/pitch_internal_unittest.cc
>@@ -26,12 +26,12 @@ TEST(PitchInternalTest, test) {
>   double lags[] = {90, 111, 122, 50};
> 
>   // Expected outputs
>-  double expected_log_pitch_gain[] = {
>-      -0.541212549898316, -1.45672279045507, -0.80471895621705};
>+  double expected_log_pitch_gain[] = {-0.541212549898316, -1.45672279045507,
>+                                      -0.80471895621705};
>   double expected_log_old_gain = log(gains[kNumInputParameters - 1]);
> 
>-  double expected_pitch_lag_hz[] = {
>-      92.3076923076923, 70.9010339734121, 93.0232558139535};
>+  double expected_pitch_lag_hz[] = {92.3076923076923, 70.9010339734121,
>+                                    93.0232558139535};
>   double expected_old_lag = lags[kNumInputParameters - 1];
> 
>   double log_pitch_gain[kNumOutputParameters];
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/pole_zero_filter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/pole_zero_filter.cc
>index fa56a3c15e9..b9967d7aea5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/pole_zero_filter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/pole_zero_filter.cc
>@@ -53,7 +53,8 @@ PoleZeroFilter::PoleZeroFilter(const float* numerator_coefficients,
> }
> 
> template <typename T>
>-static float FilterArPast(const T* past, size_t order,
>+static float FilterArPast(const T* past,
>+                          size_t order,
>                           const float* coefficients) {
>   float sum = 0.0f;
>   size_t past_index = order - 1;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/pole_zero_filter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/pole_zero_filter.h
>index 283deecde30..11a05114d1d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/pole_zero_filter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/pole_zero_filter.h
>@@ -11,9 +11,8 @@
> #ifndef MODULES_AUDIO_PROCESSING_VAD_POLE_ZERO_FILTER_H_
> #define MODULES_AUDIO_PROCESSING_VAD_POLE_ZERO_FILTER_H_
> 
>-#include <cstddef>
>-
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include <stddef.h>
>+#include <stdint.h>
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/pole_zero_filter_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/pole_zero_filter_unittest.cc
>index aef07a6e448..38ee10fc468 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/pole_zero_filter_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/pole_zero_filter_unittest.cc
>@@ -53,7 +53,7 @@ class PoleZeroFilterTest : public ::testing::Test {
>                                           kCoeffDenominator,
>                                           kFilterOrder)) {}
> 
>-  ~PoleZeroFilterTest() {}
>+  ~PoleZeroFilterTest() override {}
> 
>   void FilterSubframes(int num_subframes);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/standalone_vad.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/standalone_vad.cc
>index 004cefebb8c..19a5282c3f7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/standalone_vad.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/standalone_vad.cc
>@@ -10,18 +10,17 @@
> 
> #include "modules/audio_processing/vad/standalone_vad.h"
> 
>+#include <string.h>
>+
> #include "audio/utility/audio_frame_operations.h"
>-#include "modules/include/module_common_types.h"
> #include "rtc_base/checks.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
> static const int kDefaultStandaloneVadMode = 3;
> 
> StandaloneVad::StandaloneVad(VadInst* vad)
>-    : vad_(vad), buffer_(), index_(0), mode_(kDefaultStandaloneVadMode) {
>-}
>+    : vad_(vad), buffer_(), index_(0), mode_(kDefaultStandaloneVadMode) {}
> 
> StandaloneVad::~StandaloneVad() {
>   WebRtcVad_Free(vad_);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/standalone_vad.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/standalone_vad.h
>index b85de0a75ab..79650fbe8c6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/standalone_vad.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/standalone_vad.h
>@@ -11,9 +11,8 @@
> #ifndef MODULES_AUDIO_PROCESSING_AGC_STANDALONE_VAD_H_
> #define MODULES_AUDIO_PROCESSING_AGC_STANDALONE_VAD_H_
> 
>-#include "modules/audio_processing/vad/common.h"
> #include "common_audio/vad/include/webrtc_vad.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include "modules/audio_processing/vad/common.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/standalone_vad_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/standalone_vad_unittest.cc
>index 28d1349396e..512276ecb5a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/standalone_vad_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/standalone_vad_unittest.cc
>@@ -14,7 +14,6 @@
> 
> #include <memory>
> 
>-#include "modules/include/module_common_types.h"
> #include "test/gtest.h"
> #include "test/testsupport/fileutils.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/vad_audio_proc.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/vad_audio_proc.cc
>index b1841d0b213..53eb6de70f3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/vad_audio_proc.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/vad_audio_proc.cc
>@@ -12,19 +12,19 @@
> 
> #include <math.h>
> #include <stdio.h>
>+#include <string.h>
> 
>-#include "common_audio/fft4g.h"
>+#include "common_audio/third_party/fft4g/fft4g.h"
> #include "modules/audio_processing/vad/pitch_internal.h"
> #include "modules/audio_processing/vad/pole_zero_filter.h"
> #include "modules/audio_processing/vad/vad_audio_proc_internal.h"
> #include "rtc_base/checks.h"
> extern "C" {
>-#include "modules/audio_coding/codecs/isac/main/source/codec.h"
>-#include "modules/audio_coding/codecs/isac/main/source/lpc_analysis.h"
>+#include "modules/audio_coding/codecs/isac/main/source/filter_functions.h"
>+#include "modules/audio_coding/codecs/isac/main/source/isac_vad.h"
> #include "modules/audio_coding/codecs/isac/main/source/pitch_estimator.h"
> #include "modules/audio_coding/codecs/isac/main/source/structs.h"
> }
>-#include "modules/include/module_common_types.h"
> 
> namespace webrtc {
> 
>@@ -67,8 +67,7 @@ VadAudioProc::VadAudioProc()
>   WebRtcIsac_InitPitchAnalysis(pitch_analysis_handle_.get());
> }
> 
>-VadAudioProc::~VadAudioProc() {
>-}
>+VadAudioProc::~VadAudioProc() {}
> 
> void VadAudioProc::ResetBuffer() {
>   memcpy(audio_buffer_, &audio_buffer_[kNumSamplesToProcess],
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/vad_audio_proc.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/vad_audio_proc.h
>index b1441a08c0c..e34091bdb6a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/vad_audio_proc.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/vad_audio_proc.h
>@@ -14,7 +14,6 @@
> #include <memory>
> 
> #include "modules/audio_processing/vad/common.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/vad_audio_proc_internal.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/vad_audio_proc_internal.h
>index ab1e63651f1..915524f4748 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/vad_audio_proc_internal.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/vad_audio_proc_internal.h
>@@ -14,23 +14,10 @@
> namespace webrtc {
> 
> // These values should match MATLAB counterparts for unit-tests to pass.
>-static const double kCorrWeight[] = {1.000000,
>-                                     0.985000,
>-                                     0.970225,
>-                                     0.955672,
>-                                     0.941337,
>-                                     0.927217,
>-                                     0.913308,
>-                                     0.899609,
>-                                     0.886115,
>-                                     0.872823,
>-                                     0.859730,
>-                                     0.846834,
>-                                     0.834132,
>-                                     0.821620,
>-                                     0.809296,
>-                                     0.797156,
>-                                     0.785199};
>+static const double kCorrWeight[] = {
>+    1.000000, 0.985000, 0.970225, 0.955672, 0.941337, 0.927217,
>+    0.913308, 0.899609, 0.886115, 0.872823, 0.859730, 0.846834,
>+    0.834132, 0.821620, 0.809296, 0.797156, 0.785199};
> 
> static const double kLpcAnalWin[] = {
>     0.00000000, 0.01314436, 0.02628645, 0.03942400, 0.05255473, 0.06567639,
>@@ -75,11 +62,9 @@ static const double kLpcAnalWin[] = {
>     0.06567639, 0.05255473, 0.03942400, 0.02628645, 0.01314436, 0.00000000};
> 
> static const size_t kFilterOrder = 2;
>-static const float kCoeffNumerator[kFilterOrder + 1] = {0.974827f,
>-                                                        -1.949650f,
>+static const float kCoeffNumerator[kFilterOrder + 1] = {0.974827f, -1.949650f,
>                                                         0.974827f};
>-static const float kCoeffDenominator[kFilterOrder + 1] = {1.0f,
>-                                                          -1.971999f,
>+static const float kCoeffDenominator[kFilterOrder + 1] = {1.0f, -1.971999f,
>                                                           0.972457f};
> 
> static_assert(kFilterOrder + 1 ==
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/vad_audio_proc_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/vad_audio_proc_unittest.cc
>index c520257f8e2..5b96be624db 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/vad_audio_proc_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/vad_audio_proc_unittest.cc
>@@ -20,7 +20,6 @@
> #include <string>
> 
> #include "modules/audio_processing/vad/common.h"
>-#include "modules/include/module_common_types.h"
> #include "test/gtest.h"
> #include "test/testsupport/fileutils.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/vad_circular_buffer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/vad_circular_buffer.cc
>index 3c4d5ad0e07..31f14d7f64f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/vad_circular_buffer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/vad_circular_buffer.cc
>@@ -19,11 +19,9 @@ VadCircularBuffer::VadCircularBuffer(int buffer_size)
>       is_full_(false),
>       index_(0),
>       buffer_size_(buffer_size),
>-      sum_(0) {
>-}
>+      sum_(0) {}
> 
>-VadCircularBuffer::~VadCircularBuffer() {
>-}
>+VadCircularBuffer::~VadCircularBuffer() {}
> 
> void VadCircularBuffer::Reset() {
>   is_full_ = false;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/voice_activity_detector.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/voice_activity_detector.cc
>index 66a704f6bed..f0d34c668e2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/voice_activity_detector.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/voice_activity_detector.cc
>@@ -27,8 +27,7 @@ const double kLowProbability = 0.01;
> 
> VoiceActivityDetector::VoiceActivityDetector()
>     : last_voice_probability_(kDefaultVoiceValue),
>-      standalone_vad_(StandaloneVad::Create()) {
>-}
>+      standalone_vad_(StandaloneVad::Create()) {}
> 
> VoiceActivityDetector::~VoiceActivityDetector() = default;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/voice_activity_detector.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/voice_activity_detector.h
>index 0079cb21840..e424ac1a282 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/voice_activity_detector.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/voice_activity_detector.h
>@@ -15,10 +15,10 @@
> #include <vector>
> 
> #include "common_audio/resampler/include/resampler.h"
>-#include "modules/audio_processing/vad/vad_audio_proc.h"
> #include "modules/audio_processing/vad/common.h"
> #include "modules/audio_processing/vad/pitch_based_vad.h"
> #include "modules/audio_processing/vad/standalone_vad.h"
>+#include "modules/audio_processing/vad/vad_audio_proc.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/voice_gmm_tables.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/voice_gmm_tables.h
>index 29cc7d6cb23..ef4ad7e21ee 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/voice_gmm_tables.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/vad/voice_gmm_tables.h
>@@ -70,16 +70,8 @@ static const double kVoiceGmmMean[kVoiceGmmNumMixtures][kVoiceGmmDim] = {
>     {-7.29187507662854e-01, 5.22717685022855e+02, 1.16377942283991e+02}};
> 
> static const double kVoiceGmmWeights[kVoiceGmmNumMixtures] = {
>-    -1.39789694361035e+01,
>-    -1.19527720202104e+01,
>-    -1.32396317929055e+01,
>-    -1.09436815209238e+01,
>-    -1.13440027478149e+01,
>-    -1.12200721834504e+01,
>-    -1.02537324043693e+01,
>-    -1.60789861938302e+01,
>-    -1.03394494048344e+01,
>-    -1.83207938586818e+01,
>-    -1.31186044948288e+01,
>-    -9.52479998673554e+00};
>+    -1.39789694361035e+01, -1.19527720202104e+01, -1.32396317929055e+01,
>+    -1.09436815209238e+01, -1.13440027478149e+01, -1.12200721834504e+01,
>+    -1.02537324043693e+01, -1.60789861938302e+01, -1.03394494048344e+01,
>+    -1.83207938586818e+01, -1.31186044948288e+01, -9.52479998673554e+00};
> #endif  // MODULES_AUDIO_PROCESSING_VAD_VOICE_GMM_TABLES_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/voice_detection_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/voice_detection_impl.cc
>index 5ee0c7b9c24..9280be1ef9d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/voice_detection_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/voice_detection_impl.cc
>@@ -23,10 +23,9 @@ class VoiceDetectionImpl::Vad {
>     int error = WebRtcVad_Init(state_);
>     RTC_DCHECK_EQ(0, error);
>   }
>-  ~Vad() {
>-    WebRtcVad_Free(state_);
>-  }
>+  ~Vad() { WebRtcVad_Free(state_); }
>   VadInst* state() { return state_; }
>+
>  private:
>   VadInst* state_ = nullptr;
>   RTC_DISALLOW_COPY_AND_ASSIGN(Vad);
>@@ -65,9 +64,9 @@ void VoiceDetectionImpl::ProcessCaptureAudio(AudioBuffer* audio) {
> 
>   RTC_DCHECK_GE(160, audio->num_frames_per_band());
>   // TODO(ajm): concatenate data in frame buffer here.
>-  int vad_ret = WebRtcVad_Process(vad_->state(), sample_rate_hz_,
>-                                  audio->mixed_low_pass_data(),
>-                                  frame_size_samples_);
>+  int vad_ret =
>+      WebRtcVad_Process(vad_->state(), sample_rate_hz_,
>+                        audio->mixed_low_pass_data(), frame_size_samples_);
>   if (vad_ret == 0) {
>     stream_has_voice_ = false;
>     audio->set_activity(AudioFrame::kVadPassive);
>@@ -103,7 +102,7 @@ int VoiceDetectionImpl::set_stream_has_voice(bool has_voice) {
> bool VoiceDetectionImpl::stream_has_voice() const {
>   rtc::CritScope cs(crit_);
>   // TODO(ajm): enable this assertion?
>-  //RTC_DCHECK(using_external_vad_ || is_component_enabled());
>+  // RTC_DCHECK(using_external_vad_ || is_component_enabled());
>   return stream_has_voice_;
> }
> 
>@@ -142,7 +141,7 @@ VoiceDetection::Likelihood VoiceDetectionImpl::likelihood() const {
> 
> int VoiceDetectionImpl::set_frame_size_ms(int size) {
>   rtc::CritScope cs(crit_);
>-  RTC_DCHECK_EQ(10, size); // TODO(ajm): remove when supported.
>+  RTC_DCHECK_EQ(10, size);  // TODO(ajm): remove when supported.
>   frame_size_ms_ = size;
>   Initialize(sample_rate_hz_);
>   return AudioProcessing::kNoError;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/BUILD.gn
>index 165db62cce9..f9960cf00b4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/BUILD.gn
>@@ -11,6 +11,7 @@ import("../../webrtc.gni")
> rtc_static_library("bitrate_controller") {
>   visibility = [ "*" ]
>   sources = [
>+    "bitrate_controller.cc",
>     "bitrate_controller_impl.cc",
>     "bitrate_controller_impl.h",
>     "include/bitrate_controller.h",
>@@ -23,10 +24,6 @@ rtc_static_library("bitrate_controller") {
>   } else {
>     defines = [ "BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=0" ]
>   }
>-
>-  # TODO(jschuh): Bug 1348: fix this warning.
>-  configs += [ "//build/config/compiler:no_size_t_to_int_warning" ]
>-
>   if (!build_with_chromium && is_clang) {
>     # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>     suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>@@ -34,17 +31,19 @@ rtc_static_library("bitrate_controller") {
> 
>   deps = [
>     "..:module_api",
>+    "../../logging:rtc_event_bwe",
>     "../../logging:rtc_event_log_api",
>     "../../rtc_base:checks",
>     "../../rtc_base:rtc_base_approved",
>     "../../system_wrappers",
>     "../../system_wrappers:field_trial_api",
>     "../../system_wrappers:metrics_api",
>-    "../congestion_controller:delay_based_bwe",
>+    "../congestion_controller/goog_cc:delay_based_bwe",
>     "../pacing",
>     "../remote_bitrate_estimator:remote_bitrate_estimator",
>     "../rtp_rtcp",
>     "../rtp_rtcp:rtp_rtcp_format",
>+    "//third_party/abseil-cpp/absl/memory",
>   ]
> }
> 
>@@ -70,12 +69,12 @@ if (rtc_include_tests) {
>     deps = [
>       ":bitrate_controller",
>       "../../logging:mocks",
>+      "../../logging:rtc_event_bwe",
>       "../../logging:rtc_event_log_api",
>       "../../test:field_trial",
>       "../../test:test_support",
>       "../pacing:mock_paced_sender",
>       "../remote_bitrate_estimator:remote_bitrate_estimator",
>-      "//testing/gmock",
>     ]
>     if (!build_with_chromium && is_clang) {
>       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/bitrate_controller.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/bitrate_controller.cc
>new file mode 100644
>index 00000000000..46e908be277
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/bitrate_controller.cc
>@@ -0,0 +1,23 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ *
>+ *  Usage: this class will register multiple RtcpBitrateObserver's one at each
>+ *  RTCP module. It will aggregate the results and run one bandwidth estimation
>+ *  and push the result to the encoders via BitrateObserver(s).
>+ */
>+
>+#include "modules/bitrate_controller/include/bitrate_controller.h"
>+
>+namespace webrtc {
>+
>+size_t BitrateObserver::pacer_queue_size_in_bytes() {
>+  return 0;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/bitrate_controller_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/bitrate_controller_impl.cc
>index 5b9b9ad4627..38f97fb7e24 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/bitrate_controller_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/bitrate_controller_impl.cc
>@@ -25,8 +25,7 @@ class BitrateControllerImpl::RtcpBandwidthObserverImpl
>     : public RtcpBandwidthObserver {
>  public:
>   explicit RtcpBandwidthObserverImpl(BitrateControllerImpl* owner)
>-      : owner_(owner) {
>-  }
>+      : owner_(owner) {}
>   ~RtcpBandwidthObserverImpl() override = default;
>   // Received RTCP REMB or TMMBR.
>   void OnReceivedEstimatedBitrate(uint32_t bitrate) override {
>@@ -102,8 +101,7 @@ void BitrateControllerImpl::SetBitrates(int start_bitrate_bps,
>                                         int max_bitrate_bps) {
>   {
>     rtc::CritScope cs(&critsect_);
>-    bandwidth_estimation_.SetBitrates(start_bitrate_bps,
>-                                      min_bitrate_bps,
>+    bandwidth_estimation_.SetBitrates(start_bitrate_bps, min_bitrate_bps,
>                                       max_bitrate_bps);
>   }
>   MaybeTriggerOnNetworkChanged();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/bitrate_controller_impl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/bitrate_controller_impl.h
>index 6a678082e92..e315c5f9fd2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/bitrate_controller_impl.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/bitrate_controller_impl.h
>@@ -54,7 +54,6 @@ class BitrateControllerImpl : public BitrateController {
>                      int min_bitrate_bps,
>                      int max_bitrate_bps) override;
> 
>-
>   void SetReservedBitrate(uint32_t reserved_bitrate_bps) override;
> 
>   // Returns true if the parameters have changed since the last call.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc
>index 66ca5b9558d..ce930d10d46 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/bitrate_controller_unittest.cc
>@@ -26,28 +26,29 @@ using webrtc::BitrateObserver;
> using webrtc::PacedSender;
> using webrtc::RtcpBandwidthObserver;
> 
>-uint8_t WeightedLoss(int num_packets1, uint8_t fraction_loss1,
>-                     int num_packets2, uint8_t fraction_loss2) {
>-  int weighted_sum = num_packets1 * fraction_loss1 +
>-      num_packets2 * fraction_loss2;
>+uint8_t WeightedLoss(int num_packets1,
>+                     uint8_t fraction_loss1,
>+                     int num_packets2,
>+                     uint8_t fraction_loss2) {
>+  int weighted_sum =
>+      num_packets1 * fraction_loss1 + num_packets2 * fraction_loss2;
>   int total_num_packets = num_packets1 + num_packets2;
>   return (weighted_sum + total_num_packets / 2) / total_num_packets;
> }
> 
> webrtc::RTCPReportBlock CreateReportBlock(
>-    uint32_t remote_ssrc, uint32_t source_ssrc,
>-    uint8_t fraction_lost, uint32_t extended_high_sequence_number) {
>+    uint32_t remote_ssrc,
>+    uint32_t source_ssrc,
>+    uint8_t fraction_lost,
>+    uint32_t extended_high_sequence_number) {
>   return webrtc::RTCPReportBlock(remote_ssrc, source_ssrc, fraction_lost, 0,
>                                  extended_high_sequence_number, 0, 0, 0);
> }
> 
>-class TestBitrateObserver: public BitrateObserver {
>+class TestBitrateObserver : public BitrateObserver {
>  public:
>   TestBitrateObserver()
>-      : last_bitrate_(0),
>-        last_fraction_loss_(0),
>-        last_rtt_(0) {
>-  }
>+      : last_bitrate_(0), last_fraction_loss_(0), last_rtt_(0) {}
> 
>   virtual void OnNetworkChanged(uint32_t bitrate,
>                                 uint8_t fraction_loss,
>@@ -76,8 +77,7 @@ class BitrateControllerTest : public ::testing::Test {
>     bandwidth_observer_ = controller_.get();
>   }
> 
>-  virtual void TearDown() {
>-  }
>+  virtual void TearDown() {}
> 
>   const int kMinBitrateBps = 100000;
>   const int kStartBitrateBps = 200000;
>@@ -197,8 +197,8 @@ TEST_F(BitrateControllerTest, OneBitrateObserverTwoRtcpObservers) {
> 
>   RtcpBandwidthObserver* second_bandwidth_observer = controller_.get();
>   report_blocks = {CreateReportBlock(kSenderSsrc2, kMediaSsrc2, 0, 21)};
>-  second_bandwidth_observer->OnReceivedRtcpReceiverReport(
>-      report_blocks, 100, time_ms);
>+  second_bandwidth_observer->OnReceivedRtcpReceiverReport(report_blocks, 100,
>+                                                          time_ms);
> 
>   // Test start bitrate.
>   EXPECT_EQ(200000, bitrate_observer_.last_bitrate_);
>@@ -212,8 +212,8 @@ TEST_F(BitrateControllerTest, OneBitrateObserverTwoRtcpObservers) {
>   time_ms += 500;
> 
>   report_blocks = {CreateReportBlock(kSenderSsrc2, kMediaSsrc2, 0, 21)};
>-  second_bandwidth_observer->OnReceivedRtcpReceiverReport(
>-      report_blocks, 100, time_ms);
>+  second_bandwidth_observer->OnReceivedRtcpReceiverReport(report_blocks, 100,
>+                                                          time_ms);
>   EXPECT_EQ(217000, bitrate_observer_.last_bitrate_);
>   EXPECT_EQ(0, bitrate_observer_.last_fraction_loss_);
>   EXPECT_EQ(100, bitrate_observer_.last_rtt_);
>@@ -221,8 +221,8 @@ TEST_F(BitrateControllerTest, OneBitrateObserverTwoRtcpObservers) {
> 
>   // Extra report should not change estimate.
>   report_blocks = {CreateReportBlock(kSenderSsrc2, kMediaSsrc2, 0, 31)};
>-  second_bandwidth_observer->OnReceivedRtcpReceiverReport(
>-      report_blocks, 100, time_ms);
>+  second_bandwidth_observer->OnReceivedRtcpReceiverReport(report_blocks, 100,
>+                                                          time_ms);
>   EXPECT_EQ(217000, bitrate_observer_.last_bitrate_);
>   time_ms += 500;
> 
>@@ -232,34 +232,34 @@ TEST_F(BitrateControllerTest, OneBitrateObserverTwoRtcpObservers) {
> 
>   // Second report should not change estimate.
>   report_blocks = {CreateReportBlock(kSenderSsrc2, kMediaSsrc2, 0, 41)};
>-  second_bandwidth_observer->OnReceivedRtcpReceiverReport(
>-      report_blocks, 100, time_ms);
>+  second_bandwidth_observer->OnReceivedRtcpReceiverReport(report_blocks, 100,
>+                                                          time_ms);
>   EXPECT_EQ(235360, bitrate_observer_.last_bitrate_);
>   time_ms += 1000;
> 
>   // Reports from only one bandwidth observer is ok.
>   report_blocks = {CreateReportBlock(kSenderSsrc2, kMediaSsrc2, 0, 61)};
>-  second_bandwidth_observer->OnReceivedRtcpReceiverReport(
>-      report_blocks, 50, time_ms);
>+  second_bandwidth_observer->OnReceivedRtcpReceiverReport(report_blocks, 50,
>+                                                          time_ms);
>   EXPECT_EQ(255189, bitrate_observer_.last_bitrate_);
>   time_ms += 1000;
> 
>   report_blocks = {CreateReportBlock(kSenderSsrc2, kMediaSsrc2, 0, 81)};
>-  second_bandwidth_observer->OnReceivedRtcpReceiverReport(
>-      report_blocks, 50, time_ms);
>+  second_bandwidth_observer->OnReceivedRtcpReceiverReport(report_blocks, 50,
>+                                                          time_ms);
>   EXPECT_EQ(276604, bitrate_observer_.last_bitrate_);
>   time_ms += 1000;
> 
>   report_blocks = {CreateReportBlock(kSenderSsrc2, kMediaSsrc2, 0, 121)};
>-  second_bandwidth_observer->OnReceivedRtcpReceiverReport(
>-      report_blocks, 50, time_ms);
>+  second_bandwidth_observer->OnReceivedRtcpReceiverReport(report_blocks, 50,
>+                                                          time_ms);
>   EXPECT_EQ(299732, bitrate_observer_.last_bitrate_);
>   time_ms += 1000;
> 
>   // Reach max cap.
>   report_blocks = {CreateReportBlock(kSenderSsrc2, kMediaSsrc2, 0, 141)};
>-  second_bandwidth_observer->OnReceivedRtcpReceiverReport(
>-      report_blocks, 50, time_ms);
>+  second_bandwidth_observer->OnReceivedRtcpReceiverReport(report_blocks, 50,
>+                                                          time_ms);
>   EXPECT_EQ(300000, bitrate_observer_.last_bitrate_);
> 
>   // Test that a low REMB trigger immediately.
>@@ -340,11 +340,11 @@ TEST_F(BitrateControllerTest, OneBitrateObserverMultipleReportBlocks) {
>   report_blocks.clear();
> 
>   // All packets lost on stream with few packets, no back-off.
>-  report_blocks.push_back(CreateReportBlock(1, 2, 1, sequence_number[0]));
>+  report_blocks.push_back(CreateReportBlock(1, 2, 0, sequence_number[0]));
>   report_blocks.push_back(CreateReportBlock(1, 3, 255, sequence_number[1]));
>   bandwidth_observer_->OnReceivedRtcpReceiverReport(report_blocks, 50, time_ms);
>   EXPECT_EQ(bitrate_observer_.last_bitrate_, last_bitrate);
>-  EXPECT_EQ(WeightedLoss(20, 1, 1, 255), bitrate_observer_.last_fraction_loss_);
>+  EXPECT_EQ(WeightedLoss(20, 0, 1, 255), bitrate_observer_.last_fraction_loss_);
>   EXPECT_EQ(50, bitrate_observer_.last_rtt_);
>   last_bitrate = bitrate_observer_.last_bitrate_;
>   sequence_number[0] += 20;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/include/bitrate_controller.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/include/bitrate_controller.h
>index 5aaf88b95dc..6e692d2c249 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/include/bitrate_controller.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/include/bitrate_controller.h
>@@ -17,7 +17,7 @@
> 
> #include <map>
> 
>-#include "modules/congestion_controller/delay_based_bwe.h"
>+#include "modules/congestion_controller/goog_cc/delay_based_bwe.h"
> #include "modules/include/module.h"
> #include "modules/pacing/paced_sender.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>@@ -45,7 +45,7 @@ class BitrateObserver {
>                                 int64_t target_set_time,
>                                 uint64_t congestion_window) {}
>   virtual void OnBytesAcked(size_t bytes) {}
>-  virtual size_t pacer_queue_size_in_bytes() { return 0; }
>+  virtual size_t pacer_queue_size_in_bytes();
>   virtual ~BitrateObserver() {}
> };
> 
>@@ -67,7 +67,7 @@ class BitrateController : public Module, public RtcpBandwidthObserver {
>   static BitrateController* CreateBitrateController(const Clock* clock,
>                                                     RtcEventLog* event_log);
> 
>-  virtual ~BitrateController() {}
>+  ~BitrateController() override {}
> 
>   // Deprecated, use raw pointer to BitrateController instance instead.
>   // Creates RtcpBandwidthObserver caller responsible to delete.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
>index 86096a1b736..48eea3cd73f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
>@@ -12,15 +12,16 @@
> 
> #include <algorithm>
> #include <cmath>
>+#include <cstdio>
> #include <limits>
> #include <string>
> 
>+#include "absl/memory/memory.h"
> #include "logging/rtc_event_log/events/rtc_event_bwe_update_loss_based.h"
> #include "logging/rtc_event_log/rtc_event_log.h"
> #include "modules/remote_bitrate_estimator/include/bwe_defines.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/ptr_util.h"
> #include "system_wrappers/include/field_trial.h"
> #include "system_wrappers/include/metrics.h"
> 
>@@ -104,7 +105,7 @@ bool ReadBweLossExperimentParameters(float* low_loss_threshold,
> }  // namespace
> 
> SendSideBandwidthEstimation::SendSideBandwidthEstimation(RtcEventLog* event_log)
>-    : lost_packets_since_last_loss_update_Q8_(0),
>+    : lost_packets_since_last_loss_update_(0),
>       expected_packets_since_last_loss_update_(0),
>       current_bitrate_bps_(0),
>       min_bitrate_configured_(congestion_controller::GetMinBitrateBps()),
>@@ -124,6 +125,7 @@ SendSideBandwidthEstimation::SendSideBandwidthEstimation(RtcEventLog* event_log)
>       initially_lost_packets_(0),
>       bitrate_at_2_seconds_kbps_(0),
>       uma_update_state_(kNoUpdate),
>+      uma_rtt_state_(kNoUpdate),
>       rampup_uma_stats_updated_(kNumUmaRampupMetrics, false),
>       event_log_(event_log),
>       last_rtc_event_log_ms_(-1),
>@@ -191,8 +193,8 @@ void SendSideBandwidthEstimation::CurrentEstimate(int* bitrate,
>   *rtt = last_round_trip_time_ms_;
> }
> 
>-void SendSideBandwidthEstimation::UpdateReceiverEstimate(
>-    int64_t now_ms, uint32_t bandwidth) {
>+void SendSideBandwidthEstimation::UpdateReceiverEstimate(int64_t now_ms,
>+                                                         uint32_t bandwidth) {
>   bwe_incoming_ = bandwidth;
>   CapBitrateToThresholds(now_ms, current_bitrate_bps_);
> }
>@@ -205,24 +207,28 @@ void SendSideBandwidthEstimation::UpdateDelayBasedEstimate(
> }
> 
> void SendSideBandwidthEstimation::UpdateReceiverBlock(uint8_t fraction_loss,
>-                                                      int64_t rtt,
>+                                                      int64_t rtt_ms,
>                                                       int number_of_packets,
>                                                       int64_t now_ms) {
>+  const int kRoundingConstant = 128;
>+  int packets_lost = (static_cast<int>(fraction_loss) * number_of_packets +
>+                      kRoundingConstant) >>
>+                     8;
>+  UpdatePacketsLost(packets_lost, number_of_packets, now_ms);
>+  UpdateRtt(rtt_ms, now_ms);
>+}
>+
>+void SendSideBandwidthEstimation::UpdatePacketsLost(int packets_lost,
>+                                                    int number_of_packets,
>+                                                    int64_t now_ms) {
>   last_feedback_ms_ = now_ms;
>   if (first_report_time_ms_ == -1)
>     first_report_time_ms_ = now_ms;
> 
>-  // Update RTT if we were able to compute an RTT based on this RTCP.
>-  // FlexFEC doesn't send RTCP SR, which means we won't be able to compute RTT.
>-  if (rtt > 0)
>-    last_round_trip_time_ms_ = rtt;
>-
>   // Check sequence number diff and weight loss report
>   if (number_of_packets > 0) {
>-    // Calculate number of lost packets.
>-    const int num_lost_packets_Q8 = fraction_loss * number_of_packets;
>     // Accumulate reports.
>-    lost_packets_since_last_loss_update_Q8_ += num_lost_packets_Q8;
>+    lost_packets_since_last_loss_update_ += packets_lost;
>     expected_packets_since_last_loss_update_ += number_of_packets;
> 
>     // Don't generate a loss rate until it can be based on enough packets.
>@@ -230,21 +236,22 @@ void SendSideBandwidthEstimation::UpdateReceiverBlock(uint8_t fraction_loss,
>       return;
> 
>     has_decreased_since_last_fraction_loss_ = false;
>-    last_fraction_loss_ = lost_packets_since_last_loss_update_Q8_ /
>-                          expected_packets_since_last_loss_update_;
>+    int64_t lost_q8 = lost_packets_since_last_loss_update_ << 8;
>+    int64_t expected = expected_packets_since_last_loss_update_;
>+    last_fraction_loss_ = std::min<int>(lost_q8 / expected, 255);
> 
>     // Reset accumulators.
>-    lost_packets_since_last_loss_update_Q8_ = 0;
>+
>+    lost_packets_since_last_loss_update_ = 0;
>     expected_packets_since_last_loss_update_ = 0;
>     last_packet_report_ms_ = now_ms;
>     UpdateEstimate(now_ms);
>   }
>-  UpdateUmaStats(now_ms, rtt, (fraction_loss * number_of_packets) >> 8);
>+  UpdateUmaStatsPacketsLost(now_ms, packets_lost);
> }
> 
>-void SendSideBandwidthEstimation::UpdateUmaStats(int64_t now_ms,
>-                                                 int64_t rtt,
>-                                                 int lost_packets) {
>+void SendSideBandwidthEstimation::UpdateUmaStatsPacketsLost(int64_t now_ms,
>+                                                            int packets_lost) {
>   int bitrate_kbps = static_cast<int>((current_bitrate_bps_ + 500) / 1000);
>   for (size_t i = 0; i < kNumUmaRampupMetrics; ++i) {
>     if (!rampup_uma_stats_updated_[i] &&
>@@ -255,14 +262,12 @@ void SendSideBandwidthEstimation::UpdateUmaStats(int64_t now_ms,
>     }
>   }
>   if (IsInStartPhase(now_ms)) {
>-    initially_lost_packets_ += lost_packets;
>+    initially_lost_packets_ += packets_lost;
>   } else if (uma_update_state_ == kNoUpdate) {
>     uma_update_state_ = kFirstDone;
>     bitrate_at_2_seconds_kbps_ = bitrate_kbps;
>     RTC_HISTOGRAM_COUNTS("WebRTC.BWE.InitiallyLostPackets",
>                          initially_lost_packets_, 0, 100, 50);
>-    RTC_HISTOGRAM_COUNTS("WebRTC.BWE.InitialRtt", static_cast<int>(rtt), 0,
>-                         2000, 50);
>     RTC_HISTOGRAM_COUNTS("WebRTC.BWE.InitialBandwidthEstimate",
>                          bitrate_at_2_seconds_kbps_, 0, 2000, 50);
>   } else if (uma_update_state_ == kFirstDone &&
>@@ -275,6 +280,19 @@ void SendSideBandwidthEstimation::UpdateUmaStats(int64_t now_ms,
>   }
> }
> 
>+void SendSideBandwidthEstimation::UpdateRtt(int64_t rtt_ms, int64_t now_ms) {
>+  // Update RTT if we were able to compute an RTT based on this RTCP.
>+  // FlexFEC doesn't send RTCP SR, which means we won't be able to compute RTT.
>+  if (rtt_ms > 0)
>+    last_round_trip_time_ms_ = rtt_ms;
>+
>+  if (!IsInStartPhase(now_ms) && uma_rtt_state_ == kNoUpdate) {
>+    uma_rtt_state_ = kDone;
>+    RTC_HISTOGRAM_COUNTS("WebRTC.BWE.InitialRtt", static_cast<int>(rtt_ms), 0,
>+                         2000, 50);
>+  }
>+}
>+
> void SendSideBandwidthEstimation::UpdateEstimate(int64_t now_ms) {
>   uint32_t new_bitrate = current_bitrate_bps_;
>   // We trust the REMB and/or delay-based estimate during the first 2 seconds if
>@@ -356,7 +374,7 @@ void SendSideBandwidthEstimation::UpdateEstimate(int64_t now_ms) {
>       new_bitrate *= 0.8;
>       // Reset accumulators since we've already acted on missing feedback and
>       // shouldn't to act again on these old lost packets.
>-      lost_packets_since_last_loss_update_Q8_ = 0;
>+      lost_packets_since_last_loss_update_ = 0;
>       expected_packets_since_last_loss_update_ = 0;
>       last_timeout_ms_ = now_ms;
>     }
>@@ -416,7 +434,7 @@ void SendSideBandwidthEstimation::CapBitrateToThresholds(int64_t now_ms,
>   if (bitrate_bps != current_bitrate_bps_ ||
>       last_fraction_loss_ != last_logged_fraction_loss_ ||
>       now_ms - last_rtc_event_log_ms_ > kRtcEventLogPeriodMs) {
>-    event_log_->Log(rtc::MakeUnique<RtcEventBweUpdateLossBased>(
>+    event_log_->Log(absl::make_unique<RtcEventBweUpdateLossBased>(
>         bitrate_bps, last_fraction_loss_,
>         expected_packets_since_last_loss_update_));
>     last_logged_fraction_loss_ = last_fraction_loss_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h
>index 59d1c3280d2..54b571e96eb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.h
>@@ -42,13 +42,19 @@ class SendSideBandwidthEstimation {
> 
>   // Call when we receive a RTCP message with a ReceiveBlock.
>   void UpdateReceiverBlock(uint8_t fraction_loss,
>-                           int64_t rtt,
>+                           int64_t rtt_ms,
>                            int number_of_packets,
>                            int64_t now_ms);
> 
>-  void SetBitrates(int send_bitrate,
>-                   int min_bitrate,
>-                   int max_bitrate);
>+  // Call when we receive a RTCP message with a ReceiveBlock.
>+  void UpdatePacketsLost(int packets_lost,
>+                         int number_of_packets,
>+                         int64_t now_ms);
>+
>+  // Call when we receive a RTCP message with a ReceiveBlock.
>+  void UpdateRtt(int64_t rtt, int64_t now_ms);
>+
>+  void SetBitrates(int send_bitrate, int min_bitrate, int max_bitrate);
>   void SetSendBitrate(int bitrate);
>   void SetMinMaxBitrate(int min_bitrate, int max_bitrate);
>   int GetMinBitrate() const;
>@@ -58,7 +64,7 @@ class SendSideBandwidthEstimation {
> 
>   bool IsInStartPhase(int64_t now_ms) const;
> 
>-  void UpdateUmaStats(int64_t now_ms, int64_t rtt, int lost_packets);
>+  void UpdateUmaStatsPacketsLost(int64_t now_ms, int packets_lost);
> 
>   // Updates history of min bitrates.
>   // After this method returns min_bitrate_history_.front().second contains the
>@@ -72,7 +78,7 @@ class SendSideBandwidthEstimation {
>   std::deque<std::pair<int64_t, uint32_t> > min_bitrate_history_;
> 
>   // incoming filters
>-  int lost_packets_since_last_loss_update_Q8_;
>+  int lost_packets_since_last_loss_update_;
>   int expected_packets_since_last_loss_update_;
> 
>   uint32_t current_bitrate_bps_;
>@@ -95,6 +101,7 @@ class SendSideBandwidthEstimation {
>   int initially_lost_packets_;
>   int bitrate_at_2_seconds_kbps_;
>   UmaState uma_update_state_;
>+  UmaState uma_rtt_state_;
>   std::vector<bool> rampup_uma_stats_updated_;
>   RtcEventLog* event_log_;
>   int64_t last_rtc_event_log_ms_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation_unittest.cc
>index 567d81f610a..8d5b08bd04f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation_unittest.cc
>@@ -164,5 +164,4 @@ TEST(SendSideBweTest, SettingSendBitrateOverridesDelayBasedEstimate) {
>   EXPECT_EQ(bitrate_bps, kForcedHighBitrate);
> }
> 
>-
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/BUILD.gn
>index 84de71dd1f0..27a73693222 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/BUILD.gn
>@@ -20,30 +20,28 @@ rtc_static_library("congestion_controller") {
>   visibility = [ "*" ]
>   configs += [ ":bwe_test_logging" ]
>   sources = [
>+    "congestion_window_pushback_controller.cc",
>+    "congestion_window_pushback_controller.h",
>+    "include/network_changed_observer.h",
>     "include/receive_side_congestion_controller.h",
>     "include/send_side_congestion_controller.h",
>-    "probe_controller.cc",
>-    "probe_controller.h",
>+    "include/send_side_congestion_controller_interface.h",
>     "receive_side_congestion_controller.cc",
>     "send_side_congestion_controller.cc",
>-    "transport_feedback_adapter.cc",
>-    "transport_feedback_adapter.h",
>   ]
> 
>-  # TODO(jschuh): Bug 1348: fix this warning.
>-  configs += [ "//build/config/compiler:no_size_t_to_int_warning" ]
>-
>   if (!build_with_chromium && is_clang) {
>     # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>     suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>   }
> 
>   deps = [
>-    ":delay_based_bwe",
>-    ":estimators",
>+    ":transport_feedback",
>     "..:module_api",
>     "../..:webrtc_common",
>+    "../../api/transport:network_control",
>     "../../rtc_base:checks",
>+    "../../rtc_base:ptr_util",
>     "../../rtc_base:rate_limiter",
>     "../../system_wrappers",
>     "../../system_wrappers:field_trial_api",
>@@ -53,6 +51,10 @@ rtc_static_library("congestion_controller") {
>     "../pacing",
>     "../remote_bitrate_estimator",
>     "../rtp_rtcp:rtp_rtcp_format",
>+    "goog_cc:delay_based_bwe",
>+    "goog_cc:estimators",
>+    "goog_cc:probe_controller",
>+    "//third_party/abseil-cpp/absl/memory",
>   ]
> 
>   if (!build_with_mozilla) {
>@@ -60,90 +62,52 @@ rtc_static_library("congestion_controller") {
>   }
> }
> 
>-rtc_source_set("estimators") {
>-  configs += [ ":bwe_test_logging" ]
>+rtc_static_library("transport_feedback") {
>+  visibility = [ "*" ]
>   sources = [
>-    "acknowledged_bitrate_estimator.cc",
>-    "acknowledged_bitrate_estimator.h",
>-    "bitrate_estimator.cc",
>-    "bitrate_estimator.h",
>-    "median_slope_estimator.cc",
>-    "median_slope_estimator.h",
>-    "probe_bitrate_estimator.cc",
>-    "probe_bitrate_estimator.h",
>-    "trendline_estimator.cc",
>-    "trendline_estimator.h",
>+    "transport_feedback_adapter.cc",
>+    "transport_feedback_adapter.h",
>   ]
> 
>-  # TODO(jschuh): Bug 1348: fix this warning.
>-  configs += [ "//build/config/compiler:no_size_t_to_int_warning" ]
>-
>-  if (!build_with_chromium && is_clang) {
>-    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>-    suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-  }
>-
>   deps = [
>-    "../../api:optional",
>-    "../../logging:rtc_event_log_api",
>+    "../../modules:module_api",
>     "../../rtc_base:checks",
>     "../../rtc_base:rtc_base_approved",
>-    "../../rtc_base:rtc_numerics",
>-    "../../system_wrappers:field_trial_api",
>-    "../../system_wrappers:metrics_api",
>-    "../remote_bitrate_estimator:remote_bitrate_estimator",
>+    "../../system_wrappers:system_wrappers",
>     "../rtp_rtcp:rtp_rtcp_format",
>+    "rtp:transport_feedback",
>   ]
> }
> 
>-rtc_source_set("delay_based_bwe") {
>-  configs += [ ":bwe_test_logging" ]
>-  sources = [
>-    "delay_based_bwe.cc",
>-    "delay_based_bwe.h",
>-  ]
>-  deps = [
>-    ":estimators",
>-    "../../:typedefs",
>-    "../../logging:rtc_event_log_api",
>-    "../../rtc_base:checks",
>-    "../../rtc_base:rtc_base_approved",
>-    "../../system_wrappers:field_trial_api",
>-    "../../system_wrappers:metrics_api",
>-    "../pacing",
>-    "../remote_bitrate_estimator",
>-  ]
>-
>-  if (!build_with_chromium && is_clang) {
>-    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>-    suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-  }
>-}
>-
> if (rtc_include_tests) {
>+  rtc_source_set("test_controller_printer") {
>+    testonly = true
>+    sources = [
>+      "test/controller_printer.cc",
>+      "test/controller_printer.h",
>+    ]
>+    if (!build_with_chromium && is_clang) {
>+      # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>+      suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>+    }
>+    deps = [
>+      "../../api/transport:network_control",
>+      "../../rtc_base:checks",
>+    ]
>+  }
>   rtc_source_set("congestion_controller_unittests") {
>     testonly = true
> 
>     sources = [
>-      "acknowledged_bitrate_estimator_unittest.cc",
>-      "congestion_controller_unittests_helper.cc",
>-      "congestion_controller_unittests_helper.h",
>-      "delay_based_bwe_unittest.cc",
>-      "delay_based_bwe_unittest_helper.cc",
>-      "delay_based_bwe_unittest_helper.h",
>-      "median_slope_estimator_unittest.cc",
>-      "probe_bitrate_estimator_unittest.cc",
>-      "probe_controller_unittest.cc",
>+      "congestion_window_pushback_controller_unittest.cc",
>       "receive_side_congestion_controller_unittest.cc",
>       "send_side_congestion_controller_unittest.cc",
>       "transport_feedback_adapter_unittest.cc",
>-      "trendline_estimator_unittest.cc",
>     ]
>     deps = [
>       ":congestion_controller",
>-      ":delay_based_bwe",
>-      ":estimators",
>       ":mock_congestion_controller",
>+      ":transport_feedback",
>       "../../logging:mocks",
>       "../../rtc_base:checks",
>       "../../rtc_base:rtc_base",
>@@ -158,7 +122,10 @@ if (rtc_include_tests) {
>       "../pacing:pacing",
>       "../remote_bitrate_estimator:remote_bitrate_estimator",
>       "../rtp_rtcp:rtp_rtcp_format",
>-      "//testing/gmock",
>+      "bbr:bbr_unittests",
>+      "goog_cc:estimators",
>+      "goog_cc:goog_cc_unittests",
>+      "rtp:congestion_controller_unittests",
>     ]
>     if (!build_with_chromium && is_clang) {
>       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>@@ -170,7 +137,6 @@ if (rtc_include_tests) {
>     testonly = true
>     sources = [
>       "include/mock/mock_congestion_observer.h",
>-      "include/mock/mock_send_side_congestion_controller.h",
>     ]
>     deps = [
>       ":congestion_controller",
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/OWNERS
>index 0a94c47784b..de5159680c4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/OWNERS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/OWNERS
>@@ -1,3 +1,4 @@
>+srte@webrtc.org
> stefan@webrtc.org
> terelius@webrtc.org
> philipel@webrtc.org
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/BUILD.gn
>new file mode 100644
>index 00000000000..a424c118f43
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/BUILD.gn
>@@ -0,0 +1,160 @@
>+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+#
>+# Use of this source code is governed by a BSD-style license
>+# that can be found in the LICENSE file in the root of the source
>+# tree. An additional intellectual property rights grant can be found
>+# in the file PATENTS.  All contributing project authors may
>+# be found in the AUTHORS file in the root of the source tree.
>+
>+import("../../../webrtc.gni")
>+
>+rtc_static_library("bbr") {
>+  sources = [
>+    "bbr_factory.cc",
>+    "bbr_factory.h",
>+  ]
>+  deps = [
>+    ":bbr_controller",
>+    "../../../api/transport:network_control",
>+    "../../../rtc_base:rtc_base_approved",
>+    "//third_party/abseil-cpp/absl/memory",
>+  ]
>+}
>+
>+rtc_source_set("bbr_controller") {
>+  visibility = [ ":*" ]
>+  sources = [
>+    "bbr_network_controller.cc",
>+    "bbr_network_controller.h",
>+  ]
>+  deps = [
>+    ":bandwidth_sampler",
>+    ":loss_rate_filter",
>+    ":rtt_stats",
>+    ":windowed_filter",
>+    "../../../api/transport:network_control",
>+    "../../../rtc_base:checks",
>+    "../../../rtc_base:rtc_base_approved",
>+    "../../../rtc_base/experiments:field_trial_parser",
>+    "../../../rtc_base/system:fallthrough",
>+    "../../../system_wrappers:field_trial_api",
>+    "//third_party/abseil-cpp/absl/types:optional",
>+  ]
>+}
>+
>+rtc_source_set("bandwidth_sampler") {
>+  visibility = [ ":*" ]
>+  sources = [
>+    "bandwidth_sampler.cc",
>+    "bandwidth_sampler.h",
>+  ]
>+  deps = [
>+    ":packet_number_indexed_queue",
>+    "../../../api/units:data_rate",
>+    "../../../api/units:data_size",
>+    "../../../api/units:time_delta",
>+    "../../../api/units:timestamp",
>+    "../../../rtc_base:checks",
>+    "../../../rtc_base:rtc_base_approved",
>+    "//third_party/abseil-cpp/absl/types:optional",
>+  ]
>+}
>+
>+rtc_source_set("data_transfer_tracker") {
>+  visibility = [ ":*" ]
>+  sources = [
>+    "data_transfer_tracker.cc",
>+    "data_transfer_tracker.h",
>+  ]
>+  deps = [
>+    "../../../api/units:data_size",
>+    "../../../api/units:time_delta",
>+    "../../../api/units:timestamp",
>+    "../../../rtc_base:checks",
>+    "../../../rtc_base:rtc_base_approved",
>+  ]
>+}
>+
>+rtc_source_set("packet_number_indexed_queue") {
>+  visibility = [ ":*" ]
>+  sources = [
>+    "packet_number_indexed_queue.h",
>+  ]
>+  deps = [
>+    "../../../rtc_base:checks",
>+  ]
>+}
>+
>+rtc_source_set("loss_rate_filter") {
>+  visibility = [ ":*" ]
>+  sources = [
>+    "loss_rate_filter.cc",
>+    "loss_rate_filter.h",
>+  ]
>+  deps = [
>+    "//third_party/abseil-cpp/absl/types:optional",
>+  ]
>+}
>+rtc_source_set("rtt_stats") {
>+  visibility = [ ":*" ]
>+  sources = [
>+    "rtt_stats.cc",
>+    "rtt_stats.h",
>+  ]
>+  deps = [
>+    "../../../api/units:time_delta",
>+    "../../../api/units:timestamp",
>+    "../../../rtc_base:rtc_base_approved",
>+  ]
>+}
>+rtc_source_set("windowed_filter") {
>+  visibility = [ ":*" ]
>+  sources = [
>+    "windowed_filter.h",
>+  ]
>+}
>+if (rtc_include_tests) {
>+  rtc_source_set("test_bbr_printer") {
>+    testonly = true
>+    sources = [
>+      "test/bbr_printer.cc",
>+      "test/bbr_printer.h",
>+    ]
>+    deps = [
>+      ":bbr",
>+      ":bbr_controller",
>+      "..:test_controller_printer",
>+    ]
>+  }
>+  rtc_source_set("bbr_unittests") {
>+    testonly = true
>+    sources = [
>+      "bandwidth_sampler_unittest.cc",
>+      "bbr_network_controller_unittest.cc",
>+      "data_transfer_tracker_unittest.cc",
>+      "loss_rate_filter_unittest.cc",
>+      "packet_number_indexed_queue_unittest.cc",
>+      "rtt_stats_unittest.cc",
>+      "windowed_filter_unittest.cc",
>+    ]
>+    deps = [
>+      ":bandwidth_sampler",
>+      ":bbr",
>+      ":bbr_controller",
>+      ":data_transfer_tracker",
>+      ":loss_rate_filter",
>+      ":packet_number_indexed_queue",
>+      ":rtt_stats",
>+      ":windowed_filter",
>+      "../../../api/transport:network_control_test",
>+      "../../../api/units:data_rate",
>+      "../../../api/units:time_delta",
>+      "../../../api/units:timestamp",
>+      "../../../test:test_support",
>+    ]
>+    if (!build_with_chromium && is_clang) {
>+      # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>+      suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>+    }
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/bandwidth_sampler.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/bandwidth_sampler.cc
>new file mode 100644
>index 00000000000..b9e2c27bddf
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/bandwidth_sampler.cc
>@@ -0,0 +1,204 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+// Based on the Quic implementation in Chromium.
>+
>+#include <algorithm>
>+
>+#include "modules/congestion_controller/bbr/bandwidth_sampler.h"
>+#include "rtc_base/logging.h"
>+
>+namespace webrtc {
>+namespace bbr {
>+namespace {
>+constexpr int64_t kMaxTrackedPackets = 10000;
>+}
>+
>+BandwidthSampler::BandwidthSampler()
>+    : total_data_sent_(DataSize::Zero()),
>+      total_data_acked_(DataSize::Zero()),
>+      total_data_sent_at_last_acked_packet_(DataSize::Zero()),
>+      last_acked_packet_sent_time_(),
>+      last_acked_packet_ack_time_(),
>+      last_sent_packet_(0),
>+      is_app_limited_(false),
>+      end_of_app_limited_phase_(0),
>+      connection_state_map_() {}
>+
>+BandwidthSampler::~BandwidthSampler() {}
>+
>+void BandwidthSampler::OnPacketSent(Timestamp sent_time,
>+                                    int64_t packet_number,
>+                                    DataSize data_size,
>+                                    DataSize data_in_flight) {
>+  last_sent_packet_ = packet_number;
>+
>+  total_data_sent_ += data_size;
>+
>+  // If there are no packets in flight, the time at which the new transmission
>+  // opens can be treated as the A_0 point for the purpose of bandwidth
>+  // sampling. This underestimates bandwidth to some extent, and produces some
>+  // artificially low samples for most packets in flight, but it provides with
>+  // samples at important points where we would not have them otherwise, most
>+  // importantly at the beginning of the connection.
>+  if (data_in_flight.IsZero()) {
>+    last_acked_packet_ack_time_ = sent_time;
>+    total_data_sent_at_last_acked_packet_ = total_data_sent_;
>+
>+    // In this situation ack compression is not a concern, set send rate to
>+    // effectively infinite.
>+    last_acked_packet_sent_time_ = sent_time;
>+  }
>+
>+  if (!connection_state_map_.IsEmpty() &&
>+      packet_number >
>+          connection_state_map_.last_packet() + kMaxTrackedPackets) {
>+    RTC_LOG(LS_WARNING)
>+        << "BandwidthSampler in-flight packet map has exceeded maximum "
>+           "number "
>+           "of tracked packets.";
>+  }
>+
>+  bool success =
>+      connection_state_map_.Emplace(packet_number, sent_time, data_size, *this);
>+  if (!success)
>+    RTC_LOG(LS_WARNING) << "BandwidthSampler failed to insert the packet "
>+                           "into the map, most likely because it's already "
>+                           "in it.";
>+}
>+
>+BandwidthSample BandwidthSampler::OnPacketAcknowledged(Timestamp ack_time,
>+                                                       int64_t packet_number) {
>+  ConnectionStateOnSentPacket* sent_packet_pointer =
>+      connection_state_map_.GetEntry(packet_number);
>+  if (sent_packet_pointer == nullptr) {
>+    return BandwidthSample();
>+  }
>+  BandwidthSample sample =
>+      OnPacketAcknowledgedInner(ack_time, packet_number, *sent_packet_pointer);
>+  connection_state_map_.Remove(packet_number);
>+  return sample;
>+}
>+
>+BandwidthSample BandwidthSampler::OnPacketAcknowledgedInner(
>+    Timestamp ack_time,
>+    int64_t packet_number,
>+    const ConnectionStateOnSentPacket& sent_packet) {
>+  total_data_acked_ += sent_packet.size;
>+  total_data_sent_at_last_acked_packet_ = sent_packet.total_data_sent;
>+  last_acked_packet_sent_time_ = sent_packet.sent_time;
>+  last_acked_packet_ack_time_ = ack_time;
>+
>+  // Exit app-limited phase once a packet that was sent while the connection is
>+  // not app-limited is acknowledged.
>+  if (is_app_limited_ && packet_number > end_of_app_limited_phase_) {
>+    is_app_limited_ = false;
>+  }
>+
>+  // There might have been no packets acknowledged at the moment when the
>+  // current packet was sent. In that case, there is no bandwidth sample to
>+  // make.
>+  if (!sent_packet.last_acked_packet_sent_time ||
>+      !sent_packet.last_acked_packet_ack_time) {
>+    return BandwidthSample();
>+  }
>+
>+  // Infinite rate indicates that the sampler is supposed to discard the
>+  // current send rate sample and use only the ack rate.
>+  DataRate send_rate = DataRate::Infinity();
>+  if (sent_packet.sent_time > *sent_packet.last_acked_packet_sent_time) {
>+    DataSize sent_delta = sent_packet.total_data_sent -
>+                          sent_packet.total_data_sent_at_last_acked_packet;
>+    TimeDelta time_delta =
>+        sent_packet.sent_time - *sent_packet.last_acked_packet_sent_time;
>+    send_rate = sent_delta / time_delta;
>+  }
>+
>+  // During the slope calculation, ensure that ack time of the current packet is
>+  // always larger than the time of the previous packet, otherwise division by
>+  // zero or integer underflow can occur.
>+  if (ack_time <= *sent_packet.last_acked_packet_ack_time) {
>+    RTC_LOG(LS_WARNING)
>+        << "Time of the previously acked packet is larger than the time "
>+           "of the current packet.";
>+    return BandwidthSample();
>+  }
>+  DataSize ack_delta =
>+      total_data_acked_ - sent_packet.total_data_acked_at_the_last_acked_packet;
>+  TimeDelta time_delta = ack_time - *sent_packet.last_acked_packet_ack_time;
>+  DataRate ack_rate = ack_delta / time_delta;
>+
>+  BandwidthSample sample;
>+  sample.bandwidth = std::min(send_rate, ack_rate);
>+  // Note: this sample does not account for delayed acknowledgement time.  This
>+  // means that the RTT measurements here can be artificially high, especially
>+  // on low bandwidth connections.
>+  sample.rtt = ack_time - sent_packet.sent_time;
>+  // A sample is app-limited if the packet was sent during the app-limited
>+  // phase.
>+  sample.is_app_limited = sent_packet.is_app_limited;
>+  return sample;
>+}
>+
>+void BandwidthSampler::OnPacketLost(int64_t packet_number) {
>+  connection_state_map_.Remove(packet_number);
>+}
>+
>+void BandwidthSampler::OnAppLimited() {
>+  is_app_limited_ = true;
>+  end_of_app_limited_phase_ = last_sent_packet_;
>+}
>+
>+void BandwidthSampler::RemoveObsoletePackets(int64_t least_unacked) {
>+  while (!connection_state_map_.IsEmpty() &&
>+         connection_state_map_.first_packet() < least_unacked) {
>+    connection_state_map_.Remove(connection_state_map_.first_packet());
>+  }
>+}
>+
>+DataSize BandwidthSampler::total_data_acked() const {
>+  return total_data_acked_;
>+}
>+
>+bool BandwidthSampler::is_app_limited() const {
>+  return is_app_limited_;
>+}
>+
>+int64_t BandwidthSampler::end_of_app_limited_phase() const {
>+  return end_of_app_limited_phase_;
>+}
>+
>+BandwidthSampler::ConnectionStateOnSentPacket::ConnectionStateOnSentPacket(
>+    Timestamp sent_time,
>+    DataSize size,
>+    const BandwidthSampler& sampler)
>+    : sent_time(sent_time),
>+      size(size),
>+      total_data_sent(sampler.total_data_sent_),
>+      total_data_sent_at_last_acked_packet(
>+          sampler.total_data_sent_at_last_acked_packet_),
>+      last_acked_packet_sent_time(sampler.last_acked_packet_sent_time_),
>+      last_acked_packet_ack_time(sampler.last_acked_packet_ack_time_),
>+      total_data_acked_at_the_last_acked_packet(sampler.total_data_acked_),
>+      is_app_limited(sampler.is_app_limited_) {}
>+
>+BandwidthSampler::ConnectionStateOnSentPacket::ConnectionStateOnSentPacket()
>+    : sent_time(Timestamp::ms(0)),
>+      size(DataSize::Zero()),
>+      total_data_sent(DataSize::Zero()),
>+      total_data_sent_at_last_acked_packet(DataSize::Zero()),
>+      last_acked_packet_sent_time(),
>+      last_acked_packet_ack_time(),
>+      total_data_acked_at_the_last_acked_packet(DataSize::Zero()),
>+      is_app_limited(false) {}
>+
>+BandwidthSampler::ConnectionStateOnSentPacket::~ConnectionStateOnSentPacket() {}
>+
>+}  // namespace bbr
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/bandwidth_sampler.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/bandwidth_sampler.h
>new file mode 100644
>index 00000000000..7e0a44e61d9
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/bandwidth_sampler.h
>@@ -0,0 +1,261 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+// Based on the Quic implementation in Chromium.
>+
>+#ifndef MODULES_CONGESTION_CONTROLLER_BBR_BANDWIDTH_SAMPLER_H_
>+#define MODULES_CONGESTION_CONTROLLER_BBR_BANDWIDTH_SAMPLER_H_
>+
>+#include "absl/types/optional.h"
>+#include "api/units/data_rate.h"
>+#include "api/units/data_size.h"
>+#include "api/units/time_delta.h"
>+#include "api/units/timestamp.h"
>+#include "modules/congestion_controller/bbr/packet_number_indexed_queue.h"
>+
>+namespace webrtc {
>+namespace bbr {
>+
>+namespace test {
>+class BandwidthSamplerPeer;
>+}  // namespace test
>+
>+struct BandwidthSample {
>+  // The bandwidth at that particular sample. Zero if no valid bandwidth sample
>+  // is available.
>+  DataRate bandwidth;
>+
>+  // The RTT measurement at this particular sample.  Zero if no RTT sample is
>+  // available.  Does not correct for delayed ack time.
>+  TimeDelta rtt;
>+
>+  // Indicates whether the sample might be artificially low because the sender
>+  // did not have enough data to send in order to saturate the link.
>+  bool is_app_limited;
>+
>+  BandwidthSample()
>+      : bandwidth(DataRate::Zero()),
>+        rtt(TimeDelta::Zero()),
>+        is_app_limited(false) {}
>+};
>+
>+// BandwidthSampler keeps track of sent and acknowledged packets and outputs a
>+// bandwidth sample for every packet acknowledged. The samples are taken for
>+// individual packets, and are not filtered; the consumer has to filter the
>+// bandwidth samples itself. In certain cases, the sampler will locally severely
>+// underestimate the bandwidth, hence a maximum filter with a size of at least
>+// one RTT is recommended.
>+//
>+// This class bases its samples on the slope of two curves: the number of
>+// data_size sent over time, and the number of data_size acknowledged as
>+// received over time. It produces a sample of both slopes for every packet that
>+// gets acknowledged, based on a slope between two points on each of the
>+// corresponding curves. Note that due to the packet loss, the number of
>+// data_size on each curve might get further and further away from each other,
>+// meaning that it is not feasible to compare byte values coming from different
>+// curves with each other.
>+//
>+// The obvious points for measuring slope sample are the ones corresponding to
>+// the packet that was just acknowledged. Let us denote them as S_1 (point at
>+// which the current packet was sent) and A_1 (point at which the current packet
>+// was acknowledged). However, taking a slope requires two points on each line,
>+// so estimating bandwidth requires picking a packet in the past with respect to
>+// which the slope is measured.
>+//
>+// For that purpose, BandwidthSampler always keeps track of the most recently
>+// acknowledged packet, and records it together with every outgoing packet.
>+// When a packet gets acknowledged (A_1), it has not only information about when
>+// it itself was sent (S_1), but also the information about the latest
>+// acknowledged packet right before it was sent (S_0 and A_0).
>+//
>+// Based on that data, send and ack rate are estimated as:
>+//   send_rate = (data_size(S_1) - data_size(S_0)) / (time(S_1) - time(S_0))
>+//   ack_rate = (data_size(A_1) - data_size(A_0)) / (time(A_1) - time(A_0))
>+//
>+// Here, the ack rate is intuitively the rate we want to treat as bandwidth.
>+// However, in certain cases (e.g. ack compression) the ack rate at a point may
>+// end up higher than the rate at which the data was originally sent, which is
>+// not indicative of the real bandwidth. Hence, we use the send rate as an upper
>+// bound, and the sample value is
>+//   rate_sample = min(send_rate, ack_rate)
>+//
>+// An important edge case handled by the sampler is tracking the app-limited
>+// samples. There are multiple meaning of "app-limited" used interchangeably,
>+// hence it is important to understand and to be able to distinguish between
>+// them.
>+//
>+// Meaning 1: connection state. The connection is said to be app-limited when
>+// there is no outstanding data to send. This means that certain bandwidth
>+// samples in the future would not be an accurate indication of the link
>+// capacity, and it is important to inform consumer about that. Whenever
>+// connection becomes app-limited, the sampler is notified via OnAppLimited()
>+// method.
>+//
>+// Meaning 2: a phase in the bandwidth sampler. As soon as the bandwidth
>+// sampler becomes notified about the connection being app-limited, it enters
>+// app-limited phase. In that phase, all *sent* packets are marked as
>+// app-limited. Note that the connection itself does not have to be
>+// app-limited during the app-limited phase, and in fact it will not be
>+// (otherwise how would it send packets?). The boolean flag below indicates
>+// whether the sampler is in that phase.
>+//
>+// Meaning 3: a flag on the sent packet and on the sample. If a sent packet is
>+// sent during the app-limited phase, the resulting sample related to the
>+// packet will be marked as app-limited.
>+//
>+// With the terminology issue out of the way, let us consider the question of
>+// what kind of situation it addresses.
>+//
>+// Consider a scenario where we first send packets 1 to 20 at a regular
>+// bandwidth, and then immediately run out of data. After a few seconds, we send
>+// packets 21 to 60, and only receive ack for 21 between sending packets 40 and
>+// 41. In this case, when we sample bandwidth for packets 21 to 40, the S_0/A_0
>+// we use to compute the slope is going to be packet 20, a few seconds apart
>+// from the current packet, hence the resulting estimate would be extremely low
>+// and not indicative of anything. Only at packet 41 the S_0/A_0 will become 21,
>+// meaning that the bandwidth sample would exclude the quiescence.
>+//
>+// Based on the analysis of that scenario, we implement the following rule: once
>+// OnAppLimited() is called, all sent packets will produce app-limited samples
>+// up until an ack for a packet that was sent after OnAppLimited() was called.
>+// Note that while the scenario above is not the only scenario when the
>+// connection is app-limited, the approach works in other cases too.
>+class BandwidthSampler {
>+ public:
>+  BandwidthSampler();
>+  ~BandwidthSampler();
>+  // Inputs the sent packet information into the sampler. Assumes that all
>+  // packets are sent in order. The information about the packet will not be
>+  // released from the sampler until the packet is either acknowledged or
>+  // declared lost.
>+  void OnPacketSent(Timestamp sent_time,
>+                    int64_t packet_number,
>+                    DataSize data_size,
>+                    DataSize data_in_flight);
>+
>+  // Notifies the sampler that the |packet_number| is acknowledged. Returns a
>+  // bandwidth sample. If no bandwidth sample is available, bandwidth is set to
>+  // DataRate::Zero().
>+  BandwidthSample OnPacketAcknowledged(Timestamp ack_time,
>+                                       int64_t packet_number);
>+
>+  // Informs the sampler that a packet is considered lost and it should no
>+  // longer keep track of it.
>+  void OnPacketLost(int64_t packet_number);
>+
>+  // Informs the sampler that the connection is currently app-limited, causing
>+  // the sampler to enter the app-limited phase.  The phase will expire by
>+  // itself.
>+  void OnAppLimited();
>+
>+  // Remove all the packets lower than the specified packet number.
>+  void RemoveObsoletePackets(int64_t least_unacked);
>+
>+  // Total number of data_size currently acknowledged by the receiver.
>+  DataSize total_data_acked() const;
>+
>+  // Application-limited information exported for debugging.
>+  bool is_app_limited() const;
>+  int64_t end_of_app_limited_phase() const;
>+
>+ private:
>+  friend class test::BandwidthSamplerPeer;
>+  // ConnectionStateOnSentPacket represents the information about a sent packet
>+  // and the state of the connection at the moment the packet was sent,
>+  // specifically the information about the most recently acknowledged packet at
>+  // that moment.
>+  struct ConnectionStateOnSentPacket {
>+    // Time at which the packet is sent.
>+    Timestamp sent_time;
>+
>+    // Size of the packet.
>+    DataSize size;
>+
>+    // The value of |total_data_sent_| at the time the packet was sent.
>+    // Includes the packet itself.
>+    DataSize total_data_sent;
>+
>+    // The value of |total_data_sent_at_last_acked_packet_| at the time the
>+    // packet was sent.
>+    DataSize total_data_sent_at_last_acked_packet;
>+
>+    // The value of |last_acked_packet_sent_time_| at the time the packet was
>+    // sent.
>+    absl::optional<Timestamp> last_acked_packet_sent_time;
>+
>+    // The value of |last_acked_packet_ack_time_| at the time the packet was
>+    // sent.
>+    absl::optional<Timestamp> last_acked_packet_ack_time;
>+
>+    // The value of |total_data_acked_| at the time the packet was
>+    // sent.
>+    DataSize total_data_acked_at_the_last_acked_packet;
>+
>+    // The value of |is_app_limited_| at the time the packet was
>+    // sent.
>+    bool is_app_limited;
>+
>+    // Snapshot constructor. Records the current state of the bandwidth
>+    // sampler.
>+    ConnectionStateOnSentPacket(Timestamp sent_time,
>+                                DataSize size,
>+                                const BandwidthSampler& sampler);
>+
>+    // Default constructor.  Required to put this structure into
>+    // PacketNumberIndexedQueue.
>+    ConnectionStateOnSentPacket();
>+    ~ConnectionStateOnSentPacket();
>+  };
>+
>+  // The total number of congestion controlled data_size sent during the
>+  // connection.
>+  DataSize total_data_sent_;
>+
>+  // The total number of congestion controlled data_size which were
>+  // acknowledged.
>+  DataSize total_data_acked_;
>+
>+  // The value of |total_data_sent_| at the time the last acknowledged packet
>+  // was sent. Valid only when |last_acked_packet_sent_time_| is valid.
>+  DataSize total_data_sent_at_last_acked_packet_;
>+
>+  // The time at which the last acknowledged packet was sent. Set to
>+  // Timestamp::Zero() if no valid timestamp is available.
>+  absl::optional<Timestamp> last_acked_packet_sent_time_;
>+
>+  // The time at which the most recent packet was acknowledged.
>+  absl::optional<Timestamp> last_acked_packet_ack_time_;
>+
>+  // The most recently sent packet.
>+  int64_t last_sent_packet_;
>+
>+  // Indicates whether the bandwidth sampler is currently in an app-limited
>+  // phase.
>+  bool is_app_limited_;
>+
>+  // The packet that will be acknowledged after this one will cause the sampler
>+  // to exit the app-limited phase.
>+  int64_t end_of_app_limited_phase_;
>+
>+  // Record of the connection state at the point where each packet in flight was
>+  // sent, indexed by the packet number.
>+  PacketNumberIndexedQueue<ConnectionStateOnSentPacket> connection_state_map_;
>+
>+  // Handles the actual bandwidth calculations, whereas the outer method handles
>+  // retrieving and removing |sent_packet|.
>+  BandwidthSample OnPacketAcknowledgedInner(
>+      Timestamp ack_time,
>+      int64_t packet_number,
>+      const ConnectionStateOnSentPacket& sent_packet);
>+};
>+
>+}  // namespace bbr
>+}  // namespace webrtc
>+
>+#endif  // MODULES_CONGESTION_CONTROLLER_BBR_BANDWIDTH_SAMPLER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/bandwidth_sampler_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/bandwidth_sampler_unittest.cc
>new file mode 100644
>index 00000000000..45f3f486fd5
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/bandwidth_sampler_unittest.cc
>@@ -0,0 +1,337 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+// Based on the Quic implementation in Chromium.
>+
>+#include <algorithm>
>+
>+#include "modules/congestion_controller/bbr/bandwidth_sampler.h"
>+
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+namespace bbr {
>+namespace test {
>+
>+class BandwidthSamplerPeer {
>+ public:
>+  static size_t GetNumberOfTrackedPackets(const BandwidthSampler& sampler) {
>+    return sampler.connection_state_map_.number_of_present_entries();
>+  }
>+
>+  static DataSize GetPacketSize(const BandwidthSampler& sampler,
>+                                int64_t packet_number) {
>+    return sampler.connection_state_map_.GetEntry(packet_number)->size;
>+  }
>+};
>+
>+const int64_t kRegularPacketSizeBytes = 1280;
>+// Enforce divisibility for some of the tests.
>+static_assert((kRegularPacketSizeBytes & 31) == 0,
>+              "kRegularPacketSizeBytes has to be five times divisible by 2");
>+
>+const DataSize kRegularPacketSize = DataSize::bytes(kRegularPacketSizeBytes);
>+
>+// A test fixture with utility methods for BandwidthSampler tests.
>+class BandwidthSamplerTest : public ::testing::Test {
>+ protected:
>+  BandwidthSamplerTest()
>+      : clock_(Timestamp::seconds(100)), bytes_in_flight_(DataSize::Zero()) {}
>+
>+  Timestamp clock_;
>+  BandwidthSampler sampler_;
>+  DataSize bytes_in_flight_;
>+
>+  void SendPacketInner(int64_t packet_number, DataSize bytes) {
>+    sampler_.OnPacketSent(clock_, packet_number, bytes, bytes_in_flight_);
>+    bytes_in_flight_ += bytes;
>+  }
>+
>+  void SendPacket(int64_t packet_number) {
>+    SendPacketInner(packet_number, kRegularPacketSize);
>+  }
>+
>+  BandwidthSample AckPacketInner(int64_t packet_number) {
>+    DataSize size =
>+        BandwidthSamplerPeer::GetPacketSize(sampler_, packet_number);
>+    bytes_in_flight_ -= size;
>+    return sampler_.OnPacketAcknowledged(clock_, packet_number);
>+  }
>+
>+  // Acknowledge receipt of a packet and expect it to be not app-limited.
>+  DataRate AckPacket(int64_t packet_number) {
>+    BandwidthSample sample = AckPacketInner(packet_number);
>+    EXPECT_FALSE(sample.is_app_limited);
>+    return sample.bandwidth;
>+  }
>+
>+  void LosePacket(int64_t packet_number) {
>+    DataSize size =
>+        BandwidthSamplerPeer::GetPacketSize(sampler_, packet_number);
>+    bytes_in_flight_ -= size;
>+    sampler_.OnPacketLost(packet_number);
>+  }
>+
>+  // Sends one packet and acks it.  Then, send 20 packets.  Finally, send
>+  // another 20 packets while acknowledging previous 20.
>+  void Send40PacketsAndAckFirst20(TimeDelta time_between_packets) {
>+    // Send 20 packets at a constant inter-packet time.
>+    for (int64_t i = 1; i <= 20; i++) {
>+      SendPacket(i);
>+      clock_ += time_between_packets;
>+    }
>+
>+    // Ack packets 1 to 20, while sending new packets at the same rate as
>+    // before.
>+    for (int64_t i = 1; i <= 20; i++) {
>+      AckPacket(i);
>+      SendPacket(i + 20);
>+      clock_ += time_between_packets;
>+    }
>+  }
>+};
>+
>+// Test the sampler in a simple stop-and-wait sender setting.
>+TEST_F(BandwidthSamplerTest, SendAndWait) {
>+  TimeDelta time_between_packets = TimeDelta::ms(10);
>+  DataRate expected_bandwidth =
>+      kRegularPacketSize * 100 / TimeDelta::seconds(1);
>+
>+  // Send packets at the constant bandwidth.
>+  for (int64_t i = 1; i < 20; i++) {
>+    SendPacket(i);
>+    clock_ += time_between_packets;
>+    DataRate current_sample = AckPacket(i);
>+    EXPECT_EQ(expected_bandwidth, current_sample);
>+  }
>+
>+  // Send packets at the exponentially decreasing bandwidth.
>+  for (int64_t i = 20; i < 25; i++) {
>+    time_between_packets = time_between_packets * 2;
>+    expected_bandwidth = expected_bandwidth * 0.5;
>+
>+    SendPacket(i);
>+    clock_ += time_between_packets;
>+    DataRate current_sample = AckPacket(i);
>+    EXPECT_EQ(expected_bandwidth, current_sample);
>+  }
>+  EXPECT_EQ(0u, BandwidthSamplerPeer::GetNumberOfTrackedPackets(sampler_));
>+  EXPECT_TRUE(bytes_in_flight_.IsZero());
>+}
>+
>+// Test the sampler during regular windowed sender scenario with fixed
>+// CWND of 20.
>+TEST_F(BandwidthSamplerTest, SendPaced) {
>+  const TimeDelta time_between_packets = TimeDelta::ms(1);
>+  DataRate expected_bandwidth = kRegularPacketSize / time_between_packets;
>+
>+  Send40PacketsAndAckFirst20(time_between_packets);
>+
>+  // Ack the packets 21 to 40, arriving at the correct bandwidth.
>+  DataRate last_bandwidth = DataRate::Zero();
>+  for (int64_t i = 21; i <= 40; i++) {
>+    last_bandwidth = AckPacket(i);
>+    EXPECT_EQ(expected_bandwidth, last_bandwidth);
>+    clock_ += time_between_packets;
>+  }
>+  EXPECT_EQ(0u, BandwidthSamplerPeer::GetNumberOfTrackedPackets(sampler_));
>+  EXPECT_TRUE(bytes_in_flight_.IsZero());
>+}
>+
>+// Test the sampler in a scenario where 50% of packets is consistently lost.
>+TEST_F(BandwidthSamplerTest, SendWithLosses) {
>+  const TimeDelta time_between_packets = TimeDelta::ms(1);
>+  DataRate expected_bandwidth = kRegularPacketSize / time_between_packets * 0.5;
>+
>+  // Send 20 packets, each 1 ms apart.
>+  for (int64_t i = 1; i <= 20; i++) {
>+    SendPacket(i);
>+    clock_ += time_between_packets;
>+  }
>+
>+  // Ack packets 1 to 20, losing every even-numbered packet, while sending new
>+  // packets at the same rate as before.
>+  for (int64_t i = 1; i <= 20; i++) {
>+    if (i % 2 == 0) {
>+      AckPacket(i);
>+    } else {
>+      LosePacket(i);
>+    }
>+    SendPacket(i + 20);
>+    clock_ += time_between_packets;
>+  }
>+
>+  // Ack the packets 21 to 40 with the same loss pattern.
>+  DataRate last_bandwidth = DataRate::Zero();
>+  for (int64_t i = 21; i <= 40; i++) {
>+    if (i % 2 == 0) {
>+      last_bandwidth = AckPacket(i);
>+      EXPECT_EQ(expected_bandwidth, last_bandwidth);
>+    } else {
>+      LosePacket(i);
>+    }
>+    clock_ += time_between_packets;
>+  }
>+  EXPECT_EQ(0u, BandwidthSamplerPeer::GetNumberOfTrackedPackets(sampler_));
>+  EXPECT_TRUE(bytes_in_flight_.IsZero());
>+}
>+
>+// Simulate a situation where ACKs arrive in burst and earlier than usual, thus
>+// producing an ACK rate which is higher than the original send rate.
>+TEST_F(BandwidthSamplerTest, CompressedAck) {
>+  const TimeDelta time_between_packets = TimeDelta::ms(1);
>+  DataRate expected_bandwidth = kRegularPacketSize / time_between_packets;
>+
>+  Send40PacketsAndAckFirst20(time_between_packets);
>+
>+  // Simulate an RTT somewhat lower than the one for 1-to-21 transmission.
>+  clock_ += time_between_packets * 15;
>+
>+  // Ack the packets 21 to 40 almost immediately at once.
>+  DataRate last_bandwidth = DataRate::Zero();
>+  TimeDelta ridiculously_small_time_delta = TimeDelta::us(20);
>+  for (int64_t i = 21; i <= 40; i++) {
>+    last_bandwidth = AckPacket(i);
>+    clock_ += ridiculously_small_time_delta;
>+  }
>+  EXPECT_EQ(expected_bandwidth, last_bandwidth);
>+  EXPECT_EQ(0u, BandwidthSamplerPeer::GetNumberOfTrackedPackets(sampler_));
>+  EXPECT_TRUE(bytes_in_flight_.IsZero());
>+}
>+
>+// Tests receiving ACK packets in the reverse order.
>+TEST_F(BandwidthSamplerTest, ReorderedAck) {
>+  const TimeDelta time_between_packets = TimeDelta::ms(1);
>+  DataRate expected_bandwidth = kRegularPacketSize / time_between_packets;
>+
>+  Send40PacketsAndAckFirst20(time_between_packets);
>+
>+  // Ack the packets 21 to 40 in the reverse order, while sending packets 41 to
>+  // 60.
>+  DataRate last_bandwidth = DataRate::Zero();
>+  for (int64_t i = 0; i < 20; i++) {
>+    last_bandwidth = AckPacket(40 - i);
>+    EXPECT_EQ(expected_bandwidth, last_bandwidth);
>+    SendPacket(41 + i);
>+    clock_ += time_between_packets;
>+  }
>+
>+  // Ack the packets 41 to 60, now in the regular order.
>+  for (int64_t i = 41; i <= 60; i++) {
>+    last_bandwidth = AckPacket(i);
>+    EXPECT_EQ(expected_bandwidth, last_bandwidth);
>+    clock_ += time_between_packets;
>+  }
>+  EXPECT_EQ(0u, BandwidthSamplerPeer::GetNumberOfTrackedPackets(sampler_));
>+  EXPECT_TRUE(bytes_in_flight_.IsZero());
>+}
>+
>+// Test the app-limited logic.
>+TEST_F(BandwidthSamplerTest, AppLimited) {
>+  const TimeDelta time_between_packets = TimeDelta::ms(1);
>+  DataRate expected_bandwidth = kRegularPacketSize / time_between_packets;
>+
>+  Send40PacketsAndAckFirst20(time_between_packets);
>+
>+  // We are now app-limited. Ack 21 to 40 as usual, but do not send anything for
>+  // now.
>+  sampler_.OnAppLimited();
>+  for (int64_t i = 21; i <= 40; i++) {
>+    DataRate current_sample = AckPacket(i);
>+    EXPECT_EQ(expected_bandwidth, current_sample);
>+    clock_ += time_between_packets;
>+  }
>+
>+  // Enter quiescence.
>+  clock_ += TimeDelta::seconds(1);
>+
>+  // Send packets 41 to 60, all of which would be marked as app-limited.
>+  for (int64_t i = 41; i <= 60; i++) {
>+    SendPacket(i);
>+    clock_ += time_between_packets;
>+  }
>+
>+  // Ack packets 41 to 60, while sending packets 61 to 80.  41 to 60 should be
>+  // app-limited and underestimate the bandwidth due to that.
>+  for (int64_t i = 41; i <= 60; i++) {
>+    BandwidthSample sample = AckPacketInner(i);
>+    EXPECT_TRUE(sample.is_app_limited);
>+    EXPECT_LT(sample.bandwidth, 0.7f * expected_bandwidth);
>+
>+    SendPacket(i + 20);
>+    clock_ += time_between_packets;
>+  }
>+
>+  // Run out of packets, and then ack packet 61 to 80, all of which should have
>+  // correct non-app-limited samples.
>+  for (int64_t i = 61; i <= 80; i++) {
>+    DataRate last_bandwidth = AckPacket(i);
>+    EXPECT_EQ(expected_bandwidth, last_bandwidth);
>+    clock_ += time_between_packets;
>+  }
>+
>+  EXPECT_EQ(0u, BandwidthSamplerPeer::GetNumberOfTrackedPackets(sampler_));
>+  EXPECT_TRUE(bytes_in_flight_.IsZero());
>+}
>+
>+// Test the samples taken at the first flight of packets sent.
>+TEST_F(BandwidthSamplerTest, FirstRoundTrip) {
>+  const TimeDelta time_between_packets = TimeDelta::ms(1);
>+  const TimeDelta rtt = TimeDelta::ms(800);
>+  const int num_packets = 10;
>+  const DataSize num_bytes = kRegularPacketSize * num_packets;
>+  const DataRate real_bandwidth = num_bytes / rtt;
>+
>+  for (int64_t i = 1; i <= 10; i++) {
>+    SendPacket(i);
>+    clock_ += time_between_packets;
>+  }
>+
>+  clock_ += rtt - num_packets * time_between_packets;
>+
>+  DataRate last_sample = DataRate::Zero();
>+  for (int64_t i = 1; i <= 10; i++) {
>+    DataRate sample = AckPacket(i);
>+    EXPECT_GT(sample, last_sample);
>+    last_sample = sample;
>+    clock_ += time_between_packets;
>+  }
>+
>+  // The final measured sample for the first flight of sample is expected to be
>+  // smaller than the real bandwidth, yet it should not lose more than 10%. The
>+  // specific value of the error depends on the difference between the RTT and
>+  // the time it takes to exhaust the congestion window (i.e. in the limit when
>+  // all packets are sent simultaneously, last sample would indicate the real
>+  // bandwidth).
>+  EXPECT_LT(last_sample, real_bandwidth);
>+  EXPECT_GT(last_sample, 0.9f * real_bandwidth);
>+}
>+
>+// Test sampler's ability to remove obsolete packets.
>+TEST_F(BandwidthSamplerTest, RemoveObsoletePackets) {
>+  SendPacket(1);
>+  SendPacket(2);
>+  SendPacket(3);
>+  SendPacket(4);
>+  SendPacket(5);
>+
>+  clock_ += TimeDelta::ms(100);
>+
>+  EXPECT_EQ(5u, BandwidthSamplerPeer::GetNumberOfTrackedPackets(sampler_));
>+  sampler_.RemoveObsoletePackets(4);
>+  EXPECT_EQ(2u, BandwidthSamplerPeer::GetNumberOfTrackedPackets(sampler_));
>+  sampler_.OnPacketLost(4);
>+  EXPECT_EQ(1u, BandwidthSamplerPeer::GetNumberOfTrackedPackets(sampler_));
>+  AckPacket(5);
>+  EXPECT_EQ(0u, BandwidthSamplerPeer::GetNumberOfTrackedPackets(sampler_));
>+}
>+
>+}  // namespace test
>+}  // namespace bbr
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/bbr_factory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/bbr_factory.cc
>new file mode 100644
>index 00000000000..bf0419b36e2
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/bbr_factory.cc
>@@ -0,0 +1,30 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/congestion_controller/bbr/bbr_factory.h"
>+#include <memory>
>+
>+#include "absl/memory/memory.h"
>+#include "modules/congestion_controller/bbr/bbr_network_controller.h"
>+
>+namespace webrtc {
>+
>+BbrNetworkControllerFactory::BbrNetworkControllerFactory() {}
>+
>+std::unique_ptr<NetworkControllerInterface> BbrNetworkControllerFactory::Create(
>+    NetworkControllerConfig config) {
>+  return absl::make_unique<bbr::BbrNetworkController>(config);
>+}
>+
>+TimeDelta BbrNetworkControllerFactory::GetProcessInterval() const {
>+  return TimeDelta::PlusInfinity();
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/bbr_factory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/bbr_factory.h
>new file mode 100644
>index 00000000000..ad539c81c40
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/bbr_factory.h
>@@ -0,0 +1,29 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_CONGESTION_CONTROLLER_BBR_BBR_FACTORY_H_
>+#define MODULES_CONGESTION_CONTROLLER_BBR_BBR_FACTORY_H_
>+
>+#include <memory>
>+
>+#include "api/transport/network_control.h"
>+
>+namespace webrtc {
>+
>+class BbrNetworkControllerFactory : public NetworkControllerFactoryInterface {
>+ public:
>+  BbrNetworkControllerFactory();
>+  std::unique_ptr<NetworkControllerInterface> Create(
>+      NetworkControllerConfig config) override;
>+  TimeDelta GetProcessInterval() const override;
>+};
>+}  // namespace webrtc
>+
>+#endif  // MODULES_CONGESTION_CONTROLLER_BBR_BBR_FACTORY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/bbr_network_controller.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/bbr_network_controller.cc
>new file mode 100644
>index 00000000000..890b2855024
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/bbr_network_controller.cc
>@@ -0,0 +1,947 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/congestion_controller/bbr/bbr_network_controller.h"
>+
>+#include <algorithm>
>+#include <array>
>+#include <string>
>+#include <vector>
>+
>+#include "rtc_base/checks.h"
>+#include "rtc_base/logging.h"
>+#include "rtc_base/system/fallthrough.h"
>+#include "system_wrappers/include/field_trial.h"
>+
>+namespace webrtc {
>+namespace bbr {
>+namespace {
>+
>+// If greater than zero, mean RTT variation is multiplied by the specified
>+// factor and added to the congestion window limit.
>+const double kBbrRttVariationWeight = 0.0f;
>+
>+// Congestion window gain for QUIC BBR during PROBE_BW phase.
>+const double kProbeBWCongestionWindowGain = 2.0f;
>+
>+// The maximum packet size of any QUIC packet, based on ethernet's max size,
>+// minus the IP and UDP headers. IPv6 has a 40 byte header, UDP adds an
>+// additional 8 bytes.  This is a total overhead of 48 bytes.  Ethernet's
>+// max packet size is 1500 bytes,  1500 - 48 = 1452.
>+const DataSize kMaxPacketSize = DataSize::bytes(1452);
>+
>+// Default maximum packet size used in the Linux TCP implementation.
>+// Used in QUIC for congestion window computations in bytes.
>+const DataSize kDefaultTCPMSS = DataSize::bytes(1460);
>+// Constants based on TCP defaults.
>+const DataSize kMaxSegmentSize = kDefaultTCPMSS;
>+
>+// The gain used for the slow start, equal to 2/ln(2).
>+const double kHighGain = 2.885f;
>+// The gain used in STARTUP after loss has been detected.
>+// 1.5 is enough to allow for 25% exogenous loss and still observe a 25% growth
>+// in measured bandwidth.
>+const double kStartupAfterLossGain = 1.5;
>+// The gain used to drain the queue after the slow start.
>+const double kDrainGain = 1.f / kHighGain;
>+
>+// The length of the gain cycle.
>+const size_t kGainCycleLength = 8;
>+// The size of the bandwidth filter window, in round-trips.
>+const BbrRoundTripCount kBandwidthWindowSize = kGainCycleLength + 2;
>+
>+// The time after which the current min_rtt value expires.
>+constexpr int64_t kMinRttExpirySeconds = 10;
>+// The minimum time the connection can spend in PROBE_RTT mode.
>+constexpr int64_t kProbeRttTimeMs = 200;
>+// If the bandwidth does not increase by the factor of |kStartupGrowthTarget|
>+// within |kRoundTripsWithoutGrowthBeforeExitingStartup| rounds, the connection
>+// will exit the STARTUP mode.
>+const double kStartupGrowthTarget = 1.25;
>+// Coefficient to determine if a new RTT is sufficiently similar to min_rtt that
>+// we don't need to enter PROBE_RTT.
>+const double kSimilarMinRttThreshold = 1.125;
>+
>+constexpr int64_t kInitialBandwidthKbps = 300;
>+
>+const int64_t kInitialCongestionWindowPackets = 32;
>+// The minimum CWND to ensure delayed acks don't reduce bandwidth measurements.
>+// Does not inflate the pacing rate.
>+const int64_t kDefaultMinCongestionWindowPackets = 4;
>+const int64_t kDefaultMaxCongestionWindowPackets = 2000;
>+
>+const char kBbrConfigTrial[] = "WebRTC-BweBbrConfig";
>+
>+}  // namespace
>+
>+BbrNetworkController::BbrControllerConfig::BbrControllerConfig(
>+    std::string field_trial)
>+    : probe_bw_pacing_gain_offset("probe_bw_pacing_gain_offset", 0.25),
>+      encoder_rate_gain("encoder_rate_gain", 1),
>+      encoder_rate_gain_in_probe_rtt("encoder_rate_gain_in_probe_rtt", 1),
>+      exit_startup_rtt_threshold("exit_startup_rtt_threshold",
>+                                 TimeDelta::PlusInfinity()),
>+      initial_congestion_window(
>+          "initial_cwin",
>+          kInitialCongestionWindowPackets * kDefaultTCPMSS),
>+      min_congestion_window(
>+          "min_cwin",
>+          kDefaultMinCongestionWindowPackets * kDefaultTCPMSS),
>+      max_congestion_window(
>+          "max_cwin",
>+          kDefaultMaxCongestionWindowPackets * kDefaultTCPMSS),
>+      probe_rtt_congestion_window_gain("probe_rtt_cwin_gain", 0.75),
>+      pacing_rate_as_target("pacing_rate_as_target", false),
>+      exit_startup_on_loss("exit_startup_on_loss", true),
>+      num_startup_rtts("num_startup_rtts", 3),
>+      rate_based_recovery("rate_based_recovery", false),
>+      max_aggregation_bytes_multiplier("max_aggregation_bytes_multiplier", 0),
>+      slower_startup("slower_startup", false),
>+      rate_based_startup("rate_based_startup", false),
>+      initial_conservation_in_startup("initial_conservation",
>+                                      CONSERVATION,
>+                                      {
>+                                          {"NOT_IN_RECOVERY", NOT_IN_RECOVERY},
>+                                          {"CONSERVATION", CONSERVATION},
>+                                          {"MEDIUM_GROWTH", MEDIUM_GROWTH},
>+                                          {"GROWTH", GROWTH},
>+                                      }),
>+      fully_drain_queue("fully_drain_queue", false),
>+      max_ack_height_window_multiplier("max_ack_height_window_multiplier", 1),
>+      probe_rtt_based_on_bdp("probe_rtt_based_on_bdp", false),
>+      probe_rtt_skipped_if_similar_rtt("probe_rtt_skipped_if_similar_rtt",
>+                                       false),
>+      probe_rtt_disabled_if_app_limited("probe_rtt_disabled_if_app_limited",
>+                                        false) {
>+  ParseFieldTrial(
>+      {
>+          &exit_startup_on_loss,
>+          &encoder_rate_gain,
>+          &encoder_rate_gain_in_probe_rtt,
>+          &exit_startup_rtt_threshold,
>+          &fully_drain_queue,
>+          &initial_congestion_window,
>+          &initial_conservation_in_startup,
>+          &max_ack_height_window_multiplier,
>+          &max_aggregation_bytes_multiplier,
>+          &max_congestion_window,
>+          &min_congestion_window,
>+          &num_startup_rtts,
>+          &pacing_rate_as_target,
>+          &probe_bw_pacing_gain_offset,
>+          &probe_rtt_based_on_bdp,
>+          &probe_rtt_congestion_window_gain,
>+          &probe_rtt_disabled_if_app_limited,
>+          &probe_rtt_skipped_if_similar_rtt,
>+          &rate_based_recovery,
>+          &rate_based_startup,
>+          &slower_startup,
>+      },
>+      field_trial);
>+}
>+BbrNetworkController::BbrControllerConfig::~BbrControllerConfig() = default;
>+BbrNetworkController::BbrControllerConfig::BbrControllerConfig(
>+    const BbrControllerConfig&) = default;
>+BbrNetworkController::BbrControllerConfig
>+BbrNetworkController::BbrControllerConfig::FromTrial() {
>+  return BbrControllerConfig(
>+      webrtc::field_trial::FindFullName(kBbrConfigTrial));
>+}
>+
>+
>+BbrNetworkController::DebugState::DebugState(const BbrNetworkController& sender)
>+    : mode(sender.mode_),
>+      max_bandwidth(sender.max_bandwidth_.GetBest()),
>+      round_trip_count(sender.round_trip_count_),
>+      gain_cycle_index(sender.cycle_current_offset_),
>+      congestion_window(sender.congestion_window_),
>+      is_at_full_bandwidth(sender.is_at_full_bandwidth_),
>+      bandwidth_at_last_round(sender.bandwidth_at_last_round_),
>+      rounds_without_bandwidth_gain(sender.rounds_without_bandwidth_gain_),
>+      min_rtt(sender.min_rtt_),
>+      min_rtt_timestamp(sender.min_rtt_timestamp_),
>+      recovery_state(sender.recovery_state_),
>+      recovery_window(sender.recovery_window_),
>+      last_sample_is_app_limited(sender.last_sample_is_app_limited_),
>+      end_of_app_limited_phase(sender.sampler_->end_of_app_limited_phase()) {}
>+
>+BbrNetworkController::DebugState::DebugState(const DebugState& state) = default;
>+
>+BbrNetworkController::BbrNetworkController(NetworkControllerConfig config)
>+    : config_(BbrControllerConfig::FromTrial()),
>+      rtt_stats_(),
>+      random_(10),
>+      loss_rate_(),
>+      mode_(STARTUP),
>+      sampler_(new BandwidthSampler()),
>+      round_trip_count_(0),
>+      last_sent_packet_(0),
>+      current_round_trip_end_(0),
>+      max_bandwidth_(kBandwidthWindowSize, DataRate::Zero(), 0),
>+      default_bandwidth_(DataRate::kbps(kInitialBandwidthKbps)),
>+      max_ack_height_(kBandwidthWindowSize, DataSize::Zero(), 0),
>+      aggregation_epoch_start_time_(),
>+      aggregation_epoch_bytes_(DataSize::Zero()),
>+      bytes_acked_since_queue_drained_(DataSize::Zero()),
>+      max_aggregation_bytes_multiplier_(0),
>+      min_rtt_(TimeDelta::Zero()),
>+      last_rtt_(TimeDelta::Zero()),
>+      min_rtt_timestamp_(Timestamp::ms(0)),
>+      congestion_window_(config_.initial_congestion_window),
>+      initial_congestion_window_(config_.initial_congestion_window),
>+      min_congestion_window_(config_.min_congestion_window),
>+      max_congestion_window_(config_.max_congestion_window),
>+      pacing_rate_(DataRate::Zero()),
>+      pacing_gain_(1),
>+      congestion_window_gain_constant_(kProbeBWCongestionWindowGain),
>+      rtt_variance_weight_(kBbrRttVariationWeight),
>+      cycle_current_offset_(0),
>+      last_cycle_start_(Timestamp::ms(0)),
>+      is_at_full_bandwidth_(false),
>+      rounds_without_bandwidth_gain_(0),
>+      bandwidth_at_last_round_(DataRate::Zero()),
>+      exiting_quiescence_(false),
>+      exit_probe_rtt_at_(),
>+      probe_rtt_round_passed_(false),
>+      last_sample_is_app_limited_(false),
>+      recovery_state_(NOT_IN_RECOVERY),
>+      end_recovery_at_(),
>+      recovery_window_(max_congestion_window_),
>+      app_limited_since_last_probe_rtt_(false),
>+      min_rtt_since_last_probe_rtt_(TimeDelta::PlusInfinity()) {
>+  RTC_LOG(LS_INFO) << "Creating BBR controller";
>+  if (config.starting_bandwidth.IsFinite())
>+    default_bandwidth_ = config.starting_bandwidth;
>+  constraints_ = config.constraints;
>+  Reset();
>+}
>+
>+BbrNetworkController::~BbrNetworkController() {}
>+
>+void BbrNetworkController::Reset() {
>+  round_trip_count_ = 0;
>+  rounds_without_bandwidth_gain_ = 0;
>+  if (config_.num_startup_rtts > 0) {
>+    is_at_full_bandwidth_ = false;
>+    EnterStartupMode();
>+  } else {
>+    is_at_full_bandwidth_ = true;
>+    EnterProbeBandwidthMode(constraints_->at_time);
>+  }
>+}
>+
>+NetworkControlUpdate BbrNetworkController::CreateRateUpdate(
>+    Timestamp at_time) const {
>+  DataRate bandwidth = BandwidthEstimate();
>+  if (bandwidth.IsZero())
>+    bandwidth = default_bandwidth_;
>+  TimeDelta rtt = GetMinRtt();
>+  DataRate pacing_rate = PacingRate();
>+  DataRate target_rate =
>+      config_.pacing_rate_as_target ? pacing_rate : bandwidth;
>+
>+  if (mode_ == PROBE_RTT)
>+    target_rate = target_rate * config_.encoder_rate_gain_in_probe_rtt;
>+  else
>+    target_rate = target_rate * config_.encoder_rate_gain;
>+  target_rate = std::min(target_rate, pacing_rate);
>+
>+  if (constraints_) {
>+    if (constraints_->max_data_rate) {
>+      target_rate = std::min(target_rate, *constraints_->max_data_rate);
>+      pacing_rate = std::min(pacing_rate, *constraints_->max_data_rate);
>+    }
>+    if (constraints_->min_data_rate) {
>+      target_rate = std::max(target_rate, *constraints_->min_data_rate);
>+      pacing_rate = std::max(pacing_rate, *constraints_->min_data_rate);
>+    }
>+  }
>+
>+  NetworkControlUpdate update;
>+
>+  TargetTransferRate target_rate_msg;
>+  target_rate_msg.network_estimate.at_time = at_time;
>+  target_rate_msg.network_estimate.bandwidth = bandwidth;
>+  target_rate_msg.network_estimate.round_trip_time = rtt;
>+
>+  // TODO(srte): Fill in field below with proper value.
>+  target_rate_msg.network_estimate.loss_rate_ratio = 0;
>+  // In in PROBE_BW, target bandwidth is expected to vary over the cycle period.
>+  // In other modes the is no given period, therefore the same value as in
>+  // PROBE_BW is used for consistency.
>+  target_rate_msg.network_estimate.bwe_period =
>+      rtt * static_cast<int64_t>(kGainCycleLength);
>+
>+  target_rate_msg.target_rate = target_rate;
>+  target_rate_msg.at_time = at_time;
>+  update.target_rate = target_rate_msg;
>+
>+  PacerConfig pacer_config;
>+  // A small time window ensures an even pacing rate.
>+  pacer_config.time_window = rtt * 0.25;
>+  pacer_config.data_window = pacer_config.time_window * pacing_rate;
>+
>+  if (IsProbingForMoreBandwidth())
>+    pacer_config.pad_window = pacer_config.data_window;
>+  else
>+    pacer_config.pad_window = DataSize::Zero();
>+
>+  pacer_config.at_time = at_time;
>+  update.pacer_config = pacer_config;
>+
>+  update.congestion_window = GetCongestionWindow();
>+  return update;
>+}
>+
>+NetworkControlUpdate BbrNetworkController::OnNetworkAvailability(
>+    NetworkAvailability msg) {
>+  Reset();
>+  rtt_stats_.OnConnectionMigration();
>+  return CreateRateUpdate(msg.at_time);
>+}
>+
>+NetworkControlUpdate BbrNetworkController::OnNetworkRouteChange(
>+    NetworkRouteChange msg) {
>+  constraints_ = msg.constraints;
>+  Reset();
>+  if (msg.starting_rate)
>+    default_bandwidth_ = *msg.starting_rate;
>+
>+  rtt_stats_.OnConnectionMigration();
>+  return CreateRateUpdate(msg.at_time);
>+}
>+
>+NetworkControlUpdate BbrNetworkController::OnProcessInterval(
>+    ProcessInterval msg) {
>+  return CreateRateUpdate(msg.at_time);
>+}
>+
>+NetworkControlUpdate BbrNetworkController::OnStreamsConfig(StreamsConfig msg) {
>+  return NetworkControlUpdate();
>+}
>+
>+NetworkControlUpdate BbrNetworkController::OnTargetRateConstraints(
>+    TargetRateConstraints msg) {
>+  constraints_ = msg;
>+  return CreateRateUpdate(msg.at_time);
>+}
>+
>+bool BbrNetworkController::InSlowStart() const {
>+  return mode_ == STARTUP;
>+}
>+
>+NetworkControlUpdate BbrNetworkController::OnSentPacket(SentPacket msg) {
>+  last_sent_packet_ = msg.sequence_number;
>+
>+  if (msg.data_in_flight.IsZero() && sampler_->is_app_limited()) {
>+    exiting_quiescence_ = true;
>+  }
>+
>+  if (!aggregation_epoch_start_time_) {
>+    aggregation_epoch_start_time_ = msg.send_time;
>+  }
>+
>+  sampler_->OnPacketSent(msg.send_time, msg.sequence_number, msg.size,
>+                         msg.data_in_flight);
>+  return NetworkControlUpdate();
>+}
>+
>+bool BbrNetworkController::CanSend(DataSize bytes_in_flight) {
>+  return bytes_in_flight < GetCongestionWindow();
>+}
>+
>+DataRate BbrNetworkController::PacingRate() const {
>+  if (pacing_rate_.IsZero()) {
>+    return kHighGain * initial_congestion_window_ / GetMinRtt();
>+  }
>+  return pacing_rate_;
>+}
>+
>+DataRate BbrNetworkController::BandwidthEstimate() const {
>+  return max_bandwidth_.GetBest();
>+}
>+
>+DataSize BbrNetworkController::GetCongestionWindow() const {
>+  if (mode_ == PROBE_RTT) {
>+    return ProbeRttCongestionWindow();
>+  }
>+
>+  if (InRecovery() && !config_.rate_based_recovery &&
>+      !(config_.rate_based_startup && mode_ == STARTUP)) {
>+    return std::min(congestion_window_, recovery_window_);
>+  }
>+
>+  return congestion_window_;
>+}
>+
>+double BbrNetworkController::GetPacingGain(int round_offset) const {
>+  if (round_offset == 0)
>+    return 1 + config_.probe_bw_pacing_gain_offset;
>+  else if (round_offset == 1)
>+    return 1 - config_.probe_bw_pacing_gain_offset;
>+  else
>+    return 1;
>+}
>+
>+bool BbrNetworkController::InRecovery() const {
>+  return recovery_state_ != NOT_IN_RECOVERY;
>+}
>+
>+bool BbrNetworkController::IsProbingForMoreBandwidth() const {
>+  return (mode_ == PROBE_BW && pacing_gain_ > 1) || mode_ == STARTUP;
>+}
>+
>+NetworkControlUpdate BbrNetworkController::OnTransportPacketsFeedback(
>+    TransportPacketsFeedback msg) {
>+  Timestamp feedback_recv_time = msg.feedback_time;
>+  absl::optional<SentPacket> last_sent_packet =
>+      msg.PacketsWithFeedback().back().sent_packet;
>+  if (!last_sent_packet.has_value()) {
>+    RTC_LOG(LS_WARNING) << "Last ack packet not in history, no RTT update";
>+  } else {
>+    Timestamp send_time = last_sent_packet->send_time;
>+    TimeDelta send_delta = feedback_recv_time - send_time;
>+    rtt_stats_.UpdateRtt(send_delta, TimeDelta::Zero(), feedback_recv_time);
>+  }
>+
>+  const DataSize total_data_acked_before = sampler_->total_data_acked();
>+
>+  bool is_round_start = false;
>+  bool min_rtt_expired = false;
>+
>+  std::vector<PacketResult> lost_packets = msg.LostWithSendInfo();
>+  DiscardLostPackets(lost_packets);
>+
>+  std::vector<PacketResult> acked_packets = msg.ReceivedWithSendInfo();
>+
>+  int packets_sent =
>+      static_cast<int>(lost_packets.size() + acked_packets.size());
>+  int packets_lost = static_cast<int>(lost_packets.size());
>+  loss_rate_.UpdateWithLossStatus(msg.feedback_time.ms(), packets_sent,
>+                                  packets_lost);
>+
>+  // Input the new data into the BBR model of the connection.
>+  if (!acked_packets.empty()) {
>+    int64_t last_acked_packet =
>+        acked_packets.rbegin()->sent_packet->sequence_number;
>+
>+    is_round_start = UpdateRoundTripCounter(last_acked_packet);
>+    min_rtt_expired =
>+        UpdateBandwidthAndMinRtt(msg.feedback_time, acked_packets);
>+    UpdateRecoveryState(last_acked_packet, !lost_packets.empty(),
>+                        is_round_start);
>+
>+    const DataSize data_acked =
>+        sampler_->total_data_acked() - total_data_acked_before;
>+
>+    UpdateAckAggregationBytes(msg.feedback_time, data_acked);
>+    if (max_aggregation_bytes_multiplier_ > 0) {
>+      if (msg.data_in_flight <=
>+          1.25 * GetTargetCongestionWindow(pacing_gain_)) {
>+        bytes_acked_since_queue_drained_ = DataSize::Zero();
>+      } else {
>+        bytes_acked_since_queue_drained_ += data_acked;
>+      }
>+    }
>+  }
>+
>+  // Handle logic specific to PROBE_BW mode.
>+  if (mode_ == PROBE_BW) {
>+    UpdateGainCyclePhase(msg.feedback_time, msg.prior_in_flight,
>+                         !lost_packets.empty());
>+  }
>+
>+  // Handle logic specific to STARTUP and DRAIN modes.
>+  if (is_round_start && !is_at_full_bandwidth_) {
>+    CheckIfFullBandwidthReached();
>+  }
>+  MaybeExitStartupOrDrain(msg);
>+
>+  // Handle logic specific to PROBE_RTT.
>+  MaybeEnterOrExitProbeRtt(msg, is_round_start, min_rtt_expired);
>+
>+  // Calculate number of packets acked and lost.
>+  DataSize data_acked = sampler_->total_data_acked() - total_data_acked_before;
>+  DataSize data_lost = DataSize::Zero();
>+  for (const PacketResult& packet : lost_packets) {
>+    data_lost += packet.sent_packet->size;
>+  }
>+
>+  // After the model is updated, recalculate the pacing rate and congestion
>+  // window.
>+  CalculatePacingRate();
>+  CalculateCongestionWindow(data_acked);
>+  CalculateRecoveryWindow(data_acked, data_lost, msg.data_in_flight);
>+  // Cleanup internal state.
>+  if (!acked_packets.empty()) {
>+    sampler_->RemoveObsoletePackets(
>+        acked_packets.back().sent_packet->sequence_number);
>+  }
>+  return CreateRateUpdate(msg.feedback_time);
>+}
>+
>+NetworkControlUpdate BbrNetworkController::OnRemoteBitrateReport(
>+    RemoteBitrateReport msg) {
>+  return NetworkControlUpdate();
>+}
>+NetworkControlUpdate BbrNetworkController::OnRoundTripTimeUpdate(
>+    RoundTripTimeUpdate msg) {
>+  return NetworkControlUpdate();
>+}
>+NetworkControlUpdate BbrNetworkController::OnTransportLossReport(
>+    TransportLossReport msg) {
>+  return NetworkControlUpdate();
>+}
>+
>+TimeDelta BbrNetworkController::GetMinRtt() const {
>+  return !min_rtt_.IsZero() ? min_rtt_
>+                            : TimeDelta::us(rtt_stats_.initial_rtt_us());
>+}
>+
>+DataSize BbrNetworkController::GetTargetCongestionWindow(double gain) const {
>+  DataSize bdp = GetMinRtt() * BandwidthEstimate();
>+  DataSize congestion_window = gain * bdp;
>+
>+  // BDP estimate will be zero if no bandwidth samples are available yet.
>+  if (congestion_window.IsZero()) {
>+    congestion_window = gain * initial_congestion_window_;
>+  }
>+
>+  return std::max(congestion_window, min_congestion_window_);
>+}
>+
>+DataSize BbrNetworkController::ProbeRttCongestionWindow() const {
>+  if (config_.probe_rtt_based_on_bdp) {
>+    return GetTargetCongestionWindow(config_.probe_rtt_congestion_window_gain);
>+  }
>+  return min_congestion_window_;
>+}
>+
>+void BbrNetworkController::EnterStartupMode() {
>+  mode_ = STARTUP;
>+  pacing_gain_ = kHighGain;
>+  congestion_window_gain_ = kHighGain;
>+}
>+
>+void BbrNetworkController::EnterProbeBandwidthMode(Timestamp now) {
>+  mode_ = PROBE_BW;
>+  congestion_window_gain_ = congestion_window_gain_constant_;
>+
>+  // Pick a random offset for the gain cycle out of {0, 2..7} range. 1 is
>+  // excluded because in that case increased gain and decreased gain would not
>+  // follow each other.
>+  cycle_current_offset_ = random_.Rand(kGainCycleLength - 2);
>+  if (cycle_current_offset_ >= 1) {
>+    cycle_current_offset_ += 1;
>+  }
>+
>+  last_cycle_start_ = now;
>+  pacing_gain_ = GetPacingGain(cycle_current_offset_);
>+}
>+
>+void BbrNetworkController::DiscardLostPackets(
>+    const std::vector<PacketResult>& lost_packets) {
>+  for (const PacketResult& packet : lost_packets) {
>+    sampler_->OnPacketLost(packet.sent_packet->sequence_number);
>+  }
>+}
>+
>+bool BbrNetworkController::UpdateRoundTripCounter(int64_t last_acked_packet) {
>+  if (last_acked_packet > current_round_trip_end_) {
>+    round_trip_count_++;
>+    current_round_trip_end_ = last_sent_packet_;
>+    return true;
>+  }
>+
>+  return false;
>+}
>+
>+bool BbrNetworkController::UpdateBandwidthAndMinRtt(
>+    Timestamp now,
>+    const std::vector<PacketResult>& acked_packets) {
>+  TimeDelta sample_rtt = TimeDelta::PlusInfinity();
>+  for (const auto& packet : acked_packets) {
>+    BandwidthSample bandwidth_sample = sampler_->OnPacketAcknowledged(
>+        now, packet.sent_packet->sequence_number);
>+    last_sample_is_app_limited_ = bandwidth_sample.is_app_limited;
>+    if (!bandwidth_sample.rtt.IsZero()) {
>+      sample_rtt = std::min(sample_rtt, bandwidth_sample.rtt);
>+    }
>+
>+    if (!bandwidth_sample.is_app_limited ||
>+        bandwidth_sample.bandwidth > BandwidthEstimate()) {
>+      max_bandwidth_.Update(bandwidth_sample.bandwidth, round_trip_count_);
>+    }
>+  }
>+
>+  // If none of the RTT samples are valid, return immediately.
>+  if (sample_rtt.IsInfinite()) {
>+    return false;
>+  }
>+
>+  last_rtt_ = sample_rtt;
>+  min_rtt_since_last_probe_rtt_ =
>+      std::min(min_rtt_since_last_probe_rtt_, sample_rtt);
>+
>+  const TimeDelta kMinRttExpiry = TimeDelta::seconds(kMinRttExpirySeconds);
>+  // Do not expire min_rtt if none was ever available.
>+  bool min_rtt_expired =
>+      !min_rtt_.IsZero() && (now > (min_rtt_timestamp_ + kMinRttExpiry));
>+
>+  if (min_rtt_expired || sample_rtt < min_rtt_ || min_rtt_.IsZero()) {
>+    if (ShouldExtendMinRttExpiry()) {
>+      min_rtt_expired = false;
>+    } else {
>+      min_rtt_ = sample_rtt;
>+    }
>+    min_rtt_timestamp_ = now;
>+    // Reset since_last_probe_rtt fields.
>+    min_rtt_since_last_probe_rtt_ = TimeDelta::PlusInfinity();
>+    app_limited_since_last_probe_rtt_ = false;
>+  }
>+
>+  return min_rtt_expired;
>+}
>+
>+bool BbrNetworkController::ShouldExtendMinRttExpiry() const {
>+  if (config_.probe_rtt_disabled_if_app_limited &&
>+      app_limited_since_last_probe_rtt_) {
>+    // Extend the current min_rtt if we've been app limited recently.
>+    return true;
>+  }
>+  const bool min_rtt_increased_since_last_probe =
>+      min_rtt_since_last_probe_rtt_ > min_rtt_ * kSimilarMinRttThreshold;
>+  if (config_.probe_rtt_skipped_if_similar_rtt &&
>+      app_limited_since_last_probe_rtt_ &&
>+      !min_rtt_increased_since_last_probe) {
>+    // Extend the current min_rtt if we've been app limited recently and an rtt
>+    // has been measured in that time that's less than 12.5% more than the
>+    // current min_rtt.
>+    return true;
>+  }
>+  return false;
>+}
>+
>+void BbrNetworkController::UpdateGainCyclePhase(Timestamp now,
>+                                                DataSize prior_in_flight,
>+                                                bool has_losses) {
>+  // In most cases, the cycle is advanced after an RTT passes.
>+  bool should_advance_gain_cycling = now - last_cycle_start_ > GetMinRtt();
>+
>+  // If the pacing gain is above 1.0, the connection is trying to probe the
>+  // bandwidth by increasing the number of bytes in flight to at least
>+  // pacing_gain * BDP.  Make sure that it actually reaches the target, as long
>+  // as there are no losses suggesting that the buffers are not able to hold
>+  // that much.
>+  if (pacing_gain_ > 1.0 && !has_losses &&
>+      prior_in_flight < GetTargetCongestionWindow(pacing_gain_)) {
>+    should_advance_gain_cycling = false;
>+  }
>+
>+  // If pacing gain is below 1.0, the connection is trying to drain the extra
>+  // queue which could have been incurred by probing prior to it.  If the number
>+  // of bytes in flight falls down to the estimated BDP value earlier, conclude
>+  // that the queue has been successfully drained and exit this cycle early.
>+  if (pacing_gain_ < 1.0 && prior_in_flight <= GetTargetCongestionWindow(1)) {
>+    should_advance_gain_cycling = true;
>+  }
>+
>+  if (should_advance_gain_cycling) {
>+    cycle_current_offset_ = (cycle_current_offset_ + 1) % kGainCycleLength;
>+    last_cycle_start_ = now;
>+    // Stay in low gain mode until the target BDP is hit.
>+    // Low gain mode will be exited immediately when the target BDP is achieved.
>+    if (config_.fully_drain_queue && pacing_gain_ < 1 &&
>+        GetPacingGain(cycle_current_offset_) == 1 &&
>+        prior_in_flight > GetTargetCongestionWindow(1)) {
>+      return;
>+    }
>+    pacing_gain_ = GetPacingGain(cycle_current_offset_);
>+  }
>+}
>+
>+void BbrNetworkController::CheckIfFullBandwidthReached() {
>+  if (last_sample_is_app_limited_) {
>+    return;
>+  }
>+
>+  DataRate target = bandwidth_at_last_round_ * kStartupGrowthTarget;
>+  if (BandwidthEstimate() >= target) {
>+    bandwidth_at_last_round_ = BandwidthEstimate();
>+    rounds_without_bandwidth_gain_ = 0;
>+    return;
>+  }
>+
>+  rounds_without_bandwidth_gain_++;
>+  if ((rounds_without_bandwidth_gain_ >= config_.num_startup_rtts) ||
>+      (config_.exit_startup_on_loss && InRecovery())) {
>+    is_at_full_bandwidth_ = true;
>+  }
>+}
>+
>+void BbrNetworkController::MaybeExitStartupOrDrain(
>+    const TransportPacketsFeedback& msg) {
>+  TimeDelta exit_threshold = config_.exit_startup_rtt_threshold;
>+  TimeDelta rtt_delta = last_rtt_ - min_rtt_;
>+  if (mode_ == STARTUP &&
>+      (is_at_full_bandwidth_ || rtt_delta > exit_threshold)) {
>+    if (rtt_delta > exit_threshold)
>+      RTC_LOG(LS_INFO) << "Exiting startup due to rtt increase from: "
>+                       << ToString(min_rtt_) << " to:" << ToString(last_rtt_)
>+                       << " > " << ToString(min_rtt_ + exit_threshold);
>+    mode_ = DRAIN;
>+    pacing_gain_ = kDrainGain;
>+    congestion_window_gain_ = kHighGain;
>+  }
>+  if (mode_ == DRAIN && msg.data_in_flight <= GetTargetCongestionWindow(1)) {
>+    EnterProbeBandwidthMode(msg.feedback_time);
>+  }
>+}
>+
>+void BbrNetworkController::MaybeEnterOrExitProbeRtt(
>+    const TransportPacketsFeedback& msg,
>+    bool is_round_start,
>+    bool min_rtt_expired) {
>+  if (min_rtt_expired && !exiting_quiescence_ && mode_ != PROBE_RTT) {
>+    mode_ = PROBE_RTT;
>+    pacing_gain_ = 1;
>+    // Do not decide on the time to exit PROBE_RTT until the |bytes_in_flight|
>+    // is at the target small value.
>+    exit_probe_rtt_at_.reset();
>+  }
>+
>+  if (mode_ == PROBE_RTT) {
>+    sampler_->OnAppLimited();
>+
>+    if (!exit_probe_rtt_at_) {
>+      // If the window has reached the appropriate size, schedule exiting
>+      // PROBE_RTT.  The CWND during PROBE_RTT is kMinimumCongestionWindow, but
>+      // we allow an extra packet since QUIC checks CWND before sending a
>+      // packet.
>+      if (msg.data_in_flight < ProbeRttCongestionWindow() + kMaxPacketSize) {
>+        exit_probe_rtt_at_ = msg.feedback_time + TimeDelta::ms(kProbeRttTimeMs);
>+        probe_rtt_round_passed_ = false;
>+      }
>+    } else {
>+      if (is_round_start) {
>+        probe_rtt_round_passed_ = true;
>+      }
>+      if (msg.feedback_time >= *exit_probe_rtt_at_ && probe_rtt_round_passed_) {
>+        min_rtt_timestamp_ = msg.feedback_time;
>+        if (!is_at_full_bandwidth_) {
>+          EnterStartupMode();
>+        } else {
>+          EnterProbeBandwidthMode(msg.feedback_time);
>+        }
>+      }
>+    }
>+  }
>+
>+  exiting_quiescence_ = false;
>+}
>+
>+void BbrNetworkController::UpdateRecoveryState(int64_t last_acked_packet,
>+                                               bool has_losses,
>+                                               bool is_round_start) {
>+  // Exit recovery when there are no losses for a round.
>+  if (has_losses) {
>+    end_recovery_at_ = last_sent_packet_;
>+  }
>+
>+  switch (recovery_state_) {
>+    case NOT_IN_RECOVERY:
>+      // Enter conservation on the first loss.
>+      if (has_losses) {
>+        recovery_state_ = CONSERVATION;
>+        if (mode_ == STARTUP) {
>+          recovery_state_ = config_.initial_conservation_in_startup;
>+        }
>+        // This will cause the |recovery_window_| to be set to the correct
>+        // value in CalculateRecoveryWindow().
>+        recovery_window_ = DataSize::Zero();
>+        // Since the conservation phase is meant to be lasting for a whole
>+        // round, extend the current round as if it were started right now.
>+        current_round_trip_end_ = last_sent_packet_;
>+      }
>+      break;
>+
>+    case CONSERVATION:
>+    case MEDIUM_GROWTH:
>+      if (is_round_start) {
>+        recovery_state_ = GROWTH;
>+      }
>+      RTC_FALLTHROUGH();
>+    case GROWTH:
>+      // Exit recovery if appropriate.
>+      if (!has_losses &&
>+          (!end_recovery_at_ || last_acked_packet > *end_recovery_at_)) {
>+        recovery_state_ = NOT_IN_RECOVERY;
>+      }
>+
>+      break;
>+  }
>+}
>+
>+void BbrNetworkController::UpdateAckAggregationBytes(
>+    Timestamp ack_time,
>+    DataSize newly_acked_bytes) {
>+  if (!aggregation_epoch_start_time_) {
>+    RTC_LOG(LS_ERROR)
>+        << "Received feedback before information about sent packets.";
>+    RTC_DCHECK(aggregation_epoch_start_time_.has_value());
>+    return;
>+  }
>+  // Compute how many bytes are expected to be delivered, assuming max bandwidth
>+  // is correct.
>+  DataSize expected_bytes_acked =
>+      max_bandwidth_.GetBest() * (ack_time - *aggregation_epoch_start_time_);
>+  // Reset the current aggregation epoch as soon as the ack arrival rate is less
>+  // than or equal to the max bandwidth.
>+  if (aggregation_epoch_bytes_ <= expected_bytes_acked) {
>+    // Reset to start measuring a new aggregation epoch.
>+    aggregation_epoch_bytes_ = newly_acked_bytes;
>+    aggregation_epoch_start_time_ = ack_time;
>+    return;
>+  }
>+
>+  // Compute how many extra bytes were delivered vs max bandwidth.
>+  // Include the bytes most recently acknowledged to account for stretch acks.
>+  aggregation_epoch_bytes_ += newly_acked_bytes;
>+  max_ack_height_.Update(aggregation_epoch_bytes_ - expected_bytes_acked,
>+                         round_trip_count_);
>+}
>+
>+void BbrNetworkController::CalculatePacingRate() {
>+  if (BandwidthEstimate().IsZero()) {
>+    return;
>+  }
>+
>+  DataRate target_rate = pacing_gain_ * BandwidthEstimate();
>+  if (config_.rate_based_recovery && InRecovery()) {
>+    pacing_rate_ = pacing_gain_ * max_bandwidth_.GetThirdBest();
>+  }
>+  if (is_at_full_bandwidth_) {
>+    pacing_rate_ = target_rate;
>+    return;
>+  }
>+
>+  // Pace at the rate of initial_window / RTT as soon as RTT measurements are
>+  // available.
>+  if (pacing_rate_.IsZero() && !rtt_stats_.min_rtt().IsZero()) {
>+    pacing_rate_ = initial_congestion_window_ / rtt_stats_.min_rtt();
>+    return;
>+  }
>+  // Slow the pacing rate in STARTUP once loss has ever been detected.
>+  const bool has_ever_detected_loss = end_recovery_at_.has_value();
>+  if (config_.slower_startup && has_ever_detected_loss) {
>+    pacing_rate_ = kStartupAfterLossGain * BandwidthEstimate();
>+    return;
>+  }
>+
>+  // Do not decrease the pacing rate during the startup.
>+  pacing_rate_ = std::max(pacing_rate_, target_rate);
>+}
>+
>+void BbrNetworkController::CalculateCongestionWindow(DataSize bytes_acked) {
>+  if (mode_ == PROBE_RTT) {
>+    return;
>+  }
>+
>+  DataSize target_window = GetTargetCongestionWindow(congestion_window_gain_);
>+
>+  if (rtt_variance_weight_ > 0.f && !BandwidthEstimate().IsZero()) {
>+    target_window += rtt_variance_weight_ * rtt_stats_.mean_deviation() *
>+                     BandwidthEstimate();
>+  } else if (max_aggregation_bytes_multiplier_ > 0 && is_at_full_bandwidth_) {
>+    // Subtracting only half the bytes_acked_since_queue_drained ensures sending
>+    // doesn't completely stop for a long period of time if the queue hasn't
>+    // been drained recently.
>+    if (max_aggregation_bytes_multiplier_ * max_ack_height_.GetBest() >
>+        bytes_acked_since_queue_drained_ / 2) {
>+      target_window +=
>+          max_aggregation_bytes_multiplier_ * max_ack_height_.GetBest() -
>+          bytes_acked_since_queue_drained_ / 2;
>+    }
>+  } else if (is_at_full_bandwidth_) {
>+    target_window += max_ack_height_.GetBest();
>+  }
>+
>+  // Instead of immediately setting the target CWND as the new one, BBR grows
>+  // the CWND towards |target_window| by only increasing it |bytes_acked| at a
>+  // time.
>+  if (is_at_full_bandwidth_) {
>+    congestion_window_ =
>+        std::min(target_window, congestion_window_ + bytes_acked);
>+  } else if (congestion_window_ < target_window ||
>+             sampler_->total_data_acked() < initial_congestion_window_) {
>+    // If the connection is not yet out of startup phase, do not decrease the
>+    // window.
>+    congestion_window_ = congestion_window_ + bytes_acked;
>+  }
>+
>+  // Enforce the limits on the congestion window.
>+  congestion_window_ = std::max(congestion_window_, min_congestion_window_);
>+  congestion_window_ = std::min(congestion_window_, max_congestion_window_);
>+}
>+
>+void BbrNetworkController::CalculateRecoveryWindow(DataSize bytes_acked,
>+                                                   DataSize bytes_lost,
>+                                                   DataSize bytes_in_flight) {
>+  if (config_.rate_based_recovery ||
>+      (config_.rate_based_startup && mode_ == STARTUP)) {
>+    return;
>+  }
>+
>+  if (recovery_state_ == NOT_IN_RECOVERY) {
>+    return;
>+  }
>+
>+  // Set up the initial recovery window.
>+  if (recovery_window_.IsZero()) {
>+    recovery_window_ = bytes_in_flight + bytes_acked;
>+    recovery_window_ = std::max(min_congestion_window_, recovery_window_);
>+    return;
>+  }
>+
>+  // Remove losses from the recovery window, while accounting for a potential
>+  // integer underflow.
>+  recovery_window_ = recovery_window_ >= bytes_lost
>+                         ? recovery_window_ - bytes_lost
>+                         : kMaxSegmentSize;
>+
>+  // In CONSERVATION mode, just subtracting losses is sufficient.  In GROWTH,
>+  // release additional |bytes_acked| to achieve a slow-start-like behavior.
>+  // In MEDIUM_GROWTH, release |bytes_acked| / 2 to split the difference.
>+  if (recovery_state_ == GROWTH) {
>+    recovery_window_ += bytes_acked;
>+  } else if (recovery_state_ == MEDIUM_GROWTH) {
>+    recovery_window_ += bytes_acked / 2;
>+  }
>+
>+  // Sanity checks.  Ensure that we always allow to send at least
>+  // |bytes_acked| in response.
>+  recovery_window_ = std::max(recovery_window_, bytes_in_flight + bytes_acked);
>+  recovery_window_ = std::max(min_congestion_window_, recovery_window_);
>+}
>+
>+void BbrNetworkController::OnApplicationLimited(DataSize bytes_in_flight) {
>+  if (bytes_in_flight >= GetCongestionWindow()) {
>+    return;
>+  }
>+
>+  app_limited_since_last_probe_rtt_ = true;
>+  sampler_->OnAppLimited();
>+
>+  RTC_LOG(LS_INFO) << "Becoming application limited. Last sent packet: "
>+                   << last_sent_packet_
>+                   << ", CWND: " << ToString(GetCongestionWindow());
>+}
>+}  // namespace bbr
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/bbr_network_controller.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/bbr_network_controller.h
>new file mode 100644
>index 00000000000..8535d70d234
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/bbr_network_controller.h
>@@ -0,0 +1,395 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+// BBR (Bottleneck Bandwidth and RTT) congestion control algorithm.
>+// Based on the Quic BBR implementation in Chromium.
>+
>+#ifndef MODULES_CONGESTION_CONTROLLER_BBR_BBR_NETWORK_CONTROLLER_H_
>+#define MODULES_CONGESTION_CONTROLLER_BBR_BBR_NETWORK_CONTROLLER_H_
>+
>+#include <cstdint>
>+#include <memory>
>+#include <string>
>+#include <vector>
>+
>+#include "api/transport/network_control.h"
>+#include "api/transport/network_types.h"
>+#include "modules/congestion_controller/bbr/bandwidth_sampler.h"
>+#include "modules/congestion_controller/bbr/loss_rate_filter.h"
>+#include "modules/congestion_controller/bbr/rtt_stats.h"
>+#include "modules/congestion_controller/bbr/windowed_filter.h"
>+
>+#include "absl/types/optional.h"
>+#include "rtc_base/experiments/field_trial_parser.h"
>+#include "rtc_base/experiments/field_trial_units.h"
>+#include "rtc_base/random.h"
>+
>+namespace webrtc {
>+namespace bbr {
>+
>+typedef int64_t BbrRoundTripCount;
>+
>+// BbrSender implements BBR congestion control algorithm.  BBR aims to estimate
>+// the current available Bottleneck Bandwidth and RTT (hence the name), and
>+// regulates the pacing rate and the size of the congestion window based on
>+// those signals.
>+//
>+// BBR relies on pacing in order to function properly.  Do not use BBR when
>+// pacing is disabled.
>+class BbrNetworkController : public NetworkControllerInterface {
>+ public:
>+  enum Mode {
>+    // Startup phase of the connection.
>+    STARTUP,
>+    // After achieving the highest possible bandwidth during the startup, lower
>+    // the pacing rate in order to drain the queue.
>+    DRAIN,
>+    // Cruising mode.
>+    PROBE_BW,
>+    // Temporarily slow down sending in order to empty the buffer and measure
>+    // the real minimum RTT.
>+    PROBE_RTT,
>+  };
>+
>+  // Indicates how the congestion control limits the amount of bytes in flight.
>+  enum RecoveryState {
>+    // Do not limit.
>+    NOT_IN_RECOVERY = 0,
>+    // Allow an extra outstanding byte for each byte acknowledged.
>+    CONSERVATION = 1,
>+    // Allow 1.5 extra outstanding bytes for each byte acknowledged.
>+    MEDIUM_GROWTH = 2,
>+    // Allow two extra outstanding bytes for each byte acknowledged (slow
>+    // start).
>+    GROWTH = 3
>+  };
>+  struct BbrControllerConfig {
>+    FieldTrialParameter<double> probe_bw_pacing_gain_offset;
>+    FieldTrialParameter<double> encoder_rate_gain;
>+    FieldTrialParameter<double> encoder_rate_gain_in_probe_rtt;
>+    // RTT delta to determine if startup should be exited due to increased RTT.
>+    FieldTrialParameter<TimeDelta> exit_startup_rtt_threshold;
>+
>+    FieldTrialParameter<DataSize> initial_congestion_window;
>+    FieldTrialParameter<DataSize> min_congestion_window;
>+    FieldTrialParameter<DataSize> max_congestion_window;
>+
>+    FieldTrialParameter<double> probe_rtt_congestion_window_gain;
>+    FieldTrialParameter<bool> pacing_rate_as_target;
>+
>+    // Configurable in QUIC BBR:
>+    FieldTrialParameter<bool> exit_startup_on_loss;
>+    // The number of RTTs to stay in STARTUP mode.  Defaults to 3.
>+    FieldTrialParameter<int> num_startup_rtts;
>+    // When true, recovery is rate based rather than congestion window based.
>+    FieldTrialParameter<bool> rate_based_recovery;
>+    FieldTrialParameter<double> max_aggregation_bytes_multiplier;
>+    // When true, pace at 1.5x and disable packet conservation in STARTUP.
>+    FieldTrialParameter<bool> slower_startup;
>+    // When true, disables packet conservation in STARTUP.
>+    FieldTrialParameter<bool> rate_based_startup;
>+    // Used as the initial packet conservation mode when first entering
>+    // recovery.
>+    FieldTrialEnum<RecoveryState> initial_conservation_in_startup;
>+    // If true, will not exit low gain mode until bytes_in_flight drops below
>+    // BDP or it's time for high gain mode.
>+    FieldTrialParameter<bool> fully_drain_queue;
>+
>+    FieldTrialParameter<double> max_ack_height_window_multiplier;
>+    // If true, use a CWND of 0.75*BDP during probe_rtt instead of 4 packets.
>+    FieldTrialParameter<bool> probe_rtt_based_on_bdp;
>+    // If true, skip probe_rtt and update the timestamp of the existing min_rtt
>+    // to now if min_rtt over the last cycle is within 12.5% of the current
>+    // min_rtt. Even if the min_rtt is 12.5% too low, the 25% gain cycling and
>+    // 2x CWND gain should overcome an overly small min_rtt.
>+    FieldTrialParameter<bool> probe_rtt_skipped_if_similar_rtt;
>+    // If true, disable PROBE_RTT entirely as long as the connection was
>+    // recently app limited.
>+    FieldTrialParameter<bool> probe_rtt_disabled_if_app_limited;
>+
>+    explicit BbrControllerConfig(std::string field_trial);
>+    ~BbrControllerConfig();
>+    BbrControllerConfig(const BbrControllerConfig&);
>+    static BbrControllerConfig FromTrial();
>+  };
>+
>+  // Debug state can be exported in order to troubleshoot potential congestion
>+  // control issues.
>+  struct DebugState {
>+    explicit DebugState(const BbrNetworkController& sender);
>+    DebugState(const DebugState& state);
>+
>+    Mode mode;
>+    DataRate max_bandwidth;
>+    BbrRoundTripCount round_trip_count;
>+    int gain_cycle_index;
>+    DataSize congestion_window;
>+
>+    bool is_at_full_bandwidth;
>+    DataRate bandwidth_at_last_round;
>+    BbrRoundTripCount rounds_without_bandwidth_gain;
>+
>+    TimeDelta min_rtt;
>+    Timestamp min_rtt_timestamp;
>+
>+    RecoveryState recovery_state;
>+    DataSize recovery_window;
>+
>+    bool last_sample_is_app_limited;
>+    int64_t end_of_app_limited_phase;
>+  };
>+
>+  explicit BbrNetworkController(NetworkControllerConfig config);
>+  ~BbrNetworkController() override;
>+
>+  // NetworkControllerInterface
>+  NetworkControlUpdate OnNetworkAvailability(NetworkAvailability msg) override;
>+  NetworkControlUpdate OnNetworkRouteChange(NetworkRouteChange msg) override;
>+  NetworkControlUpdate OnProcessInterval(ProcessInterval msg) override;
>+  NetworkControlUpdate OnSentPacket(SentPacket msg) override;
>+  NetworkControlUpdate OnStreamsConfig(StreamsConfig msg) override;
>+  NetworkControlUpdate OnTargetRateConstraints(
>+      TargetRateConstraints msg) override;
>+  NetworkControlUpdate OnTransportPacketsFeedback(
>+      TransportPacketsFeedback msg) override;
>+
>+  // Part of remote bitrate estimation api, not implemented for BBR
>+  NetworkControlUpdate OnRemoteBitrateReport(RemoteBitrateReport msg) override;
>+  NetworkControlUpdate OnRoundTripTimeUpdate(RoundTripTimeUpdate msg) override;
>+  NetworkControlUpdate OnTransportLossReport(TransportLossReport msg) override;
>+
>+  NetworkControlUpdate CreateRateUpdate(Timestamp at_time) const;
>+
>+ private:
>+  void Reset();
>+  bool InSlowStart() const;
>+  bool InRecovery() const;
>+  bool IsProbingForMoreBandwidth() const;
>+
>+  bool CanSend(DataSize bytes_in_flight);
>+  DataRate PacingRate() const;
>+  DataRate BandwidthEstimate() const;
>+  DataSize GetCongestionWindow() const;
>+
>+  double GetPacingGain(int round_offset) const;
>+
>+  void OnApplicationLimited(DataSize bytes_in_flight);
>+  // End implementation of SendAlgorithmInterface.
>+
>+  typedef WindowedFilter<DataRate,
>+                         MaxFilter<DataRate>,
>+                         BbrRoundTripCount,
>+                         BbrRoundTripCount>
>+      MaxBandwidthFilter;
>+
>+  typedef WindowedFilter<TimeDelta,
>+                         MaxFilter<TimeDelta>,
>+                         BbrRoundTripCount,
>+                         BbrRoundTripCount>
>+      MaxAckDelayFilter;
>+
>+  typedef WindowedFilter<DataSize,
>+                         MaxFilter<DataSize>,
>+                         BbrRoundTripCount,
>+                         BbrRoundTripCount>
>+      MaxAckHeightFilter;
>+
>+  // Returns the current estimate of the RTT of the connection.  Outside of the
>+  // edge cases, this is minimum RTT.
>+  TimeDelta GetMinRtt() const;
>+  // Returns whether the connection has achieved full bandwidth required to exit
>+  // the slow start.
>+  bool IsAtFullBandwidth() const;
>+  // Computes the target congestion window using the specified gain.
>+  DataSize GetTargetCongestionWindow(double gain) const;
>+  // The target congestion window during PROBE_RTT.
>+  DataSize ProbeRttCongestionWindow() const;
>+  // Returns true if the current min_rtt should be kept and we should not enter
>+  // PROBE_RTT immediately.
>+  bool ShouldExtendMinRttExpiry() const;
>+
>+  // Enters the STARTUP mode.
>+  void EnterStartupMode();
>+  // Enters the PROBE_BW mode.
>+  void EnterProbeBandwidthMode(Timestamp now);
>+
>+  // Discards the lost packets from BandwidthSampler state.
>+  void DiscardLostPackets(const std::vector<PacketResult>& lost_packets);
>+  // Updates the round-trip counter if a round-trip has passed.  Returns true if
>+  // the counter has been advanced.
>+  // |last_acked_packet| is the sequence number of the last acked packet.
>+  bool UpdateRoundTripCounter(int64_t last_acked_packet);
>+  // Updates the current bandwidth and min_rtt estimate based on the samples for
>+  // the received acknowledgements.  Returns true if min_rtt has expired.
>+  bool UpdateBandwidthAndMinRtt(Timestamp now,
>+                                const std::vector<PacketResult>& acked_packets);
>+  // Updates the current gain used in PROBE_BW mode.
>+  void UpdateGainCyclePhase(Timestamp now,
>+                            DataSize prior_in_flight,
>+                            bool has_losses);
>+  // Tracks for how many round-trips the bandwidth has not increased
>+  // significantly.
>+  void CheckIfFullBandwidthReached();
>+  // Transitions from STARTUP to DRAIN and from DRAIN to PROBE_BW if
>+  // appropriate.
>+  void MaybeExitStartupOrDrain(const TransportPacketsFeedback&);
>+  // Decides whether to enter or exit PROBE_RTT.
>+  void MaybeEnterOrExitProbeRtt(const TransportPacketsFeedback& msg,
>+                                bool is_round_start,
>+                                bool min_rtt_expired);
>+  // Determines whether BBR needs to enter, exit or advance state of the
>+  // recovery.
>+  void UpdateRecoveryState(int64_t last_acked_packet,
>+                           bool has_losses,
>+                           bool is_round_start);
>+
>+  // Updates the ack aggregation max filter in bytes.
>+  void UpdateAckAggregationBytes(Timestamp ack_time,
>+                                 DataSize newly_acked_bytes);
>+
>+  // Determines the appropriate pacing rate for the connection.
>+  void CalculatePacingRate();
>+  // Determines the appropriate congestion window for the connection.
>+  void CalculateCongestionWindow(DataSize bytes_acked);
>+  // Determines the approriate window that constrains the
>+  // in-flight during recovery.
>+  void CalculateRecoveryWindow(DataSize bytes_acked,
>+                               DataSize bytes_lost,
>+                               DataSize bytes_in_flight);
>+
>+  BbrControllerConfig config_;
>+
>+  RttStats rtt_stats_;
>+  webrtc::Random random_;
>+  LossRateFilter loss_rate_;
>+
>+  absl::optional<TargetRateConstraints> constraints_;
>+
>+  Mode mode_;
>+
>+  // Bandwidth sampler provides BBR with the bandwidth measurements at
>+  // individual points.
>+  std::unique_ptr<BandwidthSampler> sampler_;
>+
>+  // The number of the round trips that have occurred during the connection.
>+  BbrRoundTripCount round_trip_count_ = 0;
>+
>+  // The packet number of the most recently sent packet.
>+  int64_t last_sent_packet_;
>+  // Acknowledgement of any packet after |current_round_trip_end_| will cause
>+  // the round trip counter to advance.
>+  int64_t current_round_trip_end_;
>+
>+  // The filter that tracks the maximum bandwidth over the multiple recent
>+  // round-trips.
>+  MaxBandwidthFilter max_bandwidth_;
>+
>+  DataRate default_bandwidth_;
>+
>+  // Tracks the maximum number of bytes acked faster than the sending rate.
>+  MaxAckHeightFilter max_ack_height_;
>+
>+  // The time this aggregation started and the number of bytes acked during it.
>+  absl::optional<Timestamp> aggregation_epoch_start_time_;
>+  DataSize aggregation_epoch_bytes_;
>+
>+  // The number of bytes acknowledged since the last time bytes in flight
>+  // dropped below the target window.
>+  DataSize bytes_acked_since_queue_drained_;
>+
>+  // The muliplier for calculating the max amount of extra CWND to add to
>+  // compensate for ack aggregation.
>+  double max_aggregation_bytes_multiplier_;
>+
>+  // Minimum RTT estimate.  Automatically expires within 10 seconds (and
>+  // triggers PROBE_RTT mode) if no new value is sampled during that period.
>+  TimeDelta min_rtt_;
>+  TimeDelta last_rtt_;
>+  // The time at which the current value of |min_rtt_| was assigned.
>+  Timestamp min_rtt_timestamp_;
>+
>+  // The maximum allowed number of bytes in flight.
>+  DataSize congestion_window_;
>+
>+  // The initial value of the |congestion_window_|.
>+  DataSize initial_congestion_window_;
>+
>+  // The smallest value the |congestion_window_| can achieve.
>+  DataSize min_congestion_window_;
>+
>+  // The largest value the |congestion_window_| can achieve.
>+  DataSize max_congestion_window_;
>+
>+  // The current pacing rate of the connection.
>+  DataRate pacing_rate_;
>+
>+  // The gain currently applied to the pacing rate.
>+  double pacing_gain_;
>+  // The gain currently applied to the congestion window.
>+  double congestion_window_gain_;
>+
>+  // The gain used for the congestion window during PROBE_BW.  Latched from
>+  // quic_bbr_cwnd_gain flag.
>+  const double congestion_window_gain_constant_;
>+  // The coefficient by which mean RTT variance is added to the congestion
>+  // window.  Latched from quic_bbr_rtt_variation_weight flag.
>+  const double rtt_variance_weight_;
>+
>+  // Number of round-trips in PROBE_BW mode, used for determining the current
>+  // pacing gain cycle.
>+  int cycle_current_offset_;
>+  // The time at which the last pacing gain cycle was started.
>+  Timestamp last_cycle_start_;
>+
>+  // Indicates whether the connection has reached the full bandwidth mode.
>+  bool is_at_full_bandwidth_;
>+  // Number of rounds during which there was no significant bandwidth increase.
>+  BbrRoundTripCount rounds_without_bandwidth_gain_;
>+  // The bandwidth compared to which the increase is measured.
>+  DataRate bandwidth_at_last_round_;
>+
>+  // Set to true upon exiting quiescence.
>+  bool exiting_quiescence_;
>+
>+  // Time at which PROBE_RTT has to be exited.  Setting it to zero indicates
>+  // that the time is yet unknown as the number of packets in flight has not
>+  // reached the required value.
>+  absl::optional<Timestamp> exit_probe_rtt_at_;
>+  // Indicates whether a round-trip has passed since PROBE_RTT became active.
>+  bool probe_rtt_round_passed_;
>+
>+  // Indicates whether the most recent bandwidth sample was marked as
>+  // app-limited.
>+  bool last_sample_is_app_limited_;
>+
>+  // Current state of recovery.
>+  RecoveryState recovery_state_;
>+  // Receiving acknowledgement of a packet after |end_recovery_at_| will cause
>+  // BBR to exit the recovery mode.  A set value indicates at least one
>+  // loss has been detected, so it must not be reset.
>+  absl::optional<int64_t> end_recovery_at_;
>+  // A window used to limit the number of bytes in flight during loss recovery.
>+  DataSize recovery_window_;
>+
>+  bool app_limited_since_last_probe_rtt_;
>+  TimeDelta min_rtt_since_last_probe_rtt_;
>+
>+  RTC_DISALLOW_COPY_AND_ASSIGN(BbrNetworkController);
>+};
>+
>+// Used in log output
>+std::ostream& operator<<(  // no-presubmit-check TODO(webrtc:8982)
>+    std::ostream& os,      // no-presubmit-check TODO(webrtc:8982)
>+    const BbrNetworkController::Mode& mode);
>+
>+}  // namespace bbr
>+}  // namespace webrtc
>+
>+#endif  // MODULES_CONGESTION_CONTROLLER_BBR_BBR_NETWORK_CONTROLLER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/bbr_network_controller_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/bbr_network_controller_unittest.cc
>new file mode 100644
>index 00000000000..68bc8ceff4d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/bbr_network_controller_unittest.cc
>@@ -0,0 +1,152 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <algorithm>
>+#include <memory>
>+
>+#include "api/transport/test/network_control_tester.h"
>+#include "modules/congestion_controller/bbr/bbr_factory.h"
>+#include "modules/congestion_controller/bbr/bbr_network_controller.h"
>+#include "test/gmock.h"
>+#include "test/gtest.h"
>+
>+using testing::Field;
>+using testing::Matcher;
>+using testing::AllOf;
>+using testing::Ge;
>+using testing::Le;
>+using testing::NiceMock;
>+using testing::Property;
>+using testing::StrictMock;
>+using testing::_;
>+
>+namespace webrtc {
>+namespace bbr {
>+namespace test {
>+namespace {
>+
>+const DataRate kInitialBitrate = DataRate::kbps(60);
>+const Timestamp kDefaultStartTime = Timestamp::ms(10000000);
>+
>+constexpr double kDataRateMargin = 0.3;
>+constexpr double kMinDataRateFactor = 1 - kDataRateMargin;
>+constexpr double kMaxDataRateFactor = 1 + kDataRateMargin;
>+inline Matcher<TargetTransferRate> TargetRateCloseTo(DataRate rate) {
>+  DataRate min_data_rate = rate * kMinDataRateFactor;
>+  DataRate max_data_rate = rate * kMaxDataRateFactor;
>+  return Field(&TargetTransferRate::target_rate,
>+               AllOf(Ge(min_data_rate), Le(max_data_rate)));
>+}
>+
>+NetworkControllerConfig InitialConfig(
>+    int starting_bandwidth_kbps = kInitialBitrate.kbps(),
>+    int min_data_rate_kbps = 0,
>+    int max_data_rate_kbps = 5 * kInitialBitrate.kbps()) {
>+  NetworkControllerConfig config;
>+  config.constraints.at_time = kDefaultStartTime;
>+  config.constraints.min_data_rate = DataRate::kbps(min_data_rate_kbps);
>+  config.constraints.max_data_rate = DataRate::kbps(max_data_rate_kbps);
>+  config.starting_bandwidth = DataRate::kbps(starting_bandwidth_kbps);
>+  return config;
>+}
>+
>+ProcessInterval InitialProcessInterval() {
>+  ProcessInterval process_interval;
>+  process_interval.at_time = kDefaultStartTime;
>+  return process_interval;
>+}
>+
>+NetworkRouteChange CreateRouteChange(Timestamp at_time,
>+                                     DataRate start_rate,
>+                                     DataRate min_rate = DataRate::Zero(),
>+                                     DataRate max_rate = DataRate::Infinity()) {
>+  NetworkRouteChange route_change;
>+  route_change.at_time = at_time;
>+  route_change.constraints.at_time = at_time;
>+  route_change.constraints.min_data_rate = min_rate;
>+  route_change.constraints.max_data_rate = max_rate;
>+  route_change.starting_rate = start_rate;
>+  return route_change;
>+}
>+}  // namespace
>+
>+class BbrNetworkControllerTest : public ::testing::Test {
>+ protected:
>+  BbrNetworkControllerTest() {}
>+  ~BbrNetworkControllerTest() override {}
>+};
>+
>+TEST_F(BbrNetworkControllerTest, SendsConfigurationOnFirstProcess) {
>+  std::unique_ptr<NetworkControllerInterface> controller_;
>+  controller_.reset(new BbrNetworkController(InitialConfig()));
>+
>+  NetworkControlUpdate update =
>+      controller_->OnProcessInterval(InitialProcessInterval());
>+  EXPECT_THAT(*update.target_rate, TargetRateCloseTo(kInitialBitrate));
>+  EXPECT_THAT(*update.pacer_config,
>+              Property(&PacerConfig::data_rate, Ge(kInitialBitrate)));
>+  EXPECT_THAT(*update.congestion_window, Property(&DataSize::IsFinite, true));
>+}
>+
>+TEST_F(BbrNetworkControllerTest, SendsConfigurationOnNetworkRouteChanged) {
>+  std::unique_ptr<NetworkControllerInterface> controller_;
>+  controller_.reset(new BbrNetworkController(InitialConfig()));
>+
>+  NetworkControlUpdate update =
>+      controller_->OnProcessInterval(InitialProcessInterval());
>+  EXPECT_TRUE(update.target_rate.has_value());
>+  EXPECT_TRUE(update.pacer_config.has_value());
>+  EXPECT_TRUE(update.congestion_window.has_value());
>+
>+  DataRate new_bitrate = DataRate::bps(200000);
>+  update = controller_->OnNetworkRouteChange(
>+      CreateRouteChange(kDefaultStartTime, new_bitrate));
>+  EXPECT_THAT(*update.target_rate, TargetRateCloseTo(new_bitrate));
>+  EXPECT_THAT(*update.pacer_config,
>+              Property(&PacerConfig::data_rate, Ge(kInitialBitrate)));
>+  EXPECT_TRUE(update.congestion_window.has_value());
>+}
>+
>+// Bandwidth estimation is updated when feedbacks are received.
>+// Feedbacks which show an increasing delay cause the estimation to be reduced.
>+TEST_F(BbrNetworkControllerTest, UpdatesTargetSendRate) {
>+  BbrNetworkControllerFactory factory;
>+  webrtc::test::NetworkControllerTester tester(&factory,
>+                                               InitialConfig(60, 0, 600));
>+  auto packet_producer = &webrtc::test::SimpleTargetRateProducer::ProduceNext;
>+
>+  tester.RunSimulation(TimeDelta::seconds(5), TimeDelta::ms(10),
>+                       DataRate::kbps(300), TimeDelta::ms(100),
>+                       packet_producer);
>+  EXPECT_GE(tester.GetState().target_rate->target_rate,
>+            DataRate::kbps(300) * kMinDataRateFactor);
>+  EXPECT_LE(tester.GetState().target_rate->target_rate,
>+            DataRate::kbps(300) * kMaxDataRateFactor);
>+
>+  tester.RunSimulation(TimeDelta::seconds(30), TimeDelta::ms(10),
>+                       DataRate::kbps(500), TimeDelta::ms(100),
>+                       packet_producer);
>+  EXPECT_GE(tester.GetState().target_rate->target_rate,
>+            DataRate::kbps(500) * kMinDataRateFactor);
>+  EXPECT_LE(tester.GetState().target_rate->target_rate,
>+            DataRate::kbps(500) * kMaxDataRateFactor);
>+
>+  tester.RunSimulation(TimeDelta::seconds(30), TimeDelta::ms(10),
>+                       DataRate::kbps(100), TimeDelta::ms(200),
>+                       packet_producer);
>+  EXPECT_GE(tester.GetState().target_rate->target_rate,
>+            DataRate::kbps(100) * kMinDataRateFactor);
>+  EXPECT_LE(tester.GetState().target_rate->target_rate,
>+            DataRate::kbps(100) * kMaxDataRateFactor);
>+}
>+
>+}  // namespace test
>+}  // namespace bbr
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/data_transfer_tracker.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/data_transfer_tracker.cc
>new file mode 100644
>index 00000000000..44e5f1d4b67
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/data_transfer_tracker.cc
>@@ -0,0 +1,84 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/congestion_controller/bbr/data_transfer_tracker.h"
>+
>+#include "rtc_base/checks.h"
>+#include "rtc_base/logging.h"
>+
>+namespace webrtc {
>+namespace bbr {
>+
>+DataTransferTracker::DataTransferTracker() {}
>+
>+DataTransferTracker::~DataTransferTracker() {}
>+
>+void DataTransferTracker::AddSample(DataSize size_delta,
>+                                    Timestamp send_time,
>+                                    Timestamp ack_time) {
>+  size_sum_ += size_delta;
>+
>+  RTC_DCHECK(samples_.empty() || ack_time >= samples_.back().ack_time);
>+
>+  if (!samples_.empty() && ack_time == samples_.back().ack_time) {
>+    samples_.back().send_time = send_time;
>+    samples_.back().size_sum = size_sum_;
>+  } else {
>+    Sample new_sample;
>+    new_sample.ack_time = ack_time;
>+    new_sample.send_time = send_time;
>+    new_sample.size_delta = size_delta;
>+    new_sample.size_sum = size_sum_;
>+    samples_.push_back(new_sample);
>+  }
>+}
>+
>+void DataTransferTracker::ClearOldSamples(Timestamp excluding_end) {
>+  while (!samples_.empty() && samples_.front().ack_time < excluding_end) {
>+    samples_.pop_front();
>+  }
>+}
>+
>+DataTransferTracker::Result DataTransferTracker::GetRatesByAckTime(
>+    Timestamp covered_start,
>+    Timestamp including_end) {
>+  Result res;
>+  // Last sample before covered_start.
>+  const Sample* window_begin = nullptr;
>+  // Sample at end time or first sample after end time-
>+  const Sample* window_end = nullptr;
>+  // To handle the case when the first sample is after covered_start.
>+  if (samples_.front().ack_time < including_end)
>+    window_begin = &samples_.front();
>+  // To handle the case when the last sample is before including_end.
>+  if (samples_.back().ack_time > covered_start)
>+    window_end = &samples_.back();
>+  for (const auto& sample : samples_) {
>+    if (sample.ack_time < covered_start) {
>+      window_begin = &sample;
>+    } else if (sample.ack_time >= including_end) {
>+      window_end = &sample;
>+      break;
>+    }
>+  }
>+  if (window_begin != nullptr && window_end != nullptr) {
>+    res.acked_data = window_end->size_sum - window_begin->size_sum;
>+    res.send_timespan = window_end->send_time - window_begin->send_time;
>+    res.ack_timespan = window_end->ack_time - window_begin->ack_time;
>+  } else {
>+    res.acked_data = DataSize::Zero();
>+    res.ack_timespan = including_end - covered_start;
>+    res.send_timespan = TimeDelta::Zero();
>+  }
>+  return res;
>+}
>+
>+}  // namespace bbr
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/data_transfer_tracker.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/data_transfer_tracker.h
>new file mode 100644
>index 00000000000..25e3997c8a0
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/data_transfer_tracker.h
>@@ -0,0 +1,50 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#ifndef MODULES_CONGESTION_CONTROLLER_BBR_DATA_TRANSFER_TRACKER_H_
>+#define MODULES_CONGESTION_CONTROLLER_BBR_DATA_TRANSFER_TRACKER_H_
>+
>+#include <deque>
>+
>+#include "api/units/data_size.h"
>+#include "api/units/time_delta.h"
>+#include "api/units/timestamp.h"
>+
>+namespace webrtc {
>+namespace bbr {
>+class DataTransferTracker {
>+ public:
>+  struct Result {
>+    TimeDelta ack_timespan = TimeDelta::Zero();
>+    TimeDelta send_timespan = TimeDelta::Zero();
>+    DataSize acked_data = DataSize::Zero();
>+  };
>+  DataTransferTracker();
>+  ~DataTransferTracker();
>+  void AddSample(DataSize size_delta, Timestamp send_time, Timestamp ack_time);
>+  void ClearOldSamples(Timestamp excluding_end);
>+
>+  // Get the average data rate in the window that starts with the last ack which
>+  // comes before covered_start and ends at the first ack that comes after or at
>+  // including_end.
>+  Result GetRatesByAckTime(Timestamp covered_start, Timestamp including_end);
>+
>+ private:
>+  struct Sample {
>+    Timestamp ack_time = Timestamp::Infinity();
>+    Timestamp send_time = Timestamp::Infinity();
>+    DataSize size_delta = DataSize::Zero();
>+    DataSize size_sum = DataSize::Zero();
>+  };
>+  std::deque<Sample> samples_;
>+  DataSize size_sum_ = DataSize::Zero();
>+};
>+}  // namespace bbr
>+}  // namespace webrtc
>+#endif  // MODULES_CONGESTION_CONTROLLER_BBR_DATA_TRANSFER_TRACKER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/data_transfer_tracker_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/data_transfer_tracker_unittest.cc
>new file mode 100644
>index 00000000000..7615a9fbb1a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/data_transfer_tracker_unittest.cc
>@@ -0,0 +1,130 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/congestion_controller/bbr/data_transfer_tracker.h"
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+namespace bbr {
>+namespace test {
>+namespace {
>+struct ResultForTest {
>+  int64_t ack_span_ms;
>+  int64_t send_span_ms;
>+  int64_t acked_bytes;
>+};
>+class DataTransferTrackerForTest : public DataTransferTracker {
>+ public:
>+  void AddSample(int bytes, int send_time_ms, int ack_time_ms) {
>+    DataTransferTracker::AddSample(DataSize::bytes(bytes),
>+                                   Timestamp::ms(send_time_ms),
>+                                   Timestamp::ms(ack_time_ms));
>+  }
>+
>+  void ClearOldSamples(int excluding_end_ms) {
>+    DataTransferTracker::ClearOldSamples(Timestamp::ms(excluding_end_ms));
>+  }
>+  ResultForTest GetRatesByAckTime(int covered_start_ms, int including_end_ms) {
>+    auto result = DataTransferTracker::GetRatesByAckTime(
>+        Timestamp::ms(covered_start_ms), Timestamp::ms(including_end_ms));
>+    return ResultForTest{result.ack_timespan.ms(), result.send_timespan.ms(),
>+                         result.acked_data.bytes()};
>+  }
>+};
>+
>+}  // namespace
>+
>+TEST(DataTransferTrackerTest, TracksData) {
>+  DataTransferTrackerForTest calc;
>+  // Since we dont have any previous reference for the first packet, it won't be
>+  // counted.
>+  calc.AddSample(5555, 100000, 100100);
>+  calc.AddSample(1000, 100020, 100120);
>+  calc.AddSample(1000, 100040, 100140);
>+  calc.AddSample(1000, 100060, 100160);
>+
>+  auto result = calc.GetRatesByAckTime(100000, 100200);
>+  EXPECT_EQ(result.acked_bytes, 3000);
>+  EXPECT_EQ(result.ack_span_ms, 60);
>+  EXPECT_EQ(result.send_span_ms, 60);
>+}
>+
>+TEST(DataTransferTrackerTest, CoversStartTime) {
>+  DataTransferTrackerForTest calc;
>+  calc.AddSample(5555, 100000, 100100);
>+  calc.AddSample(1000, 100020, 100120);
>+  calc.AddSample(1000, 100040, 100140);
>+  calc.AddSample(1000, 100060, 100160);
>+  calc.AddSample(1000, 100080, 100180);
>+
>+  auto result = calc.GetRatesByAckTime(100140, 100200);
>+  EXPECT_EQ(result.acked_bytes, 3000);
>+  EXPECT_EQ(result.ack_span_ms, 60);
>+  EXPECT_EQ(result.send_span_ms, 60);
>+}
>+
>+TEST(DataTransferTrackerTest, IncludesEndExcludesPastEnd) {
>+  DataTransferTrackerForTest calc;
>+  calc.AddSample(5555, 100000, 100100);
>+  calc.AddSample(1000, 100020, 100120);
>+  calc.AddSample(1000, 100040, 100140);
>+  calc.AddSample(1000, 100060, 100160);
>+  calc.AddSample(1000, 100080, 100180);
>+
>+  auto result = calc.GetRatesByAckTime(100120, 100160);
>+  EXPECT_EQ(result.acked_bytes, 3000);
>+  EXPECT_EQ(result.ack_span_ms, 60);
>+  EXPECT_EQ(result.send_span_ms, 60);
>+}
>+
>+TEST(DataTransferTrackerTest, AccumulatesDuplicates) {
>+  DataTransferTrackerForTest calc;
>+  calc.AddSample(5555, 100000, 100100);
>+  // Two packets at same time, should be accumulated.
>+  calc.AddSample(1000, 100020, 100120);
>+  calc.AddSample(1000, 100020, 100120);
>+  calc.AddSample(1000, 100060, 100160);
>+  // Two packets at same time, should be accumulated.
>+  calc.AddSample(1000, 100100, 100200);
>+  calc.AddSample(1000, 100100, 100200);
>+  calc.AddSample(1000, 100120, 100220);
>+
>+  auto result = calc.GetRatesByAckTime(100120, 100200);
>+  EXPECT_EQ(result.acked_bytes, 5000);
>+  EXPECT_EQ(result.ack_span_ms, 100);
>+  EXPECT_EQ(result.send_span_ms, 100);
>+}
>+
>+TEST(DataTransferTrackerTest, RemovesOldData) {
>+  DataTransferTrackerForTest calc;
>+  calc.AddSample(5555, 100000, 100100);
>+  calc.AddSample(1000, 100020, 100120);
>+  calc.AddSample(1000, 100040, 100140);
>+  calc.AddSample(1000, 100060, 100160);
>+  calc.AddSample(1000, 100080, 100180);
>+  {
>+    auto result = calc.GetRatesByAckTime(100120, 100200);
>+    EXPECT_EQ(result.acked_bytes, 4000);
>+    EXPECT_EQ(result.ack_span_ms, 80);
>+    EXPECT_EQ(result.send_span_ms, 80);
>+  }
>+  // Note that this operation means that the packet acked at 100140 will not be
>+  // counted any more, just used as time reference.
>+  calc.ClearOldSamples(100140);
>+  {
>+    auto result = calc.GetRatesByAckTime(100120, 100200);
>+    EXPECT_EQ(result.acked_bytes, 2000);
>+    EXPECT_EQ(result.ack_span_ms, 40);
>+    EXPECT_EQ(result.send_span_ms, 40);
>+  }
>+}
>+}  // namespace test
>+}  // namespace bbr
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/loss_rate_filter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/loss_rate_filter.cc
>new file mode 100644
>index 00000000000..302e64a4291
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/loss_rate_filter.cc
>@@ -0,0 +1,48 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#include "modules/congestion_controller/bbr/loss_rate_filter.h"
>+
>+namespace webrtc {
>+namespace bbr {
>+namespace {
>+// From SendSideBandwidthEstimation.
>+const int kLimitNumPackets = 20;
>+// From RTCPSender video report interval.
>+const int64_t kUpdateIntervalMs = 1000;
>+}  // namespace
>+
>+LossRateFilter::LossRateFilter()
>+    : lost_packets_since_last_loss_update_(0),
>+      expected_packets_since_last_loss_update_(0),
>+      loss_rate_estimate_(0.0),
>+      next_loss_update_ms_(0) {}
>+
>+void LossRateFilter::UpdateWithLossStatus(int64_t feedback_time,
>+                                          int packets_sent,
>+                                          int packets_lost) {
>+  lost_packets_since_last_loss_update_ += packets_lost;
>+  expected_packets_since_last_loss_update_ += packets_sent;
>+
>+  if (feedback_time >= next_loss_update_ms_ &&
>+      expected_packets_since_last_loss_update_ >= kLimitNumPackets) {
>+    int64_t lost = lost_packets_since_last_loss_update_;
>+    int64_t expected = expected_packets_since_last_loss_update_;
>+    loss_rate_estimate_ = static_cast<double>(lost) / expected;
>+    next_loss_update_ms_ = feedback_time + kUpdateIntervalMs;
>+    lost_packets_since_last_loss_update_ = 0;
>+    expected_packets_since_last_loss_update_ = 0;
>+  }
>+}
>+
>+double LossRateFilter::GetLossRate() const {
>+  return loss_rate_estimate_;
>+}
>+}  // namespace bbr
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/loss_rate_filter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/loss_rate_filter.h
>new file mode 100644
>index 00000000000..6e31b6301cc
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/loss_rate_filter.h
>@@ -0,0 +1,38 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#ifndef MODULES_CONGESTION_CONTROLLER_BBR_LOSS_RATE_FILTER_H_
>+#define MODULES_CONGESTION_CONTROLLER_BBR_LOSS_RATE_FILTER_H_
>+
>+#include "absl/types/optional.h"
>+
>+namespace webrtc {
>+namespace bbr {
>+
>+// Loss rate filter based on the implementation in SendSideBandwidthEstimation
>+// and the RTCPSender receiver report interval for video.
>+class LossRateFilter {
>+ public:
>+  LossRateFilter();
>+  void UpdateWithLossStatus(int64_t feedback_time_ms,
>+                            int packets_sent,
>+                            int packets_lost);
>+  double GetLossRate() const;
>+
>+ private:
>+  int lost_packets_since_last_loss_update_;
>+  int expected_packets_since_last_loss_update_;
>+  double loss_rate_estimate_;
>+  int64_t next_loss_update_ms_;
>+};
>+
>+}  // namespace bbr
>+}  // namespace webrtc
>+
>+#endif  // MODULES_CONGESTION_CONTROLLER_BBR_LOSS_RATE_FILTER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/loss_rate_filter_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/loss_rate_filter_unittest.cc
>new file mode 100644
>index 00000000000..c007da8b521
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/loss_rate_filter_unittest.cc
>@@ -0,0 +1,71 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#include "modules/congestion_controller/bbr/loss_rate_filter.h"
>+#include "api/units/timestamp.h"
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+namespace bbr {
>+
>+namespace {
>+const Timestamp kTestStartTime = Timestamp::seconds(100000);
>+}  // namespace
>+
>+TEST(LossRateFilterTest, AccumulatesToOne) {
>+  LossRateFilter filter;
>+  Timestamp current_time = kTestStartTime;
>+  for (int i = 0; i < 10; i++) {
>+    filter.UpdateWithLossStatus(current_time.ms(), 10, 10);
>+    current_time += TimeDelta::seconds(1);
>+  }
>+  EXPECT_NEAR(filter.GetLossRate(), 1.0, 0.01);
>+}
>+
>+TEST(LossRateFilterTest, StaysAtZero) {
>+  LossRateFilter filter;
>+  Timestamp current_time = kTestStartTime;
>+  for (int i = 0; i < 10; i++) {
>+    filter.UpdateWithLossStatus(current_time.ms(), 10, 0);
>+    current_time += TimeDelta::seconds(1);
>+  }
>+  EXPECT_NEAR(filter.GetLossRate(), 0.0, 0.01);
>+}
>+
>+TEST(LossRateFilterTest, VariesWithInput) {
>+  LossRateFilter filter;
>+  Timestamp current_time = kTestStartTime;
>+  for (int j = 0; j < 10; j++) {
>+    for (int i = 0; i < 5; i++) {
>+      filter.UpdateWithLossStatus(current_time.ms(), 10, 10);
>+      current_time += TimeDelta::seconds(1);
>+    }
>+    EXPECT_NEAR(filter.GetLossRate(), 1.0, 0.1);
>+    for (int i = 0; i < 5; i++) {
>+      filter.UpdateWithLossStatus(current_time.ms(), 10, 0);
>+      current_time += TimeDelta::seconds(1);
>+    }
>+    EXPECT_NEAR(filter.GetLossRate(), 0.0, 0.1);
>+  }
>+}
>+
>+TEST(LossRateFilterTest, DetectsChangingRate) {
>+  LossRateFilter filter;
>+  Timestamp current_time = kTestStartTime;
>+  for (int per_decile = 0; per_decile < 10; per_decile += 1) {
>+    // Update every 200 ms for 2 seconds
>+    for (int i = 0; i < 10; i++) {
>+      current_time += TimeDelta::ms(200);
>+      filter.UpdateWithLossStatus(current_time.ms(), 10, per_decile);
>+    }
>+    EXPECT_NEAR(filter.GetLossRate(), per_decile / 10.0, 0.05);
>+  }
>+}
>+}  // namespace bbr
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/packet_number_indexed_queue.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/packet_number_indexed_queue.h
>new file mode 100644
>index 00000000000..208920947e6
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/packet_number_indexed_queue.h
>@@ -0,0 +1,217 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+// Based on the Quic implementation in Chromium.
>+
>+#ifndef MODULES_CONGESTION_CONTROLLER_BBR_PACKET_NUMBER_INDEXED_QUEUE_H_
>+#define MODULES_CONGESTION_CONTROLLER_BBR_PACKET_NUMBER_INDEXED_QUEUE_H_
>+
>+#include <stdint.h>
>+#include <deque>
>+#include <utility>
>+
>+#include "rtc_base/checks.h"
>+
>+namespace webrtc {
>+namespace bbr {
>+
>+// PacketNumberIndexedQueue is a queue of mostly continuous numbered entries
>+// which supports the following operations:
>+// - adding elements to the end of the queue, or at some point past the end
>+// - removing elements in any order
>+// - retrieving elements
>+// If all elements are inserted in order, all of the operations above are
>+// amortized O(1) time.
>+//
>+// Internally, the data structure is a deque where each element is marked as
>+// present or not.  The deque starts at the lowest present index.  Whenever an
>+// element is removed, it's marked as not present, and the front of the deque is
>+// cleared of elements that are not present.
>+//
>+// The tail of the queue is not cleared due to the assumption of entries being
>+// inserted in order, though removing all elements of the queue will return it
>+// to its initial state.
>+//
>+// Note that this data structure is inherently hazardous, since an addition of
>+// just two entries will cause it to consume all of the memory available.
>+// Because of that, it is not a general-purpose container and should not be used
>+// as one.
>+template <typename T>
>+class PacketNumberIndexedQueue {
>+ public:
>+  PacketNumberIndexedQueue()
>+      : number_of_present_entries_(0), first_packet_(0) {}
>+
>+  // Retrieve the entry associated with the packet number.  Returns the pointer
>+  // to the entry in case of success, or nullptr if the entry does not exist.
>+  T* GetEntry(int64_t packet_number);
>+  const T* GetEntry(int64_t packet_number) const;
>+
>+  // Inserts data associated |packet_number| into (or past) the end of the
>+  // queue, filling up the missing intermediate entries as necessary.  Returns
>+  // true if the element has been inserted successfully, false if it was already
>+  // in the queue or inserted out of order.
>+  template <typename... Args>
>+  bool Emplace(int64_t packet_number, Args&&... args);
>+
>+  // Removes data associated with |packet_number| and frees the slots in the
>+  // queue as necessary.
>+  bool Remove(int64_t packet_number);
>+
>+  bool IsEmpty() const { return number_of_present_entries_ == 0; }
>+
>+  // Returns the number of entries in the queue.
>+  size_t number_of_present_entries() const {
>+    return number_of_present_entries_;
>+  }
>+
>+  // Returns the number of entries allocated in the underlying deque.  This is
>+  // proportional to the memory usage of the queue.
>+  size_t entry_slots_used() const { return entries_.size(); }
>+
>+  // Packet number of the first entry in the queue.  Zero if the queue is empty.
>+  int64_t first_packet() const { return first_packet_; }
>+
>+  // Packet number of the last entry ever inserted in the queue.  Note that the
>+  // entry in question may have already been removed.  Zero if the queue is
>+  // empty.
>+  int64_t last_packet() const {
>+    if (IsEmpty()) {
>+      return 0;
>+    }
>+    return first_packet_ + entries_.size() - 1;
>+  }
>+
>+ private:
>+  // Wrapper around T used to mark whether the entry is actually in the map.
>+  struct EntryWrapper {
>+    T data;
>+    bool present;
>+
>+    EntryWrapper() : data(), present(false) {}
>+
>+    template <typename... Args>
>+    explicit EntryWrapper(Args&&... args)
>+        : data(std::forward<Args>(args)...), present(true) {}
>+  };
>+
>+  // Cleans up unused slots in the front after removing an element.
>+  void Cleanup();
>+
>+  const EntryWrapper* GetEntryWrapper(int64_t offset) const;
>+  EntryWrapper* GetEntryWrapper(int64_t offset) {
>+    const auto* const_this = this;
>+    return const_cast<EntryWrapper*>(const_this->GetEntryWrapper(offset));
>+  }
>+
>+  std::deque<EntryWrapper> entries_;
>+  size_t number_of_present_entries_;
>+  int64_t first_packet_;
>+};
>+
>+template <typename T>
>+T* PacketNumberIndexedQueue<T>::GetEntry(int64_t packet_number) {
>+  EntryWrapper* entry = GetEntryWrapper(packet_number);
>+  if (entry == nullptr) {
>+    return nullptr;
>+  }
>+  return &entry->data;
>+}
>+
>+template <typename T>
>+const T* PacketNumberIndexedQueue<T>::GetEntry(int64_t packet_number) const {
>+  const EntryWrapper* entry = GetEntryWrapper(packet_number);
>+  if (entry == nullptr) {
>+    return nullptr;
>+  }
>+  return &entry->data;
>+}
>+
>+template <typename T>
>+template <typename... Args>
>+bool PacketNumberIndexedQueue<T>::Emplace(int64_t packet_number,
>+                                          Args&&... args) {
>+  if (IsEmpty()) {
>+    RTC_DCHECK(entries_.empty());
>+    RTC_DCHECK_EQ(0u, first_packet_);
>+
>+    entries_.emplace_back(std::forward<Args>(args)...);
>+    number_of_present_entries_ = 1;
>+    first_packet_ = packet_number;
>+    return true;
>+  }
>+
>+  // Do not allow insertion out-of-order.
>+  if (packet_number <= last_packet()) {
>+    return false;
>+  }
>+
>+  // Handle potentially missing elements.
>+  int64_t offset = packet_number - first_packet_;
>+  if (offset > static_cast<int64_t>(entries_.size())) {
>+    entries_.resize(offset);
>+  }
>+
>+  number_of_present_entries_++;
>+  entries_.emplace_back(std::forward<Args>(args)...);
>+  RTC_DCHECK_EQ(packet_number, last_packet());
>+  return true;
>+}
>+
>+template <typename T>
>+bool PacketNumberIndexedQueue<T>::Remove(int64_t packet_number) {
>+  EntryWrapper* entry = GetEntryWrapper(packet_number);
>+  if (entry == nullptr) {
>+    return false;
>+  }
>+  entry->present = false;
>+  number_of_present_entries_--;
>+
>+  if (packet_number == first_packet()) {
>+    Cleanup();
>+  }
>+  return true;
>+}
>+
>+template <typename T>
>+void PacketNumberIndexedQueue<T>::Cleanup() {
>+  while (!entries_.empty() && !entries_.front().present) {
>+    entries_.pop_front();
>+    first_packet_++;
>+  }
>+  if (entries_.empty()) {
>+    first_packet_ = 0;
>+  }
>+}
>+
>+template <typename T>
>+auto PacketNumberIndexedQueue<T>::GetEntryWrapper(int64_t offset) const
>+    -> const EntryWrapper* {
>+  if (offset < first_packet_) {
>+    return nullptr;
>+  }
>+
>+  offset -= first_packet_;
>+  if (offset >= static_cast<int64_t>(entries_.size())) {
>+    return nullptr;
>+  }
>+
>+  const EntryWrapper* entry = &entries_[offset];
>+  if (!entry->present) {
>+    return nullptr;
>+  }
>+
>+  return entry;
>+}
>+
>+}  // namespace bbr
>+}  // namespace webrtc
>+
>+#endif  // MODULES_CONGESTION_CONTROLLER_BBR_PACKET_NUMBER_INDEXED_QUEUE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/packet_number_indexed_queue_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/packet_number_indexed_queue_unittest.cc
>new file mode 100644
>index 00000000000..acfa8717106
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/packet_number_indexed_queue_unittest.cc
>@@ -0,0 +1,187 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/congestion_controller/bbr/packet_number_indexed_queue.h"
>+
>+#include <limits>
>+#include <map>
>+#include <string>
>+
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+namespace bbr {
>+namespace {
>+
>+class PacketNumberIndexedQueueTest : public ::testing::Test {
>+ public:
>+  PacketNumberIndexedQueueTest() {}
>+
>+ protected:
>+  PacketNumberIndexedQueue<std::string> queue_;
>+};
>+
>+TEST_F(PacketNumberIndexedQueueTest, InitialState) {
>+  EXPECT_TRUE(queue_.IsEmpty());
>+  EXPECT_EQ(0u, queue_.first_packet());
>+  EXPECT_EQ(0u, queue_.last_packet());
>+  EXPECT_EQ(0u, queue_.number_of_present_entries());
>+  EXPECT_EQ(0u, queue_.entry_slots_used());
>+}
>+
>+TEST_F(PacketNumberIndexedQueueTest, InsertingContinuousElements) {
>+  ASSERT_TRUE(queue_.Emplace(1001, "one"));
>+  EXPECT_EQ("one", *queue_.GetEntry(1001));
>+
>+  ASSERT_TRUE(queue_.Emplace(1002, "two"));
>+  EXPECT_EQ("two", *queue_.GetEntry(1002));
>+
>+  EXPECT_FALSE(queue_.IsEmpty());
>+  EXPECT_EQ(1001u, queue_.first_packet());
>+  EXPECT_EQ(1002u, queue_.last_packet());
>+  EXPECT_EQ(2u, queue_.number_of_present_entries());
>+  EXPECT_EQ(2u, queue_.entry_slots_used());
>+}
>+
>+TEST_F(PacketNumberIndexedQueueTest, InsertingOutOfOrder) {
>+  queue_.Emplace(1001, "one");
>+
>+  ASSERT_TRUE(queue_.Emplace(1003, "three"));
>+  EXPECT_EQ(nullptr, queue_.GetEntry(1002));
>+  EXPECT_EQ("three", *queue_.GetEntry(1003));
>+
>+  EXPECT_EQ(1001u, queue_.first_packet());
>+  EXPECT_EQ(1003u, queue_.last_packet());
>+  EXPECT_EQ(2u, queue_.number_of_present_entries());
>+  EXPECT_EQ(3u, queue_.entry_slots_used());
>+
>+  ASSERT_FALSE(queue_.Emplace(1002, "two"));
>+}
>+
>+TEST_F(PacketNumberIndexedQueueTest, InsertingIntoPast) {
>+  queue_.Emplace(1001, "one");
>+  EXPECT_FALSE(queue_.Emplace(1000, "zero"));
>+}
>+
>+TEST_F(PacketNumberIndexedQueueTest, InsertingDuplicate) {
>+  queue_.Emplace(1001, "one");
>+  EXPECT_FALSE(queue_.Emplace(1001, "one"));
>+}
>+
>+TEST_F(PacketNumberIndexedQueueTest, RemoveInTheMiddle) {
>+  queue_.Emplace(1001, "one");
>+  queue_.Emplace(1002, "two");
>+  queue_.Emplace(1003, "three");
>+
>+  ASSERT_TRUE(queue_.Remove(1002));
>+  EXPECT_EQ(nullptr, queue_.GetEntry(1002));
>+
>+  EXPECT_EQ(1001u, queue_.first_packet());
>+  EXPECT_EQ(1003u, queue_.last_packet());
>+  EXPECT_EQ(2u, queue_.number_of_present_entries());
>+  EXPECT_EQ(3u, queue_.entry_slots_used());
>+
>+  EXPECT_FALSE(queue_.Emplace(1002, "two"));
>+  EXPECT_TRUE(queue_.Emplace(1004, "four"));
>+}
>+
>+TEST_F(PacketNumberIndexedQueueTest, RemoveAtImmediateEdges) {
>+  queue_.Emplace(1001, "one");
>+  queue_.Emplace(1002, "two");
>+  queue_.Emplace(1003, "three");
>+  ASSERT_TRUE(queue_.Remove(1001));
>+  EXPECT_EQ(nullptr, queue_.GetEntry(1001));
>+  ASSERT_TRUE(queue_.Remove(1003));
>+  EXPECT_EQ(nullptr, queue_.GetEntry(1003));
>+
>+  EXPECT_EQ(1002u, queue_.first_packet());
>+  EXPECT_EQ(1003u, queue_.last_packet());
>+  EXPECT_EQ(1u, queue_.number_of_present_entries());
>+  EXPECT_EQ(2u, queue_.entry_slots_used());
>+
>+  EXPECT_TRUE(queue_.Emplace(1004, "four"));
>+}
>+
>+TEST_F(PacketNumberIndexedQueueTest, RemoveAtDistantFront) {
>+  queue_.Emplace(1001, "one");
>+  queue_.Emplace(1002, "one (kinda)");
>+  queue_.Emplace(2001, "two");
>+
>+  EXPECT_EQ(1001u, queue_.first_packet());
>+  EXPECT_EQ(2001u, queue_.last_packet());
>+  EXPECT_EQ(3u, queue_.number_of_present_entries());
>+  EXPECT_EQ(1001u, queue_.entry_slots_used());
>+
>+  ASSERT_TRUE(queue_.Remove(1002));
>+  EXPECT_EQ(1001u, queue_.first_packet());
>+  EXPECT_EQ(2001u, queue_.last_packet());
>+  EXPECT_EQ(2u, queue_.number_of_present_entries());
>+  EXPECT_EQ(1001u, queue_.entry_slots_used());
>+
>+  ASSERT_TRUE(queue_.Remove(1001));
>+  EXPECT_EQ(2001u, queue_.first_packet());
>+  EXPECT_EQ(2001u, queue_.last_packet());
>+  EXPECT_EQ(1u, queue_.number_of_present_entries());
>+  EXPECT_EQ(1u, queue_.entry_slots_used());
>+}
>+
>+TEST_F(PacketNumberIndexedQueueTest, RemoveAtDistantBack) {
>+  queue_.Emplace(1001, "one");
>+  queue_.Emplace(2001, "two");
>+
>+  EXPECT_EQ(1001u, queue_.first_packet());
>+  EXPECT_EQ(2001u, queue_.last_packet());
>+
>+  ASSERT_TRUE(queue_.Remove(2001));
>+  EXPECT_EQ(1001u, queue_.first_packet());
>+  EXPECT_EQ(2001u, queue_.last_packet());
>+}
>+
>+TEST_F(PacketNumberIndexedQueueTest, ClearAndRepopulate) {
>+  queue_.Emplace(1001, "one");
>+  queue_.Emplace(2001, "two");
>+
>+  ASSERT_TRUE(queue_.Remove(1001));
>+  ASSERT_TRUE(queue_.Remove(2001));
>+  EXPECT_TRUE(queue_.IsEmpty());
>+  EXPECT_EQ(0u, queue_.first_packet());
>+  EXPECT_EQ(0u, queue_.last_packet());
>+
>+  EXPECT_TRUE(queue_.Emplace(101, "one"));
>+  EXPECT_TRUE(queue_.Emplace(201, "two"));
>+  EXPECT_EQ(101u, queue_.first_packet());
>+  EXPECT_EQ(201u, queue_.last_packet());
>+}
>+
>+TEST_F(PacketNumberIndexedQueueTest, FailToRemoveElementsThatNeverExisted) {
>+  ASSERT_FALSE(queue_.Remove(1000));
>+  queue_.Emplace(1001, "one");
>+  ASSERT_FALSE(queue_.Remove(1000));
>+  ASSERT_FALSE(queue_.Remove(1002));
>+}
>+
>+TEST_F(PacketNumberIndexedQueueTest, FailToRemoveElementsTwice) {
>+  queue_.Emplace(1001, "one");
>+  ASSERT_TRUE(queue_.Remove(1001));
>+  ASSERT_FALSE(queue_.Remove(1001));
>+  ASSERT_FALSE(queue_.Remove(1001));
>+}
>+
>+TEST_F(PacketNumberIndexedQueueTest, ConstGetter) {
>+  queue_.Emplace(1001, "one");
>+  const auto& const_queue = queue_;
>+
>+  EXPECT_EQ("one", *const_queue.GetEntry(1001));
>+  EXPECT_EQ(nullptr, const_queue.GetEntry(1002));
>+}
>+
>+}  // namespace
>+}  // namespace bbr
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/rtt_stats.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/rtt_stats.cc
>new file mode 100644
>index 00000000000..e79ba9f402b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/rtt_stats.cc
>@@ -0,0 +1,95 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/congestion_controller/bbr/rtt_stats.h"
>+
>+#include <cstdlib>
>+
>+#include "rtc_base/logging.h"
>+
>+namespace webrtc {
>+namespace bbr {
>+namespace {
>+
>+// Default initial rtt used before any samples are received.
>+const int kInitialRttMs = 100;
>+const double kAlpha = 0.125;
>+const double kOneMinusAlpha = (1 - kAlpha);
>+const double kBeta = 0.25;
>+const double kOneMinusBeta = (1 - kBeta);
>+const int64_t kNumMicrosPerMilli = 1000;
>+}  // namespace
>+
>+RttStats::RttStats()
>+    : latest_rtt_(TimeDelta::Zero()),
>+      min_rtt_(TimeDelta::Zero()),
>+      smoothed_rtt_(TimeDelta::Zero()),
>+      previous_srtt_(TimeDelta::Zero()),
>+      mean_deviation_(TimeDelta::Zero()),
>+      initial_rtt_us_(kInitialRttMs * kNumMicrosPerMilli) {}
>+
>+void RttStats::ExpireSmoothedMetrics() {
>+  mean_deviation_ =
>+      std::max(mean_deviation_, (smoothed_rtt_ - latest_rtt_).Abs());
>+  smoothed_rtt_ = std::max(smoothed_rtt_, latest_rtt_);
>+}
>+
>+// Updates the RTT based on a new sample.
>+void RttStats::UpdateRtt(TimeDelta send_delta,
>+                         TimeDelta ack_delay,
>+                         Timestamp now) {
>+  if (send_delta.IsInfinite() || send_delta <= TimeDelta::Zero()) {
>+    RTC_LOG(LS_WARNING) << "Ignoring measured send_delta, because it's is "
>+                        << "either infinite, zero, or negative.  send_delta = "
>+                        << ToString(send_delta);
>+    return;
>+  }
>+
>+  // Update min_rtt_ first. min_rtt_ does not use an rtt_sample corrected for
>+  // ack_delay but the raw observed send_delta, since poor clock granularity at
>+  // the client may cause a high ack_delay to result in underestimation of the
>+  // min_rtt_.
>+  if (min_rtt_.IsZero() || min_rtt_ > send_delta) {
>+    min_rtt_ = send_delta;
>+  }
>+
>+  // Correct for ack_delay if information received from the peer results in a
>+  // positive RTT sample. Otherwise, we use the send_delta as a reasonable
>+  // measure for smoothed_rtt.
>+  TimeDelta rtt_sample = send_delta;
>+  previous_srtt_ = smoothed_rtt_;
>+
>+  if (rtt_sample > ack_delay) {
>+    rtt_sample = rtt_sample - ack_delay;
>+  }
>+  latest_rtt_ = rtt_sample;
>+  // First time call.
>+  if (smoothed_rtt_.IsZero()) {
>+    smoothed_rtt_ = rtt_sample;
>+    mean_deviation_ = rtt_sample / 2;
>+  } else {
>+    mean_deviation_ = kOneMinusBeta * mean_deviation_ +
>+                      kBeta * (smoothed_rtt_ - rtt_sample).Abs();
>+    smoothed_rtt_ = kOneMinusAlpha * smoothed_rtt_ + kAlpha * rtt_sample;
>+    RTC_LOG(LS_VERBOSE) << " smoothed_rtt(us):" << smoothed_rtt_.us()
>+                        << " mean_deviation(us):" << mean_deviation_.us();
>+  }
>+}
>+
>+void RttStats::OnConnectionMigration() {
>+  latest_rtt_ = TimeDelta::Zero();
>+  min_rtt_ = TimeDelta::Zero();
>+  smoothed_rtt_ = TimeDelta::Zero();
>+  mean_deviation_ = TimeDelta::Zero();
>+  initial_rtt_us_ = kInitialRttMs * kNumMicrosPerMilli;
>+}
>+
>+}  // namespace bbr
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/rtt_stats.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/rtt_stats.h
>new file mode 100644
>index 00000000000..bb0bda1ee1d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/rtt_stats.h
>@@ -0,0 +1,89 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+// A convenience class to store RTT samples and calculate smoothed RTT.
>+// From the Quic BBR implementation in Chromium.
>+
>+#ifndef MODULES_CONGESTION_CONTROLLER_BBR_RTT_STATS_H_
>+#define MODULES_CONGESTION_CONTROLLER_BBR_RTT_STATS_H_
>+
>+#include <algorithm>
>+#include <cstdint>
>+
>+#include "api/units/time_delta.h"
>+#include "api/units/timestamp.h"
>+#include "rtc_base/constructormagic.h"
>+#include "rtc_base/logging.h"
>+
>+namespace webrtc {
>+namespace bbr {
>+
>+class RttStats {
>+ public:
>+  RttStats();
>+
>+  // Updates the RTT from an incoming ack which is received |send_delta| after
>+  // the packet is sent and the peer reports the ack being delayed |ack_delay|.
>+  void UpdateRtt(TimeDelta send_delta, TimeDelta ack_delay, Timestamp now);
>+
>+  // Causes the smoothed_rtt to be increased to the latest_rtt if the latest_rtt
>+  // is larger. The mean deviation is increased to the most recent deviation if
>+  // it's larger.
>+  void ExpireSmoothedMetrics();
>+
>+  // Called when connection migrates and RTT measurement needs to be reset.
>+  void OnConnectionMigration();
>+
>+  // Returns the EWMA smoothed RTT for the connection.
>+  // May return Zero if no valid updates have occurred.
>+  TimeDelta smoothed_rtt() const { return smoothed_rtt_; }
>+
>+  // Returns the EWMA smoothed RTT prior to the most recent RTT sample.
>+  TimeDelta previous_srtt() const { return previous_srtt_; }
>+
>+  int64_t initial_rtt_us() const { return initial_rtt_us_; }
>+
>+  // Sets an initial RTT to be used for SmoothedRtt before any RTT updates.
>+  void set_initial_rtt_us(int64_t initial_rtt_us) {
>+    RTC_DCHECK_GE(initial_rtt_us, 0);
>+    if (initial_rtt_us <= 0) {
>+      RTC_LOG(LS_ERROR) << "Attempt to set initial rtt to <= 0.";
>+      return;
>+    }
>+    initial_rtt_us_ = initial_rtt_us;
>+  }
>+
>+  // The most recent RTT measurement.
>+  // May return Zero if no valid updates have occurred.
>+  TimeDelta latest_rtt() const { return latest_rtt_; }
>+
>+  // Returns the min_rtt for the entire connection.
>+  // May return Zero if no valid updates have occurred.
>+  TimeDelta min_rtt() const { return min_rtt_; }
>+
>+  TimeDelta mean_deviation() const { return mean_deviation_; }
>+
>+ private:
>+  TimeDelta latest_rtt_;
>+  TimeDelta min_rtt_;
>+  TimeDelta smoothed_rtt_;
>+  TimeDelta previous_srtt_;
>+  // Mean RTT deviation during this session.
>+  // Approximation of standard deviation, the error is roughly 1.25 times
>+  // larger than the standard deviation, for a normally distributed signal.
>+  TimeDelta mean_deviation_;
>+  int64_t initial_rtt_us_;
>+
>+  RTC_DISALLOW_COPY_AND_ASSIGN(RttStats);
>+};
>+
>+}  // namespace bbr
>+}  // namespace webrtc
>+
>+#endif  // MODULES_CONGESTION_CONTROLLER_BBR_RTT_STATS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/rtt_stats_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/rtt_stats_unittest.cc
>new file mode 100644
>index 00000000000..6abe70482fb
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/rtt_stats_unittest.cc
>@@ -0,0 +1,160 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/congestion_controller/bbr/rtt_stats.h"
>+
>+#include <vector>
>+
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+namespace bbr {
>+namespace test {
>+
>+class RttStatsTest : public ::testing::Test {
>+ protected:
>+  RttStats rtt_stats_;
>+};
>+
>+TEST_F(RttStatsTest, DefaultsBeforeUpdate) {
>+  EXPECT_LT(0u, rtt_stats_.initial_rtt_us());
>+  EXPECT_EQ(TimeDelta::Zero(), rtt_stats_.min_rtt());
>+  EXPECT_EQ(TimeDelta::Zero(), rtt_stats_.smoothed_rtt());
>+}
>+
>+TEST_F(RttStatsTest, SmoothedRtt) {
>+  // Verify that ack_delay is corrected for in Smoothed RTT.
>+  rtt_stats_.UpdateRtt(TimeDelta::ms(300), TimeDelta::ms(100),
>+                       Timestamp::ms(0));
>+  EXPECT_EQ(TimeDelta::ms(200), rtt_stats_.latest_rtt());
>+  EXPECT_EQ(TimeDelta::ms(200), rtt_stats_.smoothed_rtt());
>+  // Verify that effective RTT of zero does not change Smoothed RTT.
>+  rtt_stats_.UpdateRtt(TimeDelta::ms(200), TimeDelta::ms(200),
>+                       Timestamp::ms(0));
>+  EXPECT_EQ(TimeDelta::ms(200), rtt_stats_.latest_rtt());
>+  EXPECT_EQ(TimeDelta::ms(200), rtt_stats_.smoothed_rtt());
>+  // Verify that large erroneous ack_delay does not change Smoothed RTT.
>+  rtt_stats_.UpdateRtt(TimeDelta::ms(200), TimeDelta::ms(300),
>+                       Timestamp::ms(0));
>+  EXPECT_EQ(TimeDelta::ms(200), rtt_stats_.latest_rtt());
>+  EXPECT_EQ(TimeDelta::ms(200), rtt_stats_.smoothed_rtt());
>+}
>+
>+// Ensure that the potential rounding artifacts in EWMA calculation do not cause
>+// the SRTT to drift too far from the exact value.
>+TEST_F(RttStatsTest, SmoothedRttStability) {
>+  for (int64_t time = 3; time < 20000; time++) {
>+    RttStats stats;
>+    for (int64_t i = 0; i < 100; i++) {
>+      stats.UpdateRtt(TimeDelta::us(time), TimeDelta::ms(0), Timestamp::ms(0));
>+      int64_t time_delta_us = stats.smoothed_rtt().us() - time;
>+      ASSERT_LE(std::abs(time_delta_us), 1);
>+    }
>+  }
>+}
>+
>+TEST_F(RttStatsTest, PreviousSmoothedRtt) {
>+  // Verify that ack_delay is corrected for in Smoothed RTT.
>+  rtt_stats_.UpdateRtt(TimeDelta::ms(300), TimeDelta::ms(100),
>+                       Timestamp::ms(0));
>+  EXPECT_EQ(TimeDelta::ms(200), rtt_stats_.latest_rtt());
>+  EXPECT_EQ(TimeDelta::ms(200), rtt_stats_.smoothed_rtt());
>+  EXPECT_EQ(TimeDelta::Zero(), rtt_stats_.previous_srtt());
>+  // Ensure the previous SRTT is 200ms after a 100ms sample.
>+  rtt_stats_.UpdateRtt(TimeDelta::ms(100), TimeDelta::Zero(), Timestamp::ms(0));
>+  EXPECT_EQ(TimeDelta::ms(100), rtt_stats_.latest_rtt());
>+  EXPECT_EQ(TimeDelta::us(187500).us(), rtt_stats_.smoothed_rtt().us());
>+  EXPECT_EQ(TimeDelta::ms(200), rtt_stats_.previous_srtt());
>+}
>+
>+TEST_F(RttStatsTest, MinRtt) {
>+  rtt_stats_.UpdateRtt(TimeDelta::ms(200), TimeDelta::Zero(), Timestamp::ms(0));
>+  EXPECT_EQ(TimeDelta::ms(200), rtt_stats_.min_rtt());
>+  rtt_stats_.UpdateRtt(TimeDelta::ms(10), TimeDelta::Zero(),
>+                       Timestamp::ms(0) + TimeDelta::ms(10));
>+  EXPECT_EQ(TimeDelta::ms(10), rtt_stats_.min_rtt());
>+  rtt_stats_.UpdateRtt(TimeDelta::ms(50), TimeDelta::Zero(),
>+                       Timestamp::ms(0) + TimeDelta::ms(20));
>+  EXPECT_EQ(TimeDelta::ms(10), rtt_stats_.min_rtt());
>+  rtt_stats_.UpdateRtt(TimeDelta::ms(50), TimeDelta::Zero(),
>+                       Timestamp::ms(0) + TimeDelta::ms(30));
>+  EXPECT_EQ(TimeDelta::ms(10), rtt_stats_.min_rtt());
>+  rtt_stats_.UpdateRtt(TimeDelta::ms(50), TimeDelta::Zero(),
>+                       Timestamp::ms(0) + TimeDelta::ms(40));
>+  EXPECT_EQ(TimeDelta::ms(10), rtt_stats_.min_rtt());
>+  // Verify that ack_delay does not go into recording of min_rtt_.
>+  rtt_stats_.UpdateRtt(TimeDelta::ms(7), TimeDelta::ms(2),
>+                       Timestamp::ms(0) + TimeDelta::ms(50));
>+  EXPECT_EQ(TimeDelta::ms(7), rtt_stats_.min_rtt());
>+}
>+
>+TEST_F(RttStatsTest, ExpireSmoothedMetrics) {
>+  TimeDelta initial_rtt = TimeDelta::ms(10);
>+  rtt_stats_.UpdateRtt(initial_rtt, TimeDelta::Zero(), Timestamp::ms(0));
>+  EXPECT_EQ(initial_rtt, rtt_stats_.min_rtt());
>+  EXPECT_EQ(initial_rtt, rtt_stats_.smoothed_rtt());
>+
>+  EXPECT_EQ(0.5 * initial_rtt, rtt_stats_.mean_deviation());
>+
>+  // Update once with a 20ms RTT.
>+  TimeDelta doubled_rtt = 2 * initial_rtt;
>+  rtt_stats_.UpdateRtt(doubled_rtt, TimeDelta::Zero(), Timestamp::ms(0));
>+  EXPECT_EQ(1.125 * initial_rtt, rtt_stats_.smoothed_rtt());
>+
>+  // Expire the smoothed metrics, increasing smoothed rtt and mean deviation.
>+  rtt_stats_.ExpireSmoothedMetrics();
>+  EXPECT_EQ(doubled_rtt, rtt_stats_.smoothed_rtt());
>+  EXPECT_EQ(0.875 * initial_rtt, rtt_stats_.mean_deviation());
>+
>+  // Now go back down to 5ms and expire the smoothed metrics, and ensure the
>+  // mean deviation increases to 15ms.
>+  TimeDelta half_rtt = 0.5 * initial_rtt;
>+  rtt_stats_.UpdateRtt(half_rtt, TimeDelta::Zero(), Timestamp::ms(0));
>+  EXPECT_GT(doubled_rtt, rtt_stats_.smoothed_rtt());
>+  EXPECT_LT(initial_rtt, rtt_stats_.mean_deviation());
>+}
>+
>+TEST_F(RttStatsTest, UpdateRttWithBadSendDeltas) {
>+  // Make sure we ignore bad RTTs.
>+
>+  TimeDelta initial_rtt = TimeDelta::ms(10);
>+  rtt_stats_.UpdateRtt(initial_rtt, TimeDelta::Zero(), Timestamp::ms(0));
>+  EXPECT_EQ(initial_rtt, rtt_stats_.min_rtt());
>+  EXPECT_EQ(initial_rtt, rtt_stats_.smoothed_rtt());
>+
>+  std::vector<TimeDelta> bad_send_deltas;
>+  bad_send_deltas.push_back(TimeDelta::Zero());
>+  bad_send_deltas.push_back(TimeDelta::PlusInfinity());
>+  bad_send_deltas.push_back(TimeDelta::us(-1000));
>+
>+  for (TimeDelta bad_send_delta : bad_send_deltas) {
>+    rtt_stats_.UpdateRtt(bad_send_delta, TimeDelta::Zero(), Timestamp::ms(0));
>+    EXPECT_EQ(initial_rtt, rtt_stats_.min_rtt());
>+    EXPECT_EQ(initial_rtt, rtt_stats_.smoothed_rtt());
>+  }
>+}
>+
>+TEST_F(RttStatsTest, ResetAfterConnectionMigrations) {
>+  rtt_stats_.UpdateRtt(TimeDelta::ms(300), TimeDelta::ms(100),
>+                       Timestamp::ms(0));
>+  EXPECT_EQ(TimeDelta::ms(200), rtt_stats_.latest_rtt());
>+  EXPECT_EQ(TimeDelta::ms(200), rtt_stats_.smoothed_rtt());
>+  EXPECT_EQ(TimeDelta::ms(300), rtt_stats_.min_rtt());
>+
>+  // Reset rtt stats on connection migrations.
>+  rtt_stats_.OnConnectionMigration();
>+  EXPECT_EQ(TimeDelta::Zero(), rtt_stats_.latest_rtt());
>+  EXPECT_EQ(TimeDelta::Zero(), rtt_stats_.smoothed_rtt());
>+  EXPECT_EQ(TimeDelta::Zero(), rtt_stats_.min_rtt());
>+}
>+
>+}  // namespace test
>+}  // namespace bbr
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/test/bbr_printer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/test/bbr_printer.cc
>new file mode 100644
>index 00000000000..440d49d25a2
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/test/bbr_printer.cc
>@@ -0,0 +1,57 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#include "modules/congestion_controller/bbr/test/bbr_printer.h"
>+
>+namespace webrtc {
>+
>+BbrStatePrinter::BbrStatePrinter() = default;
>+BbrStatePrinter::~BbrStatePrinter() = default;
>+
>+void BbrStatePrinter::Attach(bbr::BbrNetworkController* controller) {
>+  controller_ = controller;
>+}
>+
>+bool BbrStatePrinter::Attached() const {
>+  return controller_ != nullptr;
>+}
>+
>+void BbrStatePrinter::PrintHeaders(FILE* out) {
>+  fprintf(out, "bbr_mode bbr_recovery_state round_trip_count gain_cycle_index");
>+}
>+
>+void BbrStatePrinter::PrintValues(FILE* out) {
>+  RTC_CHECK(controller_);
>+  bbr::BbrNetworkController::DebugState debug(*controller_);
>+  fprintf(out, "%i %i %i %i", debug.mode, debug.recovery_state,
>+          static_cast<int>(debug.round_trip_count), debug.gain_cycle_index);
>+}
>+
>+NetworkControlUpdate BbrStatePrinter::GetState(Timestamp at_time) const {
>+  RTC_CHECK(controller_);
>+  return controller_->CreateRateUpdate(at_time);
>+}
>+
>+BbrDebugFactory::BbrDebugFactory(BbrStatePrinter* printer)
>+    : printer_(printer) {}
>+
>+std::unique_ptr<NetworkControllerInterface> BbrDebugFactory::Create(
>+    NetworkControllerConfig config) {
>+  RTC_CHECK(controller_ == nullptr);
>+  auto controller = BbrNetworkControllerFactory::Create(config);
>+  controller_ = static_cast<bbr::BbrNetworkController*>(controller.get());
>+  printer_->Attach(controller_);
>+  return controller;
>+}
>+
>+bbr::BbrNetworkController* BbrDebugFactory::BbrController() {
>+  return controller_;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/test/bbr_printer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/test/bbr_printer.h
>new file mode 100644
>index 00000000000..75db53d866b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/test/bbr_printer.h
>@@ -0,0 +1,49 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#ifndef MODULES_CONGESTION_CONTROLLER_BBR_TEST_BBR_PRINTER_H_
>+#define MODULES_CONGESTION_CONTROLLER_BBR_TEST_BBR_PRINTER_H_
>+
>+#include <memory>
>+
>+#include "modules/congestion_controller/bbr/bbr_factory.h"
>+#include "modules/congestion_controller/bbr/bbr_network_controller.h"
>+#include "modules/congestion_controller/test/controller_printer.h"
>+
>+namespace webrtc {
>+class BbrStatePrinter : public DebugStatePrinter {
>+ public:
>+  BbrStatePrinter();
>+  ~BbrStatePrinter() override;
>+  void Attach(bbr::BbrNetworkController*);
>+  bool Attached() const override;
>+
>+  void PrintHeaders(FILE* out) override;
>+  void PrintValues(FILE* out) override;
>+
>+  NetworkControlUpdate GetState(Timestamp at_time) const override;
>+
>+ private:
>+  bbr::BbrNetworkController* controller_ = nullptr;
>+};
>+
>+class BbrDebugFactory : public BbrNetworkControllerFactory {
>+ public:
>+  explicit BbrDebugFactory(BbrStatePrinter* printer);
>+  std::unique_ptr<NetworkControllerInterface> Create(
>+      NetworkControllerConfig config) override;
>+  bbr::BbrNetworkController* BbrController();
>+
>+ private:
>+  BbrStatePrinter* printer_;
>+  bbr::BbrNetworkController* controller_ = nullptr;
>+};
>+}  // namespace webrtc
>+
>+#endif  // MODULES_CONGESTION_CONTROLLER_BBR_TEST_BBR_PRINTER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/windowed_filter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/windowed_filter.h
>new file mode 100644
>index 00000000000..14185a53065
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/windowed_filter.h
>@@ -0,0 +1,168 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#ifndef MODULES_CONGESTION_CONTROLLER_BBR_WINDOWED_FILTER_H_
>+#define MODULES_CONGESTION_CONTROLLER_BBR_WINDOWED_FILTER_H_
>+
>+// From the Quic BBR implementation in Chromium
>+
>+// Implements Kathleen Nichols' algorithm for tracking the minimum (or maximum)
>+// estimate of a stream of samples over some fixed time interval. (E.g.,
>+// the minimum RTT over the past five minutes.) The algorithm keeps track of
>+// the best, second best, and third best min (or max) estimates, maintaining an
>+// invariant that the measurement time of the n'th best >= n-1'th best.
>+
>+// The algorithm works as follows. On a reset, all three estimates are set to
>+// the same sample. The second best estimate is then recorded in the second
>+// quarter of the window, and a third best estimate is recorded in the second
>+// half of the window, bounding the worst case error when the true min is
>+// monotonically increasing (or true max is monotonically decreasing) over the
>+// window.
>+//
>+// A new best sample replaces all three estimates, since the new best is lower
>+// (or higher) than everything else in the window and it is the most recent.
>+// The window thus effectively gets reset on every new min. The same property
>+// holds true for second best and third best estimates. Specifically, when a
>+// sample arrives that is better than the second best but not better than the
>+// best, it replaces the second and third best estimates but not the best
>+// estimate. Similarly, a sample that is better than the third best estimate
>+// but not the other estimates replaces only the third best estimate.
>+//
>+// Finally, when the best expires, it is replaced by the second best, which in
>+// turn is replaced by the third best. The newest sample replaces the third
>+// best.
>+
>+namespace webrtc {
>+namespace bbr {
>+
>+// Compares two values and returns true if the first is less than or equal
>+// to the second.
>+template <class T>
>+struct MinFilter {
>+  bool operator()(const T& lhs, const T& rhs) const { return lhs <= rhs; }
>+};
>+
>+// Compares two values and returns true if the first is greater than or equal
>+// to the second.
>+template <class T>
>+struct MaxFilter {
>+  bool operator()(const T& lhs, const T& rhs) const { return lhs >= rhs; }
>+};
>+
>+// Use the following to construct a windowed filter object of type T.
>+// For example, a min filter using Timestamp as the time type:
>+//   WindowedFilter<T, MinFilter<T>, Timestamp, TimeDelta>
>+//   ObjectName;
>+// A max filter using 64-bit integers as the time type:
>+//   WindowedFilter<T, MaxFilter<T>, uint64_t, int64_t> ObjectName;
>+// Specifically, this template takes four arguments:
>+// 1. T -- type of the measurement that is being filtered.
>+// 2. Compare -- MinFilter<T> or MaxFilter<T>, depending on the type of filter
>+//    desired.
>+// 3. TimeT -- the type used to represent timestamps.
>+// 4. TimeDeltaT -- the type used to represent continuous time intervals between
>+//    two timestamps.  Has to be the type of (a - b) if both |a| and |b| are
>+//    of type TimeT.
>+template <class T, class Compare, typename TimeT, typename TimeDeltaT>
>+class WindowedFilter {
>+ public:
>+  // |window_length| is the period after which a best estimate expires.
>+  // |zero_value| is used as the uninitialized value for objects of T.
>+  // Importantly, |zero_value| should be an invalid value for a true sample.
>+  WindowedFilter(TimeDeltaT window_length, T zero_value, TimeT zero_time)
>+      : window_length_(window_length),
>+        zero_value_(zero_value),
>+        estimates_{Sample(zero_value_, zero_time),
>+                   Sample(zero_value_, zero_time),
>+                   Sample(zero_value_, zero_time)} {}
>+
>+  // Changes the window length.  Does not update any current samples.
>+  void SetWindowLength(TimeDeltaT window_length) {
>+    window_length_ = window_length;
>+  }
>+
>+  // Updates best estimates with |sample|, and expires and updates best
>+  // estimates as necessary.
>+  void Update(T new_sample, TimeT new_time) {
>+    // Reset all estimates if they have not yet been initialized, if new sample
>+    // is a new best, or if the newest recorded estimate is too old.
>+    if (estimates_[0].sample == zero_value_ ||
>+        Compare()(new_sample, estimates_[0].sample) ||
>+        new_time - estimates_[2].time > window_length_) {
>+      Reset(new_sample, new_time);
>+      return;
>+    }
>+
>+    if (Compare()(new_sample, estimates_[1].sample)) {
>+      estimates_[1] = Sample(new_sample, new_time);
>+      estimates_[2] = estimates_[1];
>+    } else if (Compare()(new_sample, estimates_[2].sample)) {
>+      estimates_[2] = Sample(new_sample, new_time);
>+    }
>+
>+    // Expire and update estimates as necessary.
>+    if (new_time - estimates_[0].time > window_length_) {
>+      // The best estimate hasn't been updated for an entire window, so promote
>+      // second and third best estimates.
>+      estimates_[0] = estimates_[1];
>+      estimates_[1] = estimates_[2];
>+      estimates_[2] = Sample(new_sample, new_time);
>+      // Need to iterate one more time. Check if the new best estimate is
>+      // outside the window as well, since it may also have been recorded a
>+      // long time ago. Don't need to iterate once more since we cover that
>+      // case at the beginning of the method.
>+      if (new_time - estimates_[0].time > window_length_) {
>+        estimates_[0] = estimates_[1];
>+        estimates_[1] = estimates_[2];
>+      }
>+      return;
>+    }
>+    if (estimates_[1].sample == estimates_[0].sample &&
>+        new_time - estimates_[1].time > window_length_ >> 2) {
>+      // A quarter of the window has passed without a better sample, so the
>+      // second-best estimate is taken from the second quarter of the window.
>+      estimates_[2] = estimates_[1] = Sample(new_sample, new_time);
>+      return;
>+    }
>+
>+    if (estimates_[2].sample == estimates_[1].sample &&
>+        new_time - estimates_[2].time > window_length_ >> 1) {
>+      // We've passed a half of the window without a better estimate, so take
>+      // a third-best estimate from the second half of the window.
>+      estimates_[2] = Sample(new_sample, new_time);
>+    }
>+  }
>+
>+  // Resets all estimates to new sample.
>+  void Reset(T new_sample, TimeT new_time) {
>+    estimates_[0] = estimates_[1] = estimates_[2] =
>+        Sample(new_sample, new_time);
>+  }
>+
>+  T GetBest() const { return estimates_[0].sample; }
>+  T GetSecondBest() const { return estimates_[1].sample; }
>+  T GetThirdBest() const { return estimates_[2].sample; }
>+
>+ private:
>+  struct Sample {
>+    T sample;
>+    TimeT time;
>+    Sample(T init_sample, TimeT init_time)
>+        : sample(init_sample), time(init_time) {}
>+  };
>+
>+  TimeDeltaT window_length_;  // Time length of window.
>+  T zero_value_;              // Uninitialized value of T.
>+  Sample estimates_[3];       // Best estimate is element 0.
>+};
>+
>+}  // namespace bbr
>+}  // namespace webrtc
>+
>+#endif  // MODULES_CONGESTION_CONTROLLER_BBR_WINDOWED_FILTER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/windowed_filter_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/windowed_filter_unittest.cc
>new file mode 100644
>index 00000000000..3eb5177b111
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bbr/windowed_filter_unittest.cc
>@@ -0,0 +1,360 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/congestion_controller/bbr/windowed_filter.h"
>+
>+#include "api/units/data_rate.h"
>+#include "api/units/time_delta.h"
>+
>+#include "modules/congestion_controller/bbr/rtt_stats.h"
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+namespace bbr {
>+namespace test {
>+class WindowedFilterTest : public ::testing::Test {
>+ public:
>+  // Set the window to 99ms, so 25ms is more than a quarter rtt.
>+  WindowedFilterTest()
>+      : windowed_min_rtt_(99, TimeDelta::Zero(), 0),
>+        windowed_max_bw_(99, DataRate::Zero(), 0) {}
>+
>+  // Sets up windowed_min_rtt_ to have the following values:
>+  // Best = 20ms, recorded at 25ms
>+  // Second best = 40ms, recorded at 75ms
>+  // Third best = 50ms, recorded at 100ms
>+  void InitializeMinFilter() {
>+    int64_t now_ms = 0;
>+    TimeDelta rtt_sample = TimeDelta::ms(10);
>+    for (int i = 0; i < 5; ++i) {
>+      windowed_min_rtt_.Update(rtt_sample, now_ms);
>+      RTC_LOG(LS_VERBOSE) << "i: " << i << " sample: " << ToString(rtt_sample)
>+                          << " mins: "
>+                          << " " << ToString(windowed_min_rtt_.GetBest()) << " "
>+                          << ToString(windowed_min_rtt_.GetSecondBest()) << " "
>+                          << ToString(windowed_min_rtt_.GetThirdBest());
>+      now_ms += 25;
>+      rtt_sample = rtt_sample + TimeDelta::ms(10);
>+    }
>+    EXPECT_EQ(TimeDelta::ms(20), windowed_min_rtt_.GetBest());
>+    EXPECT_EQ(TimeDelta::ms(40), windowed_min_rtt_.GetSecondBest());
>+    EXPECT_EQ(TimeDelta::ms(50), windowed_min_rtt_.GetThirdBest());
>+  }
>+
>+  // Sets up windowed_max_bw_ to have the following values:
>+  // Best = 900 bps, recorded at 25ms
>+  // Second best = 700 bps, recorded at 75ms
>+  // Third best = 600 bps, recorded at 100ms
>+  void InitializeMaxFilter() {
>+    int64_t now_ms = 0;
>+    DataRate bw_sample = DataRate::bps(1000);
>+    for (int i = 0; i < 5; ++i) {
>+      windowed_max_bw_.Update(bw_sample, now_ms);
>+      RTC_LOG(LS_VERBOSE) << "i: " << i << " sample: " << ToString(bw_sample)
>+                          << " maxs: "
>+                          << " " << ToString(windowed_max_bw_.GetBest()) << " "
>+                          << ToString(windowed_max_bw_.GetSecondBest()) << " "
>+                          << ToString(windowed_max_bw_.GetThirdBest());
>+      now_ms += 25;
>+      bw_sample = DataRate::bps(bw_sample.bps() - 100);
>+    }
>+    EXPECT_EQ(DataRate::bps(900), windowed_max_bw_.GetBest());
>+    EXPECT_EQ(DataRate::bps(700), windowed_max_bw_.GetSecondBest());
>+    EXPECT_EQ(DataRate::bps(600), windowed_max_bw_.GetThirdBest());
>+  }
>+
>+ protected:
>+  WindowedFilter<TimeDelta, MinFilter<TimeDelta>, int64_t, int64_t>
>+      windowed_min_rtt_;
>+  WindowedFilter<DataRate, MaxFilter<DataRate>, int64_t, int64_t>
>+      windowed_max_bw_;
>+};
>+
>+namespace {
>+// Test helper function: updates the filter with a lot of small values in order
>+// to ensure that it is not susceptible to noise.
>+void UpdateWithIrrelevantSamples(
>+    WindowedFilter<uint64_t, MaxFilter<uint64_t>, uint64_t, uint64_t>* filter,
>+    uint64_t max_value,
>+    uint64_t time) {
>+  for (uint64_t i = 0; i < 1000; i++) {
>+    filter->Update(i % max_value, time);
>+  }
>+}
>+}  // namespace
>+
>+TEST_F(WindowedFilterTest, UninitializedEstimates) {
>+  EXPECT_EQ(TimeDelta::Zero(), windowed_min_rtt_.GetBest());
>+  EXPECT_EQ(TimeDelta::Zero(), windowed_min_rtt_.GetSecondBest());
>+  EXPECT_EQ(TimeDelta::Zero(), windowed_min_rtt_.GetThirdBest());
>+  EXPECT_EQ(DataRate::Zero(), windowed_max_bw_.GetBest());
>+  EXPECT_EQ(DataRate::Zero(), windowed_max_bw_.GetSecondBest());
>+  EXPECT_EQ(DataRate::Zero(), windowed_max_bw_.GetThirdBest());
>+}
>+
>+TEST_F(WindowedFilterTest, MonotonicallyIncreasingMin) {
>+  int64_t now_ms = 0;
>+  TimeDelta rtt_sample = TimeDelta::ms(10);
>+  windowed_min_rtt_.Update(rtt_sample, now_ms);
>+  EXPECT_EQ(TimeDelta::ms(10), windowed_min_rtt_.GetBest());
>+
>+  // Gradually increase the rtt samples and ensure the windowed min rtt starts
>+  // rising.
>+  for (int i = 0; i < 6; ++i) {
>+    now_ms += 25;
>+    rtt_sample = rtt_sample + TimeDelta::ms(10);
>+    windowed_min_rtt_.Update(rtt_sample, now_ms);
>+    RTC_LOG(LS_VERBOSE) << "i: " << i << " sample: " << rtt_sample.ms()
>+                        << " mins: "
>+                        << " " << windowed_min_rtt_.GetBest().ms() << " "
>+                        << windowed_min_rtt_.GetSecondBest().ms() << " "
>+                        << windowed_min_rtt_.GetThirdBest().ms();
>+    if (i < 3) {
>+      EXPECT_EQ(TimeDelta::ms(10), windowed_min_rtt_.GetBest());
>+    } else if (i == 3) {
>+      EXPECT_EQ(TimeDelta::ms(20), windowed_min_rtt_.GetBest());
>+    } else if (i < 6) {
>+      EXPECT_EQ(TimeDelta::ms(40), windowed_min_rtt_.GetBest());
>+    }
>+  }
>+}
>+
>+TEST_F(WindowedFilterTest, MonotonicallyDecreasingMax) {
>+  int64_t now_ms = 0;
>+  DataRate bw_sample = DataRate::bps(1000);
>+  windowed_max_bw_.Update(bw_sample, now_ms);
>+  EXPECT_EQ(DataRate::bps(1000), windowed_max_bw_.GetBest());
>+
>+  // Gradually decrease the bw samples and ensure the windowed max bw starts
>+  // decreasing.
>+  for (int i = 0; i < 6; ++i) {
>+    now_ms += 25;
>+    bw_sample = DataRate::bps(bw_sample.bps() - 100);
>+    windowed_max_bw_.Update(bw_sample, now_ms);
>+    RTC_LOG(LS_VERBOSE) << "i: " << i << " sample: " << bw_sample.bps()
>+                        << " maxs: "
>+                        << " " << windowed_max_bw_.GetBest().bps() << " "
>+                        << windowed_max_bw_.GetSecondBest().bps() << " "
>+                        << windowed_max_bw_.GetThirdBest().bps();
>+    if (i < 3) {
>+      EXPECT_EQ(DataRate::bps(1000), windowed_max_bw_.GetBest());
>+    } else if (i == 3) {
>+      EXPECT_EQ(DataRate::bps(900), windowed_max_bw_.GetBest());
>+    } else if (i < 6) {
>+      EXPECT_EQ(DataRate::bps(700), windowed_max_bw_.GetBest());
>+    }
>+  }
>+}
>+
>+TEST_F(WindowedFilterTest, SampleChangesThirdBestMin) {
>+  InitializeMinFilter();
>+  // RTT sample lower than the third-choice min-rtt sets that, but nothing else.
>+  TimeDelta rtt_sample = windowed_min_rtt_.GetThirdBest() - TimeDelta::ms(5);
>+  // This assert is necessary to avoid triggering -Wstrict-overflow
>+  // See crbug/616957
>+  ASSERT_GT(windowed_min_rtt_.GetThirdBest(), TimeDelta::ms(5));
>+  // Latest sample was recorded at 100ms.
>+  int64_t now_ms = 101;
>+  windowed_min_rtt_.Update(rtt_sample, now_ms);
>+  EXPECT_EQ(rtt_sample, windowed_min_rtt_.GetThirdBest());
>+  EXPECT_EQ(TimeDelta::ms(40), windowed_min_rtt_.GetSecondBest());
>+  EXPECT_EQ(TimeDelta::ms(20), windowed_min_rtt_.GetBest());
>+}
>+
>+TEST_F(WindowedFilterTest, SampleChangesThirdBestMax) {
>+  InitializeMaxFilter();
>+  // BW sample higher than the third-choice max sets that, but nothing else.
>+  DataRate bw_sample =
>+      DataRate::bps(windowed_max_bw_.GetThirdBest().bps() + 50);
>+  // Latest sample was recorded at 100ms.
>+  int64_t now_ms = 101;
>+  windowed_max_bw_.Update(bw_sample, now_ms);
>+  EXPECT_EQ(bw_sample, windowed_max_bw_.GetThirdBest());
>+  EXPECT_EQ(DataRate::bps(700), windowed_max_bw_.GetSecondBest());
>+  EXPECT_EQ(DataRate::bps(900), windowed_max_bw_.GetBest());
>+}
>+
>+TEST_F(WindowedFilterTest, SampleChangesSecondBestMin) {
>+  InitializeMinFilter();
>+  // RTT sample lower than the second-choice min sets that and also
>+  // the third-choice min.
>+  TimeDelta rtt_sample = windowed_min_rtt_.GetSecondBest() - TimeDelta::ms(5);
>+  // This assert is necessary to avoid triggering -Wstrict-overflow
>+  // See crbug/616957
>+  ASSERT_GT(windowed_min_rtt_.GetSecondBest(), TimeDelta::ms(5));
>+  // Latest sample was recorded at 100ms.
>+  int64_t now_ms = 101;
>+  windowed_min_rtt_.Update(rtt_sample, now_ms);
>+  EXPECT_EQ(rtt_sample, windowed_min_rtt_.GetThirdBest());
>+  EXPECT_EQ(rtt_sample, windowed_min_rtt_.GetSecondBest());
>+  EXPECT_EQ(TimeDelta::ms(20), windowed_min_rtt_.GetBest());
>+}
>+
>+TEST_F(WindowedFilterTest, SampleChangesSecondBestMax) {
>+  InitializeMaxFilter();
>+  // BW sample higher than the second-choice max sets that and also
>+  // the third-choice max.
>+  DataRate bw_sample =
>+      DataRate::bps(windowed_max_bw_.GetSecondBest().bps() + 50);
>+
>+  // Latest sample was recorded at 100ms.
>+  int64_t now_ms = 101;
>+  windowed_max_bw_.Update(bw_sample, now_ms);
>+  EXPECT_EQ(bw_sample, windowed_max_bw_.GetThirdBest());
>+  EXPECT_EQ(bw_sample, windowed_max_bw_.GetSecondBest());
>+  EXPECT_EQ(DataRate::bps(900), windowed_max_bw_.GetBest());
>+}
>+
>+TEST_F(WindowedFilterTest, SampleChangesAllMins) {
>+  InitializeMinFilter();
>+  // RTT sample lower than the first-choice min-rtt sets that and also
>+  // the second and third-choice mins.
>+  TimeDelta rtt_sample = windowed_min_rtt_.GetBest() - TimeDelta::ms(5);
>+  // This assert is necessary to avoid triggering -Wstrict-overflow
>+  // See crbug/616957
>+  ASSERT_GT(windowed_min_rtt_.GetBest(), TimeDelta::ms(5));
>+  // Latest sample was recorded at 100ms.
>+  int64_t now_ms = 101;
>+  windowed_min_rtt_.Update(rtt_sample, now_ms);
>+  EXPECT_EQ(rtt_sample, windowed_min_rtt_.GetThirdBest());
>+  EXPECT_EQ(rtt_sample, windowed_min_rtt_.GetSecondBest());
>+  EXPECT_EQ(rtt_sample, windowed_min_rtt_.GetBest());
>+}
>+
>+TEST_F(WindowedFilterTest, SampleChangesAllMaxs) {
>+  InitializeMaxFilter();
>+  // BW sample higher than the first-choice max sets that and also
>+  // the second and third-choice maxs.
>+  DataRate bw_sample = DataRate::bps(windowed_max_bw_.GetBest().bps() + 50);
>+  // Latest sample was recorded at 100ms.
>+  int64_t now_ms = 101;
>+  windowed_max_bw_.Update(bw_sample, now_ms);
>+  EXPECT_EQ(bw_sample, windowed_max_bw_.GetThirdBest());
>+  EXPECT_EQ(bw_sample, windowed_max_bw_.GetSecondBest());
>+  EXPECT_EQ(bw_sample, windowed_max_bw_.GetBest());
>+}
>+
>+TEST_F(WindowedFilterTest, ExpireBestMin) {
>+  InitializeMinFilter();
>+  TimeDelta old_third_best = windowed_min_rtt_.GetThirdBest();
>+  TimeDelta old_second_best = windowed_min_rtt_.GetSecondBest();
>+  TimeDelta rtt_sample = old_third_best + TimeDelta::ms(5);
>+  // Best min sample was recorded at 25ms, so expiry time is 124ms.
>+  int64_t now_ms = 125;
>+  windowed_min_rtt_.Update(rtt_sample, now_ms);
>+  EXPECT_EQ(rtt_sample, windowed_min_rtt_.GetThirdBest());
>+  EXPECT_EQ(old_third_best, windowed_min_rtt_.GetSecondBest());
>+  EXPECT_EQ(old_second_best, windowed_min_rtt_.GetBest());
>+}
>+
>+TEST_F(WindowedFilterTest, ExpireBestMax) {
>+  InitializeMaxFilter();
>+  DataRate old_third_best = windowed_max_bw_.GetThirdBest();
>+  DataRate old_second_best = windowed_max_bw_.GetSecondBest();
>+  DataRate bw_sample = DataRate::bps(old_third_best.bps() - 50);
>+  // Best max sample was recorded at 25ms, so expiry time is 124ms.
>+  int64_t now_ms = 125;
>+  windowed_max_bw_.Update(bw_sample, now_ms);
>+  EXPECT_EQ(bw_sample, windowed_max_bw_.GetThirdBest());
>+  EXPECT_EQ(old_third_best, windowed_max_bw_.GetSecondBest());
>+  EXPECT_EQ(old_second_best, windowed_max_bw_.GetBest());
>+}
>+
>+TEST_F(WindowedFilterTest, ExpireSecondBestMin) {
>+  InitializeMinFilter();
>+  TimeDelta old_third_best = windowed_min_rtt_.GetThirdBest();
>+  TimeDelta rtt_sample = old_third_best + TimeDelta::ms(5);
>+  // Second best min sample was recorded at 75ms, so expiry time is 174ms.
>+  int64_t now_ms = 175;
>+  windowed_min_rtt_.Update(rtt_sample, now_ms);
>+  EXPECT_EQ(rtt_sample, windowed_min_rtt_.GetThirdBest());
>+  EXPECT_EQ(rtt_sample, windowed_min_rtt_.GetSecondBest());
>+  EXPECT_EQ(old_third_best, windowed_min_rtt_.GetBest());
>+}
>+
>+TEST_F(WindowedFilterTest, ExpireSecondBestMax) {
>+  InitializeMaxFilter();
>+  DataRate old_third_best = windowed_max_bw_.GetThirdBest();
>+  DataRate bw_sample = DataRate::bps(old_third_best.bps() - 50);
>+  // Second best max sample was recorded at 75ms, so expiry time is 174ms.
>+  int64_t now_ms = 175;
>+  windowed_max_bw_.Update(bw_sample, now_ms);
>+  EXPECT_EQ(bw_sample, windowed_max_bw_.GetThirdBest());
>+  EXPECT_EQ(bw_sample, windowed_max_bw_.GetSecondBest());
>+  EXPECT_EQ(old_third_best, windowed_max_bw_.GetBest());
>+}
>+
>+TEST_F(WindowedFilterTest, ExpireAllMins) {
>+  InitializeMinFilter();
>+  TimeDelta rtt_sample = windowed_min_rtt_.GetThirdBest() + TimeDelta::ms(5);
>+  // This assert is necessary to avoid triggering -Wstrict-overflow
>+  // See crbug/616957
>+  ASSERT_LT(windowed_min_rtt_.GetThirdBest(), TimeDelta::PlusInfinity());
>+  // Third best min sample was recorded at 100ms, so expiry time is 199ms.
>+  int64_t now_ms = 200;
>+  windowed_min_rtt_.Update(rtt_sample, now_ms);
>+  EXPECT_EQ(rtt_sample, windowed_min_rtt_.GetThirdBest());
>+  EXPECT_EQ(rtt_sample, windowed_min_rtt_.GetSecondBest());
>+  EXPECT_EQ(rtt_sample, windowed_min_rtt_.GetBest());
>+}
>+
>+TEST_F(WindowedFilterTest, ExpireAllMaxs) {
>+  InitializeMaxFilter();
>+  DataRate bw_sample =
>+      DataRate::bps(windowed_max_bw_.GetThirdBest().bps() - 50);
>+  // Third best max sample was recorded at 100ms, so expiry time is 199ms.
>+  int64_t now_ms = 200;
>+  windowed_max_bw_.Update(bw_sample, now_ms);
>+  EXPECT_EQ(bw_sample, windowed_max_bw_.GetThirdBest());
>+  EXPECT_EQ(bw_sample, windowed_max_bw_.GetSecondBest());
>+  EXPECT_EQ(bw_sample, windowed_max_bw_.GetBest());
>+}
>+
>+// Test the windowed filter where the time used is an exact counter instead of a
>+// timestamp.  This is useful if, for example, the time is measured in round
>+// trips.
>+TEST_F(WindowedFilterTest, ExpireCounterBasedMax) {
>+  // Create a window which starts at t = 0 and expires after two cycles.
>+  WindowedFilter<uint64_t, MaxFilter<uint64_t>, uint64_t, uint64_t> max_filter(
>+      2, 0, 0);
>+
>+  const uint64_t kBest = 50000;
>+  // Insert 50000 at t = 1.
>+  max_filter.Update(50000, 1);
>+  EXPECT_EQ(kBest, max_filter.GetBest());
>+  UpdateWithIrrelevantSamples(&max_filter, 20, 1);
>+  EXPECT_EQ(kBest, max_filter.GetBest());
>+
>+  // Insert 40000 at t = 2.  Nothing is expected to expire.
>+  max_filter.Update(40000, 2);
>+  EXPECT_EQ(kBest, max_filter.GetBest());
>+  UpdateWithIrrelevantSamples(&max_filter, 20, 2);
>+  EXPECT_EQ(kBest, max_filter.GetBest());
>+
>+  // Insert 30000 at t = 3.  Nothing is expected to expire yet.
>+  max_filter.Update(30000, 3);
>+  EXPECT_EQ(kBest, max_filter.GetBest());
>+  UpdateWithIrrelevantSamples(&max_filter, 20, 3);
>+  EXPECT_EQ(kBest, max_filter.GetBest());
>+  RTC_LOG(LS_VERBOSE) << max_filter.GetSecondBest();
>+  RTC_LOG(LS_VERBOSE) << max_filter.GetThirdBest();
>+
>+  // Insert 20000 at t = 4.  50000 at t = 1 expires, so 40000 becomes the new
>+  // maximum.
>+  const uint64_t kNewBest = 40000;
>+  max_filter.Update(20000, 4);
>+  EXPECT_EQ(kNewBest, max_filter.GetBest());
>+  UpdateWithIrrelevantSamples(&max_filter, 20, 4);
>+  EXPECT_EQ(kNewBest, max_filter.GetBest());
>+}
>+
>+}  // namespace test
>+}  // namespace bbr
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/congestion_window_pushback_controller.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/congestion_window_pushback_controller.cc
>new file mode 100644
>index 00000000000..daa51bddd30
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/congestion_window_pushback_controller.cc
>@@ -0,0 +1,93 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <string>
>+#include <algorithm>
>+
>+#include "modules/congestion_controller/congestion_window_pushback_controller.h"
>+#include "system_wrappers/include/field_trial.h"
>+
>+namespace webrtc {
>+
>+// When CongestionWindowPushback is enabled, the pacer is oblivious to
>+// the congestion window. The relation between outstanding data and
>+// the congestion window affects encoder allocations directly.
>+// This experiment is build on top of congestion window experiment.
>+const char kCongestionPushbackExperiment[] = "WebRTC-CongestionWindowPushback";
>+const uint32_t kDefaultMinPushbackTargetBitrateBps = 30000;
>+
>+bool ReadCongestionWindowPushbackExperimentParameter(
>+    uint32_t* min_pushback_target_bitrate_bps) {
>+  RTC_DCHECK(min_pushback_target_bitrate_bps);
>+  std::string experiment_string =
>+      webrtc::field_trial::FindFullName(kCongestionPushbackExperiment);
>+  int parsed_values = sscanf(experiment_string.c_str(), "Enabled-%" PRIu32,
>+                             min_pushback_target_bitrate_bps);
>+  if (parsed_values == 1) {
>+    RTC_CHECK_GE(*min_pushback_target_bitrate_bps, 0)
>+        << "Min pushback target bitrate must be greater than or equal to 0.";
>+    return true;
>+  }
>+  return false;
>+}
>+
>+CongestionWindowPushbackController::CongestionWindowPushbackController() {
>+  if (!ReadCongestionWindowPushbackExperimentParameter(
>+          &min_pushback_target_bitrate_bps_)) {
>+    min_pushback_target_bitrate_bps_ = kDefaultMinPushbackTargetBitrateBps;
>+  }
>+}
>+
>+void CongestionWindowPushbackController::UpdateOutstandingData(
>+    size_t outstanding_bytes) {
>+  outstanding_bytes_ = outstanding_bytes;
>+}
>+
>+void CongestionWindowPushbackController::UpdateMaxOutstandingData(
>+    size_t max_outstanding_bytes) {
>+  DataSize data_window = DataSize::bytes(max_outstanding_bytes);
>+  if (current_data_window_) {
>+    data_window = (data_window + current_data_window_.value()) / 2;
>+  }
>+  current_data_window_ = data_window;
>+}
>+
>+void CongestionWindowPushbackController::SetDataWindow(DataSize data_window) {
>+  current_data_window_ = data_window;
>+}
>+
>+uint32_t CongestionWindowPushbackController::UpdateTargetBitrate(
>+    uint32_t bitrate_bps) {
>+  if (!current_data_window_ || current_data_window_->IsZero())
>+    return bitrate_bps;
>+  double fill_ratio =
>+      outstanding_bytes_ / static_cast<double>(current_data_window_->bytes());
>+  if (fill_ratio > 1.5) {
>+    encoding_rate_ratio_ *= 0.9;
>+  } else if (fill_ratio > 1) {
>+    encoding_rate_ratio_ *= 0.95;
>+  } else if (fill_ratio < 0.1) {
>+    encoding_rate_ratio_ = 1.0;
>+  } else {
>+    encoding_rate_ratio_ *= 1.05;
>+    encoding_rate_ratio_ = std::min(encoding_rate_ratio_, 1.0);
>+  }
>+  uint32_t adjusted_target_bitrate_bps =
>+      static_cast<uint32_t>(bitrate_bps * encoding_rate_ratio_);
>+
>+  // Do not adjust below the minimum pushback bitrate but do obey if the
>+  // original estimate is below it.
>+  bitrate_bps = adjusted_target_bitrate_bps < min_pushback_target_bitrate_bps_
>+                    ? std::min(bitrate_bps, min_pushback_target_bitrate_bps_)
>+                    : adjusted_target_bitrate_bps;
>+  return bitrate_bps;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/congestion_window_pushback_controller.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/congestion_window_pushback_controller.h
>new file mode 100644
>index 00000000000..f608590583b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/congestion_window_pushback_controller.h
>@@ -0,0 +1,43 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_CONGESTION_CONTROLLER_CONGESTION_WINDOW_PUSHBACK_CONTROLLER_H_
>+#define MODULES_CONGESTION_CONTROLLER_CONGESTION_WINDOW_PUSHBACK_CONTROLLER_H_
>+
>+#include "api/transport/network_types.h"
>+#include "common_types.h"  // NOLINT(build/include)
>+#include "rtc_base/criticalsection.h"
>+#include "rtc_base/format_macros.h"
>+
>+namespace webrtc {
>+
>+// This class enables pushback from congestion window directly to video encoder.
>+// When the congestion window is filling up, the video encoder target bitrate
>+// will be reduced accordingly to accommodate the network changes. To avoid
>+// pausing video too frequently, a minimum encoder target bitrate threshold is
>+// used to prevent video pause due to a full congestion window.
>+class CongestionWindowPushbackController {
>+ public:
>+  CongestionWindowPushbackController();
>+  void UpdateOutstandingData(size_t outstanding_bytes);
>+  void UpdateMaxOutstandingData(size_t max_outstanding_bytes);
>+  uint32_t UpdateTargetBitrate(uint32_t bitrate_bps);
>+  void SetDataWindow(DataSize data_window);
>+
>+ private:
>+  absl::optional<DataSize> current_data_window_;
>+  size_t outstanding_bytes_ = 0;
>+  uint32_t min_pushback_target_bitrate_bps_;
>+  double encoding_rate_ratio_ = 1.0;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_CONGESTION_CONTROLLER_CONGESTION_WINDOW_PUSHBACK_CONTROLLER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/congestion_window_pushback_controller_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/congestion_window_pushback_controller_unittest.cc
>new file mode 100644
>index 00000000000..d9d6a79d62f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/congestion_window_pushback_controller_unittest.cc
>@@ -0,0 +1,65 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/congestion_controller/congestion_window_pushback_controller.h"
>+#include "test/gmock.h"
>+#include "test/gtest.h"
>+
>+using testing::_;
>+
>+namespace webrtc {
>+namespace test {
>+
>+class CongestionWindowPushbackControllerTest : public ::testing::Test {
>+ protected:
>+  CongestionWindowPushbackController cwnd_controller_;
>+};
>+
>+TEST_F(CongestionWindowPushbackControllerTest, FullCongestionWindow) {
>+  cwnd_controller_.UpdateOutstandingData(100000);
>+  cwnd_controller_.UpdateMaxOutstandingData(50000);
>+
>+  uint32_t bitrate_bps = 80000;
>+  bitrate_bps = cwnd_controller_.UpdateTargetBitrate(bitrate_bps);
>+  EXPECT_EQ(72000u, bitrate_bps);
>+
>+  cwnd_controller_.UpdateMaxOutstandingData(50000);
>+  bitrate_bps = cwnd_controller_.UpdateTargetBitrate(bitrate_bps);
>+  EXPECT_EQ(static_cast<uint32_t>(72000 * 0.9 * 0.9), bitrate_bps);
>+}
>+
>+TEST_F(CongestionWindowPushbackControllerTest, NormalCongestionWindow) {
>+  cwnd_controller_.UpdateOutstandingData(100000);
>+  cwnd_controller_.SetDataWindow(DataSize::bytes(200000));
>+
>+  uint32_t bitrate_bps = 80000;
>+  bitrate_bps = cwnd_controller_.UpdateTargetBitrate(bitrate_bps);
>+  EXPECT_EQ(80000u, bitrate_bps);
>+
>+  cwnd_controller_.UpdateMaxOutstandingData(20000);
>+  bitrate_bps = cwnd_controller_.UpdateTargetBitrate(bitrate_bps);
>+  EXPECT_EQ(80000u, bitrate_bps);
>+}
>+
>+TEST_F(CongestionWindowPushbackControllerTest, LowBitrate) {
>+  cwnd_controller_.UpdateOutstandingData(100000);
>+  cwnd_controller_.SetDataWindow(DataSize::bytes(50000));
>+
>+  uint32_t bitrate_bps = 35000;
>+  bitrate_bps = cwnd_controller_.UpdateTargetBitrate(bitrate_bps);
>+  EXPECT_EQ(static_cast<uint32_t>(35000 * 0.9), bitrate_bps);
>+
>+  cwnd_controller_.UpdateMaxOutstandingData(20000);
>+  bitrate_bps = cwnd_controller_.UpdateTargetBitrate(bitrate_bps);
>+  EXPECT_EQ(30000u, bitrate_bps);
>+}
>+
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/BUILD.gn
>new file mode 100644
>index 00000000000..9da21233427
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/BUILD.gn
>@@ -0,0 +1,194 @@
>+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+#
>+# Use of this source code is governed by a BSD-style license
>+# that can be found in the LICENSE file in the root of the source
>+# tree. An additional intellectual property rights grant can be found
>+# in the file PATENTS.  All contributing project authors may
>+# be found in the AUTHORS file in the root of the source tree.
>+
>+import("../../../webrtc.gni")
>+
>+config("bwe_test_logging") {
>+  if (rtc_enable_bwe_test_logging) {
>+    defines = [ "BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=1" ]
>+  } else {
>+    defines = [ "BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=0" ]
>+  }
>+}
>+
>+rtc_static_library("goog_cc") {
>+  configs += [ ":bwe_test_logging" ]
>+  sources = [
>+    "goog_cc_factory.cc",
>+    "goog_cc_network_control.cc",
>+    "goog_cc_network_control.h",
>+    "include/goog_cc_factory.h",
>+  ]
>+
>+  deps = [
>+    ":alr_detector",
>+    ":delay_based_bwe",
>+    ":estimators",
>+    ":probe_controller",
>+    "../..:module_api",
>+    "../../..:webrtc_common",
>+    "../../../api/transport:network_control",
>+    "../../../logging:rtc_event_log_api",
>+    "../../../logging:rtc_event_pacing",
>+    "../../../rtc_base:checks",
>+    "../../../rtc_base:rtc_base_approved",
>+    "../../../rtc_base/experiments:alr_experiment",
>+    "../../../system_wrappers",
>+    "../../../system_wrappers:field_trial_api",
>+    "../../../system_wrappers:metrics_api",
>+    "../../bitrate_controller",
>+    "../../remote_bitrate_estimator",
>+    "../../rtp_rtcp:rtp_rtcp_format",
>+    "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/types:optional",
>+  ]
>+}
>+
>+rtc_source_set("alr_detector") {
>+  sources = [
>+    "alr_detector.cc",
>+    "alr_detector.h",
>+  ]
>+  deps = [
>+    "../../..:webrtc_common",
>+    "../../../logging:rtc_event_log_api",
>+    "../../../logging:rtc_event_pacing",
>+    "../../../rtc_base:checks",
>+    "../../../rtc_base:rtc_base_approved",
>+    "../../../rtc_base/experiments:alr_experiment",
>+    "../../../system_wrappers:field_trial_api",
>+    "../../pacing:interval_budget",
>+    "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/types:optional",
>+  ]
>+}
>+rtc_source_set("estimators") {
>+  configs += [ ":bwe_test_logging" ]
>+  sources = [
>+    "acknowledged_bitrate_estimator.cc",
>+    "acknowledged_bitrate_estimator.h",
>+    "bitrate_estimator.cc",
>+    "bitrate_estimator.h",
>+    "delay_increase_detector_interface.h",
>+    "median_slope_estimator.cc",
>+    "median_slope_estimator.h",
>+    "probe_bitrate_estimator.cc",
>+    "probe_bitrate_estimator.h",
>+    "trendline_estimator.cc",
>+    "trendline_estimator.h",
>+  ]
>+
>+  # TODO(jschuh): Bug 1348: fix this warning.
>+  configs += [ "//build/config/compiler:no_size_t_to_int_warning" ]
>+
>+  deps = [
>+    "../../../logging:rtc_event_bwe",
>+    "../../../logging:rtc_event_log_api",
>+    "../../../rtc_base:checks",
>+    "../../../rtc_base:rtc_base_approved",
>+    "../../../rtc_base:rtc_numerics",
>+    "../../../rtc_base:safe_minmax",
>+    "../../../system_wrappers:field_trial_api",
>+    "../../../system_wrappers:metrics_api",
>+    "../../remote_bitrate_estimator",
>+    "../../rtp_rtcp:rtp_rtcp_format",
>+    "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/types:optional",
>+  ]
>+}
>+
>+rtc_source_set("delay_based_bwe") {
>+  configs += [ ":bwe_test_logging" ]
>+  sources = [
>+    "delay_based_bwe.cc",
>+    "delay_based_bwe.h",
>+  ]
>+
>+  deps = [
>+    ":estimators",
>+    "../../../logging:rtc_event_bwe",
>+    "../../../logging:rtc_event_log_api",
>+    "../../../rtc_base:checks",
>+    "../../../rtc_base:rtc_base_approved",
>+    "../../../system_wrappers:field_trial_api",
>+    "../../../system_wrappers:metrics_api",
>+    "../../pacing",
>+    "../../remote_bitrate_estimator",
>+    "//third_party/abseil-cpp/absl/memory",
>+  ]
>+}
>+
>+rtc_source_set("probe_controller") {
>+  sources = [
>+    "probe_controller.cc",
>+    "probe_controller.h",
>+  ]
>+
>+  deps = [
>+    "../../../api/transport:network_control",
>+    "../../../logging:rtc_event_log_api",
>+    "../../../logging:rtc_event_pacing",
>+    "../../../rtc_base:checks",
>+    "../../../rtc_base:rtc_base_approved",
>+    "../../../rtc_base/system:unused",
>+    "../../../system_wrappers:field_trial_api",
>+    "../../../system_wrappers:metrics_api",
>+    "//third_party/abseil-cpp/absl/types:optional",
>+  ]
>+}
>+
>+if (rtc_include_tests) {
>+  rtc_source_set("test_goog_cc_printer") {
>+    testonly = true
>+    sources = [
>+      "test/goog_cc_printer.cc",
>+      "test/goog_cc_printer.h",
>+    ]
>+    deps = [
>+      ":goog_cc",
>+      "..:test_controller_printer",
>+    ]
>+  }
>+  rtc_source_set("goog_cc_unittests") {
>+    testonly = true
>+
>+    sources = [
>+      "acknowledged_bitrate_estimator_unittest.cc",
>+      "alr_detector_unittest.cc",
>+      "delay_based_bwe_unittest.cc",
>+      "delay_based_bwe_unittest_helper.cc",
>+      "delay_based_bwe_unittest_helper.h",
>+      "median_slope_estimator_unittest.cc",
>+      "probe_bitrate_estimator_unittest.cc",
>+      "probe_controller_unittest.cc",
>+      "trendline_estimator_unittest.cc",
>+    ]
>+    deps = [
>+      ":alr_detector",
>+      ":delay_based_bwe",
>+      ":estimators",
>+      ":goog_cc",
>+      ":probe_controller",
>+      "../../../api/transport:network_control",
>+      "../../../api/transport:network_control_test",
>+      "../../../rtc_base:checks",
>+      "../../../rtc_base:rtc_base_approved",
>+      "../../../rtc_base:rtc_base_tests_utils",
>+      "../../../rtc_base/experiments:alr_experiment",
>+      "../../../system_wrappers",
>+      "../../../test:field_trial",
>+      "../../../test:test_support",
>+      "../../pacing",
>+      "../../remote_bitrate_estimator",
>+      "../../rtp_rtcp:rtp_rtcp_format",
>+      "//system_wrappers:field_trial_api",
>+      "//testing/gmock",
>+      "//third_party/abseil-cpp/absl/memory",
>+    ]
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/acknowledged_bitrate_estimator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.cc
>similarity index 72%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/acknowledged_bitrate_estimator.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.cc
>index 1e75ee6e925..0722b2218bd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/acknowledged_bitrate_estimator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.cc
>@@ -8,13 +8,13 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "modules/congestion_controller/acknowledged_bitrate_estimator.h"
>+#include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h"
> 
> #include <utility>
> 
>+#include "absl/memory/memory.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> #include "rtc_base/numerics/safe_conversions.h"
>-#include "rtc_base/ptr_util.h"
> 
> namespace webrtc {
> 
>@@ -25,7 +25,9 @@ bool IsInSendTimeHistory(const PacketFeedback& packet) {
> }  // namespace
> 
> AcknowledgedBitrateEstimator::AcknowledgedBitrateEstimator()
>-    : AcknowledgedBitrateEstimator(rtc::MakeUnique<BitrateEstimator>()) {}
>+    : AcknowledgedBitrateEstimator(absl::make_unique<BitrateEstimator>()) {}
>+
>+AcknowledgedBitrateEstimator::~AcknowledgedBitrateEstimator() {}
> 
> AcknowledgedBitrateEstimator::AcknowledgedBitrateEstimator(
>     std::unique_ptr<BitrateEstimator> bitrate_estimator)
>@@ -45,8 +47,12 @@ void AcknowledgedBitrateEstimator::IncomingPacketFeedbackVector(
>   }
> }
> 
>-rtc::Optional<uint32_t> AcknowledgedBitrateEstimator::bitrate_bps() const {
>-  return bitrate_estimator_->bitrate_bps();
>+absl::optional<uint32_t> AcknowledgedBitrateEstimator::bitrate_bps() const {
>+  auto estimated_bitrate = bitrate_estimator_->bitrate_bps();
>+  return estimated_bitrate
>+             ? *estimated_bitrate +
>+                   allocated_bitrate_without_feedback_bps_.value_or(0)
>+             : estimated_bitrate;
> }
> 
> void AcknowledgedBitrateEstimator::SetAlrEndedTimeMs(
>@@ -54,6 +60,11 @@ void AcknowledgedBitrateEstimator::SetAlrEndedTimeMs(
>   alr_ended_time_ms_.emplace(alr_ended_time_ms);
> }
> 
>+void AcknowledgedBitrateEstimator::SetAllocatedBitrateWithoutFeedback(
>+    uint32_t bitrate_bps) {
>+  allocated_bitrate_without_feedback_bps_.emplace(bitrate_bps);
>+}
>+
> void AcknowledgedBitrateEstimator::MaybeExpectFastRateChange(
>     int64_t packet_send_time_ms) {
>   if (alr_ended_time_ms_ && packet_send_time_ms > *alr_ended_time_ms_) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/acknowledged_bitrate_estimator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h
>similarity index 62%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/acknowledged_bitrate_estimator.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h
>index 72c1023044c..ccb9718b250 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/acknowledged_bitrate_estimator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h
>@@ -8,14 +8,14 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#ifndef MODULES_CONGESTION_CONTROLLER_ACKNOWLEDGED_BITRATE_ESTIMATOR_H_
>-#define MODULES_CONGESTION_CONTROLLER_ACKNOWLEDGED_BITRATE_ESTIMATOR_H_
>+#ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_ACKNOWLEDGED_BITRATE_ESTIMATOR_H_
>+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_ACKNOWLEDGED_BITRATE_ESTIMATOR_H_
> 
> #include <memory>
> #include <vector>
> 
>-#include "api/optional.h"
>-#include "modules/congestion_controller/bitrate_estimator.h"
>+#include "absl/types/optional.h"
>+#include "modules/congestion_controller/goog_cc/bitrate_estimator.h"
> 
> namespace webrtc {
> 
>@@ -27,18 +27,21 @@ class AcknowledgedBitrateEstimator {
>       std::unique_ptr<BitrateEstimator> bitrate_estimator);
> 
>   AcknowledgedBitrateEstimator();
>+  ~AcknowledgedBitrateEstimator();
> 
>   void IncomingPacketFeedbackVector(
>       const std::vector<PacketFeedback>& packet_feedback_vector);
>-  rtc::Optional<uint32_t> bitrate_bps() const;
>+  absl::optional<uint32_t> bitrate_bps() const;
>   void SetAlrEndedTimeMs(int64_t alr_ended_time_ms);
>+  void SetAllocatedBitrateWithoutFeedback(uint32_t bitrate_bps);
> 
>  private:
>   void MaybeExpectFastRateChange(int64_t packet_arrival_time_ms);
>-  rtc::Optional<int64_t> alr_ended_time_ms_;
>+  absl::optional<int64_t> alr_ended_time_ms_;
>   std::unique_ptr<BitrateEstimator> bitrate_estimator_;
>+  absl::optional<uint32_t> allocated_bitrate_without_feedback_bps_;
> };
> 
> }  // namespace webrtc
> 
>-#endif  // MODULES_CONGESTION_CONTROLLER_ACKNOWLEDGED_BITRATE_ESTIMATOR_H_
>+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_ACKNOWLEDGED_BITRATE_ESTIMATOR_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/acknowledged_bitrate_estimator_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_unittest.cc
>similarity index 92%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/acknowledged_bitrate_estimator_unittest.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_unittest.cc
>index d7bf7367816..4991ee69960 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/acknowledged_bitrate_estimator_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator_unittest.cc
>@@ -8,13 +8,13 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "modules/congestion_controller/acknowledged_bitrate_estimator.h"
>+#include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h"
> 
> #include <utility>
> 
>+#include "absl/memory/memory.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> #include "rtc_base/fakeclock.h"
>-#include "rtc_base/ptr_util.h"
> #include "test/gmock.h"
> #include "test/gtest.h"
> 
>@@ -35,7 +35,7 @@ constexpr size_t kPayloadSize = 10;
> class MockBitrateEstimator : public BitrateEstimator {
>  public:
>   MOCK_METHOD2(Update, void(int64_t now_ms, int bytes));
>-  MOCK_CONST_METHOD0(bitrate_bps, rtc::Optional<uint32_t>());
>+  MOCK_CONST_METHOD0(bitrate_bps, absl::optional<uint32_t>());
>   MOCK_METHOD0(ExpectFastRateChange, void());
> };
> 
>@@ -46,10 +46,10 @@ struct AcknowledgedBitrateEstimatorTestStates {
> 
> AcknowledgedBitrateEstimatorTestStates CreateTestStates() {
>   AcknowledgedBitrateEstimatorTestStates states;
>-  auto mock_bitrate_estimator = rtc::MakeUnique<MockBitrateEstimator>();
>+  auto mock_bitrate_estimator = absl::make_unique<MockBitrateEstimator>();
>   states.mock_bitrate_estimator = mock_bitrate_estimator.get();
>   states.acknowledged_bitrate_estimator =
>-      rtc::MakeUnique<AcknowledgedBitrateEstimator>(
>+      absl::make_unique<AcknowledgedBitrateEstimator>(
>           std::move(mock_bitrate_estimator));
>   return states;
> }
>@@ -124,7 +124,7 @@ TEST(TestAcknowledgedBitrateEstimator, ExpectFastRateChangeWhenLeftAlr) {
> 
> TEST(TestAcknowledgedBitrateEstimator, ReturnBitrate) {
>   auto states = CreateTestStates();
>-  rtc::Optional<uint32_t> return_value(42);
>+  absl::optional<uint32_t> return_value(42);
>   EXPECT_CALL(*states.mock_bitrate_estimator, bitrate_bps())
>       .Times(1)
>       .WillOnce(Return(return_value));
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/alr_detector.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/alr_detector.cc
>similarity index 79%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/alr_detector.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/alr_detector.cc
>index 69a0c2cb1cf..ed8f3a66f41 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/alr_detector.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/alr_detector.cc
>@@ -8,18 +8,20 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "modules/pacing/alr_detector.h"
>+#include "modules/congestion_controller/goog_cc/alr_detector.h"
> 
> #include <algorithm>
>+#include <cstdio>
> #include <string>
> 
>+#include "absl/memory/memory.h"
> #include "logging/rtc_event_log/events/rtc_event_alr_state.h"
> #include "logging/rtc_event_log/rtc_event_log.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/experiments/alr_experiment.h"
> #include "rtc_base/format_macros.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/ptr_util.h"
>+#include "rtc_base/numerics/safe_conversions.h"
> #include "rtc_base/timeutils.h"
> #include "system_wrappers/include/field_trial.h"
> 
>@@ -33,7 +35,7 @@ AlrDetector::AlrDetector(RtcEventLog* event_log)
>       alr_budget_(0, true),
>       event_log_(event_log) {
>   RTC_CHECK(AlrExperimentSettings::MaxOneFieldTrialEnabled());
>-  rtc::Optional<AlrExperimentSettings> experiment_settings =
>+  absl::optional<AlrExperimentSettings> experiment_settings =
>       AlrExperimentSettings::CreateFromFieldTrial(
>           AlrExperimentSettings::kScreenshareProbingBweExperimentName);
>   if (!experiment_settings) {
>@@ -51,7 +53,16 @@ AlrDetector::AlrDetector(RtcEventLog* event_log)
> 
> AlrDetector::~AlrDetector() {}
> 
>-void AlrDetector::OnBytesSent(size_t bytes_sent, int64_t delta_time_ms) {
>+void AlrDetector::OnBytesSent(size_t bytes_sent, int64_t send_time_ms) {
>+  if (!last_send_time_ms_.has_value()) {
>+    last_send_time_ms_ = send_time_ms;
>+    // Since the duration for sending the bytes is unknwon, return without
>+    // updating alr state.
>+    return;
>+  }
>+  int64_t delta_time_ms = send_time_ms - *last_send_time_ms_;
>+  last_send_time_ms_ = send_time_ms;
>+
>   alr_budget_.UseBudget(bytes_sent);
>   alr_budget_.IncreaseBudget(delta_time_ms);
>   bool state_changed = false;
>@@ -67,7 +78,7 @@ void AlrDetector::OnBytesSent(size_t bytes_sent, int64_t delta_time_ms) {
>   }
>   if (event_log_ && state_changed) {
>     event_log_->Log(
>-        rtc::MakeUnique<RtcEventAlrState>(alr_started_time_ms_.has_value()));
>+        absl::make_unique<RtcEventAlrState>(alr_started_time_ms_.has_value()));
>   }
> }
> 
>@@ -78,7 +89,7 @@ void AlrDetector::SetEstimatedBitrate(int bitrate_bps) {
>   alr_budget_.set_target_rate_kbps(rtc::dchecked_cast<int>(target_rate_kbps));
> }
> 
>-rtc::Optional<int64_t> AlrDetector::GetApplicationLimitedRegionStartTime()
>+absl::optional<int64_t> AlrDetector::GetApplicationLimitedRegionStartTime()
>     const {
>   return alr_started_time_ms_;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/alr_detector.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/alr_detector.h
>similarity index 82%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/alr_detector.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/alr_detector.h
>index f73bc17b3db..8d1b00699c7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/alr_detector.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/alr_detector.h
>@@ -8,15 +8,13 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#ifndef MODULES_PACING_ALR_DETECTOR_H_
>-#define MODULES_PACING_ALR_DETECTOR_H_
>+#ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_ALR_DETECTOR_H_
>+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_ALR_DETECTOR_H_
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/pacing/interval_budget.h"
>-#include "modules/pacing/paced_sender.h"
> #include "rtc_base/rate_statistics.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -35,14 +33,14 @@ class AlrDetector {
>   explicit AlrDetector(RtcEventLog* event_log);
>   ~AlrDetector();
> 
>-  void OnBytesSent(size_t bytes_sent, int64_t delta_time_ms);
>+  void OnBytesSent(size_t bytes_sent, int64_t send_time_ms);
> 
>   // Set current estimated bandwidth.
>   void SetEstimatedBitrate(int bitrate_bps);
> 
>   // Returns time in milliseconds when the current application-limited region
>   // started or empty result if the sender is currently not application-limited.
>-  rtc::Optional<int64_t> GetApplicationLimitedRegionStartTime() const;
>+  absl::optional<int64_t> GetApplicationLimitedRegionStartTime() const;
> 
>   // Sent traffic percentage as a function of network capacity used to determine
>   // application-limited region. ALR region start when bandwidth usage drops
>@@ -61,12 +59,13 @@ class AlrDetector {
>   int alr_start_budget_level_percent_;
>   int alr_stop_budget_level_percent_;
> 
>+  absl::optional<int64_t> last_send_time_ms_;
>+
>   IntervalBudget alr_budget_;
>-  rtc::Optional<int64_t> alr_started_time_ms_;
>+  absl::optional<int64_t> alr_started_time_ms_;
> 
>   RtcEventLog* event_log_;
> };
>-
> }  // namespace webrtc
> 
>-#endif  // MODULES_PACING_ALR_DETECTOR_H_
>+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_ALR_DETECTOR_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/alr_detector_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/alr_detector_unittest.cc
>similarity index 81%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/alr_detector_unittest.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/alr_detector_unittest.cc
>index 356e14e4699..3a3d002c23b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/alr_detector_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/alr_detector_unittest.cc
>@@ -8,7 +8,7 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "modules/pacing/alr_detector.h"
>+#include "modules/congestion_controller/goog_cc/alr_detector.h"
> 
> #include "rtc_base/experiments/alr_experiment.h"
> #include "test/field_trial.h"
>@@ -21,12 +21,12 @@ constexpr int kEstimatedBitrateBps = 300000;
> }  // namespace
> 
> namespace webrtc {
>-
> namespace {
> class SimulateOutgoingTrafficIn {
>  public:
>-  explicit SimulateOutgoingTrafficIn(AlrDetector* alr_detector)
>-      : alr_detector_(alr_detector) {
>+  explicit SimulateOutgoingTrafficIn(AlrDetector* alr_detector,
>+                                     int64_t* timestamp_ms)
>+      : alr_detector_(alr_detector), timestamp_ms_(timestamp_ms) {
>     RTC_CHECK(alr_detector_);
>   }
> 
>@@ -48,20 +48,23 @@ class SimulateOutgoingTrafficIn {
>       return;
>     const int kTimeStepMs = 10;
>     for (int t = 0; t < *interval_ms_; t += kTimeStepMs) {
>+      *timestamp_ms_ += kTimeStepMs;
>       alr_detector_->OnBytesSent(kEstimatedBitrateBps * *usage_percentage_ *
>                                      kTimeStepMs / (8 * 100 * 1000),
>-                                 kTimeStepMs);
>+                                 *timestamp_ms_);
>     }
>     int remainder_ms = *interval_ms_ % kTimeStepMs;
>     if (remainder_ms > 0) {
>+      *timestamp_ms_ += kTimeStepMs;
>       alr_detector_->OnBytesSent(kEstimatedBitrateBps * *usage_percentage_ *
>                                      remainder_ms / (8 * 100 * 1000),
>-                                 kTimeStepMs);
>+                                 *timestamp_ms_);
>     }
>   }
>   AlrDetector* const alr_detector_;
>-  rtc::Optional<int> interval_ms_;
>-  rtc::Optional<int> usage_percentage_;
>+  int64_t* timestamp_ms_;
>+  absl::optional<int> interval_ms_;
>+  absl::optional<int> usage_percentage_;
> };
> }  // namespace
> 
>@@ -73,6 +76,7 @@ class AlrDetectorTest : public testing::Test {
> 
>  protected:
>   AlrDetector alr_detector_;
>+  int64_t timestamp_ms_ = 1000;
> };
> 
> TEST_F(AlrDetectorTest, AlrDetection) {
>@@ -80,19 +84,19 @@ TEST_F(AlrDetectorTest, AlrDetection) {
>   EXPECT_FALSE(alr_detector_.GetApplicationLimitedRegionStartTime());
> 
>   // Stay in non-ALR state when usage is close to 100%.
>-  SimulateOutgoingTrafficIn(&alr_detector_)
>+  SimulateOutgoingTrafficIn(&alr_detector_, &timestamp_ms_)
>       .ForTimeMs(1000)
>       .AtPercentOfEstimatedBitrate(90);
>   EXPECT_FALSE(alr_detector_.GetApplicationLimitedRegionStartTime());
> 
>   // Verify that we ALR starts when bitrate drops below 20%.
>-  SimulateOutgoingTrafficIn(&alr_detector_)
>+  SimulateOutgoingTrafficIn(&alr_detector_, &timestamp_ms_)
>       .ForTimeMs(1500)
>       .AtPercentOfEstimatedBitrate(20);
>   EXPECT_TRUE(alr_detector_.GetApplicationLimitedRegionStartTime());
> 
>   // Verify that ALR ends when usage is above 65%.
>-  SimulateOutgoingTrafficIn(&alr_detector_)
>+  SimulateOutgoingTrafficIn(&alr_detector_, &timestamp_ms_)
>       .ForTimeMs(4000)
>       .AtPercentOfEstimatedBitrate(100);
>   EXPECT_FALSE(alr_detector_.GetApplicationLimitedRegionStartTime());
>@@ -103,19 +107,19 @@ TEST_F(AlrDetectorTest, ShortSpike) {
>   EXPECT_FALSE(alr_detector_.GetApplicationLimitedRegionStartTime());
> 
>   // Verify that we ALR starts when bitrate drops below 20%.
>-  SimulateOutgoingTrafficIn(&alr_detector_)
>+  SimulateOutgoingTrafficIn(&alr_detector_, &timestamp_ms_)
>       .ForTimeMs(1000)
>       .AtPercentOfEstimatedBitrate(20);
>   EXPECT_TRUE(alr_detector_.GetApplicationLimitedRegionStartTime());
> 
>   // Verify that we stay in ALR region even after a short bitrate spike.
>-  SimulateOutgoingTrafficIn(&alr_detector_)
>+  SimulateOutgoingTrafficIn(&alr_detector_, &timestamp_ms_)
>       .ForTimeMs(100)
>       .AtPercentOfEstimatedBitrate(150);
>   EXPECT_TRUE(alr_detector_.GetApplicationLimitedRegionStartTime());
> 
>   // ALR ends when usage is above 65%.
>-  SimulateOutgoingTrafficIn(&alr_detector_)
>+  SimulateOutgoingTrafficIn(&alr_detector_, &timestamp_ms_)
>       .ForTimeMs(3000)
>       .AtPercentOfEstimatedBitrate(100);
>   EXPECT_FALSE(alr_detector_.GetApplicationLimitedRegionStartTime());
>@@ -126,7 +130,7 @@ TEST_F(AlrDetectorTest, BandwidthEstimateChanges) {
>   EXPECT_FALSE(alr_detector_.GetApplicationLimitedRegionStartTime());
> 
>   // ALR starts when bitrate drops below 20%.
>-  SimulateOutgoingTrafficIn(&alr_detector_)
>+  SimulateOutgoingTrafficIn(&alr_detector_, &timestamp_ms_)
>       .ForTimeMs(1000)
>       .AtPercentOfEstimatedBitrate(20);
>   EXPECT_TRUE(alr_detector_.GetApplicationLimitedRegionStartTime());
>@@ -137,7 +141,7 @@ TEST_F(AlrDetectorTest, BandwidthEstimateChanges) {
>   // to the BWE drop by initiating a new probe.
>   alr_detector_.SetEstimatedBitrate(kEstimatedBitrateBps / 5);
>   EXPECT_TRUE(alr_detector_.GetApplicationLimitedRegionStartTime());
>-  SimulateOutgoingTrafficIn(&alr_detector_)
>+  SimulateOutgoingTrafficIn(&alr_detector_, &timestamp_ms_)
>       .ForTimeMs(1000)
>       .AtPercentOfEstimatedBitrate(50);
>   EXPECT_FALSE(alr_detector_.GetApplicationLimitedRegionStartTime());
>@@ -146,7 +150,7 @@ TEST_F(AlrDetectorTest, BandwidthEstimateChanges) {
> TEST_F(AlrDetectorTest, ParseControlFieldTrial) {
>   webrtc::test::ScopedFieldTrials field_trial(
>       "WebRTC-ProbingScreenshareBwe/Control/");
>-  rtc::Optional<AlrExperimentSettings> parsed_params =
>+  absl::optional<AlrExperimentSettings> parsed_params =
>       AlrExperimentSettings::CreateFromFieldTrial(
>           "WebRTC-ProbingScreenshareBwe");
>   EXPECT_FALSE(static_cast<bool>(parsed_params));
>@@ -155,7 +159,7 @@ TEST_F(AlrDetectorTest, ParseControlFieldTrial) {
> TEST_F(AlrDetectorTest, ParseActiveFieldTrial) {
>   webrtc::test::ScopedFieldTrials field_trial(
>       "WebRTC-ProbingScreenshareBwe/1.1,2875,85,20,-20,1/");
>-  rtc::Optional<AlrExperimentSettings> parsed_params =
>+  absl::optional<AlrExperimentSettings> parsed_params =
>       AlrExperimentSettings::CreateFromFieldTrial(
>           "WebRTC-ProbingScreenshareBwe");
>   ASSERT_TRUE(static_cast<bool>(parsed_params));
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bitrate_estimator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/bitrate_estimator.cc
>similarity index 65%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bitrate_estimator.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/bitrate_estimator.cc
>index 76c132bbe9c..c776827802c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bitrate_estimator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/bitrate_estimator.cc
>@@ -8,26 +8,60 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "modules/congestion_controller/bitrate_estimator.h"
>+#include "modules/congestion_controller/goog_cc/bitrate_estimator.h"
> 
> #include <cmath>
>+#include <string>
> 
> #include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>+#include "rtc_base/logging.h"
>+#include "system_wrappers/include/field_trial.h"
> 
> namespace webrtc {
> 
> namespace {
> constexpr int kInitialRateWindowMs = 500;
> constexpr int kRateWindowMs = 150;
>+
>+const char kBweInitialThroughputWindowExperiment[] =
>+    "WebRTC-BweInitialThroughputWindowExperiment";
>+
>+int ReadInitialThroughputWindowSizeMs() {
>+  std::string experiment_string =
>+      webrtc::field_trial::FindFullName(kBweInitialThroughputWindowExperiment);
>+  int initial_window_ms = kInitialRateWindowMs;
>+  int parsed_values =
>+      sscanf(experiment_string.c_str(), "Enabled-%d", &initial_window_ms);
>+  if (parsed_values != 1) {
>+    RTC_LOG(LS_ERROR) << "Incorrectly formatted field trial string for "
>+                      << kBweInitialThroughputWindowExperiment;
>+    return kInitialRateWindowMs;
>+  }
>+  if (kRateWindowMs <= initial_window_ms && initial_window_ms <= 1000) {
>+    RTC_LOG(LS_INFO) << kBweInitialThroughputWindowExperiment
>+                     << " enabled with " << initial_window_ms << " ms window.";
>+    return initial_window_ms;
>+  }
>+  RTC_LOG(LS_ERROR)
>+      << "Initial window for throughput estimation must be between "
>+      << kRateWindowMs << " and 1000 ms.";
>+  return kInitialRateWindowMs;
>+}
>+
> }  // namespace
> 
> BitrateEstimator::BitrateEstimator()
>     : sum_(0),
>-      current_win_ms_(0),
>+      initial_window_ms_(kInitialRateWindowMs),
>+      current_window_ms_(0),
>       prev_time_ms_(-1),
>       bitrate_estimate_(-1.0f),
>-      bitrate_estimate_var_(50.0f) {}
>+      bitrate_estimate_var_(50.0f) {
>+  if (field_trial::IsEnabled(kBweInitialThroughputWindowExperiment)) {
>+    initial_window_ms_ = ReadInitialThroughputWindowSizeMs();
>+  }
>+}
> 
> BitrateEstimator::~BitrateEstimator() = default;
> 
>@@ -36,7 +70,7 @@ void BitrateEstimator::Update(int64_t now_ms, int bytes) {
>   // We use a larger window at the beginning to get a more stable sample that
>   // we can use to initialize the estimate.
>   if (bitrate_estimate_ < 0.f)
>-    rate_window_ms = kInitialRateWindowMs;
>+    rate_window_ms = initial_window_ms_;
>   float bitrate_sample = UpdateWindow(now_ms, bytes, rate_window_ms);
>   if (bitrate_sample < 0.0f)
>     return;
>@@ -71,30 +105,30 @@ float BitrateEstimator::UpdateWindow(int64_t now_ms,
>   if (now_ms < prev_time_ms_) {
>     prev_time_ms_ = -1;
>     sum_ = 0;
>-    current_win_ms_ = 0;
>+    current_window_ms_ = 0;
>   }
>   if (prev_time_ms_ >= 0) {
>-    current_win_ms_ += now_ms - prev_time_ms_;
>+    current_window_ms_ += now_ms - prev_time_ms_;
>     // Reset if nothing has been received for more than a full window.
>     if (now_ms - prev_time_ms_ > rate_window_ms) {
>       sum_ = 0;
>-      current_win_ms_ %= rate_window_ms;
>+      current_window_ms_ %= rate_window_ms;
>     }
>   }
>   prev_time_ms_ = now_ms;
>   float bitrate_sample = -1.0f;
>-  if (current_win_ms_ >= rate_window_ms) {
>+  if (current_window_ms_ >= rate_window_ms) {
>     bitrate_sample = 8.0f * sum_ / static_cast<float>(rate_window_ms);
>-    current_win_ms_ -= rate_window_ms;
>+    current_window_ms_ -= rate_window_ms;
>     sum_ = 0;
>   }
>   sum_ += bytes;
>   return bitrate_sample;
> }
> 
>-rtc::Optional<uint32_t> BitrateEstimator::bitrate_bps() const {
>+absl::optional<uint32_t> BitrateEstimator::bitrate_bps() const {
>   if (bitrate_estimate_ < 0.f)
>-    return rtc::nullopt;
>+    return absl::nullopt;
>   return bitrate_estimate_ * 1000;
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bitrate_estimator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/bitrate_estimator.h
>similarity index 76%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bitrate_estimator.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/bitrate_estimator.h
>index 2fa1515d19f..610fa214e7a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/bitrate_estimator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/bitrate_estimator.h
>@@ -8,12 +8,12 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#ifndef MODULES_CONGESTION_CONTROLLER_BITRATE_ESTIMATOR_H_
>-#define MODULES_CONGESTION_CONTROLLER_BITRATE_ESTIMATOR_H_
>+#ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_BITRATE_ESTIMATOR_H_
>+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_BITRATE_ESTIMATOR_H_
> 
> #include <vector>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> 
> namespace webrtc {
> 
>@@ -28,14 +28,15 @@ class BitrateEstimator {
>   virtual ~BitrateEstimator();
>   virtual void Update(int64_t now_ms, int bytes);
> 
>-  virtual rtc::Optional<uint32_t> bitrate_bps() const;
>+  virtual absl::optional<uint32_t> bitrate_bps() const;
> 
>   virtual void ExpectFastRateChange();
> 
>  private:
>   float UpdateWindow(int64_t now_ms, int bytes, int rate_window_ms);
>   int sum_;
>-  int64_t current_win_ms_;
>+  int64_t initial_window_ms_;
>+  int64_t current_window_ms_;
>   int64_t prev_time_ms_;
>   float bitrate_estimate_;
>   float bitrate_estimate_var_;
>@@ -43,4 +44,4 @@ class BitrateEstimator {
> 
> }  // namespace webrtc
> 
>-#endif  // MODULES_CONGESTION_CONTROLLER_BITRATE_ESTIMATOR_H_
>+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_BITRATE_ESTIMATOR_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/delay_based_bwe.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe.cc
>similarity index 82%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/delay_based_bwe.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe.cc
>index 2092212f8bc..6d50a87276e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/delay_based_bwe.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe.cc
>@@ -8,25 +8,26 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "modules/congestion_controller/delay_based_bwe.h"
>+#include "modules/congestion_controller/goog_cc/delay_based_bwe.h"
> 
> #include <algorithm>
> #include <cmath>
>+#include <cstdio>
> #include <string>
> 
>+#include "absl/memory/memory.h"
> #include "logging/rtc_event_log/events/rtc_event_bwe_update_delay_based.h"
> #include "logging/rtc_event_log/rtc_event_log.h"
>+#include "modules/congestion_controller/goog_cc/trendline_estimator.h"
> #include "modules/pacing/paced_sender.h"
> #include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
> #include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/thread_annotations.h"
> #include "system_wrappers/include/field_trial.h"
> #include "system_wrappers/include/metrics.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace {
> constexpr int kTimestampGroupLengthMs = 5;
>@@ -83,12 +84,10 @@ DelayBasedBwe::Result::Result(bool probe, uint32_t target_bitrate_bps)
> 
> DelayBasedBwe::Result::~Result() {}
> 
>-DelayBasedBwe::DelayBasedBwe(RtcEventLog* event_log, const Clock* clock)
>+DelayBasedBwe::DelayBasedBwe(RtcEventLog* event_log)
>     : event_log_(event_log),
>-      clock_(clock),
>       inter_arrival_(),
>-      trendline_estimator_(),
>-      detector_(),
>+      delay_detector_(),
>       last_seen_packet_ms_(-1),
>       uma_recorded_(false),
>       probe_bitrate_estimator_(event_log),
>@@ -104,13 +103,17 @@ DelayBasedBwe::DelayBasedBwe(RtcEventLog* event_log, const Clock* clock)
>   RTC_LOG(LS_INFO)
>       << "Using Trendline filter for delay change estimation with window size "
>       << trendline_window_size_;
>+  delay_detector_.reset(new TrendlineEstimator(trendline_window_size_,
>+                                               trendline_smoothing_coeff_,
>+                                               trendline_threshold_gain_));
> }
> 
> DelayBasedBwe::~DelayBasedBwe() {}
> 
> DelayBasedBwe::Result DelayBasedBwe::IncomingPacketFeedbackVector(
>     const std::vector<PacketFeedback>& packet_feedback_vector,
>-    rtc::Optional<uint32_t> acked_bitrate_bps) {
>+    absl::optional<uint32_t> acked_bitrate_bps,
>+    int64_t at_time_ms) {
>   RTC_DCHECK(std::is_sorted(packet_feedback_vector.begin(),
>                             packet_feedback_vector.end(),
>                             PacketFeedbackComparator()));
>@@ -132,17 +135,17 @@ DelayBasedBwe::Result DelayBasedBwe::IncomingPacketFeedbackVector(
>   }
>   bool delayed_feedback = true;
>   bool recovered_from_overuse = false;
>-  BandwidthUsage prev_detector_state = detector_.State();
>+  BandwidthUsage prev_detector_state = delay_detector_->State();
>   for (const auto& packet_feedback : packet_feedback_vector) {
>     if (packet_feedback.send_time_ms < 0)
>       continue;
>     delayed_feedback = false;
>-    IncomingPacketFeedback(packet_feedback);
>+    IncomingPacketFeedback(packet_feedback, at_time_ms);
>     if (prev_detector_state == BandwidthUsage::kBwUnderusing &&
>-        detector_.State() == BandwidthUsage::kBwNormal) {
>+        delay_detector_->State() == BandwidthUsage::kBwNormal) {
>       recovered_from_overuse = true;
>     }
>-    prev_detector_state = detector_.State();
>+    prev_detector_state = delay_detector_->State();
>   }
> 
>   if (delayed_feedback) {
>@@ -153,7 +156,8 @@ DelayBasedBwe::Result DelayBasedBwe::IncomingPacketFeedbackVector(
>     }
>   } else {
>     consecutive_delayed_feedbacks_ = 0;
>-    return MaybeUpdateEstimate(acked_bitrate_bps, recovered_from_overuse);
>+    return MaybeUpdateEstimate(acked_bitrate_bps, recovered_from_overuse,
>+                               at_time_ms);
>   }
>   return Result();
> }
>@@ -176,17 +180,18 @@ DelayBasedBwe::Result DelayBasedBwe::OnLongFeedbackDelay(
> }
> 
> void DelayBasedBwe::IncomingPacketFeedback(
>-    const PacketFeedback& packet_feedback) {
>-  int64_t now_ms = clock_->TimeInMilliseconds();
>+    const PacketFeedback& packet_feedback,
>+    int64_t at_time_ms) {
>+  int64_t now_ms = at_time_ms;
>   // Reset if the stream has timed out.
>   if (last_seen_packet_ms_ == -1 ||
>       now_ms - last_seen_packet_ms_ > kStreamTimeOutMs) {
>     inter_arrival_.reset(
>         new InterArrival((kTimestampGroupLengthMs << kInterArrivalShift) / 1000,
>                          kTimestampToMs, true));
>-    trendline_estimator_.reset(new TrendlineEstimator(
>-        trendline_window_size_, trendline_smoothing_coeff_,
>-        trendline_threshold_gain_));
>+    delay_detector_.reset(new TrendlineEstimator(trendline_window_size_,
>+                                                 trendline_smoothing_coeff_,
>+                                                 trendline_threshold_gain_));
>   }
>   last_seen_packet_ms_ = now_ms;
> 
>@@ -208,11 +213,8 @@ void DelayBasedBwe::IncomingPacketFeedback(
>                                     now_ms, packet_feedback.payload_size,
>                                     &ts_delta, &t_delta, &size_delta)) {
>     double ts_delta_ms = (1000.0 * ts_delta) / (1 << kInterArrivalShift);
>-    trendline_estimator_->Update(t_delta, ts_delta_ms,
>-                                 packet_feedback.arrival_time_ms);
>-    detector_.Detect(trendline_estimator_->trendline_slope(), ts_delta_ms,
>-                     trendline_estimator_->num_of_deltas(),
>-                     packet_feedback.arrival_time_ms);
>+    delay_detector_->Update(t_delta, ts_delta_ms,
>+                            packet_feedback.arrival_time_ms);
>   }
>   if (packet_feedback.pacing_info.probe_cluster_id !=
>       PacedPacketInfo::kNotAProbe) {
>@@ -221,25 +223,24 @@ void DelayBasedBwe::IncomingPacketFeedback(
> }
> 
> DelayBasedBwe::Result DelayBasedBwe::MaybeUpdateEstimate(
>-    rtc::Optional<uint32_t> acked_bitrate_bps,
>-    bool recovered_from_overuse) {
>+    absl::optional<uint32_t> acked_bitrate_bps,
>+    bool recovered_from_overuse,
>+    int64_t at_time_ms) {
>   Result result;
>-  int64_t now_ms = clock_->TimeInMilliseconds();
>+  int64_t now_ms = at_time_ms;
> 
>-  rtc::Optional<int> probe_bitrate_bps =
>+  absl::optional<int> probe_bitrate_bps =
>       probe_bitrate_estimator_.FetchAndResetLastEstimatedBitrateBps();
>   // Currently overusing the bandwidth.
>-  if (detector_.State() == BandwidthUsage::kBwOverusing) {
>+  if (delay_detector_->State() == BandwidthUsage::kBwOverusing) {
>     if (acked_bitrate_bps &&
>         rate_control_.TimeToReduceFurther(now_ms, *acked_bitrate_bps)) {
>       result.updated =
>           UpdateEstimate(now_ms, acked_bitrate_bps, &result.target_bitrate_bps);
>     } else if (!acked_bitrate_bps && rate_control_.ValidEstimate() &&
>-               rate_control_.TimeToReduceFurther(
>-                   now_ms, rate_control_.LatestEstimate() / 2 - 1)) {
>-      // Overusing before we have a measured acknowledged bitrate. We check
>-      // TimeToReduceFurther (with a fake acknowledged bitrate) to avoid
>-      // reducing too often.
>+               rate_control_.InitialTimeToReduceFurther(now_ms)) {
>+      // Overusing before we have a measured acknowledged bitrate. Reduce send
>+      // rate by 50% every 200 ms.
>       // TODO(tschumim): Improve this and/or the acknowledged bitrate estimator
>       // so that we (almost) always have a bitrate estimate.
>       rate_control_.SetEstimate(rate_control_.LatestEstimate() / 2, now_ms);
>@@ -259,35 +260,34 @@ DelayBasedBwe::Result DelayBasedBwe::MaybeUpdateEstimate(
>       result.recovered_from_overuse = recovered_from_overuse;
>     }
>   }
>+  BandwidthUsage detector_state = delay_detector_->State();
>   if ((result.updated && prev_bitrate_ != result.target_bitrate_bps) ||
>-      detector_.State() != prev_state_) {
>+      detector_state != prev_state_) {
>     uint32_t bitrate_bps =
>         result.updated ? result.target_bitrate_bps : prev_bitrate_;
> 
>     BWE_TEST_LOGGING_PLOT(1, "target_bitrate_bps", now_ms, bitrate_bps);
> 
>     if (event_log_) {
>-      event_log_->Log(rtc::MakeUnique<RtcEventBweUpdateDelayBased>(
>-          bitrate_bps, detector_.State()));
>+      event_log_->Log(absl::make_unique<RtcEventBweUpdateDelayBased>(
>+          bitrate_bps, detector_state));
>     }
> 
>     prev_bitrate_ = bitrate_bps;
>-    prev_state_ = detector_.State();
>+    prev_state_ = detector_state;
>   }
>   return result;
> }
> 
> bool DelayBasedBwe::UpdateEstimate(int64_t now_ms,
>-                                   rtc::Optional<uint32_t> acked_bitrate_bps,
>+                                   absl::optional<uint32_t> acked_bitrate_bps,
>                                    uint32_t* target_bitrate_bps) {
>-  // TODO(terelius): RateControlInput::noise_var is deprecated and will be
>-  // removed. In the meantime, we set it to zero.
>-  const RateControlInput input(detector_.State(), acked_bitrate_bps, 0);
>+  const RateControlInput input(delay_detector_->State(), acked_bitrate_bps);
>   *target_bitrate_bps = rate_control_.Update(&input, now_ms);
>   return rate_control_.ValidEstimate();
> }
> 
>-void DelayBasedBwe::OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) {
>+void DelayBasedBwe::OnRttUpdate(int64_t avg_rtt_ms) {
>   rate_control_.SetRtt(avg_rtt_ms);
> }
> 
>@@ -308,7 +308,7 @@ bool DelayBasedBwe::LatestEstimate(std::vector<uint32_t>* ssrcs,
> }
> 
> void DelayBasedBwe::SetStartBitrate(int start_bitrate_bps) {
>-  RTC_LOG(LS_WARNING) << "BWE Setting start bitrate to: " << start_bitrate_bps;
>+  RTC_LOG(LS_INFO) << "BWE Setting start bitrate to: " << start_bitrate_bps;
>   rate_control_.SetStartBitrate(start_bitrate_bps);
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/delay_based_bwe.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe.h
>similarity index 70%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/delay_based_bwe.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe.h
>index 79af11bbf09..626356e7186 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/delay_based_bwe.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe.h
>@@ -8,27 +8,23 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#ifndef MODULES_CONGESTION_CONTROLLER_DELAY_BASED_BWE_H_
>-#define MODULES_CONGESTION_CONTROLLER_DELAY_BASED_BWE_H_
>+#ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_BASED_BWE_H_
>+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_BASED_BWE_H_
> 
> #include <memory>
> #include <utility>
> #include <vector>
> 
>-#include "modules/congestion_controller/median_slope_estimator.h"
>-#include "modules/congestion_controller/probe_bitrate_estimator.h"
>-#include "modules/congestion_controller/trendline_estimator.h"
>+#include "modules/congestion_controller/goog_cc/delay_increase_detector_interface.h"
>+#include "modules/congestion_controller/goog_cc/probe_bitrate_estimator.h"
> #include "modules/remote_bitrate_estimator/aimd_rate_control.h"
> #include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
> #include "modules/remote_bitrate_estimator/inter_arrival.h"
>-#include "modules/remote_bitrate_estimator/overuse_detector.h"
>-#include "modules/remote_bitrate_estimator/overuse_estimator.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/race_checker.h"
> 
> namespace webrtc {
>-
> class RtcEventLog;
> 
> class DelayBasedBwe {
>@@ -45,13 +41,14 @@ class DelayBasedBwe {
>     bool recovered_from_overuse;
>   };
> 
>-  DelayBasedBwe(RtcEventLog* event_log, const Clock* clock);
>+  explicit DelayBasedBwe(RtcEventLog* event_log);
>   virtual ~DelayBasedBwe();
> 
>   Result IncomingPacketFeedbackVector(
>       const std::vector<PacketFeedback>& packet_feedback_vector,
>-      rtc::Optional<uint32_t> acked_bitrate_bps);
>-  void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms);
>+      absl::optional<uint32_t> acked_bitrate_bps,
>+      int64_t at_time_ms);
>+  void OnRttUpdate(int64_t avg_rtt_ms);
>   bool LatestEstimate(std::vector<uint32_t>* ssrcs,
>                       uint32_t* bitrate_bps) const;
>   void SetStartBitrate(int start_bitrate_bps);
>@@ -59,22 +56,22 @@ class DelayBasedBwe {
>   int64_t GetExpectedBwePeriodMs() const;
> 
>  private:
>-  void IncomingPacketFeedback(const PacketFeedback& packet_feedback);
>+  void IncomingPacketFeedback(const PacketFeedback& packet_feedback,
>+                              int64_t at_time_ms);
>   Result OnLongFeedbackDelay(int64_t arrival_time_ms);
>-  Result MaybeUpdateEstimate(rtc::Optional<uint32_t> acked_bitrate_bps,
>-                             bool request_probe);
>+  Result MaybeUpdateEstimate(absl::optional<uint32_t> acked_bitrate_bps,
>+                             bool request_probe,
>+                             int64_t at_time_ms);
>   // Updates the current remote rate estimate and returns true if a valid
>   // estimate exists.
>   bool UpdateEstimate(int64_t now_ms,
>-                      rtc::Optional<uint32_t> acked_bitrate_bps,
>+                      absl::optional<uint32_t> acked_bitrate_bps,
>                       uint32_t* target_bitrate_bps);
> 
>   rtc::RaceChecker network_race_;
>   RtcEventLog* const event_log_;
>-  const Clock* const clock_;
>   std::unique_ptr<InterArrival> inter_arrival_;
>-  std::unique_ptr<TrendlineEstimator> trendline_estimator_;
>-  OveruseDetector detector_;
>+  std::unique_ptr<DelayIncreaseDetectorInterface> delay_detector_;
>   int64_t last_seen_packet_ms_;
>   bool uma_recorded_;
>   AimdRateControl rate_control_;
>@@ -91,4 +88,4 @@ class DelayBasedBwe {
> 
> }  // namespace webrtc
> 
>-#endif  // MODULES_CONGESTION_CONTROLLER_DELAY_BASED_BWE_H_
>+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_BASED_BWE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/delay_based_bwe_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest.cc
>similarity index 75%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/delay_based_bwe_unittest.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest.cc
>index 250cb036a32..b2add3e6e94 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/delay_based_bwe_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest.cc
>@@ -8,11 +8,12 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "modules/congestion_controller/delay_based_bwe.h"
>-#include "modules/congestion_controller/delay_based_bwe_unittest_helper.h"
>+#include "modules/congestion_controller/goog_cc/delay_based_bwe.h"
>+#include "modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.h"
> #include "modules/pacing/paced_sender.h"
> #include "rtc_base/constructormagic.h"
> #include "system_wrappers/include/clock.h"
>+#include "system_wrappers/include/field_trial.h"
> #include "test/field_trial.h"
> #include "test/gtest.h"
> 
>@@ -24,12 +25,13 @@ constexpr int kNumProbesCluster1 = 8;
> const PacedPacketInfo kPacingInfo0(0, kNumProbesCluster0, 2000);
> const PacedPacketInfo kPacingInfo1(1, kNumProbesCluster1, 4000);
> constexpr float kTargetUtilizationFraction = 0.95f;
>+constexpr int64_t kDummyTimestamp = 1000;
> }  // namespace
> 
> TEST_F(DelayBasedBweTest, NoCrashEmptyFeedback) {
>   std::vector<PacketFeedback> packet_feedback_vector;
>-  bitrate_estimator_->IncomingPacketFeedbackVector(packet_feedback_vector,
>-                                                   rtc::nullopt);
>+  bitrate_estimator_->IncomingPacketFeedbackVector(
>+      packet_feedback_vector, absl::nullopt, kDummyTimestamp);
> }
> 
> TEST_F(DelayBasedBweTest, NoCrashOnlyLostFeedback) {
>@@ -40,8 +42,8 @@ TEST_F(DelayBasedBweTest, NoCrashOnlyLostFeedback) {
>   packet_feedback_vector.push_back(PacketFeedback(PacketFeedback::kNotReceived,
>                                                   PacketFeedback::kNoSendTime,
>                                                   1, 1500, PacedPacketInfo()));
>-  bitrate_estimator_->IncomingPacketFeedbackVector(packet_feedback_vector,
>-                                                   rtc::nullopt);
>+  bitrate_estimator_->IncomingPacketFeedbackVector(
>+      packet_feedback_vector, absl::nullopt, kDummyTimestamp);
> }
> 
> TEST_F(DelayBasedBweTest, ProbeDetection) {
>@@ -233,4 +235,58 @@ TEST_F(DelayBasedBweTest, TestInitialOveruse) {
>   EXPECT_NEAR(bitrate_observer_.latest_bitrate(), kStartBitrate / 2, 15000);
> }
> 
>+class DelayBasedBweTestWithBackoffTimeoutExperiment : public DelayBasedBweTest {
>+ public:
>+  DelayBasedBweTestWithBackoffTimeoutExperiment()
>+      : DelayBasedBweTest("WebRTC-BweInitialBackOffInterval/Enabled-200/") {}
>+};
>+
>+// This test subsumes and improves DelayBasedBweTest.TestInitialOveruse above.
>+TEST_F(DelayBasedBweTestWithBackoffTimeoutExperiment, TestInitialOveruse) {
>+  const uint32_t kStartBitrate = 300e3;
>+  const uint32_t kInitialCapacityBps = 200e3;
>+  const uint32_t kDummySsrc = 0;
>+  // High FPS to ensure that we send a lot of packets in a short time.
>+  const int kFps = 90;
>+
>+  stream_generator_->AddStream(new test::RtpStream(kFps, kStartBitrate));
>+  stream_generator_->set_capacity_bps(kInitialCapacityBps);
>+
>+  // Needed to initialize the AimdRateControl.
>+  bitrate_estimator_->SetStartBitrate(kStartBitrate);
>+
>+  // Produce 30 frames (in 1/3 second) and give them to the estimator.
>+  uint32_t bitrate_bps = kStartBitrate;
>+  bool seen_overuse = false;
>+  for (int frames = 0; frames < 30 && !seen_overuse; ++frames) {
>+    bool overuse = GenerateAndProcessFrame(kDummySsrc, bitrate_bps);
>+    // The purpose of this test is to ensure that we back down even if we don't
>+    // have any acknowledged bitrate estimate yet. Hence, if the test works
>+    // as expected, we should not have a measured bitrate yet.
>+    EXPECT_FALSE(acknowledged_bitrate_estimator_->bitrate_bps().has_value());
>+    if (overuse) {
>+      EXPECT_TRUE(bitrate_observer_.updated());
>+      EXPECT_NEAR(bitrate_observer_.latest_bitrate(), kStartBitrate / 2, 15000);
>+      bitrate_bps = bitrate_observer_.latest_bitrate();
>+      seen_overuse = true;
>+    } else if (bitrate_observer_.updated()) {
>+      bitrate_bps = bitrate_observer_.latest_bitrate();
>+      bitrate_observer_.Reset();
>+    }
>+  }
>+  EXPECT_TRUE(seen_overuse);
>+  // Continue generating an additional 15 frames (equivalent to 167 ms) and
>+  // verify that we don't back down further.
>+  for (int frames = 0; frames < 15 && seen_overuse; ++frames) {
>+    bool overuse = GenerateAndProcessFrame(kDummySsrc, bitrate_bps);
>+    EXPECT_FALSE(overuse);
>+    if (bitrate_observer_.updated()) {
>+      bitrate_bps = bitrate_observer_.latest_bitrate();
>+      EXPECT_GE(bitrate_bps, kStartBitrate / 2 - 15000);
>+      EXPECT_LE(bitrate_bps, kInitialCapacityBps + 15000);
>+      bitrate_observer_.Reset();
>+    }
>+  }
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/delay_based_bwe_unittest_helper.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.cc
>similarity index 95%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/delay_based_bwe_unittest_helper.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.cc
>index b441e998cde..2a00801eb1c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/delay_based_bwe_unittest_helper.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.cc
>@@ -7,18 +7,17 @@
>  *  in the file PATENTS.  All contributing project authors may
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
>-#include "modules/congestion_controller/delay_based_bwe_unittest_helper.h"
>+#include "modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.h"
> 
> #include <algorithm>
> #include <limits>
> #include <utility>
> 
>-#include "modules/congestion_controller/delay_based_bwe.h"
>+#include "absl/memory/memory.h"
>+#include "modules/congestion_controller/goog_cc/delay_based_bwe.h"
> #include "rtc_base/checks.h"
>-#include "rtc_base/ptr_util.h"
> 
> namespace webrtc {
>-
> constexpr size_t kMtu = 1200;
> constexpr uint32_t kAcceptedBitrateErrorBps = 50000;
> 
>@@ -89,6 +88,8 @@ bool RtpStream::Compare(const std::unique_ptr<RtpStream>& lhs,
> StreamGenerator::StreamGenerator(int capacity, int64_t time_now)
>     : capacity_(capacity), prev_arrival_time_us_(time_now) {}
> 
>+StreamGenerator::~StreamGenerator() = default;
>+
> // Add a new stream.
> void StreamGenerator::AddStream(RtpStream* stream) {
>   streams_.push_back(std::unique_ptr<RtpStream>(stream));
>@@ -149,10 +150,23 @@ int64_t StreamGenerator::GenerateFrame(std::vector<PacketFeedback>* packets,
> }  // namespace test
> 
> DelayBasedBweTest::DelayBasedBweTest()
>-    : clock_(100000000),
>+    : field_trial(),
>+      clock_(100000000),
>+      acknowledged_bitrate_estimator_(
>+          absl::make_unique<AcknowledgedBitrateEstimator>()),
>+      bitrate_estimator_(new DelayBasedBwe(nullptr)),
>+      stream_generator_(new test::StreamGenerator(1e6,  // Capacity.
>+                                                  clock_.TimeInMicroseconds())),
>+      arrival_time_offset_ms_(0),
>+      first_update_(true) {}
>+
>+DelayBasedBweTest::DelayBasedBweTest(const std::string& field_trial_string)
>+    : field_trial(
>+          absl::make_unique<test::ScopedFieldTrials>(field_trial_string)),
>+      clock_(100000000),
>       acknowledged_bitrate_estimator_(
>-          rtc::MakeUnique<AcknowledgedBitrateEstimator>()),
>-      bitrate_estimator_(new DelayBasedBwe(nullptr, &clock_)),
>+          absl::make_unique<AcknowledgedBitrateEstimator>()),
>+      bitrate_estimator_(new DelayBasedBwe(nullptr)),
>       stream_generator_(new test::StreamGenerator(1e6,  // Capacity.
>                                                   clock_.TimeInMicroseconds())),
>       arrival_time_offset_ms_(0),
>@@ -187,7 +201,8 @@ void DelayBasedBweTest::IncomingFeedback(int64_t arrival_time_ms,
>   acknowledged_bitrate_estimator_->IncomingPacketFeedbackVector(packets);
>   DelayBasedBwe::Result result =
>       bitrate_estimator_->IncomingPacketFeedbackVector(
>-          packets, acknowledged_bitrate_estimator_->bitrate_bps());
>+          packets, acknowledged_bitrate_estimator_->bitrate_bps(),
>+          clock_.TimeInMilliseconds());
>   const uint32_t kDummySsrc = 0;
>   if (result.updated) {
>     bitrate_observer_.OnReceiveBitrateChanged({kDummySsrc},
>@@ -222,7 +237,8 @@ bool DelayBasedBweTest::GenerateAndProcessFrame(uint32_t ssrc,
>   acknowledged_bitrate_estimator_->IncomingPacketFeedbackVector(packets);
>   DelayBasedBwe::Result result =
>       bitrate_estimator_->IncomingPacketFeedbackVector(
>-          packets, acknowledged_bitrate_estimator_->bitrate_bps());
>+          packets, acknowledged_bitrate_estimator_->bitrate_bps(),
>+          clock_.TimeInMilliseconds());
>   const uint32_t kDummySsrc = 0;
>   if (result.updated) {
>     bitrate_observer_.OnReceiveBitrateChanged({kDummySsrc},
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/delay_based_bwe_unittest_helper.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.h
>similarity index 89%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/delay_based_bwe_unittest_helper.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.h
>index 23969a92ae6..7ba18c36752 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/delay_based_bwe_unittest_helper.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/delay_based_bwe_unittest_helper.h
>@@ -8,20 +8,22 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#ifndef MODULES_CONGESTION_CONTROLLER_DELAY_BASED_BWE_UNITTEST_HELPER_H_
>-#define MODULES_CONGESTION_CONTROLLER_DELAY_BASED_BWE_UNITTEST_HELPER_H_
>+#ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_BASED_BWE_UNITTEST_HELPER_H_
>+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_BASED_BWE_UNITTEST_HELPER_H_
> 
> #include <list>
> #include <map>
> #include <memory>
>+#include <string>
> #include <utility>
> #include <vector>
> 
>-#include "modules/congestion_controller/acknowledged_bitrate_estimator.h"
>-#include "modules/congestion_controller/delay_based_bwe.h"
>+#include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h"
>+#include "modules/congestion_controller/goog_cc/delay_based_bwe.h"
> #include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
> #include "rtc_base/constructormagic.h"
> #include "system_wrappers/include/clock.h"
>+#include "test/field_trial.h"
> #include "test/gtest.h"
> 
> namespace webrtc {
>@@ -30,7 +32,7 @@ namespace test {
> class TestBitrateObserver : public RemoteBitrateObserver {
>  public:
>   TestBitrateObserver() : updated_(false), latest_bitrate_(0) {}
>-  virtual ~TestBitrateObserver() {}
>+  ~TestBitrateObserver() override {}
> 
>   void OnReceiveBitrateChanged(const std::vector<uint32_t>& ssrcs,
>                                uint32_t bitrate) override;
>@@ -80,6 +82,7 @@ class RtpStream {
> class StreamGenerator {
>  public:
>   StreamGenerator(int capacity, int64_t time_now);
>+  ~StreamGenerator();
> 
>   // Add a new stream.
>   void AddStream(RtpStream* stream);
>@@ -114,7 +117,8 @@ class StreamGenerator {
> class DelayBasedBweTest : public ::testing::Test {
>  public:
>   DelayBasedBweTest();
>-  virtual ~DelayBasedBweTest();
>+  explicit DelayBasedBweTest(const std::string& field_trial_string);
>+  ~DelayBasedBweTest() override;
> 
>  protected:
>   void AddDefaultStream();
>@@ -163,6 +167,8 @@ class DelayBasedBweTest : public ::testing::Test {
> 
>   static const uint32_t kDefaultSsrc;
> 
>+  std::unique_ptr<test::ScopedFieldTrials>
>+      field_trial;        // Must be initialized first.
>   SimulatedClock clock_;  // Time at the receiver.
>   test::TestBitrateObserver bitrate_observer_;
>   std::unique_ptr<AcknowledgedBitrateEstimator> acknowledged_bitrate_estimator_;
>@@ -175,4 +181,4 @@ class DelayBasedBweTest : public ::testing::Test {
> };
> }  // namespace webrtc
> 
>-#endif  // MODULES_CONGESTION_CONTROLLER_DELAY_BASED_BWE_UNITTEST_HELPER_H_
>+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_BASED_BWE_UNITTEST_HELPER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/delay_increase_detector_interface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/delay_increase_detector_interface.h
>new file mode 100644
>index 00000000000..e4f899af7c0
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/delay_increase_detector_interface.h
>@@ -0,0 +1,38 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_INCREASE_DETECTOR_INTERFACE_H_
>+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_INCREASE_DETECTOR_INTERFACE_H_
>+
>+#include <stdint.h>
>+
>+#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
>+#include "rtc_base/constructormagic.h"
>+
>+namespace webrtc {
>+
>+class DelayIncreaseDetectorInterface {
>+ public:
>+  DelayIncreaseDetectorInterface() {}
>+  virtual ~DelayIncreaseDetectorInterface() {}
>+
>+  // Update the detector with a new sample. The deltas should represent deltas
>+  // between timestamp groups as defined by the InterArrival class.
>+  virtual void Update(double recv_delta_ms,
>+                      double send_delta_ms,
>+                      int64_t arrival_time_ms) = 0;
>+
>+  virtual BandwidthUsage State() const = 0;
>+
>+  RTC_DISALLOW_COPY_AND_ASSIGN(DelayIncreaseDetectorInterface);
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_DELAY_INCREASE_DETECTOR_INTERFACE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/goog_cc_factory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/goog_cc_factory.cc
>new file mode 100644
>index 00000000000..2e0ccc17ca4
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/goog_cc_factory.cc
>@@ -0,0 +1,43 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/congestion_controller/goog_cc/include/goog_cc_factory.h"
>+
>+#include "absl/memory/memory.h"
>+#include "modules/congestion_controller/goog_cc/goog_cc_network_control.h"
>+namespace webrtc {
>+GoogCcNetworkControllerFactory::GoogCcNetworkControllerFactory(
>+    RtcEventLog* event_log)
>+    : event_log_(event_log) {}
>+
>+std::unique_ptr<NetworkControllerInterface>
>+GoogCcNetworkControllerFactory::Create(NetworkControllerConfig config) {
>+  return absl::make_unique<GoogCcNetworkController>(event_log_, config, false);
>+}
>+
>+TimeDelta GoogCcNetworkControllerFactory::GetProcessInterval() const {
>+  const int64_t kUpdateIntervalMs = 25;
>+  return TimeDelta::ms(kUpdateIntervalMs);
>+}
>+
>+GoogCcFeedbackNetworkControllerFactory::GoogCcFeedbackNetworkControllerFactory(
>+    RtcEventLog* event_log)
>+    : event_log_(event_log) {}
>+
>+std::unique_ptr<NetworkControllerInterface>
>+GoogCcFeedbackNetworkControllerFactory::Create(NetworkControllerConfig config) {
>+  return absl::make_unique<GoogCcNetworkController>(event_log_, config, true);
>+}
>+
>+TimeDelta GoogCcFeedbackNetworkControllerFactory::GetProcessInterval() const {
>+  const int64_t kUpdateIntervalMs = 25;
>+  return TimeDelta::ms(kUpdateIntervalMs);
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/goog_cc_network_control.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/goog_cc_network_control.cc
>new file mode 100644
>index 00000000000..4c33054fea8
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/goog_cc_network_control.cc
>@@ -0,0 +1,545 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/congestion_controller/goog_cc/goog_cc_network_control.h"
>+
>+#include <algorithm>
>+#include <functional>
>+#include <limits>
>+#include <memory>
>+#include <numeric>
>+#include <string>
>+#include <utility>
>+#include <vector>
>+
>+#include "absl/memory/memory.h"
>+#include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h"
>+#include "modules/congestion_controller/goog_cc/alr_detector.h"
>+#include "modules/congestion_controller/goog_cc/include/goog_cc_factory.h"
>+#include "modules/congestion_controller/goog_cc/probe_controller.h"
>+#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
>+#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/format_macros.h"
>+#include "rtc_base/logging.h"
>+#include "rtc_base/timeutils.h"
>+#include "system_wrappers/include/field_trial.h"
>+
>+namespace webrtc {
>+namespace {
>+
>+const char kCwndExperiment[] = "WebRTC-CwndExperiment";
>+const int64_t kDefaultAcceptedQueueMs = 250;
>+
>+// From RTCPSender video report interval.
>+const TimeDelta kLossUpdateInterval = TimeDelta::ms(1000);
>+
>+// Pacing-rate relative to our target send rate.
>+// Multiplicative factor that is applied to the target bitrate to calculate
>+// the number of bytes that can be transmitted per interval.
>+// Increasing this factor will result in lower delays in cases of bitrate
>+// overshoots from the encoder.
>+const float kDefaultPaceMultiplier = 2.5f;
>+
>+bool CwndExperimentEnabled() {
>+  std::string experiment_string =
>+      webrtc::field_trial::FindFullName(kCwndExperiment);
>+  // The experiment is enabled iff the field trial string begins with "Enabled".
>+  return experiment_string.find("Enabled") == 0;
>+}
>+
>+bool ReadCwndExperimentParameter(int64_t* accepted_queue_ms) {
>+  RTC_DCHECK(accepted_queue_ms);
>+  std::string experiment_string =
>+      webrtc::field_trial::FindFullName(kCwndExperiment);
>+  int parsed_values =
>+      sscanf(experiment_string.c_str(), "Enabled-%" PRId64, accepted_queue_ms);
>+  if (parsed_values == 1) {
>+    RTC_CHECK_GE(*accepted_queue_ms, 0)
>+        << "Accepted must be greater than or equal to 0.";
>+    return true;
>+  }
>+  return false;
>+}
>+
>+// Makes sure that the bitrate and the min, max values are in valid range.
>+static void ClampBitrates(int64_t* bitrate_bps,
>+                          int64_t* min_bitrate_bps,
>+                          int64_t* max_bitrate_bps) {
>+  // TODO(holmer): We should make sure the default bitrates are set to 10 kbps,
>+  // and that we don't try to set the min bitrate to 0 from any applications.
>+  // The congestion controller should allow a min bitrate of 0.
>+  if (*min_bitrate_bps < congestion_controller::GetMinBitrateBps())
>+    *min_bitrate_bps = congestion_controller::GetMinBitrateBps();
>+  if (*max_bitrate_bps > 0)
>+    *max_bitrate_bps = std::max(*min_bitrate_bps, *max_bitrate_bps);
>+  if (*bitrate_bps > 0)
>+    *bitrate_bps = std::max(*min_bitrate_bps, *bitrate_bps);
>+}
>+
>+std::vector<PacketFeedback> ReceivedPacketsFeedbackAsRtp(
>+    const TransportPacketsFeedback report) {
>+  std::vector<PacketFeedback> packet_feedback_vector;
>+  for (auto& fb : report.PacketsWithFeedback()) {
>+    if (fb.receive_time.IsFinite()) {
>+      PacketFeedback pf(fb.receive_time.ms(), 0);
>+      pf.creation_time_ms = report.feedback_time.ms();
>+      if (fb.sent_packet.has_value()) {
>+        pf.payload_size = fb.sent_packet->size.bytes();
>+        pf.pacing_info = fb.sent_packet->pacing_info;
>+        pf.send_time_ms = fb.sent_packet->send_time.ms();
>+      } else {
>+        pf.send_time_ms = PacketFeedback::kNoSendTime;
>+      }
>+      packet_feedback_vector.push_back(pf);
>+    }
>+  }
>+  return packet_feedback_vector;
>+}
>+
>+int64_t GetBpsOrDefault(const absl::optional<DataRate>& rate,
>+                        int64_t fallback_bps) {
>+  if (rate && rate->IsFinite()) {
>+    return rate->bps();
>+  } else {
>+    return fallback_bps;
>+  }
>+}
>+
>+}  // namespace
>+
>+GoogCcNetworkController::GoogCcNetworkController(RtcEventLog* event_log,
>+                                                 NetworkControllerConfig config,
>+                                                 bool feedback_only)
>+    : event_log_(event_log),
>+      packet_feedback_only_(feedback_only),
>+      probe_controller_(new ProbeController()),
>+      bandwidth_estimation_(
>+          absl::make_unique<SendSideBandwidthEstimation>(event_log_)),
>+      alr_detector_(absl::make_unique<AlrDetector>()),
>+      delay_based_bwe_(new DelayBasedBwe(event_log_)),
>+      acknowledged_bitrate_estimator_(
>+          absl::make_unique<AcknowledgedBitrateEstimator>()),
>+      initial_config_(config),
>+      last_bandwidth_(config.starting_bandwidth),
>+      pacing_factor_(config.stream_based_config.pacing_factor.value_or(
>+          kDefaultPaceMultiplier)),
>+      min_pacing_rate_(config.stream_based_config.min_pacing_rate.value_or(
>+          DataRate::Zero())),
>+      max_padding_rate_(config.stream_based_config.max_padding_rate.value_or(
>+          DataRate::Zero())),
>+      max_total_allocated_bitrate_(DataRate::Zero()),
>+      in_cwnd_experiment_(CwndExperimentEnabled()),
>+      accepted_queue_ms_(kDefaultAcceptedQueueMs) {
>+  delay_based_bwe_->SetMinBitrate(congestion_controller::GetMinBitrateBps());
>+  if (in_cwnd_experiment_ &&
>+      !ReadCwndExperimentParameter(&accepted_queue_ms_)) {
>+    RTC_LOG(LS_WARNING) << "Failed to parse parameters for CwndExperiment "
>+                           "from field trial string. Experiment disabled.";
>+    in_cwnd_experiment_ = false;
>+  }
>+}
>+
>+GoogCcNetworkController::~GoogCcNetworkController() {}
>+
>+NetworkControlUpdate GoogCcNetworkController::OnNetworkAvailability(
>+    NetworkAvailability msg) {
>+  NetworkControlUpdate update;
>+  update.probe_cluster_configs = probe_controller_->OnNetworkAvailability(msg);
>+  return update;
>+}
>+
>+NetworkControlUpdate GoogCcNetworkController::OnNetworkRouteChange(
>+    NetworkRouteChange msg) {
>+  int64_t min_bitrate_bps = GetBpsOrDefault(msg.constraints.min_data_rate, -1);
>+  int64_t max_bitrate_bps = GetBpsOrDefault(msg.constraints.max_data_rate, -1);
>+  int64_t start_bitrate_bps = GetBpsOrDefault(msg.starting_rate, -1);
>+
>+  ClampBitrates(&start_bitrate_bps, &min_bitrate_bps, &max_bitrate_bps);
>+
>+  bandwidth_estimation_ =
>+      absl::make_unique<SendSideBandwidthEstimation>(event_log_);
>+  bandwidth_estimation_->SetBitrates(start_bitrate_bps, min_bitrate_bps,
>+                                     max_bitrate_bps);
>+  delay_based_bwe_.reset(new DelayBasedBwe(event_log_));
>+  acknowledged_bitrate_estimator_.reset(new AcknowledgedBitrateEstimator());
>+  delay_based_bwe_->SetStartBitrate(start_bitrate_bps);
>+  delay_based_bwe_->SetMinBitrate(min_bitrate_bps);
>+
>+  probe_controller_->Reset(msg.at_time.ms());
>+  NetworkControlUpdate update;
>+  update.probe_cluster_configs = probe_controller_->SetBitrates(
>+      min_bitrate_bps, start_bitrate_bps, max_bitrate_bps, msg.at_time.ms());
>+  MaybeTriggerOnNetworkChanged(&update, msg.at_time);
>+  return update;
>+}
>+
>+NetworkControlUpdate GoogCcNetworkController::OnProcessInterval(
>+    ProcessInterval msg) {
>+  NetworkControlUpdate update;
>+  if (initial_config_) {
>+    update.probe_cluster_configs = UpdateBitrateConstraints(
>+        initial_config_->constraints, initial_config_->starting_bandwidth);
>+    update.pacer_config = GetPacingRates(msg.at_time);
>+
>+    probe_controller_->EnablePeriodicAlrProbing(
>+        initial_config_->stream_based_config.requests_alr_probing);
>+    absl::optional<DataRate> total_bitrate =
>+        initial_config_->stream_based_config.max_total_allocated_bitrate;
>+    if (total_bitrate) {
>+      auto probes = probe_controller_->OnMaxTotalAllocatedBitrate(
>+          total_bitrate->bps(), msg.at_time.ms());
>+      update.probe_cluster_configs.insert(update.probe_cluster_configs.end(),
>+                                          probes.begin(), probes.end());
>+
>+      max_total_allocated_bitrate_ = *total_bitrate;
>+    }
>+    initial_config_.reset();
>+  }
>+
>+  bandwidth_estimation_->UpdateEstimate(msg.at_time.ms());
>+  absl::optional<int64_t> start_time_ms =
>+      alr_detector_->GetApplicationLimitedRegionStartTime();
>+  probe_controller_->SetAlrStartTimeMs(start_time_ms);
>+
>+  auto probes = probe_controller_->Process(msg.at_time.ms());
>+  update.probe_cluster_configs.insert(update.probe_cluster_configs.end(),
>+                                      probes.begin(), probes.end());
>+
>+  MaybeTriggerOnNetworkChanged(&update, msg.at_time);
>+  return update;
>+}
>+
>+NetworkControlUpdate GoogCcNetworkController::OnRemoteBitrateReport(
>+    RemoteBitrateReport msg) {
>+  if (packet_feedback_only_) {
>+    RTC_LOG(LS_ERROR) << "Received REMB for packet feedback only GoogCC";
>+    return NetworkControlUpdate();
>+  }
>+  bandwidth_estimation_->UpdateReceiverEstimate(msg.receive_time.ms(),
>+                                                msg.bandwidth.bps());
>+  BWE_TEST_LOGGING_PLOT(1, "REMB_kbps", msg.receive_time.ms(),
>+                        msg.bandwidth.bps() / 1000);
>+  return NetworkControlUpdate();
>+}
>+
>+NetworkControlUpdate GoogCcNetworkController::OnRoundTripTimeUpdate(
>+    RoundTripTimeUpdate msg) {
>+  if (packet_feedback_only_)
>+    return NetworkControlUpdate();
>+  if (msg.smoothed) {
>+    delay_based_bwe_->OnRttUpdate(msg.round_trip_time.ms());
>+  } else {
>+    bandwidth_estimation_->UpdateRtt(msg.round_trip_time.ms(),
>+                                     msg.receive_time.ms());
>+  }
>+  return NetworkControlUpdate();
>+}
>+
>+NetworkControlUpdate GoogCcNetworkController::OnSentPacket(
>+    SentPacket sent_packet) {
>+  alr_detector_->OnBytesSent(sent_packet.size.bytes(),
>+                             sent_packet.send_time.ms());
>+  return NetworkControlUpdate();
>+}
>+
>+NetworkControlUpdate GoogCcNetworkController::OnStreamsConfig(
>+    StreamsConfig msg) {
>+  NetworkControlUpdate update;
>+  probe_controller_->EnablePeriodicAlrProbing(msg.requests_alr_probing);
>+  if (msg.max_total_allocated_bitrate &&
>+      *msg.max_total_allocated_bitrate != max_total_allocated_bitrate_) {
>+    update.probe_cluster_configs =
>+        probe_controller_->OnMaxTotalAllocatedBitrate(
>+            msg.max_total_allocated_bitrate->bps(), msg.at_time.ms());
>+    max_total_allocated_bitrate_ = *msg.max_total_allocated_bitrate;
>+  }
>+  bool pacing_changed = false;
>+  if (msg.pacing_factor && *msg.pacing_factor != pacing_factor_) {
>+    pacing_factor_ = *msg.pacing_factor;
>+    pacing_changed = true;
>+  }
>+  if (msg.min_pacing_rate && *msg.min_pacing_rate != min_pacing_rate_) {
>+    min_pacing_rate_ = *msg.min_pacing_rate;
>+    pacing_changed = true;
>+  }
>+  if (msg.max_padding_rate && *msg.max_padding_rate != max_padding_rate_) {
>+    max_padding_rate_ = *msg.max_padding_rate;
>+    pacing_changed = true;
>+  }
>+  if (pacing_changed)
>+    update.pacer_config = GetPacingRates(msg.at_time);
>+  return update;
>+}
>+
>+NetworkControlUpdate GoogCcNetworkController::OnTargetRateConstraints(
>+    TargetRateConstraints constraints) {
>+  NetworkControlUpdate update;
>+  update.probe_cluster_configs =
>+      UpdateBitrateConstraints(constraints, absl::nullopt);
>+  MaybeTriggerOnNetworkChanged(&update, constraints.at_time);
>+  return update;
>+}
>+
>+std::vector<ProbeClusterConfig>
>+GoogCcNetworkController::UpdateBitrateConstraints(
>+    TargetRateConstraints constraints,
>+    absl::optional<DataRate> starting_rate) {
>+  int64_t min_bitrate_bps = GetBpsOrDefault(constraints.min_data_rate, 0);
>+  int64_t max_bitrate_bps = GetBpsOrDefault(constraints.max_data_rate, -1);
>+  int64_t start_bitrate_bps = GetBpsOrDefault(starting_rate, -1);
>+
>+  ClampBitrates(&start_bitrate_bps, &min_bitrate_bps, &max_bitrate_bps);
>+
>+  std::vector<ProbeClusterConfig> probes(probe_controller_->SetBitrates(
>+      min_bitrate_bps, start_bitrate_bps, max_bitrate_bps,
>+      constraints.at_time.ms()));
>+
>+  bandwidth_estimation_->SetBitrates(start_bitrate_bps, min_bitrate_bps,
>+                                     max_bitrate_bps);
>+  if (start_bitrate_bps > 0)
>+    delay_based_bwe_->SetStartBitrate(start_bitrate_bps);
>+  delay_based_bwe_->SetMinBitrate(min_bitrate_bps);
>+  return probes;
>+}
>+
>+NetworkControlUpdate GoogCcNetworkController::OnTransportLossReport(
>+    TransportLossReport msg) {
>+  if (packet_feedback_only_)
>+    return NetworkControlUpdate();
>+  int64_t total_packets_delta =
>+      msg.packets_received_delta + msg.packets_lost_delta;
>+  bandwidth_estimation_->UpdatePacketsLost(
>+      msg.packets_lost_delta, total_packets_delta, msg.receive_time.ms());
>+  return NetworkControlUpdate();
>+}
>+
>+NetworkControlUpdate GoogCcNetworkController::OnTransportPacketsFeedback(
>+    TransportPacketsFeedback report) {
>+  TimeDelta feedback_max_rtt = TimeDelta::MinusInfinity();
>+  Timestamp max_recv_time = Timestamp::ms(0);
>+  for (const auto& packet_feedback : report.ReceivedWithSendInfo()) {
>+    TimeDelta rtt =
>+        report.feedback_time - packet_feedback.sent_packet->send_time;
>+    // max() is used to account for feedback being delayed by the
>+    // receiver.
>+    feedback_max_rtt = std::max(feedback_max_rtt, rtt);
>+    max_recv_time = std::max(max_recv_time, packet_feedback.receive_time);
>+  }
>+  if (feedback_max_rtt.IsFinite()) {
>+    feedback_max_rtts_.push_back(feedback_max_rtt.ms());
>+    const size_t kMaxFeedbackRttWindow = 32;
>+    if (feedback_max_rtts_.size() > kMaxFeedbackRttWindow)
>+      feedback_max_rtts_.pop_front();
>+    min_feedback_max_rtt_ms_.emplace(*std::min_element(
>+        feedback_max_rtts_.begin(), feedback_max_rtts_.end()));
>+  }
>+  if (packet_feedback_only_) {
>+    if (!feedback_max_rtts_.empty()) {
>+      int64_t sum_rtt_ms = std::accumulate(feedback_max_rtts_.begin(),
>+                                           feedback_max_rtts_.end(), 0);
>+      int64_t mean_rtt_ms = sum_rtt_ms / feedback_max_rtts_.size();
>+      delay_based_bwe_->OnRttUpdate(mean_rtt_ms);
>+    }
>+
>+    TimeDelta feedback_min_rtt = TimeDelta::PlusInfinity();
>+    for (const auto& packet_feedback : report.ReceivedWithSendInfo()) {
>+      TimeDelta pending_time = packet_feedback.receive_time - max_recv_time;
>+      TimeDelta rtt = report.feedback_time -
>+                      packet_feedback.sent_packet->send_time - pending_time;
>+      // Value used for predicting NACK round trip time in FEC controller.
>+      feedback_min_rtt = std::min(rtt, feedback_min_rtt);
>+    }
>+    if (feedback_min_rtt.IsFinite()) {
>+      bandwidth_estimation_->UpdateRtt(feedback_min_rtt.ms(),
>+                                       report.feedback_time.ms());
>+    }
>+
>+    expected_packets_since_last_loss_update_ +=
>+        report.PacketsWithFeedback().size();
>+    for (const auto& packet_feedback : report.PacketsWithFeedback()) {
>+      lost_packets_since_last_loss_update_ +=
>+          packet_feedback.receive_time.IsFinite() ? 0 : 1;
>+    }
>+    if (report.feedback_time > next_loss_update_) {
>+      next_loss_update_ += kLossUpdateInterval;
>+      bandwidth_estimation_->UpdatePacketsLost(
>+          expected_packets_since_last_loss_update_,
>+          lost_packets_since_last_loss_update_, report.feedback_time.ms());
>+      expected_packets_since_last_loss_update_ = 0;
>+      lost_packets_since_last_loss_update_ = 0;
>+    }
>+  }
>+
>+  std::vector<PacketFeedback> received_feedback_vector =
>+      ReceivedPacketsFeedbackAsRtp(report);
>+
>+  absl::optional<int64_t> alr_start_time =
>+      alr_detector_->GetApplicationLimitedRegionStartTime();
>+
>+  if (previously_in_alr && !alr_start_time.has_value()) {
>+    int64_t now_ms = report.feedback_time.ms();
>+    acknowledged_bitrate_estimator_->SetAlrEndedTimeMs(now_ms);
>+    probe_controller_->SetAlrEndedTimeMs(now_ms);
>+  }
>+  previously_in_alr = alr_start_time.has_value();
>+  acknowledged_bitrate_estimator_->IncomingPacketFeedbackVector(
>+      received_feedback_vector);
>+  DelayBasedBwe::Result result;
>+  result = delay_based_bwe_->IncomingPacketFeedbackVector(
>+      received_feedback_vector, acknowledged_bitrate_estimator_->bitrate_bps(),
>+      report.feedback_time.ms());
>+  NetworkControlUpdate update;
>+  if (result.updated) {
>+    if (result.probe) {
>+      bandwidth_estimation_->SetSendBitrate(result.target_bitrate_bps);
>+    }
>+    // Since SetSendBitrate now resets the delay-based estimate, we have to call
>+    // UpdateDelayBasedEstimate after SetSendBitrate.
>+    bandwidth_estimation_->UpdateDelayBasedEstimate(report.feedback_time.ms(),
>+                                                    result.target_bitrate_bps);
>+    // Update the estimate in the ProbeController, in case we want to probe.
>+    MaybeTriggerOnNetworkChanged(&update, report.feedback_time);
>+  }
>+  if (result.recovered_from_overuse) {
>+    probe_controller_->SetAlrStartTimeMs(alr_start_time);
>+    auto probes = probe_controller_->RequestProbe(report.feedback_time.ms());
>+    update.probe_cluster_configs.insert(update.probe_cluster_configs.end(),
>+                                        probes.begin(), probes.end());
>+  }
>+  update.congestion_window = MaybeUpdateCongestionWindow();
>+  return update;
>+}
>+
>+NetworkControlUpdate GoogCcNetworkController::GetNetworkState(
>+    Timestamp at_time) const {
>+  DataRate bandwidth = DataRate::bps(last_estimated_bitrate_bps_);
>+  TimeDelta rtt = TimeDelta::ms(last_estimated_rtt_ms_);
>+  NetworkControlUpdate update;
>+  update.target_rate = TargetTransferRate();
>+  update.target_rate->network_estimate.at_time = at_time;
>+  update.target_rate->network_estimate.bandwidth = bandwidth;
>+  update.target_rate->network_estimate.loss_rate_ratio =
>+      last_estimated_fraction_loss_ / 255.0;
>+  update.target_rate->network_estimate.round_trip_time = rtt;
>+  update.target_rate->network_estimate.bwe_period =
>+      TimeDelta::ms(delay_based_bwe_->GetExpectedBwePeriodMs());
>+  update.target_rate->at_time = at_time;
>+  update.target_rate->target_rate = bandwidth;
>+  update.pacer_config = GetPacingRates(at_time);
>+  update.congestion_window = current_data_window_;
>+  return update;
>+}
>+
>+absl::optional<DataSize>
>+GoogCcNetworkController::MaybeUpdateCongestionWindow() {
>+  if (!in_cwnd_experiment_)
>+    return absl::nullopt;
>+  // No valid RTT. Could be because send-side BWE isn't used, in which case
>+  // we don't try to limit the outstanding packets.
>+  if (!min_feedback_max_rtt_ms_)
>+    return absl::nullopt;
>+
>+  const DataSize kMinCwnd = DataSize::bytes(2 * 1500);
>+  TimeDelta time_window =
>+      TimeDelta::ms(*min_feedback_max_rtt_ms_ + accepted_queue_ms_);
>+  DataSize data_window = last_bandwidth_ * time_window;
>+  if (current_data_window_) {
>+    data_window =
>+        std::max(kMinCwnd, (data_window + current_data_window_.value()) / 2);
>+  } else {
>+    data_window = std::max(kMinCwnd, data_window);
>+  }
>+  current_data_window_ = data_window;
>+  RTC_LOG(LS_INFO) << "Feedback rtt: " << *min_feedback_max_rtt_ms_
>+                   << " Bitrate: " << last_bandwidth_.bps();
>+  return data_window;
>+}
>+
>+void GoogCcNetworkController::MaybeTriggerOnNetworkChanged(
>+    NetworkControlUpdate* update,
>+    Timestamp at_time) {
>+  int32_t estimated_bitrate_bps;
>+  uint8_t fraction_loss;
>+  int64_t rtt_ms;
>+
>+  bool estimate_changed = GetNetworkParameters(
>+      &estimated_bitrate_bps, &fraction_loss, &rtt_ms, at_time);
>+  if (estimate_changed) {
>+    alr_detector_->SetEstimatedBitrate(estimated_bitrate_bps);
>+
>+    DataRate bandwidth = DataRate::bps(estimated_bitrate_bps);
>+    last_bandwidth_ = bandwidth;
>+
>+    TimeDelta bwe_period =
>+        TimeDelta::ms(delay_based_bwe_->GetExpectedBwePeriodMs());
>+
>+    TargetTransferRate target_rate;
>+    target_rate.at_time = at_time;
>+    // Set the target rate to the full estimated bandwidth since the estimation
>+    // for legacy reasons includes target rate constraints.
>+    target_rate.target_rate = bandwidth;
>+
>+    target_rate.network_estimate.at_time = at_time;
>+    target_rate.network_estimate.round_trip_time = TimeDelta::ms(rtt_ms);
>+    target_rate.network_estimate.bandwidth = bandwidth;
>+    target_rate.network_estimate.loss_rate_ratio = fraction_loss / 255.0f;
>+    target_rate.network_estimate.bwe_period = bwe_period;
>+    update->target_rate = target_rate;
>+
>+    auto probes =
>+        probe_controller_->SetEstimatedBitrate(bandwidth.bps(), at_time.ms());
>+    update->probe_cluster_configs.insert(update->probe_cluster_configs.end(),
>+                                         probes.begin(), probes.end());
>+    update->pacer_config = GetPacingRates(at_time);
>+  }
>+}
>+
>+bool GoogCcNetworkController::GetNetworkParameters(
>+    int32_t* estimated_bitrate_bps,
>+    uint8_t* fraction_loss,
>+    int64_t* rtt_ms,
>+    Timestamp at_time) {
>+  bandwidth_estimation_->CurrentEstimate(estimated_bitrate_bps, fraction_loss,
>+                                         rtt_ms);
>+  *estimated_bitrate_bps = std::max<int32_t>(
>+      *estimated_bitrate_bps, bandwidth_estimation_->GetMinBitrate());
>+
>+  bool estimate_changed = false;
>+  if ((*estimated_bitrate_bps != last_estimated_bitrate_bps_) ||
>+      (*fraction_loss != last_estimated_fraction_loss_) ||
>+      (*rtt_ms != last_estimated_rtt_ms_)) {
>+    last_estimated_bitrate_bps_ = *estimated_bitrate_bps;
>+    last_estimated_fraction_loss_ = *fraction_loss;
>+    last_estimated_rtt_ms_ = *rtt_ms;
>+    estimate_changed = true;
>+  }
>+
>+  BWE_TEST_LOGGING_PLOT(1, "fraction_loss_%", at_time.ms(),
>+                        (*fraction_loss * 100) / 256);
>+  BWE_TEST_LOGGING_PLOT(1, "rtt_ms", at_time.ms(), *rtt_ms);
>+  BWE_TEST_LOGGING_PLOT(1, "Target_bitrate_kbps", at_time.ms(),
>+                        *estimated_bitrate_bps / 1000);
>+
>+  return estimate_changed;
>+}
>+
>+PacerConfig GoogCcNetworkController::GetPacingRates(Timestamp at_time) const {
>+  DataRate pacing_rate =
>+      std::max(min_pacing_rate_, last_bandwidth_) * pacing_factor_;
>+  DataRate padding_rate = std::min(max_padding_rate_, last_bandwidth_);
>+  PacerConfig msg;
>+  msg.at_time = at_time;
>+  msg.time_window = TimeDelta::seconds(1);
>+  msg.data_window = pacing_rate * msg.time_window;
>+  msg.pad_window = padding_rate * msg.time_window;
>+  return msg;
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/goog_cc_network_control.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/goog_cc_network_control.h
>new file mode 100644
>index 00000000000..934511e2b55
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/goog_cc_network_control.h
>@@ -0,0 +1,109 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_GOOG_CC_NETWORK_CONTROL_H_
>+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_GOOG_CC_NETWORK_CONTROL_H_
>+
>+#include <stdint.h>
>+#include <deque>
>+#include <memory>
>+#include <vector>
>+
>+#include "absl/types/optional.h"
>+#include "api/transport/network_control.h"
>+#include "logging/rtc_event_log/rtc_event_log.h"
>+#include "modules/bitrate_controller/send_side_bandwidth_estimation.h"
>+#include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h"
>+#include "modules/congestion_controller/goog_cc/alr_detector.h"
>+#include "modules/congestion_controller/goog_cc/delay_based_bwe.h"
>+#include "modules/congestion_controller/goog_cc/probe_controller.h"
>+#include "rtc_base/constructormagic.h"
>+
>+namespace webrtc {
>+
>+class GoogCcNetworkController : public NetworkControllerInterface {
>+ public:
>+  GoogCcNetworkController(RtcEventLog* event_log,
>+                          NetworkControllerConfig config,
>+                          bool feedback_only);
>+  ~GoogCcNetworkController() override;
>+
>+  // NetworkControllerInterface
>+  NetworkControlUpdate OnNetworkAvailability(NetworkAvailability msg) override;
>+  NetworkControlUpdate OnNetworkRouteChange(NetworkRouteChange msg) override;
>+  NetworkControlUpdate OnProcessInterval(ProcessInterval msg) override;
>+  NetworkControlUpdate OnRemoteBitrateReport(RemoteBitrateReport msg) override;
>+  NetworkControlUpdate OnRoundTripTimeUpdate(RoundTripTimeUpdate msg) override;
>+  NetworkControlUpdate OnSentPacket(SentPacket msg) override;
>+  NetworkControlUpdate OnStreamsConfig(StreamsConfig msg) override;
>+  NetworkControlUpdate OnTargetRateConstraints(
>+      TargetRateConstraints msg) override;
>+  NetworkControlUpdate OnTransportLossReport(TransportLossReport msg) override;
>+  NetworkControlUpdate OnTransportPacketsFeedback(
>+      TransportPacketsFeedback msg) override;
>+
>+  NetworkControlUpdate GetNetworkState(Timestamp at_time) const;
>+
>+ private:
>+  std::vector<ProbeClusterConfig> UpdateBitrateConstraints(
>+      TargetRateConstraints constraints,
>+      absl::optional<DataRate> starting_rate);
>+  absl::optional<DataSize> MaybeUpdateCongestionWindow();
>+  void MaybeTriggerOnNetworkChanged(NetworkControlUpdate* update,
>+                                    Timestamp at_time);
>+  bool GetNetworkParameters(int32_t* estimated_bitrate_bps,
>+                            uint8_t* fraction_loss,
>+                            int64_t* rtt_ms,
>+                            Timestamp at_time);
>+  PacerConfig GetPacingRates(Timestamp at_time) const;
>+
>+  RtcEventLog* const event_log_;
>+  const bool packet_feedback_only_;
>+
>+  const std::unique_ptr<ProbeController> probe_controller_;
>+
>+  std::unique_ptr<SendSideBandwidthEstimation> bandwidth_estimation_;
>+  std::unique_ptr<AlrDetector> alr_detector_;
>+  std::unique_ptr<DelayBasedBwe> delay_based_bwe_;
>+  std::unique_ptr<AcknowledgedBitrateEstimator> acknowledged_bitrate_estimator_;
>+
>+  absl::optional<NetworkControllerConfig> initial_config_;
>+
>+  Timestamp next_loss_update_ = Timestamp::ms(0);
>+  int lost_packets_since_last_loss_update_ = 0;
>+  int expected_packets_since_last_loss_update_ = 0;
>+
>+  std::deque<int64_t> feedback_max_rtts_;
>+  absl::optional<int64_t> min_feedback_max_rtt_ms_;
>+
>+  DataRate last_bandwidth_;
>+  absl::optional<TargetTransferRate> last_target_rate_;
>+
>+  int32_t last_estimated_bitrate_bps_ = 0;
>+  uint8_t last_estimated_fraction_loss_ = 0;
>+  int64_t last_estimated_rtt_ms_ = 0;
>+
>+  double pacing_factor_;
>+  DataRate min_pacing_rate_;
>+  DataRate max_padding_rate_;
>+  DataRate max_total_allocated_bitrate_;
>+
>+  bool in_cwnd_experiment_;
>+  int64_t accepted_queue_ms_;
>+  bool previously_in_alr = false;
>+
>+  absl::optional<DataSize> current_data_window_;
>+
>+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(GoogCcNetworkController);
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_GOOG_CC_NETWORK_CONTROL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/include/goog_cc_factory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/include/goog_cc_factory.h
>new file mode 100644
>index 00000000000..0d91fff65fa
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/include/goog_cc_factory.h
>@@ -0,0 +1,48 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_INCLUDE_GOOG_CC_FACTORY_H_
>+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_INCLUDE_GOOG_CC_FACTORY_H_
>+#include <memory>
>+
>+#include "api/transport/network_control.h"
>+
>+namespace webrtc {
>+class Clock;
>+class RtcEventLog;
>+
>+class GoogCcNetworkControllerFactory
>+    : public NetworkControllerFactoryInterface {
>+ public:
>+  explicit GoogCcNetworkControllerFactory(RtcEventLog*);
>+  std::unique_ptr<NetworkControllerInterface> Create(
>+      NetworkControllerConfig config) override;
>+  TimeDelta GetProcessInterval() const override;
>+
>+ private:
>+  RtcEventLog* const event_log_;
>+};
>+
>+// Factory to create packet feedback only GoogCC, this can be used for
>+// connections providing packet receive time feedback but no other reports.
>+class GoogCcFeedbackNetworkControllerFactory
>+    : public NetworkControllerFactoryInterface {
>+ public:
>+  explicit GoogCcFeedbackNetworkControllerFactory(RtcEventLog*);
>+  std::unique_ptr<NetworkControllerInterface> Create(
>+      NetworkControllerConfig config) override;
>+  TimeDelta GetProcessInterval() const override;
>+
>+ private:
>+  RtcEventLog* const event_log_;
>+};
>+}  // namespace webrtc
>+
>+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_INCLUDE_GOOG_CC_FACTORY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/median_slope_estimator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/median_slope_estimator.cc
>similarity index 97%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/median_slope_estimator.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/median_slope_estimator.cc
>index ea762395775..0ad7728302e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/median_slope_estimator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/median_slope_estimator.cc
>@@ -8,7 +8,7 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "modules/congestion_controller/median_slope_estimator.h"
>+#include "modules/congestion_controller/goog_cc/median_slope_estimator.h"
> 
> #include <algorithm>
> #include <vector>
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/median_slope_estimator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/median_slope_estimator.h
>similarity index 91%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/median_slope_estimator.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/median_slope_estimator.h
>index f837b24a217..aeab4ea822c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/median_slope_estimator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/median_slope_estimator.h
>@@ -7,8 +7,8 @@
>  *  in the file PATENTS.  All contributing project authors may
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
>-#ifndef MODULES_CONGESTION_CONTROLLER_MEDIAN_SLOPE_ESTIMATOR_H_
>-#define MODULES_CONGESTION_CONTROLLER_MEDIAN_SLOPE_ESTIMATOR_H_
>+#ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_MEDIAN_SLOPE_ESTIMATOR_H_
>+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_MEDIAN_SLOPE_ESTIMATOR_H_
> 
> #include <stddef.h>
> #include <stdint.h>
>@@ -67,6 +67,7 @@ class MedianSlopeEstimator {
> 
>   RTC_DISALLOW_COPY_AND_ASSIGN(MedianSlopeEstimator);
> };
>+
> }  // namespace webrtc
> 
>-#endif  // MODULES_CONGESTION_CONTROLLER_MEDIAN_SLOPE_ESTIMATOR_H_
>+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_MEDIAN_SLOPE_ESTIMATOR_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/median_slope_estimator_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/median_slope_estimator_unittest.cc
>similarity index 96%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/median_slope_estimator_unittest.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/median_slope_estimator_unittest.cc
>index a5b90d8c354..3eb8c1fbf51 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/median_slope_estimator_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/median_slope_estimator_unittest.cc
>@@ -8,7 +8,7 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "modules/congestion_controller/median_slope_estimator.h"
>+#include "modules/congestion_controller/goog_cc/median_slope_estimator.h"
> #include "rtc_base/random.h"
> #include "test/gtest.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/probe_bitrate_estimator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/probe_bitrate_estimator.cc
>similarity index 93%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/probe_bitrate_estimator.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/probe_bitrate_estimator.cc
>index 8c754d789cd..0f0d17ea48d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/probe_bitrate_estimator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/probe_bitrate_estimator.cc
>@@ -8,17 +8,17 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "modules/congestion_controller/probe_bitrate_estimator.h"
>+#include "modules/congestion_controller/goog_cc/probe_bitrate_estimator.h"
> 
> #include <algorithm>
> 
>+#include "absl/memory/memory.h"
> #include "logging/rtc_event_log/events/rtc_event_probe_result_failure.h"
> #include "logging/rtc_event_log/events/rtc_event_probe_result_success.h"
> #include "logging/rtc_event_log/rtc_event_log.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/numerics/safe_conversions.h"
>-#include "rtc_base/ptr_util.h"
> 
> namespace {
> // The minumum number of probes we need to receive feedback about in percent
>@@ -66,8 +66,8 @@ int ProbeBitrateEstimator::HandleProbeAndEstimateBitrate(
> 
>   EraseOldClusters(packet_feedback.arrival_time_ms - kMaxClusterHistoryMs);
> 
>-  int payload_size_bits = rtc::dchecked_cast<int>(
>-      packet_feedback.payload_size * 8);
>+  int payload_size_bits =
>+      rtc::dchecked_cast<int>(packet_feedback.payload_size * 8);
>   AggregatedCluster* cluster = &clusters_[cluster_id];
> 
>   if (packet_feedback.send_time_ms < cluster->first_send_ms) {
>@@ -108,7 +108,7 @@ int ProbeBitrateEstimator::HandleProbeAndEstimateBitrate(
>                      << "] [send interval: " << send_interval_ms << " ms]"
>                      << " [receive interval: " << receive_interval_ms << " ms]";
>     if (event_log_) {
>-      event_log_->Log(rtc::MakeUnique<RtcEventProbeResultFailure>(
>+      event_log_->Log(absl::make_unique<RtcEventProbeResultFailure>(
>           cluster_id, ProbeFailureReason::kInvalidSendReceiveInterval));
>     }
>     return -1;
>@@ -141,7 +141,7 @@ int ProbeBitrateEstimator::HandleProbeAndEstimateBitrate(
>                      << send_bps / 1000 << " = " << ratio
>                      << " > kMaxValidRatio (" << kMaxValidRatio << ")]";
>     if (event_log_) {
>-      event_log_->Log(rtc::MakeUnique<RtcEventProbeResultFailure>(
>+      event_log_->Log(absl::make_unique<RtcEventProbeResultFailure>(
>           cluster_id, ProbeFailureReason::kInvalidSendReceiveRatio));
>     }
>     return -1;
>@@ -164,15 +164,15 @@ int ProbeBitrateEstimator::HandleProbeAndEstimateBitrate(
>   }
>   if (event_log_) {
>     event_log_->Log(
>-        rtc::MakeUnique<RtcEventProbeResultSuccess>(cluster_id, res));
>+        absl::make_unique<RtcEventProbeResultSuccess>(cluster_id, res));
>   }
>   estimated_bitrate_bps_ = res;
>   return *estimated_bitrate_bps_;
> }
> 
>-rtc::Optional<int>
>+absl::optional<int>
> ProbeBitrateEstimator::FetchAndResetLastEstimatedBitrateBps() {
>-  rtc::Optional<int> estimated_bitrate_bps = estimated_bitrate_bps_;
>+  absl::optional<int> estimated_bitrate_bps = estimated_bitrate_bps_;
>   estimated_bitrate_bps_.reset();
>   return estimated_bitrate_bps;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/probe_bitrate_estimator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/probe_bitrate_estimator.h
>similarity index 81%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/probe_bitrate_estimator.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/probe_bitrate_estimator.h
>index bb8a7ed534e..ed9252137fe 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/probe_bitrate_estimator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/probe_bitrate_estimator.h
>@@ -8,11 +8,11 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#ifndef MODULES_CONGESTION_CONTROLLER_PROBE_BITRATE_ESTIMATOR_H_
>-#define MODULES_CONGESTION_CONTROLLER_PROBE_BITRATE_ESTIMATOR_H_
>+#ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_PROBE_BITRATE_ESTIMATOR_H_
>+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_PROBE_BITRATE_ESTIMATOR_H_
> 
>-#include <map>
> #include <limits>
>+#include <map>
> 
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> 
>@@ -28,7 +28,7 @@ class ProbeBitrateEstimator {
>   // Returns the estimated bitrate if the probe completes a valid cluster.
>   int HandleProbeAndEstimateBitrate(const PacketFeedback& packet_feedback);
> 
>-  rtc::Optional<int> FetchAndResetLastEstimatedBitrateBps();
>+  absl::optional<int> FetchAndResetLastEstimatedBitrateBps();
> 
>  private:
>   struct AggregatedCluster {
>@@ -47,9 +47,9 @@ class ProbeBitrateEstimator {
> 
>   std::map<int, AggregatedCluster> clusters_;
>   RtcEventLog* const event_log_;
>-  rtc::Optional<int> estimated_bitrate_bps_;
>+  absl::optional<int> estimated_bitrate_bps_;
> };
> 
> }  // namespace webrtc
> 
>-#endif  // MODULES_CONGESTION_CONTROLLER_PROBE_BITRATE_ESTIMATOR_H_
>+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_PROBE_BITRATE_ESTIMATOR_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/probe_bitrate_estimator_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/probe_bitrate_estimator_unittest.cc
>similarity index 99%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/probe_bitrate_estimator_unittest.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/probe_bitrate_estimator_unittest.cc
>index 6d21de449de..f8a40f839f3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/probe_bitrate_estimator_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/probe_bitrate_estimator_unittest.cc
>@@ -8,10 +8,10 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "modules/congestion_controller/probe_bitrate_estimator.h"
>+#include "modules/congestion_controller/goog_cc/probe_bitrate_estimator.h"
> 
>-#include <vector>
> #include <utility>
>+#include <vector>
> 
> #include "modules/remote_bitrate_estimator/aimd_rate_control.h"
> #include "test/gmock.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/probe_controller.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/probe_controller.cc
>similarity index 65%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/probe_controller.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/probe_controller.cc
>index 8e6ae7cca14..c2a8e023d7d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/probe_controller.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/probe_controller.cc
>@@ -8,7 +8,7 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "modules/congestion_controller/probe_controller.h"
>+#include "modules/congestion_controller/goog_cc/probe_controller.h"
> 
> #include <algorithm>
> #include <initializer_list>
>@@ -21,6 +21,12 @@
> namespace webrtc {
> 
> namespace {
>+// The minimum number probing packets used.
>+constexpr int kMinProbePacketsSent = 5;
>+
>+// The minimum probing duration in ms.
>+constexpr int kMinProbeDurationMs = 15;
>+
> // Maximum waiting time from the time of initiating probing to getting
> // the measured results back.
> constexpr int64_t kMaxWaitingTimeForProbingResultMs = 1000;
>@@ -69,19 +75,20 @@ constexpr char kBweRapidRecoveryExperiment[] =
> 
> }  // namespace
> 
>-ProbeController::ProbeController(PacedSender* pacer, const Clock* clock)
>-    : pacer_(pacer), clock_(clock), enable_periodic_alr_probing_(false) {
>-  Reset();
>+ProbeController::ProbeController() : enable_periodic_alr_probing_(false) {
>+  Reset(0);
>   in_rapid_recovery_experiment_ = webrtc::field_trial::FindFullName(
>                                       kBweRapidRecoveryExperiment) == "Enabled";
> }
> 
>-void ProbeController::SetBitrates(int64_t min_bitrate_bps,
>-                                  int64_t start_bitrate_bps,
>-                                  int64_t max_bitrate_bps) {
>-  rtc::CritScope cs(&critsect_);
>+ProbeController::~ProbeController() {}
> 
>-  if (start_bitrate_bps > 0)  {
>+std::vector<ProbeClusterConfig> ProbeController::SetBitrates(
>+    int64_t min_bitrate_bps,
>+    int64_t start_bitrate_bps,
>+    int64_t max_bitrate_bps,
>+    int64_t at_time_ms) {
>+  if (start_bitrate_bps > 0) {
>     start_bitrate_bps_ = start_bitrate_bps;
>     estimated_bitrate_bps_ = start_bitrate_bps;
>   } else if (start_bitrate_bps_ == 0) {
>@@ -95,8 +102,8 @@ void ProbeController::SetBitrates(int64_t min_bitrate_bps,
> 
>   switch (state_) {
>     case State::kInit:
>-      if (network_state_ == kNetworkUp)
>-        InitiateExponentialProbing();
>+      if (network_available_)
>+        return InitiateExponentialProbing(at_time_ms);
>       break;
> 
>     case State::kWaitingForProbingResult:
>@@ -119,33 +126,59 @@ void ProbeController::SetBitrates(int64_t min_bitrate_bps,
>         RTC_HISTOGRAM_COUNTS_10000("WebRTC.BWE.MidCallProbing.Initiated",
>                                    max_bitrate_bps_ / 1000);
> 
>-        InitiateProbing(clock_->TimeInMilliseconds(), {max_bitrate_bps}, false);
>+        return InitiateProbing(at_time_ms, {max_bitrate_bps}, false);
>       }
>       break;
>   }
>+  return std::vector<ProbeClusterConfig>();
> }
> 
>-void ProbeController::OnNetworkStateChanged(NetworkState network_state) {
>-  rtc::CritScope cs(&critsect_);
>-  network_state_ = network_state;
>-  if (network_state_ == kNetworkUp && state_ == State::kInit)
>-    InitiateExponentialProbing();
>+std::vector<ProbeClusterConfig> ProbeController::OnMaxTotalAllocatedBitrate(
>+    int64_t max_total_allocated_bitrate,
>+    int64_t at_time_ms) {
>+  // TODO(philipel): Should |max_total_allocated_bitrate| be used as a limit for
>+  //                 ALR probing?
>+  if (state_ == State::kProbingComplete &&
>+      max_total_allocated_bitrate != max_total_allocated_bitrate_ &&
>+      estimated_bitrate_bps_ != 0 &&
>+      (max_bitrate_bps_ <= 0 || estimated_bitrate_bps_ < max_bitrate_bps_) &&
>+      estimated_bitrate_bps_ < max_total_allocated_bitrate) {
>+    max_total_allocated_bitrate_ = max_total_allocated_bitrate;
>+    return InitiateProbing(at_time_ms, {max_total_allocated_bitrate}, false);
>+  }
>+  return std::vector<ProbeClusterConfig>();
> }
> 
>-void ProbeController::InitiateExponentialProbing() {
>-  RTC_DCHECK(network_state_ == kNetworkUp);
>+std::vector<ProbeClusterConfig> ProbeController::OnNetworkAvailability(
>+    NetworkAvailability msg) {
>+  network_available_ = msg.network_available;
>+
>+  if (!network_available_ && state_ == State::kWaitingForProbingResult) {
>+    state_ = State::kProbingComplete;
>+    min_bitrate_to_probe_further_bps_ = kExponentialProbingDisabled;
>+  }
>+
>+  if (network_available_ && state_ == State::kInit && start_bitrate_bps_ > 0)
>+    return InitiateExponentialProbing(msg.at_time.ms());
>+  return std::vector<ProbeClusterConfig>();
>+}
>+
>+std::vector<ProbeClusterConfig> ProbeController::InitiateExponentialProbing(
>+    int64_t at_time_ms) {
>+  RTC_DCHECK(network_available_);
>   RTC_DCHECK(state_ == State::kInit);
>   RTC_DCHECK_GT(start_bitrate_bps_, 0);
> 
>   // When probing at 1.8 Mbps ( 6x 300), this represents a threshold of
>   // 1.2 Mbps to continue probing.
>-  InitiateProbing(clock_->TimeInMilliseconds(),
>-                  {3 * start_bitrate_bps_, 6 * start_bitrate_bps_}, true);
>+  return InitiateProbing(
>+      at_time_ms, {3 * start_bitrate_bps_, 6 * start_bitrate_bps_}, true);
> }
> 
>-void ProbeController::SetEstimatedBitrate(int64_t bitrate_bps) {
>-  rtc::CritScope cs(&critsect_);
>-  int64_t now_ms = clock_->TimeInMilliseconds();
>+std::vector<ProbeClusterConfig> ProbeController::SetEstimatedBitrate(
>+    int64_t bitrate_bps,
>+    int64_t at_time_ms) {
>+  int64_t now_ms = at_time_ms;
> 
>   if (mid_call_probing_waiting_for_result_ &&
>       bitrate_bps >= mid_call_probing_succcess_threshold_) {
>@@ -155,7 +188,7 @@ void ProbeController::SetEstimatedBitrate(int64_t bitrate_bps) {
>                                bitrate_bps / 1000);
>     mid_call_probing_waiting_for_result_ = false;
>   }
>-
>+  std::vector<ProbeClusterConfig> pending_probes;
>   if (state_ == State::kWaitingForProbingResult) {
>     // Continue probing if probing results indicate channel has greater
>     // capacity.
>@@ -166,7 +199,7 @@ void ProbeController::SetEstimatedBitrate(int64_t bitrate_bps) {
>     if (min_bitrate_to_probe_further_bps_ != kExponentialProbingDisabled &&
>         bitrate_bps > min_bitrate_to_probe_further_bps_) {
>       // Double the probing bitrate.
>-      InitiateProbing(now_ms, {2 * bitrate_bps}, true);
>+      pending_probes = InitiateProbing(now_ms, {2 * bitrate_bps}, true);
>     }
>   }
> 
>@@ -176,39 +209,41 @@ void ProbeController::SetEstimatedBitrate(int64_t bitrate_bps) {
>   }
> 
>   estimated_bitrate_bps_ = bitrate_bps;
>+  return pending_probes;
> }
> 
> void ProbeController::EnablePeriodicAlrProbing(bool enable) {
>-  rtc::CritScope cs(&critsect_);
>   enable_periodic_alr_probing_ = enable;
> }
> 
>+void ProbeController::SetAlrStartTimeMs(
>+    absl::optional<int64_t> alr_start_time_ms) {
>+  alr_start_time_ms_ = alr_start_time_ms;
>+}
> void ProbeController::SetAlrEndedTimeMs(int64_t alr_end_time_ms) {
>-  rtc::CritScope cs(&critsect_);
>   alr_end_time_ms_.emplace(alr_end_time_ms);
> }
> 
>-void ProbeController::RequestProbe() {
>-  int64_t now_ms = clock_->TimeInMilliseconds();
>-  rtc::CritScope cs(&critsect_);
>+std::vector<ProbeClusterConfig> ProbeController::RequestProbe(
>+    int64_t at_time_ms) {
>   // Called once we have returned to normal state after a large drop in
>   // estimated bandwidth. The current response is to initiate a single probe
>   // session (if not already probing) at the previous bitrate.
>   //
>   // If the probe session fails, the assumption is that this drop was a
>   // real one from a competing flow or a network change.
>-  bool in_alr = pacer_->GetApplicationLimitedRegionStartTime().has_value();
>+  bool in_alr = alr_start_time_ms_.has_value();
>   bool alr_ended_recently =
>       (alr_end_time_ms_.has_value() &&
>-       now_ms - alr_end_time_ms_.value() < kAlrEndedTimeoutMs);
>+       at_time_ms - alr_end_time_ms_.value() < kAlrEndedTimeoutMs);
>   if (in_alr || alr_ended_recently || in_rapid_recovery_experiment_) {
>     if (state_ == State::kProbingComplete) {
>       uint32_t suggested_probe_bps =
>           kProbeFractionAfterDrop * bitrate_before_last_large_drop_bps_;
>       uint32_t min_expected_probe_result_bps =
>           (1 - kProbeUncertainty) * suggested_probe_bps;
>-      int64_t time_since_drop_ms = now_ms - time_of_last_large_drop_ms_;
>-      int64_t time_since_probe_ms = now_ms - last_bwe_drop_probing_time_ms_;
>+      int64_t time_since_drop_ms = at_time_ms - time_of_last_large_drop_ms_;
>+      int64_t time_since_probe_ms = at_time_ms - last_bwe_drop_probing_time_ms_;
>       if (min_expected_probe_result_bps > estimated_bitrate_bps_ &&
>           time_since_drop_ms < kBitrateDropTimeoutMs &&
>           time_since_probe_ms > kMinTimeBetweenAlrProbesMs) {
>@@ -216,35 +251,34 @@ void ProbeController::RequestProbe() {
>         // Track how often we probe in response to bandwidth drop in ALR.
>         RTC_HISTOGRAM_COUNTS_10000(
>             "WebRTC.BWE.BweDropProbingIntervalInS",
>-            (now_ms - last_bwe_drop_probing_time_ms_) / 1000);
>-        InitiateProbing(now_ms, {suggested_probe_bps}, false);
>-        last_bwe_drop_probing_time_ms_ = now_ms;
>+            (at_time_ms - last_bwe_drop_probing_time_ms_) / 1000);
>+        return InitiateProbing(at_time_ms, {suggested_probe_bps}, false);
>+        last_bwe_drop_probing_time_ms_ = at_time_ms;
>       }
>     }
>   }
>+  return std::vector<ProbeClusterConfig>();
> }
> 
>-void ProbeController::Reset() {
>-  rtc::CritScope cs(&critsect_);
>-  network_state_ = kNetworkUp;
>+void ProbeController::Reset(int64_t at_time_ms) {
>+  network_available_ = true;
>   state_ = State::kInit;
>   min_bitrate_to_probe_further_bps_ = kExponentialProbingDisabled;
>   time_last_probing_initiated_ms_ = 0;
>   estimated_bitrate_bps_ = 0;
>   start_bitrate_bps_ = 0;
>   max_bitrate_bps_ = 0;
>-  int64_t now_ms = clock_->TimeInMilliseconds();
>+  int64_t now_ms = at_time_ms;
>   last_bwe_drop_probing_time_ms_ = now_ms;
>   alr_end_time_ms_.reset();
>   mid_call_probing_waiting_for_result_ = false;
>   time_of_last_large_drop_ms_ = now_ms;
>   bitrate_before_last_large_drop_bps_ = 0;
>+  max_total_allocated_bitrate_ = 0;
> }
> 
>-void ProbeController::Process() {
>-  rtc::CritScope cs(&critsect_);
>-
>-  int64_t now_ms = clock_->TimeInMilliseconds();
>+std::vector<ProbeClusterConfig> ProbeController::Process(int64_t at_time_ms) {
>+  int64_t now_ms = at_time_ms;
> 
>   if (now_ms - time_last_probing_initiated_ms_ >
>       kMaxWaitingTimeForProbingResultMs) {
>@@ -257,26 +291,25 @@ void ProbeController::Process() {
>     }
>   }
> 
>-  if (state_ != State::kProbingComplete || !enable_periodic_alr_probing_)
>-    return;
>-
>-  // Probe bandwidth periodically when in ALR state.
>-  rtc::Optional<int64_t> alr_start_time =
>-      pacer_->GetApplicationLimitedRegionStartTime();
>-  if (alr_start_time && estimated_bitrate_bps_ > 0) {
>-    int64_t next_probe_time_ms =
>-        std::max(*alr_start_time, time_last_probing_initiated_ms_) +
>-        kAlrPeriodicProbingIntervalMs;
>-    if (now_ms >= next_probe_time_ms) {
>-      InitiateProbing(now_ms, {estimated_bitrate_bps_ * 2}, true);
>+  if (enable_periodic_alr_probing_ && state_ == State::kProbingComplete) {
>+    // Probe bandwidth periodically when in ALR state.
>+    if (alr_start_time_ms_ && estimated_bitrate_bps_ > 0) {
>+      int64_t next_probe_time_ms =
>+          std::max(*alr_start_time_ms_, time_last_probing_initiated_ms_) +
>+          kAlrPeriodicProbingIntervalMs;
>+      if (now_ms >= next_probe_time_ms) {
>+        return InitiateProbing(now_ms, {estimated_bitrate_bps_ * 2}, true);
>+      }
>     }
>   }
>+  return std::vector<ProbeClusterConfig>();
> }
> 
>-void ProbeController::InitiateProbing(
>+std::vector<ProbeClusterConfig> ProbeController::InitiateProbing(
>     int64_t now_ms,
>     std::initializer_list<int64_t> bitrates_to_probe,
>     bool probe_further) {
>+  std::vector<ProbeClusterConfig> pending_probes;
>   for (int64_t bitrate : bitrates_to_probe) {
>     RTC_DCHECK_GT(bitrate, 0);
>     int64_t max_probe_bitrate_bps =
>@@ -285,7 +318,13 @@ void ProbeController::InitiateProbing(
>       bitrate = max_probe_bitrate_bps;
>       probe_further = false;
>     }
>-    pacer_->CreateProbeCluster(rtc::dchecked_cast<int>(bitrate));
>+
>+    ProbeClusterConfig config;
>+    config.at_time = Timestamp::ms(now_ms);
>+    config.target_data_rate = DataRate::bps(rtc::dchecked_cast<int>(bitrate));
>+    config.target_duration = TimeDelta::ms(kMinProbeDurationMs);
>+    config.target_probe_count = kMinProbePacketsSent;
>+    pending_probes.push_back(config);
>   }
>   time_last_probing_initiated_ms_ = now_ms;
>   if (probe_further) {
>@@ -296,6 +335,7 @@ void ProbeController::InitiateProbing(
>     state_ = State::kProbingComplete;
>     min_bitrate_to_probe_further_bps_ = kExponentialProbingDisabled;
>   }
>+  return pending_probes;
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/probe_controller.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/probe_controller.h
>new file mode 100644
>index 00000000000..6b6d4ae8b4b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/probe_controller.h
>@@ -0,0 +1,113 @@
>+/*
>+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_PROBE_CONTROLLER_H_
>+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_PROBE_CONTROLLER_H_
>+
>+#include <stdint.h>
>+
>+#include <initializer_list>
>+#include <vector>
>+
>+#include "absl/types/optional.h"
>+#include "api/transport/network_control.h"
>+#include "rtc_base/constructormagic.h"
>+#include "rtc_base/system/unused.h"
>+
>+namespace webrtc {
>+
>+class Clock;
>+
>+// This class controls initiation of probing to estimate initial channel
>+// capacity. There is also support for probing during a session when max
>+// bitrate is adjusted by an application.
>+class ProbeController {
>+ public:
>+  ProbeController();
>+  ~ProbeController();
>+
>+  RTC_WARN_UNUSED_RESULT std::vector<ProbeClusterConfig> SetBitrates(
>+      int64_t min_bitrate_bps,
>+      int64_t start_bitrate_bps,
>+      int64_t max_bitrate_bps,
>+      int64_t at_time_ms);
>+
>+  // The total bitrate, as opposed to the max bitrate, is the sum of the
>+  // configured bitrates for all active streams.
>+  RTC_WARN_UNUSED_RESULT std::vector<ProbeClusterConfig>
>+  OnMaxTotalAllocatedBitrate(int64_t max_total_allocated_bitrate,
>+                             int64_t at_time_ms);
>+
>+  RTC_WARN_UNUSED_RESULT std::vector<ProbeClusterConfig> OnNetworkAvailability(
>+      NetworkAvailability msg);
>+
>+  RTC_WARN_UNUSED_RESULT std::vector<ProbeClusterConfig> SetEstimatedBitrate(
>+      int64_t bitrate_bps,
>+      int64_t at_time_ms);
>+
>+  void EnablePeriodicAlrProbing(bool enable);
>+
>+  void SetAlrStartTimeMs(absl::optional<int64_t> alr_start_time);
>+  void SetAlrEndedTimeMs(int64_t alr_end_time);
>+
>+  RTC_WARN_UNUSED_RESULT std::vector<ProbeClusterConfig> RequestProbe(
>+      int64_t at_time_ms);
>+
>+  // Resets the ProbeController to a state equivalent to as if it was just
>+  // created EXCEPT for |enable_periodic_alr_probing_|.
>+  void Reset(int64_t at_time_ms);
>+
>+  RTC_WARN_UNUSED_RESULT std::vector<ProbeClusterConfig> Process(
>+      int64_t at_time_ms);
>+
>+ private:
>+  enum class State {
>+    // Initial state where no probing has been triggered yet.
>+    kInit,
>+    // Waiting for probing results to continue further probing.
>+    kWaitingForProbingResult,
>+    // Probing is complete.
>+    kProbingComplete,
>+  };
>+
>+  RTC_WARN_UNUSED_RESULT std::vector<ProbeClusterConfig>
>+  InitiateExponentialProbing(int64_t at_time_ms);
>+  RTC_WARN_UNUSED_RESULT std::vector<ProbeClusterConfig> InitiateProbing(
>+      int64_t now_ms,
>+      std::initializer_list<int64_t> bitrates_to_probe,
>+      bool probe_further);
>+
>+  bool network_available_;
>+  State state_;
>+  int64_t min_bitrate_to_probe_further_bps_;
>+  int64_t time_last_probing_initiated_ms_;
>+  int64_t estimated_bitrate_bps_;
>+  int64_t start_bitrate_bps_;
>+  int64_t max_bitrate_bps_;
>+  int64_t last_bwe_drop_probing_time_ms_;
>+  absl::optional<int64_t> alr_start_time_ms_;
>+  absl::optional<int64_t> alr_end_time_ms_;
>+  bool enable_periodic_alr_probing_;
>+  int64_t time_of_last_large_drop_ms_;
>+  int64_t bitrate_before_last_large_drop_bps_;
>+  int64_t max_total_allocated_bitrate_;
>+
>+  bool in_rapid_recovery_experiment_;
>+  // For WebRTC.BWE.MidCallProbing.* metric.
>+  bool mid_call_probing_waiting_for_result_;
>+  int64_t mid_call_probing_bitrate_bps_;
>+  int64_t mid_call_probing_succcess_threshold_;
>+
>+  RTC_DISALLOW_COPY_AND_ASSIGN(ProbeController);
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_PROBE_CONTROLLER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/probe_controller_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/probe_controller_unittest.cc
>new file mode 100644
>index 00000000000..57c81b91cc9
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/probe_controller_unittest.cc
>@@ -0,0 +1,266 @@
>+/*
>+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#include <memory>
>+
>+#include "api/transport/network_types.h"
>+#include "modules/congestion_controller/goog_cc/probe_controller.h"
>+#include "rtc_base/logging.h"
>+#include "system_wrappers/include/clock.h"
>+#include "test/gmock.h"
>+#include "test/gtest.h"
>+
>+using testing::_;
>+using testing::AtLeast;
>+using testing::Field;
>+using testing::Matcher;
>+using testing::NiceMock;
>+using testing::Return;
>+
>+namespace webrtc {
>+namespace test {
>+
>+namespace {
>+
>+constexpr int kMinBitrateBps = 100;
>+constexpr int kStartBitrateBps = 300;
>+constexpr int kMaxBitrateBps = 10000;
>+
>+constexpr int kExponentialProbingTimeoutMs = 5000;
>+
>+constexpr int kAlrProbeInterval = 5000;
>+constexpr int kAlrEndedTimeoutMs = 3000;
>+constexpr int kBitrateDropTimeoutMs = 5000;
>+}  // namespace
>+
>+class ProbeControllerTest : public ::testing::Test {
>+ protected:
>+  ProbeControllerTest() : clock_(100000000L) {
>+    probe_controller_.reset(new ProbeController());
>+  }
>+  ~ProbeControllerTest() override {}
>+
>+  std::vector<ProbeClusterConfig> SetNetworkAvailable(bool available) {
>+    NetworkAvailability msg;
>+    msg.at_time = Timestamp::ms(NowMs());
>+    msg.network_available = available;
>+    return probe_controller_->OnNetworkAvailability(msg);
>+  }
>+
>+  int64_t NowMs() { return clock_.TimeInMilliseconds(); }
>+
>+  SimulatedClock clock_;
>+  std::unique_ptr<ProbeController> probe_controller_;
>+};
>+
>+TEST_F(ProbeControllerTest, InitiatesProbingAtStart) {
>+  auto probes = probe_controller_->SetBitrates(kMinBitrateBps, kStartBitrateBps,
>+                                               kMaxBitrateBps, NowMs());
>+  EXPECT_GE(probes.size(), 2u);
>+}
>+
>+TEST_F(ProbeControllerTest, ProbeOnlyWhenNetworkIsUp) {
>+  SetNetworkAvailable(false);
>+  auto probes = probe_controller_->SetBitrates(kMinBitrateBps, kStartBitrateBps,
>+                                               kMaxBitrateBps, NowMs());
>+  EXPECT_EQ(probes.size(), 0u);
>+  probes = SetNetworkAvailable(true);
>+  EXPECT_GE(probes.size(), 2u);
>+}
>+
>+TEST_F(ProbeControllerTest, InitiatesProbingOnMaxBitrateIncrease) {
>+  auto probes = probe_controller_->SetBitrates(kMinBitrateBps, kStartBitrateBps,
>+                                               kMaxBitrateBps, NowMs());
>+  // Long enough to time out exponential probing.
>+  clock_.AdvanceTimeMilliseconds(kExponentialProbingTimeoutMs);
>+  probes = probe_controller_->SetEstimatedBitrate(kStartBitrateBps, NowMs());
>+  probes = probe_controller_->Process(NowMs());
>+  probes = probe_controller_->SetBitrates(kMinBitrateBps, kStartBitrateBps,
>+                                          kMaxBitrateBps + 100, NowMs());
>+  EXPECT_EQ(probes[0].target_data_rate.bps(), kMaxBitrateBps + 100);
>+}
>+
>+TEST_F(ProbeControllerTest, InitiatesProbingOnMaxBitrateIncreaseAtMaxBitrate) {
>+  auto probes = probe_controller_->SetBitrates(kMinBitrateBps, kStartBitrateBps,
>+                                               kMaxBitrateBps, NowMs());
>+  // Long enough to time out exponential probing.
>+  clock_.AdvanceTimeMilliseconds(kExponentialProbingTimeoutMs);
>+  probes = probe_controller_->SetEstimatedBitrate(kStartBitrateBps, NowMs());
>+  probes = probe_controller_->Process(NowMs());
>+  probes = probe_controller_->SetEstimatedBitrate(kMaxBitrateBps, NowMs());
>+  probes = probe_controller_->SetBitrates(kMinBitrateBps, kStartBitrateBps,
>+                                          kMaxBitrateBps + 100, NowMs());
>+  EXPECT_EQ(probes[0].target_data_rate.bps(), kMaxBitrateBps + 100);
>+}
>+
>+TEST_F(ProbeControllerTest, TestExponentialProbing) {
>+  auto probes = probe_controller_->SetBitrates(kMinBitrateBps, kStartBitrateBps,
>+                                               kMaxBitrateBps, NowMs());
>+
>+  // Repeated probe should only be sent when estimated bitrate climbs above
>+  // 0.7 * 6 * kStartBitrateBps = 1260.
>+  probes = probe_controller_->SetEstimatedBitrate(1000, NowMs());
>+  EXPECT_EQ(probes.size(), 0u);
>+
>+  probes = probe_controller_->SetEstimatedBitrate(1800, NowMs());
>+  EXPECT_EQ(probes[0].target_data_rate.bps(), 2 * 1800);
>+}
>+
>+TEST_F(ProbeControllerTest, TestExponentialProbingTimeout) {
>+  auto probes = probe_controller_->SetBitrates(kMinBitrateBps, kStartBitrateBps,
>+                                               kMaxBitrateBps, NowMs());
>+  // Advance far enough to cause a time out in waiting for probing result.
>+  clock_.AdvanceTimeMilliseconds(kExponentialProbingTimeoutMs);
>+  probes = probe_controller_->Process(NowMs());
>+
>+  probes = probe_controller_->SetEstimatedBitrate(1800, NowMs());
>+  EXPECT_EQ(probes.size(), 0u);
>+}
>+
>+TEST_F(ProbeControllerTest, RequestProbeInAlr) {
>+  auto probes = probe_controller_->SetBitrates(kMinBitrateBps, kStartBitrateBps,
>+                                               kMaxBitrateBps, NowMs());
>+  EXPECT_GE(probes.size(), 2u);
>+  probes = probe_controller_->SetEstimatedBitrate(500, NowMs());
>+
>+  probe_controller_->SetAlrStartTimeMs(clock_.TimeInMilliseconds());
>+  clock_.AdvanceTimeMilliseconds(kAlrProbeInterval + 1);
>+  probes = probe_controller_->Process(NowMs());
>+  probes = probe_controller_->SetEstimatedBitrate(250, NowMs());
>+  probes = probe_controller_->RequestProbe(NowMs());
>+
>+  EXPECT_EQ(probes.size(), 1u);
>+  EXPECT_EQ(probes[0].target_data_rate.bps(), 0.85 * 500);
>+}
>+
>+TEST_F(ProbeControllerTest, RequestProbeWhenAlrEndedRecently) {
>+  auto probes = probe_controller_->SetBitrates(kMinBitrateBps, kStartBitrateBps,
>+                                               kMaxBitrateBps, NowMs());
>+  EXPECT_EQ(probes.size(), 2u);
>+  probes = probe_controller_->SetEstimatedBitrate(500, NowMs());
>+
>+  probe_controller_->SetAlrStartTimeMs(absl::nullopt);
>+  clock_.AdvanceTimeMilliseconds(kAlrProbeInterval + 1);
>+  probes = probe_controller_->Process(NowMs());
>+  probes = probe_controller_->SetEstimatedBitrate(250, NowMs());
>+  probe_controller_->SetAlrEndedTimeMs(clock_.TimeInMilliseconds());
>+  clock_.AdvanceTimeMilliseconds(kAlrEndedTimeoutMs - 1);
>+  probes = probe_controller_->RequestProbe(NowMs());
>+
>+  EXPECT_EQ(probes.size(), 1u);
>+  EXPECT_EQ(probes[0].target_data_rate.bps(), 0.85 * 500);
>+}
>+
>+TEST_F(ProbeControllerTest, RequestProbeWhenAlrNotEndedRecently) {
>+  auto probes = probe_controller_->SetBitrates(kMinBitrateBps, kStartBitrateBps,
>+                                               kMaxBitrateBps, NowMs());
>+  EXPECT_EQ(probes.size(), 2u);
>+  probes = probe_controller_->SetEstimatedBitrate(500, NowMs());
>+
>+  probe_controller_->SetAlrStartTimeMs(absl::nullopt);
>+  clock_.AdvanceTimeMilliseconds(kAlrProbeInterval + 1);
>+  probes = probe_controller_->Process(NowMs());
>+  probes = probe_controller_->SetEstimatedBitrate(250, NowMs());
>+  probe_controller_->SetAlrEndedTimeMs(clock_.TimeInMilliseconds());
>+  clock_.AdvanceTimeMilliseconds(kAlrEndedTimeoutMs + 1);
>+  probes = probe_controller_->RequestProbe(NowMs());
>+  EXPECT_EQ(probes.size(), 0u);
>+}
>+
>+TEST_F(ProbeControllerTest, RequestProbeWhenBweDropNotRecent) {
>+  auto probes = probe_controller_->SetBitrates(kMinBitrateBps, kStartBitrateBps,
>+                                               kMaxBitrateBps, NowMs());
>+  EXPECT_EQ(probes.size(), 2u);
>+  probes = probe_controller_->SetEstimatedBitrate(500, NowMs());
>+
>+  probe_controller_->SetAlrStartTimeMs(clock_.TimeInMilliseconds());
>+  clock_.AdvanceTimeMilliseconds(kAlrProbeInterval + 1);
>+  probes = probe_controller_->Process(NowMs());
>+  probes = probe_controller_->SetEstimatedBitrate(250, NowMs());
>+  clock_.AdvanceTimeMilliseconds(kBitrateDropTimeoutMs + 1);
>+  probes = probe_controller_->RequestProbe(NowMs());
>+  EXPECT_EQ(probes.size(), 0u);
>+}
>+
>+TEST_F(ProbeControllerTest, PeriodicProbing) {
>+  probe_controller_->EnablePeriodicAlrProbing(true);
>+  auto probes = probe_controller_->SetBitrates(kMinBitrateBps, kStartBitrateBps,
>+                                               kMaxBitrateBps, NowMs());
>+  EXPECT_EQ(probes.size(), 2u);
>+  probes = probe_controller_->SetEstimatedBitrate(500, NowMs());
>+
>+  int64_t start_time = clock_.TimeInMilliseconds();
>+
>+  // Expect the controller to send a new probe after 5s has passed.
>+  probe_controller_->SetAlrStartTimeMs(start_time);
>+  clock_.AdvanceTimeMilliseconds(5000);
>+  probes = probe_controller_->Process(NowMs());
>+  EXPECT_EQ(probes.size(), 1u);
>+  EXPECT_EQ(probes[0].target_data_rate.bps(), 1000);
>+
>+  probes = probe_controller_->SetEstimatedBitrate(500, NowMs());
>+
>+  // The following probe should be sent at 10s into ALR.
>+  probe_controller_->SetAlrStartTimeMs(start_time);
>+  clock_.AdvanceTimeMilliseconds(4000);
>+  probes = probe_controller_->Process(NowMs());
>+  probes = probe_controller_->SetEstimatedBitrate(500, NowMs());
>+  EXPECT_EQ(probes.size(), 0u);
>+
>+  probe_controller_->SetAlrStartTimeMs(start_time);
>+  clock_.AdvanceTimeMilliseconds(1000);
>+  probes = probe_controller_->Process(NowMs());
>+  EXPECT_EQ(probes.size(), 1u);
>+  probes = probe_controller_->SetEstimatedBitrate(500, NowMs());
>+  EXPECT_EQ(probes.size(), 0u);
>+}
>+
>+TEST_F(ProbeControllerTest, PeriodicProbingAfterReset) {
>+  probe_controller_.reset(new ProbeController());
>+  int64_t alr_start_time = clock_.TimeInMilliseconds();
>+
>+  probe_controller_->SetAlrStartTimeMs(alr_start_time);
>+  probe_controller_->EnablePeriodicAlrProbing(true);
>+  auto probes = probe_controller_->SetBitrates(kMinBitrateBps, kStartBitrateBps,
>+                                               kMaxBitrateBps, NowMs());
>+  probe_controller_->Reset(NowMs());
>+
>+  clock_.AdvanceTimeMilliseconds(10000);
>+  probes = probe_controller_->Process(NowMs());
>+  // Since bitrates are not yet set, no probe is sent event though we are in ALR
>+  // mode.
>+  EXPECT_EQ(probes.size(), 0u);
>+
>+  probes = probe_controller_->SetBitrates(kMinBitrateBps, kStartBitrateBps,
>+                                          kMaxBitrateBps, NowMs());
>+  EXPECT_EQ(probes.size(), 2u);
>+
>+  // Make sure we use |kStartBitrateBps| as the estimated bitrate
>+  // until SetEstimatedBitrate is called with an updated estimate.
>+  clock_.AdvanceTimeMilliseconds(10000);
>+  probes = probe_controller_->Process(NowMs());
>+  EXPECT_EQ(probes[0].target_data_rate.bps(), kStartBitrateBps * 2);
>+}
>+
>+TEST_F(ProbeControllerTest, TestExponentialProbingOverflow) {
>+  const int64_t kMbpsMultiplier = 1000000;
>+  auto probes = probe_controller_->SetBitrates(
>+      kMinBitrateBps, 10 * kMbpsMultiplier, 100 * kMbpsMultiplier, NowMs());
>+  // Verify that probe bitrate is capped at the specified max bitrate.
>+  probes =
>+      probe_controller_->SetEstimatedBitrate(60 * kMbpsMultiplier, NowMs());
>+  EXPECT_EQ(probes[0].target_data_rate.bps(), 100 * kMbpsMultiplier);
>+  // Verify that repeated probes aren't sent.
>+  probes =
>+      probe_controller_->SetEstimatedBitrate(100 * kMbpsMultiplier, NowMs());
>+  EXPECT_EQ(probes.size(), 0u);
>+}
>+
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/test/goog_cc_printer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/test/goog_cc_printer.cc
>new file mode 100644
>index 00000000000..33310d18c65
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/test/goog_cc_printer.cc
>@@ -0,0 +1,48 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#include "modules/congestion_controller/goog_cc/test/goog_cc_printer.h"
>+namespace webrtc {
>+
>+GoogCcStatePrinter::GoogCcStatePrinter() = default;
>+GoogCcStatePrinter::~GoogCcStatePrinter() = default;
>+
>+void GoogCcStatePrinter::Attach(GoogCcNetworkController* controller) {
>+  controller_ = controller;
>+}
>+
>+bool GoogCcStatePrinter::Attached() const {
>+  return controller_ != nullptr;
>+}
>+
>+void GoogCcStatePrinter::PrintHeaders(FILE* out) {}
>+
>+void GoogCcStatePrinter::PrintValues(FILE* out) {
>+  RTC_CHECK(controller_);
>+}
>+
>+NetworkControlUpdate GoogCcStatePrinter::GetState(Timestamp at_time) const {
>+  RTC_CHECK(controller_);
>+  return controller_->GetNetworkState(at_time);
>+}
>+
>+GoogCcDebugFactory::GoogCcDebugFactory(RtcEventLog* event_log,
>+                                       GoogCcStatePrinter* printer)
>+    : GoogCcNetworkControllerFactory(event_log), printer_(printer) {}
>+
>+std::unique_ptr<NetworkControllerInterface> GoogCcDebugFactory::Create(
>+    NetworkControllerConfig config) {
>+  RTC_CHECK(controller_ == nullptr);
>+  auto controller = GoogCcNetworkControllerFactory::Create(config);
>+  controller_ = static_cast<GoogCcNetworkController*>(controller.get());
>+  printer_->Attach(controller_);
>+  return controller;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/test/goog_cc_printer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/test/goog_cc_printer.h
>new file mode 100644
>index 00000000000..e5bd891c4f0
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/test/goog_cc_printer.h
>@@ -0,0 +1,48 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_TEST_GOOG_CC_PRINTER_H_
>+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_TEST_GOOG_CC_PRINTER_H_
>+
>+#include <memory>
>+
>+#include "modules/congestion_controller/goog_cc/goog_cc_network_control.h"
>+#include "modules/congestion_controller/goog_cc/include/goog_cc_factory.h"
>+#include "modules/congestion_controller/test/controller_printer.h"
>+
>+namespace webrtc {
>+class GoogCcStatePrinter : public DebugStatePrinter {
>+ public:
>+  GoogCcStatePrinter();
>+  ~GoogCcStatePrinter() override;
>+  void Attach(GoogCcNetworkController*);
>+  bool Attached() const override;
>+
>+  void PrintHeaders(FILE* out) override;
>+  void PrintValues(FILE* out) override;
>+
>+  NetworkControlUpdate GetState(Timestamp at_time) const override;
>+
>+ private:
>+  GoogCcNetworkController* controller_ = nullptr;
>+};
>+
>+class GoogCcDebugFactory : public GoogCcNetworkControllerFactory {
>+ public:
>+  GoogCcDebugFactory(RtcEventLog* event_log, GoogCcStatePrinter* printer);
>+  std::unique_ptr<NetworkControllerInterface> Create(
>+      NetworkControllerConfig config) override;
>+
>+ private:
>+  GoogCcStatePrinter* printer_;
>+  GoogCcNetworkController* controller_ = nullptr;
>+};
>+}  // namespace webrtc
>+
>+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_TEST_GOOG_CC_PRINTER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/trendline_estimator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/trendline_estimator.cc
>new file mode 100644
>index 00000000000..14aae9519b5
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/trendline_estimator.cc
>@@ -0,0 +1,183 @@
>+/*
>+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/congestion_controller/goog_cc/trendline_estimator.h"
>+
>+#include <math.h>
>+
>+#include <algorithm>
>+
>+#include "absl/types/optional.h"
>+#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/numerics/safe_minmax.h"
>+
>+namespace webrtc {
>+
>+namespace {
>+absl::optional<double> LinearFitSlope(
>+    const std::deque<std::pair<double, double>>& points) {
>+  RTC_DCHECK(points.size() >= 2);
>+  // Compute the "center of mass".
>+  double sum_x = 0;
>+  double sum_y = 0;
>+  for (const auto& point : points) {
>+    sum_x += point.first;
>+    sum_y += point.second;
>+  }
>+  double x_avg = sum_x / points.size();
>+  double y_avg = sum_y / points.size();
>+  // Compute the slope k = \sum (x_i-x_avg)(y_i-y_avg) / \sum (x_i-x_avg)^2
>+  double numerator = 0;
>+  double denominator = 0;
>+  for (const auto& point : points) {
>+    numerator += (point.first - x_avg) * (point.second - y_avg);
>+    denominator += (point.first - x_avg) * (point.first - x_avg);
>+  }
>+  if (denominator == 0)
>+    return absl::nullopt;
>+  return numerator / denominator;
>+}
>+
>+constexpr double kMaxAdaptOffsetMs = 15.0;
>+constexpr double kOverUsingTimeThreshold = 10;
>+constexpr int kMinNumDeltas = 60;
>+
>+}  // namespace
>+
>+enum { kDeltaCounterMax = 1000 };
>+
>+TrendlineEstimator::TrendlineEstimator(size_t window_size,
>+                                       double smoothing_coef,
>+                                       double threshold_gain)
>+    : window_size_(window_size),
>+      smoothing_coef_(smoothing_coef),
>+      threshold_gain_(threshold_gain),
>+      num_of_deltas_(0),
>+      first_arrival_time_ms_(-1),
>+      accumulated_delay_(0),
>+      smoothed_delay_(0),
>+      delay_hist_(),
>+      trendline_(0),
>+      k_up_(0.0087),
>+      k_down_(0.039),
>+      overusing_time_threshold_(kOverUsingTimeThreshold),
>+      threshold_(12.5),
>+      last_update_ms_(-1),
>+      prev_offset_(0.0),
>+      time_over_using_(-1),
>+      overuse_counter_(0),
>+      hypothesis_(BandwidthUsage::kBwNormal) {}
>+
>+TrendlineEstimator::~TrendlineEstimator() {}
>+
>+void TrendlineEstimator::Update(double recv_delta_ms,
>+                                double send_delta_ms,
>+                                int64_t arrival_time_ms) {
>+  const double delta_ms = recv_delta_ms - send_delta_ms;
>+  ++num_of_deltas_;
>+  if (num_of_deltas_ > kDeltaCounterMax)
>+    num_of_deltas_ = kDeltaCounterMax;
>+  if (first_arrival_time_ms_ == -1)
>+    first_arrival_time_ms_ = arrival_time_ms;
>+
>+  // Exponential backoff filter.
>+  accumulated_delay_ += delta_ms;
>+  BWE_TEST_LOGGING_PLOT(1, "accumulated_delay_ms", arrival_time_ms,
>+                        accumulated_delay_);
>+  smoothed_delay_ = smoothing_coef_ * smoothed_delay_ +
>+                    (1 - smoothing_coef_) * accumulated_delay_;
>+  BWE_TEST_LOGGING_PLOT(1, "smoothed_delay_ms", arrival_time_ms,
>+                        smoothed_delay_);
>+
>+  // Simple linear regression.
>+  delay_hist_.push_back(std::make_pair(
>+      static_cast<double>(arrival_time_ms - first_arrival_time_ms_),
>+      smoothed_delay_));
>+  if (delay_hist_.size() > window_size_)
>+    delay_hist_.pop_front();
>+  if (delay_hist_.size() == window_size_) {
>+    // Only update trendline_ if it is possible to fit a line to the data.
>+    trendline_ = LinearFitSlope(delay_hist_).value_or(trendline_);
>+  }
>+
>+  BWE_TEST_LOGGING_PLOT(1, "trendline_slope", arrival_time_ms, trendline_);
>+
>+  Detect(trendline_slope(), send_delta_ms, num_of_deltas(), arrival_time_ms);
>+}
>+
>+BandwidthUsage TrendlineEstimator::State() const {
>+  return hypothesis_;
>+}
>+
>+void TrendlineEstimator::Detect(double offset,
>+                                double ts_delta,
>+                                int num_of_deltas,
>+                                int64_t now_ms) {
>+  if (num_of_deltas < 2) {
>+    hypothesis_ = BandwidthUsage::kBwNormal;
>+    return;
>+  }
>+  const double T = std::min(num_of_deltas, kMinNumDeltas) * offset;
>+  BWE_TEST_LOGGING_PLOT(1, "T", now_ms, T);
>+  BWE_TEST_LOGGING_PLOT(1, "threshold", now_ms, threshold_);
>+  if (T > threshold_) {
>+    if (time_over_using_ == -1) {
>+      // Initialize the timer. Assume that we've been
>+      // over-using half of the time since the previous
>+      // sample.
>+      time_over_using_ = ts_delta / 2;
>+    } else {
>+      // Increment timer
>+      time_over_using_ += ts_delta;
>+    }
>+    overuse_counter_++;
>+    if (time_over_using_ > overusing_time_threshold_ && overuse_counter_ > 1) {
>+      if (offset >= prev_offset_) {
>+        time_over_using_ = 0;
>+        overuse_counter_ = 0;
>+        hypothesis_ = BandwidthUsage::kBwOverusing;
>+      }
>+    }
>+  } else if (T < -threshold_) {
>+    time_over_using_ = -1;
>+    overuse_counter_ = 0;
>+    hypothesis_ = BandwidthUsage::kBwUnderusing;
>+  } else {
>+    time_over_using_ = -1;
>+    overuse_counter_ = 0;
>+    hypothesis_ = BandwidthUsage::kBwNormal;
>+  }
>+  prev_offset_ = offset;
>+
>+  UpdateThreshold(T, now_ms);
>+}
>+
>+void TrendlineEstimator::UpdateThreshold(double modified_offset,
>+                                         int64_t now_ms) {
>+  if (last_update_ms_ == -1)
>+    last_update_ms_ = now_ms;
>+
>+  if (fabs(modified_offset) > threshold_ + kMaxAdaptOffsetMs) {
>+    // Avoid adapting the threshold to big latency spikes, caused e.g.,
>+    // by a sudden capacity drop.
>+    last_update_ms_ = now_ms;
>+    return;
>+  }
>+
>+  const double k = fabs(modified_offset) < threshold_ ? k_down_ : k_up_;
>+  const int64_t kMaxTimeDeltaMs = 100;
>+  int64_t time_delta_ms = std::min(now_ms - last_update_ms_, kMaxTimeDeltaMs);
>+  threshold_ += k * (fabs(modified_offset) - threshold_) * time_delta_ms;
>+  threshold_ = rtc::SafeClamp(threshold_, 6.f, 600.f);
>+  last_update_ms_ = now_ms;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/trendline_estimator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/trendline_estimator.h
>similarity index 71%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/trendline_estimator.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/trendline_estimator.h
>index 7ee27cdd5d4..2841b491525 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/trendline_estimator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/trendline_estimator.h
>@@ -7,8 +7,8 @@
>  *  in the file PATENTS.  All contributing project authors may
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
>-#ifndef MODULES_CONGESTION_CONTROLLER_TRENDLINE_ESTIMATOR_H_
>-#define MODULES_CONGESTION_CONTROLLER_TRENDLINE_ESTIMATOR_H_
>+#ifndef MODULES_CONGESTION_CONTROLLER_GOOG_CC_TRENDLINE_ESTIMATOR_H_
>+#define MODULES_CONGESTION_CONTROLLER_GOOG_CC_TRENDLINE_ESTIMATOR_H_
> 
> #include <stddef.h>
> #include <stdint.h>
>@@ -16,11 +16,12 @@
> #include <deque>
> #include <utility>
> 
>+#include "modules/congestion_controller/goog_cc/delay_increase_detector_interface.h"
> #include "rtc_base/constructormagic.h"
> 
> namespace webrtc {
> 
>-class TrendlineEstimator {
>+class TrendlineEstimator : public DelayIncreaseDetectorInterface {
>  public:
>   // |window_size| is the number of points required to compute a trend line.
>   // |smoothing_coef| controls how much we smooth out the delay before fitting
>@@ -31,13 +32,16 @@ class TrendlineEstimator {
>   TrendlineEstimator(size_t window_size,
>                      double smoothing_coef,
>                      double threshold_gain);
>-  ~TrendlineEstimator();
>+
>+  ~TrendlineEstimator() override;
> 
>   // Update the estimator with a new sample. The deltas should represent deltas
>   // between timestamp groups as defined by the InterArrival class.
>   void Update(double recv_delta_ms,
>               double send_delta_ms,
>-              int64_t arrival_time_ms);
>+              int64_t arrival_time_ms) override;
>+
>+  BandwidthUsage State() const override;
> 
>   // Returns the estimated trend k multiplied by some gain.
>   // 0 < k < 1   ->  the delay increases, queues are filling up
>@@ -49,6 +53,13 @@ class TrendlineEstimator {
>   unsigned int num_of_deltas() const { return num_of_deltas_; }
> 
>  private:
>+  void Detect(double offset,
>+              double ts_delta,
>+              int num_of_deltas,
>+              int64_t now_ms);
>+
>+  void UpdateThreshold(double modified_offset, int64_t now_ms);
>+
>   // Parameters.
>   const size_t window_size_;
>   const double smoothing_coef_;
>@@ -56,7 +67,7 @@ class TrendlineEstimator {
>   // Used by the existing threshold.
>   unsigned int num_of_deltas_;
>   // Keep the arrival times small by using the change from the first packet.
>-  int64_t first_arrival_time_ms;
>+  int64_t first_arrival_time_ms_;
>   // Exponential backoff filtering.
>   double accumulated_delay_;
>   double smoothed_delay_;
>@@ -64,8 +75,18 @@ class TrendlineEstimator {
>   std::deque<std::pair<double, double>> delay_hist_;
>   double trendline_;
> 
>+  const double k_up_;
>+  const double k_down_;
>+  double overusing_time_threshold_;
>+  double threshold_;
>+  int64_t last_update_ms_;
>+  double prev_offset_;
>+  double time_over_using_;
>+  int overuse_counter_;
>+  BandwidthUsage hypothesis_;
>+
>   RTC_DISALLOW_COPY_AND_ASSIGN(TrendlineEstimator);
> };
> }  // namespace webrtc
> 
>-#endif  // MODULES_CONGESTION_CONTROLLER_TRENDLINE_ESTIMATOR_H_
>+#endif  // MODULES_CONGESTION_CONTROLLER_GOOG_CC_TRENDLINE_ESTIMATOR_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/trendline_estimator_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/trendline_estimator_unittest.cc
>similarity index 97%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/trendline_estimator_unittest.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/trendline_estimator_unittest.cc
>index e44f5032b02..aa150154b7f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/trendline_estimator_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/goog_cc/trendline_estimator_unittest.cc
>@@ -8,7 +8,7 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "modules/congestion_controller/trendline_estimator.h"
>+#include "modules/congestion_controller/goog_cc/trendline_estimator.h"
> #include "rtc_base/random.h"
> #include "test/gtest.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/include/mock/mock_congestion_observer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/include/mock/mock_congestion_observer.h
>index c6883d976b8..d9c24a23ca9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/include/mock/mock_congestion_observer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/include/mock/mock_congestion_observer.h
>@@ -11,13 +11,13 @@
> #ifndef MODULES_CONGESTION_CONTROLLER_INCLUDE_MOCK_MOCK_CONGESTION_OBSERVER_H_
> #define MODULES_CONGESTION_CONTROLLER_INCLUDE_MOCK_MOCK_CONGESTION_OBSERVER_H_
> 
>-#include "modules/congestion_controller/include/send_side_congestion_controller.h"
>+#include "modules/congestion_controller/include/network_changed_observer.h"
> #include "test/gmock.h"
> 
> namespace webrtc {
> namespace test {
> 
>-class MockCongestionObserver : public SendSideCongestionController::Observer {
>+class MockCongestionObserver : public NetworkChangedObserver {
>  public:
>   MOCK_METHOD4(OnNetworkChanged,
>                void(uint32_t bitrate_bps,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/include/mock/mock_send_side_congestion_controller.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/include/mock/mock_send_side_congestion_controller.h
>deleted file mode 100644
>index 6ec7bb1b6e5..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/include/mock/mock_send_side_congestion_controller.h
>+++ /dev/null
>@@ -1,39 +0,0 @@
>-/*
>- *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MODULES_CONGESTION_CONTROLLER_INCLUDE_MOCK_MOCK_SEND_SIDE_CONGESTION_CONTROLLER_H_
>-#define MODULES_CONGESTION_CONTROLLER_INCLUDE_MOCK_MOCK_SEND_SIDE_CONGESTION_CONTROLLER_H_
>-
>-#include "modules/congestion_controller/include/send_side_congestion_controller.h"
>-#include "test/gmock.h"
>-
>-namespace webrtc {
>-namespace test {
>-
>-class MockSendSideCongestionController : public SendSideCongestionController {
>- public:
>-  MockSendSideCongestionController(const Clock* clock,
>-                                   RtcEventLog* event_log,
>-                                   PacedSender* paced_sender)
>-      : SendSideCongestionController(clock,
>-                                     nullptr /* observer */,
>-                                     event_log,
>-                                     paced_sender) {}
>-
>-  MOCK_METHOD3(SetBweBitrates,
>-               void(int min_bitrate_bps,
>-                    int start_bitrate_bps,
>-                    int max_bitrate_bps));
>-};
>-
>-}  // namespace test
>-}  // namespace webrtc
>-
>-#endif  // MODULES_CONGESTION_CONTROLLER_INCLUDE_MOCK_MOCK_SEND_SIDE_CONGESTION_CONTROLLER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/include/network_changed_observer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/include/network_changed_observer.h
>new file mode 100644
>index 00000000000..34ac32aed15
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/include/network_changed_observer.h
>@@ -0,0 +1,41 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_CONGESTION_CONTROLLER_INCLUDE_NETWORK_CHANGED_OBSERVER_H_
>+#define MODULES_CONGESTION_CONTROLLER_INCLUDE_NETWORK_CHANGED_OBSERVER_H_
>+
>+#include <stdint.h>
>+
>+namespace webrtc {
>+
>+// Note: This interface will be deprecated in favor of the
>+// TargetTransferRateObserver interface. The new interface provides more
>+// information about the network connection and uses structs to make it easier
>+// to add fields.
>+
>+// Observer class for bitrate changes announced due to change in bandwidth
>+// estimate or due to that the send pacer is full. Fraction loss and rtt is
>+// also part of this callback to allow the observer to optimize its settings
>+// for different types of network environments. The bitrate does not include
>+// packet headers and is measured in bits per second.
>+// TODO(srte): Deprecate and remove this class when SendSideCongestionController
>+// is no longer using this as part of our public API.
>+class NetworkChangedObserver {
>+ public:
>+  virtual void OnNetworkChanged(uint32_t bitrate_bps,
>+                                uint8_t fraction_loss,  // 0 - 255.
>+                                int64_t rtt_ms,
>+                                int64_t probing_interval_ms) = 0;
>+
>+ protected:
>+  virtual ~NetworkChangedObserver() {}
>+};
>+}  // namespace webrtc
>+#endif  // MODULES_CONGESTION_CONTROLLER_INCLUDE_NETWORK_CHANGED_OBSERVER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/include/receive_side_congestion_controller.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/include/receive_side_congestion_controller.h
>index a4b593ae88b..ef3c8b16ebc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/include/receive_side_congestion_controller.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/include/receive_side_congestion_controller.h
>@@ -30,9 +30,8 @@ class RemoteBitrateObserver;
> class ReceiveSideCongestionController : public CallStatsObserver,
>                                         public Module {
>  public:
>-  ReceiveSideCongestionController(
>-      const Clock* clock,
>-      PacketRouter* packet_router);
>+  ReceiveSideCongestionController(const Clock* clock,
>+                                  PacketRouter* packet_router);
> 
>   virtual ~ReceiveSideCongestionController() {}
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/include/send_side_congestion_controller.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/include/send_side_congestion_controller.h
>index b33fad0d305..9d4f8526868 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/include/send_side_congestion_controller.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/include/send_side_congestion_controller.h
>@@ -15,7 +15,9 @@
> #include <vector>
> 
> #include "common_types.h"  // NOLINT(build/include)
>-#include "modules/congestion_controller/delay_based_bwe.h"
>+#include "modules/congestion_controller/goog_cc/delay_based_bwe.h"
>+#include "modules/congestion_controller/include/network_changed_observer.h"
>+#include "modules/congestion_controller/include/send_side_congestion_controller_interface.h"
> #include "modules/congestion_controller/transport_feedback_adapter.h"
> #include "modules/include/module.h"
> #include "modules/include/module_common_types.h"
>@@ -37,34 +39,22 @@ class AcknowledgedBitrateEstimator;
> class ProbeController;
> class RateLimiter;
> class RtcEventLog;
>+class CongestionWindowPushbackController;
> 
>-class SendSideCongestionController : public CallStatsObserver,
>-                                     public Module,
>-                                     public TransportFeedbackObserver {
>+class SendSideCongestionController
>+    : public SendSideCongestionControllerInterface {
>  public:
>-  // Observer class for bitrate changes announced due to change in bandwidth
>-  // estimate or due to that the send pacer is full. Fraction loss and rtt is
>-  // also part of this callback to allow the observer to optimize its settings
>-  // for different types of network environments. The bitrate does not include
>-  // packet headers and is measured in bits per second.
>-  class Observer {
>-   public:
>-    virtual void OnNetworkChanged(uint32_t bitrate_bps,
>-                                  uint8_t fraction_loss,  // 0 - 255.
>-                                  int64_t rtt_ms,
>-                                  int64_t probing_interval_ms) = 0;
>-
>-   protected:
>-    virtual ~Observer() {}
>-  };
>+  using Observer = NetworkChangedObserver;
>   SendSideCongestionController(const Clock* clock,
>                                Observer* observer,
>                                RtcEventLog* event_log,
>                                PacedSender* pacer);
>   ~SendSideCongestionController() override;
> 
>-  void RegisterPacketFeedbackObserver(PacketFeedbackObserver* observer);
>-  void DeRegisterPacketFeedbackObserver(PacketFeedbackObserver* observer);
>+  void RegisterPacketFeedbackObserver(
>+      PacketFeedbackObserver* observer) override;
>+  void DeRegisterPacketFeedbackObserver(
>+      PacketFeedbackObserver* observer) override;
> 
>   // Currently, there can be at most one observer.
>   // TODO(nisse): The RegisterNetworkObserver method is needed because we first
>@@ -72,35 +62,46 @@ class SendSideCongestionController : public CallStatsObserver,
>   // reference to Call, which then registers itself as the observer. We should
>   // try to break this circular chain of references, and make the observer a
>   // construction time constant.
>-  void RegisterNetworkObserver(Observer* observer);
>-  void DeRegisterNetworkObserver(Observer* observer);
>+  void RegisterNetworkObserver(Observer* observer) override;
>+  virtual void DeRegisterNetworkObserver(Observer* observer);
>+
>+  void SetBweBitrates(int min_bitrate_bps,
>+                      int start_bitrate_bps,
>+                      int max_bitrate_bps) override;
>+
>+  void SetAllocatedSendBitrateLimits(int64_t min_send_bitrate_bps,
>+                                     int64_t max_padding_bitrate_bps,
>+                                     int64_t max_total_bitrate_bps) override;
> 
>-  virtual void SetBweBitrates(int min_bitrate_bps,
>-                              int start_bitrate_bps,
>-                              int max_bitrate_bps);
>   // Resets the BWE state. Note the first argument is the bitrate_bps.
>-  virtual void OnNetworkRouteChanged(const rtc::NetworkRoute& network_route,
>-                                     int bitrate_bps,
>-                                     int min_bitrate_bps,
>-                                     int max_bitrate_bps);
>-  virtual void SignalNetworkState(NetworkState state);
>-  virtual void SetTransportOverhead(size_t transport_overhead_bytes_per_packet);
>+  void OnNetworkRouteChanged(const rtc::NetworkRoute& network_route,
>+                             int bitrate_bps,
>+                             int min_bitrate_bps,
>+                             int max_bitrate_bps) override;
>+  void SignalNetworkState(NetworkState state) override;
>+
>+  // Deprecated: Is updated by OnNetworkRouteChanged
>+  RTC_DEPRECATED virtual void SetTransportOverhead(
>+      size_t transport_overhead_bytes_per_packet);
> 
>   // Deprecated: Use GetBandwidthObserver instead.
>   RTC_DEPRECATED virtual BitrateController* GetBitrateController() const;
> 
>-  virtual RtcpBandwidthObserver* GetBandwidthObserver() const;
>+  RtcpBandwidthObserver* GetBandwidthObserver() override;
>+  RTC_DEPRECATED RtcpBandwidthObserver* GetBandwidthObserver() const;
> 
>-  virtual bool AvailableBandwidth(uint32_t* bandwidth) const;
>+  bool AvailableBandwidth(uint32_t* bandwidth) const override;
>   virtual int64_t GetPacerQueuingDelayMs() const;
>   virtual int64_t GetFirstPacketTimeMs() const;
> 
>-  virtual TransportFeedbackObserver* GetTransportFeedbackObserver();
>+  TransportFeedbackObserver* GetTransportFeedbackObserver() override;
> 
>-  RateLimiter* GetRetransmissionRateLimiter();
>-  void EnablePeriodicAlrProbing(bool enable);
>+  RTC_DEPRECATED virtual RateLimiter* GetRetransmissionRateLimiter();
> 
>-  virtual void OnSentPacket(const rtc::SentPacket& sent_packet);
>+  void SetPerPacketFeedbackAvailable(bool available) override;
>+  void EnablePeriodicAlrProbing(bool enable) override;
>+
>+  void OnSentPacket(const rtc::SentPacket& sent_packet) override;
> 
>   // Implements CallStatsObserver.
>   void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) override;
>@@ -115,7 +116,12 @@ class SendSideCongestionController : public CallStatsObserver,
>                  size_t length,
>                  const PacedPacketInfo& pacing_info) override;
>   void OnTransportFeedback(const rtcp::TransportFeedback& feedback) override;
>-  std::vector<PacketFeedback> GetTransportFeedbackVector() const override;
>+
>+  std::vector<PacketFeedback> GetTransportFeedbackVector() const;
>+
>+  void SetPacingFactor(float pacing_factor) override;
>+
>+  void SetAllocatedBitrateWithoutFeedback(uint32_t bitrate_bps) override;
> 
>  private:
>   void MaybeTriggerOnNetworkChanged();
>@@ -126,6 +132,8 @@ class SendSideCongestionController : public CallStatsObserver,
>                                            uint8_t fraction_loss,
>                                            int64_t rtt);
>   void LimitOutstandingBytes(size_t num_outstanding_bytes);
>+  void SendProbes(std::vector<ProbeClusterConfig> probe_configs)
>+      RTC_EXCLUSIVE_LOCKS_REQUIRED(&probe_lock_);
>   const Clock* const clock_;
>   rtc::CriticalSection observer_lock_;
>   Observer* observer_ RTC_GUARDED_BY(observer_lock_);
>@@ -133,7 +141,10 @@ class SendSideCongestionController : public CallStatsObserver,
>   PacedSender* const pacer_;
>   const std::unique_ptr<BitrateController> bitrate_controller_;
>   std::unique_ptr<AcknowledgedBitrateEstimator> acknowledged_bitrate_estimator_;
>-  const std::unique_ptr<ProbeController> probe_controller_;
>+  rtc::CriticalSection probe_lock_;
>+  const std::unique_ptr<ProbeController> probe_controller_
>+      RTC_GUARDED_BY(probe_lock_);
>+
>   const std::unique_ptr<RateLimiter> retransmission_rate_limiter_;
>   TransportFeedbackAdapter transport_feedback_adapter_;
>   rtc::CriticalSection network_state_lock_;
>@@ -152,12 +163,17 @@ class SendSideCongestionController : public CallStatsObserver,
>   bool in_cwnd_experiment_;
>   int64_t accepted_queue_ms_;
>   bool was_in_alr_;
>+  const bool send_side_bwe_with_overhead_;
>+  size_t transport_overhead_bytes_per_packet_ RTC_GUARDED_BY(bwe_lock_);
> 
>   rtc::RaceChecker worker_race_;
> 
>   bool pacer_pushback_experiment_ = false;
>   float encoding_rate_ = 1.0;
> 
>+  const std::unique_ptr<CongestionWindowPushbackController>
>+      congestion_window_pushback_controller_;
>+
>   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(SendSideCongestionController);
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/include/send_side_congestion_controller_interface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/include/send_side_congestion_controller_interface.h
>new file mode 100644
>index 00000000000..91866fc6488
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/include/send_side_congestion_controller_interface.h
>@@ -0,0 +1,73 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_CONGESTION_CONTROLLER_INCLUDE_SEND_SIDE_CONGESTION_CONTROLLER_INTERFACE_H_
>+#define MODULES_CONGESTION_CONTROLLER_INCLUDE_SEND_SIDE_CONGESTION_CONTROLLER_INTERFACE_H_
>+
>+#include <memory>
>+#include <vector>
>+
>+#include "modules/congestion_controller/include/network_changed_observer.h"
>+#include "modules/congestion_controller/transport_feedback_adapter.h"
>+#include "modules/include/module.h"
>+#include "modules/include/module_common_types.h"
>+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>+#include "rtc_base/constructormagic.h"
>+#include "rtc_base/networkroute.h"
>+
>+namespace rtc {
>+struct SentPacket;
>+}
>+
>+namespace webrtc {
>+
>+class BitrateController;
>+class Clock;
>+class AcknowledgedBitrateEstimator;
>+class ProbeController;
>+class RateLimiter;
>+class RtcEventLog;
>+
>+class SendSideCongestionControllerInterface : public CallStatsObserver,
>+                                              public Module,
>+                                              public TransportFeedbackObserver {
>+ public:
>+  SendSideCongestionControllerInterface() = default;
>+  ~SendSideCongestionControllerInterface() override = default;
>+  virtual void RegisterPacketFeedbackObserver(
>+      PacketFeedbackObserver* observer) = 0;
>+  virtual void DeRegisterPacketFeedbackObserver(
>+      PacketFeedbackObserver* observer) = 0;
>+  virtual void RegisterNetworkObserver(NetworkChangedObserver* observer) = 0;
>+  virtual void SetBweBitrates(int min_bitrate_bps,
>+                              int start_bitrate_bps,
>+                              int max_bitrate_bps) = 0;
>+  virtual void SetAllocatedSendBitrateLimits(int64_t min_send_bitrate_bps,
>+                                             int64_t max_padding_bitrate_bps,
>+                                             int64_t max_total_bitrate_bps) = 0;
>+  virtual void OnNetworkRouteChanged(const rtc::NetworkRoute& network_route,
>+                                     int bitrate_bps,
>+                                     int min_bitrate_bps,
>+                                     int max_bitrate_bps) = 0;
>+  virtual void SignalNetworkState(NetworkState state) = 0;
>+  virtual RtcpBandwidthObserver* GetBandwidthObserver() = 0;
>+  virtual bool AvailableBandwidth(uint32_t* bandwidth) const = 0;
>+  virtual TransportFeedbackObserver* GetTransportFeedbackObserver() = 0;
>+  virtual void SetPerPacketFeedbackAvailable(bool available) = 0;
>+  virtual void EnablePeriodicAlrProbing(bool enable) = 0;
>+  virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0;
>+  virtual void SetPacingFactor(float pacing_factor) = 0;
>+  virtual void SetAllocatedBitrateWithoutFeedback(uint32_t bitrate_bps) = 0;
>+  RTC_DISALLOW_COPY_AND_ASSIGN(SendSideCongestionControllerInterface);
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_CONGESTION_CONTROLLER_INCLUDE_SEND_SIDE_CONGESTION_CONTROLLER_INTERFACE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/probe_controller.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/probe_controller.h
>deleted file mode 100644
>index cd0e9b1b572..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/probe_controller.h
>+++ /dev/null
>@@ -1,94 +0,0 @@
>-/*
>- *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MODULES_CONGESTION_CONTROLLER_PROBE_CONTROLLER_H_
>-#define MODULES_CONGESTION_CONTROLLER_PROBE_CONTROLLER_H_
>-
>-#include <initializer_list>
>-
>-#include "common_types.h"  // NOLINT(build/include)
>-#include "modules/pacing/paced_sender.h"
>-#include "rtc_base/criticalsection.h"
>-
>-namespace webrtc {
>-
>-class Clock;
>-
>-// This class controls initiation of probing to estimate initial channel
>-// capacity. There is also support for probing during a session when max
>-// bitrate is adjusted by an application.
>-class ProbeController {
>- public:
>-  ProbeController(PacedSender* pacer, const Clock* clock);
>-
>-  void SetBitrates(int64_t min_bitrate_bps,
>-                   int64_t start_bitrate_bps,
>-                   int64_t max_bitrate_bps);
>-
>-  void OnNetworkStateChanged(NetworkState state);
>-
>-  void SetEstimatedBitrate(int64_t bitrate_bps);
>-
>-  void EnablePeriodicAlrProbing(bool enable);
>-
>-  void SetAlrEndedTimeMs(int64_t alr_end_time);
>-
>-  void RequestProbe();
>-
>-  // Resets the ProbeController to a state equivalent to as if it was just
>-  // created EXCEPT for |enable_periodic_alr_probing_|.
>-  void Reset();
>-
>-  void Process();
>-
>- private:
>-  enum class State {
>-    // Initial state where no probing has been triggered yet.
>-    kInit,
>-    // Waiting for probing results to continue further probing.
>-    kWaitingForProbingResult,
>-    // Probing is complete.
>-    kProbingComplete,
>-  };
>-
>-  void InitiateExponentialProbing() RTC_EXCLUSIVE_LOCKS_REQUIRED(critsect_);
>-  void InitiateProbing(int64_t now_ms,
>-                       std::initializer_list<int64_t> bitrates_to_probe,
>-                       bool probe_further)
>-      RTC_EXCLUSIVE_LOCKS_REQUIRED(critsect_);
>-
>-  rtc::CriticalSection critsect_;
>-  PacedSender* const pacer_;
>-  const Clock* const clock_;
>-  NetworkState network_state_ RTC_GUARDED_BY(critsect_);
>-  State state_ RTC_GUARDED_BY(critsect_);
>-  int64_t min_bitrate_to_probe_further_bps_ RTC_GUARDED_BY(critsect_);
>-  int64_t time_last_probing_initiated_ms_ RTC_GUARDED_BY(critsect_);
>-  int64_t estimated_bitrate_bps_ RTC_GUARDED_BY(critsect_);
>-  int64_t start_bitrate_bps_ RTC_GUARDED_BY(critsect_);
>-  int64_t max_bitrate_bps_ RTC_GUARDED_BY(critsect_);
>-  int64_t last_bwe_drop_probing_time_ms_ RTC_GUARDED_BY(critsect_);
>-  rtc::Optional<int64_t> alr_end_time_ms_ RTC_GUARDED_BY(critsect_);
>-  bool enable_periodic_alr_probing_ RTC_GUARDED_BY(critsect_);
>-  int64_t time_of_last_large_drop_ms_ RTC_GUARDED_BY(critsect_);
>-  int64_t bitrate_before_last_large_drop_bps_ RTC_GUARDED_BY(critsect_);
>-
>-  bool in_rapid_recovery_experiment_ RTC_GUARDED_BY(critsect_);
>-  // For WebRTC.BWE.MidCallProbing.* metric.
>-  bool mid_call_probing_waiting_for_result_ RTC_GUARDED_BY(&critsect_);
>-  int64_t mid_call_probing_bitrate_bps_ RTC_GUARDED_BY(&critsect_);
>-  int64_t mid_call_probing_succcess_threshold_ RTC_GUARDED_BY(&critsect_);
>-
>-  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(ProbeController);
>-};
>-
>-}  // namespace webrtc
>-
>-#endif  // MODULES_CONGESTION_CONTROLLER_PROBE_CONTROLLER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/probe_controller_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/probe_controller_unittest.cc
>deleted file mode 100644
>index 3477c87a26f..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/probe_controller_unittest.cc
>+++ /dev/null
>@@ -1,270 +0,0 @@
>-/*
>- *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-#include <memory>
>-
>-#include "modules/congestion_controller/probe_controller.h"
>-#include "modules/pacing/mock/mock_paced_sender.h"
>-#include "rtc_base/logging.h"
>-#include "system_wrappers/include/clock.h"
>-#include "test/gmock.h"
>-#include "test/gtest.h"
>-
>-using testing::_;
>-using testing::AtLeast;
>-using testing::NiceMock;
>-using testing::Return;
>-
>-namespace webrtc {
>-namespace test {
>-
>-namespace {
>-
>-constexpr int kMinBitrateBps = 100;
>-constexpr int kStartBitrateBps = 300;
>-constexpr int kMaxBitrateBps = 10000;
>-
>-constexpr int kExponentialProbingTimeoutMs = 5000;
>-
>-constexpr int kAlrProbeInterval = 5000;
>-constexpr int kAlrEndedTimeoutMs = 3000;
>-constexpr int kBitrateDropTimeoutMs = 5000;
>-
>-}  // namespace
>-
>-class ProbeControllerTest : public ::testing::Test {
>- protected:
>-  ProbeControllerTest() : clock_(100000000L) {
>-    probe_controller_.reset(new ProbeController(&pacer_, &clock_));
>-  }
>-  ~ProbeControllerTest() override {}
>-
>-  SimulatedClock clock_;
>-  NiceMock<MockPacedSender> pacer_;
>-  std::unique_ptr<ProbeController> probe_controller_;
>-};
>-
>-TEST_F(ProbeControllerTest, InitiatesProbingAtStart) {
>-  EXPECT_CALL(pacer_, CreateProbeCluster(_)).Times(AtLeast(2));
>-  probe_controller_->SetBitrates(kMinBitrateBps, kStartBitrateBps,
>-                                 kMaxBitrateBps);
>-}
>-
>-TEST_F(ProbeControllerTest, ProbeOnlyWhenNetworkIsUp) {
>-  probe_controller_->OnNetworkStateChanged(kNetworkDown);
>-  EXPECT_CALL(pacer_, CreateProbeCluster(_)).Times(0);
>-  probe_controller_->SetBitrates(kMinBitrateBps, kStartBitrateBps,
>-                                 kMaxBitrateBps);
>-
>-  testing::Mock::VerifyAndClearExpectations(&pacer_);
>-  EXPECT_CALL(pacer_, CreateProbeCluster(_)).Times(AtLeast(2));
>-  probe_controller_->OnNetworkStateChanged(kNetworkUp);
>-}
>-
>-TEST_F(ProbeControllerTest, InitiatesProbingOnMaxBitrateIncrease) {
>-  EXPECT_CALL(pacer_, CreateProbeCluster(_)).Times(AtLeast(2));
>-  probe_controller_->SetBitrates(kMinBitrateBps, kStartBitrateBps,
>-                                 kMaxBitrateBps);
>-  // Long enough to time out exponential probing.
>-  clock_.AdvanceTimeMilliseconds(kExponentialProbingTimeoutMs);
>-  probe_controller_->SetEstimatedBitrate(kStartBitrateBps);
>-  probe_controller_->Process();
>-
>-  EXPECT_CALL(pacer_, CreateProbeCluster(kMaxBitrateBps + 100));
>-  probe_controller_->SetBitrates(kMinBitrateBps, kStartBitrateBps,
>-                                 kMaxBitrateBps + 100);
>-}
>-
>-TEST_F(ProbeControllerTest, InitiatesProbingOnMaxBitrateIncreaseAtMaxBitrate) {
>-  EXPECT_CALL(pacer_, CreateProbeCluster(_)).Times(AtLeast(2));
>-  probe_controller_->SetBitrates(kMinBitrateBps, kStartBitrateBps,
>-                                 kMaxBitrateBps);
>-  // Long enough to time out exponential probing.
>-  clock_.AdvanceTimeMilliseconds(kExponentialProbingTimeoutMs);
>-  probe_controller_->SetEstimatedBitrate(kStartBitrateBps);
>-  probe_controller_->Process();
>-
>-  probe_controller_->SetEstimatedBitrate(kMaxBitrateBps);
>-  EXPECT_CALL(pacer_, CreateProbeCluster(kMaxBitrateBps + 100));
>-  probe_controller_->SetBitrates(kMinBitrateBps, kStartBitrateBps,
>-                                 kMaxBitrateBps + 100);
>-}
>-
>-TEST_F(ProbeControllerTest, TestExponentialProbing) {
>-  probe_controller_->SetBitrates(kMinBitrateBps, kStartBitrateBps,
>-                                 kMaxBitrateBps);
>-
>-  // Repeated probe should only be sent when estimated bitrate climbs above
>-  // 0.7 * 6 * kStartBitrateBps = 1260.
>-  EXPECT_CALL(pacer_, CreateProbeCluster(_)).Times(0);
>-  probe_controller_->SetEstimatedBitrate(1000);
>-  testing::Mock::VerifyAndClearExpectations(&pacer_);
>-
>-  EXPECT_CALL(pacer_, CreateProbeCluster(2 * 1800));
>-  probe_controller_->SetEstimatedBitrate(1800);
>-}
>-
>-TEST_F(ProbeControllerTest, TestExponentialProbingTimeout) {
>-  probe_controller_->SetBitrates(kMinBitrateBps, kStartBitrateBps,
>-                                 kMaxBitrateBps);
>-
>-  // Advance far enough to cause a time out in waiting for probing result.
>-  clock_.AdvanceTimeMilliseconds(kExponentialProbingTimeoutMs);
>-  probe_controller_->Process();
>-
>-  EXPECT_CALL(pacer_, CreateProbeCluster(_)).Times(0);
>-  probe_controller_->SetEstimatedBitrate(1800);
>-}
>-
>-TEST_F(ProbeControllerTest, RequestProbeInAlr) {
>-  EXPECT_CALL(pacer_, CreateProbeCluster(_)).Times(2);
>-  probe_controller_->SetBitrates(kMinBitrateBps, kStartBitrateBps,
>-                                 kMaxBitrateBps);
>-  probe_controller_->SetEstimatedBitrate(500);
>-  testing::Mock::VerifyAndClearExpectations(&pacer_);
>-  EXPECT_CALL(pacer_, CreateProbeCluster(0.85 * 500)).Times(1);
>-  EXPECT_CALL(pacer_, GetApplicationLimitedRegionStartTime())
>-      .WillRepeatedly(Return(clock_.TimeInMilliseconds()));
>-  clock_.AdvanceTimeMilliseconds(kAlrProbeInterval + 1);
>-  probe_controller_->Process();
>-  probe_controller_->SetEstimatedBitrate(250);
>-  probe_controller_->RequestProbe();
>-}
>-
>-TEST_F(ProbeControllerTest, RequestProbeWhenAlrEndedRecently) {
>-  EXPECT_CALL(pacer_, CreateProbeCluster(_)).Times(2);
>-  probe_controller_->SetBitrates(kMinBitrateBps, kStartBitrateBps,
>-                                 kMaxBitrateBps);
>-  probe_controller_->SetEstimatedBitrate(500);
>-  testing::Mock::VerifyAndClearExpectations(&pacer_);
>-  EXPECT_CALL(pacer_, CreateProbeCluster(0.85 * 500)).Times(1);
>-  EXPECT_CALL(pacer_, GetApplicationLimitedRegionStartTime())
>-      .WillRepeatedly(Return(rtc::nullopt));
>-  clock_.AdvanceTimeMilliseconds(kAlrProbeInterval + 1);
>-  probe_controller_->Process();
>-  probe_controller_->SetEstimatedBitrate(250);
>-  probe_controller_->SetAlrEndedTimeMs(clock_.TimeInMilliseconds());
>-  clock_.AdvanceTimeMilliseconds(kAlrEndedTimeoutMs - 1);
>-  probe_controller_->RequestProbe();
>-}
>-
>-TEST_F(ProbeControllerTest, RequestProbeWhenAlrNotEndedRecently) {
>-  EXPECT_CALL(pacer_, CreateProbeCluster(_)).Times(2);
>-  probe_controller_->SetBitrates(kMinBitrateBps, kStartBitrateBps,
>-                                 kMaxBitrateBps);
>-  probe_controller_->SetEstimatedBitrate(500);
>-  testing::Mock::VerifyAndClearExpectations(&pacer_);
>-  EXPECT_CALL(pacer_, CreateProbeCluster(_)).Times(0);
>-  EXPECT_CALL(pacer_, GetApplicationLimitedRegionStartTime())
>-      .WillRepeatedly(Return(rtc::nullopt));
>-  clock_.AdvanceTimeMilliseconds(kAlrProbeInterval + 1);
>-  probe_controller_->Process();
>-  probe_controller_->SetEstimatedBitrate(250);
>-  probe_controller_->SetAlrEndedTimeMs(clock_.TimeInMilliseconds());
>-  clock_.AdvanceTimeMilliseconds(kAlrEndedTimeoutMs + 1);
>-  probe_controller_->RequestProbe();
>-}
>-
>-TEST_F(ProbeControllerTest, RequestProbeWhenBweDropNotRecent) {
>-  EXPECT_CALL(pacer_, CreateProbeCluster(_)).Times(2);
>-  probe_controller_->SetBitrates(kMinBitrateBps, kStartBitrateBps,
>-                                 kMaxBitrateBps);
>-  probe_controller_->SetEstimatedBitrate(500);
>-  testing::Mock::VerifyAndClearExpectations(&pacer_);
>-  EXPECT_CALL(pacer_, CreateProbeCluster(_)).Times(0);
>-  EXPECT_CALL(pacer_, GetApplicationLimitedRegionStartTime())
>-      .WillRepeatedly(Return(clock_.TimeInMilliseconds()));
>-  clock_.AdvanceTimeMilliseconds(kAlrProbeInterval + 1);
>-  probe_controller_->Process();
>-  probe_controller_->SetEstimatedBitrate(250);
>-  clock_.AdvanceTimeMilliseconds(kBitrateDropTimeoutMs + 1);
>-  probe_controller_->RequestProbe();
>-}
>-
>-TEST_F(ProbeControllerTest, PeriodicProbing) {
>-  EXPECT_CALL(pacer_, CreateProbeCluster(_)).Times(2);
>-  probe_controller_->EnablePeriodicAlrProbing(true);
>-  probe_controller_->SetBitrates(kMinBitrateBps, kStartBitrateBps,
>-                                 kMaxBitrateBps);
>-  probe_controller_->SetEstimatedBitrate(500);
>-  testing::Mock::VerifyAndClearExpectations(&pacer_);
>-
>-  int64_t start_time = clock_.TimeInMilliseconds();
>-
>-  // Expect the controller to send a new probe after 5s has passed.
>-  EXPECT_CALL(pacer_, CreateProbeCluster(1000)).Times(1);
>-  EXPECT_CALL(pacer_, GetApplicationLimitedRegionStartTime())
>-      .WillRepeatedly(Return(start_time));
>-  clock_.AdvanceTimeMilliseconds(5000);
>-  probe_controller_->Process();
>-  probe_controller_->SetEstimatedBitrate(500);
>-  testing::Mock::VerifyAndClearExpectations(&pacer_);
>-
>-  // The following probe should be sent at 10s into ALR.
>-  EXPECT_CALL(pacer_, CreateProbeCluster(_)).Times(0);
>-  EXPECT_CALL(pacer_, GetApplicationLimitedRegionStartTime())
>-      .WillRepeatedly(Return(start_time));
>-  clock_.AdvanceTimeMilliseconds(4000);
>-  probe_controller_->Process();
>-  probe_controller_->SetEstimatedBitrate(500);
>-  testing::Mock::VerifyAndClearExpectations(&pacer_);
>-
>-  EXPECT_CALL(pacer_, CreateProbeCluster(_)).Times(1);
>-  EXPECT_CALL(pacer_, GetApplicationLimitedRegionStartTime())
>-      .WillRepeatedly(Return(start_time));
>-  clock_.AdvanceTimeMilliseconds(1000);
>-  probe_controller_->Process();
>-  probe_controller_->SetEstimatedBitrate(500);
>-  testing::Mock::VerifyAndClearExpectations(&pacer_);
>-}
>-
>-TEST_F(ProbeControllerTest, PeriodicProbingAfterReset) {
>-  testing::StrictMock<MockPacedSender> local_pacer;
>-  probe_controller_.reset(new ProbeController(&local_pacer, &clock_));
>-  int64_t alr_start_time = clock_.TimeInMilliseconds();
>-  EXPECT_CALL(local_pacer, GetApplicationLimitedRegionStartTime())
>-      .WillRepeatedly(Return(alr_start_time));
>-
>-  EXPECT_CALL(local_pacer, CreateProbeCluster(_)).Times(2);
>-  probe_controller_->EnablePeriodicAlrProbing(true);
>-  probe_controller_->SetBitrates(kMinBitrateBps, kStartBitrateBps,
>-                                 kMaxBitrateBps);
>-  probe_controller_->Reset();
>-
>-  clock_.AdvanceTimeMilliseconds(10000);
>-  probe_controller_->Process();
>-
>-  EXPECT_CALL(local_pacer, CreateProbeCluster(_)).Times(2);
>-  probe_controller_->SetBitrates(kMinBitrateBps, kStartBitrateBps,
>-                                 kMaxBitrateBps);
>-
>-  // Make sure we use |kStartBitrateBps| as the estimated bitrate
>-  // until SetEstimatedBitrate is called with an updated estimate.
>-  clock_.AdvanceTimeMilliseconds(10000);
>-  EXPECT_CALL(local_pacer, CreateProbeCluster(kStartBitrateBps*2));
>-  probe_controller_->Process();
>-}
>-
>-TEST_F(ProbeControllerTest, TestExponentialProbingOverflow) {
>-  const int64_t kMbpsMultiplier = 1000000;
>-  probe_controller_->SetBitrates(kMinBitrateBps, 10 * kMbpsMultiplier,
>-                                 100 * kMbpsMultiplier);
>-
>-  // Verify that probe bitrate is capped at the specified max bitrate
>-  EXPECT_CALL(pacer_, CreateProbeCluster(100 * kMbpsMultiplier));
>-  probe_controller_->SetEstimatedBitrate(60 * kMbpsMultiplier);
>-  testing::Mock::VerifyAndClearExpectations(&pacer_);
>-
>-  // Verify that repeated probes aren't sent.
>-  EXPECT_CALL(pacer_, CreateProbeCluster(_)).Times(0);
>-  probe_controller_->SetEstimatedBitrate(100 * kMbpsMultiplier);
>-}
>-
>-}  // namespace test
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/BUILD.gn
>new file mode 100644
>index 00000000000..ddfe8139acf
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/BUILD.gn
>@@ -0,0 +1,117 @@
>+# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
>+#
>+# Use of this source code is governed by a BSD-style license
>+# that can be found in the LICENSE file in the root of the source
>+# tree. An additional intellectual property rights grant can be found
>+# in the file PATENTS.  All contributing project authors may
>+# be found in the AUTHORS file in the root of the source tree.
>+
>+import("../../../webrtc.gni")
>+
>+config("bwe_test_logging") {
>+  if (rtc_enable_bwe_test_logging) {
>+    defines = [ "BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=1" ]
>+  } else {
>+    defines = [ "BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=0" ]
>+  }
>+}
>+
>+rtc_static_library("congestion_controller") {
>+  visibility = [ "*" ]
>+  configs += [ ":bwe_test_logging" ]
>+  sources = [
>+    "include/send_side_congestion_controller.h",
>+    "pacer_controller.cc",
>+    "pacer_controller.h",
>+    "send_side_congestion_controller.cc",
>+  ]
>+
>+  if (!build_with_chromium && is_clang) {
>+    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>+    suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>+  }
>+
>+  deps = [
>+    ":transport_feedback",
>+    "../:congestion_controller",
>+    "../..:module_api",
>+    "../../..:webrtc_common",
>+    "../../../api/transport:network_control",
>+    "../../../rtc_base:checks",
>+    "../../../rtc_base:rate_limiter",
>+    "../../../rtc_base:rtc_task_queue_api",
>+    "../../../rtc_base:safe_minmax",
>+    "../../../rtc_base:sequenced_task_checker",
>+    "../../../rtc_base/experiments:congestion_controller_experiment",
>+    "../../../system_wrappers",
>+    "../../../system_wrappers:field_trial_api",
>+    "../../../system_wrappers:metrics_api",
>+    "../../../system_wrappers:runtime_enabled_features_api",
>+    "../../pacing",
>+    "../../remote_bitrate_estimator",
>+    "../../rtp_rtcp:rtp_rtcp_format",
>+    "../goog_cc",
>+    "//third_party/abseil-cpp/absl/memory",
>+  ]
>+
>+  if (!build_with_mozilla) {
>+    deps += [ "../../../rtc_base:rtc_base" ]
>+  }
>+}
>+
>+rtc_static_library("transport_feedback") {
>+  visibility = [ "*" ]
>+  sources = [
>+    "send_time_history.cc",
>+    "send_time_history.h",
>+    "transport_feedback_adapter.cc",
>+    "transport_feedback_adapter.h",
>+  ]
>+
>+  deps = [
>+    "../..:module_api",
>+    "../../../rtc_base:checks",
>+    "../../../rtc_base:rtc_base_approved",
>+    "../../../system_wrappers",
>+    "../../rtp_rtcp:rtp_rtcp_format",
>+  ]
>+}
>+
>+if (rtc_include_tests) {
>+  rtc_source_set("congestion_controller_unittests") {
>+    testonly = true
>+
>+    sources = [
>+      "congestion_controller_unittests_helper.cc",
>+      "congestion_controller_unittests_helper.h",
>+      "send_side_congestion_controller_unittest.cc",
>+      "send_time_history_unittest.cc",
>+      "transport_feedback_adapter_unittest.cc",
>+    ]
>+    deps = [
>+      ":congestion_controller",
>+      ":transport_feedback",
>+      "../:congestion_controller",
>+      "../:mock_congestion_controller",
>+      "../../../api/transport:network_control",
>+      "../../../logging:mocks",
>+      "../../../rtc_base:checks",
>+      "../../../rtc_base:rtc_base",
>+      "../../../rtc_base:rtc_base_approved",
>+      "../../../rtc_base:rtc_base_tests_utils",
>+      "../../../system_wrappers",
>+      "../../../test:field_trial",
>+      "../../../test:test_support",
>+      "../../bitrate_controller:mocks",
>+      "../../pacing:mock_paced_sender",
>+      "../../pacing:pacing",
>+      "../../remote_bitrate_estimator:remote_bitrate_estimator",
>+      "../../rtp_rtcp:rtp_rtcp_format",
>+      "//testing/gmock",
>+    ]
>+    if (!build_with_chromium && is_clang) {
>+      # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>+      suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>+    }
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/congestion_controller_unittests_helper.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/congestion_controller_unittests_helper.cc
>similarity index 95%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/congestion_controller_unittests_helper.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/congestion_controller_unittests_helper.cc
>index 2ec54171c88..f2b4467dd7f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/congestion_controller_unittests_helper.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/congestion_controller_unittests_helper.cc
>@@ -8,7 +8,7 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "modules/congestion_controller/congestion_controller_unittests_helper.h"
>+#include "modules/congestion_controller/rtp/congestion_controller_unittests_helper.h"
> 
> #include "rtc_base/checks.h"
> #include "test/gtest.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/congestion_controller_unittests_helper.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/congestion_controller_unittests_helper.h
>similarity index 72%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/congestion_controller_unittests_helper.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/congestion_controller_unittests_helper.h
>index 37e3870d3eb..98dfb3ddce7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/congestion_controller_unittests_helper.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/congestion_controller_unittests_helper.h
>@@ -8,8 +8,8 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#ifndef MODULES_CONGESTION_CONTROLLER_CONGESTION_CONTROLLER_UNITTESTS_HELPER_H_
>-#define MODULES_CONGESTION_CONTROLLER_CONGESTION_CONTROLLER_UNITTESTS_HELPER_H_
>+#ifndef MODULES_CONGESTION_CONTROLLER_RTP_CONGESTION_CONTROLLER_UNITTESTS_HELPER_H_
>+#define MODULES_CONGESTION_CONTROLLER_RTP_CONGESTION_CONTROLLER_UNITTESTS_HELPER_H_
> 
> #include <vector>
> 
>@@ -20,4 +20,4 @@ void ComparePacketFeedbackVectors(const std::vector<PacketFeedback>& truth,
>                                   const std::vector<PacketFeedback>& input);
> }  // namespace webrtc
> 
>-#endif  // MODULES_CONGESTION_CONTROLLER_CONGESTION_CONTROLLER_UNITTESTS_HELPER_H_
>+#endif  // MODULES_CONGESTION_CONTROLLER_RTP_CONGESTION_CONTROLLER_UNITTESTS_HELPER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/include/send_side_congestion_controller.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/include/send_side_congestion_controller.h
>new file mode 100644
>index 00000000000..9ab19ae4471
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/include/send_side_congestion_controller.h
>@@ -0,0 +1,226 @@
>+/*
>+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_CONGESTION_CONTROLLER_RTP_INCLUDE_SEND_SIDE_CONGESTION_CONTROLLER_H_
>+#define MODULES_CONGESTION_CONTROLLER_RTP_INCLUDE_SEND_SIDE_CONGESTION_CONTROLLER_H_
>+
>+#include <atomic>
>+#include <functional>
>+#include <map>
>+#include <memory>
>+#include <vector>
>+
>+#include "api/transport/network_control.h"
>+#include "api/transport/network_types.h"
>+#include "common_types.h"  // NOLINT(build/include)
>+#include "modules/congestion_controller/include/send_side_congestion_controller_interface.h"
>+#include "modules/congestion_controller/rtp/pacer_controller.h"
>+#include "modules/congestion_controller/rtp/transport_feedback_adapter.h"
>+#include "modules/include/module.h"
>+#include "modules/include/module_common_types.h"
>+#include "modules/pacing/paced_sender.h"
>+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>+#include "rtc_base/constructormagic.h"
>+#include "rtc_base/criticalsection.h"
>+#include "rtc_base/networkroute.h"
>+#include "rtc_base/race_checker.h"
>+#include "rtc_base/task_queue.h"
>+
>+namespace rtc {
>+struct SentPacket;
>+}
>+
>+namespace webrtc {
>+
>+class Clock;
>+class RateLimiter;
>+class RtcEventLog;
>+
>+namespace webrtc_cc {
>+
>+namespace send_side_cc_internal {
>+// This is used to observe the network controller state and route calls to
>+// the proper handler. It also keeps cached values for safe asynchronous use.
>+// This makes sure that things running on the worker queue can't access state
>+// in SendSideCongestionController, which would risk causing data race on
>+// destruction unless members are properly ordered.
>+class ControlHandler;
>+
>+// TODO(srte): Make sure the PeriodicTask implementation is reusable and move it
>+// to task_queue.h.
>+class PeriodicTask : public rtc::QueuedTask {
>+ public:
>+  virtual void Stop() = 0;
>+};
>+}  // namespace send_side_cc_internal
>+
>+class SendSideCongestionController
>+    : public SendSideCongestionControllerInterface,
>+      public RtcpBandwidthObserver {
>+ public:
>+  SendSideCongestionController(
>+      const Clock* clock,
>+      rtc::TaskQueue* task_queue,
>+      RtcEventLog* event_log,
>+      PacedSender* pacer,
>+      int start_bitrate_bps,
>+      int min_bitrate_bps,
>+      int max_bitrate_bps,
>+      NetworkControllerFactoryInterface* controller_factory);
>+
>+  ~SendSideCongestionController() override;
>+
>+  void RegisterPacketFeedbackObserver(
>+      PacketFeedbackObserver* observer) override;
>+  void DeRegisterPacketFeedbackObserver(
>+      PacketFeedbackObserver* observer) override;
>+
>+  // Currently, there can be at most one observer.
>+  // TODO(nisse): The RegisterNetworkObserver method is needed because we first
>+  // construct this object (as part of RtpTransportControllerSend), then pass a
>+  // reference to Call, which then registers itself as the observer. We should
>+  // try to break this circular chain of references, and make the observer a
>+  // construction time constant.
>+  void RegisterNetworkObserver(NetworkChangedObserver* observer) override;
>+
>+  void SetBweBitrates(int min_bitrate_bps,
>+                      int start_bitrate_bps,
>+                      int max_bitrate_bps) override;
>+
>+  void SetAllocatedSendBitrateLimits(int64_t min_send_bitrate_bps,
>+                                     int64_t max_padding_bitrate_bps,
>+                                     int64_t max_total_bitrate_bps) override;
>+
>+  // Resets the BWE state. Note the first argument is the bitrate_bps.
>+  void OnNetworkRouteChanged(const rtc::NetworkRoute& network_route,
>+                             int bitrate_bps,
>+                             int min_bitrate_bps,
>+                             int max_bitrate_bps) override;
>+  void SignalNetworkState(NetworkState state) override;
>+
>+  RtcpBandwidthObserver* GetBandwidthObserver() override;
>+
>+  bool AvailableBandwidth(uint32_t* bandwidth) const override;
>+
>+  TransportFeedbackObserver* GetTransportFeedbackObserver() override;
>+
>+  void SetPerPacketFeedbackAvailable(bool available) override;
>+  void EnablePeriodicAlrProbing(bool enable) override;
>+
>+  void OnSentPacket(const rtc::SentPacket& sent_packet) override;
>+
>+  // Implements RtcpBandwidthObserver
>+  void OnReceivedEstimatedBitrate(uint32_t bitrate) override;
>+  void OnReceivedRtcpReceiverReport(const ReportBlockList& report_blocks,
>+                                    int64_t rtt,
>+                                    int64_t now_ms) override;
>+  // Implements CallStatsObserver.
>+  void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) override;
>+
>+  // Implements Module.
>+  int64_t TimeUntilNextProcess() override;
>+  void Process() override;
>+
>+  // Implements TransportFeedbackObserver.
>+  void AddPacket(uint32_t ssrc,
>+                 uint16_t sequence_number,
>+                 size_t length,
>+                 const PacedPacketInfo& pacing_info) override;
>+  void OnTransportFeedback(const rtcp::TransportFeedback& feedback) override;
>+
>+  std::vector<PacketFeedback> GetTransportFeedbackVector() const;
>+
>+  void SetPacingFactor(float pacing_factor) override;
>+
>+  void SetAllocatedBitrateWithoutFeedback(uint32_t bitrate_bps) override;
>+
>+ protected:
>+  // TODO(srte): The tests should be rewritten to not depend on internals and
>+  // these functions should be removed.
>+  // Since we can't control the timing of the internal task queue, this method
>+  // is used in unit tests to stop the periodic tasks from running unless
>+  // PostPeriodicTasksForTest is called.
>+  void DisablePeriodicTasks();
>+  // Post periodic tasks just once. This allows unit tests to trigger process
>+  // updates immediately.
>+  void PostPeriodicTasksForTest();
>+  // Waits for outstanding tasks to be finished. This allos unit tests to ensure
>+  // that expected callbacks has been called.
>+  void WaitOnTasksForTest();
>+
>+ private:
>+  void MaybeCreateControllers() RTC_RUN_ON(task_queue_);
>+  void MaybeRecreateControllers() RTC_RUN_ON(task_queue_);
>+
>+  void StartProcessPeriodicTasks() RTC_RUN_ON(task_queue_);
>+  void UpdateControllerWithTimeInterval() RTC_RUN_ON(task_queue_);
>+  void UpdatePacerQueue() RTC_RUN_ON(task_queue_);
>+
>+  void UpdateStreamsConfig() RTC_RUN_ON(task_queue_);
>+  void MaybeUpdateOutstandingData();
>+  void OnReceivedRtcpReceiverReportBlocks(const ReportBlockList& report_blocks,
>+                                          int64_t now_ms)
>+      RTC_RUN_ON(task_queue_);
>+
>+  const Clock* const clock_;
>+  // PacedSender is thread safe and doesn't need protection here.
>+  PacedSender* const pacer_;
>+  // TODO(srte): Move all access to feedback adapter to task queue.
>+  TransportFeedbackAdapter transport_feedback_adapter_;
>+
>+  NetworkControllerFactoryInterface* const controller_factory_with_feedback_
>+      RTC_GUARDED_BY(task_queue_);
>+  const std::unique_ptr<NetworkControllerFactoryInterface>
>+      controller_factory_fallback_ RTC_GUARDED_BY(task_queue_);
>+
>+  const std::unique_ptr<PacerController> pacer_controller_
>+      RTC_GUARDED_BY(task_queue_);
>+
>+  std::unique_ptr<send_side_cc_internal::ControlHandler> control_handler_
>+      RTC_GUARDED_BY(task_queue_);
>+
>+  std::unique_ptr<NetworkControllerInterface> controller_
>+      RTC_GUARDED_BY(task_queue_);
>+
>+  TimeDelta process_interval_ RTC_GUARDED_BY(task_queue_);
>+
>+  std::map<uint32_t, RTCPReportBlock> last_report_blocks_
>+      RTC_GUARDED_BY(task_queue_);
>+  Timestamp last_report_block_time_ RTC_GUARDED_BY(task_queue_);
>+
>+  NetworkChangedObserver* observer_ RTC_GUARDED_BY(task_queue_);
>+  NetworkControllerConfig initial_config_ RTC_GUARDED_BY(task_queue_);
>+  StreamsConfig streams_config_ RTC_GUARDED_BY(task_queue_);
>+
>+  const bool send_side_bwe_with_overhead_;
>+  // Transport overhead is written by OnNetworkRouteChanged and read by
>+  // AddPacket.
>+  // TODO(srte): Remove atomic when feedback adapter runs on task queue.
>+  std::atomic<size_t> transport_overhead_bytes_per_packet_;
>+  bool network_available_ RTC_GUARDED_BY(task_queue_);
>+  bool periodic_tasks_enabled_ RTC_GUARDED_BY(task_queue_);
>+  bool packet_feedback_available_ RTC_GUARDED_BY(task_queue_);
>+  send_side_cc_internal::PeriodicTask* pacer_queue_update_task_
>+      RTC_GUARDED_BY(task_queue_);
>+  send_side_cc_internal::PeriodicTask* controller_task_
>+      RTC_GUARDED_BY(task_queue_);
>+
>+  // Protects access to last_packet_feedback_vector_ in feedback adapter.
>+  // TODO(srte): Remove this checker when feedback adapter runs on task queue.
>+  rtc::RaceChecker worker_race_;
>+
>+  rtc::TaskQueue* task_queue_;
>+
>+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(SendSideCongestionController);
>+};
>+}  // namespace webrtc_cc
>+}  // namespace webrtc
>+
>+#endif  // MODULES_CONGESTION_CONTROLLER_RTP_INCLUDE_SEND_SIDE_CONGESTION_CONTROLLER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/pacer_controller.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/pacer_controller.cc
>new file mode 100644
>index 00000000000..af062e881ee
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/pacer_controller.cc
>@@ -0,0 +1,70 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/congestion_controller/rtp/pacer_controller.h"
>+
>+#include "rtc_base/checks.h"
>+#include "rtc_base/logging.h"
>+
>+namespace webrtc {
>+
>+PacerController::PacerController(PacedSender* pacer) : pacer_(pacer) {
>+  sequenced_checker_.Detach();
>+}
>+
>+PacerController::~PacerController() = default;
>+
>+void PacerController::OnCongestionWindow(DataSize congestion_window) {
>+  RTC_DCHECK_CALLED_SEQUENTIALLY(&sequenced_checker_);
>+  if (congestion_window.IsFinite())
>+    pacer_->SetCongestionWindow(congestion_window.bytes());
>+  else
>+    pacer_->SetCongestionWindow(PacedSender::kNoCongestionWindow);
>+}
>+
>+void PacerController::OnNetworkAvailability(NetworkAvailability msg) {
>+  RTC_DCHECK_CALLED_SEQUENTIALLY(&sequenced_checker_);
>+  network_available_ = msg.network_available;
>+  pacer_->UpdateOutstandingData(0);
>+  SetPacerState(!msg.network_available);
>+}
>+
>+void PacerController::OnNetworkRouteChange(NetworkRouteChange) {
>+  RTC_DCHECK_CALLED_SEQUENTIALLY(&sequenced_checker_);
>+  pacer_->UpdateOutstandingData(0);
>+}
>+
>+void PacerController::OnPacerConfig(PacerConfig msg) {
>+  RTC_DCHECK_CALLED_SEQUENTIALLY(&sequenced_checker_);
>+  DataRate pacing_rate = msg.data_window / msg.time_window;
>+  DataRate padding_rate = msg.pad_window / msg.time_window;
>+  pacer_->SetPacingRates(pacing_rate.bps(), padding_rate.bps());
>+}
>+
>+void PacerController::OnProbeClusterConfig(ProbeClusterConfig config) {
>+  RTC_DCHECK_CALLED_SEQUENTIALLY(&sequenced_checker_);
>+  int64_t bitrate_bps = config.target_data_rate.bps();
>+  pacer_->CreateProbeCluster(bitrate_bps);
>+}
>+
>+void PacerController::OnOutstandingData(DataSize in_flight_data) {
>+  RTC_DCHECK_CALLED_SEQUENTIALLY(&sequenced_checker_);
>+  pacer_->UpdateOutstandingData(in_flight_data.bytes());
>+}
>+
>+void PacerController::SetPacerState(bool paused) {
>+  if (paused && !pacer_paused_)
>+    pacer_->Pause();
>+  else if (!paused && pacer_paused_)
>+    pacer_->Resume();
>+  pacer_paused_ = paused;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/pacer_controller.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/pacer_controller.h
>new file mode 100644
>index 00000000000..b778b784890
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/pacer_controller.h
>@@ -0,0 +1,50 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_CONGESTION_CONTROLLER_RTP_PACER_CONTROLLER_H_
>+#define MODULES_CONGESTION_CONTROLLER_RTP_PACER_CONTROLLER_H_
>+
>+#include <memory>
>+
>+#include "api/transport/network_types.h"
>+#include "modules/pacing/paced_sender.h"
>+#include "rtc_base/sequenced_task_checker.h"
>+
>+namespace webrtc {
>+class Clock;
>+
>+// Wrapper class to control pacer using task queues. Note that this class is
>+// only designed to be used from a single task queue and has no built in
>+// concurrency safety.
>+// TODO(srte): Integrate this interface directly into PacedSender.
>+class PacerController {
>+ public:
>+  explicit PacerController(PacedSender* pacer);
>+  ~PacerController();
>+  void OnCongestionWindow(DataSize msg);
>+  void OnNetworkAvailability(NetworkAvailability msg);
>+  void OnNetworkRouteChange(NetworkRouteChange msg);
>+  void OnOutstandingData(DataSize in_flight_data);
>+  void OnPacerConfig(PacerConfig msg);
>+  void OnProbeClusterConfig(ProbeClusterConfig msg);
>+
>+ private:
>+  void SetPacerState(bool paused);
>+  PacedSender* const pacer_;
>+
>+  absl::optional<PacerConfig> current_pacer_config_;
>+  bool pacer_paused_ = false;
>+  bool network_available_ = true;
>+
>+  rtc::SequencedTaskChecker sequenced_checker_;
>+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(PacerController);
>+};
>+}  // namespace webrtc
>+#endif  // MODULES_CONGESTION_CONTROLLER_RTP_PACER_CONTROLLER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/send_side_congestion_controller.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/send_side_congestion_controller.cc
>new file mode 100644
>index 00000000000..5205940de1e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/send_side_congestion_controller.cc
>@@ -0,0 +1,837 @@
>+/*
>+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/congestion_controller/rtp/include/send_side_congestion_controller.h"
>+
>+#include <algorithm>
>+#include <functional>
>+#include <memory>
>+#include <vector>
>+#include "absl/memory/memory.h"
>+#include "api/transport/network_types.h"
>+#include "modules/congestion_controller/congestion_window_pushback_controller.h"
>+#include "modules/congestion_controller/goog_cc/include/goog_cc_factory.h"
>+#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
>+#include "rtc_base/bind.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/format_macros.h"
>+#include "rtc_base/logging.h"
>+#include "rtc_base/numerics/safe_conversions.h"
>+#include "rtc_base/numerics/safe_minmax.h"
>+#include "rtc_base/rate_limiter.h"
>+#include "rtc_base/sequenced_task_checker.h"
>+#include "rtc_base/socket.h"
>+#include "rtc_base/timeutils.h"
>+#include "system_wrappers/include/field_trial.h"
>+#include "system_wrappers/include/runtime_enabled_features.h"
>+
>+using absl::make_unique;
>+
>+namespace webrtc {
>+namespace webrtc_cc {
>+namespace {
>+using send_side_cc_internal::PeriodicTask;
>+
>+const char kCwndExperiment[] = "WebRTC-CwndExperiment";
>+
>+// When CongestionWindowPushback is enabled, the pacer is oblivious to
>+// the congestion window. The relation between outstanding data and
>+// the congestion window affects encoder allocations directly.
>+const char kCongestionPushbackExperiment[] = "WebRTC-CongestionWindowPushback";
>+
>+// When PacerPushbackExperiment is enabled, build-up in the pacer due to
>+// the congestion window and/or data spikes reduces encoder allocations.
>+const char kPacerPushbackExperiment[] = "WebRTC-PacerPushbackExperiment";
>+const int64_t PacerQueueUpdateIntervalMs = 25;
>+
>+bool IsPacerPushbackExperimentEnabled() {
>+  return webrtc::field_trial::IsEnabled(kPacerPushbackExperiment) ||
>+         (!webrtc::field_trial::IsDisabled(kPacerPushbackExperiment) &&
>+          webrtc::runtime_enabled_features::IsFeatureEnabled(
>+              webrtc::runtime_enabled_features::kDualStreamModeFeatureName));
>+}
>+
>+bool IsCongestionWindowPushbackExperimentEnabled() {
>+  return webrtc::field_trial::IsEnabled(kCongestionPushbackExperiment) &&
>+         webrtc::field_trial::IsEnabled(kCwndExperiment);
>+}
>+
>+std::unique_ptr<CongestionWindowPushbackController>
>+MaybeInitalizeCongestionWindowPushbackController() {
>+  return IsCongestionWindowPushbackExperimentEnabled()
>+             ? absl::make_unique<CongestionWindowPushbackController>()
>+             : nullptr;
>+}
>+
>+void SortPacketFeedbackVector(std::vector<webrtc::PacketFeedback>* input) {
>+  std::sort(input->begin(), input->end(), PacketFeedbackComparator());
>+}
>+
>+PacketResult NetworkPacketFeedbackFromRtpPacketFeedback(
>+    const webrtc::PacketFeedback& pf) {
>+  PacketResult feedback;
>+  if (pf.arrival_time_ms == webrtc::PacketFeedback::kNotReceived)
>+    feedback.receive_time = Timestamp::Infinity();
>+  else
>+    feedback.receive_time = Timestamp::ms(pf.arrival_time_ms);
>+  if (pf.send_time_ms != webrtc::PacketFeedback::kNoSendTime) {
>+    feedback.sent_packet = SentPacket();
>+    feedback.sent_packet->sequence_number = pf.long_sequence_number;
>+    feedback.sent_packet->send_time = Timestamp::ms(pf.send_time_ms);
>+    feedback.sent_packet->size = DataSize::bytes(pf.payload_size);
>+    feedback.sent_packet->pacing_info = pf.pacing_info;
>+  }
>+  return feedback;
>+}
>+
>+std::vector<PacketResult> PacketResultsFromRtpFeedbackVector(
>+    const std::vector<PacketFeedback>& feedback_vector) {
>+  RTC_DCHECK(std::is_sorted(feedback_vector.begin(), feedback_vector.end(),
>+                            PacketFeedbackComparator()));
>+
>+  std::vector<PacketResult> packet_feedbacks;
>+  packet_feedbacks.reserve(feedback_vector.size());
>+  for (const PacketFeedback& rtp_feedback : feedback_vector) {
>+    auto feedback = NetworkPacketFeedbackFromRtpPacketFeedback(rtp_feedback);
>+    packet_feedbacks.push_back(feedback);
>+  }
>+  return packet_feedbacks;
>+}
>+
>+TargetRateConstraints ConvertConstraints(int min_bitrate_bps,
>+                                         int max_bitrate_bps,
>+                                         const Clock* clock) {
>+  TargetRateConstraints msg;
>+  msg.at_time = Timestamp::ms(clock->TimeInMilliseconds());
>+  msg.min_data_rate =
>+      min_bitrate_bps >= 0 ? DataRate::bps(min_bitrate_bps) : DataRate::Zero();
>+  msg.max_data_rate = max_bitrate_bps > 0 ? DataRate::bps(max_bitrate_bps)
>+                                          : DataRate::Infinity();
>+  return msg;
>+}
>+
>+// The template closure pattern is based on rtc::ClosureTask.
>+template <class Closure>
>+class PeriodicTaskImpl final : public PeriodicTask {
>+ public:
>+  PeriodicTaskImpl(rtc::TaskQueue* task_queue,
>+                   int64_t period_ms,
>+                   Closure&& closure)
>+      : task_queue_(task_queue),
>+        period_ms_(period_ms),
>+        closure_(std::forward<Closure>(closure)) {}
>+  bool Run() override {
>+    if (!running_)
>+      return true;
>+    closure_();
>+    // absl::WrapUnique lets us repost this task on the TaskQueue.
>+    task_queue_->PostDelayedTask(absl::WrapUnique(this), period_ms_);
>+    // Return false to tell TaskQueue to not destruct this object, since we have
>+    // taken ownership with absl::WrapUnique.
>+    return false;
>+  }
>+  void Stop() override {
>+    if (task_queue_->IsCurrent()) {
>+      RTC_DCHECK(running_);
>+      running_ = false;
>+    } else {
>+      task_queue_->PostTask([this] { Stop(); });
>+    }
>+  }
>+
>+ private:
>+  rtc::TaskQueue* const task_queue_;
>+  const int64_t period_ms_;
>+  typename std::remove_const<
>+      typename std::remove_reference<Closure>::type>::type closure_;
>+  bool running_ = true;
>+};
>+
>+template <class Closure>
>+static PeriodicTask* StartPeriodicTask(rtc::TaskQueue* task_queue,
>+                                       int64_t period_ms,
>+                                       Closure&& closure) {
>+  auto periodic_task = absl::make_unique<PeriodicTaskImpl<Closure>>(
>+      task_queue, period_ms, std::forward<Closure>(closure));
>+  PeriodicTask* periodic_task_ptr = periodic_task.get();
>+  task_queue->PostDelayedTask(std::move(periodic_task), period_ms);
>+  return periodic_task_ptr;
>+}
>+
>+}  // namespace
>+
>+namespace send_side_cc_internal {
>+class ControlHandler {
>+ public:
>+  ControlHandler(NetworkChangedObserver* observer,
>+                 PacerController* pacer_controller,
>+                 const Clock* clock);
>+
>+  void PostUpdates(NetworkControlUpdate update);
>+
>+  void OnNetworkAvailability(NetworkAvailability msg);
>+  void OnOutstandingData(DataSize in_flight_data);
>+  void OnPacerQueueUpdate(TimeDelta expected_queue_time);
>+
>+  absl::optional<TargetTransferRate> last_transfer_rate();
>+
>+ private:
>+  void OnNetworkInvalidation();
>+  bool GetNetworkParameters(int32_t* estimated_bitrate_bps,
>+                            uint8_t* fraction_loss,
>+                            int64_t* rtt_ms);
>+  bool IsSendQueueFull() const;
>+  bool HasNetworkParametersToReportChanged(int64_t bitrate_bps,
>+                                           uint8_t fraction_loss,
>+                                           int64_t rtt);
>+  NetworkChangedObserver* observer_ = nullptr;
>+  PacerController* pacer_controller_;
>+
>+  absl::optional<TargetTransferRate> current_target_rate_msg_;
>+  bool network_available_ = true;
>+  int64_t last_reported_target_bitrate_bps_ = 0;
>+  uint8_t last_reported_fraction_loss_ = 0;
>+  int64_t last_reported_rtt_ms_ = 0;
>+  const bool pacer_pushback_experiment_ = false;
>+  uint32_t min_pushback_target_bitrate_bps_;
>+  int64_t pacer_expected_queue_ms_ = 0;
>+  double encoding_rate_ratio_ = 1.0;
>+  const std::unique_ptr<CongestionWindowPushbackController>
>+      congestion_window_pushback_controller_;
>+
>+  rtc::SequencedTaskChecker sequenced_checker_;
>+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(ControlHandler);
>+};
>+
>+ControlHandler::ControlHandler(NetworkChangedObserver* observer,
>+                               PacerController* pacer_controller,
>+                               const Clock* clock)
>+    : observer_(observer),
>+      pacer_controller_(pacer_controller),
>+      pacer_pushback_experiment_(IsPacerPushbackExperimentEnabled()),
>+      congestion_window_pushback_controller_(
>+          MaybeInitalizeCongestionWindowPushbackController()) {
>+  sequenced_checker_.Detach();
>+}
>+
>+void ControlHandler::PostUpdates(NetworkControlUpdate update) {
>+  RTC_DCHECK_CALLED_SEQUENTIALLY(&sequenced_checker_);
>+  if (update.congestion_window) {
>+    if (congestion_window_pushback_controller_) {
>+      congestion_window_pushback_controller_->SetDataWindow(
>+          update.congestion_window.value());
>+    } else {
>+      pacer_controller_->OnCongestionWindow(*update.congestion_window);
>+    }
>+  }
>+  if (update.pacer_config) {
>+    pacer_controller_->OnPacerConfig(*update.pacer_config);
>+  }
>+  for (const auto& probe : update.probe_cluster_configs) {
>+    pacer_controller_->OnProbeClusterConfig(probe);
>+  }
>+  if (update.target_rate) {
>+    current_target_rate_msg_ = *update.target_rate;
>+    OnNetworkInvalidation();
>+  }
>+}
>+
>+void ControlHandler::OnNetworkAvailability(NetworkAvailability msg) {
>+  RTC_DCHECK_CALLED_SEQUENTIALLY(&sequenced_checker_);
>+  network_available_ = msg.network_available;
>+  OnNetworkInvalidation();
>+}
>+
>+void ControlHandler::OnOutstandingData(DataSize in_flight_data) {
>+  RTC_DCHECK_CALLED_SEQUENTIALLY(&sequenced_checker_);
>+  if (congestion_window_pushback_controller_) {
>+    congestion_window_pushback_controller_->UpdateOutstandingData(
>+        in_flight_data.bytes());
>+  }
>+  OnNetworkInvalidation();
>+}
>+
>+void ControlHandler::OnPacerQueueUpdate(TimeDelta expected_queue_time) {
>+  RTC_DCHECK_CALLED_SEQUENTIALLY(&sequenced_checker_);
>+  pacer_expected_queue_ms_ = expected_queue_time.ms();
>+  OnNetworkInvalidation();
>+}
>+
>+void ControlHandler::OnNetworkInvalidation() {
>+  if (!current_target_rate_msg_.has_value())
>+    return;
>+
>+  uint32_t target_bitrate_bps = current_target_rate_msg_->target_rate.bps();
>+  int64_t rtt_ms =
>+      current_target_rate_msg_->network_estimate.round_trip_time.ms();
>+  float loss_rate_ratio =
>+      current_target_rate_msg_->network_estimate.loss_rate_ratio;
>+
>+  int loss_ratio_255 = loss_rate_ratio * 255;
>+  uint8_t fraction_loss =
>+      rtc::dchecked_cast<uint8_t>(rtc::SafeClamp(loss_ratio_255, 0, 255));
>+
>+  int64_t probing_interval_ms =
>+      current_target_rate_msg_->network_estimate.bwe_period.ms();
>+
>+  if (!network_available_) {
>+    target_bitrate_bps = 0;
>+  } else if (congestion_window_pushback_controller_) {
>+    target_bitrate_bps =
>+        congestion_window_pushback_controller_->UpdateTargetBitrate(
>+            target_bitrate_bps);
>+  } else if (!pacer_pushback_experiment_) {
>+    target_bitrate_bps = IsSendQueueFull() ? 0 : target_bitrate_bps;
>+  } else {
>+    int64_t queue_length_ms = pacer_expected_queue_ms_;
>+
>+    if (queue_length_ms == 0) {
>+      encoding_rate_ratio_ = 1.0;
>+    } else if (queue_length_ms > 50) {
>+      double encoding_ratio = 1.0 - queue_length_ms / 1000.0;
>+      encoding_rate_ratio_ = std::min(encoding_rate_ratio_, encoding_ratio);
>+      encoding_rate_ratio_ = std::max(encoding_rate_ratio_, 0.0);
>+    }
>+
>+    target_bitrate_bps *= encoding_rate_ratio_;
>+    target_bitrate_bps = target_bitrate_bps < 50000 ? 0 : target_bitrate_bps;
>+  }
>+  if (HasNetworkParametersToReportChanged(target_bitrate_bps, fraction_loss,
>+                                          rtt_ms)) {
>+    observer_->OnNetworkChanged(target_bitrate_bps, fraction_loss, rtt_ms,
>+                                probing_interval_ms);
>+  }
>+}
>+bool ControlHandler::HasNetworkParametersToReportChanged(
>+    int64_t target_bitrate_bps,
>+    uint8_t fraction_loss,
>+    int64_t rtt_ms) {
>+  bool changed = last_reported_target_bitrate_bps_ != target_bitrate_bps ||
>+                 (target_bitrate_bps > 0 &&
>+                  (last_reported_fraction_loss_ != fraction_loss ||
>+                   last_reported_rtt_ms_ != rtt_ms));
>+  if (changed &&
>+      (last_reported_target_bitrate_bps_ == 0 || target_bitrate_bps == 0)) {
>+    RTC_LOG(LS_INFO) << "Bitrate estimate state changed, BWE: "
>+                     << target_bitrate_bps << " bps.";
>+  }
>+  last_reported_target_bitrate_bps_ = target_bitrate_bps;
>+  last_reported_fraction_loss_ = fraction_loss;
>+  last_reported_rtt_ms_ = rtt_ms;
>+  return changed;
>+}
>+
>+bool ControlHandler::IsSendQueueFull() const {
>+  return pacer_expected_queue_ms_ > PacedSender::kMaxQueueLengthMs;
>+}
>+
>+absl::optional<TargetTransferRate> ControlHandler::last_transfer_rate() {
>+  RTC_DCHECK_CALLED_SEQUENTIALLY(&sequenced_checker_);
>+  return current_target_rate_msg_;
>+}
>+}  // namespace send_side_cc_internal
>+
>+SendSideCongestionController::SendSideCongestionController(
>+    const Clock* clock,
>+    rtc::TaskQueue* task_queue,
>+    RtcEventLog* event_log,
>+    PacedSender* pacer,
>+    int start_bitrate_bps,
>+    int min_bitrate_bps,
>+    int max_bitrate_bps,
>+    NetworkControllerFactoryInterface* controller_factory)
>+    : clock_(clock),
>+      pacer_(pacer),
>+      transport_feedback_adapter_(clock_),
>+      controller_factory_with_feedback_(controller_factory),
>+      controller_factory_fallback_(
>+          absl::make_unique<GoogCcNetworkControllerFactory>(event_log)),
>+      pacer_controller_(absl::make_unique<PacerController>(pacer_)),
>+      process_interval_(controller_factory_fallback_->GetProcessInterval()),
>+      last_report_block_time_(Timestamp::ms(clock_->TimeInMilliseconds())),
>+      observer_(nullptr),
>+      send_side_bwe_with_overhead_(
>+          webrtc::field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")),
>+      transport_overhead_bytes_per_packet_(0),
>+      network_available_(false),
>+      periodic_tasks_enabled_(true),
>+      packet_feedback_available_(false),
>+      pacer_queue_update_task_(nullptr),
>+      controller_task_(nullptr),
>+      task_queue_(task_queue) {
>+  initial_config_.constraints =
>+      ConvertConstraints(min_bitrate_bps, max_bitrate_bps, clock_);
>+  RTC_DCHECK(start_bitrate_bps > 0);
>+  initial_config_.starting_bandwidth = DataRate::bps(start_bitrate_bps);
>+}
>+
>+// There is no point in having a network controller for a network that is not
>+// yet available and if we don't have any observer of it's state.
>+// MaybeCreateControllers is used to trigger creation if those things are
>+// fulfilled. This is needed since dependent code uses the period until network
>+// is signalled to be avaliabile to set the expected start bitrate which is sent
>+// to the initializer for NetworkControllers. The observer is injected later due
>+// to a circular dependency between RtpTransportControllerSend and Call.
>+// TODO(srte): Break the circular dependency issue and make sure that starting
>+// bandwidth is set before this class is initialized so the controllers can be
>+// created in the constructor.
>+void SendSideCongestionController::MaybeCreateControllers() {
>+  if (!controller_)
>+    MaybeRecreateControllers();
>+}
>+
>+void SendSideCongestionController::MaybeRecreateControllers() {
>+  if (!network_available_ || !observer_)
>+    return;
>+  if (!control_handler_) {
>+    control_handler_ = absl::make_unique<send_side_cc_internal::ControlHandler>(
>+        observer_, pacer_controller_.get(), clock_);
>+  }
>+
>+  initial_config_.constraints.at_time =
>+      Timestamp::ms(clock_->TimeInMilliseconds());
>+  initial_config_.stream_based_config = streams_config_;
>+
>+  if (!controller_) {
>+    // TODO(srte): Use fallback controller if no feedback is available.
>+    if (controller_factory_with_feedback_) {
>+      RTC_LOG(LS_INFO) << "Creating feedback based only controller";
>+      controller_ = controller_factory_with_feedback_->Create(initial_config_);
>+      process_interval_ =
>+          controller_factory_with_feedback_->GetProcessInterval();
>+    } else {
>+      RTC_LOG(LS_INFO) << "Creating fallback controller";
>+      controller_ = controller_factory_fallback_->Create(initial_config_);
>+      process_interval_ = controller_factory_fallback_->GetProcessInterval();
>+    }
>+    UpdateControllerWithTimeInterval();
>+    StartProcessPeriodicTasks();
>+  }
>+  RTC_DCHECK(controller_);
>+}
>+
>+SendSideCongestionController::~SendSideCongestionController() = default;
>+
>+void SendSideCongestionController::RegisterPacketFeedbackObserver(
>+    PacketFeedbackObserver* observer) {
>+  transport_feedback_adapter_.RegisterPacketFeedbackObserver(observer);
>+}
>+
>+void SendSideCongestionController::DeRegisterPacketFeedbackObserver(
>+    PacketFeedbackObserver* observer) {
>+  transport_feedback_adapter_.DeRegisterPacketFeedbackObserver(observer);
>+}
>+
>+void SendSideCongestionController::RegisterNetworkObserver(
>+    NetworkChangedObserver* observer) {
>+  task_queue_->PostTask([this, observer]() {
>+    RTC_DCHECK_RUN_ON(task_queue_);
>+    RTC_DCHECK(observer_ == nullptr);
>+    observer_ = observer;
>+    MaybeCreateControllers();
>+  });
>+}
>+
>+void SendSideCongestionController::SetBweBitrates(int min_bitrate_bps,
>+                                                  int start_bitrate_bps,
>+                                                  int max_bitrate_bps) {
>+  TargetRateConstraints constraints =
>+      ConvertConstraints(min_bitrate_bps, max_bitrate_bps, clock_);
>+  task_queue_->PostTask([this, constraints, start_bitrate_bps]() {
>+    RTC_DCHECK_RUN_ON(task_queue_);
>+    if (controller_) {
>+      control_handler_->PostUpdates(
>+          controller_->OnTargetRateConstraints(constraints));
>+    } else {
>+      initial_config_.constraints = constraints;
>+      if (start_bitrate_bps > 0)
>+        initial_config_.starting_bandwidth = DataRate::bps(start_bitrate_bps);
>+    }
>+  });
>+}
>+
>+void SendSideCongestionController::SetAllocatedSendBitrateLimits(
>+    int64_t min_send_bitrate_bps,
>+    int64_t max_padding_bitrate_bps,
>+    int64_t max_total_bitrate_bps) {
>+  task_queue_->PostTask([this, min_send_bitrate_bps, max_padding_bitrate_bps,
>+                         max_total_bitrate_bps]() {
>+    RTC_DCHECK_RUN_ON(task_queue_);
>+    streams_config_.min_pacing_rate = DataRate::bps(min_send_bitrate_bps);
>+    streams_config_.max_padding_rate = DataRate::bps(max_padding_bitrate_bps);
>+    streams_config_.max_total_allocated_bitrate =
>+        DataRate::bps(max_total_bitrate_bps);
>+    UpdateStreamsConfig();
>+  });
>+}
>+
>+// TODO(holmer): Split this up and use SetBweBitrates in combination with
>+// OnNetworkRouteChanged.
>+void SendSideCongestionController::OnNetworkRouteChanged(
>+    const rtc::NetworkRoute& network_route,
>+    int start_bitrate_bps,
>+    int min_bitrate_bps,
>+    int max_bitrate_bps) {
>+  transport_feedback_adapter_.SetNetworkIds(network_route.local_network_id,
>+                                            network_route.remote_network_id);
>+  transport_overhead_bytes_per_packet_ = network_route.packet_overhead;
>+
>+  NetworkRouteChange msg;
>+  msg.at_time = Timestamp::ms(clock_->TimeInMilliseconds());
>+  msg.constraints =
>+      ConvertConstraints(min_bitrate_bps, max_bitrate_bps, clock_);
>+  if (start_bitrate_bps > 0)
>+    msg.starting_rate = DataRate::bps(start_bitrate_bps);
>+  task_queue_->PostTask([this, msg]() {
>+    RTC_DCHECK_RUN_ON(task_queue_);
>+    if (controller_) {
>+      control_handler_->PostUpdates(controller_->OnNetworkRouteChange(msg));
>+    } else {
>+      if (msg.starting_rate)
>+        initial_config_.starting_bandwidth = *msg.starting_rate;
>+      initial_config_.constraints = msg.constraints;
>+    }
>+    pacer_controller_->OnNetworkRouteChange(msg);
>+  });
>+}
>+
>+bool SendSideCongestionController::AvailableBandwidth(
>+    uint32_t* bandwidth) const {
>+  // This is only called in the OnNetworkChanged callback in
>+  // RtpTransportControllerSend which is called from ControlHandler, which is
>+  // running on the task queue.
>+  // TODO(srte): Remove this function when RtpTransportControllerSend stops
>+  // calling it.
>+  RTC_DCHECK_RUN_ON(task_queue_);
>+  if (!control_handler_) {
>+    return false;
>+  }
>+  // TODO(srte): Remove this interface and push information about bandwidth
>+  // estimation to users of this class, thereby reducing synchronous calls.
>+  if (control_handler_->last_transfer_rate().has_value()) {
>+    *bandwidth = control_handler_->last_transfer_rate()
>+                     ->network_estimate.bandwidth.bps();
>+    return true;
>+  }
>+  return false;
>+}
>+
>+RtcpBandwidthObserver* SendSideCongestionController::GetBandwidthObserver() {
>+  return this;
>+}
>+
>+void SendSideCongestionController::SetPerPacketFeedbackAvailable(
>+    bool available) {
>+  task_queue_->PostTask([this, available]() {
>+    RTC_DCHECK_RUN_ON(task_queue_);
>+    packet_feedback_available_ = available;
>+    MaybeRecreateControllers();
>+  });
>+}
>+
>+void SendSideCongestionController::EnablePeriodicAlrProbing(bool enable) {
>+  task_queue_->PostTask([this, enable]() {
>+    RTC_DCHECK_RUN_ON(task_queue_);
>+    streams_config_.requests_alr_probing = enable;
>+    UpdateStreamsConfig();
>+  });
>+}
>+
>+void SendSideCongestionController::UpdateStreamsConfig() {
>+  streams_config_.at_time = Timestamp::ms(clock_->TimeInMilliseconds());
>+  if (controller_)
>+    control_handler_->PostUpdates(
>+        controller_->OnStreamsConfig(streams_config_));
>+}
>+
>+TransportFeedbackObserver*
>+SendSideCongestionController::GetTransportFeedbackObserver() {
>+  return this;
>+}
>+
>+void SendSideCongestionController::SignalNetworkState(NetworkState state) {
>+  RTC_LOG(LS_INFO) << "SignalNetworkState "
>+                   << (state == kNetworkUp ? "Up" : "Down");
>+  NetworkAvailability msg;
>+  msg.at_time = Timestamp::ms(clock_->TimeInMilliseconds());
>+  msg.network_available = state == kNetworkUp;
>+  task_queue_->PostTask([this, msg]() {
>+    RTC_DCHECK_RUN_ON(task_queue_);
>+    network_available_ = msg.network_available;
>+    if (controller_) {
>+      control_handler_->PostUpdates(controller_->OnNetworkAvailability(msg));
>+      pacer_controller_->OnNetworkAvailability(msg);
>+      control_handler_->OnNetworkAvailability(msg);
>+    } else {
>+      MaybeCreateControllers();
>+    }
>+  });
>+}
>+
>+void SendSideCongestionController::OnSentPacket(
>+    const rtc::SentPacket& sent_packet) {
>+  // We're not interested in packets without an id, which may be stun packets,
>+  // etc, sent on the same transport.
>+  if (sent_packet.packet_id == -1)
>+    return;
>+  transport_feedback_adapter_.OnSentPacket(sent_packet.packet_id,
>+                                           sent_packet.send_time_ms);
>+  MaybeUpdateOutstandingData();
>+  auto packet = transport_feedback_adapter_.GetPacket(sent_packet.packet_id);
>+  if (packet.has_value()) {
>+    SentPacket msg;
>+    msg.size = DataSize::bytes(packet->payload_size);
>+    msg.send_time = Timestamp::ms(packet->send_time_ms);
>+    msg.sequence_number = packet->long_sequence_number;
>+    msg.data_in_flight =
>+        DataSize::bytes(transport_feedback_adapter_.GetOutstandingBytes());
>+    task_queue_->PostTask([this, msg]() {
>+      RTC_DCHECK_RUN_ON(task_queue_);
>+      if (controller_)
>+        control_handler_->PostUpdates(controller_->OnSentPacket(msg));
>+    });
>+  }
>+}
>+
>+void SendSideCongestionController::OnRttUpdate(int64_t avg_rtt_ms,
>+                                               int64_t max_rtt_ms) {
>+  int64_t now_ms = clock_->TimeInMilliseconds();
>+  RoundTripTimeUpdate report;
>+  report.receive_time = Timestamp::ms(now_ms);
>+  report.round_trip_time = TimeDelta::ms(avg_rtt_ms);
>+  report.smoothed = true;
>+  task_queue_->PostTask([this, report]() {
>+    RTC_DCHECK_RUN_ON(task_queue_);
>+    if (controller_)
>+      control_handler_->PostUpdates(controller_->OnRoundTripTimeUpdate(report));
>+  });
>+}
>+
>+int64_t SendSideCongestionController::TimeUntilNextProcess() {
>+  // Using task queue to process, just sleep long to avoid wasting resources.
>+  return 60 * 1000;
>+}
>+
>+void SendSideCongestionController::Process() {
>+  // Ignored, using task queue to process.
>+}
>+
>+void SendSideCongestionController::StartProcessPeriodicTasks() {
>+  if (!periodic_tasks_enabled_)
>+    return;
>+  if (!pacer_queue_update_task_) {
>+    pacer_queue_update_task_ =
>+        StartPeriodicTask(task_queue_, PacerQueueUpdateIntervalMs, [this]() {
>+          RTC_DCHECK_RUN_ON(task_queue_);
>+          UpdatePacerQueue();
>+        });
>+  }
>+  if (controller_task_) {
>+    // Stop is not synchronous, but is guaranteed to occur before the first
>+    // invocation of the new controller task started below.
>+    controller_task_->Stop();
>+    controller_task_ = nullptr;
>+  }
>+  if (process_interval_.IsFinite()) {
>+    // The controller task is owned by the task queue and lives until the task
>+    // queue is destroyed or some time after Stop() is called, whichever comes
>+    // first.
>+    controller_task_ =
>+        StartPeriodicTask(task_queue_, process_interval_.ms(), [this]() {
>+          RTC_DCHECK_RUN_ON(task_queue_);
>+          UpdateControllerWithTimeInterval();
>+        });
>+  }
>+}
>+
>+void SendSideCongestionController::UpdateControllerWithTimeInterval() {
>+  if (controller_) {
>+    ProcessInterval msg;
>+    msg.at_time = Timestamp::ms(clock_->TimeInMilliseconds());
>+    control_handler_->PostUpdates(controller_->OnProcessInterval(msg));
>+  }
>+}
>+
>+void SendSideCongestionController::UpdatePacerQueue() {
>+  if (control_handler_) {
>+    TimeDelta expected_queue_time =
>+        TimeDelta::ms(pacer_->ExpectedQueueTimeMs());
>+    control_handler_->OnPacerQueueUpdate(expected_queue_time);
>+  }
>+}
>+
>+void SendSideCongestionController::AddPacket(
>+    uint32_t ssrc,
>+    uint16_t sequence_number,
>+    size_t length,
>+    const PacedPacketInfo& pacing_info) {
>+  if (send_side_bwe_with_overhead_) {
>+    length += transport_overhead_bytes_per_packet_;
>+  }
>+  transport_feedback_adapter_.AddPacket(ssrc, sequence_number, length,
>+                                        pacing_info);
>+}
>+
>+void SendSideCongestionController::OnTransportFeedback(
>+    const rtcp::TransportFeedback& feedback) {
>+  RTC_DCHECK_RUNS_SERIALIZED(&worker_race_);
>+  int64_t feedback_time_ms = clock_->TimeInMilliseconds();
>+
>+  DataSize prior_in_flight =
>+      DataSize::bytes(transport_feedback_adapter_.GetOutstandingBytes());
>+  transport_feedback_adapter_.OnTransportFeedback(feedback);
>+  MaybeUpdateOutstandingData();
>+
>+  std::vector<PacketFeedback> feedback_vector =
>+      transport_feedback_adapter_.GetTransportFeedbackVector();
>+  SortPacketFeedbackVector(&feedback_vector);
>+
>+  if (!feedback_vector.empty()) {
>+    TransportPacketsFeedback msg;
>+    msg.packet_feedbacks = PacketResultsFromRtpFeedbackVector(feedback_vector);
>+    msg.feedback_time = Timestamp::ms(feedback_time_ms);
>+    msg.prior_in_flight = prior_in_flight;
>+    msg.data_in_flight =
>+        DataSize::bytes(transport_feedback_adapter_.GetOutstandingBytes());
>+    task_queue_->PostTask([this, msg]() {
>+      RTC_DCHECK_RUN_ON(task_queue_);
>+      if (controller_)
>+        control_handler_->PostUpdates(
>+            controller_->OnTransportPacketsFeedback(msg));
>+    });
>+  }
>+}
>+
>+void SendSideCongestionController::MaybeUpdateOutstandingData() {
>+  DataSize in_flight_data =
>+      DataSize::bytes(transport_feedback_adapter_.GetOutstandingBytes());
>+  task_queue_->PostTask([this, in_flight_data]() {
>+    RTC_DCHECK_RUN_ON(task_queue_);
>+    pacer_controller_->OnOutstandingData(in_flight_data);
>+    if (control_handler_)
>+      control_handler_->OnOutstandingData(in_flight_data);
>+  });
>+}
>+
>+std::vector<PacketFeedback>
>+SendSideCongestionController::GetTransportFeedbackVector() const {
>+  RTC_DCHECK_RUNS_SERIALIZED(&worker_race_);
>+  return transport_feedback_adapter_.GetTransportFeedbackVector();
>+}
>+
>+void SendSideCongestionController::PostPeriodicTasksForTest() {
>+  task_queue_->PostTask([this]() {
>+    RTC_DCHECK_RUN_ON(task_queue_);
>+    UpdateControllerWithTimeInterval();
>+    UpdatePacerQueue();
>+  });
>+}
>+
>+void SendSideCongestionController::WaitOnTasksForTest() {
>+  rtc::Event event(false, false);
>+  task_queue_->PostTask([&event]() { event.Set(); });
>+  event.Wait(rtc::Event::kForever);
>+}
>+
>+void SendSideCongestionController::SetPacingFactor(float pacing_factor) {
>+  task_queue_->PostTask([this, pacing_factor]() {
>+    RTC_DCHECK_RUN_ON(task_queue_);
>+    streams_config_.pacing_factor = pacing_factor;
>+    UpdateStreamsConfig();
>+  });
>+}
>+
>+void SendSideCongestionController::SetAllocatedBitrateWithoutFeedback(
>+    uint32_t bitrate_bps) {}
>+
>+void SendSideCongestionController::DisablePeriodicTasks() {
>+  task_queue_->PostTask([this]() {
>+    RTC_DCHECK_RUN_ON(task_queue_);
>+    periodic_tasks_enabled_ = false;
>+  });
>+}
>+
>+void SendSideCongestionController::OnReceivedEstimatedBitrate(
>+    uint32_t bitrate) {
>+  RemoteBitrateReport msg;
>+  msg.receive_time = Timestamp::ms(clock_->TimeInMilliseconds());
>+  msg.bandwidth = DataRate::bps(bitrate);
>+  task_queue_->PostTask([this, msg]() {
>+    RTC_DCHECK_RUN_ON(task_queue_);
>+    if (controller_)
>+      control_handler_->PostUpdates(controller_->OnRemoteBitrateReport(msg));
>+  });
>+}
>+
>+void SendSideCongestionController::OnReceivedRtcpReceiverReport(
>+    const webrtc::ReportBlockList& report_blocks,
>+    int64_t rtt_ms,
>+    int64_t now_ms) {
>+  task_queue_->PostTask([this, report_blocks, now_ms]() {
>+    RTC_DCHECK_RUN_ON(task_queue_);
>+    OnReceivedRtcpReceiverReportBlocks(report_blocks, now_ms);
>+  });
>+
>+  task_queue_->PostTask([this, now_ms, rtt_ms]() {
>+    RTC_DCHECK_RUN_ON(task_queue_);
>+    RoundTripTimeUpdate report;
>+    report.receive_time = Timestamp::ms(now_ms);
>+    report.round_trip_time = TimeDelta::ms(rtt_ms);
>+    report.smoothed = false;
>+    if (controller_)
>+      control_handler_->PostUpdates(controller_->OnRoundTripTimeUpdate(report));
>+  });
>+}
>+
>+void SendSideCongestionController::OnReceivedRtcpReceiverReportBlocks(
>+    const ReportBlockList& report_blocks,
>+    int64_t now_ms) {
>+  if (report_blocks.empty())
>+    return;
>+
>+  int total_packets_lost_delta = 0;
>+  int total_packets_delta = 0;
>+
>+  // Compute the packet loss from all report blocks.
>+  for (const RTCPReportBlock& report_block : report_blocks) {
>+    auto it = last_report_blocks_.find(report_block.source_ssrc);
>+    if (it != last_report_blocks_.end()) {
>+      auto number_of_packets = report_block.extended_highest_sequence_number -
>+                               it->second.extended_highest_sequence_number;
>+      total_packets_delta += number_of_packets;
>+      auto lost_delta = report_block.packets_lost - it->second.packets_lost;
>+      total_packets_lost_delta += lost_delta;
>+    }
>+    last_report_blocks_[report_block.source_ssrc] = report_block;
>+  }
>+  // Can only compute delta if there has been previous blocks to compare to. If
>+  // not, total_packets_delta will be unchanged and there's nothing more to do.
>+  if (!total_packets_delta)
>+    return;
>+  int packets_received_delta = total_packets_delta - total_packets_lost_delta;
>+  // To detect lost packets, at least one packet has to be received. This check
>+  // is needed to avoid bandwith detection update in
>+  // VideoSendStreamTest.SuspendBelowMinBitrate
>+
>+  if (packets_received_delta < 1)
>+    return;
>+  Timestamp now = Timestamp::ms(now_ms);
>+  TransportLossReport msg;
>+  msg.packets_lost_delta = total_packets_lost_delta;
>+  msg.packets_received_delta = packets_received_delta;
>+  msg.receive_time = now;
>+  msg.start_time = last_report_block_time_;
>+  msg.end_time = now;
>+  if (controller_)
>+    control_handler_->PostUpdates(controller_->OnTransportLossReport(msg));
>+  last_report_block_time_ = now;
>+}
>+}  // namespace webrtc_cc
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/send_side_congestion_controller_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/send_side_congestion_controller_unittest.cc
>new file mode 100644
>index 00000000000..d610afe13d0
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/send_side_congestion_controller_unittest.cc
>@@ -0,0 +1,520 @@
>+/*
>+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/congestion_controller/rtp/include/send_side_congestion_controller.h"
>+#include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
>+#include "modules/congestion_controller/include/mock/mock_congestion_observer.h"
>+#include "modules/congestion_controller/rtp/congestion_controller_unittests_helper.h"
>+#include "modules/pacing/mock/mock_paced_sender.h"
>+#include "modules/pacing/packet_router.h"
>+#include "modules/remote_bitrate_estimator/include/bwe_defines.h"
>+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>+#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
>+#include "rtc_base/socket.h"
>+#include "system_wrappers/include/clock.h"
>+#include "test/field_trial.h"
>+#include "test/gmock.h"
>+#include "test/gtest.h"
>+
>+using testing::_;
>+using testing::AtLeast;
>+using testing::Ge;
>+using testing::NiceMock;
>+using testing::Return;
>+using testing::SaveArg;
>+using testing::StrictMock;
>+
>+namespace webrtc {
>+namespace webrtc_cc {
>+namespace test {
>+
>+namespace {
>+using webrtc::test::MockCongestionObserver;
>+const webrtc::PacedPacketInfo kPacingInfo0(0, 5, 2000);
>+const webrtc::PacedPacketInfo kPacingInfo1(1, 8, 4000);
>+
>+const uint32_t kInitialBitrateBps = 60000;
>+const float kDefaultPacingRate = 2.5f;
>+
>+class SendSideCongestionControllerForTest
>+    : public SendSideCongestionController {
>+ public:
>+  using SendSideCongestionController::SendSideCongestionController;
>+  ~SendSideCongestionControllerForTest() {}
>+  using SendSideCongestionController::DisablePeriodicTasks;
>+  void WaitOnTasks() { SendSideCongestionController::WaitOnTasksForTest(); }
>+  void Process() override {
>+    SendSideCongestionController::PostPeriodicTasksForTest();
>+    SendSideCongestionController::WaitOnTasksForTest();
>+  }
>+};
>+}  // namespace
>+
>+class SendSideCongestionControllerTest : public ::testing::Test {
>+ protected:
>+  SendSideCongestionControllerTest()
>+      : clock_(123456),
>+        target_bitrate_observer_(this),
>+        bandwidth_observer_(nullptr) {}
>+  ~SendSideCongestionControllerTest() override {}
>+
>+  void SetUp() override {
>+    pacer_.reset(new NiceMock<MockPacedSender>());
>+    // Set the initial bitrate estimate and expect the |observer| and |pacer_|
>+    // to be updated.
>+    EXPECT_CALL(observer_, OnNetworkChanged(kInitialBitrateBps, _, _, _));
>+    EXPECT_CALL(*pacer_,
>+                SetPacingRates(kInitialBitrateBps * kDefaultPacingRate, _));
>+    EXPECT_CALL(*pacer_, CreateProbeCluster(kInitialBitrateBps * 3));
>+    EXPECT_CALL(*pacer_, CreateProbeCluster(kInitialBitrateBps * 5));
>+    task_queue_ = absl::make_unique<rtc::TaskQueue>("SSCC Test");
>+    controller_.reset(new SendSideCongestionControllerForTest(
>+        &clock_, task_queue_.get(), &event_log_, pacer_.get(),
>+        kInitialBitrateBps, 0, 5 * kInitialBitrateBps, nullptr));
>+    controller_->DisablePeriodicTasks();
>+    controller_->RegisterNetworkObserver(&observer_);
>+    controller_->SignalNetworkState(NetworkState::kNetworkUp);
>+    bandwidth_observer_ = controller_->GetBandwidthObserver();
>+    controller_->WaitOnTasks();
>+    testing::Mock::VerifyAndClearExpectations(pacer_.get());
>+    testing::Mock::VerifyAndClearExpectations(&observer_);
>+  }
>+
>+  void TearDown() override { controller_->WaitOnTasks(); }
>+
>+  // Custom setup - use an observer that tracks the target bitrate, without
>+  // prescribing on which iterations it must change (like a mock would).
>+  void TargetBitrateTrackingSetup() {
>+    bandwidth_observer_ = nullptr;
>+    pacer_.reset(new NiceMock<MockPacedSender>());
>+    task_queue_ = absl::make_unique<rtc::TaskQueue>("SSCC Test");
>+    controller_.reset(new SendSideCongestionControllerForTest(
>+        &clock_, task_queue_.get(), &event_log_, pacer_.get(),
>+        kInitialBitrateBps, 0, 5 * kInitialBitrateBps, nullptr));
>+    controller_->DisablePeriodicTasks();
>+    controller_->RegisterNetworkObserver(&target_bitrate_observer_);
>+    controller_->SignalNetworkState(NetworkState::kNetworkUp);
>+  }
>+
>+  void OnSentPacket(const PacketFeedback& packet_feedback) {
>+    constexpr uint32_t ssrc = 0;
>+    controller_->AddPacket(ssrc, packet_feedback.sequence_number,
>+                           packet_feedback.payload_size,
>+                           packet_feedback.pacing_info);
>+    controller_->OnSentPacket(rtc::SentPacket(packet_feedback.sequence_number,
>+                                              packet_feedback.send_time_ms));
>+  }
>+
>+  // Allows us to track the target bitrate, without prescribing the exact
>+  // iterations when this would hapen, like a mock would.
>+  class TargetBitrateObserver : public NetworkChangedObserver {
>+   public:
>+    explicit TargetBitrateObserver(SendSideCongestionControllerTest* owner)
>+        : owner_(owner) {}
>+    ~TargetBitrateObserver() override = default;
>+    void OnNetworkChanged(uint32_t bitrate_bps,
>+                          uint8_t fraction_loss,  // 0 - 255.
>+                          int64_t rtt_ms,
>+                          int64_t probing_interval_ms) override {
>+      owner_->target_bitrate_bps_ = bitrate_bps;
>+    }
>+
>+   private:
>+    SendSideCongestionControllerTest* owner_;
>+  };
>+
>+  void PacketTransmissionAndFeedbackBlock(uint16_t* seq_num,
>+                                          int64_t runtime_ms,
>+                                          int64_t delay) {
>+    int64_t delay_buildup = 0;
>+    int64_t start_time_ms = clock_.TimeInMilliseconds();
>+    while (clock_.TimeInMilliseconds() - start_time_ms < runtime_ms) {
>+      constexpr size_t kPayloadSize = 1000;
>+      PacketFeedback packet(clock_.TimeInMilliseconds() + delay_buildup,
>+                            clock_.TimeInMilliseconds(), *seq_num, kPayloadSize,
>+                            PacedPacketInfo());
>+      delay_buildup += delay;  // Delay has to increase, or it's just RTT.
>+      OnSentPacket(packet);
>+      // Create expected feedback and send into adapter.
>+      std::unique_ptr<rtcp::TransportFeedback> feedback(
>+          new rtcp::TransportFeedback());
>+      feedback->SetBase(packet.sequence_number, packet.arrival_time_ms * 1000);
>+      EXPECT_TRUE(feedback->AddReceivedPacket(packet.sequence_number,
>+                                              packet.arrival_time_ms * 1000));
>+      rtc::Buffer raw_packet = feedback->Build();
>+      feedback = rtcp::TransportFeedback::ParseFrom(raw_packet.data(),
>+                                                    raw_packet.size());
>+      EXPECT_TRUE(feedback.get() != nullptr);
>+      controller_->OnTransportFeedback(*feedback.get());
>+      clock_.AdvanceTimeMilliseconds(50);
>+      controller_->Process();
>+      ++(*seq_num);
>+    }
>+  }
>+
>+  SimulatedClock clock_;
>+  StrictMock<MockCongestionObserver> observer_;
>+  TargetBitrateObserver target_bitrate_observer_;
>+  NiceMock<MockRtcEventLog> event_log_;
>+  RtcpBandwidthObserver* bandwidth_observer_;
>+  PacketRouter packet_router_;
>+  std::unique_ptr<NiceMock<MockPacedSender>> pacer_;
>+  std::unique_ptr<SendSideCongestionControllerForTest> controller_;
>+  absl::optional<uint32_t> target_bitrate_bps_;
>+  std::unique_ptr<rtc::TaskQueue> task_queue_;
>+};
>+
>+TEST_F(SendSideCongestionControllerTest, OnNetworkChanged) {
>+  // Test no change.
>+  clock_.AdvanceTimeMilliseconds(25);
>+  controller_->Process();
>+
>+  EXPECT_CALL(observer_, OnNetworkChanged(kInitialBitrateBps * 2, _, _, _));
>+  EXPECT_CALL(*pacer_,
>+              SetPacingRates(kInitialBitrateBps * 2 * kDefaultPacingRate, _));
>+  bandwidth_observer_->OnReceivedEstimatedBitrate(kInitialBitrateBps * 2);
>+  clock_.AdvanceTimeMilliseconds(25);
>+  controller_->Process();
>+
>+  EXPECT_CALL(observer_, OnNetworkChanged(kInitialBitrateBps, _, _, _));
>+  EXPECT_CALL(*pacer_,
>+              SetPacingRates(kInitialBitrateBps * kDefaultPacingRate, _));
>+  bandwidth_observer_->OnReceivedEstimatedBitrate(kInitialBitrateBps);
>+  clock_.AdvanceTimeMilliseconds(25);
>+  controller_->Process();
>+}
>+
>+TEST_F(SendSideCongestionControllerTest, OnSendQueueFull) {
>+  EXPECT_CALL(*pacer_, ExpectedQueueTimeMs())
>+      .WillRepeatedly(Return(PacedSender::kMaxQueueLengthMs + 1));
>+
>+  EXPECT_CALL(observer_, OnNetworkChanged(0, _, _, _));
>+  controller_->Process();
>+
>+  // Let the pacer not be full next time the controller checks.
>+  EXPECT_CALL(*pacer_, ExpectedQueueTimeMs())
>+      .WillRepeatedly(Return(PacedSender::kMaxQueueLengthMs - 1));
>+
>+  EXPECT_CALL(observer_, OnNetworkChanged(kInitialBitrateBps, _, _, _));
>+  controller_->Process();
>+}
>+
>+TEST_F(SendSideCongestionControllerTest, OnSendQueueFullAndEstimateChange) {
>+  EXPECT_CALL(*pacer_, ExpectedQueueTimeMs())
>+      .WillRepeatedly(Return(PacedSender::kMaxQueueLengthMs + 1));
>+  EXPECT_CALL(observer_, OnNetworkChanged(0, _, _, _));
>+  controller_->Process();
>+
>+  // Receive new estimate but let the queue still be full.
>+  bandwidth_observer_->OnReceivedEstimatedBitrate(kInitialBitrateBps * 2);
>+  EXPECT_CALL(*pacer_, ExpectedQueueTimeMs())
>+      .WillRepeatedly(Return(PacedSender::kMaxQueueLengthMs + 1));
>+  //  The send pacer should get the new estimate though.
>+  EXPECT_CALL(*pacer_,
>+              SetPacingRates(kInitialBitrateBps * 2 * kDefaultPacingRate, _));
>+  clock_.AdvanceTimeMilliseconds(25);
>+  controller_->Process();
>+
>+  // Let the pacer not be full next time the controller checks.
>+  // |OnNetworkChanged| should be called with the new estimate.
>+  EXPECT_CALL(*pacer_, ExpectedQueueTimeMs())
>+      .WillRepeatedly(Return(PacedSender::kMaxQueueLengthMs - 1));
>+  EXPECT_CALL(observer_, OnNetworkChanged(kInitialBitrateBps * 2, _, _, _));
>+  clock_.AdvanceTimeMilliseconds(25);
>+  controller_->Process();
>+}
>+
>+TEST_F(SendSideCongestionControllerTest, SignalNetworkState) {
>+  EXPECT_CALL(observer_, OnNetworkChanged(0, _, _, _));
>+  controller_->SignalNetworkState(kNetworkDown);
>+  controller_->WaitOnTasks();
>+
>+  EXPECT_CALL(observer_, OnNetworkChanged(kInitialBitrateBps, _, _, _));
>+  controller_->SignalNetworkState(kNetworkUp);
>+  controller_->WaitOnTasks();
>+
>+  EXPECT_CALL(observer_, OnNetworkChanged(0, _, _, _));
>+  controller_->SignalNetworkState(kNetworkDown);
>+}
>+
>+TEST_F(SendSideCongestionControllerTest, OnNetworkRouteChanged) {
>+  int new_bitrate = 200000;
>+  EXPECT_CALL(observer_, OnNetworkChanged(new_bitrate, _, _, _));
>+  EXPECT_CALL(*pacer_, SetPacingRates(new_bitrate * kDefaultPacingRate, _));
>+  rtc::NetworkRoute route;
>+  route.local_network_id = 1;
>+  controller_->OnNetworkRouteChanged(route, new_bitrate, -1, -1);
>+  controller_->WaitOnTasks();
>+
>+  testing::Mock::VerifyAndClearExpectations(pacer_.get());
>+  testing::Mock::VerifyAndClearExpectations(&observer_);
>+  // If the bitrate is reset to -1, the new starting bitrate will be
>+  // the minimum default bitrate kMinBitrateBps.
>+  EXPECT_CALL(
>+      observer_,
>+      OnNetworkChanged(congestion_controller::GetMinBitrateBps(), _, _, _));
>+  EXPECT_CALL(
>+      *pacer_,
>+      SetPacingRates(
>+          congestion_controller::GetMinBitrateBps() * kDefaultPacingRate, _));
>+  route.local_network_id = 2;
>+  controller_->OnNetworkRouteChanged(route, -1, -1, -1);
>+}
>+
>+TEST_F(SendSideCongestionControllerTest, OldFeedback) {
>+  int new_bitrate = 200000;
>+  testing::Mock::VerifyAndClearExpectations(pacer_.get());
>+  EXPECT_CALL(observer_, OnNetworkChanged(new_bitrate, _, _, _));
>+  EXPECT_CALL(*pacer_, SetPacingRates(new_bitrate * kDefaultPacingRate, _));
>+
>+  // Send a few packets on the first network route.
>+  std::vector<PacketFeedback> packets;
>+  packets.push_back(PacketFeedback(0, 0, 0, 1500, kPacingInfo0));
>+  packets.push_back(PacketFeedback(10, 10, 1, 1500, kPacingInfo0));
>+  packets.push_back(PacketFeedback(20, 20, 2, 1500, kPacingInfo0));
>+  packets.push_back(PacketFeedback(30, 30, 3, 1500, kPacingInfo1));
>+  packets.push_back(PacketFeedback(40, 40, 4, 1500, kPacingInfo1));
>+
>+  for (const PacketFeedback& packet : packets)
>+    OnSentPacket(packet);
>+
>+  // Change route and then insert a number of feedback packets.
>+  rtc::NetworkRoute route;
>+  route.local_network_id = 1;
>+  controller_->OnNetworkRouteChanged(route, new_bitrate, -1, -1);
>+
>+  for (const PacketFeedback& packet : packets) {
>+    rtcp::TransportFeedback feedback;
>+    feedback.SetBase(packet.sequence_number, packet.arrival_time_ms * 1000);
>+
>+    EXPECT_TRUE(feedback.AddReceivedPacket(packet.sequence_number,
>+                                           packet.arrival_time_ms * 1000));
>+    feedback.Build();
>+    controller_->OnTransportFeedback(feedback);
>+  }
>+
>+  // If the bitrate is reset to -1, the new starting bitrate will be
>+  // the minimum default bitrate kMinBitrateBps.
>+  EXPECT_CALL(
>+      observer_,
>+      OnNetworkChanged(congestion_controller::GetMinBitrateBps(), _, _, _));
>+  EXPECT_CALL(
>+      *pacer_,
>+      SetPacingRates(
>+          congestion_controller::GetMinBitrateBps() * kDefaultPacingRate, _));
>+  route.local_network_id = 2;
>+  controller_->OnNetworkRouteChanged(route, -1, -1, -1);
>+}
>+
>+TEST_F(SendSideCongestionControllerTest,
>+       SignalNetworkStateAndQueueIsFullAndEstimateChange) {
>+  // Send queue is full.
>+  EXPECT_CALL(*pacer_, ExpectedQueueTimeMs())
>+      .WillRepeatedly(Return(PacedSender::kMaxQueueLengthMs + 1));
>+  EXPECT_CALL(observer_, OnNetworkChanged(0, _, _, _));
>+  controller_->Process();
>+
>+  // Queue is full and network is down. Expect no bitrate change.
>+  controller_->SignalNetworkState(kNetworkDown);
>+  controller_->Process();
>+
>+  // Queue is full but network is up. Expect no bitrate change.
>+  controller_->SignalNetworkState(kNetworkUp);
>+  controller_->Process();
>+
>+  // Receive new estimate but let the queue still be full.
>+  EXPECT_CALL(*pacer_,
>+              SetPacingRates(kInitialBitrateBps * 2 * kDefaultPacingRate, _));
>+  bandwidth_observer_->OnReceivedEstimatedBitrate(kInitialBitrateBps * 2);
>+  clock_.AdvanceTimeMilliseconds(25);
>+  controller_->Process();
>+
>+  // Let the pacer not be full next time the controller checks.
>+  EXPECT_CALL(*pacer_, ExpectedQueueTimeMs())
>+      .WillRepeatedly(Return(PacedSender::kMaxQueueLengthMs - 1));
>+  EXPECT_CALL(observer_, OnNetworkChanged(kInitialBitrateBps * 2, _, _, _));
>+  controller_->Process();
>+}
>+
>+TEST_F(SendSideCongestionControllerTest, GetProbingInterval) {
>+  clock_.AdvanceTimeMilliseconds(25);
>+  controller_->Process();
>+
>+  EXPECT_CALL(observer_, OnNetworkChanged(_, _, _, testing::Ne(0)));
>+  EXPECT_CALL(*pacer_, SetPacingRates(_, _));
>+  bandwidth_observer_->OnReceivedEstimatedBitrate(kInitialBitrateBps * 2);
>+  clock_.AdvanceTimeMilliseconds(25);
>+  controller_->Process();
>+}
>+
>+TEST_F(SendSideCongestionControllerTest, ProbeOnRouteChange) {
>+  EXPECT_CALL(*pacer_, CreateProbeCluster(kInitialBitrateBps * 6));
>+  EXPECT_CALL(*pacer_, CreateProbeCluster(kInitialBitrateBps * 12));
>+  EXPECT_CALL(observer_, OnNetworkChanged(kInitialBitrateBps * 2, _, _, _));
>+  rtc::NetworkRoute route;
>+  route.local_network_id = 1;
>+  controller_->OnNetworkRouteChanged(route, 2 * kInitialBitrateBps, 0,
>+                                     20 * kInitialBitrateBps);
>+  controller_->Process();
>+}
>+
>+// Estimated bitrate reduced when the feedbacks arrive with such a long delay,
>+// that the send-time-history no longer holds the feedbacked packets.
>+TEST_F(SendSideCongestionControllerTest, LongFeedbackDelays) {
>+  TargetBitrateTrackingSetup();
>+
>+  const int64_t kFeedbackTimeoutMs = 60001;
>+  const int kMaxConsecutiveFailedLookups = 5;
>+  for (int i = 0; i < kMaxConsecutiveFailedLookups; ++i) {
>+    std::vector<PacketFeedback> packets;
>+    packets.push_back(
>+        PacketFeedback(i * 100, 2 * i * 100, 0, 1500, kPacingInfo0));
>+    packets.push_back(
>+        PacketFeedback(i * 100 + 10, 2 * i * 100 + 10, 1, 1500, kPacingInfo0));
>+    packets.push_back(
>+        PacketFeedback(i * 100 + 20, 2 * i * 100 + 20, 2, 1500, kPacingInfo0));
>+    packets.push_back(
>+        PacketFeedback(i * 100 + 30, 2 * i * 100 + 30, 3, 1500, kPacingInfo1));
>+    packets.push_back(
>+        PacketFeedback(i * 100 + 40, 2 * i * 100 + 40, 4, 1500, kPacingInfo1));
>+
>+    for (const PacketFeedback& packet : packets)
>+      OnSentPacket(packet);
>+
>+    rtcp::TransportFeedback feedback;
>+    feedback.SetBase(packets[0].sequence_number,
>+                     packets[0].arrival_time_ms * 1000);
>+
>+    for (const PacketFeedback& packet : packets) {
>+      EXPECT_TRUE(feedback.AddReceivedPacket(packet.sequence_number,
>+                                             packet.arrival_time_ms * 1000));
>+    }
>+
>+    feedback.Build();
>+
>+    clock_.AdvanceTimeMilliseconds(kFeedbackTimeoutMs);
>+    PacketFeedback later_packet(kFeedbackTimeoutMs + i * 100 + 40,
>+                                kFeedbackTimeoutMs + i * 200 + 40, 5, 1500,
>+                                kPacingInfo1);
>+    OnSentPacket(later_packet);
>+
>+    controller_->OnTransportFeedback(feedback);
>+
>+    // Check that packets have timed out.
>+    for (PacketFeedback& packet : packets) {
>+      packet.send_time_ms = PacketFeedback::kNoSendTime;
>+      packet.payload_size = 0;
>+      packet.pacing_info = PacedPacketInfo();
>+    }
>+    ComparePacketFeedbackVectors(packets,
>+                                 controller_->GetTransportFeedbackVector());
>+  }
>+
>+  controller_->Process();
>+
>+  EXPECT_EQ(kInitialBitrateBps / 2, target_bitrate_bps_);
>+
>+  // Test with feedback that isn't late enough to time out.
>+  {
>+    std::vector<PacketFeedback> packets;
>+    packets.push_back(PacketFeedback(100, 200, 0, 1500, kPacingInfo0));
>+    packets.push_back(PacketFeedback(110, 210, 1, 1500, kPacingInfo0));
>+    packets.push_back(PacketFeedback(120, 220, 2, 1500, kPacingInfo0));
>+    packets.push_back(PacketFeedback(130, 230, 3, 1500, kPacingInfo1));
>+    packets.push_back(PacketFeedback(140, 240, 4, 1500, kPacingInfo1));
>+
>+    for (const PacketFeedback& packet : packets)
>+      OnSentPacket(packet);
>+
>+    rtcp::TransportFeedback feedback;
>+    feedback.SetBase(packets[0].sequence_number,
>+                     packets[0].arrival_time_ms * 1000);
>+
>+    for (const PacketFeedback& packet : packets) {
>+      EXPECT_TRUE(feedback.AddReceivedPacket(packet.sequence_number,
>+                                             packet.arrival_time_ms * 1000));
>+    }
>+
>+    feedback.Build();
>+
>+    clock_.AdvanceTimeMilliseconds(kFeedbackTimeoutMs - 1);
>+    PacketFeedback later_packet(kFeedbackTimeoutMs + 140,
>+                                kFeedbackTimeoutMs + 240, 5, 1500,
>+                                kPacingInfo1);
>+    OnSentPacket(later_packet);
>+
>+    controller_->OnTransportFeedback(feedback);
>+    ComparePacketFeedbackVectors(packets,
>+                                 controller_->GetTransportFeedbackVector());
>+  }
>+}
>+
>+// Bandwidth estimation is updated when feedbacks are received.
>+// Feedbacks which show an increasing delay cause the estimation to be reduced.
>+TEST_F(SendSideCongestionControllerTest, UpdatesDelayBasedEstimate) {
>+  TargetBitrateTrackingSetup();
>+
>+  const int64_t kRunTimeMs = 6000;
>+  uint16_t seq_num = 0;
>+
>+  // The test must run and insert packets/feedback long enough that the
>+  // BWE computes a valid estimate. This is first done in an environment which
>+  // simulates no bandwidth limitation, and therefore not built-up delay.
>+  PacketTransmissionAndFeedbackBlock(&seq_num, kRunTimeMs, 0);
>+  ASSERT_TRUE(target_bitrate_bps_);
>+
>+  // Repeat, but this time with a building delay, and make sure that the
>+  // estimation is adjusted downwards.
>+  uint32_t bitrate_before_delay = *target_bitrate_bps_;
>+  PacketTransmissionAndFeedbackBlock(&seq_num, kRunTimeMs, 50);
>+  EXPECT_LT(*target_bitrate_bps_, bitrate_before_delay);
>+}
>+
>+TEST_F(SendSideCongestionControllerTest, PacerQueueEncodeRatePushback) {
>+  ::webrtc::test::ScopedFieldTrials pushback_field_trial(
>+      "WebRTC-PacerPushbackExperiment/Enabled/");
>+  SetUp();
>+
>+  EXPECT_CALL(*pacer_, ExpectedQueueTimeMs()).WillOnce(Return(0));
>+  controller_->Process();
>+
>+  EXPECT_CALL(*pacer_, ExpectedQueueTimeMs()).WillOnce(Return(100));
>+  EXPECT_CALL(observer_, OnNetworkChanged(kInitialBitrateBps * 0.9, _, _, _));
>+  controller_->Process();
>+
>+  EXPECT_CALL(*pacer_, ExpectedQueueTimeMs()).WillOnce(Return(50));
>+  controller_->Process();
>+
>+  EXPECT_CALL(*pacer_, ExpectedQueueTimeMs()).WillOnce(Return(0));
>+  EXPECT_CALL(observer_, OnNetworkChanged(kInitialBitrateBps, _, _, _));
>+  controller_->Process();
>+
>+  const uint32_t kMinAdjustedBps = 50000;
>+  int expected_queue_threshold =
>+      1000 - kMinAdjustedBps * 1000.0 / kInitialBitrateBps;
>+
>+  EXPECT_CALL(*pacer_, ExpectedQueueTimeMs())
>+      .WillOnce(Return(expected_queue_threshold));
>+  EXPECT_CALL(observer_, OnNetworkChanged(Ge(kMinAdjustedBps), _, _, _));
>+  controller_->Process();
>+
>+  EXPECT_CALL(*pacer_, ExpectedQueueTimeMs())
>+      .WillOnce(Return(expected_queue_threshold + 1));
>+  EXPECT_CALL(observer_, OnNetworkChanged(0, _, _, _));
>+  controller_->Process();
>+
>+  EXPECT_CALL(*pacer_, ExpectedQueueTimeMs()).WillOnce(Return(0));
>+  EXPECT_CALL(observer_, OnNetworkChanged(kInitialBitrateBps, _, _, _));
>+  controller_->Process();
>+}
>+
>+}  // namespace test
>+}  // namespace webrtc_cc
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/send_time_history.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/send_time_history.cc
>new file mode 100644
>index 00000000000..73682cd5b24
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/send_time_history.cc
>@@ -0,0 +1,142 @@
>+/*
>+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/congestion_controller/rtp/send_time_history.h"
>+
>+#include <algorithm>
>+#include <utility>
>+
>+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>+#include "rtc_base/checks.h"
>+#include "system_wrappers/include/clock.h"
>+
>+namespace webrtc {
>+
>+SendTimeHistory::SendTimeHistory(const Clock* clock,
>+                                 int64_t packet_age_limit_ms)
>+    : clock_(clock), packet_age_limit_ms_(packet_age_limit_ms) {}
>+
>+SendTimeHistory::~SendTimeHistory() {}
>+
>+void SendTimeHistory::AddAndRemoveOld(const PacketFeedback& packet) {
>+  int64_t now_ms = clock_->TimeInMilliseconds();
>+  // Remove old.
>+  while (!history_.empty() &&
>+         now_ms - history_.begin()->second.creation_time_ms >
>+             packet_age_limit_ms_) {
>+    // TODO(sprang): Warn if erasing (too many) old items?
>+    RemovePacketBytes(history_.begin()->second);
>+    history_.erase(history_.begin());
>+  }
>+
>+  // Add new.
>+  int64_t unwrapped_seq_num = seq_num_unwrapper_.Unwrap(packet.sequence_number);
>+  PacketFeedback packet_copy = packet;
>+  packet_copy.long_sequence_number = unwrapped_seq_num;
>+  history_.insert(std::make_pair(unwrapped_seq_num, packet_copy));
>+  if (packet.send_time_ms >= 0)
>+    AddPacketBytes(packet_copy);
>+}
>+
>+bool SendTimeHistory::OnSentPacket(uint16_t sequence_number,
>+                                   int64_t send_time_ms) {
>+  int64_t unwrapped_seq_num = seq_num_unwrapper_.Unwrap(sequence_number);
>+  auto it = history_.find(unwrapped_seq_num);
>+  if (it == history_.end())
>+    return false;
>+  bool packet_retransmit = it->second.send_time_ms >= 0;
>+  it->second.send_time_ms = send_time_ms;
>+  if (!packet_retransmit)
>+    AddPacketBytes(it->second);
>+  return true;
>+}
>+
>+absl::optional<PacketFeedback> SendTimeHistory::GetPacket(
>+    uint16_t sequence_number) const {
>+  int64_t unwrapped_seq_num =
>+      seq_num_unwrapper_.UnwrapWithoutUpdate(sequence_number);
>+  absl::optional<PacketFeedback> optional_feedback;
>+  auto it = history_.find(unwrapped_seq_num);
>+  if (it != history_.end())
>+    optional_feedback.emplace(it->second);
>+  return optional_feedback;
>+}
>+
>+bool SendTimeHistory::GetFeedback(PacketFeedback* packet_feedback,
>+                                  bool remove) {
>+  RTC_DCHECK(packet_feedback);
>+  int64_t unwrapped_seq_num =
>+      seq_num_unwrapper_.Unwrap(packet_feedback->sequence_number);
>+  UpdateAckedSeqNum(unwrapped_seq_num);
>+  RTC_DCHECK_GE(*last_ack_seq_num_, 0);
>+  auto it = history_.find(unwrapped_seq_num);
>+  if (it == history_.end())
>+    return false;
>+
>+  // Save arrival_time not to overwrite it.
>+  int64_t arrival_time_ms = packet_feedback->arrival_time_ms;
>+  *packet_feedback = it->second;
>+  packet_feedback->arrival_time_ms = arrival_time_ms;
>+
>+  if (remove)
>+    history_.erase(it);
>+  return true;
>+}
>+
>+size_t SendTimeHistory::GetOutstandingBytes(uint16_t local_net_id,
>+                                            uint16_t remote_net_id) const {
>+  auto it = in_flight_bytes_.find({local_net_id, remote_net_id});
>+  if (it != in_flight_bytes_.end()) {
>+    return it->second;
>+  } else {
>+    return 0;
>+  }
>+}
>+
>+void SendTimeHistory::AddPacketBytes(const PacketFeedback& packet) {
>+  if (packet.send_time_ms < 0 || packet.payload_size == 0 ||
>+      (last_ack_seq_num_ && *last_ack_seq_num_ >= packet.long_sequence_number))
>+    return;
>+  auto it = in_flight_bytes_.find({packet.local_net_id, packet.remote_net_id});
>+  if (it != in_flight_bytes_.end()) {
>+    it->second += packet.payload_size;
>+  } else {
>+    in_flight_bytes_[{packet.local_net_id, packet.remote_net_id}] =
>+        packet.payload_size;
>+  }
>+}
>+
>+void SendTimeHistory::RemovePacketBytes(const PacketFeedback& packet) {
>+  if (packet.send_time_ms < 0 || packet.payload_size == 0 ||
>+      (last_ack_seq_num_ && *last_ack_seq_num_ >= packet.long_sequence_number))
>+    return;
>+  auto it = in_flight_bytes_.find({packet.local_net_id, packet.remote_net_id});
>+  if (it != in_flight_bytes_.end()) {
>+    it->second -= packet.payload_size;
>+    if (it->second == 0)
>+      in_flight_bytes_.erase(it);
>+  }
>+}
>+
>+void SendTimeHistory::UpdateAckedSeqNum(int64_t acked_seq_num) {
>+  if (last_ack_seq_num_ && *last_ack_seq_num_ >= acked_seq_num)
>+    return;
>+
>+  auto unacked_it = history_.begin();
>+  if (last_ack_seq_num_)
>+    unacked_it = history_.lower_bound(*last_ack_seq_num_);
>+
>+  auto newly_acked_end = history_.upper_bound(acked_seq_num);
>+  for (; unacked_it != newly_acked_end; ++unacked_it) {
>+    RemovePacketBytes(unacked_it->second);
>+  }
>+  last_ack_seq_num_.emplace(acked_seq_num);
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/include/send_time_history.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/send_time_history.h
>similarity index 71%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/include/send_time_history.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/send_time_history.h
>index 3946c4effa9..7cc6ad76414 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/include/send_time_history.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/send_time_history.h
>@@ -8,13 +8,13 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#ifndef MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_SEND_TIME_HISTORY_H_
>-#define MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_SEND_TIME_HISTORY_H_
>+#ifndef MODULES_CONGESTION_CONTROLLER_RTP_SEND_TIME_HISTORY_H_
>+#define MODULES_CONGESTION_CONTROLLER_RTP_SEND_TIME_HISTORY_H_
> 
> #include <map>
>+#include <utility>
> 
> #include "modules/include/module_common_types.h"
>-#include "rtc_base/basictypes.h"
> #include "rtc_base/constructormagic.h"
> 
> namespace webrtc {
>@@ -33,6 +33,9 @@ class SendTimeHistory {
>   // Return false if not found.
>   bool OnSentPacket(uint16_t sequence_number, int64_t send_time_ms);
> 
>+  // Retrieves packet info identified by |sequence_number|.
>+  absl::optional<PacketFeedback> GetPacket(uint16_t sequence_number) const;
>+
>   // Look up PacketFeedback for a sent packet, based on the sequence number, and
>   // populate all fields except for arrival_time. The packet parameter must
>   // thus be non-null and have the sequence_number field set.
>@@ -42,14 +45,20 @@ class SendTimeHistory {
>                              uint16_t remote_net_id) const;
> 
>  private:
>+  using RemoteAndLocalNetworkId = std::pair<uint16_t, uint16_t>;
>+
>+  void AddPacketBytes(const PacketFeedback& packet);
>+  void RemovePacketBytes(const PacketFeedback& packet);
>+  void UpdateAckedSeqNum(int64_t acked_seq_num);
>   const Clock* const clock_;
>   const int64_t packet_age_limit_ms_;
>   SequenceNumberUnwrapper seq_num_unwrapper_;
>   std::map<int64_t, PacketFeedback> history_;
>-  rtc::Optional<int64_t> latest_acked_seq_num_;
>+  absl::optional<int64_t> last_ack_seq_num_;
>+  std::map<RemoteAndLocalNetworkId, size_t> in_flight_bytes_;
> 
>   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(SendTimeHistory);
> };
> 
> }  // namespace webrtc
>-#endif  // MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_SEND_TIME_HISTORY_H_
>+#endif  // MODULES_CONGESTION_CONTROLLER_RTP_SEND_TIME_HISTORY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/send_time_history_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/send_time_history_unittest.cc
>similarity index 91%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/send_time_history_unittest.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/send_time_history_unittest.cc
>index 5e2ece6d601..3e852ba6311 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/send_time_history_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/send_time_history_unittest.cc
>@@ -13,7 +13,7 @@
> #include <random>
> #include <vector>
> 
>-#include "modules/remote_bitrate_estimator/include/send_time_history.h"
>+#include "modules/congestion_controller/rtp/send_time_history.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> #include "system_wrappers/include/clock.h"
> #include "test/gtest.h"
>@@ -52,8 +52,7 @@ TEST_F(SendTimeHistoryTest, SaveAndRestoreNetworkId) {
>   uint16_t sequence_number = 0;
>   int64_t now_ms = clock_.TimeInMilliseconds();
>   for (int i = 1; i < 5; ++i) {
>-    PacketFeedback packet(now_ms, sequence_number, 1000, i, i - 1,
>-                          kPacingInfo);
>+    PacketFeedback packet(now_ms, sequence_number, 1000, i, i - 1, kPacingInfo);
>     history_.AddAndRemoveOld(packet);
>     history_.OnSentPacket(sequence_number, now_ms);
>     PacketFeedback restored(now_ms, sequence_number);
>@@ -82,6 +81,26 @@ TEST_F(SendTimeHistoryTest, AddRemoveOne) {
>   EXPECT_FALSE(history_.GetFeedback(&received_packet3, true));
> }
> 
>+TEST_F(SendTimeHistoryTest, GetPacketReturnsSentPacket) {
>+  const uint16_t kSeqNo = 10;
>+  const PacedPacketInfo kPacingInfo(0, 5, 1200);
>+  const PacketFeedback kSentPacket(0, -1, 1, kSeqNo, 123, 0, 0, kPacingInfo);
>+  AddPacketWithSendTime(kSeqNo, 123, 1, kPacingInfo);
>+  auto sent_packet = history_.GetPacket(kSeqNo);
>+  EXPECT_EQ(kSentPacket, *sent_packet);
>+}
>+
>+TEST_F(SendTimeHistoryTest, GetPacketEmptyForRemovedPacket) {
>+  const uint16_t kSeqNo = 10;
>+  const PacedPacketInfo kPacingInfo(0, 5, 1200);
>+  AddPacketWithSendTime(kSeqNo, 123, 1, kPacingInfo);
>+  auto sent_packet = history_.GetPacket(kSeqNo);
>+  PacketFeedback received_packet(0, 0, kSeqNo, 0, kPacingInfo);
>+  EXPECT_TRUE(history_.GetFeedback(&received_packet, true));
>+  sent_packet = history_.GetPacket(kSeqNo);
>+  EXPECT_FALSE(sent_packet.has_value());
>+}
>+
> TEST_F(SendTimeHistoryTest, PopulatesExpectedFields) {
>   const uint16_t kSeqNo = 10;
>   const int64_t kSendTime = 1000;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/transport_feedback_adapter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/transport_feedback_adapter.cc
>new file mode 100644
>index 00000000000..d1b4855e369
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/transport_feedback_adapter.cc
>@@ -0,0 +1,193 @@
>+/*
>+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/congestion_controller/rtp/transport_feedback_adapter.h"
>+
>+#include <algorithm>
>+
>+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>+#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/logging.h"
>+#include "rtc_base/numerics/mod_ops.h"
>+
>+namespace webrtc {
>+namespace webrtc_cc {
>+
>+const int64_t kNoTimestamp = -1;
>+const int64_t kSendTimeHistoryWindowMs = 60000;
>+const int64_t kBaseTimestampScaleFactor =
>+    rtcp::TransportFeedback::kDeltaScaleFactor * (1 << 8);
>+const int64_t kBaseTimestampRangeSizeUs = kBaseTimestampScaleFactor * (1 << 24);
>+
>+TransportFeedbackAdapter::TransportFeedbackAdapter(const Clock* clock)
>+    : send_time_history_(clock, kSendTimeHistoryWindowMs),
>+      clock_(clock),
>+      current_offset_ms_(kNoTimestamp),
>+      last_timestamp_us_(kNoTimestamp),
>+      local_net_id_(0),
>+      remote_net_id_(0) {}
>+
>+TransportFeedbackAdapter::~TransportFeedbackAdapter() {
>+  RTC_DCHECK(observers_.empty());
>+}
>+
>+void TransportFeedbackAdapter::RegisterPacketFeedbackObserver(
>+    PacketFeedbackObserver* observer) {
>+  rtc::CritScope cs(&observers_lock_);
>+  RTC_DCHECK(observer);
>+  RTC_DCHECK(std::find(observers_.begin(), observers_.end(), observer) ==
>+             observers_.end());
>+  observers_.push_back(observer);
>+}
>+
>+void TransportFeedbackAdapter::DeRegisterPacketFeedbackObserver(
>+    PacketFeedbackObserver* observer) {
>+  rtc::CritScope cs(&observers_lock_);
>+  RTC_DCHECK(observer);
>+  const auto it = std::find(observers_.begin(), observers_.end(), observer);
>+  RTC_DCHECK(it != observers_.end());
>+  observers_.erase(it);
>+}
>+
>+void TransportFeedbackAdapter::AddPacket(uint32_t ssrc,
>+                                         uint16_t sequence_number,
>+                                         size_t length,
>+                                         const PacedPacketInfo& pacing_info) {
>+  {
>+    rtc::CritScope cs(&lock_);
>+    const int64_t creation_time_ms = clock_->TimeInMilliseconds();
>+    send_time_history_.AddAndRemoveOld(
>+        PacketFeedback(creation_time_ms, sequence_number, length, local_net_id_,
>+                       remote_net_id_, pacing_info));
>+  }
>+
>+  {
>+    rtc::CritScope cs(&observers_lock_);
>+    for (auto* observer : observers_) {
>+      observer->OnPacketAdded(ssrc, sequence_number);
>+    }
>+  }
>+}
>+
>+void TransportFeedbackAdapter::OnSentPacket(uint16_t sequence_number,
>+                                            int64_t send_time_ms) {
>+  rtc::CritScope cs(&lock_);
>+  send_time_history_.OnSentPacket(sequence_number, send_time_ms);
>+}
>+
>+absl::optional<PacketFeedback> TransportFeedbackAdapter::GetPacket(
>+    uint16_t sequence_number) const {
>+  rtc::CritScope cs(&lock_);
>+  return send_time_history_.GetPacket(sequence_number);
>+}
>+
>+void TransportFeedbackAdapter::SetNetworkIds(uint16_t local_id,
>+                                             uint16_t remote_id) {
>+  rtc::CritScope cs(&lock_);
>+  local_net_id_ = local_id;
>+  remote_net_id_ = remote_id;
>+}
>+
>+std::vector<PacketFeedback> TransportFeedbackAdapter::GetPacketFeedbackVector(
>+    const rtcp::TransportFeedback& feedback) {
>+  int64_t timestamp_us = feedback.GetBaseTimeUs();
>+  int64_t now_ms = clock_->TimeInMilliseconds();
>+  // Add timestamp deltas to a local time base selected on first packet arrival.
>+  // This won't be the true time base, but makes it easier to manually inspect
>+  // time stamps.
>+  if (last_timestamp_us_ == kNoTimestamp) {
>+    current_offset_ms_ = now_ms;
>+  } else {
>+    int64_t delta = timestamp_us - last_timestamp_us_;
>+
>+    // Detect and compensate for wrap-arounds in base time.
>+    if (std::abs(delta - kBaseTimestampRangeSizeUs) < std::abs(delta)) {
>+      delta -= kBaseTimestampRangeSizeUs;  // Wrap backwards.
>+    } else if (std::abs(delta + kBaseTimestampRangeSizeUs) < std::abs(delta)) {
>+      delta += kBaseTimestampRangeSizeUs;  // Wrap forwards.
>+    }
>+
>+    current_offset_ms_ += delta / 1000;
>+  }
>+  last_timestamp_us_ = timestamp_us;
>+
>+  std::vector<PacketFeedback> packet_feedback_vector;
>+  if (feedback.GetPacketStatusCount() == 0) {
>+    RTC_LOG(LS_INFO) << "Empty transport feedback packet received.";
>+    return packet_feedback_vector;
>+  }
>+  packet_feedback_vector.reserve(feedback.GetPacketStatusCount());
>+  {
>+    rtc::CritScope cs(&lock_);
>+    size_t failed_lookups = 0;
>+    int64_t offset_us = 0;
>+    int64_t timestamp_ms = 0;
>+    uint16_t seq_num = feedback.GetBaseSequence();
>+    for (const auto& packet : feedback.GetReceivedPackets()) {
>+      // Insert into the vector those unreceived packets which precede this
>+      // iteration's received packet.
>+      for (; seq_num != packet.sequence_number(); ++seq_num) {
>+        PacketFeedback packet_feedback(PacketFeedback::kNotReceived, seq_num);
>+        // Note: Element not removed from history because it might be reported
>+        // as received by another feedback.
>+        if (!send_time_history_.GetFeedback(&packet_feedback, false))
>+          ++failed_lookups;
>+        if (packet_feedback.local_net_id == local_net_id_ &&
>+            packet_feedback.remote_net_id == remote_net_id_) {
>+          packet_feedback_vector.push_back(packet_feedback);
>+        }
>+      }
>+
>+      // Handle this iteration's received packet.
>+      offset_us += packet.delta_us();
>+      timestamp_ms = current_offset_ms_ + (offset_us / 1000);
>+      PacketFeedback packet_feedback(timestamp_ms, packet.sequence_number());
>+      if (!send_time_history_.GetFeedback(&packet_feedback, true))
>+        ++failed_lookups;
>+      if (packet_feedback.local_net_id == local_net_id_ &&
>+          packet_feedback.remote_net_id == remote_net_id_) {
>+        packet_feedback_vector.push_back(packet_feedback);
>+      }
>+
>+      ++seq_num;
>+    }
>+
>+    if (failed_lookups > 0) {
>+      RTC_LOG(LS_WARNING) << "Failed to lookup send time for " << failed_lookups
>+                          << " packet" << (failed_lookups > 1 ? "s" : "")
>+                          << ". Send time history too small?";
>+    }
>+  }
>+  return packet_feedback_vector;
>+}
>+
>+void TransportFeedbackAdapter::OnTransportFeedback(
>+    const rtcp::TransportFeedback& feedback) {
>+  last_packet_feedback_vector_ = GetPacketFeedbackVector(feedback);
>+  {
>+    rtc::CritScope cs(&observers_lock_);
>+    for (auto* observer : observers_) {
>+      observer->OnPacketFeedbackVector(last_packet_feedback_vector_);
>+    }
>+  }
>+}
>+
>+std::vector<PacketFeedback>
>+TransportFeedbackAdapter::GetTransportFeedbackVector() const {
>+  return last_packet_feedback_vector_;
>+}
>+
>+size_t TransportFeedbackAdapter::GetOutstandingBytes() const {
>+  rtc::CritScope cs(&lock_);
>+  return send_time_history_.GetOutstandingBytes(local_net_id_, remote_net_id_);
>+}
>+}  // namespace webrtc_cc
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/transport_feedback_adapter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/transport_feedback_adapter.h
>new file mode 100644
>index 00000000000..ae38b846fc8
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/transport_feedback_adapter.h
>@@ -0,0 +1,80 @@
>+/*
>+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_CONGESTION_CONTROLLER_RTP_TRANSPORT_FEEDBACK_ADAPTER_H_
>+#define MODULES_CONGESTION_CONTROLLER_RTP_TRANSPORT_FEEDBACK_ADAPTER_H_
>+
>+#include <deque>
>+#include <vector>
>+
>+#include "modules/congestion_controller/rtp/send_time_history.h"
>+#include "rtc_base/criticalsection.h"
>+#include "rtc_base/thread_annotations.h"
>+#include "rtc_base/thread_checker.h"
>+#include "system_wrappers/include/clock.h"
>+
>+namespace webrtc {
>+
>+class PacketFeedbackObserver;
>+
>+namespace rtcp {
>+class TransportFeedback;
>+}  // namespace rtcp
>+
>+namespace webrtc_cc {
>+class TransportFeedbackAdapter {
>+ public:
>+  explicit TransportFeedbackAdapter(const Clock* clock);
>+  virtual ~TransportFeedbackAdapter();
>+
>+  void RegisterPacketFeedbackObserver(PacketFeedbackObserver* observer);
>+  void DeRegisterPacketFeedbackObserver(PacketFeedbackObserver* observer);
>+
>+  void AddPacket(uint32_t ssrc,
>+                 uint16_t sequence_number,
>+                 size_t length,
>+                 const PacedPacketInfo& pacing_info);
>+  void OnSentPacket(uint16_t sequence_number, int64_t send_time_ms);
>+
>+  // TODO(holmer): This method should return DelayBasedBwe::Result so that we
>+  // can get rid of the dependency on BitrateController. Requires changes
>+  // to the CongestionController interface.
>+  void OnTransportFeedback(const rtcp::TransportFeedback& feedback);
>+  std::vector<PacketFeedback> GetTransportFeedbackVector() const;
>+  absl::optional<PacketFeedback> GetPacket(uint16_t sequence_number) const;
>+
>+  void SetTransportOverhead(int transport_overhead_bytes_per_packet);
>+
>+  void SetNetworkIds(uint16_t local_id, uint16_t remote_id);
>+
>+  size_t GetOutstandingBytes() const;
>+
>+ private:
>+  std::vector<PacketFeedback> GetPacketFeedbackVector(
>+      const rtcp::TransportFeedback& feedback);
>+
>+  rtc::CriticalSection lock_;
>+  SendTimeHistory send_time_history_ RTC_GUARDED_BY(&lock_);
>+  const Clock* const clock_;
>+  int64_t current_offset_ms_;
>+  int64_t last_timestamp_us_;
>+  std::vector<PacketFeedback> last_packet_feedback_vector_;
>+  uint16_t local_net_id_ RTC_GUARDED_BY(&lock_);
>+  uint16_t remote_net_id_ RTC_GUARDED_BY(&lock_);
>+
>+  rtc::CriticalSection observers_lock_;
>+  std::vector<PacketFeedbackObserver*> observers_
>+      RTC_GUARDED_BY(&observers_lock_);
>+};
>+
>+}  // namespace webrtc_cc
>+}  // namespace webrtc
>+
>+#endif  // MODULES_CONGESTION_CONTROLLER_RTP_TRANSPORT_FEEDBACK_ADAPTER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/transport_feedback_adapter_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/transport_feedback_adapter_unittest.cc
>new file mode 100644
>index 00000000000..70087678a23
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/rtp/transport_feedback_adapter_unittest.cc
>@@ -0,0 +1,390 @@
>+/*
>+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <limits>
>+#include <memory>
>+#include <vector>
>+
>+#include "modules/bitrate_controller/include/mock/mock_bitrate_controller.h"
>+#include "modules/congestion_controller/rtp/congestion_controller_unittests_helper.h"
>+#include "modules/congestion_controller/rtp/transport_feedback_adapter.h"
>+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>+#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/numerics/safe_conversions.h"
>+#include "system_wrappers/include/clock.h"
>+#include "test/gmock.h"
>+#include "test/gtest.h"
>+
>+using ::testing::_;
>+using ::testing::Invoke;
>+
>+namespace webrtc {
>+namespace webrtc_cc {
>+
>+namespace {
>+const PacedPacketInfo kPacingInfo0(0, 5, 2000);
>+const PacedPacketInfo kPacingInfo1(1, 8, 4000);
>+const PacedPacketInfo kPacingInfo2(2, 14, 7000);
>+const PacedPacketInfo kPacingInfo3(3, 20, 10000);
>+const PacedPacketInfo kPacingInfo4(4, 22, 10000);
>+}  // namespace
>+
>+namespace test {
>+
>+class MockPacketFeedbackObserver : public webrtc::PacketFeedbackObserver {
>+ public:
>+  MOCK_METHOD2(OnPacketAdded, void(uint32_t ssrc, uint16_t seq_num));
>+  MOCK_METHOD1(OnPacketFeedbackVector,
>+               void(const std::vector<PacketFeedback>& packet_feedback_vector));
>+};
>+
>+class TransportFeedbackAdapterTest : public ::testing::Test {
>+ public:
>+  TransportFeedbackAdapterTest() : clock_(0) {}
>+
>+  virtual ~TransportFeedbackAdapterTest() {}
>+
>+  virtual void SetUp() {
>+    adapter_.reset(new TransportFeedbackAdapter(&clock_));
>+  }
>+
>+  virtual void TearDown() { adapter_.reset(); }
>+
>+ protected:
>+  void OnReceivedEstimatedBitrate(uint32_t bitrate) {}
>+
>+  void OnReceivedRtcpReceiverReport(const ReportBlockList& report_blocks,
>+                                    int64_t rtt,
>+                                    int64_t now_ms) {}
>+
>+  void OnSentPacket(const PacketFeedback& packet_feedback) {
>+    adapter_->AddPacket(kSsrc, packet_feedback.sequence_number,
>+                        packet_feedback.payload_size,
>+                        packet_feedback.pacing_info);
>+    adapter_->OnSentPacket(packet_feedback.sequence_number,
>+                           packet_feedback.send_time_ms);
>+  }
>+
>+  static constexpr uint32_t kSsrc = 8492;
>+
>+  SimulatedClock clock_;
>+  std::unique_ptr<TransportFeedbackAdapter> adapter_;
>+};
>+
>+TEST_F(TransportFeedbackAdapterTest, ObserverSanity) {
>+  MockPacketFeedbackObserver mock;
>+  adapter_->RegisterPacketFeedbackObserver(&mock);
>+
>+  const std::vector<PacketFeedback> packets = {
>+      PacketFeedback(100, 200, 0, 1000, kPacingInfo0),
>+      PacketFeedback(110, 210, 1, 2000, kPacingInfo0),
>+      PacketFeedback(120, 220, 2, 3000, kPacingInfo0)};
>+
>+  rtcp::TransportFeedback feedback;
>+  feedback.SetBase(packets[0].sequence_number,
>+                   packets[0].arrival_time_ms * 1000);
>+
>+  for (const PacketFeedback& packet : packets) {
>+    EXPECT_CALL(mock, OnPacketAdded(kSsrc, packet.sequence_number)).Times(1);
>+    OnSentPacket(packet);
>+    EXPECT_TRUE(feedback.AddReceivedPacket(packet.sequence_number,
>+                                           packet.arrival_time_ms * 1000));
>+  }
>+
>+  EXPECT_CALL(mock, OnPacketFeedbackVector(_)).Times(1);
>+  adapter_->OnTransportFeedback(feedback);
>+
>+  adapter_->DeRegisterPacketFeedbackObserver(&mock);
>+
>+  // After deregistration, the observer no longers gets indications.
>+  EXPECT_CALL(mock, OnPacketAdded(_, _)).Times(0);
>+  const PacketFeedback new_packet(130, 230, 3, 4000, kPacingInfo0);
>+  OnSentPacket(new_packet);
>+
>+  rtcp::TransportFeedback second_feedback;
>+  second_feedback.SetBase(new_packet.sequence_number,
>+                          new_packet.arrival_time_ms * 1000);
>+  EXPECT_TRUE(feedback.AddReceivedPacket(new_packet.sequence_number,
>+                                         new_packet.arrival_time_ms * 1000));
>+  EXPECT_CALL(mock, OnPacketFeedbackVector(_)).Times(0);
>+  adapter_->OnTransportFeedback(second_feedback);
>+}
>+
>+#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
>+TEST_F(TransportFeedbackAdapterTest, ObserverDoubleRegistrationDeathTest) {
>+  MockPacketFeedbackObserver mock;
>+  adapter_->RegisterPacketFeedbackObserver(&mock);
>+  EXPECT_DEATH(adapter_->RegisterPacketFeedbackObserver(&mock), "");
>+  adapter_->DeRegisterPacketFeedbackObserver(&mock);
>+}
>+
>+TEST_F(TransportFeedbackAdapterTest, ObserverMissingDeRegistrationDeathTest) {
>+  MockPacketFeedbackObserver mock;
>+  adapter_->RegisterPacketFeedbackObserver(&mock);
>+  EXPECT_DEATH(adapter_.reset(), "");
>+  adapter_->DeRegisterPacketFeedbackObserver(&mock);
>+}
>+#endif
>+
>+TEST_F(TransportFeedbackAdapterTest, AdaptsFeedbackAndPopulatesSendTimes) {
>+  std::vector<PacketFeedback> packets;
>+  packets.push_back(PacketFeedback(100, 200, 0, 1500, kPacingInfo0));
>+  packets.push_back(PacketFeedback(110, 210, 1, 1500, kPacingInfo0));
>+  packets.push_back(PacketFeedback(120, 220, 2, 1500, kPacingInfo0));
>+  packets.push_back(PacketFeedback(130, 230, 3, 1500, kPacingInfo1));
>+  packets.push_back(PacketFeedback(140, 240, 4, 1500, kPacingInfo1));
>+
>+  for (const PacketFeedback& packet : packets)
>+    OnSentPacket(packet);
>+
>+  rtcp::TransportFeedback feedback;
>+  feedback.SetBase(packets[0].sequence_number,
>+                   packets[0].arrival_time_ms * 1000);
>+
>+  for (const PacketFeedback& packet : packets) {
>+    EXPECT_TRUE(feedback.AddReceivedPacket(packet.sequence_number,
>+                                           packet.arrival_time_ms * 1000));
>+  }
>+
>+  feedback.Build();
>+
>+  adapter_->OnTransportFeedback(feedback);
>+  ComparePacketFeedbackVectors(packets, adapter_->GetTransportFeedbackVector());
>+}
>+
>+TEST_F(TransportFeedbackAdapterTest, FeedbackVectorReportsUnreceived) {
>+  std::vector<PacketFeedback> sent_packets = {
>+      PacketFeedback(100, 220, 0, 1500, kPacingInfo0),
>+      PacketFeedback(110, 210, 1, 1500, kPacingInfo0),
>+      PacketFeedback(120, 220, 2, 1500, kPacingInfo0),
>+      PacketFeedback(130, 230, 3, 1500, kPacingInfo0),
>+      PacketFeedback(140, 240, 4, 1500, kPacingInfo0),
>+      PacketFeedback(150, 250, 5, 1500, kPacingInfo0),
>+      PacketFeedback(160, 260, 6, 1500, kPacingInfo0)};
>+
>+  for (const PacketFeedback& packet : sent_packets)
>+    OnSentPacket(packet);
>+
>+  // Note: Important to include the last packet, as only unreceived packets in
>+  // between received packets can be inferred.
>+  std::vector<PacketFeedback> received_packets = {
>+      sent_packets[0], sent_packets[2], sent_packets[6]};
>+
>+  rtcp::TransportFeedback feedback;
>+  feedback.SetBase(received_packets[0].sequence_number,
>+                   received_packets[0].arrival_time_ms * 1000);
>+
>+  for (const PacketFeedback& packet : received_packets) {
>+    EXPECT_TRUE(feedback.AddReceivedPacket(packet.sequence_number,
>+                                           packet.arrival_time_ms * 1000));
>+  }
>+
>+  feedback.Build();
>+
>+  adapter_->OnTransportFeedback(feedback);
>+  ComparePacketFeedbackVectors(sent_packets,
>+                               adapter_->GetTransportFeedbackVector());
>+}
>+
>+TEST_F(TransportFeedbackAdapterTest, HandlesDroppedPackets) {
>+  std::vector<PacketFeedback> packets;
>+  packets.push_back(PacketFeedback(100, 200, 0, 1500, kPacingInfo0));
>+  packets.push_back(PacketFeedback(110, 210, 1, 1500, kPacingInfo1));
>+  packets.push_back(PacketFeedback(120, 220, 2, 1500, kPacingInfo2));
>+  packets.push_back(PacketFeedback(130, 230, 3, 1500, kPacingInfo3));
>+  packets.push_back(PacketFeedback(140, 240, 4, 1500, kPacingInfo4));
>+
>+  const uint16_t kSendSideDropBefore = 1;
>+  const uint16_t kReceiveSideDropAfter = 3;
>+
>+  for (const PacketFeedback& packet : packets) {
>+    if (packet.sequence_number >= kSendSideDropBefore)
>+      OnSentPacket(packet);
>+  }
>+
>+  rtcp::TransportFeedback feedback;
>+  feedback.SetBase(packets[0].sequence_number,
>+                   packets[0].arrival_time_ms * 1000);
>+
>+  for (const PacketFeedback& packet : packets) {
>+    if (packet.sequence_number <= kReceiveSideDropAfter) {
>+      EXPECT_TRUE(feedback.AddReceivedPacket(packet.sequence_number,
>+                                             packet.arrival_time_ms * 1000));
>+    }
>+  }
>+
>+  feedback.Build();
>+
>+  std::vector<PacketFeedback> expected_packets(
>+      packets.begin(), packets.begin() + kReceiveSideDropAfter + 1);
>+  // Packets that have timed out on the send-side have lost the
>+  // information stored on the send-side.
>+  for (size_t i = 0; i < kSendSideDropBefore; ++i) {
>+    expected_packets[i].send_time_ms = -1;
>+    expected_packets[i].payload_size = 0;
>+    expected_packets[i].pacing_info = PacedPacketInfo();
>+  }
>+
>+  adapter_->OnTransportFeedback(feedback);
>+  ComparePacketFeedbackVectors(expected_packets,
>+                               adapter_->GetTransportFeedbackVector());
>+}
>+
>+TEST_F(TransportFeedbackAdapterTest, SendTimeWrapsBothWays) {
>+  int64_t kHighArrivalTimeMs = rtcp::TransportFeedback::kDeltaScaleFactor *
>+                               static_cast<int64_t>(1 << 8) *
>+                               static_cast<int64_t>((1 << 23) - 1) / 1000;
>+  std::vector<PacketFeedback> packets;
>+  packets.push_back(
>+      PacketFeedback(kHighArrivalTimeMs - 64, 200, 0, 1500, PacedPacketInfo()));
>+  packets.push_back(
>+      PacketFeedback(kHighArrivalTimeMs + 64, 210, 1, 1500, PacedPacketInfo()));
>+  packets.push_back(
>+      PacketFeedback(kHighArrivalTimeMs, 220, 2, 1500, PacedPacketInfo()));
>+
>+  for (const PacketFeedback& packet : packets)
>+    OnSentPacket(packet);
>+
>+  for (size_t i = 0; i < packets.size(); ++i) {
>+    std::unique_ptr<rtcp::TransportFeedback> feedback(
>+        new rtcp::TransportFeedback());
>+    feedback->SetBase(packets[i].sequence_number,
>+                      packets[i].arrival_time_ms * 1000);
>+
>+    EXPECT_TRUE(feedback->AddReceivedPacket(packets[i].sequence_number,
>+                                            packets[i].arrival_time_ms * 1000));
>+
>+    rtc::Buffer raw_packet = feedback->Build();
>+    feedback = rtcp::TransportFeedback::ParseFrom(raw_packet.data(),
>+                                                  raw_packet.size());
>+
>+    std::vector<PacketFeedback> expected_packets;
>+    expected_packets.push_back(packets[i]);
>+
>+    adapter_->OnTransportFeedback(*feedback.get());
>+    ComparePacketFeedbackVectors(expected_packets,
>+                                 adapter_->GetTransportFeedbackVector());
>+  }
>+}
>+
>+TEST_F(TransportFeedbackAdapterTest, HandlesArrivalReordering) {
>+  std::vector<PacketFeedback> packets;
>+  packets.push_back(PacketFeedback(120, 200, 0, 1500, kPacingInfo0));
>+  packets.push_back(PacketFeedback(110, 210, 1, 1500, kPacingInfo0));
>+  packets.push_back(PacketFeedback(100, 220, 2, 1500, kPacingInfo0));
>+
>+  for (const PacketFeedback& packet : packets)
>+    OnSentPacket(packet);
>+
>+  rtcp::TransportFeedback feedback;
>+  feedback.SetBase(packets[0].sequence_number,
>+                   packets[0].arrival_time_ms * 1000);
>+
>+  for (const PacketFeedback& packet : packets) {
>+    EXPECT_TRUE(feedback.AddReceivedPacket(packet.sequence_number,
>+                                           packet.arrival_time_ms * 1000));
>+  }
>+
>+  feedback.Build();
>+
>+  // Adapter keeps the packets ordered by sequence number (which is itself
>+  // assigned by the order of transmission). Reordering by some other criteria,
>+  // eg. arrival time, is up to the observers.
>+  adapter_->OnTransportFeedback(feedback);
>+  ComparePacketFeedbackVectors(packets, adapter_->GetTransportFeedbackVector());
>+}
>+
>+TEST_F(TransportFeedbackAdapterTest, TimestampDeltas) {
>+  std::vector<PacketFeedback> sent_packets;
>+  const int64_t kSmallDeltaUs =
>+      rtcp::TransportFeedback::kDeltaScaleFactor * ((1 << 8) - 1);
>+  const int64_t kLargePositiveDeltaUs =
>+      rtcp::TransportFeedback::kDeltaScaleFactor *
>+      std::numeric_limits<int16_t>::max();
>+  const int64_t kLargeNegativeDeltaUs =
>+      rtcp::TransportFeedback::kDeltaScaleFactor *
>+      std::numeric_limits<int16_t>::min();
>+
>+  PacketFeedback packet_feedback(100, 200, 0, 1500, true, 0, 0,
>+                                 PacedPacketInfo());
>+  sent_packets.push_back(packet_feedback);
>+
>+  packet_feedback.send_time_ms += kSmallDeltaUs / 1000;
>+  packet_feedback.arrival_time_ms += kSmallDeltaUs / 1000;
>+  ++packet_feedback.sequence_number;
>+  sent_packets.push_back(packet_feedback);
>+
>+  packet_feedback.send_time_ms += kLargePositiveDeltaUs / 1000;
>+  packet_feedback.arrival_time_ms += kLargePositiveDeltaUs / 1000;
>+  ++packet_feedback.sequence_number;
>+  sent_packets.push_back(packet_feedback);
>+
>+  packet_feedback.send_time_ms += kLargeNegativeDeltaUs / 1000;
>+  packet_feedback.arrival_time_ms += kLargeNegativeDeltaUs / 1000;
>+  ++packet_feedback.sequence_number;
>+  sent_packets.push_back(packet_feedback);
>+
>+  // Too large, delta - will need two feedback messages.
>+  packet_feedback.send_time_ms += (kLargePositiveDeltaUs + 1000) / 1000;
>+  packet_feedback.arrival_time_ms += (kLargePositiveDeltaUs + 1000) / 1000;
>+  ++packet_feedback.sequence_number;
>+
>+  // Packets will be added to send history.
>+  for (const PacketFeedback& packet : sent_packets)
>+    OnSentPacket(packet);
>+  OnSentPacket(packet_feedback);
>+
>+  // Create expected feedback and send into adapter.
>+  std::unique_ptr<rtcp::TransportFeedback> feedback(
>+      new rtcp::TransportFeedback());
>+  feedback->SetBase(sent_packets[0].sequence_number,
>+                    sent_packets[0].arrival_time_ms * 1000);
>+
>+  for (const PacketFeedback& packet : sent_packets) {
>+    EXPECT_TRUE(feedback->AddReceivedPacket(packet.sequence_number,
>+                                            packet.arrival_time_ms * 1000));
>+  }
>+  EXPECT_FALSE(feedback->AddReceivedPacket(
>+      packet_feedback.sequence_number, packet_feedback.arrival_time_ms * 1000));
>+
>+  rtc::Buffer raw_packet = feedback->Build();
>+  feedback =
>+      rtcp::TransportFeedback::ParseFrom(raw_packet.data(), raw_packet.size());
>+
>+  std::vector<PacketFeedback> received_feedback;
>+
>+  EXPECT_TRUE(feedback.get() != nullptr);
>+  adapter_->OnTransportFeedback(*feedback.get());
>+  ComparePacketFeedbackVectors(sent_packets,
>+                               adapter_->GetTransportFeedbackVector());
>+
>+  // Create a new feedback message and add the trailing item.
>+  feedback.reset(new rtcp::TransportFeedback());
>+  feedback->SetBase(packet_feedback.sequence_number,
>+                    packet_feedback.arrival_time_ms * 1000);
>+  EXPECT_TRUE(feedback->AddReceivedPacket(
>+      packet_feedback.sequence_number, packet_feedback.arrival_time_ms * 1000));
>+  raw_packet = feedback->Build();
>+  feedback =
>+      rtcp::TransportFeedback::ParseFrom(raw_packet.data(), raw_packet.size());
>+
>+  EXPECT_TRUE(feedback.get() != nullptr);
>+  adapter_->OnTransportFeedback(*feedback.get());
>+  {
>+    std::vector<PacketFeedback> expected_packets;
>+    expected_packets.push_back(packet_feedback);
>+    ComparePacketFeedbackVectors(expected_packets,
>+                                 adapter_->GetTransportFeedbackVector());
>+  }
>+}
>+}  // namespace test
>+}  // namespace webrtc_cc
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/send_side_congestion_controller.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/send_side_congestion_controller.cc
>index cb3e610e38f..0439c1d567b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/send_side_congestion_controller.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/send_side_congestion_controller.cc
>@@ -11,19 +11,21 @@
> #include "modules/congestion_controller/include/send_side_congestion_controller.h"
> 
> #include <algorithm>
>+#include <cstdio>
>+#include <iterator>
> #include <memory>
> #include <vector>
> 
>+#include "absl/memory/memory.h"
> #include "modules/bitrate_controller/include/bitrate_controller.h"
>-#include "modules/congestion_controller/acknowledged_bitrate_estimator.h"
>-#include "modules/congestion_controller/probe_controller.h"
>-#include "modules/pacing/alr_detector.h"
>+#include "modules/congestion_controller/congestion_window_pushback_controller.h"
>+#include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h"
>+#include "modules/congestion_controller/goog_cc/probe_controller.h"
> #include "modules/remote_bitrate_estimator/include/bwe_defines.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/format_macros.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/numerics/safe_conversions.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/rate_limiter.h"
> #include "rtc_base/socket.h"
> #include "rtc_base/timeutils.h"
>@@ -35,6 +37,12 @@ namespace {
> 
> const char kCwndExperiment[] = "WebRTC-CwndExperiment";
> const char kPacerPushbackExperiment[] = "WebRTC-PacerPushbackExperiment";
>+
>+// When CongestionWindowPushback is enabled, the pacer is oblivious to
>+// the congestion window. The relation between outstanding data and
>+// the congestion window affects encoder allocations directly.
>+const char kCongestionPushbackExperiment[] = "WebRTC-CongestionWindowPushback";
>+
> const int64_t kDefaultAcceptedQueueMs = 250;
> 
> bool CwndExperimentEnabled() {
>@@ -58,6 +66,18 @@ bool ReadCwndExperimentParameter(int64_t* accepted_queue_ms) {
>   return false;
> }
> 
>+bool IsCongestionWindowPushbackExperimentEnabled() {
>+  return webrtc::field_trial::IsEnabled(kCongestionPushbackExperiment) &&
>+         webrtc::field_trial::IsEnabled(kCwndExperiment);
>+}
>+
>+std::unique_ptr<CongestionWindowPushbackController>
>+MaybeCreateCongestionWindowPushbackController() {
>+  return IsCongestionWindowPushbackExperimentEnabled()
>+             ? absl::make_unique<CongestionWindowPushbackController>()
>+             : nullptr;
>+}
>+
> static const int64_t kRetransmitWindowSizeMs = 500;
> 
> // Makes sure that the bitrate and the min, max values are in valid range.
>@@ -95,10 +115,10 @@ void SortPacketFeedbackVector(
> }
> 
> bool IsPacerPushbackExperimentEnabled() {
>-  return webrtc::field_trial::IsEnabled(kPacerPushbackExperiment) || (
>-      !webrtc::field_trial::IsDisabled(kPacerPushbackExperiment) &&
>-      webrtc::runtime_enabled_features::IsFeatureEnabled(
>-          webrtc::runtime_enabled_features::kDualStreamModeFeatureName));
>+  return webrtc::field_trial::IsEnabled(kPacerPushbackExperiment) ||
>+         (!webrtc::field_trial::IsDisabled(kPacerPushbackExperiment) &&
>+          webrtc::runtime_enabled_features::IsFeatureEnabled(
>+              webrtc::runtime_enabled_features::kDualStreamModeFeatureName));
> }
> 
> }  // namespace
>@@ -115,8 +135,8 @@ SendSideCongestionController::SendSideCongestionController(
>       bitrate_controller_(
>           BitrateController::CreateBitrateController(clock_, event_log)),
>       acknowledged_bitrate_estimator_(
>-          rtc::MakeUnique<AcknowledgedBitrateEstimator>()),
>-      probe_controller_(new ProbeController(pacer_, clock_)),
>+          absl::make_unique<AcknowledgedBitrateEstimator>()),
>+      probe_controller_(new ProbeController()),
>       retransmission_rate_limiter_(
>           new RateLimiter(clock, kRetransmitWindowSizeMs)),
>       transport_feedback_adapter_(clock_),
>@@ -127,12 +147,16 @@ SendSideCongestionController::SendSideCongestionController(
>       pause_pacer_(false),
>       pacer_paused_(false),
>       min_bitrate_bps_(congestion_controller::GetMinBitrateBps()),
>-      delay_based_bwe_(new DelayBasedBwe(event_log_, clock_)),
>+      delay_based_bwe_(new DelayBasedBwe(event_log_)),
>       in_cwnd_experiment_(CwndExperimentEnabled()),
>       accepted_queue_ms_(kDefaultAcceptedQueueMs),
>       was_in_alr_(false),
>-      pacer_pushback_experiment_(
>-          IsPacerPushbackExperimentEnabled()) {
>+      send_side_bwe_with_overhead_(
>+          webrtc::field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")),
>+      transport_overhead_bytes_per_packet_(0),
>+      pacer_pushback_experiment_(IsPacerPushbackExperimentEnabled()),
>+      congestion_window_pushback_controller_(
>+          MaybeCreateCongestionWindowPushbackController()) {
>   delay_based_bwe_->SetMinBitrate(min_bitrate_bps_);
>   if (in_cwnd_experiment_ &&
>       !ReadCwndExperimentParameter(&accepted_queue_ms_)) {
>@@ -174,8 +198,12 @@ void SendSideCongestionController::SetBweBitrates(int min_bitrate_bps,
>   bitrate_controller_->SetBitrates(start_bitrate_bps, min_bitrate_bps,
>                                    max_bitrate_bps);
> 
>-  probe_controller_->SetBitrates(min_bitrate_bps, start_bitrate_bps,
>-                                 max_bitrate_bps);
>+  {
>+    rtc::CritScope cs(&probe_lock_);
>+    SendProbes(probe_controller_->SetBitrates(
>+        min_bitrate_bps, start_bitrate_bps, max_bitrate_bps,
>+        clock_->TimeInMilliseconds()));
>+  }
> 
>   {
>     rtc::CritScope cs(&bwe_lock_);
>@@ -187,6 +215,17 @@ void SendSideCongestionController::SetBweBitrates(int min_bitrate_bps,
>   MaybeTriggerOnNetworkChanged();
> }
> 
>+void SendSideCongestionController::SetAllocatedSendBitrateLimits(
>+    int64_t min_send_bitrate_bps,
>+    int64_t max_padding_bitrate_bps,
>+    int64_t max_total_bitrate_bps) {
>+  pacer_->SetSendBitrateLimits(min_send_bitrate_bps, max_padding_bitrate_bps);
>+
>+  rtc::CritScope cs(&probe_lock_);
>+  SendProbes(probe_controller_->OnMaxTotalAllocatedBitrate(
>+      max_total_bitrate_bps, clock_->TimeInMilliseconds()));
>+}
>+
> // TODO(holmer): Split this up and use SetBweBitrates in combination with
> // OnNetworkRouteChanged.
> void SendSideCongestionController::OnNetworkRouteChanged(
>@@ -204,15 +243,20 @@ void SendSideCongestionController::OnNetworkRouteChanged(
>                                             network_route.remote_network_id);
>   {
>     rtc::CritScope cs(&bwe_lock_);
>+    transport_overhead_bytes_per_packet_ = network_route.packet_overhead;
>     min_bitrate_bps_ = min_bitrate_bps;
>-    delay_based_bwe_.reset(new DelayBasedBwe(event_log_, clock_));
>+    delay_based_bwe_.reset(new DelayBasedBwe(event_log_));
>     acknowledged_bitrate_estimator_.reset(new AcknowledgedBitrateEstimator());
>     delay_based_bwe_->SetStartBitrate(bitrate_bps);
>     delay_based_bwe_->SetMinBitrate(min_bitrate_bps);
>   }
>-
>-  probe_controller_->Reset();
>-  probe_controller_->SetBitrates(min_bitrate_bps, bitrate_bps, max_bitrate_bps);
>+  {
>+    rtc::CritScope cs(&probe_lock_);
>+    probe_controller_->Reset(clock_->TimeInMilliseconds());
>+    SendProbes(probe_controller_->SetBitrates(min_bitrate_bps, bitrate_bps,
>+                                              max_bitrate_bps,
>+                                              clock_->TimeInMilliseconds()));
>+  }
> 
>   MaybeTriggerOnNetworkChanged();
> }
>@@ -226,6 +270,10 @@ bool SendSideCongestionController::AvailableBandwidth(
>   return bitrate_controller_->AvailableBandwidth(bandwidth);
> }
> 
>+RtcpBandwidthObserver* SendSideCongestionController::GetBandwidthObserver() {
>+  return bitrate_controller_.get();
>+}
>+
> RtcpBandwidthObserver* SendSideCongestionController::GetBandwidthObserver()
>     const {
>   return bitrate_controller_.get();
>@@ -235,7 +283,11 @@ RateLimiter* SendSideCongestionController::GetRetransmissionRateLimiter() {
>   return retransmission_rate_limiter_.get();
> }
> 
>+void SendSideCongestionController::SetPerPacketFeedbackAvailable(
>+    bool available) {}
>+
> void SendSideCongestionController::EnablePeriodicAlrProbing(bool enable) {
>+  rtc::CritScope cs(&probe_lock_);
>   probe_controller_->EnablePeriodicAlrProbing(enable);
> }
> 
>@@ -260,14 +312,21 @@ void SendSideCongestionController::SignalNetworkState(NetworkState state) {
>     pause_pacer_ = state == kNetworkDown;
>     network_state_ = state;
>   }
>-  probe_controller_->OnNetworkStateChanged(state);
>+
>+  {
>+    rtc::CritScope cs(&probe_lock_);
>+    NetworkAvailability msg;
>+    msg.at_time = Timestamp::ms(clock_->TimeInMilliseconds());
>+    msg.network_available = state == kNetworkUp;
>+    SendProbes(probe_controller_->OnNetworkAvailability(msg));
>+  }
>   MaybeTriggerOnNetworkChanged();
> }
> 
> void SendSideCongestionController::SetTransportOverhead(
>     size_t transport_overhead_bytes_per_packet) {
>-  transport_feedback_adapter_.SetTransportOverhead(
>-      rtc::dchecked_cast<int>(transport_overhead_bytes_per_packet));
>+  rtc::CritScope cs(&bwe_lock_);
>+  transport_overhead_bytes_per_packet_ = transport_overhead_bytes_per_packet;
> }
> 
> void SendSideCongestionController::OnSentPacket(
>@@ -285,13 +344,20 @@ void SendSideCongestionController::OnSentPacket(
> void SendSideCongestionController::OnRttUpdate(int64_t avg_rtt_ms,
>                                                int64_t max_rtt_ms) {
>   rtc::CritScope cs(&bwe_lock_);
>-  delay_based_bwe_->OnRttUpdate(avg_rtt_ms, max_rtt_ms);
>+  delay_based_bwe_->OnRttUpdate(avg_rtt_ms);
> }
> 
> int64_t SendSideCongestionController::TimeUntilNextProcess() {
>   return bitrate_controller_->TimeUntilNextProcess();
> }
> 
>+void SendSideCongestionController::SendProbes(
>+    std::vector<ProbeClusterConfig> probe_configs) {
>+  for (auto probe_config : probe_configs) {
>+    pacer_->CreateProbeCluster(probe_config.target_data_rate.bps());
>+  }
>+}
>+
> void SendSideCongestionController::Process() {
>   bool pause_pacer;
>   // TODO(holmer): Once this class is running on a task queue we should
>@@ -308,7 +374,13 @@ void SendSideCongestionController::Process() {
>     pacer_paused_ = false;
>   }
>   bitrate_controller_->Process();
>-  probe_controller_->Process();
>+
>+  {
>+    rtc::CritScope cs(&probe_lock_);
>+    probe_controller_->SetAlrStartTimeMs(
>+        pacer_->GetApplicationLimitedRegionStartTime());
>+    SendProbes(probe_controller_->Process(clock_->TimeInMilliseconds()));
>+  }
>   MaybeTriggerOnNetworkChanged();
> }
> 
>@@ -317,6 +389,10 @@ void SendSideCongestionController::AddPacket(
>     uint16_t sequence_number,
>     size_t length,
>     const PacedPacketInfo& pacing_info) {
>+  if (send_side_bwe_with_overhead_) {
>+    rtc::CritScope cs(&bwe_lock_);
>+    length += transport_overhead_bytes_per_packet_;
>+  }
>   transport_feedback_adapter_.AddPacket(ssrc, sequence_number, length,
>                                         pacing_info);
> }
>@@ -334,6 +410,7 @@ void SendSideCongestionController::OnTransportFeedback(
>   if (was_in_alr_ && !currently_in_alr) {
>     int64_t now_ms = rtc::TimeMillis();
>     acknowledged_bitrate_estimator_->SetAlrEndedTimeMs(now_ms);
>+    rtc::CritScope cs(&probe_lock_);
>     probe_controller_->SetAlrEndedTimeMs(now_ms);
>   }
>   was_in_alr_ = currently_in_alr;
>@@ -344,24 +421,30 @@ void SendSideCongestionController::OnTransportFeedback(
>   {
>     rtc::CritScope cs(&bwe_lock_);
>     result = delay_based_bwe_->IncomingPacketFeedbackVector(
>-        feedback_vector, acknowledged_bitrate_estimator_->bitrate_bps());
>+        feedback_vector, acknowledged_bitrate_estimator_->bitrate_bps(),
>+        clock_->TimeInMilliseconds());
>   }
>   if (result.updated) {
>     bitrate_controller_->OnDelayBasedBweResult(result);
>     // Update the estimate in the ProbeController, in case we want to probe.
>     MaybeTriggerOnNetworkChanged();
>   }
>-  if (result.recovered_from_overuse)
>-    probe_controller_->RequestProbe();
>-  if (in_cwnd_experiment_)
>+  if (result.recovered_from_overuse) {
>+    rtc::CritScope cs(&probe_lock_);
>+    probe_controller_->SetAlrStartTimeMs(
>+        pacer_->GetApplicationLimitedRegionStartTime());
>+    SendProbes(probe_controller_->RequestProbe(clock_->TimeInMilliseconds()));
>+  }
>+  if (in_cwnd_experiment_) {
>     LimitOutstandingBytes(transport_feedback_adapter_.GetOutstandingBytes());
>+  }
> }
> 
> void SendSideCongestionController::LimitOutstandingBytes(
>     size_t num_outstanding_bytes) {
>   RTC_DCHECK(in_cwnd_experiment_);
>   rtc::CritScope lock(&network_state_lock_);
>-  rtc::Optional<int64_t> min_rtt_ms =
>+  absl::optional<int64_t> min_rtt_ms =
>       transport_feedback_adapter_.GetMinFeedbackLoopRtt();
>   // No valid RTT. Could be because send-side BWE isn't used, in which case
>   // we don't try to limit the outstanding packets.
>@@ -378,7 +461,14 @@ void SendSideCongestionController::LimitOutstandingBytes(
>                    << " max outstanding: " << max_outstanding_bytes;
>   RTC_LOG(LS_INFO) << "Feedback rtt: " << *min_rtt_ms
>                    << " Bitrate: " << last_reported_bitrate_bps_;
>-  pause_pacer_ = num_outstanding_bytes > max_outstanding_bytes;
>+  if (congestion_window_pushback_controller_) {
>+    congestion_window_pushback_controller_->UpdateOutstandingData(
>+        num_outstanding_bytes);
>+    congestion_window_pushback_controller_->UpdateMaxOutstandingData(
>+        max_outstanding_bytes);
>+  } else {
>+    pause_pacer_ = num_outstanding_bytes > max_outstanding_bytes;
>+  }
> }
> 
> std::vector<PacketFeedback>
>@@ -387,6 +477,16 @@ SendSideCongestionController::GetTransportFeedbackVector() const {
>   return transport_feedback_adapter_.GetTransportFeedbackVector();
> }
> 
>+void SendSideCongestionController::SetPacingFactor(float pacing_factor) {
>+  pacer_->SetPacingFactor(pacing_factor);
>+}
>+
>+void SendSideCongestionController::SetAllocatedBitrateWithoutFeedback(
>+    uint32_t bitrate_bps) {
>+  acknowledged_bitrate_estimator_->SetAllocatedBitrateWithoutFeedback(
>+      bitrate_bps);
>+}
>+
> void SendSideCongestionController::MaybeTriggerOnNetworkChanged() {
>   uint32_t bitrate_bps;
>   uint8_t fraction_loss;
>@@ -395,29 +495,35 @@ void SendSideCongestionController::MaybeTriggerOnNetworkChanged() {
>       &bitrate_bps, &fraction_loss, &rtt);
>   if (estimate_changed) {
>     pacer_->SetEstimatedBitrate(bitrate_bps);
>-    probe_controller_->SetEstimatedBitrate(bitrate_bps);
>+    {
>+      rtc::CritScope cs(&probe_lock_);
>+      SendProbes(probe_controller_->SetEstimatedBitrate(
>+          bitrate_bps, clock_->TimeInMilliseconds()));
>+    }
>     retransmission_rate_limiter_->SetMaxRate(bitrate_bps);
>   }
> 
>-  if (!pacer_pushback_experiment_) {
>-    bitrate_bps = IsNetworkDown() || IsSendQueueFull() ? 0 : bitrate_bps;
>+  if (IsNetworkDown()) {
>+    bitrate_bps = 0;
>+  } else if (congestion_window_pushback_controller_) {
>+    rtc::CritScope lock(&network_state_lock_);
>+    bitrate_bps = congestion_window_pushback_controller_->UpdateTargetBitrate(
>+        bitrate_bps);
>+  } else if (!pacer_pushback_experiment_) {
>+    bitrate_bps = IsSendQueueFull() ? 0 : bitrate_bps;
>   } else {
>-    if (IsNetworkDown()) {
>-      bitrate_bps = 0;
>-    } else {
>-      int64_t queue_length_ms = pacer_->ExpectedQueueTimeMs();
>-
>-      if (queue_length_ms == 0) {
>-        encoding_rate_ = 1.0;
>-      } else if (queue_length_ms > 50) {
>-        float encoding_rate = 1.0 - queue_length_ms / 1000.0;
>-        encoding_rate_ = std::min(encoding_rate_, encoding_rate);
>-        encoding_rate_ = std::max(encoding_rate_, 0.0f);
>-      }
>-
>-      bitrate_bps *= encoding_rate_;
>-      bitrate_bps = bitrate_bps < 50000 ? 0 : bitrate_bps;
>+    int64_t queue_length_ms = pacer_->ExpectedQueueTimeMs();
>+
>+    if (queue_length_ms == 0) {
>+      encoding_rate_ = 1.0;
>+    } else if (queue_length_ms > 50) {
>+      float encoding_rate = 1.0 - queue_length_ms / 1000.0;
>+      encoding_rate_ = std::min(encoding_rate_, encoding_rate);
>+      encoding_rate_ = std::max(encoding_rate_, 0.0f);
>     }
>+
>+    bitrate_bps *= encoding_rate_;
>+    bitrate_bps = bitrate_bps < 50000 ? 0 : bitrate_bps;
>   }
> 
>   if (HasNetworkParametersToReportChanged(bitrate_bps, fraction_loss, rtt)) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/send_side_congestion_controller_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/send_side_congestion_controller_unittest.cc
>index 67d2c402c4f..3653b8c22f6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/send_side_congestion_controller_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/send_side_congestion_controller_unittest.cc
>@@ -8,11 +8,11 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>+#include "modules/congestion_controller/include/send_side_congestion_controller.h"
> #include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
> #include "modules/bitrate_controller/include/bitrate_controller.h"
>-#include "modules/congestion_controller/congestion_controller_unittests_helper.h"
> #include "modules/congestion_controller/include/mock/mock_congestion_observer.h"
>-#include "modules/congestion_controller/include/send_side_congestion_controller.h"
>+#include "modules/congestion_controller/rtp/congestion_controller_unittests_helper.h"
> #include "modules/pacing/mock/mock_paced_sender.h"
> #include "modules/pacing/packet_router.h"
> #include "modules/remote_bitrate_estimator/include/bwe_defines.h"
>@@ -44,11 +44,13 @@ const uint32_t kInitialBitrateBps = 60000;
> 
> namespace test {
> 
>-class SendSideCongestionControllerTest : public ::testing::Test {
>+class LegacySendSideCongestionControllerTest : public ::testing::Test {
>  protected:
>-  SendSideCongestionControllerTest()
>-      : clock_(123456), target_bitrate_observer_(this) {}
>-  ~SendSideCongestionControllerTest() override {}
>+  LegacySendSideCongestionControllerTest()
>+      : clock_(123456),
>+        target_bitrate_observer_(this),
>+        bandwidth_observer_(nullptr) {}
>+  ~LegacySendSideCongestionControllerTest() override {}
> 
>   void SetUp() override {
>     pacer_.reset(new NiceMock<MockPacedSender>());
>@@ -85,9 +87,10 @@ class SendSideCongestionControllerTest : public ::testing::Test {
> 
>   // Allows us to track the target bitrate, without prescribing the exact
>   // iterations when this would hapen, like a mock would.
>-  class TargetBitrateObserver : public SendSideCongestionController::Observer {
>+  class TargetBitrateObserver : public NetworkChangedObserver {
>    public:
>-    explicit TargetBitrateObserver(SendSideCongestionControllerTest* owner)
>+    explicit TargetBitrateObserver(
>+        LegacySendSideCongestionControllerTest* owner)
>         : owner_(owner) {}
>     ~TargetBitrateObserver() override = default;
>     void OnNetworkChanged(uint32_t bitrate_bps,
>@@ -98,7 +101,7 @@ class SendSideCongestionControllerTest : public ::testing::Test {
>     }
> 
>    private:
>-    SendSideCongestionControllerTest* owner_;
>+    LegacySendSideCongestionControllerTest* owner_;
>   };
> 
>   void PacketTransmissionAndFeedbackBlock(uint16_t* seq_num,
>@@ -139,10 +142,10 @@ class SendSideCongestionControllerTest : public ::testing::Test {
>   std::unique_ptr<NiceMock<MockPacedSender>> pacer_;
>   std::unique_ptr<SendSideCongestionController> controller_;
> 
>-  rtc::Optional<uint32_t> target_bitrate_bps_;
>+  absl::optional<uint32_t> target_bitrate_bps_;
> };
> 
>-TEST_F(SendSideCongestionControllerTest, OnNetworkChanged) {
>+TEST_F(LegacySendSideCongestionControllerTest, OnNetworkChanged) {
>   // Test no change.
>   clock_.AdvanceTimeMilliseconds(25);
>   controller_->Process();
>@@ -160,7 +163,7 @@ TEST_F(SendSideCongestionControllerTest, OnNetworkChanged) {
>   controller_->Process();
> }
> 
>-TEST_F(SendSideCongestionControllerTest, OnSendQueueFull) {
>+TEST_F(LegacySendSideCongestionControllerTest, OnSendQueueFull) {
>   EXPECT_CALL(*pacer_, ExpectedQueueTimeMs())
>       .WillOnce(Return(PacedSender::kMaxQueueLengthMs + 1));
> 
>@@ -175,7 +178,8 @@ TEST_F(SendSideCongestionControllerTest, OnSendQueueFull) {
>   controller_->Process();
> }
> 
>-TEST_F(SendSideCongestionControllerTest, OnSendQueueFullAndEstimateChange) {
>+TEST_F(LegacySendSideCongestionControllerTest,
>+       OnSendQueueFullAndEstimateChange) {
>   EXPECT_CALL(*pacer_, ExpectedQueueTimeMs())
>       .WillOnce(Return(PacedSender::kMaxQueueLengthMs + 1));
>   EXPECT_CALL(observer_, OnNetworkChanged(0, _, _, _));
>@@ -199,7 +203,7 @@ TEST_F(SendSideCongestionControllerTest, OnSendQueueFullAndEstimateChange) {
>   controller_->Process();
> }
> 
>-TEST_F(SendSideCongestionControllerTest, SignalNetworkState) {
>+TEST_F(LegacySendSideCongestionControllerTest, SignalNetworkState) {
>   EXPECT_CALL(observer_, OnNetworkChanged(0, _, _, _));
>   controller_->SignalNetworkState(kNetworkDown);
> 
>@@ -210,7 +214,7 @@ TEST_F(SendSideCongestionControllerTest, SignalNetworkState) {
>   controller_->SignalNetworkState(kNetworkDown);
> }
> 
>-TEST_F(SendSideCongestionControllerTest, OnNetworkRouteChanged) {
>+TEST_F(LegacySendSideCongestionControllerTest, OnNetworkRouteChanged) {
>   int new_bitrate = 200000;
>   testing::Mock::VerifyAndClearExpectations(pacer_.get());
>   EXPECT_CALL(observer_, OnNetworkChanged(new_bitrate, _, _, _));
>@@ -230,7 +234,7 @@ TEST_F(SendSideCongestionControllerTest, OnNetworkRouteChanged) {
>   controller_->OnNetworkRouteChanged(route, -1, -1, -1);
> }
> 
>-TEST_F(SendSideCongestionControllerTest, OldFeedback) {
>+TEST_F(LegacySendSideCongestionControllerTest, OldFeedback) {
>   int new_bitrate = 200000;
>   testing::Mock::VerifyAndClearExpectations(pacer_.get());
>   EXPECT_CALL(observer_, OnNetworkChanged(new_bitrate, _, _, _));
>@@ -273,7 +277,7 @@ TEST_F(SendSideCongestionControllerTest, OldFeedback) {
>   controller_->OnNetworkRouteChanged(route, -1, -1, -1);
> }
> 
>-TEST_F(SendSideCongestionControllerTest,
>+TEST_F(LegacySendSideCongestionControllerTest,
>        SignalNetworkStateAndQueueIsFullAndEstimateChange) {
>   // Send queue is full
>   EXPECT_CALL(*pacer_, ExpectedQueueTimeMs())
>@@ -302,7 +306,7 @@ TEST_F(SendSideCongestionControllerTest,
>   controller_->Process();
> }
> 
>-TEST_F(SendSideCongestionControllerTest, GetPacerQueuingDelayMs) {
>+TEST_F(LegacySendSideCongestionControllerTest, GetPacerQueuingDelayMs) {
>   EXPECT_CALL(observer_, OnNetworkChanged(_, _, _, _)).Times(AtLeast(1));
> 
>   const int64_t kQueueTimeMs = 123;
>@@ -318,7 +322,7 @@ TEST_F(SendSideCongestionControllerTest, GetPacerQueuingDelayMs) {
>   EXPECT_EQ(kQueueTimeMs, controller_->GetPacerQueuingDelayMs());
> }
> 
>-TEST_F(SendSideCongestionControllerTest, GetProbingInterval) {
>+TEST_F(LegacySendSideCongestionControllerTest, GetProbingInterval) {
>   clock_.AdvanceTimeMilliseconds(25);
>   controller_->Process();
> 
>@@ -329,7 +333,7 @@ TEST_F(SendSideCongestionControllerTest, GetProbingInterval) {
>   controller_->Process();
> }
> 
>-TEST_F(SendSideCongestionControllerTest, ProbeOnRouteChange) {
>+TEST_F(LegacySendSideCongestionControllerTest, ProbeOnRouteChange) {
>   testing::Mock::VerifyAndClearExpectations(pacer_.get());
>   EXPECT_CALL(*pacer_, CreateProbeCluster(kInitialBitrateBps * 6));
>   EXPECT_CALL(*pacer_, CreateProbeCluster(kInitialBitrateBps * 12));
>@@ -342,7 +346,7 @@ TEST_F(SendSideCongestionControllerTest, ProbeOnRouteChange) {
> 
> // Estimated bitrate reduced when the feedbacks arrive with such a long delay,
> // that the send-time-history no longer holds the feedbacked packets.
>-TEST_F(SendSideCongestionControllerTest, LongFeedbackDelays) {
>+TEST_F(LegacySendSideCongestionControllerTest, LongFeedbackDelays) {
>   TargetBitrateTrackingSetup();
> 
>   const int64_t kFeedbackTimeoutMs = 60001;
>@@ -433,7 +437,7 @@ TEST_F(SendSideCongestionControllerTest, LongFeedbackDelays) {
> 
> // Bandwidth estimation is updated when feedbacks are received.
> // Feedbacks which show an increasing delay cause the estimation to be reduced.
>-TEST_F(SendSideCongestionControllerTest, UpdatesDelayBasedEstimate) {
>+TEST_F(LegacySendSideCongestionControllerTest, UpdatesDelayBasedEstimate) {
>   TargetBitrateTrackingSetup();
> 
>   const int64_t kRunTimeMs = 6000;
>@@ -452,7 +456,7 @@ TEST_F(SendSideCongestionControllerTest, UpdatesDelayBasedEstimate) {
>   EXPECT_LT(*target_bitrate_bps_, bitrate_before_delay);
> }
> 
>-TEST_F(SendSideCongestionControllerTest, PacerQueueEncodeRatePushback) {
>+TEST_F(LegacySendSideCongestionControllerTest, PacerQueueEncodeRatePushback) {
>   ScopedFieldTrials pushback_field_trial(
>       "WebRTC-PacerPushbackExperiment/Enabled/");
>   SetUp();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/test/controller_printer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/test/controller_printer.cc
>new file mode 100644
>index 00000000000..4c9c4fde8f8
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/test/controller_printer.cc
>@@ -0,0 +1,66 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#include "modules/congestion_controller/test/controller_printer.h"
>+
>+#include <cmath>
>+#include <limits>
>+#include <utility>
>+
>+#include "rtc_base/checks.h"
>+
>+namespace webrtc {
>+
>+ControlStatePrinter::ControlStatePrinter(
>+    FILE* output,
>+    std::unique_ptr<DebugStatePrinter> debug_printer)
>+    : output_(output), debug_printer_(std::move(debug_printer)) {}
>+
>+ControlStatePrinter::~ControlStatePrinter() = default;
>+
>+void ControlStatePrinter::PrintHeaders() {
>+  fprintf(output_, "time bandwidth rtt target pacing padding window");
>+  if (debug_printer_) {
>+    fprintf(output_, " ");
>+    debug_printer_->PrintHeaders(output_);
>+  }
>+  fprintf(output_, "\n");
>+  fflush(output_);
>+}
>+
>+void ControlStatePrinter::PrintState(const Timestamp time,
>+                                     const NetworkControlUpdate state) {
>+  double timestamp = time.seconds<double>();
>+  auto estimate = state.target_rate->network_estimate;
>+  double bandwidth = estimate.bandwidth.bps() / 8.0;
>+  double rtt = estimate.round_trip_time.seconds<double>();
>+  double target_rate = state.target_rate->target_rate.bps() / 8.0;
>+  double pacing_rate = state.pacer_config->data_rate().bps() / 8.0;
>+  double padding_rate = state.pacer_config->pad_rate().bps() / 8.0;
>+  double congestion_window = state.congestion_window
>+                                 ? state.congestion_window->bytes<double>()
>+                                 : std::numeric_limits<double>::infinity();
>+
>+  fprintf(output_, "%f %f %f %f %f %f %f", timestamp, bandwidth, rtt,
>+          target_rate, pacing_rate, padding_rate, congestion_window);
>+
>+  if (debug_printer_) {
>+    fprintf(output_, " ");
>+    debug_printer_->PrintValues(output_);
>+  }
>+  fprintf(output_, "\n");
>+  fflush(output_);
>+}
>+
>+void ControlStatePrinter::PrintState(const Timestamp time) {
>+  if (debug_printer_ && debug_printer_->Attached()) {
>+    PrintState(time, debug_printer_->GetState(time));
>+  }
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/test/controller_printer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/test/controller_printer.h
>new file mode 100644
>index 00000000000..3c09b9142dd
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/test/controller_printer.h
>@@ -0,0 +1,43 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#ifndef MODULES_CONGESTION_CONTROLLER_TEST_CONTROLLER_PRINTER_H_
>+#define MODULES_CONGESTION_CONTROLLER_TEST_CONTROLLER_PRINTER_H_
>+
>+#include <cstdio>
>+#include <memory>
>+
>+#include "api/transport/network_control.h"
>+
>+namespace webrtc {
>+class DebugStatePrinter {
>+ public:
>+  virtual bool Attached() const = 0;
>+  virtual void PrintHeaders(FILE* out) = 0;
>+  virtual void PrintValues(FILE* out) = 0;
>+  virtual NetworkControlUpdate GetState(Timestamp at_time) const = 0;
>+  virtual ~DebugStatePrinter() = default;
>+};
>+
>+class ControlStatePrinter {
>+ public:
>+  ControlStatePrinter(FILE* output,
>+                      std::unique_ptr<DebugStatePrinter> debug_printer);
>+  ~ControlStatePrinter();
>+  void PrintHeaders();
>+  void PrintState(const Timestamp time, const NetworkControlUpdate state);
>+  void PrintState(const Timestamp time);
>+
>+ private:
>+  FILE* output_;
>+  std::unique_ptr<DebugStatePrinter> debug_printer_;
>+};
>+}  // namespace webrtc
>+
>+#endif  // MODULES_CONGESTION_CONTROLLER_TEST_CONTROLLER_PRINTER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/transport_feedback_adapter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/transport_feedback_adapter.cc
>index 0989875df98..3baa3ad5a53 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/transport_feedback_adapter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/transport_feedback_adapter.cc
>@@ -12,12 +12,11 @@
> 
> #include <algorithm>
> 
>-#include "modules/congestion_controller/delay_based_bwe.h"
>+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> #include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/numerics/mod_ops.h"
>-#include "system_wrappers/include/field_trial.h"
> 
> namespace webrtc {
> 
>@@ -28,10 +27,7 @@ const int64_t kBaseTimestampScaleFactor =
> const int64_t kBaseTimestampRangeSizeUs = kBaseTimestampScaleFactor * (1 << 24);
> 
> TransportFeedbackAdapter::TransportFeedbackAdapter(const Clock* clock)
>-    : send_side_bwe_with_overhead_(
>-          webrtc::field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")),
>-      transport_overhead_bytes_per_packet_(0),
>-      send_time_history_(clock, kSendTimeHistoryWindowMs),
>+    : send_time_history_(clock, kSendTimeHistoryWindowMs),
>       clock_(clock),
>       current_offset_ms_(kNoTimestamp),
>       last_timestamp_us_(kNoTimestamp),
>@@ -66,9 +62,6 @@ void TransportFeedbackAdapter::AddPacket(uint32_t ssrc,
>                                          const PacedPacketInfo& pacing_info) {
>   {
>     rtc::CritScope cs(&lock_);
>-    if (send_side_bwe_with_overhead_) {
>-      length += transport_overhead_bytes_per_packet_;
>-    }
>     const int64_t creation_time_ms = clock_->TimeInMilliseconds();
>     send_time_history_.AddAndRemoveOld(
>         PacketFeedback(creation_time_ms, sequence_number, length, local_net_id_,
>@@ -77,7 +70,7 @@ void TransportFeedbackAdapter::AddPacket(uint32_t ssrc,
> 
>   {
>     rtc::CritScope cs(&observers_lock_);
>-    for (auto observer : observers_) {
>+    for (auto* observer : observers_) {
>       observer->OnPacketAdded(ssrc, sequence_number);
>     }
>   }
>@@ -89,12 +82,6 @@ void TransportFeedbackAdapter::OnSentPacket(uint16_t sequence_number,
>   send_time_history_.OnSentPacket(sequence_number, send_time_ms);
> }
> 
>-void TransportFeedbackAdapter::SetTransportOverhead(
>-    int transport_overhead_bytes_per_packet) {
>-  rtc::CritScope cs(&lock_);
>-  transport_overhead_bytes_per_packet_ = transport_overhead_bytes_per_packet;
>-}
>-
> void TransportFeedbackAdapter::SetNetworkIds(uint16_t local_id,
>                                              uint16_t remote_id) {
>   rtc::CritScope cs(&lock_);
>@@ -195,7 +182,7 @@ void TransportFeedbackAdapter::OnTransportFeedback(
>   last_packet_feedback_vector_ = GetPacketFeedbackVector(feedback);
>   {
>     rtc::CritScope cs(&observers_lock_);
>-    for (auto observer : observers_) {
>+    for (auto* observer : observers_) {
>       observer->OnPacketFeedbackVector(last_packet_feedback_vector_);
>     }
>   }
>@@ -206,7 +193,8 @@ TransportFeedbackAdapter::GetTransportFeedbackVector() const {
>   return last_packet_feedback_vector_;
> }
> 
>-rtc::Optional<int64_t> TransportFeedbackAdapter::GetMinFeedbackLoopRtt() const {
>+absl::optional<int64_t> TransportFeedbackAdapter::GetMinFeedbackLoopRtt()
>+    const {
>   rtc::CritScope cs(&lock_);
>   return min_feedback_rtt_;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/transport_feedback_adapter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/transport_feedback_adapter.h
>index 008bab87f1f..b411c26bd38 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/transport_feedback_adapter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/transport_feedback_adapter.h
>@@ -14,7 +14,7 @@
> #include <deque>
> #include <vector>
> 
>-#include "modules/remote_bitrate_estimator/include/send_time_history.h"
>+#include "modules/congestion_controller/rtp/send_time_history.h"
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/thread_annotations.h"
> #include "rtc_base/thread_checker.h"
>@@ -47,9 +47,9 @@ class TransportFeedbackAdapter {
>   // to the CongestionController interface.
>   void OnTransportFeedback(const rtcp::TransportFeedback& feedback);
>   std::vector<PacketFeedback> GetTransportFeedbackVector() const;
>-  rtc::Optional<int64_t> GetMinFeedbackLoopRtt() const;
>+  absl::optional<int64_t> GetMinFeedbackLoopRtt() const;
> 
>-  void SetTransportOverhead(int transport_overhead_bytes_per_packet);
>+  void SetTransportOverhead(size_t transport_overhead_bytes_per_packet);
> 
>   void SetNetworkIds(uint16_t local_id, uint16_t remote_id);
> 
>@@ -59,9 +59,7 @@ class TransportFeedbackAdapter {
>   std::vector<PacketFeedback> GetPacketFeedbackVector(
>       const rtcp::TransportFeedback& feedback);
> 
>-  const bool send_side_bwe_with_overhead_;
>   rtc::CriticalSection lock_;
>-  int transport_overhead_bytes_per_packet_ RTC_GUARDED_BY(&lock_);
>   SendTimeHistory send_time_history_ RTC_GUARDED_BY(&lock_);
>   const Clock* const clock_;
>   int64_t current_offset_ms_;
>@@ -70,7 +68,7 @@ class TransportFeedbackAdapter {
>   uint16_t local_net_id_ RTC_GUARDED_BY(&lock_);
>   uint16_t remote_net_id_ RTC_GUARDED_BY(&lock_);
>   std::deque<int64_t> feedback_rtts_ RTC_GUARDED_BY(&lock_);
>-  rtc::Optional<int64_t> min_feedback_rtt_ RTC_GUARDED_BY(&lock_);
>+  absl::optional<int64_t> min_feedback_rtt_ RTC_GUARDED_BY(&lock_);
> 
>   rtc::CriticalSection observers_lock_;
>   std::vector<PacketFeedbackObserver*> observers_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/transport_feedback_adapter_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/transport_feedback_adapter_unittest.cc
>index 1cff0d981f5..39aa72e8167 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/transport_feedback_adapter_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/transport_feedback_adapter_unittest.cc
>@@ -13,7 +13,7 @@
> #include <vector>
> 
> #include "modules/bitrate_controller/include/mock/mock_bitrate_controller.h"
>-#include "modules/congestion_controller/congestion_controller_unittests_helper.h"
>+#include "modules/congestion_controller/rtp/congestion_controller_unittests_helper.h"
> #include "modules/congestion_controller/transport_feedback_adapter.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> #include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
>@@ -34,7 +34,7 @@ const PacedPacketInfo kPacingInfo1(1, 8, 4000);
> const PacedPacketInfo kPacingInfo2(2, 14, 7000);
> const PacedPacketInfo kPacingInfo3(3, 20, 10000);
> const PacedPacketInfo kPacingInfo4(4, 22, 10000);
>-}
>+}  // namespace
> 
> namespace test {
> 
>@@ -45,11 +45,11 @@ class MockPacketFeedbackObserver : public webrtc::PacketFeedbackObserver {
>                void(const std::vector<PacketFeedback>& packet_feedback_vector));
> };
> 
>-class TransportFeedbackAdapterTest : public ::testing::Test {
>+class LegacyTransportFeedbackAdapterTest : public ::testing::Test {
>  public:
>-  TransportFeedbackAdapterTest() : clock_(0) {}
>+  LegacyTransportFeedbackAdapterTest() : clock_(0) {}
> 
>-  virtual ~TransportFeedbackAdapterTest() {}
>+  virtual ~LegacyTransportFeedbackAdapterTest() {}
> 
>   virtual void SetUp() {
>     adapter_.reset(new TransportFeedbackAdapter(&clock_));
>@@ -78,15 +78,14 @@ class TransportFeedbackAdapterTest : public ::testing::Test {
>   std::unique_ptr<TransportFeedbackAdapter> adapter_;
> };
> 
>-TEST_F(TransportFeedbackAdapterTest, ObserverSanity) {
>+TEST_F(LegacyTransportFeedbackAdapterTest, ObserverSanity) {
>   MockPacketFeedbackObserver mock;
>   adapter_->RegisterPacketFeedbackObserver(&mock);
> 
>   const std::vector<PacketFeedback> packets = {
>       PacketFeedback(100, 200, 0, 1000, kPacingInfo0),
>       PacketFeedback(110, 210, 1, 2000, kPacingInfo0),
>-      PacketFeedback(120, 220, 2, 3000, kPacingInfo0)
>-  };
>+      PacketFeedback(120, 220, 2, 3000, kPacingInfo0)};
> 
>   rtcp::TransportFeedback feedback;
>   feedback.SetBase(packets[0].sequence_number,
>@@ -119,14 +118,16 @@ TEST_F(TransportFeedbackAdapterTest, ObserverSanity) {
> }
> 
> #if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
>-TEST_F(TransportFeedbackAdapterTest, ObserverDoubleRegistrationDeathTest) {
>+TEST_F(LegacyTransportFeedbackAdapterTest,
>+       ObserverDoubleRegistrationDeathTest) {
>   MockPacketFeedbackObserver mock;
>   adapter_->RegisterPacketFeedbackObserver(&mock);
>   EXPECT_DEATH(adapter_->RegisterPacketFeedbackObserver(&mock), "");
>   adapter_->DeRegisterPacketFeedbackObserver(&mock);
> }
> 
>-TEST_F(TransportFeedbackAdapterTest, ObserverMissingDeRegistrationDeathTest) {
>+TEST_F(LegacyTransportFeedbackAdapterTest,
>+       ObserverMissingDeRegistrationDeathTest) {
>   MockPacketFeedbackObserver mock;
>   adapter_->RegisterPacketFeedbackObserver(&mock);
>   EXPECT_DEATH(adapter_.reset(), "");
>@@ -134,7 +135,8 @@ TEST_F(TransportFeedbackAdapterTest, ObserverMissingDeRegistrationDeathTest) {
> }
> #endif
> 
>-TEST_F(TransportFeedbackAdapterTest, AdaptsFeedbackAndPopulatesSendTimes) {
>+TEST_F(LegacyTransportFeedbackAdapterTest,
>+       AdaptsFeedbackAndPopulatesSendTimes) {
>   std::vector<PacketFeedback> packets;
>   packets.push_back(PacketFeedback(100, 200, 0, 1500, kPacingInfo0));
>   packets.push_back(PacketFeedback(110, 210, 1, 1500, kPacingInfo0));
>@@ -160,7 +162,7 @@ TEST_F(TransportFeedbackAdapterTest, AdaptsFeedbackAndPopulatesSendTimes) {
>   ComparePacketFeedbackVectors(packets, adapter_->GetTransportFeedbackVector());
> }
> 
>-TEST_F(TransportFeedbackAdapterTest, FeedbackVectorReportsUnreceived) {
>+TEST_F(LegacyTransportFeedbackAdapterTest, FeedbackVectorReportsUnreceived) {
>   std::vector<PacketFeedback> sent_packets = {
>       PacketFeedback(100, 220, 0, 1500, kPacingInfo0),
>       PacketFeedback(110, 210, 1, 1500, kPacingInfo0),
>@@ -168,8 +170,7 @@ TEST_F(TransportFeedbackAdapterTest, FeedbackVectorReportsUnreceived) {
>       PacketFeedback(130, 230, 3, 1500, kPacingInfo0),
>       PacketFeedback(140, 240, 4, 1500, kPacingInfo0),
>       PacketFeedback(150, 250, 5, 1500, kPacingInfo0),
>-      PacketFeedback(160, 260, 6, 1500, kPacingInfo0)
>-  };
>+      PacketFeedback(160, 260, 6, 1500, kPacingInfo0)};
> 
>   for (const PacketFeedback& packet : sent_packets)
>     OnSentPacket(packet);
>@@ -177,8 +178,7 @@ TEST_F(TransportFeedbackAdapterTest, FeedbackVectorReportsUnreceived) {
>   // Note: Important to include the last packet, as only unreceived packets in
>   // between received packets can be inferred.
>   std::vector<PacketFeedback> received_packets = {
>-    sent_packets[0], sent_packets[2], sent_packets[6]
>-  };
>+      sent_packets[0], sent_packets[2], sent_packets[6]};
> 
>   rtcp::TransportFeedback feedback;
>   feedback.SetBase(received_packets[0].sequence_number,
>@@ -196,7 +196,7 @@ TEST_F(TransportFeedbackAdapterTest, FeedbackVectorReportsUnreceived) {
>                                adapter_->GetTransportFeedbackVector());
> }
> 
>-TEST_F(TransportFeedbackAdapterTest, HandlesDroppedPackets) {
>+TEST_F(LegacyTransportFeedbackAdapterTest, HandlesDroppedPackets) {
>   std::vector<PacketFeedback> packets;
>   packets.push_back(PacketFeedback(100, 200, 0, 1500, kPacingInfo0));
>   packets.push_back(PacketFeedback(110, 210, 1, 1500, kPacingInfo1));
>@@ -240,7 +240,7 @@ TEST_F(TransportFeedbackAdapterTest, HandlesDroppedPackets) {
>                                adapter_->GetTransportFeedbackVector());
> }
> 
>-TEST_F(TransportFeedbackAdapterTest, SendTimeWrapsBothWays) {
>+TEST_F(LegacyTransportFeedbackAdapterTest, SendTimeWrapsBothWays) {
>   int64_t kHighArrivalTimeMs = rtcp::TransportFeedback::kDeltaScaleFactor *
>                                static_cast<int64_t>(1 << 8) *
>                                static_cast<int64_t>((1 << 23) - 1) / 1000;
>@@ -277,7 +277,7 @@ TEST_F(TransportFeedbackAdapterTest, SendTimeWrapsBothWays) {
>   }
> }
> 
>-TEST_F(TransportFeedbackAdapterTest, HandlesArrivalReordering) {
>+TEST_F(LegacyTransportFeedbackAdapterTest, HandlesArrivalReordering) {
>   std::vector<PacketFeedback> packets;
>   packets.push_back(PacketFeedback(120, 200, 0, 1500, kPacingInfo0));
>   packets.push_back(PacketFeedback(110, 210, 1, 1500, kPacingInfo0));
>@@ -304,7 +304,7 @@ TEST_F(TransportFeedbackAdapterTest, HandlesArrivalReordering) {
>   ComparePacketFeedbackVectors(packets, adapter_->GetTransportFeedbackVector());
> }
> 
>-TEST_F(TransportFeedbackAdapterTest, TimestampDeltas) {
>+TEST_F(LegacyTransportFeedbackAdapterTest, TimestampDeltas) {
>   std::vector<PacketFeedback> sent_packets;
>   const int64_t kSmallDeltaUs =
>       rtcp::TransportFeedback::kDeltaScaleFactor * ((1 << 8) - 1);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/trendline_estimator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/trendline_estimator.cc
>deleted file mode 100644
>index eeb5edf1cb0..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/congestion_controller/trendline_estimator.cc
>+++ /dev/null
>@@ -1,97 +0,0 @@
>-/*
>- *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "modules/congestion_controller/trendline_estimator.h"
>-
>-#include <algorithm>
>-
>-#include "api/optional.h"
>-#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
>-#include "rtc_base/checks.h"
>-
>-namespace webrtc {
>-
>-namespace {
>-rtc::Optional<double> LinearFitSlope(
>-    const std::deque<std::pair<double, double>>& points) {
>-  RTC_DCHECK(points.size() >= 2);
>-  // Compute the "center of mass".
>-  double sum_x = 0;
>-  double sum_y = 0;
>-  for (const auto& point : points) {
>-    sum_x += point.first;
>-    sum_y += point.second;
>-  }
>-  double x_avg = sum_x / points.size();
>-  double y_avg = sum_y / points.size();
>-  // Compute the slope k = \sum (x_i-x_avg)(y_i-y_avg) / \sum (x_i-x_avg)^2
>-  double numerator = 0;
>-  double denominator = 0;
>-  for (const auto& point : points) {
>-    numerator += (point.first - x_avg) * (point.second - y_avg);
>-    denominator += (point.first - x_avg) * (point.first - x_avg);
>-  }
>-  if (denominator == 0)
>-    return rtc::nullopt;
>-  return numerator / denominator;
>-}
>-}  // namespace
>-
>-enum { kDeltaCounterMax = 1000 };
>-
>-TrendlineEstimator::TrendlineEstimator(size_t window_size,
>-                                       double smoothing_coef,
>-                                       double threshold_gain)
>-    : window_size_(window_size),
>-      smoothing_coef_(smoothing_coef),
>-      threshold_gain_(threshold_gain),
>-      num_of_deltas_(0),
>-      first_arrival_time_ms(-1),
>-      accumulated_delay_(0),
>-      smoothed_delay_(0),
>-      delay_hist_(),
>-      trendline_(0) {}
>-
>-TrendlineEstimator::~TrendlineEstimator() {}
>-
>-void TrendlineEstimator::Update(double recv_delta_ms,
>-                                double send_delta_ms,
>-                                int64_t arrival_time_ms) {
>-  const double delta_ms = recv_delta_ms - send_delta_ms;
>-  ++num_of_deltas_;
>-  if (num_of_deltas_ > kDeltaCounterMax)
>-    num_of_deltas_ = kDeltaCounterMax;
>-  if (first_arrival_time_ms == -1)
>-    first_arrival_time_ms = arrival_time_ms;
>-
>-  // Exponential backoff filter.
>-  accumulated_delay_ += delta_ms;
>-  BWE_TEST_LOGGING_PLOT(1, "accumulated_delay_ms", arrival_time_ms,
>-                        accumulated_delay_);
>-  smoothed_delay_ = smoothing_coef_ * smoothed_delay_ +
>-                    (1 - smoothing_coef_) * accumulated_delay_;
>-  BWE_TEST_LOGGING_PLOT(1, "smoothed_delay_ms", arrival_time_ms,
>-                        smoothed_delay_);
>-
>-  // Simple linear regression.
>-  delay_hist_.push_back(std::make_pair(
>-      static_cast<double>(arrival_time_ms - first_arrival_time_ms),
>-      smoothed_delay_));
>-  if (delay_hist_.size() > window_size_)
>-    delay_hist_.pop_front();
>-  if (delay_hist_.size() == window_size_) {
>-    // Only update trendline_ if it is possible to fit a line to the data.
>-    trendline_ = LinearFitSlope(delay_hist_).value_or(trendline_);
>-  }
>-
>-  BWE_TEST_LOGGING_PLOT(1, "trendline_slope", arrival_time_ms, trendline_);
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/BUILD.gn
>index 41d9a3e8bcc..e910adf2bf7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/BUILD.gn
>@@ -29,8 +29,8 @@ rtc_static_library("primitives") {
> 
>   deps = [
>     "../..:webrtc_common",
>-    "../../:typedefs",
>     "../../rtc_base:checks",
>+    "//third_party/abseil-cpp/absl/memory",
>   ]
> 
>   if (!build_with_mozilla) {
>@@ -45,6 +45,7 @@ if (rtc_include_tests) {
>     sources = []
>     deps = [
>       "../../rtc_base:checks",
>+      "//third_party/abseil-cpp/absl/memory",
>     ]
>     if (rtc_desktop_capture_supported) {
>       deps += [
>@@ -53,6 +54,7 @@ if (rtc_include_tests) {
>         ":screen_drawer",
>         "../../rtc_base:rtc_base",
>         "../../rtc_base:rtc_base_approved",
>+        "../../rtc_base/third_party/base64",
>         "../../system_wrappers",
>         "../../test:test_support",
>         "../../test:video_test_support",
>@@ -94,12 +96,11 @@ if (rtc_include_tests) {
>       ":desktop_capture_mock",
>       ":primitives",
>       "../..:webrtc_common",
>-      "../../:typedefs",
>       "../../rtc_base:checks",
>       "../../rtc_base:rtc_base_approved",
>       "../../system_wrappers:cpu_features_api",
>       "../../test:test_support",
>-      "//testing/gmock",
>+      "//third_party/abseil-cpp/absl/memory",
>     ]
>     if (rtc_desktop_capture_supported) {
>       sources += [
>@@ -110,13 +111,9 @@ if (rtc_include_tests) {
>       ]
>       deps += [ ":desktop_capture_mock" ]
>     }
>-    if (!build_with_chromium && is_clang) {
>-      # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>-      suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-    }
>   }
> 
>-  source_set("screen_drawer") {
>+  rtc_source_set("screen_drawer") {
>     testonly = true
> 
>     public_deps = [
>@@ -127,8 +124,6 @@ if (rtc_include_tests) {
>       "screen_drawer.cc",
>       "screen_drawer.h",
>       "screen_drawer_linux.cc",
>-      "screen_drawer_lock_posix.cc",
>-      "screen_drawer_lock_posix.h",
>       "screen_drawer_mac.cc",
>       "screen_drawer_win.cc",
>     ]
>@@ -138,15 +133,22 @@ if (rtc_include_tests) {
>       "../../rtc_base:checks",
>       "../../rtc_base:rtc_base_approved",
>       "../../system_wrappers",
>+      "//third_party/abseil-cpp/absl/memory",
>     ]
>+
>+    if (is_posix || is_fuchsia) {
>+      sources += [
>+        "screen_drawer_lock_posix.cc",
>+        "screen_drawer_lock_posix.h",
>+      ]
>+    }
>   }
> 
>-  source_set("desktop_capture_mock") {
>+  rtc_source_set("desktop_capture_mock") {
>     testonly = true
> 
>     public_deps = [
>       ":desktop_capture",
>-      "//testing/gmock",
>     ]
> 
>     sources = [
>@@ -177,8 +179,16 @@ if (is_mac) {
>     visibility = [ ":desktop_capture" ]
>     sources = [
>       "mac/desktop_configuration.mm",
>+      "mac/desktop_frame_cgimage.h",
>+      "mac/desktop_frame_cgimage.mm",
>+      "mac/desktop_frame_iosurface.h",
>+      "mac/desktop_frame_iosurface.mm",
>+      "mac/desktop_frame_provider.h",
>+      "mac/desktop_frame_provider.mm",
>+      "mac/screen_capturer_mac.h",
>+      "mac/screen_capturer_mac.mm",
>       "mouse_cursor_monitor_mac.mm",
>-      "screen_capturer_mac.mm",
>+      "screen_capturer_darwin.mm",
>       "window_capturer_mac.mm",
>       "window_finder_mac.h",
>       "window_finder_mac.mm",
>@@ -189,12 +199,14 @@ if (is_mac) {
>       "../../rtc_base:checks",
>       "../../rtc_base:rtc_base",
>       "../../rtc_base:rtc_base_approved",
>+      "../../rtc_base/synchronization:rw_lock_wrapper",
>       "../../sdk:common_objc",
>+      "//third_party/abseil-cpp/absl/memory",
>     ]
>     libs = [
>       "AppKit.framework",
>       "IOKit.framework",
>-      "OpenGL.framework",
>+      "IOSurface.framework",
>     ]
>   }
> }
>@@ -241,8 +253,6 @@ rtc_static_library("desktop_capture_generic") {
>     "mac/desktop_configuration_monitor.h",
>     "mac/full_screen_chrome_window_detector.cc",
>     "mac/full_screen_chrome_window_detector.h",
>-    "mac/scoped_pixel_buffer_object.cc",
>-    "mac/scoped_pixel_buffer_object.h",
>     "mac/window_list_utils.cc",
>     "mac/window_list_utils.h",
>     "mouse_cursor.cc",
>@@ -361,13 +371,15 @@ rtc_static_library("desktop_capture_generic") {
>   deps = [
>     ":primitives",
>     "../..:webrtc_common",
>-    "../../:typedefs",
>     "../../api:refcountedbase",
>     "../../rtc_base:checks",
>     "../../rtc_base:rtc_base",  # TODO(kjellander): Cleanup in bugs.webrtc.org/3806.
>+    "../../rtc_base/synchronization:rw_lock_wrapper",
>+    "../../rtc_base/system:arch",
>     "../../system_wrappers",
>     "../../system_wrappers:cpu_features_api",
>     "../../system_wrappers:metrics_api",
>+    "//third_party/abseil-cpp/absl/memory",
>   ]
> 
>   if (build_with_mozilla) {
>@@ -391,7 +403,7 @@ if (use_desktop_capture_differ_sse2) {
>       "differ_vector_sse2.h",
>     ]
> 
>-    if (is_posix) {
>+    if (is_posix || is_fuchsia) {
>       cflags = [ "-msse2" ]
>     }
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/DEPS b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/DEPS
>index 3b8a639f5ee..8c894c44301 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/DEPS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/DEPS
>@@ -4,6 +4,15 @@ include_rules = [
> ]
> 
> specific_include_rules = {
>+  "desktop_frame_cgimage\.h": [
>+    "+sdk/objc",
>+  ],
>+  "desktop_frame_iosurface\.h": [
>+    "+sdk/objc",
>+  ],
>+  "desktop_frame_provider\.h": [
>+    "+sdk/objc",
>+  ],
>   "screen_capturer_mac\.mm": [
>     "+sdk/objc",
>   ],
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/blank_detector_desktop_capturer_wrapper.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/blank_detector_desktop_capturer_wrapper.cc
>index 722f5dc53aa..6bf8d512a84 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/blank_detector_desktop_capturer_wrapper.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/blank_detector_desktop_capturer_wrapper.cc
>@@ -22,8 +22,7 @@ namespace webrtc {
> BlankDetectorDesktopCapturerWrapper::BlankDetectorDesktopCapturerWrapper(
>     std::unique_ptr<DesktopCapturer> capturer,
>     RgbaColor blank_pixel)
>-    : capturer_(std::move(capturer)),
>-      blank_pixel_(blank_pixel) {
>+    : capturer_(std::move(capturer)), blank_pixel_(blank_pixel) {
>   RTC_DCHECK(capturer_);
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/blank_detector_desktop_capturer_wrapper_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/blank_detector_desktop_capturer_wrapper_unittest.cc
>index f9cfee026ff..d4ec0082f50 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/blank_detector_desktop_capturer_wrapper_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/blank_detector_desktop_capturer_wrapper_unittest.cc
>@@ -49,7 +49,7 @@ class BlankDetectorDesktopCapturerWrapperTest
> };
> 
> BlankDetectorDesktopCapturerWrapperTest::
>-BlankDetectorDesktopCapturerWrapperTest() {
>+    BlankDetectorDesktopCapturerWrapperTest() {
>   frame_generator_.size()->set(frame_width_, frame_height_);
>   frame_generator_.set_desktop_frame_painter(&painter_);
>   std::unique_ptr<DesktopCapturer> capturer(new FakeDesktopCapturer());
>@@ -63,7 +63,7 @@ BlankDetectorDesktopCapturerWrapperTest() {
> }
> 
> BlankDetectorDesktopCapturerWrapperTest::
>-~BlankDetectorDesktopCapturerWrapperTest() = default;
>+    ~BlankDetectorDesktopCapturerWrapperTest() = default;
> 
> void BlankDetectorDesktopCapturerWrapperTest::OnCaptureResult(
>     DesktopCapturer::Result result,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/capture_result_desktop_capturer_wrapper.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/capture_result_desktop_capturer_wrapper.cc
>index b15ce2b0fde..5e0d264cc29 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/capture_result_desktop_capturer_wrapper.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/capture_result_desktop_capturer_wrapper.cc
>@@ -20,13 +20,12 @@ namespace webrtc {
> CaptureResultDesktopCapturerWrapper::CaptureResultDesktopCapturerWrapper(
>     std::unique_ptr<DesktopCapturer> base_capturer,
>     ResultObserver* observer)
>-    : DesktopCapturerWrapper(std::move(base_capturer)),
>-      observer_(observer) {
>+    : DesktopCapturerWrapper(std::move(base_capturer)), observer_(observer) {
>   RTC_DCHECK(observer_);
> }
> 
>-CaptureResultDesktopCapturerWrapper::
>-~CaptureResultDesktopCapturerWrapper() = default;
>+CaptureResultDesktopCapturerWrapper::~CaptureResultDesktopCapturerWrapper() =
>+    default;
> 
> void CaptureResultDesktopCapturerWrapper::Start(Callback* callback) {
>   if ((callback_ == nullptr) != (callback == nullptr)) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/capture_result_desktop_capturer_wrapper.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/capture_result_desktop_capturer_wrapper.h
>index 3f97bc5fc46..ffce2961642 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/capture_result_desktop_capturer_wrapper.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/capture_result_desktop_capturer_wrapper.h
>@@ -21,9 +21,8 @@ namespace webrtc {
> // A DesktopCapturerWrapper implementation to capture the result of
> // |base_capturer|. Derived classes are expected to provide a ResultObserver
> // implementation to observe the DesktopFrame returned by |base_capturer_|.
>-class CaptureResultDesktopCapturerWrapper
>-    : public DesktopCapturerWrapper,
>-      public DesktopCapturer::Callback {
>+class CaptureResultDesktopCapturerWrapper : public DesktopCapturerWrapper,
>+                                            public DesktopCapturer::Callback {
>  public:
>   using Callback = DesktopCapturer::Callback;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/cropped_desktop_frame.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/cropped_desktop_frame.h
>index f6cac6c7a21..59dced40179 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/cropped_desktop_frame.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/cropped_desktop_frame.h
>@@ -27,4 +27,3 @@ std::unique_ptr<DesktopFrame> CreateCroppedDesktopFrame(
> }  // namespace webrtc
> 
> #endif  // MODULES_DESKTOP_CAPTURE_CROPPED_DESKTOP_FRAME_H_
>-
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/cropped_desktop_frame_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/cropped_desktop_frame_unittest.cc
>index d15f207dea9..34aff142740 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/cropped_desktop_frame_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/cropped_desktop_frame_unittest.cc
>@@ -11,15 +11,15 @@
> #include <memory>
> #include <utility>
> 
>+#include "absl/memory/memory.h"
> #include "modules/desktop_capture/cropped_desktop_frame.h"
> #include "modules/desktop_capture/desktop_frame.h"
>-#include "rtc_base/ptr_util.h"
> #include "test/gtest.h"
> 
> namespace webrtc {
> 
> std::unique_ptr<DesktopFrame> CreateTestFrame() {
>-  return rtc::MakeUnique<BasicDesktopFrame>(DesktopSize(10, 20));
>+  return absl::make_unique<BasicDesktopFrame>(DesktopSize(10, 20));
> }
> 
> TEST(CroppedDesktopFrameTest, DoNotCreateWrapperIfSizeIsNotChanged) {
>@@ -32,15 +32,16 @@ TEST(CroppedDesktopFrameTest, DoNotCreateWrapperIfSizeIsNotChanged) {
> }
> 
> TEST(CroppedDesktopFrameTest, ReturnNullptrIfSizeIsNotSufficient) {
>-  ASSERT_EQ(nullptr, CreateCroppedDesktopFrame(
>-      CreateTestFrame(), DesktopRect::MakeWH(11, 10)));
>+  ASSERT_EQ(nullptr, CreateCroppedDesktopFrame(CreateTestFrame(),
>+                                               DesktopRect::MakeWH(11, 10)));
> }
> 
> TEST(CroppedDesktopFrameTest, ReturnNullIfCropRegionIsOutOfBounds) {
>   std::unique_ptr<DesktopFrame> frame = CreateTestFrame();
>   frame->set_top_left(DesktopVector(100, 200));
>-  ASSERT_EQ(nullptr, CreateCroppedDesktopFrame(std::move(frame),
>-      DesktopRect::MakeLTRB(101, 203, 109, 218)));
>+  ASSERT_EQ(nullptr,
>+            CreateCroppedDesktopFrame(
>+                std::move(frame), DesktopRect::MakeLTRB(101, 203, 109, 218)));
> }
> 
> TEST(CroppedDesktopFrameTest, CropASubArea) {
>@@ -56,7 +57,7 @@ TEST(CroppedDesktopFrameTest, SetTopLeft) {
>   std::unique_ptr<DesktopFrame> frame = CreateTestFrame();
>   frame->set_top_left(DesktopVector(100, 200));
>   frame = CreateCroppedDesktopFrame(std::move(frame),
>-      DesktopRect::MakeLTRB(1, 3, 9, 18));
>+                                    DesktopRect::MakeLTRB(1, 3, 9, 18));
>   ASSERT_EQ(frame->size().width(), 8);
>   ASSERT_EQ(frame->size().height(), 15);
>   ASSERT_EQ(frame->top_left().x(), 101);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/cropping_window_capturer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/cropping_window_capturer.h
>index 9fbfddaa81a..c889801da3c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/cropping_window_capturer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/cropping_window_capturer.h
>@@ -13,8 +13,8 @@
> 
> #include <memory>
> 
>-#include "modules/desktop_capture/desktop_capturer.h"
> #include "modules/desktop_capture/desktop_capture_options.h"
>+#include "modules/desktop_capture/desktop_capturer.h"
> 
> namespace webrtc {
> 
>@@ -76,4 +76,3 @@ class CroppingWindowCapturer : public DesktopCapturer,
> }  // namespace webrtc
> 
> #endif  // MODULES_DESKTOP_CAPTURE_CROPPING_WINDOW_CAPTURER_H_
>-
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/cropping_window_capturer_win.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/cropping_window_capturer_win.cc
>index 6707236b3e8..20c54663cb3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/cropping_window_capturer_win.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/cropping_window_capturer_win.cc
>@@ -13,33 +13,48 @@
> #include "modules/desktop_capture/win/screen_capture_utils.h"
> #include "modules/desktop_capture/win/window_capture_utils.h"
> #include "rtc_base/logging.h"
>+#include "rtc_base/trace_event.h"
> #include "rtc_base/win32.h"
> 
> namespace webrtc {
> 
> namespace {
> 
>+const size_t kTitleLength = 256;
>+
> // Used to pass input/output data during the EnumWindow call for verifying if
> // the selected window is on top.
> struct TopWindowVerifierContext {
>   TopWindowVerifierContext(HWND selected_window,
>                            HWND excluded_window,
>-                           DesktopRect selected_window_rect)
>+                           DesktopRect selected_window_rect,
>+                           WindowCaptureHelperWin* window_capture_helper)
>       : selected_window(selected_window),
>         excluded_window(excluded_window),
>         selected_window_rect(selected_window_rect),
>+        window_capture_helper(window_capture_helper),
>         is_top_window(false) {
>     RTC_DCHECK_NE(selected_window, excluded_window);
>+
>+    GetWindowText(selected_window, selected_window_title, kTitleLength);
>+    GetWindowThreadProcessId(selected_window, &selected_window_process_id);
>   }
> 
>   const HWND selected_window;
>   const HWND excluded_window;
>   const DesktopRect selected_window_rect;
>+  WindowCaptureHelperWin* window_capture_helper;
>+  WCHAR selected_window_title[kTitleLength];
>+  DWORD selected_window_process_id;
>   bool is_top_window;
> };
> 
> // The function is called during EnumWindow for every window enumerated and is
> // responsible for verifying if the selected window is on top.
>+// Return TRUE to continue enumerating if the current window belongs to the
>+// selected window or is to be ignored.
>+// Return FALSE to stop enumerating if the selected window is found or decided
>+// if it's on top most.
> BOOL CALLBACK TopWindowVerifier(HWND hwnd, LPARAM param) {
>   TopWindowVerifierContext* context =
>       reinterpret_cast<TopWindowVerifierContext*>(param);
>@@ -54,8 +69,19 @@ BOOL CALLBACK TopWindowVerifier(HWND hwnd, LPARAM param) {
>     return TRUE;
>   }
> 
>-  // Ignore hidden or minimized window.
>-  if (IsIconic(hwnd) || !IsWindowVisible(hwnd)) {
>+  // Ignore invisible window on current desktop.
>+  if (!context->window_capture_helper->IsWindowVisibleOnCurrentDesktop(hwnd)) {
>+    return TRUE;
>+  }
>+
>+  // Ignore Chrome notification windows, especially the notification for the
>+  // ongoing window sharing.
>+  // Notes:
>+  // - This only works with notifications from Chrome, not other Apps.
>+  // - All notifications from Chrome will be ignored.
>+  // - This may cause part or whole of notification window being cropped into
>+  // the capturing of the target window if there is overlapping.
>+  if (context->window_capture_helper->IsWindowChromeNotification(hwnd)) {
>     return TRUE;
>   }
> 
>@@ -71,21 +97,20 @@ BOOL CALLBACK TopWindowVerifier(HWND hwnd, LPARAM param) {
>     return TRUE;
>   }
> 
>-  // If |hwnd| has no title and belongs to the same process, assume it's a
>-  // tooltip or context menu from the selected window and ignore it.
>+  // If |hwnd| has no title or has same title as the selected window (i.e.
>+  // Window Media Player consisting of several sibling windows) and belongs to
>+  // the same process, assume it's a tooltip or context menu or sibling window
>+  // from the selected window and ignore it.
>   // TODO(zijiehe): This check cannot cover the case where tooltip or context
>   // menu of the child-window is covering the main window. See
>   // https://bugs.chromium.org/p/webrtc/issues/detail?id=8062 for details.
>-  const size_t kTitleLength = 32;
>   WCHAR window_title[kTitleLength];
>   GetWindowText(hwnd, window_title, kTitleLength);
>-  if (wcsnlen_s(window_title, kTitleLength) == 0) {
>+  if (wcsnlen_s(window_title, kTitleLength) == 0 ||
>+      wcscmp(window_title, context->selected_window_title) == 0) {
>     DWORD enumerated_window_process_id;
>-    DWORD selected_window_process_id;
>     GetWindowThreadProcessId(hwnd, &enumerated_window_process_id);
>-    GetWindowThreadProcessId(context->selected_window,
>-                             &selected_window_process_id);
>-    if (selected_window_process_id == enumerated_window_process_id) {
>+    if (context->selected_window_process_id == enumerated_window_process_id) {
>       return TRUE;
>     }
>   }
>@@ -128,8 +153,7 @@ BOOL CALLBACK TopWindowVerifier(HWND hwnd, LPARAM param) {
> 
> class CroppingWindowCapturerWin : public CroppingWindowCapturer {
>  public:
>-  CroppingWindowCapturerWin(
>-      const DesktopCaptureOptions& options)
>+  CroppingWindowCapturerWin(const DesktopCaptureOptions& options)
>       : CroppingWindowCapturer(options) {}
> 
>  private:
>@@ -140,17 +164,17 @@ class CroppingWindowCapturerWin : public CroppingWindowCapturer {
>   // rectangular, or the rect from GetWindowRect if the region is not set.
>   DesktopRect window_region_rect_;
> 
>-  AeroChecker aero_checker_;
>+  WindowCaptureHelperWin window_capture_helper_;
> };
> 
> bool CroppingWindowCapturerWin::ShouldUseScreenCapturer() {
>-  if (!rtc::IsWindows8OrLater() && aero_checker_.IsAeroEnabled()) {
>+  if (!rtc::IsWindows8OrLater() && window_capture_helper_.IsAeroEnabled()) {
>     return false;
>   }
> 
>   const HWND selected = reinterpret_cast<HWND>(selected_window());
>-  // Check if the window is hidden or minimized.
>-  if (IsIconic(selected) || !IsWindowVisible(selected)) {
>+  // Check if the window is visible on current desktop.
>+  if (!window_capture_helper_.IsWindowVisibleOnCurrentDesktop(selected)) {
>     return false;
>   }
> 
>@@ -197,8 +221,8 @@ bool CroppingWindowCapturerWin::ShouldUseScreenCapturer() {
>   if (region_type == SIMPLEREGION) {
>     // The |region_rect| returned from GetRgnBox() is always in window
>     // coordinate.
>-    region_rect.Translate(
>-        window_region_rect_.left(), window_region_rect_.top());
>+    region_rect.Translate(window_region_rect_.left(),
>+                          window_region_rect_.top());
>     // MSDN: The window region determines the area *within* the window where the
>     // system permits drawing.
>     // https://msdn.microsoft.com/en-us/library/windows/desktop/dd144950(v=vs.85).aspx.
>@@ -223,8 +247,9 @@ bool CroppingWindowCapturerWin::ShouldUseScreenCapturer() {
>   // windows, context menus, and |excluded_window_|.
>   // |content_rect| is preferred, see the comments in TopWindowVerifier()
>   // function.
>-  TopWindowVerifierContext context(
>-      selected, reinterpret_cast<HWND>(excluded_window()), content_rect);
>+  TopWindowVerifierContext context(selected,
>+                                   reinterpret_cast<HWND>(excluded_window()),
>+                                   content_rect, &window_capture_helper_);
>   const LPARAM enum_param = reinterpret_cast<LPARAM>(&context);
>   EnumWindows(&TopWindowVerifier, enum_param);
>   if (!context.is_top_window) {
>@@ -240,6 +265,8 @@ bool CroppingWindowCapturerWin::ShouldUseScreenCapturer() {
> }
> 
> DesktopRect CroppingWindowCapturerWin::GetWindowRectInVirtualScreen() {
>+  TRACE_EVENT0("webrtc",
>+               "CroppingWindowCapturerWin::GetWindowRectInVirtualScreen");
>   DesktopRect window_rect;
>   HWND hwnd = reinterpret_cast<HWND>(selected_window());
>   if (!GetCroppedWindowRect(hwnd, &window_rect, /* original_rect */ nullptr)) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_and_cursor_composer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_and_cursor_composer.cc
>index bc46a5c528b..67a6919b466 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_and_cursor_composer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_and_cursor_composer.cc
>@@ -14,13 +14,13 @@
> 
> #include <utility>
> 
>+#include "absl/memory/memory.h"
> #include "modules/desktop_capture/desktop_capturer.h"
> #include "modules/desktop_capture/desktop_frame.h"
> #include "modules/desktop_capture/mouse_cursor.h"
> #include "modules/desktop_capture/mouse_cursor_monitor.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/constructormagic.h"
>-#include "rtc_base/ptr_util.h"
> 
> namespace webrtc {
> 
>@@ -28,8 +28,10 @@ namespace {
> 
> // Helper function that blends one image into another. Source image must be
> // pre-multiplied with the alpha channel. Destination is assumed to be opaque.
>-void AlphaBlend(uint8_t* dest, int dest_stride,
>-                const uint8_t* src, int src_stride,
>+void AlphaBlend(uint8_t* dest,
>+                int dest_stride,
>+                const uint8_t* src,
>+                int src_stride,
>                 const DesktopSize& size) {
>   for (int y = 0; y < size.height(); ++y) {
>     for (int x = 0; x < size.width(); ++x) {
>@@ -111,8 +113,7 @@ DesktopFrameWithCursor::DesktopFrameWithCursor(
>              cursor.image()->data() +
>                  origin_shift.y() * cursor.image()->stride() +
>                  origin_shift.x() * DesktopFrame::kBytesPerPixel,
>-             cursor.image()->stride(),
>-             target_rect.size());
>+             cursor.image()->stride(), target_rect.size());
> }
> 
> DesktopFrameWithCursor::~DesktopFrameWithCursor() {
>@@ -127,26 +128,16 @@ DesktopFrameWithCursor::~DesktopFrameWithCursor() {
> 
> }  // namespace
> 
>-DesktopAndCursorComposer::DesktopAndCursorComposer(
>-    DesktopCapturer* desktop_capturer,
>-    MouseCursorMonitor* mouse_monitor)
>-    : DesktopAndCursorComposer(desktop_capturer, mouse_monitor, false) {}
>-
> DesktopAndCursorComposer::DesktopAndCursorComposer(
>     std::unique_ptr<DesktopCapturer> desktop_capturer,
>     const DesktopCaptureOptions& options)
>     : DesktopAndCursorComposer(desktop_capturer.release(),
>-                               MouseCursorMonitor::Create(options).release(),
>-                               true) {}
>+                               MouseCursorMonitor::Create(options).release()) {}
> 
> DesktopAndCursorComposer::DesktopAndCursorComposer(
>     DesktopCapturer* desktop_capturer,
>-    MouseCursorMonitor* mouse_monitor,
>-    bool use_desktop_relative_cursor_position)
>-    : desktop_capturer_(desktop_capturer),
>-      mouse_monitor_(mouse_monitor),
>-      use_desktop_relative_cursor_position_(
>-          use_desktop_relative_cursor_position) {
>+    MouseCursorMonitor* mouse_monitor)
>+    : desktop_capturer_(desktop_capturer), mouse_monitor_(mouse_monitor) {
>   RTC_DCHECK(desktop_capturer_);
> }
> 
>@@ -178,19 +169,15 @@ void DesktopAndCursorComposer::OnCaptureResult(
>     DesktopCapturer::Result result,
>     std::unique_ptr<DesktopFrame> frame) {
>   if (frame && cursor_) {
>-    if (use_desktop_relative_cursor_position_) {
>-      if (frame->rect().Contains(cursor_position_) &&
>-          !desktop_capturer_->IsOccluded(cursor_position_)) {
>-        const DesktopVector relative_position =
>-            cursor_position_.subtract(frame->top_left());
>-        frame = rtc::MakeUnique<DesktopFrameWithCursor>(
>-            std::move(frame), *cursor_, relative_position);
>-      }
>-    } else {
>-      if (cursor_state_ == MouseCursorMonitor::INSIDE) {
>-        frame = rtc::MakeUnique<DesktopFrameWithCursor>(
>-            std::move(frame), *cursor_, cursor_position_);
>-      }
>+    if (frame->rect().Contains(cursor_position_) &&
>+        !desktop_capturer_->IsOccluded(cursor_position_)) {
>+      const float scale = frame->scale_factor();
>+      DesktopVector relative_position =
>+          cursor_position_.subtract(frame->top_left());
>+      relative_position.set(relative_position.x() * scale,
>+                            relative_position.y() * scale);
>+      frame = absl::make_unique<DesktopFrameWithCursor>(
>+          std::move(frame), *cursor_, relative_position);
>     }
>   }
> 
>@@ -204,17 +191,12 @@ void DesktopAndCursorComposer::OnMouseCursor(MouseCursor* cursor) {
> void DesktopAndCursorComposer::OnMouseCursorPosition(
>     MouseCursorMonitor::CursorState state,
>     const DesktopVector& position) {
>-  if (!use_desktop_relative_cursor_position_) {
>-    cursor_state_ = state;
>-    cursor_position_ = position;
>-  }
>+  RTC_NOTREACHED();
> }
> 
> void DesktopAndCursorComposer::OnMouseCursorPosition(
>     const DesktopVector& position) {
>-  if (use_desktop_relative_cursor_position_) {
>-    cursor_position_ = position;
>-  }
>+  cursor_position_ = position;
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_and_cursor_composer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_and_cursor_composer.h
>index c88075a9863..7dff7101f35 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_and_cursor_composer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_and_cursor_composer.h
>@@ -20,23 +20,12 @@
> 
> namespace webrtc {
> 
>-template <bool use_desktop_relative_cursor_position>
>-class DesktopAndCursorComposerTest;
>-
> // A wrapper for DesktopCapturer that also captures mouse using specified
> // MouseCursorMonitor and renders it on the generated streams.
> class DesktopAndCursorComposer : public DesktopCapturer,
>                                  public DesktopCapturer::Callback,
>                                  public MouseCursorMonitor::Callback {
>  public:
>-  // Creates a new blender that captures mouse cursor using |mouse_monitor| and
>-  // renders it into the frames generated by |desktop_capturer|. If
>-  // |mouse_monitor| is NULL the frames are passed unmodified. Takes ownership
>-  // of both arguments.
>-  // Deprecated: use the constructor below.
>-  DesktopAndCursorComposer(DesktopCapturer* desktop_capturer,
>-                           MouseCursorMonitor* mouse_monitor);
>-
>   // Creates a new blender that captures mouse cursor using
>   // MouseCursorMonitor::Create(options) and renders it into the frames
>   // generated by |desktop_capturer|.
>@@ -54,14 +43,12 @@ class DesktopAndCursorComposer : public DesktopCapturer,
> 
>  private:
>   // Allows test cases to use a fake MouseCursorMonitor implementation.
>-  friend class DesktopAndCursorComposerTest<true>;
>-  friend class DesktopAndCursorComposerTest<false>;
>+  friend class DesktopAndCursorComposerTest;
> 
>   // Constructor to delegate both deprecated and new constructors and allows
>   // test cases to use a fake MouseCursorMonitor implementation.
>   DesktopAndCursorComposer(DesktopCapturer* desktop_capturer,
>-                           MouseCursorMonitor* mouse_monitor,
>-                           bool use_desktop_relative_cursor_position);
>+                           MouseCursorMonitor* mouse_monitor);
> 
>   // DesktopCapturer::Callback interface.
>   void OnCaptureResult(DesktopCapturer::Result result,
>@@ -75,20 +62,10 @@ class DesktopAndCursorComposer : public DesktopCapturer,
> 
>   const std::unique_ptr<DesktopCapturer> desktop_capturer_;
>   const std::unique_ptr<MouseCursorMonitor> mouse_monitor_;
>-  // This is a temporary flag to decide how to use the |mouse_monitor_|.
>-  // If it's true, DesktopAndCursorComposer will use the absolute position from
>-  // MouseCursorMonitor but ignore the MouseCursorMonitor::CursorState.
>-  // Otherwise MouseCursorMonitor::CursorState is respected. This flag is false
>-  // when the deprecated constructor is used, and true when the new one is used.
>-  // This flag will be removed together with the deprecated constructor.
>-  const bool use_desktop_relative_cursor_position_;
> 
>   DesktopCapturer::Callback* callback_;
> 
>   std::unique_ptr<MouseCursor> cursor_;
>-  // This field is irrelevant if |use_desktop_relative_cursor_position_| is
>-  // true.
>-  MouseCursorMonitor::CursorState cursor_state_;
>   DesktopVector cursor_position_;
> 
>   RTC_DISALLOW_COPY_AND_ASSIGN(DesktopAndCursorComposer);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_and_cursor_composer_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_and_cursor_composer_unittest.cc
>index 0725facfcb1..c94a3cdd3bf 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_and_cursor_composer_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_and_cursor_composer_unittest.cc
>@@ -11,8 +11,8 @@
> #include <memory>
> 
> #include "modules/desktop_capture/desktop_and_cursor_composer.h"
>-#include "modules/desktop_capture/desktop_capturer.h"
> #include "modules/desktop_capture/desktop_capture_options.h"
>+#include "modules/desktop_capture/desktop_capturer.h"
> #include "modules/desktop_capture/desktop_frame.h"
> #include "modules/desktop_capture/mouse_cursor.h"
> #include "modules/desktop_capture/shared_desktop_frame.h"
>@@ -30,9 +30,15 @@ const int kCursorHeight = 10;
> 
> const int kTestCursorSize = 3;
> const uint32_t kTestCursorData[kTestCursorSize][kTestCursorSize] = {
>-  { 0xffffffff, 0x99990000, 0xaa222222, },
>-  { 0x88008800, 0xaa0000aa, 0xaa333333, },
>-  { 0x00000000, 0xaa0000aa, 0xaa333333, },
>+    {
>+        0xffffffff, 0x99990000, 0xaa222222,
>+    },
>+    {
>+        0x88008800, 0xaa0000aa, 0xaa333333,
>+    },
>+    {
>+        0x00000000, 0xaa0000aa, 0xaa333333,
>+    },
> };
> 
> uint32_t GetFakeFramePixelValue(const DesktopVector& p) {
>@@ -131,7 +137,6 @@ class FakeMouseMonitor : public MouseCursorMonitor {
>       callback_->OnMouseCursor(new MouseCursor(image.release(), hotspot_));
>     }
> 
>-    callback_->OnMouseCursorPosition(state_, position_);
>     callback_->OnMouseCursorPosition(position_);
>   }
> 
>@@ -167,16 +172,13 @@ void VerifyFrame(const DesktopFrame& frame,
> 
> }  // namespace
> 
>-template <bool use_desktop_relative_cursor_position>
> class DesktopAndCursorComposerTest : public testing::Test,
>                                      public DesktopCapturer::Callback {
>  public:
>   DesktopAndCursorComposerTest()
>       : fake_screen_(new FakeScreenCapturer()),
>         fake_cursor_(new FakeMouseMonitor()),
>-        blender_(fake_screen_,
>-                 fake_cursor_,
>-                 use_desktop_relative_cursor_position) {
>+        blender_(fake_screen_, fake_cursor_) {
>     blender_.Start(this);
>   }
> 
>@@ -195,40 +197,17 @@ class DesktopAndCursorComposerTest : public testing::Test,
>   std::unique_ptr<DesktopFrame> frame_;
> };
> 
>-using DesktopAndCursorComposerWithRelativePositionTest =
>-    DesktopAndCursorComposerTest<false>;
>-
>-// Verify DesktopAndCursorComposer can handle the case when the screen capturer
>-// fails.
>-TEST_F(DesktopAndCursorComposerWithRelativePositionTest, Error) {
>-  fake_cursor_->SetHotspot(DesktopVector());
>-  fake_cursor_->SetState(MouseCursorMonitor::INSIDE, DesktopVector());
>-  fake_screen_->SetNextFrame(nullptr);
>-
>-  blender_.CaptureFrame();
>-
>-  EXPECT_FALSE(frame_);
>-}
>-
>-TEST_F(DesktopAndCursorComposerWithRelativePositionTest, Blend) {
>+TEST_F(DesktopAndCursorComposerTest, CursorShouldBeIgnoredIfNoFrameCaptured) {
>   struct {
>     int x, y;
>     int hotspot_x, hotspot_y;
>     bool inside;
>   } tests[] = {
>-    {0, 0, 0, 0, true},
>-    {50, 50, 0, 0, true},
>-    {100, 50, 0, 0, true},
>-    {50, 100, 0, 0, true},
>-    {100, 100, 0, 0, true},
>-    {0, 0, 2, 5, true},
>-    {1, 1, 2, 5, true},
>-    {50, 50, 2, 5, true},
>-    {100, 100, 2, 5, true},
>-    {0, 0, 5, 2, true},
>-    {50, 50, 5, 2, true},
>-    {100, 100, 5, 2, true},
>-    {0, 0, 0, 0, false},
>+      {0, 0, 0, 0, true},    {50, 50, 0, 0, true},   {100, 50, 0, 0, true},
>+      {50, 100, 0, 0, true}, {100, 100, 0, 0, true}, {0, 0, 2, 5, true},
>+      {1, 1, 2, 5, true},    {50, 50, 2, 5, true},   {100, 100, 2, 5, true},
>+      {0, 0, 5, 2, true},    {50, 50, 5, 2, true},   {100, 100, 5, 2, true},
>+      {0, 0, 0, 0, false},
>   };
> 
>   for (size_t i = 0; i < arraysize(tests); i++) {
>@@ -245,23 +224,15 @@ TEST_F(DesktopAndCursorComposerWithRelativePositionTest, Blend) {
> 
>     std::unique_ptr<SharedDesktopFrame> frame(
>         SharedDesktopFrame::Wrap(CreateTestFrame()));
>-    fake_screen_->SetNextFrame(frame->Share());
> 
>     blender_.CaptureFrame();
>-
>-    VerifyFrame(*frame_, state, pos);
>-
>-    // Verify that the cursor is erased before the frame buffer is returned to
>-    // the screen capturer.
>-    frame_.reset();
>-    VerifyFrame(*frame, MouseCursorMonitor::OUTSIDE, DesktopVector());
>+    // If capturer captured nothing, then cursor should be ignored, not matter
>+    // its state or position.
>+    EXPECT_EQ(frame_, nullptr);
>   }
> }
> 
>-using DesktopAndCursorComposerWithAbsolutePositionTest =
>-    DesktopAndCursorComposerTest<true>;
>-
>-TEST_F(DesktopAndCursorComposerWithAbsolutePositionTest,
>+TEST_F(DesktopAndCursorComposerTest,
>        CursorShouldBeIgnoredIfItIsOutOfDesktopFrame) {
>   std::unique_ptr<SharedDesktopFrame> frame(
>       SharedDesktopFrame::Wrap(CreateTestFrame()));
>@@ -272,19 +243,9 @@ TEST_F(DesktopAndCursorComposerWithAbsolutePositionTest,
>     int x;
>     int y;
>   } tests[] = {
>-    { 0, 0 },
>-    { 50, 50 },
>-    { 50, 150 },
>-    { 100, 150 },
>-    { 50, 200 },
>-    { 99, 200 },
>-    { 100, 199 },
>-    { 200, 300 },
>-    { 200, 299 },
>-    { 199, 300 },
>-    { -1, -1 },
>-    { -10000, -10000 },
>-    { 10000, 10000 },
>+      {0, 0},    {50, 50},         {50, 150},      {100, 150}, {50, 200},
>+      {99, 200}, {100, 199},       {200, 300},     {200, 299}, {199, 300},
>+      {-1, -1},  {-10000, -10000}, {10000, 10000},
>   };
>   for (size_t i = 0; i < arraysize(tests); i++) {
>     SCOPED_TRACE(i);
>@@ -298,8 +259,7 @@ TEST_F(DesktopAndCursorComposerWithAbsolutePositionTest,
>   }
> }
> 
>-TEST_F(DesktopAndCursorComposerWithAbsolutePositionTest,
>-       IsOccludedShouldBeConsidered) {
>+TEST_F(DesktopAndCursorComposerTest, IsOccludedShouldBeConsidered) {
>   std::unique_ptr<SharedDesktopFrame> frame(
>       SharedDesktopFrame::Wrap(CreateTestFrame()));
>   frame->set_top_left(DesktopVector(100, 200));
>@@ -309,12 +269,7 @@ TEST_F(DesktopAndCursorComposerWithAbsolutePositionTest,
>     int x;
>     int y;
>   } tests[] = {
>-    { 100, 200 },
>-    { 101, 200 },
>-    { 100, 201 },
>-    { 101, 201 },
>-    { 150, 250 },
>-    { 199, 299 },
>+      {100, 200}, {101, 200}, {100, 201}, {101, 201}, {150, 250}, {199, 299},
>   };
>   fake_screen_->set_is_occluded(true);
>   for (size_t i = 0; i < arraysize(tests); i++) {
>@@ -329,7 +284,7 @@ TEST_F(DesktopAndCursorComposerWithAbsolutePositionTest,
>   }
> }
> 
>-TEST_F(DesktopAndCursorComposerWithAbsolutePositionTest, CursorIncluded) {
>+TEST_F(DesktopAndCursorComposerTest, CursorIncluded) {
>   std::unique_ptr<SharedDesktopFrame> frame(
>       SharedDesktopFrame::Wrap(CreateTestFrame()));
>   frame->set_top_left(DesktopVector(100, 200));
>@@ -339,12 +294,7 @@ TEST_F(DesktopAndCursorComposerWithAbsolutePositionTest, CursorIncluded) {
>     int x;
>     int y;
>   } tests[] = {
>-    { 100, 200 },
>-    { 101, 200 },
>-    { 100, 201 },
>-    { 101, 201 },
>-    { 150, 250 },
>-    { 199, 299 },
>+      {100, 200}, {101, 200}, {100, 201}, {101, 201}, {150, 250}, {199, 299},
>   };
>   for (size_t i = 0; i < arraysize(tests); i++) {
>     SCOPED_TRACE(i);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_capture_options.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_capture_options.h
>index fec5ff0ac5a..60aa05c3fd3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_capture_options.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_capture_options.h
>@@ -72,6 +72,9 @@ class DesktopCaptureOptions {
>       rtc::scoped_refptr<FullScreenChromeWindowDetector> detector) {
>     full_screen_window_detector_ = detector;
>   }
>+
>+  bool allow_iosurface() const { return allow_iosurface_; }
>+  void set_allow_iosurface(bool allow) { allow_iosurface_ = allow; }
> #endif
> 
>   // Flag indicating that the capturer should use screen change notifications.
>@@ -105,9 +108,7 @@ class DesktopCaptureOptions {
>   }
>   // Allowing directx based capturer or not, this capturer works on windows 7
>   // with platform update / windows 8 or upper.
>-  bool allow_directx_capturer() const {
>-    return allow_directx_capturer_;
>-  }
>+  bool allow_directx_capturer() const { return allow_directx_capturer_; }
>   void set_allow_directx_capturer(bool enabled) {
>     allow_directx_capturer_ = enabled;
>   }
>@@ -122,6 +123,7 @@ class DesktopCaptureOptions {
>   rtc::scoped_refptr<DesktopConfigurationMonitor> configuration_monitor_;
>   rtc::scoped_refptr<FullScreenChromeWindowDetector>
>       full_screen_window_detector_;
>+  bool allow_iosurface_ = false;
> #endif
> 
> #if defined(WEBRTC_WIN)
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_capture_types.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_capture_types.h
>index 5d6ab8a48c7..5031cbf3acb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_capture_types.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_capture_types.h
>@@ -13,8 +13,6 @@
> 
> #include <stdint.h>
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>-
> namespace webrtc {
> 
> // Type used to identify windows on the desktop. Values are platform-specific:
>@@ -41,17 +39,14 @@ const ScreenId kInvalidScreenId = -2;
> // An integer to attach to each DesktopFrame to differentiate the generator of
> // the frame.
> namespace DesktopCapturerId {
>-  constexpr uint32_t CreateFourCC(char a, char b, char c, char d) {
>-    return ((static_cast<uint32_t>(a)) |
>-            (static_cast<uint32_t>(b) << 8) |
>-            (static_cast<uint32_t>(c) << 16) |
>-            (static_cast<uint32_t>(d) << 24));
>-  }
>-
>-  constexpr uint32_t kUnknown = 0;
>-  constexpr uint32_t kScreenCapturerWinGdi = CreateFourCC('G', 'D', 'I', ' ');
>-  constexpr uint32_t kScreenCapturerWinDirectx =
>-      CreateFourCC('D', 'X', 'G', 'I');
>+constexpr uint32_t CreateFourCC(char a, char b, char c, char d) {
>+  return ((static_cast<uint32_t>(a)) | (static_cast<uint32_t>(b) << 8) |
>+          (static_cast<uint32_t>(c) << 16) | (static_cast<uint32_t>(d) << 24));
>+}
>+
>+constexpr uint32_t kUnknown = 0;
>+constexpr uint32_t kScreenCapturerWinGdi = CreateFourCC('G', 'D', 'I', ' ');
>+constexpr uint32_t kScreenCapturerWinDirectx = CreateFourCC('D', 'X', 'G', 'I');
> }  // namespace DesktopCapturerId
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_capturer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_capturer.h
>index 335427af23a..47196cfe491 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_capturer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_capturer.h
>@@ -19,8 +19,8 @@
> #include <type_traits>
> #include <vector>
> 
>-#include "modules/desktop_capture/desktop_frame.h"
> #include "modules/desktop_capture/desktop_capture_types.h"
>+#include "modules/desktop_capture/desktop_frame.h"
> #include "modules/desktop_capture/shared_memory.h"
> 
> namespace webrtc {
>@@ -152,4 +152,3 @@ class DesktopCapturer {
> }  // namespace webrtc
> 
> #endif  // MODULES_DESKTOP_CAPTURE_DESKTOP_CAPTURER_H_
>-
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_capturer_differ_wrapper_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_capturer_differ_wrapper_unittest.cc
>index 0a080880957..eb2b8b263ca 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_capturer_differ_wrapper_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_capturer_differ_wrapper_unittest.cc
>@@ -24,7 +24,6 @@
> #include "rtc_base/timeutils.h"
> #include "system_wrappers/include/cpu_features_wrapper.h"
> #include "test/gtest.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -122,8 +121,8 @@ void ExecuteDifferWrapperCase(BlackWhiteDesktopFramePainter* frame_painter,
>               OnCaptureResultPtr(DesktopCapturer::Result::SUCCESS, testing::_))
>       .Times(1)
>       .WillOnce(testing::Invoke([&updated_region, check_result, exactly_match](
>-          DesktopCapturer::Result result,
>-          std::unique_ptr<DesktopFrame>* frame) {
>+                                    DesktopCapturer::Result result,
>+                                    std::unique_ptr<DesktopFrame>* frame) {
>         ASSERT_EQ(result, DesktopCapturer::Result::SUCCESS);
>         if (check_result) {
>           if (exactly_match) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_frame.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_frame.cc
>index 5a62d4d9cc9..925be43ec6d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_frame.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_frame.cc
>@@ -14,9 +14,9 @@
> 
> #include <string.h>
> 
>+#include "absl/memory/memory.h"
> #include "modules/desktop_capture/desktop_geometry.h"
> #include "rtc_base/checks.h"
>-#include "rtc_base/ptr_util.h"
> 
> namespace webrtc {
> 
>@@ -36,7 +36,8 @@ DesktopFrame::DesktopFrame(DesktopSize size,
> 
> DesktopFrame::~DesktopFrame() = default;
> 
>-void DesktopFrame::CopyPixelsFrom(const uint8_t* src_buffer, int src_stride,
>+void DesktopFrame::CopyPixelsFrom(const uint8_t* src_buffer,
>+                                  int src_stride,
>                                   const DesktopRect& dest_rect) {
>   RTC_CHECK(DesktopRect::MakeSize(size()).ContainsRect(dest_rect));
> 
>@@ -51,15 +52,27 @@ void DesktopFrame::CopyPixelsFrom(const uint8_t* src_buffer, int src_stride,
> void DesktopFrame::CopyPixelsFrom(const DesktopFrame& src_frame,
>                                   const DesktopVector& src_pos,
>                                   const DesktopRect& dest_rect) {
>-  RTC_CHECK(DesktopRect::MakeSize(src_frame.size()).ContainsRect(
>-      DesktopRect::MakeOriginSize(src_pos, dest_rect.size())));
>+  RTC_CHECK(DesktopRect::MakeSize(src_frame.size())
>+                .ContainsRect(
>+                    DesktopRect::MakeOriginSize(src_pos, dest_rect.size())));
> 
>-  CopyPixelsFrom(src_frame.GetFrameDataAtPos(src_pos),
>-                 src_frame.stride(), dest_rect);
>+  CopyPixelsFrom(src_frame.GetFrameDataAtPos(src_pos), src_frame.stride(),
>+                 dest_rect);
> }
> 
> DesktopRect DesktopFrame::rect() const {
>-  return DesktopRect::MakeOriginSize(top_left(), size());
>+  const float scale = scale_factor();
>+  // Only scale the size.
>+  return DesktopRect::MakeXYWH(top_left().x(), top_left().y(),
>+                               size().width() / scale, size().height() / scale);
>+}
>+
>+float DesktopFrame::scale_factor() const {
>+  float scale = 1.0f;
>+  if (!dpi().is_zero() && dpi().x() == dpi().y())
>+    scale = dpi().x() / kStandardDPI;
>+
>+  return scale;
> }
> 
> uint8_t* DesktopFrame::GetFrameDataAtPos(const DesktopVector& pos) const {
>@@ -83,7 +96,8 @@ void DesktopFrame::MoveFrameInfoFrom(DesktopFrame* other) {
> }
> 
> BasicDesktopFrame::BasicDesktopFrame(DesktopSize size)
>-    : DesktopFrame(size, kBytesPerPixel * size.width(),
>+    : DesktopFrame(size,
>+                   kBytesPerPixel * size.width(),
>                    new uint8_t[kBytesPerPixel * size.width() * size.height()],
>                    nullptr) {}
> 
>@@ -115,7 +129,7 @@ std::unique_ptr<DesktopFrame> SharedMemoryDesktopFrame::Create(
>   if (!shared_memory)
>     return nullptr;
> 
>-  return rtc::MakeUnique<SharedMemoryDesktopFrame>(
>+  return absl::make_unique<SharedMemoryDesktopFrame>(
>       size, size.width() * kBytesPerPixel, std::move(shared_memory));
> }
> 
>@@ -131,9 +145,7 @@ SharedMemoryDesktopFrame::SharedMemoryDesktopFrame(
>     DesktopSize size,
>     int stride,
>     std::unique_ptr<SharedMemory> shared_memory)
>-    : SharedMemoryDesktopFrame(size,
>-                               stride,
>-                               shared_memory.release()) {}
>+    : SharedMemoryDesktopFrame(size, stride, shared_memory.release()) {}
> 
> SharedMemoryDesktopFrame::~SharedMemoryDesktopFrame() {
>   delete shared_memory_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_frame.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_frame.h
>index 7bd95965803..6f7edfcc4f7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_frame.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_frame.h
>@@ -18,10 +18,11 @@
> #include "modules/desktop_capture/desktop_region.h"
> #include "modules/desktop_capture/shared_memory.h"
> #include "rtc_base/constructormagic.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>+const int kStandardDPI = 96;
>+
> // DesktopFrame represents a video frame captured from the screen.
> class DesktopFrame {
>  public:
>@@ -30,15 +31,21 @@ class DesktopFrame {
> 
>   virtual ~DesktopFrame();
> 
>-  // Returns the rectangle in full desktop coordinates to indicate the area
>-  // covered by the DesktopFrame.
>+  // Returns the rectangle in full desktop coordinates to indicate it covers
>+  // the area of top_left() to top_letf() + size() / scale_factor().
>   DesktopRect rect() const;
> 
>-  // Size of the frame.
>+  // Returns the scale factor from DIPs to physical pixels of the frame.
>+  // Assumes same scale in both X and Y directions at present.
>+  float scale_factor() const;
>+
>+  // Size of the frame. In physical coordinates, mapping directly from the
>+  // underlying buffer.
>   const DesktopSize& size() const { return size_; }
> 
>   // The top-left of the frame in full desktop coordinates. E.g. the top left
>-  // monitor should start from (0, 0).
>+  // monitor should start from (0, 0). The desktop coordinates may be scaled by
>+  // OS, but this is always consistent with the MouseCursorMonitor.
>   const DesktopVector& top_left() const { return top_left_; }
>   void set_top_left(const DesktopVector& top_left) { top_left_ = top_left; }
> 
>@@ -82,9 +89,7 @@ class DesktopFrame {
>   // Not all DesktopCapturer implementations set this field; it's set to
>   // kUnknown by default.
>   uint32_t capturer_id() const { return capturer_id_; }
>-  void set_capturer_id(uint32_t capturer_id) {
>-    capturer_id_ = capturer_id;
>-  }
>+  void set_capturer_id(uint32_t capturer_id) { capturer_id_ = capturer_id; }
> 
>   // Copies various information from |other|. Anything initialized in
>   // constructor are not copied.
>@@ -185,4 +190,3 @@ class SharedMemoryDesktopFrame : public DesktopFrame {
> }  // namespace webrtc
> 
> #endif  // MODULES_DESKTOP_CAPTURE_DESKTOP_FRAME_H_
>-
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_frame_rotation.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_frame_rotation.cc
>index 7cf0348e4c2..d7ea48cf3dc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_frame_rotation.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_frame_rotation.cc
>@@ -12,8 +12,8 @@
> 
> #include <string.h>
> 
>-#include "third_party/libyuv/include/libyuv/rotate_argb.h"
> #include "rtc_base/checks.h"
>+#include "third_party/libyuv/include/libyuv/rotate_argb.h"
> 
> namespace webrtc {
> 
>@@ -109,10 +109,10 @@ void RotateDesktopFrame(const DesktopFrame& source,
>   }
> 
>   int result = libyuv::ARGBRotate(
>-       source.GetFrameDataAtPos(source_rect.top_left()), source.stride(),
>-       target->GetFrameDataAtPos(target_rect.top_left()), target->stride(),
>-       source_rect.width(), source_rect.height(),
>-       ToLibyuvRotationMode(rotation));
>+      source.GetFrameDataAtPos(source_rect.top_left()), source.stride(),
>+      target->GetFrameDataAtPos(target_rect.top_left()), target->stride(),
>+      source_rect.width(), source_rect.height(),
>+      ToLibyuvRotationMode(rotation));
>   RTC_DCHECK_EQ(result, 0);
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_frame_rotation_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_frame_rotation_unittest.cc
>index e31f47207a3..acd6db184e0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_frame_rotation_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_frame_rotation_unittest.cc
>@@ -39,8 +39,8 @@ ArrayDesktopFrame::~ArrayDesktopFrame() = default;
> TEST(DesktopFrameRotationTest, CopyRect3x4) {
>   // A DesktopFrame of 4-pixel width by 3-pixel height.
>   static uint32_t frame_pixels[] = {
>-      0, 1,  2,  3,  //
>-      4, 5,  6,  7,  //
>+      0, 1, 2,  3,   //
>+      4, 5, 6,  7,   //
>       8, 9, 10, 11,  //
>   };
>   ArrayDesktopFrame frame(DesktopSize(4, 3), frame_pixels);
>@@ -55,8 +55,8 @@ TEST(DesktopFrameRotationTest, CopyRect3x4) {
>   // After Rotating clock-wise 90 degree
>   {
>     static uint32_t expected_pixels[] = {
>-         8, 4, 0,  //
>-         9, 5, 1,  //
>+        8,  4, 0,  //
>+        9,  5, 1,  //
>         10, 6, 2,  //
>         11, 7, 3,  //
>     };
>@@ -72,8 +72,8 @@ TEST(DesktopFrameRotationTest, CopyRect3x4) {
>   {
>     static uint32_t expected_pixels[] = {
>         11, 10, 9, 8,  //
>-         7,  6, 5, 4,  //
>-         3,  2, 1, 0,  //
>+        7,  6,  5, 4,  //
>+        3,  2,  1, 0,  //
>     };
>     ArrayDesktopFrame expected(DesktopSize(4, 3), expected_pixels);
> 
>@@ -88,8 +88,8 @@ TEST(DesktopFrameRotationTest, CopyRect3x4) {
>     static uint32_t expected_pixels[] = {
>         3, 7, 11,  //
>         2, 6, 10,  //
>-        1, 5,  9,  //
>-        0, 4,  8,  //
>+        1, 5, 9,   //
>+        0, 4, 8,   //
>     };
>     ArrayDesktopFrame expected(DesktopSize(3, 4), expected_pixels);
> 
>@@ -103,8 +103,8 @@ TEST(DesktopFrameRotationTest, CopyRect3x4) {
> TEST(DesktopFrameRotationTest, CopyRect3x5) {
>   // A DesktopFrame of 5-pixel width by 3-pixel height.
>   static uint32_t frame_pixels[] = {
>-       0,  1,  2,  3,  4,  //
>-       5,  6,  7,  8,  9,  //
>+      0,  1,  2,  3,  4,   //
>+      5,  6,  7,  8,  9,   //
>       10, 11, 12, 13, 14,  //
>   };
>   ArrayDesktopFrame frame(DesktopSize(5, 3), frame_pixels);
>@@ -135,10 +135,10 @@ TEST(DesktopFrameRotationTest, CopyRect3x5) {
> 
>   // After Rotating clock-wise 180 degree
>   {
>-    static uint32_t expected_pixels[] {
>+    static uint32_t expected_pixels[]{
>         14, 13, 12, 11, 10,  //
>-         9,  8,  7,  6,  5,  //
>-         4,  3,  2,  1,  0,  //
>+        9,  8,  7,  6,  5,   //
>+        4,  3,  2,  1,  0,   //
>     };
>     ArrayDesktopFrame expected(DesktopSize(5, 3), expected_pixels);
> 
>@@ -169,8 +169,8 @@ TEST(DesktopFrameRotationTest, CopyRect3x5) {
> TEST(DesktopFrameRotationTest, PartialCopyRect3x5) {
>   // A DesktopFrame of 5-pixel width by 3-pixel height.
>   static uint32_t frame_pixels[] = {
>-       0,  1,  2,  3,  4,  //
>-       5,  6,  7,  8,  9,  //
>+      0,  1,  2,  3,  4,   //
>+      5,  6,  7,  8,  9,   //
>       10, 11, 12, 13, 14,  //
>   };
>   ArrayDesktopFrame frame(DesktopSize(5, 3), frame_pixels);
>@@ -225,11 +225,11 @@ TEST(DesktopFrameRotationTest, PartialCopyRect3x5) {
> 
>   {
>     static uint32_t expected_pixels[] = {
>-         0, 0, 0,  //
>+        0,  0, 0,  //
>         11, 6, 0,  //
>         12, 7, 0,  //
>         13, 8, 0,  //
>-         0, 0, 0,  //
>+        0,  0, 0,  //
>     };
>     ArrayDesktopFrame expected(DesktopSize(3, 5), expected_pixels);
> 
>@@ -259,8 +259,8 @@ TEST(DesktopFrameRotationTest, PartialCopyRect3x5) {
>   {
>     static uint32_t expected_pixels[] = {
>         0, 13, 12, 11, 0,  //
>-        0,  8,  7,  6, 0,  //
>-        0,  0,  0,  0, 0,  //
>+        0, 8,  7,  6,  0,  //
>+        0, 0,  0,  0,  0,  //
>     };
>     ArrayDesktopFrame expected(DesktopSize(5, 3), expected_pixels);
> 
>@@ -310,20 +310,20 @@ TEST(DesktopFrameRotationTest, PartialCopyRect3x5) {
> TEST(DesktopFrameRotationTest, WithOffset) {
>   // A DesktopFrame of 4-pixel width by 3-pixel height.
>   static uint32_t frame_pixels[] = {
>-      0, 1,  2,  3,  //
>-      4, 5,  6,  7,  //
>+      0, 1, 2,  3,   //
>+      4, 5, 6,  7,   //
>       8, 9, 10, 11,  //
>   };
>   ArrayDesktopFrame frame(DesktopSize(4, 3), frame_pixels);
> 
>   {
>     static uint32_t expected_pixels[] = {
>-      0, 0, 0,  0,  0, 0, 0, 0,  //
>-      0, 0, 1,  2,  3, 0, 0, 0,  //
>-      0, 4, 5,  6,  7, 0, 0, 0,  //
>-      0, 8, 9, 10, 11, 0, 0, 0,  //
>-      0, 0, 0,  0,  0, 0, 0, 0,  //
>-      0, 0, 0,  0,  0, 0, 0, 0,  //
>+        0, 0, 0, 0,  0,  0, 0, 0,  //
>+        0, 0, 1, 2,  3,  0, 0, 0,  //
>+        0, 4, 5, 6,  7,  0, 0, 0,  //
>+        0, 8, 9, 10, 11, 0, 0, 0,  //
>+        0, 0, 0, 0,  0,  0, 0, 0,  //
>+        0, 0, 0, 0,  0,  0, 0, 0,  //
>     };
>     ArrayDesktopFrame expected(DesktopSize(8, 6), expected_pixels);
> 
>@@ -339,12 +339,12 @@ TEST(DesktopFrameRotationTest, WithOffset) {
> 
>   {
>     static uint32_t expected_pixels[] = {
>-      0,  0,  0, 0, 0, 0, 0, 0,  //
>-      0, 11, 10, 9, 8, 0, 0, 0,  //
>-      0,  7,  6, 5, 4, 0, 0, 0,  //
>-      0,  3,  2, 1, 0, 0, 0, 0,  //
>-      0,  0,  0, 0, 0, 0, 0, 0,  //
>-      0,  0,  0, 0, 0, 0, 0, 0,  //
>+        0, 0,  0,  0, 0, 0, 0, 0,  //
>+        0, 11, 10, 9, 8, 0, 0, 0,  //
>+        0, 7,  6,  5, 4, 0, 0, 0,  //
>+        0, 3,  2,  1, 0, 0, 0, 0,  //
>+        0, 0,  0,  0, 0, 0, 0, 0,  //
>+        0, 0,  0,  0, 0, 0, 0, 0,  //
>     };
>     ArrayDesktopFrame expected(DesktopSize(8, 6), expected_pixels);
> 
>@@ -360,14 +360,14 @@ TEST(DesktopFrameRotationTest, WithOffset) {
> 
>   {
>     static uint32_t expected_pixels[] = {
>-      0,  0, 0, 0, 0, 0,  //
>-      0,  8, 4, 0, 0, 0,  //
>-      0,  9, 5, 1, 0, 0,  //
>-      0, 10, 6, 2, 0, 0,  //
>-      0, 11, 7, 3, 0, 0,  //
>-      0,  0, 0, 0, 0, 0,  //
>-      0,  0, 0, 0, 0, 0,  //
>-      0,  0, 0, 0, 0, 0,  //
>+        0, 0,  0, 0, 0, 0,  //
>+        0, 8,  4, 0, 0, 0,  //
>+        0, 9,  5, 1, 0, 0,  //
>+        0, 10, 6, 2, 0, 0,  //
>+        0, 11, 7, 3, 0, 0,  //
>+        0, 0,  0, 0, 0, 0,  //
>+        0, 0,  0, 0, 0, 0,  //
>+        0, 0,  0, 0, 0, 0,  //
>     };
>     ArrayDesktopFrame expected(DesktopSize(6, 8), expected_pixels);
> 
>@@ -383,14 +383,14 @@ TEST(DesktopFrameRotationTest, WithOffset) {
> 
>   {
>     static uint32_t expected_pixels[] = {
>-      0, 0, 0,  0, 0, 0,  //
>-      0, 3, 7, 11, 0, 0,  //
>-      0, 2, 6, 10, 0, 0,  //
>-      0, 1, 5,  9, 0, 0,  //
>-      0, 0, 4,  8, 0, 0,  //
>-      0, 0, 0,  0, 0, 0,  //
>-      0, 0, 0,  0, 0, 0,  //
>-      0, 0, 0,  0, 0, 0,  //
>+        0, 0, 0, 0,  0, 0,  //
>+        0, 3, 7, 11, 0, 0,  //
>+        0, 2, 6, 10, 0, 0,  //
>+        0, 1, 5, 9,  0, 0,  //
>+        0, 0, 4, 8,  0, 0,  //
>+        0, 0, 0, 0,  0, 0,  //
>+        0, 0, 0, 0,  0, 0,  //
>+        0, 0, 0, 0,  0, 0,  //
>     };
>     ArrayDesktopFrame expected(DesktopSize(6, 8), expected_pixels);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_frame_win.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_frame_win.cc
>index fbc1200dd78..58ebac91d5d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_frame_win.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_frame_win.cc
>@@ -53,8 +53,8 @@ std::unique_ptr<DesktopFrameWin> DesktopFrameWin::Create(
>     section_handle = shared_memory->handle();
>   }
>   void* data = nullptr;
>-  HBITMAP bitmap = CreateDIBSection(hdc, &bmi, DIB_RGB_COLORS, &data,
>-                                    section_handle, 0);
>+  HBITMAP bitmap =
>+      CreateDIBSection(hdc, &bmi, DIB_RGB_COLORS, &data, section_handle, 0);
>   if (!bitmap) {
>     RTC_LOG(LS_WARNING) << "Failed to allocate new window frame "
>                         << GetLastError();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_frame_win.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_frame_win.h
>index 9ae1dc4d95f..cd953f0123d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_frame_win.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_frame_win.h
>@@ -17,7 +17,6 @@
> 
> #include "modules/desktop_capture/desktop_frame.h"
> #include "rtc_base/constructormagic.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -48,4 +47,3 @@ class DesktopFrameWin : public DesktopFrame {
> }  // namespace webrtc
> 
> #endif  // MODULES_DESKTOP_CAPTURE_DESKTOP_FRAME_WIN_H_
>-
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_geometry.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_geometry.cc
>index 8c1b23a0066..8220e5f5938 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_geometry.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_geometry.cc
>@@ -15,8 +15,8 @@
> namespace webrtc {
> 
> bool DesktopRect::Contains(const DesktopVector& point) const {
>-  return point.x() >= left() && point.x() < right() &&
>-         point.y() >= top() && point.y() < bottom();
>+  return point.x() >= left() && point.x() < right() && point.y() >= top() &&
>+         point.y() < bottom();
> }
> 
> bool DesktopRect::ContainsRect(const DesktopRect& rect) const {
>@@ -76,4 +76,3 @@ void DesktopRect::Scale(double horizontal, double vertical) {
> }
> 
> }  // namespace webrtc
>-
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_geometry.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_geometry.h
>index c1ee1a539ef..09ad55bb4bf 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_geometry.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_geometry.h
>@@ -11,8 +11,9 @@
> #ifndef MODULES_DESKTOP_CAPTURE_DESKTOP_GEOMETRY_H_
> #define MODULES_DESKTOP_CAPTURE_DESKTOP_GEOMETRY_H_
> 
>+#include <stdint.h>
>+
> #include "rtc_base/constructormagic.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -51,9 +52,7 @@ class DesktopVector {
> class DesktopSize {
>  public:
>   DesktopSize() : width_(0), height_(0) {}
>-  DesktopSize(int32_t width, int32_t height)
>-      : width_(width), height_(height) {
>-  }
>+  DesktopSize(int32_t width, int32_t height) : width_(width), height_(height) {}
> 
>   int32_t width() const { return width_; }
>   int32_t height() const { return height_; }
>@@ -83,12 +82,16 @@ class DesktopRect {
>   static DesktopRect MakeWH(int32_t width, int32_t height) {
>     return DesktopRect(0, 0, width, height);
>   }
>-  static DesktopRect MakeXYWH(int32_t x, int32_t y,
>-                              int32_t width, int32_t height) {
>+  static DesktopRect MakeXYWH(int32_t x,
>+                              int32_t y,
>+                              int32_t width,
>+                              int32_t height) {
>     return DesktopRect(x, y, x + width, y + height);
>   }
>-  static DesktopRect MakeLTRB(int32_t left, int32_t top,
>-                              int32_t right, int32_t bottom) {
>+  static DesktopRect MakeLTRB(int32_t left,
>+                              int32_t top,
>+                              int32_t right,
>+                              int32_t bottom) {
>     return DesktopRect(left, top, right, bottom);
>   }
>   static DesktopRect MakeOriginSize(const DesktopVector& origin,
>@@ -115,7 +118,7 @@ class DesktopRect {
> 
>   bool equals(const DesktopRect& other) const {
>     return left_ == other.left_ && top_ == other.top_ &&
>-        right_ == other.right_ && bottom_ == other.bottom_;
>+           right_ == other.right_ && bottom_ == other.bottom_;
>   }
> 
>   // Returns true if |point| lies within the rectangle boundaries.
>@@ -151,8 +154,7 @@ class DesktopRect {
> 
>  private:
>   DesktopRect(int32_t left, int32_t top, int32_t right, int32_t bottom)
>-      : left_(left), top_(top), right_(right), bottom_(bottom) {
>-  }
>+      : left_(left), top_(top), right_(right), bottom_(bottom) {}
> 
>   int32_t left_;
>   int32_t top_;
>@@ -163,4 +165,3 @@ class DesktopRect {
> }  // namespace webrtc
> 
> #endif  // MODULES_DESKTOP_CAPTURE_DESKTOP_GEOMETRY_H_
>-
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_region.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_region.cc
>index fb310d43e5d..472f8155871 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_region.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_region.cc
>@@ -17,15 +17,13 @@
> namespace webrtc {
> 
> DesktopRegion::RowSpan::RowSpan(int32_t left, int32_t right)
>-    : left(left), right(right) {
>-}
>+    : left(left), right(right) {}
> 
> DesktopRegion::Row::Row(const Row&) = default;
> DesktopRegion::Row::Row(Row&&) = default;
> 
> DesktopRegion::Row::Row(int32_t top, int32_t bottom)
>-    : top(top), bottom(bottom) {
>-}
>+    : top(top), bottom(bottom) {}
> 
> DesktopRegion::Row::~Row() {}
> 
>@@ -63,8 +61,7 @@ bool DesktopRegion::Equals(const DesktopRegion& region) const {
>   Rows::const_iterator it1 = rows_.begin();
>   Rows::const_iterator it2 = region.rows_.begin();
>   while (it1 != rows_.end()) {
>-    if (it2 == region.rows_.end() ||
>-        it1->first != it2->first ||
>+    if (it2 == region.rows_.end() || it1->first != it2->first ||
>         it1->second->top != it2->second->top ||
>         it1->second->bottom != it2->second->bottom ||
>         it1->second->spans != it2->second->spans) {
>@@ -100,14 +97,13 @@ void DesktopRegion::AddRect(const DesktopRect& rect) {
>   // necessary.
>   Rows::iterator row = rows_.upper_bound(top);
>   while (top < rect.bottom()) {
>-    if (row == rows_.end() || top < row->second->top)  {
>+    if (row == rows_.end() || top < row->second->top) {
>       // If |top| is above the top of the current |row| then add a new row above
>       // the current one.
>       int32_t bottom = rect.bottom();
>       if (row != rows_.end() && row->second->top < bottom)
>         bottom = row->second->top;
>-      row = rows_.insert(
>-          row, Rows::value_type(bottom, new Row(top, bottom)));
>+      row = rows_.insert(row, Rows::value_type(bottom, new Row(top, bottom)));
>     } else if (top > row->second->top) {
>       // If the |top| falls in the middle of the |row| then split |row| into
>       // two, at |top|, and leave |row| referring to the lower of the two,
>@@ -421,9 +417,8 @@ void DesktopRegion::AddSpanToRow(Row* row, int left, int right) {
>   }
> 
>   // Find the first span that ends at or after |left|.
>-  RowSpanSet::iterator start =
>-      std::lower_bound(row->spans.begin(), row->spans.end(), left,
>-                       CompareSpanRight);
>+  RowSpanSet::iterator start = std::lower_bound(
>+      row->spans.begin(), row->spans.end(), left, CompareSpanRight);
>   assert(start < row->spans.end());
> 
>   // Find the first span that starts after |right|.
>@@ -462,9 +457,8 @@ void DesktopRegion::AddSpanToRow(Row* row, int left, int right) {
> bool DesktopRegion::IsSpanInRow(const Row& row, const RowSpan& span) {
>   // Find the first span that starts at or after |span.left| and then check if
>   // it's the same span.
>-  RowSpanSet::const_iterator it =
>-      std::lower_bound(row.spans.begin(), row.spans.end(), span.left,
>-                       CompareSpanLeft);
>+  RowSpanSet::const_iterator it = std::lower_bound(
>+      row.spans.begin(), row.spans.end(), span.left, CompareSpanLeft);
>   return it != row.spans.end() && *it == span;
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_region.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_region.h
>index 93f78a4c0c7..f4b9209e2dc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_region.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_region.h
>@@ -16,7 +16,6 @@
> 
> #include "modules/desktop_capture/desktop_geometry.h"
> #include "rtc_base/constructormagic.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -166,4 +165,3 @@ class DesktopRegion {
> }  // namespace webrtc
> 
> #endif  // MODULES_DESKTOP_CAPTURE_DESKTOP_REGION_H_
>-
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_region_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_region_unittest.cc
>index 0e94c9dee83..65fc42fd5d0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_region_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/desktop_region_unittest.cc
>@@ -23,7 +23,8 @@ int RadmonInt(int max) {
> }
> 
> void CompareRegion(const DesktopRegion& region,
>-                   const DesktopRect rects[], int rects_size) {
>+                   const DesktopRect rects[],
>+                   int rects_size) {
>   DesktopRegion::Iterator it(region);
>   for (int i = 0; i < rects_size; ++i) {
>     SCOPED_TRACE(i);
>@@ -31,8 +32,8 @@ void CompareRegion(const DesktopRegion& region,
>     EXPECT_TRUE(it.rect().equals(rects[i]))
>         << it.rect().left() << "-" << it.rect().right() << "."
>         << it.rect().top() << "-" << it.rect().bottom() << " "
>-        << rects[i].left() << "-" << rects[i].right() << "."
>-        << rects[i].top() << "-" << rects[i].bottom();
>+        << rects[i].left() << "-" << rects[i].right() << "." << rects[i].top()
>+        << "-" << rects[i].bottom();
>     it.Advance();
>   }
>   EXPECT_TRUE(it.IsAtEnd());
>@@ -68,22 +69,27 @@ TEST(DesktopRegionTest, NonOverlappingRects) {
>     int count;
>     DesktopRect rects[4];
>   } cases[] = {
>-    { 1, { DesktopRect::MakeXYWH(10, 10, 10, 10) } },
>-    { 2, { DesktopRect::MakeXYWH(10, 10, 10, 10),
>-           DesktopRect::MakeXYWH(30, 10, 10, 15) } },
>-    { 2, { DesktopRect::MakeXYWH(10, 10, 10, 10),
>-           DesktopRect::MakeXYWH(10, 30, 10, 5) } },
>-    { 3, { DesktopRect::MakeXYWH(10, 10, 10, 9),
>-           DesktopRect::MakeXYWH(30, 10, 15, 10),
>-           DesktopRect::MakeXYWH(10, 30, 8, 10) } },
>-    { 4, { DesktopRect::MakeXYWH(0, 0, 30, 10),
>-           DesktopRect::MakeXYWH(40, 0, 10, 30),
>-           DesktopRect::MakeXYWH(0, 20, 10, 30),
>-           DesktopRect::MakeXYWH(20, 40, 30, 10) } },
>-    { 4, { DesktopRect::MakeXYWH(0, 0, 10, 100),
>-           DesktopRect::MakeXYWH(20, 10, 30, 10),
>-           DesktopRect::MakeXYWH(20, 30, 30, 10),
>-           DesktopRect::MakeXYWH(20, 50, 30, 10) } },
>+      {1, {DesktopRect::MakeXYWH(10, 10, 10, 10)}},
>+      {2,
>+       {DesktopRect::MakeXYWH(10, 10, 10, 10),
>+        DesktopRect::MakeXYWH(30, 10, 10, 15)}},
>+      {2,
>+       {DesktopRect::MakeXYWH(10, 10, 10, 10),
>+        DesktopRect::MakeXYWH(10, 30, 10, 5)}},
>+      {3,
>+       {DesktopRect::MakeXYWH(10, 10, 10, 9),
>+        DesktopRect::MakeXYWH(30, 10, 15, 10),
>+        DesktopRect::MakeXYWH(10, 30, 8, 10)}},
>+      {4,
>+       {DesktopRect::MakeXYWH(0, 0, 30, 10),
>+        DesktopRect::MakeXYWH(40, 0, 10, 30),
>+        DesktopRect::MakeXYWH(0, 20, 10, 30),
>+        DesktopRect::MakeXYWH(20, 40, 30, 10)}},
>+      {4,
>+       {DesktopRect::MakeXYWH(0, 0, 10, 100),
>+        DesktopRect::MakeXYWH(20, 10, 30, 10),
>+        DesktopRect::MakeXYWH(20, 30, 30, 10),
>+        DesktopRect::MakeXYWH(20, 50, 30, 10)}},
>   };
> 
>   for (size_t i = 0; i < (sizeof(cases) / sizeof(Case)); ++i) {
>@@ -114,81 +120,97 @@ TEST(DesktopRegionTest, TwoRects) {
>     int expected_count;
>     DesktopRect expected_rects[3];
>   } cases[] = {
>-    // Touching rectangles that merge into one.
>-    { DesktopRect::MakeLTRB(100, 100, 200, 200),
>-      DesktopRect::MakeLTRB(0, 100, 100, 200),
>-      1, { DesktopRect::MakeLTRB(0, 100, 200, 200) } },
>-    { DesktopRect::MakeLTRB(100, 100, 200, 200),
>-      DesktopRect::MakeLTRB(100, 0, 200, 100),
>-      1, { DesktopRect::MakeLTRB(100, 0, 200, 200) } },
>-
>-    // Rectangles touching on the vertical edge.
>-    { DesktopRect::MakeLTRB(100, 100, 200, 200),
>-      DesktopRect::MakeLTRB(0, 150, 100, 250),
>-      3, { DesktopRect::MakeLTRB(100, 100, 200, 150),
>-           DesktopRect::MakeLTRB(0, 150, 200, 200),
>-           DesktopRect::MakeLTRB(0, 200, 100, 250) } },
>-    { DesktopRect::MakeLTRB(100, 100, 200, 200),
>-      DesktopRect::MakeLTRB(0, 50, 100, 150),
>-      3, { DesktopRect::MakeLTRB(0, 50, 100, 100),
>-           DesktopRect::MakeLTRB(0, 100, 200, 150),
>-           DesktopRect::MakeLTRB(100, 150, 200, 200) } },
>-    { DesktopRect::MakeLTRB(100, 100, 200, 200),
>-      DesktopRect::MakeLTRB(0, 120, 100, 180),
>-      3, { DesktopRect::MakeLTRB(100, 100, 200, 120),
>-           DesktopRect::MakeLTRB(0, 120, 200, 180),
>-           DesktopRect::MakeLTRB(100, 180, 200, 200) } },
>-
>-    // Rectangles touching on the horizontal edge.
>-    { DesktopRect::MakeLTRB(100, 100, 200, 200),
>-      DesktopRect::MakeLTRB(150, 0, 250, 100),
>-      2, { DesktopRect::MakeLTRB(150, 0, 250, 100),
>-           DesktopRect::MakeLTRB(100, 100, 200, 200) } },
>-    { DesktopRect::MakeLTRB(100, 100, 200, 200),
>-      DesktopRect::MakeLTRB(50, 0, 150, 100),
>-      2, { DesktopRect::MakeLTRB(50, 0, 150, 100),
>-           DesktopRect::MakeLTRB(100, 100, 200, 200) } },
>-    { DesktopRect::MakeLTRB(100, 100, 200, 200),
>-      DesktopRect::MakeLTRB(120, 0, 180, 100),
>-      2, { DesktopRect::MakeLTRB(120, 0, 180, 100),
>-           DesktopRect::MakeLTRB(100, 100, 200, 200) } },
>-
>-    // Overlapping rectangles.
>-    { DesktopRect::MakeLTRB(100, 100, 200, 200),
>-      DesktopRect::MakeLTRB(50, 50, 150, 150),
>-      3, { DesktopRect::MakeLTRB(50, 50, 150, 100),
>-           DesktopRect::MakeLTRB(50, 100, 200, 150),
>-           DesktopRect::MakeLTRB(100, 150, 200, 200) } },
>-    { DesktopRect::MakeLTRB(100, 100, 200, 200),
>-      DesktopRect::MakeLTRB(150, 50, 250, 150),
>-      3, { DesktopRect::MakeLTRB(150, 50, 250, 100),
>-           DesktopRect::MakeLTRB(100, 100, 250, 150),
>-           DesktopRect::MakeLTRB(100, 150, 200, 200) } },
>-    { DesktopRect::MakeLTRB(100, 100, 200, 200),
>-      DesktopRect::MakeLTRB(0, 120, 150, 180),
>-      3, { DesktopRect::MakeLTRB(100, 100, 200, 120),
>-           DesktopRect::MakeLTRB(0, 120, 200, 180),
>-           DesktopRect::MakeLTRB(100, 180, 200, 200) } },
>-    { DesktopRect::MakeLTRB(100, 100, 200, 200),
>-      DesktopRect::MakeLTRB(120, 0, 180, 150),
>-      2, { DesktopRect::MakeLTRB(120, 0, 180, 100),
>-           DesktopRect::MakeLTRB(100, 100, 200, 200) } },
>-    { DesktopRect::MakeLTRB(100, 0, 200, 300),
>-      DesktopRect::MakeLTRB(0, 100, 300, 200),
>-      3, { DesktopRect::MakeLTRB(100, 0, 200, 100),
>-           DesktopRect::MakeLTRB(0, 100, 300, 200),
>-           DesktopRect::MakeLTRB(100, 200, 200, 300)} },
>-
>-    // One rectangle enclosing another.
>-    { DesktopRect::MakeLTRB(100, 100, 200, 200),
>-      DesktopRect::MakeLTRB(150, 150, 180, 180),
>-      1, { DesktopRect::MakeLTRB(100, 100, 200, 200) } },
>-    { DesktopRect::MakeLTRB(100, 100, 200, 200),
>-      DesktopRect::MakeLTRB(100, 100, 180, 180),
>-      1, { DesktopRect::MakeLTRB(100, 100, 200, 200) } },
>-    { DesktopRect::MakeLTRB(100, 100, 200, 200),
>-      DesktopRect::MakeLTRB(150, 150, 200, 200),
>-      1, { DesktopRect::MakeLTRB(100, 100, 200, 200) } },
>+      // Touching rectangles that merge into one.
>+      {DesktopRect::MakeLTRB(100, 100, 200, 200),
>+       DesktopRect::MakeLTRB(0, 100, 100, 200),
>+       1,
>+       {DesktopRect::MakeLTRB(0, 100, 200, 200)}},
>+      {DesktopRect::MakeLTRB(100, 100, 200, 200),
>+       DesktopRect::MakeLTRB(100, 0, 200, 100),
>+       1,
>+       {DesktopRect::MakeLTRB(100, 0, 200, 200)}},
>+
>+      // Rectangles touching on the vertical edge.
>+      {DesktopRect::MakeLTRB(100, 100, 200, 200),
>+       DesktopRect::MakeLTRB(0, 150, 100, 250),
>+       3,
>+       {DesktopRect::MakeLTRB(100, 100, 200, 150),
>+        DesktopRect::MakeLTRB(0, 150, 200, 200),
>+        DesktopRect::MakeLTRB(0, 200, 100, 250)}},
>+      {DesktopRect::MakeLTRB(100, 100, 200, 200),
>+       DesktopRect::MakeLTRB(0, 50, 100, 150),
>+       3,
>+       {DesktopRect::MakeLTRB(0, 50, 100, 100),
>+        DesktopRect::MakeLTRB(0, 100, 200, 150),
>+        DesktopRect::MakeLTRB(100, 150, 200, 200)}},
>+      {DesktopRect::MakeLTRB(100, 100, 200, 200),
>+       DesktopRect::MakeLTRB(0, 120, 100, 180),
>+       3,
>+       {DesktopRect::MakeLTRB(100, 100, 200, 120),
>+        DesktopRect::MakeLTRB(0, 120, 200, 180),
>+        DesktopRect::MakeLTRB(100, 180, 200, 200)}},
>+
>+      // Rectangles touching on the horizontal edge.
>+      {DesktopRect::MakeLTRB(100, 100, 200, 200),
>+       DesktopRect::MakeLTRB(150, 0, 250, 100),
>+       2,
>+       {DesktopRect::MakeLTRB(150, 0, 250, 100),
>+        DesktopRect::MakeLTRB(100, 100, 200, 200)}},
>+      {DesktopRect::MakeLTRB(100, 100, 200, 200),
>+       DesktopRect::MakeLTRB(50, 0, 150, 100),
>+       2,
>+       {DesktopRect::MakeLTRB(50, 0, 150, 100),
>+        DesktopRect::MakeLTRB(100, 100, 200, 200)}},
>+      {DesktopRect::MakeLTRB(100, 100, 200, 200),
>+       DesktopRect::MakeLTRB(120, 0, 180, 100),
>+       2,
>+       {DesktopRect::MakeLTRB(120, 0, 180, 100),
>+        DesktopRect::MakeLTRB(100, 100, 200, 200)}},
>+
>+      // Overlapping rectangles.
>+      {DesktopRect::MakeLTRB(100, 100, 200, 200),
>+       DesktopRect::MakeLTRB(50, 50, 150, 150),
>+       3,
>+       {DesktopRect::MakeLTRB(50, 50, 150, 100),
>+        DesktopRect::MakeLTRB(50, 100, 200, 150),
>+        DesktopRect::MakeLTRB(100, 150, 200, 200)}},
>+      {DesktopRect::MakeLTRB(100, 100, 200, 200),
>+       DesktopRect::MakeLTRB(150, 50, 250, 150),
>+       3,
>+       {DesktopRect::MakeLTRB(150, 50, 250, 100),
>+        DesktopRect::MakeLTRB(100, 100, 250, 150),
>+        DesktopRect::MakeLTRB(100, 150, 200, 200)}},
>+      {DesktopRect::MakeLTRB(100, 100, 200, 200),
>+       DesktopRect::MakeLTRB(0, 120, 150, 180),
>+       3,
>+       {DesktopRect::MakeLTRB(100, 100, 200, 120),
>+        DesktopRect::MakeLTRB(0, 120, 200, 180),
>+        DesktopRect::MakeLTRB(100, 180, 200, 200)}},
>+      {DesktopRect::MakeLTRB(100, 100, 200, 200),
>+       DesktopRect::MakeLTRB(120, 0, 180, 150),
>+       2,
>+       {DesktopRect::MakeLTRB(120, 0, 180, 100),
>+        DesktopRect::MakeLTRB(100, 100, 200, 200)}},
>+      {DesktopRect::MakeLTRB(100, 0, 200, 300),
>+       DesktopRect::MakeLTRB(0, 100, 300, 200),
>+       3,
>+       {DesktopRect::MakeLTRB(100, 0, 200, 100),
>+        DesktopRect::MakeLTRB(0, 100, 300, 200),
>+        DesktopRect::MakeLTRB(100, 200, 200, 300)}},
>+
>+      // One rectangle enclosing another.
>+      {DesktopRect::MakeLTRB(100, 100, 200, 200),
>+       DesktopRect::MakeLTRB(150, 150, 180, 180),
>+       1,
>+       {DesktopRect::MakeLTRB(100, 100, 200, 200)}},
>+      {DesktopRect::MakeLTRB(100, 100, 200, 200),
>+       DesktopRect::MakeLTRB(100, 100, 180, 180),
>+       1,
>+       {DesktopRect::MakeLTRB(100, 100, 200, 200)}},
>+      {DesktopRect::MakeLTRB(100, 100, 200, 200),
>+       DesktopRect::MakeLTRB(150, 150, 200, 200),
>+       1,
>+       {DesktopRect::MakeLTRB(100, 100, 200, 200)}},
>   };
> 
>   for (size_t i = 0; i < (sizeof(cases) / sizeof(Case)); ++i) {
>@@ -219,7 +241,9 @@ TEST(DesktopRegionTest, SameRow) {
>   const int kLastRectSizes[] = {3, 27};
> 
>   DesktopRegion base_region;
>-  bool base_map[kMapWidth] = { false, };
>+  bool base_map[kMapWidth] = {
>+      false,
>+  };
> 
>   base_region.AddRect(DesktopRect::MakeXYWH(5, 0, 5, 1));
>   std::fill_n(base_map + 5, 5, true);
>@@ -239,13 +263,15 @@ TEST(DesktopRegionTest, SameRow) {
>       SCOPED_TRACE(x);
> 
>       DesktopRegion r = base_region;
>-      r.AddRect(DesktopRect::MakeXYWH(x, 0,  last_rect_size, 1));
>+      r.AddRect(DesktopRect::MakeXYWH(x, 0, last_rect_size, 1));
> 
>       bool expected_map[kMapWidth];
>       std::copy(base_map, base_map + kMapWidth, expected_map);
>       std::fill_n(expected_map + x, last_rect_size, true);
> 
>-      bool map[kMapWidth] = { false, };
>+      bool map[kMapWidth] = {
>+          false,
>+      };
> 
>       int pos = -1;
>       for (DesktopRegion::Iterator it(r); !it.IsAtEnd(); it.Advance()) {
>@@ -266,32 +292,42 @@ TEST(DesktopRegionTest, ComplexRegions) {
>     int expected_count;
>     DesktopRect expected_rects[6];
>   } cases[] = {
>-    { 3, { DesktopRect::MakeLTRB(100, 100, 200, 200),
>+      {3,
>+       {
>+           DesktopRect::MakeLTRB(100, 100, 200, 200),
>            DesktopRect::MakeLTRB(0, 100, 100, 200),
>-           DesktopRect::MakeLTRB(310, 110, 320, 120), },
>-      2, { DesktopRect::MakeLTRB(0, 100, 200, 200),
>-           DesktopRect::MakeLTRB(310, 110, 320, 120) } },
>-    { 3, { DesktopRect::MakeLTRB(100, 100, 200, 200),
>-           DesktopRect::MakeLTRB(50, 50, 150, 150),
>-           DesktopRect::MakeLTRB(300, 125, 350, 175) },
>-      4, { DesktopRect::MakeLTRB(50, 50, 150, 100),
>-           DesktopRect::MakeLTRB(50, 100, 200, 150),
>-           DesktopRect::MakeLTRB(300, 125, 350, 175),
>-           DesktopRect::MakeLTRB(100, 150, 200, 200) } },
>-    { 4, { DesktopRect::MakeLTRB(0, 0, 30, 30),
>-           DesktopRect::MakeLTRB(10, 10, 40, 40),
>-           DesktopRect::MakeLTRB(20, 20, 50, 50),
>-           DesktopRect::MakeLTRB(50, 0, 65, 15) },
>-      6, { DesktopRect::MakeLTRB(0, 0, 30, 10),
>-           DesktopRect::MakeLTRB(50, 0, 65, 15),
>-           DesktopRect::MakeLTRB(0, 10, 40, 20),
>-           DesktopRect::MakeLTRB(0, 20, 50, 30),
>-           DesktopRect::MakeLTRB(10, 30, 50, 40),
>-           DesktopRect::MakeLTRB(20, 40, 50, 50) } },
>-    { 3, { DesktopRect::MakeLTRB(10, 10, 40, 20),
>-           DesktopRect::MakeLTRB(10, 30, 40, 40),
>-           DesktopRect::MakeLTRB(10, 20, 40, 30) },
>-      1, { DesktopRect::MakeLTRB(10, 10, 40, 40) } },
>+           DesktopRect::MakeLTRB(310, 110, 320, 120),
>+       },
>+       2,
>+       {DesktopRect::MakeLTRB(0, 100, 200, 200),
>+        DesktopRect::MakeLTRB(310, 110, 320, 120)}},
>+      {3,
>+       {DesktopRect::MakeLTRB(100, 100, 200, 200),
>+        DesktopRect::MakeLTRB(50, 50, 150, 150),
>+        DesktopRect::MakeLTRB(300, 125, 350, 175)},
>+       4,
>+       {DesktopRect::MakeLTRB(50, 50, 150, 100),
>+        DesktopRect::MakeLTRB(50, 100, 200, 150),
>+        DesktopRect::MakeLTRB(300, 125, 350, 175),
>+        DesktopRect::MakeLTRB(100, 150, 200, 200)}},
>+      {4,
>+       {DesktopRect::MakeLTRB(0, 0, 30, 30),
>+        DesktopRect::MakeLTRB(10, 10, 40, 40),
>+        DesktopRect::MakeLTRB(20, 20, 50, 50),
>+        DesktopRect::MakeLTRB(50, 0, 65, 15)},
>+       6,
>+       {DesktopRect::MakeLTRB(0, 0, 30, 10),
>+        DesktopRect::MakeLTRB(50, 0, 65, 15),
>+        DesktopRect::MakeLTRB(0, 10, 40, 20),
>+        DesktopRect::MakeLTRB(0, 20, 50, 30),
>+        DesktopRect::MakeLTRB(10, 30, 50, 40),
>+        DesktopRect::MakeLTRB(20, 40, 50, 50)}},
>+      {3,
>+       {DesktopRect::MakeLTRB(10, 10, 40, 20),
>+        DesktopRect::MakeLTRB(10, 30, 40, 40),
>+        DesktopRect::MakeLTRB(10, 20, 40, 30)},
>+       1,
>+       {DesktopRect::MakeLTRB(10, 10, 40, 40)}},
>   };
> 
>   for (size_t i = 0; i < (sizeof(cases) / sizeof(Case)); ++i) {
>@@ -316,44 +352,62 @@ TEST(DesktopRegionTest, Equals) {
>     DesktopRect rects[4];
>     int id;
>   } regions[] = {
>-    // Same region with one of the rectangles 1 pixel wider/taller.
>-    { 2, { DesktopRect::MakeLTRB(0, 100, 200, 200),
>-           DesktopRect::MakeLTRB(310, 110, 320, 120) }, 0 },
>-    { 2, { DesktopRect::MakeLTRB(0, 100, 201, 200),
>-           DesktopRect::MakeLTRB(310, 110, 320, 120) }, 1 },
>-    { 2, { DesktopRect::MakeLTRB(0, 100, 200, 201),
>-           DesktopRect::MakeLTRB(310, 110, 320, 120) }, 2 },
>-
>-    // Same region with one of the rectangles shifted horizontally and
>-    // vertically.
>-    { 4, { DesktopRect::MakeLTRB(0, 0, 30, 30),
>-           DesktopRect::MakeLTRB(10, 10, 40, 40),
>-           DesktopRect::MakeLTRB(20, 20, 50, 50),
>-           DesktopRect::MakeLTRB(50, 0, 65, 15) }, 3 },
>-    { 4, { DesktopRect::MakeLTRB(0, 0, 30, 30),
>-           DesktopRect::MakeLTRB(10, 10, 40, 40),
>-           DesktopRect::MakeLTRB(20, 20, 50, 50),
>-           DesktopRect::MakeLTRB(50, 1, 65, 16) }, 4 },
>-    { 4, { DesktopRect::MakeLTRB(0, 0, 30, 30),
>-           DesktopRect::MakeLTRB(10, 10, 40, 40),
>-           DesktopRect::MakeLTRB(20, 20, 50, 50),
>-           DesktopRect::MakeLTRB(51, 0, 66, 15) }, 5 },
>-
>-    // Same region defined by a different set of rectangles - one of the
>-    // rectangle is split horizontally into two.
>-    { 3, { DesktopRect::MakeLTRB(100, 100, 200, 200),
>-           DesktopRect::MakeLTRB(50, 50, 150, 150),
>-           DesktopRect::MakeLTRB(300, 125, 350, 175) }, 6 },
>-    { 4, { DesktopRect::MakeLTRB(100, 100, 200, 200),
>-           DesktopRect::MakeLTRB(50, 50, 100, 150),
>-           DesktopRect::MakeLTRB(100, 50, 150, 150),
>-           DesktopRect::MakeLTRB(300, 125, 350, 175) }, 6 },
>-
>-    // Rectangle region defined by a set of rectangles that merge into one.
>-    { 3, { DesktopRect::MakeLTRB(10, 10, 40, 20),
>-           DesktopRect::MakeLTRB(10, 30, 40, 40),
>-           DesktopRect::MakeLTRB(10, 20, 40, 30) }, 7 },
>-    { 1, { DesktopRect::MakeLTRB(10, 10, 40, 40) }, 7 },
>+      // Same region with one of the rectangles 1 pixel wider/taller.
>+      {2,
>+       {DesktopRect::MakeLTRB(0, 100, 200, 200),
>+        DesktopRect::MakeLTRB(310, 110, 320, 120)},
>+       0},
>+      {2,
>+       {DesktopRect::MakeLTRB(0, 100, 201, 200),
>+        DesktopRect::MakeLTRB(310, 110, 320, 120)},
>+       1},
>+      {2,
>+       {DesktopRect::MakeLTRB(0, 100, 200, 201),
>+        DesktopRect::MakeLTRB(310, 110, 320, 120)},
>+       2},
>+
>+      // Same region with one of the rectangles shifted horizontally and
>+      // vertically.
>+      {4,
>+       {DesktopRect::MakeLTRB(0, 0, 30, 30),
>+        DesktopRect::MakeLTRB(10, 10, 40, 40),
>+        DesktopRect::MakeLTRB(20, 20, 50, 50),
>+        DesktopRect::MakeLTRB(50, 0, 65, 15)},
>+       3},
>+      {4,
>+       {DesktopRect::MakeLTRB(0, 0, 30, 30),
>+        DesktopRect::MakeLTRB(10, 10, 40, 40),
>+        DesktopRect::MakeLTRB(20, 20, 50, 50),
>+        DesktopRect::MakeLTRB(50, 1, 65, 16)},
>+       4},
>+      {4,
>+       {DesktopRect::MakeLTRB(0, 0, 30, 30),
>+        DesktopRect::MakeLTRB(10, 10, 40, 40),
>+        DesktopRect::MakeLTRB(20, 20, 50, 50),
>+        DesktopRect::MakeLTRB(51, 0, 66, 15)},
>+       5},
>+
>+      // Same region defined by a different set of rectangles - one of the
>+      // rectangle is split horizontally into two.
>+      {3,
>+       {DesktopRect::MakeLTRB(100, 100, 200, 200),
>+        DesktopRect::MakeLTRB(50, 50, 150, 150),
>+        DesktopRect::MakeLTRB(300, 125, 350, 175)},
>+       6},
>+      {4,
>+       {DesktopRect::MakeLTRB(100, 100, 200, 200),
>+        DesktopRect::MakeLTRB(50, 50, 100, 150),
>+        DesktopRect::MakeLTRB(100, 50, 150, 150),
>+        DesktopRect::MakeLTRB(300, 125, 350, 175)},
>+       6},
>+
>+      // Rectangle region defined by a set of rectangles that merge into one.
>+      {3,
>+       {DesktopRect::MakeLTRB(10, 10, 40, 20),
>+        DesktopRect::MakeLTRB(10, 30, 40, 40),
>+        DesktopRect::MakeLTRB(10, 20, 40, 30)},
>+       7},
>+      {1, {DesktopRect::MakeLTRB(10, 10, 40, 40)}, 7},
>   };
>   int kTotalRegions = sizeof(regions) / sizeof(Region);
> 
>@@ -379,15 +433,18 @@ TEST(DesktopRegionTest, Translate) {
>     int expected_count;
>     DesktopRect expected_rects[5];
>   } cases[] = {
>-    { 3, { DesktopRect::MakeLTRB(0, 0, 30, 30),
>-           DesktopRect::MakeLTRB(10, 10, 40, 40),
>-           DesktopRect::MakeLTRB(20, 20, 50, 50) },
>-      3, 5,
>-      5, { DesktopRect::MakeLTRB(3, 5, 33, 15),
>-           DesktopRect::MakeLTRB(3, 15, 43, 25),
>-           DesktopRect::MakeLTRB(3, 25, 53, 35),
>-           DesktopRect::MakeLTRB(13, 35, 53, 45),
>-           DesktopRect::MakeLTRB(23, 45, 53, 55) } },
>+      {3,
>+       {DesktopRect::MakeLTRB(0, 0, 30, 30),
>+        DesktopRect::MakeLTRB(10, 10, 40, 40),
>+        DesktopRect::MakeLTRB(20, 20, 50, 50)},
>+       3,
>+       5,
>+       5,
>+       {DesktopRect::MakeLTRB(3, 5, 33, 15),
>+        DesktopRect::MakeLTRB(3, 15, 43, 25),
>+        DesktopRect::MakeLTRB(3, 25, 53, 35),
>+        DesktopRect::MakeLTRB(13, 35, 53, 45),
>+        DesktopRect::MakeLTRB(23, 45, 53, 55)}},
>   };
> 
>   for (size_t i = 0; i < (sizeof(cases) / sizeof(Case)); ++i) {
>@@ -408,27 +465,42 @@ TEST(DesktopRegionTest, Intersect) {
>     int expected_count;
>     DesktopRect expected_rects[5];
>   } cases[] = {
>-    { 1, { DesktopRect::MakeLTRB(0, 0, 100, 100) },
>-      1, { DesktopRect::MakeLTRB(50, 50, 150, 150) },
>-      1, { DesktopRect::MakeLTRB(50, 50, 100, 100) } },
>-
>-    { 1, { DesktopRect::MakeLTRB(100, 0, 200, 300) },
>-      1, { DesktopRect::MakeLTRB(0, 100, 300, 200) },
>-      1, { DesktopRect::MakeLTRB(100, 100, 200, 200) } },
>-
>-    { 1, { DesktopRect::MakeLTRB(0, 0, 100, 100) },
>-      2, { DesktopRect::MakeLTRB(50, 10, 150, 30),
>-           DesktopRect::MakeLTRB(50, 30, 160, 50) },
>-      1, { DesktopRect::MakeLTRB(50, 10, 100, 50) } },
>-
>-    { 1, { DesktopRect::MakeLTRB(0, 0, 100, 100) },
>-      2, { DesktopRect::MakeLTRB(50, 10, 150, 30),
>-           DesktopRect::MakeLTRB(50, 30,  90, 50) },
>-      2, { DesktopRect::MakeLTRB(50, 10, 100, 30),
>-           DesktopRect::MakeLTRB(50, 30, 90, 50) } },
>-    { 1, { DesktopRect::MakeLTRB(0, 0, 100, 100) },
>-      1, { DesktopRect::MakeLTRB(100, 50, 200, 200) },
>-      0, {} },
>+      {1,
>+       {DesktopRect::MakeLTRB(0, 0, 100, 100)},
>+       1,
>+       {DesktopRect::MakeLTRB(50, 50, 150, 150)},
>+       1,
>+       {DesktopRect::MakeLTRB(50, 50, 100, 100)}},
>+
>+      {1,
>+       {DesktopRect::MakeLTRB(100, 0, 200, 300)},
>+       1,
>+       {DesktopRect::MakeLTRB(0, 100, 300, 200)},
>+       1,
>+       {DesktopRect::MakeLTRB(100, 100, 200, 200)}},
>+
>+      {1,
>+       {DesktopRect::MakeLTRB(0, 0, 100, 100)},
>+       2,
>+       {DesktopRect::MakeLTRB(50, 10, 150, 30),
>+        DesktopRect::MakeLTRB(50, 30, 160, 50)},
>+       1,
>+       {DesktopRect::MakeLTRB(50, 10, 100, 50)}},
>+
>+      {1,
>+       {DesktopRect::MakeLTRB(0, 0, 100, 100)},
>+       2,
>+       {DesktopRect::MakeLTRB(50, 10, 150, 30),
>+        DesktopRect::MakeLTRB(50, 30, 90, 50)},
>+       2,
>+       {DesktopRect::MakeLTRB(50, 10, 100, 30),
>+        DesktopRect::MakeLTRB(50, 30, 90, 50)}},
>+      {1,
>+       {DesktopRect::MakeLTRB(0, 0, 100, 100)},
>+       1,
>+       {DesktopRect::MakeLTRB(100, 50, 200, 200)},
>+       0,
>+       {}},
>   };
> 
>   for (size_t i = 0; i < (sizeof(cases) / sizeof(Case)); ++i) {
>@@ -453,81 +525,123 @@ TEST(DesktopRegionTest, Subtract) {
>     int expected_count;
>     DesktopRect expected_rects[5];
>   } cases[] = {
>-    // Subtract one rect from another.
>-    { 1, { DesktopRect::MakeLTRB(0, 0, 100, 100) },
>-      1, { DesktopRect::MakeLTRB(50, 50, 150, 150) },
>-      2, { DesktopRect::MakeLTRB(0, 0, 100, 50),
>-           DesktopRect::MakeLTRB(0, 50, 50, 100)  } },
>-
>-    { 1, { DesktopRect::MakeLTRB(0, 0, 100, 100) },
>-      1, { DesktopRect::MakeLTRB(-50, -50, 50, 50) },
>-      2, { DesktopRect::MakeLTRB(50, 0, 100, 50),
>-           DesktopRect::MakeLTRB(0, 50, 100, 100)  } },
>-
>-    { 1, { DesktopRect::MakeLTRB(0, 0, 100, 100) },
>-      1, { DesktopRect::MakeLTRB(-50, 50, 50, 150) },
>-      2, { DesktopRect::MakeLTRB(0, 0, 100, 50),
>-           DesktopRect::MakeLTRB(50, 50, 100, 100)  } },
>-
>-    { 1, { DesktopRect::MakeLTRB(0, 0, 100, 100) },
>-      1, { DesktopRect::MakeLTRB(50, 50, 150, 70) },
>-      3, { DesktopRect::MakeLTRB(0, 0, 100, 50),
>-           DesktopRect::MakeLTRB(0, 50, 50, 70),
>-           DesktopRect::MakeLTRB(0, 70, 100, 100) } },
>-
>-    { 1, { DesktopRect::MakeLTRB(0, 0, 100, 100) },
>-      1, { DesktopRect::MakeLTRB(50, 50, 70, 70) },
>-      4, { DesktopRect::MakeLTRB(0, 0, 100, 50),
>-           DesktopRect::MakeLTRB(0, 50, 50, 70),
>-           DesktopRect::MakeLTRB(70, 50, 100, 70),
>-           DesktopRect::MakeLTRB(0, 70, 100, 100) } },
>-
>-    // Empty result.
>-    { 1, { DesktopRect::MakeLTRB(0, 0, 100, 100) },
>-      1, { DesktopRect::MakeLTRB(0, 0, 100, 100) },
>-      0, {} },
>-
>-    { 1, { DesktopRect::MakeLTRB(0, 0, 100, 100) },
>-      1, { DesktopRect::MakeLTRB(-10, -10, 110, 110) },
>-      0, {} },
>-
>-    { 2, { DesktopRect::MakeLTRB(0, 0, 100, 100),
>-           DesktopRect::MakeLTRB(50, 50, 150, 150) },
>-      2, { DesktopRect::MakeLTRB(0, 0, 100, 100),
>-           DesktopRect::MakeLTRB(50, 50, 150, 150) },
>-      0, {} },
>-
>-    // One rect out of disjoint set.
>-    { 3, { DesktopRect::MakeLTRB(0, 0, 10, 10),
>-           DesktopRect::MakeLTRB(20, 20, 30, 30),
>-           DesktopRect::MakeLTRB(40, 0, 50, 10) },
>-      1, { DesktopRect::MakeLTRB(20, 20, 30, 30) },
>-      2, { DesktopRect::MakeLTRB(0, 0, 10, 10),
>-           DesktopRect::MakeLTRB(40, 0, 50, 10) } },
>-
>-    // Row merging.
>-    { 3, { DesktopRect::MakeLTRB(0, 0, 100, 50),
>-           DesktopRect::MakeLTRB(0, 50, 150, 70),
>-           DesktopRect::MakeLTRB(0, 70, 100, 100) },
>-      1, { DesktopRect::MakeLTRB(100, 50, 150, 70) },
>-      1, { DesktopRect::MakeLTRB(0, 0, 100, 100) } },
>-
>-    // No-op subtraction.
>-    { 1, { DesktopRect::MakeLTRB(0, 0, 100, 100) },
>-      1, { DesktopRect::MakeLTRB(100, 0, 200, 100) },
>-      1, { DesktopRect::MakeLTRB(0, 0, 100, 100) } },
>-
>-    { 1, { DesktopRect::MakeLTRB(0, 0, 100, 100) },
>-      1, { DesktopRect::MakeLTRB(-100, 0, 0, 100) },
>-      1, { DesktopRect::MakeLTRB(0, 0, 100, 100) } },
>-
>-    { 1, { DesktopRect::MakeLTRB(0, 0, 100, 100) },
>-      1, { DesktopRect::MakeLTRB(0, 100, 0, 200) },
>-      1, { DesktopRect::MakeLTRB(0, 0, 100, 100) } },
>-
>-    { 1, { DesktopRect::MakeLTRB(0, 0, 100, 100) },
>-      1, { DesktopRect::MakeLTRB(0, -100, 100, 0) },
>-      1, { DesktopRect::MakeLTRB(0, 0, 100, 100) } },
>+      // Subtract one rect from another.
>+      {1,
>+       {DesktopRect::MakeLTRB(0, 0, 100, 100)},
>+       1,
>+       {DesktopRect::MakeLTRB(50, 50, 150, 150)},
>+       2,
>+       {DesktopRect::MakeLTRB(0, 0, 100, 50),
>+        DesktopRect::MakeLTRB(0, 50, 50, 100)}},
>+
>+      {1,
>+       {DesktopRect::MakeLTRB(0, 0, 100, 100)},
>+       1,
>+       {DesktopRect::MakeLTRB(-50, -50, 50, 50)},
>+       2,
>+       {DesktopRect::MakeLTRB(50, 0, 100, 50),
>+        DesktopRect::MakeLTRB(0, 50, 100, 100)}},
>+
>+      {1,
>+       {DesktopRect::MakeLTRB(0, 0, 100, 100)},
>+       1,
>+       {DesktopRect::MakeLTRB(-50, 50, 50, 150)},
>+       2,
>+       {DesktopRect::MakeLTRB(0, 0, 100, 50),
>+        DesktopRect::MakeLTRB(50, 50, 100, 100)}},
>+
>+      {1,
>+       {DesktopRect::MakeLTRB(0, 0, 100, 100)},
>+       1,
>+       {DesktopRect::MakeLTRB(50, 50, 150, 70)},
>+       3,
>+       {DesktopRect::MakeLTRB(0, 0, 100, 50),
>+        DesktopRect::MakeLTRB(0, 50, 50, 70),
>+        DesktopRect::MakeLTRB(0, 70, 100, 100)}},
>+
>+      {1,
>+       {DesktopRect::MakeLTRB(0, 0, 100, 100)},
>+       1,
>+       {DesktopRect::MakeLTRB(50, 50, 70, 70)},
>+       4,
>+       {DesktopRect::MakeLTRB(0, 0, 100, 50),
>+        DesktopRect::MakeLTRB(0, 50, 50, 70),
>+        DesktopRect::MakeLTRB(70, 50, 100, 70),
>+        DesktopRect::MakeLTRB(0, 70, 100, 100)}},
>+
>+      // Empty result.
>+      {1,
>+       {DesktopRect::MakeLTRB(0, 0, 100, 100)},
>+       1,
>+       {DesktopRect::MakeLTRB(0, 0, 100, 100)},
>+       0,
>+       {}},
>+
>+      {1,
>+       {DesktopRect::MakeLTRB(0, 0, 100, 100)},
>+       1,
>+       {DesktopRect::MakeLTRB(-10, -10, 110, 110)},
>+       0,
>+       {}},
>+
>+      {2,
>+       {DesktopRect::MakeLTRB(0, 0, 100, 100),
>+        DesktopRect::MakeLTRB(50, 50, 150, 150)},
>+       2,
>+       {DesktopRect::MakeLTRB(0, 0, 100, 100),
>+        DesktopRect::MakeLTRB(50, 50, 150, 150)},
>+       0,
>+       {}},
>+
>+      // One rect out of disjoint set.
>+      {3,
>+       {DesktopRect::MakeLTRB(0, 0, 10, 10),
>+        DesktopRect::MakeLTRB(20, 20, 30, 30),
>+        DesktopRect::MakeLTRB(40, 0, 50, 10)},
>+       1,
>+       {DesktopRect::MakeLTRB(20, 20, 30, 30)},
>+       2,
>+       {DesktopRect::MakeLTRB(0, 0, 10, 10),
>+        DesktopRect::MakeLTRB(40, 0, 50, 10)}},
>+
>+      // Row merging.
>+      {3,
>+       {DesktopRect::MakeLTRB(0, 0, 100, 50),
>+        DesktopRect::MakeLTRB(0, 50, 150, 70),
>+        DesktopRect::MakeLTRB(0, 70, 100, 100)},
>+       1,
>+       {DesktopRect::MakeLTRB(100, 50, 150, 70)},
>+       1,
>+       {DesktopRect::MakeLTRB(0, 0, 100, 100)}},
>+
>+      // No-op subtraction.
>+      {1,
>+       {DesktopRect::MakeLTRB(0, 0, 100, 100)},
>+       1,
>+       {DesktopRect::MakeLTRB(100, 0, 200, 100)},
>+       1,
>+       {DesktopRect::MakeLTRB(0, 0, 100, 100)}},
>+
>+      {1,
>+       {DesktopRect::MakeLTRB(0, 0, 100, 100)},
>+       1,
>+       {DesktopRect::MakeLTRB(-100, 0, 0, 100)},
>+       1,
>+       {DesktopRect::MakeLTRB(0, 0, 100, 100)}},
>+
>+      {1,
>+       {DesktopRect::MakeLTRB(0, 0, 100, 100)},
>+       1,
>+       {DesktopRect::MakeLTRB(0, 100, 0, 200)},
>+       1,
>+       {DesktopRect::MakeLTRB(0, 0, 100, 100)}},
>+
>+      {1,
>+       {DesktopRect::MakeLTRB(0, 0, 100, 100)},
>+       1,
>+       {DesktopRect::MakeLTRB(0, -100, 100, 0)},
>+       1,
>+       {DesktopRect::MakeLTRB(0, 0, 100, 100)}},
>   };
> 
>   for (size_t i = 0; i < (sizeof(cases) / sizeof(Case)); ++i) {
>@@ -552,19 +666,21 @@ TEST(DesktopRegionTest, SubtractRectOnSameRow) {
>   struct SpanSet {
>     int count;
>     struct Range {
>-     int start;
>-     int end;
>+      int start;
>+      int end;
>     } spans[3];
>   } span_sets[] = {
>-    {1, { {0, 3} } },
>-    {1, { {0, 5} } },
>-    {1, { {0, 7} } },
>-    {1, { {0, 12} } },
>-    {2, { {0, 3}, {4, 5}, {6, 16} } },
>+      {1, {{0, 3}}},
>+      {1, {{0, 5}}},
>+      {1, {{0, 7}}},
>+      {1, {{0, 12}}},
>+      {2, {{0, 3}, {4, 5}, {6, 16}}},
>   };
> 
>   DesktopRegion base_region;
>-  bool base_map[kMapWidth] = { false, };
>+  bool base_map[kMapWidth] = {
>+      false,
>+  };
> 
>   base_region.AddRect(DesktopRect::MakeXYWH(5, 0, 5, 1));
>   std::fill_n(base_map + 5, 5, true);
>@@ -599,7 +715,9 @@ TEST(DesktopRegionTest, SubtractRectOnSameRow) {
>       }
>       r.Subtract(region2);
> 
>-      bool map[kMapWidth] = { false, };
>+      bool map[kMapWidth] = {
>+          false,
>+      };
> 
>       int pos = -1;
>       for (DesktopRegion::Iterator it(r); !it.IsAtEnd(); it.Advance()) {
>@@ -623,19 +741,21 @@ TEST(DesktopRegionTest, SubtractRectOnSameCol) {
>   struct SpanSet {
>     int count;
>     struct Range {
>-     int start;
>-     int end;
>+      int start;
>+      int end;
>     } spans[3];
>   } span_sets[] = {
>-    {1, { {0, 3} } },
>-    {1, { {0, 5} } },
>-    {1, { {0, 7} } },
>-    {1, { {0, 12} } },
>-    {2, { {0, 3}, {4, 5}, {6, 16} } },
>+      {1, {{0, 3}}},
>+      {1, {{0, 5}}},
>+      {1, {{0, 7}}},
>+      {1, {{0, 12}}},
>+      {2, {{0, 3}, {4, 5}, {6, 16}}},
>   };
> 
>   DesktopRegion base_region;
>-  bool base_map[kMapHeight] = { false, };
>+  bool base_map[kMapHeight] = {
>+      false,
>+  };
> 
>   base_region.AddRect(DesktopRect::MakeXYWH(0, 5, 1, 5));
>   std::fill_n(base_map + 5, 5, true);
>@@ -670,7 +790,9 @@ TEST(DesktopRegionTest, SubtractRectOnSameCol) {
>       }
>       r.Subtract(region2);
> 
>-      bool map[kMapHeight] = { false, };
>+      bool map[kMapHeight] = {
>+          false,
>+      };
> 
>       int pos = -1;
>       for (DesktopRegion::Iterator it(r); !it.IsAtEnd(); it.Advance()) {
>@@ -686,19 +808,18 @@ TEST(DesktopRegionTest, SubtractRectOnSameCol) {
>   }
> }
> 
>-
> TEST(DesktopRegionTest, DISABLED_Performance) {
>   for (int c = 0; c < 1000; ++c) {
>     DesktopRegion r;
>     for (int i = 0; i < 10; ++i) {
>-      r.AddRect(DesktopRect::MakeXYWH(
>-          RadmonInt(1000), RadmonInt(1000), 200, 200));
>+      r.AddRect(
>+          DesktopRect::MakeXYWH(RadmonInt(1000), RadmonInt(1000), 200, 200));
>     }
> 
>     for (int i = 0; i < 1000; ++i) {
>-      r.AddRect(DesktopRect::MakeXYWH(
>-          RadmonInt(1000), RadmonInt(1000),
>-          5 + RadmonInt(10) * 5, 5 + RadmonInt(10) * 5));
>+      r.AddRect(DesktopRect::MakeXYWH(RadmonInt(1000), RadmonInt(1000),
>+                                      5 + RadmonInt(10) * 5,
>+                                      5 + RadmonInt(10) * 5));
>     }
> 
>     // Iterate over the rectangles.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/differ_block.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/differ_block.cc
>index 3315247238f..dd9ab457e0f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/differ_block.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/differ_block.cc
>@@ -12,8 +12,8 @@
> 
> #include <string.h>
> 
>-#include "typedefs.h"  // NOLINT(build/include)
> #include "modules/desktop_capture/differ_vector_sse2.h"
>+#include "rtc_base/system/arch.h"
> #include "system_wrappers/include/cpu_features_wrapper.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/differ_block.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/differ_block.h
>index 7bf9a7b1db5..2019ae03f0f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/differ_block.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/differ_block.h
>@@ -35,9 +35,7 @@ bool BlockDifference(const uint8_t* image1,
> 
> // Low level function to compare 2 blocks of pixels of size
> // (kBlockSize, kBlockSize).  Returns whether the blocks differ.
>-bool BlockDifference(const uint8_t* image1,
>-                     const uint8_t* image2,
>-                     int stride);
>+bool BlockDifference(const uint8_t* image1, const uint8_t* image2, int stride);
> 
> }  // namespace webrtc
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/differ_block_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/differ_block_unittest.cc
>index 3f08632aaa7..ac8b3a17654 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/differ_block_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/differ_block_unittest.cc
>@@ -27,9 +27,9 @@ static void GenerateData(uint8_t* data, int size) {
> static const int kSizeOfBlock = kBlockSize * kBlockSize * kBytesPerPixel;
> uint8_t block_buffer[kSizeOfBlock * 2 + 16];
> 
>-void PrepareBuffers(uint8_t* &block1, uint8_t* &block2) {
>-  block1 = reinterpret_cast<uint8_t*>
>-      ((reinterpret_cast<uintptr_t>(&block_buffer[0]) + 15) & ~15);
>+void PrepareBuffers(uint8_t*& block1, uint8_t*& block2) {
>+  block1 = reinterpret_cast<uint8_t*>(
>+      (reinterpret_cast<uintptr_t>(&block_buffer[0]) + 15) & ~15);
>   GenerateData(block1, kSizeOfBlock);
>   block2 = block1 + kSizeOfBlock;
>   memcpy(block2, block1, kSizeOfBlock);
>@@ -51,7 +51,7 @@ TEST(BlockDifferenceTestLast, BlockDifference) {
>   uint8_t* block1;
>   uint8_t* block2;
>   PrepareBuffers(block1, block2);
>-  block2[kSizeOfBlock-2] += 1;
>+  block2[kSizeOfBlock - 2] += 1;
> 
>   for (int i = 0; i < kTimesToRun; ++i) {
>     int result = BlockDifference(block1, block2, kBlockSize * kBytesPerPixel);
>@@ -63,7 +63,7 @@ TEST(BlockDifferenceTestMid, BlockDifference) {
>   uint8_t* block1;
>   uint8_t* block2;
>   PrepareBuffers(block1, block2);
>-  block2[kSizeOfBlock/2+1] += 1;
>+  block2[kSizeOfBlock / 2 + 1] += 1;
> 
>   for (int i = 0; i < kTimesToRun; ++i) {
>     int result = BlockDifference(block1, block2, kBlockSize * kBytesPerPixel);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/differ_vector_sse2.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/differ_vector_sse2.cc
>index 9b552e0f611..1c8b602d719 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/differ_vector_sse2.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/differ_vector_sse2.cc
>@@ -13,8 +13,8 @@
> #if defined(_MSC_VER)
> #include <intrin.h>
> #else
>-#include <mmintrin.h>
> #include <emmintrin.h>
>+#include <mmintrin.h>
> #endif
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/fake_desktop_capturer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/fake_desktop_capturer.h
>index cb1cfe1ba68..82c053cbe09 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/fake_desktop_capturer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/fake_desktop_capturer.h
>@@ -13,8 +13,8 @@
> 
> #include <memory>
> 
>-#include "modules/desktop_capture/desktop_capturer.h"
> #include "modules/desktop_capture/desktop_capture_types.h"
>+#include "modules/desktop_capture/desktop_capturer.h"
> #include "modules/desktop_capture/desktop_frame_generator.h"
> #include "modules/desktop_capture/shared_memory.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/fallback_desktop_capturer_wrapper.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/fallback_desktop_capturer_wrapper.cc
>index b261a030281..2dd8ce1f5ea 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/fallback_desktop_capturer_wrapper.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/fallback_desktop_capturer_wrapper.cc
>@@ -51,16 +51,16 @@ SharedMemoryFactoryProxy::SharedMemoryFactoryProxy(
> }
> 
> // static
>-std::unique_ptr<SharedMemoryFactory>
>-SharedMemoryFactoryProxy::Create(SharedMemoryFactory* factory) {
>+std::unique_ptr<SharedMemoryFactory> SharedMemoryFactoryProxy::Create(
>+    SharedMemoryFactory* factory) {
>   return std::unique_ptr<SharedMemoryFactory>(
>       new SharedMemoryFactoryProxy(factory));
> }
> 
> SharedMemoryFactoryProxy::~SharedMemoryFactoryProxy() = default;
> 
>-std::unique_ptr<SharedMemory>
>-SharedMemoryFactoryProxy::CreateSharedMemory(size_t size) {
>+std::unique_ptr<SharedMemory> SharedMemoryFactoryProxy::CreateSharedMemory(
>+    size_t size) {
>   RTC_DCHECK(thread_checker_.CalledOnValidThread());
>   return factory_->CreateSharedMemory(size);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/fallback_desktop_capturer_wrapper.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/fallback_desktop_capturer_wrapper.h
>index a6f2ceb100d..97affbe461b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/fallback_desktop_capturer_wrapper.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/fallback_desktop_capturer_wrapper.h
>@@ -13,8 +13,8 @@
> 
> #include <memory>
> 
>-#include "modules/desktop_capture/desktop_capturer.h"
> #include "modules/desktop_capture/desktop_capture_types.h"
>+#include "modules/desktop_capture/desktop_capturer.h"
> #include "modules/desktop_capture/desktop_frame.h"
> #include "modules/desktop_capture/shared_memory.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/fallback_desktop_capturer_wrapper_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/fallback_desktop_capturer_wrapper_unittest.cc
>index ca11162dc4c..f2d6071598d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/fallback_desktop_capturer_wrapper_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/fallback_desktop_capturer_wrapper_unittest.cc
>@@ -180,8 +180,8 @@ TEST_F(FallbackDesktopCapturerWrapperTest, BothFailed) {
> }
> 
> TEST_F(FallbackDesktopCapturerWrapperTest, WithSharedMemory) {
>-  wrapper_->SetSharedMemoryFactory(std::unique_ptr<SharedMemoryFactory>(
>-      new FakeSharedMemoryFactory()));
>+  wrapper_->SetSharedMemoryFactory(
>+      std::unique_ptr<SharedMemoryFactory>(new FakeSharedMemoryFactory()));
>   wrapper_->CaptureFrame();
>   main_capturer_->set_result(DesktopCapturer::Result::ERROR_TEMPORARY);
>   wrapper_->CaptureFrame();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/desktop_configuration.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/desktop_configuration.h
>index 2cf28406fae..5d96aa9b939 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/desktop_configuration.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/desktop_configuration.h
>@@ -14,7 +14,6 @@
> #include <ApplicationServices/ApplicationServices.h>
> #include <vector>
> 
>-#include "typedefs.h"  // NOLINT(build/include)
> #include "modules/desktop_capture/desktop_geometry.h"
> 
> namespace webrtc {
>@@ -40,6 +39,9 @@ struct MacDisplayConfiguration {
> 
>   // Scale factor from DIPs to physical pixels.
>   float dip_to_pixel_scale = 1.0f;
>+
>+  // Display type, built-in or external.
>+  bool is_builtin;
> };
> 
> typedef std::vector<MacDisplayConfiguration> MacDisplayConfigurations;
>@@ -67,7 +69,9 @@ struct MacDesktopConfiguration {
>   // Returns true if the given desktop configuration equals this one.
>   bool Equals(const MacDesktopConfiguration& other);
> 
>-  // Returns the pointer to the display configuration with the specified id.
>+  // If |id| corresponds to the built-in display, return its configuration,
>+  // otherwise return the configuration for the display with the specified id,
>+  // or nullptr if no such display exists.
>   const MacDisplayConfiguration* FindDisplayConfigurationById(
>       CGDirectDisplayID id);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/desktop_configuration.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/desktop_configuration.mm
>index e6316632c5f..5962fabb70f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/desktop_configuration.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/desktop_configuration.mm
>@@ -69,6 +69,9 @@ MacDisplayConfiguration GetConfigurationForScreen(NSScreen* screen) {
>     display_config.pixel_bounds = display_config.bounds;
>   }
> 
>+  // Determine if the display is built-in or external.
>+  display_config.is_builtin = CGDisplayIsBuiltin(display_config.id);
>+
>   return display_config;
> }
> 
>@@ -164,14 +167,19 @@ bool MacDesktopConfiguration::Equals(const MacDesktopConfiguration& other) {
>       displays == other.displays;
> }
> 
>-// Finds the display configuration with the specified id.
> const MacDisplayConfiguration*
> MacDesktopConfiguration::FindDisplayConfigurationById(
>     CGDirectDisplayID id) {
>+  bool is_builtin = CGDisplayIsBuiltin(id);
>   for (MacDisplayConfigurations::const_iterator it = displays.begin();
>       it != displays.end(); ++it) {
>-    if (it->id == id)
>-      return &(*it);
>+    // The MBP having both discrete and integrated graphic cards will do
>+    // automate graphics switching by default. When it switches from discrete to
>+    // integrated one, the current display ID of the built-in display will
>+    // change and this will cause screen capture stops.
>+    // So make screen capture of built-in display continuing even if its display
>+    // ID is changed.
>+    if ((is_builtin && it->is_builtin) || (!is_builtin && it->id == id)) return &(*it);
>   }
>   return NULL;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/desktop_configuration_monitor.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/desktop_configuration_monitor.cc
>index 4eeee321066..d4364bedc2b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/desktop_configuration_monitor.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/desktop_configuration_monitor.cc
>@@ -12,6 +12,7 @@
> 
> #include "modules/desktop_capture/mac/desktop_configuration.h"
> #include "rtc_base/logging.h"
>+#include "rtc_base/trace_event.h"
> #include "system_wrappers/include/event_wrapper.h"
> 
> namespace webrtc {
>@@ -42,7 +43,7 @@ DesktopConfigurationMonitor::~DesktopConfigurationMonitor() {
> 
> void DesktopConfigurationMonitor::Lock() {
>   if (!display_configuration_capture_event_->Wait(
>-              kDisplayConfigurationEventTimeoutMs)) {
>+          kDisplayConfigurationEventTimeoutMs)) {
>     RTC_LOG_F(LS_ERROR) << "Event wait timed out.";
>     abort();
>   }
>@@ -56,7 +57,7 @@ void DesktopConfigurationMonitor::Unlock() {
> void DesktopConfigurationMonitor::DisplaysReconfiguredCallback(
>     CGDirectDisplayID display,
>     CGDisplayChangeSummaryFlags flags,
>-    void *user_parameter) {
>+    void* user_parameter) {
>   DesktopConfigurationMonitor* monitor =
>       reinterpret_cast<DesktopConfigurationMonitor*>(user_parameter);
>   monitor->DisplaysReconfigured(display, flags);
>@@ -65,6 +66,11 @@ void DesktopConfigurationMonitor::DisplaysReconfiguredCallback(
> void DesktopConfigurationMonitor::DisplaysReconfigured(
>     CGDirectDisplayID display,
>     CGDisplayChangeSummaryFlags flags) {
>+  TRACE_EVENT0("webrtc", "DesktopConfigurationMonitor::DisplaysReconfigured");
>+  RTC_LOG(LS_INFO) << "DisplaysReconfigured: "
>+                   << "DisplayID " << display << "; ChangeSummaryFlags "
>+                   << flags;
>+
>   if (flags & kCGDisplayBeginConfigurationFlag) {
>     if (reconfiguring_displays_.empty()) {
>       // If this is the first display to start reconfiguring then wait on
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/desktop_configuration_monitor.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/desktop_configuration_monitor.h
>index 746fbb271d6..5215e5496ad 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/desktop_configuration_monitor.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/desktop_configuration_monitor.h
>@@ -45,7 +45,7 @@ class DesktopConfigurationMonitor : public rtc::RefCountedBase {
>  private:
>   static void DisplaysReconfiguredCallback(CGDirectDisplayID display,
>                                            CGDisplayChangeSummaryFlags flags,
>-                                           void *user_parameter);
>+                                           void* user_parameter);
>   void DisplaysReconfigured(CGDirectDisplayID display,
>                             CGDisplayChangeSummaryFlags flags);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/desktop_frame_cgimage.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/desktop_frame_cgimage.h
>new file mode 100644
>index 00000000000..1adc8b02cfc
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/desktop_frame_cgimage.h
>@@ -0,0 +1,48 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_DESKTOP_CAPTURE_MAC_DESKTOP_FRAME_CGIMAGE_H_
>+#define MODULES_DESKTOP_CAPTURE_MAC_DESKTOP_FRAME_CGIMAGE_H_
>+
>+#include <CoreGraphics/CoreGraphics.h>
>+
>+#include <memory>
>+
>+#include "modules/desktop_capture/desktop_frame.h"
>+#include "sdk/objc/Framework/Classes/Common/scoped_cftyperef.h"
>+
>+namespace webrtc {
>+
>+class DesktopFrameCGImage final : public DesktopFrame {
>+ public:
>+  // Create an image containing a snapshot of the display at the time this is
>+  // being called.
>+  static std::unique_ptr<DesktopFrameCGImage> CreateForDisplay(
>+      CGDirectDisplayID display_id);
>+
>+  ~DesktopFrameCGImage() override;
>+
>+ private:
>+  // This constructor expects |cg_image| to hold a non-null CGImageRef.
>+  DesktopFrameCGImage(DesktopSize size,
>+                      int stride,
>+                      uint8_t* data,
>+                      rtc::ScopedCFTypeRef<CGImageRef> cg_image,
>+                      rtc::ScopedCFTypeRef<CFDataRef> cg_data);
>+
>+  const rtc::ScopedCFTypeRef<CGImageRef> cg_image_;
>+  const rtc::ScopedCFTypeRef<CFDataRef> cg_data_;
>+
>+  RTC_DISALLOW_COPY_AND_ASSIGN(DesktopFrameCGImage);
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_DESKTOP_CAPTURE_MAC_DESKTOP_FRAME_CGIMAGE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/desktop_frame_cgimage.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/desktop_frame_cgimage.mm
>new file mode 100644
>index 00000000000..23afaaa1383
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/desktop_frame_cgimage.mm
>@@ -0,0 +1,67 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/desktop_capture/mac/desktop_frame_cgimage.h"
>+
>+#include "rtc_base/checks.h"
>+#include "rtc_base/logging.h"
>+
>+namespace webrtc {
>+
>+// static
>+std::unique_ptr<DesktopFrameCGImage> DesktopFrameCGImage::CreateForDisplay(
>+    CGDirectDisplayID display_id) {
>+  // Create an image containing a snapshot of the display.
>+  rtc::ScopedCFTypeRef<CGImageRef> cg_image(CGDisplayCreateImage(display_id));
>+  if (!cg_image) {
>+    return nullptr;
>+  }
>+
>+  // Verify that the image has 32-bit depth.
>+  int bits_per_pixel = CGImageGetBitsPerPixel(cg_image.get());
>+  if (bits_per_pixel / 8 != DesktopFrame::kBytesPerPixel) {
>+    RTC_LOG(LS_ERROR) << "CGDisplayCreateImage() returned imaged with " << bits_per_pixel
>+                      << " bits per pixel. Only 32-bit depth is supported.";
>+    return nullptr;
>+  }
>+
>+  // Request access to the raw pixel data via the image's DataProvider.
>+  CGDataProviderRef cg_provider = CGImageGetDataProvider(cg_image.get());
>+  RTC_DCHECK(cg_provider);
>+
>+  // CGDataProviderCopyData returns a new data object containing a copy of the providerâs
>+  // data.
>+  rtc::ScopedCFTypeRef<CFDataRef> cg_data(CGDataProviderCopyData(cg_provider));
>+  RTC_DCHECK(cg_data);
>+
>+  // CFDataGetBytePtr returns a read-only pointer to the bytes of a CFData object.
>+  uint8_t* data = const_cast<uint8_t*>(CFDataGetBytePtr(cg_data.get()));
>+  RTC_DCHECK(data);
>+
>+  DesktopSize size(CGImageGetWidth(cg_image.get()), CGImageGetHeight(cg_image.get()));
>+  int stride = CGImageGetBytesPerRow(cg_image.get());
>+
>+  return std::unique_ptr<DesktopFrameCGImage>(
>+      new DesktopFrameCGImage(size, stride, data, cg_image, cg_data));
>+}
>+
>+DesktopFrameCGImage::DesktopFrameCGImage(DesktopSize size,
>+                                         int stride,
>+                                         uint8_t* data,
>+                                         rtc::ScopedCFTypeRef<CGImageRef> cg_image,
>+                                         rtc::ScopedCFTypeRef<CFDataRef> cg_data)
>+    : DesktopFrame(size, stride, data, nullptr), cg_image_(cg_image), cg_data_(cg_data) {
>+  RTC_DCHECK(cg_image_);
>+  RTC_DCHECK(cg_data_);
>+}
>+
>+DesktopFrameCGImage::~DesktopFrameCGImage() = default;
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/desktop_frame_iosurface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/desktop_frame_iosurface.h
>new file mode 100644
>index 00000000000..ced60f80571
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/desktop_frame_iosurface.h
>@@ -0,0 +1,44 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_DESKTOP_CAPTURE_MAC_DESKTOP_FRAME_IOSURFACE_H_
>+#define MODULES_DESKTOP_CAPTURE_MAC_DESKTOP_FRAME_IOSURFACE_H_
>+
>+#include <CoreGraphics/CoreGraphics.h>
>+#include <IOSurface/IOSurface.h>
>+
>+#include <memory>
>+
>+#include "modules/desktop_capture/desktop_frame.h"
>+#include "sdk/objc/Framework/Classes/Common/scoped_cftyperef.h"
>+
>+namespace webrtc {
>+
>+class DesktopFrameIOSurface final : public DesktopFrame {
>+ public:
>+  // Lock an IOSurfaceRef containing a snapshot of a display. Return NULL if
>+  // failed to lock.
>+  static std::unique_ptr<DesktopFrameIOSurface> Wrap(
>+      rtc::ScopedCFTypeRef<IOSurfaceRef> io_surface);
>+
>+  ~DesktopFrameIOSurface() override;
>+
>+ private:
>+  // This constructor expects |io_surface| to hold a non-null IOSurfaceRef.
>+  explicit DesktopFrameIOSurface(rtc::ScopedCFTypeRef<IOSurfaceRef> io_surface);
>+
>+  const rtc::ScopedCFTypeRef<IOSurfaceRef> io_surface_;
>+
>+  RTC_DISALLOW_COPY_AND_ASSIGN(DesktopFrameIOSurface);
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_DESKTOP_CAPTURE_MAC_DESKTOP_FRAME_IOSURFACE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/desktop_frame_iosurface.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/desktop_frame_iosurface.mm
>new file mode 100644
>index 00000000000..b59b319db9a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/desktop_frame_iosurface.mm
>@@ -0,0 +1,61 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/desktop_capture/mac/desktop_frame_iosurface.h"
>+
>+#include "rtc_base/checks.h"
>+#include "rtc_base/logging.h"
>+
>+namespace webrtc {
>+
>+// static
>+std::unique_ptr<DesktopFrameIOSurface> DesktopFrameIOSurface::Wrap(
>+    rtc::ScopedCFTypeRef<IOSurfaceRef> io_surface) {
>+  if (!io_surface) {
>+    return nullptr;
>+  }
>+
>+  IOSurfaceIncrementUseCount(io_surface.get());
>+  IOReturn status = IOSurfaceLock(io_surface.get(), kIOSurfaceLockReadOnly, nullptr);
>+  if (status != kIOReturnSuccess) {
>+    RTC_LOG(LS_ERROR) << "Failed to lock the IOSurface with status " << status;
>+    IOSurfaceDecrementUseCount(io_surface.get());
>+    return nullptr;
>+  }
>+
>+  // Verify that the image has 32-bit depth.
>+  int bytes_per_pixel = IOSurfaceGetBytesPerElement(io_surface.get());
>+  if (bytes_per_pixel != DesktopFrame::kBytesPerPixel) {
>+    RTC_LOG(LS_ERROR) << "CGDisplayStream handler returned IOSurface with " << (8 * bytes_per_pixel)
>+                      << " bits per pixel. Only 32-bit depth is supported.";
>+    IOSurfaceUnlock(io_surface.get(), kIOSurfaceLockReadOnly, nullptr);
>+    IOSurfaceDecrementUseCount(io_surface.get());
>+    return nullptr;
>+  }
>+
>+  return std::unique_ptr<DesktopFrameIOSurface>(new DesktopFrameIOSurface(io_surface));
>+}
>+
>+DesktopFrameIOSurface::DesktopFrameIOSurface(rtc::ScopedCFTypeRef<IOSurfaceRef> io_surface)
>+    : DesktopFrame(
>+          DesktopSize(IOSurfaceGetWidth(io_surface.get()), IOSurfaceGetHeight(io_surface.get())),
>+          IOSurfaceGetBytesPerRow(io_surface.get()),
>+          static_cast<uint8_t*>(IOSurfaceGetBaseAddress(io_surface.get())),
>+          nullptr),
>+      io_surface_(io_surface) {
>+  RTC_DCHECK(io_surface_);
>+}
>+
>+DesktopFrameIOSurface::~DesktopFrameIOSurface() {
>+  IOSurfaceUnlock(io_surface_.get(), kIOSurfaceLockReadOnly, nullptr);
>+  IOSurfaceDecrementUseCount(io_surface_.get());
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/desktop_frame_provider.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/desktop_frame_provider.h
>new file mode 100644
>index 00000000000..7e172716ea1
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/desktop_frame_provider.h
>@@ -0,0 +1,58 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_DESKTOP_CAPTURE_MAC_DESKTOP_FRAME_PROVIDER_H_
>+#define MODULES_DESKTOP_CAPTURE_MAC_DESKTOP_FRAME_PROVIDER_H_
>+
>+#include <CoreGraphics/CoreGraphics.h>
>+#include <IOSurface/IOSurface.h>
>+
>+#include <map>
>+#include <memory>
>+
>+#include "modules/desktop_capture/shared_desktop_frame.h"
>+#include "rtc_base/thread_checker.h"
>+#include "sdk/objc/Framework/Classes/Common/scoped_cftyperef.h"
>+
>+namespace webrtc {
>+
>+class DesktopFrameProvider {
>+ public:
>+  explicit DesktopFrameProvider(bool allow_iosurface);
>+  ~DesktopFrameProvider();
>+
>+  // The caller takes ownership of the returned desktop frame. Otherwise
>+  // returns null if |display_id| is invalid or not ready. Note that this
>+  // function does not remove the frame from the internal container. Caller
>+  // has to call the Release function.
>+  std::unique_ptr<DesktopFrame> TakeLatestFrameForDisplay(
>+      CGDirectDisplayID display_id);
>+
>+  // OS sends the latest IOSurfaceRef through
>+  // CGDisplayStreamFrameAvailableHandler callback; we store it here.
>+  void InvalidateIOSurface(CGDirectDisplayID display_id,
>+                           rtc::ScopedCFTypeRef<IOSurfaceRef> io_surface);
>+
>+  // Expected to be called before stopping the CGDisplayStreamRef streams.
>+  void Release();
>+
>+ private:
>+  rtc::ThreadChecker thread_checker_;
>+  const bool allow_iosurface_;
>+
>+  // Most recent IOSurface that contains a capture of matching display.
>+  std::map<CGDirectDisplayID, std::unique_ptr<SharedDesktopFrame>> io_surfaces_;
>+
>+  RTC_DISALLOW_COPY_AND_ASSIGN(DesktopFrameProvider);
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_DESKTOP_CAPTURE_MAC_DESKTOP_FRAME_PROVIDER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/desktop_frame_provider.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/desktop_frame_provider.mm
>new file mode 100644
>index 00000000000..42343150b5d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/desktop_frame_provider.mm
>@@ -0,0 +1,73 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/desktop_capture/mac/desktop_frame_provider.h"
>+
>+#include <utility>
>+
>+#include "modules/desktop_capture/mac/desktop_frame_cgimage.h"
>+#include "modules/desktop_capture/mac/desktop_frame_iosurface.h"
>+
>+namespace webrtc {
>+
>+DesktopFrameProvider::DesktopFrameProvider(bool allow_iosurface)
>+    : allow_iosurface_(allow_iosurface) {
>+  thread_checker_.DetachFromThread();
>+}
>+
>+DesktopFrameProvider::~DesktopFrameProvider() {
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+
>+  Release();
>+}
>+
>+std::unique_ptr<DesktopFrame> DesktopFrameProvider::TakeLatestFrameForDisplay(
>+    CGDirectDisplayID display_id) {
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+
>+  if (!allow_iosurface_) {
>+    // Regenerate a snapshot.
>+    return DesktopFrameCGImage::CreateForDisplay(display_id);
>+  }
>+
>+  if (io_surfaces_[display_id]) {
>+    return io_surfaces_[display_id]->Share();
>+  }
>+
>+  return nullptr;
>+}
>+
>+void DesktopFrameProvider::InvalidateIOSurface(CGDirectDisplayID display_id,
>+                                               rtc::ScopedCFTypeRef<IOSurfaceRef> io_surface) {
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+
>+  if (!allow_iosurface_) {
>+    return;
>+  }
>+
>+  std::unique_ptr<DesktopFrameIOSurface> desktop_frame_iosurface =
>+      DesktopFrameIOSurface::Wrap(io_surface);
>+
>+  io_surfaces_[display_id] = desktop_frame_iosurface ?
>+      SharedDesktopFrame::Wrap(std::move(desktop_frame_iosurface)) :
>+      nullptr;
>+}
>+
>+void DesktopFrameProvider::Release() {
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+
>+  if (!allow_iosurface_) {
>+    return;
>+  }
>+
>+  io_surfaces_.clear();
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/full_screen_chrome_window_detector.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/full_screen_chrome_window_detector.cc
>index a9f334fb7e5..0cda46c93fa 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/full_screen_chrome_window_detector.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/full_screen_chrome_window_detector.cc
>@@ -27,7 +27,7 @@ const int64_t kUpdateIntervalMs = 500;
> 
> std::string GetWindowTitle(CGWindowID id) {
>   CFArrayRef window_id_array =
>-      CFArrayCreate(NULL, reinterpret_cast<const void **>(&id), 1, NULL);
>+      CFArrayCreate(NULL, reinterpret_cast<const void**>(&id), 1, NULL);
>   CFArrayRef window_array =
>       CGWindowListCreateDescriptionFromArray(window_id_array);
>   std::string title;
>@@ -35,7 +35,7 @@ std::string GetWindowTitle(CGWindowID id) {
>   if (window_array && CFArrayGetCount(window_array)) {
>     CFDictionaryRef window = reinterpret_cast<CFDictionaryRef>(
>         CFArrayGetValueAtIndex(window_array, 0));
>-    CFStringRef title_ref =  reinterpret_cast<CFStringRef>(
>+    CFStringRef title_ref = reinterpret_cast<CFStringRef>(
>         CFDictionaryGetValue(window, kCGWindowName));
> 
>     if (title_ref)
>@@ -49,7 +49,7 @@ std::string GetWindowTitle(CGWindowID id) {
> 
> int GetWindowOwnerPid(CGWindowID id) {
>   CFArrayRef window_id_array =
>-      CFArrayCreate(NULL, reinterpret_cast<const void **>(&id), 1, NULL);
>+      CFArrayCreate(NULL, reinterpret_cast<const void**>(&id), 1, NULL);
>   CFArrayRef window_array =
>       CGWindowListCreateDescriptionFromArray(window_id_array);
>   int pid = 0;
>@@ -57,7 +57,7 @@ int GetWindowOwnerPid(CGWindowID id) {
>   if (window_array && CFArrayGetCount(window_array)) {
>     CFDictionaryRef window = reinterpret_cast<CFDictionaryRef>(
>         CFArrayGetValueAtIndex(window_array, 0));
>-    CFNumberRef pid_ref =  reinterpret_cast<CFNumberRef>(
>+    CFNumberRef pid_ref = reinterpret_cast<CFNumberRef>(
>         CFDictionaryGetValue(window, kCGWindowOwnerPID));
> 
>     if (pid_ref)
>@@ -97,7 +97,7 @@ CGWindowID FindFullScreenWindowWithSamePidAndTitle(CGWindowID id) {
>         CFDictionaryGetValue(window, kCGWindowName));
>     CFNumberRef window_id_ref = reinterpret_cast<CFNumberRef>(
>         CFDictionaryGetValue(window, kCGWindowNumber));
>-    CFNumberRef window_pid_ref =  reinterpret_cast<CFNumberRef>(
>+    CFNumberRef window_pid_ref = reinterpret_cast<CFNumberRef>(
>         CFDictionaryGetValue(window, kCGWindowOwnerPID));
> 
>     if (!window_title_ref || !window_id_ref || !window_pid_ref)
>@@ -170,8 +170,8 @@ CGWindowID FullScreenChromeWindowDetector::FindFullScreenWindow(
> void FullScreenChromeWindowDetector::UpdateWindowListIfNeeded(
>     CGWindowID original_window) {
>   if (IsChromeWindow(original_window) &&
>-      (rtc::TimeNanos() - last_update_time_ns_) / rtc::kNumNanosecsPerMillisec
>-          > kUpdateIntervalMs) {
>+      (rtc::TimeNanos() - last_update_time_ns_) / rtc::kNumNanosecsPerMillisec >
>+          kUpdateIntervalMs) {
>     previous_window_list_.clear();
>     previous_window_list_.swap(current_window_list_);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/scoped_pixel_buffer_object.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/scoped_pixel_buffer_object.cc
>deleted file mode 100644
>index 7ba12d757fa..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/scoped_pixel_buffer_object.cc
>+++ /dev/null
>@@ -1,55 +0,0 @@
>-/*
>- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "modules/desktop_capture/mac/scoped_pixel_buffer_object.h"
>-
>-#include <stddef.h>
>-
>-namespace webrtc {
>-
>-ScopedPixelBufferObject::ScopedPixelBufferObject()
>-    : cgl_context_(NULL),
>-      pixel_buffer_object_(0) {
>-}
>-
>-ScopedPixelBufferObject::~ScopedPixelBufferObject() {
>-  Release();
>-}
>-
>-bool ScopedPixelBufferObject::Init(CGLContextObj cgl_context,
>-                                      int size_in_bytes) {
>-  cgl_context_ = cgl_context;
>-  CGLContextObj CGL_MACRO_CONTEXT = cgl_context_;
>-  glGenBuffersARB(1, &pixel_buffer_object_);
>-  if (glGetError() == GL_NO_ERROR) {
>-    glBindBufferARB(GL_PIXEL_PACK_BUFFER_ARB, pixel_buffer_object_);
>-    glBufferDataARB(GL_PIXEL_PACK_BUFFER_ARB, size_in_bytes, NULL,
>-                    GL_STREAM_READ_ARB);
>-    glBindBufferARB(GL_PIXEL_PACK_BUFFER_ARB, 0);
>-    if (glGetError() != GL_NO_ERROR) {
>-      Release();
>-    }
>-  } else {
>-    cgl_context_ = NULL;
>-    pixel_buffer_object_ = 0;
>-  }
>-  return pixel_buffer_object_ != 0;
>-}
>-
>-void ScopedPixelBufferObject::Release() {
>-  if (pixel_buffer_object_) {
>-    CGLContextObj CGL_MACRO_CONTEXT = cgl_context_;
>-    glDeleteBuffersARB(1, &pixel_buffer_object_);
>-    cgl_context_ = NULL;
>-    pixel_buffer_object_ = 0;
>-  }
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/scoped_pixel_buffer_object.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/scoped_pixel_buffer_object.h
>deleted file mode 100644
>index 1790e873d75..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/scoped_pixel_buffer_object.h
>+++ /dev/null
>@@ -1,41 +0,0 @@
>-/*
>- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MODULES_DESKTOP_CAPTURE_SCOPED_PIXEL_BUFFER_OBJECT_H_
>-#define MODULES_DESKTOP_CAPTURE_SCOPED_PIXEL_BUFFER_OBJECT_H_
>-
>-#include <OpenGL/CGLMacro.h>
>-#include <OpenGL/OpenGL.h>
>-
>-#include "rtc_base/constructormagic.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>-
>-namespace webrtc {
>-
>-class ScopedPixelBufferObject {
>- public:
>-  ScopedPixelBufferObject();
>-  ~ScopedPixelBufferObject();
>-
>-  bool Init(CGLContextObj cgl_context, int size_in_bytes);
>-  void Release();
>-
>-  GLuint get() const { return pixel_buffer_object_; }
>-
>- private:
>-  CGLContextObj cgl_context_;
>-  GLuint pixel_buffer_object_;
>-
>-  RTC_DISALLOW_COPY_AND_ASSIGN(ScopedPixelBufferObject);
>-};
>-
>-}  // namespace webrtc
>-
>-#endif // MODULES_DESKTOP_CAPTURE_SCOPED_PIXEL_BUFFER_OBJECT_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/screen_capturer_mac.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/screen_capturer_mac.h
>new file mode 100644
>index 00000000000..8a406700eea
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/screen_capturer_mac.h
>@@ -0,0 +1,116 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_DESKTOP_CAPTURE_MAC_SCREEN_CAPTURER_MAC_H_
>+#define MODULES_DESKTOP_CAPTURE_MAC_SCREEN_CAPTURER_MAC_H_
>+
>+#include <CoreGraphics/CoreGraphics.h>
>+
>+#include <memory>
>+
>+#include "modules/desktop_capture/desktop_capture_options.h"
>+#include "modules/desktop_capture/desktop_capturer.h"
>+#include "modules/desktop_capture/desktop_frame.h"
>+#include "modules/desktop_capture/desktop_geometry.h"
>+#include "modules/desktop_capture/desktop_region.h"
>+#include "modules/desktop_capture/mac/desktop_configuration.h"
>+#include "modules/desktop_capture/mac/desktop_configuration_monitor.h"
>+#include "modules/desktop_capture/mac/desktop_frame_provider.h"
>+#include "modules/desktop_capture/screen_capture_frame_queue.h"
>+#include "modules/desktop_capture/screen_capturer_helper.h"
>+#include "modules/desktop_capture/shared_desktop_frame.h"
>+
>+namespace webrtc {
>+
>+class DisplayStreamManager;
>+
>+// A class to perform video frame capturing for mac.
>+class ScreenCapturerMac final : public DesktopCapturer {
>+ public:
>+  ScreenCapturerMac(
>+      rtc::scoped_refptr<DesktopConfigurationMonitor> desktop_config_monitor,
>+      bool detect_updated_region,
>+      bool allow_iosurface);
>+  ~ScreenCapturerMac() override;
>+
>+  // TODO(julien.isorce): Remove Init() or make it private.
>+  bool Init();
>+
>+  // DesktopCapturer interface.
>+  void Start(Callback* callback) override;
>+  void CaptureFrame() override;
>+  void SetExcludedWindow(WindowId window) override;
>+  bool GetSourceList(SourceList* screens) override;
>+  bool SelectSource(SourceId id) override;
>+
>+ private:
>+  // Returns false if the selected screen is no longer valid.
>+  bool CgBlit(const DesktopFrame& frame, const DesktopRegion& region);
>+
>+  // Called when the screen configuration is changed.
>+  void ScreenConfigurationChanged();
>+
>+  bool RegisterRefreshAndMoveHandlers();
>+  void UnregisterRefreshAndMoveHandlers();
>+
>+  void ScreenRefresh(CGDirectDisplayID display_id,
>+                     CGRectCount count,
>+                     const CGRect* rect_array,
>+                     DesktopVector display_origin,
>+                     IOSurfaceRef io_surface);
>+  void ReleaseBuffers();
>+
>+  std::unique_ptr<DesktopFrame> CreateFrame();
>+
>+  const bool detect_updated_region_;
>+
>+  Callback* callback_ = nullptr;
>+
>+  // Queue of the frames buffers.
>+  ScreenCaptureFrameQueue<SharedDesktopFrame> queue_;
>+
>+  // Current display configuration.
>+  MacDesktopConfiguration desktop_config_;
>+
>+  // Currently selected display, or 0 if the full desktop is selected. On OS X
>+  // 10.6 and before, this is always 0.
>+  CGDirectDisplayID current_display_ = 0;
>+
>+  // The physical pixel bounds of the current screen.
>+  DesktopRect screen_pixel_bounds_;
>+
>+  // The dip to physical pixel scale of the current screen.
>+  float dip_to_pixel_scale_ = 1.0f;
>+
>+  // A thread-safe list of invalid rectangles, and the size of the most
>+  // recently captured screen.
>+  ScreenCapturerHelper helper_;
>+
>+  // Contains an invalid region from the previous capture.
>+  DesktopRegion last_invalid_region_;
>+
>+  // Monitoring display reconfiguration.
>+  rtc::scoped_refptr<DesktopConfigurationMonitor> desktop_config_monitor_;
>+
>+  CGWindowID excluded_window_ = 0;
>+
>+  // A self-owned object that will destroy itself after ScreenCapturerMac and
>+  // all display streams have been destroyed..
>+  DisplayStreamManager* display_stream_manager_;
>+
>+  // Container holding latest state of the snapshot per displays.
>+  DesktopFrameProvider desktop_frame_provider_;
>+
>+  RTC_DISALLOW_COPY_AND_ASSIGN(ScreenCapturerMac);
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_DESKTOP_CAPTURE_MAC_SCREEN_CAPTURER_MAC_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_capturer_mac.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/screen_capturer_mac.mm
>similarity index 56%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_capturer_mac.mm
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/screen_capturer_mac.mm
>index 6f2690f2852..ae6c47cfa37 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_capturer_mac.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/screen_capturer_mac.mm
>@@ -8,39 +8,20 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include <stddef.h>
>-
>-#include <memory>
>-#include <set>
> #include <utility>
> 
>-#include <ApplicationServices/ApplicationServices.h>
>-#include <Cocoa/Cocoa.h>
>-#include <CoreGraphics/CoreGraphics.h>
>-#include <dlfcn.h>
>-
>-#include "modules/desktop_capture/desktop_capture_options.h"
>-#include "modules/desktop_capture/desktop_capturer.h"
>-#include "modules/desktop_capture/desktop_frame.h"
>-#include "modules/desktop_capture/desktop_geometry.h"
>-#include "modules/desktop_capture/desktop_region.h"
>-#include "modules/desktop_capture/mac/desktop_configuration.h"
>-#include "modules/desktop_capture/mac/desktop_configuration_monitor.h"
>-#include "modules/desktop_capture/mac/scoped_pixel_buffer_object.h"
>-#include "modules/desktop_capture/screen_capture_frame_queue.h"
>-#include "modules/desktop_capture/screen_capturer_helper.h"
>-#include "modules/desktop_capture/shared_desktop_frame.h"
>+#include "modules/desktop_capture/mac/screen_capturer_mac.h"
>+
>+#include "modules/desktop_capture/mac/desktop_frame_provider.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/macutils.h"
> #include "rtc_base/timeutils.h"
>+#include "rtc_base/trace_event.h"
> #include "sdk/objc/Framework/Classes/Common/scoped_cftyperef.h"
> 
> namespace webrtc {
> 
>-namespace {
>-
> // CGDisplayStreamRefs need to be destroyed asynchronously after receiving a
> // kCGDisplayStreamFrameStatusStopped callback from CoreGraphics. This may
> // happen after the ScreenCapturerMac has been destroyed. DisplayStreamManager
>@@ -56,12 +37,10 @@ class DisplayStreamManager {
>     CFRelease(it->second.stream);
>     display_stream_wrappers_.erase(it);
> 
>-    if (ready_for_self_destruction_ && display_stream_wrappers_.empty())
>-      delete this;
>+    if (ready_for_self_destruction_ && display_stream_wrappers_.empty()) delete this;
>   }
> 
>-  void SaveStream(int unique_id,
>-                  CGDisplayStreamRef stream) {
>+  void SaveStream(int unique_id, CGDisplayStreamRef stream) {
>     RTC_CHECK(unique_id <= unique_id_generator_);
>     DisplayStreamWrapper wrapper;
>     wrapper.stream = stream;
>@@ -73,10 +52,8 @@ class DisplayStreamManager {
>       DisplayStreamWrapper& wrapper = pair.second;
>       if (wrapper.active) {
>         wrapper.active = false;
>-        CFRunLoopSourceRef source =
>-            CGDisplayStreamGetRunLoopSource(wrapper.stream);
>-        CFRunLoopRemoveSource(CFRunLoopGetCurrent(), source,
>-                              kCFRunLoopCommonModes);
>+        CFRunLoopSourceRef source = CGDisplayStreamGetRunLoopSource(wrapper.stream);
>+        CFRunLoopRemoveSource(CFRunLoopGetCurrent(), source, kCFRunLoopCommonModes);
>         CGDisplayStreamStop(wrapper.stream);
>       }
>     }
>@@ -85,8 +62,7 @@ class DisplayStreamManager {
>   void PrepareForSelfDestruction() {
>     ready_for_self_destruction_ = true;
> 
>-    if (display_stream_wrappers_.empty())
>-      delete this;
>+    if (display_stream_wrappers_.empty()) delete this;
>   }
> 
>   // Once the DisplayStreamManager is ready for destruction, the
>@@ -108,17 +84,14 @@ class DisplayStreamManager {
>   bool ready_for_self_destruction_ = false;
> };
> 
>-// Standard Mac displays have 72dpi, but we report 96dpi for
>-// consistency with Windows and Linux.
>-const int kStandardDPI = 96;
>+namespace {
> 
> // Scales all coordinates of a rect by a specified factor.
> DesktopRect ScaleAndRoundCGRect(const CGRect& rect, float scale) {
>-  return DesktopRect::MakeLTRB(
>-    static_cast<int>(floor(rect.origin.x * scale)),
>-    static_cast<int>(floor(rect.origin.y * scale)),
>-    static_cast<int>(ceil((rect.origin.x + rect.size.width) * scale)),
>-    static_cast<int>(ceil((rect.origin.y + rect.size.height) * scale)));
>+  return DesktopRect::MakeLTRB(static_cast<int>(floor(rect.origin.x * scale)),
>+                               static_cast<int>(floor(rect.origin.y * scale)),
>+                               static_cast<int>(ceil((rect.origin.x + rect.size.width) * scale)),
>+                               static_cast<int>(ceil((rect.origin.y + rect.size.height) * scale)));
> }
> 
> // Copy pixels in the |rect| from |src_place| to |dest_plane|. |rect| should be
>@@ -139,7 +112,7 @@ void CopyRect(const uint8_t* src_plane,
>   // Copy pixels in the rectangle line by line.
>   const int bytes_per_line = bytes_per_pixel * rect.width();
>   const int height = rect.height();
>-  for (int i = 0 ; i < height; ++i) {
>+  for (int i = 0; i < height; ++i) {
>     memcpy(dest_plane, src_plane, bytes_per_line);
>     src_plane += src_plane_stride;
>     dest_plane += dest_plane_stride;
>@@ -150,24 +123,22 @@ void CopyRect(const uint8_t* src_plane,
> // |window_to_exclude|, or NULL if the window is not found or it fails. The
> // caller should release the returned CFArrayRef.
> CFArrayRef CreateWindowListWithExclusion(CGWindowID window_to_exclude) {
>-  if (!window_to_exclude)
>-    return nullptr;
>+  if (!window_to_exclude) return nullptr;
> 
>-  CFArrayRef all_windows = CGWindowListCopyWindowInfo(
>-      kCGWindowListOptionOnScreenOnly, kCGNullWindowID);
>-  if (!all_windows)
>-    return nullptr;
>+  CFArrayRef all_windows =
>+      CGWindowListCopyWindowInfo(kCGWindowListOptionOnScreenOnly, kCGNullWindowID);
>+  if (!all_windows) return nullptr;
> 
>   CFMutableArrayRef returned_array =
>       CFArrayCreateMutable(nullptr, CFArrayGetCount(all_windows), nullptr);
> 
>   bool found = false;
>   for (CFIndex i = 0; i < CFArrayGetCount(all_windows); ++i) {
>-    CFDictionaryRef window = reinterpret_cast<CFDictionaryRef>(
>-        CFArrayGetValueAtIndex(all_windows, i));
>+    CFDictionaryRef window =
>+        reinterpret_cast<CFDictionaryRef>(CFArrayGetValueAtIndex(all_windows, i));
> 
>-    CFNumberRef id_ref = reinterpret_cast<CFNumberRef>(
>-        CFDictionaryGetValue(window, kCGWindowNumber));
>+    CFNumberRef id_ref =
>+        reinterpret_cast<CFNumberRef>(CFDictionaryGetValue(window, kCGWindowNumber));
> 
>     CGWindowID id;
>     CFNumberGetValue(id_ref, kCFNumberIntType, &id);
>@@ -175,7 +146,7 @@ CFArrayRef CreateWindowListWithExclusion(CGWindowID window_to_exclude) {
>       found = true;
>       continue;
>     }
>-    CFArrayAppendValue(returned_array, reinterpret_cast<void *>(id));
>+    CFArrayAppendValue(returned_array, reinterpret_cast<void*>(id));
>   }
>   CFRelease(all_windows);
> 
>@@ -188,8 +159,7 @@ CFArrayRef CreateWindowListWithExclusion(CGWindowID window_to_exclude) {
> 
> // Returns the bounds of |window| in physical pixels, enlarged by a small amount
> // on four edges to take account of the border/shadow effects.
>-DesktopRect GetExcludedWindowPixelBounds(CGWindowID window,
>-                                         float dip_to_pixel_scale) {
>+DesktopRect GetExcludedWindowPixelBounds(CGWindowID window, float dip_to_pixel_scale) {
>   // The amount of pixels to add to the actual window bounds to take into
>   // account of the border/shadow effects.
>   static const int kBorderEffectSize = 20;
>@@ -199,14 +169,13 @@ DesktopRect GetExcludedWindowPixelBounds(CGWindowID window,
> 
>   CFArrayRef window_id_array =
>       CFArrayCreate(nullptr, reinterpret_cast<const void**>(&ids), 1, nullptr);
>-  CFArrayRef window_array =
>-      CGWindowListCreateDescriptionFromArray(window_id_array);
>+  CFArrayRef window_array = CGWindowListCreateDescriptionFromArray(window_id_array);
> 
>   if (CFArrayGetCount(window_array) > 0) {
>-    CFDictionaryRef window = reinterpret_cast<CFDictionaryRef>(
>-        CFArrayGetValueAtIndex(window_array, 0));
>-    CFDictionaryRef bounds_ref = reinterpret_cast<CFDictionaryRef>(
>-        CFDictionaryGetValue(window, kCGWindowBounds));
>+    CFDictionaryRef window =
>+        reinterpret_cast<CFDictionaryRef>(CFArrayGetValueAtIndex(window_array, 0));
>+    CFDictionaryRef bounds_ref =
>+        reinterpret_cast<CFDictionaryRef>(CFDictionaryGetValue(window, kCGWindowBounds));
>     CGRectMakeWithDictionaryRepresentation(bounds_ref, &rect);
>   }
> 
>@@ -223,10 +192,10 @@ DesktopRect GetExcludedWindowPixelBounds(CGWindowID window,
> 
> // Create an image of the given region using the given |window_list|.
> // |pixel_bounds| should be in the primary display's coordinate in physical
>-// pixels. The caller should release the returned CGImageRef and CFDataRef.
>-CGImageRef CreateExcludedWindowRegionImage(const DesktopRect& pixel_bounds,
>-                                           float dip_to_pixel_scale,
>-                                           CFArrayRef window_list) {
>+// pixels.
>+rtc::ScopedCFTypeRef<CGImageRef> CreateExcludedWindowRegionImage(const DesktopRect& pixel_bounds,
>+                                                                 float dip_to_pixel_scale,
>+                                                                 CFArrayRef window_list) {
>   CGRect window_bounds;
>   // The origin is in DIP while the size is in physical pixels. That's what
>   // CGWindowListCreateImageFromArray expects.
>@@ -235,112 +204,22 @@ CGImageRef CreateExcludedWindowRegionImage(const DesktopRect& pixel_bounds,
>   window_bounds.size.width = pixel_bounds.width();
>   window_bounds.size.height = pixel_bounds.height();
> 
>-  return CGWindowListCreateImageFromArray(
>-      window_bounds, window_list, kCGWindowImageDefault);
>+  return rtc::ScopedCFTypeRef<CGImageRef>(
>+      CGWindowListCreateImageFromArray(window_bounds, window_list, kCGWindowImageDefault));
> }
> 
>-// A class to perform video frame capturing for mac.
>-class ScreenCapturerMac : public DesktopCapturer {
>- public:
>-  explicit ScreenCapturerMac(
>-      rtc::scoped_refptr<DesktopConfigurationMonitor> desktop_config_monitor,
>-      bool detect_updated_region);
>-  ~ScreenCapturerMac() override;
>-
>-  bool Init();
>-
>-  // DesktopCapturer interface.
>-  void Start(Callback* callback) override;
>-  void CaptureFrame() override;
>-  void SetExcludedWindow(WindowId window) override;
>-  bool GetSourceList(SourceList* screens) override;
>-  bool SelectSource(SourceId id) override;
>-
>- private:
>-  // Returns false if the selected screen is no longer valid.
>-  bool CgBlit(const DesktopFrame& frame, const DesktopRegion& region);
>-
>-  // Called when the screen configuration is changed.
>-  void ScreenConfigurationChanged();
>-
>-  bool RegisterRefreshAndMoveHandlers();
>-  void UnregisterRefreshAndMoveHandlers();
>-
>-  void ScreenRefresh(CGRectCount count,
>-                     const CGRect *rect_array,
>-                     DesktopVector display_origin);
>-  void ReleaseBuffers();
>-
>-  std::unique_ptr<DesktopFrame> CreateFrame();
>-
>-  const bool detect_updated_region_;
>-
>-  Callback* callback_ = nullptr;
>-
>-  ScopedPixelBufferObject pixel_buffer_object_;
>-
>-  // Queue of the frames buffers.
>-  ScreenCaptureFrameQueue<SharedDesktopFrame> queue_;
>-
>-  // Current display configuration.
>-  MacDesktopConfiguration desktop_config_;
>-
>-  // Currently selected display, or 0 if the full desktop is selected. On OS X
>-  // 10.6 and before, this is always 0.
>-  CGDirectDisplayID current_display_ = 0;
>-
>-  // The physical pixel bounds of the current screen.
>-  DesktopRect screen_pixel_bounds_;
>-
>-  // The dip to physical pixel scale of the current screen.
>-  float dip_to_pixel_scale_ = 1.0f;
>-
>-  // A thread-safe list of invalid rectangles, and the size of the most
>-  // recently captured screen.
>-  ScreenCapturerHelper helper_;
>-
>-  // Contains an invalid region from the previous capture.
>-  DesktopRegion last_invalid_region_;
>-
>-  // Monitoring display reconfiguration.
>-  rtc::scoped_refptr<DesktopConfigurationMonitor> desktop_config_monitor_;
>-
>-  CGWindowID excluded_window_ = 0;
>-
>-  // A self-owned object that will destroy itself after ScreenCapturerMac and
>-  // all display streams have been destroyed..
>-  DisplayStreamManager* display_stream_manager_;
>-
>-  RTC_DISALLOW_COPY_AND_ASSIGN(ScreenCapturerMac);
>-};
>-
>-// DesktopFrame wrapper that flips wrapped frame upside down by inverting
>-// stride.
>-class InvertedDesktopFrame : public DesktopFrame {
>- public:
>-  InvertedDesktopFrame(std::unique_ptr<DesktopFrame> frame)
>-      : DesktopFrame(
>-            frame->size(),
>-            -frame->stride(),
>-            frame->data() + (frame->size().height() - 1) * frame->stride(),
>-            frame->shared_memory()) {
>-    original_frame_ = std::move(frame);
>-    MoveFrameInfoFrom(original_frame_.get());
>-  }
>-  ~InvertedDesktopFrame() override {}
>-
>- private:
>-  std::unique_ptr<DesktopFrame> original_frame_;
>-
>-  RTC_DISALLOW_COPY_AND_ASSIGN(InvertedDesktopFrame);
>-};
>+}  // namespace
> 
> ScreenCapturerMac::ScreenCapturerMac(
>     rtc::scoped_refptr<DesktopConfigurationMonitor> desktop_config_monitor,
>-    bool detect_updated_region)
>+    bool detect_updated_region,
>+    bool allow_iosurface)
>     : detect_updated_region_(detect_updated_region),
>-      desktop_config_monitor_(desktop_config_monitor) {
>+      desktop_config_monitor_(desktop_config_monitor),
>+      desktop_frame_provider_(allow_iosurface) {
>   display_stream_manager_ = new DisplayStreamManager;
>+
>+  RTC_LOG(LS_INFO) << "Allow IOSurface: " << allow_iosurface;
> }
> 
> ScreenCapturerMac::~ScreenCapturerMac() {
>@@ -350,13 +229,12 @@ ScreenCapturerMac::~ScreenCapturerMac() {
> }
> 
> bool ScreenCapturerMac::Init() {
>+  TRACE_EVENT0("webrtc", "ScreenCapturerMac::Init");
>+
>   desktop_config_monitor_->Lock();
>   desktop_config_ = desktop_config_monitor_->desktop_configuration();
>   desktop_config_monitor_->Unlock();
>-  if (!RegisterRefreshAndMoveHandlers()) {
>-    return false;
>-  }
>-  ScreenConfigurationChanged();
>+
>   return true;
> }
> 
>@@ -368,28 +246,41 @@ void ScreenCapturerMac::ReleaseBuffers() {
> }
> 
> void ScreenCapturerMac::Start(Callback* callback) {
>-  assert(!callback_);
>-  assert(callback);
>+  RTC_DCHECK(!callback_);
>+  RTC_DCHECK(callback);
>+  TRACE_EVENT_INSTANT1(
>+      "webrtc", "ScreenCapturermac::Start", "target display id ", current_display_);
> 
>   callback_ = callback;
>+  // Start and operate CGDisplayStream handler all from capture thread.
>+  if (!RegisterRefreshAndMoveHandlers()) {
>+    RTC_LOG(LS_ERROR) << "Failed to register refresh and move handlers.";
>+    callback_->OnCaptureResult(Result::ERROR_PERMANENT, nullptr);
>+    return;
>+  }
>+  ScreenConfigurationChanged();
> }
> 
> void ScreenCapturerMac::CaptureFrame() {
>+  TRACE_EVENT0("webrtc", "creenCapturerMac::CaptureFrame");
>   int64_t capture_start_time_nanos = rtc::TimeNanos();
> 
>   queue_.MoveToNextFrame();
>   RTC_DCHECK(!queue_.current_frame() || !queue_.current_frame()->IsShared());
> 
>   desktop_config_monitor_->Lock();
>-  MacDesktopConfiguration new_config =
>-      desktop_config_monitor_->desktop_configuration();
>+  MacDesktopConfiguration new_config = desktop_config_monitor_->desktop_configuration();
>   if (!desktop_config_.Equals(new_config)) {
>     desktop_config_ = new_config;
>     // If the display configuraiton has changed then refresh capturer data
>     // structures. Occasionally, the refresh and move handlers are lost when
>     // the screen mode changes, so re-register them here.
>     UnregisterRefreshAndMoveHandlers();
>-    RegisterRefreshAndMoveHandlers();
>+    if (!RegisterRefreshAndMoveHandlers()) {
>+      RTC_LOG(LS_ERROR) << "Failed to register refresh and move handlers.";
>+      callback_->OnCaptureResult(Result::ERROR_PERMANENT, nullptr);
>+      return;
>+    }
>     ScreenConfigurationChanged();
>   }
> 
>@@ -399,8 +290,7 @@ void ScreenCapturerMac::CaptureFrame() {
>   // If the current buffer is from an older generation then allocate a new one.
>   // Note that we can't reallocate other buffers at this point, since the caller
>   // may still be reading from them.
>-  if (!queue_.current_frame())
>-    queue_.ReplaceCurrentFrame(SharedDesktopFrame::Wrap(CreateFrame()));
>+  if (!queue_.current_frame()) queue_.ReplaceCurrentFrame(SharedDesktopFrame::Wrap(CreateFrame()));
> 
>   DesktopFrame* current_frame = queue_.current_frame();
> 
>@@ -413,16 +303,15 @@ void ScreenCapturerMac::CaptureFrame() {
>   if (detect_updated_region_) {
>     *new_frame->mutable_updated_region() = region;
>   } else {
>-    new_frame->mutable_updated_region()->AddRect(
>-        DesktopRect::MakeSize(new_frame->size()));
>+    new_frame->mutable_updated_region()->AddRect(DesktopRect::MakeSize(new_frame->size()));
>   }
> 
>   if (current_display_) {
>     const MacDisplayConfiguration* config =
>         desktop_config_.FindDisplayConfigurationById(current_display_);
>     if (config) {
>-      new_frame->set_top_left(config->bounds.top_left().subtract(
>-          desktop_config_.bounds.top_left()));
>+      new_frame->set_top_left(
>+          config->bounds.top_left().subtract(desktop_config_.bounds.top_left()));
>     }
>   }
> 
>@@ -442,10 +331,11 @@ void ScreenCapturerMac::SetExcludedWindow(WindowId window) {
> }
> 
> bool ScreenCapturerMac::GetSourceList(SourceList* screens) {
>-  assert(screens->size() == 0);
>+  RTC_DCHECK(screens->size() == 0);
> 
>   for (MacDisplayConfigurations::iterator it = desktop_config_.displays.begin();
>-       it != desktop_config_.displays.end(); ++it) {
>+       it != desktop_config_.displays.end();
>+       ++it) {
>     screens->push_back({it->id, std::string()});
>   }
>   return true;
>@@ -456,10 +346,8 @@ bool ScreenCapturerMac::SelectSource(SourceId id) {
>     current_display_ = 0;
>   } else {
>     const MacDisplayConfiguration* config =
>-        desktop_config_.FindDisplayConfigurationById(
>-            static_cast<CGDirectDisplayID>(id));
>-    if (!config)
>-      return false;
>+        desktop_config_.FindDisplayConfigurationById(static_cast<CGDirectDisplayID>(id));
>+    if (!config) return false;
>     current_display_ = config->id;
>   }
> 
>@@ -472,8 +360,7 @@ bool ScreenCapturerMac::CgBlit(const DesktopFrame& frame, const DesktopRegion& r
>   // TODO(wez): Get rid of this as per crbug.com/145064, or implement
>   // crbug.com/92354.
>   if (queue_.previous_frame()) {
>-    memcpy(frame.data(), queue_.previous_frame()->data(),
>-           frame.stride() * frame.size().height());
>+    memcpy(frame.data(), queue_.previous_frame()->data(), frame.stride() * frame.size().height());
>   }
> 
>   MacDisplayConfigurations displays_to_capture;
>@@ -502,89 +389,68 @@ bool ScreenCapturerMac::CgBlit(const DesktopFrame& frame, const DesktopRegion& r
>     // Capturing mixed-DPI on one surface is hard, so we only return displays
>     // that match the "primary" display's DPI. The primary display is always
>     // the first in the list.
>-    if (i > 0 && display_config.dip_to_pixel_scale !=
>-        displays_to_capture[0].dip_to_pixel_scale) {
>+    if (i > 0 && display_config.dip_to_pixel_scale != displays_to_capture[0].dip_to_pixel_scale) {
>       continue;
>     }
>     // Determine the display's position relative to the desktop, in pixels.
>     DesktopRect display_bounds = display_config.pixel_bounds;
>-    display_bounds.Translate(-screen_pixel_bounds_.left(),
>-                             -screen_pixel_bounds_.top());
>+    display_bounds.Translate(-screen_pixel_bounds_.left(), -screen_pixel_bounds_.top());
> 
>     // Determine which parts of the blit region, if any, lay within the monitor.
>     DesktopRegion copy_region = region;
>     copy_region.IntersectWith(display_bounds);
>-    if (copy_region.is_empty())
>-      continue;
>+    if (copy_region.is_empty()) continue;
> 
>     // Translate the region to be copied into display-relative coordinates.
>     copy_region.Translate(-display_bounds.left(), -display_bounds.top());
> 
>     DesktopRect excluded_window_bounds;
>-    CGImageRef excluded_image = nullptr;
>+    rtc::ScopedCFTypeRef<CGImageRef> excluded_image;
>     if (excluded_window_ && window_list) {
>       // Get the region of the excluded window relative the primary display.
>-      excluded_window_bounds = GetExcludedWindowPixelBounds(
>-          excluded_window_, display_config.dip_to_pixel_scale);
>+      excluded_window_bounds =
>+          GetExcludedWindowPixelBounds(excluded_window_, display_config.dip_to_pixel_scale);
>       excluded_window_bounds.IntersectWith(display_config.pixel_bounds);
> 
>       // Create the image under the excluded window first, because it's faster
>       // than captuing the whole display.
>       if (!excluded_window_bounds.is_empty()) {
>         excluded_image = CreateExcludedWindowRegionImage(
>-            excluded_window_bounds, display_config.dip_to_pixel_scale,
>-            window_list);
>+            excluded_window_bounds, display_config.dip_to_pixel_scale, window_list);
>       }
>     }
> 
>-    // Create an image containing a snapshot of the display.
>-    CGImageRef image = CGDisplayCreateImage(display_config.id);
>-    if (!image) {
>-      if (excluded_image)
>-        CFRelease(excluded_image);
>+    std::unique_ptr<DesktopFrame> frame_source =
>+        desktop_frame_provider_.TakeLatestFrameForDisplay(display_config.id);
>+    if (!frame_source) {
>       continue;
>     }
> 
>-    // Verify that the image has 32-bit depth.
>-    int bits_per_pixel = CGImageGetBitsPerPixel(image);
>-    if (bits_per_pixel / 8 != DesktopFrame::kBytesPerPixel) {
>-      RTC_LOG(LS_ERROR) << "CGDisplayCreateImage() returned imaged with " << bits_per_pixel
>-                        << " bits per pixel. Only 32-bit depth is supported.";
>-      CFRelease(image);
>-      if (excluded_image)
>-        CFRelease(excluded_image);
>-      return false;
>-    }
>-
>-    // Request access to the raw pixel data via the image's DataProvider.
>-    CGDataProviderRef provider = CGImageGetDataProvider(image);
>-    CFDataRef data = CGDataProviderCopyData(provider);
>-    assert(data);
>+    const uint8_t* display_base_address = frame_source->data();
>+    int src_bytes_per_row = frame_source->stride();
>+    RTC_DCHECK(display_base_address);
> 
>-    const uint8_t* display_base_address = CFDataGetBytePtr(data);
>-    int src_bytes_per_row = CGImageGetBytesPerRow(image);
>-
>-    // |image| size may be different from display_bounds in case the screen was
>+    // |frame_source| size may be different from display_bounds in case the screen was
>     // resized recently.
>-    copy_region.IntersectWith(
>-        DesktopRect::MakeWH(CGImageGetWidth(image), CGImageGetHeight(image)));
>+    copy_region.IntersectWith(frame_source->rect());
> 
>     // Copy the dirty region from the display buffer into our desktop buffer.
>     uint8_t* out_ptr = frame.GetFrameDataAtPos(display_bounds.top_left());
>     for (DesktopRegion::Iterator i(copy_region); !i.IsAtEnd(); i.Advance()) {
>-      CopyRect(display_base_address, src_bytes_per_row, out_ptr, frame.stride(),
>-               DesktopFrame::kBytesPerPixel, i.rect());
>+      CopyRect(display_base_address,
>+               src_bytes_per_row,
>+               out_ptr,
>+               frame.stride(),
>+               DesktopFrame::kBytesPerPixel,
>+               i.rect());
>     }
> 
>-    CFRelease(data);
>-    CFRelease(image);
>-
>     if (excluded_image) {
>-      CGDataProviderRef provider = CGImageGetDataProvider(excluded_image);
>-      CFDataRef excluded_image_data = CGDataProviderCopyData(provider);
>-      assert(excluded_image_data);
>-      display_base_address = CFDataGetBytePtr(excluded_image_data);
>-      src_bytes_per_row = CGImageGetBytesPerRow(excluded_image);
>+      CGDataProviderRef provider = CGImageGetDataProvider(excluded_image.get());
>+      rtc::ScopedCFTypeRef<CFDataRef> excluded_image_data(CGDataProviderCopyData(provider));
>+      RTC_DCHECK(excluded_image_data);
>+      display_base_address = CFDataGetBytePtr(excluded_image_data.get());
>+      src_bytes_per_row = CGImageGetBytesPerRow(excluded_image.get());
> 
>       // Translate the bounds relative to the desktop, because |frame| data
>       // starts from the desktop top-left corner.
>@@ -592,25 +458,21 @@ bool ScreenCapturerMac::CgBlit(const DesktopFrame& frame, const DesktopRegion& r
>       window_bounds_relative_to_desktop.Translate(-screen_pixel_bounds_.left(),
>                                                   -screen_pixel_bounds_.top());
> 
>-      DesktopRect rect_to_copy =
>-          DesktopRect::MakeSize(excluded_window_bounds.size());
>-      rect_to_copy.IntersectWith(DesktopRect::MakeWH(
>-          CGImageGetWidth(excluded_image), CGImageGetHeight(excluded_image)));
>-
>-      if (CGImageGetBitsPerPixel(excluded_image) / 8 ==
>-          DesktopFrame::kBytesPerPixel) {
>-        CopyRect(display_base_address, src_bytes_per_row,
>-                 frame.GetFrameDataAtPos(
>-                     window_bounds_relative_to_desktop.top_left()),
>-                 frame.stride(), DesktopFrame::kBytesPerPixel, rect_to_copy);
>+      DesktopRect rect_to_copy = DesktopRect::MakeSize(excluded_window_bounds.size());
>+      rect_to_copy.IntersectWith(DesktopRect::MakeWH(CGImageGetWidth(excluded_image.get()),
>+                                                     CGImageGetHeight(excluded_image.get())));
>+
>+      if (CGImageGetBitsPerPixel(excluded_image.get()) / 8 == DesktopFrame::kBytesPerPixel) {
>+        CopyRect(display_base_address,
>+                 src_bytes_per_row,
>+                 frame.GetFrameDataAtPos(window_bounds_relative_to_desktop.top_left()),
>+                 frame.stride(),
>+                 DesktopFrame::kBytesPerPixel,
>+                 rect_to_copy);
>       }
>-
>-      CFRelease(excluded_image_data);
>-      CFRelease(excluded_image);
>     }
>   }
>-  if (window_list)
>-    CFRelease(window_list);
>+  if (window_list) CFRelease(window_list);
>   return true;
> }
> 
>@@ -643,38 +505,36 @@ bool ScreenCapturerMac::RegisterRefreshAndMoveHandlers() {
>   for (const auto& config : desktop_config_.displays) {
>     size_t pixel_width = config.pixel_bounds.width();
>     size_t pixel_height = config.pixel_bounds.height();
>-    if (pixel_width == 0 || pixel_height == 0)
>-      continue;
>+    if (pixel_width == 0 || pixel_height == 0) continue;
>     // Using a local variable forces the block to capture the raw pointer.
>     DisplayStreamManager* manager = display_stream_manager_;
>     int unique_id = manager->GetUniqueId();
>     CGDirectDisplayID display_id = config.id;
>     DesktopVector display_origin = config.pixel_bounds.top_left();
> 
>-    CGDisplayStreamFrameAvailableHandler handler =
>-        ^(CGDisplayStreamFrameStatus status, uint64_t display_time,
>-          IOSurfaceRef frame_surface, CGDisplayStreamUpdateRef updateRef) {
>-          if (status == kCGDisplayStreamFrameStatusStopped) {
>-            manager->DestroyStream(unique_id);
>-            return;
>-          }
>-
>-          if (manager->ShouldIgnoreUpdates())
>-            return;
>-
>-          // Only pay attention to frame updates.
>-          if (status != kCGDisplayStreamFrameStatusFrameComplete)
>-            return;
>-
>-          size_t count = 0;
>-          const CGRect* rects = CGDisplayStreamUpdateGetRects(
>-              updateRef, kCGDisplayStreamUpdateDirtyRects, &count);
>-          if (count != 0) {
>-            // According to CGDisplayStream.h, it's safe to call
>-            // CGDisplayStreamStop() from within the callback.
>-            ScreenRefresh(count, rects, display_origin);
>-          }
>-        };
>+    CGDisplayStreamFrameAvailableHandler handler = ^(CGDisplayStreamFrameStatus status,
>+                                                     uint64_t display_time,
>+                                                     IOSurfaceRef frame_surface,
>+                                                     CGDisplayStreamUpdateRef updateRef) {
>+      if (status == kCGDisplayStreamFrameStatusStopped) {
>+        manager->DestroyStream(unique_id);
>+        return;
>+      }
>+
>+      if (manager->ShouldIgnoreUpdates()) return;
>+
>+      // Only pay attention to frame updates.
>+      if (status != kCGDisplayStreamFrameStatusFrameComplete) return;
>+
>+      size_t count = 0;
>+      const CGRect* rects =
>+          CGDisplayStreamUpdateGetRects(updateRef, kCGDisplayStreamUpdateDirtyRects, &count);
>+      if (count != 0) {
>+        // According to CGDisplayStream.h, it's safe to call
>+        // CGDisplayStreamStop() from within the callback.
>+        ScreenRefresh(display_id, count, rects, display_origin, frame_surface);
>+      }
>+    };
> 
>     rtc::ScopedCFTypeRef<CFDictionaryRef> properties_dict(
>         CFDictionaryCreate(kCFAllocatorDefault,
>@@ -689,11 +549,9 @@ bool ScreenCapturerMac::RegisterRefreshAndMoveHandlers() {
> 
>     if (display_stream) {
>       CGError error = CGDisplayStreamStart(display_stream);
>-      if (error != kCGErrorSuccess)
>-        return false;
>+      if (error != kCGErrorSuccess) return false;
> 
>-      CFRunLoopSourceRef source =
>-          CGDisplayStreamGetRunLoopSource(display_stream);
>+      CFRunLoopSourceRef source = CGDisplayStreamGetRunLoopSource(display_stream);
>       CFRunLoopAddSource(CFRunLoopGetCurrent(), source, kCFRunLoopCommonModes);
>       display_stream_manager_->SaveStream(unique_id, display_stream);
>     }
>@@ -704,60 +562,48 @@ bool ScreenCapturerMac::RegisterRefreshAndMoveHandlers() {
> 
> void ScreenCapturerMac::UnregisterRefreshAndMoveHandlers() {
>   display_stream_manager_->UnregisterActiveStreams();
>+  // Release obsolete io surfaces.
>+  desktop_frame_provider_.Release();
> }
> 
>-void ScreenCapturerMac::ScreenRefresh(CGRectCount count,
>+void ScreenCapturerMac::ScreenRefresh(CGDirectDisplayID display_id,
>+                                      CGRectCount count,
>                                       const CGRect* rect_array,
>-                                      DesktopVector display_origin) {
>-  if (screen_pixel_bounds_.is_empty())
>-    ScreenConfigurationChanged();
>+                                      DesktopVector display_origin,
>+                                      IOSurfaceRef io_surface) {
>+  if (screen_pixel_bounds_.is_empty()) ScreenConfigurationChanged();
> 
>   // The refresh rects are in display coordinates. We want to translate to
>   // framebuffer coordinates. If a specific display is being captured, then no
>   // change is necessary. If all displays are being captured, then we want to
>   // translate by the origin of the display.
>   DesktopVector translate_vector;
>-  if (!current_display_)
>-    translate_vector = display_origin;
>+  if (!current_display_) translate_vector = display_origin;
> 
>   DesktopRegion region;
>   for (CGRectCount i = 0; i < count; ++i) {
>     // All rects are already in physical pixel coordinates.
>-    DesktopRect rect = DesktopRect::MakeXYWH(
>-        rect_array[i].origin.x, rect_array[i].origin.y,
>-        rect_array[i].size.width, rect_array[i].size.height);
>+    DesktopRect rect = DesktopRect::MakeXYWH(rect_array[i].origin.x,
>+                                             rect_array[i].origin.y,
>+                                             rect_array[i].size.width,
>+                                             rect_array[i].size.height);
> 
>     rect.Translate(translate_vector);
> 
>     region.AddRect(rect);
>   }
>-
>+  // Always having the latest iosurface before invalidating a region.
>+  // See https://bugs.chromium.org/p/webrtc/issues/detail?id=8652 for details.
>+  desktop_frame_provider_.InvalidateIOSurface(
>+      display_id, rtc::ScopedCFTypeRef<IOSurfaceRef>(io_surface, rtc::RetainPolicy::RETAIN));
>   helper_.InvalidateRegion(region);
> }
> 
> std::unique_ptr<DesktopFrame> ScreenCapturerMac::CreateFrame() {
>-  std::unique_ptr<DesktopFrame> frame(
>-      new BasicDesktopFrame(screen_pixel_bounds_.size()));
>-  frame->set_dpi(DesktopVector(kStandardDPI * dip_to_pixel_scale_,
>-                               kStandardDPI * dip_to_pixel_scale_));
>+  std::unique_ptr<DesktopFrame> frame(new BasicDesktopFrame(screen_pixel_bounds_.size()));
>+  frame->set_dpi(
>+      DesktopVector(kStandardDPI * dip_to_pixel_scale_, kStandardDPI * dip_to_pixel_scale_));
>   return frame;
> }
> 
>-}  // namespace
>-
>-// static
>-std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateRawScreenCapturer(
>-    const DesktopCaptureOptions& options) {
>-  if (!options.configuration_monitor())
>-    return nullptr;
>-
>-  std::unique_ptr<ScreenCapturerMac> capturer(new ScreenCapturerMac(
>-      options.configuration_monitor(), options.detect_updated_region()));
>-  if (!capturer.get()->Init()) {
>-    return nullptr;
>-  }
>-
>-  return capturer;
>-}
>-
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/window_list_utils.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/window_list_utils.cc
>index e919fec5fc1..7f89cde1ce0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/window_list_utils.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/window_list_utils.cc
>@@ -13,14 +13,15 @@
> #include <ApplicationServices/ApplicationServices.h>
> 
> #include <algorithm>
>+#include <cmath>
> #include <iterator>
> 
> #include "rtc_base/checks.h"
> #include "rtc_base/macutils.h"
> 
>-static_assert(
>-    static_cast<webrtc::WindowId>(kCGNullWindowID) == webrtc::kNullWindowId,
>-    "kNullWindowId needs to equal to kCGNullWindowID.");
>+static_assert(static_cast<webrtc::WindowId>(kCGNullWindowID) ==
>+                  webrtc::kNullWindowId,
>+              "kNullWindowId needs to equal to kCGNullWindowID.");
> 
> namespace webrtc {
> 
>@@ -60,37 +61,6 @@ bool GetWindowRef(CGWindowID id,
>   return result;
> }
> 
>-// Scales the |rect| according to the DIP to physical pixel scale of |rect|.
>-// |rect| is in unscaled system coordinate, i.e. it's device-independent and the
>-// primary monitor starts from (0, 0). If |rect| overlaps multiple monitors, the
>-// returned size may not be accurate when monitors have different DIP settings.
>-// If |rect| is entirely out of the display, this function returns |rect|.
>-DesktopRect ApplyScaleFactorOfRect(
>-    const MacDesktopConfiguration& desktop_config,
>-    DesktopRect rect) {
>-  // TODO(http://crbug.com/778049): How does Mac OSX decide the scale factor
>-  // if one window is across two monitors with different DPIs.
>-  float scales[] = {
>-      GetScaleFactorAtPosition(desktop_config, rect.top_left()),
>-      GetScaleFactorAtPosition(desktop_config,
>-          DesktopVector(rect.left() + rect.width() / 2,
>-                        rect.top() + rect.height() / 2)),
>-      GetScaleFactorAtPosition(
>-            desktop_config, DesktopVector(rect.right(), rect.bottom())),
>-  };
>-  // Since GetScaleFactorAtPosition() returns 1 if the position is out of the
>-  // display, we always prefer a value which not equals to 1.
>-  float scale = *std::max_element(std::begin(scales), std::end(scales));
>-  if (scale == 1) {
>-    scale = *std::min_element(std::begin(scales), std::end(scales));
>-  }
>-
>-  return DesktopRect::MakeXYWH(rect.left() * scale,
>-                               rect.top() * scale,
>-                               rect.width() * scale,
>-                               rect.height() * scale);
>-}
>-
> }  // namespace
> 
> bool GetWindowList(rtc::FunctionView<bool(CFDictionaryRef)> on_window,
>@@ -99,8 +69,8 @@ bool GetWindowList(rtc::FunctionView<bool(CFDictionaryRef)> on_window,
> 
>   // Only get on screen, non-desktop windows.
>   // According to
>-  // https://developer.apple.com/documentation/coregraphics/cgwindowlistoption/1454105-optiononscreenonly ,
>-  // when kCGWindowListOptionOnScreenOnly is used, the order of windows are in
>+  // https://developer.apple.com/documentation/coregraphics/cgwindowlistoption/1454105-optiononscreenonly
>+  // , when kCGWindowListOptionOnScreenOnly is used, the order of windows are in
>   // decreasing z-order.
>   CFArrayRef window_array = CGWindowListCopyWindowInfo(
>       kCGWindowListOptionOnScreenOnly | kCGWindowListExcludeDesktopElements,
>@@ -173,7 +143,7 @@ bool GetWindowList(DesktopCapturer::SourceList* windows,
>         WindowId id = GetWindowId(window);
>         std::string title = GetWindowTitle(window);
>         if (id != kNullWindowId && !title.empty()) {
>-          windows->push_back(DesktopCapturer::Source{ id, title });
>+          windows->push_back(DesktopCapturer::Source{id, title});
>         }
>         return true;
>       },
>@@ -181,9 +151,8 @@ bool GetWindowList(DesktopCapturer::SourceList* windows,
> }
> 
> // Returns true if the window is occupying a full screen.
>-bool IsWindowFullScreen(
>-    const MacDesktopConfiguration& desktop_config,
>-    CFDictionaryRef window) {
>+bool IsWindowFullScreen(const MacDesktopConfiguration& desktop_config,
>+                        CFDictionaryRef window) {
>   bool fullscreen = false;
>   CFDictionaryRef bounds_ref = reinterpret_cast<CFDictionaryRef>(
>       CFDictionaryGetValue(window, kCGWindowBounds));
>@@ -194,10 +163,9 @@ bool IsWindowFullScreen(
>     for (MacDisplayConfigurations::const_iterator it =
>              desktop_config.displays.begin();
>          it != desktop_config.displays.end(); it++) {
>-      if (it->bounds.equals(DesktopRect::MakeXYWH(bounds.origin.x,
>-                                                  bounds.origin.y,
>-                                                  bounds.size.width,
>-                                                  bounds.size.height))) {
>+      if (it->bounds.equals(
>+              DesktopRect::MakeXYWH(bounds.origin.x, bounds.origin.y,
>+                                    bounds.size.width, bounds.size.height))) {
>         fullscreen = true;
>         break;
>       }
>@@ -215,10 +183,9 @@ bool IsWindowOnScreen(CFDictionaryRef window) {
> 
> bool IsWindowOnScreen(CGWindowID id) {
>   bool on_screen;
>-  if (GetWindowRef(id,
>-                   [&on_screen](CFDictionaryRef window) {
>-                     on_screen = IsWindowOnScreen(window);
>-                   })) {
>+  if (GetWindowRef(id, [&on_screen](CFDictionaryRef window) {
>+        on_screen = IsWindowOnScreen(window);
>+      })) {
>     return on_screen;
>   }
>   return false;
>@@ -265,6 +232,24 @@ float GetScaleFactorAtPosition(const MacDesktopConfiguration& desktop_config,
>   return 1;
> }
> 
>+float GetWindowScaleFactor(CGWindowID id, DesktopSize size) {
>+  DesktopRect window_bounds = GetWindowBounds(id);
>+  float scale = 1.0f;
>+
>+  if (!window_bounds.is_empty() && !size.is_empty()) {
>+    float scale_x = size.width() / window_bounds.width();
>+    float scale_y = size.height() / window_bounds.height();
>+    // Currently the scale in X and Y directions must be same.
>+    if ((std::fabs(scale_x - scale_y) <=
>+         std::numeric_limits<float>::epsilon() * std::max(scale_x, scale_y)) &&
>+        scale_x > 0.0f) {
>+      scale = scale_x;
>+    }
>+  }
>+
>+  return scale;
>+}
>+
> DesktopRect GetWindowBounds(CFDictionaryRef window) {
>   CFDictionaryRef window_bounds = reinterpret_cast<CFDictionaryRef>(
>       CFDictionaryGetValue(window, kCGWindowBounds));
>@@ -277,33 +262,19 @@ DesktopRect GetWindowBounds(CFDictionaryRef window) {
>     return DesktopRect();
>   }
> 
>-  return DesktopRect::MakeXYWH(gc_window_rect.origin.x,
>-                               gc_window_rect.origin.y,
>+  return DesktopRect::MakeXYWH(gc_window_rect.origin.x, gc_window_rect.origin.y,
>                                gc_window_rect.size.width,
>                                gc_window_rect.size.height);
> }
> 
>-DesktopRect GetWindowBounds(const MacDesktopConfiguration& desktop_config,
>-                            CFDictionaryRef window) {
>-  DesktopRect rect = GetWindowBounds(window);
>-  return ApplyScaleFactorOfRect(desktop_config, rect);
>-}
>-
> DesktopRect GetWindowBounds(CGWindowID id) {
>   DesktopRect result;
>-  if (GetWindowRef(id,
>-                   [&result](CFDictionaryRef window) {
>-                     result = GetWindowBounds(window);
>-                   })) {
>+  if (GetWindowRef(id, [&result](CFDictionaryRef window) {
>+        result = GetWindowBounds(window);
>+      })) {
>     return result;
>   }
>   return DesktopRect();
> }
> 
>-DesktopRect GetWindowBounds(const MacDesktopConfiguration& desktop_config,
>-                            CGWindowID id) {
>-  DesktopRect rect = GetWindowBounds(id);
>-  return ApplyScaleFactorOfRect(desktop_config, rect);
>-}
>-
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/window_list_utils.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/window_list_utils.h
>index 8a79f7ecd5e..b54ca3d3466 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/window_list_utils.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mac/window_list_utils.h
>@@ -59,6 +59,15 @@ WindowId GetWindowId(CFDictionaryRef window);
> float GetScaleFactorAtPosition(const MacDesktopConfiguration& desktop_config,
>                                DesktopVector position);
> 
>+// Returns the DIP to physical pixel scale factor of the window with |id|.
>+// The bounds of the window with |id| is in DIP coordinates and |size| is the
>+// CGImage size of the window with |id| in physical coordinates. Comparing them
>+// can give the current scale factor.
>+// If the window overlaps multiple monitors, OS will decide on which monitor the
>+// window is displayed and use its scale factor to the window. So this method
>+// still works.
>+float GetWindowScaleFactor(CGWindowID id, DesktopSize size);
>+
> // Returns the bounds of |window|. If |window| is not a window or the bounds
> // cannot be retrieved, this function returns an empty DesktopRect. The returned
> // DesktopRect is in system coordinate, i.e. the primary monitor always starts
>@@ -67,11 +76,6 @@ float GetScaleFactorAtPosition(const MacDesktopConfiguration& desktop_config,
> // MacDesktopConfiguration.
> DesktopRect GetWindowBounds(CFDictionaryRef window);
> 
>-// Same as GetWindowBounds(CFDictionaryRef), but this function stretches the
>-// result with the scale factor.
>-DesktopRect GetWindowBounds(const MacDesktopConfiguration& desktop_config,
>-                            CFDictionaryRef window);
>-
> // Returns the bounds of window with |id|. If |id| does not represent a window
> // or the bounds cannot be retrieved, this function returns an empty
> // DesktopRect. The returned DesktopRect is in system coordinates.
>@@ -79,11 +83,6 @@ DesktopRect GetWindowBounds(const MacDesktopConfiguration& desktop_config,
> // MacDesktopConfiguration.
> DesktopRect GetWindowBounds(CGWindowID id);
> 
>-// Same as GetWindowBounds(CGWindowID), but this function stretches the result
>-// with the scale factor.
>-DesktopRect GetWindowBounds(const MacDesktopConfiguration& desktop_config,
>-                            CGWindowID id);
>-
> }  // namespace webrtc
> 
> #endif  // MODULES_DESKTOP_CAPTURE_MAC_WINDOW_LIST_UTILS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mouse_cursor.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mouse_cursor.cc
>index bb9f32707b0..3b61e10a8bf 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mouse_cursor.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mouse_cursor.cc
>@@ -19,8 +19,7 @@ namespace webrtc {
> MouseCursor::MouseCursor() {}
> 
> MouseCursor::MouseCursor(DesktopFrame* image, const DesktopVector& hotspot)
>-    : image_(image),
>-      hotspot_(hotspot) {
>+    : image_(image), hotspot_(hotspot) {
>   assert(0 <= hotspot_.x() && hotspot_.x() <= image_->size().width());
>   assert(0 <= hotspot_.y() && hotspot_.y() <= image_->size().height());
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mouse_cursor.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mouse_cursor.h
>index daf6c9695e2..31fe8362321 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mouse_cursor.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mouse_cursor.h
>@@ -34,7 +34,7 @@ class MouseCursor {
>   void set_image(DesktopFrame* image) { image_.reset(image); }
>   const DesktopFrame* image() const { return image_.get(); }
> 
>-  void set_hotspot(const DesktopVector& hotspot ) { hotspot_ = hotspot; }
>+  void set_hotspot(const DesktopVector& hotspot) { hotspot_ = hotspot; }
>   const DesktopVector& hotspot() const { return hotspot_; }
> 
>  private:
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mouse_cursor_monitor.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mouse_cursor_monitor.h
>index 2e98594d63e..6ba90997574 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mouse_cursor_monitor.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mouse_cursor_monitor.h
>@@ -15,7 +15,6 @@
> 
> #include "modules/desktop_capture/desktop_capture_types.h"
> #include "modules/desktop_capture/desktop_geometry.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -59,6 +58,8 @@ class MouseCursorMonitor {
>     // Called in response to Capture(). |position| indicates cursor absolute
>     // position on the system in fullscreen coordinate, i.e. the top-left
>     // monitor always starts from (0, 0).
>+    // The coordinates of the position is controlled by OS, but it's always
>+    // consistent with DesktopFrame.rect().top_left().
>     // TODO(zijiehe): Ensure all implementations return the absolute position.
>     // TODO(zijiehe): Make this function pure virtual after Chromium changes.
>     // TODO(zijiehe): Current this overload works correctly only when capturing
>@@ -108,4 +109,3 @@ class MouseCursorMonitor {
> }  // namespace webrtc
> 
> #endif  // MODULES_DESKTOP_CAPTURE_MOUSE_CURSOR_MONITOR_H_
>-
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mouse_cursor_monitor_mac.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mouse_cursor_monitor_mac.mm
>index b4a80e9c347..de8876438f8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mouse_cursor_monitor_mac.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mouse_cursor_monitor_mac.mm
>@@ -111,8 +111,6 @@ void MouseCursorMonitorMac::Init(Callback* callback, Mode mode) {
> void MouseCursorMonitorMac::Capture() {
>   assert(callback_);
> 
>-  CursorState state = INSIDE;
>-
>   CGEventRef event = CGEventCreate(NULL);
>   CGPoint gc_position = CGEventGetLocation(event);
>   CFRelease(event);
>@@ -130,110 +128,7 @@ void MouseCursorMonitorMac::Capture() {
>   if (mode_ != SHAPE_AND_POSITION)
>     return;
> 
>-  // If we are capturing cursor for a specific window then we need to figure out
>-  // if the current mouse position is covered by another window and also adjust
>-  // |position| to make it relative to the window origin.
>-  if (window_id_ != kCGNullWindowID) {
>-    CGWindowID on_screen_window = window_id_;
>-    if (full_screen_chrome_window_detector_) {
>-      CGWindowID full_screen_window =
>-          full_screen_chrome_window_detector_->FindFullScreenWindow(window_id_);
>-
>-      if (full_screen_window != kCGNullWindowID)
>-        on_screen_window = full_screen_window;
>-    }
>-
>-    // Get list of windows that may be covering parts of |on_screen_window|.
>-    // CGWindowListCopyWindowInfo() returns windows in order from front to back,
>-    // so |on_screen_window| is expected to be the last in the list.
>-    CFArrayRef window_array =
>-        CGWindowListCopyWindowInfo(kCGWindowListOptionOnScreenOnly |
>-                                       kCGWindowListOptionOnScreenAboveWindow |
>-                                       kCGWindowListOptionIncludingWindow,
>-                                   on_screen_window);
>-    bool found_window = false;
>-    if (window_array) {
>-      CFIndex count = CFArrayGetCount(window_array);
>-      for (CFIndex i = 0; i < count; ++i) {
>-        CFDictionaryRef window = reinterpret_cast<CFDictionaryRef>(
>-            CFArrayGetValueAtIndex(window_array, i));
>-
>-        // Skip the Dock window. Dock window covers the whole screen, but it is
>-        // transparent.
>-        CFStringRef window_name = reinterpret_cast<CFStringRef>(
>-            CFDictionaryGetValue(window, kCGWindowName));
>-        if (window_name && CFStringCompare(window_name, CFSTR("Dock"), 0) == 0)
>-          continue;
>-
>-        CFDictionaryRef window_bounds = reinterpret_cast<CFDictionaryRef>(
>-            CFDictionaryGetValue(window, kCGWindowBounds));
>-        CFNumberRef window_number = reinterpret_cast<CFNumberRef>(
>-            CFDictionaryGetValue(window, kCGWindowNumber));
>-
>-        if (window_bounds && window_number) {
>-          CGRect gc_window_rect;
>-          if (!CGRectMakeWithDictionaryRepresentation(window_bounds,
>-                                                      &gc_window_rect)) {
>-            continue;
>-          }
>-          DesktopRect window_rect =
>-              DesktopRect::MakeXYWH(gc_window_rect.origin.x,
>-                                    gc_window_rect.origin.y,
>-                                    gc_window_rect.size.width,
>-                                    gc_window_rect.size.height);
>-
>-          CGWindowID window_id;
>-          if (!CFNumberGetValue(window_number, kCFNumberIntType, &window_id))
>-            continue;
>-
>-          if (window_id == on_screen_window) {
>-            found_window = true;
>-            if (!window_rect.Contains(position))
>-              state = OUTSIDE;
>-            position = position.subtract(window_rect.top_left());
>-
>-            assert(i == count - 1);
>-            break;
>-          } else if (window_rect.Contains(position)) {
>-            state = OUTSIDE;
>-            position.set(-1, -1);
>-            break;
>-          }
>-        }
>-      }
>-      CFRelease(window_array);
>-    }
>-    if (!found_window) {
>-      // If we failed to get list of windows or the window wasn't in the list
>-      // pretend that the cursor is outside the window. This can happen, e.g. if
>-      // the window was closed.
>-      state = OUTSIDE;
>-      position.set(-1, -1);
>-    }
>-  } else {
>-    assert(screen_id_ >= kFullDesktopScreenId);
>-    if (screen_id_ != kFullDesktopScreenId) {
>-      // For single screen capturing, convert the position to relative to the
>-      // target screen.
>-      const MacDisplayConfiguration* config =
>-          configuration.FindDisplayConfigurationById(
>-              static_cast<CGDirectDisplayID>(screen_id_));
>-      if (config) {
>-        if (!config->pixel_bounds.Contains(position))
>-          state = OUTSIDE;
>-        position = position.subtract(config->bounds.top_left());
>-      } else {
>-        // The target screen is no longer valid.
>-        state = OUTSIDE;
>-        position.set(-1, -1);
>-      }
>-    }
>-  }
>-  // Convert Density Independent Pixel to physical pixel.
>-  position = DesktopVector(round(position.x() * scale),
>-                           round(position.y() * scale));
>-  // TODO(zijiehe): Remove this overload.
>-  callback_->OnMouseCursorPosition(state, position);
>+  // Always report cursor position in DIP pixel.
>   callback_->OnMouseCursorPosition(
>       position.subtract(configuration.bounds.top_left()));
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mouse_cursor_monitor_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mouse_cursor_monitor_unittest.cc
>index 9af0d68efcc..ea7ba606164 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mouse_cursor_monitor_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mouse_cursor_monitor_unittest.cc
>@@ -10,11 +10,11 @@
> 
> #include <memory>
> 
>-#include "modules/desktop_capture/mouse_cursor_monitor.h"
>-#include "modules/desktop_capture/desktop_capturer.h"
> #include "modules/desktop_capture/desktop_capture_options.h"
>+#include "modules/desktop_capture/desktop_capturer.h"
> #include "modules/desktop_capture/desktop_frame.h"
> #include "modules/desktop_capture/mouse_cursor.h"
>+#include "modules/desktop_capture/mouse_cursor_monitor.h"
> #include "test/gtest.h"
> 
> namespace webrtc {
>@@ -22,9 +22,7 @@ namespace webrtc {
> class MouseCursorMonitorTest : public testing::Test,
>                                public MouseCursorMonitor::Callback {
>  public:
>-  MouseCursorMonitorTest()
>-      : position_received_(false) {
>-  }
>+  MouseCursorMonitorTest() : position_received_(false) {}
> 
>   // MouseCursorMonitor::Callback interface
>   void OnMouseCursor(MouseCursor* cursor_image) override {
>@@ -33,14 +31,16 @@ class MouseCursorMonitorTest : public testing::Test,
> 
>   void OnMouseCursorPosition(MouseCursorMonitor::CursorState state,
>                              const DesktopVector& position) override {
>-    state_ = state;
>+    RTC_NOTREACHED();
>+  }
>+
>+  void OnMouseCursorPosition(const DesktopVector& position) override {
>     position_ = position;
>     position_received_ = true;
>   }
> 
>  protected:
>   std::unique_ptr<MouseCursor> cursor_image_;
>-  MouseCursorMonitor::CursorState state_;
>   DesktopVector position_;
>   bool position_received_;
> };
>@@ -78,7 +78,6 @@ TEST_F(MouseCursorMonitorTest, MAYBE(FromScreen)) {
>             cursor_image_->image()->size().height());
> 
>   EXPECT_TRUE(position_received_);
>-  EXPECT_EQ(MouseCursorMonitor::INSIDE, state_);
> }
> 
> TEST_F(MouseCursorMonitorTest, MAYBE(FromWindow)) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mouse_cursor_monitor_win.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mouse_cursor_monitor_win.cc
>index a50743cc6e6..bc03b958683 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mouse_cursor_monitor_win.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mouse_cursor_monitor_win.cc
>@@ -31,8 +31,7 @@ namespace {
> bool IsSameCursorShape(const CURSORINFO& left, const CURSORINFO& right) {
>   // If the cursors are not showing, we do not care the hCursor handle.
>   return left.flags == right.flags &&
>-         (left.flags != CURSOR_SHOWING ||
>-          left.hCursor == right.hCursor);
>+         (left.flags != CURSOR_SHOWING || left.hCursor == right.hCursor);
> }
> 
> }  // namespace
>@@ -152,8 +151,9 @@ void MouseCursorMonitorWin::Capture() {
>     } else {
>       if (inside) {
>         HWND windowUnderCursor = WindowFromPoint(cursor_info.ptScreenPos);
>-        inside = windowUnderCursor ?
>-            (window_ == GetAncestor(windowUnderCursor, GA_ROOT)) : false;
>+        inside = windowUnderCursor
>+                     ? (window_ == GetAncestor(windowUnderCursor, GA_ROOT))
>+                     : false;
>       }
>       position = position.subtract(cropped_rect.top_left());
>     }
>@@ -165,19 +165,16 @@ void MouseCursorMonitorWin::Capture() {
>     position = position.subtract(rect.top_left());
>   }
> 
>-  // TODO(zijiehe): Remove this overload.
>-  callback_->OnMouseCursorPosition(inside ? INSIDE : OUTSIDE, position);
>   callback_->OnMouseCursorPosition(position);
> }
> 
> DesktopRect MouseCursorMonitorWin::GetScreenRect() {
>   assert(screen_ != kInvalidScreenId);
>   if (screen_ == kFullDesktopScreenId) {
>-    return DesktopRect::MakeXYWH(
>-        GetSystemMetrics(SM_XVIRTUALSCREEN),
>-        GetSystemMetrics(SM_YVIRTUALSCREEN),
>-        GetSystemMetrics(SM_CXVIRTUALSCREEN),
>-        GetSystemMetrics(SM_CYVIRTUALSCREEN));
>+    return DesktopRect::MakeXYWH(GetSystemMetrics(SM_XVIRTUALSCREEN),
>+                                 GetSystemMetrics(SM_YVIRTUALSCREEN),
>+                                 GetSystemMetrics(SM_CXVIRTUALSCREEN),
>+                                 GetSystemMetrics(SM_CYVIRTUALSCREEN));
>   }
>   DISPLAY_DEVICE device;
>   device.cb = sizeof(device);
>@@ -188,19 +185,19 @@ DesktopRect MouseCursorMonitorWin::GetScreenRect() {
>   DEVMODE device_mode;
>   device_mode.dmSize = sizeof(device_mode);
>   device_mode.dmDriverExtra = 0;
>-  result = EnumDisplaySettingsEx(
>-      device.DeviceName, ENUM_CURRENT_SETTINGS, &device_mode, 0);
>+  result = EnumDisplaySettingsEx(device.DeviceName, ENUM_CURRENT_SETTINGS,
>+                                 &device_mode, 0);
>   if (!result)
>     return DesktopRect();
> 
>-  return DesktopRect::MakeXYWH(device_mode.dmPosition.x,
>-                               device_mode.dmPosition.y,
>-                               device_mode.dmPelsWidth,
>-                               device_mode.dmPelsHeight);
>+  return DesktopRect::MakeXYWH(
>+      device_mode.dmPosition.x, device_mode.dmPosition.y,
>+      device_mode.dmPelsWidth, device_mode.dmPelsHeight);
> }
> 
> MouseCursorMonitor* MouseCursorMonitor::CreateForWindow(
>-    const DesktopCaptureOptions& options, WindowId window) {
>+    const DesktopCaptureOptions& options,
>+    WindowId window) {
>   return new MouseCursorMonitorWin(reinterpret_cast<HWND>(window));
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mouse_cursor_monitor_x11.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mouse_cursor_monitor_x11.cc
>index aeb864e2bdb..41749bd2841 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mouse_cursor_monitor_x11.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/mouse_cursor_monitor_x11.cc
>@@ -12,9 +12,9 @@
> 
> #include "modules/desktop_capture/mouse_cursor_monitor.h"
> 
>-#include <X11/extensions/Xfixes.h>
> #include <X11/Xlib.h>
> #include <X11/Xutil.h>
>+#include <X11/extensions/Xfixes.h>
> 
> #include "modules/desktop_capture/desktop_capture_options.h"
> #include "modules/desktop_capture/desktop_capture_types.h"
>@@ -35,7 +35,7 @@ Window GetTopLevelWindow(Display* display, Window window) {
>   while (true) {
>     // If the window is in WithdrawnState then look at all of its children.
>     ::Window root, parent;
>-    ::Window *children;
>+    ::Window* children;
>     unsigned int num_children;
>     if (!XQueryTree(display, window, &root, &parent, &children,
>                     &num_children)) {
>@@ -104,12 +104,9 @@ MouseCursorMonitorX11::MouseCursorMonitorX11(
>   std::unique_ptr<DesktopFrame> default_cursor(
>       new BasicDesktopFrame(DesktopSize(kSize, kSize)));
>   const uint8_t pixels[kSize * kSize] = {
>-    0x00, 0x00, 0x00, 0x00, 0x00,
>-    0x00, 0xff, 0xff, 0xff, 0x00,
>-    0x00, 0xff, 0xff, 0xff, 0x00,
>-    0x00, 0xff, 0xff, 0xff, 0x00,
>-    0x00, 0x00, 0x00, 0x00, 0x00
>-  };
>+      0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0xff, 0xff, 0xff,
>+      0x00, 0x00, 0xff, 0xff, 0xff, 0x00, 0x00, 0xff, 0xff,
>+      0xff, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00};
>   uint8_t* ptr = default_cursor->data();
>   for (int y = 0; y < kSize; ++y) {
>     for (int x = 0; x < kSize; ++x) {
>@@ -201,8 +198,6 @@ void MouseCursorMonitorX11::Capture() {
>       }
>     }
> 
>-    // TODO(zijiehe): Remove this overload.
>-    callback_->OnMouseCursorPosition(state, DesktopVector(win_x, win_y));
>     // X11 always starts the coordinate from (0, 0), so we do not need to
>     // translate here.
>     callback_->OnMouseCursorPosition(DesktopVector(root_x, root_y));
>@@ -230,11 +225,11 @@ void MouseCursorMonitorX11::CaptureCursor() {
>     XErrorTrap error_trap(display());
>     img = XFixesGetCursorImage(display());
>     if (!img || error_trap.GetLastErrorAndDisable() != 0)
>-       return;
>-   }
>+      return;
>+  }
> 
>-   std::unique_ptr<DesktopFrame> image(
>-       new BasicDesktopFrame(DesktopSize(img->width, img->height)));
>+  std::unique_ptr<DesktopFrame> image(
>+      new BasicDesktopFrame(DesktopSize(img->width, img->height)));
> 
>   // Xlib stores 32-bit data in longs, even if longs are 64-bits long.
>   unsigned long* src = img->pixels;
>@@ -254,7 +249,8 @@ void MouseCursorMonitorX11::CaptureCursor() {
> 
> // static
> MouseCursorMonitor* MouseCursorMonitor::CreateForWindow(
>-    const DesktopCaptureOptions& options, WindowId window) {
>+    const DesktopCaptureOptions& options,
>+    WindowId window) {
>   if (!options.x_display())
>     return NULL;
>   window = GetTopLevelWindow(options.x_display()->display(), window);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/rgba_color.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/rgba_color.cc
>index 08324e37507..4bef3cdf9ac 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/rgba_color.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/rgba_color.cc
>@@ -9,8 +9,7 @@
>  */
> 
> #include "modules/desktop_capture/rgba_color.h"
>-
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include "rtc_base/system/arch.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_capture_frame_queue.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_capture_frame_queue.h
>index a9a1643058e..a5bb595150f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_capture_frame_queue.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_capture_frame_queue.h
>@@ -16,10 +16,9 @@
> #include "rtc_base/constructormagic.h"
> // TODO(zijiehe): These headers are not used in this file, but to avoid build
> // break in remoting/host. We should add headers in each individual files.
>-#include "modules/desktop_capture/desktop_frame.h"  // Remove
>+#include "modules/desktop_capture/desktop_frame.h"         // Remove
> #include "modules/desktop_capture/shared_desktop_frame.h"  // Remove
> 
>-
> namespace webrtc {
> 
> // Represents a queue of reusable video frames. Provides access to the 'current'
>@@ -43,9 +42,7 @@ class ScreenCaptureFrameQueue {
> 
>   // Moves to the next frame in the queue, moving the 'current' frame to become
>   // the 'previous' one.
>-  void MoveToNextFrame() {
>-    current_ = (current_ + 1) % kQueueLength;
>-  }
>+  void MoveToNextFrame() { current_ = (current_ + 1) % kQueueLength; }
> 
>   // Replaces the current frame with a new one allocated by the caller. The
>   // existing frame (if any) is destroyed. Takes ownership of |frame|.
>@@ -62,9 +59,7 @@ class ScreenCaptureFrameQueue {
>     current_ = 0;
>   }
> 
>-  FrameType* current_frame() const {
>-    return frames_[current_].get();
>-  }
>+  FrameType* current_frame() const { return frames_[current_].get(); }
> 
>   FrameType* previous_frame() const {
>     return frames_[(current_ + kQueueLength - 1) % kQueueLength].get();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_capturer_darwin.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_capturer_darwin.mm
>new file mode 100644
>index 00000000000..d5a7bb0522b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_capturer_darwin.mm
>@@ -0,0 +1,33 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <memory>
>+
>+#include "modules/desktop_capture/mac/screen_capturer_mac.h"
>+
>+namespace webrtc {
>+
>+// static
>+std::unique_ptr<DesktopCapturer> DesktopCapturer::CreateRawScreenCapturer(
>+    const DesktopCaptureOptions& options) {
>+  if (!options.configuration_monitor()) {
>+    return nullptr;
>+  }
>+
>+  std::unique_ptr<ScreenCapturerMac> capturer(new ScreenCapturerMac(
>+      options.configuration_monitor(), options.detect_updated_region(), options.allow_iosurface()));
>+  if (!capturer.get()->Init()) {
>+    return nullptr;
>+  }
>+
>+  return capturer;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_capturer_helper.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_capturer_helper.cc
>index 4e4635eaed2..2e4fef86be6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_capturer_helper.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_capturer_helper.cc
>@@ -16,12 +16,9 @@
> namespace webrtc {
> 
> ScreenCapturerHelper::ScreenCapturerHelper()
>-    : invalid_region_lock_(RWLockWrapper::CreateRWLock()),
>-      log_grid_size_(0) {
>-}
>+    : invalid_region_lock_(RWLockWrapper::CreateRWLock()), log_grid_size_(0) {}
> 
>-ScreenCapturerHelper::~ScreenCapturerHelper() {
>-}
>+ScreenCapturerHelper::~ScreenCapturerHelper() {}
> 
> void ScreenCapturerHelper::ClearInvalidRegion() {
>   WriteLockScoped scoped_invalid_region_lock(*invalid_region_lock_);
>@@ -39,8 +36,7 @@ void ScreenCapturerHelper::InvalidateScreen(const DesktopSize& size) {
>   invalid_region_.AddRect(DesktopRect::MakeSize(size));
> }
> 
>-void ScreenCapturerHelper::TakeInvalidRegion(
>-    DesktopRegion* invalid_region) {
>+void ScreenCapturerHelper::TakeInvalidRegion(DesktopRegion* invalid_region) {
>   invalid_region->Clear();
> 
>   {
>@@ -65,8 +61,7 @@ const DesktopSize& ScreenCapturerHelper::size_most_recent() const {
>   return size_most_recent_;
> }
> 
>-void ScreenCapturerHelper::set_size_most_recent(
>-    const DesktopSize& size) {
>+void ScreenCapturerHelper::set_size_most_recent(const DesktopSize& size) {
>   size_most_recent_ = size;
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_capturer_helper.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_capturer_helper.h
>index d36be49b60b..d8fc36837b8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_capturer_helper.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_capturer_helper.h
>@@ -16,7 +16,7 @@
> #include "modules/desktop_capture/desktop_geometry.h"
> #include "modules/desktop_capture/desktop_region.h"
> #include "rtc_base/constructormagic.h"
>-#include "system_wrappers/include/rw_lock_wrapper.h"
>+#include "rtc_base/synchronization/rw_lock_wrapper.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_capturer_helper_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_capturer_helper_unittest.cc
>index f2944abf673..1ce4959e76b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_capturer_helper_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_capturer_helper_unittest.cc
>@@ -55,8 +55,7 @@ TEST_F(ScreenCapturerHelperTest, InvalidateScreen) {
> TEST_F(ScreenCapturerHelperTest, SizeMostRecent) {
>   EXPECT_TRUE(capturer_helper_.size_most_recent().is_empty());
>   capturer_helper_.set_size_most_recent(DesktopSize(12, 34));
>-  EXPECT_TRUE(
>-      DesktopSize(12, 34).equals(capturer_helper_.size_most_recent()));
>+  EXPECT_TRUE(DesktopSize(12, 34).equals(capturer_helper_.size_most_recent()));
> }
> 
> TEST_F(ScreenCapturerHelperTest, SetLogGridSize) {
>@@ -103,7 +102,8 @@ TEST_F(ScreenCapturerHelperTest, SetLogGridSize) {
>   EXPECT_TRUE(DesktopRegion(DesktopRect::MakeXYWH(7, 7, 1, 1)).Equals(region));
> }
> 
>-void TestExpandRegionToGrid(const DesktopRegion& region, int log_grid_size,
>+void TestExpandRegionToGrid(const DesktopRegion& region,
>+                            int log_grid_size,
>                             const DesktopRegion& expanded_region_expected) {
>   DesktopRegion expanded_region1;
>   ScreenCapturerHelper::ExpandToGrid(region, log_grid_size, &expanded_region1);
>@@ -115,9 +115,15 @@ void TestExpandRegionToGrid(const DesktopRegion& region, int log_grid_size,
>   EXPECT_TRUE(expanded_region1.Equals(expanded_region2));
> }
> 
>-void TestExpandRectToGrid(int l, int t, int r, int b, int log_grid_size,
>-                          int lExpanded, int tExpanded,
>-                          int rExpanded, int bExpanded) {
>+void TestExpandRectToGrid(int l,
>+                          int t,
>+                          int r,
>+                          int b,
>+                          int log_grid_size,
>+                          int lExpanded,
>+                          int tExpanded,
>+                          int rExpanded,
>+                          int bExpanded) {
>   TestExpandRegionToGrid(DesktopRegion(DesktopRect::MakeLTRB(l, t, r, b)),
>                          log_grid_size,
>                          DesktopRegion(DesktopRect::MakeLTRB(
>@@ -131,23 +137,23 @@ TEST_F(ScreenCapturerHelperTest, ExpandToGrid) {
>     int x = i * kGridSize;
>     for (int j = -2; j <= 2; j++) {
>       int y = j * kGridSize;
>-      TestExpandRectToGrid(x + 0, y + 0, x + 1, y + 1, kLogGridSize,
>-                           x + 0, y + 0, x + kGridSize, y + kGridSize);
>+      TestExpandRectToGrid(x + 0, y + 0, x + 1, y + 1, kLogGridSize, x + 0,
>+                           y + 0, x + kGridSize, y + kGridSize);
>       TestExpandRectToGrid(x + 0, y + kGridSize - 1, x + 1, y + kGridSize,
>-                           kLogGridSize,
>-                           x + 0, y + 0, x + kGridSize, y + kGridSize);
>-      TestExpandRectToGrid(x + kGridSize - 1, y + kGridSize - 1,
>-                           x + kGridSize, y + kGridSize, kLogGridSize,
>-                           x + 0, y + 0, x + kGridSize, y + kGridSize);
>-      TestExpandRectToGrid(x + kGridSize - 1, y + 0,
>-                           x + kGridSize, y + 1, kLogGridSize,
>-                           x + 0, y + 0, x + kGridSize, y + kGridSize);
>+                           kLogGridSize, x + 0, y + 0, x + kGridSize,
>+                           y + kGridSize);
>+      TestExpandRectToGrid(x + kGridSize - 1, y + kGridSize - 1, x + kGridSize,
>+                           y + kGridSize, kLogGridSize, x + 0, y + 0,
>+                           x + kGridSize, y + kGridSize);
>+      TestExpandRectToGrid(x + kGridSize - 1, y + 0, x + kGridSize, y + 1,
>+                           kLogGridSize, x + 0, y + 0, x + kGridSize,
>+                           y + kGridSize);
>       TestExpandRectToGrid(x - 1, y + 0, x + 1, y + 1, kLogGridSize,
>                            x - kGridSize, y + 0, x + kGridSize, y + kGridSize);
>       TestExpandRectToGrid(x - 1, y - 1, x + 1, y + 0, kLogGridSize,
>                            x - kGridSize, y - kGridSize, x + kGridSize, y);
>-      TestExpandRectToGrid(x + 0, y - 1, x + 1, y + 1, kLogGridSize,
>-                           x, y - kGridSize, x + kGridSize, y + kGridSize);
>+      TestExpandRectToGrid(x + 0, y - 1, x + 1, y + 1, kLogGridSize, x,
>+                           y - kGridSize, x + kGridSize, y + kGridSize);
>       TestExpandRectToGrid(x - 1, y - 1, x + 0, y + 1, kLogGridSize,
>                            x - kGridSize, y - kGridSize, x, y + kGridSize);
> 
>@@ -164,18 +170,18 @@ TEST_F(ScreenCapturerHelperTest, ExpandToGrid) {
>         }
>         if (q != 1) {
>           region.AddRect(DesktopRect::MakeXYWH(x, y - 1, 1, 1));
>-          expanded_region_expected.AddRect(DesktopRect::MakeXYWH(
>-              x, y - kGridSize, kGridSize, kGridSize));
>+          expanded_region_expected.AddRect(
>+              DesktopRect::MakeXYWH(x, y - kGridSize, kGridSize, kGridSize));
>         }
>         if (q != 2) {
>           region.AddRect(DesktopRect::MakeXYWH(x - 1, y, 1, 1));
>-          expanded_region_expected.AddRect(DesktopRect::MakeXYWH(
>-              x - kGridSize, y, kGridSize, kGridSize));
>+          expanded_region_expected.AddRect(
>+              DesktopRect::MakeXYWH(x - kGridSize, y, kGridSize, kGridSize));
>         }
>         if (q != 3) {
>           region.AddRect(DesktopRect::MakeXYWH(x, y, 1, 1));
>-          expanded_region_expected.AddRect(DesktopRect::MakeXYWH(
>-              x, y, kGridSize, kGridSize));
>+          expanded_region_expected.AddRect(
>+              DesktopRect::MakeXYWH(x, y, kGridSize, kGridSize));
>         }
> 
>         TestExpandRegionToGrid(region, kLogGridSize, expanded_region_expected);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_capturer_integration_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_capturer_integration_test.cc
>index 0104a560747..3fe38daabe0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_capturer_integration_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_capturer_integration_test.cc
>@@ -24,10 +24,10 @@
> #include "modules/desktop_capture/mock_desktop_capturer_callback.h"
> #include "modules/desktop_capture/rgba_color.h"
> #include "modules/desktop_capture/screen_drawer.h"
>-#include "rtc_base/base64.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/logging.h"
>+#include "rtc_base/third_party/base64/base64.h"
> #include "test/gmock.h"
> #include "test/gtest.h"
> 
>@@ -89,8 +89,8 @@ class ScreenCapturerIntegrationTest : public testing::Test {
>   void TestCaptureUpdatedRegion(
>       std::initializer_list<DesktopCapturer*> capturers) {
>     RTC_DCHECK(capturers.size() > 0);
>-    // A large enough area for the tests, which should be able to be fulfilled
>-    // by most systems.
>+// A large enough area for the tests, which should be able to be fulfilled
>+// by most systems.
> #if defined(WEBRTC_WIN)
>     // On Windows, an interesting warning window may pop up randomly. The root
>     // cause is still under investigation, so reduce the test area to work
>@@ -212,8 +212,8 @@ class ScreenCapturerIntegrationTest : public testing::Test {
>           return;
>         }
> 
>-        if (ArePixelsColoredBy(
>-            *frame, rect, color, drawer->MayDrawIncompleteShapes())) {
>+        if (ArePixelsColoredBy(*frame, rect, color,
>+                               drawer->MayDrawIncompleteShapes())) {
>           capturers[j] = nullptr;
>           succeeded_capturers++;
>         }
>@@ -222,17 +222,15 @@ class ScreenCapturerIntegrationTest : public testing::Test {
>         // been resolved.
>         else if (i == wait_capture_round - 1) {
>           std::string result;
>-          rtc::Base64::EncodeFromArray(frame->data(),
>-                                       frame->size().height() * frame->stride(),
>-                                       &result);
>+          rtc::Base64::EncodeFromArray(
>+              frame->data(), frame->size().height() * frame->stride(), &result);
>           std::cout << frame->size().width() << " x " << frame->size().height()
>                     << std::endl;
>           // Split the entire string (can be over 4M) into several lines to
>           // avoid browser from sticking.
>           static const size_t kLineLength = 32768;
>           const char* result_end = result.c_str() + result.length();
>-          for (const char* it = result.c_str();
>-               it < result_end;
>+          for (const char* it = result.c_str(); it < result_end;
>                it += kLineLength) {
>             const size_t max_length = result_end - it;
>             std::cout << std::string(it, std::min(kLineLength, max_length))
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_capturer_mac_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_capturer_mac_unittest.cc
>index 00d2d02a29f..812f230a068 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_capturer_mac_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_capturer_mac_unittest.cc
>@@ -13,8 +13,8 @@
> #include <memory>
> #include <ostream>
> 
>-#include "modules/desktop_capture/desktop_capturer.h"
> #include "modules/desktop_capture/desktop_capture_options.h"
>+#include "modules/desktop_capture/desktop_capturer.h"
> #include "modules/desktop_capture/desktop_frame.h"
> #include "modules/desktop_capture/desktop_geometry.h"
> #include "modules/desktop_capture/desktop_region.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_capturer_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_capturer_unittest.cc
>index 9ed4f4cb988..90197d61698 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_capturer_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_capturer_unittest.cc
>@@ -73,12 +73,9 @@ class ScreenCapturerTest : public testing::Test {
> class FakeSharedMemory : public SharedMemory {
>  public:
>   FakeSharedMemory(char* buffer, size_t size)
>-    : SharedMemory(buffer, size, 0, kTestSharedMemoryId),
>-      buffer_(buffer) {
>-  }
>-  virtual ~FakeSharedMemory() {
>-    delete[] buffer_;
>-  }
>+      : SharedMemory(buffer, size, 0, kTestSharedMemoryId), buffer_(buffer) {}
>+  ~FakeSharedMemory() override { delete[] buffer_; }
>+
>  private:
>   char* buffer_;
>   RTC_DISALLOW_COPY_AND_ASSIGN(FakeSharedMemory);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_capturer_win.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_capturer_win.cc
>index ecdc48e5b4d..be6d8a5275c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_capturer_win.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_capturer_win.cc
>@@ -12,8 +12,8 @@
> #include <utility>
> 
> #include "modules/desktop_capture/blank_detector_desktop_capturer_wrapper.h"
>-#include "modules/desktop_capture/desktop_capturer.h"
> #include "modules/desktop_capture/desktop_capture_options.h"
>+#include "modules/desktop_capture/desktop_capturer.h"
> #include "modules/desktop_capture/fallback_desktop_capturer_wrapper.h"
> #include "modules/desktop_capture/rgba_color.h"
> #include "modules/desktop_capture/win/screen_capturer_win_directx.h"
>@@ -25,8 +25,7 @@ namespace webrtc {
> namespace {
> 
> std::unique_ptr<DesktopCapturer> CreateScreenCapturerWinDirectx() {
>-  std::unique_ptr<DesktopCapturer> capturer(
>-      new ScreenCapturerWinDirectx());
>+  std::unique_ptr<DesktopCapturer> capturer(new ScreenCapturerWinDirectx());
>   capturer.reset(new BlankDetectorDesktopCapturerWrapper(
>       std::move(capturer), RgbaColor(0, 0, 0, 0)));
>   return capturer;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_capturer_x11.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_capturer_x11.cc
>index 3acd2a8de9f..e95fccc2082 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_capturer_x11.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_capturer_x11.cc
>@@ -14,10 +14,10 @@
> #include <set>
> #include <utility>
> 
>-#include <X11/extensions/Xdamage.h>
>-#include <X11/extensions/Xfixes.h>
> #include <X11/Xlib.h>
> #include <X11/Xutil.h>
>+#include <X11/extensions/Xdamage.h>
>+#include <X11/extensions/Xfixes.h>
> 
> #include "modules/desktop_capture/desktop_capture_options.h"
> #include "modules/desktop_capture/desktop_capturer.h"
>@@ -30,6 +30,7 @@
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/timeutils.h"
>+#include "rtc_base/trace_event.h"
> 
> namespace webrtc {
> namespace {
>@@ -127,13 +128,14 @@ ScreenCapturerLinux::ScreenCapturerLinux() {
> ScreenCapturerLinux::~ScreenCapturerLinux() {
>   options_.x_display()->RemoveEventHandler(ConfigureNotify, this);
>   if (use_damage_) {
>-    options_.x_display()->RemoveEventHandler(
>-        damage_event_base_ + XDamageNotify, this);
>+    options_.x_display()->RemoveEventHandler(damage_event_base_ + XDamageNotify,
>+                                             this);
>   }
>   DeinitXlib();
> }
> 
> bool ScreenCapturerLinux::Init(const DesktopCaptureOptions& options) {
>+  TRACE_EVENT0("webrtc", "ScreenCapturerLinux::Init");
>   options_ = options;
> 
>   root_window_ = RootWindow(display(), DefaultScreen(display()));
>@@ -195,8 +197,8 @@ void ScreenCapturerLinux::InitXDamage() {
>   // properly.
> 
>   // Request notifications every time the screen becomes damaged.
>-  damage_handle_ = XDamageCreate(display(), root_window_,
>-                                 XDamageReportNonEmpty);
>+  damage_handle_ =
>+      XDamageCreate(display(), root_window_, XDamageReportNonEmpty);
>   if (!damage_handle_) {
>     RTC_LOG(LS_ERROR) << "Unable to initialize XDamage.";
>     return;
>@@ -210,8 +212,8 @@ void ScreenCapturerLinux::InitXDamage() {
>     return;
>   }
> 
>-  options_.x_display()->AddEventHandler(
>-      damage_event_base_ + XDamageNotify, this);
>+  options_.x_display()->AddEventHandler(damage_event_base_ + XDamageNotify,
>+                                        this);
> 
>   use_damage_ = true;
>   RTC_LOG(LS_INFO) << "Using XDamage extension.";
>@@ -225,6 +227,7 @@ void ScreenCapturerLinux::Start(Callback* callback) {
> }
> 
> void ScreenCapturerLinux::CaptureFrame() {
>+  TRACE_EVENT0("webrtc", "ScreenCapturerLinux::CaptureFrame");
>   int64_t capture_start_time_nanos = rtc::TimeNanos();
> 
>   queue_.MoveToNextFrame();
>@@ -237,9 +240,10 @@ void ScreenCapturerLinux::CaptureFrame() {
>   // reinitializes |x_server_pixel_buffer_|. Check if the pixel buffer is still
>   // in a good shape.
>   if (!x_server_pixel_buffer_.is_initialized()) {
>-     // We failed to initialize pixel buffer.
>-     callback_->OnCaptureResult(Result::ERROR_PERMANENT, nullptr);
>-     return;
>+    // We failed to initialize pixel buffer.
>+    RTC_LOG(LS_ERROR) << "Pixel buffer is not initialized.";
>+    callback_->OnCaptureResult(Result::ERROR_PERMANENT, nullptr);
>+    return;
>   }
> 
>   // If the current frame is from an older generation then allocate a new one.
>@@ -253,6 +257,7 @@ void ScreenCapturerLinux::CaptureFrame() {
> 
>   std::unique_ptr<DesktopFrame> result = CaptureScreen();
>   if (!result) {
>+    RTC_LOG(LS_WARNING) << "Temporarily failed to capture screen.";
>     callback_->OnCaptureResult(Result::ERROR_TEMPORARY, nullptr);
>     return;
>   }
>@@ -331,8 +336,8 @@ std::unique_ptr<DesktopFrame> ScreenCapturerLinux::CaptureScreen() {
>     updated_region->IntersectWith(
>         DesktopRect::MakeSize(x_server_pixel_buffer_.window_size()));
> 
>-    for (DesktopRegion::Iterator it(*updated_region);
>-         !it.IsAtEnd(); it.Advance()) {
>+    for (DesktopRegion::Iterator it(*updated_region); !it.IsAtEnd();
>+         it.Advance()) {
>       if (!x_server_pixel_buffer_.CaptureRect(it.rect(), frame.get()))
>         return nullptr;
>     }
>@@ -349,6 +354,7 @@ std::unique_ptr<DesktopFrame> ScreenCapturerLinux::CaptureScreen() {
> }
> 
> void ScreenCapturerLinux::ScreenConfigurationChanged() {
>+  TRACE_EVENT0("webrtc", "ScreenCapturerLinux::ScreenConfigurationChanged");
>   // Make sure the frame buffers will be reallocated.
>   queue_.Reset();
> 
>@@ -373,8 +379,8 @@ void ScreenCapturerLinux::SynchronizeFrame() {
>   DesktopFrame* current = queue_.current_frame();
>   DesktopFrame* last = queue_.previous_frame();
>   RTC_DCHECK(current != last);
>-  for (DesktopRegion::Iterator it(last_invalid_region_);
>-       !it.IsAtEnd(); it.Advance()) {
>+  for (DesktopRegion::Iterator it(last_invalid_region_); !it.IsAtEnd();
>+       it.Advance()) {
>     current->CopyPixelsFrom(*last, it.rect().top_left(), it.rect());
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_drawer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_drawer.h
>index 89dfd66ddd0..0f4ee25fca3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_drawer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_drawer.h
>@@ -15,9 +15,9 @@
> 
> #include <memory>
> 
>-#include "modules/desktop_capture/rgba_color.h"
> #include "modules/desktop_capture/desktop_capture_types.h"
> #include "modules/desktop_capture/desktop_geometry.h"
>+#include "modules/desktop_capture/rgba_color.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_drawer_linux.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_drawer_linux.cc
>index cbd8ba8612b..39ceca7dc61 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_drawer_linux.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_drawer_linux.cc
>@@ -8,16 +8,16 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include <string.h>
> #include <X11/Xlib.h>
>+#include <string.h>
> 
> #include <memory>
> 
>+#include "absl/memory/memory.h"
> #include "modules/desktop_capture/screen_drawer.h"
> #include "modules/desktop_capture/screen_drawer_lock_posix.h"
> #include "modules/desktop_capture/x11/shared_x_display.h"
> #include "rtc_base/checks.h"
>-#include "rtc_base/ptr_util.h"
> #include "system_wrappers/include/sleep.h"
> 
> namespace webrtc {
>@@ -167,13 +167,13 @@ void ScreenDrawerLinux::BringToFront() {
> 
> // static
> std::unique_ptr<ScreenDrawerLock> ScreenDrawerLock::Create() {
>-  return rtc::MakeUnique<ScreenDrawerLockPosix>();
>+  return absl::make_unique<ScreenDrawerLockPosix>();
> }
> 
> // static
> std::unique_ptr<ScreenDrawer> ScreenDrawer::Create() {
>   if (SharedXDisplay::CreateDefault().get()) {
>-    return rtc::MakeUnique<ScreenDrawerLinux>();
>+    return absl::make_unique<ScreenDrawerLinux>();
>   }
>   return nullptr;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_drawer_mac.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_drawer_mac.cc
>index bfc162c6e32..156a2e993f0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_drawer_mac.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_drawer_mac.cc
>@@ -10,15 +10,15 @@
> 
> // TODO(zijiehe): Implement ScreenDrawerMac
> 
>+#include "absl/memory/memory.h"
> #include "modules/desktop_capture/screen_drawer.h"
> #include "modules/desktop_capture/screen_drawer_lock_posix.h"
>-#include "rtc_base/ptr_util.h"
> 
> namespace webrtc {
> 
> // static
> std::unique_ptr<ScreenDrawerLock> ScreenDrawerLock::Create() {
>-  return rtc::MakeUnique<ScreenDrawerLockPosix>();
>+  return absl::make_unique<ScreenDrawerLockPosix>();
> }
> 
> // static
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_drawer_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_drawer_unittest.cc
>index 349edbc58b4..5f0f3cc5733 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_drawer_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/screen_drawer_unittest.cc
>@@ -10,15 +10,15 @@
> 
> #include "modules/desktop_capture/screen_drawer.h"
> 
>-#include <atomic>
> #include <stdint.h>
>+#include <atomic>
> 
>+#include "absl/memory/memory.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/function_view.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/random.h"
> #include "rtc_base/platform_thread.h"
>-#include "rtc_base/ptr_util.h"
>+#include "rtc_base/random.h"
> #include "rtc_base/timeutils.h"
> #include "system_wrappers/include/sleep.h"
> #include "test/gtest.h"
>@@ -45,9 +45,7 @@ void TestScreenDrawerLock(
>     Task(std::atomic<bool>* created,
>          const std::atomic<bool>& ready,
>          rtc::FunctionView<std::unique_ptr<ScreenDrawerLock>()> ctor)
>-        : created_(created),
>-          ready_(ready),
>-          ctor_(ctor) {}
>+        : created_(created), ready_(ready), ctor_(ctor) {}
> 
>     ~Task() = default;
> 
>@@ -85,7 +83,8 @@ void TestScreenDrawerLock(
> 
>   // Wait for the first lock in Task::RunTask() to be created.
>   // TODO(zijiehe): Find a better solution to wait for the creation of the first
>-  // lock. See https://chromium-review.googlesource.com/c/607688/13/webrtc/modules/desktop_capture/screen_drawer_unittest.cc
>+  // lock. See
>+  // https://chromium-review.googlesource.com/c/607688/13/webrtc/modules/desktop_capture/screen_drawer_unittest.cc
>   while (!created.load()) {
>     SleepMs(1);
>   }
>@@ -149,12 +148,10 @@ TEST(ScreenDrawerTest, TwoScreenDrawerLocks) {
>   ScreenDrawerLockPosix::Unlink(semaphore_name);
> 
>   TestScreenDrawerLock([semaphore_name]() {
>-    return rtc::MakeUnique<ScreenDrawerLockPosix>(semaphore_name);
>+    return absl::make_unique<ScreenDrawerLockPosix>(semaphore_name);
>   });
> #elif defined(WEBRTC_WIN)
>-  TestScreenDrawerLock([]() {
>-    return ScreenDrawerLock::Create();
>-  });
>+  TestScreenDrawerLock([]() { return ScreenDrawerLock::Create(); });
> #endif
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/shared_desktop_frame.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/shared_desktop_frame.cc
>index 25345d79fd6..267b5c80aeb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/shared_desktop_frame.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/shared_desktop_frame.cc
>@@ -13,8 +13,8 @@
> #include <memory>
> #include <utility>
> 
>+#include "absl/memory/memory.h"
> #include "rtc_base/constructormagic.h"
>-#include "rtc_base/ptr_util.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/shared_memory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/shared_memory.cc
>index fcbdd54e28c..b4ff78b2a0c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/shared_memory.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/shared_memory.cc
>@@ -19,10 +19,6 @@ const SharedMemory::Handle SharedMemory::kInvalidHandle = -1;
> #endif
> 
> SharedMemory::SharedMemory(void* data, size_t size, Handle handle, int id)
>-  : data_(data),
>-    size_(size),
>-    handle_(handle),
>-    id_(id) {
>-}
>+    : data_(data), size_(size), handle_(handle), id_(id) {}
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/shared_memory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/shared_memory.h
>index dc72b29b400..3ecdadf4615 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/shared_memory.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/shared_memory.h
>@@ -21,7 +21,6 @@
> 
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/thread_checker.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -80,4 +79,3 @@ class SharedMemoryFactory {
> }  // namespace webrtc
> 
> #endif  // MODULES_DESKTOP_CAPTURE_SHARED_MEMORY_H_
>-
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/test_utils.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/test_utils.cc
>index 0a4749afdba..81473eec425 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/test_utils.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/test_utils.cc
>@@ -34,8 +34,7 @@ bool DesktopFrameDataEquals(const DesktopFrame& left,
>   const uint8_t* left_array = left.data();
>   const uint8_t* right_array = right.data();
>   for (int i = 0; i < left.size().height(); i++) {
>-    if (memcmp(left_array,
>-               right_array,
>+    if (memcmp(left_array, right_array,
>                DesktopFrame::kBytesPerPixel * left.size().width()) != 0) {
>       return false;
>     }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/cursor.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/cursor.cc
>index 38dc02b883d..5aba76908a4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/cursor.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/cursor.cc
>@@ -18,7 +18,7 @@
> #include "modules/desktop_capture/mouse_cursor.h"
> #include "modules/desktop_capture/win/scoped_gdi_object.h"
> #include "rtc_base/logging.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include "rtc_base/system/arch.h"
> 
> namespace webrtc {
> 
>@@ -26,19 +26,15 @@ namespace {
> 
> #if defined(WEBRTC_ARCH_LITTLE_ENDIAN)
> 
>-#define RGBA(r, g, b, a) \
>-    ((((a) << 24) & 0xff000000) | \
>-    (((b) << 16) & 0xff0000) | \
>-    (((g) << 8) & 0xff00) | \
>-    ((r) & 0xff))
>+#define RGBA(r, g, b, a)                                   \
>+  ((((a) << 24) & 0xff000000) | (((b) << 16) & 0xff0000) | \
>+   (((g) << 8) & 0xff00) | ((r)&0xff))
> 
> #else  // !defined(WEBRTC_ARCH_LITTLE_ENDIAN)
> 
>-#define RGBA(r, g, b, a) \
>-    ((((r) << 24) & 0xff000000) | \
>-    (((g) << 16) & 0xff0000) | \
>-    (((b) << 8) & 0xff00) | \
>-    ((a) & 0xff))
>+#define RGBA(r, g, b, a)                                   \
>+  ((((r) << 24) & 0xff000000) | (((g) << 16) & 0xff0000) | \
>+   (((b) << 8) & 0xff00) | ((a)&0xff))
> 
> #endif  // !defined(WEBRTC_ARCH_LITTLE_ENDIAN)
> 
>@@ -149,13 +145,8 @@ MouseCursor* CreateMouseCursorFromHCursor(HDC dc, HCURSOR cursor) {
>   bmi.bV5AlphaMask = 0xff000000;
>   bmi.bV5CSType = LCS_WINDOWS_COLOR_SPACE;
>   bmi.bV5Intent = LCS_GM_BUSINESS;
>-  if (!GetDIBits(dc,
>-                 scoped_mask,
>-                 0,
>-                 height,
>-                 mask_data.get(),
>-                 reinterpret_cast<BITMAPINFO*>(&bmi),
>-                 DIB_RGB_COLORS)) {
>+  if (!GetDIBits(dc, scoped_mask, 0, height, mask_data.get(),
>+                 reinterpret_cast<BITMAPINFO*>(&bmi), DIB_RGB_COLORS)) {
>     RTC_LOG_F(LS_ERROR) << "Unable to get bitmap bits. Error = "
>                         << GetLastError();
>     return NULL;
>@@ -169,13 +160,8 @@ MouseCursor* CreateMouseCursorFromHCursor(HDC dc, HCURSOR cursor) {
>   if (is_color) {
>     image.reset(new BasicDesktopFrame(DesktopSize(width, height)));
>     // Get the pixels from the color bitmap.
>-    if (!GetDIBits(dc,
>-                   scoped_color,
>-                   0,
>-                   height,
>-                   image->data(),
>-                   reinterpret_cast<BITMAPINFO*>(&bmi),
>-                   DIB_RGB_COLORS)) {
>+    if (!GetDIBits(dc, scoped_color, 0, height, image->data(),
>+                   reinterpret_cast<BITMAPINFO*>(&bmi), DIB_RGB_COLORS)) {
>       RTC_LOG_F(LS_ERROR) << "Unable to get bitmap bits. Error = "
>                           << GetLastError();
>       return NULL;
>@@ -194,8 +180,8 @@ MouseCursor* CreateMouseCursorFromHCursor(HDC dc, HCURSOR cursor) {
>     image.reset(new BasicDesktopFrame(DesktopSize(width, height)));
> 
>     // The XOR mask becomes the color bitmap.
>-    memcpy(
>-        image->data(), mask_plane + (width * height), image->stride() * height);
>+    memcpy(image->data(), mask_plane + (width * height),
>+           image->stride() * height);
>   }
> 
>   // Reconstruct transparency from the mask if the color image does not has
>@@ -232,8 +218,8 @@ MouseCursor* CreateMouseCursorFromHCursor(HDC dc, HCURSOR cursor) {
>       }
>     }
>     if (add_outline) {
>-      AddCursorOutline(
>-          width, height, reinterpret_cast<uint32_t*>(image->data()));
>+      AddCursorOutline(width, height,
>+                       reinterpret_cast<uint32_t*>(image->data()));
>     }
>   }
> 
>@@ -241,8 +227,7 @@ MouseCursor* CreateMouseCursorFromHCursor(HDC dc, HCURSOR cursor) {
>   // images.
>   AlphaMul(reinterpret_cast<uint32_t*>(image->data()), width, height);
> 
>-  return new MouseCursor(
>-      image.release(), DesktopVector(hotspot_x, hotspot_y));
>+  return new MouseCursor(image.release(), DesktopVector(hotspot_x, hotspot_y));
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/cursor_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/cursor_unittest.cc
>index 368a3da5871..ff7b6bf311f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/cursor_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/cursor_unittest.cc
>@@ -74,17 +74,17 @@ bool ConvertToMouseShapeAndCompare(unsigned left, unsigned right) {
> }  // namespace
> 
> TEST(MouseCursorTest, MatchCursors) {
>-  EXPECT_TRUE(ConvertToMouseShapeAndCompare(IDD_CURSOR1_24BPP,
>-                                            IDD_CURSOR1_32BPP));
>+  EXPECT_TRUE(
>+      ConvertToMouseShapeAndCompare(IDD_CURSOR1_24BPP, IDD_CURSOR1_32BPP));
> 
>-  EXPECT_TRUE(ConvertToMouseShapeAndCompare(IDD_CURSOR1_8BPP,
>-                                            IDD_CURSOR1_32BPP));
>+  EXPECT_TRUE(
>+      ConvertToMouseShapeAndCompare(IDD_CURSOR1_8BPP, IDD_CURSOR1_32BPP));
> 
>-  EXPECT_TRUE(ConvertToMouseShapeAndCompare(IDD_CURSOR2_1BPP,
>-                                            IDD_CURSOR2_32BPP));
>+  EXPECT_TRUE(
>+      ConvertToMouseShapeAndCompare(IDD_CURSOR2_1BPP, IDD_CURSOR2_32BPP));
> 
>-  EXPECT_TRUE(ConvertToMouseShapeAndCompare(IDD_CURSOR3_4BPP,
>-                                            IDD_CURSOR3_32BPP));
>+  EXPECT_TRUE(
>+      ConvertToMouseShapeAndCompare(IDD_CURSOR3_4BPP, IDD_CURSOR3_32BPP));
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/cursor_unittest_resources.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/cursor_unittest_resources.h
>index 043103dc479..f583554d685 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/cursor_unittest_resources.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/cursor_unittest_resources.h
>@@ -11,14 +11,14 @@
> #ifndef MODULES_DESKTOP_CAPTURE_WIN_CURSOR_UNITTEST_RESOURCES_H_
> #define MODULES_DESKTOP_CAPTURE_WIN_CURSOR_UNITTEST_RESOURCES_H_
> 
>-#define IDD_CURSOR1_24BPP                101
>-#define IDD_CURSOR1_32BPP                102
>-#define IDD_CURSOR1_8BPP                 103
>+#define IDD_CURSOR1_24BPP 101
>+#define IDD_CURSOR1_32BPP 102
>+#define IDD_CURSOR1_8BPP 103
> 
>-#define IDD_CURSOR2_1BPP                 104
>-#define IDD_CURSOR2_32BPP                105
>+#define IDD_CURSOR2_1BPP 104
>+#define IDD_CURSOR2_32BPP 105
> 
>-#define IDD_CURSOR3_4BPP                 106
>-#define IDD_CURSOR3_32BPP                107
>+#define IDD_CURSOR3_4BPP 106
>+#define IDD_CURSOR3_32BPP 107
> 
> #endif  // MODULES_DESKTOP_CAPTURE_WIN_CURSOR_UNITTEST_RESOURCES_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/d3d_device.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/d3d_device.cc
>index c2b78c12ccf..b220b138a56 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/d3d_device.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/d3d_device.cc
>@@ -69,8 +69,9 @@ bool D3dDevice::Initialize(const ComPtr<IDXGIAdapter>& adapter) {
> // static
> std::vector<D3dDevice> D3dDevice::EnumDevices() {
>   ComPtr<IDXGIFactory1> factory;
>-  _com_error error = CreateDXGIFactory1(__uuidof(IDXGIFactory1),
>-      reinterpret_cast<void**>(factory.GetAddressOf()));
>+  _com_error error =
>+      CreateDXGIFactory1(__uuidof(IDXGIFactory1),
>+                         reinterpret_cast<void**>(factory.GetAddressOf()));
>   if (error.Error() != S_OK || !factory) {
>     RTC_LOG(LS_WARNING) << "Cannot create IDXGIFactory1.";
>     return std::vector<D3dDevice>();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/d3d_device.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/d3d_device.h
>index 8f545d367c5..b40279cd443 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/d3d_device.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/d3d_device.h
>@@ -11,10 +11,10 @@
> #ifndef MODULES_DESKTOP_CAPTURE_WIN_D3D_DEVICE_H_
> #define MODULES_DESKTOP_CAPTURE_WIN_D3D_DEVICE_H_
> 
>-#include <comdef.h>
>-#include <wrl/client.h>
> #include <D3D11.h>
> #include <DXGI.h>
>+#include <comdef.h>
>+#include <wrl/client.h>
> 
> #include <vector>
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/desktop.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/desktop.cc
>index 34842b3c798..459b6fe3be5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/desktop.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/desktop.cc
>@@ -16,8 +16,7 @@
> 
> namespace webrtc {
> 
>-Desktop::Desktop(HDESK desktop, bool own) : desktop_(desktop), own_(own) {
>-}
>+Desktop::Desktop(HDESK desktop, bool own) : desktop_(desktop), own_(own) {}
> 
> Desktop::~Desktop() {
>   if (own_ && desktop_ != NULL) {
>@@ -72,10 +71,10 @@ bool Desktop::SetThreadDesktop() const {
> }
> 
> Desktop* Desktop::GetDesktop(const WCHAR* desktop_name) {
>-  ACCESS_MASK desired_access =
>-      DESKTOP_CREATEMENU | DESKTOP_CREATEWINDOW | DESKTOP_ENUMERATE |
>-      DESKTOP_HOOKCONTROL | DESKTOP_WRITEOBJECTS | DESKTOP_READOBJECTS |
>-      DESKTOP_SWITCHDESKTOP | GENERIC_WRITE;
>+  ACCESS_MASK desired_access = DESKTOP_CREATEMENU | DESKTOP_CREATEWINDOW |
>+                               DESKTOP_ENUMERATE | DESKTOP_HOOKCONTROL |
>+                               DESKTOP_WRITEOBJECTS | DESKTOP_READOBJECTS |
>+                               DESKTOP_SWITCHDESKTOP | GENERIC_WRITE;
>   HDESK desktop = OpenDesktop(desktop_name, 0, FALSE, desired_access);
>   if (desktop == NULL) {
>     RTC_LOG(LS_ERROR) << "Failed to open the desktop '" << desktop_name
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/dxgi_adapter_duplicator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/dxgi_adapter_duplicator.cc
>index c4cd982d27c..81354fbead5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/dxgi_adapter_duplicator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/dxgi_adapter_duplicator.cc
>@@ -10,8 +10,8 @@
> 
> #include "modules/desktop_capture/win/dxgi_adapter_duplicator.h"
> 
>-#include <comdef.h>
> #include <DXGI.h>
>+#include <comdef.h>
> 
> #include <algorithm>
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/dxgi_context.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/dxgi_context.cc
>index 6b4649e26cb..6a679cb91f7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/dxgi_context.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/dxgi_context.cc
>@@ -14,8 +14,8 @@
> namespace webrtc {
> 
> DxgiAdapterContext::DxgiAdapterContext() = default;
>-DxgiAdapterContext::DxgiAdapterContext(
>-    const DxgiAdapterContext& context) = default;
>+DxgiAdapterContext::DxgiAdapterContext(const DxgiAdapterContext& context) =
>+    default;
> DxgiAdapterContext::~DxgiAdapterContext() = default;
> 
> DxgiFrameContext::DxgiFrameContext() = default;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/dxgi_duplicator_controller.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/dxgi_duplicator_controller.cc
>index 2da12a676ac..2029811b2ab 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/dxgi_duplicator_controller.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/dxgi_duplicator_controller.cc
>@@ -67,8 +67,7 @@ bool DxgiDuplicatorController::IsCurrentSessionSupported() {
>   return session_id != 0;
> }
> 
>-DxgiDuplicatorController::DxgiDuplicatorController()
>-    : refcount_(0) {}
>+DxgiDuplicatorController::DxgiDuplicatorController() : refcount_(0) {}
> 
> void DxgiDuplicatorController::AddRef() {
>   int refcount = (++refcount_);
>@@ -104,13 +103,14 @@ bool DxgiDuplicatorController::RetrieveD3dInfo(D3dInfo* info) {
>   return result;
> }
> 
>-DxgiDuplicatorController::Result
>-DxgiDuplicatorController::Duplicate(DxgiFrame* frame) {
>+DxgiDuplicatorController::Result DxgiDuplicatorController::Duplicate(
>+    DxgiFrame* frame) {
>   return DoDuplicate(frame, -1);
> }
> 
>-DxgiDuplicatorController::Result
>-DxgiDuplicatorController::DuplicateMonitor(DxgiFrame* frame, int monitor_id) {
>+DxgiDuplicatorController::Result DxgiDuplicatorController::DuplicateMonitor(
>+    DxgiFrame* frame,
>+    int monitor_id) {
>   RTC_DCHECK_GE(monitor_id, 0);
>   return DoDuplicate(frame, monitor_id);
> }
>@@ -141,8 +141,9 @@ bool DxgiDuplicatorController::GetDeviceNames(
>   return false;
> }
> 
>-DxgiDuplicatorController::Result
>-DxgiDuplicatorController::DoDuplicate(DxgiFrame* frame, int monitor_id) {
>+DxgiDuplicatorController::Result DxgiDuplicatorController::DoDuplicate(
>+    DxgiFrame* frame,
>+    int monitor_id) {
>   RTC_DCHECK(frame);
>   rtc::CritScope lock(&lock_);
> 
>@@ -443,8 +444,8 @@ bool DxgiDuplicatorController::EnsureFrameCaptured(Context* context,
>     // |fallback_frame|.
>     shared_frame = target;
>   } else {
>-    fallback_frame = SharedDesktopFrame::Wrap(std::unique_ptr<DesktopFrame>(
>-        new BasicDesktopFrame(desktop_size())));
>+    fallback_frame = SharedDesktopFrame::Wrap(
>+        std::unique_ptr<DesktopFrame>(new BasicDesktopFrame(desktop_size())));
>     shared_frame = fallback_frame.get();
>   }
> 
>@@ -454,8 +455,7 @@ bool DxgiDuplicatorController::EnsureFrameCaptured(Context* context,
>     if (GetNumFramesCaptured() > 0) {
>       // Sleep |ms_per_frame| before capturing next frame to ensure the screen
>       // has been updated by the video adapter.
>-      webrtc::SleepMs(
>-          ms_per_frame - (rtc::TimeMillis() - last_frame_start_ms));
>+      webrtc::SleepMs(ms_per_frame - (rtc::TimeMillis() - last_frame_start_ms));
>     }
>     last_frame_start_ms = rtc::TimeMillis();
>     if (!DoDuplicateAll(context, shared_frame)) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/dxgi_frame.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/dxgi_frame.cc
>index b51fcca73c9..85d708c82cf 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/dxgi_frame.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/dxgi_frame.cc
>@@ -21,8 +21,7 @@
> 
> namespace webrtc {
> 
>-DxgiFrame::DxgiFrame(SharedMemoryFactory* factory)
>-    : factory_(factory) {}
>+DxgiFrame::DxgiFrame(SharedMemoryFactory* factory) : factory_(factory) {}
> 
> DxgiFrame::~DxgiFrame() = default;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/dxgi_frame.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/dxgi_frame.h
>index 49f2320c31b..c87f42e01eb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/dxgi_frame.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/dxgi_frame.h
>@@ -14,8 +14,8 @@
> #include <memory>
> #include <vector>
> 
>-#include "modules/desktop_capture/desktop_capturer.h"
> #include "modules/desktop_capture/desktop_capture_types.h"
>+#include "modules/desktop_capture/desktop_capturer.h"
> #include "modules/desktop_capture/desktop_geometry.h"
> #include "modules/desktop_capture/resolution_tracker.h"
> #include "modules/desktop_capture/shared_desktop_frame.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/dxgi_output_duplicator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/dxgi_output_duplicator.cc
>index b920821dea4..f1f6f420662 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/dxgi_output_duplicator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/dxgi_output_duplicator.cc
>@@ -12,10 +12,10 @@
> 
> #include <string.h>
> 
>-#include <unknwn.h>
> #include <DXGI.h>
> #include <DXGIFormat.h>
> #include <Windows.h>
>+#include <unknwn.h>
> 
> #include <algorithm>
> 
>@@ -23,6 +23,7 @@
> #include "modules/desktop_capture/win/dxgi_texture_staging.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>+#include "rtc_base/stringutils.h"
> #include "rtc_base/win32.h"
> 
> namespace webrtc {
>@@ -176,9 +177,7 @@ bool DxgiOutputDuplicator::Duplicate(Context* context,
>   // context here. The |updated_region| always starts from (0, 0).
>   DesktopRegion updated_region;
>   updated_region.Swap(&context->updated_region);
>-  if (error.Error() == S_OK &&
>-      frame_info.AccumulatedFrames > 0 &&
>-      resource) {
>+  if (error.Error() == S_OK && frame_info.AccumulatedFrames > 0 && resource) {
>     DetectUpdatedRegion(frame_info, &context->updated_region);
>     SpreadContextChange(context);
>     if (!texture_->CopyFrom(frame_info, resource.Get())) {
>@@ -194,8 +193,8 @@ bool DxgiOutputDuplicator::Duplicate(Context* context,
>            it.Advance()) {
>         // The |updated_region| returned by Windows is rotated, but the |source|
>         // frame is not. So we need to rotate it reversely.
>-        const DesktopRect source_rect = RotateRect(
>-            it.rect(), desktop_size(), ReverseRotation(rotation_));
>+        const DesktopRect source_rect =
>+            RotateRect(it.rect(), desktop_size(), ReverseRotation(rotation_));
>         RotateDesktopFrame(source, source_rect, rotation_, offset, target);
>       }
>     } else {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/dxgi_output_duplicator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/dxgi_output_duplicator.h
>index 3f10533a224..d2a019d35b2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/dxgi_output_duplicator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/dxgi_output_duplicator.h
>@@ -11,10 +11,10 @@
> #ifndef MODULES_DESKTOP_CAPTURE_WIN_DXGI_OUTPUT_DUPLICATOR_H_
> #define MODULES_DESKTOP_CAPTURE_WIN_DXGI_OUTPUT_DUPLICATOR_H_
> 
>-#include <comdef.h>
>-#include <wrl/client.h>
> #include <DXGI.h>
> #include <DXGI1_2.h>
>+#include <comdef.h>
>+#include <wrl/client.h>
> 
> #include <memory>
> #include <string>
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/dxgi_texture.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/dxgi_texture.cc
>index 8f1d034a06b..c8081fb69ea 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/dxgi_texture.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/dxgi_texture.cc
>@@ -10,9 +10,9 @@
> 
> #include "modules/desktop_capture/win/dxgi_texture.h"
> 
>+#include <D3D11.h>
> #include <comdef.h>
> #include <wrl/client.h>
>-#include <D3D11.h>
> 
> #include "modules/desktop_capture/desktop_region.h"
> #include "rtc_base/checks.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/dxgi_texture_mapping.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/dxgi_texture_mapping.cc
>index dfe8259dd05..01bcabafc72 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/dxgi_texture_mapping.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/dxgi_texture_mapping.cc
>@@ -10,9 +10,9 @@
> 
> #include "modules/desktop_capture/win/dxgi_texture_mapping.h"
> 
>-#include <comdef.h>
> #include <DXGI.h>
> #include <DXGI1_2.h>
>+#include <comdef.h>
> 
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/dxgi_texture_staging.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/dxgi_texture_staging.cc
>index e3e915726ef..f2b35833206 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/dxgi_texture_staging.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/dxgi_texture_staging.cc
>@@ -10,10 +10,10 @@
> 
> #include "modules/desktop_capture/win/dxgi_texture_staging.h"
> 
>-#include <comdef.h>
>-#include <unknwn.h>
> #include <DXGI.h>
> #include <DXGI1_2.h>
>+#include <comdef.h>
>+#include <unknwn.h>
> 
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>@@ -45,8 +45,8 @@ bool DxgiTextureStaging::InitializeStage(ID3D11Texture2D* texture) {
>     AssertStageAndSurfaceAreSameObject();
>     D3D11_TEXTURE2D_DESC current_desc;
>     stage_->GetDesc(&current_desc);
>-    const bool recreate_needed = (
>-        memcmp(&desc, &current_desc, sizeof(D3D11_TEXTURE2D_DESC)) != 0);
>+    const bool recreate_needed =
>+        (memcmp(&desc, &current_desc, sizeof(D3D11_TEXTURE2D_DESC)) != 0);
>     RTC_HISTOGRAM_BOOLEAN("WebRTC.DesktopCapture.StagingTextureRecreate",
>                           recreate_needed);
>     if (!recreate_needed) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/dxgi_texture_staging.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/dxgi_texture_staging.h
>index 65d56baed29..998b79f8a4b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/dxgi_texture_staging.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/dxgi_texture_staging.h
>@@ -11,9 +11,9 @@
> #ifndef MODULES_DESKTOP_CAPTURE_WIN_DXGI_TEXTURE_STAGING_H_
> #define MODULES_DESKTOP_CAPTURE_WIN_DXGI_TEXTURE_STAGING_H_
> 
>-#include <wrl/client.h>
> #include <D3D11.h>
> #include <DXGI1_2.h>
>+#include <wrl/client.h>
> 
> #include "modules/desktop_capture/desktop_geometry.h"
> #include "modules/desktop_capture/desktop_region.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/scoped_gdi_object.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/scoped_gdi_object.h
>index 60a8b5ca8ee..0de316c694f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/scoped_gdi_object.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/scoped_gdi_object.h
>@@ -14,25 +14,20 @@
> #include <windows.h>
> 
> #include "rtc_base/constructormagic.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> namespace win {
> 
> // Scoper for GDI objects.
>-template<class T, class Traits>
>+template <class T, class Traits>
> class ScopedGDIObject {
>  public:
>   ScopedGDIObject() : handle_(NULL) {}
>   explicit ScopedGDIObject(T object) : handle_(object) {}
> 
>-  ~ScopedGDIObject() {
>-    Traits::Close(handle_);
>-  }
>+  ~ScopedGDIObject() { Traits::Close(handle_); }
> 
>-  T Get() {
>-    return handle_;
>-  }
>+  T Get() { return handle_; }
> 
>   void Set(T object) {
>     if (handle_ && object != handle_)
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/scoped_thread_desktop.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/scoped_thread_desktop.cc
>index bbc00efbbb8..22e8e7bc8fe 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/scoped_thread_desktop.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/scoped_thread_desktop.cc
>@@ -15,8 +15,7 @@
> namespace webrtc {
> 
> ScopedThreadDesktop::ScopedThreadDesktop()
>-    : initial_(Desktop::GetThreadDesktop()) {
>-}
>+    : initial_(Desktop::GetThreadDesktop()) {}
> 
> ScopedThreadDesktop::~ScopedThreadDesktop() {
>   Revert();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/screen_capture_utils.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/screen_capture_utils.cc
>index a76a7f527a0..e294497e553 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/screen_capture_utils.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/screen_capture_utils.cc
>@@ -17,6 +17,7 @@
> 
> #include "modules/desktop_capture/desktop_capturer.h"
> #include "rtc_base/checks.h"
>+#include "rtc_base/stringutils.h"
> #include "rtc_base/win32.h"
> 
> namespace webrtc {
>@@ -94,15 +95,14 @@ DesktopRect GetScreenRect(DesktopCapturer::SourceId screen,
>   DEVMODE device_mode;
>   device_mode.dmSize = sizeof(device_mode);
>   device_mode.dmDriverExtra = 0;
>-  result = EnumDisplaySettingsEx(
>-      device.DeviceName, ENUM_CURRENT_SETTINGS, &device_mode, 0);
>+  result = EnumDisplaySettingsEx(device.DeviceName, ENUM_CURRENT_SETTINGS,
>+                                 &device_mode, 0);
>   if (!result)
>     return DesktopRect();
> 
>-  return DesktopRect::MakeXYWH(device_mode.dmPosition.x,
>-                               device_mode.dmPosition.y,
>-                               device_mode.dmPelsWidth,
>-                               device_mode.dmPelsHeight);
>+  return DesktopRect::MakeXYWH(
>+      device_mode.dmPosition.x, device_mode.dmPosition.y,
>+      device_mode.dmPelsWidth, device_mode.dmPelsHeight);
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/screen_capture_utils.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/screen_capture_utils.h
>index 2e7d1fff164..f494dd2176e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/screen_capture_utils.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/screen_capture_utils.h
>@@ -11,8 +11,8 @@
> #ifndef MODULES_DESKTOP_CAPTURE_WIN_SCREEN_CAPTURE_UTILS_H_
> #define MODULES_DESKTOP_CAPTURE_WIN_SCREEN_CAPTURE_UTILS_H_
> 
>-#include <vector>
> #include <string>
>+#include <vector>
> 
> #include "modules/desktop_capture/desktop_capturer.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/screen_capture_utils_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/screen_capture_utils_unittest.cc
>index b9dbf3d05bd..a71c4f7610d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/screen_capture_utils_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/screen_capture_utils_unittest.cc
>@@ -10,8 +10,8 @@
> 
> #include "modules/desktop_capture/win/screen_capture_utils.h"
> 
>-#include <vector>
> #include <string>
>+#include <vector>
> 
> #include "modules/desktop_capture/desktop_capturer.h"
> #include "test/gtest.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/screen_capturer_win_directx.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/screen_capturer_win_directx.cc
>index b03d2e17dee..f038c70cf54 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/screen_capturer_win_directx.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/screen_capturer_win_directx.cc
>@@ -15,12 +15,13 @@
> #include <utility>
> #include <vector>
> 
>+#include "absl/memory/memory.h"
> #include "modules/desktop_capture/desktop_frame.h"
> #include "modules/desktop_capture/win/screen_capture_utils.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/timeutils.h"
>+#include "rtc_base/trace_event.h"
> 
> namespace webrtc {
> 
>@@ -64,14 +65,13 @@ bool ScreenCapturerWinDirectx::GetScreenListFromDeviceNames(
>   }
> 
>   for (const auto& device_name : device_names) {
>-    const auto it = std::find(
>-        gdi_names.begin(), gdi_names.end(), device_name);
>+    const auto it = std::find(gdi_names.begin(), gdi_names.end(), device_name);
>     if (it == gdi_names.end()) {
>       // devices_names[i] has not been found in gdi_names, so use max_screen_id.
>       max_screen_id++;
>-      screens->push_back({ max_screen_id });
>+      screens->push_back({max_screen_id});
>     } else {
>-      screens->push_back({ gdi_screens[it - gdi_names.begin()] });
>+      screens->push_back({gdi_screens[it - gdi_names.begin()]});
>     }
>   }
> 
>@@ -117,21 +117,22 @@ void ScreenCapturerWinDirectx::SetSharedMemoryFactory(
> 
> void ScreenCapturerWinDirectx::CaptureFrame() {
>   RTC_DCHECK(callback_);
>+  TRACE_EVENT0("webrtc", "ScreenCapturerWinDirectx::CaptureFrame");
> 
>   int64_t capture_start_time_nanos = rtc::TimeNanos();
> 
>   frames_.MoveToNextFrame();
>   if (!frames_.current_frame()) {
>     frames_.ReplaceCurrentFrame(
>-        rtc::MakeUnique<DxgiFrame>(shared_memory_factory_.get()));
>+        absl::make_unique<DxgiFrame>(shared_memory_factory_.get()));
>   }
> 
>   DxgiDuplicatorController::Result result;
>   if (current_screen_id_ == kFullDesktopScreenId) {
>     result = controller_->Duplicate(frames_.current_frame());
>   } else {
>-    result = controller_->DuplicateMonitor(
>-        frames_.current_frame(), current_screen_id_);
>+    result = controller_->DuplicateMonitor(frames_.current_frame(),
>+                                           current_screen_id_);
>   }
> 
>   using DuplicateResult = DxgiDuplicatorController::Result;
>@@ -168,9 +169,8 @@ void ScreenCapturerWinDirectx::CaptureFrame() {
>     case DuplicateResult::SUCCEEDED: {
>       std::unique_ptr<DesktopFrame> frame =
>           frames_.current_frame()->frame()->Share();
>-      frame->set_capture_time_ms(
>-          (rtc::TimeNanos() - capture_start_time_nanos) /
>-          rtc::kNumNanosecsPerMillisec);
>+      frame->set_capture_time_ms((rtc::TimeNanos() - capture_start_time_nanos) /
>+                                 rtc::kNumNanosecsPerMillisec);
>       frame->set_capturer_id(DesktopCapturerId::kScreenCapturerWinDirectx);
>       callback_->OnCaptureResult(Result::SUCCESS, std::move(frame));
>       break;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/screen_capturer_win_directx_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/screen_capturer_win_directx_unittest.cc
>index d07a6482eec..5874cb820ee 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/screen_capturer_win_directx_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/screen_capturer_win_directx_unittest.cc
>@@ -10,8 +10,8 @@
> 
> #include "modules/desktop_capture/win/screen_capturer_win_directx.h"
> 
>-#include <vector>
> #include <string>
>+#include <vector>
> 
> #include "modules/desktop_capture/desktop_capturer.h"
> #include "test/gtest.h"
>@@ -22,9 +22,7 @@ namespace webrtc {
> // devices in its output, since the device name is missing.
> TEST(ScreenCaptureUtilsTest, GetScreenListFromDeviceNamesAndGetIndex) {
>   const std::vector<std::string> device_names = {
>-    "\\\\.\\DISPLAY0",
>-    "\\\\.\\DISPLAY1",
>-    "\\\\.\\DISPLAY2",
>+      "\\\\.\\DISPLAY0", "\\\\.\\DISPLAY1", "\\\\.\\DISPLAY2",
>   };
>   DesktopCapturer::SourceList screens;
>   ASSERT_TRUE(ScreenCapturerWinDirectx::GetScreenListFromDeviceNames(
>@@ -32,8 +30,8 @@ TEST(ScreenCaptureUtilsTest, GetScreenListFromDeviceNamesAndGetIndex) {
>   ASSERT_EQ(device_names.size(), screens.size());
> 
>   for (size_t i = 0; i < screens.size(); i++) {
>-    ASSERT_EQ(ScreenCapturerWinDirectx::GetIndexFromScreenId(
>-                  screens[i].id, device_names),
>+    ASSERT_EQ(ScreenCapturerWinDirectx::GetIndexFromScreenId(screens[i].id,
>+                                                             device_names),
>               static_cast<int>(i));
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/screen_capturer_win_gdi.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/screen_capturer_win_gdi.cc
>index 2c299fc72a6..03f8340bbc1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/screen_capturer_win_gdi.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/screen_capturer_win_gdi.cc
>@@ -23,6 +23,7 @@
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/timeutils.h"
>+#include "rtc_base/trace_event.h"
> 
> namespace webrtc {
> 
>@@ -70,6 +71,7 @@ void ScreenCapturerWinGdi::SetSharedMemoryFactory(
> }
> 
> void ScreenCapturerWinGdi::CaptureFrame() {
>+  TRACE_EVENT0("webrtc", "ScreenCapturerWinGdi::CaptureFrame");
>   int64_t capture_start_time_nanos = rtc::TimeNanos();
> 
>   queue_.MoveToNextFrame();
>@@ -79,20 +81,19 @@ void ScreenCapturerWinGdi::CaptureFrame() {
>   PrepareCaptureResources();
> 
>   if (!CaptureImage()) {
>+    RTC_LOG(WARNING) << "Failed to capture screen by GDI.";
>     callback_->OnCaptureResult(Result::ERROR_TEMPORARY, nullptr);
>     return;
>   }
> 
>   // Emit the current frame.
>   std::unique_ptr<DesktopFrame> frame = queue_.current_frame()->Share();
>-  frame->set_dpi(DesktopVector(
>-      GetDeviceCaps(desktop_dc_, LOGPIXELSX),
>-      GetDeviceCaps(desktop_dc_, LOGPIXELSY)));
>+  frame->set_dpi(DesktopVector(GetDeviceCaps(desktop_dc_, LOGPIXELSX),
>+                               GetDeviceCaps(desktop_dc_, LOGPIXELSY)));
>   frame->mutable_updated_region()->SetRect(
>       DesktopRect::MakeSize(frame->size()));
>-  frame->set_capture_time_ms(
>-      (rtc::TimeNanos() - capture_start_time_nanos) /
>-      rtc::kNumNanosecsPerMillisec);
>+  frame->set_capture_time_ms((rtc::TimeNanos() - capture_start_time_nanos) /
>+                             rtc::kNumNanosecsPerMillisec);
>   frame->set_capturer_id(DesktopCapturerId::kScreenCapturerWinGdi);
>   callback_->OnCaptureResult(Result::SUCCESS, std::move(frame));
> }
>@@ -177,8 +178,10 @@ void ScreenCapturerWinGdi::PrepareCaptureResources() {
> bool ScreenCapturerWinGdi::CaptureImage() {
>   DesktopRect screen_rect =
>       GetScreenRect(current_screen_id_, current_device_key_);
>-  if (screen_rect.is_empty())
>+  if (screen_rect.is_empty()) {
>+    RTC_LOG(LS_WARNING) << "Failed to get screen rect.";
>     return false;
>+  }
> 
>   DesktopSize size = screen_rect.size();
>   // If the current buffer is from an older generation then allocate a new one.
>@@ -191,8 +194,10 @@ bool ScreenCapturerWinGdi::CaptureImage() {
> 
>     std::unique_ptr<DesktopFrame> buffer = DesktopFrameWin::Create(
>         size, shared_memory_factory_.get(), desktop_dc_);
>-    if (!buffer)
>+    if (!buffer) {
>+      RTC_LOG(LS_WARNING) << "Failed to create frame buffer.";
>       return false;
>+    }
>     queue_.ReplaceCurrentFrame(SharedDesktopFrame::Wrap(std::move(buffer)));
>   }
>   queue_.current_frame()->set_top_left(
>@@ -204,12 +209,13 @@ bool ScreenCapturerWinGdi::CaptureImage() {
>       queue_.current_frame()->GetUnderlyingFrame());
>   HGDIOBJ previous_object = SelectObject(memory_dc_, current->bitmap());
>   if (!previous_object || previous_object == HGDI_ERROR) {
>+    RTC_LOG(LS_WARNING) << "Failed to select current bitmap into memery dc.";
>     return false;
>   }
> 
>   bool result = (BitBlt(memory_dc_, 0, 0, screen_rect.width(),
>-      screen_rect.height(), desktop_dc_, screen_rect.left(), screen_rect.top(),
>-      SRCCOPY | CAPTUREBLT) != FALSE);
>+                        screen_rect.height(), desktop_dc_, screen_rect.left(),
>+                        screen_rect.top(), SRCCOPY | CAPTUREBLT) != FALSE);
>   if (!result) {
>     RTC_LOG_GLE(LS_WARNING) << "BitBlt failed";
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/screen_capturer_win_gdi.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/screen_capturer_win_gdi.h
>index 2e30d59e0bb..be4794efd8e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/screen_capturer_win_gdi.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/screen_capturer_win_gdi.h
>@@ -44,7 +44,7 @@ class ScreenCapturerWinGdi : public DesktopCapturer {
>   bool SelectSource(SourceId id) override;
> 
>  private:
>-  typedef HRESULT (WINAPI * DwmEnableCompositionFunc)(UINT);
>+  typedef HRESULT(WINAPI* DwmEnableCompositionFunc)(UINT);
> 
>   // Make sure that the device contexts match the screen configuration.
>   void PrepareCaptureResources();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.cc
>index d52b028814b..c7b07d15b85 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.cc
>@@ -136,8 +136,8 @@ bool ScreenCapturerWinMagnifier::SelectSource(SourceId id) {
> void ScreenCapturerWinMagnifier::SetExcludedWindow(WindowId excluded_window) {
>   excluded_window_ = (HWND)excluded_window;
>   if (excluded_window_ && magnifier_initialized_) {
>-    set_window_filter_list_func_(
>-        magnifier_window_, MW_FILTERMODE_EXCLUDE, 1, &excluded_window_);
>+    set_window_filter_list_func_(magnifier_window_, MW_FILTERMODE_EXCLUDE, 1,
>+                                 &excluded_window_);
>   }
> }
> 
>@@ -186,8 +186,7 @@ BOOL ScreenCapturerWinMagnifier::OnMagImageScalingCallback(
>     RECT clipped,
>     HRGN dirty) {
>   ScreenCapturerWinMagnifier* owner =
>-      reinterpret_cast<ScreenCapturerWinMagnifier*>(
>-          TlsGetValue(GetTlsIndex()));
>+      reinterpret_cast<ScreenCapturerWinMagnifier*>(TlsGetValue(GetTlsIndex()));
>   TlsSetValue(GetTlsIndex(), nullptr);
>   owner->OnCaptured(srcdata, srcheader);
> 
>@@ -243,10 +242,10 @@ bool ScreenCapturerWinMagnifier::InitializeMagnifier() {
>   }
> 
>   HMODULE hInstance = nullptr;
>-  result = GetModuleHandleExA(GET_MODULE_HANDLE_EX_FLAG_FROM_ADDRESS |
>-                                  GET_MODULE_HANDLE_EX_FLAG_UNCHANGED_REFCOUNT,
>-                              reinterpret_cast<char*>(&DefWindowProc),
>-                              &hInstance);
>+  result =
>+      GetModuleHandleExA(GET_MODULE_HANDLE_EX_FLAG_FROM_ADDRESS |
>+                             GET_MODULE_HANDLE_EX_FLAG_UNCHANGED_REFCOUNT,
>+                         reinterpret_cast<char*>(&DefWindowProc), &hInstance);
>   if (!result) {
>     mag_uninitialize_func_();
>     RTC_LOG_F(LS_WARNING) << "Failed to initialize ScreenCapturerWinMagnifier: "
>@@ -346,8 +345,7 @@ void ScreenCapturerWinMagnifier::OnCaptured(void* data,
> 
>   // Copy the data into the frame.
>   current_frame->CopyPixelsFrom(
>-      reinterpret_cast<uint8_t*>(data),
>-      header.stride,
>+      reinterpret_cast<uint8_t*>(data), header.stride,
>       DesktopRect::MakeXYWH(0, 0, header.width, header.height));
> 
>   magnifier_capture_succeeded_ = true;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.h
>index d994b1e0597..28f29bce158 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.h
>@@ -13,9 +13,9 @@
> 
> #include <memory>
> 
>-#include <windows.h>
> #include <magnification.h>
> #include <wincodec.h>
>+#include <windows.h>
> 
> #include "modules/desktop_capture/desktop_capturer.h"
> #include "modules/desktop_capture/screen_capture_frame_queue.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/window_capture_utils.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/window_capture_utils.cc
>index 41c62dca598..4c5489fa71f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/window_capture_utils.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/window_capture_utils.cc
>@@ -12,17 +12,20 @@
> 
> #include "modules/desktop_capture/win/scoped_gdi_object.h"
> #include "rtc_base/checks.h"
>+#include "rtc_base/logging.h"
> #include "rtc_base/win32.h"
> 
> namespace webrtc {
> 
>+// Prefix used to match the window class for Chrome windows.
>+const wchar_t kChromeWindowClassPrefix[] = L"Chrome_WidgetWin_";
>+
> bool GetWindowRect(HWND window, DesktopRect* result) {
>   RECT rect;
>   if (!::GetWindowRect(window, &rect)) {
>     return false;
>   }
>-  *result = DesktopRect::MakeLTRB(
>-      rect.left, rect.top, rect.right, rect.bottom);
>+  *result = DesktopRect::MakeLTRB(rect.left, rect.top, rect.right, rect.bottom);
>   return true;
> }
> 
>@@ -93,22 +96,22 @@ bool GetWindowContentRect(HWND window, DesktopRect* result) {
> }
> 
> int GetWindowRegionTypeWithBoundary(HWND window, DesktopRect* result) {
>-  win::ScopedGDIObject<HRGN, win::DeleteObjectTraits<HRGN>>
>-      scoped_hrgn(CreateRectRgn(0, 0, 0, 0));
>+  win::ScopedGDIObject<HRGN, win::DeleteObjectTraits<HRGN>> scoped_hrgn(
>+      CreateRectRgn(0, 0, 0, 0));
>   const int region_type = GetWindowRgn(window, scoped_hrgn.Get());
> 
>   if (region_type == SIMPLEREGION) {
>     RECT rect;
>     GetRgnBox(scoped_hrgn.Get(), &rect);
>-    *result = DesktopRect::MakeLTRB(
>-        rect.left, rect.top, rect.right, rect.bottom);
>+    *result =
>+        DesktopRect::MakeLTRB(rect.left, rect.top, rect.right, rect.bottom);
>   }
>   return region_type;
> }
> 
> bool GetDcSize(HDC hdc, DesktopSize* size) {
>-  win::ScopedGDIObject<HGDIOBJ, win::DeleteObjectTraits<HGDIOBJ>>
>-      scoped_hgdi(GetCurrentObject(hdc, OBJ_BITMAP));
>+  win::ScopedGDIObject<HGDIOBJ, win::DeleteObjectTraits<HGDIOBJ>> scoped_hgdi(
>+      GetCurrentObject(hdc, OBJ_BITMAP));
>   BITMAP bitmap;
>   memset(&bitmap, 0, sizeof(BITMAP));
>   if (GetObject(scoped_hgdi.Get(), sizeof(BITMAP), &bitmap) == 0) {
>@@ -130,22 +133,34 @@ bool IsWindowMaximized(HWND window, bool* result) {
>   return true;
> }
> 
>-AeroChecker::AeroChecker() : dwmapi_library_(nullptr), func_(nullptr) {
>+// WindowCaptureHelperWin implementation.
>+WindowCaptureHelperWin::WindowCaptureHelperWin()
>+    : dwmapi_library_(nullptr),
>+      func_(nullptr),
>+      virtual_desktop_manager_(nullptr) {
>   // Try to load dwmapi.dll dynamically since it is not available on XP.
>   dwmapi_library_ = LoadLibrary(L"dwmapi.dll");
>   if (dwmapi_library_) {
>     func_ = reinterpret_cast<DwmIsCompositionEnabledFunc>(
>         GetProcAddress(dwmapi_library_, "DwmIsCompositionEnabled"));
>   }
>+
>+  if (rtc::IsWindows10OrLater()) {
>+    if (FAILED(::CoCreateInstance(__uuidof(VirtualDesktopManager), nullptr,
>+                                  CLSCTX_ALL,
>+                                  IID_PPV_ARGS(&virtual_desktop_manager_)))) {
>+      RTC_LOG(LS_WARNING) << "Fail to create instance of VirtualDesktopManager";
>+    }
>+  }
> }
> 
>-AeroChecker::~AeroChecker() {
>+WindowCaptureHelperWin::~WindowCaptureHelperWin() {
>   if (dwmapi_library_) {
>     FreeLibrary(dwmapi_library_);
>   }
> }
> 
>-bool AeroChecker::IsAeroEnabled() {
>+bool WindowCaptureHelperWin::IsAeroEnabled() {
>   BOOL result = FALSE;
>   if (func_) {
>     func_(&result);
>@@ -153,4 +168,53 @@ bool AeroChecker::IsAeroEnabled() {
>   return result != FALSE;
> }
> 
>+// This is just a best guess of a notification window. Chrome uses the Windows
>+// native framework for showing notifications. So far what we know about such a
>+// window includes: no title, class name with prefix "Chrome_WidgetWin_" and
>+// with certain extended styles.
>+bool WindowCaptureHelperWin::IsWindowChromeNotification(HWND hwnd) {
>+  const size_t kTitleLength = 32;
>+  WCHAR window_title[kTitleLength];
>+  GetWindowText(hwnd, window_title, kTitleLength);
>+  if (wcsnlen_s(window_title, kTitleLength) != 0) {
>+    return false;
>+  }
>+
>+  const size_t kClassLength = 256;
>+  WCHAR class_name[kClassLength];
>+  const int class_name_length = GetClassName(hwnd, class_name, kClassLength);
>+  RTC_DCHECK(class_name_length)
>+      << "Error retrieving the application's class name";
>+  if (wcsncmp(class_name, kChromeWindowClassPrefix,
>+              wcsnlen_s(kChromeWindowClassPrefix, kClassLength)) != 0) {
>+    return false;
>+  }
>+
>+  const LONG exstyle = GetWindowLong(hwnd, GWL_EXSTYLE);
>+  if ((exstyle & WS_EX_NOACTIVATE) && (exstyle & WS_EX_TOOLWINDOW) &&
>+      (exstyle & WS_EX_TOPMOST)) {
>+    return true;
>+  }
>+
>+  return false;
>+}
>+
>+bool WindowCaptureHelperWin::IsWindowOnCurrentDesktop(HWND hwnd) {
>+  // Make sure the window is on the current virtual desktop.
>+  if (virtual_desktop_manager_) {
>+    BOOL on_current_desktop;
>+    if (SUCCEEDED(virtual_desktop_manager_->IsWindowOnCurrentVirtualDesktop(
>+            hwnd, &on_current_desktop)) &&
>+        !on_current_desktop) {
>+      return false;
>+    }
>+  }
>+  return true;
>+}
>+
>+bool WindowCaptureHelperWin::IsWindowVisibleOnCurrentDesktop(HWND hwnd) {
>+  return !::IsIconic(hwnd) && ::IsWindowVisible(hwnd) &&
>+         IsWindowOnCurrentDesktop(hwnd);
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/window_capture_utils.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/window_capture_utils.h
>index c38995a82ad..024c7eb860c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/window_capture_utils.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/win/window_capture_utils.h
>@@ -8,7 +8,9 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>+#include <shlobj.h>
> #include <windows.h>
>+#include <wrl/client.h>
> 
> #include "modules/desktop_capture/desktop_geometry.h"
> #include "rtc_base/constructormagic.h"
>@@ -57,19 +59,25 @@ bool GetDcSize(HDC hdc, DesktopSize* size);
> // function returns false if native APIs fail.
> bool IsWindowMaximized(HWND window, bool* result);
> 
>-typedef HRESULT (WINAPI *DwmIsCompositionEnabledFunc)(BOOL* enabled);
>-class AeroChecker {
>+typedef HRESULT(WINAPI* DwmIsCompositionEnabledFunc)(BOOL* enabled);
>+class WindowCaptureHelperWin {
>  public:
>-  AeroChecker();
>-  ~AeroChecker();
>+  WindowCaptureHelperWin();
>+  ~WindowCaptureHelperWin();
> 
>   bool IsAeroEnabled();
>+  bool IsWindowChromeNotification(HWND hwnd);
>+  bool IsWindowOnCurrentDesktop(HWND hwnd);
>+  bool IsWindowVisibleOnCurrentDesktop(HWND hwnd);
> 
>  private:
>   HMODULE dwmapi_library_;
>   DwmIsCompositionEnabledFunc func_;
> 
>-  RTC_DISALLOW_COPY_AND_ASSIGN(AeroChecker);
>+  // Only used on Win10+.
>+  Microsoft::WRL::ComPtr<IVirtualDesktopManager> virtual_desktop_manager_;
>+
>+  RTC_DISALLOW_COPY_AND_ASSIGN(WindowCaptureHelperWin);
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/window_capturer_mac.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/window_capturer_mac.mm
>index bbdcdd52836..95e622a77a1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/window_capturer_mac.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/window_capturer_mac.mm
>@@ -18,15 +18,16 @@
> #include "modules/desktop_capture/desktop_capture_options.h"
> #include "modules/desktop_capture/desktop_capturer.h"
> #include "modules/desktop_capture/desktop_frame.h"
>-#include "modules/desktop_capture/window_finder_mac.h"
> #include "modules/desktop_capture/mac/desktop_configuration.h"
> #include "modules/desktop_capture/mac/desktop_configuration_monitor.h"
> #include "modules/desktop_capture/mac/full_screen_chrome_window_detector.h"
> #include "modules/desktop_capture/mac/window_list_utils.h"
>+#include "modules/desktop_capture/window_finder_mac.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/macutils.h"
> #include "rtc_base/scoped_ref_ptr.h"
>+#include "rtc_base/trace_event.h"
> 
> namespace webrtc {
> 
>@@ -155,7 +156,10 @@ void WindowCapturerMac::Start(Callback* callback) {
> }
> 
> void WindowCapturerMac::CaptureFrame() {
>+  TRACE_EVENT0("webrtc", "WindowCapturerMac::CaptureFrame");
>+
>   if (!IsWindowValid(window_id_)) {
>+    RTC_LOG(LS_ERROR) << "The window is not valid any longer.";
>     callback_->OnCaptureResult(Result::ERROR_PERMANENT, nullptr);
>     return;
>   }
>@@ -174,6 +178,7 @@ void WindowCapturerMac::CaptureFrame() {
>       on_screen_window, kCGWindowImageBoundsIgnoreFraming);
> 
>   if (!window_image) {
>+    RTC_LOG(LS_WARNING) << "Temporarily failed to capture window.";
>     callback_->OnCaptureResult(Result::ERROR_TEMPORARY, nullptr);
>     return;
>   }
>@@ -205,17 +210,10 @@ void WindowCapturerMac::CaptureFrame() {
> 
>   frame->mutable_updated_region()->SetRect(
>       DesktopRect::MakeSize(frame->size()));
>-  DesktopVector top_left;
>-  if (configuration_monitor_) {
>-    configuration_monitor_->Lock();
>-    auto configuration = configuration_monitor_->desktop_configuration();
>-    configuration_monitor_->Unlock();
>-    top_left = GetWindowBounds(configuration, on_screen_window).top_left();
>-    top_left = top_left.subtract(configuration.bounds.top_left());
>-  } else {
>-    top_left = GetWindowBounds(on_screen_window).top_left();
>-  }
>-  frame->set_top_left(top_left);
>+  frame->set_top_left(GetWindowBounds(on_screen_window).top_left());
>+
>+  float scale_factor = GetWindowScaleFactor(window_id_, frame->size());
>+  frame->set_dpi(DesktopVector(kStandardDPI * scale_factor, kStandardDPI * scale_factor));
> 
>   callback_->OnCaptureResult(Result::SUCCESS, std::move(frame));
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/window_capturer_null.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/window_capturer_null.cc
>old mode 100755
>new mode 100644
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/window_capturer_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/window_capturer_unittest.cc
>index 0c31fb7823a..ac51d15c7a3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/window_capturer_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/window_capturer_unittest.cc
>@@ -10,8 +10,8 @@
> 
> #include <memory>
> 
>-#include "modules/desktop_capture/desktop_capturer.h"
> #include "modules/desktop_capture/desktop_capture_options.h"
>+#include "modules/desktop_capture/desktop_capturer.h"
> #include "modules/desktop_capture/desktop_frame.h"
> #include "modules/desktop_capture/desktop_region.h"
> #include "test/gtest.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/window_capturer_win.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/window_capturer_win.cc
>index 414eddf64b0..f345a5e2e4d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/window_capturer_win.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/window_capturer_win.cc
>@@ -14,12 +14,14 @@
> 
> #include "modules/desktop_capture/desktop_capturer.h"
> #include "modules/desktop_capture/desktop_frame_win.h"
>-#include "modules/desktop_capture/window_finder_win.h"
> #include "modules/desktop_capture/win/screen_capture_utils.h"
> #include "modules/desktop_capture/win/window_capture_utils.h"
>+#include "modules/desktop_capture/window_finder_win.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/logging.h"
>+#include "rtc_base/stringutils.h"
>+#include "rtc_base/trace_event.h"
> #include "rtc_base/win32.h"
> 
> namespace webrtc {
>@@ -102,8 +104,8 @@ bool GetWindowDrawableRect(HWND window,
>   }
> 
>   if (is_maximized) {
>-    return GetCroppedWindowRect(
>-        window, drawable_rect, /* original_rect */ nullptr);
>+    return GetCroppedWindowRect(window, drawable_rect,
>+                                /* original_rect */ nullptr);
>   }
>   *drawable_rect = *original_rect;
>   return true;
>@@ -131,7 +133,7 @@ class WindowCapturerWin : public DesktopCapturer {
> 
>   DesktopSize previous_size_;
> 
>-  AeroChecker aero_checker_;
>+  WindowCaptureHelperWin window_capture_helper_;
> 
>   // This map is used to avoid flickering for the case when SelectWindow() calls
>   // are interleaved with Capture() calls.
>@@ -151,6 +153,15 @@ bool WindowCapturerWin::GetSourceList(SourceList* sources) {
>   // EnumWindows only enumerates root windows.
>   if (!EnumWindows(&WindowsEnumerationHandler, param))
>     return false;
>+
>+  for (auto it = result.begin(); it != result.end();) {
>+    if (!window_capture_helper_.IsWindowOnCurrentDesktop(
>+            reinterpret_cast<HWND>(it->id))) {
>+      it = result.erase(it);
>+    } else {
>+      ++it;
>+    }
>+  }
>   sources->swap(result);
> 
>   std::map<HWND, DesktopSize> new_map;
>@@ -187,8 +198,8 @@ bool WindowCapturerWin::FocusOnSelectedSource() {
> 
> bool WindowCapturerWin::IsOccluded(const DesktopVector& pos) {
>   DesktopVector sys_pos = pos.add(GetFullscreenRect().top_left());
>-  return reinterpret_cast<HWND>(window_finder_.GetWindowUnderPoint(sys_pos))
>-      != window_;
>+  return reinterpret_cast<HWND>(window_finder_.GetWindowUnderPoint(sys_pos)) !=
>+         window_;
> }
> 
> void WindowCapturerWin::Start(Callback* callback) {
>@@ -199,6 +210,8 @@ void WindowCapturerWin::Start(Callback* callback) {
> }
> 
> void WindowCapturerWin::CaptureFrame() {
>+  TRACE_EVENT0("webrtc", "WindowCapturerWin::CaptureFrame");
>+
>   if (!window_) {
>     RTC_LOG(LS_ERROR) << "Window hasn't been selected: " << GetLastError();
>     callback_->OnCaptureResult(Result::ERROR_PERMANENT, nullptr);
>@@ -207,6 +220,7 @@ void WindowCapturerWin::CaptureFrame() {
> 
>   // Stop capturing if the window has been closed.
>   if (!IsWindow(window_)) {
>+    RTC_LOG(LS_ERROR) << "target window has been closed";
>     callback_->OnCaptureResult(Result::ERROR_PERMANENT, nullptr);
>     return;
>   }
>@@ -220,12 +234,11 @@ void WindowCapturerWin::CaptureFrame() {
>     return;
>   }
> 
>-  // Return a 1x1 black frame if the window is minimized or invisible, to match
>-  // behavior on mace. Window can be temporarily invisible during the
>-  // transition of full screen mode on/off.
>+  // Return a 1x1 black frame if the window is minimized or invisible on current
>+  // desktop, to match behavior on mace. Window can be temporarily invisible
>+  // during the transition of full screen mode on/off.
>   if (original_rect.is_empty() ||
>-      IsIconic(window_) ||
>-      !IsWindowVisible(window_)) {
>+      !window_capture_helper_.IsWindowVisibleOnCurrentDesktop(window_)) {
>     std::unique_ptr<DesktopFrame> frame(
>         new BasicDesktopFrame(DesktopSize(1, 1)));
>     memset(frame->data(), 0, frame->stride() * frame->size().height());
>@@ -269,6 +282,7 @@ void WindowCapturerWin::CaptureFrame() {
>   std::unique_ptr<DesktopFrameWin> frame(
>       DesktopFrameWin::Create(cropped_rect.size(), nullptr, window_dc));
>   if (!frame.get()) {
>+    RTC_LOG(LS_WARNING) << "Failed to create frame.";
>     ReleaseDC(window_, window_dc);
>     callback_->OnCaptureResult(Result::ERROR_TEMPORARY, nullptr);
>     return;
>@@ -295,17 +309,16 @@ void WindowCapturerWin::CaptureFrame() {
>   // capturing - it somehow affects what we get from BitBlt() on the subsequent
>   // captures.
> 
>-  if (!aero_checker_.IsAeroEnabled() || !previous_size_.equals(frame->size())) {
>+  if (!window_capture_helper_.IsAeroEnabled() ||
>+      !previous_size_.equals(frame->size())) {
>     result = PrintWindow(window_, mem_dc, 0);
>   }
> 
>   // Aero is enabled or PrintWindow() failed, use BitBlt.
>   if (!result) {
>     result = BitBlt(mem_dc, 0, 0, frame->size().width(), frame->size().height(),
>-                    window_dc,
>-                    cropped_rect.left() - original_rect.left(),
>-                    cropped_rect.top() - original_rect.top(),
>-                    SRCCOPY);
>+                    window_dc, cropped_rect.left() - original_rect.left(),
>+                    cropped_rect.top() - original_rect.top(), SRCCOPY);
>   }
> 
>   SelectObject(mem_dc, previous_object);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/window_capturer_x11.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/window_capturer_x11.cc
>index 8985ab163ec..c4ca3ae5e8c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/window_capturer_x11.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/window_capturer_x11.cc
>@@ -10,9 +10,9 @@
> 
> #include <string.h>
> 
>+#include <X11/Xutil.h>
> #include <X11/extensions/Xcomposite.h>
> #include <X11/extensions/Xrender.h>
>-#include <X11/Xutil.h>
> 
> #include <utility>
> 
>@@ -28,6 +28,7 @@
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/scoped_ref_ptr.h"
>+#include "rtc_base/trace_event.h"
> 
> namespace webrtc {
> 
>@@ -92,15 +93,14 @@ WindowCapturerLinux::~WindowCapturerLinux() {
> }
> 
> bool WindowCapturerLinux::GetSourceList(SourceList* sources) {
>-  return GetWindowList(&atom_cache_,
>-                       [this, sources](::Window window) {
>-                         Source w;
>-                         w.id = window;
>-                         if (this->GetWindowTitle(window, &w.title)) {
>-                           sources->push_back(w);
>-                         }
>-                         return true;
>-                       });
>+  return GetWindowList(&atom_cache_, [this, sources](::Window window) {
>+    Source w;
>+    w.id = window;
>+    if (this->GetWindowTitle(window, &w.title)) {
>+      sources->push_back(w);
>+    }
>+    return true;
>+  });
> }
> 
> bool WindowCapturerLinux::SelectSource(SourceId id) {
>@@ -133,8 +133,8 @@ bool WindowCapturerLinux::FocusOnSelectedSource() {
>   ::Window parent;
>   ::Window root;
>   // Find the root window to pass event to.
>-  int status = XQueryTree(
>-      display(), selected_window_, &root, &parent, &children, &num_children);
>+  int status = XQueryTree(display(), selected_window_, &root, &parent,
>+                          &children, &num_children);
>   if (status == 0) {
>     RTC_LOG(LS_ERROR) << "Failed to query for the root window.";
>     return false;
>@@ -163,11 +163,8 @@ bool WindowCapturerLinux::FocusOnSelectedSource() {
> 
>     memset(xev.xclient.data.l, 0, sizeof(xev.xclient.data.l));
> 
>-    XSendEvent(display(),
>-               root,
>-               False,
>-               SubstructureRedirectMask | SubstructureNotifyMask,
>-               &xev);
>+    XSendEvent(display(), root, False,
>+               SubstructureRedirectMask | SubstructureNotifyMask, &xev);
>   }
>   XFlush(display());
>   return true;
>@@ -181,8 +178,10 @@ void WindowCapturerLinux::Start(Callback* callback) {
> }
> 
> void WindowCapturerLinux::CaptureFrame() {
>+  TRACE_EVENT0("webrtc", "WindowCapturerLinux::CaptureFrame");
>+
>   if (!x_server_pixel_buffer_.IsWindowValid()) {
>-    RTC_LOG(LS_INFO) << "The window is no longer valid.";
>+    RTC_LOG(LS_ERROR) << "The window is no longer valid.";
>     callback_->OnCaptureResult(Result::ERROR_PERMANENT, nullptr);
>     return;
>   }
>@@ -193,7 +192,7 @@ void WindowCapturerLinux::CaptureFrame() {
>     // Without the Xcomposite extension we capture when the whole window is
>     // visible on screen and not covered by any other window. This is not
>     // something we want so instead, just bail out.
>-    RTC_LOG(LS_INFO) << "No Xcomposite extension detected.";
>+    RTC_LOG(LS_ERROR) << "No Xcomposite extension detected.";
>     callback_->OnCaptureResult(Result::ERROR_PERMANENT, nullptr);
>     return;
>   }
>@@ -212,6 +211,7 @@ void WindowCapturerLinux::CaptureFrame() {
>   x_server_pixel_buffer_.Synchronize();
>   if (!x_server_pixel_buffer_.CaptureRect(DesktopRect::MakeSize(frame->size()),
>                                           frame.get())) {
>+    RTC_LOG(LS_WARNING) << "Temporarily failed to capture winodw.";
>     callback_->OnCaptureResult(Result::ERROR_TEMPORARY, nullptr);
>     return;
>   }
>@@ -225,7 +225,7 @@ void WindowCapturerLinux::CaptureFrame() {
> 
> bool WindowCapturerLinux::IsOccluded(const DesktopVector& pos) {
>   return window_finder_.GetWindowUnderPoint(pos) !=
>-      static_cast<WindowId>(selected_window_);
>+         static_cast<WindowId>(selected_window_);
> }
> 
> bool WindowCapturerLinux::HandleXEvent(const XEvent& event) {
>@@ -233,7 +233,7 @@ bool WindowCapturerLinux::HandleXEvent(const XEvent& event) {
>     XConfigureEvent xce = event.xconfigure;
>     if (xce.window == selected_window_) {
>       if (!DesktopRectFromXAttributes(xce).equals(
>-            x_server_pixel_buffer_.window_rect())) {
>+              x_server_pixel_buffer_.window_rect())) {
>         if (!x_server_pixel_buffer_.Init(display(), selected_window_)) {
>           RTC_LOG(LS_ERROR)
>               << "Failed to initialize pixel buffer after resizing.";
>@@ -256,8 +256,8 @@ bool WindowCapturerLinux::GetWindowTitle(::Window window, std::string* title) {
>     if (status && window_name.value && window_name.nitems) {
>       int cnt;
>       char** list = nullptr;
>-      status = Xutf8TextPropertyToTextList(display(), &window_name, &list,
>-                                           &cnt);
>+      status =
>+          Xutf8TextPropertyToTextList(display(), &window_name, &list, &cnt);
>       if (status >= Success && cnt && *list) {
>         if (cnt > 1) {
>           RTC_LOG(LS_INFO) << "Window has " << cnt
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/window_finder_mac.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/window_finder_mac.mm
>index 6df0d4d9b7e..f75822a1a3f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/window_finder_mac.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/window_finder_mac.mm
>@@ -14,10 +14,10 @@
> 
> #include <utility>
> 
>-#include "modules/desktop_capture/mac/window_list_utils.h"
>+#include "absl/memory/memory.h"
> #include "modules/desktop_capture/mac/desktop_configuration.h"
> #include "modules/desktop_capture/mac/desktop_configuration_monitor.h"
>-#include "rtc_base/ptr_util.h"
>+#include "modules/desktop_capture/mac/window_list_utils.h"
> 
> namespace webrtc {
> 
>@@ -28,35 +28,24 @@ WindowFinderMac::~WindowFinderMac() = default;
> 
> WindowId WindowFinderMac::GetWindowUnderPoint(DesktopVector point) {
>   WindowId id = kNullWindowId;
>-  MacDesktopConfiguration configuration_holder;
>-  MacDesktopConfiguration* configuration = nullptr;
>-  if (configuration_monitor_) {
>-    configuration_monitor_->Lock();
>-    configuration_holder = configuration_monitor_->desktop_configuration();
>-    configuration_monitor_->Unlock();
>-    configuration = &configuration_holder;
>-  }
>-  GetWindowList([&id, point, configuration](CFDictionaryRef window) {
>-                  DesktopRect bounds;
>-                  if (configuration) {
>-                    bounds = GetWindowBounds(*configuration, window);
>-                  } else {
>-                    bounds = GetWindowBounds(window);
>-                  }
>-                  if (bounds.Contains(point)) {
>-                    id = GetWindowId(window);
>-                    return false;
>-                  }
>-                  return true;
>-                },
>-                true);
>+  GetWindowList(
>+      [&id, point](CFDictionaryRef window) {
>+        DesktopRect bounds;
>+        bounds = GetWindowBounds(window);
>+        if (bounds.Contains(point)) {
>+          id = GetWindowId(window);
>+          return false;
>+        }
>+        return true;
>+      },
>+      true);
>   return id;
> }
> 
> // static
> std::unique_ptr<WindowFinder> WindowFinder::Create(
>     const WindowFinder::Options& options) {
>-  return rtc::MakeUnique<WindowFinderMac>(options.configuration_monitor);
>+  return absl::make_unique<WindowFinderMac>(options.configuration_monitor);
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/window_finder_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/window_finder_unittest.cc
>index ccbc46956ab..88327aaac54 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/window_finder_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/window_finder_unittest.cc
>@@ -20,16 +20,16 @@
> #include "test/gtest.h"
> 
> #if defined(USE_X11)
>+#include "absl/memory/memory.h"
> #include "modules/desktop_capture/x11/shared_x_display.h"
> #include "modules/desktop_capture/x11/x_atom_cache.h"
>-#include "rtc_base/ptr_util.h"
> #endif
> 
> #if defined(WEBRTC_WIN)
> #include <windows.h>
> 
>-#include "modules/desktop_capture/window_finder_win.h"
> #include "modules/desktop_capture/win/window_capture_utils.h"
>+#include "modules/desktop_capture/window_finder_win.h"
> #endif
> 
> namespace webrtc {
>@@ -64,8 +64,8 @@ TEST(WindowFinderTest, FindConsoleWindow) {
>   MoveWindow(console_window, 0, 0, kMaxSize, kMaxSize, true);
> 
>   // Brings console window to top.
>-  SetWindowPos(
>-      console_window, HWND_TOPMOST, 0, 0, 0, 0, SWP_NOMOVE | SWP_NOSIZE);
>+  SetWindowPos(console_window, HWND_TOPMOST, 0, 0, 0, 0,
>+               SWP_NOMOVE | SWP_NOSIZE);
>   BringWindowToTop(console_window);
> 
>   WindowFinderWin finder;
>@@ -87,7 +87,7 @@ TEST(WindowFinderTest, FindDrawerWindow) {
>   std::unique_ptr<XAtomCache> cache;
>   const auto shared_x_display = SharedXDisplay::CreateDefault();
>   if (shared_x_display) {
>-    cache = rtc::MakeUnique<XAtomCache>(shared_x_display->display());
>+    cache = absl::make_unique<XAtomCache>(shared_x_display->display());
>     options.cache = cache.get();
>   }
> #endif
>@@ -143,7 +143,7 @@ TEST(WindowFinderTest, ShouldReturnNullWindowIfSpotIsOutOfScreen) {
>   std::unique_ptr<XAtomCache> cache;
>   const auto shared_x_display = SharedXDisplay::CreateDefault();
>   if (shared_x_display) {
>-    cache = rtc::MakeUnique<XAtomCache>(shared_x_display->display());
>+    cache = absl::make_unique<XAtomCache>(shared_x_display->display());
>     options.cache = cache.get();
>   }
> #endif
>@@ -154,14 +154,14 @@ TEST(WindowFinderTest, ShouldReturnNullWindowIfSpotIsOutOfScreen) {
>     return;
>   }
> 
>-  ASSERT_EQ(kNullWindowId, finder->GetWindowUnderPoint(
>-      DesktopVector(INT16_MAX, INT16_MAX)));
>-  ASSERT_EQ(kNullWindowId, finder->GetWindowUnderPoint(
>-      DesktopVector(INT16_MAX, INT16_MIN)));
>-  ASSERT_EQ(kNullWindowId, finder->GetWindowUnderPoint(
>-      DesktopVector(INT16_MIN, INT16_MAX)));
>-  ASSERT_EQ(kNullWindowId, finder->GetWindowUnderPoint(
>-      DesktopVector(INT16_MIN, INT16_MIN)));
>+  ASSERT_EQ(kNullWindowId,
>+            finder->GetWindowUnderPoint(DesktopVector(INT16_MAX, INT16_MAX)));
>+  ASSERT_EQ(kNullWindowId,
>+            finder->GetWindowUnderPoint(DesktopVector(INT16_MAX, INT16_MIN)));
>+  ASSERT_EQ(kNullWindowId,
>+            finder->GetWindowUnderPoint(DesktopVector(INT16_MIN, INT16_MAX)));
>+  ASSERT_EQ(kNullWindowId,
>+            finder->GetWindowUnderPoint(DesktopVector(INT16_MIN, INT16_MIN)));
> }
> 
> }  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/window_finder_win.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/window_finder_win.cc
>index 9f87106a30c..10ffe0d9792 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/window_finder_win.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/window_finder_win.cc
>@@ -12,7 +12,7 @@
> 
> #include <windows.h>
> 
>-#include "rtc_base/ptr_util.h"
>+#include "absl/memory/memory.h"
> 
> namespace webrtc {
> 
>@@ -20,7 +20,7 @@ WindowFinderWin::WindowFinderWin() = default;
> WindowFinderWin::~WindowFinderWin() = default;
> 
> WindowId WindowFinderWin::GetWindowUnderPoint(DesktopVector point) {
>-  HWND window = WindowFromPoint(POINT { point.x(), point.y() });
>+  HWND window = WindowFromPoint(POINT{point.x(), point.y()});
>   if (!window) {
>     return kNullWindowId;
>   }
>@@ -40,7 +40,7 @@ WindowId WindowFinderWin::GetWindowUnderPoint(DesktopVector point) {
> // static
> std::unique_ptr<WindowFinder> WindowFinder::Create(
>     const WindowFinder::Options& options) {
>-  return rtc::MakeUnique<WindowFinderWin>();
>+  return absl::make_unique<WindowFinderWin>();
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/window_finder_x11.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/window_finder_x11.cc
>index 3096217e5ba..03694326d2f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/window_finder_x11.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/window_finder_x11.cc
>@@ -10,14 +10,13 @@
> 
> #include "modules/desktop_capture/window_finder_x11.h"
> 
>+#include "absl/memory/memory.h"
> #include "modules/desktop_capture/x11/window_list_utils.h"
> #include "rtc_base/checks.h"
>-#include "rtc_base/ptr_util.h"
> 
> namespace webrtc {
> 
>-WindowFinderX11::WindowFinderX11(XAtomCache* cache)
>-    : cache_(cache) {
>+WindowFinderX11::WindowFinderX11(XAtomCache* cache) : cache_(cache) {
>   RTC_DCHECK(cache_);
> }
> 
>@@ -25,16 +24,15 @@ WindowFinderX11::~WindowFinderX11() = default;
> 
> WindowId WindowFinderX11::GetWindowUnderPoint(DesktopVector point) {
>   WindowId id = kNullWindowId;
>-  GetWindowList(cache_,
>-                [&id, this, point](::Window window) {
>-                  DesktopRect rect;
>-                  if (GetWindowRect(this->cache_->display(), window, &rect) &&
>-                      rect.Contains(point)) {
>-                    id = window;
>-                    return false;
>-                  }
>-                  return true;
>-                });
>+  GetWindowList(cache_, [&id, this, point](::Window window) {
>+    DesktopRect rect;
>+    if (GetWindowRect(this->cache_->display(), window, &rect) &&
>+        rect.Contains(point)) {
>+      id = window;
>+      return false;
>+    }
>+    return true;
>+  });
>   return id;
> }
> 
>@@ -45,7 +43,7 @@ std::unique_ptr<WindowFinder> WindowFinder::Create(
>     return nullptr;
>   }
> 
>-  return rtc::MakeUnique<WindowFinderX11>(options.cache);
>+  return absl::make_unique<WindowFinderX11>(options.cache);
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/x11/shared_x_display.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/x11/shared_x_display.cc
>index ff2cba5b4a3..db6a64b81d9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/x11/shared_x_display.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/x11/shared_x_display.cc
>@@ -19,8 +19,7 @@
> 
> namespace webrtc {
> 
>-SharedXDisplay::SharedXDisplay(Display* display)
>-  : display_(display) {
>+SharedXDisplay::SharedXDisplay(Display* display) : display_(display) {
>   RTC_DCHECK(display_);
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/x11/window_list_utils.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/x11/window_list_utils.cc
>index 65c7edffbef..fe070d88782 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/x11/window_list_utils.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/x11/window_list_utils.cc
>@@ -10,10 +10,10 @@
> 
> #include "modules/desktop_capture/x11/window_list_utils.h"
> 
>-#include <string.h>
>-#include <X11/Xlib.h>
> #include <X11/Xatom.h>
>+#include <X11/Xlib.h>
> #include <X11/Xutil.h>
>+#include <string.h>
> 
> #include <algorithm>
> 
>@@ -49,10 +49,9 @@ class XWindowProperty {
>     Atom actual_type;
>     int actual_format;
>     unsigned long bytes_after;  // NOLINT: type required by XGetWindowProperty
>-    int status = XGetWindowProperty(display, window, property, 0L, ~0L, False,
>-                                    AnyPropertyType, &actual_type,
>-                                    &actual_format, &size_,
>-                                    &bytes_after, &data_);
>+    int status = XGetWindowProperty(
>+        display, window, property, 0L, ~0L, False, AnyPropertyType,
>+        &actual_type, &actual_format, &size_, &bytes_after, &data_);
>     if (status != Success) {
>       data_ = nullptr;
>       return;
>@@ -78,9 +77,7 @@ class XWindowProperty {
>   const PropertyType* data() const {
>     return reinterpret_cast<PropertyType*>(data_);
>   }
>-  PropertyType* data() {
>-    return reinterpret_cast<PropertyType*>(data_);
>-  }
>+  PropertyType* data() { return reinterpret_cast<PropertyType*>(data_); }
> 
>  private:
>   bool is_valid_ = false;
>@@ -106,7 +103,7 @@ class XWindowProperty {
>   RTC_DCHECK_EQ(state, WithdrawnState);
>   // If the window is in WithdrawnState then look at all of its children.
>   ::Window root, parent;
>-  ::Window *children;
>+  ::Window* children;
>   unsigned int num_children;
>   if (!XQueryTree(cache->display(), window, &root, &parent, &children,
>                   &num_children)) {
>@@ -137,14 +134,12 @@ bool IsDesktopElement(XAtomCache* cache, ::Window window) {
>   // says this hint *should* be present on all windows, and we use the existence
>   // of _NET_WM_WINDOW_TYPE_NORMAL in the property to indicate a window is not
>   // a desktop element (that is, only "normal" windows should be shareable).
>-  XWindowProperty<uint32_t> window_type(
>-      cache->display(), window, cache->WindowType());
>+  XWindowProperty<uint32_t> window_type(cache->display(), window,
>+                                        cache->WindowType());
>   if (window_type.is_valid() && window_type.size() > 0) {
>     uint32_t* end = window_type.data() + window_type.size();
>-    bool is_normal = (end != std::find(
>-        window_type.data(),
>-        end,
>-        cache->WindowTypeNormal()));
>+    bool is_normal =
>+        (end != std::find(window_type.data(), end, cache->WindowTypeNormal()));
>     return !is_normal;
>   }
> 
>@@ -166,8 +161,8 @@ bool IsDesktopElement(XAtomCache* cache, ::Window window) {
> 
> int32_t GetWindowState(XAtomCache* cache, ::Window window) {
>   // Get WM_STATE property of the window.
>-  XWindowProperty<uint32_t> window_state(
>-      cache->display(), window, cache->WmState());
>+  XWindowProperty<uint32_t> window_state(cache->display(), window,
>+                                         cache->WmState());
> 
>   // WM_STATE is considered to be set to WithdrawnState when it missing.
>   return window_state.is_valid() ? *window_state.data() : WithdrawnState;
>@@ -188,11 +183,7 @@ bool GetWindowList(XAtomCache* cache,
>     unsigned int num_children;
>     {
>       XErrorTrap error_trap(display);
>-      if (XQueryTree(display,
>-                     root_window,
>-                     &root_window,
>-                     &parent,
>-                     &children,
>+      if (XQueryTree(display, root_window, &root_window, &parent, &children,
>                      &num_children) == 0 ||
>           error_trap.GetLastErrorAndDisable() != 0) {
>         failed_screens++;
>@@ -242,14 +233,8 @@ bool GetWindowRect(::Display* display,
>   {
>     XErrorTrap error_trap(display);
>     ::Window child;
>-    if (!XTranslateCoordinates(display,
>-                               window,
>-                               attributes->root,
>-                               -rect->left(),
>-                               -rect->top(),
>-                               &offset_x,
>-                               &offset_y,
>-                               &child) ||
>+    if (!XTranslateCoordinates(display, window, attributes->root, -rect->left(),
>+                               -rect->top(), &offset_x, &offset_y, &child) ||
>         error_trap.GetLastErrorAndDisable() != 0) {
>       return false;
>     }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/x11/window_list_utils.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/x11/window_list_utils.h
>index 92571b2ca62..72d2a70d210 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/x11/window_list_utils.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/x11/window_list_utils.h
>@@ -45,9 +45,7 @@ bool GetWindowRect(::Display* display,
> // Creates a DesktopRect from |attributes|.
> template <typename T>
> DesktopRect DesktopRectFromXAttributes(const T& attributes) {
>-  return DesktopRect::MakeXYWH(attributes.x,
>-                               attributes.y,
>-                               attributes.width,
>+  return DesktopRect::MakeXYWH(attributes.x, attributes.y, attributes.width,
>                                attributes.height);
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/x11/x_atom_cache.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/x11/x_atom_cache.h
>index d77d2fb524d..5246c2bb5c9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/x11/x_atom_cache.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/x11/x_atom_cache.h
>@@ -11,8 +11,8 @@
> #ifndef MODULES_DESKTOP_CAPTURE_X11_X_ATOM_CACHE_H_
> #define MODULES_DESKTOP_CAPTURE_X11_X_ATOM_CACHE_H_
> 
>-#include <X11/Xlib.h>
> #include <X11/Xatom.h>
>+#include <X11/Xlib.h>
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/x11/x_error_trap.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/x11/x_error_trap.cc
>index a0dbac6c46b..6559c3dbe9f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/x11/x_error_trap.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/x11/x_error_trap.cc
>@@ -37,11 +37,10 @@ int XServerErrorHandler(Display* display, XErrorEvent* error_event) {
> }  // namespace
> 
> XErrorTrap::XErrorTrap(Display* display)
>-    : original_error_handler_(NULL),
>-      enabled_(true) {
>+    : original_error_handler_(NULL), enabled_(true) {
> #if defined(TOOLKIT_GTK)
>   gdk_error_trap_push();
>-#else  // !defined(TOOLKIT_GTK)
>+#else   // !defined(TOOLKIT_GTK)
>   assert(!g_xserver_error_trap_enabled);
>   original_error_handler_ = XSetErrorHandler(&XServerErrorHandler);
>   g_xserver_error_trap_enabled = true;
>@@ -53,7 +52,7 @@ int XErrorTrap::GetLastErrorAndDisable() {
>   enabled_ = false;
> #if defined(TOOLKIT_GTK)
>   return gdk_error_trap_push();
>-#else  // !defined(TOOLKIT_GTK)
>+#else   // !defined(TOOLKIT_GTK)
>   assert(g_xserver_error_trap_enabled);
>   XSetErrorHandler(original_error_handler_);
>   g_xserver_error_trap_enabled = false;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/x11/x_server_pixel_buffer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/x11/x_server_pixel_buffer.cc
>index ab1b95add40..e327d3fc7dd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/x11/x_server_pixel_buffer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/desktop_capture/x11/x_server_pixel_buffer.cc
>@@ -52,10 +52,8 @@ uint32_t MaskToShift(uint32_t mask) {
> 
> // Returns true if |image| is in RGB format.
> bool IsXImageRGBFormat(XImage* image) {
>-  return image->bits_per_pixel == 32 &&
>-      image->red_mask == 0xff0000 &&
>-      image->green_mask == 0xff00 &&
>-      image->blue_mask == 0xff;
>+  return image->bits_per_pixel == 32 && image->red_mask == 0xff0000 &&
>+         image->green_mask == 0xff00 && image->blue_mask == 0xff;
> }
> 
> // We expose two forms of blitting to handle variations in the pixel format.
>@@ -255,11 +253,9 @@ bool XServerPixelBuffer::InitPixmaps(int depth) {
> 
>   {
>     XErrorTrap error_trap(display_);
>-    shm_pixmap_ = XShmCreatePixmap(display_, window_,
>-                                   shm_segment_info_->shmaddr,
>-                                   shm_segment_info_,
>-                                   window_rect_.width(),
>-                                   window_rect_.height(), depth);
>+    shm_pixmap_ = XShmCreatePixmap(
>+        display_, window_, shm_segment_info_->shmaddr, shm_segment_info_,
>+        window_rect_.width(), window_rect_.height(), depth);
>     XSync(display_, False);
>     if (error_trap.GetLastErrorAndDisable() != 0) {
>       // |shm_pixmap_| is not not valid because the request was not processed
>@@ -275,8 +271,7 @@ bool XServerPixelBuffer::InitPixmaps(int depth) {
>     shm_gc_values.subwindow_mode = IncludeInferiors;
>     shm_gc_values.graphics_exposures = False;
>     shm_gc_ = XCreateGC(display_, window_,
>-                        GCSubwindowMode | GCGraphicsExposures,
>-                        &shm_gc_values);
>+                        GCSubwindowMode | GCGraphicsExposures, &shm_gc_values);
>     XSync(display_, False);
>     if (error_trap.GetLastErrorAndDisable() != 0) {
>       XFreePixmap(display_, shm_pixmap_);
>@@ -321,9 +316,9 @@ bool XServerPixelBuffer::CaptureRect(const DesktopRect& rect,
> 
>   if (shm_segment_info_ && (shm_pixmap_ || xshm_get_image_succeeded_)) {
>     if (shm_pixmap_) {
>-      XCopyArea(display_, window_, shm_pixmap_, shm_gc_,
>-                rect.left(), rect.top(), rect.width(), rect.height(),
>-                rect.left(), rect.top());
>+      XCopyArea(display_, window_, shm_pixmap_, shm_gc_, rect.left(),
>+                rect.top(), rect.width(), rect.height(), rect.left(),
>+                rect.top());
>       XSync(display_, False);
>     }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/include/module.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/include/module.h
>index fc2a1b5fc96..3b1e046ab12 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/include/module.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/include/module.h
>@@ -11,7 +11,7 @@
> #ifndef MODULES_INCLUDE_MODULE_H_
> #define MODULES_INCLUDE_MODULE_H_
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include <stdint.h>
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/include/module_common_types.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/include/module_common_types.h
>index 94ccbdc1580..4175c411d5f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/include/module_common_types.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/include/module_common_types.h
>@@ -17,79 +17,31 @@
> #include <algorithm>
> #include <limits>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> #include "api/rtp_headers.h"
>+#include "api/transport/network_types.h"
> #include "api/video/video_rotation.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/include/module_common_types_public.h"
>-#include "modules/video_coding/codecs/h264/include/h264_globals.h"
>-#include "modules/video_coding/codecs/stereo/include/stereo_globals.h"
>-#include "modules/video_coding/codecs/vp8/include/vp8_globals.h"
>-#include "modules/video_coding/codecs/vp9/include/vp9_globals.h"
>+#include "modules/include/module_fec_types.h"
>+#include "modules/rtp_rtcp/source/rtp_video_header.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/deprecation.h"
> #include "rtc_base/numerics/safe_conversions.h"
> #include "rtc_base/timeutils.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>-struct RTPAudioHeader {
>-  uint8_t numEnergy;                  // number of valid entries in arrOfEnergy
>-  uint8_t arrOfEnergy[kRtpCsrcSize];  // one energy byte (0-9) per channel
>-  bool isCNG;                         // is this CNG
>-  size_t channel;                     // number of channels 2 = stereo
>-};
>-
>-enum RtpVideoCodecTypes {
>-  kRtpVideoNone = 0,
>-  kRtpVideoGeneric = 1,
>-  kRtpVideoVp8 = 2,
>-  kRtpVideoVp9 = 3,
>-  kRtpVideoH264 = 4,
>-  kRtpVideoStereo = 5
>-};
>-
>-struct RTPVideoHeaderStereo {
>-  RtpVideoCodecTypes associated_codec_type;
>-  StereoIndices indices;
>-};
>-
>-union RTPVideoTypeHeader {
>-  RTPVideoHeaderVP8 VP8;
>-  RTPVideoHeaderVP9 VP9;
>-  RTPVideoHeaderH264 H264;
>-  RTPVideoHeaderStereo stereo;
>-};
>-
>-// Since RTPVideoHeader is used as a member of a union, it can't have a
>-// non-trivial default constructor.
>-struct RTPVideoHeader {
>-  uint16_t width;  // size
>-  uint16_t height;
>-  VideoRotation rotation;
>-
>-  PlayoutDelay playout_delay;
>-
>-  VideoContentType content_type;
>-
>-  VideoSendTiming video_timing;
>-
>-  bool is_first_packet_in_frame;
>-  uint8_t simulcastIdx;  // Index if the simulcast encoder creating
>-                         // this frame, 0 if not using simulcast.
>-  RtpVideoCodecTypes codec;
>-  RTPVideoTypeHeader codecHeader;
>-};
>-union RTPTypeHeader {
>-  RTPAudioHeader Audio;
>-  RTPVideoHeader Video;
>-};
>+// TODO(nisse): Deprecated, use webrtc::VideoCodecType instead.
>+using RtpVideoCodecTypes = VideoCodecType;
> 
> struct WebRtcRTPHeader {
>+  RTPVideoHeader& video_header() { return video; }
>+  const RTPVideoHeader& video_header() const { return video; }
>+  RTPVideoHeader video;
>+
>   RTPHeader header;
>   FrameType frameType;
>-  RTPTypeHeader type;
>   // NTP time of the capture time in local timebase in milliseconds.
>   int64_t ntp_time_ms;
> };
>@@ -196,8 +148,7 @@ class RTPFragmentationHeader {
>         memset(fragmentationOffset + oldVectorSize, 0,
>                sizeof(size_t) * (size16 - oldVectorSize));
>         // copy old values
>-        memcpy(fragmentationOffset, oldOffsets,
>-               sizeof(size_t) * oldVectorSize);
>+        memcpy(fragmentationOffset, oldOffsets, sizeof(size_t) * oldVectorSize);
>         delete[] oldOffsets;
>       }
>       // length
>@@ -206,8 +157,7 @@ class RTPFragmentationHeader {
>         fragmentationLength = new size_t[size16];
>         memset(fragmentationLength + oldVectorSize, 0,
>                sizeof(size_t) * (size16 - oldVectorSize));
>-        memcpy(fragmentationLength, oldLengths,
>-               sizeof(size_t) * oldVectorSize);
>+        memcpy(fragmentationLength, oldLengths, sizeof(size_t) * oldVectorSize);
>         delete[] oldLengths;
>       }
>       // time diff
>@@ -270,22 +220,6 @@ struct RTCPVoIPMetric {
>   uint16_t JBabsMax;
> };
> 
>-// Types for the FEC packet masks. The type |kFecMaskRandom| is based on a
>-// random loss model. The type |kFecMaskBursty| is based on a bursty/consecutive
>-// loss model. The packet masks are defined in
>-// modules/rtp_rtcp/fec_private_tables_random(bursty).h
>-enum FecMaskType {
>-  kFecMaskRandom,
>-  kFecMaskBursty,
>-};
>-
>-// Struct containing forward error correction settings.
>-struct FecProtectionParams {
>-  int fec_rate;
>-  int max_fec_frames;
>-  FecMaskType fec_mask_type;
>-};
>-
> // Interface used by the CallStats class to distribute call statistics.
> // Callbacks will be triggered as soon as the class has been registered to a
> // CallStats object using RegisterStatsObserver.
>@@ -296,312 +230,32 @@ class CallStatsObserver {
>   virtual ~CallStatsObserver() {}
> };
> 
>-/* This class holds up to 60 ms of super-wideband (32 kHz) stereo audio. It
>- * allows for adding and subtracting frames while keeping track of the resulting
>- * states.
>- *
>- * Notes
>- * - The total number of samples is samples_per_channel_ * num_channels_
>- * - Stereo data is interleaved starting with the left channel.
>- */
>-class AudioFrame {
>+// Interface used by NackModule and JitterBuffer.
>+class NackSender {
>  public:
>-  // Using constexpr here causes linker errors unless the variable also has an
>-  // out-of-class definition, which is impractical in this header-only class.
>-  // (This makes no sense because it compiles as an enum value, which we most
>-  // certainly cannot take the address of, just fine.) C++17 introduces inline
>-  // variables which should allow us to switch to constexpr and keep this a
>-  // header-only class.
>-  enum : size_t {
>-    // Stereo, 32 kHz, 60 ms (2 * 32 * 60)
>-    kMaxDataSizeSamples = 3840,
>-    kMaxDataSizeBytes = kMaxDataSizeSamples * sizeof(int16_t),
>-  };
>-
>-  enum VADActivity {
>-    kVadActive = 0,
>-    kVadPassive = 1,
>-    kVadUnknown = 2
>-  };
>-  enum SpeechType {
>-    kNormalSpeech = 0,
>-    kPLC = 1,
>-    kCNG = 2,
>-    kPLCCNG = 3,
>-    kUndefined = 4
>-  };
>-
>-  AudioFrame();
>-
>-  // Resets all members to their default state.
>-  void Reset();
>-  // Same as Reset(), but leaves mute state unchanged. Muting a frame requires
>-  // the buffer to be zeroed on the next call to mutable_data(). Callers
>-  // intending to write to the buffer immediately after Reset() can instead use
>-  // ResetWithoutMuting() to skip this wasteful zeroing.
>-  void ResetWithoutMuting();
>-
>-  // TODO(solenberg): Remove once downstream users of AudioFrame have updated.
>-  RTC_DEPRECATED
>-      void UpdateFrame(int id, uint32_t timestamp, const int16_t* data,
>-                       size_t samples_per_channel, int sample_rate_hz,
>-                       SpeechType speech_type, VADActivity vad_activity,
>-                       size_t num_channels = 1) {
>-    RTC_UNUSED(id);
>-    UpdateFrame(timestamp, data, samples_per_channel, sample_rate_hz,
>-                speech_type, vad_activity, num_channels);
>-  }
>-
>-  void UpdateFrame(uint32_t timestamp, const int16_t* data,
>-                   size_t samples_per_channel, int sample_rate_hz,
>-                   SpeechType speech_type, VADActivity vad_activity,
>-                   size_t num_channels = 1);
>-
>-  void CopyFrom(const AudioFrame& src);
>-
>-  // Sets a wall-time clock timestamp in milliseconds to be used for profiling
>-  // of time between two points in the audio chain.
>-  // Example:
>-  //   t0: UpdateProfileTimeStamp()
>-  //   t1: ElapsedProfileTimeMs() => t1 - t0 [msec]
>-  void UpdateProfileTimeStamp();
>-  // Returns the time difference between now and when UpdateProfileTimeStamp()
>-  // was last called. Returns -1 if UpdateProfileTimeStamp() has not yet been
>-  // called.
>-  int64_t ElapsedProfileTimeMs() const;
>-
>-  // data() returns a zeroed static buffer if the frame is muted.
>-  // mutable_frame() always returns a non-static buffer; the first call to
>-  // mutable_frame() zeros the non-static buffer and marks the frame unmuted.
>-  const int16_t* data() const;
>-  int16_t* mutable_data();
>-
>-  // Prefer to mute frames using AudioFrameOperations::Mute.
>-  void Mute();
>-  // Frame is muted by default.
>-  bool muted() const;
>-
>-  // These methods are deprecated. Use the functions in
>-  // webrtc/audio/utility instead. These methods will exists for a
>-  // short period of time until webrtc clients have updated. See
>-  // webrtc:6548 for details.
>-  RTC_DEPRECATED AudioFrame& operator>>=(const int rhs);
>-  RTC_DEPRECATED AudioFrame& operator+=(const AudioFrame& rhs);
>-
>-  // RTP timestamp of the first sample in the AudioFrame.
>-  uint32_t timestamp_ = 0;
>-  // Time since the first frame in milliseconds.
>-  // -1 represents an uninitialized value.
>-  int64_t elapsed_time_ms_ = -1;
>-  // NTP time of the estimated capture time in local timebase in milliseconds.
>-  // -1 represents an uninitialized value.
>-  int64_t ntp_time_ms_ = -1;
>-  size_t samples_per_channel_ = 0;
>-  int sample_rate_hz_ = 0;
>-  size_t num_channels_ = 0;
>-  SpeechType speech_type_ = kUndefined;
>-  VADActivity vad_activity_ = kVadUnknown;
>-  // Monotonically increasing timestamp intended for profiling of audio frames.
>-  // Typically used for measuring elapsed time between two different points in
>-  // the audio path. No lock is used to save resources and we are thread safe
>-  // by design. Also, rtc::Optional is not used since it will cause a "complex
>-  // class/struct needs an explicit out-of-line destructor" build error.
>-  int64_t profile_timestamp_ms_ = 0;
>-
>- private:
>-  // A permamently zeroed out buffer to represent muted frames. This is a
>-  // header-only class, so the only way to avoid creating a separate empty
>-  // buffer per translation unit is to wrap a static in an inline function.
>-  static const int16_t* empty_data() {
>-    static const int16_t kEmptyData[kMaxDataSizeSamples] = {0};
>-    static_assert(sizeof(kEmptyData) == kMaxDataSizeBytes, "kMaxDataSizeBytes");
>-    return kEmptyData;
>-  }
>-
>-  int16_t data_[kMaxDataSizeSamples];
>-  bool muted_ = true;
>+  virtual void SendNack(const std::vector<uint16_t>& sequence_numbers) = 0;
> 
>-  RTC_DISALLOW_COPY_AND_ASSIGN(AudioFrame);
>+ protected:
>+  virtual ~NackSender() {}
> };
> 
>-inline AudioFrame::AudioFrame() {
>-  // Visual Studio doesn't like this in the class definition.
>-  static_assert(sizeof(data_) == kMaxDataSizeBytes, "kMaxDataSizeBytes");
>-}
>-
>-inline void AudioFrame::Reset() {
>-  ResetWithoutMuting();
>-  muted_ = true;
>-}
>-
>-inline void AudioFrame::ResetWithoutMuting() {
>-  // TODO(wu): Zero is a valid value for |timestamp_|. We should initialize
>-  // to an invalid value, or add a new member to indicate invalidity.
>-  timestamp_ = 0;
>-  elapsed_time_ms_ = -1;
>-  ntp_time_ms_ = -1;
>-  samples_per_channel_ = 0;
>-  sample_rate_hz_ = 0;
>-  num_channels_ = 0;
>-  speech_type_ = kUndefined;
>-  vad_activity_ = kVadUnknown;
>-  profile_timestamp_ms_ = 0;
>-}
>-
>-inline void AudioFrame::UpdateFrame(uint32_t timestamp,
>-                                    const int16_t* data,
>-                                    size_t samples_per_channel,
>-                                    int sample_rate_hz,
>-                                    SpeechType speech_type,
>-                                    VADActivity vad_activity,
>-                                    size_t num_channels) {
>-  timestamp_ = timestamp;
>-  samples_per_channel_ = samples_per_channel;
>-  sample_rate_hz_ = sample_rate_hz;
>-  speech_type_ = speech_type;
>-  vad_activity_ = vad_activity;
>-  num_channels_ = num_channels;
>-
>-  const size_t length = samples_per_channel * num_channels;
>-  assert(length <= kMaxDataSizeSamples);
>-  if (data != nullptr) {
>-    memcpy(data_, data, sizeof(int16_t) * length);
>-    muted_ = false;
>-  } else {
>-    muted_ = true;
>-  }
>-}
>-
>-inline void AudioFrame::CopyFrom(const AudioFrame& src) {
>-  if (this == &src) return;
>-
>-  timestamp_ = src.timestamp_;
>-  elapsed_time_ms_ = src.elapsed_time_ms_;
>-  ntp_time_ms_ = src.ntp_time_ms_;
>-  muted_ = src.muted();
>-  samples_per_channel_ = src.samples_per_channel_;
>-  sample_rate_hz_ = src.sample_rate_hz_;
>-  speech_type_ = src.speech_type_;
>-  vad_activity_ = src.vad_activity_;
>-  num_channels_ = src.num_channels_;
>-
>-  const size_t length = samples_per_channel_ * num_channels_;
>-  assert(length <= kMaxDataSizeSamples);
>-  if (!src.muted()) {
>-    memcpy(data_, src.data(), sizeof(int16_t) * length);
>-    muted_ = false;
>-  }
>-}
>-
>-inline void AudioFrame::UpdateProfileTimeStamp() {
>-  profile_timestamp_ms_ = rtc::TimeMillis();
>-}
>-
>-inline int64_t AudioFrame::ElapsedProfileTimeMs() const {
>-  if (profile_timestamp_ms_ == 0) {
>-    // Profiling has not been activated.
>-    return -1;
>-  }
>-  return rtc::TimeSince(profile_timestamp_ms_);
>-}
>-
>-inline const int16_t* AudioFrame::data() const {
>-  return muted_ ? empty_data() : data_;
>-}
>-
>-// TODO(henrik.lundin) Can we skip zeroing the buffer?
>-// See https://bugs.chromium.org/p/webrtc/issues/detail?id=5647.
>-inline int16_t* AudioFrame::mutable_data() {
>-  if (muted_) {
>-    memset(data_, 0, kMaxDataSizeBytes);
>-    muted_ = false;
>-  }
>-  return data_;
>-}
>-
>-inline void AudioFrame::Mute() {
>-  muted_ = true;
>-}
>-
>-inline bool AudioFrame::muted() const { return muted_; }
>-
>-inline AudioFrame& AudioFrame::operator>>=(const int rhs) {
>-  assert((num_channels_ > 0) && (num_channels_ < 3));
>-  if ((num_channels_ > 2) || (num_channels_ < 1)) return *this;
>-  if (muted_) return *this;
>-
>-  for (size_t i = 0; i < samples_per_channel_ * num_channels_; i++) {
>-    data_[i] = static_cast<int16_t>(data_[i] >> rhs);
>-  }
>-  return *this;
>-}
>-
>-inline AudioFrame& AudioFrame::operator+=(const AudioFrame& rhs) {
>-  // Sanity check
>-  assert((num_channels_ > 0) && (num_channels_ < 3));
>-  if ((num_channels_ > 2) || (num_channels_ < 1)) return *this;
>-  if (num_channels_ != rhs.num_channels_) return *this;
>-
>-  bool noPrevData = muted_;
>-  if (samples_per_channel_ != rhs.samples_per_channel_) {
>-    if (samples_per_channel_ == 0) {
>-      // special case we have no data to start with
>-      samples_per_channel_ = rhs.samples_per_channel_;
>-      noPrevData = true;
>-    } else {
>-      return *this;
>-    }
>-  }
>-
>-  if ((vad_activity_ == kVadActive) || rhs.vad_activity_ == kVadActive) {
>-    vad_activity_ = kVadActive;
>-  } else if (vad_activity_ == kVadUnknown || rhs.vad_activity_ == kVadUnknown) {
>-    vad_activity_ = kVadUnknown;
>-  }
>-
>-  if (speech_type_ != rhs.speech_type_) speech_type_ = kUndefined;
>-
>-  if (!rhs.muted()) {
>-    muted_ = false;
>-    if (noPrevData) {
>-      memcpy(data_, rhs.data(),
>-             sizeof(int16_t) * rhs.samples_per_channel_ * num_channels_);
>-    } else {
>-      // IMPROVEMENT this can be done very fast in assembly
>-      for (size_t i = 0; i < samples_per_channel_ * num_channels_; i++) {
>-        int32_t wrap_guard =
>-            static_cast<int32_t>(data_[i]) + static_cast<int32_t>(rhs.data_[i]);
>-        data_[i] = rtc::saturated_cast<int16_t>(wrap_guard);
>-      }
>-    }
>-  }
>-
>-  return *this;
>-}
>-
>-struct PacedPacketInfo {
>-  PacedPacketInfo() {}
>-  PacedPacketInfo(int probe_cluster_id,
>-                  int probe_cluster_min_probes,
>-                  int probe_cluster_min_bytes)
>-      : probe_cluster_id(probe_cluster_id),
>-        probe_cluster_min_probes(probe_cluster_min_probes),
>-        probe_cluster_min_bytes(probe_cluster_min_bytes) {}
>-
>-  bool operator==(const PacedPacketInfo& rhs) const {
>-    return send_bitrate_bps == rhs.send_bitrate_bps &&
>-           probe_cluster_id == rhs.probe_cluster_id &&
>-           probe_cluster_min_probes == rhs.probe_cluster_min_probes &&
>-           probe_cluster_min_bytes == rhs.probe_cluster_min_bytes;
>-  }
>+// Interface used by NackModule and JitterBuffer.
>+class KeyFrameRequestSender {
>+ public:
>+  virtual void RequestKeyFrame() = 0;
> 
>-  static constexpr int kNotAProbe = -1;
>-  int send_bitrate_bps = -1;
>-  int probe_cluster_id = kNotAProbe;
>-  int probe_cluster_min_probes = -1;
>-  int probe_cluster_min_bytes = -1;
>+ protected:
>+  virtual ~KeyFrameRequestSender() {}
> };
> 
>+// Used to indicate if a received packet contain a complete NALU (or equivalent)
>+enum VCMNaluCompleteness {
>+  kNaluUnset = 0,     // Packet has not been filled.
>+  kNaluComplete = 1,  // Packet can be decoded as is.
>+  kNaluStart,         // Packet contain beginning of NALU
>+  kNaluIncomplete,    // Packet is not beginning or end of NALU
>+  kNaluEnd,           // Packet is the end of a NALU
>+};
> }  // namespace webrtc
> 
> #endif  // MODULES_INCLUDE_MODULE_COMMON_TYPES_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/include/module_common_types_public.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/include/module_common_types_public.h
>index 2fbb49ae9c9..345e45ce127 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/include/module_common_types_public.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/include/module_common_types_public.h
>@@ -13,8 +13,7 @@
> 
> #include <limits>
> 
>-#include "api/optional.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include "absl/types/optional.h"
> 
> namespace webrtc {
> 
>@@ -68,9 +67,7 @@ class Unwrapper {
>   }
> 
>   // Only update the internal state to the specified last (unwrapped) value.
>-  void UpdateLast(int64_t last_value) {
>-    last_value_ = last_value;
>-  }
>+  void UpdateLast(int64_t last_value) { last_value_ = last_value; }
> 
>   // Unwrap the value and update the internal state.
>   int64_t Unwrap(U value) {
>@@ -80,17 +77,21 @@ class Unwrapper {
>   }
> 
>  private:
>-  rtc::Optional<int64_t> last_value_;
>+  absl::optional<int64_t> last_value_;
> };
> 
> using SequenceNumberUnwrapper = Unwrapper<uint16_t>;
> using TimestampUnwrapper = Unwrapper<uint32_t>;
> 
>+// NB: Doesn't fulfill strict weak ordering requirements.
>+//     Mustn't be used as std::map Compare function.
> inline bool IsNewerSequenceNumber(uint16_t sequence_number,
>                                   uint16_t prev_sequence_number) {
>   return IsNewer(sequence_number, prev_sequence_number);
> }
> 
>+// NB: Doesn't fulfill strict weak ordering requirements.
>+//     Mustn't be used as std::map Compare function.
> inline bool IsNewerTimestamp(uint32_t timestamp, uint32_t prev_timestamp) {
>   return IsNewer(timestamp, prev_timestamp);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/include/module_fec_types.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/include/module_fec_types.h
>new file mode 100644
>index 00000000000..25d6bc5714d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/include/module_fec_types.h
>@@ -0,0 +1,34 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_INCLUDE_MODULE_FEC_TYPES_H_
>+#define MODULES_INCLUDE_MODULE_FEC_TYPES_H_
>+
>+namespace webrtc {
>+
>+// Types for the FEC packet masks. The type |kFecMaskRandom| is based on a
>+// random loss model. The type |kFecMaskBursty| is based on a bursty/consecutive
>+// loss model. The packet masks are defined in
>+// modules/rtp_rtcp/fec_private_tables_random(bursty).h
>+enum FecMaskType {
>+  kFecMaskRandom,
>+  kFecMaskBursty,
>+};
>+
>+// Struct containing forward error correction settings.
>+struct FecProtectionParams {
>+  int fec_rate;
>+  int max_fec_frames;
>+  FecMaskType fec_mask_type;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_INCLUDE_MODULE_FEC_TYPES_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/media_file/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/media_file/BUILD.gn
>deleted file mode 100644
>index 4fff59de246..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/media_file/BUILD.gn
>+++ /dev/null
>@@ -1,67 +0,0 @@
>-# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
>-#
>-# Use of this source code is governed by a BSD-style license
>-# that can be found in the LICENSE file in the root of the source
>-# tree. An additional intellectual property rights grant can be found
>-# in the file PATENTS.  All contributing project authors may
>-# be found in the AUTHORS file in the root of the source tree.
>-
>-import("../../webrtc.gni")
>-
>-config("media_file_config") {
>-  visibility = [ ":*" ]  # Only targets in this file can depend on this.
>-}
>-
>-rtc_static_library("media_file") {
>-  visibility = [ "*" ]
>-  sources = [
>-    "media_file.h",
>-    "media_file_defines.h",
>-    "media_file_impl.cc",
>-    "media_file_impl.h",
>-    "media_file_utility.cc",
>-    "media_file_utility.h",
>-  ]
>-
>-  public_configs = [ ":media_file_config" ]
>-
>-  # TODO(jschuh): Bug 1348: fix this warning.
>-  configs += [ "//build/config/compiler:no_size_t_to_int_warning" ]
>-
>-  if (!build_with_chromium && is_clang) {
>-    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>-    suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-  }
>-
>-  deps = [
>-    "..:module_api",
>-    "../..:webrtc_common",
>-    "../../:typedefs",
>-    "../../common_audio",
>-    "../../rtc_base:rtc_base_approved",
>-  ]
>-}
>-
>-if (rtc_include_tests) {
>-  rtc_source_set("media_file_unittests") {
>-    testonly = true
>-
>-    sources = [
>-      "media_file_unittest.cc",
>-    ]
>-    deps = [
>-      ":media_file",
>-      "../../test:test_support",
>-    ]
>-    if (is_win) {
>-      cflags = [
>-        # TODO(kjellander): bugs.webrtc.org/261: Fix this warning.
>-        "/wd4373",  # virtual function override.
>-      ]
>-    }
>-    if (!build_with_chromium && is_clang) {
>-      # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>-      suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-    }
>-  }
>-}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/media_file/DEPS b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/media_file/DEPS
>deleted file mode 100644
>index dc17ef44fe7..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/media_file/DEPS
>+++ /dev/null
>@@ -1,4 +0,0 @@
>-include_rules = [
>-  "+common_audio",
>-  "+system_wrappers",
>-]
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/media_file/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/media_file/OWNERS
>deleted file mode 100644
>index f5e93f01381..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/media_file/OWNERS
>+++ /dev/null
>@@ -1,8 +0,0 @@
>-mflodman@webrtc.org
>-perkj@webrtc.org
>-niklas.enbom@webrtc.org
>-
>-# These are for the common case of adding or renaming files. If you're doing
>-# structural changes, please get a review from a reviewer in this file.
>-per-file *.gn=*
>-per-file *.gni=*
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/media_file/media_file.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/media_file/media_file.h
>deleted file mode 100644
>index d27e00a3af1..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/media_file/media_file.h
>+++ /dev/null
>@@ -1,84 +0,0 @@
>-/*
>- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MODULES_MEDIA_FILE_MEDIA_FILE_H_
>-#define MODULES_MEDIA_FILE_MEDIA_FILE_H_
>-
>-#include "common_types.h"  // NOLINT(build/include)
>-#include "modules/include/module.h"
>-#include "modules/include/module_common_types.h"
>-#include "modules/media_file/media_file_defines.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>-
>-namespace webrtc {
>-class MediaFile : public Module
>-{
>-public:
>-    // Factory method. Constructor disabled. id is the identifier for the
>-    // MediaFile instance.
>-    static MediaFile* CreateMediaFile(const int32_t id);
>-    static void DestroyMediaFile(MediaFile* module);
>-
>-    // Put 10-60ms of audio data from file into the audioBuffer depending on
>-    // codec frame size. dataLengthInBytes is both an input and output
>-    // parameter. As input parameter it indicates the size of audioBuffer.
>-    // As output parameter it indicates the number of bytes written to
>-    // audioBuffer.
>-    // Note: This API only play mono audio but can be used on file containing
>-    // audio with more channels (in which case the audio will be converted to
>-    // mono).
>-    virtual int32_t PlayoutAudioData(
>-        int8_t* audioBuffer,
>-        size_t& dataLengthInBytes) = 0;
>-
>-    // Prepare for playing audio from stream.
>-    // FileCallback::PlayNotification(..) will be called after
>-    // notificationTimeMs of the file has been played if notificationTimeMs is
>-    // greater than zero. format specifies the type of file fileName refers to.
>-    // codecInst specifies the encoding of the audio data. Note that
>-    // file formats that contain this information (like WAV files) don't need to
>-    // provide a non-NULL codecInst. startPointMs and stopPointMs, unless zero,
>-    // specify what part of the file should be read. From startPointMs ms to
>-    // stopPointMs ms.
>-    // Note: codecInst.channels should be set to 2 for stereo (and 1 for
>-    // mono). Stereo audio is only supported for WAV files.
>-    virtual int32_t StartPlayingAudioStream(
>-        InStream& stream,
>-        const uint32_t notificationTimeMs = 0,
>-        const FileFormats    format             = kFileFormatPcm16kHzFile,
>-        const CodecInst*     codecInst          = NULL,
>-        const uint32_t startPointMs       = 0,
>-        const uint32_t stopPointMs        = 0) = 0;
>-
>-    // Stop playing from file or stream.
>-    virtual int32_t StopPlaying() = 0;
>-
>-    // Return true if playing.
>-    virtual bool IsPlaying() = 0;
>-
>-
>-    // Set durationMs to the number of ms that has been played from file.
>-    virtual int32_t PlayoutPositionMs(
>-        uint32_t& durationMs) const = 0;
>-
>-    // Register callback to receive media file related notifications. Disables
>-    // callbacks if callback is NULL.
>-    virtual int32_t SetModuleFileCallback(FileCallback* callback) = 0;
>-
>-    // Update codecInst according to the current audio codec being used for
>-    // reading or writing.
>-    virtual int32_t codec_info(CodecInst& codecInst) const = 0;
>-
>-protected:
>-    MediaFile() {}
>-    virtual ~MediaFile() {}
>-};
>-}  // namespace webrtc
>-#endif // MODULES_MEDIA_FILE_MEDIA_FILE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/media_file/media_file_defines.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/media_file/media_file_defines.h
>deleted file mode 100644
>index 73b3b059ce1..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/media_file/media_file_defines.h
>+++ /dev/null
>@@ -1,50 +0,0 @@
>-/*
>- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MODULES_MEDIA_FILE_MEDIA_FILE_DEFINES_H_
>-#define MODULES_MEDIA_FILE_MEDIA_FILE_DEFINES_H_
>-
>-#include "modules/include/module_common_types.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>-
>-namespace webrtc {
>-// Callback class for the MediaFile class.
>-class FileCallback
>-{
>-public:
>-    virtual ~FileCallback(){}
>-
>-    // This function is called by MediaFile when a file has been playing for
>-    // durationMs ms. id is the identifier for the MediaFile instance calling
>-    // the callback.
>-    virtual void PlayNotification(const int32_t id,
>-                                  const uint32_t durationMs) = 0;
>-
>-    // This function is called by MediaFile when a file has been recording for
>-    // durationMs ms. id is the identifier for the MediaFile instance calling
>-    // the callback.
>-    virtual void RecordNotification(const int32_t id,
>-                                    const uint32_t durationMs) = 0;
>-
>-    // This function is called by MediaFile when a file has been stopped
>-    // playing. id is the identifier for the MediaFile instance calling the
>-    // callback.
>-    virtual void PlayFileEnded(const int32_t id) = 0;
>-
>-    // This function is called by MediaFile when a file has been stopped
>-    // recording. id is the identifier for the MediaFile instance calling the
>-    // callback.
>-    virtual void RecordFileEnded(const int32_t id) = 0;
>-
>-protected:
>-    FileCallback() {}
>-};
>-}  // namespace webrtc
>-#endif // MODULES_MEDIA_FILE_MEDIA_FILE_DEFINES_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/media_file/media_file_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/media_file/media_file_impl.cc
>deleted file mode 100644
>index 28271396575..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/media_file/media_file_impl.cc
>+++ /dev/null
>@@ -1,356 +0,0 @@
>-/*
>- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include <assert.h>
>-
>-#include "modules/media_file/media_file_impl.h"
>-#include "rtc_base/format_macros.h"
>-#include "rtc_base/logging.h"
>-
>-namespace webrtc {
>-MediaFile* MediaFile::CreateMediaFile(const int32_t id) {
>-  return new MediaFileImpl(id);
>-}
>-
>-void MediaFile::DestroyMediaFile(MediaFile* module) {
>-  delete static_cast<MediaFileImpl*>(module);
>-}
>-
>-MediaFileImpl::MediaFileImpl(const int32_t id)
>-    : _id(id),
>-      _ptrFileUtilityObj(NULL),
>-      codec_info_(),
>-      _ptrInStream(NULL),
>-      _fileFormat((FileFormats)-1),
>-      _playoutPositionMs(0),
>-      _notificationMs(0),
>-      _playingActive(false),
>-      _fileName(),
>-      _ptrCallback(NULL) {
>-  RTC_LOG(LS_INFO) << "MediaFileImpl()";
>-
>-  codec_info_.plname[0] = '\0';
>-  _fileName[0] = '\0';
>-}
>-
>-MediaFileImpl::~MediaFileImpl() {
>-  RTC_LOG(LS_INFO) << "~MediaFileImpl()";
>-  {
>-    rtc::CritScope lock(&_crit);
>-
>-    if (_playingActive) {
>-      StopPlaying();
>-    }
>-
>-    delete _ptrFileUtilityObj;
>-  }
>-}
>-
>-int64_t MediaFileImpl::TimeUntilNextProcess() {
>-  RTC_LOG(LS_WARNING)
>-      << "TimeUntilNextProcess: This method is not used by MediaFile class.";
>-  return -1;
>-}
>-
>-void MediaFileImpl::Process() {
>-  RTC_LOG(LS_WARNING) << "Process: This method is not used by MediaFile class.";
>-}
>-
>-int32_t MediaFileImpl::PlayoutAudioData(int8_t* buffer,
>-                                        size_t& dataLengthInBytes) {
>-  RTC_LOG(LS_INFO) << "MediaFileImpl::PlayoutData(buffer= "
>-                   << static_cast<void*>(buffer)
>-                   << ", bufLen= " << dataLengthInBytes << ")";
>-
>-  const size_t bufferLengthInBytes = dataLengthInBytes;
>-  dataLengthInBytes = 0;
>-
>-  if (buffer == NULL || bufferLengthInBytes == 0) {
>-    RTC_LOG(LS_ERROR) << "Buffer pointer or length is NULL!";
>-    return -1;
>-  }
>-
>-  int32_t bytesRead = 0;
>-  {
>-    rtc::CritScope lock(&_crit);
>-
>-    if (!_playingActive) {
>-      RTC_LOG(LS_WARNING) << "Not currently playing!";
>-      return -1;
>-    }
>-
>-    if (!_ptrFileUtilityObj) {
>-      RTC_LOG(LS_ERROR) << "Playing, but no FileUtility object!";
>-      StopPlaying();
>-      return -1;
>-    }
>-
>-    switch (_fileFormat) {
>-      case kFileFormatPcm32kHzFile:
>-      case kFileFormatPcm16kHzFile:
>-      case kFileFormatPcm8kHzFile:
>-        bytesRead = _ptrFileUtilityObj->ReadPCMData(*_ptrInStream, buffer,
>-                                                    bufferLengthInBytes);
>-        break;
>-      case kFileFormatWavFile:
>-        bytesRead = _ptrFileUtilityObj->ReadWavDataAsMono(*_ptrInStream, buffer,
>-                                                          bufferLengthInBytes);
>-        break;
>-      default: {
>-        RTC_LOG(LS_ERROR) << "Invalid file format: " << _fileFormat;
>-        assert(false);
>-        break;
>-      }
>-    }
>-
>-    if (bytesRead > 0) {
>-      dataLengthInBytes = static_cast<size_t>(bytesRead);
>-    }
>-  }
>-  HandlePlayCallbacks(bytesRead);
>-  return 0;
>-}
>-
>-void MediaFileImpl::HandlePlayCallbacks(int32_t bytesRead) {
>-  bool playEnded = false;
>-  uint32_t callbackNotifyMs = 0;
>-
>-  if (bytesRead > 0) {
>-    // Check if it's time for PlayNotification(..).
>-    _playoutPositionMs = _ptrFileUtilityObj->PlayoutPositionMs();
>-    if (_notificationMs) {
>-      if (_playoutPositionMs >= _notificationMs) {
>-        _notificationMs = 0;
>-        callbackNotifyMs = _playoutPositionMs;
>-      }
>-    }
>-  } else {
>-    // If no bytes were read assume end of file.
>-    StopPlaying();
>-    playEnded = true;
>-  }
>-
>-  // Only _callbackCrit may and should be taken when making callbacks.
>-  rtc::CritScope lock(&_callbackCrit);
>-  if (_ptrCallback) {
>-    if (callbackNotifyMs) {
>-      _ptrCallback->PlayNotification(_id, callbackNotifyMs);
>-    }
>-    if (playEnded) {
>-      _ptrCallback->PlayFileEnded(_id);
>-    }
>-  }
>-}
>-
>-int32_t MediaFileImpl::StartPlayingAudioStream(
>-    InStream& stream,
>-    const uint32_t notificationTimeMs,
>-    const FileFormats format,
>-    const CodecInst* codecInst,
>-    const uint32_t startPointMs,
>-    const uint32_t stopPointMs) {
>-  return StartPlayingStream(stream, false, notificationTimeMs, format,
>-                            codecInst, startPointMs, stopPointMs);
>-}
>-
>-int32_t MediaFileImpl::StartPlayingStream(InStream& stream,
>-                                          bool loop,
>-                                          const uint32_t notificationTimeMs,
>-                                          const FileFormats format,
>-                                          const CodecInst* codecInst,
>-                                          const uint32_t startPointMs,
>-                                          const uint32_t stopPointMs) {
>-  if (!ValidFileFormat(format, codecInst)) {
>-    return -1;
>-  }
>-
>-  if (!ValidFilePositions(startPointMs, stopPointMs)) {
>-    return -1;
>-  }
>-
>-  rtc::CritScope lock(&_crit);
>-  if (_playingActive) {
>-    RTC_LOG(LS_ERROR)
>-        << "StartPlaying called, but already playing file "
>-        << ((_fileName[0] == '\0') ? "(name not set)" : _fileName);
>-    return -1;
>-  }
>-
>-  if (_ptrFileUtilityObj != NULL) {
>-    RTC_LOG(LS_ERROR)
>-        << "StartPlaying called, but FileUtilityObj already exists!";
>-    StopPlaying();
>-    return -1;
>-  }
>-
>-  _ptrFileUtilityObj = new ModuleFileUtility();
>-  if (_ptrFileUtilityObj == NULL) {
>-    RTC_LOG(LS_INFO) << "Failed to create FileUtilityObj!";
>-    return -1;
>-  }
>-
>-  switch (format) {
>-    case kFileFormatWavFile: {
>-      if (_ptrFileUtilityObj->InitWavReading(stream, startPointMs,
>-                                             stopPointMs) == -1) {
>-        RTC_LOG(LS_ERROR) << "Not a valid WAV file!";
>-        StopPlaying();
>-        return -1;
>-      }
>-      _fileFormat = kFileFormatWavFile;
>-      break;
>-    }
>-    case kFileFormatPcm8kHzFile:
>-    case kFileFormatPcm16kHzFile:
>-    case kFileFormatPcm32kHzFile: {
>-      // ValidFileFormat() called in the beginneing of this function
>-      // prevents codecInst from being NULL here.
>-      assert(codecInst != NULL);
>-      if (!ValidFrequency(codecInst->plfreq) ||
>-          _ptrFileUtilityObj->InitPCMReading(stream, startPointMs, stopPointMs,
>-                                             codecInst->plfreq) == -1) {
>-        RTC_LOG(LS_ERROR) << "Not a valid raw 8 or 16 KHz PCM file!";
>-        StopPlaying();
>-        return -1;
>-      }
>-
>-      _fileFormat = format;
>-      break;
>-    }
>-    default: {
>-      RTC_LOG(LS_ERROR) << "Invalid file format: " << format;
>-      assert(false);
>-      break;
>-    }
>-  }
>-  if (_ptrFileUtilityObj->codec_info(codec_info_) == -1) {
>-    RTC_LOG(LS_ERROR) << "Failed to retrieve codec info!";
>-    StopPlaying();
>-    return -1;
>-  }
>-
>-  if ((codec_info_.channels == 2) && (_fileFormat != kFileFormatWavFile)) {
>-    RTC_LOG(LS_WARNING) << "Stereo is only allowed for WAV files";
>-    StopPlaying();
>-    return -1;
>-  }
>-  _playingActive = true;
>-  _playoutPositionMs = _ptrFileUtilityObj->PlayoutPositionMs();
>-  _ptrInStream = &stream;
>-  _notificationMs = notificationTimeMs;
>-
>-  return 0;
>-}
>-
>-int32_t MediaFileImpl::StopPlaying() {
>-  rtc::CritScope lock(&_crit);
>-  if (_ptrFileUtilityObj) {
>-    delete _ptrFileUtilityObj;
>-    _ptrFileUtilityObj = NULL;
>-  }
>-  if (_ptrInStream) {
>-    _ptrInStream = NULL;
>-  }
>-
>-  codec_info_.pltype = 0;
>-  codec_info_.plname[0] = '\0';
>-
>-  if (!_playingActive) {
>-    RTC_LOG(LS_WARNING) << "playing is not active!";
>-    return -1;
>-  }
>-
>-  _playingActive = false;
>-  return 0;
>-}
>-
>-bool MediaFileImpl::IsPlaying() {
>-  RTC_LOG(LS_VERBOSE) << "MediaFileImpl::IsPlaying()";
>-  rtc::CritScope lock(&_crit);
>-  return _playingActive;
>-}
>-
>-int32_t MediaFileImpl::SetModuleFileCallback(FileCallback* callback) {
>-  rtc::CritScope lock(&_callbackCrit);
>-
>-  _ptrCallback = callback;
>-  return 0;
>-}
>-
>-int32_t MediaFileImpl::PlayoutPositionMs(uint32_t& positionMs) const {
>-  rtc::CritScope lock(&_crit);
>-  if (!_playingActive) {
>-    positionMs = 0;
>-    return -1;
>-  }
>-  positionMs = _playoutPositionMs;
>-  return 0;
>-}
>-
>-int32_t MediaFileImpl::codec_info(CodecInst& codecInst) const {
>-  rtc::CritScope lock(&_crit);
>-  if (!_playingActive) {
>-    RTC_LOG(LS_ERROR) << "Playout has not been initialized!";
>-    return -1;
>-  }
>-  if (codec_info_.pltype == 0 && codec_info_.plname[0] == '\0') {
>-    RTC_LOG(LS_ERROR) << "The CodecInst for Playback is unknown!";
>-    return -1;
>-  }
>-  memcpy(&codecInst, &codec_info_, sizeof(CodecInst));
>-  return 0;
>-}
>-
>-bool MediaFileImpl::ValidFileFormat(const FileFormats format,
>-                                    const CodecInst* codecInst) {
>-  if (codecInst == NULL) {
>-    if (format == kFileFormatPcm8kHzFile || format == kFileFormatPcm16kHzFile ||
>-        format == kFileFormatPcm32kHzFile) {
>-      RTC_LOG(LS_ERROR) << "Codec info required for file format specified!";
>-      return false;
>-    }
>-  }
>-  return true;
>-}
>-
>-bool MediaFileImpl::ValidFileName(const char* fileName) {
>-  if ((fileName == NULL) || (fileName[0] == '\0')) {
>-    RTC_LOG(LS_ERROR) << "FileName not specified!";
>-    return false;
>-  }
>-  return true;
>-}
>-
>-bool MediaFileImpl::ValidFilePositions(const uint32_t startPointMs,
>-                                       const uint32_t stopPointMs) {
>-  if (startPointMs == 0 && stopPointMs == 0)  // Default values
>-  {
>-    return true;
>-  }
>-  if (stopPointMs && (startPointMs >= stopPointMs)) {
>-    RTC_LOG(LS_ERROR) << "startPointMs must be less than stopPointMs!";
>-    return false;
>-  }
>-  if (stopPointMs && ((stopPointMs - startPointMs) < 20)) {
>-    RTC_LOG(LS_ERROR) << "minimum play duration for files is 20 ms!";
>-    return false;
>-  }
>-  return true;
>-}
>-
>-bool MediaFileImpl::ValidFrequency(const uint32_t frequency) {
>-  if ((frequency == 8000) || (frequency == 16000) || (frequency == 32000) ||
>-      (frequency == 48000)) {
>-    return true;
>-  }
>-  RTC_LOG(LS_ERROR) << "Frequency should be 8000, 16000, 32000, or 48000 (Hz)";
>-  return false;
>-}
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/media_file/media_file_impl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/media_file/media_file_impl.h
>deleted file mode 100644
>index 40d9d71cfe0..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/media_file/media_file_impl.h
>+++ /dev/null
>@@ -1,102 +0,0 @@
>-/*
>- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MODULES_MEDIA_FILE_MEDIA_FILE_IMPL_H_
>-#define MODULES_MEDIA_FILE_MEDIA_FILE_IMPL_H_
>-
>-#include "common_types.h"  // NOLINT(build/include)
>-#include "modules/include/module_common_types.h"
>-#include "modules/media_file/media_file.h"
>-#include "modules/media_file/media_file_defines.h"
>-#include "modules/media_file/media_file_utility.h"
>-#include "rtc_base/criticalsection.h"
>-
>-namespace webrtc {
>-class MediaFileImpl : public MediaFile
>-{
>-
>-public:
>-    MediaFileImpl(const int32_t id);
>-    ~MediaFileImpl();
>-
>-    void Process() override;
>-    int64_t TimeUntilNextProcess() override;
>-
>-    // MediaFile functions
>-    int32_t PlayoutAudioData(int8_t* audioBuffer,
>-                             size_t& dataLengthInBytes) override;
>-
>-    int32_t StartPlayingAudioStream(
>-        InStream& stream,
>-        const uint32_t notificationTimeMs = 0,
>-        const FileFormats format = kFileFormatPcm16kHzFile,
>-        const CodecInst* codecInst = NULL,
>-        const uint32_t startPointMs = 0,
>-        const uint32_t stopPointMs = 0) override;
>-
>-    int32_t StopPlaying() override;
>-
>-    bool IsPlaying() override;
>-
>-    int32_t PlayoutPositionMs(uint32_t& positionMs) const override;
>-
>-    int32_t SetModuleFileCallback(FileCallback* callback) override;
>-
>-    int32_t codec_info(CodecInst& codecInst) const override;
>-
>-private:
>-    // Returns true if the combination of format and codecInst is valid.
>-    static bool ValidFileFormat(const FileFormats format,
>-                                const CodecInst*  codecInst);
>-
>-
>-    // Returns true if the filename is valid
>-    static bool ValidFileName(const char* fileName);
>-
>-    // Returns true if the combination of startPointMs and stopPointMs is valid.
>-    static bool ValidFilePositions(const uint32_t startPointMs,
>-                                   const uint32_t stopPointMs);
>-
>-    // Returns true if frequencyInHz is a supported frequency.
>-    static bool ValidFrequency(const uint32_t frequencyInHz);
>-
>-    void HandlePlayCallbacks(int32_t bytesRead);
>-
>-    int32_t StartPlayingStream(
>-        InStream& stream,
>-        bool loop,
>-        const uint32_t notificationTimeMs,
>-        const FileFormats format,
>-        const CodecInst*  codecInst,
>-        const uint32_t startPointMs,
>-        const uint32_t stopPointMs);
>-
>-    int32_t _id;
>-    rtc::CriticalSection _crit;
>-    rtc::CriticalSection _callbackCrit;
>-
>-    ModuleFileUtility* _ptrFileUtilityObj;
>-    CodecInst codec_info_;
>-
>-    InStream*  _ptrInStream;
>-
>-    FileFormats _fileFormat;
>-    uint32_t _playoutPositionMs;
>-    uint32_t _notificationMs;
>-
>-    bool _playingActive;
>-
>-    char _fileName[512];
>-
>-    FileCallback* _ptrCallback;
>-};
>-}  // namespace webrtc
>-
>-#endif // MODULES_MEDIA_FILE_MEDIA_FILE_IMPL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/media_file/media_file_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/media_file/media_file_unittest.cc
>deleted file mode 100644
>index 28ce1e2c0c2..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/media_file/media_file_unittest.cc
>+++ /dev/null
>@@ -1,26 +0,0 @@
>-/*
>- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "modules/media_file/media_file.h"
>-#include "test/gtest.h"
>-
>-class MediaFileTest : public testing::Test {
>- protected:
>-  void SetUp() {
>-    // Use number 0 as the the identifier and pass to CreateMediaFile.
>-    media_file_ = webrtc::MediaFile::CreateMediaFile(0);
>-    ASSERT_TRUE(media_file_ != NULL);
>-  }
>-  void TearDown() {
>-    webrtc::MediaFile::DestroyMediaFile(media_file_);
>-    media_file_ = NULL;
>-  }
>-  webrtc::MediaFile* media_file_;
>-};
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/media_file/media_file_utility.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/media_file/media_file_utility.cc
>deleted file mode 100644
>index d8cc6d9440b..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/media_file/media_file_utility.cc
>+++ /dev/null
>@@ -1,629 +0,0 @@
>-/*
>- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "modules/media_file/media_file_utility.h"
>-
>-#include <assert.h>
>-#include <sys/stat.h>
>-#include <sys/types.h>
>-#include <limits>
>-
>-#include "common_audio/wav_header.h"
>-#include "common_types.h"  // NOLINT(build/include)
>-#include "modules/include/module_common_types.h"
>-#include "rtc_base/format_macros.h"
>-#include "rtc_base/logging.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>-
>-namespace {
>-
>-// First 16 bytes the WAVE header. ckID should be "RIFF", wave_ckID should be
>-// "WAVE" and ckSize is the chunk size (4 + n)
>-struct WAVE_RIFF_header {
>-  int8_t ckID[4];
>-  int32_t ckSize;
>-  int8_t wave_ckID[4];
>-};
>-
>-// First 8 byte of the format chunk. fmt_ckID should be "fmt ". fmt_ckSize is
>-// the chunk size (16, 18 or 40 byte)
>-struct WAVE_CHUNK_header {
>-  int8_t fmt_ckID[4];
>-  uint32_t fmt_ckSize;
>-};
>-}  // unnamed namespace
>-
>-namespace webrtc {
>-ModuleFileUtility::ModuleFileUtility()
>-    : _wavFormatObj(),
>-      _dataSize(0),
>-      _readSizeBytes(0),
>-      _stopPointInMs(0),
>-      _startPointInMs(0),
>-      _playoutPositionMs(0),
>-      codec_info_(),
>-      _codecId(kCodecNoCodec),
>-      _bytesPerSample(0),
>-      _readPos(0),
>-      _reading(false),
>-      _tempData() {
>-  RTC_LOG(LS_INFO) << "ModuleFileUtility::ModuleFileUtility()";
>-  memset(&codec_info_, 0, sizeof(CodecInst));
>-  codec_info_.pltype = -1;
>-}
>-
>-ModuleFileUtility::~ModuleFileUtility() {
>-  RTC_LOG(LS_INFO) << "ModuleFileUtility::~ModuleFileUtility()";
>-}
>-
>-int32_t ModuleFileUtility::ReadWavHeader(InStream& wav) {
>-  WAVE_RIFF_header RIFFheaderObj;
>-  WAVE_CHUNK_header CHUNKheaderObj;
>-  // TODO (hellner): tmpStr and tmpStr2 seems unnecessary here.
>-  char tmpStr[6] = "FOUR";
>-  unsigned char tmpStr2[4];
>-  size_t i;
>-  bool dataFound = false;
>-  bool fmtFound = false;
>-  int8_t dummyRead;
>-
>-  _dataSize = 0;
>-  int len = wav.Read(&RIFFheaderObj, sizeof(WAVE_RIFF_header));
>-  if (len != static_cast<int>(sizeof(WAVE_RIFF_header))) {
>-    RTC_LOG(LS_ERROR) << "Not a wave file (too short)";
>-    return -1;
>-  }
>-
>-  for (i = 0; i < 4; i++) {
>-    tmpStr[i] = RIFFheaderObj.ckID[i];
>-  }
>-  if (strcmp(tmpStr, "RIFF") != 0) {
>-    RTC_LOG(LS_ERROR) << "Not a wave file (does not have RIFF)";
>-    return -1;
>-  }
>-  for (i = 0; i < 4; i++) {
>-    tmpStr[i] = RIFFheaderObj.wave_ckID[i];
>-  }
>-  if (strcmp(tmpStr, "WAVE") != 0) {
>-    RTC_LOG(LS_ERROR) << "Not a wave file (does not have WAVE)";
>-    return -1;
>-  }
>-
>-  len = wav.Read(&CHUNKheaderObj, sizeof(WAVE_CHUNK_header));
>-
>-  // WAVE files are stored in little endian byte order. Make sure that the
>-  // data can be read on big endian as well.
>-  // TODO (hellner): little endian to system byte order should be done in
>-  //                 in a subroutine.
>-  memcpy(tmpStr2, &CHUNKheaderObj.fmt_ckSize, 4);
>-  CHUNKheaderObj.fmt_ckSize =
>-      (uint32_t)tmpStr2[0] + (((uint32_t)tmpStr2[1]) << 8) +
>-      (((uint32_t)tmpStr2[2]) << 16) + (((uint32_t)tmpStr2[3]) << 24);
>-
>-  memcpy(tmpStr, CHUNKheaderObj.fmt_ckID, 4);
>-
>-  while ((len == static_cast<int>(sizeof(WAVE_CHUNK_header))) &&
>-         (!fmtFound || !dataFound)) {
>-    if (strcmp(tmpStr, "fmt ") == 0) {
>-      len = wav.Read(&_wavFormatObj, sizeof(WAVE_FMTINFO_header));
>-
>-      memcpy(tmpStr2, &_wavFormatObj.formatTag, 2);
>-      _wavFormatObj.formatTag =
>-          (uint32_t)tmpStr2[0] + (((uint32_t)tmpStr2[1]) << 8);
>-      memcpy(tmpStr2, &_wavFormatObj.nChannels, 2);
>-      _wavFormatObj.nChannels =
>-          (int16_t)((uint32_t)tmpStr2[0] + (((uint32_t)tmpStr2[1]) << 8));
>-      memcpy(tmpStr2, &_wavFormatObj.nSamplesPerSec, 4);
>-      _wavFormatObj.nSamplesPerSec = (int32_t)(
>-          (uint32_t)tmpStr2[0] + (((uint32_t)tmpStr2[1]) << 8) +
>-          (((uint32_t)tmpStr2[2]) << 16) + (((uint32_t)tmpStr2[3]) << 24));
>-      memcpy(tmpStr2, &_wavFormatObj.nAvgBytesPerSec, 4);
>-      _wavFormatObj.nAvgBytesPerSec = (int32_t)(
>-          (uint32_t)tmpStr2[0] + (((uint32_t)tmpStr2[1]) << 8) +
>-          (((uint32_t)tmpStr2[2]) << 16) + (((uint32_t)tmpStr2[3]) << 24));
>-      memcpy(tmpStr2, &_wavFormatObj.nBlockAlign, 2);
>-      _wavFormatObj.nBlockAlign =
>-          (int16_t)((uint32_t)tmpStr2[0] + (((uint32_t)tmpStr2[1]) << 8));
>-      memcpy(tmpStr2, &_wavFormatObj.nBitsPerSample, 2);
>-      _wavFormatObj.nBitsPerSample =
>-          (int16_t)((uint32_t)tmpStr2[0] + (((uint32_t)tmpStr2[1]) << 8));
>-
>-      if (CHUNKheaderObj.fmt_ckSize < sizeof(WAVE_FMTINFO_header)) {
>-        RTC_LOG(LS_ERROR) << "Chunk size is too small";
>-        return -1;
>-      }
>-      for (i = 0; i < CHUNKheaderObj.fmt_ckSize - sizeof(WAVE_FMTINFO_header);
>-           i++) {
>-        len = wav.Read(&dummyRead, 1);
>-        if (len != 1) {
>-          RTC_LOG(LS_ERROR) << "File corrupted, reached EOF (reading fmt)";
>-          return -1;
>-        }
>-      }
>-      fmtFound = true;
>-    } else if (strcmp(tmpStr, "data") == 0) {
>-      _dataSize = CHUNKheaderObj.fmt_ckSize;
>-      dataFound = true;
>-      break;
>-    } else {
>-      for (i = 0; i < CHUNKheaderObj.fmt_ckSize; i++) {
>-        len = wav.Read(&dummyRead, 1);
>-        if (len != 1) {
>-          RTC_LOG(LS_ERROR) << "File corrupted, reached EOF (reading other)";
>-          return -1;
>-        }
>-      }
>-    }
>-
>-    len = wav.Read(&CHUNKheaderObj, sizeof(WAVE_CHUNK_header));
>-
>-    memcpy(tmpStr2, &CHUNKheaderObj.fmt_ckSize, 4);
>-    CHUNKheaderObj.fmt_ckSize =
>-        (uint32_t)tmpStr2[0] + (((uint32_t)tmpStr2[1]) << 8) +
>-        (((uint32_t)tmpStr2[2]) << 16) + (((uint32_t)tmpStr2[3]) << 24);
>-
>-    memcpy(tmpStr, CHUNKheaderObj.fmt_ckID, 4);
>-  }
>-
>-  // Either a proper format chunk has been read or a data chunk was come
>-  // across.
>-  if ((_wavFormatObj.formatTag != kWavFormatPcm) &&
>-      (_wavFormatObj.formatTag != kWavFormatALaw) &&
>-      (_wavFormatObj.formatTag != kWavFormatMuLaw)) {
>-    RTC_LOG(LS_ERROR) << "Coding formatTag value=" << _wavFormatObj.formatTag
>-                      << " not supported!";
>-    return -1;
>-  }
>-  if ((_wavFormatObj.nChannels < 1) || (_wavFormatObj.nChannels > 2)) {
>-    RTC_LOG(LS_ERROR) << "nChannels value=" << _wavFormatObj.nChannels
>-                      << " not supported!";
>-    return -1;
>-  }
>-
>-  if ((_wavFormatObj.nBitsPerSample != 8) &&
>-      (_wavFormatObj.nBitsPerSample != 16)) {
>-    RTC_LOG(LS_ERROR) << "nBitsPerSample value=" << _wavFormatObj.nBitsPerSample
>-                      << " not supported!";
>-    return -1;
>-  }
>-
>-  // Calculate the number of bytes that 10 ms of audio data correspond to.
>-  size_t samples_per_10ms =
>-      ((_wavFormatObj.formatTag == kWavFormatPcm) &&
>-       (_wavFormatObj.nSamplesPerSec == 44100))
>-          ? 440
>-          : static_cast<size_t>(_wavFormatObj.nSamplesPerSec / 100);
>-  _readSizeBytes = samples_per_10ms * _wavFormatObj.nChannels *
>-                   (_wavFormatObj.nBitsPerSample / 8);
>-  return 0;
>-}
>-
>-int32_t ModuleFileUtility::InitWavCodec(uint32_t samplesPerSec,
>-                                        size_t channels,
>-                                        uint32_t bitsPerSample,
>-                                        uint32_t formatTag) {
>-  codec_info_.pltype = -1;
>-  codec_info_.plfreq = samplesPerSec;
>-  codec_info_.channels = channels;
>-  codec_info_.rate = bitsPerSample * samplesPerSec;
>-
>-  // Calculate the packet size for 10ms frames
>-  switch (formatTag) {
>-    case kWavFormatALaw:
>-      strcpy(codec_info_.plname, "PCMA");
>-      _codecId = kCodecPcma;
>-      codec_info_.pltype = 8;
>-      codec_info_.pacsize = codec_info_.plfreq / 100;
>-      break;
>-    case kWavFormatMuLaw:
>-      strcpy(codec_info_.plname, "PCMU");
>-      _codecId = kCodecPcmu;
>-      codec_info_.pltype = 0;
>-      codec_info_.pacsize = codec_info_.plfreq / 100;
>-      break;
>-    case kWavFormatPcm:
>-      codec_info_.pacsize = (bitsPerSample * (codec_info_.plfreq / 100)) / 8;
>-      if (samplesPerSec == 8000) {
>-        strcpy(codec_info_.plname, "L16");
>-        _codecId = kCodecL16_8Khz;
>-      } else if (samplesPerSec == 16000) {
>-        strcpy(codec_info_.plname, "L16");
>-        _codecId = kCodecL16_16kHz;
>-      } else if (samplesPerSec == 32000) {
>-        strcpy(codec_info_.plname, "L16");
>-        _codecId = kCodecL16_32Khz;
>-      }
>-      // Set the packet size for "odd" sampling frequencies so that it
>-      // properly corresponds to _readSizeBytes.
>-      else if (samplesPerSec == 11025) {
>-        strcpy(codec_info_.plname, "L16");
>-        _codecId = kCodecL16_16kHz;
>-        codec_info_.pacsize = 110;
>-        codec_info_.plfreq = 11000;
>-      } else if (samplesPerSec == 22050) {
>-        strcpy(codec_info_.plname, "L16");
>-        _codecId = kCodecL16_16kHz;
>-        codec_info_.pacsize = 220;
>-        codec_info_.plfreq = 22000;
>-      } else if (samplesPerSec == 44100) {
>-        strcpy(codec_info_.plname, "L16");
>-        _codecId = kCodecL16_16kHz;
>-        codec_info_.pacsize = 440;
>-        codec_info_.plfreq = 44000;
>-      } else if (samplesPerSec == 48000) {
>-        strcpy(codec_info_.plname, "L16");
>-        _codecId = kCodecL16_16kHz;
>-        codec_info_.pacsize = 480;
>-        codec_info_.plfreq = 48000;
>-      } else {
>-        RTC_LOG(LS_ERROR) << "Unsupported PCM frequency!";
>-        return -1;
>-      }
>-      break;
>-    default:
>-      RTC_LOG(LS_ERROR) << "unknown WAV format TAG!";
>-      return -1;
>-      break;
>-  }
>-  return 0;
>-}
>-
>-int32_t ModuleFileUtility::InitWavReading(InStream& wav,
>-                                          const uint32_t start,
>-                                          const uint32_t stop) {
>-  _reading = false;
>-
>-  if (ReadWavHeader(wav) == -1) {
>-    RTC_LOG(LS_ERROR) << "failed to read WAV header!";
>-    return -1;
>-  }
>-
>-  _playoutPositionMs = 0;
>-  _readPos = 0;
>-
>-  if (start > 0) {
>-    uint8_t dummy[WAV_MAX_BUFFER_SIZE];
>-    int readLength;
>-    if (_readSizeBytes <= WAV_MAX_BUFFER_SIZE) {
>-      while (_playoutPositionMs < start) {
>-        readLength = wav.Read(dummy, _readSizeBytes);
>-        if (readLength == static_cast<int>(_readSizeBytes)) {
>-          _readPos += _readSizeBytes;
>-          _playoutPositionMs += 10;
>-        } else  // Must have reached EOF before start position!
>-        {
>-          RTC_LOG(LS_ERROR) << "InitWavReading(), EOF before start position";
>-          return -1;
>-        }
>-      }
>-    } else {
>-      return -1;
>-    }
>-  }
>-  if (InitWavCodec(_wavFormatObj.nSamplesPerSec, _wavFormatObj.nChannels,
>-                   _wavFormatObj.nBitsPerSample,
>-                   _wavFormatObj.formatTag) != 0) {
>-    return -1;
>-  }
>-  _bytesPerSample = static_cast<size_t>(_wavFormatObj.nBitsPerSample / 8);
>-
>-  _startPointInMs = start;
>-  _stopPointInMs = stop;
>-  _reading = true;
>-  return 0;
>-}
>-
>-int32_t ModuleFileUtility::ReadWavDataAsMono(InStream& wav,
>-                                             int8_t* outData,
>-                                             const size_t bufferSize) {
>-  RTC_LOG(LS_VERBOSE) << "ModuleFileUtility::ReadWavDataAsMono(wav= " << &wav
>-                      << ", outData= " << static_cast<void*>(outData)
>-                      << ", bufSize= " << bufferSize << ")";
>-
>-  // The number of bytes that should be read from file.
>-  const size_t totalBytesNeeded = _readSizeBytes;
>-  // The number of bytes that will be written to outData.
>-  const size_t bytesRequested =
>-      (codec_info_.channels == 2) ? totalBytesNeeded >> 1 : totalBytesNeeded;
>-  if (bufferSize < bytesRequested) {
>-    RTC_LOG(LS_ERROR) << "ReadWavDataAsMono: output buffer is too short!";
>-    return -1;
>-  }
>-  if (outData == NULL) {
>-    RTC_LOG(LS_ERROR) << "ReadWavDataAsMono: output buffer NULL!";
>-    return -1;
>-  }
>-
>-  if (!_reading) {
>-    RTC_LOG(LS_ERROR) << "ReadWavDataAsMono: no longer reading file.";
>-    return -1;
>-  }
>-
>-  int32_t bytesRead = ReadWavData(
>-      wav, (codec_info_.channels == 2) ? _tempData : (uint8_t*)outData,
>-      totalBytesNeeded);
>-  if (bytesRead == 0) {
>-    return 0;
>-  }
>-  if (bytesRead < 0) {
>-    RTC_LOG(LS_ERROR)
>-        << "ReadWavDataAsMono: failed to read data from WAV file.";
>-    return -1;
>-  }
>-  // Output data is should be mono.
>-  if (codec_info_.channels == 2) {
>-    for (size_t i = 0; i < bytesRequested / _bytesPerSample; i++) {
>-      // Sample value is the average of left and right buffer rounded to
>-      // closest integer value. Note samples can be either 1 or 2 byte.
>-      if (_bytesPerSample == 1) {
>-        _tempData[i] = ((_tempData[2 * i] + _tempData[(2 * i) + 1] + 1) >> 1);
>-      } else {
>-        int16_t* sampleData = (int16_t*)_tempData;
>-        sampleData[i] =
>-            ((sampleData[2 * i] + sampleData[(2 * i) + 1] + 1) >> 1);
>-      }
>-    }
>-    memcpy(outData, _tempData, bytesRequested);
>-  }
>-  return static_cast<int32_t>(bytesRequested);
>-}
>-
>-int32_t ModuleFileUtility::ReadWavData(InStream& wav,
>-                                       uint8_t* buffer,
>-                                       size_t dataLengthInBytes) {
>-  RTC_LOG(LS_VERBOSE) << "ModuleFileUtility::ReadWavData(wav= " << &wav
>-                      << ", buffer= " << static_cast<void*>(buffer)
>-                      << ", dataLen= " << dataLengthInBytes << ")";
>-
>-  if (buffer == NULL) {
>-    RTC_LOG(LS_ERROR) << "ReadWavDataAsMono: output buffer NULL!";
>-    return -1;
>-  }
>-
>-  // Make sure that a read won't return too few samples.
>-  // TODO (hellner): why not read the remaining bytes needed from the start
>-  //                 of the file?
>-  if (_dataSize < (_readPos + dataLengthInBytes)) {
>-    // Rewind() being -1 may be due to the file not supposed to be looped.
>-    if (wav.Rewind() == -1) {
>-      _reading = false;
>-      return 0;
>-    }
>-    if (InitWavReading(wav, _startPointInMs, _stopPointInMs) == -1) {
>-      _reading = false;
>-      return -1;
>-    }
>-  }
>-
>-  int32_t bytesRead = wav.Read(buffer, dataLengthInBytes);
>-  if (bytesRead < 0) {
>-    _reading = false;
>-    return -1;
>-  }
>-
>-  // This should never happen due to earlier sanity checks.
>-  // TODO (hellner): change to an assert and fail here since this should
>-  //                 never happen...
>-  if (bytesRead < (int32_t)dataLengthInBytes) {
>-    if ((wav.Rewind() == -1) ||
>-        (InitWavReading(wav, _startPointInMs, _stopPointInMs) == -1)) {
>-      _reading = false;
>-      return -1;
>-    } else {
>-      bytesRead = wav.Read(buffer, dataLengthInBytes);
>-      if (bytesRead < (int32_t)dataLengthInBytes) {
>-        _reading = false;
>-        return -1;
>-      }
>-    }
>-  }
>-
>-  _readPos += bytesRead;
>-
>-  // TODO (hellner): Why is dataLengthInBytes let dictate the number of bytes
>-  //                 to read when exactly 10ms should be read?!
>-  _playoutPositionMs += 10;
>-  if ((_stopPointInMs > 0) && (_playoutPositionMs >= _stopPointInMs)) {
>-    if ((wav.Rewind() == -1) ||
>-        (InitWavReading(wav, _startPointInMs, _stopPointInMs) == -1)) {
>-      _reading = false;
>-    }
>-  }
>-  return bytesRead;
>-}
>-
>-int32_t ModuleFileUtility::InitPCMReading(InStream& pcm,
>-                                          const uint32_t start,
>-                                          const uint32_t stop,
>-                                          uint32_t freq) {
>-  RTC_LOG(LS_VERBOSE) << "ModuleFileUtility::InitPCMReading(pcm= " << &pcm
>-                      << ", start=" << start << ", stop=" << stop
>-                      << ", freq=" << freq << ")";
>-
>-  int8_t dummy[320];
>-  int read_len;
>-
>-  _playoutPositionMs = 0;
>-  _startPointInMs = start;
>-  _stopPointInMs = stop;
>-  _reading = false;
>-
>-  if (freq == 8000) {
>-    strcpy(codec_info_.plname, "L16");
>-    codec_info_.pltype = -1;
>-    codec_info_.plfreq = 8000;
>-    codec_info_.pacsize = 160;
>-    codec_info_.channels = 1;
>-    codec_info_.rate = 128000;
>-    _codecId = kCodecL16_8Khz;
>-  } else if (freq == 16000) {
>-    strcpy(codec_info_.plname, "L16");
>-    codec_info_.pltype = -1;
>-    codec_info_.plfreq = 16000;
>-    codec_info_.pacsize = 320;
>-    codec_info_.channels = 1;
>-    codec_info_.rate = 256000;
>-    _codecId = kCodecL16_16kHz;
>-  } else if (freq == 32000) {
>-    strcpy(codec_info_.plname, "L16");
>-    codec_info_.pltype = -1;
>-    codec_info_.plfreq = 32000;
>-    codec_info_.pacsize = 320;
>-    codec_info_.channels = 1;
>-    codec_info_.rate = 512000;
>-    _codecId = kCodecL16_32Khz;
>-  } else if (freq == 48000) {
>-    strcpy(codec_info_.plname, "L16");
>-    codec_info_.pltype = -1;
>-    codec_info_.plfreq = 48000;
>-    codec_info_.pacsize = 480;
>-    codec_info_.channels = 1;
>-    codec_info_.rate = 768000;
>-    _codecId = kCodecL16_48Khz;
>-  }
>-
>-  // Readsize for 10ms of audio data (2 bytes per sample).
>-  _readSizeBytes = 2 * codec_info_.plfreq / 100;
>-  if (_startPointInMs > 0) {
>-    while (_playoutPositionMs < _startPointInMs) {
>-      read_len = pcm.Read(dummy, _readSizeBytes);
>-      if (read_len != static_cast<int>(_readSizeBytes)) {
>-        return -1;  // Must have reached EOF before start position!
>-      }
>-      _playoutPositionMs += 10;
>-    }
>-  }
>-  _reading = true;
>-  return 0;
>-}
>-
>-int32_t ModuleFileUtility::ReadPCMData(InStream& pcm,
>-                                       int8_t* outData,
>-                                       size_t bufferSize) {
>-  RTC_LOG(LS_VERBOSE) << "ModuleFileUtility::ReadPCMData(pcm= " << &pcm
>-                      << ", outData= " << static_cast<void*>(outData)
>-                      << ", bufSize= " << bufferSize << ")";
>-
>-  if (outData == NULL) {
>-    RTC_LOG(LS_ERROR) << "buffer NULL";
>-  }
>-
>-  // Readsize for 10ms of audio data (2 bytes per sample).
>-  size_t bytesRequested = static_cast<size_t>(2 * codec_info_.plfreq / 100);
>-  if (bufferSize < bytesRequested) {
>-    RTC_LOG(LS_ERROR)
>-        << "ReadPCMData: buffer not long enough for a 10ms frame.";
>-    assert(false);
>-    return -1;
>-  }
>-
>-  int bytesRead = pcm.Read(outData, bytesRequested);
>-  if (bytesRead < static_cast<int>(bytesRequested)) {
>-    if (pcm.Rewind() == -1) {
>-      _reading = false;
>-    } else {
>-      if (InitPCMReading(pcm, _startPointInMs, _stopPointInMs,
>-                         codec_info_.plfreq) == -1) {
>-        _reading = false;
>-      } else {
>-        size_t rest = bytesRequested - bytesRead;
>-        int len = pcm.Read(&(outData[bytesRead]), rest);
>-        if (len == static_cast<int>(rest)) {
>-          bytesRead += len;
>-        } else {
>-          _reading = false;
>-        }
>-      }
>-      if (bytesRead <= 0) {
>-        RTC_LOG(LS_ERROR) << "ReadPCMData: Failed to rewind audio file.";
>-        return -1;
>-      }
>-    }
>-  }
>-
>-  if (bytesRead <= 0) {
>-    RTC_LOG(LS_VERBOSE) << "ReadPCMData: end of file";
>-    return -1;
>-  }
>-  _playoutPositionMs += 10;
>-  if (_stopPointInMs && _playoutPositionMs >= _stopPointInMs) {
>-    if (!pcm.Rewind()) {
>-      if (InitPCMReading(pcm, _startPointInMs, _stopPointInMs,
>-                         codec_info_.plfreq) == -1) {
>-        _reading = false;
>-      }
>-    }
>-  }
>-  return bytesRead;
>-}
>-
>-int32_t ModuleFileUtility::codec_info(CodecInst& codecInst) {
>-  RTC_LOG(LS_VERBOSE) << "ModuleFileUtility::codec_info(codecInst= "
>-                      << &codecInst << ")";
>-
>-  if (!_reading) {
>-    RTC_LOG(LS_ERROR) << "CodecInst: not currently reading audio file!";
>-    return -1;
>-  }
>-  memcpy(&codecInst, &codec_info_, sizeof(CodecInst));
>-  return 0;
>-}
>-
>-int32_t ModuleFileUtility::set_codec_info(const CodecInst& codecInst) {
>-  _codecId = kCodecNoCodec;
>-  if (STR_CASE_CMP(codecInst.plname, "PCMU") == 0) {
>-    _codecId = kCodecPcmu;
>-  } else if (STR_CASE_CMP(codecInst.plname, "PCMA") == 0) {
>-    _codecId = kCodecPcma;
>-  } else if (STR_CASE_CMP(codecInst.plname, "L16") == 0) {
>-    if (codecInst.plfreq == 8000) {
>-      _codecId = kCodecL16_8Khz;
>-    } else if (codecInst.plfreq == 16000) {
>-      _codecId = kCodecL16_16kHz;
>-    } else if (codecInst.plfreq == 32000) {
>-      _codecId = kCodecL16_32Khz;
>-    } else if (codecInst.plfreq == 48000) {
>-      _codecId = kCodecL16_48Khz;
>-    }
>-  }
>-#ifdef WEBRTC_CODEC_ILBC
>-  else if (STR_CASE_CMP(codecInst.plname, "ilbc") == 0) {
>-    if (codecInst.pacsize == 160) {
>-      _codecId = kCodecIlbc20Ms;
>-    } else if (codecInst.pacsize == 240) {
>-      _codecId = kCodecIlbc30Ms;
>-    }
>-  }
>-#endif
>-#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
>-  else if (STR_CASE_CMP(codecInst.plname, "isac") == 0) {
>-    if (codecInst.plfreq == 16000) {
>-      _codecId = kCodecIsac;
>-    } else if (codecInst.plfreq == 32000) {
>-      _codecId = kCodecIsacSwb;
>-    }
>-  }
>-#endif
>-  else if (STR_CASE_CMP(codecInst.plname, "G722") == 0) {
>-    _codecId = kCodecG722;
>-  }
>-  if (_codecId == kCodecNoCodec) {
>-    return -1;
>-  }
>-  memcpy(&codec_info_, &codecInst, sizeof(CodecInst));
>-  return 0;
>-}
>-
>-uint32_t ModuleFileUtility::PlayoutPositionMs() {
>-  RTC_LOG(LS_VERBOSE) << "ModuleFileUtility::PlayoutPosition()";
>-
>-  return _reading ? _playoutPositionMs : 0;
>-}
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/media_file/media_file_utility.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/media_file/media_file_utility.h
>deleted file mode 100644
>index 79436d7ee6c..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/media_file/media_file_utility.h
>+++ /dev/null
>@@ -1,158 +0,0 @@
>-/*
>- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-// Note: the class cannot be used for reading and writing at the same time.
>-#ifndef MODULES_MEDIA_FILE_MEDIA_FILE_UTILITY_H_
>-#define MODULES_MEDIA_FILE_MEDIA_FILE_UTILITY_H_
>-
>-#include <stdio.h>
>-
>-#include "common_types.h"  // NOLINT(build/include)
>-#include "modules/media_file/media_file_defines.h"
>-
>-namespace webrtc {
>-class InStream;
>-class OutStream;
>-
>-class ModuleFileUtility
>-{
>-public:
>-
>-    ModuleFileUtility();
>-    ~ModuleFileUtility();
>-
>-    // Prepare for playing audio from stream.
>-    // startPointMs and stopPointMs, unless zero, specify what part of the file
>-    // should be read. From startPointMs ms to stopPointMs ms.
>-    int32_t InitWavReading(InStream& stream,
>-                           const uint32_t startPointMs = 0,
>-                           const uint32_t stopPointMs = 0);
>-
>-    // Put 10-60ms of audio data from stream into the audioBuffer depending on
>-    // codec frame size. dataLengthInBytes indicates the size of audioBuffer.
>-    // The return value is the number of bytes written to audioBuffer.
>-    // Note: This API only play mono audio but can be used on file containing
>-    // audio with more channels (in which case the audio will be converted to
>-    // mono).
>-    int32_t ReadWavDataAsMono(InStream& stream, int8_t* audioBuffer,
>-                              const size_t dataLengthInBytes);
>-
>-    // Prepare for playing audio from stream.
>-    // startPointMs and stopPointMs, unless zero, specify what part of the file
>-    // should be read. From startPointMs ms to stopPointMs ms.
>-    // freqInHz is the PCM sampling frequency.
>-    // NOTE, allowed frequencies are 8000, 16000 and 32000 (Hz)
>-    int32_t InitPCMReading(InStream& stream,
>-                           const uint32_t startPointMs = 0,
>-                           const uint32_t stopPointMs = 0,
>-                           const uint32_t freqInHz = 16000);
>-
>-    // Put 10-60ms of audio data from stream into the audioBuffer depending on
>-    // codec frame size. dataLengthInBytes indicates the size of audioBuffer.
>-    // The return value is the number of bytes written to audioBuffer.
>-    int32_t ReadPCMData(InStream& stream, int8_t* audioBuffer,
>-                        const size_t dataLengthInBytes);
>-
>-    // Return the number of ms that have been played so far.
>-    uint32_t PlayoutPositionMs();
>-
>-    // Update codecInst according to the current audio codec being used for
>-    // reading or writing.
>-    int32_t codec_info(CodecInst& codecInst);
>-
>-private:
>-    // Biggest WAV frame supported is 10 ms at 48kHz of 2 channel, 16 bit audio.
>-    static const size_t WAV_MAX_BUFFER_SIZE = 480 * 2 * 2;
>-
>-
>-    int32_t InitWavCodec(uint32_t samplesPerSec,
>-                         size_t channels,
>-                         uint32_t bitsPerSample,
>-                         uint32_t formatTag);
>-
>-    // Parse the WAV header in stream.
>-    int32_t ReadWavHeader(InStream& stream);
>-
>-    // Put dataLengthInBytes of audio data from stream into the audioBuffer.
>-    // The return value is the number of bytes written to audioBuffer.
>-    int32_t ReadWavData(InStream& stream, uint8_t* audioBuffer,
>-                        size_t dataLengthInBytes);
>-
>-    // Update the current audio codec being used for reading or writing
>-    // according to codecInst.
>-    int32_t set_codec_info(const CodecInst& codecInst);
>-
>-    struct WAVE_FMTINFO_header
>-    {
>-        int16_t formatTag;
>-        int16_t nChannels;
>-        int32_t nSamplesPerSec;
>-        int32_t nAvgBytesPerSec;
>-        int16_t nBlockAlign;
>-        int16_t nBitsPerSample;
>-    };
>-    // Identifiers for preencoded files.
>-    enum MediaFileUtility_CodecType
>-    {
>-        kCodecNoCodec  = 0,
>-        kCodecIsac,
>-        kCodecIsacSwb,
>-        kCodecIsacLc,
>-        kCodecL16_8Khz,
>-        kCodecL16_16kHz,
>-        kCodecL16_32Khz,
>-        kCodecL16_48Khz,
>-        kCodecPcmu,
>-        kCodecPcma,
>-        kCodecIlbc20Ms,
>-        kCodecIlbc30Ms,
>-        kCodecG722,
>-        kCodecG722_1_32Kbps,
>-        kCodecG722_1_24Kbps,
>-        kCodecG722_1_16Kbps,
>-        kCodecG722_1c_48,
>-        kCodecG722_1c_32,
>-        kCodecG722_1c_24,
>-        kCodecAmr,
>-        kCodecAmrWb,
>-        kCodecG729,
>-        kCodecG729_1,
>-        kCodecG726_40,
>-        kCodecG726_32,
>-        kCodecG726_24,
>-        kCodecG726_16
>-    };
>-
>-    // TODO (hellner): why store multiple formats. Just store either codec_info_
>-    //                 or _wavFormatObj and supply conversion functions.
>-    WAVE_FMTINFO_header _wavFormatObj;
>-    size_t _dataSize;      // Chunk size if reading a WAV file
>-    // Number of bytes to read. I.e. frame size in bytes. May be multiple
>-    // chunks if reading WAV.
>-    size_t _readSizeBytes;
>-
>-    uint32_t _stopPointInMs;
>-    uint32_t _startPointInMs;
>-    uint32_t _playoutPositionMs;
>-
>-    CodecInst codec_info_;
>-    MediaFileUtility_CodecType _codecId;
>-
>-    // The amount of bytes, on average, used for one audio sample.
>-    size_t _bytesPerSample;
>-    size_t _readPos;
>-
>-    bool _reading;
>-
>-    // Scratch buffer used for turning stereo audio to mono.
>-    uint8_t _tempData[WAV_MAX_BUFFER_SIZE];
>-};
>-}  // namespace webrtc
>-#endif // MODULES_MEDIA_FILE_MEDIA_FILE_UTILITY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/module_common_types_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/module_common_types_unittest.cc
>index c8bb5f98509..8289bea17fd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/module_common_types_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/module_common_types_unittest.cc
>@@ -10,108 +10,10 @@
> 
> #include "modules/include/module_common_types.h"
> 
>-#include <string.h>  // memcmp
>-
> #include "test/gtest.h"
> 
> namespace webrtc {
> 
>-namespace {
>-
>-bool AllSamplesAre(int16_t sample, const AudioFrame& frame) {
>-  const int16_t* frame_data = frame.data();
>-  for (size_t i = 0; i < AudioFrame::kMaxDataSizeSamples; i++) {
>-    if (frame_data[i] != sample) {
>-      return false;
>-    }
>-  }
>-  return true;
>-}
>-
>-constexpr uint32_t kTimestamp = 27;
>-constexpr int kSampleRateHz = 16000;
>-constexpr size_t kNumChannels = 1;
>-constexpr size_t kSamplesPerChannel = kSampleRateHz / 100;
>-
>-}  // namespace
>-
>-TEST(AudioFrameTest, FrameStartsMuted) {
>-  AudioFrame frame;
>-  EXPECT_TRUE(frame.muted());
>-  EXPECT_TRUE(AllSamplesAre(0, frame));
>-}
>-
>-TEST(AudioFrameTest, UnmutedFrameIsInitiallyZeroed) {
>-  AudioFrame frame;
>-  frame.mutable_data();
>-  EXPECT_FALSE(frame.muted());
>-  EXPECT_TRUE(AllSamplesAre(0, frame));
>-}
>-
>-TEST(AudioFrameTest, MutedFrameBufferIsZeroed) {
>-  AudioFrame frame;
>-  int16_t* frame_data = frame.mutable_data();
>-  for (size_t i = 0; i < AudioFrame::kMaxDataSizeSamples; i++) {
>-    frame_data[i] = 17;
>-  }
>-  ASSERT_TRUE(AllSamplesAre(17, frame));
>-  frame.Mute();
>-  EXPECT_TRUE(frame.muted());
>-  EXPECT_TRUE(AllSamplesAre(0, frame));
>-}
>-
>-TEST(AudioFrameTest, UpdateFrame) {
>-  AudioFrame frame;
>-  int16_t samples[kNumChannels * kSamplesPerChannel] = {17};
>-  frame.UpdateFrame(kTimestamp, samples, kSamplesPerChannel, kSampleRateHz,
>-                    AudioFrame::kPLC, AudioFrame::kVadActive, kNumChannels);
>-
>-  EXPECT_EQ(kTimestamp, frame.timestamp_);
>-  EXPECT_EQ(kSamplesPerChannel, frame.samples_per_channel_);
>-  EXPECT_EQ(kSampleRateHz, frame.sample_rate_hz_);
>-  EXPECT_EQ(AudioFrame::kPLC, frame.speech_type_);
>-  EXPECT_EQ(AudioFrame::kVadActive, frame.vad_activity_);
>-  EXPECT_EQ(kNumChannels, frame.num_channels_);
>-
>-  EXPECT_FALSE(frame.muted());
>-  EXPECT_EQ(0, memcmp(samples, frame.data(), sizeof(samples)));
>-
>-  frame.UpdateFrame(kTimestamp, nullptr /* data*/, kSamplesPerChannel,
>-                    kSampleRateHz, AudioFrame::kPLC, AudioFrame::kVadActive,
>-                    kNumChannels);
>-  EXPECT_TRUE(frame.muted());
>-  EXPECT_TRUE(AllSamplesAre(0, frame));
>-}
>-
>-TEST(AudioFrameTest, CopyFrom) {
>-  AudioFrame frame1;
>-  AudioFrame frame2;
>-
>-  int16_t samples[kNumChannels * kSamplesPerChannel] = {17};
>-  frame2.UpdateFrame(kTimestamp, samples, kSamplesPerChannel,
>-                     kSampleRateHz, AudioFrame::kPLC, AudioFrame::kVadActive,
>-                     kNumChannels);
>-  frame1.CopyFrom(frame2);
>-
>-  EXPECT_EQ(frame2.timestamp_, frame1.timestamp_);
>-  EXPECT_EQ(frame2.samples_per_channel_, frame1.samples_per_channel_);
>-  EXPECT_EQ(frame2.sample_rate_hz_, frame1.sample_rate_hz_);
>-  EXPECT_EQ(frame2.speech_type_, frame1.speech_type_);
>-  EXPECT_EQ(frame2.vad_activity_, frame1.vad_activity_);
>-  EXPECT_EQ(frame2.num_channels_, frame1.num_channels_);
>-
>-  EXPECT_EQ(frame2.muted(), frame1.muted());
>-  EXPECT_EQ(0, memcmp(frame2.data(), frame1.data(), sizeof(samples)));
>-
>-  frame2.UpdateFrame(kTimestamp, nullptr /* data */, kSamplesPerChannel,
>-                     kSampleRateHz, AudioFrame::kPLC, AudioFrame::kVadActive,
>-                     kNumChannels);
>-  frame1.CopyFrom(frame2);
>-
>-  EXPECT_EQ(frame2.muted(), frame1.muted());
>-  EXPECT_EQ(0, memcmp(frame2.data(), frame1.data(), sizeof(samples)));
>-}
>-
> TEST(IsNewerSequenceNumber, Equal) {
>   EXPECT_FALSE(IsNewerSequenceNumber(0x0001, 0x0001));
> }
>@@ -210,7 +112,6 @@ TEST(LatestTimestamp, Wrap) {
>   EXPECT_EQ(0x0000FFFFu, LatestTimestamp(0xFFFF0000, 0x0000FFFF));
> }
> 
>-
> TEST(SequenceNumberUnwrapper, Limits) {
>   SequenceNumberUnwrapper unwrapper;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/BUILD.gn
>index df83aa55826..7f7767f4115 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/BUILD.gn
>@@ -10,44 +10,54 @@ import("../../webrtc.gni")
> 
> rtc_static_library("pacing") {
>   sources = [
>-    "alr_detector.cc",
>-    "alr_detector.h",
>     "bitrate_prober.cc",
>     "bitrate_prober.h",
>-    "interval_budget.cc",
>-    "interval_budget.h",
>     "paced_sender.cc",
>     "paced_sender.h",
>     "pacer.h",
>     "packet_queue.cc",
>     "packet_queue.h",
>-    "packet_queue2.cc",
>-    "packet_queue2.h",
>+    "packet_queue_interface.cc",
>+    "packet_queue_interface.h",
>     "packet_router.cc",
>     "packet_router.h",
>+    "round_robin_packet_queue.cc",
>+    "round_robin_packet_queue.h",
>   ]
> 
>-  if (!build_with_chromium && is_clang) {
>-    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>-    suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-  }
>-
>   deps = [
>+    ":interval_budget",
>     "..:module_api",
>-    "../../:typedefs",
>     "../../:webrtc_common",
>-    "../../api:optional",
>+    "../../logging:rtc_event_bwe",
>     "../../logging:rtc_event_log_api",
>+    "../../logging:rtc_event_pacing",
>     "../../rtc_base:checks",
>     "../../rtc_base:rtc_base_approved",
>     "../../rtc_base/experiments:alr_experiment",
>     "../../system_wrappers",
>     "../../system_wrappers:field_trial_api",
>     "../../system_wrappers:runtime_enabled_features_api",
>+    "../congestion_controller/goog_cc:alr_detector",
>     "../remote_bitrate_estimator",
>     "../rtp_rtcp",
>     "../rtp_rtcp:rtp_rtcp_format",
>     "../utility",
>+    "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/types:optional",
>+  ]
>+}
>+
>+rtc_source_set("interval_budget") {
>+  sources = [
>+    "interval_budget.cc",
>+    "interval_budget.h",
>+  ]
>+
>+  deps = [
>+    "../../:webrtc_common",
>+    "../../rtc_base:checks",
>+    "../../rtc_base:rtc_base_approved",
>   ]
> }
> 
>@@ -56,13 +66,13 @@ if (rtc_include_tests) {
>     testonly = true
> 
>     sources = [
>-      "alr_detector_unittest.cc",
>       "bitrate_prober_unittest.cc",
>       "interval_budget_unittest.cc",
>       "paced_sender_unittest.cc",
>       "packet_router_unittest.cc",
>     ]
>     deps = [
>+      ":interval_budget",
>       ":pacing",
>       "../../rtc_base:checks",
>       "../../rtc_base:rtc_base_approved",
>@@ -76,15 +86,7 @@ if (rtc_include_tests) {
>       "../rtp_rtcp",
>       "../rtp_rtcp:mock_rtp_rtcp",
>       "../rtp_rtcp:rtp_rtcp_format",
>-      "//testing/gmock",
>     ]
>-
>-    # TODO(jschuh): bugs.webrtc.org/1348: fix this warning.
>-    configs += [ "//build/config/compiler:no_size_t_to_int_warning" ]
>-    if (!build_with_chromium && is_clang) {
>-      # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>-      suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-    }
>   }
> 
>   rtc_source_set("mock_paced_sender") {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/OWNERS
>index 1528d7663fc..883ee0620d9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/OWNERS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/OWNERS
>@@ -2,6 +2,7 @@ stefan@webrtc.org
> mflodman@webrtc.org
> asapersson@webrtc.org
> philipel@webrtc.org
>+srte@webrtc.org
> 
> # These are for the common case of adding or renaming files. If you're doing
> # structural changes, please get a review from a reviewer in this file.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/bitrate_prober.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/bitrate_prober.cc
>index 7f60207f3e6..cbf1d2b97ef 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/bitrate_prober.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/bitrate_prober.cc
>@@ -12,12 +12,12 @@
> 
> #include <algorithm>
> 
>+#include "absl/memory/memory.h"
> #include "logging/rtc_event_log/events/rtc_event_probe_cluster_created.h"
> #include "logging/rtc_event_log/rtc_event_log.h"
> #include "modules/pacing/paced_sender.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/ptr_util.h"
> 
> namespace webrtc {
> 
>@@ -47,6 +47,8 @@ constexpr int64_t kProbeClusterTimeoutMs = 5000;
> 
> BitrateProber::BitrateProber() : BitrateProber(nullptr) {}
> 
>+BitrateProber::~BitrateProber() = default;
>+
> BitrateProber::BitrateProber(RtcEventLog* event_log)
>     : probing_state_(ProbingState::kDisabled),
>       next_probe_time_ms_(-1),
>@@ -100,7 +102,7 @@ void BitrateProber::CreateProbeCluster(int bitrate_bps, int64_t now_ms) {
>   cluster.pace_info.probe_cluster_id = next_cluster_id_++;
>   clusters_.push(cluster);
>   if (event_log_)
>-    event_log_->Log(rtc::MakeUnique<RtcEventProbeClusterCreated>(
>+    event_log_->Log(absl::make_unique<RtcEventProbeClusterCreated>(
>         cluster.pace_info.probe_cluster_id, cluster.pace_info.send_bitrate_bps,
>         cluster.pace_info.probe_cluster_min_probes,
>         cluster.pace_info.probe_cluster_min_bytes));
>@@ -124,9 +126,9 @@ int BitrateProber::TimeUntilNextProbe(int64_t now_ms) {
>   if (next_probe_time_ms_ >= 0) {
>     time_until_probe_ms = next_probe_time_ms_ - now_ms;
>     if (time_until_probe_ms < -kMaxProbeDelayMs) {
>-      RTC_LOG(LS_WARNING)<<"Probe delay too high"<<
>-                           " (next_ms:"<<next_probe_time_ms_<<
>-                           ", now_ms: "<<now_ms<<")";
>+      RTC_LOG(LS_WARNING) << "Probe delay too high"
>+                          << " (next_ms:" << next_probe_time_ms_
>+                          << ", now_ms: " << now_ms << ")";
>       return -1;
>     }
>   }
>@@ -183,5 +185,4 @@ int64_t BitrateProber::GetNextProbeTime(const ProbeCluster& cluster) {
>   return cluster.time_started_ms + delta_ms;
> }
> 
>-
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/bitrate_prober.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/bitrate_prober.h
>index c98de83a348..f0ceab27a05 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/bitrate_prober.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/bitrate_prober.h
>@@ -14,8 +14,6 @@
> #include <queue>
> 
> #include "modules/include/module_common_types.h"
>-#include "rtc_base/basictypes.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> class RtcEventLog;
>@@ -26,6 +24,7 @@ class BitrateProber {
>  public:
>   BitrateProber();
>   explicit BitrateProber(RtcEventLog* event_log);
>+  ~BitrateProber();
> 
>   void SetEnabled(bool enable);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/bitrate_prober_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/bitrate_prober_unittest.cc
>index 184ac063884..dc596982db0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/bitrate_prober_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/bitrate_prober_unittest.cc
>@@ -49,7 +49,7 @@ TEST(BitrateProberTest, VerifyStatesAndTimeBetweenProbes) {
>   // Verify that the actual bitrate is withing 10% of the target.
>   double bitrate = kProbeSize * (kClusterSize - 1) * 8 * 1000.0 / now_ms;
>   EXPECT_GT(bitrate, kTestBitrate1 * 0.9);
>-  EXPECT_LT(bitrate,  kTestBitrate1 * 1.1);
>+  EXPECT_LT(bitrate, kTestBitrate1 * 1.1);
> 
>   now_ms += prober.TimeUntilNextProbe(now_ms);
>   int64_t probe2_started = now_ms;
>@@ -66,7 +66,7 @@ TEST(BitrateProberTest, VerifyStatesAndTimeBetweenProbes) {
>   EXPECT_GE(duration, kMinProbeDurationMs);
>   bitrate = kProbeSize * (kClusterSize - 1) * 8 * 1000.0 / duration;
>   EXPECT_GT(bitrate, kTestBitrate2 * 0.9);
>-  EXPECT_LT(bitrate,  kTestBitrate2 * 1.1);
>+  EXPECT_LT(bitrate, kTestBitrate2 * 1.1);
> 
>   EXPECT_EQ(-1, prober.TimeUntilNextProbe(now_ms));
>   EXPECT_FALSE(prober.IsProbing());
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/interval_budget.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/interval_budget.cc
>index b63bc37149e..21cc2949f54 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/interval_budget.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/interval_budget.cc
>@@ -8,14 +8,14 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "modules/pacing/interval_budget.h"
>-
> #include <algorithm>
> 
>+#include "modules/pacing/interval_budget.h"
>+#include "rtc_base/numerics/safe_conversions.h"
>+
> namespace webrtc {
> namespace {
> constexpr int kWindowMs = 500;
>-constexpr int kDeltaTimeMs = 2000;
> }
> 
> IntervalBudget::IntervalBudget(int initial_target_rate_kbps)
>@@ -35,8 +35,7 @@ void IntervalBudget::set_target_rate_kbps(int target_rate_kbps) {
> }
> 
> void IntervalBudget::IncreaseBudget(int64_t delta_time_ms) {
>-  RTC_DCHECK_LT(delta_time_ms, kDeltaTimeMs);
>-  int bytes = target_rate_kbps_ * delta_time_ms / 8;
>+  int bytes = rtc::dchecked_cast<int>(target_rate_kbps_ * delta_time_ms / 8);
>   if (bytes_remaining_ < 0 || can_build_up_underuse_) {
>     // We overused last interval, compensate this interval.
>     bytes_remaining_ = std::min(bytes_remaining_ + bytes, max_bytes_in_budget_);
>@@ -56,7 +55,10 @@ size_t IntervalBudget::bytes_remaining() const {
> }
> 
> int IntervalBudget::budget_level_percent() const {
>-  return bytes_remaining_ * 100 / max_bytes_in_budget_;
>+  if (max_bytes_in_budget_ == 0)
>+    return 0;
>+  return rtc::dchecked_cast<int>(int64_t{bytes_remaining_} * 100 /
>+                                 max_bytes_in_budget_);
> }
> 
> int IntervalBudget::target_rate_kbps() const {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/mock/mock_paced_sender.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/mock/mock_paced_sender.h
>index 3366aa85ed1..4a5337d70b4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/mock/mock_paced_sender.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/mock/mock_paced_sender.h
>@@ -23,19 +23,21 @@ class MockPacedSender : public PacedSender {
>  public:
>   MockPacedSender()
>       : PacedSender(Clock::GetRealTimeClock(), nullptr, nullptr) {}
>-  MOCK_METHOD6(SendPacket, bool(Priority priority,
>-                                uint32_t ssrc,
>-                                uint16_t sequence_number,
>-                                int64_t capture_time_ms,
>-                                size_t bytes,
>-                                bool retransmission));
>+  MOCK_METHOD6(SendPacket,
>+               bool(Priority priority,
>+                    uint32_t ssrc,
>+                    uint16_t sequence_number,
>+                    int64_t capture_time_ms,
>+                    size_t bytes,
>+                    bool retransmission));
>   MOCK_METHOD1(CreateProbeCluster, void(int));
>   MOCK_METHOD1(SetEstimatedBitrate, void(uint32_t));
>+  MOCK_METHOD2(SetPacingRates, void(uint32_t, uint32_t));
>   MOCK_CONST_METHOD0(QueueInMs, int64_t());
>   MOCK_CONST_METHOD0(QueueInPackets, int());
>   MOCK_CONST_METHOD0(ExpectedQueueTimeMs, int64_t());
>   MOCK_CONST_METHOD0(GetApplicationLimitedRegionStartTime,
>-                     rtc::Optional<int64_t>());
>+                     absl::optional<int64_t>());
>   MOCK_METHOD0(Process, void());
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/paced_sender.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/paced_sender.cc
>index f559ffc2c24..c64b161478b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/paced_sender.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/paced_sender.cc
>@@ -14,17 +14,18 @@
> #include <map>
> #include <queue>
> #include <set>
>-#include <vector>
> #include <utility>
>+#include <vector>
> 
>+#include "absl/memory/memory.h"
>+#include "modules/congestion_controller/goog_cc/alr_detector.h"
> #include "modules/include/module_common_types.h"
>-#include "modules/pacing/alr_detector.h"
> #include "modules/pacing/bitrate_prober.h"
> #include "modules/pacing/interval_budget.h"
>+#include "modules/pacing/round_robin_packet_queue.h"
> #include "modules/utility/include/process_thread.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/ptr_util.h"
> #include "system_wrappers/include/clock.h"
> #include "system_wrappers/include/field_trial.h"
> #include "system_wrappers/include/runtime_enabled_features.h"
>@@ -32,21 +33,14 @@
> namespace {
> // Time limit in milliseconds between packet bursts.
> const int64_t kMinPacketLimitMs = 5;
>-const int64_t kPausedPacketIntervalMs = 500;
>+const int64_t kCongestedPacketIntervalMs = 500;
>+const int64_t kPausedProcessIntervalMs = kCongestedPacketIntervalMs;
>+const int64_t kMaxElapsedTimeMs = 2000;
> 
> // Upper cap on process interval, in case process has not been called in a long
> // time.
> const int64_t kMaxIntervalTimeMs = 30;
> 
>-const char kRoundRobinExperimentName[] = "WebRTC-RoundRobinPacing";
>-
>-bool IsRoundRobinPacingEnabled() {
>-  return webrtc::field_trial::IsEnabled(kRoundRobinExperimentName) || (
>-      !webrtc::field_trial::IsDisabled(kRoundRobinExperimentName) &&
>-      webrtc::runtime_enabled_features::IsFeatureEnabled(
>-          webrtc::runtime_enabled_features::kDualStreamModeFeatureName));
>-}
>-
> }  // namespace
> 
> namespace webrtc {
>@@ -56,35 +50,43 @@ const float PacedSender::kDefaultPaceMultiplier = 2.5f;
> 
> PacedSender::PacedSender(const Clock* clock,
>                          PacketSender* packet_sender,
>-                         RtcEventLog* event_log) :
>-    PacedSender(clock, packet_sender, event_log,
>-                IsRoundRobinPacingEnabled()
>-                    ? rtc::MakeUnique<PacketQueue2>(clock)
>-                    : rtc::MakeUnique<PacketQueue>(clock)) {}
>+                         RtcEventLog* event_log)
>+    : PacedSender(clock,
>+                  packet_sender,
>+                  event_log,
>+                  absl::make_unique<RoundRobinPacketQueue>(clock)) {}
> 
> PacedSender::PacedSender(const Clock* clock,
>                          PacketSender* packet_sender,
>                          RtcEventLog* event_log,
>-                         std::unique_ptr<PacketQueue> packets)
>+                         std::unique_ptr<PacketQueueInterface> packets)
>     : clock_(clock),
>       packet_sender_(packet_sender),
>-      alr_detector_(rtc::MakeUnique<AlrDetector>(event_log)),
>+      alr_detector_(absl::make_unique<AlrDetector>(event_log)),
>+      drain_large_queues_(!field_trial::IsDisabled("WebRTC-Pacer-DrainQueue")),
>+      send_padding_if_silent_(
>+          field_trial::IsEnabled("WebRTC-Pacer-PadInSilence")),
>+      video_blocks_audio_(!field_trial::IsDisabled("WebRTC-Pacer-BlockAudio")),
>       paused_(false),
>-      media_budget_(rtc::MakeUnique<IntervalBudget>(0)),
>-      padding_budget_(rtc::MakeUnique<IntervalBudget>(0)),
>-      prober_(rtc::MakeUnique<BitrateProber>(event_log)),
>+      media_budget_(absl::make_unique<IntervalBudget>(0)),
>+      padding_budget_(absl::make_unique<IntervalBudget>(0)),
>+      prober_(absl::make_unique<BitrateProber>(event_log)),
>       probing_send_failure_(false),
>       estimated_bitrate_bps_(0),
>       min_send_bitrate_kbps_(0u),
>       max_padding_bitrate_kbps_(0u),
>       pacing_bitrate_kbps_(0),
>-      time_last_update_us_(clock->TimeInMicroseconds()),
>+      time_last_process_us_(clock->TimeInMicroseconds()),
>+      last_send_time_us_(clock->TimeInMicroseconds()),
>       first_sent_packet_ms_(-1),
>       packets_(std::move(packets)),
>       packet_counter_(0),
>       pacing_factor_(kDefaultPaceMultiplier),
>       queue_time_limit(kMaxQueueLengthMs),
>       account_for_audio_(false) {
>+  if (!drain_large_queues_)
>+    RTC_LOG(LS_WARNING) << "Pacer queues will not be drained,"
>+                           "pushback experiment must be enabled.";
>   UpdateBudgetWithElapsedTime(kMinPacketLimitMs);
> }
> 
>@@ -103,6 +105,7 @@ void PacedSender::Pause() {
>     paused_ = true;
>     packets_->SetPauseState(true, clock_->TimeInMilliseconds());
>   }
>+  rtc::CritScope cs(&process_thread_lock_);
>   // Tell the process thread to call our TimeUntilNextProcess() method to get
>   // a new (longer) estimate for when to call Process().
>   if (process_thread_)
>@@ -117,12 +120,29 @@ void PacedSender::Resume() {
>     paused_ = false;
>     packets_->SetPauseState(false, clock_->TimeInMilliseconds());
>   }
>+  rtc::CritScope cs(&process_thread_lock_);
>   // Tell the process thread to call our TimeUntilNextProcess() method to
>   // refresh the estimate for when to call Process().
>   if (process_thread_)
>     process_thread_->WakeUp(this);
> }
> 
>+void PacedSender::SetCongestionWindow(int64_t congestion_window_bytes) {
>+  rtc::CritScope cs(&critsect_);
>+  congestion_window_bytes_ = congestion_window_bytes;
>+}
>+
>+void PacedSender::UpdateOutstandingData(int64_t outstanding_bytes) {
>+  rtc::CritScope cs(&critsect_);
>+  outstanding_bytes_ = outstanding_bytes;
>+}
>+
>+bool PacedSender::Congested() const {
>+  if (congestion_window_bytes_ == kNoCongestionWindow)
>+    return false;
>+  return outstanding_bytes_ >= congestion_window_bytes_;
>+}
>+
> void PacedSender::SetProbingEnabled(bool enabled) {
>   rtc::CritScope cs(&critsect_);
>   RTC_CHECK_EQ(0, packet_counter_);
>@@ -154,6 +174,14 @@ void PacedSender::SetSendBitrateLimits(int min_send_bitrate_bps,
>       std::min(estimated_bitrate_bps_ / 1000, max_padding_bitrate_kbps_));
> }
> 
>+void PacedSender::SetPacingRates(uint32_t pacing_rate_bps,
>+                                 uint32_t padding_rate_bps) {
>+  rtc::CritScope cs(&critsect_);
>+  RTC_DCHECK(pacing_rate_bps > 0);
>+  pacing_bitrate_kbps_ = pacing_rate_bps / 1000;
>+  padding_budget_->set_target_rate_kbps(padding_rate_bps / 1000);
>+}
>+
> void PacedSender::InsertPacket(RtpPacketSender::Priority priority,
>                                uint32_t ssrc,
>                                uint16_t sequence_number,
>@@ -161,8 +189,8 @@ void PacedSender::InsertPacket(RtpPacketSender::Priority priority,
>                                size_t bytes,
>                                bool retransmission) {
>   rtc::CritScope cs(&critsect_);
>-  RTC_DCHECK(estimated_bitrate_bps_ > 0)
>-        << "SetEstimatedBitrate must be called before InsertPacket.";
>+  RTC_DCHECK(pacing_bitrate_kbps_ > 0)
>+      << "SetPacingRate must be called before InsertPacket.";
> 
>   int64_t now_ms = clock_->TimeInMilliseconds();
>   prober_->OnIncomingPacket(bytes);
>@@ -170,9 +198,9 @@ void PacedSender::InsertPacket(RtpPacketSender::Priority priority,
>   if (capture_time_ms < 0)
>     capture_time_ms = now_ms;
> 
>-  packets_->Push(PacketQueue::Packet(priority, ssrc, sequence_number,
>-                                     capture_time_ms, now_ms, bytes,
>-                                     retransmission, packet_counter_++));
>+  packets_->Push(PacketQueueInterface::Packet(
>+      priority, ssrc, sequence_number, capture_time_ms, now_ms, bytes,
>+      retransmission, packet_counter_++));
> }
> 
> void PacedSender::SetAccountForAudioPackets(bool account_for_audio) {
>@@ -187,7 +215,7 @@ int64_t PacedSender::ExpectedQueueTimeMs() const {
>                               pacing_bitrate_kbps_);
> }
> 
>-rtc::Optional<int64_t> PacedSender::GetApplicationLimitedRegionStartTime()
>+absl::optional<int64_t> PacedSender::GetApplicationLimitedRegionStartTime()
>     const {
>   rtc::CritScope cs(&critsect_);
>   return alr_detector_->GetApplicationLimitedRegionStartTime();
>@@ -215,12 +243,13 @@ int64_t PacedSender::QueueInMs() const {
> 
> int64_t PacedSender::TimeUntilNextProcess() {
>   rtc::CritScope cs(&critsect_);
>-  int64_t elapsed_time_us = clock_->TimeInMicroseconds() - time_last_update_us_;
>+  int64_t elapsed_time_us =
>+      clock_->TimeInMicroseconds() - time_last_process_us_;
>   int64_t elapsed_time_ms = (elapsed_time_us + 500) / 1000;
>   // When paused we wake up every 500 ms to send a padding packet to ensure
>   // we won't get stuck in the paused state due to no feedback being received.
>   if (paused_)
>-    return std::max<int64_t>(kPausedPacketIntervalMs - elapsed_time_ms, 0);
>+    return std::max<int64_t>(kPausedProcessIntervalMs - elapsed_time_ms, 0);
> 
>   if (prober_->IsProbing()) {
>     int64_t ret = prober_->TimeUntilNextProbe(clock_->TimeInMilliseconds());
>@@ -233,43 +262,53 @@ int64_t PacedSender::TimeUntilNextProcess() {
> void PacedSender::Process() {
>   int64_t now_us = clock_->TimeInMicroseconds();
>   rtc::CritScope cs(&critsect_);
>-  int64_t elapsed_time_ms = std::min(
>-      kMaxIntervalTimeMs, (now_us - time_last_update_us_ + 500) / 1000);
>-  int target_bitrate_kbps = pacing_bitrate_kbps_;
>-
>-  if (paused_) {
>-    PacedPacketInfo pacing_info;
>-    time_last_update_us_ = now_us;
>-    // We can not send padding unless a normal packet has first been sent. If we
>-    // do, timestamps get messed up.
>-    if (packet_counter_ == 0)
>-      return;
>-    size_t bytes_sent = SendPadding(1, pacing_info);
>-    alr_detector_->OnBytesSent(bytes_sent, elapsed_time_ms);
>-    return;
>+  int64_t elapsed_time_ms = (now_us - time_last_process_us_ + 500) / 1000;
>+  time_last_process_us_ = now_us;
>+  if (elapsed_time_ms > kMaxElapsedTimeMs) {
>+    RTC_LOG(LS_WARNING) << "Elapsed time (" << elapsed_time_ms
>+                        << " ms) longer than expected, limiting to "
>+                        << kMaxElapsedTimeMs << " ms";
>+    elapsed_time_ms = kMaxElapsedTimeMs;
>+  }
>+  if (send_padding_if_silent_ || paused_ || Congested()) {
>+    // We send a padding packet every 500 ms to ensure we won't get stuck in
>+    // congested state due to no feedback being received.
>+    int64_t elapsed_since_last_send_us = now_us - last_send_time_us_;
>+    if (elapsed_since_last_send_us >= kCongestedPacketIntervalMs * 1000) {
>+      // We can not send padding unless a normal packet has first been sent. If
>+      // we do, timestamps get messed up.
>+      if (packet_counter_ > 0) {
>+        PacedPacketInfo pacing_info;
>+        size_t bytes_sent = SendPadding(1, pacing_info);
>+        alr_detector_->OnBytesSent(bytes_sent, now_us / 1000);
>+      }
>+    }
>   }
>+  if (paused_)
>+    return;
> 
>   if (elapsed_time_ms > 0) {
>+    int target_bitrate_kbps = pacing_bitrate_kbps_;
>     size_t queue_size_bytes = packets_->SizeInBytes();
>     if (queue_size_bytes > 0) {
>       // Assuming equal size packets and input/output rate, the average packet
>       // has avg_time_left_ms left to get queue_size_bytes out of the queue, if
>       // time constraint shall be met. Determine bitrate needed for that.
>       packets_->UpdateQueueTime(clock_->TimeInMilliseconds());
>-      int64_t avg_time_left_ms = std::max<int64_t>(
>-          1, queue_time_limit - packets_->AverageQueueTimeMs());
>-      int min_bitrate_needed_kbps =
>-          static_cast<int>(queue_size_bytes * 8 / avg_time_left_ms);
>-      if (min_bitrate_needed_kbps > target_bitrate_kbps)
>-        target_bitrate_kbps = min_bitrate_needed_kbps;
>+      if (drain_large_queues_) {
>+        int64_t avg_time_left_ms = std::max<int64_t>(
>+            1, queue_time_limit - packets_->AverageQueueTimeMs());
>+        int min_bitrate_needed_kbps =
>+            static_cast<int>(queue_size_bytes * 8 / avg_time_left_ms);
>+        if (min_bitrate_needed_kbps > target_bitrate_kbps)
>+          target_bitrate_kbps = min_bitrate_needed_kbps;
>+      }
>     }
> 
>     media_budget_->set_target_rate_kbps(target_bitrate_kbps);
>     UpdateBudgetWithElapsedTime(elapsed_time_ms);
>   }
> 
>-  time_last_update_us_ = now_us;
>-
>   bool is_probing = prober_->IsProbing();
>   PacedPacketInfo pacing_info;
>   size_t bytes_sent = 0;
>@@ -278,17 +317,17 @@ void PacedSender::Process() {
>     pacing_info = prober_->CurrentCluster();
>     recommended_probe_size = prober_->RecommendedMinProbeSize();
>   }
>-  while (!packets_->Empty()) {
>+  // The paused state is checked in the loop since SendPacket leaves the
>+  // critical section allowing the paused state to be changed from other code.
>+  while (!packets_->Empty() && !paused_) {
>     // Since we need to release the lock in order to send, we first pop the
>     // element from the priority queue but keep it in storage, so that we can
>     // reinsert it if send fails.
>-    const PacketQueue::Packet& packet = packets_->BeginPop();
>+    const PacketQueueInterface::Packet& packet = packets_->BeginPop();
> 
>     if (SendPacket(packet, pacing_info)) {
>-      // Send succeeded, remove it from the queue.
>-      if (first_sent_packet_ms_ == -1)
>-        first_sent_packet_ms_ = clock_->TimeInMilliseconds();
>       bytes_sent += packet.bytes;
>+      // Send succeeded, remove it from the queue.
>       packets_->FinalizePop(packet);
>       if (is_probing && bytes_sent > recommended_probe_size)
>         break;
>@@ -299,15 +338,16 @@ void PacedSender::Process() {
>     }
>   }
> 
>-  if (packets_->Empty()) {
>+  if (packets_->Empty() && !Congested()) {
>     // We can not send padding unless a normal packet has first been sent. If we
>     // do, timestamps get messed up.
>     if (packet_counter_ > 0) {
>       int padding_needed =
>           static_cast<int>(is_probing ? (recommended_probe_size - bytes_sent)
>                                       : padding_budget_->bytes_remaining());
>-      if (padding_needed > 0)
>+      if (padding_needed > 0) {
>         bytes_sent += SendPadding(padding_needed, pacing_info);
>+      }
>     }
>   }
>   if (is_probing) {
>@@ -315,19 +355,24 @@ void PacedSender::Process() {
>     if (!probing_send_failure_)
>       prober_->ProbeSent(clock_->TimeInMilliseconds(), bytes_sent);
>   }
>-  alr_detector_->OnBytesSent(bytes_sent, elapsed_time_ms);
>+  alr_detector_->OnBytesSent(bytes_sent, now_us / 1000);
> }
> 
> void PacedSender::ProcessThreadAttached(ProcessThread* process_thread) {
>-  RTC_LOG(LS_INFO) << "ProcessThreadAttached 0x" << std::hex << process_thread;
>+  RTC_LOG(LS_INFO) << "ProcessThreadAttached 0x" << process_thread;
>+  rtc::CritScope cs(&process_thread_lock_);
>   process_thread_ = process_thread;
> }
> 
>-bool PacedSender::SendPacket(const PacketQueue::Packet& packet,
>+bool PacedSender::SendPacket(const PacketQueueInterface::Packet& packet,
>                              const PacedPacketInfo& pacing_info) {
>   RTC_DCHECK(!paused_);
>-  if (media_budget_->bytes_remaining() == 0 &&
>-      pacing_info.probe_cluster_id == PacedPacketInfo::kNotAProbe) {
>+  bool audio_packet = packet.priority == kHighPriority;
>+  bool apply_pacing =
>+      !audio_packet || account_for_audio_ || video_blocks_audio_;
>+  if (apply_pacing && (Congested() || (media_budget_->bytes_remaining() == 0 &&
>+                                       pacing_info.probe_cluster_id ==
>+                                           PacedPacketInfo::kNotAProbe))) {
>     return false;
>   }
> 
>@@ -338,13 +383,16 @@ bool PacedSender::SendPacket(const PacketQueue::Packet& packet,
>   critsect_.Enter();
> 
>   if (success) {
>-    if (packet.priority != kHighPriority || account_for_audio_) {
>+    if (first_sent_packet_ms_ == -1)
>+      first_sent_packet_ms_ = clock_->TimeInMilliseconds();
>+    if (!audio_packet || account_for_audio_) {
>       // Update media bytes sent.
>       // TODO(eladalon): TimeToSendPacket() can also return |true| in some
>       // situations where nothing actually ended up being sent to the network,
>       // and we probably don't want to update the budget in such cases.
>       // https://bugs.chromium.org/p/webrtc/issues/detail?id=8052
>       UpdateBudgetWithBytesSent(packet.bytes);
>+      last_send_time_us_ = clock_->TimeInMicroseconds();
>     }
>   }
> 
>@@ -362,15 +410,18 @@ size_t PacedSender::SendPadding(size_t padding_needed,
>   if (bytes_sent > 0) {
>     UpdateBudgetWithBytesSent(bytes_sent);
>   }
>+  last_send_time_us_ = clock_->TimeInMicroseconds();
>   return bytes_sent;
> }
> 
> void PacedSender::UpdateBudgetWithElapsedTime(int64_t delta_time_ms) {
>+  delta_time_ms = std::min(kMaxIntervalTimeMs, delta_time_ms);
>   media_budget_->IncreaseBudget(delta_time_ms);
>   padding_budget_->IncreaseBudget(delta_time_ms);
> }
> 
> void PacedSender::UpdateBudgetWithBytesSent(size_t bytes_sent) {
>+  outstanding_bytes_ += bytes_sent;
>   media_budget_->UseBudget(bytes_sent);
>   padding_budget_->UseBudget(bytes_sent);
> }
>@@ -383,11 +434,6 @@ void PacedSender::SetPacingFactor(float pacing_factor) {
>   SetEstimatedBitrate(estimated_bitrate_bps_);
> }
> 
>-float PacedSender::GetPacingFactor() const {
>-  rtc::CritScope cs(&critsect_);
>-  return pacing_factor_;
>-}
>-
> void PacedSender::SetQueueTimeLimit(int limit_ms) {
>   rtc::CritScope cs(&critsect_);
>   queue_time_limit = limit_ms;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/paced_sender.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/paced_sender.h
>index c1090ecbc02..2fcc3a3c0d2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/paced_sender.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/paced_sender.h
>@@ -13,18 +13,16 @@
> 
> #include <memory>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> #include "modules/pacing/pacer.h"
>-#include "modules/pacing/packet_queue2.h"
>+#include "modules/pacing/packet_queue_interface.h"
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/thread_annotations.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> class AlrDetector;
> class BitrateProber;
> class Clock;
>-class ProbeClusterCreatedObserver;
> class RtcEventLog;
> class IntervalBudget;
> 
>@@ -49,6 +47,7 @@ class PacedSender : public Pacer {
>    protected:
>     virtual ~PacketSender() {}
>   };
>+  static constexpr int64_t kNoCongestionWindow = -1;
> 
>   // Expected max pacer delay in ms. If ExpectedQueueTimeMs() is higher than
>   // this value, the packet producers should wait (eg drop frames rather than
>@@ -69,7 +68,7 @@ class PacedSender : public Pacer {
>   PacedSender(const Clock* clock,
>               PacketSender* packet_sender,
>               RtcEventLog* event_log,
>-              std::unique_ptr<PacketQueue> packets);
>+              std::unique_ptr<PacketQueueInterface> packets);
> 
>   ~PacedSender() override;
> 
>@@ -81,28 +80,24 @@ class PacedSender : public Pacer {
>   // Resume sending packets.
>   void Resume();
> 
>+  void SetCongestionWindow(int64_t congestion_window_bytes);
>+  void UpdateOutstandingData(int64_t outstanding_bytes);
>+
>   // Enable bitrate probing. Enabled by default, mostly here to simplify
>   // testing. Must be called before any packets are being sent to have an
>   // effect.
>   void SetProbingEnabled(bool enabled);
> 
>-  // Sets the estimated capacity of the network. Must be called once before
>-  // packets can be sent.
>-  // |bitrate_bps| is our estimate of what we are allowed to send on average.
>-  // We will pace out bursts of packets at a bitrate of
>-  // |bitrate_bps| * kDefaultPaceMultiplier.
>+  // Deprecated, SetPacingRates should be used instead.
>   void SetEstimatedBitrate(uint32_t bitrate_bps) override;
>-
>-  // Sets the minimum send bitrate and maximum padding bitrate requested by send
>-  // streams.
>-  // |min_send_bitrate_bps| might be higher that the estimated available network
>-  // bitrate and if so, the pacer will send with |min_send_bitrate_bps|.
>-  // |max_padding_bitrate_bps| might be higher than the estimate available
>-  // network bitrate and if so, the pacer will send padding packets to reach
>-  // the min of the estimated available bitrate and |max_padding_bitrate_bps|.
>+  // Deprecated, SetPacingRates should be used instead.
>   void SetSendBitrateLimits(int min_send_bitrate_bps,
>                             int max_padding_bitrate_bps);
> 
>+  // Sets the pacing rates. Must be called once before packets can be sent.
>+  void SetPacingRates(uint32_t pacing_rate_bps,
>+                      uint32_t padding_rate_bps) override;
>+
>   // Returns true if we send the packet now, else it will add the packet
>   // information to the queue and call TimeToSendPacket when it's time to send.
>   void InsertPacket(RtpPacketSender::Priority priority,
>@@ -131,13 +126,8 @@ class PacedSender : public Pacer {
>   // packets in the queue, given the current size and bitrate, ignoring prio.
>   virtual int64_t ExpectedQueueTimeMs() const;
> 
>-  // Returns time in milliseconds when the current application-limited region
>-  // started or empty result if the sender is currently not application-limited.
>-  //
>-  // Application Limited Region (ALR) refers to operating in a state where the
>-  // traffic on network is limited due to application not having enough
>-  // traffic to meet the current channel capacity.
>-  virtual rtc::Optional<int64_t> GetApplicationLimitedRegionStartTime() const;
>+  // Deprecated, alr detection will be moved out of the pacer.
>+  virtual absl::optional<int64_t> GetApplicationLimitedRegionStartTime() const;
> 
>   // Returns the number of milliseconds until the module want a worker thread
>   // to call Process.
>@@ -148,8 +138,8 @@ class PacedSender : public Pacer {
> 
>   // Called when the prober is associated with a process thread.
>   void ProcessThreadAttached(ProcessThread* process_thread) override;
>+  // Deprecated, SetPacingRates should be used instead.
>   void SetPacingFactor(float pacing_factor);
>-  float GetPacingFactor() const;
>   void SetQueueTimeLimit(int limit_ms);
> 
>  private:
>@@ -159,16 +149,22 @@ class PacedSender : public Pacer {
>   void UpdateBudgetWithBytesSent(size_t bytes)
>       RTC_EXCLUSIVE_LOCKS_REQUIRED(critsect_);
> 
>-  bool SendPacket(const PacketQueue::Packet& packet,
>+  bool SendPacket(const PacketQueueInterface::Packet& packet,
>                   const PacedPacketInfo& cluster_info)
>       RTC_EXCLUSIVE_LOCKS_REQUIRED(critsect_);
>   size_t SendPadding(size_t padding_needed, const PacedPacketInfo& cluster_info)
>       RTC_EXCLUSIVE_LOCKS_REQUIRED(critsect_);
> 
>+  void OnBytesSent(size_t bytes_sent) RTC_EXCLUSIVE_LOCKS_REQUIRED(critsect_);
>+  bool Congested() const RTC_EXCLUSIVE_LOCKS_REQUIRED(critsect_);
>+
>   const Clock* const clock_;
>   PacketSender* const packet_sender_;
>   const std::unique_ptr<AlrDetector> alr_detector_ RTC_PT_GUARDED_BY(critsect_);
> 
>+  const bool drain_large_queues_;
>+  const bool send_padding_if_silent_;
>+  const bool video_blocks_audio_;
>   rtc::CriticalSection critsect_;
>   bool paused_ RTC_GUARDED_BY(critsect_);
>   // This is the media budget, keeping track of how many bits of media
>@@ -190,14 +186,26 @@ class PacedSender : public Pacer {
>   uint32_t max_padding_bitrate_kbps_ RTC_GUARDED_BY(critsect_);
>   uint32_t pacing_bitrate_kbps_ RTC_GUARDED_BY(critsect_);
> 
>-  int64_t time_last_update_us_ RTC_GUARDED_BY(critsect_);
>+  int64_t time_last_process_us_ RTC_GUARDED_BY(critsect_);
>+  int64_t last_send_time_us_ RTC_GUARDED_BY(critsect_);
>   int64_t first_sent_packet_ms_ RTC_GUARDED_BY(critsect_);
> 
>-  const std::unique_ptr<PacketQueue> packets_ RTC_PT_GUARDED_BY(critsect_);
>+  const std::unique_ptr<PacketQueueInterface> packets_
>+      RTC_PT_GUARDED_BY(critsect_);
>   uint64_t packet_counter_ RTC_GUARDED_BY(critsect_);
>-  ProcessThread* process_thread_ = nullptr;
> 
>+  int64_t congestion_window_bytes_ RTC_GUARDED_BY(critsect_) =
>+      kNoCongestionWindow;
>+  int64_t outstanding_bytes_ RTC_GUARDED_BY(critsect_) = 0;
>   float pacing_factor_ RTC_GUARDED_BY(critsect_);
>+  // Lock to avoid race when attaching process thread. This can happen due to
>+  // the Call class setting network state on SendSideCongestionController, which
>+  // in turn calls Pause/Resume on Pacedsender, before actually starting the
>+  // pacer process thread. If SendSideCongestionController is running on a task
>+  // queue separate from the thread used by Call, this causes a race.
>+  rtc::CriticalSection process_thread_lock_;
>+  ProcessThread* process_thread_ RTC_GUARDED_BY(process_thread_lock_) = nullptr;
>+
>   int64_t queue_time_limit RTC_GUARDED_BY(critsect_);
>   bool account_for_audio_ RTC_GUARDED_BY(critsect_);
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/paced_sender_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/paced_sender_unittest.cc
>index 4281ec220aa..fb79ebcbaf5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/paced_sender_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/paced_sender_unittest.cc
>@@ -31,6 +31,8 @@ constexpr unsigned kSecondClusterBps = 1800000;
> // values. This results in probing slightly higher than the target bitrate.
> // For 1.8 Mbps, this comes to be about 120 kbps with 1200 probe packets.
> constexpr int kBitrateProbingError = 150000;
>+
>+const float kPaceMultiplier = 2.5f;
> }  // namespace
> 
> namespace webrtc {
>@@ -106,7 +108,7 @@ class PacedSenderProbing : public PacedSender::PacketSender {
> 
> class PacedSenderTest : public testing::TestWithParam<std::string> {
>  protected:
>-  PacedSenderTest() : clock_(123456), field_trial_(GetParam()) {
>+  PacedSenderTest() : clock_(123456) {
>     srand(0);
>     // Need to initialize PacedSender after we initialize clock.
>     send_bucket_.reset(new PacedSender(&clock_, &callback_, nullptr));
>@@ -116,7 +118,7 @@ class PacedSenderTest : public testing::TestWithParam<std::string> {
>     // have to enable probing, either by creating a new PacedSender instance or
>     // by calling SetProbingEnabled(true).
>     send_bucket_->SetProbingEnabled(false);
>-    send_bucket_->SetEstimatedBitrate(kTargetBitrateBps);
>+    send_bucket_->SetPacingRates(kTargetBitrateBps * kPaceMultiplier, 0);
> 
>     clock_.AdvanceTimeMilliseconds(send_bucket_->TimeUntilNextProcess());
>   }
>@@ -137,15 +139,142 @@ class PacedSenderTest : public testing::TestWithParam<std::string> {
>   SimulatedClock clock_;
>   MockPacedSenderCallback callback_;
>   std::unique_ptr<PacedSender> send_bucket_;
>-  test::ScopedFieldTrials field_trial_;
> };
> 
>-INSTANTIATE_TEST_CASE_P(RoundRobin,
>-                        PacedSenderTest,
>-                        ::testing::Values("WebRTC-RoundRobinPacing/Disabled/",
>-                                          "WebRTC-RoundRobinPacing/Enabled/"));
>+class PacedSenderFieldTrialTest : public testing::Test {
>+ protected:
>+  struct MediaStream {
>+    const RtpPacketSender::Priority priority;
>+    const uint32_t ssrc;
>+    const size_t packet_size;
>+    uint16_t seq_num;
>+  };
>+
>+  const int kProcessIntervalsPerSecond = 1000 / 5;
>+
>+  PacedSenderFieldTrialTest() : clock_(123456) {}
>+  void InsertPacket(PacedSender* pacer, MediaStream* stream) {
>+    pacer->InsertPacket(stream->priority, stream->ssrc, stream->seq_num++,
>+                        clock_.TimeInMilliseconds(), stream->packet_size,
>+                        false);
>+  }
>+  void ProcessNext(PacedSender* pacer) {
>+    clock_.AdvanceTimeMilliseconds(5);
>+    pacer->Process();
>+  }
>+  MediaStream audio{/*priority*/ PacedSender::kHighPriority,
>+                    /*ssrc*/ 3333, /*packet_size*/ 100, /*seq_num*/ 1000};
>+  MediaStream video{/*priority*/ PacedSender::kNormalPriority,
>+                    /*ssrc*/ 4444, /*packet_size*/ 1000, /*seq_num*/ 1000};
>+  SimulatedClock clock_;
>+  MockPacedSenderCallback callback_;
>+};
> 
>-TEST_P(PacedSenderTest, FirstSentPacketTimeIsSet) {
>+TEST_F(PacedSenderFieldTrialTest, DefaultNoPaddingInSilence) {
>+  PacedSender pacer(&clock_, &callback_, nullptr);
>+  pacer.SetPacingRates(kTargetBitrateBps, 0);
>+  // Video packet to reset last send time and provide padding data.
>+  InsertPacket(&pacer, &video);
>+  EXPECT_CALL(callback_, TimeToSendPacket).WillOnce(Return(true));
>+  clock_.AdvanceTimeMilliseconds(5);
>+  pacer.Process();
>+  EXPECT_CALL(callback_, TimeToSendPadding).Times(0);
>+  // Waiting 500 ms should not trigger sending of padding.
>+  clock_.AdvanceTimeMilliseconds(500);
>+  pacer.Process();
>+}
>+
>+TEST_F(PacedSenderFieldTrialTest, PaddingInSilenceWithTrial) {
>+  ScopedFieldTrials trial("WebRTC-Pacer-PadInSilence/Enabled/");
>+  PacedSender pacer(&clock_, &callback_, nullptr);
>+  pacer.SetPacingRates(kTargetBitrateBps, 0);
>+  // Video packet to reset last send time and provide padding data.
>+  InsertPacket(&pacer, &video);
>+  EXPECT_CALL(callback_, TimeToSendPacket).WillOnce(Return(true));
>+  clock_.AdvanceTimeMilliseconds(5);
>+  pacer.Process();
>+  EXPECT_CALL(callback_, TimeToSendPadding).WillOnce(Return(1000));
>+  // Waiting 500 ms should trigger sending of padding.
>+  clock_.AdvanceTimeMilliseconds(500);
>+  pacer.Process();
>+}
>+
>+TEST_F(PacedSenderFieldTrialTest, DefaultCongestionWindowAffectsAudio) {
>+  EXPECT_CALL(callback_, TimeToSendPadding).Times(0);
>+  PacedSender pacer(&clock_, &callback_, nullptr);
>+  pacer.SetPacingRates(10000000, 0);
>+  pacer.SetCongestionWindow(800);
>+  pacer.UpdateOutstandingData(0);
>+  // Video packet fills congestion window.
>+  InsertPacket(&pacer, &video);
>+  EXPECT_CALL(callback_, TimeToSendPacket).WillOnce(Return(true));
>+  ProcessNext(&pacer);
>+  // Audio packet blocked due to congestion.
>+  InsertPacket(&pacer, &audio);
>+  EXPECT_CALL(callback_, TimeToSendPacket).Times(0);
>+  ProcessNext(&pacer);
>+  ProcessNext(&pacer);
>+  // Audio packet unblocked when congestion window clear.
>+  testing::Mock::VerifyAndClearExpectations(&callback_);
>+  pacer.UpdateOutstandingData(0);
>+  EXPECT_CALL(callback_, TimeToSendPacket).WillOnce(Return(true));
>+  ProcessNext(&pacer);
>+}
>+
>+TEST_F(PacedSenderFieldTrialTest, CongestionWindowDoesNotAffectAudioInTrial) {
>+  ScopedFieldTrials trial("WebRTC-Pacer-BlockAudio/Disabled/");
>+  EXPECT_CALL(callback_, TimeToSendPadding).Times(0);
>+  PacedSender pacer(&clock_, &callback_, nullptr);
>+  pacer.SetPacingRates(10000000, 0);
>+  pacer.SetCongestionWindow(800);
>+  pacer.UpdateOutstandingData(0);
>+  // Video packet fills congestion window.
>+  InsertPacket(&pacer, &video);
>+  EXPECT_CALL(callback_, TimeToSendPacket).WillOnce(Return(true));
>+  ProcessNext(&pacer);
>+  // Audio not blocked due to congestion.
>+  InsertPacket(&pacer, &audio);
>+  EXPECT_CALL(callback_, TimeToSendPacket).WillOnce(Return(true));
>+  ProcessNext(&pacer);
>+}
>+
>+TEST_F(PacedSenderFieldTrialTest, DefaultBudgetAffectsAudio) {
>+  PacedSender pacer(&clock_, &callback_, nullptr);
>+  pacer.SetPacingRates(video.packet_size / 3 * 8 * kProcessIntervalsPerSecond,
>+                       0);
>+  // Video fills budget for following process periods.
>+  InsertPacket(&pacer, &video);
>+  EXPECT_CALL(callback_, TimeToSendPacket).WillOnce(Return(true));
>+  ProcessNext(&pacer);
>+  // Audio packet blocked due to budget limit.
>+  EXPECT_CALL(callback_, TimeToSendPacket).Times(0);
>+  InsertPacket(&pacer, &audio);
>+  ProcessNext(&pacer);
>+  ProcessNext(&pacer);
>+  testing::Mock::VerifyAndClearExpectations(&callback_);
>+  // Audio packet unblocked when the budget has recovered.
>+  EXPECT_CALL(callback_, TimeToSendPacket).WillOnce(Return(true));
>+  ProcessNext(&pacer);
>+  ProcessNext(&pacer);
>+}
>+
>+TEST_F(PacedSenderFieldTrialTest, BudgetDoesNotAffectAudioInTrial) {
>+  ScopedFieldTrials trial("WebRTC-Pacer-BlockAudio/Disabled/");
>+  EXPECT_CALL(callback_, TimeToSendPadding).Times(0);
>+  PacedSender pacer(&clock_, &callback_, nullptr);
>+  pacer.SetPacingRates(video.packet_size / 3 * 8 * kProcessIntervalsPerSecond,
>+                       0);
>+  // Video fills budget for following process periods.
>+  InsertPacket(&pacer, &video);
>+  EXPECT_CALL(callback_, TimeToSendPacket).WillOnce(Return(true));
>+  ProcessNext(&pacer);
>+  // Audio packet not blocked due to budget limit.
>+  EXPECT_CALL(callback_, TimeToSendPacket).WillOnce(Return(true));
>+  InsertPacket(&pacer, &audio);
>+  ProcessNext(&pacer);
>+}
>+
>+TEST_F(PacedSenderTest, FirstSentPacketTimeIsSet) {
>   uint16_t sequence_number = 1234;
>   const uint32_t kSsrc = 12345;
>   const size_t kSizeBytes = 250;
>@@ -164,14 +293,14 @@ TEST_P(PacedSenderTest, FirstSentPacketTimeIsSet) {
>   EXPECT_EQ(kStartMs, send_bucket_->FirstSentPacketTimeMs());
> }
> 
>-TEST_P(PacedSenderTest, QueuePacket) {
>+TEST_F(PacedSenderTest, QueuePacket) {
>   uint32_t ssrc = 12345;
>   uint16_t sequence_number = 1234;
>   // Due to the multiplicative factor we can send 5 packets during a send
>   // interval. (network capacity * multiplier / (8 bits per byte *
>   // (packet size * #send intervals per second)
>   const size_t packets_to_send =
>-      kTargetBitrateBps * PacedSender::kDefaultPaceMultiplier / (8 * 250 * 200);
>+      kTargetBitrateBps * kPaceMultiplier / (8 * 250 * 200);
>   for (size_t i = 0; i < packets_to_send; ++i) {
>     SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number++,
>                         clock_.TimeInMilliseconds(), 250, false);
>@@ -212,7 +341,7 @@ TEST_P(PacedSenderTest, QueuePacket) {
>   EXPECT_EQ(1u, send_bucket_->QueueSizePackets());
> }
> 
>-TEST_P(PacedSenderTest, PaceQueuedPackets) {
>+TEST_F(PacedSenderTest, PaceQueuedPackets) {
>   uint32_t ssrc = 12345;
>   uint16_t sequence_number = 1234;
> 
>@@ -220,7 +349,7 @@ TEST_P(PacedSenderTest, PaceQueuedPackets) {
>   // interval. (network capacity * multiplier / (8 bits per byte *
>   // (packet size * #send intervals per second)
>   const size_t packets_to_send_per_interval =
>-      kTargetBitrateBps * PacedSender::kDefaultPaceMultiplier / (8 * 250 * 200);
>+      kTargetBitrateBps * kPaceMultiplier / (8 * 250 * 200);
>   for (size_t i = 0; i < packets_to_send_per_interval; ++i) {
>     SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number++,
>                         clock_.TimeInMilliseconds(), 250, false);
>@@ -264,7 +393,7 @@ TEST_P(PacedSenderTest, PaceQueuedPackets) {
>   EXPECT_EQ(1u, send_bucket_->QueueSizePackets());
> }
> 
>-TEST_P(PacedSenderTest, RepeatedRetransmissionsAllowed) {
>+TEST_F(PacedSenderTest, RepeatedRetransmissionsAllowed) {
>   // Send one packet, then two retransmissions of that packet.
>   for (size_t i = 0; i < 3; i++) {
>     constexpr uint32_t ssrc = 333;
>@@ -278,41 +407,33 @@ TEST_P(PacedSenderTest, RepeatedRetransmissionsAllowed) {
>   send_bucket_->Process();
> }
> 
>-TEST_P(PacedSenderTest, CanQueuePacketsWithSameSequenceNumberOnDifferentSsrcs) {
>+TEST_F(PacedSenderTest, CanQueuePacketsWithSameSequenceNumberOnDifferentSsrcs) {
>   uint32_t ssrc = 12345;
>   uint16_t sequence_number = 1234;
> 
>-  SendAndExpectPacket(PacedSender::kNormalPriority,
>-                      ssrc,
>-                      sequence_number,
>-                      clock_.TimeInMilliseconds(),
>-                      250,
>-                      false);
>+  SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number,
>+                      clock_.TimeInMilliseconds(), 250, false);
> 
>   // Expect packet on second ssrc to be queued and sent as well.
>-  SendAndExpectPacket(PacedSender::kNormalPriority,
>-                      ssrc + 1,
>-                      sequence_number,
>-                      clock_.TimeInMilliseconds(),
>-                      250,
>-                      false);
>+  SendAndExpectPacket(PacedSender::kNormalPriority, ssrc + 1, sequence_number,
>+                      clock_.TimeInMilliseconds(), 250, false);
> 
>   clock_.AdvanceTimeMilliseconds(1000);
>   send_bucket_->Process();
> }
> 
>-TEST_P(PacedSenderTest, Padding) {
>+TEST_F(PacedSenderTest, Padding) {
>   uint32_t ssrc = 12345;
>   uint16_t sequence_number = 1234;
> 
>-  send_bucket_->SetEstimatedBitrate(kTargetBitrateBps);
>-  send_bucket_->SetSendBitrateLimits(kTargetBitrateBps, kTargetBitrateBps);
>+  send_bucket_->SetPacingRates(kTargetBitrateBps * kPaceMultiplier,
>+                               kTargetBitrateBps);
> 
>   // Due to the multiplicative factor we can send 5 packets during a send
>   // interval. (network capacity * multiplier / (8 bits per byte *
>   // (packet size * #send intervals per second)
>   const size_t packets_to_send_per_interval =
>-      kTargetBitrateBps * PacedSender::kDefaultPaceMultiplier / (8 * 250 * 200);
>+      kTargetBitrateBps * kPaceMultiplier / (8 * 250 * 200);
>   for (size_t i = 0; i < packets_to_send_per_interval; ++i) {
>     SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number++,
>                         clock_.TimeInMilliseconds(), 250, false);
>@@ -341,9 +462,9 @@ TEST_P(PacedSenderTest, Padding) {
>   send_bucket_->Process();
> }
> 
>-TEST_P(PacedSenderTest, NoPaddingBeforeNormalPacket) {
>-  send_bucket_->SetEstimatedBitrate(kTargetBitrateBps);
>-  send_bucket_->SetSendBitrateLimits(kTargetBitrateBps, kTargetBitrateBps);
>+TEST_F(PacedSenderTest, NoPaddingBeforeNormalPacket) {
>+  send_bucket_->SetPacingRates(kTargetBitrateBps * kPaceMultiplier,
>+                               kTargetBitrateBps);
> 
>   EXPECT_CALL(callback_, TimeToSendPadding(_, _)).Times(0);
>   send_bucket_->Process();
>@@ -364,23 +485,19 @@ TEST_P(PacedSenderTest, NoPaddingBeforeNormalPacket) {
>   send_bucket_->Process();
> }
> 
>-TEST_P(PacedSenderTest, VerifyPaddingUpToBitrate) {
>+TEST_F(PacedSenderTest, VerifyPaddingUpToBitrate) {
>   uint32_t ssrc = 12345;
>   uint16_t sequence_number = 1234;
>   int64_t capture_time_ms = 56789;
>   const int kTimeStep = 5;
>   const int64_t kBitrateWindow = 100;
>-  send_bucket_->SetEstimatedBitrate(kTargetBitrateBps);
>-  send_bucket_->SetSendBitrateLimits(kTargetBitrateBps, kTargetBitrateBps);
>+  send_bucket_->SetPacingRates(kTargetBitrateBps * kPaceMultiplier,
>+                               kTargetBitrateBps);
> 
>   int64_t start_time = clock_.TimeInMilliseconds();
>   while (clock_.TimeInMilliseconds() - start_time < kBitrateWindow) {
>-    SendAndExpectPacket(PacedSender::kNormalPriority,
>-                        ssrc,
>-                        sequence_number++,
>-                        capture_time_ms,
>-                        250,
>-                        false);
>+    SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number++,
>+                        capture_time_ms, 250, false);
>     EXPECT_CALL(callback_, TimeToSendPadding(250, _))
>         .Times(1)
>         .WillOnce(Return(250));
>@@ -389,7 +506,7 @@ TEST_P(PacedSenderTest, VerifyPaddingUpToBitrate) {
>   }
> }
> 
>-TEST_P(PacedSenderTest, VerifyAverageBitrateVaryingMediaPayload) {
>+TEST_F(PacedSenderTest, VerifyAverageBitrateVaryingMediaPayload) {
>   uint32_t ssrc = 12345;
>   uint16_t sequence_number = 1234;
>   int64_t capture_time_ms = 56789;
>@@ -398,11 +515,8 @@ TEST_P(PacedSenderTest, VerifyAverageBitrateVaryingMediaPayload) {
>   PacedSenderPadding callback;
>   send_bucket_.reset(new PacedSender(&clock_, &callback, nullptr));
>   send_bucket_->SetProbingEnabled(false);
>-  send_bucket_->SetEstimatedBitrate(kTargetBitrateBps);
>-
>-  send_bucket_->SetSendBitrateLimits(
>-      0 /*allocated_bitrate_bps*/,
>-      kTargetBitrateBps * 2 /* max_padding_bitrate_bps */);
>+  send_bucket_->SetPacingRates(kTargetBitrateBps * kPaceMultiplier,
>+                               kTargetBitrateBps);
> 
>   int64_t start_time = clock_.TimeInMilliseconds();
>   size_t media_bytes = 0;
>@@ -422,7 +536,7 @@ TEST_P(PacedSenderTest, VerifyAverageBitrateVaryingMediaPayload) {
>               1);
> }
> 
>-TEST_P(PacedSenderTest, Priority) {
>+TEST_F(PacedSenderTest, Priority) {
>   uint32_t ssrc_low_priority = 12345;
>   uint32_t ssrc = 12346;
>   uint16_t sequence_number = 1234;
>@@ -433,7 +547,7 @@ TEST_P(PacedSenderTest, Priority) {
>   // interval. (network capacity * multiplier / (8 bits per byte *
>   // (packet size * #send intervals per second)
>   const size_t packets_to_send_per_interval =
>-      kTargetBitrateBps * PacedSender::kDefaultPaceMultiplier / (8 * 250 * 200);
>+      kTargetBitrateBps * kPaceMultiplier / (8 * 250 * 200);
>   for (size_t i = 0; i < packets_to_send_per_interval; ++i) {
>     SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number++,
>                         clock_.TimeInMilliseconds(), 250, false);
>@@ -477,7 +591,7 @@ TEST_P(PacedSenderTest, Priority) {
>   send_bucket_->Process();
> }
> 
>-TEST_P(PacedSenderTest, RetransmissionPriority) {
>+TEST_F(PacedSenderTest, RetransmissionPriority) {
>   uint32_t ssrc = 12345;
>   uint16_t sequence_number = 1234;
>   int64_t capture_time_ms = 45678;
>@@ -487,7 +601,7 @@ TEST_P(PacedSenderTest, RetransmissionPriority) {
>   // interval. (network capacity * multiplier / (8 bits per byte *
>   // (packet size * #send intervals per second)
>   const size_t packets_to_send_per_interval =
>-      kTargetBitrateBps * PacedSender::kDefaultPaceMultiplier / (8 * 250 * 200);
>+      kTargetBitrateBps * kPaceMultiplier / (8 * 250 * 200);
>   send_bucket_->Process();
>   EXPECT_EQ(0u, send_bucket_->QueueSizePackets());
> 
>@@ -531,7 +645,7 @@ TEST_P(PacedSenderTest, RetransmissionPriority) {
>   EXPECT_EQ(0u, send_bucket_->QueueSizePackets());
> }
> 
>-TEST_P(PacedSenderTest, HighPrioDoesntAffectBudget) {
>+TEST_F(PacedSenderTest, HighPrioDoesntAffectBudget) {
>   uint32_t ssrc = 12346;
>   uint16_t sequence_number = 1234;
>   int64_t capture_time_ms = 56789;
>@@ -548,7 +662,7 @@ TEST_P(PacedSenderTest, HighPrioDoesntAffectBudget) {
>   // interval. (network capacity * multiplier / (8 bits per byte *
>   // (packet size * #send intervals per second)
>   const size_t packets_to_send_per_interval =
>-      kTargetBitrateBps * PacedSender::kDefaultPaceMultiplier / (8 * 250 * 200);
>+      kTargetBitrateBps * kPaceMultiplier / (8 * 250 * 200);
>   for (size_t i = 0; i < packets_to_send_per_interval; ++i) {
>     SendAndExpectPacket(PacedSender::kLowPriority, ssrc, sequence_number++,
>                         clock_.TimeInMilliseconds(), 250, false);
>@@ -569,7 +683,146 @@ TEST_P(PacedSenderTest, HighPrioDoesntAffectBudget) {
>   EXPECT_EQ(0u, send_bucket_->QueueSizePackets());
> }
> 
>-TEST_P(PacedSenderTest, Pause) {
>+TEST_F(PacedSenderTest, SendsOnlyPaddingWhenCongested) {
>+  uint32_t ssrc = 202020;
>+  uint16_t sequence_number = 1000;
>+  int kPacketSize = 250;
>+  int kCongestionWindow = kPacketSize * 10;
>+
>+  send_bucket_->UpdateOutstandingData(0);
>+  send_bucket_->SetCongestionWindow(kCongestionWindow);
>+  int sent_data = 0;
>+  while (sent_data < kCongestionWindow) {
>+    sent_data += kPacketSize;
>+    SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number++,
>+                        clock_.TimeInMilliseconds(), kPacketSize, false);
>+    clock_.AdvanceTimeMilliseconds(5);
>+    send_bucket_->Process();
>+  }
>+  testing::Mock::VerifyAndClearExpectations(&callback_);
>+  EXPECT_CALL(callback_, TimeToSendPacket(_, _, _, _, _)).Times(0);
>+  EXPECT_CALL(callback_, TimeToSendPadding(_, _)).Times(0);
>+
>+  size_t blocked_packets = 0;
>+  int64_t expected_time_until_padding = 500;
>+  while (expected_time_until_padding > 5) {
>+    send_bucket_->InsertPacket(PacedSender::kNormalPriority, ssrc,
>+                               sequence_number++, clock_.TimeInMilliseconds(),
>+                               kPacketSize, false);
>+    blocked_packets++;
>+    clock_.AdvanceTimeMilliseconds(5);
>+    send_bucket_->Process();
>+    expected_time_until_padding -= 5;
>+  }
>+  testing::Mock::VerifyAndClearExpectations(&callback_);
>+  EXPECT_CALL(callback_, TimeToSendPadding(1, _)).Times(1);
>+  clock_.AdvanceTimeMilliseconds(5);
>+  send_bucket_->Process();
>+  EXPECT_EQ(blocked_packets, send_bucket_->QueueSizePackets());
>+}
>+
>+TEST_F(PacedSenderTest, DoesNotAllowOveruseAfterCongestion) {
>+  uint32_t ssrc = 202020;
>+  uint16_t seq_num = 1000;
>+  RtpPacketSender::Priority prio = PacedSender::kNormalPriority;
>+  int size = 1000;
>+  auto now_ms = [this] { return clock_.TimeInMilliseconds(); };
>+  EXPECT_CALL(callback_, TimeToSendPadding).Times(0);
>+  // The pacing rate is low enough that the budget should not allow two packets
>+  // to be sent in a row.
>+  send_bucket_->SetPacingRates(400 * 8 * 1000 / 5, 0);
>+  // The congestion window is small enough to only let one packet through.
>+  send_bucket_->SetCongestionWindow(800);
>+  send_bucket_->UpdateOutstandingData(0);
>+  // Not yet budget limited or congested, packet is sent.
>+  send_bucket_->InsertPacket(prio, ssrc, seq_num++, now_ms(), size, false);
>+  EXPECT_CALL(callback_, TimeToSendPacket).WillOnce(Return(true));
>+  clock_.AdvanceTimeMilliseconds(5);
>+  send_bucket_->Process();
>+  // Packet blocked due to congestion.
>+  send_bucket_->InsertPacket(prio, ssrc, seq_num++, now_ms(), size, false);
>+  EXPECT_CALL(callback_, TimeToSendPacket).Times(0);
>+  clock_.AdvanceTimeMilliseconds(5);
>+  send_bucket_->Process();
>+  // Packet blocked due to congestion.
>+  send_bucket_->InsertPacket(prio, ssrc, seq_num++, now_ms(), size, false);
>+  EXPECT_CALL(callback_, TimeToSendPacket).Times(0);
>+  clock_.AdvanceTimeMilliseconds(5);
>+  send_bucket_->Process();
>+  send_bucket_->UpdateOutstandingData(0);
>+  // Congestion removed and budget has recovered, packet is sent.
>+  send_bucket_->InsertPacket(prio, ssrc, seq_num++, now_ms(), size, false);
>+  EXPECT_CALL(callback_, TimeToSendPacket).WillOnce(Return(true));
>+  clock_.AdvanceTimeMilliseconds(5);
>+  send_bucket_->Process();
>+  send_bucket_->UpdateOutstandingData(0);
>+  // Should be blocked due to budget limitation as congestion has be removed.
>+  send_bucket_->InsertPacket(prio, ssrc, seq_num++, now_ms(), size, false);
>+  EXPECT_CALL(callback_, TimeToSendPacket).Times(0);
>+  clock_.AdvanceTimeMilliseconds(5);
>+  send_bucket_->Process();
>+}
>+
>+TEST_F(PacedSenderTest, ResumesSendingWhenCongestionEnds) {
>+  uint32_t ssrc = 202020;
>+  uint16_t sequence_number = 1000;
>+  int64_t kPacketSize = 250;
>+  int64_t kCongestionCount = 10;
>+  int64_t kCongestionWindow = kPacketSize * kCongestionCount;
>+  int64_t kCongestionTimeMs = 1000;
>+
>+  send_bucket_->UpdateOutstandingData(0);
>+  send_bucket_->SetCongestionWindow(kCongestionWindow);
>+  int sent_data = 0;
>+  while (sent_data < kCongestionWindow) {
>+    sent_data += kPacketSize;
>+    SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number++,
>+                        clock_.TimeInMilliseconds(), kPacketSize, false);
>+    clock_.AdvanceTimeMilliseconds(5);
>+    send_bucket_->Process();
>+  }
>+  testing::Mock::VerifyAndClearExpectations(&callback_);
>+  EXPECT_CALL(callback_, TimeToSendPacket(_, _, _, _, _)).Times(0);
>+  int unacked_packets = 0;
>+  for (int duration = 0; duration < kCongestionTimeMs; duration += 5) {
>+    send_bucket_->InsertPacket(PacedSender::kNormalPriority, ssrc,
>+                               sequence_number++, clock_.TimeInMilliseconds(),
>+                               kPacketSize, false);
>+    unacked_packets++;
>+    clock_.AdvanceTimeMilliseconds(5);
>+    send_bucket_->Process();
>+  }
>+  testing::Mock::VerifyAndClearExpectations(&callback_);
>+
>+  // First mark half of the congested packets as cleared and make sure that just
>+  // as many are sent
>+  int ack_count = kCongestionCount / 2;
>+  EXPECT_CALL(callback_, TimeToSendPacket(ssrc, _, _, false, _))
>+      .Times(ack_count)
>+      .WillRepeatedly(Return(true));
>+  send_bucket_->UpdateOutstandingData(kCongestionWindow -
>+                                      kPacketSize * ack_count);
>+
>+  for (int duration = 0; duration < kCongestionTimeMs; duration += 5) {
>+    clock_.AdvanceTimeMilliseconds(5);
>+    send_bucket_->Process();
>+  }
>+  unacked_packets -= ack_count;
>+  testing::Mock::VerifyAndClearExpectations(&callback_);
>+
>+  // Second make sure all packets are sent if sent packets are continuously
>+  // marked as acked.
>+  EXPECT_CALL(callback_, TimeToSendPacket(ssrc, _, _, false, _))
>+      .Times(unacked_packets)
>+      .WillRepeatedly(Return(true));
>+  for (int duration = 0; duration < kCongestionTimeMs; duration += 5) {
>+    send_bucket_->UpdateOutstandingData(0);
>+    clock_.AdvanceTimeMilliseconds(5);
>+    send_bucket_->Process();
>+  }
>+}
>+
>+TEST_F(PacedSenderTest, Pause) {
>   uint32_t ssrc_low_priority = 12345;
>   uint32_t ssrc = 12346;
>   uint32_t ssrc_high_priority = 12347;
>@@ -582,7 +835,7 @@ TEST_P(PacedSenderTest, Pause) {
>   // interval. (network capacity * multiplier / (8 bits per byte *
>   // (packet size * #send intervals per second)
>   const size_t packets_to_send_per_interval =
>-      kTargetBitrateBps * PacedSender::kDefaultPaceMultiplier / (8 * 250 * 200);
>+      kTargetBitrateBps * kPaceMultiplier / (8 * 250 * 200);
>   for (size_t i = 0; i < packets_to_send_per_interval; ++i) {
>     SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number++,
>                         clock_.TimeInMilliseconds(), 250, false);
>@@ -623,16 +876,17 @@ TEST_P(PacedSenderTest, Pause) {
>   send_bucket_->Process();
> 
>   int64_t expected_time_until_send = 500;
>-  EXPECT_CALL(callback_, TimeToSendPadding(1, _)).Times(1);
>-  while (expected_time_until_send >= 0) {
>-    // TimeUntilNextProcess must not return 0 when paused.  If it does,
>-    // we risk running a busy loop, so ideally it should return a large value.
>-    EXPECT_EQ(expected_time_until_send, send_bucket_->TimeUntilNextProcess());
>-    if (expected_time_until_send == 0)
>-      send_bucket_->Process();
>+  EXPECT_CALL(callback_, TimeToSendPadding(_, _)).Times(0);
>+  while (expected_time_until_send >= 5) {
>+    send_bucket_->Process();
>     clock_.AdvanceTimeMilliseconds(5);
>     expected_time_until_send -= 5;
>   }
>+  testing::Mock::VerifyAndClearExpectations(&callback_);
>+  EXPECT_CALL(callback_, TimeToSendPadding(1, _)).Times(1);
>+  clock_.AdvanceTimeMilliseconds(5);
>+  send_bucket_->Process();
>+  testing::Mock::VerifyAndClearExpectations(&callback_);
> 
>   // Expect high prio packets to come out first followed by normal
>   // prio packets and low prio packets (all in capture order).
>@@ -673,6 +927,11 @@ TEST_P(PacedSenderTest, Pause) {
>   }
>   send_bucket_->Resume();
> 
>+  // The pacer was resumed directly after the previous process call finished. It
>+  // will therefore wait 5 ms until next process.
>+  EXPECT_EQ(5, send_bucket_->TimeUntilNextProcess());
>+  clock_.AdvanceTimeMilliseconds(5);
>+
>   for (size_t i = 0; i < 4; i++) {
>     EXPECT_EQ(0, send_bucket_->TimeUntilNextProcess());
>     send_bucket_->Process();
>@@ -683,7 +942,7 @@ TEST_P(PacedSenderTest, Pause) {
>   EXPECT_EQ(0, send_bucket_->QueueInMs());
> }
> 
>-TEST_P(PacedSenderTest, ResendPacket) {
>+TEST_F(PacedSenderTest, ResendPacket) {
>   uint32_t ssrc = 12346;
>   uint16_t sequence_number = 1234;
>   int64_t capture_time_ms = clock_.TimeInMilliseconds();
>@@ -736,15 +995,15 @@ TEST_P(PacedSenderTest, ResendPacket) {
>   EXPECT_EQ(0, send_bucket_->QueueInMs());
> }
> 
>-TEST_P(PacedSenderTest, ExpectedQueueTimeMs) {
>+TEST_F(PacedSenderTest, ExpectedQueueTimeMs) {
>   uint32_t ssrc = 12346;
>   uint16_t sequence_number = 1234;
>   const size_t kNumPackets = 60;
>   const size_t kPacketSize = 1200;
>-  const int32_t kMaxBitrate = PacedSender::kDefaultPaceMultiplier * 30000;
>+  const int32_t kMaxBitrate = kPaceMultiplier * 30000;
>   EXPECT_EQ(0, send_bucket_->ExpectedQueueTimeMs());
> 
>-  send_bucket_->SetEstimatedBitrate(30000);
>+  send_bucket_->SetPacingRates(30000 * kPaceMultiplier, 0);
>   for (size_t i = 0; i < kNumPackets; ++i) {
>     SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number++,
>                         clock_.TimeInMilliseconds(), kPacketSize, false);
>@@ -773,18 +1032,14 @@ TEST_P(PacedSenderTest, ExpectedQueueTimeMs) {
>               static_cast<int64_t>(1000 * kPacketSize * 8 / kMaxBitrate));
> }
> 
>-TEST_P(PacedSenderTest, QueueTimeGrowsOverTime) {
>+TEST_F(PacedSenderTest, QueueTimeGrowsOverTime) {
>   uint32_t ssrc = 12346;
>   uint16_t sequence_number = 1234;
>   EXPECT_EQ(0, send_bucket_->QueueInMs());
> 
>-  send_bucket_->SetEstimatedBitrate(30000);
>-  SendAndExpectPacket(PacedSender::kNormalPriority,
>-                      ssrc,
>-                      sequence_number,
>-                      clock_.TimeInMilliseconds(),
>-                      1200,
>-                      false);
>+  send_bucket_->SetPacingRates(30000 * kPaceMultiplier, 0);
>+  SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number,
>+                      clock_.TimeInMilliseconds(), 1200, false);
> 
>   clock_.AdvanceTimeMilliseconds(500);
>   EXPECT_EQ(500, send_bucket_->QueueInMs());
>@@ -792,7 +1047,7 @@ TEST_P(PacedSenderTest, QueueTimeGrowsOverTime) {
>   EXPECT_EQ(0, send_bucket_->QueueInMs());
> }
> 
>-TEST_P(PacedSenderTest, ProbingWithInsertedPackets) {
>+TEST_F(PacedSenderTest, ProbingWithInsertedPackets) {
>   const size_t kPacketSize = 1200;
>   const int kInitialBitrateBps = 300000;
>   uint32_t ssrc = 12346;
>@@ -802,7 +1057,7 @@ TEST_P(PacedSenderTest, ProbingWithInsertedPackets) {
>   send_bucket_.reset(new PacedSender(&clock_, &packet_sender, nullptr));
>   send_bucket_->CreateProbeCluster(kFirstClusterBps);
>   send_bucket_->CreateProbeCluster(kSecondClusterBps);
>-  send_bucket_->SetEstimatedBitrate(kInitialBitrateBps);
>+  send_bucket_->SetPacingRates(kInitialBitrateBps * kPaceMultiplier, 0);
> 
>   for (int i = 0; i < 10; ++i) {
>     send_bucket_->InsertPacket(PacedSender::kNormalPriority, ssrc,
>@@ -838,7 +1093,7 @@ TEST_P(PacedSenderTest, ProbingWithInsertedPackets) {
>               kSecondClusterBps, kBitrateProbingError);
> }
> 
>-TEST_P(PacedSenderTest, ProbingWithPaddingSupport) {
>+TEST_F(PacedSenderTest, ProbingWithPaddingSupport) {
>   const size_t kPacketSize = 1200;
>   const int kInitialBitrateBps = 300000;
>   uint32_t ssrc = 12346;
>@@ -847,7 +1102,7 @@ TEST_P(PacedSenderTest, ProbingWithPaddingSupport) {
>   PacedSenderProbing packet_sender;
>   send_bucket_.reset(new PacedSender(&clock_, &packet_sender, nullptr));
>   send_bucket_->CreateProbeCluster(kFirstClusterBps);
>-  send_bucket_->SetEstimatedBitrate(kInitialBitrateBps);
>+  send_bucket_->SetPacingRates(kInitialBitrateBps * kPaceMultiplier, 0);
> 
>   for (int i = 0; i < 3; ++i) {
>     send_bucket_->InsertPacket(PacedSender::kNormalPriority, ssrc,
>@@ -874,69 +1129,13 @@ TEST_P(PacedSenderTest, ProbingWithPaddingSupport) {
>               kFirstClusterBps, kBitrateProbingError);
> }
> 
>-TEST_P(PacedSenderTest, PriorityInversion) {
>-  // In this test capture timestamps are used to order packets, capture
>-  // timestamps are not used in PacketQueue2.
>-  if (webrtc::field_trial::IsEnabled("WebRTC-RoundRobinPacing"))
>-    return;
>-  uint32_t ssrc = 12346;
>-  uint16_t sequence_number = 1234;
>-  const size_t kPacketSize = 1200;
>-
>-  send_bucket_->InsertPacket(
>-      PacedSender::kHighPriority, ssrc, sequence_number + 3,
>-      clock_.TimeInMilliseconds() + 33, kPacketSize, true);
>-
>-  send_bucket_->InsertPacket(
>-      PacedSender::kHighPriority, ssrc, sequence_number + 2,
>-      clock_.TimeInMilliseconds() + 33, kPacketSize, true);
>-
>-  send_bucket_->InsertPacket(PacedSender::kHighPriority, ssrc, sequence_number,
>-                             clock_.TimeInMilliseconds(), kPacketSize, true);
>-
>-  send_bucket_->InsertPacket(PacedSender::kHighPriority, ssrc,
>-                             sequence_number + 1, clock_.TimeInMilliseconds(),
>-                             kPacketSize, true);
>-
>-  // Packets from earlier frames should be sent first.
>-  {
>-    ::testing::InSequence sequence;
>-    EXPECT_CALL(callback_,
>-                TimeToSendPacket(ssrc, sequence_number,
>-                                 clock_.TimeInMilliseconds(), true, _))
>-        .WillOnce(Return(true));
>-    EXPECT_CALL(callback_,
>-                TimeToSendPacket(ssrc, sequence_number + 1,
>-                                 clock_.TimeInMilliseconds(), true, _))
>-        .WillOnce(Return(true));
>-    EXPECT_CALL(callback_,
>-                TimeToSendPacket(ssrc, sequence_number + 3,
>-                                 clock_.TimeInMilliseconds() + 33, true, _))
>-        .WillOnce(Return(true));
>-    EXPECT_CALL(callback_,
>-                TimeToSendPacket(ssrc, sequence_number + 2,
>-                                 clock_.TimeInMilliseconds() + 33, true, _))
>-        .WillOnce(Return(true));
>-
>-    while (send_bucket_->QueueSizePackets() > 0) {
>-      int time_until_process = send_bucket_->TimeUntilNextProcess();
>-      if (time_until_process <= 0) {
>-        send_bucket_->Process();
>-      } else {
>-        clock_.AdvanceTimeMilliseconds(time_until_process);
>-      }
>-    }
>-  }
>-}
>-
>-TEST_P(PacedSenderTest, PaddingOveruse) {
>+TEST_F(PacedSenderTest, PaddingOveruse) {
>   uint32_t ssrc = 12346;
>   uint16_t sequence_number = 1234;
>   const size_t kPacketSize = 1200;
> 
>   send_bucket_->Process();
>-  send_bucket_->SetEstimatedBitrate(60000);
>-  send_bucket_->SetSendBitrateLimits(60000, 0);
>+  send_bucket_->SetPacingRates(60000 * kPaceMultiplier, 0);
> 
>   SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number++,
>                       clock_.TimeInMilliseconds(), kPacketSize, false);
>@@ -945,7 +1144,7 @@ TEST_P(PacedSenderTest, PaddingOveruse) {
>   // Add 30kbit padding. When increasing budget, media budget will increase from
>   // negative (overuse) while padding budget will increase from 0.
>   clock_.AdvanceTimeMilliseconds(5);
>-  send_bucket_->SetSendBitrateLimits(60000, 30000);
>+  send_bucket_->SetPacingRates(60000 * kPaceMultiplier, 30000);
> 
>   SendAndExpectPacket(PacedSender::kNormalPriority, ssrc, sequence_number++,
>                       clock_.TimeInMilliseconds(), kPacketSize, false);
>@@ -963,7 +1162,7 @@ TEST_F(PacedSenderTest, AverageQueueTime) {
>   const size_t kPacketSize = 1200;
>   const int kBitrateBps = 10 * kPacketSize * 8;  // 10 packets per second.
> 
>-  send_bucket_->SetEstimatedBitrate(kBitrateBps);
>+  send_bucket_->SetPacingRates(kBitrateBps * kPaceMultiplier, 0);
> 
>   EXPECT_EQ(0, send_bucket_->AverageQueueTimeMs());
> 
>@@ -1003,12 +1202,13 @@ TEST_F(PacedSenderTest, AverageQueueTime) {
> }
> #endif
> 
>-TEST_P(PacedSenderTest, ProbeClusterId) {
>+TEST_F(PacedSenderTest, ProbeClusterId) {
>   uint32_t ssrc = 12346;
>   uint16_t sequence_number = 1234;
>   const size_t kPacketSize = 1200;
> 
>-  send_bucket_->SetSendBitrateLimits(kTargetBitrateBps, kTargetBitrateBps);
>+  send_bucket_->SetPacingRates(kTargetBitrateBps * kPaceMultiplier,
>+                               kTargetBitrateBps);
>   send_bucket_->SetProbingEnabled(true);
>   for (int i = 0; i < 10; ++i) {
>     send_bucket_->InsertPacket(PacedSender::kNormalPriority, ssrc,
>@@ -1049,12 +1249,13 @@ TEST_P(PacedSenderTest, ProbeClusterId) {
>   send_bucket_->Process();
> }
> 
>-TEST_P(PacedSenderTest, AvoidBusyLoopOnSendFailure) {
>+TEST_F(PacedSenderTest, AvoidBusyLoopOnSendFailure) {
>   uint32_t ssrc = 12346;
>   uint16_t sequence_number = 1234;
>   const size_t kPacketSize = kFirstClusterBps / (8000 / 10);
> 
>-  send_bucket_->SetSendBitrateLimits(kTargetBitrateBps, kTargetBitrateBps);
>+  send_bucket_->SetPacingRates(kTargetBitrateBps * kPaceMultiplier,
>+                               kTargetBitrateBps);
>   send_bucket_->SetProbingEnabled(true);
>   send_bucket_->InsertPacket(PacedSender::kNormalPriority, ssrc,
>                              sequence_number, clock_.TimeInMilliseconds(),
>@@ -1107,7 +1308,7 @@ TEST_F(PacedSenderTest, QueueTimeWithPause) {
>   EXPECT_EQ(200, send_bucket_->AverageQueueTimeMs());
> }
> 
>-TEST_P(PacedSenderTest, QueueTimePausedDuringPush) {
>+TEST_F(PacedSenderTest, QueueTimePausedDuringPush) {
>   const size_t kPacketSize = 1200;
>   const uint32_t kSsrc = 12346;
>   uint16_t sequence_number = 1234;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/pacer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/pacer.h
>index 8b43e1851ed..d7d5ea5fb44 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/pacer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/pacer.h
>@@ -18,6 +18,8 @@ namespace webrtc {
> class Pacer : public Module, public RtpPacketSender {
>  public:
>   virtual void SetEstimatedBitrate(uint32_t bitrate_bps) {}
>+  virtual void SetPacingRates(uint32_t pacing_rate_bps,
>+                              uint32_t padding_rate_bps) {}
>   virtual void SetEstimatedBitrateAndCongestionWindow(
>       uint32_t bitrate_bps,
>       bool in_probe_rtt,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/packet_queue.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/packet_queue.cc
>index e0308d160c5..eb0f822d506 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/packet_queue.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/packet_queue.cc
>@@ -15,7 +15,6 @@
> #include <vector>
> 
> #include "modules/include/module_common_types.h"
>-#include "modules/pacing/alr_detector.h"
> #include "modules/pacing/bitrate_prober.h"
> #include "modules/pacing/interval_budget.h"
> #include "modules/utility/include/process_thread.h"
>@@ -26,28 +25,6 @@
> 
> namespace webrtc {
> 
>-PacketQueue::Packet::Packet(RtpPacketSender::Priority priority,
>-                            uint32_t ssrc,
>-                            uint16_t seq_number,
>-                            int64_t capture_time_ms,
>-                            int64_t enqueue_time_ms,
>-                            size_t length_in_bytes,
>-                            bool retransmission,
>-                            uint64_t enqueue_order)
>-    : priority(priority),
>-      ssrc(ssrc),
>-      sequence_number(seq_number),
>-      capture_time_ms(capture_time_ms),
>-      enqueue_time_ms(enqueue_time_ms),
>-      sum_paused_ms(0),
>-      bytes(length_in_bytes),
>-      retransmission(retransmission),
>-      enqueue_order(enqueue_order) {}
>-
>-PacketQueue::Packet::Packet(const Packet& other) = default;
>-
>-PacketQueue::Packet::~Packet() {}
>-
> PacketQueue::PacketQueue(const Clock* clock)
>     : bytes_(0),
>       clock_(clock),
>@@ -70,17 +47,17 @@ void PacketQueue::Push(const Packet& packet) {
>   bytes_ += packet.bytes;
> }
> 
>-const PacketQueue::Packet& PacketQueue::BeginPop() {
>-  const PacketQueue::Packet& packet = *prio_queue_.top();
>+const PacketQueueInterface::Packet& PacketQueue::BeginPop() {
>+  const Packet& packet = *prio_queue_.top();
>   prio_queue_.pop();
>   return packet;
> }
> 
>-void PacketQueue::CancelPop(const PacketQueue::Packet& packet) {
>+void PacketQueue::CancelPop(const Packet& packet) {
>   prio_queue_.push(&(*packet.this_it));
> }
> 
>-void PacketQueue::FinalizePop(const PacketQueue::Packet& packet) {
>+void PacketQueue::FinalizePop(const Packet& packet) {
>   bytes_ -= packet.bytes;
>   int64_t packet_queue_time_ms = time_last_updated_ - packet.enqueue_time_ms;
>   RTC_DCHECK_LE(packet.sum_paused_ms, packet_queue_time_ms);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/packet_queue.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/packet_queue.h
>index 240961601c3..8041e080578 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/packet_queue.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/packet_queue.h
>@@ -16,62 +16,29 @@
> #include <set>
> #include <vector>
> 
>+#include "modules/pacing/packet_queue_interface.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> 
> namespace webrtc {
> 
>-class PacketQueue {
>+class PacketQueue : public PacketQueueInterface {
>  public:
>   explicit PacketQueue(const Clock* clock);
>-  virtual ~PacketQueue();
>-
>-  struct Packet {
>-    Packet(RtpPacketSender::Priority priority,
>-           uint32_t ssrc,
>-           uint16_t seq_number,
>-           int64_t capture_time_ms,
>-           int64_t enqueue_time_ms,
>-           size_t length_in_bytes,
>-           bool retransmission,
>-           uint64_t enqueue_order);
>-
>-    Packet(const Packet& other);
>-
>-    virtual ~Packet();
>-
>-    bool operator<(const Packet& other) const {
>-      if (priority != other.priority)
>-        return priority > other.priority;
>-      if (retransmission != other.retransmission)
>-        return other.retransmission;
>-
>-      return enqueue_order > other.enqueue_order;
>-    }
>-
>-    RtpPacketSender::Priority priority;
>-    uint32_t ssrc;
>-    uint16_t sequence_number;
>-    int64_t capture_time_ms;  // Absolute time of frame capture.
>-    int64_t enqueue_time_ms;  // Absolute time of pacer queue entry.
>-    int64_t sum_paused_ms;
>-    size_t bytes;
>-    bool retransmission;
>-    uint64_t enqueue_order;
>-    std::list<Packet>::iterator this_it;
>-    std::multiset<int64_t>::iterator enqueue_time_it;
>-  };
>-
>-  virtual void Push(const Packet& packet);
>-  virtual const Packet& BeginPop();
>-  virtual void CancelPop(const Packet& packet);
>-  virtual void FinalizePop(const Packet& packet);
>-  virtual bool Empty() const;
>-  virtual size_t SizeInPackets() const;
>-  virtual uint64_t SizeInBytes() const;
>-  virtual int64_t OldestEnqueueTimeMs() const;
>-  virtual void UpdateQueueTime(int64_t timestamp_ms);
>-  virtual void SetPauseState(bool paused, int64_t timestamp_ms);
>-  virtual int64_t AverageQueueTimeMs() const;
>+  ~PacketQueue() override;
>+
>+  using Packet = PacketQueueInterface::Packet;
>+
>+  void Push(const Packet& packet) override;
>+  const Packet& BeginPop() override;
>+  void CancelPop(const Packet& packet) override;
>+  void FinalizePop(const Packet& packet) override;
>+  bool Empty() const override;
>+  size_t SizeInPackets() const override;
>+  uint64_t SizeInBytes() const override;
>+  int64_t OldestEnqueueTimeMs() const override;
>+  void UpdateQueueTime(int64_t timestamp_ms) override;
>+  void SetPauseState(bool paused, int64_t timestamp_ms) override;
>+  int64_t AverageQueueTimeMs() const override;
> 
>  private:
>   // Try to add a packet to the set of ssrc/seqno identifiers currently in the
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/packet_queue_interface.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/packet_queue_interface.cc
>new file mode 100644
>index 00000000000..a82d7df4f2f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/packet_queue_interface.cc
>@@ -0,0 +1,46 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/pacing/packet_queue_interface.h"
>+
>+namespace webrtc {
>+
>+PacketQueueInterface::Packet::Packet(RtpPacketSender::Priority priority,
>+                                     uint32_t ssrc,
>+                                     uint16_t seq_number,
>+                                     int64_t capture_time_ms,
>+                                     int64_t enqueue_time_ms,
>+                                     size_t length_in_bytes,
>+                                     bool retransmission,
>+                                     uint64_t enqueue_order)
>+    : priority(priority),
>+      ssrc(ssrc),
>+      sequence_number(seq_number),
>+      capture_time_ms(capture_time_ms),
>+      enqueue_time_ms(enqueue_time_ms),
>+      sum_paused_ms(0),
>+      bytes(length_in_bytes),
>+      retransmission(retransmission),
>+      enqueue_order(enqueue_order) {}
>+
>+PacketQueueInterface::Packet::Packet(const Packet& other) = default;
>+
>+PacketQueueInterface::Packet::~Packet() {}
>+
>+bool PacketQueueInterface::Packet::operator<(
>+    const PacketQueueInterface::Packet& other) const {
>+  if (priority != other.priority)
>+    return priority > other.priority;
>+  if (retransmission != other.retransmission)
>+    return other.retransmission;
>+
>+  return enqueue_order > other.enqueue_order;
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/packet_queue_interface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/packet_queue_interface.h
>new file mode 100644
>index 00000000000..81a8b05b1b9
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/packet_queue_interface.h
>@@ -0,0 +1,69 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_PACING_PACKET_QUEUE_INTERFACE_H_
>+#define MODULES_PACING_PACKET_QUEUE_INTERFACE_H_
>+
>+#include <stdint.h>
>+
>+#include <list>
>+#include <queue>
>+#include <set>
>+
>+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>+
>+namespace webrtc {
>+
>+class PacketQueueInterface {
>+ public:
>+  PacketQueueInterface() = default;
>+  virtual ~PacketQueueInterface() = default;
>+
>+  struct Packet {
>+    Packet(RtpPacketSender::Priority priority,
>+           uint32_t ssrc,
>+           uint16_t seq_number,
>+           int64_t capture_time_ms,
>+           int64_t enqueue_time_ms,
>+           size_t length_in_bytes,
>+           bool retransmission,
>+           uint64_t enqueue_order);
>+    Packet(const Packet& other);
>+    virtual ~Packet();
>+    bool operator<(const Packet& other) const;
>+
>+    RtpPacketSender::Priority priority;
>+    uint32_t ssrc;
>+    uint16_t sequence_number;
>+    int64_t capture_time_ms;  // Absolute time of frame capture.
>+    int64_t enqueue_time_ms;  // Absolute time of pacer queue entry.
>+    int64_t sum_paused_ms;
>+    size_t bytes;
>+    bool retransmission;
>+    uint64_t enqueue_order;
>+    std::list<Packet>::iterator this_it;
>+    std::multiset<int64_t>::iterator enqueue_time_it;
>+  };
>+
>+  virtual void Push(const Packet& packet) = 0;
>+  virtual const Packet& BeginPop() = 0;
>+  virtual void CancelPop(const Packet& packet) = 0;
>+  virtual void FinalizePop(const Packet& packet) = 0;
>+  virtual bool Empty() const = 0;
>+  virtual size_t SizeInPackets() const = 0;
>+  virtual uint64_t SizeInBytes() const = 0;
>+  virtual int64_t OldestEnqueueTimeMs() const = 0;
>+  virtual void UpdateQueueTime(int64_t timestamp_ms) = 0;
>+  virtual void SetPauseState(bool paused, int64_t timestamp_ms) = 0;
>+  virtual int64_t AverageQueueTimeMs() const = 0;
>+};
>+}  // namespace webrtc
>+
>+#endif  // MODULES_PACING_PACKET_QUEUE_INTERFACE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/packet_router.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/packet_router.cc
>index e3d935e65c8..b27c73c2f0c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/packet_router.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/packet_router.cc
>@@ -28,7 +28,8 @@ constexpr int kRembSendIntervalMs = 200;
> }  // namespace
> 
> PacketRouter::PacketRouter()
>-    : last_remb_time_ms_(rtc::TimeMillis()),
>+    : last_send_module_(nullptr),
>+      last_remb_time_ms_(rtc::TimeMillis()),
>       last_send_bitrate_bps_(0),
>       bitrate_bps_(0),
>       max_bitrate_bps_(std::numeric_limits<decltype(max_bitrate_bps_)>::max()),
>@@ -67,6 +68,9 @@ void PacketRouter::RemoveSendRtpModule(RtpRtcp* rtp_module) {
>       std::find(rtp_send_modules_.begin(), rtp_send_modules_.end(), rtp_module);
>   RTC_DCHECK(it != rtp_send_modules_.end());
>   rtp_send_modules_.erase(it);
>+  if (last_send_module_ == rtp_module) {
>+    last_send_module_ = nullptr;
>+  }
> }
> 
> void PacketRouter::AddReceiveRtpModule(RtcpFeedbackSenderInterface* rtcp_sender,
>@@ -98,12 +102,18 @@ bool PacketRouter::TimeToSendPacket(uint32_t ssrc,
>                                     int64_t capture_timestamp,
>                                     bool retransmission,
>                                     const PacedPacketInfo& pacing_info) {
>-  RTC_DCHECK_RUNS_SERIALIZED(&pacer_race_);
>   rtc::CritScope cs(&modules_crit_);
>   for (auto* rtp_module : rtp_send_modules_) {
>-    if (!rtp_module->SendingMedia())
>+    if (!rtp_module->SendingMedia()) {
>       continue;
>+    }
>     if (ssrc == rtp_module->SSRC() || ssrc == rtp_module->FlexfecSsrc()) {
>+      if ((rtp_module->RtxSendStatus() & kRtxRedundantPayloads) &&
>+          rtp_module->HasBweExtensions()) {
>+        // This is now the last module to send media, and has the desired
>+        // properties needed for payload based padding. Cache it for later use.
>+        last_send_module_ = rtp_module;
>+      }
>       return rtp_module->TimeToSendPacket(ssrc, sequence_number,
>                                           capture_timestamp, retransmission,
>                                           pacing_info);
>@@ -114,9 +124,25 @@ bool PacketRouter::TimeToSendPacket(uint32_t ssrc,
> 
> size_t PacketRouter::TimeToSendPadding(size_t bytes_to_send,
>                                        const PacedPacketInfo& pacing_info) {
>-  RTC_DCHECK_RUNS_SERIALIZED(&pacer_race_);
>   size_t total_bytes_sent = 0;
>   rtc::CritScope cs(&modules_crit_);
>+  // First try on the last rtp module to have sent media. This increases the
>+  // the chance that any payload based padding will be useful as it will be
>+  // somewhat distributed over modules according the packet rate, even if it
>+  // will be more skewed towards the highest bitrate stream. At the very least
>+  // this prevents sending payload padding on a disabled stream where it's
>+  // guaranteed not to be useful.
>+  if (last_send_module_ != nullptr) {
>+    RTC_DCHECK(std::find(rtp_send_modules_.begin(), rtp_send_modules_.end(),
>+                         last_send_module_) != rtp_send_modules_.end());
>+    RTC_DCHECK(last_send_module_->HasBweExtensions());
>+    total_bytes_sent += last_send_module_->TimeToSendPadding(
>+        bytes_to_send - total_bytes_sent, pacing_info);
>+    if (total_bytes_sent >= bytes_to_send) {
>+      return total_bytes_sent;
>+    }
>+  }
>+
>   // Rtp modules are ordered by which stream can most benefit from padding.
>   for (RtpRtcp* module : rtp_send_modules_) {
>     if (module->SendingMedia() && module->HasBweExtensions()) {
>@@ -223,18 +249,19 @@ bool PacketRouter::SendRemb(int64_t bitrate_bps,
> }
> 
> bool PacketRouter::SendTransportFeedback(rtcp::TransportFeedback* packet) {
>-  RTC_DCHECK_RUNS_SERIALIZED(&pacer_race_);
>   rtc::CritScope cs(&modules_crit_);
>   // Prefer send modules.
>   for (auto* rtp_module : rtp_send_modules_) {
>     packet->SetSenderSsrc(rtp_module->SSRC());
>-    if (rtp_module->SendFeedbackPacket(*packet))
>+    if (rtp_module->SendFeedbackPacket(*packet)) {
>       return true;
>+    }
>   }
>   for (auto* rtcp_sender : rtcp_feedback_senders_) {
>     packet->SetSenderSsrc(rtcp_sender->SSRC());
>-    if (rtcp_sender->SendFeedbackPacket(*packet))
>+    if (rtcp_sender->SendFeedbackPacket(*packet)) {
>       return true;
>+    }
>   }
>   return false;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/packet_router.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/packet_router.h
>index 9597896353f..f01e9ef4933 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/packet_router.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/packet_router.h
>@@ -91,10 +91,11 @@ class PacketRouter : public PacedSender::PacketSender,
>   void UnsetActiveRembModule() RTC_EXCLUSIVE_LOCKS_REQUIRED(modules_crit_);
>   void DetermineActiveRembModule() RTC_EXCLUSIVE_LOCKS_REQUIRED(modules_crit_);
> 
>-  rtc::RaceChecker pacer_race_;
>   rtc::CriticalSection modules_crit_;
>   // Rtp and Rtcp modules of the rtp senders.
>   std::list<RtpRtcp*> rtp_send_modules_ RTC_GUARDED_BY(modules_crit_);
>+  // The last module used to send media.
>+  RtpRtcp* last_send_module_ RTC_GUARDED_BY(modules_crit_);
>   // Rtcp modules of the rtp receivers.
>   std::vector<RtcpFeedbackSenderInterface*> rtcp_feedback_senders_
>       RTC_GUARDED_BY(modules_crit_);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/packet_router_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/packet_router_unittest.cc
>index 9acd080dd56..3da134a8040 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/packet_router_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/packet_router_unittest.cc
>@@ -234,11 +234,10 @@ TEST(PacketRouterTest, TimeToSendPadding) {
>   EXPECT_CALL(rtp_1, TimeToSendPadding(requested_padding_bytes, _)).Times(0);
>   EXPECT_CALL(rtp_2, SendingMedia()).Times(1).WillOnce(Return(false));
>   EXPECT_CALL(rtp_2, TimeToSendPadding(_, _)).Times(0);
>-  EXPECT_EQ(0u,
>-            packet_router.TimeToSendPadding(
>-                requested_padding_bytes,
>-                PacedPacketInfo(PacedPacketInfo::kNotAProbe, kProbeMinBytes,
>-                                kProbeMinBytes)));
>+  EXPECT_EQ(0u, packet_router.TimeToSendPadding(
>+                    requested_padding_bytes,
>+                    PacedPacketInfo(PacedPacketInfo::kNotAProbe, kProbeMinBytes,
>+                                    kProbeMinBytes)));
> 
>   // Only one module has BWE extensions.
>   EXPECT_CALL(rtp_1, SendingMedia()).Times(1).WillOnce(Return(true));
>@@ -262,13 +261,90 @@ TEST(PacketRouterTest, TimeToSendPadding) {
>   EXPECT_CALL(rtp_2, SendingMedia()).Times(1).WillOnce(Return(true));
>   EXPECT_CALL(rtp_2, HasBweExtensions()).Times(1).WillOnce(Return(true));
>   EXPECT_CALL(rtp_2, TimeToSendPadding(requested_padding_bytes, _)).Times(1);
>-  EXPECT_EQ(0u,
>-            packet_router.TimeToSendPadding(
>-                requested_padding_bytes,
>-                PacedPacketInfo(PacedPacketInfo::kNotAProbe, kProbeMinBytes,
>-                                kProbeMinBytes)));
>+  EXPECT_EQ(0u, packet_router.TimeToSendPadding(
>+                    requested_padding_bytes,
>+                    PacedPacketInfo(PacedPacketInfo::kNotAProbe, kProbeMinBytes,
>+                                    kProbeMinBytes)));
>+
>+  packet_router.RemoveSendRtpModule(&rtp_2);
>+}
>+
>+TEST(PacketRouterTest, PadsOnLastActiveMediaStream) {
>+  PacketRouter packet_router;
>+
>+  const uint16_t kSsrc1 = 1234;
>+  const uint16_t kSsrc2 = 4567;
>+  const uint16_t kSsrc3 = 8901;
>+
>+  // First two rtp modules send media and have rtx.
>+  NiceMock<MockRtpRtcp> rtp_1;
>+  EXPECT_CALL(rtp_1, RtxSendStatus())
>+      .WillRepeatedly(Return(kRtxRedundantPayloads));
>+  EXPECT_CALL(rtp_1, SSRC()).WillRepeatedly(Return(kSsrc1));
>+  EXPECT_CALL(rtp_1, SendingMedia()).WillRepeatedly(Return(true));
>+  EXPECT_CALL(rtp_1, HasBweExtensions()).WillRepeatedly(Return(true));
>+
>+  NiceMock<MockRtpRtcp> rtp_2;
>+  EXPECT_CALL(rtp_2, RtxSendStatus())
>+      .WillRepeatedly(Return(kRtxRedundantPayloads));
>+  EXPECT_CALL(rtp_2, SSRC()).WillRepeatedly(Return(kSsrc2));
>+  EXPECT_CALL(rtp_2, SendingMedia()).WillRepeatedly(Return(true));
>+  EXPECT_CALL(rtp_2, HasBweExtensions()).WillRepeatedly(Return(true));
> 
>+  // Third module is sending media, but does not support rtx.
>+  NiceMock<MockRtpRtcp> rtp_3;
>+  EXPECT_CALL(rtp_3, RtxSendStatus()).WillRepeatedly(Return(kRtxOff));
>+  EXPECT_CALL(rtp_3, SSRC()).WillRepeatedly(Return(kSsrc3));
>+  EXPECT_CALL(rtp_3, SendingMedia()).WillRepeatedly(Return(true));
>+  EXPECT_CALL(rtp_3, HasBweExtensions()).WillRepeatedly(Return(true));
>+
>+  packet_router.AddSendRtpModule(&rtp_1, false);
>+  packet_router.AddSendRtpModule(&rtp_2, false);
>+  packet_router.AddSendRtpModule(&rtp_3, false);
>+
>+  const size_t kPaddingBytes = 100;
>+
>+  // Initially, padding will be sent on last added rtp module that sends media
>+  // and supports rtx.
>+  EXPECT_CALL(rtp_2, TimeToSendPadding(kPaddingBytes, _))
>+      .Times(1)
>+      .WillOnce(Return(kPaddingBytes));
>+  packet_router.TimeToSendPadding(kPaddingBytes, PacedPacketInfo());
>+
>+  // Send media on first module. Padding should be sent on that module.
>+  EXPECT_CALL(rtp_1, TimeToSendPacket(kSsrc1, _, _, _, _));
>+  packet_router.TimeToSendPacket(kSsrc1, 0, 0, false, PacedPacketInfo());
>+
>+  EXPECT_CALL(rtp_1, TimeToSendPadding(kPaddingBytes, _))
>+      .Times(1)
>+      .WillOnce(Return(kPaddingBytes));
>+  packet_router.TimeToSendPadding(kPaddingBytes, PacedPacketInfo());
>+
>+  // Send media on second module. Padding should be sent there.
>+  EXPECT_CALL(rtp_2, TimeToSendPacket(kSsrc2, _, _, _, _));
>+  packet_router.TimeToSendPacket(kSsrc2, 0, 0, false, PacedPacketInfo());
>+
>+  EXPECT_CALL(rtp_2, TimeToSendPadding(kPaddingBytes, _))
>+      .Times(1)
>+      .WillOnce(Return(kPaddingBytes));
>+  packet_router.TimeToSendPadding(kPaddingBytes, PacedPacketInfo());
>+
>+  // Remove second module, padding should now fall back to first module.
>   packet_router.RemoveSendRtpModule(&rtp_2);
>+  EXPECT_CALL(rtp_1, TimeToSendPadding(kPaddingBytes, _))
>+      .Times(1)
>+      .WillOnce(Return(kPaddingBytes));
>+  packet_router.TimeToSendPadding(kPaddingBytes, PacedPacketInfo());
>+
>+  // Remove first module too, leaving only the one without rtx.
>+  packet_router.RemoveSendRtpModule(&rtp_1);
>+
>+  EXPECT_CALL(rtp_3, TimeToSendPadding(kPaddingBytes, _))
>+      .Times(1)
>+      .WillOnce(Return(kPaddingBytes));
>+  packet_router.TimeToSendPadding(kPaddingBytes, PacedPacketInfo());
>+
>+  packet_router.RemoveSendRtpModule(&rtp_3);
> }
> 
> TEST(PacketRouterTest, SenderOnlyFunctionsRespectSendingMedia) {
>@@ -282,14 +358,14 @@ TEST(PacketRouterTest, SenderOnlyFunctionsRespectSendingMedia) {
>   // Verify that TimeToSendPacket does not end up in a receiver.
>   EXPECT_CALL(rtp, TimeToSendPacket(_, _, _, _, _)).Times(0);
>   EXPECT_TRUE(packet_router.TimeToSendPacket(
>-      kSsrc, 1, 1, false, PacedPacketInfo(PacedPacketInfo::kNotAProbe,
>-                                          kProbeMinBytes, kProbeMinBytes)));
>+      kSsrc, 1, 1, false,
>+      PacedPacketInfo(PacedPacketInfo::kNotAProbe, kProbeMinBytes,
>+                      kProbeMinBytes)));
>   // Verify that TimeToSendPadding does not end up in a receiver.
>   EXPECT_CALL(rtp, TimeToSendPadding(_, _)).Times(0);
>-  EXPECT_EQ(0u,
>-            packet_router.TimeToSendPadding(
>-                200, PacedPacketInfo(PacedPacketInfo::kNotAProbe,
>-                                     kProbeMinBytes, kProbeMinBytes)));
>+  EXPECT_EQ(0u, packet_router.TimeToSendPadding(
>+                    200, PacedPacketInfo(PacedPacketInfo::kNotAProbe,
>+                                         kProbeMinBytes, kProbeMinBytes)));
> 
>   packet_router.RemoveSendRtpModule(&rtp);
> }
>@@ -379,7 +455,7 @@ TEST(PacketRouterRembTest, LowerEstimateToSendRemb) {
>   packet_router.OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
> 
>   // Call OnReceiveBitrateChanged twice to get a first estimate.
>-  clock.AdvanceTime(rtc::TimeDelta::FromMilliseconds(1000));
>+  clock.AdvanceTime(TimeDelta::ms(1000));
>   EXPECT_CALL(rtp, SetRemb(bitrate_estimate, ssrcs)).Times(1);
>   packet_router.OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
> 
>@@ -405,7 +481,7 @@ TEST(PacketRouterRembTest, VerifyIncreasingAndDecreasing) {
> 
>   // Call OnReceiveBitrateChanged twice to get a first estimate.
>   EXPECT_CALL(rtp, SetRemb(bitrate_estimate[0], ssrcs)).Times(1);
>-  clock.AdvanceTime(rtc::TimeDelta::FromMilliseconds(1000));
>+  clock.AdvanceTime(TimeDelta::ms(1000));
>   packet_router.OnReceiveBitrateChanged(ssrcs, bitrate_estimate[0]);
> 
>   packet_router.OnReceiveBitrateChanged(ssrcs, bitrate_estimate[1] + 100);
>@@ -430,7 +506,7 @@ TEST(PacketRouterRembTest, NoRembForIncreasedBitrate) {
> 
>   // Call OnReceiveBitrateChanged twice to get a first estimate.
>   EXPECT_CALL(rtp, SetRemb(bitrate_estimate, ssrcs)).Times(1);
>-  clock.AdvanceTime(rtc::TimeDelta::FromMilliseconds(1000));
>+  clock.AdvanceTime(TimeDelta::ms(1000));
>   packet_router.OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
> 
>   // Increased estimate shouldn't trigger a callback right away.
>@@ -459,7 +535,7 @@ TEST(PacketRouterRembTest, ChangeSendRtpModule) {
>   packet_router.OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
> 
>   // Call OnReceiveBitrateChanged twice to get a first estimate.
>-  clock.AdvanceTime(rtc::TimeDelta::FromMilliseconds(1000));
>+  clock.AdvanceTime(TimeDelta::ms(1000));
>   EXPECT_CALL(rtp_send, SetRemb(bitrate_estimate, ssrcs)).Times(1);
>   packet_router.OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
> 
>@@ -492,7 +568,7 @@ TEST(PacketRouterRembTest, OnlyOneRembForRepeatedOnReceiveBitrateChanged) {
>   packet_router.OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
> 
>   // Call OnReceiveBitrateChanged twice to get a first estimate.
>-  clock.AdvanceTime(rtc::TimeDelta::FromMilliseconds(1000));
>+  clock.AdvanceTime(TimeDelta::ms(1000));
>   EXPECT_CALL(rtp, SetRemb(_, _)).Times(1);
>   packet_router.OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
> 
>@@ -510,7 +586,7 @@ TEST(PacketRouterRembTest, OnlyOneRembForRepeatedOnReceiveBitrateChanged) {
> TEST(PacketRouterRembTest, SetMaxDesiredReceiveBitrateLimitsSetRemb) {
>   rtc::ScopedFakeClock clock;
>   PacketRouter packet_router;
>-  clock.AdvanceTime(rtc::TimeDelta::FromMilliseconds(1000));
>+  clock.AdvanceTime(TimeDelta::ms(1000));
>   NiceMock<MockRtpRtcp> remb_sender;
>   constexpr bool remb_candidate = true;
>   packet_router.AddSendRtpModule(&remb_sender, remb_candidate);
>@@ -522,7 +598,7 @@ TEST(PacketRouterRembTest, SetMaxDesiredReceiveBitrateLimitsSetRemb) {
>   const std::vector<uint32_t> ssrcs = {1234};
>   packet_router.SetMaxDesiredReceiveBitrate(cap_bitrate);
>   packet_router.OnReceiveBitrateChanged(ssrcs, cap_bitrate + 5000);
>-  clock.AdvanceTime(rtc::TimeDelta::FromMilliseconds(1000));
>+  clock.AdvanceTime(TimeDelta::ms(1000));
>   packet_router.OnReceiveBitrateChanged(ssrcs, cap_bitrate - 5000);
> 
>   // Test tear-down.
>@@ -533,7 +609,7 @@ TEST(PacketRouterRembTest,
>      SetMaxDesiredReceiveBitrateTriggersRembWhenMoreRestrictive) {
>   rtc::ScopedFakeClock clock;
>   PacketRouter packet_router;
>-  clock.AdvanceTime(rtc::TimeDelta::FromMilliseconds(1000));
>+  clock.AdvanceTime(TimeDelta::ms(1000));
>   NiceMock<MockRtpRtcp> remb_sender;
>   constexpr bool remb_candidate = true;
>   packet_router.AddSendRtpModule(&remb_sender, remb_candidate);
>@@ -555,7 +631,7 @@ TEST(PacketRouterRembTest,
>      SetMaxDesiredReceiveBitrateDoesNotTriggerRembWhenAsRestrictive) {
>   rtc::ScopedFakeClock clock;
>   PacketRouter packet_router;
>-  clock.AdvanceTime(rtc::TimeDelta::FromMilliseconds(1000));
>+  clock.AdvanceTime(TimeDelta::ms(1000));
>   NiceMock<MockRtpRtcp> remb_sender;
>   constexpr bool remb_candidate = true;
>   packet_router.AddSendRtpModule(&remb_sender, remb_candidate);
>@@ -577,7 +653,7 @@ TEST(PacketRouterRembTest,
>      SetMaxDesiredReceiveBitrateDoesNotTriggerRembWhenLessRestrictive) {
>   rtc::ScopedFakeClock clock;
>   PacketRouter packet_router;
>-  clock.AdvanceTime(rtc::TimeDelta::FromMilliseconds(1000));
>+  clock.AdvanceTime(TimeDelta::ms(1000));
>   NiceMock<MockRtpRtcp> remb_sender;
>   constexpr bool remb_candidate = true;
>   packet_router.AddSendRtpModule(&remb_sender, remb_candidate);
>@@ -599,7 +675,7 @@ TEST(PacketRouterRembTest,
>      SetMaxDesiredReceiveBitrateTriggersRembWhenNoRecentMeasure) {
>   rtc::ScopedFakeClock clock;
>   PacketRouter packet_router;
>-  clock.AdvanceTime(rtc::TimeDelta::FromMilliseconds(1000));
>+  clock.AdvanceTime(TimeDelta::ms(1000));
>   NiceMock<MockRtpRtcp> remb_sender;
>   constexpr bool remb_candidate = true;
>   packet_router.AddSendRtpModule(&remb_sender, remb_candidate);
>@@ -609,7 +685,7 @@ TEST(PacketRouterRembTest,
>   const std::vector<uint32_t> ssrcs = {1234};
>   EXPECT_CALL(remb_sender, SetRemb(measured_bitrate_bps, _));
>   packet_router.OnReceiveBitrateChanged(ssrcs, measured_bitrate_bps);
>-  clock.AdvanceTime(rtc::TimeDelta::FromMilliseconds(1000));
>+  clock.AdvanceTime(TimeDelta::ms(1000));
> 
>   EXPECT_CALL(remb_sender, SetRemb(cap_bitrate_bps, _));
>   packet_router.SetMaxDesiredReceiveBitrate(cap_bitrate_bps);
>@@ -622,7 +698,7 @@ TEST(PacketRouterRembTest,
>      SetMaxDesiredReceiveBitrateTriggersRembWhenNoMeasures) {
>   rtc::ScopedFakeClock clock;
>   PacketRouter packet_router;
>-  clock.AdvanceTime(rtc::TimeDelta::FromMilliseconds(1000));
>+  clock.AdvanceTime(TimeDelta::ms(1000));
>   NiceMock<MockRtpRtcp> remb_sender;
>   constexpr bool remb_candidate = true;
>   packet_router.AddSendRtpModule(&remb_sender, remb_candidate);
>@@ -656,7 +732,7 @@ TEST(PacketRouterRembTest, NoSendingRtpModule) {
>   packet_router.OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
> 
>   // Call OnReceiveBitrateChanged twice to get a first estimate.
>-  clock.AdvanceTime(rtc::TimeDelta::FromMilliseconds(1000));
>+  clock.AdvanceTime(TimeDelta::ms(1000));
>   EXPECT_CALL(rtp, SetRemb(bitrate_estimate, ssrcs)).Times(1);
>   packet_router.OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
> 
>@@ -680,7 +756,7 @@ TEST(PacketRouterRembTest, NonCandidateSendRtpModuleNotUsedForRemb) {
>   constexpr uint32_t bitrate_estimate = 456;
>   const std::vector<uint32_t> ssrcs = {1234};
>   EXPECT_CALL(module, SetRemb(_, _)).Times(0);
>-  clock.AdvanceTime(rtc::TimeDelta::FromMilliseconds(1000));
>+  clock.AdvanceTime(TimeDelta::ms(1000));
>   packet_router.OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
> 
>   // Test tear-down
>@@ -699,7 +775,7 @@ TEST(PacketRouterRembTest, CandidateSendRtpModuleUsedForRemb) {
>   constexpr uint32_t bitrate_estimate = 456;
>   const std::vector<uint32_t> ssrcs = {1234};
>   EXPECT_CALL(module, SetRemb(bitrate_estimate, ssrcs)).Times(1);
>-  clock.AdvanceTime(rtc::TimeDelta::FromMilliseconds(1000));
>+  clock.AdvanceTime(TimeDelta::ms(1000));
>   packet_router.OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
> 
>   // Test tear-down
>@@ -718,7 +794,7 @@ TEST(PacketRouterRembTest, NonCandidateReceiveRtpModuleNotUsedForRemb) {
>   constexpr uint32_t bitrate_estimate = 456;
>   const std::vector<uint32_t> ssrcs = {1234};
>   EXPECT_CALL(module, SetRemb(_, _)).Times(0);
>-  clock.AdvanceTime(rtc::TimeDelta::FromMilliseconds(1000));
>+  clock.AdvanceTime(TimeDelta::ms(1000));
>   packet_router.OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
> 
>   // Test tear-down
>@@ -737,7 +813,7 @@ TEST(PacketRouterRembTest, CandidateReceiveRtpModuleUsedForRemb) {
>   constexpr uint32_t bitrate_estimate = 456;
>   const std::vector<uint32_t> ssrcs = {1234};
>   EXPECT_CALL(module, SetRemb(bitrate_estimate, ssrcs)).Times(1);
>-  clock.AdvanceTime(rtc::TimeDelta::FromMilliseconds(1000));
>+  clock.AdvanceTime(TimeDelta::ms(1000));
>   packet_router.OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
> 
>   // Test tear-down
>@@ -764,7 +840,7 @@ TEST(PacketRouterRembTest,
>   EXPECT_CALL(send_module, SetRemb(bitrate_estimate, ssrcs)).Times(1);
>   EXPECT_CALL(receive_module, SetRemb(_, _)).Times(0);
> 
>-  clock.AdvanceTime(rtc::TimeDelta::FromMilliseconds(1000));
>+  clock.AdvanceTime(TimeDelta::ms(1000));
>   packet_router.OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
> 
>   // Test tear-down
>@@ -792,7 +868,7 @@ TEST(PacketRouterRembTest,
>   EXPECT_CALL(send_module, SetRemb(bitrate_estimate, ssrcs)).Times(1);
>   EXPECT_CALL(receive_module, SetRemb(_, _)).Times(0);
> 
>-  clock.AdvanceTime(rtc::TimeDelta::FromMilliseconds(1000));
>+  clock.AdvanceTime(TimeDelta::ms(1000));
>   packet_router.OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
> 
>   // Test tear-down
>@@ -819,7 +895,7 @@ TEST(PacketRouterRembTest, ReceiveModuleTakesOverWhenLastSendModuleRemoved) {
>   EXPECT_CALL(send_module, SetRemb(_, _)).Times(0);
>   EXPECT_CALL(receive_module, SetRemb(bitrate_estimate, ssrcs)).Times(1);
> 
>-  clock.AdvanceTime(rtc::TimeDelta::FromMilliseconds(1000));
>+  clock.AdvanceTime(TimeDelta::ms(1000));
>   packet_router.OnReceiveBitrateChanged(ssrcs, bitrate_estimate);
> 
>   // Test tear-down
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/packet_queue2.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/round_robin_packet_queue.cc
>similarity index 83%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/packet_queue2.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/round_robin_packet_queue.cc
>index 6aee807af36..a7cb8befda4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/packet_queue2.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/round_robin_packet_queue.cc
>@@ -8,7 +8,7 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "modules/pacing/packet_queue2.h"
>+#include "modules/pacing/round_robin_packet_queue.h"
> 
> #include <algorithm>
> 
>@@ -17,17 +17,16 @@
> 
> namespace webrtc {
> 
>-PacketQueue2::Stream::Stream() : bytes(0) {}
>-PacketQueue2::Stream::~Stream() {}
>+RoundRobinPacketQueue::Stream::Stream() : bytes(0) {}
>+RoundRobinPacketQueue::Stream::Stream(const Stream& stream) = default;
>+RoundRobinPacketQueue::Stream::~Stream() {}
> 
>-PacketQueue2::PacketQueue2(const Clock* clock)
>-    : PacketQueue(clock),
>-      clock_(clock),
>-      time_last_updated_(clock_->TimeInMilliseconds()) {}
>+RoundRobinPacketQueue::RoundRobinPacketQueue(const Clock* clock)
>+    : clock_(clock), time_last_updated_(clock_->TimeInMilliseconds()) {}
> 
>-PacketQueue2::~PacketQueue2() {}
>+RoundRobinPacketQueue::~RoundRobinPacketQueue() {}
> 
>-void PacketQueue2::Push(const Packet& packet_to_insert) {
>+void RoundRobinPacketQueue::Push(const Packet& packet_to_insert) {
>   Packet packet(packet_to_insert);
> 
>   auto stream_info_it = streams_.find(packet.ssrc);
>@@ -70,7 +69,7 @@ void PacketQueue2::Push(const Packet& packet_to_insert) {
>   size_bytes_ += packet.bytes;
> }
> 
>-const PacketQueue2::Packet& PacketQueue2::BeginPop() {
>+const PacketQueueInterface::Packet& RoundRobinPacketQueue::BeginPop() {
>   RTC_CHECK(!pop_packet_ && !pop_stream_);
> 
>   Stream* stream = GetHighestPriorityStream();
>@@ -81,15 +80,14 @@ const PacketQueue2::Packet& PacketQueue2::BeginPop() {
>   return *pop_packet_;
> }
> 
>-void PacketQueue2::CancelPop(const Packet& packet) {
>+void RoundRobinPacketQueue::CancelPop(const Packet& packet) {
>   RTC_CHECK(pop_packet_ && pop_stream_);
>   (*pop_stream_)->packet_queue.push(*pop_packet_);
>   pop_packet_.reset();
>   pop_stream_.reset();
> }
> 
>-void PacketQueue2::FinalizePop(const Packet& packet) {
>-  RTC_CHECK(!paused_);
>+void RoundRobinPacketQueue::FinalizePop(const Packet& packet) {
>   if (!Empty()) {
>     RTC_CHECK(pop_packet_ && pop_stream_);
>     Stream* stream = *pop_stream_;
>@@ -137,28 +135,28 @@ void PacketQueue2::FinalizePop(const Packet& packet) {
>   }
> }
> 
>-bool PacketQueue2::Empty() const {
>+bool RoundRobinPacketQueue::Empty() const {
>   RTC_CHECK((!stream_priorities_.empty() && size_packets_ > 0) ||
>             (stream_priorities_.empty() && size_packets_ == 0));
>   return stream_priorities_.empty();
> }
> 
>-size_t PacketQueue2::SizeInPackets() const {
>+size_t RoundRobinPacketQueue::SizeInPackets() const {
>   return size_packets_;
> }
> 
>-uint64_t PacketQueue2::SizeInBytes() const {
>+uint64_t RoundRobinPacketQueue::SizeInBytes() const {
>   return size_bytes_;
> }
> 
>-int64_t PacketQueue2::OldestEnqueueTimeMs() const {
>+int64_t RoundRobinPacketQueue::OldestEnqueueTimeMs() const {
>   if (Empty())
>     return 0;
>   RTC_CHECK(!enqueue_times_.empty());
>   return *enqueue_times_.begin();
> }
> 
>-void PacketQueue2::UpdateQueueTime(int64_t timestamp_ms) {
>+void RoundRobinPacketQueue::UpdateQueueTime(int64_t timestamp_ms) {
>   RTC_CHECK_GE(timestamp_ms, time_last_updated_);
>   if (timestamp_ms == time_last_updated_)
>     return;
>@@ -174,20 +172,21 @@ void PacketQueue2::UpdateQueueTime(int64_t timestamp_ms) {
>   time_last_updated_ = timestamp_ms;
> }
> 
>-void PacketQueue2::SetPauseState(bool paused, int64_t timestamp_ms) {
>+void RoundRobinPacketQueue::SetPauseState(bool paused, int64_t timestamp_ms) {
>   if (paused_ == paused)
>     return;
>   UpdateQueueTime(timestamp_ms);
>   paused_ = paused;
> }
> 
>-int64_t PacketQueue2::AverageQueueTimeMs() const {
>+int64_t RoundRobinPacketQueue::AverageQueueTimeMs() const {
>   if (Empty())
>     return 0;
>   return queue_time_sum_ms_ / size_packets_;
> }
> 
>-PacketQueue2::Stream* PacketQueue2::GetHighestPriorityStream() {
>+RoundRobinPacketQueue::Stream*
>+RoundRobinPacketQueue::GetHighestPriorityStream() {
>   RTC_CHECK(!stream_priorities_.empty());
>   uint32_t ssrc = stream_priorities_.begin()->second;
> 
>@@ -198,7 +197,7 @@ PacketQueue2::Stream* PacketQueue2::GetHighestPriorityStream() {
>   return &stream_info_it->second;
> }
> 
>-bool PacketQueue2::IsSsrcScheduled(uint32_t ssrc) const {
>+bool RoundRobinPacketQueue::IsSsrcScheduled(uint32_t ssrc) const {
>   for (const auto& scheduled_stream : stream_priorities_) {
>     if (scheduled_stream.second == ssrc)
>       return true;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/packet_queue2.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/round_robin_packet_queue.h
>similarity index 85%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/packet_queue2.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/round_robin_packet_queue.h
>index 8c18a0fb60b..5dd68cd0ece 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/packet_queue2.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/pacing/round_robin_packet_queue.h
>@@ -8,24 +8,24 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#ifndef MODULES_PACING_PACKET_QUEUE2_H_
>-#define MODULES_PACING_PACKET_QUEUE2_H_
>+#ifndef MODULES_PACING_ROUND_ROBIN_PACKET_QUEUE_H_
>+#define MODULES_PACING_ROUND_ROBIN_PACKET_QUEUE_H_
> 
> #include <map>
> #include <queue>
> #include <set>
> 
>-#include "modules/pacing/packet_queue.h"
>+#include "modules/pacing/packet_queue_interface.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> 
> namespace webrtc {
> 
>-class PacketQueue2 : public PacketQueue {
>+class RoundRobinPacketQueue : public PacketQueueInterface {
>  public:
>-  explicit PacketQueue2(const Clock* clock);
>-  ~PacketQueue2() override;
>+  explicit RoundRobinPacketQueue(const Clock* clock);
>+  ~RoundRobinPacketQueue() override;
> 
>-  using Packet = PacketQueue::Packet;
>+  using Packet = PacketQueueInterface::Packet;
> 
>   void Push(const Packet& packet) override;
>   const Packet& BeginPop() override;
>@@ -41,6 +41,7 @@ class PacketQueue2 : public PacketQueue {
>   void UpdateQueueTime(int64_t timestamp_ms) override;
>   void SetPauseState(bool paused, int64_t timestamp_ms) override;
> 
>+ private:
>   struct StreamPrioKey {
>     StreamPrioKey() = default;
>     StreamPrioKey(RtpPacketSender::Priority priority, int64_t bytes)
>@@ -58,6 +59,7 @@ class PacketQueue2 : public PacketQueue {
> 
>   struct Stream {
>     Stream();
>+    Stream(const Stream&);
> 
>     virtual ~Stream();
> 
>@@ -73,7 +75,6 @@ class PacketQueue2 : public PacketQueue {
>     std::multimap<StreamPrioKey, uint32_t>::iterator priority_it;
>   };
> 
>- private:
>   static constexpr size_t kMaxLeadingBytes = 1400;
> 
>   Stream* GetHighestPriorityStream();
>@@ -83,8 +84,8 @@ class PacketQueue2 : public PacketQueue {
> 
>   const Clock* const clock_;
>   int64_t time_last_updated_;
>-  rtc::Optional<Packet> pop_packet_;
>-  rtc::Optional<Stream*> pop_stream_;
>+  absl::optional<Packet> pop_packet_;
>+  absl::optional<Stream*> pop_stream_;
> 
>   bool paused_ = false;
>   size_t size_packets_ = 0;
>@@ -108,4 +109,4 @@ class PacketQueue2 : public PacketQueue {
> };
> }  // namespace webrtc
> 
>-#endif  // MODULES_PACING_PACKET_QUEUE2_H_
>+#endif  // MODULES_PACING_ROUND_ROBIN_PACKET_QUEUE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/BUILD.gn
>index 6bca26dfca5..12612032bff 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/BUILD.gn
>@@ -16,7 +16,6 @@ rtc_static_library("remote_bitrate_estimator") {
>     "bwe_defines.cc",
>     "include/bwe_defines.h",
>     "include/remote_bitrate_estimator.h",
>-    "include/send_time_history.h",
>     "inter_arrival.cc",
>     "inter_arrival.h",
>     "overuse_detector.cc",
>@@ -29,7 +28,6 @@ rtc_static_library("remote_bitrate_estimator") {
>     "remote_bitrate_estimator_single_stream.h",
>     "remote_estimator_proxy.cc",
>     "remote_estimator_proxy.h",
>-    "send_time_history.cc",
>     "test/bwe_test_logging.h",
>   ]
> 
>@@ -40,22 +38,17 @@ rtc_static_library("remote_bitrate_estimator") {
>     defines = [ "BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=0" ]
>   }
> 
>-  if (!build_with_chromium && is_clang) {
>-    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>-    suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-  }
>-
>   deps = [
>     "../..:webrtc_common",
>-    "../../:typedefs",
>-    "../../api:optional",
>     "../../modules:module_api",
>     "../../modules/rtp_rtcp:rtp_rtcp_format",
>     "../../rtc_base:checks",
>     "../../rtc_base:rtc_base_approved",
>+    "../../rtc_base:safe_minmax",
>     "../../system_wrappers",
>     "../../system_wrappers:field_trial_api",
>     "../../system_wrappers:metrics_api",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
> 
>@@ -72,10 +65,6 @@ if (!build_with_chromium) {
>       "../../test:rtp_test_utils",
>       "../rtp_rtcp",
>     ]
>-    if (is_clang) {
>-      # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>-      suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-    }
>   }
> }
> 
>@@ -102,6 +91,7 @@ if (rtc_include_tests) {
>       "test/estimators/congestion_window.h",
>       "test/estimators/max_bandwidth_filter.cc",
>       "test/estimators/max_bandwidth_filter.h",
>+      "test/estimators/min_rtt_filter.cc",
>       "test/estimators/min_rtt_filter.h",
>       "test/estimators/nada.cc",
>       "test/estimators/nada.h",
>@@ -113,6 +103,7 @@ if (rtc_include_tests) {
>       "test/estimators/tcp.h",
>       "test/metric_recorder.cc",
>       "test/metric_recorder.h",
>+      "test/packet.cc",
>       "test/packet.h",
>       "test/packet_receiver.cc",
>       "test/packet_receiver.h",
>@@ -126,46 +117,38 @@ if (rtc_include_tests) {
>       defines = [ "BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=0" ]
>     }
> 
>-    if (!build_with_chromium && is_clang) {
>-      # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>-      suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-    }
>-
>-    if (is_win) {
>-      cflags = [
>-        # TODO(kjellander): Bug 261: fix this warning.
>-        "/wd4373",  # virtual function override.
>-      ]
>-    }
>-
>     deps = [
>       ":remote_bitrate_estimator",
>       "..:module_api",
>       "../..:webrtc_common",
>-      "../../:typedefs",
>-      "../../api:optional",
>       "../../logging:mocks",
>       "../../rtc_base:checks",
>       "../../rtc_base:gtest_prod",
>       "../../rtc_base:rtc_base",
>       "../../rtc_base:rtc_base_approved",
>       "../../rtc_base:rtc_numerics",
>+      "../../rtc_base:safe_minmax",
>       "../../rtc_base/experiments:alr_experiment",
>+      "../../rtc_base/system:fallthrough",
>+      "../../rtc_base/system:unused",
>       "../../system_wrappers",
>       "../../system_wrappers:field_trial_api",
>+      "../../test:fileutils",
>       "../../test:perf_test",
>       "../../test:test_support",
>-      "../../voice_engine",
>       "../bitrate_controller",
>       "../congestion_controller",
>-      "../congestion_controller:delay_based_bwe",
>-      "../congestion_controller:estimators",
>+      "../congestion_controller:transport_feedback",
>+      "../congestion_controller/goog_cc:delay_based_bwe",
>+      "../congestion_controller/goog_cc:estimators",
>+      "../congestion_controller/rtp:transport_feedback",
>       "../pacing",
>       "../rtp_rtcp",
>       "../rtp_rtcp:rtp_rtcp_format",
>       "../video_coding:video_coding_utility",
>-      "//testing/gmock",
>       "//testing/gtest",
>+      "//third_party/abseil-cpp/absl/memory",
>+      "//third_party/abseil-cpp/absl/types:optional",
>     ]
>   }
> 
>@@ -180,6 +163,7 @@ if (rtc_include_tests) {
>       ":remote_bitrate_estimator",
>       "../../rtc_base:rtc_base_approved",
>       "../../test:field_trial",
>+      "../../test:fileutils",
>       "../../test:test_support",
>     ]
>     if (!build_with_chromium && is_clang) {
>@@ -200,7 +184,6 @@ if (rtc_include_tests) {
>       "remote_bitrate_estimator_unittest_helper.cc",
>       "remote_bitrate_estimator_unittest_helper.h",
>       "remote_estimator_proxy_unittest.cc",
>-      "send_time_history_unittest.cc",
>       "test/bwe_test_framework_unittest.cc",
>       "test/bwe_unittest.cc",
>       "test/estimators/congestion_window_unittest.cc",
>@@ -218,17 +201,11 @@ if (rtc_include_tests) {
>       "../../rtc_base:rtc_base_approved",
>       "../../system_wrappers",
>       "../../test:field_trial",
>+      "../../test:fileutils",
>       "../../test:test_support",
>       "../pacing:pacing",
>       "../rtp_rtcp:rtp_rtcp_format",
>-      "//testing/gmock",
>     ]
>-    if (is_win) {
>-      cflags = [
>-        # TODO(kjellander): bugs.webrtc.org/261: Fix this warning.
>-        "/wd4373",  # virtual function override.
>-      ]
>-    }
>     if (!build_with_chromium && is_clang) {
>       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>@@ -246,8 +223,8 @@ if (rtc_include_tests) {
>       ":remote_bitrate_estimator",
>       "../..:webrtc_common",
>       "../../rtc_base:rtc_base_approved",
>+      "../../test:fileutils",
>       "../../test:test_main",
>-      "//testing/gmock",
>       "//testing/gtest",
>     ]
>     data = [
>@@ -260,12 +237,5 @@ if (rtc_include_tests) {
>       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>     }
>-
>-    if (is_win) {
>-      cflags = [
>-        # TODO(kjellander): bugs.webrtc.org/261: Fix this warning.
>-        "/wd4373",  # virtual function override.
>-      ]
>-    }
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/DEPS b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/DEPS
>index d6a2fbafc9f..8499c256137 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/DEPS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/DEPS
>@@ -2,9 +2,3 @@ include_rules = [
>   "+logging/rtc_event_log",
>   "+system_wrappers",
> ]
>-
>-specific_include_rules = {
>-  "nada\.h": [
>-    "+voice_engine",
>-  ],
>-}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/aimd_rate_control.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/aimd_rate_control.cc
>index 22c2cc6ba15..8db2a6102a6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/aimd_rate_control.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/aimd_rate_control.cc
>@@ -10,19 +10,21 @@
> 
> #include "modules/remote_bitrate_estimator/aimd_rate_control.h"
> 
>+#include <inttypes.h>
>+
> #include <algorithm>
> #include <cassert>
> #include <cmath>
>+#include <cstdio>
> #include <string>
> 
>-#include "rtc_base/checks.h"
>-#include "rtc_base/numerics/safe_minmax.h"
>-
> #include "modules/remote_bitrate_estimator/include/bwe_defines.h"
> #include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
> #include "modules/remote_bitrate_estimator/overuse_detector.h"
> #include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
>+#include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>+#include "rtc_base/numerics/safe_minmax.h"
> #include "system_wrappers/include/field_trial.h"
> 
> namespace webrtc {
>@@ -30,10 +32,13 @@ namespace webrtc {
> static const int64_t kDefaultRttMs = 200;
> static const int64_t kMaxFeedbackIntervalMs = 1000;
> static const float kDefaultBackoffFactor = 0.85f;
>+static const int64_t kDefaultInitialBackOffIntervalMs = 200;
> 
> const char kBweBackOffFactorExperiment[] = "WebRTC-BweBackOffFactor";
>+const char kBweInitialBackOffIntervalExperiment[] =
>+    "WebRTC-BweInitialBackOffInterval";
> 
>-float ReadTrendlineFilterWindowSize() {
>+float ReadBackoffFactor() {
>   std::string experiment_string =
>       webrtc::field_trial::FindFullName(kBweBackOffFactorExperiment);
>   float backoff_factor;
>@@ -53,24 +58,53 @@ float ReadTrendlineFilterWindowSize() {
>   return kDefaultBackoffFactor;
> }
> 
>+int64_t ReadInitialBackoffIntervalMs() {
>+  std::string experiment_string =
>+      webrtc::field_trial::FindFullName(kBweInitialBackOffIntervalExperiment);
>+  int64_t backoff_interval;
>+  int parsed_values =
>+      sscanf(experiment_string.c_str(), "Enabled-%" SCNd64, &backoff_interval);
>+  if (parsed_values == 1) {
>+    if (10 <= backoff_interval && backoff_interval <= 200) {
>+      return backoff_interval;
>+    }
>+    RTC_LOG(WARNING)
>+        << "Initial back-off interval must be between 10 and 200 ms.";
>+  }
>+  RTC_LOG(LS_WARNING) << "Failed to parse parameters for "
>+                      << kBweInitialBackOffIntervalExperiment
>+                      << " experiment. Using default.";
>+  return kDefaultInitialBackOffIntervalMs;
>+}
>+
> AimdRateControl::AimdRateControl()
>     : min_configured_bitrate_bps_(congestion_controller::GetMinBitrateBps()),
>       max_configured_bitrate_bps_(30000000),
>       current_bitrate_bps_(max_configured_bitrate_bps_),
>+      latest_estimated_throughput_bps_(current_bitrate_bps_),
>       avg_max_bitrate_kbps_(-1.0f),
>       var_max_bitrate_kbps_(0.4f),
>       rate_control_state_(kRcHold),
>       rate_control_region_(kRcMaxUnknown),
>       time_last_bitrate_change_(-1),
>-      time_first_incoming_estimate_(-1),
>+      time_last_bitrate_decrease_(-1),
>+      time_first_throughput_estimate_(-1),
>       bitrate_is_initialized_(false),
>       beta_(webrtc::field_trial::IsEnabled(kBweBackOffFactorExperiment)
>-                ? ReadTrendlineFilterWindowSize()
>+                ? ReadBackoffFactor()
>                 : kDefaultBackoffFactor),
>       rtt_(kDefaultRttMs),
>       in_experiment_(!AdaptiveThresholdExperimentIsDisabled()),
>       smoothing_experiment_(
>-          webrtc::field_trial::IsEnabled("WebRTC-Audio-BandwidthSmoothing")) {
>+          webrtc::field_trial::IsEnabled("WebRTC-Audio-BandwidthSmoothing")),
>+      in_initial_backoff_interval_experiment_(
>+          webrtc::field_trial::IsEnabled(kBweInitialBackOffIntervalExperiment)),
>+      initial_backoff_interval_ms_(kDefaultInitialBackOffIntervalMs) {
>+  if (in_initial_backoff_interval_experiment_) {
>+    initial_backoff_interval_ms_ = ReadInitialBackoffIntervalMs();
>+    RTC_LOG(LS_INFO) << "Using aimd rate control with initial back-off interval"
>+                     << " " << initial_backoff_interval_ms_ << " ms.";
>+  }
>   RTC_LOG(LS_INFO) << "Using aimd rate control with back off factor " << beta_;
> }
> 
>@@ -78,6 +112,7 @@ AimdRateControl::~AimdRateControl() {}
> 
> void AimdRateControl::SetStartBitrate(int start_bitrate_bps) {
>   current_bitrate_bps_ = start_bitrate_bps;
>+  latest_estimated_throughput_bps_ = current_bitrate_bps_;
>   bitrate_is_initialized_ = true;
> }
> 
>@@ -101,18 +136,33 @@ int64_t AimdRateControl::GetFeedbackInterval() const {
>                         kMaxFeedbackIntervalMs);
> }
> 
>-bool AimdRateControl::TimeToReduceFurther(int64_t time_now,
>-                                          uint32_t incoming_bitrate_bps) const {
>+bool AimdRateControl::TimeToReduceFurther(
>+    int64_t now_ms,
>+    uint32_t estimated_throughput_bps) const {
>   const int64_t bitrate_reduction_interval =
>       std::max<int64_t>(std::min<int64_t>(rtt_, 200), 10);
>-  if (time_now - time_last_bitrate_change_ >= bitrate_reduction_interval) {
>+  if (now_ms - time_last_bitrate_change_ >= bitrate_reduction_interval) {
>     return true;
>   }
>   if (ValidEstimate()) {
>     // TODO(terelius/holmer): Investigate consequences of increasing
>     // the threshold to 0.95 * LatestEstimate().
>-    const uint32_t threshold = static_cast<uint32_t> (0.5 * LatestEstimate());
>-    return incoming_bitrate_bps < threshold;
>+    const uint32_t threshold = static_cast<uint32_t>(0.5 * LatestEstimate());
>+    return estimated_throughput_bps < threshold;
>+  }
>+  return false;
>+}
>+
>+bool AimdRateControl::InitialTimeToReduceFurther(int64_t now_ms) const {
>+  if (!in_initial_backoff_interval_experiment_) {
>+    return ValidEstimate() &&
>+           TimeToReduceFurther(now_ms, LatestEstimate() / 2 - 1);
>+  }
>+  // TODO(terelius): We could use the RTT (clamped to suitable limits) instead
>+  // of a fixed bitrate_reduction_interval.
>+  if (time_last_bitrate_decrease_ == -1 ||
>+      now_ms - time_last_bitrate_decrease_ >= initial_backoff_interval_ms_) {
>+    return true;
>   }
>   return false;
> }
>@@ -131,15 +181,17 @@ uint32_t AimdRateControl::Update(const RateControlInput* input,
> 
>   // Set the initial bit rate value to what we're receiving the first half
>   // second.
>+  // TODO(bugs.webrtc.org/9379): The comment above doesn't match to the code.
>   if (!bitrate_is_initialized_) {
>     const int64_t kInitializationTimeMs = 5000;
>     RTC_DCHECK_LE(kBitrateWindowMs, kInitializationTimeMs);
>-    if (time_first_incoming_estimate_ < 0) {
>-      if (input->incoming_bitrate)
>-        time_first_incoming_estimate_ = now_ms;
>-    } else if (now_ms - time_first_incoming_estimate_ > kInitializationTimeMs &&
>-               input->incoming_bitrate) {
>-      current_bitrate_bps_ = *input->incoming_bitrate;
>+    if (time_first_throughput_estimate_ < 0) {
>+      if (input->estimated_throughput_bps)
>+        time_first_throughput_estimate_ = now_ms;
>+    } else if (now_ms - time_first_throughput_estimate_ >
>+                   kInitializationTimeMs &&
>+               input->estimated_throughput_bps) {
>+      current_bitrate_bps_ = *input->estimated_throughput_bps;
>       bitrate_is_initialized_ = true;
>     }
>   }
>@@ -150,8 +202,12 @@ uint32_t AimdRateControl::Update(const RateControlInput* input,
> 
> void AimdRateControl::SetEstimate(int bitrate_bps, int64_t now_ms) {
>   bitrate_is_initialized_ = true;
>+  uint32_t prev_bitrate_bps = current_bitrate_bps_;
>   current_bitrate_bps_ = ClampBitrate(bitrate_bps, bitrate_bps);
>   time_last_bitrate_change_ = now_ms;
>+  if (current_bitrate_bps_ < prev_bitrate_bps) {
>+    time_last_bitrate_decrease_ = now_ms;
>+  }
> }
> 
> int AimdRateControl::GetNearMaxIncreaseRateBps() const {
>@@ -185,8 +241,10 @@ int AimdRateControl::GetExpectedBandwidthPeriodMs() const {
> uint32_t AimdRateControl::ChangeBitrate(uint32_t new_bitrate_bps,
>                                         const RateControlInput& input,
>                                         int64_t now_ms) {
>-  uint32_t incoming_bitrate_bps =
>-      input.incoming_bitrate.value_or(current_bitrate_bps_);
>+  uint32_t estimated_throughput_bps =
>+      input.estimated_throughput_bps.value_or(latest_estimated_throughput_bps_);
>+  if (input.estimated_throughput_bps)
>+    latest_estimated_throughput_bps_ = *input.estimated_throughput_bps;
> 
>   // An over-use should always trigger us to reduce the bitrate, even though
>   // we have not yet established our first estimate. By acting on the over-use,
>@@ -197,18 +255,18 @@ uint32_t AimdRateControl::ChangeBitrate(uint32_t new_bitrate_bps,
> 
>   ChangeState(input, now_ms);
>   // Calculated here because it's used in multiple places.
>-  const float incoming_bitrate_kbps = incoming_bitrate_bps / 1000.0f;
>+  const float estimated_throughput_kbps = estimated_throughput_bps / 1000.0f;
>   // Calculate the max bit rate std dev given the normalized
>-  // variance and the current incoming bit rate.
>-  const float std_max_bit_rate = sqrt(var_max_bitrate_kbps_ *
>-                                      avg_max_bitrate_kbps_);
>+  // variance and the current throughput bitrate.
>+  const float std_max_bit_rate =
>+      sqrt(var_max_bitrate_kbps_ * avg_max_bitrate_kbps_);
>   switch (rate_control_state_) {
>     case kRcHold:
>       break;
> 
>     case kRcIncrease:
>       if (avg_max_bitrate_kbps_ >= 0 &&
>-          incoming_bitrate_kbps >
>+          estimated_throughput_kbps >
>               avg_max_bitrate_kbps_ + 3 * std_max_bit_rate) {
>         ChangeRegion(kRcMaxUnknown);
>         avg_max_bitrate_kbps_ = -1.0;
>@@ -230,7 +288,7 @@ uint32_t AimdRateControl::ChangeBitrate(uint32_t new_bitrate_bps,
>       // Set bit rate to something slightly lower than max
>       // to get rid of any self-induced delay.
>       new_bitrate_bps =
>-          static_cast<uint32_t>(beta_ * incoming_bitrate_bps + 0.5);
>+          static_cast<uint32_t>(beta_ * estimated_throughput_bps + 0.5);
>       if (new_bitrate_bps > current_bitrate_bps_) {
>         // Avoid increasing the rate when over-using.
>         if (rate_control_region_ != kRcMaxUnknown) {
>@@ -242,7 +300,7 @@ uint32_t AimdRateControl::ChangeBitrate(uint32_t new_bitrate_bps,
>       ChangeRegion(kRcNearMax);
> 
>       if (bitrate_is_initialized_ &&
>-          incoming_bitrate_bps < current_bitrate_bps_) {
>+          estimated_throughput_bps < current_bitrate_bps_) {
>         constexpr float kDegradationFactor = 0.9f;
>         if (smoothing_experiment_ &&
>             new_bitrate_bps <
>@@ -250,36 +308,38 @@ uint32_t AimdRateControl::ChangeBitrate(uint32_t new_bitrate_bps,
>           // If bitrate decreases more than a normal back off after overuse, it
>           // indicates a real network degradation. We do not let such a decrease
>           // to determine the bandwidth estimation period.
>-          last_decrease_ = rtc::nullopt;
>+          last_decrease_ = absl::nullopt;
>         } else {
>           last_decrease_ = current_bitrate_bps_ - new_bitrate_bps;
>         }
>       }
>-      if (incoming_bitrate_kbps <
>+      if (estimated_throughput_kbps <
>           avg_max_bitrate_kbps_ - 3 * std_max_bit_rate) {
>         avg_max_bitrate_kbps_ = -1.0f;
>       }
> 
>       bitrate_is_initialized_ = true;
>-      UpdateMaxBitRateEstimate(incoming_bitrate_kbps);
>+      UpdateMaxThroughputEstimate(estimated_throughput_kbps);
>       // Stay on hold until the pipes are cleared.
>       rate_control_state_ = kRcHold;
>       time_last_bitrate_change_ = now_ms;
>+      time_last_bitrate_decrease_ = now_ms;
>       break;
> 
>     default:
>       assert(false);
>   }
>-  return ClampBitrate(new_bitrate_bps, incoming_bitrate_bps);
>+  return ClampBitrate(new_bitrate_bps, estimated_throughput_bps);
> }
> 
>-uint32_t AimdRateControl::ClampBitrate(uint32_t new_bitrate_bps,
>-                                       uint32_t incoming_bitrate_bps) const {
>+uint32_t AimdRateControl::ClampBitrate(
>+    uint32_t new_bitrate_bps,
>+    uint32_t estimated_throughput_bps) const {
>   // Don't change the bit rate if the send side is too far off.
>   // We allow a bit more lag at very low rates to not too easily get stuck if
>   // the encoder produces uneven outputs.
>   const uint32_t max_bitrate_bps =
>-      static_cast<uint32_t>(1.5f * incoming_bitrate_bps) + 10000;
>+      static_cast<uint32_t>(1.5f * estimated_throughput_bps) + 10000;
>   if (new_bitrate_bps > current_bitrate_bps_ &&
>       new_bitrate_bps > max_bitrate_bps) {
>     new_bitrate_bps = std::max(current_bitrate_bps_, max_bitrate_bps);
>@@ -289,15 +349,17 @@ uint32_t AimdRateControl::ClampBitrate(uint32_t new_bitrate_bps,
> }
> 
> uint32_t AimdRateControl::MultiplicativeRateIncrease(
>-    int64_t now_ms, int64_t last_ms, uint32_t current_bitrate_bps) const {
>+    int64_t now_ms,
>+    int64_t last_ms,
>+    uint32_t current_bitrate_bps) const {
>   double alpha = 1.08;
>   if (last_ms > -1) {
>     auto time_since_last_update_ms =
>         rtc::SafeMin<int64_t>(now_ms - last_ms, 1000);
>-    alpha = pow(alpha,  time_since_last_update_ms / 1000.0);
>+    alpha = pow(alpha, time_since_last_update_ms / 1000.0);
>   }
>-  uint32_t multiplicative_increase_bps = std::max(
>-      current_bitrate_bps * (alpha - 1.0), 1000.0);
>+  uint32_t multiplicative_increase_bps =
>+      std::max(current_bitrate_bps * (alpha - 1.0), 1000.0);
>   return multiplicative_increase_bps;
> }
> 
>@@ -307,20 +369,22 @@ uint32_t AimdRateControl::AdditiveRateIncrease(int64_t now_ms,
>                                GetNearMaxIncreaseRateBps() / 1000);
> }
> 
>-void AimdRateControl::UpdateMaxBitRateEstimate(float incoming_bitrate_kbps) {
>+void AimdRateControl::UpdateMaxThroughputEstimate(
>+    float estimated_throughput_kbps) {
>   const float alpha = 0.05f;
>   if (avg_max_bitrate_kbps_ == -1.0f) {
>-    avg_max_bitrate_kbps_ = incoming_bitrate_kbps;
>+    avg_max_bitrate_kbps_ = estimated_throughput_kbps;
>   } else {
>-    avg_max_bitrate_kbps_ = (1 - alpha) * avg_max_bitrate_kbps_ +
>-        alpha * incoming_bitrate_kbps;
>+    avg_max_bitrate_kbps_ =
>+        (1 - alpha) * avg_max_bitrate_kbps_ + alpha * estimated_throughput_kbps;
>   }
>   // Estimate the max bit rate variance and normalize the variance
>   // with the average max bit rate.
>   const float norm = std::max(avg_max_bitrate_kbps_, 1.0f);
>-  var_max_bitrate_kbps_ = (1 - alpha) * var_max_bitrate_kbps_ +
>-      alpha * (avg_max_bitrate_kbps_ - incoming_bitrate_kbps) *
>-          (avg_max_bitrate_kbps_ - incoming_bitrate_kbps) / norm;
>+  var_max_bitrate_kbps_ =
>+      (1 - alpha) * var_max_bitrate_kbps_ +
>+      alpha * (avg_max_bitrate_kbps_ - estimated_throughput_kbps) *
>+          (avg_max_bitrate_kbps_ - estimated_throughput_kbps) / norm;
>   // 0.4 ~= 14 kbit/s at 500 kbit/s
>   if (var_max_bitrate_kbps_ < 0.4f) {
>     var_max_bitrate_kbps_ = 0.4f;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/aimd_rate_control.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/aimd_rate_control.h
>index a9ecb67c34d..8ec43576946 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/aimd_rate_control.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/aimd_rate_control.h
>@@ -26,18 +26,23 @@ class AimdRateControl {
>   AimdRateControl();
>   ~AimdRateControl();
> 
>-  // Returns true if there is a valid estimate of the incoming bitrate, false
>-  // otherwise.
>+  // Returns true if the target bitrate has been initialized. This happens
>+  // either if it has been explicitly set via SetStartBitrate/SetEstimate, or if
>+  // we have measured a throughput.
>   bool ValidEstimate() const;
>   void SetStartBitrate(int start_bitrate_bps);
>   void SetMinBitrate(int min_bitrate_bps);
>   int64_t GetFeedbackInterval() const;
>+
>   // Returns true if the bitrate estimate hasn't been changed for more than
>-  // an RTT, or if the incoming_bitrate is less than half of the current
>+  // an RTT, or if the estimated_throughput is less than half of the current
>   // estimate. Should be used to decide if we should reduce the rate further
>   // when over-using.
>-  bool TimeToReduceFurther(int64_t time_now,
>-                           uint32_t incoming_bitrate_bps) const;
>+  bool TimeToReduceFurther(int64_t now_ms,
>+                           uint32_t estimated_throughput_bps) const;
>+  // As above. To be used if overusing before we have measured a throughput.
>+  bool InitialTimeToReduceFurther(int64_t now_ms) const;
>+
>   uint32_t LatestEstimate() const;
>   void SetRtt(int64_t rtt);
>   uint32_t Update(const RateControlInput* input, int64_t now_ms);
>@@ -49,44 +54,49 @@ class AimdRateControl {
>   int GetExpectedBandwidthPeriodMs() const;
> 
>  private:
>-  // Update the target bitrate according based on, among other things,
>-  // the current rate control state, the current target bitrate and the incoming
>-  // bitrate. When in the "increase" state the bitrate will be increased either
>+  // Update the target bitrate based on, among other things, the current rate
>+  // control state, the current target bitrate and the estimated throughput.
>+  // When in the "increase" state the bitrate will be increased either
>   // additively or multiplicatively depending on the rate control region. When
>   // in the "decrease" state the bitrate will be decreased to slightly below the
>-  // incoming bitrate. When in the "hold" state the bitrate will be kept
>+  // current throughput. When in the "hold" state the bitrate will be kept
>   // constant to allow built up queues to drain.
>   uint32_t ChangeBitrate(uint32_t current_bitrate,
>                          const RateControlInput& input,
>                          int64_t now_ms);
>   // Clamps new_bitrate_bps to within the configured min bitrate and a linear
>-  // function of the incoming bitrate, so that the new bitrate can't grow too
>+  // function of the throughput, so that the new bitrate can't grow too
>   // large compared to the bitrate actually being received by the other end.
>   uint32_t ClampBitrate(uint32_t new_bitrate_bps,
>-                        uint32_t incoming_bitrate_bps) const;
>-  uint32_t MultiplicativeRateIncrease(int64_t now_ms, int64_t last_ms,
>+                        uint32_t estimated_throughput_bps) const;
>+  uint32_t MultiplicativeRateIncrease(int64_t now_ms,
>+                                      int64_t last_ms,
>                                       uint32_t current_bitrate_bps) const;
>   uint32_t AdditiveRateIncrease(int64_t now_ms, int64_t last_ms) const;
>   void UpdateChangePeriod(int64_t now_ms);
>-  void UpdateMaxBitRateEstimate(float incoming_bit_rate_kbps);
>+  void UpdateMaxThroughputEstimate(float estimated_throughput_kbps);
>   void ChangeState(const RateControlInput& input, int64_t now_ms);
>   void ChangeRegion(RateControlRegion region);
> 
>   uint32_t min_configured_bitrate_bps_;
>   uint32_t max_configured_bitrate_bps_;
>   uint32_t current_bitrate_bps_;
>+  uint32_t latest_estimated_throughput_bps_;
>   float avg_max_bitrate_kbps_;
>   float var_max_bitrate_kbps_;
>   RateControlState rate_control_state_;
>   RateControlRegion rate_control_region_;
>   int64_t time_last_bitrate_change_;
>-  int64_t time_first_incoming_estimate_;
>+  int64_t time_last_bitrate_decrease_;
>+  int64_t time_first_throughput_estimate_;
>   bool bitrate_is_initialized_;
>   float beta_;
>   int64_t rtt_;
>-  bool in_experiment_;
>-  bool smoothing_experiment_;
>-  rtc::Optional<int> last_decrease_;
>+  const bool in_experiment_;
>+  const bool smoothing_experiment_;
>+  const bool in_initial_backoff_interval_experiment_;
>+  int64_t initial_backoff_interval_ms_;
>+  absl::optional<int> last_decrease_;
> };
> }  // namespace webrtc
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/aimd_rate_control_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/aimd_rate_control_unittest.cc
>index f6844b0e946..5cde7cedaf4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/aimd_rate_control_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/aimd_rate_control_unittest.cc
>@@ -43,9 +43,9 @@ AimdRateControlStates CreateAimdRateControlStates() {
> 
> void UpdateRateControl(const AimdRateControlStates& states,
>                        const BandwidthUsage& bandwidth_usage,
>-                       int bitrate,
>+                       absl::optional<uint32_t> throughput_estimate,
>                        int64_t now_ms) {
>-  RateControlInput input(bandwidth_usage, bitrate, now_ms);
>+  RateControlInput input(bandwidth_usage, throughput_estimate);
>   states.aimd_rate_control->Update(&input, now_ms);
> }
> 
>@@ -247,4 +247,26 @@ TEST(AimdRateControlTest, BandwidthPeriodIsNotAboveMaxNoSmoothingExp) {
>             states.aimd_rate_control->GetExpectedBandwidthPeriodMs());
> }
> 
>+TEST(AimdRateControlTest, SendingRateBoundedWhenThroughputNotEstimated) {
>+  auto states = CreateAimdRateControlStates();
>+  constexpr int kInitialBitrateBps = 123000;
>+  UpdateRateControl(states, BandwidthUsage::kBwNormal, kInitialBitrateBps,
>+                    states.simulated_clock->TimeInMilliseconds());
>+  // AimdRateControl sets the initial bit rate to what it receives after
>+  // five seconds has passed.
>+  // TODO(bugs.webrtc.org/9379): The comment in the AimdRateControl does not
>+  // match the constant.
>+  constexpr int kInitializationTimeMs = 5000;
>+  states.simulated_clock->AdvanceTimeMilliseconds(kInitializationTimeMs + 1);
>+  UpdateRateControl(states, BandwidthUsage::kBwNormal, kInitialBitrateBps,
>+                    states.simulated_clock->TimeInMilliseconds());
>+  for (int i = 0; i < 100; ++i) {
>+    UpdateRateControl(states, BandwidthUsage::kBwNormal, absl::nullopt,
>+                      states.simulated_clock->TimeInMilliseconds());
>+    states.simulated_clock->AdvanceTimeMilliseconds(100);
>+  }
>+  EXPECT_LE(states.aimd_rate_control->LatestEstimate(),
>+            kInitialBitrateBps * 1.5 + 10000);
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/bwe_defines.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/bwe_defines.cc
>index 2c55e324dbd..35f1d19cc3d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/bwe_defines.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/bwe_defines.cc
>@@ -19,7 +19,8 @@ namespace congestion_controller {
> int GetMinBitrateBps() {
>   constexpr int kAudioMinBitrateBps = 5000;
>   constexpr int kMinBitrateBps = 10000;
>-  if (webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe")) {
>+  if (webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe") &&
>+      !webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe-For-Video")) {
>     return kAudioMinBitrateBps;
>   }
>   return kMinBitrateBps;
>@@ -29,11 +30,8 @@ int GetMinBitrateBps() {
> 
> RateControlInput::RateControlInput(
>     BandwidthUsage bw_state,
>-    const rtc::Optional<uint32_t>& incoming_bitrate,
>-    double noise_var)
>-    : bw_state(bw_state),
>-      incoming_bitrate(incoming_bitrate),
>-      noise_var(noise_var) {}
>+    const absl::optional<uint32_t>& estimated_throughput_bps)
>+    : bw_state(bw_state), estimated_throughput_bps(estimated_throughput_bps) {}
> 
> RateControlInput::~RateControlInput() = default;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/bwe_simulations.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/bwe_simulations.cc
>index f8866d81cb1..de6d2d99274 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/bwe_simulations.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/bwe_simulations.cc
>@@ -477,47 +477,47 @@ TEST_P(BweSimulation, Evaluation8) {
>   RunPauseResumeFlows(GetParam());
> }
> 
>-// Following test cases begin with "GccComparison" run the
>-// evaluation test cases for both GCC and other calling RMCAT.
>+// Following test cases begin with "GoogCcComparison" run the
>+// evaluation test cases for both GoogCc and other calling RMCAT.
> 
>-TEST_P(BweSimulation, GccComparison1) {
>+TEST_P(BweSimulation, GoogCcComparison1) {
>   RunVariableCapacity1SingleFlow(GetParam());
>-  BweTest gcc_test(false);
>-  gcc_test.RunVariableCapacity1SingleFlow(kSendSideEstimator);
>+  BweTest goog_cc_test(false);
>+  goog_cc_test.RunVariableCapacity1SingleFlow(kSendSideEstimator);
> }
> 
>-TEST_P(BweSimulation, GccComparison2) {
>+TEST_P(BweSimulation, GoogCcComparison2) {
>   const size_t kNumFlows = 2;
>   RunVariableCapacity2MultipleFlows(GetParam(), kNumFlows);
>-  BweTest gcc_test(false);
>-  gcc_test.RunVariableCapacity2MultipleFlows(kSendSideEstimator, kNumFlows);
>+  BweTest goog_cc_test(false);
>+  goog_cc_test.RunVariableCapacity2MultipleFlows(kSendSideEstimator, kNumFlows);
> }
> 
>-TEST_P(BweSimulation, GccComparison3) {
>+TEST_P(BweSimulation, GoogCcComparison3) {
>   RunBidirectionalFlow(GetParam());
>-  BweTest gcc_test(false);
>-  gcc_test.RunBidirectionalFlow(kSendSideEstimator);
>+  BweTest goog_cc_test(false);
>+  goog_cc_test.RunBidirectionalFlow(kSendSideEstimator);
> }
> 
>-TEST_P(BweSimulation, GccComparison4) {
>+TEST_P(BweSimulation, GoogCcComparison4) {
>   RunSelfFairness(GetParam());
>-  BweTest gcc_test(false);
>-  gcc_test.RunSelfFairness(GetParam());
>+  BweTest goog_cc_test(false);
>+  goog_cc_test.RunSelfFairness(GetParam());
> }
> 
>-TEST_P(BweSimulation, GccComparison5) {
>+TEST_P(BweSimulation, GoogCcComparison5) {
>   RunRoundTripTimeFairness(GetParam());
>-  BweTest gcc_test(false);
>-  gcc_test.RunRoundTripTimeFairness(kSendSideEstimator);
>+  BweTest goog_cc_test(false);
>+  goog_cc_test.RunRoundTripTimeFairness(kSendSideEstimator);
> }
> 
>-TEST_P(BweSimulation, GccComparison6) {
>+TEST_P(BweSimulation, GoogCcComparison6) {
>   RunLongTcpFairness(GetParam());
>-  BweTest gcc_test(false);
>-  gcc_test.RunLongTcpFairness(kSendSideEstimator);
>+  BweTest goog_cc_test(false);
>+  goog_cc_test.RunLongTcpFairness(kSendSideEstimator);
> }
> 
>-TEST_P(BweSimulation, GccComparison7) {
>+TEST_P(BweSimulation, GoogCcComparison7) {
>   const int kNumTcpFiles = 10;
> 
>   std::vector<int> tcp_file_sizes_bytes =
>@@ -528,27 +528,26 @@ TEST_P(BweSimulation, GccComparison7) {
>   RunMultipleShortTcpFairness(GetParam(), tcp_file_sizes_bytes,
>                               tcp_starting_times_ms);
> 
>-  BweTest gcc_test(false);
>-  gcc_test.RunMultipleShortTcpFairness(kSendSideEstimator, tcp_file_sizes_bytes,
>-                                       tcp_starting_times_ms);
>+  BweTest goog_cc_test(false);
>+  goog_cc_test.RunMultipleShortTcpFairness(
>+      kSendSideEstimator, tcp_file_sizes_bytes, tcp_starting_times_ms);
> }
> 
>-TEST_P(BweSimulation, GccComparison8) {
>+TEST_P(BweSimulation, GoogCcComparison8) {
>   RunPauseResumeFlows(GetParam());
>-  BweTest gcc_test(false);
>-  gcc_test.RunPauseResumeFlows(kSendSideEstimator);
>+  BweTest goog_cc_test(false);
>+  goog_cc_test.RunPauseResumeFlows(kSendSideEstimator);
> }
> 
>-TEST_P(BweSimulation, GccComparisonChoke) {
>+TEST_P(BweSimulation, GoogCcComparisonChoke) {
>   int array[] = {1000, 500, 1000};
>   std::vector<int> capacities_kbps(array, array + 3);
>   RunChoke(GetParam(), capacities_kbps);
> 
>-  BweTest gcc_test(false);
>-  gcc_test.RunChoke(kSendSideEstimator, capacities_kbps);
>+  BweTest goog_cc_test(false);
>+  goog_cc_test.RunChoke(kSendSideEstimator, capacities_kbps);
> }
> 
> }  // namespace bwe
> }  // namespace testing
> }  // namespace webrtc
>-
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h
>index 9b5b48b4fa4..0d8c49bda68 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h
>@@ -11,8 +11,7 @@
> #ifndef MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_BWE_DEFINES_H_
> #define MODULES_REMOTE_BITRATE_ESTIMATOR_INCLUDE_BWE_DEFINES_H_
> 
>-#include "api/optional.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include "absl/types/optional.h"
> 
> #define BWE_MAX(a, b) ((a) > (b) ? (a) : (b))
> #define BWE_MIN(a, b) ((a) < (b) ? (a) : (b))
>@@ -48,13 +47,11 @@ enum RateControlRegion { kRcNearMax, kRcAboveMax, kRcMaxUnknown };
> 
> struct RateControlInput {
>   RateControlInput(BandwidthUsage bw_state,
>-                   const rtc::Optional<uint32_t>& incoming_bitrate,
>-                   double noise_var);
>+                   const absl::optional<uint32_t>& estimated_throughput_bps);
>   ~RateControlInput();
> 
>   BandwidthUsage bw_state;
>-  rtc::Optional<uint32_t> incoming_bitrate;
>-  double noise_var;
>+  absl::optional<uint32_t> estimated_throughput_bps;
> };
> }  // namespace webrtc
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h
>index 99f4ab82505..6bf0ff3906b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h
>@@ -20,7 +20,6 @@
> #include "modules/include/module.h"
> #include "modules/include/module_common_types.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> namespace rtcp {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/inter_arrival.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/inter_arrival.cc
>index ef3ca8b624c..b779e417971 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/inter_arrival.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/inter_arrival.cc
>@@ -51,8 +51,8 @@ bool InterArrival::ComputeDeltas(uint32_t timestamp,
>   } else if (NewTimestampGroup(arrival_time_ms, timestamp)) {
>     // First packet of a later frame, the previous frame sample is ready.
>     if (prev_timestamp_group_.complete_time_ms >= 0) {
>-      *timestamp_delta = current_timestamp_group_.timestamp -
>-                         prev_timestamp_group_.timestamp;
>+      *timestamp_delta =
>+          current_timestamp_group_.timestamp - prev_timestamp_group_.timestamp;
>       *arrival_time_delta_ms = current_timestamp_group_.complete_time_ms -
>                                prev_timestamp_group_.complete_time_ms;
>       // Check system time differences to see if we have an unproportional jump
>@@ -86,7 +86,7 @@ bool InterArrival::ComputeDeltas(uint32_t timestamp,
>       }
>       assert(*arrival_time_delta_ms >= 0);
>       *packet_size_delta = static_cast<int>(current_timestamp_group_.size) -
>-          static_cast<int>(prev_timestamp_group_.size);
>+                           static_cast<int>(prev_timestamp_group_.size);
>       calculated_deltas = true;
>     }
>     prev_timestamp_group_ = current_timestamp_group_;
>@@ -95,8 +95,8 @@ bool InterArrival::ComputeDeltas(uint32_t timestamp,
>     current_timestamp_group_.timestamp = timestamp;
>     current_timestamp_group_.size = 0;
>   } else {
>-    current_timestamp_group_.timestamp = LatestTimestamp(
>-        current_timestamp_group_.timestamp, timestamp);
>+    current_timestamp_group_.timestamp =
>+        LatestTimestamp(current_timestamp_group_.timestamp, timestamp);
>   }
>   // Accumulate the frame size.
>   current_timestamp_group_.size += packet_size;
>@@ -113,8 +113,8 @@ bool InterArrival::PacketInOrder(uint32_t timestamp) {
>     // Assume that a diff which is bigger than half the timestamp interval
>     // (32 bits) must be due to reordering. This code is almost identical to
>     // that in IsNewerTimestamp() in module_common_types.h.
>-    uint32_t timestamp_diff = timestamp -
>-        current_timestamp_group_.first_timestamp;
>+    uint32_t timestamp_diff =
>+        timestamp - current_timestamp_group_.first_timestamp;
>     return timestamp_diff < 0x80000000;
>   }
> }
>@@ -128,8 +128,8 @@ bool InterArrival::NewTimestampGroup(int64_t arrival_time_ms,
>   } else if (BelongsToBurst(arrival_time_ms, timestamp)) {
>     return false;
>   } else {
>-    uint32_t timestamp_diff = timestamp -
>-        current_timestamp_group_.first_timestamp;
>+    uint32_t timestamp_diff =
>+        timestamp - current_timestamp_group_.first_timestamp;
>     return timestamp_diff > kTimestampGroupLengthTicks;
>   }
> }
>@@ -140,15 +140,15 @@ bool InterArrival::BelongsToBurst(int64_t arrival_time_ms,
>     return false;
>   }
>   assert(current_timestamp_group_.complete_time_ms >= 0);
>-  int64_t arrival_time_delta_ms = arrival_time_ms -
>-      current_timestamp_group_.complete_time_ms;
>+  int64_t arrival_time_delta_ms =
>+      arrival_time_ms - current_timestamp_group_.complete_time_ms;
>   uint32_t timestamp_diff = timestamp - current_timestamp_group_.timestamp;
>   int64_t ts_delta_ms = timestamp_to_ms_coeff_ * timestamp_diff + 0.5;
>   if (ts_delta_ms == 0)
>     return true;
>   int propagation_delta_ms = arrival_time_delta_ms - ts_delta_ms;
>   return propagation_delta_ms < 0 &&
>-      arrival_time_delta_ms <= kBurstDeltaThresholdMs;
>+         arrival_time_delta_ms <= kBurstDeltaThresholdMs;
> }
> 
> void InterArrival::Reset() {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/inter_arrival.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/inter_arrival.h
>index f6ab280d344..f46bf4425d7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/inter_arrival.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/inter_arrival.h
>@@ -11,10 +11,10 @@
> #ifndef MODULES_REMOTE_BITRATE_ESTIMATOR_INTER_ARRIVAL_H_
> #define MODULES_REMOTE_BITRATE_ESTIMATOR_INTER_ARRIVAL_H_
> 
>-#include <cstddef>
>+#include <stddef.h>
>+#include <stdint.h>
> 
> #include "rtc_base/constructormagic.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -54,14 +54,9 @@ class InterArrival {
>  private:
>   struct TimestampGroup {
>     TimestampGroup()
>-        : size(0),
>-          first_timestamp(0),
>-          timestamp(0),
>-          complete_time_ms(-1) {}
>-
>-    bool IsFirstPacket() const {
>-      return complete_time_ms == -1;
>-    }
>+        : size(0), first_timestamp(0), timestamp(0), complete_time_ms(-1) {}
>+
>+    bool IsFirstPacket() const { return complete_time_ms == -1; }
> 
>     size_t size;
>     uint32_t first_timestamp;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/inter_arrival_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/inter_arrival_unittest.cc
>index cd038c42ace..545b19600d7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/inter_arrival_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/inter_arrival_unittest.cc
>@@ -36,18 +36,15 @@ class InterArrivalTest : public ::testing::Test {
>     inter_arrival_.reset(
>         new InterArrival(kTimestampGroupLengthUs / 1000, 1.0, true));
>     inter_arrival_rtp_.reset(new InterArrival(
>-        MakeRtpTimestamp(kTimestampGroupLengthUs),
>-        kRtpTimestampToMs,
>-        true));
>+        MakeRtpTimestamp(kTimestampGroupLengthUs), kRtpTimestampToMs, true));
>     inter_arrival_ast_.reset(new InterArrival(
>-        MakeAbsSendTime(kTimestampGroupLengthUs),
>-        kAstToMs,
>-        true));
>+        MakeAbsSendTime(kTimestampGroupLengthUs), kAstToMs, true));
>   }
> 
>   // Test that neither inter_arrival instance complete the timestamp group from
>   // the given data.
>-  void ExpectFalse(int64_t timestamp_us, int64_t arrival_time_ms,
>+  void ExpectFalse(int64_t timestamp_us,
>+                   int64_t arrival_time_ms,
>                    size_t packet_size) {
>     InternalExpectFalse(inter_arrival_rtp_.get(),
>                         MakeRtpTimestamp(timestamp_us), arrival_time_ms,
>@@ -60,8 +57,10 @@ class InterArrivalTest : public ::testing::Test {
>   // the given data and that all returned deltas are as expected (except
>   // timestamp delta, which is rounded from us to different ranges and must
>   // match within an interval, given in |timestamp_near].
>-  void ExpectTrue(int64_t timestamp_us, int64_t arrival_time_ms,
>-                  size_t packet_size, int64_t expected_timestamp_delta_us,
>+  void ExpectTrue(int64_t timestamp_us,
>+                  int64_t arrival_time_ms,
>+                  size_t packet_size,
>+                  int64_t expected_timestamp_delta_us,
>                   int64_t expected_arrival_time_delta_ms,
>                   int expected_packet_size_delta,
>                   uint32_t timestamp_near) {
>@@ -77,7 +76,8 @@ class InterArrivalTest : public ::testing::Test {
>                        expected_packet_size_delta, timestamp_near << 8);
>   }
> 
>-  void WrapTestHelper(int64_t wrap_start_us, uint32_t timestamp_near,
>+  void WrapTestHelper(int64_t wrap_start_us,
>+                      uint32_t timestamp_near,
>                       bool unorderly_within_group) {
>     // Step through the range of a 32 bit int, 1/4 at a time to not cause
>     // packets close to wraparound to be judged as out of order.
>@@ -92,21 +92,21 @@ class InterArrivalTest : public ::testing::Test {
> 
>     // G3
>     arrival_time += kBurstThresholdMs + 1;
>-    ExpectTrue(wrap_start_us / 2, arrival_time, 1,
>-               wrap_start_us / 4, 6, 0,   // Delta G2-G1
>+    ExpectTrue(wrap_start_us / 2, arrival_time, 1, wrap_start_us / 4, 6,
>+               0,  // Delta G2-G1
>                0);
> 
>     // G4
>     arrival_time += kBurstThresholdMs + 1;
>     int64_t g4_arrival_time = arrival_time;
>     ExpectTrue(wrap_start_us / 2 + wrap_start_us / 4, arrival_time, 1,
>-               wrap_start_us / 4, 6, 0,   // Delta G3-G2
>+               wrap_start_us / 4, 6, 0,  // Delta G3-G2
>                timestamp_near);
> 
>     // G5
>     arrival_time += kBurstThresholdMs + 1;
>-    ExpectTrue(wrap_start_us, arrival_time, 2,
>-               wrap_start_us / 4, 6, 0,   // Delta G4-G3
>+    ExpectTrue(wrap_start_us, arrival_time, 2, wrap_start_us / 4, 6,
>+               0,  // Delta G4-G3
>                timestamp_near);
>     for (int i = 0; i < 10; ++i) {
>       // Slowly step across the wrap point.
>@@ -141,12 +141,10 @@ class InterArrivalTest : public ::testing::Test {
> 
>     // G7
>     arrival_time += kBurstThresholdMs + 1;
>-    ExpectTrue(wrap_start_us + 2 * kTriggerNewGroupUs,
>-               arrival_time, 100,
>+    ExpectTrue(wrap_start_us + 2 * kTriggerNewGroupUs, arrival_time, 100,
>                // Delta G6-G5
>                kTriggerNewGroupUs - 9 * kMinStep,
>-               g6_arrival_time - g5_arrival_time,
>-               10 - (2 + 10),
>+               g6_arrival_time - g5_arrival_time, 10 - (2 + 10),
>                timestamp_near);
>   }
> 
>@@ -158,13 +156,16 @@ class InterArrivalTest : public ::testing::Test {
>   }
> 
>   static uint32_t MakeAbsSendTime(int64_t us) {
>-    uint32_t absolute_send_time = static_cast<uint32_t>(
>-        ((static_cast<uint64_t>(us) << 18) + 500000) / 1000000) & 0x00FFFFFFul;
>+    uint32_t absolute_send_time =
>+        static_cast<uint32_t>(((static_cast<uint64_t>(us) << 18) + 500000) /
>+                              1000000) &
>+        0x00FFFFFFul;
>     return absolute_send_time << 8;
>   }
> 
>   static void InternalExpectFalse(InterArrival* inter_arrival,
>-                                  uint32_t timestamp, int64_t arrival_time_ms,
>+                                  uint32_t timestamp,
>+                                  int64_t arrival_time_ms,
>                                   size_t packet_size) {
>     uint32_t dummy_timestamp = 101;
>     int64_t dummy_arrival_time_ms = 303;
>@@ -179,7 +180,8 @@ class InterArrivalTest : public ::testing::Test {
>   }
> 
>   static void InternalExpectTrue(InterArrival* inter_arrival,
>-                                 uint32_t timestamp, int64_t arrival_time_ms,
>+                                 uint32_t timestamp,
>+                                 int64_t arrival_time_ms,
>                                  size_t packet_size,
>                                  uint32_t expected_timestamp_delta,
>                                  int64_t expected_arrival_time_delta_ms,
>@@ -222,8 +224,7 @@ TEST_F(InterArrivalTest, FirstGroup) {
>   arrival_time += kBurstThresholdMs + 1;
>   ExpectTrue(2 * kTriggerNewGroupUs, arrival_time, 1,
>              // Delta G2-G1
>-             kTriggerNewGroupUs, g2_arrival_time - g1_arrival_time, 1,
>-             0);
>+             kTriggerNewGroupUs, g2_arrival_time - g1_arrival_time, 1, 0);
> }
> 
> TEST_F(InterArrivalTest, SecondGroup) {
>@@ -242,16 +243,14 @@ TEST_F(InterArrivalTest, SecondGroup) {
>   int64_t g3_arrival_time = arrival_time;
>   ExpectTrue(2 * kTriggerNewGroupUs, arrival_time, 1,
>              // Delta G2-G1
>-             kTriggerNewGroupUs, g2_arrival_time - g1_arrival_time, 1,
>-             0);
>+             kTriggerNewGroupUs, g2_arrival_time - g1_arrival_time, 1, 0);
> 
>   // G4
>   // First packet of 4th group yields deltas between group 2 and 3.
>   arrival_time += kBurstThresholdMs + 1;
>   ExpectTrue(3 * kTriggerNewGroupUs, arrival_time, 2,
>              // Delta G3-G2
>-             kTriggerNewGroupUs, g3_arrival_time - g2_arrival_time, -1,
>-             0);
>+             kTriggerNewGroupUs, g3_arrival_time - g2_arrival_time, -1, 0);
> }
> 
> TEST_F(InterArrivalTest, AccumulatedGroup) {
>@@ -275,9 +274,9 @@ TEST_F(InterArrivalTest, AccumulatedGroup) {
> 
>   // G3
>   arrival_time = 500;
>-  ExpectTrue(2 * kTriggerNewGroupUs, arrival_time, 100,
>-             g2_timestamp, g2_arrival_time - g1_arrival_time,
>-             (2 + 10) - 1,   // Delta G2-G1
>+  ExpectTrue(2 * kTriggerNewGroupUs, arrival_time, 100, g2_timestamp,
>+             g2_arrival_time - g1_arrival_time,
>+             (2 + 10) - 1,  // Delta G2-G1
>              0);
> }
> 
>@@ -311,8 +310,7 @@ TEST_F(InterArrivalTest, OutOfOrderPacket) {
>   ExpectTrue(timestamp, arrival_time, 100,
>              // Delta G2-G1
>              g2_timestamp - g1_timestamp, g2_arrival_time - g1_arrival_time,
>-             (2 + 10) - 1,
>-             0);
>+             (2 + 10) - 1, 0);
> }
> 
> TEST_F(InterArrivalTest, OutOfOrderWithinGroup) {
>@@ -347,10 +345,8 @@ TEST_F(InterArrivalTest, OutOfOrderWithinGroup) {
>   // G3
>   timestamp = 2 * kTriggerNewGroupUs;
>   arrival_time = 500;
>-  ExpectTrue(timestamp, arrival_time, 100,
>-             g2_timestamp - g1_timestamp, g2_arrival_time - g1_arrival_time,
>-             (2 + 10) - 1,
>-             0);
>+  ExpectTrue(timestamp, arrival_time, 100, g2_timestamp - g1_timestamp,
>+             g2_arrival_time - g1_arrival_time, (2 + 10) - 1, 0);
> }
> 
> TEST_F(InterArrivalTest, TwoBursts) {
>@@ -373,13 +369,12 @@ TEST_F(InterArrivalTest, TwoBursts) {
>   // G3
>   timestamp += 30000;
>   arrival_time += kBurstThresholdMs + 1;
>-  ExpectTrue(timestamp, arrival_time, 100,
>-             g2_timestamp, g2_arrival_time - g1_arrival_time,
>+  ExpectTrue(timestamp, arrival_time, 100, g2_timestamp,
>+             g2_arrival_time - g1_arrival_time,
>              10 - 1,  // Delta G2-G1
>              0);
> }
> 
>-
> TEST_F(InterArrivalTest, NoBursts) {
>   // G1
>   ExpectFalse(0, 17, 1);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/overuse_detector.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/overuse_detector.cc
>index 7110d356424..776ca42a56f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/overuse_detector.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/overuse_detector.cc
>@@ -11,10 +11,10 @@
> #include "modules/remote_bitrate_estimator/overuse_detector.h"
> 
> #include <math.h>
>+#include <stdio.h>
> #include <stdlib.h>
> 
> #include <algorithm>
>-#include <sstream>
> #include <string>
> 
> #include "modules/remote_bitrate_estimator/include/bwe_defines.h"
>@@ -142,8 +142,7 @@ void OveruseDetector::UpdateThreshold(double modified_offset, int64_t now_ms) {
>   const double k = fabs(modified_offset) < threshold_ ? k_down_ : k_up_;
>   const int64_t kMaxTimeDeltaMs = 100;
>   int64_t time_delta_ms = std::min(now_ms - last_update_ms_, kMaxTimeDeltaMs);
>-  threshold_ +=
>-      k * (fabs(modified_offset) - threshold_) * time_delta_ms;
>+  threshold_ += k * (fabs(modified_offset) - threshold_) * time_delta_ms;
>   threshold_ = rtc::SafeClamp(threshold_, 6.f, 600.f);
>   last_update_ms_ = now_ms;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/overuse_detector.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/overuse_detector.h
>index 69e5c13d732..f322c304960 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/overuse_detector.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/overuse_detector.h
>@@ -15,7 +15,6 @@
> #include "modules/include/module_common_types.h"
> #include "modules/remote_bitrate_estimator/include/bwe_defines.h"
> #include "rtc_base/constructormagic.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/overuse_detector_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/overuse_detector_unittest.cc
>index 2b50d70b65f..366902c268f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/overuse_detector_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/overuse_detector_unittest.cc
>@@ -41,11 +41,11 @@ class OveruseDetectorTest : public ::testing::Test {
>         random_(123456789) {}
> 
>  protected:
>-  void SetUp() override {
>-    overuse_detector_.reset(new OveruseDetector());
>-  }
>+  void SetUp() override { overuse_detector_.reset(new OveruseDetector()); }
> 
>-  int Run100000Samples(int packets_per_frame, size_t packet_size, int mean_ms,
>+  int Run100000Samples(int packets_per_frame,
>+                       size_t packet_size,
>+                       int mean_ms,
>                        int standard_deviation_ms) {
>     int unique_overuse = 0;
>     int last_overuse = -1;
>@@ -69,8 +69,11 @@ class OveruseDetectorTest : public ::testing::Test {
>     return unique_overuse;
>   }
> 
>-  int RunUntilOveruse(int packets_per_frame, size_t packet_size, int mean_ms,
>-                      int standard_deviation_ms, int drift_per_frame_ms) {
>+  int RunUntilOveruse(int packets_per_frame,
>+                      size_t packet_size,
>+                      int mean_ms,
>+                      int standard_deviation_ms,
>+                      int drift_per_frame_ms) {
>     // Simulate a higher send pace, that is too high.
>     for (int i = 0; i < 1000; ++i) {
>       for (int j = 0; j < packets_per_frame; ++j) {
>@@ -89,7 +92,8 @@ class OveruseDetectorTest : public ::testing::Test {
>     return -1;
>   }
> 
>-  void UpdateDetector(uint32_t rtp_timestamp, int64_t receive_time_ms,
>+  void UpdateDetector(uint32_t rtp_timestamp,
>+                      int64_t receive_time_ms,
>                       size_t packet_size) {
>     uint32_t timestamp_delta;
>     int64_t time_delta;
>@@ -189,8 +193,9 @@ TEST_F(OveruseDetectorTest, SimpleOveruse2000Kbit30fps) {
>                                         frame_duration_ms, sigma_ms);
> 
>   EXPECT_EQ(0, unique_overuse);
>-  int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
>-      frame_duration_ms, sigma_ms, drift_per_frame_ms);
>+  int frames_until_overuse =
>+      RunUntilOveruse(packets_per_frame, packet_size, frame_duration_ms,
>+                      sigma_ms, drift_per_frame_ms);
>   EXPECT_EQ(7, frames_until_overuse);
> }
> 
>@@ -204,8 +209,9 @@ TEST_F(OveruseDetectorTest, SimpleOveruse100kbit10fps) {
>                                         frame_duration_ms, sigma_ms);
> 
>   EXPECT_EQ(0, unique_overuse);
>-  int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
>-      frame_duration_ms, sigma_ms, drift_per_frame_ms);
>+  int frames_until_overuse =
>+      RunUntilOveruse(packets_per_frame, packet_size, frame_duration_ms,
>+                      sigma_ms, drift_per_frame_ms);
>   EXPECT_EQ(7, frames_until_overuse);
> }
> 
>@@ -328,8 +334,9 @@ TEST_F(OveruseDetectorTest, MAYBE_LowGaussianVariance30Kbit3fps) {
>   int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
>                                         frame_duration_ms, sigma_ms);
>   EXPECT_EQ(0, unique_overuse);
>-  int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
>-      frame_duration_ms, sigma_ms, drift_per_frame_ms);
>+  int frames_until_overuse =
>+      RunUntilOveruse(packets_per_frame, packet_size, frame_duration_ms,
>+                      sigma_ms, drift_per_frame_ms);
>   EXPECT_EQ(20, frames_until_overuse);
> }
> 
>@@ -342,8 +349,9 @@ TEST_F(OveruseDetectorTest, LowGaussianVarianceFastDrift30Kbit3fps) {
>   int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
>                                         frame_duration_ms, sigma_ms);
>   EXPECT_EQ(0, unique_overuse);
>-  int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
>-      frame_duration_ms, sigma_ms, drift_per_frame_ms);
>+  int frames_until_overuse =
>+      RunUntilOveruse(packets_per_frame, packet_size, frame_duration_ms,
>+                      sigma_ms, drift_per_frame_ms);
>   EXPECT_EQ(4, frames_until_overuse);
> }
> 
>@@ -356,8 +364,9 @@ TEST_F(OveruseDetectorTest, HighGaussianVariance30Kbit3fps) {
>   int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
>                                         frame_duration_ms, sigma_ms);
>   EXPECT_EQ(0, unique_overuse);
>-  int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
>-      frame_duration_ms, sigma_ms, drift_per_frame_ms);
>+  int frames_until_overuse =
>+      RunUntilOveruse(packets_per_frame, packet_size, frame_duration_ms,
>+                      sigma_ms, drift_per_frame_ms);
>   EXPECT_EQ(44, frames_until_overuse);
> }
> 
>@@ -370,8 +379,9 @@ TEST_F(OveruseDetectorTest, HighGaussianVarianceFastDrift30Kbit3fps) {
>   int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
>                                         frame_duration_ms, sigma_ms);
>   EXPECT_EQ(0, unique_overuse);
>-  int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
>-      frame_duration_ms, sigma_ms, drift_per_frame_ms);
>+  int frames_until_overuse =
>+      RunUntilOveruse(packets_per_frame, packet_size, frame_duration_ms,
>+                      sigma_ms, drift_per_frame_ms);
>   EXPECT_EQ(4, frames_until_overuse);
> }
> 
>@@ -390,8 +400,9 @@ TEST_F(OveruseDetectorTest, MAYBE_LowGaussianVariance100Kbit5fps) {
>   int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
>                                         frame_duration_ms, sigma_ms);
>   EXPECT_EQ(0, unique_overuse);
>-  int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
>-      frame_duration_ms, sigma_ms, drift_per_frame_ms);
>+  int frames_until_overuse =
>+      RunUntilOveruse(packets_per_frame, packet_size, frame_duration_ms,
>+                      sigma_ms, drift_per_frame_ms);
>   EXPECT_EQ(20, frames_until_overuse);
> }
> 
>@@ -410,8 +421,9 @@ TEST_F(OveruseDetectorTest, MAYBE_HighGaussianVariance100Kbit5fps) {
>   int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
>                                         frame_duration_ms, sigma_ms);
>   EXPECT_EQ(0, unique_overuse);
>-  int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
>-      frame_duration_ms, sigma_ms, drift_per_frame_ms);
>+  int frames_until_overuse =
>+      RunUntilOveruse(packets_per_frame, packet_size, frame_duration_ms,
>+                      sigma_ms, drift_per_frame_ms);
>   EXPECT_EQ(44, frames_until_overuse);
> }
> 
>@@ -430,8 +442,9 @@ TEST_F(OveruseDetectorTest, MAYBE_LowGaussianVariance100Kbit10fps) {
>   int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
>                                         frame_duration_ms, sigma_ms);
>   EXPECT_EQ(0, unique_overuse);
>-  int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
>-      frame_duration_ms, sigma_ms, drift_per_frame_ms);
>+  int frames_until_overuse =
>+      RunUntilOveruse(packets_per_frame, packet_size, frame_duration_ms,
>+                      sigma_ms, drift_per_frame_ms);
>   EXPECT_EQ(20, frames_until_overuse);
> }
> 
>@@ -450,8 +463,9 @@ TEST_F(OveruseDetectorTest, MAYBE_HighGaussianVariance100Kbit10fps) {
>   int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
>                                         frame_duration_ms, sigma_ms);
>   EXPECT_EQ(0, unique_overuse);
>-  int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
>-      frame_duration_ms, sigma_ms, drift_per_frame_ms);
>+  int frames_until_overuse =
>+      RunUntilOveruse(packets_per_frame, packet_size, frame_duration_ms,
>+                      sigma_ms, drift_per_frame_ms);
>   EXPECT_EQ(44, frames_until_overuse);
> }
> 
>@@ -470,8 +484,9 @@ TEST_F(OveruseDetectorTest, MAYBE_LowGaussianVariance300Kbit30fps) {
>   int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
>                                         frame_duration_ms, sigma_ms);
>   EXPECT_EQ(0, unique_overuse);
>-  int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
>-      frame_duration_ms, sigma_ms, drift_per_frame_ms);
>+  int frames_until_overuse =
>+      RunUntilOveruse(packets_per_frame, packet_size, frame_duration_ms,
>+                      sigma_ms, drift_per_frame_ms);
>   EXPECT_EQ(19, frames_until_overuse);
> }
> 
>@@ -484,8 +499,9 @@ TEST_F(OveruseDetectorTest, LowGaussianVarianceFastDrift300Kbit30fps) {
>   int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
>                                         frame_duration_ms, sigma_ms);
>   EXPECT_EQ(0, unique_overuse);
>-  int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
>-      frame_duration_ms, sigma_ms, drift_per_frame_ms);
>+  int frames_until_overuse =
>+      RunUntilOveruse(packets_per_frame, packet_size, frame_duration_ms,
>+                      sigma_ms, drift_per_frame_ms);
>   EXPECT_EQ(5, frames_until_overuse);
> }
> 
>@@ -498,8 +514,9 @@ TEST_F(OveruseDetectorTest, HighGaussianVariance300Kbit30fps) {
>   int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
>                                         frame_duration_ms, sigma_ms);
>   EXPECT_EQ(0, unique_overuse);
>-  int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
>-      frame_duration_ms, sigma_ms, drift_per_frame_ms);
>+  int frames_until_overuse =
>+      RunUntilOveruse(packets_per_frame, packet_size, frame_duration_ms,
>+                      sigma_ms, drift_per_frame_ms);
>   EXPECT_EQ(44, frames_until_overuse);
> }
> 
>@@ -512,8 +529,9 @@ TEST_F(OveruseDetectorTest, HighGaussianVarianceFastDrift300Kbit30fps) {
>   int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
>                                         frame_duration_ms, sigma_ms);
>   EXPECT_EQ(0, unique_overuse);
>-  int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
>-      frame_duration_ms, sigma_ms, drift_per_frame_ms);
>+  int frames_until_overuse =
>+      RunUntilOveruse(packets_per_frame, packet_size, frame_duration_ms,
>+                      sigma_ms, drift_per_frame_ms);
>   EXPECT_EQ(10, frames_until_overuse);
> }
> 
>@@ -532,8 +550,9 @@ TEST_F(OveruseDetectorTest, MAYBE_LowGaussianVariance1000Kbit30fps) {
>   int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
>                                         frame_duration_ms, sigma_ms);
>   EXPECT_EQ(0, unique_overuse);
>-  int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
>-      frame_duration_ms, sigma_ms, drift_per_frame_ms);
>+  int frames_until_overuse =
>+      RunUntilOveruse(packets_per_frame, packet_size, frame_duration_ms,
>+                      sigma_ms, drift_per_frame_ms);
>   EXPECT_EQ(19, frames_until_overuse);
> }
> 
>@@ -546,8 +565,9 @@ TEST_F(OveruseDetectorTest, LowGaussianVarianceFastDrift1000Kbit30fps) {
>   int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
>                                         frame_duration_ms, sigma_ms);
>   EXPECT_EQ(0, unique_overuse);
>-  int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
>-      frame_duration_ms, sigma_ms, drift_per_frame_ms);
>+  int frames_until_overuse =
>+      RunUntilOveruse(packets_per_frame, packet_size, frame_duration_ms,
>+                      sigma_ms, drift_per_frame_ms);
>   EXPECT_EQ(5, frames_until_overuse);
> }
> 
>@@ -560,8 +580,9 @@ TEST_F(OveruseDetectorTest, HighGaussianVariance1000Kbit30fps) {
>   int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
>                                         frame_duration_ms, sigma_ms);
>   EXPECT_EQ(0, unique_overuse);
>-  int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
>-      frame_duration_ms, sigma_ms, drift_per_frame_ms);
>+  int frames_until_overuse =
>+      RunUntilOveruse(packets_per_frame, packet_size, frame_duration_ms,
>+                      sigma_ms, drift_per_frame_ms);
>   EXPECT_EQ(44, frames_until_overuse);
> }
> 
>@@ -574,8 +595,9 @@ TEST_F(OveruseDetectorTest, HighGaussianVarianceFastDrift1000Kbit30fps) {
>   int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
>                                         frame_duration_ms, sigma_ms);
>   EXPECT_EQ(0, unique_overuse);
>-  int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
>-      frame_duration_ms, sigma_ms, drift_per_frame_ms);
>+  int frames_until_overuse =
>+      RunUntilOveruse(packets_per_frame, packet_size, frame_duration_ms,
>+                      sigma_ms, drift_per_frame_ms);
>   EXPECT_EQ(10, frames_until_overuse);
> }
> 
>@@ -594,8 +616,9 @@ TEST_F(OveruseDetectorTest, MAYBE_LowGaussianVariance2000Kbit30fps) {
>   int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
>                                         frame_duration_ms, sigma_ms);
>   EXPECT_EQ(0, unique_overuse);
>-  int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
>-      frame_duration_ms, sigma_ms, drift_per_frame_ms);
>+  int frames_until_overuse =
>+      RunUntilOveruse(packets_per_frame, packet_size, frame_duration_ms,
>+                      sigma_ms, drift_per_frame_ms);
>   EXPECT_EQ(19, frames_until_overuse);
> }
> 
>@@ -608,8 +631,9 @@ TEST_F(OveruseDetectorTest, LowGaussianVarianceFastDrift2000Kbit30fps) {
>   int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
>                                         frame_duration_ms, sigma_ms);
>   EXPECT_EQ(0, unique_overuse);
>-  int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
>-      frame_duration_ms, sigma_ms, drift_per_frame_ms);
>+  int frames_until_overuse =
>+      RunUntilOveruse(packets_per_frame, packet_size, frame_duration_ms,
>+                      sigma_ms, drift_per_frame_ms);
>   EXPECT_EQ(5, frames_until_overuse);
> }
> 
>@@ -622,8 +646,9 @@ TEST_F(OveruseDetectorTest, HighGaussianVariance2000Kbit30fps) {
>   int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
>                                         frame_duration_ms, sigma_ms);
>   EXPECT_EQ(0, unique_overuse);
>-  int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
>-      frame_duration_ms, sigma_ms, drift_per_frame_ms);
>+  int frames_until_overuse =
>+      RunUntilOveruse(packets_per_frame, packet_size, frame_duration_ms,
>+                      sigma_ms, drift_per_frame_ms);
>   EXPECT_EQ(44, frames_until_overuse);
> }
> 
>@@ -636,8 +661,9 @@ TEST_F(OveruseDetectorTest, HighGaussianVarianceFastDrift2000Kbit30fps) {
>   int unique_overuse = Run100000Samples(packets_per_frame, packet_size,
>                                         frame_duration_ms, sigma_ms);
>   EXPECT_EQ(0, unique_overuse);
>-  int frames_until_overuse = RunUntilOveruse(packets_per_frame, packet_size,
>-      frame_duration_ms, sigma_ms, drift_per_frame_ms);
>+  int frames_until_overuse =
>+      RunUntilOveruse(packets_per_frame, packet_size, frame_duration_ms,
>+                      sigma_ms, drift_per_frame_ms);
>   EXPECT_EQ(10, frames_until_overuse);
> }
> 
>@@ -648,9 +674,7 @@ class OveruseDetectorExperimentTest : public OveruseDetectorTest {
>             "WebRTC-AdaptiveBweThreshold/Enabled-0.01,0.00018/") {}
> 
>  protected:
>-  void SetUp() override {
>-    overuse_detector_.reset(new OveruseDetector());
>-  }
>+  void SetUp() override { overuse_detector_.reset(new OveruseDetector()); }
> 
>   test::ScopedFieldTrials override_field_trials_;
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/overuse_estimator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/overuse_estimator.cc
>index 3d8ab2ad60e..09de5c6e7c1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/overuse_estimator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/overuse_estimator.cc
>@@ -73,12 +73,12 @@ void OveruseEstimator::Update(int64_t t_delta,
>   }
> 
>   const double h[2] = {fs_delta, 1.0};
>-  const double Eh[2] = {E_[0][0]*h[0] + E_[0][1]*h[1],
>-                        E_[1][0]*h[0] + E_[1][1]*h[1]};
>+  const double Eh[2] = {E_[0][0] * h[0] + E_[0][1] * h[1],
>+                        E_[1][0] * h[0] + E_[1][1] * h[1]};
> 
>   BWE_TEST_LOGGING_PLOT(1, "d_ms", now_ms, slope_ * h[0] - offset_);
> 
>-  const double residual = t_ts_delta - slope_*h[0] - offset_;
>+  const double residual = t_ts_delta - slope_ * h[0] - offset_;
> 
>   const bool in_stable_state =
>       (current_hypothesis == BandwidthUsage::kBwNormal);
>@@ -92,13 +92,12 @@ void OveruseEstimator::Update(int64_t t_delta,
>                         min_frame_period, in_stable_state);
>   }
> 
>-  const double denom = var_noise_ + h[0]*Eh[0] + h[1]*Eh[1];
>+  const double denom = var_noise_ + h[0] * Eh[0] + h[1] * Eh[1];
> 
>-  const double K[2] = {Eh[0] / denom,
>-                       Eh[1] / denom};
>+  const double K[2] = {Eh[0] / denom, Eh[1] / denom};
> 
>-  const double IKh[2][2] = {{1.0 - K[0]*h[0], -K[0]*h[1]},
>-                            {-K[1]*h[0], 1.0 - K[1]*h[1]}};
>+  const double IKh[2][2] = {{1.0 - K[0] * h[0], -K[0] * h[1]},
>+                            {-K[1] * h[0], 1.0 - K[1] * h[1]}};
>   const double e00 = E_[0][0];
>   const double e01 = E_[0][1];
> 
>@@ -109,7 +108,8 @@ void OveruseEstimator::Update(int64_t t_delta,
>   E_[1][1] = e01 * IKh[1][0] + E_[1][1] * IKh[1][1];
> 
>   // The covariance matrix must be positive semi-definite.
>-  bool positive_semi_definite = E_[0][0] + E_[1][1] >= 0 &&
>+  bool positive_semi_definite =
>+      E_[0][0] + E_[1][1] >= 0 &&
>       E_[0][0] * E_[1][1] - E_[0][1] * E_[1][0] >= 0 && E_[0][0] >= 0;
>   assert(positive_semi_definite);
>   if (!positive_semi_definite) {
>@@ -150,16 +150,15 @@ void OveruseEstimator::UpdateNoiseEstimate(double residual,
>   // of the network. |alpha| is tuned for 30 frames per second, but is scaled
>   // according to |ts_delta|.
>   double alpha = 0.01;
>-  if (num_of_deltas_ > 10*30) {
>+  if (num_of_deltas_ > 10 * 30) {
>     alpha = 0.002;
>   }
>   // Only update the noise estimate if we're not over-using. |beta| is a
>   // function of alpha and the time delta since the previous update.
>   const double beta = pow(1 - alpha, ts_delta * 30.0 / 1000.0);
>-  avg_noise_ = beta * avg_noise_
>-              + (1 - beta) * residual;
>-  var_noise_ = beta * var_noise_
>-              + (1 - beta) * (avg_noise_ - residual) * (avg_noise_ - residual);
>+  avg_noise_ = beta * avg_noise_ + (1 - beta) * residual;
>+  var_noise_ = beta * var_noise_ +
>+               (1 - beta) * (avg_noise_ - residual) * (avg_noise_ - residual);
>   if (var_noise_ < 1) {
>     var_noise_ = 1;
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/overuse_estimator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/overuse_estimator.h
>index 61a7150a920..4c924938aa0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/overuse_estimator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/overuse_estimator.h
>@@ -34,20 +34,14 @@ class OveruseEstimator {
>               int64_t now_ms);
> 
>   // Returns the estimated noise/jitter variance in ms^2.
>-  double var_noise() const {
>-    return var_noise_;
>-  }
>+  double var_noise() const { return var_noise_; }
> 
>   // Returns the estimated inter-arrival time delta offset in ms.
>-  double offset() const {
>-    return offset_;
>-  }
>+  double offset() const { return offset_; }
> 
>   // Returns the number of deltas which the current over-use estimator state is
>   // based on.
>-  unsigned int num_of_deltas() const {
>-    return num_of_deltas_;
>-  }
>+  unsigned int num_of_deltas() const { return num_of_deltas_; }
> 
>  private:
>   double UpdateMinFramePeriod(double ts_delta);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc
>index fdc15237003..a7cfe4c90ac 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc
>@@ -20,7 +20,6 @@
> #include "rtc_base/logging.h"
> #include "rtc_base/thread_annotations.h"
> #include "system_wrappers/include/metrics.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -35,15 +34,15 @@ enum {
>   kExpectedNumberOfProbes = 3
> };
> 
>-static const double kTimestampToMs = 1000.0 /
>-    static_cast<double>(1 << kInterArrivalShift);
>+static const double kTimestampToMs =
>+    1000.0 / static_cast<double>(1 << kInterArrivalShift);
> 
>-template<typename K, typename V>
>+template <typename K, typename V>
> std::vector<K> Keys(const std::map<K, V>& map) {
>   std::vector<K> keys;
>   keys.reserve(map.size());
>   for (typename std::map<K, V>::const_iterator it = map.begin();
>-      it != map.end(); ++it) {
>+       it != map.end(); ++it) {
>     keys.push_back(it->first);
>   }
>   return keys;
>@@ -58,42 +57,44 @@ uint32_t ConvertMsTo24Bits(int64_t time_ms) {
>   return time_24_bits;
> }
> 
>+RemoteBitrateEstimatorAbsSendTime::~RemoteBitrateEstimatorAbsSendTime() =
>+    default;
>+
> bool RemoteBitrateEstimatorAbsSendTime::IsWithinClusterBounds(
>     int send_delta_ms,
>     const Cluster& cluster_aggregate) {
>-    if (cluster_aggregate.count == 0)
>-      return true;
>-    float cluster_mean = cluster_aggregate.send_mean_ms /
>-                         static_cast<float>(cluster_aggregate.count);
>-    return fabs(static_cast<float>(send_delta_ms) - cluster_mean) < 2.5f;
>-  }
>+  if (cluster_aggregate.count == 0)
>+    return true;
>+  float cluster_mean = cluster_aggregate.send_mean_ms /
>+                       static_cast<float>(cluster_aggregate.count);
>+  return fabs(static_cast<float>(send_delta_ms) - cluster_mean) < 2.5f;
>+}
> 
>-  void RemoteBitrateEstimatorAbsSendTime::AddCluster(
>-      std::list<Cluster>* clusters,
>-      Cluster* cluster) {
>-    cluster->send_mean_ms /= static_cast<float>(cluster->count);
>-    cluster->recv_mean_ms /= static_cast<float>(cluster->count);
>-    cluster->mean_size /= cluster->count;
>-    clusters->push_back(*cluster);
>-  }
>+void RemoteBitrateEstimatorAbsSendTime::AddCluster(std::list<Cluster>* clusters,
>+                                                   Cluster* cluster) {
>+  cluster->send_mean_ms /= static_cast<float>(cluster->count);
>+  cluster->recv_mean_ms /= static_cast<float>(cluster->count);
>+  cluster->mean_size /= cluster->count;
>+  clusters->push_back(*cluster);
>+}
> 
>-  RemoteBitrateEstimatorAbsSendTime::RemoteBitrateEstimatorAbsSendTime(
>-      RemoteBitrateObserver* observer,
>-      const Clock* clock)
>-      : clock_(clock),
>-        observer_(observer),
>-        inter_arrival_(),
>-        estimator_(),
>-        detector_(),
>-        incoming_bitrate_(kBitrateWindowMs, 8000),
>-        incoming_bitrate_initialized_(false),
>-        total_probes_received_(0),
>-        first_packet_time_ms_(-1),
>-        last_update_ms_(-1),
>-        uma_recorded_(false) {
>-    RTC_DCHECK(clock_);
>-    RTC_DCHECK(observer_);
>-    RTC_LOG(LS_INFO) << "RemoteBitrateEstimatorAbsSendTime: Instantiating.";
>+RemoteBitrateEstimatorAbsSendTime::RemoteBitrateEstimatorAbsSendTime(
>+    RemoteBitrateObserver* observer,
>+    const Clock* clock)
>+    : clock_(clock),
>+      observer_(observer),
>+      inter_arrival_(),
>+      estimator_(),
>+      detector_(),
>+      incoming_bitrate_(kBitrateWindowMs, 8000),
>+      incoming_bitrate_initialized_(false),
>+      total_probes_received_(0),
>+      first_packet_time_ms_(-1),
>+      last_update_ms_(-1),
>+      uma_recorded_(false) {
>+  RTC_DCHECK(clock_);
>+  RTC_DCHECK(observer_);
>+  RTC_LOG(LS_INFO) << "RemoteBitrateEstimatorAbsSendTime: Instantiating.";
> }
> 
> void RemoteBitrateEstimatorAbsSendTime::ComputeClusters(
>@@ -102,8 +103,7 @@ void RemoteBitrateEstimatorAbsSendTime::ComputeClusters(
>   int64_t prev_send_time = -1;
>   int64_t prev_recv_time = -1;
>   for (std::list<Probe>::const_iterator it = probes_.begin();
>-       it != probes_.end();
>-       ++it) {
>+       it != probes_.end(); ++it) {
>     if (prev_send_time >= 0) {
>       int send_delta_ms = it->send_time_ms - prev_send_time;
>       int recv_delta_ms = it->recv_time_ms - prev_recv_time;
>@@ -111,8 +111,7 @@ void RemoteBitrateEstimatorAbsSendTime::ComputeClusters(
>         ++current.num_above_min_delta;
>       }
>       if (!IsWithinClusterBounds(send_delta_ms, current)) {
>-        if (current.count >= kMinClusterSize &&
>-            current.send_mean_ms > 0.0f &&
>+        if (current.count >= kMinClusterSize && current.send_mean_ms > 0.0f &&
>             current.recv_mean_ms > 0.0f) {
>           AddCluster(clusters, &current);
>         }
>@@ -126,8 +125,7 @@ void RemoteBitrateEstimatorAbsSendTime::ComputeClusters(
>     prev_send_time = it->send_time_ms;
>     prev_recv_time = it->recv_time_ms;
>   }
>-  if (current.count >= kMinClusterSize &&
>-      current.send_mean_ms > 0.0f &&
>+  if (current.count >= kMinClusterSize && current.send_mean_ms > 0.0f &&
>       current.recv_mean_ms > 0.0f) {
>     AddCluster(clusters, &current);
>   }
>@@ -139,8 +137,7 @@ RemoteBitrateEstimatorAbsSendTime::FindBestProbe(
>   int highest_probe_bitrate_bps = 0;
>   std::list<Cluster>::const_iterator best_it = clusters.end();
>   for (std::list<Cluster>::const_iterator it = clusters.begin();
>-       it != clusters.end();
>-       ++it) {
>+       it != clusters.end(); ++it) {
>     if (it->send_mean_ms == 0 || it->recv_mean_ms == 0)
>       continue;
>     if (it->num_above_min_delta > it->count / 2 &&
>@@ -248,7 +245,7 @@ void RemoteBitrateEstimatorAbsSendTime::IncomingPacketInfo(
>   // here.
> 
>   // Check if incoming bitrate estimate is valid, and if it needs to be reset.
>-  rtc::Optional<uint32_t> incoming_bitrate =
>+  absl::optional<uint32_t> incoming_bitrate =
>       incoming_bitrate_.Rate(arrival_time_ms);
>   if (incoming_bitrate) {
>     incoming_bitrate_initialized_ = true;
>@@ -322,7 +319,7 @@ void RemoteBitrateEstimatorAbsSendTime::IncomingPacketInfo(
>           now_ms - last_update_ms_ > remote_rate_.GetFeedbackInterval()) {
>         update_estimate = true;
>       } else if (detector_.State() == BandwidthUsage::kBwOverusing) {
>-        rtc::Optional<uint32_t> incoming_rate =
>+        absl::optional<uint32_t> incoming_rate =
>             incoming_bitrate_.Rate(arrival_time_ms);
>         if (incoming_rate &&
>             remote_rate_.TimeToReduceFurther(now_ms, *incoming_rate)) {
>@@ -336,8 +333,7 @@ void RemoteBitrateEstimatorAbsSendTime::IncomingPacketInfo(
>       // We also have to update the estimate immediately if we are overusing
>       // and the target bitrate is too high compared to what we are receiving.
>       const RateControlInput input(detector_.State(),
>-                                   incoming_bitrate_.Rate(arrival_time_ms),
>-                                   estimator_->var_noise());
>+                                   incoming_bitrate_.Rate(arrival_time_ms));
>       target_bitrate_bps = remote_rate_.Update(&input, now_ms);
>       update_estimate = remote_rate_.ValidEstimate();
>       ssrcs = Keys(ssrcs_);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h
>index 9bf8d9a45c8..8e5aacbf55c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h
>@@ -69,7 +69,7 @@ class RemoteBitrateEstimatorAbsSendTime : public RemoteBitrateEstimator {
>  public:
>   RemoteBitrateEstimatorAbsSendTime(RemoteBitrateObserver* observer,
>                                     const Clock* clock);
>-  virtual ~RemoteBitrateEstimatorAbsSendTime() {}
>+  ~RemoteBitrateEstimatorAbsSendTime() override;
> 
>   void IncomingPacket(int64_t arrival_time_ms,
>                       size_t payload_size,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time_unittest.cc
>index ccfb64645b9..bf7f5940519 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time_unittest.cc
>@@ -14,14 +14,15 @@
> 
> namespace webrtc {
> 
>-class RemoteBitrateEstimatorAbsSendTimeTest :
>-    public RemoteBitrateEstimatorTest {
>+class RemoteBitrateEstimatorAbsSendTimeTest
>+    : public RemoteBitrateEstimatorTest {
>  public:
>   RemoteBitrateEstimatorAbsSendTimeTest() {}
>   virtual void SetUp() {
>     bitrate_estimator_.reset(new RemoteBitrateEstimatorAbsSendTime(
>         bitrate_observer_.get(), &clock_));
>   }
>+
>  protected:
>   RTC_DISALLOW_COPY_AND_ASSIGN(RemoteBitrateEstimatorAbsSendTimeTest);
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc
>index a914a846731..3d791f78487 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc
>@@ -19,7 +19,6 @@
> #include "rtc_base/logging.h"
> #include "system_wrappers/include/clock.h"
> #include "system_wrappers/include/metrics.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -76,8 +75,8 @@ void RemoteBitrateEstimatorSingleStream::IncomingPacket(
>     uma_recorded_ = true;
>   }
>   uint32_t ssrc = header.ssrc;
>-  uint32_t rtp_timestamp = header.timestamp +
>-      header.extension.transmissionTimeOffset;
>+  uint32_t rtp_timestamp =
>+      header.timestamp + header.extension.transmissionTimeOffset;
>   int64_t now_ms = clock_->TimeInMilliseconds();
>   rtc::CritScope cs(&crit_sect_);
>   SsrcOveruseEstimatorMap::iterator it = overuse_detectors_.find(ssrc);
>@@ -97,7 +96,7 @@ void RemoteBitrateEstimatorSingleStream::IncomingPacket(
>   estimator->last_packet_time_ms = now_ms;
> 
>   // Check if incoming bitrate estimate is valid, and if it needs to be reset.
>-  rtc::Optional<uint32_t> incoming_bitrate = incoming_bitrate_.Rate(now_ms);
>+  absl::optional<uint32_t> incoming_bitrate = incoming_bitrate_.Rate(now_ms);
>   if (incoming_bitrate) {
>     last_valid_incoming_bitrate_ = *incoming_bitrate;
>   } else if (last_valid_incoming_bitrate_ > 0) {
>@@ -124,7 +123,7 @@ void RemoteBitrateEstimatorSingleStream::IncomingPacket(
>                                estimator->estimator.num_of_deltas(), now_ms);
>   }
>   if (estimator->detector.State() == BandwidthUsage::kBwOverusing) {
>-    rtc::Optional<uint32_t> incoming_bitrate_bps =
>+    absl::optional<uint32_t> incoming_bitrate_bps =
>         incoming_bitrate_.Rate(now_ms);
>     if (incoming_bitrate_bps &&
>         (prior_state != BandwidthUsage::kBwOverusing ||
>@@ -152,12 +151,11 @@ int64_t RemoteBitrateEstimatorSingleStream::TimeUntilNextProcess() {
>   rtc::CritScope cs_(&crit_sect_);
>   RTC_DCHECK_GT(process_interval_ms_, 0);
>   return last_process_time_ + process_interval_ms_ -
>-      clock_->TimeInMilliseconds();
>+         clock_->TimeInMilliseconds();
> }
> 
> void RemoteBitrateEstimatorSingleStream::UpdateEstimate(int64_t now_ms) {
>   BandwidthUsage bw_state = BandwidthUsage::kBwNormal;
>-  double sum_var_noise = 0.0;
>   SsrcOveruseEstimatorMap::iterator it = overuse_detectors_.begin();
>   while (it != overuse_detectors_.end()) {
>     const int64_t time_of_last_received_packet =
>@@ -169,7 +167,6 @@ void RemoteBitrateEstimatorSingleStream::UpdateEstimate(int64_t now_ms) {
>       delete it->second;
>       overuse_detectors_.erase(it++);
>     } else {
>-      sum_var_noise += it->second->estimator.var_noise();
>       // Make sure that we trigger an over-use if any of the over-use detectors
>       // is detecting over-use.
>       if (it->second->detector.State() > bw_state) {
>@@ -184,11 +181,7 @@ void RemoteBitrateEstimatorSingleStream::UpdateEstimate(int64_t now_ms) {
>   }
>   AimdRateControl* remote_rate = GetRemoteRate();
> 
>-  double mean_noise_var = sum_var_noise /
>-      static_cast<double>(overuse_detectors_.size());
>-  const RateControlInput input(bw_state,
>-                               incoming_bitrate_.Rate(now_ms),
>-                               mean_noise_var);
>+  const RateControlInput input(bw_state, incoming_bitrate_.Rate(now_ms));
>   uint32_t target_bitrate = remote_rate->Update(&input, now_ms);
>   if (remote_rate->ValidEstimate()) {
>     process_interval_ms_ = remote_rate->GetFeedbackInterval();
>@@ -237,7 +230,7 @@ void RemoteBitrateEstimatorSingleStream::GetSsrcs(
>   ssrcs->resize(overuse_detectors_.size());
>   int i = 0;
>   for (SsrcOveruseEstimatorMap::const_iterator it = overuse_detectors_.begin();
>-      it != overuse_detectors_.end(); ++it, ++i) {
>+       it != overuse_detectors_.end(); ++it, ++i) {
>     (*ssrcs)[i] = it->first;
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h
>index fd4ff329799..18b435affdd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h
>@@ -27,7 +27,7 @@ class RemoteBitrateEstimatorSingleStream : public RemoteBitrateEstimator {
>  public:
>   RemoteBitrateEstimatorSingleStream(RemoteBitrateObserver* observer,
>                                      const Clock* clock);
>-  virtual ~RemoteBitrateEstimatorSingleStream();
>+  ~RemoteBitrateEstimatorSingleStream() override;
> 
>   void IncomingPacket(int64_t arrival_time_ms,
>                       size_t payload_size,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream_unittest.cc
>index c4bb76bb23f..120db13c7f2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream_unittest.cc
>@@ -14,14 +14,14 @@
> 
> namespace webrtc {
> 
>-class RemoteBitrateEstimatorSingleTest :
>-    public RemoteBitrateEstimatorTest {
>+class RemoteBitrateEstimatorSingleTest : public RemoteBitrateEstimatorTest {
>  public:
>   RemoteBitrateEstimatorSingleTest() {}
>   virtual void SetUp() {
>     bitrate_estimator_.reset(new RemoteBitrateEstimatorSingleStream(
>         bitrate_observer_.get(), &clock_));
>   }
>+
>  protected:
>   RTC_DISALLOW_COPY_AND_ASSIGN(RemoteBitrateEstimatorSingleTest);
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc
>index 0b3ff439d81..5e117942c1a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc
>@@ -69,8 +69,10 @@ int64_t RtpStream::GenerateFrame(int64_t time_now_us, PacketList* packets) {
>     RtpPacket* packet = new RtpPacket;
>     packet->send_time = time_now_us + kSendSideOffsetUs;
>     packet->size = packet_size;
>-    packet->rtp_timestamp = rtp_timestamp_offset_ + static_cast<uint32_t>(
>-        ((frequency_ / 1000) * packet->send_time + 500) / 1000);
>+    packet->rtp_timestamp =
>+        rtp_timestamp_offset_ +
>+        static_cast<uint32_t>(((frequency_ / 1000) * packet->send_time + 500) /
>+                              1000);
>     packet->ssrc = ssrc_;
>     packets->push_back(packet);
>   }
>@@ -90,11 +92,12 @@ RtpStream::RtcpPacket* RtpStream::Rtcp(int64_t time_now_us) {
>   }
>   RtcpPacket* rtcp = new RtcpPacket;
>   int64_t send_time_us = time_now_us + kSendSideOffsetUs;
>-  rtcp->timestamp = rtp_timestamp_offset_ + static_cast<uint32_t>(
>-      ((frequency_ / 1000) * send_time_us + 500) / 1000);
>+  rtcp->timestamp =
>+      rtp_timestamp_offset_ +
>+      static_cast<uint32_t>(((frequency_ / 1000) * send_time_us + 500) / 1000);
>   rtcp->ntp_secs = send_time_us / 1000000;
>-  rtcp->ntp_frac = static_cast<int64_t>((send_time_us % 1000000) *
>-      kNtpFracPerMs);
>+  rtcp->ntp_frac =
>+      static_cast<int64_t>((send_time_us % 1000000) * kNtpFracPerMs);
>   rtcp->ssrc = ssrc_;
>   next_rtcp_time_ = time_now_us + kRtcpIntervalUs;
>   return rtcp;
>@@ -119,12 +122,10 @@ bool RtpStream::Compare(const std::pair<uint32_t, RtpStream*>& left,
> }
> 
> StreamGenerator::StreamGenerator(int capacity, int64_t time_now)
>-    : capacity_(capacity),
>-      prev_arrival_time_us_(time_now) {}
>+    : capacity_(capacity), prev_arrival_time_us_(time_now) {}
> 
> StreamGenerator::~StreamGenerator() {
>-  for (StreamMap::iterator it = streams_.begin(); it != streams_.end();
>-      ++it) {
>+  for (StreamMap::iterator it = streams_.begin(); it != streams_.end(); ++it) {
>     delete it->second;
>   }
>   streams_.clear();
>@@ -153,8 +154,9 @@ void StreamGenerator::SetBitrateBps(int bitrate_bps) {
>   int total_bitrate_after = 0;
>   for (StreamMap::iterator it = streams_.begin(); it != streams_.end(); ++it) {
>     bitrate_before += it->second->bitrate_bps();
>-    int64_t bitrate_after = (bitrate_before * bitrate_bps +
>-        total_bitrate_before / 2) / total_bitrate_before;
>+    int64_t bitrate_after =
>+        (bitrate_before * bitrate_bps + total_bitrate_before / 2) /
>+        total_bitrate_before;
>     it->second->set_bitrate_bps(bitrate_after - total_bitrate_after);
>     total_bitrate_after += it->second->bitrate_bps();
>   }
>@@ -174,17 +176,18 @@ int64_t StreamGenerator::GenerateFrame(RtpStream::PacketList* packets,
>   assert(packets != NULL);
>   assert(packets->empty());
>   assert(capacity_ > 0);
>-  StreamMap::iterator it = std::min_element(streams_.begin(), streams_.end(),
>-                                            RtpStream::Compare);
>+  StreamMap::iterator it =
>+      std::min_element(streams_.begin(), streams_.end(), RtpStream::Compare);
>   (*it).second->GenerateFrame(time_now_us, packets);
>   int i = 0;
>   for (RtpStream::PacketList::iterator packet_it = packets->begin();
>-      packet_it != packets->end(); ++packet_it) {
>+       packet_it != packets->end(); ++packet_it) {
>     int capacity_bpus = capacity_ / 1000;
>     int64_t required_network_time_us =
>         (8 * 1000 * (*packet_it)->size + capacity_bpus / 2) / capacity_bpus;
>-    prev_arrival_time_us_ = std::max(time_now_us + required_network_time_us,
>-        prev_arrival_time_us_ + required_network_time_us);
>+    prev_arrival_time_us_ =
>+        std::max(time_now_us + required_network_time_us,
>+                 prev_arrival_time_us_ + required_network_time_us);
>     (*packet_it)->arrival_time = prev_arrival_time_us_;
>     ++i;
>   }
>@@ -196,21 +199,21 @@ int64_t StreamGenerator::GenerateFrame(RtpStream::PacketList* packets,
> RemoteBitrateEstimatorTest::RemoteBitrateEstimatorTest()
>     : clock_(100000000),
>       bitrate_observer_(new testing::TestBitrateObserver),
>-      stream_generator_(new testing::StreamGenerator(
>-          1e6,  // Capacity.
>-          clock_.TimeInMicroseconds())),
>+      stream_generator_(
>+          new testing::StreamGenerator(1e6,  // Capacity.
>+                                       clock_.TimeInMicroseconds())),
>       arrival_time_offset_ms_(0) {}
> 
> RemoteBitrateEstimatorTest::~RemoteBitrateEstimatorTest() {}
> 
> void RemoteBitrateEstimatorTest::AddDefaultStream() {
>-  stream_generator_->AddStream(new testing::RtpStream(
>-    30,          // Frames per second.
>-    3e5,         // Bitrate.
>-    1,           // SSRC.
>-    90000,       // RTP frequency.
>-    0xFFFFF000,  // Timestamp offset.
>-    0));         // RTCP receive time.
>+  stream_generator_->AddStream(
>+      new testing::RtpStream(30,          // Frames per second.
>+                             3e5,         // Bitrate.
>+                             1,           // SSRC.
>+                             90000,       // RTP frequency.
>+                             0xFFFFF000,  // Timestamp offset.
>+                             0));         // RTCP receive time.
> }
> 
> uint32_t RemoteBitrateEstimatorTest::AbsSendTime(int64_t t, int64_t denom) {
>@@ -250,8 +253,8 @@ bool RemoteBitrateEstimatorTest::GenerateAndProcessFrame(uint32_t ssrc,
>   RTC_DCHECK_GT(bitrate_bps, 0);
>   stream_generator_->SetBitrateBps(bitrate_bps);
>   testing::RtpStream::PacketList packets;
>-  int64_t next_time_us = stream_generator_->GenerateFrame(
>-      &packets, clock_.TimeInMicroseconds());
>+  int64_t next_time_us =
>+      stream_generator_->GenerateFrame(&packets, clock_.TimeInMicroseconds());
>   bool overuse = false;
>   while (!packets.empty()) {
>     testing::RtpStream::RtpPacket* packet = packets.front();
>@@ -339,8 +342,8 @@ void RemoteBitrateEstimatorTest::InitialBehaviorTestHelper(
>                    absolute_send_time);
>     clock_.AdvanceTimeMilliseconds(1000 / kFramerate);
>     timestamp += 90 * kFrameIntervalMs;
>-    absolute_send_time = AddAbsSendTime(absolute_send_time,
>-                                        kFrameIntervalAbsSendTime);
>+    absolute_send_time =
>+        AddAbsSendTime(absolute_send_time, kFrameIntervalAbsSendTime);
>   }
>   bitrate_estimator_->Process();
>   EXPECT_TRUE(bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_bps));
>@@ -377,19 +380,18 @@ void RemoteBitrateEstimatorTest::RateIncreaseReorderingTestHelper(
>                    absolute_send_time);
>     clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
>     timestamp += 90 * kFrameIntervalMs;
>-    absolute_send_time = AddAbsSendTime(absolute_send_time,
>-                                        kFrameIntervalAbsSendTime);
>+    absolute_send_time =
>+        AddAbsSendTime(absolute_send_time, kFrameIntervalAbsSendTime);
>   }
>   bitrate_estimator_->Process();
>   EXPECT_TRUE(bitrate_observer_->updated());
>-  EXPECT_NEAR(expected_bitrate_bps,
>-              bitrate_observer_->latest_bitrate(),
>+  EXPECT_NEAR(expected_bitrate_bps, bitrate_observer_->latest_bitrate(),
>               kAcceptedBitrateErrorBps);
>   for (int i = 0; i < 10; ++i) {
>     clock_.AdvanceTimeMilliseconds(2 * kFrameIntervalMs);
>     timestamp += 2 * 90 * kFrameIntervalMs;
>-    absolute_send_time = AddAbsSendTime(absolute_send_time,
>-                                        2 * kFrameIntervalAbsSendTime);
>+    absolute_send_time =
>+        AddAbsSendTime(absolute_send_time, 2 * kFrameIntervalAbsSendTime);
>     IncomingPacket(kDefaultSsrc, 1000, clock_.TimeInMilliseconds(), timestamp,
>                    absolute_send_time);
>     IncomingPacket(
>@@ -400,8 +402,7 @@ void RemoteBitrateEstimatorTest::RateIncreaseReorderingTestHelper(
>   }
>   bitrate_estimator_->Process();
>   EXPECT_TRUE(bitrate_observer_->updated());
>-  EXPECT_NEAR(expected_bitrate_bps,
>-              bitrate_observer_->latest_bitrate(),
>+  EXPECT_NEAR(expected_bitrate_bps, bitrate_observer_->latest_bitrate(),
>               kAcceptedBitrateErrorBps);
> }
> 
>@@ -472,7 +473,8 @@ void RemoteBitrateEstimatorTest::CapacityDropTestHelper(
>     ASSERT_EQ(bitrate_sum, kStartBitrate);
>   }
>   if (wrap_time_stamp) {
>-    stream_generator_->set_rtp_timestamp_offset(kDefaultSsrc,
>+    stream_generator_->set_rtp_timestamp_offset(
>+        kDefaultSsrc,
>         std::numeric_limits<uint32_t>::max() - steady_state_time * 90000);
>   }
> 
>@@ -538,8 +540,8 @@ void RemoteBitrateEstimatorTest::TestTimestampGroupingTestHelper() {
>     bitrate_estimator_->Process();
>     clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
>     timestamp += 90 * kFrameIntervalMs;
>-    absolute_send_time = AddAbsSendTime(absolute_send_time,
>-                                        kFrameIntervalAbsSendTime);
>+    absolute_send_time =
>+        AddAbsSendTime(absolute_send_time, kFrameIntervalAbsSendTime);
>   }
>   EXPECT_TRUE(bitrate_observer_->updated());
>   EXPECT_GE(bitrate_observer_->latest_bitrate(), 400000u);
>@@ -558,8 +560,8 @@ void RemoteBitrateEstimatorTest::TestTimestampGroupingTestHelper() {
>                      absolute_send_time);
>       clock_.AdvanceTimeMilliseconds(kFrameIntervalMs / kTimestampGroupLength);
>       timestamp += 1;
>-      absolute_send_time = AddAbsSendTime(absolute_send_time,
>-                                          kSingleRtpTickAbsSendTime);
>+      absolute_send_time =
>+          AddAbsSendTime(absolute_send_time, kSingleRtpTickAbsSendTime);
>     }
>     // Increase time until next batch to simulate over-use.
>     clock_.AdvanceTimeMilliseconds(10);
>@@ -575,8 +577,7 @@ void RemoteBitrateEstimatorTest::TestTimestampGroupingTestHelper() {
>   EXPECT_LT(bitrate_observer_->latest_bitrate(), 400000u);
> }
> 
>-void RemoteBitrateEstimatorTest::TestWrappingHelper(
>-    int silence_time_s) {
>+void RemoteBitrateEstimatorTest::TestWrappingHelper(int silence_time_s) {
>   const int kFramerate = 100;
>   const int kFrameIntervalMs = 1000 / kFramerate;
>   const uint32_t kFrameIntervalAbsSendTime = AbsSendTime(1, kFramerate);
>@@ -588,8 +589,8 @@ void RemoteBitrateEstimatorTest::TestWrappingHelper(
>                    absolute_send_time);
>     timestamp += kFrameIntervalMs;
>     clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
>-    absolute_send_time = AddAbsSendTime(absolute_send_time,
>-                                        kFrameIntervalAbsSendTime);
>+    absolute_send_time =
>+        AddAbsSendTime(absolute_send_time, kFrameIntervalAbsSendTime);
>     bitrate_estimator_->Process();
>   }
>   uint32_t bitrate_before = 0;
>@@ -597,16 +598,16 @@ void RemoteBitrateEstimatorTest::TestWrappingHelper(
>   bitrate_estimator_->LatestEstimate(&ssrcs, &bitrate_before);
> 
>   clock_.AdvanceTimeMilliseconds(silence_time_s * 1000);
>-  absolute_send_time = AddAbsSendTime(absolute_send_time,
>-                                      AbsSendTime(silence_time_s, 1));
>+  absolute_send_time =
>+      AddAbsSendTime(absolute_send_time, AbsSendTime(silence_time_s, 1));
>   bitrate_estimator_->Process();
>   for (size_t i = 0; i < 21; ++i) {
>     IncomingPacket(kDefaultSsrc, 1000, clock_.TimeInMilliseconds(), timestamp,
>                    absolute_send_time);
>     timestamp += kFrameIntervalMs;
>     clock_.AdvanceTimeMilliseconds(2 * kFrameIntervalMs);
>-    absolute_send_time = AddAbsSendTime(absolute_send_time,
>-                                        kFrameIntervalAbsSendTime);
>+    absolute_send_time =
>+        AddAbsSendTime(absolute_send_time, kFrameIntervalAbsSendTime);
>     bitrate_estimator_->Process();
>   }
>   uint32_t bitrate_after = 0;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.cc
>index 1e42266f7de..8cde28d0278 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.cc
>@@ -10,8 +10,8 @@
> 
> #include "modules/remote_bitrate_estimator/remote_estimator_proxy.h"
> 
>-#include <limits>
> #include <algorithm>
>+#include <limits>
> 
> #include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
> #include "rtc_base/checks.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.h
>index 985ddd3eaf9..63994c49573 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy.h
>@@ -34,7 +34,7 @@ class RemoteEstimatorProxy : public RemoteBitrateEstimator {
>  public:
>   RemoteEstimatorProxy(const Clock* clock,
>                        TransportFeedbackSenderInterface* feedback_sender);
>-  virtual ~RemoteEstimatorProxy();
>+  ~RemoteEstimatorProxy() override;
> 
>   void IncomingPacket(int64_t arrival_time_ms,
>                       size_t payload_size,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy_unittest.cc
>index 4197fcc7279..6c1fc8cd934 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/remote_estimator_proxy_unittest.cc
>@@ -8,8 +8,8 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "modules/pacing/packet_router.h"
> #include "modules/remote_bitrate_estimator/remote_estimator_proxy.h"
>+#include "modules/pacing/packet_router.h"
> #include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
> #include "system_wrappers/include/clock.h"
> #include "test/gmock.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/send_time_history.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/send_time_history.cc
>deleted file mode 100644
>index 014080530a0..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/send_time_history.cc
>+++ /dev/null
>@@ -1,89 +0,0 @@
>-/*
>- *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "modules/remote_bitrate_estimator/include/send_time_history.h"
>-
>-#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>-#include "rtc_base/checks.h"
>-#include "system_wrappers/include/clock.h"
>-
>-namespace webrtc {
>-
>-SendTimeHistory::SendTimeHistory(const Clock* clock,
>-                                 int64_t packet_age_limit_ms)
>-    : clock_(clock), packet_age_limit_ms_(packet_age_limit_ms) {}
>-
>-SendTimeHistory::~SendTimeHistory() {}
>-
>-void SendTimeHistory::AddAndRemoveOld(const PacketFeedback& packet) {
>-  int64_t now_ms = clock_->TimeInMilliseconds();
>-  // Remove old.
>-  while (!history_.empty() &&
>-         now_ms - history_.begin()->second.creation_time_ms >
>-             packet_age_limit_ms_) {
>-    // TODO(sprang): Warn if erasing (too many) old items?
>-    history_.erase(history_.begin());
>-  }
>-
>-  // Add new.
>-  int64_t unwrapped_seq_num = seq_num_unwrapper_.Unwrap(packet.sequence_number);
>-  history_.insert(std::make_pair(unwrapped_seq_num, packet));
>-}
>-
>-bool SendTimeHistory::OnSentPacket(uint16_t sequence_number,
>-                                   int64_t send_time_ms) {
>-  int64_t unwrapped_seq_num = seq_num_unwrapper_.Unwrap(sequence_number);
>-  auto it = history_.find(unwrapped_seq_num);
>-  if (it == history_.end())
>-    return false;
>-  it->second.send_time_ms = send_time_ms;
>-  return true;
>-}
>-
>-bool SendTimeHistory::GetFeedback(PacketFeedback* packet_feedback,
>-                                  bool remove) {
>-  RTC_DCHECK(packet_feedback);
>-  int64_t unwrapped_seq_num =
>-      seq_num_unwrapper_.Unwrap(packet_feedback->sequence_number);
>-  latest_acked_seq_num_.emplace(
>-      std::max(unwrapped_seq_num, latest_acked_seq_num_.value_or(0)));
>-  RTC_DCHECK_GE(*latest_acked_seq_num_, 0);
>-  auto it = history_.find(unwrapped_seq_num);
>-  if (it == history_.end())
>-    return false;
>-
>-  // Save arrival_time not to overwrite it.
>-  int64_t arrival_time_ms = packet_feedback->arrival_time_ms;
>-  *packet_feedback = it->second;
>-  packet_feedback->arrival_time_ms = arrival_time_ms;
>-
>-  if (remove)
>-    history_.erase(it);
>-  return true;
>-}
>-
>-size_t SendTimeHistory::GetOutstandingBytes(uint16_t local_net_id,
>-                                            uint16_t remote_net_id) const {
>-  size_t outstanding_bytes = 0;
>-  auto unacked_it = history_.begin();
>-  if (latest_acked_seq_num_) {
>-    unacked_it = history_.lower_bound(*latest_acked_seq_num_);
>-  }
>-  for (; unacked_it != history_.end(); ++unacked_it) {
>-    if (unacked_it->second.local_net_id == local_net_id &&
>-        unacked_it->second.remote_net_id == remote_net_id &&
>-        unacked_it->second.send_time_ms >= 0) {
>-      outstanding_bytes += unacked_it->second.payload_size;
>-    }
>-  }
>-  return outstanding_bytes;
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/bwe.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/bwe.cc
>index cf72f29d243..1d9b7278f59 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/bwe.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/bwe.cc
>@@ -10,6 +10,7 @@
> 
> #include "modules/remote_bitrate_estimator/test/bwe.h"
> 
>+#include <algorithm>
> #include <limits>
> 
> #include "modules/remote_bitrate_estimator/test/estimators/bbr.h"
>@@ -18,30 +19,26 @@
> #include "modules/remote_bitrate_estimator/test/estimators/send_side.h"
> #include "modules/remote_bitrate_estimator/test/estimators/tcp.h"
> #include "rtc_base/constructormagic.h"
>+#include "rtc_base/numerics/safe_conversions.h"
>+#include "rtc_base/system/fallthrough.h"
> 
> namespace webrtc {
> namespace testing {
> namespace bwe {
> 
>-// With the assumption that packet loss is lower than 97%, the max gap
>-// between elements in the set is lower than 0x8000, hence we have a
>-// total order in the set. For (x,y,z) subset of the LinkedSet,
>-// (x<=y and y<=z) ==> x<=z so the set can be sorted.
> const int kSetCapacity = 1000;
> 
> BweReceiver::BweReceiver(int flow_id)
>     : flow_id_(flow_id),
>       received_packets_(kSetCapacity),
>       rate_counter_(),
>-      loss_account_() {
>-}
>+      loss_account_() {}
> 
> BweReceiver::BweReceiver(int flow_id, int64_t window_size_ms)
>     : flow_id_(flow_id),
>       received_packets_(kSetCapacity),
>       rate_counter_(window_size_ms),
>-      loss_account_() {
>-}
>+      loss_account_() {}
> 
> void BweReceiver::ReceivePacket(int64_t arrival_time_ms,
>                                 const MediaPacket& media_packet) {
>@@ -57,10 +54,14 @@ void BweReceiver::ReceivePacket(int64_t arrival_time_ms,
>                             static_cast<uint32_t>(media_packet.payload_size()));
> }
> 
>+FeedbackPacket* BweReceiver::GetFeedback(int64_t now_ms) {
>+  return nullptr;
>+}
>+
> class NullBweSender : public BweSender {
>  public:
>   NullBweSender() {}
>-  virtual ~NullBweSender() {}
>+  ~NullBweSender() override {}
> 
>   int GetFeedbackIntervalMs() const override { return 1000; }
>   void GiveFeedback(const FeedbackPacket& feedback) override {}
>@@ -97,7 +98,7 @@ BweSender* CreateBweSender(BandwidthEstimatorType estimator,
>     case kBbrEstimator:
>       return new BbrBweSender(observer, clock);
>     case kTcpEstimator:
>-      FALLTHROUGH();
>+      RTC_FALLTHROUGH();
>     case kNullEstimator:
>       return new NullBweSender();
>   }
>@@ -160,12 +161,7 @@ LossAccount BweReceiver::LinkedSetPacketLossRatio() {
>     return LossAccount();
>   }
> 
>-  uint16_t oldest_seq_num = received_packets_.OldestSeqNumber();
>-  uint16_t newest_seq_num = received_packets_.NewestSeqNumber();
>-
>-  size_t set_total_packets =
>-      static_cast<uint16_t>(newest_seq_num - oldest_seq_num + 1);
>-
>+  size_t set_total_packets = received_packets_.Range();
>   size_t set_received_packets = received_packets_.size();
>   size_t set_lost_packets = set_total_packets - set_received_packets;
> 
>@@ -190,78 +186,63 @@ float BweReceiver::RecentPacketLossRatio() {
>   PacketNodeIt node_it = received_packets_.begin();  // Latest.
> 
>   // Lowest timestamp limit, oldest one that should be checked.
>-  int64_t time_limit_ms = (*node_it)->arrival_time_ms - kPacketLossTimeWindowMs;
>+  int64_t time_limit_ms = node_it->arrival_time_ms - kPacketLossTimeWindowMs;
>   // Oldest and newest values found within the given time window.
>-  uint16_t oldest_seq_num = (*node_it)->sequence_number;
>-  uint16_t newest_seq_num = oldest_seq_num;
>+  int64_t oldest_seq_num = node_it->unwrapped_sequence_number;
>+  int64_t newest_seq_num = oldest_seq_num;
> 
>   while (node_it != received_packets_.end()) {
>-    if ((*node_it)->arrival_time_ms < time_limit_ms) {
>+    if (node_it->arrival_time_ms < time_limit_ms) {
>       break;
>     }
>-    uint16_t seq_num = (*node_it)->sequence_number;
>-    if (IsNewerSequenceNumber(seq_num, newest_seq_num)) {
>-      newest_seq_num = seq_num;
>-    }
>-    if (IsNewerSequenceNumber(oldest_seq_num, seq_num)) {
>-      oldest_seq_num = seq_num;
>-    }
>+    int64_t seq_num = node_it->unwrapped_sequence_number;
>+    newest_seq_num = std::max(newest_seq_num, seq_num);
>+    oldest_seq_num = std::min(oldest_seq_num, seq_num);
>     ++node_it;
>     ++number_packets_received;
>   }
>   // Interval width between oldest and newest sequence number.
>-  // There was an overflow if newest_seq_num < oldest_seq_num.
>-  int gap = static_cast<uint16_t>(newest_seq_num - oldest_seq_num + 1);
>+  int gap = rtc::dchecked_cast<int>(newest_seq_num - oldest_seq_num + 1);
> 
>   return static_cast<float>(gap - number_packets_received) / gap;
> }
> 
>+LinkedSet::LinkedSet(int capacity) : capacity_(capacity) {}
>+
> LinkedSet::~LinkedSet() {
>-  while (!empty())
>-    RemoveTail();
> }
> 
> void LinkedSet::Insert(uint16_t sequence_number,
>                        int64_t send_time_ms,
>                        int64_t arrival_time_ms,
>                        size_t payload_size) {
>-  auto it = map_.find(sequence_number);
>+  auto unwrapped_sequence_number = unwrapper_.Unwrap(sequence_number);
>+  auto it = map_.find(unwrapped_sequence_number);
>+  // Handle duplicate unwrapped sequence number.
>   if (it != map_.end()) {
>     PacketNodeIt node_it = it->second;
>-    PacketIdentifierNode* node = *node_it;
>-    node->arrival_time_ms = arrival_time_ms;
>-    if (node_it != list_.begin()) {
>-      list_.erase(node_it);
>-      list_.push_front(node);
>-      map_[sequence_number] = list_.begin();
>-    }
>+    PacketIdentifierNode& node = *node_it;
>+    node.arrival_time_ms = arrival_time_ms;
>+    // Move to front, without invalidating iterators.
>+    list_.splice(list_.begin(), list_, node_it);
>   } else {
>     if (size() == capacity_) {
>       RemoveTail();
>     }
>-    UpdateHead(new PacketIdentifierNode(sequence_number, send_time_ms,
>-                                        arrival_time_ms, payload_size));
>+    // Push new element to the front of the list and insert it in the map.
>+    PacketIdentifierNode new_head(unwrapped_sequence_number, send_time_ms,
>+                                  arrival_time_ms, payload_size);
>+    list_.push_front(new_head);
>+    map_[unwrapped_sequence_number] = list_.begin();
>   }
> }
> 
>-void LinkedSet::Insert(PacketIdentifierNode packet_identifier) {
>-  Insert(packet_identifier.sequence_number, packet_identifier.send_time_ms,
>-         packet_identifier.arrival_time_ms, packet_identifier.payload_size);
>-}
>-
> void LinkedSet::RemoveTail() {
>-  map_.erase(list_.back()->sequence_number);
>-  delete list_.back();
>-  list_.pop_back();
>-}
>-void LinkedSet::UpdateHead(PacketIdentifierNode* new_head) {
>-  list_.push_front(new_head);
>-  map_[new_head->sequence_number] = list_.begin();
>+  Erase(--list_.end());
> }
> 
> void LinkedSet::Erase(PacketNodeIt node_it) {
>-  map_.erase((*node_it)->sequence_number);
>-  delete (*node_it);
>+  map_.erase(node_it->unwrapped_sequence_number);
>   list_.erase(node_it);
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/bwe.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/bwe.h
>index 96aed21c6b4..350a249294c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/bwe.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/bwe.h
>@@ -21,19 +21,12 @@
> #include "modules/remote_bitrate_estimator/test/packet.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/gtest_prod_util.h"
>+#include "rtc_base/numerics/sequence_number_util.h"
> 
> namespace webrtc {
> namespace testing {
> namespace bwe {
> 
>-// Overload map comparator.
>-class SequenceNumberOlderThan {
>- public:
>-  bool operator()(uint16_t seq_num_1, uint16_t seq_num_2) const {
>-    return IsNewerSequenceNumber(seq_num_2, seq_num_1);
>-  }
>-};
>-
> // Holds information for computing global packet loss.
> struct LossAccount {
>   LossAccount() : num_total(0), num_lost(0) {}
>@@ -49,42 +42,40 @@ struct LossAccount {
> // Holds only essential information about packets to be saved for
> // further use, e.g. for calculating packet loss and receiving rate.
> struct PacketIdentifierNode {
>-  PacketIdentifierNode(uint16_t sequence_number,
>+  PacketIdentifierNode(int64_t unwrapped_sequence_number,
>                        int64_t send_time_ms,
>                        int64_t arrival_time_ms,
>                        size_t payload_size)
>-      : sequence_number(sequence_number),
>+      : unwrapped_sequence_number(unwrapped_sequence_number),
>         send_time_ms(send_time_ms),
>         arrival_time_ms(arrival_time_ms),
>         payload_size(payload_size) {}
> 
>-  uint16_t sequence_number;
>+  int64_t unwrapped_sequence_number;
>   int64_t send_time_ms;
>   int64_t arrival_time_ms;
>   size_t payload_size;
> };
> 
>-typedef std::list<PacketIdentifierNode*>::iterator PacketNodeIt;
>+typedef std::list<PacketIdentifierNode>::iterator PacketNodeIt;
> 
> // FIFO implementation for a limited capacity set.
> // Used for keeping the latest arrived packets while avoiding duplicates.
> // Allows efficient insertion, deletion and search.
> class LinkedSet {
>  public:
>-  explicit LinkedSet(int capacity) : capacity_(capacity) {}
>+  explicit LinkedSet(int capacity);
>   ~LinkedSet();
> 
>   // If the arriving packet (identified by its sequence number) is already
>-  // in the LinkedSet, move its Node to the head of the list. Else, create
>-  // a PacketIdentifierNode n_ and then UpdateHead(n_), calling RemoveTail()
>-  // if the LinkedSet reached its maximum capacity.
>+  // in the LinkedSet, move its Node to the head of the list.
>+  // Else, add a PacketIdentifierNode n_ at the head of the list,
>+  // calling RemoveTail() if the LinkedSet reached its maximum capacity.
>   void Insert(uint16_t sequence_number,
>               int64_t send_time_ms,
>               int64_t arrival_time_ms,
>               size_t payload_size);
> 
>-  void Insert(PacketIdentifierNode packet_identifier);
>-
>   PacketNodeIt begin() { return list_.begin(); }
>   PacketNodeIt end() { return list_.end(); }
> 
>@@ -92,9 +83,10 @@ class LinkedSet {
>   size_t size() const { return list_.size(); }
>   size_t capacity() const { return capacity_; }
> 
>-  uint16_t OldestSeqNumber() const { return empty() ? 0 : map_.begin()->first; }
>-  uint16_t NewestSeqNumber() const {
>-    return empty() ? 0 : map_.rbegin()->first;
>+  // Return size of interval covering current set, i.e.:
>+  // unwrapped newest seq number - unwrapped oldest seq number + 1
>+  int64_t Range() const {
>+    return empty() ? 0 : map_.rbegin()->first - map_.begin()->first + 1;
>   }
> 
>   void Erase(PacketNodeIt node_it);
>@@ -102,11 +94,12 @@ class LinkedSet {
>  private:
>   // Pop oldest element from the back of the list and remove it from the map.
>   void RemoveTail();
>-  // Add new element to the front of the list and insert it in the map.
>-  void UpdateHead(PacketIdentifierNode* new_head);
>   size_t capacity_;
>-  std::map<uint16_t, PacketNodeIt, SequenceNumberOlderThan> map_;
>-  std::list<PacketIdentifierNode*> list_;
>+  // We want to keep track of the current oldest and newest sequence_numbers.
>+  // To get strict weak ordering, we unwrap uint16_t into an int64_t.
>+  SeqNumUnwrapper<uint16_t> unwrapper_;
>+  std::map<int64_t, PacketNodeIt> map_;
>+  std::list<PacketIdentifierNode> list_;
> };
> 
> const int kMinBitrateKbps = 10;
>@@ -116,7 +109,7 @@ class BweSender : public Module {
>  public:
>   BweSender() {}
>   explicit BweSender(int bitrate_kbps) : bitrate_kbps_(bitrate_kbps) {}
>-  virtual ~BweSender() {}
>+  ~BweSender() override {}
> 
>   virtual int GetFeedbackIntervalMs() const = 0;
>   virtual void GiveFeedback(const FeedbackPacket& feedback) = 0;
>@@ -138,7 +131,7 @@ class BweReceiver {
> 
>   virtual void ReceivePacket(int64_t arrival_time_ms,
>                              const MediaPacket& media_packet);
>-  virtual FeedbackPacket* GetFeedback(int64_t now_ms) { return NULL; }
>+  virtual FeedbackPacket* GetFeedback(int64_t now_ms);
> 
>   size_t GetSetCapacity() { return received_packets_.capacity(); }
>   double BitrateWindowS() const { return rate_counter_.BitrateWindowS(); }
>@@ -181,7 +174,8 @@ enum BandwidthEstimatorType {
>   kBbrEstimator
> };
> 
>-const char* const bwe_names[] = {"Null", "NADA", "REMB", "GCC", "TCP", "BBR"};
>+const char* const bwe_names[] = {"Null",   "NADA", "REMB",
>+                                 "GoogCc", "TCP",  "BBR"};
> 
> int64_t GetAbsSendTimeInMs(uint32_t abs_send_time);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/bwe_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/bwe_test.cc
>index 71088ff56ee..817a5151007 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/bwe_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/bwe_test.cc
>@@ -34,8 +34,10 @@ namespace testing {
> namespace bwe {
> 
> PacketProcessorRunner::PacketProcessorRunner(PacketProcessor* processor)
>-    : processor_(processor) {
>-}
>+    : processor_(processor) {}
>+
>+PacketProcessorRunner::PacketProcessorRunner(
>+    const PacketProcessorRunner& runner) = default;
> 
> PacketProcessorRunner::~PacketProcessorRunner() {
>   for (Packet* packet : queue_)
>@@ -97,8 +99,7 @@ void PacketProcessorRunner::QueuePackets(Packets* batch,
> }
> 
> // Plot link capacity by default.
>-BweTest::BweTest() : BweTest(true) {
>-}
>+BweTest::BweTest() : BweTest(true) {}
> 
> BweTest::BweTest(bool plot_capacity)
>     : run_time_ms_(0),
>@@ -117,13 +118,15 @@ BweTest::~BweTest() {
> void BweTest::SetUp() {
>   const ::testing::TestInfo* const test_info =
>       ::testing::UnitTest::GetInstance()->current_test_info();
>-  std::string test_name =
>-      std::string(test_info->test_case_name()) + "_" +
>-      std::string(test_info->name());
>+  std::string test_name = std::string(test_info->test_case_name()) + "_" +
>+                          std::string(test_info->name());
>   BWE_TEST_LOGGING_GLOBAL_CONTEXT(test_name);
>   BWE_TEST_LOGGING_GLOBAL_ENABLE(false);
> }
> 
>+Link::Link() = default;
>+Link::~Link() = default;
>+
> void Link::AddPacketProcessor(PacketProcessor* processor,
>                               ProcessorType processor_type) {
>   assert(processor);
>@@ -769,8 +772,8 @@ void BweTest::RunLongTcpFairness(BandwidthEstimatorType bwe_type) {
>   // max_delay_ms = 1000;
> 
>   std::string title("5.6_Long_TCP_Fairness");
>-  std::string flow_name = std::string() +
>-      bwe_names[bwe_type] + 'x' + bwe_names[kTcpEstimator];
>+  std::string flow_name =
>+      std::string() + bwe_names[bwe_type] + 'x' + bwe_names[kTcpEstimator];
> 
>   RunFairnessTest(bwe_type, kNumRmcatFlows, kNumTcpFlows, kRunTimeS,
>                   kCapacityKbps, max_delay_ms, rtt_ms, kMaxJitterMs, kOffSetsMs,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/bwe_test.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/bwe_test.h
>index 43156e7fc7e..34030e99a64 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/bwe_test.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/bwe_test.h
>@@ -33,6 +33,7 @@ class PacketSender;
> class PacketProcessorRunner {
>  public:
>   explicit PacketProcessorRunner(PacketProcessor* processor);
>+  PacketProcessorRunner(const PacketProcessorRunner&);
>   ~PacketProcessorRunner();
> 
>   bool RunsProcessor(const PacketProcessor* processor) const;
>@@ -48,11 +49,12 @@ class PacketProcessorRunner {
> 
> class Link : public PacketProcessorListener {
>  public:
>-  virtual ~Link() {}
>+  Link();
>+  ~Link() override;
> 
>-  virtual void AddPacketProcessor(PacketProcessor* processor,
>-                                  ProcessorType type);
>-  virtual void RemovePacketProcessor(PacketProcessor* processor);
>+  void AddPacketProcessor(PacketProcessor* processor,
>+                          ProcessorType type) override;
>+  void RemovePacketProcessor(PacketProcessor* processor) override;
> 
>   void Run(int64_t run_for_ms, int64_t now_ms, Packets* packets);
> 
>@@ -133,8 +135,7 @@ class BweTest {
>   Link uplink_;
> 
>  private:
>-  void FindPacketsToProcess(const FlowIds& flow_ids, Packets* in,
>-                            Packets* out);
>+  void FindPacketsToProcess(const FlowIds& flow_ids, Packets* in, Packets* out);
>   void GiveFeedbackToAffectedSenders(PacketReceiver* receiver);
> 
>   int64_t run_time_ms_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/bwe_test_baselinefile.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/bwe_test_baselinefile.cc
>index 7b12f067cfd..9f2bc5a68ed 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/bwe_test_baselinefile.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/bwe_test_baselinefile.cc
>@@ -42,8 +42,7 @@ class BaseLineFileVerify : public BaseLineFileInterface {
>   // the baseline file is missing. This is the default when verifying files, but
>   // not when updating (i.e. we always write it out if missing).
>   BaseLineFileVerify(const std::string& filepath, bool allow_missing_file)
>-      : reader_(),
>-        fail_to_read_response_(false) {
>+      : reader_(), fail_to_read_response_(false) {
>     std::unique_ptr<ResourceFileReader> reader;
>     reader.reset(ResourceFileReader::Create(filepath, "bin"));
>     if (!reader.get()) {
>@@ -62,9 +61,9 @@ class BaseLineFileVerify : public BaseLineFileInterface {
>       }
>     }
>   }
>-  virtual ~BaseLineFileVerify() {}
>+  ~BaseLineFileVerify() override {}
> 
>-  virtual void Estimate(int64_t time_ms, uint32_t estimate_bps) {
>+  void Estimate(int64_t time_ms, uint32_t estimate_bps) override {
>     if (reader_.get()) {
>       uint32_t read_ms = 0;
>       uint32_t read_bps = 0;
>@@ -72,13 +71,13 @@ class BaseLineFileVerify : public BaseLineFileInterface {
>           reader_->Read(&read_bps) && read_bps == estimate_bps) {
>       } else {
>         printf("ERROR: Baseline differs starting at: %d ms (%d vs %d)!\n",
>-            static_cast<uint32_t>(time_ms), estimate_bps, read_bps);
>+               static_cast<uint32_t>(time_ms), estimate_bps, read_bps);
>         reader_.reset(NULL);
>       }
>     }
>   }
> 
>-  virtual bool VerifyOrWrite() {
>+  bool VerifyOrWrite() override {
>     if (reader_.get()) {
>       if (reader_->IsAtEnd()) {
>         return true;
>@@ -101,21 +100,19 @@ class BaseLineFileUpdate : public BaseLineFileInterface {
>  public:
>   BaseLineFileUpdate(const std::string& filepath,
>                      BaseLineFileInterface* verifier)
>-      : verifier_(verifier),
>-        output_content_(),
>-        filepath_(filepath) {
>+      : verifier_(verifier), output_content_(), filepath_(filepath) {
>     output_content_.push_back(kMagicMarker);
>     output_content_.push_back(kFileVersion1);
>   }
>-  virtual ~BaseLineFileUpdate() {}
>+  ~BaseLineFileUpdate() override {}
> 
>-  virtual void Estimate(int64_t time_ms, uint32_t estimate_bps) {
>+  void Estimate(int64_t time_ms, uint32_t estimate_bps) override {
>     verifier_->Estimate(time_ms, estimate_bps);
>     output_content_.push_back(static_cast<uint32_t>(time_ms));
>     output_content_.push_back(estimate_bps);
>   }
> 
>-  virtual bool VerifyOrWrite() {
>+  bool VerifyOrWrite() override {
>     if (!verifier_->VerifyOrWrite()) {
>       std::string dir_path = webrtc::test::OutputPath() + kResourceSubDir;
>       if (!webrtc::test::CreateDir(dir_path)) {
>@@ -126,13 +123,13 @@ class BaseLineFileUpdate : public BaseLineFileInterface {
>       writer.reset(OutputFileWriter::Create(filepath_, "bin"));
>       if (!writer.get()) {
>         printf("WARNING: Cannot create output file: %s.bin\n",
>-            filepath_.c_str());
>+               filepath_.c_str());
>         return false;
>       }
>       printf("NOTE: Writing baseline file for BWE test: %s.bin\n",
>              filepath_.c_str());
>       for (std::vector<uint32_t>::iterator it = output_content_.begin();
>-          it != output_content_.end(); ++it) {
>+           it != output_content_.end(); ++it) {
>         writer->Write(*it);
>       }
>       return true;
>@@ -150,7 +147,8 @@ class BaseLineFileUpdate : public BaseLineFileInterface {
> };
> 
> BaseLineFileInterface* BaseLineFileInterface::Create(
>-    const std::string& filename, bool write_output_file) {
>+    const std::string& filename,
>+    bool write_output_file) {
>   std::string filepath = filename;
>   std::replace(filepath.begin(), filepath.end(), '/', '_');
>   filepath = std::string(kResourceSubDir) + "/" + filepath;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/bwe_test_fileutils.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/bwe_test_fileutils.cc
>index ffd410ef371..afea787b1ac 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/bwe_test_fileutils.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/bwe_test_fileutils.cc
>@@ -80,8 +80,8 @@ bool OutputFileWriter::Write(uint32_t value) {
> 
> OutputFileWriter* OutputFileWriter::Create(const std::string& filename,
>                                            const std::string& extension) {
>-  std::string filepath = webrtc::test::OutputPath() + filename + "." +
>-      extension;
>+  std::string filepath =
>+      webrtc::test::OutputPath() + filename + "." + extension;
>   FILE* file = fopen(filepath.c_str(), "wb");
>   if (file == NULL) {
>     BWE_TEST_LOGGING_CONTEXT("OutputFileWriter");
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.cc
>index 2a6614fc0f7..1e513ee3151 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.cc
>@@ -16,11 +16,23 @@
> 
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/numerics/safe_minmax.h"
>+#include "rtc_base/system/unused.h"
> 
> namespace webrtc {
> namespace testing {
> namespace bwe {
> 
>+RateCounter::RateCounter(int64_t window_size_ms)
>+    : window_size_us_(1000 * window_size_ms),
>+      recently_received_packets_(0),
>+      recently_received_bytes_(0),
>+      last_accumulated_us_(0),
>+      window_() {}
>+
>+RateCounter::RateCounter() : RateCounter(1000) {}
>+
>+RateCounter::~RateCounter() = default;
>+
> class DelayCapHelper {
>  public:
>   // Max delay = 0 stands for +infinite.
>@@ -39,9 +51,7 @@ class DelayCapHelper {
>     return (max_delay_us_ == 0 || max_delay_us_ >= packet_delay_us);
>   }
> 
>-  const Stats<double>& delay_stats() const {
>-    return delay_stats_;
>-  }
>+  const Stats<double>& delay_stats() const { return delay_stats_; }
> 
>  private:
>   int64_t max_delay_us_;
>@@ -50,7 +60,7 @@ class DelayCapHelper {
>   RTC_DISALLOW_COPY_AND_ASSIGN(DelayCapHelper);
> };
> 
>-const FlowIds CreateFlowIds(const int *flow_ids_array, size_t num_flow_ids) {
>+const FlowIds CreateFlowIds(const int* flow_ids_array, size_t num_flow_ids) {
>   FlowIds flow_ids(&flow_ids_array[0], flow_ids_array + num_flow_ids);
>   return flow_ids;
> }
>@@ -95,32 +105,6 @@ double RateCounter::BitrateWindowS() const {
>   return static_cast<double>(window_size_us_) / (1000 * 1000);
> }
> 
>-Packet::Packet()
>-    : flow_id_(0),
>-      creation_time_us_(-1),
>-      send_time_us_(-1),
>-      sender_timestamp_us_(-1),
>-      payload_size_(0) {}
>-
>-Packet::Packet(int flow_id, int64_t send_time_us, size_t payload_size)
>-    : flow_id_(flow_id),
>-      creation_time_us_(send_time_us),
>-      send_time_us_(send_time_us),
>-      sender_timestamp_us_(send_time_us),
>-      payload_size_(payload_size) {}
>-
>-Packet::~Packet() {
>-}
>-
>-bool Packet::operator<(const Packet& rhs) const {
>-  return send_time_us_ < rhs.send_time_us_;
>-}
>-
>-void Packet::set_send_time_us(int64_t send_time_us) {
>-  assert(send_time_us >= 0);
>-  send_time_us_ = send_time_us;
>-}
>-
> MediaPacket::MediaPacket() {
>   memset(&header_, 0, sizeof(header_));
> }
>@@ -138,8 +122,7 @@ MediaPacket::MediaPacket(int flow_id,
>                          int64_t send_time_us,
>                          size_t payload_size,
>                          const RTPHeader& header)
>-    : Packet(flow_id, send_time_us, payload_size), header_(header) {
>-}
>+    : Packet(flow_id, send_time_us, payload_size), header_(header) {}
> 
> MediaPacket::MediaPacket(int64_t send_time_us, uint16_t sequence_number)
>     : Packet(0, send_time_us, 0) {
>@@ -149,8 +132,9 @@ MediaPacket::MediaPacket(int64_t send_time_us, uint16_t sequence_number)
> 
> void MediaPacket::SetAbsSendTimeMs(int64_t abs_send_time_ms) {
>   header_.extension.hasAbsoluteSendTime = true;
>-  header_.extension.absoluteSendTime = ((static_cast<int64_t>(abs_send_time_ms *
>-    (1 << 18)) + 500) / 1000) & 0x00fffffful;
>+  header_.extension.absoluteSendTime =
>+      ((static_cast<int64_t>(abs_send_time_ms * (1 << 18)) + 500) / 1000) &
>+      0x00fffffful;
> }
> 
> BbrBweFeedback::BbrBweFeedback(
>@@ -168,8 +152,7 @@ RembFeedback::RembFeedback(int flow_id,
>                            RTCPReportBlock report_block)
>     : FeedbackPacket(flow_id, send_time_us, last_send_time_ms),
>       estimated_bps_(estimated_bps),
>-      report_block_(report_block) {
>-}
>+      report_block_(report_block) {}
> 
> SendSideBweFeedback::SendSideBweFeedback(
>     int flow_id,
>@@ -272,7 +255,6 @@ RateCounterFilter::~RateCounterFilter() {
>   LogStats();
> }
> 
>-
> void RateCounterFilter::LogStats() {
>   BWE_TEST_LOGGING_CONTEXT("RateCounterFilter");
>   packets_per_second_stats_.Log("pps");
>@@ -315,15 +297,13 @@ void RateCounterFilter::RunFor(int64_t /*time_ms*/, Packets* in_out) {
> LossFilter::LossFilter(PacketProcessorListener* listener, int flow_id)
>     : PacketProcessor(listener, flow_id, kRegular),
>       random_(0x12345678),
>-      loss_fraction_(0.0f) {
>-}
>+      loss_fraction_(0.0f) {}
> 
> LossFilter::LossFilter(PacketProcessorListener* listener,
>                        const FlowIds& flow_ids)
>     : PacketProcessor(listener, flow_ids, kRegular),
>       random_(0x12345678),
>-      loss_fraction_(0.0f) {
>-}
>+      loss_fraction_(0.0f) {}
> 
> void LossFilter::SetLoss(float loss_percent) {
>   BWE_TEST_LOGGING_ENABLE(false);
>@@ -335,7 +315,7 @@ void LossFilter::SetLoss(float loss_percent) {
> 
> void LossFilter::RunFor(int64_t /*time_ms*/, Packets* in_out) {
>   assert(in_out);
>-  for (PacketsIt it = in_out->begin(); it != in_out->end(); ) {
>+  for (PacketsIt it = in_out->begin(); it != in_out->end();) {
>     if (random_.Rand<float>() < loss_fraction_) {
>       delete *it;
>       it = in_out->erase(it);
>@@ -350,15 +330,13 @@ const int64_t kDefaultOneWayDelayUs = 0;
> DelayFilter::DelayFilter(PacketProcessorListener* listener, int flow_id)
>     : PacketProcessor(listener, flow_id, kRegular),
>       one_way_delay_us_(kDefaultOneWayDelayUs),
>-      last_send_time_us_(0) {
>-}
>+      last_send_time_us_(0) {}
> 
> DelayFilter::DelayFilter(PacketProcessorListener* listener,
>                          const FlowIds& flow_ids)
>     : PacketProcessor(listener, flow_ids, kRegular),
>       one_way_delay_us_(kDefaultOneWayDelayUs),
>-      last_send_time_us_(0) {
>-}
>+      last_send_time_us_(0) {}
> 
> void DelayFilter::SetOneWayDelayMs(int64_t one_way_delay_ms) {
>   BWE_TEST_LOGGING_ENABLE(false);
>@@ -381,8 +359,7 @@ JitterFilter::JitterFilter(PacketProcessorListener* listener, int flow_id)
>       random_(0x89674523),
>       stddev_jitter_us_(0),
>       last_send_time_us_(0),
>-      reordering_(false) {
>-}
>+      reordering_(false) {}
> 
> JitterFilter::JitterFilter(PacketProcessorListener* listener,
>                            const FlowIds& flow_ids)
>@@ -390,8 +367,7 @@ JitterFilter::JitterFilter(PacketProcessorListener* listener,
>       random_(0x89674523),
>       stddev_jitter_us_(0),
>       last_send_time_us_(0),
>-      reordering_(false) {
>-}
>+      reordering_(false) {}
> 
> const int kN = 3;  // Truncated N sigma gaussian.
> 
>@@ -410,7 +386,7 @@ inline int64_t TruncatedNSigmaGaussian(Random* const random,
>   const int64_t gaussian_random = random->Gaussian(mean, std_dev);
>   return rtc::SafeClamp(gaussian_random, -kN * std_dev, kN * std_dev);
> }
>-}
>+}  // namespace
> 
> void JitterFilter::RunFor(int64_t /*time_ms*/, Packets* in_out) {
>   assert(in_out);
>@@ -448,15 +424,13 @@ int64_t JitterFilter::MeanUs() {
> ReorderFilter::ReorderFilter(PacketProcessorListener* listener, int flow_id)
>     : PacketProcessor(listener, flow_id, kRegular),
>       random_(0x27452389),
>-      reorder_fraction_(0.0f) {
>-}
>+      reorder_fraction_(0.0f) {}
> 
> ReorderFilter::ReorderFilter(PacketProcessorListener* listener,
>                              const FlowIds& flow_ids)
>     : PacketProcessor(listener, flow_ids, kRegular),
>       random_(0x27452389),
>-      reorder_fraction_(0.0f) {
>-}
>+      reorder_fraction_(0.0f) {}
> 
> void ReorderFilter::SetReorder(float reorder_percent) {
>   BWE_TEST_LOGGING_ENABLE(false);
>@@ -490,16 +464,14 @@ ChokeFilter::ChokeFilter(PacketProcessorListener* listener, int flow_id)
>     : PacketProcessor(listener, flow_id, kRegular),
>       capacity_kbps_(kDefaultKbps),
>       last_send_time_us_(0),
>-      delay_cap_helper_(new DelayCapHelper()) {
>-}
>+      delay_cap_helper_(new DelayCapHelper()) {}
> 
> ChokeFilter::ChokeFilter(PacketProcessorListener* listener,
>                          const FlowIds& flow_ids)
>     : PacketProcessor(listener, flow_ids, kRegular),
>       capacity_kbps_(kDefaultKbps),
>       last_send_time_us_(0),
>-      delay_cap_helper_(new DelayCapHelper()) {
>-}
>+      delay_cap_helper_(new DelayCapHelper()) {}
> 
> ChokeFilter::~ChokeFilter() {}
> 
>@@ -515,7 +487,7 @@ uint32_t ChokeFilter::capacity_kbps() {
> 
> void ChokeFilter::RunFor(int64_t /*time_ms*/, Packets* in_out) {
>   assert(in_out);
>-  for (PacketsIt it = in_out->begin(); it != in_out->end(); ) {
>+  for (PacketsIt it = in_out->begin(); it != in_out->end();) {
>     int64_t earliest_send_time_us =
>         std::max(last_send_time_us_, (*it)->send_time_us());
>     int64_t new_send_time_us =
>@@ -556,8 +528,7 @@ TraceBasedDeliveryFilter::TraceBasedDeliveryFilter(
>       name_(""),
>       delay_cap_helper_(new DelayCapHelper()),
>       packets_per_second_stats_(),
>-      kbps_stats_() {
>-}
>+      kbps_stats_() {}
> 
> TraceBasedDeliveryFilter::TraceBasedDeliveryFilter(
>     PacketProcessorListener* listener,
>@@ -571,8 +542,7 @@ TraceBasedDeliveryFilter::TraceBasedDeliveryFilter(
>       name_(""),
>       delay_cap_helper_(new DelayCapHelper()),
>       packets_per_second_stats_(),
>-      kbps_stats_() {
>-}
>+      kbps_stats_() {}
> 
> TraceBasedDeliveryFilter::TraceBasedDeliveryFilter(
>     PacketProcessorListener* listener,
>@@ -587,11 +557,9 @@ TraceBasedDeliveryFilter::TraceBasedDeliveryFilter(
>       name_(name),
>       delay_cap_helper_(new DelayCapHelper()),
>       packets_per_second_stats_(),
>-      kbps_stats_() {
>-}
>+      kbps_stats_() {}
> 
>-TraceBasedDeliveryFilter::~TraceBasedDeliveryFilter() {
>-}
>+TraceBasedDeliveryFilter::~TraceBasedDeliveryFilter() {}
> 
> bool TraceBasedDeliveryFilter::Init(const std::string& filename) {
>   FILE* trace_file = fopen(filename.c_str(), "r");
>@@ -704,6 +672,10 @@ VideoSource::VideoSource(int flow_id,
>   prototype_header_.sequenceNumber = 0xf000u;
> }
> 
>+int VideoSource::flow_id() const {
>+  return flow_id_;
>+}
>+
> uint32_t VideoSource::NextFrameSize() {
>   return frame_size_bytes_;
> }
>@@ -769,8 +741,7 @@ AdaptiveVideoSource::AdaptiveVideoSource(int flow_id,
>                                          uint32_t kbps,
>                                          uint32_t ssrc,
>                                          int64_t first_frame_offset_ms)
>-    : VideoSource(flow_id, fps, kbps, ssrc, first_frame_offset_ms) {
>-}
>+    : VideoSource(flow_id, fps, kbps, ssrc, first_frame_offset_ms) {}
> 
> void AdaptiveVideoSource::SetBitrateBps(int bitrate_bps) {
>   bits_per_second_ = bitrate_bps;
>@@ -787,8 +758,7 @@ PeriodicKeyFrameSource::PeriodicKeyFrameSource(int flow_id,
>       key_frame_interval_(key_frame_interval),
>       frame_counter_(0),
>       compensation_bytes_(0),
>-      compensation_per_frame_(0) {
>-}
>+      compensation_per_frame_(0) {}
> 
> uint32_t PeriodicKeyFrameSource::NextFrameSize() {
>   uint32_t payload_size = frame_size_bytes_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.h
>index 983a79a0931..b67a218bc27 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework.h
>@@ -34,7 +34,6 @@
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/random.h"
> #include "system_wrappers/include/clock.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -47,14 +46,9 @@ class DelayCapHelper;
> 
> class RateCounter {
>  public:
>-  explicit RateCounter(int64_t window_size_ms)
>-      : window_size_us_(1000 * window_size_ms),
>-        recently_received_packets_(0),
>-        recently_received_bytes_(0),
>-        last_accumulated_us_(0),
>-        window_() {}
>-
>-  RateCounter() : RateCounter(1000) {}
>+  explicit RateCounter(int64_t window_size_ms);
>+  RateCounter();
>+  ~RateCounter();
> 
>   void UpdateRates(int64_t send_time_us, uint32_t payload_size);
> 
>@@ -75,7 +69,7 @@ class RateCounter {
> };
> 
> typedef std::set<int> FlowIds;
>-const FlowIds CreateFlowIds(const int *flow_ids_array, size_t num_flow_ids);
>+const FlowIds CreateFlowIds(const int* flow_ids_array, size_t num_flow_ids);
> const FlowIds CreateFlowIdRange(int initial_value, int last_value);
> 
> template <typename T>
>@@ -85,7 +79,8 @@ bool DereferencingComparator(const T* const& a, const T* const& b) {
>   return *a < *b;
> }
> 
>-template<typename T> class Stats {
>+template <typename T>
>+class Stats {
>  public:
>   Stats()
>       : data_(),
>@@ -95,12 +90,9 @@ template<typename T> class Stats {
>         mean_(0),
>         variance_(0),
>         min_(0),
>-        max_(0) {
>-  }
>+        max_(0) {}
> 
>-  void Push(T data_point) {
>-    data_.push_back(data_point);
>-  }
>+  void Push(T data_point) { data_.push_back(data_point); }
> 
>   T GetMean() {
>     if (last_mean_count_ != data_.size()) {
>@@ -125,9 +117,7 @@ template<typename T> class Stats {
>     }
>     return variance_;
>   }
>-  T GetStdDev() {
>-    return sqrt(static_cast<double>(GetVariance()));
>-  }
>+  T GetStdDev() { return sqrt(static_cast<double>(GetVariance())); }
>   T GetMin() {
>     RefreshMinMax();
>     return min_;
>@@ -139,14 +129,14 @@ template<typename T> class Stats {
> 
>   std::string AsString() {
>     std::stringstream ss;
>-    ss << (GetMean() >= 0 ? GetMean() : -1) << ", " <<
>-        (GetStdDev() >= 0 ? GetStdDev() : -1);
>+    ss << (GetMean() >= 0 ? GetMean() : -1) << ", "
>+       << (GetStdDev() >= 0 ? GetStdDev() : -1);
>     return ss.str();
>   }
> 
>   void Log(const std::string& units) {
>-    BWE_TEST_LOGGING_LOG5("", "%f %s\t+/-%f\t[%f,%f]",
>-        GetMean(), units.c_str(), GetStdDev(), GetMin(), GetMax());
>+    BWE_TEST_LOGGING_LOG5("", "%f %s\t+/-%f\t[%f,%f]", GetMean(), units.c_str(),
>+                          GetStdDev(), GetMin(), GetMax());
>   }
> 
>  private:
>@@ -240,12 +230,12 @@ class RateCounterFilter : public PacketProcessor {
>                     const char* name,
>                     int64_t start_plotting_time_ms,
>                     const std::string& algorithm_name);
>-  virtual ~RateCounterFilter();
>+  ~RateCounterFilter() override;
> 
>   void LogStats();
>   Stats<double> GetBitrateStats() const;
>-  virtual void Plot(int64_t timestamp_ms);
>-  virtual void RunFor(int64_t time_ms, Packets* in_out);
>+  void Plot(int64_t timestamp_ms) override;
>+  void RunFor(int64_t time_ms, Packets* in_out) override;
> 
>  private:
>   Stats<double> packets_per_second_stats_;
>@@ -263,10 +253,10 @@ class LossFilter : public PacketProcessor {
>  public:
>   LossFilter(PacketProcessorListener* listener, int flow_id);
>   LossFilter(PacketProcessorListener* listener, const FlowIds& flow_ids);
>-  virtual ~LossFilter() {}
>+  ~LossFilter() override {}
> 
>   void SetLoss(float loss_percent);
>-  virtual void RunFor(int64_t time_ms, Packets* in_out);
>+  void RunFor(int64_t time_ms, Packets* in_out) override;
> 
>  private:
>   Random random_;
>@@ -279,10 +269,10 @@ class DelayFilter : public PacketProcessor {
>  public:
>   DelayFilter(PacketProcessorListener* listener, int flow_id);
>   DelayFilter(PacketProcessorListener* listener, const FlowIds& flow_ids);
>-  virtual ~DelayFilter() {}
>+  ~DelayFilter() override {}
> 
>   void SetOneWayDelayMs(int64_t one_way_delay_ms);
>-  virtual void RunFor(int64_t time_ms, Packets* in_out);
>+  void RunFor(int64_t time_ms, Packets* in_out) override;
> 
>  private:
>   int64_t one_way_delay_us_;
>@@ -295,10 +285,10 @@ class JitterFilter : public PacketProcessor {
>  public:
>   JitterFilter(PacketProcessorListener* listener, int flow_id);
>   JitterFilter(PacketProcessorListener* listener, const FlowIds& flow_ids);
>-  virtual ~JitterFilter() {}
>+  ~JitterFilter() override {}
> 
>   void SetMaxJitter(int64_t stddev_jitter_ms);
>-  virtual void RunFor(int64_t time_ms, Packets* in_out);
>+  void RunFor(int64_t time_ms, Packets* in_out) override;
>   void set_reorderdering(bool reordering) { reordering_ = reordering; }
>   int64_t MeanUs();
> 
>@@ -316,10 +306,10 @@ class ReorderFilter : public PacketProcessor {
>  public:
>   ReorderFilter(PacketProcessorListener* listener, int flow_id);
>   ReorderFilter(PacketProcessorListener* listener, const FlowIds& flow_ids);
>-  virtual ~ReorderFilter() {}
>+  ~ReorderFilter() override {}
> 
>   void SetReorder(float reorder_percent);
>-  virtual void RunFor(int64_t time_ms, Packets* in_out);
>+  void RunFor(int64_t time_ms, Packets* in_out) override;
> 
>  private:
>   Random random_;
>@@ -333,14 +323,14 @@ class ChokeFilter : public PacketProcessor {
>  public:
>   ChokeFilter(PacketProcessorListener* listener, int flow_id);
>   ChokeFilter(PacketProcessorListener* listener, const FlowIds& flow_ids);
>-  virtual ~ChokeFilter();
>+  ~ChokeFilter() override;
> 
>   void set_capacity_kbps(uint32_t kbps);
>   void set_max_delay_ms(int64_t max_queueing_delay_ms);
> 
>   uint32_t capacity_kbps();
> 
>-  virtual void RunFor(int64_t time_ms, Packets* in_out);
>+  void RunFor(int64_t time_ms, Packets* in_out) override;
> 
>   Stats<double> GetDelayStats() const;
> 
>@@ -360,14 +350,14 @@ class TraceBasedDeliveryFilter : public PacketProcessor {
>   TraceBasedDeliveryFilter(PacketProcessorListener* listener,
>                            int flow_id,
>                            const char* name);
>-  virtual ~TraceBasedDeliveryFilter();
>+  ~TraceBasedDeliveryFilter() override;
> 
>   // The file should contain nanosecond timestamps corresponding to the time
>   // when the network can accept another packet. The timestamps should be
>   // separated by new lines, e.g., "100000000\n125000000\n321000000\n..."
>   bool Init(const std::string& filename);
>-  virtual void Plot(int64_t timestamp_ms);
>-  virtual void RunFor(int64_t time_ms, Packets* in_out);
>+  void Plot(int64_t timestamp_ms) override;
>+  void RunFor(int64_t time_ms, Packets* in_out) override;
> 
>   void set_max_delay_ms(int64_t max_delay_ms);
>   Stats<double> GetDelayStats() const;
>@@ -401,7 +391,7 @@ class VideoSource {
> 
>   virtual void RunFor(int64_t time_ms, Packets* in_out);
> 
>-  virtual int flow_id() const { return flow_id_; }
>+  int flow_id() const;
>   virtual void SetBitrateBps(int bitrate_bps) {}
>   uint32_t bits_per_second() const { return bits_per_second_; }
>   uint32_t max_payload_size_bytes() const { return kMaxPayloadSizeBytes; }
>@@ -436,7 +426,7 @@ class AdaptiveVideoSource : public VideoSource {
>                       uint32_t kbps,
>                       uint32_t ssrc,
>                       int64_t first_frame_offset_ms);
>-  virtual ~AdaptiveVideoSource() {}
>+  ~AdaptiveVideoSource() override {}
> 
>   void SetBitrateBps(int bitrate_bps) override;
> 
>@@ -452,7 +442,7 @@ class PeriodicKeyFrameSource : public AdaptiveVideoSource {
>                          uint32_t ssrc,
>                          int64_t first_frame_offset_ms,
>                          int key_frame_interval);
>-  virtual ~PeriodicKeyFrameSource() {}
>+  ~PeriodicKeyFrameSource() override {}
> 
>  protected:
>   uint32_t NextFrameSize() override;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework_unittest.cc
>index 2562732a1b3..3e41ac1952e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework_unittest.cc
>@@ -153,8 +153,10 @@ class BweTestFramework_RateCounterFilterTest : public ::testing::Test {
>   virtual ~BweTestFramework_RateCounterFilterTest() {}
> 
>  protected:
>-  void TestRateCounter(int64_t run_for_ms, uint32_t payload_bits,
>-                       uint32_t expected_pps, uint32_t expected_bps) {
>+  void TestRateCounter(int64_t run_for_ms,
>+                       uint32_t payload_bits,
>+                       uint32_t expected_pps,
>+                       uint32_t expected_bps) {
>     Packets packets;
>     RTPHeader header;
>     // "Send" a packet every 10 ms.
>@@ -280,7 +282,8 @@ class BweTestFramework_DelayFilterTest : public ::testing::Test {
>   }
> 
>  protected:
>-  void TestDelayFilter(int64_t run_for_ms, uint32_t in_packets,
>+  void TestDelayFilter(int64_t run_for_ms,
>+                       uint32_t in_packets,
>                        uint32_t out_packets) {
>     Packets packets;
>     for (uint32_t i = 0; i < in_packets; ++i) {
>@@ -301,7 +304,7 @@ class BweTestFramework_DelayFilterTest : public ::testing::Test {
> 
>   void TestDelayFilter(int64_t delay_ms) {
>     filter_.SetOneWayDelayMs(delay_ms);
>-    TestDelayFilter(1, 0, 0);    // No input should yield no output
>+    TestDelayFilter(1, 0, 0);  // No input should yield no output
> 
>     // Single packet
>     TestDelayFilter(0, 1, 1);
>@@ -357,8 +360,8 @@ TEST_F(BweTestFramework_DelayFilterTest, Delay0) {
>   TestDelayFilter(1, 10, 10);  // Expect no delay (delay time is zero)
>   TestDelayFilter(1, 0, 0);    // Check no packets are still in buffer
>   filter_.SetOneWayDelayMs(0);
>-  TestDelayFilter(1, 5, 5);    // Expect no delay (delay time is zero)
>-  TestDelayFilter(1, 0, 0);    // Check no packets are still in buffer
>+  TestDelayFilter(1, 5, 5);  // Expect no delay (delay time is zero)
>+  TestDelayFilter(1, 0, 0);  // Check no packets are still in buffer
> }
> 
> TEST_F(BweTestFramework_DelayFilterTest, Delay1) {
>@@ -580,11 +583,7 @@ TEST(BweTestFramework_ReorderFilterTest, Reorder100) {
> class BweTestFramework_ChokeFilterTest : public ::testing::Test {
>  public:
>   BweTestFramework_ChokeFilterTest()
>-    : now_ms_(0),
>-      sequence_number_(0),
>-      output_packets_(),
>-      send_times_us_() {
>-  }
>+      : now_ms_(0), sequence_number_(0), output_packets_(), send_times_us_() {}
>   virtual ~BweTestFramework_ChokeFilterTest() {
>     for (auto* packet : output_packets_)
>       delete packet;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.cc
>index fb9955ae326..571ed50c0e7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.cc
>@@ -34,16 +34,21 @@ static std::string ToString(uint32_t v) {
>   return ss.str();
> }
> 
>+Logging::ThreadState::ThreadState() = default;
>+Logging::ThreadState::~ThreadState() = default;
>+
> Logging::Context::Context(uint32_t name, int64_t timestamp_ms, bool enabled) {
>   Logging::GetInstance()->PushState(ToString(name), timestamp_ms, enabled);
> }
> 
>-Logging::Context::Context(const std::string& name, int64_t timestamp_ms,
>+Logging::Context::Context(const std::string& name,
>+                          int64_t timestamp_ms,
>                           bool enabled) {
>   Logging::GetInstance()->PushState(name, timestamp_ms, enabled);
> }
> 
>-Logging::Context::Context(const char* name, int64_t timestamp_ms,
>+Logging::Context::Context(const char* name,
>+                          int64_t timestamp_ms,
>                           bool enabled) {
>   Logging::GetInstance()->PushState(name, timestamp_ms, enabled);
> }
>@@ -201,18 +206,16 @@ void Logging::PlotLabel(int figure,
>   }
> }
> 
>-Logging::Logging()
>-    : thread_map_() {
>-}
>+Logging::Logging() : thread_map_() {}
>+
>+Logging::~Logging() = default;
> 
> Logging::State::State() : tag(""), timestamp_ms(0), enabled(true) {}
> 
>-Logging::State::State(const std::string& tag, int64_t timestamp_ms,
>+Logging::State::State(const std::string& tag,
>+                      int64_t timestamp_ms,
>                       bool enabled)
>-    : tag(tag),
>-      timestamp_ms(timestamp_ms),
>-      enabled(enabled) {
>-}
>+    : tag(tag), timestamp_ms(timestamp_ms), enabled(enabled) {}
> 
> void Logging::State::MergePrevious(const State& previous) {
>   if (tag.empty()) {
>@@ -224,7 +227,8 @@ void Logging::State::MergePrevious(const State& previous) {
>   enabled = previous.enabled && enabled;
> }
> 
>-void Logging::PushState(const std::string& append_to_tag, int64_t timestamp_ms,
>+void Logging::PushState(const std::string& append_to_tag,
>+                        int64_t timestamp_ms,
>                         bool enabled) {
>   rtc::CritScope cs(&crit_sect_);
>   State new_state(append_to_tag, timestamp_ms, enabled);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h
>index 325beb7a1ad..20a29cc7937 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/bwe_test_logging.h
>@@ -132,57 +132,56 @@
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/criticalsection.h"
> 
>-#define BWE_TEST_LOGGING_GLOBAL_CONTEXT(name) \
>-    do { \
>-      webrtc::testing::bwe::Logging::GetInstance()->SetGlobalContext(name); \
>-    } while (0)
>+#define BWE_TEST_LOGGING_GLOBAL_CONTEXT(name)                             \
>+  do {                                                                    \
>+    webrtc::testing::bwe::Logging::GetInstance()->SetGlobalContext(name); \
>+  } while (0)
> 
>-#define BWE_TEST_LOGGING_GLOBAL_ENABLE(enabled) \
>-    do { \
>-      webrtc::testing::bwe::Logging::GetInstance()->SetGlobalEnable(enabled); \
>-    } while (0)
>+#define BWE_TEST_LOGGING_GLOBAL_ENABLE(enabled)                             \
>+  do {                                                                      \
>+    webrtc::testing::bwe::Logging::GetInstance()->SetGlobalEnable(enabled); \
>+  } while (0)
> 
>-#define __BWE_TEST_LOGGING_CONTEXT_NAME(ctx, line) ctx ## line
>+#define __BWE_TEST_LOGGING_CONTEXT_NAME(ctx, line) ctx##line
> #define __BWE_TEST_LOGGING_CONTEXT_DECLARE(ctx, line, name, time, enabled) \
>-    webrtc::testing::bwe::Logging::Context \
>-        __BWE_TEST_LOGGING_CONTEXT_NAME(ctx, line)(name, time, enabled)
>+  webrtc::testing::bwe::Logging::Context __BWE_TEST_LOGGING_CONTEXT_NAME(  \
>+      ctx, line)(name, time, enabled)
> 
> #define BWE_TEST_LOGGING_CONTEXT(name) \
>-    __BWE_TEST_LOGGING_CONTEXT_DECLARE(__bwe_log_, __LINE__, name, -1, true)
>-#define BWE_TEST_LOGGING_ENABLE(enabled) \
>-    __BWE_TEST_LOGGING_CONTEXT_DECLARE(__bwe_log_, __LINE__, "", -1, \
>-                                       static_cast<bool>(enabled))
>-#define BWE_TEST_LOGGING_TIME(time) \
>-    __BWE_TEST_LOGGING_CONTEXT_DECLARE(__bwe_log_, __LINE__, "", \
>-                                       static_cast<int64_t>(time), true)
>-
>-#define BWE_TEST_LOGGING_LOG1(name, format, _1) \
>-    do { \
>-      BWE_TEST_LOGGING_CONTEXT(name); \
>-      webrtc::testing::bwe::Logging::GetInstance()->Log(format, _1); \
>-    } while (0)
>-#define BWE_TEST_LOGGING_LOG2(name, format, _1, _2) \
>-    do { \
>-      BWE_TEST_LOGGING_CONTEXT(name); \
>-      webrtc::testing::bwe::Logging::GetInstance()->Log(format, _1, _2); \
>-    } while (0)
>-#define BWE_TEST_LOGGING_LOG3(name, format, _1, _2, _3) \
>-    do { \
>-      BWE_TEST_LOGGING_CONTEXT(name); \
>-      webrtc::testing::bwe::Logging::GetInstance()->Log(format, _1, _2, _3); \
>-    } while (0)
>-#define BWE_TEST_LOGGING_LOG4(name, format, _1, _2, _3, _4) \
>-    do { \
>-      BWE_TEST_LOGGING_CONTEXT(name); \
>-      webrtc::testing::bwe::Logging::GetInstance()->Log(format, _1, _2, _3, \
>-                                                        _4); \
>-    } while (0)
>-#define BWE_TEST_LOGGING_LOG5(name, format, _1, _2, _3, _4, _5) \
>-    do {\
>-      BWE_TEST_LOGGING_CONTEXT(name); \
>-      webrtc::testing::bwe::Logging::GetInstance()->Log(format, _1, _2, _3, \
>-                                                        _4, _5); \
>-    } while (0)
>+  __BWE_TEST_LOGGING_CONTEXT_DECLARE(__bwe_log_, __LINE__, name, -1, true)
>+#define BWE_TEST_LOGGING_ENABLE(enabled)                           \
>+  __BWE_TEST_LOGGING_CONTEXT_DECLARE(__bwe_log_, __LINE__, "", -1, \
>+                                     static_cast<bool>(enabled))
>+#define BWE_TEST_LOGGING_TIME(time)                            \
>+  __BWE_TEST_LOGGING_CONTEXT_DECLARE(__bwe_log_, __LINE__, "", \
>+                                     static_cast<int64_t>(time), true)
>+
>+#define BWE_TEST_LOGGING_LOG1(name, format, _1)                    \
>+  do {                                                             \
>+    BWE_TEST_LOGGING_CONTEXT(name);                                \
>+    webrtc::testing::bwe::Logging::GetInstance()->Log(format, _1); \
>+  } while (0)
>+#define BWE_TEST_LOGGING_LOG2(name, format, _1, _2)                    \
>+  do {                                                                 \
>+    BWE_TEST_LOGGING_CONTEXT(name);                                    \
>+    webrtc::testing::bwe::Logging::GetInstance()->Log(format, _1, _2); \
>+  } while (0)
>+#define BWE_TEST_LOGGING_LOG3(name, format, _1, _2, _3)                    \
>+  do {                                                                     \
>+    BWE_TEST_LOGGING_CONTEXT(name);                                        \
>+    webrtc::testing::bwe::Logging::GetInstance()->Log(format, _1, _2, _3); \
>+  } while (0)
>+#define BWE_TEST_LOGGING_LOG4(name, format, _1, _2, _3, _4)                    \
>+  do {                                                                         \
>+    BWE_TEST_LOGGING_CONTEXT(name);                                            \
>+    webrtc::testing::bwe::Logging::GetInstance()->Log(format, _1, _2, _3, _4); \
>+  } while (0)
>+#define BWE_TEST_LOGGING_LOG5(name, format, _1, _2, _3, _4, _5)               \
>+  do {                                                                        \
>+    BWE_TEST_LOGGING_CONTEXT(name);                                           \
>+    webrtc::testing::bwe::Logging::GetInstance()->Log(format, _1, _2, _3, _4, \
>+                                                      _5);                    \
>+  } while (0)
> 
> #define BWE_TEST_LOGGING_PLOT(figure, name, time, value)                     \
>   do {                                                                       \
>@@ -266,6 +265,7 @@ class Logging {
>     Context(const std::string& name, int64_t timestamp_ms, bool enabled);
>     Context(const char* name, int64_t timestamp_ms, bool enabled);
>     ~Context();
>+
>    private:
>     RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(Context);
>   };
>@@ -281,7 +281,8 @@ class Logging {
>   // Note: Implicit |this| argument counts as the first argument.
>   __attribute__((__format__(__printf__, 2, 3)))
> #endif
>-  void Log(const char format[], ...);
>+  void
>+  Log(const char format[], ...);
>   void Plot(int figure, const std::string& name, double value);
>   void Plot(int figure,
>             const std::string& name,
>@@ -331,13 +332,17 @@ class Logging {
>     bool enabled;
>   };
>   struct ThreadState {
>+    ThreadState();
>+    ~ThreadState();
>     State global_state;
>     std::stack<State> stack;
>   };
>   typedef std::map<uint32_t, ThreadState> ThreadMap;
> 
>   Logging();
>-  void PushState(const std::string& append_to_tag, int64_t timestamp_ms,
>+  ~Logging();
>+  void PushState(const std::string& append_to_tag,
>+                 int64_t timestamp_ms,
>                  bool enabled);
>   void PopState();
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/bwe_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/bwe_unittest.cc
>index 892020c3c4b..fd4d4e789cc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/bwe_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/bwe_unittest.cc
>@@ -33,8 +33,7 @@ class LinkedSetTest : public ::testing::Test {
> };
> 
> TEST_F(LinkedSetTest, EmptySet) {
>-  EXPECT_EQ(linked_set_.OldestSeqNumber(), 0);
>-  EXPECT_EQ(linked_set_.NewestSeqNumber(), 0);
>+  EXPECT_EQ(linked_set_.Range(), 0);
> }
> 
> TEST_F(LinkedSetTest, SinglePacket) {
>@@ -42,8 +41,7 @@ TEST_F(LinkedSetTest, SinglePacket) {
>   // Other parameters don't matter here.
>   linked_set_.Insert(kSeqNumber, 0, 0, 0);
> 
>-  EXPECT_EQ(linked_set_.OldestSeqNumber(), kSeqNumber);
>-  EXPECT_EQ(linked_set_.NewestSeqNumber(), kSeqNumber);
>+  EXPECT_EQ(linked_set_.Range(), 1);
> }
> 
> TEST_F(LinkedSetTest, MultiplePackets) {
>@@ -61,9 +59,8 @@ TEST_F(LinkedSetTest, MultiplePackets) {
>     linked_set_.Insert(static_cast<uint16_t>(i), 0, 0, 0);
>   }
> 
>-  // Packets arriving out of order should not affect the following values:
>-  EXPECT_EQ(linked_set_.OldestSeqNumber(), 0);
>-  EXPECT_EQ(linked_set_.NewestSeqNumber(), kNumberPackets - 1);
>+  // Packets arriving out of order should not affect the following value:
>+  EXPECT_EQ(linked_set_.Range(), kNumberPackets);
> }
> 
> TEST_F(LinkedSetTest, Overflow) {
>@@ -75,32 +72,29 @@ TEST_F(LinkedSetTest, Overflow) {
>     linked_set_.Insert(static_cast<uint16_t>(i), 0, 0, 0);
>   }
> 
>-  // Packets arriving out of order should not affect the following values:
>-  EXPECT_EQ(linked_set_.OldestSeqNumber(),
>-            static_cast<uint16_t>(kFirstSeqNumber));
>-  EXPECT_EQ(linked_set_.NewestSeqNumber(),
>-            static_cast<uint16_t>(kLastSeqNumber));
>+  // Wrapping shouldn't matter
>+  EXPECT_EQ(linked_set_.Range(), kLastSeqNumber - kFirstSeqNumber + 1);
> }
> 
>-class SequenceNumberOlderThanTest : public ::testing::Test {
>- public:
>-  SequenceNumberOlderThanTest() {}
>-  ~SequenceNumberOlderThanTest() {}
>+TEST_F(LinkedSetTest, SameSequenceNumbers) {
>+  // Test correct behavior when
>+  // sequence numbers wrap (after 0xFFFF).
> 
>- protected:
>-  SequenceNumberOlderThan comparator_;
>-};
>+  // Choose step such as step*capacity < 0x8000
>+  // (received packets in a reasonable window)
>+  const int kStep = 0x20;
>+  // Choose iteration such as step*iteration > 0x10000
>+  // (imply wrap)
>+  const int kIterations = 0x1000;
>+
>+  int kSeqNumber = 1;
>+  for (int i = 0; i < kIterations; ++i) {
>+    // Other parameters don't matter here.
>+    linked_set_.Insert(static_cast<uint16_t>(kSeqNumber), 0, 0, 0);
>+    kSeqNumber += kStep;
>+  }
> 
>-TEST_F(SequenceNumberOlderThanTest, Operator) {
>-  // Operator()(x, y) returns true <==> y is newer than x.
>-  EXPECT_TRUE(comparator_.operator()(0x0000, 0x0001));
>-  EXPECT_TRUE(comparator_.operator()(0x0001, 0x1000));
>-  EXPECT_FALSE(comparator_.operator()(0x0001, 0x0000));
>-  EXPECT_FALSE(comparator_.operator()(0x0002, 0x0002));
>-  EXPECT_TRUE(comparator_.operator()(0xFFF6, 0x000A));
>-  EXPECT_FALSE(comparator_.operator()(0x000A, 0xFFF6));
>-  EXPECT_TRUE(comparator_.operator()(0x0000, 0x8000));
>-  EXPECT_FALSE(comparator_.operator()(0x8000, 0x0000));
>+  EXPECT_EQ(linked_set_.Range(), (kSetCapacity - 1) * kStep + 1);
> }
> 
> class LossAccountTest : public ::testing::Test {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/bbr.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/bbr.cc
>index c9ed3aea2cb..1f01c6dec8e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/bbr.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/bbr.cc
>@@ -123,6 +123,34 @@ BbrBweSender::BbrBweSender(BitrateObserver* observer, Clock* clock)
> 
> BbrBweSender::~BbrBweSender() {}
> 
>+BbrBweSender::PacketStats::PacketStats() = default;
>+
>+BbrBweSender::PacketStats::PacketStats(
>+    uint16_t sequence_number_,
>+    int64_t last_sent_packet_send_time_ms_,
>+    int64_t send_time_ms_,
>+    int64_t ack_time_ms_,
>+    int64_t last_acked_packet_ack_time_ms_,
>+    size_t payload_size_bytes_,
>+    size_t data_sent_bytes_,
>+    size_t data_sent_before_last_sent_packet_bytes_,
>+    size_t data_acked_bytes_,
>+    size_t data_acked_before_last_acked_packet_bytes_)
>+    : sequence_number(sequence_number_),
>+      last_sent_packet_send_time_ms(last_sent_packet_send_time_ms_),
>+      send_time_ms(send_time_ms_),
>+      ack_time_ms(ack_time_ms_),
>+      last_acked_packet_ack_time_ms(last_acked_packet_ack_time_ms_),
>+      payload_size_bytes(payload_size_bytes_),
>+      data_sent_bytes(data_sent_bytes_),
>+      data_sent_before_last_sent_packet_bytes(
>+          data_sent_before_last_sent_packet_bytes_),
>+      data_acked_bytes(data_acked_bytes_),
>+      data_acked_before_last_acked_packet_bytes(
>+          data_acked_before_last_acked_packet_bytes_) {}
>+
>+BbrBweSender::PacketStats::PacketStats(const PacketStats&) = default;
>+
> int BbrBweSender::GetFeedbackIntervalMs() const {
>   return kFeedbackIntervalsMs;
> }
>@@ -277,15 +305,15 @@ size_t BbrBweSender::TargetCongestionWindow(float gain) {
>   return target_congestion_window;
> }
> 
>-rtc::Optional<int64_t> BbrBweSender::CalculateBandwidthSample(
>+absl::optional<int64_t> BbrBweSender::CalculateBandwidthSample(
>     size_t data_sent_bytes,
>     int64_t send_time_delta_ms,
>     size_t data_acked_bytes,
>     int64_t ack_time_delta_ms) {
>-  rtc::Optional<int64_t> bandwidth_sample;
>+  absl::optional<int64_t> bandwidth_sample;
>   if (send_time_delta_ms > 0)
>     bandwidth_sample.emplace(data_sent_bytes * 8000 / send_time_delta_ms);
>-  rtc::Optional<int64_t> ack_rate;
>+  absl::optional<int64_t> ack_rate;
>   if (ack_time_delta_ms > 0)
>     ack_rate.emplace(data_acked_bytes * 8000 / ack_time_delta_ms);
>   // If send rate couldn't be calculated automaticaly set |bandwidth_sample| to
>@@ -311,7 +339,7 @@ void BbrBweSender::AddSampleForHighGain() {
>                             data_acked_before_high_gain_started_bytes_;
>   int64_t ack_time_delta_ms = last_packet_ack_time_during_high_gain_ms_ -
>                               first_packet_ack_time_during_high_gain_ms_;
>-  rtc::Optional<int64_t> bandwidth_sample = CalculateBandwidthSample(
>+  absl::optional<int64_t> bandwidth_sample = CalculateBandwidthSample(
>       data_sent_bytes, send_time_delta_ms, data_acked_bytes, ack_time_delta_ms);
>   if (bandwidth_sample)
>     max_bandwidth_filter_->AddBandwidthSample(*bandwidth_sample, round_count_);
>@@ -322,7 +350,7 @@ void BbrBweSender::UpdateBandwidthAndMinRtt(
>     int64_t now_ms,
>     const std::vector<uint16_t>& feedback_vector,
>     int64_t bytes_acked) {
>-  rtc::Optional<int64_t> min_rtt_sample_ms;
>+  absl::optional<int64_t> min_rtt_sample_ms;
>   for (uint16_t f : feedback_vector) {
>     PacketStats packet = packet_stats_[f];
>     size_t data_sent_bytes =
>@@ -333,7 +361,7 @@ void BbrBweSender::UpdateBandwidthAndMinRtt(
>                               packet.data_acked_before_last_acked_packet_bytes;
>     int64_t ack_time_delta_ms =
>         packet.ack_time_ms - packet.last_acked_packet_ack_time_ms;
>-    rtc::Optional<int64_t> bandwidth_sample =
>+    absl::optional<int64_t> bandwidth_sample =
>         CalculateBandwidthSample(data_sent_bytes, send_time_delta_ms,
>                                  data_acked_bytes, ack_time_delta_ms);
>     if (bandwidth_sample)
>@@ -539,9 +567,9 @@ FeedbackPacket* BbrBweReceiver::GetFeedback(int64_t now_ms) {
>   last_feedback_ms_ = now_ms;
>   int64_t corrected_send_time_ms = 0L;
>   if (!received_packets_.empty()) {
>-    PacketIdentifierNode* latest = *(received_packets_.begin());
>+    PacketIdentifierNode& latest = *(received_packets_.begin());
>     corrected_send_time_ms =
>-        latest->send_time_ms + now_ms - latest->arrival_time_ms;
>+        latest.send_time_ms + now_ms - latest.arrival_time_ms;
>   }
>   FeedbackPacket* fb = new BbrBweFeedback(
>       flow_id_, now_ms * 1000, corrected_send_time_ms, packet_feedbacks_);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/bbr.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/bbr.h
>index 966f1e2ff41..f715b96bca6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/bbr.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/bbr.h
>@@ -17,7 +17,7 @@
> #include <memory>
> #include <vector>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> #include "modules/remote_bitrate_estimator/test/bwe.h"
> #include "rtc_base/numerics/sequence_number_util.h"
> #include "rtc_base/random.h"
>@@ -31,7 +31,7 @@ class CongestionWindow;
> class BbrBweSender : public BweSender {
>  public:
>   explicit BbrBweSender(BitrateObserver* observer, Clock* clock);
>-  virtual ~BbrBweSender();
>+  ~BbrBweSender() override;
>   enum Mode {
>     // Startup phase.
>     STARTUP,
>@@ -48,7 +48,7 @@ class BbrBweSender : public BweSender {
>   };
> 
>   struct PacketStats {
>-    PacketStats() {}
>+    PacketStats();
>     PacketStats(uint16_t sequence_number_,
>                 int64_t last_sent_packet_send_time_ms_,
>                 int64_t send_time_ms_,
>@@ -58,19 +58,9 @@ class BbrBweSender : public BweSender {
>                 size_t data_sent_bytes_,
>                 size_t data_sent_before_last_sent_packet_bytes_,
>                 size_t data_acked_bytes_,
>-                size_t data_acked_before_last_acked_packet_bytes_)
>-        : sequence_number(sequence_number_),
>-          last_sent_packet_send_time_ms(last_sent_packet_send_time_ms_),
>-          send_time_ms(send_time_ms_),
>-          ack_time_ms(ack_time_ms_),
>-          last_acked_packet_ack_time_ms(last_acked_packet_ack_time_ms_),
>-          payload_size_bytes(payload_size_bytes_),
>-          data_sent_bytes(data_sent_bytes_),
>-          data_sent_before_last_sent_packet_bytes(
>-              data_sent_before_last_sent_packet_bytes_),
>-          data_acked_bytes(data_acked_bytes_),
>-          data_acked_before_last_acked_packet_bytes(
>-              data_acked_before_last_acked_packet_bytes_) {}
>+                size_t data_acked_before_last_acked_packet_bytes_);
>+    PacketStats(const PacketStats&);
>+
>     // Sequence number of this packet.
>     uint16_t sequence_number;
>     // Send time of the last sent packet at ack time of this packet.
>@@ -129,10 +119,10 @@ class BbrBweSender : public BweSender {
> 
>   // Calculates and returns bandwidth sample as minimum between send rate and
>   // ack rate, returns nothing if sample cannot be calculated.
>-  rtc::Optional<int64_t> CalculateBandwidthSample(size_t data_sent,
>-                                                  int64_t send_time_delta_ms,
>-                                                  size_t data_acked,
>-                                                  int64_t ack_time_delta_ms);
>+  absl::optional<int64_t> CalculateBandwidthSample(size_t data_sent,
>+                                                   int64_t send_time_delta_ms,
>+                                                   size_t data_acked,
>+                                                   int64_t ack_time_delta_ms);
> 
>   // Calculate and add bandwidth sample only for packets' sent during high gain
>   // phase. Motivation of having a seperate bucket for high gain phase is to
>@@ -174,7 +164,7 @@ class BbrBweSender : public BweSender {
> 
>   // First moment of time when data inflight decreased below
>   // kMinimumCongestionWindow in PROBE_RTT mode.
>-  rtc::Optional<int64_t> minimum_congestion_window_start_time_ms_;
>+  absl::optional<int64_t> minimum_congestion_window_start_time_ms_;
> 
>   // First round when data inflight decreased below kMinimumCongestionWindow in
>   // PROBE_RTT mode.
>@@ -188,7 +178,7 @@ class BbrBweSender : public BweSender {
> 
>   // Send time of a packet sent first during high gain phase. Also serves as a
>   // flag, holding value means that we are already in high gain.
>-  rtc::Optional<int64_t> first_packet_send_time_during_high_gain_ms_;
>+  absl::optional<int64_t> first_packet_send_time_during_high_gain_ms_;
> 
>   // Send time of a packet sent last during high gain phase.
>   int64_t last_packet_send_time_during_high_gain_ms_;
>@@ -226,10 +216,11 @@ class BbrBweSender : public BweSender {
> class BbrBweReceiver : public BweReceiver {
>  public:
>   explicit BbrBweReceiver(int flow_id);
>-  virtual ~BbrBweReceiver();
>+  ~BbrBweReceiver() override;
>   void ReceivePacket(int64_t arrival_time_ms,
>                      const MediaPacket& media_packet) override;
>   FeedbackPacket* GetFeedback(int64_t now_ms) override;
>+
>  private:
>   SimulatedClock clock_;
>   std::vector<uint16_t> packet_feedbacks_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/congestion_window.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/congestion_window.cc
>index f3f66a0e40e..ac22cd07c98 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/congestion_window.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/congestion_window.cc
>@@ -33,7 +33,7 @@ CongestionWindow::~CongestionWindow() {}
> 
> int CongestionWindow::GetCongestionWindow(BbrBweSender::Mode mode,
>                                           int64_t bandwidth_estimate_bps,
>-                                          rtc::Optional<int64_t> min_rtt_ms,
>+                                          absl::optional<int64_t> min_rtt_ms,
>                                           float gain) {
>   return GetTargetCongestionWindow(bandwidth_estimate_bps, min_rtt_ms, gain);
> }
>@@ -49,7 +49,7 @@ void CongestionWindow::AckReceived(size_t received_packet_size_bytes) {
> 
> int CongestionWindow::GetTargetCongestionWindow(
>     int64_t bandwidth_estimate_bps,
>-    rtc::Optional<int64_t> min_rtt_ms,
>+    absl::optional<int64_t> min_rtt_ms,
>     float gain) {
>   // If we have no rtt sample yet, return the starting congestion window size.
>   if (!min_rtt_ms)
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/congestion_window.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/congestion_window.h
>index f344adb739f..a0952af1770 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/congestion_window.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/congestion_window.h
>@@ -14,7 +14,7 @@
> 
> #include "modules/remote_bitrate_estimator/test/estimators/bbr.h"
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> 
> namespace webrtc {
> namespace testing {
>@@ -25,10 +25,10 @@ class CongestionWindow {
>   ~CongestionWindow();
>   int GetCongestionWindow(BbrBweSender::Mode mode,
>                           int64_t bandwidth_estimate,
>-                          rtc::Optional<int64_t> min_rtt,
>+                          absl::optional<int64_t> min_rtt,
>                           float gain);
>   int GetTargetCongestionWindow(int64_t bandwidth_estimate,
>-                                rtc::Optional<int64_t> min_rtt,
>+                                absl::optional<int64_t> min_rtt,
>                                 float gain);
>   // Packet sent from sender, meaning it is inflight until we receive it and we
>   // should add packet's size to data_inflight.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/max_bandwidth_filter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/max_bandwidth_filter.h
>index 011934d9cf8..489b47efe57 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/max_bandwidth_filter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/max_bandwidth_filter.h
>@@ -20,7 +20,6 @@
> #include <vector>
> 
> #include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
>-#include "modules/remote_bitrate_estimator/include/send_time_history.h"
> #include "modules/remote_bitrate_estimator/test/bwe.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/min_rtt_filter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/min_rtt_filter.cc
>new file mode 100644
>index 00000000000..8f699ce1693
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/min_rtt_filter.cc
>@@ -0,0 +1,23 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ *
>+ */
>+
>+#include "modules/remote_bitrate_estimator/test/estimators/min_rtt_filter.h"
>+
>+namespace webrtc {
>+namespace testing {
>+namespace bwe {
>+
>+MinRttFilter::MinRttFilter() = default;
>+MinRttFilter::~MinRttFilter() = default;
>+
>+}  // namespace bwe
>+}  // namespace testing
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/min_rtt_filter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/min_rtt_filter.h
>index ed8ec8601e1..300ba34897e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/min_rtt_filter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/min_rtt_filter.h
>@@ -16,7 +16,7 @@
> #include <limits>
> #include <list>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> 
> namespace webrtc {
> namespace testing {
>@@ -33,10 +33,10 @@ class MinRttFilter {
>   // This class implements a simple filter to ensure that PROBE_RTT is only
>   // entered when RTTs start to increase, instead of fixed 10 second window as
>   // in orginal BBR design doc, to avoid unnecessary freezes in stream.
>-  MinRttFilter() {}
>-  ~MinRttFilter() {}
>+  MinRttFilter();
>+  ~MinRttFilter();
> 
>-  rtc::Optional<int64_t> min_rtt_ms() { return min_rtt_ms_; }
>+  absl::optional<int64_t> min_rtt_ms() { return min_rtt_ms_; }
>   void AddRttSample(int64_t rtt_ms, int64_t now_ms) {
>     if (!min_rtt_ms_ || rtt_ms <= *min_rtt_ms_ || MinRttExpired(now_ms)) {
>       min_rtt_ms_.emplace(rtt_ms);
>@@ -63,7 +63,7 @@ class MinRttFilter {
>   }
> 
>  private:
>-  rtc::Optional<int64_t> min_rtt_ms_;
>+  absl::optional<int64_t> min_rtt_ms_;
>   std::list<int64_t> rtt_samples_;
> };
> }  // namespace bwe
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/nada.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/nada.cc
>index d3d0baea0e5..0d81878c7c0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/nada.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/nada.cc
>@@ -46,11 +46,9 @@ NadaBweReceiver::NadaBweReceiver(int flow_id)
>       last_congestion_signal_ms_(0),
>       last_delays_index_(0),
>       exp_smoothed_delay_ms_(-1),
>-      est_queuing_delay_signal_ms_(0) {
>-}
>+      est_queuing_delay_signal_ms_(0) {}
> 
>-NadaBweReceiver::~NadaBweReceiver() {
>-}
>+NadaBweReceiver::~NadaBweReceiver() {}
> 
> void NadaBweReceiver::ReceivePacket(int64_t arrival_time_ms,
>                                     const MediaPacket& media_packet) {
>@@ -119,9 +117,9 @@ FeedbackPacket* NadaBweReceiver::GetFeedback(int64_t now_ms) {
>   int64_t corrected_send_time_ms = 0L;
> 
>   if (!received_packets_.empty()) {
>-    PacketIdentifierNode* latest = *(received_packets_.begin());
>+    PacketIdentifierNode& latest = *(received_packets_.begin());
>     corrected_send_time_ms =
>-        latest->send_time_ms + now_ms - latest->arrival_time_ms;
>+        latest.send_time_ms + now_ms - latest.arrival_time_ms;
>   }
> 
>   // Sends a tuple containing latest values of <d_hat_n, d_tilde_n, x_n, x'_n,
>@@ -162,8 +160,7 @@ NadaBweSender::NadaBweSender(int kbps, BitrateObserver* observer, Clock* clock)
>     : BweSender(kbps),  // Referred as "Reference Rate" = R_n.,
>       clock_(clock),
>       observer_(observer),
>-      original_operating_mode_(true) {
>-}
>+      original_operating_mode_(true) {}
> 
> NadaBweSender::NadaBweSender(BitrateObserver* observer, Clock* clock)
>     : BweSender(kMinNadaBitrateKbps),  // Referred as "Reference Rate" = R_n.
>@@ -171,8 +168,7 @@ NadaBweSender::NadaBweSender(BitrateObserver* observer, Clock* clock)
>       observer_(observer),
>       original_operating_mode_(true) {}
> 
>-NadaBweSender::~NadaBweSender() {
>-}
>+NadaBweSender::~NadaBweSender() {}
> 
> int NadaBweSender::GetFeedbackIntervalMs() const {
>   return 100;
>@@ -243,8 +239,7 @@ int64_t NadaBweSender::TimeUntilNextProcess() {
>   return 100;
> }
> 
>-void NadaBweSender::Process() {
>-}
>+void NadaBweSender::Process() {}
> 
> void NadaBweSender::AcceleratedRampUp(const NadaFeedback& fb) {
>   const int kMaxRampUpQueuingDelayMs = 50;  // Referred as T_th.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/nada.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/nada.h
>index 590175c7842..7faedc6bef4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/nada.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/nada.h
>@@ -7,7 +7,7 @@
>  *  in the file PATENTS.  All contributing project authors may
>  *  be found in the AUTHORS file in the root of the source tree.
>  *
>-*/
>+ */
> 
> //  Implementation of Network-Assisted Dynamic Adaptation's (NADA's) proposal
> //  Version according to Draft Document (mentioned in references)
>@@ -24,7 +24,6 @@
> #include "modules/include/module_common_types.h"
> #include "modules/remote_bitrate_estimator/test/bwe.h"
> #include "rtc_base/constructormagic.h"
>-#include "voice_engine/channel.h"
> 
> namespace webrtc {
> 
>@@ -36,7 +35,7 @@ namespace bwe {
> class NadaBweReceiver : public BweReceiver {
>  public:
>   explicit NadaBweReceiver(int flow_id);
>-  virtual ~NadaBweReceiver();
>+  ~NadaBweReceiver() override;
> 
>   void ReceivePacket(int64_t arrival_time_ms,
>                      const MediaPacket& media_packet) override;
>@@ -68,7 +67,7 @@ class NadaBweSender : public BweSender {
> 
>   NadaBweSender(int kbps, BitrateObserver* observer, Clock* clock);
>   NadaBweSender(BitrateObserver* observer, Clock* clock);
>-  virtual ~NadaBweSender();
>+  ~NadaBweSender() override;
> 
>   int GetFeedbackIntervalMs() const override;
>   // Updates the min_feedback_delay_ms_ and the min_round_trip_time_ms_.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/nada_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/nada_unittest.cc
>index eac9e9349a1..2b48528a10b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/nada_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/nada_unittest.cc
>@@ -426,8 +426,8 @@ TEST_F(NadaReceiverSideTest, FeedbackWarpedDelay) {
>   // Raw delays are = [50 250 450 650 850 1050 1250 1450] ms.
>   // Baseline delay will be 50 ms.
>   // Delay signals should be: [0 200 400 600 800 1000 1200 1400] ms.
>-  const int64_t kMedianFilteredDelaysMs[] = {
>-      0, 100, 200, 300, 400, 600, 800, 1000};
>+  const int64_t kMedianFilteredDelaysMs[] = {0,   100, 200, 300,
>+                                             400, 600, 800, 1000};
>   const int kNumPackets = arraysize(kMedianFilteredDelaysMs);
>   const float kAlpha = 0.1f;  // Used for exponential smoothing.
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/remb.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/remb.cc
>index ae9638923ad..34e60a6fa5e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/remb.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/remb.cc
>@@ -16,8 +16,8 @@
> #include "modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.h"
> #include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
> #include "modules/rtp_rtcp/include/receive_statistics.h"
>+#include "rtc_base/system/unused.h"
> #include "test/gtest.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> namespace testing {
>@@ -37,8 +37,7 @@ RembBweSender::RembBweSender(int kbps, BitrateObserver* observer, Clock* clock)
>                                         1000 * kMaxBitrateKbps);
> }
> 
>-RembBweSender::~RembBweSender() {
>-}
>+RembBweSender::~RembBweSender() {}
> 
> void RembBweSender::GiveFeedback(const FeedbackPacket& feedback) {
>   const RembFeedback& remb_feedback =
>@@ -80,8 +79,7 @@ RembReceiver::RembReceiver(int flow_id, bool plot)
>   estimator_->SetMinBitrate(kRemoteBitrateEstimatorMinBitrateBps);
> }
> 
>-RembReceiver::~RembReceiver() {
>-}
>+RembReceiver::~RembReceiver() {}
> 
> void RembReceiver::ReceivePacket(int64_t arrival_time_ms,
>                                  const MediaPacket& media_packet) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/remb.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/remb.h
>index f90b926976a..34c57b8b9cf 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/remb.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/remb.h
>@@ -56,7 +56,7 @@ class RembReceiver : public BweReceiver, public RemoteBitrateObserver {
>   static const uint32_t kRemoteBitrateEstimatorMinBitrateBps = 30000;
> 
>   RembReceiver(int flow_id, bool plot);
>-  virtual ~RembReceiver();
>+  ~RembReceiver() override;
> 
>   void ReceivePacket(int64_t arrival_time_ms,
>                      const MediaPacket& media_packet) override;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/send_side.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/send_side.cc
>index 6e03eee942d..a4444375805 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/send_side.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/send_side.cc
>@@ -12,10 +12,10 @@
> 
> #include <algorithm>
> 
>-#include "modules/congestion_controller/delay_based_bwe.h"
>+#include "absl/memory/memory.h"
>+#include "modules/congestion_controller/goog_cc/delay_based_bwe.h"
> #include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/ptr_util.h"
> 
> namespace webrtc {
> namespace testing {
>@@ -31,8 +31,8 @@ SendSideBweSender::SendSideBweSender(int kbps,
>                                                      observer,
>                                                      &event_log_)),
>       acknowledged_bitrate_estimator_(
>-          rtc::MakeUnique<AcknowledgedBitrateEstimator>()),
>-      bwe_(new DelayBasedBwe(nullptr, clock)),
>+          absl::make_unique<AcknowledgedBitrateEstimator>()),
>+      bwe_(new DelayBasedBwe(nullptr)),
>       feedback_observer_(bitrate_controller_.get()),
>       clock_(clock),
>       send_time_history_(clock_, 10000),
>@@ -72,7 +72,7 @@ void SendSideBweSender::GiveFeedback(const FeedbackPacket& feedback) {
> 
>   int64_t rtt_ms =
>       clock_->TimeInMilliseconds() - feedback.latest_send_time_ms();
>-  bwe_->OnRttUpdate(rtt_ms, rtt_ms);
>+  bwe_->OnRttUpdate(rtt_ms);
>   BWE_TEST_LOGGING_PLOT(1, "RTT", clock_->TimeInMilliseconds(), rtt_ms);
> 
>   std::sort(packet_feedback_vector.begin(), packet_feedback_vector.end(),
>@@ -80,7 +80,8 @@ void SendSideBweSender::GiveFeedback(const FeedbackPacket& feedback) {
>   acknowledged_bitrate_estimator_->IncomingPacketFeedbackVector(
>       packet_feedback_vector);
>   DelayBasedBwe::Result result = bwe_->IncomingPacketFeedbackVector(
>-      packet_feedback_vector, acknowledged_bitrate_estimator_->bitrate_bps());
>+      packet_feedback_vector, acknowledged_bitrate_estimator_->bitrate_bps(),
>+      clock_->TimeInMilliseconds());
>   if (result.updated)
>     bitrate_controller_->OnDelayBasedBweResult(result);
> 
>@@ -143,11 +144,9 @@ void SendSideBweSender::Process() {
> }
> 
> SendSideBweReceiver::SendSideBweReceiver(int flow_id)
>-    : BweReceiver(flow_id), last_feedback_ms_(0) {
>-}
>+    : BweReceiver(flow_id), last_feedback_ms_(0) {}
> 
>-SendSideBweReceiver::~SendSideBweReceiver() {
>-}
>+SendSideBweReceiver::~SendSideBweReceiver() {}
> 
> void SendSideBweReceiver::ReceivePacket(int64_t arrival_time_ms,
>                                         const MediaPacket& media_packet) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/send_side.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/send_side.h
>index 6816223cf3f..6e939c1f035 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/send_side.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/send_side.h
>@@ -15,8 +15,8 @@
> #include <vector>
> 
> #include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
>-#include "modules/congestion_controller/acknowledged_bitrate_estimator.h"
>-#include "modules/remote_bitrate_estimator/include/send_time_history.h"
>+#include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h"
>+#include "modules/congestion_controller/rtp/send_time_history.h"
> #include "modules/remote_bitrate_estimator/test/bwe.h"
> 
> namespace webrtc {
>@@ -58,7 +58,7 @@ class SendSideBweSender : public BweSender, public RemoteBitrateObserver {
> class SendSideBweReceiver : public BweReceiver {
>  public:
>   explicit SendSideBweReceiver(int flow_id);
>-  virtual ~SendSideBweReceiver();
>+  ~SendSideBweReceiver() override;
> 
>   void ReceivePacket(int64_t arrival_time_ms,
>                      const MediaPacket& media_packet) override;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/tcp.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/tcp.cc
>index 58998379854..759c2f62a7a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/tcp.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/tcp.cc
>@@ -22,13 +22,9 @@ namespace testing {
> namespace bwe {
> 
> TcpBweReceiver::TcpBweReceiver(int flow_id)
>-    : BweReceiver(flow_id),
>-      last_feedback_ms_(0),
>-      latest_owd_ms_(0) {
>-}
>+    : BweReceiver(flow_id), last_feedback_ms_(0), latest_owd_ms_(0) {}
> 
>-TcpBweReceiver::~TcpBweReceiver() {
>-}
>+TcpBweReceiver::~TcpBweReceiver() {}
> 
> void TcpBweReceiver::ReceivePacket(int64_t arrival_time_ms,
>                                    const MediaPacket& media_packet) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/tcp.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/tcp.h
>index 472aaf823a3..aa4584c85a7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/tcp.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/estimators/tcp.h
>@@ -21,7 +21,7 @@ namespace bwe {
> class TcpBweReceiver : public BweReceiver {
>  public:
>   explicit TcpBweReceiver(int flow_id);
>-  virtual ~TcpBweReceiver();
>+  ~TcpBweReceiver() override;
> 
>   void ReceivePacket(int64_t arrival_time_ms,
>                      const MediaPacket& media_packet) override;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/metric_recorder.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/metric_recorder.cc
>index 612ad4ab467..fc2296e2795 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/metric_recorder.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/metric_recorder.cc
>@@ -15,7 +15,7 @@
> #include <algorithm>
> 
> #include "modules/remote_bitrate_estimator/test/packet_sender.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include "rtc_base/system/unused.h"
> 
> namespace webrtc {
> namespace testing {
>@@ -27,13 +27,14 @@ template <typename T>
> double NormLp(T sum, size_t size, double p) {
>   return pow(sum / size, 1.0 / p);
> }
>-}
>+}  // namespace
> 
> const double kP = 1.0;  // Used for Norm Lp.
> 
> LinkShare::LinkShare(ChokeFilter* choke_filter)
>-    : choke_filter_(choke_filter), running_flows_(choke_filter->flow_ids()) {
>-}
>+    : choke_filter_(choke_filter), running_flows_(choke_filter->flow_ids()) {}
>+
>+LinkShare::~LinkShare() = default;
> 
> void LinkShare::PauseFlow(int flow_id) {
>   running_flows_.erase(flow_id);
>@@ -79,6 +80,8 @@ MetricRecorder::MetricRecorder(const std::string algorithm_name,
>     packet_sender->set_metric_recorder(this);
> }
> 
>+MetricRecorder::~MetricRecorder() = default;
>+
> void MetricRecorder::SetPlotInformation(
>     const std::vector<std::string>& prefixes,
>     bool plot_delay,
>@@ -288,10 +291,10 @@ void MetricRecorder::PlotThroughputHistogram(
>       average_bitrate_kbps + pos_error + extra_error, "estimate_error",
>       optimal_bitrate_per_flow_kbps, optimum_title, flow_id_);
> 
>-  BWE_TEST_LOGGING_LOG1("RESULTS >>> " + bwe_name + " Channel utilization : ",
>-                        "%lf %%",
>-                        100.0 * static_cast<double>(average_bitrate_kbps) /
>-                            optimal_bitrate_per_flow_kbps);
>+  BWE_TEST_LOGGING_LOG1(
>+      "RESULTS >>> " + bwe_name + " Channel utilization : ", "%lf %%",
>+      100.0 * static_cast<double>(average_bitrate_kbps) /
>+          optimal_bitrate_per_flow_kbps);
> 
>   RTC_UNUSED(pos_error);
>   RTC_UNUSED(neg_error);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/metric_recorder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/metric_recorder.h
>index 4df504a24ad..d0ba4e31634 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/metric_recorder.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/metric_recorder.h
>@@ -28,6 +28,7 @@ class PacketSender;
> class LinkShare {
>  public:
>   explicit LinkShare(ChokeFilter* choke_filter);
>+  ~LinkShare();
> 
>   void PauseFlow(int flow_id);   // Increases available capacity per flow.
>   void ResumeFlow(int flow_id);  // Decreases available capacity per flow.
>@@ -67,6 +68,7 @@ class MetricRecorder {
>                  int flow_id,
>                  PacketSender* packet_sender,
>                  LinkShare* link_share);
>+  ~MetricRecorder();
> 
>   void SetPlotInformation(const std::vector<std::string>& prefixes,
>                           bool plot_delay,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/packet.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/packet.cc
>new file mode 100644
>index 00000000000..910f63839c0
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/packet.cc
>@@ -0,0 +1,97 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/remote_bitrate_estimator/test/packet.h"
>+
>+namespace webrtc {
>+namespace testing {
>+namespace bwe {
>+
>+Packet::Packet()
>+    : flow_id_(0),
>+      creation_time_us_(-1),
>+      send_time_us_(-1),
>+      sender_timestamp_us_(-1),
>+      payload_size_(0) {}
>+
>+Packet::Packet(int flow_id, int64_t send_time_us, size_t payload_size)
>+    : flow_id_(flow_id),
>+      creation_time_us_(send_time_us),
>+      send_time_us_(send_time_us),
>+      sender_timestamp_us_(send_time_us),
>+      payload_size_(payload_size) {}
>+
>+Packet::~Packet() {}
>+
>+bool Packet::operator<(const Packet& rhs) const {
>+  return send_time_us_ < rhs.send_time_us_;
>+}
>+
>+void Packet::set_send_time_us(int64_t send_time_us) {
>+  assert(send_time_us >= 0);
>+  send_time_us_ = send_time_us;
>+}
>+
>+int Packet::flow_id() const {
>+  return flow_id_;
>+}
>+
>+int64_t Packet::send_time_us() const {
>+  return send_time_us_;
>+}
>+
>+int64_t Packet::sender_timestamp_us() const {
>+  return sender_timestamp_us_;
>+}
>+
>+size_t Packet::payload_size() const {
>+  return payload_size_;
>+}
>+
>+void Packet::set_sender_timestamp_us(int64_t sender_timestamp_us) {
>+  sender_timestamp_us_ = sender_timestamp_us;
>+}
>+
>+int64_t Packet::creation_time_ms() const {
>+  return (creation_time_us_ + 500) / 1000;
>+}
>+
>+int64_t Packet::sender_timestamp_ms() const {
>+  return (sender_timestamp_us_ + 500) / 1000;
>+}
>+
>+int64_t Packet::send_time_ms() const {
>+  return (send_time_us_ + 500) / 1000;
>+}
>+
>+Packet::Type MediaPacket::GetPacketType() const {
>+  return kMedia;
>+}
>+
>+Packet::Type FeedbackPacket::GetPacketType() const {
>+  return kFeedback;
>+}
>+
>+BbrBweFeedback::~BbrBweFeedback() = default;
>+
>+SendSideBweFeedback::~SendSideBweFeedback() = default;
>+
>+TcpFeedback::TcpFeedback(int flow_id,
>+                         int64_t send_time_us,
>+                         int64_t latest_send_time_ms,
>+                         const std::vector<uint16_t>& acked_packets)
>+    : FeedbackPacket(flow_id, send_time_us, latest_send_time_ms),
>+      acked_packets_(acked_packets) {}
>+
>+TcpFeedback::~TcpFeedback() = default;
>+
>+}  // namespace bwe
>+}  // namespace testing
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/packet.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/packet.h
>index 0a0e0ca7b55..5e45eb52d6f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/packet.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/packet.h
>@@ -17,8 +17,8 @@
> #include <vector>
> 
> #include "common_types.h"  // NOLINT(build/include)
>-#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> #include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
>+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> 
> namespace webrtc {
> namespace testing {
>@@ -34,22 +34,16 @@ class Packet {
> 
>   virtual bool operator<(const Packet& rhs) const;
> 
>-  virtual int flow_id() const { return flow_id_; }
>+  virtual int flow_id() const;
>   virtual void set_send_time_us(int64_t send_time_us);
>-  virtual int64_t send_time_us() const { return send_time_us_; }
>-  virtual int64_t sender_timestamp_us() const { return sender_timestamp_us_; }
>-  virtual size_t payload_size() const { return payload_size_; }
>+  virtual int64_t send_time_us() const;
>+  virtual int64_t sender_timestamp_us() const;
>+  virtual size_t payload_size() const;
>   virtual Packet::Type GetPacketType() const = 0;
>-  virtual void set_sender_timestamp_us(int64_t sender_timestamp_us) {
>-    sender_timestamp_us_ = sender_timestamp_us;
>-  }
>-  virtual int64_t creation_time_ms() const {
>-    return (creation_time_us_ + 500) / 1000;
>-  }
>-  virtual int64_t sender_timestamp_ms() const {
>-    return (sender_timestamp_us_ + 500) / 1000;
>-  }
>-  virtual int64_t send_time_ms() const { return (send_time_us_ + 500) / 1000; }
>+  virtual void set_sender_timestamp_us(int64_t sender_timestamp_us);
>+  virtual int64_t creation_time_ms() const;
>+  virtual int64_t sender_timestamp_ms() const;
>+  virtual int64_t send_time_ms() const;
> 
>  protected:
>   int flow_id_;
>@@ -72,7 +66,7 @@ class MediaPacket : public Packet {
>               const RTPHeader& header);
>   MediaPacket(int64_t send_time_us, uint16_t sequence_number);
> 
>-  virtual ~MediaPacket() {}
>+  ~MediaPacket() override {}
> 
>   int64_t GetAbsSendTimeInMs() const {
>     int64_t timestamp = header_.extension.absoluteSendTime
>@@ -81,7 +75,7 @@ class MediaPacket : public Packet {
>   }
>   void SetAbsSendTimeMs(int64_t abs_send_time_ms);
>   const RTPHeader& header() const { return header_; }
>-  virtual Packet::Type GetPacketType() const { return kMedia; }
>+  Packet::Type GetPacketType() const override;
>   uint16_t sequence_number() const { return header_.sequenceNumber; }
> 
>  private:
>@@ -100,9 +94,9 @@ class FeedbackPacket : public Packet {
>                  int64_t latest_send_time_ms)
>       : Packet(flow_id, this_send_time_us, 0),
>         latest_send_time_ms_(latest_send_time_ms) {}
>-  virtual ~FeedbackPacket() {}
>+  ~FeedbackPacket() override {}
> 
>-  virtual Packet::Type GetPacketType() const { return kFeedback; }
>+  Packet::Type GetPacketType() const override;
>   int64_t latest_send_time_ms() const { return latest_send_time_ms_; }
> 
>  private:
>@@ -115,7 +109,7 @@ class BbrBweFeedback : public FeedbackPacket {
>                  int64_t send_time_us,
>                  int64_t latest_send_time_ms,
>                  const std::vector<uint16_t>& packet_feedback_vector);
>-  virtual ~BbrBweFeedback() {}
>+  ~BbrBweFeedback() override;
> 
>   const std::vector<uint16_t>& packet_feedback_vector() const {
>     return packet_feedback_vector_;
>@@ -132,7 +126,7 @@ class RembFeedback : public FeedbackPacket {
>                int64_t latest_send_time_ms,
>                uint32_t estimated_bps,
>                RTCPReportBlock report_block);
>-  virtual ~RembFeedback() {}
>+  ~RembFeedback() override {}
> 
>   uint32_t estimated_bps() const { return estimated_bps_; }
>   RTCPReportBlock report_block() const { return report_block_; }
>@@ -150,7 +144,7 @@ class SendSideBweFeedback : public FeedbackPacket {
>       int64_t send_time_us,
>       int64_t latest_send_time_ms,
>       const std::vector<PacketFeedback>& packet_feedback_vector);
>-  virtual ~SendSideBweFeedback() {}
>+  ~SendSideBweFeedback() override;
> 
>   const std::vector<PacketFeedback>& packet_feedback_vector() const {
>     return packet_feedback_vector_;
>@@ -176,7 +170,7 @@ class NadaFeedback : public FeedbackPacket {
>         congestion_signal_(congestion_signal),
>         derivative_(derivative),
>         receiving_rate_(receiving_rate) {}
>-  virtual ~NadaFeedback() {}
>+  ~NadaFeedback() override {}
> 
>   int64_t exp_smoothed_delay_ms() const { return exp_smoothed_delay_ms_; }
>   int64_t est_queuing_delay_signal_ms() const {
>@@ -199,10 +193,8 @@ class TcpFeedback : public FeedbackPacket {
>   TcpFeedback(int flow_id,
>               int64_t send_time_us,
>               int64_t latest_send_time_ms,
>-              const std::vector<uint16_t>& acked_packets)
>-      : FeedbackPacket(flow_id, send_time_us, latest_send_time_ms),
>-        acked_packets_(acked_packets) {}
>-  virtual ~TcpFeedback() {}
>+              const std::vector<uint16_t>& acked_packets);
>+  ~TcpFeedback() override;
> 
>   const std::vector<uint16_t>& acked_packets() const { return acked_packets_; }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/packet_receiver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/packet_receiver.cc
>index f4a42ed1827..d68ea9a37b8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/packet_receiver.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/packet_receiver.cc
>@@ -68,11 +68,9 @@ PacketReceiver::PacketReceiver(PacketProcessorListener* listener,
>                      bwe_type,
>                      plot_delay,
>                      plot_bwe,
>-                     nullptr) {
>-}
>+                     nullptr) {}
> 
>-PacketReceiver::~PacketReceiver() {
>-}
>+PacketReceiver::~PacketReceiver() {}
> 
> void PacketReceiver::RunFor(int64_t time_ms, Packets* in_out) {
>   Packets feedback;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/packet_receiver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/packet_receiver.h
>index a4b26daf1ed..af9dadb9937 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/packet_receiver.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/packet_receiver.h
>@@ -36,7 +36,7 @@ class PacketReceiver : public PacketProcessor {
>                  bool plot_delay,
>                  bool plot_bwe,
>                  MetricRecorder* metric_recorder);
>-  ~PacketReceiver();
>+  ~PacketReceiver() override;
> 
>   // Implements PacketProcessor.
>   void RunFor(int64_t time_ms, Packets* in_out) override;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc
>index 08db1a3f551..06e5505c18c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/packet_sender.cc
>@@ -24,6 +24,13 @@
> namespace webrtc {
> namespace testing {
> namespace bwe {
>+namespace {
>+const float kPaceMultiplier = 2.5f;
>+}
>+
>+uint32_t PacketSender::TargetBitrateKbps() {
>+  return 0;
>+}
> 
> void PacketSender::Pause() {
>   running_ = false;
>@@ -83,8 +90,7 @@ VideoSender::VideoSender(PacketProcessorListener* listener,
>   modules_.push_back(bwe_.get());
> }
> 
>-VideoSender::~VideoSender() {
>-}
>+VideoSender::~VideoSender() {}
> 
> void VideoSender::Pause() {
>   previous_sending_bitrate_ = TargetBitrateKbps();
>@@ -102,6 +108,10 @@ void VideoSender::RunFor(int64_t time_ms, Packets* in_out) {
>   ProcessFeedbackAndGeneratePackets(time_ms, &feedbacks, in_out);
> }
> 
>+VideoSource* VideoSender::source() const {
>+  return source_;
>+}
>+
> void VideoSender::ProcessFeedbackAndGeneratePackets(
>     int64_t time_ms,
>     std::list<FeedbackPacket*>* feedbacks,
>@@ -164,7 +174,7 @@ PacedVideoSender::PacedVideoSender(PacketProcessorListener* listener,
>               ? static_cast<Pacer*>(new BbrPacedSender(&clock_, this, nullptr))
>               : static_cast<Pacer*>(new PacedSender(&clock_, this, nullptr))) {
>   modules_.push_back(pacer_.get());
>-  pacer_->SetEstimatedBitrate(source->bits_per_second());
>+  pacer_->SetPacingRates(source->bits_per_second() * kPaceMultiplier, 0);
> }
> 
> PacedVideoSender::~PacedVideoSender() {
>@@ -312,7 +322,7 @@ void PacedVideoSender::OnNetworkChanged(uint32_t target_bitrate_bps,
>                                         uint8_t fraction_lost,
>                                         int64_t rtt) {
>   VideoSender::OnNetworkChanged(target_bitrate_bps, fraction_lost, rtt);
>-  pacer_->SetEstimatedBitrate(target_bitrate_bps);
>+  pacer_->SetPacingRates(target_bitrate_bps * kPaceMultiplier, 0);
> }
> 
> void PacedVideoSender::OnNetworkChanged(uint32_t bitrate_for_encoder_bps,
>@@ -325,6 +335,10 @@ void PacedVideoSender::OnNetworkChanged(uint32_t bitrate_for_encoder_bps,
>       bitrate_for_pacer_bps, in_probe_rtt, congestion_window);
> }
> 
>+size_t PacedVideoSender::pacer_queue_size_in_bytes() {
>+  return pacer_queue_size_in_bytes_;
>+}
>+
> void PacedVideoSender::OnBytesAcked(size_t bytes) {
>   pacer_->OnBytesAcked(bytes);
> }
>@@ -335,8 +349,7 @@ const int kPacketSizeBytes = 1200;
> TcpSender::TcpSender(PacketProcessorListener* listener,
>                      int flow_id,
>                      int64_t offset_ms)
>-    : TcpSender(listener, flow_id, offset_ms, kNoLimit) {
>-}
>+    : TcpSender(listener, flow_id, offset_ms, kNoLimit) {}
> 
> TcpSender::TcpSender(PacketProcessorListener* listener,
>                      int flow_id,
>@@ -355,8 +368,9 @@ TcpSender::TcpSender(PacketProcessorListener* listener,
>       send_limit_bytes_(send_limit_bytes),
>       last_generated_packets_ms_(0),
>       num_recent_sent_packets_(0),
>-      bitrate_kbps_(0) {
>-}
>+      bitrate_kbps_(0) {}
>+
>+TcpSender::~TcpSender() = default;
> 
> void TcpSender::RunFor(int64_t time_ms, Packets* in_out) {
>   if (clock_.TimeInMilliseconds() + time_ms < offset_ms_) {
>@@ -398,6 +412,10 @@ void TcpSender::RunFor(int64_t time_ms, Packets* in_out) {
>   SendPackets(in_out);
> }
> 
>+int TcpSender::GetFeedbackIntervalMs() const {
>+  return 10;
>+}
>+
> void TcpSender::SendPackets(Packets* in_out) {
>   int cwnd = ceil(cwnd_);
>   int packets_to_send = std::max(cwnd - static_cast<int>(in_flight_.size()), 0);
>@@ -474,8 +492,8 @@ Packets TcpSender::GeneratePackets(size_t num_packets) {
>     generated.push_back(
>         new MediaPacket(*flow_ids().begin(), 1000 * clock_.TimeInMilliseconds(),
>                         kPacketSizeBytes, next_sequence_number_++));
>-    generated.back()->set_sender_timestamp_us(
>-        1000 * clock_.TimeInMilliseconds());
>+    generated.back()->set_sender_timestamp_us(1000 *
>+                                              clock_.TimeInMilliseconds());
> 
>     total_sent_bytes_ += kPacketSizeBytes;
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/packet_sender.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/packet_sender.h
>index 81a466bca46..42e98b6a8a6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/packet_sender.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/packet_sender.h
>@@ -11,8 +11,8 @@
> #ifndef MODULES_REMOTE_BITRATE_ESTIMATOR_TEST_PACKET_SENDER_H_
> #define MODULES_REMOTE_BITRATE_ESTIMATOR_TEST_PACKET_SENDER_H_
> 
>-#include <list>
> #include <limits>
>+#include <list>
> #include <memory>
> #include <set>
> #include <string>
>@@ -38,7 +38,7 @@ class PacketSender : public PacketProcessor {
>         // We assume that both start at time 0.
>         clock_(0),
>         metric_recorder_(nullptr) {}
>-  virtual ~PacketSender() {}
>+  ~PacketSender() override {}
>   // Call GiveFeedback() with the returned interval in milliseconds, provided
>   // there is a new estimate available.
>   // Note that changing the feedback interval affects the timing of when the
>@@ -47,7 +47,7 @@ class PacketSender : public PacketProcessor {
>   virtual int GetFeedbackIntervalMs() const = 0;
>   void SetSenderTimestamps(Packets* in_out);
> 
>-  virtual uint32_t TargetBitrateKbps() { return 0; }
>+  virtual uint32_t TargetBitrateKbps();
> 
>   virtual void Pause();
>   virtual void Resume(int64_t paused_time_ms);
>@@ -68,12 +68,12 @@ class VideoSender : public PacketSender, public BitrateObserver {
>   VideoSender(PacketProcessorListener* listener,
>               VideoSource* source,
>               BandwidthEstimatorType estimator);
>-  virtual ~VideoSender();
>+  ~VideoSender() override;
> 
>   int GetFeedbackIntervalMs() const override;
>   void RunFor(int64_t time_ms, Packets* in_out) override;
> 
>-  virtual VideoSource* source() const { return source_; }
>+  virtual VideoSource* source() const;
> 
>   uint32_t TargetBitrateKbps() override;
> 
>@@ -104,7 +104,7 @@ class PacedVideoSender : public VideoSender, public PacedSender::PacketSender {
>   PacedVideoSender(PacketProcessorListener* listener,
>                    VideoSource* source,
>                    BandwidthEstimatorType estimator);
>-  virtual ~PacedVideoSender();
>+  ~PacedVideoSender() override;
> 
>   void RunFor(int64_t time_ms, Packets* in_out) override;
> 
>@@ -127,9 +127,7 @@ class PacedVideoSender : public VideoSender, public PacedSender::PacketSender {
>                         bool in_probe_rtt,
>                         int64_t rtt,
>                         uint64_t congestion_window) override;
>-  size_t pacer_queue_size_in_bytes() override {
>-    return pacer_queue_size_in_bytes_;
>-  }
>+  size_t pacer_queue_size_in_bytes() override;
>   void OnBytesAcked(size_t bytes) override;
> 
>  private:
>@@ -152,10 +150,10 @@ class TcpSender : public PacketSender {
>             int flow_id,
>             int64_t offset_ms,
>             int send_limit_bytes);
>-  virtual ~TcpSender() {}
>+  ~TcpSender() override;
> 
>   void RunFor(int64_t time_ms, Packets* in_out) override;
>-  int GetFeedbackIntervalMs() const override { return 10; }
>+  int GetFeedbackIntervalMs() const override;
> 
>   uint32_t TargetBitrateKbps() override;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/plot_dynamics.py b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/plot_dynamics.py
>index 02b20523e8d..e5180634cb7 100755
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/plot_dynamics.py
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/test/plot_dynamics.py
>@@ -41,7 +41,7 @@ def ParsePlotLine(line):
>   # The variable name can contain any non-whitespace character except "#:@"
>   match = re.match(r'([^\s#:@]+)(?:#\d)?:(\d+)@(\S+)', annotated_var)
> 
>-  if match == None:
>+  if match is None:
>     raise ParsePlotLineException("Could not parse variable name, ssrc and \
>                                  algorithm name", annotated_var)
>   var_name = match.group(1)
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.cc
>index 2fcb4241e3b..47e3cfa761e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.cc
>@@ -128,13 +128,13 @@ bool ParseArgsAndSetupEstimator(int argc,
>             new webrtc::RemoteBitrateEstimatorAbsSendTime(observer, clock);
>         *estimator_used = "AbsoluteSendTimeRemoteBitrateEstimator";
>         break;
>-        }
>+      }
>       case webrtc::kRtpExtensionTransmissionTimeOffset: {
>         *estimator =
>             new webrtc::RemoteBitrateEstimatorSingleStream(observer, clock);
>-          *estimator_used = "RemoteBitrateEstimator";
>-          break;
>-        }
>+        *estimator_used = "RemoteBitrateEstimator";
>+        break;
>+      }
>       default:
>         assert(false);
>     }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.h
>index 9a4723b7950..57484fd95d3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp.h
>@@ -21,16 +21,15 @@ class RtpHeaderParser;
> namespace test {
> class RtpFileReader;
> }
>-}
>+}  // namespace webrtc
> 
>-bool ParseArgsAndSetupEstimator(
>-    int argc,
>-    char** argv,
>-    webrtc::Clock* clock,
>-    webrtc::RemoteBitrateObserver* observer,
>-    webrtc::test::RtpFileReader** rtp_reader,
>-    webrtc::RtpHeaderParser** parser,
>-    webrtc::RemoteBitrateEstimator** estimator,
>-    std::string* estimator_used);
>+bool ParseArgsAndSetupEstimator(int argc,
>+                                char** argv,
>+                                webrtc::Clock* clock,
>+                                webrtc::RemoteBitrateObserver* observer,
>+                                webrtc::test::RtpFileReader** rtp_reader,
>+                                webrtc::RtpHeaderParser** parser,
>+                                webrtc::RemoteBitrateEstimator** estimator,
>+                                std::string* estimator_used);
> 
> #endif  // MODULES_REMOTE_BITRATE_ESTIMATOR_TOOLS_BWE_RTP_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp_play.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp_play.cc
>index 887921b6ae8..0594a335001 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp_play.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/tools/bwe_rtp_play.cc
>@@ -104,10 +104,8 @@ int main(int argc, char* argv[]) {
>   printf("Parsed %d packets\nTime passed: %" PRId64 " ms\n", packet_counter,
>          clock.TimeInMilliseconds());
>   printf("Estimator used: %s\n", estimator_used.c_str());
>-  printf("Packets with absolute send time: %d\n",
>-         abs_send_time_count);
>-  printf("Packets with timestamp offset: %d\n",
>-         ts_offset_count);
>+  printf("Packets with absolute send time: %d\n", abs_send_time_count);
>+  printf("Packets with timestamp offset: %d\n", ts_offset_count);
>   printf("Packets with no extension: %d\n",
>          packet_counter - ts_offset_count - abs_send_time_count);
>   return 0;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/tools/rtp_to_text.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/tools/rtp_to_text.cc
>index 54f027a069e..9e82da47141 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/tools/rtp_to_text.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/remote_bitrate_estimator/tools/rtp_to_text.cc
>@@ -29,7 +29,8 @@ int main(int argc, char* argv[]) {
>   bool arrival_time_only = (argc >= 5 && strncmp(argv[4], "-t", 2) == 0);
>   std::unique_ptr<webrtc::test::RtpFileReader> rtp_reader(reader);
>   std::unique_ptr<webrtc::RtpHeaderParser> rtp_parser(parser);
>-  fprintf(stdout, "seqnum timestamp ts_offset abs_sendtime recvtime "
>+  fprintf(stdout,
>+          "seqnum timestamp ts_offset abs_sendtime recvtime "
>           "markerbit ssrc size original_size\n");
>   int packet_counter = 0;
>   int non_zero_abs_send_time = 0;
>@@ -47,24 +48,19 @@ int main(int argc, char* argv[]) {
>       ss << static_cast<int64_t>(packet.time_ms) * 1000000;
>       fprintf(stdout, "%s\n", ss.str().c_str());
>     } else {
>-      fprintf(stdout,
>-              "%u %u %d %u %u %d %u %" PRIuS " %" PRIuS "\n",
>-              header.sequenceNumber,
>-              header.timestamp,
>+      fprintf(stdout, "%u %u %d %u %u %d %u %" PRIuS " %" PRIuS "\n",
>+              header.sequenceNumber, header.timestamp,
>               header.extension.transmissionTimeOffset,
>-              header.extension.absoluteSendTime,
>-              packet.time_ms,
>-              header.markerBit,
>-              header.ssrc,
>-              packet.length,
>+              header.extension.absoluteSendTime, packet.time_ms,
>+              header.markerBit, header.ssrc, packet.length,
>               packet.original_length);
>     }
>     ++packet_counter;
>   }
>   fprintf(stderr, "Parsed %d packets\n", packet_counter);
>   fprintf(stderr, "Packets with non-zero absolute send time: %d\n",
>-         non_zero_abs_send_time);
>+          non_zero_abs_send_time);
>   fprintf(stderr, "Packets with non-zero timestamp offset: %d\n",
>-         non_zero_ts_offsets);
>+          non_zero_ts_offsets);
>   return 0;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/BUILD.gn
>index bf94013bf8c..8df53805f85 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/BUILD.gn
>@@ -40,6 +40,8 @@ rtc_source_set("rtp_rtcp_format") {
>     "source/rtcp_packet/tmmbr.h",
>     "source/rtcp_packet/transport_feedback.h",
>     "source/rtcp_packet/voip_metric.h",
>+    "source/rtp_generic_frame_descriptor.h",
>+    "source/rtp_generic_frame_descriptor_extension.h",
>     "source/rtp_header_extensions.h",
>     "source/rtp_packet.h",
>     "source/rtp_packet_received.h",
>@@ -73,26 +75,29 @@ rtc_source_set("rtp_rtcp_format") {
>     "source/rtcp_packet/tmmbr.cc",
>     "source/rtcp_packet/transport_feedback.cc",
>     "source/rtcp_packet/voip_metric.cc",
>+    "source/rtp_generic_frame_descriptor.cc",
>+    "source/rtp_generic_frame_descriptor_extension.cc",
>     "source/rtp_header_extension_map.cc",
>     "source/rtp_header_extensions.cc",
>     "source/rtp_packet.cc",
>     "source/rtp_packet_received.cc",
>+    "source/rtp_packet_to_send.cc",
>   ]
> 
>   deps = [
>     "..:module_api",
>     "../..:webrtc_common",
>-    "../../:typedefs",
>     "../../api:array_view",
>     "../../api:libjingle_peerconnection_api",
>-    "../../api:optional",
>-    "../../api:video_frame_api",
>     "../../api/audio_codecs:audio_codecs_api",
>+    "../../api/video:video_frame",
>     "../../common_video",
>     "../../rtc_base:checks",
>     "../../rtc_base:deprecation",
>     "../../rtc_base:rtc_base_approved",
>     "../../system_wrappers",
>+    "//third_party/abseil-cpp/absl/types:optional",
>+    "//third_party/abseil-cpp/absl/types:variant",
>   ]
> }
> 
>@@ -108,9 +113,13 @@ rtc_static_library("rtp_rtcp") {
>     "include/rtp_receiver.h",
>     "include/rtp_rtcp.h",
>     "include/ulpfec_receiver.h",
>+    "source/contributing_sources.cc",
>+    "source/contributing_sources.h",
>     "source/dtmf_queue.cc",
>     "source/dtmf_queue.h",
>+    "source/fec_private_tables_bursty.cc",
>     "source/fec_private_tables_bursty.h",
>+    "source/fec_private_tables_random.cc",
>     "source/fec_private_tables_random.h",
>     "source/flexfec_header_reader_writer.cc",
>     "source/flexfec_header_reader_writer.h",
>@@ -139,8 +148,6 @@ rtc_static_library("rtp_rtcp") {
>     "source/rtp_format_h264.h",
>     "source/rtp_format_video_generic.cc",
>     "source/rtp_format_video_generic.h",
>-    "source/rtp_format_video_stereo.cc",
>-    "source/rtp_format_video_stereo.h",
>     "source/rtp_format_vp8.cc",
>     "source/rtp_format_vp8.h",
>     "source/rtp_format_vp9.cc",
>@@ -178,7 +185,6 @@ rtc_static_library("rtp_rtcp") {
>     "source/ulpfec_header_reader_writer.h",
>     "source/ulpfec_receiver_impl.cc",
>     "source/ulpfec_receiver_impl.h",
>-    "source/video_codec_information.h",
>   ]
> 
>   if (rtc_enable_bwe_test_logging) {
>@@ -187,47 +193,40 @@ rtc_static_library("rtp_rtcp") {
>     defines = [ "BWE_TEST_LOGGING_COMPILE_TIME_ENABLE=0" ]
>   }
> 
>-  if (!build_with_chromium && is_clang) {
>-    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>-    suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-  }
>-
>   deps = [
>     ":rtp_rtcp_format",
>     "..:module_api",
>     "../..:webrtc_common",
>-    "../../:typedefs",
>     "../../api:array_view",
>     "../../api:libjingle_peerconnection_api",
>-    "../../api:optional",
>     "../../api:transport_api",
>     "../../api/audio_codecs:audio_codecs_api",
>+    "../../api/video:video_bitrate_allocation",
>+    "../../api/video:video_bitrate_allocator",
>+    "../../api/video_codecs:video_codecs_api",
>     "../../common_video",
>+    "../../logging:rtc_event_audio",
>     "../../logging:rtc_event_log_api",
>+    "../../logging:rtc_event_rtp_rtcp",
>     "../../rtc_base:checks",
>     "../../rtc_base:deprecation",
>     "../../rtc_base:gtest_prod",
>     "../../rtc_base:rate_limiter",
>     "../../rtc_base:rtc_base_approved",
>     "../../rtc_base:rtc_numerics",
>+    "../../rtc_base:safe_minmax",
>     "../../rtc_base:sequenced_task_checker",
>     "../../rtc_base:stringutils",
>+    "../../rtc_base/system:fallthrough",
>+    "../../rtc_base/time:timestamp_extrapolator",
>     "../../system_wrappers",
>     "../../system_wrappers:field_trial_api",
>     "../../system_wrappers:metrics_api",
>     "../audio_coding:audio_format_conversion",
>     "../remote_bitrate_estimator",
>+    "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
>-
>-  # TODO(jschuh): Bug 1348: fix this warning.
>-  configs += [ "//build/config/compiler:no_size_t_to_int_warning" ]
>-
>-  if (is_win) {
>-    cflags = [
>-      # TODO(kjellander): Bug 261: fix this warning.
>-      "/wd4373",  # virtual function override.
>-    ]
>-  }
> }
> 
> rtc_source_set("rtcp_transceiver") {
>@@ -247,13 +246,31 @@ rtc_source_set("rtcp_transceiver") {
>     ":rtp_rtcp_format",
>     "../../:webrtc_common",
>     "../../api:array_view",
>-    "../../api:optional",
>+    "../../api:libjingle_peerconnection_api",
>     "../../api:transport_api",
>+    "../../api/video:video_bitrate_allocation",
>     "../../rtc_base:checks",
>     "../../rtc_base:rtc_base_approved",
>     "../../rtc_base:rtc_task_queue",
>     "../../rtc_base:weak_ptr",
>     "../../system_wrappers",
>+    "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/types:optional",
>+  ]
>+}
>+
>+rtc_source_set("rtp_video_header") {
>+  visibility = [ "*" ]
>+  sources = [
>+    "source/rtp_video_header.cc",
>+    "source/rtp_video_header.h",
>+  ]
>+  deps = [
>+    "../../:webrtc_common",
>+    "../../api/video:video_frame",
>+    "../../modules/video_coding:codec_globals_headers",
>+    "//third_party/abseil-cpp/absl/container:inlined_vector",
>+    "//third_party/abseil-cpp/absl/types:variant",
>   ]
> }
> 
>@@ -270,19 +287,19 @@ rtc_source_set("fec_test_helper") {
>     "../../rtc_base:checks",
>     "../../rtc_base:rtc_base_approved",
>   ]
>-
>-  # TODO(jschuh): bugs.webrtc.org/1348: fix this warning.
>-  configs += [ "//build/config/compiler:no_size_t_to_int_warning" ]
>-  if (!build_with_chromium && is_clang) {
>-    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>-    suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-  }
> }
> 
> rtc_source_set("mock_rtp_rtcp") {
>   testonly = true
>   sources = [
>+    "mocks/mock_recovered_packet_receiver.cc",
>+    "mocks/mock_rtcp_bandwidth_observer.cc",
>+    "mocks/mock_rtcp_rtt_stats.cc",
>+    "mocks/mock_rtp_rtcp.cc",
>+  ]
>+  public = [
>     "mocks/mock_recovered_packet_receiver.h",
>+    "mocks/mock_rtcp_bandwidth_observer.h",
>     "mocks/mock_rtcp_rtt_stats.h",
>     "mocks/mock_rtp_rtcp.h",
>   ]
>@@ -290,10 +307,11 @@ rtc_source_set("mock_rtp_rtcp") {
>     ":rtp_rtcp",
>     ":rtp_rtcp_format",
>     "..:module_api",
>-    "../../api:optional",
>+    "../../api/video:video_bitrate_allocation",
>     "../../rtc_base:checks",
>     "../../rtc_base:rtc_base_approved",
>     "../../test:test_support",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
> 
>@@ -308,6 +326,7 @@ if (rtc_include_tests) {
> 
>     deps = [
>       ":rtp_rtcp",
>+      "../../test:fileutils",
>       "../../test:test_main",
>       "//testing/gtest",
>     ]
>@@ -323,12 +342,9 @@ if (rtc_include_tests) {
>       ":rtp_rtcp",
>       ":rtp_rtcp_format",
>       "../../rtc_base:rtc_base_approved",
>+      "../../test:fileutils",
>       "../../test:test_support",
>     ]
>-    if (!build_with_chromium && is_clang) {
>-      # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>-      suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-    }
>   }
> 
>   rtc_source_set("rtp_rtcp_unittests") {
>@@ -336,6 +352,8 @@ if (rtc_include_tests) {
> 
>     sources = [
>       "source/byte_io_unittest.cc",
>+      "source/contributing_sources_unittest.cc",
>+      "source/fec_private_tables_bursty_unittest.cc",
>       "source/flexfec_header_reader_writer_unittest.cc",
>       "source/flexfec_receiver_unittest.cc",
>       "source/flexfec_sender_unittest.cc",
>@@ -375,11 +393,11 @@ if (rtc_include_tests) {
>       "source/rtp_fec_unittest.cc",
>       "source/rtp_format_h264_unittest.cc",
>       "source/rtp_format_video_generic_unittest.cc",
>-      "source/rtp_format_video_stereo_unittest.cc",
>       "source/rtp_format_vp8_test_helper.cc",
>       "source/rtp_format_vp8_test_helper.h",
>       "source/rtp_format_vp8_unittest.cc",
>       "source/rtp_format_vp9_unittest.cc",
>+      "source/rtp_generic_frame_descriptor_extension_unittest.cc",
>       "source/rtp_header_extension_map_unittest.cc",
>       "source/rtp_packet_history_unittest.cc",
>       "source/rtp_packet_unittest.cc",
>@@ -406,12 +424,13 @@ if (rtc_include_tests) {
>       ":rtp_rtcp_format",
>       "..:module_api",
>       "../..:webrtc_common",
>-      "../../:typedefs",
>       "../../api:array_view",
>       "../../api:libjingle_peerconnection_api",
>-      "../../api:optional",
>       "../../api:transport_api",
>-      "../../api:video_frame_api",
>+      "../../api/video:video_bitrate_allocation",
>+      "../../api/video:video_bitrate_allocator",
>+      "../../api/video:video_frame",
>+      "../../api/video_codecs:video_codecs_api",
>       "../../call:rtp_receiver",
>       "../../common_video:common_video",
>       "../../logging:mocks",
>@@ -427,14 +446,8 @@ if (rtc_include_tests) {
>       "../../test:test_common",
>       "../../test:test_support",
>       "../audio_coding:audio_format_conversion",
>-      "//testing/gmock",
>+      "//third_party/abseil-cpp/absl/memory",
>+      "//third_party/abseil-cpp/absl/types:optional",
>     ]
>-
>-    # TODO(jschuh): bugs.webrtc.org/1348: fix this warning.
>-    configs += [ "//build/config/compiler:no_size_t_to_int_warning" ]
>-    if (!build_with_chromium && is_clang) {
>-      # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>-      suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-    }
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/flexfec_receiver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/flexfec_receiver.h
>index 024d691bf9c..f006111e117 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/flexfec_receiver.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/flexfec_receiver.h
>@@ -16,7 +16,6 @@
> #include "modules/rtp_rtcp/include/ulpfec_receiver.h"
> #include "modules/rtp_rtcp/source/forward_error_correction.h"
> #include "modules/rtp_rtcp/source/rtp_packet_received.h"
>-#include "rtc_base/basictypes.h"
> #include "rtc_base/sequenced_task_checker.h"
> #include "system_wrappers/include/clock.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/flexfec_sender.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/flexfec_sender.h
>index 5b939d534ba..57ba84124f6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/flexfec_sender.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/flexfec_sender.h
>@@ -12,6 +12,7 @@
> #define MODULES_RTP_RTCP_INCLUDE_FLEXFEC_SENDER_H_
> 
> #include <memory>
>+#include <string>
> #include <vector>
> 
> #include "api/array_view.h"
>@@ -22,7 +23,6 @@
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> #include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
> #include "modules/rtp_rtcp/source/ulpfec_generator.h"
>-#include "rtc_base/basictypes.h"
> #include "rtc_base/random.h"
> #include "system_wrappers/include/clock.h"
> 
>@@ -38,6 +38,7 @@ class FlexfecSender {
>   FlexfecSender(int payload_type,
>                 uint32_t ssrc,
>                 uint32_t protected_media_ssrc,
>+                const std::string& mid,
>                 const std::vector<RtpExtension>& rtp_header_extensions,
>                 rtc::ArrayView<const RtpExtensionSize> extension_sizes,
>                 const RtpState* rtp_state,
>@@ -79,6 +80,8 @@ class FlexfecSender {
>   const uint32_t timestamp_offset_;
>   const uint32_t ssrc_;
>   const uint32_t protected_media_ssrc_;
>+  // MID value to send in the MID header extension.
>+  const std::string mid_;
>   // Sequence number of next packet to generate.
>   uint16_t seq_num_;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/receive_statistics.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/receive_statistics.h
>index 55c2c056e72..6c51e8502ba 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/receive_statistics.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/receive_statistics.h
>@@ -16,9 +16,8 @@
> 
> #include "modules/include/module.h"
> #include "modules/include/module_common_types.h"
>-#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
> 
> namespace webrtc {
> 
>@@ -37,7 +36,17 @@ class StreamStatistician {
>  public:
>   virtual ~StreamStatistician();
> 
>-  virtual bool GetStatistics(RtcpStatistics* statistics, bool reset) = 0;
>+  // If |update_fraction_lost| is true, |fraction_lost| will be recomputed
>+  // between now and the last time |update_fraction_lost| was true. Otherwise
>+  // the last-computed value of |fraction_lost| will be returned.
>+  //
>+  // |update_fraction_lost| should be true any time an RTCP SR or RR is being
>+  // generated, since RFC3550 defines it as the fraction of packets lost since
>+  // the previous SR or RR packet was sent.
>+  //
>+  // Aside from |fraction_lost|, every other value will be freshly computed.
>+  virtual bool GetStatistics(RtcpStatistics* statistics,
>+                             bool update_fraction_lost) = 0;
>   virtual void GetDataCounters(size_t* bytes_received,
>                                uint32_t* packets_received) const = 0;
> 
>@@ -49,11 +58,7 @@ class StreamStatistician {
> 
>   // Returns true if the packet with RTP header |header| is likely to be a
>   // retransmitted packet, false otherwise.
>-  virtual bool IsRetransmitOfOldPacket(const RTPHeader& header,
>-                                       int64_t min_rtt) const = 0;
>-
>-  // Returns true if |sequence_number| is received in order, false otherwise.
>-  virtual bool IsPacketInOrder(uint16_t sequence_number) const = 0;
>+  virtual bool IsRetransmitOfOldPacket(const RTPHeader& header) const = 0;
> };
> 
> class ReceiveStatistics : public ReceiveStatisticsProvider {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h
>index e3d95b20ca1..5195e8ab7e3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h
>@@ -33,7 +33,9 @@ class RemoteNtpTimeEstimator {
> 
>   // Updates the estimator with round trip time |rtt|, NTP seconds |ntp_secs|,
>   // NTP fraction |ntp_frac| and RTP timestamp |rtcp_timestamp|.
>-  bool UpdateRtcpTimestamp(int64_t rtt, uint32_t ntp_secs, uint32_t ntp_frac,
>+  bool UpdateRtcpTimestamp(int64_t rtt,
>+                           uint32_t ntp_secs,
>+                           uint32_t ntp_frac,
>                            uint32_t rtp_timestamp);
> 
>   // Estimates the NTP timestamp in local timebase from |rtp_timestamp|.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/rtp_header_extension_map.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/rtp_header_extension_map.h
>index efac056322b..89a0ed526fa 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/rtp_header_extension_map.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/rtp_header_extension_map.h
>@@ -16,7 +16,6 @@
> #include "api/array_view.h"
> #include "api/rtpparameters.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>-#include "rtc_base/basictypes.h"
> #include "rtc_base/checks.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/rtp_header_parser.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/rtp_header_parser.h
>index 59012ce0a44..2d84fc1e690 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/rtp_header_parser.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/rtp_header_parser.h
>@@ -11,7 +11,6 @@
> #define MODULES_RTP_RTCP_INCLUDE_RTP_HEADER_PARSER_H_
> 
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h
>index 942c4ff1e7b..229c71b0a2c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h
>@@ -14,15 +14,14 @@
> #include <map>
> #include <set>
> 
>+#include "absl/types/optional.h"
> #include "api/audio_codecs/audio_format.h"
>-#include "api/optional.h"
>+#include "api/video_codecs/video_codec.h"
> #include "modules/rtp_rtcp/source/rtp_utility.h"
> #include "rtc_base/criticalsection.h"
> 
> namespace webrtc {
> 
>-class VideoCodec;
>-
> class RTPPayloadRegistry {
>  public:
>   RTPPayloadRegistry();
>@@ -41,84 +40,21 @@ class RTPPayloadRegistry {
> 
>   int32_t DeRegisterReceivePayload(int8_t payload_type);
> 
>-  int32_t ReceivePayloadType(const SdpAudioFormat& audio_format,
>-                             int8_t* payload_type) const;
>-  int32_t ReceivePayloadType(const VideoCodec& video_codec,
>-                             int8_t* payload_type) const;
>-
>-  bool RtxEnabled() const;
>-
>-  void SetRtxSsrc(uint32_t ssrc);
>-
>-  bool GetRtxSsrc(uint32_t* ssrc) const;
>-
>-  void SetRtxPayloadType(int payload_type, int associated_payload_type);
>-
>-  bool IsRed(const RTPHeader& header) const;
>-
>   int GetPayloadTypeFrequency(uint8_t payload_type) const;
> 
>-  rtc::Optional<RtpUtility::Payload> PayloadTypeToPayload(
>+  absl::optional<RtpUtility::Payload> PayloadTypeToPayload(
>       uint8_t payload_type) const;
> 
>-  void ResetLastReceivedPayloadTypes() {
>-    rtc::CritScope cs(&crit_sect_);
>-    last_received_payload_type_ = -1;
>-    last_received_media_payload_type_ = -1;
>-  }
>-
>-  // This sets the payload type of the packets being received from the network
>-  // on the media SSRC. For instance if packets are encapsulated with RED, this
>-  // payload type will be the RED payload type.
>-  void SetIncomingPayloadType(const RTPHeader& header);
>-
>-  // Returns true if the new media payload type has not changed.
>-  bool ReportMediaPayloadType(uint8_t media_payload_type);
>-
>-  int8_t red_payload_type() const { return GetPayloadTypeWithName("red"); }
>-  int8_t ulpfec_payload_type() const {
>-    return GetPayloadTypeWithName("ulpfec");
>-  }
>-  int8_t last_received_payload_type() const {
>-    rtc::CritScope cs(&crit_sect_);
>-    return last_received_payload_type_;
>-  }
>-  void set_last_received_payload_type(int8_t last_received_payload_type) {
>-    rtc::CritScope cs(&crit_sect_);
>-    last_received_payload_type_ = last_received_payload_type;
>-  }
>-
>-  int8_t last_received_media_payload_type() const {
>-    rtc::CritScope cs(&crit_sect_);
>-    return last_received_media_payload_type_;
>-  }
>-
>  private:
>   // Prunes the payload type map of the specific payload type, if it exists.
>   void DeregisterAudioCodecOrRedTypeRegardlessOfPayloadType(
>       const SdpAudioFormat& audio_format);
> 
>-  bool IsRtxInternal(const RTPHeader& header) const;
>-  // Returns the payload type for the payload with name |payload_name|, or -1 if
>-  // no such payload is registered.
>-  int8_t GetPayloadTypeWithName(const char* payload_name) const;
>-
>   rtc::CriticalSection crit_sect_;
>   std::map<int, RtpUtility::Payload> payload_type_map_;
>-  int8_t incoming_payload_type_;
>-  int8_t last_received_payload_type_;
>-  int8_t last_received_media_payload_type_;
>-  bool rtx_;
>-  // Mapping rtx_payload_type_map_[rtx] = associated.
>-  std::map<int, int> rtx_payload_type_map_;
>-  uint32_t ssrc_rtx_;
>-  // Only warn once per payload type, if an RTX packet is received but
>-  // no associated payload type found in |rtx_payload_type_map_|.
>-  std::set<int> payload_types_with_suppressed_warnings_
>-      RTC_GUARDED_BY(crit_sect_);
>-
>-  // As a first step in splitting this class up in separate cases for audio and
>-  // video, DCHECK that no instance is used for both audio and video.
>+
>+// As a first step in splitting this class up in separate cases for audio and
>+// video, DCHECK that no instance is used for both audio and video.
> #if RTC_DCHECK_IS_ON
>   bool used_for_audio_ RTC_GUARDED_BY(crit_sect_) = false;
>   bool used_for_video_ RTC_GUARDED_BY(crit_sect_) = false;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/rtp_receiver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/rtp_receiver.h
>index 9738116a1a6..d93d6566ef4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/rtp_receiver.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/rtp_receiver.h
>@@ -15,49 +15,28 @@
> 
> #include "api/rtpreceiverinterface.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
> class RTPPayloadRegistry;
> class VideoCodec;
> 
>-class TelephoneEventHandler {
>- public:
>-  virtual ~TelephoneEventHandler() {}
>-
>-  // The following three methods implement the TelephoneEventHandler interface.
>-  // Forward DTMFs to decoder for playout.
>-  virtual void SetTelephoneEventForwardToDecoder(bool forward_to_decoder) = 0;
>-
>-  // Is forwarding of outband telephone events turned on/off?
>-  virtual bool TelephoneEventForwardToDecoder() const = 0;
>-
>-  // Is TelephoneEvent configured with payload type payload_type
>-  virtual bool TelephoneEventPayloadType(const int8_t payload_type) const = 0;
>-};
>-
> class RtpReceiver {
>  public:
>   // Creates a video-enabled RTP receiver.
>   static RtpReceiver* CreateVideoReceiver(
>       Clock* clock,
>       RtpData* incoming_payload_callback,
>-      RtpFeedback* incoming_messages_callback,
>       RTPPayloadRegistry* rtp_payload_registry);
> 
>   // Creates an audio-enabled RTP receiver.
>   static RtpReceiver* CreateAudioReceiver(
>       Clock* clock,
>       RtpData* incoming_payload_callback,
>-      RtpFeedback* incoming_messages_callback,
>       RTPPayloadRegistry* rtp_payload_registry);
> 
>   virtual ~RtpReceiver() {}
> 
>-  // Returns a TelephoneEventHandler if available.
>-  virtual TelephoneEventHandler* GetTelephoneEventHandler() = 0;
>-
>   // Registers a receive payload in the payload registry and notifies the media
>   // receiver strategy.
>   virtual int32_t RegisterReceivePayload(
>@@ -100,12 +79,6 @@ class RtpReceiver {
>   // Returns the remote SSRC of the currently received RTP stream.
>   virtual uint32_t SSRC() const = 0;
> 
>-  // Returns the current remote CSRCs.
>-  virtual int32_t CSRCs(uint32_t array_of_csrc[kRtpCsrcSize]) const = 0;
>-
>-  // Returns the current energy of the RTP stream received.
>-  virtual int32_t Energy(uint8_t array_of_energy[kRtpCsrcSize]) const = 0;
>-
>   virtual std::vector<RtpSource> GetSources() const = 0;
> };
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
>index 7e85cb69830..94237e0acf3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
>@@ -16,7 +16,8 @@
> #include <utility>
> #include <vector>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
>+#include "api/video/video_bitrate_allocation.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/include/module.h"
> #include "modules/rtp_rtcp/include/flexfec_sender.h"
>@@ -94,6 +95,10 @@ class RtpRtcp : public Module, public RtcpFeedbackSenderInterface {
>     RateLimiter* retransmission_rate_limiter = nullptr;
>     OverheadObserver* overhead_observer = nullptr;
>     RtpKeepAliveConfig keepalive_config;
>+    RtcpIntervalConfig rtcp_interval_config;
>+
>+    // Update network2 instead of pacer_exit field of video timing extension.
>+    bool populate_network2_timestamp = false;
> 
>    private:
>     RTC_DISALLOW_COPY_AND_ASSIGN(Configuration);
>@@ -126,9 +131,6 @@ class RtpRtcp : public Module, public RtcpFeedbackSenderInterface {
>   // Sets codec name and payload type. Returns -1 on failure else 0.
>   virtual int32_t RegisterSendPayload(const CodecInst& voice_codec) = 0;
> 
>-  // Sets codec name and payload type. Return -1 on failure else 0.
>-  virtual int32_t RegisterSendPayload(const VideoCodec& video_codec) = 0;
>-
>   virtual void RegisterVideoSendPayload(int payload_type,
>                                         const char* payload_name) = 0;
> 
>@@ -170,6 +172,11 @@ class RtpRtcp : public Module, public RtcpFeedbackSenderInterface {
>   // Sets SSRC, default is a random number.
>   virtual void SetSSRC(uint32_t ssrc) = 0;
> 
>+  // Sets the value for sending in the MID RTP header extension.
>+  // The MID RTP header extension should be registered for this to do anything.
>+  // Once set, this value can not be changed or removed.
>+  virtual void SetMid(const std::string& mid) = 0;
>+
>   // Sets CSRC.
>   // |csrcs| - vector of CSRCs
>   virtual void SetCsrcs(const std::vector<uint32_t>& csrcs) = 0;
>@@ -192,7 +199,7 @@ class RtpRtcp : public Module, public RtcpFeedbackSenderInterface {
>                                      int associated_payload_type) = 0;
> 
>   // Returns the FlexFEC SSRC, if there is one.
>-  virtual rtc::Optional<uint32_t> FlexfecSsrc() const = 0;
>+  virtual absl::optional<uint32_t> FlexfecSsrc() const = 0;
> 
>   // Sets sending status. Sends kRtcpByeCode when going from true to false.
>   // Returns -1 on failure else 0.
>@@ -393,7 +400,8 @@ class RtpRtcp : public Module, public RtcpFeedbackSenderInterface {
>   // BWE feedback packets.
>   bool SendFeedbackPacket(const rtcp::TransportFeedback& packet) override = 0;
> 
>-  virtual void SetVideoBitrateAllocation(const BitrateAllocation& bitrate) = 0;
>+  virtual void SetVideoBitrateAllocation(
>+      const VideoBitrateAllocation& bitrate) = 0;
> 
>   // **************************************************************************
>   // Audio
>@@ -418,13 +426,8 @@ class RtpRtcp : public Module, public RtcpFeedbackSenderInterface {
> 
>   // Set RED and ULPFEC payload types. A payload type of -1 means that the
>   // corresponding feature is turned off. Note that we DO NOT support enabling
>-  // ULPFEC without enabling RED. However, we DO support enabling RED without
>-  // enabling ULPFEC. This is due to an RED/RTX workaround, where the receiver
>-  // assumes that RTX packets carry RED if RED has been configured in the SDP,
>-  // regardless of what RTX payload type mapping was negotiated in the SDP.
>-  // TODO(brandtr): Update this comment when we have removed the RED/RTX
>-  // send-side workaround, i.e., when we do not support enabling RED without
>-  // enabling ULPFEC.
>+  // ULPFEC without enabling RED, and RED is only ever used when ULPFEC is
>+  // enabled.
>   virtual void SetUlpfecConfig(int red_payload_type,
>                                int ulpfec_payload_type) = 0;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.cc
>index d9b082bd903..f86b238b7da 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.cc
>@@ -16,7 +16,19 @@ StreamDataCounters::StreamDataCounters() : first_packet_time_ms(-1) {}
> 
> constexpr size_t StreamId::kMaxSize;
> 
>-bool StreamId::IsLegalName(rtc::ArrayView<const char> name) {
>+// Check if passed character is a "token-char" from RFC 4566.
>+static bool IsTokenChar(char ch) {
>+  return ch == 0x21 || (ch >= 0x23 && ch <= 0x27) || ch == 0x2a || ch == 0x2b ||
>+         ch == 0x2d || ch == 0x2e || (ch >= 0x30 && ch <= 0x39) ||
>+         (ch >= 0x41 && ch <= 0x5a) || (ch >= 0x5e && ch <= 0x7e);
>+}
>+
>+bool StreamId::IsLegalMidName(rtc::ArrayView<const char> name) {
>+  return (name.size() <= kMaxSize && name.size() > 0 &&
>+          std::all_of(name.data(), name.data() + name.size(), IsTokenChar));
>+}
>+
>+bool StreamId::IsLegalRsidName(rtc::ArrayView<const char> name) {
>   return (name.size() <= kMaxSize && name.size() > 0 &&
>           std::all_of(name.data(), name.data() + name.size(), isalnum));
> }
>@@ -33,10 +45,8 @@ void StreamId::Set(const char* data, size_t size) {
> // and thus assume trivial destructibility.
> static_assert(std::is_trivially_destructible<StreamId>::value, "");
> 
>-PayloadUnion::PayloadUnion(const AudioPayload& payload)
>-    : audio_payload_(payload) {}
>-PayloadUnion::PayloadUnion(const VideoPayload& payload)
>-    : video_payload_(payload) {}
>+PayloadUnion::PayloadUnion(const AudioPayload& payload) : payload_(payload) {}
>+PayloadUnion::PayloadUnion(const VideoPayload& payload) : payload_(payload) {}
> PayloadUnion::PayloadUnion(const PayloadUnion&) = default;
> PayloadUnion::PayloadUnion(PayloadUnion&&) = default;
> PayloadUnion::~PayloadUnion() = default;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h
>index bc32ed7ccd0..34c74287900 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h
>@@ -15,17 +15,15 @@
> #include <list>
> #include <vector>
> 
>+#include "absl/types/variant.h"
> #include "api/audio_codecs/audio_format.h"
> #include "api/rtp_headers.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/include/module_common_types.h"
>-#include "rtc_base/deprecation.h"
> #include "system_wrappers/include/clock.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
>-#define RTCP_CNAME_SIZE 256    // RFC 3550 page 44, including null termination
>-#define IP_PACKET_SIZE 1500    // we assume ethernet
>-#define MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS 10
>+#define RTCP_CNAME_SIZE 256  // RFC 3550 page 44, including null termination
>+#define IP_PACKET_SIZE 1500  // we assume ethernet
> 
> namespace webrtc {
> namespace rtcp {
>@@ -41,14 +39,23 @@ const int kBogusRtpRateForAudioRtcp = 8000;
> // Minimum RTP header size in bytes.
> const uint8_t kRtpHeaderSize = 12;
> 
>+struct RtcpIntervalConfig final {
>+  RtcpIntervalConfig() = default;
>+  RtcpIntervalConfig(int64_t video_interval_ms, int64_t audio_interval_ms)
>+      : video_interval_ms(video_interval_ms),
>+        audio_interval_ms(audio_interval_ms) {}
>+  int64_t video_interval_ms = 1000;
>+  int64_t audio_interval_ms = 5000;
>+};
>+
> struct AudioPayload {
>   SdpAudioFormat format;
>   uint32_t rate;
> };
> 
> struct VideoPayload {
>-  RtpVideoCodecTypes videoCodecType;
>-  // The H264 profile only matters if videoCodecType == kRtpVideoH264.
>+  VideoCodecType videoCodecType;
>+  // The H264 profile only matters if videoCodecType == kVideoCodecH264.
>   H264::Profile h264_profile;
> };
> 
>@@ -63,41 +70,28 @@ class PayloadUnion {
>   PayloadUnion& operator=(const PayloadUnion&);
>   PayloadUnion& operator=(PayloadUnion&&);
> 
>-  bool is_audio() const { return audio_payload_.has_value(); }
>-  bool is_video() const { return video_payload_.has_value(); }
>-  const AudioPayload& audio_payload() const {
>-    RTC_DCHECK(audio_payload_);
>-    return *audio_payload_;
>+  bool is_audio() const {
>+    return absl::holds_alternative<AudioPayload>(payload_);
>   }
>-  const VideoPayload& video_payload() const {
>-    RTC_DCHECK(video_payload_);
>-    return *video_payload_;
>+  bool is_video() const {
>+    return absl::holds_alternative<VideoPayload>(payload_);
>   }
>-  AudioPayload& audio_payload() {
>-    RTC_DCHECK(audio_payload_);
>-    return *audio_payload_;
>+  const AudioPayload& audio_payload() const {
>+    return absl::get<AudioPayload>(payload_);
>   }
>-  VideoPayload& video_payload() {
>-    RTC_DCHECK(video_payload_);
>-    return *video_payload_;
>+  const VideoPayload& video_payload() const {
>+    return absl::get<VideoPayload>(payload_);
>   }
>+  AudioPayload& audio_payload() { return absl::get<AudioPayload>(payload_); }
>+  VideoPayload& video_payload() { return absl::get<VideoPayload>(payload_); }
> 
>  private:
>-  rtc::Optional<AudioPayload> audio_payload_;
>-  rtc::Optional<VideoPayload> video_payload_;
>+  absl::variant<AudioPayload, VideoPayload> payload_;
> };
> 
>-enum RTPAliveType { kRtpDead = 0, kRtpNoRtp = 1, kRtpAlive = 2 };
>+enum ProtectionType { kUnprotectedPacket, kProtectedPacket };
> 
>-enum ProtectionType {
>-  kUnprotectedPacket,
>-  kProtectedPacket
>-};
>-
>-enum StorageType {
>-  kDontRetransmit,
>-  kAllowRetransmission
>-};
>+enum StorageType { kDontRetransmit, kAllowRetransmission };
> 
> enum RTPExtensionType {
>   kRtpExtensionNone,
>@@ -112,6 +106,7 @@ enum RTPExtensionType {
>   kRtpExtensionRtpStreamId,
>   kRtpExtensionRepairedRtpStreamId,
>   kRtpExtensionMid,
>+  kRtpExtensionGenericFrameDescriptor,
>   kRtpExtensionNumberOfExtensions  // Must be the last entity in the enum.
> };
> 
>@@ -158,10 +153,10 @@ enum RetransmissionMode : uint8_t {
> };
> 
> enum RtxMode {
>-  kRtxOff                 = 0x0,
>-  kRtxRetransmitted       = 0x1,  // Only send retransmissions over RTX.
>-  kRtxRedundantPayloads   = 0x2   // Preventively send redundant payloads
>-                                  // instead of padding.
>+  kRtxOff = 0x0,
>+  kRtxRetransmitted = 0x1,     // Only send retransmissions over RTX.
>+  kRtxRedundantPayloads = 0x2  // Preventively send redundant payloads
>+                               // instead of padding.
> };
> 
> const size_t kRtxHeaderSize = 2;
>@@ -198,7 +193,7 @@ struct RTCPReportBlock {
>   uint32_t sender_ssrc;  // SSRC of sender of this report.
>   uint32_t source_ssrc;  // SSRC of the RTP packet sender.
>   uint8_t fraction_lost;
>-  uint32_t packets_lost;  // 24 bits valid.
>+  int32_t packets_lost;  // 24 bits valid.
>   uint32_t extended_highest_sequence_number;
>   uint32_t jitter;
>   uint32_t last_sender_report_timestamp;
>@@ -243,34 +238,11 @@ class RecoveredPacketReceiver {
>   virtual ~RecoveredPacketReceiver() = default;
> };
> 
>-class RtpFeedback {
>- public:
>-  virtual ~RtpFeedback() {}
>-
>-  // Receiving payload change or SSRC change. (return success!)
>-  /*
>-  *   channels    - number of channels in codec (1 = mono, 2 = stereo)
>-  */
>-  virtual int32_t OnInitializeDecoder(int payload_type,
>-                                      const SdpAudioFormat& audio_format,
>-                                      uint32_t rate) = 0;
>-
>-  virtual void OnIncomingSSRCChanged(uint32_t ssrc) = 0;
>-
>-  virtual void OnIncomingCSRCChanged(uint32_t csrc, bool added) = 0;
>-};
>-
> class RtcpIntraFrameObserver {
>  public:
>-  virtual void OnReceivedIntraFrameRequest(uint32_t ssrc) = 0;
>-
>-  RTC_DEPRECATED virtual void OnReceivedSLI(uint32_t /* ssrc */,
>-                             uint8_t /* picture_id */) {}
>-
>-  RTC_DEPRECATED virtual void OnReceivedRPSI(uint32_t /* ssrc */,
>-                              uint64_t /* picture_id */) {}
>-
>   virtual ~RtcpIntraFrameObserver() {}
>+
>+  virtual void OnReceivedIntraFrameRequest(uint32_t ssrc) = 0;
> };
> 
> class RtcpBandwidthObserver {
>@@ -371,6 +343,9 @@ struct PacketFeedback {
>   // Packet identifier, incremented with 1 for every packet generated by the
>   // sender.
>   uint16_t sequence_number;
>+  // Session unique packet identifier, incremented with 1 for every packet
>+  // generated by the sender.
>+  int64_t long_sequence_number;
>   // Size of the packet excluding RTP headers.
>   size_t payload_size;
>   // The network route ids that this packet is associated with.
>@@ -403,8 +378,6 @@ class TransportFeedbackObserver {
>                          const PacedPacketInfo& pacing_info) = 0;
> 
>   virtual void OnTransportFeedback(const rtcp::TransportFeedback& feedback) = 0;
>-
>-  virtual std::vector<PacketFeedback> GetTransportFeedbackVector() const = 0;
> };
> 
> // Interface for PacketRouter to send rtcp feedback on behalf of
>@@ -438,26 +411,6 @@ class RtcpRttStats {
>   virtual ~RtcpRttStats() {}
> };
> 
>-// Null object version of RtpFeedback.
>-class NullRtpFeedback : public RtpFeedback {
>- public:
>-  ~NullRtpFeedback() override {}
>-
>-  int32_t OnInitializeDecoder(int payload_type,
>-                              const SdpAudioFormat& audio_format,
>-                              uint32_t rate) override;
>-
>-  void OnIncomingSSRCChanged(uint32_t /* ssrc */) override {}
>-  void OnIncomingCSRCChanged(uint32_t /* csrc */, bool /* added */) override {}
>-};
>-
>-inline int32_t NullRtpFeedback::OnInitializeDecoder(
>-    int /* payload_type */,
>-    const SdpAudioFormat&,
>-    uint32_t /* rate */) {
>-  return 0;
>-}
>-
> // Statistics about packet loss for a single directional connection. All values
> // are totals since the connection initiated.
> struct RtpPacketLossStats {
>@@ -499,7 +452,7 @@ class RtpPacketSender {
>   // at high priority.
>   // TODO(alexnarest): Make it pure virtual after rtp_sender_unittest will be
>   // updated to support it
>-  virtual void SetAccountForAudioPackets(bool  /* account_for_audio */) {}
>+  virtual void SetAccountForAudioPackets(bool account_for_audio) {}
> };
> 
> class TransportSequenceNumberAllocator {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/ulpfec_receiver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/ulpfec_receiver.h
>index 6ea8496f831..fce6e888cc7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/ulpfec_receiver.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/include/ulpfec_receiver.h
>@@ -12,7 +12,6 @@
> #define MODULES_RTP_RTCP_INCLUDE_ULPFEC_RECEIVER_H_
> 
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/mocks/mock_recovered_packet_receiver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/mocks/mock_recovered_packet_receiver.cc
>new file mode 100644
>index 00000000000..e0183f89c02
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/mocks/mock_recovered_packet_receiver.cc
>@@ -0,0 +1,18 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/rtp_rtcp/mocks/mock_recovered_packet_receiver.h"
>+
>+namespace webrtc {
>+
>+MockRecoveredPacketReceiver::MockRecoveredPacketReceiver() = default;
>+MockRecoveredPacketReceiver::~MockRecoveredPacketReceiver() = default;
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/mocks/mock_recovered_packet_receiver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/mocks/mock_recovered_packet_receiver.h
>index 501c5c3298a..d6442ad58a4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/mocks/mock_recovered_packet_receiver.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/mocks/mock_recovered_packet_receiver.h
>@@ -12,13 +12,14 @@
> #define MODULES_RTP_RTCP_MOCKS_MOCK_RECOVERED_PACKET_RECEIVER_H_
> 
> #include "modules/rtp_rtcp/include/flexfec_receiver.h"
>-#include "rtc_base/basictypes.h"
> #include "test/gmock.h"
> 
> namespace webrtc {
> 
> class MockRecoveredPacketReceiver : public RecoveredPacketReceiver {
>  public:
>+  MockRecoveredPacketReceiver();
>+  ~MockRecoveredPacketReceiver();
>   MOCK_METHOD2(OnRecoveredPacket, void(const uint8_t* packet, size_t length));
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/mocks/mock_rtcp_bandwidth_observer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/mocks/mock_rtcp_bandwidth_observer.cc
>new file mode 100644
>index 00000000000..abaf129d8b8
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/mocks/mock_rtcp_bandwidth_observer.cc
>@@ -0,0 +1,18 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/rtp_rtcp/mocks/mock_rtcp_bandwidth_observer.h"
>+
>+namespace webrtc {
>+
>+MockRtcpBandwidthObserver::MockRtcpBandwidthObserver() = default;
>+MockRtcpBandwidthObserver::~MockRtcpBandwidthObserver() = default;
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/mocks/mock_rtcp_bandwidth_observer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/mocks/mock_rtcp_bandwidth_observer.h
>new file mode 100644
>index 00000000000..b9a8f79a1d0
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/mocks/mock_rtcp_bandwidth_observer.h
>@@ -0,0 +1,29 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_RTP_RTCP_MOCKS_MOCK_RTCP_BANDWIDTH_OBSERVER_H_
>+#define MODULES_RTP_RTCP_MOCKS_MOCK_RTCP_BANDWIDTH_OBSERVER_H_
>+
>+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>+#include "test/gmock.h"
>+
>+namespace webrtc {
>+
>+class MockRtcpBandwidthObserver : public RtcpBandwidthObserver {
>+ public:
>+  MockRtcpBandwidthObserver();
>+  ~MockRtcpBandwidthObserver();
>+
>+  MOCK_METHOD1(OnReceivedEstimatedBitrate, void(uint32_t));
>+  MOCK_METHOD3(OnReceivedRtcpReceiverReport,
>+               void(const ReportBlockList&, int64_t, int64_t));
>+};
>+}  // namespace webrtc
>+#endif  // MODULES_RTP_RTCP_MOCKS_MOCK_RTCP_BANDWIDTH_OBSERVER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.cc
>new file mode 100644
>index 00000000000..1aca566bb68
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.cc
>@@ -0,0 +1,18 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h"
>+
>+namespace webrtc {
>+
>+MockRtcpRttStats::MockRtcpRttStats() = default;
>+MockRtcpRttStats::~MockRtcpRttStats() = default;
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h
>index 8f2bb6e1e17..6ccef612c20 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h
>@@ -18,6 +18,9 @@ namespace webrtc {
> 
> class MockRtcpRttStats : public RtcpRttStats {
>  public:
>+  MockRtcpRttStats();
>+  ~MockRtcpRttStats();
>+
>   MOCK_METHOD1(OnRttUpdate, void(int64_t rtt));
>   MOCK_CONST_METHOD0(LastProcessedRtt, int64_t());
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.cc
>new file mode 100644
>index 00000000000..d24c1b094c5
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.cc
>@@ -0,0 +1,21 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
>+
>+namespace webrtc {
>+
>+MockRtpData::MockRtpData() = default;
>+MockRtpData::~MockRtpData() = default;
>+
>+MockRtpRtcp::MockRtpRtcp() = default;
>+MockRtpRtcp::~MockRtpRtcp() = default;
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
>index 05eea81cd5f..d1769304d8b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
>@@ -12,10 +12,12 @@
> #define MODULES_RTP_RTCP_MOCKS_MOCK_RTP_RTCP_H_
> 
> #include <set>
>+#include <string>
> #include <utility>
> #include <vector>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
>+#include "api/video/video_bitrate_allocation.h"
> #include "modules/include/module.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>@@ -27,6 +29,9 @@ namespace webrtc {
> 
> class MockRtpData : public RtpData {
>  public:
>+  MockRtpData();
>+  ~MockRtpData();
>+
>   MOCK_METHOD3(OnReceivedPayloadData,
>                int32_t(const uint8_t* payload_data,
>                        size_t payload_size,
>@@ -35,27 +40,15 @@ class MockRtpData : public RtpData {
> 
> class MockRtpRtcp : public RtpRtcp {
>  public:
>-  MOCK_METHOD1(RegisterDefaultModule, int32_t(RtpRtcp* module));
>-  MOCK_METHOD0(DeRegisterDefaultModule, int32_t());
>-  MOCK_METHOD0(DefaultModuleRegistered, bool());
>-  MOCK_METHOD0(NumberChildModules, uint32_t());
>-  MOCK_METHOD1(RegisterSyncModule, int32_t(RtpRtcp* module));
>-  MOCK_METHOD0(DeRegisterSyncModule, int32_t());
>+  MockRtpRtcp();
>+  ~MockRtpRtcp();
>+
>   MOCK_METHOD2(IncomingRtcpPacket,
>                void(const uint8_t* incoming_packet, size_t packet_length));
>   MOCK_METHOD1(SetRemoteSSRC, void(uint32_t ssrc));
>-  MOCK_METHOD4(IncomingAudioNTP,
>-               int32_t(uint32_t audio_received_ntp_secs,
>-                       uint32_t audio_received_ntp_frac,
>-                       uint32_t audio_rtcp_arrival_time_secs,
>-                       uint32_t audio_rtcp_arrival_time_frac));
>-  MOCK_METHOD0(InitSender, int32_t());
>-  MOCK_METHOD1(RegisterSendTransport, int32_t(Transport* outgoing_transport));
>   MOCK_METHOD1(SetMaxRtpPacketSize, void(size_t size));
>-  MOCK_METHOD1(SetTransportOverhead, void(int transport_overhead_per_packet));
>   MOCK_CONST_METHOD0(MaxRtpPacketSize, size_t());
>   MOCK_METHOD1(RegisterSendPayload, int32_t(const CodecInst& voice_codec));
>-  MOCK_METHOD1(RegisterSendPayload, int32_t(const VideoCodec& video_codec));
>   MOCK_METHOD2(RegisterVideoSendPayload,
>                void(int payload_type, const char* payload_name));
>   MOCK_METHOD1(DeRegisterSendPayload, int32_t(int8_t payload_type));
>@@ -74,6 +67,7 @@ class MockRtpRtcp : public RtpRtcp {
>   MOCK_CONST_METHOD0(GetRtxState, RtpState());
>   MOCK_CONST_METHOD0(SSRC, uint32_t());
>   MOCK_METHOD1(SetSSRC, void(uint32_t ssrc));
>+  MOCK_METHOD1(SetMid, void(const std::string& mid));
>   MOCK_CONST_METHOD1(CSRCs, int32_t(uint32_t csrcs[kRtpCsrcSize]));
>   MOCK_METHOD1(SetCsrcs, void(const std::vector<uint32_t>& csrcs));
>   MOCK_METHOD1(SetCSRCStatus, int32_t(bool include));
>@@ -81,7 +75,7 @@ class MockRtpRtcp : public RtpRtcp {
>   MOCK_CONST_METHOD0(RtxSendStatus, int());
>   MOCK_METHOD1(SetRtxSsrc, void(uint32_t));
>   MOCK_METHOD2(SetRtxSendPayloadType, void(int, int));
>-  MOCK_CONST_METHOD0(FlexfecSsrc, rtc::Optional<uint32_t>());
>+  MOCK_CONST_METHOD0(FlexfecSsrc, absl::optional<uint32_t>());
>   MOCK_CONST_METHOD0(RtxSendPayloadType, std::pair<int, int>());
>   MOCK_METHOD1(SetSendingStatus, int32_t(bool sending));
>   MOCK_CONST_METHOD0(Sending, bool());
>@@ -92,8 +86,6 @@ class MockRtpRtcp : public RtpRtcp {
>                           uint32_t* video_rate,
>                           uint32_t* fec_rate,
>                           uint32_t* nack_rate));
>-  MOCK_METHOD1(RegisterVideoBitrateObserver, void(BitrateStatisticsObserver*));
>-  MOCK_CONST_METHOD0(GetVideoBitrateObserver, BitrateStatisticsObserver*(void));
>   MOCK_CONST_METHOD1(EstimatedReceiveBandwidth,
>                      int(uint32_t* available_bandwidth));
>   MOCK_METHOD9(SendOutgoingData,
>@@ -174,10 +166,6 @@ class MockRtpRtcp : public RtpRtcp {
>   MOCK_METHOD1(SendFeedbackPacket, bool(const rtcp::TransportFeedback& packet));
>   MOCK_METHOD3(SendTelephoneEventOutband,
>                int32_t(uint8_t key, uint16_t time_ms, uint8_t level));
>-  MOCK_METHOD1(SetSendREDPayloadType, int32_t(int8_t payload_type));
>-  MOCK_CONST_METHOD1(SendREDPayloadType, int32_t(int8_t* payload_type));
>-  MOCK_METHOD2(SetRTPAudioLevelIndicationStatus,
>-               int32_t(bool enable, uint8_t id));
>   MOCK_METHOD1(SetAudioLevel, int32_t(uint8_t level_dbov));
>   MOCK_METHOD1(SetTargetSendBitrate, void(uint32_t bitrate_bps));
>   MOCK_METHOD2(SetUlpfecConfig,
>@@ -188,13 +176,11 @@ class MockRtpRtcp : public RtpRtcp {
>   MOCK_METHOD1(SetKeyFrameRequestMethod, int32_t(KeyFrameRequestMethod method));
>   MOCK_METHOD0(RequestKeyFrame, int32_t());
>   MOCK_METHOD0(Process, void());
>-  MOCK_METHOD1(RegisterSendFrameCountObserver, void(FrameCountObserver*));
>-  MOCK_CONST_METHOD0(GetSendFrameCountObserver, FrameCountObserver*(void));
>   MOCK_METHOD1(RegisterSendChannelRtpStatisticsCallback,
>                void(StreamDataCountersCallback*));
>   MOCK_CONST_METHOD0(GetSendChannelRtpStatisticsCallback,
>                      StreamDataCountersCallback*(void));
>-  MOCK_METHOD1(SetVideoBitrateAllocation, void(const BitrateAllocation&));
>+  MOCK_METHOD1(SetVideoBitrateAllocation, void(const VideoBitrateAllocation&));
>   // Members.
>   unsigned int remote_ssrc_;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/byte_io.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/byte_io.h
>index e482452b9fd..955bce746e7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/byte_io.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/byte_io.h
>@@ -11,7 +11,6 @@
> #ifndef MODULES_RTP_RTCP_SOURCE_BYTE_IO_H_
> #define MODULES_RTP_RTCP_SOURCE_BYTE_IO_H_
> 
>-
> // This file contains classes for reading and writing integer types from/to
> // byte array representations. Signed/unsigned, partial (whole byte) sizes,
> // and big/little endian byte order is all supported.
>@@ -35,11 +34,9 @@
> // These classes are implemented as recursive templetizations, inteded to make
> // it easy for the compiler to completely inline the reading/writing.
> 
>-
>+#include <stdint.h>
> #include <limits>
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>-
> namespace webrtc {
> 
> // According to ISO C standard ISO/IEC 9899, section 6.2.6.2 (2), the three
>@@ -129,8 +126,8 @@ class ByteReader<T, B, true> {
>   // two's complement for is used.
>   static T ReinterpretAsSigned(U unsigned_val) {
>     // An unsigned value with only the highest order bit set (ex 0x80).
>-    const U kUnsignedHighestBitMask =
>-        static_cast<U>(1) << ((sizeof(U) * 8) - 1);
>+    const U kUnsignedHighestBitMask = static_cast<U>(1)
>+                                      << ((sizeof(U) * 8) - 1);
>     // A signed value with only the highest bit set. Since this is two's
>     // complement form, we can use the min value from std::numeric_limits.
>     const T kSignedHighestBitMask = std::numeric_limits<T>::min();
>@@ -353,20 +350,16 @@ class ByteReader<T, 8, false> {
>  public:
>   static T ReadBigEndian(const uint8_t* data) {
>     static_assert(sizeof(T) >= 8, kSizeErrorMsg);
>-    return
>-        (Get(data, 0) << 56) | (Get(data, 1) << 48) |
>-        (Get(data, 2) << 40) | (Get(data, 3) << 32) |
>-        (Get(data, 4) << 24) | (Get(data, 5) << 16) |
>-        (Get(data, 6) << 8)  |  Get(data, 7);
>+    return (Get(data, 0) << 56) | (Get(data, 1) << 48) | (Get(data, 2) << 40) |
>+           (Get(data, 3) << 32) | (Get(data, 4) << 24) | (Get(data, 5) << 16) |
>+           (Get(data, 6) << 8) | Get(data, 7);
>   }
> 
>   static T ReadLittleEndian(const uint8_t* data) {
>     static_assert(sizeof(T) >= 8, kSizeErrorMsg);
>-    return
>-         Get(data, 0)        | (Get(data, 1) << 8)  |
>-        (Get(data, 2) << 16) | (Get(data, 3) << 24) |
>-        (Get(data, 4) << 32) | (Get(data, 5) << 40) |
>-        (Get(data, 6) << 48) | (Get(data, 7) << 56);
>+    return Get(data, 0) | (Get(data, 1) << 8) | (Get(data, 2) << 16) |
>+           (Get(data, 3) << 24) | (Get(data, 4) << 32) | (Get(data, 5) << 40) |
>+           (Get(data, 6) << 48) | (Get(data, 7) << 56);
>   }
> 
>  private:
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/byte_io_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/byte_io_unittest.cc
>index 3931a04fe3b..20b66362fc2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/byte_io_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/byte_io_unittest.cc
>@@ -18,8 +18,8 @@ namespace {
> 
> class ByteIoTest : public ::testing::Test {
>  protected:
>-  ByteIoTest() {}
>-  virtual ~ByteIoTest() {}
>+  ByteIoTest() = default;
>+  ~ByteIoTest() override = default;
> 
>   enum { kAlignments = sizeof(uint64_t) - 1 };
> 
>@@ -127,10 +127,10 @@ class ByteIoTest : public ::testing::Test {
> };
> 
> TEST_F(ByteIoTest, Test16UBitBigEndian) {
>-  TestRead<uint16_t, ByteReader<uint16_t>::ReadBigEndian,
>-      sizeof(uint16_t)>(true);
>-  TestWrite<uint16_t, ByteWriter<uint16_t>::WriteBigEndian,
>-    sizeof(uint16_t)>(true);
>+  TestRead<uint16_t, ByteReader<uint16_t>::ReadBigEndian, sizeof(uint16_t)>(
>+      true);
>+  TestWrite<uint16_t, ByteWriter<uint16_t>::WriteBigEndian, sizeof(uint16_t)>(
>+      true);
> }
> 
> TEST_F(ByteIoTest, Test24UBitBigEndian) {
>@@ -139,24 +139,23 @@ TEST_F(ByteIoTest, Test24UBitBigEndian) {
> }
> 
> TEST_F(ByteIoTest, Test32UBitBigEndian) {
>-  TestRead<uint32_t, ByteReader<uint32_t>::ReadBigEndian,
>-      sizeof(uint32_t)>(true);
>-  TestWrite<uint32_t, ByteWriter<uint32_t>::WriteBigEndian,
>-      sizeof(uint32_t)>(true);
>+  TestRead<uint32_t, ByteReader<uint32_t>::ReadBigEndian, sizeof(uint32_t)>(
>+      true);
>+  TestWrite<uint32_t, ByteWriter<uint32_t>::WriteBigEndian, sizeof(uint32_t)>(
>+      true);
> }
> 
> TEST_F(ByteIoTest, Test64UBitBigEndian) {
>-  TestRead<uint64_t, ByteReader<uint64_t>::ReadBigEndian,
>-      sizeof(uint64_t)>(true);
>-  TestWrite<uint64_t, ByteWriter<uint64_t>::WriteBigEndian,
>-      sizeof(uint64_t)>(true);
>+  TestRead<uint64_t, ByteReader<uint64_t>::ReadBigEndian, sizeof(uint64_t)>(
>+      true);
>+  TestWrite<uint64_t, ByteWriter<uint64_t>::WriteBigEndian, sizeof(uint64_t)>(
>+      true);
> }
> 
> TEST_F(ByteIoTest, Test16SBitBigEndian) {
>-  TestRead<int16_t, ByteReader<int16_t>::ReadBigEndian,
>-      sizeof(int16_t)>(true);
>-  TestWrite<int16_t, ByteWriter<int16_t>::WriteBigEndian,
>-      sizeof(int16_t)>(true);
>+  TestRead<int16_t, ByteReader<int16_t>::ReadBigEndian, sizeof(int16_t)>(true);
>+  TestWrite<int16_t, ByteWriter<int16_t>::WriteBigEndian, sizeof(int16_t)>(
>+      true);
> }
> 
> TEST_F(ByteIoTest, Test24SBitBigEndian) {
>@@ -165,24 +164,22 @@ TEST_F(ByteIoTest, Test24SBitBigEndian) {
> }
> 
> TEST_F(ByteIoTest, Test32SBitBigEndian) {
>-  TestRead<int32_t, ByteReader<int32_t>::ReadBigEndian,
>-      sizeof(int32_t)>(true);
>-  TestWrite<int32_t, ByteWriter<int32_t>::WriteBigEndian,
>-      sizeof(int32_t)>(true);
>+  TestRead<int32_t, ByteReader<int32_t>::ReadBigEndian, sizeof(int32_t)>(true);
>+  TestWrite<int32_t, ByteWriter<int32_t>::WriteBigEndian, sizeof(int32_t)>(
>+      true);
> }
> 
> TEST_F(ByteIoTest, Test64SBitBigEndian) {
>-  TestRead<int64_t, ByteReader<int64_t>::ReadBigEndian,
>-      sizeof(int64_t)>(true);
>-  TestWrite<int64_t, ByteWriter<int64_t>::WriteBigEndian,
>-      sizeof(int64_t)>(true);
>+  TestRead<int64_t, ByteReader<int64_t>::ReadBigEndian, sizeof(int64_t)>(true);
>+  TestWrite<int64_t, ByteWriter<int64_t>::WriteBigEndian, sizeof(int64_t)>(
>+      true);
> }
> 
> TEST_F(ByteIoTest, Test16UBitLittleEndian) {
>-  TestRead<uint16_t, ByteReader<uint16_t>::ReadLittleEndian,
>-      sizeof(uint16_t)>(false);
>+  TestRead<uint16_t, ByteReader<uint16_t>::ReadLittleEndian, sizeof(uint16_t)>(
>+      false);
>   TestWrite<uint16_t, ByteWriter<uint16_t>::WriteLittleEndian,
>-      sizeof(uint16_t)>(false);
>+            sizeof(uint16_t)>(false);
> }
> 
> TEST_F(ByteIoTest, Test24UBitLittleEndian) {
>@@ -191,24 +188,24 @@ TEST_F(ByteIoTest, Test24UBitLittleEndian) {
> }
> 
> TEST_F(ByteIoTest, Test32UBitLittleEndian) {
>-  TestRead<uint32_t, ByteReader<uint32_t>::ReadLittleEndian,
>-      sizeof(uint32_t)>(false);
>+  TestRead<uint32_t, ByteReader<uint32_t>::ReadLittleEndian, sizeof(uint32_t)>(
>+      false);
>   TestWrite<uint32_t, ByteWriter<uint32_t>::WriteLittleEndian,
>-      sizeof(uint32_t)>(false);
>+            sizeof(uint32_t)>(false);
> }
> 
> TEST_F(ByteIoTest, Test64UBitLittleEndian) {
>-  TestRead<uint64_t, ByteReader<uint64_t>::ReadLittleEndian,
>-      sizeof(uint64_t)>(false);
>+  TestRead<uint64_t, ByteReader<uint64_t>::ReadLittleEndian, sizeof(uint64_t)>(
>+      false);
>   TestWrite<uint64_t, ByteWriter<uint64_t>::WriteLittleEndian,
>-      sizeof(uint64_t)>(false);
>+            sizeof(uint64_t)>(false);
> }
> 
> TEST_F(ByteIoTest, Test16SBitLittleEndian) {
>-  TestRead<int16_t, ByteReader<int16_t>::ReadLittleEndian,
>-      sizeof(int16_t)>(false);
>-  TestWrite<int16_t, ByteWriter<int16_t>::WriteLittleEndian,
>-      sizeof(int16_t)>(false);
>+  TestRead<int16_t, ByteReader<int16_t>::ReadLittleEndian, sizeof(int16_t)>(
>+      false);
>+  TestWrite<int16_t, ByteWriter<int16_t>::WriteLittleEndian, sizeof(int16_t)>(
>+      false);
> }
> 
> TEST_F(ByteIoTest, Test24SBitLittleEndian) {
>@@ -217,17 +214,17 @@ TEST_F(ByteIoTest, Test24SBitLittleEndian) {
> }
> 
> TEST_F(ByteIoTest, Test32SBitLittleEndian) {
>-  TestRead<int32_t, ByteReader<int32_t>::ReadLittleEndian,
>-      sizeof(int32_t)>(false);
>-  TestWrite<int32_t, ByteWriter<int32_t>::WriteLittleEndian,
>-      sizeof(int32_t)>(false);
>+  TestRead<int32_t, ByteReader<int32_t>::ReadLittleEndian, sizeof(int32_t)>(
>+      false);
>+  TestWrite<int32_t, ByteWriter<int32_t>::WriteLittleEndian, sizeof(int32_t)>(
>+      false);
> }
> 
> TEST_F(ByteIoTest, Test64SBitLittleEndian) {
>-  TestRead<int64_t, ByteReader<int64_t>::ReadLittleEndian,
>-      sizeof(int64_t)>(false);
>-  TestWrite<int64_t, ByteWriter<int64_t>::WriteLittleEndian,
>-      sizeof(int64_t)>(false);
>+  TestRead<int64_t, ByteReader<int64_t>::ReadLittleEndian, sizeof(int64_t)>(
>+      false);
>+  TestWrite<int64_t, ByteWriter<int64_t>::WriteLittleEndian, sizeof(int64_t)>(
>+      false);
> }
> 
> // Sets up a fixed byte array and converts N bytes from the array into a
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/contributing_sources.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/contributing_sources.cc
>new file mode 100644
>index 00000000000..853706c4f5d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/contributing_sources.cc
>@@ -0,0 +1,68 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/rtp_rtcp/source/contributing_sources.h"
>+
>+namespace webrtc {
>+
>+namespace {
>+
>+// Allow some stale records to accumulate before cleaning.
>+constexpr int64_t kPruningIntervalMs = 15 * rtc::kNumMillisecsPerSec;
>+
>+}  // namespace
>+
>+constexpr int64_t ContributingSources::kHistoryMs;
>+
>+ContributingSources::ContributingSources() = default;
>+ContributingSources::~ContributingSources() = default;
>+
>+void ContributingSources::Update(int64_t now_ms,
>+                                 rtc::ArrayView<const uint32_t> csrcs) {
>+  for (uint32_t csrc : csrcs) {
>+    last_seen_ms_[csrc] = now_ms;
>+  }
>+  if (!next_pruning_ms_) {
>+    next_pruning_ms_ = now_ms + kPruningIntervalMs;
>+  } else if (now_ms > next_pruning_ms_) {
>+    // To prevent unlimited growth, prune it every 15 seconds.
>+    DeleteOldEntries(now_ms);
>+  }
>+}
>+
>+// Return contributing sources seen the last 10 s.
>+// TODO(nisse): It would be more efficient to delete any stale entries while
>+// iterating over the mapping, but then we'd have to make the method
>+// non-const.
>+std::vector<RtpSource> ContributingSources::GetSources(int64_t now_ms) const {
>+  std::vector<RtpSource> sources;
>+  for (auto& record : last_seen_ms_) {
>+    if (record.second >= now_ms - kHistoryMs) {
>+      sources.emplace_back(record.second, record.first, RtpSourceType::CSRC);
>+    }
>+  }
>+
>+  return sources;
>+}
>+
>+// Delete stale entries.
>+void ContributingSources::DeleteOldEntries(int64_t now_ms) {
>+  for (auto it = last_seen_ms_.begin(); it != last_seen_ms_.end();) {
>+    if (it->second >= now_ms - kHistoryMs) {
>+      // Still relevant.
>+      ++it;
>+    } else {
>+      it = last_seen_ms_.erase(it);
>+    }
>+  }
>+  next_pruning_ms_ = now_ms + kPruningIntervalMs;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/contributing_sources.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/contributing_sources.h
>new file mode 100644
>index 00000000000..1a4a572d8f5
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/contributing_sources.h
>@@ -0,0 +1,51 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_RTP_RTCP_SOURCE_CONTRIBUTING_SOURCES_H_
>+#define MODULES_RTP_RTCP_SOURCE_CONTRIBUTING_SOURCES_H_
>+
>+#include <stdint.h>
>+
>+#include <map>
>+#include <vector>
>+
>+#include "absl/types/optional.h"
>+#include "api/array_view.h"
>+#include "api/rtpreceiverinterface.h"
>+
>+namespace webrtc {
>+
>+class ContributingSources {
>+ public:
>+  // Set by the spec, see
>+  // https://www.w3.org/TR/webrtc/#dom-rtcrtpreceiver-getcontributingsources
>+  static constexpr int64_t kHistoryMs = 10 * rtc::kNumMillisecsPerSec;
>+
>+  ContributingSources();
>+  ~ContributingSources();
>+
>+  // TODO(bugs.webrtc.org/3333): Needs to be extended with audio-level, to
>+  // support RFC6465.
>+  void Update(int64_t now_ms, rtc::ArrayView<const uint32_t> csrcs);
>+
>+  // Returns contributing sources seen the last 10 s.
>+  std::vector<RtpSource> GetSources(int64_t now_ms) const;
>+
>+ private:
>+  void DeleteOldEntries(int64_t now_ms);
>+
>+  // Indexed by csrc.
>+  std::map<uint32_t, int64_t> last_seen_ms_;
>+  absl::optional<int64_t> next_pruning_ms_;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_RTP_RTCP_SOURCE_CONTRIBUTING_SOURCES_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/contributing_sources_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/contributing_sources_unittest.cc
>new file mode 100644
>index 00000000000..8b22d26c948
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/contributing_sources_unittest.cc
>@@ -0,0 +1,111 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/rtp_rtcp/source/contributing_sources.h"
>+
>+#include "rtc_base/timeutils.h"
>+
>+#include "test/gmock.h"
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+namespace {
>+
>+using ::testing::UnorderedElementsAre;
>+
>+constexpr uint32_t kCsrc1 = 111;
>+constexpr uint32_t kCsrc2 = 222;
>+constexpr uint32_t kCsrc3 = 333;
>+
>+}  // namespace
>+
>+TEST(ContributingSourcesTest, RecordSources) {
>+  ContributingSources csrcs;
>+  constexpr uint32_t kCsrcs[] = {kCsrc1, kCsrc2};
>+  constexpr int64_t kTime1 = 10;
>+  csrcs.Update(kTime1, kCsrcs);
>+  EXPECT_THAT(
>+      csrcs.GetSources(kTime1),
>+      UnorderedElementsAre(RtpSource(kTime1, kCsrc1, RtpSourceType::CSRC),
>+                           RtpSource(kTime1, kCsrc2, RtpSourceType::CSRC)));
>+}
>+
>+TEST(ContributingSourcesTest, UpdateSources) {
>+  ContributingSources csrcs;
>+  // TODO(nisse): When migrating to absl::Span, the named constant arrays should
>+  // be replaced by unnamed literals where they are passed to csrcs.Update(...).
>+  constexpr uint32_t kCsrcs1[] = {kCsrc1, kCsrc2};
>+  constexpr uint32_t kCsrcs2[] = {kCsrc2, kCsrc3};
>+  constexpr int64_t kTime1 = 10;
>+  constexpr int64_t kTime2 = kTime1 + 5 * rtc::kNumMillisecsPerSec;
>+  csrcs.Update(kTime1, kCsrcs1);
>+  EXPECT_THAT(
>+      csrcs.GetSources(kTime1),
>+      UnorderedElementsAre(RtpSource(kTime1, kCsrc1, RtpSourceType::CSRC),
>+                           RtpSource(kTime1, kCsrc2, RtpSourceType::CSRC)));
>+  csrcs.Update(kTime2, kCsrcs2);
>+  EXPECT_THAT(
>+      csrcs.GetSources(kTime2),
>+      UnorderedElementsAre(RtpSource(kTime1, kCsrc1, RtpSourceType::CSRC),
>+                           RtpSource(kTime2, kCsrc2, RtpSourceType::CSRC),
>+                           RtpSource(kTime2, kCsrc3, RtpSourceType::CSRC)));
>+}
>+
>+TEST(ContributingSourcesTest, ReturnRecentOnly) {
>+  ContributingSources csrcs;
>+  constexpr uint32_t kCsrcs1[] = {kCsrc1, kCsrc2};
>+  constexpr uint32_t kCsrcs2[] = {kCsrc2, kCsrc3};
>+  constexpr int64_t kTime1 = 10;
>+  constexpr int64_t kTime2 = kTime1 + 5 * rtc::kNumMillisecsPerSec;
>+  constexpr int64_t kTime3 = kTime1 + 12 * rtc::kNumMillisecsPerSec;
>+  csrcs.Update(kTime1, kCsrcs1);
>+  EXPECT_THAT(
>+      csrcs.GetSources(kTime1),
>+      UnorderedElementsAre(RtpSource(kTime1, kCsrc1, RtpSourceType::CSRC),
>+                           RtpSource(kTime1, kCsrc2, RtpSourceType::CSRC)));
>+  csrcs.Update(kTime2, kCsrcs2);
>+  EXPECT_THAT(
>+      csrcs.GetSources(kTime3),
>+      UnorderedElementsAre(RtpSource(kTime2, kCsrc2, RtpSourceType::CSRC),
>+                           RtpSource(kTime2, kCsrc3, RtpSourceType::CSRC)));
>+}
>+
>+TEST(ContributingSourcesTest, PurgeOldSources) {
>+  ContributingSources csrcs;
>+  constexpr uint32_t kCsrcs1[] = {kCsrc1, kCsrc2};
>+  constexpr uint32_t kCsrcs2[] = {kCsrc2, kCsrc3};
>+  constexpr int64_t kTime1 = 10;
>+  constexpr int64_t kTime2 = kTime1 + 10 * rtc::kNumMillisecsPerSec;
>+  constexpr int64_t kTime3 = kTime1 + 20 * rtc::kNumMillisecsPerSec;
>+  csrcs.Update(kTime1, kCsrcs1);
>+  EXPECT_THAT(
>+      csrcs.GetSources(kTime2),
>+      UnorderedElementsAre(RtpSource(kTime1, kCsrc1, RtpSourceType::CSRC),
>+                           RtpSource(kTime1, kCsrc2, RtpSourceType::CSRC)));
>+  csrcs.Update(kTime2, kCsrcs2);
>+  EXPECT_THAT(
>+      csrcs.GetSources(kTime2),
>+      UnorderedElementsAre(RtpSource(kTime1, kCsrc1, RtpSourceType::CSRC),
>+                           RtpSource(kTime2, kCsrc2, RtpSourceType::CSRC),
>+                           RtpSource(kTime2, kCsrc3, RtpSourceType::CSRC)));
>+  csrcs.Update(kTime3, kCsrcs2);
>+  EXPECT_THAT(
>+      csrcs.GetSources(kTime3),
>+      UnorderedElementsAre(RtpSource(kTime3, kCsrc2, RtpSourceType::CSRC),
>+                           RtpSource(kTime3, kCsrc3, RtpSourceType::CSRC)));
>+  // Query at an earlier time; check that old sources really have been purged
>+  // and don't reappear.
>+  EXPECT_THAT(
>+      csrcs.GetSources(kTime2),
>+      UnorderedElementsAre(RtpSource(kTime3, kCsrc2, RtpSourceType::CSRC),
>+                           RtpSource(kTime3, kCsrc3, RtpSourceType::CSRC)));
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/fec_private_tables_bursty.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/fec_private_tables_bursty.cc
>new file mode 100644
>index 00000000000..9dbc012368b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/fec_private_tables_bursty.cc
>@@ -0,0 +1,660 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/rtp_rtcp/source/fec_private_tables_bursty.h"
>+
>+namespace {
>+// clang-format off
>+#define kMaskBursty1_1 \
>+  0x80, 0x00
>+
>+#define kMaskBursty2_1 \
>+  0xc0, 0x00
>+
>+#define kMaskBursty2_2 \
>+  0x80, 0x00, \
>+  0xc0, 0x00
>+
>+#define kMaskBursty3_1 \
>+  0xe0, 0x00
>+
>+#define kMaskBursty3_2 \
>+  0xc0, 0x00, \
>+  0xa0, 0x00
>+
>+#define kMaskBursty3_3 \
>+  0x80, 0x00, \
>+  0xc0, 0x00, \
>+  0x60, 0x00
>+
>+#define kMaskBursty4_1 \
>+  0xf0, 0x00
>+
>+#define kMaskBursty4_2 \
>+  0xa0, 0x00, \
>+  0xd0, 0x00
>+
>+#define kMaskBursty4_3 \
>+  0xc0, 0x00, \
>+  0x60, 0x00, \
>+  0x90, 0x00
>+
>+#define kMaskBursty4_4 \
>+  0x80, 0x00, \
>+  0xc0, 0x00, \
>+  0x60, 0x00, \
>+  0x30, 0x00
>+
>+#define kMaskBursty5_1 \
>+  0xf8, 0x00
>+
>+#define kMaskBursty5_2 \
>+  0xd0, 0x00, \
>+  0xa8, 0x00
>+
>+#define kMaskBursty5_3 \
>+  0x70, 0x00, \
>+  0x90, 0x00, \
>+  0xc8, 0x00
>+
>+#define kMaskBursty5_4 \
>+  0xc0, 0x00, \
>+  0x60, 0x00, \
>+  0x30, 0x00, \
>+  0x88, 0x00
>+
>+#define kMaskBursty5_5 \
>+  0x80, 0x00, \
>+  0xc0, 0x00, \
>+  0x60, 0x00, \
>+  0x30, 0x00, \
>+  0x18, 0x00
>+
>+#define kMaskBursty6_1 \
>+  0xfc, 0x00
>+
>+#define kMaskBursty6_2 \
>+  0xa8, 0x00, \
>+  0xd4, 0x00
>+
>+#define kMaskBursty6_3 \
>+  0x94, 0x00, \
>+  0xc8, 0x00, \
>+  0x64, 0x00
>+
>+#define kMaskBursty6_4 \
>+  0x60, 0x00, \
>+  0x38, 0x00, \
>+  0x88, 0x00, \
>+  0xc4, 0x00
>+
>+#define kMaskBursty6_5 \
>+  0xc0, 0x00, \
>+  0x60, 0x00, \
>+  0x30, 0x00, \
>+  0x18, 0x00, \
>+  0x84, 0x00
>+
>+#define kMaskBursty6_6 \
>+  0x80, 0x00, \
>+  0xc0, 0x00, \
>+  0x60, 0x00, \
>+  0x30, 0x00, \
>+  0x18, 0x00, \
>+  0x0c, 0x00
>+
>+#define kMaskBursty7_1 \
>+  0xfe, 0x00
>+
>+#define kMaskBursty7_2 \
>+  0xd4, 0x00, \
>+  0xaa, 0x00
>+
>+#define kMaskBursty7_3 \
>+  0xc8, 0x00, \
>+  0x74, 0x00, \
>+  0x92, 0x00
>+
>+#define kMaskBursty7_4 \
>+  0x38, 0x00, \
>+  0x8a, 0x00, \
>+  0xc4, 0x00, \
>+  0x62, 0x00
>+
>+#define kMaskBursty7_5 \
>+  0x60, 0x00, \
>+  0x30, 0x00, \
>+  0x1c, 0x00, \
>+  0x84, 0x00, \
>+  0xc2, 0x00
>+
>+#define kMaskBursty7_6 \
>+  0xc0, 0x00, \
>+  0x60, 0x00, \
>+  0x30, 0x00, \
>+  0x18, 0x00, \
>+  0x0c, 0x00, \
>+  0x82, 0x00
>+
>+#define kMaskBursty7_7 \
>+  0x80, 0x00, \
>+  0xc0, 0x00, \
>+  0x60, 0x00, \
>+  0x30, 0x00, \
>+  0x18, 0x00, \
>+  0x0c, 0x00, \
>+  0x06, 0x00
>+
>+#define kMaskBursty8_1 \
>+  0xff, 0x00
>+
>+#define kMaskBursty8_2 \
>+  0xaa, 0x00, \
>+  0xd5, 0x00
>+
>+#define kMaskBursty8_3 \
>+  0x74, 0x00, \
>+  0x92, 0x00, \
>+  0xc9, 0x00
>+
>+#define kMaskBursty8_4 \
>+  0x8a, 0x00, \
>+  0xc5, 0x00, \
>+  0x62, 0x00, \
>+  0x31, 0x00
>+
>+#define kMaskBursty8_5 \
>+  0x30, 0x00, \
>+  0x1c, 0x00, \
>+  0x85, 0x00, \
>+  0xc2, 0x00, \
>+  0x61, 0x00
>+
>+#define kMaskBursty8_6 \
>+  0x60, 0x00, \
>+  0x30, 0x00, \
>+  0x18, 0x00, \
>+  0x0e, 0x00, \
>+  0x82, 0x00, \
>+  0xc1, 0x00
>+
>+#define kMaskBursty8_7 \
>+  0xc0, 0x00, \
>+  0x60, 0x00, \
>+  0x30, 0x00, \
>+  0x18, 0x00, \
>+  0x0c, 0x00, \
>+  0x06, 0x00, \
>+  0x81, 0x00
>+
>+#define kMaskBursty8_8 \
>+  0x80, 0x00, \
>+  0xc0, 0x00, \
>+  0x60, 0x00, \
>+  0x30, 0x00, \
>+  0x18, 0x00, \
>+  0x0c, 0x00, \
>+  0x06, 0x00, \
>+  0x03, 0x00
>+
>+#define kMaskBursty9_1 \
>+  0xff, 0x80
>+
>+#define kMaskBursty9_2 \
>+  0xd5, 0x00, \
>+  0xaa, 0x80
>+
>+#define kMaskBursty9_3 \
>+  0x92, 0x00, \
>+  0xc9, 0x00, \
>+  0x74, 0x80
>+
>+#define kMaskBursty9_4 \
>+  0xc5, 0x00, \
>+  0x62, 0x00, \
>+  0x39, 0x00, \
>+  0x8a, 0x80
>+
>+#define kMaskBursty9_5 \
>+  0x1c, 0x00, \
>+  0x85, 0x00, \
>+  0xc2, 0x80, \
>+  0x61, 0x00, \
>+  0x30, 0x80
>+
>+#define kMaskBursty9_6 \
>+  0x30, 0x00, \
>+  0x18, 0x00, \
>+  0x0e, 0x00, \
>+  0x82, 0x80, \
>+  0xc1, 0x00, \
>+  0x60, 0x80
>+
>+#define kMaskBursty9_7 \
>+  0x60, 0x00, \
>+  0x30, 0x00, \
>+  0x18, 0x00, \
>+  0x0c, 0x00, \
>+  0x07, 0x00, \
>+  0x81, 0x00, \
>+  0xc0, 0x80
>+
>+#define kMaskBursty9_8 \
>+  0xc0, 0x00, \
>+  0x60, 0x00, \
>+  0x30, 0x00, \
>+  0x18, 0x00, \
>+  0x0c, 0x00, \
>+  0x06, 0x00, \
>+  0x03, 0x00, \
>+  0x80, 0x80
>+
>+#define kMaskBursty9_9 \
>+  0x80, 0x00, \
>+  0xc0, 0x00, \
>+  0x60, 0x00, \
>+  0x30, 0x00, \
>+  0x18, 0x00, \
>+  0x0c, 0x00, \
>+  0x06, 0x00, \
>+  0x03, 0x00, \
>+  0x01, 0x80
>+
>+#define kMaskBursty10_1 \
>+  0xff, 0xc0
>+
>+#define kMaskBursty10_2 \
>+  0xaa, 0x80, \
>+  0xd5, 0x40
>+
>+#define kMaskBursty10_3 \
>+  0xc9, 0x00, \
>+  0x74, 0x80, \
>+  0x92, 0x40
>+
>+#define kMaskBursty10_4 \
>+  0x62, 0x00, \
>+  0x39, 0x00, \
>+  0x8a, 0x80, \
>+  0xc5, 0x40
>+
>+#define kMaskBursty10_5 \
>+  0x85, 0x00, \
>+  0xc2, 0x80, \
>+  0x61, 0x40, \
>+  0x30, 0x80, \
>+  0x18, 0x40
>+
>+#define kMaskBursty10_6 \
>+  0x18, 0x00, \
>+  0x0e, 0x00, \
>+  0x82, 0x80, \
>+  0xc1, 0x40, \
>+  0x60, 0x80, \
>+  0x30, 0x40
>+
>+#define kMaskBursty10_7 \
>+  0x30, 0x00, \
>+  0x18, 0x00, \
>+  0x0c, 0x00, \
>+  0x07, 0x00, \
>+  0x81, 0x40, \
>+  0xc0, 0x80, \
>+  0x60, 0x40
>+
>+#define kMaskBursty10_8 \
>+  0x60, 0x00, \
>+  0x30, 0x00, \
>+  0x18, 0x00, \
>+  0x0c, 0x00, \
>+  0x06, 0x00, \
>+  0x03, 0x00, \
>+  0x80, 0x80, \
>+  0xc0, 0x40
>+
>+#define kMaskBursty10_9 \
>+  0xc0, 0x00, \
>+  0x60, 0x00, \
>+  0x30, 0x00, \
>+  0x18, 0x00, \
>+  0x0c, 0x00, \
>+  0x06, 0x00, \
>+  0x03, 0x00, \
>+  0x01, 0x80, \
>+  0x80, 0x40
>+
>+#define kMaskBursty10_10 \
>+  0x80, 0x00, \
>+  0xc0, 0x00, \
>+  0x60, 0x00, \
>+  0x30, 0x00, \
>+  0x18, 0x00, \
>+  0x0c, 0x00, \
>+  0x06, 0x00, \
>+  0x03, 0x00, \
>+  0x01, 0x80, \
>+  0x00, 0xc0
>+
>+#define kMaskBursty11_1 \
>+  0xff, 0xe0
>+
>+#define kMaskBursty11_2 \
>+  0xd5, 0x40, \
>+  0xaa, 0xa0
>+
>+#define kMaskBursty11_3 \
>+  0x74, 0x80, \
>+  0x92, 0x40, \
>+  0xc9, 0x20
>+
>+#define kMaskBursty11_4 \
>+  0x39, 0x00, \
>+  0x8a, 0x80, \
>+  0xc5, 0x40, \
>+  0x62, 0x20
>+
>+#define kMaskBursty11_5 \
>+  0xc2, 0xc0, \
>+  0x61, 0x00, \
>+  0x30, 0xa0, \
>+  0x1c, 0x40, \
>+  0x85, 0x20
>+
>+#define kMaskBursty11_6 \
>+  0x0e, 0x00, \
>+  0x82, 0x80, \
>+  0xc1, 0x40, \
>+  0x60, 0xa0, \
>+  0x30, 0x40, \
>+  0x18, 0x20
>+
>+#define kMaskBursty11_7 \
>+  0x18, 0x00, \
>+  0x0c, 0x00, \
>+  0x07, 0x00, \
>+  0x81, 0x40, \
>+  0xc0, 0xa0, \
>+  0x60, 0x40, \
>+  0x30, 0x20
>+
>+#define kMaskBursty11_8 \
>+  0x30, 0x00, \
>+  0x18, 0x00, \
>+  0x0c, 0x00, \
>+  0x06, 0x00, \
>+  0x03, 0x40, \
>+  0x80, 0xa0, \
>+  0xc0, 0x40, \
>+  0x60, 0x20
>+
>+#define kMaskBursty11_9 \
>+  0x60, 0x00, \
>+  0x30, 0x00, \
>+  0x18, 0x00, \
>+  0x0c, 0x00, \
>+  0x06, 0x00, \
>+  0x03, 0x00, \
>+  0x01, 0x80, \
>+  0x80, 0x40, \
>+  0xc0, 0x20
>+
>+#define kMaskBursty11_10 \
>+  0xc0, 0x00, \
>+  0x60, 0x00, \
>+  0x30, 0x00, \
>+  0x18, 0x00, \
>+  0x0c, 0x00, \
>+  0x06, 0x00, \
>+  0x03, 0x00, \
>+  0x01, 0x80, \
>+  0x00, 0xc0, \
>+  0x80, 0x20
>+
>+#define kMaskBursty11_11 \
>+  0x80, 0x00, \
>+  0xc0, 0x00, \
>+  0x60, 0x00, \
>+  0x30, 0x00, \
>+  0x18, 0x00, \
>+  0x0c, 0x00, \
>+  0x06, 0x00, \
>+  0x03, 0x00, \
>+  0x01, 0x80, \
>+  0x00, 0xc0, \
>+  0x00, 0x60
>+
>+#define kMaskBursty12_1 \
>+  0xff, 0xf0
>+
>+#define kMaskBursty12_2 \
>+  0xaa, 0xa0, \
>+  0xd5, 0x50
>+
>+#define kMaskBursty12_3 \
>+  0x92, 0x40, \
>+  0xc9, 0x20, \
>+  0x74, 0x90
>+
>+#define kMaskBursty12_4 \
>+  0x8a, 0x80, \
>+  0xc5, 0x40, \
>+  0x62, 0x20, \
>+  0x39, 0x10
>+
>+#define kMaskBursty12_5 \
>+  0x61, 0x00, \
>+  0x30, 0xa0, \
>+  0x1c, 0x50, \
>+  0x85, 0x20, \
>+  0xc2, 0x90
>+
>+#define kMaskBursty12_6 \
>+  0x82, 0x90, \
>+  0xc1, 0x40, \
>+  0x60, 0xa0, \
>+  0x30, 0x50, \
>+  0x18, 0x20, \
>+  0x0c, 0x10
>+
>+#define kMaskBursty12_7 \
>+  0x0c, 0x00, \
>+  0x07, 0x00, \
>+  0x81, 0x40, \
>+  0xc0, 0xa0, \
>+  0x60, 0x50, \
>+  0x30, 0x20, \
>+  0x18, 0x10
>+
>+#define kMaskBursty12_8 \
>+  0x18, 0x00, \
>+  0x0c, 0x00, \
>+  0x06, 0x00, \
>+  0x03, 0x00, \
>+  0x80, 0xa0, \
>+  0xc0, 0x50, \
>+  0x60, 0x20, \
>+  0x30, 0x10
>+
>+#define kMaskBursty12_9 \
>+  0x30, 0x00, \
>+  0x18, 0x00, \
>+  0x0c, 0x00, \
>+  0x06, 0x00, \
>+  0x03, 0x00, \
>+  0x01, 0x80, \
>+  0x80, 0x50, \
>+  0xc0, 0x20, \
>+  0x60, 0x10
>+
>+#define kMaskBursty12_10 \
>+  0x60, 0x00, \
>+  0x30, 0x00, \
>+  0x18, 0x00, \
>+  0x0c, 0x00, \
>+  0x06, 0x00, \
>+  0x03, 0x00, \
>+  0x01, 0x80, \
>+  0x00, 0xc0, \
>+  0x80, 0x20, \
>+  0xc0, 0x10
>+
>+#define kMaskBursty12_11 \
>+  0xc0, 0x00, \
>+  0x60, 0x00, \
>+  0x30, 0x00, \
>+  0x18, 0x00, \
>+  0x0c, 0x00, \
>+  0x06, 0x00, \
>+  0x03, 0x00, \
>+  0x01, 0x80, \
>+  0x00, 0xc0, \
>+  0x00, 0x60, \
>+  0x80, 0x10
>+
>+#define kMaskBursty12_12 \
>+  0x80, 0x00, \
>+  0xc0, 0x00, \
>+  0x60, 0x00, \
>+  0x30, 0x00, \
>+  0x18, 0x00, \
>+  0x0c, 0x00, \
>+  0x06, 0x00, \
>+  0x03, 0x00, \
>+  0x01, 0x80, \
>+  0x00, 0xc0, \
>+  0x00, 0x60, \
>+  0x00, 0x30
>+
>+#define kPacketMaskBursty1 1, \
>+  kMaskBursty1_1
>+
>+#define kPacketMaskBursty2 2, \
>+  kMaskBursty2_1, \
>+  kMaskBursty2_2
>+
>+#define kPacketMaskBursty3 3, \
>+  kMaskBursty3_1, \
>+  kMaskBursty3_2, \
>+  kMaskBursty3_3
>+
>+#define kPacketMaskBursty4 4, \
>+  kMaskBursty4_1, \
>+  kMaskBursty4_2, \
>+  kMaskBursty4_3, \
>+  kMaskBursty4_4
>+
>+#define kPacketMaskBursty5 5, \
>+  kMaskBursty5_1, \
>+  kMaskBursty5_2, \
>+  kMaskBursty5_3, \
>+  kMaskBursty5_4, \
>+  kMaskBursty5_5
>+
>+#define kPacketMaskBursty6 6, \
>+  kMaskBursty6_1, \
>+  kMaskBursty6_2, \
>+  kMaskBursty6_3, \
>+  kMaskBursty6_4, \
>+  kMaskBursty6_5, \
>+  kMaskBursty6_6
>+
>+#define kPacketMaskBursty7 7, \
>+  kMaskBursty7_1, \
>+  kMaskBursty7_2, \
>+  kMaskBursty7_3, \
>+  kMaskBursty7_4, \
>+  kMaskBursty7_5, \
>+  kMaskBursty7_6, \
>+  kMaskBursty7_7
>+
>+#define kPacketMaskBursty8 8, \
>+  kMaskBursty8_1, \
>+  kMaskBursty8_2, \
>+  kMaskBursty8_3, \
>+  kMaskBursty8_4, \
>+  kMaskBursty8_5, \
>+  kMaskBursty8_6, \
>+  kMaskBursty8_7, \
>+  kMaskBursty8_8
>+
>+#define kPacketMaskBursty9 9, \
>+  kMaskBursty9_1, \
>+  kMaskBursty9_2, \
>+  kMaskBursty9_3, \
>+  kMaskBursty9_4, \
>+  kMaskBursty9_5, \
>+  kMaskBursty9_6, \
>+  kMaskBursty9_7, \
>+  kMaskBursty9_8, \
>+  kMaskBursty9_9
>+
>+#define kPacketMaskBursty10 10, \
>+  kMaskBursty10_1, \
>+  kMaskBursty10_2, \
>+  kMaskBursty10_3, \
>+  kMaskBursty10_4, \
>+  kMaskBursty10_5, \
>+  kMaskBursty10_6, \
>+  kMaskBursty10_7, \
>+  kMaskBursty10_8, \
>+  kMaskBursty10_9, \
>+  kMaskBursty10_10
>+
>+#define kPacketMaskBursty11 11, \
>+  kMaskBursty11_1, \
>+  kMaskBursty11_2, \
>+  kMaskBursty11_3, \
>+  kMaskBursty11_4, \
>+  kMaskBursty11_5, \
>+  kMaskBursty11_6, \
>+  kMaskBursty11_7, \
>+  kMaskBursty11_8, \
>+  kMaskBursty11_9, \
>+  kMaskBursty11_10, \
>+  kMaskBursty11_11
>+
>+#define kPacketMaskBursty12 12, \
>+  kMaskBursty12_1, \
>+  kMaskBursty12_2, \
>+  kMaskBursty12_3, \
>+  kMaskBursty12_4, \
>+  kMaskBursty12_5, \
>+  kMaskBursty12_6, \
>+  kMaskBursty12_7, \
>+  kMaskBursty12_8, \
>+  kMaskBursty12_9, \
>+  kMaskBursty12_10, \
>+  kMaskBursty12_11, \
>+  kMaskBursty12_12
>+
>+// clang-format on
>+}  // namespace
>+
>+namespace webrtc {
>+namespace fec_private_tables {
>+
>+const uint8_t kPacketMaskBurstyTbl[] = {
>+    12,
>+    kPacketMaskBursty1,
>+    kPacketMaskBursty2,
>+    kPacketMaskBursty3,
>+    kPacketMaskBursty4,
>+    kPacketMaskBursty5,
>+    kPacketMaskBursty6,
>+    kPacketMaskBursty7,
>+    kPacketMaskBursty8,
>+    kPacketMaskBursty9,
>+    kPacketMaskBursty10,
>+    kPacketMaskBursty11,
>+    kPacketMaskBursty12,
>+};
>+
>+}  // namespace fec_private_tables
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/fec_private_tables_bursty.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/fec_private_tables_bursty.h
>index 7fba7aa5500..5d67292b939 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/fec_private_tables_bursty.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/fec_private_tables_bursty.h
>@@ -25,737 +25,12 @@
> // (i.e., more packets/symbols in the code, so larger (k,m), i.e.,  k > 4,
> // m > 3).
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include <stdint.h>
> 
> namespace webrtc {
> namespace fec_private_tables {
> 
>-const uint8_t kMaskBursty1_1[2] = {
>-  0x80, 0x00
>-};
>-
>-const uint8_t kMaskBursty2_1[2] = {
>-  0xc0, 0x00
>-};
>-
>-const uint8_t kMaskBursty2_2[4] = {
>-  0x80, 0x00,
>-  0xc0, 0x00
>-};
>-
>-const uint8_t kMaskBursty3_1[2] = {
>-  0xe0, 0x00
>-};
>-
>-const uint8_t kMaskBursty3_2[4] = {
>-  0xc0, 0x00,
>-  0xa0, 0x00
>-};
>-
>-const uint8_t kMaskBursty3_3[6] = {
>-  0x80, 0x00,
>-  0xc0, 0x00,
>-  0x60, 0x00
>-};
>-
>-const uint8_t kMaskBursty4_1[2] = {
>-  0xf0, 0x00
>-};
>-
>-const uint8_t kMaskBursty4_2[4] = {
>-  0xa0, 0x00,
>-  0xd0, 0x00
>-};
>-
>-const uint8_t kMaskBursty4_3[6] = {
>-  0xc0, 0x00,
>-  0x60, 0x00,
>-  0x90, 0x00
>-};
>-
>-const uint8_t kMaskBursty4_4[8] = {
>-  0x80, 0x00,
>-  0xc0, 0x00,
>-  0x60, 0x00,
>-  0x30, 0x00
>-};
>-
>-const uint8_t kMaskBursty5_1[2] = {
>-  0xf8, 0x00
>-};
>-
>-const uint8_t kMaskBursty5_2[4] = {
>-  0xd0, 0x00,
>-  0xa8, 0x00
>-};
>-
>-const uint8_t kMaskBursty5_3[6] = {
>-  0x70, 0x00,
>-  0x90, 0x00,
>-  0xc8, 0x00
>-};
>-
>-const uint8_t kMaskBursty5_4[8] = {
>-  0xc0, 0x00,
>-  0x60, 0x00,
>-  0x30, 0x00,
>-  0x88, 0x00
>-};
>-
>-const uint8_t kMaskBursty5_5[10] = {
>-  0x80, 0x00,
>-  0xc0, 0x00,
>-  0x60, 0x00,
>-  0x30, 0x00,
>-  0x18, 0x00
>-};
>-
>-const uint8_t kMaskBursty6_1[2] = {
>-  0xfc, 0x00
>-};
>-
>-const uint8_t kMaskBursty6_2[4] = {
>-  0xa8, 0x00,
>-  0xd4, 0x00
>-};
>-
>-const uint8_t kMaskBursty6_3[6] = {
>-  0x94, 0x00,
>-  0xc8, 0x00,
>-  0x64, 0x00
>-};
>-
>-const uint8_t kMaskBursty6_4[8] = {
>-  0x60, 0x00,
>-  0x38, 0x00,
>-  0x88, 0x00,
>-  0xc4, 0x00
>-};
>-
>-const uint8_t kMaskBursty6_5[10] = {
>-  0xc0, 0x00,
>-  0x60, 0x00,
>-  0x30, 0x00,
>-  0x18, 0x00,
>-  0x84, 0x00
>-};
>-
>-const uint8_t kMaskBursty6_6[12] = {
>-  0x80, 0x00,
>-  0xc0, 0x00,
>-  0x60, 0x00,
>-  0x30, 0x00,
>-  0x18, 0x00,
>-  0x0c, 0x00
>-};
>-
>-const uint8_t kMaskBursty7_1[2] = {
>-  0xfe, 0x00
>-};
>-
>-const uint8_t kMaskBursty7_2[4] = {
>-  0xd4, 0x00,
>-  0xaa, 0x00
>-};
>-
>-const uint8_t kMaskBursty7_3[6] = {
>-  0xc8, 0x00,
>-  0x74, 0x00,
>-  0x92, 0x00
>-};
>-
>-const uint8_t kMaskBursty7_4[8] = {
>-  0x38, 0x00,
>-  0x8a, 0x00,
>-  0xc4, 0x00,
>-  0x62, 0x00
>-};
>-
>-const uint8_t kMaskBursty7_5[10] = {
>-  0x60, 0x00,
>-  0x30, 0x00,
>-  0x1c, 0x00,
>-  0x84, 0x00,
>-  0xc2, 0x00
>-};
>-
>-const uint8_t kMaskBursty7_6[12] = {
>-  0xc0, 0x00,
>-  0x60, 0x00,
>-  0x30, 0x00,
>-  0x18, 0x00,
>-  0x0c, 0x00,
>-  0x82, 0x00
>-};
>-
>-const uint8_t kMaskBursty7_7[14] = {
>-  0x80, 0x00,
>-  0xc0, 0x00,
>-  0x60, 0x00,
>-  0x30, 0x00,
>-  0x18, 0x00,
>-  0x0c, 0x00,
>-  0x06, 0x00
>-};
>-
>-const uint8_t kMaskBursty8_1[2] = {
>-  0xff, 0x00
>-};
>-
>-const uint8_t kMaskBursty8_2[4] = {
>-  0xaa, 0x00,
>-  0xd5, 0x00
>-};
>-
>-const uint8_t kMaskBursty8_3[6] = {
>-  0x74, 0x00,
>-  0x92, 0x00,
>-  0xc9, 0x00
>-};
>-
>-const uint8_t kMaskBursty8_4[8] = {
>-  0x8a, 0x00,
>-  0xc5, 0x00,
>-  0x62, 0x00,
>-  0x31, 0x00
>-};
>-
>-const uint8_t kMaskBursty8_5[10] = {
>-  0x30, 0x00,
>-  0x1c, 0x00,
>-  0x85, 0x00,
>-  0xc2, 0x00,
>-  0x61, 0x00
>-};
>-
>-const uint8_t kMaskBursty8_6[12] = {
>-  0x60, 0x00,
>-  0x30, 0x00,
>-  0x18, 0x00,
>-  0x0e, 0x00,
>-  0x82, 0x00,
>-  0xc1, 0x00
>-};
>-
>-const uint8_t kMaskBursty8_7[14] = {
>-  0xc0, 0x00,
>-  0x60, 0x00,
>-  0x30, 0x00,
>-  0x18, 0x00,
>-  0x0c, 0x00,
>-  0x06, 0x00,
>-  0x81, 0x00
>-};
>-
>-const uint8_t kMaskBursty8_8[16] = {
>-  0x80, 0x00,
>-  0xc0, 0x00,
>-  0x60, 0x00,
>-  0x30, 0x00,
>-  0x18, 0x00,
>-  0x0c, 0x00,
>-  0x06, 0x00,
>-  0x03, 0x00
>-};
>-
>-const uint8_t kMaskBursty9_1[2] = {
>-  0xff, 0x80
>-};
>-
>-const uint8_t kMaskBursty9_2[4] = {
>-  0xd5, 0x00,
>-  0xaa, 0x80
>-};
>-
>-const uint8_t kMaskBursty9_3[6] = {
>-  0x92, 0x00,
>-  0xc9, 0x00,
>-  0x74, 0x80
>-};
>-
>-const uint8_t kMaskBursty9_4[8] = {
>-  0xc5, 0x00,
>-  0x62, 0x00,
>-  0x39, 0x00,
>-  0x8a, 0x80
>-};
>-
>-const uint8_t kMaskBursty9_5[10] = {
>-  0x1c, 0x00,
>-  0x85, 0x00,
>-  0xc2, 0x80,
>-  0x61, 0x00,
>-  0x30, 0x80
>-};
>-
>-const uint8_t kMaskBursty9_6[12] = {
>-  0x30, 0x00,
>-  0x18, 0x00,
>-  0x0e, 0x00,
>-  0x82, 0x80,
>-  0xc1, 0x00,
>-  0x60, 0x80
>-};
>-
>-const uint8_t kMaskBursty9_7[14] = {
>-  0x60, 0x00,
>-  0x30, 0x00,
>-  0x18, 0x00,
>-  0x0c, 0x00,
>-  0x07, 0x00,
>-  0x81, 0x00,
>-  0xc0, 0x80
>-};
>-
>-const uint8_t kMaskBursty9_8[16] = {
>-  0xc0, 0x00,
>-  0x60, 0x00,
>-  0x30, 0x00,
>-  0x18, 0x00,
>-  0x0c, 0x00,
>-  0x06, 0x00,
>-  0x03, 0x00,
>-  0x80, 0x80
>-};
>-
>-const uint8_t kMaskBursty9_9[18] = {
>-  0x80, 0x00,
>-  0xc0, 0x00,
>-  0x60, 0x00,
>-  0x30, 0x00,
>-  0x18, 0x00,
>-  0x0c, 0x00,
>-  0x06, 0x00,
>-  0x03, 0x00,
>-  0x01, 0x80
>-};
>-
>-const uint8_t kMaskBursty10_1[2] = {
>-  0xff, 0xc0
>-};
>-
>-const uint8_t kMaskBursty10_2[4] = {
>-  0xaa, 0x80,
>-  0xd5, 0x40
>-};
>-
>-const uint8_t kMaskBursty10_3[6] = {
>-  0xc9, 0x00,
>-  0x74, 0x80,
>-  0x92, 0x40
>-};
>-
>-const uint8_t kMaskBursty10_4[8] = {
>-  0x62, 0x00,
>-  0x39, 0x00,
>-  0x8a, 0x80,
>-  0xc5, 0x40
>-};
>-
>-const uint8_t kMaskBursty10_5[10] = {
>-  0x85, 0x00,
>-  0xc2, 0x80,
>-  0x61, 0x40,
>-  0x30, 0x80,
>-  0x18, 0x40
>-};
>-
>-const uint8_t kMaskBursty10_6[12] = {
>-  0x18, 0x00,
>-  0x0e, 0x00,
>-  0x82, 0x80,
>-  0xc1, 0x40,
>-  0x60, 0x80,
>-  0x30, 0x40
>-};
>-
>-const uint8_t kMaskBursty10_7[14] = {
>-  0x30, 0x00,
>-  0x18, 0x00,
>-  0x0c, 0x00,
>-  0x07, 0x00,
>-  0x81, 0x40,
>-  0xc0, 0x80,
>-  0x60, 0x40
>-};
>-
>-const uint8_t kMaskBursty10_8[16] = {
>-  0x60, 0x00,
>-  0x30, 0x00,
>-  0x18, 0x00,
>-  0x0c, 0x00,
>-  0x06, 0x00,
>-  0x03, 0x00,
>-  0x80, 0x80,
>-  0xc0, 0x40
>-};
>-
>-const uint8_t kMaskBursty10_9[18] = {
>-  0xc0, 0x00,
>-  0x60, 0x00,
>-  0x30, 0x00,
>-  0x18, 0x00,
>-  0x0c, 0x00,
>-  0x06, 0x00,
>-  0x03, 0x00,
>-  0x01, 0x80,
>-  0x80, 0x40
>-};
>-
>-const uint8_t kMaskBursty10_10[20] = {
>-  0x80, 0x00,
>-  0xc0, 0x00,
>-  0x60, 0x00,
>-  0x30, 0x00,
>-  0x18, 0x00,
>-  0x0c, 0x00,
>-  0x06, 0x00,
>-  0x03, 0x00,
>-  0x01, 0x80,
>-  0x00, 0xc0
>-};
>-
>-const uint8_t kMaskBursty11_1[2] = {
>-  0xff, 0xe0
>-};
>-
>-const uint8_t kMaskBursty11_2[4] = {
>-  0xd5, 0x40,
>-  0xaa, 0xa0
>-};
>-
>-const uint8_t kMaskBursty11_3[6] = {
>-  0x74, 0x80,
>-  0x92, 0x40,
>-  0xc9, 0x20
>-};
>-
>-const uint8_t kMaskBursty11_4[8] = {
>-  0x39, 0x00,
>-  0x8a, 0x80,
>-  0xc5, 0x40,
>-  0x62, 0x20
>-};
>-
>-const uint8_t kMaskBursty11_5[10] = {
>-  0xc2, 0xc0,
>-  0x61, 0x00,
>-  0x30, 0xa0,
>-  0x1c, 0x40,
>-  0x85, 0x20
>-};
>-
>-const uint8_t kMaskBursty11_6[12] = {
>-  0x0e, 0x00,
>-  0x82, 0x80,
>-  0xc1, 0x40,
>-  0x60, 0xa0,
>-  0x30, 0x40,
>-  0x18, 0x20
>-};
>-
>-const uint8_t kMaskBursty11_7[14] = {
>-  0x18, 0x00,
>-  0x0c, 0x00,
>-  0x07, 0x00,
>-  0x81, 0x40,
>-  0xc0, 0xa0,
>-  0x60, 0x40,
>-  0x30, 0x20
>-};
>-
>-const uint8_t kMaskBursty11_8[16] = {
>-  0x30, 0x00,
>-  0x18, 0x00,
>-  0x0c, 0x00,
>-  0x06, 0x00,
>-  0x03, 0x40,
>-  0x80, 0xa0,
>-  0xc0, 0x40,
>-  0x60, 0x20
>-};
>-
>-const uint8_t kMaskBursty11_9[18] = {
>-  0x60, 0x00,
>-  0x30, 0x00,
>-  0x18, 0x00,
>-  0x0c, 0x00,
>-  0x06, 0x00,
>-  0x03, 0x00,
>-  0x01, 0x80,
>-  0x80, 0x40,
>-  0xc0, 0x20
>-};
>-
>-const uint8_t kMaskBursty11_10[20] = {
>-  0xc0, 0x00,
>-  0x60, 0x00,
>-  0x30, 0x00,
>-  0x18, 0x00,
>-  0x0c, 0x00,
>-  0x06, 0x00,
>-  0x03, 0x00,
>-  0x01, 0x80,
>-  0x00, 0xc0,
>-  0x80, 0x20
>-};
>-
>-const uint8_t kMaskBursty11_11[22] = {
>-  0x80, 0x00,
>-  0xc0, 0x00,
>-  0x60, 0x00,
>-  0x30, 0x00,
>-  0x18, 0x00,
>-  0x0c, 0x00,
>-  0x06, 0x00,
>-  0x03, 0x00,
>-  0x01, 0x80,
>-  0x00, 0xc0,
>-  0x00, 0x60
>-};
>-
>-const uint8_t kMaskBursty12_1[2] = {
>-  0xff, 0xf0
>-};
>-
>-const uint8_t kMaskBursty12_2[4] = {
>-  0xaa, 0xa0,
>-  0xd5, 0x50
>-};
>-
>-const uint8_t kMaskBursty12_3[6] = {
>-  0x92, 0x40,
>-  0xc9, 0x20,
>-  0x74, 0x90
>-};
>-
>-const uint8_t kMaskBursty12_4[8] = {
>-  0x8a, 0x80,
>-  0xc5, 0x40,
>-  0x62, 0x20,
>-  0x39, 0x10
>-};
>-
>-const uint8_t kMaskBursty12_5[10] = {
>-  0x61, 0x00,
>-  0x30, 0xa0,
>-  0x1c, 0x50,
>-  0x85, 0x20,
>-  0xc2, 0x90
>-};
>-
>-const uint8_t kMaskBursty12_6[12] = {
>-  0x82, 0x90,
>-  0xc1, 0x40,
>-  0x60, 0xa0,
>-  0x30, 0x50,
>-  0x18, 0x20,
>-  0x0c, 0x10
>-};
>-
>-const uint8_t kMaskBursty12_7[14] = {
>-  0x0c, 0x00,
>-  0x07, 0x00,
>-  0x81, 0x40,
>-  0xc0, 0xa0,
>-  0x60, 0x50,
>-  0x30, 0x20,
>-  0x18, 0x10
>-};
>-
>-const uint8_t kMaskBursty12_8[16] = {
>-  0x18, 0x00,
>-  0x0c, 0x00,
>-  0x06, 0x00,
>-  0x03, 0x00,
>-  0x80, 0xa0,
>-  0xc0, 0x50,
>-  0x60, 0x20,
>-  0x30, 0x10
>-};
>-
>-const uint8_t kMaskBursty12_9[18] = {
>-  0x30, 0x00,
>-  0x18, 0x00,
>-  0x0c, 0x00,
>-  0x06, 0x00,
>-  0x03, 0x00,
>-  0x01, 0x80,
>-  0x80, 0x50,
>-  0xc0, 0x20,
>-  0x60, 0x10
>-};
>-
>-const uint8_t kMaskBursty12_10[20] = {
>-  0x60, 0x00,
>-  0x30, 0x00,
>-  0x18, 0x00,
>-  0x0c, 0x00,
>-  0x06, 0x00,
>-  0x03, 0x00,
>-  0x01, 0x80,
>-  0x00, 0xc0,
>-  0x80, 0x20,
>-  0xc0, 0x10
>-};
>-
>-const uint8_t kMaskBursty12_11[22] = {
>-  0xc0, 0x00,
>-  0x60, 0x00,
>-  0x30, 0x00,
>-  0x18, 0x00,
>-  0x0c, 0x00,
>-  0x06, 0x00,
>-  0x03, 0x00,
>-  0x01, 0x80,
>-  0x00, 0xc0,
>-  0x00, 0x60,
>-  0x80, 0x10
>-};
>-
>-const uint8_t kMaskBursty12_12[24] = {
>-  0x80, 0x00,
>-  0xc0, 0x00,
>-  0x60, 0x00,
>-  0x30, 0x00,
>-  0x18, 0x00,
>-  0x0c, 0x00,
>-  0x06, 0x00,
>-  0x03, 0x00,
>-  0x01, 0x80,
>-  0x00, 0xc0,
>-  0x00, 0x60,
>-  0x00, 0x30
>-};
>-
>-const uint8_t* const kPacketMaskBursty1[1] = {
>-  kMaskBursty1_1
>-};
>-
>-const uint8_t* const kPacketMaskBursty2[2] = {
>-  kMaskBursty2_1,
>-  kMaskBursty2_2
>-};
>-
>-const uint8_t* const kPacketMaskBursty3[3] = {
>-  kMaskBursty3_1,
>-  kMaskBursty3_2,
>-  kMaskBursty3_3
>-};
>-
>-const uint8_t* const kPacketMaskBursty4[4] = {
>-  kMaskBursty4_1,
>-  kMaskBursty4_2,
>-  kMaskBursty4_3,
>-  kMaskBursty4_4
>-};
>-
>-const uint8_t* const kPacketMaskBursty5[5] = {
>-  kMaskBursty5_1,
>-  kMaskBursty5_2,
>-  kMaskBursty5_3,
>-  kMaskBursty5_4,
>-  kMaskBursty5_5
>-};
>-
>-const uint8_t* const kPacketMaskBursty6[6] = {
>-  kMaskBursty6_1,
>-  kMaskBursty6_2,
>-  kMaskBursty6_3,
>-  kMaskBursty6_4,
>-  kMaskBursty6_5,
>-  kMaskBursty6_6
>-};
>-
>-const uint8_t* const kPacketMaskBursty7[7] = {
>-  kMaskBursty7_1,
>-  kMaskBursty7_2,
>-  kMaskBursty7_3,
>-  kMaskBursty7_4,
>-  kMaskBursty7_5,
>-  kMaskBursty7_6,
>-  kMaskBursty7_7
>-};
>-
>-const uint8_t* const kPacketMaskBursty8[8] = {
>-  kMaskBursty8_1,
>-  kMaskBursty8_2,
>-  kMaskBursty8_3,
>-  kMaskBursty8_4,
>-  kMaskBursty8_5,
>-  kMaskBursty8_6,
>-  kMaskBursty8_7,
>-  kMaskBursty8_8
>-};
>-
>-const uint8_t* const kPacketMaskBursty9[9] = {
>-  kMaskBursty9_1,
>-  kMaskBursty9_2,
>-  kMaskBursty9_3,
>-  kMaskBursty9_4,
>-  kMaskBursty9_5,
>-  kMaskBursty9_6,
>-  kMaskBursty9_7,
>-  kMaskBursty9_8,
>-  kMaskBursty9_9
>-};
>-
>-const uint8_t* const kPacketMaskBursty10[10] = {
>-  kMaskBursty10_1,
>-  kMaskBursty10_2,
>-  kMaskBursty10_3,
>-  kMaskBursty10_4,
>-  kMaskBursty10_5,
>-  kMaskBursty10_6,
>-  kMaskBursty10_7,
>-  kMaskBursty10_8,
>-  kMaskBursty10_9,
>-  kMaskBursty10_10
>-};
>-
>-const uint8_t* const kPacketMaskBursty11[11] = {
>-  kMaskBursty11_1,
>-  kMaskBursty11_2,
>-  kMaskBursty11_3,
>-  kMaskBursty11_4,
>-  kMaskBursty11_5,
>-  kMaskBursty11_6,
>-  kMaskBursty11_7,
>-  kMaskBursty11_8,
>-  kMaskBursty11_9,
>-  kMaskBursty11_10,
>-  kMaskBursty11_11
>-};
>-
>-const uint8_t* const kPacketMaskBursty12[12] = {
>-  kMaskBursty12_1,
>-  kMaskBursty12_2,
>-  kMaskBursty12_3,
>-  kMaskBursty12_4,
>-  kMaskBursty12_5,
>-  kMaskBursty12_6,
>-  kMaskBursty12_7,
>-  kMaskBursty12_8,
>-  kMaskBursty12_9,
>-  kMaskBursty12_10,
>-  kMaskBursty12_11,
>-  kMaskBursty12_12
>-};
>-
>-const uint8_t* const* const kPacketMaskBurstyTbl[12] = {
>-  kPacketMaskBursty1,
>-  kPacketMaskBursty2,
>-  kPacketMaskBursty3,
>-  kPacketMaskBursty4,
>-  kPacketMaskBursty5,
>-  kPacketMaskBursty6,
>-  kPacketMaskBursty7,
>-  kPacketMaskBursty8,
>-  kPacketMaskBursty9,
>-  kPacketMaskBursty10,
>-  kPacketMaskBursty11,
>-  kPacketMaskBursty12
>-};
>+extern const uint8_t kPacketMaskBurstyTbl[];
> 
> }  // namespace fec_private_tables
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/fec_private_tables_bursty_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/fec_private_tables_bursty_unittest.cc
>new file mode 100644
>index 00000000000..fd116eb3f0c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/fec_private_tables_bursty_unittest.cc
>@@ -0,0 +1,82 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/rtp_rtcp/source/fec_private_tables_bursty.h"
>+#include "modules/rtp_rtcp/source/fec_private_tables_random.h"
>+#include "modules/rtp_rtcp/source/forward_error_correction_internal.h"
>+
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+namespace {
>+constexpr uint8_t kMaskRandom15_6[] = {0x82, 0x08, 0x41, 0x04, 0x20, 0x82,
>+                                       0x10, 0x40, 0x08, 0x20, 0x04, 0x10};
>+}
>+
>+namespace fec_private_tables {
>+
>+using internal::LookUpInFecTable;
>+
>+TEST(FecTable, TestBurstyLookup) {
>+  rtc::ArrayView<const uint8_t> result;
>+  result = LookUpInFecTable(&kPacketMaskBurstyTbl[0], 0, 0);
>+  // Should match kMaskBursty1_1.
>+  EXPECT_EQ(2u, result.size());
>+  EXPECT_EQ(0x80u, result[0]);
>+
>+  result = LookUpInFecTable(&kPacketMaskBurstyTbl[0], 3, 0);
>+  // Should match kMaskBursty4_1.
>+  EXPECT_EQ(2u, result.size());
>+  EXPECT_EQ(0xf0u, result[0]);
>+  EXPECT_EQ(0x00u, result[1]);
>+
>+  result = LookUpInFecTable(&kPacketMaskBurstyTbl[0], 1, 1);
>+  // Should match kMaskBursty2_2.
>+  EXPECT_EQ(4u, result.size());
>+  EXPECT_EQ(0x80u, result[0]);
>+  EXPECT_EQ(0xc0u, result[2]);
>+
>+  result = LookUpInFecTable(&kPacketMaskBurstyTbl[0], 11, 11);
>+  // Should match kMaskBursty12_12.
>+  EXPECT_EQ(24u, result.size());
>+  EXPECT_EQ(0x80u, result[0]);
>+  EXPECT_EQ(0x30u, result[23]);
>+}
>+
>+TEST(FecTable, TestRandomLookup) {
>+  rtc::ArrayView<const uint8_t> result;
>+  result = LookUpInFecTable(&kPacketMaskRandomTbl[0], 0, 0);
>+  EXPECT_EQ(2u, result.size());
>+  EXPECT_EQ(0x80u, result[0]);
>+  EXPECT_EQ(0x00u, result[1]);
>+
>+  result = LookUpInFecTable(&kPacketMaskRandomTbl[0], 4, 1);
>+  // kMaskRandom5_2.
>+  EXPECT_EQ(4u, result.size());
>+  EXPECT_EQ(0xa8u, result[0]);
>+  EXPECT_EQ(0xd0u, result[2]);
>+}
>+
>+TEST(FecTable, TestRandomGenerated) {
>+  FecMaskType fec_mask_type = kFecMaskRandom;
>+  int num_media_packets = 15;
>+  int num_fec_packets = 6;
>+  size_t mask_size = sizeof(kMaskRandom15_6) / sizeof(uint8_t);
>+  internal::PacketMaskTable mask_table(fec_mask_type, num_media_packets);
>+  rtc::ArrayView<const uint8_t> mask =
>+      mask_table.LookUp(num_media_packets, num_fec_packets);
>+  EXPECT_EQ(mask.size(), mask_size);
>+  for (size_t i = 0; i < mask_size; ++i) {
>+    EXPECT_EQ(mask[i], kMaskRandom15_6[i]);
>+  }
>+}
>+
>+}  // namespace fec_private_tables
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/fec_private_tables_random.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/fec_private_tables_random.cc
>new file mode 100644
>index 00000000000..3cac5db17b7
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/fec_private_tables_random.cc
>@@ -0,0 +1,660 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/rtp_rtcp/source/fec_private_tables_random.h"
>+
>+namespace {
>+// clang-format off
>+#define kMaskRandom1_1 \
>+  0x80, 0x00
>+
>+#define kMaskRandom2_1 \
>+  0xc0, 0x00
>+
>+#define kMaskRandom2_2 \
>+  0xc0, 0x00, \
>+  0x80, 0x00
>+
>+#define kMaskRandom3_1 \
>+  0xe0, 0x00
>+
>+#define kMaskRandom3_2 \
>+  0xc0, 0x00, \
>+  0xa0, 0x00
>+
>+#define kMaskRandom3_3 \
>+  0xc0, 0x00, \
>+  0xa0, 0x00, \
>+  0x60, 0x00
>+
>+#define kMaskRandom4_1 \
>+  0xf0, 0x00
>+
>+#define kMaskRandom4_2 \
>+  0xc0, 0x00, \
>+  0xb0, 0x00
>+
>+#define kMaskRandom4_3 \
>+  0xc0, 0x00, \
>+  0xb0, 0x00, \
>+  0x60, 0x00
>+
>+#define kMaskRandom4_4 \
>+  0xc0, 0x00, \
>+  0xa0, 0x00, \
>+  0x30, 0x00, \
>+  0x50, 0x00
>+
>+#define kMaskRandom5_1 \
>+  0xf8, 0x00
>+
>+#define kMaskRandom5_2 \
>+  0xa8, 0x00, \
>+  0xd0, 0x00
>+
>+#define kMaskRandom5_3 \
>+  0xb0, 0x00, \
>+  0xc8, 0x00, \
>+  0x50, 0x00
>+
>+#define kMaskRandom5_4 \
>+  0xc8, 0x00, \
>+  0xb0, 0x00, \
>+  0x50, 0x00, \
>+  0x28, 0x00
>+
>+#define kMaskRandom5_5 \
>+  0xc0, 0x00, \
>+  0x30, 0x00, \
>+  0x18, 0x00, \
>+  0xa0, 0x00, \
>+  0x48, 0x00
>+
>+#define kMaskRandom6_1 \
>+  0xfc, 0x00
>+
>+#define kMaskRandom6_2 \
>+  0xa8, 0x00, \
>+  0xd4, 0x00
>+
>+#define kMaskRandom6_3 \
>+  0xd0, 0x00, \
>+  0x68, 0x00, \
>+  0xa4, 0x00
>+
>+#define kMaskRandom6_4 \
>+  0xa8, 0x00, \
>+  0x58, 0x00, \
>+  0x64, 0x00, \
>+  0x94, 0x00
>+
>+#define kMaskRandom6_5 \
>+  0xa8, 0x00, \
>+  0x84, 0x00, \
>+  0x64, 0x00, \
>+  0x90, 0x00, \
>+  0x58, 0x00
>+
>+#define kMaskRandom6_6 \
>+  0x98, 0x00, \
>+  0x64, 0x00, \
>+  0x50, 0x00, \
>+  0x14, 0x00, \
>+  0xa8, 0x00, \
>+  0xe0, 0x00
>+
>+#define kMaskRandom7_1 \
>+  0xfe, 0x00
>+
>+#define kMaskRandom7_2 \
>+  0xd4, 0x00, \
>+  0xaa, 0x00
>+
>+#define kMaskRandom7_3 \
>+  0xd0, 0x00, \
>+  0xaa, 0x00, \
>+  0x64, 0x00
>+
>+#define kMaskRandom7_4 \
>+  0xd0, 0x00, \
>+  0xaa, 0x00, \
>+  0x64, 0x00, \
>+  0x1c, 0x00
>+
>+#define kMaskRandom7_5 \
>+  0x0c, 0x00, \
>+  0xb0, 0x00, \
>+  0x1a, 0x00, \
>+  0xc4, 0x00, \
>+  0x62, 0x00
>+
>+#define kMaskRandom7_6 \
>+  0x8c, 0x00, \
>+  0x4a, 0x00, \
>+  0x64, 0x00, \
>+  0xd0, 0x00, \
>+  0xa0, 0x00, \
>+  0x32, 0x00
>+
>+#define kMaskRandom7_7 \
>+  0x4a, 0x00, \
>+  0x94, 0x00, \
>+  0x1a, 0x00, \
>+  0xc4, 0x00, \
>+  0x28, 0x00, \
>+  0xc2, 0x00, \
>+  0x34, 0x00
>+
>+#define kMaskRandom8_1 \
>+  0xff, 0x00
>+
>+#define kMaskRandom8_2 \
>+  0xaa, 0x00, \
>+  0xd5, 0x00
>+
>+#define kMaskRandom8_3 \
>+  0xc5, 0x00, \
>+  0x92, 0x00, \
>+  0x6a, 0x00
>+
>+#define kMaskRandom8_4 \
>+  0x45, 0x00, \
>+  0xb4, 0x00, \
>+  0x6a, 0x00, \
>+  0x89, 0x00
>+
>+#define kMaskRandom8_5 \
>+  0x8c, 0x00, \
>+  0x92, 0x00, \
>+  0x2b, 0x00, \
>+  0x51, 0x00, \
>+  0x64, 0x00
>+
>+#define kMaskRandom8_6 \
>+  0xa1, 0x00, \
>+  0x52, 0x00, \
>+  0x91, 0x00, \
>+  0x2a, 0x00, \
>+  0xc4, 0x00, \
>+  0x4c, 0x00
>+
>+#define kMaskRandom8_7 \
>+  0x15, 0x00, \
>+  0xc2, 0x00, \
>+  0x25, 0x00, \
>+  0x62, 0x00, \
>+  0x58, 0x00, \
>+  0x8c, 0x00, \
>+  0xa3, 0x00
>+
>+#define kMaskRandom8_8 \
>+  0x25, 0x00, \
>+  0x8a, 0x00, \
>+  0x91, 0x00, \
>+  0x68, 0x00, \
>+  0x32, 0x00, \
>+  0x43, 0x00, \
>+  0xc4, 0x00, \
>+  0x1c, 0x00
>+
>+#define kMaskRandom9_1 \
>+  0xff, 0x80
>+
>+#define kMaskRandom9_2 \
>+  0xaa, 0x80, \
>+  0xd5, 0x00
>+
>+#define kMaskRandom9_3 \
>+  0xa5, 0x00, \
>+  0xc8, 0x00, \
>+  0x52, 0x80
>+
>+#define kMaskRandom9_4 \
>+  0xa2, 0x00, \
>+  0xc9, 0x00, \
>+  0x52, 0x80, \
>+  0x24, 0x80
>+
>+#define kMaskRandom9_5 \
>+  0x8c, 0x00, \
>+  0x25, 0x00, \
>+  0x92, 0x80, \
>+  0x41, 0x80, \
>+  0x58, 0x00
>+
>+#define kMaskRandom9_6 \
>+  0x84, 0x80, \
>+  0x27, 0x00, \
>+  0x51, 0x80, \
>+  0x1a, 0x00, \
>+  0x68, 0x00, \
>+  0x89, 0x00
>+
>+#define kMaskRandom9_7 \
>+  0x8c, 0x00, \
>+  0x47, 0x00, \
>+  0x81, 0x80, \
>+  0x12, 0x80, \
>+  0x58, 0x00, \
>+  0x28, 0x80, \
>+  0xb4, 0x00
>+
>+#define kMaskRandom9_8 \
>+  0x2c, 0x00, \
>+  0x91, 0x00, \
>+  0x40, 0x80, \
>+  0x06, 0x80, \
>+  0xc8, 0x00, \
>+  0x45, 0x00, \
>+  0x30, 0x80, \
>+  0xa2, 0x00
>+
>+#define kMaskRandom9_9 \
>+  0x4c, 0x00, \
>+  0x62, 0x00, \
>+  0x91, 0x00, \
>+  0x42, 0x80, \
>+  0xa4, 0x00, \
>+  0x13, 0x00, \
>+  0x30, 0x80, \
>+  0x88, 0x80, \
>+  0x09, 0x00
>+
>+#define kMaskRandom10_1 \
>+  0xff, 0xc0
>+
>+#define kMaskRandom10_10 \
>+  0x4c, 0x00, \
>+  0x51, 0x00, \
>+  0xa0, 0x40, \
>+  0x04, 0xc0, \
>+  0x03, 0x80, \
>+  0x86, 0x00, \
>+  0x29, 0x00, \
>+  0x42, 0x40, \
>+  0x98, 0x00, \
>+  0x30, 0x80
>+
>+#define kMaskRandom10_2 \
>+  0xaa, 0x80, \
>+  0xd5, 0x40
>+
>+#define kMaskRandom10_3 \
>+  0xa4, 0x40, \
>+  0xc9, 0x00, \
>+  0x52, 0x80
>+
>+#define kMaskRandom10_4 \
>+  0xca, 0x00, \
>+  0x32, 0x80, \
>+  0xa1, 0x40, \
>+  0x55, 0x00
>+
>+#define kMaskRandom10_5 \
>+  0xca, 0x00, \
>+  0x32, 0x80, \
>+  0xa1, 0x40, \
>+  0x55, 0x00, \
>+  0x08, 0xc0
>+
>+#define kMaskRandom10_6 \
>+  0x0e, 0x00, \
>+  0x33, 0x00, \
>+  0x10, 0xc0, \
>+  0x45, 0x40, \
>+  0x88, 0x80, \
>+  0xe0, 0x00
>+
>+#define kMaskRandom10_7 \
>+  0x46, 0x00, \
>+  0x33, 0x00, \
>+  0x80, 0xc0, \
>+  0x0c, 0x40, \
>+  0x28, 0x80, \
>+  0x94, 0x00, \
>+  0xc1, 0x00
>+
>+#define kMaskRandom10_8 \
>+  0x2c, 0x00, \
>+  0x81, 0x80, \
>+  0xa0, 0x40, \
>+  0x05, 0x40, \
>+  0x18, 0x80, \
>+  0xc2, 0x00, \
>+  0x22, 0x80, \
>+  0x50, 0x40
>+
>+#define kMaskRandom10_9 \
>+  0x4c, 0x00, \
>+  0x23, 0x00, \
>+  0x88, 0xc0, \
>+  0x21, 0x40, \
>+  0x52, 0x80, \
>+  0x94, 0x00, \
>+  0x26, 0x00, \
>+  0x48, 0x40, \
>+  0x91, 0x80
>+
>+#define kMaskRandom11_1 \
>+  0xff, 0xe0
>+
>+#define kMaskRandom11_10 \
>+  0x64, 0x40, \
>+  0x51, 0x40, \
>+  0xa9, 0x00, \
>+  0x04, 0xc0, \
>+  0xd0, 0x00, \
>+  0x82, 0x40, \
>+  0x21, 0x20, \
>+  0x0c, 0x20, \
>+  0x4a, 0x00, \
>+  0x12, 0xa0
>+
>+#define kMaskRandom11_11 \
>+  0x46, 0x40, \
>+  0x33, 0x20, \
>+  0x99, 0x00, \
>+  0x05, 0x80, \
>+  0x80, 0xa0, \
>+  0x84, 0x40, \
>+  0x40, 0x60, \
>+  0x0a, 0x80, \
>+  0x68, 0x00, \
>+  0x10, 0x20, \
>+  0x30, 0x40
>+
>+#define kMaskRandom11_2 \
>+  0xec, 0xc0, \
>+  0x9b, 0xa0
>+
>+#define kMaskRandom11_3 \
>+  0xca, 0xc0, \
>+  0xf1, 0x40, \
>+  0xb6, 0x20
>+
>+#define kMaskRandom11_4 \
>+  0xc4, 0xc0, \
>+  0x31, 0x60, \
>+  0x4b, 0x20, \
>+  0x2c, 0xa0
>+
>+#define kMaskRandom11_5 \
>+  0x86, 0x80, \
>+  0x23, 0x20, \
>+  0x16, 0x20, \
>+  0x4c, 0x20, \
>+  0x41, 0xc0
>+
>+#define kMaskRandom11_6 \
>+  0x64, 0x40, \
>+  0x51, 0x40, \
>+  0x0c, 0xa0, \
>+  0xa1, 0x20, \
>+  0x12, 0xa0, \
>+  0x8a, 0x40
>+
>+#define kMaskRandom11_7 \
>+  0x46, 0x40, \
>+  0x33, 0x20, \
>+  0x91, 0x80, \
>+  0xa4, 0x20, \
>+  0x50, 0xa0, \
>+  0x84, 0xc0, \
>+  0x09, 0x60
>+
>+#define kMaskRandom11_8 \
>+  0x0c, 0x80, \
>+  0x80, 0x60, \
>+  0xa0, 0x80, \
>+  0x05, 0x40, \
>+  0x43, 0x00, \
>+  0x1a, 0x00, \
>+  0x60, 0x20, \
>+  0x14, 0x20
>+
>+#define kMaskRandom11_9 \
>+  0x46, 0x40, \
>+  0x62, 0x60, \
>+  0x8c, 0x00, \
>+  0x01, 0x60, \
>+  0x07, 0x80, \
>+  0xa0, 0x80, \
>+  0x18, 0xa0, \
>+  0x91, 0x00, \
>+  0x78, 0x00
>+
>+#define kMaskRandom12_1 \
>+  0xff, 0xf0
>+
>+#define kMaskRandom12_10 \
>+  0x51, 0x40, \
>+  0x45, 0x10, \
>+  0x80, 0xd0, \
>+  0x24, 0x20, \
>+  0x0a, 0x20, \
>+  0x00, 0xe0, \
>+  0xb8, 0x00, \
>+  0x09, 0x10, \
>+  0x56, 0x00, \
>+  0xa2, 0x80
>+
>+#define kMaskRandom12_11 \
>+  0x53, 0x60, \
>+  0x21, 0x30, \
>+  0x10, 0x90, \
>+  0x00, 0x70, \
>+  0x0c, 0x10, \
>+  0x40, 0xc0, \
>+  0x6a, 0x00, \
>+  0x86, 0x00, \
>+  0x24, 0x80, \
>+  0x89, 0x00, \
>+  0xc0, 0x20
>+
>+#define kMaskRandom12_12 \
>+  0x10, 0x60, \
>+  0x02, 0x30, \
>+  0x40, 0x50, \
>+  0x21, 0x80, \
>+  0x81, 0x10, \
>+  0x14, 0x80, \
>+  0x98, 0x00, \
>+  0x08, 0x90, \
>+  0x62, 0x00, \
>+  0x24, 0x20, \
>+  0x8a, 0x00, \
>+  0x84, 0x40
>+
>+#define kMaskRandom12_2 \
>+  0xec, 0xc0, \
>+  0x93, 0xb0
>+
>+#define kMaskRandom12_3 \
>+  0x9b, 0x80, \
>+  0x4f, 0x10, \
>+  0x3c, 0x60
>+
>+#define kMaskRandom12_4 \
>+  0x8b, 0x20, \
>+  0x14, 0xb0, \
>+  0x22, 0xd0, \
>+  0x45, 0x50
>+
>+#define kMaskRandom12_5 \
>+  0x53, 0x60, \
>+  0x64, 0x20, \
>+  0x0c, 0xc0, \
>+  0x82, 0xa0, \
>+  0x09, 0x30
>+
>+#define kMaskRandom12_6 \
>+  0x51, 0x40, \
>+  0xc5, 0x10, \
>+  0x21, 0x80, \
>+  0x12, 0x30, \
>+  0x08, 0xe0, \
>+  0x2e, 0x00
>+
>+#define kMaskRandom12_7 \
>+  0x53, 0x60, \
>+  0x21, 0x30, \
>+  0x90, 0x90, \
>+  0x02, 0x50, \
>+  0x06, 0xa0, \
>+  0x2c, 0x00, \
>+  0x88, 0x60
>+
>+#define kMaskRandom12_8 \
>+  0x20, 0x60, \
>+  0x80, 0x30, \
>+  0x42, 0x40, \
>+  0x01, 0x90, \
>+  0x14, 0x10, \
>+  0x0a, 0x80, \
>+  0x38, 0x00, \
>+  0xc5, 0x00
>+
>+#define kMaskRandom12_9 \
>+  0x53, 0x60, \
>+  0xe4, 0x20, \
>+  0x24, 0x40, \
>+  0xa1, 0x10, \
>+  0x18, 0x30, \
>+  0x03, 0x90, \
>+  0x8a, 0x10, \
>+  0x04, 0x90, \
>+  0x00, 0xe0
>+
>+#define kPacketMaskRandom1 1, \
>+  kMaskRandom1_1
>+
>+#define kPacketMaskRandom2 2, \
>+  kMaskRandom2_1, \
>+  kMaskRandom2_2
>+
>+#define kPacketMaskRandom3 3, \
>+  kMaskRandom3_1, \
>+  kMaskRandom3_2, \
>+  kMaskRandom3_3
>+
>+#define kPacketMaskRandom4 4, \
>+  kMaskRandom4_1, \
>+  kMaskRandom4_2, \
>+  kMaskRandom4_3, \
>+  kMaskRandom4_4
>+
>+#define kPacketMaskRandom5 5, \
>+  kMaskRandom5_1, \
>+  kMaskRandom5_2, \
>+  kMaskRandom5_3, \
>+  kMaskRandom5_4, \
>+  kMaskRandom5_5
>+
>+#define kPacketMaskRandom6 6, \
>+  kMaskRandom6_1, \
>+  kMaskRandom6_2, \
>+  kMaskRandom6_3, \
>+  kMaskRandom6_4, \
>+  kMaskRandom6_5, \
>+  kMaskRandom6_6
>+
>+#define kPacketMaskRandom7 7, \
>+  kMaskRandom7_1, \
>+  kMaskRandom7_2, \
>+  kMaskRandom7_3, \
>+  kMaskRandom7_4, \
>+  kMaskRandom7_5, \
>+  kMaskRandom7_6, \
>+  kMaskRandom7_7
>+
>+#define kPacketMaskRandom8 8, \
>+  kMaskRandom8_1, \
>+  kMaskRandom8_2, \
>+  kMaskRandom8_3, \
>+  kMaskRandom8_4, \
>+  kMaskRandom8_5, \
>+  kMaskRandom8_6, \
>+  kMaskRandom8_7, \
>+  kMaskRandom8_8
>+
>+#define kPacketMaskRandom9 9, \
>+  kMaskRandom9_1, \
>+  kMaskRandom9_2, \
>+  kMaskRandom9_3, \
>+  kMaskRandom9_4, \
>+  kMaskRandom9_5, \
>+  kMaskRandom9_6, \
>+  kMaskRandom9_7, \
>+  kMaskRandom9_8, \
>+  kMaskRandom9_9
>+
>+#define kPacketMaskRandom10 10, \
>+  kMaskRandom10_1, \
>+  kMaskRandom10_2, \
>+  kMaskRandom10_3, \
>+  kMaskRandom10_4, \
>+  kMaskRandom10_5, \
>+  kMaskRandom10_6, \
>+  kMaskRandom10_7, \
>+  kMaskRandom10_8, \
>+  kMaskRandom10_9, \
>+  kMaskRandom10_10
>+
>+#define kPacketMaskRandom11 11, \
>+  kMaskRandom11_1, \
>+  kMaskRandom11_2, \
>+  kMaskRandom11_3, \
>+  kMaskRandom11_4, \
>+  kMaskRandom11_5, \
>+  kMaskRandom11_6, \
>+  kMaskRandom11_7, \
>+  kMaskRandom11_8, \
>+  kMaskRandom11_9, \
>+  kMaskRandom11_10, \
>+  kMaskRandom11_11
>+
>+#define kPacketMaskRandom12 12, \
>+  kMaskRandom12_1, \
>+  kMaskRandom12_2, \
>+  kMaskRandom12_3, \
>+  kMaskRandom12_4, \
>+  kMaskRandom12_5, \
>+  kMaskRandom12_6, \
>+  kMaskRandom12_7, \
>+  kMaskRandom12_8, \
>+  kMaskRandom12_9, \
>+  kMaskRandom12_10, \
>+  kMaskRandom12_11, \
>+  kMaskRandom12_12
>+
>+// clang-format on
>+}  // namespace
>+
>+namespace webrtc {
>+namespace fec_private_tables {
>+
>+const uint8_t kPacketMaskRandomTbl[] = {
>+    12,
>+    kPacketMaskRandom1,  // 2 byte entries.
>+    kPacketMaskRandom2,
>+    kPacketMaskRandom3,
>+    kPacketMaskRandom4,
>+    kPacketMaskRandom5,
>+    kPacketMaskRandom6,
>+    kPacketMaskRandom7,
>+    kPacketMaskRandom8,
>+    kPacketMaskRandom9,
>+    kPacketMaskRandom10,
>+    kPacketMaskRandom11,
>+    kPacketMaskRandom12,
>+};
>+
>+}  // namespace fec_private_tables
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/fec_private_tables_random.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/fec_private_tables_random.h
>index 4d490a182ac..cc7b92984cc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/fec_private_tables_random.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/fec_private_tables_random.h
>@@ -15,24509 +15,12 @@
> // this table are specifically designed to favor recovery to random loss.
> // These packet masks are defined to protect up to maximum of 48 media packets.
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include <stdint.h>
> 
> namespace webrtc {
> namespace fec_private_tables {
> 
>-const uint8_t kMaskRandom10_1[2] = {
>-  0xff, 0xc0
>-};
>-
>-const uint8_t kMaskRandom10_10[20] = {
>-  0x4c, 0x00,
>-  0x51, 0x00,
>-  0xa0, 0x40,
>-  0x04, 0xc0,
>-  0x03, 0x80,
>-  0x86, 0x00,
>-  0x29, 0x00,
>-  0x42, 0x40,
>-  0x98, 0x00,
>-  0x30, 0x80
>-};
>-
>-const uint8_t kMaskRandom10_2[4] = {
>-  0xaa, 0x80,
>-  0xd5, 0x40
>-};
>-
>-const uint8_t kMaskRandom10_3[6] = {
>-  0xa4, 0x40,
>-  0xc9, 0x00,
>-  0x52, 0x80
>-};
>-
>-const uint8_t kMaskRandom10_4[8] = {
>-  0xca, 0x00,
>-  0x32, 0x80,
>-  0xa1, 0x40,
>-  0x55, 0x00
>-};
>-
>-const uint8_t kMaskRandom10_5[10] = {
>-  0xca, 0x00,
>-  0x32, 0x80,
>-  0xa1, 0x40,
>-  0x55, 0x00,
>-  0x08, 0xc0
>-};
>-
>-const uint8_t kMaskRandom10_6[12] = {
>-  0x0e, 0x00,
>-  0x33, 0x00,
>-  0x10, 0xc0,
>-  0x45, 0x40,
>-  0x88, 0x80,
>-  0xe0, 0x00
>-};
>-
>-const uint8_t kMaskRandom10_7[14] = {
>-  0x46, 0x00,
>-  0x33, 0x00,
>-  0x80, 0xc0,
>-  0x0c, 0x40,
>-  0x28, 0x80,
>-  0x94, 0x00,
>-  0xc1, 0x00
>-};
>-
>-const uint8_t kMaskRandom10_8[16] = {
>-  0x2c, 0x00,
>-  0x81, 0x80,
>-  0xa0, 0x40,
>-  0x05, 0x40,
>-  0x18, 0x80,
>-  0xc2, 0x00,
>-  0x22, 0x80,
>-  0x50, 0x40
>-};
>-
>-const uint8_t kMaskRandom10_9[18] = {
>-  0x4c, 0x00,
>-  0x23, 0x00,
>-  0x88, 0xc0,
>-  0x21, 0x40,
>-  0x52, 0x80,
>-  0x94, 0x00,
>-  0x26, 0x00,
>-  0x48, 0x40,
>-  0x91, 0x80
>-};
>-
>-const uint8_t kMaskRandom11_1[2] = {
>-  0xff, 0xe0
>-};
>-
>-const uint8_t kMaskRandom11_10[20] = {
>-  0x64, 0x40,
>-  0x51, 0x40,
>-  0xa9, 0x00,
>-  0x04, 0xc0,
>-  0xd0, 0x00,
>-  0x82, 0x40,
>-  0x21, 0x20,
>-  0x0c, 0x20,
>-  0x4a, 0x00,
>-  0x12, 0xa0
>-};
>-
>-const uint8_t kMaskRandom11_11[22] = {
>-  0x46, 0x40,
>-  0x33, 0x20,
>-  0x99, 0x00,
>-  0x05, 0x80,
>-  0x80, 0xa0,
>-  0x84, 0x40,
>-  0x40, 0x60,
>-  0x0a, 0x80,
>-  0x68, 0x00,
>-  0x10, 0x20,
>-  0x30, 0x40
>-};
>-
>-const uint8_t kMaskRandom11_2[4] = {
>-  0xec, 0xc0,
>-  0x9b, 0xa0
>-};
>-
>-const uint8_t kMaskRandom11_3[6] = {
>-  0xca, 0xc0,
>-  0xf1, 0x40,
>-  0xb6, 0x20
>-};
>-
>-const uint8_t kMaskRandom11_4[8] = {
>-  0xc4, 0xc0,
>-  0x31, 0x60,
>-  0x4b, 0x20,
>-  0x2c, 0xa0
>-};
>-
>-const uint8_t kMaskRandom11_5[10] = {
>-  0x86, 0x80,
>-  0x23, 0x20,
>-  0x16, 0x20,
>-  0x4c, 0x20,
>-  0x41, 0xc0
>-};
>-
>-const uint8_t kMaskRandom11_6[12] = {
>-  0x64, 0x40,
>-  0x51, 0x40,
>-  0x0c, 0xa0,
>-  0xa1, 0x20,
>-  0x12, 0xa0,
>-  0x8a, 0x40
>-};
>-
>-const uint8_t kMaskRandom11_7[14] = {
>-  0x46, 0x40,
>-  0x33, 0x20,
>-  0x91, 0x80,
>-  0xa4, 0x20,
>-  0x50, 0xa0,
>-  0x84, 0xc0,
>-  0x09, 0x60
>-};
>-
>-const uint8_t kMaskRandom11_8[16] = {
>-  0x0c, 0x80,
>-  0x80, 0x60,
>-  0xa0, 0x80,
>-  0x05, 0x40,
>-  0x43, 0x00,
>-  0x1a, 0x00,
>-  0x60, 0x20,
>-  0x14, 0x20
>-};
>-
>-const uint8_t kMaskRandom11_9[18] = {
>-  0x46, 0x40,
>-  0x62, 0x60,
>-  0x8c, 0x00,
>-  0x01, 0x60,
>-  0x07, 0x80,
>-  0xa0, 0x80,
>-  0x18, 0xa0,
>-  0x91, 0x00,
>-  0x78, 0x00
>-};
>-
>-const uint8_t kMaskRandom12_1[2] = {
>-  0xff, 0xf0
>-};
>-
>-const uint8_t kMaskRandom12_10[20] = {
>-  0x51, 0x40,
>-  0x45, 0x10,
>-  0x80, 0xd0,
>-  0x24, 0x20,
>-  0x0a, 0x20,
>-  0x00, 0xe0,
>-  0xb8, 0x00,
>-  0x09, 0x10,
>-  0x56, 0x00,
>-  0xa2, 0x80
>-};
>-
>-const uint8_t kMaskRandom12_11[22] = {
>-  0x53, 0x60,
>-  0x21, 0x30,
>-  0x10, 0x90,
>-  0x00, 0x70,
>-  0x0c, 0x10,
>-  0x40, 0xc0,
>-  0x6a, 0x00,
>-  0x86, 0x00,
>-  0x24, 0x80,
>-  0x89, 0x00,
>-  0xc0, 0x20
>-};
>-
>-const uint8_t kMaskRandom12_12[24] = {
>-  0x10, 0x60,
>-  0x02, 0x30,
>-  0x40, 0x50,
>-  0x21, 0x80,
>-  0x81, 0x10,
>-  0x14, 0x80,
>-  0x98, 0x00,
>-  0x08, 0x90,
>-  0x62, 0x00,
>-  0x24, 0x20,
>-  0x8a, 0x00,
>-  0x84, 0x40
>-};
>-
>-const uint8_t kMaskRandom12_2[4] = {
>-  0xec, 0xc0,
>-  0x93, 0xb0
>-};
>-
>-const uint8_t kMaskRandom12_3[6] = {
>-  0x9b, 0x80,
>-  0x4f, 0x10,
>-  0x3c, 0x60
>-};
>-
>-const uint8_t kMaskRandom12_4[8] = {
>-  0x8b, 0x20,
>-  0x14, 0xb0,
>-  0x22, 0xd0,
>-  0x45, 0x50
>-};
>-
>-const uint8_t kMaskRandom12_5[10] = {
>-  0x53, 0x60,
>-  0x64, 0x20,
>-  0x0c, 0xc0,
>-  0x82, 0xa0,
>-  0x09, 0x30
>-};
>-
>-const uint8_t kMaskRandom12_6[12] = {
>-  0x51, 0x40,
>-  0xc5, 0x10,
>-  0x21, 0x80,
>-  0x12, 0x30,
>-  0x08, 0xe0,
>-  0x2e, 0x00
>-};
>-
>-const uint8_t kMaskRandom12_7[14] = {
>-  0x53, 0x60,
>-  0x21, 0x30,
>-  0x90, 0x90,
>-  0x02, 0x50,
>-  0x06, 0xa0,
>-  0x2c, 0x00,
>-  0x88, 0x60
>-};
>-
>-const uint8_t kMaskRandom12_8[16] = {
>-  0x20, 0x60,
>-  0x80, 0x30,
>-  0x42, 0x40,
>-  0x01, 0x90,
>-  0x14, 0x10,
>-  0x0a, 0x80,
>-  0x38, 0x00,
>-  0xc5, 0x00
>-};
>-
>-const uint8_t kMaskRandom12_9[18] = {
>-  0x53, 0x60,
>-  0xe4, 0x20,
>-  0x24, 0x40,
>-  0xa1, 0x10,
>-  0x18, 0x30,
>-  0x03, 0x90,
>-  0x8a, 0x10,
>-  0x04, 0x90,
>-  0x00, 0xe0
>-};
>-
>-const uint8_t kMaskRandom13_1[2] = {
>-  0xff, 0xf8
>-};
>-
>-const uint8_t kMaskRandom13_10[20] = {
>-  0xd1, 0x00,
>-  0x44, 0x50,
>-  0x10, 0x98,
>-  0xa0, 0x50,
>-  0x4a, 0x08,
>-  0x40, 0x30,
>-  0x80, 0x28,
>-  0x0c, 0x90,
>-  0x05, 0x88,
>-  0x62, 0x20
>-};
>-
>-const uint8_t kMaskRandom13_11[22] = {
>-  0x51, 0x20,
>-  0x22, 0x10,
>-  0x13, 0x40,
>-  0x25, 0x00,
>-  0x18, 0x18,
>-  0x0a, 0x20,
>-  0x88, 0x88,
>-  0x06, 0x80,
>-  0xe0, 0x20,
>-  0x84, 0x40,
>-  0x44, 0x18
>-};
>-
>-const uint8_t kMaskRandom13_12[24] = {
>-  0x28, 0x28,
>-  0x84, 0x50,
>-  0x60, 0x40,
>-  0x05, 0x48,
>-  0x02, 0x98,
>-  0x01, 0x30,
>-  0x48, 0x10,
>-  0x24, 0x80,
>-  0x94, 0x00,
>-  0x8a, 0x00,
>-  0x11, 0x80,
>-  0x52, 0x20
>-};
>-
>-const uint8_t kMaskRandom13_13[26] = {
>-  0x51, 0x20,
>-  0x66, 0x40,
>-  0x05, 0x48,
>-  0x81, 0x20,
>-  0x94, 0x00,
>-  0x30, 0x80,
>-  0x21, 0x10,
>-  0x03, 0xc0,
>-  0xe8, 0x00,
>-  0x0a, 0x10,
>-  0x80, 0x18,
>-  0x04, 0x90,
>-  0x08, 0xa8
>-};
>-
>-const uint8_t kMaskRandom13_2[4] = {
>-  0xec, 0xc0,
>-  0x1b, 0x38
>-};
>-
>-const uint8_t kMaskRandom13_3[6] = {
>-  0x99, 0xb0,
>-  0x46, 0xd8,
>-  0x37, 0x28
>-};
>-
>-const uint8_t kMaskRandom13_4[8] = {
>-  0x49, 0xb0,
>-  0x26, 0xd0,
>-  0x85, 0x68,
>-  0x52, 0x58
>-};
>-
>-const uint8_t kMaskRandom13_5[10] = {
>-  0x51, 0x30,
>-  0x66, 0x40,
>-  0x0c, 0x68,
>-  0xa1, 0xc0,
>-  0x22, 0x98
>-};
>-
>-const uint8_t kMaskRandom13_6[12] = {
>-  0xd1, 0x20,
>-  0x46, 0xd0,
>-  0x15, 0x48,
>-  0x21, 0x70,
>-  0x28, 0xc8,
>-  0xaa, 0x20
>-};
>-
>-const uint8_t kMaskRandom13_7[14] = {
>-  0x59, 0x20,
>-  0x26, 0x50,
>-  0xb1, 0x40,
>-  0x2b, 0x08,
>-  0x14, 0xc8,
>-  0xc8, 0x88,
>-  0x84, 0xb0
>-};
>-
>-const uint8_t kMaskRandom13_8[16] = {
>-  0x80, 0xa8,
>-  0x30, 0x90,
>-  0x16, 0x08,
>-  0x03, 0x30,
>-  0x44, 0x60,
>-  0x08, 0x18,
>-  0xd8, 0x00,
>-  0xa1, 0x40
>-};
>-
>-const uint8_t kMaskRandom13_9[18] = {
>-  0x59, 0x20,
>-  0x66, 0x40,
>-  0x14, 0x40,
>-  0x21, 0x48,
>-  0x02, 0xc8,
>-  0x94, 0x10,
>-  0x80, 0xa8,
>-  0x0a, 0x90,
>-  0x40, 0x18
>-};
>-
>-const uint8_t kMaskRandom14_1[2] = {
>-  0xff, 0xfc
>-};
>-
>-const uint8_t kMaskRandom14_10[20] = {
>-  0xc0, 0xd4,
>-  0x1d, 0x40,
>-  0xd4, 0x08,
>-  0x02, 0x60,
>-  0x04, 0x28,
>-  0x20, 0x98,
>-  0x40, 0x44,
>-  0x08, 0x84,
>-  0x68, 0x00,
>-  0x23, 0x10
>-};
>-
>-const uint8_t kMaskRandom14_11[22] = {
>-  0x62, 0xd0,
>-  0x35, 0x20,
>-  0x14, 0x14,
>-  0xc5, 0x08,
>-  0x22, 0x0c,
>-  0x88, 0xb8,
>-  0x42, 0x54,
>-  0x28, 0xa4,
>-  0x94, 0x20,
>-  0x1b, 0x04,
>-  0x22, 0xc0
>-};
>-
>-const uint8_t kMaskRandom14_12[24] = {
>-  0x81, 0x04,
>-  0x40, 0x68,
>-  0x90, 0x24,
>-  0x28, 0x28,
>-  0x52, 0x10,
>-  0x41, 0x88,
>-  0x09, 0x30,
>-  0x48, 0x44,
>-  0x04, 0x44,
>-  0x0e, 0x80,
>-  0xa5, 0x90,
>-  0x12, 0x0c
>-};
>-
>-const uint8_t kMaskRandom14_13[26] = {
>-  0x62, 0x54,
>-  0x34, 0x60,
>-  0x48, 0x04,
>-  0x00, 0xac,
>-  0x28, 0x08,
>-  0x81, 0x08,
>-  0x23, 0x04,
>-  0x06, 0x80,
>-  0x80, 0x14,
>-  0x30, 0x10,
>-  0x8c, 0x20,
>-  0x54, 0x00,
>-  0x80, 0xc0
>-};
>-
>-const uint8_t kMaskRandom14_14[28] = {
>-  0x40, 0x54,
>-  0x15, 0x40,
>-  0xc0, 0x04,
>-  0x28, 0x10,
>-  0x05, 0x0c,
>-  0x64, 0x80,
>-  0x81, 0x80,
>-  0x10, 0x98,
>-  0x84, 0x20,
>-  0x12, 0x30,
>-  0x62, 0x00,
>-  0x28, 0x60,
>-  0x0e, 0x08,
>-  0x10, 0x84
>-};
>-
>-const uint8_t kMaskRandom14_2[4] = {
>-  0xec, 0xe8,
>-  0x3b, 0x9c
>-};
>-
>-const uint8_t kMaskRandom14_3[6] = {
>-  0xac, 0xd8,
>-  0x55, 0x6c,
>-  0x27, 0xb4
>-};
>-
>-const uint8_t kMaskRandom14_4[8] = {
>-  0x2c, 0xd8,
>-  0x93, 0x68,
>-  0x1a, 0xb4,
>-  0x47, 0x2c
>-};
>-
>-const uint8_t kMaskRandom14_5[10] = {
>-  0x64, 0xd8,
>-  0xa5, 0x68,
>-  0x52, 0xb4,
>-  0x1d, 0xa8,
>-  0x9c, 0x54
>-};
>-
>-const uint8_t kMaskRandom14_6[12] = {
>-  0x4a, 0x54,
>-  0x95, 0x48,
>-  0x14, 0xb4,
>-  0x51, 0xa8,
>-  0x22, 0x6c,
>-  0x88, 0x8c
>-};
>-
>-const uint8_t kMaskRandom14_7[14] = {
>-  0x62, 0x54,
>-  0xb9, 0x20,
>-  0x18, 0xb4,
>-  0x54, 0x98,
>-  0x06, 0x6c,
>-  0x85, 0x54,
>-  0xaa, 0x88
>-};
>-
>-const uint8_t kMaskRandom14_8[16] = {
>-  0xc0, 0x14,
>-  0x41, 0x60,
>-  0x88, 0x30,
>-  0x20, 0xa4,
>-  0x0a, 0x48,
>-  0x04, 0x98,
>-  0x94, 0x40,
>-  0x72, 0x00
>-};
>-
>-const uint8_t kMaskRandom14_9[18] = {
>-  0xa2, 0x54,
>-  0x34, 0x60,
>-  0x4a, 0x24,
>-  0x20, 0xa8,
>-  0x11, 0x84,
>-  0x49, 0x08,
>-  0x86, 0x0c,
>-  0x20, 0xd4,
>-  0x88, 0x48
>-};
>-
>-const uint8_t kMaskRandom15_1[2] = {
>-  0xff, 0xfe
>-};
>-
>-const uint8_t kMaskRandom15_10[20] = {
>-  0xc0, 0xa0,
>-  0x15, 0x56,
>-  0x74, 0x40,
>-  0x00, 0x9c,
>-  0x01, 0x2c,
>-  0x44, 0x92,
>-  0x88, 0x50,
>-  0x20, 0xa4,
>-  0xaa, 0x04,
>-  0x02, 0x62
>-};
>-
>-const uint8_t kMaskRandom15_11[22] = {
>-  0x62, 0x22,
>-  0xf1, 0x10,
>-  0x10, 0x0e,
>-  0x10, 0xb0,
>-  0x24, 0x24,
>-  0x01, 0x12,
>-  0x00, 0xc4,
>-  0x04, 0xa2,
>-  0x02, 0x58,
>-  0x2b, 0x00,
>-  0x98, 0x40
>-};
>-
>-const uint8_t kMaskRandom15_12[24] = {
>-  0x88, 0x90,
>-  0x40, 0x54,
>-  0x82, 0x62,
>-  0x21, 0xa4,
>-  0x10, 0x64,
>-  0x44, 0x0a,
>-  0x10, 0xc8,
>-  0x4d, 0x2a,
>-  0x38, 0x02,
>-  0x17, 0x48,
>-  0x90, 0x84,
>-  0x72, 0x14
>-};
>-
>-const uint8_t kMaskRandom15_13[26] = {
>-  0x62, 0xa2,
>-  0x34, 0x44,
>-  0x40, 0x4a,
>-  0xc4, 0x04,
>-  0x08, 0x60,
>-  0x94, 0x12,
>-  0x88, 0xc0,
>-  0x21, 0x32,
>-  0xc1, 0x40,
>-  0x10, 0x68,
>-  0x06, 0x90,
>-  0x59, 0x00,
>-  0x0a, 0x0c
>-};
>-
>-const uint8_t kMaskRandom15_14[28] = {
>-  0x40, 0x82,
>-  0x15, 0x54,
>-  0x88, 0x12,
>-  0xc0, 0x10,
>-  0x80, 0xa0,
>-  0x01, 0x22,
>-  0x40, 0x2c,
>-  0x22, 0x02,
>-  0x90, 0x04,
>-  0x12, 0x40,
>-  0x5d, 0x00,
>-  0x20, 0x54,
>-  0x86, 0x08,
>-  0x28, 0x88
>-};
>-
>-const uint8_t kMaskRandom15_15[30] = {
>-  0x62, 0x22,
>-  0x31, 0x10,
>-  0x58, 0x00,
>-  0x01, 0x12,
>-  0x88, 0x20,
>-  0x44, 0x02,
>-  0x29, 0x04,
>-  0x82, 0xa0,
>-  0x0a, 0x1a,
>-  0x11, 0xe0,
>-  0x84, 0x04,
>-  0x86, 0x40,
>-  0x00, 0x86,
>-  0x44, 0x48,
>-  0x10, 0x98
>-};
>-
>-const uint8_t kMaskRandom15_2[4] = {
>-  0xec, 0xea,
>-  0xbb, 0x9c
>-};
>-
>-const uint8_t kMaskRandom15_3[6] = {
>-  0xac, 0x92,
>-  0x55, 0x4a,
>-  0x43, 0x36
>-};
>-
>-const uint8_t kMaskRandom15_4[8] = {
>-  0x25, 0xaa,
>-  0x95, 0x54,
>-  0x1a, 0x6a,
>-  0x43, 0xd4
>-};
>-
>-const uint8_t kMaskRandom15_5[10] = {
>-  0x64, 0xa2,
>-  0x25, 0x54,
>-  0x49, 0x68,
>-  0x53, 0x90,
>-  0x8e, 0x30
>-};
>-
>-const uint8_t kMaskRandom15_6[12] = {
>-  0x62, 0x8a,
>-  0x15, 0x54,
>-  0x4c, 0x46,
>-  0x52, 0x94,
>-  0x23, 0x64,
>-  0x8a, 0x58
>-};
>-
>-const uint8_t kMaskRandom15_7[14] = {
>-  0x62, 0xa2,
>-  0xb1, 0x14,
>-  0x18, 0x6a,
>-  0x44, 0xd4,
>-  0x13, 0x64,
>-  0x49, 0x1a,
>-  0x86, 0x8c
>-};
>-
>-const uint8_t kMaskRandom15_8[16] = {
>-  0x90, 0x22,
>-  0x09, 0x50,
>-  0x00, 0x6a,
>-  0x20, 0x34,
>-  0x14, 0x44,
>-  0xc2, 0x10,
>-  0x00, 0xc6,
>-  0x65, 0x80
>-};
>-
>-const uint8_t kMaskRandom15_9[18] = {
>-  0x62, 0x22,
>-  0x24, 0x44,
>-  0xc0, 0x50,
>-  0x03, 0x0c,
>-  0x16, 0x28,
>-  0x89, 0x00,
>-  0x82, 0x90,
>-  0x08, 0xa4,
>-  0x90, 0x48
>-};
>-
>-const uint8_t kMaskRandom16_1[2] = {
>-  0xff, 0xff
>-};
>-
>-const uint8_t kMaskRandom16_10[20] = {
>-  0x45, 0x51,
>-  0x10, 0xa2,
>-  0x01, 0x25,
>-  0x0b, 0x42,
>-  0xd8, 0x20,
>-  0x82, 0x8c,
>-  0x24, 0x4a,
>-  0x38, 0x18,
>-  0x2a, 0x25,
>-  0x84, 0x92
>-};
>-
>-const uint8_t kMaskRandom16_11[22] = {
>-  0x55, 0x55,
>-  0x2a, 0x22,
>-  0x31, 0x11,
>-  0x83, 0x42,
>-  0x06, 0x98,
>-  0x40, 0xe1,
>-  0x2c, 0x44,
>-  0xd8, 0x28,
>-  0x92, 0x81,
>-  0x84, 0x32,
>-  0x68, 0x0c
>-};
>-
>-const uint8_t kMaskRandom16_12[24] = {
>-  0x84, 0x31,
>-  0x18, 0xa2,
>-  0x4e, 0x01,
>-  0x44, 0xc8,
>-  0x0e, 0x90,
>-  0x20, 0xcc,
>-  0x93, 0x40,
>-  0x2d, 0x10,
>-  0x31, 0x44,
>-  0xc0, 0x23,
>-  0x11, 0x25,
>-  0xe8, 0x80
>-};
>-
>-const uint8_t kMaskRandom16_13[26] = {
>-  0x45, 0x15,
>-  0x22, 0x22,
>-  0x96, 0x0c,
>-  0x0c, 0x50,
>-  0x62, 0x04,
>-  0x49, 0x06,
>-  0x11, 0x82,
>-  0x12, 0x38,
>-  0x40, 0x71,
>-  0xa8, 0x8a,
>-  0x08, 0xa1,
>-  0xa0, 0xc0,
>-  0xc5, 0x10
>-};
>-
>-const uint8_t kMaskRandom16_14[28] = {
>-  0x45, 0x51,
>-  0x22, 0x0a,
>-  0x84, 0xd0,
>-  0x0c, 0x8a,
>-  0x18, 0x06,
>-  0x30, 0x03,
>-  0x61, 0x08,
>-  0x40, 0x11,
>-  0x10, 0x2c,
>-  0x09, 0x60,
>-  0x00, 0x94,
>-  0x52, 0x40,
>-  0xa4, 0x24,
>-  0x82, 0x88
>-};
>-
>-const uint8_t kMaskRandom16_15[30] = {
>-  0x55, 0x11,
>-  0x22, 0x22,
>-  0x11, 0x11,
>-  0x80, 0x45,
>-  0x20, 0x1a,
>-  0x08, 0x68,
>-  0x22, 0x84,
>-  0x48, 0x09,
>-  0x07, 0x01,
>-  0x94, 0x20,
>-  0x82, 0x06,
>-  0x60, 0x48,
>-  0x89, 0x80,
>-  0x00, 0x8e,
>-  0x18, 0x22
>-};
>-
>-const uint8_t kMaskRandom16_16[32] = {
>-  0xa4, 0x10,
>-  0x01, 0x2a,
>-  0x06, 0x42,
>-  0x08, 0x68,
>-  0x81, 0x90,
>-  0x00, 0xf0,
>-  0x50, 0x05,
>-  0x20, 0x51,
>-  0x43, 0x08,
>-  0x68, 0x80,
>-  0x80, 0x0b,
>-  0x10, 0x4c,
>-  0x12, 0x30,
>-  0x40, 0x85,
>-  0x0e, 0x04,
>-  0x18, 0x12
>-};
>-
>-const uint8_t kMaskRandom16_2[4] = {
>-  0xae, 0xae,
>-  0x79, 0x79
>-};
>-
>-const uint8_t kMaskRandom16_3[6] = {
>-  0xad, 0x2d,
>-  0x76, 0x36,
>-  0x26, 0xdb
>-};
>-
>-const uint8_t kMaskRandom16_4[8] = {
>-  0x55, 0x55,
>-  0xaa, 0xaa,
>-  0x35, 0x35,
>-  0xca, 0xca
>-};
>-
>-const uint8_t kMaskRandom16_5[10] = {
>-  0x55, 0x55,
>-  0x2a, 0x2a,
>-  0x24, 0x25,
>-  0x84, 0xc8,
>-  0x10, 0xb6
>-};
>-
>-const uint8_t kMaskRandom16_6[12] = {
>-  0x51, 0x51,
>-  0x0a, 0x2a,
>-  0xa2, 0x15,
>-  0x84, 0x4a,
>-  0x30, 0x92,
>-  0x04, 0xac
>-};
>-
>-const uint8_t kMaskRandom16_7[14] = {
>-  0x45, 0x51,
>-  0x22, 0x2a,
>-  0x91, 0x11,
>-  0x2e, 0x08,
>-  0x48, 0x34,
>-  0x90, 0x29,
>-  0x09, 0x86
>-};
>-
>-const uint8_t kMaskRandom16_8[16] = {
>-  0x20, 0x54,
>-  0x18, 0x88,
>-  0x84, 0x07,
>-  0x60, 0x48,
>-  0x12, 0x82,
>-  0x81, 0x41,
>-  0x40, 0x62,
>-  0x16, 0x30
>-};
>-
>-const uint8_t kMaskRandom16_9[18] = {
>-  0x55, 0x51,
>-  0x22, 0x2a,
>-  0x05, 0x85,
>-  0x09, 0x4a,
>-  0x84, 0x32,
>-  0xc0, 0x0d,
>-  0x20, 0xa6,
>-  0x1a, 0x09,
>-  0x44, 0x64
>-};
>-
>-const uint8_t kMaskRandom17_1[6] = {
>-  0xff, 0xff, 0x80, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom17_10[60] = {
>-  0x55, 0x8c, 0x80, 0x00, 0x00, 0x00,
>-  0xaa, 0x27, 0x00, 0x00, 0x00, 0x00,
>-  0xa5, 0x32, 0x80, 0x00, 0x00, 0x00,
>-  0x62, 0x61, 0x80, 0x00, 0x00, 0x00,
>-  0x3c, 0x5c, 0x00, 0x00, 0x00, 0x00,
>-  0x8e, 0xcc, 0x00, 0x00, 0x00, 0x00,
>-  0x6a, 0x2b, 0x00, 0x00, 0x00, 0x00,
>-  0x36, 0x32, 0x80, 0x00, 0x00, 0x00,
>-  0xd1, 0x25, 0x80, 0x00, 0x00, 0x00,
>-  0xc8, 0x02, 0x80, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom17_11[66] = {
>-  0x55, 0x8c, 0x80, 0x00, 0x00, 0x00,
>-  0xaa, 0x27, 0x00, 0x00, 0x00, 0x00,
>-  0xa5, 0x32, 0x80, 0x00, 0x00, 0x00,
>-  0x62, 0x61, 0x80, 0x00, 0x00, 0x00,
>-  0x3c, 0x5c, 0x00, 0x00, 0x00, 0x00,
>-  0x51, 0x84, 0x80, 0x00, 0x00, 0x00,
>-  0xa2, 0x27, 0x00, 0x00, 0x00, 0x00,
>-  0x95, 0x51, 0x80, 0x00, 0x00, 0x00,
>-  0x4a, 0x1a, 0x00, 0x00, 0x00, 0x00,
>-  0x30, 0x68, 0x00, 0x00, 0x00, 0x00,
>-  0x2c, 0x89, 0x00, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom17_12[72] = {
>-  0x51, 0x84, 0x80, 0x00, 0x00, 0x00,
>-  0xa2, 0x27, 0x00, 0x00, 0x00, 0x00,
>-  0x95, 0x51, 0x80, 0x00, 0x00, 0x00,
>-  0x4a, 0x1a, 0x00, 0x00, 0x00, 0x00,
>-  0x30, 0x68, 0x00, 0x00, 0x00, 0x00,
>-  0x2c, 0x89, 0x00, 0x00, 0x00, 0x00,
>-  0x55, 0x8c, 0x80, 0x00, 0x00, 0x00,
>-  0xaa, 0x27, 0x00, 0x00, 0x00, 0x00,
>-  0xa5, 0x32, 0x80, 0x00, 0x00, 0x00,
>-  0x62, 0x61, 0x80, 0x00, 0x00, 0x00,
>-  0x3c, 0x5c, 0x00, 0x00, 0x00, 0x00,
>-  0x51, 0x35, 0x00, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom17_13[78] = {
>-  0x51, 0x84, 0x80, 0x00, 0x00, 0x00,
>-  0xa2, 0x27, 0x00, 0x00, 0x00, 0x00,
>-  0x95, 0x51, 0x80, 0x00, 0x00, 0x00,
>-  0x4a, 0x1a, 0x00, 0x00, 0x00, 0x00,
>-  0x30, 0x68, 0x00, 0x00, 0x00, 0x00,
>-  0x2c, 0x89, 0x00, 0x00, 0x00, 0x00,
>-  0x15, 0x8c, 0x00, 0x00, 0x00, 0x00,
>-  0x8a, 0x47, 0x00, 0x00, 0x00, 0x00,
>-  0x25, 0x81, 0x80, 0x00, 0x00, 0x00,
>-  0x62, 0x12, 0x80, 0x00, 0x00, 0x00,
>-  0x58, 0x58, 0x00, 0x00, 0x00, 0x00,
>-  0x0e, 0x28, 0x80, 0x00, 0x00, 0x00,
>-  0x83, 0x34, 0x00, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom17_14[84] = {
>-  0x15, 0x8c, 0x00, 0x00, 0x00, 0x00,
>-  0x8a, 0x47, 0x00, 0x00, 0x00, 0x00,
>-  0x25, 0x81, 0x80, 0x00, 0x00, 0x00,
>-  0x62, 0x12, 0x80, 0x00, 0x00, 0x00,
>-  0x58, 0x58, 0x00, 0x00, 0x00, 0x00,
>-  0x0e, 0x28, 0x80, 0x00, 0x00, 0x00,
>-  0x83, 0x34, 0x00, 0x00, 0x00, 0x00,
>-  0x51, 0x84, 0x80, 0x00, 0x00, 0x00,
>-  0xa2, 0x27, 0x00, 0x00, 0x00, 0x00,
>-  0x95, 0x51, 0x80, 0x00, 0x00, 0x00,
>-  0x4a, 0x1a, 0x00, 0x00, 0x00, 0x00,
>-  0x30, 0x68, 0x00, 0x00, 0x00, 0x00,
>-  0x2c, 0x89, 0x00, 0x00, 0x00, 0x00,
>-  0xb0, 0xde, 0x80, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom17_15[90] = {
>-  0x15, 0x8c, 0x00, 0x00, 0x00, 0x00,
>-  0x8a, 0x47, 0x00, 0x00, 0x00, 0x00,
>-  0x25, 0x81, 0x80, 0x00, 0x00, 0x00,
>-  0x62, 0x12, 0x80, 0x00, 0x00, 0x00,
>-  0x58, 0x58, 0x00, 0x00, 0x00, 0x00,
>-  0x0e, 0x28, 0x80, 0x00, 0x00, 0x00,
>-  0x83, 0x34, 0x00, 0x00, 0x00, 0x00,
>-  0x25, 0x2c, 0x00, 0x00, 0x00, 0x00,
>-  0x8a, 0x91, 0x00, 0x00, 0x00, 0x00,
>-  0x91, 0xc0, 0x80, 0x00, 0x00, 0x00,
>-  0x68, 0x06, 0x80, 0x00, 0x00, 0x00,
>-  0x32, 0xc8, 0x00, 0x00, 0x00, 0x00,
>-  0x43, 0x45, 0x00, 0x00, 0x00, 0x00,
>-  0xc4, 0x30, 0x80, 0x00, 0x00, 0x00,
>-  0x1c, 0xa2, 0x00, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom17_16[96] = {
>-  0x25, 0x2c, 0x00, 0x00, 0x00, 0x00,
>-  0x8a, 0x91, 0x00, 0x00, 0x00, 0x00,
>-  0x91, 0xc0, 0x80, 0x00, 0x00, 0x00,
>-  0x68, 0x06, 0x80, 0x00, 0x00, 0x00,
>-  0x32, 0xc8, 0x00, 0x00, 0x00, 0x00,
>-  0x43, 0x45, 0x00, 0x00, 0x00, 0x00,
>-  0xc4, 0x30, 0x80, 0x00, 0x00, 0x00,
>-  0x1c, 0xa2, 0x00, 0x00, 0x00, 0x00,
>-  0x15, 0x8c, 0x00, 0x00, 0x00, 0x00,
>-  0x8a, 0x47, 0x00, 0x00, 0x00, 0x00,
>-  0x25, 0x81, 0x80, 0x00, 0x00, 0x00,
>-  0x62, 0x12, 0x80, 0x00, 0x00, 0x00,
>-  0x58, 0x58, 0x00, 0x00, 0x00, 0x00,
>-  0x0e, 0x28, 0x80, 0x00, 0x00, 0x00,
>-  0x83, 0x34, 0x00, 0x00, 0x00, 0x00,
>-  0x0a, 0x1c, 0x00, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom17_17[102] = {
>-  0x25, 0x2c, 0x00, 0x00, 0x00, 0x00,
>-  0x8a, 0x91, 0x00, 0x00, 0x00, 0x00,
>-  0x91, 0xc0, 0x80, 0x00, 0x00, 0x00,
>-  0x68, 0x06, 0x80, 0x00, 0x00, 0x00,
>-  0x32, 0xc8, 0x00, 0x00, 0x00, 0x00,
>-  0x43, 0x45, 0x00, 0x00, 0x00, 0x00,
>-  0xc4, 0x30, 0x80, 0x00, 0x00, 0x00,
>-  0x1c, 0xa2, 0x00, 0x00, 0x00, 0x00,
>-  0x25, 0x4c, 0x00, 0x00, 0x00, 0x00,
>-  0x8a, 0x66, 0x00, 0x00, 0x00, 0x00,
>-  0x91, 0x91, 0x00, 0x00, 0x00, 0x00,
>-  0x68, 0x42, 0x80, 0x00, 0x00, 0x00,
>-  0x32, 0xa4, 0x00, 0x00, 0x00, 0x00,
>-  0x43, 0x13, 0x00, 0x00, 0x00, 0x00,
>-  0xc4, 0x30, 0x80, 0x00, 0x00, 0x00,
>-  0x1c, 0x88, 0x80, 0x00, 0x00, 0x00,
>-  0x3c, 0x09, 0x00, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom17_2[12] = {
>-  0xce, 0xce, 0x00, 0x00, 0x00, 0x00,
>-  0xb9, 0x39, 0x80, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom17_3[18] = {
>-  0xcd, 0xcc, 0x00, 0x00, 0x00, 0x00,
>-  0x97, 0x27, 0x00, 0x00, 0x00, 0x00,
>-  0xb8, 0xd1, 0x80, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom17_4[24] = {
>-  0xca, 0xec, 0x00, 0x00, 0x00, 0x00,
>-  0xa9, 0x67, 0x00, 0x00, 0x00, 0x00,
>-  0x3a, 0xb1, 0x80, 0x00, 0x00, 0x00,
>-  0x55, 0x5a, 0x80, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom17_5[30] = {
>-  0x55, 0x44, 0x80, 0x00, 0x00, 0x00,
>-  0x2a, 0x66, 0x00, 0x00, 0x00, 0x00,
>-  0x25, 0xa1, 0x80, 0x00, 0x00, 0x00,
>-  0xe2, 0x12, 0x80, 0x00, 0x00, 0x00,
>-  0x99, 0x98, 0x00, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom17_6[36] = {
>-  0xd1, 0x4c, 0x00, 0x00, 0x00, 0x00,
>-  0xa2, 0xc5, 0x00, 0x00, 0x00, 0x00,
>-  0x95, 0x30, 0x80, 0x00, 0x00, 0x00,
>-  0xca, 0x0a, 0x80, 0x00, 0x00, 0x00,
>-  0xa4, 0xaa, 0x00, 0x00, 0x00, 0x00,
>-  0x78, 0x15, 0x00, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom17_7[42] = {
>-  0x15, 0x44, 0x80, 0x00, 0x00, 0x00,
>-  0x8a, 0x23, 0x00, 0x00, 0x00, 0x00,
>-  0x85, 0x91, 0x00, 0x00, 0x00, 0x00,
>-  0x32, 0x0a, 0x80, 0x00, 0x00, 0x00,
>-  0x58, 0x34, 0x00, 0x00, 0x00, 0x00,
>-  0x2c, 0x0d, 0x00, 0x00, 0x00, 0x00,
>-  0x43, 0xc8, 0x00, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom17_8[48] = {
>-  0x64, 0x16, 0x00, 0x00, 0x00, 0x00,
>-  0xa2, 0xc2, 0x00, 0x00, 0x00, 0x00,
>-  0x51, 0x60, 0x80, 0x00, 0x00, 0x00,
>-  0x4a, 0x85, 0x00, 0x00, 0x00, 0x00,
>-  0x38, 0x4c, 0x00, 0x00, 0x00, 0x00,
>-  0x89, 0x29, 0x00, 0x00, 0x00, 0x00,
>-  0x07, 0x11, 0x80, 0x00, 0x00, 0x00,
>-  0x94, 0xb0, 0x00, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom17_9[54] = {
>-  0x8e, 0xcc, 0x00, 0x00, 0x00, 0x00,
>-  0x6a, 0x2b, 0x00, 0x00, 0x00, 0x00,
>-  0x36, 0x32, 0x80, 0x00, 0x00, 0x00,
>-  0xd1, 0x25, 0x80, 0x00, 0x00, 0x00,
>-  0x55, 0x8c, 0x80, 0x00, 0x00, 0x00,
>-  0xaa, 0x27, 0x00, 0x00, 0x00, 0x00,
>-  0xa5, 0x32, 0x80, 0x00, 0x00, 0x00,
>-  0x62, 0x61, 0x80, 0x00, 0x00, 0x00,
>-  0x3c, 0x5c, 0x00, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom18_1[6] = {
>-  0xff, 0xff, 0xc0, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom18_10[60] = {
>-  0x8c, 0xc6, 0x40, 0x00, 0x00, 0x00,
>-  0x27, 0x13, 0x80, 0x00, 0x00, 0x00,
>-  0x32, 0x99, 0x40, 0x00, 0x00, 0x00,
>-  0x61, 0xb0, 0xc0, 0x00, 0x00, 0x00,
>-  0x5c, 0x2e, 0x00, 0x00, 0x00, 0x00,
>-  0xcc, 0x66, 0x00, 0x00, 0x00, 0x00,
>-  0x2b, 0x15, 0x80, 0x00, 0x00, 0x00,
>-  0x32, 0x99, 0x40, 0x00, 0x00, 0x00,
>-  0x25, 0x92, 0xc0, 0x00, 0x00, 0x00,
>-  0xfd, 0x9d, 0xc0, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom18_11[66] = {
>-  0x8c, 0xc6, 0x40, 0x00, 0x00, 0x00,
>-  0x27, 0x13, 0x80, 0x00, 0x00, 0x00,
>-  0x32, 0x99, 0x40, 0x00, 0x00, 0x00,
>-  0x61, 0xb0, 0xc0, 0x00, 0x00, 0x00,
>-  0x5c, 0x2e, 0x00, 0x00, 0x00, 0x00,
>-  0x84, 0xc2, 0x40, 0x00, 0x00, 0x00,
>-  0x27, 0x13, 0x80, 0x00, 0x00, 0x00,
>-  0x51, 0xa8, 0xc0, 0x00, 0x00, 0x00,
>-  0x1a, 0x0d, 0x00, 0x00, 0x00, 0x00,
>-  0x68, 0x34, 0x00, 0x00, 0x00, 0x00,
>-  0x89, 0x44, 0x80, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom18_12[72] = {
>-  0x84, 0xc2, 0x40, 0x00, 0x00, 0x00,
>-  0x27, 0x13, 0x80, 0x00, 0x00, 0x00,
>-  0x51, 0xa8, 0xc0, 0x00, 0x00, 0x00,
>-  0x1a, 0x0d, 0x00, 0x00, 0x00, 0x00,
>-  0x68, 0x34, 0x00, 0x00, 0x00, 0x00,
>-  0x89, 0x44, 0x80, 0x00, 0x00, 0x00,
>-  0x8c, 0xc6, 0x40, 0x00, 0x00, 0x00,
>-  0x27, 0x13, 0x80, 0x00, 0x00, 0x00,
>-  0x32, 0x99, 0x40, 0x00, 0x00, 0x00,
>-  0x61, 0xb0, 0xc0, 0x00, 0x00, 0x00,
>-  0x5c, 0x2e, 0x00, 0x00, 0x00, 0x00,
>-  0x5b, 0x0c, 0x40, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom18_13[78] = {
>-  0x84, 0xc2, 0x40, 0x00, 0x00, 0x00,
>-  0x27, 0x13, 0x80, 0x00, 0x00, 0x00,
>-  0x51, 0xa8, 0xc0, 0x00, 0x00, 0x00,
>-  0x1a, 0x0d, 0x00, 0x00, 0x00, 0x00,
>-  0x68, 0x34, 0x00, 0x00, 0x00, 0x00,
>-  0x89, 0x44, 0x80, 0x00, 0x00, 0x00,
>-  0x8c, 0x46, 0x00, 0x00, 0x00, 0x00,
>-  0x47, 0x23, 0x80, 0x00, 0x00, 0x00,
>-  0x81, 0xc0, 0xc0, 0x00, 0x00, 0x00,
>-  0x12, 0x89, 0x40, 0x00, 0x00, 0x00,
>-  0x58, 0x2c, 0x00, 0x00, 0x00, 0x00,
>-  0x28, 0x94, 0x40, 0x00, 0x00, 0x00,
>-  0x34, 0x1a, 0x00, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom18_14[84] = {
>-  0x8c, 0x46, 0x00, 0x00, 0x00, 0x00,
>-  0x47, 0x23, 0x80, 0x00, 0x00, 0x00,
>-  0x81, 0xc0, 0xc0, 0x00, 0x00, 0x00,
>-  0x12, 0x89, 0x40, 0x00, 0x00, 0x00,
>-  0x58, 0x2c, 0x00, 0x00, 0x00, 0x00,
>-  0x28, 0x94, 0x40, 0x00, 0x00, 0x00,
>-  0x34, 0x1a, 0x00, 0x00, 0x00, 0x00,
>-  0x84, 0xc2, 0x40, 0x00, 0x00, 0x00,
>-  0x27, 0x13, 0x80, 0x00, 0x00, 0x00,
>-  0x51, 0xa8, 0xc0, 0x00, 0x00, 0x00,
>-  0x1a, 0x0d, 0x00, 0x00, 0x00, 0x00,
>-  0x68, 0x34, 0x00, 0x00, 0x00, 0x00,
>-  0x89, 0x44, 0x80, 0x00, 0x00, 0x00,
>-  0x7f, 0x4f, 0xc0, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom18_15[90] = {
>-  0x8c, 0x46, 0x00, 0x00, 0x00, 0x00,
>-  0x47, 0x23, 0x80, 0x00, 0x00, 0x00,
>-  0x81, 0xc0, 0xc0, 0x00, 0x00, 0x00,
>-  0x12, 0x89, 0x40, 0x00, 0x00, 0x00,
>-  0x58, 0x2c, 0x00, 0x00, 0x00, 0x00,
>-  0x28, 0x94, 0x40, 0x00, 0x00, 0x00,
>-  0x34, 0x1a, 0x00, 0x00, 0x00, 0x00,
>-  0x2c, 0x16, 0x00, 0x00, 0x00, 0x00,
>-  0x91, 0x48, 0x80, 0x00, 0x00, 0x00,
>-  0xc0, 0xe0, 0x40, 0x00, 0x00, 0x00,
>-  0x06, 0x83, 0x40, 0x00, 0x00, 0x00,
>-  0xc8, 0x64, 0x00, 0x00, 0x00, 0x00,
>-  0x45, 0x22, 0x80, 0x00, 0x00, 0x00,
>-  0x30, 0x98, 0x40, 0x00, 0x00, 0x00,
>-  0xa2, 0x51, 0x00, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom18_16[96] = {
>-  0x2c, 0x16, 0x00, 0x00, 0x00, 0x00,
>-  0x91, 0x48, 0x80, 0x00, 0x00, 0x00,
>-  0xc0, 0xe0, 0x40, 0x00, 0x00, 0x00,
>-  0x06, 0x83, 0x40, 0x00, 0x00, 0x00,
>-  0xc8, 0x64, 0x00, 0x00, 0x00, 0x00,
>-  0x45, 0x22, 0x80, 0x00, 0x00, 0x00,
>-  0x30, 0x98, 0x40, 0x00, 0x00, 0x00,
>-  0xa2, 0x51, 0x00, 0x00, 0x00, 0x00,
>-  0x8c, 0x46, 0x00, 0x00, 0x00, 0x00,
>-  0x47, 0x23, 0x80, 0x00, 0x00, 0x00,
>-  0x81, 0xc0, 0xc0, 0x00, 0x00, 0x00,
>-  0x12, 0x89, 0x40, 0x00, 0x00, 0x00,
>-  0x58, 0x2c, 0x00, 0x00, 0x00, 0x00,
>-  0x28, 0x94, 0x40, 0x00, 0x00, 0x00,
>-  0x34, 0x1a, 0x00, 0x00, 0x00, 0x00,
>-  0xef, 0xf2, 0x00, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom18_17[102] = {
>-  0x2c, 0x16, 0x00, 0x00, 0x00, 0x00,
>-  0x91, 0x48, 0x80, 0x00, 0x00, 0x00,
>-  0xc0, 0xe0, 0x40, 0x00, 0x00, 0x00,
>-  0x06, 0x83, 0x40, 0x00, 0x00, 0x00,
>-  0xc8, 0x64, 0x00, 0x00, 0x00, 0x00,
>-  0x45, 0x22, 0x80, 0x00, 0x00, 0x00,
>-  0x30, 0x98, 0x40, 0x00, 0x00, 0x00,
>-  0xa2, 0x51, 0x00, 0x00, 0x00, 0x00,
>-  0x4c, 0x26, 0x00, 0x00, 0x00, 0x00,
>-  0x66, 0x33, 0x00, 0x00, 0x00, 0x00,
>-  0x91, 0x48, 0x80, 0x00, 0x00, 0x00,
>-  0x42, 0xa1, 0x40, 0x00, 0x00, 0x00,
>-  0xa4, 0x52, 0x00, 0x00, 0x00, 0x00,
>-  0x13, 0x09, 0x80, 0x00, 0x00, 0x00,
>-  0x30, 0x98, 0x40, 0x00, 0x00, 0x00,
>-  0x88, 0xc4, 0x40, 0x00, 0x00, 0x00,
>-  0x09, 0x04, 0x80, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom18_18[108] = {
>-  0x4c, 0x26, 0x00, 0x00, 0x00, 0x00,
>-  0x66, 0x33, 0x00, 0x00, 0x00, 0x00,
>-  0x91, 0x48, 0x80, 0x00, 0x00, 0x00,
>-  0x42, 0xa1, 0x40, 0x00, 0x00, 0x00,
>-  0xa4, 0x52, 0x00, 0x00, 0x00, 0x00,
>-  0x13, 0x09, 0x80, 0x00, 0x00, 0x00,
>-  0x30, 0x98, 0x40, 0x00, 0x00, 0x00,
>-  0x88, 0xc4, 0x40, 0x00, 0x00, 0x00,
>-  0x09, 0x04, 0x80, 0x00, 0x00, 0x00,
>-  0x2c, 0x16, 0x00, 0x00, 0x00, 0x00,
>-  0x91, 0x48, 0x80, 0x00, 0x00, 0x00,
>-  0xc0, 0xe0, 0x40, 0x00, 0x00, 0x00,
>-  0x06, 0x83, 0x40, 0x00, 0x00, 0x00,
>-  0xc8, 0x64, 0x00, 0x00, 0x00, 0x00,
>-  0x45, 0x22, 0x80, 0x00, 0x00, 0x00,
>-  0x30, 0x98, 0x40, 0x00, 0x00, 0x00,
>-  0xa2, 0x51, 0x00, 0x00, 0x00, 0x00,
>-  0xd0, 0x03, 0x40, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom18_2[12] = {
>-  0xce, 0x67, 0x00, 0x00, 0x00, 0x00,
>-  0x39, 0x9c, 0xc0, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom18_3[18] = {
>-  0xcc, 0x66, 0x00, 0x00, 0x00, 0x00,
>-  0x27, 0x15, 0x80, 0x00, 0x00, 0x00,
>-  0x92, 0xc9, 0x40, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom18_4[24] = {
>-  0xec, 0x76, 0x00, 0x00, 0x00, 0x00,
>-  0x67, 0x33, 0x80, 0x00, 0x00, 0x00,
>-  0xb1, 0xd8, 0xc0, 0x00, 0x00, 0x00,
>-  0x5a, 0xad, 0x40, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom18_5[30] = {
>-  0x4c, 0xa6, 0x40, 0x00, 0x00, 0x00,
>-  0x66, 0x33, 0x00, 0x00, 0x00, 0x00,
>-  0x19, 0xd0, 0xc0, 0x00, 0x00, 0x00,
>-  0x9c, 0x89, 0x40, 0x00, 0x00, 0x00,
>-  0xe3, 0x4c, 0x00, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom18_6[36] = {
>-  0xcc, 0x26, 0x00, 0x00, 0x00, 0x00,
>-  0x45, 0x62, 0x80, 0x00, 0x00, 0x00,
>-  0xb0, 0x98, 0x40, 0x00, 0x00, 0x00,
>-  0x8a, 0x85, 0x40, 0x00, 0x00, 0x00,
>-  0x29, 0x53, 0x00, 0x00, 0x00, 0x00,
>-  0xa6, 0x0a, 0x80, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom18_7[42] = {
>-  0x44, 0xa2, 0x40, 0x00, 0x00, 0x00,
>-  0x23, 0x11, 0x80, 0x00, 0x00, 0x00,
>-  0x91, 0x48, 0x80, 0x00, 0x00, 0x00,
>-  0x0a, 0x85, 0x40, 0x00, 0x00, 0x00,
>-  0x34, 0x1a, 0x00, 0x00, 0x00, 0x00,
>-  0x0b, 0x06, 0x80, 0x00, 0x00, 0x00,
>-  0xe0, 0x64, 0x00, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom18_8[48] = {
>-  0x16, 0x0b, 0x00, 0x00, 0x00, 0x00,
>-  0xc2, 0x61, 0x00, 0x00, 0x00, 0x00,
>-  0x60, 0xb0, 0x40, 0x00, 0x00, 0x00,
>-  0x85, 0x42, 0x80, 0x00, 0x00, 0x00,
>-  0x4c, 0x26, 0x00, 0x00, 0x00, 0x00,
>-  0x29, 0x14, 0x80, 0x00, 0x00, 0x00,
>-  0x11, 0x88, 0xc0, 0x00, 0x00, 0x00,
>-  0xb0, 0x58, 0x00, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom18_9[54] = {
>-  0x44, 0xa2, 0x40, 0x00, 0x00, 0x00,
>-  0x66, 0x26, 0x00, 0x00, 0x00, 0x00,
>-  0x90, 0x49, 0x40, 0x00, 0x00, 0x00,
>-  0x01, 0xa5, 0x80, 0x00, 0x00, 0x00,
>-  0x0e, 0x12, 0x80, 0x00, 0x00, 0x00,
>-  0x13, 0x0b, 0x00, 0x00, 0x00, 0x00,
>-  0x20, 0xd0, 0x40, 0x00, 0x00, 0x00,
>-  0xc2, 0x51, 0x00, 0x00, 0x00, 0x00,
>-  0x29, 0x0c, 0x80, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom19_1[6] = {
>-  0xff, 0xff, 0xe0, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom19_10[60] = {
>-  0x8c, 0xe3, 0x00, 0x00, 0x00, 0x00,
>-  0x27, 0x11, 0xc0, 0x00, 0x00, 0x00,
>-  0x32, 0x8d, 0x20, 0x00, 0x00, 0x00,
>-  0x61, 0x92, 0x60, 0x00, 0x00, 0x00,
>-  0x5c, 0x38, 0x80, 0x00, 0x00, 0x00,
>-  0xcc, 0x75, 0x00, 0x00, 0x00, 0x00,
>-  0x2b, 0x19, 0xc0, 0x00, 0x00, 0x00,
>-  0x32, 0xd2, 0x60, 0x00, 0x00, 0x00,
>-  0x25, 0x8e, 0xa0, 0x00, 0x00, 0x00,
>-  0x50, 0x88, 0xc0, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom19_11[66] = {
>-  0x8c, 0xe3, 0x00, 0x00, 0x00, 0x00,
>-  0x27, 0x11, 0xc0, 0x00, 0x00, 0x00,
>-  0x32, 0x8d, 0x20, 0x00, 0x00, 0x00,
>-  0x61, 0x92, 0x60, 0x00, 0x00, 0x00,
>-  0x5c, 0x38, 0x80, 0x00, 0x00, 0x00,
>-  0x84, 0x87, 0x00, 0x00, 0x00, 0x00,
>-  0x27, 0x19, 0x80, 0x00, 0x00, 0x00,
>-  0x51, 0x88, 0x60, 0x00, 0x00, 0x00,
>-  0x1a, 0x22, 0xa0, 0x00, 0x00, 0x00,
>-  0x68, 0x44, 0x40, 0x00, 0x00, 0x00,
>-  0x89, 0x70, 0x00, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom19_12[72] = {
>-  0x84, 0x87, 0x00, 0x00, 0x00, 0x00,
>-  0x27, 0x19, 0x80, 0x00, 0x00, 0x00,
>-  0x51, 0x88, 0x60, 0x00, 0x00, 0x00,
>-  0x1a, 0x22, 0xa0, 0x00, 0x00, 0x00,
>-  0x68, 0x44, 0x40, 0x00, 0x00, 0x00,
>-  0x89, 0x70, 0x00, 0x00, 0x00, 0x00,
>-  0x8c, 0xe3, 0x00, 0x00, 0x00, 0x00,
>-  0x27, 0x11, 0xc0, 0x00, 0x00, 0x00,
>-  0x32, 0x8d, 0x20, 0x00, 0x00, 0x00,
>-  0x61, 0x92, 0x60, 0x00, 0x00, 0x00,
>-  0x5c, 0x38, 0x80, 0x00, 0x00, 0x00,
>-  0x90, 0xc8, 0x80, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom19_13[78] = {
>-  0x84, 0x87, 0x00, 0x00, 0x00, 0x00,
>-  0x27, 0x19, 0x80, 0x00, 0x00, 0x00,
>-  0x51, 0x88, 0x60, 0x00, 0x00, 0x00,
>-  0x1a, 0x22, 0xa0, 0x00, 0x00, 0x00,
>-  0x68, 0x44, 0x40, 0x00, 0x00, 0x00,
>-  0x89, 0x70, 0x00, 0x00, 0x00, 0x00,
>-  0x8c, 0x23, 0x00, 0x00, 0x00, 0x00,
>-  0x47, 0x19, 0x80, 0x00, 0x00, 0x00,
>-  0x81, 0x88, 0x60, 0x00, 0x00, 0x00,
>-  0x12, 0x86, 0x20, 0x00, 0x00, 0x00,
>-  0x58, 0x14, 0x40, 0x00, 0x00, 0x00,
>-  0x28, 0xca, 0x00, 0x00, 0x00, 0x00,
>-  0x34, 0x60, 0x80, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom19_14[84] = {
>-  0x8c, 0x23, 0x00, 0x00, 0x00, 0x00,
>-  0x47, 0x19, 0x80, 0x00, 0x00, 0x00,
>-  0x81, 0x88, 0x60, 0x00, 0x00, 0x00,
>-  0x12, 0x86, 0x20, 0x00, 0x00, 0x00,
>-  0x58, 0x14, 0x40, 0x00, 0x00, 0x00,
>-  0x28, 0xca, 0x00, 0x00, 0x00, 0x00,
>-  0x34, 0x60, 0x80, 0x00, 0x00, 0x00,
>-  0x84, 0x87, 0x00, 0x00, 0x00, 0x00,
>-  0x27, 0x19, 0x80, 0x00, 0x00, 0x00,
>-  0x51, 0x88, 0x60, 0x00, 0x00, 0x00,
>-  0x1a, 0x22, 0xa0, 0x00, 0x00, 0x00,
>-  0x68, 0x44, 0x40, 0x00, 0x00, 0x00,
>-  0x89, 0x70, 0x00, 0x00, 0x00, 0x00,
>-  0x6e, 0x27, 0x60, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom19_15[90] = {
>-  0x8c, 0x23, 0x00, 0x00, 0x00, 0x00,
>-  0x47, 0x19, 0x80, 0x00, 0x00, 0x00,
>-  0x81, 0x88, 0x60, 0x00, 0x00, 0x00,
>-  0x12, 0x86, 0x20, 0x00, 0x00, 0x00,
>-  0x58, 0x14, 0x40, 0x00, 0x00, 0x00,
>-  0x28, 0xca, 0x00, 0x00, 0x00, 0x00,
>-  0x34, 0x60, 0x80, 0x00, 0x00, 0x00,
>-  0x2c, 0x16, 0x00, 0x00, 0x00, 0x00,
>-  0x91, 0x40, 0xc0, 0x00, 0x00, 0x00,
>-  0xc0, 0xd0, 0x20, 0x00, 0x00, 0x00,
>-  0x06, 0x82, 0xa0, 0x00, 0x00, 0x00,
>-  0xc8, 0x0c, 0x40, 0x00, 0x00, 0x00,
>-  0x45, 0x61, 0x00, 0x00, 0x00, 0x00,
>-  0x30, 0x91, 0x40, 0x00, 0x00, 0x00,
>-  0xa2, 0x28, 0x20, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom19_16[96] = {
>-  0x2c, 0x16, 0x00, 0x00, 0x00, 0x00,
>-  0x91, 0x40, 0xc0, 0x00, 0x00, 0x00,
>-  0xc0, 0xd0, 0x20, 0x00, 0x00, 0x00,
>-  0x06, 0x82, 0xa0, 0x00, 0x00, 0x00,
>-  0xc8, 0x0c, 0x40, 0x00, 0x00, 0x00,
>-  0x45, 0x61, 0x00, 0x00, 0x00, 0x00,
>-  0x30, 0x91, 0x40, 0x00, 0x00, 0x00,
>-  0xa2, 0x28, 0x20, 0x00, 0x00, 0x00,
>-  0x8c, 0x23, 0x00, 0x00, 0x00, 0x00,
>-  0x47, 0x19, 0x80, 0x00, 0x00, 0x00,
>-  0x81, 0x88, 0x60, 0x00, 0x00, 0x00,
>-  0x12, 0x86, 0x20, 0x00, 0x00, 0x00,
>-  0x58, 0x14, 0x40, 0x00, 0x00, 0x00,
>-  0x28, 0xca, 0x00, 0x00, 0x00, 0x00,
>-  0x34, 0x60, 0x80, 0x00, 0x00, 0x00,
>-  0x7e, 0x75, 0xe0, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom19_17[102] = {
>-  0x2c, 0x16, 0x00, 0x00, 0x00, 0x00,
>-  0x91, 0x40, 0xc0, 0x00, 0x00, 0x00,
>-  0xc0, 0xd0, 0x20, 0x00, 0x00, 0x00,
>-  0x06, 0x82, 0xa0, 0x00, 0x00, 0x00,
>-  0xc8, 0x0c, 0x40, 0x00, 0x00, 0x00,
>-  0x45, 0x61, 0x00, 0x00, 0x00, 0x00,
>-  0x30, 0x91, 0x40, 0x00, 0x00, 0x00,
>-  0xa2, 0x28, 0x20, 0x00, 0x00, 0x00,
>-  0x4c, 0x27, 0x00, 0x00, 0x00, 0x00,
>-  0x66, 0x71, 0x80, 0x00, 0x00, 0x00,
>-  0x91, 0x40, 0xe0, 0x00, 0x00, 0x00,
>-  0x42, 0x90, 0xa0, 0x00, 0x00, 0x00,
>-  0xa4, 0x29, 0x40, 0x00, 0x00, 0x00,
>-  0x13, 0x5a, 0x00, 0x00, 0x00, 0x00,
>-  0x30, 0x93, 0x40, 0x00, 0x00, 0x00,
>-  0x88, 0xac, 0x20, 0x00, 0x00, 0x00,
>-  0x09, 0x0c, 0xc0, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom19_18[108] = {
>-  0x4c, 0x27, 0x00, 0x00, 0x00, 0x00,
>-  0x66, 0x71, 0x80, 0x00, 0x00, 0x00,
>-  0x91, 0x40, 0xe0, 0x00, 0x00, 0x00,
>-  0x42, 0x90, 0xa0, 0x00, 0x00, 0x00,
>-  0xa4, 0x29, 0x40, 0x00, 0x00, 0x00,
>-  0x13, 0x5a, 0x00, 0x00, 0x00, 0x00,
>-  0x30, 0x93, 0x40, 0x00, 0x00, 0x00,
>-  0x88, 0xac, 0x20, 0x00, 0x00, 0x00,
>-  0x09, 0x0c, 0xc0, 0x00, 0x00, 0x00,
>-  0x2c, 0x16, 0x00, 0x00, 0x00, 0x00,
>-  0x91, 0x40, 0xc0, 0x00, 0x00, 0x00,
>-  0xc0, 0xd0, 0x20, 0x00, 0x00, 0x00,
>-  0x06, 0x82, 0xa0, 0x00, 0x00, 0x00,
>-  0xc8, 0x0c, 0x40, 0x00, 0x00, 0x00,
>-  0x45, 0x61, 0x00, 0x00, 0x00, 0x00,
>-  0x30, 0x91, 0x40, 0x00, 0x00, 0x00,
>-  0xa2, 0x28, 0x20, 0x00, 0x00, 0x00,
>-  0x51, 0x97, 0x20, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom19_19[114] = {
>-  0x4c, 0x27, 0x00, 0x00, 0x00, 0x00,
>-  0x66, 0x71, 0x80, 0x00, 0x00, 0x00,
>-  0x91, 0x40, 0xe0, 0x00, 0x00, 0x00,
>-  0x42, 0x90, 0xa0, 0x00, 0x00, 0x00,
>-  0xa4, 0x29, 0x40, 0x00, 0x00, 0x00,
>-  0x13, 0x5a, 0x00, 0x00, 0x00, 0x00,
>-  0x30, 0x93, 0x40, 0x00, 0x00, 0x00,
>-  0x88, 0xac, 0x20, 0x00, 0x00, 0x00,
>-  0x09, 0x0c, 0xc0, 0x00, 0x00, 0x00,
>-  0x4c, 0x26, 0x00, 0x00, 0x00, 0x00,
>-  0x66, 0x28, 0x80, 0x00, 0x00, 0x00,
>-  0x91, 0x50, 0x20, 0x00, 0x00, 0x00,
>-  0x42, 0x82, 0x60, 0x00, 0x00, 0x00,
>-  0xa4, 0x01, 0xc0, 0x00, 0x00, 0x00,
>-  0x13, 0x43, 0x00, 0x00, 0x00, 0x00,
>-  0x30, 0x94, 0x80, 0x00, 0x00, 0x00,
>-  0x88, 0xa1, 0x20, 0x00, 0x00, 0x00,
>-  0x09, 0x4c, 0x00, 0x00, 0x00, 0x00,
>-  0xcd, 0x98, 0x40, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom19_2[12] = {
>-  0xce, 0x77, 0x00, 0x00, 0x00, 0x00,
>-  0x39, 0xcc, 0xe0, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom19_3[18] = {
>-  0xcc, 0x67, 0x00, 0x00, 0x00, 0x00,
>-  0x27, 0x2c, 0xc0, 0x00, 0x00, 0x00,
>-  0x92, 0xd2, 0x60, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom19_4[24] = {
>-  0xec, 0x73, 0x00, 0x00, 0x00, 0x00,
>-  0x67, 0x19, 0xc0, 0x00, 0x00, 0x00,
>-  0xb1, 0xcc, 0x60, 0x00, 0x00, 0x00,
>-  0x5a, 0x96, 0xa0, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom19_5[30] = {
>-  0x4c, 0xe7, 0x00, 0x00, 0x00, 0x00,
>-  0x66, 0x31, 0xc0, 0x00, 0x00, 0x00,
>-  0xa1, 0xcc, 0x60, 0x00, 0x00, 0x00,
>-  0x92, 0xa6, 0xa0, 0x00, 0x00, 0x00,
>-  0xb8, 0x99, 0x80, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom19_6[36] = {
>-  0x4c, 0x36, 0x00, 0x00, 0x00, 0x00,
>-  0x45, 0x68, 0x80, 0x00, 0x00, 0x00,
>-  0x30, 0xd0, 0x60, 0x00, 0x00, 0x00,
>-  0x8a, 0x82, 0xa0, 0x00, 0x00, 0x00,
>-  0x26, 0x0b, 0x40, 0x00, 0x00, 0x00,
>-  0x95, 0x45, 0x00, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom19_7[42] = {
>-  0xc4, 0xa3, 0x00, 0x00, 0x00, 0x00,
>-  0x23, 0x19, 0x80, 0x00, 0x00, 0x00,
>-  0x91, 0x1c, 0x20, 0x00, 0x00, 0x00,
>-  0x4a, 0x82, 0xa0, 0x00, 0x00, 0x00,
>-  0x34, 0x49, 0x40, 0x00, 0x00, 0x00,
>-  0x8b, 0x4a, 0x00, 0x00, 0x00, 0x00,
>-  0xc8, 0x24, 0xc0, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom19_8[48] = {
>-  0x16, 0x13, 0x80, 0x00, 0x00, 0x00,
>-  0xc2, 0x44, 0xc0, 0x00, 0x00, 0x00,
>-  0x60, 0xe8, 0x20, 0x00, 0x00, 0x00,
>-  0x85, 0x12, 0x60, 0x00, 0x00, 0x00,
>-  0xcc, 0x21, 0x40, 0x00, 0x00, 0x00,
>-  0x29, 0x63, 0x00, 0x00, 0x00, 0x00,
>-  0x11, 0x98, 0xc0, 0x00, 0x00, 0x00,
>-  0xb0, 0x0c, 0x60, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom19_9[54] = {
>-  0x44, 0xa7, 0x00, 0x00, 0x00, 0x00,
>-  0x66, 0x70, 0x80, 0x00, 0x00, 0x00,
>-  0x12, 0xc0, 0xe0, 0x00, 0x00, 0x00,
>-  0xc3, 0x10, 0xa0, 0x00, 0x00, 0x00,
>-  0x8c, 0x29, 0x40, 0x00, 0x00, 0x00,
>-  0x11, 0x5b, 0x00, 0x00, 0x00, 0x00,
>-  0x21, 0x93, 0x40, 0x00, 0x00, 0x00,
>-  0xa2, 0x2c, 0x00, 0x00, 0x00, 0x00,
>-  0x18, 0x0c, 0xe0, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom1_1[2] = {
>-  0x80, 0x00
>-};
>-
>-const uint8_t kMaskRandom20_1[6] = {
>-  0xff, 0xff, 0xf0, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom20_10[60] = {
>-  0x4c, 0x13, 0x00, 0x00, 0x00, 0x00,
>-  0x51, 0x14, 0x40, 0x00, 0x00, 0x00,
>-  0xa0, 0x68, 0x10, 0x00, 0x00, 0x00,
>-  0x04, 0xc1, 0x30, 0x00, 0x00, 0x00,
>-  0x03, 0x80, 0xe0, 0x00, 0x00, 0x00,
>-  0x86, 0x21, 0x80, 0x00, 0x00, 0x00,
>-  0x29, 0x0a, 0x40, 0x00, 0x00, 0x00,
>-  0x42, 0x50, 0x90, 0x00, 0x00, 0x00,
>-  0x98, 0x26, 0x00, 0x00, 0x00, 0x00,
>-  0x30, 0x8c, 0x20, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom20_11[66] = {
>-  0xc6, 0x31, 0x80, 0x00, 0x00, 0x00,
>-  0x23, 0x88, 0xe0, 0x00, 0x00, 0x00,
>-  0x1a, 0x46, 0x90, 0x00, 0x00, 0x00,
>-  0x24, 0xc9, 0x30, 0x00, 0x00, 0x00,
>-  0x71, 0x1c, 0x40, 0x00, 0x00, 0x00,
>-  0x0e, 0x03, 0x80, 0x00, 0x00, 0x00,
>-  0x33, 0x0c, 0xc0, 0x00, 0x00, 0x00,
>-  0x10, 0xc4, 0x30, 0x00, 0x00, 0x00,
>-  0x45, 0x51, 0x50, 0x00, 0x00, 0x00,
>-  0x88, 0xa2, 0x20, 0x00, 0x00, 0x00,
>-  0xe0, 0x38, 0x00, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom20_12[72] = {
>-  0x0e, 0x03, 0x80, 0x00, 0x00, 0x00,
>-  0x33, 0x0c, 0xc0, 0x00, 0x00, 0x00,
>-  0x10, 0xc4, 0x30, 0x00, 0x00, 0x00,
>-  0x45, 0x51, 0x50, 0x00, 0x00, 0x00,
>-  0x88, 0xa2, 0x20, 0x00, 0x00, 0x00,
>-  0xe0, 0x38, 0x00, 0x00, 0x00, 0x00,
>-  0xc6, 0x31, 0x80, 0x00, 0x00, 0x00,
>-  0x23, 0x88, 0xe0, 0x00, 0x00, 0x00,
>-  0x1a, 0x46, 0x90, 0x00, 0x00, 0x00,
>-  0x24, 0xc9, 0x30, 0x00, 0x00, 0x00,
>-  0x71, 0x1c, 0x40, 0x00, 0x00, 0x00,
>-  0xf5, 0xdc, 0x40, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom20_13[78] = {
>-  0x0e, 0x03, 0x80, 0x00, 0x00, 0x00,
>-  0x33, 0x0c, 0xc0, 0x00, 0x00, 0x00,
>-  0x10, 0xc4, 0x30, 0x00, 0x00, 0x00,
>-  0x45, 0x51, 0x50, 0x00, 0x00, 0x00,
>-  0x88, 0xa2, 0x20, 0x00, 0x00, 0x00,
>-  0xe0, 0x38, 0x00, 0x00, 0x00, 0x00,
>-  0x46, 0x11, 0x80, 0x00, 0x00, 0x00,
>-  0x33, 0x0c, 0xc0, 0x00, 0x00, 0x00,
>-  0x10, 0xc4, 0x30, 0x00, 0x00, 0x00,
>-  0x0c, 0x43, 0x10, 0x00, 0x00, 0x00,
>-  0x28, 0x8a, 0x20, 0x00, 0x00, 0x00,
>-  0x94, 0x25, 0x00, 0x00, 0x00, 0x00,
>-  0xc1, 0x30, 0x40, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom20_14[84] = {
>-  0x46, 0x11, 0x80, 0x00, 0x00, 0x00,
>-  0x33, 0x0c, 0xc0, 0x00, 0x00, 0x00,
>-  0x10, 0xc4, 0x30, 0x00, 0x00, 0x00,
>-  0x0c, 0x43, 0x10, 0x00, 0x00, 0x00,
>-  0x28, 0x8a, 0x20, 0x00, 0x00, 0x00,
>-  0x94, 0x25, 0x00, 0x00, 0x00, 0x00,
>-  0xc1, 0x30, 0x40, 0x00, 0x00, 0x00,
>-  0x0e, 0x03, 0x80, 0x00, 0x00, 0x00,
>-  0x33, 0x0c, 0xc0, 0x00, 0x00, 0x00,
>-  0x10, 0xc4, 0x30, 0x00, 0x00, 0x00,
>-  0x45, 0x51, 0x50, 0x00, 0x00, 0x00,
>-  0x88, 0xa2, 0x20, 0x00, 0x00, 0x00,
>-  0xe0, 0x38, 0x00, 0x00, 0x00, 0x00,
>-  0x56, 0x3e, 0x20, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom20_15[90] = {
>-  0x46, 0x11, 0x80, 0x00, 0x00, 0x00,
>-  0x33, 0x0c, 0xc0, 0x00, 0x00, 0x00,
>-  0x10, 0xc4, 0x30, 0x00, 0x00, 0x00,
>-  0x0c, 0x43, 0x10, 0x00, 0x00, 0x00,
>-  0x28, 0x8a, 0x20, 0x00, 0x00, 0x00,
>-  0x94, 0x25, 0x00, 0x00, 0x00, 0x00,
>-  0xc1, 0x30, 0x40, 0x00, 0x00, 0x00,
>-  0x2c, 0x0b, 0x00, 0x00, 0x00, 0x00,
>-  0x81, 0xa0, 0x60, 0x00, 0x00, 0x00,
>-  0xa0, 0x68, 0x10, 0x00, 0x00, 0x00,
>-  0x05, 0x41, 0x50, 0x00, 0x00, 0x00,
>-  0x18, 0x86, 0x20, 0x00, 0x00, 0x00,
>-  0xc2, 0x30, 0x80, 0x00, 0x00, 0x00,
>-  0x22, 0x88, 0xa0, 0x00, 0x00, 0x00,
>-  0x50, 0x54, 0x10, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom20_16[96] = {
>-  0x2c, 0x0b, 0x00, 0x00, 0x00, 0x00,
>-  0x81, 0xa0, 0x60, 0x00, 0x00, 0x00,
>-  0xa0, 0x68, 0x10, 0x00, 0x00, 0x00,
>-  0x05, 0x41, 0x50, 0x00, 0x00, 0x00,
>-  0x18, 0x86, 0x20, 0x00, 0x00, 0x00,
>-  0xc2, 0x30, 0x80, 0x00, 0x00, 0x00,
>-  0x22, 0x88, 0xa0, 0x00, 0x00, 0x00,
>-  0x50, 0x54, 0x10, 0x00, 0x00, 0x00,
>-  0x46, 0x11, 0x80, 0x00, 0x00, 0x00,
>-  0x33, 0x0c, 0xc0, 0x00, 0x00, 0x00,
>-  0x10, 0xc4, 0x30, 0x00, 0x00, 0x00,
>-  0x0c, 0x43, 0x10, 0x00, 0x00, 0x00,
>-  0x28, 0x8a, 0x20, 0x00, 0x00, 0x00,
>-  0x94, 0x25, 0x00, 0x00, 0x00, 0x00,
>-  0xc1, 0x30, 0x40, 0x00, 0x00, 0x00,
>-  0x28, 0x1c, 0x60, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom20_17[102] = {
>-  0x2c, 0x0b, 0x00, 0x00, 0x00, 0x00,
>-  0x81, 0xa0, 0x60, 0x00, 0x00, 0x00,
>-  0xa0, 0x68, 0x10, 0x00, 0x00, 0x00,
>-  0x05, 0x41, 0x50, 0x00, 0x00, 0x00,
>-  0x18, 0x86, 0x20, 0x00, 0x00, 0x00,
>-  0xc2, 0x30, 0x80, 0x00, 0x00, 0x00,
>-  0x22, 0x88, 0xa0, 0x00, 0x00, 0x00,
>-  0x50, 0x54, 0x10, 0x00, 0x00, 0x00,
>-  0x4e, 0x13, 0x80, 0x00, 0x00, 0x00,
>-  0xe3, 0x38, 0xc0, 0x00, 0x00, 0x00,
>-  0x81, 0xe0, 0x70, 0x00, 0x00, 0x00,
>-  0x21, 0x48, 0x50, 0x00, 0x00, 0x00,
>-  0x52, 0x94, 0xa0, 0x00, 0x00, 0x00,
>-  0xb4, 0x2d, 0x00, 0x00, 0x00, 0x00,
>-  0x26, 0x89, 0xa0, 0x00, 0x00, 0x00,
>-  0x58, 0x56, 0x10, 0x00, 0x00, 0x00,
>-  0x19, 0x86, 0x60, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom20_18[108] = {
>-  0x4e, 0x13, 0x80, 0x00, 0x00, 0x00,
>-  0xe3, 0x38, 0xc0, 0x00, 0x00, 0x00,
>-  0x81, 0xe0, 0x70, 0x00, 0x00, 0x00,
>-  0x21, 0x48, 0x50, 0x00, 0x00, 0x00,
>-  0x52, 0x94, 0xa0, 0x00, 0x00, 0x00,
>-  0xb4, 0x2d, 0x00, 0x00, 0x00, 0x00,
>-  0x26, 0x89, 0xa0, 0x00, 0x00, 0x00,
>-  0x58, 0x56, 0x10, 0x00, 0x00, 0x00,
>-  0x19, 0x86, 0x60, 0x00, 0x00, 0x00,
>-  0x2c, 0x0b, 0x00, 0x00, 0x00, 0x00,
>-  0x81, 0xa0, 0x60, 0x00, 0x00, 0x00,
>-  0xa0, 0x68, 0x10, 0x00, 0x00, 0x00,
>-  0x05, 0x41, 0x50, 0x00, 0x00, 0x00,
>-  0x18, 0x86, 0x20, 0x00, 0x00, 0x00,
>-  0xc2, 0x30, 0x80, 0x00, 0x00, 0x00,
>-  0x22, 0x88, 0xa0, 0x00, 0x00, 0x00,
>-  0x50, 0x54, 0x10, 0x00, 0x00, 0x00,
>-  0x21, 0x7b, 0xf0, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom20_19[114] = {
>-  0x4e, 0x13, 0x80, 0x00, 0x00, 0x00,
>-  0xe3, 0x38, 0xc0, 0x00, 0x00, 0x00,
>-  0x81, 0xe0, 0x70, 0x00, 0x00, 0x00,
>-  0x21, 0x48, 0x50, 0x00, 0x00, 0x00,
>-  0x52, 0x94, 0xa0, 0x00, 0x00, 0x00,
>-  0xb4, 0x2d, 0x00, 0x00, 0x00, 0x00,
>-  0x26, 0x89, 0xa0, 0x00, 0x00, 0x00,
>-  0x58, 0x56, 0x10, 0x00, 0x00, 0x00,
>-  0x19, 0x86, 0x60, 0x00, 0x00, 0x00,
>-  0x4c, 0x13, 0x00, 0x00, 0x00, 0x00,
>-  0x51, 0x14, 0x40, 0x00, 0x00, 0x00,
>-  0xa0, 0x68, 0x10, 0x00, 0x00, 0x00,
>-  0x04, 0xc1, 0x30, 0x00, 0x00, 0x00,
>-  0x03, 0x80, 0xe0, 0x00, 0x00, 0x00,
>-  0x86, 0x21, 0x80, 0x00, 0x00, 0x00,
>-  0x29, 0x0a, 0x40, 0x00, 0x00, 0x00,
>-  0x42, 0x50, 0x90, 0x00, 0x00, 0x00,
>-  0x98, 0x26, 0x00, 0x00, 0x00, 0x00,
>-  0x30, 0x8c, 0x20, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom20_2[12] = {
>-  0xee, 0x3b, 0x80, 0x00, 0x00, 0x00,
>-  0x99, 0xe6, 0x70, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom20_20[120] = {
>-  0x4c, 0x13, 0x00, 0x00, 0x00, 0x00,
>-  0x51, 0x14, 0x40, 0x00, 0x00, 0x00,
>-  0xa0, 0x68, 0x10, 0x00, 0x00, 0x00,
>-  0x04, 0xc1, 0x30, 0x00, 0x00, 0x00,
>-  0x03, 0x80, 0xe0, 0x00, 0x00, 0x00,
>-  0x86, 0x21, 0x80, 0x00, 0x00, 0x00,
>-  0x29, 0x0a, 0x40, 0x00, 0x00, 0x00,
>-  0x42, 0x50, 0x90, 0x00, 0x00, 0x00,
>-  0x98, 0x26, 0x00, 0x00, 0x00, 0x00,
>-  0x30, 0x8c, 0x20, 0x00, 0x00, 0x00,
>-  0x4e, 0x13, 0x80, 0x00, 0x00, 0x00,
>-  0xe3, 0x38, 0xc0, 0x00, 0x00, 0x00,
>-  0x81, 0xe0, 0x70, 0x00, 0x00, 0x00,
>-  0x21, 0x48, 0x50, 0x00, 0x00, 0x00,
>-  0x52, 0x94, 0xa0, 0x00, 0x00, 0x00,
>-  0xb4, 0x2d, 0x00, 0x00, 0x00, 0x00,
>-  0x26, 0x89, 0xa0, 0x00, 0x00, 0x00,
>-  0x58, 0x56, 0x10, 0x00, 0x00, 0x00,
>-  0x19, 0x86, 0x60, 0x00, 0x00, 0x00,
>-  0xf7, 0x8d, 0xa0, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom20_3[18] = {
>-  0xce, 0x33, 0x80, 0x00, 0x00, 0x00,
>-  0x55, 0x95, 0x60, 0x00, 0x00, 0x00,
>-  0xb1, 0x6a, 0x30, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom20_4[24] = {
>-  0xe6, 0x39, 0x80, 0x00, 0x00, 0x00,
>-  0x33, 0x8c, 0xe0, 0x00, 0x00, 0x00,
>-  0x98, 0xe6, 0x30, 0x00, 0x00, 0x00,
>-  0x2d, 0x4b, 0x50, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom20_5[30] = {
>-  0xce, 0x33, 0x80, 0x00, 0x00, 0x00,
>-  0x63, 0x98, 0xe0, 0x00, 0x00, 0x00,
>-  0x98, 0xe5, 0x30, 0x00, 0x00, 0x00,
>-  0x2b, 0x53, 0x50, 0x00, 0x00, 0x00,
>-  0xb4, 0x5c, 0xa0, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom20_6[36] = {
>-  0x4c, 0x1b, 0x00, 0x00, 0x00, 0x00,
>-  0x51, 0x34, 0x40, 0x00, 0x00, 0x00,
>-  0x20, 0xe8, 0x30, 0x00, 0x00, 0x00,
>-  0x85, 0x41, 0x50, 0x00, 0x00, 0x00,
>-  0x06, 0x86, 0xa0, 0x00, 0x00, 0x00,
>-  0x9a, 0x21, 0x80, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom20_7[42] = {
>-  0x4e, 0x11, 0x80, 0x00, 0x00, 0x00,
>-  0x33, 0x2c, 0x00, 0x00, 0x00, 0x00,
>-  0x10, 0x0e, 0xb0, 0x00, 0x00, 0x00,
>-  0x81, 0x51, 0x50, 0x00, 0x00, 0x00,
>-  0x24, 0xc4, 0xa0, 0x00, 0x00, 0x00,
>-  0xd4, 0x23, 0x00, 0x00, 0x00, 0x00,
>-  0x0c, 0xa2, 0x60, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom20_8[48] = {
>-  0x27, 0x09, 0xc0, 0x00, 0x00, 0x00,
>-  0x89, 0xa2, 0x60, 0x00, 0x00, 0x00,
>-  0xd0, 0x74, 0x10, 0x00, 0x00, 0x00,
>-  0x24, 0xc9, 0x30, 0x00, 0x00, 0x00,
>-  0xe2, 0x90, 0xa0, 0x00, 0x00, 0x00,
>-  0xc6, 0x31, 0x80, 0x00, 0x00, 0x00,
>-  0x31, 0x8c, 0x60, 0x00, 0x00, 0x00,
>-  0x18, 0xc6, 0x30, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom20_9[54] = {
>-  0x4e, 0x13, 0x80, 0x00, 0x00, 0x00,
>-  0x62, 0x38, 0xc0, 0x00, 0x00, 0x00,
>-  0x81, 0xe0, 0x70, 0x00, 0x00, 0x00,
>-  0xe1, 0x48, 0x50, 0x00, 0x00, 0x00,
>-  0x13, 0x94, 0xa0, 0x00, 0x00, 0x00,
>-  0xb4, 0x2d, 0x00, 0x00, 0x00, 0x00,
>-  0x26, 0x89, 0xa0, 0x00, 0x00, 0x00,
>-  0x58, 0x56, 0x10, 0x00, 0x00, 0x00,
>-  0x49, 0x86, 0x50, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom21_1[6] = {
>-  0xff, 0xff, 0xf8, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom21_10[60] = {
>-  0x4c, 0x19, 0x10, 0x00, 0x00, 0x00,
>-  0x51, 0x14, 0x50, 0x00, 0x00, 0x00,
>-  0xa0, 0x6a, 0x40, 0x00, 0x00, 0x00,
>-  0x04, 0xc1, 0x30, 0x00, 0x00, 0x00,
>-  0x03, 0xb4, 0x00, 0x00, 0x00, 0x00,
>-  0x86, 0x20, 0x90, 0x00, 0x00, 0x00,
>-  0x29, 0x08, 0x48, 0x00, 0x00, 0x00,
>-  0x42, 0x43, 0x08, 0x00, 0x00, 0x00,
>-  0x98, 0x12, 0x80, 0x00, 0x00, 0x00,
>-  0x30, 0x84, 0xa8, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom21_11[66] = {
>-  0xc6, 0x21, 0xa0, 0x00, 0x00, 0x00,
>-  0x23, 0x88, 0xc8, 0x00, 0x00, 0x00,
>-  0x1a, 0x45, 0x88, 0x00, 0x00, 0x00,
>-  0x24, 0xd3, 0x08, 0x00, 0x00, 0x00,
>-  0x71, 0x10, 0x70, 0x00, 0x00, 0x00,
>-  0x0e, 0x19, 0x10, 0x00, 0x00, 0x00,
>-  0x33, 0x14, 0x50, 0x00, 0x00, 0x00,
>-  0x10, 0xc3, 0x28, 0x00, 0x00, 0x00,
>-  0x45, 0x68, 0x48, 0x00, 0x00, 0x00,
>-  0x88, 0x84, 0xa8, 0x00, 0x00, 0x00,
>-  0xe0, 0x22, 0x90, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom21_12[72] = {
>-  0x0e, 0x19, 0x10, 0x00, 0x00, 0x00,
>-  0x33, 0x14, 0x50, 0x00, 0x00, 0x00,
>-  0x10, 0xc3, 0x28, 0x00, 0x00, 0x00,
>-  0x45, 0x68, 0x48, 0x00, 0x00, 0x00,
>-  0x88, 0x84, 0xa8, 0x00, 0x00, 0x00,
>-  0xe0, 0x22, 0x90, 0x00, 0x00, 0x00,
>-  0xc6, 0x21, 0xa0, 0x00, 0x00, 0x00,
>-  0x23, 0x88, 0xc8, 0x00, 0x00, 0x00,
>-  0x1a, 0x45, 0x88, 0x00, 0x00, 0x00,
>-  0x24, 0xd3, 0x08, 0x00, 0x00, 0x00,
>-  0x71, 0x10, 0x70, 0x00, 0x00, 0x00,
>-  0xa0, 0x65, 0x18, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom21_13[78] = {
>-  0x0e, 0x19, 0x10, 0x00, 0x00, 0x00,
>-  0x33, 0x14, 0x50, 0x00, 0x00, 0x00,
>-  0x10, 0xc3, 0x28, 0x00, 0x00, 0x00,
>-  0x45, 0x68, 0x48, 0x00, 0x00, 0x00,
>-  0x88, 0x84, 0xa8, 0x00, 0x00, 0x00,
>-  0xe0, 0x22, 0x90, 0x00, 0x00, 0x00,
>-  0x46, 0x11, 0x90, 0x00, 0x00, 0x00,
>-  0x33, 0x0c, 0xc8, 0x00, 0x00, 0x00,
>-  0x10, 0xe4, 0x60, 0x00, 0x00, 0x00,
>-  0x0c, 0x69, 0x08, 0x00, 0x00, 0x00,
>-  0x28, 0x94, 0x28, 0x00, 0x00, 0x00,
>-  0x94, 0x21, 0x30, 0x00, 0x00, 0x00,
>-  0xc1, 0x02, 0x58, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom21_14[84] = {
>-  0x46, 0x11, 0x90, 0x00, 0x00, 0x00,
>-  0x33, 0x0c, 0xc8, 0x00, 0x00, 0x00,
>-  0x10, 0xe4, 0x60, 0x00, 0x00, 0x00,
>-  0x0c, 0x69, 0x08, 0x00, 0x00, 0x00,
>-  0x28, 0x94, 0x28, 0x00, 0x00, 0x00,
>-  0x94, 0x21, 0x30, 0x00, 0x00, 0x00,
>-  0xc1, 0x02, 0x58, 0x00, 0x00, 0x00,
>-  0x0e, 0x19, 0x10, 0x00, 0x00, 0x00,
>-  0x33, 0x14, 0x50, 0x00, 0x00, 0x00,
>-  0x10, 0xc3, 0x28, 0x00, 0x00, 0x00,
>-  0x45, 0x68, 0x48, 0x00, 0x00, 0x00,
>-  0x88, 0x84, 0xa8, 0x00, 0x00, 0x00,
>-  0xe0, 0x22, 0x90, 0x00, 0x00, 0x00,
>-  0x4d, 0xd0, 0xc0, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom21_15[90] = {
>-  0x46, 0x11, 0x90, 0x00, 0x00, 0x00,
>-  0x33, 0x0c, 0xc8, 0x00, 0x00, 0x00,
>-  0x10, 0xe4, 0x60, 0x00, 0x00, 0x00,
>-  0x0c, 0x69, 0x08, 0x00, 0x00, 0x00,
>-  0x28, 0x94, 0x28, 0x00, 0x00, 0x00,
>-  0x94, 0x21, 0x30, 0x00, 0x00, 0x00,
>-  0xc1, 0x02, 0x58, 0x00, 0x00, 0x00,
>-  0x2c, 0x03, 0x20, 0x00, 0x00, 0x00,
>-  0x81, 0xa0, 0x18, 0x00, 0x00, 0x00,
>-  0xa0, 0x68, 0x20, 0x00, 0x00, 0x00,
>-  0x05, 0x41, 0x50, 0x00, 0x00, 0x00,
>-  0x18, 0x90, 0xc0, 0x00, 0x00, 0x00,
>-  0xc2, 0x06, 0x80, 0x00, 0x00, 0x00,
>-  0x22, 0x98, 0x08, 0x00, 0x00, 0x00,
>-  0x50, 0x45, 0x08, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom21_16[96] = {
>-  0x2c, 0x03, 0x20, 0x00, 0x00, 0x00,
>-  0x81, 0xa0, 0x18, 0x00, 0x00, 0x00,
>-  0xa0, 0x68, 0x20, 0x00, 0x00, 0x00,
>-  0x05, 0x41, 0x50, 0x00, 0x00, 0x00,
>-  0x18, 0x90, 0xc0, 0x00, 0x00, 0x00,
>-  0xc2, 0x06, 0x80, 0x00, 0x00, 0x00,
>-  0x22, 0x98, 0x08, 0x00, 0x00, 0x00,
>-  0x50, 0x45, 0x08, 0x00, 0x00, 0x00,
>-  0x46, 0x11, 0x90, 0x00, 0x00, 0x00,
>-  0x33, 0x0c, 0xc8, 0x00, 0x00, 0x00,
>-  0x10, 0xe4, 0x60, 0x00, 0x00, 0x00,
>-  0x0c, 0x69, 0x08, 0x00, 0x00, 0x00,
>-  0x28, 0x94, 0x28, 0x00, 0x00, 0x00,
>-  0x94, 0x21, 0x30, 0x00, 0x00, 0x00,
>-  0xc1, 0x02, 0x58, 0x00, 0x00, 0x00,
>-  0x3b, 0xf5, 0x38, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom21_17[102] = {
>-  0x2c, 0x03, 0x20, 0x00, 0x00, 0x00,
>-  0x81, 0xa0, 0x18, 0x00, 0x00, 0x00,
>-  0xa0, 0x68, 0x20, 0x00, 0x00, 0x00,
>-  0x05, 0x41, 0x50, 0x00, 0x00, 0x00,
>-  0x18, 0x90, 0xc0, 0x00, 0x00, 0x00,
>-  0xc2, 0x06, 0x80, 0x00, 0x00, 0x00,
>-  0x22, 0x98, 0x08, 0x00, 0x00, 0x00,
>-  0x50, 0x45, 0x08, 0x00, 0x00, 0x00,
>-  0x4e, 0x11, 0x90, 0x00, 0x00, 0x00,
>-  0xe3, 0x18, 0x98, 0x00, 0x00, 0x00,
>-  0x81, 0xe3, 0x00, 0x00, 0x00, 0x00,
>-  0x21, 0x40, 0x58, 0x00, 0x00, 0x00,
>-  0x52, 0x81, 0xe0, 0x00, 0x00, 0x00,
>-  0xb4, 0x28, 0x20, 0x00, 0x00, 0x00,
>-  0x26, 0x86, 0x28, 0x00, 0x00, 0x00,
>-  0x58, 0x64, 0x40, 0x00, 0x00, 0x00,
>-  0x19, 0x9e, 0x00, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom21_18[108] = {
>-  0x4e, 0x11, 0x90, 0x00, 0x00, 0x00,
>-  0xe3, 0x18, 0x98, 0x00, 0x00, 0x00,
>-  0x81, 0xe3, 0x00, 0x00, 0x00, 0x00,
>-  0x21, 0x40, 0x58, 0x00, 0x00, 0x00,
>-  0x52, 0x81, 0xe0, 0x00, 0x00, 0x00,
>-  0xb4, 0x28, 0x20, 0x00, 0x00, 0x00,
>-  0x26, 0x86, 0x28, 0x00, 0x00, 0x00,
>-  0x58, 0x64, 0x40, 0x00, 0x00, 0x00,
>-  0x19, 0x9e, 0x00, 0x00, 0x00, 0x00,
>-  0x2c, 0x03, 0x20, 0x00, 0x00, 0x00,
>-  0x81, 0xa0, 0x18, 0x00, 0x00, 0x00,
>-  0xa0, 0x68, 0x20, 0x00, 0x00, 0x00,
>-  0x05, 0x41, 0x50, 0x00, 0x00, 0x00,
>-  0x18, 0x90, 0xc0, 0x00, 0x00, 0x00,
>-  0xc2, 0x06, 0x80, 0x00, 0x00, 0x00,
>-  0x22, 0x98, 0x08, 0x00, 0x00, 0x00,
>-  0x50, 0x45, 0x08, 0x00, 0x00, 0x00,
>-  0x5a, 0x56, 0x58, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom21_19[114] = {
>-  0x4e, 0x11, 0x90, 0x00, 0x00, 0x00,
>-  0xe3, 0x18, 0x98, 0x00, 0x00, 0x00,
>-  0x81, 0xe3, 0x00, 0x00, 0x00, 0x00,
>-  0x21, 0x40, 0x58, 0x00, 0x00, 0x00,
>-  0x52, 0x81, 0xe0, 0x00, 0x00, 0x00,
>-  0xb4, 0x28, 0x20, 0x00, 0x00, 0x00,
>-  0x26, 0x86, 0x28, 0x00, 0x00, 0x00,
>-  0x58, 0x64, 0x40, 0x00, 0x00, 0x00,
>-  0x19, 0x9e, 0x00, 0x00, 0x00, 0x00,
>-  0x4c, 0x19, 0x10, 0x00, 0x00, 0x00,
>-  0x51, 0x14, 0x50, 0x00, 0x00, 0x00,
>-  0xa0, 0x6a, 0x40, 0x00, 0x00, 0x00,
>-  0x04, 0xc1, 0x30, 0x00, 0x00, 0x00,
>-  0x03, 0xb4, 0x00, 0x00, 0x00, 0x00,
>-  0x86, 0x20, 0x90, 0x00, 0x00, 0x00,
>-  0x29, 0x08, 0x48, 0x00, 0x00, 0x00,
>-  0x42, 0x43, 0x08, 0x00, 0x00, 0x00,
>-  0x98, 0x12, 0x80, 0x00, 0x00, 0x00,
>-  0x30, 0x84, 0xa8, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom21_2[12] = {
>-  0xee, 0x3b, 0x30, 0x00, 0x00, 0x00,
>-  0x99, 0xe6, 0xe8, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom21_20[120] = {
>-  0x4c, 0x19, 0x10, 0x00, 0x00, 0x00,
>-  0x51, 0x14, 0x50, 0x00, 0x00, 0x00,
>-  0xa0, 0x6a, 0x40, 0x00, 0x00, 0x00,
>-  0x04, 0xc1, 0x30, 0x00, 0x00, 0x00,
>-  0x03, 0xb4, 0x00, 0x00, 0x00, 0x00,
>-  0x86, 0x20, 0x90, 0x00, 0x00, 0x00,
>-  0x29, 0x08, 0x48, 0x00, 0x00, 0x00,
>-  0x42, 0x43, 0x08, 0x00, 0x00, 0x00,
>-  0x98, 0x12, 0x80, 0x00, 0x00, 0x00,
>-  0x30, 0x84, 0xa8, 0x00, 0x00, 0x00,
>-  0x4e, 0x11, 0x90, 0x00, 0x00, 0x00,
>-  0xe3, 0x18, 0x98, 0x00, 0x00, 0x00,
>-  0x81, 0xe3, 0x00, 0x00, 0x00, 0x00,
>-  0x21, 0x40, 0x58, 0x00, 0x00, 0x00,
>-  0x52, 0x81, 0xe0, 0x00, 0x00, 0x00,
>-  0xb4, 0x28, 0x20, 0x00, 0x00, 0x00,
>-  0x26, 0x86, 0x28, 0x00, 0x00, 0x00,
>-  0x58, 0x64, 0x40, 0x00, 0x00, 0x00,
>-  0x19, 0x9e, 0x00, 0x00, 0x00, 0x00,
>-  0x2a, 0x03, 0x30, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom21_21[126] = {
>-  0x4c, 0x19, 0x10, 0x00, 0x00, 0x00,
>-  0x51, 0x14, 0x50, 0x00, 0x00, 0x00,
>-  0xa0, 0x6a, 0x40, 0x00, 0x00, 0x00,
>-  0x04, 0xc1, 0x30, 0x00, 0x00, 0x00,
>-  0x03, 0xb4, 0x00, 0x00, 0x00, 0x00,
>-  0x86, 0x20, 0x90, 0x00, 0x00, 0x00,
>-  0x29, 0x08, 0x48, 0x00, 0x00, 0x00,
>-  0x42, 0x43, 0x08, 0x00, 0x00, 0x00,
>-  0x98, 0x12, 0x80, 0x00, 0x00, 0x00,
>-  0x30, 0x84, 0xa8, 0x00, 0x00, 0x00,
>-  0x4c, 0x11, 0x90, 0x00, 0x00, 0x00,
>-  0x51, 0x0c, 0xc8, 0x00, 0x00, 0x00,
>-  0xa0, 0x66, 0x40, 0x00, 0x00, 0x00,
>-  0x04, 0xc1, 0x60, 0x00, 0x00, 0x00,
>-  0x03, 0xa0, 0x28, 0x00, 0x00, 0x00,
>-  0x86, 0x21, 0x10, 0x00, 0x00, 0x00,
>-  0x29, 0x10, 0x18, 0x00, 0x00, 0x00,
>-  0x42, 0x42, 0xa0, 0x00, 0x00, 0x00,
>-  0x98, 0x1a, 0x00, 0x00, 0x00, 0x00,
>-  0x30, 0x84, 0x08, 0x00, 0x00, 0x00,
>-  0xdf, 0x4c, 0x10, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom21_3[18] = {
>-  0xce, 0x32, 0xb0, 0x00, 0x00, 0x00,
>-  0x55, 0xdc, 0x50, 0x00, 0x00, 0x00,
>-  0xa8, 0xed, 0x88, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom21_4[24] = {
>-  0xe6, 0x31, 0x30, 0x00, 0x00, 0x00,
>-  0x33, 0x8c, 0x58, 0x00, 0x00, 0x00,
>-  0x98, 0xd2, 0xc8, 0x00, 0x00, 0x00,
>-  0x2d, 0x4b, 0x28, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom21_5[30] = {
>-  0xce, 0x31, 0xb0, 0x00, 0x00, 0x00,
>-  0x63, 0x98, 0xd8, 0x00, 0x00, 0x00,
>-  0x98, 0xc7, 0x68, 0x00, 0x00, 0x00,
>-  0x4d, 0x6b, 0x50, 0x00, 0x00, 0x00,
>-  0xb2, 0x6c, 0xa8, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom21_6[36] = {
>-  0x4c, 0x19, 0x10, 0x00, 0x00, 0x00,
>-  0x51, 0x14, 0x50, 0x00, 0x00, 0x00,
>-  0x20, 0xea, 0x08, 0x00, 0x00, 0x00,
>-  0x85, 0x41, 0x28, 0x00, 0x00, 0x00,
>-  0x06, 0x80, 0xd8, 0x00, 0x00, 0x00,
>-  0x8a, 0x24, 0x30, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom21_7[42] = {
>-  0xc6, 0x11, 0x90, 0x00, 0x00, 0x00,
>-  0x33, 0x04, 0xc8, 0x00, 0x00, 0x00,
>-  0x18, 0x67, 0x40, 0x00, 0x00, 0x00,
>-  0x45, 0x42, 0xd0, 0x00, 0x00, 0x00,
>-  0x12, 0xd4, 0x28, 0x00, 0x00, 0x00,
>-  0xb4, 0x28, 0x30, 0x00, 0x00, 0x00,
>-  0x29, 0x92, 0x18, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom21_8[48] = {
>-  0x07, 0x0a, 0x70, 0x00, 0x00, 0x00,
>-  0x49, 0xa8, 0x28, 0x00, 0x00, 0x00,
>-  0xb0, 0x7a, 0x00, 0x00, 0x00, 0x00,
>-  0x24, 0xc5, 0xc0, 0x00, 0x00, 0x00,
>-  0x52, 0x80, 0xe8, 0x00, 0x00, 0x00,
>-  0xc6, 0x31, 0x80, 0x00, 0x00, 0x00,
>-  0x31, 0x94, 0x18, 0x00, 0x00, 0x00,
>-  0x18, 0xc7, 0x08, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom21_9[54] = {
>-  0x4e, 0x11, 0x10, 0x00, 0x00, 0x00,
>-  0x62, 0x1a, 0x08, 0x00, 0x00, 0x00,
>-  0x80, 0xe9, 0x40, 0x00, 0x00, 0x00,
>-  0xa1, 0x50, 0x50, 0x00, 0x00, 0x00,
>-  0x53, 0x00, 0x68, 0x00, 0x00, 0x00,
>-  0xa4, 0x24, 0x30, 0x00, 0x00, 0x00,
>-  0x16, 0xa0, 0x88, 0x00, 0x00, 0x00,
>-  0x58, 0x45, 0x20, 0x00, 0x00, 0x00,
>-  0x29, 0x86, 0x80, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom22_1[6] = {
>-  0xff, 0xff, 0xfc, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom22_10[60] = {
>-  0xc0, 0x38, 0x88, 0x00, 0x00, 0x00,
>-  0x30, 0x0e, 0x28, 0x00, 0x00, 0x00,
>-  0xe8, 0x07, 0x00, 0x00, 0x00, 0x00,
>-  0x85, 0x08, 0xa8, 0x00, 0x00, 0x00,
>-  0xd0, 0x92, 0x10, 0x00, 0x00, 0x00,
>-  0x86, 0x50, 0x48, 0x00, 0x00, 0x00,
>-  0x4a, 0x68, 0x0c, 0x00, 0x00, 0x00,
>-  0x01, 0xa0, 0x74, 0x00, 0x00, 0x00,
>-  0x4c, 0x81, 0x90, 0x00, 0x00, 0x00,
>-  0x62, 0x24, 0x04, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom22_11[66] = {
>-  0x46, 0x48, 0xc8, 0x00, 0x00, 0x00,
>-  0x33, 0x26, 0x64, 0x00, 0x00, 0x00,
>-  0x99, 0x13, 0x20, 0x00, 0x00, 0x00,
>-  0x05, 0x80, 0xb0, 0x00, 0x00, 0x00,
>-  0x80, 0xb0, 0x14, 0x00, 0x00, 0x00,
>-  0x84, 0x50, 0x88, 0x00, 0x00, 0x00,
>-  0x40, 0x68, 0x0c, 0x00, 0x00, 0x00,
>-  0x0a, 0x81, 0x50, 0x00, 0x00, 0x00,
>-  0x68, 0x0d, 0x00, 0x00, 0x00, 0x00,
>-  0x10, 0x22, 0x04, 0x00, 0x00, 0x00,
>-  0x30, 0x46, 0x08, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom22_12[72] = {
>-  0x64, 0x4c, 0x88, 0x00, 0x00, 0x00,
>-  0x51, 0x4a, 0x28, 0x00, 0x00, 0x00,
>-  0x0c, 0xa1, 0x94, 0x00, 0x00, 0x00,
>-  0xa1, 0x34, 0x24, 0x00, 0x00, 0x00,
>-  0x12, 0xa2, 0x54, 0x00, 0x00, 0x00,
>-  0x8a, 0x51, 0x48, 0x00, 0x00, 0x00,
>-  0x86, 0x90, 0xd0, 0x00, 0x00, 0x00,
>-  0x23, 0x24, 0x64, 0x00, 0x00, 0x00,
>-  0x16, 0x22, 0xc4, 0x00, 0x00, 0x00,
>-  0x4c, 0x29, 0x84, 0x00, 0x00, 0x00,
>-  0x41, 0xc8, 0x38, 0x00, 0x00, 0x00,
>-  0xf4, 0x18, 0x9c, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom22_13[78] = {
>-  0x64, 0x4c, 0x88, 0x00, 0x00, 0x00,
>-  0x51, 0x4a, 0x28, 0x00, 0x00, 0x00,
>-  0x0c, 0xa1, 0x94, 0x00, 0x00, 0x00,
>-  0xa1, 0x34, 0x24, 0x00, 0x00, 0x00,
>-  0x12, 0xa2, 0x54, 0x00, 0x00, 0x00,
>-  0x8a, 0x51, 0x48, 0x00, 0x00, 0x00,
>-  0x46, 0x48, 0xc8, 0x00, 0x00, 0x00,
>-  0x33, 0x26, 0x64, 0x00, 0x00, 0x00,
>-  0x91, 0x92, 0x30, 0x00, 0x00, 0x00,
>-  0xa4, 0x34, 0x84, 0x00, 0x00, 0x00,
>-  0x50, 0xaa, 0x14, 0x00, 0x00, 0x00,
>-  0x84, 0xd0, 0x98, 0x00, 0x00, 0x00,
>-  0x09, 0x61, 0x2c, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom22_14[84] = {
>-  0x46, 0x48, 0xc8, 0x00, 0x00, 0x00,
>-  0x33, 0x26, 0x64, 0x00, 0x00, 0x00,
>-  0x91, 0x92, 0x30, 0x00, 0x00, 0x00,
>-  0xa4, 0x34, 0x84, 0x00, 0x00, 0x00,
>-  0x50, 0xaa, 0x14, 0x00, 0x00, 0x00,
>-  0x84, 0xd0, 0x98, 0x00, 0x00, 0x00,
>-  0x09, 0x61, 0x2c, 0x00, 0x00, 0x00,
>-  0x64, 0x4c, 0x88, 0x00, 0x00, 0x00,
>-  0x51, 0x4a, 0x28, 0x00, 0x00, 0x00,
>-  0x0c, 0xa1, 0x94, 0x00, 0x00, 0x00,
>-  0xa1, 0x34, 0x24, 0x00, 0x00, 0x00,
>-  0x12, 0xa2, 0x54, 0x00, 0x00, 0x00,
>-  0x8a, 0x51, 0x48, 0x00, 0x00, 0x00,
>-  0xc6, 0xca, 0xe8, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom22_15[90] = {
>-  0x46, 0x48, 0xc8, 0x00, 0x00, 0x00,
>-  0x33, 0x26, 0x64, 0x00, 0x00, 0x00,
>-  0x91, 0x92, 0x30, 0x00, 0x00, 0x00,
>-  0xa4, 0x34, 0x84, 0x00, 0x00, 0x00,
>-  0x50, 0xaa, 0x14, 0x00, 0x00, 0x00,
>-  0x84, 0xd0, 0x98, 0x00, 0x00, 0x00,
>-  0x09, 0x61, 0x2c, 0x00, 0x00, 0x00,
>-  0x0c, 0x81, 0x90, 0x00, 0x00, 0x00,
>-  0x80, 0x70, 0x0c, 0x00, 0x00, 0x00,
>-  0xa0, 0x94, 0x10, 0x00, 0x00, 0x00,
>-  0x05, 0x40, 0xa8, 0x00, 0x00, 0x00,
>-  0x43, 0x08, 0x60, 0x00, 0x00, 0x00,
>-  0x1a, 0x03, 0x40, 0x00, 0x00, 0x00,
>-  0x60, 0x2c, 0x04, 0x00, 0x00, 0x00,
>-  0x14, 0x22, 0x84, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom22_16[96] = {
>-  0x0c, 0x81, 0x90, 0x00, 0x00, 0x00,
>-  0x80, 0x70, 0x0c, 0x00, 0x00, 0x00,
>-  0xa0, 0x94, 0x10, 0x00, 0x00, 0x00,
>-  0x05, 0x40, 0xa8, 0x00, 0x00, 0x00,
>-  0x43, 0x08, 0x60, 0x00, 0x00, 0x00,
>-  0x1a, 0x03, 0x40, 0x00, 0x00, 0x00,
>-  0x60, 0x2c, 0x04, 0x00, 0x00, 0x00,
>-  0x14, 0x22, 0x84, 0x00, 0x00, 0x00,
>-  0x46, 0x48, 0xc8, 0x00, 0x00, 0x00,
>-  0x33, 0x26, 0x64, 0x00, 0x00, 0x00,
>-  0x91, 0x92, 0x30, 0x00, 0x00, 0x00,
>-  0xa4, 0x34, 0x84, 0x00, 0x00, 0x00,
>-  0x50, 0xaa, 0x14, 0x00, 0x00, 0x00,
>-  0x84, 0xd0, 0x98, 0x00, 0x00, 0x00,
>-  0x09, 0x61, 0x2c, 0x00, 0x00, 0x00,
>-  0x86, 0xc1, 0x44, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom22_17[102] = {
>-  0x0c, 0x81, 0x90, 0x00, 0x00, 0x00,
>-  0x80, 0x70, 0x0c, 0x00, 0x00, 0x00,
>-  0xa0, 0x94, 0x10, 0x00, 0x00, 0x00,
>-  0x05, 0x40, 0xa8, 0x00, 0x00, 0x00,
>-  0x43, 0x08, 0x60, 0x00, 0x00, 0x00,
>-  0x1a, 0x03, 0x40, 0x00, 0x00, 0x00,
>-  0x60, 0x2c, 0x04, 0x00, 0x00, 0x00,
>-  0x14, 0x22, 0x84, 0x00, 0x00, 0x00,
>-  0x46, 0x48, 0xc8, 0x00, 0x00, 0x00,
>-  0x62, 0x6c, 0x4c, 0x00, 0x00, 0x00,
>-  0x8c, 0x11, 0x80, 0x00, 0x00, 0x00,
>-  0x01, 0x60, 0x2c, 0x00, 0x00, 0x00,
>-  0x07, 0x80, 0xf0, 0x00, 0x00, 0x00,
>-  0xa0, 0x94, 0x10, 0x00, 0x00, 0x00,
>-  0x18, 0xa3, 0x14, 0x00, 0x00, 0x00,
>-  0x91, 0x12, 0x20, 0x00, 0x00, 0x00,
>-  0x78, 0x0f, 0x00, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom22_18[108] = {
>-  0x46, 0x48, 0xc8, 0x00, 0x00, 0x00,
>-  0x62, 0x6c, 0x4c, 0x00, 0x00, 0x00,
>-  0x8c, 0x11, 0x80, 0x00, 0x00, 0x00,
>-  0x01, 0x60, 0x2c, 0x00, 0x00, 0x00,
>-  0x07, 0x80, 0xf0, 0x00, 0x00, 0x00,
>-  0xa0, 0x94, 0x10, 0x00, 0x00, 0x00,
>-  0x18, 0xa3, 0x14, 0x00, 0x00, 0x00,
>-  0x91, 0x12, 0x20, 0x00, 0x00, 0x00,
>-  0x78, 0x0f, 0x00, 0x00, 0x00, 0x00,
>-  0x0c, 0x81, 0x90, 0x00, 0x00, 0x00,
>-  0x80, 0x70, 0x0c, 0x00, 0x00, 0x00,
>-  0xa0, 0x94, 0x10, 0x00, 0x00, 0x00,
>-  0x05, 0x40, 0xa8, 0x00, 0x00, 0x00,
>-  0x43, 0x08, 0x60, 0x00, 0x00, 0x00,
>-  0x1a, 0x03, 0x40, 0x00, 0x00, 0x00,
>-  0x60, 0x2c, 0x04, 0x00, 0x00, 0x00,
>-  0x14, 0x22, 0x84, 0x00, 0x00, 0x00,
>-  0xe4, 0xd4, 0x6c, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom22_19[114] = {
>-  0x46, 0x48, 0xc8, 0x00, 0x00, 0x00,
>-  0x62, 0x6c, 0x4c, 0x00, 0x00, 0x00,
>-  0x8c, 0x11, 0x80, 0x00, 0x00, 0x00,
>-  0x01, 0x60, 0x2c, 0x00, 0x00, 0x00,
>-  0x07, 0x80, 0xf0, 0x00, 0x00, 0x00,
>-  0xa0, 0x94, 0x10, 0x00, 0x00, 0x00,
>-  0x18, 0xa3, 0x14, 0x00, 0x00, 0x00,
>-  0x91, 0x12, 0x20, 0x00, 0x00, 0x00,
>-  0x78, 0x0f, 0x00, 0x00, 0x00, 0x00,
>-  0x64, 0x4c, 0x88, 0x00, 0x00, 0x00,
>-  0x51, 0x4a, 0x28, 0x00, 0x00, 0x00,
>-  0xa9, 0x15, 0x20, 0x00, 0x00, 0x00,
>-  0x04, 0xc0, 0x98, 0x00, 0x00, 0x00,
>-  0xd0, 0x1a, 0x00, 0x00, 0x00, 0x00,
>-  0x82, 0x50, 0x48, 0x00, 0x00, 0x00,
>-  0x21, 0x24, 0x24, 0x00, 0x00, 0x00,
>-  0x0c, 0x21, 0x84, 0x00, 0x00, 0x00,
>-  0x4a, 0x09, 0x40, 0x00, 0x00, 0x00,
>-  0x12, 0xa2, 0x54, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom22_2[12] = {
>-  0xec, 0xdd, 0x98, 0x00, 0x00, 0x00,
>-  0x9b, 0xb3, 0x74, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom22_20[120] = {
>-  0x64, 0x4c, 0x88, 0x00, 0x00, 0x00,
>-  0x51, 0x4a, 0x28, 0x00, 0x00, 0x00,
>-  0xa9, 0x15, 0x20, 0x00, 0x00, 0x00,
>-  0x04, 0xc0, 0x98, 0x00, 0x00, 0x00,
>-  0xd0, 0x1a, 0x00, 0x00, 0x00, 0x00,
>-  0x82, 0x50, 0x48, 0x00, 0x00, 0x00,
>-  0x21, 0x24, 0x24, 0x00, 0x00, 0x00,
>-  0x0c, 0x21, 0x84, 0x00, 0x00, 0x00,
>-  0x4a, 0x09, 0x40, 0x00, 0x00, 0x00,
>-  0x12, 0xa2, 0x54, 0x00, 0x00, 0x00,
>-  0x46, 0x48, 0xc8, 0x00, 0x00, 0x00,
>-  0x62, 0x6c, 0x4c, 0x00, 0x00, 0x00,
>-  0x8c, 0x11, 0x80, 0x00, 0x00, 0x00,
>-  0x01, 0x60, 0x2c, 0x00, 0x00, 0x00,
>-  0x07, 0x80, 0xf0, 0x00, 0x00, 0x00,
>-  0xa0, 0x94, 0x10, 0x00, 0x00, 0x00,
>-  0x18, 0xa3, 0x14, 0x00, 0x00, 0x00,
>-  0x91, 0x12, 0x20, 0x00, 0x00, 0x00,
>-  0x78, 0x0f, 0x00, 0x00, 0x00, 0x00,
>-  0x3b, 0x48, 0xc4, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom22_21[126] = {
>-  0x64, 0x4c, 0x88, 0x00, 0x00, 0x00,
>-  0x51, 0x4a, 0x28, 0x00, 0x00, 0x00,
>-  0xa9, 0x15, 0x20, 0x00, 0x00, 0x00,
>-  0x04, 0xc0, 0x98, 0x00, 0x00, 0x00,
>-  0xd0, 0x1a, 0x00, 0x00, 0x00, 0x00,
>-  0x82, 0x50, 0x48, 0x00, 0x00, 0x00,
>-  0x21, 0x24, 0x24, 0x00, 0x00, 0x00,
>-  0x0c, 0x21, 0x84, 0x00, 0x00, 0x00,
>-  0x4a, 0x09, 0x40, 0x00, 0x00, 0x00,
>-  0x12, 0xa2, 0x54, 0x00, 0x00, 0x00,
>-  0x46, 0x48, 0xc8, 0x00, 0x00, 0x00,
>-  0x33, 0x26, 0x64, 0x00, 0x00, 0x00,
>-  0x99, 0x13, 0x20, 0x00, 0x00, 0x00,
>-  0x05, 0x80, 0xb0, 0x00, 0x00, 0x00,
>-  0x80, 0xb0, 0x14, 0x00, 0x00, 0x00,
>-  0x84, 0x50, 0x88, 0x00, 0x00, 0x00,
>-  0x40, 0x68, 0x0c, 0x00, 0x00, 0x00,
>-  0x0a, 0x81, 0x50, 0x00, 0x00, 0x00,
>-  0x68, 0x0d, 0x00, 0x00, 0x00, 0x00,
>-  0x10, 0x22, 0x04, 0x00, 0x00, 0x00,
>-  0x30, 0x46, 0x08, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom22_22[132] = {
>-  0x46, 0x48, 0xc8, 0x00, 0x00, 0x00,
>-  0x33, 0x26, 0x64, 0x00, 0x00, 0x00,
>-  0x99, 0x13, 0x20, 0x00, 0x00, 0x00,
>-  0x05, 0x80, 0xb0, 0x00, 0x00, 0x00,
>-  0x80, 0xb0, 0x14, 0x00, 0x00, 0x00,
>-  0x84, 0x50, 0x88, 0x00, 0x00, 0x00,
>-  0x40, 0x68, 0x0c, 0x00, 0x00, 0x00,
>-  0x0a, 0x81, 0x50, 0x00, 0x00, 0x00,
>-  0x68, 0x0d, 0x00, 0x00, 0x00, 0x00,
>-  0x10, 0x22, 0x04, 0x00, 0x00, 0x00,
>-  0x30, 0x46, 0x08, 0x00, 0x00, 0x00,
>-  0x64, 0x4c, 0x88, 0x00, 0x00, 0x00,
>-  0x51, 0x4a, 0x28, 0x00, 0x00, 0x00,
>-  0xa9, 0x15, 0x20, 0x00, 0x00, 0x00,
>-  0x04, 0xc0, 0x98, 0x00, 0x00, 0x00,
>-  0xd0, 0x1a, 0x00, 0x00, 0x00, 0x00,
>-  0x82, 0x50, 0x48, 0x00, 0x00, 0x00,
>-  0x21, 0x24, 0x24, 0x00, 0x00, 0x00,
>-  0x0c, 0x21, 0x84, 0x00, 0x00, 0x00,
>-  0x4a, 0x09, 0x40, 0x00, 0x00, 0x00,
>-  0x12, 0xa2, 0x54, 0x00, 0x00, 0x00,
>-  0x9e, 0xce, 0x88, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom22_3[18] = {
>-  0xca, 0xd9, 0x58, 0x00, 0x00, 0x00,
>-  0xf1, 0x5e, 0x28, 0x00, 0x00, 0x00,
>-  0xb6, 0x35, 0xc4, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom22_4[24] = {
>-  0xc4, 0xd8, 0x98, 0x00, 0x00, 0x00,
>-  0x31, 0x66, 0x2c, 0x00, 0x00, 0x00,
>-  0x4b, 0x29, 0x64, 0x00, 0x00, 0x00,
>-  0x2c, 0xa5, 0x94, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom22_5[30] = {
>-  0xc6, 0xd8, 0xd8, 0x00, 0x00, 0x00,
>-  0x63, 0x6c, 0x6c, 0x00, 0x00, 0x00,
>-  0x1d, 0xa3, 0xb4, 0x00, 0x00, 0x00,
>-  0xad, 0x55, 0xa8, 0x00, 0x00, 0x00,
>-  0xb2, 0xb6, 0x54, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom22_6[36] = {
>-  0x64, 0x4c, 0x88, 0x00, 0x00, 0x00,
>-  0x51, 0x4a, 0x28, 0x00, 0x00, 0x00,
>-  0xa8, 0x35, 0x04, 0x00, 0x00, 0x00,
>-  0xc4, 0xa0, 0x94, 0x00, 0x00, 0x00,
>-  0x03, 0x60, 0x6c, 0x00, 0x00, 0x00,
>-  0x90, 0xd2, 0x18, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom22_7[42] = {
>-  0xc6, 0x48, 0xc8, 0x00, 0x00, 0x00,
>-  0x13, 0x26, 0x64, 0x00, 0x00, 0x00,
>-  0x8d, 0x13, 0xa0, 0x00, 0x00, 0x00,
>-  0x8b, 0x41, 0x68, 0x00, 0x00, 0x00,
>-  0x52, 0xaa, 0x14, 0x00, 0x00, 0x00,
>-  0xa2, 0xd4, 0x18, 0x00, 0x00, 0x00,
>-  0x61, 0xa8, 0x2c, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom22_8[48] = {
>-  0x28, 0x85, 0x38, 0x00, 0x00, 0x00,
>-  0x21, 0xf4, 0x04, 0x00, 0x00, 0x00,
>-  0xe9, 0x1d, 0x00, 0x00, 0x00, 0x00,
>-  0x17, 0x02, 0xe0, 0x00, 0x00, 0x00,
>-  0x83, 0xa0, 0x54, 0x00, 0x00, 0x00,
>-  0x46, 0x18, 0xe8, 0x00, 0x00, 0x00,
>-  0x50, 0x6a, 0x0c, 0x00, 0x00, 0x00,
>-  0x1c, 0x23, 0x84, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom22_9[54] = {
>-  0x44, 0x48, 0xc8, 0x00, 0x00, 0x00,
>-  0x28, 0x2d, 0x0c, 0x00, 0x00, 0x00,
>-  0x25, 0x14, 0xa0, 0x00, 0x00, 0x00,
>-  0x59, 0x0a, 0x20, 0x00, 0x00, 0x00,
>-  0x03, 0xa0, 0x34, 0x00, 0x00, 0x00,
>-  0xc0, 0xd0, 0x18, 0x00, 0x00, 0x00,
>-  0xa2, 0x30, 0x44, 0x00, 0x00, 0x00,
>-  0x14, 0x82, 0xd0, 0x00, 0x00, 0x00,
>-  0x9a, 0x03, 0x80, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom23_1[6] = {
>-  0xff, 0xff, 0xfe, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom23_10[60] = {
>-  0x64, 0x4a, 0x28, 0x00, 0x00, 0x00,
>-  0x51, 0x48, 0xa2, 0x00, 0x00, 0x00,
>-  0xa9, 0x10, 0x1a, 0x00, 0x00, 0x00,
>-  0x04, 0xc4, 0x84, 0x00, 0x00, 0x00,
>-  0xd0, 0x01, 0x44, 0x00, 0x00, 0x00,
>-  0x82, 0x40, 0x1c, 0x00, 0x00, 0x00,
>-  0x21, 0x37, 0x00, 0x00, 0x00, 0x00,
>-  0x0c, 0x21, 0x22, 0x00, 0x00, 0x00,
>-  0x4a, 0x0a, 0xc0, 0x00, 0x00, 0x00,
>-  0x12, 0xb4, 0x50, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom23_11[66] = {
>-  0x46, 0x4a, 0x6c, 0x00, 0x00, 0x00,
>-  0x33, 0x24, 0x26, 0x00, 0x00, 0x00,
>-  0x99, 0x02, 0x12, 0x00, 0x00, 0x00,
>-  0x05, 0x80, 0x0e, 0x00, 0x00, 0x00,
>-  0x80, 0xa1, 0x82, 0x00, 0x00, 0x00,
>-  0x84, 0x48, 0x18, 0x00, 0x00, 0x00,
>-  0x40, 0x6d, 0x40, 0x00, 0x00, 0x00,
>-  0x0a, 0x90, 0xc0, 0x00, 0x00, 0x00,
>-  0x68, 0x04, 0x90, 0x00, 0x00, 0x00,
>-  0x10, 0x31, 0x20, 0x00, 0x00, 0x00,
>-  0x30, 0x58, 0x04, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom23_12[72] = {
>-  0x64, 0x4a, 0x28, 0x00, 0x00, 0x00,
>-  0x51, 0x58, 0xa2, 0x00, 0x00, 0x00,
>-  0x0c, 0xa4, 0x30, 0x00, 0x00, 0x00,
>-  0xa1, 0x22, 0x46, 0x00, 0x00, 0x00,
>-  0x12, 0xa1, 0x1c, 0x00, 0x00, 0x00,
>-  0x8a, 0x45, 0xc0, 0x00, 0x00, 0x00,
>-  0x86, 0x8a, 0x6c, 0x00, 0x00, 0x00,
>-  0x23, 0x2c, 0x84, 0x00, 0x00, 0x00,
>-  0x16, 0x21, 0x98, 0x00, 0x00, 0x00,
>-  0x4c, 0x30, 0x54, 0x00, 0x00, 0x00,
>-  0x41, 0xc1, 0x26, 0x00, 0x00, 0x00,
>-  0x19, 0x56, 0xe4, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom23_13[78] = {
>-  0x64, 0x4a, 0x28, 0x00, 0x00, 0x00,
>-  0x51, 0x58, 0xa2, 0x00, 0x00, 0x00,
>-  0x0c, 0xa4, 0x30, 0x00, 0x00, 0x00,
>-  0xa1, 0x22, 0x46, 0x00, 0x00, 0x00,
>-  0x12, 0xa1, 0x1c, 0x00, 0x00, 0x00,
>-  0x8a, 0x45, 0xc0, 0x00, 0x00, 0x00,
>-  0x46, 0x4a, 0x6c, 0x00, 0x00, 0x00,
>-  0x33, 0x24, 0x26, 0x00, 0x00, 0x00,
>-  0x91, 0x92, 0x12, 0x00, 0x00, 0x00,
>-  0xa4, 0x20, 0x4a, 0x00, 0x00, 0x00,
>-  0x50, 0xa0, 0xd4, 0x00, 0x00, 0x00,
>-  0x84, 0xc5, 0x80, 0x00, 0x00, 0x00,
>-  0x09, 0x71, 0x0c, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom23_14[84] = {
>-  0x46, 0x4a, 0x6c, 0x00, 0x00, 0x00,
>-  0x33, 0x24, 0x26, 0x00, 0x00, 0x00,
>-  0x91, 0x92, 0x12, 0x00, 0x00, 0x00,
>-  0xa4, 0x20, 0x4a, 0x00, 0x00, 0x00,
>-  0x50, 0xa0, 0xd4, 0x00, 0x00, 0x00,
>-  0x84, 0xc5, 0x80, 0x00, 0x00, 0x00,
>-  0x09, 0x71, 0x0c, 0x00, 0x00, 0x00,
>-  0x64, 0x4a, 0x28, 0x00, 0x00, 0x00,
>-  0x51, 0x58, 0xa2, 0x00, 0x00, 0x00,
>-  0x0c, 0xa4, 0x30, 0x00, 0x00, 0x00,
>-  0xa1, 0x22, 0x46, 0x00, 0x00, 0x00,
>-  0x12, 0xa1, 0x1c, 0x00, 0x00, 0x00,
>-  0x8a, 0x45, 0xc0, 0x00, 0x00, 0x00,
>-  0x9c, 0x3f, 0xb2, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom23_15[90] = {
>-  0x46, 0x4a, 0x6c, 0x00, 0x00, 0x00,
>-  0x33, 0x24, 0x26, 0x00, 0x00, 0x00,
>-  0x91, 0x92, 0x12, 0x00, 0x00, 0x00,
>-  0xa4, 0x20, 0x4a, 0x00, 0x00, 0x00,
>-  0x50, 0xa0, 0xd4, 0x00, 0x00, 0x00,
>-  0x84, 0xc5, 0x80, 0x00, 0x00, 0x00,
>-  0x09, 0x71, 0x0c, 0x00, 0x00, 0x00,
>-  0x0c, 0x84, 0x0c, 0x00, 0x00, 0x00,
>-  0x80, 0x70, 0x06, 0x00, 0x00, 0x00,
>-  0xa0, 0x88, 0x48, 0x00, 0x00, 0x00,
>-  0x05, 0x40, 0x32, 0x00, 0x00, 0x00,
>-  0x43, 0x02, 0x82, 0x00, 0x00, 0x00,
>-  0x1a, 0x01, 0x50, 0x00, 0x00, 0x00,
>-  0x60, 0x27, 0x00, 0x00, 0x00, 0x00,
>-  0x14, 0x38, 0xa0, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom23_16[96] = {
>-  0x0c, 0x84, 0x0c, 0x00, 0x00, 0x00,
>-  0x80, 0x70, 0x06, 0x00, 0x00, 0x00,
>-  0xa0, 0x88, 0x48, 0x00, 0x00, 0x00,
>-  0x05, 0x40, 0x32, 0x00, 0x00, 0x00,
>-  0x43, 0x02, 0x82, 0x00, 0x00, 0x00,
>-  0x1a, 0x01, 0x50, 0x00, 0x00, 0x00,
>-  0x60, 0x27, 0x00, 0x00, 0x00, 0x00,
>-  0x14, 0x38, 0xa0, 0x00, 0x00, 0x00,
>-  0x46, 0x4a, 0x6c, 0x00, 0x00, 0x00,
>-  0x33, 0x24, 0x26, 0x00, 0x00, 0x00,
>-  0x91, 0x92, 0x12, 0x00, 0x00, 0x00,
>-  0xa4, 0x20, 0x4a, 0x00, 0x00, 0x00,
>-  0x50, 0xa0, 0xd4, 0x00, 0x00, 0x00,
>-  0x84, 0xc5, 0x80, 0x00, 0x00, 0x00,
>-  0x09, 0x71, 0x0c, 0x00, 0x00, 0x00,
>-  0xfa, 0xd9, 0xf4, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom23_17[102] = {
>-  0x0c, 0x84, 0x0c, 0x00, 0x00, 0x00,
>-  0x80, 0x70, 0x06, 0x00, 0x00, 0x00,
>-  0xa0, 0x88, 0x48, 0x00, 0x00, 0x00,
>-  0x05, 0x40, 0x32, 0x00, 0x00, 0x00,
>-  0x43, 0x02, 0x82, 0x00, 0x00, 0x00,
>-  0x1a, 0x01, 0x50, 0x00, 0x00, 0x00,
>-  0x60, 0x27, 0x00, 0x00, 0x00, 0x00,
>-  0x14, 0x38, 0xa0, 0x00, 0x00, 0x00,
>-  0x46, 0x4a, 0x6c, 0x00, 0x00, 0x00,
>-  0x62, 0x7c, 0x84, 0x00, 0x00, 0x00,
>-  0x8c, 0x04, 0x88, 0x00, 0x00, 0x00,
>-  0x01, 0x74, 0x22, 0x00, 0x00, 0x00,
>-  0x07, 0x83, 0x06, 0x00, 0x00, 0x00,
>-  0xa0, 0x80, 0x72, 0x00, 0x00, 0x00,
>-  0x18, 0xb1, 0x42, 0x00, 0x00, 0x00,
>-  0x91, 0x00, 0x92, 0x00, 0x00, 0x00,
>-  0x78, 0x00, 0x1c, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom23_18[108] = {
>-  0x46, 0x4a, 0x6c, 0x00, 0x00, 0x00,
>-  0x62, 0x7c, 0x84, 0x00, 0x00, 0x00,
>-  0x8c, 0x04, 0x88, 0x00, 0x00, 0x00,
>-  0x01, 0x74, 0x22, 0x00, 0x00, 0x00,
>-  0x07, 0x83, 0x06, 0x00, 0x00, 0x00,
>-  0xa0, 0x80, 0x72, 0x00, 0x00, 0x00,
>-  0x18, 0xb1, 0x42, 0x00, 0x00, 0x00,
>-  0x91, 0x00, 0x92, 0x00, 0x00, 0x00,
>-  0x78, 0x00, 0x1c, 0x00, 0x00, 0x00,
>-  0x0c, 0x84, 0x0c, 0x00, 0x00, 0x00,
>-  0x80, 0x70, 0x06, 0x00, 0x00, 0x00,
>-  0xa0, 0x88, 0x48, 0x00, 0x00, 0x00,
>-  0x05, 0x40, 0x32, 0x00, 0x00, 0x00,
>-  0x43, 0x02, 0x82, 0x00, 0x00, 0x00,
>-  0x1a, 0x01, 0x50, 0x00, 0x00, 0x00,
>-  0x60, 0x27, 0x00, 0x00, 0x00, 0x00,
>-  0x14, 0x38, 0xa0, 0x00, 0x00, 0x00,
>-  0x82, 0x32, 0x56, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom23_19[114] = {
>-  0x46, 0x4a, 0x6c, 0x00, 0x00, 0x00,
>-  0x62, 0x7c, 0x84, 0x00, 0x00, 0x00,
>-  0x8c, 0x04, 0x88, 0x00, 0x00, 0x00,
>-  0x01, 0x74, 0x22, 0x00, 0x00, 0x00,
>-  0x07, 0x83, 0x06, 0x00, 0x00, 0x00,
>-  0xa0, 0x80, 0x72, 0x00, 0x00, 0x00,
>-  0x18, 0xb1, 0x42, 0x00, 0x00, 0x00,
>-  0x91, 0x00, 0x92, 0x00, 0x00, 0x00,
>-  0x78, 0x00, 0x1c, 0x00, 0x00, 0x00,
>-  0x64, 0x4a, 0x28, 0x00, 0x00, 0x00,
>-  0x51, 0x48, 0xa2, 0x00, 0x00, 0x00,
>-  0xa9, 0x10, 0x1a, 0x00, 0x00, 0x00,
>-  0x04, 0xc4, 0x84, 0x00, 0x00, 0x00,
>-  0xd0, 0x01, 0x44, 0x00, 0x00, 0x00,
>-  0x82, 0x40, 0x1c, 0x00, 0x00, 0x00,
>-  0x21, 0x37, 0x00, 0x00, 0x00, 0x00,
>-  0x0c, 0x21, 0x22, 0x00, 0x00, 0x00,
>-  0x4a, 0x0a, 0xc0, 0x00, 0x00, 0x00,
>-  0x12, 0xb4, 0x50, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom23_2[12] = {
>-  0xec, 0xdd, 0x98, 0x00, 0x00, 0x00,
>-  0x9b, 0xb2, 0x76, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom23_20[120] = {
>-  0x64, 0x4a, 0x28, 0x00, 0x00, 0x00,
>-  0x51, 0x48, 0xa2, 0x00, 0x00, 0x00,
>-  0xa9, 0x10, 0x1a, 0x00, 0x00, 0x00,
>-  0x04, 0xc4, 0x84, 0x00, 0x00, 0x00,
>-  0xd0, 0x01, 0x44, 0x00, 0x00, 0x00,
>-  0x82, 0x40, 0x1c, 0x00, 0x00, 0x00,
>-  0x21, 0x37, 0x00, 0x00, 0x00, 0x00,
>-  0x0c, 0x21, 0x22, 0x00, 0x00, 0x00,
>-  0x4a, 0x0a, 0xc0, 0x00, 0x00, 0x00,
>-  0x12, 0xb4, 0x50, 0x00, 0x00, 0x00,
>-  0x46, 0x4a, 0x6c, 0x00, 0x00, 0x00,
>-  0x62, 0x7c, 0x84, 0x00, 0x00, 0x00,
>-  0x8c, 0x04, 0x88, 0x00, 0x00, 0x00,
>-  0x01, 0x74, 0x22, 0x00, 0x00, 0x00,
>-  0x07, 0x83, 0x06, 0x00, 0x00, 0x00,
>-  0xa0, 0x80, 0x72, 0x00, 0x00, 0x00,
>-  0x18, 0xb1, 0x42, 0x00, 0x00, 0x00,
>-  0x91, 0x00, 0x92, 0x00, 0x00, 0x00,
>-  0x78, 0x00, 0x1c, 0x00, 0x00, 0x00,
>-  0xdb, 0x4a, 0x7a, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom23_21[126] = {
>-  0x64, 0x4a, 0x28, 0x00, 0x00, 0x00,
>-  0x51, 0x48, 0xa2, 0x00, 0x00, 0x00,
>-  0xa9, 0x10, 0x1a, 0x00, 0x00, 0x00,
>-  0x04, 0xc4, 0x84, 0x00, 0x00, 0x00,
>-  0xd0, 0x01, 0x44, 0x00, 0x00, 0x00,
>-  0x82, 0x40, 0x1c, 0x00, 0x00, 0x00,
>-  0x21, 0x37, 0x00, 0x00, 0x00, 0x00,
>-  0x0c, 0x21, 0x22, 0x00, 0x00, 0x00,
>-  0x4a, 0x0a, 0xc0, 0x00, 0x00, 0x00,
>-  0x12, 0xb4, 0x50, 0x00, 0x00, 0x00,
>-  0x46, 0x4a, 0x6c, 0x00, 0x00, 0x00,
>-  0x33, 0x24, 0x26, 0x00, 0x00, 0x00,
>-  0x99, 0x02, 0x12, 0x00, 0x00, 0x00,
>-  0x05, 0x80, 0x0e, 0x00, 0x00, 0x00,
>-  0x80, 0xa1, 0x82, 0x00, 0x00, 0x00,
>-  0x84, 0x48, 0x18, 0x00, 0x00, 0x00,
>-  0x40, 0x6d, 0x40, 0x00, 0x00, 0x00,
>-  0x0a, 0x90, 0xc0, 0x00, 0x00, 0x00,
>-  0x68, 0x04, 0x90, 0x00, 0x00, 0x00,
>-  0x10, 0x31, 0x20, 0x00, 0x00, 0x00,
>-  0x30, 0x58, 0x04, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom23_22[132] = {
>-  0x46, 0x4a, 0x6c, 0x00, 0x00, 0x00,
>-  0x33, 0x24, 0x26, 0x00, 0x00, 0x00,
>-  0x99, 0x02, 0x12, 0x00, 0x00, 0x00,
>-  0x05, 0x80, 0x0e, 0x00, 0x00, 0x00,
>-  0x80, 0xa1, 0x82, 0x00, 0x00, 0x00,
>-  0x84, 0x48, 0x18, 0x00, 0x00, 0x00,
>-  0x40, 0x6d, 0x40, 0x00, 0x00, 0x00,
>-  0x0a, 0x90, 0xc0, 0x00, 0x00, 0x00,
>-  0x68, 0x04, 0x90, 0x00, 0x00, 0x00,
>-  0x10, 0x31, 0x20, 0x00, 0x00, 0x00,
>-  0x30, 0x58, 0x04, 0x00, 0x00, 0x00,
>-  0x64, 0x4a, 0x28, 0x00, 0x00, 0x00,
>-  0x51, 0x48, 0xa2, 0x00, 0x00, 0x00,
>-  0xa9, 0x10, 0x1a, 0x00, 0x00, 0x00,
>-  0x04, 0xc4, 0x84, 0x00, 0x00, 0x00,
>-  0xd0, 0x01, 0x44, 0x00, 0x00, 0x00,
>-  0x82, 0x40, 0x1c, 0x00, 0x00, 0x00,
>-  0x21, 0x37, 0x00, 0x00, 0x00, 0x00,
>-  0x0c, 0x21, 0x22, 0x00, 0x00, 0x00,
>-  0x4a, 0x0a, 0xc0, 0x00, 0x00, 0x00,
>-  0x12, 0xb4, 0x50, 0x00, 0x00, 0x00,
>-  0xea, 0x8d, 0x1a, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom23_23[138] = {
>-  0x46, 0x4a, 0x6c, 0x00, 0x00, 0x00,
>-  0x33, 0x24, 0x26, 0x00, 0x00, 0x00,
>-  0x99, 0x02, 0x12, 0x00, 0x00, 0x00,
>-  0x05, 0x80, 0x0e, 0x00, 0x00, 0x00,
>-  0x80, 0xa1, 0x82, 0x00, 0x00, 0x00,
>-  0x84, 0x48, 0x18, 0x00, 0x00, 0x00,
>-  0x40, 0x6d, 0x40, 0x00, 0x00, 0x00,
>-  0x0a, 0x90, 0xc0, 0x00, 0x00, 0x00,
>-  0x68, 0x04, 0x90, 0x00, 0x00, 0x00,
>-  0x10, 0x31, 0x20, 0x00, 0x00, 0x00,
>-  0x30, 0x58, 0x04, 0x00, 0x00, 0x00,
>-  0x46, 0x42, 0x0c, 0x00, 0x00, 0x00,
>-  0x33, 0x20, 0x46, 0x00, 0x00, 0x00,
>-  0x99, 0x08, 0x0a, 0x00, 0x00, 0x00,
>-  0x05, 0x84, 0x30, 0x00, 0x00, 0x00,
>-  0x80, 0xb0, 0x22, 0x00, 0x00, 0x00,
>-  0x84, 0x42, 0x90, 0x00, 0x00, 0x00,
>-  0x40, 0x73, 0x00, 0x00, 0x00, 0x00,
>-  0x0a, 0x81, 0x12, 0x00, 0x00, 0x00,
>-  0x68, 0x0c, 0x40, 0x00, 0x00, 0x00,
>-  0x10, 0x24, 0x84, 0x00, 0x00, 0x00,
>-  0x30, 0x51, 0x40, 0x00, 0x00, 0x00,
>-  0x5f, 0x50, 0x88, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom23_3[18] = {
>-  0xca, 0xd3, 0x64, 0x00, 0x00, 0x00,
>-  0xf1, 0x49, 0x3a, 0x00, 0x00, 0x00,
>-  0x76, 0x27, 0xd0, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom23_4[24] = {
>-  0xc4, 0xd1, 0x64, 0x00, 0x00, 0x00,
>-  0x31, 0x62, 0x96, 0x00, 0x00, 0x00,
>-  0x4b, 0x24, 0x5a, 0x00, 0x00, 0x00,
>-  0x2c, 0xa8, 0xaa, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom23_5[30] = {
>-  0xc6, 0xca, 0x6c, 0x00, 0x00, 0x00,
>-  0x63, 0x6c, 0x96, 0x00, 0x00, 0x00,
>-  0x1d, 0xa1, 0xdc, 0x00, 0x00, 0x00,
>-  0xad, 0x55, 0x38, 0x00, 0x00, 0x00,
>-  0xb2, 0xb7, 0x06, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom23_6[36] = {
>-  0x64, 0x4a, 0x28, 0x00, 0x00, 0x00,
>-  0x51, 0x58, 0xa2, 0x00, 0x00, 0x00,
>-  0x0c, 0xa4, 0x30, 0x00, 0x00, 0x00,
>-  0xa1, 0x22, 0x46, 0x00, 0x00, 0x00,
>-  0x12, 0xa1, 0x1c, 0x00, 0x00, 0x00,
>-  0x8a, 0x45, 0xc0, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom23_7[42] = {
>-  0x46, 0x4a, 0x6c, 0x00, 0x00, 0x00,
>-  0x33, 0x24, 0x26, 0x00, 0x00, 0x00,
>-  0x91, 0x92, 0x12, 0x00, 0x00, 0x00,
>-  0xa4, 0x20, 0x4a, 0x00, 0x00, 0x00,
>-  0x50, 0xa0, 0xd4, 0x00, 0x00, 0x00,
>-  0x84, 0xc5, 0x80, 0x00, 0x00, 0x00,
>-  0x09, 0x71, 0x0c, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom23_8[48] = {
>-  0x0c, 0x84, 0x0c, 0x00, 0x00, 0x00,
>-  0x80, 0x70, 0x06, 0x00, 0x00, 0x00,
>-  0xa0, 0x88, 0x48, 0x00, 0x00, 0x00,
>-  0x05, 0x40, 0x32, 0x00, 0x00, 0x00,
>-  0x43, 0x02, 0x82, 0x00, 0x00, 0x00,
>-  0x1a, 0x01, 0x50, 0x00, 0x00, 0x00,
>-  0x60, 0x27, 0x00, 0x00, 0x00, 0x00,
>-  0x14, 0x38, 0xa0, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom23_9[54] = {
>-  0x46, 0x4a, 0x6c, 0x00, 0x00, 0x00,
>-  0x62, 0x7c, 0x84, 0x00, 0x00, 0x00,
>-  0x8c, 0x04, 0x88, 0x00, 0x00, 0x00,
>-  0x01, 0x74, 0x22, 0x00, 0x00, 0x00,
>-  0x07, 0x83, 0x06, 0x00, 0x00, 0x00,
>-  0xa0, 0x80, 0x72, 0x00, 0x00, 0x00,
>-  0x18, 0xb1, 0x42, 0x00, 0x00, 0x00,
>-  0x91, 0x00, 0x92, 0x00, 0x00, 0x00,
>-  0x78, 0x00, 0x1c, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom24_1[6] = {
>-  0xff, 0xff, 0xff, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom24_10[60] = {
>-  0x11, 0x45, 0x14, 0x00, 0x00, 0x00,
>-  0x45, 0x34, 0x53, 0x00, 0x00, 0x00,
>-  0x00, 0x48, 0x05, 0x00, 0x00, 0x00,
>-  0x10, 0x83, 0x09, 0x00, 0x00, 0x00,
>-  0x4a, 0x14, 0xa1, 0x00, 0x00, 0x00,
>-  0x40, 0xa4, 0x0a, 0x00, 0x00, 0x00,
>-  0xa0, 0x6a, 0x02, 0x00, 0x00, 0x00,
>-  0x88, 0x80, 0x8c, 0x00, 0x00, 0x00,
>-  0x86, 0x08, 0x60, 0x00, 0x00, 0x00,
>-  0x54, 0x0d, 0x40, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom24_11[66] = {
>-  0x53, 0x65, 0x34, 0x00, 0x00, 0x00,
>-  0xa0, 0x32, 0x11, 0x00, 0x00, 0x00,
>-  0x15, 0x11, 0x41, 0x00, 0x00, 0x00,
>-  0x03, 0x50, 0x15, 0x00, 0x00, 0x00,
>-  0x8c, 0x88, 0xc8, 0x00, 0x00, 0x00,
>-  0x28, 0x82, 0x88, 0x00, 0x00, 0x00,
>-  0x08, 0x48, 0x84, 0x00, 0x00, 0x00,
>-  0x99, 0x01, 0x90, 0x00, 0x00, 0x00,
>-  0x22, 0x92, 0x29, 0x00, 0x00, 0x00,
>-  0x46, 0x04, 0x60, 0x00, 0x00, 0x00,
>-  0x8c, 0x2c, 0x02, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom24_12[72] = {
>-  0x10, 0x61, 0x06, 0x00, 0x00, 0x00,
>-  0x02, 0x30, 0x23, 0x00, 0x00, 0x00,
>-  0x40, 0x54, 0x05, 0x00, 0x00, 0x00,
>-  0x21, 0x82, 0x18, 0x00, 0x00, 0x00,
>-  0x81, 0x18, 0x11, 0x00, 0x00, 0x00,
>-  0x14, 0x81, 0x48, 0x00, 0x00, 0x00,
>-  0x98, 0x09, 0x80, 0x00, 0x00, 0x00,
>-  0x08, 0x90, 0x89, 0x00, 0x00, 0x00,
>-  0x62, 0x06, 0x20, 0x00, 0x00, 0x00,
>-  0x24, 0x22, 0x42, 0x00, 0x00, 0x00,
>-  0x8a, 0x08, 0xa0, 0x00, 0x00, 0x00,
>-  0x84, 0x48, 0x44, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom24_13[78] = {
>-  0x51, 0x45, 0x14, 0x00, 0x00, 0x00,
>-  0xc5, 0x1c, 0x51, 0x00, 0x00, 0x00,
>-  0x21, 0x82, 0x18, 0x00, 0x00, 0x00,
>-  0x12, 0x31, 0x23, 0x00, 0x00, 0x00,
>-  0x08, 0xe0, 0x8e, 0x00, 0x00, 0x00,
>-  0x2e, 0x02, 0xe0, 0x00, 0x00, 0x00,
>-  0x53, 0x65, 0x36, 0x00, 0x00, 0x00,
>-  0x21, 0x32, 0x13, 0x00, 0x00, 0x00,
>-  0x90, 0x99, 0x09, 0x00, 0x00, 0x00,
>-  0x02, 0x50, 0x25, 0x00, 0x00, 0x00,
>-  0x06, 0xa0, 0x6a, 0x00, 0x00, 0x00,
>-  0x2c, 0x02, 0xc0, 0x00, 0x00, 0x00,
>-  0x88, 0x68, 0x86, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom24_14[84] = {
>-  0x53, 0x65, 0x36, 0x00, 0x00, 0x00,
>-  0x21, 0x32, 0x13, 0x00, 0x00, 0x00,
>-  0x90, 0x99, 0x09, 0x00, 0x00, 0x00,
>-  0x02, 0x50, 0x25, 0x00, 0x00, 0x00,
>-  0x06, 0xa0, 0x6a, 0x00, 0x00, 0x00,
>-  0x2c, 0x02, 0xc0, 0x00, 0x00, 0x00,
>-  0x88, 0x68, 0x86, 0x00, 0x00, 0x00,
>-  0x51, 0x45, 0x14, 0x00, 0x00, 0x00,
>-  0xc5, 0x1c, 0x51, 0x00, 0x00, 0x00,
>-  0x21, 0x82, 0x18, 0x00, 0x00, 0x00,
>-  0x12, 0x31, 0x23, 0x00, 0x00, 0x00,
>-  0x08, 0xe0, 0x8e, 0x00, 0x00, 0x00,
>-  0x2e, 0x02, 0xe0, 0x00, 0x00, 0x00,
>-  0xf2, 0xd6, 0x8e, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom24_15[90] = {
>-  0x53, 0x65, 0x36, 0x00, 0x00, 0x00,
>-  0x21, 0x32, 0x13, 0x00, 0x00, 0x00,
>-  0x90, 0x99, 0x09, 0x00, 0x00, 0x00,
>-  0x02, 0x50, 0x25, 0x00, 0x00, 0x00,
>-  0x06, 0xa0, 0x6a, 0x00, 0x00, 0x00,
>-  0x2c, 0x02, 0xc0, 0x00, 0x00, 0x00,
>-  0x88, 0x68, 0x86, 0x00, 0x00, 0x00,
>-  0x20, 0x62, 0x06, 0x00, 0x00, 0x00,
>-  0x80, 0x38, 0x03, 0x00, 0x00, 0x00,
>-  0x42, 0x44, 0x24, 0x00, 0x00, 0x00,
>-  0x01, 0x90, 0x19, 0x00, 0x00, 0x00,
>-  0x14, 0x11, 0x41, 0x00, 0x00, 0x00,
>-  0x0a, 0x80, 0xa8, 0x00, 0x00, 0x00,
>-  0x38, 0x03, 0x80, 0x00, 0x00, 0x00,
>-  0xc5, 0x0c, 0x50, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom24_16[96] = {
>-  0x20, 0x62, 0x06, 0x00, 0x00, 0x00,
>-  0x80, 0x38, 0x03, 0x00, 0x00, 0x00,
>-  0x42, 0x44, 0x24, 0x00, 0x00, 0x00,
>-  0x01, 0x90, 0x19, 0x00, 0x00, 0x00,
>-  0x14, 0x11, 0x41, 0x00, 0x00, 0x00,
>-  0x0a, 0x80, 0xa8, 0x00, 0x00, 0x00,
>-  0x38, 0x03, 0x80, 0x00, 0x00, 0x00,
>-  0xc5, 0x0c, 0x50, 0x00, 0x00, 0x00,
>-  0x53, 0x65, 0x36, 0x00, 0x00, 0x00,
>-  0x21, 0x32, 0x13, 0x00, 0x00, 0x00,
>-  0x90, 0x99, 0x09, 0x00, 0x00, 0x00,
>-  0x02, 0x50, 0x25, 0x00, 0x00, 0x00,
>-  0x06, 0xa0, 0x6a, 0x00, 0x00, 0x00,
>-  0x2c, 0x02, 0xc0, 0x00, 0x00, 0x00,
>-  0x88, 0x68, 0x86, 0x00, 0x00, 0x00,
>-  0xff, 0x6e, 0x0a, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom24_17[102] = {
>-  0x20, 0x62, 0x06, 0x00, 0x00, 0x00,
>-  0x80, 0x38, 0x03, 0x00, 0x00, 0x00,
>-  0x42, 0x44, 0x24, 0x00, 0x00, 0x00,
>-  0x01, 0x90, 0x19, 0x00, 0x00, 0x00,
>-  0x14, 0x11, 0x41, 0x00, 0x00, 0x00,
>-  0x0a, 0x80, 0xa8, 0x00, 0x00, 0x00,
>-  0x38, 0x03, 0x80, 0x00, 0x00, 0x00,
>-  0xc5, 0x0c, 0x50, 0x00, 0x00, 0x00,
>-  0x53, 0x65, 0x36, 0x00, 0x00, 0x00,
>-  0xe4, 0x2e, 0x42, 0x00, 0x00, 0x00,
>-  0x24, 0x42, 0x44, 0x00, 0x00, 0x00,
>-  0xa1, 0x1a, 0x11, 0x00, 0x00, 0x00,
>-  0x18, 0x31, 0x83, 0x00, 0x00, 0x00,
>-  0x03, 0x90, 0x39, 0x00, 0x00, 0x00,
>-  0x8a, 0x18, 0xa1, 0x00, 0x00, 0x00,
>-  0x04, 0x90, 0x49, 0x00, 0x00, 0x00,
>-  0x00, 0xe0, 0x0e, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom24_18[108] = {
>-  0x53, 0x65, 0x36, 0x00, 0x00, 0x00,
>-  0xe4, 0x2e, 0x42, 0x00, 0x00, 0x00,
>-  0x24, 0x42, 0x44, 0x00, 0x00, 0x00,
>-  0xa1, 0x1a, 0x11, 0x00, 0x00, 0x00,
>-  0x18, 0x31, 0x83, 0x00, 0x00, 0x00,
>-  0x03, 0x90, 0x39, 0x00, 0x00, 0x00,
>-  0x8a, 0x18, 0xa1, 0x00, 0x00, 0x00,
>-  0x04, 0x90, 0x49, 0x00, 0x00, 0x00,
>-  0x00, 0xe0, 0x0e, 0x00, 0x00, 0x00,
>-  0x20, 0x62, 0x06, 0x00, 0x00, 0x00,
>-  0x80, 0x38, 0x03, 0x00, 0x00, 0x00,
>-  0x42, 0x44, 0x24, 0x00, 0x00, 0x00,
>-  0x01, 0x90, 0x19, 0x00, 0x00, 0x00,
>-  0x14, 0x11, 0x41, 0x00, 0x00, 0x00,
>-  0x0a, 0x80, 0xa8, 0x00, 0x00, 0x00,
>-  0x38, 0x03, 0x80, 0x00, 0x00, 0x00,
>-  0xc5, 0x0c, 0x50, 0x00, 0x00, 0x00,
>-  0x34, 0x50, 0xae, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom24_19[114] = {
>-  0x53, 0x65, 0x36, 0x00, 0x00, 0x00,
>-  0xe4, 0x2e, 0x42, 0x00, 0x00, 0x00,
>-  0x24, 0x42, 0x44, 0x00, 0x00, 0x00,
>-  0xa1, 0x1a, 0x11, 0x00, 0x00, 0x00,
>-  0x18, 0x31, 0x83, 0x00, 0x00, 0x00,
>-  0x03, 0x90, 0x39, 0x00, 0x00, 0x00,
>-  0x8a, 0x18, 0xa1, 0x00, 0x00, 0x00,
>-  0x04, 0x90, 0x49, 0x00, 0x00, 0x00,
>-  0x00, 0xe0, 0x0e, 0x00, 0x00, 0x00,
>-  0x51, 0x45, 0x14, 0x00, 0x00, 0x00,
>-  0x45, 0x14, 0x51, 0x00, 0x00, 0x00,
>-  0x80, 0xd8, 0x0d, 0x00, 0x00, 0x00,
>-  0x24, 0x22, 0x42, 0x00, 0x00, 0x00,
>-  0x0a, 0x20, 0xa2, 0x00, 0x00, 0x00,
>-  0x00, 0xe0, 0x0e, 0x00, 0x00, 0x00,
>-  0xb8, 0x0b, 0x80, 0x00, 0x00, 0x00,
>-  0x09, 0x10, 0x91, 0x00, 0x00, 0x00,
>-  0x56, 0x05, 0x60, 0x00, 0x00, 0x00,
>-  0xa2, 0x8a, 0x28, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom24_2[12] = {
>-  0xec, 0xce, 0xcc, 0x00, 0x00, 0x00,
>-  0x93, 0xb9, 0x3b, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom24_20[120] = {
>-  0x51, 0x45, 0x14, 0x00, 0x00, 0x00,
>-  0x45, 0x14, 0x51, 0x00, 0x00, 0x00,
>-  0x80, 0xd8, 0x0d, 0x00, 0x00, 0x00,
>-  0x24, 0x22, 0x42, 0x00, 0x00, 0x00,
>-  0x0a, 0x20, 0xa2, 0x00, 0x00, 0x00,
>-  0x00, 0xe0, 0x0e, 0x00, 0x00, 0x00,
>-  0xb8, 0x0b, 0x80, 0x00, 0x00, 0x00,
>-  0x09, 0x10, 0x91, 0x00, 0x00, 0x00,
>-  0x56, 0x05, 0x60, 0x00, 0x00, 0x00,
>-  0xa2, 0x8a, 0x28, 0x00, 0x00, 0x00,
>-  0x53, 0x65, 0x36, 0x00, 0x00, 0x00,
>-  0xe4, 0x2e, 0x42, 0x00, 0x00, 0x00,
>-  0x24, 0x42, 0x44, 0x00, 0x00, 0x00,
>-  0xa1, 0x1a, 0x11, 0x00, 0x00, 0x00,
>-  0x18, 0x31, 0x83, 0x00, 0x00, 0x00,
>-  0x03, 0x90, 0x39, 0x00, 0x00, 0x00,
>-  0x8a, 0x18, 0xa1, 0x00, 0x00, 0x00,
>-  0x04, 0x90, 0x49, 0x00, 0x00, 0x00,
>-  0x00, 0xe0, 0x0e, 0x00, 0x00, 0x00,
>-  0x98, 0xa2, 0x95, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom24_21[126] = {
>-  0x51, 0x45, 0x14, 0x00, 0x00, 0x00,
>-  0x45, 0x14, 0x51, 0x00, 0x00, 0x00,
>-  0x80, 0xd8, 0x0d, 0x00, 0x00, 0x00,
>-  0x24, 0x22, 0x42, 0x00, 0x00, 0x00,
>-  0x0a, 0x20, 0xa2, 0x00, 0x00, 0x00,
>-  0x00, 0xe0, 0x0e, 0x00, 0x00, 0x00,
>-  0xb8, 0x0b, 0x80, 0x00, 0x00, 0x00,
>-  0x09, 0x10, 0x91, 0x00, 0x00, 0x00,
>-  0x56, 0x05, 0x60, 0x00, 0x00, 0x00,
>-  0xa2, 0x8a, 0x28, 0x00, 0x00, 0x00,
>-  0x53, 0x65, 0x36, 0x00, 0x00, 0x00,
>-  0x21, 0x32, 0x13, 0x00, 0x00, 0x00,
>-  0x10, 0x91, 0x09, 0x00, 0x00, 0x00,
>-  0x00, 0x70, 0x07, 0x00, 0x00, 0x00,
>-  0x0c, 0x10, 0xc1, 0x00, 0x00, 0x00,
>-  0x40, 0xc4, 0x0c, 0x00, 0x00, 0x00,
>-  0x6a, 0x06, 0xa0, 0x00, 0x00, 0x00,
>-  0x86, 0x08, 0x60, 0x00, 0x00, 0x00,
>-  0x24, 0x82, 0x48, 0x00, 0x00, 0x00,
>-  0x89, 0x08, 0x90, 0x00, 0x00, 0x00,
>-  0xc0, 0x2c, 0x02, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom24_22[132] = {
>-  0x53, 0x65, 0x36, 0x00, 0x00, 0x00,
>-  0x21, 0x32, 0x13, 0x00, 0x00, 0x00,
>-  0x10, 0x91, 0x09, 0x00, 0x00, 0x00,
>-  0x00, 0x70, 0x07, 0x00, 0x00, 0x00,
>-  0x0c, 0x10, 0xc1, 0x00, 0x00, 0x00,
>-  0x40, 0xc4, 0x0c, 0x00, 0x00, 0x00,
>-  0x6a, 0x06, 0xa0, 0x00, 0x00, 0x00,
>-  0x86, 0x08, 0x60, 0x00, 0x00, 0x00,
>-  0x24, 0x82, 0x48, 0x00, 0x00, 0x00,
>-  0x89, 0x08, 0x90, 0x00, 0x00, 0x00,
>-  0xc0, 0x2c, 0x02, 0x00, 0x00, 0x00,
>-  0x51, 0x45, 0x14, 0x00, 0x00, 0x00,
>-  0x45, 0x14, 0x51, 0x00, 0x00, 0x00,
>-  0x80, 0xd8, 0x0d, 0x00, 0x00, 0x00,
>-  0x24, 0x22, 0x42, 0x00, 0x00, 0x00,
>-  0x0a, 0x20, 0xa2, 0x00, 0x00, 0x00,
>-  0x00, 0xe0, 0x0e, 0x00, 0x00, 0x00,
>-  0xb8, 0x0b, 0x80, 0x00, 0x00, 0x00,
>-  0x09, 0x10, 0x91, 0x00, 0x00, 0x00,
>-  0x56, 0x05, 0x60, 0x00, 0x00, 0x00,
>-  0xa2, 0x8a, 0x28, 0x00, 0x00, 0x00,
>-  0x1a, 0xaa, 0xee, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom24_23[138] = {
>-  0x53, 0x65, 0x36, 0x00, 0x00, 0x00,
>-  0x21, 0x32, 0x13, 0x00, 0x00, 0x00,
>-  0x10, 0x91, 0x09, 0x00, 0x00, 0x00,
>-  0x00, 0x70, 0x07, 0x00, 0x00, 0x00,
>-  0x0c, 0x10, 0xc1, 0x00, 0x00, 0x00,
>-  0x40, 0xc4, 0x0c, 0x00, 0x00, 0x00,
>-  0x6a, 0x06, 0xa0, 0x00, 0x00, 0x00,
>-  0x86, 0x08, 0x60, 0x00, 0x00, 0x00,
>-  0x24, 0x82, 0x48, 0x00, 0x00, 0x00,
>-  0x89, 0x08, 0x90, 0x00, 0x00, 0x00,
>-  0xc0, 0x2c, 0x02, 0x00, 0x00, 0x00,
>-  0x10, 0x61, 0x06, 0x00, 0x00, 0x00,
>-  0x02, 0x30, 0x23, 0x00, 0x00, 0x00,
>-  0x40, 0x54, 0x05, 0x00, 0x00, 0x00,
>-  0x21, 0x82, 0x18, 0x00, 0x00, 0x00,
>-  0x81, 0x18, 0x11, 0x00, 0x00, 0x00,
>-  0x14, 0x81, 0x48, 0x00, 0x00, 0x00,
>-  0x98, 0x09, 0x80, 0x00, 0x00, 0x00,
>-  0x08, 0x90, 0x89, 0x00, 0x00, 0x00,
>-  0x62, 0x06, 0x20, 0x00, 0x00, 0x00,
>-  0x24, 0x22, 0x42, 0x00, 0x00, 0x00,
>-  0x8a, 0x08, 0xa0, 0x00, 0x00, 0x00,
>-  0x84, 0x48, 0x44, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom24_24[144] = {
>-  0x10, 0x61, 0x06, 0x00, 0x00, 0x00,
>-  0x02, 0x30, 0x23, 0x00, 0x00, 0x00,
>-  0x40, 0x54, 0x05, 0x00, 0x00, 0x00,
>-  0x21, 0x82, 0x18, 0x00, 0x00, 0x00,
>-  0x81, 0x18, 0x11, 0x00, 0x00, 0x00,
>-  0x14, 0x81, 0x48, 0x00, 0x00, 0x00,
>-  0x98, 0x09, 0x80, 0x00, 0x00, 0x00,
>-  0x08, 0x90, 0x89, 0x00, 0x00, 0x00,
>-  0x62, 0x06, 0x20, 0x00, 0x00, 0x00,
>-  0x24, 0x22, 0x42, 0x00, 0x00, 0x00,
>-  0x8a, 0x08, 0xa0, 0x00, 0x00, 0x00,
>-  0x84, 0x48, 0x44, 0x00, 0x00, 0x00,
>-  0x53, 0x65, 0x36, 0x00, 0x00, 0x00,
>-  0x21, 0x32, 0x13, 0x00, 0x00, 0x00,
>-  0x10, 0x91, 0x09, 0x00, 0x00, 0x00,
>-  0x00, 0x70, 0x07, 0x00, 0x00, 0x00,
>-  0x0c, 0x10, 0xc1, 0x00, 0x00, 0x00,
>-  0x40, 0xc4, 0x0c, 0x00, 0x00, 0x00,
>-  0x6a, 0x06, 0xa0, 0x00, 0x00, 0x00,
>-  0x86, 0x08, 0x60, 0x00, 0x00, 0x00,
>-  0x24, 0x82, 0x48, 0x00, 0x00, 0x00,
>-  0x89, 0x08, 0x90, 0x00, 0x00, 0x00,
>-  0xc0, 0x2c, 0x02, 0x00, 0x00, 0x00,
>-  0x88, 0x32, 0x59, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom24_3[18] = {
>-  0x9b, 0x29, 0xb2, 0x00, 0x00, 0x00,
>-  0x49, 0xd4, 0x9d, 0x00, 0x00, 0x00,
>-  0x3e, 0x83, 0xe8, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom24_4[24] = {
>-  0x8b, 0x28, 0xb2, 0x00, 0x00, 0x00,
>-  0x14, 0xb1, 0x4b, 0x00, 0x00, 0x00,
>-  0x22, 0xd2, 0x2d, 0x00, 0x00, 0x00,
>-  0x45, 0x54, 0x55, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom24_5[30] = {
>-  0x53, 0x65, 0x36, 0x00, 0x00, 0x00,
>-  0x64, 0xb6, 0x4b, 0x00, 0x00, 0x00,
>-  0x0e, 0xe0, 0xee, 0x00, 0x00, 0x00,
>-  0xa9, 0xca, 0x9c, 0x00, 0x00, 0x00,
>-  0xb8, 0x3b, 0x83, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom24_6[36] = {
>-  0xd1, 0x4d, 0x14, 0x00, 0x00, 0x00,
>-  0x45, 0x34, 0x53, 0x00, 0x00, 0x00,
>-  0x22, 0xd2, 0x2d, 0x00, 0x00, 0x00,
>-  0x16, 0xc1, 0x6c, 0x00, 0x00, 0x00,
>-  0x0b, 0xa0, 0xba, 0x00, 0x00, 0x00,
>-  0xe8, 0x8e, 0x88, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom24_7[42] = {
>-  0xd3, 0x65, 0x36, 0x00, 0x00, 0x00,
>-  0x25, 0x32, 0x53, 0x00, 0x00, 0x00,
>-  0x30, 0xd3, 0x05, 0x00, 0x00, 0x00,
>-  0x06, 0x48, 0x6c, 0x00, 0x00, 0x00,
>-  0xc0, 0xb8, 0x1b, 0x00, 0x00, 0x00,
>-  0x2a, 0xa2, 0xaa, 0x00, 0x00, 0x00,
>-  0xa8, 0x4e, 0x84, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom24_8[48] = {
>-  0x81, 0x60, 0x16, 0x00, 0x00, 0x00,
>-  0x40, 0x3c, 0x03, 0x00, 0x00, 0x00,
>-  0x10, 0x91, 0x09, 0x00, 0x00, 0x00,
>-  0x06, 0x50, 0x65, 0x00, 0x00, 0x00,
>-  0x20, 0x4a, 0x84, 0x00, 0x00, 0x00,
>-  0x8a, 0xa0, 0xaa, 0x00, 0x00, 0x00,
>-  0x33, 0x03, 0x30, 0x00, 0x00, 0x00,
>-  0x4c, 0x84, 0xc8, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom24_9[54] = {
>-  0xd3, 0x65, 0x36, 0x00, 0x00, 0x00,
>-  0x64, 0x26, 0x42, 0x00, 0x00, 0x00,
>-  0x18, 0x41, 0xc4, 0x00, 0x00, 0x00,
>-  0xa0, 0x4a, 0x04, 0x00, 0x00, 0x00,
>-  0x81, 0x38, 0x13, 0x00, 0x00, 0x00,
>-  0x22, 0xa2, 0x2a, 0x00, 0x00, 0x00,
>-  0x08, 0x70, 0x87, 0x00, 0x00, 0x00,
>-  0x04, 0x90, 0x49, 0x00, 0x00, 0x00,
>-  0x01, 0xc0, 0x1c, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom25_1[6] = {
>-  0xff, 0xff, 0xff, 0x80, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom25_10[60] = {
>-  0x51, 0x4d, 0x10, 0x00, 0x00, 0x00,
>-  0x45, 0x14, 0x45, 0x00, 0x00, 0x00,
>-  0x80, 0xd1, 0x09, 0x80, 0x00, 0x00,
>-  0x24, 0x2a, 0x05, 0x00, 0x00, 0x00,
>-  0x0a, 0x24, 0xa0, 0x80, 0x00, 0x00,
>-  0x00, 0xe4, 0x03, 0x00, 0x00, 0x00,
>-  0xb8, 0x08, 0x02, 0x80, 0x00, 0x00,
>-  0x09, 0x10, 0xc9, 0x00, 0x00, 0x00,
>-  0x56, 0x00, 0x58, 0x80, 0x00, 0x00,
>-  0xa2, 0x86, 0x22, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom25_11[66] = {
>-  0x53, 0x65, 0x12, 0x00, 0x00, 0x00,
>-  0x21, 0x32, 0x21, 0x00, 0x00, 0x00,
>-  0x10, 0x91, 0x34, 0x00, 0x00, 0x00,
>-  0x00, 0x72, 0x50, 0x00, 0x00, 0x00,
>-  0x0c, 0x11, 0x81, 0x80, 0x00, 0x00,
>-  0x40, 0xc0, 0xa2, 0x00, 0x00, 0x00,
>-  0x6a, 0x08, 0x88, 0x80, 0x00, 0x00,
>-  0x86, 0x00, 0x68, 0x00, 0x00, 0x00,
>-  0x24, 0x8e, 0x02, 0x00, 0x00, 0x00,
>-  0x89, 0x08, 0x44, 0x00, 0x00, 0x00,
>-  0xc0, 0x24, 0x41, 0x80, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom25_12[72] = {
>-  0x10, 0x62, 0x82, 0x80, 0x00, 0x00,
>-  0x02, 0x38, 0x45, 0x00, 0x00, 0x00,
>-  0x40, 0x56, 0x04, 0x00, 0x00, 0x00,
>-  0x21, 0x80, 0x54, 0x80, 0x00, 0x00,
>-  0x81, 0x10, 0x29, 0x80, 0x00, 0x00,
>-  0x14, 0x80, 0x13, 0x00, 0x00, 0x00,
>-  0x98, 0x04, 0x81, 0x00, 0x00, 0x00,
>-  0x08, 0x92, 0x48, 0x00, 0x00, 0x00,
>-  0x62, 0x09, 0x40, 0x00, 0x00, 0x00,
>-  0x24, 0x28, 0xa0, 0x00, 0x00, 0x00,
>-  0x8a, 0x01, 0x18, 0x00, 0x00, 0x00,
>-  0x84, 0x45, 0x22, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom25_13[78] = {
>-  0x51, 0x4d, 0x12, 0x00, 0x00, 0x00,
>-  0xc5, 0x14, 0x6d, 0x00, 0x00, 0x00,
>-  0x21, 0x81, 0x54, 0x80, 0x00, 0x00,
>-  0x12, 0x32, 0x17, 0x00, 0x00, 0x00,
>-  0x08, 0xe2, 0x8c, 0x80, 0x00, 0x00,
>-  0x2e, 0x0a, 0xa2, 0x00, 0x00, 0x00,
>-  0x53, 0x65, 0x92, 0x00, 0x00, 0x00,
>-  0x21, 0x32, 0x65, 0x00, 0x00, 0x00,
>-  0x90, 0x9b, 0x14, 0x00, 0x00, 0x00,
>-  0x02, 0x52, 0xb0, 0x80, 0x00, 0x00,
>-  0x06, 0xa1, 0x4c, 0x80, 0x00, 0x00,
>-  0x2c, 0x0c, 0x88, 0x80, 0x00, 0x00,
>-  0x88, 0x68, 0x4b, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom25_14[84] = {
>-  0x53, 0x65, 0x92, 0x00, 0x00, 0x00,
>-  0x21, 0x32, 0x65, 0x00, 0x00, 0x00,
>-  0x90, 0x9b, 0x14, 0x00, 0x00, 0x00,
>-  0x02, 0x52, 0xb0, 0x80, 0x00, 0x00,
>-  0x06, 0xa1, 0x4c, 0x80, 0x00, 0x00,
>-  0x2c, 0x0c, 0x88, 0x80, 0x00, 0x00,
>-  0x88, 0x68, 0x4b, 0x00, 0x00, 0x00,
>-  0x51, 0x4d, 0x12, 0x00, 0x00, 0x00,
>-  0xc5, 0x14, 0x6d, 0x00, 0x00, 0x00,
>-  0x21, 0x81, 0x54, 0x80, 0x00, 0x00,
>-  0x12, 0x32, 0x17, 0x00, 0x00, 0x00,
>-  0x08, 0xe2, 0x8c, 0x80, 0x00, 0x00,
>-  0x2e, 0x0a, 0xa2, 0x00, 0x00, 0x00,
>-  0x73, 0x76, 0x61, 0x80, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom25_15[90] = {
>-  0x53, 0x65, 0x92, 0x00, 0x00, 0x00,
>-  0x21, 0x32, 0x65, 0x00, 0x00, 0x00,
>-  0x90, 0x9b, 0x14, 0x00, 0x00, 0x00,
>-  0x02, 0x52, 0xb0, 0x80, 0x00, 0x00,
>-  0x06, 0xa1, 0x4c, 0x80, 0x00, 0x00,
>-  0x2c, 0x0c, 0x88, 0x80, 0x00, 0x00,
>-  0x88, 0x68, 0x4b, 0x00, 0x00, 0x00,
>-  0x20, 0x68, 0x0a, 0x80, 0x00, 0x00,
>-  0x80, 0x33, 0x09, 0x00, 0x00, 0x00,
>-  0x42, 0x41, 0x60, 0x80, 0x00, 0x00,
>-  0x01, 0x90, 0x33, 0x00, 0x00, 0x00,
>-  0x14, 0x14, 0x46, 0x00, 0x00, 0x00,
>-  0x0a, 0x80, 0x81, 0x80, 0x00, 0x00,
>-  0x38, 0x0d, 0x80, 0x00, 0x00, 0x00,
>-  0xc5, 0x0a, 0x14, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom25_16[96] = {
>-  0x20, 0x68, 0x0a, 0x80, 0x00, 0x00,
>-  0x80, 0x33, 0x09, 0x00, 0x00, 0x00,
>-  0x42, 0x41, 0x60, 0x80, 0x00, 0x00,
>-  0x01, 0x90, 0x33, 0x00, 0x00, 0x00,
>-  0x14, 0x14, 0x46, 0x00, 0x00, 0x00,
>-  0x0a, 0x80, 0x81, 0x80, 0x00, 0x00,
>-  0x38, 0x0d, 0x80, 0x00, 0x00, 0x00,
>-  0xc5, 0x0a, 0x14, 0x00, 0x00, 0x00,
>-  0x53, 0x65, 0x92, 0x00, 0x00, 0x00,
>-  0x21, 0x32, 0x65, 0x00, 0x00, 0x00,
>-  0x90, 0x9b, 0x14, 0x00, 0x00, 0x00,
>-  0x02, 0x52, 0xb0, 0x80, 0x00, 0x00,
>-  0x06, 0xa1, 0x4c, 0x80, 0x00, 0x00,
>-  0x2c, 0x0c, 0x88, 0x80, 0x00, 0x00,
>-  0x88, 0x68, 0x4b, 0x00, 0x00, 0x00,
>-  0x16, 0xe8, 0xdc, 0x80, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom25_17[102] = {
>-  0x20, 0x68, 0x0a, 0x80, 0x00, 0x00,
>-  0x80, 0x33, 0x09, 0x00, 0x00, 0x00,
>-  0x42, 0x41, 0x60, 0x80, 0x00, 0x00,
>-  0x01, 0x90, 0x33, 0x00, 0x00, 0x00,
>-  0x14, 0x14, 0x46, 0x00, 0x00, 0x00,
>-  0x0a, 0x80, 0x81, 0x80, 0x00, 0x00,
>-  0x38, 0x0d, 0x80, 0x00, 0x00, 0x00,
>-  0xc5, 0x0a, 0x14, 0x00, 0x00, 0x00,
>-  0x53, 0x65, 0x92, 0x00, 0x00, 0x00,
>-  0xe4, 0x26, 0x64, 0x00, 0x00, 0x00,
>-  0x24, 0x41, 0x44, 0x00, 0x00, 0x00,
>-  0xa1, 0x12, 0x14, 0x80, 0x00, 0x00,
>-  0x18, 0x30, 0x2c, 0x80, 0x00, 0x00,
>-  0x03, 0x99, 0x41, 0x00, 0x00, 0x00,
>-  0x8a, 0x18, 0x0a, 0x80, 0x00, 0x00,
>-  0x04, 0x90, 0xa9, 0x00, 0x00, 0x00,
>-  0x00, 0xe4, 0x01, 0x80, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom25_18[108] = {
>-  0x53, 0x65, 0x92, 0x00, 0x00, 0x00,
>-  0xe4, 0x26, 0x64, 0x00, 0x00, 0x00,
>-  0x24, 0x41, 0x44, 0x00, 0x00, 0x00,
>-  0xa1, 0x12, 0x14, 0x80, 0x00, 0x00,
>-  0x18, 0x30, 0x2c, 0x80, 0x00, 0x00,
>-  0x03, 0x99, 0x41, 0x00, 0x00, 0x00,
>-  0x8a, 0x18, 0x0a, 0x80, 0x00, 0x00,
>-  0x04, 0x90, 0xa9, 0x00, 0x00, 0x00,
>-  0x00, 0xe4, 0x01, 0x80, 0x00, 0x00,
>-  0x20, 0x68, 0x0a, 0x80, 0x00, 0x00,
>-  0x80, 0x33, 0x09, 0x00, 0x00, 0x00,
>-  0x42, 0x41, 0x60, 0x80, 0x00, 0x00,
>-  0x01, 0x90, 0x33, 0x00, 0x00, 0x00,
>-  0x14, 0x14, 0x46, 0x00, 0x00, 0x00,
>-  0x0a, 0x80, 0x81, 0x80, 0x00, 0x00,
>-  0x38, 0x0d, 0x80, 0x00, 0x00, 0x00,
>-  0xc5, 0x0a, 0x14, 0x00, 0x00, 0x00,
>-  0xce, 0x9b, 0xe1, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom25_19[114] = {
>-  0x53, 0x65, 0x92, 0x00, 0x00, 0x00,
>-  0xe4, 0x26, 0x64, 0x00, 0x00, 0x00,
>-  0x24, 0x41, 0x44, 0x00, 0x00, 0x00,
>-  0xa1, 0x12, 0x14, 0x80, 0x00, 0x00,
>-  0x18, 0x30, 0x2c, 0x80, 0x00, 0x00,
>-  0x03, 0x99, 0x41, 0x00, 0x00, 0x00,
>-  0x8a, 0x18, 0x0a, 0x80, 0x00, 0x00,
>-  0x04, 0x90, 0xa9, 0x00, 0x00, 0x00,
>-  0x00, 0xe4, 0x01, 0x80, 0x00, 0x00,
>-  0x51, 0x4d, 0x10, 0x00, 0x00, 0x00,
>-  0x45, 0x14, 0x45, 0x00, 0x00, 0x00,
>-  0x80, 0xd1, 0x09, 0x80, 0x00, 0x00,
>-  0x24, 0x2a, 0x05, 0x00, 0x00, 0x00,
>-  0x0a, 0x24, 0xa0, 0x80, 0x00, 0x00,
>-  0x00, 0xe4, 0x03, 0x00, 0x00, 0x00,
>-  0xb8, 0x08, 0x02, 0x80, 0x00, 0x00,
>-  0x09, 0x10, 0xc9, 0x00, 0x00, 0x00,
>-  0x56, 0x00, 0x58, 0x80, 0x00, 0x00,
>-  0xa2, 0x86, 0x22, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom25_2[12] = {
>-  0xec, 0xce, 0xcc, 0x00, 0x00, 0x00,
>-  0x93, 0xb1, 0xb3, 0x80, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom25_20[120] = {
>-  0x51, 0x4d, 0x10, 0x00, 0x00, 0x00,
>-  0x45, 0x14, 0x45, 0x00, 0x00, 0x00,
>-  0x80, 0xd1, 0x09, 0x80, 0x00, 0x00,
>-  0x24, 0x2a, 0x05, 0x00, 0x00, 0x00,
>-  0x0a, 0x24, 0xa0, 0x80, 0x00, 0x00,
>-  0x00, 0xe4, 0x03, 0x00, 0x00, 0x00,
>-  0xb8, 0x08, 0x02, 0x80, 0x00, 0x00,
>-  0x09, 0x10, 0xc9, 0x00, 0x00, 0x00,
>-  0x56, 0x00, 0x58, 0x80, 0x00, 0x00,
>-  0xa2, 0x86, 0x22, 0x00, 0x00, 0x00,
>-  0x53, 0x65, 0x92, 0x00, 0x00, 0x00,
>-  0xe4, 0x26, 0x64, 0x00, 0x00, 0x00,
>-  0x24, 0x41, 0x44, 0x00, 0x00, 0x00,
>-  0xa1, 0x12, 0x14, 0x80, 0x00, 0x00,
>-  0x18, 0x30, 0x2c, 0x80, 0x00, 0x00,
>-  0x03, 0x99, 0x41, 0x00, 0x00, 0x00,
>-  0x8a, 0x18, 0x0a, 0x80, 0x00, 0x00,
>-  0x04, 0x90, 0xa9, 0x00, 0x00, 0x00,
>-  0x00, 0xe4, 0x01, 0x80, 0x00, 0x00,
>-  0x1b, 0x8a, 0xa0, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom25_21[126] = {
>-  0x51, 0x4d, 0x10, 0x00, 0x00, 0x00,
>-  0x45, 0x14, 0x45, 0x00, 0x00, 0x00,
>-  0x80, 0xd1, 0x09, 0x80, 0x00, 0x00,
>-  0x24, 0x2a, 0x05, 0x00, 0x00, 0x00,
>-  0x0a, 0x24, 0xa0, 0x80, 0x00, 0x00,
>-  0x00, 0xe4, 0x03, 0x00, 0x00, 0x00,
>-  0xb8, 0x08, 0x02, 0x80, 0x00, 0x00,
>-  0x09, 0x10, 0xc9, 0x00, 0x00, 0x00,
>-  0x56, 0x00, 0x58, 0x80, 0x00, 0x00,
>-  0xa2, 0x86, 0x22, 0x00, 0x00, 0x00,
>-  0x53, 0x65, 0x12, 0x00, 0x00, 0x00,
>-  0x21, 0x32, 0x21, 0x00, 0x00, 0x00,
>-  0x10, 0x91, 0x34, 0x00, 0x00, 0x00,
>-  0x00, 0x72, 0x50, 0x00, 0x00, 0x00,
>-  0x0c, 0x11, 0x81, 0x80, 0x00, 0x00,
>-  0x40, 0xc0, 0xa2, 0x00, 0x00, 0x00,
>-  0x6a, 0x08, 0x88, 0x80, 0x00, 0x00,
>-  0x86, 0x00, 0x68, 0x00, 0x00, 0x00,
>-  0x24, 0x8e, 0x02, 0x00, 0x00, 0x00,
>-  0x89, 0x08, 0x44, 0x00, 0x00, 0x00,
>-  0xc0, 0x24, 0x41, 0x80, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom25_22[132] = {
>-  0x53, 0x65, 0x12, 0x00, 0x00, 0x00,
>-  0x21, 0x32, 0x21, 0x00, 0x00, 0x00,
>-  0x10, 0x91, 0x34, 0x00, 0x00, 0x00,
>-  0x00, 0x72, 0x50, 0x00, 0x00, 0x00,
>-  0x0c, 0x11, 0x81, 0x80, 0x00, 0x00,
>-  0x40, 0xc0, 0xa2, 0x00, 0x00, 0x00,
>-  0x6a, 0x08, 0x88, 0x80, 0x00, 0x00,
>-  0x86, 0x00, 0x68, 0x00, 0x00, 0x00,
>-  0x24, 0x8e, 0x02, 0x00, 0x00, 0x00,
>-  0x89, 0x08, 0x44, 0x00, 0x00, 0x00,
>-  0xc0, 0x24, 0x41, 0x80, 0x00, 0x00,
>-  0x51, 0x4d, 0x10, 0x00, 0x00, 0x00,
>-  0x45, 0x14, 0x45, 0x00, 0x00, 0x00,
>-  0x80, 0xd1, 0x09, 0x80, 0x00, 0x00,
>-  0x24, 0x2a, 0x05, 0x00, 0x00, 0x00,
>-  0x0a, 0x24, 0xa0, 0x80, 0x00, 0x00,
>-  0x00, 0xe4, 0x03, 0x00, 0x00, 0x00,
>-  0xb8, 0x08, 0x02, 0x80, 0x00, 0x00,
>-  0x09, 0x10, 0xc9, 0x00, 0x00, 0x00,
>-  0x56, 0x00, 0x58, 0x80, 0x00, 0x00,
>-  0xa2, 0x86, 0x22, 0x00, 0x00, 0x00,
>-  0x15, 0xa2, 0x99, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom25_23[138] = {
>-  0x53, 0x65, 0x12, 0x00, 0x00, 0x00,
>-  0x21, 0x32, 0x21, 0x00, 0x00, 0x00,
>-  0x10, 0x91, 0x34, 0x00, 0x00, 0x00,
>-  0x00, 0x72, 0x50, 0x00, 0x00, 0x00,
>-  0x0c, 0x11, 0x81, 0x80, 0x00, 0x00,
>-  0x40, 0xc0, 0xa2, 0x00, 0x00, 0x00,
>-  0x6a, 0x08, 0x88, 0x80, 0x00, 0x00,
>-  0x86, 0x00, 0x68, 0x00, 0x00, 0x00,
>-  0x24, 0x8e, 0x02, 0x00, 0x00, 0x00,
>-  0x89, 0x08, 0x44, 0x00, 0x00, 0x00,
>-  0xc0, 0x24, 0x41, 0x80, 0x00, 0x00,
>-  0x10, 0x62, 0x82, 0x80, 0x00, 0x00,
>-  0x02, 0x38, 0x45, 0x00, 0x00, 0x00,
>-  0x40, 0x56, 0x04, 0x00, 0x00, 0x00,
>-  0x21, 0x80, 0x54, 0x80, 0x00, 0x00,
>-  0x81, 0x10, 0x29, 0x80, 0x00, 0x00,
>-  0x14, 0x80, 0x13, 0x00, 0x00, 0x00,
>-  0x98, 0x04, 0x81, 0x00, 0x00, 0x00,
>-  0x08, 0x92, 0x48, 0x00, 0x00, 0x00,
>-  0x62, 0x09, 0x40, 0x00, 0x00, 0x00,
>-  0x24, 0x28, 0xa0, 0x00, 0x00, 0x00,
>-  0x8a, 0x01, 0x18, 0x00, 0x00, 0x00,
>-  0x84, 0x45, 0x22, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom25_24[144] = {
>-  0x10, 0x62, 0x82, 0x80, 0x00, 0x00,
>-  0x02, 0x38, 0x45, 0x00, 0x00, 0x00,
>-  0x40, 0x56, 0x04, 0x00, 0x00, 0x00,
>-  0x21, 0x80, 0x54, 0x80, 0x00, 0x00,
>-  0x81, 0x10, 0x29, 0x80, 0x00, 0x00,
>-  0x14, 0x80, 0x13, 0x00, 0x00, 0x00,
>-  0x98, 0x04, 0x81, 0x00, 0x00, 0x00,
>-  0x08, 0x92, 0x48, 0x00, 0x00, 0x00,
>-  0x62, 0x09, 0x40, 0x00, 0x00, 0x00,
>-  0x24, 0x28, 0xa0, 0x00, 0x00, 0x00,
>-  0x8a, 0x01, 0x18, 0x00, 0x00, 0x00,
>-  0x84, 0x45, 0x22, 0x00, 0x00, 0x00,
>-  0x53, 0x65, 0x12, 0x00, 0x00, 0x00,
>-  0x21, 0x32, 0x21, 0x00, 0x00, 0x00,
>-  0x10, 0x91, 0x34, 0x00, 0x00, 0x00,
>-  0x00, 0x72, 0x50, 0x00, 0x00, 0x00,
>-  0x0c, 0x11, 0x81, 0x80, 0x00, 0x00,
>-  0x40, 0xc0, 0xa2, 0x00, 0x00, 0x00,
>-  0x6a, 0x08, 0x88, 0x80, 0x00, 0x00,
>-  0x86, 0x00, 0x68, 0x00, 0x00, 0x00,
>-  0x24, 0x8e, 0x02, 0x00, 0x00, 0x00,
>-  0x89, 0x08, 0x44, 0x00, 0x00, 0x00,
>-  0xc0, 0x24, 0x41, 0x80, 0x00, 0x00,
>-  0xf9, 0x0c, 0x14, 0x80, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom25_25[150] = {
>-  0x10, 0x62, 0x82, 0x80, 0x00, 0x00,
>-  0x02, 0x38, 0x45, 0x00, 0x00, 0x00,
>-  0x40, 0x56, 0x04, 0x00, 0x00, 0x00,
>-  0x21, 0x80, 0x54, 0x80, 0x00, 0x00,
>-  0x81, 0x10, 0x29, 0x80, 0x00, 0x00,
>-  0x14, 0x80, 0x13, 0x00, 0x00, 0x00,
>-  0x98, 0x04, 0x81, 0x00, 0x00, 0x00,
>-  0x08, 0x92, 0x48, 0x00, 0x00, 0x00,
>-  0x62, 0x09, 0x40, 0x00, 0x00, 0x00,
>-  0x24, 0x28, 0xa0, 0x00, 0x00, 0x00,
>-  0x8a, 0x01, 0x18, 0x00, 0x00, 0x00,
>-  0x84, 0x45, 0x22, 0x00, 0x00, 0x00,
>-  0x10, 0x65, 0x12, 0x00, 0x00, 0x00,
>-  0x02, 0x36, 0x64, 0x00, 0x00, 0x00,
>-  0x40, 0x50, 0x54, 0x80, 0x00, 0x00,
>-  0x21, 0x88, 0x12, 0x00, 0x00, 0x00,
>-  0x81, 0x19, 0x40, 0x00, 0x00, 0x00,
>-  0x14, 0x83, 0x08, 0x00, 0x00, 0x00,
>-  0x98, 0x02, 0x11, 0x00, 0x00, 0x00,
>-  0x08, 0x90, 0x3c, 0x00, 0x00, 0x00,
>-  0x62, 0x0e, 0x80, 0x00, 0x00, 0x00,
>-  0x24, 0x20, 0xa1, 0x00, 0x00, 0x00,
>-  0x8a, 0x08, 0x01, 0x80, 0x00, 0x00,
>-  0x84, 0x40, 0x49, 0x00, 0x00, 0x00,
>-  0x1c, 0x20, 0x8a, 0x80, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom25_3[18] = {
>-  0x9b, 0x89, 0x9b, 0x00, 0x00, 0x00,
>-  0x4f, 0x14, 0x6d, 0x80, 0x00, 0x00,
>-  0x3c, 0x63, 0x72, 0x80, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom25_4[24] = {
>-  0x8b, 0x24, 0x9b, 0x00, 0x00, 0x00,
>-  0x14, 0xb2, 0x6d, 0x00, 0x00, 0x00,
>-  0x22, 0xd8, 0x56, 0x80, 0x00, 0x00,
>-  0x45, 0x55, 0x25, 0x80, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom25_5[30] = {
>-  0x53, 0x65, 0x13, 0x00, 0x00, 0x00,
>-  0x64, 0x26, 0x64, 0x00, 0x00, 0x00,
>-  0x0c, 0xc0, 0xc6, 0x80, 0x00, 0x00,
>-  0x82, 0xaa, 0x1c, 0x00, 0x00, 0x00,
>-  0x09, 0x32, 0x29, 0x80, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom25_6[36] = {
>-  0x51, 0x4d, 0x12, 0x00, 0x00, 0x00,
>-  0xc5, 0x14, 0x6d, 0x00, 0x00, 0x00,
>-  0x21, 0x81, 0x54, 0x80, 0x00, 0x00,
>-  0x12, 0x32, 0x17, 0x00, 0x00, 0x00,
>-  0x08, 0xe2, 0x8c, 0x80, 0x00, 0x00,
>-  0x2e, 0x0a, 0xa2, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom25_7[42] = {
>-  0x53, 0x65, 0x92, 0x00, 0x00, 0x00,
>-  0x21, 0x32, 0x65, 0x00, 0x00, 0x00,
>-  0x90, 0x9b, 0x14, 0x00, 0x00, 0x00,
>-  0x02, 0x52, 0xb0, 0x80, 0x00, 0x00,
>-  0x06, 0xa1, 0x4c, 0x80, 0x00, 0x00,
>-  0x2c, 0x0c, 0x88, 0x80, 0x00, 0x00,
>-  0x88, 0x68, 0x4b, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom25_8[48] = {
>-  0x20, 0x68, 0x0a, 0x80, 0x00, 0x00,
>-  0x80, 0x33, 0x09, 0x00, 0x00, 0x00,
>-  0x42, 0x41, 0x60, 0x80, 0x00, 0x00,
>-  0x01, 0x90, 0x33, 0x00, 0x00, 0x00,
>-  0x14, 0x14, 0x46, 0x00, 0x00, 0x00,
>-  0x0a, 0x80, 0x81, 0x80, 0x00, 0x00,
>-  0x38, 0x0d, 0x80, 0x00, 0x00, 0x00,
>-  0xc5, 0x0a, 0x14, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom25_9[54] = {
>-  0x53, 0x65, 0x92, 0x00, 0x00, 0x00,
>-  0xe4, 0x26, 0x64, 0x00, 0x00, 0x00,
>-  0x24, 0x41, 0x44, 0x00, 0x00, 0x00,
>-  0xa1, 0x12, 0x14, 0x80, 0x00, 0x00,
>-  0x18, 0x30, 0x2c, 0x80, 0x00, 0x00,
>-  0x03, 0x99, 0x41, 0x00, 0x00, 0x00,
>-  0x8a, 0x18, 0x0a, 0x80, 0x00, 0x00,
>-  0x04, 0x90, 0xa9, 0x00, 0x00, 0x00,
>-  0x00, 0xe4, 0x01, 0x80, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom26_1[6] = {
>-  0xff, 0xff, 0xff, 0xc0, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom26_10[60] = {
>-  0xd1, 0x06, 0x88, 0x00, 0x00, 0x00,
>-  0x44, 0x52, 0x22, 0x80, 0x00, 0x00,
>-  0x10, 0x98, 0x84, 0xc0, 0x00, 0x00,
>-  0xa0, 0x55, 0x02, 0x80, 0x00, 0x00,
>-  0x4a, 0x0a, 0x50, 0x40, 0x00, 0x00,
>-  0x40, 0x32, 0x01, 0x80, 0x00, 0x00,
>-  0x80, 0x2c, 0x01, 0x40, 0x00, 0x00,
>-  0x0c, 0x90, 0x64, 0x80, 0x00, 0x00,
>-  0x05, 0x88, 0x2c, 0x40, 0x00, 0x00,
>-  0x62, 0x23, 0x11, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom26_11[66] = {
>-  0x51, 0x22, 0x89, 0x00, 0x00, 0x00,
>-  0x22, 0x11, 0x10, 0x80, 0x00, 0x00,
>-  0x13, 0x40, 0x9a, 0x00, 0x00, 0x00,
>-  0x25, 0x01, 0x28, 0x00, 0x00, 0x00,
>-  0x18, 0x18, 0xc0, 0xc0, 0x00, 0x00,
>-  0x0a, 0x20, 0x51, 0x00, 0x00, 0x00,
>-  0x88, 0x8c, 0x44, 0x40, 0x00, 0x00,
>-  0x06, 0x80, 0x34, 0x00, 0x00, 0x00,
>-  0xe0, 0x27, 0x01, 0x00, 0x00, 0x00,
>-  0x84, 0x44, 0x22, 0x00, 0x00, 0x00,
>-  0x44, 0x1a, 0x20, 0xc0, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom26_12[72] = {
>-  0x28, 0x29, 0x41, 0x40, 0x00, 0x00,
>-  0x84, 0x54, 0x22, 0x80, 0x00, 0x00,
>-  0x60, 0x43, 0x02, 0x00, 0x00, 0x00,
>-  0x05, 0x48, 0x2a, 0x40, 0x00, 0x00,
>-  0x02, 0x98, 0x14, 0xc0, 0x00, 0x00,
>-  0x01, 0x30, 0x09, 0x80, 0x00, 0x00,
>-  0x48, 0x12, 0x40, 0x80, 0x00, 0x00,
>-  0x24, 0x81, 0x24, 0x00, 0x00, 0x00,
>-  0x94, 0x04, 0xa0, 0x00, 0x00, 0x00,
>-  0x8a, 0x04, 0x50, 0x00, 0x00, 0x00,
>-  0x11, 0x80, 0x8c, 0x00, 0x00, 0x00,
>-  0x52, 0x22, 0x91, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom26_13[78] = {
>-  0x51, 0x22, 0x89, 0x00, 0x00, 0x00,
>-  0x66, 0x43, 0x32, 0x00, 0x00, 0x00,
>-  0x05, 0x48, 0x2a, 0x40, 0x00, 0x00,
>-  0x81, 0x24, 0x09, 0x00, 0x00, 0x00,
>-  0x94, 0x04, 0xa0, 0x00, 0x00, 0x00,
>-  0x30, 0x81, 0x84, 0x00, 0x00, 0x00,
>-  0x21, 0x11, 0x08, 0x80, 0x00, 0x00,
>-  0x03, 0xc0, 0x1e, 0x00, 0x00, 0x00,
>-  0xe8, 0x07, 0x40, 0x00, 0x00, 0x00,
>-  0x0a, 0x10, 0x50, 0x80, 0x00, 0x00,
>-  0x80, 0x1c, 0x00, 0xc0, 0x00, 0x00,
>-  0x04, 0x90, 0x24, 0x80, 0x00, 0x00,
>-  0x08, 0xa8, 0x45, 0x40, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom26_14[84] = {
>-  0x59, 0x22, 0xc9, 0x00, 0x00, 0x00,
>-  0x26, 0x51, 0x32, 0x80, 0x00, 0x00,
>-  0xb1, 0x45, 0x8a, 0x00, 0x00, 0x00,
>-  0x2b, 0x09, 0x58, 0x40, 0x00, 0x00,
>-  0x14, 0xc8, 0xa6, 0x40, 0x00, 0x00,
>-  0xc8, 0x8e, 0x44, 0x40, 0x00, 0x00,
>-  0x84, 0xb4, 0x25, 0x80, 0x00, 0x00,
>-  0xd1, 0x26, 0x89, 0x00, 0x00, 0x00,
>-  0x46, 0xd2, 0x36, 0x80, 0x00, 0x00,
>-  0x15, 0x48, 0xaa, 0x40, 0x00, 0x00,
>-  0x21, 0x71, 0x0b, 0x80, 0x00, 0x00,
>-  0x28, 0xc9, 0x46, 0x40, 0x00, 0x00,
>-  0xaa, 0x25, 0x51, 0x00, 0x00, 0x00,
>-  0x5d, 0xa7, 0x78, 0x40, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom26_15[90] = {
>-  0x59, 0x22, 0xc9, 0x00, 0x00, 0x00,
>-  0x26, 0x51, 0x32, 0x80, 0x00, 0x00,
>-  0xb1, 0x45, 0x8a, 0x00, 0x00, 0x00,
>-  0x2b, 0x09, 0x58, 0x40, 0x00, 0x00,
>-  0x14, 0xc8, 0xa6, 0x40, 0x00, 0x00,
>-  0xc8, 0x8e, 0x44, 0x40, 0x00, 0x00,
>-  0x84, 0xb4, 0x25, 0x80, 0x00, 0x00,
>-  0x80, 0xac, 0x05, 0x40, 0x00, 0x00,
>-  0x30, 0x91, 0x84, 0x80, 0x00, 0x00,
>-  0x16, 0x08, 0xb0, 0x40, 0x00, 0x00,
>-  0x03, 0x30, 0x19, 0x80, 0x00, 0x00,
>-  0x44, 0x62, 0x23, 0x00, 0x00, 0x00,
>-  0x08, 0x18, 0x40, 0xc0, 0x00, 0x00,
>-  0xd8, 0x06, 0xc0, 0x00, 0x00, 0x00,
>-  0xa1, 0x45, 0x0a, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom26_16[96] = {
>-  0x80, 0xac, 0x05, 0x40, 0x00, 0x00,
>-  0x30, 0x91, 0x84, 0x80, 0x00, 0x00,
>-  0x16, 0x08, 0xb0, 0x40, 0x00, 0x00,
>-  0x03, 0x30, 0x19, 0x80, 0x00, 0x00,
>-  0x44, 0x62, 0x23, 0x00, 0x00, 0x00,
>-  0x08, 0x18, 0x40, 0xc0, 0x00, 0x00,
>-  0xd8, 0x06, 0xc0, 0x00, 0x00, 0x00,
>-  0xa1, 0x45, 0x0a, 0x00, 0x00, 0x00,
>-  0x59, 0x22, 0xc9, 0x00, 0x00, 0x00,
>-  0x26, 0x51, 0x32, 0x80, 0x00, 0x00,
>-  0xb1, 0x45, 0x8a, 0x00, 0x00, 0x00,
>-  0x2b, 0x09, 0x58, 0x40, 0x00, 0x00,
>-  0x14, 0xc8, 0xa6, 0x40, 0x00, 0x00,
>-  0xc8, 0x8e, 0x44, 0x40, 0x00, 0x00,
>-  0x84, 0xb4, 0x25, 0x80, 0x00, 0x00,
>-  0x3c, 0xaf, 0x88, 0x80, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom26_17[102] = {
>-  0x80, 0xac, 0x05, 0x40, 0x00, 0x00,
>-  0x30, 0x91, 0x84, 0x80, 0x00, 0x00,
>-  0x16, 0x08, 0xb0, 0x40, 0x00, 0x00,
>-  0x03, 0x30, 0x19, 0x80, 0x00, 0x00,
>-  0x44, 0x62, 0x23, 0x00, 0x00, 0x00,
>-  0x08, 0x18, 0x40, 0xc0, 0x00, 0x00,
>-  0xd8, 0x06, 0xc0, 0x00, 0x00, 0x00,
>-  0xa1, 0x45, 0x0a, 0x00, 0x00, 0x00,
>-  0x59, 0x22, 0xc9, 0x00, 0x00, 0x00,
>-  0x66, 0x43, 0x32, 0x00, 0x00, 0x00,
>-  0x14, 0x40, 0xa2, 0x00, 0x00, 0x00,
>-  0x21, 0x49, 0x0a, 0x40, 0x00, 0x00,
>-  0x02, 0xc8, 0x16, 0x40, 0x00, 0x00,
>-  0x94, 0x14, 0xa0, 0x80, 0x00, 0x00,
>-  0x80, 0xac, 0x05, 0x40, 0x00, 0x00,
>-  0x0a, 0x90, 0x54, 0x80, 0x00, 0x00,
>-  0x40, 0x1a, 0x00, 0xc0, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom26_18[108] = {
>-  0x59, 0x22, 0xc9, 0x00, 0x00, 0x00,
>-  0x66, 0x43, 0x32, 0x00, 0x00, 0x00,
>-  0x14, 0x40, 0xa2, 0x00, 0x00, 0x00,
>-  0x21, 0x49, 0x0a, 0x40, 0x00, 0x00,
>-  0x02, 0xc8, 0x16, 0x40, 0x00, 0x00,
>-  0x94, 0x14, 0xa0, 0x80, 0x00, 0x00,
>-  0x80, 0xac, 0x05, 0x40, 0x00, 0x00,
>-  0x0a, 0x90, 0x54, 0x80, 0x00, 0x00,
>-  0x40, 0x1a, 0x00, 0xc0, 0x00, 0x00,
>-  0x80, 0xac, 0x05, 0x40, 0x00, 0x00,
>-  0x30, 0x91, 0x84, 0x80, 0x00, 0x00,
>-  0x16, 0x08, 0xb0, 0x40, 0x00, 0x00,
>-  0x03, 0x30, 0x19, 0x80, 0x00, 0x00,
>-  0x44, 0x62, 0x23, 0x00, 0x00, 0x00,
>-  0x08, 0x18, 0x40, 0xc0, 0x00, 0x00,
>-  0xd8, 0x06, 0xc0, 0x00, 0x00, 0x00,
>-  0xa1, 0x45, 0x0a, 0x00, 0x00, 0x00,
>-  0xaa, 0x0c, 0x83, 0x80, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom26_19[114] = {
>-  0x59, 0x22, 0xc9, 0x00, 0x00, 0x00,
>-  0x66, 0x43, 0x32, 0x00, 0x00, 0x00,
>-  0x14, 0x40, 0xa2, 0x00, 0x00, 0x00,
>-  0x21, 0x49, 0x0a, 0x40, 0x00, 0x00,
>-  0x02, 0xc8, 0x16, 0x40, 0x00, 0x00,
>-  0x94, 0x14, 0xa0, 0x80, 0x00, 0x00,
>-  0x80, 0xac, 0x05, 0x40, 0x00, 0x00,
>-  0x0a, 0x90, 0x54, 0x80, 0x00, 0x00,
>-  0x40, 0x1a, 0x00, 0xc0, 0x00, 0x00,
>-  0xd1, 0x06, 0x88, 0x00, 0x00, 0x00,
>-  0x44, 0x52, 0x22, 0x80, 0x00, 0x00,
>-  0x10, 0x98, 0x84, 0xc0, 0x00, 0x00,
>-  0xa0, 0x55, 0x02, 0x80, 0x00, 0x00,
>-  0x4a, 0x0a, 0x50, 0x40, 0x00, 0x00,
>-  0x40, 0x32, 0x01, 0x80, 0x00, 0x00,
>-  0x80, 0x2c, 0x01, 0x40, 0x00, 0x00,
>-  0x0c, 0x90, 0x64, 0x80, 0x00, 0x00,
>-  0x05, 0x88, 0x2c, 0x40, 0x00, 0x00,
>-  0x62, 0x23, 0x11, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom26_2[12] = {
>-  0xec, 0xc7, 0x66, 0x00, 0x00, 0x00,
>-  0x1b, 0x38, 0xd9, 0xc0, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom26_20[120] = {
>-  0xd1, 0x06, 0x88, 0x00, 0x00, 0x00,
>-  0x44, 0x52, 0x22, 0x80, 0x00, 0x00,
>-  0x10, 0x98, 0x84, 0xc0, 0x00, 0x00,
>-  0xa0, 0x55, 0x02, 0x80, 0x00, 0x00,
>-  0x4a, 0x0a, 0x50, 0x40, 0x00, 0x00,
>-  0x40, 0x32, 0x01, 0x80, 0x00, 0x00,
>-  0x80, 0x2c, 0x01, 0x40, 0x00, 0x00,
>-  0x0c, 0x90, 0x64, 0x80, 0x00, 0x00,
>-  0x05, 0x88, 0x2c, 0x40, 0x00, 0x00,
>-  0x62, 0x23, 0x11, 0x00, 0x00, 0x00,
>-  0x59, 0x22, 0xc9, 0x00, 0x00, 0x00,
>-  0x66, 0x43, 0x32, 0x00, 0x00, 0x00,
>-  0x14, 0x40, 0xa2, 0x00, 0x00, 0x00,
>-  0x21, 0x49, 0x0a, 0x40, 0x00, 0x00,
>-  0x02, 0xc8, 0x16, 0x40, 0x00, 0x00,
>-  0x94, 0x14, 0xa0, 0x80, 0x00, 0x00,
>-  0x80, 0xac, 0x05, 0x40, 0x00, 0x00,
>-  0x0a, 0x90, 0x54, 0x80, 0x00, 0x00,
>-  0x40, 0x1a, 0x00, 0xc0, 0x00, 0x00,
>-  0xf4, 0x08, 0xec, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom26_21[126] = {
>-  0xd1, 0x06, 0x88, 0x00, 0x00, 0x00,
>-  0x44, 0x52, 0x22, 0x80, 0x00, 0x00,
>-  0x10, 0x98, 0x84, 0xc0, 0x00, 0x00,
>-  0xa0, 0x55, 0x02, 0x80, 0x00, 0x00,
>-  0x4a, 0x0a, 0x50, 0x40, 0x00, 0x00,
>-  0x40, 0x32, 0x01, 0x80, 0x00, 0x00,
>-  0x80, 0x2c, 0x01, 0x40, 0x00, 0x00,
>-  0x0c, 0x90, 0x64, 0x80, 0x00, 0x00,
>-  0x05, 0x88, 0x2c, 0x40, 0x00, 0x00,
>-  0x62, 0x23, 0x11, 0x00, 0x00, 0x00,
>-  0x51, 0x22, 0x89, 0x00, 0x00, 0x00,
>-  0x22, 0x11, 0x10, 0x80, 0x00, 0x00,
>-  0x13, 0x40, 0x9a, 0x00, 0x00, 0x00,
>-  0x25, 0x01, 0x28, 0x00, 0x00, 0x00,
>-  0x18, 0x18, 0xc0, 0xc0, 0x00, 0x00,
>-  0x0a, 0x20, 0x51, 0x00, 0x00, 0x00,
>-  0x88, 0x8c, 0x44, 0x40, 0x00, 0x00,
>-  0x06, 0x80, 0x34, 0x00, 0x00, 0x00,
>-  0xe0, 0x27, 0x01, 0x00, 0x00, 0x00,
>-  0x84, 0x44, 0x22, 0x00, 0x00, 0x00,
>-  0x44, 0x1a, 0x20, 0xc0, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom26_22[132] = {
>-  0x51, 0x22, 0x89, 0x00, 0x00, 0x00,
>-  0x22, 0x11, 0x10, 0x80, 0x00, 0x00,
>-  0x13, 0x40, 0x9a, 0x00, 0x00, 0x00,
>-  0x25, 0x01, 0x28, 0x00, 0x00, 0x00,
>-  0x18, 0x18, 0xc0, 0xc0, 0x00, 0x00,
>-  0x0a, 0x20, 0x51, 0x00, 0x00, 0x00,
>-  0x88, 0x8c, 0x44, 0x40, 0x00, 0x00,
>-  0x06, 0x80, 0x34, 0x00, 0x00, 0x00,
>-  0xe0, 0x27, 0x01, 0x00, 0x00, 0x00,
>-  0x84, 0x44, 0x22, 0x00, 0x00, 0x00,
>-  0x44, 0x1a, 0x20, 0xc0, 0x00, 0x00,
>-  0xd1, 0x06, 0x88, 0x00, 0x00, 0x00,
>-  0x44, 0x52, 0x22, 0x80, 0x00, 0x00,
>-  0x10, 0x98, 0x84, 0xc0, 0x00, 0x00,
>-  0xa0, 0x55, 0x02, 0x80, 0x00, 0x00,
>-  0x4a, 0x0a, 0x50, 0x40, 0x00, 0x00,
>-  0x40, 0x32, 0x01, 0x80, 0x00, 0x00,
>-  0x80, 0x2c, 0x01, 0x40, 0x00, 0x00,
>-  0x0c, 0x90, 0x64, 0x80, 0x00, 0x00,
>-  0x05, 0x88, 0x2c, 0x40, 0x00, 0x00,
>-  0x62, 0x23, 0x11, 0x00, 0x00, 0x00,
>-  0x13, 0xc6, 0x6b, 0x40, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom26_23[138] = {
>-  0x51, 0x22, 0x89, 0x00, 0x00, 0x00,
>-  0x22, 0x11, 0x10, 0x80, 0x00, 0x00,
>-  0x13, 0x40, 0x9a, 0x00, 0x00, 0x00,
>-  0x25, 0x01, 0x28, 0x00, 0x00, 0x00,
>-  0x18, 0x18, 0xc0, 0xc0, 0x00, 0x00,
>-  0x0a, 0x20, 0x51, 0x00, 0x00, 0x00,
>-  0x88, 0x8c, 0x44, 0x40, 0x00, 0x00,
>-  0x06, 0x80, 0x34, 0x00, 0x00, 0x00,
>-  0xe0, 0x27, 0x01, 0x00, 0x00, 0x00,
>-  0x84, 0x44, 0x22, 0x00, 0x00, 0x00,
>-  0x44, 0x1a, 0x20, 0xc0, 0x00, 0x00,
>-  0x28, 0x29, 0x41, 0x40, 0x00, 0x00,
>-  0x84, 0x54, 0x22, 0x80, 0x00, 0x00,
>-  0x60, 0x43, 0x02, 0x00, 0x00, 0x00,
>-  0x05, 0x48, 0x2a, 0x40, 0x00, 0x00,
>-  0x02, 0x98, 0x14, 0xc0, 0x00, 0x00,
>-  0x01, 0x30, 0x09, 0x80, 0x00, 0x00,
>-  0x48, 0x12, 0x40, 0x80, 0x00, 0x00,
>-  0x24, 0x81, 0x24, 0x00, 0x00, 0x00,
>-  0x94, 0x04, 0xa0, 0x00, 0x00, 0x00,
>-  0x8a, 0x04, 0x50, 0x00, 0x00, 0x00,
>-  0x11, 0x80, 0x8c, 0x00, 0x00, 0x00,
>-  0x52, 0x22, 0x91, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom26_24[144] = {
>-  0x28, 0x29, 0x41, 0x40, 0x00, 0x00,
>-  0x84, 0x54, 0x22, 0x80, 0x00, 0x00,
>-  0x60, 0x43, 0x02, 0x00, 0x00, 0x00,
>-  0x05, 0x48, 0x2a, 0x40, 0x00, 0x00,
>-  0x02, 0x98, 0x14, 0xc0, 0x00, 0x00,
>-  0x01, 0x30, 0x09, 0x80, 0x00, 0x00,
>-  0x48, 0x12, 0x40, 0x80, 0x00, 0x00,
>-  0x24, 0x81, 0x24, 0x00, 0x00, 0x00,
>-  0x94, 0x04, 0xa0, 0x00, 0x00, 0x00,
>-  0x8a, 0x04, 0x50, 0x00, 0x00, 0x00,
>-  0x11, 0x80, 0x8c, 0x00, 0x00, 0x00,
>-  0x52, 0x22, 0x91, 0x00, 0x00, 0x00,
>-  0x51, 0x22, 0x89, 0x00, 0x00, 0x00,
>-  0x22, 0x11, 0x10, 0x80, 0x00, 0x00,
>-  0x13, 0x40, 0x9a, 0x00, 0x00, 0x00,
>-  0x25, 0x01, 0x28, 0x00, 0x00, 0x00,
>-  0x18, 0x18, 0xc0, 0xc0, 0x00, 0x00,
>-  0x0a, 0x20, 0x51, 0x00, 0x00, 0x00,
>-  0x88, 0x8c, 0x44, 0x40, 0x00, 0x00,
>-  0x06, 0x80, 0x34, 0x00, 0x00, 0x00,
>-  0xe0, 0x27, 0x01, 0x00, 0x00, 0x00,
>-  0x84, 0x44, 0x22, 0x00, 0x00, 0x00,
>-  0x44, 0x1a, 0x20, 0xc0, 0x00, 0x00,
>-  0xdb, 0x4d, 0xd8, 0x40, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom26_25[150] = {
>-  0x28, 0x29, 0x41, 0x40, 0x00, 0x00,
>-  0x84, 0x54, 0x22, 0x80, 0x00, 0x00,
>-  0x60, 0x43, 0x02, 0x00, 0x00, 0x00,
>-  0x05, 0x48, 0x2a, 0x40, 0x00, 0x00,
>-  0x02, 0x98, 0x14, 0xc0, 0x00, 0x00,
>-  0x01, 0x30, 0x09, 0x80, 0x00, 0x00,
>-  0x48, 0x12, 0x40, 0x80, 0x00, 0x00,
>-  0x24, 0x81, 0x24, 0x00, 0x00, 0x00,
>-  0x94, 0x04, 0xa0, 0x00, 0x00, 0x00,
>-  0x8a, 0x04, 0x50, 0x00, 0x00, 0x00,
>-  0x11, 0x80, 0x8c, 0x00, 0x00, 0x00,
>-  0x52, 0x22, 0x91, 0x00, 0x00, 0x00,
>-  0x51, 0x22, 0x89, 0x00, 0x00, 0x00,
>-  0x66, 0x43, 0x32, 0x00, 0x00, 0x00,
>-  0x05, 0x48, 0x2a, 0x40, 0x00, 0x00,
>-  0x81, 0x24, 0x09, 0x00, 0x00, 0x00,
>-  0x94, 0x04, 0xa0, 0x00, 0x00, 0x00,
>-  0x30, 0x81, 0x84, 0x00, 0x00, 0x00,
>-  0x21, 0x11, 0x08, 0x80, 0x00, 0x00,
>-  0x03, 0xc0, 0x1e, 0x00, 0x00, 0x00,
>-  0xe8, 0x07, 0x40, 0x00, 0x00, 0x00,
>-  0x0a, 0x10, 0x50, 0x80, 0x00, 0x00,
>-  0x80, 0x1c, 0x00, 0xc0, 0x00, 0x00,
>-  0x04, 0x90, 0x24, 0x80, 0x00, 0x00,
>-  0x08, 0xa8, 0x45, 0x40, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom26_26[156] = {
>-  0x51, 0x22, 0x89, 0x00, 0x00, 0x00,
>-  0x66, 0x43, 0x32, 0x00, 0x00, 0x00,
>-  0x05, 0x48, 0x2a, 0x40, 0x00, 0x00,
>-  0x81, 0x24, 0x09, 0x00, 0x00, 0x00,
>-  0x94, 0x04, 0xa0, 0x00, 0x00, 0x00,
>-  0x30, 0x81, 0x84, 0x00, 0x00, 0x00,
>-  0x21, 0x11, 0x08, 0x80, 0x00, 0x00,
>-  0x03, 0xc0, 0x1e, 0x00, 0x00, 0x00,
>-  0xe8, 0x07, 0x40, 0x00, 0x00, 0x00,
>-  0x0a, 0x10, 0x50, 0x80, 0x00, 0x00,
>-  0x80, 0x1c, 0x00, 0xc0, 0x00, 0x00,
>-  0x04, 0x90, 0x24, 0x80, 0x00, 0x00,
>-  0x08, 0xa8, 0x45, 0x40, 0x00, 0x00,
>-  0x28, 0x29, 0x41, 0x40, 0x00, 0x00,
>-  0x84, 0x54, 0x22, 0x80, 0x00, 0x00,
>-  0x60, 0x43, 0x02, 0x00, 0x00, 0x00,
>-  0x05, 0x48, 0x2a, 0x40, 0x00, 0x00,
>-  0x02, 0x98, 0x14, 0xc0, 0x00, 0x00,
>-  0x01, 0x30, 0x09, 0x80, 0x00, 0x00,
>-  0x48, 0x12, 0x40, 0x80, 0x00, 0x00,
>-  0x24, 0x81, 0x24, 0x00, 0x00, 0x00,
>-  0x94, 0x04, 0xa0, 0x00, 0x00, 0x00,
>-  0x8a, 0x04, 0x50, 0x00, 0x00, 0x00,
>-  0x11, 0x80, 0x8c, 0x00, 0x00, 0x00,
>-  0x52, 0x22, 0x91, 0x00, 0x00, 0x00,
>-  0xf9, 0x13, 0x51, 0x80, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom26_3[18] = {
>-  0x99, 0xb4, 0xcd, 0x80, 0x00, 0x00,
>-  0x46, 0xda, 0x36, 0xc0, 0x00, 0x00,
>-  0x37, 0x29, 0xb9, 0x40, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom26_4[24] = {
>-  0x49, 0xb2, 0x4d, 0x80, 0x00, 0x00,
>-  0x26, 0xd1, 0x36, 0x80, 0x00, 0x00,
>-  0x85, 0x6c, 0x2b, 0x40, 0x00, 0x00,
>-  0x52, 0x5a, 0x92, 0xc0, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom26_5[30] = {
>-  0x51, 0x32, 0x89, 0x80, 0x00, 0x00,
>-  0x66, 0x43, 0x32, 0x00, 0x00, 0x00,
>-  0x0c, 0x68, 0x63, 0x40, 0x00, 0x00,
>-  0xa1, 0xc5, 0x0e, 0x00, 0x00, 0x00,
>-  0x22, 0x99, 0x14, 0xc0, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom26_6[36] = {
>-  0xd1, 0x26, 0x89, 0x00, 0x00, 0x00,
>-  0x46, 0xd2, 0x36, 0x80, 0x00, 0x00,
>-  0x15, 0x48, 0xaa, 0x40, 0x00, 0x00,
>-  0x21, 0x71, 0x0b, 0x80, 0x00, 0x00,
>-  0x28, 0xc9, 0x46, 0x40, 0x00, 0x00,
>-  0xaa, 0x25, 0x51, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom26_7[42] = {
>-  0x59, 0x22, 0xc9, 0x00, 0x00, 0x00,
>-  0x26, 0x51, 0x32, 0x80, 0x00, 0x00,
>-  0xb1, 0x45, 0x8a, 0x00, 0x00, 0x00,
>-  0x2b, 0x09, 0x58, 0x40, 0x00, 0x00,
>-  0x14, 0xc8, 0xa6, 0x40, 0x00, 0x00,
>-  0xc8, 0x8e, 0x44, 0x40, 0x00, 0x00,
>-  0x84, 0xb4, 0x25, 0x80, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom26_8[48] = {
>-  0x80, 0xac, 0x05, 0x40, 0x00, 0x00,
>-  0x30, 0x91, 0x84, 0x80, 0x00, 0x00,
>-  0x16, 0x08, 0xb0, 0x40, 0x00, 0x00,
>-  0x03, 0x30, 0x19, 0x80, 0x00, 0x00,
>-  0x44, 0x62, 0x23, 0x00, 0x00, 0x00,
>-  0x08, 0x18, 0x40, 0xc0, 0x00, 0x00,
>-  0xd8, 0x06, 0xc0, 0x00, 0x00, 0x00,
>-  0xa1, 0x45, 0x0a, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom26_9[54] = {
>-  0x59, 0x22, 0xc9, 0x00, 0x00, 0x00,
>-  0x66, 0x43, 0x32, 0x00, 0x00, 0x00,
>-  0x14, 0x40, 0xa2, 0x00, 0x00, 0x00,
>-  0x21, 0x49, 0x0a, 0x40, 0x00, 0x00,
>-  0x02, 0xc8, 0x16, 0x40, 0x00, 0x00,
>-  0x94, 0x14, 0xa0, 0x80, 0x00, 0x00,
>-  0x80, 0xac, 0x05, 0x40, 0x00, 0x00,
>-  0x0a, 0x90, 0x54, 0x80, 0x00, 0x00,
>-  0x40, 0x1a, 0x00, 0xc0, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom27_1[6] = {
>-  0xff, 0xff, 0xff, 0xe0, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom27_10[60] = {
>-  0xd1, 0x06, 0x06, 0xa0, 0x00, 0x00,
>-  0x44, 0x50, 0xea, 0x00, 0x00, 0x00,
>-  0x10, 0x9e, 0xa0, 0x40, 0x00, 0x00,
>-  0xa0, 0x50, 0x13, 0x00, 0x00, 0x00,
>-  0x4a, 0x08, 0x21, 0x40, 0x00, 0x00,
>-  0x40, 0x31, 0x04, 0xc0, 0x00, 0x00,
>-  0x80, 0x2a, 0x02, 0x20, 0x00, 0x00,
>-  0x0c, 0x90, 0x44, 0x20, 0x00, 0x00,
>-  0x05, 0x8b, 0x40, 0x00, 0x00, 0x00,
>-  0x62, 0x21, 0x18, 0x80, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom27_11[66] = {
>-  0x51, 0x23, 0x16, 0x80, 0x00, 0x00,
>-  0x22, 0x11, 0xa9, 0x00, 0x00, 0x00,
>-  0x13, 0x40, 0xa0, 0xa0, 0x00, 0x00,
>-  0x25, 0x06, 0x28, 0x40, 0x00, 0x00,
>-  0x18, 0x19, 0x10, 0x60, 0x00, 0x00,
>-  0x0a, 0x24, 0x45, 0xc0, 0x00, 0x00,
>-  0x88, 0x8a, 0x12, 0xa0, 0x00, 0x00,
>-  0x06, 0x81, 0x45, 0x20, 0x00, 0x00,
>-  0xe0, 0x24, 0xa1, 0x00, 0x00, 0x00,
>-  0x84, 0x40, 0xd8, 0x20, 0x00, 0x00,
>-  0x44, 0x19, 0x16, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom27_12[72] = {
>-  0x28, 0x2c, 0x08, 0x20, 0x00, 0x00,
>-  0x84, 0x52, 0x03, 0x40, 0x00, 0x00,
>-  0x60, 0x44, 0x81, 0x20, 0x00, 0x00,
>-  0x05, 0x49, 0x41, 0x40, 0x00, 0x00,
>-  0x02, 0x9a, 0x90, 0x80, 0x00, 0x00,
>-  0x01, 0x32, 0x0c, 0x40, 0x00, 0x00,
>-  0x48, 0x10, 0x49, 0x80, 0x00, 0x00,
>-  0x24, 0x82, 0x42, 0x20, 0x00, 0x00,
>-  0x94, 0x00, 0x22, 0x20, 0x00, 0x00,
>-  0x8a, 0x00, 0x74, 0x00, 0x00, 0x00,
>-  0x11, 0x85, 0x2c, 0x80, 0x00, 0x00,
>-  0x52, 0x20, 0x90, 0x60, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom27_13[78] = {
>-  0x51, 0x23, 0x12, 0xa0, 0x00, 0x00,
>-  0x66, 0x41, 0xa3, 0x00, 0x00, 0x00,
>-  0x05, 0x4a, 0x40, 0x20, 0x00, 0x00,
>-  0x81, 0x20, 0x05, 0x60, 0x00, 0x00,
>-  0x94, 0x01, 0x40, 0x40, 0x00, 0x00,
>-  0x30, 0x84, 0x08, 0x40, 0x00, 0x00,
>-  0x21, 0x11, 0x18, 0x20, 0x00, 0x00,
>-  0x03, 0xc0, 0x34, 0x00, 0x00, 0x00,
>-  0xe8, 0x04, 0x00, 0xa0, 0x00, 0x00,
>-  0x0a, 0x11, 0x80, 0x80, 0x00, 0x00,
>-  0x80, 0x1c, 0x61, 0x00, 0x00, 0x00,
>-  0x04, 0x92, 0xa0, 0x00, 0x00, 0x00,
>-  0x08, 0xac, 0x06, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom27_14[84] = {
>-  0x59, 0x23, 0x12, 0xa0, 0x00, 0x00,
>-  0x26, 0x55, 0xc9, 0x00, 0x00, 0x00,
>-  0xb1, 0x40, 0xc5, 0xa0, 0x00, 0x00,
>-  0x2b, 0x0a, 0xa4, 0xc0, 0x00, 0x00,
>-  0x14, 0xc8, 0x33, 0x60, 0x00, 0x00,
>-  0xc8, 0x8c, 0x2a, 0xa0, 0x00, 0x00,
>-  0x84, 0xb5, 0x54, 0x40, 0x00, 0x00,
>-  0xd1, 0x22, 0x52, 0xa0, 0x00, 0x00,
>-  0x46, 0xd4, 0xaa, 0x40, 0x00, 0x00,
>-  0x15, 0x48, 0xa5, 0xa0, 0x00, 0x00,
>-  0x21, 0x72, 0x8d, 0x40, 0x00, 0x00,
>-  0x28, 0xc9, 0x13, 0x60, 0x00, 0x00,
>-  0xaa, 0x24, 0x44, 0x60, 0x00, 0x00,
>-  0x0a, 0xe7, 0x3b, 0x20, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom27_15[90] = {
>-  0x59, 0x23, 0x12, 0xa0, 0x00, 0x00,
>-  0x26, 0x55, 0xc9, 0x00, 0x00, 0x00,
>-  0xb1, 0x40, 0xc5, 0xa0, 0x00, 0x00,
>-  0x2b, 0x0a, 0xa4, 0xc0, 0x00, 0x00,
>-  0x14, 0xc8, 0x33, 0x60, 0x00, 0x00,
>-  0xc8, 0x8c, 0x2a, 0xa0, 0x00, 0x00,
>-  0x84, 0xb5, 0x54, 0x40, 0x00, 0x00,
>-  0x80, 0xae, 0x00, 0xa0, 0x00, 0x00,
>-  0x30, 0x92, 0x0b, 0x00, 0x00, 0x00,
>-  0x16, 0x0c, 0x41, 0x80, 0x00, 0x00,
>-  0x03, 0x31, 0x05, 0x20, 0x00, 0x00,
>-  0x44, 0x60, 0x52, 0x40, 0x00, 0x00,
>-  0x08, 0x18, 0x24, 0xc0, 0x00, 0x00,
>-  0xd8, 0x04, 0xa2, 0x00, 0x00, 0x00,
>-  0xa1, 0x43, 0x90, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom27_16[96] = {
>-  0x80, 0xae, 0x00, 0xa0, 0x00, 0x00,
>-  0x30, 0x92, 0x0b, 0x00, 0x00, 0x00,
>-  0x16, 0x0c, 0x41, 0x80, 0x00, 0x00,
>-  0x03, 0x31, 0x05, 0x20, 0x00, 0x00,
>-  0x44, 0x60, 0x52, 0x40, 0x00, 0x00,
>-  0x08, 0x18, 0x24, 0xc0, 0x00, 0x00,
>-  0xd8, 0x04, 0xa2, 0x00, 0x00, 0x00,
>-  0xa1, 0x43, 0x90, 0x00, 0x00, 0x00,
>-  0x59, 0x23, 0x12, 0xa0, 0x00, 0x00,
>-  0x26, 0x55, 0xc9, 0x00, 0x00, 0x00,
>-  0xb1, 0x40, 0xc5, 0xa0, 0x00, 0x00,
>-  0x2b, 0x0a, 0xa4, 0xc0, 0x00, 0x00,
>-  0x14, 0xc8, 0x33, 0x60, 0x00, 0x00,
>-  0xc8, 0x8c, 0x2a, 0xa0, 0x00, 0x00,
>-  0x84, 0xb5, 0x54, 0x40, 0x00, 0x00,
>-  0x01, 0x50, 0xfb, 0xe0, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom27_17[102] = {
>-  0x80, 0xae, 0x00, 0xa0, 0x00, 0x00,
>-  0x30, 0x92, 0x0b, 0x00, 0x00, 0x00,
>-  0x16, 0x0c, 0x41, 0x80, 0x00, 0x00,
>-  0x03, 0x31, 0x05, 0x20, 0x00, 0x00,
>-  0x44, 0x60, 0x52, 0x40, 0x00, 0x00,
>-  0x08, 0x18, 0x24, 0xc0, 0x00, 0x00,
>-  0xd8, 0x04, 0xa2, 0x00, 0x00, 0x00,
>-  0xa1, 0x43, 0x90, 0x00, 0x00, 0x00,
>-  0x59, 0x25, 0x12, 0xa0, 0x00, 0x00,
>-  0x66, 0x41, 0xa3, 0x00, 0x00, 0x00,
>-  0x14, 0x42, 0x51, 0x20, 0x00, 0x00,
>-  0x21, 0x49, 0x05, 0x40, 0x00, 0x00,
>-  0x02, 0xc8, 0x8c, 0x20, 0x00, 0x00,
>-  0x94, 0x12, 0x48, 0x40, 0x00, 0x00,
>-  0x80, 0xac, 0x30, 0x60, 0x00, 0x00,
>-  0x0a, 0x91, 0x06, 0xa0, 0x00, 0x00,
>-  0x40, 0x1c, 0x42, 0x40, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom27_18[108] = {
>-  0x59, 0x25, 0x12, 0xa0, 0x00, 0x00,
>-  0x66, 0x41, 0xa3, 0x00, 0x00, 0x00,
>-  0x14, 0x42, 0x51, 0x20, 0x00, 0x00,
>-  0x21, 0x49, 0x05, 0x40, 0x00, 0x00,
>-  0x02, 0xc8, 0x8c, 0x20, 0x00, 0x00,
>-  0x94, 0x12, 0x48, 0x40, 0x00, 0x00,
>-  0x80, 0xac, 0x30, 0x60, 0x00, 0x00,
>-  0x0a, 0x91, 0x06, 0xa0, 0x00, 0x00,
>-  0x40, 0x1c, 0x42, 0x40, 0x00, 0x00,
>-  0x80, 0xae, 0x00, 0xa0, 0x00, 0x00,
>-  0x30, 0x92, 0x0b, 0x00, 0x00, 0x00,
>-  0x16, 0x0c, 0x41, 0x80, 0x00, 0x00,
>-  0x03, 0x31, 0x05, 0x20, 0x00, 0x00,
>-  0x44, 0x60, 0x52, 0x40, 0x00, 0x00,
>-  0x08, 0x18, 0x24, 0xc0, 0x00, 0x00,
>-  0xd8, 0x04, 0xa2, 0x00, 0x00, 0x00,
>-  0xa1, 0x43, 0x90, 0x00, 0x00, 0x00,
>-  0x53, 0xc3, 0x33, 0x80, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom27_19[114] = {
>-  0x59, 0x25, 0x12, 0xa0, 0x00, 0x00,
>-  0x66, 0x41, 0xa3, 0x00, 0x00, 0x00,
>-  0x14, 0x42, 0x51, 0x20, 0x00, 0x00,
>-  0x21, 0x49, 0x05, 0x40, 0x00, 0x00,
>-  0x02, 0xc8, 0x8c, 0x20, 0x00, 0x00,
>-  0x94, 0x12, 0x48, 0x40, 0x00, 0x00,
>-  0x80, 0xac, 0x30, 0x60, 0x00, 0x00,
>-  0x0a, 0x91, 0x06, 0xa0, 0x00, 0x00,
>-  0x40, 0x1c, 0x42, 0x40, 0x00, 0x00,
>-  0xd1, 0x06, 0x06, 0xa0, 0x00, 0x00,
>-  0x44, 0x50, 0xea, 0x00, 0x00, 0x00,
>-  0x10, 0x9e, 0xa0, 0x40, 0x00, 0x00,
>-  0xa0, 0x50, 0x13, 0x00, 0x00, 0x00,
>-  0x4a, 0x08, 0x21, 0x40, 0x00, 0x00,
>-  0x40, 0x31, 0x04, 0xc0, 0x00, 0x00,
>-  0x80, 0x2a, 0x02, 0x20, 0x00, 0x00,
>-  0x0c, 0x90, 0x44, 0x20, 0x00, 0x00,
>-  0x05, 0x8b, 0x40, 0x00, 0x00, 0x00,
>-  0x62, 0x21, 0x18, 0x80, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom27_2[12] = {
>-  0xec, 0xc7, 0x67, 0x40, 0x00, 0x00,
>-  0x1b, 0x39, 0xdc, 0xe0, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom27_20[120] = {
>-  0xd1, 0x06, 0x06, 0xa0, 0x00, 0x00,
>-  0x44, 0x50, 0xea, 0x00, 0x00, 0x00,
>-  0x10, 0x9e, 0xa0, 0x40, 0x00, 0x00,
>-  0xa0, 0x50, 0x13, 0x00, 0x00, 0x00,
>-  0x4a, 0x08, 0x21, 0x40, 0x00, 0x00,
>-  0x40, 0x31, 0x04, 0xc0, 0x00, 0x00,
>-  0x80, 0x2a, 0x02, 0x20, 0x00, 0x00,
>-  0x0c, 0x90, 0x44, 0x20, 0x00, 0x00,
>-  0x05, 0x8b, 0x40, 0x00, 0x00, 0x00,
>-  0x62, 0x21, 0x18, 0x80, 0x00, 0x00,
>-  0x59, 0x25, 0x12, 0xa0, 0x00, 0x00,
>-  0x66, 0x41, 0xa3, 0x00, 0x00, 0x00,
>-  0x14, 0x42, 0x51, 0x20, 0x00, 0x00,
>-  0x21, 0x49, 0x05, 0x40, 0x00, 0x00,
>-  0x02, 0xc8, 0x8c, 0x20, 0x00, 0x00,
>-  0x94, 0x12, 0x48, 0x40, 0x00, 0x00,
>-  0x80, 0xac, 0x30, 0x60, 0x00, 0x00,
>-  0x0a, 0x91, 0x06, 0xa0, 0x00, 0x00,
>-  0x40, 0x1c, 0x42, 0x40, 0x00, 0x00,
>-  0xcb, 0xff, 0x6f, 0xc0, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom27_21[126] = {
>-  0xd1, 0x06, 0x06, 0xa0, 0x00, 0x00,
>-  0x44, 0x50, 0xea, 0x00, 0x00, 0x00,
>-  0x10, 0x9e, 0xa0, 0x40, 0x00, 0x00,
>-  0xa0, 0x50, 0x13, 0x00, 0x00, 0x00,
>-  0x4a, 0x08, 0x21, 0x40, 0x00, 0x00,
>-  0x40, 0x31, 0x04, 0xc0, 0x00, 0x00,
>-  0x80, 0x2a, 0x02, 0x20, 0x00, 0x00,
>-  0x0c, 0x90, 0x44, 0x20, 0x00, 0x00,
>-  0x05, 0x8b, 0x40, 0x00, 0x00, 0x00,
>-  0x62, 0x21, 0x18, 0x80, 0x00, 0x00,
>-  0x51, 0x23, 0x16, 0x80, 0x00, 0x00,
>-  0x22, 0x11, 0xa9, 0x00, 0x00, 0x00,
>-  0x13, 0x40, 0xa0, 0xa0, 0x00, 0x00,
>-  0x25, 0x06, 0x28, 0x40, 0x00, 0x00,
>-  0x18, 0x19, 0x10, 0x60, 0x00, 0x00,
>-  0x0a, 0x24, 0x45, 0xc0, 0x00, 0x00,
>-  0x88, 0x8a, 0x12, 0xa0, 0x00, 0x00,
>-  0x06, 0x81, 0x45, 0x20, 0x00, 0x00,
>-  0xe0, 0x24, 0xa1, 0x00, 0x00, 0x00,
>-  0x84, 0x40, 0xd8, 0x20, 0x00, 0x00,
>-  0x44, 0x19, 0x16, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom27_22[132] = {
>-  0x51, 0x23, 0x16, 0x80, 0x00, 0x00,
>-  0x22, 0x11, 0xa9, 0x00, 0x00, 0x00,
>-  0x13, 0x40, 0xa0, 0xa0, 0x00, 0x00,
>-  0x25, 0x06, 0x28, 0x40, 0x00, 0x00,
>-  0x18, 0x19, 0x10, 0x60, 0x00, 0x00,
>-  0x0a, 0x24, 0x45, 0xc0, 0x00, 0x00,
>-  0x88, 0x8a, 0x12, 0xa0, 0x00, 0x00,
>-  0x06, 0x81, 0x45, 0x20, 0x00, 0x00,
>-  0xe0, 0x24, 0xa1, 0x00, 0x00, 0x00,
>-  0x84, 0x40, 0xd8, 0x20, 0x00, 0x00,
>-  0x44, 0x19, 0x16, 0x00, 0x00, 0x00,
>-  0xd1, 0x06, 0x06, 0xa0, 0x00, 0x00,
>-  0x44, 0x50, 0xea, 0x00, 0x00, 0x00,
>-  0x10, 0x9e, 0xa0, 0x40, 0x00, 0x00,
>-  0xa0, 0x50, 0x13, 0x00, 0x00, 0x00,
>-  0x4a, 0x08, 0x21, 0x40, 0x00, 0x00,
>-  0x40, 0x31, 0x04, 0xc0, 0x00, 0x00,
>-  0x80, 0x2a, 0x02, 0x20, 0x00, 0x00,
>-  0x0c, 0x90, 0x44, 0x20, 0x00, 0x00,
>-  0x05, 0x8b, 0x40, 0x00, 0x00, 0x00,
>-  0x62, 0x21, 0x18, 0x80, 0x00, 0x00,
>-  0xf5, 0x2d, 0x52, 0x40, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom27_23[138] = {
>-  0x51, 0x23, 0x16, 0x80, 0x00, 0x00,
>-  0x22, 0x11, 0xa9, 0x00, 0x00, 0x00,
>-  0x13, 0x40, 0xa0, 0xa0, 0x00, 0x00,
>-  0x25, 0x06, 0x28, 0x40, 0x00, 0x00,
>-  0x18, 0x19, 0x10, 0x60, 0x00, 0x00,
>-  0x0a, 0x24, 0x45, 0xc0, 0x00, 0x00,
>-  0x88, 0x8a, 0x12, 0xa0, 0x00, 0x00,
>-  0x06, 0x81, 0x45, 0x20, 0x00, 0x00,
>-  0xe0, 0x24, 0xa1, 0x00, 0x00, 0x00,
>-  0x84, 0x40, 0xd8, 0x20, 0x00, 0x00,
>-  0x44, 0x19, 0x16, 0x00, 0x00, 0x00,
>-  0x28, 0x2c, 0x08, 0x20, 0x00, 0x00,
>-  0x84, 0x52, 0x03, 0x40, 0x00, 0x00,
>-  0x60, 0x44, 0x81, 0x20, 0x00, 0x00,
>-  0x05, 0x49, 0x41, 0x40, 0x00, 0x00,
>-  0x02, 0x9a, 0x90, 0x80, 0x00, 0x00,
>-  0x01, 0x32, 0x0c, 0x40, 0x00, 0x00,
>-  0x48, 0x10, 0x49, 0x80, 0x00, 0x00,
>-  0x24, 0x82, 0x42, 0x20, 0x00, 0x00,
>-  0x94, 0x00, 0x22, 0x20, 0x00, 0x00,
>-  0x8a, 0x00, 0x74, 0x00, 0x00, 0x00,
>-  0x11, 0x85, 0x2c, 0x80, 0x00, 0x00,
>-  0x52, 0x20, 0x90, 0x60, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom27_24[144] = {
>-  0x28, 0x2c, 0x08, 0x20, 0x00, 0x00,
>-  0x84, 0x52, 0x03, 0x40, 0x00, 0x00,
>-  0x60, 0x44, 0x81, 0x20, 0x00, 0x00,
>-  0x05, 0x49, 0x41, 0x40, 0x00, 0x00,
>-  0x02, 0x9a, 0x90, 0x80, 0x00, 0x00,
>-  0x01, 0x32, 0x0c, 0x40, 0x00, 0x00,
>-  0x48, 0x10, 0x49, 0x80, 0x00, 0x00,
>-  0x24, 0x82, 0x42, 0x20, 0x00, 0x00,
>-  0x94, 0x00, 0x22, 0x20, 0x00, 0x00,
>-  0x8a, 0x00, 0x74, 0x00, 0x00, 0x00,
>-  0x11, 0x85, 0x2c, 0x80, 0x00, 0x00,
>-  0x52, 0x20, 0x90, 0x60, 0x00, 0x00,
>-  0x51, 0x23, 0x16, 0x80, 0x00, 0x00,
>-  0x22, 0x11, 0xa9, 0x00, 0x00, 0x00,
>-  0x13, 0x40, 0xa0, 0xa0, 0x00, 0x00,
>-  0x25, 0x06, 0x28, 0x40, 0x00, 0x00,
>-  0x18, 0x19, 0x10, 0x60, 0x00, 0x00,
>-  0x0a, 0x24, 0x45, 0xc0, 0x00, 0x00,
>-  0x88, 0x8a, 0x12, 0xa0, 0x00, 0x00,
>-  0x06, 0x81, 0x45, 0x20, 0x00, 0x00,
>-  0xe0, 0x24, 0xa1, 0x00, 0x00, 0x00,
>-  0x84, 0x40, 0xd8, 0x20, 0x00, 0x00,
>-  0x44, 0x19, 0x16, 0x00, 0x00, 0x00,
>-  0xa2, 0x85, 0xdb, 0xa0, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom27_25[150] = {
>-  0x28, 0x2c, 0x08, 0x20, 0x00, 0x00,
>-  0x84, 0x52, 0x03, 0x40, 0x00, 0x00,
>-  0x60, 0x44, 0x81, 0x20, 0x00, 0x00,
>-  0x05, 0x49, 0x41, 0x40, 0x00, 0x00,
>-  0x02, 0x9a, 0x90, 0x80, 0x00, 0x00,
>-  0x01, 0x32, 0x0c, 0x40, 0x00, 0x00,
>-  0x48, 0x10, 0x49, 0x80, 0x00, 0x00,
>-  0x24, 0x82, 0x42, 0x20, 0x00, 0x00,
>-  0x94, 0x00, 0x22, 0x20, 0x00, 0x00,
>-  0x8a, 0x00, 0x74, 0x00, 0x00, 0x00,
>-  0x11, 0x85, 0x2c, 0x80, 0x00, 0x00,
>-  0x52, 0x20, 0x90, 0x60, 0x00, 0x00,
>-  0x51, 0x23, 0x12, 0xa0, 0x00, 0x00,
>-  0x66, 0x41, 0xa3, 0x00, 0x00, 0x00,
>-  0x05, 0x4a, 0x40, 0x20, 0x00, 0x00,
>-  0x81, 0x20, 0x05, 0x60, 0x00, 0x00,
>-  0x94, 0x01, 0x40, 0x40, 0x00, 0x00,
>-  0x30, 0x84, 0x08, 0x40, 0x00, 0x00,
>-  0x21, 0x11, 0x18, 0x20, 0x00, 0x00,
>-  0x03, 0xc0, 0x34, 0x00, 0x00, 0x00,
>-  0xe8, 0x04, 0x00, 0xa0, 0x00, 0x00,
>-  0x0a, 0x11, 0x80, 0x80, 0x00, 0x00,
>-  0x80, 0x1c, 0x61, 0x00, 0x00, 0x00,
>-  0x04, 0x92, 0xa0, 0x00, 0x00, 0x00,
>-  0x08, 0xac, 0x06, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom27_26[156] = {
>-  0x51, 0x23, 0x12, 0xa0, 0x00, 0x00,
>-  0x66, 0x41, 0xa3, 0x00, 0x00, 0x00,
>-  0x05, 0x4a, 0x40, 0x20, 0x00, 0x00,
>-  0x81, 0x20, 0x05, 0x60, 0x00, 0x00,
>-  0x94, 0x01, 0x40, 0x40, 0x00, 0x00,
>-  0x30, 0x84, 0x08, 0x40, 0x00, 0x00,
>-  0x21, 0x11, 0x18, 0x20, 0x00, 0x00,
>-  0x03, 0xc0, 0x34, 0x00, 0x00, 0x00,
>-  0xe8, 0x04, 0x00, 0xa0, 0x00, 0x00,
>-  0x0a, 0x11, 0x80, 0x80, 0x00, 0x00,
>-  0x80, 0x1c, 0x61, 0x00, 0x00, 0x00,
>-  0x04, 0x92, 0xa0, 0x00, 0x00, 0x00,
>-  0x08, 0xac, 0x06, 0x00, 0x00, 0x00,
>-  0x28, 0x2c, 0x08, 0x20, 0x00, 0x00,
>-  0x84, 0x52, 0x03, 0x40, 0x00, 0x00,
>-  0x60, 0x44, 0x81, 0x20, 0x00, 0x00,
>-  0x05, 0x49, 0x41, 0x40, 0x00, 0x00,
>-  0x02, 0x9a, 0x90, 0x80, 0x00, 0x00,
>-  0x01, 0x32, 0x0c, 0x40, 0x00, 0x00,
>-  0x48, 0x10, 0x49, 0x80, 0x00, 0x00,
>-  0x24, 0x82, 0x42, 0x20, 0x00, 0x00,
>-  0x94, 0x00, 0x22, 0x20, 0x00, 0x00,
>-  0x8a, 0x00, 0x74, 0x00, 0x00, 0x00,
>-  0x11, 0x85, 0x2c, 0x80, 0x00, 0x00,
>-  0x52, 0x20, 0x90, 0x60, 0x00, 0x00,
>-  0xcd, 0x41, 0xa2, 0x40, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom27_27[162] = {
>-  0x51, 0x23, 0x12, 0xa0, 0x00, 0x00,
>-  0x66, 0x41, 0xa3, 0x00, 0x00, 0x00,
>-  0x05, 0x4a, 0x40, 0x20, 0x00, 0x00,
>-  0x81, 0x20, 0x05, 0x60, 0x00, 0x00,
>-  0x94, 0x01, 0x40, 0x40, 0x00, 0x00,
>-  0x30, 0x84, 0x08, 0x40, 0x00, 0x00,
>-  0x21, 0x11, 0x18, 0x20, 0x00, 0x00,
>-  0x03, 0xc0, 0x34, 0x00, 0x00, 0x00,
>-  0xe8, 0x04, 0x00, 0xa0, 0x00, 0x00,
>-  0x0a, 0x11, 0x80, 0x80, 0x00, 0x00,
>-  0x80, 0x1c, 0x61, 0x00, 0x00, 0x00,
>-  0x04, 0x92, 0xa0, 0x00, 0x00, 0x00,
>-  0x08, 0xac, 0x06, 0x00, 0x00, 0x00,
>-  0x51, 0x22, 0x02, 0xa0, 0x00, 0x00,
>-  0x66, 0x40, 0xaa, 0x00, 0x00, 0x00,
>-  0x05, 0x4e, 0x00, 0x20, 0x00, 0x00,
>-  0x81, 0x21, 0x40, 0x80, 0x00, 0x00,
>-  0x94, 0x00, 0x28, 0x60, 0x00, 0x00,
>-  0x30, 0x83, 0x24, 0x00, 0x00, 0x00,
>-  0x21, 0x14, 0x0c, 0x00, 0x00, 0x00,
>-  0x03, 0xc0, 0x84, 0xc0, 0x00, 0x00,
>-  0xe8, 0x04, 0x21, 0x00, 0x00, 0x00,
>-  0x0a, 0x10, 0x91, 0x80, 0x00, 0x00,
>-  0x80, 0x1b, 0x10, 0x00, 0x00, 0x00,
>-  0x04, 0x91, 0x43, 0x00, 0x00, 0x00,
>-  0x08, 0xa8, 0x70, 0x40, 0x00, 0x00,
>-  0x9c, 0xc0, 0x84, 0x20, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom27_3[18] = {
>-  0x99, 0xb5, 0x66, 0xc0, 0x00, 0x00,
>-  0x46, 0xda, 0xab, 0x60, 0x00, 0x00,
>-  0x37, 0x29, 0x3d, 0xa0, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom27_4[24] = {
>-  0x49, 0xb1, 0x66, 0xc0, 0x00, 0x00,
>-  0x26, 0xd4, 0x9b, 0x40, 0x00, 0x00,
>-  0x85, 0x68, 0xd5, 0xa0, 0x00, 0x00,
>-  0x52, 0x5a, 0x39, 0x60, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom27_5[30] = {
>-  0x51, 0x33, 0x26, 0xc0, 0x00, 0x00,
>-  0x66, 0x45, 0x2b, 0x40, 0x00, 0x00,
>-  0x0c, 0x6a, 0x95, 0xa0, 0x00, 0x00,
>-  0xa1, 0xc0, 0xed, 0x40, 0x00, 0x00,
>-  0x22, 0x9c, 0xe2, 0xa0, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom27_6[36] = {
>-  0xd1, 0x22, 0x52, 0xa0, 0x00, 0x00,
>-  0x46, 0xd4, 0xaa, 0x40, 0x00, 0x00,
>-  0x15, 0x48, 0xa5, 0xa0, 0x00, 0x00,
>-  0x21, 0x72, 0x8d, 0x40, 0x00, 0x00,
>-  0x28, 0xc9, 0x13, 0x60, 0x00, 0x00,
>-  0xaa, 0x24, 0x44, 0x60, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom27_7[42] = {
>-  0x59, 0x23, 0x12, 0xa0, 0x00, 0x00,
>-  0x26, 0x55, 0xc9, 0x00, 0x00, 0x00,
>-  0xb1, 0x40, 0xc5, 0xa0, 0x00, 0x00,
>-  0x2b, 0x0a, 0xa4, 0xc0, 0x00, 0x00,
>-  0x14, 0xc8, 0x33, 0x60, 0x00, 0x00,
>-  0xc8, 0x8c, 0x2a, 0xa0, 0x00, 0x00,
>-  0x84, 0xb5, 0x54, 0x40, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom27_8[48] = {
>-  0x80, 0xae, 0x00, 0xa0, 0x00, 0x00,
>-  0x30, 0x92, 0x0b, 0x00, 0x00, 0x00,
>-  0x16, 0x0c, 0x41, 0x80, 0x00, 0x00,
>-  0x03, 0x31, 0x05, 0x20, 0x00, 0x00,
>-  0x44, 0x60, 0x52, 0x40, 0x00, 0x00,
>-  0x08, 0x18, 0x24, 0xc0, 0x00, 0x00,
>-  0xd8, 0x04, 0xa2, 0x00, 0x00, 0x00,
>-  0xa1, 0x43, 0x90, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom27_9[54] = {
>-  0x59, 0x25, 0x12, 0xa0, 0x00, 0x00,
>-  0x66, 0x41, 0xa3, 0x00, 0x00, 0x00,
>-  0x14, 0x42, 0x51, 0x20, 0x00, 0x00,
>-  0x21, 0x49, 0x05, 0x40, 0x00, 0x00,
>-  0x02, 0xc8, 0x8c, 0x20, 0x00, 0x00,
>-  0x94, 0x12, 0x48, 0x40, 0x00, 0x00,
>-  0x80, 0xac, 0x30, 0x60, 0x00, 0x00,
>-  0x0a, 0x91, 0x06, 0xa0, 0x00, 0x00,
>-  0x40, 0x1c, 0x42, 0x40, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom28_1[6] = {
>-  0xff, 0xff, 0xff, 0xf0, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom28_10[60] = {
>-  0xc0, 0xd7, 0x03, 0x50, 0x00, 0x00,
>-  0x1d, 0x40, 0x75, 0x00, 0x00, 0x00,
>-  0xd4, 0x0b, 0x50, 0x20, 0x00, 0x00,
>-  0x02, 0x60, 0x09, 0x80, 0x00, 0x00,
>-  0x04, 0x28, 0x10, 0xa0, 0x00, 0x00,
>-  0x20, 0x98, 0x82, 0x60, 0x00, 0x00,
>-  0x40, 0x45, 0x01, 0x10, 0x00, 0x00,
>-  0x08, 0x84, 0x22, 0x10, 0x00, 0x00,
>-  0x68, 0x01, 0xa0, 0x00, 0x00, 0x00,
>-  0x23, 0x10, 0x8c, 0x40, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom28_11[66] = {
>-  0x62, 0xd1, 0x8b, 0x40, 0x00, 0x00,
>-  0x35, 0x20, 0xd4, 0x80, 0x00, 0x00,
>-  0x14, 0x14, 0x50, 0x50, 0x00, 0x00,
>-  0xc5, 0x0b, 0x14, 0x20, 0x00, 0x00,
>-  0x22, 0x0c, 0x88, 0x30, 0x00, 0x00,
>-  0x88, 0xba, 0x22, 0xe0, 0x00, 0x00,
>-  0x42, 0x55, 0x09, 0x50, 0x00, 0x00,
>-  0x28, 0xa4, 0xa2, 0x90, 0x00, 0x00,
>-  0x94, 0x22, 0x50, 0x80, 0x00, 0x00,
>-  0x1b, 0x04, 0x6c, 0x10, 0x00, 0x00,
>-  0x22, 0xc0, 0x8b, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom28_12[72] = {
>-  0x81, 0x06, 0x04, 0x10, 0x00, 0x00,
>-  0x40, 0x69, 0x01, 0xa0, 0x00, 0x00,
>-  0x90, 0x26, 0x40, 0x90, 0x00, 0x00,
>-  0x28, 0x28, 0xa0, 0xa0, 0x00, 0x00,
>-  0x52, 0x11, 0x48, 0x40, 0x00, 0x00,
>-  0x41, 0x89, 0x06, 0x20, 0x00, 0x00,
>-  0x09, 0x30, 0x24, 0xc0, 0x00, 0x00,
>-  0x48, 0x45, 0x21, 0x10, 0x00, 0x00,
>-  0x04, 0x44, 0x11, 0x10, 0x00, 0x00,
>-  0x0e, 0x80, 0x3a, 0x00, 0x00, 0x00,
>-  0xa5, 0x92, 0x96, 0x40, 0x00, 0x00,
>-  0x12, 0x0c, 0x48, 0x30, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom28_13[78] = {
>-  0x62, 0x55, 0x89, 0x50, 0x00, 0x00,
>-  0x34, 0x60, 0xd1, 0x80, 0x00, 0x00,
>-  0x48, 0x05, 0x20, 0x10, 0x00, 0x00,
>-  0x00, 0xac, 0x02, 0xb0, 0x00, 0x00,
>-  0x28, 0x08, 0xa0, 0x20, 0x00, 0x00,
>-  0x81, 0x0a, 0x04, 0x20, 0x00, 0x00,
>-  0x23, 0x04, 0x8c, 0x10, 0x00, 0x00,
>-  0x06, 0x80, 0x1a, 0x00, 0x00, 0x00,
>-  0x80, 0x16, 0x00, 0x50, 0x00, 0x00,
>-  0x30, 0x10, 0xc0, 0x40, 0x00, 0x00,
>-  0x8c, 0x22, 0x30, 0x80, 0x00, 0x00,
>-  0x54, 0x01, 0x50, 0x00, 0x00, 0x00,
>-  0x80, 0xc2, 0x03, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom28_14[84] = {
>-  0x40, 0x55, 0x01, 0x50, 0x00, 0x00,
>-  0x15, 0x40, 0x55, 0x00, 0x00, 0x00,
>-  0xc0, 0x07, 0x00, 0x10, 0x00, 0x00,
>-  0x28, 0x10, 0xa0, 0x40, 0x00, 0x00,
>-  0x05, 0x0c, 0x14, 0x30, 0x00, 0x00,
>-  0x64, 0x81, 0x92, 0x00, 0x00, 0x00,
>-  0x81, 0x82, 0x06, 0x00, 0x00, 0x00,
>-  0x10, 0x98, 0x42, 0x60, 0x00, 0x00,
>-  0x84, 0x22, 0x10, 0x80, 0x00, 0x00,
>-  0x12, 0x30, 0x48, 0xc0, 0x00, 0x00,
>-  0x62, 0x01, 0x88, 0x00, 0x00, 0x00,
>-  0x28, 0x60, 0xa1, 0x80, 0x00, 0x00,
>-  0x0e, 0x08, 0x38, 0x20, 0x00, 0x00,
>-  0x10, 0x84, 0x42, 0x10, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom28_15[90] = {
>-  0x62, 0x55, 0x89, 0x50, 0x00, 0x00,
>-  0xb9, 0x22, 0xe4, 0x80, 0x00, 0x00,
>-  0x18, 0xb4, 0x62, 0xd0, 0x00, 0x00,
>-  0x54, 0x99, 0x52, 0x60, 0x00, 0x00,
>-  0x06, 0x6c, 0x19, 0xb0, 0x00, 0x00,
>-  0x85, 0x56, 0x15, 0x50, 0x00, 0x00,
>-  0xaa, 0x8a, 0xaa, 0x20, 0x00, 0x00,
>-  0xc0, 0x17, 0x00, 0x50, 0x00, 0x00,
>-  0x41, 0x61, 0x05, 0x80, 0x00, 0x00,
>-  0x88, 0x32, 0x20, 0xc0, 0x00, 0x00,
>-  0x20, 0xa4, 0x82, 0x90, 0x00, 0x00,
>-  0x0a, 0x48, 0x29, 0x20, 0x00, 0x00,
>-  0x04, 0x98, 0x12, 0x60, 0x00, 0x00,
>-  0x94, 0x42, 0x51, 0x00, 0x00, 0x00,
>-  0x72, 0x01, 0xc8, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom28_16[96] = {
>-  0xc0, 0x17, 0x00, 0x50, 0x00, 0x00,
>-  0x41, 0x61, 0x05, 0x80, 0x00, 0x00,
>-  0x88, 0x32, 0x20, 0xc0, 0x00, 0x00,
>-  0x20, 0xa4, 0x82, 0x90, 0x00, 0x00,
>-  0x0a, 0x48, 0x29, 0x20, 0x00, 0x00,
>-  0x04, 0x98, 0x12, 0x60, 0x00, 0x00,
>-  0x94, 0x42, 0x51, 0x00, 0x00, 0x00,
>-  0x72, 0x01, 0xc8, 0x00, 0x00, 0x00,
>-  0x62, 0x55, 0x89, 0x50, 0x00, 0x00,
>-  0xb9, 0x22, 0xe4, 0x80, 0x00, 0x00,
>-  0x18, 0xb4, 0x62, 0xd0, 0x00, 0x00,
>-  0x54, 0x99, 0x52, 0x60, 0x00, 0x00,
>-  0x06, 0x6c, 0x19, 0xb0, 0x00, 0x00,
>-  0x85, 0x56, 0x15, 0x50, 0x00, 0x00,
>-  0xaa, 0x8a, 0xaa, 0x20, 0x00, 0x00,
>-  0xed, 0x76, 0x36, 0x50, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom28_17[102] = {
>-  0xc0, 0x17, 0x00, 0x50, 0x00, 0x00,
>-  0x41, 0x61, 0x05, 0x80, 0x00, 0x00,
>-  0x88, 0x32, 0x20, 0xc0, 0x00, 0x00,
>-  0x20, 0xa4, 0x82, 0x90, 0x00, 0x00,
>-  0x0a, 0x48, 0x29, 0x20, 0x00, 0x00,
>-  0x04, 0x98, 0x12, 0x60, 0x00, 0x00,
>-  0x94, 0x42, 0x51, 0x00, 0x00, 0x00,
>-  0x72, 0x01, 0xc8, 0x00, 0x00, 0x00,
>-  0xa2, 0x56, 0x89, 0x50, 0x00, 0x00,
>-  0x34, 0x60, 0xd1, 0x80, 0x00, 0x00,
>-  0x4a, 0x25, 0x28, 0x90, 0x00, 0x00,
>-  0x20, 0xa8, 0x82, 0xa0, 0x00, 0x00,
>-  0x11, 0x84, 0x46, 0x10, 0x00, 0x00,
>-  0x49, 0x09, 0x24, 0x20, 0x00, 0x00,
>-  0x86, 0x0e, 0x18, 0x30, 0x00, 0x00,
>-  0x20, 0xd4, 0x83, 0x50, 0x00, 0x00,
>-  0x88, 0x4a, 0x21, 0x20, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom28_18[108] = {
>-  0xa2, 0x56, 0x89, 0x50, 0x00, 0x00,
>-  0x34, 0x60, 0xd1, 0x80, 0x00, 0x00,
>-  0x4a, 0x25, 0x28, 0x90, 0x00, 0x00,
>-  0x20, 0xa8, 0x82, 0xa0, 0x00, 0x00,
>-  0x11, 0x84, 0x46, 0x10, 0x00, 0x00,
>-  0x49, 0x09, 0x24, 0x20, 0x00, 0x00,
>-  0x86, 0x0e, 0x18, 0x30, 0x00, 0x00,
>-  0x20, 0xd4, 0x83, 0x50, 0x00, 0x00,
>-  0x88, 0x4a, 0x21, 0x20, 0x00, 0x00,
>-  0xc0, 0x17, 0x00, 0x50, 0x00, 0x00,
>-  0x41, 0x61, 0x05, 0x80, 0x00, 0x00,
>-  0x88, 0x32, 0x20, 0xc0, 0x00, 0x00,
>-  0x20, 0xa4, 0x82, 0x90, 0x00, 0x00,
>-  0x0a, 0x48, 0x29, 0x20, 0x00, 0x00,
>-  0x04, 0x98, 0x12, 0x60, 0x00, 0x00,
>-  0x94, 0x42, 0x51, 0x00, 0x00, 0x00,
>-  0x72, 0x01, 0xc8, 0x00, 0x00, 0x00,
>-  0x6e, 0x9f, 0x98, 0x10, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom28_19[114] = {
>-  0xa2, 0x56, 0x89, 0x50, 0x00, 0x00,
>-  0x34, 0x60, 0xd1, 0x80, 0x00, 0x00,
>-  0x4a, 0x25, 0x28, 0x90, 0x00, 0x00,
>-  0x20, 0xa8, 0x82, 0xa0, 0x00, 0x00,
>-  0x11, 0x84, 0x46, 0x10, 0x00, 0x00,
>-  0x49, 0x09, 0x24, 0x20, 0x00, 0x00,
>-  0x86, 0x0e, 0x18, 0x30, 0x00, 0x00,
>-  0x20, 0xd4, 0x83, 0x50, 0x00, 0x00,
>-  0x88, 0x4a, 0x21, 0x20, 0x00, 0x00,
>-  0xc0, 0xd7, 0x03, 0x50, 0x00, 0x00,
>-  0x1d, 0x40, 0x75, 0x00, 0x00, 0x00,
>-  0xd4, 0x0b, 0x50, 0x20, 0x00, 0x00,
>-  0x02, 0x60, 0x09, 0x80, 0x00, 0x00,
>-  0x04, 0x28, 0x10, 0xa0, 0x00, 0x00,
>-  0x20, 0x98, 0x82, 0x60, 0x00, 0x00,
>-  0x40, 0x45, 0x01, 0x10, 0x00, 0x00,
>-  0x08, 0x84, 0x22, 0x10, 0x00, 0x00,
>-  0x68, 0x01, 0xa0, 0x00, 0x00, 0x00,
>-  0x23, 0x10, 0x8c, 0x40, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom28_2[12] = {
>-  0xec, 0xeb, 0xb3, 0xa0, 0x00, 0x00,
>-  0x3b, 0x9c, 0xee, 0x70, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom28_20[120] = {
>-  0xc0, 0xd7, 0x03, 0x50, 0x00, 0x00,
>-  0x1d, 0x40, 0x75, 0x00, 0x00, 0x00,
>-  0xd4, 0x0b, 0x50, 0x20, 0x00, 0x00,
>-  0x02, 0x60, 0x09, 0x80, 0x00, 0x00,
>-  0x04, 0x28, 0x10, 0xa0, 0x00, 0x00,
>-  0x20, 0x98, 0x82, 0x60, 0x00, 0x00,
>-  0x40, 0x45, 0x01, 0x10, 0x00, 0x00,
>-  0x08, 0x84, 0x22, 0x10, 0x00, 0x00,
>-  0x68, 0x01, 0xa0, 0x00, 0x00, 0x00,
>-  0x23, 0x10, 0x8c, 0x40, 0x00, 0x00,
>-  0xa2, 0x56, 0x89, 0x50, 0x00, 0x00,
>-  0x34, 0x60, 0xd1, 0x80, 0x00, 0x00,
>-  0x4a, 0x25, 0x28, 0x90, 0x00, 0x00,
>-  0x20, 0xa8, 0x82, 0xa0, 0x00, 0x00,
>-  0x11, 0x84, 0x46, 0x10, 0x00, 0x00,
>-  0x49, 0x09, 0x24, 0x20, 0x00, 0x00,
>-  0x86, 0x0e, 0x18, 0x30, 0x00, 0x00,
>-  0x20, 0xd4, 0x83, 0x50, 0x00, 0x00,
>-  0x88, 0x4a, 0x21, 0x20, 0x00, 0x00,
>-  0xea, 0x1b, 0x3a, 0x10, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom28_21[126] = {
>-  0xc0, 0xd7, 0x03, 0x50, 0x00, 0x00,
>-  0x1d, 0x40, 0x75, 0x00, 0x00, 0x00,
>-  0xd4, 0x0b, 0x50, 0x20, 0x00, 0x00,
>-  0x02, 0x60, 0x09, 0x80, 0x00, 0x00,
>-  0x04, 0x28, 0x10, 0xa0, 0x00, 0x00,
>-  0x20, 0x98, 0x82, 0x60, 0x00, 0x00,
>-  0x40, 0x45, 0x01, 0x10, 0x00, 0x00,
>-  0x08, 0x84, 0x22, 0x10, 0x00, 0x00,
>-  0x68, 0x01, 0xa0, 0x00, 0x00, 0x00,
>-  0x23, 0x10, 0x8c, 0x40, 0x00, 0x00,
>-  0x62, 0xd1, 0x8b, 0x40, 0x00, 0x00,
>-  0x35, 0x20, 0xd4, 0x80, 0x00, 0x00,
>-  0x14, 0x14, 0x50, 0x50, 0x00, 0x00,
>-  0xc5, 0x0b, 0x14, 0x20, 0x00, 0x00,
>-  0x22, 0x0c, 0x88, 0x30, 0x00, 0x00,
>-  0x88, 0xba, 0x22, 0xe0, 0x00, 0x00,
>-  0x42, 0x55, 0x09, 0x50, 0x00, 0x00,
>-  0x28, 0xa4, 0xa2, 0x90, 0x00, 0x00,
>-  0x94, 0x22, 0x50, 0x80, 0x00, 0x00,
>-  0x1b, 0x04, 0x6c, 0x10, 0x00, 0x00,
>-  0x22, 0xc0, 0x8b, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom28_22[132] = {
>-  0x62, 0xd1, 0x8b, 0x40, 0x00, 0x00,
>-  0x35, 0x20, 0xd4, 0x80, 0x00, 0x00,
>-  0x14, 0x14, 0x50, 0x50, 0x00, 0x00,
>-  0xc5, 0x0b, 0x14, 0x20, 0x00, 0x00,
>-  0x22, 0x0c, 0x88, 0x30, 0x00, 0x00,
>-  0x88, 0xba, 0x22, 0xe0, 0x00, 0x00,
>-  0x42, 0x55, 0x09, 0x50, 0x00, 0x00,
>-  0x28, 0xa4, 0xa2, 0x90, 0x00, 0x00,
>-  0x94, 0x22, 0x50, 0x80, 0x00, 0x00,
>-  0x1b, 0x04, 0x6c, 0x10, 0x00, 0x00,
>-  0x22, 0xc0, 0x8b, 0x00, 0x00, 0x00,
>-  0xc0, 0xd7, 0x03, 0x50, 0x00, 0x00,
>-  0x1d, 0x40, 0x75, 0x00, 0x00, 0x00,
>-  0xd4, 0x0b, 0x50, 0x20, 0x00, 0x00,
>-  0x02, 0x60, 0x09, 0x80, 0x00, 0x00,
>-  0x04, 0x28, 0x10, 0xa0, 0x00, 0x00,
>-  0x20, 0x98, 0x82, 0x60, 0x00, 0x00,
>-  0x40, 0x45, 0x01, 0x10, 0x00, 0x00,
>-  0x08, 0x84, 0x22, 0x10, 0x00, 0x00,
>-  0x68, 0x01, 0xa0, 0x00, 0x00, 0x00,
>-  0x23, 0x10, 0x8c, 0x40, 0x00, 0x00,
>-  0x45, 0x05, 0x10, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom28_23[138] = {
>-  0x62, 0xd1, 0x8b, 0x40, 0x00, 0x00,
>-  0x35, 0x20, 0xd4, 0x80, 0x00, 0x00,
>-  0x14, 0x14, 0x50, 0x50, 0x00, 0x00,
>-  0xc5, 0x0b, 0x14, 0x20, 0x00, 0x00,
>-  0x22, 0x0c, 0x88, 0x30, 0x00, 0x00,
>-  0x88, 0xba, 0x22, 0xe0, 0x00, 0x00,
>-  0x42, 0x55, 0x09, 0x50, 0x00, 0x00,
>-  0x28, 0xa4, 0xa2, 0x90, 0x00, 0x00,
>-  0x94, 0x22, 0x50, 0x80, 0x00, 0x00,
>-  0x1b, 0x04, 0x6c, 0x10, 0x00, 0x00,
>-  0x22, 0xc0, 0x8b, 0x00, 0x00, 0x00,
>-  0x81, 0x06, 0x04, 0x10, 0x00, 0x00,
>-  0x40, 0x69, 0x01, 0xa0, 0x00, 0x00,
>-  0x90, 0x26, 0x40, 0x90, 0x00, 0x00,
>-  0x28, 0x28, 0xa0, 0xa0, 0x00, 0x00,
>-  0x52, 0x11, 0x48, 0x40, 0x00, 0x00,
>-  0x41, 0x89, 0x06, 0x20, 0x00, 0x00,
>-  0x09, 0x30, 0x24, 0xc0, 0x00, 0x00,
>-  0x48, 0x45, 0x21, 0x10, 0x00, 0x00,
>-  0x04, 0x44, 0x11, 0x10, 0x00, 0x00,
>-  0x0e, 0x80, 0x3a, 0x00, 0x00, 0x00,
>-  0xa5, 0x92, 0x96, 0x40, 0x00, 0x00,
>-  0x12, 0x0c, 0x48, 0x30, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom28_24[144] = {
>-  0x81, 0x06, 0x04, 0x10, 0x00, 0x00,
>-  0x40, 0x69, 0x01, 0xa0, 0x00, 0x00,
>-  0x90, 0x26, 0x40, 0x90, 0x00, 0x00,
>-  0x28, 0x28, 0xa0, 0xa0, 0x00, 0x00,
>-  0x52, 0x11, 0x48, 0x40, 0x00, 0x00,
>-  0x41, 0x89, 0x06, 0x20, 0x00, 0x00,
>-  0x09, 0x30, 0x24, 0xc0, 0x00, 0x00,
>-  0x48, 0x45, 0x21, 0x10, 0x00, 0x00,
>-  0x04, 0x44, 0x11, 0x10, 0x00, 0x00,
>-  0x0e, 0x80, 0x3a, 0x00, 0x00, 0x00,
>-  0xa5, 0x92, 0x96, 0x40, 0x00, 0x00,
>-  0x12, 0x0c, 0x48, 0x30, 0x00, 0x00,
>-  0x62, 0xd1, 0x8b, 0x40, 0x00, 0x00,
>-  0x35, 0x20, 0xd4, 0x80, 0x00, 0x00,
>-  0x14, 0x14, 0x50, 0x50, 0x00, 0x00,
>-  0xc5, 0x0b, 0x14, 0x20, 0x00, 0x00,
>-  0x22, 0x0c, 0x88, 0x30, 0x00, 0x00,
>-  0x88, 0xba, 0x22, 0xe0, 0x00, 0x00,
>-  0x42, 0x55, 0x09, 0x50, 0x00, 0x00,
>-  0x28, 0xa4, 0xa2, 0x90, 0x00, 0x00,
>-  0x94, 0x22, 0x50, 0x80, 0x00, 0x00,
>-  0x1b, 0x04, 0x6c, 0x10, 0x00, 0x00,
>-  0x22, 0xc0, 0x8b, 0x00, 0x00, 0x00,
>-  0x6f, 0xd8, 0xee, 0xa0, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom28_25[150] = {
>-  0x81, 0x06, 0x04, 0x10, 0x00, 0x00,
>-  0x40, 0x69, 0x01, 0xa0, 0x00, 0x00,
>-  0x90, 0x26, 0x40, 0x90, 0x00, 0x00,
>-  0x28, 0x28, 0xa0, 0xa0, 0x00, 0x00,
>-  0x52, 0x11, 0x48, 0x40, 0x00, 0x00,
>-  0x41, 0x89, 0x06, 0x20, 0x00, 0x00,
>-  0x09, 0x30, 0x24, 0xc0, 0x00, 0x00,
>-  0x48, 0x45, 0x21, 0x10, 0x00, 0x00,
>-  0x04, 0x44, 0x11, 0x10, 0x00, 0x00,
>-  0x0e, 0x80, 0x3a, 0x00, 0x00, 0x00,
>-  0xa5, 0x92, 0x96, 0x40, 0x00, 0x00,
>-  0x12, 0x0c, 0x48, 0x30, 0x00, 0x00,
>-  0x62, 0x55, 0x89, 0x50, 0x00, 0x00,
>-  0x34, 0x60, 0xd1, 0x80, 0x00, 0x00,
>-  0x48, 0x05, 0x20, 0x10, 0x00, 0x00,
>-  0x00, 0xac, 0x02, 0xb0, 0x00, 0x00,
>-  0x28, 0x08, 0xa0, 0x20, 0x00, 0x00,
>-  0x81, 0x0a, 0x04, 0x20, 0x00, 0x00,
>-  0x23, 0x04, 0x8c, 0x10, 0x00, 0x00,
>-  0x06, 0x80, 0x1a, 0x00, 0x00, 0x00,
>-  0x80, 0x16, 0x00, 0x50, 0x00, 0x00,
>-  0x30, 0x10, 0xc0, 0x40, 0x00, 0x00,
>-  0x8c, 0x22, 0x30, 0x80, 0x00, 0x00,
>-  0x54, 0x01, 0x50, 0x00, 0x00, 0x00,
>-  0x80, 0xc2, 0x03, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom28_26[156] = {
>-  0x62, 0x55, 0x89, 0x50, 0x00, 0x00,
>-  0x34, 0x60, 0xd1, 0x80, 0x00, 0x00,
>-  0x48, 0x05, 0x20, 0x10, 0x00, 0x00,
>-  0x00, 0xac, 0x02, 0xb0, 0x00, 0x00,
>-  0x28, 0x08, 0xa0, 0x20, 0x00, 0x00,
>-  0x81, 0x0a, 0x04, 0x20, 0x00, 0x00,
>-  0x23, 0x04, 0x8c, 0x10, 0x00, 0x00,
>-  0x06, 0x80, 0x1a, 0x00, 0x00, 0x00,
>-  0x80, 0x16, 0x00, 0x50, 0x00, 0x00,
>-  0x30, 0x10, 0xc0, 0x40, 0x00, 0x00,
>-  0x8c, 0x22, 0x30, 0x80, 0x00, 0x00,
>-  0x54, 0x01, 0x50, 0x00, 0x00, 0x00,
>-  0x80, 0xc2, 0x03, 0x00, 0x00, 0x00,
>-  0x81, 0x06, 0x04, 0x10, 0x00, 0x00,
>-  0x40, 0x69, 0x01, 0xa0, 0x00, 0x00,
>-  0x90, 0x26, 0x40, 0x90, 0x00, 0x00,
>-  0x28, 0x28, 0xa0, 0xa0, 0x00, 0x00,
>-  0x52, 0x11, 0x48, 0x40, 0x00, 0x00,
>-  0x41, 0x89, 0x06, 0x20, 0x00, 0x00,
>-  0x09, 0x30, 0x24, 0xc0, 0x00, 0x00,
>-  0x48, 0x45, 0x21, 0x10, 0x00, 0x00,
>-  0x04, 0x44, 0x11, 0x10, 0x00, 0x00,
>-  0x0e, 0x80, 0x3a, 0x00, 0x00, 0x00,
>-  0xa5, 0x92, 0x96, 0x40, 0x00, 0x00,
>-  0x12, 0x0c, 0x48, 0x30, 0x00, 0x00,
>-  0xf1, 0x64, 0xbe, 0x40, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom28_27[162] = {
>-  0x62, 0x55, 0x89, 0x50, 0x00, 0x00,
>-  0x34, 0x60, 0xd1, 0x80, 0x00, 0x00,
>-  0x48, 0x05, 0x20, 0x10, 0x00, 0x00,
>-  0x00, 0xac, 0x02, 0xb0, 0x00, 0x00,
>-  0x28, 0x08, 0xa0, 0x20, 0x00, 0x00,
>-  0x81, 0x0a, 0x04, 0x20, 0x00, 0x00,
>-  0x23, 0x04, 0x8c, 0x10, 0x00, 0x00,
>-  0x06, 0x80, 0x1a, 0x00, 0x00, 0x00,
>-  0x80, 0x16, 0x00, 0x50, 0x00, 0x00,
>-  0x30, 0x10, 0xc0, 0x40, 0x00, 0x00,
>-  0x8c, 0x22, 0x30, 0x80, 0x00, 0x00,
>-  0x54, 0x01, 0x50, 0x00, 0x00, 0x00,
>-  0x80, 0xc2, 0x03, 0x00, 0x00, 0x00,
>-  0x40, 0x55, 0x01, 0x50, 0x00, 0x00,
>-  0x15, 0x40, 0x55, 0x00, 0x00, 0x00,
>-  0xc0, 0x07, 0x00, 0x10, 0x00, 0x00,
>-  0x28, 0x10, 0xa0, 0x40, 0x00, 0x00,
>-  0x05, 0x0c, 0x14, 0x30, 0x00, 0x00,
>-  0x64, 0x81, 0x92, 0x00, 0x00, 0x00,
>-  0x81, 0x82, 0x06, 0x00, 0x00, 0x00,
>-  0x10, 0x98, 0x42, 0x60, 0x00, 0x00,
>-  0x84, 0x22, 0x10, 0x80, 0x00, 0x00,
>-  0x12, 0x30, 0x48, 0xc0, 0x00, 0x00,
>-  0x62, 0x01, 0x88, 0x00, 0x00, 0x00,
>-  0x28, 0x60, 0xa1, 0x80, 0x00, 0x00,
>-  0x0e, 0x08, 0x38, 0x20, 0x00, 0x00,
>-  0x10, 0x84, 0x42, 0x10, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom28_28[168] = {
>-  0x40, 0x55, 0x01, 0x50, 0x00, 0x00,
>-  0x15, 0x40, 0x55, 0x00, 0x00, 0x00,
>-  0xc0, 0x07, 0x00, 0x10, 0x00, 0x00,
>-  0x28, 0x10, 0xa0, 0x40, 0x00, 0x00,
>-  0x05, 0x0c, 0x14, 0x30, 0x00, 0x00,
>-  0x64, 0x81, 0x92, 0x00, 0x00, 0x00,
>-  0x81, 0x82, 0x06, 0x00, 0x00, 0x00,
>-  0x10, 0x98, 0x42, 0x60, 0x00, 0x00,
>-  0x84, 0x22, 0x10, 0x80, 0x00, 0x00,
>-  0x12, 0x30, 0x48, 0xc0, 0x00, 0x00,
>-  0x62, 0x01, 0x88, 0x00, 0x00, 0x00,
>-  0x28, 0x60, 0xa1, 0x80, 0x00, 0x00,
>-  0x0e, 0x08, 0x38, 0x20, 0x00, 0x00,
>-  0x10, 0x84, 0x42, 0x10, 0x00, 0x00,
>-  0x62, 0x55, 0x89, 0x50, 0x00, 0x00,
>-  0x34, 0x60, 0xd1, 0x80, 0x00, 0x00,
>-  0x48, 0x05, 0x20, 0x10, 0x00, 0x00,
>-  0x00, 0xac, 0x02, 0xb0, 0x00, 0x00,
>-  0x28, 0x08, 0xa0, 0x20, 0x00, 0x00,
>-  0x81, 0x0a, 0x04, 0x20, 0x00, 0x00,
>-  0x23, 0x04, 0x8c, 0x10, 0x00, 0x00,
>-  0x06, 0x80, 0x1a, 0x00, 0x00, 0x00,
>-  0x80, 0x16, 0x00, 0x50, 0x00, 0x00,
>-  0x30, 0x10, 0xc0, 0x40, 0x00, 0x00,
>-  0x8c, 0x22, 0x30, 0x80, 0x00, 0x00,
>-  0x54, 0x01, 0x50, 0x00, 0x00, 0x00,
>-  0x80, 0xc2, 0x03, 0x00, 0x00, 0x00,
>-  0x36, 0x4f, 0x1f, 0xb0, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom28_3[18] = {
>-  0xac, 0xda, 0xb3, 0x60, 0x00, 0x00,
>-  0x55, 0x6d, 0x55, 0xb0, 0x00, 0x00,
>-  0x27, 0xb4, 0x9e, 0xd0, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom28_4[24] = {
>-  0x2c, 0xd8, 0xb3, 0x60, 0x00, 0x00,
>-  0x93, 0x6a, 0x4d, 0xa0, 0x00, 0x00,
>-  0x1a, 0xb4, 0x6a, 0xd0, 0x00, 0x00,
>-  0x47, 0x2d, 0x1c, 0xb0, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom28_5[30] = {
>-  0x64, 0xd9, 0x93, 0x60, 0x00, 0x00,
>-  0xa5, 0x6a, 0x95, 0xa0, 0x00, 0x00,
>-  0x52, 0xb5, 0x4a, 0xd0, 0x00, 0x00,
>-  0x1d, 0xa8, 0x76, 0xa0, 0x00, 0x00,
>-  0x9c, 0x56, 0x71, 0x50, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom28_6[36] = {
>-  0x4a, 0x55, 0x29, 0x50, 0x00, 0x00,
>-  0x95, 0x4a, 0x55, 0x20, 0x00, 0x00,
>-  0x14, 0xb4, 0x52, 0xd0, 0x00, 0x00,
>-  0x51, 0xa9, 0x46, 0xa0, 0x00, 0x00,
>-  0x22, 0x6c, 0x89, 0xb0, 0x00, 0x00,
>-  0x88, 0x8e, 0x22, 0x30, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom28_7[42] = {
>-  0x62, 0x55, 0x89, 0x50, 0x00, 0x00,
>-  0xb9, 0x22, 0xe4, 0x80, 0x00, 0x00,
>-  0x18, 0xb4, 0x62, 0xd0, 0x00, 0x00,
>-  0x54, 0x99, 0x52, 0x60, 0x00, 0x00,
>-  0x06, 0x6c, 0x19, 0xb0, 0x00, 0x00,
>-  0x85, 0x56, 0x15, 0x50, 0x00, 0x00,
>-  0xaa, 0x8a, 0xaa, 0x20, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom28_8[48] = {
>-  0xc0, 0x17, 0x00, 0x50, 0x00, 0x00,
>-  0x41, 0x61, 0x05, 0x80, 0x00, 0x00,
>-  0x88, 0x32, 0x20, 0xc0, 0x00, 0x00,
>-  0x20, 0xa4, 0x82, 0x90, 0x00, 0x00,
>-  0x0a, 0x48, 0x29, 0x20, 0x00, 0x00,
>-  0x04, 0x98, 0x12, 0x60, 0x00, 0x00,
>-  0x94, 0x42, 0x51, 0x00, 0x00, 0x00,
>-  0x72, 0x01, 0xc8, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom28_9[54] = {
>-  0xa2, 0x56, 0x89, 0x50, 0x00, 0x00,
>-  0x34, 0x60, 0xd1, 0x80, 0x00, 0x00,
>-  0x4a, 0x25, 0x28, 0x90, 0x00, 0x00,
>-  0x20, 0xa8, 0x82, 0xa0, 0x00, 0x00,
>-  0x11, 0x84, 0x46, 0x10, 0x00, 0x00,
>-  0x49, 0x09, 0x24, 0x20, 0x00, 0x00,
>-  0x86, 0x0e, 0x18, 0x30, 0x00, 0x00,
>-  0x20, 0xd4, 0x83, 0x50, 0x00, 0x00,
>-  0x88, 0x4a, 0x21, 0x20, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom29_1[6] = {
>-  0xff, 0xff, 0xff, 0xf8, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom29_10[60] = {
>-  0xc0, 0xd7, 0x02, 0x80, 0x00, 0x00,
>-  0x1d, 0x40, 0x55, 0x58, 0x00, 0x00,
>-  0xd4, 0x09, 0xd1, 0x00, 0x00, 0x00,
>-  0x02, 0x60, 0x02, 0x70, 0x00, 0x00,
>-  0x04, 0x28, 0x04, 0xb0, 0x00, 0x00,
>-  0x20, 0x99, 0x12, 0x48, 0x00, 0x00,
>-  0x40, 0x46, 0x21, 0x40, 0x00, 0x00,
>-  0x08, 0x84, 0x82, 0x90, 0x00, 0x00,
>-  0x68, 0x02, 0xa8, 0x10, 0x00, 0x00,
>-  0x23, 0x10, 0x09, 0x88, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom29_11[66] = {
>-  0x62, 0xd1, 0x88, 0x88, 0x00, 0x00,
>-  0x35, 0x23, 0xc4, 0x40, 0x00, 0x00,
>-  0x14, 0x14, 0x40, 0x38, 0x00, 0x00,
>-  0xc5, 0x08, 0x42, 0xc0, 0x00, 0x00,
>-  0x22, 0x0c, 0x90, 0x90, 0x00, 0x00,
>-  0x88, 0xb8, 0x04, 0x48, 0x00, 0x00,
>-  0x42, 0x54, 0x03, 0x10, 0x00, 0x00,
>-  0x28, 0xa4, 0x12, 0x88, 0x00, 0x00,
>-  0x94, 0x20, 0x09, 0x60, 0x00, 0x00,
>-  0x1b, 0x04, 0xac, 0x00, 0x00, 0x00,
>-  0x22, 0xc2, 0x61, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom29_12[72] = {
>-  0x81, 0x06, 0x22, 0x40, 0x00, 0x00,
>-  0x40, 0x69, 0x01, 0x50, 0x00, 0x00,
>-  0x90, 0x26, 0x09, 0x88, 0x00, 0x00,
>-  0x28, 0x28, 0x86, 0x90, 0x00, 0x00,
>-  0x52, 0x10, 0x41, 0x90, 0x00, 0x00,
>-  0x41, 0x89, 0x10, 0x28, 0x00, 0x00,
>-  0x09, 0x30, 0x43, 0x20, 0x00, 0x00,
>-  0x48, 0x45, 0x34, 0xa8, 0x00, 0x00,
>-  0x04, 0x44, 0xe0, 0x08, 0x00, 0x00,
>-  0x0e, 0x80, 0x5d, 0x20, 0x00, 0x00,
>-  0xa5, 0x92, 0x42, 0x10, 0x00, 0x00,
>-  0x12, 0x0d, 0xc8, 0x50, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom29_13[78] = {
>-  0x62, 0x55, 0x8a, 0x88, 0x00, 0x00,
>-  0x34, 0x60, 0xd1, 0x10, 0x00, 0x00,
>-  0x48, 0x05, 0x01, 0x28, 0x00, 0x00,
>-  0x00, 0xaf, 0x10, 0x10, 0x00, 0x00,
>-  0x28, 0x08, 0x21, 0x80, 0x00, 0x00,
>-  0x81, 0x0a, 0x50, 0x48, 0x00, 0x00,
>-  0x23, 0x06, 0x23, 0x00, 0x00, 0x00,
>-  0x06, 0x80, 0x84, 0xc8, 0x00, 0x00,
>-  0x80, 0x17, 0x05, 0x00, 0x00, 0x00,
>-  0x30, 0x10, 0x41, 0xa0, 0x00, 0x00,
>-  0x8c, 0x20, 0x1a, 0x40, 0x00, 0x00,
>-  0x54, 0x01, 0x64, 0x00, 0x00, 0x00,
>-  0x80, 0xc0, 0x28, 0x30, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom29_14[84] = {
>-  0x40, 0x55, 0x02, 0x08, 0x00, 0x00,
>-  0x15, 0x40, 0x55, 0x50, 0x00, 0x00,
>-  0xc0, 0x06, 0x20, 0x48, 0x00, 0x00,
>-  0x28, 0x13, 0x00, 0x40, 0x00, 0x00,
>-  0x05, 0x0e, 0x02, 0x80, 0x00, 0x00,
>-  0x64, 0x80, 0x04, 0x88, 0x00, 0x00,
>-  0x81, 0x81, 0x00, 0xb0, 0x00, 0x00,
>-  0x10, 0x98, 0x88, 0x08, 0x00, 0x00,
>-  0x84, 0x22, 0x40, 0x10, 0x00, 0x00,
>-  0x12, 0x30, 0x49, 0x00, 0x00, 0x00,
>-  0x62, 0x01, 0x74, 0x00, 0x00, 0x00,
>-  0x28, 0x60, 0x81, 0x50, 0x00, 0x00,
>-  0x0e, 0x0a, 0x18, 0x20, 0x00, 0x00,
>-  0x10, 0x84, 0xa2, 0x20, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom29_15[90] = {
>-  0x62, 0x55, 0x8a, 0x88, 0x00, 0x00,
>-  0xb9, 0x22, 0xc4, 0x50, 0x00, 0x00,
>-  0x18, 0xb4, 0x61, 0xa8, 0x00, 0x00,
>-  0x54, 0x99, 0x13, 0x50, 0x00, 0x00,
>-  0x06, 0x6c, 0x4d, 0x90, 0x00, 0x00,
>-  0x85, 0x55, 0x24, 0x68, 0x00, 0x00,
>-  0xaa, 0x8a, 0x1a, 0x30, 0x00, 0x00,
>-  0xc0, 0x16, 0x40, 0x88, 0x00, 0x00,
>-  0x41, 0x60, 0x25, 0x40, 0x00, 0x00,
>-  0x88, 0x30, 0x01, 0xa8, 0x00, 0x00,
>-  0x20, 0xa4, 0x80, 0xd0, 0x00, 0x00,
>-  0x0a, 0x48, 0x51, 0x10, 0x00, 0x00,
>-  0x04, 0x9b, 0x08, 0x40, 0x00, 0x00,
>-  0x94, 0x40, 0x03, 0x18, 0x00, 0x00,
>-  0x72, 0x01, 0x96, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom29_16[96] = {
>-  0xc0, 0x16, 0x40, 0x88, 0x00, 0x00,
>-  0x41, 0x60, 0x25, 0x40, 0x00, 0x00,
>-  0x88, 0x30, 0x01, 0xa8, 0x00, 0x00,
>-  0x20, 0xa4, 0x80, 0xd0, 0x00, 0x00,
>-  0x0a, 0x48, 0x51, 0x10, 0x00, 0x00,
>-  0x04, 0x9b, 0x08, 0x40, 0x00, 0x00,
>-  0x94, 0x40, 0x03, 0x18, 0x00, 0x00,
>-  0x72, 0x01, 0x96, 0x00, 0x00, 0x00,
>-  0x62, 0x55, 0x8a, 0x88, 0x00, 0x00,
>-  0xb9, 0x22, 0xc4, 0x50, 0x00, 0x00,
>-  0x18, 0xb4, 0x61, 0xa8, 0x00, 0x00,
>-  0x54, 0x99, 0x13, 0x50, 0x00, 0x00,
>-  0x06, 0x6c, 0x4d, 0x90, 0x00, 0x00,
>-  0x85, 0x55, 0x24, 0x68, 0x00, 0x00,
>-  0xaa, 0x8a, 0x1a, 0x30, 0x00, 0x00,
>-  0x0d, 0x2c, 0xf2, 0x20, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom29_17[102] = {
>-  0xc0, 0x16, 0x40, 0x88, 0x00, 0x00,
>-  0x41, 0x60, 0x25, 0x40, 0x00, 0x00,
>-  0x88, 0x30, 0x01, 0xa8, 0x00, 0x00,
>-  0x20, 0xa4, 0x80, 0xd0, 0x00, 0x00,
>-  0x0a, 0x48, 0x51, 0x10, 0x00, 0x00,
>-  0x04, 0x9b, 0x08, 0x40, 0x00, 0x00,
>-  0x94, 0x40, 0x03, 0x18, 0x00, 0x00,
>-  0x72, 0x01, 0x96, 0x00, 0x00, 0x00,
>-  0xa2, 0x55, 0x88, 0x88, 0x00, 0x00,
>-  0x34, 0x60, 0x91, 0x10, 0x00, 0x00,
>-  0x4a, 0x27, 0x01, 0x40, 0x00, 0x00,
>-  0x20, 0xa8, 0x0c, 0x30, 0x00, 0x00,
>-  0x11, 0x84, 0x58, 0xa0, 0x00, 0x00,
>-  0x49, 0x0a, 0x24, 0x00, 0x00, 0x00,
>-  0x86, 0x0e, 0x0a, 0x40, 0x00, 0x00,
>-  0x20, 0xd4, 0x22, 0x90, 0x00, 0x00,
>-  0x88, 0x4a, 0x41, 0x20, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom29_18[108] = {
>-  0xa2, 0x55, 0x88, 0x88, 0x00, 0x00,
>-  0x34, 0x60, 0x91, 0x10, 0x00, 0x00,
>-  0x4a, 0x27, 0x01, 0x40, 0x00, 0x00,
>-  0x20, 0xa8, 0x0c, 0x30, 0x00, 0x00,
>-  0x11, 0x84, 0x58, 0xa0, 0x00, 0x00,
>-  0x49, 0x0a, 0x24, 0x00, 0x00, 0x00,
>-  0x86, 0x0e, 0x0a, 0x40, 0x00, 0x00,
>-  0x20, 0xd4, 0x22, 0x90, 0x00, 0x00,
>-  0x88, 0x4a, 0x41, 0x20, 0x00, 0x00,
>-  0xc0, 0x16, 0x40, 0x88, 0x00, 0x00,
>-  0x41, 0x60, 0x25, 0x40, 0x00, 0x00,
>-  0x88, 0x30, 0x01, 0xa8, 0x00, 0x00,
>-  0x20, 0xa4, 0x80, 0xd0, 0x00, 0x00,
>-  0x0a, 0x48, 0x51, 0x10, 0x00, 0x00,
>-  0x04, 0x9b, 0x08, 0x40, 0x00, 0x00,
>-  0x94, 0x40, 0x03, 0x18, 0x00, 0x00,
>-  0x72, 0x01, 0x96, 0x00, 0x00, 0x00,
>-  0x71, 0x36, 0xf2, 0xb0, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom29_19[114] = {
>-  0xa2, 0x55, 0x88, 0x88, 0x00, 0x00,
>-  0x34, 0x60, 0x91, 0x10, 0x00, 0x00,
>-  0x4a, 0x27, 0x01, 0x40, 0x00, 0x00,
>-  0x20, 0xa8, 0x0c, 0x30, 0x00, 0x00,
>-  0x11, 0x84, 0x58, 0xa0, 0x00, 0x00,
>-  0x49, 0x0a, 0x24, 0x00, 0x00, 0x00,
>-  0x86, 0x0e, 0x0a, 0x40, 0x00, 0x00,
>-  0x20, 0xd4, 0x22, 0x90, 0x00, 0x00,
>-  0x88, 0x4a, 0x41, 0x20, 0x00, 0x00,
>-  0xc0, 0xd7, 0x02, 0x80, 0x00, 0x00,
>-  0x1d, 0x40, 0x55, 0x58, 0x00, 0x00,
>-  0xd4, 0x09, 0xd1, 0x00, 0x00, 0x00,
>-  0x02, 0x60, 0x02, 0x70, 0x00, 0x00,
>-  0x04, 0x28, 0x04, 0xb0, 0x00, 0x00,
>-  0x20, 0x99, 0x12, 0x48, 0x00, 0x00,
>-  0x40, 0x46, 0x21, 0x40, 0x00, 0x00,
>-  0x08, 0x84, 0x82, 0x90, 0x00, 0x00,
>-  0x68, 0x02, 0xa8, 0x10, 0x00, 0x00,
>-  0x23, 0x10, 0x09, 0x88, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom29_2[12] = {
>-  0xec, 0xeb, 0xb3, 0xa8, 0x00, 0x00,
>-  0x3b, 0x9e, 0xee, 0x70, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom29_20[120] = {
>-  0xc0, 0xd7, 0x02, 0x80, 0x00, 0x00,
>-  0x1d, 0x40, 0x55, 0x58, 0x00, 0x00,
>-  0xd4, 0x09, 0xd1, 0x00, 0x00, 0x00,
>-  0x02, 0x60, 0x02, 0x70, 0x00, 0x00,
>-  0x04, 0x28, 0x04, 0xb0, 0x00, 0x00,
>-  0x20, 0x99, 0x12, 0x48, 0x00, 0x00,
>-  0x40, 0x46, 0x21, 0x40, 0x00, 0x00,
>-  0x08, 0x84, 0x82, 0x90, 0x00, 0x00,
>-  0x68, 0x02, 0xa8, 0x10, 0x00, 0x00,
>-  0x23, 0x10, 0x09, 0x88, 0x00, 0x00,
>-  0xa2, 0x55, 0x88, 0x88, 0x00, 0x00,
>-  0x34, 0x60, 0x91, 0x10, 0x00, 0x00,
>-  0x4a, 0x27, 0x01, 0x40, 0x00, 0x00,
>-  0x20, 0xa8, 0x0c, 0x30, 0x00, 0x00,
>-  0x11, 0x84, 0x58, 0xa0, 0x00, 0x00,
>-  0x49, 0x0a, 0x24, 0x00, 0x00, 0x00,
>-  0x86, 0x0e, 0x0a, 0x40, 0x00, 0x00,
>-  0x20, 0xd4, 0x22, 0x90, 0x00, 0x00,
>-  0x88, 0x4a, 0x41, 0x20, 0x00, 0x00,
>-  0xe7, 0xec, 0xdc, 0xb0, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom29_21[126] = {
>-  0xc0, 0xd7, 0x02, 0x80, 0x00, 0x00,
>-  0x1d, 0x40, 0x55, 0x58, 0x00, 0x00,
>-  0xd4, 0x09, 0xd1, 0x00, 0x00, 0x00,
>-  0x02, 0x60, 0x02, 0x70, 0x00, 0x00,
>-  0x04, 0x28, 0x04, 0xb0, 0x00, 0x00,
>-  0x20, 0x99, 0x12, 0x48, 0x00, 0x00,
>-  0x40, 0x46, 0x21, 0x40, 0x00, 0x00,
>-  0x08, 0x84, 0x82, 0x90, 0x00, 0x00,
>-  0x68, 0x02, 0xa8, 0x10, 0x00, 0x00,
>-  0x23, 0x10, 0x09, 0x88, 0x00, 0x00,
>-  0x62, 0xd1, 0x88, 0x88, 0x00, 0x00,
>-  0x35, 0x23, 0xc4, 0x40, 0x00, 0x00,
>-  0x14, 0x14, 0x40, 0x38, 0x00, 0x00,
>-  0xc5, 0x08, 0x42, 0xc0, 0x00, 0x00,
>-  0x22, 0x0c, 0x90, 0x90, 0x00, 0x00,
>-  0x88, 0xb8, 0x04, 0x48, 0x00, 0x00,
>-  0x42, 0x54, 0x03, 0x10, 0x00, 0x00,
>-  0x28, 0xa4, 0x12, 0x88, 0x00, 0x00,
>-  0x94, 0x20, 0x09, 0x60, 0x00, 0x00,
>-  0x1b, 0x04, 0xac, 0x00, 0x00, 0x00,
>-  0x22, 0xc2, 0x61, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom29_22[132] = {
>-  0x62, 0xd1, 0x88, 0x88, 0x00, 0x00,
>-  0x35, 0x23, 0xc4, 0x40, 0x00, 0x00,
>-  0x14, 0x14, 0x40, 0x38, 0x00, 0x00,
>-  0xc5, 0x08, 0x42, 0xc0, 0x00, 0x00,
>-  0x22, 0x0c, 0x90, 0x90, 0x00, 0x00,
>-  0x88, 0xb8, 0x04, 0x48, 0x00, 0x00,
>-  0x42, 0x54, 0x03, 0x10, 0x00, 0x00,
>-  0x28, 0xa4, 0x12, 0x88, 0x00, 0x00,
>-  0x94, 0x20, 0x09, 0x60, 0x00, 0x00,
>-  0x1b, 0x04, 0xac, 0x00, 0x00, 0x00,
>-  0x22, 0xc2, 0x61, 0x00, 0x00, 0x00,
>-  0xc0, 0xd7, 0x02, 0x80, 0x00, 0x00,
>-  0x1d, 0x40, 0x55, 0x58, 0x00, 0x00,
>-  0xd4, 0x09, 0xd1, 0x00, 0x00, 0x00,
>-  0x02, 0x60, 0x02, 0x70, 0x00, 0x00,
>-  0x04, 0x28, 0x04, 0xb0, 0x00, 0x00,
>-  0x20, 0x99, 0x12, 0x48, 0x00, 0x00,
>-  0x40, 0x46, 0x21, 0x40, 0x00, 0x00,
>-  0x08, 0x84, 0x82, 0x90, 0x00, 0x00,
>-  0x68, 0x02, 0xa8, 0x10, 0x00, 0x00,
>-  0x23, 0x10, 0x09, 0x88, 0x00, 0x00,
>-  0x1c, 0x90, 0xa9, 0xa0, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom29_23[138] = {
>-  0x62, 0xd1, 0x88, 0x88, 0x00, 0x00,
>-  0x35, 0x23, 0xc4, 0x40, 0x00, 0x00,
>-  0x14, 0x14, 0x40, 0x38, 0x00, 0x00,
>-  0xc5, 0x08, 0x42, 0xc0, 0x00, 0x00,
>-  0x22, 0x0c, 0x90, 0x90, 0x00, 0x00,
>-  0x88, 0xb8, 0x04, 0x48, 0x00, 0x00,
>-  0x42, 0x54, 0x03, 0x10, 0x00, 0x00,
>-  0x28, 0xa4, 0x12, 0x88, 0x00, 0x00,
>-  0x94, 0x20, 0x09, 0x60, 0x00, 0x00,
>-  0x1b, 0x04, 0xac, 0x00, 0x00, 0x00,
>-  0x22, 0xc2, 0x61, 0x00, 0x00, 0x00,
>-  0x81, 0x06, 0x22, 0x40, 0x00, 0x00,
>-  0x40, 0x69, 0x01, 0x50, 0x00, 0x00,
>-  0x90, 0x26, 0x09, 0x88, 0x00, 0x00,
>-  0x28, 0x28, 0x86, 0x90, 0x00, 0x00,
>-  0x52, 0x10, 0x41, 0x90, 0x00, 0x00,
>-  0x41, 0x89, 0x10, 0x28, 0x00, 0x00,
>-  0x09, 0x30, 0x43, 0x20, 0x00, 0x00,
>-  0x48, 0x45, 0x34, 0xa8, 0x00, 0x00,
>-  0x04, 0x44, 0xe0, 0x08, 0x00, 0x00,
>-  0x0e, 0x80, 0x5d, 0x20, 0x00, 0x00,
>-  0xa5, 0x92, 0x42, 0x10, 0x00, 0x00,
>-  0x12, 0x0d, 0xc8, 0x50, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom29_24[144] = {
>-  0x81, 0x06, 0x22, 0x40, 0x00, 0x00,
>-  0x40, 0x69, 0x01, 0x50, 0x00, 0x00,
>-  0x90, 0x26, 0x09, 0x88, 0x00, 0x00,
>-  0x28, 0x28, 0x86, 0x90, 0x00, 0x00,
>-  0x52, 0x10, 0x41, 0x90, 0x00, 0x00,
>-  0x41, 0x89, 0x10, 0x28, 0x00, 0x00,
>-  0x09, 0x30, 0x43, 0x20, 0x00, 0x00,
>-  0x48, 0x45, 0x34, 0xa8, 0x00, 0x00,
>-  0x04, 0x44, 0xe0, 0x08, 0x00, 0x00,
>-  0x0e, 0x80, 0x5d, 0x20, 0x00, 0x00,
>-  0xa5, 0x92, 0x42, 0x10, 0x00, 0x00,
>-  0x12, 0x0d, 0xc8, 0x50, 0x00, 0x00,
>-  0x62, 0xd1, 0x88, 0x88, 0x00, 0x00,
>-  0x35, 0x23, 0xc4, 0x40, 0x00, 0x00,
>-  0x14, 0x14, 0x40, 0x38, 0x00, 0x00,
>-  0xc5, 0x08, 0x42, 0xc0, 0x00, 0x00,
>-  0x22, 0x0c, 0x90, 0x90, 0x00, 0x00,
>-  0x88, 0xb8, 0x04, 0x48, 0x00, 0x00,
>-  0x42, 0x54, 0x03, 0x10, 0x00, 0x00,
>-  0x28, 0xa4, 0x12, 0x88, 0x00, 0x00,
>-  0x94, 0x20, 0x09, 0x60, 0x00, 0x00,
>-  0x1b, 0x04, 0xac, 0x00, 0x00, 0x00,
>-  0x22, 0xc2, 0x61, 0x00, 0x00, 0x00,
>-  0xbd, 0x86, 0x97, 0xc0, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom29_25[150] = {
>-  0x81, 0x06, 0x22, 0x40, 0x00, 0x00,
>-  0x40, 0x69, 0x01, 0x50, 0x00, 0x00,
>-  0x90, 0x26, 0x09, 0x88, 0x00, 0x00,
>-  0x28, 0x28, 0x86, 0x90, 0x00, 0x00,
>-  0x52, 0x10, 0x41, 0x90, 0x00, 0x00,
>-  0x41, 0x89, 0x10, 0x28, 0x00, 0x00,
>-  0x09, 0x30, 0x43, 0x20, 0x00, 0x00,
>-  0x48, 0x45, 0x34, 0xa8, 0x00, 0x00,
>-  0x04, 0x44, 0xe0, 0x08, 0x00, 0x00,
>-  0x0e, 0x80, 0x5d, 0x20, 0x00, 0x00,
>-  0xa5, 0x92, 0x42, 0x10, 0x00, 0x00,
>-  0x12, 0x0d, 0xc8, 0x50, 0x00, 0x00,
>-  0x62, 0x55, 0x8a, 0x88, 0x00, 0x00,
>-  0x34, 0x60, 0xd1, 0x10, 0x00, 0x00,
>-  0x48, 0x05, 0x01, 0x28, 0x00, 0x00,
>-  0x00, 0xaf, 0x10, 0x10, 0x00, 0x00,
>-  0x28, 0x08, 0x21, 0x80, 0x00, 0x00,
>-  0x81, 0x0a, 0x50, 0x48, 0x00, 0x00,
>-  0x23, 0x06, 0x23, 0x00, 0x00, 0x00,
>-  0x06, 0x80, 0x84, 0xc8, 0x00, 0x00,
>-  0x80, 0x17, 0x05, 0x00, 0x00, 0x00,
>-  0x30, 0x10, 0x41, 0xa0, 0x00, 0x00,
>-  0x8c, 0x20, 0x1a, 0x40, 0x00, 0x00,
>-  0x54, 0x01, 0x64, 0x00, 0x00, 0x00,
>-  0x80, 0xc0, 0x28, 0x30, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom29_26[156] = {
>-  0x62, 0x55, 0x8a, 0x88, 0x00, 0x00,
>-  0x34, 0x60, 0xd1, 0x10, 0x00, 0x00,
>-  0x48, 0x05, 0x01, 0x28, 0x00, 0x00,
>-  0x00, 0xaf, 0x10, 0x10, 0x00, 0x00,
>-  0x28, 0x08, 0x21, 0x80, 0x00, 0x00,
>-  0x81, 0x0a, 0x50, 0x48, 0x00, 0x00,
>-  0x23, 0x06, 0x23, 0x00, 0x00, 0x00,
>-  0x06, 0x80, 0x84, 0xc8, 0x00, 0x00,
>-  0x80, 0x17, 0x05, 0x00, 0x00, 0x00,
>-  0x30, 0x10, 0x41, 0xa0, 0x00, 0x00,
>-  0x8c, 0x20, 0x1a, 0x40, 0x00, 0x00,
>-  0x54, 0x01, 0x64, 0x00, 0x00, 0x00,
>-  0x80, 0xc0, 0x28, 0x30, 0x00, 0x00,
>-  0x81, 0x06, 0x22, 0x40, 0x00, 0x00,
>-  0x40, 0x69, 0x01, 0x50, 0x00, 0x00,
>-  0x90, 0x26, 0x09, 0x88, 0x00, 0x00,
>-  0x28, 0x28, 0x86, 0x90, 0x00, 0x00,
>-  0x52, 0x10, 0x41, 0x90, 0x00, 0x00,
>-  0x41, 0x89, 0x10, 0x28, 0x00, 0x00,
>-  0x09, 0x30, 0x43, 0x20, 0x00, 0x00,
>-  0x48, 0x45, 0x34, 0xa8, 0x00, 0x00,
>-  0x04, 0x44, 0xe0, 0x08, 0x00, 0x00,
>-  0x0e, 0x80, 0x5d, 0x20, 0x00, 0x00,
>-  0xa5, 0x92, 0x42, 0x10, 0x00, 0x00,
>-  0x12, 0x0d, 0xc8, 0x50, 0x00, 0x00,
>-  0xb5, 0x4c, 0xa9, 0x70, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom29_27[162] = {
>-  0x62, 0x55, 0x8a, 0x88, 0x00, 0x00,
>-  0x34, 0x60, 0xd1, 0x10, 0x00, 0x00,
>-  0x48, 0x05, 0x01, 0x28, 0x00, 0x00,
>-  0x00, 0xaf, 0x10, 0x10, 0x00, 0x00,
>-  0x28, 0x08, 0x21, 0x80, 0x00, 0x00,
>-  0x81, 0x0a, 0x50, 0x48, 0x00, 0x00,
>-  0x23, 0x06, 0x23, 0x00, 0x00, 0x00,
>-  0x06, 0x80, 0x84, 0xc8, 0x00, 0x00,
>-  0x80, 0x17, 0x05, 0x00, 0x00, 0x00,
>-  0x30, 0x10, 0x41, 0xa0, 0x00, 0x00,
>-  0x8c, 0x20, 0x1a, 0x40, 0x00, 0x00,
>-  0x54, 0x01, 0x64, 0x00, 0x00, 0x00,
>-  0x80, 0xc0, 0x28, 0x30, 0x00, 0x00,
>-  0x40, 0x55, 0x02, 0x08, 0x00, 0x00,
>-  0x15, 0x40, 0x55, 0x50, 0x00, 0x00,
>-  0xc0, 0x06, 0x20, 0x48, 0x00, 0x00,
>-  0x28, 0x13, 0x00, 0x40, 0x00, 0x00,
>-  0x05, 0x0e, 0x02, 0x80, 0x00, 0x00,
>-  0x64, 0x80, 0x04, 0x88, 0x00, 0x00,
>-  0x81, 0x81, 0x00, 0xb0, 0x00, 0x00,
>-  0x10, 0x98, 0x88, 0x08, 0x00, 0x00,
>-  0x84, 0x22, 0x40, 0x10, 0x00, 0x00,
>-  0x12, 0x30, 0x49, 0x00, 0x00, 0x00,
>-  0x62, 0x01, 0x74, 0x00, 0x00, 0x00,
>-  0x28, 0x60, 0x81, 0x50, 0x00, 0x00,
>-  0x0e, 0x0a, 0x18, 0x20, 0x00, 0x00,
>-  0x10, 0x84, 0xa2, 0x20, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom29_28[168] = {
>-  0x40, 0x55, 0x02, 0x08, 0x00, 0x00,
>-  0x15, 0x40, 0x55, 0x50, 0x00, 0x00,
>-  0xc0, 0x06, 0x20, 0x48, 0x00, 0x00,
>-  0x28, 0x13, 0x00, 0x40, 0x00, 0x00,
>-  0x05, 0x0e, 0x02, 0x80, 0x00, 0x00,
>-  0x64, 0x80, 0x04, 0x88, 0x00, 0x00,
>-  0x81, 0x81, 0x00, 0xb0, 0x00, 0x00,
>-  0x10, 0x98, 0x88, 0x08, 0x00, 0x00,
>-  0x84, 0x22, 0x40, 0x10, 0x00, 0x00,
>-  0x12, 0x30, 0x49, 0x00, 0x00, 0x00,
>-  0x62, 0x01, 0x74, 0x00, 0x00, 0x00,
>-  0x28, 0x60, 0x81, 0x50, 0x00, 0x00,
>-  0x0e, 0x0a, 0x18, 0x20, 0x00, 0x00,
>-  0x10, 0x84, 0xa2, 0x20, 0x00, 0x00,
>-  0x62, 0x55, 0x8a, 0x88, 0x00, 0x00,
>-  0x34, 0x60, 0xd1, 0x10, 0x00, 0x00,
>-  0x48, 0x05, 0x01, 0x28, 0x00, 0x00,
>-  0x00, 0xaf, 0x10, 0x10, 0x00, 0x00,
>-  0x28, 0x08, 0x21, 0x80, 0x00, 0x00,
>-  0x81, 0x0a, 0x50, 0x48, 0x00, 0x00,
>-  0x23, 0x06, 0x23, 0x00, 0x00, 0x00,
>-  0x06, 0x80, 0x84, 0xc8, 0x00, 0x00,
>-  0x80, 0x17, 0x05, 0x00, 0x00, 0x00,
>-  0x30, 0x10, 0x41, 0xa0, 0x00, 0x00,
>-  0x8c, 0x20, 0x1a, 0x40, 0x00, 0x00,
>-  0x54, 0x01, 0x64, 0x00, 0x00, 0x00,
>-  0x80, 0xc0, 0x28, 0x30, 0x00, 0x00,
>-  0xbe, 0x1f, 0x99, 0xb0, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom29_29[174] = {
>-  0x40, 0x55, 0x02, 0x08, 0x00, 0x00,
>-  0x15, 0x40, 0x55, 0x50, 0x00, 0x00,
>-  0xc0, 0x06, 0x20, 0x48, 0x00, 0x00,
>-  0x28, 0x13, 0x00, 0x40, 0x00, 0x00,
>-  0x05, 0x0e, 0x02, 0x80, 0x00, 0x00,
>-  0x64, 0x80, 0x04, 0x88, 0x00, 0x00,
>-  0x81, 0x81, 0x00, 0xb0, 0x00, 0x00,
>-  0x10, 0x98, 0x88, 0x08, 0x00, 0x00,
>-  0x84, 0x22, 0x40, 0x10, 0x00, 0x00,
>-  0x12, 0x30, 0x49, 0x00, 0x00, 0x00,
>-  0x62, 0x01, 0x74, 0x00, 0x00, 0x00,
>-  0x28, 0x60, 0x81, 0x50, 0x00, 0x00,
>-  0x0e, 0x0a, 0x18, 0x20, 0x00, 0x00,
>-  0x10, 0x84, 0xa2, 0x20, 0x00, 0x00,
>-  0x40, 0x55, 0x88, 0x88, 0x00, 0x00,
>-  0x15, 0x40, 0xc4, 0x40, 0x00, 0x00,
>-  0xc0, 0x05, 0x60, 0x00, 0x00, 0x00,
>-  0x28, 0x10, 0x04, 0x48, 0x00, 0x00,
>-  0x05, 0x0e, 0x20, 0x80, 0x00, 0x00,
>-  0x64, 0x81, 0x10, 0x08, 0x00, 0x00,
>-  0x81, 0x80, 0xa4, 0x10, 0x00, 0x00,
>-  0x10, 0x9a, 0x0a, 0x80, 0x00, 0x00,
>-  0x84, 0x20, 0x28, 0x68, 0x00, 0x00,
>-  0x12, 0x30, 0x47, 0x80, 0x00, 0x00,
>-  0x62, 0x02, 0x10, 0x10, 0x00, 0x00,
>-  0x28, 0x62, 0x19, 0x00, 0x00, 0x00,
>-  0x0e, 0x08, 0x02, 0x18, 0x00, 0x00,
>-  0x10, 0x85, 0x11, 0x20, 0x00, 0x00,
>-  0x29, 0x50, 0x42, 0x60, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom29_3[18] = {
>-  0xac, 0xda, 0xb2, 0x48, 0x00, 0x00,
>-  0x55, 0x6d, 0x55, 0x28, 0x00, 0x00,
>-  0x27, 0xb5, 0x0c, 0xd8, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom29_4[24] = {
>-  0x2c, 0xd8, 0x96, 0xa8, 0x00, 0x00,
>-  0x93, 0x6a, 0x55, 0x50, 0x00, 0x00,
>-  0x1a, 0xb4, 0x69, 0xa8, 0x00, 0x00,
>-  0x47, 0x2d, 0x0f, 0x50, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom29_5[30] = {
>-  0x64, 0xd9, 0x92, 0x88, 0x00, 0x00,
>-  0xa5, 0x68, 0x95, 0x50, 0x00, 0x00,
>-  0x52, 0xb5, 0x25, 0xa0, 0x00, 0x00,
>-  0x1d, 0xa9, 0x4e, 0x40, 0x00, 0x00,
>-  0x9c, 0x56, 0x38, 0xc0, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom29_6[36] = {
>-  0x4a, 0x55, 0x8a, 0x28, 0x00, 0x00,
>-  0x95, 0x48, 0x55, 0x50, 0x00, 0x00,
>-  0x14, 0xb5, 0x31, 0x18, 0x00, 0x00,
>-  0x51, 0xa9, 0x4a, 0x50, 0x00, 0x00,
>-  0x22, 0x6c, 0x8d, 0x90, 0x00, 0x00,
>-  0x88, 0x8e, 0x29, 0x60, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom29_7[42] = {
>-  0x62, 0x55, 0x8a, 0x88, 0x00, 0x00,
>-  0xb9, 0x22, 0xc4, 0x50, 0x00, 0x00,
>-  0x18, 0xb4, 0x61, 0xa8, 0x00, 0x00,
>-  0x54, 0x99, 0x13, 0x50, 0x00, 0x00,
>-  0x06, 0x6c, 0x4d, 0x90, 0x00, 0x00,
>-  0x85, 0x55, 0x24, 0x68, 0x00, 0x00,
>-  0xaa, 0x8a, 0x1a, 0x30, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom29_8[48] = {
>-  0xc0, 0x16, 0x40, 0x88, 0x00, 0x00,
>-  0x41, 0x60, 0x25, 0x40, 0x00, 0x00,
>-  0x88, 0x30, 0x01, 0xa8, 0x00, 0x00,
>-  0x20, 0xa4, 0x80, 0xd0, 0x00, 0x00,
>-  0x0a, 0x48, 0x51, 0x10, 0x00, 0x00,
>-  0x04, 0x9b, 0x08, 0x40, 0x00, 0x00,
>-  0x94, 0x40, 0x03, 0x18, 0x00, 0x00,
>-  0x72, 0x01, 0x96, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom29_9[54] = {
>-  0xa2, 0x55, 0x88, 0x88, 0x00, 0x00,
>-  0x34, 0x60, 0x91, 0x10, 0x00, 0x00,
>-  0x4a, 0x27, 0x01, 0x40, 0x00, 0x00,
>-  0x20, 0xa8, 0x0c, 0x30, 0x00, 0x00,
>-  0x11, 0x84, 0x58, 0xa0, 0x00, 0x00,
>-  0x49, 0x0a, 0x24, 0x00, 0x00, 0x00,
>-  0x86, 0x0e, 0x0a, 0x40, 0x00, 0x00,
>-  0x20, 0xd4, 0x22, 0x90, 0x00, 0x00,
>-  0x88, 0x4a, 0x41, 0x20, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom2_1[2] = {
>-  0xc0, 0x00
>-};
>-
>-const uint8_t kMaskRandom2_2[4] = {
>-  0xc0, 0x00,
>-  0x80, 0x00
>-};
>-
>-const uint8_t kMaskRandom30_1[6] = {
>-  0xff, 0xff, 0xff, 0xfc, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom30_10[60] = {
>-  0xc0, 0xa1, 0x81, 0x40, 0x00, 0x00,
>-  0x15, 0x56, 0x2a, 0xac, 0x00, 0x00,
>-  0x74, 0x40, 0xe8, 0x80, 0x00, 0x00,
>-  0x00, 0x9c, 0x01, 0x38, 0x00, 0x00,
>-  0x01, 0x2c, 0x02, 0x58, 0x00, 0x00,
>-  0x44, 0x92, 0x89, 0x24, 0x00, 0x00,
>-  0x88, 0x51, 0x10, 0xa0, 0x00, 0x00,
>-  0x20, 0xa4, 0x41, 0x48, 0x00, 0x00,
>-  0xaa, 0x05, 0x54, 0x08, 0x00, 0x00,
>-  0x02, 0x62, 0x04, 0xc4, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom30_11[66] = {
>-  0x62, 0x22, 0xc4, 0x44, 0x00, 0x00,
>-  0xf1, 0x11, 0xe2, 0x20, 0x00, 0x00,
>-  0x10, 0x0e, 0x20, 0x1c, 0x00, 0x00,
>-  0x10, 0xb0, 0x21, 0x60, 0x00, 0x00,
>-  0x24, 0x24, 0x48, 0x48, 0x00, 0x00,
>-  0x01, 0x12, 0x02, 0x24, 0x00, 0x00,
>-  0x00, 0xc4, 0x01, 0x88, 0x00, 0x00,
>-  0x04, 0xa2, 0x09, 0x44, 0x00, 0x00,
>-  0x02, 0x58, 0x04, 0xb0, 0x00, 0x00,
>-  0x2b, 0x00, 0x56, 0x00, 0x00, 0x00,
>-  0x98, 0x41, 0x30, 0x80, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom30_12[72] = {
>-  0x88, 0x91, 0x11, 0x20, 0x00, 0x00,
>-  0x40, 0x54, 0x80, 0xa8, 0x00, 0x00,
>-  0x82, 0x63, 0x04, 0xc4, 0x00, 0x00,
>-  0x21, 0xa4, 0x43, 0x48, 0x00, 0x00,
>-  0x10, 0x64, 0x20, 0xc8, 0x00, 0x00,
>-  0x44, 0x0a, 0x88, 0x14, 0x00, 0x00,
>-  0x10, 0xc8, 0x21, 0x90, 0x00, 0x00,
>-  0x4d, 0x2a, 0x9a, 0x54, 0x00, 0x00,
>-  0x38, 0x02, 0x70, 0x04, 0x00, 0x00,
>-  0x17, 0x48, 0x2e, 0x90, 0x00, 0x00,
>-  0x90, 0x85, 0x21, 0x08, 0x00, 0x00,
>-  0x72, 0x14, 0xe4, 0x28, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom30_13[78] = {
>-  0x62, 0xa2, 0xc5, 0x44, 0x00, 0x00,
>-  0x34, 0x44, 0x68, 0x88, 0x00, 0x00,
>-  0x40, 0x4a, 0x80, 0x94, 0x00, 0x00,
>-  0xc4, 0x05, 0x88, 0x08, 0x00, 0x00,
>-  0x08, 0x60, 0x10, 0xc0, 0x00, 0x00,
>-  0x94, 0x13, 0x28, 0x24, 0x00, 0x00,
>-  0x88, 0xc1, 0x11, 0x80, 0x00, 0x00,
>-  0x21, 0x32, 0x42, 0x64, 0x00, 0x00,
>-  0xc1, 0x41, 0x82, 0x80, 0x00, 0x00,
>-  0x10, 0x68, 0x20, 0xd0, 0x00, 0x00,
>-  0x06, 0x90, 0x0d, 0x20, 0x00, 0x00,
>-  0x59, 0x00, 0xb2, 0x00, 0x00, 0x00,
>-  0x0a, 0x0c, 0x14, 0x18, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom30_14[84] = {
>-  0x40, 0x82, 0x81, 0x04, 0x00, 0x00,
>-  0x15, 0x54, 0x2a, 0xa8, 0x00, 0x00,
>-  0x88, 0x13, 0x10, 0x24, 0x00, 0x00,
>-  0xc0, 0x11, 0x80, 0x20, 0x00, 0x00,
>-  0x80, 0xa1, 0x01, 0x40, 0x00, 0x00,
>-  0x01, 0x22, 0x02, 0x44, 0x00, 0x00,
>-  0x40, 0x2c, 0x80, 0x58, 0x00, 0x00,
>-  0x22, 0x02, 0x44, 0x04, 0x00, 0x00,
>-  0x90, 0x05, 0x20, 0x08, 0x00, 0x00,
>-  0x12, 0x40, 0x24, 0x80, 0x00, 0x00,
>-  0x5d, 0x00, 0xba, 0x00, 0x00, 0x00,
>-  0x20, 0x54, 0x40, 0xa8, 0x00, 0x00,
>-  0x86, 0x09, 0x0c, 0x10, 0x00, 0x00,
>-  0x28, 0x88, 0x51, 0x10, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom30_15[90] = {
>-  0x62, 0x22, 0xc4, 0x44, 0x00, 0x00,
>-  0x31, 0x10, 0x62, 0x20, 0x00, 0x00,
>-  0x58, 0x00, 0xb0, 0x00, 0x00, 0x00,
>-  0x01, 0x12, 0x02, 0x24, 0x00, 0x00,
>-  0x88, 0x21, 0x10, 0x40, 0x00, 0x00,
>-  0x44, 0x02, 0x88, 0x04, 0x00, 0x00,
>-  0x29, 0x04, 0x52, 0x08, 0x00, 0x00,
>-  0x82, 0xa1, 0x05, 0x40, 0x00, 0x00,
>-  0x0a, 0x1a, 0x14, 0x34, 0x00, 0x00,
>-  0x11, 0xe0, 0x23, 0xc0, 0x00, 0x00,
>-  0x84, 0x05, 0x08, 0x08, 0x00, 0x00,
>-  0x86, 0x41, 0x0c, 0x80, 0x00, 0x00,
>-  0x00, 0x86, 0x01, 0x0c, 0x00, 0x00,
>-  0x44, 0x48, 0x88, 0x90, 0x00, 0x00,
>-  0x10, 0x98, 0x21, 0x30, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom30_16[96] = {
>-  0x90, 0x23, 0x20, 0x44, 0x00, 0x00,
>-  0x09, 0x50, 0x12, 0xa0, 0x00, 0x00,
>-  0x00, 0x6a, 0x00, 0xd4, 0x00, 0x00,
>-  0x20, 0x34, 0x40, 0x68, 0x00, 0x00,
>-  0x14, 0x44, 0x28, 0x88, 0x00, 0x00,
>-  0xc2, 0x11, 0x84, 0x20, 0x00, 0x00,
>-  0x00, 0xc6, 0x01, 0x8c, 0x00, 0x00,
>-  0x65, 0x80, 0xcb, 0x00, 0x00, 0x00,
>-  0x62, 0xa2, 0xc5, 0x44, 0x00, 0x00,
>-  0xb1, 0x15, 0x62, 0x28, 0x00, 0x00,
>-  0x18, 0x6a, 0x30, 0xd4, 0x00, 0x00,
>-  0x44, 0xd4, 0x89, 0xa8, 0x00, 0x00,
>-  0x13, 0x64, 0x26, 0xc8, 0x00, 0x00,
>-  0x49, 0x1a, 0x92, 0x34, 0x00, 0x00,
>-  0x86, 0x8d, 0x0d, 0x18, 0x00, 0x00,
>-  0xce, 0x58, 0xa0, 0x14, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom30_17[102] = {
>-  0x90, 0x23, 0x20, 0x44, 0x00, 0x00,
>-  0x09, 0x50, 0x12, 0xa0, 0x00, 0x00,
>-  0x00, 0x6a, 0x00, 0xd4, 0x00, 0x00,
>-  0x20, 0x34, 0x40, 0x68, 0x00, 0x00,
>-  0x14, 0x44, 0x28, 0x88, 0x00, 0x00,
>-  0xc2, 0x11, 0x84, 0x20, 0x00, 0x00,
>-  0x00, 0xc6, 0x01, 0x8c, 0x00, 0x00,
>-  0x65, 0x80, 0xcb, 0x00, 0x00, 0x00,
>-  0x62, 0x22, 0xc4, 0x44, 0x00, 0x00,
>-  0x24, 0x44, 0x48, 0x88, 0x00, 0x00,
>-  0xc0, 0x51, 0x80, 0xa0, 0x00, 0x00,
>-  0x03, 0x0c, 0x06, 0x18, 0x00, 0x00,
>-  0x16, 0x28, 0x2c, 0x50, 0x00, 0x00,
>-  0x89, 0x01, 0x12, 0x00, 0x00, 0x00,
>-  0x82, 0x91, 0x05, 0x20, 0x00, 0x00,
>-  0x08, 0xa4, 0x11, 0x48, 0x00, 0x00,
>-  0x90, 0x49, 0x20, 0x90, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom30_18[108] = {
>-  0x62, 0x22, 0xc4, 0x44, 0x00, 0x00,
>-  0x24, 0x44, 0x48, 0x88, 0x00, 0x00,
>-  0xc0, 0x51, 0x80, 0xa0, 0x00, 0x00,
>-  0x03, 0x0c, 0x06, 0x18, 0x00, 0x00,
>-  0x16, 0x28, 0x2c, 0x50, 0x00, 0x00,
>-  0x89, 0x01, 0x12, 0x00, 0x00, 0x00,
>-  0x82, 0x91, 0x05, 0x20, 0x00, 0x00,
>-  0x08, 0xa4, 0x11, 0x48, 0x00, 0x00,
>-  0x90, 0x49, 0x20, 0x90, 0x00, 0x00,
>-  0x90, 0x23, 0x20, 0x44, 0x00, 0x00,
>-  0x09, 0x50, 0x12, 0xa0, 0x00, 0x00,
>-  0x00, 0x6a, 0x00, 0xd4, 0x00, 0x00,
>-  0x20, 0x34, 0x40, 0x68, 0x00, 0x00,
>-  0x14, 0x44, 0x28, 0x88, 0x00, 0x00,
>-  0xc2, 0x11, 0x84, 0x20, 0x00, 0x00,
>-  0x00, 0xc6, 0x01, 0x8c, 0x00, 0x00,
>-  0x65, 0x80, 0xcb, 0x00, 0x00, 0x00,
>-  0x00, 0xb2, 0x47, 0x80, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom30_19[114] = {
>-  0x62, 0x22, 0xc4, 0x44, 0x00, 0x00,
>-  0x24, 0x44, 0x48, 0x88, 0x00, 0x00,
>-  0xc0, 0x51, 0x80, 0xa0, 0x00, 0x00,
>-  0x03, 0x0c, 0x06, 0x18, 0x00, 0x00,
>-  0x16, 0x28, 0x2c, 0x50, 0x00, 0x00,
>-  0x89, 0x01, 0x12, 0x00, 0x00, 0x00,
>-  0x82, 0x91, 0x05, 0x20, 0x00, 0x00,
>-  0x08, 0xa4, 0x11, 0x48, 0x00, 0x00,
>-  0x90, 0x49, 0x20, 0x90, 0x00, 0x00,
>-  0xc0, 0xa1, 0x81, 0x40, 0x00, 0x00,
>-  0x15, 0x56, 0x2a, 0xac, 0x00, 0x00,
>-  0x74, 0x40, 0xe8, 0x80, 0x00, 0x00,
>-  0x00, 0x9c, 0x01, 0x38, 0x00, 0x00,
>-  0x01, 0x2c, 0x02, 0x58, 0x00, 0x00,
>-  0x44, 0x92, 0x89, 0x24, 0x00, 0x00,
>-  0x88, 0x51, 0x10, 0xa0, 0x00, 0x00,
>-  0x20, 0xa4, 0x41, 0x48, 0x00, 0x00,
>-  0xaa, 0x05, 0x54, 0x08, 0x00, 0x00,
>-  0x02, 0x62, 0x04, 0xc4, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom30_2[12] = {
>-  0xec, 0xeb, 0xd9, 0xd4, 0x00, 0x00,
>-  0xbb, 0x9d, 0x77, 0x38, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom30_20[120] = {
>-  0xc0, 0xa1, 0x81, 0x40, 0x00, 0x00,
>-  0x15, 0x56, 0x2a, 0xac, 0x00, 0x00,
>-  0x74, 0x40, 0xe8, 0x80, 0x00, 0x00,
>-  0x00, 0x9c, 0x01, 0x38, 0x00, 0x00,
>-  0x01, 0x2c, 0x02, 0x58, 0x00, 0x00,
>-  0x44, 0x92, 0x89, 0x24, 0x00, 0x00,
>-  0x88, 0x51, 0x10, 0xa0, 0x00, 0x00,
>-  0x20, 0xa4, 0x41, 0x48, 0x00, 0x00,
>-  0xaa, 0x05, 0x54, 0x08, 0x00, 0x00,
>-  0x02, 0x62, 0x04, 0xc4, 0x00, 0x00,
>-  0x62, 0x22, 0xc4, 0x44, 0x00, 0x00,
>-  0x24, 0x44, 0x48, 0x88, 0x00, 0x00,
>-  0xc0, 0x51, 0x80, 0xa0, 0x00, 0x00,
>-  0x03, 0x0c, 0x06, 0x18, 0x00, 0x00,
>-  0x16, 0x28, 0x2c, 0x50, 0x00, 0x00,
>-  0x89, 0x01, 0x12, 0x00, 0x00, 0x00,
>-  0x82, 0x91, 0x05, 0x20, 0x00, 0x00,
>-  0x08, 0xa4, 0x11, 0x48, 0x00, 0x00,
>-  0x90, 0x49, 0x20, 0x90, 0x00, 0x00,
>-  0x51, 0x88, 0xd1, 0x78, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom30_21[126] = {
>-  0xc0, 0xa1, 0x81, 0x40, 0x00, 0x00,
>-  0x15, 0x56, 0x2a, 0xac, 0x00, 0x00,
>-  0x74, 0x40, 0xe8, 0x80, 0x00, 0x00,
>-  0x00, 0x9c, 0x01, 0x38, 0x00, 0x00,
>-  0x01, 0x2c, 0x02, 0x58, 0x00, 0x00,
>-  0x44, 0x92, 0x89, 0x24, 0x00, 0x00,
>-  0x88, 0x51, 0x10, 0xa0, 0x00, 0x00,
>-  0x20, 0xa4, 0x41, 0x48, 0x00, 0x00,
>-  0xaa, 0x05, 0x54, 0x08, 0x00, 0x00,
>-  0x02, 0x62, 0x04, 0xc4, 0x00, 0x00,
>-  0x62, 0x22, 0xc4, 0x44, 0x00, 0x00,
>-  0xf1, 0x11, 0xe2, 0x20, 0x00, 0x00,
>-  0x10, 0x0e, 0x20, 0x1c, 0x00, 0x00,
>-  0x10, 0xb0, 0x21, 0x60, 0x00, 0x00,
>-  0x24, 0x24, 0x48, 0x48, 0x00, 0x00,
>-  0x01, 0x12, 0x02, 0x24, 0x00, 0x00,
>-  0x00, 0xc4, 0x01, 0x88, 0x00, 0x00,
>-  0x04, 0xa2, 0x09, 0x44, 0x00, 0x00,
>-  0x02, 0x58, 0x04, 0xb0, 0x00, 0x00,
>-  0x2b, 0x00, 0x56, 0x00, 0x00, 0x00,
>-  0x98, 0x41, 0x30, 0x80, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom30_22[132] = {
>-  0x62, 0x22, 0xc4, 0x44, 0x00, 0x00,
>-  0xf1, 0x11, 0xe2, 0x20, 0x00, 0x00,
>-  0x10, 0x0e, 0x20, 0x1c, 0x00, 0x00,
>-  0x10, 0xb0, 0x21, 0x60, 0x00, 0x00,
>-  0x24, 0x24, 0x48, 0x48, 0x00, 0x00,
>-  0x01, 0x12, 0x02, 0x24, 0x00, 0x00,
>-  0x00, 0xc4, 0x01, 0x88, 0x00, 0x00,
>-  0x04, 0xa2, 0x09, 0x44, 0x00, 0x00,
>-  0x02, 0x58, 0x04, 0xb0, 0x00, 0x00,
>-  0x2b, 0x00, 0x56, 0x00, 0x00, 0x00,
>-  0x98, 0x41, 0x30, 0x80, 0x00, 0x00,
>-  0xc0, 0xa1, 0x81, 0x40, 0x00, 0x00,
>-  0x15, 0x56, 0x2a, 0xac, 0x00, 0x00,
>-  0x74, 0x40, 0xe8, 0x80, 0x00, 0x00,
>-  0x00, 0x9c, 0x01, 0x38, 0x00, 0x00,
>-  0x01, 0x2c, 0x02, 0x58, 0x00, 0x00,
>-  0x44, 0x92, 0x89, 0x24, 0x00, 0x00,
>-  0x88, 0x51, 0x10, 0xa0, 0x00, 0x00,
>-  0x20, 0xa4, 0x41, 0x48, 0x00, 0x00,
>-  0xaa, 0x05, 0x54, 0x08, 0x00, 0x00,
>-  0x02, 0x62, 0x04, 0xc4, 0x00, 0x00,
>-  0x03, 0x10, 0x18, 0x74, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom30_23[138] = {
>-  0x62, 0x22, 0xc4, 0x44, 0x00, 0x00,
>-  0xf1, 0x11, 0xe2, 0x20, 0x00, 0x00,
>-  0x10, 0x0e, 0x20, 0x1c, 0x00, 0x00,
>-  0x10, 0xb0, 0x21, 0x60, 0x00, 0x00,
>-  0x24, 0x24, 0x48, 0x48, 0x00, 0x00,
>-  0x01, 0x12, 0x02, 0x24, 0x00, 0x00,
>-  0x00, 0xc4, 0x01, 0x88, 0x00, 0x00,
>-  0x04, 0xa2, 0x09, 0x44, 0x00, 0x00,
>-  0x02, 0x58, 0x04, 0xb0, 0x00, 0x00,
>-  0x2b, 0x00, 0x56, 0x00, 0x00, 0x00,
>-  0x98, 0x41, 0x30, 0x80, 0x00, 0x00,
>-  0x88, 0x91, 0x11, 0x20, 0x00, 0x00,
>-  0x40, 0x54, 0x80, 0xa8, 0x00, 0x00,
>-  0x82, 0x63, 0x04, 0xc4, 0x00, 0x00,
>-  0x21, 0xa4, 0x43, 0x48, 0x00, 0x00,
>-  0x10, 0x64, 0x20, 0xc8, 0x00, 0x00,
>-  0x44, 0x0a, 0x88, 0x14, 0x00, 0x00,
>-  0x10, 0xc8, 0x21, 0x90, 0x00, 0x00,
>-  0x4d, 0x2a, 0x9a, 0x54, 0x00, 0x00,
>-  0x38, 0x02, 0x70, 0x04, 0x00, 0x00,
>-  0x17, 0x48, 0x2e, 0x90, 0x00, 0x00,
>-  0x90, 0x85, 0x21, 0x08, 0x00, 0x00,
>-  0x72, 0x14, 0xe4, 0x28, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom30_24[144] = {
>-  0x88, 0x91, 0x11, 0x20, 0x00, 0x00,
>-  0x40, 0x54, 0x80, 0xa8, 0x00, 0x00,
>-  0x82, 0x63, 0x04, 0xc4, 0x00, 0x00,
>-  0x21, 0xa4, 0x43, 0x48, 0x00, 0x00,
>-  0x10, 0x64, 0x20, 0xc8, 0x00, 0x00,
>-  0x44, 0x0a, 0x88, 0x14, 0x00, 0x00,
>-  0x10, 0xc8, 0x21, 0x90, 0x00, 0x00,
>-  0x4d, 0x2a, 0x9a, 0x54, 0x00, 0x00,
>-  0x38, 0x02, 0x70, 0x04, 0x00, 0x00,
>-  0x17, 0x48, 0x2e, 0x90, 0x00, 0x00,
>-  0x90, 0x85, 0x21, 0x08, 0x00, 0x00,
>-  0x72, 0x14, 0xe4, 0x28, 0x00, 0x00,
>-  0x62, 0x22, 0xc4, 0x44, 0x00, 0x00,
>-  0xf1, 0x11, 0xe2, 0x20, 0x00, 0x00,
>-  0x10, 0x0e, 0x20, 0x1c, 0x00, 0x00,
>-  0x10, 0xb0, 0x21, 0x60, 0x00, 0x00,
>-  0x24, 0x24, 0x48, 0x48, 0x00, 0x00,
>-  0x01, 0x12, 0x02, 0x24, 0x00, 0x00,
>-  0x00, 0xc4, 0x01, 0x88, 0x00, 0x00,
>-  0x04, 0xa2, 0x09, 0x44, 0x00, 0x00,
>-  0x02, 0x58, 0x04, 0xb0, 0x00, 0x00,
>-  0x2b, 0x00, 0x56, 0x00, 0x00, 0x00,
>-  0x98, 0x41, 0x30, 0x80, 0x00, 0x00,
>-  0xf3, 0x4d, 0x1c, 0x70, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom30_25[150] = {
>-  0x88, 0x91, 0x11, 0x20, 0x00, 0x00,
>-  0x40, 0x54, 0x80, 0xa8, 0x00, 0x00,
>-  0x82, 0x63, 0x04, 0xc4, 0x00, 0x00,
>-  0x21, 0xa4, 0x43, 0x48, 0x00, 0x00,
>-  0x10, 0x64, 0x20, 0xc8, 0x00, 0x00,
>-  0x44, 0x0a, 0x88, 0x14, 0x00, 0x00,
>-  0x10, 0xc8, 0x21, 0x90, 0x00, 0x00,
>-  0x4d, 0x2a, 0x9a, 0x54, 0x00, 0x00,
>-  0x38, 0x02, 0x70, 0x04, 0x00, 0x00,
>-  0x17, 0x48, 0x2e, 0x90, 0x00, 0x00,
>-  0x90, 0x85, 0x21, 0x08, 0x00, 0x00,
>-  0x72, 0x14, 0xe4, 0x28, 0x00, 0x00,
>-  0x62, 0xa2, 0xc5, 0x44, 0x00, 0x00,
>-  0x34, 0x44, 0x68, 0x88, 0x00, 0x00,
>-  0x40, 0x4a, 0x80, 0x94, 0x00, 0x00,
>-  0xc4, 0x05, 0x88, 0x08, 0x00, 0x00,
>-  0x08, 0x60, 0x10, 0xc0, 0x00, 0x00,
>-  0x94, 0x13, 0x28, 0x24, 0x00, 0x00,
>-  0x88, 0xc1, 0x11, 0x80, 0x00, 0x00,
>-  0x21, 0x32, 0x42, 0x64, 0x00, 0x00,
>-  0xc1, 0x41, 0x82, 0x80, 0x00, 0x00,
>-  0x10, 0x68, 0x20, 0xd0, 0x00, 0x00,
>-  0x06, 0x90, 0x0d, 0x20, 0x00, 0x00,
>-  0x59, 0x00, 0xb2, 0x00, 0x00, 0x00,
>-  0x0a, 0x0c, 0x14, 0x18, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom30_26[156] = {
>-  0x62, 0xa2, 0xc5, 0x44, 0x00, 0x00,
>-  0x34, 0x44, 0x68, 0x88, 0x00, 0x00,
>-  0x40, 0x4a, 0x80, 0x94, 0x00, 0x00,
>-  0xc4, 0x05, 0x88, 0x08, 0x00, 0x00,
>-  0x08, 0x60, 0x10, 0xc0, 0x00, 0x00,
>-  0x94, 0x13, 0x28, 0x24, 0x00, 0x00,
>-  0x88, 0xc1, 0x11, 0x80, 0x00, 0x00,
>-  0x21, 0x32, 0x42, 0x64, 0x00, 0x00,
>-  0xc1, 0x41, 0x82, 0x80, 0x00, 0x00,
>-  0x10, 0x68, 0x20, 0xd0, 0x00, 0x00,
>-  0x06, 0x90, 0x0d, 0x20, 0x00, 0x00,
>-  0x59, 0x00, 0xb2, 0x00, 0x00, 0x00,
>-  0x0a, 0x0c, 0x14, 0x18, 0x00, 0x00,
>-  0x88, 0x91, 0x11, 0x20, 0x00, 0x00,
>-  0x40, 0x54, 0x80, 0xa8, 0x00, 0x00,
>-  0x82, 0x63, 0x04, 0xc4, 0x00, 0x00,
>-  0x21, 0xa4, 0x43, 0x48, 0x00, 0x00,
>-  0x10, 0x64, 0x20, 0xc8, 0x00, 0x00,
>-  0x44, 0x0a, 0x88, 0x14, 0x00, 0x00,
>-  0x10, 0xc8, 0x21, 0x90, 0x00, 0x00,
>-  0x4d, 0x2a, 0x9a, 0x54, 0x00, 0x00,
>-  0x38, 0x02, 0x70, 0x04, 0x00, 0x00,
>-  0x17, 0x48, 0x2e, 0x90, 0x00, 0x00,
>-  0x90, 0x85, 0x21, 0x08, 0x00, 0x00,
>-  0x72, 0x14, 0xe4, 0x28, 0x00, 0x00,
>-  0x83, 0x11, 0xad, 0xe8, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom30_27[162] = {
>-  0x62, 0xa2, 0xc5, 0x44, 0x00, 0x00,
>-  0x34, 0x44, 0x68, 0x88, 0x00, 0x00,
>-  0x40, 0x4a, 0x80, 0x94, 0x00, 0x00,
>-  0xc4, 0x05, 0x88, 0x08, 0x00, 0x00,
>-  0x08, 0x60, 0x10, 0xc0, 0x00, 0x00,
>-  0x94, 0x13, 0x28, 0x24, 0x00, 0x00,
>-  0x88, 0xc1, 0x11, 0x80, 0x00, 0x00,
>-  0x21, 0x32, 0x42, 0x64, 0x00, 0x00,
>-  0xc1, 0x41, 0x82, 0x80, 0x00, 0x00,
>-  0x10, 0x68, 0x20, 0xd0, 0x00, 0x00,
>-  0x06, 0x90, 0x0d, 0x20, 0x00, 0x00,
>-  0x59, 0x00, 0xb2, 0x00, 0x00, 0x00,
>-  0x0a, 0x0c, 0x14, 0x18, 0x00, 0x00,
>-  0x40, 0x82, 0x81, 0x04, 0x00, 0x00,
>-  0x15, 0x54, 0x2a, 0xa8, 0x00, 0x00,
>-  0x88, 0x13, 0x10, 0x24, 0x00, 0x00,
>-  0xc0, 0x11, 0x80, 0x20, 0x00, 0x00,
>-  0x80, 0xa1, 0x01, 0x40, 0x00, 0x00,
>-  0x01, 0x22, 0x02, 0x44, 0x00, 0x00,
>-  0x40, 0x2c, 0x80, 0x58, 0x00, 0x00,
>-  0x22, 0x02, 0x44, 0x04, 0x00, 0x00,
>-  0x90, 0x05, 0x20, 0x08, 0x00, 0x00,
>-  0x12, 0x40, 0x24, 0x80, 0x00, 0x00,
>-  0x5d, 0x00, 0xba, 0x00, 0x00, 0x00,
>-  0x20, 0x54, 0x40, 0xa8, 0x00, 0x00,
>-  0x86, 0x09, 0x0c, 0x10, 0x00, 0x00,
>-  0x28, 0x88, 0x51, 0x10, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom30_28[168] = {
>-  0x40, 0x82, 0x81, 0x04, 0x00, 0x00,
>-  0x15, 0x54, 0x2a, 0xa8, 0x00, 0x00,
>-  0x88, 0x13, 0x10, 0x24, 0x00, 0x00,
>-  0xc0, 0x11, 0x80, 0x20, 0x00, 0x00,
>-  0x80, 0xa1, 0x01, 0x40, 0x00, 0x00,
>-  0x01, 0x22, 0x02, 0x44, 0x00, 0x00,
>-  0x40, 0x2c, 0x80, 0x58, 0x00, 0x00,
>-  0x22, 0x02, 0x44, 0x04, 0x00, 0x00,
>-  0x90, 0x05, 0x20, 0x08, 0x00, 0x00,
>-  0x12, 0x40, 0x24, 0x80, 0x00, 0x00,
>-  0x5d, 0x00, 0xba, 0x00, 0x00, 0x00,
>-  0x20, 0x54, 0x40, 0xa8, 0x00, 0x00,
>-  0x86, 0x09, 0x0c, 0x10, 0x00, 0x00,
>-  0x28, 0x88, 0x51, 0x10, 0x00, 0x00,
>-  0x62, 0xa2, 0xc5, 0x44, 0x00, 0x00,
>-  0x34, 0x44, 0x68, 0x88, 0x00, 0x00,
>-  0x40, 0x4a, 0x80, 0x94, 0x00, 0x00,
>-  0xc4, 0x05, 0x88, 0x08, 0x00, 0x00,
>-  0x08, 0x60, 0x10, 0xc0, 0x00, 0x00,
>-  0x94, 0x13, 0x28, 0x24, 0x00, 0x00,
>-  0x88, 0xc1, 0x11, 0x80, 0x00, 0x00,
>-  0x21, 0x32, 0x42, 0x64, 0x00, 0x00,
>-  0xc1, 0x41, 0x82, 0x80, 0x00, 0x00,
>-  0x10, 0x68, 0x20, 0xd0, 0x00, 0x00,
>-  0x06, 0x90, 0x0d, 0x20, 0x00, 0x00,
>-  0x59, 0x00, 0xb2, 0x00, 0x00, 0x00,
>-  0x0a, 0x0c, 0x14, 0x18, 0x00, 0x00,
>-  0x94, 0x59, 0x03, 0x18, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom30_29[174] = {
>-  0x40, 0x82, 0x81, 0x04, 0x00, 0x00,
>-  0x15, 0x54, 0x2a, 0xa8, 0x00, 0x00,
>-  0x88, 0x13, 0x10, 0x24, 0x00, 0x00,
>-  0xc0, 0x11, 0x80, 0x20, 0x00, 0x00,
>-  0x80, 0xa1, 0x01, 0x40, 0x00, 0x00,
>-  0x01, 0x22, 0x02, 0x44, 0x00, 0x00,
>-  0x40, 0x2c, 0x80, 0x58, 0x00, 0x00,
>-  0x22, 0x02, 0x44, 0x04, 0x00, 0x00,
>-  0x90, 0x05, 0x20, 0x08, 0x00, 0x00,
>-  0x12, 0x40, 0x24, 0x80, 0x00, 0x00,
>-  0x5d, 0x00, 0xba, 0x00, 0x00, 0x00,
>-  0x20, 0x54, 0x40, 0xa8, 0x00, 0x00,
>-  0x86, 0x09, 0x0c, 0x10, 0x00, 0x00,
>-  0x28, 0x88, 0x51, 0x10, 0x00, 0x00,
>-  0x62, 0x22, 0xc4, 0x44, 0x00, 0x00,
>-  0x31, 0x10, 0x62, 0x20, 0x00, 0x00,
>-  0x58, 0x00, 0xb0, 0x00, 0x00, 0x00,
>-  0x01, 0x12, 0x02, 0x24, 0x00, 0x00,
>-  0x88, 0x21, 0x10, 0x40, 0x00, 0x00,
>-  0x44, 0x02, 0x88, 0x04, 0x00, 0x00,
>-  0x29, 0x04, 0x52, 0x08, 0x00, 0x00,
>-  0x82, 0xa1, 0x05, 0x40, 0x00, 0x00,
>-  0x0a, 0x1a, 0x14, 0x34, 0x00, 0x00,
>-  0x11, 0xe0, 0x23, 0xc0, 0x00, 0x00,
>-  0x84, 0x05, 0x08, 0x08, 0x00, 0x00,
>-  0x86, 0x41, 0x0c, 0x80, 0x00, 0x00,
>-  0x00, 0x86, 0x01, 0x0c, 0x00, 0x00,
>-  0x44, 0x48, 0x88, 0x90, 0x00, 0x00,
>-  0x10, 0x98, 0x21, 0x30, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom30_3[18] = {
>-  0xac, 0x93, 0x59, 0x24, 0x00, 0x00,
>-  0x55, 0x4a, 0xaa, 0x94, 0x00, 0x00,
>-  0x43, 0x36, 0x86, 0x6c, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom30_30[180] = {
>-  0x62, 0x22, 0xc4, 0x44, 0x00, 0x00,
>-  0x31, 0x10, 0x62, 0x20, 0x00, 0x00,
>-  0x58, 0x00, 0xb0, 0x00, 0x00, 0x00,
>-  0x01, 0x12, 0x02, 0x24, 0x00, 0x00,
>-  0x88, 0x21, 0x10, 0x40, 0x00, 0x00,
>-  0x44, 0x02, 0x88, 0x04, 0x00, 0x00,
>-  0x29, 0x04, 0x52, 0x08, 0x00, 0x00,
>-  0x82, 0xa1, 0x05, 0x40, 0x00, 0x00,
>-  0x0a, 0x1a, 0x14, 0x34, 0x00, 0x00,
>-  0x11, 0xe0, 0x23, 0xc0, 0x00, 0x00,
>-  0x84, 0x05, 0x08, 0x08, 0x00, 0x00,
>-  0x86, 0x41, 0x0c, 0x80, 0x00, 0x00,
>-  0x00, 0x86, 0x01, 0x0c, 0x00, 0x00,
>-  0x44, 0x48, 0x88, 0x90, 0x00, 0x00,
>-  0x10, 0x98, 0x21, 0x30, 0x00, 0x00,
>-  0x40, 0x82, 0x81, 0x04, 0x00, 0x00,
>-  0x15, 0x54, 0x2a, 0xa8, 0x00, 0x00,
>-  0x88, 0x13, 0x10, 0x24, 0x00, 0x00,
>-  0xc0, 0x11, 0x80, 0x20, 0x00, 0x00,
>-  0x80, 0xa1, 0x01, 0x40, 0x00, 0x00,
>-  0x01, 0x22, 0x02, 0x44, 0x00, 0x00,
>-  0x40, 0x2c, 0x80, 0x58, 0x00, 0x00,
>-  0x22, 0x02, 0x44, 0x04, 0x00, 0x00,
>-  0x90, 0x05, 0x20, 0x08, 0x00, 0x00,
>-  0x12, 0x40, 0x24, 0x80, 0x00, 0x00,
>-  0x5d, 0x00, 0xba, 0x00, 0x00, 0x00,
>-  0x20, 0x54, 0x40, 0xa8, 0x00, 0x00,
>-  0x86, 0x09, 0x0c, 0x10, 0x00, 0x00,
>-  0x28, 0x88, 0x51, 0x10, 0x00, 0x00,
>-  0x46, 0xf1, 0xef, 0xec, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom30_4[24] = {
>-  0x25, 0xaa, 0x4b, 0x54, 0x00, 0x00,
>-  0x95, 0x55, 0x2a, 0xa8, 0x00, 0x00,
>-  0x1a, 0x6a, 0x34, 0xd4, 0x00, 0x00,
>-  0x43, 0xd4, 0x87, 0xa8, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom30_5[30] = {
>-  0x64, 0xa2, 0xc9, 0x44, 0x00, 0x00,
>-  0x25, 0x54, 0x4a, 0xa8, 0x00, 0x00,
>-  0x49, 0x68, 0x92, 0xd0, 0x00, 0x00,
>-  0x53, 0x90, 0xa7, 0x20, 0x00, 0x00,
>-  0x8e, 0x31, 0x1c, 0x60, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom30_6[36] = {
>-  0x62, 0x8a, 0xc5, 0x14, 0x00, 0x00,
>-  0x15, 0x54, 0x2a, 0xa8, 0x00, 0x00,
>-  0x4c, 0x46, 0x98, 0x8c, 0x00, 0x00,
>-  0x52, 0x94, 0xa5, 0x28, 0x00, 0x00,
>-  0x23, 0x64, 0x46, 0xc8, 0x00, 0x00,
>-  0x8a, 0x59, 0x14, 0xb0, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom30_7[42] = {
>-  0x62, 0xa2, 0xc5, 0x44, 0x00, 0x00,
>-  0xb1, 0x15, 0x62, 0x28, 0x00, 0x00,
>-  0x18, 0x6a, 0x30, 0xd4, 0x00, 0x00,
>-  0x44, 0xd4, 0x89, 0xa8, 0x00, 0x00,
>-  0x13, 0x64, 0x26, 0xc8, 0x00, 0x00,
>-  0x49, 0x1a, 0x92, 0x34, 0x00, 0x00,
>-  0x86, 0x8d, 0x0d, 0x18, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom30_8[48] = {
>-  0x90, 0x23, 0x20, 0x44, 0x00, 0x00,
>-  0x09, 0x50, 0x12, 0xa0, 0x00, 0x00,
>-  0x00, 0x6a, 0x00, 0xd4, 0x00, 0x00,
>-  0x20, 0x34, 0x40, 0x68, 0x00, 0x00,
>-  0x14, 0x44, 0x28, 0x88, 0x00, 0x00,
>-  0xc2, 0x11, 0x84, 0x20, 0x00, 0x00,
>-  0x00, 0xc6, 0x01, 0x8c, 0x00, 0x00,
>-  0x65, 0x80, 0xcb, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom30_9[54] = {
>-  0x62, 0x22, 0xc4, 0x44, 0x00, 0x00,
>-  0x24, 0x44, 0x48, 0x88, 0x00, 0x00,
>-  0xc0, 0x51, 0x80, 0xa0, 0x00, 0x00,
>-  0x03, 0x0c, 0x06, 0x18, 0x00, 0x00,
>-  0x16, 0x28, 0x2c, 0x50, 0x00, 0x00,
>-  0x89, 0x01, 0x12, 0x00, 0x00, 0x00,
>-  0x82, 0x91, 0x05, 0x20, 0x00, 0x00,
>-  0x08, 0xa4, 0x11, 0x48, 0x00, 0x00,
>-  0x90, 0x49, 0x20, 0x90, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom31_1[6] = {
>-  0xff, 0xff, 0xff, 0xfe, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom31_10[60] = {
>-  0xc0, 0xa0, 0x8a, 0xa2, 0x00, 0x00,
>-  0x15, 0x56, 0x21, 0x44, 0x00, 0x00,
>-  0x74, 0x40, 0x02, 0x4a, 0x00, 0x00,
>-  0x00, 0x9c, 0x16, 0x84, 0x00, 0x00,
>-  0x01, 0x2d, 0xb0, 0x40, 0x00, 0x00,
>-  0x44, 0x93, 0x05, 0x18, 0x00, 0x00,
>-  0x88, 0x50, 0x48, 0x94, 0x00, 0x00,
>-  0x20, 0xa4, 0x70, 0x30, 0x00, 0x00,
>-  0xaa, 0x04, 0x54, 0x4a, 0x00, 0x00,
>-  0x02, 0x63, 0x09, 0x24, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom31_11[66] = {
>-  0x62, 0x22, 0xaa, 0xaa, 0x00, 0x00,
>-  0xf1, 0x10, 0x54, 0x44, 0x00, 0x00,
>-  0x10, 0x0e, 0x62, 0x22, 0x00, 0x00,
>-  0x10, 0xb1, 0x06, 0x84, 0x00, 0x00,
>-  0x24, 0x24, 0x0d, 0x30, 0x00, 0x00,
>-  0x01, 0x12, 0x81, 0xc2, 0x00, 0x00,
>-  0x00, 0xc4, 0x58, 0x88, 0x00, 0x00,
>-  0x04, 0xa3, 0xb0, 0x50, 0x00, 0x00,
>-  0x02, 0x59, 0x25, 0x02, 0x00, 0x00,
>-  0x2b, 0x01, 0x08, 0x64, 0x00, 0x00,
>-  0x98, 0x40, 0xd0, 0x18, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom31_12[72] = {
>-  0x88, 0x91, 0x08, 0x62, 0x00, 0x00,
>-  0x40, 0x54, 0x31, 0x44, 0x00, 0x00,
>-  0x82, 0x62, 0x9c, 0x02, 0x00, 0x00,
>-  0x21, 0xa4, 0x89, 0x90, 0x00, 0x00,
>-  0x10, 0x64, 0x1d, 0x20, 0x00, 0x00,
>-  0x44, 0x0a, 0x41, 0x98, 0x00, 0x00,
>-  0x10, 0xc9, 0x26, 0x80, 0x00, 0x00,
>-  0x4d, 0x2a, 0x5a, 0x20, 0x00, 0x00,
>-  0x38, 0x02, 0x62, 0x88, 0x00, 0x00,
>-  0x17, 0x49, 0x80, 0x46, 0x00, 0x00,
>-  0x90, 0x84, 0x22, 0x4a, 0x00, 0x00,
>-  0x72, 0x15, 0xd1, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom31_13[78] = {
>-  0x62, 0xa2, 0x8a, 0x2a, 0x00, 0x00,
>-  0x34, 0x44, 0x44, 0x44, 0x00, 0x00,
>-  0x40, 0x4b, 0x2c, 0x18, 0x00, 0x00,
>-  0xc4, 0x04, 0x18, 0xa0, 0x00, 0x00,
>-  0x08, 0x60, 0xc4, 0x08, 0x00, 0x00,
>-  0x94, 0x12, 0x92, 0x0c, 0x00, 0x00,
>-  0x88, 0xc0, 0x23, 0x04, 0x00, 0x00,
>-  0x21, 0x32, 0x24, 0x70, 0x00, 0x00,
>-  0xc1, 0x40, 0x80, 0xe2, 0x00, 0x00,
>-  0x10, 0x69, 0x51, 0x14, 0x00, 0x00,
>-  0x06, 0x90, 0x11, 0x42, 0x00, 0x00,
>-  0x59, 0x01, 0x41, 0x80, 0x00, 0x00,
>-  0x0a, 0x0d, 0x8a, 0x20, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom31_14[84] = {
>-  0x40, 0x82, 0x8a, 0xa2, 0x00, 0x00,
>-  0x15, 0x54, 0x44, 0x14, 0x00, 0x00,
>-  0x88, 0x13, 0x09, 0xa0, 0x00, 0x00,
>-  0xc0, 0x10, 0x19, 0x14, 0x00, 0x00,
>-  0x80, 0xa0, 0x30, 0x0c, 0x00, 0x00,
>-  0x01, 0x22, 0x60, 0x06, 0x00, 0x00,
>-  0x40, 0x2c, 0xc2, 0x10, 0x00, 0x00,
>-  0x22, 0x02, 0x80, 0x22, 0x00, 0x00,
>-  0x90, 0x04, 0x20, 0x58, 0x00, 0x00,
>-  0x12, 0x40, 0x12, 0xc0, 0x00, 0x00,
>-  0x5d, 0x00, 0x01, 0x28, 0x00, 0x00,
>-  0x20, 0x54, 0xa4, 0x80, 0x00, 0x00,
>-  0x86, 0x09, 0x48, 0x48, 0x00, 0x00,
>-  0x28, 0x89, 0x05, 0x10, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom31_15[90] = {
>-  0x62, 0x22, 0xaa, 0x22, 0x00, 0x00,
>-  0x31, 0x10, 0x44, 0x44, 0x00, 0x00,
>-  0x58, 0x00, 0x22, 0x22, 0x00, 0x00,
>-  0x01, 0x13, 0x00, 0x8a, 0x00, 0x00,
>-  0x88, 0x20, 0x40, 0x34, 0x00, 0x00,
>-  0x44, 0x02, 0x10, 0xd0, 0x00, 0x00,
>-  0x29, 0x04, 0x45, 0x08, 0x00, 0x00,
>-  0x82, 0xa0, 0x90, 0x12, 0x00, 0x00,
>-  0x0a, 0x1a, 0x0e, 0x02, 0x00, 0x00,
>-  0x11, 0xe1, 0x28, 0x40, 0x00, 0x00,
>-  0x84, 0x05, 0x04, 0x0c, 0x00, 0x00,
>-  0x86, 0x40, 0xc0, 0x90, 0x00, 0x00,
>-  0x00, 0x87, 0x13, 0x00, 0x00, 0x00,
>-  0x44, 0x48, 0x01, 0x1c, 0x00, 0x00,
>-  0x10, 0x98, 0x30, 0x44, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom31_16[96] = {
>-  0x90, 0x22, 0x40, 0xa8, 0x00, 0x00,
>-  0x09, 0x50, 0x31, 0x10, 0x00, 0x00,
>-  0x00, 0x6b, 0x08, 0x0e, 0x00, 0x00,
>-  0x20, 0x34, 0xc0, 0x90, 0x00, 0x00,
>-  0x14, 0x44, 0x25, 0x04, 0x00, 0x00,
>-  0xc2, 0x11, 0x02, 0x82, 0x00, 0x00,
>-  0x00, 0xc6, 0x80, 0xc4, 0x00, 0x00,
>-  0x65, 0x80, 0x2c, 0x60, 0x00, 0x00,
>-  0x62, 0xa2, 0x8a, 0xa2, 0x00, 0x00,
>-  0xb1, 0x14, 0x44, 0x54, 0x00, 0x00,
>-  0x18, 0x6b, 0x22, 0x22, 0x00, 0x00,
>-  0x44, 0xd4, 0x5c, 0x10, 0x00, 0x00,
>-  0x13, 0x64, 0x90, 0x68, 0x00, 0x00,
>-  0x49, 0x1b, 0x20, 0x52, 0x00, 0x00,
>-  0x86, 0x8c, 0x13, 0x0c, 0x00, 0x00,
>-  0x8d, 0x94, 0xa9, 0xe0, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom31_17[102] = {
>-  0x90, 0x22, 0x40, 0xa8, 0x00, 0x00,
>-  0x09, 0x50, 0x31, 0x10, 0x00, 0x00,
>-  0x00, 0x6b, 0x08, 0x0e, 0x00, 0x00,
>-  0x20, 0x34, 0xc0, 0x90, 0x00, 0x00,
>-  0x14, 0x44, 0x25, 0x04, 0x00, 0x00,
>-  0xc2, 0x11, 0x02, 0x82, 0x00, 0x00,
>-  0x00, 0xc6, 0x80, 0xc4, 0x00, 0x00,
>-  0x65, 0x80, 0x2c, 0x60, 0x00, 0x00,
>-  0x62, 0x22, 0xaa, 0xa2, 0x00, 0x00,
>-  0x24, 0x44, 0x44, 0x54, 0x00, 0x00,
>-  0xc0, 0x50, 0x0b, 0x0a, 0x00, 0x00,
>-  0x03, 0x0c, 0x12, 0x94, 0x00, 0x00,
>-  0x16, 0x29, 0x08, 0x64, 0x00, 0x00,
>-  0x89, 0x01, 0x80, 0x1a, 0x00, 0x00,
>-  0x82, 0x90, 0x41, 0x4c, 0x00, 0x00,
>-  0x08, 0xa4, 0x34, 0x12, 0x00, 0x00,
>-  0x90, 0x48, 0x88, 0xc8, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom31_18[108] = {
>-  0x62, 0x22, 0xaa, 0xa2, 0x00, 0x00,
>-  0x24, 0x44, 0x44, 0x54, 0x00, 0x00,
>-  0xc0, 0x50, 0x0b, 0x0a, 0x00, 0x00,
>-  0x03, 0x0c, 0x12, 0x94, 0x00, 0x00,
>-  0x16, 0x29, 0x08, 0x64, 0x00, 0x00,
>-  0x89, 0x01, 0x80, 0x1a, 0x00, 0x00,
>-  0x82, 0x90, 0x41, 0x4c, 0x00, 0x00,
>-  0x08, 0xa4, 0x34, 0x12, 0x00, 0x00,
>-  0x90, 0x48, 0x88, 0xc8, 0x00, 0x00,
>-  0x90, 0x22, 0x40, 0xa8, 0x00, 0x00,
>-  0x09, 0x50, 0x31, 0x10, 0x00, 0x00,
>-  0x00, 0x6b, 0x08, 0x0e, 0x00, 0x00,
>-  0x20, 0x34, 0xc0, 0x90, 0x00, 0x00,
>-  0x14, 0x44, 0x25, 0x04, 0x00, 0x00,
>-  0xc2, 0x11, 0x02, 0x82, 0x00, 0x00,
>-  0x00, 0xc6, 0x80, 0xc4, 0x00, 0x00,
>-  0x65, 0x80, 0x2c, 0x60, 0x00, 0x00,
>-  0xe3, 0xd1, 0x2e, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom31_19[114] = {
>-  0x62, 0x22, 0xaa, 0xa2, 0x00, 0x00,
>-  0x24, 0x44, 0x44, 0x54, 0x00, 0x00,
>-  0xc0, 0x50, 0x0b, 0x0a, 0x00, 0x00,
>-  0x03, 0x0c, 0x12, 0x94, 0x00, 0x00,
>-  0x16, 0x29, 0x08, 0x64, 0x00, 0x00,
>-  0x89, 0x01, 0x80, 0x1a, 0x00, 0x00,
>-  0x82, 0x90, 0x41, 0x4c, 0x00, 0x00,
>-  0x08, 0xa4, 0x34, 0x12, 0x00, 0x00,
>-  0x90, 0x48, 0x88, 0xc8, 0x00, 0x00,
>-  0xc0, 0xa0, 0x8a, 0xa2, 0x00, 0x00,
>-  0x15, 0x56, 0x21, 0x44, 0x00, 0x00,
>-  0x74, 0x40, 0x02, 0x4a, 0x00, 0x00,
>-  0x00, 0x9c, 0x16, 0x84, 0x00, 0x00,
>-  0x01, 0x2d, 0xb0, 0x40, 0x00, 0x00,
>-  0x44, 0x93, 0x05, 0x18, 0x00, 0x00,
>-  0x88, 0x50, 0x48, 0x94, 0x00, 0x00,
>-  0x20, 0xa4, 0x70, 0x30, 0x00, 0x00,
>-  0xaa, 0x04, 0x54, 0x4a, 0x00, 0x00,
>-  0x02, 0x63, 0x09, 0x24, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom31_2[12] = {
>-  0xec, 0xeb, 0x5d, 0x5c, 0x00, 0x00,
>-  0xbb, 0x9c, 0xf2, 0xf2, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom31_20[120] = {
>-  0xc0, 0xa0, 0x8a, 0xa2, 0x00, 0x00,
>-  0x15, 0x56, 0x21, 0x44, 0x00, 0x00,
>-  0x74, 0x40, 0x02, 0x4a, 0x00, 0x00,
>-  0x00, 0x9c, 0x16, 0x84, 0x00, 0x00,
>-  0x01, 0x2d, 0xb0, 0x40, 0x00, 0x00,
>-  0x44, 0x93, 0x05, 0x18, 0x00, 0x00,
>-  0x88, 0x50, 0x48, 0x94, 0x00, 0x00,
>-  0x20, 0xa4, 0x70, 0x30, 0x00, 0x00,
>-  0xaa, 0x04, 0x54, 0x4a, 0x00, 0x00,
>-  0x02, 0x63, 0x09, 0x24, 0x00, 0x00,
>-  0x62, 0x22, 0xaa, 0xa2, 0x00, 0x00,
>-  0x24, 0x44, 0x44, 0x54, 0x00, 0x00,
>-  0xc0, 0x50, 0x0b, 0x0a, 0x00, 0x00,
>-  0x03, 0x0c, 0x12, 0x94, 0x00, 0x00,
>-  0x16, 0x29, 0x08, 0x64, 0x00, 0x00,
>-  0x89, 0x01, 0x80, 0x1a, 0x00, 0x00,
>-  0x82, 0x90, 0x41, 0x4c, 0x00, 0x00,
>-  0x08, 0xa4, 0x34, 0x12, 0x00, 0x00,
>-  0x90, 0x48, 0x88, 0xc8, 0x00, 0x00,
>-  0x9a, 0xd4, 0x6a, 0x36, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom31_21[126] = {
>-  0xc0, 0xa0, 0x8a, 0xa2, 0x00, 0x00,
>-  0x15, 0x56, 0x21, 0x44, 0x00, 0x00,
>-  0x74, 0x40, 0x02, 0x4a, 0x00, 0x00,
>-  0x00, 0x9c, 0x16, 0x84, 0x00, 0x00,
>-  0x01, 0x2d, 0xb0, 0x40, 0x00, 0x00,
>-  0x44, 0x93, 0x05, 0x18, 0x00, 0x00,
>-  0x88, 0x50, 0x48, 0x94, 0x00, 0x00,
>-  0x20, 0xa4, 0x70, 0x30, 0x00, 0x00,
>-  0xaa, 0x04, 0x54, 0x4a, 0x00, 0x00,
>-  0x02, 0x63, 0x09, 0x24, 0x00, 0x00,
>-  0x62, 0x22, 0xaa, 0xaa, 0x00, 0x00,
>-  0xf1, 0x10, 0x54, 0x44, 0x00, 0x00,
>-  0x10, 0x0e, 0x62, 0x22, 0x00, 0x00,
>-  0x10, 0xb1, 0x06, 0x84, 0x00, 0x00,
>-  0x24, 0x24, 0x0d, 0x30, 0x00, 0x00,
>-  0x01, 0x12, 0x81, 0xc2, 0x00, 0x00,
>-  0x00, 0xc4, 0x58, 0x88, 0x00, 0x00,
>-  0x04, 0xa3, 0xb0, 0x50, 0x00, 0x00,
>-  0x02, 0x59, 0x25, 0x02, 0x00, 0x00,
>-  0x2b, 0x01, 0x08, 0x64, 0x00, 0x00,
>-  0x98, 0x40, 0xd0, 0x18, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom31_22[132] = {
>-  0x62, 0x22, 0xaa, 0xaa, 0x00, 0x00,
>-  0xf1, 0x10, 0x54, 0x44, 0x00, 0x00,
>-  0x10, 0x0e, 0x62, 0x22, 0x00, 0x00,
>-  0x10, 0xb1, 0x06, 0x84, 0x00, 0x00,
>-  0x24, 0x24, 0x0d, 0x30, 0x00, 0x00,
>-  0x01, 0x12, 0x81, 0xc2, 0x00, 0x00,
>-  0x00, 0xc4, 0x58, 0x88, 0x00, 0x00,
>-  0x04, 0xa3, 0xb0, 0x50, 0x00, 0x00,
>-  0x02, 0x59, 0x25, 0x02, 0x00, 0x00,
>-  0x2b, 0x01, 0x08, 0x64, 0x00, 0x00,
>-  0x98, 0x40, 0xd0, 0x18, 0x00, 0x00,
>-  0xc0, 0xa0, 0x8a, 0xa2, 0x00, 0x00,
>-  0x15, 0x56, 0x21, 0x44, 0x00, 0x00,
>-  0x74, 0x40, 0x02, 0x4a, 0x00, 0x00,
>-  0x00, 0x9c, 0x16, 0x84, 0x00, 0x00,
>-  0x01, 0x2d, 0xb0, 0x40, 0x00, 0x00,
>-  0x44, 0x93, 0x05, 0x18, 0x00, 0x00,
>-  0x88, 0x50, 0x48, 0x94, 0x00, 0x00,
>-  0x20, 0xa4, 0x70, 0x30, 0x00, 0x00,
>-  0xaa, 0x04, 0x54, 0x4a, 0x00, 0x00,
>-  0x02, 0x63, 0x09, 0x24, 0x00, 0x00,
>-  0x32, 0x23, 0x73, 0x8e, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom31_23[138] = {
>-  0x62, 0x22, 0xaa, 0xaa, 0x00, 0x00,
>-  0xf1, 0x10, 0x54, 0x44, 0x00, 0x00,
>-  0x10, 0x0e, 0x62, 0x22, 0x00, 0x00,
>-  0x10, 0xb1, 0x06, 0x84, 0x00, 0x00,
>-  0x24, 0x24, 0x0d, 0x30, 0x00, 0x00,
>-  0x01, 0x12, 0x81, 0xc2, 0x00, 0x00,
>-  0x00, 0xc4, 0x58, 0x88, 0x00, 0x00,
>-  0x04, 0xa3, 0xb0, 0x50, 0x00, 0x00,
>-  0x02, 0x59, 0x25, 0x02, 0x00, 0x00,
>-  0x2b, 0x01, 0x08, 0x64, 0x00, 0x00,
>-  0x98, 0x40, 0xd0, 0x18, 0x00, 0x00,
>-  0x88, 0x91, 0x08, 0x62, 0x00, 0x00,
>-  0x40, 0x54, 0x31, 0x44, 0x00, 0x00,
>-  0x82, 0x62, 0x9c, 0x02, 0x00, 0x00,
>-  0x21, 0xa4, 0x89, 0x90, 0x00, 0x00,
>-  0x10, 0x64, 0x1d, 0x20, 0x00, 0x00,
>-  0x44, 0x0a, 0x41, 0x98, 0x00, 0x00,
>-  0x10, 0xc9, 0x26, 0x80, 0x00, 0x00,
>-  0x4d, 0x2a, 0x5a, 0x20, 0x00, 0x00,
>-  0x38, 0x02, 0x62, 0x88, 0x00, 0x00,
>-  0x17, 0x49, 0x80, 0x46, 0x00, 0x00,
>-  0x90, 0x84, 0x22, 0x4a, 0x00, 0x00,
>-  0x72, 0x15, 0xd1, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom31_24[144] = {
>-  0x88, 0x91, 0x08, 0x62, 0x00, 0x00,
>-  0x40, 0x54, 0x31, 0x44, 0x00, 0x00,
>-  0x82, 0x62, 0x9c, 0x02, 0x00, 0x00,
>-  0x21, 0xa4, 0x89, 0x90, 0x00, 0x00,
>-  0x10, 0x64, 0x1d, 0x20, 0x00, 0x00,
>-  0x44, 0x0a, 0x41, 0x98, 0x00, 0x00,
>-  0x10, 0xc9, 0x26, 0x80, 0x00, 0x00,
>-  0x4d, 0x2a, 0x5a, 0x20, 0x00, 0x00,
>-  0x38, 0x02, 0x62, 0x88, 0x00, 0x00,
>-  0x17, 0x49, 0x80, 0x46, 0x00, 0x00,
>-  0x90, 0x84, 0x22, 0x4a, 0x00, 0x00,
>-  0x72, 0x15, 0xd1, 0x00, 0x00, 0x00,
>-  0x62, 0x22, 0xaa, 0xaa, 0x00, 0x00,
>-  0xf1, 0x10, 0x54, 0x44, 0x00, 0x00,
>-  0x10, 0x0e, 0x62, 0x22, 0x00, 0x00,
>-  0x10, 0xb1, 0x06, 0x84, 0x00, 0x00,
>-  0x24, 0x24, 0x0d, 0x30, 0x00, 0x00,
>-  0x01, 0x12, 0x81, 0xc2, 0x00, 0x00,
>-  0x00, 0xc4, 0x58, 0x88, 0x00, 0x00,
>-  0x04, 0xa3, 0xb0, 0x50, 0x00, 0x00,
>-  0x02, 0x59, 0x25, 0x02, 0x00, 0x00,
>-  0x2b, 0x01, 0x08, 0x64, 0x00, 0x00,
>-  0x98, 0x40, 0xd0, 0x18, 0x00, 0x00,
>-  0xf0, 0xdf, 0x91, 0xb6, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom31_25[150] = {
>-  0x88, 0x91, 0x08, 0x62, 0x00, 0x00,
>-  0x40, 0x54, 0x31, 0x44, 0x00, 0x00,
>-  0x82, 0x62, 0x9c, 0x02, 0x00, 0x00,
>-  0x21, 0xa4, 0x89, 0x90, 0x00, 0x00,
>-  0x10, 0x64, 0x1d, 0x20, 0x00, 0x00,
>-  0x44, 0x0a, 0x41, 0x98, 0x00, 0x00,
>-  0x10, 0xc9, 0x26, 0x80, 0x00, 0x00,
>-  0x4d, 0x2a, 0x5a, 0x20, 0x00, 0x00,
>-  0x38, 0x02, 0x62, 0x88, 0x00, 0x00,
>-  0x17, 0x49, 0x80, 0x46, 0x00, 0x00,
>-  0x90, 0x84, 0x22, 0x4a, 0x00, 0x00,
>-  0x72, 0x15, 0xd1, 0x00, 0x00, 0x00,
>-  0x62, 0xa2, 0x8a, 0x2a, 0x00, 0x00,
>-  0x34, 0x44, 0x44, 0x44, 0x00, 0x00,
>-  0x40, 0x4b, 0x2c, 0x18, 0x00, 0x00,
>-  0xc4, 0x04, 0x18, 0xa0, 0x00, 0x00,
>-  0x08, 0x60, 0xc4, 0x08, 0x00, 0x00,
>-  0x94, 0x12, 0x92, 0x0c, 0x00, 0x00,
>-  0x88, 0xc0, 0x23, 0x04, 0x00, 0x00,
>-  0x21, 0x32, 0x24, 0x70, 0x00, 0x00,
>-  0xc1, 0x40, 0x80, 0xe2, 0x00, 0x00,
>-  0x10, 0x69, 0x51, 0x14, 0x00, 0x00,
>-  0x06, 0x90, 0x11, 0x42, 0x00, 0x00,
>-  0x59, 0x01, 0x41, 0x80, 0x00, 0x00,
>-  0x0a, 0x0d, 0x8a, 0x20, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom31_26[156] = {
>-  0x62, 0xa2, 0x8a, 0x2a, 0x00, 0x00,
>-  0x34, 0x44, 0x44, 0x44, 0x00, 0x00,
>-  0x40, 0x4b, 0x2c, 0x18, 0x00, 0x00,
>-  0xc4, 0x04, 0x18, 0xa0, 0x00, 0x00,
>-  0x08, 0x60, 0xc4, 0x08, 0x00, 0x00,
>-  0x94, 0x12, 0x92, 0x0c, 0x00, 0x00,
>-  0x88, 0xc0, 0x23, 0x04, 0x00, 0x00,
>-  0x21, 0x32, 0x24, 0x70, 0x00, 0x00,
>-  0xc1, 0x40, 0x80, 0xe2, 0x00, 0x00,
>-  0x10, 0x69, 0x51, 0x14, 0x00, 0x00,
>-  0x06, 0x90, 0x11, 0x42, 0x00, 0x00,
>-  0x59, 0x01, 0x41, 0x80, 0x00, 0x00,
>-  0x0a, 0x0d, 0x8a, 0x20, 0x00, 0x00,
>-  0x88, 0x91, 0x08, 0x62, 0x00, 0x00,
>-  0x40, 0x54, 0x31, 0x44, 0x00, 0x00,
>-  0x82, 0x62, 0x9c, 0x02, 0x00, 0x00,
>-  0x21, 0xa4, 0x89, 0x90, 0x00, 0x00,
>-  0x10, 0x64, 0x1d, 0x20, 0x00, 0x00,
>-  0x44, 0x0a, 0x41, 0x98, 0x00, 0x00,
>-  0x10, 0xc9, 0x26, 0x80, 0x00, 0x00,
>-  0x4d, 0x2a, 0x5a, 0x20, 0x00, 0x00,
>-  0x38, 0x02, 0x62, 0x88, 0x00, 0x00,
>-  0x17, 0x49, 0x80, 0x46, 0x00, 0x00,
>-  0x90, 0x84, 0x22, 0x4a, 0x00, 0x00,
>-  0x72, 0x15, 0xd1, 0x00, 0x00, 0x00,
>-  0xc5, 0x75, 0x48, 0xba, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom31_27[162] = {
>-  0x62, 0xa2, 0x8a, 0x2a, 0x00, 0x00,
>-  0x34, 0x44, 0x44, 0x44, 0x00, 0x00,
>-  0x40, 0x4b, 0x2c, 0x18, 0x00, 0x00,
>-  0xc4, 0x04, 0x18, 0xa0, 0x00, 0x00,
>-  0x08, 0x60, 0xc4, 0x08, 0x00, 0x00,
>-  0x94, 0x12, 0x92, 0x0c, 0x00, 0x00,
>-  0x88, 0xc0, 0x23, 0x04, 0x00, 0x00,
>-  0x21, 0x32, 0x24, 0x70, 0x00, 0x00,
>-  0xc1, 0x40, 0x80, 0xe2, 0x00, 0x00,
>-  0x10, 0x69, 0x51, 0x14, 0x00, 0x00,
>-  0x06, 0x90, 0x11, 0x42, 0x00, 0x00,
>-  0x59, 0x01, 0x41, 0x80, 0x00, 0x00,
>-  0x0a, 0x0d, 0x8a, 0x20, 0x00, 0x00,
>-  0x40, 0x82, 0x8a, 0xa2, 0x00, 0x00,
>-  0x15, 0x54, 0x44, 0x14, 0x00, 0x00,
>-  0x88, 0x13, 0x09, 0xa0, 0x00, 0x00,
>-  0xc0, 0x10, 0x19, 0x14, 0x00, 0x00,
>-  0x80, 0xa0, 0x30, 0x0c, 0x00, 0x00,
>-  0x01, 0x22, 0x60, 0x06, 0x00, 0x00,
>-  0x40, 0x2c, 0xc2, 0x10, 0x00, 0x00,
>-  0x22, 0x02, 0x80, 0x22, 0x00, 0x00,
>-  0x90, 0x04, 0x20, 0x58, 0x00, 0x00,
>-  0x12, 0x40, 0x12, 0xc0, 0x00, 0x00,
>-  0x5d, 0x00, 0x01, 0x28, 0x00, 0x00,
>-  0x20, 0x54, 0xa4, 0x80, 0x00, 0x00,
>-  0x86, 0x09, 0x48, 0x48, 0x00, 0x00,
>-  0x28, 0x89, 0x05, 0x10, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom31_28[168] = {
>-  0x40, 0x82, 0x8a, 0xa2, 0x00, 0x00,
>-  0x15, 0x54, 0x44, 0x14, 0x00, 0x00,
>-  0x88, 0x13, 0x09, 0xa0, 0x00, 0x00,
>-  0xc0, 0x10, 0x19, 0x14, 0x00, 0x00,
>-  0x80, 0xa0, 0x30, 0x0c, 0x00, 0x00,
>-  0x01, 0x22, 0x60, 0x06, 0x00, 0x00,
>-  0x40, 0x2c, 0xc2, 0x10, 0x00, 0x00,
>-  0x22, 0x02, 0x80, 0x22, 0x00, 0x00,
>-  0x90, 0x04, 0x20, 0x58, 0x00, 0x00,
>-  0x12, 0x40, 0x12, 0xc0, 0x00, 0x00,
>-  0x5d, 0x00, 0x01, 0x28, 0x00, 0x00,
>-  0x20, 0x54, 0xa4, 0x80, 0x00, 0x00,
>-  0x86, 0x09, 0x48, 0x48, 0x00, 0x00,
>-  0x28, 0x89, 0x05, 0x10, 0x00, 0x00,
>-  0x62, 0xa2, 0x8a, 0x2a, 0x00, 0x00,
>-  0x34, 0x44, 0x44, 0x44, 0x00, 0x00,
>-  0x40, 0x4b, 0x2c, 0x18, 0x00, 0x00,
>-  0xc4, 0x04, 0x18, 0xa0, 0x00, 0x00,
>-  0x08, 0x60, 0xc4, 0x08, 0x00, 0x00,
>-  0x94, 0x12, 0x92, 0x0c, 0x00, 0x00,
>-  0x88, 0xc0, 0x23, 0x04, 0x00, 0x00,
>-  0x21, 0x32, 0x24, 0x70, 0x00, 0x00,
>-  0xc1, 0x40, 0x80, 0xe2, 0x00, 0x00,
>-  0x10, 0x69, 0x51, 0x14, 0x00, 0x00,
>-  0x06, 0x90, 0x11, 0x42, 0x00, 0x00,
>-  0x59, 0x01, 0x41, 0x80, 0x00, 0x00,
>-  0x0a, 0x0d, 0x8a, 0x20, 0x00, 0x00,
>-  0xbc, 0x0d, 0xca, 0x28, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom31_29[174] = {
>-  0x40, 0x82, 0x8a, 0xa2, 0x00, 0x00,
>-  0x15, 0x54, 0x44, 0x14, 0x00, 0x00,
>-  0x88, 0x13, 0x09, 0xa0, 0x00, 0x00,
>-  0xc0, 0x10, 0x19, 0x14, 0x00, 0x00,
>-  0x80, 0xa0, 0x30, 0x0c, 0x00, 0x00,
>-  0x01, 0x22, 0x60, 0x06, 0x00, 0x00,
>-  0x40, 0x2c, 0xc2, 0x10, 0x00, 0x00,
>-  0x22, 0x02, 0x80, 0x22, 0x00, 0x00,
>-  0x90, 0x04, 0x20, 0x58, 0x00, 0x00,
>-  0x12, 0x40, 0x12, 0xc0, 0x00, 0x00,
>-  0x5d, 0x00, 0x01, 0x28, 0x00, 0x00,
>-  0x20, 0x54, 0xa4, 0x80, 0x00, 0x00,
>-  0x86, 0x09, 0x48, 0x48, 0x00, 0x00,
>-  0x28, 0x89, 0x05, 0x10, 0x00, 0x00,
>-  0x62, 0x22, 0xaa, 0x22, 0x00, 0x00,
>-  0x31, 0x10, 0x44, 0x44, 0x00, 0x00,
>-  0x58, 0x00, 0x22, 0x22, 0x00, 0x00,
>-  0x01, 0x13, 0x00, 0x8a, 0x00, 0x00,
>-  0x88, 0x20, 0x40, 0x34, 0x00, 0x00,
>-  0x44, 0x02, 0x10, 0xd0, 0x00, 0x00,
>-  0x29, 0x04, 0x45, 0x08, 0x00, 0x00,
>-  0x82, 0xa0, 0x90, 0x12, 0x00, 0x00,
>-  0x0a, 0x1a, 0x0e, 0x02, 0x00, 0x00,
>-  0x11, 0xe1, 0x28, 0x40, 0x00, 0x00,
>-  0x84, 0x05, 0x04, 0x0c, 0x00, 0x00,
>-  0x86, 0x40, 0xc0, 0x90, 0x00, 0x00,
>-  0x00, 0x87, 0x13, 0x00, 0x00, 0x00,
>-  0x44, 0x48, 0x01, 0x1c, 0x00, 0x00,
>-  0x10, 0x98, 0x30, 0x44, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom31_3[18] = {
>-  0xac, 0x93, 0x5a, 0x5a, 0x00, 0x00,
>-  0x55, 0x4a, 0xec, 0x6c, 0x00, 0x00,
>-  0x43, 0x36, 0x4d, 0xb6, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom31_30[180] = {
>-  0x62, 0x22, 0xaa, 0x22, 0x00, 0x00,
>-  0x31, 0x10, 0x44, 0x44, 0x00, 0x00,
>-  0x58, 0x00, 0x22, 0x22, 0x00, 0x00,
>-  0x01, 0x13, 0x00, 0x8a, 0x00, 0x00,
>-  0x88, 0x20, 0x40, 0x34, 0x00, 0x00,
>-  0x44, 0x02, 0x10, 0xd0, 0x00, 0x00,
>-  0x29, 0x04, 0x45, 0x08, 0x00, 0x00,
>-  0x82, 0xa0, 0x90, 0x12, 0x00, 0x00,
>-  0x0a, 0x1a, 0x0e, 0x02, 0x00, 0x00,
>-  0x11, 0xe1, 0x28, 0x40, 0x00, 0x00,
>-  0x84, 0x05, 0x04, 0x0c, 0x00, 0x00,
>-  0x86, 0x40, 0xc0, 0x90, 0x00, 0x00,
>-  0x00, 0x87, 0x13, 0x00, 0x00, 0x00,
>-  0x44, 0x48, 0x01, 0x1c, 0x00, 0x00,
>-  0x10, 0x98, 0x30, 0x44, 0x00, 0x00,
>-  0x40, 0x82, 0x8a, 0xa2, 0x00, 0x00,
>-  0x15, 0x54, 0x44, 0x14, 0x00, 0x00,
>-  0x88, 0x13, 0x09, 0xa0, 0x00, 0x00,
>-  0xc0, 0x10, 0x19, 0x14, 0x00, 0x00,
>-  0x80, 0xa0, 0x30, 0x0c, 0x00, 0x00,
>-  0x01, 0x22, 0x60, 0x06, 0x00, 0x00,
>-  0x40, 0x2c, 0xc2, 0x10, 0x00, 0x00,
>-  0x22, 0x02, 0x80, 0x22, 0x00, 0x00,
>-  0x90, 0x04, 0x20, 0x58, 0x00, 0x00,
>-  0x12, 0x40, 0x12, 0xc0, 0x00, 0x00,
>-  0x5d, 0x00, 0x01, 0x28, 0x00, 0x00,
>-  0x20, 0x54, 0xa4, 0x80, 0x00, 0x00,
>-  0x86, 0x09, 0x48, 0x48, 0x00, 0x00,
>-  0x28, 0x89, 0x05, 0x10, 0x00, 0x00,
>-  0xe1, 0x4f, 0xe0, 0x80, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom31_31[186] = {
>-  0x62, 0x22, 0xaa, 0x22, 0x00, 0x00,
>-  0x31, 0x10, 0x44, 0x44, 0x00, 0x00,
>-  0x58, 0x00, 0x22, 0x22, 0x00, 0x00,
>-  0x01, 0x13, 0x00, 0x8a, 0x00, 0x00,
>-  0x88, 0x20, 0x40, 0x34, 0x00, 0x00,
>-  0x44, 0x02, 0x10, 0xd0, 0x00, 0x00,
>-  0x29, 0x04, 0x45, 0x08, 0x00, 0x00,
>-  0x82, 0xa0, 0x90, 0x12, 0x00, 0x00,
>-  0x0a, 0x1a, 0x0e, 0x02, 0x00, 0x00,
>-  0x11, 0xe1, 0x28, 0x40, 0x00, 0x00,
>-  0x84, 0x05, 0x04, 0x0c, 0x00, 0x00,
>-  0x86, 0x40, 0xc0, 0x90, 0x00, 0x00,
>-  0x00, 0x87, 0x13, 0x00, 0x00, 0x00,
>-  0x44, 0x48, 0x01, 0x1c, 0x00, 0x00,
>-  0x10, 0x98, 0x30, 0x44, 0x00, 0x00,
>-  0x62, 0x23, 0x48, 0x20, 0x00, 0x00,
>-  0x31, 0x10, 0x02, 0x54, 0x00, 0x00,
>-  0x58, 0x00, 0x0c, 0x84, 0x00, 0x00,
>-  0x01, 0x12, 0x10, 0xd0, 0x00, 0x00,
>-  0x88, 0x21, 0x03, 0x20, 0x00, 0x00,
>-  0x44, 0x02, 0x01, 0xe0, 0x00, 0x00,
>-  0x29, 0x04, 0xa0, 0x0a, 0x00, 0x00,
>-  0x82, 0xa0, 0x40, 0xa2, 0x00, 0x00,
>-  0x0a, 0x1a, 0x86, 0x10, 0x00, 0x00,
>-  0x11, 0xe0, 0xd1, 0x00, 0x00, 0x00,
>-  0x84, 0x05, 0x00, 0x16, 0x00, 0x00,
>-  0x86, 0x40, 0x20, 0x98, 0x00, 0x00,
>-  0x00, 0x86, 0x24, 0x60, 0x00, 0x00,
>-  0x44, 0x48, 0x81, 0x0a, 0x00, 0x00,
>-  0x10, 0x98, 0x1c, 0x08, 0x00, 0x00,
>-  0x87, 0x74, 0x30, 0x24, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom31_4[24] = {
>-  0x25, 0xaa, 0xaa, 0xaa, 0x00, 0x00,
>-  0x95, 0x55, 0x55, 0x54, 0x00, 0x00,
>-  0x1a, 0x6a, 0x6a, 0x6a, 0x00, 0x00,
>-  0x43, 0xd5, 0x95, 0x94, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom31_5[30] = {
>-  0x64, 0xa2, 0xaa, 0xaa, 0x00, 0x00,
>-  0x25, 0x54, 0x54, 0x54, 0x00, 0x00,
>-  0x49, 0x68, 0x48, 0x4a, 0x00, 0x00,
>-  0x53, 0x91, 0x09, 0x90, 0x00, 0x00,
>-  0x8e, 0x30, 0x21, 0x6c, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom31_6[36] = {
>-  0x62, 0x8a, 0xa2, 0xa2, 0x00, 0x00,
>-  0x15, 0x54, 0x14, 0x54, 0x00, 0x00,
>-  0x4c, 0x47, 0x44, 0x2a, 0x00, 0x00,
>-  0x52, 0x95, 0x08, 0x94, 0x00, 0x00,
>-  0x23, 0x64, 0x61, 0x24, 0x00, 0x00,
>-  0x8a, 0x58, 0x09, 0x58, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom31_7[42] = {
>-  0x62, 0xa2, 0x8a, 0xa2, 0x00, 0x00,
>-  0xb1, 0x14, 0x44, 0x54, 0x00, 0x00,
>-  0x18, 0x6b, 0x22, 0x22, 0x00, 0x00,
>-  0x44, 0xd4, 0x5c, 0x10, 0x00, 0x00,
>-  0x13, 0x64, 0x90, 0x68, 0x00, 0x00,
>-  0x49, 0x1b, 0x20, 0x52, 0x00, 0x00,
>-  0x86, 0x8c, 0x13, 0x0c, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom31_8[48] = {
>-  0x90, 0x22, 0x40, 0xa8, 0x00, 0x00,
>-  0x09, 0x50, 0x31, 0x10, 0x00, 0x00,
>-  0x00, 0x6b, 0x08, 0x0e, 0x00, 0x00,
>-  0x20, 0x34, 0xc0, 0x90, 0x00, 0x00,
>-  0x14, 0x44, 0x25, 0x04, 0x00, 0x00,
>-  0xc2, 0x11, 0x02, 0x82, 0x00, 0x00,
>-  0x00, 0xc6, 0x80, 0xc4, 0x00, 0x00,
>-  0x65, 0x80, 0x2c, 0x60, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom31_9[54] = {
>-  0x62, 0x22, 0xaa, 0xa2, 0x00, 0x00,
>-  0x24, 0x44, 0x44, 0x54, 0x00, 0x00,
>-  0xc0, 0x50, 0x0b, 0x0a, 0x00, 0x00,
>-  0x03, 0x0c, 0x12, 0x94, 0x00, 0x00,
>-  0x16, 0x29, 0x08, 0x64, 0x00, 0x00,
>-  0x89, 0x01, 0x80, 0x1a, 0x00, 0x00,
>-  0x82, 0x90, 0x41, 0x4c, 0x00, 0x00,
>-  0x08, 0xa4, 0x34, 0x12, 0x00, 0x00,
>-  0x90, 0x48, 0x88, 0xc8, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom32_1[6] = {
>-  0xff, 0xff, 0xff, 0xff, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom32_10[60] = {
>-  0x45, 0x51, 0x45, 0x51, 0x00, 0x00,
>-  0x10, 0xa2, 0x10, 0xa2, 0x00, 0x00,
>-  0x01, 0x25, 0x01, 0x25, 0x00, 0x00,
>-  0x0b, 0x42, 0x0b, 0x42, 0x00, 0x00,
>-  0xd8, 0x20, 0xd8, 0x20, 0x00, 0x00,
>-  0x82, 0x8c, 0x82, 0x8c, 0x00, 0x00,
>-  0x24, 0x4a, 0x24, 0x4a, 0x00, 0x00,
>-  0x38, 0x18, 0x38, 0x18, 0x00, 0x00,
>-  0x2a, 0x25, 0x2a, 0x25, 0x00, 0x00,
>-  0x84, 0x92, 0x84, 0x92, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom32_11[66] = {
>-  0x55, 0x55, 0x55, 0x55, 0x00, 0x00,
>-  0x2a, 0x22, 0x2a, 0x22, 0x00, 0x00,
>-  0x31, 0x11, 0x31, 0x11, 0x00, 0x00,
>-  0x83, 0x42, 0x83, 0x42, 0x00, 0x00,
>-  0x06, 0x98, 0x06, 0x98, 0x00, 0x00,
>-  0x40, 0xe1, 0x40, 0xe1, 0x00, 0x00,
>-  0x2c, 0x44, 0x2c, 0x44, 0x00, 0x00,
>-  0xd8, 0x28, 0xd8, 0x28, 0x00, 0x00,
>-  0x92, 0x81, 0x92, 0x81, 0x00, 0x00,
>-  0x84, 0x32, 0x84, 0x32, 0x00, 0x00,
>-  0x68, 0x0c, 0x68, 0x0c, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom32_12[72] = {
>-  0x84, 0x31, 0x84, 0x31, 0x00, 0x00,
>-  0x18, 0xa2, 0x18, 0xa2, 0x00, 0x00,
>-  0x4e, 0x01, 0x4e, 0x01, 0x00, 0x00,
>-  0x44, 0xc8, 0x44, 0xc8, 0x00, 0x00,
>-  0x0e, 0x90, 0x0e, 0x90, 0x00, 0x00,
>-  0x20, 0xcc, 0x20, 0xcc, 0x00, 0x00,
>-  0x93, 0x40, 0x93, 0x40, 0x00, 0x00,
>-  0x2d, 0x10, 0x2d, 0x10, 0x00, 0x00,
>-  0x31, 0x44, 0x31, 0x44, 0x00, 0x00,
>-  0xc0, 0x23, 0xc0, 0x23, 0x00, 0x00,
>-  0x11, 0x25, 0x11, 0x25, 0x00, 0x00,
>-  0xe8, 0x80, 0xe8, 0x80, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom32_13[78] = {
>-  0x45, 0x15, 0x45, 0x15, 0x00, 0x00,
>-  0x22, 0x22, 0x22, 0x22, 0x00, 0x00,
>-  0x96, 0x0c, 0x96, 0x0c, 0x00, 0x00,
>-  0x0c, 0x50, 0x0c, 0x50, 0x00, 0x00,
>-  0x62, 0x04, 0x62, 0x04, 0x00, 0x00,
>-  0x49, 0x06, 0x49, 0x06, 0x00, 0x00,
>-  0x11, 0x82, 0x11, 0x82, 0x00, 0x00,
>-  0x12, 0x38, 0x12, 0x38, 0x00, 0x00,
>-  0x40, 0x71, 0x40, 0x71, 0x00, 0x00,
>-  0xa8, 0x8a, 0xa8, 0x8a, 0x00, 0x00,
>-  0x08, 0xa1, 0x08, 0xa1, 0x00, 0x00,
>-  0xa0, 0xc0, 0xa0, 0xc0, 0x00, 0x00,
>-  0xc5, 0x10, 0xc5, 0x10, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom32_14[84] = {
>-  0x45, 0x51, 0x45, 0x51, 0x00, 0x00,
>-  0x22, 0x0a, 0x22, 0x0a, 0x00, 0x00,
>-  0x84, 0xd0, 0x84, 0xd0, 0x00, 0x00,
>-  0x0c, 0x8a, 0x0c, 0x8a, 0x00, 0x00,
>-  0x18, 0x06, 0x18, 0x06, 0x00, 0x00,
>-  0x30, 0x03, 0x30, 0x03, 0x00, 0x00,
>-  0x61, 0x08, 0x61, 0x08, 0x00, 0x00,
>-  0x40, 0x11, 0x40, 0x11, 0x00, 0x00,
>-  0x10, 0x2c, 0x10, 0x2c, 0x00, 0x00,
>-  0x09, 0x60, 0x09, 0x60, 0x00, 0x00,
>-  0x00, 0x94, 0x00, 0x94, 0x00, 0x00,
>-  0x52, 0x40, 0x52, 0x40, 0x00, 0x00,
>-  0xa4, 0x24, 0xa4, 0x24, 0x00, 0x00,
>-  0x82, 0x88, 0x82, 0x88, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom32_15[90] = {
>-  0x55, 0x11, 0x55, 0x11, 0x00, 0x00,
>-  0x22, 0x22, 0x22, 0x22, 0x00, 0x00,
>-  0x11, 0x11, 0x11, 0x11, 0x00, 0x00,
>-  0x80, 0x45, 0x80, 0x45, 0x00, 0x00,
>-  0x20, 0x1a, 0x20, 0x1a, 0x00, 0x00,
>-  0x08, 0x68, 0x08, 0x68, 0x00, 0x00,
>-  0x22, 0x84, 0x22, 0x84, 0x00, 0x00,
>-  0x48, 0x09, 0x48, 0x09, 0x00, 0x00,
>-  0x07, 0x01, 0x07, 0x01, 0x00, 0x00,
>-  0x94, 0x20, 0x94, 0x20, 0x00, 0x00,
>-  0x82, 0x06, 0x82, 0x06, 0x00, 0x00,
>-  0x60, 0x48, 0x60, 0x48, 0x00, 0x00,
>-  0x89, 0x80, 0x89, 0x80, 0x00, 0x00,
>-  0x00, 0x8e, 0x00, 0x8e, 0x00, 0x00,
>-  0x18, 0x22, 0x18, 0x22, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom32_16[96] = {
>-  0xa4, 0x10, 0xa4, 0x10, 0x00, 0x00,
>-  0x01, 0x2a, 0x01, 0x2a, 0x00, 0x00,
>-  0x06, 0x42, 0x06, 0x42, 0x00, 0x00,
>-  0x08, 0x68, 0x08, 0x68, 0x00, 0x00,
>-  0x81, 0x90, 0x81, 0x90, 0x00, 0x00,
>-  0x00, 0xf0, 0x00, 0xf0, 0x00, 0x00,
>-  0x50, 0x05, 0x50, 0x05, 0x00, 0x00,
>-  0x20, 0x51, 0x20, 0x51, 0x00, 0x00,
>-  0x43, 0x08, 0x43, 0x08, 0x00, 0x00,
>-  0x68, 0x80, 0x68, 0x80, 0x00, 0x00,
>-  0x80, 0x0b, 0x80, 0x0b, 0x00, 0x00,
>-  0x10, 0x4c, 0x10, 0x4c, 0x00, 0x00,
>-  0x12, 0x30, 0x12, 0x30, 0x00, 0x00,
>-  0x40, 0x85, 0x40, 0x85, 0x00, 0x00,
>-  0x0e, 0x04, 0x0e, 0x04, 0x00, 0x00,
>-  0x18, 0x12, 0x18, 0x12, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom32_17[102] = {
>-  0x20, 0x54, 0x20, 0x54, 0x00, 0x00,
>-  0x18, 0x88, 0x18, 0x88, 0x00, 0x00,
>-  0x84, 0x07, 0x84, 0x07, 0x00, 0x00,
>-  0x60, 0x48, 0x60, 0x48, 0x00, 0x00,
>-  0x12, 0x82, 0x12, 0x82, 0x00, 0x00,
>-  0x81, 0x41, 0x81, 0x41, 0x00, 0x00,
>-  0x40, 0x62, 0x40, 0x62, 0x00, 0x00,
>-  0x16, 0x30, 0x16, 0x30, 0x00, 0x00,
>-  0x55, 0x51, 0x55, 0x51, 0x00, 0x00,
>-  0x22, 0x2a, 0x22, 0x2a, 0x00, 0x00,
>-  0x05, 0x85, 0x05, 0x85, 0x00, 0x00,
>-  0x09, 0x4a, 0x09, 0x4a, 0x00, 0x00,
>-  0x84, 0x32, 0x84, 0x32, 0x00, 0x00,
>-  0xc0, 0x0d, 0xc0, 0x0d, 0x00, 0x00,
>-  0x20, 0xa6, 0x20, 0xa6, 0x00, 0x00,
>-  0x1a, 0x09, 0x1a, 0x09, 0x00, 0x00,
>-  0x44, 0x64, 0x44, 0x64, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom32_18[108] = {
>-  0x55, 0x51, 0x55, 0x51, 0x00, 0x00,
>-  0x22, 0x2a, 0x22, 0x2a, 0x00, 0x00,
>-  0x05, 0x85, 0x05, 0x85, 0x00, 0x00,
>-  0x09, 0x4a, 0x09, 0x4a, 0x00, 0x00,
>-  0x84, 0x32, 0x84, 0x32, 0x00, 0x00,
>-  0xc0, 0x0d, 0xc0, 0x0d, 0x00, 0x00,
>-  0x20, 0xa6, 0x20, 0xa6, 0x00, 0x00,
>-  0x1a, 0x09, 0x1a, 0x09, 0x00, 0x00,
>-  0x44, 0x64, 0x44, 0x64, 0x00, 0x00,
>-  0x20, 0x54, 0x20, 0x54, 0x00, 0x00,
>-  0x18, 0x88, 0x18, 0x88, 0x00, 0x00,
>-  0x84, 0x07, 0x84, 0x07, 0x00, 0x00,
>-  0x60, 0x48, 0x60, 0x48, 0x00, 0x00,
>-  0x12, 0x82, 0x12, 0x82, 0x00, 0x00,
>-  0x81, 0x41, 0x81, 0x41, 0x00, 0x00,
>-  0x40, 0x62, 0x40, 0x62, 0x00, 0x00,
>-  0x16, 0x30, 0x16, 0x30, 0x00, 0x00,
>-  0x1e, 0xb2, 0xd8, 0x53, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom32_19[114] = {
>-  0x55, 0x51, 0x55, 0x51, 0x00, 0x00,
>-  0x22, 0x2a, 0x22, 0x2a, 0x00, 0x00,
>-  0x05, 0x85, 0x05, 0x85, 0x00, 0x00,
>-  0x09, 0x4a, 0x09, 0x4a, 0x00, 0x00,
>-  0x84, 0x32, 0x84, 0x32, 0x00, 0x00,
>-  0xc0, 0x0d, 0xc0, 0x0d, 0x00, 0x00,
>-  0x20, 0xa6, 0x20, 0xa6, 0x00, 0x00,
>-  0x1a, 0x09, 0x1a, 0x09, 0x00, 0x00,
>-  0x44, 0x64, 0x44, 0x64, 0x00, 0x00,
>-  0x45, 0x51, 0x45, 0x51, 0x00, 0x00,
>-  0x10, 0xa2, 0x10, 0xa2, 0x00, 0x00,
>-  0x01, 0x25, 0x01, 0x25, 0x00, 0x00,
>-  0x0b, 0x42, 0x0b, 0x42, 0x00, 0x00,
>-  0xd8, 0x20, 0xd8, 0x20, 0x00, 0x00,
>-  0x82, 0x8c, 0x82, 0x8c, 0x00, 0x00,
>-  0x24, 0x4a, 0x24, 0x4a, 0x00, 0x00,
>-  0x38, 0x18, 0x38, 0x18, 0x00, 0x00,
>-  0x2a, 0x25, 0x2a, 0x25, 0x00, 0x00,
>-  0x84, 0x92, 0x84, 0x92, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom32_2[12] = {
>-  0xae, 0xae, 0xae, 0xae, 0x00, 0x00,
>-  0x79, 0x79, 0x79, 0x79, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom32_20[120] = {
>-  0x45, 0x51, 0x45, 0x51, 0x00, 0x00,
>-  0x10, 0xa2, 0x10, 0xa2, 0x00, 0x00,
>-  0x01, 0x25, 0x01, 0x25, 0x00, 0x00,
>-  0x0b, 0x42, 0x0b, 0x42, 0x00, 0x00,
>-  0xd8, 0x20, 0xd8, 0x20, 0x00, 0x00,
>-  0x82, 0x8c, 0x82, 0x8c, 0x00, 0x00,
>-  0x24, 0x4a, 0x24, 0x4a, 0x00, 0x00,
>-  0x38, 0x18, 0x38, 0x18, 0x00, 0x00,
>-  0x2a, 0x25, 0x2a, 0x25, 0x00, 0x00,
>-  0x84, 0x92, 0x84, 0x92, 0x00, 0x00,
>-  0x55, 0x51, 0x55, 0x51, 0x00, 0x00,
>-  0x22, 0x2a, 0x22, 0x2a, 0x00, 0x00,
>-  0x05, 0x85, 0x05, 0x85, 0x00, 0x00,
>-  0x09, 0x4a, 0x09, 0x4a, 0x00, 0x00,
>-  0x84, 0x32, 0x84, 0x32, 0x00, 0x00,
>-  0xc0, 0x0d, 0xc0, 0x0d, 0x00, 0x00,
>-  0x20, 0xa6, 0x20, 0xa6, 0x00, 0x00,
>-  0x1a, 0x09, 0x1a, 0x09, 0x00, 0x00,
>-  0x44, 0x64, 0x44, 0x64, 0x00, 0x00,
>-  0x96, 0xd3, 0xf6, 0xac, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom32_21[126] = {
>-  0x45, 0x51, 0x45, 0x51, 0x00, 0x00,
>-  0x10, 0xa2, 0x10, 0xa2, 0x00, 0x00,
>-  0x01, 0x25, 0x01, 0x25, 0x00, 0x00,
>-  0x0b, 0x42, 0x0b, 0x42, 0x00, 0x00,
>-  0xd8, 0x20, 0xd8, 0x20, 0x00, 0x00,
>-  0x82, 0x8c, 0x82, 0x8c, 0x00, 0x00,
>-  0x24, 0x4a, 0x24, 0x4a, 0x00, 0x00,
>-  0x38, 0x18, 0x38, 0x18, 0x00, 0x00,
>-  0x2a, 0x25, 0x2a, 0x25, 0x00, 0x00,
>-  0x84, 0x92, 0x84, 0x92, 0x00, 0x00,
>-  0x55, 0x55, 0x55, 0x55, 0x00, 0x00,
>-  0x2a, 0x22, 0x2a, 0x22, 0x00, 0x00,
>-  0x31, 0x11, 0x31, 0x11, 0x00, 0x00,
>-  0x83, 0x42, 0x83, 0x42, 0x00, 0x00,
>-  0x06, 0x98, 0x06, 0x98, 0x00, 0x00,
>-  0x40, 0xe1, 0x40, 0xe1, 0x00, 0x00,
>-  0x2c, 0x44, 0x2c, 0x44, 0x00, 0x00,
>-  0xd8, 0x28, 0xd8, 0x28, 0x00, 0x00,
>-  0x92, 0x81, 0x92, 0x81, 0x00, 0x00,
>-  0x84, 0x32, 0x84, 0x32, 0x00, 0x00,
>-  0x68, 0x0c, 0x68, 0x0c, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom32_22[132] = {
>-  0x55, 0x55, 0x55, 0x55, 0x00, 0x00,
>-  0x2a, 0x22, 0x2a, 0x22, 0x00, 0x00,
>-  0x31, 0x11, 0x31, 0x11, 0x00, 0x00,
>-  0x83, 0x42, 0x83, 0x42, 0x00, 0x00,
>-  0x06, 0x98, 0x06, 0x98, 0x00, 0x00,
>-  0x40, 0xe1, 0x40, 0xe1, 0x00, 0x00,
>-  0x2c, 0x44, 0x2c, 0x44, 0x00, 0x00,
>-  0xd8, 0x28, 0xd8, 0x28, 0x00, 0x00,
>-  0x92, 0x81, 0x92, 0x81, 0x00, 0x00,
>-  0x84, 0x32, 0x84, 0x32, 0x00, 0x00,
>-  0x68, 0x0c, 0x68, 0x0c, 0x00, 0x00,
>-  0x45, 0x51, 0x45, 0x51, 0x00, 0x00,
>-  0x10, 0xa2, 0x10, 0xa2, 0x00, 0x00,
>-  0x01, 0x25, 0x01, 0x25, 0x00, 0x00,
>-  0x0b, 0x42, 0x0b, 0x42, 0x00, 0x00,
>-  0xd8, 0x20, 0xd8, 0x20, 0x00, 0x00,
>-  0x82, 0x8c, 0x82, 0x8c, 0x00, 0x00,
>-  0x24, 0x4a, 0x24, 0x4a, 0x00, 0x00,
>-  0x38, 0x18, 0x38, 0x18, 0x00, 0x00,
>-  0x2a, 0x25, 0x2a, 0x25, 0x00, 0x00,
>-  0x84, 0x92, 0x84, 0x92, 0x00, 0x00,
>-  0xeb, 0xb2, 0x22, 0x89, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom32_23[138] = {
>-  0x55, 0x55, 0x55, 0x55, 0x00, 0x00,
>-  0x2a, 0x22, 0x2a, 0x22, 0x00, 0x00,
>-  0x31, 0x11, 0x31, 0x11, 0x00, 0x00,
>-  0x83, 0x42, 0x83, 0x42, 0x00, 0x00,
>-  0x06, 0x98, 0x06, 0x98, 0x00, 0x00,
>-  0x40, 0xe1, 0x40, 0xe1, 0x00, 0x00,
>-  0x2c, 0x44, 0x2c, 0x44, 0x00, 0x00,
>-  0xd8, 0x28, 0xd8, 0x28, 0x00, 0x00,
>-  0x92, 0x81, 0x92, 0x81, 0x00, 0x00,
>-  0x84, 0x32, 0x84, 0x32, 0x00, 0x00,
>-  0x68, 0x0c, 0x68, 0x0c, 0x00, 0x00,
>-  0x84, 0x31, 0x84, 0x31, 0x00, 0x00,
>-  0x18, 0xa2, 0x18, 0xa2, 0x00, 0x00,
>-  0x4e, 0x01, 0x4e, 0x01, 0x00, 0x00,
>-  0x44, 0xc8, 0x44, 0xc8, 0x00, 0x00,
>-  0x0e, 0x90, 0x0e, 0x90, 0x00, 0x00,
>-  0x20, 0xcc, 0x20, 0xcc, 0x00, 0x00,
>-  0x93, 0x40, 0x93, 0x40, 0x00, 0x00,
>-  0x2d, 0x10, 0x2d, 0x10, 0x00, 0x00,
>-  0x31, 0x44, 0x31, 0x44, 0x00, 0x00,
>-  0xc0, 0x23, 0xc0, 0x23, 0x00, 0x00,
>-  0x11, 0x25, 0x11, 0x25, 0x00, 0x00,
>-  0xe8, 0x80, 0xe8, 0x80, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom32_24[144] = {
>-  0x84, 0x31, 0x84, 0x31, 0x00, 0x00,
>-  0x18, 0xa2, 0x18, 0xa2, 0x00, 0x00,
>-  0x4e, 0x01, 0x4e, 0x01, 0x00, 0x00,
>-  0x44, 0xc8, 0x44, 0xc8, 0x00, 0x00,
>-  0x0e, 0x90, 0x0e, 0x90, 0x00, 0x00,
>-  0x20, 0xcc, 0x20, 0xcc, 0x00, 0x00,
>-  0x93, 0x40, 0x93, 0x40, 0x00, 0x00,
>-  0x2d, 0x10, 0x2d, 0x10, 0x00, 0x00,
>-  0x31, 0x44, 0x31, 0x44, 0x00, 0x00,
>-  0xc0, 0x23, 0xc0, 0x23, 0x00, 0x00,
>-  0x11, 0x25, 0x11, 0x25, 0x00, 0x00,
>-  0xe8, 0x80, 0xe8, 0x80, 0x00, 0x00,
>-  0x55, 0x55, 0x55, 0x55, 0x00, 0x00,
>-  0x2a, 0x22, 0x2a, 0x22, 0x00, 0x00,
>-  0x31, 0x11, 0x31, 0x11, 0x00, 0x00,
>-  0x83, 0x42, 0x83, 0x42, 0x00, 0x00,
>-  0x06, 0x98, 0x06, 0x98, 0x00, 0x00,
>-  0x40, 0xe1, 0x40, 0xe1, 0x00, 0x00,
>-  0x2c, 0x44, 0x2c, 0x44, 0x00, 0x00,
>-  0xd8, 0x28, 0xd8, 0x28, 0x00, 0x00,
>-  0x92, 0x81, 0x92, 0x81, 0x00, 0x00,
>-  0x84, 0x32, 0x84, 0x32, 0x00, 0x00,
>-  0x68, 0x0c, 0x68, 0x0c, 0x00, 0x00,
>-  0xf3, 0x5a, 0x2f, 0x5d, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom32_25[150] = {
>-  0x84, 0x31, 0x84, 0x31, 0x00, 0x00,
>-  0x18, 0xa2, 0x18, 0xa2, 0x00, 0x00,
>-  0x4e, 0x01, 0x4e, 0x01, 0x00, 0x00,
>-  0x44, 0xc8, 0x44, 0xc8, 0x00, 0x00,
>-  0x0e, 0x90, 0x0e, 0x90, 0x00, 0x00,
>-  0x20, 0xcc, 0x20, 0xcc, 0x00, 0x00,
>-  0x93, 0x40, 0x93, 0x40, 0x00, 0x00,
>-  0x2d, 0x10, 0x2d, 0x10, 0x00, 0x00,
>-  0x31, 0x44, 0x31, 0x44, 0x00, 0x00,
>-  0xc0, 0x23, 0xc0, 0x23, 0x00, 0x00,
>-  0x11, 0x25, 0x11, 0x25, 0x00, 0x00,
>-  0xe8, 0x80, 0xe8, 0x80, 0x00, 0x00,
>-  0x45, 0x15, 0x45, 0x15, 0x00, 0x00,
>-  0x22, 0x22, 0x22, 0x22, 0x00, 0x00,
>-  0x96, 0x0c, 0x96, 0x0c, 0x00, 0x00,
>-  0x0c, 0x50, 0x0c, 0x50, 0x00, 0x00,
>-  0x62, 0x04, 0x62, 0x04, 0x00, 0x00,
>-  0x49, 0x06, 0x49, 0x06, 0x00, 0x00,
>-  0x11, 0x82, 0x11, 0x82, 0x00, 0x00,
>-  0x12, 0x38, 0x12, 0x38, 0x00, 0x00,
>-  0x40, 0x71, 0x40, 0x71, 0x00, 0x00,
>-  0xa8, 0x8a, 0xa8, 0x8a, 0x00, 0x00,
>-  0x08, 0xa1, 0x08, 0xa1, 0x00, 0x00,
>-  0xa0, 0xc0, 0xa0, 0xc0, 0x00, 0x00,
>-  0xc5, 0x10, 0xc5, 0x10, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom32_26[156] = {
>-  0x45, 0x15, 0x45, 0x15, 0x00, 0x00,
>-  0x22, 0x22, 0x22, 0x22, 0x00, 0x00,
>-  0x96, 0x0c, 0x96, 0x0c, 0x00, 0x00,
>-  0x0c, 0x50, 0x0c, 0x50, 0x00, 0x00,
>-  0x62, 0x04, 0x62, 0x04, 0x00, 0x00,
>-  0x49, 0x06, 0x49, 0x06, 0x00, 0x00,
>-  0x11, 0x82, 0x11, 0x82, 0x00, 0x00,
>-  0x12, 0x38, 0x12, 0x38, 0x00, 0x00,
>-  0x40, 0x71, 0x40, 0x71, 0x00, 0x00,
>-  0xa8, 0x8a, 0xa8, 0x8a, 0x00, 0x00,
>-  0x08, 0xa1, 0x08, 0xa1, 0x00, 0x00,
>-  0xa0, 0xc0, 0xa0, 0xc0, 0x00, 0x00,
>-  0xc5, 0x10, 0xc5, 0x10, 0x00, 0x00,
>-  0x84, 0x31, 0x84, 0x31, 0x00, 0x00,
>-  0x18, 0xa2, 0x18, 0xa2, 0x00, 0x00,
>-  0x4e, 0x01, 0x4e, 0x01, 0x00, 0x00,
>-  0x44, 0xc8, 0x44, 0xc8, 0x00, 0x00,
>-  0x0e, 0x90, 0x0e, 0x90, 0x00, 0x00,
>-  0x20, 0xcc, 0x20, 0xcc, 0x00, 0x00,
>-  0x93, 0x40, 0x93, 0x40, 0x00, 0x00,
>-  0x2d, 0x10, 0x2d, 0x10, 0x00, 0x00,
>-  0x31, 0x44, 0x31, 0x44, 0x00, 0x00,
>-  0xc0, 0x23, 0xc0, 0x23, 0x00, 0x00,
>-  0x11, 0x25, 0x11, 0x25, 0x00, 0x00,
>-  0xe8, 0x80, 0xe8, 0x80, 0x00, 0x00,
>-  0x52, 0x15, 0x62, 0x0a, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom32_27[162] = {
>-  0x45, 0x15, 0x45, 0x15, 0x00, 0x00,
>-  0x22, 0x22, 0x22, 0x22, 0x00, 0x00,
>-  0x96, 0x0c, 0x96, 0x0c, 0x00, 0x00,
>-  0x0c, 0x50, 0x0c, 0x50, 0x00, 0x00,
>-  0x62, 0x04, 0x62, 0x04, 0x00, 0x00,
>-  0x49, 0x06, 0x49, 0x06, 0x00, 0x00,
>-  0x11, 0x82, 0x11, 0x82, 0x00, 0x00,
>-  0x12, 0x38, 0x12, 0x38, 0x00, 0x00,
>-  0x40, 0x71, 0x40, 0x71, 0x00, 0x00,
>-  0xa8, 0x8a, 0xa8, 0x8a, 0x00, 0x00,
>-  0x08, 0xa1, 0x08, 0xa1, 0x00, 0x00,
>-  0xa0, 0xc0, 0xa0, 0xc0, 0x00, 0x00,
>-  0xc5, 0x10, 0xc5, 0x10, 0x00, 0x00,
>-  0x45, 0x51, 0x45, 0x51, 0x00, 0x00,
>-  0x22, 0x0a, 0x22, 0x0a, 0x00, 0x00,
>-  0x84, 0xd0, 0x84, 0xd0, 0x00, 0x00,
>-  0x0c, 0x8a, 0x0c, 0x8a, 0x00, 0x00,
>-  0x18, 0x06, 0x18, 0x06, 0x00, 0x00,
>-  0x30, 0x03, 0x30, 0x03, 0x00, 0x00,
>-  0x61, 0x08, 0x61, 0x08, 0x00, 0x00,
>-  0x40, 0x11, 0x40, 0x11, 0x00, 0x00,
>-  0x10, 0x2c, 0x10, 0x2c, 0x00, 0x00,
>-  0x09, 0x60, 0x09, 0x60, 0x00, 0x00,
>-  0x00, 0x94, 0x00, 0x94, 0x00, 0x00,
>-  0x52, 0x40, 0x52, 0x40, 0x00, 0x00,
>-  0xa4, 0x24, 0xa4, 0x24, 0x00, 0x00,
>-  0x82, 0x88, 0x82, 0x88, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom32_28[168] = {
>-  0x45, 0x51, 0x45, 0x51, 0x00, 0x00,
>-  0x22, 0x0a, 0x22, 0x0a, 0x00, 0x00,
>-  0x84, 0xd0, 0x84, 0xd0, 0x00, 0x00,
>-  0x0c, 0x8a, 0x0c, 0x8a, 0x00, 0x00,
>-  0x18, 0x06, 0x18, 0x06, 0x00, 0x00,
>-  0x30, 0x03, 0x30, 0x03, 0x00, 0x00,
>-  0x61, 0x08, 0x61, 0x08, 0x00, 0x00,
>-  0x40, 0x11, 0x40, 0x11, 0x00, 0x00,
>-  0x10, 0x2c, 0x10, 0x2c, 0x00, 0x00,
>-  0x09, 0x60, 0x09, 0x60, 0x00, 0x00,
>-  0x00, 0x94, 0x00, 0x94, 0x00, 0x00,
>-  0x52, 0x40, 0x52, 0x40, 0x00, 0x00,
>-  0xa4, 0x24, 0xa4, 0x24, 0x00, 0x00,
>-  0x82, 0x88, 0x82, 0x88, 0x00, 0x00,
>-  0x45, 0x15, 0x45, 0x15, 0x00, 0x00,
>-  0x22, 0x22, 0x22, 0x22, 0x00, 0x00,
>-  0x96, 0x0c, 0x96, 0x0c, 0x00, 0x00,
>-  0x0c, 0x50, 0x0c, 0x50, 0x00, 0x00,
>-  0x62, 0x04, 0x62, 0x04, 0x00, 0x00,
>-  0x49, 0x06, 0x49, 0x06, 0x00, 0x00,
>-  0x11, 0x82, 0x11, 0x82, 0x00, 0x00,
>-  0x12, 0x38, 0x12, 0x38, 0x00, 0x00,
>-  0x40, 0x71, 0x40, 0x71, 0x00, 0x00,
>-  0xa8, 0x8a, 0xa8, 0x8a, 0x00, 0x00,
>-  0x08, 0xa1, 0x08, 0xa1, 0x00, 0x00,
>-  0xa0, 0xc0, 0xa0, 0xc0, 0x00, 0x00,
>-  0xc5, 0x10, 0xc5, 0x10, 0x00, 0x00,
>-  0x7f, 0xe2, 0xbc, 0x01, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom32_29[174] = {
>-  0x45, 0x51, 0x45, 0x51, 0x00, 0x00,
>-  0x22, 0x0a, 0x22, 0x0a, 0x00, 0x00,
>-  0x84, 0xd0, 0x84, 0xd0, 0x00, 0x00,
>-  0x0c, 0x8a, 0x0c, 0x8a, 0x00, 0x00,
>-  0x18, 0x06, 0x18, 0x06, 0x00, 0x00,
>-  0x30, 0x03, 0x30, 0x03, 0x00, 0x00,
>-  0x61, 0x08, 0x61, 0x08, 0x00, 0x00,
>-  0x40, 0x11, 0x40, 0x11, 0x00, 0x00,
>-  0x10, 0x2c, 0x10, 0x2c, 0x00, 0x00,
>-  0x09, 0x60, 0x09, 0x60, 0x00, 0x00,
>-  0x00, 0x94, 0x00, 0x94, 0x00, 0x00,
>-  0x52, 0x40, 0x52, 0x40, 0x00, 0x00,
>-  0xa4, 0x24, 0xa4, 0x24, 0x00, 0x00,
>-  0x82, 0x88, 0x82, 0x88, 0x00, 0x00,
>-  0x55, 0x11, 0x55, 0x11, 0x00, 0x00,
>-  0x22, 0x22, 0x22, 0x22, 0x00, 0x00,
>-  0x11, 0x11, 0x11, 0x11, 0x00, 0x00,
>-  0x80, 0x45, 0x80, 0x45, 0x00, 0x00,
>-  0x20, 0x1a, 0x20, 0x1a, 0x00, 0x00,
>-  0x08, 0x68, 0x08, 0x68, 0x00, 0x00,
>-  0x22, 0x84, 0x22, 0x84, 0x00, 0x00,
>-  0x48, 0x09, 0x48, 0x09, 0x00, 0x00,
>-  0x07, 0x01, 0x07, 0x01, 0x00, 0x00,
>-  0x94, 0x20, 0x94, 0x20, 0x00, 0x00,
>-  0x82, 0x06, 0x82, 0x06, 0x00, 0x00,
>-  0x60, 0x48, 0x60, 0x48, 0x00, 0x00,
>-  0x89, 0x80, 0x89, 0x80, 0x00, 0x00,
>-  0x00, 0x8e, 0x00, 0x8e, 0x00, 0x00,
>-  0x18, 0x22, 0x18, 0x22, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom32_3[18] = {
>-  0xad, 0x2d, 0xad, 0x2d, 0x00, 0x00,
>-  0x76, 0x36, 0x76, 0x36, 0x00, 0x00,
>-  0x26, 0xdb, 0x26, 0xdb, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom32_30[180] = {
>-  0x55, 0x11, 0x55, 0x11, 0x00, 0x00,
>-  0x22, 0x22, 0x22, 0x22, 0x00, 0x00,
>-  0x11, 0x11, 0x11, 0x11, 0x00, 0x00,
>-  0x80, 0x45, 0x80, 0x45, 0x00, 0x00,
>-  0x20, 0x1a, 0x20, 0x1a, 0x00, 0x00,
>-  0x08, 0x68, 0x08, 0x68, 0x00, 0x00,
>-  0x22, 0x84, 0x22, 0x84, 0x00, 0x00,
>-  0x48, 0x09, 0x48, 0x09, 0x00, 0x00,
>-  0x07, 0x01, 0x07, 0x01, 0x00, 0x00,
>-  0x94, 0x20, 0x94, 0x20, 0x00, 0x00,
>-  0x82, 0x06, 0x82, 0x06, 0x00, 0x00,
>-  0x60, 0x48, 0x60, 0x48, 0x00, 0x00,
>-  0x89, 0x80, 0x89, 0x80, 0x00, 0x00,
>-  0x00, 0x8e, 0x00, 0x8e, 0x00, 0x00,
>-  0x18, 0x22, 0x18, 0x22, 0x00, 0x00,
>-  0x45, 0x51, 0x45, 0x51, 0x00, 0x00,
>-  0x22, 0x0a, 0x22, 0x0a, 0x00, 0x00,
>-  0x84, 0xd0, 0x84, 0xd0, 0x00, 0x00,
>-  0x0c, 0x8a, 0x0c, 0x8a, 0x00, 0x00,
>-  0x18, 0x06, 0x18, 0x06, 0x00, 0x00,
>-  0x30, 0x03, 0x30, 0x03, 0x00, 0x00,
>-  0x61, 0x08, 0x61, 0x08, 0x00, 0x00,
>-  0x40, 0x11, 0x40, 0x11, 0x00, 0x00,
>-  0x10, 0x2c, 0x10, 0x2c, 0x00, 0x00,
>-  0x09, 0x60, 0x09, 0x60, 0x00, 0x00,
>-  0x00, 0x94, 0x00, 0x94, 0x00, 0x00,
>-  0x52, 0x40, 0x52, 0x40, 0x00, 0x00,
>-  0xa4, 0x24, 0xa4, 0x24, 0x00, 0x00,
>-  0x82, 0x88, 0x82, 0x88, 0x00, 0x00,
>-  0x1e, 0x27, 0xe2, 0xd8, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom32_31[186] = {
>-  0x55, 0x11, 0x55, 0x11, 0x00, 0x00,
>-  0x22, 0x22, 0x22, 0x22, 0x00, 0x00,
>-  0x11, 0x11, 0x11, 0x11, 0x00, 0x00,
>-  0x80, 0x45, 0x80, 0x45, 0x00, 0x00,
>-  0x20, 0x1a, 0x20, 0x1a, 0x00, 0x00,
>-  0x08, 0x68, 0x08, 0x68, 0x00, 0x00,
>-  0x22, 0x84, 0x22, 0x84, 0x00, 0x00,
>-  0x48, 0x09, 0x48, 0x09, 0x00, 0x00,
>-  0x07, 0x01, 0x07, 0x01, 0x00, 0x00,
>-  0x94, 0x20, 0x94, 0x20, 0x00, 0x00,
>-  0x82, 0x06, 0x82, 0x06, 0x00, 0x00,
>-  0x60, 0x48, 0x60, 0x48, 0x00, 0x00,
>-  0x89, 0x80, 0x89, 0x80, 0x00, 0x00,
>-  0x00, 0x8e, 0x00, 0x8e, 0x00, 0x00,
>-  0x18, 0x22, 0x18, 0x22, 0x00, 0x00,
>-  0xa4, 0x10, 0xa4, 0x10, 0x00, 0x00,
>-  0x01, 0x2a, 0x01, 0x2a, 0x00, 0x00,
>-  0x06, 0x42, 0x06, 0x42, 0x00, 0x00,
>-  0x08, 0x68, 0x08, 0x68, 0x00, 0x00,
>-  0x81, 0x90, 0x81, 0x90, 0x00, 0x00,
>-  0x00, 0xf0, 0x00, 0xf0, 0x00, 0x00,
>-  0x50, 0x05, 0x50, 0x05, 0x00, 0x00,
>-  0x20, 0x51, 0x20, 0x51, 0x00, 0x00,
>-  0x43, 0x08, 0x43, 0x08, 0x00, 0x00,
>-  0x68, 0x80, 0x68, 0x80, 0x00, 0x00,
>-  0x80, 0x0b, 0x80, 0x0b, 0x00, 0x00,
>-  0x10, 0x4c, 0x10, 0x4c, 0x00, 0x00,
>-  0x12, 0x30, 0x12, 0x30, 0x00, 0x00,
>-  0x40, 0x85, 0x40, 0x85, 0x00, 0x00,
>-  0x0e, 0x04, 0x0e, 0x04, 0x00, 0x00,
>-  0x18, 0x12, 0x18, 0x12, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom32_32[192] = {
>-  0xa4, 0x10, 0xa4, 0x10, 0x00, 0x00,
>-  0x01, 0x2a, 0x01, 0x2a, 0x00, 0x00,
>-  0x06, 0x42, 0x06, 0x42, 0x00, 0x00,
>-  0x08, 0x68, 0x08, 0x68, 0x00, 0x00,
>-  0x81, 0x90, 0x81, 0x90, 0x00, 0x00,
>-  0x00, 0xf0, 0x00, 0xf0, 0x00, 0x00,
>-  0x50, 0x05, 0x50, 0x05, 0x00, 0x00,
>-  0x20, 0x51, 0x20, 0x51, 0x00, 0x00,
>-  0x43, 0x08, 0x43, 0x08, 0x00, 0x00,
>-  0x68, 0x80, 0x68, 0x80, 0x00, 0x00,
>-  0x80, 0x0b, 0x80, 0x0b, 0x00, 0x00,
>-  0x10, 0x4c, 0x10, 0x4c, 0x00, 0x00,
>-  0x12, 0x30, 0x12, 0x30, 0x00, 0x00,
>-  0x40, 0x85, 0x40, 0x85, 0x00, 0x00,
>-  0x0e, 0x04, 0x0e, 0x04, 0x00, 0x00,
>-  0x18, 0x12, 0x18, 0x12, 0x00, 0x00,
>-  0x55, 0x11, 0x55, 0x11, 0x00, 0x00,
>-  0x22, 0x22, 0x22, 0x22, 0x00, 0x00,
>-  0x11, 0x11, 0x11, 0x11, 0x00, 0x00,
>-  0x80, 0x45, 0x80, 0x45, 0x00, 0x00,
>-  0x20, 0x1a, 0x20, 0x1a, 0x00, 0x00,
>-  0x08, 0x68, 0x08, 0x68, 0x00, 0x00,
>-  0x22, 0x84, 0x22, 0x84, 0x00, 0x00,
>-  0x48, 0x09, 0x48, 0x09, 0x00, 0x00,
>-  0x07, 0x01, 0x07, 0x01, 0x00, 0x00,
>-  0x94, 0x20, 0x94, 0x20, 0x00, 0x00,
>-  0x82, 0x06, 0x82, 0x06, 0x00, 0x00,
>-  0x60, 0x48, 0x60, 0x48, 0x00, 0x00,
>-  0x89, 0x80, 0x89, 0x80, 0x00, 0x00,
>-  0x00, 0x8e, 0x00, 0x8e, 0x00, 0x00,
>-  0x18, 0x22, 0x18, 0x22, 0x00, 0x00,
>-  0x60, 0xc4, 0x02, 0x02, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom32_4[24] = {
>-  0x55, 0x55, 0x55, 0x55, 0x00, 0x00,
>-  0xaa, 0xaa, 0xaa, 0xaa, 0x00, 0x00,
>-  0x35, 0x35, 0x35, 0x35, 0x00, 0x00,
>-  0xca, 0xca, 0xca, 0xca, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom32_5[30] = {
>-  0x55, 0x55, 0x55, 0x55, 0x00, 0x00,
>-  0x2a, 0x2a, 0x2a, 0x2a, 0x00, 0x00,
>-  0x24, 0x25, 0x24, 0x25, 0x00, 0x00,
>-  0x84, 0xc8, 0x84, 0xc8, 0x00, 0x00,
>-  0x10, 0xb6, 0x10, 0xb6, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom32_6[36] = {
>-  0x51, 0x51, 0x51, 0x51, 0x00, 0x00,
>-  0x0a, 0x2a, 0x0a, 0x2a, 0x00, 0x00,
>-  0xa2, 0x15, 0xa2, 0x15, 0x00, 0x00,
>-  0x84, 0x4a, 0x84, 0x4a, 0x00, 0x00,
>-  0x30, 0x92, 0x30, 0x92, 0x00, 0x00,
>-  0x04, 0xac, 0x04, 0xac, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom32_7[42] = {
>-  0x45, 0x51, 0x45, 0x51, 0x00, 0x00,
>-  0x22, 0x2a, 0x22, 0x2a, 0x00, 0x00,
>-  0x91, 0x11, 0x91, 0x11, 0x00, 0x00,
>-  0x2e, 0x08, 0x2e, 0x08, 0x00, 0x00,
>-  0x48, 0x34, 0x48, 0x34, 0x00, 0x00,
>-  0x90, 0x29, 0x90, 0x29, 0x00, 0x00,
>-  0x09, 0x86, 0x09, 0x86, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom32_8[48] = {
>-  0x20, 0x54, 0x20, 0x54, 0x00, 0x00,
>-  0x18, 0x88, 0x18, 0x88, 0x00, 0x00,
>-  0x84, 0x07, 0x84, 0x07, 0x00, 0x00,
>-  0x60, 0x48, 0x60, 0x48, 0x00, 0x00,
>-  0x12, 0x82, 0x12, 0x82, 0x00, 0x00,
>-  0x81, 0x41, 0x81, 0x41, 0x00, 0x00,
>-  0x40, 0x62, 0x40, 0x62, 0x00, 0x00,
>-  0x16, 0x30, 0x16, 0x30, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom32_9[54] = {
>-  0x55, 0x51, 0x55, 0x51, 0x00, 0x00,
>-  0x22, 0x2a, 0x22, 0x2a, 0x00, 0x00,
>-  0x05, 0x85, 0x05, 0x85, 0x00, 0x00,
>-  0x09, 0x4a, 0x09, 0x4a, 0x00, 0x00,
>-  0x84, 0x32, 0x84, 0x32, 0x00, 0x00,
>-  0xc0, 0x0d, 0xc0, 0x0d, 0x00, 0x00,
>-  0x20, 0xa6, 0x20, 0xa6, 0x00, 0x00,
>-  0x1a, 0x09, 0x1a, 0x09, 0x00, 0x00,
>-  0x44, 0x64, 0x44, 0x64, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom33_1[6] = {
>-  0xff, 0xff, 0xff, 0xff, 0x80, 0x00
>-};
>-
>-const uint8_t kMaskRandom33_10[60] = {
>-  0x45, 0x51, 0x55, 0x8c, 0x80, 0x00,
>-  0x10, 0xa2, 0xaa, 0x27, 0x00, 0x00,
>-  0x01, 0x25, 0xa5, 0x32, 0x80, 0x00,
>-  0x0b, 0x42, 0x62, 0x61, 0x80, 0x00,
>-  0xd8, 0x20, 0x3c, 0x5c, 0x00, 0x00,
>-  0x82, 0x8c, 0x8e, 0xcc, 0x00, 0x00,
>-  0x24, 0x4a, 0x6a, 0x2b, 0x00, 0x00,
>-  0x38, 0x18, 0x36, 0x32, 0x80, 0x00,
>-  0x2a, 0x25, 0xd1, 0x25, 0x80, 0x00,
>-  0x84, 0x92, 0xc8, 0x02, 0x80, 0x00
>-};
>-
>-const uint8_t kMaskRandom33_11[66] = {
>-  0x55, 0x55, 0x55, 0x8c, 0x80, 0x00,
>-  0x2a, 0x22, 0xaa, 0x27, 0x00, 0x00,
>-  0x31, 0x11, 0xa5, 0x32, 0x80, 0x00,
>-  0x83, 0x42, 0x62, 0x61, 0x80, 0x00,
>-  0x06, 0x98, 0x3c, 0x5c, 0x00, 0x00,
>-  0x40, 0xe1, 0x51, 0x84, 0x80, 0x00,
>-  0x2c, 0x44, 0xa2, 0x27, 0x00, 0x00,
>-  0xd8, 0x28, 0x95, 0x51, 0x80, 0x00,
>-  0x92, 0x81, 0x4a, 0x1a, 0x00, 0x00,
>-  0x84, 0x32, 0x30, 0x68, 0x00, 0x00,
>-  0x68, 0x0c, 0x2c, 0x89, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom33_12[72] = {
>-  0x84, 0x31, 0x51, 0x84, 0x80, 0x00,
>-  0x18, 0xa2, 0xa2, 0x27, 0x00, 0x00,
>-  0x4e, 0x01, 0x95, 0x51, 0x80, 0x00,
>-  0x44, 0xc8, 0x4a, 0x1a, 0x00, 0x00,
>-  0x0e, 0x90, 0x30, 0x68, 0x00, 0x00,
>-  0x20, 0xcc, 0x2c, 0x89, 0x00, 0x00,
>-  0x93, 0x40, 0x55, 0x8c, 0x80, 0x00,
>-  0x2d, 0x10, 0xaa, 0x27, 0x00, 0x00,
>-  0x31, 0x44, 0xa5, 0x32, 0x80, 0x00,
>-  0xc0, 0x23, 0x62, 0x61, 0x80, 0x00,
>-  0x11, 0x25, 0x3c, 0x5c, 0x00, 0x00,
>-  0xe8, 0x80, 0x51, 0x35, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom33_13[78] = {
>-  0x45, 0x15, 0x51, 0x84, 0x80, 0x00,
>-  0x22, 0x22, 0xa2, 0x27, 0x00, 0x00,
>-  0x96, 0x0c, 0x95, 0x51, 0x80, 0x00,
>-  0x0c, 0x50, 0x4a, 0x1a, 0x00, 0x00,
>-  0x62, 0x04, 0x30, 0x68, 0x00, 0x00,
>-  0x49, 0x06, 0x2c, 0x89, 0x00, 0x00,
>-  0x11, 0x82, 0x15, 0x8c, 0x00, 0x00,
>-  0x12, 0x38, 0x8a, 0x47, 0x00, 0x00,
>-  0x40, 0x71, 0x25, 0x81, 0x80, 0x00,
>-  0xa8, 0x8a, 0x62, 0x12, 0x80, 0x00,
>-  0x08, 0xa1, 0x58, 0x58, 0x00, 0x00,
>-  0xa0, 0xc0, 0x0e, 0x28, 0x80, 0x00,
>-  0xc5, 0x10, 0x83, 0x34, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom33_14[84] = {
>-  0x45, 0x51, 0x15, 0x8c, 0x00, 0x00,
>-  0x22, 0x0a, 0x8a, 0x47, 0x00, 0x00,
>-  0x84, 0xd0, 0x25, 0x81, 0x80, 0x00,
>-  0x0c, 0x8a, 0x62, 0x12, 0x80, 0x00,
>-  0x18, 0x06, 0x58, 0x58, 0x00, 0x00,
>-  0x30, 0x03, 0x0e, 0x28, 0x80, 0x00,
>-  0x61, 0x08, 0x83, 0x34, 0x00, 0x00,
>-  0x40, 0x11, 0x51, 0x84, 0x80, 0x00,
>-  0x10, 0x2c, 0xa2, 0x27, 0x00, 0x00,
>-  0x09, 0x60, 0x95, 0x51, 0x80, 0x00,
>-  0x00, 0x94, 0x4a, 0x1a, 0x00, 0x00,
>-  0x52, 0x40, 0x30, 0x68, 0x00, 0x00,
>-  0xa4, 0x24, 0x2c, 0x89, 0x00, 0x00,
>-  0x82, 0x88, 0xb0, 0xde, 0x80, 0x00
>-};
>-
>-const uint8_t kMaskRandom33_15[90] = {
>-  0x55, 0x11, 0x15, 0x8c, 0x00, 0x00,
>-  0x22, 0x22, 0x8a, 0x47, 0x00, 0x00,
>-  0x11, 0x11, 0x25, 0x81, 0x80, 0x00,
>-  0x80, 0x45, 0x62, 0x12, 0x80, 0x00,
>-  0x20, 0x1a, 0x58, 0x58, 0x00, 0x00,
>-  0x08, 0x68, 0x0e, 0x28, 0x80, 0x00,
>-  0x22, 0x84, 0x83, 0x34, 0x00, 0x00,
>-  0x48, 0x09, 0x25, 0x2c, 0x00, 0x00,
>-  0x07, 0x01, 0x8a, 0x91, 0x00, 0x00,
>-  0x94, 0x20, 0x91, 0xc0, 0x80, 0x00,
>-  0x82, 0x06, 0x68, 0x06, 0x80, 0x00,
>-  0x60, 0x48, 0x32, 0xc8, 0x00, 0x00,
>-  0x89, 0x80, 0x43, 0x45, 0x00, 0x00,
>-  0x00, 0x8e, 0xc4, 0x30, 0x80, 0x00,
>-  0x18, 0x22, 0x1c, 0xa2, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom33_16[96] = {
>-  0xa4, 0x10, 0x25, 0x2c, 0x00, 0x00,
>-  0x01, 0x2a, 0x8a, 0x91, 0x00, 0x00,
>-  0x06, 0x42, 0x91, 0xc0, 0x80, 0x00,
>-  0x08, 0x68, 0x68, 0x06, 0x80, 0x00,
>-  0x81, 0x90, 0x32, 0xc8, 0x00, 0x00,
>-  0x00, 0xf0, 0x43, 0x45, 0x00, 0x00,
>-  0x50, 0x05, 0xc4, 0x30, 0x80, 0x00,
>-  0x20, 0x51, 0x1c, 0xa2, 0x00, 0x00,
>-  0x43, 0x08, 0x15, 0x8c, 0x00, 0x00,
>-  0x68, 0x80, 0x8a, 0x47, 0x00, 0x00,
>-  0x80, 0x0b, 0x25, 0x81, 0x80, 0x00,
>-  0x10, 0x4c, 0x62, 0x12, 0x80, 0x00,
>-  0x12, 0x30, 0x58, 0x58, 0x00, 0x00,
>-  0x40, 0x85, 0x0e, 0x28, 0x80, 0x00,
>-  0x0e, 0x04, 0x83, 0x34, 0x00, 0x00,
>-  0x18, 0x12, 0x0a, 0x1c, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom33_17[102] = {
>-  0x20, 0x54, 0x64, 0x16, 0x00, 0x00,
>-  0x18, 0x88, 0xa2, 0xc2, 0x00, 0x00,
>-  0x84, 0x07, 0x51, 0x60, 0x80, 0x00,
>-  0x60, 0x48, 0x4a, 0x85, 0x00, 0x00,
>-  0x12, 0x82, 0x38, 0x4c, 0x00, 0x00,
>-  0x81, 0x41, 0x89, 0x29, 0x00, 0x00,
>-  0x40, 0x62, 0x07, 0x11, 0x80, 0x00,
>-  0x16, 0x30, 0x94, 0xb0, 0x00, 0x00,
>-  0x55, 0x51, 0x8e, 0xcc, 0x00, 0x00,
>-  0x22, 0x2a, 0x6a, 0x2b, 0x00, 0x00,
>-  0x05, 0x85, 0x36, 0x32, 0x80, 0x00,
>-  0x09, 0x4a, 0xd1, 0x25, 0x80, 0x00,
>-  0x84, 0x32, 0x55, 0x8c, 0x80, 0x00,
>-  0xc0, 0x0d, 0xaa, 0x27, 0x00, 0x00,
>-  0x20, 0xa6, 0xa5, 0x32, 0x80, 0x00,
>-  0x1a, 0x09, 0x62, 0x61, 0x80, 0x00,
>-  0x44, 0x64, 0x3c, 0x5c, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom33_18[108] = {
>-  0x55, 0x51, 0x8e, 0xcc, 0x00, 0x00,
>-  0x22, 0x2a, 0x6a, 0x2b, 0x00, 0x00,
>-  0x05, 0x85, 0x36, 0x32, 0x80, 0x00,
>-  0x09, 0x4a, 0xd1, 0x25, 0x80, 0x00,
>-  0x84, 0x32, 0x55, 0x8c, 0x80, 0x00,
>-  0xc0, 0x0d, 0xaa, 0x27, 0x00, 0x00,
>-  0x20, 0xa6, 0xa5, 0x32, 0x80, 0x00,
>-  0x1a, 0x09, 0x62, 0x61, 0x80, 0x00,
>-  0x44, 0x64, 0x3c, 0x5c, 0x00, 0x00,
>-  0x20, 0x54, 0x64, 0x16, 0x00, 0x00,
>-  0x18, 0x88, 0xa2, 0xc2, 0x00, 0x00,
>-  0x84, 0x07, 0x51, 0x60, 0x80, 0x00,
>-  0x60, 0x48, 0x4a, 0x85, 0x00, 0x00,
>-  0x12, 0x82, 0x38, 0x4c, 0x00, 0x00,
>-  0x81, 0x41, 0x89, 0x29, 0x00, 0x00,
>-  0x40, 0x62, 0x07, 0x11, 0x80, 0x00,
>-  0x16, 0x30, 0x94, 0xb0, 0x00, 0x00,
>-  0x89, 0x53, 0x03, 0xad, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom33_19[114] = {
>-  0x55, 0x51, 0x8e, 0xcc, 0x00, 0x00,
>-  0x22, 0x2a, 0x6a, 0x2b, 0x00, 0x00,
>-  0x05, 0x85, 0x36, 0x32, 0x80, 0x00,
>-  0x09, 0x4a, 0xd1, 0x25, 0x80, 0x00,
>-  0x84, 0x32, 0x55, 0x8c, 0x80, 0x00,
>-  0xc0, 0x0d, 0xaa, 0x27, 0x00, 0x00,
>-  0x20, 0xa6, 0xa5, 0x32, 0x80, 0x00,
>-  0x1a, 0x09, 0x62, 0x61, 0x80, 0x00,
>-  0x44, 0x64, 0x3c, 0x5c, 0x00, 0x00,
>-  0x45, 0x51, 0x55, 0x8c, 0x80, 0x00,
>-  0x10, 0xa2, 0xaa, 0x27, 0x00, 0x00,
>-  0x01, 0x25, 0xa5, 0x32, 0x80, 0x00,
>-  0x0b, 0x42, 0x62, 0x61, 0x80, 0x00,
>-  0xd8, 0x20, 0x3c, 0x5c, 0x00, 0x00,
>-  0x82, 0x8c, 0x8e, 0xcc, 0x00, 0x00,
>-  0x24, 0x4a, 0x6a, 0x2b, 0x00, 0x00,
>-  0x38, 0x18, 0x36, 0x32, 0x80, 0x00,
>-  0x2a, 0x25, 0xd1, 0x25, 0x80, 0x00,
>-  0x84, 0x92, 0xc8, 0x02, 0x80, 0x00
>-};
>-
>-const uint8_t kMaskRandom33_2[12] = {
>-  0xae, 0xae, 0xce, 0xce, 0x00, 0x00,
>-  0x79, 0x79, 0xb9, 0x39, 0x80, 0x00
>-};
>-
>-const uint8_t kMaskRandom33_20[120] = {
>-  0x45, 0x51, 0x55, 0x8c, 0x80, 0x00,
>-  0x10, 0xa2, 0xaa, 0x27, 0x00, 0x00,
>-  0x01, 0x25, 0xa5, 0x32, 0x80, 0x00,
>-  0x0b, 0x42, 0x62, 0x61, 0x80, 0x00,
>-  0xd8, 0x20, 0x3c, 0x5c, 0x00, 0x00,
>-  0x82, 0x8c, 0x8e, 0xcc, 0x00, 0x00,
>-  0x24, 0x4a, 0x6a, 0x2b, 0x00, 0x00,
>-  0x38, 0x18, 0x36, 0x32, 0x80, 0x00,
>-  0x2a, 0x25, 0xd1, 0x25, 0x80, 0x00,
>-  0x84, 0x92, 0xc8, 0x02, 0x80, 0x00,
>-  0x55, 0x51, 0x8e, 0xcc, 0x00, 0x00,
>-  0x22, 0x2a, 0x6a, 0x2b, 0x00, 0x00,
>-  0x05, 0x85, 0x36, 0x32, 0x80, 0x00,
>-  0x09, 0x4a, 0xd1, 0x25, 0x80, 0x00,
>-  0x84, 0x32, 0x55, 0x8c, 0x80, 0x00,
>-  0xc0, 0x0d, 0xaa, 0x27, 0x00, 0x00,
>-  0x20, 0xa6, 0xa5, 0x32, 0x80, 0x00,
>-  0x1a, 0x09, 0x62, 0x61, 0x80, 0x00,
>-  0x44, 0x64, 0x3c, 0x5c, 0x00, 0x00,
>-  0x73, 0x5f, 0x5b, 0x0e, 0x80, 0x00
>-};
>-
>-const uint8_t kMaskRandom33_21[126] = {
>-  0x45, 0x51, 0x55, 0x8c, 0x80, 0x00,
>-  0x10, 0xa2, 0xaa, 0x27, 0x00, 0x00,
>-  0x01, 0x25, 0xa5, 0x32, 0x80, 0x00,
>-  0x0b, 0x42, 0x62, 0x61, 0x80, 0x00,
>-  0xd8, 0x20, 0x3c, 0x5c, 0x00, 0x00,
>-  0x82, 0x8c, 0x8e, 0xcc, 0x00, 0x00,
>-  0x24, 0x4a, 0x6a, 0x2b, 0x00, 0x00,
>-  0x38, 0x18, 0x36, 0x32, 0x80, 0x00,
>-  0x2a, 0x25, 0xd1, 0x25, 0x80, 0x00,
>-  0x84, 0x92, 0xc8, 0x02, 0x80, 0x00,
>-  0x55, 0x55, 0x55, 0x8c, 0x80, 0x00,
>-  0x2a, 0x22, 0xaa, 0x27, 0x00, 0x00,
>-  0x31, 0x11, 0xa5, 0x32, 0x80, 0x00,
>-  0x83, 0x42, 0x62, 0x61, 0x80, 0x00,
>-  0x06, 0x98, 0x3c, 0x5c, 0x00, 0x00,
>-  0x40, 0xe1, 0x51, 0x84, 0x80, 0x00,
>-  0x2c, 0x44, 0xa2, 0x27, 0x00, 0x00,
>-  0xd8, 0x28, 0x95, 0x51, 0x80, 0x00,
>-  0x92, 0x81, 0x4a, 0x1a, 0x00, 0x00,
>-  0x84, 0x32, 0x30, 0x68, 0x00, 0x00,
>-  0x68, 0x0c, 0x2c, 0x89, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom33_22[132] = {
>-  0x55, 0x55, 0x55, 0x8c, 0x80, 0x00,
>-  0x2a, 0x22, 0xaa, 0x27, 0x00, 0x00,
>-  0x31, 0x11, 0xa5, 0x32, 0x80, 0x00,
>-  0x83, 0x42, 0x62, 0x61, 0x80, 0x00,
>-  0x06, 0x98, 0x3c, 0x5c, 0x00, 0x00,
>-  0x40, 0xe1, 0x51, 0x84, 0x80, 0x00,
>-  0x2c, 0x44, 0xa2, 0x27, 0x00, 0x00,
>-  0xd8, 0x28, 0x95, 0x51, 0x80, 0x00,
>-  0x92, 0x81, 0x4a, 0x1a, 0x00, 0x00,
>-  0x84, 0x32, 0x30, 0x68, 0x00, 0x00,
>-  0x68, 0x0c, 0x2c, 0x89, 0x00, 0x00,
>-  0x45, 0x51, 0x55, 0x8c, 0x80, 0x00,
>-  0x10, 0xa2, 0xaa, 0x27, 0x00, 0x00,
>-  0x01, 0x25, 0xa5, 0x32, 0x80, 0x00,
>-  0x0b, 0x42, 0x62, 0x61, 0x80, 0x00,
>-  0xd8, 0x20, 0x3c, 0x5c, 0x00, 0x00,
>-  0x82, 0x8c, 0x8e, 0xcc, 0x00, 0x00,
>-  0x24, 0x4a, 0x6a, 0x2b, 0x00, 0x00,
>-  0x38, 0x18, 0x36, 0x32, 0x80, 0x00,
>-  0x2a, 0x25, 0xd1, 0x25, 0x80, 0x00,
>-  0x84, 0x92, 0xc8, 0x02, 0x80, 0x00,
>-  0xcc, 0xe3, 0x42, 0x6b, 0x80, 0x00
>-};
>-
>-const uint8_t kMaskRandom33_23[138] = {
>-  0x55, 0x55, 0x55, 0x8c, 0x80, 0x00,
>-  0x2a, 0x22, 0xaa, 0x27, 0x00, 0x00,
>-  0x31, 0x11, 0xa5, 0x32, 0x80, 0x00,
>-  0x83, 0x42, 0x62, 0x61, 0x80, 0x00,
>-  0x06, 0x98, 0x3c, 0x5c, 0x00, 0x00,
>-  0x40, 0xe1, 0x51, 0x84, 0x80, 0x00,
>-  0x2c, 0x44, 0xa2, 0x27, 0x00, 0x00,
>-  0xd8, 0x28, 0x95, 0x51, 0x80, 0x00,
>-  0x92, 0x81, 0x4a, 0x1a, 0x00, 0x00,
>-  0x84, 0x32, 0x30, 0x68, 0x00, 0x00,
>-  0x68, 0x0c, 0x2c, 0x89, 0x00, 0x00,
>-  0x84, 0x31, 0x51, 0x84, 0x80, 0x00,
>-  0x18, 0xa2, 0xa2, 0x27, 0x00, 0x00,
>-  0x4e, 0x01, 0x95, 0x51, 0x80, 0x00,
>-  0x44, 0xc8, 0x4a, 0x1a, 0x00, 0x00,
>-  0x0e, 0x90, 0x30, 0x68, 0x00, 0x00,
>-  0x20, 0xcc, 0x2c, 0x89, 0x00, 0x00,
>-  0x93, 0x40, 0x55, 0x8c, 0x80, 0x00,
>-  0x2d, 0x10, 0xaa, 0x27, 0x00, 0x00,
>-  0x31, 0x44, 0xa5, 0x32, 0x80, 0x00,
>-  0xc0, 0x23, 0x62, 0x61, 0x80, 0x00,
>-  0x11, 0x25, 0x3c, 0x5c, 0x00, 0x00,
>-  0xe8, 0x80, 0x51, 0x35, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom33_24[144] = {
>-  0x84, 0x31, 0x51, 0x84, 0x80, 0x00,
>-  0x18, 0xa2, 0xa2, 0x27, 0x00, 0x00,
>-  0x4e, 0x01, 0x95, 0x51, 0x80, 0x00,
>-  0x44, 0xc8, 0x4a, 0x1a, 0x00, 0x00,
>-  0x0e, 0x90, 0x30, 0x68, 0x00, 0x00,
>-  0x20, 0xcc, 0x2c, 0x89, 0x00, 0x00,
>-  0x93, 0x40, 0x55, 0x8c, 0x80, 0x00,
>-  0x2d, 0x10, 0xaa, 0x27, 0x00, 0x00,
>-  0x31, 0x44, 0xa5, 0x32, 0x80, 0x00,
>-  0xc0, 0x23, 0x62, 0x61, 0x80, 0x00,
>-  0x11, 0x25, 0x3c, 0x5c, 0x00, 0x00,
>-  0xe8, 0x80, 0x51, 0x35, 0x00, 0x00,
>-  0x55, 0x55, 0x55, 0x8c, 0x80, 0x00,
>-  0x2a, 0x22, 0xaa, 0x27, 0x00, 0x00,
>-  0x31, 0x11, 0xa5, 0x32, 0x80, 0x00,
>-  0x83, 0x42, 0x62, 0x61, 0x80, 0x00,
>-  0x06, 0x98, 0x3c, 0x5c, 0x00, 0x00,
>-  0x40, 0xe1, 0x51, 0x84, 0x80, 0x00,
>-  0x2c, 0x44, 0xa2, 0x27, 0x00, 0x00,
>-  0xd8, 0x28, 0x95, 0x51, 0x80, 0x00,
>-  0x92, 0x81, 0x4a, 0x1a, 0x00, 0x00,
>-  0x84, 0x32, 0x30, 0x68, 0x00, 0x00,
>-  0x68, 0x0c, 0x2c, 0x89, 0x00, 0x00,
>-  0xdc, 0x4e, 0xfc, 0x70, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom33_25[150] = {
>-  0x84, 0x31, 0x51, 0x84, 0x80, 0x00,
>-  0x18, 0xa2, 0xa2, 0x27, 0x00, 0x00,
>-  0x4e, 0x01, 0x95, 0x51, 0x80, 0x00,
>-  0x44, 0xc8, 0x4a, 0x1a, 0x00, 0x00,
>-  0x0e, 0x90, 0x30, 0x68, 0x00, 0x00,
>-  0x20, 0xcc, 0x2c, 0x89, 0x00, 0x00,
>-  0x93, 0x40, 0x55, 0x8c, 0x80, 0x00,
>-  0x2d, 0x10, 0xaa, 0x27, 0x00, 0x00,
>-  0x31, 0x44, 0xa5, 0x32, 0x80, 0x00,
>-  0xc0, 0x23, 0x62, 0x61, 0x80, 0x00,
>-  0x11, 0x25, 0x3c, 0x5c, 0x00, 0x00,
>-  0xe8, 0x80, 0x51, 0x35, 0x00, 0x00,
>-  0x45, 0x15, 0x51, 0x84, 0x80, 0x00,
>-  0x22, 0x22, 0xa2, 0x27, 0x00, 0x00,
>-  0x96, 0x0c, 0x95, 0x51, 0x80, 0x00,
>-  0x0c, 0x50, 0x4a, 0x1a, 0x00, 0x00,
>-  0x62, 0x04, 0x30, 0x68, 0x00, 0x00,
>-  0x49, 0x06, 0x2c, 0x89, 0x00, 0x00,
>-  0x11, 0x82, 0x15, 0x8c, 0x00, 0x00,
>-  0x12, 0x38, 0x8a, 0x47, 0x00, 0x00,
>-  0x40, 0x71, 0x25, 0x81, 0x80, 0x00,
>-  0xa8, 0x8a, 0x62, 0x12, 0x80, 0x00,
>-  0x08, 0xa1, 0x58, 0x58, 0x00, 0x00,
>-  0xa0, 0xc0, 0x0e, 0x28, 0x80, 0x00,
>-  0xc5, 0x10, 0x83, 0x34, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom33_26[156] = {
>-  0x45, 0x15, 0x51, 0x84, 0x80, 0x00,
>-  0x22, 0x22, 0xa2, 0x27, 0x00, 0x00,
>-  0x96, 0x0c, 0x95, 0x51, 0x80, 0x00,
>-  0x0c, 0x50, 0x4a, 0x1a, 0x00, 0x00,
>-  0x62, 0x04, 0x30, 0x68, 0x00, 0x00,
>-  0x49, 0x06, 0x2c, 0x89, 0x00, 0x00,
>-  0x11, 0x82, 0x15, 0x8c, 0x00, 0x00,
>-  0x12, 0x38, 0x8a, 0x47, 0x00, 0x00,
>-  0x40, 0x71, 0x25, 0x81, 0x80, 0x00,
>-  0xa8, 0x8a, 0x62, 0x12, 0x80, 0x00,
>-  0x08, 0xa1, 0x58, 0x58, 0x00, 0x00,
>-  0xa0, 0xc0, 0x0e, 0x28, 0x80, 0x00,
>-  0xc5, 0x10, 0x83, 0x34, 0x00, 0x00,
>-  0x84, 0x31, 0x51, 0x84, 0x80, 0x00,
>-  0x18, 0xa2, 0xa2, 0x27, 0x00, 0x00,
>-  0x4e, 0x01, 0x95, 0x51, 0x80, 0x00,
>-  0x44, 0xc8, 0x4a, 0x1a, 0x00, 0x00,
>-  0x0e, 0x90, 0x30, 0x68, 0x00, 0x00,
>-  0x20, 0xcc, 0x2c, 0x89, 0x00, 0x00,
>-  0x93, 0x40, 0x55, 0x8c, 0x80, 0x00,
>-  0x2d, 0x10, 0xaa, 0x27, 0x00, 0x00,
>-  0x31, 0x44, 0xa5, 0x32, 0x80, 0x00,
>-  0xc0, 0x23, 0x62, 0x61, 0x80, 0x00,
>-  0x11, 0x25, 0x3c, 0x5c, 0x00, 0x00,
>-  0xe8, 0x80, 0x51, 0x35, 0x00, 0x00,
>-  0xa4, 0xa4, 0xfc, 0x91, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom33_27[162] = {
>-  0x45, 0x15, 0x51, 0x84, 0x80, 0x00,
>-  0x22, 0x22, 0xa2, 0x27, 0x00, 0x00,
>-  0x96, 0x0c, 0x95, 0x51, 0x80, 0x00,
>-  0x0c, 0x50, 0x4a, 0x1a, 0x00, 0x00,
>-  0x62, 0x04, 0x30, 0x68, 0x00, 0x00,
>-  0x49, 0x06, 0x2c, 0x89, 0x00, 0x00,
>-  0x11, 0x82, 0x15, 0x8c, 0x00, 0x00,
>-  0x12, 0x38, 0x8a, 0x47, 0x00, 0x00,
>-  0x40, 0x71, 0x25, 0x81, 0x80, 0x00,
>-  0xa8, 0x8a, 0x62, 0x12, 0x80, 0x00,
>-  0x08, 0xa1, 0x58, 0x58, 0x00, 0x00,
>-  0xa0, 0xc0, 0x0e, 0x28, 0x80, 0x00,
>-  0xc5, 0x10, 0x83, 0x34, 0x00, 0x00,
>-  0x45, 0x51, 0x15, 0x8c, 0x00, 0x00,
>-  0x22, 0x0a, 0x8a, 0x47, 0x00, 0x00,
>-  0x84, 0xd0, 0x25, 0x81, 0x80, 0x00,
>-  0x0c, 0x8a, 0x62, 0x12, 0x80, 0x00,
>-  0x18, 0x06, 0x58, 0x58, 0x00, 0x00,
>-  0x30, 0x03, 0x0e, 0x28, 0x80, 0x00,
>-  0x61, 0x08, 0x83, 0x34, 0x00, 0x00,
>-  0x40, 0x11, 0x51, 0x84, 0x80, 0x00,
>-  0x10, 0x2c, 0xa2, 0x27, 0x00, 0x00,
>-  0x09, 0x60, 0x95, 0x51, 0x80, 0x00,
>-  0x00, 0x94, 0x4a, 0x1a, 0x00, 0x00,
>-  0x52, 0x40, 0x30, 0x68, 0x00, 0x00,
>-  0xa4, 0x24, 0x2c, 0x89, 0x00, 0x00,
>-  0x82, 0x88, 0xb0, 0xde, 0x80, 0x00
>-};
>-
>-const uint8_t kMaskRandom33_28[168] = {
>-  0x45, 0x51, 0x15, 0x8c, 0x00, 0x00,
>-  0x22, 0x0a, 0x8a, 0x47, 0x00, 0x00,
>-  0x84, 0xd0, 0x25, 0x81, 0x80, 0x00,
>-  0x0c, 0x8a, 0x62, 0x12, 0x80, 0x00,
>-  0x18, 0x06, 0x58, 0x58, 0x00, 0x00,
>-  0x30, 0x03, 0x0e, 0x28, 0x80, 0x00,
>-  0x61, 0x08, 0x83, 0x34, 0x00, 0x00,
>-  0x40, 0x11, 0x51, 0x84, 0x80, 0x00,
>-  0x10, 0x2c, 0xa2, 0x27, 0x00, 0x00,
>-  0x09, 0x60, 0x95, 0x51, 0x80, 0x00,
>-  0x00, 0x94, 0x4a, 0x1a, 0x00, 0x00,
>-  0x52, 0x40, 0x30, 0x68, 0x00, 0x00,
>-  0xa4, 0x24, 0x2c, 0x89, 0x00, 0x00,
>-  0x82, 0x88, 0xb0, 0xde, 0x80, 0x00,
>-  0x45, 0x15, 0x51, 0x84, 0x80, 0x00,
>-  0x22, 0x22, 0xa2, 0x27, 0x00, 0x00,
>-  0x96, 0x0c, 0x95, 0x51, 0x80, 0x00,
>-  0x0c, 0x50, 0x4a, 0x1a, 0x00, 0x00,
>-  0x62, 0x04, 0x30, 0x68, 0x00, 0x00,
>-  0x49, 0x06, 0x2c, 0x89, 0x00, 0x00,
>-  0x11, 0x82, 0x15, 0x8c, 0x00, 0x00,
>-  0x12, 0x38, 0x8a, 0x47, 0x00, 0x00,
>-  0x40, 0x71, 0x25, 0x81, 0x80, 0x00,
>-  0xa8, 0x8a, 0x62, 0x12, 0x80, 0x00,
>-  0x08, 0xa1, 0x58, 0x58, 0x00, 0x00,
>-  0xa0, 0xc0, 0x0e, 0x28, 0x80, 0x00,
>-  0xc5, 0x10, 0x83, 0x34, 0x00, 0x00,
>-  0x1b, 0xf4, 0xaa, 0xec, 0x80, 0x00
>-};
>-
>-const uint8_t kMaskRandom33_29[174] = {
>-  0x45, 0x51, 0x15, 0x8c, 0x00, 0x00,
>-  0x22, 0x0a, 0x8a, 0x47, 0x00, 0x00,
>-  0x84, 0xd0, 0x25, 0x81, 0x80, 0x00,
>-  0x0c, 0x8a, 0x62, 0x12, 0x80, 0x00,
>-  0x18, 0x06, 0x58, 0x58, 0x00, 0x00,
>-  0x30, 0x03, 0x0e, 0x28, 0x80, 0x00,
>-  0x61, 0x08, 0x83, 0x34, 0x00, 0x00,
>-  0x40, 0x11, 0x51, 0x84, 0x80, 0x00,
>-  0x10, 0x2c, 0xa2, 0x27, 0x00, 0x00,
>-  0x09, 0x60, 0x95, 0x51, 0x80, 0x00,
>-  0x00, 0x94, 0x4a, 0x1a, 0x00, 0x00,
>-  0x52, 0x40, 0x30, 0x68, 0x00, 0x00,
>-  0xa4, 0x24, 0x2c, 0x89, 0x00, 0x00,
>-  0x82, 0x88, 0xb0, 0xde, 0x80, 0x00,
>-  0x55, 0x11, 0x15, 0x8c, 0x00, 0x00,
>-  0x22, 0x22, 0x8a, 0x47, 0x00, 0x00,
>-  0x11, 0x11, 0x25, 0x81, 0x80, 0x00,
>-  0x80, 0x45, 0x62, 0x12, 0x80, 0x00,
>-  0x20, 0x1a, 0x58, 0x58, 0x00, 0x00,
>-  0x08, 0x68, 0x0e, 0x28, 0x80, 0x00,
>-  0x22, 0x84, 0x83, 0x34, 0x00, 0x00,
>-  0x48, 0x09, 0x25, 0x2c, 0x00, 0x00,
>-  0x07, 0x01, 0x8a, 0x91, 0x00, 0x00,
>-  0x94, 0x20, 0x91, 0xc0, 0x80, 0x00,
>-  0x82, 0x06, 0x68, 0x06, 0x80, 0x00,
>-  0x60, 0x48, 0x32, 0xc8, 0x00, 0x00,
>-  0x89, 0x80, 0x43, 0x45, 0x00, 0x00,
>-  0x00, 0x8e, 0xc4, 0x30, 0x80, 0x00,
>-  0x18, 0x22, 0x1c, 0xa2, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom33_3[18] = {
>-  0xad, 0x2d, 0xcd, 0xcc, 0x00, 0x00,
>-  0x76, 0x36, 0x97, 0x27, 0x00, 0x00,
>-  0x26, 0xdb, 0xb8, 0xd1, 0x80, 0x00
>-};
>-
>-const uint8_t kMaskRandom33_30[180] = {
>-  0x55, 0x11, 0x15, 0x8c, 0x00, 0x00,
>-  0x22, 0x22, 0x8a, 0x47, 0x00, 0x00,
>-  0x11, 0x11, 0x25, 0x81, 0x80, 0x00,
>-  0x80, 0x45, 0x62, 0x12, 0x80, 0x00,
>-  0x20, 0x1a, 0x58, 0x58, 0x00, 0x00,
>-  0x08, 0x68, 0x0e, 0x28, 0x80, 0x00,
>-  0x22, 0x84, 0x83, 0x34, 0x00, 0x00,
>-  0x48, 0x09, 0x25, 0x2c, 0x00, 0x00,
>-  0x07, 0x01, 0x8a, 0x91, 0x00, 0x00,
>-  0x94, 0x20, 0x91, 0xc0, 0x80, 0x00,
>-  0x82, 0x06, 0x68, 0x06, 0x80, 0x00,
>-  0x60, 0x48, 0x32, 0xc8, 0x00, 0x00,
>-  0x89, 0x80, 0x43, 0x45, 0x00, 0x00,
>-  0x00, 0x8e, 0xc4, 0x30, 0x80, 0x00,
>-  0x18, 0x22, 0x1c, 0xa2, 0x00, 0x00,
>-  0x45, 0x51, 0x15, 0x8c, 0x00, 0x00,
>-  0x22, 0x0a, 0x8a, 0x47, 0x00, 0x00,
>-  0x84, 0xd0, 0x25, 0x81, 0x80, 0x00,
>-  0x0c, 0x8a, 0x62, 0x12, 0x80, 0x00,
>-  0x18, 0x06, 0x58, 0x58, 0x00, 0x00,
>-  0x30, 0x03, 0x0e, 0x28, 0x80, 0x00,
>-  0x61, 0x08, 0x83, 0x34, 0x00, 0x00,
>-  0x40, 0x11, 0x51, 0x84, 0x80, 0x00,
>-  0x10, 0x2c, 0xa2, 0x27, 0x00, 0x00,
>-  0x09, 0x60, 0x95, 0x51, 0x80, 0x00,
>-  0x00, 0x94, 0x4a, 0x1a, 0x00, 0x00,
>-  0x52, 0x40, 0x30, 0x68, 0x00, 0x00,
>-  0xa4, 0x24, 0x2c, 0x89, 0x00, 0x00,
>-  0x82, 0x88, 0xb0, 0xde, 0x80, 0x00,
>-  0x6d, 0xd2, 0x8c, 0x00, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom33_31[186] = {
>-  0x55, 0x11, 0x15, 0x8c, 0x00, 0x00,
>-  0x22, 0x22, 0x8a, 0x47, 0x00, 0x00,
>-  0x11, 0x11, 0x25, 0x81, 0x80, 0x00,
>-  0x80, 0x45, 0x62, 0x12, 0x80, 0x00,
>-  0x20, 0x1a, 0x58, 0x58, 0x00, 0x00,
>-  0x08, 0x68, 0x0e, 0x28, 0x80, 0x00,
>-  0x22, 0x84, 0x83, 0x34, 0x00, 0x00,
>-  0x48, 0x09, 0x25, 0x2c, 0x00, 0x00,
>-  0x07, 0x01, 0x8a, 0x91, 0x00, 0x00,
>-  0x94, 0x20, 0x91, 0xc0, 0x80, 0x00,
>-  0x82, 0x06, 0x68, 0x06, 0x80, 0x00,
>-  0x60, 0x48, 0x32, 0xc8, 0x00, 0x00,
>-  0x89, 0x80, 0x43, 0x45, 0x00, 0x00,
>-  0x00, 0x8e, 0xc4, 0x30, 0x80, 0x00,
>-  0x18, 0x22, 0x1c, 0xa2, 0x00, 0x00,
>-  0xa4, 0x10, 0x25, 0x2c, 0x00, 0x00,
>-  0x01, 0x2a, 0x8a, 0x91, 0x00, 0x00,
>-  0x06, 0x42, 0x91, 0xc0, 0x80, 0x00,
>-  0x08, 0x68, 0x68, 0x06, 0x80, 0x00,
>-  0x81, 0x90, 0x32, 0xc8, 0x00, 0x00,
>-  0x00, 0xf0, 0x43, 0x45, 0x00, 0x00,
>-  0x50, 0x05, 0xc4, 0x30, 0x80, 0x00,
>-  0x20, 0x51, 0x1c, 0xa2, 0x00, 0x00,
>-  0x43, 0x08, 0x15, 0x8c, 0x00, 0x00,
>-  0x68, 0x80, 0x8a, 0x47, 0x00, 0x00,
>-  0x80, 0x0b, 0x25, 0x81, 0x80, 0x00,
>-  0x10, 0x4c, 0x62, 0x12, 0x80, 0x00,
>-  0x12, 0x30, 0x58, 0x58, 0x00, 0x00,
>-  0x40, 0x85, 0x0e, 0x28, 0x80, 0x00,
>-  0x0e, 0x04, 0x83, 0x34, 0x00, 0x00,
>-  0x18, 0x12, 0x0a, 0x1c, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom33_32[192] = {
>-  0xa4, 0x10, 0x25, 0x2c, 0x00, 0x00,
>-  0x01, 0x2a, 0x8a, 0x91, 0x00, 0x00,
>-  0x06, 0x42, 0x91, 0xc0, 0x80, 0x00,
>-  0x08, 0x68, 0x68, 0x06, 0x80, 0x00,
>-  0x81, 0x90, 0x32, 0xc8, 0x00, 0x00,
>-  0x00, 0xf0, 0x43, 0x45, 0x00, 0x00,
>-  0x50, 0x05, 0xc4, 0x30, 0x80, 0x00,
>-  0x20, 0x51, 0x1c, 0xa2, 0x00, 0x00,
>-  0x43, 0x08, 0x15, 0x8c, 0x00, 0x00,
>-  0x68, 0x80, 0x8a, 0x47, 0x00, 0x00,
>-  0x80, 0x0b, 0x25, 0x81, 0x80, 0x00,
>-  0x10, 0x4c, 0x62, 0x12, 0x80, 0x00,
>-  0x12, 0x30, 0x58, 0x58, 0x00, 0x00,
>-  0x40, 0x85, 0x0e, 0x28, 0x80, 0x00,
>-  0x0e, 0x04, 0x83, 0x34, 0x00, 0x00,
>-  0x18, 0x12, 0x0a, 0x1c, 0x00, 0x00,
>-  0x55, 0x11, 0x15, 0x8c, 0x00, 0x00,
>-  0x22, 0x22, 0x8a, 0x47, 0x00, 0x00,
>-  0x11, 0x11, 0x25, 0x81, 0x80, 0x00,
>-  0x80, 0x45, 0x62, 0x12, 0x80, 0x00,
>-  0x20, 0x1a, 0x58, 0x58, 0x00, 0x00,
>-  0x08, 0x68, 0x0e, 0x28, 0x80, 0x00,
>-  0x22, 0x84, 0x83, 0x34, 0x00, 0x00,
>-  0x48, 0x09, 0x25, 0x2c, 0x00, 0x00,
>-  0x07, 0x01, 0x8a, 0x91, 0x00, 0x00,
>-  0x94, 0x20, 0x91, 0xc0, 0x80, 0x00,
>-  0x82, 0x06, 0x68, 0x06, 0x80, 0x00,
>-  0x60, 0x48, 0x32, 0xc8, 0x00, 0x00,
>-  0x89, 0x80, 0x43, 0x45, 0x00, 0x00,
>-  0x00, 0x8e, 0xc4, 0x30, 0x80, 0x00,
>-  0x18, 0x22, 0x1c, 0xa2, 0x00, 0x00,
>-  0x73, 0x8e, 0x12, 0xca, 0x80, 0x00
>-};
>-
>-const uint8_t kMaskRandom33_33[198] = {
>-  0xa4, 0x10, 0x25, 0x2c, 0x00, 0x00,
>-  0x01, 0x2a, 0x8a, 0x91, 0x00, 0x00,
>-  0x06, 0x42, 0x91, 0xc0, 0x80, 0x00,
>-  0x08, 0x68, 0x68, 0x06, 0x80, 0x00,
>-  0x81, 0x90, 0x32, 0xc8, 0x00, 0x00,
>-  0x00, 0xf0, 0x43, 0x45, 0x00, 0x00,
>-  0x50, 0x05, 0xc4, 0x30, 0x80, 0x00,
>-  0x20, 0x51, 0x1c, 0xa2, 0x00, 0x00,
>-  0x43, 0x08, 0x15, 0x8c, 0x00, 0x00,
>-  0x68, 0x80, 0x8a, 0x47, 0x00, 0x00,
>-  0x80, 0x0b, 0x25, 0x81, 0x80, 0x00,
>-  0x10, 0x4c, 0x62, 0x12, 0x80, 0x00,
>-  0x12, 0x30, 0x58, 0x58, 0x00, 0x00,
>-  0x40, 0x85, 0x0e, 0x28, 0x80, 0x00,
>-  0x0e, 0x04, 0x83, 0x34, 0x00, 0x00,
>-  0x18, 0x12, 0x0a, 0x1c, 0x00, 0x00,
>-  0xa4, 0x10, 0x25, 0x2c, 0x00, 0x00,
>-  0x01, 0x2a, 0x8a, 0x91, 0x00, 0x00,
>-  0x06, 0x42, 0x91, 0xc0, 0x80, 0x00,
>-  0x08, 0x68, 0x68, 0x06, 0x80, 0x00,
>-  0x81, 0x90, 0x32, 0xc8, 0x00, 0x00,
>-  0x00, 0xf0, 0x43, 0x45, 0x00, 0x00,
>-  0x50, 0x05, 0xc4, 0x30, 0x80, 0x00,
>-  0x20, 0x51, 0x1c, 0xa2, 0x00, 0x00,
>-  0x43, 0x08, 0x25, 0x4c, 0x00, 0x00,
>-  0x68, 0x80, 0x8a, 0x66, 0x00, 0x00,
>-  0x80, 0x0b, 0x91, 0x91, 0x00, 0x00,
>-  0x10, 0x4c, 0x68, 0x42, 0x80, 0x00,
>-  0x12, 0x30, 0x32, 0xa4, 0x00, 0x00,
>-  0x40, 0x85, 0x43, 0x13, 0x00, 0x00,
>-  0x0e, 0x04, 0xc4, 0x30, 0x80, 0x00,
>-  0x18, 0x12, 0x1c, 0x88, 0x80, 0x00,
>-  0xdb, 0x10, 0x3c, 0x09, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom33_4[24] = {
>-  0x55, 0x55, 0xca, 0xec, 0x00, 0x00,
>-  0xaa, 0xaa, 0xa9, 0x67, 0x00, 0x00,
>-  0x35, 0x35, 0x3a, 0xb1, 0x80, 0x00,
>-  0xca, 0xca, 0x55, 0x5a, 0x80, 0x00
>-};
>-
>-const uint8_t kMaskRandom33_5[30] = {
>-  0x55, 0x55, 0x55, 0x44, 0x80, 0x00,
>-  0x2a, 0x2a, 0x2a, 0x66, 0x00, 0x00,
>-  0x24, 0x25, 0x25, 0xa1, 0x80, 0x00,
>-  0x84, 0xc8, 0xe2, 0x12, 0x80, 0x00,
>-  0x10, 0xb6, 0x99, 0x98, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom33_6[36] = {
>-  0x51, 0x51, 0xd1, 0x4c, 0x00, 0x00,
>-  0x0a, 0x2a, 0xa2, 0xc5, 0x00, 0x00,
>-  0xa2, 0x15, 0x95, 0x30, 0x80, 0x00,
>-  0x84, 0x4a, 0xca, 0x0a, 0x80, 0x00,
>-  0x30, 0x92, 0xa4, 0xaa, 0x00, 0x00,
>-  0x04, 0xac, 0x78, 0x15, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom33_7[42] = {
>-  0x45, 0x51, 0x15, 0x44, 0x80, 0x00,
>-  0x22, 0x2a, 0x8a, 0x23, 0x00, 0x00,
>-  0x91, 0x11, 0x85, 0x91, 0x00, 0x00,
>-  0x2e, 0x08, 0x32, 0x0a, 0x80, 0x00,
>-  0x48, 0x34, 0x58, 0x34, 0x00, 0x00,
>-  0x90, 0x29, 0x2c, 0x0d, 0x00, 0x00,
>-  0x09, 0x86, 0x43, 0xc8, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom33_8[48] = {
>-  0x20, 0x54, 0x64, 0x16, 0x00, 0x00,
>-  0x18, 0x88, 0xa2, 0xc2, 0x00, 0x00,
>-  0x84, 0x07, 0x51, 0x60, 0x80, 0x00,
>-  0x60, 0x48, 0x4a, 0x85, 0x00, 0x00,
>-  0x12, 0x82, 0x38, 0x4c, 0x00, 0x00,
>-  0x81, 0x41, 0x89, 0x29, 0x00, 0x00,
>-  0x40, 0x62, 0x07, 0x11, 0x80, 0x00,
>-  0x16, 0x30, 0x94, 0xb0, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom33_9[54] = {
>-  0x55, 0x51, 0x8e, 0xcc, 0x00, 0x00,
>-  0x22, 0x2a, 0x6a, 0x2b, 0x00, 0x00,
>-  0x05, 0x85, 0x36, 0x32, 0x80, 0x00,
>-  0x09, 0x4a, 0xd1, 0x25, 0x80, 0x00,
>-  0x84, 0x32, 0x55, 0x8c, 0x80, 0x00,
>-  0xc0, 0x0d, 0xaa, 0x27, 0x00, 0x00,
>-  0x20, 0xa6, 0xa5, 0x32, 0x80, 0x00,
>-  0x1a, 0x09, 0x62, 0x61, 0x80, 0x00,
>-  0x44, 0x64, 0x3c, 0x5c, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom34_1[6] = {
>-  0xff, 0xff, 0xff, 0xff, 0xc0, 0x00
>-};
>-
>-const uint8_t kMaskRandom34_10[60] = {
>-  0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
>-  0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
>-  0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
>-  0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
>-  0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00,
>-  0x8e, 0xcc, 0x47, 0x66, 0x00, 0x00,
>-  0x6a, 0x2b, 0x35, 0x15, 0x80, 0x00,
>-  0x36, 0x32, 0x9b, 0x19, 0x40, 0x00,
>-  0xd1, 0x25, 0xe8, 0x92, 0xc0, 0x00,
>-  0xc8, 0x02, 0xe4, 0x01, 0x40, 0x00
>-};
>-
>-const uint8_t kMaskRandom34_11[66] = {
>-  0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
>-  0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
>-  0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
>-  0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
>-  0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00,
>-  0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
>-  0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
>-  0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
>-  0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
>-  0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
>-  0x2c, 0x89, 0x16, 0x44, 0x80, 0x00
>-};
>-
>-const uint8_t kMaskRandom34_12[72] = {
>-  0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
>-  0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
>-  0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
>-  0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
>-  0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
>-  0x2c, 0x89, 0x16, 0x44, 0x80, 0x00,
>-  0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
>-  0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
>-  0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
>-  0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
>-  0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00,
>-  0x51, 0x35, 0x28, 0x9a, 0x80, 0x00
>-};
>-
>-const uint8_t kMaskRandom34_13[78] = {
>-  0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
>-  0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
>-  0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
>-  0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
>-  0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
>-  0x2c, 0x89, 0x16, 0x44, 0x80, 0x00,
>-  0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
>-  0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
>-  0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
>-  0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
>-  0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
>-  0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
>-  0x83, 0x34, 0x41, 0x9a, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom34_14[84] = {
>-  0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
>-  0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
>-  0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
>-  0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
>-  0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
>-  0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
>-  0x83, 0x34, 0x41, 0x9a, 0x00, 0x00,
>-  0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
>-  0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
>-  0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
>-  0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
>-  0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
>-  0x2c, 0x89, 0x16, 0x44, 0x80, 0x00,
>-  0xb0, 0xde, 0xd8, 0x6f, 0x40, 0x00
>-};
>-
>-const uint8_t kMaskRandom34_15[90] = {
>-  0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
>-  0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
>-  0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
>-  0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
>-  0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
>-  0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
>-  0x83, 0x34, 0x41, 0x9a, 0x00, 0x00,
>-  0x25, 0x2c, 0x12, 0x96, 0x00, 0x00,
>-  0x8a, 0x91, 0x45, 0x48, 0x80, 0x00,
>-  0x91, 0xc0, 0xc8, 0xe0, 0x40, 0x00,
>-  0x68, 0x06, 0xb4, 0x03, 0x40, 0x00,
>-  0x32, 0xc8, 0x19, 0x64, 0x00, 0x00,
>-  0x43, 0x45, 0x21, 0xa2, 0x80, 0x00,
>-  0xc4, 0x30, 0xe2, 0x18, 0x40, 0x00,
>-  0x1c, 0xa2, 0x0e, 0x51, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom34_16[96] = {
>-  0x25, 0x2c, 0x12, 0x96, 0x00, 0x00,
>-  0x8a, 0x91, 0x45, 0x48, 0x80, 0x00,
>-  0x91, 0xc0, 0xc8, 0xe0, 0x40, 0x00,
>-  0x68, 0x06, 0xb4, 0x03, 0x40, 0x00,
>-  0x32, 0xc8, 0x19, 0x64, 0x00, 0x00,
>-  0x43, 0x45, 0x21, 0xa2, 0x80, 0x00,
>-  0xc4, 0x30, 0xe2, 0x18, 0x40, 0x00,
>-  0x1c, 0xa2, 0x0e, 0x51, 0x00, 0x00,
>-  0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
>-  0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
>-  0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
>-  0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
>-  0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
>-  0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
>-  0x83, 0x34, 0x41, 0x9a, 0x00, 0x00,
>-  0x0a, 0x1c, 0x05, 0x0e, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom34_17[102] = {
>-  0x25, 0x2c, 0x12, 0x96, 0x00, 0x00,
>-  0x8a, 0x91, 0x45, 0x48, 0x80, 0x00,
>-  0x91, 0xc0, 0xc8, 0xe0, 0x40, 0x00,
>-  0x68, 0x06, 0xb4, 0x03, 0x40, 0x00,
>-  0x32, 0xc8, 0x19, 0x64, 0x00, 0x00,
>-  0x43, 0x45, 0x21, 0xa2, 0x80, 0x00,
>-  0xc4, 0x30, 0xe2, 0x18, 0x40, 0x00,
>-  0x1c, 0xa2, 0x0e, 0x51, 0x00, 0x00,
>-  0x25, 0x4c, 0x12, 0xa6, 0x00, 0x00,
>-  0x8a, 0x66, 0x45, 0x33, 0x00, 0x00,
>-  0x91, 0x91, 0x48, 0xc8, 0x80, 0x00,
>-  0x68, 0x42, 0xb4, 0x21, 0x40, 0x00,
>-  0x32, 0xa4, 0x19, 0x52, 0x00, 0x00,
>-  0x43, 0x13, 0x21, 0x89, 0x80, 0x00,
>-  0xc4, 0x30, 0xe2, 0x18, 0x40, 0x00,
>-  0x1c, 0x88, 0x8e, 0x44, 0x40, 0x00,
>-  0x3c, 0x09, 0x1e, 0x04, 0x80, 0x00
>-};
>-
>-const uint8_t kMaskRandom34_18[108] = {
>-  0x8e, 0xcc, 0x47, 0x66, 0x00, 0x00,
>-  0x6a, 0x2b, 0x35, 0x15, 0x80, 0x00,
>-  0x36, 0x32, 0x9b, 0x19, 0x40, 0x00,
>-  0xd1, 0x25, 0xe8, 0x92, 0xc0, 0x00,
>-  0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
>-  0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
>-  0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
>-  0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
>-  0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00,
>-  0x64, 0x16, 0x32, 0x0b, 0x00, 0x00,
>-  0xa2, 0xc2, 0x51, 0x61, 0x00, 0x00,
>-  0x51, 0x60, 0xa8, 0xb0, 0x40, 0x00,
>-  0x4a, 0x85, 0x25, 0x42, 0x80, 0x00,
>-  0x38, 0x4c, 0x1c, 0x26, 0x00, 0x00,
>-  0x89, 0x29, 0x44, 0x94, 0x80, 0x00,
>-  0x07, 0x11, 0x83, 0x88, 0xc0, 0x00,
>-  0x94, 0xb0, 0x4a, 0x58, 0x00, 0x00,
>-  0x89, 0x70, 0xf3, 0xf7, 0x40, 0x00
>-};
>-
>-const uint8_t kMaskRandom34_19[114] = {
>-  0x8e, 0xcc, 0x47, 0x66, 0x00, 0x00,
>-  0x6a, 0x2b, 0x35, 0x15, 0x80, 0x00,
>-  0x36, 0x32, 0x9b, 0x19, 0x40, 0x00,
>-  0xd1, 0x25, 0xe8, 0x92, 0xc0, 0x00,
>-  0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
>-  0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
>-  0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
>-  0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
>-  0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00,
>-  0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
>-  0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
>-  0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
>-  0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
>-  0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00,
>-  0x8e, 0xcc, 0x47, 0x66, 0x00, 0x00,
>-  0x6a, 0x2b, 0x35, 0x15, 0x80, 0x00,
>-  0x36, 0x32, 0x9b, 0x19, 0x40, 0x00,
>-  0xd1, 0x25, 0xe8, 0x92, 0xc0, 0x00,
>-  0xc8, 0x02, 0xe4, 0x01, 0x40, 0x00
>-};
>-
>-const uint8_t kMaskRandom34_2[12] = {
>-  0xce, 0xce, 0x67, 0x67, 0x00, 0x00,
>-  0xb9, 0x39, 0xdc, 0x9c, 0xc0, 0x00
>-};
>-
>-const uint8_t kMaskRandom34_20[120] = {
>-  0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
>-  0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
>-  0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
>-  0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
>-  0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00,
>-  0x8e, 0xcc, 0x47, 0x66, 0x00, 0x00,
>-  0x6a, 0x2b, 0x35, 0x15, 0x80, 0x00,
>-  0x36, 0x32, 0x9b, 0x19, 0x40, 0x00,
>-  0xd1, 0x25, 0xe8, 0x92, 0xc0, 0x00,
>-  0xc8, 0x02, 0xe4, 0x01, 0x40, 0x00,
>-  0x8e, 0xcc, 0x47, 0x66, 0x00, 0x00,
>-  0x6a, 0x2b, 0x35, 0x15, 0x80, 0x00,
>-  0x36, 0x32, 0x9b, 0x19, 0x40, 0x00,
>-  0xd1, 0x25, 0xe8, 0x92, 0xc0, 0x00,
>-  0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
>-  0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
>-  0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
>-  0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
>-  0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00,
>-  0x5d, 0xc5, 0xfe, 0xd8, 0x40, 0x00
>-};
>-
>-const uint8_t kMaskRandom34_21[126] = {
>-  0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
>-  0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
>-  0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
>-  0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
>-  0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00,
>-  0x8e, 0xcc, 0x47, 0x66, 0x00, 0x00,
>-  0x6a, 0x2b, 0x35, 0x15, 0x80, 0x00,
>-  0x36, 0x32, 0x9b, 0x19, 0x40, 0x00,
>-  0xd1, 0x25, 0xe8, 0x92, 0xc0, 0x00,
>-  0xc8, 0x02, 0xe4, 0x01, 0x40, 0x00,
>-  0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
>-  0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
>-  0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
>-  0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
>-  0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00,
>-  0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
>-  0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
>-  0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
>-  0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
>-  0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
>-  0x2c, 0x89, 0x16, 0x44, 0x80, 0x00
>-};
>-
>-const uint8_t kMaskRandom34_22[132] = {
>-  0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
>-  0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
>-  0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
>-  0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
>-  0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00,
>-  0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
>-  0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
>-  0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
>-  0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
>-  0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
>-  0x2c, 0x89, 0x16, 0x44, 0x80, 0x00,
>-  0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
>-  0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
>-  0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
>-  0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
>-  0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00,
>-  0x8e, 0xcc, 0x47, 0x66, 0x00, 0x00,
>-  0x6a, 0x2b, 0x35, 0x15, 0x80, 0x00,
>-  0x36, 0x32, 0x9b, 0x19, 0x40, 0x00,
>-  0xd1, 0x25, 0xe8, 0x92, 0xc0, 0x00,
>-  0xc8, 0x02, 0xe4, 0x01, 0x40, 0x00,
>-  0x2a, 0xf7, 0x4f, 0xf5, 0x80, 0x00
>-};
>-
>-const uint8_t kMaskRandom34_23[138] = {
>-  0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
>-  0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
>-  0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
>-  0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
>-  0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00,
>-  0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
>-  0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
>-  0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
>-  0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
>-  0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
>-  0x2c, 0x89, 0x16, 0x44, 0x80, 0x00,
>-  0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
>-  0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
>-  0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
>-  0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
>-  0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
>-  0x2c, 0x89, 0x16, 0x44, 0x80, 0x00,
>-  0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
>-  0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
>-  0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
>-  0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
>-  0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00,
>-  0x51, 0x35, 0x28, 0x9a, 0x80, 0x00
>-};
>-
>-const uint8_t kMaskRandom34_24[144] = {
>-  0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
>-  0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
>-  0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
>-  0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
>-  0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
>-  0x2c, 0x89, 0x16, 0x44, 0x80, 0x00,
>-  0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
>-  0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
>-  0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
>-  0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
>-  0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00,
>-  0x51, 0x35, 0x28, 0x9a, 0x80, 0x00,
>-  0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
>-  0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
>-  0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
>-  0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
>-  0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00,
>-  0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
>-  0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
>-  0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
>-  0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
>-  0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
>-  0x2c, 0x89, 0x16, 0x44, 0x80, 0x00,
>-  0x4c, 0xb8, 0x04, 0x74, 0xc0, 0x00
>-};
>-
>-const uint8_t kMaskRandom34_25[150] = {
>-  0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
>-  0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
>-  0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
>-  0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
>-  0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
>-  0x2c, 0x89, 0x16, 0x44, 0x80, 0x00,
>-  0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
>-  0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
>-  0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
>-  0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
>-  0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00,
>-  0x51, 0x35, 0x28, 0x9a, 0x80, 0x00,
>-  0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
>-  0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
>-  0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
>-  0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
>-  0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
>-  0x2c, 0x89, 0x16, 0x44, 0x80, 0x00,
>-  0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
>-  0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
>-  0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
>-  0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
>-  0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
>-  0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
>-  0x83, 0x34, 0x41, 0x9a, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom34_26[156] = {
>-  0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
>-  0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
>-  0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
>-  0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
>-  0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
>-  0x2c, 0x89, 0x16, 0x44, 0x80, 0x00,
>-  0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
>-  0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
>-  0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
>-  0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
>-  0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
>-  0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
>-  0x83, 0x34, 0x41, 0x9a, 0x00, 0x00,
>-  0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
>-  0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
>-  0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
>-  0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
>-  0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
>-  0x2c, 0x89, 0x16, 0x44, 0x80, 0x00,
>-  0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
>-  0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
>-  0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
>-  0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
>-  0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00,
>-  0x51, 0x35, 0x28, 0x9a, 0x80, 0x00,
>-  0x95, 0x20, 0xe9, 0xef, 0xc0, 0x00
>-};
>-
>-const uint8_t kMaskRandom34_27[162] = {
>-  0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
>-  0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
>-  0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
>-  0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
>-  0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
>-  0x2c, 0x89, 0x16, 0x44, 0x80, 0x00,
>-  0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
>-  0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
>-  0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
>-  0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
>-  0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
>-  0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
>-  0x83, 0x34, 0x41, 0x9a, 0x00, 0x00,
>-  0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
>-  0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
>-  0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
>-  0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
>-  0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
>-  0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
>-  0x83, 0x34, 0x41, 0x9a, 0x00, 0x00,
>-  0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
>-  0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
>-  0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
>-  0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
>-  0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
>-  0x2c, 0x89, 0x16, 0x44, 0x80, 0x00,
>-  0xb0, 0xde, 0xd8, 0x6f, 0x40, 0x00
>-};
>-
>-const uint8_t kMaskRandom34_28[168] = {
>-  0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
>-  0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
>-  0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
>-  0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
>-  0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
>-  0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
>-  0x83, 0x34, 0x41, 0x9a, 0x00, 0x00,
>-  0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
>-  0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
>-  0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
>-  0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
>-  0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
>-  0x2c, 0x89, 0x16, 0x44, 0x80, 0x00,
>-  0xb0, 0xde, 0xd8, 0x6f, 0x40, 0x00,
>-  0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
>-  0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
>-  0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
>-  0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
>-  0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
>-  0x2c, 0x89, 0x16, 0x44, 0x80, 0x00,
>-  0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
>-  0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
>-  0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
>-  0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
>-  0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
>-  0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
>-  0x83, 0x34, 0x41, 0x9a, 0x00, 0x00,
>-  0x10, 0x6c, 0xff, 0x60, 0x80, 0x00
>-};
>-
>-const uint8_t kMaskRandom34_29[174] = {
>-  0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
>-  0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
>-  0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
>-  0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
>-  0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
>-  0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
>-  0x83, 0x34, 0x41, 0x9a, 0x00, 0x00,
>-  0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
>-  0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
>-  0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
>-  0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
>-  0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
>-  0x2c, 0x89, 0x16, 0x44, 0x80, 0x00,
>-  0xb0, 0xde, 0xd8, 0x6f, 0x40, 0x00,
>-  0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
>-  0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
>-  0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
>-  0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
>-  0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
>-  0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
>-  0x83, 0x34, 0x41, 0x9a, 0x00, 0x00,
>-  0x25, 0x2c, 0x12, 0x96, 0x00, 0x00,
>-  0x8a, 0x91, 0x45, 0x48, 0x80, 0x00,
>-  0x91, 0xc0, 0xc8, 0xe0, 0x40, 0x00,
>-  0x68, 0x06, 0xb4, 0x03, 0x40, 0x00,
>-  0x32, 0xc8, 0x19, 0x64, 0x00, 0x00,
>-  0x43, 0x45, 0x21, 0xa2, 0x80, 0x00,
>-  0xc4, 0x30, 0xe2, 0x18, 0x40, 0x00,
>-  0x1c, 0xa2, 0x0e, 0x51, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom34_3[18] = {
>-  0xcd, 0xcc, 0x66, 0xe6, 0x00, 0x00,
>-  0x97, 0x27, 0x4b, 0x93, 0x80, 0x00,
>-  0xb8, 0xd1, 0xdc, 0x68, 0xc0, 0x00
>-};
>-
>-const uint8_t kMaskRandom34_30[180] = {
>-  0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
>-  0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
>-  0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
>-  0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
>-  0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
>-  0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
>-  0x83, 0x34, 0x41, 0x9a, 0x00, 0x00,
>-  0x25, 0x2c, 0x12, 0x96, 0x00, 0x00,
>-  0x8a, 0x91, 0x45, 0x48, 0x80, 0x00,
>-  0x91, 0xc0, 0xc8, 0xe0, 0x40, 0x00,
>-  0x68, 0x06, 0xb4, 0x03, 0x40, 0x00,
>-  0x32, 0xc8, 0x19, 0x64, 0x00, 0x00,
>-  0x43, 0x45, 0x21, 0xa2, 0x80, 0x00,
>-  0xc4, 0x30, 0xe2, 0x18, 0x40, 0x00,
>-  0x1c, 0xa2, 0x0e, 0x51, 0x00, 0x00,
>-  0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
>-  0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
>-  0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
>-  0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
>-  0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
>-  0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
>-  0x83, 0x34, 0x41, 0x9a, 0x00, 0x00,
>-  0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
>-  0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
>-  0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
>-  0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
>-  0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
>-  0x2c, 0x89, 0x16, 0x44, 0x80, 0x00,
>-  0xb0, 0xde, 0xd8, 0x6f, 0x40, 0x00,
>-  0x87, 0x93, 0x96, 0xc7, 0x80, 0x00
>-};
>-
>-const uint8_t kMaskRandom34_31[186] = {
>-  0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
>-  0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
>-  0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
>-  0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
>-  0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
>-  0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
>-  0x83, 0x34, 0x41, 0x9a, 0x00, 0x00,
>-  0x25, 0x2c, 0x12, 0x96, 0x00, 0x00,
>-  0x8a, 0x91, 0x45, 0x48, 0x80, 0x00,
>-  0x91, 0xc0, 0xc8, 0xe0, 0x40, 0x00,
>-  0x68, 0x06, 0xb4, 0x03, 0x40, 0x00,
>-  0x32, 0xc8, 0x19, 0x64, 0x00, 0x00,
>-  0x43, 0x45, 0x21, 0xa2, 0x80, 0x00,
>-  0xc4, 0x30, 0xe2, 0x18, 0x40, 0x00,
>-  0x1c, 0xa2, 0x0e, 0x51, 0x00, 0x00,
>-  0x25, 0x2c, 0x12, 0x96, 0x00, 0x00,
>-  0x8a, 0x91, 0x45, 0x48, 0x80, 0x00,
>-  0x91, 0xc0, 0xc8, 0xe0, 0x40, 0x00,
>-  0x68, 0x06, 0xb4, 0x03, 0x40, 0x00,
>-  0x32, 0xc8, 0x19, 0x64, 0x00, 0x00,
>-  0x43, 0x45, 0x21, 0xa2, 0x80, 0x00,
>-  0xc4, 0x30, 0xe2, 0x18, 0x40, 0x00,
>-  0x1c, 0xa2, 0x0e, 0x51, 0x00, 0x00,
>-  0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
>-  0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
>-  0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
>-  0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
>-  0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
>-  0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
>-  0x83, 0x34, 0x41, 0x9a, 0x00, 0x00,
>-  0x0a, 0x1c, 0x05, 0x0e, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom34_32[192] = {
>-  0x25, 0x2c, 0x12, 0x96, 0x00, 0x00,
>-  0x8a, 0x91, 0x45, 0x48, 0x80, 0x00,
>-  0x91, 0xc0, 0xc8, 0xe0, 0x40, 0x00,
>-  0x68, 0x06, 0xb4, 0x03, 0x40, 0x00,
>-  0x32, 0xc8, 0x19, 0x64, 0x00, 0x00,
>-  0x43, 0x45, 0x21, 0xa2, 0x80, 0x00,
>-  0xc4, 0x30, 0xe2, 0x18, 0x40, 0x00,
>-  0x1c, 0xa2, 0x0e, 0x51, 0x00, 0x00,
>-  0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
>-  0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
>-  0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
>-  0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
>-  0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
>-  0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
>-  0x83, 0x34, 0x41, 0x9a, 0x00, 0x00,
>-  0x0a, 0x1c, 0x05, 0x0e, 0x00, 0x00,
>-  0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
>-  0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
>-  0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
>-  0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
>-  0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
>-  0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
>-  0x83, 0x34, 0x41, 0x9a, 0x00, 0x00,
>-  0x25, 0x2c, 0x12, 0x96, 0x00, 0x00,
>-  0x8a, 0x91, 0x45, 0x48, 0x80, 0x00,
>-  0x91, 0xc0, 0xc8, 0xe0, 0x40, 0x00,
>-  0x68, 0x06, 0xb4, 0x03, 0x40, 0x00,
>-  0x32, 0xc8, 0x19, 0x64, 0x00, 0x00,
>-  0x43, 0x45, 0x21, 0xa2, 0x80, 0x00,
>-  0xc4, 0x30, 0xe2, 0x18, 0x40, 0x00,
>-  0x1c, 0xa2, 0x0e, 0x51, 0x00, 0x00,
>-  0xa6, 0x27, 0xa9, 0x4a, 0x40, 0x00
>-};
>-
>-const uint8_t kMaskRandom34_33[198] = {
>-  0x25, 0x2c, 0x12, 0x96, 0x00, 0x00,
>-  0x8a, 0x91, 0x45, 0x48, 0x80, 0x00,
>-  0x91, 0xc0, 0xc8, 0xe0, 0x40, 0x00,
>-  0x68, 0x06, 0xb4, 0x03, 0x40, 0x00,
>-  0x32, 0xc8, 0x19, 0x64, 0x00, 0x00,
>-  0x43, 0x45, 0x21, 0xa2, 0x80, 0x00,
>-  0xc4, 0x30, 0xe2, 0x18, 0x40, 0x00,
>-  0x1c, 0xa2, 0x0e, 0x51, 0x00, 0x00,
>-  0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
>-  0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
>-  0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
>-  0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
>-  0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
>-  0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
>-  0x83, 0x34, 0x41, 0x9a, 0x00, 0x00,
>-  0x0a, 0x1c, 0x05, 0x0e, 0x00, 0x00,
>-  0x25, 0x2c, 0x12, 0x96, 0x00, 0x00,
>-  0x8a, 0x91, 0x45, 0x48, 0x80, 0x00,
>-  0x91, 0xc0, 0xc8, 0xe0, 0x40, 0x00,
>-  0x68, 0x06, 0xb4, 0x03, 0x40, 0x00,
>-  0x32, 0xc8, 0x19, 0x64, 0x00, 0x00,
>-  0x43, 0x45, 0x21, 0xa2, 0x80, 0x00,
>-  0xc4, 0x30, 0xe2, 0x18, 0x40, 0x00,
>-  0x1c, 0xa2, 0x0e, 0x51, 0x00, 0x00,
>-  0x25, 0x4c, 0x12, 0xa6, 0x00, 0x00,
>-  0x8a, 0x66, 0x45, 0x33, 0x00, 0x00,
>-  0x91, 0x91, 0x48, 0xc8, 0x80, 0x00,
>-  0x68, 0x42, 0xb4, 0x21, 0x40, 0x00,
>-  0x32, 0xa4, 0x19, 0x52, 0x00, 0x00,
>-  0x43, 0x13, 0x21, 0x89, 0x80, 0x00,
>-  0xc4, 0x30, 0xe2, 0x18, 0x40, 0x00,
>-  0x1c, 0x88, 0x8e, 0x44, 0x40, 0x00,
>-  0x3c, 0x09, 0x1e, 0x04, 0x80, 0x00
>-};
>-
>-const uint8_t kMaskRandom34_34[204] = {
>-  0x25, 0x2c, 0x12, 0x96, 0x00, 0x00,
>-  0x8a, 0x91, 0x45, 0x48, 0x80, 0x00,
>-  0x91, 0xc0, 0xc8, 0xe0, 0x40, 0x00,
>-  0x68, 0x06, 0xb4, 0x03, 0x40, 0x00,
>-  0x32, 0xc8, 0x19, 0x64, 0x00, 0x00,
>-  0x43, 0x45, 0x21, 0xa2, 0x80, 0x00,
>-  0xc4, 0x30, 0xe2, 0x18, 0x40, 0x00,
>-  0x1c, 0xa2, 0x0e, 0x51, 0x00, 0x00,
>-  0x25, 0x4c, 0x12, 0xa6, 0x00, 0x00,
>-  0x8a, 0x66, 0x45, 0x33, 0x00, 0x00,
>-  0x91, 0x91, 0x48, 0xc8, 0x80, 0x00,
>-  0x68, 0x42, 0xb4, 0x21, 0x40, 0x00,
>-  0x32, 0xa4, 0x19, 0x52, 0x00, 0x00,
>-  0x43, 0x13, 0x21, 0x89, 0x80, 0x00,
>-  0xc4, 0x30, 0xe2, 0x18, 0x40, 0x00,
>-  0x1c, 0x88, 0x8e, 0x44, 0x40, 0x00,
>-  0x3c, 0x09, 0x1e, 0x04, 0x80, 0x00,
>-  0x25, 0x2c, 0x12, 0x96, 0x00, 0x00,
>-  0x8a, 0x91, 0x45, 0x48, 0x80, 0x00,
>-  0x91, 0xc0, 0xc8, 0xe0, 0x40, 0x00,
>-  0x68, 0x06, 0xb4, 0x03, 0x40, 0x00,
>-  0x32, 0xc8, 0x19, 0x64, 0x00, 0x00,
>-  0x43, 0x45, 0x21, 0xa2, 0x80, 0x00,
>-  0xc4, 0x30, 0xe2, 0x18, 0x40, 0x00,
>-  0x1c, 0xa2, 0x0e, 0x51, 0x00, 0x00,
>-  0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
>-  0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
>-  0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
>-  0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
>-  0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
>-  0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
>-  0x83, 0x34, 0x41, 0x9a, 0x00, 0x00,
>-  0x0a, 0x1c, 0x05, 0x0e, 0x00, 0x00,
>-  0x30, 0x3c, 0xb3, 0xe6, 0x80, 0x00
>-};
>-
>-const uint8_t kMaskRandom34_4[24] = {
>-  0xca, 0xec, 0x65, 0x76, 0x00, 0x00,
>-  0xa9, 0x67, 0x54, 0xb3, 0x80, 0x00,
>-  0x3a, 0xb1, 0x9d, 0x58, 0xc0, 0x00,
>-  0x55, 0x5a, 0xaa, 0xad, 0x40, 0x00
>-};
>-
>-const uint8_t kMaskRandom34_5[30] = {
>-  0x55, 0x44, 0xaa, 0xa2, 0x40, 0x00,
>-  0x2a, 0x66, 0x15, 0x33, 0x00, 0x00,
>-  0x25, 0xa1, 0x92, 0xd0, 0xc0, 0x00,
>-  0xe2, 0x12, 0xf1, 0x09, 0x40, 0x00,
>-  0x99, 0x98, 0x4c, 0xcc, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom34_6[36] = {
>-  0xd1, 0x4c, 0x68, 0xa6, 0x00, 0x00,
>-  0xa2, 0xc5, 0x51, 0x62, 0x80, 0x00,
>-  0x95, 0x30, 0xca, 0x98, 0x40, 0x00,
>-  0xca, 0x0a, 0xe5, 0x05, 0x40, 0x00,
>-  0xa4, 0xaa, 0x52, 0x55, 0x00, 0x00,
>-  0x78, 0x15, 0x3c, 0x0a, 0x80, 0x00
>-};
>-
>-const uint8_t kMaskRandom34_7[42] = {
>-  0x15, 0x44, 0x8a, 0xa2, 0x40, 0x00,
>-  0x8a, 0x23, 0x45, 0x11, 0x80, 0x00,
>-  0x85, 0x91, 0x42, 0xc8, 0x80, 0x00,
>-  0x32, 0x0a, 0x99, 0x05, 0x40, 0x00,
>-  0x58, 0x34, 0x2c, 0x1a, 0x00, 0x00,
>-  0x2c, 0x0d, 0x16, 0x06, 0x80, 0x00,
>-  0x43, 0xc8, 0x21, 0xe4, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom34_8[48] = {
>-  0x64, 0x16, 0x32, 0x0b, 0x00, 0x00,
>-  0xa2, 0xc2, 0x51, 0x61, 0x00, 0x00,
>-  0x51, 0x60, 0xa8, 0xb0, 0x40, 0x00,
>-  0x4a, 0x85, 0x25, 0x42, 0x80, 0x00,
>-  0x38, 0x4c, 0x1c, 0x26, 0x00, 0x00,
>-  0x89, 0x29, 0x44, 0x94, 0x80, 0x00,
>-  0x07, 0x11, 0x83, 0x88, 0xc0, 0x00,
>-  0x94, 0xb0, 0x4a, 0x58, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom34_9[54] = {
>-  0x8e, 0xcc, 0x47, 0x66, 0x00, 0x00,
>-  0x6a, 0x2b, 0x35, 0x15, 0x80, 0x00,
>-  0x36, 0x32, 0x9b, 0x19, 0x40, 0x00,
>-  0xd1, 0x25, 0xe8, 0x92, 0xc0, 0x00,
>-  0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
>-  0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
>-  0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
>-  0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
>-  0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom35_1[6] = {
>-  0xff, 0xff, 0xff, 0xff, 0xe0, 0x00
>-};
>-
>-const uint8_t kMaskRandom35_10[60] = {
>-  0x55, 0x8c, 0xc6, 0x63, 0x20, 0x00,
>-  0xaa, 0x27, 0x13, 0x89, 0xc0, 0x00,
>-  0xa5, 0x32, 0x99, 0x4c, 0xa0, 0x00,
>-  0x62, 0x61, 0xb0, 0xd8, 0x60, 0x00,
>-  0x3c, 0x5c, 0x2e, 0x17, 0x00, 0x00,
>-  0x8e, 0xcc, 0x66, 0x33, 0x00, 0x00,
>-  0x6a, 0x2b, 0x15, 0x8a, 0xc0, 0x00,
>-  0x36, 0x32, 0x99, 0x4c, 0xa0, 0x00,
>-  0xd1, 0x25, 0x92, 0xc9, 0x60, 0x00,
>-  0xc8, 0x02, 0xfe, 0xce, 0xe0, 0x00
>-};
>-
>-const uint8_t kMaskRandom35_11[66] = {
>-  0x55, 0x8c, 0xc6, 0x63, 0x20, 0x00,
>-  0xaa, 0x27, 0x13, 0x89, 0xc0, 0x00,
>-  0xa5, 0x32, 0x99, 0x4c, 0xa0, 0x00,
>-  0x62, 0x61, 0xb0, 0xd8, 0x60, 0x00,
>-  0x3c, 0x5c, 0x2e, 0x17, 0x00, 0x00,
>-  0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
>-  0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
>-  0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
>-  0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
>-  0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
>-  0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00
>-};
>-
>-const uint8_t kMaskRandom35_12[72] = {
>-  0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
>-  0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
>-  0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
>-  0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
>-  0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
>-  0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00,
>-  0x55, 0x8c, 0xc6, 0x63, 0x20, 0x00,
>-  0xaa, 0x27, 0x13, 0x89, 0xc0, 0x00,
>-  0xa5, 0x32, 0x99, 0x4c, 0xa0, 0x00,
>-  0x62, 0x61, 0xb0, 0xd8, 0x60, 0x00,
>-  0x3c, 0x5c, 0x2e, 0x17, 0x00, 0x00,
>-  0x51, 0x35, 0x2d, 0x86, 0x20, 0x00
>-};
>-
>-const uint8_t kMaskRandom35_13[78] = {
>-  0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
>-  0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
>-  0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
>-  0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
>-  0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
>-  0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00,
>-  0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
>-  0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
>-  0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
>-  0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
>-  0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
>-  0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
>-  0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom35_14[84] = {
>-  0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
>-  0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
>-  0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
>-  0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
>-  0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
>-  0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
>-  0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00,
>-  0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
>-  0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
>-  0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
>-  0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
>-  0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
>-  0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00,
>-  0xb0, 0xde, 0xbf, 0xa7, 0xe0, 0x00
>-};
>-
>-const uint8_t kMaskRandom35_15[90] = {
>-  0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
>-  0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
>-  0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
>-  0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
>-  0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
>-  0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
>-  0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00,
>-  0x25, 0x2c, 0x16, 0x0b, 0x00, 0x00,
>-  0x8a, 0x91, 0x48, 0xa4, 0x40, 0x00,
>-  0x91, 0xc0, 0xe0, 0x70, 0x20, 0x00,
>-  0x68, 0x06, 0x83, 0x41, 0xa0, 0x00,
>-  0x32, 0xc8, 0x64, 0x32, 0x00, 0x00,
>-  0x43, 0x45, 0x22, 0x91, 0x40, 0x00,
>-  0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
>-  0x1c, 0xa2, 0x51, 0x28, 0x80, 0x00
>-};
>-
>-const uint8_t kMaskRandom35_16[96] = {
>-  0x25, 0x2c, 0x16, 0x0b, 0x00, 0x00,
>-  0x8a, 0x91, 0x48, 0xa4, 0x40, 0x00,
>-  0x91, 0xc0, 0xe0, 0x70, 0x20, 0x00,
>-  0x68, 0x06, 0x83, 0x41, 0xa0, 0x00,
>-  0x32, 0xc8, 0x64, 0x32, 0x00, 0x00,
>-  0x43, 0x45, 0x22, 0x91, 0x40, 0x00,
>-  0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
>-  0x1c, 0xa2, 0x51, 0x28, 0x80, 0x00,
>-  0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
>-  0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
>-  0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
>-  0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
>-  0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
>-  0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
>-  0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00,
>-  0x0a, 0x1c, 0x77, 0xf9, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom35_17[102] = {
>-  0x25, 0x2c, 0x16, 0x0b, 0x00, 0x00,
>-  0x8a, 0x91, 0x48, 0xa4, 0x40, 0x00,
>-  0x91, 0xc0, 0xe0, 0x70, 0x20, 0x00,
>-  0x68, 0x06, 0x83, 0x41, 0xa0, 0x00,
>-  0x32, 0xc8, 0x64, 0x32, 0x00, 0x00,
>-  0x43, 0x45, 0x22, 0x91, 0x40, 0x00,
>-  0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
>-  0x1c, 0xa2, 0x51, 0x28, 0x80, 0x00,
>-  0x25, 0x4c, 0x26, 0x13, 0x00, 0x00,
>-  0x8a, 0x66, 0x33, 0x19, 0x80, 0x00,
>-  0x91, 0x91, 0x48, 0xa4, 0x40, 0x00,
>-  0x68, 0x42, 0xa1, 0x50, 0xa0, 0x00,
>-  0x32, 0xa4, 0x52, 0x29, 0x00, 0x00,
>-  0x43, 0x13, 0x09, 0x84, 0xc0, 0x00,
>-  0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
>-  0x1c, 0x88, 0xc4, 0x62, 0x20, 0x00,
>-  0x3c, 0x09, 0x04, 0x82, 0x40, 0x00
>-};
>-
>-const uint8_t kMaskRandom35_18[108] = {
>-  0x8e, 0xcc, 0x22, 0x51, 0x20, 0x00,
>-  0x6a, 0x2b, 0x33, 0x13, 0x00, 0x00,
>-  0x36, 0x32, 0xc8, 0x24, 0xa0, 0x00,
>-  0xd1, 0x25, 0x80, 0xd2, 0xc0, 0x00,
>-  0x55, 0x8c, 0x87, 0x09, 0x40, 0x00,
>-  0xaa, 0x27, 0x09, 0x85, 0x80, 0x00,
>-  0xa5, 0x32, 0x90, 0x68, 0x20, 0x00,
>-  0x62, 0x61, 0xe1, 0x28, 0x80, 0x00,
>-  0x3c, 0x5c, 0x14, 0x86, 0x40, 0x00,
>-  0x64, 0x16, 0x0b, 0x05, 0x80, 0x00,
>-  0xa2, 0xc2, 0x61, 0x30, 0x80, 0x00,
>-  0x51, 0x60, 0xb0, 0x58, 0x20, 0x00,
>-  0x4a, 0x85, 0x42, 0xa1, 0x40, 0x00,
>-  0x38, 0x4c, 0x26, 0x13, 0x00, 0x00,
>-  0x89, 0x29, 0x14, 0x8a, 0x40, 0x00,
>-  0x07, 0x11, 0x88, 0xc4, 0x60, 0x00,
>-  0x94, 0xb0, 0x58, 0x2c, 0x00, 0x00,
>-  0x40, 0xc9, 0x65, 0xbe, 0xc0, 0x00
>-};
>-
>-const uint8_t kMaskRandom35_19[114] = {
>-  0x8e, 0xcc, 0x22, 0x51, 0x20, 0x00,
>-  0x6a, 0x2b, 0x33, 0x13, 0x00, 0x00,
>-  0x36, 0x32, 0xc8, 0x24, 0xa0, 0x00,
>-  0xd1, 0x25, 0x80, 0xd2, 0xc0, 0x00,
>-  0x55, 0x8c, 0x87, 0x09, 0x40, 0x00,
>-  0xaa, 0x27, 0x09, 0x85, 0x80, 0x00,
>-  0xa5, 0x32, 0x90, 0x68, 0x20, 0x00,
>-  0x62, 0x61, 0xe1, 0x28, 0x80, 0x00,
>-  0x3c, 0x5c, 0x14, 0x86, 0x40, 0x00,
>-  0x55, 0x8c, 0xc6, 0x63, 0x20, 0x00,
>-  0xaa, 0x27, 0x13, 0x89, 0xc0, 0x00,
>-  0xa5, 0x32, 0x99, 0x4c, 0xa0, 0x00,
>-  0x62, 0x61, 0xb0, 0xd8, 0x60, 0x00,
>-  0x3c, 0x5c, 0x2e, 0x17, 0x00, 0x00,
>-  0x8e, 0xcc, 0x66, 0x33, 0x00, 0x00,
>-  0x6a, 0x2b, 0x15, 0x8a, 0xc0, 0x00,
>-  0x36, 0x32, 0x99, 0x4c, 0xa0, 0x00,
>-  0xd1, 0x25, 0x92, 0xc9, 0x60, 0x00,
>-  0xc8, 0x02, 0xfe, 0xce, 0xe0, 0x00
>-};
>-
>-const uint8_t kMaskRandom35_2[12] = {
>-  0xce, 0xce, 0x67, 0x33, 0x80, 0x00,
>-  0xb9, 0x39, 0x9c, 0xce, 0x60, 0x00
>-};
>-
>-const uint8_t kMaskRandom35_20[120] = {
>-  0x55, 0x8c, 0xc6, 0x63, 0x20, 0x00,
>-  0xaa, 0x27, 0x13, 0x89, 0xc0, 0x00,
>-  0xa5, 0x32, 0x99, 0x4c, 0xa0, 0x00,
>-  0x62, 0x61, 0xb0, 0xd8, 0x60, 0x00,
>-  0x3c, 0x5c, 0x2e, 0x17, 0x00, 0x00,
>-  0x8e, 0xcc, 0x66, 0x33, 0x00, 0x00,
>-  0x6a, 0x2b, 0x15, 0x8a, 0xc0, 0x00,
>-  0x36, 0x32, 0x99, 0x4c, 0xa0, 0x00,
>-  0xd1, 0x25, 0x92, 0xc9, 0x60, 0x00,
>-  0xc8, 0x02, 0xfe, 0xce, 0xe0, 0x00,
>-  0x8e, 0xcc, 0x22, 0x51, 0x20, 0x00,
>-  0x6a, 0x2b, 0x33, 0x13, 0x00, 0x00,
>-  0x36, 0x32, 0xc8, 0x24, 0xa0, 0x00,
>-  0xd1, 0x25, 0x80, 0xd2, 0xc0, 0x00,
>-  0x55, 0x8c, 0x87, 0x09, 0x40, 0x00,
>-  0xaa, 0x27, 0x09, 0x85, 0x80, 0x00,
>-  0xa5, 0x32, 0x90, 0x68, 0x20, 0x00,
>-  0x62, 0x61, 0xe1, 0x28, 0x80, 0x00,
>-  0x3c, 0x5c, 0x14, 0x86, 0x40, 0x00,
>-  0x63, 0x36, 0x5c, 0xd3, 0x80, 0x00
>-};
>-
>-const uint8_t kMaskRandom35_21[126] = {
>-  0x55, 0x8c, 0xc6, 0x63, 0x20, 0x00,
>-  0xaa, 0x27, 0x13, 0x89, 0xc0, 0x00,
>-  0xa5, 0x32, 0x99, 0x4c, 0xa0, 0x00,
>-  0x62, 0x61, 0xb0, 0xd8, 0x60, 0x00,
>-  0x3c, 0x5c, 0x2e, 0x17, 0x00, 0x00,
>-  0x8e, 0xcc, 0x66, 0x33, 0x00, 0x00,
>-  0x6a, 0x2b, 0x15, 0x8a, 0xc0, 0x00,
>-  0x36, 0x32, 0x99, 0x4c, 0xa0, 0x00,
>-  0xd1, 0x25, 0x92, 0xc9, 0x60, 0x00,
>-  0xc8, 0x02, 0xfe, 0xce, 0xe0, 0x00,
>-  0x55, 0x8c, 0xc6, 0x63, 0x20, 0x00,
>-  0xaa, 0x27, 0x13, 0x89, 0xc0, 0x00,
>-  0xa5, 0x32, 0x99, 0x4c, 0xa0, 0x00,
>-  0x62, 0x61, 0xb0, 0xd8, 0x60, 0x00,
>-  0x3c, 0x5c, 0x2e, 0x17, 0x00, 0x00,
>-  0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
>-  0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
>-  0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
>-  0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
>-  0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
>-  0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00
>-};
>-
>-const uint8_t kMaskRandom35_22[132] = {
>-  0x55, 0x8c, 0xc6, 0x63, 0x20, 0x00,
>-  0xaa, 0x27, 0x13, 0x89, 0xc0, 0x00,
>-  0xa5, 0x32, 0x99, 0x4c, 0xa0, 0x00,
>-  0x62, 0x61, 0xb0, 0xd8, 0x60, 0x00,
>-  0x3c, 0x5c, 0x2e, 0x17, 0x00, 0x00,
>-  0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
>-  0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
>-  0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
>-  0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
>-  0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
>-  0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00,
>-  0x55, 0x8c, 0xc6, 0x63, 0x20, 0x00,
>-  0xaa, 0x27, 0x13, 0x89, 0xc0, 0x00,
>-  0xa5, 0x32, 0x99, 0x4c, 0xa0, 0x00,
>-  0x62, 0x61, 0xb0, 0xd8, 0x60, 0x00,
>-  0x3c, 0x5c, 0x2e, 0x17, 0x00, 0x00,
>-  0x8e, 0xcc, 0x66, 0x33, 0x00, 0x00,
>-  0x6a, 0x2b, 0x15, 0x8a, 0xc0, 0x00,
>-  0x36, 0x32, 0x99, 0x4c, 0xa0, 0x00,
>-  0xd1, 0x25, 0x92, 0xc9, 0x60, 0x00,
>-  0xc8, 0x02, 0xfe, 0xce, 0xe0, 0x00,
>-  0x84, 0xc7, 0xbc, 0xcc, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom35_23[138] = {
>-  0x55, 0x8c, 0xc6, 0x63, 0x20, 0x00,
>-  0xaa, 0x27, 0x13, 0x89, 0xc0, 0x00,
>-  0xa5, 0x32, 0x99, 0x4c, 0xa0, 0x00,
>-  0x62, 0x61, 0xb0, 0xd8, 0x60, 0x00,
>-  0x3c, 0x5c, 0x2e, 0x17, 0x00, 0x00,
>-  0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
>-  0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
>-  0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
>-  0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
>-  0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
>-  0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00,
>-  0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
>-  0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
>-  0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
>-  0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
>-  0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
>-  0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00,
>-  0x55, 0x8c, 0xc6, 0x63, 0x20, 0x00,
>-  0xaa, 0x27, 0x13, 0x89, 0xc0, 0x00,
>-  0xa5, 0x32, 0x99, 0x4c, 0xa0, 0x00,
>-  0x62, 0x61, 0xb0, 0xd8, 0x60, 0x00,
>-  0x3c, 0x5c, 0x2e, 0x17, 0x00, 0x00,
>-  0x51, 0x35, 0x2d, 0x86, 0x20, 0x00
>-};
>-
>-const uint8_t kMaskRandom35_24[144] = {
>-  0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
>-  0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
>-  0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
>-  0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
>-  0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
>-  0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00,
>-  0x55, 0x8c, 0xc6, 0x63, 0x20, 0x00,
>-  0xaa, 0x27, 0x13, 0x89, 0xc0, 0x00,
>-  0xa5, 0x32, 0x99, 0x4c, 0xa0, 0x00,
>-  0x62, 0x61, 0xb0, 0xd8, 0x60, 0x00,
>-  0x3c, 0x5c, 0x2e, 0x17, 0x00, 0x00,
>-  0x51, 0x35, 0x2d, 0x86, 0x20, 0x00,
>-  0x55, 0x8c, 0xc6, 0x63, 0x20, 0x00,
>-  0xaa, 0x27, 0x13, 0x89, 0xc0, 0x00,
>-  0xa5, 0x32, 0x99, 0x4c, 0xa0, 0x00,
>-  0x62, 0x61, 0xb0, 0xd8, 0x60, 0x00,
>-  0x3c, 0x5c, 0x2e, 0x17, 0x00, 0x00,
>-  0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
>-  0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
>-  0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
>-  0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
>-  0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
>-  0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00,
>-  0x0d, 0xfb, 0x06, 0x89, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom35_25[150] = {
>-  0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
>-  0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
>-  0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
>-  0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
>-  0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
>-  0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00,
>-  0x55, 0x8c, 0xc6, 0x63, 0x20, 0x00,
>-  0xaa, 0x27, 0x13, 0x89, 0xc0, 0x00,
>-  0xa5, 0x32, 0x99, 0x4c, 0xa0, 0x00,
>-  0x62, 0x61, 0xb0, 0xd8, 0x60, 0x00,
>-  0x3c, 0x5c, 0x2e, 0x17, 0x00, 0x00,
>-  0x51, 0x35, 0x2d, 0x86, 0x20, 0x00,
>-  0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
>-  0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
>-  0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
>-  0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
>-  0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
>-  0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00,
>-  0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
>-  0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
>-  0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
>-  0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
>-  0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
>-  0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
>-  0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom35_26[156] = {
>-  0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
>-  0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
>-  0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
>-  0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
>-  0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
>-  0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00,
>-  0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
>-  0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
>-  0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
>-  0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
>-  0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
>-  0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
>-  0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00,
>-  0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
>-  0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
>-  0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
>-  0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
>-  0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
>-  0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00,
>-  0x55, 0x8c, 0xc6, 0x63, 0x20, 0x00,
>-  0xaa, 0x27, 0x13, 0x89, 0xc0, 0x00,
>-  0xa5, 0x32, 0x99, 0x4c, 0xa0, 0x00,
>-  0x62, 0x61, 0xb0, 0xd8, 0x60, 0x00,
>-  0x3c, 0x5c, 0x2e, 0x17, 0x00, 0x00,
>-  0x51, 0x35, 0x2d, 0x86, 0x20, 0x00,
>-  0xc4, 0x57, 0x70, 0x47, 0x40, 0x00
>-};
>-
>-const uint8_t kMaskRandom35_27[162] = {
>-  0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
>-  0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
>-  0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
>-  0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
>-  0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
>-  0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00,
>-  0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
>-  0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
>-  0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
>-  0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
>-  0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
>-  0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
>-  0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00,
>-  0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
>-  0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
>-  0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
>-  0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
>-  0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
>-  0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
>-  0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00,
>-  0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
>-  0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
>-  0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
>-  0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
>-  0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
>-  0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00,
>-  0xb0, 0xde, 0xbf, 0xa7, 0xe0, 0x00
>-};
>-
>-const uint8_t kMaskRandom35_28[168] = {
>-  0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
>-  0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
>-  0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
>-  0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
>-  0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
>-  0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
>-  0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00,
>-  0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
>-  0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
>-  0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
>-  0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
>-  0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
>-  0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00,
>-  0xb0, 0xde, 0xbf, 0xa7, 0xe0, 0x00,
>-  0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
>-  0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
>-  0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
>-  0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
>-  0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
>-  0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00,
>-  0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
>-  0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
>-  0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
>-  0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
>-  0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
>-  0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
>-  0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00,
>-  0x34, 0x4a, 0x80, 0x94, 0x40, 0x00
>-};
>-
>-const uint8_t kMaskRandom35_29[174] = {
>-  0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
>-  0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
>-  0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
>-  0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
>-  0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
>-  0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
>-  0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00,
>-  0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
>-  0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
>-  0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
>-  0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
>-  0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
>-  0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00,
>-  0xb0, 0xde, 0xbf, 0xa7, 0xe0, 0x00,
>-  0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
>-  0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
>-  0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
>-  0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
>-  0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
>-  0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
>-  0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00,
>-  0x25, 0x2c, 0x16, 0x0b, 0x00, 0x00,
>-  0x8a, 0x91, 0x48, 0xa4, 0x40, 0x00,
>-  0x91, 0xc0, 0xe0, 0x70, 0x20, 0x00,
>-  0x68, 0x06, 0x83, 0x41, 0xa0, 0x00,
>-  0x32, 0xc8, 0x64, 0x32, 0x00, 0x00,
>-  0x43, 0x45, 0x22, 0x91, 0x40, 0x00,
>-  0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
>-  0x1c, 0xa2, 0x51, 0x28, 0x80, 0x00
>-};
>-
>-const uint8_t kMaskRandom35_3[18] = {
>-  0xcd, 0xcc, 0x66, 0x33, 0x00, 0x00,
>-  0x97, 0x27, 0x13, 0x8a, 0xc0, 0x00,
>-  0xb8, 0xd1, 0xc9, 0x64, 0xa0, 0x00
>-};
>-
>-const uint8_t kMaskRandom35_30[180] = {
>-  0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
>-  0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
>-  0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
>-  0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
>-  0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
>-  0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
>-  0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00,
>-  0x25, 0x2c, 0x16, 0x0b, 0x00, 0x00,
>-  0x8a, 0x91, 0x48, 0xa4, 0x40, 0x00,
>-  0x91, 0xc0, 0xe0, 0x70, 0x20, 0x00,
>-  0x68, 0x06, 0x83, 0x41, 0xa0, 0x00,
>-  0x32, 0xc8, 0x64, 0x32, 0x00, 0x00,
>-  0x43, 0x45, 0x22, 0x91, 0x40, 0x00,
>-  0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
>-  0x1c, 0xa2, 0x51, 0x28, 0x80, 0x00,
>-  0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
>-  0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
>-  0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
>-  0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
>-  0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
>-  0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
>-  0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00,
>-  0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
>-  0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
>-  0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
>-  0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
>-  0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
>-  0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00,
>-  0xb0, 0xde, 0xbf, 0xa7, 0xe0, 0x00,
>-  0x32, 0x1b, 0x9f, 0x09, 0x20, 0x00
>-};
>-
>-const uint8_t kMaskRandom35_31[186] = {
>-  0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
>-  0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
>-  0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
>-  0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
>-  0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
>-  0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
>-  0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00,
>-  0x25, 0x2c, 0x16, 0x0b, 0x00, 0x00,
>-  0x8a, 0x91, 0x48, 0xa4, 0x40, 0x00,
>-  0x91, 0xc0, 0xe0, 0x70, 0x20, 0x00,
>-  0x68, 0x06, 0x83, 0x41, 0xa0, 0x00,
>-  0x32, 0xc8, 0x64, 0x32, 0x00, 0x00,
>-  0x43, 0x45, 0x22, 0x91, 0x40, 0x00,
>-  0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
>-  0x1c, 0xa2, 0x51, 0x28, 0x80, 0x00,
>-  0x25, 0x2c, 0x16, 0x0b, 0x00, 0x00,
>-  0x8a, 0x91, 0x48, 0xa4, 0x40, 0x00,
>-  0x91, 0xc0, 0xe0, 0x70, 0x20, 0x00,
>-  0x68, 0x06, 0x83, 0x41, 0xa0, 0x00,
>-  0x32, 0xc8, 0x64, 0x32, 0x00, 0x00,
>-  0x43, 0x45, 0x22, 0x91, 0x40, 0x00,
>-  0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
>-  0x1c, 0xa2, 0x51, 0x28, 0x80, 0x00,
>-  0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
>-  0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
>-  0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
>-  0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
>-  0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
>-  0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
>-  0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00,
>-  0x0a, 0x1c, 0x77, 0xf9, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom35_32[192] = {
>-  0x25, 0x2c, 0x16, 0x0b, 0x00, 0x00,
>-  0x8a, 0x91, 0x48, 0xa4, 0x40, 0x00,
>-  0x91, 0xc0, 0xe0, 0x70, 0x20, 0x00,
>-  0x68, 0x06, 0x83, 0x41, 0xa0, 0x00,
>-  0x32, 0xc8, 0x64, 0x32, 0x00, 0x00,
>-  0x43, 0x45, 0x22, 0x91, 0x40, 0x00,
>-  0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
>-  0x1c, 0xa2, 0x51, 0x28, 0x80, 0x00,
>-  0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
>-  0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
>-  0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
>-  0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
>-  0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
>-  0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
>-  0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00,
>-  0x0a, 0x1c, 0x77, 0xf9, 0x00, 0x00,
>-  0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
>-  0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
>-  0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
>-  0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
>-  0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
>-  0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
>-  0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00,
>-  0x25, 0x2c, 0x16, 0x0b, 0x00, 0x00,
>-  0x8a, 0x91, 0x48, 0xa4, 0x40, 0x00,
>-  0x91, 0xc0, 0xe0, 0x70, 0x20, 0x00,
>-  0x68, 0x06, 0x83, 0x41, 0xa0, 0x00,
>-  0x32, 0xc8, 0x64, 0x32, 0x00, 0x00,
>-  0x43, 0x45, 0x22, 0x91, 0x40, 0x00,
>-  0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
>-  0x1c, 0xa2, 0x51, 0x28, 0x80, 0x00,
>-  0xeb, 0x31, 0x7b, 0x80, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom35_33[198] = {
>-  0x25, 0x2c, 0x16, 0x0b, 0x00, 0x00,
>-  0x8a, 0x91, 0x48, 0xa4, 0x40, 0x00,
>-  0x91, 0xc0, 0xe0, 0x70, 0x20, 0x00,
>-  0x68, 0x06, 0x83, 0x41, 0xa0, 0x00,
>-  0x32, 0xc8, 0x64, 0x32, 0x00, 0x00,
>-  0x43, 0x45, 0x22, 0x91, 0x40, 0x00,
>-  0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
>-  0x1c, 0xa2, 0x51, 0x28, 0x80, 0x00,
>-  0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
>-  0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
>-  0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
>-  0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
>-  0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
>-  0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
>-  0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00,
>-  0x0a, 0x1c, 0x77, 0xf9, 0x00, 0x00,
>-  0x25, 0x2c, 0x16, 0x0b, 0x00, 0x00,
>-  0x8a, 0x91, 0x48, 0xa4, 0x40, 0x00,
>-  0x91, 0xc0, 0xe0, 0x70, 0x20, 0x00,
>-  0x68, 0x06, 0x83, 0x41, 0xa0, 0x00,
>-  0x32, 0xc8, 0x64, 0x32, 0x00, 0x00,
>-  0x43, 0x45, 0x22, 0x91, 0x40, 0x00,
>-  0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
>-  0x1c, 0xa2, 0x51, 0x28, 0x80, 0x00,
>-  0x25, 0x4c, 0x26, 0x13, 0x00, 0x00,
>-  0x8a, 0x66, 0x33, 0x19, 0x80, 0x00,
>-  0x91, 0x91, 0x48, 0xa4, 0x40, 0x00,
>-  0x68, 0x42, 0xa1, 0x50, 0xa0, 0x00,
>-  0x32, 0xa4, 0x52, 0x29, 0x00, 0x00,
>-  0x43, 0x13, 0x09, 0x84, 0xc0, 0x00,
>-  0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
>-  0x1c, 0x88, 0xc4, 0x62, 0x20, 0x00,
>-  0x3c, 0x09, 0x04, 0x82, 0x40, 0x00
>-};
>-
>-const uint8_t kMaskRandom35_34[204] = {
>-  0x25, 0x2c, 0x16, 0x0b, 0x00, 0x00,
>-  0x8a, 0x91, 0x48, 0xa4, 0x40, 0x00,
>-  0x91, 0xc0, 0xe0, 0x70, 0x20, 0x00,
>-  0x68, 0x06, 0x83, 0x41, 0xa0, 0x00,
>-  0x32, 0xc8, 0x64, 0x32, 0x00, 0x00,
>-  0x43, 0x45, 0x22, 0x91, 0x40, 0x00,
>-  0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
>-  0x1c, 0xa2, 0x51, 0x28, 0x80, 0x00,
>-  0x25, 0x4c, 0x26, 0x13, 0x00, 0x00,
>-  0x8a, 0x66, 0x33, 0x19, 0x80, 0x00,
>-  0x91, 0x91, 0x48, 0xa4, 0x40, 0x00,
>-  0x68, 0x42, 0xa1, 0x50, 0xa0, 0x00,
>-  0x32, 0xa4, 0x52, 0x29, 0x00, 0x00,
>-  0x43, 0x13, 0x09, 0x84, 0xc0, 0x00,
>-  0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
>-  0x1c, 0x88, 0xc4, 0x62, 0x20, 0x00,
>-  0x3c, 0x09, 0x04, 0x82, 0x40, 0x00,
>-  0x25, 0x2c, 0x16, 0x0b, 0x00, 0x00,
>-  0x8a, 0x91, 0x48, 0xa4, 0x40, 0x00,
>-  0x91, 0xc0, 0xe0, 0x70, 0x20, 0x00,
>-  0x68, 0x06, 0x83, 0x41, 0xa0, 0x00,
>-  0x32, 0xc8, 0x64, 0x32, 0x00, 0x00,
>-  0x43, 0x45, 0x22, 0x91, 0x40, 0x00,
>-  0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
>-  0x1c, 0xa2, 0x51, 0x28, 0x80, 0x00,
>-  0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
>-  0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
>-  0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
>-  0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
>-  0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
>-  0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
>-  0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00,
>-  0x0a, 0x1c, 0x77, 0xf9, 0x00, 0x00,
>-  0x70, 0x07, 0xcd, 0x8c, 0xc0, 0x00
>-};
>-
>-const uint8_t kMaskRandom35_35[210] = {
>-  0x25, 0x2c, 0x16, 0x0b, 0x00, 0x00,
>-  0x8a, 0x91, 0x48, 0xa4, 0x40, 0x00,
>-  0x91, 0xc0, 0xe0, 0x70, 0x20, 0x00,
>-  0x68, 0x06, 0x83, 0x41, 0xa0, 0x00,
>-  0x32, 0xc8, 0x64, 0x32, 0x00, 0x00,
>-  0x43, 0x45, 0x22, 0x91, 0x40, 0x00,
>-  0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
>-  0x1c, 0xa2, 0x51, 0x28, 0x80, 0x00,
>-  0x25, 0x4c, 0x26, 0x13, 0x00, 0x00,
>-  0x8a, 0x66, 0x33, 0x19, 0x80, 0x00,
>-  0x91, 0x91, 0x48, 0xa4, 0x40, 0x00,
>-  0x68, 0x42, 0xa1, 0x50, 0xa0, 0x00,
>-  0x32, 0xa4, 0x52, 0x29, 0x00, 0x00,
>-  0x43, 0x13, 0x09, 0x84, 0xc0, 0x00,
>-  0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
>-  0x1c, 0x88, 0xc4, 0x62, 0x20, 0x00,
>-  0x3c, 0x09, 0x04, 0x82, 0x40, 0x00,
>-  0x25, 0x2c, 0x26, 0x13, 0x00, 0x00,
>-  0x8a, 0x91, 0x33, 0x19, 0x80, 0x00,
>-  0x91, 0xc0, 0xc8, 0xa4, 0x40, 0x00,
>-  0x68, 0x06, 0xa1, 0x50, 0xa0, 0x00,
>-  0x32, 0xc8, 0x52, 0x29, 0x00, 0x00,
>-  0x43, 0x45, 0x09, 0x84, 0xc0, 0x00,
>-  0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
>-  0x1c, 0xa2, 0x44, 0x62, 0x20, 0x00,
>-  0x25, 0x4c, 0x04, 0x82, 0x40, 0x00,
>-  0x8a, 0x66, 0x16, 0x0b, 0x00, 0x00,
>-  0x91, 0x91, 0x48, 0xa4, 0x40, 0x00,
>-  0x68, 0x42, 0xe0, 0x70, 0x20, 0x00,
>-  0x32, 0xa4, 0x03, 0x41, 0xa0, 0x00,
>-  0x43, 0x13, 0x64, 0x32, 0x00, 0x00,
>-  0xc4, 0x30, 0xa2, 0x91, 0x40, 0x00,
>-  0x1c, 0x88, 0x98, 0x4c, 0x20, 0x00,
>-  0x3c, 0x09, 0x51, 0x28, 0x80, 0x00,
>-  0xc2, 0x1c, 0x68, 0x01, 0xa0, 0x00
>-};
>-
>-const uint8_t kMaskRandom35_4[24] = {
>-  0xca, 0xec, 0x76, 0x3b, 0x00, 0x00,
>-  0xa9, 0x67, 0x33, 0x99, 0xc0, 0x00,
>-  0x3a, 0xb1, 0xd8, 0xec, 0x60, 0x00,
>-  0x55, 0x5a, 0xad, 0x56, 0xa0, 0x00
>-};
>-
>-const uint8_t kMaskRandom35_5[30] = {
>-  0x55, 0x44, 0xa6, 0x53, 0x20, 0x00,
>-  0x2a, 0x66, 0x33, 0x19, 0x80, 0x00,
>-  0x25, 0xa1, 0x8c, 0xe8, 0x60, 0x00,
>-  0xe2, 0x12, 0xce, 0x44, 0xa0, 0x00,
>-  0x99, 0x98, 0x71, 0xa6, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom35_6[36] = {
>-  0xd1, 0x4c, 0x66, 0x13, 0x00, 0x00,
>-  0xa2, 0xc5, 0x22, 0xb1, 0x40, 0x00,
>-  0x95, 0x30, 0xd8, 0x4c, 0x20, 0x00,
>-  0xca, 0x0a, 0xc5, 0x42, 0xa0, 0x00,
>-  0xa4, 0xaa, 0x14, 0xa9, 0x80, 0x00,
>-  0x78, 0x15, 0x53, 0x05, 0x40, 0x00
>-};
>-
>-const uint8_t kMaskRandom35_7[42] = {
>-  0x15, 0x44, 0xa2, 0x51, 0x20, 0x00,
>-  0x8a, 0x23, 0x11, 0x88, 0xc0, 0x00,
>-  0x85, 0x91, 0x48, 0xa4, 0x40, 0x00,
>-  0x32, 0x0a, 0x85, 0x42, 0xa0, 0x00,
>-  0x58, 0x34, 0x1a, 0x0d, 0x00, 0x00,
>-  0x2c, 0x0d, 0x05, 0x83, 0x40, 0x00,
>-  0x43, 0xc8, 0x70, 0x32, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom35_8[48] = {
>-  0x64, 0x16, 0x0b, 0x05, 0x80, 0x00,
>-  0xa2, 0xc2, 0x61, 0x30, 0x80, 0x00,
>-  0x51, 0x60, 0xb0, 0x58, 0x20, 0x00,
>-  0x4a, 0x85, 0x42, 0xa1, 0x40, 0x00,
>-  0x38, 0x4c, 0x26, 0x13, 0x00, 0x00,
>-  0x89, 0x29, 0x14, 0x8a, 0x40, 0x00,
>-  0x07, 0x11, 0x88, 0xc4, 0x60, 0x00,
>-  0x94, 0xb0, 0x58, 0x2c, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom35_9[54] = {
>-  0x8e, 0xcc, 0x22, 0x51, 0x20, 0x00,
>-  0x6a, 0x2b, 0x33, 0x13, 0x00, 0x00,
>-  0x36, 0x32, 0xc8, 0x24, 0xa0, 0x00,
>-  0xd1, 0x25, 0x80, 0xd2, 0xc0, 0x00,
>-  0x55, 0x8c, 0x87, 0x09, 0x40, 0x00,
>-  0xaa, 0x27, 0x09, 0x85, 0x80, 0x00,
>-  0xa5, 0x32, 0x90, 0x68, 0x20, 0x00,
>-  0x62, 0x61, 0xe1, 0x28, 0x80, 0x00,
>-  0x3c, 0x5c, 0x14, 0x86, 0x40, 0x00
>-};
>-
>-const uint8_t kMaskRandom36_1[6] = {
>-  0xff, 0xff, 0xff, 0xff, 0xf0, 0x00
>-};
>-
>-const uint8_t kMaskRandom36_10[60] = {
>-  0x8c, 0xc6, 0x63, 0x31, 0x90, 0x00,
>-  0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
>-  0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
>-  0x61, 0xb0, 0xd8, 0x6c, 0x30, 0x00,
>-  0x5c, 0x2e, 0x17, 0x0b, 0x80, 0x00,
>-  0xcc, 0x66, 0x33, 0x19, 0x80, 0x00,
>-  0x2b, 0x15, 0x8a, 0xc5, 0x60, 0x00,
>-  0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
>-  0x25, 0x92, 0xc9, 0x64, 0xb0, 0x00,
>-  0xfd, 0x9d, 0xff, 0x67, 0x70, 0x00
>-};
>-
>-const uint8_t kMaskRandom36_11[66] = {
>-  0x8c, 0xc6, 0x63, 0x31, 0x90, 0x00,
>-  0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
>-  0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
>-  0x61, 0xb0, 0xd8, 0x6c, 0x30, 0x00,
>-  0x5c, 0x2e, 0x17, 0x0b, 0x80, 0x00,
>-  0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
>-  0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
>-  0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
>-  0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
>-  0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
>-  0x89, 0x44, 0xa2, 0x51, 0x20, 0x00
>-};
>-
>-const uint8_t kMaskRandom36_12[72] = {
>-  0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
>-  0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
>-  0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
>-  0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
>-  0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
>-  0x89, 0x44, 0xa2, 0x51, 0x20, 0x00,
>-  0x8c, 0xc6, 0x63, 0x31, 0x90, 0x00,
>-  0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
>-  0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
>-  0x61, 0xb0, 0xd8, 0x6c, 0x30, 0x00,
>-  0x5c, 0x2e, 0x17, 0x0b, 0x80, 0x00,
>-  0x5b, 0x0c, 0x56, 0xc3, 0x10, 0x00
>-};
>-
>-const uint8_t kMaskRandom36_13[78] = {
>-  0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
>-  0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
>-  0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
>-  0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
>-  0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
>-  0x89, 0x44, 0xa2, 0x51, 0x20, 0x00,
>-  0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
>-  0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
>-  0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
>-  0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
>-  0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
>-  0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
>-  0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00
>-};
>-
>-const uint8_t kMaskRandom36_14[84] = {
>-  0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
>-  0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
>-  0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
>-  0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
>-  0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
>-  0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
>-  0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
>-  0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
>-  0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
>-  0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
>-  0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
>-  0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
>-  0x89, 0x44, 0xa2, 0x51, 0x20, 0x00,
>-  0x7f, 0x4f, 0xdf, 0xd3, 0xf0, 0x00
>-};
>-
>-const uint8_t kMaskRandom36_15[90] = {
>-  0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
>-  0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
>-  0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
>-  0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
>-  0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
>-  0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
>-  0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
>-  0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
>-  0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
>-  0xc0, 0xe0, 0x70, 0x38, 0x10, 0x00,
>-  0x06, 0x83, 0x41, 0xa0, 0xd0, 0x00,
>-  0xc8, 0x64, 0x32, 0x19, 0x00, 0x00,
>-  0x45, 0x22, 0x91, 0x48, 0xa0, 0x00,
>-  0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
>-  0xa2, 0x51, 0x28, 0x94, 0x40, 0x00
>-};
>-
>-const uint8_t kMaskRandom36_16[96] = {
>-  0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
>-  0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
>-  0xc0, 0xe0, 0x70, 0x38, 0x10, 0x00,
>-  0x06, 0x83, 0x41, 0xa0, 0xd0, 0x00,
>-  0xc8, 0x64, 0x32, 0x19, 0x00, 0x00,
>-  0x45, 0x22, 0x91, 0x48, 0xa0, 0x00,
>-  0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
>-  0xa2, 0x51, 0x28, 0x94, 0x40, 0x00,
>-  0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
>-  0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
>-  0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
>-  0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
>-  0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
>-  0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
>-  0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
>-  0xef, 0xf2, 0x3b, 0xfc, 0x80, 0x00
>-};
>-
>-const uint8_t kMaskRandom36_17[102] = {
>-  0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
>-  0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
>-  0xc0, 0xe0, 0x70, 0x38, 0x10, 0x00,
>-  0x06, 0x83, 0x41, 0xa0, 0xd0, 0x00,
>-  0xc8, 0x64, 0x32, 0x19, 0x00, 0x00,
>-  0x45, 0x22, 0x91, 0x48, 0xa0, 0x00,
>-  0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
>-  0xa2, 0x51, 0x28, 0x94, 0x40, 0x00,
>-  0x4c, 0x26, 0x13, 0x09, 0x80, 0x00,
>-  0x66, 0x33, 0x19, 0x8c, 0xc0, 0x00,
>-  0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
>-  0x42, 0xa1, 0x50, 0xa8, 0x50, 0x00,
>-  0xa4, 0x52, 0x29, 0x14, 0x80, 0x00,
>-  0x13, 0x09, 0x84, 0xc2, 0x60, 0x00,
>-  0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
>-  0x88, 0xc4, 0x62, 0x31, 0x10, 0x00,
>-  0x09, 0x04, 0x82, 0x41, 0x20, 0x00
>-};
>-
>-const uint8_t kMaskRandom36_18[108] = {
>-  0x4c, 0x26, 0x13, 0x09, 0x80, 0x00,
>-  0x66, 0x33, 0x19, 0x8c, 0xc0, 0x00,
>-  0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
>-  0x42, 0xa1, 0x50, 0xa8, 0x50, 0x00,
>-  0xa4, 0x52, 0x29, 0x14, 0x80, 0x00,
>-  0x13, 0x09, 0x84, 0xc2, 0x60, 0x00,
>-  0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
>-  0x88, 0xc4, 0x62, 0x31, 0x10, 0x00,
>-  0x09, 0x04, 0x82, 0x41, 0x20, 0x00,
>-  0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
>-  0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
>-  0xc0, 0xe0, 0x70, 0x38, 0x10, 0x00,
>-  0x06, 0x83, 0x41, 0xa0, 0xd0, 0x00,
>-  0xc8, 0x64, 0x32, 0x19, 0x00, 0x00,
>-  0x45, 0x22, 0x91, 0x48, 0xa0, 0x00,
>-  0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
>-  0xa2, 0x51, 0x28, 0x94, 0x40, 0x00,
>-  0xd0, 0x03, 0x74, 0x00, 0xd0, 0x00
>-};
>-
>-const uint8_t kMaskRandom36_19[114] = {
>-  0x44, 0xa2, 0x51, 0x28, 0x90, 0x00,
>-  0x66, 0x26, 0x19, 0x89, 0x80, 0x00,
>-  0x90, 0x49, 0x64, 0x12, 0x50, 0x00,
>-  0x01, 0xa5, 0x80, 0x69, 0x60, 0x00,
>-  0x0e, 0x12, 0x83, 0x84, 0xa0, 0x00,
>-  0x13, 0x0b, 0x04, 0xc2, 0xc0, 0x00,
>-  0x20, 0xd0, 0x48, 0x34, 0x10, 0x00,
>-  0xc2, 0x51, 0x30, 0x94, 0x40, 0x00,
>-  0x29, 0x0c, 0x8a, 0x43, 0x20, 0x00,
>-  0x8c, 0xc6, 0x63, 0x31, 0x90, 0x00,
>-  0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
>-  0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
>-  0x61, 0xb0, 0xd8, 0x6c, 0x30, 0x00,
>-  0x5c, 0x2e, 0x17, 0x0b, 0x80, 0x00,
>-  0xcc, 0x66, 0x33, 0x19, 0x80, 0x00,
>-  0x2b, 0x15, 0x8a, 0xc5, 0x60, 0x00,
>-  0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
>-  0x25, 0x92, 0xc9, 0x64, 0xb0, 0x00,
>-  0xfd, 0x9d, 0xff, 0x67, 0x70, 0x00
>-};
>-
>-const uint8_t kMaskRandom36_2[12] = {
>-  0xce, 0x67, 0x33, 0x99, 0xc0, 0x00,
>-  0x39, 0x9c, 0xce, 0x67, 0x30, 0x00
>-};
>-
>-const uint8_t kMaskRandom36_20[120] = {
>-  0x8c, 0xc6, 0x63, 0x31, 0x90, 0x00,
>-  0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
>-  0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
>-  0x61, 0xb0, 0xd8, 0x6c, 0x30, 0x00,
>-  0x5c, 0x2e, 0x17, 0x0b, 0x80, 0x00,
>-  0xcc, 0x66, 0x33, 0x19, 0x80, 0x00,
>-  0x2b, 0x15, 0x8a, 0xc5, 0x60, 0x00,
>-  0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
>-  0x25, 0x92, 0xc9, 0x64, 0xb0, 0x00,
>-  0xfd, 0x9d, 0xff, 0x67, 0x70, 0x00,
>-  0x44, 0xa2, 0x51, 0x28, 0x90, 0x00,
>-  0x66, 0x26, 0x19, 0x89, 0x80, 0x00,
>-  0x90, 0x49, 0x64, 0x12, 0x50, 0x00,
>-  0x01, 0xa5, 0x80, 0x69, 0x60, 0x00,
>-  0x0e, 0x12, 0x83, 0x84, 0xa0, 0x00,
>-  0x13, 0x0b, 0x04, 0xc2, 0xc0, 0x00,
>-  0x20, 0xd0, 0x48, 0x34, 0x10, 0x00,
>-  0xc2, 0x51, 0x30, 0x94, 0x40, 0x00,
>-  0x29, 0x0c, 0x8a, 0x43, 0x20, 0x00,
>-  0x45, 0xb9, 0x08, 0x16, 0x30, 0x00
>-};
>-
>-const uint8_t kMaskRandom36_21[126] = {
>-  0x8c, 0xc6, 0x63, 0x31, 0x90, 0x00,
>-  0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
>-  0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
>-  0x61, 0xb0, 0xd8, 0x6c, 0x30, 0x00,
>-  0x5c, 0x2e, 0x17, 0x0b, 0x80, 0x00,
>-  0xcc, 0x66, 0x33, 0x19, 0x80, 0x00,
>-  0x2b, 0x15, 0x8a, 0xc5, 0x60, 0x00,
>-  0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
>-  0x25, 0x92, 0xc9, 0x64, 0xb0, 0x00,
>-  0xfd, 0x9d, 0xff, 0x67, 0x70, 0x00,
>-  0x8c, 0xc6, 0x63, 0x31, 0x90, 0x00,
>-  0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
>-  0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
>-  0x61, 0xb0, 0xd8, 0x6c, 0x30, 0x00,
>-  0x5c, 0x2e, 0x17, 0x0b, 0x80, 0x00,
>-  0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
>-  0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
>-  0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
>-  0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
>-  0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
>-  0x89, 0x44, 0xa2, 0x51, 0x20, 0x00
>-};
>-
>-const uint8_t kMaskRandom36_22[132] = {
>-  0x8c, 0xc6, 0x63, 0x31, 0x90, 0x00,
>-  0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
>-  0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
>-  0x61, 0xb0, 0xd8, 0x6c, 0x30, 0x00,
>-  0x5c, 0x2e, 0x17, 0x0b, 0x80, 0x00,
>-  0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
>-  0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
>-  0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
>-  0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
>-  0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
>-  0x89, 0x44, 0xa2, 0x51, 0x20, 0x00,
>-  0x8c, 0xc6, 0x63, 0x31, 0x90, 0x00,
>-  0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
>-  0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
>-  0x61, 0xb0, 0xd8, 0x6c, 0x30, 0x00,
>-  0x5c, 0x2e, 0x17, 0x0b, 0x80, 0x00,
>-  0xcc, 0x66, 0x33, 0x19, 0x80, 0x00,
>-  0x2b, 0x15, 0x8a, 0xc5, 0x60, 0x00,
>-  0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
>-  0x25, 0x92, 0xc9, 0x64, 0xb0, 0x00,
>-  0xfd, 0x9d, 0xff, 0x67, 0x70, 0x00,
>-  0x71, 0x04, 0xba, 0x7b, 0xe0, 0x00
>-};
>-
>-const uint8_t kMaskRandom36_23[138] = {
>-  0x8c, 0xc6, 0x63, 0x31, 0x90, 0x00,
>-  0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
>-  0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
>-  0x61, 0xb0, 0xd8, 0x6c, 0x30, 0x00,
>-  0x5c, 0x2e, 0x17, 0x0b, 0x80, 0x00,
>-  0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
>-  0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
>-  0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
>-  0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
>-  0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
>-  0x89, 0x44, 0xa2, 0x51, 0x20, 0x00,
>-  0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
>-  0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
>-  0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
>-  0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
>-  0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
>-  0x89, 0x44, 0xa2, 0x51, 0x20, 0x00,
>-  0x8c, 0xc6, 0x63, 0x31, 0x90, 0x00,
>-  0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
>-  0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
>-  0x61, 0xb0, 0xd8, 0x6c, 0x30, 0x00,
>-  0x5c, 0x2e, 0x17, 0x0b, 0x80, 0x00,
>-  0x5b, 0x0c, 0x56, 0xc3, 0x10, 0x00
>-};
>-
>-const uint8_t kMaskRandom36_24[144] = {
>-  0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
>-  0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
>-  0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
>-  0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
>-  0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
>-  0x89, 0x44, 0xa2, 0x51, 0x20, 0x00,
>-  0x8c, 0xc6, 0x63, 0x31, 0x90, 0x00,
>-  0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
>-  0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
>-  0x61, 0xb0, 0xd8, 0x6c, 0x30, 0x00,
>-  0x5c, 0x2e, 0x17, 0x0b, 0x80, 0x00,
>-  0x5b, 0x0c, 0x56, 0xc3, 0x10, 0x00,
>-  0x8c, 0xc6, 0x63, 0x31, 0x90, 0x00,
>-  0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
>-  0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
>-  0x61, 0xb0, 0xd8, 0x6c, 0x30, 0x00,
>-  0x5c, 0x2e, 0x17, 0x0b, 0x80, 0x00,
>-  0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
>-  0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
>-  0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
>-  0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
>-  0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
>-  0x89, 0x44, 0xa2, 0x51, 0x20, 0x00,
>-  0x76, 0x3a, 0xeb, 0x17, 0xc0, 0x00
>-};
>-
>-const uint8_t kMaskRandom36_25[150] = {
>-  0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
>-  0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
>-  0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
>-  0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
>-  0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
>-  0x89, 0x44, 0xa2, 0x51, 0x20, 0x00,
>-  0x8c, 0xc6, 0x63, 0x31, 0x90, 0x00,
>-  0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
>-  0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
>-  0x61, 0xb0, 0xd8, 0x6c, 0x30, 0x00,
>-  0x5c, 0x2e, 0x17, 0x0b, 0x80, 0x00,
>-  0x5b, 0x0c, 0x56, 0xc3, 0x10, 0x00,
>-  0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
>-  0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
>-  0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
>-  0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
>-  0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
>-  0x89, 0x44, 0xa2, 0x51, 0x20, 0x00,
>-  0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
>-  0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
>-  0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
>-  0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
>-  0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
>-  0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
>-  0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00
>-};
>-
>-const uint8_t kMaskRandom36_26[156] = {
>-  0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
>-  0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
>-  0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
>-  0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
>-  0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
>-  0x89, 0x44, 0xa2, 0x51, 0x20, 0x00,
>-  0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
>-  0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
>-  0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
>-  0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
>-  0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
>-  0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
>-  0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
>-  0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
>-  0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
>-  0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
>-  0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
>-  0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
>-  0x89, 0x44, 0xa2, 0x51, 0x20, 0x00,
>-  0x8c, 0xc6, 0x63, 0x31, 0x90, 0x00,
>-  0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
>-  0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
>-  0x61, 0xb0, 0xd8, 0x6c, 0x30, 0x00,
>-  0x5c, 0x2e, 0x17, 0x0b, 0x80, 0x00,
>-  0x5b, 0x0c, 0x56, 0xc3, 0x10, 0x00,
>-  0xec, 0x58, 0x0e, 0x6c, 0xe0, 0x00
>-};
>-
>-const uint8_t kMaskRandom36_27[162] = {
>-  0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
>-  0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
>-  0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
>-  0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
>-  0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
>-  0x89, 0x44, 0xa2, 0x51, 0x20, 0x00,
>-  0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
>-  0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
>-  0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
>-  0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
>-  0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
>-  0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
>-  0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
>-  0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
>-  0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
>-  0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
>-  0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
>-  0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
>-  0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
>-  0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
>-  0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
>-  0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
>-  0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
>-  0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
>-  0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
>-  0x89, 0x44, 0xa2, 0x51, 0x20, 0x00,
>-  0x7f, 0x4f, 0xdf, 0xd3, 0xf0, 0x00
>-};
>-
>-const uint8_t kMaskRandom36_28[168] = {
>-  0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
>-  0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
>-  0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
>-  0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
>-  0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
>-  0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
>-  0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
>-  0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
>-  0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
>-  0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
>-  0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
>-  0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
>-  0x89, 0x44, 0xa2, 0x51, 0x20, 0x00,
>-  0x7f, 0x4f, 0xdf, 0xd3, 0xf0, 0x00,
>-  0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
>-  0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
>-  0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
>-  0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
>-  0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
>-  0x89, 0x44, 0xa2, 0x51, 0x20, 0x00,
>-  0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
>-  0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
>-  0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
>-  0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
>-  0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
>-  0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
>-  0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
>-  0x29, 0xfd, 0x91, 0x6f, 0xd0, 0x00
>-};
>-
>-const uint8_t kMaskRandom36_29[174] = {
>-  0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
>-  0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
>-  0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
>-  0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
>-  0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
>-  0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
>-  0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
>-  0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
>-  0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
>-  0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
>-  0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
>-  0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
>-  0x89, 0x44, 0xa2, 0x51, 0x20, 0x00,
>-  0x7f, 0x4f, 0xdf, 0xd3, 0xf0, 0x00,
>-  0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
>-  0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
>-  0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
>-  0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
>-  0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
>-  0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
>-  0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
>-  0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
>-  0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
>-  0xc0, 0xe0, 0x70, 0x38, 0x10, 0x00,
>-  0x06, 0x83, 0x41, 0xa0, 0xd0, 0x00,
>-  0xc8, 0x64, 0x32, 0x19, 0x00, 0x00,
>-  0x45, 0x22, 0x91, 0x48, 0xa0, 0x00,
>-  0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
>-  0xa2, 0x51, 0x28, 0x94, 0x40, 0x00
>-};
>-
>-const uint8_t kMaskRandom36_3[18] = {
>-  0xcc, 0x66, 0x33, 0x19, 0x80, 0x00,
>-  0x27, 0x15, 0x89, 0xc5, 0x60, 0x00,
>-  0x92, 0xc9, 0x64, 0xb2, 0x50, 0x00
>-};
>-
>-const uint8_t kMaskRandom36_30[180] = {
>-  0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
>-  0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
>-  0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
>-  0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
>-  0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
>-  0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
>-  0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
>-  0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
>-  0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
>-  0xc0, 0xe0, 0x70, 0x38, 0x10, 0x00,
>-  0x06, 0x83, 0x41, 0xa0, 0xd0, 0x00,
>-  0xc8, 0x64, 0x32, 0x19, 0x00, 0x00,
>-  0x45, 0x22, 0x91, 0x48, 0xa0, 0x00,
>-  0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
>-  0xa2, 0x51, 0x28, 0x94, 0x40, 0x00,
>-  0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
>-  0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
>-  0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
>-  0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
>-  0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
>-  0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
>-  0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
>-  0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
>-  0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
>-  0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
>-  0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
>-  0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
>-  0x89, 0x44, 0xa2, 0x51, 0x20, 0x00,
>-  0x7f, 0x4f, 0xdf, 0xd3, 0xf0, 0x00,
>-  0xc5, 0x38, 0xbb, 0x98, 0x80, 0x00
>-};
>-
>-const uint8_t kMaskRandom36_31[186] = {
>-  0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
>-  0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
>-  0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
>-  0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
>-  0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
>-  0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
>-  0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
>-  0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
>-  0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
>-  0xc0, 0xe0, 0x70, 0x38, 0x10, 0x00,
>-  0x06, 0x83, 0x41, 0xa0, 0xd0, 0x00,
>-  0xc8, 0x64, 0x32, 0x19, 0x00, 0x00,
>-  0x45, 0x22, 0x91, 0x48, 0xa0, 0x00,
>-  0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
>-  0xa2, 0x51, 0x28, 0x94, 0x40, 0x00,
>-  0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
>-  0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
>-  0xc0, 0xe0, 0x70, 0x38, 0x10, 0x00,
>-  0x06, 0x83, 0x41, 0xa0, 0xd0, 0x00,
>-  0xc8, 0x64, 0x32, 0x19, 0x00, 0x00,
>-  0x45, 0x22, 0x91, 0x48, 0xa0, 0x00,
>-  0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
>-  0xa2, 0x51, 0x28, 0x94, 0x40, 0x00,
>-  0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
>-  0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
>-  0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
>-  0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
>-  0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
>-  0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
>-  0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
>-  0xef, 0xf2, 0x3b, 0xfc, 0x80, 0x00
>-};
>-
>-const uint8_t kMaskRandom36_32[192] = {
>-  0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
>-  0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
>-  0xc0, 0xe0, 0x70, 0x38, 0x10, 0x00,
>-  0x06, 0x83, 0x41, 0xa0, 0xd0, 0x00,
>-  0xc8, 0x64, 0x32, 0x19, 0x00, 0x00,
>-  0x45, 0x22, 0x91, 0x48, 0xa0, 0x00,
>-  0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
>-  0xa2, 0x51, 0x28, 0x94, 0x40, 0x00,
>-  0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
>-  0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
>-  0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
>-  0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
>-  0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
>-  0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
>-  0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
>-  0xef, 0xf2, 0x3b, 0xfc, 0x80, 0x00,
>-  0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
>-  0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
>-  0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
>-  0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
>-  0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
>-  0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
>-  0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
>-  0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
>-  0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
>-  0xc0, 0xe0, 0x70, 0x38, 0x10, 0x00,
>-  0x06, 0x83, 0x41, 0xa0, 0xd0, 0x00,
>-  0xc8, 0x64, 0x32, 0x19, 0x00, 0x00,
>-  0x45, 0x22, 0x91, 0x48, 0xa0, 0x00,
>-  0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
>-  0xa2, 0x51, 0x28, 0x94, 0x40, 0x00,
>-  0x3a, 0x28, 0x9c, 0x2f, 0xc0, 0x00
>-};
>-
>-const uint8_t kMaskRandom36_33[198] = {
>-  0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
>-  0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
>-  0xc0, 0xe0, 0x70, 0x38, 0x10, 0x00,
>-  0x06, 0x83, 0x41, 0xa0, 0xd0, 0x00,
>-  0xc8, 0x64, 0x32, 0x19, 0x00, 0x00,
>-  0x45, 0x22, 0x91, 0x48, 0xa0, 0x00,
>-  0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
>-  0xa2, 0x51, 0x28, 0x94, 0x40, 0x00,
>-  0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
>-  0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
>-  0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
>-  0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
>-  0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
>-  0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
>-  0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
>-  0xef, 0xf2, 0x3b, 0xfc, 0x80, 0x00,
>-  0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
>-  0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
>-  0xc0, 0xe0, 0x70, 0x38, 0x10, 0x00,
>-  0x06, 0x83, 0x41, 0xa0, 0xd0, 0x00,
>-  0xc8, 0x64, 0x32, 0x19, 0x00, 0x00,
>-  0x45, 0x22, 0x91, 0x48, 0xa0, 0x00,
>-  0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
>-  0xa2, 0x51, 0x28, 0x94, 0x40, 0x00,
>-  0x4c, 0x26, 0x13, 0x09, 0x80, 0x00,
>-  0x66, 0x33, 0x19, 0x8c, 0xc0, 0x00,
>-  0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
>-  0x42, 0xa1, 0x50, 0xa8, 0x50, 0x00,
>-  0xa4, 0x52, 0x29, 0x14, 0x80, 0x00,
>-  0x13, 0x09, 0x84, 0xc2, 0x60, 0x00,
>-  0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
>-  0x88, 0xc4, 0x62, 0x31, 0x10, 0x00,
>-  0x09, 0x04, 0x82, 0x41, 0x20, 0x00
>-};
>-
>-const uint8_t kMaskRandom36_34[204] = {
>-  0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
>-  0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
>-  0xc0, 0xe0, 0x70, 0x38, 0x10, 0x00,
>-  0x06, 0x83, 0x41, 0xa0, 0xd0, 0x00,
>-  0xc8, 0x64, 0x32, 0x19, 0x00, 0x00,
>-  0x45, 0x22, 0x91, 0x48, 0xa0, 0x00,
>-  0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
>-  0xa2, 0x51, 0x28, 0x94, 0x40, 0x00,
>-  0x4c, 0x26, 0x13, 0x09, 0x80, 0x00,
>-  0x66, 0x33, 0x19, 0x8c, 0xc0, 0x00,
>-  0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
>-  0x42, 0xa1, 0x50, 0xa8, 0x50, 0x00,
>-  0xa4, 0x52, 0x29, 0x14, 0x80, 0x00,
>-  0x13, 0x09, 0x84, 0xc2, 0x60, 0x00,
>-  0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
>-  0x88, 0xc4, 0x62, 0x31, 0x10, 0x00,
>-  0x09, 0x04, 0x82, 0x41, 0x20, 0x00,
>-  0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
>-  0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
>-  0xc0, 0xe0, 0x70, 0x38, 0x10, 0x00,
>-  0x06, 0x83, 0x41, 0xa0, 0xd0, 0x00,
>-  0xc8, 0x64, 0x32, 0x19, 0x00, 0x00,
>-  0x45, 0x22, 0x91, 0x48, 0xa0, 0x00,
>-  0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
>-  0xa2, 0x51, 0x28, 0x94, 0x40, 0x00,
>-  0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
>-  0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
>-  0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
>-  0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
>-  0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
>-  0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
>-  0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
>-  0xef, 0xf2, 0x3b, 0xfc, 0x80, 0x00,
>-  0xf7, 0x5e, 0x66, 0x5b, 0x60, 0x00
>-};
>-
>-const uint8_t kMaskRandom36_35[210] = {
>-  0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
>-  0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
>-  0xc0, 0xe0, 0x70, 0x38, 0x10, 0x00,
>-  0x06, 0x83, 0x41, 0xa0, 0xd0, 0x00,
>-  0xc8, 0x64, 0x32, 0x19, 0x00, 0x00,
>-  0x45, 0x22, 0x91, 0x48, 0xa0, 0x00,
>-  0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
>-  0xa2, 0x51, 0x28, 0x94, 0x40, 0x00,
>-  0x4c, 0x26, 0x13, 0x09, 0x80, 0x00,
>-  0x66, 0x33, 0x19, 0x8c, 0xc0, 0x00,
>-  0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
>-  0x42, 0xa1, 0x50, 0xa8, 0x50, 0x00,
>-  0xa4, 0x52, 0x29, 0x14, 0x80, 0x00,
>-  0x13, 0x09, 0x84, 0xc2, 0x60, 0x00,
>-  0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
>-  0x88, 0xc4, 0x62, 0x31, 0x10, 0x00,
>-  0x09, 0x04, 0x82, 0x41, 0x20, 0x00,
>-  0x4c, 0x26, 0x13, 0x09, 0x80, 0x00,
>-  0x66, 0x33, 0x19, 0x8c, 0xc0, 0x00,
>-  0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
>-  0x42, 0xa1, 0x50, 0xa8, 0x50, 0x00,
>-  0xa4, 0x52, 0x29, 0x14, 0x80, 0x00,
>-  0x13, 0x09, 0x84, 0xc2, 0x60, 0x00,
>-  0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
>-  0x88, 0xc4, 0x62, 0x31, 0x10, 0x00,
>-  0x09, 0x04, 0x82, 0x41, 0x20, 0x00,
>-  0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
>-  0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
>-  0xc0, 0xe0, 0x70, 0x38, 0x10, 0x00,
>-  0x06, 0x83, 0x41, 0xa0, 0xd0, 0x00,
>-  0xc8, 0x64, 0x32, 0x19, 0x00, 0x00,
>-  0x45, 0x22, 0x91, 0x48, 0xa0, 0x00,
>-  0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
>-  0xa2, 0x51, 0x28, 0x94, 0x40, 0x00,
>-  0xd0, 0x03, 0x74, 0x00, 0xd0, 0x00
>-};
>-
>-const uint8_t kMaskRandom36_36[216] = {
>-  0x4c, 0x26, 0x13, 0x09, 0x80, 0x00,
>-  0x66, 0x33, 0x19, 0x8c, 0xc0, 0x00,
>-  0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
>-  0x42, 0xa1, 0x50, 0xa8, 0x50, 0x00,
>-  0xa4, 0x52, 0x29, 0x14, 0x80, 0x00,
>-  0x13, 0x09, 0x84, 0xc2, 0x60, 0x00,
>-  0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
>-  0x88, 0xc4, 0x62, 0x31, 0x10, 0x00,
>-  0x09, 0x04, 0x82, 0x41, 0x20, 0x00,
>-  0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
>-  0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
>-  0xc0, 0xe0, 0x70, 0x38, 0x10, 0x00,
>-  0x06, 0x83, 0x41, 0xa0, 0xd0, 0x00,
>-  0xc8, 0x64, 0x32, 0x19, 0x00, 0x00,
>-  0x45, 0x22, 0x91, 0x48, 0xa0, 0x00,
>-  0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
>-  0xa2, 0x51, 0x28, 0x94, 0x40, 0x00,
>-  0xd0, 0x03, 0x74, 0x00, 0xd0, 0x00,
>-  0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
>-  0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
>-  0xc0, 0xe0, 0x70, 0x38, 0x10, 0x00,
>-  0x06, 0x83, 0x41, 0xa0, 0xd0, 0x00,
>-  0xc8, 0x64, 0x32, 0x19, 0x00, 0x00,
>-  0x45, 0x22, 0x91, 0x48, 0xa0, 0x00,
>-  0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
>-  0xa2, 0x51, 0x28, 0x94, 0x40, 0x00,
>-  0x4c, 0x26, 0x13, 0x09, 0x80, 0x00,
>-  0x66, 0x33, 0x19, 0x8c, 0xc0, 0x00,
>-  0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
>-  0x42, 0xa1, 0x50, 0xa8, 0x50, 0x00,
>-  0xa4, 0x52, 0x29, 0x14, 0x80, 0x00,
>-  0x13, 0x09, 0x84, 0xc2, 0x60, 0x00,
>-  0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
>-  0x88, 0xc4, 0x62, 0x31, 0x10, 0x00,
>-  0x09, 0x04, 0x82, 0x41, 0x20, 0x00,
>-  0xa4, 0x9c, 0x31, 0x13, 0x80, 0x00
>-};
>-
>-const uint8_t kMaskRandom36_4[24] = {
>-  0xec, 0x76, 0x3b, 0x1d, 0x80, 0x00,
>-  0x67, 0x33, 0x99, 0xcc, 0xe0, 0x00,
>-  0xb1, 0xd8, 0xec, 0x76, 0x30, 0x00,
>-  0x5a, 0xad, 0x56, 0xab, 0x50, 0x00
>-};
>-
>-const uint8_t kMaskRandom36_5[30] = {
>-  0x4c, 0xa6, 0x53, 0x29, 0x90, 0x00,
>-  0x66, 0x33, 0x19, 0x8c, 0xc0, 0x00,
>-  0x19, 0xd0, 0xc6, 0x74, 0x30, 0x00,
>-  0x9c, 0x89, 0x67, 0x22, 0x50, 0x00,
>-  0xe3, 0x4c, 0x38, 0xd3, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom36_6[36] = {
>-  0xcc, 0x26, 0x33, 0x09, 0x80, 0x00,
>-  0x45, 0x62, 0x91, 0x58, 0xa0, 0x00,
>-  0xb0, 0x98, 0x6c, 0x26, 0x10, 0x00,
>-  0x8a, 0x85, 0x62, 0xa1, 0x50, 0x00,
>-  0x29, 0x53, 0x0a, 0x54, 0xc0, 0x00,
>-  0xa6, 0x0a, 0xa9, 0x82, 0xa0, 0x00
>-};
>-
>-const uint8_t kMaskRandom36_7[42] = {
>-  0x44, 0xa2, 0x51, 0x28, 0x90, 0x00,
>-  0x23, 0x11, 0x88, 0xc4, 0x60, 0x00,
>-  0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
>-  0x0a, 0x85, 0x42, 0xa1, 0x50, 0x00,
>-  0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
>-  0x0b, 0x06, 0x82, 0xc1, 0xa0, 0x00,
>-  0xe0, 0x64, 0x38, 0x19, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom36_8[48] = {
>-  0x16, 0x0b, 0x05, 0x82, 0xc0, 0x00,
>-  0xc2, 0x61, 0x30, 0x98, 0x40, 0x00,
>-  0x60, 0xb0, 0x58, 0x2c, 0x10, 0x00,
>-  0x85, 0x42, 0xa1, 0x50, 0xa0, 0x00,
>-  0x4c, 0x26, 0x13, 0x09, 0x80, 0x00,
>-  0x29, 0x14, 0x8a, 0x45, 0x20, 0x00,
>-  0x11, 0x88, 0xc4, 0x62, 0x30, 0x00,
>-  0xb0, 0x58, 0x2c, 0x16, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom36_9[54] = {
>-  0x44, 0xa2, 0x51, 0x28, 0x90, 0x00,
>-  0x66, 0x26, 0x19, 0x89, 0x80, 0x00,
>-  0x90, 0x49, 0x64, 0x12, 0x50, 0x00,
>-  0x01, 0xa5, 0x80, 0x69, 0x60, 0x00,
>-  0x0e, 0x12, 0x83, 0x84, 0xa0, 0x00,
>-  0x13, 0x0b, 0x04, 0xc2, 0xc0, 0x00,
>-  0x20, 0xd0, 0x48, 0x34, 0x10, 0x00,
>-  0xc2, 0x51, 0x30, 0x94, 0x40, 0x00,
>-  0x29, 0x0c, 0x8a, 0x43, 0x20, 0x00
>-};
>-
>-const uint8_t kMaskRandom37_1[6] = {
>-  0xff, 0xff, 0xff, 0xff, 0xf8, 0x00
>-};
>-
>-const uint8_t kMaskRandom37_10[60] = {
>-  0x8c, 0xc6, 0x63, 0x38, 0xc0, 0x00,
>-  0x27, 0x13, 0x89, 0xc4, 0x70, 0x00,
>-  0x32, 0x99, 0x4c, 0xa3, 0x48, 0x00,
>-  0x61, 0xb0, 0xd8, 0x64, 0x98, 0x00,
>-  0x5c, 0x2e, 0x17, 0x0e, 0x20, 0x00,
>-  0xcc, 0x66, 0x33, 0x1d, 0x40, 0x00,
>-  0x2b, 0x15, 0x8a, 0xc6, 0x70, 0x00,
>-  0x32, 0x99, 0x4c, 0xb4, 0x98, 0x00,
>-  0x25, 0x92, 0xc9, 0x63, 0xa8, 0x00,
>-  0xfd, 0x9d, 0xd4, 0x22, 0x30, 0x00
>-};
>-
>-const uint8_t kMaskRandom37_11[66] = {
>-  0x8c, 0xc6, 0x63, 0x38, 0xc0, 0x00,
>-  0x27, 0x13, 0x89, 0xc4, 0x70, 0x00,
>-  0x32, 0x99, 0x4c, 0xa3, 0x48, 0x00,
>-  0x61, 0xb0, 0xd8, 0x64, 0x98, 0x00,
>-  0x5c, 0x2e, 0x17, 0x0e, 0x20, 0x00,
>-  0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
>-  0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
>-  0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
>-  0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
>-  0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
>-  0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom37_12[72] = {
>-  0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
>-  0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
>-  0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
>-  0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
>-  0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
>-  0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00,
>-  0x8c, 0xc6, 0x63, 0x38, 0xc0, 0x00,
>-  0x27, 0x13, 0x89, 0xc4, 0x70, 0x00,
>-  0x32, 0x99, 0x4c, 0xa3, 0x48, 0x00,
>-  0x61, 0xb0, 0xd8, 0x64, 0x98, 0x00,
>-  0x5c, 0x2e, 0x17, 0x0e, 0x20, 0x00,
>-  0x5b, 0x0c, 0x64, 0x32, 0x20, 0x00
>-};
>-
>-const uint8_t kMaskRandom37_13[78] = {
>-  0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
>-  0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
>-  0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
>-  0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
>-  0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
>-  0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00,
>-  0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
>-  0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
>-  0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
>-  0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
>-  0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
>-  0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
>-  0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00
>-};
>-
>-const uint8_t kMaskRandom37_14[84] = {
>-  0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
>-  0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
>-  0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
>-  0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
>-  0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
>-  0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
>-  0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00,
>-  0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
>-  0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
>-  0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
>-  0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
>-  0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
>-  0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00,
>-  0x7f, 0x4f, 0xdb, 0x89, 0xd8, 0x00
>-};
>-
>-const uint8_t kMaskRandom37_15[90] = {
>-  0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
>-  0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
>-  0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
>-  0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
>-  0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
>-  0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
>-  0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00,
>-  0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
>-  0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
>-  0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
>-  0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
>-  0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
>-  0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
>-  0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
>-  0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00
>-};
>-
>-const uint8_t kMaskRandom37_16[96] = {
>-  0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
>-  0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
>-  0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
>-  0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
>-  0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
>-  0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
>-  0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
>-  0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00,
>-  0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
>-  0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
>-  0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
>-  0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
>-  0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
>-  0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
>-  0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00,
>-  0xef, 0xf2, 0x1f, 0x9d, 0x78, 0x00
>-};
>-
>-const uint8_t kMaskRandom37_17[102] = {
>-  0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
>-  0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
>-  0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
>-  0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
>-  0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
>-  0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
>-  0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
>-  0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00,
>-  0x4c, 0x26, 0x13, 0x09, 0xc0, 0x00,
>-  0x66, 0x33, 0x19, 0x9c, 0x60, 0x00,
>-  0x91, 0x48, 0xa4, 0x50, 0x38, 0x00,
>-  0x42, 0xa1, 0x50, 0xa4, 0x28, 0x00,
>-  0xa4, 0x52, 0x29, 0x0a, 0x50, 0x00,
>-  0x13, 0x09, 0x84, 0xd6, 0x80, 0x00,
>-  0x30, 0x98, 0x4c, 0x24, 0xd0, 0x00,
>-  0x88, 0xc4, 0x62, 0x2b, 0x08, 0x00,
>-  0x09, 0x04, 0x82, 0x43, 0x30, 0x00
>-};
>-
>-const uint8_t kMaskRandom37_18[108] = {
>-  0x4c, 0x26, 0x13, 0x09, 0xc0, 0x00,
>-  0x66, 0x33, 0x19, 0x9c, 0x60, 0x00,
>-  0x91, 0x48, 0xa4, 0x50, 0x38, 0x00,
>-  0x42, 0xa1, 0x50, 0xa4, 0x28, 0x00,
>-  0xa4, 0x52, 0x29, 0x0a, 0x50, 0x00,
>-  0x13, 0x09, 0x84, 0xd6, 0x80, 0x00,
>-  0x30, 0x98, 0x4c, 0x24, 0xd0, 0x00,
>-  0x88, 0xc4, 0x62, 0x2b, 0x08, 0x00,
>-  0x09, 0x04, 0x82, 0x43, 0x30, 0x00,
>-  0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
>-  0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
>-  0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
>-  0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
>-  0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
>-  0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
>-  0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
>-  0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00,
>-  0xd0, 0x03, 0x54, 0x65, 0xc8, 0x00
>-};
>-
>-const uint8_t kMaskRandom37_19[114] = {
>-  0x44, 0xa2, 0x51, 0x29, 0xc0, 0x00,
>-  0x66, 0x26, 0x19, 0x9c, 0x20, 0x00,
>-  0x90, 0x49, 0x44, 0xb0, 0x38, 0x00,
>-  0x01, 0xa5, 0xb0, 0xc4, 0x28, 0x00,
>-  0x0e, 0x12, 0xa3, 0x0a, 0x50, 0x00,
>-  0x13, 0x0b, 0x04, 0x56, 0xc0, 0x00,
>-  0x20, 0xd0, 0x48, 0x64, 0xd0, 0x00,
>-  0xc2, 0x51, 0x28, 0x8b, 0x00, 0x00,
>-  0x29, 0x0c, 0x86, 0x03, 0x38, 0x00,
>-  0x8c, 0xc6, 0x63, 0x38, 0xc0, 0x00,
>-  0x27, 0x13, 0x89, 0xc4, 0x70, 0x00,
>-  0x32, 0x99, 0x4c, 0xa3, 0x48, 0x00,
>-  0x61, 0xb0, 0xd8, 0x64, 0x98, 0x00,
>-  0x5c, 0x2e, 0x17, 0x0e, 0x20, 0x00,
>-  0xcc, 0x66, 0x33, 0x1d, 0x40, 0x00,
>-  0x2b, 0x15, 0x8a, 0xc6, 0x70, 0x00,
>-  0x32, 0x99, 0x4c, 0xb4, 0x98, 0x00,
>-  0x25, 0x92, 0xc9, 0x63, 0xa8, 0x00,
>-  0xfd, 0x9d, 0xd4, 0x22, 0x30, 0x00
>-};
>-
>-const uint8_t kMaskRandom37_2[12] = {
>-  0xce, 0x67, 0x33, 0x9d, 0xc0, 0x00,
>-  0x39, 0x9c, 0xce, 0x73, 0x38, 0x00
>-};
>-
>-const uint8_t kMaskRandom37_20[120] = {
>-  0x8c, 0xc6, 0x63, 0x38, 0xc0, 0x00,
>-  0x27, 0x13, 0x89, 0xc4, 0x70, 0x00,
>-  0x32, 0x99, 0x4c, 0xa3, 0x48, 0x00,
>-  0x61, 0xb0, 0xd8, 0x64, 0x98, 0x00,
>-  0x5c, 0x2e, 0x17, 0x0e, 0x20, 0x00,
>-  0xcc, 0x66, 0x33, 0x1d, 0x40, 0x00,
>-  0x2b, 0x15, 0x8a, 0xc6, 0x70, 0x00,
>-  0x32, 0x99, 0x4c, 0xb4, 0x98, 0x00,
>-  0x25, 0x92, 0xc9, 0x63, 0xa8, 0x00,
>-  0xfd, 0x9d, 0xd4, 0x22, 0x30, 0x00,
>-  0x44, 0xa2, 0x51, 0x29, 0xc0, 0x00,
>-  0x66, 0x26, 0x19, 0x9c, 0x20, 0x00,
>-  0x90, 0x49, 0x44, 0xb0, 0x38, 0x00,
>-  0x01, 0xa5, 0xb0, 0xc4, 0x28, 0x00,
>-  0x0e, 0x12, 0xa3, 0x0a, 0x50, 0x00,
>-  0x13, 0x0b, 0x04, 0x56, 0xc0, 0x00,
>-  0x20, 0xd0, 0x48, 0x64, 0xd0, 0x00,
>-  0xc2, 0x51, 0x28, 0x8b, 0x00, 0x00,
>-  0x29, 0x0c, 0x86, 0x03, 0x38, 0x00,
>-  0xe5, 0x44, 0xda, 0x3a, 0xc8, 0x00
>-};
>-
>-const uint8_t kMaskRandom37_21[126] = {
>-  0x8c, 0xc6, 0x63, 0x38, 0xc0, 0x00,
>-  0x27, 0x13, 0x89, 0xc4, 0x70, 0x00,
>-  0x32, 0x99, 0x4c, 0xa3, 0x48, 0x00,
>-  0x61, 0xb0, 0xd8, 0x64, 0x98, 0x00,
>-  0x5c, 0x2e, 0x17, 0x0e, 0x20, 0x00,
>-  0xcc, 0x66, 0x33, 0x1d, 0x40, 0x00,
>-  0x2b, 0x15, 0x8a, 0xc6, 0x70, 0x00,
>-  0x32, 0x99, 0x4c, 0xb4, 0x98, 0x00,
>-  0x25, 0x92, 0xc9, 0x63, 0xa8, 0x00,
>-  0xfd, 0x9d, 0xd4, 0x22, 0x30, 0x00,
>-  0x8c, 0xc6, 0x63, 0x38, 0xc0, 0x00,
>-  0x27, 0x13, 0x89, 0xc4, 0x70, 0x00,
>-  0x32, 0x99, 0x4c, 0xa3, 0x48, 0x00,
>-  0x61, 0xb0, 0xd8, 0x64, 0x98, 0x00,
>-  0x5c, 0x2e, 0x17, 0x0e, 0x20, 0x00,
>-  0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
>-  0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
>-  0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
>-  0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
>-  0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
>-  0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom37_22[132] = {
>-  0x8c, 0xc6, 0x63, 0x38, 0xc0, 0x00,
>-  0x27, 0x13, 0x89, 0xc4, 0x70, 0x00,
>-  0x32, 0x99, 0x4c, 0xa3, 0x48, 0x00,
>-  0x61, 0xb0, 0xd8, 0x64, 0x98, 0x00,
>-  0x5c, 0x2e, 0x17, 0x0e, 0x20, 0x00,
>-  0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
>-  0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
>-  0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
>-  0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
>-  0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
>-  0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00,
>-  0x8c, 0xc6, 0x63, 0x38, 0xc0, 0x00,
>-  0x27, 0x13, 0x89, 0xc4, 0x70, 0x00,
>-  0x32, 0x99, 0x4c, 0xa3, 0x48, 0x00,
>-  0x61, 0xb0, 0xd8, 0x64, 0x98, 0x00,
>-  0x5c, 0x2e, 0x17, 0x0e, 0x20, 0x00,
>-  0xcc, 0x66, 0x33, 0x1d, 0x40, 0x00,
>-  0x2b, 0x15, 0x8a, 0xc6, 0x70, 0x00,
>-  0x32, 0x99, 0x4c, 0xb4, 0x98, 0x00,
>-  0x25, 0x92, 0xc9, 0x63, 0xa8, 0x00,
>-  0xfd, 0x9d, 0xd4, 0x22, 0x30, 0x00,
>-  0xe4, 0xd3, 0xff, 0x5a, 0x28, 0x00
>-};
>-
>-const uint8_t kMaskRandom37_23[138] = {
>-  0x8c, 0xc6, 0x63, 0x38, 0xc0, 0x00,
>-  0x27, 0x13, 0x89, 0xc4, 0x70, 0x00,
>-  0x32, 0x99, 0x4c, 0xa3, 0x48, 0x00,
>-  0x61, 0xb0, 0xd8, 0x64, 0x98, 0x00,
>-  0x5c, 0x2e, 0x17, 0x0e, 0x20, 0x00,
>-  0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
>-  0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
>-  0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
>-  0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
>-  0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
>-  0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00,
>-  0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
>-  0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
>-  0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
>-  0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
>-  0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
>-  0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00,
>-  0x8c, 0xc6, 0x63, 0x38, 0xc0, 0x00,
>-  0x27, 0x13, 0x89, 0xc4, 0x70, 0x00,
>-  0x32, 0x99, 0x4c, 0xa3, 0x48, 0x00,
>-  0x61, 0xb0, 0xd8, 0x64, 0x98, 0x00,
>-  0x5c, 0x2e, 0x17, 0x0e, 0x20, 0x00,
>-  0x5b, 0x0c, 0x64, 0x32, 0x20, 0x00
>-};
>-
>-const uint8_t kMaskRandom37_24[144] = {
>-  0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
>-  0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
>-  0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
>-  0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
>-  0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
>-  0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00,
>-  0x8c, 0xc6, 0x63, 0x38, 0xc0, 0x00,
>-  0x27, 0x13, 0x89, 0xc4, 0x70, 0x00,
>-  0x32, 0x99, 0x4c, 0xa3, 0x48, 0x00,
>-  0x61, 0xb0, 0xd8, 0x64, 0x98, 0x00,
>-  0x5c, 0x2e, 0x17, 0x0e, 0x20, 0x00,
>-  0x5b, 0x0c, 0x64, 0x32, 0x20, 0x00,
>-  0x8c, 0xc6, 0x63, 0x38, 0xc0, 0x00,
>-  0x27, 0x13, 0x89, 0xc4, 0x70, 0x00,
>-  0x32, 0x99, 0x4c, 0xa3, 0x48, 0x00,
>-  0x61, 0xb0, 0xd8, 0x64, 0x98, 0x00,
>-  0x5c, 0x2e, 0x17, 0x0e, 0x20, 0x00,
>-  0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
>-  0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
>-  0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
>-  0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
>-  0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
>-  0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00,
>-  0xad, 0x58, 0xb2, 0x36, 0x68, 0x00
>-};
>-
>-const uint8_t kMaskRandom37_25[150] = {
>-  0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
>-  0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
>-  0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
>-  0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
>-  0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
>-  0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00,
>-  0x8c, 0xc6, 0x63, 0x38, 0xc0, 0x00,
>-  0x27, 0x13, 0x89, 0xc4, 0x70, 0x00,
>-  0x32, 0x99, 0x4c, 0xa3, 0x48, 0x00,
>-  0x61, 0xb0, 0xd8, 0x64, 0x98, 0x00,
>-  0x5c, 0x2e, 0x17, 0x0e, 0x20, 0x00,
>-  0x5b, 0x0c, 0x64, 0x32, 0x20, 0x00,
>-  0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
>-  0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
>-  0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
>-  0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
>-  0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
>-  0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00,
>-  0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
>-  0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
>-  0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
>-  0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
>-  0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
>-  0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
>-  0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00
>-};
>-
>-const uint8_t kMaskRandom37_26[156] = {
>-  0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
>-  0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
>-  0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
>-  0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
>-  0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
>-  0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00,
>-  0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
>-  0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
>-  0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
>-  0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
>-  0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
>-  0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
>-  0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00,
>-  0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
>-  0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
>-  0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
>-  0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
>-  0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
>-  0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00,
>-  0x8c, 0xc6, 0x63, 0x38, 0xc0, 0x00,
>-  0x27, 0x13, 0x89, 0xc4, 0x70, 0x00,
>-  0x32, 0x99, 0x4c, 0xa3, 0x48, 0x00,
>-  0x61, 0xb0, 0xd8, 0x64, 0x98, 0x00,
>-  0x5c, 0x2e, 0x17, 0x0e, 0x20, 0x00,
>-  0x5b, 0x0c, 0x64, 0x32, 0x20, 0x00,
>-  0x7f, 0xb2, 0x5a, 0xaa, 0x20, 0x00
>-};
>-
>-const uint8_t kMaskRandom37_27[162] = {
>-  0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
>-  0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
>-  0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
>-  0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
>-  0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
>-  0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00,
>-  0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
>-  0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
>-  0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
>-  0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
>-  0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
>-  0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
>-  0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00,
>-  0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
>-  0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
>-  0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
>-  0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
>-  0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
>-  0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
>-  0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00,
>-  0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
>-  0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
>-  0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
>-  0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
>-  0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
>-  0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00,
>-  0x7f, 0x4f, 0xdb, 0x89, 0xd8, 0x00
>-};
>-
>-const uint8_t kMaskRandom37_28[168] = {
>-  0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
>-  0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
>-  0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
>-  0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
>-  0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
>-  0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
>-  0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00,
>-  0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
>-  0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
>-  0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
>-  0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
>-  0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
>-  0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00,
>-  0x7f, 0x4f, 0xdb, 0x89, 0xd8, 0x00,
>-  0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
>-  0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
>-  0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
>-  0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
>-  0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
>-  0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00,
>-  0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
>-  0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
>-  0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
>-  0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
>-  0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
>-  0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
>-  0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00,
>-  0x7b, 0xc4, 0x24, 0xbf, 0x10, 0x00
>-};
>-
>-const uint8_t kMaskRandom37_29[174] = {
>-  0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
>-  0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
>-  0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
>-  0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
>-  0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
>-  0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
>-  0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00,
>-  0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
>-  0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
>-  0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
>-  0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
>-  0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
>-  0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00,
>-  0x7f, 0x4f, 0xdb, 0x89, 0xd8, 0x00,
>-  0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
>-  0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
>-  0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
>-  0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
>-  0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
>-  0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
>-  0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00,
>-  0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
>-  0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
>-  0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
>-  0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
>-  0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
>-  0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
>-  0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
>-  0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00
>-};
>-
>-const uint8_t kMaskRandom37_3[18] = {
>-  0xcc, 0x66, 0x33, 0x19, 0xc0, 0x00,
>-  0x27, 0x15, 0x89, 0xcb, 0x30, 0x00,
>-  0x92, 0xc9, 0x64, 0xb4, 0x98, 0x00
>-};
>-
>-const uint8_t kMaskRandom37_30[180] = {
>-  0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
>-  0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
>-  0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
>-  0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
>-  0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
>-  0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
>-  0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00,
>-  0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
>-  0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
>-  0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
>-  0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
>-  0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
>-  0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
>-  0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
>-  0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00,
>-  0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
>-  0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
>-  0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
>-  0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
>-  0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
>-  0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
>-  0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00,
>-  0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
>-  0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
>-  0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
>-  0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
>-  0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
>-  0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00,
>-  0x7f, 0x4f, 0xdb, 0x89, 0xd8, 0x00,
>-  0x1d, 0x8e, 0x11, 0xb0, 0xe8, 0x00
>-};
>-
>-const uint8_t kMaskRandom37_31[186] = {
>-  0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
>-  0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
>-  0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
>-  0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
>-  0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
>-  0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
>-  0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00,
>-  0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
>-  0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
>-  0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
>-  0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
>-  0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
>-  0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
>-  0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
>-  0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00,
>-  0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
>-  0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
>-  0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
>-  0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
>-  0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
>-  0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
>-  0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
>-  0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00,
>-  0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
>-  0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
>-  0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
>-  0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
>-  0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
>-  0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
>-  0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00,
>-  0xef, 0xf2, 0x1f, 0x9d, 0x78, 0x00
>-};
>-
>-const uint8_t kMaskRandom37_32[192] = {
>-  0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
>-  0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
>-  0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
>-  0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
>-  0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
>-  0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
>-  0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
>-  0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00,
>-  0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
>-  0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
>-  0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
>-  0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
>-  0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
>-  0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
>-  0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00,
>-  0xef, 0xf2, 0x1f, 0x9d, 0x78, 0x00,
>-  0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
>-  0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
>-  0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
>-  0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
>-  0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
>-  0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
>-  0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00,
>-  0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
>-  0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
>-  0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
>-  0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
>-  0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
>-  0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
>-  0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
>-  0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00,
>-  0xf7, 0x95, 0x57, 0x8c, 0x40, 0x00
>-};
>-
>-const uint8_t kMaskRandom37_33[198] = {
>-  0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
>-  0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
>-  0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
>-  0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
>-  0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
>-  0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
>-  0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
>-  0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00,
>-  0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
>-  0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
>-  0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
>-  0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
>-  0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
>-  0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
>-  0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00,
>-  0xef, 0xf2, 0x1f, 0x9d, 0x78, 0x00,
>-  0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
>-  0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
>-  0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
>-  0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
>-  0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
>-  0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
>-  0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
>-  0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00,
>-  0x4c, 0x26, 0x13, 0x09, 0xc0, 0x00,
>-  0x66, 0x33, 0x19, 0x9c, 0x60, 0x00,
>-  0x91, 0x48, 0xa4, 0x50, 0x38, 0x00,
>-  0x42, 0xa1, 0x50, 0xa4, 0x28, 0x00,
>-  0xa4, 0x52, 0x29, 0x0a, 0x50, 0x00,
>-  0x13, 0x09, 0x84, 0xd6, 0x80, 0x00,
>-  0x30, 0x98, 0x4c, 0x24, 0xd0, 0x00,
>-  0x88, 0xc4, 0x62, 0x2b, 0x08, 0x00,
>-  0x09, 0x04, 0x82, 0x43, 0x30, 0x00
>-};
>-
>-const uint8_t kMaskRandom37_34[204] = {
>-  0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
>-  0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
>-  0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
>-  0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
>-  0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
>-  0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
>-  0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
>-  0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00,
>-  0x4c, 0x26, 0x13, 0x09, 0xc0, 0x00,
>-  0x66, 0x33, 0x19, 0x9c, 0x60, 0x00,
>-  0x91, 0x48, 0xa4, 0x50, 0x38, 0x00,
>-  0x42, 0xa1, 0x50, 0xa4, 0x28, 0x00,
>-  0xa4, 0x52, 0x29, 0x0a, 0x50, 0x00,
>-  0x13, 0x09, 0x84, 0xd6, 0x80, 0x00,
>-  0x30, 0x98, 0x4c, 0x24, 0xd0, 0x00,
>-  0x88, 0xc4, 0x62, 0x2b, 0x08, 0x00,
>-  0x09, 0x04, 0x82, 0x43, 0x30, 0x00,
>-  0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
>-  0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
>-  0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
>-  0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
>-  0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
>-  0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
>-  0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
>-  0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00,
>-  0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
>-  0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
>-  0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
>-  0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
>-  0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
>-  0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
>-  0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00,
>-  0xef, 0xf2, 0x1f, 0x9d, 0x78, 0x00,
>-  0x31, 0x9c, 0xfb, 0x37, 0xc0, 0x00
>-};
>-
>-const uint8_t kMaskRandom37_35[210] = {
>-  0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
>-  0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
>-  0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
>-  0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
>-  0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
>-  0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
>-  0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
>-  0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00,
>-  0x4c, 0x26, 0x13, 0x09, 0xc0, 0x00,
>-  0x66, 0x33, 0x19, 0x9c, 0x60, 0x00,
>-  0x91, 0x48, 0xa4, 0x50, 0x38, 0x00,
>-  0x42, 0xa1, 0x50, 0xa4, 0x28, 0x00,
>-  0xa4, 0x52, 0x29, 0x0a, 0x50, 0x00,
>-  0x13, 0x09, 0x84, 0xd6, 0x80, 0x00,
>-  0x30, 0x98, 0x4c, 0x24, 0xd0, 0x00,
>-  0x88, 0xc4, 0x62, 0x2b, 0x08, 0x00,
>-  0x09, 0x04, 0x82, 0x43, 0x30, 0x00,
>-  0x4c, 0x26, 0x13, 0x09, 0xc0, 0x00,
>-  0x66, 0x33, 0x19, 0x9c, 0x60, 0x00,
>-  0x91, 0x48, 0xa4, 0x50, 0x38, 0x00,
>-  0x42, 0xa1, 0x50, 0xa4, 0x28, 0x00,
>-  0xa4, 0x52, 0x29, 0x0a, 0x50, 0x00,
>-  0x13, 0x09, 0x84, 0xd6, 0x80, 0x00,
>-  0x30, 0x98, 0x4c, 0x24, 0xd0, 0x00,
>-  0x88, 0xc4, 0x62, 0x2b, 0x08, 0x00,
>-  0x09, 0x04, 0x82, 0x43, 0x30, 0x00,
>-  0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
>-  0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
>-  0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
>-  0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
>-  0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
>-  0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
>-  0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
>-  0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00,
>-  0xd0, 0x03, 0x54, 0x65, 0xc8, 0x00
>-};
>-
>-const uint8_t kMaskRandom37_36[216] = {
>-  0x4c, 0x26, 0x13, 0x09, 0xc0, 0x00,
>-  0x66, 0x33, 0x19, 0x9c, 0x60, 0x00,
>-  0x91, 0x48, 0xa4, 0x50, 0x38, 0x00,
>-  0x42, 0xa1, 0x50, 0xa4, 0x28, 0x00,
>-  0xa4, 0x52, 0x29, 0x0a, 0x50, 0x00,
>-  0x13, 0x09, 0x84, 0xd6, 0x80, 0x00,
>-  0x30, 0x98, 0x4c, 0x24, 0xd0, 0x00,
>-  0x88, 0xc4, 0x62, 0x2b, 0x08, 0x00,
>-  0x09, 0x04, 0x82, 0x43, 0x30, 0x00,
>-  0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
>-  0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
>-  0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
>-  0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
>-  0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
>-  0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
>-  0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
>-  0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00,
>-  0xd0, 0x03, 0x54, 0x65, 0xc8, 0x00,
>-  0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
>-  0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
>-  0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
>-  0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
>-  0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
>-  0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
>-  0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
>-  0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00,
>-  0x4c, 0x26, 0x13, 0x09, 0xc0, 0x00,
>-  0x66, 0x33, 0x19, 0x9c, 0x60, 0x00,
>-  0x91, 0x48, 0xa4, 0x50, 0x38, 0x00,
>-  0x42, 0xa1, 0x50, 0xa4, 0x28, 0x00,
>-  0xa4, 0x52, 0x29, 0x0a, 0x50, 0x00,
>-  0x13, 0x09, 0x84, 0xd6, 0x80, 0x00,
>-  0x30, 0x98, 0x4c, 0x24, 0xd0, 0x00,
>-  0x88, 0xc4, 0x62, 0x2b, 0x08, 0x00,
>-  0x09, 0x04, 0x82, 0x43, 0x30, 0x00,
>-  0xc3, 0xc7, 0xce, 0xd8, 0x50, 0x00
>-};
>-
>-const uint8_t kMaskRandom37_37[222] = {
>-  0x4c, 0x26, 0x13, 0x09, 0xc0, 0x00,
>-  0x66, 0x33, 0x19, 0x9c, 0x60, 0x00,
>-  0x91, 0x48, 0xa4, 0x50, 0x38, 0x00,
>-  0x42, 0xa1, 0x50, 0xa4, 0x28, 0x00,
>-  0xa4, 0x52, 0x29, 0x0a, 0x50, 0x00,
>-  0x13, 0x09, 0x84, 0xd6, 0x80, 0x00,
>-  0x30, 0x98, 0x4c, 0x24, 0xd0, 0x00,
>-  0x88, 0xc4, 0x62, 0x2b, 0x08, 0x00,
>-  0x09, 0x04, 0x82, 0x43, 0x30, 0x00,
>-  0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
>-  0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
>-  0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
>-  0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
>-  0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
>-  0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
>-  0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
>-  0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00,
>-  0xd0, 0x03, 0x54, 0x65, 0xc8, 0x00,
>-  0x4c, 0x26, 0x13, 0x09, 0xc0, 0x00,
>-  0x66, 0x33, 0x19, 0x9c, 0x60, 0x00,
>-  0x91, 0x48, 0xa4, 0x50, 0x38, 0x00,
>-  0x42, 0xa1, 0x50, 0xa4, 0x28, 0x00,
>-  0xa4, 0x52, 0x29, 0x0a, 0x50, 0x00,
>-  0x13, 0x09, 0x84, 0xd6, 0x80, 0x00,
>-  0x30, 0x98, 0x4c, 0x24, 0xd0, 0x00,
>-  0x88, 0xc4, 0x62, 0x2b, 0x08, 0x00,
>-  0x09, 0x04, 0x82, 0x43, 0x30, 0x00,
>-  0x2c, 0x16, 0x13, 0x09, 0x80, 0x00,
>-  0x91, 0x48, 0x99, 0x8a, 0x20, 0x00,
>-  0xc0, 0xe0, 0x64, 0x54, 0x08, 0x00,
>-  0x06, 0x83, 0x50, 0xa0, 0x98, 0x00,
>-  0xc8, 0x64, 0x29, 0x00, 0x70, 0x00,
>-  0x45, 0x22, 0x84, 0xd0, 0xc0, 0x00,
>-  0x30, 0x98, 0x4c, 0x25, 0x20, 0x00,
>-  0xa2, 0x51, 0x22, 0x28, 0x48, 0x00,
>-  0xd0, 0x03, 0x42, 0x53, 0x00, 0x00,
>-  0xee, 0xf5, 0xb3, 0x66, 0x10, 0x00
>-};
>-
>-const uint8_t kMaskRandom37_4[24] = {
>-  0xec, 0x76, 0x3b, 0x1c, 0xc0, 0x00,
>-  0x67, 0x33, 0x99, 0xc6, 0x70, 0x00,
>-  0xb1, 0xd8, 0xec, 0x73, 0x18, 0x00,
>-  0x5a, 0xad, 0x56, 0xa5, 0xa8, 0x00
>-};
>-
>-const uint8_t kMaskRandom37_5[30] = {
>-  0x4c, 0xa6, 0x53, 0x39, 0xc0, 0x00,
>-  0x66, 0x33, 0x19, 0x8c, 0x70, 0x00,
>-  0x19, 0xd0, 0xe8, 0x73, 0x18, 0x00,
>-  0x9c, 0x89, 0x64, 0xa9, 0xa8, 0x00,
>-  0xe3, 0x4c, 0x2e, 0x26, 0x60, 0x00
>-};
>-
>-const uint8_t kMaskRandom37_6[36] = {
>-  0xcc, 0x26, 0x13, 0x0d, 0x80, 0x00,
>-  0x45, 0x62, 0x91, 0x5a, 0x20, 0x00,
>-  0xb0, 0x98, 0x4c, 0x34, 0x18, 0x00,
>-  0x8a, 0x85, 0x62, 0xa0, 0xa8, 0x00,
>-  0x29, 0x53, 0x09, 0x82, 0xd0, 0x00,
>-  0xa6, 0x0a, 0xa5, 0x51, 0x40, 0x00
>-};
>-
>-const uint8_t kMaskRandom37_7[42] = {
>-  0x44, 0xa2, 0x71, 0x28, 0xc0, 0x00,
>-  0x23, 0x11, 0x88, 0xc6, 0x60, 0x00,
>-  0x91, 0x48, 0xa4, 0x47, 0x08, 0x00,
>-  0x0a, 0x85, 0x52, 0xa0, 0xa8, 0x00,
>-  0x34, 0x1a, 0x0d, 0x12, 0x50, 0x00,
>-  0x0b, 0x06, 0xa2, 0xd2, 0x80, 0x00,
>-  0xe0, 0x64, 0x32, 0x09, 0x30, 0x00
>-};
>-
>-const uint8_t kMaskRandom37_8[48] = {
>-  0x16, 0x0b, 0x05, 0x84, 0xe0, 0x00,
>-  0xc2, 0x61, 0x30, 0x91, 0x30, 0x00,
>-  0x60, 0xb0, 0x58, 0x3a, 0x08, 0x00,
>-  0x85, 0x42, 0xa1, 0x44, 0x98, 0x00,
>-  0x4c, 0x26, 0x33, 0x08, 0x50, 0x00,
>-  0x29, 0x14, 0x8a, 0x58, 0xc0, 0x00,
>-  0x11, 0x88, 0xc4, 0x66, 0x30, 0x00,
>-  0xb0, 0x58, 0x2c, 0x03, 0x18, 0x00
>-};
>-
>-const uint8_t kMaskRandom37_9[54] = {
>-  0x44, 0xa2, 0x51, 0x29, 0xc0, 0x00,
>-  0x66, 0x26, 0x19, 0x9c, 0x20, 0x00,
>-  0x90, 0x49, 0x44, 0xb0, 0x38, 0x00,
>-  0x01, 0xa5, 0xb0, 0xc4, 0x28, 0x00,
>-  0x0e, 0x12, 0xa3, 0x0a, 0x50, 0x00,
>-  0x13, 0x0b, 0x04, 0x56, 0xc0, 0x00,
>-  0x20, 0xd0, 0x48, 0x64, 0xd0, 0x00,
>-  0xc2, 0x51, 0x28, 0x8b, 0x00, 0x00,
>-  0x29, 0x0c, 0x86, 0x03, 0x38, 0x00
>-};
>-
>-const uint8_t kMaskRandom38_1[6] = {
>-  0xff, 0xff, 0xff, 0xff, 0xfc, 0x00
>-};
>-
>-const uint8_t kMaskRandom38_10[60] = {
>-  0x8c, 0xe3, 0x11, 0x9c, 0x60, 0x00,
>-  0x27, 0x11, 0xc4, 0xe2, 0x38, 0x00,
>-  0x32, 0x8d, 0x26, 0x51, 0xa4, 0x00,
>-  0x61, 0x92, 0x6c, 0x32, 0x4c, 0x00,
>-  0x5c, 0x38, 0x8b, 0x87, 0x10, 0x00,
>-  0xcc, 0x75, 0x19, 0x8e, 0xa0, 0x00,
>-  0x2b, 0x19, 0xc5, 0x63, 0x38, 0x00,
>-  0x32, 0xd2, 0x66, 0x5a, 0x4c, 0x00,
>-  0x25, 0x8e, 0xa4, 0xb1, 0xd4, 0x00,
>-  0x50, 0x88, 0xca, 0x11, 0x18, 0x00
>-};
>-
>-const uint8_t kMaskRandom38_11[66] = {
>-  0x8c, 0xe3, 0x11, 0x9c, 0x60, 0x00,
>-  0x27, 0x11, 0xc4, 0xe2, 0x38, 0x00,
>-  0x32, 0x8d, 0x26, 0x51, 0xa4, 0x00,
>-  0x61, 0x92, 0x6c, 0x32, 0x4c, 0x00,
>-  0x5c, 0x38, 0x8b, 0x87, 0x10, 0x00,
>-  0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
>-  0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
>-  0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
>-  0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
>-  0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
>-  0x89, 0x70, 0x11, 0x2e, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom38_12[72] = {
>-  0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
>-  0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
>-  0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
>-  0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
>-  0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
>-  0x89, 0x70, 0x11, 0x2e, 0x00, 0x00,
>-  0x8c, 0xe3, 0x11, 0x9c, 0x60, 0x00,
>-  0x27, 0x11, 0xc4, 0xe2, 0x38, 0x00,
>-  0x32, 0x8d, 0x26, 0x51, 0xa4, 0x00,
>-  0x61, 0x92, 0x6c, 0x32, 0x4c, 0x00,
>-  0x5c, 0x38, 0x8b, 0x87, 0x10, 0x00,
>-  0x90, 0xc8, 0x92, 0x19, 0x10, 0x00
>-};
>-
>-const uint8_t kMaskRandom38_13[78] = {
>-  0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
>-  0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
>-  0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
>-  0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
>-  0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
>-  0x89, 0x70, 0x11, 0x2e, 0x00, 0x00,
>-  0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
>-  0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
>-  0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
>-  0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
>-  0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
>-  0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
>-  0x34, 0x60, 0x86, 0x8c, 0x10, 0x00
>-};
>-
>-const uint8_t kMaskRandom38_14[84] = {
>-  0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
>-  0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
>-  0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
>-  0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
>-  0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
>-  0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
>-  0x34, 0x60, 0x86, 0x8c, 0x10, 0x00,
>-  0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
>-  0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
>-  0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
>-  0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
>-  0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
>-  0x89, 0x70, 0x11, 0x2e, 0x00, 0x00,
>-  0x6e, 0x27, 0x6d, 0xc4, 0xec, 0x00
>-};
>-
>-const uint8_t kMaskRandom38_15[90] = {
>-  0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
>-  0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
>-  0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
>-  0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
>-  0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
>-  0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
>-  0x34, 0x60, 0x86, 0x8c, 0x10, 0x00,
>-  0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
>-  0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
>-  0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
>-  0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
>-  0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
>-  0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
>-  0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
>-  0xa2, 0x28, 0x34, 0x45, 0x04, 0x00
>-};
>-
>-const uint8_t kMaskRandom38_16[96] = {
>-  0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
>-  0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
>-  0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
>-  0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
>-  0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
>-  0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
>-  0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
>-  0xa2, 0x28, 0x34, 0x45, 0x04, 0x00,
>-  0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
>-  0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
>-  0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
>-  0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
>-  0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
>-  0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
>-  0x34, 0x60, 0x86, 0x8c, 0x10, 0x00,
>-  0x7e, 0x75, 0xef, 0xce, 0xbc, 0x00
>-};
>-
>-const uint8_t kMaskRandom38_17[102] = {
>-  0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
>-  0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
>-  0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
>-  0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
>-  0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
>-  0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
>-  0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
>-  0xa2, 0x28, 0x34, 0x45, 0x04, 0x00,
>-  0x4c, 0x27, 0x09, 0x84, 0xe0, 0x00,
>-  0x66, 0x71, 0x8c, 0xce, 0x30, 0x00,
>-  0x91, 0x40, 0xf2, 0x28, 0x1c, 0x00,
>-  0x42, 0x90, 0xa8, 0x52, 0x14, 0x00,
>-  0xa4, 0x29, 0x54, 0x85, 0x28, 0x00,
>-  0x13, 0x5a, 0x02, 0x6b, 0x40, 0x00,
>-  0x30, 0x93, 0x46, 0x12, 0x68, 0x00,
>-  0x88, 0xac, 0x31, 0x15, 0x84, 0x00,
>-  0x09, 0x0c, 0xc1, 0x21, 0x98, 0x00
>-};
>-
>-const uint8_t kMaskRandom38_18[108] = {
>-  0x4c, 0x27, 0x09, 0x84, 0xe0, 0x00,
>-  0x66, 0x71, 0x8c, 0xce, 0x30, 0x00,
>-  0x91, 0x40, 0xf2, 0x28, 0x1c, 0x00,
>-  0x42, 0x90, 0xa8, 0x52, 0x14, 0x00,
>-  0xa4, 0x29, 0x54, 0x85, 0x28, 0x00,
>-  0x13, 0x5a, 0x02, 0x6b, 0x40, 0x00,
>-  0x30, 0x93, 0x46, 0x12, 0x68, 0x00,
>-  0x88, 0xac, 0x31, 0x15, 0x84, 0x00,
>-  0x09, 0x0c, 0xc1, 0x21, 0x98, 0x00,
>-  0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
>-  0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
>-  0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
>-  0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
>-  0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
>-  0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
>-  0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
>-  0xa2, 0x28, 0x34, 0x45, 0x04, 0x00,
>-  0x51, 0x97, 0x2a, 0x32, 0xe4, 0x00
>-};
>-
>-const uint8_t kMaskRandom38_19[114] = {
>-  0x4c, 0x27, 0x09, 0x84, 0xe0, 0x00,
>-  0x66, 0x71, 0x8c, 0xce, 0x30, 0x00,
>-  0x91, 0x40, 0xf2, 0x28, 0x1c, 0x00,
>-  0x42, 0x90, 0xa8, 0x52, 0x14, 0x00,
>-  0xa4, 0x29, 0x54, 0x85, 0x28, 0x00,
>-  0x13, 0x5a, 0x02, 0x6b, 0x40, 0x00,
>-  0x30, 0x93, 0x46, 0x12, 0x68, 0x00,
>-  0x88, 0xac, 0x31, 0x15, 0x84, 0x00,
>-  0x09, 0x0c, 0xc1, 0x21, 0x98, 0x00,
>-  0x4c, 0x26, 0x09, 0x84, 0xc0, 0x00,
>-  0x66, 0x28, 0x8c, 0xc5, 0x10, 0x00,
>-  0x91, 0x50, 0x32, 0x2a, 0x04, 0x00,
>-  0x42, 0x82, 0x68, 0x50, 0x4c, 0x00,
>-  0xa4, 0x01, 0xd4, 0x80, 0x38, 0x00,
>-  0x13, 0x43, 0x02, 0x68, 0x60, 0x00,
>-  0x30, 0x94, 0x86, 0x12, 0x90, 0x00,
>-  0x88, 0xa1, 0x31, 0x14, 0x24, 0x00,
>-  0x09, 0x4c, 0x01, 0x29, 0x80, 0x00,
>-  0xcd, 0x98, 0x59, 0xb3, 0x08, 0x00
>-};
>-
>-const uint8_t kMaskRandom38_2[12] = {
>-  0xce, 0x77, 0x19, 0xce, 0xe0, 0x00,
>-  0x39, 0xcc, 0xe7, 0x39, 0x9c, 0x00
>-};
>-
>-const uint8_t kMaskRandom38_20[120] = {
>-  0x8c, 0xe3, 0x11, 0x9c, 0x60, 0x00,
>-  0x27, 0x11, 0xc4, 0xe2, 0x38, 0x00,
>-  0x32, 0x8d, 0x26, 0x51, 0xa4, 0x00,
>-  0x61, 0x92, 0x6c, 0x32, 0x4c, 0x00,
>-  0x5c, 0x38, 0x8b, 0x87, 0x10, 0x00,
>-  0xcc, 0x75, 0x19, 0x8e, 0xa0, 0x00,
>-  0x2b, 0x19, 0xc5, 0x63, 0x38, 0x00,
>-  0x32, 0xd2, 0x66, 0x5a, 0x4c, 0x00,
>-  0x25, 0x8e, 0xa4, 0xb1, 0xd4, 0x00,
>-  0x50, 0x88, 0xca, 0x11, 0x18, 0x00,
>-  0x44, 0xa7, 0x08, 0x94, 0xe0, 0x00,
>-  0x66, 0x70, 0x8c, 0xce, 0x10, 0x00,
>-  0x12, 0xc0, 0xe2, 0x58, 0x1c, 0x00,
>-  0xc3, 0x10, 0xb8, 0x62, 0x14, 0x00,
>-  0x8c, 0x29, 0x51, 0x85, 0x28, 0x00,
>-  0x11, 0x5b, 0x02, 0x2b, 0x60, 0x00,
>-  0x21, 0x93, 0x44, 0x32, 0x68, 0x00,
>-  0xa2, 0x2c, 0x14, 0x45, 0x80, 0x00,
>-  0x18, 0x0c, 0xe3, 0x01, 0x9c, 0x00,
>-  0xe6, 0xbc, 0x88, 0xe3, 0x78, 0x00
>-};
>-
>-const uint8_t kMaskRandom38_21[126] = {
>-  0x8c, 0xe3, 0x11, 0x9c, 0x60, 0x00,
>-  0x27, 0x11, 0xc4, 0xe2, 0x38, 0x00,
>-  0x32, 0x8d, 0x26, 0x51, 0xa4, 0x00,
>-  0x61, 0x92, 0x6c, 0x32, 0x4c, 0x00,
>-  0x5c, 0x38, 0x8b, 0x87, 0x10, 0x00,
>-  0xcc, 0x75, 0x19, 0x8e, 0xa0, 0x00,
>-  0x2b, 0x19, 0xc5, 0x63, 0x38, 0x00,
>-  0x32, 0xd2, 0x66, 0x5a, 0x4c, 0x00,
>-  0x25, 0x8e, 0xa4, 0xb1, 0xd4, 0x00,
>-  0x50, 0x88, 0xca, 0x11, 0x18, 0x00,
>-  0x8c, 0xe3, 0x11, 0x9c, 0x60, 0x00,
>-  0x27, 0x11, 0xc4, 0xe2, 0x38, 0x00,
>-  0x32, 0x8d, 0x26, 0x51, 0xa4, 0x00,
>-  0x61, 0x92, 0x6c, 0x32, 0x4c, 0x00,
>-  0x5c, 0x38, 0x8b, 0x87, 0x10, 0x00,
>-  0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
>-  0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
>-  0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
>-  0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
>-  0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
>-  0x89, 0x70, 0x11, 0x2e, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom38_22[132] = {
>-  0x8c, 0xe3, 0x11, 0x9c, 0x60, 0x00,
>-  0x27, 0x11, 0xc4, 0xe2, 0x38, 0x00,
>-  0x32, 0x8d, 0x26, 0x51, 0xa4, 0x00,
>-  0x61, 0x92, 0x6c, 0x32, 0x4c, 0x00,
>-  0x5c, 0x38, 0x8b, 0x87, 0x10, 0x00,
>-  0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
>-  0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
>-  0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
>-  0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
>-  0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
>-  0x89, 0x70, 0x11, 0x2e, 0x00, 0x00,
>-  0x8c, 0xe3, 0x11, 0x9c, 0x60, 0x00,
>-  0x27, 0x11, 0xc4, 0xe2, 0x38, 0x00,
>-  0x32, 0x8d, 0x26, 0x51, 0xa4, 0x00,
>-  0x61, 0x92, 0x6c, 0x32, 0x4c, 0x00,
>-  0x5c, 0x38, 0x8b, 0x87, 0x10, 0x00,
>-  0xcc, 0x75, 0x19, 0x8e, 0xa0, 0x00,
>-  0x2b, 0x19, 0xc5, 0x63, 0x38, 0x00,
>-  0x32, 0xd2, 0x66, 0x5a, 0x4c, 0x00,
>-  0x25, 0x8e, 0xa4, 0xb1, 0xd4, 0x00,
>-  0x50, 0x88, 0xca, 0x11, 0x18, 0x00,
>-  0x0c, 0x3c, 0x48, 0x3d, 0x58, 0x00
>-};
>-
>-const uint8_t kMaskRandom38_23[138] = {
>-  0x8c, 0xe3, 0x11, 0x9c, 0x60, 0x00,
>-  0x27, 0x11, 0xc4, 0xe2, 0x38, 0x00,
>-  0x32, 0x8d, 0x26, 0x51, 0xa4, 0x00,
>-  0x61, 0x92, 0x6c, 0x32, 0x4c, 0x00,
>-  0x5c, 0x38, 0x8b, 0x87, 0x10, 0x00,
>-  0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
>-  0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
>-  0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
>-  0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
>-  0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
>-  0x89, 0x70, 0x11, 0x2e, 0x00, 0x00,
>-  0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
>-  0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
>-  0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
>-  0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
>-  0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
>-  0x89, 0x70, 0x11, 0x2e, 0x00, 0x00,
>-  0x8c, 0xe3, 0x11, 0x9c, 0x60, 0x00,
>-  0x27, 0x11, 0xc4, 0xe2, 0x38, 0x00,
>-  0x32, 0x8d, 0x26, 0x51, 0xa4, 0x00,
>-  0x61, 0x92, 0x6c, 0x32, 0x4c, 0x00,
>-  0x5c, 0x38, 0x8b, 0x87, 0x10, 0x00,
>-  0x90, 0xc8, 0x92, 0x19, 0x10, 0x00
>-};
>-
>-const uint8_t kMaskRandom38_24[144] = {
>-  0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
>-  0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
>-  0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
>-  0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
>-  0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
>-  0x89, 0x70, 0x11, 0x2e, 0x00, 0x00,
>-  0x8c, 0xe3, 0x11, 0x9c, 0x60, 0x00,
>-  0x27, 0x11, 0xc4, 0xe2, 0x38, 0x00,
>-  0x32, 0x8d, 0x26, 0x51, 0xa4, 0x00,
>-  0x61, 0x92, 0x6c, 0x32, 0x4c, 0x00,
>-  0x5c, 0x38, 0x8b, 0x87, 0x10, 0x00,
>-  0x90, 0xc8, 0x92, 0x19, 0x10, 0x00,
>-  0x8c, 0xe3, 0x11, 0x9c, 0x60, 0x00,
>-  0x27, 0x11, 0xc4, 0xe2, 0x38, 0x00,
>-  0x32, 0x8d, 0x26, 0x51, 0xa4, 0x00,
>-  0x61, 0x92, 0x6c, 0x32, 0x4c, 0x00,
>-  0x5c, 0x38, 0x8b, 0x87, 0x10, 0x00,
>-  0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
>-  0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
>-  0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
>-  0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
>-  0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
>-  0x89, 0x70, 0x11, 0x2e, 0x00, 0x00,
>-  0x93, 0xc8, 0xb3, 0xbe, 0x5c, 0x00
>-};
>-
>-const uint8_t kMaskRandom38_25[150] = {
>-  0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
>-  0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
>-  0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
>-  0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
>-  0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
>-  0x89, 0x70, 0x11, 0x2e, 0x00, 0x00,
>-  0x8c, 0xe3, 0x11, 0x9c, 0x60, 0x00,
>-  0x27, 0x11, 0xc4, 0xe2, 0x38, 0x00,
>-  0x32, 0x8d, 0x26, 0x51, 0xa4, 0x00,
>-  0x61, 0x92, 0x6c, 0x32, 0x4c, 0x00,
>-  0x5c, 0x38, 0x8b, 0x87, 0x10, 0x00,
>-  0x90, 0xc8, 0x92, 0x19, 0x10, 0x00,
>-  0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
>-  0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
>-  0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
>-  0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
>-  0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
>-  0x89, 0x70, 0x11, 0x2e, 0x00, 0x00,
>-  0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
>-  0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
>-  0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
>-  0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
>-  0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
>-  0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
>-  0x34, 0x60, 0x86, 0x8c, 0x10, 0x00
>-};
>-
>-const uint8_t kMaskRandom38_26[156] = {
>-  0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
>-  0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
>-  0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
>-  0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
>-  0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
>-  0x89, 0x70, 0x11, 0x2e, 0x00, 0x00,
>-  0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
>-  0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
>-  0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
>-  0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
>-  0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
>-  0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
>-  0x34, 0x60, 0x86, 0x8c, 0x10, 0x00,
>-  0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
>-  0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
>-  0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
>-  0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
>-  0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
>-  0x89, 0x70, 0x11, 0x2e, 0x00, 0x00,
>-  0x8c, 0xe3, 0x11, 0x9c, 0x60, 0x00,
>-  0x27, 0x11, 0xc4, 0xe2, 0x38, 0x00,
>-  0x32, 0x8d, 0x26, 0x51, 0xa4, 0x00,
>-  0x61, 0x92, 0x6c, 0x32, 0x4c, 0x00,
>-  0x5c, 0x38, 0x8b, 0x87, 0x10, 0x00,
>-  0x90, 0xc8, 0x92, 0x19, 0x10, 0x00,
>-  0x4b, 0xab, 0xfc, 0xe6, 0xe8, 0x00
>-};
>-
>-const uint8_t kMaskRandom38_27[162] = {
>-  0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
>-  0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
>-  0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
>-  0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
>-  0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
>-  0x89, 0x70, 0x11, 0x2e, 0x00, 0x00,
>-  0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
>-  0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
>-  0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
>-  0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
>-  0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
>-  0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
>-  0x34, 0x60, 0x86, 0x8c, 0x10, 0x00,
>-  0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
>-  0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
>-  0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
>-  0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
>-  0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
>-  0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
>-  0x34, 0x60, 0x86, 0x8c, 0x10, 0x00,
>-  0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
>-  0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
>-  0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
>-  0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
>-  0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
>-  0x89, 0x70, 0x11, 0x2e, 0x00, 0x00,
>-  0x6e, 0x27, 0x6d, 0xc4, 0xec, 0x00
>-};
>-
>-const uint8_t kMaskRandom38_28[168] = {
>-  0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
>-  0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
>-  0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
>-  0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
>-  0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
>-  0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
>-  0x34, 0x60, 0x86, 0x8c, 0x10, 0x00,
>-  0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
>-  0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
>-  0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
>-  0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
>-  0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
>-  0x89, 0x70, 0x11, 0x2e, 0x00, 0x00,
>-  0x6e, 0x27, 0x6d, 0xc4, 0xec, 0x00,
>-  0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
>-  0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
>-  0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
>-  0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
>-  0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
>-  0x89, 0x70, 0x11, 0x2e, 0x00, 0x00,
>-  0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
>-  0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
>-  0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
>-  0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
>-  0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
>-  0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
>-  0x34, 0x60, 0x86, 0x8c, 0x10, 0x00,
>-  0x70, 0x1b, 0x5b, 0x2c, 0x0c, 0x00
>-};
>-
>-const uint8_t kMaskRandom38_29[174] = {
>-  0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
>-  0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
>-  0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
>-  0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
>-  0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
>-  0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
>-  0x34, 0x60, 0x86, 0x8c, 0x10, 0x00,
>-  0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
>-  0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
>-  0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
>-  0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
>-  0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
>-  0x89, 0x70, 0x11, 0x2e, 0x00, 0x00,
>-  0x6e, 0x27, 0x6d, 0xc4, 0xec, 0x00,
>-  0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
>-  0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
>-  0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
>-  0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
>-  0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
>-  0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
>-  0x34, 0x60, 0x86, 0x8c, 0x10, 0x00,
>-  0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
>-  0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
>-  0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
>-  0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
>-  0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
>-  0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
>-  0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
>-  0xa2, 0x28, 0x34, 0x45, 0x04, 0x00
>-};
>-
>-const uint8_t kMaskRandom38_3[18] = {
>-  0xcc, 0x67, 0x19, 0x8c, 0xe0, 0x00,
>-  0x27, 0x2c, 0xc4, 0xe5, 0x98, 0x00,
>-  0x92, 0xd2, 0x72, 0x5a, 0x4c, 0x00
>-};
>-
>-const uint8_t kMaskRandom38_30[180] = {
>-  0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
>-  0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
>-  0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
>-  0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
>-  0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
>-  0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
>-  0x34, 0x60, 0x86, 0x8c, 0x10, 0x00,
>-  0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
>-  0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
>-  0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
>-  0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
>-  0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
>-  0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
>-  0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
>-  0xa2, 0x28, 0x34, 0x45, 0x04, 0x00,
>-  0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
>-  0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
>-  0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
>-  0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
>-  0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
>-  0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
>-  0x34, 0x60, 0x86, 0x8c, 0x10, 0x00,
>-  0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
>-  0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
>-  0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
>-  0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
>-  0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
>-  0x89, 0x70, 0x11, 0x2e, 0x00, 0x00,
>-  0x6e, 0x27, 0x6d, 0xc4, 0xec, 0x00,
>-  0x5b, 0x16, 0xdf, 0xb8, 0xd0, 0x00
>-};
>-
>-const uint8_t kMaskRandom38_31[186] = {
>-  0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
>-  0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
>-  0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
>-  0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
>-  0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
>-  0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
>-  0x34, 0x60, 0x86, 0x8c, 0x10, 0x00,
>-  0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
>-  0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
>-  0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
>-  0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
>-  0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
>-  0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
>-  0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
>-  0xa2, 0x28, 0x34, 0x45, 0x04, 0x00,
>-  0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
>-  0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
>-  0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
>-  0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
>-  0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
>-  0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
>-  0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
>-  0xa2, 0x28, 0x34, 0x45, 0x04, 0x00,
>-  0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
>-  0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
>-  0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
>-  0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
>-  0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
>-  0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
>-  0x34, 0x60, 0x86, 0x8c, 0x10, 0x00,
>-  0x7e, 0x75, 0xef, 0xce, 0xbc, 0x00
>-};
>-
>-const uint8_t kMaskRandom38_32[192] = {
>-  0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
>-  0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
>-  0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
>-  0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
>-  0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
>-  0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
>-  0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
>-  0xa2, 0x28, 0x34, 0x45, 0x04, 0x00,
>-  0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
>-  0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
>-  0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
>-  0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
>-  0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
>-  0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
>-  0x34, 0x60, 0x86, 0x8c, 0x10, 0x00,
>-  0x7e, 0x75, 0xef, 0xce, 0xbc, 0x00,
>-  0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
>-  0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
>-  0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
>-  0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
>-  0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
>-  0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
>-  0x34, 0x60, 0x86, 0x8c, 0x10, 0x00,
>-  0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
>-  0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
>-  0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
>-  0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
>-  0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
>-  0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
>-  0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
>-  0xa2, 0x28, 0x34, 0x45, 0x04, 0x00,
>-  0x33, 0x10, 0x02, 0x4e, 0x54, 0x00
>-};
>-
>-const uint8_t kMaskRandom38_33[198] = {
>-  0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
>-  0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
>-  0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
>-  0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
>-  0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
>-  0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
>-  0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
>-  0xa2, 0x28, 0x34, 0x45, 0x04, 0x00,
>-  0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
>-  0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
>-  0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
>-  0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
>-  0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
>-  0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
>-  0x34, 0x60, 0x86, 0x8c, 0x10, 0x00,
>-  0x7e, 0x75, 0xef, 0xce, 0xbc, 0x00,
>-  0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
>-  0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
>-  0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
>-  0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
>-  0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
>-  0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
>-  0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
>-  0xa2, 0x28, 0x34, 0x45, 0x04, 0x00,
>-  0x4c, 0x27, 0x09, 0x84, 0xe0, 0x00,
>-  0x66, 0x71, 0x8c, 0xce, 0x30, 0x00,
>-  0x91, 0x40, 0xf2, 0x28, 0x1c, 0x00,
>-  0x42, 0x90, 0xa8, 0x52, 0x14, 0x00,
>-  0xa4, 0x29, 0x54, 0x85, 0x28, 0x00,
>-  0x13, 0x5a, 0x02, 0x6b, 0x40, 0x00,
>-  0x30, 0x93, 0x46, 0x12, 0x68, 0x00,
>-  0x88, 0xac, 0x31, 0x15, 0x84, 0x00,
>-  0x09, 0x0c, 0xc1, 0x21, 0x98, 0x00
>-};
>-
>-const uint8_t kMaskRandom38_34[204] = {
>-  0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
>-  0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
>-  0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
>-  0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
>-  0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
>-  0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
>-  0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
>-  0xa2, 0x28, 0x34, 0x45, 0x04, 0x00,
>-  0x4c, 0x27, 0x09, 0x84, 0xe0, 0x00,
>-  0x66, 0x71, 0x8c, 0xce, 0x30, 0x00,
>-  0x91, 0x40, 0xf2, 0x28, 0x1c, 0x00,
>-  0x42, 0x90, 0xa8, 0x52, 0x14, 0x00,
>-  0xa4, 0x29, 0x54, 0x85, 0x28, 0x00,
>-  0x13, 0x5a, 0x02, 0x6b, 0x40, 0x00,
>-  0x30, 0x93, 0x46, 0x12, 0x68, 0x00,
>-  0x88, 0xac, 0x31, 0x15, 0x84, 0x00,
>-  0x09, 0x0c, 0xc1, 0x21, 0x98, 0x00,
>-  0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
>-  0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
>-  0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
>-  0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
>-  0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
>-  0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
>-  0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
>-  0xa2, 0x28, 0x34, 0x45, 0x04, 0x00,
>-  0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
>-  0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
>-  0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
>-  0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
>-  0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
>-  0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
>-  0x34, 0x60, 0x86, 0x8c, 0x10, 0x00,
>-  0x7e, 0x75, 0xef, 0xce, 0xbc, 0x00,
>-  0x91, 0x48, 0xfa, 0xf0, 0xd8, 0x00
>-};
>-
>-const uint8_t kMaskRandom38_35[210] = {
>-  0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
>-  0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
>-  0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
>-  0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
>-  0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
>-  0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
>-  0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
>-  0xa2, 0x28, 0x34, 0x45, 0x04, 0x00,
>-  0x4c, 0x27, 0x09, 0x84, 0xe0, 0x00,
>-  0x66, 0x71, 0x8c, 0xce, 0x30, 0x00,
>-  0x91, 0x40, 0xf2, 0x28, 0x1c, 0x00,
>-  0x42, 0x90, 0xa8, 0x52, 0x14, 0x00,
>-  0xa4, 0x29, 0x54, 0x85, 0x28, 0x00,
>-  0x13, 0x5a, 0x02, 0x6b, 0x40, 0x00,
>-  0x30, 0x93, 0x46, 0x12, 0x68, 0x00,
>-  0x88, 0xac, 0x31, 0x15, 0x84, 0x00,
>-  0x09, 0x0c, 0xc1, 0x21, 0x98, 0x00,
>-  0x4c, 0x27, 0x09, 0x84, 0xe0, 0x00,
>-  0x66, 0x71, 0x8c, 0xce, 0x30, 0x00,
>-  0x91, 0x40, 0xf2, 0x28, 0x1c, 0x00,
>-  0x42, 0x90, 0xa8, 0x52, 0x14, 0x00,
>-  0xa4, 0x29, 0x54, 0x85, 0x28, 0x00,
>-  0x13, 0x5a, 0x02, 0x6b, 0x40, 0x00,
>-  0x30, 0x93, 0x46, 0x12, 0x68, 0x00,
>-  0x88, 0xac, 0x31, 0x15, 0x84, 0x00,
>-  0x09, 0x0c, 0xc1, 0x21, 0x98, 0x00,
>-  0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
>-  0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
>-  0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
>-  0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
>-  0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
>-  0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
>-  0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
>-  0xa2, 0x28, 0x34, 0x45, 0x04, 0x00,
>-  0x51, 0x97, 0x2a, 0x32, 0xe4, 0x00
>-};
>-
>-const uint8_t kMaskRandom38_36[216] = {
>-  0x4c, 0x27, 0x09, 0x84, 0xe0, 0x00,
>-  0x66, 0x71, 0x8c, 0xce, 0x30, 0x00,
>-  0x91, 0x40, 0xf2, 0x28, 0x1c, 0x00,
>-  0x42, 0x90, 0xa8, 0x52, 0x14, 0x00,
>-  0xa4, 0x29, 0x54, 0x85, 0x28, 0x00,
>-  0x13, 0x5a, 0x02, 0x6b, 0x40, 0x00,
>-  0x30, 0x93, 0x46, 0x12, 0x68, 0x00,
>-  0x88, 0xac, 0x31, 0x15, 0x84, 0x00,
>-  0x09, 0x0c, 0xc1, 0x21, 0x98, 0x00,
>-  0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
>-  0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
>-  0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
>-  0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
>-  0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
>-  0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
>-  0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
>-  0xa2, 0x28, 0x34, 0x45, 0x04, 0x00,
>-  0x51, 0x97, 0x2a, 0x32, 0xe4, 0x00,
>-  0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
>-  0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
>-  0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
>-  0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
>-  0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
>-  0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
>-  0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
>-  0xa2, 0x28, 0x34, 0x45, 0x04, 0x00,
>-  0x4c, 0x27, 0x09, 0x84, 0xe0, 0x00,
>-  0x66, 0x71, 0x8c, 0xce, 0x30, 0x00,
>-  0x91, 0x40, 0xf2, 0x28, 0x1c, 0x00,
>-  0x42, 0x90, 0xa8, 0x52, 0x14, 0x00,
>-  0xa4, 0x29, 0x54, 0x85, 0x28, 0x00,
>-  0x13, 0x5a, 0x02, 0x6b, 0x40, 0x00,
>-  0x30, 0x93, 0x46, 0x12, 0x68, 0x00,
>-  0x88, 0xac, 0x31, 0x15, 0x84, 0x00,
>-  0x09, 0x0c, 0xc1, 0x21, 0x98, 0x00,
>-  0x80, 0x95, 0xc2, 0x68, 0x28, 0x00
>-};
>-
>-const uint8_t kMaskRandom38_37[222] = {
>-  0x4c, 0x27, 0x09, 0x84, 0xe0, 0x00,
>-  0x66, 0x71, 0x8c, 0xce, 0x30, 0x00,
>-  0x91, 0x40, 0xf2, 0x28, 0x1c, 0x00,
>-  0x42, 0x90, 0xa8, 0x52, 0x14, 0x00,
>-  0xa4, 0x29, 0x54, 0x85, 0x28, 0x00,
>-  0x13, 0x5a, 0x02, 0x6b, 0x40, 0x00,
>-  0x30, 0x93, 0x46, 0x12, 0x68, 0x00,
>-  0x88, 0xac, 0x31, 0x15, 0x84, 0x00,
>-  0x09, 0x0c, 0xc1, 0x21, 0x98, 0x00,
>-  0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
>-  0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
>-  0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
>-  0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
>-  0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
>-  0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
>-  0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
>-  0xa2, 0x28, 0x34, 0x45, 0x04, 0x00,
>-  0x51, 0x97, 0x2a, 0x32, 0xe4, 0x00,
>-  0x4c, 0x27, 0x09, 0x84, 0xe0, 0x00,
>-  0x66, 0x71, 0x8c, 0xce, 0x30, 0x00,
>-  0x91, 0x40, 0xf2, 0x28, 0x1c, 0x00,
>-  0x42, 0x90, 0xa8, 0x52, 0x14, 0x00,
>-  0xa4, 0x29, 0x54, 0x85, 0x28, 0x00,
>-  0x13, 0x5a, 0x02, 0x6b, 0x40, 0x00,
>-  0x30, 0x93, 0x46, 0x12, 0x68, 0x00,
>-  0x88, 0xac, 0x31, 0x15, 0x84, 0x00,
>-  0x09, 0x0c, 0xc1, 0x21, 0x98, 0x00,
>-  0x4c, 0x26, 0x09, 0x84, 0xc0, 0x00,
>-  0x66, 0x28, 0x8c, 0xc5, 0x10, 0x00,
>-  0x91, 0x50, 0x32, 0x2a, 0x04, 0x00,
>-  0x42, 0x82, 0x68, 0x50, 0x4c, 0x00,
>-  0xa4, 0x01, 0xd4, 0x80, 0x38, 0x00,
>-  0x13, 0x43, 0x02, 0x68, 0x60, 0x00,
>-  0x30, 0x94, 0x86, 0x12, 0x90, 0x00,
>-  0x88, 0xa1, 0x31, 0x14, 0x24, 0x00,
>-  0x09, 0x4c, 0x01, 0x29, 0x80, 0x00,
>-  0xcd, 0x98, 0x59, 0xb3, 0x08, 0x00
>-};
>-
>-const uint8_t kMaskRandom38_38[228] = {
>-  0x4c, 0x27, 0x09, 0x84, 0xe0, 0x00,
>-  0x66, 0x71, 0x8c, 0xce, 0x30, 0x00,
>-  0x91, 0x40, 0xf2, 0x28, 0x1c, 0x00,
>-  0x42, 0x90, 0xa8, 0x52, 0x14, 0x00,
>-  0xa4, 0x29, 0x54, 0x85, 0x28, 0x00,
>-  0x13, 0x5a, 0x02, 0x6b, 0x40, 0x00,
>-  0x30, 0x93, 0x46, 0x12, 0x68, 0x00,
>-  0x88, 0xac, 0x31, 0x15, 0x84, 0x00,
>-  0x09, 0x0c, 0xc1, 0x21, 0x98, 0x00,
>-  0x4c, 0x26, 0x09, 0x84, 0xc0, 0x00,
>-  0x66, 0x28, 0x8c, 0xc5, 0x10, 0x00,
>-  0x91, 0x50, 0x32, 0x2a, 0x04, 0x00,
>-  0x42, 0x82, 0x68, 0x50, 0x4c, 0x00,
>-  0xa4, 0x01, 0xd4, 0x80, 0x38, 0x00,
>-  0x13, 0x43, 0x02, 0x68, 0x60, 0x00,
>-  0x30, 0x94, 0x86, 0x12, 0x90, 0x00,
>-  0x88, 0xa1, 0x31, 0x14, 0x24, 0x00,
>-  0x09, 0x4c, 0x01, 0x29, 0x80, 0x00,
>-  0xcd, 0x98, 0x59, 0xb3, 0x08, 0x00,
>-  0x4c, 0x27, 0x09, 0x84, 0xe0, 0x00,
>-  0x66, 0x71, 0x8c, 0xce, 0x30, 0x00,
>-  0x91, 0x40, 0xf2, 0x28, 0x1c, 0x00,
>-  0x42, 0x90, 0xa8, 0x52, 0x14, 0x00,
>-  0xa4, 0x29, 0x54, 0x85, 0x28, 0x00,
>-  0x13, 0x5a, 0x02, 0x6b, 0x40, 0x00,
>-  0x30, 0x93, 0x46, 0x12, 0x68, 0x00,
>-  0x88, 0xac, 0x31, 0x15, 0x84, 0x00,
>-  0x09, 0x0c, 0xc1, 0x21, 0x98, 0x00,
>-  0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
>-  0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
>-  0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
>-  0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
>-  0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
>-  0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
>-  0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
>-  0xa2, 0x28, 0x34, 0x45, 0x04, 0x00,
>-  0x51, 0x97, 0x2a, 0x32, 0xe4, 0x00,
>-  0x8c, 0xed, 0x11, 0x5f, 0x24, 0x00
>-};
>-
>-const uint8_t kMaskRandom38_4[24] = {
>-  0xec, 0x73, 0x1d, 0x8e, 0x60, 0x00,
>-  0x67, 0x19, 0xcc, 0xe3, 0x38, 0x00,
>-  0xb1, 0xcc, 0x76, 0x39, 0x8c, 0x00,
>-  0x5a, 0x96, 0xab, 0x52, 0xd4, 0x00
>-};
>-
>-const uint8_t kMaskRandom38_5[30] = {
>-  0x4c, 0xe7, 0x09, 0x9c, 0xe0, 0x00,
>-  0x66, 0x31, 0xcc, 0xc6, 0x38, 0x00,
>-  0xa1, 0xcc, 0x74, 0x39, 0x8c, 0x00,
>-  0x92, 0xa6, 0xb2, 0x54, 0xd4, 0x00,
>-  0xb8, 0x99, 0x97, 0x13, 0x30, 0x00
>-};
>-
>-const uint8_t kMaskRandom38_6[36] = {
>-  0x4c, 0x36, 0x09, 0x86, 0xc0, 0x00,
>-  0x45, 0x68, 0x88, 0xad, 0x10, 0x00,
>-  0x30, 0xd0, 0x66, 0x1a, 0x0c, 0x00,
>-  0x8a, 0x82, 0xb1, 0x50, 0x54, 0x00,
>-  0x26, 0x0b, 0x44, 0xc1, 0x68, 0x00,
>-  0x95, 0x45, 0x12, 0xa8, 0xa0, 0x00
>-};
>-
>-const uint8_t kMaskRandom38_7[42] = {
>-  0xc4, 0xa3, 0x18, 0x94, 0x60, 0x00,
>-  0x23, 0x19, 0x84, 0x63, 0x30, 0x00,
>-  0x91, 0x1c, 0x32, 0x23, 0x84, 0x00,
>-  0x4a, 0x82, 0xa9, 0x50, 0x54, 0x00,
>-  0x34, 0x49, 0x46, 0x89, 0x28, 0x00,
>-  0x8b, 0x4a, 0x11, 0x69, 0x40, 0x00,
>-  0xc8, 0x24, 0xd9, 0x04, 0x98, 0x00
>-};
>-
>-const uint8_t kMaskRandom38_8[48] = {
>-  0x16, 0x13, 0x82, 0xc2, 0x70, 0x00,
>-  0xc2, 0x44, 0xd8, 0x48, 0x98, 0x00,
>-  0x60, 0xe8, 0x2c, 0x1d, 0x04, 0x00,
>-  0x85, 0x12, 0x70, 0xa2, 0x4c, 0x00,
>-  0xcc, 0x21, 0x59, 0x84, 0x28, 0x00,
>-  0x29, 0x63, 0x05, 0x2c, 0x60, 0x00,
>-  0x11, 0x98, 0xc2, 0x33, 0x18, 0x00,
>-  0xb0, 0x0c, 0x76, 0x01, 0x8c, 0x00
>-};
>-
>-const uint8_t kMaskRandom38_9[54] = {
>-  0x44, 0xa7, 0x08, 0x94, 0xe0, 0x00,
>-  0x66, 0x70, 0x8c, 0xce, 0x10, 0x00,
>-  0x12, 0xc0, 0xe2, 0x58, 0x1c, 0x00,
>-  0xc3, 0x10, 0xb8, 0x62, 0x14, 0x00,
>-  0x8c, 0x29, 0x51, 0x85, 0x28, 0x00,
>-  0x11, 0x5b, 0x02, 0x2b, 0x60, 0x00,
>-  0x21, 0x93, 0x44, 0x32, 0x68, 0x00,
>-  0xa2, 0x2c, 0x14, 0x45, 0x80, 0x00,
>-  0x18, 0x0c, 0xe3, 0x01, 0x9c, 0x00
>-};
>-
>-const uint8_t kMaskRandom39_1[6] = {
>-  0xff, 0xff, 0xff, 0xff, 0xfe, 0x00
>-};
>-
>-const uint8_t kMaskRandom39_10[60] = {
>-  0x8c, 0xe3, 0x09, 0x82, 0x60, 0x00,
>-  0x27, 0x11, 0xca, 0x22, 0x88, 0x00,
>-  0x32, 0x8d, 0x34, 0x0d, 0x02, 0x00,
>-  0x61, 0x92, 0x60, 0x98, 0x26, 0x00,
>-  0x5c, 0x38, 0x80, 0x70, 0x1c, 0x00,
>-  0xcc, 0x75, 0x10, 0xc4, 0x30, 0x00,
>-  0x2b, 0x19, 0xc5, 0x21, 0x48, 0x00,
>-  0x32, 0xd2, 0x68, 0x4a, 0x12, 0x00,
>-  0x25, 0x8e, 0xb3, 0x04, 0xc0, 0x00,
>-  0x50, 0x88, 0xc6, 0x11, 0x84, 0x00
>-};
>-
>-const uint8_t kMaskRandom39_11[66] = {
>-  0x8c, 0xe3, 0x18, 0xc6, 0x30, 0x00,
>-  0x27, 0x11, 0xc4, 0x71, 0x1c, 0x00,
>-  0x32, 0x8d, 0x23, 0x48, 0xd2, 0x00,
>-  0x61, 0x92, 0x64, 0x99, 0x26, 0x00,
>-  0x5c, 0x38, 0x8e, 0x23, 0x88, 0x00,
>-  0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
>-  0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
>-  0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
>-  0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
>-  0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
>-  0x89, 0x70, 0x1c, 0x07, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom39_12[72] = {
>-  0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
>-  0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
>-  0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
>-  0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
>-  0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
>-  0x89, 0x70, 0x1c, 0x07, 0x00, 0x00,
>-  0x8c, 0xe3, 0x18, 0xc6, 0x30, 0x00,
>-  0x27, 0x11, 0xc4, 0x71, 0x1c, 0x00,
>-  0x32, 0x8d, 0x23, 0x48, 0xd2, 0x00,
>-  0x61, 0x92, 0x64, 0x99, 0x26, 0x00,
>-  0x5c, 0x38, 0x8e, 0x23, 0x88, 0x00,
>-  0x90, 0xc8, 0x9e, 0xbb, 0x88, 0x00
>-};
>-
>-const uint8_t kMaskRandom39_13[78] = {
>-  0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
>-  0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
>-  0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
>-  0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
>-  0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
>-  0x89, 0x70, 0x1c, 0x07, 0x00, 0x00,
>-  0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
>-  0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
>-  0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
>-  0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
>-  0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
>-  0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
>-  0x34, 0x60, 0x98, 0x26, 0x08, 0x00
>-};
>-
>-const uint8_t kMaskRandom39_14[84] = {
>-  0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
>-  0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
>-  0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
>-  0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
>-  0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
>-  0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
>-  0x34, 0x60, 0x98, 0x26, 0x08, 0x00,
>-  0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
>-  0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
>-  0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
>-  0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
>-  0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
>-  0x89, 0x70, 0x1c, 0x07, 0x00, 0x00,
>-  0x6e, 0x27, 0x6a, 0xc7, 0xc4, 0x00
>-};
>-
>-const uint8_t kMaskRandom39_15[90] = {
>-  0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
>-  0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
>-  0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
>-  0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
>-  0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
>-  0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
>-  0x34, 0x60, 0x98, 0x26, 0x08, 0x00,
>-  0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
>-  0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
>-  0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
>-  0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
>-  0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
>-  0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
>-  0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
>-  0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00
>-};
>-
>-const uint8_t kMaskRandom39_16[96] = {
>-  0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
>-  0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
>-  0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
>-  0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
>-  0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
>-  0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
>-  0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
>-  0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00,
>-  0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
>-  0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
>-  0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
>-  0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
>-  0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
>-  0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
>-  0x34, 0x60, 0x98, 0x26, 0x08, 0x00,
>-  0x7e, 0x75, 0xe5, 0x03, 0x8c, 0x00
>-};
>-
>-const uint8_t kMaskRandom39_17[102] = {
>-  0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
>-  0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
>-  0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
>-  0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
>-  0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
>-  0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
>-  0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
>-  0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00,
>-  0x4c, 0x27, 0x09, 0xc2, 0x70, 0x00,
>-  0x66, 0x71, 0x9c, 0x67, 0x18, 0x00,
>-  0x91, 0x40, 0xf0, 0x3c, 0x0e, 0x00,
>-  0x42, 0x90, 0xa4, 0x29, 0x0a, 0x00,
>-  0xa4, 0x29, 0x4a, 0x52, 0x94, 0x00,
>-  0x13, 0x5a, 0x16, 0x85, 0xa0, 0x00,
>-  0x30, 0x93, 0x44, 0xd1, 0x34, 0x00,
>-  0x88, 0xac, 0x2b, 0x0a, 0xc2, 0x00,
>-  0x09, 0x0c, 0xc3, 0x30, 0xcc, 0x00
>-};
>-
>-const uint8_t kMaskRandom39_18[108] = {
>-  0x4c, 0x27, 0x09, 0xc2, 0x70, 0x00,
>-  0x66, 0x71, 0x9c, 0x67, 0x18, 0x00,
>-  0x91, 0x40, 0xf0, 0x3c, 0x0e, 0x00,
>-  0x42, 0x90, 0xa4, 0x29, 0x0a, 0x00,
>-  0xa4, 0x29, 0x4a, 0x52, 0x94, 0x00,
>-  0x13, 0x5a, 0x16, 0x85, 0xa0, 0x00,
>-  0x30, 0x93, 0x44, 0xd1, 0x34, 0x00,
>-  0x88, 0xac, 0x2b, 0x0a, 0xc2, 0x00,
>-  0x09, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
>-  0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
>-  0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
>-  0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
>-  0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
>-  0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
>-  0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
>-  0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
>-  0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00,
>-  0x51, 0x97, 0x24, 0x2f, 0x7e, 0x00
>-};
>-
>-const uint8_t kMaskRandom39_19[114] = {
>-  0x4c, 0x27, 0x09, 0xc2, 0x70, 0x00,
>-  0x66, 0x71, 0x9c, 0x67, 0x18, 0x00,
>-  0x91, 0x40, 0xf0, 0x3c, 0x0e, 0x00,
>-  0x42, 0x90, 0xa4, 0x29, 0x0a, 0x00,
>-  0xa4, 0x29, 0x4a, 0x52, 0x94, 0x00,
>-  0x13, 0x5a, 0x16, 0x85, 0xa0, 0x00,
>-  0x30, 0x93, 0x44, 0xd1, 0x34, 0x00,
>-  0x88, 0xac, 0x2b, 0x0a, 0xc2, 0x00,
>-  0x09, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
>-  0x4c, 0x26, 0x09, 0x82, 0x60, 0x00,
>-  0x66, 0x28, 0x8a, 0x22, 0x88, 0x00,
>-  0x91, 0x50, 0x34, 0x0d, 0x02, 0x00,
>-  0x42, 0x82, 0x60, 0x98, 0x26, 0x00,
>-  0xa4, 0x01, 0xc0, 0x70, 0x1c, 0x00,
>-  0x13, 0x43, 0x10, 0xc4, 0x30, 0x00,
>-  0x30, 0x94, 0x85, 0x21, 0x48, 0x00,
>-  0x88, 0xa1, 0x28, 0x4a, 0x12, 0x00,
>-  0x09, 0x4c, 0x13, 0x04, 0xc0, 0x00,
>-  0xcd, 0x98, 0x46, 0x11, 0x84, 0x00
>-};
>-
>-const uint8_t kMaskRandom39_2[12] = {
>-  0xce, 0x77, 0x1d, 0xc7, 0x70, 0x00,
>-  0x39, 0xcc, 0xf3, 0x3c, 0xce, 0x00
>-};
>-
>-const uint8_t kMaskRandom39_20[120] = {
>-  0x8c, 0xe3, 0x09, 0x82, 0x60, 0x00,
>-  0x27, 0x11, 0xca, 0x22, 0x88, 0x00,
>-  0x32, 0x8d, 0x34, 0x0d, 0x02, 0x00,
>-  0x61, 0x92, 0x60, 0x98, 0x26, 0x00,
>-  0x5c, 0x38, 0x80, 0x70, 0x1c, 0x00,
>-  0xcc, 0x75, 0x10, 0xc4, 0x30, 0x00,
>-  0x2b, 0x19, 0xc5, 0x21, 0x48, 0x00,
>-  0x32, 0xd2, 0x68, 0x4a, 0x12, 0x00,
>-  0x25, 0x8e, 0xb3, 0x04, 0xc0, 0x00,
>-  0x50, 0x88, 0xc6, 0x11, 0x84, 0x00,
>-  0x44, 0xa7, 0x09, 0xc2, 0x70, 0x00,
>-  0x66, 0x70, 0x8c, 0x47, 0x18, 0x00,
>-  0x12, 0xc0, 0xf0, 0x3c, 0x0e, 0x00,
>-  0xc3, 0x10, 0xbc, 0x29, 0x0a, 0x00,
>-  0x8c, 0x29, 0x42, 0x72, 0x94, 0x00,
>-  0x11, 0x5b, 0x16, 0x85, 0xa0, 0x00,
>-  0x21, 0x93, 0x44, 0xd1, 0x34, 0x00,
>-  0xa2, 0x2c, 0x0b, 0x0a, 0xc2, 0x00,
>-  0x18, 0x0c, 0xe9, 0x30, 0xca, 0x00,
>-  0x0d, 0xba, 0x52, 0x38, 0xbc, 0x00
>-};
>-
>-const uint8_t kMaskRandom39_21[126] = {
>-  0x8c, 0xe3, 0x09, 0x82, 0x60, 0x00,
>-  0x27, 0x11, 0xca, 0x22, 0x88, 0x00,
>-  0x32, 0x8d, 0x34, 0x0d, 0x02, 0x00,
>-  0x61, 0x92, 0x60, 0x98, 0x26, 0x00,
>-  0x5c, 0x38, 0x80, 0x70, 0x1c, 0x00,
>-  0xcc, 0x75, 0x10, 0xc4, 0x30, 0x00,
>-  0x2b, 0x19, 0xc5, 0x21, 0x48, 0x00,
>-  0x32, 0xd2, 0x68, 0x4a, 0x12, 0x00,
>-  0x25, 0x8e, 0xb3, 0x04, 0xc0, 0x00,
>-  0x50, 0x88, 0xc6, 0x11, 0x84, 0x00,
>-  0x8c, 0xe3, 0x18, 0xc6, 0x30, 0x00,
>-  0x27, 0x11, 0xc4, 0x71, 0x1c, 0x00,
>-  0x32, 0x8d, 0x23, 0x48, 0xd2, 0x00,
>-  0x61, 0x92, 0x64, 0x99, 0x26, 0x00,
>-  0x5c, 0x38, 0x8e, 0x23, 0x88, 0x00,
>-  0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
>-  0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
>-  0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
>-  0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
>-  0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
>-  0x89, 0x70, 0x1c, 0x07, 0x00, 0x00
>-};
>-
>-const uint8_t kMaskRandom39_22[132] = {
>-  0x8c, 0xe3, 0x18, 0xc6, 0x30, 0x00,
>-  0x27, 0x11, 0xc4, 0x71, 0x1c, 0x00,
>-  0x32, 0x8d, 0x23, 0x48, 0xd2, 0x00,
>-  0x61, 0x92, 0x64, 0x99, 0x26, 0x00,
>-  0x5c, 0x38, 0x8e, 0x23, 0x88, 0x00,
>-  0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
>-  0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
>-  0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
>-  0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
>-  0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
>-  0x89, 0x70, 0x1c, 0x07, 0x00, 0x00,
>-  0x8c, 0xe3, 0x09, 0x82, 0x60, 0x00,
>-  0x27, 0x11, 0xca, 0x22, 0x88, 0x00,
>-  0x32, 0x8d, 0x34, 0x0d, 0x02, 0x00,
>-  0x61, 0x92, 0x60, 0x98, 0x26, 0x00,
>-  0x5c, 0x38, 0x80, 0x70, 0x1c, 0x00,
>-  0xcc, 0x75, 0x10, 0xc4, 0x30, 0x00,
>-  0x2b, 0x19, 0xc5, 0x21, 0x48, 0x00,
>-  0x32, 0xd2, 0x68, 0x4a, 0x12, 0x00,
>-  0x25, 0x8e, 0xb3, 0x04, 0xc0, 0x00,
>-  0x50, 0x88, 0xc6, 0x11, 0x84, 0x00,
>-  0xfc, 0x5a, 0xb2, 0x13, 0x12, 0x00
>-};
>-
>-const uint8_t kMaskRandom39_23[138] = {
>-  0x8c, 0xe3, 0x18, 0xc6, 0x30, 0x00,
>-  0x27, 0x11, 0xc4, 0x71, 0x1c, 0x00,
>-  0x32, 0x8d, 0x23, 0x48, 0xd2, 0x00,
>-  0x61, 0x92, 0x64, 0x99, 0x26, 0x00,
>-  0x5c, 0x38, 0x8e, 0x23, 0x88, 0x00,
>-  0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
>-  0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
>-  0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
>-  0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
>-  0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
>-  0x89, 0x70, 0x1c, 0x07, 0x00, 0x00,
>-  0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
>-  0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
>-  0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
>-  0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
>-  0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
>-  0x89, 0x70, 0x1c, 0x07, 0x00, 0x00,
>-  0x8c, 0xe3, 0x18, 0xc6, 0x30, 0x00,
>-  0x27, 0x11, 0xc4, 0x71, 0x1c, 0x00,
>-  0x32, 0x8d, 0x23, 0x48, 0xd2, 0x00,
>-  0x61, 0x92, 0x64, 0x99, 0x26, 0x00,
>-  0x5c, 0x38, 0x8e, 0x23, 0x88, 0x00,
>-  0x90, 0xc8, 0x9e, 0xbb, 0x88, 0x00
>-};
>-
>-const uint8_t kMaskRandom39_24[144] = {
>-  0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
>-  0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
>-  0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
>-  0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
>-  0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
>-  0x89, 0x70, 0x1c, 0x07, 0x00, 0x00,
>-  0x8c, 0xe3, 0x18, 0xc6, 0x30, 0x00,
>-  0x27, 0x11, 0xc4, 0x71, 0x1c, 0x00,
>-  0x32, 0x8d, 0x23, 0x48, 0xd2, 0x00,
>-  0x61, 0x92, 0x64, 0x99, 0x26, 0x00,
>-  0x5c, 0x38, 0x8e, 0x23, 0x88, 0x00,
>-  0x90, 0xc8, 0x9e, 0xbb, 0x88, 0x00,
>-  0x8c, 0xe3, 0x18, 0xc6, 0x30, 0x00,
>-  0x27, 0x11, 0xc4, 0x71, 0x1c, 0x00,
>-  0x32, 0x8d, 0x23, 0x48, 0xd2, 0x00,
>-  0x61, 0x92, 0x64, 0x99, 0x26, 0x00,
>-  0x5c, 0x38, 0x8e, 0x23, 0x88, 0x00,
>-  0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
>-  0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
>-  0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
>-  0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
>-  0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
>-  0x89, 0x70, 0x1c, 0x07, 0x00, 0x00,
>-  0xac, 0xbc, 0xf0, 0xff, 0x62, 0x00
>-};
>-
>-const uint8_t kMaskRandom39_25[150] = {
>-  0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
>-  0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
>-  0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
>-  0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
>-  0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
>-  0x89, 0x70, 0x1c, 0x07, 0x00, 0x00,
>-  0x8c, 0xe3, 0x18, 0xc6, 0x30, 0x00,
>-  0x27, 0x11, 0xc4, 0x71, 0x1c, 0x00,
>-  0x32, 0x8d, 0x23, 0x48, 0xd2, 0x00,
>-  0x61, 0x92, 0x64, 0x99, 0x26, 0x00,
>-  0x5c, 0x38, 0x8e, 0x23, 0x88, 0x00,
>-  0x90, 0xc8, 0x9e, 0xbb, 0x88, 0x00,
>-  0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
>-  0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
>-  0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
>-  0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
>-  0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
>-  0x89, 0x70, 0x1c, 0x07, 0x00, 0x00,
>-  0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
>-  0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
>-  0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
>-  0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
>-  0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
>-  0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
>-  0x34, 0x60, 0x98, 0x26, 0x08, 0x00
>-};
>-
>-const uint8_t kMaskRandom39_26[156] = {
>-  0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
>-  0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
>-  0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
>-  0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
>-  0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
>-  0x89, 0x70, 0x1c, 0x07, 0x00, 0x00,
>-  0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
>-  0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
>-  0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
>-  0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
>-  0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
>-  0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
>-  0x34, 0x60, 0x98, 0x26, 0x08, 0x00,
>-  0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
>-  0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
>-  0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
>-  0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
>-  0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
>-  0x89, 0x70, 0x1c, 0x07, 0x00, 0x00,
>-  0x8c, 0xe3, 0x18, 0xc6, 0x30, 0x00,
>-  0x27, 0x11, 0xc4, 0x71, 0x1c, 0x00,
>-  0x32, 0x8d, 0x23, 0x48, 0xd2, 0x00,
>-  0x61, 0x92, 0x64, 0x99, 0x26, 0x00,
>-  0x5c, 0x38, 0x8e, 0x23, 0x88, 0x00,
>-  0x90, 0xc8, 0x9e, 0xbb, 0x88, 0x00,
>-  0x10, 0x17, 0x44, 0x72, 0xec, 0x00
>-};
>-
>-const uint8_t kMaskRandom39_27[162] = {
>-  0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
>-  0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
>-  0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
>-  0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
>-  0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
>-  0x89, 0x70, 0x1c, 0x07, 0x00, 0x00,
>-  0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
>-  0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
>-  0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
>-  0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
>-  0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
>-  0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
>-  0x34, 0x60, 0x98, 0x26, 0x08, 0x00,
>-  0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
>-  0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
>-  0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
>-  0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
>-  0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
>-  0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
>-  0x34, 0x60, 0x98, 0x26, 0x08, 0x00,
>-  0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
>-  0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
>-  0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
>-  0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
>-  0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
>-  0x89, 0x70, 0x1c, 0x07, 0x00, 0x00,
>-  0x6e, 0x27, 0x6a, 0xc7, 0xc4, 0x00
>-};
>-
>-const uint8_t kMaskRandom39_28[168] = {
>-  0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
>-  0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
>-  0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
>-  0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
>-  0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
>-  0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
>-  0x34, 0x60, 0x98, 0x26, 0x08, 0x00,
>-  0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
>-  0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
>-  0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
>-  0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
>-  0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
>-  0x89, 0x70, 0x1c, 0x07, 0x00, 0x00,
>-  0x6e, 0x27, 0x6a, 0xc7, 0xc4, 0x00,
>-  0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
>-  0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
>-  0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
>-  0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
>-  0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
>-  0x89, 0x70, 0x1c, 0x07, 0x00, 0x00,
>-  0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
>-  0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
>-  0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
>-  0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
>-  0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
>-  0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
>-  0x34, 0x60, 0x98, 0x26, 0x08, 0x00,
>-  0x86, 0xb6, 0x04, 0xbc, 0x1e, 0x00
>-};
>-
>-const uint8_t kMaskRandom39_29[174] = {
>-  0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
>-  0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
>-  0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
>-  0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
>-  0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
>-  0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
>-  0x34, 0x60, 0x98, 0x26, 0x08, 0x00,
>-  0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
>-  0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
>-  0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
>-  0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
>-  0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
>-  0x89, 0x70, 0x1c, 0x07, 0x00, 0x00,
>-  0x6e, 0x27, 0x6a, 0xc7, 0xc4, 0x00,
>-  0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
>-  0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
>-  0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
>-  0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
>-  0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
>-  0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
>-  0x34, 0x60, 0x98, 0x26, 0x08, 0x00,
>-  0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
>-  0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
>-  0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
>-  0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
>-  0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
>-  0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
>-  0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
>-  0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00
>-};
>-
>-const uint8_t kMaskRandom39_3[18] = {
>-  0xcc, 0x67, 0x19, 0xc6, 0x70, 0x00,
>-  0x27, 0x2c, 0xca, 0xb2, 0xac, 0x00,
>-  0x92, 0xd2, 0x76, 0x2d, 0x46, 0x00
>-};
>-
>-const uint8_t kMaskRandom39_30[180] = {
>-  0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
>-  0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
>-  0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
>-  0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
>-  0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
>-  0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
>-  0x34, 0x60, 0x98, 0x26, 0x08, 0x00,
>-  0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
>-  0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
>-  0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
>-  0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
>-  0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
>-  0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
>-  0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
>-  0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00,
>-  0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
>-  0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
>-  0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
>-  0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
>-  0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
>-  0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
>-  0x34, 0x60, 0x98, 0x26, 0x08, 0x00,
>-  0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
>-  0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
>-  0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
>-  0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
>-  0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
>-  0x89, 0x70, 0x1c, 0x07, 0x00, 0x00,
>-  0x6e, 0x27, 0x6a, 0xc7, 0xc4, 0x00,
>-  0xb3, 0x1d, 0x13, 0x03, 0x5a, 0x00
>-};
>-
>-const uint8_t kMaskRandom39_31[186] = {
>-  0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
>-  0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
>-  0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
>-  0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
>-  0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
>-  0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
>-  0x34, 0x60, 0x98, 0x26, 0x08, 0x00,
>-  0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
>-  0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
>-  0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
>-  0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
>-  0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
>-  0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
>-  0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
>-  0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00,
>-  0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
>-  0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
>-  0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
>-  0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
>-  0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
>-  0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
>-  0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
>-  0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00,
>-  0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
>-  0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
>-  0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
>-  0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
>-  0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
>-  0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
>-  0x34, 0x60, 0x98, 0x26, 0x08, 0x00,
>-  0x7e, 0x75, 0xe5, 0x03, 0x8c, 0x00
>-};
>-
>-const uint8_t kMaskRandom39_32[192] = {
>-  0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
>-  0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
>-  0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
>-  0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
>-  0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
>-  0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
>-  0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
>-  0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00,
>-  0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
>-  0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
>-  0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
>-  0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
>-  0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
>-  0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
>-  0x34, 0x60, 0x98, 0x26, 0x08, 0x00,
>-  0x7e, 0x75, 0xe5, 0x03, 0x8c, 0x00,
>-  0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
>-  0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
>-  0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
>-  0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
>-  0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
>-  0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
>-  0x34, 0x60, 0x98, 0x26, 0x08, 0x00,
>-  0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
>-  0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
>-  0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
>-  0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
>-  0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
>-  0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
>-  0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
>-  0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00,
>-  0x83, 0x1a, 0x3c, 0x2a, 0x7a, 0x00
>-};
>-
>-const uint8_t kMaskRandom39_33[198] = {
>-  0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
>-  0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
>-  0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
>-  0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
>-  0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
>-  0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
>-  0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
>-  0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00,
>-  0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
>-  0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
>-  0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
>-  0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
>-  0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
>-  0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
>-  0x34, 0x60, 0x98, 0x26, 0x08, 0x00,
>-  0x7e, 0x75, 0xe5, 0x03, 0x8c, 0x00,
>-  0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
>-  0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
>-  0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
>-  0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
>-  0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
>-  0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
>-  0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
>-  0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00,
>-  0x4c, 0x27, 0x09, 0xc2, 0x70, 0x00,
>-  0x66, 0x71, 0x9c, 0x67, 0x18, 0x00,
>-  0x91, 0x40, 0xf0, 0x3c, 0x0e, 0x00,
>-  0x42, 0x90, 0xa4, 0x29, 0x0a, 0x00,
>-  0xa4, 0x29, 0x4a, 0x52, 0x94, 0x00,
>-  0x13, 0x5a, 0x16, 0x85, 0xa0, 0x00,
>-  0x30, 0x93, 0x44, 0xd1, 0x34, 0x00,
>-  0x88, 0xac, 0x2b, 0x0a, 0xc2, 0x00,
>-  0x09, 0x0c, 0xc3, 0x30, 0xcc, 0x00
>-};
>-
>-const uint8_t kMaskRandom39_34[204] = {
>-  0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
>-  0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
>-  0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
>-  0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
>-  0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
>-  0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
>-  0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
>-  0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00,
>-  0x4c, 0x27, 0x09, 0xc2, 0x70, 0x00,
>-  0x66, 0x71, 0x9c, 0x67, 0x18, 0x00,
>-  0x91, 0x40, 0xf0, 0x3c, 0x0e, 0x00,
>-  0x42, 0x90, 0xa4, 0x29, 0x0a, 0x00,
>-  0xa4, 0x29, 0x4a, 0x52, 0x94, 0x00,
>-  0x13, 0x5a, 0x16, 0x85, 0xa0, 0x00,
>-  0x30, 0x93, 0x44, 0xd1, 0x34, 0x00,
>-  0x88, 0xac, 0x2b, 0x0a, 0xc2, 0x00,
>-  0x09, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
>-  0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
>-  0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
>-  0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
>-  0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
>-  0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
>-  0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
>-  0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
>-  0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00,
>-  0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
>-  0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
>-  0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
>-  0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
>-  0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
>-  0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
>-  0x34, 0x60, 0x98, 0x26, 0x08, 0x00,
>-  0x7e, 0x75, 0xe5, 0x03, 0x8c, 0x00,
>-  0xc6, 0xbb, 0x7e, 0xd9, 0x80, 0x00
>-};
>-
>-const uint8_t kMaskRandom39_35[210] = {
>-  0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
>-  0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
>-  0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
>-  0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
>-  0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
>-  0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
>-  0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
>-  0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00,
>-  0x4c, 0x27, 0x09, 0xc2, 0x70, 0x00,
>-  0x66, 0x71, 0x9c, 0x67, 0x18, 0x00,
>-  0x91, 0x40, 0xf0, 0x3c, 0x0e, 0x00,
>-  0x42, 0x90, 0xa4, 0x29, 0x0a, 0x00,
>-  0xa4, 0x29, 0x4a, 0x52, 0x94, 0x00,
>-  0x13, 0x5a, 0x16, 0x85, 0xa0, 0x00,
>-  0x30, 0x93, 0x44, 0xd1, 0x34, 0x00,
>-  0x88, 0xac, 0x2b, 0x0a, 0xc2, 0x00,
>-  0x09, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
>-  0x4c, 0x27, 0x09, 0xc2, 0x70, 0x00,
>-  0x66, 0x71, 0x9c, 0x67, 0x18, 0x00,
>-  0x91, 0x40, 0xf0, 0x3c, 0x0e, 0x00,
>-  0x42, 0x90, 0xa4, 0x29, 0x0a, 0x00,
>-  0xa4, 0x29, 0x4a, 0x52, 0x94, 0x00,
>-  0x13, 0x5a, 0x16, 0x85, 0xa0, 0x00,
>-  0x30, 0x93, 0x44, 0xd1, 0x34, 0x00,
>-  0x88, 0xac, 0x2b, 0x0a, 0xc2, 0x00,
>-  0x09, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
>-  0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
>-  0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
>-  0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
>-  0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
>-  0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
>-  0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
>-  0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
>-  0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00,
>-  0x51, 0x97, 0x24, 0x2f, 0x7e, 0x00
>-};
>-
>-const uint8_t kMaskRandom39_36[216] = {
>-  0x4c, 0x27, 0x09, 0xc2, 0x70, 0x00,
>-  0x66, 0x71, 0x9c, 0x67, 0x18, 0x00,
>-  0x91, 0x40, 0xf0, 0x3c, 0x0e, 0x00,
>-  0x42, 0x90, 0xa4, 0x29, 0x0a, 0x00,
>-  0xa4, 0x29, 0x4a, 0x52, 0x94, 0x00,
>-  0x13, 0x5a, 0x16, 0x85, 0xa0, 0x00,
>-  0x30, 0x93, 0x44, 0xd1, 0x34, 0x00,
>-  0x88, 0xac, 0x2b, 0x0a, 0xc2, 0x00,
>-  0x09, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
>-  0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
>-  0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
>-  0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
>-  0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
>-  0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
>-  0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
>-  0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
>-  0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00,
>-  0x51, 0x97, 0x24, 0x2f, 0x7e, 0x00,
>-  0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
>-  0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
>-  0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
>-  0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
>-  0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
>-  0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
>-  0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
>-  0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00,
>-  0x4c, 0x27, 0x09, 0xc2, 0x70, 0x00,
>-  0x66, 0x71, 0x9c, 0x67, 0x18, 0x00,
>-  0x91, 0x40, 0xf0, 0x3c, 0x0e, 0x00,
>-  0x42, 0x90, 0xa4, 0x29, 0x0a, 0x00,
>-  0xa4, 0x29, 0x4a, 0x52, 0x94, 0x00,
>-  0x13, 0x5a, 0x16, 0x85, 0xa0, 0x00,
>-  0x30, 0x93, 0x44, 0xd1, 0x34, 0x00,
>-  0x88, 0xac, 0x2b, 0x0a, 0xc2, 0x00,
>-  0x09, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
>-  0x11, 0x78, 0xfe, 0x43, 0xd6, 0x00
>-};
>-
>-const uint8_t kMaskRandom39_37[222] = {
>-  0x4c, 0x27, 0x09, 0xc2, 0x70, 0x00,
>-  0x66, 0x71, 0x9c, 0x67, 0x18, 0x00,
>-  0x91, 0x40, 0xf0, 0x3c, 0x0e, 0x00,
>-  0x42, 0x90, 0xa4, 0x29, 0x0a, 0x00,
>-  0xa4, 0x29, 0x4a, 0x52, 0x94, 0x00,
>-  0x13, 0x5a, 0x16, 0x85, 0xa0, 0x00,
>-  0x30, 0x93, 0x44, 0xd1, 0x34, 0x00,
>-  0x88, 0xac, 0x2b, 0x0a, 0xc2, 0x00,
>-  0x09, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
>-  0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
>-  0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
>-  0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
>-  0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
>-  0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
>-  0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
>-  0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
>-  0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00,
>-  0x51, 0x97, 0x24, 0x2f, 0x7e, 0x00,
>-  0x4c, 0x27, 0x09, 0xc2, 0x70, 0x00,
>-  0x66, 0x71, 0x9c, 0x67, 0x18, 0x00,
>-  0x91, 0x40, 0xf0, 0x3c, 0x0e, 0x00,
>-  0x42, 0x90, 0xa4, 0x29, 0x0a, 0x00,
>-  0xa4, 0x29, 0x4a, 0x52, 0x94, 0x00,
>-  0x13, 0x5a, 0x16, 0x85, 0xa0, 0x00,
>-  0x30, 0x93, 0x44, 0xd1, 0x34, 0x00,
>-  0x88, 0xac, 0x2b, 0x0a, 0xc2, 0x00,
>-  0x09, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
>-  0x4c, 0x26, 0x09, 0x82, 0x60, 0x00,
>-  0x66, 0x28, 0x8a, 0x22, 0x88, 0x00,
>-  0x91, 0x50, 0x34, 0x0d, 0x02, 0x00,
>-  0x42, 0x82, 0x60, 0x98, 0x26, 0x00,
>-  0xa4, 0x01, 0xc0, 0x70, 0x1c, 0x00,
>-  0x13, 0x43, 0x10, 0xc4, 0x30, 0x00,
>-  0x30, 0x94, 0x85, 0x21, 0x48, 0x00,
>-  0x88, 0xa1, 0x28, 0x4a, 0x12, 0x00,
>-  0x09, 0x4c, 0x13, 0x04, 0xc0, 0x00,
>-  0xcd, 0x98, 0x46, 0x11, 0x84, 0x00
>-};
>-
>-const uint8_t kMaskRandom39_38[228] = {
>-  0x4c, 0x27, 0x09, 0xc2, 0x70, 0x00,
>-  0x66, 0x71, 0x9c, 0x67, 0x18, 0x00,
>-  0x91, 0x40, 0xf0, 0x3c, 0x0e, 0x00,
>-  0x42, 0x90, 0xa4, 0x29, 0x0a, 0x00,
>-  0xa4, 0x29, 0x4a, 0x52, 0x94, 0x00,
>-  0x13, 0x5a, 0x16, 0x85, 0xa0, 0x00,
>-  0x30, 0x93, 0x44, 0xd1, 0x34, 0x00,
>-  0x88, 0xac, 0x2b, 0x0a, 0xc2, 0x00,
>-  0x09, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
>-  0x4c, 0x26, 0x09, 0x82, 0x60, 0x00,
>-  0x66, 0x28, 0x8a, 0x22, 0x88, 0x00,
>-  0x91, 0x50, 0x34, 0x0d, 0x02, 0x00,
>-  0x42, 0x82, 0x60, 0x98, 0x26, 0x00,
>-  0xa4, 0x01, 0xc0, 0x70, 0x1c, 0x00,
>-  0x13, 0x43, 0x10, 0xc4, 0x30, 0x00,
>-  0x30, 0x94, 0x85, 0x21, 0x48, 0x00,
>-  0x88, 0xa1, 0x28, 0x4a, 0x12, 0x00,
>-  0x09, 0x4c, 0x13, 0x04, 0xc0, 0x00,
>-  0xcd, 0x98, 0x46, 0x11, 0x84, 0x00,
>-  0x4c, 0x27, 0x09, 0xc2, 0x70, 0x00,
>-  0x66, 0x71, 0x9c, 0x67, 0x18, 0x00,
>-  0x91, 0x40, 0xf0, 0x3c, 0x0e, 0x00,
>-  0x42, 0x90, 0xa4, 0x29, 0x0a, 0x00,
>-  0xa4, 0x29, 0x4a, 0x52, 0x94, 0x00,
>-  0x13, 0x5a, 0x16, 0x85, 0xa0, 0x00,
>-  0x30, 0x93, 0x44, 0xd1, 0x34, 0x00,
>-  0x88, 0xac, 0x2b, 0x0a, 0xc2, 0x00,
>-  0x09, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
>-  0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
>-  0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
>-  0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
>-  0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
>-  0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
>-  0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
>-  0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
>-  0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00,
>-  0x51, 0x97, 0x24, 0x2f, 0x7e, 0x00,
>-  0x9e, 0xd8, 0x3c, 0x7e, 0x2e, 0x00
>-};
>-
>-const uint8_t kMaskRandom39_39[234] = {
>-  0x4c, 0x27, 0x09, 0xc2, 0x70, 0x00,
>-  0x66, 0x71, 0x9c, 0x67, 0x18, 0x00,
>-  0x91, 0x40, 0xf0, 0x3c, 0x0e, 0x00,
>-  0x42, 0x90, 0xa4, 0x29, 0x0a, 0x00,
>-  0xa4, 0x29, 0x4a, 0x52, 0x94, 0x00,
>-  0x13, 0x5a, 0x16, 0x85, 0xa0, 0x00,
>-  0x30, 0x93, 0x44, 0xd1, 0x34, 0x00,
>-  0x88, 0xac, 0x2b, 0x0a, 0xc2, 0x00,
>-  0x09, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
>-  0x4c, 0x26, 0x09, 0x82, 0x60, 0x00,
>-  0x66, 0x28, 0x8a, 0x22, 0x88, 0x00,
>-  0x91, 0x50, 0x34, 0x0d, 0x02, 0x00,
>-  0x42, 0x82, 0x60, 0x98, 0x26, 0x00,
>-  0xa4, 0x01, 0xc0, 0x70, 0x1c, 0x00,
>-  0x13, 0x43, 0x10, 0xc4, 0x30, 0x00,
>-  0x30, 0x94, 0x85, 0x21, 0x48, 0x00,
>-  0x88, 0xa1, 0x28, 0x4a, 0x12, 0x00,
>-  0x09, 0x4c, 0x13, 0x04, 0xc0, 0x00,
>-  0xcd, 0x98, 0x46, 0x11, 0x84, 0x00,
>-  0x4c, 0x27, 0x09, 0x82, 0x60, 0x00,
>-  0x66, 0x71, 0x8a, 0x22, 0x88, 0x00,
>-  0x91, 0x40, 0xf4, 0x0d, 0x02, 0x00,
>-  0x42, 0x90, 0xa0, 0x98, 0x26, 0x00,
>-  0xa4, 0x29, 0x40, 0x70, 0x1c, 0x00,
>-  0x13, 0x5a, 0x10, 0xc4, 0x30, 0x00,
>-  0x30, 0x93, 0x45, 0x21, 0x48, 0x00,
>-  0x88, 0xac, 0x28, 0x4a, 0x12, 0x00,
>-  0x09, 0x0c, 0xd3, 0x04, 0xc0, 0x00,
>-  0x4c, 0x26, 0x06, 0x11, 0x84, 0x00,
>-  0x66, 0x28, 0x89, 0xc2, 0x70, 0x00,
>-  0x91, 0x50, 0x3c, 0x67, 0x18, 0x00,
>-  0x42, 0x82, 0x70, 0x3c, 0x0e, 0x00,
>-  0xa4, 0x01, 0xc4, 0x29, 0x0a, 0x00,
>-  0x13, 0x43, 0x0a, 0x52, 0x94, 0x00,
>-  0x30, 0x94, 0x96, 0x85, 0xa0, 0x00,
>-  0x88, 0xa1, 0x24, 0xd1, 0x34, 0x00,
>-  0x09, 0x4c, 0x0b, 0x0a, 0xc2, 0x00,
>-  0xcd, 0x98, 0x43, 0x30, 0xcc, 0x00,
>-  0x1d, 0x04, 0x3e, 0xf1, 0xb4, 0x00
>-};
>-
>-const uint8_t kMaskRandom39_4[24] = {
>-  0xec, 0x73, 0x1c, 0xc7, 0x30, 0x00,
>-  0x67, 0x19, 0xc6, 0x71, 0x9c, 0x00,
>-  0xb1, 0xcc, 0x73, 0x1c, 0xc6, 0x00,
>-  0x5a, 0x96, 0xa5, 0xa9, 0x6a, 0x00
>-};
>-
>-const uint8_t kMaskRandom39_5[30] = {
>-  0x4c, 0xe7, 0x19, 0xc6, 0x70, 0x00,
>-  0x66, 0x31, 0xcc, 0x73, 0x1c, 0x00,
>-  0xa1, 0xcc, 0x73, 0x1c, 0xa6, 0x00,
>-  0x92, 0xa6, 0xa5, 0x6a, 0x6a, 0x00,
>-  0xb8, 0x99, 0x96, 0x8b, 0x94, 0x00
>-};
>-
>-const uint8_t kMaskRandom39_6[36] = {
>-  0x4c, 0x36, 0x09, 0x83, 0x60, 0x00,
>-  0x45, 0x68, 0x8a, 0x26, 0x88, 0x00,
>-  0x30, 0xd0, 0x64, 0x1d, 0x06, 0x00,
>-  0x8a, 0x82, 0xb0, 0xa8, 0x2a, 0x00,
>-  0x26, 0x0b, 0x40, 0xd0, 0xd4, 0x00,
>-  0x95, 0x45, 0x13, 0x44, 0x30, 0x00
>-};
>-
>-const uint8_t kMaskRandom39_7[42] = {
>-  0xc4, 0xa3, 0x09, 0xc2, 0x30, 0x00,
>-  0x23, 0x19, 0x86, 0x65, 0x80, 0x00,
>-  0x91, 0x1c, 0x22, 0x01, 0xd6, 0x00,
>-  0x4a, 0x82, 0xb0, 0x2a, 0x2a, 0x00,
>-  0x34, 0x49, 0x44, 0x98, 0x94, 0x00,
>-  0x8b, 0x4a, 0x1a, 0x84, 0x60, 0x00,
>-  0xc8, 0x24, 0xc1, 0x94, 0x4c, 0x00
>-};
>-
>-const uint8_t kMaskRandom39_8[48] = {
>-  0x16, 0x13, 0x84, 0xe1, 0x38, 0x00,
>-  0xc2, 0x44, 0xd1, 0x34, 0x4c, 0x00,
>-  0x60, 0xe8, 0x3a, 0x0e, 0x82, 0x00,
>-  0x85, 0x12, 0x64, 0x99, 0x26, 0x00,
>-  0xcc, 0x21, 0x5c, 0x52, 0x14, 0x00,
>-  0x29, 0x63, 0x18, 0xc6, 0x30, 0x00,
>-  0x11, 0x98, 0xc6, 0x31, 0x8c, 0x00,
>-  0xb0, 0x0c, 0x63, 0x18, 0xc6, 0x00
>-};
>-
>-const uint8_t kMaskRandom39_9[54] = {
>-  0x44, 0xa7, 0x09, 0xc2, 0x70, 0x00,
>-  0x66, 0x70, 0x8c, 0x47, 0x18, 0x00,
>-  0x12, 0xc0, 0xf0, 0x3c, 0x0e, 0x00,
>-  0xc3, 0x10, 0xbc, 0x29, 0x0a, 0x00,
>-  0x8c, 0x29, 0x42, 0x72, 0x94, 0x00,
>-  0x11, 0x5b, 0x16, 0x85, 0xa0, 0x00,
>-  0x21, 0x93, 0x44, 0xd1, 0x34, 0x00,
>-  0xa2, 0x2c, 0x0b, 0x0a, 0xc2, 0x00,
>-  0x18, 0x0c, 0xe9, 0x30, 0xca, 0x00
>-};
>-
>-const uint8_t kMaskRandom3_1[2] = {
>-  0xe0, 0x00
>-};
>-
>-const uint8_t kMaskRandom3_2[4] = {
>-  0xc0, 0x00,
>-  0xa0, 0x00
>-};
>-
>-const uint8_t kMaskRandom3_3[6] = {
>-  0xc0, 0x00,
>-  0xa0, 0x00,
>-  0x60, 0x00
>-};
>-
>-const uint8_t kMaskRandom40_1[6] = {
>-  0xff, 0xff, 0xff, 0xff, 0xff, 0x00
>-};
>-
>-const uint8_t kMaskRandom40_10[60] = {
>-  0x4c, 0x13, 0x04, 0xc1, 0x30, 0x00,
>-  0x51, 0x14, 0x45, 0x11, 0x44, 0x00,
>-  0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
>-  0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
>-  0x03, 0x80, 0xe0, 0x38, 0x0e, 0x00,
>-  0x86, 0x21, 0x88, 0x62, 0x18, 0x00,
>-  0x29, 0x0a, 0x42, 0x90, 0xa4, 0x00,
>-  0x42, 0x50, 0x94, 0x25, 0x09, 0x00,
>-  0x98, 0x26, 0x09, 0x82, 0x60, 0x00,
>-  0x30, 0x8c, 0x23, 0x08, 0xc2, 0x00
>-};
>-
>-const uint8_t kMaskRandom40_11[66] = {
>-  0xc6, 0x31, 0x8c, 0x63, 0x18, 0x00,
>-  0x23, 0x88, 0xe2, 0x38, 0x8e, 0x00,
>-  0x1a, 0x46, 0x91, 0xa4, 0x69, 0x00,
>-  0x24, 0xc9, 0x32, 0x4c, 0x93, 0x00,
>-  0x71, 0x1c, 0x47, 0x11, 0xc4, 0x00,
>-  0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
>-  0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
>-  0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
>-  0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00
>-};
>-
>-const uint8_t kMaskRandom40_12[72] = {
>-  0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
>-  0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
>-  0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
>-  0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00,
>-  0xc6, 0x31, 0x8c, 0x63, 0x18, 0x00,
>-  0x23, 0x88, 0xe2, 0x38, 0x8e, 0x00,
>-  0x1a, 0x46, 0x91, 0xa4, 0x69, 0x00,
>-  0x24, 0xc9, 0x32, 0x4c, 0x93, 0x00,
>-  0x71, 0x1c, 0x47, 0x11, 0xc4, 0x00,
>-  0xf5, 0xdc, 0x4f, 0x5d, 0xc4, 0x00
>-};
>-
>-const uint8_t kMaskRandom40_13[78] = {
>-  0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
>-  0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
>-  0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
>-  0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00,
>-  0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
>-  0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
>-  0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
>-  0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
>-  0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00
>-};
>-
>-const uint8_t kMaskRandom40_14[84] = {
>-  0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
>-  0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
>-  0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
>-  0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
>-  0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00,
>-  0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
>-  0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
>-  0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
>-  0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00,
>-  0x56, 0x3e, 0x25, 0x63, 0xe2, 0x00
>-};
>-
>-const uint8_t kMaskRandom40_15[90] = {
>-  0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
>-  0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
>-  0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
>-  0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
>-  0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00,
>-  0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
>-  0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
>-  0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
>-  0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
>-  0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
>-  0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
>-  0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
>-  0x50, 0x54, 0x15, 0x05, 0x41, 0x00
>-};
>-
>-const uint8_t kMaskRandom40_16[96] = {
>-  0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
>-  0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
>-  0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
>-  0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
>-  0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
>-  0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
>-  0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
>-  0x50, 0x54, 0x15, 0x05, 0x41, 0x00,
>-  0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
>-  0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
>-  0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
>-  0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
>-  0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00,
>-  0x28, 0x1c, 0x62, 0x81, 0xc6, 0x00
>-};
>-
>-const uint8_t kMaskRandom40_17[102] = {
>-  0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
>-  0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
>-  0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
>-  0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
>-  0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
>-  0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
>-  0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
>-  0x50, 0x54, 0x15, 0x05, 0x41, 0x00,
>-  0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
>-  0xe3, 0x38, 0xce, 0x33, 0x8c, 0x00,
>-  0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
>-  0x21, 0x48, 0x52, 0x14, 0x85, 0x00,
>-  0x52, 0x94, 0xa5, 0x29, 0x4a, 0x00,
>-  0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
>-  0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
>-  0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
>-  0x19, 0x86, 0x61, 0x98, 0x66, 0x00
>-};
>-
>-const uint8_t kMaskRandom40_18[108] = {
>-  0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
>-  0xe3, 0x38, 0xce, 0x33, 0x8c, 0x00,
>-  0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
>-  0x21, 0x48, 0x52, 0x14, 0x85, 0x00,
>-  0x52, 0x94, 0xa5, 0x29, 0x4a, 0x00,
>-  0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
>-  0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
>-  0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
>-  0x19, 0x86, 0x61, 0x98, 0x66, 0x00,
>-  0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
>-  0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
>-  0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
>-  0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
>-  0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
>-  0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
>-  0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
>-  0x50, 0x54, 0x15, 0x05, 0x41, 0x00,
>-  0x21, 0x7b, 0xf2, 0x17, 0xbf, 0x00
>-};
>-
>-const uint8_t kMaskRandom40_19[114] = {
>-  0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
>-  0xe3, 0x38, 0xce, 0x33, 0x8c, 0x00,
>-  0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
>-  0x21, 0x48, 0x52, 0x14, 0x85, 0x00,
>-  0x52, 0x94, 0xa5, 0x29, 0x4a, 0x00,
>-  0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
>-  0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
>-  0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
>-  0x19, 0x86, 0x61, 0x98, 0x66, 0x00,
>-  0x4c, 0x13, 0x04, 0xc1, 0x30, 0x00,
>-  0x51, 0x14, 0x45, 0x11, 0x44, 0x00,
>-  0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
>-  0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
>-  0x03, 0x80, 0xe0, 0x38, 0x0e, 0x00,
>-  0x86, 0x21, 0x88, 0x62, 0x18, 0x00,
>-  0x29, 0x0a, 0x42, 0x90, 0xa4, 0x00,
>-  0x42, 0x50, 0x94, 0x25, 0x09, 0x00,
>-  0x98, 0x26, 0x09, 0x82, 0x60, 0x00,
>-  0x30, 0x8c, 0x23, 0x08, 0xc2, 0x00
>-};
>-
>-const uint8_t kMaskRandom40_2[12] = {
>-  0xee, 0x3b, 0x8e, 0xe3, 0xb8, 0x00,
>-  0x99, 0xe6, 0x79, 0x9e, 0x67, 0x00
>-};
>-
>-const uint8_t kMaskRandom40_20[120] = {
>-  0x4c, 0x13, 0x04, 0xc1, 0x30, 0x00,
>-  0x51, 0x14, 0x45, 0x11, 0x44, 0x00,
>-  0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
>-  0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
>-  0x03, 0x80, 0xe0, 0x38, 0x0e, 0x00,
>-  0x86, 0x21, 0x88, 0x62, 0x18, 0x00,
>-  0x29, 0x0a, 0x42, 0x90, 0xa4, 0x00,
>-  0x42, 0x50, 0x94, 0x25, 0x09, 0x00,
>-  0x98, 0x26, 0x09, 0x82, 0x60, 0x00,
>-  0x30, 0x8c, 0x23, 0x08, 0xc2, 0x00,
>-  0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
>-  0xe3, 0x38, 0xce, 0x33, 0x8c, 0x00,
>-  0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
>-  0x21, 0x48, 0x52, 0x14, 0x85, 0x00,
>-  0x52, 0x94, 0xa5, 0x29, 0x4a, 0x00,
>-  0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
>-  0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
>-  0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
>-  0x19, 0x86, 0x61, 0x98, 0x66, 0x00,
>-  0xf7, 0x8d, 0xaf, 0x78, 0xda, 0x00
>-};
>-
>-const uint8_t kMaskRandom40_21[126] = {
>-  0x4c, 0x13, 0x04, 0xc1, 0x30, 0x00,
>-  0x51, 0x14, 0x45, 0x11, 0x44, 0x00,
>-  0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
>-  0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
>-  0x03, 0x80, 0xe0, 0x38, 0x0e, 0x00,
>-  0x86, 0x21, 0x88, 0x62, 0x18, 0x00,
>-  0x29, 0x0a, 0x42, 0x90, 0xa4, 0x00,
>-  0x42, 0x50, 0x94, 0x25, 0x09, 0x00,
>-  0x98, 0x26, 0x09, 0x82, 0x60, 0x00,
>-  0x30, 0x8c, 0x23, 0x08, 0xc2, 0x00,
>-  0xc6, 0x31, 0x8c, 0x63, 0x18, 0x00,
>-  0x23, 0x88, 0xe2, 0x38, 0x8e, 0x00,
>-  0x1a, 0x46, 0x91, 0xa4, 0x69, 0x00,
>-  0x24, 0xc9, 0x32, 0x4c, 0x93, 0x00,
>-  0x71, 0x1c, 0x47, 0x11, 0xc4, 0x00,
>-  0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
>-  0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
>-  0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
>-  0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00
>-};
>-
>-const uint8_t kMaskRandom40_22[132] = {
>-  0xc6, 0x31, 0x8c, 0x63, 0x18, 0x00,
>-  0x23, 0x88, 0xe2, 0x38, 0x8e, 0x00,
>-  0x1a, 0x46, 0x91, 0xa4, 0x69, 0x00,
>-  0x24, 0xc9, 0x32, 0x4c, 0x93, 0x00,
>-  0x71, 0x1c, 0x47, 0x11, 0xc4, 0x00,
>-  0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
>-  0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
>-  0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
>-  0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00,
>-  0x4c, 0x13, 0x04, 0xc1, 0x30, 0x00,
>-  0x51, 0x14, 0x45, 0x11, 0x44, 0x00,
>-  0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
>-  0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
>-  0x03, 0x80, 0xe0, 0x38, 0x0e, 0x00,
>-  0x86, 0x21, 0x88, 0x62, 0x18, 0x00,
>-  0x29, 0x0a, 0x42, 0x90, 0xa4, 0x00,
>-  0x42, 0x50, 0x94, 0x25, 0x09, 0x00,
>-  0x98, 0x26, 0x09, 0x82, 0x60, 0x00,
>-  0x30, 0x8c, 0x23, 0x08, 0xc2, 0x00,
>-  0x89, 0xee, 0x1f, 0x38, 0xca, 0x00
>-};
>-
>-const uint8_t kMaskRandom40_23[138] = {
>-  0xc6, 0x31, 0x8c, 0x63, 0x18, 0x00,
>-  0x23, 0x88, 0xe2, 0x38, 0x8e, 0x00,
>-  0x1a, 0x46, 0x91, 0xa4, 0x69, 0x00,
>-  0x24, 0xc9, 0x32, 0x4c, 0x93, 0x00,
>-  0x71, 0x1c, 0x47, 0x11, 0xc4, 0x00,
>-  0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
>-  0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
>-  0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
>-  0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00,
>-  0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
>-  0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
>-  0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
>-  0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00,
>-  0xc6, 0x31, 0x8c, 0x63, 0x18, 0x00,
>-  0x23, 0x88, 0xe2, 0x38, 0x8e, 0x00,
>-  0x1a, 0x46, 0x91, 0xa4, 0x69, 0x00,
>-  0x24, 0xc9, 0x32, 0x4c, 0x93, 0x00,
>-  0x71, 0x1c, 0x47, 0x11, 0xc4, 0x00,
>-  0xf5, 0xdc, 0x4f, 0x5d, 0xc4, 0x00
>-};
>-
>-const uint8_t kMaskRandom40_24[144] = {
>-  0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
>-  0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
>-  0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
>-  0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00,
>-  0xc6, 0x31, 0x8c, 0x63, 0x18, 0x00,
>-  0x23, 0x88, 0xe2, 0x38, 0x8e, 0x00,
>-  0x1a, 0x46, 0x91, 0xa4, 0x69, 0x00,
>-  0x24, 0xc9, 0x32, 0x4c, 0x93, 0x00,
>-  0x71, 0x1c, 0x47, 0x11, 0xc4, 0x00,
>-  0xf5, 0xdc, 0x4f, 0x5d, 0xc4, 0x00,
>-  0xc6, 0x31, 0x8c, 0x63, 0x18, 0x00,
>-  0x23, 0x88, 0xe2, 0x38, 0x8e, 0x00,
>-  0x1a, 0x46, 0x91, 0xa4, 0x69, 0x00,
>-  0x24, 0xc9, 0x32, 0x4c, 0x93, 0x00,
>-  0x71, 0x1c, 0x47, 0x11, 0xc4, 0x00,
>-  0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
>-  0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
>-  0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
>-  0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00,
>-  0x68, 0xde, 0x83, 0xa9, 0xcf, 0x00
>-};
>-
>-const uint8_t kMaskRandom40_25[150] = {
>-  0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
>-  0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
>-  0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
>-  0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00,
>-  0xc6, 0x31, 0x8c, 0x63, 0x18, 0x00,
>-  0x23, 0x88, 0xe2, 0x38, 0x8e, 0x00,
>-  0x1a, 0x46, 0x91, 0xa4, 0x69, 0x00,
>-  0x24, 0xc9, 0x32, 0x4c, 0x93, 0x00,
>-  0x71, 0x1c, 0x47, 0x11, 0xc4, 0x00,
>-  0xf5, 0xdc, 0x4f, 0x5d, 0xc4, 0x00,
>-  0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
>-  0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
>-  0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
>-  0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00,
>-  0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
>-  0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
>-  0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
>-  0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
>-  0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00
>-};
>-
>-const uint8_t kMaskRandom40_26[156] = {
>-  0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
>-  0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
>-  0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
>-  0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00,
>-  0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
>-  0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
>-  0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
>-  0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
>-  0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00,
>-  0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
>-  0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
>-  0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
>-  0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00,
>-  0xc6, 0x31, 0x8c, 0x63, 0x18, 0x00,
>-  0x23, 0x88, 0xe2, 0x38, 0x8e, 0x00,
>-  0x1a, 0x46, 0x91, 0xa4, 0x69, 0x00,
>-  0x24, 0xc9, 0x32, 0x4c, 0x93, 0x00,
>-  0x71, 0x1c, 0x47, 0x11, 0xc4, 0x00,
>-  0xf5, 0xdc, 0x4f, 0x5d, 0xc4, 0x00,
>-  0x06, 0x8e, 0x8c, 0x1a, 0xd2, 0x00
>-};
>-
>-const uint8_t kMaskRandom40_27[162] = {
>-  0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
>-  0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
>-  0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
>-  0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00,
>-  0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
>-  0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
>-  0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
>-  0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
>-  0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00,
>-  0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
>-  0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
>-  0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
>-  0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
>-  0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00,
>-  0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
>-  0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
>-  0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
>-  0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00,
>-  0x56, 0x3e, 0x25, 0x63, 0xe2, 0x00
>-};
>-
>-const uint8_t kMaskRandom40_28[168] = {
>-  0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
>-  0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
>-  0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
>-  0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
>-  0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00,
>-  0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
>-  0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
>-  0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
>-  0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00,
>-  0x56, 0x3e, 0x25, 0x63, 0xe2, 0x00,
>-  0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
>-  0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
>-  0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
>-  0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00,
>-  0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
>-  0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
>-  0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
>-  0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
>-  0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00,
>-  0x68, 0x0e, 0x9b, 0x52, 0xb6, 0x00
>-};
>-
>-const uint8_t kMaskRandom40_29[174] = {
>-  0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
>-  0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
>-  0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
>-  0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
>-  0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00,
>-  0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
>-  0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
>-  0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
>-  0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00,
>-  0x56, 0x3e, 0x25, 0x63, 0xe2, 0x00,
>-  0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
>-  0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
>-  0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
>-  0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
>-  0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00,
>-  0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
>-  0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
>-  0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
>-  0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
>-  0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
>-  0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
>-  0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
>-  0x50, 0x54, 0x15, 0x05, 0x41, 0x00
>-};
>-
>-const uint8_t kMaskRandom40_3[18] = {
>-  0xce, 0x33, 0x8c, 0xe3, 0x38, 0x00,
>-  0x55, 0x95, 0x65, 0x59, 0x56, 0x00,
>-  0xb1, 0x6a, 0x3b, 0x16, 0xa3, 0x00
>-};
>-
>-const uint8_t kMaskRandom40_30[180] = {
>-  0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
>-  0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
>-  0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
>-  0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
>-  0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00,
>-  0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
>-  0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
>-  0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
>-  0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
>-  0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
>-  0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
>-  0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
>-  0x50, 0x54, 0x15, 0x05, 0x41, 0x00,
>-  0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
>-  0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
>-  0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
>-  0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
>-  0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00,
>-  0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
>-  0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
>-  0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
>-  0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00,
>-  0x56, 0x3e, 0x25, 0x63, 0xe2, 0x00,
>-  0xe1, 0x47, 0x04, 0x05, 0x47, 0x00
>-};
>-
>-const uint8_t kMaskRandom40_31[186] = {
>-  0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
>-  0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
>-  0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
>-  0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
>-  0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00,
>-  0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
>-  0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
>-  0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
>-  0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
>-  0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
>-  0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
>-  0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
>-  0x50, 0x54, 0x15, 0x05, 0x41, 0x00,
>-  0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
>-  0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
>-  0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
>-  0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
>-  0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
>-  0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
>-  0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
>-  0x50, 0x54, 0x15, 0x05, 0x41, 0x00,
>-  0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
>-  0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
>-  0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
>-  0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
>-  0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00,
>-  0x28, 0x1c, 0x62, 0x81, 0xc6, 0x00
>-};
>-
>-const uint8_t kMaskRandom40_32[192] = {
>-  0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
>-  0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
>-  0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
>-  0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
>-  0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
>-  0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
>-  0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
>-  0x50, 0x54, 0x15, 0x05, 0x41, 0x00,
>-  0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
>-  0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
>-  0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
>-  0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
>-  0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00,
>-  0x28, 0x1c, 0x62, 0x81, 0xc6, 0x00,
>-  0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
>-  0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
>-  0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
>-  0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
>-  0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00,
>-  0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
>-  0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
>-  0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
>-  0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
>-  0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
>-  0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
>-  0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
>-  0x50, 0x54, 0x15, 0x05, 0x41, 0x00,
>-  0x03, 0x0c, 0x46, 0x10, 0xc5, 0x00
>-};
>-
>-const uint8_t kMaskRandom40_33[198] = {
>-  0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
>-  0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
>-  0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
>-  0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
>-  0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
>-  0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
>-  0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
>-  0x50, 0x54, 0x15, 0x05, 0x41, 0x00,
>-  0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
>-  0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
>-  0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
>-  0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
>-  0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00,
>-  0x28, 0x1c, 0x62, 0x81, 0xc6, 0x00,
>-  0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
>-  0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
>-  0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
>-  0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
>-  0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
>-  0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
>-  0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
>-  0x50, 0x54, 0x15, 0x05, 0x41, 0x00,
>-  0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
>-  0xe3, 0x38, 0xce, 0x33, 0x8c, 0x00,
>-  0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
>-  0x21, 0x48, 0x52, 0x14, 0x85, 0x00,
>-  0x52, 0x94, 0xa5, 0x29, 0x4a, 0x00,
>-  0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
>-  0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
>-  0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
>-  0x19, 0x86, 0x61, 0x98, 0x66, 0x00
>-};
>-
>-const uint8_t kMaskRandom40_34[204] = {
>-  0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
>-  0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
>-  0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
>-  0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
>-  0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
>-  0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
>-  0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
>-  0x50, 0x54, 0x15, 0x05, 0x41, 0x00,
>-  0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
>-  0xe3, 0x38, 0xce, 0x33, 0x8c, 0x00,
>-  0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
>-  0x21, 0x48, 0x52, 0x14, 0x85, 0x00,
>-  0x52, 0x94, 0xa5, 0x29, 0x4a, 0x00,
>-  0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
>-  0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
>-  0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
>-  0x19, 0x86, 0x61, 0x98, 0x66, 0x00,
>-  0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
>-  0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
>-  0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
>-  0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
>-  0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
>-  0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
>-  0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
>-  0x50, 0x54, 0x15, 0x05, 0x41, 0x00,
>-  0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
>-  0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
>-  0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
>-  0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
>-  0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00,
>-  0x28, 0x1c, 0x62, 0x81, 0xc6, 0x00,
>-  0x87, 0x3c, 0x08, 0x19, 0x31, 0x00
>-};
>-
>-const uint8_t kMaskRandom40_35[210] = {
>-  0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
>-  0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
>-  0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
>-  0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
>-  0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
>-  0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
>-  0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
>-  0x50, 0x54, 0x15, 0x05, 0x41, 0x00,
>-  0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
>-  0xe3, 0x38, 0xce, 0x33, 0x8c, 0x00,
>-  0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
>-  0x21, 0x48, 0x52, 0x14, 0x85, 0x00,
>-  0x52, 0x94, 0xa5, 0x29, 0x4a, 0x00,
>-  0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
>-  0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
>-  0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
>-  0x19, 0x86, 0x61, 0x98, 0x66, 0x00,
>-  0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
>-  0xe3, 0x38, 0xce, 0x33, 0x8c, 0x00,
>-  0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
>-  0x21, 0x48, 0x52, 0x14, 0x85, 0x00,
>-  0x52, 0x94, 0xa5, 0x29, 0x4a, 0x00,
>-  0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
>-  0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
>-  0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
>-  0x19, 0x86, 0x61, 0x98, 0x66, 0x00,
>-  0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
>-  0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
>-  0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
>-  0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
>-  0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
>-  0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
>-  0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
>-  0x50, 0x54, 0x15, 0x05, 0x41, 0x00,
>-  0x21, 0x7b, 0xf2, 0x17, 0xbf, 0x00
>-};
>-
>-const uint8_t kMaskRandom40_36[216] = {
>-  0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
>-  0xe3, 0x38, 0xce, 0x33, 0x8c, 0x00,
>-  0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
>-  0x21, 0x48, 0x52, 0x14, 0x85, 0x00,
>-  0x52, 0x94, 0xa5, 0x29, 0x4a, 0x00,
>-  0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
>-  0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
>-  0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
>-  0x19, 0x86, 0x61, 0x98, 0x66, 0x00,
>-  0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
>-  0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
>-  0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
>-  0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
>-  0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
>-  0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
>-  0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
>-  0x50, 0x54, 0x15, 0x05, 0x41, 0x00,
>-  0x21, 0x7b, 0xf2, 0x17, 0xbf, 0x00,
>-  0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
>-  0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
>-  0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
>-  0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
>-  0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
>-  0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
>-  0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
>-  0x50, 0x54, 0x15, 0x05, 0x41, 0x00,
>-  0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
>-  0xe3, 0x38, 0xce, 0x33, 0x8c, 0x00,
>-  0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
>-  0x21, 0x48, 0x52, 0x14, 0x85, 0x00,
>-  0x52, 0x94, 0xa5, 0x29, 0x4a, 0x00,
>-  0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
>-  0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
>-  0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
>-  0x19, 0x86, 0x61, 0x98, 0x66, 0x00,
>-  0x1e, 0xb9, 0x3d, 0x25, 0xcc, 0x00
>-};
>-
>-const uint8_t kMaskRandom40_37[222] = {
>-  0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
>-  0xe3, 0x38, 0xce, 0x33, 0x8c, 0x00,
>-  0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
>-  0x21, 0x48, 0x52, 0x14, 0x85, 0x00,
>-  0x52, 0x94, 0xa5, 0x29, 0x4a, 0x00,
>-  0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
>-  0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
>-  0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
>-  0x19, 0x86, 0x61, 0x98, 0x66, 0x00,
>-  0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
>-  0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
>-  0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
>-  0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
>-  0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
>-  0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
>-  0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
>-  0x50, 0x54, 0x15, 0x05, 0x41, 0x00,
>-  0x21, 0x7b, 0xf2, 0x17, 0xbf, 0x00,
>-  0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
>-  0xe3, 0x38, 0xce, 0x33, 0x8c, 0x00,
>-  0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
>-  0x21, 0x48, 0x52, 0x14, 0x85, 0x00,
>-  0x52, 0x94, 0xa5, 0x29, 0x4a, 0x00,
>-  0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
>-  0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
>-  0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
>-  0x19, 0x86, 0x61, 0x98, 0x66, 0x00,
>-  0x4c, 0x13, 0x04, 0xc1, 0x30, 0x00,
>-  0x51, 0x14, 0x45, 0x11, 0x44, 0x00,
>-  0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
>-  0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
>-  0x03, 0x80, 0xe0, 0x38, 0x0e, 0x00,
>-  0x86, 0x21, 0x88, 0x62, 0x18, 0x00,
>-  0x29, 0x0a, 0x42, 0x90, 0xa4, 0x00,
>-  0x42, 0x50, 0x94, 0x25, 0x09, 0x00,
>-  0x98, 0x26, 0x09, 0x82, 0x60, 0x00,
>-  0x30, 0x8c, 0x23, 0x08, 0xc2, 0x00
>-};
>-
>-const uint8_t kMaskRandom40_38[228] = {
>-  0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
>-  0xe3, 0x38, 0xce, 0x33, 0x8c, 0x00,
>-  0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
>-  0x21, 0x48, 0x52, 0x14, 0x85, 0x00,
>-  0x52, 0x94, 0xa5, 0x29, 0x4a, 0x00,
>-  0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
>-  0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
>-  0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
>-  0x19, 0x86, 0x61, 0x98, 0x66, 0x00,
>-  0x4c, 0x13, 0x04, 0xc1, 0x30, 0x00,
>-  0x51, 0x14, 0x45, 0x11, 0x44, 0x00,
>-  0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
>-  0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
>-  0x03, 0x80, 0xe0, 0x38, 0x0e, 0x00,
>-  0x86, 0x21, 0x88, 0x62, 0x18, 0x00,
>-  0x29, 0x0a, 0x42, 0x90, 0xa4, 0x00,
>-  0x42, 0x50, 0x94, 0x25, 0x09, 0x00,
>-  0x98, 0x26, 0x09, 0x82, 0x60, 0x00,
>-  0x30, 0x8c, 0x23, 0x08, 0xc2, 0x00,
>-  0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
>-  0xe3, 0x38, 0xce, 0x33, 0x8c, 0x00,
>-  0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
>-  0x21, 0x48, 0x52, 0x14, 0x85, 0x00,
>-  0x52, 0x94, 0xa5, 0x29, 0x4a, 0x00,
>-  0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
>-  0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
>-  0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
>-  0x19, 0x86, 0x61, 0x98, 0x66, 0x00,
>-  0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
>-  0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
>-  0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
>-  0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
>-  0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
>-  0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
>-  0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
>-  0x50, 0x54, 0x15, 0x05, 0x41, 0x00,
>-  0x21, 0x7b, 0xf2, 0x17, 0xbf, 0x00,
>-  0xea, 0xaa, 0x20, 0xa2, 0x1b, 0x00
>-};
>-
>-const uint8_t kMaskRandom40_39[234] = {
>-  0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
>-  0xe3, 0x38, 0xce, 0x33, 0x8c, 0x00,
>-  0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
>-  0x21, 0x48, 0x52, 0x14, 0x85, 0x00,
>-  0x52, 0x94, 0xa5, 0x29, 0x4a, 0x00,
>-  0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
>-  0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
>-  0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
>-  0x19, 0x86, 0x61, 0x98, 0x66, 0x00,
>-  0x4c, 0x13, 0x04, 0xc1, 0x30, 0x00,
>-  0x51, 0x14, 0x45, 0x11, 0x44, 0x00,
>-  0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
>-  0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
>-  0x03, 0x80, 0xe0, 0x38, 0x0e, 0x00,
>-  0x86, 0x21, 0x88, 0x62, 0x18, 0x00,
>-  0x29, 0x0a, 0x42, 0x90, 0xa4, 0x00,
>-  0x42, 0x50, 0x94, 0x25, 0x09, 0x00,
>-  0x98, 0x26, 0x09, 0x82, 0x60, 0x00,
>-  0x30, 0x8c, 0x23, 0x08, 0xc2, 0x00,
>-  0x4c, 0x13, 0x04, 0xc1, 0x30, 0x00,
>-  0x51, 0x14, 0x45, 0x11, 0x44, 0x00,
>-  0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
>-  0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
>-  0x03, 0x80, 0xe0, 0x38, 0x0e, 0x00,
>-  0x86, 0x21, 0x88, 0x62, 0x18, 0x00,
>-  0x29, 0x0a, 0x42, 0x90, 0xa4, 0x00,
>-  0x42, 0x50, 0x94, 0x25, 0x09, 0x00,
>-  0x98, 0x26, 0x09, 0x82, 0x60, 0x00,
>-  0x30, 0x8c, 0x23, 0x08, 0xc2, 0x00,
>-  0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
>-  0xe3, 0x38, 0xce, 0x33, 0x8c, 0x00,
>-  0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
>-  0x21, 0x48, 0x52, 0x14, 0x85, 0x00,
>-  0x52, 0x94, 0xa5, 0x29, 0x4a, 0x00,
>-  0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
>-  0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
>-  0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
>-  0x19, 0x86, 0x61, 0x98, 0x66, 0x00,
>-  0xf7, 0x8d, 0xaf, 0x78, 0xda, 0x00
>-};
>-
>-const uint8_t kMaskRandom40_4[24] = {
>-  0xe6, 0x39, 0x8e, 0x63, 0x98, 0x00,
>-  0x33, 0x8c, 0xe3, 0x38, 0xce, 0x00,
>-  0x98, 0xe6, 0x39, 0x8e, 0x63, 0x00,
>-  0x2d, 0x4b, 0x52, 0xd4, 0xb5, 0x00
>-};
>-
>-const uint8_t kMaskRandom40_40[240] = {
>-  0x4c, 0x13, 0x04, 0xc1, 0x30, 0x00,
>-  0x51, 0x14, 0x45, 0x11, 0x44, 0x00,
>-  0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
>-  0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
>-  0x03, 0x80, 0xe0, 0x38, 0x0e, 0x00,
>-  0x86, 0x21, 0x88, 0x62, 0x18, 0x00,
>-  0x29, 0x0a, 0x42, 0x90, 0xa4, 0x00,
>-  0x42, 0x50, 0x94, 0x25, 0x09, 0x00,
>-  0x98, 0x26, 0x09, 0x82, 0x60, 0x00,
>-  0x30, 0x8c, 0x23, 0x08, 0xc2, 0x00,
>-  0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
>-  0xe3, 0x38, 0xce, 0x33, 0x8c, 0x00,
>-  0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
>-  0x21, 0x48, 0x52, 0x14, 0x85, 0x00,
>-  0x52, 0x94, 0xa5, 0x29, 0x4a, 0x00,
>-  0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
>-  0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
>-  0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
>-  0x19, 0x86, 0x61, 0x98, 0x66, 0x00,
>-  0xf7, 0x8d, 0xaf, 0x78, 0xda, 0x00,
>-  0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
>-  0xe3, 0x38, 0xce, 0x33, 0x8c, 0x00,
>-  0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
>-  0x21, 0x48, 0x52, 0x14, 0x85, 0x00,
>-  0x52, 0x94, 0xa5, 0x29, 0x4a, 0x00,
>-  0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
>-  0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
>-  0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
>-  0x19, 0x86, 0x61, 0x98, 0x66, 0x00,
>-  0x4c, 0x13, 0x04, 0xc1, 0x30, 0x00,
>-  0x51, 0x14, 0x45, 0x11, 0x44, 0x00,
>-  0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
>-  0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
>-  0x03, 0x80, 0xe0, 0x38, 0x0e, 0x00,
>-  0x86, 0x21, 0x88, 0x62, 0x18, 0x00,
>-  0x29, 0x0a, 0x42, 0x90, 0xa4, 0x00,
>-  0x42, 0x50, 0x94, 0x25, 0x09, 0x00,
>-  0x98, 0x26, 0x09, 0x82, 0x60, 0x00,
>-  0x30, 0x8c, 0x23, 0x08, 0xc2, 0x00,
>-  0xa6, 0xf3, 0xab, 0x1b, 0x87, 0x00
>-};
>-
>-const uint8_t kMaskRandom40_5[30] = {
>-  0xce, 0x33, 0x8c, 0xe3, 0x38, 0x00,
>-  0x63, 0x98, 0xe6, 0x39, 0x8e, 0x00,
>-  0x98, 0xe5, 0x39, 0x8e, 0x53, 0x00,
>-  0x2b, 0x53, 0x52, 0xb5, 0x35, 0x00,
>-  0xb4, 0x5c, 0xab, 0x45, 0xca, 0x00
>-};
>-
>-const uint8_t kMaskRandom40_6[36] = {
>-  0x4c, 0x1b, 0x04, 0xc1, 0xb0, 0x00,
>-  0x51, 0x34, 0x45, 0x13, 0x44, 0x00,
>-  0x20, 0xe8, 0x32, 0x0e, 0x83, 0x00,
>-  0x85, 0x41, 0x58, 0x54, 0x15, 0x00,
>-  0x06, 0x86, 0xa0, 0x68, 0x6a, 0x00,
>-  0x9a, 0x21, 0x89, 0xa2, 0x18, 0x00
>-};
>-
>-const uint8_t kMaskRandom40_7[42] = {
>-  0x4e, 0x11, 0x84, 0xe1, 0x18, 0x00,
>-  0x33, 0x2c, 0x03, 0x32, 0xc0, 0x00,
>-  0x10, 0x0e, 0xb1, 0x00, 0xeb, 0x00,
>-  0x81, 0x51, 0x58, 0x15, 0x15, 0x00,
>-  0x24, 0xc4, 0xa2, 0x4c, 0x4a, 0x00,
>-  0xd4, 0x23, 0x0d, 0x42, 0x30, 0x00,
>-  0x0c, 0xa2, 0x60, 0xca, 0x26, 0x00
>-};
>-
>-const uint8_t kMaskRandom40_8[48] = {
>-  0x27, 0x09, 0xc2, 0x70, 0x9c, 0x00,
>-  0x89, 0xa2, 0x68, 0x9a, 0x26, 0x00,
>-  0xd0, 0x74, 0x1d, 0x07, 0x41, 0x00,
>-  0x24, 0xc9, 0x32, 0x4c, 0x93, 0x00,
>-  0xe2, 0x90, 0xae, 0x29, 0x0a, 0x00,
>-  0xc6, 0x31, 0x8c, 0x63, 0x18, 0x00,
>-  0x31, 0x8c, 0x63, 0x18, 0xc6, 0x00,
>-  0x18, 0xc6, 0x31, 0x8c, 0x63, 0x00
>-};
>-
>-const uint8_t kMaskRandom40_9[54] = {
>-  0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
>-  0x62, 0x38, 0xc6, 0x23, 0x8c, 0x00,
>-  0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
>-  0xe1, 0x48, 0x5e, 0x14, 0x85, 0x00,
>-  0x13, 0x94, 0xa1, 0x39, 0x4a, 0x00,
>-  0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
>-  0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
>-  0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
>-  0x49, 0x86, 0x54, 0x98, 0x65, 0x00
>-};
>-
>-const uint8_t kMaskRandom41_1[6] = {
>-  0xff, 0xff, 0xff, 0xff, 0xff, 0x80
>-};
>-
>-const uint8_t kMaskRandom41_10[60] = {
>-  0x4c, 0x13, 0x04, 0xc1, 0x91, 0x00,
>-  0x51, 0x14, 0x45, 0x11, 0x45, 0x00,
>-  0xa0, 0x68, 0x1a, 0x06, 0xa4, 0x00,
>-  0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
>-  0x03, 0x80, 0xe0, 0x3b, 0x40, 0x00,
>-  0x86, 0x21, 0x88, 0x62, 0x09, 0x00,
>-  0x29, 0x0a, 0x42, 0x90, 0x84, 0x80,
>-  0x42, 0x50, 0x94, 0x24, 0x30, 0x80,
>-  0x98, 0x26, 0x09, 0x81, 0x28, 0x00,
>-  0x30, 0x8c, 0x23, 0x08, 0x4a, 0x80
>-};
>-
>-const uint8_t kMaskRandom41_11[66] = {
>-  0xc6, 0x31, 0x8c, 0x62, 0x1a, 0x00,
>-  0x23, 0x88, 0xe2, 0x38, 0x8c, 0x80,
>-  0x1a, 0x46, 0x91, 0xa4, 0x58, 0x80,
>-  0x24, 0xc9, 0x32, 0x4d, 0x30, 0x80,
>-  0x71, 0x1c, 0x47, 0x11, 0x07, 0x00,
>-  0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
>-  0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
>-  0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
>-  0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
>-  0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00
>-};
>-
>-const uint8_t kMaskRandom41_12[72] = {
>-  0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
>-  0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
>-  0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
>-  0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
>-  0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00,
>-  0xc6, 0x31, 0x8c, 0x62, 0x1a, 0x00,
>-  0x23, 0x88, 0xe2, 0x38, 0x8c, 0x80,
>-  0x1a, 0x46, 0x91, 0xa4, 0x58, 0x80,
>-  0x24, 0xc9, 0x32, 0x4d, 0x30, 0x80,
>-  0x71, 0x1c, 0x47, 0x11, 0x07, 0x00,
>-  0xf5, 0xdc, 0x4a, 0x06, 0x51, 0x80
>-};
>-
>-const uint8_t kMaskRandom41_13[78] = {
>-  0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
>-  0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
>-  0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
>-  0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
>-  0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00,
>-  0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
>-  0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
>-  0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
>-  0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
>-  0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
>-  0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80
>-};
>-
>-const uint8_t kMaskRandom41_14[84] = {
>-  0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
>-  0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
>-  0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
>-  0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
>-  0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
>-  0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80,
>-  0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
>-  0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
>-  0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
>-  0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
>-  0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00,
>-  0x56, 0x3e, 0x24, 0xdd, 0x0c, 0x00
>-};
>-
>-const uint8_t kMaskRandom41_15[90] = {
>-  0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
>-  0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
>-  0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
>-  0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
>-  0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
>-  0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80,
>-  0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
>-  0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
>-  0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
>-  0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
>-  0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
>-  0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
>-  0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
>-  0x50, 0x54, 0x15, 0x04, 0x50, 0x80
>-};
>-
>-const uint8_t kMaskRandom41_16[96] = {
>-  0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
>-  0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
>-  0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
>-  0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
>-  0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
>-  0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
>-  0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
>-  0x50, 0x54, 0x15, 0x04, 0x50, 0x80,
>-  0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
>-  0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
>-  0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
>-  0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
>-  0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
>-  0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80,
>-  0x28, 0x1c, 0x63, 0xbf, 0x53, 0x80
>-};
>-
>-const uint8_t kMaskRandom41_17[102] = {
>-  0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
>-  0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
>-  0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
>-  0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
>-  0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
>-  0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
>-  0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
>-  0x50, 0x54, 0x15, 0x04, 0x50, 0x80,
>-  0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
>-  0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
>-  0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
>-  0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
>-  0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
>-  0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
>-  0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
>-  0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
>-  0x19, 0x86, 0x61, 0x99, 0xe0, 0x00
>-};
>-
>-const uint8_t kMaskRandom41_18[108] = {
>-  0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
>-  0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
>-  0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
>-  0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
>-  0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
>-  0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
>-  0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
>-  0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
>-  0x19, 0x86, 0x61, 0x99, 0xe0, 0x00,
>-  0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
>-  0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
>-  0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
>-  0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
>-  0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
>-  0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
>-  0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
>-  0x50, 0x54, 0x15, 0x04, 0x50, 0x80,
>-  0x21, 0x7b, 0xf5, 0xa5, 0x65, 0x80
>-};
>-
>-const uint8_t kMaskRandom41_19[114] = {
>-  0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
>-  0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
>-  0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
>-  0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
>-  0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
>-  0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
>-  0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
>-  0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
>-  0x19, 0x86, 0x61, 0x99, 0xe0, 0x00,
>-  0x4c, 0x13, 0x04, 0xc1, 0x91, 0x00,
>-  0x51, 0x14, 0x45, 0x11, 0x45, 0x00,
>-  0xa0, 0x68, 0x1a, 0x06, 0xa4, 0x00,
>-  0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
>-  0x03, 0x80, 0xe0, 0x3b, 0x40, 0x00,
>-  0x86, 0x21, 0x88, 0x62, 0x09, 0x00,
>-  0x29, 0x0a, 0x42, 0x90, 0x84, 0x80,
>-  0x42, 0x50, 0x94, 0x24, 0x30, 0x80,
>-  0x98, 0x26, 0x09, 0x81, 0x28, 0x00,
>-  0x30, 0x8c, 0x23, 0x08, 0x4a, 0x80
>-};
>-
>-const uint8_t kMaskRandom41_2[12] = {
>-  0xee, 0x3b, 0x8e, 0xe3, 0xb3, 0x00,
>-  0x99, 0xe6, 0x79, 0x9e, 0x6e, 0x80
>-};
>-
>-const uint8_t kMaskRandom41_20[120] = {
>-  0x4c, 0x13, 0x04, 0xc1, 0x91, 0x00,
>-  0x51, 0x14, 0x45, 0x11, 0x45, 0x00,
>-  0xa0, 0x68, 0x1a, 0x06, 0xa4, 0x00,
>-  0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
>-  0x03, 0x80, 0xe0, 0x3b, 0x40, 0x00,
>-  0x86, 0x21, 0x88, 0x62, 0x09, 0x00,
>-  0x29, 0x0a, 0x42, 0x90, 0x84, 0x80,
>-  0x42, 0x50, 0x94, 0x24, 0x30, 0x80,
>-  0x98, 0x26, 0x09, 0x81, 0x28, 0x00,
>-  0x30, 0x8c, 0x23, 0x08, 0x4a, 0x80,
>-  0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
>-  0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
>-  0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
>-  0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
>-  0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
>-  0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
>-  0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
>-  0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
>-  0x19, 0x86, 0x61, 0x99, 0xe0, 0x00,
>-  0xf7, 0x8d, 0xa2, 0xa0, 0x33, 0x00
>-};
>-
>-const uint8_t kMaskRandom41_21[126] = {
>-  0x4c, 0x13, 0x04, 0xc1, 0x91, 0x00,
>-  0x51, 0x14, 0x45, 0x11, 0x45, 0x00,
>-  0xa0, 0x68, 0x1a, 0x06, 0xa4, 0x00,
>-  0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
>-  0x03, 0x80, 0xe0, 0x3b, 0x40, 0x00,
>-  0x86, 0x21, 0x88, 0x62, 0x09, 0x00,
>-  0x29, 0x0a, 0x42, 0x90, 0x84, 0x80,
>-  0x42, 0x50, 0x94, 0x24, 0x30, 0x80,
>-  0x98, 0x26, 0x09, 0x81, 0x28, 0x00,
>-  0x30, 0x8c, 0x23, 0x08, 0x4a, 0x80,
>-  0xc6, 0x31, 0x8c, 0x62, 0x1a, 0x00,
>-  0x23, 0x88, 0xe2, 0x38, 0x8c, 0x80,
>-  0x1a, 0x46, 0x91, 0xa4, 0x58, 0x80,
>-  0x24, 0xc9, 0x32, 0x4d, 0x30, 0x80,
>-  0x71, 0x1c, 0x47, 0x11, 0x07, 0x00,
>-  0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
>-  0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
>-  0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
>-  0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
>-  0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00
>-};
>-
>-const uint8_t kMaskRandom41_22[132] = {
>-  0xc6, 0x31, 0x8c, 0x62, 0x1a, 0x00,
>-  0x23, 0x88, 0xe2, 0x38, 0x8c, 0x80,
>-  0x1a, 0x46, 0x91, 0xa4, 0x58, 0x80,
>-  0x24, 0xc9, 0x32, 0x4d, 0x30, 0x80,
>-  0x71, 0x1c, 0x47, 0x11, 0x07, 0x00,
>-  0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
>-  0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
>-  0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
>-  0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
>-  0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00,
>-  0x4c, 0x13, 0x04, 0xc1, 0x91, 0x00,
>-  0x51, 0x14, 0x45, 0x11, 0x45, 0x00,
>-  0xa0, 0x68, 0x1a, 0x06, 0xa4, 0x00,
>-  0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
>-  0x03, 0x80, 0xe0, 0x3b, 0x40, 0x00,
>-  0x86, 0x21, 0x88, 0x62, 0x09, 0x00,
>-  0x29, 0x0a, 0x42, 0x90, 0x84, 0x80,
>-  0x42, 0x50, 0x94, 0x24, 0x30, 0x80,
>-  0x98, 0x26, 0x09, 0x81, 0x28, 0x00,
>-  0x30, 0x8c, 0x23, 0x08, 0x4a, 0x80,
>-  0x33, 0x09, 0x6e, 0x49, 0x6b, 0x80
>-};
>-
>-const uint8_t kMaskRandom41_23[138] = {
>-  0xc6, 0x31, 0x8c, 0x62, 0x1a, 0x00,
>-  0x23, 0x88, 0xe2, 0x38, 0x8c, 0x80,
>-  0x1a, 0x46, 0x91, 0xa4, 0x58, 0x80,
>-  0x24, 0xc9, 0x32, 0x4d, 0x30, 0x80,
>-  0x71, 0x1c, 0x47, 0x11, 0x07, 0x00,
>-  0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
>-  0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
>-  0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
>-  0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
>-  0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00,
>-  0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
>-  0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
>-  0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
>-  0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
>-  0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00,
>-  0xc6, 0x31, 0x8c, 0x62, 0x1a, 0x00,
>-  0x23, 0x88, 0xe2, 0x38, 0x8c, 0x80,
>-  0x1a, 0x46, 0x91, 0xa4, 0x58, 0x80,
>-  0x24, 0xc9, 0x32, 0x4d, 0x30, 0x80,
>-  0x71, 0x1c, 0x47, 0x11, 0x07, 0x00,
>-  0xf5, 0xdc, 0x4a, 0x06, 0x51, 0x80
>-};
>-
>-const uint8_t kMaskRandom41_24[144] = {
>-  0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
>-  0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
>-  0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
>-  0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
>-  0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00,
>-  0xc6, 0x31, 0x8c, 0x62, 0x1a, 0x00,
>-  0x23, 0x88, 0xe2, 0x38, 0x8c, 0x80,
>-  0x1a, 0x46, 0x91, 0xa4, 0x58, 0x80,
>-  0x24, 0xc9, 0x32, 0x4d, 0x30, 0x80,
>-  0x71, 0x1c, 0x47, 0x11, 0x07, 0x00,
>-  0xf5, 0xdc, 0x4a, 0x06, 0x51, 0x80,
>-  0xc6, 0x31, 0x8c, 0x62, 0x1a, 0x00,
>-  0x23, 0x88, 0xe2, 0x38, 0x8c, 0x80,
>-  0x1a, 0x46, 0x91, 0xa4, 0x58, 0x80,
>-  0x24, 0xc9, 0x32, 0x4d, 0x30, 0x80,
>-  0x71, 0x1c, 0x47, 0x11, 0x07, 0x00,
>-  0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
>-  0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
>-  0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
>-  0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
>-  0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00,
>-  0x45, 0xa6, 0xef, 0xc9, 0xc3, 0x00
>-};
>-
>-const uint8_t kMaskRandom41_25[150] = {
>-  0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
>-  0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
>-  0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
>-  0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
>-  0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00,
>-  0xc6, 0x31, 0x8c, 0x62, 0x1a, 0x00,
>-  0x23, 0x88, 0xe2, 0x38, 0x8c, 0x80,
>-  0x1a, 0x46, 0x91, 0xa4, 0x58, 0x80,
>-  0x24, 0xc9, 0x32, 0x4d, 0x30, 0x80,
>-  0x71, 0x1c, 0x47, 0x11, 0x07, 0x00,
>-  0xf5, 0xdc, 0x4a, 0x06, 0x51, 0x80,
>-  0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
>-  0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
>-  0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
>-  0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
>-  0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00,
>-  0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
>-  0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
>-  0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
>-  0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
>-  0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
>-  0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80
>-};
>-
>-const uint8_t kMaskRandom41_26[156] = {
>-  0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
>-  0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
>-  0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
>-  0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
>-  0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00,
>-  0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
>-  0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
>-  0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
>-  0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
>-  0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
>-  0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80,
>-  0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
>-  0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
>-  0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
>-  0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
>-  0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00,
>-  0xc6, 0x31, 0x8c, 0x62, 0x1a, 0x00,
>-  0x23, 0x88, 0xe2, 0x38, 0x8c, 0x80,
>-  0x1a, 0x46, 0x91, 0xa4, 0x58, 0x80,
>-  0x24, 0xc9, 0x32, 0x4d, 0x30, 0x80,
>-  0x71, 0x1c, 0x47, 0x11, 0x07, 0x00,
>-  0xf5, 0xdc, 0x4a, 0x06, 0x51, 0x80,
>-  0x6f, 0x72, 0xf1, 0xe7, 0x1a, 0x80
>-};
>-
>-const uint8_t kMaskRandom41_27[162] = {
>-  0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
>-  0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
>-  0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
>-  0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
>-  0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00,
>-  0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
>-  0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
>-  0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
>-  0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
>-  0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
>-  0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80,
>-  0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
>-  0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
>-  0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
>-  0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
>-  0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
>-  0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80,
>-  0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
>-  0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
>-  0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
>-  0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
>-  0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00,
>-  0x56, 0x3e, 0x24, 0xdd, 0x0c, 0x00
>-};
>-
>-const uint8_t kMaskRandom41_28[168] = {
>-  0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
>-  0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
>-  0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
>-  0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
>-  0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
>-  0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80,
>-  0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
>-  0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
>-  0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
>-  0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
>-  0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00,
>-  0x56, 0x3e, 0x24, 0xdd, 0x0c, 0x00,
>-  0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
>-  0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
>-  0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
>-  0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
>-  0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00,
>-  0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
>-  0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
>-  0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
>-  0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
>-  0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
>-  0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80,
>-  0x61, 0x2c, 0xfa, 0x25, 0x38, 0x00
>-};
>-
>-const uint8_t kMaskRandom41_29[174] = {
>-  0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
>-  0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
>-  0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
>-  0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
>-  0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
>-  0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80,
>-  0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
>-  0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
>-  0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
>-  0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
>-  0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00,
>-  0x56, 0x3e, 0x24, 0xdd, 0x0c, 0x00,
>-  0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
>-  0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
>-  0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
>-  0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
>-  0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
>-  0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80,
>-  0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
>-  0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
>-  0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
>-  0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
>-  0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
>-  0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
>-  0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
>-  0x50, 0x54, 0x15, 0x04, 0x50, 0x80
>-};
>-
>-const uint8_t kMaskRandom41_3[18] = {
>-  0xce, 0x33, 0x8c, 0xe3, 0x2b, 0x00,
>-  0x55, 0x95, 0x65, 0x5d, 0xc5, 0x00,
>-  0xb1, 0x6a, 0x3a, 0x8e, 0xd8, 0x80
>-};
>-
>-const uint8_t kMaskRandom41_30[180] = {
>-  0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
>-  0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
>-  0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
>-  0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
>-  0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
>-  0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80,
>-  0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
>-  0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
>-  0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
>-  0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
>-  0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
>-  0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
>-  0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
>-  0x50, 0x54, 0x15, 0x04, 0x50, 0x80,
>-  0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
>-  0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
>-  0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
>-  0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
>-  0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
>-  0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80,
>-  0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
>-  0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
>-  0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
>-  0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
>-  0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
>-  0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00,
>-  0x56, 0x3e, 0x24, 0xdd, 0x0c, 0x00,
>-  0x59, 0x53, 0x31, 0x62, 0x15, 0x00
>-};
>-
>-const uint8_t kMaskRandom41_31[186] = {
>-  0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
>-  0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
>-  0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
>-  0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
>-  0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
>-  0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80,
>-  0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
>-  0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
>-  0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
>-  0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
>-  0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
>-  0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
>-  0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
>-  0x50, 0x54, 0x15, 0x04, 0x50, 0x80,
>-  0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
>-  0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
>-  0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
>-  0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
>-  0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
>-  0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
>-  0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
>-  0x50, 0x54, 0x15, 0x04, 0x50, 0x80,
>-  0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
>-  0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
>-  0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
>-  0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
>-  0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
>-  0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80,
>-  0x28, 0x1c, 0x63, 0xbf, 0x53, 0x80
>-};
>-
>-const uint8_t kMaskRandom41_32[192] = {
>-  0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
>-  0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
>-  0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
>-  0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
>-  0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
>-  0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
>-  0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
>-  0x50, 0x54, 0x15, 0x04, 0x50, 0x80,
>-  0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
>-  0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
>-  0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
>-  0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
>-  0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
>-  0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80,
>-  0x28, 0x1c, 0x63, 0xbf, 0x53, 0x80,
>-  0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
>-  0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
>-  0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
>-  0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
>-  0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
>-  0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80,
>-  0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
>-  0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
>-  0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
>-  0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
>-  0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
>-  0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
>-  0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
>-  0x50, 0x54, 0x15, 0x04, 0x50, 0x80,
>-  0xca, 0xbb, 0xcb, 0x6d, 0xaa, 0x00
>-};
>-
>-const uint8_t kMaskRandom41_33[198] = {
>-  0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
>-  0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
>-  0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
>-  0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
>-  0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
>-  0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
>-  0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
>-  0x50, 0x54, 0x15, 0x04, 0x50, 0x80,
>-  0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
>-  0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
>-  0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
>-  0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
>-  0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
>-  0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80,
>-  0x28, 0x1c, 0x63, 0xbf, 0x53, 0x80,
>-  0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
>-  0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
>-  0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
>-  0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
>-  0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
>-  0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
>-  0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
>-  0x50, 0x54, 0x15, 0x04, 0x50, 0x80,
>-  0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
>-  0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
>-  0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
>-  0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
>-  0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
>-  0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
>-  0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
>-  0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
>-  0x19, 0x86, 0x61, 0x99, 0xe0, 0x00
>-};
>-
>-const uint8_t kMaskRandom41_34[204] = {
>-  0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
>-  0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
>-  0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
>-  0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
>-  0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
>-  0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
>-  0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
>-  0x50, 0x54, 0x15, 0x04, 0x50, 0x80,
>-  0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
>-  0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
>-  0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
>-  0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
>-  0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
>-  0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
>-  0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
>-  0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
>-  0x19, 0x86, 0x61, 0x99, 0xe0, 0x00,
>-  0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
>-  0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
>-  0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
>-  0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
>-  0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
>-  0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
>-  0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
>-  0x50, 0x54, 0x15, 0x04, 0x50, 0x80,
>-  0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
>-  0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
>-  0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
>-  0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
>-  0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
>-  0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
>-  0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80,
>-  0x28, 0x1c, 0x63, 0xbf, 0x53, 0x80,
>-  0xbd, 0x37, 0x3f, 0x75, 0x36, 0x80
>-};
>-
>-const uint8_t kMaskRandom41_35[210] = {
>-  0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
>-  0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
>-  0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
>-  0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
>-  0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
>-  0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
>-  0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
>-  0x50, 0x54, 0x15, 0x04, 0x50, 0x80,
>-  0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
>-  0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
>-  0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
>-  0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
>-  0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
>-  0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
>-  0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
>-  0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
>-  0x19, 0x86, 0x61, 0x99, 0xe0, 0x00,
>-  0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
>-  0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
>-  0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
>-  0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
>-  0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
>-  0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
>-  0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
>-  0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
>-  0x19, 0x86, 0x61, 0x99, 0xe0, 0x00,
>-  0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
>-  0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
>-  0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
>-  0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
>-  0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
>-  0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
>-  0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
>-  0x50, 0x54, 0x15, 0x04, 0x50, 0x80,
>-  0x21, 0x7b, 0xf5, 0xa5, 0x65, 0x80
>-};
>-
>-const uint8_t kMaskRandom41_36[216] = {
>-  0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
>-  0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
>-  0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
>-  0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
>-  0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
>-  0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
>-  0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
>-  0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
>-  0x19, 0x86, 0x61, 0x99, 0xe0, 0x00,
>-  0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
>-  0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
>-  0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
>-  0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
>-  0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
>-  0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
>-  0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
>-  0x50, 0x54, 0x15, 0x04, 0x50, 0x80,
>-  0x21, 0x7b, 0xf5, 0xa5, 0x65, 0x80,
>-  0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
>-  0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
>-  0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
>-  0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
>-  0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
>-  0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
>-  0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
>-  0x50, 0x54, 0x15, 0x04, 0x50, 0x80,
>-  0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
>-  0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
>-  0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
>-  0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
>-  0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
>-  0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
>-  0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
>-  0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
>-  0x19, 0x86, 0x61, 0x99, 0xe0, 0x00,
>-  0xc1, 0xb1, 0x80, 0xbe, 0x3e, 0x00
>-};
>-
>-const uint8_t kMaskRandom41_37[222] = {
>-  0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
>-  0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
>-  0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
>-  0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
>-  0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
>-  0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
>-  0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
>-  0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
>-  0x19, 0x86, 0x61, 0x99, 0xe0, 0x00,
>-  0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
>-  0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
>-  0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
>-  0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
>-  0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
>-  0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
>-  0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
>-  0x50, 0x54, 0x15, 0x04, 0x50, 0x80,
>-  0x21, 0x7b, 0xf5, 0xa5, 0x65, 0x80,
>-  0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
>-  0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
>-  0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
>-  0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
>-  0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
>-  0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
>-  0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
>-  0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
>-  0x19, 0x86, 0x61, 0x99, 0xe0, 0x00,
>-  0x4c, 0x13, 0x04, 0xc1, 0x91, 0x00,
>-  0x51, 0x14, 0x45, 0x11, 0x45, 0x00,
>-  0xa0, 0x68, 0x1a, 0x06, 0xa4, 0x00,
>-  0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
>-  0x03, 0x80, 0xe0, 0x3b, 0x40, 0x00,
>-  0x86, 0x21, 0x88, 0x62, 0x09, 0x00,
>-  0x29, 0x0a, 0x42, 0x90, 0x84, 0x80,
>-  0x42, 0x50, 0x94, 0x24, 0x30, 0x80,
>-  0x98, 0x26, 0x09, 0x81, 0x28, 0x00,
>-  0x30, 0x8c, 0x23, 0x08, 0x4a, 0x80
>-};
>-
>-const uint8_t kMaskRandom41_38[228] = {
>-  0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
>-  0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
>-  0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
>-  0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
>-  0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
>-  0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
>-  0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
>-  0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
>-  0x19, 0x86, 0x61, 0x99, 0xe0, 0x00,
>-  0x4c, 0x13, 0x04, 0xc1, 0x91, 0x00,
>-  0x51, 0x14, 0x45, 0x11, 0x45, 0x00,
>-  0xa0, 0x68, 0x1a, 0x06, 0xa4, 0x00,
>-  0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
>-  0x03, 0x80, 0xe0, 0x3b, 0x40, 0x00,
>-  0x86, 0x21, 0x88, 0x62, 0x09, 0x00,
>-  0x29, 0x0a, 0x42, 0x90, 0x84, 0x80,
>-  0x42, 0x50, 0x94, 0x24, 0x30, 0x80,
>-  0x98, 0x26, 0x09, 0x81, 0x28, 0x00,
>-  0x30, 0x8c, 0x23, 0x08, 0x4a, 0x80,
>-  0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
>-  0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
>-  0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
>-  0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
>-  0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
>-  0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
>-  0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
>-  0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
>-  0x19, 0x86, 0x61, 0x99, 0xe0, 0x00,
>-  0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
>-  0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
>-  0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
>-  0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
>-  0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
>-  0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
>-  0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
>-  0x50, 0x54, 0x15, 0x04, 0x50, 0x80,
>-  0x21, 0x7b, 0xf5, 0xa5, 0x65, 0x80,
>-  0xea, 0xc8, 0xbb, 0xd4, 0x5d, 0x00
>-};
>-
>-const uint8_t kMaskRandom41_39[234] = {
>-  0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
>-  0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
>-  0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
>-  0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
>-  0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
>-  0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
>-  0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
>-  0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
>-  0x19, 0x86, 0x61, 0x99, 0xe0, 0x00,
>-  0x4c, 0x13, 0x04, 0xc1, 0x91, 0x00,
>-  0x51, 0x14, 0x45, 0x11, 0x45, 0x00,
>-  0xa0, 0x68, 0x1a, 0x06, 0xa4, 0x00,
>-  0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
>-  0x03, 0x80, 0xe0, 0x3b, 0x40, 0x00,
>-  0x86, 0x21, 0x88, 0x62, 0x09, 0x00,
>-  0x29, 0x0a, 0x42, 0x90, 0x84, 0x80,
>-  0x42, 0x50, 0x94, 0x24, 0x30, 0x80,
>-  0x98, 0x26, 0x09, 0x81, 0x28, 0x00,
>-  0x30, 0x8c, 0x23, 0x08, 0x4a, 0x80,
>-  0x4c, 0x13, 0x04, 0xc1, 0x91, 0x00,
>-  0x51, 0x14, 0x45, 0x11, 0x45, 0x00,
>-  0xa0, 0x68, 0x1a, 0x06, 0xa4, 0x00,
>-  0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
>-  0x03, 0x80, 0xe0, 0x3b, 0x40, 0x00,
>-  0x86, 0x21, 0x88, 0x62, 0x09, 0x00,
>-  0x29, 0x0a, 0x42, 0x90, 0x84, 0x80,
>-  0x42, 0x50, 0x94, 0x24, 0x30, 0x80,
>-  0x98, 0x26, 0x09, 0x81, 0x28, 0x00,
>-  0x30, 0x8c, 0x23, 0x08, 0x4a, 0x80,
>-  0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
>-  0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
>-  0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
>-  0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
>-  0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
>-  0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
>-  0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
>-  0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
>-  0x19, 0x86, 0x61, 0x99, 0xe0, 0x00,
>-  0xf7, 0x8d, 0xa2, 0xa0, 0x33, 0x00
>-};
>-
>-const uint8_t kMaskRandom41_4[24] = {
>-  0xe6, 0x39, 0x8e, 0x63, 0x13, 0x00,
>-  0x33, 0x8c, 0xe3, 0x38, 0xc5, 0x80,
>-  0x98, 0xe6, 0x39, 0x8d, 0x2c, 0x80,
>-  0x2d, 0x4b, 0x52, 0xd4, 0xb2, 0x80
>-};
>-
>-const uint8_t kMaskRandom41_40[240] = {
>-  0x4c, 0x13, 0x04, 0xc1, 0x91, 0x00,
>-  0x51, 0x14, 0x45, 0x11, 0x45, 0x00,
>-  0xa0, 0x68, 0x1a, 0x06, 0xa4, 0x00,
>-  0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
>-  0x03, 0x80, 0xe0, 0x3b, 0x40, 0x00,
>-  0x86, 0x21, 0x88, 0x62, 0x09, 0x00,
>-  0x29, 0x0a, 0x42, 0x90, 0x84, 0x80,
>-  0x42, 0x50, 0x94, 0x24, 0x30, 0x80,
>-  0x98, 0x26, 0x09, 0x81, 0x28, 0x00,
>-  0x30, 0x8c, 0x23, 0x08, 0x4a, 0x80,
>-  0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
>-  0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
>-  0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
>-  0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
>-  0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
>-  0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
>-  0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
>-  0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
>-  0x19, 0x86, 0x61, 0x99, 0xe0, 0x00,
>-  0xf7, 0x8d, 0xa2, 0xa0, 0x33, 0x00,
>-  0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
>-  0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
>-  0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
>-  0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
>-  0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
>-  0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
>-  0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
>-  0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
>-  0x19, 0x86, 0x61, 0x99, 0xe0, 0x00,
>-  0x4c, 0x13, 0x04, 0xc1, 0x91, 0x00,
>-  0x51, 0x14, 0x45, 0x11, 0x45, 0x00,
>-  0xa0, 0x68, 0x1a, 0x06, 0xa4, 0x00,
>-  0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
>-  0x03, 0x80, 0xe0, 0x3b, 0x40, 0x00,
>-  0x86, 0x21, 0x88, 0x62, 0x09, 0x00,
>-  0x29, 0x0a, 0x42, 0x90, 0x84, 0x80,
>-  0x42, 0x50, 0x94, 0x24, 0x30, 0x80,
>-  0x98, 0x26, 0x09, 0x81, 0x28, 0x00,
>-  0x30, 0x8c, 0x23, 0x08, 0x4a, 0x80,
>-  0xe8, 0x07, 0x18, 0x9a, 0x02, 0x00
>-};
>-
>-const uint8_t kMaskRandom41_41[246] = {
>-  0x4c, 0x13, 0x04, 0xc1, 0x91, 0x00,
>-  0x51, 0x14, 0x45, 0x11, 0x45, 0x00,
>-  0xa0, 0x68, 0x1a, 0x06, 0xa4, 0x00,
>-  0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
>-  0x03, 0x80, 0xe0, 0x3b, 0x40, 0x00,
>-  0x86, 0x21, 0x88, 0x62, 0x09, 0x00,
>-  0x29, 0x0a, 0x42, 0x90, 0x84, 0x80,
>-  0x42, 0x50, 0x94, 0x24, 0x30, 0x80,
>-  0x98, 0x26, 0x09, 0x81, 0x28, 0x00,
>-  0x30, 0x8c, 0x23, 0x08, 0x4a, 0x80,
>-  0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
>-  0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
>-  0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
>-  0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
>-  0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
>-  0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
>-  0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
>-  0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
>-  0x19, 0x86, 0x61, 0x99, 0xe0, 0x00,
>-  0xf7, 0x8d, 0xa2, 0xa0, 0x33, 0x00,
>-  0x4c, 0x13, 0x04, 0xc1, 0x91, 0x00,
>-  0x51, 0x14, 0x45, 0x11, 0x45, 0x00,
>-  0xa0, 0x68, 0x1a, 0x06, 0xa4, 0x00,
>-  0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
>-  0x03, 0x80, 0xe0, 0x3b, 0x40, 0x00,
>-  0x86, 0x21, 0x88, 0x62, 0x09, 0x00,
>-  0x29, 0x0a, 0x42, 0x90, 0x84, 0x80,
>-  0x42, 0x50, 0x94, 0x24, 0x30, 0x80,
>-  0x98, 0x26, 0x09, 0x81, 0x28, 0x00,
>-  0x30, 0x8c, 0x23, 0x08, 0x4a, 0x80,
>-  0x4e, 0x13, 0x84, 0xc1, 0x19, 0x00,
>-  0xe3, 0x38, 0xc5, 0x10, 0xcc, 0x80,
>-  0x81, 0xe0, 0x7a, 0x06, 0x64, 0x00,
>-  0x21, 0x48, 0x50, 0x4c, 0x16, 0x00,
>-  0x52, 0x94, 0xa0, 0x3a, 0x02, 0x80,
>-  0xb4, 0x2d, 0x08, 0x62, 0x11, 0x00,
>-  0x26, 0x89, 0xa2, 0x91, 0x01, 0x80,
>-  0x58, 0x56, 0x14, 0x24, 0x2a, 0x00,
>-  0x19, 0x86, 0x69, 0x81, 0xa0, 0x00,
>-  0xf7, 0x8d, 0xa3, 0x08, 0x40, 0x80,
>-  0x2b, 0xea, 0x4d, 0xf4, 0xc1, 0x00
>-};
>-
>-const uint8_t kMaskRandom41_5[30] = {
>-  0xce, 0x33, 0x8c, 0xe3, 0x1b, 0x00,
>-  0x63, 0x98, 0xe6, 0x39, 0x8d, 0x80,
>-  0x98, 0xe5, 0x39, 0x8c, 0x76, 0x80,
>-  0x2b, 0x53, 0x54, 0xd6, 0xb5, 0x00,
>-  0xb4, 0x5c, 0xab, 0x26, 0xca, 0x80
>-};
>-
>-const uint8_t kMaskRandom41_6[36] = {
>-  0x4c, 0x1b, 0x04, 0xc1, 0x91, 0x00,
>-  0x51, 0x34, 0x45, 0x11, 0x45, 0x00,
>-  0x20, 0xe8, 0x32, 0x0e, 0xa0, 0x80,
>-  0x85, 0x41, 0x58, 0x54, 0x12, 0x80,
>-  0x06, 0x86, 0xa0, 0x68, 0x0d, 0x80,
>-  0x9a, 0x21, 0x88, 0xa2, 0x43, 0x00
>-};
>-
>-const uint8_t kMaskRandom41_7[42] = {
>-  0x4e, 0x11, 0x8c, 0x61, 0x19, 0x00,
>-  0x33, 0x2c, 0x03, 0x30, 0x4c, 0x80,
>-  0x10, 0x0e, 0xb1, 0x86, 0x74, 0x00,
>-  0x81, 0x51, 0x54, 0x54, 0x2d, 0x00,
>-  0x24, 0xc4, 0xa1, 0x2d, 0x42, 0x80,
>-  0xd4, 0x23, 0x0b, 0x42, 0x83, 0x00,
>-  0x0c, 0xa2, 0x62, 0x99, 0x21, 0x80
>-};
>-
>-const uint8_t kMaskRandom41_8[48] = {
>-  0x27, 0x09, 0xc0, 0x70, 0xa7, 0x00,
>-  0x89, 0xa2, 0x64, 0x9a, 0x82, 0x80,
>-  0xd0, 0x74, 0x1b, 0x07, 0xa0, 0x00,
>-  0x24, 0xc9, 0x32, 0x4c, 0x5c, 0x00,
>-  0xe2, 0x90, 0xa5, 0x28, 0x0e, 0x80,
>-  0xc6, 0x31, 0x8c, 0x63, 0x18, 0x00,
>-  0x31, 0x8c, 0x63, 0x19, 0x41, 0x80,
>-  0x18, 0xc6, 0x31, 0x8c, 0x70, 0x80
>-};
>-
>-const uint8_t kMaskRandom41_9[54] = {
>-  0x4e, 0x13, 0x84, 0xe1, 0x11, 0x00,
>-  0x62, 0x38, 0xc6, 0x21, 0xa0, 0x80,
>-  0x81, 0xe0, 0x78, 0x0e, 0x94, 0x00,
>-  0xe1, 0x48, 0x5a, 0x15, 0x05, 0x00,
>-  0x13, 0x94, 0xa5, 0x30, 0x06, 0x80,
>-  0xb4, 0x2d, 0x0a, 0x42, 0x43, 0x00,
>-  0x26, 0x89, 0xa1, 0x6a, 0x08, 0x80,
>-  0x58, 0x56, 0x15, 0x84, 0x52, 0x00,
>-  0x49, 0x86, 0x52, 0x98, 0x68, 0x00
>-};
>-
>-const uint8_t kMaskRandom42_1[6] = {
>-  0xff, 0xff, 0xff, 0xff, 0xff, 0xc0
>-};
>-
>-const uint8_t kMaskRandom42_10[60] = {
>-  0x4c, 0x19, 0x12, 0x60, 0xc8, 0x80,
>-  0x51, 0x14, 0x52, 0x88, 0xa2, 0x80,
>-  0xa0, 0x6a, 0x45, 0x03, 0x52, 0x00,
>-  0x04, 0xc1, 0x30, 0x26, 0x09, 0x80,
>-  0x03, 0xb4, 0x00, 0x1d, 0xa0, 0x00,
>-  0x86, 0x20, 0x94, 0x31, 0x04, 0x80,
>-  0x29, 0x08, 0x49, 0x48, 0x42, 0x40,
>-  0x42, 0x43, 0x0a, 0x12, 0x18, 0x40,
>-  0x98, 0x12, 0x84, 0xc0, 0x94, 0x00,
>-  0x30, 0x84, 0xa9, 0x84, 0x25, 0x40
>-};
>-
>-const uint8_t kMaskRandom42_11[66] = {
>-  0xc6, 0x21, 0xa6, 0x31, 0x0d, 0x00,
>-  0x23, 0x88, 0xc9, 0x1c, 0x46, 0x40,
>-  0x1a, 0x45, 0x88, 0xd2, 0x2c, 0x40,
>-  0x24, 0xd3, 0x09, 0x26, 0x98, 0x40,
>-  0x71, 0x10, 0x73, 0x88, 0x83, 0x80,
>-  0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
>-  0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
>-  0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
>-  0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
>-  0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
>-  0xe0, 0x22, 0x97, 0x01, 0x14, 0x80
>-};
>-
>-const uint8_t kMaskRandom42_12[72] = {
>-  0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
>-  0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
>-  0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
>-  0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
>-  0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
>-  0xe0, 0x22, 0x97, 0x01, 0x14, 0x80,
>-  0xc6, 0x21, 0xa6, 0x31, 0x0d, 0x00,
>-  0x23, 0x88, 0xc9, 0x1c, 0x46, 0x40,
>-  0x1a, 0x45, 0x88, 0xd2, 0x2c, 0x40,
>-  0x24, 0xd3, 0x09, 0x26, 0x98, 0x40,
>-  0x71, 0x10, 0x73, 0x88, 0x83, 0x80,
>-  0xa0, 0x65, 0x1d, 0x03, 0x28, 0xc0
>-};
>-
>-const uint8_t kMaskRandom42_13[78] = {
>-  0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
>-  0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
>-  0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
>-  0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
>-  0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
>-  0xe0, 0x22, 0x97, 0x01, 0x14, 0x80,
>-  0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
>-  0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
>-  0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
>-  0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
>-  0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
>-  0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
>-  0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0
>-};
>-
>-const uint8_t kMaskRandom42_14[84] = {
>-  0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
>-  0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
>-  0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
>-  0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
>-  0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
>-  0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
>-  0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0,
>-  0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
>-  0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
>-  0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
>-  0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
>-  0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
>-  0xe0, 0x22, 0x97, 0x01, 0x14, 0x80,
>-  0x4d, 0xd0, 0xc2, 0x6e, 0x86, 0x00
>-};
>-
>-const uint8_t kMaskRandom42_15[90] = {
>-  0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
>-  0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
>-  0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
>-  0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
>-  0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
>-  0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
>-  0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0,
>-  0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
>-  0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
>-  0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
>-  0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
>-  0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
>-  0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
>-  0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
>-  0x50, 0x45, 0x0a, 0x82, 0x28, 0x40
>-};
>-
>-const uint8_t kMaskRandom42_16[96] = {
>-  0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
>-  0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
>-  0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
>-  0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
>-  0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
>-  0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
>-  0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
>-  0x50, 0x45, 0x0a, 0x82, 0x28, 0x40,
>-  0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
>-  0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
>-  0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
>-  0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
>-  0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
>-  0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
>-  0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0,
>-  0x3b, 0xf5, 0x39, 0xdf, 0xa9, 0xc0
>-};
>-
>-const uint8_t kMaskRandom42_17[102] = {
>-  0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
>-  0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
>-  0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
>-  0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
>-  0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
>-  0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
>-  0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
>-  0x50, 0x45, 0x0a, 0x82, 0x28, 0x40,
>-  0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
>-  0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
>-  0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
>-  0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
>-  0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
>-  0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
>-  0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
>-  0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
>-  0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00
>-};
>-
>-const uint8_t kMaskRandom42_18[108] = {
>-  0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
>-  0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
>-  0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
>-  0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
>-  0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
>-  0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
>-  0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
>-  0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
>-  0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00,
>-  0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
>-  0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
>-  0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
>-  0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
>-  0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
>-  0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
>-  0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
>-  0x50, 0x45, 0x0a, 0x82, 0x28, 0x40,
>-  0x5a, 0x56, 0x5a, 0xd2, 0xb2, 0xc0
>-};
>-
>-const uint8_t kMaskRandom42_19[114] = {
>-  0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
>-  0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
>-  0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
>-  0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
>-  0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
>-  0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
>-  0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
>-  0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
>-  0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00,
>-  0x4c, 0x19, 0x12, 0x60, 0xc8, 0x80,
>-  0x51, 0x14, 0x52, 0x88, 0xa2, 0x80,
>-  0xa0, 0x6a, 0x45, 0x03, 0x52, 0x00,
>-  0x04, 0xc1, 0x30, 0x26, 0x09, 0x80,
>-  0x03, 0xb4, 0x00, 0x1d, 0xa0, 0x00,
>-  0x86, 0x20, 0x94, 0x31, 0x04, 0x80,
>-  0x29, 0x08, 0x49, 0x48, 0x42, 0x40,
>-  0x42, 0x43, 0x0a, 0x12, 0x18, 0x40,
>-  0x98, 0x12, 0x84, 0xc0, 0x94, 0x00,
>-  0x30, 0x84, 0xa9, 0x84, 0x25, 0x40
>-};
>-
>-const uint8_t kMaskRandom42_2[12] = {
>-  0xee, 0x3b, 0x37, 0x71, 0xd9, 0x80,
>-  0x99, 0xe6, 0xec, 0xcf, 0x37, 0x40
>-};
>-
>-const uint8_t kMaskRandom42_20[120] = {
>-  0x4c, 0x19, 0x12, 0x60, 0xc8, 0x80,
>-  0x51, 0x14, 0x52, 0x88, 0xa2, 0x80,
>-  0xa0, 0x6a, 0x45, 0x03, 0x52, 0x00,
>-  0x04, 0xc1, 0x30, 0x26, 0x09, 0x80,
>-  0x03, 0xb4, 0x00, 0x1d, 0xa0, 0x00,
>-  0x86, 0x20, 0x94, 0x31, 0x04, 0x80,
>-  0x29, 0x08, 0x49, 0x48, 0x42, 0x40,
>-  0x42, 0x43, 0x0a, 0x12, 0x18, 0x40,
>-  0x98, 0x12, 0x84, 0xc0, 0x94, 0x00,
>-  0x30, 0x84, 0xa9, 0x84, 0x25, 0x40,
>-  0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
>-  0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
>-  0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
>-  0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
>-  0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
>-  0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
>-  0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
>-  0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
>-  0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00,
>-  0x2a, 0x03, 0x31, 0x50, 0x19, 0x80
>-};
>-
>-const uint8_t kMaskRandom42_21[126] = {
>-  0x4c, 0x19, 0x12, 0x60, 0xc8, 0x80,
>-  0x51, 0x14, 0x52, 0x88, 0xa2, 0x80,
>-  0xa0, 0x6a, 0x45, 0x03, 0x52, 0x00,
>-  0x04, 0xc1, 0x30, 0x26, 0x09, 0x80,
>-  0x03, 0xb4, 0x00, 0x1d, 0xa0, 0x00,
>-  0x86, 0x20, 0x94, 0x31, 0x04, 0x80,
>-  0x29, 0x08, 0x49, 0x48, 0x42, 0x40,
>-  0x42, 0x43, 0x0a, 0x12, 0x18, 0x40,
>-  0x98, 0x12, 0x84, 0xc0, 0x94, 0x00,
>-  0x30, 0x84, 0xa9, 0x84, 0x25, 0x40,
>-  0x4c, 0x11, 0x92, 0x60, 0x8c, 0x80,
>-  0x51, 0x0c, 0xca, 0x88, 0x66, 0x40,
>-  0xa0, 0x66, 0x45, 0x03, 0x32, 0x00,
>-  0x04, 0xc1, 0x60, 0x26, 0x0b, 0x00,
>-  0x03, 0xa0, 0x28, 0x1d, 0x01, 0x40,
>-  0x86, 0x21, 0x14, 0x31, 0x08, 0x80,
>-  0x29, 0x10, 0x19, 0x48, 0x80, 0xc0,
>-  0x42, 0x42, 0xa2, 0x12, 0x15, 0x00,
>-  0x98, 0x1a, 0x04, 0xc0, 0xd0, 0x00,
>-  0x30, 0x84, 0x09, 0x84, 0x20, 0x40,
>-  0xdf, 0x4c, 0x16, 0xfa, 0x60, 0x80
>-};
>-
>-const uint8_t kMaskRandom42_22[132] = {
>-  0xc6, 0x21, 0xa6, 0x31, 0x0d, 0x00,
>-  0x23, 0x88, 0xc9, 0x1c, 0x46, 0x40,
>-  0x1a, 0x45, 0x88, 0xd2, 0x2c, 0x40,
>-  0x24, 0xd3, 0x09, 0x26, 0x98, 0x40,
>-  0x71, 0x10, 0x73, 0x88, 0x83, 0x80,
>-  0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
>-  0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
>-  0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
>-  0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
>-  0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
>-  0xe0, 0x22, 0x97, 0x01, 0x14, 0x80,
>-  0x4c, 0x19, 0x12, 0x60, 0xc8, 0x80,
>-  0x51, 0x14, 0x52, 0x88, 0xa2, 0x80,
>-  0xa0, 0x6a, 0x45, 0x03, 0x52, 0x00,
>-  0x04, 0xc1, 0x30, 0x26, 0x09, 0x80,
>-  0x03, 0xb4, 0x00, 0x1d, 0xa0, 0x00,
>-  0x86, 0x20, 0x94, 0x31, 0x04, 0x80,
>-  0x29, 0x08, 0x49, 0x48, 0x42, 0x40,
>-  0x42, 0x43, 0x0a, 0x12, 0x18, 0x40,
>-  0x98, 0x12, 0x84, 0xc0, 0x94, 0x00,
>-  0x30, 0x84, 0xa9, 0x84, 0x25, 0x40,
>-  0xdb, 0x36, 0xb0, 0x33, 0x14, 0x80
>-};
>-
>-const uint8_t kMaskRandom42_23[138] = {
>-  0xc6, 0x21, 0xa6, 0x31, 0x0d, 0x00,
>-  0x23, 0x88, 0xc9, 0x1c, 0x46, 0x40,
>-  0x1a, 0x45, 0x88, 0xd2, 0x2c, 0x40,
>-  0x24, 0xd3, 0x09, 0x26, 0x98, 0x40,
>-  0x71, 0x10, 0x73, 0x88, 0x83, 0x80,
>-  0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
>-  0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
>-  0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
>-  0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
>-  0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
>-  0xe0, 0x22, 0x97, 0x01, 0x14, 0x80,
>-  0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
>-  0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
>-  0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
>-  0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
>-  0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
>-  0xe0, 0x22, 0x97, 0x01, 0x14, 0x80,
>-  0xc6, 0x21, 0xa6, 0x31, 0x0d, 0x00,
>-  0x23, 0x88, 0xc9, 0x1c, 0x46, 0x40,
>-  0x1a, 0x45, 0x88, 0xd2, 0x2c, 0x40,
>-  0x24, 0xd3, 0x09, 0x26, 0x98, 0x40,
>-  0x71, 0x10, 0x73, 0x88, 0x83, 0x80,
>-  0xa0, 0x65, 0x1d, 0x03, 0x28, 0xc0
>-};
>-
>-const uint8_t kMaskRandom42_24[144] = {
>-  0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
>-  0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
>-  0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
>-  0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
>-  0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
>-  0xe0, 0x22, 0x97, 0x01, 0x14, 0x80,
>-  0xc6, 0x21, 0xa6, 0x31, 0x0d, 0x00,
>-  0x23, 0x88, 0xc9, 0x1c, 0x46, 0x40,
>-  0x1a, 0x45, 0x88, 0xd2, 0x2c, 0x40,
>-  0x24, 0xd3, 0x09, 0x26, 0x98, 0x40,
>-  0x71, 0x10, 0x73, 0x88, 0x83, 0x80,
>-  0xa0, 0x65, 0x1d, 0x03, 0x28, 0xc0,
>-  0xc6, 0x21, 0xa6, 0x31, 0x0d, 0x00,
>-  0x23, 0x88, 0xc9, 0x1c, 0x46, 0x40,
>-  0x1a, 0x45, 0x88, 0xd2, 0x2c, 0x40,
>-  0x24, 0xd3, 0x09, 0x26, 0x98, 0x40,
>-  0x71, 0x10, 0x73, 0x88, 0x83, 0x80,
>-  0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
>-  0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
>-  0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
>-  0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
>-  0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
>-  0xe0, 0x22, 0x97, 0x01, 0x14, 0x80,
>-  0x2e, 0x1c, 0x92, 0xbb, 0x07, 0xc0
>-};
>-
>-const uint8_t kMaskRandom42_25[150] = {
>-  0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
>-  0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
>-  0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
>-  0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
>-  0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
>-  0xe0, 0x22, 0x97, 0x01, 0x14, 0x80,
>-  0xc6, 0x21, 0xa6, 0x31, 0x0d, 0x00,
>-  0x23, 0x88, 0xc9, 0x1c, 0x46, 0x40,
>-  0x1a, 0x45, 0x88, 0xd2, 0x2c, 0x40,
>-  0x24, 0xd3, 0x09, 0x26, 0x98, 0x40,
>-  0x71, 0x10, 0x73, 0x88, 0x83, 0x80,
>-  0xa0, 0x65, 0x1d, 0x03, 0x28, 0xc0,
>-  0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
>-  0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
>-  0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
>-  0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
>-  0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
>-  0xe0, 0x22, 0x97, 0x01, 0x14, 0x80,
>-  0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
>-  0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
>-  0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
>-  0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
>-  0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
>-  0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
>-  0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0
>-};
>-
>-const uint8_t kMaskRandom42_26[156] = {
>-  0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
>-  0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
>-  0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
>-  0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
>-  0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
>-  0xe0, 0x22, 0x97, 0x01, 0x14, 0x80,
>-  0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
>-  0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
>-  0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
>-  0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
>-  0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
>-  0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
>-  0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0,
>-  0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
>-  0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
>-  0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
>-  0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
>-  0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
>-  0xe0, 0x22, 0x97, 0x01, 0x14, 0x80,
>-  0xc6, 0x21, 0xa6, 0x31, 0x0d, 0x00,
>-  0x23, 0x88, 0xc9, 0x1c, 0x46, 0x40,
>-  0x1a, 0x45, 0x88, 0xd2, 0x2c, 0x40,
>-  0x24, 0xd3, 0x09, 0x26, 0x98, 0x40,
>-  0x71, 0x10, 0x73, 0x88, 0x83, 0x80,
>-  0xa0, 0x65, 0x1d, 0x03, 0x28, 0xc0,
>-  0xb8, 0x41, 0xed, 0xa3, 0x77, 0xc0
>-};
>-
>-const uint8_t kMaskRandom42_27[162] = {
>-  0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
>-  0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
>-  0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
>-  0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
>-  0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
>-  0xe0, 0x22, 0x97, 0x01, 0x14, 0x80,
>-  0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
>-  0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
>-  0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
>-  0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
>-  0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
>-  0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
>-  0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0,
>-  0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
>-  0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
>-  0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
>-  0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
>-  0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
>-  0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
>-  0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0,
>-  0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
>-  0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
>-  0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
>-  0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
>-  0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
>-  0xe0, 0x22, 0x97, 0x01, 0x14, 0x80,
>-  0x4d, 0xd0, 0xc2, 0x6e, 0x86, 0x00
>-};
>-
>-const uint8_t kMaskRandom42_28[168] = {
>-  0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
>-  0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
>-  0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
>-  0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
>-  0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
>-  0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
>-  0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0,
>-  0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
>-  0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
>-  0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
>-  0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
>-  0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
>-  0xe0, 0x22, 0x97, 0x01, 0x14, 0x80,
>-  0x4d, 0xd0, 0xc2, 0x6e, 0x86, 0x00,
>-  0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
>-  0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
>-  0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
>-  0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
>-  0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
>-  0xe0, 0x22, 0x97, 0x01, 0x14, 0x80,
>-  0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
>-  0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
>-  0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
>-  0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
>-  0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
>-  0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
>-  0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0,
>-  0xc3, 0x3c, 0x56, 0xc2, 0x30, 0x40
>-};
>-
>-const uint8_t kMaskRandom42_29[174] = {
>-  0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
>-  0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
>-  0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
>-  0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
>-  0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
>-  0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
>-  0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0,
>-  0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
>-  0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
>-  0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
>-  0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
>-  0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
>-  0xe0, 0x22, 0x97, 0x01, 0x14, 0x80,
>-  0x4d, 0xd0, 0xc2, 0x6e, 0x86, 0x00,
>-  0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
>-  0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
>-  0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
>-  0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
>-  0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
>-  0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
>-  0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0,
>-  0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
>-  0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
>-  0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
>-  0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
>-  0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
>-  0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
>-  0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
>-  0x50, 0x45, 0x0a, 0x82, 0x28, 0x40
>-};
>-
>-const uint8_t kMaskRandom42_3[18] = {
>-  0xce, 0x32, 0xb6, 0x71, 0x95, 0x80,
>-  0x55, 0xdc, 0x52, 0xae, 0xe2, 0x80,
>-  0xa8, 0xed, 0x8d, 0x47, 0x6c, 0x40
>-};
>-
>-const uint8_t kMaskRandom42_30[180] = {
>-  0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
>-  0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
>-  0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
>-  0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
>-  0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
>-  0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
>-  0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0,
>-  0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
>-  0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
>-  0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
>-  0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
>-  0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
>-  0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
>-  0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
>-  0x50, 0x45, 0x0a, 0x82, 0x28, 0x40,
>-  0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
>-  0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
>-  0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
>-  0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
>-  0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
>-  0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
>-  0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0,
>-  0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
>-  0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
>-  0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
>-  0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
>-  0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
>-  0xe0, 0x22, 0x97, 0x01, 0x14, 0x80,
>-  0x4d, 0xd0, 0xc2, 0x6e, 0x86, 0x00,
>-  0xf5, 0xdd, 0x0d, 0x58, 0xeb, 0x00
>-};
>-
>-const uint8_t kMaskRandom42_31[186] = {
>-  0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
>-  0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
>-  0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
>-  0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
>-  0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
>-  0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
>-  0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0,
>-  0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
>-  0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
>-  0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
>-  0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
>-  0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
>-  0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
>-  0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
>-  0x50, 0x45, 0x0a, 0x82, 0x28, 0x40,
>-  0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
>-  0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
>-  0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
>-  0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
>-  0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
>-  0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
>-  0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
>-  0x50, 0x45, 0x0a, 0x82, 0x28, 0x40,
>-  0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
>-  0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
>-  0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
>-  0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
>-  0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
>-  0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
>-  0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0,
>-  0x3b, 0xf5, 0x39, 0xdf, 0xa9, 0xc0
>-};
>-
>-const uint8_t kMaskRandom42_32[192] = {
>-  0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
>-  0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
>-  0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
>-  0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
>-  0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
>-  0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
>-  0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
>-  0x50, 0x45, 0x0a, 0x82, 0x28, 0x40,
>-  0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
>-  0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
>-  0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
>-  0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
>-  0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
>-  0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
>-  0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0,
>-  0x3b, 0xf5, 0x39, 0xdf, 0xa9, 0xc0,
>-  0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
>-  0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
>-  0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
>-  0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
>-  0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
>-  0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
>-  0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0,
>-  0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
>-  0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
>-  0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
>-  0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
>-  0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
>-  0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
>-  0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
>-  0x50, 0x45, 0x0a, 0x82, 0x28, 0x40,
>-  0xf9, 0x1f, 0xb6, 0xe1, 0x09, 0xc0
>-};
>-
>-const uint8_t kMaskRandom42_33[198] = {
>-  0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
>-  0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
>-  0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
>-  0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
>-  0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
>-  0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
>-  0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
>-  0x50, 0x45, 0x0a, 0x82, 0x28, 0x40,
>-  0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
>-  0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
>-  0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
>-  0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
>-  0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
>-  0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
>-  0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0,
>-  0x3b, 0xf5, 0x39, 0xdf, 0xa9, 0xc0,
>-  0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
>-  0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
>-  0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
>-  0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
>-  0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
>-  0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
>-  0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
>-  0x50, 0x45, 0x0a, 0x82, 0x28, 0x40,
>-  0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
>-  0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
>-  0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
>-  0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
>-  0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
>-  0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
>-  0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
>-  0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
>-  0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00
>-};
>-
>-const uint8_t kMaskRandom42_34[204] = {
>-  0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
>-  0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
>-  0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
>-  0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
>-  0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
>-  0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
>-  0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
>-  0x50, 0x45, 0x0a, 0x82, 0x28, 0x40,
>-  0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
>-  0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
>-  0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
>-  0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
>-  0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
>-  0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
>-  0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
>-  0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
>-  0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00,
>-  0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
>-  0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
>-  0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
>-  0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
>-  0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
>-  0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
>-  0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
>-  0x50, 0x45, 0x0a, 0x82, 0x28, 0x40,
>-  0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
>-  0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
>-  0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
>-  0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
>-  0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
>-  0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
>-  0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0,
>-  0x3b, 0xf5, 0x39, 0xdf, 0xa9, 0xc0,
>-  0xf8, 0xbf, 0xf6, 0x76, 0x1b, 0x80
>-};
>-
>-const uint8_t kMaskRandom42_35[210] = {
>-  0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
>-  0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
>-  0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
>-  0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
>-  0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
>-  0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
>-  0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
>-  0x50, 0x45, 0x0a, 0x82, 0x28, 0x40,
>-  0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
>-  0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
>-  0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
>-  0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
>-  0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
>-  0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
>-  0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
>-  0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
>-  0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00,
>-  0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
>-  0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
>-  0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
>-  0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
>-  0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
>-  0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
>-  0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
>-  0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
>-  0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00,
>-  0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
>-  0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
>-  0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
>-  0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
>-  0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
>-  0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
>-  0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
>-  0x50, 0x45, 0x0a, 0x82, 0x28, 0x40,
>-  0x5a, 0x56, 0x5a, 0xd2, 0xb2, 0xc0
>-};
>-
>-const uint8_t kMaskRandom42_36[216] = {
>-  0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
>-  0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
>-  0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
>-  0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
>-  0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
>-  0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
>-  0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
>-  0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
>-  0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00,
>-  0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
>-  0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
>-  0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
>-  0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
>-  0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
>-  0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
>-  0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
>-  0x50, 0x45, 0x0a, 0x82, 0x28, 0x40,
>-  0x5a, 0x56, 0x5a, 0xd2, 0xb2, 0xc0,
>-  0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
>-  0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
>-  0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
>-  0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
>-  0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
>-  0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
>-  0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
>-  0x50, 0x45, 0x0a, 0x82, 0x28, 0x40,
>-  0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
>-  0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
>-  0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
>-  0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
>-  0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
>-  0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
>-  0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
>-  0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
>-  0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00,
>-  0x57, 0xc7, 0x03, 0xf9, 0xc6, 0x00
>-};
>-
>-const uint8_t kMaskRandom42_37[222] = {
>-  0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
>-  0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
>-  0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
>-  0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
>-  0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
>-  0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
>-  0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
>-  0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
>-  0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00,
>-  0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
>-  0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
>-  0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
>-  0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
>-  0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
>-  0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
>-  0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
>-  0x50, 0x45, 0x0a, 0x82, 0x28, 0x40,
>-  0x5a, 0x56, 0x5a, 0xd2, 0xb2, 0xc0,
>-  0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
>-  0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
>-  0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
>-  0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
>-  0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
>-  0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
>-  0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
>-  0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
>-  0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00,
>-  0x4c, 0x19, 0x12, 0x60, 0xc8, 0x80,
>-  0x51, 0x14, 0x52, 0x88, 0xa2, 0x80,
>-  0xa0, 0x6a, 0x45, 0x03, 0x52, 0x00,
>-  0x04, 0xc1, 0x30, 0x26, 0x09, 0x80,
>-  0x03, 0xb4, 0x00, 0x1d, 0xa0, 0x00,
>-  0x86, 0x20, 0x94, 0x31, 0x04, 0x80,
>-  0x29, 0x08, 0x49, 0x48, 0x42, 0x40,
>-  0x42, 0x43, 0x0a, 0x12, 0x18, 0x40,
>-  0x98, 0x12, 0x84, 0xc0, 0x94, 0x00,
>-  0x30, 0x84, 0xa9, 0x84, 0x25, 0x40
>-};
>-
>-const uint8_t kMaskRandom42_38[228] = {
>-  0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
>-  0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
>-  0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
>-  0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
>-  0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
>-  0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
>-  0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
>-  0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
>-  0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00,
>-  0x4c, 0x19, 0x12, 0x60, 0xc8, 0x80,
>-  0x51, 0x14, 0x52, 0x88, 0xa2, 0x80,
>-  0xa0, 0x6a, 0x45, 0x03, 0x52, 0x00,
>-  0x04, 0xc1, 0x30, 0x26, 0x09, 0x80,
>-  0x03, 0xb4, 0x00, 0x1d, 0xa0, 0x00,
>-  0x86, 0x20, 0x94, 0x31, 0x04, 0x80,
>-  0x29, 0x08, 0x49, 0x48, 0x42, 0x40,
>-  0x42, 0x43, 0x0a, 0x12, 0x18, 0x40,
>-  0x98, 0x12, 0x84, 0xc0, 0x94, 0x00,
>-  0x30, 0x84, 0xa9, 0x84, 0x25, 0x40,
>-  0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
>-  0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
>-  0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
>-  0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
>-  0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
>-  0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
>-  0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
>-  0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
>-  0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00,
>-  0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
>-  0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
>-  0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
>-  0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
>-  0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
>-  0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
>-  0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
>-  0x50, 0x45, 0x0a, 0x82, 0x28, 0x40,
>-  0x5a, 0x56, 0x5a, 0xd2, 0xb2, 0xc0,
>-  0x05, 0x19, 0x55, 0xee, 0xe2, 0xc0
>-};
>-
>-const uint8_t kMaskRandom42_39[234] = {
>-  0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
>-  0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
>-  0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
>-  0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
>-  0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
>-  0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
>-  0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
>-  0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
>-  0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00,
>-  0x4c, 0x19, 0x12, 0x60, 0xc8, 0x80,
>-  0x51, 0x14, 0x52, 0x88, 0xa2, 0x80,
>-  0xa0, 0x6a, 0x45, 0x03, 0x52, 0x00,
>-  0x04, 0xc1, 0x30, 0x26, 0x09, 0x80,
>-  0x03, 0xb4, 0x00, 0x1d, 0xa0, 0x00,
>-  0x86, 0x20, 0x94, 0x31, 0x04, 0x80,
>-  0x29, 0x08, 0x49, 0x48, 0x42, 0x40,
>-  0x42, 0x43, 0x0a, 0x12, 0x18, 0x40,
>-  0x98, 0x12, 0x84, 0xc0, 0x94, 0x00,
>-  0x30, 0x84, 0xa9, 0x84, 0x25, 0x40,
>-  0x4c, 0x19, 0x12, 0x60, 0xc8, 0x80,
>-  0x51, 0x14, 0x52, 0x88, 0xa2, 0x80,
>-  0xa0, 0x6a, 0x45, 0x03, 0x52, 0x00,
>-  0x04, 0xc1, 0x30, 0x26, 0x09, 0x80,
>-  0x03, 0xb4, 0x00, 0x1d, 0xa0, 0x00,
>-  0x86, 0x20, 0x94, 0x31, 0x04, 0x80,
>-  0x29, 0x08, 0x49, 0x48, 0x42, 0x40,
>-  0x42, 0x43, 0x0a, 0x12, 0x18, 0x40,
>-  0x98, 0x12, 0x84, 0xc0, 0x94, 0x00,
>-  0x30, 0x84, 0xa9, 0x84, 0x25, 0x40,
>-  0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
>-  0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
>-  0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
>-  0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
>-  0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
>-  0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
>-  0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
>-  0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
>-  0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00,
>-  0x2a, 0x03, 0x31, 0x50, 0x19, 0x80
>-};
>-
>-const uint8_t kMaskRandom42_4[24] = {
>-  0xe6, 0x31, 0x37, 0x31, 0x89, 0x80,
>-  0x33, 0x8c, 0x59, 0x9c, 0x62, 0xc0,
>-  0x98, 0xd2, 0xcc, 0xc6, 0x96, 0x40,
>-  0x2d, 0x4b, 0x29, 0x6a, 0x59, 0x40
>-};
>-
>-const uint8_t kMaskRandom42_40[240] = {
>-  0x4c, 0x19, 0x12, 0x60, 0xc8, 0x80,
>-  0x51, 0x14, 0x52, 0x88, 0xa2, 0x80,
>-  0xa0, 0x6a, 0x45, 0x03, 0x52, 0x00,
>-  0x04, 0xc1, 0x30, 0x26, 0x09, 0x80,
>-  0x03, 0xb4, 0x00, 0x1d, 0xa0, 0x00,
>-  0x86, 0x20, 0x94, 0x31, 0x04, 0x80,
>-  0x29, 0x08, 0x49, 0x48, 0x42, 0x40,
>-  0x42, 0x43, 0x0a, 0x12, 0x18, 0x40,
>-  0x98, 0x12, 0x84, 0xc0, 0x94, 0x00,
>-  0x30, 0x84, 0xa9, 0x84, 0x25, 0x40,
>-  0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
>-  0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
>-  0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
>-  0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
>-  0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
>-  0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
>-  0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
>-  0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
>-  0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00,
>-  0x2a, 0x03, 0x31, 0x50, 0x19, 0x80,
>-  0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
>-  0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
>-  0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
>-  0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
>-  0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
>-  0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
>-  0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
>-  0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
>-  0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00,
>-  0x4c, 0x19, 0x12, 0x60, 0xc8, 0x80,
>-  0x51, 0x14, 0x52, 0x88, 0xa2, 0x80,
>-  0xa0, 0x6a, 0x45, 0x03, 0x52, 0x00,
>-  0x04, 0xc1, 0x30, 0x26, 0x09, 0x80,
>-  0x03, 0xb4, 0x00, 0x1d, 0xa0, 0x00,
>-  0x86, 0x20, 0x94, 0x31, 0x04, 0x80,
>-  0x29, 0x08, 0x49, 0x48, 0x42, 0x40,
>-  0x42, 0x43, 0x0a, 0x12, 0x18, 0x40,
>-  0x98, 0x12, 0x84, 0xc0, 0x94, 0x00,
>-  0x30, 0x84, 0xa9, 0x84, 0x25, 0x40,
>-  0xf9, 0xdb, 0x5d, 0x7a, 0xd4, 0x40
>-};
>-
>-const uint8_t kMaskRandom42_41[246] = {
>-  0x4c, 0x19, 0x12, 0x60, 0xc8, 0x80,
>-  0x51, 0x14, 0x52, 0x88, 0xa2, 0x80,
>-  0xa0, 0x6a, 0x45, 0x03, 0x52, 0x00,
>-  0x04, 0xc1, 0x30, 0x26, 0x09, 0x80,
>-  0x03, 0xb4, 0x00, 0x1d, 0xa0, 0x00,
>-  0x86, 0x20, 0x94, 0x31, 0x04, 0x80,
>-  0x29, 0x08, 0x49, 0x48, 0x42, 0x40,
>-  0x42, 0x43, 0x0a, 0x12, 0x18, 0x40,
>-  0x98, 0x12, 0x84, 0xc0, 0x94, 0x00,
>-  0x30, 0x84, 0xa9, 0x84, 0x25, 0x40,
>-  0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
>-  0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
>-  0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
>-  0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
>-  0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
>-  0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
>-  0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
>-  0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
>-  0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00,
>-  0x2a, 0x03, 0x31, 0x50, 0x19, 0x80,
>-  0x4c, 0x19, 0x12, 0x60, 0xc8, 0x80,
>-  0x51, 0x14, 0x52, 0x88, 0xa2, 0x80,
>-  0xa0, 0x6a, 0x45, 0x03, 0x52, 0x00,
>-  0x04, 0xc1, 0x30, 0x26, 0x09, 0x80,
>-  0x03, 0xb4, 0x00, 0x1d, 0xa0, 0x00,
>-  0x86, 0x20, 0x94, 0x31, 0x04, 0x80,
>-  0x29, 0x08, 0x49, 0x48, 0x42, 0x40,
>-  0x42, 0x43, 0x0a, 0x12, 0x18, 0x40,
>-  0x98, 0x12, 0x84, 0xc0, 0x94, 0x00,
>-  0x30, 0x84, 0xa9, 0x84, 0x25, 0x40,
>-  0x4c, 0x11, 0x92, 0x60, 0x8c, 0x80,
>-  0x51, 0x0c, 0xca, 0x88, 0x66, 0x40,
>-  0xa0, 0x66, 0x45, 0x03, 0x32, 0x00,
>-  0x04, 0xc1, 0x60, 0x26, 0x0b, 0x00,
>-  0x03, 0xa0, 0x28, 0x1d, 0x01, 0x40,
>-  0x86, 0x21, 0x14, 0x31, 0x08, 0x80,
>-  0x29, 0x10, 0x19, 0x48, 0x80, 0xc0,
>-  0x42, 0x42, 0xa2, 0x12, 0x15, 0x00,
>-  0x98, 0x1a, 0x04, 0xc0, 0xd0, 0x00,
>-  0x30, 0x84, 0x09, 0x84, 0x20, 0x40,
>-  0xdf, 0x4c, 0x16, 0xfa, 0x60, 0x80
>-};
>-
>-const uint8_t kMaskRandom42_42[252] = {
>-  0x4c, 0x19, 0x12, 0x60, 0xc8, 0x80,
>-  0x51, 0x14, 0x52, 0x88, 0xa2, 0x80,
>-  0xa0, 0x6a, 0x45, 0x03, 0x52, 0x00,
>-  0x04, 0xc1, 0x30, 0x26, 0x09, 0x80,
>-  0x03, 0xb4, 0x00, 0x1d, 0xa0, 0x00,
>-  0x86, 0x20, 0x94, 0x31, 0x04, 0x80,
>-  0x29, 0x08, 0x49, 0x48, 0x42, 0x40,
>-  0x42, 0x43, 0x0a, 0x12, 0x18, 0x40,
>-  0x98, 0x12, 0x84, 0xc0, 0x94, 0x00,
>-  0x30, 0x84, 0xa9, 0x84, 0x25, 0x40,
>-  0x4c, 0x11, 0x92, 0x60, 0x8c, 0x80,
>-  0x51, 0x0c, 0xca, 0x88, 0x66, 0x40,
>-  0xa0, 0x66, 0x45, 0x03, 0x32, 0x00,
>-  0x04, 0xc1, 0x60, 0x26, 0x0b, 0x00,
>-  0x03, 0xa0, 0x28, 0x1d, 0x01, 0x40,
>-  0x86, 0x21, 0x14, 0x31, 0x08, 0x80,
>-  0x29, 0x10, 0x19, 0x48, 0x80, 0xc0,
>-  0x42, 0x42, 0xa2, 0x12, 0x15, 0x00,
>-  0x98, 0x1a, 0x04, 0xc0, 0xd0, 0x00,
>-  0x30, 0x84, 0x09, 0x84, 0x20, 0x40,
>-  0xdf, 0x4c, 0x16, 0xfa, 0x60, 0x80,
>-  0x4c, 0x19, 0x12, 0x60, 0xc8, 0x80,
>-  0x51, 0x14, 0x52, 0x88, 0xa2, 0x80,
>-  0xa0, 0x6a, 0x45, 0x03, 0x52, 0x00,
>-  0x04, 0xc1, 0x30, 0x26, 0x09, 0x80,
>-  0x03, 0xb4, 0x00, 0x1d, 0xa0, 0x00,
>-  0x86, 0x20, 0x94, 0x31, 0x04, 0x80,
>-  0x29, 0x08, 0x49, 0x48, 0x42, 0x40,
>-  0x42, 0x43, 0x0a, 0x12, 0x18, 0x40,
>-  0x98, 0x12, 0x84, 0xc0, 0x94, 0x00,
>-  0x30, 0x84, 0xa9, 0x84, 0x25, 0x40,
>-  0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
>-  0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
>-  0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
>-  0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
>-  0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
>-  0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
>-  0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
>-  0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
>-  0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00,
>-  0x2a, 0x03, 0x31, 0x50, 0x19, 0x80,
>-  0xea, 0x9e, 0x23, 0xb3, 0x65, 0x00
>-};
>-
>-const uint8_t kMaskRandom42_5[30] = {
>-  0xce, 0x31, 0xb6, 0x71, 0x8d, 0x80,
>-  0x63, 0x98, 0xdb, 0x1c, 0xc6, 0xc0,
>-  0x98, 0xc7, 0x6c, 0xc6, 0x3b, 0x40,
>-  0x4d, 0x6b, 0x52, 0x6b, 0x5a, 0x80,
>-  0xb2, 0x6c, 0xad, 0x93, 0x65, 0x40
>-};
>-
>-const uint8_t kMaskRandom42_6[36] = {
>-  0x4c, 0x19, 0x12, 0x60, 0xc8, 0x80,
>-  0x51, 0x14, 0x52, 0x88, 0xa2, 0x80,
>-  0x20, 0xea, 0x09, 0x07, 0x50, 0x40,
>-  0x85, 0x41, 0x2c, 0x2a, 0x09, 0x40,
>-  0x06, 0x80, 0xd8, 0x34, 0x06, 0xc0,
>-  0x8a, 0x24, 0x34, 0x51, 0x21, 0x80
>-};
>-
>-const uint8_t kMaskRandom42_7[42] = {
>-  0xc6, 0x11, 0x96, 0x30, 0x8c, 0x80,
>-  0x33, 0x04, 0xc9, 0x98, 0x26, 0x40,
>-  0x18, 0x67, 0x40, 0xc3, 0x3a, 0x00,
>-  0x45, 0x42, 0xd2, 0x2a, 0x16, 0x80,
>-  0x12, 0xd4, 0x28, 0x96, 0xa1, 0x40,
>-  0xb4, 0x28, 0x35, 0xa1, 0x41, 0x80,
>-  0x29, 0x92, 0x19, 0x4c, 0x90, 0xc0
>-};
>-
>-const uint8_t kMaskRandom42_8[48] = {
>-  0x07, 0x0a, 0x70, 0x38, 0x53, 0x80,
>-  0x49, 0xa8, 0x2a, 0x4d, 0x41, 0x40,
>-  0xb0, 0x7a, 0x05, 0x83, 0xd0, 0x00,
>-  0x24, 0xc5, 0xc1, 0x26, 0x2e, 0x00,
>-  0x52, 0x80, 0xea, 0x94, 0x07, 0x40,
>-  0xc6, 0x31, 0x86, 0x31, 0x8c, 0x00,
>-  0x31, 0x94, 0x19, 0x8c, 0xa0, 0xc0,
>-  0x18, 0xc7, 0x08, 0xc6, 0x38, 0x40
>-};
>-
>-const uint8_t kMaskRandom42_9[54] = {
>-  0x4e, 0x11, 0x12, 0x70, 0x88, 0x80,
>-  0x62, 0x1a, 0x0b, 0x10, 0xd0, 0x40,
>-  0x80, 0xe9, 0x44, 0x07, 0x4a, 0x00,
>-  0xa1, 0x50, 0x55, 0x0a, 0x82, 0x80,
>-  0x53, 0x00, 0x6a, 0x98, 0x03, 0x40,
>-  0xa4, 0x24, 0x35, 0x21, 0x21, 0x80,
>-  0x16, 0xa0, 0x88, 0xb5, 0x04, 0x40,
>-  0x58, 0x45, 0x22, 0xc2, 0x29, 0x00,
>-  0x29, 0x86, 0x81, 0x4c, 0x34, 0x00
>-};
>-
>-const uint8_t kMaskRandom43_1[6] = {
>-  0xff, 0xff, 0xff, 0xff, 0xff, 0xe0
>-};
>-
>-const uint8_t kMaskRandom43_10[60] = {
>-  0x4c, 0x19, 0x16, 0x01, 0xc4, 0x40,
>-  0x51, 0x14, 0x51, 0x80, 0x71, 0x40,
>-  0xa0, 0x6a, 0x47, 0x40, 0x38, 0x00,
>-  0x04, 0xc1, 0x34, 0x28, 0x45, 0x40,
>-  0x03, 0xb4, 0x06, 0x84, 0x90, 0x80,
>-  0x86, 0x20, 0x94, 0x32, 0x82, 0x40,
>-  0x29, 0x08, 0x4a, 0x53, 0x40, 0x60,
>-  0x42, 0x43, 0x08, 0x0d, 0x03, 0xa0,
>-  0x98, 0x12, 0x82, 0x64, 0x0c, 0x80,
>-  0x30, 0x84, 0xab, 0x11, 0x20, 0x20
>-};
>-
>-const uint8_t kMaskRandom43_11[66] = {
>-  0xc6, 0x21, 0xa2, 0x32, 0x46, 0x40,
>-  0x23, 0x88, 0xc9, 0x99, 0x33, 0x20,
>-  0x1a, 0x45, 0x8c, 0xc8, 0x99, 0x00,
>-  0x24, 0xd3, 0x08, 0x2c, 0x05, 0x80,
>-  0x71, 0x10, 0x74, 0x05, 0x80, 0xa0,
>-  0x0e, 0x19, 0x14, 0x22, 0x84, 0x40,
>-  0x33, 0x14, 0x52, 0x03, 0x40, 0x60,
>-  0x10, 0xc3, 0x28, 0x54, 0x0a, 0x80,
>-  0x45, 0x68, 0x4b, 0x40, 0x68, 0x00,
>-  0x88, 0x84, 0xa8, 0x81, 0x10, 0x20,
>-  0xe0, 0x22, 0x91, 0x82, 0x30, 0x40
>-};
>-
>-const uint8_t kMaskRandom43_12[72] = {
>-  0x0e, 0x19, 0x13, 0x22, 0x64, 0x40,
>-  0x33, 0x14, 0x52, 0x8a, 0x51, 0x40,
>-  0x10, 0xc3, 0x28, 0x65, 0x0c, 0xa0,
>-  0x45, 0x68, 0x4d, 0x09, 0xa1, 0x20,
>-  0x88, 0x84, 0xa8, 0x95, 0x12, 0xa0,
>-  0xe0, 0x22, 0x94, 0x52, 0x8a, 0x40,
>-  0xc6, 0x21, 0xa4, 0x34, 0x86, 0x80,
>-  0x23, 0x88, 0xc9, 0x19, 0x23, 0x20,
>-  0x1a, 0x45, 0x88, 0xb1, 0x16, 0x20,
>-  0x24, 0xd3, 0x0a, 0x61, 0x4c, 0x20,
>-  0x71, 0x10, 0x72, 0x0e, 0x41, 0xc0,
>-  0xa0, 0x65, 0x1f, 0xa0, 0xc4, 0xe0
>-};
>-
>-const uint8_t kMaskRandom43_13[78] = {
>-  0x0e, 0x19, 0x13, 0x22, 0x64, 0x40,
>-  0x33, 0x14, 0x52, 0x8a, 0x51, 0x40,
>-  0x10, 0xc3, 0x28, 0x65, 0x0c, 0xa0,
>-  0x45, 0x68, 0x4d, 0x09, 0xa1, 0x20,
>-  0x88, 0x84, 0xa8, 0x95, 0x12, 0xa0,
>-  0xe0, 0x22, 0x94, 0x52, 0x8a, 0x40,
>-  0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
>-  0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
>-  0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
>-  0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
>-  0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
>-  0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
>-  0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60
>-};
>-
>-const uint8_t kMaskRandom43_14[84] = {
>-  0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
>-  0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
>-  0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
>-  0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
>-  0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
>-  0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
>-  0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60,
>-  0x0e, 0x19, 0x13, 0x22, 0x64, 0x40,
>-  0x33, 0x14, 0x52, 0x8a, 0x51, 0x40,
>-  0x10, 0xc3, 0x28, 0x65, 0x0c, 0xa0,
>-  0x45, 0x68, 0x4d, 0x09, 0xa1, 0x20,
>-  0x88, 0x84, 0xa8, 0x95, 0x12, 0xa0,
>-  0xe0, 0x22, 0x94, 0x52, 0x8a, 0x40,
>-  0x4d, 0xd0, 0xc6, 0x36, 0x57, 0x40
>-};
>-
>-const uint8_t kMaskRandom43_15[90] = {
>-  0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
>-  0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
>-  0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
>-  0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
>-  0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
>-  0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
>-  0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60,
>-  0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
>-  0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
>-  0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
>-  0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
>-  0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
>-  0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
>-  0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
>-  0x50, 0x45, 0x08, 0xa1, 0x14, 0x20
>-};
>-
>-const uint8_t kMaskRandom43_16[96] = {
>-  0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
>-  0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
>-  0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
>-  0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
>-  0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
>-  0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
>-  0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
>-  0x50, 0x45, 0x08, 0xa1, 0x14, 0x20,
>-  0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
>-  0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
>-  0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
>-  0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
>-  0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
>-  0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
>-  0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60,
>-  0x3b, 0xf5, 0x3c, 0x36, 0x0a, 0x20
>-};
>-
>-const uint8_t kMaskRandom43_17[102] = {
>-  0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
>-  0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
>-  0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
>-  0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
>-  0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
>-  0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
>-  0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
>-  0x50, 0x45, 0x08, 0xa1, 0x14, 0x20,
>-  0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
>-  0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
>-  0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
>-  0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
>-  0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
>-  0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
>-  0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
>-  0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
>-  0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00
>-};
>-
>-const uint8_t kMaskRandom43_18[108] = {
>-  0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
>-  0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
>-  0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
>-  0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
>-  0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
>-  0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
>-  0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
>-  0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
>-  0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00,
>-  0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
>-  0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
>-  0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
>-  0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
>-  0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
>-  0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
>-  0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
>-  0x50, 0x45, 0x08, 0xa1, 0x14, 0x20,
>-  0x5a, 0x56, 0x5f, 0x26, 0xa3, 0x60
>-};
>-
>-const uint8_t kMaskRandom43_19[114] = {
>-  0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
>-  0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
>-  0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
>-  0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
>-  0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
>-  0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
>-  0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
>-  0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
>-  0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00,
>-  0x4c, 0x19, 0x13, 0x22, 0x64, 0x40,
>-  0x51, 0x14, 0x52, 0x8a, 0x51, 0x40,
>-  0xa0, 0x6a, 0x45, 0x48, 0xa9, 0x00,
>-  0x04, 0xc1, 0x30, 0x26, 0x04, 0xc0,
>-  0x03, 0xb4, 0x06, 0x80, 0xd0, 0x00,
>-  0x86, 0x20, 0x94, 0x12, 0x82, 0x40,
>-  0x29, 0x08, 0x49, 0x09, 0x21, 0x20,
>-  0x42, 0x43, 0x08, 0x61, 0x0c, 0x20,
>-  0x98, 0x12, 0x82, 0x50, 0x4a, 0x00,
>-  0x30, 0x84, 0xa8, 0x95, 0x12, 0xa0
>-};
>-
>-const uint8_t kMaskRandom43_2[12] = {
>-  0xee, 0x3b, 0x37, 0x66, 0xec, 0xc0,
>-  0x99, 0xe6, 0xec, 0xdd, 0x9b, 0xa0
>-};
>-
>-const uint8_t kMaskRandom43_20[120] = {
>-  0x4c, 0x19, 0x13, 0x22, 0x64, 0x40,
>-  0x51, 0x14, 0x52, 0x8a, 0x51, 0x40,
>-  0xa0, 0x6a, 0x45, 0x48, 0xa9, 0x00,
>-  0x04, 0xc1, 0x30, 0x26, 0x04, 0xc0,
>-  0x03, 0xb4, 0x06, 0x80, 0xd0, 0x00,
>-  0x86, 0x20, 0x94, 0x12, 0x82, 0x40,
>-  0x29, 0x08, 0x49, 0x09, 0x21, 0x20,
>-  0x42, 0x43, 0x08, 0x61, 0x0c, 0x20,
>-  0x98, 0x12, 0x82, 0x50, 0x4a, 0x00,
>-  0x30, 0x84, 0xa8, 0x95, 0x12, 0xa0,
>-  0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
>-  0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
>-  0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
>-  0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
>-  0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
>-  0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
>-  0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
>-  0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
>-  0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00,
>-  0x2a, 0x03, 0x31, 0xda, 0x46, 0x20
>-};
>-
>-const uint8_t kMaskRandom43_21[126] = {
>-  0x4c, 0x19, 0x13, 0x22, 0x64, 0x40,
>-  0x51, 0x14, 0x52, 0x8a, 0x51, 0x40,
>-  0xa0, 0x6a, 0x45, 0x48, 0xa9, 0x00,
>-  0x04, 0xc1, 0x30, 0x26, 0x04, 0xc0,
>-  0x03, 0xb4, 0x06, 0x80, 0xd0, 0x00,
>-  0x86, 0x20, 0x94, 0x12, 0x82, 0x40,
>-  0x29, 0x08, 0x49, 0x09, 0x21, 0x20,
>-  0x42, 0x43, 0x08, 0x61, 0x0c, 0x20,
>-  0x98, 0x12, 0x82, 0x50, 0x4a, 0x00,
>-  0x30, 0x84, 0xa8, 0x95, 0x12, 0xa0,
>-  0x4c, 0x11, 0x92, 0x32, 0x46, 0x40,
>-  0x51, 0x0c, 0xc9, 0x99, 0x33, 0x20,
>-  0xa0, 0x66, 0x44, 0xc8, 0x99, 0x00,
>-  0x04, 0xc1, 0x60, 0x2c, 0x05, 0x80,
>-  0x03, 0xa0, 0x2c, 0x05, 0x80, 0xa0,
>-  0x86, 0x21, 0x14, 0x22, 0x84, 0x40,
>-  0x29, 0x10, 0x1a, 0x03, 0x40, 0x60,
>-  0x42, 0x42, 0xa0, 0x54, 0x0a, 0x80,
>-  0x98, 0x1a, 0x03, 0x40, 0x68, 0x00,
>-  0x30, 0x84, 0x08, 0x81, 0x10, 0x20,
>-  0xdf, 0x4c, 0x11, 0x82, 0x30, 0x40
>-};
>-
>-const uint8_t kMaskRandom43_22[132] = {
>-  0xc6, 0x21, 0xa2, 0x32, 0x46, 0x40,
>-  0x23, 0x88, 0xc9, 0x99, 0x33, 0x20,
>-  0x1a, 0x45, 0x8c, 0xc8, 0x99, 0x00,
>-  0x24, 0xd3, 0x08, 0x2c, 0x05, 0x80,
>-  0x71, 0x10, 0x74, 0x05, 0x80, 0xa0,
>-  0x0e, 0x19, 0x14, 0x22, 0x84, 0x40,
>-  0x33, 0x14, 0x52, 0x03, 0x40, 0x60,
>-  0x10, 0xc3, 0x28, 0x54, 0x0a, 0x80,
>-  0x45, 0x68, 0x4b, 0x40, 0x68, 0x00,
>-  0x88, 0x84, 0xa8, 0x81, 0x10, 0x20,
>-  0xe0, 0x22, 0x91, 0x82, 0x30, 0x40,
>-  0x4c, 0x19, 0x16, 0x01, 0xc4, 0x40,
>-  0x51, 0x14, 0x51, 0x80, 0x71, 0x40,
>-  0xa0, 0x6a, 0x47, 0x40, 0x38, 0x00,
>-  0x04, 0xc1, 0x34, 0x28, 0x45, 0x40,
>-  0x03, 0xb4, 0x06, 0x84, 0x90, 0x80,
>-  0x86, 0x20, 0x94, 0x32, 0x82, 0x40,
>-  0x29, 0x08, 0x4a, 0x53, 0x40, 0x60,
>-  0x42, 0x43, 0x08, 0x0d, 0x03, 0xa0,
>-  0x98, 0x12, 0x82, 0x64, 0x0c, 0x80,
>-  0x30, 0x84, 0xab, 0x11, 0x20, 0x20,
>-  0xfe, 0x2c, 0x85, 0xcc, 0x24, 0x80
>-};
>-
>-const uint8_t kMaskRandom43_23[138] = {
>-  0xc6, 0x21, 0xa2, 0x32, 0x46, 0x40,
>-  0x23, 0x88, 0xc9, 0x99, 0x33, 0x20,
>-  0x1a, 0x45, 0x8c, 0xc8, 0x99, 0x00,
>-  0x24, 0xd3, 0x08, 0x2c, 0x05, 0x80,
>-  0x71, 0x10, 0x74, 0x05, 0x80, 0xa0,
>-  0x0e, 0x19, 0x14, 0x22, 0x84, 0x40,
>-  0x33, 0x14, 0x52, 0x03, 0x40, 0x60,
>-  0x10, 0xc3, 0x28, 0x54, 0x0a, 0x80,
>-  0x45, 0x68, 0x4b, 0x40, 0x68, 0x00,
>-  0x88, 0x84, 0xa8, 0x81, 0x10, 0x20,
>-  0xe0, 0x22, 0x91, 0x82, 0x30, 0x40,
>-  0x0e, 0x19, 0x13, 0x22, 0x64, 0x40,
>-  0x33, 0x14, 0x52, 0x8a, 0x51, 0x40,
>-  0x10, 0xc3, 0x28, 0x65, 0x0c, 0xa0,
>-  0x45, 0x68, 0x4d, 0x09, 0xa1, 0x20,
>-  0x88, 0x84, 0xa8, 0x95, 0x12, 0xa0,
>-  0xe0, 0x22, 0x94, 0x52, 0x8a, 0x40,
>-  0xc6, 0x21, 0xa4, 0x34, 0x86, 0x80,
>-  0x23, 0x88, 0xc9, 0x19, 0x23, 0x20,
>-  0x1a, 0x45, 0x88, 0xb1, 0x16, 0x20,
>-  0x24, 0xd3, 0x0a, 0x61, 0x4c, 0x20,
>-  0x71, 0x10, 0x72, 0x0e, 0x41, 0xc0,
>-  0xa0, 0x65, 0x1f, 0xa0, 0xc4, 0xe0
>-};
>-
>-const uint8_t kMaskRandom43_24[144] = {
>-  0x0e, 0x19, 0x13, 0x22, 0x64, 0x40,
>-  0x33, 0x14, 0x52, 0x8a, 0x51, 0x40,
>-  0x10, 0xc3, 0x28, 0x65, 0x0c, 0xa0,
>-  0x45, 0x68, 0x4d, 0x09, 0xa1, 0x20,
>-  0x88, 0x84, 0xa8, 0x95, 0x12, 0xa0,
>-  0xe0, 0x22, 0x94, 0x52, 0x8a, 0x40,
>-  0xc6, 0x21, 0xa4, 0x34, 0x86, 0x80,
>-  0x23, 0x88, 0xc9, 0x19, 0x23, 0x20,
>-  0x1a, 0x45, 0x88, 0xb1, 0x16, 0x20,
>-  0x24, 0xd3, 0x0a, 0x61, 0x4c, 0x20,
>-  0x71, 0x10, 0x72, 0x0e, 0x41, 0xc0,
>-  0xa0, 0x65, 0x1f, 0xa0, 0xc4, 0xe0,
>-  0xc6, 0x21, 0xa2, 0x32, 0x46, 0x40,
>-  0x23, 0x88, 0xc9, 0x99, 0x33, 0x20,
>-  0x1a, 0x45, 0x8c, 0xc8, 0x99, 0x00,
>-  0x24, 0xd3, 0x08, 0x2c, 0x05, 0x80,
>-  0x71, 0x10, 0x74, 0x05, 0x80, 0xa0,
>-  0x0e, 0x19, 0x14, 0x22, 0x84, 0x40,
>-  0x33, 0x14, 0x52, 0x03, 0x40, 0x60,
>-  0x10, 0xc3, 0x28, 0x54, 0x0a, 0x80,
>-  0x45, 0x68, 0x4b, 0x40, 0x68, 0x00,
>-  0x88, 0x84, 0xa8, 0x81, 0x10, 0x20,
>-  0xe0, 0x22, 0x91, 0x82, 0x30, 0x40,
>-  0xf9, 0xb1, 0x26, 0x6c, 0x51, 0xe0
>-};
>-
>-const uint8_t kMaskRandom43_25[150] = {
>-  0x0e, 0x19, 0x13, 0x22, 0x64, 0x40,
>-  0x33, 0x14, 0x52, 0x8a, 0x51, 0x40,
>-  0x10, 0xc3, 0x28, 0x65, 0x0c, 0xa0,
>-  0x45, 0x68, 0x4d, 0x09, 0xa1, 0x20,
>-  0x88, 0x84, 0xa8, 0x95, 0x12, 0xa0,
>-  0xe0, 0x22, 0x94, 0x52, 0x8a, 0x40,
>-  0xc6, 0x21, 0xa4, 0x34, 0x86, 0x80,
>-  0x23, 0x88, 0xc9, 0x19, 0x23, 0x20,
>-  0x1a, 0x45, 0x88, 0xb1, 0x16, 0x20,
>-  0x24, 0xd3, 0x0a, 0x61, 0x4c, 0x20,
>-  0x71, 0x10, 0x72, 0x0e, 0x41, 0xc0,
>-  0xa0, 0x65, 0x1f, 0xa0, 0xc4, 0xe0,
>-  0x0e, 0x19, 0x13, 0x22, 0x64, 0x40,
>-  0x33, 0x14, 0x52, 0x8a, 0x51, 0x40,
>-  0x10, 0xc3, 0x28, 0x65, 0x0c, 0xa0,
>-  0x45, 0x68, 0x4d, 0x09, 0xa1, 0x20,
>-  0x88, 0x84, 0xa8, 0x95, 0x12, 0xa0,
>-  0xe0, 0x22, 0x94, 0x52, 0x8a, 0x40,
>-  0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
>-  0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
>-  0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
>-  0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
>-  0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
>-  0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
>-  0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60
>-};
>-
>-const uint8_t kMaskRandom43_26[156] = {
>-  0x0e, 0x19, 0x13, 0x22, 0x64, 0x40,
>-  0x33, 0x14, 0x52, 0x8a, 0x51, 0x40,
>-  0x10, 0xc3, 0x28, 0x65, 0x0c, 0xa0,
>-  0x45, 0x68, 0x4d, 0x09, 0xa1, 0x20,
>-  0x88, 0x84, 0xa8, 0x95, 0x12, 0xa0,
>-  0xe0, 0x22, 0x94, 0x52, 0x8a, 0x40,
>-  0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
>-  0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
>-  0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
>-  0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
>-  0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
>-  0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
>-  0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60,
>-  0x0e, 0x19, 0x13, 0x22, 0x64, 0x40,
>-  0x33, 0x14, 0x52, 0x8a, 0x51, 0x40,
>-  0x10, 0xc3, 0x28, 0x65, 0x0c, 0xa0,
>-  0x45, 0x68, 0x4d, 0x09, 0xa1, 0x20,
>-  0x88, 0x84, 0xa8, 0x95, 0x12, 0xa0,
>-  0xe0, 0x22, 0x94, 0x52, 0x8a, 0x40,
>-  0xc6, 0x21, 0xa4, 0x34, 0x86, 0x80,
>-  0x23, 0x88, 0xc9, 0x19, 0x23, 0x20,
>-  0x1a, 0x45, 0x88, 0xb1, 0x16, 0x20,
>-  0x24, 0xd3, 0x0a, 0x61, 0x4c, 0x20,
>-  0x71, 0x10, 0x72, 0x0e, 0x41, 0xc0,
>-  0xa0, 0x65, 0x1f, 0xa0, 0xc4, 0xe0,
>-  0xef, 0x84, 0x77, 0xca, 0x0d, 0x40
>-};
>-
>-const uint8_t kMaskRandom43_27[162] = {
>-  0x0e, 0x19, 0x13, 0x22, 0x64, 0x40,
>-  0x33, 0x14, 0x52, 0x8a, 0x51, 0x40,
>-  0x10, 0xc3, 0x28, 0x65, 0x0c, 0xa0,
>-  0x45, 0x68, 0x4d, 0x09, 0xa1, 0x20,
>-  0x88, 0x84, 0xa8, 0x95, 0x12, 0xa0,
>-  0xe0, 0x22, 0x94, 0x52, 0x8a, 0x40,
>-  0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
>-  0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
>-  0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
>-  0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
>-  0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
>-  0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
>-  0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60,
>-  0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
>-  0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
>-  0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
>-  0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
>-  0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
>-  0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
>-  0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60,
>-  0x0e, 0x19, 0x13, 0x22, 0x64, 0x40,
>-  0x33, 0x14, 0x52, 0x8a, 0x51, 0x40,
>-  0x10, 0xc3, 0x28, 0x65, 0x0c, 0xa0,
>-  0x45, 0x68, 0x4d, 0x09, 0xa1, 0x20,
>-  0x88, 0x84, 0xa8, 0x95, 0x12, 0xa0,
>-  0xe0, 0x22, 0x94, 0x52, 0x8a, 0x40,
>-  0x4d, 0xd0, 0xc6, 0x36, 0x57, 0x40
>-};
>-
>-const uint8_t kMaskRandom43_28[168] = {
>-  0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
>-  0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
>-  0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
>-  0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
>-  0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
>-  0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
>-  0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60,
>-  0x0e, 0x19, 0x13, 0x22, 0x64, 0x40,
>-  0x33, 0x14, 0x52, 0x8a, 0x51, 0x40,
>-  0x10, 0xc3, 0x28, 0x65, 0x0c, 0xa0,
>-  0x45, 0x68, 0x4d, 0x09, 0xa1, 0x20,
>-  0x88, 0x84, 0xa8, 0x95, 0x12, 0xa0,
>-  0xe0, 0x22, 0x94, 0x52, 0x8a, 0x40,
>-  0x4d, 0xd0, 0xc6, 0x36, 0x57, 0x40,
>-  0x0e, 0x19, 0x13, 0x22, 0x64, 0x40,
>-  0x33, 0x14, 0x52, 0x8a, 0x51, 0x40,
>-  0x10, 0xc3, 0x28, 0x65, 0x0c, 0xa0,
>-  0x45, 0x68, 0x4d, 0x09, 0xa1, 0x20,
>-  0x88, 0x84, 0xa8, 0x95, 0x12, 0xa0,
>-  0xe0, 0x22, 0x94, 0x52, 0x8a, 0x40,
>-  0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
>-  0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
>-  0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
>-  0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
>-  0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
>-  0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
>-  0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60,
>-  0x16, 0xc9, 0x53, 0x1e, 0xc4, 0x00
>-};
>-
>-const uint8_t kMaskRandom43_29[174] = {
>-  0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
>-  0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
>-  0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
>-  0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
>-  0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
>-  0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
>-  0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60,
>-  0x0e, 0x19, 0x13, 0x22, 0x64, 0x40,
>-  0x33, 0x14, 0x52, 0x8a, 0x51, 0x40,
>-  0x10, 0xc3, 0x28, 0x65, 0x0c, 0xa0,
>-  0x45, 0x68, 0x4d, 0x09, 0xa1, 0x20,
>-  0x88, 0x84, 0xa8, 0x95, 0x12, 0xa0,
>-  0xe0, 0x22, 0x94, 0x52, 0x8a, 0x40,
>-  0x4d, 0xd0, 0xc6, 0x36, 0x57, 0x40,
>-  0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
>-  0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
>-  0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
>-  0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
>-  0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
>-  0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
>-  0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60,
>-  0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
>-  0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
>-  0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
>-  0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
>-  0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
>-  0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
>-  0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
>-  0x50, 0x45, 0x08, 0xa1, 0x14, 0x20
>-};
>-
>-const uint8_t kMaskRandom43_3[18] = {
>-  0xce, 0x32, 0xb6, 0x56, 0xca, 0xc0,
>-  0x55, 0xdc, 0x57, 0x8a, 0xf1, 0x40,
>-  0xa8, 0xed, 0x8d, 0xb1, 0xae, 0x20
>-};
>-
>-const uint8_t kMaskRandom43_30[180] = {
>-  0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
>-  0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
>-  0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
>-  0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
>-  0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
>-  0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
>-  0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60,
>-  0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
>-  0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
>-  0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
>-  0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
>-  0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
>-  0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
>-  0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
>-  0x50, 0x45, 0x08, 0xa1, 0x14, 0x20,
>-  0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
>-  0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
>-  0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
>-  0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
>-  0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
>-  0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
>-  0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60,
>-  0x0e, 0x19, 0x13, 0x22, 0x64, 0x40,
>-  0x33, 0x14, 0x52, 0x8a, 0x51, 0x40,
>-  0x10, 0xc3, 0x28, 0x65, 0x0c, 0xa0,
>-  0x45, 0x68, 0x4d, 0x09, 0xa1, 0x20,
>-  0x88, 0x84, 0xa8, 0x95, 0x12, 0xa0,
>-  0xe0, 0x22, 0x94, 0x52, 0x8a, 0x40,
>-  0x4d, 0xd0, 0xc6, 0x36, 0x57, 0x40,
>-  0x79, 0x4a, 0x8f, 0x42, 0x79, 0x40
>-};
>-
>-const uint8_t kMaskRandom43_31[186] = {
>-  0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
>-  0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
>-  0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
>-  0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
>-  0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
>-  0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
>-  0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60,
>-  0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
>-  0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
>-  0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
>-  0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
>-  0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
>-  0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
>-  0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
>-  0x50, 0x45, 0x08, 0xa1, 0x14, 0x20,
>-  0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
>-  0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
>-  0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
>-  0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
>-  0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
>-  0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
>-  0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
>-  0x50, 0x45, 0x08, 0xa1, 0x14, 0x20,
>-  0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
>-  0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
>-  0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
>-  0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
>-  0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
>-  0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
>-  0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60,
>-  0x3b, 0xf5, 0x3c, 0x36, 0x0a, 0x20
>-};
>-
>-const uint8_t kMaskRandom43_32[192] = {
>-  0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
>-  0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
>-  0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
>-  0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
>-  0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
>-  0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
>-  0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
>-  0x50, 0x45, 0x08, 0xa1, 0x14, 0x20,
>-  0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
>-  0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
>-  0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
>-  0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
>-  0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
>-  0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
>-  0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60,
>-  0x3b, 0xf5, 0x3c, 0x36, 0x0a, 0x20,
>-  0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
>-  0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
>-  0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
>-  0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
>-  0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
>-  0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
>-  0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60,
>-  0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
>-  0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
>-  0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
>-  0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
>-  0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
>-  0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
>-  0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
>-  0x50, 0x45, 0x08, 0xa1, 0x14, 0x20,
>-  0xd1, 0xd1, 0x11, 0xa4, 0xed, 0xc0
>-};
>-
>-const uint8_t kMaskRandom43_33[198] = {
>-  0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
>-  0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
>-  0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
>-  0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
>-  0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
>-  0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
>-  0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
>-  0x50, 0x45, 0x08, 0xa1, 0x14, 0x20,
>-  0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
>-  0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
>-  0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
>-  0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
>-  0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
>-  0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
>-  0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60,
>-  0x3b, 0xf5, 0x3c, 0x36, 0x0a, 0x20,
>-  0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
>-  0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
>-  0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
>-  0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
>-  0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
>-  0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
>-  0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
>-  0x50, 0x45, 0x08, 0xa1, 0x14, 0x20,
>-  0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
>-  0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
>-  0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
>-  0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
>-  0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
>-  0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
>-  0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
>-  0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
>-  0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00
>-};
>-
>-const uint8_t kMaskRandom43_34[204] = {
>-  0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
>-  0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
>-  0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
>-  0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
>-  0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
>-  0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
>-  0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
>-  0x50, 0x45, 0x08, 0xa1, 0x14, 0x20,
>-  0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
>-  0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
>-  0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
>-  0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
>-  0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
>-  0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
>-  0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
>-  0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
>-  0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00,
>-  0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
>-  0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
>-  0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
>-  0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
>-  0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
>-  0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
>-  0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
>-  0x50, 0x45, 0x08, 0xa1, 0x14, 0x20,
>-  0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
>-  0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
>-  0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
>-  0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
>-  0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
>-  0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
>-  0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60,
>-  0x3b, 0xf5, 0x3c, 0x36, 0x0a, 0x20,
>-  0x76, 0x81, 0x4d, 0x33, 0x66, 0x00
>-};
>-
>-const uint8_t kMaskRandom43_35[210] = {
>-  0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
>-  0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
>-  0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
>-  0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
>-  0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
>-  0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
>-  0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
>-  0x50, 0x45, 0x08, 0xa1, 0x14, 0x20,
>-  0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
>-  0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
>-  0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
>-  0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
>-  0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
>-  0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
>-  0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
>-  0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
>-  0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00,
>-  0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
>-  0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
>-  0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
>-  0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
>-  0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
>-  0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
>-  0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
>-  0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
>-  0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00,
>-  0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
>-  0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
>-  0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
>-  0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
>-  0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
>-  0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
>-  0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
>-  0x50, 0x45, 0x08, 0xa1, 0x14, 0x20,
>-  0x5a, 0x56, 0x5f, 0x26, 0xa3, 0x60
>-};
>-
>-const uint8_t kMaskRandom43_36[216] = {
>-  0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
>-  0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
>-  0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
>-  0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
>-  0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
>-  0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
>-  0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
>-  0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
>-  0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00,
>-  0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
>-  0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
>-  0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
>-  0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
>-  0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
>-  0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
>-  0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
>-  0x50, 0x45, 0x08, 0xa1, 0x14, 0x20,
>-  0x5a, 0x56, 0x5f, 0x26, 0xa3, 0x60,
>-  0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
>-  0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
>-  0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
>-  0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
>-  0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
>-  0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
>-  0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
>-  0x50, 0x45, 0x08, 0xa1, 0x14, 0x20,
>-  0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
>-  0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
>-  0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
>-  0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
>-  0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
>-  0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
>-  0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
>-  0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
>-  0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00,
>-  0xa3, 0x85, 0x0a, 0xb5, 0x11, 0x60
>-};
>-
>-const uint8_t kMaskRandom43_37[222] = {
>-  0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
>-  0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
>-  0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
>-  0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
>-  0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
>-  0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
>-  0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
>-  0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
>-  0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00,
>-  0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
>-  0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
>-  0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
>-  0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
>-  0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
>-  0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
>-  0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
>-  0x50, 0x45, 0x08, 0xa1, 0x14, 0x20,
>-  0x5a, 0x56, 0x5f, 0x26, 0xa3, 0x60,
>-  0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
>-  0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
>-  0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
>-  0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
>-  0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
>-  0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
>-  0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
>-  0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
>-  0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00,
>-  0x4c, 0x19, 0x13, 0x22, 0x64, 0x40,
>-  0x51, 0x14, 0x52, 0x8a, 0x51, 0x40,
>-  0xa0, 0x6a, 0x45, 0x48, 0xa9, 0x00,
>-  0x04, 0xc1, 0x30, 0x26, 0x04, 0xc0,
>-  0x03, 0xb4, 0x06, 0x80, 0xd0, 0x00,
>-  0x86, 0x20, 0x94, 0x12, 0x82, 0x40,
>-  0x29, 0x08, 0x49, 0x09, 0x21, 0x20,
>-  0x42, 0x43, 0x08, 0x61, 0x0c, 0x20,
>-  0x98, 0x12, 0x82, 0x50, 0x4a, 0x00,
>-  0x30, 0x84, 0xa8, 0x95, 0x12, 0xa0
>-};
>-
>-const uint8_t kMaskRandom43_38[228] = {
>-  0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
>-  0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
>-  0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
>-  0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
>-  0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
>-  0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
>-  0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
>-  0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
>-  0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00,
>-  0x4c, 0x19, 0x13, 0x22, 0x64, 0x40,
>-  0x51, 0x14, 0x52, 0x8a, 0x51, 0x40,
>-  0xa0, 0x6a, 0x45, 0x48, 0xa9, 0x00,
>-  0x04, 0xc1, 0x30, 0x26, 0x04, 0xc0,
>-  0x03, 0xb4, 0x06, 0x80, 0xd0, 0x00,
>-  0x86, 0x20, 0x94, 0x12, 0x82, 0x40,
>-  0x29, 0x08, 0x49, 0x09, 0x21, 0x20,
>-  0x42, 0x43, 0x08, 0x61, 0x0c, 0x20,
>-  0x98, 0x12, 0x82, 0x50, 0x4a, 0x00,
>-  0x30, 0x84, 0xa8, 0x95, 0x12, 0xa0,
>-  0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
>-  0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
>-  0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
>-  0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
>-  0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
>-  0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
>-  0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
>-  0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
>-  0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00,
>-  0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
>-  0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
>-  0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
>-  0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
>-  0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
>-  0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
>-  0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
>-  0x50, 0x45, 0x08, 0xa1, 0x14, 0x20,
>-  0x5a, 0x56, 0x5f, 0x26, 0xa3, 0x60,
>-  0x9a, 0x16, 0x97, 0x21, 0xb9, 0x80
>-};
>-
>-const uint8_t kMaskRandom43_39[234] = {
>-  0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
>-  0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
>-  0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
>-  0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
>-  0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
>-  0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
>-  0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
>-  0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
>-  0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00,
>-  0x4c, 0x19, 0x13, 0x22, 0x64, 0x40,
>-  0x51, 0x14, 0x52, 0x8a, 0x51, 0x40,
>-  0xa0, 0x6a, 0x45, 0x48, 0xa9, 0x00,
>-  0x04, 0xc1, 0x30, 0x26, 0x04, 0xc0,
>-  0x03, 0xb4, 0x06, 0x80, 0xd0, 0x00,
>-  0x86, 0x20, 0x94, 0x12, 0x82, 0x40,
>-  0x29, 0x08, 0x49, 0x09, 0x21, 0x20,
>-  0x42, 0x43, 0x08, 0x61, 0x0c, 0x20,
>-  0x98, 0x12, 0x82, 0x50, 0x4a, 0x00,
>-  0x30, 0x84, 0xa8, 0x95, 0x12, 0xa0,
>-  0x4c, 0x19, 0x13, 0x22, 0x64, 0x40,
>-  0x51, 0x14, 0x52, 0x8a, 0x51, 0x40,
>-  0xa0, 0x6a, 0x45, 0x48, 0xa9, 0x00,
>-  0x04, 0xc1, 0x30, 0x26, 0x04, 0xc0,
>-  0x03, 0xb4, 0x06, 0x80, 0xd0, 0x00,
>-  0x86, 0x20, 0x94, 0x12, 0x82, 0x40,
>-  0x29, 0x08, 0x49, 0x09, 0x21, 0x20,
>-  0x42, 0x43, 0x08, 0x61, 0x0c, 0x20,
>-  0x98, 0x12, 0x82, 0x50, 0x4a, 0x00,
>-  0x30, 0x84, 0xa8, 0x95, 0x12, 0xa0,
>-  0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
>-  0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
>-  0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
>-  0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
>-  0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
>-  0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
>-  0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
>-  0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
>-  0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00,
>-  0x2a, 0x03, 0x31, 0xda, 0x46, 0x20
>-};
>-
>-const uint8_t kMaskRandom43_4[24] = {
>-  0xe6, 0x31, 0x36, 0x26, 0xc4, 0xc0,
>-  0x33, 0x8c, 0x59, 0x8b, 0x31, 0x60,
>-  0x98, 0xd2, 0xca, 0x59, 0x4b, 0x20,
>-  0x2d, 0x4b, 0x29, 0x65, 0x2c, 0xa0
>-};
>-
>-const uint8_t kMaskRandom43_40[240] = {
>-  0x4c, 0x19, 0x13, 0x22, 0x64, 0x40,
>-  0x51, 0x14, 0x52, 0x8a, 0x51, 0x40,
>-  0xa0, 0x6a, 0x45, 0x48, 0xa9, 0x00,
>-  0x04, 0xc1, 0x30, 0x26, 0x04, 0xc0,
>-  0x03, 0xb4, 0x06, 0x80, 0xd0, 0x00,
>-  0x86, 0x20, 0x94, 0x12, 0x82, 0x40,
>-  0x29, 0x08, 0x49, 0x09, 0x21, 0x20,
>-  0x42, 0x43, 0x08, 0x61, 0x0c, 0x20,
>-  0x98, 0x12, 0x82, 0x50, 0x4a, 0x00,
>-  0x30, 0x84, 0xa8, 0x95, 0x12, 0xa0,
>-  0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
>-  0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
>-  0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
>-  0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
>-  0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
>-  0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
>-  0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
>-  0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
>-  0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00,
>-  0x2a, 0x03, 0x31, 0xda, 0x46, 0x20,
>-  0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
>-  0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
>-  0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
>-  0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
>-  0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
>-  0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
>-  0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
>-  0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
>-  0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00,
>-  0x4c, 0x19, 0x13, 0x22, 0x64, 0x40,
>-  0x51, 0x14, 0x52, 0x8a, 0x51, 0x40,
>-  0xa0, 0x6a, 0x45, 0x48, 0xa9, 0x00,
>-  0x04, 0xc1, 0x30, 0x26, 0x04, 0xc0,
>-  0x03, 0xb4, 0x06, 0x80, 0xd0, 0x00,
>-  0x86, 0x20, 0x94, 0x12, 0x82, 0x40,
>-  0x29, 0x08, 0x49, 0x09, 0x21, 0x20,
>-  0x42, 0x43, 0x08, 0x61, 0x0c, 0x20,
>-  0x98, 0x12, 0x82, 0x50, 0x4a, 0x00,
>-  0x30, 0x84, 0xa8, 0x95, 0x12, 0xa0,
>-  0x3a, 0xab, 0x77, 0x63, 0xef, 0x60
>-};
>-
>-const uint8_t kMaskRandom43_41[246] = {
>-  0x4c, 0x19, 0x13, 0x22, 0x64, 0x40,
>-  0x51, 0x14, 0x52, 0x8a, 0x51, 0x40,
>-  0xa0, 0x6a, 0x45, 0x48, 0xa9, 0x00,
>-  0x04, 0xc1, 0x30, 0x26, 0x04, 0xc0,
>-  0x03, 0xb4, 0x06, 0x80, 0xd0, 0x00,
>-  0x86, 0x20, 0x94, 0x12, 0x82, 0x40,
>-  0x29, 0x08, 0x49, 0x09, 0x21, 0x20,
>-  0x42, 0x43, 0x08, 0x61, 0x0c, 0x20,
>-  0x98, 0x12, 0x82, 0x50, 0x4a, 0x00,
>-  0x30, 0x84, 0xa8, 0x95, 0x12, 0xa0,
>-  0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
>-  0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
>-  0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
>-  0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
>-  0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
>-  0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
>-  0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
>-  0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
>-  0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00,
>-  0x2a, 0x03, 0x31, 0xda, 0x46, 0x20,
>-  0x4c, 0x19, 0x13, 0x22, 0x64, 0x40,
>-  0x51, 0x14, 0x52, 0x8a, 0x51, 0x40,
>-  0xa0, 0x6a, 0x45, 0x48, 0xa9, 0x00,
>-  0x04, 0xc1, 0x30, 0x26, 0x04, 0xc0,
>-  0x03, 0xb4, 0x06, 0x80, 0xd0, 0x00,
>-  0x86, 0x20, 0x94, 0x12, 0x82, 0x40,
>-  0x29, 0x08, 0x49, 0x09, 0x21, 0x20,
>-  0x42, 0x43, 0x08, 0x61, 0x0c, 0x20,
>-  0x98, 0x12, 0x82, 0x50, 0x4a, 0x00,
>-  0x30, 0x84, 0xa8, 0x95, 0x12, 0xa0,
>-  0x4c, 0x11, 0x92, 0x32, 0x46, 0x40,
>-  0x51, 0x0c, 0xc9, 0x99, 0x33, 0x20,
>-  0xa0, 0x66, 0x44, 0xc8, 0x99, 0x00,
>-  0x04, 0xc1, 0x60, 0x2c, 0x05, 0x80,
>-  0x03, 0xa0, 0x2c, 0x05, 0x80, 0xa0,
>-  0x86, 0x21, 0x14, 0x22, 0x84, 0x40,
>-  0x29, 0x10, 0x1a, 0x03, 0x40, 0x60,
>-  0x42, 0x42, 0xa0, 0x54, 0x0a, 0x80,
>-  0x98, 0x1a, 0x03, 0x40, 0x68, 0x00,
>-  0x30, 0x84, 0x08, 0x81, 0x10, 0x20,
>-  0xdf, 0x4c, 0x11, 0x82, 0x30, 0x40
>-};
>-
>-const uint8_t kMaskRandom43_42[252] = {
>-  0x4c, 0x19, 0x13, 0x22, 0x64, 0x40,
>-  0x51, 0x14, 0x52, 0x8a, 0x51, 0x40,
>-  0xa0, 0x6a, 0x45, 0x48, 0xa9, 0x00,
>-  0x04, 0xc1, 0x30, 0x26, 0x04, 0xc0,
>-  0x03, 0xb4, 0x06, 0x80, 0xd0, 0x00,
>-  0x86, 0x20, 0x94, 0x12, 0x82, 0x40,
>-  0x29, 0x08, 0x49, 0x09, 0x21, 0x20,
>-  0x42, 0x43, 0x08, 0x61, 0x0c, 0x20,
>-  0x98, 0x12, 0x82, 0x50, 0x4a, 0x00,
>-  0x30, 0x84, 0xa8, 0x95, 0x12, 0xa0,
>-  0x4c, 0x11, 0x92, 0x32, 0x46, 0x40,
>-  0x51, 0x0c, 0xc9, 0x99, 0x33, 0x20,
>-  0xa0, 0x66, 0x44, 0xc8, 0x99, 0x00,
>-  0x04, 0xc1, 0x60, 0x2c, 0x05, 0x80,
>-  0x03, 0xa0, 0x2c, 0x05, 0x80, 0xa0,
>-  0x86, 0x21, 0x14, 0x22, 0x84, 0x40,
>-  0x29, 0x10, 0x1a, 0x03, 0x40, 0x60,
>-  0x42, 0x42, 0xa0, 0x54, 0x0a, 0x80,
>-  0x98, 0x1a, 0x03, 0x40, 0x68, 0x00,
>-  0x30, 0x84, 0x08, 0x81, 0x10, 0x20,
>-  0xdf, 0x4c, 0x11, 0x82, 0x30, 0x40,
>-  0x4c, 0x19, 0x13, 0x22, 0x64, 0x40,
>-  0x51, 0x14, 0x52, 0x8a, 0x51, 0x40,
>-  0xa0, 0x6a, 0x45, 0x48, 0xa9, 0x00,
>-  0x04, 0xc1, 0x30, 0x26, 0x04, 0xc0,
>-  0x03, 0xb4, 0x06, 0x80, 0xd0, 0x00,
>-  0x86, 0x20, 0x94, 0x12, 0x82, 0x40,
>-  0x29, 0x08, 0x49, 0x09, 0x21, 0x20,
>-  0x42, 0x43, 0x08, 0x61, 0x0c, 0x20,
>-  0x98, 0x12, 0x82, 0x50, 0x4a, 0x00,
>-  0x30, 0x84, 0xa8, 0x95, 0x12, 0xa0,
>-  0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
>-  0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
>-  0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
>-  0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
>-  0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
>-  0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
>-  0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
>-  0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
>-  0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00,
>-  0x2a, 0x03, 0x31, 0xda, 0x46, 0x20,
>-  0x26, 0x84, 0x10, 0xcd, 0xf7, 0x60
>-};
>-
>-const uint8_t kMaskRandom43_43[258] = {
>-  0x4c, 0x19, 0x13, 0x22, 0x64, 0x40,
>-  0x51, 0x14, 0x52, 0x8a, 0x51, 0x40,
>-  0xa0, 0x6a, 0x45, 0x48, 0xa9, 0x00,
>-  0x04, 0xc1, 0x30, 0x26, 0x04, 0xc0,
>-  0x03, 0xb4, 0x06, 0x80, 0xd0, 0x00,
>-  0x86, 0x20, 0x94, 0x12, 0x82, 0x40,
>-  0x29, 0x08, 0x49, 0x09, 0x21, 0x20,
>-  0x42, 0x43, 0x08, 0x61, 0x0c, 0x20,
>-  0x98, 0x12, 0x82, 0x50, 0x4a, 0x00,
>-  0x30, 0x84, 0xa8, 0x95, 0x12, 0xa0,
>-  0x4c, 0x11, 0x92, 0x32, 0x46, 0x40,
>-  0x51, 0x0c, 0xc9, 0x99, 0x33, 0x20,
>-  0xa0, 0x66, 0x44, 0xc8, 0x99, 0x00,
>-  0x04, 0xc1, 0x60, 0x2c, 0x05, 0x80,
>-  0x03, 0xa0, 0x2c, 0x05, 0x80, 0xa0,
>-  0x86, 0x21, 0x14, 0x22, 0x84, 0x40,
>-  0x29, 0x10, 0x1a, 0x03, 0x40, 0x60,
>-  0x42, 0x42, 0xa0, 0x54, 0x0a, 0x80,
>-  0x98, 0x1a, 0x03, 0x40, 0x68, 0x00,
>-  0x30, 0x84, 0x08, 0x81, 0x10, 0x20,
>-  0xdf, 0x4c, 0x11, 0x82, 0x30, 0x40,
>-  0x4c, 0x19, 0x12, 0x32, 0x46, 0x40,
>-  0x51, 0x14, 0x51, 0x99, 0x33, 0x20,
>-  0xa0, 0x6a, 0x44, 0xc8, 0x99, 0x00,
>-  0x04, 0xc1, 0x30, 0x2c, 0x05, 0x80,
>-  0x03, 0xb4, 0x04, 0x05, 0x80, 0xa0,
>-  0x86, 0x20, 0x94, 0x22, 0x84, 0x40,
>-  0x29, 0x08, 0x4a, 0x03, 0x40, 0x60,
>-  0x42, 0x43, 0x08, 0x54, 0x0a, 0x80,
>-  0x98, 0x12, 0x83, 0x40, 0x68, 0x00,
>-  0x30, 0x84, 0xa8, 0x81, 0x10, 0x20,
>-  0x4c, 0x11, 0x91, 0x82, 0x30, 0x40,
>-  0x51, 0x0c, 0xcb, 0x22, 0x64, 0x40,
>-  0xa0, 0x66, 0x42, 0x8a, 0x51, 0x40,
>-  0x04, 0xc1, 0x65, 0x48, 0xa9, 0x00,
>-  0x03, 0xa0, 0x28, 0x26, 0x04, 0xc0,
>-  0x86, 0x21, 0x16, 0x80, 0xd0, 0x00,
>-  0x29, 0x10, 0x1c, 0x12, 0x82, 0x40,
>-  0x42, 0x42, 0xa1, 0x09, 0x21, 0x20,
>-  0x98, 0x1a, 0x00, 0x61, 0x0c, 0x20,
>-  0x30, 0x84, 0x0a, 0x50, 0x4a, 0x00,
>-  0xdf, 0x4c, 0x10, 0x95, 0x12, 0xa0,
>-  0x72, 0x06, 0x94, 0xf6, 0x74, 0x40
>-};
>-
>-const uint8_t kMaskRandom43_5[30] = {
>-  0xce, 0x31, 0xb6, 0x36, 0xc6, 0xc0,
>-  0x63, 0x98, 0xdb, 0x1b, 0x63, 0x60,
>-  0x98, 0xc7, 0x68, 0xed, 0x1d, 0xa0,
>-  0x4d, 0x6b, 0x55, 0x6a, 0xad, 0x40,
>-  0xb2, 0x6c, 0xad, 0x95, 0xb2, 0xa0
>-};
>-
>-const uint8_t kMaskRandom43_6[36] = {
>-  0x4c, 0x19, 0x13, 0x22, 0x64, 0x40,
>-  0x51, 0x14, 0x52, 0x8a, 0x51, 0x40,
>-  0x20, 0xea, 0x0d, 0x41, 0xa8, 0x20,
>-  0x85, 0x41, 0x2e, 0x25, 0x04, 0xa0,
>-  0x06, 0x80, 0xd8, 0x1b, 0x03, 0x60,
>-  0x8a, 0x24, 0x34, 0x86, 0x90, 0xc0
>-};
>-
>-const uint8_t kMaskRandom43_7[42] = {
>-  0xc6, 0x11, 0x96, 0x32, 0x46, 0x40,
>-  0x33, 0x04, 0xc8, 0x99, 0x33, 0x20,
>-  0x18, 0x67, 0x44, 0x68, 0x9d, 0x00,
>-  0x45, 0x42, 0xd4, 0x5a, 0x0b, 0x40,
>-  0x12, 0xd4, 0x2a, 0x95, 0x50, 0xa0,
>-  0xb4, 0x28, 0x35, 0x16, 0xa0, 0xc0,
>-  0x29, 0x92, 0x1b, 0x0d, 0x41, 0x60
>-};
>-
>-const uint8_t kMaskRandom43_8[48] = {
>-  0x07, 0x0a, 0x71, 0x44, 0x29, 0xc0,
>-  0x49, 0xa8, 0x29, 0x0f, 0xa0, 0x20,
>-  0xb0, 0x7a, 0x07, 0x48, 0xe8, 0x00,
>-  0x24, 0xc5, 0xc0, 0xb8, 0x17, 0x00,
>-  0x52, 0x80, 0xec, 0x1d, 0x02, 0xa0,
>-  0xc6, 0x31, 0x82, 0x30, 0xc7, 0x40,
>-  0x31, 0x94, 0x1a, 0x83, 0x50, 0x60,
>-  0x18, 0xc7, 0x08, 0xe1, 0x1c, 0x20
>-};
>-
>-const uint8_t kMaskRandom43_9[54] = {
>-  0x4e, 0x11, 0x12, 0x22, 0x46, 0x40,
>-  0x62, 0x1a, 0x09, 0x41, 0x68, 0x60,
>-  0x80, 0xe9, 0x41, 0x28, 0xa5, 0x00,
>-  0xa1, 0x50, 0x52, 0xc8, 0x51, 0x00,
>-  0x53, 0x00, 0x68, 0x1d, 0x01, 0xa0,
>-  0xa4, 0x24, 0x36, 0x06, 0x80, 0xc0,
>-  0x16, 0xa0, 0x8d, 0x11, 0x82, 0x20,
>-  0x58, 0x45, 0x20, 0xa4, 0x16, 0x80,
>-  0x29, 0x86, 0x84, 0xd0, 0x1c, 0x00
>-};
>-
>-const uint8_t kMaskRandom44_1[6] = {
>-  0xff, 0xff, 0xff, 0xff, 0xff, 0xf0
>-};
>-
>-const uint8_t kMaskRandom44_10[60] = {
>-  0xc0, 0x38, 0x8b, 0x00, 0xe2, 0x20,
>-  0x30, 0x0e, 0x28, 0xc0, 0x38, 0xa0,
>-  0xe8, 0x07, 0x03, 0xa0, 0x1c, 0x00,
>-  0x85, 0x08, 0xaa, 0x14, 0x22, 0xa0,
>-  0xd0, 0x92, 0x13, 0x42, 0x48, 0x40,
>-  0x86, 0x50, 0x4a, 0x19, 0x41, 0x20,
>-  0x4a, 0x68, 0x0d, 0x29, 0xa0, 0x30,
>-  0x01, 0xa0, 0x74, 0x06, 0x81, 0xd0,
>-  0x4c, 0x81, 0x91, 0x32, 0x06, 0x40,
>-  0x62, 0x24, 0x05, 0x88, 0x90, 0x10
>-};
>-
>-const uint8_t kMaskRandom44_11[66] = {
>-  0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
>-  0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
>-  0x99, 0x13, 0x22, 0x64, 0x4c, 0x80,
>-  0x05, 0x80, 0xb0, 0x16, 0x02, 0xc0,
>-  0x80, 0xb0, 0x16, 0x02, 0xc0, 0x50,
>-  0x84, 0x50, 0x8a, 0x11, 0x42, 0x20,
>-  0x40, 0x68, 0x0d, 0x01, 0xa0, 0x30,
>-  0x0a, 0x81, 0x50, 0x2a, 0x05, 0x40,
>-  0x68, 0x0d, 0x01, 0xa0, 0x34, 0x00,
>-  0x10, 0x22, 0x04, 0x40, 0x88, 0x10,
>-  0x30, 0x46, 0x08, 0xc1, 0x18, 0x20
>-};
>-
>-const uint8_t kMaskRandom44_12[72] = {
>-  0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
>-  0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
>-  0x0c, 0xa1, 0x94, 0x32, 0x86, 0x50,
>-  0xa1, 0x34, 0x26, 0x84, 0xd0, 0x90,
>-  0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
>-  0x8a, 0x51, 0x4a, 0x29, 0x45, 0x20,
>-  0x86, 0x90, 0xd2, 0x1a, 0x43, 0x40,
>-  0x23, 0x24, 0x64, 0x8c, 0x91, 0x90,
>-  0x16, 0x22, 0xc4, 0x58, 0x8b, 0x10,
>-  0x4c, 0x29, 0x85, 0x30, 0xa6, 0x10,
>-  0x41, 0xc8, 0x39, 0x07, 0x20, 0xe0,
>-  0xf4, 0x18, 0x9f, 0xd0, 0x62, 0x70
>-};
>-
>-const uint8_t kMaskRandom44_13[78] = {
>-  0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
>-  0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
>-  0x0c, 0xa1, 0x94, 0x32, 0x86, 0x50,
>-  0xa1, 0x34, 0x26, 0x84, 0xd0, 0x90,
>-  0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
>-  0x8a, 0x51, 0x4a, 0x29, 0x45, 0x20,
>-  0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
>-  0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
>-  0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
>-  0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
>-  0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
>-  0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
>-  0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0
>-};
>-
>-const uint8_t kMaskRandom44_14[84] = {
>-  0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
>-  0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
>-  0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
>-  0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
>-  0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
>-  0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
>-  0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0,
>-  0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
>-  0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
>-  0x0c, 0xa1, 0x94, 0x32, 0x86, 0x50,
>-  0xa1, 0x34, 0x26, 0x84, 0xd0, 0x90,
>-  0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
>-  0x8a, 0x51, 0x4a, 0x29, 0x45, 0x20,
>-  0xc6, 0xca, 0xeb, 0x1b, 0x2b, 0xa0
>-};
>-
>-const uint8_t kMaskRandom44_15[90] = {
>-  0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
>-  0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
>-  0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
>-  0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
>-  0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
>-  0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
>-  0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0,
>-  0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
>-  0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
>-  0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
>-  0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
>-  0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
>-  0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
>-  0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
>-  0x14, 0x22, 0x84, 0x50, 0x8a, 0x10
>-};
>-
>-const uint8_t kMaskRandom44_16[96] = {
>-  0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
>-  0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
>-  0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
>-  0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
>-  0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
>-  0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
>-  0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
>-  0x14, 0x22, 0x84, 0x50, 0x8a, 0x10,
>-  0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
>-  0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
>-  0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
>-  0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
>-  0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
>-  0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
>-  0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0,
>-  0x86, 0xc1, 0x46, 0x1b, 0x05, 0x10
>-};
>-
>-const uint8_t kMaskRandom44_17[102] = {
>-  0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
>-  0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
>-  0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
>-  0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
>-  0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
>-  0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
>-  0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
>-  0x14, 0x22, 0x84, 0x50, 0x8a, 0x10,
>-  0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
>-  0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
>-  0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
>-  0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
>-  0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
>-  0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
>-  0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
>-  0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
>-  0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00
>-};
>-
>-const uint8_t kMaskRandom44_18[108] = {
>-  0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
>-  0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
>-  0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
>-  0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
>-  0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
>-  0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
>-  0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
>-  0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
>-  0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00,
>-  0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
>-  0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
>-  0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
>-  0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
>-  0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
>-  0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
>-  0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
>-  0x14, 0x22, 0x84, 0x50, 0x8a, 0x10,
>-  0xe4, 0xd4, 0x6f, 0x93, 0x51, 0xb0
>-};
>-
>-const uint8_t kMaskRandom44_19[114] = {
>-  0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
>-  0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
>-  0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
>-  0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
>-  0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
>-  0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
>-  0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
>-  0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
>-  0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00,
>-  0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
>-  0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
>-  0xa9, 0x15, 0x22, 0xa4, 0x54, 0x80,
>-  0x04, 0xc0, 0x98, 0x13, 0x02, 0x60,
>-  0xd0, 0x1a, 0x03, 0x40, 0x68, 0x00,
>-  0x82, 0x50, 0x4a, 0x09, 0x41, 0x20,
>-  0x21, 0x24, 0x24, 0x84, 0x90, 0x90,
>-  0x0c, 0x21, 0x84, 0x30, 0x86, 0x10,
>-  0x4a, 0x09, 0x41, 0x28, 0x25, 0x00,
>-  0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50
>-};
>-
>-const uint8_t kMaskRandom44_2[12] = {
>-  0xec, 0xdd, 0x9b, 0xb3, 0x76, 0x60,
>-  0x9b, 0xb3, 0x76, 0x6e, 0xcd, 0xd0
>-};
>-
>-const uint8_t kMaskRandom44_20[120] = {
>-  0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
>-  0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
>-  0xa9, 0x15, 0x22, 0xa4, 0x54, 0x80,
>-  0x04, 0xc0, 0x98, 0x13, 0x02, 0x60,
>-  0xd0, 0x1a, 0x03, 0x40, 0x68, 0x00,
>-  0x82, 0x50, 0x4a, 0x09, 0x41, 0x20,
>-  0x21, 0x24, 0x24, 0x84, 0x90, 0x90,
>-  0x0c, 0x21, 0x84, 0x30, 0x86, 0x10,
>-  0x4a, 0x09, 0x41, 0x28, 0x25, 0x00,
>-  0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
>-  0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
>-  0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
>-  0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
>-  0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
>-  0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
>-  0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
>-  0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
>-  0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
>-  0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00,
>-  0x3b, 0x48, 0xc4, 0xed, 0x23, 0x10
>-};
>-
>-const uint8_t kMaskRandom44_21[126] = {
>-  0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
>-  0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
>-  0xa9, 0x15, 0x22, 0xa4, 0x54, 0x80,
>-  0x04, 0xc0, 0x98, 0x13, 0x02, 0x60,
>-  0xd0, 0x1a, 0x03, 0x40, 0x68, 0x00,
>-  0x82, 0x50, 0x4a, 0x09, 0x41, 0x20,
>-  0x21, 0x24, 0x24, 0x84, 0x90, 0x90,
>-  0x0c, 0x21, 0x84, 0x30, 0x86, 0x10,
>-  0x4a, 0x09, 0x41, 0x28, 0x25, 0x00,
>-  0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
>-  0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
>-  0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
>-  0x99, 0x13, 0x22, 0x64, 0x4c, 0x80,
>-  0x05, 0x80, 0xb0, 0x16, 0x02, 0xc0,
>-  0x80, 0xb0, 0x16, 0x02, 0xc0, 0x50,
>-  0x84, 0x50, 0x8a, 0x11, 0x42, 0x20,
>-  0x40, 0x68, 0x0d, 0x01, 0xa0, 0x30,
>-  0x0a, 0x81, 0x50, 0x2a, 0x05, 0x40,
>-  0x68, 0x0d, 0x01, 0xa0, 0x34, 0x00,
>-  0x10, 0x22, 0x04, 0x40, 0x88, 0x10,
>-  0x30, 0x46, 0x08, 0xc1, 0x18, 0x20
>-};
>-
>-const uint8_t kMaskRandom44_22[132] = {
>-  0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
>-  0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
>-  0x99, 0x13, 0x22, 0x64, 0x4c, 0x80,
>-  0x05, 0x80, 0xb0, 0x16, 0x02, 0xc0,
>-  0x80, 0xb0, 0x16, 0x02, 0xc0, 0x50,
>-  0x84, 0x50, 0x8a, 0x11, 0x42, 0x20,
>-  0x40, 0x68, 0x0d, 0x01, 0xa0, 0x30,
>-  0x0a, 0x81, 0x50, 0x2a, 0x05, 0x40,
>-  0x68, 0x0d, 0x01, 0xa0, 0x34, 0x00,
>-  0x10, 0x22, 0x04, 0x40, 0x88, 0x10,
>-  0x30, 0x46, 0x08, 0xc1, 0x18, 0x20,
>-  0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
>-  0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
>-  0xa9, 0x15, 0x22, 0xa4, 0x54, 0x80,
>-  0x04, 0xc0, 0x98, 0x13, 0x02, 0x60,
>-  0xd0, 0x1a, 0x03, 0x40, 0x68, 0x00,
>-  0x82, 0x50, 0x4a, 0x09, 0x41, 0x20,
>-  0x21, 0x24, 0x24, 0x84, 0x90, 0x90,
>-  0x0c, 0x21, 0x84, 0x30, 0x86, 0x10,
>-  0x4a, 0x09, 0x41, 0x28, 0x25, 0x00,
>-  0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
>-  0x9e, 0xce, 0x8a, 0x7b, 0x3a, 0x20
>-};
>-
>-const uint8_t kMaskRandom44_23[138] = {
>-  0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
>-  0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
>-  0x99, 0x13, 0x22, 0x64, 0x4c, 0x80,
>-  0x05, 0x80, 0xb0, 0x16, 0x02, 0xc0,
>-  0x80, 0xb0, 0x16, 0x02, 0xc0, 0x50,
>-  0x84, 0x50, 0x8a, 0x11, 0x42, 0x20,
>-  0x40, 0x68, 0x0d, 0x01, 0xa0, 0x30,
>-  0x0a, 0x81, 0x50, 0x2a, 0x05, 0x40,
>-  0x68, 0x0d, 0x01, 0xa0, 0x34, 0x00,
>-  0x10, 0x22, 0x04, 0x40, 0x88, 0x10,
>-  0x30, 0x46, 0x08, 0xc1, 0x18, 0x20,
>-  0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
>-  0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
>-  0x0c, 0xa1, 0x94, 0x32, 0x86, 0x50,
>-  0xa1, 0x34, 0x26, 0x84, 0xd0, 0x90,
>-  0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
>-  0x8a, 0x51, 0x4a, 0x29, 0x45, 0x20,
>-  0x86, 0x90, 0xd2, 0x1a, 0x43, 0x40,
>-  0x23, 0x24, 0x64, 0x8c, 0x91, 0x90,
>-  0x16, 0x22, 0xc4, 0x58, 0x8b, 0x10,
>-  0x4c, 0x29, 0x85, 0x30, 0xa6, 0x10,
>-  0x41, 0xc8, 0x39, 0x07, 0x20, 0xe0,
>-  0xf4, 0x18, 0x9f, 0xd0, 0x62, 0x70
>-};
>-
>-const uint8_t kMaskRandom44_24[144] = {
>-  0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
>-  0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
>-  0x0c, 0xa1, 0x94, 0x32, 0x86, 0x50,
>-  0xa1, 0x34, 0x26, 0x84, 0xd0, 0x90,
>-  0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
>-  0x8a, 0x51, 0x4a, 0x29, 0x45, 0x20,
>-  0x86, 0x90, 0xd2, 0x1a, 0x43, 0x40,
>-  0x23, 0x24, 0x64, 0x8c, 0x91, 0x90,
>-  0x16, 0x22, 0xc4, 0x58, 0x8b, 0x10,
>-  0x4c, 0x29, 0x85, 0x30, 0xa6, 0x10,
>-  0x41, 0xc8, 0x39, 0x07, 0x20, 0xe0,
>-  0xf4, 0x18, 0x9f, 0xd0, 0x62, 0x70,
>-  0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
>-  0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
>-  0x99, 0x13, 0x22, 0x64, 0x4c, 0x80,
>-  0x05, 0x80, 0xb0, 0x16, 0x02, 0xc0,
>-  0x80, 0xb0, 0x16, 0x02, 0xc0, 0x50,
>-  0x84, 0x50, 0x8a, 0x11, 0x42, 0x20,
>-  0x40, 0x68, 0x0d, 0x01, 0xa0, 0x30,
>-  0x0a, 0x81, 0x50, 0x2a, 0x05, 0x40,
>-  0x68, 0x0d, 0x01, 0xa0, 0x34, 0x00,
>-  0x10, 0x22, 0x04, 0x40, 0x88, 0x10,
>-  0x30, 0x46, 0x08, 0xc1, 0x18, 0x20,
>-  0x15, 0x0f, 0x44, 0x6d, 0x9d, 0xa0
>-};
>-
>-const uint8_t kMaskRandom44_25[150] = {
>-  0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
>-  0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
>-  0x0c, 0xa1, 0x94, 0x32, 0x86, 0x50,
>-  0xa1, 0x34, 0x26, 0x84, 0xd0, 0x90,
>-  0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
>-  0x8a, 0x51, 0x4a, 0x29, 0x45, 0x20,
>-  0x86, 0x90, 0xd2, 0x1a, 0x43, 0x40,
>-  0x23, 0x24, 0x64, 0x8c, 0x91, 0x90,
>-  0x16, 0x22, 0xc4, 0x58, 0x8b, 0x10,
>-  0x4c, 0x29, 0x85, 0x30, 0xa6, 0x10,
>-  0x41, 0xc8, 0x39, 0x07, 0x20, 0xe0,
>-  0xf4, 0x18, 0x9f, 0xd0, 0x62, 0x70,
>-  0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
>-  0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
>-  0x0c, 0xa1, 0x94, 0x32, 0x86, 0x50,
>-  0xa1, 0x34, 0x26, 0x84, 0xd0, 0x90,
>-  0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
>-  0x8a, 0x51, 0x4a, 0x29, 0x45, 0x20,
>-  0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
>-  0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
>-  0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
>-  0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
>-  0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
>-  0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
>-  0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0
>-};
>-
>-const uint8_t kMaskRandom44_26[156] = {
>-  0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
>-  0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
>-  0x0c, 0xa1, 0x94, 0x32, 0x86, 0x50,
>-  0xa1, 0x34, 0x26, 0x84, 0xd0, 0x90,
>-  0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
>-  0x8a, 0x51, 0x4a, 0x29, 0x45, 0x20,
>-  0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
>-  0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
>-  0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
>-  0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
>-  0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
>-  0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
>-  0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0,
>-  0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
>-  0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
>-  0x0c, 0xa1, 0x94, 0x32, 0x86, 0x50,
>-  0xa1, 0x34, 0x26, 0x84, 0xd0, 0x90,
>-  0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
>-  0x8a, 0x51, 0x4a, 0x29, 0x45, 0x20,
>-  0x86, 0x90, 0xd2, 0x1a, 0x43, 0x40,
>-  0x23, 0x24, 0x64, 0x8c, 0x91, 0x90,
>-  0x16, 0x22, 0xc4, 0x58, 0x8b, 0x10,
>-  0x4c, 0x29, 0x85, 0x30, 0xa6, 0x10,
>-  0x41, 0xc8, 0x39, 0x07, 0x20, 0xe0,
>-  0xf4, 0x18, 0x9f, 0xd0, 0x62, 0x70,
>-  0x02, 0xcb, 0x64, 0xb8, 0x55, 0x80
>-};
>-
>-const uint8_t kMaskRandom44_27[162] = {
>-  0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
>-  0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
>-  0x0c, 0xa1, 0x94, 0x32, 0x86, 0x50,
>-  0xa1, 0x34, 0x26, 0x84, 0xd0, 0x90,
>-  0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
>-  0x8a, 0x51, 0x4a, 0x29, 0x45, 0x20,
>-  0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
>-  0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
>-  0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
>-  0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
>-  0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
>-  0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
>-  0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0,
>-  0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
>-  0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
>-  0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
>-  0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
>-  0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
>-  0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
>-  0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0,
>-  0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
>-  0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
>-  0x0c, 0xa1, 0x94, 0x32, 0x86, 0x50,
>-  0xa1, 0x34, 0x26, 0x84, 0xd0, 0x90,
>-  0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
>-  0x8a, 0x51, 0x4a, 0x29, 0x45, 0x20,
>-  0xc6, 0xca, 0xeb, 0x1b, 0x2b, 0xa0
>-};
>-
>-const uint8_t kMaskRandom44_28[168] = {
>-  0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
>-  0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
>-  0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
>-  0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
>-  0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
>-  0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
>-  0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0,
>-  0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
>-  0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
>-  0x0c, 0xa1, 0x94, 0x32, 0x86, 0x50,
>-  0xa1, 0x34, 0x26, 0x84, 0xd0, 0x90,
>-  0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
>-  0x8a, 0x51, 0x4a, 0x29, 0x45, 0x20,
>-  0xc6, 0xca, 0xeb, 0x1b, 0x2b, 0xa0,
>-  0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
>-  0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
>-  0x0c, 0xa1, 0x94, 0x32, 0x86, 0x50,
>-  0xa1, 0x34, 0x26, 0x84, 0xd0, 0x90,
>-  0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
>-  0x8a, 0x51, 0x4a, 0x29, 0x45, 0x20,
>-  0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
>-  0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
>-  0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
>-  0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
>-  0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
>-  0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
>-  0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0,
>-  0x66, 0x26, 0x6c, 0x91, 0xc7, 0x20
>-};
>-
>-const uint8_t kMaskRandom44_29[174] = {
>-  0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
>-  0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
>-  0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
>-  0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
>-  0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
>-  0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
>-  0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0,
>-  0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
>-  0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
>-  0x0c, 0xa1, 0x94, 0x32, 0x86, 0x50,
>-  0xa1, 0x34, 0x26, 0x84, 0xd0, 0x90,
>-  0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
>-  0x8a, 0x51, 0x4a, 0x29, 0x45, 0x20,
>-  0xc6, 0xca, 0xeb, 0x1b, 0x2b, 0xa0,
>-  0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
>-  0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
>-  0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
>-  0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
>-  0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
>-  0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
>-  0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0,
>-  0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
>-  0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
>-  0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
>-  0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
>-  0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
>-  0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
>-  0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
>-  0x14, 0x22, 0x84, 0x50, 0x8a, 0x10
>-};
>-
>-const uint8_t kMaskRandom44_3[18] = {
>-  0xca, 0xd9, 0x5b, 0x2b, 0x65, 0x60,
>-  0xf1, 0x5e, 0x2b, 0xc5, 0x78, 0xa0,
>-  0xb6, 0x35, 0xc6, 0xd8, 0xd7, 0x10
>-};
>-
>-const uint8_t kMaskRandom44_30[180] = {
>-  0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
>-  0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
>-  0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
>-  0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
>-  0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
>-  0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
>-  0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0,
>-  0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
>-  0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
>-  0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
>-  0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
>-  0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
>-  0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
>-  0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
>-  0x14, 0x22, 0x84, 0x50, 0x8a, 0x10,
>-  0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
>-  0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
>-  0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
>-  0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
>-  0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
>-  0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
>-  0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0,
>-  0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
>-  0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
>-  0x0c, 0xa1, 0x94, 0x32, 0x86, 0x50,
>-  0xa1, 0x34, 0x26, 0x84, 0xd0, 0x90,
>-  0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
>-  0x8a, 0x51, 0x4a, 0x29, 0x45, 0x20,
>-  0xc6, 0xca, 0xeb, 0x1b, 0x2b, 0xa0,
>-  0x60, 0xf4, 0x75, 0x84, 0x90, 0xc0
>-};
>-
>-const uint8_t kMaskRandom44_31[186] = {
>-  0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
>-  0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
>-  0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
>-  0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
>-  0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
>-  0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
>-  0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0,
>-  0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
>-  0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
>-  0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
>-  0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
>-  0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
>-  0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
>-  0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
>-  0x14, 0x22, 0x84, 0x50, 0x8a, 0x10,
>-  0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
>-  0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
>-  0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
>-  0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
>-  0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
>-  0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
>-  0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
>-  0x14, 0x22, 0x84, 0x50, 0x8a, 0x10,
>-  0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
>-  0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
>-  0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
>-  0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
>-  0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
>-  0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
>-  0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0,
>-  0x86, 0xc1, 0x46, 0x1b, 0x05, 0x10
>-};
>-
>-const uint8_t kMaskRandom44_32[192] = {
>-  0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
>-  0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
>-  0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
>-  0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
>-  0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
>-  0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
>-  0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
>-  0x14, 0x22, 0x84, 0x50, 0x8a, 0x10,
>-  0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
>-  0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
>-  0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
>-  0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
>-  0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
>-  0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
>-  0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0,
>-  0x86, 0xc1, 0x46, 0x1b, 0x05, 0x10,
>-  0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
>-  0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
>-  0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
>-  0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
>-  0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
>-  0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
>-  0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0,
>-  0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
>-  0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
>-  0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
>-  0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
>-  0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
>-  0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
>-  0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
>-  0x14, 0x22, 0x84, 0x50, 0x8a, 0x10,
>-  0x3e, 0x39, 0x86, 0x5c, 0xd9, 0xd0
>-};
>-
>-const uint8_t kMaskRandom44_33[198] = {
>-  0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
>-  0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
>-  0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
>-  0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
>-  0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
>-  0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
>-  0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
>-  0x14, 0x22, 0x84, 0x50, 0x8a, 0x10,
>-  0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
>-  0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
>-  0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
>-  0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
>-  0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
>-  0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
>-  0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0,
>-  0x86, 0xc1, 0x46, 0x1b, 0x05, 0x10,
>-  0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
>-  0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
>-  0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
>-  0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
>-  0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
>-  0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
>-  0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
>-  0x14, 0x22, 0x84, 0x50, 0x8a, 0x10,
>-  0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
>-  0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
>-  0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
>-  0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
>-  0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
>-  0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
>-  0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
>-  0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
>-  0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00
>-};
>-
>-const uint8_t kMaskRandom44_34[204] = {
>-  0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
>-  0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
>-  0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
>-  0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
>-  0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
>-  0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
>-  0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
>-  0x14, 0x22, 0x84, 0x50, 0x8a, 0x10,
>-  0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
>-  0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
>-  0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
>-  0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
>-  0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
>-  0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
>-  0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
>-  0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
>-  0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00,
>-  0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
>-  0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
>-  0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
>-  0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
>-  0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
>-  0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
>-  0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
>-  0x14, 0x22, 0x84, 0x50, 0x8a, 0x10,
>-  0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
>-  0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
>-  0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
>-  0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
>-  0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
>-  0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
>-  0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0,
>-  0x86, 0xc1, 0x46, 0x1b, 0x05, 0x10,
>-  0xb5, 0xc7, 0xe8, 0x0c, 0xb9, 0x90
>-};
>-
>-const uint8_t kMaskRandom44_35[210] = {
>-  0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
>-  0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
>-  0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
>-  0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
>-  0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
>-  0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
>-  0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
>-  0x14, 0x22, 0x84, 0x50, 0x8a, 0x10,
>-  0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
>-  0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
>-  0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
>-  0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
>-  0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
>-  0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
>-  0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
>-  0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
>-  0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00,
>-  0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
>-  0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
>-  0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
>-  0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
>-  0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
>-  0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
>-  0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
>-  0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
>-  0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00,
>-  0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
>-  0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
>-  0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
>-  0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
>-  0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
>-  0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
>-  0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
>-  0x14, 0x22, 0x84, 0x50, 0x8a, 0x10,
>-  0xe4, 0xd4, 0x6f, 0x93, 0x51, 0xb0
>-};
>-
>-const uint8_t kMaskRandom44_36[216] = {
>-  0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
>-  0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
>-  0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
>-  0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
>-  0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
>-  0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
>-  0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
>-  0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
>-  0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00,
>-  0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
>-  0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
>-  0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
>-  0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
>-  0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
>-  0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
>-  0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
>-  0x14, 0x22, 0x84, 0x50, 0x8a, 0x10,
>-  0xe4, 0xd4, 0x6f, 0x93, 0x51, 0xb0,
>-  0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
>-  0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
>-  0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
>-  0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
>-  0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
>-  0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
>-  0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
>-  0x14, 0x22, 0x84, 0x50, 0x8a, 0x10,
>-  0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
>-  0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
>-  0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
>-  0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
>-  0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
>-  0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
>-  0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
>-  0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
>-  0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00,
>-  0xa6, 0x92, 0x01, 0x65, 0x91, 0x20
>-};
>-
>-const uint8_t kMaskRandom44_37[222] = {
>-  0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
>-  0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
>-  0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
>-  0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
>-  0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
>-  0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
>-  0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
>-  0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
>-  0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00,
>-  0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
>-  0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
>-  0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
>-  0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
>-  0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
>-  0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
>-  0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
>-  0x14, 0x22, 0x84, 0x50, 0x8a, 0x10,
>-  0xe4, 0xd4, 0x6f, 0x93, 0x51, 0xb0,
>-  0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
>-  0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
>-  0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
>-  0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
>-  0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
>-  0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
>-  0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
>-  0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
>-  0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00,
>-  0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
>-  0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
>-  0xa9, 0x15, 0x22, 0xa4, 0x54, 0x80,
>-  0x04, 0xc0, 0x98, 0x13, 0x02, 0x60,
>-  0xd0, 0x1a, 0x03, 0x40, 0x68, 0x00,
>-  0x82, 0x50, 0x4a, 0x09, 0x41, 0x20,
>-  0x21, 0x24, 0x24, 0x84, 0x90, 0x90,
>-  0x0c, 0x21, 0x84, 0x30, 0x86, 0x10,
>-  0x4a, 0x09, 0x41, 0x28, 0x25, 0x00,
>-  0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50
>-};
>-
>-const uint8_t kMaskRandom44_38[228] = {
>-  0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
>-  0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
>-  0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
>-  0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
>-  0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
>-  0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
>-  0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
>-  0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
>-  0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00,
>-  0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
>-  0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
>-  0xa9, 0x15, 0x22, 0xa4, 0x54, 0x80,
>-  0x04, 0xc0, 0x98, 0x13, 0x02, 0x60,
>-  0xd0, 0x1a, 0x03, 0x40, 0x68, 0x00,
>-  0x82, 0x50, 0x4a, 0x09, 0x41, 0x20,
>-  0x21, 0x24, 0x24, 0x84, 0x90, 0x90,
>-  0x0c, 0x21, 0x84, 0x30, 0x86, 0x10,
>-  0x4a, 0x09, 0x41, 0x28, 0x25, 0x00,
>-  0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
>-  0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
>-  0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
>-  0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
>-  0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
>-  0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
>-  0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
>-  0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
>-  0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
>-  0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00,
>-  0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
>-  0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
>-  0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
>-  0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
>-  0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
>-  0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
>-  0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
>-  0x14, 0x22, 0x84, 0x50, 0x8a, 0x10,
>-  0xe4, 0xd4, 0x6f, 0x93, 0x51, 0xb0,
>-  0x43, 0x64, 0xf2, 0xe5, 0x5d, 0x10
>-};
>-
>-const uint8_t kMaskRandom44_39[234] = {
>-  0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
>-  0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
>-  0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
>-  0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
>-  0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
>-  0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
>-  0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
>-  0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
>-  0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00,
>-  0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
>-  0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
>-  0xa9, 0x15, 0x22, 0xa4, 0x54, 0x80,
>-  0x04, 0xc0, 0x98, 0x13, 0x02, 0x60,
>-  0xd0, 0x1a, 0x03, 0x40, 0x68, 0x00,
>-  0x82, 0x50, 0x4a, 0x09, 0x41, 0x20,
>-  0x21, 0x24, 0x24, 0x84, 0x90, 0x90,
>-  0x0c, 0x21, 0x84, 0x30, 0x86, 0x10,
>-  0x4a, 0x09, 0x41, 0x28, 0x25, 0x00,
>-  0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
>-  0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
>-  0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
>-  0xa9, 0x15, 0x22, 0xa4, 0x54, 0x80,
>-  0x04, 0xc0, 0x98, 0x13, 0x02, 0x60,
>-  0xd0, 0x1a, 0x03, 0x40, 0x68, 0x00,
>-  0x82, 0x50, 0x4a, 0x09, 0x41, 0x20,
>-  0x21, 0x24, 0x24, 0x84, 0x90, 0x90,
>-  0x0c, 0x21, 0x84, 0x30, 0x86, 0x10,
>-  0x4a, 0x09, 0x41, 0x28, 0x25, 0x00,
>-  0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
>-  0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
>-  0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
>-  0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
>-  0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
>-  0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
>-  0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
>-  0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
>-  0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
>-  0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00,
>-  0x3b, 0x48, 0xc4, 0xed, 0x23, 0x10
>-};
>-
>-const uint8_t kMaskRandom44_4[24] = {
>-  0xc4, 0xd8, 0x9b, 0x13, 0x62, 0x60,
>-  0x31, 0x66, 0x2c, 0xc5, 0x98, 0xb0,
>-  0x4b, 0x29, 0x65, 0x2c, 0xa5, 0x90,
>-  0x2c, 0xa5, 0x94, 0xb2, 0x96, 0x50
>-};
>-
>-const uint8_t kMaskRandom44_40[240] = {
>-  0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
>-  0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
>-  0xa9, 0x15, 0x22, 0xa4, 0x54, 0x80,
>-  0x04, 0xc0, 0x98, 0x13, 0x02, 0x60,
>-  0xd0, 0x1a, 0x03, 0x40, 0x68, 0x00,
>-  0x82, 0x50, 0x4a, 0x09, 0x41, 0x20,
>-  0x21, 0x24, 0x24, 0x84, 0x90, 0x90,
>-  0x0c, 0x21, 0x84, 0x30, 0x86, 0x10,
>-  0x4a, 0x09, 0x41, 0x28, 0x25, 0x00,
>-  0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
>-  0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
>-  0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
>-  0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
>-  0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
>-  0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
>-  0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
>-  0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
>-  0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
>-  0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00,
>-  0x3b, 0x48, 0xc4, 0xed, 0x23, 0x10,
>-  0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
>-  0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
>-  0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
>-  0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
>-  0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
>-  0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
>-  0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
>-  0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
>-  0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00,
>-  0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
>-  0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
>-  0xa9, 0x15, 0x22, 0xa4, 0x54, 0x80,
>-  0x04, 0xc0, 0x98, 0x13, 0x02, 0x60,
>-  0xd0, 0x1a, 0x03, 0x40, 0x68, 0x00,
>-  0x82, 0x50, 0x4a, 0x09, 0x41, 0x20,
>-  0x21, 0x24, 0x24, 0x84, 0x90, 0x90,
>-  0x0c, 0x21, 0x84, 0x30, 0x86, 0x10,
>-  0x4a, 0x09, 0x41, 0x28, 0x25, 0x00,
>-  0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
>-  0xd8, 0x2a, 0x16, 0x26, 0x51, 0x40
>-};
>-
>-const uint8_t kMaskRandom44_41[246] = {
>-  0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
>-  0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
>-  0xa9, 0x15, 0x22, 0xa4, 0x54, 0x80,
>-  0x04, 0xc0, 0x98, 0x13, 0x02, 0x60,
>-  0xd0, 0x1a, 0x03, 0x40, 0x68, 0x00,
>-  0x82, 0x50, 0x4a, 0x09, 0x41, 0x20,
>-  0x21, 0x24, 0x24, 0x84, 0x90, 0x90,
>-  0x0c, 0x21, 0x84, 0x30, 0x86, 0x10,
>-  0x4a, 0x09, 0x41, 0x28, 0x25, 0x00,
>-  0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
>-  0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
>-  0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
>-  0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
>-  0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
>-  0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
>-  0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
>-  0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
>-  0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
>-  0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00,
>-  0x3b, 0x48, 0xc4, 0xed, 0x23, 0x10,
>-  0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
>-  0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
>-  0xa9, 0x15, 0x22, 0xa4, 0x54, 0x80,
>-  0x04, 0xc0, 0x98, 0x13, 0x02, 0x60,
>-  0xd0, 0x1a, 0x03, 0x40, 0x68, 0x00,
>-  0x82, 0x50, 0x4a, 0x09, 0x41, 0x20,
>-  0x21, 0x24, 0x24, 0x84, 0x90, 0x90,
>-  0x0c, 0x21, 0x84, 0x30, 0x86, 0x10,
>-  0x4a, 0x09, 0x41, 0x28, 0x25, 0x00,
>-  0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
>-  0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
>-  0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
>-  0x99, 0x13, 0x22, 0x64, 0x4c, 0x80,
>-  0x05, 0x80, 0xb0, 0x16, 0x02, 0xc0,
>-  0x80, 0xb0, 0x16, 0x02, 0xc0, 0x50,
>-  0x84, 0x50, 0x8a, 0x11, 0x42, 0x20,
>-  0x40, 0x68, 0x0d, 0x01, 0xa0, 0x30,
>-  0x0a, 0x81, 0x50, 0x2a, 0x05, 0x40,
>-  0x68, 0x0d, 0x01, 0xa0, 0x34, 0x00,
>-  0x10, 0x22, 0x04, 0x40, 0x88, 0x10,
>-  0x30, 0x46, 0x08, 0xc1, 0x18, 0x20
>-};
>-
>-const uint8_t kMaskRandom44_42[252] = {
>-  0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
>-  0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
>-  0xa9, 0x15, 0x22, 0xa4, 0x54, 0x80,
>-  0x04, 0xc0, 0x98, 0x13, 0x02, 0x60,
>-  0xd0, 0x1a, 0x03, 0x40, 0x68, 0x00,
>-  0x82, 0x50, 0x4a, 0x09, 0x41, 0x20,
>-  0x21, 0x24, 0x24, 0x84, 0x90, 0x90,
>-  0x0c, 0x21, 0x84, 0x30, 0x86, 0x10,
>-  0x4a, 0x09, 0x41, 0x28, 0x25, 0x00,
>-  0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
>-  0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
>-  0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
>-  0x99, 0x13, 0x22, 0x64, 0x4c, 0x80,
>-  0x05, 0x80, 0xb0, 0x16, 0x02, 0xc0,
>-  0x80, 0xb0, 0x16, 0x02, 0xc0, 0x50,
>-  0x84, 0x50, 0x8a, 0x11, 0x42, 0x20,
>-  0x40, 0x68, 0x0d, 0x01, 0xa0, 0x30,
>-  0x0a, 0x81, 0x50, 0x2a, 0x05, 0x40,
>-  0x68, 0x0d, 0x01, 0xa0, 0x34, 0x00,
>-  0x10, 0x22, 0x04, 0x40, 0x88, 0x10,
>-  0x30, 0x46, 0x08, 0xc1, 0x18, 0x20,
>-  0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
>-  0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
>-  0xa9, 0x15, 0x22, 0xa4, 0x54, 0x80,
>-  0x04, 0xc0, 0x98, 0x13, 0x02, 0x60,
>-  0xd0, 0x1a, 0x03, 0x40, 0x68, 0x00,
>-  0x82, 0x50, 0x4a, 0x09, 0x41, 0x20,
>-  0x21, 0x24, 0x24, 0x84, 0x90, 0x90,
>-  0x0c, 0x21, 0x84, 0x30, 0x86, 0x10,
>-  0x4a, 0x09, 0x41, 0x28, 0x25, 0x00,
>-  0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
>-  0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
>-  0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
>-  0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
>-  0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
>-  0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
>-  0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
>-  0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
>-  0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
>-  0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00,
>-  0x3b, 0x48, 0xc4, 0xed, 0x23, 0x10,
>-  0xd9, 0xc1, 0x6f, 0xa8, 0x1c, 0x90
>-};
>-
>-const uint8_t kMaskRandom44_43[258] = {
>-  0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
>-  0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
>-  0xa9, 0x15, 0x22, 0xa4, 0x54, 0x80,
>-  0x04, 0xc0, 0x98, 0x13, 0x02, 0x60,
>-  0xd0, 0x1a, 0x03, 0x40, 0x68, 0x00,
>-  0x82, 0x50, 0x4a, 0x09, 0x41, 0x20,
>-  0x21, 0x24, 0x24, 0x84, 0x90, 0x90,
>-  0x0c, 0x21, 0x84, 0x30, 0x86, 0x10,
>-  0x4a, 0x09, 0x41, 0x28, 0x25, 0x00,
>-  0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
>-  0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
>-  0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
>-  0x99, 0x13, 0x22, 0x64, 0x4c, 0x80,
>-  0x05, 0x80, 0xb0, 0x16, 0x02, 0xc0,
>-  0x80, 0xb0, 0x16, 0x02, 0xc0, 0x50,
>-  0x84, 0x50, 0x8a, 0x11, 0x42, 0x20,
>-  0x40, 0x68, 0x0d, 0x01, 0xa0, 0x30,
>-  0x0a, 0x81, 0x50, 0x2a, 0x05, 0x40,
>-  0x68, 0x0d, 0x01, 0xa0, 0x34, 0x00,
>-  0x10, 0x22, 0x04, 0x40, 0x88, 0x10,
>-  0x30, 0x46, 0x08, 0xc1, 0x18, 0x20,
>-  0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
>-  0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
>-  0x99, 0x13, 0x22, 0x64, 0x4c, 0x80,
>-  0x05, 0x80, 0xb0, 0x16, 0x02, 0xc0,
>-  0x80, 0xb0, 0x16, 0x02, 0xc0, 0x50,
>-  0x84, 0x50, 0x8a, 0x11, 0x42, 0x20,
>-  0x40, 0x68, 0x0d, 0x01, 0xa0, 0x30,
>-  0x0a, 0x81, 0x50, 0x2a, 0x05, 0x40,
>-  0x68, 0x0d, 0x01, 0xa0, 0x34, 0x00,
>-  0x10, 0x22, 0x04, 0x40, 0x88, 0x10,
>-  0x30, 0x46, 0x08, 0xc1, 0x18, 0x20,
>-  0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
>-  0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
>-  0xa9, 0x15, 0x22, 0xa4, 0x54, 0x80,
>-  0x04, 0xc0, 0x98, 0x13, 0x02, 0x60,
>-  0xd0, 0x1a, 0x03, 0x40, 0x68, 0x00,
>-  0x82, 0x50, 0x4a, 0x09, 0x41, 0x20,
>-  0x21, 0x24, 0x24, 0x84, 0x90, 0x90,
>-  0x0c, 0x21, 0x84, 0x30, 0x86, 0x10,
>-  0x4a, 0x09, 0x41, 0x28, 0x25, 0x00,
>-  0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
>-  0x9e, 0xce, 0x8a, 0x7b, 0x3a, 0x20
>-};
>-
>-const uint8_t kMaskRandom44_44[264] = {
>-  0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
>-  0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
>-  0x99, 0x13, 0x22, 0x64, 0x4c, 0x80,
>-  0x05, 0x80, 0xb0, 0x16, 0x02, 0xc0,
>-  0x80, 0xb0, 0x16, 0x02, 0xc0, 0x50,
>-  0x84, 0x50, 0x8a, 0x11, 0x42, 0x20,
>-  0x40, 0x68, 0x0d, 0x01, 0xa0, 0x30,
>-  0x0a, 0x81, 0x50, 0x2a, 0x05, 0x40,
>-  0x68, 0x0d, 0x01, 0xa0, 0x34, 0x00,
>-  0x10, 0x22, 0x04, 0x40, 0x88, 0x10,
>-  0x30, 0x46, 0x08, 0xc1, 0x18, 0x20,
>-  0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
>-  0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
>-  0xa9, 0x15, 0x22, 0xa4, 0x54, 0x80,
>-  0x04, 0xc0, 0x98, 0x13, 0x02, 0x60,
>-  0xd0, 0x1a, 0x03, 0x40, 0x68, 0x00,
>-  0x82, 0x50, 0x4a, 0x09, 0x41, 0x20,
>-  0x21, 0x24, 0x24, 0x84, 0x90, 0x90,
>-  0x0c, 0x21, 0x84, 0x30, 0x86, 0x10,
>-  0x4a, 0x09, 0x41, 0x28, 0x25, 0x00,
>-  0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
>-  0x9e, 0xce, 0x8a, 0x7b, 0x3a, 0x20,
>-  0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
>-  0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
>-  0xa9, 0x15, 0x22, 0xa4, 0x54, 0x80,
>-  0x04, 0xc0, 0x98, 0x13, 0x02, 0x60,
>-  0xd0, 0x1a, 0x03, 0x40, 0x68, 0x00,
>-  0x82, 0x50, 0x4a, 0x09, 0x41, 0x20,
>-  0x21, 0x24, 0x24, 0x84, 0x90, 0x90,
>-  0x0c, 0x21, 0x84, 0x30, 0x86, 0x10,
>-  0x4a, 0x09, 0x41, 0x28, 0x25, 0x00,
>-  0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
>-  0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
>-  0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
>-  0x99, 0x13, 0x22, 0x64, 0x4c, 0x80,
>-  0x05, 0x80, 0xb0, 0x16, 0x02, 0xc0,
>-  0x80, 0xb0, 0x16, 0x02, 0xc0, 0x50,
>-  0x84, 0x50, 0x8a, 0x11, 0x42, 0x20,
>-  0x40, 0x68, 0x0d, 0x01, 0xa0, 0x30,
>-  0x0a, 0x81, 0x50, 0x2a, 0x05, 0x40,
>-  0x68, 0x0d, 0x01, 0xa0, 0x34, 0x00,
>-  0x10, 0x22, 0x04, 0x40, 0x88, 0x10,
>-  0x30, 0x46, 0x08, 0xc1, 0x18, 0x20,
>-  0xb5, 0x1c, 0x1c, 0x21, 0xac, 0xa0
>-};
>-
>-const uint8_t kMaskRandom44_5[30] = {
>-  0xc6, 0xd8, 0xdb, 0x1b, 0x63, 0x60,
>-  0x63, 0x6c, 0x6d, 0x8d, 0xb1, 0xb0,
>-  0x1d, 0xa3, 0xb4, 0x76, 0x8e, 0xd0,
>-  0xad, 0x55, 0xaa, 0xb5, 0x56, 0xa0,
>-  0xb2, 0xb6, 0x56, 0xca, 0xd9, 0x50
>-};
>-
>-const uint8_t kMaskRandom44_6[36] = {
>-  0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
>-  0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
>-  0xa8, 0x35, 0x06, 0xa0, 0xd4, 0x10,
>-  0xc4, 0xa0, 0x97, 0x12, 0x82, 0x50,
>-  0x03, 0x60, 0x6c, 0x0d, 0x81, 0xb0,
>-  0x90, 0xd2, 0x1a, 0x43, 0x48, 0x60
>-};
>-
>-const uint8_t kMaskRandom44_7[42] = {
>-  0xc6, 0x48, 0xcb, 0x19, 0x23, 0x20,
>-  0x13, 0x26, 0x64, 0x4c, 0x99, 0x90,
>-  0x8d, 0x13, 0xa2, 0x34, 0x4e, 0x80,
>-  0x8b, 0x41, 0x6a, 0x2d, 0x05, 0xa0,
>-  0x52, 0xaa, 0x15, 0x4a, 0xa8, 0x50,
>-  0xa2, 0xd4, 0x1a, 0x8b, 0x50, 0x60,
>-  0x61, 0xa8, 0x2d, 0x86, 0xa0, 0xb0
>-};
>-
>-const uint8_t kMaskRandom44_8[48] = {
>-  0x28, 0x85, 0x38, 0xa2, 0x14, 0xe0,
>-  0x21, 0xf4, 0x04, 0x87, 0xd0, 0x10,
>-  0xe9, 0x1d, 0x03, 0xa4, 0x74, 0x00,
>-  0x17, 0x02, 0xe0, 0x5c, 0x0b, 0x80,
>-  0x83, 0xa0, 0x56, 0x0e, 0x81, 0x50,
>-  0x46, 0x18, 0xe9, 0x18, 0x63, 0xa0,
>-  0x50, 0x6a, 0x0d, 0x41, 0xa8, 0x30,
>-  0x1c, 0x23, 0x84, 0x70, 0x8e, 0x10
>-};
>-
>-const uint8_t kMaskRandom44_9[54] = {
>-  0x44, 0x48, 0xc9, 0x11, 0x23, 0x20,
>-  0x28, 0x2d, 0x0c, 0xa0, 0xb4, 0x30,
>-  0x25, 0x14, 0xa0, 0x94, 0x52, 0x80,
>-  0x59, 0x0a, 0x21, 0x64, 0x28, 0x80,
>-  0x03, 0xa0, 0x34, 0x0e, 0x80, 0xd0,
>-  0xc0, 0xd0, 0x1b, 0x03, 0x40, 0x60,
>-  0xa2, 0x30, 0x46, 0x88, 0xc1, 0x10,
>-  0x14, 0x82, 0xd0, 0x52, 0x0b, 0x40,
>-  0x9a, 0x03, 0x82, 0x68, 0x0e, 0x00
>-};
>-
>-const uint8_t kMaskRandom45_1[6] = {
>-  0xff, 0xff, 0xff, 0xff, 0xff, 0xf8
>-};
>-
>-const uint8_t kMaskRandom45_10[60] = {
>-  0xc0, 0x38, 0x89, 0x91, 0x28, 0xa0,
>-  0x30, 0x0e, 0x29, 0x45, 0x22, 0x88,
>-  0xe8, 0x07, 0x02, 0xa4, 0x40, 0x68,
>-  0x85, 0x08, 0xa8, 0x13, 0x12, 0x10,
>-  0xd0, 0x92, 0x13, 0x40, 0x05, 0x10,
>-  0x86, 0x50, 0x4a, 0x09, 0x00, 0x70,
>-  0x4a, 0x68, 0x0c, 0x84, 0xdc, 0x00,
>-  0x01, 0xa0, 0x74, 0x30, 0x84, 0x88,
>-  0x4c, 0x81, 0x91, 0x28, 0x2b, 0x00,
>-  0x62, 0x24, 0x04, 0x4a, 0xd1, 0x40
>-};
>-
>-const uint8_t kMaskRandom45_11[66] = {
>-  0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
>-  0x99, 0x13, 0x22, 0x64, 0x08, 0x48,
>-  0x05, 0x80, 0xb0, 0x16, 0x00, 0x38,
>-  0x80, 0xb0, 0x16, 0x02, 0x86, 0x08,
>-  0x84, 0x50, 0x8a, 0x11, 0x20, 0x60,
>-  0x40, 0x68, 0x0d, 0x01, 0xb5, 0x00,
>-  0x0a, 0x81, 0x50, 0x2a, 0x43, 0x00,
>-  0x68, 0x0d, 0x01, 0xa0, 0x12, 0x40,
>-  0x10, 0x22, 0x04, 0x40, 0xc4, 0x80,
>-  0x30, 0x46, 0x08, 0xc1, 0x60, 0x10
>-};
>-
>-const uint8_t kMaskRandom45_12[72] = {
>-  0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
>-  0x51, 0x4a, 0x29, 0x45, 0x62, 0x88,
>-  0x0c, 0xa1, 0x94, 0x32, 0x90, 0xc0,
>-  0xa1, 0x34, 0x26, 0x84, 0x89, 0x18,
>-  0x12, 0xa2, 0x54, 0x4a, 0x84, 0x70,
>-  0x8a, 0x51, 0x4a, 0x29, 0x17, 0x00,
>-  0x86, 0x90, 0xd2, 0x1a, 0x29, 0xb0,
>-  0x23, 0x24, 0x64, 0x8c, 0xb2, 0x10,
>-  0x16, 0x22, 0xc4, 0x58, 0x86, 0x60,
>-  0x4c, 0x29, 0x85, 0x30, 0xc1, 0x50,
>-  0x41, 0xc8, 0x39, 0x07, 0x04, 0x98,
>-  0xf4, 0x18, 0x9c, 0x65, 0x5b, 0x90
>-};
>-
>-const uint8_t kMaskRandom45_13[78] = {
>-  0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
>-  0x51, 0x4a, 0x29, 0x45, 0x62, 0x88,
>-  0x0c, 0xa1, 0x94, 0x32, 0x90, 0xc0,
>-  0xa1, 0x34, 0x26, 0x84, 0x89, 0x18,
>-  0x12, 0xa2, 0x54, 0x4a, 0x84, 0x70,
>-  0x8a, 0x51, 0x4a, 0x29, 0x17, 0x00,
>-  0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
>-  0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
>-  0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
>-  0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
>-  0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
>-  0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30
>-};
>-
>-const uint8_t kMaskRandom45_14[84] = {
>-  0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
>-  0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
>-  0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
>-  0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
>-  0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
>-  0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30,
>-  0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
>-  0x51, 0x4a, 0x29, 0x45, 0x62, 0x88,
>-  0x0c, 0xa1, 0x94, 0x32, 0x90, 0xc0,
>-  0xa1, 0x34, 0x26, 0x84, 0x89, 0x18,
>-  0x12, 0xa2, 0x54, 0x4a, 0x84, 0x70,
>-  0x8a, 0x51, 0x4a, 0x29, 0x17, 0x00,
>-  0xc6, 0xca, 0xea, 0x70, 0xfe, 0xc8
>-};
>-
>-const uint8_t kMaskRandom45_15[90] = {
>-  0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
>-  0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
>-  0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
>-  0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
>-  0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
>-  0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30,
>-  0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
>-  0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
>-  0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
>-  0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
>-  0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
>-  0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
>-  0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
>-  0x14, 0x22, 0x84, 0x50, 0xe2, 0x80
>-};
>-
>-const uint8_t kMaskRandom45_16[96] = {
>-  0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
>-  0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
>-  0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
>-  0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
>-  0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
>-  0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
>-  0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
>-  0x14, 0x22, 0x84, 0x50, 0xe2, 0x80,
>-  0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
>-  0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
>-  0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
>-  0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
>-  0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
>-  0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30,
>-  0x86, 0xc1, 0x47, 0xeb, 0x67, 0xd0
>-};
>-
>-const uint8_t kMaskRandom45_17[102] = {
>-  0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
>-  0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
>-  0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
>-  0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
>-  0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
>-  0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
>-  0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
>-  0x14, 0x22, 0x84, 0x50, 0xe2, 0x80,
>-  0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
>-  0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
>-  0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
>-  0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
>-  0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
>-  0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
>-  0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
>-  0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70
>-};
>-
>-const uint8_t kMaskRandom45_18[108] = {
>-  0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
>-  0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
>-  0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
>-  0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
>-  0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
>-  0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
>-  0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
>-  0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70,
>-  0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
>-  0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
>-  0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
>-  0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
>-  0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
>-  0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
>-  0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
>-  0x14, 0x22, 0x84, 0x50, 0xe2, 0x80,
>-  0xe4, 0xd4, 0x6e, 0x08, 0xc9, 0x58
>-};
>-
>-const uint8_t kMaskRandom45_19[114] = {
>-  0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
>-  0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
>-  0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
>-  0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
>-  0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
>-  0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
>-  0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
>-  0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70,
>-  0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
>-  0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
>-  0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
>-  0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
>-  0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
>-  0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
>-  0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
>-  0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
>-  0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
>-  0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40
>-};
>-
>-const uint8_t kMaskRandom45_2[12] = {
>-  0xec, 0xdd, 0x9b, 0xb3, 0x76, 0x60,
>-  0x9b, 0xb3, 0x76, 0x6e, 0xc9, 0xd8
>-};
>-
>-const uint8_t kMaskRandom45_20[120] = {
>-  0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
>-  0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
>-  0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
>-  0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
>-  0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
>-  0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
>-  0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
>-  0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
>-  0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
>-  0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40,
>-  0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
>-  0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
>-  0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
>-  0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
>-  0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
>-  0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
>-  0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
>-  0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70,
>-  0x3b, 0x48, 0xc7, 0x6d, 0x29, 0xe8
>-};
>-
>-const uint8_t kMaskRandom45_21[126] = {
>-  0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
>-  0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
>-  0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
>-  0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
>-  0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
>-  0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
>-  0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
>-  0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
>-  0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
>-  0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40,
>-  0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
>-  0x99, 0x13, 0x22, 0x64, 0x08, 0x48,
>-  0x05, 0x80, 0xb0, 0x16, 0x00, 0x38,
>-  0x80, 0xb0, 0x16, 0x02, 0x86, 0x08,
>-  0x84, 0x50, 0x8a, 0x11, 0x20, 0x60,
>-  0x40, 0x68, 0x0d, 0x01, 0xb5, 0x00,
>-  0x0a, 0x81, 0x50, 0x2a, 0x43, 0x00,
>-  0x68, 0x0d, 0x01, 0xa0, 0x12, 0x40,
>-  0x10, 0x22, 0x04, 0x40, 0xc4, 0x80,
>-  0x30, 0x46, 0x08, 0xc1, 0x60, 0x10
>-};
>-
>-const uint8_t kMaskRandom45_22[132] = {
>-  0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
>-  0x99, 0x13, 0x22, 0x64, 0x08, 0x48,
>-  0x05, 0x80, 0xb0, 0x16, 0x00, 0x38,
>-  0x80, 0xb0, 0x16, 0x02, 0x86, 0x08,
>-  0x84, 0x50, 0x8a, 0x11, 0x20, 0x60,
>-  0x40, 0x68, 0x0d, 0x01, 0xb5, 0x00,
>-  0x0a, 0x81, 0x50, 0x2a, 0x43, 0x00,
>-  0x68, 0x0d, 0x01, 0xa0, 0x12, 0x40,
>-  0x10, 0x22, 0x04, 0x40, 0xc4, 0x80,
>-  0x30, 0x46, 0x08, 0xc1, 0x60, 0x10,
>-  0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
>-  0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
>-  0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
>-  0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
>-  0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
>-  0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
>-  0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
>-  0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
>-  0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
>-  0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40,
>-  0x9e, 0xce, 0x8b, 0xaa, 0x34, 0x68
>-};
>-
>-const uint8_t kMaskRandom45_23[138] = {
>-  0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
>-  0x99, 0x13, 0x22, 0x64, 0x08, 0x48,
>-  0x05, 0x80, 0xb0, 0x16, 0x00, 0x38,
>-  0x80, 0xb0, 0x16, 0x02, 0x86, 0x08,
>-  0x84, 0x50, 0x8a, 0x11, 0x20, 0x60,
>-  0x40, 0x68, 0x0d, 0x01, 0xb5, 0x00,
>-  0x0a, 0x81, 0x50, 0x2a, 0x43, 0x00,
>-  0x68, 0x0d, 0x01, 0xa0, 0x12, 0x40,
>-  0x10, 0x22, 0x04, 0x40, 0xc4, 0x80,
>-  0x30, 0x46, 0x08, 0xc1, 0x60, 0x10,
>-  0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
>-  0x51, 0x4a, 0x29, 0x45, 0x62, 0x88,
>-  0x0c, 0xa1, 0x94, 0x32, 0x90, 0xc0,
>-  0xa1, 0x34, 0x26, 0x84, 0x89, 0x18,
>-  0x12, 0xa2, 0x54, 0x4a, 0x84, 0x70,
>-  0x8a, 0x51, 0x4a, 0x29, 0x17, 0x00,
>-  0x86, 0x90, 0xd2, 0x1a, 0x29, 0xb0,
>-  0x23, 0x24, 0x64, 0x8c, 0xb2, 0x10,
>-  0x16, 0x22, 0xc4, 0x58, 0x86, 0x60,
>-  0x4c, 0x29, 0x85, 0x30, 0xc1, 0x50,
>-  0x41, 0xc8, 0x39, 0x07, 0x04, 0x98,
>-  0xf4, 0x18, 0x9c, 0x65, 0x5b, 0x90
>-};
>-
>-const uint8_t kMaskRandom45_24[144] = {
>-  0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
>-  0x51, 0x4a, 0x29, 0x45, 0x62, 0x88,
>-  0x0c, 0xa1, 0x94, 0x32, 0x90, 0xc0,
>-  0xa1, 0x34, 0x26, 0x84, 0x89, 0x18,
>-  0x12, 0xa2, 0x54, 0x4a, 0x84, 0x70,
>-  0x8a, 0x51, 0x4a, 0x29, 0x17, 0x00,
>-  0x86, 0x90, 0xd2, 0x1a, 0x29, 0xb0,
>-  0x23, 0x24, 0x64, 0x8c, 0xb2, 0x10,
>-  0x16, 0x22, 0xc4, 0x58, 0x86, 0x60,
>-  0x4c, 0x29, 0x85, 0x30, 0xc1, 0x50,
>-  0x41, 0xc8, 0x39, 0x07, 0x04, 0x98,
>-  0xf4, 0x18, 0x9c, 0x65, 0x5b, 0x90,
>-  0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
>-  0x99, 0x13, 0x22, 0x64, 0x08, 0x48,
>-  0x05, 0x80, 0xb0, 0x16, 0x00, 0x38,
>-  0x80, 0xb0, 0x16, 0x02, 0x86, 0x08,
>-  0x84, 0x50, 0x8a, 0x11, 0x20, 0x60,
>-  0x40, 0x68, 0x0d, 0x01, 0xb5, 0x00,
>-  0x0a, 0x81, 0x50, 0x2a, 0x43, 0x00,
>-  0x68, 0x0d, 0x01, 0xa0, 0x12, 0x40,
>-  0x10, 0x22, 0x04, 0x40, 0xc4, 0x80,
>-  0x30, 0x46, 0x08, 0xc1, 0x60, 0x10,
>-  0x95, 0x91, 0xad, 0xd9, 0x86, 0x98
>-};
>-
>-const uint8_t kMaskRandom45_25[150] = {
>-  0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
>-  0x51, 0x4a, 0x29, 0x45, 0x62, 0x88,
>-  0x0c, 0xa1, 0x94, 0x32, 0x90, 0xc0,
>-  0xa1, 0x34, 0x26, 0x84, 0x89, 0x18,
>-  0x12, 0xa2, 0x54, 0x4a, 0x84, 0x70,
>-  0x8a, 0x51, 0x4a, 0x29, 0x17, 0x00,
>-  0x86, 0x90, 0xd2, 0x1a, 0x29, 0xb0,
>-  0x23, 0x24, 0x64, 0x8c, 0xb2, 0x10,
>-  0x16, 0x22, 0xc4, 0x58, 0x86, 0x60,
>-  0x4c, 0x29, 0x85, 0x30, 0xc1, 0x50,
>-  0x41, 0xc8, 0x39, 0x07, 0x04, 0x98,
>-  0xf4, 0x18, 0x9c, 0x65, 0x5b, 0x90,
>-  0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
>-  0x51, 0x4a, 0x29, 0x45, 0x62, 0x88,
>-  0x0c, 0xa1, 0x94, 0x32, 0x90, 0xc0,
>-  0xa1, 0x34, 0x26, 0x84, 0x89, 0x18,
>-  0x12, 0xa2, 0x54, 0x4a, 0x84, 0x70,
>-  0x8a, 0x51, 0x4a, 0x29, 0x17, 0x00,
>-  0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
>-  0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
>-  0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
>-  0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
>-  0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
>-  0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30
>-};
>-
>-const uint8_t kMaskRandom45_26[156] = {
>-  0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
>-  0x51, 0x4a, 0x29, 0x45, 0x62, 0x88,
>-  0x0c, 0xa1, 0x94, 0x32, 0x90, 0xc0,
>-  0xa1, 0x34, 0x26, 0x84, 0x89, 0x18,
>-  0x12, 0xa2, 0x54, 0x4a, 0x84, 0x70,
>-  0x8a, 0x51, 0x4a, 0x29, 0x17, 0x00,
>-  0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
>-  0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
>-  0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
>-  0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
>-  0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
>-  0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30,
>-  0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
>-  0x51, 0x4a, 0x29, 0x45, 0x62, 0x88,
>-  0x0c, 0xa1, 0x94, 0x32, 0x90, 0xc0,
>-  0xa1, 0x34, 0x26, 0x84, 0x89, 0x18,
>-  0x12, 0xa2, 0x54, 0x4a, 0x84, 0x70,
>-  0x8a, 0x51, 0x4a, 0x29, 0x17, 0x00,
>-  0x86, 0x90, 0xd2, 0x1a, 0x29, 0xb0,
>-  0x23, 0x24, 0x64, 0x8c, 0xb2, 0x10,
>-  0x16, 0x22, 0xc4, 0x58, 0x86, 0x60,
>-  0x4c, 0x29, 0x85, 0x30, 0xc1, 0x50,
>-  0x41, 0xc8, 0x39, 0x07, 0x04, 0x98,
>-  0xf4, 0x18, 0x9c, 0x65, 0x5b, 0x90,
>-  0xb0, 0xfd, 0xb2, 0xf3, 0x8a, 0xc0
>-};
>-
>-const uint8_t kMaskRandom45_27[162] = {
>-  0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
>-  0x51, 0x4a, 0x29, 0x45, 0x62, 0x88,
>-  0x0c, 0xa1, 0x94, 0x32, 0x90, 0xc0,
>-  0xa1, 0x34, 0x26, 0x84, 0x89, 0x18,
>-  0x12, 0xa2, 0x54, 0x4a, 0x84, 0x70,
>-  0x8a, 0x51, 0x4a, 0x29, 0x17, 0x00,
>-  0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
>-  0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
>-  0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
>-  0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
>-  0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
>-  0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30,
>-  0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
>-  0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
>-  0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
>-  0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
>-  0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
>-  0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30,
>-  0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
>-  0x51, 0x4a, 0x29, 0x45, 0x62, 0x88,
>-  0x0c, 0xa1, 0x94, 0x32, 0x90, 0xc0,
>-  0xa1, 0x34, 0x26, 0x84, 0x89, 0x18,
>-  0x12, 0xa2, 0x54, 0x4a, 0x84, 0x70,
>-  0x8a, 0x51, 0x4a, 0x29, 0x17, 0x00,
>-  0xc6, 0xca, 0xea, 0x70, 0xfe, 0xc8
>-};
>-
>-const uint8_t kMaskRandom45_28[168] = {
>-  0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
>-  0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
>-  0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
>-  0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
>-  0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
>-  0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30,
>-  0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
>-  0x51, 0x4a, 0x29, 0x45, 0x62, 0x88,
>-  0x0c, 0xa1, 0x94, 0x32, 0x90, 0xc0,
>-  0xa1, 0x34, 0x26, 0x84, 0x89, 0x18,
>-  0x12, 0xa2, 0x54, 0x4a, 0x84, 0x70,
>-  0x8a, 0x51, 0x4a, 0x29, 0x17, 0x00,
>-  0xc6, 0xca, 0xea, 0x70, 0xfe, 0xc8,
>-  0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
>-  0x51, 0x4a, 0x29, 0x45, 0x62, 0x88,
>-  0x0c, 0xa1, 0x94, 0x32, 0x90, 0xc0,
>-  0xa1, 0x34, 0x26, 0x84, 0x89, 0x18,
>-  0x12, 0xa2, 0x54, 0x4a, 0x84, 0x70,
>-  0x8a, 0x51, 0x4a, 0x29, 0x17, 0x00,
>-  0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
>-  0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
>-  0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
>-  0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
>-  0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
>-  0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30,
>-  0x44, 0x46, 0x28, 0xfb, 0x66, 0x80
>-};
>-
>-const uint8_t kMaskRandom45_29[174] = {
>-  0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
>-  0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
>-  0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
>-  0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
>-  0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
>-  0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30,
>-  0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
>-  0x51, 0x4a, 0x29, 0x45, 0x62, 0x88,
>-  0x0c, 0xa1, 0x94, 0x32, 0x90, 0xc0,
>-  0xa1, 0x34, 0x26, 0x84, 0x89, 0x18,
>-  0x12, 0xa2, 0x54, 0x4a, 0x84, 0x70,
>-  0x8a, 0x51, 0x4a, 0x29, 0x17, 0x00,
>-  0xc6, 0xca, 0xea, 0x70, 0xfe, 0xc8,
>-  0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
>-  0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
>-  0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
>-  0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
>-  0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
>-  0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30,
>-  0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
>-  0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
>-  0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
>-  0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
>-  0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
>-  0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
>-  0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
>-  0x14, 0x22, 0x84, 0x50, 0xe2, 0x80
>-};
>-
>-const uint8_t kMaskRandom45_3[18] = {
>-  0xca, 0xd9, 0x5b, 0x2b, 0x4d, 0x90,
>-  0xf1, 0x5e, 0x2b, 0xc5, 0x24, 0xe8,
>-  0xb6, 0x35, 0xc5, 0xd8, 0x9f, 0x40
>-};
>-
>-const uint8_t kMaskRandom45_30[180] = {
>-  0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
>-  0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
>-  0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
>-  0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
>-  0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
>-  0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30,
>-  0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
>-  0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
>-  0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
>-  0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
>-  0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
>-  0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
>-  0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
>-  0x14, 0x22, 0x84, 0x50, 0xe2, 0x80,
>-  0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
>-  0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
>-  0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
>-  0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
>-  0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
>-  0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30,
>-  0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
>-  0x51, 0x4a, 0x29, 0x45, 0x62, 0x88,
>-  0x0c, 0xa1, 0x94, 0x32, 0x90, 0xc0,
>-  0xa1, 0x34, 0x26, 0x84, 0x89, 0x18,
>-  0x12, 0xa2, 0x54, 0x4a, 0x84, 0x70,
>-  0x8a, 0x51, 0x4a, 0x29, 0x17, 0x00,
>-  0xc6, 0xca, 0xea, 0x70, 0xfe, 0xc8,
>-  0x1c, 0xc9, 0x43, 0x25, 0xa7, 0x00
>-};
>-
>-const uint8_t kMaskRandom45_31[186] = {
>-  0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
>-  0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
>-  0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
>-  0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
>-  0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
>-  0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30,
>-  0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
>-  0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
>-  0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
>-  0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
>-  0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
>-  0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
>-  0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
>-  0x14, 0x22, 0x84, 0x50, 0xe2, 0x80,
>-  0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
>-  0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
>-  0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
>-  0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
>-  0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
>-  0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
>-  0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
>-  0x14, 0x22, 0x84, 0x50, 0xe2, 0x80,
>-  0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
>-  0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
>-  0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
>-  0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
>-  0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
>-  0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30,
>-  0x86, 0xc1, 0x47, 0xeb, 0x67, 0xd0
>-};
>-
>-const uint8_t kMaskRandom45_32[192] = {
>-  0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
>-  0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
>-  0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
>-  0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
>-  0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
>-  0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
>-  0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
>-  0x14, 0x22, 0x84, 0x50, 0xe2, 0x80,
>-  0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
>-  0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
>-  0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
>-  0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
>-  0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
>-  0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30,
>-  0x86, 0xc1, 0x47, 0xeb, 0x67, 0xd0,
>-  0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
>-  0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
>-  0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
>-  0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
>-  0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
>-  0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30,
>-  0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
>-  0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
>-  0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
>-  0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
>-  0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
>-  0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
>-  0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
>-  0x14, 0x22, 0x84, 0x50, 0xe2, 0x80,
>-  0x40, 0x7e, 0xc1, 0x30, 0x29, 0x50
>-};
>-
>-const uint8_t kMaskRandom45_33[198] = {
>-  0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
>-  0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
>-  0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
>-  0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
>-  0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
>-  0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
>-  0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
>-  0x14, 0x22, 0x84, 0x50, 0xe2, 0x80,
>-  0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
>-  0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
>-  0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
>-  0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
>-  0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
>-  0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30,
>-  0x86, 0xc1, 0x47, 0xeb, 0x67, 0xd0,
>-  0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
>-  0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
>-  0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
>-  0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
>-  0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
>-  0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
>-  0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
>-  0x14, 0x22, 0x84, 0x50, 0xe2, 0x80,
>-  0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
>-  0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
>-  0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
>-  0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
>-  0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
>-  0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
>-  0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
>-  0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70
>-};
>-
>-const uint8_t kMaskRandom45_34[204] = {
>-  0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
>-  0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
>-  0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
>-  0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
>-  0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
>-  0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
>-  0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
>-  0x14, 0x22, 0x84, 0x50, 0xe2, 0x80,
>-  0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
>-  0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
>-  0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
>-  0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
>-  0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
>-  0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
>-  0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
>-  0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70,
>-  0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
>-  0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
>-  0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
>-  0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
>-  0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
>-  0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
>-  0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
>-  0x14, 0x22, 0x84, 0x50, 0xe2, 0x80,
>-  0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
>-  0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
>-  0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
>-  0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
>-  0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
>-  0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30,
>-  0x86, 0xc1, 0x47, 0xeb, 0x67, 0xd0,
>-  0x1f, 0x78, 0x45, 0x5e, 0x46, 0x50
>-};
>-
>-const uint8_t kMaskRandom45_35[210] = {
>-  0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
>-  0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
>-  0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
>-  0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
>-  0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
>-  0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
>-  0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
>-  0x14, 0x22, 0x84, 0x50, 0xe2, 0x80,
>-  0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
>-  0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
>-  0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
>-  0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
>-  0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
>-  0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
>-  0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
>-  0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70,
>-  0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
>-  0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
>-  0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
>-  0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
>-  0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
>-  0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
>-  0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
>-  0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70,
>-  0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
>-  0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
>-  0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
>-  0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
>-  0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
>-  0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
>-  0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
>-  0x14, 0x22, 0x84, 0x50, 0xe2, 0x80,
>-  0xe4, 0xd4, 0x6e, 0x08, 0xc9, 0x58
>-};
>-
>-const uint8_t kMaskRandom45_36[216] = {
>-  0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
>-  0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
>-  0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
>-  0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
>-  0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
>-  0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
>-  0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
>-  0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70,
>-  0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
>-  0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
>-  0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
>-  0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
>-  0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
>-  0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
>-  0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
>-  0x14, 0x22, 0x84, 0x50, 0xe2, 0x80,
>-  0xe4, 0xd4, 0x6e, 0x08, 0xc9, 0x58,
>-  0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
>-  0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
>-  0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
>-  0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
>-  0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
>-  0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
>-  0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
>-  0x14, 0x22, 0x84, 0x50, 0xe2, 0x80,
>-  0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
>-  0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
>-  0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
>-  0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
>-  0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
>-  0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
>-  0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
>-  0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70,
>-  0xd0, 0x1a, 0xf0, 0x14, 0xf0, 0xe8
>-};
>-
>-const uint8_t kMaskRandom45_37[222] = {
>-  0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
>-  0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
>-  0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
>-  0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
>-  0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
>-  0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
>-  0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
>-  0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70,
>-  0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
>-  0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
>-  0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
>-  0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
>-  0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
>-  0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
>-  0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
>-  0x14, 0x22, 0x84, 0x50, 0xe2, 0x80,
>-  0xe4, 0xd4, 0x6e, 0x08, 0xc9, 0x58,
>-  0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
>-  0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
>-  0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
>-  0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
>-  0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
>-  0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
>-  0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
>-  0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70,
>-  0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
>-  0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
>-  0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
>-  0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
>-  0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
>-  0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
>-  0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
>-  0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
>-  0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
>-  0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40
>-};
>-
>-const uint8_t kMaskRandom45_38[228] = {
>-  0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
>-  0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
>-  0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
>-  0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
>-  0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
>-  0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
>-  0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
>-  0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70,
>-  0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
>-  0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
>-  0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
>-  0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
>-  0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
>-  0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
>-  0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
>-  0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
>-  0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
>-  0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40,
>-  0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
>-  0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
>-  0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
>-  0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
>-  0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
>-  0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
>-  0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
>-  0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70,
>-  0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
>-  0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
>-  0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
>-  0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
>-  0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
>-  0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
>-  0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
>-  0x14, 0x22, 0x84, 0x50, 0xe2, 0x80,
>-  0xe4, 0xd4, 0x6e, 0x08, 0xc9, 0x58,
>-  0x04, 0x67, 0x1b, 0xba, 0x1d, 0xa0
>-};
>-
>-const uint8_t kMaskRandom45_39[234] = {
>-  0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
>-  0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
>-  0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
>-  0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
>-  0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
>-  0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
>-  0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
>-  0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70,
>-  0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
>-  0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
>-  0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
>-  0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
>-  0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
>-  0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
>-  0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
>-  0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
>-  0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
>-  0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40,
>-  0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
>-  0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
>-  0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
>-  0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
>-  0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
>-  0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
>-  0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
>-  0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
>-  0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
>-  0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40,
>-  0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
>-  0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
>-  0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
>-  0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
>-  0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
>-  0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
>-  0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
>-  0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70,
>-  0x3b, 0x48, 0xc7, 0x6d, 0x29, 0xe8
>-};
>-
>-const uint8_t kMaskRandom45_4[24] = {
>-  0xc4, 0xd8, 0x9b, 0x13, 0x45, 0x90,
>-  0x31, 0x66, 0x2c, 0xc5, 0x8a, 0x58,
>-  0x4b, 0x29, 0x65, 0x2c, 0x91, 0x68,
>-  0x2c, 0xa5, 0x94, 0xb2, 0xa2, 0xa8
>-};
>-
>-const uint8_t kMaskRandom45_40[240] = {
>-  0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
>-  0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
>-  0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
>-  0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
>-  0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
>-  0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
>-  0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
>-  0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
>-  0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
>-  0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40,
>-  0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
>-  0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
>-  0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
>-  0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
>-  0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
>-  0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
>-  0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
>-  0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70,
>-  0x3b, 0x48, 0xc7, 0x6d, 0x29, 0xe8,
>-  0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
>-  0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
>-  0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
>-  0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
>-  0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
>-  0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
>-  0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
>-  0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70,
>-  0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
>-  0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
>-  0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
>-  0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
>-  0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
>-  0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
>-  0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
>-  0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
>-  0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
>-  0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40,
>-  0xd9, 0x40, 0x46, 0xe6, 0x4f, 0xd8
>-};
>-
>-const uint8_t kMaskRandom45_41[246] = {
>-  0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
>-  0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
>-  0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
>-  0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
>-  0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
>-  0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
>-  0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
>-  0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
>-  0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
>-  0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40,
>-  0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
>-  0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
>-  0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
>-  0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
>-  0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
>-  0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
>-  0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
>-  0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70,
>-  0x3b, 0x48, 0xc7, 0x6d, 0x29, 0xe8,
>-  0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
>-  0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
>-  0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
>-  0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
>-  0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
>-  0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
>-  0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
>-  0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
>-  0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
>-  0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40,
>-  0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
>-  0x99, 0x13, 0x22, 0x64, 0x08, 0x48,
>-  0x05, 0x80, 0xb0, 0x16, 0x00, 0x38,
>-  0x80, 0xb0, 0x16, 0x02, 0x86, 0x08,
>-  0x84, 0x50, 0x8a, 0x11, 0x20, 0x60,
>-  0x40, 0x68, 0x0d, 0x01, 0xb5, 0x00,
>-  0x0a, 0x81, 0x50, 0x2a, 0x43, 0x00,
>-  0x68, 0x0d, 0x01, 0xa0, 0x12, 0x40,
>-  0x10, 0x22, 0x04, 0x40, 0xc4, 0x80,
>-  0x30, 0x46, 0x08, 0xc1, 0x60, 0x10
>-};
>-
>-const uint8_t kMaskRandom45_42[252] = {
>-  0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
>-  0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
>-  0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
>-  0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
>-  0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
>-  0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
>-  0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
>-  0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
>-  0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
>-  0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40,
>-  0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
>-  0x99, 0x13, 0x22, 0x64, 0x08, 0x48,
>-  0x05, 0x80, 0xb0, 0x16, 0x00, 0x38,
>-  0x80, 0xb0, 0x16, 0x02, 0x86, 0x08,
>-  0x84, 0x50, 0x8a, 0x11, 0x20, 0x60,
>-  0x40, 0x68, 0x0d, 0x01, 0xb5, 0x00,
>-  0x0a, 0x81, 0x50, 0x2a, 0x43, 0x00,
>-  0x68, 0x0d, 0x01, 0xa0, 0x12, 0x40,
>-  0x10, 0x22, 0x04, 0x40, 0xc4, 0x80,
>-  0x30, 0x46, 0x08, 0xc1, 0x60, 0x10,
>-  0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
>-  0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
>-  0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
>-  0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
>-  0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
>-  0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
>-  0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
>-  0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
>-  0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
>-  0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40,
>-  0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
>-  0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
>-  0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
>-  0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
>-  0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
>-  0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
>-  0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
>-  0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70,
>-  0x3b, 0x48, 0xc7, 0x6d, 0x29, 0xe8,
>-  0xac, 0xcc, 0x04, 0x41, 0x97, 0x30
>-};
>-
>-const uint8_t kMaskRandom45_43[258] = {
>-  0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
>-  0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
>-  0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
>-  0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
>-  0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
>-  0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
>-  0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
>-  0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
>-  0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
>-  0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40,
>-  0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
>-  0x99, 0x13, 0x22, 0x64, 0x08, 0x48,
>-  0x05, 0x80, 0xb0, 0x16, 0x00, 0x38,
>-  0x80, 0xb0, 0x16, 0x02, 0x86, 0x08,
>-  0x84, 0x50, 0x8a, 0x11, 0x20, 0x60,
>-  0x40, 0x68, 0x0d, 0x01, 0xb5, 0x00,
>-  0x0a, 0x81, 0x50, 0x2a, 0x43, 0x00,
>-  0x68, 0x0d, 0x01, 0xa0, 0x12, 0x40,
>-  0x10, 0x22, 0x04, 0x40, 0xc4, 0x80,
>-  0x30, 0x46, 0x08, 0xc1, 0x60, 0x10,
>-  0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
>-  0x99, 0x13, 0x22, 0x64, 0x08, 0x48,
>-  0x05, 0x80, 0xb0, 0x16, 0x00, 0x38,
>-  0x80, 0xb0, 0x16, 0x02, 0x86, 0x08,
>-  0x84, 0x50, 0x8a, 0x11, 0x20, 0x60,
>-  0x40, 0x68, 0x0d, 0x01, 0xb5, 0x00,
>-  0x0a, 0x81, 0x50, 0x2a, 0x43, 0x00,
>-  0x68, 0x0d, 0x01, 0xa0, 0x12, 0x40,
>-  0x10, 0x22, 0x04, 0x40, 0xc4, 0x80,
>-  0x30, 0x46, 0x08, 0xc1, 0x60, 0x10,
>-  0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
>-  0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
>-  0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
>-  0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
>-  0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
>-  0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
>-  0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
>-  0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
>-  0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
>-  0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40,
>-  0x9e, 0xce, 0x8b, 0xaa, 0x34, 0x68
>-};
>-
>-const uint8_t kMaskRandom45_44[264] = {
>-  0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
>-  0x99, 0x13, 0x22, 0x64, 0x08, 0x48,
>-  0x05, 0x80, 0xb0, 0x16, 0x00, 0x38,
>-  0x80, 0xb0, 0x16, 0x02, 0x86, 0x08,
>-  0x84, 0x50, 0x8a, 0x11, 0x20, 0x60,
>-  0x40, 0x68, 0x0d, 0x01, 0xb5, 0x00,
>-  0x0a, 0x81, 0x50, 0x2a, 0x43, 0x00,
>-  0x68, 0x0d, 0x01, 0xa0, 0x12, 0x40,
>-  0x10, 0x22, 0x04, 0x40, 0xc4, 0x80,
>-  0x30, 0x46, 0x08, 0xc1, 0x60, 0x10,
>-  0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
>-  0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
>-  0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
>-  0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
>-  0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
>-  0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
>-  0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
>-  0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
>-  0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
>-  0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40,
>-  0x9e, 0xce, 0x8b, 0xaa, 0x34, 0x68,
>-  0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
>-  0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
>-  0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
>-  0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
>-  0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
>-  0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
>-  0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
>-  0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
>-  0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
>-  0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40,
>-  0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
>-  0x99, 0x13, 0x22, 0x64, 0x08, 0x48,
>-  0x05, 0x80, 0xb0, 0x16, 0x00, 0x38,
>-  0x80, 0xb0, 0x16, 0x02, 0x86, 0x08,
>-  0x84, 0x50, 0x8a, 0x11, 0x20, 0x60,
>-  0x40, 0x68, 0x0d, 0x01, 0xb5, 0x00,
>-  0x0a, 0x81, 0x50, 0x2a, 0x43, 0x00,
>-  0x68, 0x0d, 0x01, 0xa0, 0x12, 0x40,
>-  0x10, 0x22, 0x04, 0x40, 0xc4, 0x80,
>-  0x30, 0x46, 0x08, 0xc1, 0x60, 0x10,
>-  0xf8, 0x40, 0xe3, 0x2e, 0x16, 0x00
>-};
>-
>-const uint8_t kMaskRandom45_45[270] = {
>-  0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
>-  0x99, 0x13, 0x22, 0x64, 0x08, 0x48,
>-  0x05, 0x80, 0xb0, 0x16, 0x00, 0x38,
>-  0x80, 0xb0, 0x16, 0x02, 0x86, 0x08,
>-  0x84, 0x50, 0x8a, 0x11, 0x20, 0x60,
>-  0x40, 0x68, 0x0d, 0x01, 0xb5, 0x00,
>-  0x0a, 0x81, 0x50, 0x2a, 0x43, 0x00,
>-  0x68, 0x0d, 0x01, 0xa0, 0x12, 0x40,
>-  0x10, 0x22, 0x04, 0x40, 0xc4, 0x80,
>-  0x30, 0x46, 0x08, 0xc1, 0x60, 0x10,
>-  0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
>-  0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
>-  0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
>-  0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
>-  0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
>-  0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
>-  0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
>-  0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
>-  0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
>-  0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40,
>-  0x9e, 0xce, 0x8b, 0xaa, 0x34, 0x68,
>-  0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
>-  0x99, 0x13, 0x22, 0x64, 0x08, 0x48,
>-  0x05, 0x80, 0xb0, 0x16, 0x00, 0x38,
>-  0x80, 0xb0, 0x16, 0x02, 0x86, 0x08,
>-  0x84, 0x50, 0x8a, 0x11, 0x20, 0x60,
>-  0x40, 0x68, 0x0d, 0x01, 0xb5, 0x00,
>-  0x0a, 0x81, 0x50, 0x2a, 0x43, 0x00,
>-  0x68, 0x0d, 0x01, 0xa0, 0x12, 0x40,
>-  0x10, 0x22, 0x04, 0x40, 0xc4, 0x80,
>-  0x30, 0x46, 0x08, 0xc1, 0x60, 0x10,
>-  0x64, 0x4c, 0x89, 0x19, 0x08, 0x30,
>-  0x51, 0x4a, 0x28, 0xcc, 0x81, 0x18,
>-  0xa9, 0x15, 0x22, 0x64, 0x20, 0x28,
>-  0x04, 0xc0, 0x98, 0x16, 0x10, 0xc0,
>-  0xd0, 0x1a, 0x02, 0x02, 0xc0, 0x88,
>-  0x82, 0x50, 0x4a, 0x11, 0x0a, 0x40,
>-  0x21, 0x24, 0x25, 0x01, 0xcc, 0x00,
>-  0x0c, 0x21, 0x84, 0x2a, 0x04, 0x48,
>-  0x4a, 0x09, 0x41, 0xa0, 0x31, 0x00,
>-  0x12, 0xa2, 0x54, 0x40, 0x92, 0x10,
>-  0x9e, 0xce, 0x88, 0xc1, 0x45, 0x00,
>-  0xfb, 0x97, 0x5d, 0x7d, 0x42, 0x20
>-};
>-
>-const uint8_t kMaskRandom45_5[30] = {
>-  0xc6, 0xd8, 0xdb, 0x1b, 0x29, 0xb0,
>-  0x63, 0x6c, 0x6d, 0x8d, 0xb2, 0x58,
>-  0x1d, 0xa3, 0xb4, 0x76, 0x87, 0x70,
>-  0xad, 0x55, 0xaa, 0xb5, 0x54, 0xe0,
>-  0xb2, 0xb6, 0x56, 0xca, 0xdc, 0x18
>-};
>-
>-const uint8_t kMaskRandom45_6[36] = {
>-  0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
>-  0x51, 0x4a, 0x29, 0x45, 0x62, 0x88,
>-  0xa8, 0x35, 0x04, 0x32, 0x90, 0xc0,
>-  0xc4, 0xa0, 0x96, 0x84, 0x89, 0x18,
>-  0x03, 0x60, 0x6c, 0x4a, 0x84, 0x70,
>-  0x90, 0xd2, 0x1a, 0x29, 0x17, 0x00
>-};
>-
>-const uint8_t kMaskRandom45_7[42] = {
>-  0xc6, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x13, 0x26, 0x64, 0xcc, 0x90, 0x98,
>-  0x8d, 0x13, 0xa2, 0x46, 0x48, 0x48,
>-  0x8b, 0x41, 0x6a, 0x90, 0x81, 0x28,
>-  0x52, 0xaa, 0x15, 0x42, 0x83, 0x50,
>-  0xa2, 0xd4, 0x1a, 0x13, 0x16, 0x00,
>-  0x61, 0xa8, 0x2c, 0x25, 0xc4, 0x30
>-};
>-
>-const uint8_t kMaskRandom45_8[48] = {
>-  0x28, 0x85, 0x38, 0x32, 0x10, 0x30,
>-  0x21, 0xf4, 0x06, 0x01, 0xc0, 0x18,
>-  0xe9, 0x1d, 0x02, 0x82, 0x21, 0x20,
>-  0x17, 0x02, 0xe0, 0x15, 0x00, 0xc8,
>-  0x83, 0xa0, 0x55, 0x0c, 0x0a, 0x08,
>-  0x46, 0x18, 0xe8, 0x68, 0x05, 0x40,
>-  0x50, 0x6a, 0x0d, 0x80, 0x9c, 0x00,
>-  0x1c, 0x23, 0x84, 0x50, 0xe2, 0x80
>-};
>-
>-const uint8_t kMaskRandom45_9[54] = {
>-  0x44, 0x48, 0xc9, 0x19, 0x29, 0xb0,
>-  0x28, 0x2d, 0x0d, 0x89, 0xf2, 0x10,
>-  0x25, 0x14, 0xa2, 0x30, 0x12, 0x20,
>-  0x59, 0x0a, 0x20, 0x05, 0xd0, 0x88,
>-  0x03, 0xa0, 0x34, 0x1e, 0x0c, 0x18,
>-  0xc0, 0xd0, 0x1a, 0x82, 0x01, 0xc8,
>-  0xa2, 0x30, 0x44, 0x62, 0xc5, 0x08,
>-  0x14, 0x82, 0xd2, 0x44, 0x02, 0x48,
>-  0x9a, 0x03, 0x81, 0xe0, 0x00, 0x70
>-};
>-
>-const uint8_t kMaskRandom46_1[6] = {
>-  0xff, 0xff, 0xff, 0xff, 0xff, 0xfc
>-};
>-
>-const uint8_t kMaskRandom46_10[60] = {
>-  0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
>-  0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
>-  0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
>-  0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
>-  0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
>-  0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
>-  0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
>-  0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
>-  0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
>-  0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0
>-};
>-
>-const uint8_t kMaskRandom46_11[66] = {
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
>-  0x99, 0x02, 0x13, 0x32, 0x04, 0x24,
>-  0x05, 0x80, 0x0e, 0x0b, 0x00, 0x1c,
>-  0x80, 0xa1, 0x83, 0x01, 0x43, 0x04,
>-  0x84, 0x48, 0x19, 0x08, 0x90, 0x30,
>-  0x40, 0x6d, 0x40, 0x80, 0xda, 0x80,
>-  0x0a, 0x90, 0xc0, 0x15, 0x21, 0x80,
>-  0x68, 0x04, 0x90, 0xd0, 0x09, 0x20,
>-  0x10, 0x31, 0x20, 0x20, 0x62, 0x40,
>-  0x30, 0x58, 0x04, 0x60, 0xb0, 0x08
>-};
>-
>-const uint8_t kMaskRandom46_12[72] = {
>-  0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
>-  0x51, 0x58, 0xa2, 0xa2, 0xb1, 0x44,
>-  0x0c, 0xa4, 0x30, 0x19, 0x48, 0x60,
>-  0xa1, 0x22, 0x47, 0x42, 0x44, 0x8c,
>-  0x12, 0xa1, 0x1c, 0x25, 0x42, 0x38,
>-  0x8a, 0x45, 0xc1, 0x14, 0x8b, 0x80,
>-  0x86, 0x8a, 0x6d, 0x0d, 0x14, 0xd8,
>-  0x23, 0x2c, 0x84, 0x46, 0x59, 0x08,
>-  0x16, 0x21, 0x98, 0x2c, 0x43, 0x30,
>-  0x4c, 0x30, 0x54, 0x98, 0x60, 0xa8,
>-  0x41, 0xc1, 0x26, 0x83, 0x82, 0x4c,
>-  0x19, 0x56, 0xe4, 0x32, 0xad, 0xc8
>-};
>-
>-const uint8_t kMaskRandom46_13[78] = {
>-  0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
>-  0x51, 0x58, 0xa2, 0xa2, 0xb1, 0x44,
>-  0x0c, 0xa4, 0x30, 0x19, 0x48, 0x60,
>-  0xa1, 0x22, 0x47, 0x42, 0x44, 0x8c,
>-  0x12, 0xa1, 0x1c, 0x25, 0x42, 0x38,
>-  0x8a, 0x45, 0xc1, 0x14, 0x8b, 0x80,
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
>-  0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
>-  0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
>-  0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
>-  0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
>-  0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18
>-};
>-
>-const uint8_t kMaskRandom46_14[84] = {
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
>-  0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
>-  0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
>-  0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
>-  0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
>-  0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18,
>-  0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
>-  0x51, 0x58, 0xa2, 0xa2, 0xb1, 0x44,
>-  0x0c, 0xa4, 0x30, 0x19, 0x48, 0x60,
>-  0xa1, 0x22, 0x47, 0x42, 0x44, 0x8c,
>-  0x12, 0xa1, 0x1c, 0x25, 0x42, 0x38,
>-  0x8a, 0x45, 0xc1, 0x14, 0x8b, 0x80,
>-  0x9c, 0x3f, 0xb3, 0x38, 0x7f, 0x64
>-};
>-
>-const uint8_t kMaskRandom46_15[90] = {
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
>-  0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
>-  0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
>-  0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
>-  0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
>-  0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18,
>-  0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
>-  0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
>-  0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
>-  0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
>-  0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
>-  0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
>-  0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
>-  0x14, 0x38, 0xa0, 0x28, 0x71, 0x40
>-};
>-
>-const uint8_t kMaskRandom46_16[96] = {
>-  0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
>-  0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
>-  0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
>-  0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
>-  0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
>-  0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
>-  0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
>-  0x14, 0x38, 0xa0, 0x28, 0x71, 0x40,
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
>-  0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
>-  0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
>-  0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
>-  0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
>-  0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18,
>-  0xfa, 0xd9, 0xf5, 0xf5, 0xb3, 0xe8
>-};
>-
>-const uint8_t kMaskRandom46_17[102] = {
>-  0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
>-  0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
>-  0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
>-  0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
>-  0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
>-  0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
>-  0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
>-  0x14, 0x38, 0xa0, 0x28, 0x71, 0x40,
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
>-  0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
>-  0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
>-  0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
>-  0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
>-  0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
>-  0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
>-  0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38
>-};
>-
>-const uint8_t kMaskRandom46_18[108] = {
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
>-  0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
>-  0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
>-  0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
>-  0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
>-  0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
>-  0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
>-  0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38,
>-  0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
>-  0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
>-  0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
>-  0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
>-  0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
>-  0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
>-  0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
>-  0x14, 0x38, 0xa0, 0x28, 0x71, 0x40,
>-  0x82, 0x32, 0x57, 0x04, 0x64, 0xac
>-};
>-
>-const uint8_t kMaskRandom46_19[114] = {
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
>-  0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
>-  0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
>-  0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
>-  0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
>-  0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
>-  0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
>-  0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38,
>-  0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
>-  0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
>-  0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
>-  0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
>-  0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
>-  0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
>-  0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
>-  0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
>-  0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
>-  0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0
>-};
>-
>-const uint8_t kMaskRandom46_2[12] = {
>-  0xec, 0xdd, 0x99, 0xd9, 0xbb, 0x30,
>-  0x9b, 0xb2, 0x77, 0x37, 0x64, 0xec
>-};
>-
>-const uint8_t kMaskRandom46_20[120] = {
>-  0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
>-  0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
>-  0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
>-  0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
>-  0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
>-  0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
>-  0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
>-  0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
>-  0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
>-  0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0,
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
>-  0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
>-  0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
>-  0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
>-  0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
>-  0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
>-  0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
>-  0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38,
>-  0xdb, 0x4a, 0x7b, 0xb6, 0x94, 0xf4
>-};
>-
>-const uint8_t kMaskRandom46_21[126] = {
>-  0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
>-  0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
>-  0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
>-  0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
>-  0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
>-  0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
>-  0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
>-  0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
>-  0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
>-  0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0,
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
>-  0x99, 0x02, 0x13, 0x32, 0x04, 0x24,
>-  0x05, 0x80, 0x0e, 0x0b, 0x00, 0x1c,
>-  0x80, 0xa1, 0x83, 0x01, 0x43, 0x04,
>-  0x84, 0x48, 0x19, 0x08, 0x90, 0x30,
>-  0x40, 0x6d, 0x40, 0x80, 0xda, 0x80,
>-  0x0a, 0x90, 0xc0, 0x15, 0x21, 0x80,
>-  0x68, 0x04, 0x90, 0xd0, 0x09, 0x20,
>-  0x10, 0x31, 0x20, 0x20, 0x62, 0x40,
>-  0x30, 0x58, 0x04, 0x60, 0xb0, 0x08
>-};
>-
>-const uint8_t kMaskRandom46_22[132] = {
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
>-  0x99, 0x02, 0x13, 0x32, 0x04, 0x24,
>-  0x05, 0x80, 0x0e, 0x0b, 0x00, 0x1c,
>-  0x80, 0xa1, 0x83, 0x01, 0x43, 0x04,
>-  0x84, 0x48, 0x19, 0x08, 0x90, 0x30,
>-  0x40, 0x6d, 0x40, 0x80, 0xda, 0x80,
>-  0x0a, 0x90, 0xc0, 0x15, 0x21, 0x80,
>-  0x68, 0x04, 0x90, 0xd0, 0x09, 0x20,
>-  0x10, 0x31, 0x20, 0x20, 0x62, 0x40,
>-  0x30, 0x58, 0x04, 0x60, 0xb0, 0x08,
>-  0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
>-  0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
>-  0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
>-  0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
>-  0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
>-  0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
>-  0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
>-  0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
>-  0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
>-  0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0,
>-  0xea, 0x8d, 0x1b, 0xd5, 0x1a, 0x34
>-};
>-
>-const uint8_t kMaskRandom46_23[138] = {
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
>-  0x99, 0x02, 0x13, 0x32, 0x04, 0x24,
>-  0x05, 0x80, 0x0e, 0x0b, 0x00, 0x1c,
>-  0x80, 0xa1, 0x83, 0x01, 0x43, 0x04,
>-  0x84, 0x48, 0x19, 0x08, 0x90, 0x30,
>-  0x40, 0x6d, 0x40, 0x80, 0xda, 0x80,
>-  0x0a, 0x90, 0xc0, 0x15, 0x21, 0x80,
>-  0x68, 0x04, 0x90, 0xd0, 0x09, 0x20,
>-  0x10, 0x31, 0x20, 0x20, 0x62, 0x40,
>-  0x30, 0x58, 0x04, 0x60, 0xb0, 0x08,
>-  0x46, 0x42, 0x0c, 0x8c, 0x84, 0x18,
>-  0x33, 0x20, 0x46, 0x66, 0x40, 0x8c,
>-  0x99, 0x08, 0x0b, 0x32, 0x10, 0x14,
>-  0x05, 0x84, 0x30, 0x0b, 0x08, 0x60,
>-  0x80, 0xb0, 0x23, 0x01, 0x60, 0x44,
>-  0x84, 0x42, 0x91, 0x08, 0x85, 0x20,
>-  0x40, 0x73, 0x00, 0x80, 0xe6, 0x00,
>-  0x0a, 0x81, 0x12, 0x15, 0x02, 0x24,
>-  0x68, 0x0c, 0x40, 0xd0, 0x18, 0x80,
>-  0x10, 0x24, 0x84, 0x20, 0x49, 0x08,
>-  0x30, 0x51, 0x40, 0x60, 0xa2, 0x80,
>-  0x5f, 0x50, 0x88, 0xbe, 0xa1, 0x10
>-};
>-
>-const uint8_t kMaskRandom46_24[144] = {
>-  0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
>-  0x51, 0x58, 0xa2, 0xa2, 0xb1, 0x44,
>-  0x0c, 0xa4, 0x30, 0x19, 0x48, 0x60,
>-  0xa1, 0x22, 0x47, 0x42, 0x44, 0x8c,
>-  0x12, 0xa1, 0x1c, 0x25, 0x42, 0x38,
>-  0x8a, 0x45, 0xc1, 0x14, 0x8b, 0x80,
>-  0x86, 0x8a, 0x6d, 0x0d, 0x14, 0xd8,
>-  0x23, 0x2c, 0x84, 0x46, 0x59, 0x08,
>-  0x16, 0x21, 0x98, 0x2c, 0x43, 0x30,
>-  0x4c, 0x30, 0x54, 0x98, 0x60, 0xa8,
>-  0x41, 0xc1, 0x26, 0x83, 0x82, 0x4c,
>-  0x19, 0x56, 0xe4, 0x32, 0xad, 0xc8,
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
>-  0x99, 0x02, 0x13, 0x32, 0x04, 0x24,
>-  0x05, 0x80, 0x0e, 0x0b, 0x00, 0x1c,
>-  0x80, 0xa1, 0x83, 0x01, 0x43, 0x04,
>-  0x84, 0x48, 0x19, 0x08, 0x90, 0x30,
>-  0x40, 0x6d, 0x40, 0x80, 0xda, 0x80,
>-  0x0a, 0x90, 0xc0, 0x15, 0x21, 0x80,
>-  0x68, 0x04, 0x90, 0xd0, 0x09, 0x20,
>-  0x10, 0x31, 0x20, 0x20, 0x62, 0x40,
>-  0x30, 0x58, 0x04, 0x60, 0xb0, 0x08,
>-  0x18, 0x8b, 0x03, 0xb4, 0x3b, 0x10
>-};
>-
>-const uint8_t kMaskRandom46_25[150] = {
>-  0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
>-  0x51, 0x58, 0xa2, 0xa2, 0xb1, 0x44,
>-  0x0c, 0xa4, 0x30, 0x19, 0x48, 0x60,
>-  0xa1, 0x22, 0x47, 0x42, 0x44, 0x8c,
>-  0x12, 0xa1, 0x1c, 0x25, 0x42, 0x38,
>-  0x8a, 0x45, 0xc1, 0x14, 0x8b, 0x80,
>-  0x86, 0x8a, 0x6d, 0x0d, 0x14, 0xd8,
>-  0x23, 0x2c, 0x84, 0x46, 0x59, 0x08,
>-  0x16, 0x21, 0x98, 0x2c, 0x43, 0x30,
>-  0x4c, 0x30, 0x54, 0x98, 0x60, 0xa8,
>-  0x41, 0xc1, 0x26, 0x83, 0x82, 0x4c,
>-  0x19, 0x56, 0xe4, 0x32, 0xad, 0xc8,
>-  0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
>-  0x51, 0x58, 0xa2, 0xa2, 0xb1, 0x44,
>-  0x0c, 0xa4, 0x30, 0x19, 0x48, 0x60,
>-  0xa1, 0x22, 0x47, 0x42, 0x44, 0x8c,
>-  0x12, 0xa1, 0x1c, 0x25, 0x42, 0x38,
>-  0x8a, 0x45, 0xc1, 0x14, 0x8b, 0x80,
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
>-  0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
>-  0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
>-  0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
>-  0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
>-  0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18
>-};
>-
>-const uint8_t kMaskRandom46_26[156] = {
>-  0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
>-  0x51, 0x58, 0xa2, 0xa2, 0xb1, 0x44,
>-  0x0c, 0xa4, 0x30, 0x19, 0x48, 0x60,
>-  0xa1, 0x22, 0x47, 0x42, 0x44, 0x8c,
>-  0x12, 0xa1, 0x1c, 0x25, 0x42, 0x38,
>-  0x8a, 0x45, 0xc1, 0x14, 0x8b, 0x80,
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
>-  0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
>-  0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
>-  0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
>-  0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
>-  0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18,
>-  0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
>-  0x51, 0x58, 0xa2, 0xa2, 0xb1, 0x44,
>-  0x0c, 0xa4, 0x30, 0x19, 0x48, 0x60,
>-  0xa1, 0x22, 0x47, 0x42, 0x44, 0x8c,
>-  0x12, 0xa1, 0x1c, 0x25, 0x42, 0x38,
>-  0x8a, 0x45, 0xc1, 0x14, 0x8b, 0x80,
>-  0x86, 0x8a, 0x6d, 0x0d, 0x14, 0xd8,
>-  0x23, 0x2c, 0x84, 0x46, 0x59, 0x08,
>-  0x16, 0x21, 0x98, 0x2c, 0x43, 0x30,
>-  0x4c, 0x30, 0x54, 0x98, 0x60, 0xa8,
>-  0x41, 0xc1, 0x26, 0x83, 0x82, 0x4c,
>-  0x19, 0x56, 0xe4, 0x32, 0xad, 0xc8,
>-  0x2d, 0x6d, 0xd2, 0x57, 0xd6, 0x2c
>-};
>-
>-const uint8_t kMaskRandom46_27[162] = {
>-  0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
>-  0x51, 0x58, 0xa2, 0xa2, 0xb1, 0x44,
>-  0x0c, 0xa4, 0x30, 0x19, 0x48, 0x60,
>-  0xa1, 0x22, 0x47, 0x42, 0x44, 0x8c,
>-  0x12, 0xa1, 0x1c, 0x25, 0x42, 0x38,
>-  0x8a, 0x45, 0xc1, 0x14, 0x8b, 0x80,
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
>-  0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
>-  0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
>-  0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
>-  0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
>-  0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18,
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
>-  0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
>-  0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
>-  0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
>-  0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
>-  0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18,
>-  0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
>-  0x51, 0x58, 0xa2, 0xa2, 0xb1, 0x44,
>-  0x0c, 0xa4, 0x30, 0x19, 0x48, 0x60,
>-  0xa1, 0x22, 0x47, 0x42, 0x44, 0x8c,
>-  0x12, 0xa1, 0x1c, 0x25, 0x42, 0x38,
>-  0x8a, 0x45, 0xc1, 0x14, 0x8b, 0x80,
>-  0x9c, 0x3f, 0xb3, 0x38, 0x7f, 0x64
>-};
>-
>-const uint8_t kMaskRandom46_28[168] = {
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
>-  0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
>-  0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
>-  0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
>-  0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
>-  0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18,
>-  0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
>-  0x51, 0x58, 0xa2, 0xa2, 0xb1, 0x44,
>-  0x0c, 0xa4, 0x30, 0x19, 0x48, 0x60,
>-  0xa1, 0x22, 0x47, 0x42, 0x44, 0x8c,
>-  0x12, 0xa1, 0x1c, 0x25, 0x42, 0x38,
>-  0x8a, 0x45, 0xc1, 0x14, 0x8b, 0x80,
>-  0x9c, 0x3f, 0xb3, 0x38, 0x7f, 0x64,
>-  0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
>-  0x51, 0x58, 0xa2, 0xa2, 0xb1, 0x44,
>-  0x0c, 0xa4, 0x30, 0x19, 0x48, 0x60,
>-  0xa1, 0x22, 0x47, 0x42, 0x44, 0x8c,
>-  0x12, 0xa1, 0x1c, 0x25, 0x42, 0x38,
>-  0x8a, 0x45, 0xc1, 0x14, 0x8b, 0x80,
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
>-  0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
>-  0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
>-  0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
>-  0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
>-  0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18,
>-  0xfa, 0x52, 0xf9, 0x72, 0xd9, 0x68
>-};
>-
>-const uint8_t kMaskRandom46_29[174] = {
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
>-  0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
>-  0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
>-  0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
>-  0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
>-  0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18,
>-  0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
>-  0x51, 0x58, 0xa2, 0xa2, 0xb1, 0x44,
>-  0x0c, 0xa4, 0x30, 0x19, 0x48, 0x60,
>-  0xa1, 0x22, 0x47, 0x42, 0x44, 0x8c,
>-  0x12, 0xa1, 0x1c, 0x25, 0x42, 0x38,
>-  0x8a, 0x45, 0xc1, 0x14, 0x8b, 0x80,
>-  0x9c, 0x3f, 0xb3, 0x38, 0x7f, 0x64,
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
>-  0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
>-  0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
>-  0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
>-  0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
>-  0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18,
>-  0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
>-  0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
>-  0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
>-  0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
>-  0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
>-  0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
>-  0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
>-  0x14, 0x38, 0xa0, 0x28, 0x71, 0x40
>-};
>-
>-const uint8_t kMaskRandom46_3[18] = {
>-  0xca, 0xd3, 0x65, 0x95, 0xa6, 0xc8,
>-  0xf1, 0x49, 0x3b, 0xe2, 0x92, 0x74,
>-  0x76, 0x27, 0xd0, 0xec, 0x4f, 0xa0
>-};
>-
>-const uint8_t kMaskRandom46_30[180] = {
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
>-  0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
>-  0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
>-  0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
>-  0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
>-  0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18,
>-  0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
>-  0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
>-  0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
>-  0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
>-  0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
>-  0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
>-  0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
>-  0x14, 0x38, 0xa0, 0x28, 0x71, 0x40,
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
>-  0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
>-  0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
>-  0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
>-  0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
>-  0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18,
>-  0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
>-  0x51, 0x58, 0xa2, 0xa2, 0xb1, 0x44,
>-  0x0c, 0xa4, 0x30, 0x19, 0x48, 0x60,
>-  0xa1, 0x22, 0x47, 0x42, 0x44, 0x8c,
>-  0x12, 0xa1, 0x1c, 0x25, 0x42, 0x38,
>-  0x8a, 0x45, 0xc1, 0x14, 0x8b, 0x80,
>-  0x9c, 0x3f, 0xb3, 0x38, 0x7f, 0x64,
>-  0x99, 0xf6, 0x0a, 0xdd, 0x16, 0xb0
>-};
>-
>-const uint8_t kMaskRandom46_31[186] = {
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
>-  0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
>-  0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
>-  0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
>-  0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
>-  0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18,
>-  0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
>-  0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
>-  0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
>-  0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
>-  0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
>-  0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
>-  0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
>-  0x14, 0x38, 0xa0, 0x28, 0x71, 0x40,
>-  0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
>-  0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
>-  0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
>-  0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
>-  0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
>-  0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
>-  0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
>-  0x14, 0x38, 0xa0, 0x28, 0x71, 0x40,
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
>-  0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
>-  0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
>-  0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
>-  0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
>-  0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18,
>-  0xfa, 0xd9, 0xf5, 0xf5, 0xb3, 0xe8
>-};
>-
>-const uint8_t kMaskRandom46_32[192] = {
>-  0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
>-  0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
>-  0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
>-  0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
>-  0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
>-  0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
>-  0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
>-  0x14, 0x38, 0xa0, 0x28, 0x71, 0x40,
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
>-  0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
>-  0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
>-  0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
>-  0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
>-  0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18,
>-  0xfa, 0xd9, 0xf5, 0xf5, 0xb3, 0xe8,
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
>-  0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
>-  0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
>-  0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
>-  0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
>-  0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18,
>-  0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
>-  0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
>-  0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
>-  0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
>-  0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
>-  0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
>-  0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
>-  0x14, 0x38, 0xa0, 0x28, 0x71, 0x40,
>-  0x69, 0xcd, 0xeb, 0x51, 0xc9, 0xa8
>-};
>-
>-const uint8_t kMaskRandom46_33[198] = {
>-  0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
>-  0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
>-  0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
>-  0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
>-  0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
>-  0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
>-  0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
>-  0x14, 0x38, 0xa0, 0x28, 0x71, 0x40,
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
>-  0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
>-  0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
>-  0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
>-  0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
>-  0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18,
>-  0xfa, 0xd9, 0xf5, 0xf5, 0xb3, 0xe8,
>-  0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
>-  0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
>-  0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
>-  0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
>-  0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
>-  0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
>-  0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
>-  0x14, 0x38, 0xa0, 0x28, 0x71, 0x40,
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
>-  0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
>-  0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
>-  0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
>-  0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
>-  0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
>-  0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
>-  0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38
>-};
>-
>-const uint8_t kMaskRandom46_34[204] = {
>-  0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
>-  0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
>-  0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
>-  0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
>-  0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
>-  0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
>-  0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
>-  0x14, 0x38, 0xa0, 0x28, 0x71, 0x40,
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
>-  0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
>-  0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
>-  0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
>-  0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
>-  0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
>-  0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
>-  0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38,
>-  0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
>-  0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
>-  0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
>-  0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
>-  0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
>-  0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
>-  0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
>-  0x14, 0x38, 0xa0, 0x28, 0x71, 0x40,
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
>-  0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
>-  0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
>-  0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
>-  0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
>-  0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18,
>-  0xfa, 0xd9, 0xf5, 0xf5, 0xb3, 0xe8,
>-  0x60, 0xf0, 0x13, 0xf0, 0x4d, 0xe0
>-};
>-
>-const uint8_t kMaskRandom46_35[210] = {
>-  0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
>-  0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
>-  0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
>-  0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
>-  0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
>-  0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
>-  0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
>-  0x14, 0x38, 0xa0, 0x28, 0x71, 0x40,
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
>-  0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
>-  0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
>-  0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
>-  0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
>-  0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
>-  0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
>-  0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38,
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
>-  0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
>-  0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
>-  0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
>-  0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
>-  0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
>-  0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
>-  0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38,
>-  0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
>-  0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
>-  0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
>-  0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
>-  0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
>-  0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
>-  0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
>-  0x14, 0x38, 0xa0, 0x28, 0x71, 0x40,
>-  0x82, 0x32, 0x57, 0x04, 0x64, 0xac
>-};
>-
>-const uint8_t kMaskRandom46_36[216] = {
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
>-  0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
>-  0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
>-  0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
>-  0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
>-  0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
>-  0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
>-  0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38,
>-  0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
>-  0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
>-  0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
>-  0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
>-  0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
>-  0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
>-  0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
>-  0x14, 0x38, 0xa0, 0x28, 0x71, 0x40,
>-  0x82, 0x32, 0x57, 0x04, 0x64, 0xac,
>-  0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
>-  0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
>-  0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
>-  0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
>-  0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
>-  0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
>-  0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
>-  0x14, 0x38, 0xa0, 0x28, 0x71, 0x40,
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
>-  0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
>-  0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
>-  0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
>-  0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
>-  0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
>-  0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
>-  0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38,
>-  0x6c, 0x3a, 0x45, 0x70, 0xd7, 0x00
>-};
>-
>-const uint8_t kMaskRandom46_37[222] = {
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
>-  0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
>-  0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
>-  0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
>-  0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
>-  0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
>-  0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
>-  0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38,
>-  0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
>-  0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
>-  0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
>-  0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
>-  0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
>-  0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
>-  0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
>-  0x14, 0x38, 0xa0, 0x28, 0x71, 0x40,
>-  0x82, 0x32, 0x57, 0x04, 0x64, 0xac,
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
>-  0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
>-  0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
>-  0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
>-  0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
>-  0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
>-  0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
>-  0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38,
>-  0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
>-  0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
>-  0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
>-  0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
>-  0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
>-  0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
>-  0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
>-  0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
>-  0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
>-  0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0
>-};
>-
>-const uint8_t kMaskRandom46_38[228] = {
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
>-  0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
>-  0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
>-  0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
>-  0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
>-  0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
>-  0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
>-  0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38,
>-  0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
>-  0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
>-  0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
>-  0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
>-  0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
>-  0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
>-  0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
>-  0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
>-  0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
>-  0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0,
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
>-  0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
>-  0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
>-  0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
>-  0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
>-  0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
>-  0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
>-  0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38,
>-  0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
>-  0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
>-  0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
>-  0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
>-  0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
>-  0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
>-  0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
>-  0x14, 0x38, 0xa0, 0x28, 0x71, 0x40,
>-  0x82, 0x32, 0x57, 0x04, 0x64, 0xac,
>-  0x72, 0x2b, 0xa5, 0xd4, 0xb9, 0x30
>-};
>-
>-const uint8_t kMaskRandom46_39[234] = {
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
>-  0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
>-  0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
>-  0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
>-  0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
>-  0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
>-  0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
>-  0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38,
>-  0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
>-  0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
>-  0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
>-  0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
>-  0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
>-  0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
>-  0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
>-  0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
>-  0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
>-  0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0,
>-  0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
>-  0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
>-  0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
>-  0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
>-  0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
>-  0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
>-  0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
>-  0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
>-  0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
>-  0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0,
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
>-  0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
>-  0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
>-  0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
>-  0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
>-  0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
>-  0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
>-  0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38,
>-  0xdb, 0x4a, 0x7b, 0xb6, 0x94, 0xf4
>-};
>-
>-const uint8_t kMaskRandom46_4[24] = {
>-  0xc4, 0xd1, 0x65, 0x89, 0xa2, 0xc8,
>-  0x31, 0x62, 0x96, 0x62, 0xc5, 0x2c,
>-  0x4b, 0x24, 0x5a, 0x96, 0x48, 0xb4,
>-  0x2c, 0xa8, 0xaa, 0x59, 0x51, 0x54
>-};
>-
>-const uint8_t kMaskRandom46_40[240] = {
>-  0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
>-  0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
>-  0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
>-  0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
>-  0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
>-  0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
>-  0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
>-  0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
>-  0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
>-  0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0,
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
>-  0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
>-  0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
>-  0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
>-  0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
>-  0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
>-  0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
>-  0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38,
>-  0xdb, 0x4a, 0x7b, 0xb6, 0x94, 0xf4,
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
>-  0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
>-  0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
>-  0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
>-  0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
>-  0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
>-  0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
>-  0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38,
>-  0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
>-  0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
>-  0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
>-  0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
>-  0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
>-  0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
>-  0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
>-  0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
>-  0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
>-  0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0,
>-  0x7c, 0xc8, 0x93, 0x63, 0x3c, 0x80
>-};
>-
>-const uint8_t kMaskRandom46_41[246] = {
>-  0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
>-  0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
>-  0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
>-  0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
>-  0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
>-  0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
>-  0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
>-  0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
>-  0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
>-  0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0,
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
>-  0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
>-  0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
>-  0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
>-  0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
>-  0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
>-  0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
>-  0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38,
>-  0xdb, 0x4a, 0x7b, 0xb6, 0x94, 0xf4,
>-  0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
>-  0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
>-  0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
>-  0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
>-  0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
>-  0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
>-  0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
>-  0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
>-  0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
>-  0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0,
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
>-  0x99, 0x02, 0x13, 0x32, 0x04, 0x24,
>-  0x05, 0x80, 0x0e, 0x0b, 0x00, 0x1c,
>-  0x80, 0xa1, 0x83, 0x01, 0x43, 0x04,
>-  0x84, 0x48, 0x19, 0x08, 0x90, 0x30,
>-  0x40, 0x6d, 0x40, 0x80, 0xda, 0x80,
>-  0x0a, 0x90, 0xc0, 0x15, 0x21, 0x80,
>-  0x68, 0x04, 0x90, 0xd0, 0x09, 0x20,
>-  0x10, 0x31, 0x20, 0x20, 0x62, 0x40,
>-  0x30, 0x58, 0x04, 0x60, 0xb0, 0x08
>-};
>-
>-const uint8_t kMaskRandom46_42[252] = {
>-  0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
>-  0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
>-  0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
>-  0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
>-  0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
>-  0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
>-  0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
>-  0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
>-  0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
>-  0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0,
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
>-  0x99, 0x02, 0x13, 0x32, 0x04, 0x24,
>-  0x05, 0x80, 0x0e, 0x0b, 0x00, 0x1c,
>-  0x80, 0xa1, 0x83, 0x01, 0x43, 0x04,
>-  0x84, 0x48, 0x19, 0x08, 0x90, 0x30,
>-  0x40, 0x6d, 0x40, 0x80, 0xda, 0x80,
>-  0x0a, 0x90, 0xc0, 0x15, 0x21, 0x80,
>-  0x68, 0x04, 0x90, 0xd0, 0x09, 0x20,
>-  0x10, 0x31, 0x20, 0x20, 0x62, 0x40,
>-  0x30, 0x58, 0x04, 0x60, 0xb0, 0x08,
>-  0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
>-  0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
>-  0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
>-  0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
>-  0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
>-  0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
>-  0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
>-  0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
>-  0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
>-  0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0,
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
>-  0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
>-  0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
>-  0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
>-  0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
>-  0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
>-  0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
>-  0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38,
>-  0xdb, 0x4a, 0x7b, 0xb6, 0x94, 0xf4,
>-  0xfc, 0x6e, 0x89, 0x54, 0x4f, 0x00
>-};
>-
>-const uint8_t kMaskRandom46_43[258] = {
>-  0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
>-  0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
>-  0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
>-  0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
>-  0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
>-  0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
>-  0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
>-  0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
>-  0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
>-  0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0,
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
>-  0x99, 0x02, 0x13, 0x32, 0x04, 0x24,
>-  0x05, 0x80, 0x0e, 0x0b, 0x00, 0x1c,
>-  0x80, 0xa1, 0x83, 0x01, 0x43, 0x04,
>-  0x84, 0x48, 0x19, 0x08, 0x90, 0x30,
>-  0x40, 0x6d, 0x40, 0x80, 0xda, 0x80,
>-  0x0a, 0x90, 0xc0, 0x15, 0x21, 0x80,
>-  0x68, 0x04, 0x90, 0xd0, 0x09, 0x20,
>-  0x10, 0x31, 0x20, 0x20, 0x62, 0x40,
>-  0x30, 0x58, 0x04, 0x60, 0xb0, 0x08,
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
>-  0x99, 0x02, 0x13, 0x32, 0x04, 0x24,
>-  0x05, 0x80, 0x0e, 0x0b, 0x00, 0x1c,
>-  0x80, 0xa1, 0x83, 0x01, 0x43, 0x04,
>-  0x84, 0x48, 0x19, 0x08, 0x90, 0x30,
>-  0x40, 0x6d, 0x40, 0x80, 0xda, 0x80,
>-  0x0a, 0x90, 0xc0, 0x15, 0x21, 0x80,
>-  0x68, 0x04, 0x90, 0xd0, 0x09, 0x20,
>-  0x10, 0x31, 0x20, 0x20, 0x62, 0x40,
>-  0x30, 0x58, 0x04, 0x60, 0xb0, 0x08,
>-  0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
>-  0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
>-  0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
>-  0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
>-  0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
>-  0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
>-  0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
>-  0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
>-  0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
>-  0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0,
>-  0xea, 0x8d, 0x1b, 0xd5, 0x1a, 0x34
>-};
>-
>-const uint8_t kMaskRandom46_44[264] = {
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
>-  0x99, 0x02, 0x13, 0x32, 0x04, 0x24,
>-  0x05, 0x80, 0x0e, 0x0b, 0x00, 0x1c,
>-  0x80, 0xa1, 0x83, 0x01, 0x43, 0x04,
>-  0x84, 0x48, 0x19, 0x08, 0x90, 0x30,
>-  0x40, 0x6d, 0x40, 0x80, 0xda, 0x80,
>-  0x0a, 0x90, 0xc0, 0x15, 0x21, 0x80,
>-  0x68, 0x04, 0x90, 0xd0, 0x09, 0x20,
>-  0x10, 0x31, 0x20, 0x20, 0x62, 0x40,
>-  0x30, 0x58, 0x04, 0x60, 0xb0, 0x08,
>-  0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
>-  0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
>-  0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
>-  0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
>-  0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
>-  0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
>-  0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
>-  0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
>-  0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
>-  0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0,
>-  0xea, 0x8d, 0x1b, 0xd5, 0x1a, 0x34,
>-  0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
>-  0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
>-  0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
>-  0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
>-  0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
>-  0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
>-  0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
>-  0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
>-  0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
>-  0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0,
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
>-  0x99, 0x02, 0x13, 0x32, 0x04, 0x24,
>-  0x05, 0x80, 0x0e, 0x0b, 0x00, 0x1c,
>-  0x80, 0xa1, 0x83, 0x01, 0x43, 0x04,
>-  0x84, 0x48, 0x19, 0x08, 0x90, 0x30,
>-  0x40, 0x6d, 0x40, 0x80, 0xda, 0x80,
>-  0x0a, 0x90, 0xc0, 0x15, 0x21, 0x80,
>-  0x68, 0x04, 0x90, 0xd0, 0x09, 0x20,
>-  0x10, 0x31, 0x20, 0x20, 0x62, 0x40,
>-  0x30, 0x58, 0x04, 0x60, 0xb0, 0x08,
>-  0x1a, 0x8a, 0x00, 0x1c, 0x89, 0x54
>-};
>-
>-const uint8_t kMaskRandom46_45[270] = {
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
>-  0x99, 0x02, 0x13, 0x32, 0x04, 0x24,
>-  0x05, 0x80, 0x0e, 0x0b, 0x00, 0x1c,
>-  0x80, 0xa1, 0x83, 0x01, 0x43, 0x04,
>-  0x84, 0x48, 0x19, 0x08, 0x90, 0x30,
>-  0x40, 0x6d, 0x40, 0x80, 0xda, 0x80,
>-  0x0a, 0x90, 0xc0, 0x15, 0x21, 0x80,
>-  0x68, 0x04, 0x90, 0xd0, 0x09, 0x20,
>-  0x10, 0x31, 0x20, 0x20, 0x62, 0x40,
>-  0x30, 0x58, 0x04, 0x60, 0xb0, 0x08,
>-  0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
>-  0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
>-  0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
>-  0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
>-  0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
>-  0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
>-  0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
>-  0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
>-  0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
>-  0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0,
>-  0xea, 0x8d, 0x1b, 0xd5, 0x1a, 0x34,
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
>-  0x99, 0x02, 0x13, 0x32, 0x04, 0x24,
>-  0x05, 0x80, 0x0e, 0x0b, 0x00, 0x1c,
>-  0x80, 0xa1, 0x83, 0x01, 0x43, 0x04,
>-  0x84, 0x48, 0x19, 0x08, 0x90, 0x30,
>-  0x40, 0x6d, 0x40, 0x80, 0xda, 0x80,
>-  0x0a, 0x90, 0xc0, 0x15, 0x21, 0x80,
>-  0x68, 0x04, 0x90, 0xd0, 0x09, 0x20,
>-  0x10, 0x31, 0x20, 0x20, 0x62, 0x40,
>-  0x30, 0x58, 0x04, 0x60, 0xb0, 0x08,
>-  0x46, 0x42, 0x0c, 0x8c, 0x84, 0x18,
>-  0x33, 0x20, 0x46, 0x66, 0x40, 0x8c,
>-  0x99, 0x08, 0x0b, 0x32, 0x10, 0x14,
>-  0x05, 0x84, 0x30, 0x0b, 0x08, 0x60,
>-  0x80, 0xb0, 0x23, 0x01, 0x60, 0x44,
>-  0x84, 0x42, 0x91, 0x08, 0x85, 0x20,
>-  0x40, 0x73, 0x00, 0x80, 0xe6, 0x00,
>-  0x0a, 0x81, 0x12, 0x15, 0x02, 0x24,
>-  0x68, 0x0c, 0x40, 0xd0, 0x18, 0x80,
>-  0x10, 0x24, 0x84, 0x20, 0x49, 0x08,
>-  0x30, 0x51, 0x40, 0x60, 0xa2, 0x80,
>-  0x5f, 0x50, 0x88, 0xbe, 0xa1, 0x10
>-};
>-
>-const uint8_t kMaskRandom46_46[276] = {
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
>-  0x99, 0x02, 0x13, 0x32, 0x04, 0x24,
>-  0x05, 0x80, 0x0e, 0x0b, 0x00, 0x1c,
>-  0x80, 0xa1, 0x83, 0x01, 0x43, 0x04,
>-  0x84, 0x48, 0x19, 0x08, 0x90, 0x30,
>-  0x40, 0x6d, 0x40, 0x80, 0xda, 0x80,
>-  0x0a, 0x90, 0xc0, 0x15, 0x21, 0x80,
>-  0x68, 0x04, 0x90, 0xd0, 0x09, 0x20,
>-  0x10, 0x31, 0x20, 0x20, 0x62, 0x40,
>-  0x30, 0x58, 0x04, 0x60, 0xb0, 0x08,
>-  0x46, 0x42, 0x0c, 0x8c, 0x84, 0x18,
>-  0x33, 0x20, 0x46, 0x66, 0x40, 0x8c,
>-  0x99, 0x08, 0x0b, 0x32, 0x10, 0x14,
>-  0x05, 0x84, 0x30, 0x0b, 0x08, 0x60,
>-  0x80, 0xb0, 0x23, 0x01, 0x60, 0x44,
>-  0x84, 0x42, 0x91, 0x08, 0x85, 0x20,
>-  0x40, 0x73, 0x00, 0x80, 0xe6, 0x00,
>-  0x0a, 0x81, 0x12, 0x15, 0x02, 0x24,
>-  0x68, 0x0c, 0x40, 0xd0, 0x18, 0x80,
>-  0x10, 0x24, 0x84, 0x20, 0x49, 0x08,
>-  0x30, 0x51, 0x40, 0x60, 0xa2, 0x80,
>-  0x5f, 0x50, 0x88, 0xbe, 0xa1, 0x10,
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
>-  0x99, 0x02, 0x13, 0x32, 0x04, 0x24,
>-  0x05, 0x80, 0x0e, 0x0b, 0x00, 0x1c,
>-  0x80, 0xa1, 0x83, 0x01, 0x43, 0x04,
>-  0x84, 0x48, 0x19, 0x08, 0x90, 0x30,
>-  0x40, 0x6d, 0x40, 0x80, 0xda, 0x80,
>-  0x0a, 0x90, 0xc0, 0x15, 0x21, 0x80,
>-  0x68, 0x04, 0x90, 0xd0, 0x09, 0x20,
>-  0x10, 0x31, 0x20, 0x20, 0x62, 0x40,
>-  0x30, 0x58, 0x04, 0x60, 0xb0, 0x08,
>-  0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
>-  0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
>-  0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
>-  0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
>-  0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
>-  0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
>-  0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
>-  0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
>-  0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
>-  0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0,
>-  0xea, 0x8d, 0x1b, 0xd5, 0x1a, 0x34,
>-  0xd5, 0xdf, 0x59, 0xb9, 0xba, 0x10
>-};
>-
>-const uint8_t kMaskRandom46_5[30] = {
>-  0xc6, 0xca, 0x6d, 0x8d, 0x94, 0xd8,
>-  0x63, 0x6c, 0x96, 0xc6, 0xd9, 0x2c,
>-  0x1d, 0xa1, 0xdc, 0x3b, 0x43, 0xb8,
>-  0xad, 0x55, 0x39, 0x5a, 0xaa, 0x70,
>-  0xb2, 0xb7, 0x07, 0x65, 0x6e, 0x0c
>-};
>-
>-const uint8_t kMaskRandom46_6[36] = {
>-  0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
>-  0x51, 0x58, 0xa2, 0xa2, 0xb1, 0x44,
>-  0x0c, 0xa4, 0x30, 0x19, 0x48, 0x60,
>-  0xa1, 0x22, 0x47, 0x42, 0x44, 0x8c,
>-  0x12, 0xa1, 0x1c, 0x25, 0x42, 0x38,
>-  0x8a, 0x45, 0xc1, 0x14, 0x8b, 0x80
>-};
>-
>-const uint8_t kMaskRandom46_7[42] = {
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
>-  0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
>-  0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
>-  0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
>-  0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
>-  0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18
>-};
>-
>-const uint8_t kMaskRandom46_8[48] = {
>-  0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
>-  0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
>-  0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
>-  0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
>-  0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
>-  0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
>-  0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
>-  0x14, 0x38, 0xa0, 0x28, 0x71, 0x40
>-};
>-
>-const uint8_t kMaskRandom46_9[54] = {
>-  0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
>-  0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
>-  0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
>-  0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
>-  0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
>-  0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
>-  0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
>-  0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
>-  0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38
>-};
>-
>-const uint8_t kMaskRandom47_1[6] = {
>-  0xff, 0xff, 0xff, 0xff, 0xff, 0xfe
>-};
>-
>-const uint8_t kMaskRandom47_10[60] = {
>-  0x64, 0x4a, 0x28, 0x22, 0x8a, 0x28,
>-  0x51, 0x48, 0xa2, 0x8a, 0x68, 0xa6,
>-  0xa9, 0x10, 0x1a, 0x00, 0x90, 0x0a,
>-  0x04, 0xc4, 0x84, 0x21, 0x06, 0x12,
>-  0xd0, 0x01, 0x44, 0x94, 0x29, 0x42,
>-  0x82, 0x40, 0x1c, 0x81, 0x48, 0x14,
>-  0x21, 0x37, 0x01, 0x40, 0xd4, 0x04,
>-  0x0c, 0x21, 0x23, 0x11, 0x01, 0x18,
>-  0x4a, 0x0a, 0xc1, 0x0c, 0x10, 0xc0,
>-  0x12, 0xb4, 0x50, 0xa8, 0x1a, 0x80
>-};
>-
>-const uint8_t kMaskRandom47_11[66] = {
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x68,
>-  0x33, 0x24, 0x27, 0x40, 0x64, 0x22,
>-  0x99, 0x02, 0x12, 0x2a, 0x22, 0x82,
>-  0x05, 0x80, 0x0e, 0x06, 0xa0, 0x2a,
>-  0x80, 0xa1, 0x83, 0x19, 0x11, 0x90,
>-  0x84, 0x48, 0x18, 0x51, 0x05, 0x10,
>-  0x40, 0x6d, 0x40, 0x10, 0x91, 0x08,
>-  0x0a, 0x90, 0xc1, 0x32, 0x03, 0x20,
>-  0x68, 0x04, 0x90, 0x45, 0x24, 0x52,
>-  0x10, 0x31, 0x20, 0x8c, 0x08, 0xc0,
>-  0x30, 0x58, 0x05, 0x18, 0x58, 0x04
>-};
>-
>-const uint8_t kMaskRandom47_12[72] = {
>-  0x64, 0x4a, 0x28, 0x20, 0xc2, 0x0c,
>-  0x51, 0x58, 0xa2, 0x04, 0x60, 0x46,
>-  0x0c, 0xa4, 0x30, 0x80, 0xa8, 0x0a,
>-  0xa1, 0x22, 0x46, 0x43, 0x04, 0x30,
>-  0x12, 0xa1, 0x1d, 0x02, 0x30, 0x22,
>-  0x8a, 0x45, 0xc0, 0x29, 0x02, 0x90,
>-  0x86, 0x8a, 0x6d, 0x30, 0x13, 0x00,
>-  0x23, 0x2c, 0x84, 0x11, 0x21, 0x12,
>-  0x16, 0x21, 0x98, 0xc4, 0x0c, 0x40,
>-  0x4c, 0x30, 0x54, 0x48, 0x44, 0x84,
>-  0x41, 0xc1, 0x27, 0x14, 0x11, 0x40,
>-  0x19, 0x56, 0xe5, 0x08, 0x90, 0x88
>-};
>-
>-const uint8_t kMaskRandom47_13[78] = {
>-  0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
>-  0x51, 0x58, 0xa3, 0x8a, 0x38, 0xa2,
>-  0x0c, 0xa4, 0x30, 0x43, 0x04, 0x30,
>-  0xa1, 0x22, 0x46, 0x24, 0x62, 0x46,
>-  0x12, 0xa1, 0x1c, 0x11, 0xc1, 0x1c,
>-  0x8a, 0x45, 0xc0, 0x5c, 0x05, 0xc0,
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
>-  0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
>-  0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
>-  0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
>-  0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
>-  0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c
>-};
>-
>-const uint8_t kMaskRandom47_14[84] = {
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
>-  0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
>-  0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
>-  0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
>-  0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
>-  0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c,
>-  0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
>-  0x51, 0x58, 0xa3, 0x8a, 0x38, 0xa2,
>-  0x0c, 0xa4, 0x30, 0x43, 0x04, 0x30,
>-  0xa1, 0x22, 0x46, 0x24, 0x62, 0x46,
>-  0x12, 0xa1, 0x1c, 0x11, 0xc1, 0x1c,
>-  0x8a, 0x45, 0xc0, 0x5c, 0x05, 0xc0,
>-  0x9c, 0x3f, 0xb3, 0xe5, 0xad, 0x1c
>-};
>-
>-const uint8_t kMaskRandom47_15[90] = {
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
>-  0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
>-  0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
>-  0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
>-  0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
>-  0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c,
>-  0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
>-  0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
>-  0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
>-  0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
>-  0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
>-  0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
>-  0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
>-  0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0
>-};
>-
>-const uint8_t kMaskRandom47_16[96] = {
>-  0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
>-  0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
>-  0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
>-  0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
>-  0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
>-  0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
>-  0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
>-  0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0,
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
>-  0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
>-  0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
>-  0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
>-  0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
>-  0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c,
>-  0xfa, 0xd9, 0xf5, 0xfe, 0xdc, 0x14
>-};
>-
>-const uint8_t kMaskRandom47_17[102] = {
>-  0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
>-  0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
>-  0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
>-  0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
>-  0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
>-  0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
>-  0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
>-  0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0,
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
>-  0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
>-  0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
>-  0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
>-  0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
>-  0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
>-  0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
>-  0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c
>-};
>-
>-const uint8_t kMaskRandom47_18[108] = {
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
>-  0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
>-  0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
>-  0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
>-  0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
>-  0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
>-  0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
>-  0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c,
>-  0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
>-  0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
>-  0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
>-  0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
>-  0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
>-  0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
>-  0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
>-  0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0,
>-  0x82, 0x32, 0x56, 0x68, 0xa1, 0x5c
>-};
>-
>-const uint8_t kMaskRandom47_19[114] = {
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
>-  0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
>-  0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
>-  0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
>-  0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
>-  0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
>-  0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
>-  0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c,
>-  0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
>-  0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
>-  0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
>-  0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
>-  0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
>-  0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
>-  0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
>-  0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
>-  0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
>-  0x12, 0xb4, 0x51, 0x45, 0x14, 0x50
>-};
>-
>-const uint8_t kMaskRandom47_2[12] = {
>-  0xec, 0xdd, 0x99, 0xd9, 0x9d, 0x98,
>-  0x9b, 0xb2, 0x77, 0x27, 0x72, 0x76
>-};
>-
>-const uint8_t kMaskRandom47_20[120] = {
>-  0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
>-  0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
>-  0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
>-  0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
>-  0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
>-  0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
>-  0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
>-  0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
>-  0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
>-  0x12, 0xb4, 0x51, 0x45, 0x14, 0x50,
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
>-  0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
>-  0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
>-  0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
>-  0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
>-  0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
>-  0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
>-  0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c,
>-  0xdb, 0x4a, 0x7b, 0x31, 0x45, 0x2a
>-};
>-
>-const uint8_t kMaskRandom47_21[126] = {
>-  0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
>-  0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
>-  0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
>-  0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
>-  0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
>-  0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
>-  0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
>-  0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
>-  0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
>-  0x12, 0xb4, 0x51, 0x45, 0x14, 0x50,
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
>-  0x99, 0x02, 0x12, 0x21, 0x22, 0x12,
>-  0x05, 0x80, 0x0e, 0x00, 0xe0, 0x0e,
>-  0x80, 0xa1, 0x82, 0x18, 0x21, 0x82,
>-  0x84, 0x48, 0x18, 0x81, 0x88, 0x18,
>-  0x40, 0x6d, 0x40, 0xd4, 0x0d, 0x40,
>-  0x0a, 0x90, 0xc1, 0x0c, 0x10, 0xc0,
>-  0x68, 0x04, 0x90, 0x49, 0x04, 0x90,
>-  0x10, 0x31, 0x21, 0x12, 0x11, 0x20,
>-  0x30, 0x58, 0x05, 0x80, 0x58, 0x04
>-};
>-
>-const uint8_t kMaskRandom47_22[132] = {
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
>-  0x99, 0x02, 0x12, 0x21, 0x22, 0x12,
>-  0x05, 0x80, 0x0e, 0x00, 0xe0, 0x0e,
>-  0x80, 0xa1, 0x82, 0x18, 0x21, 0x82,
>-  0x84, 0x48, 0x18, 0x81, 0x88, 0x18,
>-  0x40, 0x6d, 0x40, 0xd4, 0x0d, 0x40,
>-  0x0a, 0x90, 0xc1, 0x0c, 0x10, 0xc0,
>-  0x68, 0x04, 0x90, 0x49, 0x04, 0x90,
>-  0x10, 0x31, 0x21, 0x12, 0x11, 0x20,
>-  0x30, 0x58, 0x05, 0x80, 0x58, 0x04,
>-  0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
>-  0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
>-  0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
>-  0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
>-  0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
>-  0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
>-  0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
>-  0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
>-  0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
>-  0x12, 0xb4, 0x51, 0x45, 0x14, 0x50,
>-  0xea, 0x8d, 0x1a, 0x35, 0x55, 0xdc
>-};
>-
>-const uint8_t kMaskRandom47_23[138] = {
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
>-  0x99, 0x02, 0x12, 0x21, 0x22, 0x12,
>-  0x05, 0x80, 0x0e, 0x00, 0xe0, 0x0e,
>-  0x80, 0xa1, 0x82, 0x18, 0x21, 0x82,
>-  0x84, 0x48, 0x18, 0x81, 0x88, 0x18,
>-  0x40, 0x6d, 0x40, 0xd4, 0x0d, 0x40,
>-  0x0a, 0x90, 0xc1, 0x0c, 0x10, 0xc0,
>-  0x68, 0x04, 0x90, 0x49, 0x04, 0x90,
>-  0x10, 0x31, 0x21, 0x12, 0x11, 0x20,
>-  0x30, 0x58, 0x05, 0x80, 0x58, 0x04,
>-  0x46, 0x42, 0x0c, 0x20, 0xc2, 0x0c,
>-  0x33, 0x20, 0x46, 0x04, 0x60, 0x46,
>-  0x99, 0x08, 0x0a, 0x80, 0xa8, 0x0a,
>-  0x05, 0x84, 0x30, 0x43, 0x04, 0x30,
>-  0x80, 0xb0, 0x23, 0x02, 0x30, 0x22,
>-  0x84, 0x42, 0x90, 0x29, 0x02, 0x90,
>-  0x40, 0x73, 0x01, 0x30, 0x13, 0x00,
>-  0x0a, 0x81, 0x12, 0x11, 0x21, 0x12,
>-  0x68, 0x0c, 0x40, 0xc4, 0x0c, 0x40,
>-  0x10, 0x24, 0x84, 0x48, 0x44, 0x84,
>-  0x30, 0x51, 0x41, 0x14, 0x11, 0x40,
>-  0x5f, 0x50, 0x89, 0x08, 0x90, 0x88
>-};
>-
>-const uint8_t kMaskRandom47_24[144] = {
>-  0x64, 0x4a, 0x28, 0x20, 0xc2, 0x0c,
>-  0x51, 0x58, 0xa2, 0x04, 0x60, 0x46,
>-  0x0c, 0xa4, 0x30, 0x80, 0xa8, 0x0a,
>-  0xa1, 0x22, 0x46, 0x43, 0x04, 0x30,
>-  0x12, 0xa1, 0x1d, 0x02, 0x30, 0x22,
>-  0x8a, 0x45, 0xc0, 0x29, 0x02, 0x90,
>-  0x86, 0x8a, 0x6d, 0x30, 0x13, 0x00,
>-  0x23, 0x2c, 0x84, 0x11, 0x21, 0x12,
>-  0x16, 0x21, 0x98, 0xc4, 0x0c, 0x40,
>-  0x4c, 0x30, 0x54, 0x48, 0x44, 0x84,
>-  0x41, 0xc1, 0x27, 0x14, 0x11, 0x40,
>-  0x19, 0x56, 0xe5, 0x08, 0x90, 0x88,
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x68,
>-  0x33, 0x24, 0x27, 0x40, 0x64, 0x22,
>-  0x99, 0x02, 0x12, 0x2a, 0x22, 0x82,
>-  0x05, 0x80, 0x0e, 0x06, 0xa0, 0x2a,
>-  0x80, 0xa1, 0x83, 0x19, 0x11, 0x90,
>-  0x84, 0x48, 0x18, 0x51, 0x05, 0x10,
>-  0x40, 0x6d, 0x40, 0x10, 0x91, 0x08,
>-  0x0a, 0x90, 0xc1, 0x32, 0x03, 0x20,
>-  0x68, 0x04, 0x90, 0x45, 0x24, 0x52,
>-  0x10, 0x31, 0x20, 0x8c, 0x08, 0xc0,
>-  0x30, 0x58, 0x05, 0x18, 0x58, 0x04,
>-  0x27, 0x41, 0x35, 0x57, 0x7e, 0x6a
>-};
>-
>-const uint8_t kMaskRandom47_25[150] = {
>-  0x64, 0x4a, 0x28, 0x20, 0xc2, 0x0c,
>-  0x51, 0x58, 0xa2, 0x04, 0x60, 0x46,
>-  0x0c, 0xa4, 0x30, 0x80, 0xa8, 0x0a,
>-  0xa1, 0x22, 0x46, 0x43, 0x04, 0x30,
>-  0x12, 0xa1, 0x1d, 0x02, 0x30, 0x22,
>-  0x8a, 0x45, 0xc0, 0x29, 0x02, 0x90,
>-  0x86, 0x8a, 0x6d, 0x30, 0x13, 0x00,
>-  0x23, 0x2c, 0x84, 0x11, 0x21, 0x12,
>-  0x16, 0x21, 0x98, 0xc4, 0x0c, 0x40,
>-  0x4c, 0x30, 0x54, 0x48, 0x44, 0x84,
>-  0x41, 0xc1, 0x27, 0x14, 0x11, 0x40,
>-  0x19, 0x56, 0xe5, 0x08, 0x90, 0x88,
>-  0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
>-  0x51, 0x58, 0xa3, 0x8a, 0x38, 0xa2,
>-  0x0c, 0xa4, 0x30, 0x43, 0x04, 0x30,
>-  0xa1, 0x22, 0x46, 0x24, 0x62, 0x46,
>-  0x12, 0xa1, 0x1c, 0x11, 0xc1, 0x1c,
>-  0x8a, 0x45, 0xc0, 0x5c, 0x05, 0xc0,
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
>-  0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
>-  0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
>-  0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
>-  0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
>-  0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c
>-};
>-
>-const uint8_t kMaskRandom47_26[156] = {
>-  0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
>-  0x51, 0x58, 0xa3, 0x8a, 0x38, 0xa2,
>-  0x0c, 0xa4, 0x30, 0x43, 0x04, 0x30,
>-  0xa1, 0x22, 0x46, 0x24, 0x62, 0x46,
>-  0x12, 0xa1, 0x1c, 0x11, 0xc1, 0x1c,
>-  0x8a, 0x45, 0xc0, 0x5c, 0x05, 0xc0,
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
>-  0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
>-  0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
>-  0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
>-  0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
>-  0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c,
>-  0x64, 0x4a, 0x28, 0x20, 0xc2, 0x0c,
>-  0x51, 0x58, 0xa2, 0x04, 0x60, 0x46,
>-  0x0c, 0xa4, 0x30, 0x80, 0xa8, 0x0a,
>-  0xa1, 0x22, 0x46, 0x43, 0x04, 0x30,
>-  0x12, 0xa1, 0x1d, 0x02, 0x30, 0x22,
>-  0x8a, 0x45, 0xc0, 0x29, 0x02, 0x90,
>-  0x86, 0x8a, 0x6d, 0x30, 0x13, 0x00,
>-  0x23, 0x2c, 0x84, 0x11, 0x21, 0x12,
>-  0x16, 0x21, 0x98, 0xc4, 0x0c, 0x40,
>-  0x4c, 0x30, 0x54, 0x48, 0x44, 0x84,
>-  0x41, 0xc1, 0x27, 0x14, 0x11, 0x40,
>-  0x19, 0x56, 0xe5, 0x08, 0x90, 0x88,
>-  0x6c, 0xea, 0xc4, 0x42, 0x20, 0x9e
>-};
>-
>-const uint8_t kMaskRandom47_27[162] = {
>-  0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
>-  0x51, 0x58, 0xa3, 0x8a, 0x38, 0xa2,
>-  0x0c, 0xa4, 0x30, 0x43, 0x04, 0x30,
>-  0xa1, 0x22, 0x46, 0x24, 0x62, 0x46,
>-  0x12, 0xa1, 0x1c, 0x11, 0xc1, 0x1c,
>-  0x8a, 0x45, 0xc0, 0x5c, 0x05, 0xc0,
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
>-  0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
>-  0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
>-  0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
>-  0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
>-  0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c,
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
>-  0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
>-  0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
>-  0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
>-  0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
>-  0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c,
>-  0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
>-  0x51, 0x58, 0xa3, 0x8a, 0x38, 0xa2,
>-  0x0c, 0xa4, 0x30, 0x43, 0x04, 0x30,
>-  0xa1, 0x22, 0x46, 0x24, 0x62, 0x46,
>-  0x12, 0xa1, 0x1c, 0x11, 0xc1, 0x1c,
>-  0x8a, 0x45, 0xc0, 0x5c, 0x05, 0xc0,
>-  0x9c, 0x3f, 0xb3, 0xe5, 0xad, 0x1c
>-};
>-
>-const uint8_t kMaskRandom47_28[168] = {
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
>-  0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
>-  0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
>-  0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
>-  0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
>-  0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c,
>-  0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
>-  0x51, 0x58, 0xa3, 0x8a, 0x38, 0xa2,
>-  0x0c, 0xa4, 0x30, 0x43, 0x04, 0x30,
>-  0xa1, 0x22, 0x46, 0x24, 0x62, 0x46,
>-  0x12, 0xa1, 0x1c, 0x11, 0xc1, 0x1c,
>-  0x8a, 0x45, 0xc0, 0x5c, 0x05, 0xc0,
>-  0x9c, 0x3f, 0xb3, 0xe5, 0xad, 0x1c,
>-  0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
>-  0x51, 0x58, 0xa3, 0x8a, 0x38, 0xa2,
>-  0x0c, 0xa4, 0x30, 0x43, 0x04, 0x30,
>-  0xa1, 0x22, 0x46, 0x24, 0x62, 0x46,
>-  0x12, 0xa1, 0x1c, 0x11, 0xc1, 0x1c,
>-  0x8a, 0x45, 0xc0, 0x5c, 0x05, 0xc0,
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
>-  0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
>-  0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
>-  0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
>-  0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
>-  0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c,
>-  0x86, 0x1e, 0xa6, 0xaf, 0x3d, 0x04
>-};
>-
>-const uint8_t kMaskRandom47_29[174] = {
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
>-  0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
>-  0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
>-  0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
>-  0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
>-  0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c,
>-  0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
>-  0x51, 0x58, 0xa3, 0x8a, 0x38, 0xa2,
>-  0x0c, 0xa4, 0x30, 0x43, 0x04, 0x30,
>-  0xa1, 0x22, 0x46, 0x24, 0x62, 0x46,
>-  0x12, 0xa1, 0x1c, 0x11, 0xc1, 0x1c,
>-  0x8a, 0x45, 0xc0, 0x5c, 0x05, 0xc0,
>-  0x9c, 0x3f, 0xb3, 0xe5, 0xad, 0x1c,
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
>-  0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
>-  0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
>-  0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
>-  0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
>-  0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c,
>-  0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
>-  0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
>-  0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
>-  0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
>-  0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
>-  0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
>-  0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
>-  0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0
>-};
>-
>-const uint8_t kMaskRandom47_3[18] = {
>-  0xca, 0xd3, 0x65, 0x36, 0x53, 0x64,
>-  0xf1, 0x49, 0x3a, 0x93, 0xa9, 0x3a,
>-  0x76, 0x27, 0xd0, 0x7d, 0x07, 0xd0
>-};
>-
>-const uint8_t kMaskRandom47_30[180] = {
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
>-  0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
>-  0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
>-  0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
>-  0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
>-  0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c,
>-  0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
>-  0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
>-  0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
>-  0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
>-  0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
>-  0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
>-  0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
>-  0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0,
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
>-  0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
>-  0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
>-  0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
>-  0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
>-  0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c,
>-  0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
>-  0x51, 0x58, 0xa3, 0x8a, 0x38, 0xa2,
>-  0x0c, 0xa4, 0x30, 0x43, 0x04, 0x30,
>-  0xa1, 0x22, 0x46, 0x24, 0x62, 0x46,
>-  0x12, 0xa1, 0x1c, 0x11, 0xc1, 0x1c,
>-  0x8a, 0x45, 0xc0, 0x5c, 0x05, 0xc0,
>-  0x9c, 0x3f, 0xb3, 0xe5, 0xad, 0x1c,
>-  0x97, 0x43, 0x63, 0xc6, 0x09, 0x9c
>-};
>-
>-const uint8_t kMaskRandom47_31[186] = {
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
>-  0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
>-  0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
>-  0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
>-  0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
>-  0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c,
>-  0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
>-  0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
>-  0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
>-  0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
>-  0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
>-  0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
>-  0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
>-  0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0,
>-  0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
>-  0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
>-  0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
>-  0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
>-  0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
>-  0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
>-  0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
>-  0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0,
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
>-  0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
>-  0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
>-  0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
>-  0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
>-  0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c,
>-  0xfa, 0xd9, 0xf5, 0xfe, 0xdc, 0x14
>-};
>-
>-const uint8_t kMaskRandom47_32[192] = {
>-  0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
>-  0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
>-  0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
>-  0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
>-  0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
>-  0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
>-  0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
>-  0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0,
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
>-  0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
>-  0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
>-  0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
>-  0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
>-  0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c,
>-  0xfa, 0xd9, 0xf5, 0xfe, 0xdc, 0x14,
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
>-  0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
>-  0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
>-  0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
>-  0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
>-  0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c,
>-  0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
>-  0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
>-  0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
>-  0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
>-  0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
>-  0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
>-  0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
>-  0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0,
>-  0xe5, 0x50, 0x45, 0x63, 0xc2, 0xf4
>-};
>-
>-const uint8_t kMaskRandom47_33[198] = {
>-  0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
>-  0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
>-  0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
>-  0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
>-  0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
>-  0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
>-  0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
>-  0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0,
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
>-  0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
>-  0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
>-  0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
>-  0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
>-  0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c,
>-  0xfa, 0xd9, 0xf5, 0xfe, 0xdc, 0x14,
>-  0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
>-  0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
>-  0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
>-  0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
>-  0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
>-  0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
>-  0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
>-  0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0,
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
>-  0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
>-  0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
>-  0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
>-  0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
>-  0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
>-  0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
>-  0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c
>-};
>-
>-const uint8_t kMaskRandom47_34[204] = {
>-  0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
>-  0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
>-  0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
>-  0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
>-  0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
>-  0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
>-  0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
>-  0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0,
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
>-  0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
>-  0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
>-  0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
>-  0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
>-  0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
>-  0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
>-  0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c,
>-  0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
>-  0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
>-  0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
>-  0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
>-  0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
>-  0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
>-  0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
>-  0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0,
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
>-  0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
>-  0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
>-  0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
>-  0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
>-  0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c,
>-  0xfa, 0xd9, 0xf5, 0xfe, 0xdc, 0x14,
>-  0xef, 0xbb, 0xa6, 0x23, 0x5c, 0xbe
>-};
>-
>-const uint8_t kMaskRandom47_35[210] = {
>-  0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
>-  0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
>-  0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
>-  0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
>-  0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
>-  0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
>-  0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
>-  0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0,
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
>-  0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
>-  0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
>-  0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
>-  0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
>-  0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
>-  0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
>-  0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c,
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
>-  0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
>-  0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
>-  0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
>-  0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
>-  0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
>-  0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
>-  0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c,
>-  0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
>-  0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
>-  0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
>-  0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
>-  0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
>-  0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
>-  0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
>-  0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0,
>-  0x82, 0x32, 0x56, 0x68, 0xa1, 0x5c
>-};
>-
>-const uint8_t kMaskRandom47_36[216] = {
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
>-  0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
>-  0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
>-  0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
>-  0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
>-  0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
>-  0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
>-  0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c,
>-  0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
>-  0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
>-  0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
>-  0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
>-  0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
>-  0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
>-  0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
>-  0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0,
>-  0x82, 0x32, 0x56, 0x68, 0xa1, 0x5c,
>-  0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
>-  0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
>-  0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
>-  0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
>-  0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
>-  0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
>-  0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
>-  0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0,
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
>-  0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
>-  0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
>-  0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
>-  0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
>-  0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
>-  0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
>-  0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c,
>-  0x0e, 0xd7, 0x38, 0x20, 0x87, 0x66
>-};
>-
>-const uint8_t kMaskRandom47_37[222] = {
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
>-  0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
>-  0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
>-  0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
>-  0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
>-  0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
>-  0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
>-  0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c,
>-  0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
>-  0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
>-  0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
>-  0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
>-  0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
>-  0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
>-  0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
>-  0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0,
>-  0x82, 0x32, 0x56, 0x68, 0xa1, 0x5c,
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
>-  0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
>-  0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
>-  0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
>-  0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
>-  0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
>-  0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
>-  0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c,
>-  0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
>-  0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
>-  0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
>-  0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
>-  0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
>-  0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
>-  0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
>-  0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
>-  0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
>-  0x12, 0xb4, 0x51, 0x45, 0x14, 0x50
>-};
>-
>-const uint8_t kMaskRandom47_38[228] = {
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
>-  0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
>-  0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
>-  0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
>-  0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
>-  0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
>-  0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
>-  0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c,
>-  0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
>-  0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
>-  0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
>-  0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
>-  0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
>-  0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
>-  0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
>-  0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
>-  0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
>-  0x12, 0xb4, 0x51, 0x45, 0x14, 0x50,
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
>-  0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
>-  0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
>-  0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
>-  0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
>-  0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
>-  0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
>-  0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c,
>-  0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
>-  0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
>-  0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
>-  0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
>-  0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
>-  0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
>-  0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
>-  0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0,
>-  0x82, 0x32, 0x56, 0x68, 0xa1, 0x5c,
>-  0x7b, 0x47, 0xa5, 0xde, 0x9a, 0xd4
>-};
>-
>-const uint8_t kMaskRandom47_39[234] = {
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
>-  0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
>-  0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
>-  0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
>-  0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
>-  0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
>-  0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
>-  0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c,
>-  0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
>-  0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
>-  0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
>-  0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
>-  0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
>-  0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
>-  0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
>-  0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
>-  0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
>-  0x12, 0xb4, 0x51, 0x45, 0x14, 0x50,
>-  0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
>-  0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
>-  0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
>-  0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
>-  0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
>-  0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
>-  0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
>-  0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
>-  0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
>-  0x12, 0xb4, 0x51, 0x45, 0x14, 0x50,
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
>-  0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
>-  0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
>-  0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
>-  0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
>-  0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
>-  0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
>-  0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c,
>-  0xdb, 0x4a, 0x7b, 0x31, 0x45, 0x2a
>-};
>-
>-const uint8_t kMaskRandom47_4[24] = {
>-  0xc4, 0xd1, 0x65, 0x16, 0x51, 0x64,
>-  0x31, 0x62, 0x96, 0x29, 0x62, 0x96,
>-  0x4b, 0x24, 0x5a, 0x45, 0xa4, 0x5a,
>-  0x2c, 0xa8, 0xaa, 0x8a, 0xa8, 0xaa
>-};
>-
>-const uint8_t kMaskRandom47_40[240] = {
>-  0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
>-  0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
>-  0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
>-  0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
>-  0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
>-  0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
>-  0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
>-  0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
>-  0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
>-  0x12, 0xb4, 0x51, 0x45, 0x14, 0x50,
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
>-  0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
>-  0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
>-  0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
>-  0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
>-  0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
>-  0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
>-  0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c,
>-  0xdb, 0x4a, 0x7b, 0x31, 0x45, 0x2a,
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
>-  0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
>-  0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
>-  0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
>-  0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
>-  0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
>-  0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
>-  0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c,
>-  0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
>-  0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
>-  0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
>-  0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
>-  0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
>-  0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
>-  0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
>-  0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
>-  0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
>-  0x12, 0xb4, 0x51, 0x45, 0x14, 0x50,
>-  0xc4, 0xae, 0x5e, 0x33, 0xf5, 0x1a
>-};
>-
>-const uint8_t kMaskRandom47_41[246] = {
>-  0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
>-  0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
>-  0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
>-  0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
>-  0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
>-  0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
>-  0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
>-  0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
>-  0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
>-  0x12, 0xb4, 0x51, 0x45, 0x14, 0x50,
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
>-  0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
>-  0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
>-  0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
>-  0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
>-  0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
>-  0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
>-  0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c,
>-  0xdb, 0x4a, 0x7b, 0x31, 0x45, 0x2a,
>-  0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
>-  0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
>-  0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
>-  0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
>-  0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
>-  0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
>-  0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
>-  0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
>-  0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
>-  0x12, 0xb4, 0x51, 0x45, 0x14, 0x50,
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
>-  0x99, 0x02, 0x12, 0x21, 0x22, 0x12,
>-  0x05, 0x80, 0x0e, 0x00, 0xe0, 0x0e,
>-  0x80, 0xa1, 0x82, 0x18, 0x21, 0x82,
>-  0x84, 0x48, 0x18, 0x81, 0x88, 0x18,
>-  0x40, 0x6d, 0x40, 0xd4, 0x0d, 0x40,
>-  0x0a, 0x90, 0xc1, 0x0c, 0x10, 0xc0,
>-  0x68, 0x04, 0x90, 0x49, 0x04, 0x90,
>-  0x10, 0x31, 0x21, 0x12, 0x11, 0x20,
>-  0x30, 0x58, 0x05, 0x80, 0x58, 0x04
>-};
>-
>-const uint8_t kMaskRandom47_42[252] = {
>-  0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
>-  0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
>-  0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
>-  0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
>-  0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
>-  0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
>-  0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
>-  0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
>-  0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
>-  0x12, 0xb4, 0x51, 0x45, 0x14, 0x50,
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
>-  0x99, 0x02, 0x12, 0x21, 0x22, 0x12,
>-  0x05, 0x80, 0x0e, 0x00, 0xe0, 0x0e,
>-  0x80, 0xa1, 0x82, 0x18, 0x21, 0x82,
>-  0x84, 0x48, 0x18, 0x81, 0x88, 0x18,
>-  0x40, 0x6d, 0x40, 0xd4, 0x0d, 0x40,
>-  0x0a, 0x90, 0xc1, 0x0c, 0x10, 0xc0,
>-  0x68, 0x04, 0x90, 0x49, 0x04, 0x90,
>-  0x10, 0x31, 0x21, 0x12, 0x11, 0x20,
>-  0x30, 0x58, 0x05, 0x80, 0x58, 0x04,
>-  0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
>-  0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
>-  0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
>-  0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
>-  0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
>-  0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
>-  0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
>-  0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
>-  0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
>-  0x12, 0xb4, 0x51, 0x45, 0x14, 0x50,
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
>-  0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
>-  0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
>-  0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
>-  0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
>-  0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
>-  0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
>-  0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c,
>-  0xdb, 0x4a, 0x7b, 0x31, 0x45, 0x2a,
>-  0x3c, 0xb0, 0x36, 0x3b, 0x14, 0xa2
>-};
>-
>-const uint8_t kMaskRandom47_43[258] = {
>-  0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
>-  0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
>-  0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
>-  0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
>-  0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
>-  0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
>-  0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
>-  0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
>-  0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
>-  0x12, 0xb4, 0x51, 0x45, 0x14, 0x50,
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
>-  0x99, 0x02, 0x12, 0x21, 0x22, 0x12,
>-  0x05, 0x80, 0x0e, 0x00, 0xe0, 0x0e,
>-  0x80, 0xa1, 0x82, 0x18, 0x21, 0x82,
>-  0x84, 0x48, 0x18, 0x81, 0x88, 0x18,
>-  0x40, 0x6d, 0x40, 0xd4, 0x0d, 0x40,
>-  0x0a, 0x90, 0xc1, 0x0c, 0x10, 0xc0,
>-  0x68, 0x04, 0x90, 0x49, 0x04, 0x90,
>-  0x10, 0x31, 0x21, 0x12, 0x11, 0x20,
>-  0x30, 0x58, 0x05, 0x80, 0x58, 0x04,
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
>-  0x99, 0x02, 0x12, 0x21, 0x22, 0x12,
>-  0x05, 0x80, 0x0e, 0x00, 0xe0, 0x0e,
>-  0x80, 0xa1, 0x82, 0x18, 0x21, 0x82,
>-  0x84, 0x48, 0x18, 0x81, 0x88, 0x18,
>-  0x40, 0x6d, 0x40, 0xd4, 0x0d, 0x40,
>-  0x0a, 0x90, 0xc1, 0x0c, 0x10, 0xc0,
>-  0x68, 0x04, 0x90, 0x49, 0x04, 0x90,
>-  0x10, 0x31, 0x21, 0x12, 0x11, 0x20,
>-  0x30, 0x58, 0x05, 0x80, 0x58, 0x04,
>-  0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
>-  0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
>-  0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
>-  0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
>-  0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
>-  0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
>-  0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
>-  0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
>-  0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
>-  0x12, 0xb4, 0x51, 0x45, 0x14, 0x50,
>-  0xea, 0x8d, 0x1a, 0x35, 0x55, 0xdc
>-};
>-
>-const uint8_t kMaskRandom47_44[264] = {
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
>-  0x99, 0x02, 0x12, 0x21, 0x22, 0x12,
>-  0x05, 0x80, 0x0e, 0x00, 0xe0, 0x0e,
>-  0x80, 0xa1, 0x82, 0x18, 0x21, 0x82,
>-  0x84, 0x48, 0x18, 0x81, 0x88, 0x18,
>-  0x40, 0x6d, 0x40, 0xd4, 0x0d, 0x40,
>-  0x0a, 0x90, 0xc1, 0x0c, 0x10, 0xc0,
>-  0x68, 0x04, 0x90, 0x49, 0x04, 0x90,
>-  0x10, 0x31, 0x21, 0x12, 0x11, 0x20,
>-  0x30, 0x58, 0x05, 0x80, 0x58, 0x04,
>-  0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
>-  0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
>-  0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
>-  0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
>-  0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
>-  0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
>-  0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
>-  0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
>-  0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
>-  0x12, 0xb4, 0x51, 0x45, 0x14, 0x50,
>-  0xea, 0x8d, 0x1a, 0x35, 0x55, 0xdc,
>-  0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
>-  0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
>-  0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
>-  0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
>-  0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
>-  0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
>-  0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
>-  0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
>-  0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
>-  0x12, 0xb4, 0x51, 0x45, 0x14, 0x50,
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
>-  0x99, 0x02, 0x12, 0x21, 0x22, 0x12,
>-  0x05, 0x80, 0x0e, 0x00, 0xe0, 0x0e,
>-  0x80, 0xa1, 0x82, 0x18, 0x21, 0x82,
>-  0x84, 0x48, 0x18, 0x81, 0x88, 0x18,
>-  0x40, 0x6d, 0x40, 0xd4, 0x0d, 0x40,
>-  0x0a, 0x90, 0xc1, 0x0c, 0x10, 0xc0,
>-  0x68, 0x04, 0x90, 0x49, 0x04, 0x90,
>-  0x10, 0x31, 0x21, 0x12, 0x11, 0x20,
>-  0x30, 0x58, 0x05, 0x80, 0x58, 0x04,
>-  0xd4, 0x8a, 0xd4, 0xd3, 0x3f, 0xe6
>-};
>-
>-const uint8_t kMaskRandom47_45[270] = {
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
>-  0x99, 0x02, 0x12, 0x21, 0x22, 0x12,
>-  0x05, 0x80, 0x0e, 0x00, 0xe0, 0x0e,
>-  0x80, 0xa1, 0x82, 0x18, 0x21, 0x82,
>-  0x84, 0x48, 0x18, 0x81, 0x88, 0x18,
>-  0x40, 0x6d, 0x40, 0xd4, 0x0d, 0x40,
>-  0x0a, 0x90, 0xc1, 0x0c, 0x10, 0xc0,
>-  0x68, 0x04, 0x90, 0x49, 0x04, 0x90,
>-  0x10, 0x31, 0x21, 0x12, 0x11, 0x20,
>-  0x30, 0x58, 0x05, 0x80, 0x58, 0x04,
>-  0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
>-  0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
>-  0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
>-  0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
>-  0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
>-  0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
>-  0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
>-  0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
>-  0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
>-  0x12, 0xb4, 0x51, 0x45, 0x14, 0x50,
>-  0xea, 0x8d, 0x1a, 0x35, 0x55, 0xdc,
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
>-  0x99, 0x02, 0x12, 0x21, 0x22, 0x12,
>-  0x05, 0x80, 0x0e, 0x00, 0xe0, 0x0e,
>-  0x80, 0xa1, 0x82, 0x18, 0x21, 0x82,
>-  0x84, 0x48, 0x18, 0x81, 0x88, 0x18,
>-  0x40, 0x6d, 0x40, 0xd4, 0x0d, 0x40,
>-  0x0a, 0x90, 0xc1, 0x0c, 0x10, 0xc0,
>-  0x68, 0x04, 0x90, 0x49, 0x04, 0x90,
>-  0x10, 0x31, 0x21, 0x12, 0x11, 0x20,
>-  0x30, 0x58, 0x05, 0x80, 0x58, 0x04,
>-  0x46, 0x42, 0x0c, 0x20, 0xc2, 0x0c,
>-  0x33, 0x20, 0x46, 0x04, 0x60, 0x46,
>-  0x99, 0x08, 0x0a, 0x80, 0xa8, 0x0a,
>-  0x05, 0x84, 0x30, 0x43, 0x04, 0x30,
>-  0x80, 0xb0, 0x23, 0x02, 0x30, 0x22,
>-  0x84, 0x42, 0x90, 0x29, 0x02, 0x90,
>-  0x40, 0x73, 0x01, 0x30, 0x13, 0x00,
>-  0x0a, 0x81, 0x12, 0x11, 0x21, 0x12,
>-  0x68, 0x0c, 0x40, 0xc4, 0x0c, 0x40,
>-  0x10, 0x24, 0x84, 0x48, 0x44, 0x84,
>-  0x30, 0x51, 0x41, 0x14, 0x11, 0x40,
>-  0x5f, 0x50, 0x89, 0x08, 0x90, 0x88
>-};
>-
>-const uint8_t kMaskRandom47_46[276] = {
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
>-  0x99, 0x02, 0x12, 0x21, 0x22, 0x12,
>-  0x05, 0x80, 0x0e, 0x00, 0xe0, 0x0e,
>-  0x80, 0xa1, 0x82, 0x18, 0x21, 0x82,
>-  0x84, 0x48, 0x18, 0x81, 0x88, 0x18,
>-  0x40, 0x6d, 0x40, 0xd4, 0x0d, 0x40,
>-  0x0a, 0x90, 0xc1, 0x0c, 0x10, 0xc0,
>-  0x68, 0x04, 0x90, 0x49, 0x04, 0x90,
>-  0x10, 0x31, 0x21, 0x12, 0x11, 0x20,
>-  0x30, 0x58, 0x05, 0x80, 0x58, 0x04,
>-  0x46, 0x42, 0x0c, 0x20, 0xc2, 0x0c,
>-  0x33, 0x20, 0x46, 0x04, 0x60, 0x46,
>-  0x99, 0x08, 0x0a, 0x80, 0xa8, 0x0a,
>-  0x05, 0x84, 0x30, 0x43, 0x04, 0x30,
>-  0x80, 0xb0, 0x23, 0x02, 0x30, 0x22,
>-  0x84, 0x42, 0x90, 0x29, 0x02, 0x90,
>-  0x40, 0x73, 0x01, 0x30, 0x13, 0x00,
>-  0x0a, 0x81, 0x12, 0x11, 0x21, 0x12,
>-  0x68, 0x0c, 0x40, 0xc4, 0x0c, 0x40,
>-  0x10, 0x24, 0x84, 0x48, 0x44, 0x84,
>-  0x30, 0x51, 0x41, 0x14, 0x11, 0x40,
>-  0x5f, 0x50, 0x89, 0x08, 0x90, 0x88,
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
>-  0x99, 0x02, 0x12, 0x21, 0x22, 0x12,
>-  0x05, 0x80, 0x0e, 0x00, 0xe0, 0x0e,
>-  0x80, 0xa1, 0x82, 0x18, 0x21, 0x82,
>-  0x84, 0x48, 0x18, 0x81, 0x88, 0x18,
>-  0x40, 0x6d, 0x40, 0xd4, 0x0d, 0x40,
>-  0x0a, 0x90, 0xc1, 0x0c, 0x10, 0xc0,
>-  0x68, 0x04, 0x90, 0x49, 0x04, 0x90,
>-  0x10, 0x31, 0x21, 0x12, 0x11, 0x20,
>-  0x30, 0x58, 0x05, 0x80, 0x58, 0x04,
>-  0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
>-  0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
>-  0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
>-  0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
>-  0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
>-  0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
>-  0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
>-  0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
>-  0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
>-  0x12, 0xb4, 0x51, 0x45, 0x14, 0x50,
>-  0xea, 0x8d, 0x1a, 0x35, 0x55, 0xdc,
>-  0x37, 0x9d, 0xcf, 0xe0, 0xe4, 0x20
>-};
>-
>-const uint8_t kMaskRandom47_47[282] = {
>-  0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
>-  0x99, 0x02, 0x12, 0x21, 0x22, 0x12,
>-  0x05, 0x80, 0x0e, 0x00, 0xe0, 0x0e,
>-  0x80, 0xa1, 0x82, 0x18, 0x21, 0x82,
>-  0x84, 0x48, 0x18, 0x81, 0x88, 0x18,
>-  0x40, 0x6d, 0x40, 0xd4, 0x0d, 0x40,
>-  0x0a, 0x90, 0xc1, 0x0c, 0x10, 0xc0,
>-  0x68, 0x04, 0x90, 0x49, 0x04, 0x90,
>-  0x10, 0x31, 0x21, 0x12, 0x11, 0x20,
>-  0x30, 0x58, 0x05, 0x80, 0x58, 0x04,
>-  0x46, 0x42, 0x0c, 0x20, 0xc2, 0x0c,
>-  0x33, 0x20, 0x46, 0x04, 0x60, 0x46,
>-  0x99, 0x08, 0x0a, 0x80, 0xa8, 0x0a,
>-  0x05, 0x84, 0x30, 0x43, 0x04, 0x30,
>-  0x80, 0xb0, 0x23, 0x02, 0x30, 0x22,
>-  0x84, 0x42, 0x90, 0x29, 0x02, 0x90,
>-  0x40, 0x73, 0x01, 0x30, 0x13, 0x00,
>-  0x0a, 0x81, 0x12, 0x11, 0x21, 0x12,
>-  0x68, 0x0c, 0x40, 0xc4, 0x0c, 0x40,
>-  0x10, 0x24, 0x84, 0x48, 0x44, 0x84,
>-  0x30, 0x51, 0x41, 0x14, 0x11, 0x40,
>-  0x5f, 0x50, 0x89, 0x08, 0x90, 0x88,
>-  0x46, 0x4a, 0x6c, 0x20, 0xc2, 0x0c,
>-  0x33, 0x24, 0x26, 0x04, 0x60, 0x46,
>-  0x99, 0x02, 0x12, 0x80, 0xa8, 0x0a,
>-  0x05, 0x80, 0x0e, 0x43, 0x04, 0x30,
>-  0x80, 0xa1, 0x83, 0x02, 0x30, 0x22,
>-  0x84, 0x48, 0x18, 0x29, 0x02, 0x90,
>-  0x40, 0x6d, 0x41, 0x30, 0x13, 0x00,
>-  0x0a, 0x90, 0xc0, 0x11, 0x21, 0x12,
>-  0x68, 0x04, 0x90, 0xc4, 0x0c, 0x40,
>-  0x10, 0x31, 0x20, 0x48, 0x44, 0x84,
>-  0x30, 0x58, 0x05, 0x14, 0x11, 0x40,
>-  0x46, 0x42, 0x0d, 0x08, 0x90, 0x88,
>-  0x33, 0x20, 0x46, 0xa6, 0xca, 0x6c,
>-  0x99, 0x08, 0x0a, 0x42, 0x64, 0x26,
>-  0x05, 0x84, 0x30, 0x21, 0x22, 0x12,
>-  0x80, 0xb0, 0x22, 0x00, 0xe0, 0x0e,
>-  0x84, 0x42, 0x90, 0x18, 0x21, 0x82,
>-  0x40, 0x73, 0x00, 0x81, 0x88, 0x18,
>-  0x0a, 0x81, 0x12, 0xd4, 0x0d, 0x40,
>-  0x68, 0x0c, 0x41, 0x0c, 0x10, 0xc0,
>-  0x10, 0x24, 0x84, 0x49, 0x04, 0x90,
>-  0x30, 0x51, 0x41, 0x12, 0x11, 0x20,
>-  0x5f, 0x50, 0x89, 0x80, 0x58, 0x04,
>-  0x1f, 0x2f, 0x63, 0x10, 0x64, 0xb2
>-};
>-
>-const uint8_t kMaskRandom47_5[30] = {
>-  0xc6, 0xca, 0x6c, 0xa6, 0xca, 0x6c,
>-  0x63, 0x6c, 0x96, 0xc9, 0x6c, 0x96,
>-  0x1d, 0xa1, 0xdc, 0x1d, 0xc1, 0xdc,
>-  0xad, 0x55, 0x39, 0x53, 0x95, 0x38,
>-  0xb2, 0xb7, 0x07, 0x70, 0x77, 0x06
>-};
>-
>-const uint8_t kMaskRandom47_6[36] = {
>-  0x64, 0x4a, 0x29, 0xa2, 0x9a, 0x28,
>-  0x51, 0x58, 0xa2, 0x8a, 0x68, 0xa6,
>-  0x0c, 0xa4, 0x30, 0x45, 0xa4, 0x5a,
>-  0xa1, 0x22, 0x46, 0x2d, 0x82, 0xd8,
>-  0x12, 0xa1, 0x1c, 0x17, 0x41, 0x74,
>-  0x8a, 0x45, 0xc1, 0xd1, 0x1d, 0x10
>-};
>-
>-const uint8_t kMaskRandom47_7[42] = {
>-  0x46, 0x4a, 0x6d, 0xa6, 0xca, 0x6c,
>-  0x33, 0x24, 0x26, 0x4a, 0x64, 0xa6,
>-  0x91, 0x92, 0x12, 0x61, 0xa6, 0x0a,
>-  0xa4, 0x20, 0x4a, 0x0c, 0x90, 0xd8,
>-  0x50, 0xa0, 0xd5, 0x81, 0x70, 0x36,
>-  0x84, 0xc5, 0x80, 0x55, 0x45, 0x54,
>-  0x09, 0x71, 0x0d, 0x50, 0x9d, 0x08
>-};
>-
>-const uint8_t kMaskRandom47_8[48] = {
>-  0x0c, 0x84, 0x0d, 0x02, 0xc0, 0x2c,
>-  0x80, 0x70, 0x06, 0x80, 0x78, 0x06,
>-  0xa0, 0x88, 0x48, 0x21, 0x22, 0x12,
>-  0x05, 0x40, 0x32, 0x0c, 0xa0, 0xca,
>-  0x43, 0x02, 0x82, 0x40, 0x95, 0x08,
>-  0x1a, 0x01, 0x51, 0x15, 0x41, 0x54,
>-  0x60, 0x27, 0x00, 0x66, 0x06, 0x60,
>-  0x14, 0x38, 0xa0, 0x99, 0x09, 0x90
>-};
>-
>-const uint8_t kMaskRandom47_9[54] = {
>-  0x46, 0x4a, 0x6d, 0xa6, 0xca, 0x6c,
>-  0x62, 0x7c, 0x84, 0xc8, 0x4c, 0x84,
>-  0x8c, 0x04, 0x88, 0x30, 0x83, 0x88,
>-  0x01, 0x74, 0x23, 0x40, 0x94, 0x08,
>-  0x07, 0x83, 0x07, 0x02, 0x70, 0x26,
>-  0xa0, 0x80, 0x72, 0x45, 0x44, 0x54,
>-  0x18, 0xb1, 0x42, 0x10, 0xe1, 0x0e,
>-  0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
>-  0x78, 0x00, 0x1c, 0x03, 0x80, 0x38
>-};
>-
>-const uint8_t kMaskRandom48_1[6] = {
>-  0xff, 0xff, 0xff, 0xff, 0xff, 0xff
>-};
>-
>-const uint8_t kMaskRandom48_10[60] = {
>-  0x11, 0x45, 0x14, 0x11, 0x45, 0x14,
>-  0x45, 0x34, 0x53, 0x45, 0x34, 0x53,
>-  0x00, 0x48, 0x05, 0x00, 0x48, 0x05,
>-  0x10, 0x83, 0x09, 0x10, 0x83, 0x09,
>-  0x4a, 0x14, 0xa1, 0x4a, 0x14, 0xa1,
>-  0x40, 0xa4, 0x0a, 0x40, 0xa4, 0x0a,
>-  0xa0, 0x6a, 0x02, 0xa0, 0x6a, 0x02,
>-  0x88, 0x80, 0x8c, 0x88, 0x80, 0x8c,
>-  0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
>-  0x54, 0x0d, 0x40, 0x54, 0x0d, 0x40
>-};
>-
>-const uint8_t kMaskRandom48_11[66] = {
>-  0x53, 0x65, 0x34, 0x53, 0x65, 0x34,
>-  0xa0, 0x32, 0x11, 0xa0, 0x32, 0x11,
>-  0x15, 0x11, 0x41, 0x15, 0x11, 0x41,
>-  0x03, 0x50, 0x15, 0x03, 0x50, 0x15,
>-  0x8c, 0x88, 0xc8, 0x8c, 0x88, 0xc8,
>-  0x28, 0x82, 0x88, 0x28, 0x82, 0x88,
>-  0x08, 0x48, 0x84, 0x08, 0x48, 0x84,
>-  0x99, 0x01, 0x90, 0x99, 0x01, 0x90,
>-  0x22, 0x92, 0x29, 0x22, 0x92, 0x29,
>-  0x46, 0x04, 0x60, 0x46, 0x04, 0x60,
>-  0x8c, 0x2c, 0x02, 0x8c, 0x2c, 0x02
>-};
>-
>-const uint8_t kMaskRandom48_12[72] = {
>-  0x10, 0x61, 0x06, 0x10, 0x61, 0x06,
>-  0x02, 0x30, 0x23, 0x02, 0x30, 0x23,
>-  0x40, 0x54, 0x05, 0x40, 0x54, 0x05,
>-  0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
>-  0x81, 0x18, 0x11, 0x81, 0x18, 0x11,
>-  0x14, 0x81, 0x48, 0x14, 0x81, 0x48,
>-  0x98, 0x09, 0x80, 0x98, 0x09, 0x80,
>-  0x08, 0x90, 0x89, 0x08, 0x90, 0x89,
>-  0x62, 0x06, 0x20, 0x62, 0x06, 0x20,
>-  0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
>-  0x8a, 0x08, 0xa0, 0x8a, 0x08, 0xa0,
>-  0x84, 0x48, 0x44, 0x84, 0x48, 0x44
>-};
>-
>-const uint8_t kMaskRandom48_13[78] = {
>-  0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
>-  0xc5, 0x1c, 0x51, 0xc5, 0x1c, 0x51,
>-  0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
>-  0x12, 0x31, 0x23, 0x12, 0x31, 0x23,
>-  0x08, 0xe0, 0x8e, 0x08, 0xe0, 0x8e,
>-  0x2e, 0x02, 0xe0, 0x2e, 0x02, 0xe0,
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
>-  0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
>-  0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
>-  0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
>-  0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
>-  0x88, 0x68, 0x86, 0x88, 0x68, 0x86
>-};
>-
>-const uint8_t kMaskRandom48_14[84] = {
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
>-  0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
>-  0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
>-  0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
>-  0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
>-  0x88, 0x68, 0x86, 0x88, 0x68, 0x86,
>-  0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
>-  0xc5, 0x1c, 0x51, 0xc5, 0x1c, 0x51,
>-  0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
>-  0x12, 0x31, 0x23, 0x12, 0x31, 0x23,
>-  0x08, 0xe0, 0x8e, 0x08, 0xe0, 0x8e,
>-  0x2e, 0x02, 0xe0, 0x2e, 0x02, 0xe0,
>-  0xf2, 0xd6, 0x8e, 0xf2, 0xd6, 0x8e
>-};
>-
>-const uint8_t kMaskRandom48_15[90] = {
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
>-  0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
>-  0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
>-  0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
>-  0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
>-  0x88, 0x68, 0x86, 0x88, 0x68, 0x86,
>-  0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
>-  0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
>-  0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
>-  0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
>-  0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
>-  0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
>-  0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
>-  0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50
>-};
>-
>-const uint8_t kMaskRandom48_16[96] = {
>-  0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
>-  0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
>-  0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
>-  0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
>-  0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
>-  0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
>-  0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
>-  0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50,
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
>-  0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
>-  0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
>-  0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
>-  0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
>-  0x88, 0x68, 0x86, 0x88, 0x68, 0x86,
>-  0xff, 0x6e, 0x0a, 0xff, 0x6e, 0x0a
>-};
>-
>-const uint8_t kMaskRandom48_17[102] = {
>-  0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
>-  0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
>-  0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
>-  0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
>-  0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
>-  0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
>-  0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
>-  0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50,
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
>-  0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
>-  0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
>-  0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
>-  0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
>-  0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
>-  0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
>-  0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e
>-};
>-
>-const uint8_t kMaskRandom48_18[108] = {
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
>-  0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
>-  0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
>-  0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
>-  0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
>-  0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
>-  0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
>-  0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
>-  0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
>-  0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
>-  0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
>-  0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
>-  0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
>-  0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
>-  0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
>-  0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50,
>-  0x34, 0x50, 0xae, 0x34, 0x50, 0xae
>-};
>-
>-const uint8_t kMaskRandom48_19[114] = {
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
>-  0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
>-  0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
>-  0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
>-  0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
>-  0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
>-  0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
>-  0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
>-  0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
>-  0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
>-  0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
>-  0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
>-  0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
>-  0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
>-  0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
>-  0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
>-  0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
>-  0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28
>-};
>-
>-const uint8_t kMaskRandom48_2[12] = {
>-  0xec, 0xce, 0xcc, 0xec, 0xce, 0xcc,
>-  0x93, 0xb9, 0x3b, 0x93, 0xb9, 0x3b
>-};
>-
>-const uint8_t kMaskRandom48_20[120] = {
>-  0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
>-  0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
>-  0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
>-  0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
>-  0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
>-  0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
>-  0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
>-  0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
>-  0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
>-  0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28,
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
>-  0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
>-  0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
>-  0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
>-  0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
>-  0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
>-  0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
>-  0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
>-  0x98, 0xa2, 0x95, 0x98, 0xa2, 0x95
>-};
>-
>-const uint8_t kMaskRandom48_21[126] = {
>-  0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
>-  0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
>-  0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
>-  0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
>-  0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
>-  0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
>-  0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
>-  0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
>-  0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
>-  0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28,
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
>-  0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
>-  0x00, 0x70, 0x07, 0x00, 0x70, 0x07,
>-  0x0c, 0x10, 0xc1, 0x0c, 0x10, 0xc1,
>-  0x40, 0xc4, 0x0c, 0x40, 0xc4, 0x0c,
>-  0x6a, 0x06, 0xa0, 0x6a, 0x06, 0xa0,
>-  0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
>-  0x24, 0x82, 0x48, 0x24, 0x82, 0x48,
>-  0x89, 0x08, 0x90, 0x89, 0x08, 0x90,
>-  0xc0, 0x2c, 0x02, 0xc0, 0x2c, 0x02
>-};
>-
>-const uint8_t kMaskRandom48_22[132] = {
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
>-  0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
>-  0x00, 0x70, 0x07, 0x00, 0x70, 0x07,
>-  0x0c, 0x10, 0xc1, 0x0c, 0x10, 0xc1,
>-  0x40, 0xc4, 0x0c, 0x40, 0xc4, 0x0c,
>-  0x6a, 0x06, 0xa0, 0x6a, 0x06, 0xa0,
>-  0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
>-  0x24, 0x82, 0x48, 0x24, 0x82, 0x48,
>-  0x89, 0x08, 0x90, 0x89, 0x08, 0x90,
>-  0xc0, 0x2c, 0x02, 0xc0, 0x2c, 0x02,
>-  0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
>-  0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
>-  0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
>-  0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
>-  0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
>-  0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
>-  0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
>-  0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
>-  0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
>-  0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28,
>-  0x1a, 0xaa, 0xee, 0x1a, 0xaa, 0xee
>-};
>-
>-const uint8_t kMaskRandom48_23[138] = {
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
>-  0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
>-  0x00, 0x70, 0x07, 0x00, 0x70, 0x07,
>-  0x0c, 0x10, 0xc1, 0x0c, 0x10, 0xc1,
>-  0x40, 0xc4, 0x0c, 0x40, 0xc4, 0x0c,
>-  0x6a, 0x06, 0xa0, 0x6a, 0x06, 0xa0,
>-  0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
>-  0x24, 0x82, 0x48, 0x24, 0x82, 0x48,
>-  0x89, 0x08, 0x90, 0x89, 0x08, 0x90,
>-  0xc0, 0x2c, 0x02, 0xc0, 0x2c, 0x02,
>-  0x10, 0x61, 0x06, 0x10, 0x61, 0x06,
>-  0x02, 0x30, 0x23, 0x02, 0x30, 0x23,
>-  0x40, 0x54, 0x05, 0x40, 0x54, 0x05,
>-  0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
>-  0x81, 0x18, 0x11, 0x81, 0x18, 0x11,
>-  0x14, 0x81, 0x48, 0x14, 0x81, 0x48,
>-  0x98, 0x09, 0x80, 0x98, 0x09, 0x80,
>-  0x08, 0x90, 0x89, 0x08, 0x90, 0x89,
>-  0x62, 0x06, 0x20, 0x62, 0x06, 0x20,
>-  0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
>-  0x8a, 0x08, 0xa0, 0x8a, 0x08, 0xa0,
>-  0x84, 0x48, 0x44, 0x84, 0x48, 0x44
>-};
>-
>-const uint8_t kMaskRandom48_24[144] = {
>-  0x10, 0x61, 0x06, 0x10, 0x61, 0x06,
>-  0x02, 0x30, 0x23, 0x02, 0x30, 0x23,
>-  0x40, 0x54, 0x05, 0x40, 0x54, 0x05,
>-  0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
>-  0x81, 0x18, 0x11, 0x81, 0x18, 0x11,
>-  0x14, 0x81, 0x48, 0x14, 0x81, 0x48,
>-  0x98, 0x09, 0x80, 0x98, 0x09, 0x80,
>-  0x08, 0x90, 0x89, 0x08, 0x90, 0x89,
>-  0x62, 0x06, 0x20, 0x62, 0x06, 0x20,
>-  0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
>-  0x8a, 0x08, 0xa0, 0x8a, 0x08, 0xa0,
>-  0x84, 0x48, 0x44, 0x84, 0x48, 0x44,
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
>-  0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
>-  0x00, 0x70, 0x07, 0x00, 0x70, 0x07,
>-  0x0c, 0x10, 0xc1, 0x0c, 0x10, 0xc1,
>-  0x40, 0xc4, 0x0c, 0x40, 0xc4, 0x0c,
>-  0x6a, 0x06, 0xa0, 0x6a, 0x06, 0xa0,
>-  0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
>-  0x24, 0x82, 0x48, 0x24, 0x82, 0x48,
>-  0x89, 0x08, 0x90, 0x89, 0x08, 0x90,
>-  0xc0, 0x2c, 0x02, 0xc0, 0x2c, 0x02,
>-  0x88, 0x32, 0x59, 0x88, 0x32, 0x59
>-};
>-
>-const uint8_t kMaskRandom48_25[150] = {
>-  0x10, 0x61, 0x06, 0x10, 0x61, 0x06,
>-  0x02, 0x30, 0x23, 0x02, 0x30, 0x23,
>-  0x40, 0x54, 0x05, 0x40, 0x54, 0x05,
>-  0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
>-  0x81, 0x18, 0x11, 0x81, 0x18, 0x11,
>-  0x14, 0x81, 0x48, 0x14, 0x81, 0x48,
>-  0x98, 0x09, 0x80, 0x98, 0x09, 0x80,
>-  0x08, 0x90, 0x89, 0x08, 0x90, 0x89,
>-  0x62, 0x06, 0x20, 0x62, 0x06, 0x20,
>-  0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
>-  0x8a, 0x08, 0xa0, 0x8a, 0x08, 0xa0,
>-  0x84, 0x48, 0x44, 0x84, 0x48, 0x44,
>-  0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
>-  0xc5, 0x1c, 0x51, 0xc5, 0x1c, 0x51,
>-  0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
>-  0x12, 0x31, 0x23, 0x12, 0x31, 0x23,
>-  0x08, 0xe0, 0x8e, 0x08, 0xe0, 0x8e,
>-  0x2e, 0x02, 0xe0, 0x2e, 0x02, 0xe0,
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
>-  0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
>-  0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
>-  0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
>-  0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
>-  0x88, 0x68, 0x86, 0x88, 0x68, 0x86
>-};
>-
>-const uint8_t kMaskRandom48_26[156] = {
>-  0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
>-  0xc5, 0x1c, 0x51, 0xc5, 0x1c, 0x51,
>-  0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
>-  0x12, 0x31, 0x23, 0x12, 0x31, 0x23,
>-  0x08, 0xe0, 0x8e, 0x08, 0xe0, 0x8e,
>-  0x2e, 0x02, 0xe0, 0x2e, 0x02, 0xe0,
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
>-  0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
>-  0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
>-  0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
>-  0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
>-  0x88, 0x68, 0x86, 0x88, 0x68, 0x86,
>-  0x10, 0x61, 0x06, 0x10, 0x61, 0x06,
>-  0x02, 0x30, 0x23, 0x02, 0x30, 0x23,
>-  0x40, 0x54, 0x05, 0x40, 0x54, 0x05,
>-  0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
>-  0x81, 0x18, 0x11, 0x81, 0x18, 0x11,
>-  0x14, 0x81, 0x48, 0x14, 0x81, 0x48,
>-  0x98, 0x09, 0x80, 0x98, 0x09, 0x80,
>-  0x08, 0x90, 0x89, 0x08, 0x90, 0x89,
>-  0x62, 0x06, 0x20, 0x62, 0x06, 0x20,
>-  0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
>-  0x8a, 0x08, 0xa0, 0x8a, 0x08, 0xa0,
>-  0x84, 0x48, 0x44, 0x84, 0x48, 0x44,
>-  0x3e, 0x20, 0x79, 0xe5, 0x55, 0x70
>-};
>-
>-const uint8_t kMaskRandom48_27[162] = {
>-  0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
>-  0xc5, 0x1c, 0x51, 0xc5, 0x1c, 0x51,
>-  0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
>-  0x12, 0x31, 0x23, 0x12, 0x31, 0x23,
>-  0x08, 0xe0, 0x8e, 0x08, 0xe0, 0x8e,
>-  0x2e, 0x02, 0xe0, 0x2e, 0x02, 0xe0,
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
>-  0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
>-  0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
>-  0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
>-  0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
>-  0x88, 0x68, 0x86, 0x88, 0x68, 0x86,
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
>-  0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
>-  0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
>-  0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
>-  0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
>-  0x88, 0x68, 0x86, 0x88, 0x68, 0x86,
>-  0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
>-  0xc5, 0x1c, 0x51, 0xc5, 0x1c, 0x51,
>-  0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
>-  0x12, 0x31, 0x23, 0x12, 0x31, 0x23,
>-  0x08, 0xe0, 0x8e, 0x08, 0xe0, 0x8e,
>-  0x2e, 0x02, 0xe0, 0x2e, 0x02, 0xe0,
>-  0xf2, 0xd6, 0x8e, 0xf2, 0xd6, 0x8e
>-};
>-
>-const uint8_t kMaskRandom48_28[168] = {
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
>-  0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
>-  0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
>-  0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
>-  0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
>-  0x88, 0x68, 0x86, 0x88, 0x68, 0x86,
>-  0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
>-  0xc5, 0x1c, 0x51, 0xc5, 0x1c, 0x51,
>-  0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
>-  0x12, 0x31, 0x23, 0x12, 0x31, 0x23,
>-  0x08, 0xe0, 0x8e, 0x08, 0xe0, 0x8e,
>-  0x2e, 0x02, 0xe0, 0x2e, 0x02, 0xe0,
>-  0xf2, 0xd6, 0x8e, 0xf2, 0xd6, 0x8e,
>-  0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
>-  0xc5, 0x1c, 0x51, 0xc5, 0x1c, 0x51,
>-  0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
>-  0x12, 0x31, 0x23, 0x12, 0x31, 0x23,
>-  0x08, 0xe0, 0x8e, 0x08, 0xe0, 0x8e,
>-  0x2e, 0x02, 0xe0, 0x2e, 0x02, 0xe0,
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
>-  0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
>-  0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
>-  0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
>-  0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
>-  0x88, 0x68, 0x86, 0x88, 0x68, 0x86,
>-  0x32, 0xe3, 0xc0, 0x4a, 0xf2, 0x2a
>-};
>-
>-const uint8_t kMaskRandom48_29[174] = {
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
>-  0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
>-  0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
>-  0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
>-  0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
>-  0x88, 0x68, 0x86, 0x88, 0x68, 0x86,
>-  0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
>-  0xc5, 0x1c, 0x51, 0xc5, 0x1c, 0x51,
>-  0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
>-  0x12, 0x31, 0x23, 0x12, 0x31, 0x23,
>-  0x08, 0xe0, 0x8e, 0x08, 0xe0, 0x8e,
>-  0x2e, 0x02, 0xe0, 0x2e, 0x02, 0xe0,
>-  0xf2, 0xd6, 0x8e, 0xf2, 0xd6, 0x8e,
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
>-  0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
>-  0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
>-  0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
>-  0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
>-  0x88, 0x68, 0x86, 0x88, 0x68, 0x86,
>-  0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
>-  0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
>-  0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
>-  0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
>-  0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
>-  0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
>-  0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
>-  0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50
>-};
>-
>-const uint8_t kMaskRandom48_3[18] = {
>-  0x9b, 0x29, 0xb2, 0x9b, 0x29, 0xb2,
>-  0x49, 0xd4, 0x9d, 0x49, 0xd4, 0x9d,
>-  0x3e, 0x83, 0xe8, 0x3e, 0x83, 0xe8
>-};
>-
>-const uint8_t kMaskRandom48_30[180] = {
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
>-  0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
>-  0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
>-  0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
>-  0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
>-  0x88, 0x68, 0x86, 0x88, 0x68, 0x86,
>-  0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
>-  0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
>-  0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
>-  0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
>-  0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
>-  0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
>-  0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
>-  0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50,
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
>-  0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
>-  0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
>-  0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
>-  0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
>-  0x88, 0x68, 0x86, 0x88, 0x68, 0x86,
>-  0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
>-  0xc5, 0x1c, 0x51, 0xc5, 0x1c, 0x51,
>-  0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
>-  0x12, 0x31, 0x23, 0x12, 0x31, 0x23,
>-  0x08, 0xe0, 0x8e, 0x08, 0xe0, 0x8e,
>-  0x2e, 0x02, 0xe0, 0x2e, 0x02, 0xe0,
>-  0xf2, 0xd6, 0x8e, 0xf2, 0xd6, 0x8e,
>-  0x66, 0xf3, 0x9a, 0xdd, 0x68, 0x93
>-};
>-
>-const uint8_t kMaskRandom48_31[186] = {
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
>-  0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
>-  0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
>-  0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
>-  0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
>-  0x88, 0x68, 0x86, 0x88, 0x68, 0x86,
>-  0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
>-  0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
>-  0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
>-  0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
>-  0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
>-  0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
>-  0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
>-  0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50,
>-  0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
>-  0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
>-  0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
>-  0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
>-  0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
>-  0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
>-  0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
>-  0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50,
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
>-  0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
>-  0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
>-  0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
>-  0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
>-  0x88, 0x68, 0x86, 0x88, 0x68, 0x86,
>-  0xff, 0x6e, 0x0a, 0xff, 0x6e, 0x0a
>-};
>-
>-const uint8_t kMaskRandom48_32[192] = {
>-  0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
>-  0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
>-  0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
>-  0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
>-  0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
>-  0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
>-  0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
>-  0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50,
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
>-  0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
>-  0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
>-  0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
>-  0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
>-  0x88, 0x68, 0x86, 0x88, 0x68, 0x86,
>-  0xff, 0x6e, 0x0a, 0xff, 0x6e, 0x0a,
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
>-  0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
>-  0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
>-  0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
>-  0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
>-  0x88, 0x68, 0x86, 0x88, 0x68, 0x86,
>-  0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
>-  0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
>-  0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
>-  0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
>-  0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
>-  0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
>-  0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
>-  0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50,
>-  0xd5, 0x4a, 0x4f, 0x48, 0xb5, 0x31
>-};
>-
>-const uint8_t kMaskRandom48_33[198] = {
>-  0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
>-  0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
>-  0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
>-  0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
>-  0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
>-  0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
>-  0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
>-  0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50,
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
>-  0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
>-  0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
>-  0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
>-  0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
>-  0x88, 0x68, 0x86, 0x88, 0x68, 0x86,
>-  0xff, 0x6e, 0x0a, 0xff, 0x6e, 0x0a,
>-  0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
>-  0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
>-  0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
>-  0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
>-  0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
>-  0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
>-  0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
>-  0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50,
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
>-  0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
>-  0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
>-  0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
>-  0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
>-  0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
>-  0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
>-  0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e
>-};
>-
>-const uint8_t kMaskRandom48_34[204] = {
>-  0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
>-  0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
>-  0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
>-  0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
>-  0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
>-  0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
>-  0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
>-  0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50,
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
>-  0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
>-  0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
>-  0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
>-  0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
>-  0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
>-  0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
>-  0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
>-  0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
>-  0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
>-  0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
>-  0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
>-  0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
>-  0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
>-  0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
>-  0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50,
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
>-  0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
>-  0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
>-  0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
>-  0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
>-  0x88, 0x68, 0x86, 0x88, 0x68, 0x86,
>-  0xff, 0x6e, 0x0a, 0xff, 0x6e, 0x0a,
>-  0x40, 0x72, 0x4c, 0xe8, 0xf2, 0x42
>-};
>-
>-const uint8_t kMaskRandom48_35[210] = {
>-  0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
>-  0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
>-  0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
>-  0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
>-  0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
>-  0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
>-  0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
>-  0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50,
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
>-  0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
>-  0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
>-  0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
>-  0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
>-  0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
>-  0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
>-  0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
>-  0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
>-  0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
>-  0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
>-  0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
>-  0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
>-  0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
>-  0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
>-  0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
>-  0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
>-  0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
>-  0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
>-  0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
>-  0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
>-  0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
>-  0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50,
>-  0x34, 0x50, 0xae, 0x34, 0x50, 0xae
>-};
>-
>-const uint8_t kMaskRandom48_36[216] = {
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
>-  0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
>-  0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
>-  0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
>-  0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
>-  0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
>-  0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
>-  0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
>-  0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
>-  0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
>-  0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
>-  0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
>-  0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
>-  0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
>-  0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
>-  0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50,
>-  0x34, 0x50, 0xae, 0x34, 0x50, 0xae,
>-  0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
>-  0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
>-  0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
>-  0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
>-  0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
>-  0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
>-  0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
>-  0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50,
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
>-  0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
>-  0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
>-  0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
>-  0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
>-  0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
>-  0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
>-  0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
>-  0x71, 0xba, 0x8b, 0xf3, 0xfa, 0x9d
>-};
>-
>-const uint8_t kMaskRandom48_37[222] = {
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
>-  0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
>-  0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
>-  0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
>-  0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
>-  0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
>-  0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
>-  0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
>-  0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
>-  0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
>-  0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
>-  0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
>-  0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
>-  0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
>-  0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
>-  0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50,
>-  0x34, 0x50, 0xae, 0x34, 0x50, 0xae,
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
>-  0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
>-  0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
>-  0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
>-  0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
>-  0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
>-  0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
>-  0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
>-  0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
>-  0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
>-  0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
>-  0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
>-  0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
>-  0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
>-  0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
>-  0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
>-  0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
>-  0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28
>-};
>-
>-const uint8_t kMaskRandom48_38[228] = {
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
>-  0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
>-  0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
>-  0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
>-  0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
>-  0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
>-  0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
>-  0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
>-  0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
>-  0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
>-  0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
>-  0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
>-  0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
>-  0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
>-  0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
>-  0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
>-  0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
>-  0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28,
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
>-  0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
>-  0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
>-  0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
>-  0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
>-  0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
>-  0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
>-  0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
>-  0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
>-  0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
>-  0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
>-  0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
>-  0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
>-  0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
>-  0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
>-  0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50,
>-  0x34, 0x50, 0xae, 0x34, 0x50, 0xae,
>-  0x2a, 0x7a, 0xf6, 0x8c, 0xde, 0x51
>-};
>-
>-const uint8_t kMaskRandom48_39[234] = {
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
>-  0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
>-  0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
>-  0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
>-  0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
>-  0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
>-  0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
>-  0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
>-  0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
>-  0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
>-  0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
>-  0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
>-  0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
>-  0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
>-  0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
>-  0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
>-  0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
>-  0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28,
>-  0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
>-  0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
>-  0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
>-  0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
>-  0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
>-  0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
>-  0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
>-  0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
>-  0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
>-  0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28,
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
>-  0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
>-  0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
>-  0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
>-  0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
>-  0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
>-  0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
>-  0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
>-  0x98, 0xa2, 0x95, 0x98, 0xa2, 0x95
>-};
>-
>-const uint8_t kMaskRandom48_4[24] = {
>-  0x8b, 0x28, 0xb2, 0x8b, 0x28, 0xb2,
>-  0x14, 0xb1, 0x4b, 0x14, 0xb1, 0x4b,
>-  0x22, 0xd2, 0x2d, 0x22, 0xd2, 0x2d,
>-  0x45, 0x54, 0x55, 0x45, 0x54, 0x55
>-};
>-
>-const uint8_t kMaskRandom48_40[240] = {
>-  0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
>-  0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
>-  0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
>-  0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
>-  0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
>-  0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
>-  0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
>-  0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
>-  0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
>-  0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28,
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
>-  0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
>-  0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
>-  0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
>-  0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
>-  0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
>-  0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
>-  0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
>-  0x98, 0xa2, 0x95, 0x98, 0xa2, 0x95,
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
>-  0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
>-  0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
>-  0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
>-  0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
>-  0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
>-  0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
>-  0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
>-  0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
>-  0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
>-  0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
>-  0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
>-  0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
>-  0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
>-  0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
>-  0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
>-  0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
>-  0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28,
>-  0x20, 0x5f, 0x68, 0xd5, 0xa2, 0x1b
>-};
>-
>-const uint8_t kMaskRandom48_41[246] = {
>-  0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
>-  0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
>-  0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
>-  0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
>-  0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
>-  0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
>-  0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
>-  0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
>-  0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
>-  0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28,
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
>-  0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
>-  0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
>-  0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
>-  0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
>-  0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
>-  0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
>-  0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
>-  0x98, 0xa2, 0x95, 0x98, 0xa2, 0x95,
>-  0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
>-  0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
>-  0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
>-  0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
>-  0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
>-  0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
>-  0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
>-  0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
>-  0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
>-  0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28,
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
>-  0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
>-  0x00, 0x70, 0x07, 0x00, 0x70, 0x07,
>-  0x0c, 0x10, 0xc1, 0x0c, 0x10, 0xc1,
>-  0x40, 0xc4, 0x0c, 0x40, 0xc4, 0x0c,
>-  0x6a, 0x06, 0xa0, 0x6a, 0x06, 0xa0,
>-  0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
>-  0x24, 0x82, 0x48, 0x24, 0x82, 0x48,
>-  0x89, 0x08, 0x90, 0x89, 0x08, 0x90,
>-  0xc0, 0x2c, 0x02, 0xc0, 0x2c, 0x02
>-};
>-
>-const uint8_t kMaskRandom48_42[252] = {
>-  0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
>-  0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
>-  0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
>-  0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
>-  0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
>-  0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
>-  0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
>-  0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
>-  0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
>-  0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28,
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
>-  0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
>-  0x00, 0x70, 0x07, 0x00, 0x70, 0x07,
>-  0x0c, 0x10, 0xc1, 0x0c, 0x10, 0xc1,
>-  0x40, 0xc4, 0x0c, 0x40, 0xc4, 0x0c,
>-  0x6a, 0x06, 0xa0, 0x6a, 0x06, 0xa0,
>-  0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
>-  0x24, 0x82, 0x48, 0x24, 0x82, 0x48,
>-  0x89, 0x08, 0x90, 0x89, 0x08, 0x90,
>-  0xc0, 0x2c, 0x02, 0xc0, 0x2c, 0x02,
>-  0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
>-  0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
>-  0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
>-  0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
>-  0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
>-  0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
>-  0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
>-  0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
>-  0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
>-  0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28,
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
>-  0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
>-  0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
>-  0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
>-  0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
>-  0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
>-  0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
>-  0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
>-  0x98, 0xa2, 0x95, 0x98, 0xa2, 0x95,
>-  0x66, 0xcf, 0xa3, 0x47, 0x69, 0x00
>-};
>-
>-const uint8_t kMaskRandom48_43[258] = {
>-  0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
>-  0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
>-  0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
>-  0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
>-  0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
>-  0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
>-  0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
>-  0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
>-  0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
>-  0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28,
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
>-  0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
>-  0x00, 0x70, 0x07, 0x00, 0x70, 0x07,
>-  0x0c, 0x10, 0xc1, 0x0c, 0x10, 0xc1,
>-  0x40, 0xc4, 0x0c, 0x40, 0xc4, 0x0c,
>-  0x6a, 0x06, 0xa0, 0x6a, 0x06, 0xa0,
>-  0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
>-  0x24, 0x82, 0x48, 0x24, 0x82, 0x48,
>-  0x89, 0x08, 0x90, 0x89, 0x08, 0x90,
>-  0xc0, 0x2c, 0x02, 0xc0, 0x2c, 0x02,
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
>-  0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
>-  0x00, 0x70, 0x07, 0x00, 0x70, 0x07,
>-  0x0c, 0x10, 0xc1, 0x0c, 0x10, 0xc1,
>-  0x40, 0xc4, 0x0c, 0x40, 0xc4, 0x0c,
>-  0x6a, 0x06, 0xa0, 0x6a, 0x06, 0xa0,
>-  0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
>-  0x24, 0x82, 0x48, 0x24, 0x82, 0x48,
>-  0x89, 0x08, 0x90, 0x89, 0x08, 0x90,
>-  0xc0, 0x2c, 0x02, 0xc0, 0x2c, 0x02,
>-  0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
>-  0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
>-  0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
>-  0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
>-  0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
>-  0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
>-  0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
>-  0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
>-  0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
>-  0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28,
>-  0x1a, 0xaa, 0xee, 0x1a, 0xaa, 0xee
>-};
>-
>-const uint8_t kMaskRandom48_44[264] = {
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
>-  0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
>-  0x00, 0x70, 0x07, 0x00, 0x70, 0x07,
>-  0x0c, 0x10, 0xc1, 0x0c, 0x10, 0xc1,
>-  0x40, 0xc4, 0x0c, 0x40, 0xc4, 0x0c,
>-  0x6a, 0x06, 0xa0, 0x6a, 0x06, 0xa0,
>-  0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
>-  0x24, 0x82, 0x48, 0x24, 0x82, 0x48,
>-  0x89, 0x08, 0x90, 0x89, 0x08, 0x90,
>-  0xc0, 0x2c, 0x02, 0xc0, 0x2c, 0x02,
>-  0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
>-  0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
>-  0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
>-  0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
>-  0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
>-  0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
>-  0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
>-  0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
>-  0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
>-  0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28,
>-  0x1a, 0xaa, 0xee, 0x1a, 0xaa, 0xee,
>-  0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
>-  0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
>-  0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
>-  0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
>-  0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
>-  0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
>-  0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
>-  0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
>-  0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
>-  0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28,
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
>-  0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
>-  0x00, 0x70, 0x07, 0x00, 0x70, 0x07,
>-  0x0c, 0x10, 0xc1, 0x0c, 0x10, 0xc1,
>-  0x40, 0xc4, 0x0c, 0x40, 0xc4, 0x0c,
>-  0x6a, 0x06, 0xa0, 0x6a, 0x06, 0xa0,
>-  0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
>-  0x24, 0x82, 0x48, 0x24, 0x82, 0x48,
>-  0x89, 0x08, 0x90, 0x89, 0x08, 0x90,
>-  0xc0, 0x2c, 0x02, 0xc0, 0x2c, 0x02,
>-  0xc6, 0x40, 0x1f, 0x57, 0xc6, 0xe6
>-};
>-
>-const uint8_t kMaskRandom48_45[270] = {
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
>-  0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
>-  0x00, 0x70, 0x07, 0x00, 0x70, 0x07,
>-  0x0c, 0x10, 0xc1, 0x0c, 0x10, 0xc1,
>-  0x40, 0xc4, 0x0c, 0x40, 0xc4, 0x0c,
>-  0x6a, 0x06, 0xa0, 0x6a, 0x06, 0xa0,
>-  0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
>-  0x24, 0x82, 0x48, 0x24, 0x82, 0x48,
>-  0x89, 0x08, 0x90, 0x89, 0x08, 0x90,
>-  0xc0, 0x2c, 0x02, 0xc0, 0x2c, 0x02,
>-  0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
>-  0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
>-  0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
>-  0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
>-  0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
>-  0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
>-  0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
>-  0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
>-  0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
>-  0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28,
>-  0x1a, 0xaa, 0xee, 0x1a, 0xaa, 0xee,
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
>-  0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
>-  0x00, 0x70, 0x07, 0x00, 0x70, 0x07,
>-  0x0c, 0x10, 0xc1, 0x0c, 0x10, 0xc1,
>-  0x40, 0xc4, 0x0c, 0x40, 0xc4, 0x0c,
>-  0x6a, 0x06, 0xa0, 0x6a, 0x06, 0xa0,
>-  0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
>-  0x24, 0x82, 0x48, 0x24, 0x82, 0x48,
>-  0x89, 0x08, 0x90, 0x89, 0x08, 0x90,
>-  0xc0, 0x2c, 0x02, 0xc0, 0x2c, 0x02,
>-  0x10, 0x61, 0x06, 0x10, 0x61, 0x06,
>-  0x02, 0x30, 0x23, 0x02, 0x30, 0x23,
>-  0x40, 0x54, 0x05, 0x40, 0x54, 0x05,
>-  0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
>-  0x81, 0x18, 0x11, 0x81, 0x18, 0x11,
>-  0x14, 0x81, 0x48, 0x14, 0x81, 0x48,
>-  0x98, 0x09, 0x80, 0x98, 0x09, 0x80,
>-  0x08, 0x90, 0x89, 0x08, 0x90, 0x89,
>-  0x62, 0x06, 0x20, 0x62, 0x06, 0x20,
>-  0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
>-  0x8a, 0x08, 0xa0, 0x8a, 0x08, 0xa0,
>-  0x84, 0x48, 0x44, 0x84, 0x48, 0x44
>-};
>-
>-const uint8_t kMaskRandom48_46[276] = {
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
>-  0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
>-  0x00, 0x70, 0x07, 0x00, 0x70, 0x07,
>-  0x0c, 0x10, 0xc1, 0x0c, 0x10, 0xc1,
>-  0x40, 0xc4, 0x0c, 0x40, 0xc4, 0x0c,
>-  0x6a, 0x06, 0xa0, 0x6a, 0x06, 0xa0,
>-  0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
>-  0x24, 0x82, 0x48, 0x24, 0x82, 0x48,
>-  0x89, 0x08, 0x90, 0x89, 0x08, 0x90,
>-  0xc0, 0x2c, 0x02, 0xc0, 0x2c, 0x02,
>-  0x10, 0x61, 0x06, 0x10, 0x61, 0x06,
>-  0x02, 0x30, 0x23, 0x02, 0x30, 0x23,
>-  0x40, 0x54, 0x05, 0x40, 0x54, 0x05,
>-  0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
>-  0x81, 0x18, 0x11, 0x81, 0x18, 0x11,
>-  0x14, 0x81, 0x48, 0x14, 0x81, 0x48,
>-  0x98, 0x09, 0x80, 0x98, 0x09, 0x80,
>-  0x08, 0x90, 0x89, 0x08, 0x90, 0x89,
>-  0x62, 0x06, 0x20, 0x62, 0x06, 0x20,
>-  0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
>-  0x8a, 0x08, 0xa0, 0x8a, 0x08, 0xa0,
>-  0x84, 0x48, 0x44, 0x84, 0x48, 0x44,
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
>-  0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
>-  0x00, 0x70, 0x07, 0x00, 0x70, 0x07,
>-  0x0c, 0x10, 0xc1, 0x0c, 0x10, 0xc1,
>-  0x40, 0xc4, 0x0c, 0x40, 0xc4, 0x0c,
>-  0x6a, 0x06, 0xa0, 0x6a, 0x06, 0xa0,
>-  0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
>-  0x24, 0x82, 0x48, 0x24, 0x82, 0x48,
>-  0x89, 0x08, 0x90, 0x89, 0x08, 0x90,
>-  0xc0, 0x2c, 0x02, 0xc0, 0x2c, 0x02,
>-  0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
>-  0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
>-  0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
>-  0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
>-  0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
>-  0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
>-  0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
>-  0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
>-  0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
>-  0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28,
>-  0x1a, 0xaa, 0xee, 0x1a, 0xaa, 0xee,
>-  0x10, 0xf9, 0xab, 0x12, 0x14, 0xef
>-};
>-
>-const uint8_t kMaskRandom48_47[282] = {
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
>-  0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
>-  0x00, 0x70, 0x07, 0x00, 0x70, 0x07,
>-  0x0c, 0x10, 0xc1, 0x0c, 0x10, 0xc1,
>-  0x40, 0xc4, 0x0c, 0x40, 0xc4, 0x0c,
>-  0x6a, 0x06, 0xa0, 0x6a, 0x06, 0xa0,
>-  0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
>-  0x24, 0x82, 0x48, 0x24, 0x82, 0x48,
>-  0x89, 0x08, 0x90, 0x89, 0x08, 0x90,
>-  0xc0, 0x2c, 0x02, 0xc0, 0x2c, 0x02,
>-  0x10, 0x61, 0x06, 0x10, 0x61, 0x06,
>-  0x02, 0x30, 0x23, 0x02, 0x30, 0x23,
>-  0x40, 0x54, 0x05, 0x40, 0x54, 0x05,
>-  0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
>-  0x81, 0x18, 0x11, 0x81, 0x18, 0x11,
>-  0x14, 0x81, 0x48, 0x14, 0x81, 0x48,
>-  0x98, 0x09, 0x80, 0x98, 0x09, 0x80,
>-  0x08, 0x90, 0x89, 0x08, 0x90, 0x89,
>-  0x62, 0x06, 0x20, 0x62, 0x06, 0x20,
>-  0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
>-  0x8a, 0x08, 0xa0, 0x8a, 0x08, 0xa0,
>-  0x84, 0x48, 0x44, 0x84, 0x48, 0x44,
>-  0x10, 0x61, 0x06, 0x10, 0x61, 0x06,
>-  0x02, 0x30, 0x23, 0x02, 0x30, 0x23,
>-  0x40, 0x54, 0x05, 0x40, 0x54, 0x05,
>-  0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
>-  0x81, 0x18, 0x11, 0x81, 0x18, 0x11,
>-  0x14, 0x81, 0x48, 0x14, 0x81, 0x48,
>-  0x98, 0x09, 0x80, 0x98, 0x09, 0x80,
>-  0x08, 0x90, 0x89, 0x08, 0x90, 0x89,
>-  0x62, 0x06, 0x20, 0x62, 0x06, 0x20,
>-  0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
>-  0x8a, 0x08, 0xa0, 0x8a, 0x08, 0xa0,
>-  0x84, 0x48, 0x44, 0x84, 0x48, 0x44,
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
>-  0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
>-  0x00, 0x70, 0x07, 0x00, 0x70, 0x07,
>-  0x0c, 0x10, 0xc1, 0x0c, 0x10, 0xc1,
>-  0x40, 0xc4, 0x0c, 0x40, 0xc4, 0x0c,
>-  0x6a, 0x06, 0xa0, 0x6a, 0x06, 0xa0,
>-  0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
>-  0x24, 0x82, 0x48, 0x24, 0x82, 0x48,
>-  0x89, 0x08, 0x90, 0x89, 0x08, 0x90,
>-  0xc0, 0x2c, 0x02, 0xc0, 0x2c, 0x02,
>-  0x88, 0x32, 0x59, 0x88, 0x32, 0x59
>-};
>-
>-const uint8_t kMaskRandom48_48[288] = {
>-  0x10, 0x61, 0x06, 0x10, 0x61, 0x06,
>-  0x02, 0x30, 0x23, 0x02, 0x30, 0x23,
>-  0x40, 0x54, 0x05, 0x40, 0x54, 0x05,
>-  0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
>-  0x81, 0x18, 0x11, 0x81, 0x18, 0x11,
>-  0x14, 0x81, 0x48, 0x14, 0x81, 0x48,
>-  0x98, 0x09, 0x80, 0x98, 0x09, 0x80,
>-  0x08, 0x90, 0x89, 0x08, 0x90, 0x89,
>-  0x62, 0x06, 0x20, 0x62, 0x06, 0x20,
>-  0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
>-  0x8a, 0x08, 0xa0, 0x8a, 0x08, 0xa0,
>-  0x84, 0x48, 0x44, 0x84, 0x48, 0x44,
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
>-  0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
>-  0x00, 0x70, 0x07, 0x00, 0x70, 0x07,
>-  0x0c, 0x10, 0xc1, 0x0c, 0x10, 0xc1,
>-  0x40, 0xc4, 0x0c, 0x40, 0xc4, 0x0c,
>-  0x6a, 0x06, 0xa0, 0x6a, 0x06, 0xa0,
>-  0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
>-  0x24, 0x82, 0x48, 0x24, 0x82, 0x48,
>-  0x89, 0x08, 0x90, 0x89, 0x08, 0x90,
>-  0xc0, 0x2c, 0x02, 0xc0, 0x2c, 0x02,
>-  0x88, 0x32, 0x59, 0x88, 0x32, 0x59,
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
>-  0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
>-  0x00, 0x70, 0x07, 0x00, 0x70, 0x07,
>-  0x0c, 0x10, 0xc1, 0x0c, 0x10, 0xc1,
>-  0x40, 0xc4, 0x0c, 0x40, 0xc4, 0x0c,
>-  0x6a, 0x06, 0xa0, 0x6a, 0x06, 0xa0,
>-  0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
>-  0x24, 0x82, 0x48, 0x24, 0x82, 0x48,
>-  0x89, 0x08, 0x90, 0x89, 0x08, 0x90,
>-  0xc0, 0x2c, 0x02, 0xc0, 0x2c, 0x02,
>-  0x10, 0x61, 0x06, 0x10, 0x61, 0x06,
>-  0x02, 0x30, 0x23, 0x02, 0x30, 0x23,
>-  0x40, 0x54, 0x05, 0x40, 0x54, 0x05,
>-  0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
>-  0x81, 0x18, 0x11, 0x81, 0x18, 0x11,
>-  0x14, 0x81, 0x48, 0x14, 0x81, 0x48,
>-  0x98, 0x09, 0x80, 0x98, 0x09, 0x80,
>-  0x08, 0x90, 0x89, 0x08, 0x90, 0x89,
>-  0x62, 0x06, 0x20, 0x62, 0x06, 0x20,
>-  0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
>-  0x8a, 0x08, 0xa0, 0x8a, 0x08, 0xa0,
>-  0x84, 0x48, 0x44, 0x84, 0x48, 0x44,
>-  0xff, 0x9b, 0xdf, 0xec, 0xae, 0x0e
>-};
>-
>-const uint8_t kMaskRandom48_5[30] = {
>-  0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
>-  0x64, 0xb6, 0x4b, 0x64, 0xb6, 0x4b,
>-  0x0e, 0xe0, 0xee, 0x0e, 0xe0, 0xee,
>-  0xa9, 0xca, 0x9c, 0xa9, 0xca, 0x9c,
>-  0xb8, 0x3b, 0x83, 0xb8, 0x3b, 0x83
>-};
>-
>-const uint8_t kMaskRandom48_6[36] = {
>-  0xd1, 0x4d, 0x14, 0xd1, 0x4d, 0x14,
>-  0x45, 0x34, 0x53, 0x45, 0x34, 0x53,
>-  0x22, 0xd2, 0x2d, 0x22, 0xd2, 0x2d,
>-  0x16, 0xc1, 0x6c, 0x16, 0xc1, 0x6c,
>-  0x0b, 0xa0, 0xba, 0x0b, 0xa0, 0xba,
>-  0xe8, 0x8e, 0x88, 0xe8, 0x8e, 0x88
>-};
>-
>-const uint8_t kMaskRandom48_7[42] = {
>-  0xd3, 0x65, 0x36, 0xd3, 0x65, 0x36,
>-  0x25, 0x32, 0x53, 0x25, 0x32, 0x53,
>-  0x30, 0xd3, 0x05, 0x30, 0xd3, 0x05,
>-  0x06, 0x48, 0x6c, 0x06, 0x48, 0x6c,
>-  0xc0, 0xb8, 0x1b, 0xc0, 0xb8, 0x1b,
>-  0x2a, 0xa2, 0xaa, 0x2a, 0xa2, 0xaa,
>-  0xa8, 0x4e, 0x84, 0xa8, 0x4e, 0x84
>-};
>-
>-const uint8_t kMaskRandom48_8[48] = {
>-  0x81, 0x60, 0x16, 0x81, 0x60, 0x16,
>-  0x40, 0x3c, 0x03, 0x40, 0x3c, 0x03,
>-  0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
>-  0x06, 0x50, 0x65, 0x06, 0x50, 0x65,
>-  0x20, 0x4a, 0x84, 0x20, 0x4a, 0x84,
>-  0x8a, 0xa0, 0xaa, 0x8a, 0xa0, 0xaa,
>-  0x33, 0x03, 0x30, 0x33, 0x03, 0x30,
>-  0x4c, 0x84, 0xc8, 0x4c, 0x84, 0xc8
>-};
>-
>-const uint8_t kMaskRandom48_9[54] = {
>-  0xd3, 0x65, 0x36, 0xd3, 0x65, 0x36,
>-  0x64, 0x26, 0x42, 0x64, 0x26, 0x42,
>-  0x18, 0x41, 0xc4, 0x18, 0x41, 0xc4,
>-  0xa0, 0x4a, 0x04, 0xa0, 0x4a, 0x04,
>-  0x81, 0x38, 0x13, 0x81, 0x38, 0x13,
>-  0x22, 0xa2, 0x2a, 0x22, 0xa2, 0x2a,
>-  0x08, 0x70, 0x87, 0x08, 0x70, 0x87,
>-  0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
>-  0x01, 0xc0, 0x1c, 0x01, 0xc0, 0x1c
>-};
>-
>-const uint8_t kMaskRandom4_1[2] = {
>-  0xf0, 0x00
>-};
>-
>-const uint8_t kMaskRandom4_2[4] = {
>-  0xc0, 0x00,
>-  0xb0, 0x00
>-};
>-
>-const uint8_t kMaskRandom4_3[6] = {
>-  0xc0, 0x00,
>-  0xb0, 0x00,
>-  0x60, 0x00
>-};
>-
>-const uint8_t kMaskRandom4_4[8] = {
>-  0xc0, 0x00,
>-  0xa0, 0x00,
>-  0x30, 0x00,
>-  0x50, 0x00
>-};
>-
>-const uint8_t kMaskRandom5_1[2] = {
>-  0xf8, 0x00
>-};
>-
>-const uint8_t kMaskRandom5_2[4] = {
>-  0xa8, 0x00,
>-  0xd0, 0x00
>-};
>-
>-const uint8_t kMaskRandom5_3[6] = {
>-  0xb0, 0x00,
>-  0xc8, 0x00,
>-  0x50, 0x00
>-};
>-
>-const uint8_t kMaskRandom5_4[8] = {
>-  0xc8, 0x00,
>-  0xb0, 0x00,
>-  0x50, 0x00,
>-  0x28, 0x00
>-};
>-
>-const uint8_t kMaskRandom5_5[10] = {
>-  0xc0, 0x00,
>-  0x30, 0x00,
>-  0x18, 0x00,
>-  0xa0, 0x00,
>-  0x48, 0x00
>-};
>-
>-const uint8_t kMaskRandom6_1[2] = {
>-  0xfc, 0x00
>-};
>-
>-const uint8_t kMaskRandom6_2[4] = {
>-  0xa8, 0x00,
>-  0xd4, 0x00
>-};
>-
>-const uint8_t kMaskRandom6_3[6] = {
>-  0xd0, 0x00,
>-  0x68, 0x00,
>-  0xa4, 0x00
>-};
>-
>-const uint8_t kMaskRandom6_4[8] = {
>-  0xa8, 0x00,
>-  0x58, 0x00,
>-  0x64, 0x00,
>-  0x94, 0x00
>-};
>-
>-const uint8_t kMaskRandom6_5[10] = {
>-  0xa8, 0x00,
>-  0x84, 0x00,
>-  0x64, 0x00,
>-  0x90, 0x00,
>-  0x58, 0x00
>-};
>-
>-const uint8_t kMaskRandom6_6[12] = {
>-  0x98, 0x00,
>-  0x64, 0x00,
>-  0x50, 0x00,
>-  0x14, 0x00,
>-  0xa8, 0x00,
>-  0xe0, 0x00
>-};
>-
>-const uint8_t kMaskRandom7_1[2] = {
>-  0xfe, 0x00
>-};
>-
>-const uint8_t kMaskRandom7_2[4] = {
>-  0xd4, 0x00,
>-  0xaa, 0x00
>-};
>-
>-const uint8_t kMaskRandom7_3[6] = {
>-  0xd0, 0x00,
>-  0xaa, 0x00,
>-  0x64, 0x00
>-};
>-
>-const uint8_t kMaskRandom7_4[8] = {
>-  0xd0, 0x00,
>-  0xaa, 0x00,
>-  0x64, 0x00,
>-  0x1c, 0x00
>-};
>-
>-const uint8_t kMaskRandom7_5[10] = {
>-  0x0c, 0x00,
>-  0xb0, 0x00,
>-  0x1a, 0x00,
>-  0xc4, 0x00,
>-  0x62, 0x00
>-};
>-
>-const uint8_t kMaskRandom7_6[12] = {
>-  0x8c, 0x00,
>-  0x4a, 0x00,
>-  0x64, 0x00,
>-  0xd0, 0x00,
>-  0xa0, 0x00,
>-  0x32, 0x00
>-};
>-
>-const uint8_t kMaskRandom7_7[14] = {
>-  0x4a, 0x00,
>-  0x94, 0x00,
>-  0x1a, 0x00,
>-  0xc4, 0x00,
>-  0x28, 0x00,
>-  0xc2, 0x00,
>-  0x34, 0x00
>-};
>-
>-const uint8_t kMaskRandom8_1[2] = {
>-  0xff, 0x00
>-};
>-
>-const uint8_t kMaskRandom8_2[4] = {
>-  0xaa, 0x00,
>-  0xd5, 0x00
>-};
>-
>-const uint8_t kMaskRandom8_3[6] = {
>-  0xc5, 0x00,
>-  0x92, 0x00,
>-  0x6a, 0x00
>-};
>-
>-const uint8_t kMaskRandom8_4[8] = {
>-  0x45, 0x00,
>-  0xb4, 0x00,
>-  0x6a, 0x00,
>-  0x89, 0x00
>-};
>-
>-const uint8_t kMaskRandom8_5[10] = {
>-  0x8c, 0x00,
>-  0x92, 0x00,
>-  0x2b, 0x00,
>-  0x51, 0x00,
>-  0x64, 0x00
>-};
>-
>-const uint8_t kMaskRandom8_6[12] = {
>-  0xa1, 0x00,
>-  0x52, 0x00,
>-  0x91, 0x00,
>-  0x2a, 0x00,
>-  0xc4, 0x00,
>-  0x4c, 0x00
>-};
>-
>-const uint8_t kMaskRandom8_7[14] = {
>-  0x15, 0x00,
>-  0xc2, 0x00,
>-  0x25, 0x00,
>-  0x62, 0x00,
>-  0x58, 0x00,
>-  0x8c, 0x00,
>-  0xa3, 0x00
>-};
>-
>-const uint8_t kMaskRandom8_8[16] = {
>-  0x25, 0x00,
>-  0x8a, 0x00,
>-  0x91, 0x00,
>-  0x68, 0x00,
>-  0x32, 0x00,
>-  0x43, 0x00,
>-  0xc4, 0x00,
>-  0x1c, 0x00
>-};
>-
>-const uint8_t kMaskRandom9_1[2] = {
>-  0xff, 0x80
>-};
>-
>-const uint8_t kMaskRandom9_2[4] = {
>-  0xaa, 0x80,
>-  0xd5, 0x00
>-};
>-
>-const uint8_t kMaskRandom9_3[6] = {
>-  0xa5, 0x00,
>-  0xc8, 0x00,
>-  0x52, 0x80
>-};
>-
>-const uint8_t kMaskRandom9_4[8] = {
>-  0xa2, 0x00,
>-  0xc9, 0x00,
>-  0x52, 0x80,
>-  0x24, 0x80
>-};
>-
>-const uint8_t kMaskRandom9_5[10] = {
>-  0x8c, 0x00,
>-  0x25, 0x00,
>-  0x92, 0x80,
>-  0x41, 0x80,
>-  0x58, 0x00
>-};
>-
>-const uint8_t kMaskRandom9_6[12] = {
>-  0x84, 0x80,
>-  0x27, 0x00,
>-  0x51, 0x80,
>-  0x1a, 0x00,
>-  0x68, 0x00,
>-  0x89, 0x00
>-};
>-
>-const uint8_t kMaskRandom9_7[14] = {
>-  0x8c, 0x00,
>-  0x47, 0x00,
>-  0x81, 0x80,
>-  0x12, 0x80,
>-  0x58, 0x00,
>-  0x28, 0x80,
>-  0xb4, 0x00
>-};
>-
>-const uint8_t kMaskRandom9_8[16] = {
>-  0x2c, 0x00,
>-  0x91, 0x00,
>-  0x40, 0x80,
>-  0x06, 0x80,
>-  0xc8, 0x00,
>-  0x45, 0x00,
>-  0x30, 0x80,
>-  0xa2, 0x00
>-};
>-
>-const uint8_t kMaskRandom9_9[18] = {
>-  0x4c, 0x00,
>-  0x62, 0x00,
>-  0x91, 0x00,
>-  0x42, 0x80,
>-  0xa4, 0x00,
>-  0x13, 0x00,
>-  0x30, 0x80,
>-  0x88, 0x80,
>-  0x09, 0x00
>-};
>-
>-const uint8_t* const kPacketMaskRandom1[1] = {
>-  kMaskRandom1_1
>-};
>-
>-const uint8_t* const kPacketMaskRandom2[2] = {
>-  kMaskRandom2_1,
>-  kMaskRandom2_2
>-};
>-
>-const uint8_t* const kPacketMaskRandom3[3] = {
>-  kMaskRandom3_1,
>-  kMaskRandom3_2,
>-  kMaskRandom3_3
>-};
>-
>-const uint8_t* const kPacketMaskRandom4[4] = {
>-  kMaskRandom4_1,
>-  kMaskRandom4_2,
>-  kMaskRandom4_3,
>-  kMaskRandom4_4
>-};
>-
>-const uint8_t* const kPacketMaskRandom5[5] = {
>-  kMaskRandom5_1,
>-  kMaskRandom5_2,
>-  kMaskRandom5_3,
>-  kMaskRandom5_4,
>-  kMaskRandom5_5
>-};
>-
>-const uint8_t* const kPacketMaskRandom6[6] = {
>-  kMaskRandom6_1,
>-  kMaskRandom6_2,
>-  kMaskRandom6_3,
>-  kMaskRandom6_4,
>-  kMaskRandom6_5,
>-  kMaskRandom6_6
>-};
>-
>-const uint8_t* const kPacketMaskRandom7[7] = {
>-  kMaskRandom7_1,
>-  kMaskRandom7_2,
>-  kMaskRandom7_3,
>-  kMaskRandom7_4,
>-  kMaskRandom7_5,
>-  kMaskRandom7_6,
>-  kMaskRandom7_7
>-};
>-
>-const uint8_t* const kPacketMaskRandom8[8] = {
>-  kMaskRandom8_1,
>-  kMaskRandom8_2,
>-  kMaskRandom8_3,
>-  kMaskRandom8_4,
>-  kMaskRandom8_5,
>-  kMaskRandom8_6,
>-  kMaskRandom8_7,
>-  kMaskRandom8_8
>-};
>-
>-const uint8_t* const kPacketMaskRandom9[9] = {
>-  kMaskRandom9_1,
>-  kMaskRandom9_2,
>-  kMaskRandom9_3,
>-  kMaskRandom9_4,
>-  kMaskRandom9_5,
>-  kMaskRandom9_6,
>-  kMaskRandom9_7,
>-  kMaskRandom9_8,
>-  kMaskRandom9_9
>-};
>-
>-const uint8_t* const kPacketMaskRandom10[10] = {
>-  kMaskRandom10_1,
>-  kMaskRandom10_2,
>-  kMaskRandom10_3,
>-  kMaskRandom10_4,
>-  kMaskRandom10_5,
>-  kMaskRandom10_6,
>-  kMaskRandom10_7,
>-  kMaskRandom10_8,
>-  kMaskRandom10_9,
>-  kMaskRandom10_10
>-};
>-
>-const uint8_t* const kPacketMaskRandom11[11] = {
>-  kMaskRandom11_1,
>-  kMaskRandom11_2,
>-  kMaskRandom11_3,
>-  kMaskRandom11_4,
>-  kMaskRandom11_5,
>-  kMaskRandom11_6,
>-  kMaskRandom11_7,
>-  kMaskRandom11_8,
>-  kMaskRandom11_9,
>-  kMaskRandom11_10,
>-  kMaskRandom11_11
>-};
>-
>-const uint8_t* const kPacketMaskRandom12[12] = {
>-  kMaskRandom12_1,
>-  kMaskRandom12_2,
>-  kMaskRandom12_3,
>-  kMaskRandom12_4,
>-  kMaskRandom12_5,
>-  kMaskRandom12_6,
>-  kMaskRandom12_7,
>-  kMaskRandom12_8,
>-  kMaskRandom12_9,
>-  kMaskRandom12_10,
>-  kMaskRandom12_11,
>-  kMaskRandom12_12
>-};
>-
>-const uint8_t* const kPacketMaskRandom13[13] = {
>-  kMaskRandom13_1,
>-  kMaskRandom13_2,
>-  kMaskRandom13_3,
>-  kMaskRandom13_4,
>-  kMaskRandom13_5,
>-  kMaskRandom13_6,
>-  kMaskRandom13_7,
>-  kMaskRandom13_8,
>-  kMaskRandom13_9,
>-  kMaskRandom13_10,
>-  kMaskRandom13_11,
>-  kMaskRandom13_12,
>-  kMaskRandom13_13
>-};
>-
>-const uint8_t* const kPacketMaskRandom14[14] = {
>-  kMaskRandom14_1,
>-  kMaskRandom14_2,
>-  kMaskRandom14_3,
>-  kMaskRandom14_4,
>-  kMaskRandom14_5,
>-  kMaskRandom14_6,
>-  kMaskRandom14_7,
>-  kMaskRandom14_8,
>-  kMaskRandom14_9,
>-  kMaskRandom14_10,
>-  kMaskRandom14_11,
>-  kMaskRandom14_12,
>-  kMaskRandom14_13,
>-  kMaskRandom14_14
>-};
>-
>-const uint8_t* const kPacketMaskRandom15[15] = {
>-  kMaskRandom15_1,
>-  kMaskRandom15_2,
>-  kMaskRandom15_3,
>-  kMaskRandom15_4,
>-  kMaskRandom15_5,
>-  kMaskRandom15_6,
>-  kMaskRandom15_7,
>-  kMaskRandom15_8,
>-  kMaskRandom15_9,
>-  kMaskRandom15_10,
>-  kMaskRandom15_11,
>-  kMaskRandom15_12,
>-  kMaskRandom15_13,
>-  kMaskRandom15_14,
>-  kMaskRandom15_15
>-};
>-
>-const uint8_t* const kPacketMaskRandom16[16] = {
>-  kMaskRandom16_1,
>-  kMaskRandom16_2,
>-  kMaskRandom16_3,
>-  kMaskRandom16_4,
>-  kMaskRandom16_5,
>-  kMaskRandom16_6,
>-  kMaskRandom16_7,
>-  kMaskRandom16_8,
>-  kMaskRandom16_9,
>-  kMaskRandom16_10,
>-  kMaskRandom16_11,
>-  kMaskRandom16_12,
>-  kMaskRandom16_13,
>-  kMaskRandom16_14,
>-  kMaskRandom16_15,
>-  kMaskRandom16_16
>-};
>-
>-const uint8_t* const kPacketMaskRandom17[17] = {
>-  kMaskRandom17_1,
>-  kMaskRandom17_2,
>-  kMaskRandom17_3,
>-  kMaskRandom17_4,
>-  kMaskRandom17_5,
>-  kMaskRandom17_6,
>-  kMaskRandom17_7,
>-  kMaskRandom17_8,
>-  kMaskRandom17_9,
>-  kMaskRandom17_10,
>-  kMaskRandom17_11,
>-  kMaskRandom17_12,
>-  kMaskRandom17_13,
>-  kMaskRandom17_14,
>-  kMaskRandom17_15,
>-  kMaskRandom17_16,
>-  kMaskRandom17_17
>-};
>-
>-const uint8_t* const kPacketMaskRandom18[18] = {
>-  kMaskRandom18_1,
>-  kMaskRandom18_2,
>-  kMaskRandom18_3,
>-  kMaskRandom18_4,
>-  kMaskRandom18_5,
>-  kMaskRandom18_6,
>-  kMaskRandom18_7,
>-  kMaskRandom18_8,
>-  kMaskRandom18_9,
>-  kMaskRandom18_10,
>-  kMaskRandom18_11,
>-  kMaskRandom18_12,
>-  kMaskRandom18_13,
>-  kMaskRandom18_14,
>-  kMaskRandom18_15,
>-  kMaskRandom18_16,
>-  kMaskRandom18_17,
>-  kMaskRandom18_18
>-};
>-
>-const uint8_t* const kPacketMaskRandom19[19] = {
>-  kMaskRandom19_1,
>-  kMaskRandom19_2,
>-  kMaskRandom19_3,
>-  kMaskRandom19_4,
>-  kMaskRandom19_5,
>-  kMaskRandom19_6,
>-  kMaskRandom19_7,
>-  kMaskRandom19_8,
>-  kMaskRandom19_9,
>-  kMaskRandom19_10,
>-  kMaskRandom19_11,
>-  kMaskRandom19_12,
>-  kMaskRandom19_13,
>-  kMaskRandom19_14,
>-  kMaskRandom19_15,
>-  kMaskRandom19_16,
>-  kMaskRandom19_17,
>-  kMaskRandom19_18,
>-  kMaskRandom19_19
>-};
>-
>-const uint8_t* const kPacketMaskRandom20[20] = {
>-  kMaskRandom20_1,
>-  kMaskRandom20_2,
>-  kMaskRandom20_3,
>-  kMaskRandom20_4,
>-  kMaskRandom20_5,
>-  kMaskRandom20_6,
>-  kMaskRandom20_7,
>-  kMaskRandom20_8,
>-  kMaskRandom20_9,
>-  kMaskRandom20_10,
>-  kMaskRandom20_11,
>-  kMaskRandom20_12,
>-  kMaskRandom20_13,
>-  kMaskRandom20_14,
>-  kMaskRandom20_15,
>-  kMaskRandom20_16,
>-  kMaskRandom20_17,
>-  kMaskRandom20_18,
>-  kMaskRandom20_19,
>-  kMaskRandom20_20
>-};
>-
>-const uint8_t* const kPacketMaskRandom21[21] = {
>-  kMaskRandom21_1,
>-  kMaskRandom21_2,
>-  kMaskRandom21_3,
>-  kMaskRandom21_4,
>-  kMaskRandom21_5,
>-  kMaskRandom21_6,
>-  kMaskRandom21_7,
>-  kMaskRandom21_8,
>-  kMaskRandom21_9,
>-  kMaskRandom21_10,
>-  kMaskRandom21_11,
>-  kMaskRandom21_12,
>-  kMaskRandom21_13,
>-  kMaskRandom21_14,
>-  kMaskRandom21_15,
>-  kMaskRandom21_16,
>-  kMaskRandom21_17,
>-  kMaskRandom21_18,
>-  kMaskRandom21_19,
>-  kMaskRandom21_20,
>-  kMaskRandom21_21
>-};
>-
>-const uint8_t* const kPacketMaskRandom22[22] = {
>-  kMaskRandom22_1,
>-  kMaskRandom22_2,
>-  kMaskRandom22_3,
>-  kMaskRandom22_4,
>-  kMaskRandom22_5,
>-  kMaskRandom22_6,
>-  kMaskRandom22_7,
>-  kMaskRandom22_8,
>-  kMaskRandom22_9,
>-  kMaskRandom22_10,
>-  kMaskRandom22_11,
>-  kMaskRandom22_12,
>-  kMaskRandom22_13,
>-  kMaskRandom22_14,
>-  kMaskRandom22_15,
>-  kMaskRandom22_16,
>-  kMaskRandom22_17,
>-  kMaskRandom22_18,
>-  kMaskRandom22_19,
>-  kMaskRandom22_20,
>-  kMaskRandom22_21,
>-  kMaskRandom22_22
>-};
>-
>-const uint8_t* const kPacketMaskRandom23[23] = {
>-  kMaskRandom23_1,
>-  kMaskRandom23_2,
>-  kMaskRandom23_3,
>-  kMaskRandom23_4,
>-  kMaskRandom23_5,
>-  kMaskRandom23_6,
>-  kMaskRandom23_7,
>-  kMaskRandom23_8,
>-  kMaskRandom23_9,
>-  kMaskRandom23_10,
>-  kMaskRandom23_11,
>-  kMaskRandom23_12,
>-  kMaskRandom23_13,
>-  kMaskRandom23_14,
>-  kMaskRandom23_15,
>-  kMaskRandom23_16,
>-  kMaskRandom23_17,
>-  kMaskRandom23_18,
>-  kMaskRandom23_19,
>-  kMaskRandom23_20,
>-  kMaskRandom23_21,
>-  kMaskRandom23_22,
>-  kMaskRandom23_23
>-};
>-
>-const uint8_t* const kPacketMaskRandom24[24] = {
>-  kMaskRandom24_1,
>-  kMaskRandom24_2,
>-  kMaskRandom24_3,
>-  kMaskRandom24_4,
>-  kMaskRandom24_5,
>-  kMaskRandom24_6,
>-  kMaskRandom24_7,
>-  kMaskRandom24_8,
>-  kMaskRandom24_9,
>-  kMaskRandom24_10,
>-  kMaskRandom24_11,
>-  kMaskRandom24_12,
>-  kMaskRandom24_13,
>-  kMaskRandom24_14,
>-  kMaskRandom24_15,
>-  kMaskRandom24_16,
>-  kMaskRandom24_17,
>-  kMaskRandom24_18,
>-  kMaskRandom24_19,
>-  kMaskRandom24_20,
>-  kMaskRandom24_21,
>-  kMaskRandom24_22,
>-  kMaskRandom24_23,
>-  kMaskRandom24_24
>-};
>-
>-const uint8_t* const kPacketMaskRandom25[25] = {
>-  kMaskRandom25_1,
>-  kMaskRandom25_2,
>-  kMaskRandom25_3,
>-  kMaskRandom25_4,
>-  kMaskRandom25_5,
>-  kMaskRandom25_6,
>-  kMaskRandom25_7,
>-  kMaskRandom25_8,
>-  kMaskRandom25_9,
>-  kMaskRandom25_10,
>-  kMaskRandom25_11,
>-  kMaskRandom25_12,
>-  kMaskRandom25_13,
>-  kMaskRandom25_14,
>-  kMaskRandom25_15,
>-  kMaskRandom25_16,
>-  kMaskRandom25_17,
>-  kMaskRandom25_18,
>-  kMaskRandom25_19,
>-  kMaskRandom25_20,
>-  kMaskRandom25_21,
>-  kMaskRandom25_22,
>-  kMaskRandom25_23,
>-  kMaskRandom25_24,
>-  kMaskRandom25_25
>-};
>-
>-const uint8_t* const kPacketMaskRandom26[26] = {
>-  kMaskRandom26_1,
>-  kMaskRandom26_2,
>-  kMaskRandom26_3,
>-  kMaskRandom26_4,
>-  kMaskRandom26_5,
>-  kMaskRandom26_6,
>-  kMaskRandom26_7,
>-  kMaskRandom26_8,
>-  kMaskRandom26_9,
>-  kMaskRandom26_10,
>-  kMaskRandom26_11,
>-  kMaskRandom26_12,
>-  kMaskRandom26_13,
>-  kMaskRandom26_14,
>-  kMaskRandom26_15,
>-  kMaskRandom26_16,
>-  kMaskRandom26_17,
>-  kMaskRandom26_18,
>-  kMaskRandom26_19,
>-  kMaskRandom26_20,
>-  kMaskRandom26_21,
>-  kMaskRandom26_22,
>-  kMaskRandom26_23,
>-  kMaskRandom26_24,
>-  kMaskRandom26_25,
>-  kMaskRandom26_26
>-};
>-
>-const uint8_t* const kPacketMaskRandom27[27] = {
>-  kMaskRandom27_1,
>-  kMaskRandom27_2,
>-  kMaskRandom27_3,
>-  kMaskRandom27_4,
>-  kMaskRandom27_5,
>-  kMaskRandom27_6,
>-  kMaskRandom27_7,
>-  kMaskRandom27_8,
>-  kMaskRandom27_9,
>-  kMaskRandom27_10,
>-  kMaskRandom27_11,
>-  kMaskRandom27_12,
>-  kMaskRandom27_13,
>-  kMaskRandom27_14,
>-  kMaskRandom27_15,
>-  kMaskRandom27_16,
>-  kMaskRandom27_17,
>-  kMaskRandom27_18,
>-  kMaskRandom27_19,
>-  kMaskRandom27_20,
>-  kMaskRandom27_21,
>-  kMaskRandom27_22,
>-  kMaskRandom27_23,
>-  kMaskRandom27_24,
>-  kMaskRandom27_25,
>-  kMaskRandom27_26,
>-  kMaskRandom27_27
>-};
>-
>-const uint8_t* const kPacketMaskRandom28[28] = {
>-  kMaskRandom28_1,
>-  kMaskRandom28_2,
>-  kMaskRandom28_3,
>-  kMaskRandom28_4,
>-  kMaskRandom28_5,
>-  kMaskRandom28_6,
>-  kMaskRandom28_7,
>-  kMaskRandom28_8,
>-  kMaskRandom28_9,
>-  kMaskRandom28_10,
>-  kMaskRandom28_11,
>-  kMaskRandom28_12,
>-  kMaskRandom28_13,
>-  kMaskRandom28_14,
>-  kMaskRandom28_15,
>-  kMaskRandom28_16,
>-  kMaskRandom28_17,
>-  kMaskRandom28_18,
>-  kMaskRandom28_19,
>-  kMaskRandom28_20,
>-  kMaskRandom28_21,
>-  kMaskRandom28_22,
>-  kMaskRandom28_23,
>-  kMaskRandom28_24,
>-  kMaskRandom28_25,
>-  kMaskRandom28_26,
>-  kMaskRandom28_27,
>-  kMaskRandom28_28
>-};
>-
>-const uint8_t* const kPacketMaskRandom29[29] = {
>-  kMaskRandom29_1,
>-  kMaskRandom29_2,
>-  kMaskRandom29_3,
>-  kMaskRandom29_4,
>-  kMaskRandom29_5,
>-  kMaskRandom29_6,
>-  kMaskRandom29_7,
>-  kMaskRandom29_8,
>-  kMaskRandom29_9,
>-  kMaskRandom29_10,
>-  kMaskRandom29_11,
>-  kMaskRandom29_12,
>-  kMaskRandom29_13,
>-  kMaskRandom29_14,
>-  kMaskRandom29_15,
>-  kMaskRandom29_16,
>-  kMaskRandom29_17,
>-  kMaskRandom29_18,
>-  kMaskRandom29_19,
>-  kMaskRandom29_20,
>-  kMaskRandom29_21,
>-  kMaskRandom29_22,
>-  kMaskRandom29_23,
>-  kMaskRandom29_24,
>-  kMaskRandom29_25,
>-  kMaskRandom29_26,
>-  kMaskRandom29_27,
>-  kMaskRandom29_28,
>-  kMaskRandom29_29
>-};
>-
>-const uint8_t* const kPacketMaskRandom30[30] = {
>-  kMaskRandom30_1,
>-  kMaskRandom30_2,
>-  kMaskRandom30_3,
>-  kMaskRandom30_4,
>-  kMaskRandom30_5,
>-  kMaskRandom30_6,
>-  kMaskRandom30_7,
>-  kMaskRandom30_8,
>-  kMaskRandom30_9,
>-  kMaskRandom30_10,
>-  kMaskRandom30_11,
>-  kMaskRandom30_12,
>-  kMaskRandom30_13,
>-  kMaskRandom30_14,
>-  kMaskRandom30_15,
>-  kMaskRandom30_16,
>-  kMaskRandom30_17,
>-  kMaskRandom30_18,
>-  kMaskRandom30_19,
>-  kMaskRandom30_20,
>-  kMaskRandom30_21,
>-  kMaskRandom30_22,
>-  kMaskRandom30_23,
>-  kMaskRandom30_24,
>-  kMaskRandom30_25,
>-  kMaskRandom30_26,
>-  kMaskRandom30_27,
>-  kMaskRandom30_28,
>-  kMaskRandom30_29,
>-  kMaskRandom30_30
>-};
>-
>-const uint8_t* const kPacketMaskRandom31[31] = {
>-  kMaskRandom31_1,
>-  kMaskRandom31_2,
>-  kMaskRandom31_3,
>-  kMaskRandom31_4,
>-  kMaskRandom31_5,
>-  kMaskRandom31_6,
>-  kMaskRandom31_7,
>-  kMaskRandom31_8,
>-  kMaskRandom31_9,
>-  kMaskRandom31_10,
>-  kMaskRandom31_11,
>-  kMaskRandom31_12,
>-  kMaskRandom31_13,
>-  kMaskRandom31_14,
>-  kMaskRandom31_15,
>-  kMaskRandom31_16,
>-  kMaskRandom31_17,
>-  kMaskRandom31_18,
>-  kMaskRandom31_19,
>-  kMaskRandom31_20,
>-  kMaskRandom31_21,
>-  kMaskRandom31_22,
>-  kMaskRandom31_23,
>-  kMaskRandom31_24,
>-  kMaskRandom31_25,
>-  kMaskRandom31_26,
>-  kMaskRandom31_27,
>-  kMaskRandom31_28,
>-  kMaskRandom31_29,
>-  kMaskRandom31_30,
>-  kMaskRandom31_31
>-};
>-
>-const uint8_t* const kPacketMaskRandom32[32] = {
>-  kMaskRandom32_1,
>-  kMaskRandom32_2,
>-  kMaskRandom32_3,
>-  kMaskRandom32_4,
>-  kMaskRandom32_5,
>-  kMaskRandom32_6,
>-  kMaskRandom32_7,
>-  kMaskRandom32_8,
>-  kMaskRandom32_9,
>-  kMaskRandom32_10,
>-  kMaskRandom32_11,
>-  kMaskRandom32_12,
>-  kMaskRandom32_13,
>-  kMaskRandom32_14,
>-  kMaskRandom32_15,
>-  kMaskRandom32_16,
>-  kMaskRandom32_17,
>-  kMaskRandom32_18,
>-  kMaskRandom32_19,
>-  kMaskRandom32_20,
>-  kMaskRandom32_21,
>-  kMaskRandom32_22,
>-  kMaskRandom32_23,
>-  kMaskRandom32_24,
>-  kMaskRandom32_25,
>-  kMaskRandom32_26,
>-  kMaskRandom32_27,
>-  kMaskRandom32_28,
>-  kMaskRandom32_29,
>-  kMaskRandom32_30,
>-  kMaskRandom32_31,
>-  kMaskRandom32_32
>-};
>-
>-const uint8_t* const kPacketMaskRandom33[33] = {
>-  kMaskRandom33_1,
>-  kMaskRandom33_2,
>-  kMaskRandom33_3,
>-  kMaskRandom33_4,
>-  kMaskRandom33_5,
>-  kMaskRandom33_6,
>-  kMaskRandom33_7,
>-  kMaskRandom33_8,
>-  kMaskRandom33_9,
>-  kMaskRandom33_10,
>-  kMaskRandom33_11,
>-  kMaskRandom33_12,
>-  kMaskRandom33_13,
>-  kMaskRandom33_14,
>-  kMaskRandom33_15,
>-  kMaskRandom33_16,
>-  kMaskRandom33_17,
>-  kMaskRandom33_18,
>-  kMaskRandom33_19,
>-  kMaskRandom33_20,
>-  kMaskRandom33_21,
>-  kMaskRandom33_22,
>-  kMaskRandom33_23,
>-  kMaskRandom33_24,
>-  kMaskRandom33_25,
>-  kMaskRandom33_26,
>-  kMaskRandom33_27,
>-  kMaskRandom33_28,
>-  kMaskRandom33_29,
>-  kMaskRandom33_30,
>-  kMaskRandom33_31,
>-  kMaskRandom33_32,
>-  kMaskRandom33_33
>-};
>-
>-const uint8_t* const kPacketMaskRandom34[34] = {
>-  kMaskRandom34_1,
>-  kMaskRandom34_2,
>-  kMaskRandom34_3,
>-  kMaskRandom34_4,
>-  kMaskRandom34_5,
>-  kMaskRandom34_6,
>-  kMaskRandom34_7,
>-  kMaskRandom34_8,
>-  kMaskRandom34_9,
>-  kMaskRandom34_10,
>-  kMaskRandom34_11,
>-  kMaskRandom34_12,
>-  kMaskRandom34_13,
>-  kMaskRandom34_14,
>-  kMaskRandom34_15,
>-  kMaskRandom34_16,
>-  kMaskRandom34_17,
>-  kMaskRandom34_18,
>-  kMaskRandom34_19,
>-  kMaskRandom34_20,
>-  kMaskRandom34_21,
>-  kMaskRandom34_22,
>-  kMaskRandom34_23,
>-  kMaskRandom34_24,
>-  kMaskRandom34_25,
>-  kMaskRandom34_26,
>-  kMaskRandom34_27,
>-  kMaskRandom34_28,
>-  kMaskRandom34_29,
>-  kMaskRandom34_30,
>-  kMaskRandom34_31,
>-  kMaskRandom34_32,
>-  kMaskRandom34_33,
>-  kMaskRandom34_34
>-};
>-
>-const uint8_t* const kPacketMaskRandom35[35] = {
>-  kMaskRandom35_1,
>-  kMaskRandom35_2,
>-  kMaskRandom35_3,
>-  kMaskRandom35_4,
>-  kMaskRandom35_5,
>-  kMaskRandom35_6,
>-  kMaskRandom35_7,
>-  kMaskRandom35_8,
>-  kMaskRandom35_9,
>-  kMaskRandom35_10,
>-  kMaskRandom35_11,
>-  kMaskRandom35_12,
>-  kMaskRandom35_13,
>-  kMaskRandom35_14,
>-  kMaskRandom35_15,
>-  kMaskRandom35_16,
>-  kMaskRandom35_17,
>-  kMaskRandom35_18,
>-  kMaskRandom35_19,
>-  kMaskRandom35_20,
>-  kMaskRandom35_21,
>-  kMaskRandom35_22,
>-  kMaskRandom35_23,
>-  kMaskRandom35_24,
>-  kMaskRandom35_25,
>-  kMaskRandom35_26,
>-  kMaskRandom35_27,
>-  kMaskRandom35_28,
>-  kMaskRandom35_29,
>-  kMaskRandom35_30,
>-  kMaskRandom35_31,
>-  kMaskRandom35_32,
>-  kMaskRandom35_33,
>-  kMaskRandom35_34,
>-  kMaskRandom35_35
>-};
>-
>-const uint8_t* const kPacketMaskRandom36[36] = {
>-  kMaskRandom36_1,
>-  kMaskRandom36_2,
>-  kMaskRandom36_3,
>-  kMaskRandom36_4,
>-  kMaskRandom36_5,
>-  kMaskRandom36_6,
>-  kMaskRandom36_7,
>-  kMaskRandom36_8,
>-  kMaskRandom36_9,
>-  kMaskRandom36_10,
>-  kMaskRandom36_11,
>-  kMaskRandom36_12,
>-  kMaskRandom36_13,
>-  kMaskRandom36_14,
>-  kMaskRandom36_15,
>-  kMaskRandom36_16,
>-  kMaskRandom36_17,
>-  kMaskRandom36_18,
>-  kMaskRandom36_19,
>-  kMaskRandom36_20,
>-  kMaskRandom36_21,
>-  kMaskRandom36_22,
>-  kMaskRandom36_23,
>-  kMaskRandom36_24,
>-  kMaskRandom36_25,
>-  kMaskRandom36_26,
>-  kMaskRandom36_27,
>-  kMaskRandom36_28,
>-  kMaskRandom36_29,
>-  kMaskRandom36_30,
>-  kMaskRandom36_31,
>-  kMaskRandom36_32,
>-  kMaskRandom36_33,
>-  kMaskRandom36_34,
>-  kMaskRandom36_35,
>-  kMaskRandom36_36
>-};
>-
>-const uint8_t* const kPacketMaskRandom37[37] = {
>-  kMaskRandom37_1,
>-  kMaskRandom37_2,
>-  kMaskRandom37_3,
>-  kMaskRandom37_4,
>-  kMaskRandom37_5,
>-  kMaskRandom37_6,
>-  kMaskRandom37_7,
>-  kMaskRandom37_8,
>-  kMaskRandom37_9,
>-  kMaskRandom37_10,
>-  kMaskRandom37_11,
>-  kMaskRandom37_12,
>-  kMaskRandom37_13,
>-  kMaskRandom37_14,
>-  kMaskRandom37_15,
>-  kMaskRandom37_16,
>-  kMaskRandom37_17,
>-  kMaskRandom37_18,
>-  kMaskRandom37_19,
>-  kMaskRandom37_20,
>-  kMaskRandom37_21,
>-  kMaskRandom37_22,
>-  kMaskRandom37_23,
>-  kMaskRandom37_24,
>-  kMaskRandom37_25,
>-  kMaskRandom37_26,
>-  kMaskRandom37_27,
>-  kMaskRandom37_28,
>-  kMaskRandom37_29,
>-  kMaskRandom37_30,
>-  kMaskRandom37_31,
>-  kMaskRandom37_32,
>-  kMaskRandom37_33,
>-  kMaskRandom37_34,
>-  kMaskRandom37_35,
>-  kMaskRandom37_36,
>-  kMaskRandom37_37
>-};
>-
>-const uint8_t* const kPacketMaskRandom38[38] = {
>-  kMaskRandom38_1,
>-  kMaskRandom38_2,
>-  kMaskRandom38_3,
>-  kMaskRandom38_4,
>-  kMaskRandom38_5,
>-  kMaskRandom38_6,
>-  kMaskRandom38_7,
>-  kMaskRandom38_8,
>-  kMaskRandom38_9,
>-  kMaskRandom38_10,
>-  kMaskRandom38_11,
>-  kMaskRandom38_12,
>-  kMaskRandom38_13,
>-  kMaskRandom38_14,
>-  kMaskRandom38_15,
>-  kMaskRandom38_16,
>-  kMaskRandom38_17,
>-  kMaskRandom38_18,
>-  kMaskRandom38_19,
>-  kMaskRandom38_20,
>-  kMaskRandom38_21,
>-  kMaskRandom38_22,
>-  kMaskRandom38_23,
>-  kMaskRandom38_24,
>-  kMaskRandom38_25,
>-  kMaskRandom38_26,
>-  kMaskRandom38_27,
>-  kMaskRandom38_28,
>-  kMaskRandom38_29,
>-  kMaskRandom38_30,
>-  kMaskRandom38_31,
>-  kMaskRandom38_32,
>-  kMaskRandom38_33,
>-  kMaskRandom38_34,
>-  kMaskRandom38_35,
>-  kMaskRandom38_36,
>-  kMaskRandom38_37,
>-  kMaskRandom38_38
>-};
>-
>-const uint8_t* const kPacketMaskRandom39[39] = {
>-  kMaskRandom39_1,
>-  kMaskRandom39_2,
>-  kMaskRandom39_3,
>-  kMaskRandom39_4,
>-  kMaskRandom39_5,
>-  kMaskRandom39_6,
>-  kMaskRandom39_7,
>-  kMaskRandom39_8,
>-  kMaskRandom39_9,
>-  kMaskRandom39_10,
>-  kMaskRandom39_11,
>-  kMaskRandom39_12,
>-  kMaskRandom39_13,
>-  kMaskRandom39_14,
>-  kMaskRandom39_15,
>-  kMaskRandom39_16,
>-  kMaskRandom39_17,
>-  kMaskRandom39_18,
>-  kMaskRandom39_19,
>-  kMaskRandom39_20,
>-  kMaskRandom39_21,
>-  kMaskRandom39_22,
>-  kMaskRandom39_23,
>-  kMaskRandom39_24,
>-  kMaskRandom39_25,
>-  kMaskRandom39_26,
>-  kMaskRandom39_27,
>-  kMaskRandom39_28,
>-  kMaskRandom39_29,
>-  kMaskRandom39_30,
>-  kMaskRandom39_31,
>-  kMaskRandom39_32,
>-  kMaskRandom39_33,
>-  kMaskRandom39_34,
>-  kMaskRandom39_35,
>-  kMaskRandom39_36,
>-  kMaskRandom39_37,
>-  kMaskRandom39_38,
>-  kMaskRandom39_39
>-};
>-
>-const uint8_t* const kPacketMaskRandom40[40] = {
>-  kMaskRandom40_1,
>-  kMaskRandom40_2,
>-  kMaskRandom40_3,
>-  kMaskRandom40_4,
>-  kMaskRandom40_5,
>-  kMaskRandom40_6,
>-  kMaskRandom40_7,
>-  kMaskRandom40_8,
>-  kMaskRandom40_9,
>-  kMaskRandom40_10,
>-  kMaskRandom40_11,
>-  kMaskRandom40_12,
>-  kMaskRandom40_13,
>-  kMaskRandom40_14,
>-  kMaskRandom40_15,
>-  kMaskRandom40_16,
>-  kMaskRandom40_17,
>-  kMaskRandom40_18,
>-  kMaskRandom40_19,
>-  kMaskRandom40_20,
>-  kMaskRandom40_21,
>-  kMaskRandom40_22,
>-  kMaskRandom40_23,
>-  kMaskRandom40_24,
>-  kMaskRandom40_25,
>-  kMaskRandom40_26,
>-  kMaskRandom40_27,
>-  kMaskRandom40_28,
>-  kMaskRandom40_29,
>-  kMaskRandom40_30,
>-  kMaskRandom40_31,
>-  kMaskRandom40_32,
>-  kMaskRandom40_33,
>-  kMaskRandom40_34,
>-  kMaskRandom40_35,
>-  kMaskRandom40_36,
>-  kMaskRandom40_37,
>-  kMaskRandom40_38,
>-  kMaskRandom40_39,
>-  kMaskRandom40_40
>-};
>-
>-const uint8_t* const kPacketMaskRandom41[41] = {
>-  kMaskRandom41_1,
>-  kMaskRandom41_2,
>-  kMaskRandom41_3,
>-  kMaskRandom41_4,
>-  kMaskRandom41_5,
>-  kMaskRandom41_6,
>-  kMaskRandom41_7,
>-  kMaskRandom41_8,
>-  kMaskRandom41_9,
>-  kMaskRandom41_10,
>-  kMaskRandom41_11,
>-  kMaskRandom41_12,
>-  kMaskRandom41_13,
>-  kMaskRandom41_14,
>-  kMaskRandom41_15,
>-  kMaskRandom41_16,
>-  kMaskRandom41_17,
>-  kMaskRandom41_18,
>-  kMaskRandom41_19,
>-  kMaskRandom41_20,
>-  kMaskRandom41_21,
>-  kMaskRandom41_22,
>-  kMaskRandom41_23,
>-  kMaskRandom41_24,
>-  kMaskRandom41_25,
>-  kMaskRandom41_26,
>-  kMaskRandom41_27,
>-  kMaskRandom41_28,
>-  kMaskRandom41_29,
>-  kMaskRandom41_30,
>-  kMaskRandom41_31,
>-  kMaskRandom41_32,
>-  kMaskRandom41_33,
>-  kMaskRandom41_34,
>-  kMaskRandom41_35,
>-  kMaskRandom41_36,
>-  kMaskRandom41_37,
>-  kMaskRandom41_38,
>-  kMaskRandom41_39,
>-  kMaskRandom41_40,
>-  kMaskRandom41_41
>-};
>-
>-const uint8_t* const kPacketMaskRandom42[42] = {
>-  kMaskRandom42_1,
>-  kMaskRandom42_2,
>-  kMaskRandom42_3,
>-  kMaskRandom42_4,
>-  kMaskRandom42_5,
>-  kMaskRandom42_6,
>-  kMaskRandom42_7,
>-  kMaskRandom42_8,
>-  kMaskRandom42_9,
>-  kMaskRandom42_10,
>-  kMaskRandom42_11,
>-  kMaskRandom42_12,
>-  kMaskRandom42_13,
>-  kMaskRandom42_14,
>-  kMaskRandom42_15,
>-  kMaskRandom42_16,
>-  kMaskRandom42_17,
>-  kMaskRandom42_18,
>-  kMaskRandom42_19,
>-  kMaskRandom42_20,
>-  kMaskRandom42_21,
>-  kMaskRandom42_22,
>-  kMaskRandom42_23,
>-  kMaskRandom42_24,
>-  kMaskRandom42_25,
>-  kMaskRandom42_26,
>-  kMaskRandom42_27,
>-  kMaskRandom42_28,
>-  kMaskRandom42_29,
>-  kMaskRandom42_30,
>-  kMaskRandom42_31,
>-  kMaskRandom42_32,
>-  kMaskRandom42_33,
>-  kMaskRandom42_34,
>-  kMaskRandom42_35,
>-  kMaskRandom42_36,
>-  kMaskRandom42_37,
>-  kMaskRandom42_38,
>-  kMaskRandom42_39,
>-  kMaskRandom42_40,
>-  kMaskRandom42_41,
>-  kMaskRandom42_42
>-};
>-
>-const uint8_t* const kPacketMaskRandom43[43] = {
>-  kMaskRandom43_1,
>-  kMaskRandom43_2,
>-  kMaskRandom43_3,
>-  kMaskRandom43_4,
>-  kMaskRandom43_5,
>-  kMaskRandom43_6,
>-  kMaskRandom43_7,
>-  kMaskRandom43_8,
>-  kMaskRandom43_9,
>-  kMaskRandom43_10,
>-  kMaskRandom43_11,
>-  kMaskRandom43_12,
>-  kMaskRandom43_13,
>-  kMaskRandom43_14,
>-  kMaskRandom43_15,
>-  kMaskRandom43_16,
>-  kMaskRandom43_17,
>-  kMaskRandom43_18,
>-  kMaskRandom43_19,
>-  kMaskRandom43_20,
>-  kMaskRandom43_21,
>-  kMaskRandom43_22,
>-  kMaskRandom43_23,
>-  kMaskRandom43_24,
>-  kMaskRandom43_25,
>-  kMaskRandom43_26,
>-  kMaskRandom43_27,
>-  kMaskRandom43_28,
>-  kMaskRandom43_29,
>-  kMaskRandom43_30,
>-  kMaskRandom43_31,
>-  kMaskRandom43_32,
>-  kMaskRandom43_33,
>-  kMaskRandom43_34,
>-  kMaskRandom43_35,
>-  kMaskRandom43_36,
>-  kMaskRandom43_37,
>-  kMaskRandom43_38,
>-  kMaskRandom43_39,
>-  kMaskRandom43_40,
>-  kMaskRandom43_41,
>-  kMaskRandom43_42,
>-  kMaskRandom43_43
>-};
>-
>-const uint8_t* const kPacketMaskRandom44[44] = {
>-  kMaskRandom44_1,
>-  kMaskRandom44_2,
>-  kMaskRandom44_3,
>-  kMaskRandom44_4,
>-  kMaskRandom44_5,
>-  kMaskRandom44_6,
>-  kMaskRandom44_7,
>-  kMaskRandom44_8,
>-  kMaskRandom44_9,
>-  kMaskRandom44_10,
>-  kMaskRandom44_11,
>-  kMaskRandom44_12,
>-  kMaskRandom44_13,
>-  kMaskRandom44_14,
>-  kMaskRandom44_15,
>-  kMaskRandom44_16,
>-  kMaskRandom44_17,
>-  kMaskRandom44_18,
>-  kMaskRandom44_19,
>-  kMaskRandom44_20,
>-  kMaskRandom44_21,
>-  kMaskRandom44_22,
>-  kMaskRandom44_23,
>-  kMaskRandom44_24,
>-  kMaskRandom44_25,
>-  kMaskRandom44_26,
>-  kMaskRandom44_27,
>-  kMaskRandom44_28,
>-  kMaskRandom44_29,
>-  kMaskRandom44_30,
>-  kMaskRandom44_31,
>-  kMaskRandom44_32,
>-  kMaskRandom44_33,
>-  kMaskRandom44_34,
>-  kMaskRandom44_35,
>-  kMaskRandom44_36,
>-  kMaskRandom44_37,
>-  kMaskRandom44_38,
>-  kMaskRandom44_39,
>-  kMaskRandom44_40,
>-  kMaskRandom44_41,
>-  kMaskRandom44_42,
>-  kMaskRandom44_43,
>-  kMaskRandom44_44
>-};
>-
>-const uint8_t* const kPacketMaskRandom45[45] = {
>-  kMaskRandom45_1,
>-  kMaskRandom45_2,
>-  kMaskRandom45_3,
>-  kMaskRandom45_4,
>-  kMaskRandom45_5,
>-  kMaskRandom45_6,
>-  kMaskRandom45_7,
>-  kMaskRandom45_8,
>-  kMaskRandom45_9,
>-  kMaskRandom45_10,
>-  kMaskRandom45_11,
>-  kMaskRandom45_12,
>-  kMaskRandom45_13,
>-  kMaskRandom45_14,
>-  kMaskRandom45_15,
>-  kMaskRandom45_16,
>-  kMaskRandom45_17,
>-  kMaskRandom45_18,
>-  kMaskRandom45_19,
>-  kMaskRandom45_20,
>-  kMaskRandom45_21,
>-  kMaskRandom45_22,
>-  kMaskRandom45_23,
>-  kMaskRandom45_24,
>-  kMaskRandom45_25,
>-  kMaskRandom45_26,
>-  kMaskRandom45_27,
>-  kMaskRandom45_28,
>-  kMaskRandom45_29,
>-  kMaskRandom45_30,
>-  kMaskRandom45_31,
>-  kMaskRandom45_32,
>-  kMaskRandom45_33,
>-  kMaskRandom45_34,
>-  kMaskRandom45_35,
>-  kMaskRandom45_36,
>-  kMaskRandom45_37,
>-  kMaskRandom45_38,
>-  kMaskRandom45_39,
>-  kMaskRandom45_40,
>-  kMaskRandom45_41,
>-  kMaskRandom45_42,
>-  kMaskRandom45_43,
>-  kMaskRandom45_44,
>-  kMaskRandom45_45
>-};
>-
>-const uint8_t* const kPacketMaskRandom46[46] = {
>-  kMaskRandom46_1,
>-  kMaskRandom46_2,
>-  kMaskRandom46_3,
>-  kMaskRandom46_4,
>-  kMaskRandom46_5,
>-  kMaskRandom46_6,
>-  kMaskRandom46_7,
>-  kMaskRandom46_8,
>-  kMaskRandom46_9,
>-  kMaskRandom46_10,
>-  kMaskRandom46_11,
>-  kMaskRandom46_12,
>-  kMaskRandom46_13,
>-  kMaskRandom46_14,
>-  kMaskRandom46_15,
>-  kMaskRandom46_16,
>-  kMaskRandom46_17,
>-  kMaskRandom46_18,
>-  kMaskRandom46_19,
>-  kMaskRandom46_20,
>-  kMaskRandom46_21,
>-  kMaskRandom46_22,
>-  kMaskRandom46_23,
>-  kMaskRandom46_24,
>-  kMaskRandom46_25,
>-  kMaskRandom46_26,
>-  kMaskRandom46_27,
>-  kMaskRandom46_28,
>-  kMaskRandom46_29,
>-  kMaskRandom46_30,
>-  kMaskRandom46_31,
>-  kMaskRandom46_32,
>-  kMaskRandom46_33,
>-  kMaskRandom46_34,
>-  kMaskRandom46_35,
>-  kMaskRandom46_36,
>-  kMaskRandom46_37,
>-  kMaskRandom46_38,
>-  kMaskRandom46_39,
>-  kMaskRandom46_40,
>-  kMaskRandom46_41,
>-  kMaskRandom46_42,
>-  kMaskRandom46_43,
>-  kMaskRandom46_44,
>-  kMaskRandom46_45,
>-  kMaskRandom46_46
>-};
>-
>-const uint8_t* const kPacketMaskRandom47[47] = {
>-  kMaskRandom47_1,
>-  kMaskRandom47_2,
>-  kMaskRandom47_3,
>-  kMaskRandom47_4,
>-  kMaskRandom47_5,
>-  kMaskRandom47_6,
>-  kMaskRandom47_7,
>-  kMaskRandom47_8,
>-  kMaskRandom47_9,
>-  kMaskRandom47_10,
>-  kMaskRandom47_11,
>-  kMaskRandom47_12,
>-  kMaskRandom47_13,
>-  kMaskRandom47_14,
>-  kMaskRandom47_15,
>-  kMaskRandom47_16,
>-  kMaskRandom47_17,
>-  kMaskRandom47_18,
>-  kMaskRandom47_19,
>-  kMaskRandom47_20,
>-  kMaskRandom47_21,
>-  kMaskRandom47_22,
>-  kMaskRandom47_23,
>-  kMaskRandom47_24,
>-  kMaskRandom47_25,
>-  kMaskRandom47_26,
>-  kMaskRandom47_27,
>-  kMaskRandom47_28,
>-  kMaskRandom47_29,
>-  kMaskRandom47_30,
>-  kMaskRandom47_31,
>-  kMaskRandom47_32,
>-  kMaskRandom47_33,
>-  kMaskRandom47_34,
>-  kMaskRandom47_35,
>-  kMaskRandom47_36,
>-  kMaskRandom47_37,
>-  kMaskRandom47_38,
>-  kMaskRandom47_39,
>-  kMaskRandom47_40,
>-  kMaskRandom47_41,
>-  kMaskRandom47_42,
>-  kMaskRandom47_43,
>-  kMaskRandom47_44,
>-  kMaskRandom47_45,
>-  kMaskRandom47_46,
>-  kMaskRandom47_47
>-};
>-
>-const uint8_t* const kPacketMaskRandom48[48] = {
>-  kMaskRandom48_1,
>-  kMaskRandom48_2,
>-  kMaskRandom48_3,
>-  kMaskRandom48_4,
>-  kMaskRandom48_5,
>-  kMaskRandom48_6,
>-  kMaskRandom48_7,
>-  kMaskRandom48_8,
>-  kMaskRandom48_9,
>-  kMaskRandom48_10,
>-  kMaskRandom48_11,
>-  kMaskRandom48_12,
>-  kMaskRandom48_13,
>-  kMaskRandom48_14,
>-  kMaskRandom48_15,
>-  kMaskRandom48_16,
>-  kMaskRandom48_17,
>-  kMaskRandom48_18,
>-  kMaskRandom48_19,
>-  kMaskRandom48_20,
>-  kMaskRandom48_21,
>-  kMaskRandom48_22,
>-  kMaskRandom48_23,
>-  kMaskRandom48_24,
>-  kMaskRandom48_25,
>-  kMaskRandom48_26,
>-  kMaskRandom48_27,
>-  kMaskRandom48_28,
>-  kMaskRandom48_29,
>-  kMaskRandom48_30,
>-  kMaskRandom48_31,
>-  kMaskRandom48_32,
>-  kMaskRandom48_33,
>-  kMaskRandom48_34,
>-  kMaskRandom48_35,
>-  kMaskRandom48_36,
>-  kMaskRandom48_37,
>-  kMaskRandom48_38,
>-  kMaskRandom48_39,
>-  kMaskRandom48_40,
>-  kMaskRandom48_41,
>-  kMaskRandom48_42,
>-  kMaskRandom48_43,
>-  kMaskRandom48_44,
>-  kMaskRandom48_45,
>-  kMaskRandom48_46,
>-  kMaskRandom48_47,
>-  kMaskRandom48_48
>-};
>-
>-const uint8_t* const* const kPacketMaskRandomTbl[48] = {
>-  kPacketMaskRandom1,
>-  kPacketMaskRandom2,
>-  kPacketMaskRandom3,
>-  kPacketMaskRandom4,
>-  kPacketMaskRandom5,
>-  kPacketMaskRandom6,
>-  kPacketMaskRandom7,
>-  kPacketMaskRandom8,
>-  kPacketMaskRandom9,
>-  kPacketMaskRandom10,
>-  kPacketMaskRandom11,
>-  kPacketMaskRandom12,
>-  kPacketMaskRandom13,
>-  kPacketMaskRandom14,
>-  kPacketMaskRandom15,
>-  kPacketMaskRandom16,
>-  kPacketMaskRandom17,
>-  kPacketMaskRandom18,
>-  kPacketMaskRandom19,
>-  kPacketMaskRandom20,
>-  kPacketMaskRandom21,
>-  kPacketMaskRandom22,
>-  kPacketMaskRandom23,
>-  kPacketMaskRandom24,
>-  kPacketMaskRandom25,
>-  kPacketMaskRandom26,
>-  kPacketMaskRandom27,
>-  kPacketMaskRandom28,
>-  kPacketMaskRandom29,
>-  kPacketMaskRandom30,
>-  kPacketMaskRandom31,
>-  kPacketMaskRandom32,
>-  kPacketMaskRandom33,
>-  kPacketMaskRandom34,
>-  kPacketMaskRandom35,
>-  kPacketMaskRandom36,
>-  kPacketMaskRandom37,
>-  kPacketMaskRandom38,
>-  kPacketMaskRandom39,
>-  kPacketMaskRandom40,
>-  kPacketMaskRandom41,
>-  kPacketMaskRandom42,
>-  kPacketMaskRandom43,
>-  kPacketMaskRandom44,
>-  kPacketMaskRandom45,
>-  kPacketMaskRandom46,
>-  kPacketMaskRandom47,
>-  kPacketMaskRandom48
>-};
>+extern const uint8_t kPacketMaskRandomTbl[];
> 
> }  // namespace fec_private_tables
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/fec_test_helper.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/fec_test_helper.cc
>index d73af5a31c8..b325d6750bd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/fec_test_helper.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/fec_test_helper.cc
>@@ -30,6 +30,17 @@ constexpr uint8_t kVp8PayloadType = 120;
> constexpr int kPacketTimestampIncrement = 3000;
> }  // namespace
> 
>+MediaPacketGenerator::MediaPacketGenerator(uint32_t min_packet_size,
>+                                           uint32_t max_packet_size,
>+                                           uint32_t ssrc,
>+                                           Random* random)
>+    : min_packet_size_(min_packet_size),
>+      max_packet_size_(max_packet_size),
>+      ssrc_(ssrc),
>+      random_(random) {}
>+
>+MediaPacketGenerator::~MediaPacketGenerator() = default;
>+
> ForwardErrorCorrection::PacketList MediaPacketGenerator::ConstructMediaPackets(
>     int num_media_packets,
>     uint16_t start_seq_num) {
>@@ -112,7 +123,6 @@ std::unique_ptr<AugmentedPacket> AugmentedPacketGenerator::NextPacket(
>   for (size_t i = 0; i < length; ++i)
>     packet->data[i + kRtpHeaderSize] = offset + i;
>   packet->length = length + kRtpHeaderSize;
>-  memset(&packet->header, 0, sizeof(WebRtcRTPHeader));
>   packet->header.frameType = kVideoFrameDelta;
>   packet->header.header.headerLength = kRtpHeaderSize;
>   packet->header.header.markerBit = (num_packets_ == 1);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/fec_test_helper.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/fec_test_helper.h
>index 712b20143b6..004edf77d46 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/fec_test_helper.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/fec_test_helper.h
>@@ -15,7 +15,6 @@
> 
> #include "modules/include/module_common_types.h"
> #include "modules/rtp_rtcp/source/forward_error_correction.h"
>-#include "rtc_base/basictypes.h"
> #include "rtc_base/random.h"
> 
> namespace webrtc {
>@@ -36,11 +35,8 @@ class MediaPacketGenerator {
>   MediaPacketGenerator(uint32_t min_packet_size,
>                        uint32_t max_packet_size,
>                        uint32_t ssrc,
>-                       Random* random)
>-      : min_packet_size_(min_packet_size),
>-        max_packet_size_(max_packet_size),
>-        ssrc_(ssrc),
>-        random_(random) {}
>+                       Random* random);
>+  ~MediaPacketGenerator();
> 
>   // Construct the media packets, up to |num_media_packets| packets.
>   ForwardErrorCorrection::PacketList ConstructMediaPackets(
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/flexfec_header_reader_writer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/flexfec_header_reader_writer.h
>index 3bb73f97bc9..1d6ddda4b40 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/flexfec_header_reader_writer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/flexfec_header_reader_writer.h
>@@ -12,7 +12,6 @@
> #define MODULES_RTP_RTCP_SOURCE_FLEXFEC_HEADER_READER_WRITER_H_
> 
> #include "modules/rtp_rtcp/source/forward_error_correction.h"
>-#include "rtc_base/basictypes.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/flexfec_header_reader_writer_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/flexfec_header_reader_writer_unittest.cc
>index 0f0bf274fa4..5da378cb015 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/flexfec_header_reader_writer_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/flexfec_header_reader_writer_unittest.cc
>@@ -17,7 +17,6 @@
> #include "modules/rtp_rtcp/source/flexfec_header_reader_writer.h"
> #include "modules/rtp_rtcp/source/forward_error_correction.h"
> #include "modules/rtp_rtcp/source/forward_error_correction_internal.h"
>-#include "rtc_base/basictypes.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/random.h"
> #include "rtc_base/scoped_ref_ptr.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/flexfec_receiver_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/flexfec_receiver_unittest.cc
>index af9a9983b97..378cf7d6dff 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/flexfec_receiver_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/flexfec_receiver_unittest.cc
>@@ -16,7 +16,6 @@
> #include "modules/rtp_rtcp/source/fec_test_helper.h"
> #include "modules/rtp_rtcp/source/forward_error_correction.h"
> #include "modules/rtp_rtcp/source/rtp_packet_received.h"
>-#include "rtc_base/basictypes.h"
> #include "test/gmock.h"
> #include "test/gtest.h"
> 
>@@ -211,7 +210,7 @@ TEST_F(FlexfecReceiverTest, ReceivesMultiplePackets) {
>   }
> 
>   // Receive FEC packet.
>-  auto fec_packet = fec_packets.front();
>+  auto* fec_packet = fec_packets.front();
>   std::unique_ptr<Packet> packet_with_rtp_header =
>       packet_generator_.BuildFlexfecPacket(*fec_packet);
>   std::unique_ptr<ForwardErrorCorrection::ReceivedPacket> received_packet =
>@@ -354,7 +353,7 @@ TEST_F(FlexfecReceiverTest, RecoversFrom50PercentLoss) {
> 
>   // Receive all FEC packets.
>   media_it = media_packets.begin();
>-  for (const auto& fec_packet : fec_packets) {
>+  for (const auto* fec_packet : fec_packets) {
>     std::unique_ptr<Packet> fec_packet_with_rtp_header =
>         packet_generator_.BuildFlexfecPacket(*fec_packet);
>     ++media_it;
>@@ -453,7 +452,7 @@ TEST_F(FlexfecReceiverTest, SurvivesOldRecoveredPacketBeingReinserted) {
>     void SetReceiver(FlexfecReceiver* receiver) { receiver_ = receiver; }
> 
>     // Implements RecoveredPacketReceiver.
>-    void OnRecoveredPacket(const uint8_t* packet, size_t length) {
>+    void OnRecoveredPacket(const uint8_t* packet, size_t length) override {
>       RtpPacketReceived parsed_packet;
>       EXPECT_TRUE(parsed_packet.Parse(packet, length));
>       parsed_packet.set_recovered(true);
>@@ -567,7 +566,7 @@ TEST_F(FlexfecReceiverTest, RecoveryCallbackDoesNotLoopInfinitely) {
>     bool DeepRecursion() const { return deep_recursion_; }
> 
>     // Implements RecoveredPacketReceiver.
>-    void OnRecoveredPacket(const uint8_t* packet, size_t length) {
>+    void OnRecoveredPacket(const uint8_t* packet, size_t length) override {
>       RtpPacketReceived parsed_packet;
>       EXPECT_TRUE(parsed_packet.Parse(packet, length));
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/flexfec_sender.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/flexfec_sender.cc
>index f046a349eea..9704096ff7d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/flexfec_sender.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/flexfec_sender.cc
>@@ -38,7 +38,7 @@ const int kMsToRtpTimestamp = kVideoPayloadTypeFrequency / 1000;
> // How often to log the generated FEC packets to the text log.
> constexpr int64_t kPacketLogIntervalMs = 10000;
> 
>-RtpHeaderExtensionMap RegisterBweExtensions(
>+RtpHeaderExtensionMap RegisterSupportedExtensions(
>     const std::vector<RtpExtension>& rtp_header_extensions) {
>   RtpHeaderExtensionMap map;
>   for (const auto& extension : rtp_header_extensions) {
>@@ -48,10 +48,12 @@ RtpHeaderExtensionMap RegisterBweExtensions(
>       map.Register<AbsoluteSendTime>(extension.id);
>     } else if (extension.uri == TransmissionOffset::kUri) {
>       map.Register<TransmissionOffset>(extension.id);
>+    } else if (extension.uri == RtpMid::kUri) {
>+      map.Register<RtpMid>(extension.id);
>     } else {
>       RTC_LOG(LS_INFO)
>           << "FlexfecSender only supports RTP header extensions for "
>-          << "BWE, so the extension " << extension.ToString()
>+          << "BWE and MID, so the extension " << extension.ToString()
>           << " will not be used.";
>     }
>   }
>@@ -64,6 +66,7 @@ FlexfecSender::FlexfecSender(
>     int payload_type,
>     uint32_t ssrc,
>     uint32_t protected_media_ssrc,
>+    const std::string& mid,
>     const std::vector<RtpExtension>& rtp_header_extensions,
>     rtc::ArrayView<const RtpExtensionSize> extension_sizes,
>     const RtpState* rtp_state,
>@@ -79,11 +82,13 @@ FlexfecSender::FlexfecSender(
>                                   : random_.Rand<uint32_t>()),
>       ssrc_(ssrc),
>       protected_media_ssrc_(protected_media_ssrc),
>+      mid_(mid),
>       seq_num_(rtp_state ? rtp_state->sequence_number
>                          : random_.Rand(1, kMaxInitRtpSeqNumber)),
>       ulpfec_generator_(
>           ForwardErrorCorrection::CreateFlexfec(ssrc, protected_media_ssrc)),
>-      rtp_header_extension_map_(RegisterBweExtensions(rtp_header_extensions)),
>+      rtp_header_extension_map_(
>+          RegisterSupportedExtensions(rtp_header_extensions)),
>       header_extensions_size_(
>           rtp_header_extension_map_.GetTotalLengthInBytes(extension_sizes)) {
>   // This object should not have been instantiated if FlexFEC is disabled.
>@@ -114,7 +119,7 @@ bool FlexfecSender::FecAvailable() const {
> std::vector<std::unique_ptr<RtpPacketToSend>> FlexfecSender::GetFecPackets() {
>   std::vector<std::unique_ptr<RtpPacketToSend>> fec_packets_to_send;
>   fec_packets_to_send.reserve(ulpfec_generator_.generated_fec_packets_.size());
>-  for (const auto& fec_packet : ulpfec_generator_.generated_fec_packets_) {
>+  for (const auto* fec_packet : ulpfec_generator_.generated_fec_packets_) {
>     std::unique_ptr<RtpPacketToSend> fec_packet_to_send(
>         new RtpPacketToSend(&rtp_header_extension_map_));
> 
>@@ -134,6 +139,11 @@ std::vector<std::unique_ptr<RtpPacketToSend>> FlexfecSender::GetFecPackets() {
>     fec_packet_to_send->ReserveExtension<AbsoluteSendTime>();
>     fec_packet_to_send->ReserveExtension<TransmissionOffset>();
>     fec_packet_to_send->ReserveExtension<TransportSequenceNumber>();
>+    // Possibly include the MID header extension.
>+    if (!mid_.empty()) {
>+      // This is a no-op if the MID header extension is not registered.
>+      fec_packet_to_send->SetExtension<RtpMid>(mid_);
>+    }
> 
>     // RTP payload.
>     uint8_t* payload = fec_packet_to_send->AllocatePayload(fec_packet->length);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/flexfec_sender_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/flexfec_sender_unittest.cc
>index aa729ed0b87..ddc2805209d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/flexfec_sender_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/flexfec_sender_unittest.cc
>@@ -32,6 +32,7 @@ using test::fec::AugmentedPacketGenerator;
> constexpr int kFlexfecPayloadType = 123;
> constexpr uint32_t kMediaSsrc = 1234;
> constexpr uint32_t kFlexfecSsrc = 5678;
>+const char kNoMid[] = "";
> const std::vector<RtpExtension> kNoRtpHeaderExtensions;
> const std::vector<RtpExtensionSize> kNoRtpHeaderExtensionSizes;
> // Assume a single protected media SSRC.
>@@ -76,7 +77,7 @@ std::unique_ptr<RtpPacketToSend> GenerateSingleFlexfecPacket(
> 
> TEST(FlexfecSenderTest, Ssrc) {
>   SimulatedClock clock(kInitialSimulatedClockTime);
>-  FlexfecSender sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc,
>+  FlexfecSender sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc, kNoMid,
>                        kNoRtpHeaderExtensions, kNoRtpHeaderExtensionSizes,
>                        nullptr /* rtp_state */, &clock);
> 
>@@ -85,7 +86,7 @@ TEST(FlexfecSenderTest, Ssrc) {
> 
> TEST(FlexfecSenderTest, NoFecAvailableBeforeMediaAdded) {
>   SimulatedClock clock(kInitialSimulatedClockTime);
>-  FlexfecSender sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc,
>+  FlexfecSender sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc, kNoMid,
>                        kNoRtpHeaderExtensions, kNoRtpHeaderExtensionSizes,
>                        nullptr /* rtp_state */, &clock);
> 
>@@ -96,7 +97,7 @@ TEST(FlexfecSenderTest, NoFecAvailableBeforeMediaAdded) {
> 
> TEST(FlexfecSenderTest, ProtectOneFrameWithOneFecPacket) {
>   SimulatedClock clock(kInitialSimulatedClockTime);
>-  FlexfecSender sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc,
>+  FlexfecSender sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc, kNoMid,
>                        kNoRtpHeaderExtensions, kNoRtpHeaderExtensionSizes,
>                        nullptr /* rtp_state */, &clock);
>   auto fec_packet = GenerateSingleFlexfecPacket(&sender);
>@@ -119,7 +120,7 @@ TEST(FlexfecSenderTest, ProtectTwoFramesWithOneFecPacket) {
>   constexpr size_t kNumFrames = 2;
>   constexpr size_t kNumPacketsPerFrame = 2;
>   SimulatedClock clock(kInitialSimulatedClockTime);
>-  FlexfecSender sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc,
>+  FlexfecSender sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc, kNoMid,
>                        kNoRtpHeaderExtensions, kNoRtpHeaderExtensionSizes,
>                        nullptr /* rtp_state */, &clock);
>   sender.SetFecParameters(params);
>@@ -159,7 +160,7 @@ TEST(FlexfecSenderTest, ProtectTwoFramesWithTwoFecPackets) {
>   constexpr size_t kNumFrames = 2;
>   constexpr size_t kNumPacketsPerFrame = 2;
>   SimulatedClock clock(kInitialSimulatedClockTime);
>-  FlexfecSender sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc,
>+  FlexfecSender sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc, kNoMid,
>                        kNoRtpHeaderExtensions, kNoRtpHeaderExtensionSizes,
>                        nullptr /* rtp_state */, &clock);
>   sender.SetFecParameters(params);
>@@ -195,7 +196,7 @@ TEST(FlexfecSenderTest, ProtectTwoFramesWithTwoFecPackets) {
> TEST(FlexfecSenderTest, NoRtpHeaderExtensionsForBweByDefault) {
>   const std::vector<RtpExtension> kRtpHeaderExtensions{};
>   SimulatedClock clock(kInitialSimulatedClockTime);
>-  FlexfecSender sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc,
>+  FlexfecSender sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc, kNoMid,
>                        kRtpHeaderExtensions, kNoRtpHeaderExtensionSizes,
>                        nullptr /* rtp_state */, &clock);
>   auto fec_packet = GenerateSingleFlexfecPacket(&sender);
>@@ -209,7 +210,7 @@ TEST(FlexfecSenderTest, RegisterAbsoluteSendTimeRtpHeaderExtension) {
>   const std::vector<RtpExtension> kRtpHeaderExtensions{
>       {RtpExtension::kAbsSendTimeUri, 1}};
>   SimulatedClock clock(kInitialSimulatedClockTime);
>-  FlexfecSender sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc,
>+  FlexfecSender sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc, kNoMid,
>                        kRtpHeaderExtensions, kNoRtpHeaderExtensionSizes,
>                        nullptr /* rtp_state */, &clock);
>   auto fec_packet = GenerateSingleFlexfecPacket(&sender);
>@@ -223,7 +224,7 @@ TEST(FlexfecSenderTest, RegisterTransmissionOffsetRtpHeaderExtension) {
>   const std::vector<RtpExtension> kRtpHeaderExtensions{
>       {RtpExtension::kTimestampOffsetUri, 1}};
>   SimulatedClock clock(kInitialSimulatedClockTime);
>-  FlexfecSender sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc,
>+  FlexfecSender sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc, kNoMid,
>                        kRtpHeaderExtensions, kNoRtpHeaderExtensionSizes,
>                        nullptr /* rtp_state */, &clock);
>   auto fec_packet = GenerateSingleFlexfecPacket(&sender);
>@@ -237,7 +238,7 @@ TEST(FlexfecSenderTest, RegisterTransportSequenceNumberRtpHeaderExtension) {
>   const std::vector<RtpExtension> kRtpHeaderExtensions{
>       {RtpExtension::kTransportSequenceNumberUri, 1}};
>   SimulatedClock clock(kInitialSimulatedClockTime);
>-  FlexfecSender sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc,
>+  FlexfecSender sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc, kNoMid,
>                        kRtpHeaderExtensions, kNoRtpHeaderExtensionSizes,
>                        nullptr /* rtp_state */, &clock);
>   auto fec_packet = GenerateSingleFlexfecPacket(&sender);
>@@ -253,7 +254,7 @@ TEST(FlexfecSenderTest, RegisterAllRtpHeaderExtensionsForBwe) {
>       {RtpExtension::kTimestampOffsetUri, 2},
>       {RtpExtension::kTransportSequenceNumberUri, 3}};
>   SimulatedClock clock(kInitialSimulatedClockTime);
>-  FlexfecSender sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc,
>+  FlexfecSender sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc, kNoMid,
>                        kRtpHeaderExtensions, kNoRtpHeaderExtensionSizes,
>                        nullptr /* rtp_state */, &clock);
>   auto fec_packet = GenerateSingleFlexfecPacket(&sender);
>@@ -265,7 +266,7 @@ TEST(FlexfecSenderTest, RegisterAllRtpHeaderExtensionsForBwe) {
> 
> TEST(FlexfecSenderTest, MaxPacketOverhead) {
>   SimulatedClock clock(kInitialSimulatedClockTime);
>-  FlexfecSender sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc,
>+  FlexfecSender sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc, kNoMid,
>                        kNoRtpHeaderExtensions, kNoRtpHeaderExtensionSizes,
>                        nullptr /* rtp_state */, &clock);
> 
>@@ -280,12 +281,12 @@ TEST(FlexfecSenderTest, MaxPacketOverheadWithExtensions) {
>   SimulatedClock clock(kInitialSimulatedClockTime);
>   const size_t kExtensionHeaderLength = 1;
>   const size_t kRtpOneByteHeaderLength = 4;
>-  const size_t kExtensionsTotalSize = Word32Align(
>-      kRtpOneByteHeaderLength +
>-      kExtensionHeaderLength + AbsoluteSendTime::kValueSizeBytes +
>-      kExtensionHeaderLength + TransmissionOffset::kValueSizeBytes +
>-      kExtensionHeaderLength + TransportSequenceNumber::kValueSizeBytes);
>-  FlexfecSender sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc,
>+  const size_t kExtensionsTotalSize =
>+      Word32Align(kRtpOneByteHeaderLength + kExtensionHeaderLength +
>+                  AbsoluteSendTime::kValueSizeBytes + kExtensionHeaderLength +
>+                  TransmissionOffset::kValueSizeBytes + kExtensionHeaderLength +
>+                  TransportSequenceNumber::kValueSizeBytes);
>+  FlexfecSender sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc, kNoMid,
>                        kRtpHeaderExtensions, RTPSender::FecExtensionSizes(),
>                        nullptr /* rtp_state */, &clock);
> 
>@@ -293,12 +294,28 @@ TEST(FlexfecSenderTest, MaxPacketOverheadWithExtensions) {
>             sender.MaxPacketOverhead());
> }
> 
>+TEST(FlexfecSenderTest, MidIncludedInPacketsWhenSet) {
>+  const std::vector<RtpExtension> kRtpHeaderExtensions{
>+      {RtpExtension::kMidUri, 1}};
>+  const char kMid[] = "mid";
>+  SimulatedClock clock(kInitialSimulatedClockTime);
>+  FlexfecSender sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc, kMid,
>+                       kRtpHeaderExtensions, RTPSender::FecExtensionSizes(),
>+                       nullptr /* rtp_state */, &clock);
>+
>+  auto fec_packet = GenerateSingleFlexfecPacket(&sender);
>+
>+  std::string mid;
>+  ASSERT_TRUE(fec_packet->GetExtension<RtpMid>(&mid));
>+  EXPECT_EQ(kMid, mid);
>+}
>+
> TEST(FlexfecSenderTest, SetsAndGetsRtpState) {
>   RtpState initial_rtp_state;
>   initial_rtp_state.sequence_number = 100;
>   initial_rtp_state.start_timestamp = 200;
>   SimulatedClock clock(kInitialSimulatedClockTime);
>-  FlexfecSender sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc,
>+  FlexfecSender sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc, kNoMid,
>                        kNoRtpHeaderExtensions, kNoRtpHeaderExtensionSizes,
>                        &initial_rtp_state, &clock);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc
>index 33829c4fb22..b743110b2d6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/forward_error_correction.cc
>@@ -52,7 +52,7 @@ int32_t ForwardErrorCorrection::Packet::Release() {
> // the std::unique_ptr's are not covariant w.r.t. the types that
> // they are pointing to.
> template <typename S, typename T>
>-bool ForwardErrorCorrection::SortablePacket::LessThan::operator() (
>+bool ForwardErrorCorrection::SortablePacket::LessThan::operator()(
>     const S& first,
>     const T& second) {
>   RTC_DCHECK_EQ(first->ssrc, second->ssrc);
>@@ -155,12 +155,12 @@ int ForwardErrorCorrection::EncodeFec(const PacketList& media_packets,
>     fec_packets->push_back(&generated_fec_packets_[i]);
>   }
> 
>-  const internal::PacketMaskTable mask_table(fec_mask_type, num_media_packets);
>+  internal::PacketMaskTable mask_table(fec_mask_type, num_media_packets);
>   packet_mask_size_ = internal::PacketMaskSize(num_media_packets);
>   memset(packet_masks_, 0, num_fec_packets * packet_mask_size_);
>   internal::GeneratePacketMasks(num_media_packets, num_fec_packets,
>                                 num_important_packets, use_unequal_protection,
>-                                mask_table, packet_masks_);
>+                                &mask_table, packet_masks_);
> 
>   // Adapt packet masks to missing media packets.
>   int num_mask_bits = InsertZerosInPacketMasks(media_packets, num_fec_packets);
>@@ -655,7 +655,7 @@ void ForwardErrorCorrection::AttemptRecovery(
>         continue;
>       }
> 
>-      auto recovered_packet_ptr = recovered_packet.get();
>+      auto* recovered_packet_ptr = recovered_packet.get();
>       // Add recovered packet to the list of recovered packets and update any
>       // FEC packets covering this packet with a pointer to the data.
>       // TODO(holmer): Consider replacing this with a binary search for the
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/forward_error_correction.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/forward_error_correction.h
>index 97acb393f35..819f6bcec56 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/forward_error_correction.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/forward_error_correction.h
>@@ -19,7 +19,6 @@
> 
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> #include "modules/rtp_rtcp/source/forward_error_correction_internal.h"
>-#include "rtc_base/basictypes.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/refcount.h"
> #include "rtc_base/scoped_ref_ptr.h"
>@@ -68,7 +67,7 @@ class ForwardErrorCorrection {
>     // packets belonging to the same SSRC.
>     struct LessThan {
>       template <typename S, typename T>
>-      bool operator() (const S& first, const T& second);
>+      bool operator()(const S& first, const T& second);
>     };
> 
>     uint32_t ssrc;
>@@ -84,8 +83,8 @@ class ForwardErrorCorrection {
>     ReceivedPacket();
>     ~ReceivedPacket();
> 
>-    bool is_fec;    // Set to true if this is an FEC packet and false
>-                    // otherwise.
>+    bool is_fec;  // Set to true if this is an FEC packet and false
>+                  // otherwise.
>     rtc::scoped_refptr<Packet> pkt;  // Pointer to the packet storage.
>   };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/forward_error_correction_internal.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/forward_error_correction_internal.cc
>index 60b403bb781..7e5fd91e035 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/forward_error_correction_internal.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/forward_error_correction_internal.cc
>@@ -10,9 +10,6 @@
> 
> #include "modules/rtp_rtcp/source/forward_error_correction_internal.h"
> 
>-#include <assert.h>
>-#include <string.h>
>-
> #include <algorithm>
> 
> #include "modules/rtp_rtcp/source/fec_private_tables_bursty.h"
>@@ -20,9 +17,6 @@
> #include "rtc_base/checks.h"
> 
> namespace {
>-using webrtc::fec_private_tables::kPacketMaskBurstyTbl;
>-using webrtc::fec_private_tables::kPacketMaskRandomTbl;
>-
> // Allow for different modes of protection for packets in UEP case.
> enum ProtectionMode {
>   kModeNoOverlap,
>@@ -140,6 +134,7 @@ void ShiftFitSubMask(int num_mask_bytes,
>     packet_mask[pkt_mask_idx] = shift_right_curr_byte;
>   }
> }
>+
> }  // namespace
> 
> namespace webrtc {
>@@ -147,50 +142,89 @@ namespace internal {
> 
> PacketMaskTable::PacketMaskTable(FecMaskType fec_mask_type,
>                                  int num_media_packets)
>-    : fec_mask_type_(InitMaskType(fec_mask_type, num_media_packets)),
>-      fec_packet_mask_table_(InitMaskTable(fec_mask_type_)) {}
>+    : table_(PickTable(fec_mask_type, num_media_packets)) {}
> 
>-// Sets |fec_mask_type_| to the type of packet mask selected. The type of
>-// packet mask selected is based on |fec_mask_type| and |num_media_packets|.
>-// If |num_media_packets| is larger than the maximum allowed by |fec_mask_type|
>-// for the bursty type, then the random type is selected.
>-FecMaskType PacketMaskTable::InitMaskType(FecMaskType fec_mask_type,
>-                                          int num_media_packets) {
>-  // The mask should not be bigger than |packetMaskTbl|.
>-  assert(num_media_packets <= static_cast<int>(sizeof(kPacketMaskRandomTbl) /
>-                                               sizeof(*kPacketMaskRandomTbl)));
>-  switch (fec_mask_type) {
>-    case kFecMaskRandom: {
>-      return kFecMaskRandom;
>-    }
>-    case kFecMaskBursty: {
>-      int max_media_packets = static_cast<int>(sizeof(kPacketMaskBurstyTbl) /
>-                                               sizeof(*kPacketMaskBurstyTbl));
>-      if (num_media_packets > max_media_packets) {
>-        return kFecMaskRandom;
>-      } else {
>-        return kFecMaskBursty;
>-      }
>+PacketMaskTable::~PacketMaskTable() = default;
>+
>+rtc::ArrayView<const uint8_t> PacketMaskTable::LookUp(int num_media_packets,
>+                                                      int num_fec_packets) {
>+  RTC_DCHECK_GT(num_media_packets, 0);
>+  RTC_DCHECK_GT(num_fec_packets, 0);
>+  RTC_DCHECK_LE(num_media_packets, kUlpfecMaxMediaPackets);
>+  RTC_DCHECK_LE(num_fec_packets, num_media_packets);
>+
>+  if (num_media_packets <= 12) {
>+    return LookUpInFecTable(table_, num_media_packets - 1, num_fec_packets - 1);
>+  }
>+  int mask_length =
>+      static_cast<int>(PacketMaskSize(static_cast<size_t>(num_media_packets)));
>+
>+  // Generate FEC code mask for {num_media_packets(M), num_fec_packets(N)} (use
>+  // N FEC packets to protect M media packets) In the mask, each FEC packet
>+  // occupies one row, each bit / coloumn represent one media packet. E.g. Row
>+  // A, Col/Bit B is set to 1, means FEC packet A will have protection for media
>+  // packet B.
>+
>+  // Loop through each fec packet.
>+  for (int row = 0; row < num_fec_packets; row++) {
>+    // Loop through each fec code in a row, one code has 8 bits.
>+    // Bit X will be set to 1 if media packet X shall be protected by current
>+    // FEC packet. In this implementation, the protection is interleaved, thus
>+    // media packet X will be protected by FEC packet (X % N)
>+    for (int col = 0; col < mask_length; col++) {
>+      fec_packet_mask_[row * mask_length + col] =
>+          ((col * 8) % num_fec_packets == row && (col * 8) < num_media_packets
>+               ? 0x80
>+               : 0x00) |
>+          ((col * 8 + 1) % num_fec_packets == row &&
>+                   (col * 8 + 1) < num_media_packets
>+               ? 0x40
>+               : 0x00) |
>+          ((col * 8 + 2) % num_fec_packets == row &&
>+                   (col * 8 + 2) < num_media_packets
>+               ? 0x20
>+               : 0x00) |
>+          ((col * 8 + 3) % num_fec_packets == row &&
>+                   (col * 8 + 3) < num_media_packets
>+               ? 0x10
>+               : 0x00) |
>+          ((col * 8 + 4) % num_fec_packets == row &&
>+                   (col * 8 + 4) < num_media_packets
>+               ? 0x08
>+               : 0x00) |
>+          ((col * 8 + 5) % num_fec_packets == row &&
>+                   (col * 8 + 5) < num_media_packets
>+               ? 0x04
>+               : 0x00) |
>+          ((col * 8 + 6) % num_fec_packets == row &&
>+                   (col * 8 + 6) < num_media_packets
>+               ? 0x02
>+               : 0x00) |
>+          ((col * 8 + 7) % num_fec_packets == row &&
>+                   (col * 8 + 7) < num_media_packets
>+               ? 0x01
>+               : 0x00);
>     }
>   }
>-  assert(false);
>-  return kFecMaskRandom;
>+  return {&fec_packet_mask_[0],
>+          static_cast<size_t>(num_fec_packets * mask_length)};
> }
> 
>-// Returns the pointer to the packet mask tables corresponding to type
>-// |fec_mask_type|.
>-const uint8_t* const* const* PacketMaskTable::InitMaskTable(
>-    FecMaskType fec_mask_type) {
>-  switch (fec_mask_type) {
>-    case kFecMaskRandom: {
>-      return kPacketMaskRandomTbl;
>-    }
>-    case kFecMaskBursty: {
>-      return kPacketMaskBurstyTbl;
>-    }
>+// If |num_media_packets| is larger than the maximum allowed by |fec_mask_type|
>+// for the bursty type, or the random table is explicitly asked for, then the
>+// random type is selected. Otherwise the bursty table callback is returned.
>+const uint8_t* PacketMaskTable::PickTable(FecMaskType fec_mask_type,
>+                                          int num_media_packets) {
>+  RTC_DCHECK_GE(num_media_packets, 0);
>+  RTC_DCHECK_LE(static_cast<size_t>(num_media_packets), kUlpfecMaxMediaPackets);
>+
>+  if (fec_mask_type != kFecMaskRandom &&
>+      num_media_packets <=
>+          static_cast<int>(fec_private_tables::kPacketMaskBurstyTbl[0])) {
>+    return &fec_private_tables::kPacketMaskBurstyTbl[0];
>   }
>-  assert(false);
>-  return kPacketMaskRandomTbl;
>+
>+  return &fec_private_tables::kPacketMaskRandomTbl[0];
> }
> 
> // Remaining protection after important (first partition) packet protection
>@@ -200,31 +234,27 @@ void RemainingPacketProtection(int num_media_packets,
>                                int num_mask_bytes,
>                                ProtectionMode mode,
>                                uint8_t* packet_mask,
>-                               const PacketMaskTable& mask_table) {
>+                               PacketMaskTable* mask_table) {
>   if (mode == kModeNoOverlap) {
>     // sub_mask21
> 
>     const int res_mask_bytes =
>         PacketMaskSize(num_media_packets - num_fec_for_imp_packets);
> 
>-    const uint8_t* packet_mask_sub_21 =
>-        mask_table.fec_packet_mask_table()[num_media_packets -
>-                                           num_fec_for_imp_packets -
>-                                           1][num_fec_remaining - 1];
>+    auto end_row = (num_fec_for_imp_packets + num_fec_remaining);
>+    rtc::ArrayView<const uint8_t> packet_mask_sub_21 = mask_table->LookUp(
>+        num_media_packets - num_fec_for_imp_packets, num_fec_remaining);
> 
>     ShiftFitSubMask(num_mask_bytes, res_mask_bytes, num_fec_for_imp_packets,
>-                    (num_fec_for_imp_packets + num_fec_remaining),
>-                    packet_mask_sub_21, packet_mask);
>+                    end_row, &packet_mask_sub_21[0], packet_mask);
> 
>   } else if (mode == kModeOverlap || mode == kModeBiasFirstPacket) {
>     // sub_mask22
>-
>-    const uint8_t* packet_mask_sub_22 =
>-        mask_table.fec_packet_mask_table()[num_media_packets -
>-                                           1][num_fec_remaining - 1];
>+    rtc::ArrayView<const uint8_t> packet_mask_sub_22 =
>+        mask_table->LookUp(num_media_packets, num_fec_remaining);
> 
>     FitSubMask(num_mask_bytes, num_mask_bytes, num_fec_remaining,
>-               packet_mask_sub_22,
>+               &packet_mask_sub_22[0],
>                &packet_mask[num_fec_for_imp_packets * num_mask_bytes]);
> 
>     if (mode == kModeBiasFirstPacket) {
>@@ -234,7 +264,7 @@ void RemainingPacketProtection(int num_media_packets,
>       }
>     }
>   } else {
>-    assert(false);
>+    RTC_NOTREACHED();
>   }
> }
> 
>@@ -243,16 +273,15 @@ void ImportantPacketProtection(int num_fec_for_imp_packets,
>                                int num_imp_packets,
>                                int num_mask_bytes,
>                                uint8_t* packet_mask,
>-                               const PacketMaskTable& mask_table) {
>+                               PacketMaskTable* mask_table) {
>   const int num_imp_mask_bytes = PacketMaskSize(num_imp_packets);
> 
>   // Get sub_mask1 from table
>-  const uint8_t* packet_mask_sub_1 =
>-      mask_table.fec_packet_mask_table()[num_imp_packets -
>-                                         1][num_fec_for_imp_packets - 1];
>+  rtc::ArrayView<const uint8_t> packet_mask_sub_1 =
>+      mask_table->LookUp(num_imp_packets, num_fec_for_imp_packets);
> 
>   FitSubMask(num_mask_bytes, num_imp_mask_bytes, num_fec_for_imp_packets,
>-             packet_mask_sub_1, packet_mask);
>+             &packet_mask_sub_1[0], packet_mask);
> }
> 
> // This function sets the protection allocation: i.e., how many FEC packets
>@@ -329,7 +358,7 @@ void UnequalProtectionMask(int num_media_packets,
>                            int num_imp_packets,
>                            int num_mask_bytes,
>                            uint8_t* packet_mask,
>-                           const PacketMaskTable& mask_table) {
>+                           PacketMaskTable* mask_table) {
>   // Set Protection type and allocation
>   // TODO(marpan): test/update for best mode and some combinations thereof.
> 
>@@ -362,15 +391,69 @@ void UnequalProtectionMask(int num_media_packets,
>   }
> }
> 
>+// This algorithm is tailored to look up data in the |kPacketMaskRandomTbl| and
>+// |kPacketMaskBurstyTbl| tables. These tables only cover fec code for up to 12
>+// media packets. Starting from 13 media packets, the fec code will be generated
>+// at runtime. The format of those arrays is that they're essentially a 3
>+// dimensional array with the following dimensions: * media packet
>+//   * Size for kPacketMaskRandomTbl: 12
>+//   * Size for kPacketMaskBurstyTbl: 12
>+// * fec index
>+//   * Size for both random and bursty table increases from 1 to number of rows.
>+//     (i.e. 1-48, or 1-12 respectively).
>+// * Fec data (what actually gets returned)
>+//   * Size for kPacketMaskRandomTbl: 2 bytes.
>+//     * For all entries: 2 * fec index (1 based)
>+//   * Size for kPacketMaskBurstyTbl: 2 bytes.
>+//     * For all entries: 2 * fec index (1 based)
>+rtc::ArrayView<const uint8_t> LookUpInFecTable(const uint8_t* table,
>+                                               int media_packet_index,
>+                                               int fec_index) {
>+  RTC_DCHECK_LT(media_packet_index, table[0]);
>+
>+  // Skip over the table size.
>+  const uint8_t* entry = &table[1];
>+
>+  uint8_t entry_size_increment = 2;  // 0-16 are 2 byte wide, then changes to 6.
>+
>+  // Hop over un-interesting array entries.
>+  for (int i = 0; i < media_packet_index; ++i) {
>+    if (i == 16)
>+      entry_size_increment = 6;
>+    uint8_t count = entry[0];
>+    ++entry;  // skip over the count.
>+    for (int j = 0; j < count; ++j) {
>+      entry += entry_size_increment * (j + 1);  // skip over the data.
>+    }
>+  }
>+
>+  if (media_packet_index == 16)
>+    entry_size_increment = 6;
>+
>+  RTC_DCHECK_LT(fec_index, entry[0]);
>+  ++entry;  // Skip over the size.
>+
>+  // Find the appropriate data in the second dimension.
>+
>+  // Find the specific data we're looking for.
>+  for (int i = 0; i < fec_index; ++i)
>+    entry += entry_size_increment * (i + 1);  // skip over the data.
>+
>+  size_t size = entry_size_increment * (fec_index + 1);
>+  return {&entry[0], size};
>+}
>+
> void GeneratePacketMasks(int num_media_packets,
>                          int num_fec_packets,
>                          int num_imp_packets,
>                          bool use_unequal_protection,
>-                         const PacketMaskTable& mask_table,
>+                         PacketMaskTable* mask_table,
>                          uint8_t* packet_mask) {
>-  assert(num_media_packets > 0);
>-  assert(num_fec_packets <= num_media_packets && num_fec_packets > 0);
>-  assert(num_imp_packets <= num_media_packets && num_imp_packets >= 0);
>+  RTC_DCHECK_GT(num_media_packets, 0);
>+  RTC_DCHECK_GT(num_fec_packets, 0);
>+  RTC_DCHECK_LE(num_fec_packets, num_media_packets);
>+  RTC_DCHECK_LE(num_imp_packets, num_media_packets);
>+  RTC_DCHECK_GE(num_imp_packets, 0);
> 
>   const int num_mask_bytes = PacketMaskSize(num_media_packets);
> 
>@@ -379,10 +462,9 @@ void GeneratePacketMasks(int num_media_packets,
>     // Retrieve corresponding mask table directly:for equal-protection case.
>     // Mask = (k,n-k), with protection factor = (n-k)/k,
>     // where k = num_media_packets, n=total#packets, (n-k)=num_fec_packets.
>-    memcpy(packet_mask,
>-           mask_table.fec_packet_mask_table()[num_media_packets -
>-                                              1][num_fec_packets - 1],
>-           num_fec_packets * num_mask_bytes);
>+    rtc::ArrayView<const uint8_t> mask =
>+        mask_table->LookUp(num_media_packets, num_fec_packets);
>+    memcpy(packet_mask, &mask[0], mask.size());
>   } else {  // UEP case
>     UnequalProtectionMask(num_media_packets, num_fec_packets, num_imp_packets,
>                           num_mask_bytes, packet_mask, mask_table);
>@@ -416,6 +498,8 @@ void CopyColumn(uint8_t* new_mask,
>                 int num_fec_packets,
>                 int new_bit_index,
>                 int old_bit_index) {
>+  RTC_CHECK_LT(new_bit_index, 8 * new_mask_bytes);
>+
>   // Copy column from the old mask to the beginning of the new mask and shift it
>   // out from the old mask.
>   for (uint16_t row = 0; row < num_fec_packets; ++row) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/forward_error_correction_internal.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/forward_error_correction_internal.h
>index 0c8f0a3e3cc..2e8a202113a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/forward_error_correction_internal.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/forward_error_correction_internal.h
>@@ -12,7 +12,8 @@
> #define MODULES_RTP_RTCP_SOURCE_FORWARD_ERROR_CORRECTION_INTERNAL_H_
> 
> #include "modules/include/module_common_types.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>+
>+#include "api/array_view.h"
> 
> namespace webrtc {
> 
>@@ -24,6 +25,11 @@ constexpr size_t kUlpfecMaxMediaPackets = 48;
> constexpr size_t kUlpfecPacketMaskSizeLBitClear = 2;
> constexpr size_t kUlpfecPacketMaskSizeLBitSet = 6;
> 
>+// Packet code mask maximum length. kFECPacketMaskMaxSize = MaxNumFECPackets *
>+// (kUlpfecMaxMediaPackets / 8), and MaxNumFECPackets is equal to maximum number
>+// of media packets (kUlpfecMaxMediaPackets)
>+constexpr size_t kFECPacketMaskMaxSize = 288;
>+
> // Convenience constants.
> constexpr size_t kUlpfecMinPacketMaskSize = kUlpfecPacketMaskSizeLBitClear;
> constexpr size_t kUlpfecMaxPacketMaskSize = kUlpfecPacketMaskSizeLBitSet;
>@@ -33,19 +39,22 @@ namespace internal {
> class PacketMaskTable {
>  public:
>   PacketMaskTable(FecMaskType fec_mask_type, int num_media_packets);
>-  ~PacketMaskTable() {}
>-  FecMaskType fec_mask_type() const { return fec_mask_type_; }
>-  const uint8_t* const* const* fec_packet_mask_table() const {
>-    return fec_packet_mask_table_;
>-  }
>+  ~PacketMaskTable();
>+
>+  rtc::ArrayView<const uint8_t> LookUp(int num_media_packets,
>+                                       int num_fec_packets);
> 
>  private:
>-  FecMaskType InitMaskType(FecMaskType fec_mask_type, int num_media_packets);
>-  const uint8_t* const* const* InitMaskTable(FecMaskType fec_mask_type_);
>-  const FecMaskType fec_mask_type_;
>-  const uint8_t* const* const* fec_packet_mask_table_;
>+  static const uint8_t* PickTable(FecMaskType fec_mask_type,
>+                                  int num_media_packets);
>+  const uint8_t* table_;
>+  uint8_t fec_packet_mask_[kFECPacketMaskMaxSize];
> };
> 
>+rtc::ArrayView<const uint8_t> LookUpInFecTable(const uint8_t* table,
>+                                               int media_packet_index,
>+                                               int fec_index);
>+
> // Returns an array of packet masks. The mask of a single FEC packet
> // corresponds to a number of mask bytes. The mask indicates which
> // media packets should be protected by the FEC packet.
>@@ -67,9 +76,11 @@ class PacketMaskTable {
> // \param[out] packet_mask             A pointer to hold the packet mask array,
> //                                     of size: num_fec_packets *
> //                                     "number of mask bytes".
>-void GeneratePacketMasks(int num_media_packets, int num_fec_packets,
>-                         int num_imp_packets, bool use_unequal_protection,
>-                         const PacketMaskTable& mask_table,
>+void GeneratePacketMasks(int num_media_packets,
>+                         int num_fec_packets,
>+                         int num_imp_packets,
>+                         bool use_unequal_protection,
>+                         PacketMaskTable* mask_table,
>                          uint8_t* packet_mask);
> 
> // Returns the required packet mask size, given the number of sequence numbers
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
>index f3b0becd905..7ecec764746 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/nack_rtx_unittest.cc
>@@ -1,12 +1,12 @@
> /*
>-*  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
>-*
>-*  Use of this source code is governed by a BSD-style license
>-*  that can be found in the LICENSE file in the root of the source
>-*  tree. An additional intellectual property rights grant can be found
>-*  in the file PATENTS.  All contributing project authors may
>-*  be found in the AUTHORS file in the root of the source tree.
>-*/
>+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
> 
> #include <algorithm>
> #include <iterator>
>@@ -62,9 +62,7 @@ class RtxLoopBackTransport : public webrtc::Transport {
>         count_rtx_ssrc_(0),
>         module_(NULL) {}
> 
>-  void SetSendModule(RtpRtcp* rtpRtcpModule) {
>-    module_ = rtpRtcpModule;
>-  }
>+  void SetSendModule(RtpRtcp* rtpRtcpModule) { module_ = rtpRtcpModule; }
> 
>   void DropEveryNthPacket(int n) { packet_loss_ = n; }
> 
>@@ -120,13 +118,11 @@ class RtpRtcpRtxNackTest : public ::testing::Test {
>   RtpRtcpRtxNackTest()
>       : rtp_rtcp_module_(nullptr),
>         transport_(kTestRtxSsrc),
>-        rtx_stream_(&media_stream_,
>-                    rtx_associated_payload_types_,
>-                    kTestSsrc),
>+        rtx_stream_(&media_stream_, rtx_associated_payload_types_, kTestSsrc),
>         payload_data_length(sizeof(payload_data)),
>         fake_clock(123456),
>         retransmission_rate_limiter_(&fake_clock, kMaxRttMs) {}
>-  ~RtpRtcpRtxNackTest() {}
>+  ~RtpRtcpRtxNackTest() override {}
> 
>   void SetUp() override {
>     RtpRtcp::Configuration configuration;
>@@ -206,9 +202,10 @@ class RtpRtcpRtxNackTest : public ::testing::Test {
>     uint32_t timestamp = 3000;
>     uint16_t nack_list[kVideoNackListSize];
>     for (int frame = 0; frame < kNumFrames; ++frame) {
>+      RTPVideoHeader video_header;
>       EXPECT_TRUE(rtp_rtcp_module_->SendOutgoingData(
>           webrtc::kVideoFrameDelta, kPayloadType, timestamp, timestamp / 90,
>-          payload_data, payload_data_length, nullptr, nullptr, nullptr));
>+          payload_data, payload_data_length, nullptr, &video_header, nullptr));
>       // Min required delay until retransmit = 5 + RTT ms (RTT = 0).
>       fake_clock.AdvanceTimeMilliseconds(5);
>       int length = BuildNackList(nack_list);
>@@ -227,8 +224,8 @@ class RtpRtcpRtxNackTest : public ::testing::Test {
>   std::unique_ptr<ReceiveStatistics> receive_statistics_;
>   RtpRtcp* rtp_rtcp_module_;
>   RtxLoopBackTransport transport_;
>-  const std::map<int, int> rtx_associated_payload_types_ =
>-      {{kRtxPayloadType, kPayloadType}};
>+  const std::map<int, int> rtx_associated_payload_types_ = {
>+      {kRtxPayloadType, kPayloadType}};
>   VerifyingMediaStream media_stream_;
>   RtxReceiveStream rtx_stream_;
>   uint8_t payload_data[65000];
>@@ -254,9 +251,10 @@ TEST_F(RtpRtcpRtxNackTest, LongNackList) {
>   // Send 30 frames which at the default size is roughly what we need to get
>   // enough packets.
>   for (int frame = 0; frame < kNumFrames; ++frame) {
>+    RTPVideoHeader video_header;
>     EXPECT_TRUE(rtp_rtcp_module_->SendOutgoingData(
>         webrtc::kVideoFrameDelta, kPayloadType, timestamp, timestamp / 90,
>-        payload_data, payload_data_length, nullptr, nullptr, nullptr));
>+        payload_data, payload_data_length, nullptr, &video_header, nullptr));
>     // Prepare next frame.
>     timestamp += 3000;
>     fake_clock.AdvanceTimeMilliseconds(33);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/packet_loss_stats.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/packet_loss_stats.cc
>index 69c20c4cf84..076348d2da3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/packet_loss_stats.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/packet_loss_stats.cc
>@@ -23,22 +23,24 @@ namespace webrtc {
> PacketLossStats::PacketLossStats()
>     : single_loss_historic_count_(0),
>       multiple_loss_historic_event_count_(0),
>-      multiple_loss_historic_packet_count_(0) {
>-}
>+      multiple_loss_historic_packet_count_(0) {}
>+
>+PacketLossStats::~PacketLossStats() = default;
> 
> void PacketLossStats::AddLostPacket(uint16_t sequence_number) {
>   // Detect sequence number wrap around.
>   if (!lost_packets_buffer_.empty() &&
>-      static_cast<int>(*(lost_packets_buffer_.rbegin())) - sequence_number
>-      > 0x8000) {
>+      static_cast<int>(*(lost_packets_buffer_.rbegin())) - sequence_number >
>+          0x8000) {
>     // The buffer contains large numbers and this is a small number.
>     lost_packets_wrapped_buffer_.insert(sequence_number);
>   } else {
>     lost_packets_buffer_.insert(sequence_number);
>   }
>-  if (lost_packets_wrapped_buffer_.size() + lost_packets_buffer_.size()
>-      > kBufferSize || (!lost_packets_wrapped_buffer_.empty() &&
>-                        *(lost_packets_wrapped_buffer_.rbegin()) > 0x4000)) {
>+  if (lost_packets_wrapped_buffer_.size() + lost_packets_buffer_.size() >
>+          kBufferSize ||
>+      (!lost_packets_wrapped_buffer_.empty() &&
>+       *(lost_packets_wrapped_buffer_.rbegin()) > 0x4000)) {
>     PruneBuffer();
>   }
> }
>@@ -77,7 +79,7 @@ void PacketLossStats::ComputeLossCounts(
>   std::vector<const std::set<uint16_t>*> buffers;
>   buffers.push_back(&lost_packets_buffer_);
>   buffers.push_back(&lost_packets_wrapped_buffer_);
>-  for (auto buffer : buffers) {
>+  for (const auto* buffer : buffers) {
>     for (auto it = buffer->begin(); it != buffer->end(); ++it) {
>       uint16_t current_num = *it;
>       if (sequential_count > 0 && current_num != ((last_num + 1) & 0xFFFF)) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/packet_loss_stats.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/packet_loss_stats.h
>index 683399f1232..7c4f65855bb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/packet_loss_stats.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/packet_loss_stats.h
>@@ -21,7 +21,7 @@ namespace webrtc {
> class PacketLossStats {
>  public:
>   PacketLossStats();
>-  ~PacketLossStats() {}
>+  ~PacketLossStats();
> 
>   // Adds a lost packet to the stats by sequence number.
>   void AddLostPacket(uint16_t sequence_number);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc
>index 0e40aacec2a..f6bcddb5c81 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.cc
>@@ -39,31 +39,38 @@ StreamStatisticianImpl::StreamStatisticianImpl(
>                         RateStatistics::kBpsScale),
>       max_reordering_threshold_(kDefaultMaxReorderingThreshold),
>       jitter_q4_(0),
>-      cumulative_loss_(0),
>       last_receive_time_ms_(0),
>       last_received_timestamp_(0),
>       received_seq_first_(0),
>       received_seq_max_(0),
>       received_seq_wraps_(0),
>       received_packet_overhead_(12),
>-      last_report_inorder_packets_(0),
>-      last_report_old_packets_(0),
>-      last_report_seq_max_(0),
>       rtcp_callback_(rtcp_callback),
>       rtp_callback_(rtp_callback) {}
> 
>+StreamStatisticianImpl::~StreamStatisticianImpl() = default;
>+
> void StreamStatisticianImpl::IncomingPacket(const RTPHeader& header,
>                                             size_t packet_length,
>                                             bool retransmitted) {
>-  auto counters = UpdateCounters(header, packet_length, retransmitted);
>+  StreamDataCounters counters;
>+  RtcpStatistics rtcp_stats;
>+  {
>+    rtc::CritScope cs(&stream_lock_);
>+    counters = UpdateCounters(header, packet_length, retransmitted);
>+    // We only want to recalculate |fraction_lost| when sending an RTCP SR or
>+    // RR.
>+    rtcp_stats = CalculateRtcpStatistics(/*update_fraction_lost=*/false);
>+  }
>+
>   rtp_callback_->DataCountersUpdated(counters, ssrc_);
>+  rtcp_callback_->StatisticsUpdated(rtcp_stats, ssrc_);
> }
> 
> StreamDataCounters StreamStatisticianImpl::UpdateCounters(
>     const RTPHeader& header,
>     size_t packet_length,
>     bool retransmitted) {
>-  rtc::CritScope cs(&stream_lock_);
>   bool in_order = InOrderPacketInternal(header.sequenceNumber);
>   RTC_DCHECK_EQ(ssrc_, header.ssrc);
>   incoming_bitrate_.Update(packet_length, clock_->TimeInMilliseconds());
>@@ -119,7 +126,7 @@ void StreamStatisticianImpl::UpdateJitter(const RTPHeader& header,
>   uint32_t last_receive_time_rtp =
>       NtpToRtp(last_receive_time_ntp_, header.payload_type_frequency);
>   int32_t time_diff_samples = (receive_time_rtp - last_receive_time_rtp) -
>-      (header.timestamp - last_received_timestamp_);
>+                              (header.timestamp - last_received_timestamp_);
> 
>   time_diff_samples = std::abs(time_diff_samples);
> 
>@@ -151,7 +158,7 @@ void StreamStatisticianImpl::SetMaxReorderingThreshold(
> }
> 
> bool StreamStatisticianImpl::GetStatistics(RtcpStatistics* statistics,
>-                                           bool reset) {
>+                                           bool update_fraction_lost) {
>   {
>     rtc::CritScope cs(&stream_lock_);
>     if (received_seq_first_ == 0 &&
>@@ -160,20 +167,12 @@ bool StreamStatisticianImpl::GetStatistics(RtcpStatistics* statistics,
>       return false;
>     }
> 
>-    if (!reset) {
>-      if (last_report_inorder_packets_ == 0) {
>-        // No report.
>-        return false;
>-      }
>-      // Just get last report.
>-      *statistics = last_reported_statistics_;
>-      return true;
>-    }
>-
>-    *statistics = CalculateRtcpStatistics();
>+    *statistics = CalculateRtcpStatistics(update_fraction_lost);
>   }
> 
>-  rtcp_callback_->StatisticsUpdated(*statistics, ssrc_);
>+  if (update_fraction_lost) {
>+    rtcp_callback_->StatisticsUpdated(*statistics, ssrc_);
>+  }
>   return true;
> }
> 
>@@ -192,90 +191,75 @@ bool StreamStatisticianImpl::GetActiveStatisticsAndReset(
>       return false;
>     }
> 
>-    *statistics = CalculateRtcpStatistics();
>+    *statistics = CalculateRtcpStatistics(/*update_fraction_lost=*/true);
>   }
> 
>   rtcp_callback_->StatisticsUpdated(*statistics, ssrc_);
>   return true;
> }
> 
>-RtcpStatistics StreamStatisticianImpl::CalculateRtcpStatistics() {
>-  RtcpStatistics stats;
>+RtcpStatistics StreamStatisticianImpl::CalculateRtcpStatistics(
>+    bool update_fraction_lost) {
>+  RtcpStatistics statistics;
> 
>-  if (last_report_inorder_packets_ == 0) {
>-    // First time we send a report.
>-    last_report_seq_max_ = received_seq_first_ - 1;
>-  }
>+  uint32_t extended_seq_max = (received_seq_wraps_ << 16) + received_seq_max_;
> 
>-  // Calculate fraction lost.
>-  uint16_t exp_since_last = (received_seq_max_ - last_report_seq_max_);
>+  if (update_fraction_lost) {
>+    if (last_report_received_packets_ == 0) {
>+      // First time we're calculating fraction lost.
>+      last_report_extended_seq_max_ = received_seq_first_ - 1;
>+    }
> 
>-  if (last_report_seq_max_ > received_seq_max_) {
>-    // Can we assume that the seq_num can't go decrease over a full RTCP period?
>-    exp_since_last = 0;
>-  }
>+    uint32_t exp_since_last =
>+        (extended_seq_max - last_report_extended_seq_max_);
>+
>+    // Number of received RTP packets since last report; counts all packets
>+    // including retransmissions.
>+    uint32_t rec_since_last =
>+        receive_counters_.transmitted.packets - last_report_received_packets_;
>+
>+    // Calculate fraction lost according to RFC3550 Appendix A.3. Snap to 0 if
>+    // negative (which is possible with duplicate packets).
>+    uint8_t local_fraction_lost = 0;
>+    if (exp_since_last > rec_since_last) {
>+      // Scale 0 to 255, where 255 is 100% loss.
>+      local_fraction_lost = static_cast<uint8_t>(
>+          255 * (exp_since_last - rec_since_last) / exp_since_last);
>+    }
> 
>-  // Number of received RTP packets since last report, counts all packets but
>-  // not re-transmissions.
>-  uint32_t rec_since_last =
>-      (receive_counters_.transmitted.packets -
>-       receive_counters_.retransmitted.packets) - last_report_inorder_packets_;
>-
>-  // With NACK we don't know the expected retransmissions during the last
>-  // second. We know how many "old" packets we have received. We just count
>-  // the number of old received to estimate the loss, but it still does not
>-  // guarantee an exact number since we run this based on time triggered by
>-  // sending of an RTP packet. This should have a minimum effect.
>-
>-  // With NACK we don't count old packets as received since they are
>-  // re-transmitted. We use RTT to decide if a packet is re-ordered or
>-  // re-transmitted.
>-  uint32_t retransmitted_packets =
>-      receive_counters_.retransmitted.packets - last_report_old_packets_;
>-  rec_since_last += retransmitted_packets;
>-
>-  int32_t missing = 0;
>-  if (exp_since_last > rec_since_last) {
>-    missing = (exp_since_last - rec_since_last);
>+    last_fraction_lost_ = local_fraction_lost;
>+    last_report_received_packets_ = receive_counters_.transmitted.packets;
>+    last_report_extended_seq_max_ = extended_seq_max;
>   }
>-  uint8_t local_fraction_lost = 0;
>-  if (exp_since_last) {
>-    // Scale 0 to 255, where 255 is 100% loss.
>-    local_fraction_lost =
>-        static_cast<uint8_t>(255 * missing / exp_since_last);
>-  }
>-  stats.fraction_lost = local_fraction_lost;
>-
>-  // We need a counter for cumulative loss too.
>-  // TODO(danilchap): Ensure cumulative loss is below maximum value of 2^24.
>-  cumulative_loss_ += missing;
>-  stats.packets_lost = cumulative_loss_;
>-  stats.extended_highest_sequence_number =
>-      (received_seq_wraps_ << 16) + received_seq_max_;
>-  // Note: internal jitter value is in Q4 and needs to be scaled by 1/16.
>-  stats.jitter = jitter_q4_ >> 4;
> 
>-  // Store this report.
>-  last_reported_statistics_ = stats;
>+  statistics.fraction_lost = last_fraction_lost_;
>+  // Calculate cumulative loss, according to RFC3550 Appendix A.3.
>+  uint32_t total_expected_packets = extended_seq_max - received_seq_first_ + 1;
>+  statistics.packets_lost =
>+      total_expected_packets - receive_counters_.transmitted.packets;
>+  // Since cumulative loss is carried in a signed 24-bit field in RTCP, we may
>+  // need to clamp it.
>+  statistics.packets_lost = std::min(statistics.packets_lost, 0x7fffff);
>+  // TODO(bugs.webrtc.org/9598): This packets_lost should be signed according to
>+  // RFC3550. However, old WebRTC implementations reads it as unsigned.
>+  // Therefore we limit this to 0.
>+  statistics.packets_lost = std::max(statistics.packets_lost, 0);
>+  statistics.extended_highest_sequence_number = extended_seq_max;
>+  // Note: internal jitter value is in Q4 and needs to be scaled by 1/16.
>+  statistics.jitter = jitter_q4_ >> 4;
> 
>-  // Only for report blocks in RTCP SR and RR.
>-  last_report_inorder_packets_ =
>-      receive_counters_.transmitted.packets -
>-      receive_counters_.retransmitted.packets;
>-  last_report_old_packets_ = receive_counters_.retransmitted.packets;
>-  last_report_seq_max_ = received_seq_max_;
>   BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "cumulative_loss_pkts",
>                                   clock_->TimeInMilliseconds(),
>-                                  cumulative_loss_, ssrc_);
>+                                  statistics.packets_lost, ssrc_);
>   BWE_TEST_LOGGING_PLOT_WITH_SSRC(
>       1, "received_seq_max_pkts", clock_->TimeInMilliseconds(),
>       (received_seq_max_ - received_seq_first_), ssrc_);
> 
>-  return stats;
>+  return statistics;
> }
> 
>-void StreamStatisticianImpl::GetDataCounters(
>-    size_t* bytes_received, uint32_t* packets_received) const {
>+void StreamStatisticianImpl::GetDataCounters(size_t* bytes_received,
>+                                             uint32_t* packets_received) const {
>   rtc::CritScope cs(&stream_lock_);
>   if (bytes_received) {
>     *bytes_received = receive_counters_.transmitted.payload_bytes +
>@@ -299,7 +283,7 @@ uint32_t StreamStatisticianImpl::BitrateReceived() const {
> }
> 
> bool StreamStatisticianImpl::IsRetransmitOfOldPacket(
>-    const RTPHeader& header, int64_t min_rtt) const {
>+    const RTPHeader& header) const {
>   rtc::CritScope cs(&stream_lock_);
>   if (InOrderPacketInternal(header.sequenceNumber)) {
>     return false;
>@@ -307,49 +291,40 @@ bool StreamStatisticianImpl::IsRetransmitOfOldPacket(
>   uint32_t frequency_khz = header.payload_type_frequency / 1000;
>   assert(frequency_khz > 0);
> 
>-  int64_t time_diff_ms = clock_->TimeInMilliseconds() -
>-      last_receive_time_ms_;
>+  int64_t time_diff_ms = clock_->TimeInMilliseconds() - last_receive_time_ms_;
> 
>   // Diff in time stamp since last received in order.
>   uint32_t timestamp_diff = header.timestamp - last_received_timestamp_;
>   uint32_t rtp_time_stamp_diff_ms = timestamp_diff / frequency_khz;
> 
>   int64_t max_delay_ms = 0;
>-  if (min_rtt == 0) {
>-    // Jitter standard deviation in samples.
>-    float jitter_std = sqrt(static_cast<float>(jitter_q4_ >> 4));
> 
>-    // 2 times the standard deviation => 95% confidence.
>-    // And transform to milliseconds by dividing by the frequency in kHz.
>-    max_delay_ms = static_cast<int64_t>((2 * jitter_std) / frequency_khz);
>+  // Jitter standard deviation in samples.
>+  float jitter_std = sqrt(static_cast<float>(jitter_q4_ >> 4));
> 
>-    // Min max_delay_ms is 1.
>-    if (max_delay_ms == 0) {
>-      max_delay_ms = 1;
>-    }
>-  } else {
>-    max_delay_ms = (min_rtt / 3) + 1;
>+  // 2 times the standard deviation => 95% confidence.
>+  // And transform to milliseconds by dividing by the frequency in kHz.
>+  max_delay_ms = static_cast<int64_t>((2 * jitter_std) / frequency_khz);
>+
>+  // Min max_delay_ms is 1.
>+  if (max_delay_ms == 0) {
>+    max_delay_ms = 1;
>   }
>   return time_diff_ms > rtp_time_stamp_diff_ms + max_delay_ms;
> }
> 
>-bool StreamStatisticianImpl::IsPacketInOrder(uint16_t sequence_number) const {
>-  rtc::CritScope cs(&stream_lock_);
>-  return InOrderPacketInternal(sequence_number);
>-}
>-
> bool StreamStatisticianImpl::InOrderPacketInternal(
>     uint16_t sequence_number) const {
>   // First packet is always in order.
>-  if (last_receive_time_ms_ == 0)
>+  if (receive_counters_.transmitted.packets == 0)
>     return true;
> 
>   if (IsNewerSequenceNumber(sequence_number, received_seq_max_)) {
>     return true;
>   } else {
>     // If we have a restart of the remote side this packet is still in order.
>-    return !IsNewerSequenceNumber(sequence_number, received_seq_max_ -
>-                                  max_reordering_threshold_);
>+    return !IsNewerSequenceNumber(
>+        sequence_number, received_seq_max_ - max_reordering_threshold_);
>   }
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
>index 1f827f62a93..35869f1f1e5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/receive_statistics_impl.h
>@@ -13,6 +13,8 @@
> 
> #include "modules/rtp_rtcp/include/receive_statistics.h"
> 
>+#include <math.h>
>+
> #include <algorithm>
> #include <map>
> #include <vector>
>@@ -29,19 +31,17 @@ class StreamStatisticianImpl : public StreamStatistician {
>                          Clock* clock,
>                          RtcpStatisticsCallback* rtcp_callback,
>                          StreamDataCountersCallback* rtp_callback);
>-  virtual ~StreamStatisticianImpl() {}
>+  ~StreamStatisticianImpl() override;
> 
>-  // |reset| here and in next method restarts calculation of fraction_lost stat.
>-  bool GetStatistics(RtcpStatistics* statistics, bool reset) override;
>+  bool GetStatistics(RtcpStatistics* statistics,
>+                     bool update_fraction_lost) override;
>   bool GetActiveStatisticsAndReset(RtcpStatistics* statistics);
>   void GetDataCounters(size_t* bytes_received,
>                        uint32_t* packets_received) const override;
>   void GetReceiveStreamDataCounters(
>       StreamDataCounters* data_counters) const override;
>   uint32_t BitrateReceived() const override;
>-  bool IsRetransmitOfOldPacket(const RTPHeader& header,
>-                               int64_t min_rtt) const override;
>-  bool IsPacketInOrder(uint16_t sequence_number) const override;
>+  bool IsRetransmitOfOldPacket(const RTPHeader& header) const override;
> 
>   void IncomingPacket(const RTPHeader& rtp_header,
>                       size_t packet_length,
>@@ -50,13 +50,15 @@ class StreamStatisticianImpl : public StreamStatistician {
>   void SetMaxReorderingThreshold(int max_reordering_threshold);
> 
>  private:
>-  bool InOrderPacketInternal(uint16_t sequence_number) const;
>-  RtcpStatistics CalculateRtcpStatistics()
>+  bool InOrderPacketInternal(uint16_t sequence_number) const
>+      RTC_EXCLUSIVE_LOCKS_REQUIRED(stream_lock_);
>+  RtcpStatistics CalculateRtcpStatistics(bool update_fraction_lost)
>       RTC_EXCLUSIVE_LOCKS_REQUIRED(stream_lock_);
>   void UpdateJitter(const RTPHeader& header, NtpTime receive_time);
>   StreamDataCounters UpdateCounters(const RTPHeader& rtp_header,
>                                     size_t packet_length,
>-                                    bool retransmitted);
>+                                    bool retransmitted)
>+      RTC_EXCLUSIVE_LOCKS_REQUIRED(stream_lock_);
> 
>   const uint32_t ssrc_;
>   Clock* const clock_;
>@@ -66,7 +68,6 @@ class StreamStatisticianImpl : public StreamStatistician {
> 
>   // Stats on received RTP packets.
>   uint32_t jitter_q4_;
>-  uint32_t cumulative_loss_;
> 
>   int64_t last_receive_time_ms_;
>   NtpTime last_receive_time_ntp_;
>@@ -77,13 +78,12 @@ class StreamStatisticianImpl : public StreamStatistician {
> 
>   // Current counter values.
>   size_t received_packet_overhead_;
>-  StreamDataCounters receive_counters_;
>+  StreamDataCounters receive_counters_ RTC_GUARDED_BY(stream_lock_);
> 
>-  // Counter values when we sent the last report.
>-  uint32_t last_report_inorder_packets_;
>-  uint32_t last_report_old_packets_;
>-  uint16_t last_report_seq_max_;
>-  RtcpStatistics last_reported_statistics_;
>+  // Used to calculate fraction_lost between reports.
>+  uint32_t last_report_received_packets_ = 0;
>+  uint32_t last_report_extended_seq_max_ = 0;
>+  uint8_t last_fraction_lost_ = 0;
> 
>   // stream_lock_ shouldn't be held when calling callbacks.
>   RtcpStatisticsCallback* const rtcp_callback_;
>@@ -96,7 +96,7 @@ class ReceiveStatisticsImpl : public ReceiveStatistics,
>  public:
>   explicit ReceiveStatisticsImpl(Clock* clock);
> 
>-  ~ReceiveStatisticsImpl();
>+  ~ReceiveStatisticsImpl() override;
> 
>   // Implement ReceiveStatisticsProvider.
>   std::vector<rtcp::ReportBlock> RtcpReportBlocks(size_t max_blocks) override;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc
>index 001ae2e1197..eedcf46df7c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/receive_statistics_unittest.cc
>@@ -19,6 +19,8 @@
> namespace webrtc {
> namespace {
> 
>+using ::testing::_;
>+using ::testing::SaveArg;
> using ::testing::SizeIs;
> using ::testing::UnorderedElementsAre;
> 
>@@ -39,9 +41,8 @@ RTPHeader CreateRtpHeader(uint32_t ssrc) {
> 
> class ReceiveStatisticsTest : public ::testing::Test {
>  public:
>-  ReceiveStatisticsTest() :
>-      clock_(0),
>-      receive_statistics_(ReceiveStatistics::Create(&clock_)) {
>+  ReceiveStatisticsTest()
>+      : clock_(0), receive_statistics_(ReceiveStatistics::Create(&clock_)) {
>     header1_ = CreateRtpHeader(kSsrc1);
>     header2_ = CreateRtpHeader(kSsrc2);
>   }
>@@ -74,8 +75,7 @@ TEST_F(ReceiveStatisticsTest, TwoIncomingSsrcs) {
>   EXPECT_EQ(200u, bytes_received);
>   EXPECT_EQ(2u, packets_received);
> 
>-  statistician =
>-      receive_statistics_->GetStatistician(kSsrc2);
>+  statistician = receive_statistics_->GetStatistician(kSsrc2);
>   ASSERT_TRUE(statistician != NULL);
>   EXPECT_GT(statistician->BitrateReceived(), 0u);
>   statistician->GetDataCounters(&bytes_received, &packets_received);
>@@ -190,29 +190,82 @@ TEST_F(ReceiveStatisticsTest, GetReceiveStreamDataCounters) {
>   EXPECT_EQ(2u, counters.transmitted.packets);
> }
> 
>-TEST_F(ReceiveStatisticsTest, RtcpCallbacks) {
>-  class TestCallback : public RtcpStatisticsCallback {
>-   public:
>-    TestCallback()
>-        : RtcpStatisticsCallback(), num_calls_(0), ssrc_(0), stats_() {}
>-    virtual ~TestCallback() {}
>+class MockRtcpCallback : public RtcpStatisticsCallback {
>+ public:
>+  MOCK_METHOD2(StatisticsUpdated,
>+               void(const RtcpStatistics& statistics, uint32_t ssrc));
>+  MOCK_METHOD2(CNameChanged, void(const char* cname, uint32_t ssrc));
>+};
> 
>-    void StatisticsUpdated(const RtcpStatistics& statistics,
>-                           uint32_t ssrc) override {
>-      ssrc_ = ssrc;
>-      stats_ = statistics;
>-      ++num_calls_;
>-    }
>+// Test that the RTCP statistics callback is invoked every time a packet is
>+// received (so that at the application level, GetStats will return up-to-date
>+// stats, not just stats from the last generated RTCP SR or RR).
>+TEST_F(ReceiveStatisticsTest,
>+       RtcpStatisticsCallbackInvokedForEveryPacketReceived) {
>+  MockRtcpCallback callback;
>+  receive_statistics_->RegisterRtcpStatisticsCallback(&callback);
>+
>+  // Just receive the same packet multiple times; doesn't really matter for the
>+  // purposes of this test.
>+  EXPECT_CALL(callback, StatisticsUpdated(_, _)).Times(3);
>+  receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
>+  receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
>+  receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
>+}
>+
>+// The callback should also be invoked when |fraction_lost| is updated due to
>+// GetStatistics being called.
>+TEST_F(ReceiveStatisticsTest,
>+       RtcpStatisticsCallbackInvokedWhenFractionLostUpdated) {
>+  MockRtcpCallback callback;
>+  receive_statistics_->RegisterRtcpStatisticsCallback(&callback);
>+
>+  EXPECT_CALL(callback, StatisticsUpdated(_, _)).Times(2);
>+  receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
>+  receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
>+
>+  // This just returns the current statistics without updating anything, so no
>+  // need to invoke the callback.
>+  RtcpStatistics statistics;
>+  receive_statistics_->GetStatistician(kSsrc1)->GetStatistics(
>+      &statistics, /*update_fraction_lost=*/false);
>+
>+  // Update fraction lost, expecting a new callback.
>+  EXPECT_CALL(callback, StatisticsUpdated(_, _)).Times(1);
>+  receive_statistics_->GetStatistician(kSsrc1)->GetStatistics(
>+      &statistics, /*update_fraction_lost=*/true);
>+}
> 
>-    void CNameChanged(const char* cname, uint32_t ssrc) override {}
>+TEST_F(ReceiveStatisticsTest,
>+       RtcpStatisticsCallbackNotInvokedAfterDeregistered) {
>+  // Register the callback and receive a couple packets.
>+  MockRtcpCallback callback;
>+  receive_statistics_->RegisterRtcpStatisticsCallback(&callback);
>+  EXPECT_CALL(callback, StatisticsUpdated(_, _)).Times(2);
>+  receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
>+  receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
> 
>-    uint32_t num_calls_;
>-    uint32_t ssrc_;
>-    RtcpStatistics stats_;
>-  } callback;
>+  // Dereigster the callback. Neither receiving a packet nor generating a
>+  // report (calling GetStatistics) should result in another callback.
>+  receive_statistics_->RegisterRtcpStatisticsCallback(nullptr);
>+  receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
>+  RtcpStatistics statistics;
>+  receive_statistics_->GetStatistician(kSsrc1)->GetStatistics(
>+      &statistics, /*update_fraction_lost=*/true);
>+}
> 
>+// Test that the RtcpStatisticsCallback sees the exact same values as returned
>+// from GetStatistics.
>+TEST_F(ReceiveStatisticsTest,
>+       RtcpStatisticsFromCallbackMatchThoseFromGetStatistics) {
>+  MockRtcpCallback callback;
>+  RtcpStatistics stats_from_callback;
>+  EXPECT_CALL(callback, StatisticsUpdated(_, _))
>+      .WillRepeatedly(SaveArg<0>(&stats_from_callback));
>   receive_statistics_->RegisterRtcpStatisticsCallback(&callback);
> 
>+  // Using units of milliseconds.
>+  header1_.payload_type_frequency = 1000;
>   // Add some arbitrary data, with loss and jitter.
>   header1_.sequenceNumber = 1;
>   clock_.AdvanceTimeMilliseconds(7);
>@@ -230,63 +283,394 @@ TEST_F(ReceiveStatisticsTest, RtcpCallbacks) {
>   clock_.AdvanceTimeMilliseconds(11);
>   header1_.timestamp += 17;
>   receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
>-  ++header1_.sequenceNumber;
> 
>-  EXPECT_EQ(0u, callback.num_calls_);
>+  // The stats from the last callback due to IncomingPacket should match
>+  // those returned by GetStatistics afterwards.
>+  RtcpStatistics stats_from_getstatistics;
>+  receive_statistics_->GetStatistician(kSsrc1)->GetStatistics(
>+      &stats_from_getstatistics, /*update_fraction_lost=*/false);
>+
>+  EXPECT_EQ(stats_from_getstatistics.packets_lost,
>+            stats_from_callback.packets_lost);
>+  EXPECT_EQ(stats_from_getstatistics.extended_highest_sequence_number,
>+            stats_from_callback.extended_highest_sequence_number);
>+  EXPECT_EQ(stats_from_getstatistics.fraction_lost,
>+            stats_from_callback.fraction_lost);
>+  EXPECT_EQ(stats_from_getstatistics.jitter, stats_from_callback.jitter);
>+
>+  // Now update fraction lost, and check that we got matching values from the
>+  // new callback.
>+  receive_statistics_->GetStatistician(kSsrc1)->GetStatistics(
>+      &stats_from_getstatistics, /*update_fraction_lost=*/true);
>+  EXPECT_EQ(stats_from_getstatistics.packets_lost,
>+            stats_from_callback.packets_lost);
>+  EXPECT_EQ(stats_from_getstatistics.extended_highest_sequence_number,
>+            stats_from_callback.extended_highest_sequence_number);
>+  EXPECT_EQ(stats_from_getstatistics.fraction_lost,
>+            stats_from_callback.fraction_lost);
>+  EXPECT_EQ(stats_from_getstatistics.jitter, stats_from_callback.jitter);
>+}
>+
>+// Test that |fraction_lost| is only updated when a report is generated (when
>+// GetStatistics is called with |update_fraction_lost| set to true). Meaning
>+// that it will always represent a value computed between two RTCP SR or RRs.
>+TEST_F(ReceiveStatisticsTest, FractionLostOnlyUpdatedWhenReportGenerated) {
>+  MockRtcpCallback callback;
>+  RtcpStatistics stats_from_callback;
>+  EXPECT_CALL(callback, StatisticsUpdated(_, _))
>+      .WillRepeatedly(SaveArg<0>(&stats_from_callback));
>+  receive_statistics_->RegisterRtcpStatisticsCallback(&callback);
>+
>+  // Simulate losing one packet.
>+  header1_.sequenceNumber = 1;
>+  receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
>+  header1_.sequenceNumber = 2;
>+  receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
>+  header1_.sequenceNumber = 4;
>+  receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
>+  // Haven't generated a report yet, so |fraction_lost| should still be 0.
>+  EXPECT_EQ(0u, stats_from_callback.fraction_lost);
>+
>+  // Call GetStatistics with |update_fraction_lost| set to false; should be a
>+  // no-op.
>+  RtcpStatistics statistics;
>+  receive_statistics_->GetStatistician(kSsrc1)->GetStatistics(
>+      &statistics, /*update_fraction_lost=*/false);
>+  EXPECT_EQ(0u, stats_from_callback.fraction_lost);
>+
>+  // Call GetStatistics with |update_fraction_lost| set to true, simulating a
>+  // report being generated.
>+  receive_statistics_->GetStatistician(kSsrc1)->GetStatistics(
>+      &statistics, /*update_fraction_lost=*/true);
>+  // 25% = 63/255.
>+  EXPECT_EQ(63u, stats_from_callback.fraction_lost);
>+
>+  // Lose another packet.
>+  header1_.sequenceNumber = 6;
>+  receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
>+  // Should return same value as before since we haven't generated a new report
>+  // yet.
>+  EXPECT_EQ(63u, stats_from_callback.fraction_lost);
>+
>+  // Simulate another report being generated.
>+  receive_statistics_->GetStatistician(kSsrc1)->GetStatistics(
>+      &statistics, /*update_fraction_lost=*/true);
>+  // 50% = 127/255.
>+  EXPECT_EQ(127, stats_from_callback.fraction_lost);
>+}
>+
>+// Simple test for fraction/cumulative loss computation, with only loss, no
>+// duplicates or reordering.
>+TEST_F(ReceiveStatisticsTest, SimpleLossComputation) {
>+  header1_.sequenceNumber = 1;
>+  receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
>+  header1_.sequenceNumber = 3;
>+  receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
>+  header1_.sequenceNumber = 4;
>+  receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
>+  header1_.sequenceNumber = 5;
>+  receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
>+
>+  RtcpStatistics statistics;
>+  receive_statistics_->GetStatistician(kSsrc1)->GetStatistics(
>+      &statistics, /*update_fraction_lost=*/true);
>+  // 20% = 51/255.
>+  EXPECT_EQ(51u, statistics.fraction_lost);
>+  EXPECT_EQ(1, statistics.packets_lost);
>+}
>+
>+// Test that fraction/cumulative loss is computed correctly when there's some
>+// reordering.
>+TEST_F(ReceiveStatisticsTest, LossComputationWithReordering) {
>+  header1_.sequenceNumber = 1;
>+  receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
>+  header1_.sequenceNumber = 3;
>+  receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
>+  header1_.sequenceNumber = 2;
>+  receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
>+  header1_.sequenceNumber = 5;
>+  receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
>+
>+  RtcpStatistics statistics;
>+  receive_statistics_->GetStatistician(kSsrc1)->GetStatistics(
>+      &statistics, /*update_fraction_lost=*/true);
>+  // 20% = 51/255.
>+  EXPECT_EQ(51u, statistics.fraction_lost);
>+}
>+
>+// Somewhat unintuitively, duplicate packets count against lost packets
>+// according to RFC3550.
>+TEST_F(ReceiveStatisticsTest, LossComputationWithDuplicates) {
>+  // Lose 2 packets, but also receive 1 duplicate. Should actually count as
>+  // only 1 packet being lost.
>+  header1_.sequenceNumber = 1;
>+  receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
>+  header1_.sequenceNumber = 4;
>+  receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
>+  header1_.sequenceNumber = 4;
>+  receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
>+  header1_.sequenceNumber = 5;
>+  receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
>+
>+  RtcpStatistics statistics;
>+  receive_statistics_->GetStatistician(kSsrc1)->GetStatistics(
>+      &statistics, /*update_fraction_lost=*/true);
>+  // 20% = 51/255.
>+  EXPECT_EQ(51u, statistics.fraction_lost);
>+  EXPECT_EQ(1, statistics.packets_lost);
>+}
>+
>+// Test that sequence numbers wrapping around doesn't screw up loss
>+// computations.
>+TEST_F(ReceiveStatisticsTest, LossComputationWithSequenceNumberWrapping) {
>+  // First, test loss computation over a period that included a sequence number
>+  // rollover.
>+  header1_.sequenceNumber = 65533;
>+  receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
>+  header1_.sequenceNumber = 0;
>+  receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
>+  header1_.sequenceNumber = 65534;
>+  receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
>+  header1_.sequenceNumber = 1;
>+  receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
>+
>+  // Only one packet was actually lost, 65535.
>+  RtcpStatistics statistics;
>+  receive_statistics_->GetStatistician(kSsrc1)->GetStatistics(
>+      &statistics, /*update_fraction_lost=*/true);
>+  // 20% = 51/255.
>+  EXPECT_EQ(51u, statistics.fraction_lost);
>+  EXPECT_EQ(1, statistics.packets_lost);
> 
>-  // Call GetStatistics, simulating a timed rtcp sender thread.
>+  // Now test losing one packet *after* the rollover.
>+  header1_.sequenceNumber = 3;
>+  receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
>+  receive_statistics_->GetStatistician(kSsrc1)->GetStatistics(
>+      &statistics, /*update_fraction_lost=*/true);
>+  // 50% = 127/255.
>+  EXPECT_EQ(127u, statistics.fraction_lost);
>+  EXPECT_EQ(2, statistics.packets_lost);
>+}
>+
>+// Somewhat unintuitively, since duplicate packets count against loss, you can
>+// actually end up with negative loss. |fraction_lost| should be clamped to
>+// zero in this case, since it's signed, while |packets_lost| is signed so it
>+// should be negative.
>+TEST_F(ReceiveStatisticsTest, NegativeLoss) {
>+  // Receive one packet and simulate a report being generated by calling
>+  // GetStatistics, to establish a baseline for |fraction_lost|.
>+  header1_.sequenceNumber = 1;
>+  receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
>   RtcpStatistics statistics;
>-  receive_statistics_->GetStatistician(kSsrc1)
>-      ->GetStatistics(&statistics, true);
>-
>-  EXPECT_EQ(1u, callback.num_calls_);
>-  EXPECT_EQ(callback.ssrc_, kSsrc1);
>-  EXPECT_EQ(statistics.packets_lost, callback.stats_.packets_lost);
>-  EXPECT_EQ(statistics.extended_highest_sequence_number,
>-            callback.stats_.extended_highest_sequence_number);
>-  EXPECT_EQ(statistics.fraction_lost, callback.stats_.fraction_lost);
>-  EXPECT_EQ(statistics.jitter, callback.stats_.jitter);
>-  EXPECT_EQ(51, statistics.fraction_lost);
>+  receive_statistics_->GetStatistician(kSsrc1)->GetStatistics(
>+      &statistics, /*update_fraction_lost=*/true);
>+
>+  // Receive some duplicate packets. Results in "negative" loss, since
>+  // "expected packets since last report" is 3 and "received" is 4, and 3 minus
>+  // 4 is -1. See RFC3550 Appendix A.3.
>+  header1_.sequenceNumber = 4;
>+  receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
>+  header1_.sequenceNumber = 2;
>+  receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
>+  header1_.sequenceNumber = 2;
>+  receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
>+  header1_.sequenceNumber = 2;
>+  receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
>+  receive_statistics_->GetStatistician(kSsrc1)->GetStatistics(
>+      &statistics, /*update_fraction_lost=*/true);
>+  EXPECT_EQ(0u, statistics.fraction_lost);
>+  // TODO(bugs.webrtc.org/9598): Since old WebRTC implementations reads this
>+  // value as unsigned we currently limit it to 0.
>+  EXPECT_EQ(0, statistics.packets_lost);
>+
>+  // Lose 2 packets; now cumulative loss should become positive again.
>+  header1_.sequenceNumber = 7;
>+  receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
>+  receive_statistics_->GetStatistician(kSsrc1)->GetStatistics(
>+      &statistics, /*update_fraction_lost=*/true);
>+  // 66% = 170/255.
>+  EXPECT_EQ(170u, statistics.fraction_lost);
>   EXPECT_EQ(1, statistics.packets_lost);
>-  EXPECT_EQ(5u, statistics.extended_highest_sequence_number);
>-  EXPECT_EQ(4u, statistics.jitter);
>+}
>+
>+// Since cumulative loss is carried in a signed 24-bit field, it should be
>+// clamped to 0x7fffff in the positive direction, 0x800000 in the negative
>+// direction.
>+TEST_F(ReceiveStatisticsTest, PositiveCumulativeLossClamped) {
>+  header1_.sequenceNumber = 1;
>+  receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
>+
>+  // Lose 2^23 packets, expecting loss to be clamped to 2^23-1.
>+  for (int i = 0; i < 0x800000; ++i) {
>+    header1_.sequenceNumber = (header1_.sequenceNumber + 2 % 65536);
>+    receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
>+  }
>+  RtcpStatistics statistics;
>+  receive_statistics_->GetStatistician(kSsrc1)->GetStatistics(
>+      &statistics, /*update_fraction_lost=*/false);
>+  EXPECT_EQ(0x7fffff, statistics.packets_lost);
>+}
> 
>-  receive_statistics_->RegisterRtcpStatisticsCallback(NULL);
>+TEST_F(ReceiveStatisticsTest, NegativeCumulativeLossClamped) {
>+  header1_.sequenceNumber = 1;
>+  receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
>+
>+  // Receive 2^23+1 duplicate packets (counted as negative loss), expecting
>+  // loss to be clamped to -2^23.
>+  for (int i = 0; i < 0x800001; ++i) {
>+    receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
>+  }
>+  RtcpStatistics statistics;
>+  receive_statistics_->GetStatistician(kSsrc1)->GetStatistics(
>+      &statistics, /*update_fraction_lost=*/false);
>+  // TODO(bugs.webrtc.org/9598): Since old WebRTC implementations reads this
>+  // value as unsigned we currently limit it to 0.
>+  EXPECT_EQ(0, statistics.packets_lost);
>+}
>+
>+// Test that the extended highest sequence number is computed correctly when
>+// sequence numbers wrap around or packets are received out of order.
>+TEST_F(ReceiveStatisticsTest, ExtendedHighestSequenceNumberComputation) {
>+  MockRtcpCallback callback;
>+  RtcpStatistics stats_from_callback;
>+  EXPECT_CALL(callback, StatisticsUpdated(_, _))
>+      .WillRepeatedly(SaveArg<0>(&stats_from_callback));
>+  receive_statistics_->RegisterRtcpStatisticsCallback(&callback);
>+
>+  header1_.sequenceNumber = 65535;
>+  receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
>+  EXPECT_EQ(65535u, stats_from_callback.extended_highest_sequence_number);
> 
>-  // Add some more data.
>+  // Wrap around.
>+  header1_.sequenceNumber = 1;
>+  receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
>+  EXPECT_EQ(65536u + 1u, stats_from_callback.extended_highest_sequence_number);
>+
>+  // Should be treated as out of order; shouldn't increment highest extended
>+  // sequence number.
>+  header1_.sequenceNumber = 65530;
>+  receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
>+  EXPECT_EQ(65536u + 1u, stats_from_callback.extended_highest_sequence_number);
>+
>+  // Receive a couple packets then wrap around again.
>+  // TODO(bugs.webrtc.org/9445): With large jumps like this, RFC3550 suggests
>+  // for the receiver to assume the other side restarted, and reset all its
>+  // sequence number counters. Why aren't we doing this?
>+  header1_.sequenceNumber = 30000;
>+  receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
>+  EXPECT_EQ(65536u + 30000u,
>+            stats_from_callback.extended_highest_sequence_number);
>+
>+  header1_.sequenceNumber = 50000;
>+  receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
>+  EXPECT_EQ(65536u + 50000u,
>+            stats_from_callback.extended_highest_sequence_number);
>+
>+  header1_.sequenceNumber = 10000;
>+  receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
>+  EXPECT_EQ(2 * 65536u + 10000u,
>+            stats_from_callback.extended_highest_sequence_number);
>+
>+  // If a packet is received more than "MaxReorderingThreshold" packets out of
>+  // order (defaults to 50), it's assumed to be in order.
>+  // TODO(bugs.webrtc.org/9445): RFC3550 would recommend treating this as a
>+  // restart as mentioned above.
>+  header1_.sequenceNumber = 9900;
>+  receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
>+  EXPECT_EQ(3 * 65536u + 9900u,
>+            stats_from_callback.extended_highest_sequence_number);
>+}
>+
>+// Test jitter computation with no loss/reordering/etc.
>+TEST_F(ReceiveStatisticsTest, SimpleJitterComputation) {
>+  MockRtcpCallback callback;
>+  RtcpStatistics stats_from_callback;
>+  EXPECT_CALL(callback, StatisticsUpdated(_, _))
>+      .WillRepeatedly(SaveArg<0>(&stats_from_callback));
>+  receive_statistics_->RegisterRtcpStatisticsCallback(&callback);
>+
>+  // Using units of milliseconds.
>+  header1_.payload_type_frequency = 1000;
>+
>+  // Regardless of initial timestamps, jitter should start at 0.
>   header1_.sequenceNumber = 1;
>   clock_.AdvanceTimeMilliseconds(7);
>   header1_.timestamp += 3;
>   receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
>-  header1_.sequenceNumber += 2;
>-  clock_.AdvanceTimeMilliseconds(9);
>-  header1_.timestamp += 9;
>+  EXPECT_EQ(0u, stats_from_callback.jitter);
>+
>+  // Incrementing timestamps by the same amount shouldn't increase jitter.
>+  ++header1_.sequenceNumber;
>+  clock_.AdvanceTimeMilliseconds(50);
>+  header1_.timestamp += 50;
>   receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
>-  --header1_.sequenceNumber;
>-  clock_.AdvanceTimeMilliseconds(13);
>-  header1_.timestamp += 47;
>+  EXPECT_EQ(0u, stats_from_callback.jitter);
>+
>+  // Difference of 16ms, divided by 16 yields exactly 1.
>+  ++header1_.sequenceNumber;
>+  clock_.AdvanceTimeMilliseconds(32);
>+  header1_.timestamp += 16;
>   receive_statistics_->IncomingPacket(header1_, kPacketSize1, true);
>-  header1_.sequenceNumber += 3;
>-  clock_.AdvanceTimeMilliseconds(11);
>-  header1_.timestamp += 17;
>-  receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
>+  EXPECT_EQ(1u, stats_from_callback.jitter);
>+
>+  // (90 + 1 * 15) / 16 = 6.5625; should round down to 6.
>+  // TODO(deadbeef): Why don't we round to the nearest integer?
>   ++header1_.sequenceNumber;
>+  clock_.AdvanceTimeMilliseconds(10);
>+  header1_.timestamp += 100;
>+  receive_statistics_->IncomingPacket(header1_, kPacketSize1, true);
>+  EXPECT_EQ(6u, stats_from_callback.jitter);
>+
>+  // (30 + 6.5625 * 15) / 16 = 8.0273; should round down to 8.
>+  ++header1_.sequenceNumber;
>+  clock_.AdvanceTimeMilliseconds(50);
>+  header1_.timestamp += 20;
>+  receive_statistics_->IncomingPacket(header1_, kPacketSize1, true);
>+  EXPECT_EQ(8u, stats_from_callback.jitter);
>+}
>+
>+// TODO(deadbeef): Why do we do this? It goes against RFC3550, which explicitly
>+// says the calculation should be based on order of arrival and packets may not
>+// necessarily arrive in sequence.
>+TEST_F(ReceiveStatisticsTest, JitterComputationIgnoresReorderedPackets) {
>+  MockRtcpCallback callback;
>+  RtcpStatistics stats_from_callback;
>+  EXPECT_CALL(callback, StatisticsUpdated(_, _))
>+      .WillRepeatedly(SaveArg<0>(&stats_from_callback));
>+  receive_statistics_->RegisterRtcpStatisticsCallback(&callback);
> 
>-  receive_statistics_->GetStatistician(kSsrc1)
>-      ->GetStatistics(&statistics, true);
>+  // Using units of milliseconds.
>+  header1_.payload_type_frequency = 1000;
> 
>-  // Should not have been called after deregister.
>-  EXPECT_EQ(1u, callback.num_calls_);
>+  // Regardless of initial timestamps, jitter should start at 0.
>+  header1_.sequenceNumber = 1;
>+  receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
>+  EXPECT_EQ(0u, stats_from_callback.jitter);
>+
>+  // This should be ignored, even though there's a difference of 70 here.
>+  header1_.sequenceNumber = 0;
>+  clock_.AdvanceTimeMilliseconds(50);
>+  header1_.timestamp -= 20;
>+  receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
>+  EXPECT_EQ(0u, stats_from_callback.jitter);
>+
>+  // Relative to the first packet there's a difference of 181ms in arrival
>+  // time, 20ms in timestamp, so jitter should be 161/16 = 10.
>+  header1_.sequenceNumber = 2;
>+  clock_.AdvanceTimeMilliseconds(131);
>+  header1_.timestamp += 40;
>+  receive_statistics_->IncomingPacket(header1_, kPacketSize1, false);
>+  EXPECT_EQ(10u, stats_from_callback.jitter);
> }
> 
> class RtpTestCallback : public StreamDataCountersCallback {
>  public:
>   RtpTestCallback()
>       : StreamDataCountersCallback(), num_calls_(0), ssrc_(0), stats_() {}
>-  virtual ~RtpTestCallback() {}
>+  ~RtpTestCallback() override = default;
> 
>-  virtual void DataCountersUpdated(const StreamDataCounters& counters,
>-                                   uint32_t ssrc) {
>+  void DataCountersUpdated(const StreamDataCounters& counters,
>+                           uint32_t ssrc) override {
>     ssrc_ = ssrc;
>     stats_ = counters;
>     ++num_calls_;
>@@ -324,8 +708,8 @@ TEST_F(ReceiveStatisticsTest, RtpCallbacks) {
> 
>   // One packet of size kPacketSize1.
>   header1_.headerLength = kHeaderLength;
>-  receive_statistics_->IncomingPacket(
>-      header1_, kPacketSize1 + kHeaderLength, false);
>+  receive_statistics_->IncomingPacket(header1_, kPacketSize1 + kHeaderLength,
>+                                      false);
>   StreamDataCounters expected;
>   expected.transmitted.payload_bytes = kPacketSize1;
>   expected.transmitted.header_bytes = kHeaderLength;
>@@ -368,8 +752,8 @@ TEST_F(ReceiveStatisticsTest, RtpCallbacks) {
>   ++header1_.sequenceNumber;
>   clock_.AdvanceTimeMilliseconds(5);
>   // One FEC packet.
>-  receive_statistics_->IncomingPacket(
>-      header1_, kPacketSize1 + kHeaderLength, false);
>+  receive_statistics_->IncomingPacket(header1_, kPacketSize1 + kHeaderLength,
>+                                      false);
>   receive_statistics_->FecPacketReceived(header1_,
>                                          kPacketSize1 + kHeaderLength);
>   expected.transmitted.payload_bytes = kPacketSize1 * 4;
>@@ -385,8 +769,8 @@ TEST_F(ReceiveStatisticsTest, RtpCallbacks) {
>   // New stats, but callback should not be called.
>   ++header1_.sequenceNumber;
>   clock_.AdvanceTimeMilliseconds(5);
>-  receive_statistics_->IncomingPacket(
>-      header1_, kPacketSize1 + kHeaderLength, true);
>+  receive_statistics_->IncomingPacket(header1_, kPacketSize1 + kHeaderLength,
>+                                      true);
>   callback.Matches(5, kSsrc1, expected);
> }
> 
>@@ -402,8 +786,8 @@ TEST_F(ReceiveStatisticsTest, RtpCallbacksFecFirst) {
>                                          kPacketSize1 + kHeaderLength);
>   EXPECT_EQ(0u, callback.num_calls_);
> 
>-  receive_statistics_->IncomingPacket(
>-      header1_, kPacketSize1 + kHeaderLength, false);
>+  receive_statistics_->IncomingPacket(header1_, kPacketSize1 + kHeaderLength,
>+                                      false);
>   StreamDataCounters expected;
>   expected.transmitted.payload_bytes = kPacketSize1;
>   expected.transmitted.header_bytes = kHeaderLength;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator.cc
>index e3a4ed91e53..fc867a491a3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator.cc
>@@ -11,6 +11,7 @@
> #include "modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
> 
> #include "rtc_base/logging.h"
>+#include "rtc_base/time/timestamp_extrapolator.h"
> #include "system_wrappers/include/clock.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator_unittest.cc
>index 24eb46f767c..5254cd57bb9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/remote_ntp_time_estimator_unittest.cc
>@@ -1,12 +1,12 @@
> /*
>-*  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
>-*
>-*  Use of this source code is governed by a BSD-style license
>-*  that can be found in the LICENSE file in the root of the source
>-*  tree. An additional intellectual property rights grant can be found
>-*  in the file PATENTS.  All contributing project authors may
>-*  be found in the AUTHORS file in the root of the source tree.
>-*/
>+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
> 
> #include "modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
> #include "common_types.h"  // NOLINT(build/include)
>@@ -32,7 +32,7 @@ class RemoteNtpTimeEstimatorTest : public ::testing::Test {
>       : local_clock_(kLocalClockInitialTimeMs * 1000),
>         remote_clock_(kRemoteClockInitialTimeMs * 1000),
>         estimator_(new RemoteNtpTimeEstimator(&local_clock_)) {}
>-  ~RemoteNtpTimeEstimatorTest() {}
>+  ~RemoteNtpTimeEstimatorTest() override = default;
> 
>   void AdvanceTimeMilliseconds(int64_t ms) {
>     local_clock_.AdvanceTimeMilliseconds(ms);
>@@ -63,8 +63,11 @@ class RemoteNtpTimeEstimatorTest : public ::testing::Test {
>     ReceiveRtcpSr(kTestRtt, rtcp_timestamp, ntp.seconds(), ntp.fractions());
>   }
> 
>-  void UpdateRtcpTimestamp(int64_t rtt, uint32_t ntp_secs, uint32_t ntp_frac,
>-                           uint32_t rtp_timestamp, bool expected_result) {
>+  void UpdateRtcpTimestamp(int64_t rtt,
>+                           uint32_t ntp_secs,
>+                           uint32_t ntp_frac,
>+                           uint32_t rtp_timestamp,
>+                           bool expected_result) {
>     EXPECT_EQ(expected_result, estimator_->UpdateRtcpTimestamp(
>                                    rtt, ntp_secs, ntp_frac, rtp_timestamp));
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_nack_stats.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_nack_stats.cc
>index 8700b045b9a..24b708502ee 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_nack_stats.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_nack_stats.cc
>@@ -15,14 +15,12 @@
> namespace webrtc {
> 
> RtcpNackStats::RtcpNackStats()
>-    : max_sequence_number_(0),
>-      requests_(0),
>-      unique_requests_(0) {}
>+    : max_sequence_number_(0), requests_(0), unique_requests_(0) {}
> 
> void RtcpNackStats::ReportRequest(uint16_t sequence_number) {
>   if (requests_ == 0 ||
>       IsNewerSequenceNumber(sequence_number, max_sequence_number_)) {
>-    max_sequence_number_ =  sequence_number;
>+    max_sequence_number_ = sequence_number;
>     ++unique_requests_;
>   }
>   ++requests_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_nack_stats.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_nack_stats.h
>index d9e2622abc5..9da4351a59b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_nack_stats.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_nack_stats.h
>@@ -1,12 +1,12 @@
> /*
>-*  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>-*
>-*  Use of this source code is governed by a BSD-style license
>-*  that can be found in the LICENSE file in the root of the source
>-*  tree. An additional intellectual property rights grant can be found
>-*  in the file PATENTS.  All contributing project authors may
>-*  be found in the AUTHORS file in the root of the source tree.
>-*/
>+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
> 
> #ifndef MODULES_RTP_RTCP_SOURCE_RTCP_NACK_STATS_H_
> #define MODULES_RTP_RTCP_SOURCE_RTCP_NACK_STATS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_nack_stats_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_nack_stats_unittest.cc
>index 5b4def5476c..1c3017375d8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_nack_stats_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_nack_stats_unittest.cc
>@@ -61,4 +61,3 @@ TEST(RtcpNackStatsTest, RequestsWithWrap) {
> }
> 
> }  // namespace webrtc
>-
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc
>index ecce5ef34c5..194b992ef8f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet.cc
>@@ -75,8 +75,8 @@ void RtcpPacket::CreateHeader(
>   RTC_DCHECK_LE(count_or_format, 0x1f);
>   constexpr uint8_t kVersionBits = 2 << 6;
>   constexpr uint8_t kNoPaddingBit = 0 << 5;
>-  buffer[*pos + 0] = kVersionBits | kNoPaddingBit |
>-                     static_cast<uint8_t>(count_or_format);
>+  buffer[*pos + 0] =
>+      kVersionBits | kNoPaddingBit | static_cast<uint8_t>(count_or_format);
>   buffer[*pos + 1] = packet_type;
>   buffer[*pos + 2] = (length >> 8) & 0xff;
>   buffer[*pos + 3] = length & 0xff;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet.h
>index e773398553f..11037cbd82d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet.h
>@@ -11,7 +11,6 @@
> #ifndef MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_H_
> #define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_H_
> 
>-#include "rtc_base/basictypes.h"
> #include "rtc_base/buffer.h"
> #include "rtc_base/function_view.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/app.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/app.cc
>index 10d2a771fee..4e21bc93e48 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/app.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/app.cc
>@@ -64,9 +64,9 @@ void App::SetSubType(uint8_t subtype) {
> void App::SetData(const uint8_t* data, size_t data_length) {
>   RTC_DCHECK(data);
>   RTC_DCHECK_EQ(data_length % 4, 0) << "Data must be 32 bits aligned.";
>-  RTC_DCHECK_LE(data_length, kMaxDataSize) << "App data size " << data_length
>-                                           << " exceed maximum of "
>-                                           << kMaxDataSize << " bytes.";
>+  RTC_DCHECK_LE(data_length, kMaxDataSize)
>+      << "App data size " << data_length << " exceed maximum of "
>+      << kMaxDataSize << " bytes.";
>   data_.SetData(data, data_length);
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/common_header.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/common_header.h
>index b760d169f7c..54164060910 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/common_header.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/common_header.h
>@@ -10,7 +10,8 @@
> #ifndef MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_COMMON_HEADER_H_
> #define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_COMMON_HEADER_H_
> 
>-#include "rtc_base/basictypes.h"
>+#include <stddef.h>
>+#include <stdint.h>
> 
> namespace webrtc {
> namespace rtcp {
>@@ -20,7 +21,7 @@ class CommonHeader {
> 
>   CommonHeader() {}
>   CommonHeader(const CommonHeader&) = default;
>-  CommonHeader& operator =(const CommonHeader&) = default;
>+  CommonHeader& operator=(const CommonHeader&) = default;
> 
>   bool Parse(const uint8_t* buffer, size_t size_bytes);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/common_header_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/common_header_unittest.cc
>index a2846424515..e8b4c52c68a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/common_header_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/common_header_unittest.cc
>@@ -42,9 +42,8 @@ TEST(RtcpCommonHeaderTest, Version) {
> }
> 
> TEST(RtcpCommonHeaderTest, PacketSize) {
>-  uint8_t buffer[] = {0x80, 0x00, 0x00, 0x02,
>-                      0x00, 0x00, 0x00, 0x00,
>-                      0x00, 0x00, 0x00, 0x00};
>+  uint8_t buffer[] = {0x80, 0x00, 0x00, 0x02, 0x00, 0x00,
>+                      0x00, 0x00, 0x00, 0x00, 0x00, 0x00};
>   CommonHeader header;
>   EXPECT_FALSE(header.Parse(buffer, sizeof(buffer) - 1));
>   EXPECT_TRUE(header.Parse(buffer, sizeof(buffer)));
>@@ -55,9 +54,8 @@ TEST(RtcpCommonHeaderTest, PacketSize) {
> 
> TEST(RtcpCommonHeaderTest, PaddingAndPayloadSize) {
>   // Set v = 2, p = 1, but leave fmt, pt as 0.
>-  uint8_t buffer[] = {0xa0, 0x00, 0x00, 0x00,
>-                      0x00, 0x00, 0x00, 0x00,
>-                      0x00, 0x00, 0x00, 0x00};
>+  uint8_t buffer[] = {0xa0, 0x00, 0x00, 0x00, 0x00, 0x00,
>+                      0x00, 0x00, 0x00, 0x00, 0x00, 0x00};
>   CommonHeader header;
>   // Padding bit set, but no byte for padding (can't specify padding length).
>   EXPECT_FALSE(header.Parse(buffer, 4));
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/compound_packet.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/compound_packet.h
>index fdf0a4fa9a1..89b78148f7b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/compound_packet.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/compound_packet.h
>@@ -15,7 +15,6 @@
> #include <vector>
> 
> #include "modules/rtp_rtcp/source/rtcp_packet.h"
>-#include "rtc_base/basictypes.h"
> #include "rtc_base/constructormagic.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/dlrr.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/dlrr.h
>index f0f23022f53..6fe2099fd91 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/dlrr.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/dlrr.h
>@@ -12,9 +12,10 @@
> #ifndef MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_DLRR_H_
> #define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_DLRR_H_
> 
>-#include <vector>
>+#include <stddef.h>
>+#include <stdint.h>
> 
>-#include "rtc_base/basictypes.h"
>+#include <vector>
> 
> namespace webrtc {
> namespace rtcp {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.cc
>index d9ee5997ca4..06472ef4835 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.cc
>@@ -18,6 +18,7 @@
> namespace webrtc {
> namespace rtcp {
> constexpr uint8_t ExtendedReports::kPacketType;
>+constexpr size_t ExtendedReports::kMaxNumberOfDlrrItems;
> // From RFC 3611: RTP Control Protocol Extended Reports (RTCP XR).
> //
> // Format for XR packets:
>@@ -56,7 +57,7 @@ bool ExtendedReports::Parse(const CommonHeader& packet) {
>   rrtr_block_.reset();
>   dlrr_block_.ClearItems();
>   voip_metric_block_.reset();
>-  target_bitrate_ = rtc::nullopt;
>+  target_bitrate_ = absl::nullopt;
> 
>   const uint8_t* current_block = packet.payload() + kXrBaseLength;
>   const uint8_t* const packet_end =
>@@ -104,8 +105,13 @@ void ExtendedReports::SetRrtr(const Rrtr& rrtr) {
>   rrtr_block_.emplace(rrtr);
> }
> 
>-void ExtendedReports::AddDlrrItem(const ReceiveTimeInfo& time_info) {
>+bool ExtendedReports::AddDlrrItem(const ReceiveTimeInfo& time_info) {
>+  if (dlrr_block_.sub_blocks().size() >= kMaxNumberOfDlrrItems) {
>+    RTC_LOG(LS_WARNING) << "Reached maximum number of DLRR items.";
>+    return false;
>+  }
>   dlrr_block_.AddDlrrItem(time_info);
>+  return true;
> }
> 
> void ExtendedReports::SetVoipMetric(const VoipMetric& voip_metric) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.h
>index a1da4dc01f4..78555022c63 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.h
>@@ -13,7 +13,7 @@
> 
> #include <vector>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> #include "modules/rtp_rtcp/source/rtcp_packet.h"
> #include "modules/rtp_rtcp/source/rtcp_packet/dlrr.h"
> #include "modules/rtp_rtcp/source/rtcp_packet/rrtr.h"
>@@ -28,6 +28,7 @@ class CommonHeader;
> class ExtendedReports : public RtcpPacket {
>  public:
>   static constexpr uint8_t kPacketType = 207;
>+  static constexpr size_t kMaxNumberOfDlrrItems = 50;
> 
>   ExtendedReports();
>   ~ExtendedReports() override;
>@@ -38,17 +39,17 @@ class ExtendedReports : public RtcpPacket {
>   void SetSenderSsrc(uint32_t ssrc) { sender_ssrc_ = ssrc; }
> 
>   void SetRrtr(const Rrtr& rrtr);
>-  void AddDlrrItem(const ReceiveTimeInfo& time_info);
>+  bool AddDlrrItem(const ReceiveTimeInfo& time_info);
>   void SetVoipMetric(const VoipMetric& voip_metric);
>   void SetTargetBitrate(const TargetBitrate& target_bitrate);
> 
>   uint32_t sender_ssrc() const { return sender_ssrc_; }
>-  const rtc::Optional<Rrtr>& rrtr() const { return rrtr_block_; }
>+  const absl::optional<Rrtr>& rrtr() const { return rrtr_block_; }
>   const Dlrr& dlrr() const { return dlrr_block_; }
>-  const rtc::Optional<VoipMetric>& voip_metric() const {
>+  const absl::optional<VoipMetric>& voip_metric() const {
>     return voip_metric_block_;
>   }
>-  const rtc::Optional<TargetBitrate>& target_bitrate() const {
>+  const absl::optional<TargetBitrate>& target_bitrate() const {
>     return target_bitrate_;
>   }
> 
>@@ -75,10 +76,10 @@ class ExtendedReports : public RtcpPacket {
>   void ParseTargetBitrateBlock(const uint8_t* block, uint16_t block_length);
> 
>   uint32_t sender_ssrc_;
>-  rtc::Optional<Rrtr> rrtr_block_;
>+  absl::optional<Rrtr> rrtr_block_;
>   Dlrr dlrr_block_;  // Dlrr without items treated same as no dlrr block.
>-  rtc::Optional<VoipMetric> voip_metric_block_;
>-  rtc::Optional<TargetBitrate> target_bitrate_;
>+  absl::optional<VoipMetric> voip_metric_block_;
>+  absl::optional<TargetBitrate> target_bitrate_;
> };
> }  // namespace rtcp
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports_unittest.cc
>index 088b363c54d..c37f77e812b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports_unittest.cc
>@@ -18,6 +18,7 @@
> using testing::ElementsAre;
> using testing::ElementsAreArray;
> using testing::make_tuple;
>+using testing::SizeIs;
> using webrtc::rtcp::Dlrr;
> using webrtc::rtcp::ExtendedReports;
> using webrtc::rtcp::ReceiveTimeInfo;
>@@ -38,16 +39,13 @@ bool operator==(const RTCPVoIPMetric& metric1, const RTCPVoIPMetric& metric2) {
>          metric1.endSystemDelay == metric2.endSystemDelay &&
>          metric1.signalLevel == metric2.signalLevel &&
>          metric1.noiseLevel == metric2.noiseLevel &&
>-         metric1.RERL == metric2.RERL &&
>-         metric1.Gmin == metric2.Gmin &&
>+         metric1.RERL == metric2.RERL && metric1.Gmin == metric2.Gmin &&
>          metric1.Rfactor == metric2.Rfactor &&
>          metric1.extRfactor == metric2.extRfactor &&
>-         metric1.MOSLQ == metric2.MOSLQ &&
>-         metric1.MOSCQ == metric2.MOSCQ &&
>+         metric1.MOSLQ == metric2.MOSLQ && metric1.MOSCQ == metric2.MOSCQ &&
>          metric1.RXconfig == metric2.RXconfig &&
>          metric1.JBnominal == metric2.JBnominal &&
>-         metric1.JBmax == metric2.JBmax &&
>-         metric1.JBabsMax == metric2.JBabsMax;
>+         metric1.JBmax == metric2.JBmax && metric1.JBabsMax == metric2.JBabsMax;
> }
> 
> namespace rtcp {
>@@ -56,8 +54,7 @@ bool operator==(const Rrtr& rrtr1, const Rrtr& rrtr2) {
> }
> 
> bool operator==(const ReceiveTimeInfo& time1, const ReceiveTimeInfo& time2) {
>-  return time1.ssrc == time2.ssrc &&
>-         time1.last_rr == time2.last_rr &&
>+  return time1.ssrc == time2.ssrc && time1.last_rr == time2.last_rr &&
>          time1.delay_since_last_rr == time2.delay_since_last_rr;
> }
> 
>@@ -112,26 +109,26 @@ Rrtr RtcpPacketExtendedReportsTest::Rand<Rrtr>() {
> template <>
> RTCPVoIPMetric RtcpPacketExtendedReportsTest::Rand<RTCPVoIPMetric>() {
>   RTCPVoIPMetric metric;
>-  metric.lossRate       = Rand<uint8_t>();
>-  metric.discardRate    = Rand<uint8_t>();
>-  metric.burstDensity   = Rand<uint8_t>();
>-  metric.gapDensity     = Rand<uint8_t>();
>-  metric.burstDuration  = Rand<uint16_t>();
>-  metric.gapDuration    = Rand<uint16_t>();
>+  metric.lossRate = Rand<uint8_t>();
>+  metric.discardRate = Rand<uint8_t>();
>+  metric.burstDensity = Rand<uint8_t>();
>+  metric.gapDensity = Rand<uint8_t>();
>+  metric.burstDuration = Rand<uint16_t>();
>+  metric.gapDuration = Rand<uint16_t>();
>   metric.roundTripDelay = Rand<uint16_t>();
>   metric.endSystemDelay = Rand<uint16_t>();
>-  metric.signalLevel    = Rand<uint8_t>();
>-  metric.noiseLevel     = Rand<uint8_t>();
>-  metric.RERL           = Rand<uint8_t>();
>-  metric.Gmin           = Rand<uint8_t>();
>-  metric.Rfactor        = Rand<uint8_t>();
>-  metric.extRfactor     = Rand<uint8_t>();
>-  metric.MOSLQ          = Rand<uint8_t>();
>-  metric.MOSCQ          = Rand<uint8_t>();
>-  metric.RXconfig       = Rand<uint8_t>();
>-  metric.JBnominal      = Rand<uint16_t>();
>-  metric.JBmax          = Rand<uint16_t>();
>-  metric.JBabsMax       = Rand<uint16_t>();
>+  metric.signalLevel = Rand<uint8_t>();
>+  metric.noiseLevel = Rand<uint8_t>();
>+  metric.RERL = Rand<uint8_t>();
>+  metric.Gmin = Rand<uint8_t>();
>+  metric.Rfactor = Rand<uint8_t>();
>+  metric.extRfactor = Rand<uint8_t>();
>+  metric.MOSLQ = Rand<uint8_t>();
>+  metric.MOSCQ = Rand<uint8_t>();
>+  metric.RXconfig = Rand<uint8_t>();
>+  metric.JBnominal = Rand<uint16_t>();
>+  metric.JBmax = Rand<uint16_t>();
>+  metric.JBabsMax = Rand<uint16_t>();
>   return metric;
> }
> 
>@@ -211,6 +208,18 @@ TEST_F(RtcpPacketExtendedReportsTest, CreateAndParseWithDlrrWithTwoSubBlocks) {
>   EXPECT_THAT(parsed.dlrr().sub_blocks(), ElementsAre(kTimeInfo1, kTimeInfo2));
> }
> 
>+TEST_F(RtcpPacketExtendedReportsTest, CreateLimitsTheNumberOfDlrrSubBlocks) {
>+  const ReceiveTimeInfo kTimeInfo = Rand<ReceiveTimeInfo>();
>+  ExtendedReports xr;
>+
>+  for (size_t i = 0; i < ExtendedReports::kMaxNumberOfDlrrItems; ++i)
>+    EXPECT_TRUE(xr.AddDlrrItem(kTimeInfo));
>+  EXPECT_FALSE(xr.AddDlrrItem(kTimeInfo));
>+
>+  EXPECT_THAT(xr.dlrr().sub_blocks(),
>+              SizeIs(ExtendedReports::kMaxNumberOfDlrrItems));
>+}
>+
> TEST_F(RtcpPacketExtendedReportsTest, CreateAndParseWithVoipMetric) {
>   const VoipMetric kVoipMetric = Rand<VoipMetric>();
> 
>@@ -228,15 +237,15 @@ TEST_F(RtcpPacketExtendedReportsTest, CreateAndParseWithVoipMetric) {
>   EXPECT_EQ(kVoipMetric, parsed.voip_metric());
> }
> 
>-TEST_F(RtcpPacketExtendedReportsTest, CreateAndParseWithMultipleReportBlocks) {
>+TEST_F(RtcpPacketExtendedReportsTest, CreateAndParseWithMaximumReportBlocks) {
>   const Rrtr kRrtr = Rand<Rrtr>();
>-  const ReceiveTimeInfo kTimeInfo = Rand<ReceiveTimeInfo>();
>   const VoipMetric kVoipMetric = Rand<VoipMetric>();
> 
>   ExtendedReports xr;
>   xr.SetSenderSsrc(kSenderSsrc);
>   xr.SetRrtr(kRrtr);
>-  xr.AddDlrrItem(kTimeInfo);
>+  for (size_t i = 0; i < ExtendedReports::kMaxNumberOfDlrrItems; ++i)
>+    xr.AddDlrrItem(Rand<ReceiveTimeInfo>());
>   xr.SetVoipMetric(kVoipMetric);
> 
>   rtc::Buffer packet = xr.Build();
>@@ -247,7 +256,8 @@ TEST_F(RtcpPacketExtendedReportsTest, CreateAndParseWithMultipleReportBlocks) {
> 
>   EXPECT_EQ(kSenderSsrc, parsed.sender_ssrc());
>   EXPECT_EQ(kRrtr, parsed.rrtr());
>-  EXPECT_THAT(parsed.dlrr().sub_blocks(), ElementsAre(kTimeInfo));
>+  EXPECT_THAT(parsed.dlrr().sub_blocks(),
>+              ElementsAreArray(xr.dlrr().sub_blocks()));
>   EXPECT_EQ(kVoipMetric, parsed.voip_metric());
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/fir.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/fir.h
>index b37c4dacd72..6fbc54c7f22 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/fir.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/fir.h
>@@ -14,7 +14,6 @@
> #include <vector>
> 
> #include "modules/rtp_rtcp/source/rtcp_packet/psfb.h"
>-#include "rtc_base/basictypes.h"
> 
> namespace webrtc {
> namespace rtcp {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/fir_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/fir_unittest.cc
>index 2800fbc79e2..d9eb465fd97 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/fir_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/fir_unittest.cc
>@@ -29,11 +29,9 @@ constexpr uint32_t kSenderSsrc = 0x12345678;
> constexpr uint32_t kRemoteSsrc = 0x23456789;
> constexpr uint8_t kSeqNr = 13;
> // Manually created Fir packet matching constants above.
>-constexpr uint8_t kPacket[] = {0x84,  206, 0x00, 0x04,
>-                               0x12, 0x34, 0x56, 0x78,
>-                               0x00, 0x00, 0x00, 0x00,
>-                               0x23, 0x45, 0x67, 0x89,
>-                               0x0d, 0x00, 0x00, 0x00};
>+constexpr uint8_t kPacket[] = {0x84, 206,  0x00, 0x04, 0x12, 0x34, 0x56,
>+                               0x78, 0x00, 0x00, 0x00, 0x00, 0x23, 0x45,
>+                               0x67, 0x89, 0x0d, 0x00, 0x00, 0x00};
> }  // namespace
> 
> TEST(RtcpPacketFirTest, Parse) {
>@@ -77,20 +75,16 @@ TEST(RtcpPacketFirTest, TwoFciEntries) {
> }
> 
> TEST(RtcpPacketFirTest, ParseFailsOnZeroFciEntries) {
>-  constexpr uint8_t kPacketWithoutFci[] = {0x84,  206, 0x00, 0x02,
>-                                           0x12, 0x34, 0x56, 0x78,
>-                                           0x00, 0x00, 0x00, 0x00};
>+  constexpr uint8_t kPacketWithoutFci[] = {0x84, 206,  0x00, 0x02, 0x12, 0x34,
>+                                           0x56, 0x78, 0x00, 0x00, 0x00, 0x00};
>   Fir parsed;
>   EXPECT_FALSE(test::ParseSinglePacket(kPacketWithoutFci, &parsed));
> }
> 
> TEST(RtcpPacketFirTest, ParseFailsOnFractionalFciEntries) {
>-  constexpr uint8_t kPacketWithOneAndHalfFci[] = {0x84,  206, 0x00, 0x05,
>-                                                  0x12, 0x34, 0x56, 0x78,
>-                                                  0x00, 0x00, 0x00, 0x00,
>-                                                  0x23, 0x45, 0x67, 0x89,
>-                                                  0x0d, 0x00, 0x00, 0x00,
>-                                                   'h',  'a',  'l',  'f'};
>+  constexpr uint8_t kPacketWithOneAndHalfFci[] = {
>+      0x84, 206,  0x00, 0x05, 0x12, 0x34, 0x56, 0x78, 0x00, 0x00, 0x00, 0x00,
>+      0x23, 0x45, 0x67, 0x89, 0x0d, 0x00, 0x00, 0x00, 'h',  'a',  'l',  'f'};
> 
>   Fir parsed;
>   EXPECT_FALSE(test::ParseSinglePacket(kPacketWithOneAndHalfFci, &parsed));
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/nack.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/nack.h
>index cab77d4ff12..0e462f98925 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/nack.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/nack.h
>@@ -14,7 +14,6 @@
> #include <vector>
> 
> #include "modules/rtp_rtcp/source/rtcp_packet/rtpfb.h"
>-#include "rtc_base/basictypes.h"
> 
> namespace webrtc {
> namespace rtcp {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/pli.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/pli.h
>index e64b877a4ad..fc3c20a8a85 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/pli.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/pli.h
>@@ -11,7 +11,6 @@
> #define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_PLI_H_
> 
> #include "modules/rtp_rtcp/source/rtcp_packet/psfb.h"
>-#include "rtc_base/basictypes.h"
> 
> namespace webrtc {
> namespace rtcp {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/pli_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/pli_unittest.cc
>index e6cf7bef4a1..67e614dd3c0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/pli_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/pli_unittest.cc
>@@ -23,9 +23,8 @@ namespace {
> const uint32_t kSenderSsrc = 0x12345678;
> const uint32_t kRemoteSsrc = 0x23456789;
> // Manually created Pli packet matching constants above.
>-const uint8_t kPacket[] = {0x81, 206,  0x00, 0x02,
>-                           0x12, 0x34, 0x56, 0x78,
>-                           0x23, 0x45, 0x67, 0x89};
>+const uint8_t kPacket[] = {0x81, 206,  0x00, 0x02, 0x12, 0x34,
>+                           0x56, 0x78, 0x23, 0x45, 0x67, 0x89};
> }  // namespace
> 
> TEST(RtcpPacketPliTest, Parse) {
>@@ -45,7 +44,7 @@ TEST(RtcpPacketPliTest, Create) {
>   rtc::Buffer packet = pli.Build();
> 
>   EXPECT_THAT(make_tuple(packet.data(), packet.size()),
>-      ElementsAreArray(kPacket));
>+              ElementsAreArray(kPacket));
> }
> 
> TEST(RtcpPacketPliTest, ParseFailsOnTooSmallPacket) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.h
>index 33b7f1c9ef9..ae66a172387 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.h
>@@ -13,7 +13,6 @@
> #define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_PSFB_H_
> 
> #include "modules/rtp_rtcp/source/rtcp_packet.h"
>-#include "rtc_base/basictypes.h"
> 
> namespace webrtc {
> namespace rtcp {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h
>index 9f1772ad1a8..1955b98f5c0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h
>@@ -12,7 +12,6 @@
> #define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RAPID_RESYNC_REQUEST_H_
> 
> #include "modules/rtp_rtcp/source/rtcp_packet/rtpfb.h"
>-#include "rtc_base/basictypes.h"
> 
> namespace webrtc {
> namespace rtcp {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request_unittest.cc
>index e80736900d9..f10f93ea209 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request_unittest.cc
>@@ -23,9 +23,8 @@ namespace {
> const uint32_t kSenderSsrc = 0x12345678;
> const uint32_t kRemoteSsrc = 0x23456789;
> // Manually created packet matching constants above.
>-const uint8_t kPacket[] = {0x85, 205,  0x00, 0x02,
>-                           0x12, 0x34, 0x56, 0x78,
>-                           0x23, 0x45, 0x67, 0x89};
>+const uint8_t kPacket[] = {0x85, 205,  0x00, 0x02, 0x12, 0x34,
>+                           0x56, 0x78, 0x23, 0x45, 0x67, 0x89};
> }  // namespace
> 
> TEST(RtcpPacketRapidResyncRequestTest, Parse) {
>@@ -56,9 +55,8 @@ TEST(RtcpPacketRapidResyncRequestTest, ParseFailsOnTooSmallPacket) {
> }
> 
> TEST(RtcpPacketRapidResyncRequestTest, ParseFailsOnTooLargePacket) {
>-  const uint8_t kTooLargePacket[] = {0x85, 205,  0x00, 0x03,
>-                                     0x12, 0x34, 0x56, 0x78,
>-                                     0x32, 0x21, 0x65, 0x87,
>+  const uint8_t kTooLargePacket[] = {0x85, 205,  0x00, 0x03, 0x12, 0x34,
>+                                     0x56, 0x78, 0x32, 0x21, 0x65, 0x87,
>                                      0x23, 0x45, 0x67, 0x89};
>   RapidResyncRequest parsed;
>   EXPECT_FALSE(test::ParseSinglePacket(kTooLargePacket, &parsed));
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h
>index 894e14e5c3e..3dc79205856 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h
>@@ -15,7 +15,6 @@
> 
> #include "modules/rtp_rtcp/source/rtcp_packet.h"
> #include "modules/rtp_rtcp/source/rtcp_packet/report_block.h"
>-#include "rtc_base/basictypes.h"
> 
> namespace webrtc {
> namespace rtcp {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/remb.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/remb.h
>index b034a964fd0..5f4aef8ea4d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/remb.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/remb.h
>@@ -14,7 +14,6 @@
> #include <vector>
> 
> #include "modules/rtp_rtcp/source/rtcp_packet/psfb.h"
>-#include "rtc_base/basictypes.h"
> 
> namespace webrtc {
> namespace rtcp {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/report_block.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/report_block.cc
>index 4f989637666..d4579fc8d65 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/report_block.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/report_block.cc
>@@ -54,7 +54,7 @@ bool ReportBlock::Parse(const uint8_t* buffer, size_t length) {
> 
>   source_ssrc_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[0]);
>   fraction_lost_ = buffer[4];
>-  cumulative_lost_ = ByteReader<uint32_t, 3>::ReadBigEndian(&buffer[5]);
>+  cumulative_lost_ = ByteReader<int32_t, 3>::ReadBigEndian(&buffer[5]);
>   extended_high_seq_num_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[8]);
>   jitter_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[12]);
>   last_sr_ = ByteReader<uint32_t>::ReadBigEndian(&buffer[16]);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/report_block.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/report_block.h
>index 766d59e26b6..eb16640ae28 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/report_block.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/report_block.h
>@@ -12,7 +12,8 @@
> #ifndef MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_REPORT_BLOCK_H_
> #define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_REPORT_BLOCK_H_
> 
>-#include "rtc_base/basictypes.h"
>+#include <stddef.h>
>+#include <stdint.h>
> 
> namespace webrtc {
> namespace rtcp {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/report_block_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/report_block_unittest.cc
>index a074a2f4f4d..5cc102fed0a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/report_block_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/report_block_unittest.cc
>@@ -91,5 +91,20 @@ TEST(RtcpPacketReportBlockTest, ValidateCumulativeLost) {
>   EXPECT_EQ(0u, rb.cumulative_lost());
> }
> 
>+TEST(RtcpPacketReportBlockTest, ParseNegativeCumulativeLost) {
>+  // CumulativeLost is a signed 24-bit integer.
>+  const int32_t kNegativeCumulativeLost = -123;
>+  ReportBlock rb;
>+  EXPECT_TRUE(rb.SetCumulativeLost(kNegativeCumulativeLost));
>+
>+  uint8_t buffer[kBufferLength];
>+  rb.Create(buffer);
>+
>+  ReportBlock parsed;
>+  EXPECT_TRUE(parsed.Parse(buffer, kBufferLength));
>+
>+  EXPECT_EQ(kNegativeCumulativeLost, parsed.cumulative_lost_signed());
>+}
>+
> }  // namespace
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/rrtr.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/rrtr.h
>index 9eae8c89229..a470b1a217c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/rrtr.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/rrtr.h
>@@ -12,7 +12,6 @@
> #ifndef MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RRTR_H_
> #define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RRTR_H_
> 
>-#include "rtc_base/basictypes.h"
> #include "system_wrappers/include/ntp_time.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/rrtr_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/rrtr_unittest.cc
>index 7bd79b311cc..56622ea81a7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/rrtr_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/rrtr_unittest.cc
>@@ -19,9 +19,8 @@ namespace {
> 
> const uint32_t kNtpSec = 0x12345678;
> const uint32_t kNtpFrac = 0x23456789;
>-const uint8_t kBlock[] = {0x04, 0x00, 0x00, 0x02,
>-                          0x12, 0x34, 0x56, 0x78,
>-                          0x23, 0x45, 0x67, 0x89};
>+const uint8_t kBlock[] = {0x04, 0x00, 0x00, 0x02, 0x12, 0x34,
>+                          0x56, 0x78, 0x23, 0x45, 0x67, 0x89};
> const size_t kBlockSizeBytes = sizeof(kBlock);
> static_assert(
>     kBlockSizeBytes == Rrtr::kLength,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/rtpfb.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/rtpfb.h
>index 4b08106af40..a0407411ab3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/rtpfb.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/rtpfb.h
>@@ -13,7 +13,6 @@
> #define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_RTPFB_H_
> 
> #include "modules/rtp_rtcp/source/rtcp_packet.h"
>-#include "rtc_base/basictypes.h"
> 
> namespace webrtc {
> namespace rtcp {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/sdes.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/sdes.h
>index 9010686b1ac..70e9385b9be 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/sdes.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/sdes.h
>@@ -15,7 +15,6 @@
> #include <vector>
> 
> #include "modules/rtp_rtcp/source/rtcp_packet.h"
>-#include "rtc_base/basictypes.h"
> 
> namespace webrtc {
> namespace rtcp {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/sdes_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/sdes_unittest.cc
>index 7896f2d0bc3..b903a4e3834 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/sdes_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/sdes_unittest.cc
>@@ -97,11 +97,10 @@ TEST(RtcpPacketSdesTest, CreateAndParseCnameItemWithEmptyString) {
> TEST(RtcpPacketSdesTest, ParseSkipsNonCNameField) {
>   const uint8_t kName[] = "abc";
>   const uint8_t kCname[] = "de";
>-  const uint8_t kValidPacket[] = {0x81,  202, 0x00, 0x04,
>-                                  0x12, 0x34, 0x56, 0x78,
>-                                  kNameTag,  3, kName[0],  kName[1], kName[2],
>-                                  kCnameTag, 2, kCname[0], kCname[1],
>-                                  kTerminatorTag, kPadding, kPadding};
>+  const uint8_t kValidPacket[] = {
>+      0x81, 202,       0x00,      0x04,           0x12,     0x34,     0x56,
>+      0x78, kNameTag,  3,         kName[0],       kName[1], kName[2], kCnameTag,
>+      2,    kCname[0], kCname[1], kTerminatorTag, kPadding, kPadding};
>   // Sanity checks packet was assembled correctly.
>   ASSERT_EQ(0u, sizeof(kValidPacket) % 4);
>   ASSERT_EQ(kValidPacket[3] + 1u, sizeof(kValidPacket) / 4);
>@@ -118,13 +117,14 @@ TEST(RtcpPacketSdesTest, ParseSkipsChunksWithoutCName) {
>   const uint8_t kName[] = "ab";
>   const uint8_t kEmail[] = "de";
>   const uint8_t kCname[] = "def";
>-  const uint8_t kPacket[] = {0x82,  202, 0x00, 0x07,
>-      0x12, 0x34, 0x56, 0x78,  // 1st chunk.
>-      kNameTag,  3, kName[0],  kName[1], kName[2],
>-      kEmailTag, 2, kEmail[0], kEmail[1],
>-      kTerminatorTag, kPadding, kPadding,
>-      0x23, 0x45, 0x67, 0x89,  // 2nd chunk.
>-      kCnameTag, 3, kCname[0], kCname[1], kCname[2],
>+  const uint8_t kPacket[] = {
>+      0x82,           202,      0x00,      0x07,      0x12,
>+      0x34,           0x56,     0x78,  // 1st chunk.
>+      kNameTag,       3,        kName[0],  kName[1],  kName[2],
>+      kEmailTag,      2,        kEmail[0], kEmail[1], kTerminatorTag,
>+      kPadding,       kPadding, 0x23,      0x45,      0x67,
>+      0x89,  // 2nd chunk.
>+      kCnameTag,      3,        kCname[0], kCname[1], kCname[2],
>       kTerminatorTag, kPadding, kPadding};
>   // Sanity checks packet was assembled correctly.
>   ASSERT_EQ(0u, sizeof(kPacket) % 4);
>@@ -141,10 +141,9 @@ TEST(RtcpPacketSdesTest, ParseFailsWithoutChunkItemTerminator) {
>   const uint8_t kName[] = "abc";
>   const uint8_t kCname[] = "d";
>   // No place for next chunk item.
>-  const uint8_t kInvalidPacket[] = {0x81,  202, 0x00, 0x03,
>-                                    0x12, 0x34, 0x56, 0x78,
>-                                    kNameTag,  3, kName[0], kName[1], kName[2],
>-                                    kCnameTag, 1, kCname[0]};
>+  const uint8_t kInvalidPacket[] = {
>+      0x81,     202, 0x00,     0x03,     0x12,     0x34,      0x56, 0x78,
>+      kNameTag, 3,   kName[0], kName[1], kName[2], kCnameTag, 1,    kCname[0]};
>   // Sanity checks packet was assembled correctly.
>   ASSERT_EQ(0u, sizeof(kInvalidPacket) % 4);
>   ASSERT_EQ(kInvalidPacket[3] + 1u, sizeof(kInvalidPacket) / 4);
>@@ -157,11 +156,9 @@ TEST(RtcpPacketSdesTest, ParseFailsWithDamagedChunkItem) {
>   const uint8_t kName[] = "ab";
>   const uint8_t kCname[] = "d";
>   // Next chunk item has non-terminator type, but not the size.
>-  const uint8_t kInvalidPacket[] = {0x81,  202, 0x00, 0x03,
>-                                    0x12, 0x34, 0x56, 0x78,
>-                                    kNameTag,  2, kName[0], kName[1],
>-                                    kCnameTag, 1, kCname[0],
>-                                    kEmailTag};
>+  const uint8_t kInvalidPacket[] = {
>+      0x81,     202, 0x00,     0x03,     0x12,      0x34, 0x56,      0x78,
>+      kNameTag, 2,   kName[0], kName[1], kCnameTag, 1,    kCname[0], kEmailTag};
>   // Sanity checks packet was assembled correctly.
>   ASSERT_EQ(0u, sizeof(kInvalidPacket) % 4);
>   ASSERT_EQ(kInvalidPacket[3] + 1u, sizeof(kInvalidPacket) / 4);
>@@ -174,10 +171,9 @@ TEST(RtcpPacketSdesTest, ParseFailsWithTooLongChunkItem) {
>   const uint8_t kName[] = "abc";
>   const uint8_t kCname[] = "d";
>   // Last chunk item has length that goes beyond the buffer end.
>-  const uint8_t kInvalidPacket[] = {0x81,  202, 0x00, 0x03,
>-                                    0x12, 0x34, 0x56, 0x78,
>-                                    kNameTag,  3, kName[0], kName[1], kName[2],
>-                                    kCnameTag, 2, kCname[0]};
>+  const uint8_t kInvalidPacket[] = {
>+      0x81,     202, 0x00,     0x03,     0x12,     0x34,      0x56, 0x78,
>+      kNameTag, 3,   kName[0], kName[1], kName[2], kCnameTag, 2,    kCname[0]};
>   // Sanity checks packet was assembled correctly.
>   ASSERT_EQ(0u, sizeof(kInvalidPacket) % 4);
>   ASSERT_EQ(kInvalidPacket[3] + 1u, sizeof(kInvalidPacket) / 4);
>@@ -189,11 +185,10 @@ TEST(RtcpPacketSdesTest, ParseFailsWithTooLongChunkItem) {
> TEST(RtcpPacketSdesTest, ParseFailsWithTwoCNames) {
>   const uint8_t kCname1[] = "a";
>   const uint8_t kCname2[] = "de";
>-  const uint8_t kInvalidPacket[] = {0x81,  202, 0x00, 0x03,
>-                                    0x12, 0x34, 0x56, 0x78,
>-                                    kCnameTag, 1, kCname1[0],
>-                                    kCnameTag, 2, kCname2[0], kCname2[1],
>-                                    kTerminatorTag};
>+  const uint8_t kInvalidPacket[] = {
>+      0x81,       202,           0x00, 0x03,       0x12,      0x34, 0x56,
>+      0x78,       kCnameTag,     1,    kCname1[0], kCnameTag, 2,    kCname2[0],
>+      kCname2[1], kTerminatorTag};
>   // Sanity checks packet was assembled correctly.
>   ASSERT_EQ(0u, sizeof(kInvalidPacket) % 4);
>   ASSERT_EQ(kInvalidPacket[3] + 1u, sizeof(kInvalidPacket) / 4);
>@@ -206,12 +201,11 @@ TEST(RtcpPacketSdesTest, ParseFailsWithTooLittleSpaceForNextChunk) {
>   const uint8_t kCname[] = "a";
>   const uint8_t kEmail[] = "de";
>   // Two chunks are promised in the header, but no place for the second chunk.
>-  const uint8_t kInvalidPacket[] = {0x82,  202, 0x00, 0x04,
>-                                    0x12, 0x34, 0x56, 0x78,  // 1st chunk.
>-                                    kCnameTag, 1, kCname[0],
>-                                    kEmailTag, 2, kEmail[0], kEmail[1],
>-                                    kTerminatorTag,
>-                                    0x23, 0x45, 0x67, 0x89};  // 2nd chunk.
>+  const uint8_t kInvalidPacket[] = {
>+      0x82,           202,  0x00,      0x04,      0x12, 0x34,      0x56,
>+      0x78,  // 1st chunk.
>+      kCnameTag,      1,    kCname[0], kEmailTag, 2,    kEmail[0], kEmail[1],
>+      kTerminatorTag, 0x23, 0x45,      0x67,      0x89};  // 2nd chunk.
>   // Sanity checks packet was assembled correctly.
>   ASSERT_EQ(0u, sizeof(kInvalidPacket) % 4);
>   ASSERT_EQ(kInvalidPacket[3] + 1u, sizeof(kInvalidPacket) / 4);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.cc
>index 444ba68a987..e524aeadaa5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.cc
>@@ -21,6 +21,7 @@ namespace webrtc {
> namespace rtcp {
> constexpr uint8_t SenderReport::kPacketType;
> constexpr size_t SenderReport::kMaxNumberOfReportBlocks;
>+constexpr size_t SenderReport::kSenderBaseLength;
> //    Sender report (SR) (RFC 3550).
> //     0                   1                   2                   3
> //     0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
>@@ -46,6 +47,10 @@ SenderReport::SenderReport()
>       sender_packet_count_(0),
>       sender_octet_count_(0) {}
> 
>+SenderReport::SenderReport(const SenderReport&) = default;
>+SenderReport::SenderReport(SenderReport&&) = default;
>+SenderReport& SenderReport::operator=(const SenderReport&) = default;
>+SenderReport& SenderReport::operator=(SenderReport&&) = default;
> SenderReport::~SenderReport() = default;
> 
> bool SenderReport::Parse(const CommonHeader& packet) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h
>index 844459a3a1b..f1ee5250da4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h
>@@ -27,6 +27,10 @@ class SenderReport : public RtcpPacket {
>   static constexpr size_t kMaxNumberOfReportBlocks = 0x1f;
> 
>   SenderReport();
>+  SenderReport(const SenderReport&);
>+  SenderReport(SenderReport&&);
>+  SenderReport& operator=(const SenderReport&);
>+  SenderReport& operator=(SenderReport&&);
>   ~SenderReport() override;
> 
>   // Parse assumes header is already parsed and validated.
>@@ -65,7 +69,7 @@ class SenderReport : public RtcpPacket {
>               PacketReadyCallback callback) const override;
> 
>  private:
>-  const size_t kSenderBaseLength = 24;
>+  static constexpr size_t kSenderBaseLength = 24;
> 
>   uint32_t sender_ssrc_;
>   NtpTime ntp_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report_unittest.cc
>index 30edc7ecdb3..a0c6e725f06 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report_unittest.cc
>@@ -29,13 +29,10 @@ const NtpTime kNtp(0x11121418, 0x22242628);
> const uint32_t kRtpTimestamp = 0x33343536;
> const uint32_t kPacketCount = 0x44454647;
> const uint32_t kOctetCount = 0x55565758;
>-const uint8_t kPacket[] = {0x80,  200, 0x00, 0x06,
>-                           0x12, 0x34, 0x56, 0x78,
>-                           0x11, 0x12, 0x14, 0x18,
>-                           0x22, 0x24, 0x26, 0x28,
>-                           0x33, 0x34, 0x35, 0x36,
>-                           0x44, 0x45, 0x46, 0x47,
>-                           0x55, 0x56, 0x57, 0x58};
>+const uint8_t kPacket[] = {0x80, 200,  0x00, 0x06, 0x12, 0x34, 0x56,
>+                           0x78, 0x11, 0x12, 0x14, 0x18, 0x22, 0x24,
>+                           0x26, 0x28, 0x33, 0x34, 0x35, 0x36, 0x44,
>+                           0x45, 0x46, 0x47, 0x55, 0x56, 0x57, 0x58};
> }  // namespace
> 
> TEST(RtcpPacketSenderReportTest, CreateWithoutReportBlocks) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/target_bitrate.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/target_bitrate.h
>index a3793e320b9..07e5da1a49d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/target_bitrate.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/target_bitrate.h
>@@ -11,9 +11,10 @@
> #ifndef MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_TARGET_BITRATE_H_
> #define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_TARGET_BITRATE_H_
> 
>-#include <vector>
>+#include <stddef.h>
>+#include <stdint.h>
> 
>-#include "rtc_base/basictypes.h"
>+#include <vector>
> 
> namespace webrtc {
> namespace rtcp {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/target_bitrate_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/target_bitrate_unittest.cc
>index 5fec75fe441..b16bb5beaa5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/target_bitrate_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/target_bitrate_unittest.cc
>@@ -67,7 +67,7 @@ TEST(TargetBitrateTest, FullPacket) {
>   rtcp::ExtendedReports xr;
>   EXPECT_TRUE(ParseSinglePacket(kRtcpPacket, &xr));
>   EXPECT_EQ(kSsrc, xr.sender_ssrc());
>-  const rtc::Optional<TargetBitrate>& target_bitrate = xr.target_bitrate();
>+  const absl::optional<TargetBitrate>& target_bitrate = xr.target_bitrate();
>   ASSERT_TRUE(static_cast<bool>(target_bitrate));
>   CheckBitrateItems(target_bitrate->GetTargetBitrates());
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmb_item.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmb_item.h
>index 7185976fe05..dc5d1b2c2d9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmb_item.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmb_item.h
>@@ -11,7 +11,8 @@
> #ifndef MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_TMMB_ITEM_H_
> #define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_TMMB_ITEM_H_
> 
>-#include "rtc_base/basictypes.h"
>+#include <stddef.h>
>+#include <stdint.h>
> 
> namespace webrtc {
> namespace rtcp {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.h
>index 75a349b6cde..ff7779d8ace 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn.h
>@@ -15,7 +15,6 @@
> 
> #include "modules/rtp_rtcp/source/rtcp_packet/rtpfb.h"
> #include "modules/rtp_rtcp/source/rtcp_packet/tmmb_item.h"
>-#include "rtc_base/basictypes.h"
> 
> namespace webrtc {
> namespace rtcp {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn_unittest.cc
>index 5a85d8a7bbf..e5d2e0bdd62 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbn_unittest.cc
>@@ -26,11 +26,9 @@ const uint32_t kSenderSsrc = 0x12345678;
> const uint32_t kRemoteSsrc = 0x23456789;
> const uint32_t kBitrateBps = 312000;
> const uint16_t kOverhead = 0x1fe;
>-const uint8_t kPacket[] = {0x84, 205,  0x00, 0x04,
>-                           0x12, 0x34, 0x56, 0x78,
>-                           0x00, 0x00, 0x00, 0x00,
>-                           0x23, 0x45, 0x67, 0x89,
>-                           0x0a, 0x61, 0x61, 0xfe};
>+const uint8_t kPacket[] = {0x84, 205,  0x00, 0x04, 0x12, 0x34, 0x56,
>+                           0x78, 0x00, 0x00, 0x00, 0x00, 0x23, 0x45,
>+                           0x67, 0x89, 0x0a, 0x61, 0x61, 0xfe};
> }  // namespace
> 
> TEST(RtcpPacketTmmbnTest, Create) {
>@@ -97,9 +95,8 @@ TEST(RtcpPacketTmmbnTest, ParseFailsOnTooSmallPacket) {
> }
> 
> TEST(RtcpPacketTmmbnTest, ParseFailsOnUnAlignedPacket) {
>-  const uint8_t kUnalignedPacket[] = {0x84,  205, 0x00, 0x03,
>-                                      0x12, 0x34, 0x56, 0x78,
>-                                      0x00, 0x00, 0x00, 0x00,
>+  const uint8_t kUnalignedPacket[] = {0x84, 205,  0x00, 0x03, 0x12, 0x34,
>+                                      0x56, 0x78, 0x00, 0x00, 0x00, 0x00,
>                                       0x23, 0x45, 0x67, 0x89};
> 
>   Tmmbn tmmbn;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.h
>index bc3ecde2a00..7482cb75cc5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr.h
>@@ -15,7 +15,6 @@
> 
> #include "modules/rtp_rtcp/source/rtcp_packet/rtpfb.h"
> #include "modules/rtp_rtcp/source/rtcp_packet/tmmb_item.h"
>-#include "rtc_base/basictypes.h"
> 
> namespace webrtc {
> namespace rtcp {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr_unittest.cc
>index d3fe72ac720..63fa9116ffa 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/tmmbr_unittest.cc
>@@ -26,11 +26,9 @@ const uint32_t kSenderSsrc = 0x12345678;
> const uint32_t kRemoteSsrc = 0x23456789;
> const uint32_t kBitrateBps = 312000;
> const uint16_t kOverhead = 0x1fe;
>-const uint8_t kPacket[] = {0x83,  205, 0x00, 0x04,
>-                           0x12, 0x34, 0x56, 0x78,
>-                           0x00, 0x00, 0x00, 0x00,
>-                           0x23, 0x45, 0x67, 0x89,
>-                           0x0a, 0x61, 0x61, 0xfe};
>+const uint8_t kPacket[] = {0x83, 205,  0x00, 0x04, 0x12, 0x34, 0x56,
>+                           0x78, 0x00, 0x00, 0x00, 0x00, 0x23, 0x45,
>+                           0x67, 0x89, 0x0a, 0x61, 0x61, 0xfe};
> }  // namespace
> 
> TEST(RtcpPacketTmmbrTest, Create) {
>@@ -74,21 +72,17 @@ TEST(RtcpPacketTmmbrTest, CreateAndParseWithTwoEntries) {
> }
> 
> TEST(RtcpPacketTmmbrTest, ParseFailsWithoutItems) {
>-  const uint8_t kZeroItemsPacket[] = {0x83,  205, 0x00, 0x02,
>-                                      0x12, 0x34, 0x56, 0x78,
>-                                      0x00, 0x00, 0x00, 0x00};
>+  const uint8_t kZeroItemsPacket[] = {0x83, 205,  0x00, 0x02, 0x12, 0x34,
>+                                      0x56, 0x78, 0x00, 0x00, 0x00, 0x00};
> 
>   Tmmbr tmmbr;
>   EXPECT_FALSE(test::ParseSinglePacket(kZeroItemsPacket, &tmmbr));
> }
> 
> TEST(RtcpPacketTmmbrTest, ParseFailsOnUnAlignedPacket) {
>-  const uint8_t kUnalignedPacket[] = {0x83,  205, 0x00, 0x05,
>-                                      0x12, 0x34, 0x56, 0x78,
>-                                      0x00, 0x00, 0x00, 0x00,
>-                                      0x23, 0x45, 0x67, 0x89,
>-                                      0x0a, 0x61, 0x61, 0xfe,
>-                                      0x34, 0x56, 0x78, 0x9a};
>+  const uint8_t kUnalignedPacket[] = {
>+      0x83, 205,  0x00, 0x05, 0x12, 0x34, 0x56, 0x78, 0x00, 0x00, 0x00, 0x00,
>+      0x23, 0x45, 0x67, 0x89, 0x0a, 0x61, 0x61, 0xfe, 0x34, 0x56, 0x78, 0x9a};
> 
>   Tmmbr tmmbr;
>   EXPECT_FALSE(test::ParseSinglePacket(kUnalignedPacket, &tmmbr));
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.cc
>index 05596de5cc8..4703d3132ee 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.cc
>@@ -11,6 +11,7 @@
> #include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
> 
> #include <algorithm>
>+#include <utility>
> 
> #include "modules/include/module_common_types.h"
> #include "modules/rtp_rtcp/source/byte_io.h"
>@@ -73,53 +74,6 @@ constexpr int64_t kTimeWrapPeriodUs = (1ll << 24) * kBaseScaleFactor;
> constexpr uint8_t TransportFeedback::kFeedbackMessageType;
> constexpr size_t TransportFeedback::kMaxReportedPackets;
> 
>-// Keep delta_sizes that can be encoded into single chunk if it is last chunk.
>-class TransportFeedback::LastChunk {
>- public:
>-  using DeltaSize = TransportFeedback::DeltaSize;
>-
>-  LastChunk();
>-
>-  bool Empty() const;
>-  void Clear();
>-  // Return if delta sizes still can be encoded into single chunk with added
>-  // |delta_size|.
>-  bool CanAdd(DeltaSize delta_size) const;
>-  // Add |delta_size|, assumes |CanAdd(delta_size)|,
>-  void Add(DeltaSize delta_size);
>-
>-  // Encode chunk as large as possible removing encoded delta sizes.
>-  // Assume CanAdd() == false for some valid delta_size.
>-  uint16_t Emit();
>-  // Encode all stored delta_sizes into single chunk, pad with 0s if needed.
>-  uint16_t EncodeLast() const;
>-
>-  // Decode up to |max_size| delta sizes from |chunk|.
>-  void Decode(uint16_t chunk, size_t max_size);
>-  // Appends content of the Lastchunk to |deltas|.
>-  void AppendTo(std::vector<DeltaSize>* deltas) const;
>-
>- private:
>-  static constexpr size_t kMaxRunLengthCapacity = 0x1fff;
>-  static constexpr size_t kMaxOneBitCapacity = 14;
>-  static constexpr size_t kMaxTwoBitCapacity = 7;
>-  static constexpr size_t kMaxVectorCapacity = kMaxOneBitCapacity;
>-  static constexpr DeltaSize kLarge = 2;
>-
>-  uint16_t EncodeOneBit() const;
>-  void DecodeOneBit(uint16_t chunk, size_t max_size);
>-
>-  uint16_t EncodeTwoBit(size_t size) const;
>-  void DecodeTwoBit(uint16_t chunk, size_t max_size);
>-
>-  uint16_t EncodeRunLength() const;
>-  void DecodeRunLength(uint16_t chunk, size_t max_size);
>-
>-  DeltaSize delta_sizes_[kMaxVectorCapacity];
>-  size_t size_;
>-  bool all_same_;
>-  bool has_large_delta_;
>-};
> constexpr size_t TransportFeedback::LastChunk::kMaxRunLengthCapacity;
> constexpr size_t TransportFeedback::LastChunk::kMaxOneBitCapacity;
> constexpr size_t TransportFeedback::LastChunk::kMaxTwoBitCapacity;
>@@ -312,9 +266,23 @@ TransportFeedback::TransportFeedback()
>       base_time_ticks_(0),
>       feedback_seq_(0),
>       last_timestamp_us_(0),
>-      last_chunk_(new LastChunk()),
>       size_bytes_(kTransportFeedbackHeaderSizeBytes) {}
> 
>+TransportFeedback::TransportFeedback(const TransportFeedback&) = default;
>+
>+TransportFeedback::TransportFeedback(TransportFeedback&& other)
>+    : base_seq_no_(other.base_seq_no_),
>+      num_seq_no_(other.num_seq_no_),
>+      base_time_ticks_(other.base_time_ticks_),
>+      feedback_seq_(other.feedback_seq_),
>+      last_timestamp_us_(other.last_timestamp_us_),
>+      packets_(std::move(other.packets_)),
>+      encoded_chunks_(std::move(other.encoded_chunks_)),
>+      last_chunk_(other.last_chunk_),
>+      size_bytes_(other.size_bytes_) {
>+  other.Clear();
>+}
>+
> TransportFeedback::~TransportFeedback() {}
> 
> void TransportFeedback::SetBase(uint16_t base_sequence,
>@@ -422,8 +390,8 @@ bool TransportFeedback::Parse(const CommonHeader& packet) {
>     uint16_t chunk = ByteReader<uint16_t>::ReadBigEndian(&payload[index]);
>     index += kChunkSizeBytes;
>     encoded_chunks_.push_back(chunk);
>-    last_chunk_->Decode(chunk, status_count - delta_sizes.size());
>-    last_chunk_->AppendTo(&delta_sizes);
>+    last_chunk_.Decode(chunk, status_count - delta_sizes.size());
>+    last_chunk_.AppendTo(&delta_sizes);
>   }
>   // Last chunk is stored in the |last_chunk_|.
>   encoded_chunks_.pop_back();
>@@ -492,8 +460,8 @@ bool TransportFeedback::IsConsistent() const {
>     chunk_decoder.AppendTo(&delta_sizes);
>     packet_size += kChunkSizeBytes;
>   }
>-  if (!last_chunk_->Empty()) {
>-    last_chunk_->AppendTo(&delta_sizes);
>+  if (!last_chunk_.Empty()) {
>+    last_chunk_.AppendTo(&delta_sizes);
>     packet_size += kChunkSizeBytes;
>   }
>   if (num_seq_no_ != delta_sizes.size()) {
>@@ -586,8 +554,8 @@ bool TransportFeedback::Create(uint8_t* packet,
>     ByteWriter<uint16_t>::WriteBigEndian(&packet[*position], chunk);
>     *position += 2;
>   }
>-  if (!last_chunk_->Empty()) {
>-    uint16_t chunk = last_chunk_->EncodeLast();
>+  if (!last_chunk_.Empty()) {
>+    uint16_t chunk = last_chunk_.EncodeLast();
>     ByteWriter<uint16_t>::WriteBigEndian(&packet[*position], chunk);
>     *position += 2;
>   }
>@@ -614,29 +582,29 @@ void TransportFeedback::Clear() {
>   last_timestamp_us_ = GetBaseTimeUs();
>   packets_.clear();
>   encoded_chunks_.clear();
>-  last_chunk_->Clear();
>+  last_chunk_.Clear();
>   size_bytes_ = kTransportFeedbackHeaderSizeBytes;
> }
> 
> bool TransportFeedback::AddDeltaSize(DeltaSize delta_size) {
>   if (num_seq_no_ == kMaxReportedPackets)
>     return false;
>-  size_t add_chunk_size = last_chunk_->Empty() ? kChunkSizeBytes : 0;
>+  size_t add_chunk_size = last_chunk_.Empty() ? kChunkSizeBytes : 0;
>   if (size_bytes_ + delta_size + add_chunk_size > kMaxSizeBytes)
>     return false;
> 
>-  if (last_chunk_->CanAdd(delta_size)) {
>+  if (last_chunk_.CanAdd(delta_size)) {
>     size_bytes_ += add_chunk_size;
>-    last_chunk_->Add(delta_size);
>+    last_chunk_.Add(delta_size);
>     ++num_seq_no_;
>     return true;
>   }
>   if (size_bytes_ + delta_size + kChunkSizeBytes > kMaxSizeBytes)
>     return false;
> 
>-  encoded_chunks_.push_back(last_chunk_->Emit());
>+  encoded_chunks_.push_back(last_chunk_.Emit());
>   size_bytes_ += kChunkSizeBytes;
>-  last_chunk_->Add(delta_size);
>+  last_chunk_.Add(delta_size);
>   ++num_seq_no_;
>   return true;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h
>index 5ec83a7c4d8..fbdc38e9264 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h
>@@ -45,6 +45,9 @@ class TransportFeedback : public Rtpfb {
>   static constexpr size_t kMaxReportedPackets = 0xffff;
> 
>   TransportFeedback();
>+  TransportFeedback(const TransportFeedback&);
>+  TransportFeedback(TransportFeedback&&);
>+
>   ~TransportFeedback() override;
> 
>   void SetBase(uint16_t base_sequence,     // Seq# of first packet in this msg.
>@@ -81,7 +84,52 @@ class TransportFeedback : public Rtpfb {
>   // Valid values are 0 (packet wasn't received), 1 or 2.
>   using DeltaSize = uint8_t;
>   // Keeps DeltaSizes that can be encoded into single chunk if it is last chunk.
>-  class LastChunk;
>+  class LastChunk {
>+   public:
>+    using DeltaSize = TransportFeedback::DeltaSize;
>+
>+    LastChunk();
>+
>+    bool Empty() const;
>+    void Clear();
>+    // Return if delta sizes still can be encoded into single chunk with added
>+    // |delta_size|.
>+    bool CanAdd(DeltaSize delta_size) const;
>+    // Add |delta_size|, assumes |CanAdd(delta_size)|,
>+    void Add(DeltaSize delta_size);
>+
>+    // Encode chunk as large as possible removing encoded delta sizes.
>+    // Assume CanAdd() == false for some valid delta_size.
>+    uint16_t Emit();
>+    // Encode all stored delta_sizes into single chunk, pad with 0s if needed.
>+    uint16_t EncodeLast() const;
>+
>+    // Decode up to |max_size| delta sizes from |chunk|.
>+    void Decode(uint16_t chunk, size_t max_size);
>+    // Appends content of the Lastchunk to |deltas|.
>+    void AppendTo(std::vector<DeltaSize>* deltas) const;
>+
>+   private:
>+    static constexpr size_t kMaxRunLengthCapacity = 0x1fff;
>+    static constexpr size_t kMaxOneBitCapacity = 14;
>+    static constexpr size_t kMaxTwoBitCapacity = 7;
>+    static constexpr size_t kMaxVectorCapacity = kMaxOneBitCapacity;
>+    static constexpr DeltaSize kLarge = 2;
>+
>+    uint16_t EncodeOneBit() const;
>+    void DecodeOneBit(uint16_t chunk, size_t max_size);
>+
>+    uint16_t EncodeTwoBit(size_t size) const;
>+    void DecodeTwoBit(uint16_t chunk, size_t max_size);
>+
>+    uint16_t EncodeRunLength() const;
>+    void DecodeRunLength(uint16_t chunk, size_t max_size);
>+
>+    DeltaSize delta_sizes_[kMaxVectorCapacity];
>+    size_t size_;
>+    bool all_same_;
>+    bool has_large_delta_;
>+  };
> 
>   // Reset packet to consistent empty state.
>   void Clear();
>@@ -97,7 +145,7 @@ class TransportFeedback : public Rtpfb {
>   std::vector<ReceivedPacket> packets_;
>   // All but last encoded packet chunks.
>   std::vector<uint16_t> encoded_chunks_;
>-  const std::unique_ptr<LastChunk> last_chunk_;
>+  LastChunk last_chunk_;
>   size_t size_bytes_;
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback_unittest.cc
>index 1a4cf7874df..04965256bd7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback_unittest.cc
>@@ -12,6 +12,7 @@
> 
> #include <limits>
> #include <memory>
>+#include <utility>
> 
> #include "modules/rtp_rtcp/source/byte_io.h"
> #include "test/gmock.h"
>@@ -78,8 +79,8 @@ class FeedbackTester {
>     ASSERT_TRUE(feedback_->IsConsistent());
>     serialized_ = feedback_->Build();
>     VerifyInternal();
>-    feedback_ = TransportFeedback::ParseFrom(serialized_.data(),
>-                                             serialized_.size());
>+    feedback_ =
>+        TransportFeedback::ParseFrom(serialized_.data(), serialized_.size());
>     ASSERT_TRUE(feedback_->IsConsistent());
>     ASSERT_NE(nullptr, feedback_.get());
>     VerifyInternal();
>@@ -208,10 +209,7 @@ TEST(RtcpPacketTest, TransportFeedback_TwoBitVectorFull) {
> TEST(RtcpPacketTest, TransportFeedback_LargeAndNegativeDeltas) {
>   const uint16_t kReceived[] = {1, 2, 6, 7, 8};
>   const int64_t kReceiveTimes[] = {
>-      2000,
>-      1000,
>-      4000,
>-      3000,
>+      2000, 1000, 4000, 3000,
>       3000 + TransportFeedback::kDeltaScaleFactor * (1 << 8)};
>   const size_t kLength = sizeof(kReceived) / sizeof(uint16_t);
>   const size_t kExpectedSizeBytes =
>@@ -466,5 +464,31 @@ TEST(RtcpPacketTest, TransportFeedback_CorrectlySplitsVectorChunks) {
>   }
> }
> 
>+TEST(RtcpPacketTest, TransportFeedback_MoveConstructor) {
>+  const int kSamples = 100;
>+  const int64_t kDelta = TransportFeedback::kDeltaScaleFactor;
>+  const uint16_t kBaseSeqNo = 7531;
>+  const int64_t kBaseTimestampUs = 123456789;
>+  const uint8_t kFeedbackSeqNo = 90;
>+
>+  TransportFeedback feedback;
>+  feedback.SetBase(kBaseSeqNo, kBaseTimestampUs);
>+  feedback.SetFeedbackSequenceNumber(kFeedbackSeqNo);
>+  for (int i = 0; i < kSamples; ++i) {
>+    feedback.AddReceivedPacket(kBaseSeqNo + i, kBaseTimestampUs + i * kDelta);
>+  }
>+  EXPECT_TRUE(feedback.IsConsistent());
>+
>+  TransportFeedback feedback_copy(feedback);
>+  EXPECT_TRUE(feedback_copy.IsConsistent());
>+  EXPECT_TRUE(feedback.IsConsistent());
>+  EXPECT_EQ(feedback_copy.Build(), feedback.Build());
>+
>+  TransportFeedback moved(std::move(feedback));
>+  EXPECT_TRUE(moved.IsConsistent());
>+  EXPECT_TRUE(feedback.IsConsistent());
>+  EXPECT_EQ(moved.Build(), feedback_copy.Build());
>+}
>+
> }  // namespace
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/voip_metric.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/voip_metric.h
>index 135dc45c760..c5588a47cef 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/voip_metric.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/voip_metric.h
>@@ -13,7 +13,6 @@
> #define MODULES_RTP_RTCP_SOURCE_RTCP_PACKET_VOIP_METRIC_H_
> 
> #include "modules/include/module_common_types.h"
>-#include "rtc_base/basictypes.h"
> 
> namespace webrtc {
> namespace rtcp {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/voip_metric_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/voip_metric_unittest.cc
>index a2fbcc5f758..598b8c6c6cf 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/voip_metric_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_packet/voip_metric_unittest.cc
>@@ -17,11 +17,10 @@ namespace rtcp {
> namespace {
> 
> const uint32_t kRemoteSsrc = 0x23456789;
>-const uint8_t kBlock[] = {0x07, 0x00, 0x00, 0x08, 0x23, 0x45, 0x67, 0x89,
>-                          0x01, 0x02, 0x03, 0x04, 0x11, 0x12, 0x22, 0x23,
>-                          0x33, 0x34, 0x44, 0x45, 0x05, 0x06, 0x07, 0x08,
>-                          0x09, 0x0a, 0x0b, 0x0c, 0x0d, 0x00, 0x55, 0x56,
>-                          0x66, 0x67, 0x77, 0x78};
>+const uint8_t kBlock[] = {0x07, 0x00, 0x00, 0x08, 0x23, 0x45, 0x67, 0x89, 0x01,
>+                          0x02, 0x03, 0x04, 0x11, 0x12, 0x22, 0x23, 0x33, 0x34,
>+                          0x44, 0x45, 0x05, 0x06, 0x07, 0x08, 0x09, 0x0a, 0x0b,
>+                          0x0c, 0x0d, 0x00, 0x55, 0x56, 0x66, 0x67, 0x77, 0x78};
> const size_t kBlockSizeBytes = sizeof(kBlock);
> static_assert(
>     kBlockSizeBytes == VoipMetric::kLength,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
>index cff7468af65..496675497ad 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc
>@@ -18,8 +18,9 @@
> #include <utility>
> #include <vector>
> 
>+#include "api/video/video_bitrate_allocation.h"
>+#include "api/video/video_bitrate_allocator.h"
> #include "common_types.h"  // NOLINT(build/include)
>-#include "common_video/include/video_bitrate_allocator.h"
> #include "modules/rtp_rtcp/source/rtcp_packet/bye.h"
> #include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
> #include "modules/rtp_rtcp/source/rtcp_packet/compound_packet.h"
>@@ -52,9 +53,14 @@ using rtcp::ReportBlock;
> // The number of RTCP time intervals needed to trigger a timeout.
> const int kRrTimeoutIntervals = 3;
> 
>+const int64_t kTmmbrTimeoutIntervalMs = 5 * 5000;
>+
> const int64_t kMaxWarningLogIntervalMs = 10000;
> const int64_t kRtcpMinFrameLengthMs = 17;
> 
>+// Maximum number of received RRTRs that will be stored.
>+const size_t kMaxNumberOfStoredRrtrs = 200;
>+
> }  // namespace
> 
> struct RTCPReceiver::PacketInformation {
>@@ -66,7 +72,7 @@ struct RTCPReceiver::PacketInformation {
>   int64_t rtt_ms = 0;
>   uint32_t receiver_estimated_max_bitrate_bps = 0;
>   std::unique_ptr<rtcp::TransportFeedback> transport_feedback;
>-  rtc::Optional<BitrateAllocation> target_bitrate_allocation;
>+  absl::optional<VideoBitrateAllocation> target_bitrate_allocation;
> };
> 
> // Structure for handing TMMBR and TMMBN rtcp messages (RFC5104, section 3.5.4).
>@@ -84,6 +90,22 @@ struct RTCPReceiver::TmmbrInformation {
>   std::map<uint32_t, TimedTmmbrItem> tmmbr;
> };
> 
>+// Structure for storing received RRTR RTCP messages (RFC3611, section 4.4).
>+struct RTCPReceiver::RrtrInformation {
>+  RrtrInformation(uint32_t ssrc,
>+                  uint32_t received_remote_mid_ntp_time,
>+                  uint32_t local_receive_mid_ntp_time)
>+      : ssrc(ssrc),
>+        received_remote_mid_ntp_time(received_remote_mid_ntp_time),
>+        local_receive_mid_ntp_time(local_receive_mid_ntp_time) {}
>+
>+  uint32_t ssrc;
>+  // Received NTP timestamp in compact representation.
>+  uint32_t received_remote_mid_ntp_time;
>+  // NTP time when the report was received in compact representation.
>+  uint32_t local_receive_mid_ntp_time;
>+};
>+
> struct RTCPReceiver::ReportBlockWithRtt {
>   RTCPReportBlock report_block;
> 
>@@ -249,22 +271,26 @@ bool RTCPReceiver::NTP(uint32_t* received_ntp_secs,
>   return true;
> }
> 
>-bool RTCPReceiver::LastReceivedXrReferenceTimeInfo(
>-    rtcp::ReceiveTimeInfo* info) const {
>-  RTC_DCHECK(info);
>+std::vector<rtcp::ReceiveTimeInfo>
>+RTCPReceiver::ConsumeReceivedXrReferenceTimeInfo() {
>   rtc::CritScope lock(&rtcp_receiver_lock_);
>-  if (!last_received_xr_ntp_.Valid())
>-    return false;
> 
>-  info->ssrc = remote_time_info_.ssrc;
>-  info->last_rr = remote_time_info_.last_rr;
>+  const size_t last_xr_rtis_size = std::min(
>+      received_rrtrs_.size(), rtcp::ExtendedReports::kMaxNumberOfDlrrItems);
>+  std::vector<rtcp::ReceiveTimeInfo> last_xr_rtis;
>+  last_xr_rtis.reserve(last_xr_rtis_size);
> 
>-  // Get the delay since last received report (RFC 3611).
>-  uint32_t receive_time_ntp = CompactNtp(last_received_xr_ntp_);
>-  uint32_t now_ntp = CompactNtp(clock_->CurrentNtpTime());
>+  const uint32_t now_ntp = CompactNtp(clock_->CurrentNtpTime());
> 
>-  info->delay_since_last_rr = now_ntp - receive_time_ntp;
>-  return true;
>+  for (size_t i = 0; i < last_xr_rtis_size; ++i) {
>+    RrtrInformation& rrtr = received_rrtrs_.front();
>+    last_xr_rtis.emplace_back(rrtr.ssrc, rrtr.received_remote_mid_ntp_time,
>+                              now_ntp - rrtr.local_receive_mid_ntp_time);
>+    received_rrtrs_ssrc_it_.erase(rrtr.ssrc);
>+    received_rrtrs_.pop_front();
>+  }
>+
>+  return last_xr_rtis;
> }
> 
> // We can get multiple receive reports when we receive the report from a CE.
>@@ -394,9 +420,6 @@ void RTCPReceiver::HandleSenderReport(const CommonHeader& rtcp_block,
> 
>   UpdateTmmbrRemoteIsAlive(remote_ssrc);
> 
>-  TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "SR",
>-                       "remote_ssrc", remote_ssrc, "ssrc", main_ssrc_);
>-
>   // Have I received RTP packets from this party?
>   if (remote_ssrc_ == remote_ssrc) {
>     // Only signal that we have received a SR when we accept one.
>@@ -429,9 +452,6 @@ void RTCPReceiver::HandleReceiverReport(const CommonHeader& rtcp_block,
> 
>   UpdateTmmbrRemoteIsAlive(remote_ssrc);
> 
>-  TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "RR",
>-                       "remote_ssrc", remote_ssrc, "ssrc", main_ssrc_);
>-
>   packet_information->packet_type_flags |= kRtcpRr;
> 
>   for (const ReportBlock& report_block : receiver_report.report_blocks())
>@@ -502,12 +522,10 @@ void RTCPReceiver::HandleReportBlock(const ReportBlock& report_block,
>     report_block_info->last_rtt_ms = rtt_ms;
>     report_block_info->sum_rtt_ms += rtt_ms;
>     ++report_block_info->num_rtts;
>-  }
> 
>-  TRACE_COUNTER_ID1(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "RR_RTT",
>-                    report_block.source_ssrc(), rtt_ms);
>+    packet_information->rtt_ms = rtt_ms;
>+  }
> 
>-  packet_information->rtt_ms = rtt_ms;
>   packet_information->report_blocks.push_back(report_block_info->report_block);
> }
> 
>@@ -567,8 +585,7 @@ bool RTCPReceiver::UpdateTmmbrTimers() {
>   rtc::CritScope lock(&rtcp_receiver_lock_);
> 
>   int64_t now_ms = clock_->TimeInMilliseconds();
>-  // Use audio define since we don't know what interval the remote peer use.
>-  int64_t timeout_ms = now_ms - 5 * RTCP_INTERVAL_AUDIO_MS;
>+  int64_t timeout_ms = now_ms - kTmmbrTimeoutIntervalMs;
> 
>   if (oldest_tmmbr_info_ms_ >= timeout_ms)
>     return false;
>@@ -672,6 +689,11 @@ void RTCPReceiver::HandleBye(const CommonHeader& rtcp_block) {
> 
>   last_fir_.erase(bye.sender_ssrc());
>   received_cnames_.erase(bye.sender_ssrc());
>+  auto it = received_rrtrs_ssrc_it_.find(bye.sender_ssrc());
>+  if (it != received_rrtrs_ssrc_it_.end()) {
>+    received_rrtrs_.erase(it->second);
>+    received_rrtrs_ssrc_it_.erase(it);
>+  }
>   xr_rr_rtt_ms_ = 0;
> }
> 
>@@ -697,9 +719,23 @@ void RTCPReceiver::HandleXr(const CommonHeader& rtcp_block,
> 
> void RTCPReceiver::HandleXrReceiveReferenceTime(uint32_t sender_ssrc,
>                                                 const rtcp::Rrtr& rrtr) {
>-  remote_time_info_.ssrc = sender_ssrc;
>-  remote_time_info_.last_rr = CompactNtp(rrtr.ntp());
>-  last_received_xr_ntp_ = clock_->CurrentNtpTime();
>+  uint32_t received_remote_mid_ntp_time = CompactNtp(rrtr.ntp());
>+  uint32_t local_receive_mid_ntp_time = CompactNtp(clock_->CurrentNtpTime());
>+
>+  auto it = received_rrtrs_ssrc_it_.find(sender_ssrc);
>+  if (it != received_rrtrs_ssrc_it_.end()) {
>+    it->second->received_remote_mid_ntp_time = received_remote_mid_ntp_time;
>+    it->second->local_receive_mid_ntp_time = local_receive_mid_ntp_time;
>+  } else {
>+    if (received_rrtrs_.size() < kMaxNumberOfStoredRrtrs) {
>+      received_rrtrs_.emplace_back(sender_ssrc, received_remote_mid_ntp_time,
>+                                   local_receive_mid_ntp_time);
>+      received_rrtrs_ssrc_it_[sender_ssrc] = std::prev(received_rrtrs_.end());
>+    } else {
>+      RTC_LOG(LS_WARNING) << "Discarding received RRTR for ssrc " << sender_ssrc
>+                          << ", reached maximum number of stored RRTRs.";
>+    }
>+  }
> }
> 
> void RTCPReceiver::HandleXrDlrrReportBlock(const rtcp::ReceiveTimeInfo& rti) {
>@@ -732,7 +768,7 @@ void RTCPReceiver::HandleXrTargetBitrate(
>     return;  // Not for us.
>   }
> 
>-  BitrateAllocation bitrate_allocation;
>+  VideoBitrateAllocation bitrate_allocation;
>   for (const auto& item : target_bitrate.GetTargetBitrates()) {
>     if (item.spatial_layer >= kMaxSpatialLayers ||
>         item.temporal_layer >= kMaxTemporalStreams) {
>@@ -757,8 +793,6 @@ void RTCPReceiver::HandlePli(const CommonHeader& rtcp_block,
>   }
> 
>   if (main_ssrc_ == pli.media_ssrc()) {
>-    TRACE_EVENT_INSTANT0(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "PLI");
>-
>     ++packet_type_counter_.pli_packets;
>     // Received a signal that we need to send a new key frame.
>     packet_information->packet_type_flags |= kRtcpPli;
>@@ -786,8 +820,7 @@ void RTCPReceiver::HandleTmmbr(const CommonHeader& rtcp_block,
> 
>     TmmbrInformation* tmmbr_info = FindOrCreateTmmbrInfo(tmmbr.sender_ssrc());
>     auto* entry = &tmmbr_info->tmmbr[sender_ssrc];
>-    entry->tmmbr_item = rtcp::TmmbItem(sender_ssrc,
>-                                       request.bitrate_bps(),
>+    entry->tmmbr_item = rtcp::TmmbItem(sender_ssrc, request.bitrate_bps(),
>                                        request.packet_overhead());
>     entry->last_updated_ms = clock_->TimeInMilliseconds();
> 
>@@ -1032,8 +1065,7 @@ std::vector<rtcp::TmmbItem> RTCPReceiver::TmmbrReceived() {
>   std::vector<rtcp::TmmbItem> candidates;
> 
>   int64_t now_ms = clock_->TimeInMilliseconds();
>-  // Use audio define since we don't know what interval the remote peer use.
>-  int64_t timeout_ms = now_ms - 5 * RTCP_INTERVAL_AUDIO_MS;
>+  int64_t timeout_ms = now_ms - kTmmbrTimeoutIntervalMs;
> 
>   for (auto& kv : tmmbr_infos_) {
>     for (auto it = kv.second.tmmbr.begin(); it != kv.second.tmmbr.end();) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
>index c3de36c54f4..a863caeff7b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_receiver.h
>@@ -11,6 +11,7 @@
> #ifndef MODULES_RTP_RTCP_SOURCE_RTCP_RECEIVER_H_
> #define MODULES_RTP_RTCP_SOURCE_RTCP_RECEIVER_H_
> 
>+#include <list>
> #include <map>
> #include <set>
> #include <string>
>@@ -22,7 +23,6 @@
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/thread_annotations.h"
> #include "system_wrappers/include/ntp_time.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> class VideoBitrateAllocationObserver;
>@@ -77,7 +77,7 @@ class RTCPReceiver {
>            uint32_t* rtcp_arrival_time_frac,
>            uint32_t* rtcp_timestamp) const;
> 
>-  bool LastReceivedXrReferenceTimeInfo(rtcp::ReceiveTimeInfo* info) const;
>+  std::vector<rtcp::ReceiveTimeInfo> ConsumeReceivedXrReferenceTimeInfo();
> 
>   // Get rtt.
>   int32_t RTT(uint32_t remote_ssrc,
>@@ -115,6 +115,7 @@ class RTCPReceiver {
>  private:
>   struct PacketInformation;
>   struct TmmbrInformation;
>+  struct RrtrInformation;
>   struct ReportBlockWithRtt;
>   struct LastFirStatus;
>   // RTCP report blocks mapped by remote SSRC.
>@@ -226,10 +227,13 @@ class RTCPReceiver {
>   // When did we receive the last send report.
>   NtpTime last_received_sr_ntp_ RTC_GUARDED_BY(rtcp_receiver_lock_);
> 
>-  // Received XR receive time report.
>-  rtcp::ReceiveTimeInfo remote_time_info_;
>-  // Time when the report was received.
>-  NtpTime last_received_xr_ntp_;
>+  // Received RRTR information in ascending receive time order.
>+  std::list<RrtrInformation> received_rrtrs_
>+      RTC_GUARDED_BY(rtcp_receiver_lock_);
>+  // Received RRTR information mapped by remote ssrc.
>+  std::map<uint32_t, std::list<RrtrInformation>::iterator>
>+      received_rrtrs_ssrc_it_ RTC_GUARDED_BY(rtcp_receiver_lock_);
>+
>   // Estimated rtt, zero when there is no valid estimate.
>   bool xr_rrtr_status_ RTC_GUARDED_BY(rtcp_receiver_lock_);
>   int64_t xr_rr_rtt_ms_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
>index fb09ee1b7bf..6cd953c28fc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc
>@@ -11,8 +11,10 @@
> #include <memory>
> 
> #include "api/array_view.h"
>+#include "api/video/video_bitrate_allocation.h"
>+#include "api/video/video_bitrate_allocator.h"
> #include "common_types.h"  // NOLINT(build/include)
>-#include "common_video/include/video_bitrate_allocator.h"
>+#include "modules/rtp_rtcp/mocks/mock_rtcp_bandwidth_observer.h"
> #include "modules/rtp_rtcp/source/byte_io.h"
> #include "modules/rtp_rtcp/source/rtcp_packet.h"
> #include "modules/rtp_rtcp/source/rtcp_packet/app.h"
>@@ -80,13 +82,6 @@ class MockTransportFeedbackObserver : public TransportFeedbackObserver {
>   MOCK_CONST_METHOD0(GetTransportFeedbackVector, std::vector<PacketFeedback>());
> };
> 
>-class MockRtcpBandwidthObserver : public RtcpBandwidthObserver {
>- public:
>-  MOCK_METHOD1(OnReceivedEstimatedBitrate, void(uint32_t));
>-  MOCK_METHOD3(OnReceivedRtcpReceiverReport,
>-               void(const ReportBlockList&, int64_t, int64_t));
>-};
>-
> class MockModuleRtpRtcp : public RTCPReceiver::ModuleRtpRtcp {
>  public:
>   MOCK_METHOD1(SetTmmbn, void(std::vector<rtcp::TmmbItem>));
>@@ -99,7 +94,7 @@ class MockVideoBitrateAllocationObserver
>     : public VideoBitrateAllocationObserver {
>  public:
>   MOCK_METHOD1(OnBitrateAllocationUpdated,
>-               void(const BitrateAllocation& allocation));
>+               void(const VideoBitrateAllocation& allocation));
> };
> 
> // SSRC of remote peer, that sends rtcp packet to the rtcp receiver under test.
>@@ -264,6 +259,33 @@ TEST_F(RtcpReceiverTest, InjectSrPacketCalculatesNegativeRTTAsOne) {
>   EXPECT_EQ(1, rtt_ms);
> }
> 
>+TEST_F(
>+    RtcpReceiverTest,
>+    TwoReportBlocksWithLastOneWithoutLastSrCalculatesRttForBandwidthObserver) {
>+  const int64_t kRttMs = 120;
>+  const uint32_t kDelayNtp = 123000;
>+  const int64_t kDelayMs = CompactNtpRttToMs(kDelayNtp);
>+
>+  uint32_t sent_ntp = CompactNtp(system_clock_.CurrentNtpTime());
>+  system_clock_.AdvanceTimeMilliseconds(kRttMs + kDelayMs);
>+
>+  rtcp::SenderReport sr;
>+  sr.SetSenderSsrc(kSenderSsrc);
>+  rtcp::ReportBlock block;
>+  block.SetMediaSsrc(kReceiverMainSsrc);
>+  block.SetLastSr(sent_ntp);
>+  block.SetDelayLastSr(kDelayNtp);
>+  sr.AddReportBlock(block);
>+  block.SetMediaSsrc(kReceiverExtraSsrc);
>+  block.SetLastSr(0);
>+  sr.AddReportBlock(block);
>+
>+  EXPECT_CALL(rtp_rtcp_impl_, OnReceivedRtcpReportBlocks(SizeIs(2)));
>+  EXPECT_CALL(bandwidth_observer_,
>+              OnReceivedRtcpReceiverReport(SizeIs(2), kRttMs, _));
>+  InjectRtcpPacket(sr);
>+}
>+
> TEST_F(RtcpReceiverTest, InjectRrPacket) {
>   int64_t now = system_clock_.TimeInMilliseconds();
>   rtcp::ReceiverReport rr;
>@@ -419,7 +441,7 @@ TEST_F(RtcpReceiverTest, InjectRrPacketWithTwoReportBlocks) {
> TEST_F(RtcpReceiverTest, InjectRrPacketsFromTwoRemoteSsrcs) {
>   const uint32_t kSenderSsrc2 = 0x20304;
>   const uint16_t kSequenceNumbers[] = {10, 12423};
>-  const uint32_t kCumLost[] = {13, 555};
>+  const int32_t kCumLost[] = {13, 555};
>   const uint8_t kFracLost[] = {20, 11};
> 
>   rtcp::ReportBlock rb1;
>@@ -599,6 +621,21 @@ TEST_F(RtcpReceiverTest, InjectByePacket_RemovesReportBlocks) {
>   EXPECT_EQ(2u, received_blocks.size());
> }
> 
>+TEST_F(RtcpReceiverTest, InjectByePacketRemovesReferenceTimeInfo) {
>+  rtcp::ExtendedReports xr;
>+  xr.SetSenderSsrc(kSenderSsrc);
>+  rtcp::Rrtr rrtr;
>+  rrtr.SetNtp(NtpTime(0x10203, 0x40506));
>+  xr.SetRrtr(rrtr);
>+  InjectRtcpPacket(xr);
>+
>+  rtcp::Bye bye;
>+  bye.SetSenderSsrc(kSenderSsrc);
>+  InjectRtcpPacket(bye);
>+
>+  EXPECT_THAT(rtcp_receiver_.ConsumeReceivedXrReferenceTimeInfo(), IsEmpty());
>+}
>+
> TEST_F(RtcpReceiverTest, InjectPliPacket) {
>   rtcp::Pli pli;
>   pli.SetMediaSsrc(kReceiverMainSsrc);
>@@ -685,19 +722,17 @@ TEST_F(RtcpReceiverTest, InjectExtendedReportsReceiverReferenceTimePacket) {
>   xr.SetSenderSsrc(kSenderSsrc);
>   xr.SetRrtr(rrtr);
> 
>-  ReceiveTimeInfo rrtime;
>-  EXPECT_FALSE(rtcp_receiver_.LastReceivedXrReferenceTimeInfo(&rrtime));
>+  std::vector<rtcp::ReceiveTimeInfo> last_xr_rtis =
>+      rtcp_receiver_.ConsumeReceivedXrReferenceTimeInfo();
>+  EXPECT_THAT(last_xr_rtis, IsEmpty());
> 
>   InjectRtcpPacket(xr);
> 
>-  EXPECT_TRUE(rtcp_receiver_.LastReceivedXrReferenceTimeInfo(&rrtime));
>-  EXPECT_EQ(rrtime.ssrc, kSenderSsrc);
>-  EXPECT_EQ(rrtime.last_rr, CompactNtp(kNtp));
>-  EXPECT_EQ(0U, rrtime.delay_since_last_rr);
>-
>-  system_clock_.AdvanceTimeMilliseconds(1500);
>-  EXPECT_TRUE(rtcp_receiver_.LastReceivedXrReferenceTimeInfo(&rrtime));
>-  EXPECT_NEAR(1500, CompactNtpRttToMs(rrtime.delay_since_last_rr), 1);
>+  last_xr_rtis = rtcp_receiver_.ConsumeReceivedXrReferenceTimeInfo();
>+  ASSERT_THAT(last_xr_rtis, SizeIs(1));
>+  EXPECT_EQ(kSenderSsrc, last_xr_rtis[0].ssrc);
>+  EXPECT_EQ(CompactNtp(kNtp), last_xr_rtis[0].last_rr);
>+  EXPECT_EQ(0U, last_xr_rtis[0].delay_since_last_rr);
> }
> 
> TEST_F(RtcpReceiverTest, ExtendedReportsDlrrPacketNotToUsIgnored) {
>@@ -767,8 +802,9 @@ TEST_F(RtcpReceiverTest, InjectExtendedReportsPacketWithMultipleReportBlocks) {
> 
>   InjectRtcpPacket(xr);
> 
>-  ReceiveTimeInfo rrtime;
>-  EXPECT_TRUE(rtcp_receiver_.LastReceivedXrReferenceTimeInfo(&rrtime));
>+  std::vector<rtcp::ReceiveTimeInfo> last_xr_rtis =
>+      rtcp_receiver_.ConsumeReceivedXrReferenceTimeInfo();
>+  EXPECT_THAT(last_xr_rtis, SizeIs(1));
>   int64_t rtt_ms = 0;
>   EXPECT_TRUE(rtcp_receiver_.GetAndResetXrRrRtt(&rtt_ms));
> }
>@@ -792,8 +828,9 @@ TEST_F(RtcpReceiverTest, InjectExtendedReportsPacketWithUnknownReportBlock) {
>   InjectRtcpPacket(packet);
> 
>   // Validate Rrtr was received and processed.
>-  ReceiveTimeInfo rrtime;
>-  EXPECT_TRUE(rtcp_receiver_.LastReceivedXrReferenceTimeInfo(&rrtime));
>+  std::vector<rtcp::ReceiveTimeInfo> last_xr_rtis =
>+      rtcp_receiver_.ConsumeReceivedXrReferenceTimeInfo();
>+  EXPECT_THAT(last_xr_rtis, SizeIs(1));
>   // Validate Dlrr report wasn't processed.
>   int64_t rtt_ms = 0;
>   EXPECT_FALSE(rtcp_receiver_.GetAndResetXrRrRtt(&rtt_ms));
>@@ -848,12 +885,11 @@ TEST_F(RtcpReceiverTest, XrDlrrCalculatesNegativeRttAsOne) {
>   EXPECT_EQ(1, rtt_ms);
> }
> 
>-TEST_F(RtcpReceiverTest, LastReceivedXrReferenceTimeInfoInitiallyFalse) {
>-  ReceiveTimeInfo info;
>-  EXPECT_FALSE(rtcp_receiver_.LastReceivedXrReferenceTimeInfo(&info));
>+TEST_F(RtcpReceiverTest, ConsumeReceivedXrReferenceTimeInfoInitiallyEmpty) {
>+  EXPECT_THAT(rtcp_receiver_.ConsumeReceivedXrReferenceTimeInfo(), IsEmpty());
> }
> 
>-TEST_F(RtcpReceiverTest, GetLastReceivedExtendedReportsReferenceTimeInfo) {
>+TEST_F(RtcpReceiverTest, ConsumeReceivedXrReferenceTimeInfo) {
>   const NtpTime kNtp(0x10203, 0x40506);
>   const uint32_t kNtpMid = CompactNtp(kNtp);
> 
>@@ -865,15 +901,69 @@ TEST_F(RtcpReceiverTest, GetLastReceivedExtendedReportsReferenceTimeInfo) {
> 
>   InjectRtcpPacket(xr);
> 
>-  ReceiveTimeInfo info;
>-  EXPECT_TRUE(rtcp_receiver_.LastReceivedXrReferenceTimeInfo(&info));
>-  EXPECT_EQ(kSenderSsrc, info.ssrc);
>-  EXPECT_EQ(kNtpMid, info.last_rr);
>-  EXPECT_EQ(0U, info.delay_since_last_rr);
>-
>   system_clock_.AdvanceTimeMilliseconds(1000);
>-  EXPECT_TRUE(rtcp_receiver_.LastReceivedXrReferenceTimeInfo(&info));
>-  EXPECT_EQ(65536U, info.delay_since_last_rr);
>+
>+  std::vector<rtcp::ReceiveTimeInfo> last_xr_rtis =
>+      rtcp_receiver_.ConsumeReceivedXrReferenceTimeInfo();
>+  ASSERT_THAT(last_xr_rtis, SizeIs(1));
>+  EXPECT_EQ(kSenderSsrc, last_xr_rtis[0].ssrc);
>+  EXPECT_EQ(kNtpMid, last_xr_rtis[0].last_rr);
>+  EXPECT_EQ(65536U, last_xr_rtis[0].delay_since_last_rr);
>+}
>+
>+TEST_F(RtcpReceiverTest,
>+       ReceivedRrtrFromSameSsrcUpdatesReceivedReferenceTimeInfo) {
>+  const NtpTime kNtp1(0x10203, 0x40506);
>+  const NtpTime kNtp2(0x11223, 0x44556);
>+  const int64_t kDelayMs = 2000;
>+
>+  rtcp::ExtendedReports xr;
>+  xr.SetSenderSsrc(kSenderSsrc);
>+  rtcp::Rrtr rrtr1;
>+  rrtr1.SetNtp(kNtp1);
>+  xr.SetRrtr(rrtr1);
>+  InjectRtcpPacket(xr);
>+  system_clock_.AdvanceTimeMilliseconds(kDelayMs);
>+  rtcp::Rrtr rrtr2;
>+  rrtr2.SetNtp(kNtp2);
>+  xr.SetRrtr(rrtr2);
>+  InjectRtcpPacket(xr);
>+  system_clock_.AdvanceTimeMilliseconds(kDelayMs);
>+
>+  std::vector<rtcp::ReceiveTimeInfo> last_xr_rtis =
>+      rtcp_receiver_.ConsumeReceivedXrReferenceTimeInfo();
>+  ASSERT_THAT(last_xr_rtis, SizeIs(1));
>+  EXPECT_EQ(kSenderSsrc, last_xr_rtis[0].ssrc);
>+  EXPECT_EQ(CompactNtp(kNtp2), last_xr_rtis[0].last_rr);
>+  EXPECT_EQ(kDelayMs * 65536 / 1000, last_xr_rtis[0].delay_since_last_rr);
>+}
>+
>+TEST_F(RtcpReceiverTest, StoresLastReceivedRrtrPerSsrc) {
>+  const size_t kNumBufferedReports = 1;
>+  const size_t kNumReports =
>+      rtcp::ExtendedReports::kMaxNumberOfDlrrItems + kNumBufferedReports;
>+  for (size_t i = 0; i < kNumReports; ++i) {
>+    rtcp::ExtendedReports xr;
>+    xr.SetSenderSsrc(i * 100);
>+    rtcp::Rrtr rrtr;
>+    rrtr.SetNtp(NtpTime(i * 200, i * 300));
>+    xr.SetRrtr(rrtr);
>+    InjectRtcpPacket(xr);
>+    system_clock_.AdvanceTimeMilliseconds(1000);
>+  }
>+
>+  std::vector<rtcp::ReceiveTimeInfo> last_xr_rtis =
>+      rtcp_receiver_.ConsumeReceivedXrReferenceTimeInfo();
>+  ASSERT_THAT(last_xr_rtis,
>+              SizeIs(rtcp::ExtendedReports::kMaxNumberOfDlrrItems));
>+  for (size_t i = 0; i < rtcp::ExtendedReports::kMaxNumberOfDlrrItems; ++i) {
>+    EXPECT_EQ(i * 100, last_xr_rtis[i].ssrc);
>+    EXPECT_EQ(CompactNtp(NtpTime(i * 200, i * 300)), last_xr_rtis[i].last_rr);
>+    EXPECT_EQ(65536U * (kNumReports - i), last_xr_rtis[i].delay_since_last_rr);
>+  }
>+
>+  last_xr_rtis = rtcp_receiver_.ConsumeReceivedXrReferenceTimeInfo();
>+  ASSERT_THAT(last_xr_rtis, SizeIs(kNumBufferedReports));
> }
> 
> TEST_F(RtcpReceiverTest, ReceiveReportTimeout) {
>@@ -1145,8 +1235,8 @@ TEST_F(RtcpReceiverTest, HandlesInvalidTransportFeedback) {
> 
>   // Modify the TransportFeedback packet so that it is invalid.
>   const size_t kStatusCountOffset = 14;
>-  ByteWriter<uint16_t>::WriteBigEndian(
>-      &built_packet.data()[kStatusCountOffset], 42);
>+  ByteWriter<uint16_t>::WriteBigEndian(&built_packet.data()[kStatusCountOffset],
>+                                       42);
> 
>   // Stress no transport feedback is expected.
>   EXPECT_CALL(transport_feedback_observer_, OnTransportFeedback(_)).Times(0);
>@@ -1229,7 +1319,7 @@ TEST_F(RtcpReceiverTest, ForceSenderReport) {
> }
> 
> TEST_F(RtcpReceiverTest, ReceivesTargetBitrate) {
>-  BitrateAllocation expected_allocation;
>+  VideoBitrateAllocation expected_allocation;
>   expected_allocation.SetBitrate(0, 0, 10000);
>   expected_allocation.SetBitrate(0, 1, 20000);
>   expected_allocation.SetBitrate(1, 0, 40000);
>@@ -1259,7 +1349,7 @@ TEST_F(RtcpReceiverTest, ReceivesTargetBitrate) {
> }
> 
> TEST_F(RtcpReceiverTest, HandlesIncorrectTargetBitrate) {
>-  BitrateAllocation expected_allocation;
>+  VideoBitrateAllocation expected_allocation;
>   expected_allocation.SetBitrate(0, 0, 10000);
> 
>   rtcp::TargetBitrate bitrate;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
>index bd1a5d4b62d..55f545c9675 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_sender.cc
>@@ -14,6 +14,7 @@
> 
> #include <utility>
> 
>+#include "absl/memory/memory.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "logging/rtc_event_log/events/rtc_event_rtcp_packet_outgoing.h"
> #include "logging/rtc_event_log/rtc_event_log.h"
>@@ -37,7 +38,7 @@
> #include "rtc_base/checks.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/ptr_util.h"
>+#include "rtc_base/numerics/safe_conversions.h"
> #include "rtc_base/trace_event.h"
> 
> namespace webrtc {
>@@ -48,35 +49,6 @@ const uint32_t kRtcpAnyExtendedReports =
>     kRtcpXrTargetBitrate;
> }  // namespace
> 
>-NACKStringBuilder::NACKStringBuilder()
>-    : stream_(""), count_(0), prevNack_(0), consecutive_(false) {}
>-
>-NACKStringBuilder::~NACKStringBuilder() {}
>-
>-void NACKStringBuilder::PushNACK(uint16_t nack) {
>-  if (count_ == 0) {
>-    stream_ << nack;
>-  } else if (nack == prevNack_ + 1) {
>-    consecutive_ = true;
>-  } else {
>-    if (consecutive_) {
>-      stream_ << "-" << prevNack_;
>-      consecutive_ = false;
>-    }
>-    stream_ << "," << nack;
>-  }
>-  count_++;
>-  prevNack_ = nack;
>-}
>-
>-std::string NACKStringBuilder::GetResult() {
>-  if (consecutive_) {
>-    stream_ << "-" << prevNack_;
>-    consecutive_ = false;
>-  }
>-  return stream_.str();
>-}
>-
> RTCPSender::FeedbackState::FeedbackState()
>     : packets_sent(0),
>       media_bytes_sent(0),
>@@ -84,14 +56,19 @@ RTCPSender::FeedbackState::FeedbackState()
>       last_rr_ntp_secs(0),
>       last_rr_ntp_frac(0),
>       remote_sr(0),
>-      has_last_xr_rr(false),
>       module(nullptr) {}
> 
>+RTCPSender::FeedbackState::FeedbackState(const FeedbackState&) = default;
>+
>+RTCPSender::FeedbackState::FeedbackState(FeedbackState&&) = default;
>+
>+RTCPSender::FeedbackState::~FeedbackState() = default;
>+
> class PacketContainer : public rtcp::CompoundPacket {
>  public:
>   PacketContainer(Transport* transport, RtcEventLog* event_log)
>       : transport_(transport), event_log_(event_log) {}
>-  virtual ~PacketContainer() {
>+  ~PacketContainer() override {
>     for (RtcpPacket* packet : appended_packets_)
>       delete packet;
>   }
>@@ -102,7 +79,8 @@ class PacketContainer : public rtcp::CompoundPacket {
>       if (transport_->SendRtcp(packet.data(), packet.size())) {
>         bytes_sent += packet.size();
>         if (event_log_) {
>-          event_log_->Log(rtc::MakeUnique<RtcEventRtcpPacketOutgoing>(packet));
>+          event_log_->Log(
>+              absl::make_unique<RtcEventRtcpPacketOutgoing>(packet));
>         }
>       }
>     });
>@@ -139,13 +117,15 @@ RTCPSender::RTCPSender(
>     ReceiveStatisticsProvider* receive_statistics,
>     RtcpPacketTypeCounterObserver* packet_type_counter_observer,
>     RtcEventLog* event_log,
>-    Transport* outgoing_transport)
>+    Transport* outgoing_transport,
>+    RtcpIntervalConfig interval_config)
>     : audio_(audio),
>       clock_(clock),
>       random_(clock_->TimeInMicroseconds()),
>       method_(RtcpMode::kOff),
>       event_log_(event_log),
>       transport_(outgoing_transport),
>+      interval_config_(interval_config),
>       using_nack_(false),
>       sending_(false),
>       next_time_to_send_rtcp_(0),
>@@ -199,9 +179,9 @@ void RTCPSender::SetRTCPStatus(RtcpMode new_method) {
> 
>   if (method_ == RtcpMode::kOff && new_method != RtcpMode::kOff) {
>     // When switching on, reschedule the next packet
>-    next_time_to_send_rtcp_ =
>-      clock_->TimeInMilliseconds() +
>-      (audio_ ? RTCP_INTERVAL_AUDIO_MS / 2 : RTCP_INTERVAL_VIDEO_MS / 2);
>+    int64_t interval_ms = audio_ ? interval_config_.audio_interval_ms
>+                                 : interval_config_.video_interval_ms;
>+    next_time_to_send_rtcp_ = clock_->TimeInMilliseconds() + (interval_ms / 2);
>   }
>   method_ = new_method;
> }
>@@ -343,11 +323,11 @@ int32_t RTCPSender::RemoveMixedCNAME(uint32_t SSRC) {
> 
> bool RTCPSender::TimeToSendRTCPReport(bool sendKeyframeBeforeRTP) const {
>   /*
>-      For audio we use a fix 5 sec interval
>+      For audio we use a configurable interval (default: 5 seconds)
> 
>-      For video we use 1 sec interval fo a BW smaller than 360 kbit/s,
>-          technicaly we break the max 5% RTCP BW for video below 10 kbit/s but
>-          that should be extremely rare
>+      For video we use a configurable interval (default: 1 second) for a BW
>+          smaller than 360 kbit/s, technicaly we break the max 5% RTCP BW for
>+          video below 10 kbit/s but that should be extremely rare
> 
> 
>   From RFC 3550
>@@ -361,8 +341,8 @@ bool RTCPSender::TimeToSendRTCPReport(bool sendKeyframeBeforeRTP) const {
>         is smaller than 5 seconds for bandwidths greater than 72 kb/s.
> 
>       If the participant has not yet sent an RTCP packet (the variable
>-        initial is true), the constant Tmin is set to 2.5 seconds, else it
>-        is set to 5 seconds.
>+        initial is true), the constant Tmin is set to half of the configured
>+        interval.
> 
>       The interval between RTCP packets is varied randomly over the
>         range [0.5,1.5] times the calculated interval to avoid unintended
>@@ -474,11 +454,7 @@ std::unique_ptr<rtcp::RtcpPacket> RTCPSender::BuildPLI(const RtcpContext& ctx) {
>   pli->SetSenderSsrc(ssrc_);
>   pli->SetMediaSsrc(remote_ssrc_);
> 
>-  TRACE_EVENT_INSTANT0(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
>-                       "RTCPSender::PLI");
>   ++packet_type_counter_.pli_packets;
>-  TRACE_COUNTER_ID1(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "RTCP_PLICount",
>-                    ssrc_, packet_type_counter_.pli_packets);
> 
>   return std::unique_ptr<rtcp::RtcpPacket>(pli);
> }
>@@ -490,11 +466,7 @@ std::unique_ptr<rtcp::RtcpPacket> RTCPSender::BuildFIR(const RtcpContext& ctx) {
>   fir->SetSenderSsrc(ssrc_);
>   fir->AddRequestTo(remote_ssrc_, sequence_number_fir_);
> 
>-  TRACE_EVENT_INSTANT0(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
>-                       "RTCPSender::FIR");
>   ++packet_type_counter_.fir_packets;
>-  TRACE_COUNTER_ID1(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "RTCP_FIRCount",
>-                    ssrc_, packet_type_counter_.fir_packets);
> 
>   return std::unique_ptr<rtcp::RtcpPacket>(fir);
> }
>@@ -506,9 +478,6 @@ std::unique_ptr<rtcp::RtcpPacket> RTCPSender::BuildREMB(
>   remb->SetBitrateBps(remb_bitrate_);
>   remb->SetSsrcs(remb_ssrcs_);
> 
>-  TRACE_EVENT_INSTANT0(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
>-                       "RTCPSender::REMB");
>-
>   return std::unique_ptr<rtcp::RtcpPacket>(remb);
> }
> 
>@@ -604,20 +573,13 @@ std::unique_ptr<rtcp::RtcpPacket> RTCPSender::BuildNACK(
>   nack->SetPacketIds(ctx.nack_list_, ctx.nack_size_);
> 
>   // Report stats.
>-  NACKStringBuilder stringBuilder;
>   for (int idx = 0; idx < ctx.nack_size_; ++idx) {
>-    stringBuilder.PushNACK(ctx.nack_list_[idx]);
>     nack_stats_.ReportRequest(ctx.nack_list_[idx]);
>   }
>   packet_type_counter_.nack_requests = nack_stats_.requests();
>   packet_type_counter_.unique_nack_requests = nack_stats_.unique_requests();
> 
>-  TRACE_EVENT_INSTANT1(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
>-                       "RTCPSender::NACK", "nacks",
>-                       TRACE_STR_COPY(stringBuilder.GetResult().c_str()));
>   ++packet_type_counter_.nack_packets;
>-  TRACE_COUNTER_ID1(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "RTCP_NACKCount",
>-                    ssrc_, packet_type_counter_.nack_packets);
> 
>   return std::unique_ptr<rtcp::RtcpPacket>(nack);
> }
>@@ -641,8 +603,8 @@ std::unique_ptr<rtcp::RtcpPacket> RTCPSender::BuildExtendedReports(
>     xr->SetRrtr(rrtr);
>   }
> 
>-  if (ctx.feedback_state_.has_last_xr_rr) {
>-    xr->AddDlrrItem(ctx.feedback_state_.last_xr_rr);
>+  for (const rtcp::ReceiveTimeInfo& rti : ctx.feedback_state_.last_xr_rtis) {
>+    xr->AddDlrrItem(rti);
>   }
> 
>   if (video_bitrate_allocation_) {
>@@ -788,12 +750,13 @@ void RTCPSender::PrepareReport(const FeedbackState& feedback_state) {
> 
>   if (generate_report) {
>     if ((!sending_ && xr_send_receiver_reference_time_enabled_) ||
>-        feedback_state.has_last_xr_rr || video_bitrate_allocation_) {
>+        !feedback_state.last_xr_rtis.empty() || video_bitrate_allocation_) {
>       SetFlag(kRtcpAnyExtendedReports, true);
>     }
> 
>     // generate next time to send an RTCP report
>-    uint32_t minIntervalMs = RTCP_INTERVAL_AUDIO_MS;
>+    uint32_t minIntervalMs =
>+        rtc::dchecked_cast<uint32_t>(interval_config_.audio_interval_ms);
> 
>     if (!audio_) {
>       if (sending_) {
>@@ -802,9 +765,13 @@ void RTCPSender::PrepareReport(const FeedbackState& feedback_state) {
>         if (send_bitrate_kbit != 0)
>           minIntervalMs = 360000 / send_bitrate_kbit;
>       }
>-      if (minIntervalMs > RTCP_INTERVAL_VIDEO_MS)
>-        minIntervalMs = RTCP_INTERVAL_VIDEO_MS;
>+      if (minIntervalMs >
>+          rtc::dchecked_cast<uint32_t>(interval_config_.video_interval_ms)) {
>+        minIntervalMs =
>+            rtc::dchecked_cast<uint32_t>(interval_config_.video_interval_ms);
>+      }
>     }
>+
>     // The interval between RTCP packets is varied randomly over the
>     // range [1/2,3/2] times the calculated interval.
>     uint32_t timeToNext =
>@@ -934,7 +901,8 @@ bool RTCPSender::AllVolatileFlagsConsumed() const {
>   return true;
> }
> 
>-void RTCPSender::SetVideoBitrateAllocation(const BitrateAllocation& bitrate) {
>+void RTCPSender::SetVideoBitrateAllocation(
>+    const VideoBitrateAllocation& bitrate) {
>   rtc::CritScope lock(&critical_section_rtcp_sender_);
>   video_bitrate_allocation_.emplace(bitrate);
>   SetFlag(kRtcpAnyExtendedReports, true);
>@@ -954,7 +922,7 @@ bool RTCPSender::SendFeedbackPacket(const rtcp::TransportFeedback& packet) {
>   auto callback = [&](rtc::ArrayView<const uint8_t> packet) {
>     if (transport_->SendRtcp(packet.data(), packet.size())) {
>       if (event_log_)
>-        event_log_->Log(rtc::MakeUnique<RtcEventRtcpPacketOutgoing>(packet));
>+        event_log_->Log(absl::make_unique<RtcEventRtcpPacketOutgoing>(packet));
>     } else {
>       send_failure = true;
>     }
>@@ -962,4 +930,12 @@ bool RTCPSender::SendFeedbackPacket(const rtcp::TransportFeedback& packet) {
>   return packet.Build(max_packet_size, callback) && !send_failure;
> }
> 
>+int64_t RTCPSender::RtcpAudioReportInverval() const {
>+  return interval_config_.audio_interval_ms;
>+}
>+
>+int64_t RTCPSender::RtcpVideoReportInverval() const {
>+  return interval_config_.video_interval_ms;
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_sender.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
>index e56f66ce628..133bf6809c2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_sender.h
>@@ -14,12 +14,12 @@
> #include <map>
> #include <memory>
> #include <set>
>-#include <sstream>
> #include <string>
> #include <vector>
> 
>+#include "absl/types/optional.h"
> #include "api/call/transport.h"
>-#include "api/optional.h"
>+#include "api/video/video_bitrate_allocation.h"
> #include "modules/remote_bitrate_estimator/include/bwe_defines.h"
> #include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
> #include "modules/rtp_rtcp/include/receive_statistics.h"
>@@ -33,32 +33,20 @@
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/random.h"
> #include "rtc_base/thread_annotations.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
> class ModuleRtpRtcpImpl;
> class RtcEventLog;
> 
>-class NACKStringBuilder {
>- public:
>-  NACKStringBuilder();
>-  ~NACKStringBuilder();
>-
>-  void PushNACK(uint16_t nack);
>-  std::string GetResult();
>-
>- private:
>-  std::ostringstream stream_;
>-  int count_;
>-  uint16_t prevNack_;
>-  bool consecutive_;
>-};
>-
> class RTCPSender {
>  public:
>   struct FeedbackState {
>     FeedbackState();
>+    FeedbackState(const FeedbackState&);
>+    FeedbackState(FeedbackState&&);
>+
>+    ~FeedbackState();
> 
>     uint32_t packets_sent;
>     size_t media_bytes_sent;
>@@ -68,8 +56,7 @@ class RTCPSender {
>     uint32_t last_rr_ntp_frac;
>     uint32_t remote_sr;
> 
>-    bool has_last_xr_rr;
>-    rtcp::ReceiveTimeInfo last_xr_rr;
>+    std::vector<rtcp::ReceiveTimeInfo> last_xr_rtis;
> 
>     // Used when generating TMMBR.
>     ModuleRtpRtcpImpl* module;
>@@ -80,7 +67,8 @@ class RTCPSender {
>              ReceiveStatisticsProvider* receive_statistics,
>              RtcpPacketTypeCounterObserver* packet_type_counter_observer,
>              RtcEventLog* event_log,
>-             Transport* outgoing_transport);
>+             Transport* outgoing_transport,
>+             RtcpIntervalConfig interval_config);
>   virtual ~RTCPSender();
> 
>   RtcpMode Status() const;
>@@ -145,9 +133,12 @@ class RTCPSender {
>   void SetCsrcs(const std::vector<uint32_t>& csrcs);
> 
>   void SetTargetBitrate(unsigned int target_bitrate);
>-  void SetVideoBitrateAllocation(const BitrateAllocation& bitrate);
>+  void SetVideoBitrateAllocation(const VideoBitrateAllocation& bitrate);
>   bool SendFeedbackPacket(const rtcp::TransportFeedback& packet);
> 
>+  int64_t RtcpAudioReportInverval() const;
>+  int64_t RtcpVideoReportInverval() const;
>+
>  private:
>   class RtcpContext;
> 
>@@ -194,6 +185,8 @@ class RTCPSender {
>   RtcEventLog* const event_log_;
>   Transport* const transport_;
> 
>+  const RtcpIntervalConfig interval_config_;
>+
>   rtc::CriticalSection critical_section_rtcp_sender_;
>   bool using_nack_ RTC_GUARDED_BY(critical_section_rtcp_sender_);
>   bool sending_ RTC_GUARDED_BY(critical_section_rtcp_sender_);
>@@ -243,7 +236,7 @@ class RTCPSender {
>       RTC_GUARDED_BY(critical_section_rtcp_sender_);
> 
>   // XR VoIP metric
>-  rtc::Optional<RTCPVoIPMetric> xr_voip_metric_
>+  absl::optional<RTCPVoIPMetric> xr_voip_metric_
>       RTC_GUARDED_BY(critical_section_rtcp_sender_);
> 
>   RtcpPacketTypeCounterObserver* const packet_type_counter_observer_;
>@@ -252,7 +245,7 @@ class RTCPSender {
> 
>   RtcpNackStats nack_stats_ RTC_GUARDED_BY(critical_section_rtcp_sender_);
> 
>-  rtc::Optional<BitrateAllocation> video_bitrate_allocation_
>+  absl::optional<VideoBitrateAllocation> video_bitrate_allocation_
>       RTC_GUARDED_BY(critical_section_rtcp_sender_);
> 
>   void SetFlag(uint32_t type, bool is_volatile)
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
>index 4549c674d89..974f19f5ee0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_sender_unittest.cc
>@@ -24,168 +24,14 @@
> using ::testing::_;
> using ::testing::ElementsAre;
> using ::testing::Invoke;
>+using ::testing::SizeIs;
> 
> namespace webrtc {
> 
>-TEST(NACKStringBuilderTest, TestCase1) {
>-  NACKStringBuilder builder;
>-  builder.PushNACK(5);
>-  builder.PushNACK(7);
>-  builder.PushNACK(9);
>-  builder.PushNACK(10);
>-  builder.PushNACK(11);
>-  builder.PushNACK(12);
>-  builder.PushNACK(15);
>-  builder.PushNACK(18);
>-  builder.PushNACK(19);
>-  EXPECT_EQ(std::string("5,7,9-12,15,18-19"), builder.GetResult());
>-}
>-
>-TEST(NACKStringBuilderTest, TestCase2) {
>-  NACKStringBuilder builder;
>-  builder.PushNACK(5);
>-  builder.PushNACK(6);
>-  builder.PushNACK(7);
>-  builder.PushNACK(9);
>-  builder.PushNACK(10);
>-  builder.PushNACK(11);
>-  builder.PushNACK(12);
>-  builder.PushNACK(15);
>-  builder.PushNACK(18);
>-  builder.PushNACK(19);
>-  EXPECT_EQ(std::string("5-7,9-12,15,18-19"), builder.GetResult());
>-}
>-
>-TEST(NACKStringBuilderTest, TestCase3) {
>-  NACKStringBuilder builder;
>-  builder.PushNACK(5);
>-  builder.PushNACK(7);
>-  builder.PushNACK(9);
>-  builder.PushNACK(10);
>-  builder.PushNACK(11);
>-  builder.PushNACK(12);
>-  builder.PushNACK(15);
>-  builder.PushNACK(18);
>-  builder.PushNACK(19);
>-  builder.PushNACK(21);
>-  EXPECT_EQ(std::string("5,7,9-12,15,18-19,21"), builder.GetResult());
>-}
>-
>-TEST(NACKStringBuilderTest, TestCase4) {
>-  NACKStringBuilder builder;
>-  builder.PushNACK(5);
>-  builder.PushNACK(7);
>-  builder.PushNACK(8);
>-  builder.PushNACK(9);
>-  builder.PushNACK(10);
>-  builder.PushNACK(11);
>-  builder.PushNACK(12);
>-  builder.PushNACK(15);
>-  builder.PushNACK(18);
>-  builder.PushNACK(19);
>-  EXPECT_EQ(std::string("5,7-12,15,18-19"), builder.GetResult());
>-}
>-
>-TEST(NACKStringBuilderTest, TestCase5) {
>-  NACKStringBuilder builder;
>-  builder.PushNACK(5);
>-  builder.PushNACK(7);
>-  builder.PushNACK(9);
>-  builder.PushNACK(10);
>-  builder.PushNACK(11);
>-  builder.PushNACK(12);
>-  builder.PushNACK(15);
>-  builder.PushNACK(16);
>-  builder.PushNACK(18);
>-  builder.PushNACK(19);
>-  EXPECT_EQ(std::string("5,7,9-12,15-16,18-19"), builder.GetResult());
>-}
>-
>-TEST(NACKStringBuilderTest, TestCase6) {
>-  NACKStringBuilder builder;
>-  builder.PushNACK(5);
>-  builder.PushNACK(7);
>-  builder.PushNACK(9);
>-  builder.PushNACK(10);
>-  builder.PushNACK(11);
>-  builder.PushNACK(12);
>-  builder.PushNACK(15);
>-  builder.PushNACK(16);
>-  builder.PushNACK(17);
>-  builder.PushNACK(18);
>-  builder.PushNACK(19);
>-  EXPECT_EQ(std::string("5,7,9-12,15-19"), builder.GetResult());
>-}
>-
>-TEST(NACKStringBuilderTest, TestCase7) {
>-  NACKStringBuilder builder;
>-  builder.PushNACK(5);
>-  builder.PushNACK(6);
>-  builder.PushNACK(7);
>-  builder.PushNACK(8);
>-  builder.PushNACK(11);
>-  builder.PushNACK(12);
>-  builder.PushNACK(13);
>-  builder.PushNACK(14);
>-  builder.PushNACK(15);
>-  EXPECT_EQ(std::string("5-8,11-15"), builder.GetResult());
>-}
>-
>-TEST(NACKStringBuilderTest, TestCase8) {
>-  NACKStringBuilder builder;
>-  builder.PushNACK(5);
>-  builder.PushNACK(7);
>-  builder.PushNACK(9);
>-  builder.PushNACK(11);
>-  builder.PushNACK(15);
>-  builder.PushNACK(17);
>-  builder.PushNACK(19);
>-  EXPECT_EQ(std::string("5,7,9,11,15,17,19"), builder.GetResult());
>-}
>-
>-TEST(NACKStringBuilderTest, TestCase9) {
>-  NACKStringBuilder builder;
>-  builder.PushNACK(5);
>-  builder.PushNACK(6);
>-  builder.PushNACK(7);
>-  builder.PushNACK(8);
>-  builder.PushNACK(9);
>-  builder.PushNACK(10);
>-  builder.PushNACK(11);
>-  builder.PushNACK(12);
>-  EXPECT_EQ(std::string("5-12"), builder.GetResult());
>-}
>-
>-TEST(NACKStringBuilderTest, TestCase10) {
>-  NACKStringBuilder builder;
>-  builder.PushNACK(5);
>-  EXPECT_EQ(std::string("5"), builder.GetResult());
>-}
>-
>-TEST(NACKStringBuilderTest, TestCase11) {
>-  NACKStringBuilder builder;
>-  EXPECT_EQ(std::string(""), builder.GetResult());
>-}
>-
>-TEST(NACKStringBuilderTest, TestCase12) {
>-  NACKStringBuilder builder;
>-  builder.PushNACK(5);
>-  builder.PushNACK(6);
>-  EXPECT_EQ(std::string("5-6"), builder.GetResult());
>-}
>-
>-TEST(NACKStringBuilderTest, TestCase13) {
>-  NACKStringBuilder builder;
>-  builder.PushNACK(5);
>-  builder.PushNACK(6);
>-  builder.PushNACK(9);
>-  EXPECT_EQ(std::string("5-6,9"), builder.GetResult());
>-}
>-
> class RtcpPacketTypeCounterObserverImpl : public RtcpPacketTypeCounterObserver {
>  public:
>   RtcpPacketTypeCounterObserverImpl() : ssrc_(0) {}
>-  virtual ~RtcpPacketTypeCounterObserverImpl() {}
>+  ~RtcpPacketTypeCounterObserverImpl() override = default;
>   void RtcpPacketTypesCounterUpdated(
>       uint32_t ssrc,
>       const RtcpPacketTypeCounter& packet_counter) override {
>@@ -196,8 +42,7 @@ class RtcpPacketTypeCounterObserverImpl : public RtcpPacketTypeCounterObserver {
>   RtcpPacketTypeCounter counter_;
> };
> 
>-class TestTransport : public Transport,
>-                      public RtpData {
>+class TestTransport : public Transport, public RtpData {
>  public:
>   TestTransport() {}
> 
>@@ -223,7 +68,7 @@ static const uint32_t kSenderSsrc = 0x11111111;
> static const uint32_t kRemoteSsrc = 0x22222222;
> static const uint32_t kStartRtpTimestamp = 0x34567;
> static const uint32_t kRtpTimestamp = 0x45678;
>-}
>+}  // namespace
> 
> class RtcpSenderTest : public ::testing::Test {
>  protected:
>@@ -239,7 +84,8 @@ class RtcpSenderTest : public ::testing::Test {
> 
>     rtp_rtcp_impl_.reset(new ModuleRtpRtcpImpl(configuration));
>     rtcp_sender_.reset(new RTCPSender(false, &clock_, receive_statistics_.get(),
>-                                      nullptr, nullptr, &test_transport_));
>+                                      nullptr, nullptr, &test_transport_,
>+                                      configuration.rtcp_interval_config));
>     rtcp_sender_->SetSSRC(kSenderSsrc);
>     rtcp_sender_->SetRemoteSSRC(kRemoteSsrc);
>     rtcp_sender_->SetTimestampOffset(kStartRtpTimestamp);
>@@ -309,7 +155,8 @@ TEST_F(RtcpSenderTest, SendSr) {
> 
> TEST_F(RtcpSenderTest, DoNotSendSrBeforeRtp) {
>   rtcp_sender_.reset(new RTCPSender(false, &clock_, receive_statistics_.get(),
>-                                    nullptr, nullptr, &test_transport_));
>+                                    nullptr, nullptr, &test_transport_,
>+                                    RtcpIntervalConfig{}));
>   rtcp_sender_->SetSSRC(kSenderSsrc);
>   rtcp_sender_->SetRemoteSSRC(kRemoteSsrc);
>   rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize);
>@@ -327,7 +174,8 @@ TEST_F(RtcpSenderTest, DoNotSendSrBeforeRtp) {
> 
> TEST_F(RtcpSenderTest, DoNotSendCompundBeforeRtp) {
>   rtcp_sender_.reset(new RTCPSender(false, &clock_, receive_statistics_.get(),
>-                                    nullptr, nullptr, &test_transport_));
>+                                    nullptr, nullptr, &test_transport_,
>+                                    RtcpIntervalConfig{}));
>   rtcp_sender_->SetSSRC(kSenderSsrc);
>   rtcp_sender_->SetRemoteSSRC(kRemoteSsrc);
>   rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound);
>@@ -460,8 +308,9 @@ TEST_F(RtcpSenderTest, SendEmptyApp) {
> TEST_F(RtcpSenderTest, SetInvalidApplicationSpecificData) {
>   const uint8_t kData[] = {'t', 'e', 's', 't', 'd', 'a', 't'};
>   const uint16_t kInvalidDataLength = sizeof(kData) / sizeof(kData[0]);
>-  EXPECT_EQ(-1, rtcp_sender_->SetApplicationSpecificData(
>-      0, 0, kData, kInvalidDataLength));  // Should by multiple of 4.
>+  EXPECT_EQ(-1,
>+            rtcp_sender_->SetApplicationSpecificData(
>+                0, 0, kData, kInvalidDataLength));  // Should by multiple of 4.
> }
> 
> TEST_F(RtcpSenderTest, SendFir) {
>@@ -605,22 +454,47 @@ TEST_F(RtcpSenderTest, SendXrWithVoipMetric) {
> TEST_F(RtcpSenderTest, SendXrWithDlrr) {
>   rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound);
>   RTCPSender::FeedbackState feedback_state = rtp_rtcp_impl_->GetFeedbackState();
>-  feedback_state.has_last_xr_rr = true;
>   rtcp::ReceiveTimeInfo last_xr_rr;
>   last_xr_rr.ssrc = 0x11111111;
>   last_xr_rr.last_rr = 0x22222222;
>   last_xr_rr.delay_since_last_rr = 0x33333333;
>-  feedback_state.last_xr_rr = last_xr_rr;
>+  feedback_state.last_xr_rtis.push_back(last_xr_rr);
>   EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state, kRtcpReport));
>   EXPECT_EQ(1, parser()->xr()->num_packets());
>   EXPECT_EQ(kSenderSsrc, parser()->xr()->sender_ssrc());
>-  EXPECT_EQ(1U, parser()->xr()->dlrr().sub_blocks().size());
>+  ASSERT_THAT(parser()->xr()->dlrr().sub_blocks(), SizeIs(1));
>   EXPECT_EQ(last_xr_rr.ssrc, parser()->xr()->dlrr().sub_blocks()[0].ssrc);
>   EXPECT_EQ(last_xr_rr.last_rr, parser()->xr()->dlrr().sub_blocks()[0].last_rr);
>   EXPECT_EQ(last_xr_rr.delay_since_last_rr,
>             parser()->xr()->dlrr().sub_blocks()[0].delay_since_last_rr);
> }
> 
>+TEST_F(RtcpSenderTest, SendXrWithMultipleDlrrSubBlocks) {
>+  const size_t kNumReceivers = 2;
>+  rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound);
>+  RTCPSender::FeedbackState feedback_state = rtp_rtcp_impl_->GetFeedbackState();
>+  for (size_t i = 0; i < kNumReceivers; ++i) {
>+    rtcp::ReceiveTimeInfo last_xr_rr;
>+    last_xr_rr.ssrc = i;
>+    last_xr_rr.last_rr = (i + 1) * 100;
>+    last_xr_rr.delay_since_last_rr = (i + 2) * 200;
>+    feedback_state.last_xr_rtis.push_back(last_xr_rr);
>+  }
>+
>+  EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state, kRtcpReport));
>+  EXPECT_EQ(1, parser()->xr()->num_packets());
>+  EXPECT_EQ(kSenderSsrc, parser()->xr()->sender_ssrc());
>+  ASSERT_THAT(parser()->xr()->dlrr().sub_blocks(), SizeIs(kNumReceivers));
>+  for (size_t i = 0; i < kNumReceivers; ++i) {
>+    EXPECT_EQ(feedback_state.last_xr_rtis[i].ssrc,
>+              parser()->xr()->dlrr().sub_blocks()[i].ssrc);
>+    EXPECT_EQ(feedback_state.last_xr_rtis[i].last_rr,
>+              parser()->xr()->dlrr().sub_blocks()[i].last_rr);
>+    EXPECT_EQ(feedback_state.last_xr_rtis[i].delay_since_last_rr,
>+              parser()->xr()->dlrr().sub_blocks()[i].delay_since_last_rr);
>+  }
>+}
>+
> TEST_F(RtcpSenderTest, SendXrWithRrtr) {
>   rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound);
>   EXPECT_EQ(0, rtcp_sender_->SetSendingStatus(feedback_state(), false));
>@@ -654,7 +528,8 @@ TEST_F(RtcpSenderTest, TestNoXrRrtrSentIfNotEnabled) {
> TEST_F(RtcpSenderTest, TestRegisterRtcpPacketTypeObserver) {
>   RtcpPacketTypeCounterObserverImpl observer;
>   rtcp_sender_.reset(new RTCPSender(false, &clock_, receive_statistics_.get(),
>-                                    &observer, nullptr, &test_transport_));
>+                                    &observer, nullptr, &test_transport_,
>+                                    RtcpIntervalConfig{}));
>   rtcp_sender_->SetRemoteSSRC(kRemoteSsrc);
>   rtcp_sender_->SetRTCPStatus(RtcpMode::kReducedSize);
>   EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpPli));
>@@ -755,27 +630,29 @@ TEST_F(RtcpSenderTest, SendCompoundPliRemb) {
> TEST_F(RtcpSenderTest, ByeMustBeLast) {
>   MockTransport mock_transport;
>   EXPECT_CALL(mock_transport, SendRtcp(_, _))
>-    .WillOnce(Invoke([](const uint8_t* data, size_t len) {
>-    const uint8_t* next_packet = data;
>-    const uint8_t* const packet_end = data + len;
>-    rtcp::CommonHeader packet;
>-    while (next_packet < packet_end) {
>-      EXPECT_TRUE(packet.Parse(next_packet, packet_end - next_packet));
>-      next_packet = packet.NextPacket();
>-      if (packet.type() == rtcp::Bye::kPacketType)  // Main test expectation.
>-        EXPECT_EQ(0, packet_end - next_packet)
>-            << "Bye packet should be last in a compound RTCP packet.";
>-      if (next_packet == packet_end)  // Validate test was set correctly.
>-        EXPECT_EQ(packet.type(), rtcp::Bye::kPacketType)
>-            << "Last packet in this test expected to be Bye.";
>-    }
>-
>-    return true;
>-  }));
>+      .WillOnce(Invoke([](const uint8_t* data, size_t len) {
>+        const uint8_t* next_packet = data;
>+        const uint8_t* const packet_end = data + len;
>+        rtcp::CommonHeader packet;
>+        while (next_packet < packet_end) {
>+          EXPECT_TRUE(packet.Parse(next_packet, packet_end - next_packet));
>+          next_packet = packet.NextPacket();
>+          if (packet.type() ==
>+              rtcp::Bye::kPacketType)  // Main test expectation.
>+            EXPECT_EQ(0, packet_end - next_packet)
>+                << "Bye packet should be last in a compound RTCP packet.";
>+          if (next_packet == packet_end)  // Validate test was set correctly.
>+            EXPECT_EQ(packet.type(), rtcp::Bye::kPacketType)
>+                << "Last packet in this test expected to be Bye.";
>+        }
>+
>+        return true;
>+      }));
> 
>   // Re-configure rtcp_sender_ with mock_transport_
>   rtcp_sender_.reset(new RTCPSender(false, &clock_, receive_statistics_.get(),
>-                                    nullptr, nullptr, &mock_transport));
>+                                    nullptr, nullptr, &mock_transport,
>+                                    RtcpIntervalConfig{}));
>   rtcp_sender_->SetSSRC(kSenderSsrc);
>   rtcp_sender_->SetRemoteSSRC(kRemoteSsrc);
>   rtcp_sender_->SetTimestampOffset(kStartRtpTimestamp);
>@@ -792,7 +669,7 @@ TEST_F(RtcpSenderTest, SendXrWithTargetBitrate) {
>   rtcp_sender_->SetRTCPStatus(RtcpMode::kCompound);
>   const size_t kNumSpatialLayers = 2;
>   const size_t kNumTemporalLayers = 2;
>-  BitrateAllocation allocation;
>+  VideoBitrateAllocation allocation;
>   for (size_t sl = 0; sl < kNumSpatialLayers; ++sl) {
>     uint32_t start_bitrate_bps = (sl + 1) * 100000;
>     for (size_t tl = 0; tl < kNumTemporalLayers; ++tl)
>@@ -803,7 +680,7 @@ TEST_F(RtcpSenderTest, SendXrWithTargetBitrate) {
>   EXPECT_EQ(0, rtcp_sender_->SendRTCP(feedback_state(), kRtcpReport));
>   EXPECT_EQ(1, parser()->xr()->num_packets());
>   EXPECT_EQ(kSenderSsrc, parser()->xr()->sender_ssrc());
>-  const rtc::Optional<rtcp::TargetBitrate>& target_bitrate =
>+  const absl::optional<rtcp::TargetBitrate>& target_bitrate =
>       parser()->xr()->target_bitrate();
>   ASSERT_TRUE(target_bitrate);
>   const std::vector<rtcp::TargetBitrate::BitrateItem>& bitrates =
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_transceiver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_transceiver.cc
>index a937131e414..9a179295b29 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_transceiver.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_transceiver.cc
>@@ -12,17 +12,17 @@
> 
> #include <utility>
> 
>+#include "absl/memory/memory.h"
> #include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/event.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/timeutils.h"
> 
> namespace webrtc {
> 
> RtcpTransceiver::RtcpTransceiver(const RtcpTransceiverConfig& config)
>     : task_queue_(config.task_queue),
>-      rtcp_transceiver_(rtc::MakeUnique<RtcpTransceiverImpl>(config)),
>+      rtcp_transceiver_(absl::make_unique<RtcpTransceiverImpl>(config)),
>       ptr_factory_(rtcp_transceiver_.get()),
>       // Creating first weak ptr can be done on any thread, but is not
>       // thread-safe, thus do it at construction. Creating second (e.g. making a
>@@ -73,6 +73,14 @@ void RtcpTransceiver::RemoveMediaReceiverRtcpObserver(
>   task_queue_->PostTaskAndReply(std::move(remove), std::move(on_removed));
> }
> 
>+void RtcpTransceiver::SetReadyToSend(bool ready) {
>+  rtc::WeakPtr<RtcpTransceiverImpl> ptr = ptr_;
>+  task_queue_->PostTask([ptr, ready] {
>+    if (ptr)
>+      ptr->SetReadyToSend(ready);
>+  });
>+}
>+
> void RtcpTransceiver::ReceivePacket(rtc::CopyOnWriteBuffer packet) {
>   rtc::WeakPtr<RtcpTransceiverImpl> ptr = ptr_;
>   int64_t now_us = rtc::TimeMicros();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_transceiver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_transceiver.h
>index 9df56d14e64..9125781ea56 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_transceiver.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_transceiver.h
>@@ -43,6 +43,11 @@ class RtcpTransceiver : public RtcpFeedbackSenderInterface {
>       MediaReceiverRtcpObserver* observer,
>       std::unique_ptr<rtc::QueuedTask> on_removed);
> 
>+  // Enables/disables sending rtcp packets eventually.
>+  // Packets may be sent after the SetReadyToSend(false) returns, but no new
>+  // packets will be scheduled.
>+  void SetReadyToSend(bool ready);
>+
>   // Handles incoming rtcp packets.
>   void ReceivePacket(rtc::CopyOnWriteBuffer packet);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_transceiver_config.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_transceiver_config.h
>index 4b08d89e0e0..0da18e2cc1b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_transceiver_config.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_transceiver_config.h
>@@ -13,9 +13,11 @@
> 
> #include <string>
> 
>-#include "common_types.h"  // NOLINT(build/include)
>+#include "api/rtp_headers.h"
>+#include "api/video/video_bitrate_allocation.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> #include "rtc_base/task_queue.h"
>+#include "system_wrappers/include/ntp_time.h"
> 
> namespace webrtc {
> class ReceiveStatisticsProvider;
>@@ -33,7 +35,7 @@ class MediaReceiverRtcpObserver {
>                               uint32_t rtp_time) {}
>   virtual void OnBye(uint32_t sender_ssrc) {}
>   virtual void OnBitrateAllocation(uint32_t sender_ssrc,
>-                                   const BitrateAllocation& allocation) {}
>+                                   const VideoBitrateAllocation& allocation) {}
> };
> 
> struct RtcpTransceiverConfig {
>@@ -80,6 +82,8 @@ struct RtcpTransceiverConfig {
>   //
>   // Tuning parameters.
>   //
>+  // Initial state if |outgoing_transport| ready to accept packets.
>+  bool initial_ready_to_send = true;
>   // Delay before 1st periodic compound packet.
>   int initial_report_delay_ms = 500;
> 
>@@ -93,6 +97,11 @@ struct RtcpTransceiverConfig {
>   // Estimate RTT as non-sender as described in
>   // https://tools.ietf.org/html/rfc3611#section-4.4 and #section-4.5
>   bool non_sender_rtt_measurement = false;
>+  // Copies LastSR/DelaySinceLastSR for previous report block to avoid
>+  // triggering bug in older version of RtcpReceiver.
>+  // TODO(bugs.webrtc.org/8805): Change to false by default then remove when
>+  // all major webrtc clients updated with the fix in RtcpReceiver.
>+  bool avoid_zero_last_sr_in_last_report_block = true;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_transceiver_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_transceiver_impl.cc
>index 2ac1c6bbc49..a4da63a10f0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_transceiver_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_transceiver_impl.cc
>@@ -12,7 +12,9 @@
> 
> #include <utility>
> 
>+#include "absl/memory/memory.h"
> #include "api/call/transport.h"
>+#include "api/video/video_bitrate_allocation.h"
> #include "modules/rtp_rtcp/include/receive_statistics.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> #include "modules/rtp_rtcp/source/rtcp_packet.h"
>@@ -29,7 +31,6 @@
> #include "modules/rtp_rtcp/source/time_util.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/task_queue.h"
> #include "rtc_base/timeutils.h"
> 
>@@ -45,7 +46,7 @@ struct SenderReportTimes {
> 
> struct RtcpTransceiverImpl::RemoteSenderState {
>   uint8_t fir_sequence_number = 0;
>-  rtc::Optional<SenderReportTimes> last_received_sender_report;
>+  absl::optional<SenderReportTimes> last_received_sender_report;
>   std::vector<MediaReceiverRtcpObserver*> observers;
> };
> 
>@@ -86,9 +87,11 @@ class RtcpTransceiverImpl::PacketSender {
> };
> 
> RtcpTransceiverImpl::RtcpTransceiverImpl(const RtcpTransceiverConfig& config)
>-    : config_(config), ptr_factory_(this) {
>+    : config_(config),
>+      ready_to_send_(config.initial_ready_to_send),
>+      ptr_factory_(this) {
>   RTC_CHECK(config_.Validate());
>-  if (config_.schedule_periodic_compound_packets)
>+  if (ready_to_send_ && config_.schedule_periodic_compound_packets)
>     SchedulePeriodicCompoundPackets(config_.initial_report_delay_ms);
> }
> 
>@@ -115,6 +118,17 @@ void RtcpTransceiverImpl::RemoveMediaReceiverRtcpObserver(
>   stored.erase(it);
> }
> 
>+void RtcpTransceiverImpl::SetReadyToSend(bool ready) {
>+  if (config_.schedule_periodic_compound_packets) {
>+    if (ready_to_send_ && !ready)  // Stop existent send task.
>+      ptr_factory_.InvalidateWeakPtrs();
>+
>+    if (!ready_to_send_ && ready)  // Restart periodic sending.
>+      SchedulePeriodicCompoundPackets(config_.report_period_ms / 2);
>+  }
>+  ready_to_send_ = ready;
>+}
>+
> void RtcpTransceiverImpl::ReceivePacket(rtc::ArrayView<const uint8_t> packet,
>                                         int64_t now_us) {
>   while (!packet.empty()) {
>@@ -130,6 +144,8 @@ void RtcpTransceiverImpl::ReceivePacket(rtc::ArrayView<const uint8_t> packet,
> }
> 
> void RtcpTransceiverImpl::SendCompoundPacket() {
>+  if (!ready_to_send_)
>+    return;
>   SendPeriodicCompoundPacket();
>   ReschedulePeriodicCompoundPackets();
> }
>@@ -150,6 +166,8 @@ void RtcpTransceiverImpl::UnsetRemb() {
> }
> 
> void RtcpTransceiverImpl::SendRawPacket(rtc::ArrayView<const uint8_t> packet) {
>+  if (!ready_to_send_)
>+    return;
>   // Unlike other senders, this functions just tries to send packet away and
>   // disregard rtcp_mode, max_packet_size or anything else.
>   // TODO(bugs.webrtc.org/8239): respect config_ by creating the
>@@ -160,6 +178,8 @@ void RtcpTransceiverImpl::SendRawPacket(rtc::ArrayView<const uint8_t> packet) {
> void RtcpTransceiverImpl::SendNack(uint32_t ssrc,
>                                    std::vector<uint16_t> sequence_numbers) {
>   RTC_DCHECK(!sequence_numbers.empty());
>+  if (!ready_to_send_)
>+    return;
>   rtcp::Nack nack;
>   nack.SetSenderSsrc(config_.feedback_ssrc);
>   nack.SetMediaSsrc(ssrc);
>@@ -168,6 +188,8 @@ void RtcpTransceiverImpl::SendNack(uint32_t ssrc,
> }
> 
> void RtcpTransceiverImpl::SendPictureLossIndication(uint32_t ssrc) {
>+  if (!ready_to_send_)
>+    return;
>   rtcp::Pli pli;
>   pli.SetSenderSsrc(config_.feedback_ssrc);
>   pli.SetMediaSsrc(ssrc);
>@@ -177,6 +199,8 @@ void RtcpTransceiverImpl::SendPictureLossIndication(uint32_t ssrc) {
> void RtcpTransceiverImpl::SendFullIntraRequest(
>     rtc::ArrayView<const uint32_t> ssrcs) {
>   RTC_DCHECK(!ssrcs.empty());
>+  if (!ready_to_send_)
>+    return;
>   rtcp::Fir fir;
>   fir.SetSenderSsrc(config_.feedback_ssrc);
>   for (uint32_t media_ssrc : ssrcs)
>@@ -221,7 +245,7 @@ void RtcpTransceiverImpl::HandleSenderReport(
>     return;
>   RemoteSenderState& remote_sender =
>       remote_senders_[sender_report.sender_ssrc()];
>-  rtc::Optional<SenderReportTimes>& last =
>+  absl::optional<SenderReportTimes>& last =
>       remote_sender.last_received_sender_report;
>   last.emplace();
>   last->local_received_time_us = now_us;
>@@ -271,8 +295,8 @@ void RtcpTransceiverImpl::HandleTargetBitrate(
>       remote_sender_it->second.observers.empty())
>     return;
> 
>-  // Convert rtcp::TargetBitrate to BitrateAllocation from common types.
>-  BitrateAllocation bitrate_allocation;
>+  // Convert rtcp::TargetBitrate to VideoBitrateAllocation.
>+  VideoBitrateAllocation bitrate_allocation;
>   for (const rtcp::TargetBitrate::BitrateItem& item :
>        target_bitrate.GetTargetBitrates()) {
>     if (item.spatial_layer >= kMaxSpatialLayers ||
>@@ -310,7 +334,7 @@ void RtcpTransceiverImpl::SchedulePeriodicCompoundPackets(int64_t delay_ms) {
>       if (!ptr_)
>         return true;
>       ptr_->SendPeriodicCompoundPacket();
>-      task_queue_->PostDelayedTask(rtc::WrapUnique(this),
>+      task_queue_->PostDelayedTask(absl::WrapUnique(this),
>                                    ptr_->config_.report_period_ms);
>       return false;
>     }
>@@ -322,7 +346,7 @@ void RtcpTransceiverImpl::SchedulePeriodicCompoundPackets(int64_t delay_ms) {
> 
>   RTC_DCHECK(config_.schedule_periodic_compound_packets);
> 
>-  auto task = rtc::MakeUnique<SendPeriodicCompoundPacketTask>(
>+  auto task = absl::make_unique<SendPeriodicCompoundPacketTask>(
>       config_.task_queue, ptr_factory_.GetWeakPtr());
>   if (delay_ms > 0)
>     config_.task_queue->PostDelayedTask(std::move(task), delay_ms);
>@@ -402,15 +426,27 @@ std::vector<rtcp::ReportBlock> RtcpTransceiverImpl::CreateReportBlocks(
>   std::vector<rtcp::ReportBlock> report_blocks =
>       config_.receive_statistics->RtcpReportBlocks(
>           rtcp::ReceiverReport::kMaxNumberOfReportBlocks);
>+  uint32_t last_sr = 0;
>+  uint32_t last_delay = 0;
>   for (rtcp::ReportBlock& report_block : report_blocks) {
>     auto it = remote_senders_.find(report_block.source_ssrc());
>-    if (it == remote_senders_.end() || !it->second.last_received_sender_report)
>+    if (it == remote_senders_.end() ||
>+        !it->second.last_received_sender_report) {
>+      if (config_.avoid_zero_last_sr_in_last_report_block && last_sr != 0) {
>+        // Simulate behaviour of the RtcpSender to avoid hitting bug in
>+        // RtcpReceiver.
>+        report_block.SetLastSr(last_sr);
>+        report_block.SetDelayLastSr(last_delay);
>+      }
>       continue;
>+    }
>     const SenderReportTimes& last_sender_report =
>         *it->second.last_received_sender_report;
>-    report_block.SetLastSr(CompactNtp(last_sender_report.remote_sent_time));
>-    report_block.SetDelayLastSr(SaturatedUsToCompactNtp(
>-        now_us - last_sender_report.local_received_time_us));
>+    last_sr = CompactNtp(last_sender_report.remote_sent_time);
>+    last_delay = SaturatedUsToCompactNtp(
>+        now_us - last_sender_report.local_received_time_us);
>+    report_block.SetLastSr(last_sr);
>+    report_block.SetDelayLastSr(last_delay);
>   }
>   return report_blocks;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_transceiver_impl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_transceiver_impl.h
>index abbc40fe030..7bcd16fa718 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_transceiver_impl.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_transceiver_impl.h
>@@ -16,8 +16,8 @@
> #include <string>
> #include <vector>
> 
>+#include "absl/types/optional.h"
> #include "api/array_view.h"
>-#include "api/optional.h"
> #include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
> #include "modules/rtp_rtcp/source/rtcp_packet/dlrr.h"
> #include "modules/rtp_rtcp/source/rtcp_packet/remb.h"
>@@ -43,6 +43,8 @@ class RtcpTransceiverImpl {
>   void RemoveMediaReceiverRtcpObserver(uint32_t remote_ssrc,
>                                        MediaReceiverRtcpObserver* observer);
> 
>+  void SetReadyToSend(bool ready);
>+
>   void ReceivePacket(rtc::ArrayView<const uint8_t> packet, int64_t now_us);
> 
>   void SendCompoundPacket();
>@@ -88,7 +90,8 @@ class RtcpTransceiverImpl {
> 
>   const RtcpTransceiverConfig config_;
> 
>-  rtc::Optional<rtcp::Remb> remb_;
>+  bool ready_to_send_;
>+  absl::optional<rtcp::Remb> remb_;
>   // TODO(danilchap): Remove entries from remote_senders_ that are no longer
>   // needed.
>   std::map<uint32_t, RemoteSenderState> remote_senders_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_transceiver_impl_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_transceiver_impl_unittest.cc
>index 77d8e0cd3ea..1675b668e96 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_transceiver_impl_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_transceiver_impl_unittest.cc
>@@ -12,6 +12,8 @@
> 
> #include <vector>
> 
>+#include "absl/memory/memory.h"
>+#include "api/video/video_bitrate_allocation.h"
> #include "modules/rtp_rtcp/include/receive_statistics.h"
> #include "modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h"
> #include "modules/rtp_rtcp/source/rtcp_packet/app.h"
>@@ -20,7 +22,6 @@
> #include "modules/rtp_rtcp/source/time_util.h"
> #include "rtc_base/event.h"
> #include "rtc_base/fakeclock.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/task_queue.h"
> #include "test/gmock.h"
> #include "test/gtest.h"
>@@ -35,7 +36,7 @@ using ::testing::Invoke;
> using ::testing::Return;
> using ::testing::SizeIs;
> using ::testing::StrictMock;
>-using ::webrtc::BitrateAllocation;
>+using ::webrtc::VideoBitrateAllocation;
> using ::webrtc::CompactNtp;
> using ::webrtc::CompactNtpRttToMs;
> using ::webrtc::MockRtcpRttStats;
>@@ -60,7 +61,8 @@ class MockMediaReceiverRtcpObserver : public webrtc::MediaReceiverRtcpObserver {
>  public:
>   MOCK_METHOD3(OnSenderReport, void(uint32_t, NtpTime, uint32_t));
>   MOCK_METHOD1(OnBye, void(uint32_t));
>-  MOCK_METHOD2(OnBitrateAllocation, void(uint32_t, const BitrateAllocation&));
>+  MOCK_METHOD2(OnBitrateAllocation,
>+               void(uint32_t, const VideoBitrateAllocation&));
> };
> 
> // Since some tests will need to wait for this period, make it small to avoid
>@@ -130,6 +132,8 @@ RtcpTransceiverConfig DefaultTestConfig() {
>   RtcpTransceiverConfig config;
>   config.outgoing_transport = &null_transport;
>   config.schedule_periodic_compound_packets = false;
>+  config.initial_report_delay_ms = 10;
>+  config.report_period_ms = kReportPeriodMs;
>   return config;
> }
> 
>@@ -140,7 +144,7 @@ TEST(RtcpTransceiverImplTest, DelaysSendingFirstCompondPacket) {
>   config.outgoing_transport = &transport;
>   config.initial_report_delay_ms = 10;
>   config.task_queue = &queue;
>-  rtc::Optional<RtcpTransceiverImpl> rtcp_transceiver;
>+  absl::optional<RtcpTransceiverImpl> rtcp_transceiver;
> 
>   int64_t started_ms = rtc::TimeMillis();
>   queue.PostTask([&] { rtcp_transceiver.emplace(config); });
>@@ -165,7 +169,7 @@ TEST(RtcpTransceiverImplTest, PeriodicallySendsPackets) {
>   config.initial_report_delay_ms = 0;
>   config.report_period_ms = kReportPeriodMs;
>   config.task_queue = &queue;
>-  rtc::Optional<RtcpTransceiverImpl> rtcp_transceiver;
>+  absl::optional<RtcpTransceiverImpl> rtcp_transceiver;
>   int64_t time_just_before_1st_packet_ms = 0;
>   queue.PostTask([&] {
>     // Because initial_report_delay_ms is set to 0, time_just_before_the_packet
>@@ -179,7 +183,7 @@ TEST(RtcpTransceiverImplTest, PeriodicallySendsPackets) {
>   int64_t time_just_after_2nd_packet_ms = rtc::TimeMillis();
> 
>   EXPECT_GE(time_just_after_2nd_packet_ms - time_just_before_1st_packet_ms,
>-            config.report_period_ms);
>+            config.report_period_ms - 1);
> 
>   // Cleanup.
>   rtc::Event done(false, false);
>@@ -198,7 +202,7 @@ TEST(RtcpTransceiverImplTest, SendCompoundPacketDelaysPeriodicSendPackets) {
>   config.initial_report_delay_ms = 0;
>   config.report_period_ms = kReportPeriodMs;
>   config.task_queue = &queue;
>-  rtc::Optional<RtcpTransceiverImpl> rtcp_transceiver;
>+  absl::optional<RtcpTransceiverImpl> rtcp_transceiver;
>   queue.PostTask([&] { rtcp_transceiver.emplace(config); });
> 
>   // Wait for first packet.
>@@ -224,7 +228,71 @@ TEST(RtcpTransceiverImplTest, SendCompoundPacketDelaysPeriodicSendPackets) {
>   int64_t time_of_last_periodic_packet_ms = rtc::TimeMillis();
> 
>   EXPECT_GE(time_of_last_periodic_packet_ms - time_of_non_periodic_packet_ms,
>-            config.report_period_ms);
>+            config.report_period_ms - 1);
>+
>+  // Cleanup.
>+  rtc::Event done(false, false);
>+  queue.PostTask([&] {
>+    rtcp_transceiver.reset();
>+    done.Set();
>+  });
>+  ASSERT_TRUE(done.Wait(kAlmostForeverMs));
>+}
>+
>+TEST(RtcpTransceiverImplTest, SendsNoRtcpWhenNetworkStateIsDown) {
>+  MockTransport mock_transport;
>+  RtcpTransceiverConfig config = DefaultTestConfig();
>+  config.initial_ready_to_send = false;
>+  config.outgoing_transport = &mock_transport;
>+  RtcpTransceiverImpl rtcp_transceiver(config);
>+
>+  EXPECT_CALL(mock_transport, SendRtcp(_, _)).Times(0);
>+
>+  const uint8_t raw[] = {1, 2, 3, 4};
>+  const std::vector<uint16_t> sequence_numbers = {45, 57};
>+  const uint32_t ssrcs[] = {123};
>+  rtcp_transceiver.SendCompoundPacket();
>+  rtcp_transceiver.SendRawPacket(raw);
>+  rtcp_transceiver.SendNack(ssrcs[0], sequence_numbers);
>+  rtcp_transceiver.SendPictureLossIndication(ssrcs[0]);
>+  rtcp_transceiver.SendFullIntraRequest(ssrcs);
>+}
>+
>+TEST(RtcpTransceiverImplTest, SendsRtcpWhenNetworkStateIsUp) {
>+  MockTransport mock_transport;
>+  RtcpTransceiverConfig config = DefaultTestConfig();
>+  config.initial_ready_to_send = false;
>+  config.outgoing_transport = &mock_transport;
>+  RtcpTransceiverImpl rtcp_transceiver(config);
>+
>+  rtcp_transceiver.SetReadyToSend(true);
>+
>+  EXPECT_CALL(mock_transport, SendRtcp(_, _)).Times(5);
>+
>+  const uint8_t raw[] = {1, 2, 3, 4};
>+  const std::vector<uint16_t> sequence_numbers = {45, 57};
>+  const uint32_t ssrcs[] = {123};
>+  rtcp_transceiver.SendCompoundPacket();
>+  rtcp_transceiver.SendRawPacket(raw);
>+  rtcp_transceiver.SendNack(ssrcs[0], sequence_numbers);
>+  rtcp_transceiver.SendPictureLossIndication(ssrcs[0]);
>+  rtcp_transceiver.SendFullIntraRequest(ssrcs);
>+}
>+
>+TEST(RtcpTransceiverImplTest, SendsPeriodicRtcpWhenNetworkStateIsUp) {
>+  rtc::TaskQueue queue("rtcp");
>+  FakeRtcpTransport transport;
>+  RtcpTransceiverConfig config = DefaultTestConfig();
>+  config.schedule_periodic_compound_packets = true;
>+  config.initial_ready_to_send = false;
>+  config.outgoing_transport = &transport;
>+  config.task_queue = &queue;
>+  absl::optional<RtcpTransceiverImpl> rtcp_transceiver;
>+  rtcp_transceiver.emplace(config);
>+
>+  rtcp_transceiver->SetReadyToSend(true);
>+
>+  EXPECT_TRUE(transport.WaitPacket());
> 
>   // Cleanup.
>   rtc::Event done(false, false);
>@@ -485,7 +553,7 @@ TEST(RtcpTransceiverImplTest, CallsObserverOnTargetBitrateBySenderSsrc) {
>   xr.SetTargetBitrate(target_bitrate);
>   auto raw_packet = xr.Build();
> 
>-  BitrateAllocation bitrate_allocation;
>+  VideoBitrateAllocation bitrate_allocation;
>   bitrate_allocation.SetBitrate(0, 0, /*bitrate_bps=*/10000);
>   bitrate_allocation.SetBitrate(0, 1, /*bitrate_bps=*/20000);
>   bitrate_allocation.SetBitrate(1, 0, /*bitrate_bps=*/40000);
>@@ -511,7 +579,7 @@ TEST(RtcpTransceiverImplTest, SkipsIncorrectTargetBitrateEntries) {
>   xr.SetSenderSsrc(kRemoteSsrc);
>   auto raw_packet = xr.Build();
> 
>-  BitrateAllocation expected_allocation;
>+  VideoBitrateAllocation expected_allocation;
>   expected_allocation.SetBitrate(0, 0, /*bitrate_bps=*/10000);
>   EXPECT_CALL(observer, OnBitrateAllocation(kRemoteSsrc, expected_allocation));
>   rtcp_transceiver.ReceivePacket(raw_packet, /*now_us=*/0);
>@@ -569,6 +637,7 @@ TEST(RtcpTransceiverImplTest,
> 
>   RtcpTransceiverConfig config;
>   config.schedule_periodic_compound_packets = false;
>+  config.avoid_zero_last_sr_in_last_report_block = false;
>   RtcpPacketParser rtcp_parser;
>   RtcpParserTransport transport(&rtcp_parser);
>   config.outgoing_transport = &transport;
>@@ -576,9 +645,53 @@ TEST(RtcpTransceiverImplTest,
>   RtcpTransceiverImpl rtcp_transceiver(config);
> 
>   const NtpTime kRemoteNtp(0x9876543211);
>-  // Receive SenderReport for RemoteSsrc2, but no report for RemoteSsrc1.
>+  // Receive SenderReport for RemoteSsrc1, but no report for RemoteSsrc2.
>   SenderReport sr;
>-  sr.SetSenderSsrc(kRemoteSsrc2);
>+  sr.SetSenderSsrc(kRemoteSsrc1);
>+  sr.SetNtp(kRemoteNtp);
>+  auto raw_packet = sr.Build();
>+  rtcp_transceiver.ReceivePacket(raw_packet, /*now_us=*/0);
>+
>+  // Trigger sending ReceiverReport.
>+  rtcp_transceiver.SendCompoundPacket();
>+
>+  EXPECT_GT(rtcp_parser.receiver_report()->num_packets(), 0);
>+  const auto& report_blocks = rtcp_parser.receiver_report()->report_blocks();
>+  ASSERT_EQ(report_blocks.size(), 2u);
>+  // RtcpTransceiverImpl doesn't guarantee order of the report blocks
>+  // match result of ReceiveStatisticsProvider::RtcpReportBlocks callback,
>+  // but for simplicity of the test asume it is the same.
>+  ASSERT_EQ(report_blocks[0].source_ssrc(), kRemoteSsrc1);
>+  EXPECT_EQ(report_blocks[0].last_sr(), CompactNtp(kRemoteNtp));
>+
>+  ASSERT_EQ(report_blocks[1].source_ssrc(), kRemoteSsrc2);
>+  // No matching Sender Report for kRemoteSsrc2, LastSR fields has to be 0.
>+  EXPECT_EQ(report_blocks[1].last_sr(), 0u);
>+}
>+
>+TEST(RtcpTransceiverImplTest, AvoidLastReportBlockToHaveZeroLastSrField) {
>+  const uint32_t kRemoteSsrc1 = 54321;
>+  const uint32_t kRemoteSsrc2 = 54323;
>+  MockReceiveStatisticsProvider receive_statistics;
>+  std::vector<ReportBlock> statistics_report_blocks(2);
>+  statistics_report_blocks[0].SetMediaSsrc(kRemoteSsrc1);
>+  statistics_report_blocks[1].SetMediaSsrc(kRemoteSsrc2);
>+  ON_CALL(receive_statistics, RtcpReportBlocks(_))
>+      .WillByDefault(Return(statistics_report_blocks));
>+
>+  RtcpTransceiverConfig config;
>+  config.schedule_periodic_compound_packets = false;
>+  config.avoid_zero_last_sr_in_last_report_block = true;
>+  RtcpPacketParser rtcp_parser;
>+  RtcpParserTransport transport(&rtcp_parser);
>+  config.outgoing_transport = &transport;
>+  config.receive_statistics = &receive_statistics;
>+  RtcpTransceiverImpl rtcp_transceiver(config);
>+
>+  const NtpTime kRemoteNtp(0x9876543211);
>+  // Receive SenderReport for RemoteSsrc1, but no report for RemoteSsrc2.
>+  SenderReport sr;
>+  sr.SetSenderSsrc(kRemoteSsrc1);
>   sr.SetNtp(kRemoteNtp);
>   auto raw_packet = sr.Build();
>   rtcp_transceiver.ReceivePacket(raw_packet, /*now_us=*/0);
>@@ -593,11 +706,13 @@ TEST(RtcpTransceiverImplTest,
>   // match result of ReceiveStatisticsProvider::RtcpReportBlocks callback,
>   // but for simplicity of the test asume it is the same.
>   ASSERT_EQ(report_blocks[0].source_ssrc(), kRemoteSsrc1);
>-  // No matching Sender Report for kRemoteSsrc1, LastSR fields has to be 0.
>-  EXPECT_EQ(report_blocks[0].last_sr(), 0u);
>+  EXPECT_NE(report_blocks[0].last_sr(), 0u);
> 
>   ASSERT_EQ(report_blocks[1].source_ssrc(), kRemoteSsrc2);
>-  EXPECT_EQ(report_blocks[1].last_sr(), CompactNtp(kRemoteNtp));
>+  // No Sender Report for kRemoteSsrc2, use same LastSR as for kRemoteSsrc1
>+  EXPECT_EQ(report_blocks[1].last_sr(), report_blocks[0].last_sr());
>+  EXPECT_EQ(report_blocks[1].delay_since_last_sr(),
>+            report_blocks[0].delay_since_last_sr());
> }
> 
> TEST(RtcpTransceiverImplTest,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_transceiver_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_transceiver_unittest.cc
>index 74986bc6cf4..1305d9a47bf 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_transceiver_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtcp_transceiver_unittest.cc
>@@ -12,10 +12,10 @@
> 
> #include <memory>
> 
>+#include "absl/memory/memory.h"
> #include "modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
> #include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
> #include "rtc_base/event.h"
>-#include "rtc_base/ptr_util.h"
> #include "test/gmock.h"
> #include "test/gtest.h"
> #include "test/mock_transport.h"
>@@ -77,7 +77,7 @@ TEST(RtcpTransceiverTest, SendsRtcpOnTaskQueueWhenCreatedOnTaskQueue) {
> 
>   std::unique_ptr<RtcpTransceiver> rtcp_transceiver;
>   queue.PostTask([&] {
>-    rtcp_transceiver = rtc::MakeUnique<RtcpTransceiver>(config);
>+    rtcp_transceiver = absl::make_unique<RtcpTransceiver>(config);
>     rtcp_transceiver->SendCompoundPacket();
>   });
>   WaitPostedTasks(&queue);
>@@ -89,7 +89,7 @@ TEST(RtcpTransceiverTest, CanBeDestoryedOnTaskQueue) {
>   RtcpTransceiverConfig config;
>   config.outgoing_transport = &outgoing_transport;
>   config.task_queue = &queue;
>-  auto rtcp_transceiver = rtc::MakeUnique<RtcpTransceiver>(config);
>+  auto rtcp_transceiver = absl::make_unique<RtcpTransceiver>(config);
> 
>   queue.PostTask([&] { rtcp_transceiver.reset(); });
>   WaitPostedTasks(&queue);
>@@ -116,7 +116,7 @@ TEST(RtcpTransceiverTest, DoesntPostToRtcpObserverAfterCallToRemove) {
>   RtcpTransceiver rtcp_transceiver(config);
>   rtc::Event observer_deleted(false, false);
> 
>-  auto observer = rtc::MakeUnique<MockMediaReceiverRtcpObserver>();
>+  auto observer = absl::make_unique<MockMediaReceiverRtcpObserver>();
>   EXPECT_CALL(*observer, OnSenderReport(kRemoteSsrc, _, 1));
>   EXPECT_CALL(*observer, OnSenderReport(kRemoteSsrc, _, 2)).Times(0);
> 
>@@ -142,7 +142,7 @@ TEST(RtcpTransceiverTest, RemoveMediaReceiverRtcpObserverIsNonBlocking) {
>   config.outgoing_transport = &null_transport;
>   config.task_queue = &queue;
>   RtcpTransceiver rtcp_transceiver(config);
>-  auto observer = rtc::MakeUnique<MockMediaReceiverRtcpObserver>();
>+  auto observer = absl::make_unique<MockMediaReceiverRtcpObserver>();
>   rtcp_transceiver.AddMediaReceiverRtcpObserver(kRemoteSsrc, observer.get());
> 
>   rtc::Event queue_blocker(false, false);
>@@ -199,7 +199,7 @@ TEST(RtcpTransceiverTest, DoesntSendPacketsAfterDestruction) {
> 
>   EXPECT_CALL(outgoing_transport, SendRtcp(_, _)).Times(0);
> 
>-  auto rtcp_transceiver = rtc::MakeUnique<RtcpTransceiver>(config);
>+  auto rtcp_transceiver = absl::make_unique<RtcpTransceiver>(config);
>   rtc::Event pause(false, false);
>   queue.PostTask([&] {
>     pause.Wait(rtc::Event::kForever);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_fec_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_fec_unittest.cc
>index 1abed150f19..cbad0aed4a3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_fec_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_fec_unittest.cc
>@@ -17,7 +17,6 @@
> #include "modules/rtp_rtcp/source/flexfec_header_reader_writer.h"
> #include "modules/rtp_rtcp/source/forward_error_correction.h"
> #include "modules/rtp_rtcp/source/ulpfec_header_reader_writer.h"
>-#include "rtc_base/basictypes.h"
> #include "rtc_base/random.h"
> #include "test/gtest.h"
> 
>@@ -156,19 +155,19 @@ bool RtpFecTest<ForwardErrorCorrectionType>::IsRecoveryComplete() {
> 
>   // All recovered packets must be identical to the corresponding
>   // original packets.
>-  auto cmp = [](
>-      const std::unique_ptr<ForwardErrorCorrection::Packet>& media_packet,
>-      const std::unique_ptr<ForwardErrorCorrection::RecoveredPacket>&
>-          recovered_packet) {
>-    if (media_packet->length != recovered_packet->pkt->length) {
>-      return false;
>-    }
>-    if (memcmp(media_packet->data, recovered_packet->pkt->data,
>-               media_packet->length) != 0) {
>-      return false;
>-    }
>-    return true;
>-  };
>+  auto cmp =
>+      [](const std::unique_ptr<ForwardErrorCorrection::Packet>& media_packet,
>+         const std::unique_ptr<ForwardErrorCorrection::RecoveredPacket>&
>+             recovered_packet) {
>+        if (media_packet->length != recovered_packet->pkt->length) {
>+          return false;
>+        }
>+        if (memcmp(media_packet->data, recovered_packet->pkt->data,
>+                   media_packet->length) != 0) {
>+          return false;
>+        }
>+        return true;
>+      };
>   return std::equal(media_packets_.cbegin(), media_packets_.cend(),
>                     recovered_packets_.cbegin(), cmp);
> }
>@@ -287,8 +286,7 @@ TYPED_TEST(RtpFecTest, FecRecoveryNoLoss) {
>   this->NetworkReceivedPackets(this->media_loss_mask_, this->fec_loss_mask_);
> 
>   for (const auto& received_packet : this->received_packets_) {
>-    this->fec_.DecodeFec(*received_packet,
>-                         &this->recovered_packets_);
>+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
>   }
> 
>   // No packets lost, expect complete recovery.
>@@ -319,8 +317,7 @@ TYPED_TEST(RtpFecTest, FecRecoveryWithLoss) {
>   this->NetworkReceivedPackets(this->media_loss_mask_, this->fec_loss_mask_);
> 
>   for (const auto& received_packet : this->received_packets_) {
>-    this->fec_.DecodeFec(*received_packet,
>-                         &this->recovered_packets_);
>+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
>   }
> 
>   // One packet lost, one FEC packet, expect complete recovery.
>@@ -335,8 +332,7 @@ TYPED_TEST(RtpFecTest, FecRecoveryWithLoss) {
>   this->NetworkReceivedPackets(this->media_loss_mask_, this->fec_loss_mask_);
> 
>   for (const auto& received_packet : this->received_packets_) {
>-    this->fec_.DecodeFec(*received_packet,
>-                         &this->recovered_packets_);
>+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
>   }
> 
>   // 2 packets lost, one FEC packet, cannot get complete recovery.
>@@ -399,8 +395,7 @@ TYPED_TEST(RtpFecTest, NoFecRecoveryWithOldFecPacket) {
>   this->ReceivedPackets(this->media_packets_, this->media_loss_mask_, false);
> 
>   for (const auto& received_packet : this->received_packets_) {
>-    this->fec_.DecodeFec(*received_packet,
>-                         &this->recovered_packets_);
>+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
>   }
> 
>   // Expect that no decoding is done to get missing packet (seq#0) of third
>@@ -442,8 +437,7 @@ TYPED_TEST(RtpFecTest, FecRecoveryWithSeqNumGapOneFrameRecovery) {
>                         true);
> 
>   for (const auto& received_packet : this->received_packets_) {
>-    this->fec_.DecodeFec(*received_packet,
>-                         &this->recovered_packets_);
>+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
>   }
> 
>   // Expect 3 media packets in recovered list, and complete recovery.
>@@ -494,8 +488,7 @@ TEST_F(RtpFecTestUlpfecOnly, FecRecoveryWithSeqNumGapOneFrameRecovery) {
>                         true);
> 
>   for (const auto& received_packet : this->received_packets_) {
>-    this->fec_.DecodeFec(*received_packet,
>-                         &this->recovered_packets_);
>+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
>   }
> 
>   // The two FEC packets are received and should allow for complete recovery,
>@@ -553,8 +546,7 @@ TEST_F(RtpFecTestFlexfecOnly, FecRecoveryWithSeqNumGapOneFrameNoRecovery) {
>                         true);
> 
>   for (const auto& received_packet : this->received_packets_) {
>-    this->fec_.DecodeFec(*received_packet,
>-                         &this->recovered_packets_);
>+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
>   }
> 
>   // The two FEC packets are received and should allow for complete recovery,
>@@ -599,8 +591,7 @@ TYPED_TEST(RtpFecTest, FecRecoveryWithMediaOutOfOrder) {
>   std::swap(*it0, *it1);
> 
>   for (const auto& received_packet : this->received_packets_) {
>-    this->fec_.DecodeFec(*received_packet,
>-                         &this->recovered_packets_);
>+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
>   }
> 
>   // Expect 3 media packets in recovered list, and complete recovery.
>@@ -639,8 +630,7 @@ TYPED_TEST(RtpFecTest, FecRecoveryWithFecOutOfOrder) {
>   this->ReceivedPackets(this->media_packets_, this->media_loss_mask_, false);
> 
>   for (const auto& received_packet : this->received_packets_) {
>-    this->fec_.DecodeFec(*received_packet,
>-                         &this->recovered_packets_);
>+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
>   }
> 
>   // Expect 3 media packets in recovered list, and complete recovery.
>@@ -688,8 +678,7 @@ TYPED_TEST(RtpFecTest, FecRecoveryWithLoss50percRandomMask) {
>   this->NetworkReceivedPackets(this->media_loss_mask_, this->fec_loss_mask_);
> 
>   for (const auto& received_packet : this->received_packets_) {
>-    this->fec_.DecodeFec(*received_packet,
>-                         &this->recovered_packets_);
>+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
>   }
> 
>   // With media packet#1 and FEC packets #1, #2, #3, expect complete recovery.
>@@ -706,8 +695,7 @@ TYPED_TEST(RtpFecTest, FecRecoveryWithLoss50percRandomMask) {
>   this->NetworkReceivedPackets(this->media_loss_mask_, this->fec_loss_mask_);
> 
>   for (const auto& received_packet : this->received_packets_) {
>-    this->fec_.DecodeFec(*received_packet,
>-                         &this->recovered_packets_);
>+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
>   }
> 
>   // Cannot get complete recovery for this loss configuration with random mask.
>@@ -754,8 +742,7 @@ TYPED_TEST(RtpFecTest, FecRecoveryWithLoss50percBurstyMask) {
>   this->NetworkReceivedPackets(this->media_loss_mask_, this->fec_loss_mask_);
> 
>   for (const auto& received_packet : this->received_packets_) {
>-    this->fec_.DecodeFec(*received_packet,
>-                         &this->recovered_packets_);
>+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
>   }
> 
>   // Expect complete recovery for consecutive packet loss <= 50%.
>@@ -772,8 +759,7 @@ TYPED_TEST(RtpFecTest, FecRecoveryWithLoss50percBurstyMask) {
>   this->NetworkReceivedPackets(this->media_loss_mask_, this->fec_loss_mask_);
> 
>   for (const auto& received_packet : this->received_packets_) {
>-    this->fec_.DecodeFec(*received_packet,
>-                         &this->recovered_packets_);
>+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
>   }
> 
>   // Expect complete recovery for consecutive packet loss <= 50%.
>@@ -790,8 +776,7 @@ TYPED_TEST(RtpFecTest, FecRecoveryWithLoss50percBurstyMask) {
>   this->NetworkReceivedPackets(this->media_loss_mask_, this->fec_loss_mask_);
> 
>   for (const auto& received_packet : this->received_packets_) {
>-    this->fec_.DecodeFec(*received_packet,
>-                         &this->recovered_packets_);
>+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
>   }
> 
>   // Cannot get complete recovery for this loss configuration.
>@@ -821,8 +806,7 @@ TYPED_TEST(RtpFecTest, FecRecoveryNoLossUep) {
>   this->NetworkReceivedPackets(this->media_loss_mask_, this->fec_loss_mask_);
> 
>   for (const auto& received_packet : this->received_packets_) {
>-    this->fec_.DecodeFec(*received_packet,
>-                         &this->recovered_packets_);
>+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
>   }
> 
>   // No packets lost, expect complete recovery.
>@@ -853,8 +837,7 @@ TYPED_TEST(RtpFecTest, FecRecoveryWithLossUep) {
>   this->NetworkReceivedPackets(this->media_loss_mask_, this->fec_loss_mask_);
> 
>   for (const auto& received_packet : this->received_packets_) {
>-    this->fec_.DecodeFec(*received_packet,
>-                         &this->recovered_packets_);
>+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
>   }
> 
>   // One packet lost, one FEC packet, expect complete recovery.
>@@ -869,8 +852,7 @@ TYPED_TEST(RtpFecTest, FecRecoveryWithLossUep) {
>   this->NetworkReceivedPackets(this->media_loss_mask_, this->fec_loss_mask_);
> 
>   for (const auto& received_packet : this->received_packets_) {
>-    this->fec_.DecodeFec(*received_packet,
>-                         &this->recovered_packets_);
>+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
>   }
> 
>   // 2 packets lost, one FEC packet, cannot get complete recovery.
>@@ -915,8 +897,7 @@ TYPED_TEST(RtpFecTest, FecRecoveryWithLoss50percUepRandomMask) {
>   this->NetworkReceivedPackets(this->media_loss_mask_, this->fec_loss_mask_);
> 
>   for (const auto& received_packet : this->received_packets_) {
>-    this->fec_.DecodeFec(*received_packet,
>-                         &this->recovered_packets_);
>+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
>   }
> 
>   // With media packet#3 and FEC packets #0, #1, #3, expect complete recovery.
>@@ -934,8 +915,7 @@ TYPED_TEST(RtpFecTest, FecRecoveryWithLoss50percUepRandomMask) {
>   this->NetworkReceivedPackets(this->media_loss_mask_, this->fec_loss_mask_);
> 
>   for (const auto& received_packet : this->received_packets_) {
>-    this->fec_.DecodeFec(*received_packet,
>-                         &this->recovered_packets_);
>+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
>   }
> 
>   // Cannot get complete recovery for this loss configuration.
>@@ -971,8 +951,7 @@ TYPED_TEST(RtpFecTest, FecRecoveryNonConsecutivePackets) {
>   this->NetworkReceivedPackets(this->media_loss_mask_, this->fec_loss_mask_);
> 
>   for (const auto& received_packet : this->received_packets_) {
>-    this->fec_.DecodeFec(*received_packet,
>-                         &this->recovered_packets_);
>+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
>   }
> 
>   // One packet lost, one FEC packet, expect complete recovery.
>@@ -986,8 +965,7 @@ TYPED_TEST(RtpFecTest, FecRecoveryNonConsecutivePackets) {
>   this->NetworkReceivedPackets(this->media_loss_mask_, this->fec_loss_mask_);
> 
>   for (const auto& received_packet : this->received_packets_) {
>-    this->fec_.DecodeFec(*received_packet,
>-                         &this->recovered_packets_);
>+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
>   }
> 
>   // Unprotected packet lost. Recovery not possible.
>@@ -1002,8 +980,7 @@ TYPED_TEST(RtpFecTest, FecRecoveryNonConsecutivePackets) {
>   this->NetworkReceivedPackets(this->media_loss_mask_, this->fec_loss_mask_);
> 
>   for (const auto& received_packet : this->received_packets_) {
>-    this->fec_.DecodeFec(*received_packet,
>-                         &this->recovered_packets_);
>+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
>   }
> 
>   // 2 protected packets lost, one FEC packet, cannot get complete recovery.
>@@ -1042,8 +1019,7 @@ TYPED_TEST(RtpFecTest, FecRecoveryNonConsecutivePacketsExtension) {
>   this->NetworkReceivedPackets(this->media_loss_mask_, this->fec_loss_mask_);
> 
>   for (const auto& received_packet : this->received_packets_) {
>-    this->fec_.DecodeFec(*received_packet,
>-                         &this->recovered_packets_);
>+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
>   }
> 
>   // One packet lost, one FEC packet, expect complete recovery.
>@@ -1057,8 +1033,7 @@ TYPED_TEST(RtpFecTest, FecRecoveryNonConsecutivePacketsExtension) {
>   this->NetworkReceivedPackets(this->media_loss_mask_, this->fec_loss_mask_);
> 
>   for (const auto& received_packet : this->received_packets_) {
>-    this->fec_.DecodeFec(*received_packet,
>-                         &this->recovered_packets_);
>+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
>   }
> 
>   // Unprotected packet lost. Recovery not possible.
>@@ -1077,8 +1052,7 @@ TYPED_TEST(RtpFecTest, FecRecoveryNonConsecutivePacketsExtension) {
>   this->NetworkReceivedPackets(this->media_loss_mask_, this->fec_loss_mask_);
> 
>   for (const auto& received_packet : this->received_packets_) {
>-    this->fec_.DecodeFec(*received_packet,
>-                         &this->recovered_packets_);
>+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
>   }
> 
>   // 5 protected packets lost, one FEC packet, cannot get complete recovery.
>@@ -1117,8 +1091,7 @@ TYPED_TEST(RtpFecTest, FecRecoveryNonConsecutivePacketsWrap) {
>   this->NetworkReceivedPackets(this->media_loss_mask_, this->fec_loss_mask_);
> 
>   for (const auto& received_packet : this->received_packets_) {
>-    this->fec_.DecodeFec(*received_packet,
>-                         &this->recovered_packets_);
>+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
>   }
> 
>   // One packet lost, one FEC packet, expect complete recovery.
>@@ -1132,8 +1105,7 @@ TYPED_TEST(RtpFecTest, FecRecoveryNonConsecutivePacketsWrap) {
>   this->NetworkReceivedPackets(this->media_loss_mask_, this->fec_loss_mask_);
> 
>   for (const auto& received_packet : this->received_packets_) {
>-    this->fec_.DecodeFec(*received_packet,
>-                         &this->recovered_packets_);
>+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
>   }
> 
>   // Unprotected packet lost. Recovery not possible.
>@@ -1152,8 +1124,7 @@ TYPED_TEST(RtpFecTest, FecRecoveryNonConsecutivePacketsWrap) {
>   this->NetworkReceivedPackets(this->media_loss_mask_, this->fec_loss_mask_);
> 
>   for (const auto& received_packet : this->received_packets_) {
>-    this->fec_.DecodeFec(*received_packet,
>-                         &this->recovered_packets_);
>+    this->fec_.DecodeFec(*received_packet, &this->recovered_packets_);
>   }
> 
>   // 5 protected packets lost, one FEC packet, cannot get complete recovery.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format.cc
>index c5f57503636..1078a668100 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format.cc
>@@ -14,57 +14,53 @@
> 
> #include "modules/rtp_rtcp/source/rtp_format_h264.h"
> #include "modules/rtp_rtcp/source/rtp_format_video_generic.h"
>-#include "modules/rtp_rtcp/source/rtp_format_video_stereo.h"
> #include "modules/rtp_rtcp/source/rtp_format_vp8.h"
> #include "modules/rtp_rtcp/source/rtp_format_vp9.h"
> 
> namespace webrtc {
>-RtpPacketizer* RtpPacketizer::Create(RtpVideoCodecTypes type,
>+RtpPacketizer* RtpPacketizer::Create(VideoCodecType type,
>                                      size_t max_payload_len,
>                                      size_t last_packet_reduction_len,
>-                                     const RTPVideoTypeHeader* rtp_type_header,
>+                                     const RTPVideoHeader* rtp_video_header,
>                                      FrameType frame_type) {
>+  RTC_CHECK(type == kVideoCodecGeneric || rtp_video_header);
>   switch (type) {
>-    case kRtpVideoH264:
>-      RTC_CHECK(rtp_type_header);
>+    case kVideoCodecH264: {
>+      const auto& h264 =
>+          absl::get<RTPVideoHeaderH264>(rtp_video_header->video_type_header);
>       return new RtpPacketizerH264(max_payload_len, last_packet_reduction_len,
>-                                   rtp_type_header->H264.packetization_mode);
>-    case kRtpVideoVp8:
>-      RTC_CHECK(rtp_type_header);
>-      return new RtpPacketizerVp8(rtp_type_header->VP8, max_payload_len,
>+                                   h264.packetization_mode);
>+    }
>+    case kVideoCodecVP8:
>+      return new RtpPacketizerVp8(rtp_video_header->vp8(), max_payload_len,
>                                   last_packet_reduction_len);
>-    case kRtpVideoVp9:
>-      RTC_CHECK(rtp_type_header);
>-      return new RtpPacketizerVp9(rtp_type_header->VP9, max_payload_len,
>+    case kVideoCodecVP9: {
>+      const auto& vp9 =
>+          absl::get<RTPVideoHeaderVP9>(rtp_video_header->video_type_header);
>+      return new RtpPacketizerVp9(vp9, max_payload_len,
>                                   last_packet_reduction_len);
>-    case kRtpVideoStereo:
>-      return new RtpPacketizerStereo(rtp_type_header->stereo, frame_type,
>-                                     max_payload_len,
>-                                     last_packet_reduction_len);
>-    case kRtpVideoGeneric:
>-      return new RtpPacketizerGeneric(frame_type, max_payload_len,
>+    }
>+    case kVideoCodecGeneric:
>+      RTC_CHECK(rtp_video_header);
>+      return new RtpPacketizerGeneric(*rtp_video_header, frame_type,
>+                                      max_payload_len,
>                                       last_packet_reduction_len);
>-    case kRtpVideoNone:
>+    default:
>       RTC_NOTREACHED();
>   }
>   return nullptr;
> }
> 
>-RtpDepacketizer* RtpDepacketizer::Create(RtpVideoCodecTypes type) {
>+RtpDepacketizer* RtpDepacketizer::Create(VideoCodecType type) {
>   switch (type) {
>-    case kRtpVideoH264:
>+    case kVideoCodecH264:
>       return new RtpDepacketizerH264();
>-    case kRtpVideoVp8:
>+    case kVideoCodecVP8:
>       return new RtpDepacketizerVp8();
>-    case kRtpVideoVp9:
>+    case kVideoCodecVP9:
>       return new RtpDepacketizerVp9();
>-    case kRtpVideoStereo:
>-      return new RtpDepacketizerStereo();
>-    case kRtpVideoGeneric:
>+    default:
>       return new RtpDepacketizerGeneric();
>-    case kRtpVideoNone:
>-      assert(false);
>   }
>-  return nullptr;
> }
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format.h
>index ea098da710b..76d7f47ddbf 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format.h
>@@ -13,6 +13,7 @@
> 
> #include <string>
> 
>+#include "common_types.h"  // NOLINT(build/include)
> #include "modules/include/module_common_types.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> #include "rtc_base/constructormagic.h"
>@@ -22,10 +23,10 @@ class RtpPacketToSend;
> 
> class RtpPacketizer {
>  public:
>-  static RtpPacketizer* Create(RtpVideoCodecTypes type,
>+  static RtpPacketizer* Create(VideoCodecType type,
>                                size_t max_payload_len,
>                                size_t last_packet_reduction_len,
>-                               const RTPVideoTypeHeader* rtp_type_header,
>+                               const RTPVideoHeader* rtp_video_header,
>                                FrameType frame_type);
> 
>   virtual ~RtpPacketizer() {}
>@@ -51,13 +52,16 @@ class RtpPacketizer {
> class RtpDepacketizer {
>  public:
>   struct ParsedPayload {
>+    RTPVideoHeader& video_header() { return video; }
>+    const RTPVideoHeader& video_header() const { return video; }
>+    RTPVideoHeader video;
>+
>     const uint8_t* payload;
>     size_t payload_length;
>     FrameType frame_type;
>-    RTPTypeHeader type;
>   };
> 
>-  static RtpDepacketizer* Create(RtpVideoCodecTypes type);
>+  static RtpDepacketizer* Create(VideoCodecType type);
> 
>   virtual ~RtpDepacketizer() {}
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
>index 57a79557e9d..45e8d11fe2f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_h264.cc
>@@ -24,6 +24,7 @@
> #include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>+#include "rtc_base/system/fallthrough.h"
> #include "system_wrappers/include/metrics.h"
> 
> namespace webrtc {
>@@ -91,8 +92,9 @@ RtpPacketizerH264::RtpPacketizerH264(size_t max_payload_len,
>   RTC_CHECK_GT(max_payload_len, last_packet_reduction_len);
> }
> 
>-RtpPacketizerH264::~RtpPacketizerH264() {
>-}
>+RtpPacketizerH264::~RtpPacketizerH264() {}
>+
>+RtpPacketizerH264::Fragment::~Fragment() = default;
> 
> RtpPacketizerH264::Fragment::Fragment(const uint8_t* buffer, size_t length)
>     : buffer(buffer), length(length) {}
>@@ -126,7 +128,7 @@ size_t RtpPacketizerH264::SetPayloadData(
>       // RtpDepacketizerH264::ParseSingleNalu (receive side, in orderer to
>       // protect us from unknown or legacy send clients).
> 
>-      rtc::Optional<SpsParser::SpsState> sps;
>+      absl::optional<SpsParser::SpsState> sps;
> 
>       std::unique_ptr<rtc::Buffer> output_buffer(new rtc::Buffer());
>       // Add the type header to the output buffer first, so that the rewriter
>@@ -167,15 +169,25 @@ size_t RtpPacketizerH264::SetPayloadData(
>     if (!updated_sps)
>       input_fragments_.push_back(Fragment(buffer, length));
>   }
>-  GeneratePackets();
>+  if (!GeneratePackets()) {
>+    // If failed to generate all the packets, discard already generated
>+    // packets in case the caller would ignore return value and still try to
>+    // call NextPacket().
>+    num_packets_left_ = 0;
>+    while (!packets_.empty()) {
>+      packets_.pop();
>+    }
>+    return 0;
>+  }
>   return num_packets_left_;
> }
> 
>-void RtpPacketizerH264::GeneratePackets() {
>+bool RtpPacketizerH264::GeneratePackets() {
>   for (size_t i = 0; i < input_fragments_.size();) {
>     switch (packetization_mode_) {
>       case H264PacketizationMode::SingleNalUnit:
>-        PacketizeSingleNalu(i);
>+        if (!PacketizeSingleNalu(i))
>+          return false;
>         ++i;
>         break;
>       case H264PacketizationMode::NonInterleaved:
>@@ -194,6 +206,7 @@ void RtpPacketizerH264::GeneratePackets() {
>         break;
>     }
>   }
>+  return true;
> }
> 
> void RtpPacketizerH264::PacketizeFuA(size_t fragment_index) {
>@@ -287,19 +300,25 @@ size_t RtpPacketizerH264::PacketizeStapA(size_t fragment_index) {
>   return fragment_index;
> }
> 
>-void RtpPacketizerH264::PacketizeSingleNalu(size_t fragment_index) {
>+bool RtpPacketizerH264::PacketizeSingleNalu(size_t fragment_index) {
>   // Add a single NALU to the queue, no aggregation.
>   size_t payload_size_left = max_payload_len_;
>   if (fragment_index + 1 == input_fragments_.size())
>     payload_size_left -= last_packet_reduction_len_;
>   const Fragment* fragment = &input_fragments_[fragment_index];
>-  RTC_CHECK_GE(payload_size_left, fragment->length)
>-      << "Payload size left " << payload_size_left << ", fragment length "
>-      << fragment->length << ", packetization mode " << packetization_mode_;
>+  if (payload_size_left < fragment->length) {
>+    RTC_LOG(LS_ERROR) << "Failed to fit a fragment to packet in SingleNalu "
>+                         "packetization mode. Payload size left "
>+                      << payload_size_left << ", fragment length "
>+                      << fragment->length << ", packet capacity "
>+                      << max_payload_len_;
>+    return false;
>+  }
>   RTC_CHECK_GT(fragment->length, 0u);
>   packets_.push(PacketUnit(*fragment, true /* first */, true /* last */,
>                            false /* aggregated */, fragment->buffer[0]));
>   ++num_packets_left_;
>+  return true;
> }
> 
> bool RtpPacketizerH264::NextPacket(RtpPacketToSend* rtp_packet) {
>@@ -317,11 +336,11 @@ bool RtpPacketizerH264::NextPacket(RtpPacketToSend* rtp_packet) {
>     packets_.pop();
>     input_fragments_.pop_front();
>   } else if (packet.aggregated) {
>-    RTC_CHECK_EQ(H264PacketizationMode::NonInterleaved, packetization_mode_);
>+    RTC_CHECK(H264PacketizationMode::NonInterleaved == packetization_mode_);
>     bool is_last_packet = num_packets_left_ == 1;
>     NextAggregatePacket(rtp_packet, is_last_packet);
>   } else {
>-    RTC_CHECK_EQ(H264PacketizationMode::NonInterleaved, packetization_mode_);
>+    RTC_CHECK(H264PacketizationMode::NonInterleaved == packetization_mode_);
>     NextFragmentPacket(rtp_packet);
>   }
>   RTC_DCHECK_LE(rtp_packet->payload_size(), max_payload_len_);
>@@ -410,7 +429,8 @@ bool RtpDepacketizerH264::Parse(ParsedPayload* parsed_payload,
>   modified_buffer_.reset();
> 
>   uint8_t nal_type = payload_data[0] & kTypeMask;
>-  parsed_payload->type.Video.codecHeader.H264.nalus_length = 0;
>+  parsed_payload->video_header()
>+      .video_type_header.emplace<RTPVideoHeaderH264>();
>   if (nal_type == H264::NaluType::kFuA) {
>     // Fragmented NAL units (FU-A).
>     if (!ParseFuaNalu(parsed_payload, payload_data))
>@@ -434,12 +454,13 @@ bool RtpDepacketizerH264::Parse(ParsedPayload* parsed_payload,
> bool RtpDepacketizerH264::ProcessStapAOrSingleNalu(
>     ParsedPayload* parsed_payload,
>     const uint8_t* payload_data) {
>-  parsed_payload->type.Video.width = 0;
>-  parsed_payload->type.Video.height = 0;
>-  parsed_payload->type.Video.codec = kRtpVideoH264;
>-  parsed_payload->type.Video.is_first_packet_in_frame = true;
>-  RTPVideoHeaderH264* h264_header =
>-      &parsed_payload->type.Video.codecHeader.H264;
>+  parsed_payload->video_header().width = 0;
>+  parsed_payload->video_header().height = 0;
>+  parsed_payload->video_header().codec = kVideoCodecH264;
>+  parsed_payload->video_header().simulcastIdx = 0;
>+  parsed_payload->video_header().is_first_packet_in_frame = true;
>+  auto& h264_header = absl::get<RTPVideoHeaderH264>(
>+      parsed_payload->video_header().video_type_header);
> 
>   const uint8_t* nalu_start = payload_data + kNalHeaderSize;
>   const size_t nalu_length = length_ - kNalHeaderSize;
>@@ -457,13 +478,13 @@ bool RtpDepacketizerH264::ProcessStapAOrSingleNalu(
>       return false;
>     }
> 
>-    h264_header->packetization_type = kH264StapA;
>+    h264_header.packetization_type = kH264StapA;
>     nal_type = payload_data[kStapAHeaderSize] & kTypeMask;
>   } else {
>-    h264_header->packetization_type = kH264SingleNalu;
>+    h264_header.packetization_type = kH264SingleNalu;
>     nalu_start_offsets.push_back(0);
>   }
>-  h264_header->nalu_type = nal_type;
>+  h264_header.nalu_type = nal_type;
>   parsed_payload->frame_type = kVideoFrameDelta;
> 
>   nalu_start_offsets.push_back(length_ + kLengthFieldSize);  // End offset.
>@@ -494,7 +515,7 @@ bool RtpDepacketizerH264::ProcessStapAOrSingleNalu(
>         if (start_offset)
>           output_buffer->AppendData(payload_data, start_offset);
> 
>-        rtc::Optional<SpsParser::SpsState> sps;
>+        absl::optional<SpsParser::SpsState> sps;
> 
>         SpsVuiRewriter::ParseResult result = SpsVuiRewriter::ParseAndRewriteSps(
>             &payload_data[start_offset], end_offset - start_offset, &sps,
>@@ -509,7 +530,7 @@ bool RtpDepacketizerH264::ProcessStapAOrSingleNalu(
>             }
> 
>             // Rewrite length field to new SPS size.
>-            if (h264_header->packetization_type == kH264StapA) {
>+            if (h264_header.packetization_type == kH264StapA) {
>               size_t length_field_offset =
>                   start_offset - (H264::kNaluTypeSize + kLengthFieldSize);
>               // Stap-A Length includes payload data and type header.
>@@ -549,8 +570,8 @@ bool RtpDepacketizerH264::ProcessStapAOrSingleNalu(
>         }
> 
>         if (sps) {
>-          parsed_payload->type.Video.width = sps->width;
>-          parsed_payload->type.Video.height = sps->height;
>+          parsed_payload->video_header().width = sps->width;
>+          parsed_payload->video_header().height = sps->height;
>           nalu.sps_id = sps->id;
>         } else {
>           RTC_LOG(LS_WARNING) << "Failed to parse SPS id from SPS slice.";
>@@ -562,8 +583,8 @@ bool RtpDepacketizerH264::ProcessStapAOrSingleNalu(
>         uint32_t pps_id;
>         uint32_t sps_id;
>         if (PpsParser::ParsePpsIds(&payload_data[start_offset],
>-                                    end_offset - start_offset, &pps_id,
>-                                    &sps_id)) {
>+                                   end_offset - start_offset, &pps_id,
>+                                   &sps_id)) {
>           nalu.pps_id = pps_id;
>           nalu.sps_id = sps_id;
>         } else {
>@@ -574,9 +595,9 @@ bool RtpDepacketizerH264::ProcessStapAOrSingleNalu(
>       }
>       case H264::NaluType::kIdr:
>         parsed_payload->frame_type = kVideoFrameKey;
>-        FALLTHROUGH();
>+        RTC_FALLTHROUGH();
>       case H264::NaluType::kSlice: {
>-        rtc::Optional<uint32_t> pps_id = PpsParser::ParsePpsIdFromSlice(
>+        absl::optional<uint32_t> pps_id = PpsParser::ParsePpsIdFromSlice(
>             &payload_data[start_offset], end_offset - start_offset);
>         if (pps_id) {
>           nalu.pps_id = *pps_id;
>@@ -598,13 +619,13 @@ bool RtpDepacketizerH264::ProcessStapAOrSingleNalu(
>         RTC_LOG(LS_WARNING) << "Unexpected STAP-A or FU-A received.";
>         return false;
>     }
>-    RTPVideoHeaderH264* h264 = &parsed_payload->type.Video.codecHeader.H264;
>-    if (h264->nalus_length == kMaxNalusPerPacket) {
>+
>+    if (h264_header.nalus_length == kMaxNalusPerPacket) {
>       RTC_LOG(LS_WARNING)
>           << "Received packet containing more than " << kMaxNalusPerPacket
>           << " NAL units. Will not keep track sps and pps ids for all of them.";
>     } else {
>-      h264->nalus[h264->nalus_length++] = nalu;
>+      h264_header.nalus[h264_header.nalus_length++] = nalu;
>     }
>   }
> 
>@@ -628,7 +649,7 @@ bool RtpDepacketizerH264::ParseFuaNalu(
>   if (first_fragment) {
>     offset_ = 0;
>     length_ -= kNalHeaderSize;
>-    rtc::Optional<uint32_t> pps_id = PpsParser::ParsePpsIdFromSlice(
>+    absl::optional<uint32_t> pps_id = PpsParser::ParsePpsIdFromSlice(
>         payload_data + 2 * kNalHeaderSize, length_ - kNalHeaderSize);
>     if (pps_id) {
>       nalu.pps_id = *pps_id;
>@@ -652,16 +673,18 @@ bool RtpDepacketizerH264::ParseFuaNalu(
>   } else {
>     parsed_payload->frame_type = kVideoFrameDelta;
>   }
>-  parsed_payload->type.Video.width = 0;
>-  parsed_payload->type.Video.height = 0;
>-  parsed_payload->type.Video.codec = kRtpVideoH264;
>-  parsed_payload->type.Video.is_first_packet_in_frame = first_fragment;
>-  RTPVideoHeaderH264* h264 = &parsed_payload->type.Video.codecHeader.H264;
>-  h264->packetization_type = kH264FuA;
>-  h264->nalu_type = original_nal_type;
>+  parsed_payload->video_header().width = 0;
>+  parsed_payload->video_header().height = 0;
>+  parsed_payload->video_header().codec = kVideoCodecH264;
>+  parsed_payload->video_header().simulcastIdx = 0;
>+  parsed_payload->video_header().is_first_packet_in_frame = first_fragment;
>+  auto& h264_header = absl::get<RTPVideoHeaderH264>(
>+      parsed_payload->video_header().video_type_header);
>+  h264_header.packetization_type = kH264FuA;
>+  h264_header.nalu_type = original_nal_type;
>   if (first_fragment) {
>-    h264->nalus[h264->nalus_length] = nalu;
>-    h264->nalus_length = 1;
>+    h264_header.nalus[h264_header.nalus_length] = nalu;
>+    h264_header.nalus_length = 1;
>   }
>   return true;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_h264.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_h264.h
>index 5b6fe022950..99b080b9ecc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_h264.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_h264.h
>@@ -30,7 +30,7 @@ class RtpPacketizerH264 : public RtpPacketizer {
>                     size_t last_packet_reduction_len,
>                     H264PacketizationMode packetization_mode);
> 
>-  virtual ~RtpPacketizerH264();
>+  ~RtpPacketizerH264() override;
> 
>   size_t SetPayloadData(const uint8_t* payload_data,
>                         size_t payload_size,
>@@ -49,6 +49,7 @@ class RtpPacketizerH264 : public RtpPacketizer {
>   struct Fragment {
>     Fragment(const uint8_t* buffer, size_t length);
>     explicit Fragment(const Fragment& fragment);
>+    ~Fragment();
>     const uint8_t* buffer = nullptr;
>     size_t length = 0;
>     std::unique_ptr<rtc::Buffer> tmp_buffer;
>@@ -79,10 +80,10 @@ class RtpPacketizerH264 : public RtpPacketizer {
>     uint8_t header;
>   };
> 
>-  void GeneratePackets();
>+  bool GeneratePackets();
>   void PacketizeFuA(size_t fragment_index);
>   size_t PacketizeStapA(size_t fragment_index);
>-  void PacketizeSingleNalu(size_t fragment_index);
>+  bool PacketizeSingleNalu(size_t fragment_index);
>   void NextAggregatePacket(RtpPacketToSend* rtp_packet, bool last);
>   void NextFragmentPacket(RtpPacketToSend* rtp_packet);
> 
>@@ -100,7 +101,7 @@ class RtpPacketizerH264 : public RtpPacketizer {
> class RtpDepacketizerH264 : public RtpDepacketizer {
>  public:
>   RtpDepacketizerH264();
>-  virtual ~RtpDepacketizerH264();
>+  ~RtpDepacketizerH264() override;
> 
>   bool Parse(ParsedPayload* parsed_payload,
>              const uint8_t* payload_data,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
>index 0eef2bcd85e..1f69fcb3ba5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc
>@@ -26,6 +26,12 @@ namespace {
> 
> using ::testing::ElementsAreArray;
> 
>+struct H264ParsedPayload : public RtpDepacketizer::ParsedPayload {
>+  RTPVideoHeaderH264& h264() {
>+    return absl::get<RTPVideoHeaderH264>(video.video_type_header);
>+  }
>+};
>+
> constexpr RtpPacketToSend::ExtensionManager* kNoExtensions = nullptr;
> const size_t kMaxPayloadSize = 1200;
> const size_t kLengthFieldLength = 2;
>@@ -65,11 +71,11 @@ void CreateThreeFragments(RTPFragmentationHeader* fragmentation,
> RtpPacketizer* CreateH264Packetizer(H264PacketizationMode mode,
>                                     size_t max_payload_size,
>                                     size_t last_packet_reduction) {
>-  RTPVideoTypeHeader type_header;
>-  type_header.H264.packetization_mode = mode;
>-  return RtpPacketizer::Create(kRtpVideoH264, max_payload_size,
>-                               last_packet_reduction, &type_header,
>-                               kEmptyFrame);
>+  RTPVideoHeader header;
>+  header.video_type_header.emplace<RTPVideoHeaderH264>().packetization_mode =
>+      mode;
>+  return RtpPacketizer::Create(kVideoCodecH264, max_payload_size,
>+                               last_packet_reduction, &header, kEmptyFrame);
> }
> 
> void VerifyFua(size_t fua_index,
>@@ -468,9 +474,7 @@ TEST(RtpPacketizerH264Test, TestFUABig) {
>                               sizeof(kExpectedPayloadSizes) / sizeof(size_t)));
> }
> 
>-#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
>-
>-TEST(RtpPacketizerH264DeathTest, SendOverlongDataInPacketizationMode0) {
>+TEST(RtpPacketizerH264Test, SendOverlongDataInPacketizationMode0) {
>   const size_t kFrameSize = kMaxPayloadSize + 1;
>   uint8_t frame[kFrameSize] = {0};
>   for (size_t i = 0; i < kFrameSize; ++i)
>@@ -484,12 +488,9 @@ TEST(RtpPacketizerH264DeathTest, SendOverlongDataInPacketizationMode0) {
> 
>   std::unique_ptr<RtpPacketizer> packetizer(CreateH264Packetizer(
>       H264PacketizationMode::SingleNalUnit, kMaxPayloadSize, 0));
>-  EXPECT_DEATH(packetizer->SetPayloadData(frame, kFrameSize, &fragmentation),
>-               "payload_size");
>+  EXPECT_EQ(0u, packetizer->SetPayloadData(frame, kFrameSize, &fragmentation));
> }
> 
>-#endif  // RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
>-
> namespace {
> const uint8_t kStartSequence[] = {0x00, 0x00, 0x00, 0x01};
> const uint8_t kOriginalSps[] = {kSps, 0x00, 0x00, 0x03, 0x03,
>@@ -498,7 +499,7 @@ const uint8_t kRewrittenSps[] = {kSps, 0x00, 0x00, 0x03, 0x03, 0xF4, 0x05, 0x03,
>                                  0xC7, 0xE0, 0x1B, 0x41, 0x10, 0x8D, 0x00};
> const uint8_t kIdrOne[] = {kIdr, 0xFF, 0x00, 0x00, 0x04};
> const uint8_t kIdrTwo[] = {kIdr, 0xFF, 0x00, 0x11};
>-}
>+}  // namespace
> 
> class RtpPacketizerH264TestSpsRewriting : public ::testing::Test {
>  public:
>@@ -583,9 +584,9 @@ TEST_F(RtpPacketizerH264TestSpsRewriting, StapASps) {
> class RtpDepacketizerH264Test : public ::testing::Test {
>  protected:
>   RtpDepacketizerH264Test()
>-      : depacketizer_(RtpDepacketizer::Create(kRtpVideoH264)) {}
>+      : depacketizer_(RtpDepacketizer::Create(kVideoCodecH264)) {}
> 
>-  void ExpectPacket(RtpDepacketizer::ParsedPayload* parsed_payload,
>+  void ExpectPacket(H264ParsedPayload* parsed_payload,
>                     const uint8_t* data,
>                     size_t length) {
>     ASSERT_TRUE(parsed_payload != NULL);
>@@ -600,33 +601,31 @@ class RtpDepacketizerH264Test : public ::testing::Test {
> 
> TEST_F(RtpDepacketizerH264Test, TestSingleNalu) {
>   uint8_t packet[2] = {0x05, 0xFF};  // F=0, NRI=0, Type=5 (IDR).
>-  RtpDepacketizer::ParsedPayload payload;
>+  H264ParsedPayload payload;
> 
>   ASSERT_TRUE(depacketizer_->Parse(&payload, packet, sizeof(packet)));
>   ExpectPacket(&payload, packet, sizeof(packet));
>   EXPECT_EQ(kVideoFrameKey, payload.frame_type);
>-  EXPECT_EQ(kRtpVideoH264, payload.type.Video.codec);
>-  EXPECT_TRUE(payload.type.Video.is_first_packet_in_frame);
>-  EXPECT_EQ(kH264SingleNalu,
>-            payload.type.Video.codecHeader.H264.packetization_type);
>-  EXPECT_EQ(kIdr, payload.type.Video.codecHeader.H264.nalu_type);
>+  EXPECT_EQ(kVideoCodecH264, payload.video_header().codec);
>+  EXPECT_TRUE(payload.video_header().is_first_packet_in_frame);
>+  EXPECT_EQ(kH264SingleNalu, payload.h264().packetization_type);
>+  EXPECT_EQ(kIdr, payload.h264().nalu_type);
> }
> 
> TEST_F(RtpDepacketizerH264Test, TestSingleNaluSpsWithResolution) {
>   uint8_t packet[] = {kSps, 0x7A, 0x00, 0x1F, 0xBC, 0xD9, 0x40, 0x50,
>                       0x05, 0xBA, 0x10, 0x00, 0x00, 0x03, 0x00, 0xC0,
>                       0x00, 0x00, 0x03, 0x2A, 0xE0, 0xF1, 0x83, 0x25};
>-  RtpDepacketizer::ParsedPayload payload;
>+  H264ParsedPayload payload;
> 
>   ASSERT_TRUE(depacketizer_->Parse(&payload, packet, sizeof(packet)));
>   ExpectPacket(&payload, packet, sizeof(packet));
>   EXPECT_EQ(kVideoFrameKey, payload.frame_type);
>-  EXPECT_EQ(kRtpVideoH264, payload.type.Video.codec);
>-  EXPECT_TRUE(payload.type.Video.is_first_packet_in_frame);
>-  EXPECT_EQ(kH264SingleNalu,
>-            payload.type.Video.codecHeader.H264.packetization_type);
>-  EXPECT_EQ(1280u, payload.type.Video.width);
>-  EXPECT_EQ(720u, payload.type.Video.height);
>+  EXPECT_EQ(kVideoCodecH264, payload.video_header().codec);
>+  EXPECT_TRUE(payload.video_header().is_first_packet_in_frame);
>+  EXPECT_EQ(kH264SingleNalu, payload.h264().packetization_type);
>+  EXPECT_EQ(1280u, payload.video_header().width);
>+  EXPECT_EQ(720u, payload.video_header().height);
> }
> 
> TEST_F(RtpDepacketizerH264Test, TestStapAKey) {
>@@ -647,13 +646,13 @@ TEST_F(RtpDepacketizerH264Test, TestStapAKey) {
>                         0x85, 0xB8, 0x0, 0x4, 0x0, 0x0, 0x13, 0x93, 0x12, 0x0};
>   // clang-format on
> 
>-  RtpDepacketizer::ParsedPayload payload;
>+  H264ParsedPayload payload;
>   ASSERT_TRUE(depacketizer_->Parse(&payload, packet, sizeof(packet)));
>   ExpectPacket(&payload, packet, sizeof(packet));
>   EXPECT_EQ(kVideoFrameKey, payload.frame_type);
>-  EXPECT_EQ(kRtpVideoH264, payload.type.Video.codec);
>-  EXPECT_TRUE(payload.type.Video.is_first_packet_in_frame);
>-  const RTPVideoHeaderH264& h264 = payload.type.Video.codecHeader.H264;
>+  EXPECT_EQ(kVideoCodecH264, payload.video_header().codec);
>+  EXPECT_TRUE(payload.video_header().is_first_packet_in_frame);
>+  const RTPVideoHeaderH264& h264 = payload.h264();
>   EXPECT_EQ(kH264StapA, h264.packetization_type);
>   // NALU type for aggregated packets is the type of the first packet only.
>   EXPECT_EQ(kSps, h264.nalu_type);
>@@ -670,23 +669,23 @@ TEST_F(RtpDepacketizerH264Test, TestStapAKey) {
> 
> TEST_F(RtpDepacketizerH264Test, TestStapANaluSpsWithResolution) {
>   uint8_t packet[] = {kStapA,  // F=0, NRI=0, Type=24.
>-                      // Length (2 bytes), nal header, payload.
>+                               // Length (2 bytes), nal header, payload.
>                       0x00, 0x19, kSps, 0x7A, 0x00, 0x1F, 0xBC, 0xD9, 0x40,
>                       0x50, 0x05, 0xBA, 0x10, 0x00, 0x00, 0x03, 0x00, 0xC0,
>                       0x00, 0x00, 0x03, 0x2A, 0xE0, 0xF1, 0x83, 0x25, 0x80,
>                       0x00, 0x03, kIdr, 0xFF, 0x00, 0x00, 0x04, kIdr, 0xFF,
>                       0x00, 0x11};
> 
>-  RtpDepacketizer::ParsedPayload payload;
>+  H264ParsedPayload payload;
> 
>   ASSERT_TRUE(depacketizer_->Parse(&payload, packet, sizeof(packet)));
>   ExpectPacket(&payload, packet, sizeof(packet));
>   EXPECT_EQ(kVideoFrameKey, payload.frame_type);
>-  EXPECT_EQ(kRtpVideoH264, payload.type.Video.codec);
>-  EXPECT_TRUE(payload.type.Video.is_first_packet_in_frame);
>-  EXPECT_EQ(kH264StapA, payload.type.Video.codecHeader.H264.packetization_type);
>-  EXPECT_EQ(1280u, payload.type.Video.width);
>-  EXPECT_EQ(720u, payload.type.Video.height);
>+  EXPECT_EQ(kVideoCodecH264, payload.video_header().codec);
>+  EXPECT_TRUE(payload.video_header().is_first_packet_in_frame);
>+  EXPECT_EQ(kH264StapA, payload.h264().packetization_type);
>+  EXPECT_EQ(1280u, payload.video_header().width);
>+  EXPECT_EQ(720u, payload.video_header().height);
> }
> 
> TEST_F(RtpDepacketizerH264Test, TestEmptyStapARejected) {
>@@ -701,7 +700,7 @@ TEST_F(RtpDepacketizerH264Test, TestEmptyStapARejected) {
>   uint8_t trailing_empty_packet[] = {kStapA, 0x00, 0x03, kIdr,
>                                      0xFF,   0x00, 0x00, 0x00};
> 
>-  RtpDepacketizer::ParsedPayload payload;
>+  H264ParsedPayload payload;
> 
>   EXPECT_FALSE(depacketizer_->Parse(&payload, lone_empty_packet,
>                                     sizeof(lone_empty_packet)));
>@@ -740,7 +739,7 @@ TEST_F(RtpDepacketizerH264Test, DepacketizeWithRewriting) {
>   out_buffer.AppendData(kHeader, 2);
>   out_buffer.AppendData(kIdrTwo);
> 
>-  RtpDepacketizer::ParsedPayload payload;
>+  H264ParsedPayload payload;
>   EXPECT_TRUE(
>       depacketizer_->Parse(&payload, in_buffer.data(), in_buffer.size()));
> 
>@@ -787,7 +786,7 @@ TEST_F(RtpDepacketizerH264Test, DepacketizeWithDoubleRewriting) {
>   out_buffer.AppendData(kHeader, 2);
>   out_buffer.AppendData(kIdrTwo);
> 
>-  RtpDepacketizer::ParsedPayload payload;
>+  H264ParsedPayload payload;
>   EXPECT_TRUE(
>       depacketizer_->Parse(&payload, in_buffer.data(), in_buffer.size()));
> 
>@@ -801,19 +800,19 @@ TEST_F(RtpDepacketizerH264Test, DepacketizeWithDoubleRewriting) {
> 
> TEST_F(RtpDepacketizerH264Test, TestStapADelta) {
>   uint8_t packet[16] = {kStapA,  // F=0, NRI=0, Type=24.
>-                        // Length, nal header, payload.
>-                        0,      0x02, kSlice, 0xFF,   0,    0x03, kSlice, 0xFF,
>-                        0x00,   0,    0x04,   kSlice, 0xFF, 0x00, 0x11};
>-  RtpDepacketizer::ParsedPayload payload;
>+                                 // Length, nal header, payload.
>+                        0, 0x02, kSlice, 0xFF, 0, 0x03, kSlice, 0xFF, 0x00, 0,
>+                        0x04, kSlice, 0xFF, 0x00, 0x11};
>+  H264ParsedPayload payload;
> 
>   ASSERT_TRUE(depacketizer_->Parse(&payload, packet, sizeof(packet)));
>   ExpectPacket(&payload, packet, sizeof(packet));
>   EXPECT_EQ(kVideoFrameDelta, payload.frame_type);
>-  EXPECT_EQ(kRtpVideoH264, payload.type.Video.codec);
>-  EXPECT_TRUE(payload.type.Video.is_first_packet_in_frame);
>-  EXPECT_EQ(kH264StapA, payload.type.Video.codecHeader.H264.packetization_type);
>+  EXPECT_EQ(kVideoCodecH264, payload.video_header().codec);
>+  EXPECT_TRUE(payload.video_header().is_first_packet_in_frame);
>+  EXPECT_EQ(kH264StapA, payload.h264().packetization_type);
>   // NALU type for aggregated packets is the type of the first packet only.
>-  EXPECT_EQ(kSlice, payload.type.Video.codecHeader.H264.nalu_type);
>+  EXPECT_EQ(kSlice, payload.h264().nalu_type);
> }
> 
> TEST_F(RtpDepacketizerH264Test, TestFuA) {
>@@ -841,16 +840,16 @@ TEST_F(RtpDepacketizerH264Test, TestFuA) {
>   };
>   const uint8_t kExpected3[] = {0x03};
> 
>-  RtpDepacketizer::ParsedPayload payload;
>+  H264ParsedPayload payload;
> 
>   // We expect that the first packet is one byte shorter since the FU-A header
>   // has been replaced by the original nal header.
>   ASSERT_TRUE(depacketizer_->Parse(&payload, packet1, sizeof(packet1)));
>   ExpectPacket(&payload, kExpected1, sizeof(kExpected1));
>   EXPECT_EQ(kVideoFrameKey, payload.frame_type);
>-  EXPECT_EQ(kRtpVideoH264, payload.type.Video.codec);
>-  EXPECT_TRUE(payload.type.Video.is_first_packet_in_frame);
>-  const RTPVideoHeaderH264& h264 = payload.type.Video.codecHeader.H264;
>+  EXPECT_EQ(kVideoCodecH264, payload.video_header().codec);
>+  EXPECT_TRUE(payload.video_header().is_first_packet_in_frame);
>+  const RTPVideoHeaderH264& h264 = payload.h264();
>   EXPECT_EQ(kH264FuA, h264.packetization_type);
>   EXPECT_EQ(kIdr, h264.nalu_type);
>   ASSERT_EQ(1u, h264.nalus_length);
>@@ -860,28 +859,28 @@ TEST_F(RtpDepacketizerH264Test, TestFuA) {
> 
>   // Following packets will be 2 bytes shorter since they will only be appended
>   // onto the first packet.
>-  payload = RtpDepacketizer::ParsedPayload();
>+  payload = H264ParsedPayload();
>   ASSERT_TRUE(depacketizer_->Parse(&payload, packet2, sizeof(packet2)));
>   ExpectPacket(&payload, kExpected2, sizeof(kExpected2));
>   EXPECT_EQ(kVideoFrameKey, payload.frame_type);
>-  EXPECT_EQ(kRtpVideoH264, payload.type.Video.codec);
>-  EXPECT_FALSE(payload.type.Video.is_first_packet_in_frame);
>+  EXPECT_EQ(kVideoCodecH264, payload.video_header().codec);
>+  EXPECT_FALSE(payload.video_header().is_first_packet_in_frame);
>   {
>-    const RTPVideoHeaderH264& h264 = payload.type.Video.codecHeader.H264;
>+    const RTPVideoHeaderH264& h264 = payload.h264();
>     EXPECT_EQ(kH264FuA, h264.packetization_type);
>     EXPECT_EQ(kIdr, h264.nalu_type);
>     // NALU info is only expected for the first FU-A packet.
>     EXPECT_EQ(0u, h264.nalus_length);
>   }
> 
>-  payload = RtpDepacketizer::ParsedPayload();
>+  payload = H264ParsedPayload();
>   ASSERT_TRUE(depacketizer_->Parse(&payload, packet3, sizeof(packet3)));
>   ExpectPacket(&payload, kExpected3, sizeof(kExpected3));
>   EXPECT_EQ(kVideoFrameKey, payload.frame_type);
>-  EXPECT_EQ(kRtpVideoH264, payload.type.Video.codec);
>-  EXPECT_FALSE(payload.type.Video.is_first_packet_in_frame);
>+  EXPECT_EQ(kVideoCodecH264, payload.video_header().codec);
>+  EXPECT_FALSE(payload.video_header().is_first_packet_in_frame);
>   {
>-    const RTPVideoHeaderH264& h264 = payload.type.Video.codecHeader.H264;
>+    const RTPVideoHeaderH264& h264 = payload.h264();
>     EXPECT_EQ(kH264FuA, h264.packetization_type);
>     EXPECT_EQ(kIdr, h264.nalu_type);
>     // NALU info is only expected for the first FU-A packet.
>@@ -892,48 +891,48 @@ TEST_F(RtpDepacketizerH264Test, TestFuA) {
> TEST_F(RtpDepacketizerH264Test, TestEmptyPayload) {
>   // Using a wild pointer to crash on accesses from inside the depacketizer.
>   uint8_t* garbage_ptr = reinterpret_cast<uint8_t*>(0x4711);
>-  RtpDepacketizer::ParsedPayload payload;
>+  H264ParsedPayload payload;
>   EXPECT_FALSE(depacketizer_->Parse(&payload, garbage_ptr, 0));
> }
> 
> TEST_F(RtpDepacketizerH264Test, TestTruncatedFuaNalu) {
>   const uint8_t kPayload[] = {0x9c};
>-  RtpDepacketizer::ParsedPayload payload;
>+  H264ParsedPayload payload;
>   EXPECT_FALSE(depacketizer_->Parse(&payload, kPayload, sizeof(kPayload)));
> }
> 
> TEST_F(RtpDepacketizerH264Test, TestTruncatedSingleStapANalu) {
>   const uint8_t kPayload[] = {0xd8, 0x27};
>-  RtpDepacketizer::ParsedPayload payload;
>+  H264ParsedPayload payload;
>   EXPECT_FALSE(depacketizer_->Parse(&payload, kPayload, sizeof(kPayload)));
> }
> 
> TEST_F(RtpDepacketizerH264Test, TestStapAPacketWithTruncatedNalUnits) {
>-  const uint8_t kPayload[] = { 0x58, 0xCB, 0xED, 0xDF};
>-  RtpDepacketizer::ParsedPayload payload;
>+  const uint8_t kPayload[] = {0x58, 0xCB, 0xED, 0xDF};
>+  H264ParsedPayload payload;
>   EXPECT_FALSE(depacketizer_->Parse(&payload, kPayload, sizeof(kPayload)));
> }
> 
> TEST_F(RtpDepacketizerH264Test, TestTruncationJustAfterSingleStapANalu) {
>   const uint8_t kPayload[] = {0x38, 0x27, 0x27};
>-  RtpDepacketizer::ParsedPayload payload;
>+  H264ParsedPayload payload;
>   EXPECT_FALSE(depacketizer_->Parse(&payload, kPayload, sizeof(kPayload)));
> }
> 
> TEST_F(RtpDepacketizerH264Test, TestShortSpsPacket) {
>   const uint8_t kPayload[] = {0x27, 0x80, 0x00};
>-  RtpDepacketizer::ParsedPayload payload;
>+  H264ParsedPayload payload;
>   EXPECT_TRUE(depacketizer_->Parse(&payload, kPayload, sizeof(kPayload)));
> }
> 
> TEST_F(RtpDepacketizerH264Test, TestSeiPacket) {
>   const uint8_t kPayload[] = {
>-      kSei,  // F=0, NRI=0, Type=6.
>+      kSei,                   // F=0, NRI=0, Type=6.
>       0x03, 0x03, 0x03, 0x03  // Payload.
>   };
>-  RtpDepacketizer::ParsedPayload payload;
>+  H264ParsedPayload payload;
>   ASSERT_TRUE(depacketizer_->Parse(&payload, kPayload, sizeof(kPayload)));
>-  const RTPVideoHeaderH264& h264 = payload.type.Video.codecHeader.H264;
>+  const RTPVideoHeaderH264& h264 = payload.h264();
>   EXPECT_EQ(kVideoFrameDelta, payload.frame_type);
>   EXPECT_EQ(kH264SingleNalu, h264.packetization_type);
>   EXPECT_EQ(kSei, h264.nalu_type);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.cc
>index 0c9bb439e08..1a8efc1289b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.cc
>@@ -18,20 +18,27 @@
> namespace webrtc {
> 
> static const size_t kGenericHeaderLength = 1;
>-
>-RtpPacketizerGeneric::RtpPacketizerGeneric(FrameType frame_type,
>-                                           size_t max_payload_len,
>-                                           size_t last_packet_reduction_len)
>-    : payload_data_(NULL),
>+static const size_t kExtendedHeaderLength = 2;
>+
>+RtpPacketizerGeneric::RtpPacketizerGeneric(
>+    const RTPVideoHeader& rtp_video_header,
>+    FrameType frame_type,
>+    size_t max_payload_len,
>+    size_t last_packet_reduction_len)
>+    : picture_id_(
>+          rtp_video_header.frame_id != kNoPictureId
>+              ? absl::optional<uint16_t>(rtp_video_header.frame_id & 0x7FFF)
>+              : absl::nullopt),
>+      payload_data_(nullptr),
>       payload_size_(0),
>-      max_payload_len_(max_payload_len - kGenericHeaderLength),
>+      max_payload_len_(max_payload_len - kGenericHeaderLength -
>+                       (picture_id_.has_value() ? kExtendedHeaderLength : 0)),
>       last_packet_reduction_len_(last_packet_reduction_len),
>       frame_type_(frame_type),
>       num_packets_left_(0),
>       num_larger_packets_(0) {}
> 
>-RtpPacketizerGeneric::~RtpPacketizerGeneric() {
>-}
>+RtpPacketizerGeneric::~RtpPacketizerGeneric() {}
> 
> size_t RtpPacketizerGeneric::SetPayloadData(
>     const uint8_t* payload_data,
>@@ -63,6 +70,10 @@ size_t RtpPacketizerGeneric::SetPayloadData(
>   if (frame_type_ == kVideoFrameKey) {
>     generic_header_ |= RtpFormatVideoGeneric::kKeyFrameBit;
>   }
>+  if (picture_id_.has_value()) {
>+    generic_header_ |= RtpFormatVideoGeneric::kExtendedHeaderBit;
>+  }
>+
>   return num_packets_left_;
> }
> 
>@@ -87,16 +98,24 @@ bool RtpPacketizerGeneric::NextPacket(RtpPacketToSend* packet) {
>   }
>   RTC_DCHECK_LE(next_packet_payload_len, max_payload_len_);
> 
>-  uint8_t* out_ptr =
>-      packet->AllocatePayload(kGenericHeaderLength + next_packet_payload_len);
>+  size_t total_length = next_packet_payload_len + kGenericHeaderLength +
>+                        (picture_id_.has_value() ? kExtendedHeaderLength : 0);
>+  uint8_t* out_ptr = packet->AllocatePayload(total_length);
>+
>   // Put generic header in packet.
>   out_ptr[0] = generic_header_;
>+  out_ptr += kGenericHeaderLength;
>+
>+  if (picture_id_.has_value()) {
>+    WriteExtendedHeader(out_ptr);
>+    out_ptr += kExtendedHeaderLength;
>+  }
>+
>   // Remove first-packet bit, following packets are intermediate.
>   generic_header_ &= ~RtpFormatVideoGeneric::kFirstPacketBit;
> 
>   // Put payload in packet.
>-  memcpy(out_ptr + kGenericHeaderLength, payload_data_,
>-         next_packet_payload_len);
>+  memcpy(out_ptr, payload_data_, next_packet_payload_len);
>   payload_data_ += next_packet_payload_len;
>   payload_size_ -= next_packet_payload_len;
>   --num_packets_left_;
>@@ -112,12 +131,21 @@ std::string RtpPacketizerGeneric::ToString() {
>   return "RtpPacketizerGeneric";
> }
> 
>+void RtpPacketizerGeneric::WriteExtendedHeader(uint8_t* out_ptr) {
>+  // Store bottom 15 bits of the the sequence number. Only 15 bits are used for
>+  // compatibility with other packetizer implemenetations that also use 15 bits.
>+  out_ptr[0] = (*picture_id_ >> 8) & 0x7F;
>+  out_ptr[1] = *picture_id_ & 0xFF;
>+}
>+
>+RtpDepacketizerGeneric::~RtpDepacketizerGeneric() = default;
>+
> bool RtpDepacketizerGeneric::Parse(ParsedPayload* parsed_payload,
>                                    const uint8_t* payload_data,
>                                    size_t payload_data_length) {
>   assert(parsed_payload != NULL);
>   if (payload_data_length == 0) {
>-    RTC_LOG(LS_ERROR) << "Empty payload.";
>+    RTC_LOG(LS_WARNING) << "Empty payload.";
>     return false;
>   }
> 
>@@ -128,11 +156,22 @@ bool RtpDepacketizerGeneric::Parse(ParsedPayload* parsed_payload,
>       ((generic_header & RtpFormatVideoGeneric::kKeyFrameBit) != 0)
>           ? kVideoFrameKey
>           : kVideoFrameDelta;
>-  parsed_payload->type.Video.is_first_packet_in_frame =
>+  parsed_payload->video_header().is_first_packet_in_frame =
>       (generic_header & RtpFormatVideoGeneric::kFirstPacketBit) != 0;
>-  parsed_payload->type.Video.codec = kRtpVideoGeneric;
>-  parsed_payload->type.Video.width = 0;
>-  parsed_payload->type.Video.height = 0;
>+  parsed_payload->video_header().codec = kVideoCodecGeneric;
>+  parsed_payload->video_header().width = 0;
>+  parsed_payload->video_header().height = 0;
>+
>+  if (generic_header & RtpFormatVideoGeneric::kExtendedHeaderBit) {
>+    if (payload_data_length < kExtendedHeaderLength) {
>+      RTC_LOG(LS_WARNING) << "Too short payload for generic header.";
>+      return false;
>+    }
>+    parsed_payload->video_header().frame_id =
>+        ((payload_data[0] & 0x7F) << 8) | payload_data[1];
>+    payload_data += kExtendedHeaderLength;
>+    payload_data_length -= kExtendedHeaderLength;
>+  }
> 
>   parsed_payload->payload = payload_data;
>   parsed_payload->payload_length = payload_data_length;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h
>index a9da5da1d58..e608db59211 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic.h
>@@ -15,23 +15,26 @@
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/rtp_rtcp/source/rtp_format.h"
> #include "rtc_base/constructormagic.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> namespace RtpFormatVideoGeneric {
> static const uint8_t kKeyFrameBit = 0x01;
> static const uint8_t kFirstPacketBit = 0x02;
>+// If this bit is set, there will be an extended header contained in this
>+// packet. This was added later so old clients will not send this.
>+static const uint8_t kExtendedHeaderBit = 0x04;
> }  // namespace RtpFormatVideoGeneric
> 
> class RtpPacketizerGeneric : public RtpPacketizer {
>  public:
>   // Initialize with payload from encoder.
>   // The payload_data must be exactly one encoded generic frame.
>-  RtpPacketizerGeneric(FrameType frametype,
>+  RtpPacketizerGeneric(const RTPVideoHeader& rtp_video_header,
>+                       FrameType frametype,
>                        size_t max_payload_len,
>                        size_t last_packet_reduction_len);
> 
>-  virtual ~RtpPacketizerGeneric();
>+  ~RtpPacketizerGeneric() override;
> 
>   // Returns total number of packets to be generated.
>   size_t SetPayloadData(const uint8_t* payload_data,
>@@ -46,6 +49,7 @@ class RtpPacketizerGeneric : public RtpPacketizer {
>   std::string ToString() override;
> 
>  private:
>+  const absl::optional<uint16_t> picture_id_;
>   const uint8_t* payload_data_;
>   size_t payload_size_;
>   const size_t max_payload_len_;
>@@ -58,13 +62,15 @@ class RtpPacketizerGeneric : public RtpPacketizer {
>   // Number of packets, which will be 1 byte more than the rest.
>   size_t num_larger_packets_;
> 
>+  void WriteExtendedHeader(uint8_t* out_ptr);
>+
>   RTC_DISALLOW_COPY_AND_ASSIGN(RtpPacketizerGeneric);
> };
> 
> // Depacketizer for generic codec.
> class RtpDepacketizerGeneric : public RtpDepacketizer {
>  public:
>-  virtual ~RtpDepacketizerGeneric() {}
>+  ~RtpDepacketizerGeneric() override;
> 
>   bool Parse(ParsedPayload* parsed_payload,
>              const uint8_t* payload_data,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic_unittest.cc
>index 983bd8fc99c..35ffa670bce 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_video_generic_unittest.cc
>@@ -63,8 +63,8 @@ TEST(RtpPacketizerVideoGeneric, AllPacketsMayBeEqualAndRespectMaxPayloadSize) {
>   const size_t kMaxPayloadLen = 6;
>   const size_t kLastPacketReductionLen = 2;
>   const size_t kPayloadSize = 13;
>-  RtpPacketizerGeneric packetizer(kVideoFrameKey, kMaxPayloadLen,
>-                                  kLastPacketReductionLen);
>+  RtpPacketizerGeneric packetizer(RTPVideoHeader(), kVideoFrameKey,
>+                                  kMaxPayloadLen, kLastPacketReductionLen);
>   size_t num_packets =
>       packetizer.SetPayloadData(kTestPayload, kPayloadSize, nullptr);
>   std::vector<size_t> payload_sizes = NextPacketFillPayloadSizes(&packetizer);
>@@ -78,8 +78,8 @@ TEST(RtpPacketizerVideoGeneric,
>   const size_t kMaxPayloadLen = 6;
>   const size_t kLastPacketReductionLen = 2;
>   const size_t kPayloadSize = 13;
>-  RtpPacketizerGeneric packetizer(kVideoFrameKey, kMaxPayloadLen,
>-                                  kLastPacketReductionLen);
>+  RtpPacketizerGeneric packetizer(RTPVideoHeader(), kVideoFrameKey,
>+                                  kMaxPayloadLen, kLastPacketReductionLen);
>   size_t num_packets =
>       packetizer.SetPayloadData(kTestPayload, kPayloadSize, nullptr);
>   std::vector<size_t> payload_sizes = NextPacketFillPayloadSizes(&packetizer);
>@@ -93,8 +93,8 @@ TEST(RtpPacketizerVideoGeneric,
>   const size_t kMaxPayloadLen = 6;
>   const size_t kLastPacketReductionLen = 2;
>   const size_t kPayloadSize = 13;
>-  RtpPacketizerGeneric packetizer(kVideoFrameKey, kMaxPayloadLen,
>-                                  kLastPacketReductionLen);
>+  RtpPacketizerGeneric packetizer(RTPVideoHeader(), kVideoFrameKey,
>+                                  kMaxPayloadLen, kLastPacketReductionLen);
>   size_t num_packets =
>       packetizer.SetPayloadData(kTestPayload, kPayloadSize, nullptr);
>   std::vector<size_t> payload_sizes = NextPacketFillPayloadSizes(&packetizer);
>@@ -114,8 +114,8 @@ TEST(RtpPacketizerVideoGeneric,
>   // generic header lengh for each packet minus last packet reduction).
>   // 4 packets is enough for kPayloadSize.
>   const size_t kMinNumPackets = 4;
>-  RtpPacketizerGeneric packetizer(kVideoFrameKey, kMaxPayloadLen,
>-                                  kLastPacketReductionLen);
>+  RtpPacketizerGeneric packetizer(RTPVideoHeader(), kVideoFrameKey,
>+                                  kMaxPayloadLen, kLastPacketReductionLen);
>   size_t num_packets =
>       packetizer.SetPayloadData(kTestPayload, kPayloadSize, nullptr);
>   std::vector<size_t> payload_sizes = NextPacketFillPayloadSizes(&packetizer);
>@@ -128,8 +128,8 @@ TEST(RtpPacketizerVideoGeneric, SomePacketsAreSmaller_RespectsMaxPayloadSize) {
>   const size_t kMaxPayloadLen = 8;
>   const size_t kLastPacketReductionLen = 5;
>   const size_t kPayloadSize = 28;
>-  RtpPacketizerGeneric packetizer(kVideoFrameKey, kMaxPayloadLen,
>-                                  kLastPacketReductionLen);
>+  RtpPacketizerGeneric packetizer(RTPVideoHeader(), kVideoFrameKey,
>+                                  kMaxPayloadLen, kLastPacketReductionLen);
>   size_t num_packets =
>       packetizer.SetPayloadData(kTestPayload, kPayloadSize, nullptr);
>   std::vector<size_t> payload_sizes = NextPacketFillPayloadSizes(&packetizer);
>@@ -143,8 +143,8 @@ TEST(RtpPacketizerVideoGeneric,
>   const size_t kMaxPayloadLen = 8;
>   const size_t kLastPacketReductionLen = 5;
>   const size_t kPayloadSize = 28;
>-  RtpPacketizerGeneric packetizer(kVideoFrameKey, kMaxPayloadLen,
>-                                  kLastPacketReductionLen);
>+  RtpPacketizerGeneric packetizer(RTPVideoHeader(), kVideoFrameKey,
>+                                  kMaxPayloadLen, kLastPacketReductionLen);
>   size_t num_packets =
>       packetizer.SetPayloadData(kTestPayload, kPayloadSize, nullptr);
>   std::vector<size_t> payload_sizes = NextPacketFillPayloadSizes(&packetizer);
>@@ -158,8 +158,8 @@ TEST(RtpPacketizerVideoGeneric,
>   const size_t kMaxPayloadLen = 8;
>   const size_t kLastPacketReductionLen = 5;
>   const size_t kPayloadSize = 28;
>-  RtpPacketizerGeneric packetizer(kVideoFrameKey, kMaxPayloadLen,
>-                                  kLastPacketReductionLen);
>+  RtpPacketizerGeneric packetizer(RTPVideoHeader(), kVideoFrameKey,
>+                                  kMaxPayloadLen, kLastPacketReductionLen);
>   size_t num_packets =
>       packetizer.SetPayloadData(kTestPayload, kPayloadSize, nullptr);
>   std::vector<size_t> payload_sizes = NextPacketFillPayloadSizes(&packetizer);
>@@ -179,8 +179,8 @@ TEST(RtpPacketizerVideoGeneric,
>   // generic header lengh for each packet minus last packet reduction).
>   // 5 packets is enough for kPayloadSize.
>   const size_t kMinNumPackets = 5;
>-  RtpPacketizerGeneric packetizer(kVideoFrameKey, kMaxPayloadLen,
>-                                  kLastPacketReductionLen);
>+  RtpPacketizerGeneric packetizer(RTPVideoHeader(), kVideoFrameKey,
>+                                  kMaxPayloadLen, kLastPacketReductionLen);
>   size_t num_packets =
>       packetizer.SetPayloadData(kTestPayload, kPayloadSize, nullptr);
>   std::vector<size_t> payload_sizes = NextPacketFillPayloadSizes(&packetizer);
>@@ -189,4 +189,84 @@ TEST(RtpPacketizerVideoGeneric,
>   EXPECT_EQ(num_packets, kMinNumPackets);
> }
> 
>+TEST(RtpPacketizerVideoGeneric, HasFrameIdWritesExtendedHeader) {
>+  const size_t kMaxPayloadLen = 6;
>+  const size_t kLastPacketReductionLen = 2;
>+  const size_t kPayloadSize = 13;
>+
>+  RTPVideoHeader rtp_video_header;
>+  rtp_video_header.frame_id = 37;
>+  RtpPacketizerGeneric packetizer(rtp_video_header, kVideoFrameKey,
>+                                  kMaxPayloadLen, kLastPacketReductionLen);
>+  packetizer.SetPayloadData(kTestPayload, kPayloadSize, nullptr);
>+
>+  RtpPacketToSend packet(nullptr);
>+  packetizer.NextPacket(&packet);
>+
>+  rtc::ArrayView<const uint8_t> payload = packet.payload();
>+  EXPECT_TRUE(payload[0] & 0x04);  // Extended header bit is set.
>+  // Frame id is 37.
>+  EXPECT_EQ(0u, payload[1]);
>+  EXPECT_EQ(37u, payload[2]);
>+}
>+
>+TEST(RtpPacketizerVideoGeneric, FrameIdOver15bitsWrapsAround) {
>+  const size_t kMaxPayloadLen = 6;
>+  const size_t kLastPacketReductionLen = 2;
>+  const size_t kPayloadSize = 13;
>+
>+  RTPVideoHeader rtp_video_header;
>+  rtp_video_header.frame_id = 0x8137;
>+  RtpPacketizerGeneric packetizer(rtp_video_header, kVideoFrameKey,
>+                                  kMaxPayloadLen, kLastPacketReductionLen);
>+  packetizer.SetPayloadData(kTestPayload, kPayloadSize, nullptr);
>+
>+  RtpPacketToSend packet(nullptr);
>+  packetizer.NextPacket(&packet);
>+
>+  rtc::ArrayView<const uint8_t> payload = packet.payload();
>+  EXPECT_TRUE(payload[0] & 0x04);  // Extended header bit is set.
>+  // Frame id is 0x137.
>+  EXPECT_EQ(0x01u, payload[1]);
>+  EXPECT_EQ(0x37u, payload[2]);
>+}
>+
>+TEST(RtpPacketizerVideoGeneric, NoFrameIdDoesNotWriteExtendedHeader) {
>+  const size_t kMaxPayloadLen = 6;
>+  const size_t kLastPacketReductionLen = 2;
>+  const size_t kPayloadSize = 13;
>+
>+  RtpPacketizerGeneric packetizer(RTPVideoHeader(), kVideoFrameKey,
>+                                  kMaxPayloadLen, kLastPacketReductionLen);
>+  packetizer.SetPayloadData(kTestPayload, kPayloadSize, nullptr);
>+
>+  RtpPacketToSend packet(nullptr);
>+  packetizer.NextPacket(&packet);
>+
>+  rtc::ArrayView<const uint8_t> payload = packet.payload();
>+  EXPECT_FALSE(payload[0] & 0x04);
>+}
>+
>+TEST(RtpDepacketizerVideoGeneric, NonExtendedHeaderNoFrameId) {
>+  const size_t kPayloadLen = 1;
>+  uint8_t payload[kPayloadLen] = {0x01};
>+
>+  RtpDepacketizerGeneric depacketizer;
>+  RtpDepacketizer::ParsedPayload parsed_payload;
>+  depacketizer.Parse(&parsed_payload, payload, kPayloadLen);
>+
>+  EXPECT_EQ(kNoPictureId, parsed_payload.video_header().frame_id);
>+}
>+
>+TEST(RtpDepacketizerVideoGeneric, ExtendedHeaderParsesFrameId) {
>+  const size_t kPayloadLen = 3;
>+  uint8_t payload[kPayloadLen] = {0x05, 0x13, 0x37};
>+
>+  RtpDepacketizerGeneric depacketizer;
>+  RtpDepacketizer::ParsedPayload parsed_payload;
>+  depacketizer.Parse(&parsed_payload, payload, kPayloadLen);
>+
>+  EXPECT_EQ(0x1337, parsed_payload.video_header().frame_id);
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_video_stereo.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_video_stereo.cc
>deleted file mode 100644
>index 7c3a93362c5..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_video_stereo.cc
>+++ /dev/null
>@@ -1,164 +0,0 @@
>-/*
>- *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include <memory>
>-#include <string>
>-
>-#include "modules/include/module_common_types.h"
>-#include "modules/rtp_rtcp/source/byte_io.h"
>-#include "modules/rtp_rtcp/source/rtp_format_video_stereo.h"
>-#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
>-#include "rtc_base/logging.h"
>-
>-namespace webrtc {
>-
>-namespace {
>-// Write the Stereo header descriptor.
>-//       0 1 2 3 4 5 6 7 8
>-//      +-+-+-+-+-+-+-+-+-+
>-//      | VideoCodecType  | (optional)
>-//      +-+-+-+-+-+-+-+-+-+
>-//      |   frame_index   | (optional)
>-//      +-+-+-+-+-+-+-+-+-+
>-//      |   frame_count   | (optional)
>-//      +-+-+-+-+-+-+-+-+-+
>-//      |  picture_index  | (optional)
>-//      |    (16 bits)    |
>-//      +-+-+-+-+-+-+-+-+-+
>-//      |  HeaderMarker   | (mandatory)
>-//      +-+-+-+-+-+-+-+-+-+
>-constexpr size_t kStereoHeaderLength =
>-    sizeof(uint8_t) + sizeof(uint8_t) + sizeof(uint8_t) + sizeof(uint16_t);
>-
>-constexpr size_t kHeaderMarkerLength = 1;
>-constexpr uint8_t kHeaderMarkerBit = 0x02;
>-}  // namespace
>-
>-RtpPacketizerStereo::RtpPacketizerStereo(const RTPVideoHeaderStereo& header,
>-                                         FrameType frame_type,
>-                                         size_t max_payload_len,
>-                                         size_t last_packet_reduction_len)
>-    : header_(header),
>-      max_payload_len_(max_payload_len - kHeaderMarkerLength),
>-      last_packet_reduction_len_(last_packet_reduction_len +
>-                                 kStereoHeaderLength),
>-      packetizer_(frame_type, max_payload_len_, last_packet_reduction_len_) {}
>-
>-RtpPacketizerStereo::~RtpPacketizerStereo() {}
>-
>-size_t RtpPacketizerStereo::SetPayloadData(
>-    const uint8_t* payload_data,
>-    size_t payload_size,
>-    const RTPFragmentationHeader* fragmentation) {
>-  num_packets_remaining_ =
>-      packetizer_.SetPayloadData(payload_data, payload_size, fragmentation);
>-  return num_packets_remaining_;
>-}
>-
>-bool RtpPacketizerStereo::NextPacket(RtpPacketToSend* packet) {
>-  if (max_payload_len_ <= last_packet_reduction_len_) {
>-    RTC_LOG(LS_ERROR) << "Payload length not large enough.";
>-    return false;
>-  }
>-
>-  RTC_DCHECK(packet);
>-  if (!packetizer_.NextPacket(packet))
>-    return false;
>-
>-  RTC_DCHECK_GT(num_packets_remaining_, 0);
>-  const bool last_packet = --num_packets_remaining_ == 0;
>-  const size_t header_length = last_packet
>-                                   ? kHeaderMarkerLength + kStereoHeaderLength
>-                                   : kHeaderMarkerLength;
>-
>-  const uint8_t* payload_ptr = packet->payload().data();
>-  const size_t payload_size = packet->payload_size();
>-  uint8_t* padded_payload_ptr =
>-      packet->SetPayloadSize(header_length + packet->payload_size());
>-  RTC_DCHECK(padded_payload_ptr);
>-
>-  padded_payload_ptr += payload_size;
>-  if (last_packet) {
>-    ByteWriter<uint8_t>::WriteBigEndian(
>-        padded_payload_ptr,
>-        static_cast<uint8_t>(header_.associated_codec_type));
>-    padded_payload_ptr += sizeof(uint8_t);
>-    ByteWriter<uint8_t>::WriteBigEndian(padded_payload_ptr,
>-                                        header_.indices.frame_index);
>-    padded_payload_ptr += sizeof(uint8_t);
>-    ByteWriter<uint8_t>::WriteBigEndian(padded_payload_ptr,
>-                                        header_.indices.frame_count);
>-    padded_payload_ptr += sizeof(uint8_t);
>-    ByteWriter<uint16_t>::WriteBigEndian(padded_payload_ptr,
>-                                         header_.indices.picture_index);
>-    padded_payload_ptr += sizeof(uint16_t);
>-    RTC_DCHECK_EQ(payload_size + kStereoHeaderLength,
>-                  padded_payload_ptr - payload_ptr);
>-  }
>-  padded_payload_ptr[0] = last_packet ? kHeaderMarkerBit : 0;
>-  return true;
>-}
>-
>-std::string RtpPacketizerStereo::ToString() {
>-  return "RtpPacketizerStereo";
>-}
>-
>-RtpDepacketizerStereo::~RtpDepacketizerStereo() {}
>-
>-bool RtpDepacketizerStereo::Parse(ParsedPayload* parsed_payload,
>-                                  const uint8_t* payload_data,
>-                                  size_t payload_data_length) {
>-  RTC_DCHECK(parsed_payload);
>-  if (payload_data_length == 0) {
>-    RTC_LOG(LS_ERROR) << "Empty payload.";
>-    return false;
>-  }
>-
>-  uint8_t header_marker = payload_data[payload_data_length - 1];
>-  --payload_data_length;
>-  const bool last_packet = (header_marker & kHeaderMarkerBit) != 0;
>-
>-  if (last_packet) {
>-    if (payload_data_length <= kStereoHeaderLength) {
>-      RTC_LOG(LS_WARNING) << "Payload not large enough.";
>-      return false;
>-    }
>-    size_t offset = payload_data_length - kStereoHeaderLength;
>-    uint8_t associated_codec_type =
>-        ByteReader<uint8_t>::ReadBigEndian(&payload_data[offset]);
>-    switch (associated_codec_type) {
>-      case kRtpVideoVp8:
>-      case kRtpVideoVp9:
>-      case kRtpVideoH264:
>-        break;
>-      default:
>-        RTC_LOG(LS_WARNING) << "Unexpected codec type.";
>-        return false;
>-    }
>-    parsed_payload->type.Video.codecHeader.stereo.associated_codec_type =
>-        static_cast<RtpVideoCodecTypes>(associated_codec_type);
>-    offset += sizeof(uint8_t);
>-    parsed_payload->type.Video.codecHeader.stereo.indices.frame_index =
>-        ByteReader<uint8_t>::ReadBigEndian(&payload_data[offset]);
>-    offset += sizeof(uint8_t);
>-    parsed_payload->type.Video.codecHeader.stereo.indices.frame_count =
>-        ByteReader<uint8_t>::ReadBigEndian(&payload_data[offset]);
>-    offset += sizeof(uint8_t);
>-    parsed_payload->type.Video.codecHeader.stereo.indices.picture_index =
>-        ByteReader<uint16_t>::ReadBigEndian(&payload_data[offset]);
>-    RTC_DCHECK_EQ(payload_data_length, offset + sizeof(uint16_t));
>-    payload_data_length -= kStereoHeaderLength;
>-  }
>-  if (!depacketizer_.Parse(parsed_payload, payload_data, payload_data_length))
>-    return false;
>-  parsed_payload->type.Video.codec = kRtpVideoStereo;
>-  return true;
>-}
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_video_stereo.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_video_stereo.h
>deleted file mode 100644
>index f054498d860..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_video_stereo.h
>+++ /dev/null
>@@ -1,65 +0,0 @@
>-/*
>- *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-#ifndef MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VIDEO_STEREO_H_
>-#define MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VIDEO_STEREO_H_
>-
>-#include <string>
>-
>-#include "modules/rtp_rtcp/source/rtp_format.h"
>-#include "modules/rtp_rtcp/source/rtp_format_video_generic.h"
>-#include "rtc_base/constructormagic.h"
>-
>-namespace webrtc {
>-
>-class RtpPacketizerStereo : public RtpPacketizer {
>- public:
>-  RtpPacketizerStereo(const RTPVideoHeaderStereo& header,
>-                      FrameType frame_type,
>-                      size_t max_payload_len,
>-                      size_t last_packet_reduction_len);
>-
>-  ~RtpPacketizerStereo() override;
>-
>-  size_t SetPayloadData(const uint8_t* payload_data,
>-                        size_t payload_size,
>-                        const RTPFragmentationHeader* fragmentation) override;
>-
>-  // Get the next payload with generic payload header.
>-  // Write payload and set marker bit of the |packet|.
>-  // Returns true on success, false otherwise.
>-  bool NextPacket(RtpPacketToSend* packet) override;
>-
>-  std::string ToString() override;
>-
>- private:
>-  const RTPVideoHeaderStereo header_;
>-  const size_t max_payload_len_;
>-  const size_t last_packet_reduction_len_;
>-  size_t num_packets_remaining_ = 0;
>-  // TODO(emircan): Use codec specific packetizers. If not possible, refactor
>-  // this class to have similar logic to generic packetizer.
>-  RtpPacketizerGeneric packetizer_;
>-
>-  RTC_DISALLOW_COPY_AND_ASSIGN(RtpPacketizerStereo);
>-};
>-
>-class RtpDepacketizerStereo : public RtpDepacketizer {
>- public:
>-  ~RtpDepacketizerStereo() override;
>-
>-  bool Parse(ParsedPayload* parsed_payload,
>-             const uint8_t* payload_data,
>-             size_t payload_data_length) override;
>-
>- private:
>-  RtpDepacketizerGeneric depacketizer_;
>-};
>-}  // namespace webrtc
>-#endif  // MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VIDEO_STEREO_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_video_stereo_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_video_stereo_unittest.cc
>deleted file mode 100644
>index f75dbce2813..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_video_stereo_unittest.cc
>+++ /dev/null
>@@ -1,118 +0,0 @@
>-/*
>- *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include <vector>
>-
>-#include "modules/rtp_rtcp/source/rtp_format_video_stereo.h"
>-#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
>-#include "test/gmock.h"
>-#include "test/gtest.h"
>-
>-namespace webrtc {
>-
>-namespace {
>-
>-using ::testing::Each;
>-using ::testing::ElementsAreArray;
>-using ::testing::Le;
>-using ::testing::SizeIs;
>-
>-constexpr RtpVideoCodecTypes kTestAssociatedCodecType = kRtpVideoVp9;
>-constexpr uint8_t kTestFrameIndex = 23;
>-constexpr uint8_t kTestFrameCount = 34;
>-constexpr uint16_t kTestPictureIndex = 123;
>-
>-RTPVideoHeaderStereo GenerateTestStereoHeader() {
>-  RTPVideoHeaderStereo header;
>-  header.associated_codec_type = kTestAssociatedCodecType;
>-  header.indices.frame_index = kTestFrameIndex;
>-  header.indices.frame_count = kTestFrameCount;
>-  header.indices.picture_index = kTestPictureIndex;
>-  return header;
>-}
>-
>-std::vector<uint8_t> GenerateTestPayload() {
>-  const size_t kPayloadSize = 68;
>-  return std::vector<uint8_t>(kPayloadSize, 0);
>-}
>-
>-std::vector<size_t> NextPacketFillPayloadSizes(
>-    RtpPacketizerStereo* packetizer) {
>-  RtpPacketToSend packet(nullptr);
>-  std::vector<size_t> result;
>-  while (packetizer->NextPacket(&packet))
>-    result.push_back(packet.payload_size());
>-  return result;
>-}
>-
>-}  // namespace
>-
>-TEST(RtpPacketizerVideoStereo, SmallMaxPayloadSizeThrowsErrors) {
>-  const size_t kMaxPayloadLen = 7;
>-  const size_t kLastPacketReductionLen = 2;
>-  RtpPacketizerStereo packetizer(GenerateTestStereoHeader(), kVideoFrameKey,
>-                                 kMaxPayloadLen, kLastPacketReductionLen);
>-  const std::vector<uint8_t>& test_payload = GenerateTestPayload();
>-  packetizer.SetPayloadData(test_payload.data(), test_payload.size(), nullptr);
>-  RtpPacketToSend packet(nullptr);
>-  EXPECT_FALSE(packetizer.NextPacket(&packet));
>-}
>-
>-TEST(RtpPacketizerVideoStereo, AllPacketsRespectMaxPayloadSize) {
>-  const size_t kMaxPayloadLen = 34;
>-  const size_t kLastPacketReductionLen = 2;
>-  RtpPacketizerStereo packetizer(GenerateTestStereoHeader(), kVideoFrameKey,
>-                                 kMaxPayloadLen, kLastPacketReductionLen);
>-  const std::vector<uint8_t>& test_payload = GenerateTestPayload();
>-  size_t num_packets = packetizer.SetPayloadData(test_payload.data(),
>-                                                 test_payload.size(), nullptr);
>-  std::vector<size_t> payload_sizes = NextPacketFillPayloadSizes(&packetizer);
>-  EXPECT_THAT(payload_sizes, SizeIs(num_packets));
>-  EXPECT_THAT(payload_sizes, Each(Le(kMaxPayloadLen)));
>-}
>-
>-TEST(RtpPacketizerVideoStereo, PreservesTypeAndHeader) {
>-  const size_t kMaxPayloadLen = 34;
>-  const size_t kLastPacketReductionLen = 2;
>-  const auto kFrameType = kVideoFrameKey;
>-  RtpPacketizerStereo packetizer(GenerateTestStereoHeader(), kFrameType,
>-                                 kMaxPayloadLen, kLastPacketReductionLen);
>-  const std::vector<uint8_t>& test_payload = GenerateTestPayload();
>-  packetizer.SetPayloadData(test_payload.data(), test_payload.size(), nullptr);
>-  RtpPacketToSend packet(nullptr);
>-  std::vector<RtpPacketToSend> result;
>-  while (packetizer.NextPacket(&packet)) {
>-    result.push_back(packet);
>-    packet = RtpPacketToSend(nullptr);
>-  }
>-
>-  RtpDepacketizerStereo depacketizer;
>-  const auto& first_payload = result.front().payload();
>-  RtpDepacketizer::ParsedPayload parsed_payload;
>-  ASSERT_TRUE(depacketizer.Parse(&parsed_payload, first_payload.data(),
>-                                 first_payload.size()));
>-  EXPECT_TRUE(parsed_payload.type.Video.is_first_packet_in_frame);
>-
>-  const auto& last_payload = result.back().payload();
>-  ASSERT_TRUE(depacketizer.Parse(&parsed_payload, last_payload.data(),
>-                                 last_payload.size()));
>-  EXPECT_EQ(kFrameType, parsed_payload.frame_type);
>-  EXPECT_EQ(kRtpVideoStereo, parsed_payload.type.Video.codec);
>-  EXPECT_EQ(kTestAssociatedCodecType,
>-            parsed_payload.type.Video.codecHeader.stereo.associated_codec_type);
>-  EXPECT_EQ(kTestFrameIndex,
>-            parsed_payload.type.Video.codecHeader.stereo.indices.frame_index);
>-  EXPECT_EQ(kTestFrameCount,
>-            parsed_payload.type.Video.codecHeader.stereo.indices.frame_count);
>-  EXPECT_EQ(kTestPictureIndex,
>-            parsed_payload.type.Video.codecHeader.stereo.indices.picture_index);
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.cc
>index 74d20c50b2c..48c73512abb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.cc
>@@ -109,9 +109,8 @@ int ParseVP8Extension(RTPVideoHeaderVP8* vp8,
>   }
> 
>   if (has_tid || has_key_idx) {
>-    if (ParseVP8TIDAndKeyIdx(
>-            vp8, &data, &data_length, &parsed_bytes, has_tid, has_key_idx) !=
>-        0) {
>+    if (ParseVP8TIDAndKeyIdx(vp8, &data, &data_length, &parsed_bytes, has_tid,
>+                             has_key_idx) != 0) {
>       return -1;
>     }
>   }
>@@ -130,8 +129,8 @@ int ParseVP8FrameSize(RtpDepacketizer::ParsedPayload* parsed_payload,
>     // in the beginning of the partition.
>     return -1;
>   }
>-  parsed_payload->type.Video.width = ((data[7] << 8) + data[6]) & 0x3FFF;
>-  parsed_payload->type.Video.height = ((data[9] << 8) + data[8]) & 0x3FFF;
>+  parsed_payload->video_header().width = ((data[7] << 8) + data[6]) & 0x3FFF;
>+  parsed_payload->video_header().height = ((data[9] << 8) + data[8]) & 0x3FFF;
>   return 0;
> }
> 
>@@ -171,8 +170,7 @@ RtpPacketizerVp8::RtpPacketizerVp8(const RTPVideoHeaderVP8& hdr_info,
>   RTC_DCHECK(ValidateHeader(hdr_info));
> }
> 
>-RtpPacketizerVp8::~RtpPacketizerVp8() {
>-}
>+RtpPacketizerVp8::~RtpPacketizerVp8() {}
> 
> size_t RtpPacketizerVp8::SetPayloadData(
>     const uint8_t* payload_data,
>@@ -252,8 +250,8 @@ void RtpPacketizerVp8::GeneratePacketsSplitPayloadBalanced(size_t payload_len,
>     if (num_packets_left == 2 && current_packet_bytes == remaining_data) {
>       --current_packet_bytes;
>     }
>-    QueuePacket(payload_len - remaining_data,
>-                current_packet_bytes, remaining_data == payload_len);
>+    QueuePacket(payload_len - remaining_data, current_packet_bytes,
>+                remaining_data == payload_len);
>     remaining_data -= current_packet_bytes;
>     --num_packets_left;
>   }
>@@ -302,8 +300,7 @@ int RtpPacketizerVp8::WriteHeaderAndPayload(const InfoStruct& packet_info,
>     return -1;
> 
>   memcpy(&buffer[vp8_fixed_payload_descriptor_bytes_ + extension_length],
>-         &payload_data_[packet_info.payload_start_pos],
>-         packet_info.size);
>+         &payload_data_[packet_info.payload_start_pos], packet_info.size);
> 
>   // Return total length of written data.
>   return packet_info.size + vp8_fixed_payload_descriptor_bytes_ +
>@@ -318,20 +315,20 @@ int RtpPacketizerVp8::WriteExtensionFields(uint8_t* buffer,
>     *x_field = 0;
>     extension_length = 1;  // One octet for the X field.
>     if (PictureIdPresent()) {
>-      if (WritePictureIDFields(
>-              x_field, buffer, buffer_length, &extension_length) < 0) {
>+      if (WritePictureIDFields(x_field, buffer, buffer_length,
>+                               &extension_length) < 0) {
>         return -1;
>       }
>     }
>     if (TL0PicIdxFieldPresent()) {
>-      if (WriteTl0PicIdxFields(
>-              x_field, buffer, buffer_length, &extension_length) < 0) {
>+      if (WriteTl0PicIdxFields(x_field, buffer, buffer_length,
>+                               &extension_length) < 0) {
>         return -1;
>       }
>     }
>     if (TIDFieldPresent() || KeyIdxFieldPresent()) {
>-      if (WriteTIDAndKeyIdxFields(
>-              x_field, buffer, buffer_length, &extension_length) < 0) {
>+      if (WriteTIDAndKeyIdxFields(x_field, buffer, buffer_length,
>+                                  &extension_length) < 0) {
>         return -1;
>       }
>     }
>@@ -483,22 +480,22 @@ bool RtpDepacketizerVp8::Parse(ParsedPayload* parsed_payload,
>   bool beginning_of_partition = (*payload_data & 0x10) ? true : false;  // S bit
>   int partition_id = (*payload_data & 0x0F);  // PartID field
> 
>-  parsed_payload->type.Video.width = 0;
>-  parsed_payload->type.Video.height = 0;
>-  parsed_payload->type.Video.is_first_packet_in_frame =
>+  parsed_payload->video_header().width = 0;
>+  parsed_payload->video_header().height = 0;
>+  parsed_payload->video_header().is_first_packet_in_frame =
>       beginning_of_partition && (partition_id == 0);
>-  parsed_payload->type.Video.simulcastIdx = 0;
>-  parsed_payload->type.Video.codec = kRtpVideoVp8;
>-  parsed_payload->type.Video.codecHeader.VP8.nonReference =
>+  parsed_payload->video_header().simulcastIdx = 0;
>+  parsed_payload->video_header().codec = kVideoCodecVP8;
>+  parsed_payload->video_header().vp8().nonReference =
>       (*payload_data & 0x20) ? true : false;  // N bit
>-  parsed_payload->type.Video.codecHeader.VP8.partitionId = partition_id;
>-  parsed_payload->type.Video.codecHeader.VP8.beginningOfPartition =
>+  parsed_payload->video_header().vp8().partitionId = partition_id;
>+  parsed_payload->video_header().vp8().beginningOfPartition =
>       beginning_of_partition;
>-  parsed_payload->type.Video.codecHeader.VP8.pictureId = kNoPictureId;
>-  parsed_payload->type.Video.codecHeader.VP8.tl0PicIdx = kNoTl0PicIdx;
>-  parsed_payload->type.Video.codecHeader.VP8.temporalIdx = kNoTemporalIdx;
>-  parsed_payload->type.Video.codecHeader.VP8.layerSync = false;
>-  parsed_payload->type.Video.codecHeader.VP8.keyIdx = kNoKeyIdx;
>+  parsed_payload->video_header().vp8().pictureId = kNoPictureId;
>+  parsed_payload->video_header().vp8().tl0PicIdx = kNoTl0PicIdx;
>+  parsed_payload->video_header().vp8().temporalIdx = kNoTemporalIdx;
>+  parsed_payload->video_header().vp8().layerSync = false;
>+  parsed_payload->video_header().vp8().keyIdx = kNoKeyIdx;
> 
>   if (partition_id > 8) {
>     // Weak check for corrupt payload_data: PartID MUST NOT be larger than 8.
>@@ -515,8 +512,7 @@ bool RtpDepacketizerVp8::Parse(ParsedPayload* parsed_payload,
> 
>   if (extension) {
>     const int parsed_bytes =
>-        ParseVP8Extension(&parsed_payload->type.Video.codecHeader.VP8,
>-                          payload_data,
>+        ParseVP8Extension(&parsed_payload->video_header().vp8(), payload_data,
>                           payload_data_length);
>     if (parsed_bytes < 0)
>       return false;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h
>index 893086430a5..2b4468abe48 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_vp8.h
>@@ -32,7 +32,6 @@
> #include "modules/include/module_common_types.h"
> #include "modules/rtp_rtcp/source/rtp_format.h"
> #include "rtc_base/constructormagic.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -45,7 +44,7 @@ class RtpPacketizerVp8 : public RtpPacketizer {
>                    size_t max_payload_len,
>                    size_t last_packet_reduction_len);
> 
>-  virtual ~RtpPacketizerVp8();
>+  ~RtpPacketizerVp8() override;
> 
>   size_t SetPayloadData(const uint8_t* payload_data,
>                         size_t payload_size,
>@@ -82,13 +81,10 @@ class RtpPacketizerVp8 : public RtpPacketizer {
> 
>   // Splits given part of payload to packets with a given capacity. The last
>   // packet should be reduced by last_packet_reduction_len_.
>-  void GeneratePacketsSplitPayloadBalanced(size_t payload_len,
>-                                           size_t capacity);
>+  void GeneratePacketsSplitPayloadBalanced(size_t payload_len, size_t capacity);
> 
>   // Insert packet into packet queue.
>-  void QueuePacket(size_t start_pos,
>-                   size_t packet_size,
>-                   bool first_packet);
>+  void QueuePacket(size_t start_pos, size_t packet_size, bool first_packet);
> 
>   // Write the payload header and copy the payload to the buffer.
>   // The info in packet_info determines which part of the payload is written
>@@ -159,7 +155,7 @@ class RtpPacketizerVp8 : public RtpPacketizer {
> // Depacketizer for VP8.
> class RtpDepacketizerVp8 : public RtpDepacketizer {
>  public:
>-  virtual ~RtpDepacketizerVp8() {}
>+  ~RtpDepacketizerVp8() override = default;
> 
>   bool Parse(ParsedPayload* parsed_payload,
>              const uint8_t* payload_data,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.cc
>index 62d7c3c3bfd..8dc7ba0a4c1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.cc
>@@ -8,7 +8,6 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-
> #include "modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h"
> 
> #include "test/gtest.h"
>@@ -32,12 +31,13 @@ RtpFormatVp8TestHelper::RtpFormatVp8TestHelper(const RTPVideoHeaderVP8* hdr)
> 
> RtpFormatVp8TestHelper::~RtpFormatVp8TestHelper() {
>   delete fragmentation_;
>-  delete [] payload_data_;
>+  delete[] payload_data_;
> }
> 
> bool RtpFormatVp8TestHelper::Init(const size_t* partition_sizes,
>                                   size_t num_partitions) {
>-  if (inited_) return false;
>+  if (inited_)
>+    return false;
>   fragmentation_ = new RTPFragmentationHeader;
>   fragmentation_->VerifyAndAllocateFragmentationHeader(num_partitions);
>   payload_size_ = 0;
>@@ -95,7 +95,7 @@ void RtpFormatVp8TestHelper::GetAllPacketsAndCheck(
> // First octet tests.
> #define EXPECT_BIT_EQ(x, n, a) EXPECT_EQ((((x) >> (n)) & 0x1), a)
> 
>-#define EXPECT_RSV_ZERO(x) EXPECT_EQ(((x) & 0xE0), 0)
>+#define EXPECT_RSV_ZERO(x) EXPECT_EQ(((x)&0xE0), 0)
> 
> #define EXPECT_BIT_X_EQ(x, a) EXPECT_BIT_EQ(x, 7, a)
> 
>@@ -103,7 +103,7 @@ void RtpFormatVp8TestHelper::GetAllPacketsAndCheck(
> 
> #define EXPECT_BIT_S_EQ(x, a) EXPECT_BIT_EQ(x, 4, a)
> 
>-#define EXPECT_PART_ID_EQ(x, a) EXPECT_EQ(((x) & 0x0F), a)
>+#define EXPECT_PART_ID_EQ(x, a) EXPECT_EQ(((x)&0x0F), a)
> 
> // Extension fields tests
> #define EXPECT_BIT_I_EQ(x, a) EXPECT_BIT_EQ(x, 7, a)
>@@ -114,11 +114,11 @@ void RtpFormatVp8TestHelper::GetAllPacketsAndCheck(
> 
> #define EXPECT_BIT_K_EQ(x, a) EXPECT_BIT_EQ(x, 4, a)
> 
>-#define EXPECT_TID_EQ(x, a) EXPECT_EQ((((x) & 0xC0) >> 6), a)
>+#define EXPECT_TID_EQ(x, a) EXPECT_EQ((((x)&0xC0) >> 6), a)
> 
> #define EXPECT_BIT_Y_EQ(x, a) EXPECT_BIT_EQ(x, 5, a)
> 
>-#define EXPECT_KEYIDX_EQ(x, a) EXPECT_EQ(((x) & 0x1F), a)
>+#define EXPECT_KEYIDX_EQ(x, a) EXPECT_EQ(((x)&0x1F), a)
> 
> void RtpFormatVp8TestHelper::CheckHeader(bool frag_start) {
>   payload_start_ = 1;
>@@ -127,8 +127,7 @@ void RtpFormatVp8TestHelper::CheckHeader(bool frag_start) {
> 
>   if (hdr_info_->pictureId != kNoPictureId ||
>       hdr_info_->temporalIdx != kNoTemporalIdx ||
>-      hdr_info_->tl0PicIdx != kNoTl0PicIdx ||
>-      hdr_info_->keyIdx != kNoKeyIdx) {
>+      hdr_info_->tl0PicIdx != kNoTl0PicIdx || hdr_info_->keyIdx != kNoKeyIdx) {
>     EXPECT_BIT_X_EQ(buffer[0], 1);
>     ++payload_start_;
>     CheckPictureID();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h
>index 1ed63abdee2..f8d5c34f9d6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h
>@@ -22,7 +22,6 @@
> #include "modules/rtp_rtcp/source/rtp_format_vp8.h"
> #include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
> #include "rtc_base/constructormagic.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_unittest.cc
>index 0095560e5d9..7da320c1b57 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_vp8_unittest.cc
>@@ -15,7 +15,6 @@
> #include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
> #include "test/gmock.h"
> #include "test/gtest.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> #define CHECK_ARRAY_SIZE(expected_size, array)                     \
>   static_assert(expected_size == sizeof(array) / sizeof(array[0]), \
>@@ -59,30 +58,30 @@ constexpr RtpPacketToSend::ExtensionManager* kNoExtensions = nullptr;
> //      | padding       |
> //      :               :
> //      +-+-+-+-+-+-+-+-+
>-void VerifyBasicHeader(RTPTypeHeader* type, bool N, bool S, int part_id) {
>-  ASSERT_TRUE(type != NULL);
>-  EXPECT_EQ(N, type->Video.codecHeader.VP8.nonReference);
>-  EXPECT_EQ(S, type->Video.codecHeader.VP8.beginningOfPartition);
>-  EXPECT_EQ(part_id, type->Video.codecHeader.VP8.partitionId);
>+void VerifyBasicHeader(RTPVideoHeader* header, bool N, bool S, int part_id) {
>+  ASSERT_TRUE(header != NULL);
>+  EXPECT_EQ(N, header->vp8().nonReference);
>+  EXPECT_EQ(S, header->vp8().beginningOfPartition);
>+  EXPECT_EQ(part_id, header->vp8().partitionId);
> }
> 
>-void VerifyExtensions(RTPTypeHeader* type,
>+void VerifyExtensions(RTPVideoHeader* header,
>                       int16_t picture_id,   /* I */
>                       int16_t tl0_pic_idx,  /* L */
>                       uint8_t temporal_idx, /* T */
>                       int key_idx /* K */) {
>-  ASSERT_TRUE(type != NULL);
>-  EXPECT_EQ(picture_id, type->Video.codecHeader.VP8.pictureId);
>-  EXPECT_EQ(tl0_pic_idx, type->Video.codecHeader.VP8.tl0PicIdx);
>-  EXPECT_EQ(temporal_idx, type->Video.codecHeader.VP8.temporalIdx);
>-  EXPECT_EQ(key_idx, type->Video.codecHeader.VP8.keyIdx);
>+  ASSERT_TRUE(header != NULL);
>+  EXPECT_EQ(picture_id, header->vp8().pictureId);
>+  EXPECT_EQ(tl0_pic_idx, header->vp8().tl0PicIdx);
>+  EXPECT_EQ(temporal_idx, header->vp8().temporalIdx);
>+  EXPECT_EQ(key_idx, header->vp8().keyIdx);
> }
> }  // namespace
> 
> class RtpPacketizerVp8Test : public ::testing::Test {
>  protected:
>   RtpPacketizerVp8Test() : helper_(NULL) {}
>-  virtual void TearDown() { delete helper_; }
>+  void TearDown() override { delete helper_; }
>   bool Init(const size_t* partition_sizes, size_t num_partitions) {
>     hdr_info_.pictureId = kNoPictureId;
>     hdr_info_.nonReference = false;
>@@ -123,11 +122,8 @@ TEST_F(RtpPacketizerVp8Test, TestEqualSizeModeFallback) {
>   ASSERT_EQ(num_packets, kExpectedNum);
> 
>   helper_->set_sloppy_partitioning(true);
>-  helper_->GetAllPacketsAndCheck(&packetizer,
>-                                 kExpectedSizes,
>-                                 kExpectedPart,
>-                                 kExpectedFragStart,
>-                                 kExpectedNum);
>+  helper_->GetAllPacketsAndCheck(&packetizer, kExpectedSizes, kExpectedPart,
>+                                 kExpectedFragStart, kExpectedNum);
> }
> 
> TEST_F(RtpPacketizerVp8Test, TestEqualSizeWithLastPacketReduction) {
>@@ -184,11 +180,8 @@ TEST_F(RtpPacketizerVp8Test, TestNonReferenceBit) {
>   ASSERT_EQ(num_packets, kExpectedNum);
> 
>   helper_->set_sloppy_partitioning(true);
>-  helper_->GetAllPacketsAndCheck(&packetizer,
>-                                 kExpectedSizes,
>-                                 kExpectedPart,
>-                                 kExpectedFragStart,
>-                                 kExpectedNum);
>+  helper_->GetAllPacketsAndCheck(&packetizer, kExpectedSizes, kExpectedPart,
>+                                 kExpectedFragStart, kExpectedNum);
> }
> 
> // Verify Tl0PicIdx and TID fields, and layerSync bit.
>@@ -216,11 +209,8 @@ TEST_F(RtpPacketizerVp8Test, TestTl0PicIdxAndTID) {
>   CHECK_ARRAY_SIZE(kExpectedNum, kExpectedFragStart);
>   ASSERT_EQ(num_packets, kExpectedNum);
> 
>-  helper_->GetAllPacketsAndCheck(&packetizer,
>-                                 kExpectedSizes,
>-                                 kExpectedPart,
>-                                 kExpectedFragStart,
>-                                 kExpectedNum);
>+  helper_->GetAllPacketsAndCheck(&packetizer, kExpectedSizes, kExpectedPart,
>+                                 kExpectedFragStart, kExpectedNum);
> }
> 
> // Verify KeyIdx field.
>@@ -246,11 +236,8 @@ TEST_F(RtpPacketizerVp8Test, TestKeyIdx) {
>   CHECK_ARRAY_SIZE(kExpectedNum, kExpectedFragStart);
>   ASSERT_EQ(num_packets, kExpectedNum);
> 
>-  helper_->GetAllPacketsAndCheck(&packetizer,
>-                                 kExpectedSizes,
>-                                 kExpectedPart,
>-                                 kExpectedFragStart,
>-                                 kExpectedNum);
>+  helper_->GetAllPacketsAndCheck(&packetizer, kExpectedSizes, kExpectedPart,
>+                                 kExpectedFragStart, kExpectedNum);
> }
> 
> // Verify TID field and KeyIdx field in combination.
>@@ -277,17 +264,14 @@ TEST_F(RtpPacketizerVp8Test, TestTIDAndKeyIdx) {
>   CHECK_ARRAY_SIZE(kExpectedNum, kExpectedFragStart);
>   ASSERT_EQ(num_packets, kExpectedNum);
> 
>-  helper_->GetAllPacketsAndCheck(&packetizer,
>-                                 kExpectedSizes,
>-                                 kExpectedPart,
>-                                 kExpectedFragStart,
>-                                 kExpectedNum);
>+  helper_->GetAllPacketsAndCheck(&packetizer, kExpectedSizes, kExpectedPart,
>+                                 kExpectedFragStart, kExpectedNum);
> }
> 
> class RtpDepacketizerVp8Test : public ::testing::Test {
>  protected:
>   RtpDepacketizerVp8Test()
>-      : depacketizer_(RtpDepacketizer::Create(kRtpVideoVp8)) {}
>+      : depacketizer_(RtpDepacketizer::Create(kVideoCodecVP8)) {}
> 
>   void ExpectPacket(RtpDepacketizer::ParsedPayload* parsed_payload,
>                     const uint8_t* data,
>@@ -310,14 +294,14 @@ TEST_F(RtpDepacketizerVp8Test, BasicHeader) {
>   RtpDepacketizer::ParsedPayload payload;
> 
>   ASSERT_TRUE(depacketizer_->Parse(&payload, packet, sizeof(packet)));
>-  ExpectPacket(
>-      &payload, packet + kHeaderLength, sizeof(packet) - kHeaderLength);
>+  ExpectPacket(&payload, packet + kHeaderLength,
>+               sizeof(packet) - kHeaderLength);
> 
>   EXPECT_EQ(kVideoFrameDelta, payload.frame_type);
>-  EXPECT_EQ(kRtpVideoVp8, payload.type.Video.codec);
>-  VerifyBasicHeader(&payload.type, 0, 1, 4);
>-  VerifyExtensions(
>-      &payload.type, kNoPictureId, kNoTl0PicIdx, kNoTemporalIdx, kNoKeyIdx);
>+  EXPECT_EQ(kVideoCodecVP8, payload.video_header().codec);
>+  VerifyBasicHeader(&payload.video_header(), 0, 1, 4);
>+  VerifyExtensions(&payload.video_header(), kNoPictureId, kNoTl0PicIdx,
>+                   kNoTemporalIdx, kNoKeyIdx);
> }
> 
> TEST_F(RtpDepacketizerVp8Test, PictureID) {
>@@ -331,13 +315,13 @@ TEST_F(RtpDepacketizerVp8Test, PictureID) {
>   RtpDepacketizer::ParsedPayload payload;
> 
>   ASSERT_TRUE(depacketizer_->Parse(&payload, packet, sizeof(packet)));
>-  ExpectPacket(
>-      &payload, packet + kHeaderLength1, sizeof(packet) - kHeaderLength1);
>+  ExpectPacket(&payload, packet + kHeaderLength1,
>+               sizeof(packet) - kHeaderLength1);
>   EXPECT_EQ(kVideoFrameDelta, payload.frame_type);
>-  EXPECT_EQ(kRtpVideoVp8, payload.type.Video.codec);
>-  VerifyBasicHeader(&payload.type, 1, 0, 0);
>-  VerifyExtensions(
>-      &payload.type, kPictureId, kNoTl0PicIdx, kNoTemporalIdx, kNoKeyIdx);
>+  EXPECT_EQ(kVideoCodecVP8, payload.video_header().codec);
>+  VerifyBasicHeader(&payload.video_header(), 1, 0, 0);
>+  VerifyExtensions(&payload.video_header(), kPictureId, kNoTl0PicIdx,
>+                   kNoTemporalIdx, kNoKeyIdx);
> 
>   // Re-use packet, but change to long PictureID.
>   packet[2] = 0x80 | kPictureId;
>@@ -345,14 +329,11 @@ TEST_F(RtpDepacketizerVp8Test, PictureID) {
> 
>   payload = RtpDepacketizer::ParsedPayload();
>   ASSERT_TRUE(depacketizer_->Parse(&payload, packet, sizeof(packet)));
>-  ExpectPacket(
>-      &payload, packet + kHeaderLength2, sizeof(packet) - kHeaderLength2);
>-  VerifyBasicHeader(&payload.type, 1, 0, 0);
>-  VerifyExtensions(&payload.type,
>-                   (kPictureId << 8) + kPictureId,
>-                   kNoTl0PicIdx,
>-                   kNoTemporalIdx,
>-                   kNoKeyIdx);
>+  ExpectPacket(&payload, packet + kHeaderLength2,
>+               sizeof(packet) - kHeaderLength2);
>+  VerifyBasicHeader(&payload.video_header(), 1, 0, 0);
>+  VerifyExtensions(&payload.video_header(), (kPictureId << 8) + kPictureId,
>+                   kNoTl0PicIdx, kNoTemporalIdx, kNoKeyIdx);
> }
> 
> TEST_F(RtpDepacketizerVp8Test, Tl0PicIdx) {
>@@ -365,13 +346,13 @@ TEST_F(RtpDepacketizerVp8Test, Tl0PicIdx) {
>   RtpDepacketizer::ParsedPayload payload;
> 
>   ASSERT_TRUE(depacketizer_->Parse(&payload, packet, sizeof(packet)));
>-  ExpectPacket(
>-      &payload, packet + kHeaderLength, sizeof(packet) - kHeaderLength);
>+  ExpectPacket(&payload, packet + kHeaderLength,
>+               sizeof(packet) - kHeaderLength);
>   EXPECT_EQ(kVideoFrameKey, payload.frame_type);
>-  EXPECT_EQ(kRtpVideoVp8, payload.type.Video.codec);
>-  VerifyBasicHeader(&payload.type, 0, 1, 0);
>-  VerifyExtensions(
>-      &payload.type, kNoPictureId, kTl0PicIdx, kNoTemporalIdx, kNoKeyIdx);
>+  EXPECT_EQ(kVideoCodecVP8, payload.video_header().codec);
>+  VerifyBasicHeader(&payload.video_header(), 0, 1, 0);
>+  VerifyExtensions(&payload.video_header(), kNoPictureId, kTl0PicIdx,
>+                   kNoTemporalIdx, kNoKeyIdx);
> }
> 
> TEST_F(RtpDepacketizerVp8Test, TIDAndLayerSync) {
>@@ -383,13 +364,14 @@ TEST_F(RtpDepacketizerVp8Test, TIDAndLayerSync) {
>   RtpDepacketizer::ParsedPayload payload;
> 
>   ASSERT_TRUE(depacketizer_->Parse(&payload, packet, sizeof(packet)));
>-  ExpectPacket(
>-      &payload, packet + kHeaderLength, sizeof(packet) - kHeaderLength);
>+  ExpectPacket(&payload, packet + kHeaderLength,
>+               sizeof(packet) - kHeaderLength);
>   EXPECT_EQ(kVideoFrameDelta, payload.frame_type);
>-  EXPECT_EQ(kRtpVideoVp8, payload.type.Video.codec);
>-  VerifyBasicHeader(&payload.type, 0, 0, 8);
>-  VerifyExtensions(&payload.type, kNoPictureId, kNoTl0PicIdx, 2, kNoKeyIdx);
>-  EXPECT_FALSE(payload.type.Video.codecHeader.VP8.layerSync);
>+  EXPECT_EQ(kVideoCodecVP8, payload.video_header().codec);
>+  VerifyBasicHeader(&payload.video_header(), 0, 0, 8);
>+  VerifyExtensions(&payload.video_header(), kNoPictureId, kNoTl0PicIdx, 2,
>+                   kNoKeyIdx);
>+  EXPECT_FALSE(payload.video_header().vp8().layerSync);
> }
> 
> TEST_F(RtpDepacketizerVp8Test, KeyIdx) {
>@@ -402,13 +384,13 @@ TEST_F(RtpDepacketizerVp8Test, KeyIdx) {
>   RtpDepacketizer::ParsedPayload payload;
> 
>   ASSERT_TRUE(depacketizer_->Parse(&payload, packet, sizeof(packet)));
>-  ExpectPacket(
>-      &payload, packet + kHeaderLength, sizeof(packet) - kHeaderLength);
>+  ExpectPacket(&payload, packet + kHeaderLength,
>+               sizeof(packet) - kHeaderLength);
>   EXPECT_EQ(kVideoFrameDelta, payload.frame_type);
>-  EXPECT_EQ(kRtpVideoVp8, payload.type.Video.codec);
>-  VerifyBasicHeader(&payload.type, 0, 0, 8);
>-  VerifyExtensions(
>-      &payload.type, kNoPictureId, kNoTl0PicIdx, kNoTemporalIdx, kKeyIdx);
>+  EXPECT_EQ(kVideoCodecVP8, payload.video_header().codec);
>+  VerifyBasicHeader(&payload.video_header(), 0, 0, 8);
>+  VerifyExtensions(&payload.video_header(), kNoPictureId, kNoTl0PicIdx,
>+                   kNoTemporalIdx, kKeyIdx);
> }
> 
> TEST_F(RtpDepacketizerVp8Test, MultipleExtensions) {
>@@ -423,12 +405,12 @@ TEST_F(RtpDepacketizerVp8Test, MultipleExtensions) {
>   RtpDepacketizer::ParsedPayload payload;
> 
>   ASSERT_TRUE(depacketizer_->Parse(&payload, packet, sizeof(packet)));
>-  ExpectPacket(
>-      &payload, packet + kHeaderLength, sizeof(packet) - kHeaderLength);
>+  ExpectPacket(&payload, packet + kHeaderLength,
>+               sizeof(packet) - kHeaderLength);
>   EXPECT_EQ(kVideoFrameDelta, payload.frame_type);
>-  EXPECT_EQ(kRtpVideoVp8, payload.type.Video.codec);
>-  VerifyBasicHeader(&payload.type, 0, 0, 8);
>-  VerifyExtensions(&payload.type, (17 << 8) + 17, 42, 1, 17);
>+  EXPECT_EQ(kVideoCodecVP8, payload.video_header().codec);
>+  VerifyBasicHeader(&payload.video_header(), 0, 0, 8);
>+  VerifyExtensions(&payload.video_header(), (17 << 8) + 17, 42, 1, 17);
> }
> 
> TEST_F(RtpDepacketizerVp8Test, TooShortHeader) {
>@@ -465,15 +447,12 @@ TEST_F(RtpDepacketizerVp8Test, TestWithPacketizer) {
>   auto vp8_payload = rtp_payload.subview(kHeaderLength);
>   ExpectPacket(&payload, vp8_payload.data(), vp8_payload.size());
>   EXPECT_EQ(kVideoFrameKey, payload.frame_type);
>-  EXPECT_EQ(kRtpVideoVp8, payload.type.Video.codec);
>-  VerifyBasicHeader(&payload.type, 1, 1, 0);
>-  VerifyExtensions(&payload.type,
>-                   input_header.pictureId,
>-                   input_header.tl0PicIdx,
>-                   input_header.temporalIdx,
>+  EXPECT_EQ(kVideoCodecVP8, payload.video_header().codec);
>+  VerifyBasicHeader(&payload.video_header(), 1, 1, 0);
>+  VerifyExtensions(&payload.video_header(), input_header.pictureId,
>+                   input_header.tl0PicIdx, input_header.temporalIdx,
>                    input_header.keyIdx);
>-  EXPECT_EQ(payload.type.Video.codecHeader.VP8.layerSync,
>-            input_header.layerSync);
>+  EXPECT_EQ(payload.video_header().vp8().layerSync, input_header.layerSync);
> }
> 
> TEST_F(RtpDepacketizerVp8Test, TestEmptyPayload) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_vp9.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_vp9.cc
>index bb3edfced7c..974df8f5a9c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_vp9.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_vp9.cc
>@@ -73,8 +73,7 @@ bool PictureIdPresent(const RTPVideoHeaderVP9& hdr) {
> //                          +-+-+-+-+-+-+-+-+
> //
> size_t LayerInfoLength(const RTPVideoHeaderVP9& hdr) {
>-  if (hdr.temporal_idx == kNoTemporalIdx &&
>-      hdr.spatial_idx == kNoSpatialIdx) {
>+  if (hdr.temporal_idx == kNoTemporalIdx && hdr.spatial_idx == kNoSpatialIdx) {
>     return 0;
>   }
>   return hdr.flexible_mode ? 1 : 2;
>@@ -127,12 +126,12 @@ size_t SsDataLength(const RTPVideoHeaderVP9& hdr) {
>   RTC_DCHECK_GT(hdr.num_spatial_layers, 0U);
>   RTC_DCHECK_LE(hdr.num_spatial_layers, kMaxVp9NumberOfSpatialLayers);
>   RTC_DCHECK_LE(hdr.gof.num_frames_in_gof, kMaxVp9FramesInGof);
>-  size_t length = 1;                           // V
>+  size_t length = 1;  // V
>   if (hdr.spatial_layer_resolution_present) {
>-    length += 4 * hdr.num_spatial_layers;      // Y
>+    length += 4 * hdr.num_spatial_layers;  // Y
>   }
>   if (hdr.gof.num_frames_in_gof > 0) {
>-    ++length;                                  // G
>+    ++length;  // G
>   }
>   // N_G
>   length += hdr.gof.num_frames_in_gof;  // T, U, R
>@@ -194,7 +193,8 @@ bool WriteLayerInfoCommon(const RTPVideoHeaderVP9& vp9,
>   RETURN_FALSE_ON_ERROR(writer->WriteBits(TemporalIdxField(vp9, 0), 3));
>   RETURN_FALSE_ON_ERROR(writer->WriteBits(vp9.temporal_up_switch ? 1 : 0, 1));
>   RETURN_FALSE_ON_ERROR(writer->WriteBits(SpatialIdxField(vp9, 0), 3));
>-  RETURN_FALSE_ON_ERROR(writer->WriteBits(vp9.inter_layer_predicted ? 1: 0, 1));
>+  RETURN_FALSE_ON_ERROR(
>+      writer->WriteBits(vp9.inter_layer_predicted ? 1 : 0, 1));
>   return true;
> }
> 
>@@ -232,8 +232,8 @@ bool WriteLayerInfo(const RTPVideoHeaderVP9& vp9,
> //
> bool WriteRefIndices(const RTPVideoHeaderVP9& vp9,
>                      rtc::BitBufferWriter* writer) {
>-  if (!PictureIdPresent(vp9) ||
>-      vp9.num_ref_pics == 0 || vp9.num_ref_pics > kMaxVp9RefPics) {
>+  if (!PictureIdPresent(vp9) || vp9.num_ref_pics == 0 ||
>+      vp9.num_ref_pics > kMaxVp9RefPics) {
>     return false;
>   }
>   for (uint8_t i = 0; i < vp9.num_ref_pics; ++i) {
>@@ -469,8 +469,7 @@ RtpPacketizerVp9::RtpPacketizerVp9(const RTPVideoHeaderVP9& hdr,
>       payload_size_(0),
>       last_packet_reduction_len_(last_packet_reduction_len) {}
> 
>-RtpPacketizerVp9::~RtpPacketizerVp9() {
>-}
>+RtpPacketizerVp9::~RtpPacketizerVp9() {}
> 
> std::string RtpPacketizerVp9::ToString() {
>   return "RtpPacketizerVp9";
>@@ -576,9 +575,13 @@ bool RtpPacketizerVp9::NextPacket(RtpPacketToSend* packet) {
>   if (!WriteHeaderAndPayload(packet_info, packet, packets_.empty())) {
>     return false;
>   }
>-  packet->SetMarker(packets_.empty() &&
>-                    (hdr_.spatial_idx == kNoSpatialIdx ||
>-                     hdr_.spatial_idx == hdr_.num_spatial_layers - 1));
>+
>+  // Ensure end_of_picture is always set on top spatial layer when it is not
>+  // dropped.
>+  RTC_DCHECK(hdr_.spatial_idx < hdr_.num_spatial_layers - 1 ||
>+             hdr_.end_of_picture);
>+
>+  packet->SetMarker(packets_.empty() && hdr_.end_of_picture);
>   return true;
> }
> 
>@@ -587,7 +590,7 @@ bool RtpPacketizerVp9::NextPacket(RtpPacketToSend* packet) {
> // Payload descriptor for F = 1 (flexible mode)
> //       0 1 2 3 4 5 6 7
> //      +-+-+-+-+-+-+-+-+
>-//      |I|P|L|F|B|E|V|-| (REQUIRED)
>+//      |I|P|L|F|B|E|V|Z| (REQUIRED)
> //      +-+-+-+-+-+-+-+-+
> // I:   |M| PICTURE ID  | (RECOMMENDED)
> //      +-+-+-+-+-+-+-+-+
>@@ -604,7 +607,7 @@ bool RtpPacketizerVp9::NextPacket(RtpPacketToSend* packet) {
> // Payload descriptor for F = 0 (non-flexible mode)
> //       0 1 2 3 4 5 6 7
> //      +-+-+-+-+-+-+-+-+
>-//      |I|P|L|F|B|E|V|-| (REQUIRED)
>+//      |I|P|L|F|B|E|V|Z| (REQUIRED)
> //      +-+-+-+-+-+-+-+-+
> // I:   |M| PICTURE ID  | (RECOMMENDED)
> //      +-+-+-+-+-+-+-+-+
>@@ -630,8 +633,8 @@ bool RtpPacketizerVp9::WriteHeaderAndPayload(const PacketInfo& packet_info,
>     return false;
> 
>   // Copy payload data.
>-  memcpy(&buffer[header_length],
>-         &payload_[packet_info.payload_start_pos], packet_info.size);
>+  memcpy(&buffer[header_length], &payload_[packet_info.payload_start_pos],
>+         packet_info.size);
> 
>   packet->SetPayloadSize(header_length + packet_info.size);
>   return true;
>@@ -648,6 +651,7 @@ bool RtpPacketizerVp9::WriteHeader(const PacketInfo& packet_info,
>   bool b_bit = packet_info.layer_begin;
>   bool e_bit = packet_info.layer_end;
>   bool v_bit = hdr_.ss_data_available && b_bit;
>+  bool z_bit = hdr_.non_ref_for_inter_layer_pred;
> 
>   rtc::BitBufferWriter writer(buffer, max_payload_length_);
>   RETURN_FALSE_ON_ERROR(writer.WriteBits(i_bit ? 1 : 0, 1));
>@@ -657,7 +661,7 @@ bool RtpPacketizerVp9::WriteHeader(const PacketInfo& packet_info,
>   RETURN_FALSE_ON_ERROR(writer.WriteBits(b_bit ? 1 : 0, 1));
>   RETURN_FALSE_ON_ERROR(writer.WriteBits(e_bit ? 1 : 0, 1));
>   RETURN_FALSE_ON_ERROR(writer.WriteBits(v_bit ? 1 : 0, 1));
>-  RETURN_FALSE_ON_ERROR(writer.WriteBits(kReservedBitValue0, 1));
>+  RETURN_FALSE_ON_ERROR(writer.WriteBits(z_bit ? 1 : 0, 1));
> 
>   // Add fields that are present.
>   if (i_bit && !WritePictureId(hdr_, &writer)) {
>@@ -697,7 +701,7 @@ bool RtpDepacketizerVp9::Parse(ParsedPayload* parsed_payload,
> 
>   // Parse mandatory first byte of payload descriptor.
>   rtc::BitBuffer parser(payload, payload_length);
>-  uint32_t i_bit, p_bit, l_bit, f_bit, b_bit, e_bit, v_bit;
>+  uint32_t i_bit, p_bit, l_bit, f_bit, b_bit, e_bit, v_bit, z_bit;
>   RETURN_FALSE_ON_ERROR(parser.ReadBits(&i_bit, 1));
>   RETURN_FALSE_ON_ERROR(parser.ReadBits(&p_bit, 1));
>   RETURN_FALSE_ON_ERROR(parser.ReadBits(&l_bit, 1));
>@@ -705,50 +709,52 @@ bool RtpDepacketizerVp9::Parse(ParsedPayload* parsed_payload,
>   RETURN_FALSE_ON_ERROR(parser.ReadBits(&b_bit, 1));
>   RETURN_FALSE_ON_ERROR(parser.ReadBits(&e_bit, 1));
>   RETURN_FALSE_ON_ERROR(parser.ReadBits(&v_bit, 1));
>-  RETURN_FALSE_ON_ERROR(parser.ConsumeBits(1));
>+  RETURN_FALSE_ON_ERROR(parser.ReadBits(&z_bit, 1));
> 
>   // Parsed payload.
>-  parsed_payload->type.Video.width = 0;
>-  parsed_payload->type.Video.height = 0;
>-  parsed_payload->type.Video.simulcastIdx = 0;
>-  parsed_payload->type.Video.codec = kRtpVideoVp9;
>+  parsed_payload->video_header().width = 0;
>+  parsed_payload->video_header().height = 0;
>+  parsed_payload->video_header().simulcastIdx = 0;
>+  parsed_payload->video_header().codec = kVideoCodecVP9;
> 
>   parsed_payload->frame_type = p_bit ? kVideoFrameDelta : kVideoFrameKey;
> 
>-  RTPVideoHeaderVP9* vp9 = &parsed_payload->type.Video.codecHeader.VP9;
>-  vp9->InitRTPVideoHeaderVP9();
>-  vp9->inter_pic_predicted = p_bit ? true : false;
>-  vp9->flexible_mode = f_bit ? true : false;
>-  vp9->beginning_of_frame = b_bit ? true : false;
>-  vp9->end_of_frame = e_bit ? true : false;
>-  vp9->ss_data_available = v_bit ? true : false;
>+  auto& vp9_header = parsed_payload->video_header()
>+                         .video_type_header.emplace<RTPVideoHeaderVP9>();
>+  vp9_header.InitRTPVideoHeaderVP9();
>+  vp9_header.inter_pic_predicted = p_bit ? true : false;
>+  vp9_header.flexible_mode = f_bit ? true : false;
>+  vp9_header.beginning_of_frame = b_bit ? true : false;
>+  vp9_header.end_of_frame = e_bit ? true : false;
>+  vp9_header.ss_data_available = v_bit ? true : false;
>+  vp9_header.non_ref_for_inter_layer_pred = z_bit ? true : false;
> 
>   // Parse fields that are present.
>-  if (i_bit && !ParsePictureId(&parser, vp9)) {
>+  if (i_bit && !ParsePictureId(&parser, &vp9_header)) {
>     RTC_LOG(LS_ERROR) << "Failed parsing VP9 picture id.";
>     return false;
>   }
>-  if (l_bit && !ParseLayerInfo(&parser, vp9)) {
>+  if (l_bit && !ParseLayerInfo(&parser, &vp9_header)) {
>     RTC_LOG(LS_ERROR) << "Failed parsing VP9 layer info.";
>     return false;
>   }
>-  if (p_bit && f_bit && !ParseRefIndices(&parser, vp9)) {
>+  if (p_bit && f_bit && !ParseRefIndices(&parser, &vp9_header)) {
>     RTC_LOG(LS_ERROR) << "Failed parsing VP9 ref indices.";
>     return false;
>   }
>   if (v_bit) {
>-    if (!ParseSsData(&parser, vp9)) {
>+    if (!ParseSsData(&parser, &vp9_header)) {
>       RTC_LOG(LS_ERROR) << "Failed parsing VP9 SS data.";
>       return false;
>     }
>-    if (vp9->spatial_layer_resolution_present) {
>+    if (vp9_header.spatial_layer_resolution_present) {
>       // TODO(asapersson): Add support for spatial layers.
>-      parsed_payload->type.Video.width = vp9->width[0];
>-      parsed_payload->type.Video.height = vp9->height[0];
>+      parsed_payload->video_header().width = vp9_header.width[0];
>+      parsed_payload->video_header().height = vp9_header.height[0];
>     }
>   }
>-  parsed_payload->type.Video.is_first_packet_in_frame =
>-      b_bit && (!l_bit || !vp9->inter_layer_predicted);
>+  parsed_payload->video_header().is_first_packet_in_frame =
>+      b_bit && (!l_bit || !vp9_header.inter_layer_predicted);
> 
>   uint64_t rem_bits = parser.RemainingBitCount();
>   assert(rem_bits % 8 == 0);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_vp9.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_vp9.h
>index aa86ccdfa43..9052c21d678 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_vp9.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_vp9.h
>@@ -27,7 +27,6 @@
> #include "modules/include/module_common_types.h"
> #include "modules/rtp_rtcp/source/rtp_format.h"
> #include "rtc_base/constructormagic.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -37,7 +36,7 @@ class RtpPacketizerVp9 : public RtpPacketizer {
>                    size_t max_payload_length,
>                    size_t last_packet_reduction_len);
> 
>-  virtual ~RtpPacketizerVp9();
>+  ~RtpPacketizerVp9() override;
> 
>   std::string ToString() override;
> 
>@@ -87,10 +86,9 @@ class RtpPacketizerVp9 : public RtpPacketizer {
>   RTC_DISALLOW_COPY_AND_ASSIGN(RtpPacketizerVp9);
> };
> 
>-
> class RtpDepacketizerVp9 : public RtpDepacketizer {
>  public:
>-  virtual ~RtpDepacketizerVp9() {}
>+  ~RtpDepacketizerVp9() override = default;
> 
>   bool Parse(ParsedPayload* parsed_payload,
>              const uint8_t* payload,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_vp9_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_vp9_unittest.cc
>index c522c4dc8a9..66c6091938e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_vp9_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_format_vp9_unittest.cc
>@@ -15,7 +15,6 @@
> #include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
> #include "test/gmock.h"
> #include "test/gtest.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> namespace {
>@@ -27,6 +26,8 @@ void VerifyHeader(const RTPVideoHeaderVP9& expected,
>   EXPECT_EQ(expected.beginning_of_frame, actual.beginning_of_frame);
>   EXPECT_EQ(expected.end_of_frame, actual.end_of_frame);
>   EXPECT_EQ(expected.ss_data_available, actual.ss_data_available);
>+  EXPECT_EQ(expected.non_ref_for_inter_layer_pred,
>+            actual.non_ref_for_inter_layer_pred);
>   EXPECT_EQ(expected.picture_id, actual.picture_id);
>   EXPECT_EQ(expected.max_picture_id, actual.max_picture_id);
>   EXPECT_EQ(expected.temporal_idx, actual.temporal_idx);
>@@ -80,8 +81,10 @@ void ParseAndCheckPacket(const uint8_t* packet,
>   std::unique_ptr<RtpDepacketizer> depacketizer(new RtpDepacketizerVp9());
>   RtpDepacketizer::ParsedPayload parsed;
>   ASSERT_TRUE(depacketizer->Parse(&parsed, packet, expected_length));
>-  EXPECT_EQ(kRtpVideoVp9, parsed.type.Video.codec);
>-  VerifyHeader(expected, parsed.type.Video.codecHeader.VP9);
>+  EXPECT_EQ(kVideoCodecVP9, parsed.video_header().codec);
>+  auto& vp9_header =
>+      absl::get<RTPVideoHeaderVP9>(parsed.video_header().video_type_header);
>+  VerifyHeader(expected, vp9_header);
>   const size_t kExpectedPayloadLength = expected_length - expected_hdr_length;
>   VerifyPayload(parsed, packet + expected_hdr_length, kExpectedPayloadLength);
> }
>@@ -90,7 +93,7 @@ void ParseAndCheckPacket(const uint8_t* packet,
> // Payload descriptor for flexible mode
> //        0 1 2 3 4 5 6 7
> //        +-+-+-+-+-+-+-+-+
>-//        |I|P|L|F|B|E|V|-| (REQUIRED)
>+//        |I|P|L|F|B|E|V|Z| (REQUIRED)
> //        +-+-+-+-+-+-+-+-+
> //   I:   |M| PICTURE ID  | (RECOMMENDED)
> //        +-+-+-+-+-+-+-+-+
>@@ -107,7 +110,7 @@ void ParseAndCheckPacket(const uint8_t* packet,
> // Payload descriptor for non-flexible mode
> //        0 1 2 3 4 5 6 7
> //        +-+-+-+-+-+-+-+-+
>-//        |I|P|L|F|B|E|V|-| (REQUIRED)
>+//        |I|P|L|F|B|E|V|Z| (REQUIRED)
> //        +-+-+-+-+-+-+-+-+
> //   I:   |M| PICTURE ID  | (RECOMMENDED)
> //        +-+-+-+-+-+-+-+-+
>@@ -127,9 +130,7 @@ class RtpPacketizerVp9Test : public ::testing::Test {
>   static constexpr size_t kMaxPacketSize = 1200;
> 
>   RtpPacketizerVp9Test() : packet_(kNoExtensions, kMaxPacketSize) {}
>-  virtual void SetUp() {
>-    expected_.InitRTPVideoHeaderVP9();
>-  }
>+  void SetUp() override { expected_.InitRTPVideoHeaderVP9(); }
> 
>   RtpPacketToSend packet_;
>   std::unique_ptr<uint8_t[]> payload_;
>@@ -191,7 +192,7 @@ TEST_F(RtpPacketizerVp9Test, TestEqualSizedMode_OnePacket) {
>   Init(kFrameSize, kPacketSize);
> 
>   // One packet:
>-  // I:0, P:0, L:0, F:0, B:1, E:1, V:0  (1hdr + 25 payload)
>+  // I:0, P:0, L:0, F:0, B:1, E:1, V:0, Z:0  (1hdr + 25 payload)
>   const size_t kExpectedHdrSizes[] = {1};
>   const size_t kExpectedSizes[] = {26};
>   const size_t kExpectedNum = GTEST_ARRAY_SIZE_(kExpectedSizes);
>@@ -204,8 +205,8 @@ TEST_F(RtpPacketizerVp9Test, TestEqualSizedMode_TwoPackets) {
>   Init(kFrameSize, kPacketSize);
> 
>   // Two packets:
>-  // I:0, P:0, L:0, F:0, B:1, E:0, V:0  (1hdr + 14 payload)
>-  // I:0, P:0, L:0, F:0, B:0, E:1, V:0  (1hdr + 13 payload)
>+  // I:0, P:0, L:0, F:0, B:1, E:0, V:0, Z:0  (1hdr + 14 payload)
>+  // I:0, P:0, L:0, F:0, B:0, E:1, V:0, Z:0  (1hdr + 13 payload)
>   const size_t kExpectedHdrSizes[] = {1, 1};
>   const size_t kExpectedSizes[] = {14, 15};
>   const size_t kExpectedNum = GTEST_ARRAY_SIZE_(kExpectedSizes);
>@@ -225,14 +226,14 @@ TEST_F(RtpPacketizerVp9Test, TestOneBytePictureId) {
>   const size_t kFrameSize = 30;
>   const size_t kPacketSize = 12;
> 
>-  expected_.picture_id = kMaxOneBytePictureId;   // 2 byte payload descriptor
>+  expected_.picture_id = kMaxOneBytePictureId;  // 2 byte payload descriptor
>   expected_.max_picture_id = kMaxOneBytePictureId;
>   Init(kFrameSize, kPacketSize);
> 
>   // Three packets:
>-  // I:1, P:0, L:0, F:0, B:1, E:0, V:0 (2hdr + 10 payload)
>-  // I:1, P:0, L:0, F:0, B:0, E:0, V:0 (2hdr + 10 payload)
>-  // I:1, P:0, L:0, F:0, B:0, E:1, V:0 (2hdr + 10 payload)
>+  // I:1, P:0, L:0, F:0, B:1, E:0, V:0, Z:0 (2hdr + 10 payload)
>+  // I:1, P:0, L:0, F:0, B:0, E:0, V:0, Z:0 (2hdr + 10 payload)
>+  // I:1, P:0, L:0, F:0, B:0, E:1, V:0, Z:0 (2hdr + 10 payload)
>   const size_t kExpectedHdrSizes[] = {2, 2, 2};
>   const size_t kExpectedSizes[] = {12, 12, 12};
>   const size_t kExpectedNum = GTEST_ARRAY_SIZE_(kExpectedSizes);
>@@ -247,10 +248,10 @@ TEST_F(RtpPacketizerVp9Test, TestTwoBytePictureId) {
>   Init(kFrameSize, kPacketSize);
> 
>   // Four packets:
>-  // I:1, P:0, L:0, F:0, B:1, E:0, V:0 (3hdr + 8 payload)
>-  // I:1, P:0, L:0, F:0, B:0, E:0, V:0 (3hdr + 8 payload)
>-  // I:1, P:0, L:0, F:0, B:0, E:0, V:0 (3hdr + 8 payload)
>-  // I:1, P:0, L:0, F:0, B:0, E:1, V:0 (3hdr + 7 payload)
>+  // I:1, P:0, L:0, F:0, B:1, E:0, V:0, Z:0 (3hdr + 8 payload)
>+  // I:1, P:0, L:0, F:0, B:0, E:0, V:0, Z:0 (3hdr + 8 payload)
>+  // I:1, P:0, L:0, F:0, B:0, E:0, V:0, Z:0 (3hdr + 8 payload)
>+  // I:1, P:0, L:0, F:0, B:0, E:1, V:0, Z:0 (3hdr + 7 payload)
>   const size_t kExpectedHdrSizes[] = {3, 3, 3, 3};
>   const size_t kExpectedSizes[] = {10, 11, 11, 11};
>   const size_t kExpectedNum = GTEST_ARRAY_SIZE_(kExpectedSizes);
>@@ -270,9 +271,9 @@ TEST_F(RtpPacketizerVp9Test, TestLayerInfoWithNonFlexibleMode) {
>   Init(kFrameSize, kPacketSize);
> 
>   // Two packets:
>-  //    | I:0, P:0, L:1, F:0, B:1, E:0, V:0 | (3hdr + 15 payload)
>+  //    | I:0, P:0, L:1, F:0, B:1, E:0, V:0 Z:0 | (3hdr + 15 payload)
>   // L: | T:3, U:1, S:2, D:1 | TL0PICIDX:117 |
>-  //    | I:0, P:0, L:1, F:0, B:0, E:1, V:0 | (3hdr + 15 payload)
>+  //    | I:0, P:0, L:1, F:0, B:0, E:1, V:0 Z:0 | (3hdr + 15 payload)
>   // L: | T:3, U:1, S:2, D:1 | TL0PICIDX:117 |
>   const size_t kExpectedHdrSizes[] = {3, 3};
>   const size_t kExpectedSizes[] = {18, 18};
>@@ -320,7 +321,7 @@ TEST_F(RtpPacketizerVp9Test, TestRefIdx) {
>   Init(kFrameSize, kPacketSize);
> 
>   // Two packets:
>-  // I:1, P:1, L:0, F:1, B:1, E:1, V:0 (5hdr + 16 payload)
>+  // I:1, P:1, L:0, F:1, B:1, E:1, V:0, Z:0 (5hdr + 16 payload)
>   // I:   2
>   // P,F: P_DIFF:1,   N:1
>   //      P_DIFF:3,   N:1
>@@ -360,7 +361,7 @@ TEST_F(RtpPacketizerVp9Test, TestSsDataWithoutSpatialResolutionPresent) {
>   Init(kFrameSize, kPacketSize);
> 
>   // One packet:
>-  // I:0, P:0, L:0, F:0, B:1, E:1, V:1 (5hdr + 21 payload)
>+  // I:0, P:0, L:0, F:0, B:1, E:1, V:1, Z:0 (5hdr + 21 payload)
>   // N_S:0, Y:0, G:1
>   // N_G:1
>   // T:0, U:1, R:1 | P_DIFF[0][0]:4
>@@ -381,7 +382,7 @@ TEST_F(RtpPacketizerVp9Test, TestSsDataWithoutGbitPresent) {
>   Init(kFrameSize, kPacketSize);
> 
>   // One packet:
>-  // I:0, P:0, L:0, F:0, B:1, E:1, V:1 (2hdr + 21 payload)
>+  // I:0, P:0, L:0, F:0, B:1, E:1, V:1, Z:0 (2hdr + 21 payload)
>   // N_S:0, Y:0, G:0
>   const size_t kExpectedHdrSizes[] = {2};
>   const size_t kExpectedSizes[] = {23};
>@@ -418,7 +419,7 @@ TEST_F(RtpPacketizerVp9Test, TestSsData) {
>   Init(kFrameSize, kPacketSize);
> 
>   // One packet:
>-  // I:0, P:0, L:0, F:0, B:1, E:1, V:1 (19hdr + 21 payload)
>+  // I:0, P:0, L:0, F:0, B:1, E:1, V:1, Z:0 (19hdr + 21 payload)
>   // N_S:1, Y:1, G:1
>   // WIDTH:640   // 2 bytes
>   // HEIGHT:360  // 2 bytes
>@@ -463,7 +464,7 @@ TEST_F(RtpPacketizerVp9Test, TestSsDataDoesNotFitInAveragePacket) {
>   Init(kFrameSize, kPacketSize);
> 
>   // Three packets:
>-  // I:0, P:0, L:0, F:0, B:1, E:1, V:1 (19hdr + 1 payload)
>+  // I:0, P:0, L:0, F:0, B:1, E:1, V:1, Z:0 (19hdr + 1 payload)
>   // N_S:1, Y:1, G:1
>   // WIDTH:640   // 2 bytes
>   // HEIGHT:360  // 2 bytes
>@@ -480,7 +481,7 @@ TEST_F(RtpPacketizerVp9Test, TestSsDataDoesNotFitInAveragePacket) {
>   CreateParseAndCheckPackets(kExpectedHdrSizes, kExpectedSizes, kExpectedNum);
> }
> 
>-TEST_F(RtpPacketizerVp9Test, TestOnlyHighestSpatialLayerSetMarker) {
>+TEST_F(RtpPacketizerVp9Test, EndOfPictureSetsSetMarker) {
>   const size_t kFrameSize = 10;
>   const size_t kPacketSize = 8;
>   const size_t kLastPacketReductionLen = 0;
>@@ -494,32 +495,21 @@ TEST_F(RtpPacketizerVp9Test, TestOnlyHighestSpatialLayerSetMarker) {
> 
>   RtpPacketToSend packet(kNoExtensions);
> 
>-  vp9_header.spatial_idx = 0;
>-  RtpPacketizerVp9 packetizer0(vp9_header, kPacketSize,
>-                               kLastPacketReductionLen);
>-  packetizer0.SetPayloadData(kFrame, sizeof(kFrame), kNoFragmentation);
>-  ASSERT_TRUE(packetizer0.NextPacket(&packet));
>-  EXPECT_FALSE(packet.Marker());
>-  ASSERT_TRUE(packetizer0.NextPacket(&packet));
>-  EXPECT_FALSE(packet.Marker());
>-
>-  vp9_header.spatial_idx = 1;
>-  RtpPacketizerVp9 packetizer1(vp9_header, kPacketSize,
>-                               kLastPacketReductionLen);
>-  packetizer1.SetPayloadData(kFrame, sizeof(kFrame), kNoFragmentation);
>-  ASSERT_TRUE(packetizer1.NextPacket(&packet));
>-  EXPECT_FALSE(packet.Marker());
>-  ASSERT_TRUE(packetizer1.NextPacket(&packet));
>-  EXPECT_FALSE(packet.Marker());
>-
>-  vp9_header.spatial_idx = 2;
>-  RtpPacketizerVp9 packetizer2(vp9_header, kPacketSize,
>-                               kLastPacketReductionLen);
>-  packetizer2.SetPayloadData(kFrame, sizeof(kFrame), kNoFragmentation);
>-  ASSERT_TRUE(packetizer2.NextPacket(&packet));
>-  EXPECT_FALSE(packet.Marker());
>-  ASSERT_TRUE(packetizer2.NextPacket(&packet));
>-  EXPECT_TRUE(packet.Marker());
>+  // Drop top layer and ensure that marker bit is set on last encoded layer.
>+  for (size_t spatial_idx = 0; spatial_idx < vp9_header.num_spatial_layers - 1;
>+       ++spatial_idx) {
>+    const bool end_of_picture =
>+        spatial_idx + 1 == vp9_header.num_spatial_layers - 1;
>+    vp9_header.spatial_idx = spatial_idx;
>+    vp9_header.end_of_picture = end_of_picture;
>+    RtpPacketizerVp9 packetizer(vp9_header, kPacketSize,
>+                                kLastPacketReductionLen);
>+    packetizer.SetPayloadData(kFrame, sizeof(kFrame), kNoFragmentation);
>+    ASSERT_TRUE(packetizer.NextPacket(&packet));
>+    EXPECT_FALSE(packet.Marker());
>+    ASSERT_TRUE(packetizer.NextPacket(&packet));
>+    EXPECT_EQ(packet.Marker(), end_of_picture);
>+  }
> }
> 
> TEST_F(RtpPacketizerVp9Test, TestGeneratesMinimumNumberOfPackets) {
>@@ -579,14 +569,25 @@ TEST_F(RtpPacketizerVp9Test, TestRespectsLastPacketReductionLen) {
>   EXPECT_TRUE(packet.Marker());
> }
> 
>+TEST_F(RtpPacketizerVp9Test, TestNonRefForInterLayerPred) {
>+  const size_t kFrameSize = 25;
>+  const size_t kPacketSize = 26;
>+
>+  expected_.non_ref_for_inter_layer_pred = true;
>+  Init(kFrameSize, kPacketSize);
>+
>+  // I:0, P:0, L:0, F:0, B:1, E:1, V:0, Z:1  (1hdr + 25 payload)
>+  const size_t kExpectedHdrSizes[] = {1};
>+  const size_t kExpectedSizes[] = {26};
>+  const size_t kExpectedNum = GTEST_ARRAY_SIZE_(kExpectedSizes);
>+  CreateParseAndCheckPackets(kExpectedHdrSizes, kExpectedSizes, kExpectedNum);
>+}
>+
> class RtpDepacketizerVp9Test : public ::testing::Test {
>  protected:
>-  RtpDepacketizerVp9Test()
>-      : depacketizer_(new RtpDepacketizerVp9()) {}
>+  RtpDepacketizerVp9Test() : depacketizer_(new RtpDepacketizerVp9()) {}
> 
>-  virtual void SetUp() {
>-    expected_.InitRTPVideoHeaderVP9();
>-  }
>+  void SetUp() override { expected_.InitRTPVideoHeaderVP9(); }
> 
>   RTPVideoHeaderVP9 expected_;
>   std::unique_ptr<RtpDepacketizer> depacketizer_;
>@@ -595,7 +596,7 @@ class RtpDepacketizerVp9Test : public ::testing::Test {
> TEST_F(RtpDepacketizerVp9Test, ParseBasicHeader) {
>   const uint8_t kHeaderLength = 1;
>   uint8_t packet[4] = {0};
>-  packet[0] = 0x0C;  // I:0 P:0 L:0 F:0 B:1 E:1 V:0 R:0
>+  packet[0] = 0x0C;  // I:0 P:0 L:0 F:0 B:1 E:1 V:0 Z:0
>   expected_.beginning_of_frame = true;
>   expected_.end_of_frame = true;
>   ParseAndCheckPacket(packet, expected_, kHeaderLength, sizeof(packet));
>@@ -604,7 +605,7 @@ TEST_F(RtpDepacketizerVp9Test, ParseBasicHeader) {
> TEST_F(RtpDepacketizerVp9Test, ParseOneBytePictureId) {
>   const uint8_t kHeaderLength = 2;
>   uint8_t packet[10] = {0};
>-  packet[0] = 0x80;  // I:1 P:0 L:0 F:0 B:0 E:0 V:0 R:0
>+  packet[0] = 0x80;  // I:1 P:0 L:0 F:0 B:0 E:0 V:0 Z:0
>   packet[1] = kMaxOneBytePictureId;
> 
>   expected_.picture_id = kMaxOneBytePictureId;
>@@ -615,7 +616,7 @@ TEST_F(RtpDepacketizerVp9Test, ParseOneBytePictureId) {
> TEST_F(RtpDepacketizerVp9Test, ParseTwoBytePictureId) {
>   const uint8_t kHeaderLength = 3;
>   uint8_t packet[10] = {0};
>-  packet[0] = 0x80;  // I:1 P:0 L:0 F:0 B:0 E:0 V:0 R:0
>+  packet[0] = 0x80;  // I:1 P:0 L:0 F:0 B:0 E:0 V:0 Z:0
>   packet[1] = 0x80 | ((kMaxTwoBytePictureId >> 8) & 0x7F);
>   packet[2] = kMaxTwoBytePictureId & 0xFF;
> 
>@@ -632,7 +633,7 @@ TEST_F(RtpDepacketizerVp9Test, ParseLayerInfoWithNonFlexibleMode) {
>   const uint8_t kDbit = 1;
>   const uint8_t kTl0PicIdx = 17;
>   uint8_t packet[13] = {0};
>-  packet[0] = 0x20;  // I:0 P:0 L:1 F:0 B:0 E:0 V:0 R:0
>+  packet[0] = 0x20;  // I:0 P:0 L:1 F:0 B:0 E:0 V:0 Z:0
>   packet[1] = (kTemporalIdx << 5) | (kUbit << 4) | (kSpatialIdx << 1) | kDbit;
>   packet[2] = kTl0PicIdx;
> 
>@@ -653,10 +654,10 @@ TEST_F(RtpDepacketizerVp9Test, ParseLayerInfoWithFlexibleMode) {
>   const uint8_t kSpatialIdx = 0;
>   const uint8_t kDbit = 0;
>   uint8_t packet[13] = {0};
>-  packet[0] = 0x38;  // I:0 P:0 L:1 F:1 B:1 E:0 V:0 R:0
>+  packet[0] = 0x38;  // I:0 P:0 L:1 F:1 B:1 E:0 V:0 Z:0
>   packet[1] = (kTemporalIdx << 5) | (kUbit << 4) | (kSpatialIdx << 1) | kDbit;
> 
>-  // I:0 P:0 L:1 F:1 B:1 E:0 V:0
>+  // I:0 P:0 L:1 F:1 B:1 E:0 V:0 Z:0
>   // L:   T:2 U:1 S:0 D:0
>   expected_.beginning_of_frame = true;
>   expected_.flexible_mode = true;
>@@ -674,14 +675,14 @@ TEST_F(RtpDepacketizerVp9Test, ParseRefIdx) {
>   const uint8_t kPdiff2 = 18;
>   const uint8_t kPdiff3 = 127;
>   uint8_t packet[13] = {0};
>-  packet[0] = 0xD8;  // I:1 P:1 L:0 F:1 B:1 E:0 V:0 R:0
>+  packet[0] = 0xD8;  // I:1 P:1 L:0 F:1 B:1 E:0 V:0 Z:0
>   packet[1] = 0x80 | ((kPictureId >> 8) & 0x7F);  // Two byte pictureID.
>   packet[2] = kPictureId;
>   packet[3] = (kPdiff1 << 1) | 1;  // P_DIFF N:1
>   packet[4] = (kPdiff2 << 1) | 1;  // P_DIFF N:1
>   packet[5] = (kPdiff3 << 1) | 0;  // P_DIFF N:0
> 
>-  // I:1 P:1 L:0 F:1 B:1 E:0 V:0
>+  // I:1 P:1 L:0 F:1 B:1 E:0 V:0 Z:0
>   // I:    PICTURE ID:17
>   // I:
>   // P,F:  P_DIFF:17  N:1 => refPicId = 17 - 17 = 0
>@@ -704,8 +705,8 @@ TEST_F(RtpDepacketizerVp9Test, ParseRefIdx) {
> TEST_F(RtpDepacketizerVp9Test, ParseRefIdxFailsWithNoPictureId) {
>   const uint8_t kPdiff = 3;
>   uint8_t packet[13] = {0};
>-  packet[0] = 0x58;            // I:0 P:1 L:0 F:1 B:1 E:0 V:0 R:0
>-  packet[1] = (kPdiff << 1);   // P,F:  P_DIFF:3 N:0
>+  packet[0] = 0x58;           // I:0 P:1 L:0 F:1 B:1 E:0 V:0 Z:0
>+  packet[1] = (kPdiff << 1);  // P,F:  P_DIFF:3 N:0
> 
>   RtpDepacketizer::ParsedPayload parsed;
>   EXPECT_FALSE(depacketizer_->Parse(&parsed, packet, sizeof(packet)));
>@@ -714,7 +715,7 @@ TEST_F(RtpDepacketizerVp9Test, ParseRefIdxFailsWithNoPictureId) {
> TEST_F(RtpDepacketizerVp9Test, ParseRefIdxFailsWithTooManyRefPics) {
>   const uint8_t kPdiff = 3;
>   uint8_t packet[13] = {0};
>-  packet[0] = 0xD8;                  // I:1 P:1 L:0 F:1 B:1 E:0 V:0 R:0
>+  packet[0] = 0xD8;                  // I:1 P:1 L:0 F:1 B:1 E:0 V:0 Z:0
>   packet[1] = kMaxOneBytePictureId;  // I:    PICTURE ID:127
>   packet[2] = (kPdiff << 1) | 1;     // P,F:  P_DIFF:3 N:1
>   packet[3] = (kPdiff << 1) | 1;     // P,F:  P_DIFF:3 N:1
>@@ -731,7 +732,7 @@ TEST_F(RtpDepacketizerVp9Test, ParseSsData) {
>   const size_t kNs = 2;
>   const size_t kNg = 2;
>   uint8_t packet[23] = {0};
>-  packet[0] = 0x0A;  // I:0 P:0 L:0 F:0 B:1 E:0 V:1 R:0
>+  packet[0] = 0x0A;  // I:0 P:0 L:0 F:0 B:1 E:0 V:1 Z:0
>   packet[1] = ((kNs - 1) << 5) | (kYbit << 4) | (1 << 3);  // N_S Y G:1 -
>   packet[2] = kNg;                                         // N_G
>   packet[3] = (0 << 5) | (1 << 4) | (0 << 2) | 0;          // T:0 U:1 R:0 -
>@@ -755,29 +756,29 @@ TEST_F(RtpDepacketizerVp9Test, ParseSsData) {
> 
> TEST_F(RtpDepacketizerVp9Test, ParseFirstPacketInKeyFrame) {
>   uint8_t packet[2] = {0};
>-  packet[0] = 0x08;  // I:0 P:0 L:0 F:0 B:1 E:0 V:0 R:0
>+  packet[0] = 0x08;  // I:0 P:0 L:0 F:0 B:1 E:0 V:0 Z:0
> 
>   RtpDepacketizer::ParsedPayload parsed;
>   ASSERT_TRUE(depacketizer_->Parse(&parsed, packet, sizeof(packet)));
>   EXPECT_EQ(kVideoFrameKey, parsed.frame_type);
>-  EXPECT_TRUE(parsed.type.Video.is_first_packet_in_frame);
>+  EXPECT_TRUE(parsed.video_header().is_first_packet_in_frame);
> }
> 
> TEST_F(RtpDepacketizerVp9Test, ParseLastPacketInDeltaFrame) {
>   uint8_t packet[2] = {0};
>-  packet[0] = 0x44;  // I:0 P:1 L:0 F:0 B:0 E:1 V:0 R:0
>+  packet[0] = 0x44;  // I:0 P:1 L:0 F:0 B:0 E:1 V:0 Z:0
> 
>   RtpDepacketizer::ParsedPayload parsed;
>   ASSERT_TRUE(depacketizer_->Parse(&parsed, packet, sizeof(packet)));
>   EXPECT_EQ(kVideoFrameDelta, parsed.frame_type);
>-  EXPECT_FALSE(parsed.type.Video.is_first_packet_in_frame);
>+  EXPECT_FALSE(parsed.video_header().is_first_packet_in_frame);
> }
> 
> TEST_F(RtpDepacketizerVp9Test, ParseResolution) {
>   const uint16_t kWidth[2] = {640, 1280};
>   const uint16_t kHeight[2] = {360, 720};
>   uint8_t packet[20] = {0};
>-  packet[0] = 0x0A;  // I:0 P:0 L:0 F:0 B:1 E:0 V:1 R:0
>+  packet[0] = 0x0A;                     // I:0 P:0 L:0 F:0 B:1 E:0 V:1 Z:0
>   packet[1] = (1 << 5) | (1 << 4) | 0;  // N_S:1 Y:1 G:0
>   packet[2] = kWidth[0] >> 8;
>   packet[3] = kWidth[0] & 0xFF;
>@@ -790,8 +791,8 @@ TEST_F(RtpDepacketizerVp9Test, ParseResolution) {
> 
>   RtpDepacketizer::ParsedPayload parsed;
>   ASSERT_TRUE(depacketizer_->Parse(&parsed, packet, sizeof(packet)));
>-  EXPECT_EQ(kWidth[0], parsed.type.Video.width);
>-  EXPECT_EQ(kHeight[0], parsed.type.Video.height);
>+  EXPECT_EQ(kWidth[0], parsed.video_header().width);
>+  EXPECT_EQ(kHeight[0], parsed.video_header().height);
> }
> 
> TEST_F(RtpDepacketizerVp9Test, ParseFailsForNoPayloadLength) {
>@@ -807,4 +808,19 @@ TEST_F(RtpDepacketizerVp9Test, ParseFailsForTooShortBufferToFitPayload) {
>   EXPECT_FALSE(depacketizer_->Parse(&parsed, packet, sizeof(packet)));
> }
> 
>+TEST_F(RtpDepacketizerVp9Test, ParseNonRefForInterLayerPred) {
>+  uint8_t packet[2] = {0};
>+
>+  packet[0] = 0x08;  // I:0 P:0 L:0 F:0 B:1 E:0 V:0 Z:0
>+  expected_.beginning_of_frame = true;
>+  expected_.non_ref_for_inter_layer_pred = false;
>+  ParseAndCheckPacket(packet, expected_, 1, sizeof(packet));
>+
>+  packet[0] = 0x05;  // I:0 P:0 L:0 F:0 B:0 E:1 V:0 Z:1
>+  expected_.beginning_of_frame = false;
>+  expected_.end_of_frame = true;
>+  expected_.non_ref_for_inter_layer_pred = true;
>+  ParseAndCheckPacket(packet, expected_, 1, sizeof(packet));
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor.cc
>new file mode 100644
>index 00000000000..080cac7a8d9
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor.cc
>@@ -0,0 +1,72 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h"
>+
>+#include "rtc_base/checks.h"
>+
>+namespace webrtc {
>+
>+constexpr size_t RtpGenericFrameDescriptor::kMaxNumFrameDependencies;
>+
>+RtpGenericFrameDescriptor::RtpGenericFrameDescriptor() = default;
>+
>+int RtpGenericFrameDescriptor::TemporalLayer() const {
>+  RTC_DCHECK(FirstPacketInSubFrame());
>+  return temporal_layer_;
>+}
>+
>+void RtpGenericFrameDescriptor::SetTemporalLayer(int temporal_layer) {
>+  RTC_DCHECK_GE(temporal_layer, 0);
>+  RTC_DCHECK_LE(temporal_layer, 7);
>+  temporal_layer_ = temporal_layer;
>+}
>+
>+uint8_t RtpGenericFrameDescriptor::SpatialLayersBitmask() const {
>+  RTC_DCHECK(FirstPacketInSubFrame());
>+  return spatial_layers_;
>+}
>+
>+void RtpGenericFrameDescriptor::SetSpatialLayersBitmask(
>+    uint8_t spatial_layers) {
>+  RTC_DCHECK(FirstPacketInSubFrame());
>+  spatial_layers_ = spatial_layers;
>+}
>+
>+uint16_t RtpGenericFrameDescriptor::FrameId() const {
>+  RTC_DCHECK(FirstPacketInSubFrame());
>+  return frame_id_;
>+}
>+
>+void RtpGenericFrameDescriptor::SetFrameId(uint16_t frame_id) {
>+  RTC_DCHECK(FirstPacketInSubFrame());
>+  frame_id_ = frame_id;
>+}
>+
>+rtc::ArrayView<const uint16_t>
>+RtpGenericFrameDescriptor::FrameDependenciesDiffs() const {
>+  RTC_DCHECK(FirstPacketInSubFrame());
>+  return rtc::MakeArrayView(frame_deps_id_diffs_, num_frame_deps_);
>+}
>+
>+bool RtpGenericFrameDescriptor::AddFrameDependencyDiff(uint16_t fdiff) {
>+  RTC_DCHECK(FirstPacketInSubFrame());
>+  if (num_frame_deps_ == kMaxNumFrameDependencies)
>+    return false;
>+  if (fdiff == 0)
>+    return false;
>+  RTC_DCHECK_LT(fdiff, 1 << 14);
>+  RTC_DCHECK_GT(fdiff, 0);
>+  frame_deps_id_diffs_[num_frame_deps_] = fdiff;
>+  num_frame_deps_++;
>+  return true;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h
>new file mode 100644
>index 00000000000..51a9ac00220
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h
>@@ -0,0 +1,70 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#ifndef MODULES_RTP_RTCP_SOURCE_RTP_GENERIC_FRAME_DESCRIPTOR_H_
>+#define MODULES_RTP_RTCP_SOURCE_RTP_GENERIC_FRAME_DESCRIPTOR_H_
>+
>+#include <stddef.h>
>+#include <stdint.h>
>+
>+#include "api/array_view.h"
>+
>+namespace webrtc {
>+
>+// Data to put on the wire for FrameDescriptor rtp header extension.
>+class RtpGenericFrameDescriptor {
>+ public:
>+  RtpGenericFrameDescriptor();
>+
>+  bool FirstPacketInSubFrame() const { return beginning_of_subframe_; }
>+  void SetFirstPacketInSubFrame(bool first) { beginning_of_subframe_ = first; }
>+  bool LastPacketInSubFrame() const { return end_of_subframe_; }
>+  void SetLastPacketInSubFrame(bool last) { end_of_subframe_ = last; }
>+
>+  bool FirstSubFrameInFrame() const { return beginning_of_frame_; }
>+  void SetFirstSubFrameInFrame(bool first) { beginning_of_frame_ = first; }
>+  bool LastSubFrameInFrame() const { return end_of_frame_; }
>+  void SetLastSubFrameInFrame(bool last) { end_of_frame_ = last; }
>+
>+  // Properties below undefined if !FirstPacketInSubFrame()
>+  // Valid range for temporal layer: [0, 7]
>+  int TemporalLayer() const;
>+  void SetTemporalLayer(int temporal_layer);
>+
>+  // Frame might by used, possible indrectly, for spatial layer sid iff
>+  // (bitmask & (1 << sid)) != 0
>+  uint8_t SpatialLayersBitmask() const;
>+  void SetSpatialLayersBitmask(uint8_t spatial_layers);
>+
>+  uint16_t FrameId() const;
>+  void SetFrameId(uint16_t frame_id);
>+
>+  rtc::ArrayView<const uint16_t> FrameDependenciesDiffs() const;
>+  void ClearFrameDependencies() { num_frame_deps_ = 0; }
>+  // Returns false on failure, i.e. number of dependencies is too large.
>+  bool AddFrameDependencyDiff(uint16_t fdiff);
>+
>+ private:
>+  static constexpr size_t kMaxNumFrameDependencies = 8;
>+
>+  bool beginning_of_subframe_ = false;
>+  bool end_of_subframe_ = false;
>+  bool beginning_of_frame_ = false;
>+  bool end_of_frame_ = false;
>+
>+  uint16_t frame_id_ = 0;
>+  uint8_t spatial_layers_ = 1;
>+  uint8_t temporal_layer_ = 0;
>+  size_t num_frame_deps_ = 0;
>+  uint16_t frame_deps_id_diffs_[kMaxNumFrameDependencies];
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_RTP_RTCP_SOURCE_RTP_GENERIC_FRAME_DESCRIPTOR_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.cc
>new file mode 100644
>index 00000000000..c7b52d5ba68
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.cc
>@@ -0,0 +1,145 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.h"
>+
>+#include "rtc_base/checks.h"
>+
>+namespace webrtc {
>+namespace {
>+
>+constexpr uint8_t kFlagBeginOfSubframe = 0x80;
>+constexpr uint8_t kFlagEndOfSubframe = 0x40;
>+constexpr uint8_t kFlagFirstSubframe = 0x20;
>+constexpr uint8_t kFlagLastSubframe = 0x10;
>+constexpr uint8_t kFlagDependencies = 0x08;
>+constexpr uint8_t kMaskTemporalLayer = 0x07;
>+
>+constexpr uint8_t kFlagMoreDependencies = 0x01;
>+constexpr uint8_t kFlageXtendedOffset = 0x02;
>+
>+}  // namespace
>+//       0 1 2 3 4 5 6 7
>+//      +-+-+-+-+-+-+-+-+
>+//      |B|E|F|L|D|  T  |
>+//      +-+-+-+-+-+-+-+-+
>+// B:   |       S       |
>+//      +-+-+-+-+-+-+-+-+
>+//      |               |
>+// B:   +      FID      +
>+//      |               |
>+//      +-+-+-+-+-+-+-+-+
>+// D:   |    FDIFF  |X|M|
>+//      +---------------+
>+// X:   |      ...      |
>+//      +-+-+-+-+-+-+-+-+
>+// M:   |    FDIFF  |X|M|
>+//      +---------------+
>+//      |      ...      |
>+//      +-+-+-+-+-+-+-+-+
>+constexpr RTPExtensionType RtpGenericFrameDescriptorExtension::kId;
>+constexpr char RtpGenericFrameDescriptorExtension::kUri[];
>+
>+bool RtpGenericFrameDescriptorExtension::Parse(
>+    rtc::ArrayView<const uint8_t> data,
>+    RtpGenericFrameDescriptor* descriptor) {
>+  if (data.empty()) {
>+    return false;
>+  }
>+
>+  bool begins_subframe = (data[0] & kFlagBeginOfSubframe) != 0;
>+  descriptor->SetFirstPacketInSubFrame(begins_subframe);
>+  descriptor->SetLastPacketInSubFrame((data[0] & kFlagEndOfSubframe) != 0);
>+  descriptor->SetFirstSubFrameInFrame((data[0] & kFlagFirstSubframe) != 0);
>+  descriptor->SetLastSubFrameInFrame((data[0] & kFlagLastSubframe) != 0);
>+
>+  // Parse Subframe details provided in 1st packet of subframe.
>+  if (!begins_subframe) {
>+    return data.size() == 1;
>+  }
>+  if (data.size() < 4) {
>+    return false;
>+  }
>+  descriptor->SetTemporalLayer(data[0] & kMaskTemporalLayer);
>+  descriptor->SetSpatialLayersBitmask(data[1]);
>+  descriptor->SetFrameId(data[2] | (data[3] << 8));
>+
>+  // Parse dependencies.
>+  descriptor->ClearFrameDependencies();
>+  size_t offset = 4;
>+  bool has_more_dependencies = (data[0] & kFlagDependencies) != 0;
>+  while (has_more_dependencies) {
>+    if (data.size() == offset)
>+      return false;
>+    has_more_dependencies = (data[offset] & kFlagMoreDependencies) != 0;
>+    bool extended = (data[offset] & kFlageXtendedOffset) != 0;
>+    uint16_t fdiff = data[offset] >> 2;
>+    offset++;
>+    if (extended) {
>+      if (data.size() == offset)
>+        return false;
>+      fdiff |= (data[offset] << 6);
>+      offset++;
>+    }
>+    if (!descriptor->AddFrameDependencyDiff(fdiff))
>+      return false;
>+  }
>+  return data.size() == offset;
>+}
>+
>+size_t RtpGenericFrameDescriptorExtension::ValueSize(
>+    const RtpGenericFrameDescriptor& descriptor) {
>+  if (!descriptor.FirstPacketInSubFrame())
>+    return 1;
>+
>+  size_t size = 4;
>+  for (uint16_t fdiff : descriptor.FrameDependenciesDiffs()) {
>+    size += (fdiff >= (1 << 6)) ? 2 : 1;
>+  }
>+  return size;
>+}
>+
>+bool RtpGenericFrameDescriptorExtension::Write(
>+    rtc::ArrayView<uint8_t> data,
>+    const RtpGenericFrameDescriptor& descriptor) {
>+  RTC_CHECK_EQ(data.size(), ValueSize(descriptor));
>+  uint8_t base_header =
>+      (descriptor.FirstPacketInSubFrame() ? kFlagBeginOfSubframe : 0) |
>+      (descriptor.LastPacketInSubFrame() ? kFlagEndOfSubframe : 0) |
>+      (descriptor.FirstSubFrameInFrame() ? kFlagFirstSubframe : 0) |
>+      (descriptor.LastSubFrameInFrame() ? kFlagLastSubframe : 0);
>+  if (!descriptor.FirstPacketInSubFrame()) {
>+    data[0] = base_header;
>+    return true;
>+  }
>+  data[0] =
>+      base_header |
>+      (descriptor.FrameDependenciesDiffs().empty() ? 0 : kFlagDependencies) |
>+      descriptor.TemporalLayer();
>+  data[1] = descriptor.SpatialLayersBitmask();
>+  uint16_t frame_id = descriptor.FrameId();
>+  data[2] = frame_id & 0xff;
>+  data[3] = frame_id >> 8;
>+  rtc::ArrayView<const uint16_t> fdiffs = descriptor.FrameDependenciesDiffs();
>+  size_t offset = 4;
>+  for (size_t i = 0; i < fdiffs.size(); i++) {
>+    bool extended = fdiffs[i] >= (1 << 6);
>+    bool more = i < fdiffs.size() - 1;
>+    data[offset++] = ((fdiffs[i] & 0x3f) << 2) |
>+                     (extended ? kFlageXtendedOffset : 0) |
>+                     (more ? kFlagMoreDependencies : 0);
>+    if (extended) {
>+      data[offset++] = fdiffs[i] >> 6;
>+    }
>+  }
>+  return true;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.h
>new file mode 100644
>index 00000000000..d6acbe562b8
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.h
>@@ -0,0 +1,37 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#ifndef MODULES_RTP_RTCP_SOURCE_RTP_GENERIC_FRAME_DESCRIPTOR_EXTENSION_H_
>+#define MODULES_RTP_RTCP_SOURCE_RTP_GENERIC_FRAME_DESCRIPTOR_EXTENSION_H_
>+
>+#include <stddef.h>
>+#include <stdint.h>
>+
>+#include "api/array_view.h"
>+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>+#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor.h"
>+
>+namespace webrtc {
>+
>+class RtpGenericFrameDescriptorExtension {
>+ public:
>+  static constexpr RTPExtensionType kId = kRtpExtensionGenericFrameDescriptor;
>+  static constexpr char kUri[] =
>+      "http://www.webrtc.org/experiments/rtp-hdrext/generic-frame-descriptor";
>+
>+  static bool Parse(rtc::ArrayView<const uint8_t> data,
>+                    RtpGenericFrameDescriptor* descriptor);
>+  static size_t ValueSize(const RtpGenericFrameDescriptor&);
>+  static bool Write(rtc::ArrayView<uint8_t> data,
>+                    const RtpGenericFrameDescriptor& descriptor);
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_RTP_RTCP_SOURCE_RTP_GENERIC_FRAME_DESCRIPTOR_EXTENSION_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension_unittest.cc
>new file mode 100644
>index 00000000000..bb69e1455dd
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension_unittest.cc
>@@ -0,0 +1,283 @@
>+/*
>+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.h"
>+
>+#include "test/gmock.h"
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+namespace {
>+
>+using ::testing::ElementsAre;
>+using ::testing::ElementsAreArray;
>+using ::testing::IsEmpty;
>+
>+// TODO(danilchap): Add fuzzer to test for various invalid inputs.
>+
>+TEST(RtpGenericFrameDescriptorExtensionTest,
>+     ParseFirstPacketOfIndependenSubFrame) {
>+  const int kTemporalLayer = 5;
>+  constexpr uint8_t kRaw[] = {0x80 | kTemporalLayer, 0x49, 0x12, 0x34};
>+  RtpGenericFrameDescriptor descriptor;
>+
>+  ASSERT_TRUE(RtpGenericFrameDescriptorExtension::Parse(kRaw, &descriptor));
>+
>+  EXPECT_TRUE(descriptor.FirstPacketInSubFrame());
>+  EXPECT_FALSE(descriptor.LastPacketInSubFrame());
>+  EXPECT_FALSE(descriptor.FirstSubFrameInFrame());
>+  EXPECT_FALSE(descriptor.LastSubFrameInFrame());
>+  EXPECT_THAT(descriptor.FrameDependenciesDiffs(), IsEmpty());
>+  EXPECT_EQ(descriptor.TemporalLayer(), kTemporalLayer);
>+  EXPECT_EQ(descriptor.SpatialLayersBitmask(), 0x49);
>+  EXPECT_EQ(descriptor.FrameId(), 0x3412);
>+}
>+
>+TEST(RtpGenericFrameDescriptorExtensionTest,
>+     WriteFirstPacketOfIndependenSubFrame) {
>+  const int kTemporalLayer = 5;
>+  constexpr uint8_t kRaw[] = {0x80 | kTemporalLayer, 0x49, 0x12, 0x34};
>+  RtpGenericFrameDescriptor descriptor;
>+
>+  descriptor.SetFirstPacketInSubFrame(true);
>+  descriptor.SetTemporalLayer(kTemporalLayer);
>+  descriptor.SetSpatialLayersBitmask(0x49);
>+  descriptor.SetFrameId(0x3412);
>+
>+  ASSERT_EQ(RtpGenericFrameDescriptorExtension::ValueSize(descriptor),
>+            sizeof(kRaw));
>+  uint8_t buffer[sizeof(kRaw)];
>+  EXPECT_TRUE(RtpGenericFrameDescriptorExtension::Write(buffer, descriptor));
>+  EXPECT_THAT(buffer, ElementsAreArray(kRaw));
>+}
>+
>+TEST(RtpGenericFrameDescriptorExtensionTest, ParseLastPacketOfSubFrame) {
>+  constexpr uint8_t kRaw[] = {0x40};
>+  RtpGenericFrameDescriptor descriptor;
>+
>+  ASSERT_TRUE(RtpGenericFrameDescriptorExtension::Parse(kRaw, &descriptor));
>+
>+  EXPECT_FALSE(descriptor.FirstPacketInSubFrame());
>+  EXPECT_TRUE(descriptor.LastPacketInSubFrame());
>+  EXPECT_FALSE(descriptor.FirstSubFrameInFrame());
>+  EXPECT_FALSE(descriptor.LastSubFrameInFrame());
>+}
>+
>+TEST(RtpGenericFrameDescriptorExtensionTest, WriteLastPacketOfSubFrame) {
>+  constexpr uint8_t kRaw[] = {0x40};
>+  RtpGenericFrameDescriptor descriptor;
>+  descriptor.SetLastPacketInSubFrame(true);
>+
>+  ASSERT_EQ(RtpGenericFrameDescriptorExtension::ValueSize(descriptor),
>+            sizeof(kRaw));
>+  uint8_t buffer[sizeof(kRaw)];
>+  EXPECT_TRUE(RtpGenericFrameDescriptorExtension::Write(buffer, descriptor));
>+  EXPECT_THAT(buffer, ElementsAreArray(kRaw));
>+}
>+TEST(RtpGenericFrameDescriptorExtensionTest, ParseFirstSubFrameInFrame) {
>+  constexpr uint8_t kRaw[] = {0x20};
>+  RtpGenericFrameDescriptor descriptor;
>+
>+  ASSERT_TRUE(RtpGenericFrameDescriptorExtension::Parse(kRaw, &descriptor));
>+
>+  EXPECT_FALSE(descriptor.FirstPacketInSubFrame());
>+  EXPECT_FALSE(descriptor.LastPacketInSubFrame());
>+  EXPECT_TRUE(descriptor.FirstSubFrameInFrame());
>+  EXPECT_FALSE(descriptor.LastSubFrameInFrame());
>+}
>+
>+TEST(RtpGenericFrameDescriptorExtensionTest, WriteFirstSubFrameInFrame) {
>+  constexpr uint8_t kRaw[] = {0x20};
>+  RtpGenericFrameDescriptor descriptor;
>+  descriptor.SetFirstSubFrameInFrame(true);
>+
>+  ASSERT_EQ(RtpGenericFrameDescriptorExtension::ValueSize(descriptor),
>+            sizeof(kRaw));
>+  uint8_t buffer[sizeof(kRaw)];
>+  EXPECT_TRUE(RtpGenericFrameDescriptorExtension::Write(buffer, descriptor));
>+  EXPECT_THAT(buffer, ElementsAreArray(kRaw));
>+}
>+
>+TEST(RtpGenericFrameDescriptorExtensionTest, ParseLastSubFrameInFrame) {
>+  constexpr uint8_t kRaw[] = {0x10};
>+  RtpGenericFrameDescriptor descriptor;
>+
>+  ASSERT_TRUE(RtpGenericFrameDescriptorExtension::Parse(kRaw, &descriptor));
>+
>+  EXPECT_FALSE(descriptor.FirstPacketInSubFrame());
>+  EXPECT_FALSE(descriptor.LastPacketInSubFrame());
>+  EXPECT_FALSE(descriptor.FirstSubFrameInFrame());
>+  EXPECT_TRUE(descriptor.LastSubFrameInFrame());
>+}
>+
>+TEST(RtpGenericFrameDescriptorExtensionTest, WriteLastSubFrameInFrame) {
>+  constexpr uint8_t kRaw[] = {0x10};
>+  RtpGenericFrameDescriptor descriptor;
>+  descriptor.SetLastSubFrameInFrame(true);
>+
>+  ASSERT_EQ(RtpGenericFrameDescriptorExtension::ValueSize(descriptor),
>+            sizeof(kRaw));
>+  uint8_t buffer[sizeof(kRaw)];
>+  EXPECT_TRUE(RtpGenericFrameDescriptorExtension::Write(buffer, descriptor));
>+  EXPECT_THAT(buffer, ElementsAreArray(kRaw));
>+}
>+
>+TEST(RtpGenericFrameDescriptorExtensionTest, ParseMinShortFrameDependencies) {
>+  constexpr uint16_t kDiff = 1;
>+  constexpr uint8_t kRaw[] = {0x88, 0x01, 0x00, 0x00, 0x04};
>+  RtpGenericFrameDescriptor descriptor;
>+
>+  ASSERT_TRUE(RtpGenericFrameDescriptorExtension::Parse(kRaw, &descriptor));
>+  ASSERT_TRUE(descriptor.FirstPacketInSubFrame());
>+  EXPECT_THAT(descriptor.FrameDependenciesDiffs(), ElementsAre(kDiff));
>+}
>+
>+TEST(RtpGenericFrameDescriptorExtensionTest, WriteMinShortFrameDependencies) {
>+  constexpr uint16_t kDiff = 1;
>+  constexpr uint8_t kRaw[] = {0x88, 0x01, 0x00, 0x00, 0x04};
>+  RtpGenericFrameDescriptor descriptor;
>+  descriptor.SetFirstPacketInSubFrame(true);
>+  descriptor.AddFrameDependencyDiff(kDiff);
>+
>+  ASSERT_EQ(RtpGenericFrameDescriptorExtension::ValueSize(descriptor),
>+            sizeof(kRaw));
>+  uint8_t buffer[sizeof(kRaw)];
>+  EXPECT_TRUE(RtpGenericFrameDescriptorExtension::Write(buffer, descriptor));
>+  EXPECT_THAT(buffer, ElementsAreArray(kRaw));
>+}
>+
>+TEST(RtpGenericFrameDescriptorExtensionTest, ParseMaxShortFrameDependencies) {
>+  constexpr uint16_t kDiff = 0x3f;
>+  constexpr uint8_t kRaw[] = {0x88, 0x01, 0x00, 0x00, 0xfc};
>+  RtpGenericFrameDescriptor descriptor;
>+
>+  ASSERT_TRUE(RtpGenericFrameDescriptorExtension::Parse(kRaw, &descriptor));
>+  ASSERT_TRUE(descriptor.FirstPacketInSubFrame());
>+  EXPECT_THAT(descriptor.FrameDependenciesDiffs(), ElementsAre(kDiff));
>+}
>+
>+TEST(RtpGenericFrameDescriptorExtensionTest, WriteMaxShortFrameDependencies) {
>+  constexpr uint16_t kDiff = 0x3f;
>+  constexpr uint8_t kRaw[] = {0x88, 0x01, 0x00, 0x00, 0xfc};
>+  RtpGenericFrameDescriptor descriptor;
>+  descriptor.SetFirstPacketInSubFrame(true);
>+  descriptor.AddFrameDependencyDiff(kDiff);
>+
>+  ASSERT_EQ(RtpGenericFrameDescriptorExtension::ValueSize(descriptor),
>+            sizeof(kRaw));
>+  uint8_t buffer[sizeof(kRaw)];
>+  EXPECT_TRUE(RtpGenericFrameDescriptorExtension::Write(buffer, descriptor));
>+  EXPECT_THAT(buffer, ElementsAreArray(kRaw));
>+}
>+
>+TEST(RtpGenericFrameDescriptorExtensionTest, ParseMinLongFrameDependencies) {
>+  constexpr uint16_t kDiff = 0x40;
>+  constexpr uint8_t kRaw[] = {0x88, 0x01, 0x00, 0x00, 0x02, 0x01};
>+  RtpGenericFrameDescriptor descriptor;
>+
>+  ASSERT_TRUE(RtpGenericFrameDescriptorExtension::Parse(kRaw, &descriptor));
>+  ASSERT_TRUE(descriptor.FirstPacketInSubFrame());
>+  EXPECT_THAT(descriptor.FrameDependenciesDiffs(), ElementsAre(kDiff));
>+}
>+
>+TEST(RtpGenericFrameDescriptorExtensionTest, WriteMinLongFrameDependencies) {
>+  constexpr uint16_t kDiff = 0x40;
>+  constexpr uint8_t kRaw[] = {0x88, 0x01, 0x00, 0x00, 0x02, 0x01};
>+  RtpGenericFrameDescriptor descriptor;
>+  descriptor.SetFirstPacketInSubFrame(true);
>+  descriptor.AddFrameDependencyDiff(kDiff);
>+
>+  ASSERT_EQ(RtpGenericFrameDescriptorExtension::ValueSize(descriptor),
>+            sizeof(kRaw));
>+  uint8_t buffer[sizeof(kRaw)];
>+  EXPECT_TRUE(RtpGenericFrameDescriptorExtension::Write(buffer, descriptor));
>+  EXPECT_THAT(buffer, ElementsAreArray(kRaw));
>+}
>+
>+TEST(RtpGenericFrameDescriptorExtensionTest,
>+     ParseLongFrameDependenciesAsBigEndian) {
>+  constexpr uint16_t kDiff = 0x7654 >> 2;
>+  constexpr uint8_t kRaw[] = {0x88, 0x01, 0x00, 0x00, 0x54 | 0x02, 0x76};
>+  RtpGenericFrameDescriptor descriptor;
>+
>+  ASSERT_TRUE(RtpGenericFrameDescriptorExtension::Parse(kRaw, &descriptor));
>+  ASSERT_TRUE(descriptor.FirstPacketInSubFrame());
>+  EXPECT_THAT(descriptor.FrameDependenciesDiffs(), ElementsAre(kDiff));
>+}
>+
>+TEST(RtpGenericFrameDescriptorExtensionTest,
>+     WriteLongFrameDependenciesAsBigEndian) {
>+  constexpr uint16_t kDiff = 0x7654 >> 2;
>+  constexpr uint8_t kRaw[] = {0x88, 0x01, 0x00, 0x00, 0x54 | 0x02, 0x76};
>+  RtpGenericFrameDescriptor descriptor;
>+  descriptor.SetFirstPacketInSubFrame(true);
>+  descriptor.AddFrameDependencyDiff(kDiff);
>+
>+  ASSERT_EQ(RtpGenericFrameDescriptorExtension::ValueSize(descriptor),
>+            sizeof(kRaw));
>+  uint8_t buffer[sizeof(kRaw)];
>+  EXPECT_TRUE(RtpGenericFrameDescriptorExtension::Write(buffer, descriptor));
>+  EXPECT_THAT(buffer, ElementsAreArray(kRaw));
>+}
>+
>+TEST(RtpGenericFrameDescriptorExtensionTest, ParseMaxLongFrameDependencies) {
>+  constexpr uint16_t kDiff = 0x3fff;
>+  constexpr uint8_t kRaw[] = {0x88, 0x01, 0x00, 0x00, 0xfe, 0xff};
>+  RtpGenericFrameDescriptor descriptor;
>+
>+  ASSERT_TRUE(RtpGenericFrameDescriptorExtension::Parse(kRaw, &descriptor));
>+  ASSERT_TRUE(descriptor.FirstPacketInSubFrame());
>+  EXPECT_THAT(descriptor.FrameDependenciesDiffs(), ElementsAre(kDiff));
>+}
>+
>+TEST(RtpGenericFrameDescriptorExtensionTest, WriteMaxLongFrameDependencies) {
>+  constexpr uint16_t kDiff = 0x3fff;
>+  constexpr uint8_t kRaw[] = {0x88, 0x01, 0x00, 0x00, 0xfe, 0xff};
>+  RtpGenericFrameDescriptor descriptor;
>+  descriptor.SetFirstPacketInSubFrame(true);
>+  descriptor.AddFrameDependencyDiff(kDiff);
>+
>+  ASSERT_EQ(RtpGenericFrameDescriptorExtension::ValueSize(descriptor),
>+            sizeof(kRaw));
>+  uint8_t buffer[sizeof(kRaw)];
>+  EXPECT_TRUE(RtpGenericFrameDescriptorExtension::Write(buffer, descriptor));
>+  EXPECT_THAT(buffer, ElementsAreArray(kRaw));
>+}
>+
>+TEST(RtpGenericFrameDescriptorExtensionTest, ParseTwoFrameDependencies) {
>+  constexpr uint16_t kDiff1 = 9;
>+  constexpr uint16_t kDiff2 = 15;
>+  constexpr uint8_t kRaw[] = {
>+      0x88, 0x01, 0x00, 0x00, (kDiff1 << 2) | 0x01, kDiff2 << 2};
>+  RtpGenericFrameDescriptor descriptor;
>+
>+  ASSERT_TRUE(RtpGenericFrameDescriptorExtension::Parse(kRaw, &descriptor));
>+  ASSERT_TRUE(descriptor.FirstPacketInSubFrame());
>+  EXPECT_THAT(descriptor.FrameDependenciesDiffs(), ElementsAre(kDiff1, kDiff2));
>+}
>+
>+TEST(RtpGenericFrameDescriptorExtensionTest, WriteTwoFrameDependencies) {
>+  constexpr uint16_t kDiff1 = 9;
>+  constexpr uint16_t kDiff2 = 15;
>+  constexpr uint8_t kRaw[] = {
>+      0x88, 0x01, 0x00, 0x00, (kDiff1 << 2) | 0x01, kDiff2 << 2};
>+  RtpGenericFrameDescriptor descriptor;
>+  descriptor.SetFirstPacketInSubFrame(true);
>+  descriptor.AddFrameDependencyDiff(kDiff1);
>+  descriptor.AddFrameDependencyDiff(kDiff2);
>+
>+  ASSERT_EQ(RtpGenericFrameDescriptorExtension::ValueSize(descriptor),
>+            sizeof(kRaw));
>+  uint8_t buffer[sizeof(kRaw)];
>+  EXPECT_TRUE(RtpGenericFrameDescriptorExtension::Write(buffer, descriptor));
>+  EXPECT_THAT(buffer, ElementsAreArray(kRaw));
>+}
>+
>+}  // namespace
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_header_extension_map.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_header_extension_map.cc
>index f044e9b0393..9fa66c08192 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_header_extension_map.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_header_extension_map.cc
>@@ -10,6 +10,7 @@
> 
> #include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
> 
>+#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.h"
> #include "modules/rtp_rtcp/source/rtp_header_extensions.h"
> #include "rtc_base/arraysize.h"
> #include "rtc_base/checks.h"
>@@ -40,6 +41,7 @@ constexpr ExtensionInfo kExtensions[] = {
>     CreateExtensionInfo<RtpStreamId>(),
>     CreateExtensionInfo<RepairedRtpStreamId>(),
>     CreateExtensionInfo<RtpMid>(),
>+    CreateExtensionInfo<RtpGenericFrameDescriptorExtension>(),
> };
> 
> // Because of kRtpExtensionNone, NumberOfExtension is 1 bigger than the actual
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_header_extension_map_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_header_extension_map_unittest.cc
>index 5eb8f1a62d0..ad9f79c509c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_header_extension_map_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_header_extension_map_unittest.cc
>@@ -14,7 +14,6 @@
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> #include "modules/rtp_rtcp/source/rtp_header_extensions.h"
> #include "test/gtest.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.cc
>index 958ad028a96..2dba4d7cfac 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.cc
>@@ -43,9 +43,11 @@ bool AbsoluteSendTime::Parse(rtc::ArrayView<const uint8_t> data,
>   return true;
> }
> 
>-bool AbsoluteSendTime::Write(uint8_t* data, uint32_t time_24bits) {
>+bool AbsoluteSendTime::Write(rtc::ArrayView<uint8_t> data,
>+                             uint32_t time_24bits) {
>+  RTC_DCHECK_EQ(data.size(), 3);
>   RTC_DCHECK_LE(time_24bits, 0x00FFFFFF);
>-  ByteWriter<uint32_t, 3>::WriteBigEndian(data, time_24bits);
>+  ByteWriter<uint32_t, 3>::WriteBigEndian(data.data(), time_24bits);
>   return true;
> }
> 
>@@ -75,9 +77,10 @@ bool AudioLevel::Parse(rtc::ArrayView<const uint8_t> data,
>   return true;
> }
> 
>-bool AudioLevel::Write(uint8_t* data,
>+bool AudioLevel::Write(rtc::ArrayView<uint8_t> data,
>                        bool voice_activity,
>                        uint8_t audio_level) {
>+  RTC_DCHECK_EQ(data.size(), 1);
>   RTC_CHECK_LE(audio_level, 0x7f);
>   data[0] = (voice_activity ? 0x80 : 0x00) | audio_level;
>   return true;
>@@ -111,9 +114,10 @@ bool TransmissionOffset::Parse(rtc::ArrayView<const uint8_t> data,
>   return true;
> }
> 
>-bool TransmissionOffset::Write(uint8_t* data, int32_t rtp_time) {
>+bool TransmissionOffset::Write(rtc::ArrayView<uint8_t> data, int32_t rtp_time) {
>+  RTC_DCHECK_EQ(data.size(), 3);
>   RTC_DCHECK_LE(rtp_time, 0x00ffffff);
>-  ByteWriter<int32_t, 3>::WriteBigEndian(data, rtp_time);
>+  ByteWriter<int32_t, 3>::WriteBigEndian(data.data(), rtp_time);
>   return true;
> }
> 
>@@ -134,8 +138,10 @@ bool TransportSequenceNumber::Parse(rtc::ArrayView<const uint8_t> data,
>   return true;
> }
> 
>-bool TransportSequenceNumber::Write(uint8_t* data, uint16_t value) {
>-  ByteWriter<uint16_t>::WriteBigEndian(data, value);
>+bool TransportSequenceNumber::Write(rtc::ArrayView<uint8_t> data,
>+                                    uint16_t value) {
>+  RTC_DCHECK_EQ(data.size(), 2);
>+  ByteWriter<uint16_t>::WriteBigEndian(data.data(), value);
>   return true;
> }
> 
>@@ -162,7 +168,9 @@ bool VideoOrientation::Parse(rtc::ArrayView<const uint8_t> data,
>   return true;
> }
> 
>-bool VideoOrientation::Write(uint8_t* data, VideoRotation rotation) {
>+bool VideoOrientation::Write(rtc::ArrayView<uint8_t> data,
>+                             VideoRotation rotation) {
>+  RTC_DCHECK_EQ(data.size(), 1);
>   data[0] = ConvertVideoRotationToCVOByte(rotation);
>   return true;
> }
>@@ -175,7 +183,8 @@ bool VideoOrientation::Parse(rtc::ArrayView<const uint8_t> data,
>   return true;
> }
> 
>-bool VideoOrientation::Write(uint8_t* data, uint8_t value) {
>+bool VideoOrientation::Write(rtc::ArrayView<uint8_t> data, uint8_t value) {
>+  RTC_DCHECK_EQ(data.size(), 1);
>   data[0] = value;
>   return true;
> }
>@@ -204,15 +213,17 @@ bool PlayoutDelayLimits::Parse(rtc::ArrayView<const uint8_t> data,
>   return true;
> }
> 
>-bool PlayoutDelayLimits::Write(uint8_t* data,
>+bool PlayoutDelayLimits::Write(rtc::ArrayView<uint8_t> data,
>                                const PlayoutDelay& playout_delay) {
>+  RTC_DCHECK_EQ(data.size(), 3);
>   RTC_DCHECK_LE(0, playout_delay.min_ms);
>   RTC_DCHECK_LE(playout_delay.min_ms, playout_delay.max_ms);
>   RTC_DCHECK_LE(playout_delay.max_ms, kMaxMs);
>   // Convert MS to value to be sent on extension header.
>   uint32_t min_delay = playout_delay.min_ms / kGranularityMs;
>   uint32_t max_delay = playout_delay.max_ms / kGranularityMs;
>-  ByteWriter<uint32_t, 3>::WriteBigEndian(data, (min_delay << 12) | max_delay);
>+  ByteWriter<uint32_t, 3>::WriteBigEndian(data.data(),
>+                                          (min_delay << 12) | max_delay);
>   return true;
> }
> 
>@@ -239,8 +250,9 @@ bool VideoContentTypeExtension::Parse(rtc::ArrayView<const uint8_t> data,
>   return false;
> }
> 
>-bool VideoContentTypeExtension::Write(uint8_t* data,
>+bool VideoContentTypeExtension::Write(rtc::ArrayView<uint8_t> data,
>                                       VideoContentType content_type) {
>+  RTC_DCHECK_EQ(data.size(), 1);
>   data[0] = static_cast<uint8_t>(content_type);
>   return true;
> }
>@@ -303,35 +315,38 @@ bool VideoTimingExtension::Parse(rtc::ArrayView<const uint8_t> data,
>   return true;
> }
> 
>-bool VideoTimingExtension::Write(uint8_t* data, const VideoSendTiming& timing) {
>-  ByteWriter<uint8_t>::WriteBigEndian(data + VideoSendTiming::kFlagsOffset,
>-                                      timing.flags);
>+bool VideoTimingExtension::Write(rtc::ArrayView<uint8_t> data,
>+                                 const VideoSendTiming& timing) {
>+  RTC_DCHECK_EQ(data.size(), 1 + 2 * 6);
>+  ByteWriter<uint8_t>::WriteBigEndian(
>+      data.data() + VideoSendTiming::kFlagsOffset, timing.flags);
>   ByteWriter<uint16_t>::WriteBigEndian(
>-      data + VideoSendTiming::kEncodeStartDeltaOffset,
>+      data.data() + VideoSendTiming::kEncodeStartDeltaOffset,
>       timing.encode_start_delta_ms);
>   ByteWriter<uint16_t>::WriteBigEndian(
>-      data + VideoSendTiming::kEncodeFinishDeltaOffset,
>+      data.data() + VideoSendTiming::kEncodeFinishDeltaOffset,
>       timing.encode_finish_delta_ms);
>   ByteWriter<uint16_t>::WriteBigEndian(
>-      data + VideoSendTiming::kPacketizationFinishDeltaOffset,
>+      data.data() + VideoSendTiming::kPacketizationFinishDeltaOffset,
>       timing.packetization_finish_delta_ms);
>   ByteWriter<uint16_t>::WriteBigEndian(
>-      data + VideoSendTiming::kPacerExitDeltaOffset,
>+      data.data() + VideoSendTiming::kPacerExitDeltaOffset,
>       timing.pacer_exit_delta_ms);
>   ByteWriter<uint16_t>::WriteBigEndian(
>-      data + VideoSendTiming::kNetworkTimestampDeltaOffset,
>+      data.data() + VideoSendTiming::kNetworkTimestampDeltaOffset,
>       timing.network_timestamp_delta_ms);
>   ByteWriter<uint16_t>::WriteBigEndian(
>-      data + VideoSendTiming::kNetwork2TimestampDeltaOffset,
>+      data.data() + VideoSendTiming::kNetwork2TimestampDeltaOffset,
>       timing.network2_timestamp_delta_ms);
>   return true;
> }
> 
>-bool VideoTimingExtension::Write(uint8_t* data,
>+bool VideoTimingExtension::Write(rtc::ArrayView<uint8_t> data,
>                                  uint16_t time_delta_ms,
>                                  uint8_t offset) {
>+  RTC_DCHECK_GE(data.size(), offset + 2);
>   RTC_DCHECK_LE(offset, kValueSizeBytes - sizeof(uint16_t));
>-  ByteWriter<uint16_t>::WriteBigEndian(data + offset, time_delta_ms);
>+  ByteWriter<uint16_t>::WriteBigEndian(data.data() + offset, time_delta_ms);
>   return true;
> }
> 
>@@ -344,11 +359,12 @@ bool BaseRtpStringExtension::Parse(rtc::ArrayView<const uint8_t> data,
>   return true;
> }
> 
>-bool BaseRtpStringExtension::Write(uint8_t* data,
>+bool BaseRtpStringExtension::Write(rtc::ArrayView<uint8_t> data,
>                                    const StringRtpHeaderExtension& str) {
>+  RTC_DCHECK_EQ(data.size(), str.size());
>   RTC_DCHECK_GE(str.size(), 1);
>   RTC_DCHECK_LE(str.size(), StringRtpHeaderExtension::kMaxSize);
>-  memcpy(data, str.data(), str.size());
>+  memcpy(data.data(), str.data(), str.size());
>   return true;
> }
> 
>@@ -364,10 +380,12 @@ bool BaseRtpStringExtension::Parse(rtc::ArrayView<const uint8_t> data,
>   return true;
> }
> 
>-bool BaseRtpStringExtension::Write(uint8_t* data, const std::string& str) {
>+bool BaseRtpStringExtension::Write(rtc::ArrayView<uint8_t> data,
>+                                   const std::string& str) {
>+  RTC_DCHECK_EQ(data.size(), str.size());
>   RTC_DCHECK_GE(str.size(), 1);
>   RTC_DCHECK_LE(str.size(), StringRtpHeaderExtension::kMaxSize);
>-  memcpy(data, str.data(), str.size());
>+  memcpy(data.data(), str.data(), str.size());
>   return true;
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h
>index 6ef5040db57..4e1afc1d979 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h
>@@ -30,7 +30,7 @@ class AbsoluteSendTime {
> 
>   static bool Parse(rtc::ArrayView<const uint8_t> data, uint32_t* time_24bits);
>   static size_t ValueSize(uint32_t time_24bits) { return kValueSizeBytes; }
>-  static bool Write(uint8_t* data, uint32_t time_24bits);
>+  static bool Write(rtc::ArrayView<uint8_t> data, uint32_t time_24bits);
> 
>   static constexpr uint32_t MsTo24Bits(int64_t time_ms) {
>     return static_cast<uint32_t>(((time_ms << 18) + 500) / 1000) & 0x00FFFFFF;
>@@ -50,7 +50,9 @@ class AudioLevel {
>   static size_t ValueSize(bool voice_activity, uint8_t audio_level) {
>     return kValueSizeBytes;
>   }
>-  static bool Write(uint8_t* data, bool voice_activity, uint8_t audio_level);
>+  static bool Write(rtc::ArrayView<uint8_t> data,
>+                    bool voice_activity,
>+                    uint8_t audio_level);
> };
> 
> class TransmissionOffset {
>@@ -61,7 +63,7 @@ class TransmissionOffset {
> 
>   static bool Parse(rtc::ArrayView<const uint8_t> data, int32_t* rtp_time);
>   static size_t ValueSize(int32_t rtp_time) { return kValueSizeBytes; }
>-  static bool Write(uint8_t* data, int32_t rtp_time);
>+  static bool Write(rtc::ArrayView<uint8_t> data, int32_t rtp_time);
> };
> 
> class TransportSequenceNumber {
>@@ -73,7 +75,7 @@ class TransportSequenceNumber {
>       "draft-holmer-rmcat-transport-wide-cc-extensions-01";
>   static bool Parse(rtc::ArrayView<const uint8_t> data, uint16_t* value);
>   static size_t ValueSize(uint16_t value) { return kValueSizeBytes; }
>-  static bool Write(uint8_t* data, uint16_t value);
>+  static bool Write(rtc::ArrayView<uint8_t> data, uint16_t value);
> };
> 
> class VideoOrientation {
>@@ -84,10 +86,10 @@ class VideoOrientation {
> 
>   static bool Parse(rtc::ArrayView<const uint8_t> data, VideoRotation* value);
>   static size_t ValueSize(VideoRotation) { return kValueSizeBytes; }
>-  static bool Write(uint8_t* data, VideoRotation value);
>+  static bool Write(rtc::ArrayView<uint8_t> data, VideoRotation value);
>   static bool Parse(rtc::ArrayView<const uint8_t> data, uint8_t* value);
>   static size_t ValueSize(uint8_t value) { return kValueSizeBytes; }
>-  static bool Write(uint8_t* data, uint8_t value);
>+  static bool Write(rtc::ArrayView<uint8_t> data, uint8_t value);
> };
> 
> class PlayoutDelayLimits {
>@@ -109,7 +111,8 @@ class PlayoutDelayLimits {
>   static size_t ValueSize(const PlayoutDelay&) {
>     return kValueSizeBytes;
>   }
>-  static bool Write(uint8_t* data, const PlayoutDelay& playout_delay);
>+  static bool Write(rtc::ArrayView<uint8_t> data,
>+                    const PlayoutDelay& playout_delay);
> };
> 
> class VideoContentTypeExtension {
>@@ -124,7 +127,8 @@ class VideoContentTypeExtension {
>   static size_t ValueSize(VideoContentType) {
>     return kValueSizeBytes;
>   }
>-  static bool Write(uint8_t* data, VideoContentType content_type);
>+  static bool Write(rtc::ArrayView<uint8_t> data,
>+                    VideoContentType content_type);
> };
> 
> class VideoTimingExtension {
>@@ -137,29 +141,37 @@ class VideoTimingExtension {
>   static bool Parse(rtc::ArrayView<const uint8_t> data,
>                     VideoSendTiming* timing);
>   static size_t ValueSize(const VideoSendTiming&) { return kValueSizeBytes; }
>-  static bool Write(uint8_t* data, const VideoSendTiming& timing);
>+  static bool Write(rtc::ArrayView<uint8_t> data,
>+                    const VideoSendTiming& timing);
> 
>   static size_t ValueSize(uint16_t time_delta_ms, uint8_t idx) {
>     return kValueSizeBytes;
>   }
>   // Writes only single time delta to position idx.
>-  static bool Write(uint8_t* data, uint16_t time_delta_ms, uint8_t idx);
>+  static bool Write(rtc::ArrayView<uint8_t> data,
>+                    uint16_t time_delta_ms,
>+                    uint8_t idx);
> };
> 
> // Base extension class for RTP header extensions which are strings.
> // Subclasses must defined kId and kUri static constexpr members.
> class BaseRtpStringExtension {
>  public:
>+  // String RTP header extensions are limited to 16 bytes because it is the
>+  // maximum length that can be encoded with one-byte header extensions.
>+  static constexpr uint8_t kMaxValueSizeBytes = 16;
>+
>   static bool Parse(rtc::ArrayView<const uint8_t> data,
>                     StringRtpHeaderExtension* str);
>   static size_t ValueSize(const StringRtpHeaderExtension& str) {
>     return str.size();
>   }
>-  static bool Write(uint8_t* data, const StringRtpHeaderExtension& str);
>+  static bool Write(rtc::ArrayView<uint8_t> data,
>+                    const StringRtpHeaderExtension& str);
> 
>   static bool Parse(rtc::ArrayView<const uint8_t> data, std::string* str);
>   static size_t ValueSize(const std::string& str) { return str.size(); }
>-  static bool Write(uint8_t* data, const std::string& str);
>+  static bool Write(rtc::ArrayView<uint8_t> data, const std::string& str);
> };
> 
> class RtpStreamId : public BaseRtpStringExtension {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc
>index 5d2971a0c61..df68f74dd93 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_header_parser.cc
>@@ -18,7 +18,7 @@ namespace webrtc {
> class RtpHeaderParserImpl : public RtpHeaderParser {
>  public:
>   RtpHeaderParserImpl();
>-  virtual ~RtpHeaderParserImpl() {}
>+  ~RtpHeaderParserImpl() override = default;
> 
>   bool Parse(const uint8_t* packet,
>              size_t length,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet.cc
>index f2a9709ca44..4387fe74c3e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet.cc
>@@ -107,32 +107,6 @@ bool RtpPacket::Parse(rtc::CopyOnWriteBuffer buffer) {
>   return true;
> }
> 
>-bool RtpPacket::Marker() const {
>-  RTC_DCHECK_EQ(marker_, (data()[1] & 0x80) != 0);
>-  return marker_;
>-}
>-
>-uint8_t RtpPacket::PayloadType() const {
>-  RTC_DCHECK_EQ(payload_type_, data()[1] & 0x7f);
>-  return payload_type_;
>-}
>-
>-uint16_t RtpPacket::SequenceNumber() const {
>-  RTC_DCHECK_EQ(sequence_number_,
>-                ByteReader<uint16_t>::ReadBigEndian(data() + 2));
>-  return sequence_number_;
>-}
>-
>-uint32_t RtpPacket::Timestamp() const {
>-  RTC_DCHECK_EQ(timestamp_, ByteReader<uint32_t>::ReadBigEndian(data() + 4));
>-  return timestamp_;
>-}
>-
>-uint32_t RtpPacket::Ssrc() const {
>-  RTC_DCHECK_EQ(ssrc_, ByteReader<uint32_t>::ReadBigEndian(data() + 8));
>-  return ssrc_;
>-}
>-
> std::vector<uint32_t> RtpPacket::Csrcs() const {
>   size_t num_csrc = data()[0] & 0x0F;
>   RTC_DCHECK_GE(capacity(), kFixedHeaderSize + num_csrc * 4);
>@@ -144,48 +118,6 @@ std::vector<uint32_t> RtpPacket::Csrcs() const {
>   return csrcs;
> }
> 
>-size_t RtpPacket::headers_size() const {
>-  return payload_offset_;
>-}
>-
>-size_t RtpPacket::payload_size() const {
>-  return payload_size_;
>-}
>-
>-size_t RtpPacket::padding_size() const {
>-  return padding_size_;
>-}
>-
>-rtc::ArrayView<const uint8_t> RtpPacket::payload() const {
>-  return rtc::MakeArrayView(data() + payload_offset_, payload_size_);
>-}
>-
>-rtc::CopyOnWriteBuffer RtpPacket::Buffer() const {
>-  return buffer_;
>-}
>-
>-size_t RtpPacket::capacity() const {
>-  return buffer_.capacity();
>-}
>-
>-size_t RtpPacket::size() const {
>-  size_t ret = payload_offset_ + payload_size_ + padding_size_;
>-  RTC_DCHECK_EQ(buffer_.size(), ret);
>-  return ret;
>-}
>-
>-const uint8_t* RtpPacket::data() const {
>-  return buffer_.cdata();
>-}
>-
>-size_t RtpPacket::FreeCapacity() const {
>-  return capacity() - size();
>-}
>-
>-size_t RtpPacket::MaxPayloadSize() const {
>-  return capacity() - payload_offset_;
>-}
>-
> void RtpPacket::CopyHeaderFrom(const RtpPacket& packet) {
>   RTC_DCHECK_GE(capacity(), packet.headers_size());
> 
>@@ -235,7 +167,7 @@ void RtpPacket::SetSsrc(uint32_t ssrc) {
>   ByteWriter<uint32_t>::WriteBigEndian(WriteAt(8), ssrc);
> }
> 
>-void RtpPacket::SetCsrcs(const std::vector<uint32_t>& csrcs) {
>+void RtpPacket::SetCsrcs(rtc::ArrayView<const uint32_t> csrcs) {
>   RTC_DCHECK_EQ(extensions_size_, 0);
>   RTC_DCHECK_EQ(payload_size_, 0);
>   RTC_DCHECK_EQ(padding_size_, 0);
>@@ -251,37 +183,7 @@ void RtpPacket::SetCsrcs(const std::vector<uint32_t>& csrcs) {
>   buffer_.SetSize(payload_offset_);
> }
> 
>-bool RtpPacket::HasRawExtension(int id) const {
>-  if (id == ExtensionManager::kInvalidId)
>-    return false;
>-  RTC_DCHECK_GE(id, kMinExtensionId);
>-  RTC_DCHECK_LE(id, kMaxExtensionId);
>-  return extension_entries_[id - 1].offset != 0;
>-}
>-
>-rtc::ArrayView<const uint8_t> RtpPacket::GetRawExtension(int id) const {
>-  if (id == ExtensionManager::kInvalidId)
>-    return nullptr;
>-  RTC_DCHECK_GE(id, kMinExtensionId);
>-  RTC_DCHECK_LE(id, kMaxExtensionId);
>-  const ExtensionInfo& extension = extension_entries_[id - 1];
>-  if (extension.offset == 0)
>-    return nullptr;
>-  return rtc::MakeArrayView(data() + extension.offset, extension.length);
>-}
>-
>-bool RtpPacket::SetRawExtension(int id, rtc::ArrayView<const uint8_t> data) {
>-  auto buffer = AllocateRawExtension(id, data.size());
>-  if (buffer.empty())
>-    return false;
>-  RTC_DCHECK_EQ(buffer.size(), data.size());
>-  memcpy(buffer.data(), data.data(), data.size());
>-  return true;
>-}
>-
> rtc::ArrayView<uint8_t> RtpPacket::AllocateRawExtension(int id, size_t length) {
>-  if (id == ExtensionManager::kInvalidId)
>-    return nullptr;
>   RTC_DCHECK_GE(id, kMinExtensionId);
>   RTC_DCHECK_LE(id, kMaxExtensionId);
>   RTC_DCHECK_GE(length, 1);
>@@ -336,10 +238,11 @@ rtc::ArrayView<uint8_t> RtpPacket::AllocateRawExtension(int id, size_t length) {
>   extension_entry->offset = rtc::dchecked_cast<uint16_t>(
>       extensions_offset + extensions_size_ + kOneByteHeaderSize);
>   extension_entry->length = rtc::dchecked_cast<uint8_t>(length);
>-  extensions_size_ = rtc::dchecked_cast<uint16_t>(new_extensions_size);
>+  extensions_size_ = new_extensions_size;
> 
>   // Update header length field.
>-  uint16_t extensions_words = (extensions_size_ + 3) / 4;  // Wrap up to 32bit.
>+  uint16_t extensions_words = rtc::dchecked_cast<uint16_t>(
>+      (extensions_size_ + 3) / 4);  // Wrap up to 32bit.
>   ByteWriter<uint16_t>::WriteBigEndian(WriteAt(extensions_offset - 2),
>                                        extensions_words);
>   // Fill extension padding place with zeroes.
>@@ -499,11 +402,15 @@ bool RtpPacket::ParseBuffer(const uint8_t* buffer, size_t size) {
>               << "Duplicate rtp header extension id " << id << ". Overwriting.";
>         }
> 
>-        extensions_size_ += kOneByteHeaderSize;
>-        extension_entries_[idx].offset =
>-            rtc::dchecked_cast<uint16_t>(extension_offset + extensions_size_);
>-        extension_entries_[idx].length = rtc::dchecked_cast<uint16_t>(length);
>-        extensions_size_ += length;
>+        size_t offset =
>+            extension_offset + extensions_size_ + kOneByteHeaderSize;
>+        if (!rtc::IsValueInRangeForNumericType<uint16_t>(offset)) {
>+          RTC_DLOG(LS_WARNING) << "Oversized rtp header extension.";
>+          break;
>+        }
>+        extension_entries_[idx].offset = static_cast<uint16_t>(offset);
>+        extension_entries_[idx].length = length;
>+        extensions_size_ += kOneByteHeaderSize + length;
>       }
>     }
>     payload_offset_ = extension_offset + extensions_capacity;
>@@ -542,12 +449,4 @@ rtc::ArrayView<uint8_t> RtpPacket::AllocateExtension(ExtensionType type,
>   return nullptr;
> }
> 
>-uint8_t* RtpPacket::WriteAt(size_t offset) {
>-  return buffer_.data() + offset;
>-}
>-
>-void RtpPacket::WriteAt(size_t offset, uint8_t byte) {
>-  buffer_.data()[offset] = byte;
>-}
>-
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet.h
>index 8bdb8c6ad9f..21adf6229b4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet.h
>@@ -14,7 +14,6 @@
> 
> #include "api/array_view.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>-#include "rtc_base/basictypes.h"
> #include "rtc_base/copyonwritebuffer.h"
> 
> namespace webrtc {
>@@ -52,27 +51,31 @@ class RtpPacket {
>   void IdentifyExtensions(const ExtensionManager& extensions);
> 
>   // Header.
>-  bool Marker() const;
>-  uint8_t PayloadType() const;
>-  uint16_t SequenceNumber() const;
>-  uint32_t Timestamp() const;
>-  uint32_t Ssrc() const;
>+  bool Marker() const { return marker_; }
>+  uint8_t PayloadType() const { return payload_type_; }
>+  uint16_t SequenceNumber() const { return sequence_number_; }
>+  uint32_t Timestamp() const { return timestamp_; }
>+  uint32_t Ssrc() const { return ssrc_; }
>   std::vector<uint32_t> Csrcs() const;
> 
>-  size_t headers_size() const;
>+  size_t headers_size() const { return payload_offset_; }
> 
>   // Payload.
>-  size_t payload_size() const;
>-  size_t padding_size() const;
>-  rtc::ArrayView<const uint8_t> payload() const;
>+  size_t payload_size() const { return payload_size_; }
>+  size_t padding_size() const { return padding_size_; }
>+  rtc::ArrayView<const uint8_t> payload() const {
>+    return rtc::MakeArrayView(data() + payload_offset_, payload_size_);
>+  }
> 
>   // Buffer.
>-  rtc::CopyOnWriteBuffer Buffer() const;
>-  size_t capacity() const;
>-  size_t size() const;
>-  const uint8_t* data() const;
>-  size_t FreeCapacity() const;
>-  size_t MaxPayloadSize() const;
>+  rtc::CopyOnWriteBuffer Buffer() const { return buffer_; }
>+  size_t capacity() const { return buffer_.capacity(); }
>+  size_t size() const {
>+    return payload_offset_ + payload_size_ + padding_size_;
>+  }
>+  const uint8_t* data() const { return buffer_.cdata(); }
>+  size_t FreeCapacity() const { return capacity() - size(); }
>+  size_t MaxPayloadSize() const { return capacity() - headers_size(); }
> 
>   // Reset fields and buffer.
>   void Clear();
>@@ -88,7 +91,7 @@ class RtpPacket {
>   // Writes csrc list. Assumes:
>   // a) There is enough room left in buffer.
>   // b) Extension headers, payload or padding data has not already been added.
>-  void SetCsrcs(const std::vector<uint32_t>& csrcs);
>+  void SetCsrcs(rtc::ArrayView<const uint32_t> csrcs);
> 
>   // Header extensions.
>   template <typename Extension>
>@@ -103,21 +106,6 @@ class RtpPacket {
>   template <typename Extension>
>   bool ReserveExtension();
> 
>-  // Following 4 helpers identify rtp header extension by |id| negotiated with
>-  // remote peer and written in an rtp packet.
>-  bool HasRawExtension(int id) const;
>-
>-  // Returns place where extension with |id| is stored.
>-  // Returns empty arrayview if extension is not present.
>-  rtc::ArrayView<const uint8_t> GetRawExtension(int id) const;
>-
>-  // Allocates and store header extension. Returns true on success.
>-  bool SetRawExtension(int id, rtc::ArrayView<const uint8_t> data);
>-
>-  // Allocates and returns place to store rtp header extension.
>-  // Returns empty arrayview on failure.
>-  rtc::ArrayView<uint8_t> AllocateRawExtension(int id, size_t length);
>-
>   // Reserve size_bytes for payload. Returns nullptr on failure.
>   uint8_t* SetPayloadSize(size_t size_bytes);
>   // Same as SetPayloadSize but doesn't guarantee to keep current payload.
>@@ -139,12 +127,16 @@ class RtpPacket {
>   // Returns view of the raw extension or empty view on failure.
>   rtc::ArrayView<const uint8_t> FindExtension(ExtensionType type) const;
> 
>+  // Allocates and returns place to store rtp header extension.
>+  // Returns empty arrayview on failure.
>+  rtc::ArrayView<uint8_t> AllocateRawExtension(int id, size_t length);
>+
>   // Find or allocate an extension |type|. Returns view of size |length|
>   // to write raw extension to or an empty view on failure.
>   rtc::ArrayView<uint8_t> AllocateExtension(ExtensionType type, size_t length);
> 
>-  uint8_t* WriteAt(size_t offset);
>-  void WriteAt(size_t offset, uint8_t byte);
>+  uint8_t* WriteAt(size_t offset) { return buffer_.data() + offset; }
>+  void WriteAt(size_t offset, uint8_t byte) { buffer_.data()[offset] = byte; }
> 
>   // Header.
>   bool marker_;
>@@ -157,7 +149,7 @@ class RtpPacket {
>   size_t payload_size_;
> 
>   ExtensionInfo extension_entries_[kMaxExtensionHeaders];
>-  uint16_t extensions_size_ = 0;  // Unaligned.
>+  size_t extensions_size_ = 0;  // Unaligned.
>   rtc::CopyOnWriteBuffer buffer_;
> };
> 
>@@ -182,7 +174,7 @@ bool RtpPacket::SetExtension(Values... values) {
>   auto buffer = AllocateExtension(Extension::kId, value_size);
>   if (buffer.empty())
>     return false;
>-  return Extension::Write(buffer.data(), values...);
>+  return Extension::Write(buffer, values...);
> }
> 
> template <typename Extension>
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc
>index 887406ad04c..62f1c824dc8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_history.cc
>@@ -14,6 +14,7 @@
> #include <limits>
> #include <utility>
> 
>+#include "absl/memory/memory.h"
> #include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>@@ -21,218 +22,269 @@
> 
> namespace webrtc {
> namespace {
>+// Min packet size for BestFittingPacket() to honor.
> constexpr size_t kMinPacketRequestBytes = 50;
>-// Don't overwrite a packet within one second, or three RTTs, after transmission
>-// whichever is larger. Instead try to dynamically expand history.
>-constexpr int64_t kMinPacketDurationMs = 1000;
>-constexpr int kMinPacketDurationRtt = 3;
>+
>+// Utility function to get the absolute difference in size between the provided
>+// target size and the size of packet.
>+size_t SizeDiff(const std::unique_ptr<RtpPacketToSend>& packet, size_t size) {
>+  size_t packet_size = packet->size();
>+  if (packet_size > size) {
>+    return packet_size - size;
>+  }
>+  return size - packet_size;
>+}
> }  // namespace
>+
> constexpr size_t RtpPacketHistory::kMaxCapacity;
>+constexpr int64_t RtpPacketHistory::kMinPacketDurationMs;
>+constexpr int RtpPacketHistory::kMinPacketDurationRtt;
>+constexpr int RtpPacketHistory::kPacketCullingDelayFactor;
>+
>+RtpPacketHistory::PacketState::PacketState() = default;
>+RtpPacketHistory::PacketState::PacketState(const PacketState&) = default;
>+RtpPacketHistory::PacketState::~PacketState() = default;
>+
>+RtpPacketHistory::StoredPacket::StoredPacket() = default;
>+RtpPacketHistory::StoredPacket::StoredPacket(StoredPacket&&) = default;
>+RtpPacketHistory::StoredPacket& RtpPacketHistory::StoredPacket::operator=(
>+    RtpPacketHistory::StoredPacket&&) = default;
>+RtpPacketHistory::StoredPacket::~StoredPacket() = default;
> 
> RtpPacketHistory::RtpPacketHistory(Clock* clock)
>-    : clock_(clock), store_(false), prev_index_(0), rtt_ms_(-1) {}
>+    : clock_(clock),
>+      number_to_store_(0),
>+      mode_(StorageMode::kDisabled),
>+      rtt_ms_(-1) {}
> 
> RtpPacketHistory::~RtpPacketHistory() {}
> 
>-void RtpPacketHistory::SetStorePacketsStatus(bool enable,
>-                                             uint16_t number_to_store) {
>-  rtc::CritScope cs(&critsect_);
>-  if (enable) {
>-    if (store_) {
>-      RTC_LOG(LS_WARNING)
>-          << "Purging packet history in order to re-set status.";
>-      Free();
>-    }
>-    RTC_DCHECK(!store_);
>-    Allocate(number_to_store);
>-  } else {
>-    Free();
>-  }
>-}
>-
>-void RtpPacketHistory::Allocate(size_t number_to_store) {
>-  RTC_DCHECK_GT(number_to_store, 0);
>+void RtpPacketHistory::SetStorePacketsStatus(StorageMode mode,
>+                                             size_t number_to_store) {
>   RTC_DCHECK_LE(number_to_store, kMaxCapacity);
>-  store_ = true;
>-  stored_packets_.resize(number_to_store);
>-}
>-
>-void RtpPacketHistory::Free() {
>-  if (!store_) {
>-    return;
>+  rtc::CritScope cs(&lock_);
>+  if (mode != StorageMode::kDisabled && mode_ != StorageMode::kDisabled) {
>+    RTC_LOG(LS_WARNING) << "Purging packet history in order to re-set status.";
>   }
>+  Reset();
>+  mode_ = mode;
>+  number_to_store_ = std::min(kMaxCapacity, number_to_store);
>+}
> 
>-  stored_packets_.clear();
>-
>-  store_ = false;
>-  prev_index_ = 0;
>+RtpPacketHistory::StorageMode RtpPacketHistory::GetStorageMode() const {
>+  rtc::CritScope cs(&lock_);
>+  return mode_;
> }
> 
>-bool RtpPacketHistory::StorePackets() const {
>-  rtc::CritScope cs(&critsect_);
>-  return store_;
>+void RtpPacketHistory::SetRtt(int64_t rtt_ms) {
>+  rtc::CritScope cs(&lock_);
>+  RTC_DCHECK_GE(rtt_ms, 0);
>+  rtt_ms_ = rtt_ms;
> }
> 
> void RtpPacketHistory::PutRtpPacket(std::unique_ptr<RtpPacketToSend> packet,
>                                     StorageType type,
>-                                    bool sent) {
>+                                    absl::optional<int64_t> send_time_ms) {
>   RTC_DCHECK(packet);
>-  rtc::CritScope cs(&critsect_);
>-  if (!store_) {
>+  rtc::CritScope cs(&lock_);
>+  int64_t now_ms = clock_->TimeInMilliseconds();
>+  if (mode_ == StorageMode::kDisabled) {
>     return;
>   }
> 
>-  int64_t now_ms = clock_->TimeInMilliseconds();
>-
>-  // If index we're about to overwrite contains a packet that has not
>-  // yet been sent (probably pending in paced sender), or if the send time is
>-  // less than 3 round trip times ago, expand the buffer to avoid overwriting
>-  // valid data.
>-  StoredPacket* stored_packet = &stored_packets_[prev_index_];
>-  int64_t packet_duration_ms =
>-      std::max(kMinPacketDurationRtt * rtt_ms_, kMinPacketDurationMs);
>-  if (stored_packet->packet &&
>-      (stored_packet->send_time == 0 ||
>-       (rtt_ms_ >= 0 &&
>-        now_ms - stored_packet->send_time <= packet_duration_ms))) {
>-    size_t current_size = stored_packets_.size();
>-    if (current_size < kMaxCapacity) {
>-      size_t expanded_size = std::max(current_size * 3 / 2, current_size + 1);
>-      expanded_size = std::min(expanded_size, kMaxCapacity);
>-      Allocate(expanded_size);
>-      // Causes discontinuity, but that's OK-ish. FindSeqNum() will still work,
>-      // but may be slower - at least until buffer has wrapped around once.
>-      prev_index_ = current_size;
>-      stored_packet = &stored_packets_[prev_index_];
>-    }
>-  }
>+  CullOldPackets(now_ms);
> 
>   // Store packet.
>-  if (packet->capture_time_ms() <= 0)
>-    packet->set_capture_time_ms(now_ms);
>-  stored_packet->sequence_number = packet->SequenceNumber();
>-  stored_packet->send_time = sent ? now_ms : 0;
>-  stored_packet->storage_type = type;
>-  stored_packet->has_been_retransmitted = false;
>-  stored_packet->packet = std::move(packet);
>-
>-  prev_index_ = (prev_index_ + 1) % stored_packets_.size();
>-}
>+  const uint16_t rtp_seq_no = packet->SequenceNumber();
>+  StoredPacket& stored_packet = packet_history_[rtp_seq_no];
>+  RTC_DCHECK(stored_packet.packet == nullptr);
>+  stored_packet.packet = std::move(packet);
> 
>-bool RtpPacketHistory::HasRtpPacket(uint16_t sequence_number) const {
>-  rtc::CritScope cs(&critsect_);
>-  if (!store_) {
>-    return false;
>+  if (stored_packet.packet->capture_time_ms() <= 0) {
>+    stored_packet.packet->set_capture_time_ms(now_ms);
>   }
>+  stored_packet.send_time_ms = send_time_ms;
>+  stored_packet.storage_type = type;
>+  stored_packet.times_retransmitted = 0;
> 
>-  int unused_index = 0;
>-  return FindSeqNum(sequence_number, &unused_index);
>+  if (!start_seqno_) {
>+    start_seqno_ = rtp_seq_no;
>+  }
> }
> 
> std::unique_ptr<RtpPacketToSend> RtpPacketHistory::GetPacketAndSetSendTime(
>     uint16_t sequence_number,
>-    int64_t min_elapsed_time_ms,
>-    bool retransmit) {
>-  rtc::CritScope cs(&critsect_);
>-  if (!store_) {
>+    bool verify_rtt) {
>+  rtc::CritScope cs(&lock_);
>+  if (mode_ == StorageMode::kDisabled) {
>     return nullptr;
>   }
> 
>-  int index = 0;
>-  if (!FindSeqNum(sequence_number, &index)) {
>-    RTC_LOG(LS_WARNING) << "No match for getting seqNum " << sequence_number;
>+  int64_t now_ms = clock_->TimeInMilliseconds();
>+  StoredPacketIterator rtp_it = packet_history_.find(sequence_number);
>+  if (rtp_it == packet_history_.end()) {
>     return nullptr;
>   }
>-  RTC_DCHECK_EQ(sequence_number,
>-                stored_packets_[index].packet->SequenceNumber());
> 
>-  // Verify elapsed time since last retrieve, but only for retransmissions and
>-  // always send packet upon first retransmission request.
>-  int64_t now = clock_->TimeInMilliseconds();
>-  if (min_elapsed_time_ms > 0 && retransmit &&
>-      stored_packets_[index].has_been_retransmitted &&
>-      ((now - stored_packets_[index].send_time) < min_elapsed_time_ms)) {
>+  StoredPacket& packet = rtp_it->second;
>+  if (verify_rtt && !VerifyRtt(rtp_it->second, now_ms)) {
>     return nullptr;
>   }
> 
>-  if (retransmit) {
>-    if (stored_packets_[index].storage_type == kDontRetransmit) {
>-      // No bytes copied since this packet shouldn't be retransmitted.
>-      return nullptr;
>-    }
>-    stored_packets_[index].has_been_retransmitted = true;
>+  if (packet.send_time_ms) {
>+    ++packet.times_retransmitted;
>   }
>-  stored_packets_[index].send_time = clock_->TimeInMilliseconds();
>-  return GetPacket(index);
>-}
> 
>-std::unique_ptr<RtpPacketToSend> RtpPacketHistory::GetPacket(int index) const {
>-  const RtpPacketToSend& stored = *stored_packets_[index].packet;
>-  return std::unique_ptr<RtpPacketToSend>(new RtpPacketToSend(stored));
>+  // Update send-time and return copy of packet instance.
>+  packet.send_time_ms = now_ms;
>+
>+  if (packet.storage_type == StorageType::kDontRetransmit) {
>+    // Non retransmittable packet, so call must come from paced sender.
>+    // Remove from history and return actual packet instance.
>+    return RemovePacket(rtp_it);
>+  }
>+  return absl::make_unique<RtpPacketToSend>(*packet.packet);
> }
> 
>-std::unique_ptr<RtpPacketToSend> RtpPacketHistory::GetBestFittingPacket(
>-    size_t packet_length) const {
>-  rtc::CritScope cs(&critsect_);
>-  if (!store_)
>-    return nullptr;
>-  int index = FindBestFittingPacket(packet_length);
>-  if (index < 0)
>-    return nullptr;
>-  return GetPacket(index);
>+absl::optional<RtpPacketHistory::PacketState> RtpPacketHistory::GetPacketState(
>+    uint16_t sequence_number,
>+    bool verify_rtt) const {
>+  rtc::CritScope cs(&lock_);
>+  if (mode_ == StorageMode::kDisabled) {
>+    return absl::nullopt;
>+  }
>+
>+  auto rtp_it = packet_history_.find(sequence_number);
>+  if (rtp_it == packet_history_.end()) {
>+    return absl::nullopt;
>+  }
>+
>+  if (verify_rtt && !VerifyRtt(rtp_it->second, clock_->TimeInMilliseconds())) {
>+    return absl::nullopt;
>+  }
>+
>+  return StoredPacketToPacketState(rtp_it->second);
> }
> 
>-bool RtpPacketHistory::FindSeqNum(uint16_t sequence_number, int* index) const {
>-  if (prev_index_ > 0) {
>-    *index = prev_index_ - 1;
>-  } else {
>-    *index = stored_packets_.size() - 1;  // Wrap.
>+bool RtpPacketHistory::VerifyRtt(const RtpPacketHistory::StoredPacket& packet,
>+                                 int64_t now_ms) const {
>+  if (packet.send_time_ms) {
>+    // Send-time already set, this check must be for a retransmission.
>+    if (packet.times_retransmitted > 0 &&
>+        now_ms < *packet.send_time_ms + rtt_ms_) {
>+      // This packet has already been retransmitted once, and the time since
>+      // that even is lower than on RTT. Ignore request as this packet is
>+      // likely already in the network pipe.
>+      return false;
>+    }
>   }
>-  uint16_t temp_sequence_number = stored_packets_[*index].sequence_number;
> 
>-  int idx = *index - (temp_sequence_number - sequence_number);
>-  if (idx >= 0 && idx < static_cast<int>(stored_packets_.size())) {
>-    *index = idx;
>-    temp_sequence_number = stored_packets_[*index].sequence_number;
>+  return true;
>+}
>+
>+std::unique_ptr<RtpPacketToSend> RtpPacketHistory::GetBestFittingPacket(
>+    size_t packet_length) const {
>+  // TODO(sprang): Make this smarter, taking retransmit count etc into account.
>+  rtc::CritScope cs(&lock_);
>+  if (packet_length < kMinPacketRequestBytes || packet_history_.empty()) {
>+    return nullptr;
>   }
> 
>-  if (temp_sequence_number != sequence_number) {
>-    // We did not found a match, search all.
>-    for (uint16_t m = 0; m < stored_packets_.size(); m++) {
>-      if (stored_packets_[m].sequence_number == sequence_number) {
>-        *index = m;
>-        temp_sequence_number = stored_packets_[*index].sequence_number;
>+  size_t min_diff = std::numeric_limits<size_t>::max();
>+  RtpPacketToSend* best_packet = nullptr;
>+  for (auto& it : packet_history_) {
>+    size_t diff = SizeDiff(it.second.packet, packet_length);
>+    if (!min_diff || diff < min_diff) {
>+      min_diff = diff;
>+      best_packet = it.second.packet.get();
>+      if (diff == 0) {
>         break;
>       }
>     }
>   }
>-  return temp_sequence_number == sequence_number &&
>-         stored_packets_[*index].packet;
>+
>+  return absl::make_unique<RtpPacketToSend>(*best_packet);
> }
> 
>-int RtpPacketHistory::FindBestFittingPacket(size_t size) const {
>-  if (size < kMinPacketRequestBytes || stored_packets_.empty())
>-    return -1;
>-  size_t min_diff = std::numeric_limits<size_t>::max();
>-  int best_index = -1;  // Returned unchanged if we don't find anything.
>-  for (size_t i = 0; i < stored_packets_.size(); ++i) {
>-    if (!stored_packets_[i].packet)
>+void RtpPacketHistory::Reset() {
>+  packet_history_.clear();
>+  start_seqno_.reset();
>+}
>+
>+void RtpPacketHistory::CullOldPackets(int64_t now_ms) {
>+  int64_t packet_duration_ms =
>+      std::max(kMinPacketDurationRtt * rtt_ms_, kMinPacketDurationMs);
>+  while (!packet_history_.empty()) {
>+    auto stored_packet_it = packet_history_.find(*start_seqno_);
>+    RTC_DCHECK(stored_packet_it != packet_history_.end());
>+
>+    if (packet_history_.size() >= kMaxCapacity) {
>+      // We have reached the absolute max capacity, remove one packet
>+      // unconditionally.
>+      RemovePacket(stored_packet_it);
>       continue;
>-    size_t stored_size = stored_packets_[i].packet->size();
>-    size_t diff =
>-        (stored_size > size) ? (stored_size - size) : (size - stored_size);
>-    if (diff < min_diff) {
>-      min_diff = diff;
>-      best_index = static_cast<int>(i);
>+    }
>+
>+    const StoredPacket& stored_packet = stored_packet_it->second;
>+    if (!stored_packet.send_time_ms) {
>+      // Don't remove packets that have not been sent.
>+      return;
>+    }
>+
>+    if (*stored_packet.send_time_ms + packet_duration_ms > now_ms) {
>+      // Don't cull packets too early to avoid failed retransmission requests.
>+      return;
>+    }
>+
>+    if (packet_history_.size() >= number_to_store_ ||
>+        (mode_ == StorageMode::kStoreAndCull &&
>+         *stored_packet.send_time_ms +
>+                 (packet_duration_ms * kPacketCullingDelayFactor) <=
>+             now_ms)) {
>+      // Too many packets in history, or this packet has timed out. Remove it
>+      // and continue.
>+      RemovePacket(stored_packet_it);
>+    } else {
>+      // No more packets can be removed right now.
>+      return;
>     }
>   }
>-  return best_index;
> }
> 
>-void RtpPacketHistory::SetRtt(int64_t rtt_ms) {
>-  rtc::CritScope cs(&critsect_);
>-  RTC_DCHECK_GE(rtt_ms, 0);
>-  rtt_ms_ = rtt_ms;
>+std::unique_ptr<RtpPacketToSend> RtpPacketHistory::RemovePacket(
>+    StoredPacketIterator packet_it) {
>+  // Move the packet out from the StoredPacket container.
>+  std::unique_ptr<RtpPacketToSend> rtp_packet =
>+      std::move(packet_it->second.packet);
>+  // Erase the packet from the map, and capture iterator to the next one.
>+  StoredPacketIterator next_it = packet_history_.erase(packet_it);
>+
>+  // |next_it| now points to the next element, or to the end. If the end,
>+  // check if we can wrap around.
>+  if (next_it == packet_history_.end()) {
>+    next_it = packet_history_.begin();
>+  }
>+
>+  // Update |start_seq_no| to the new oldest item.
>+  if (next_it != packet_history_.end()) {
>+    start_seqno_ = next_it->first;
>+  } else {
>+    start_seqno_.reset();
>+  }
>+
>+  return rtp_packet;
>+}
>+
>+RtpPacketHistory::PacketState RtpPacketHistory::StoredPacketToPacketState(
>+    const RtpPacketHistory::StoredPacket& stored_packet) {
>+  RtpPacketHistory::PacketState state;
>+  state.rtp_sequence_number = stored_packet.packet->SequenceNumber();
>+  state.send_time_ms = stored_packet.send_time_ms;
>+  state.capture_time_ms = stored_packet.packet->capture_time_ms();
>+  state.ssrc = stored_packet.packet->Ssrc();
>+  state.payload_size = stored_packet.packet->size();
>+  state.times_retransmitted = stored_packet.times_retransmitted;
>+  return state;
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h
>index 91134bf117f..433da0e7eee 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_history.h
>@@ -11,6 +11,7 @@
> #ifndef MODULES_RTP_RTCP_SOURCE_RTP_PACKET_HISTORY_H_
> #define MODULES_RTP_RTCP_SOURCE_RTP_PACKET_HISTORY_H_
> 
>+#include <map>
> #include <memory>
> #include <vector>
> 
>@@ -18,7 +19,6 @@
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/thread_annotations.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -27,62 +27,120 @@ class RtpPacketToSend;
> 
> class RtpPacketHistory {
>  public:
>+  enum class StorageMode {
>+    kDisabled,     // Don't store any packets.
>+    kStore,        // Store and keep at least |number_to_store| packets.
>+    kStoreAndCull  // Store up to |number_to_store| packets, but try to remove
>+                   // packets as they time out or as signaled as received.
>+  };
>+
>+  // Snapshot indicating the state of a packet in the history.
>+  struct PacketState {
>+    PacketState();
>+    PacketState(const PacketState&);
>+    ~PacketState();
>+
>+    uint16_t rtp_sequence_number = 0;
>+    absl::optional<int64_t> send_time_ms;
>+    int64_t capture_time_ms = 0;
>+    uint32_t ssrc = 0;
>+    size_t payload_size = 0;
>+    // Number of times RE-transmitted, ie not including the first transmission.
>+    size_t times_retransmitted = 0;
>+  };
>+
>+  // Maximum number of packets we ever allow in the history.
>   static constexpr size_t kMaxCapacity = 9600;
>+  // Don't remove packets within max(1000ms, 3x RTT).
>+  static constexpr int64_t kMinPacketDurationMs = 1000;
>+  static constexpr int kMinPacketDurationRtt = 3;
>+  // With kStoreAndCull, always remove packets after 3x max(1000ms, 3x rtt).
>+  static constexpr int kPacketCullingDelayFactor = 3;
>+
>   explicit RtpPacketHistory(Clock* clock);
>   ~RtpPacketHistory();
> 
>-  void SetStorePacketsStatus(bool enable, uint16_t number_to_store);
>-  bool StorePackets() const;
>-
>-  void PutRtpPacket(std::unique_ptr<RtpPacketToSend> packet,
>-                    StorageType type,
>-                    bool sent);
>+  // Set/get storage mode. Note that setting the state will clear the history,
>+  // even if setting the same state as is currently used.
>+  void SetStorePacketsStatus(StorageMode mode, size_t number_to_store);
>+  StorageMode GetStorageMode() const;
> 
>   // Set RTT, used to avoid premature retransmission and to prevent over-writing
>   // a packet in the history before we are reasonably sure it has been received.
>   void SetRtt(int64_t rtt_ms);
> 
>+  // If |send_time| is set, packet was sent without using pacer, so state will
>+  // be set accordingly.
>+  void PutRtpPacket(std::unique_ptr<RtpPacketToSend> packet,
>+                    StorageType type,
>+                    absl::optional<int64_t> send_time_ms);
>+
>   // Gets stored RTP packet corresponding to the input |sequence number|.
>-  // Returns nullptr if packet is not found.
>-  // |min_elapsed_time_ms| is the minimum time that must have elapsed since
>-  // the last time the packet was resent (parameter is ignored if set to zero).
>-  // If the packet is found but the minimum time has not elapsed, returns
>-  // nullptr.
>+  // Returns nullptr if packet is not found. If |verify_rtt| is true, doesn't
>+  // return packet that was (re)sent too recently.
>   std::unique_ptr<RtpPacketToSend> GetPacketAndSetSendTime(
>       uint16_t sequence_number,
>-      int64_t min_elapsed_time_ms,
>-      bool retransmit);
>+      bool verify_rtt);
> 
>+  // Similar to GetPacketAndSetSendTime(), but only returns a snapshot of the
>+  // current state for packet, and never updates internal state.
>+  absl::optional<PacketState> GetPacketState(uint16_t sequence_number,
>+                                             bool verify_rtt) const;
>+
>+  // Get the packet (if any) from the history, with size closest to
>+  // |packet_size|. The exact size of the packet is not guaranteed.
>   std::unique_ptr<RtpPacketToSend> GetBestFittingPacket(
>       size_t packet_size) const;
> 
>-  bool HasRtpPacket(uint16_t sequence_number) const;
>-
>  private:
>   struct StoredPacket {
>-    uint16_t sequence_number = 0;
>-    int64_t send_time = 0;
>+    StoredPacket();
>+    StoredPacket(StoredPacket&&);
>+    StoredPacket& operator=(StoredPacket&&);
>+    ~StoredPacket();
>+
>+    // The time of last transmission, including retransmissions.
>+    absl::optional<int64_t> send_time_ms;
>+
>+    // Number of times RE-transmitted, ie excluding the first transmission.
>+    size_t times_retransmitted = 0;
>+
>+    // Storing a packet with |storage_type| = kDontRetransmit indicates this is
>+    // only used as temporary storage until sent by the pacer sender.
>     StorageType storage_type = kDontRetransmit;
>-    bool has_been_retransmitted = false;
> 
>+    // The actual packet.
>     std::unique_ptr<RtpPacketToSend> packet;
>   };
> 
>-  std::unique_ptr<RtpPacketToSend> GetPacket(int index) const
>-      RTC_EXCLUSIVE_LOCKS_REQUIRED(critsect_);
>-  void Allocate(size_t number_to_store) RTC_EXCLUSIVE_LOCKS_REQUIRED(critsect_);
>-  void Free() RTC_EXCLUSIVE_LOCKS_REQUIRED(critsect_);
>-  bool FindSeqNum(uint16_t sequence_number, int* index) const
>-      RTC_EXCLUSIVE_LOCKS_REQUIRED(critsect_);
>-  int FindBestFittingPacket(size_t size) const
>-      RTC_EXCLUSIVE_LOCKS_REQUIRED(critsect_);
>-
>-  Clock* clock_;
>-  rtc::CriticalSection critsect_;
>-  bool store_ RTC_GUARDED_BY(critsect_);
>-  size_t prev_index_ RTC_GUARDED_BY(critsect_);
>-  std::vector<StoredPacket> stored_packets_ RTC_GUARDED_BY(critsect_);
>-  int64_t rtt_ms_ RTC_GUARDED_BY(critsect_);
>+  using StoredPacketIterator = std::map<uint16_t, StoredPacket>::iterator;
>+
>+  // Helper method used by GetPacketAndSetSendTime() and GetPacketState() to
>+  // check if packet has too recently been sent.
>+  bool VerifyRtt(const StoredPacket& packet, int64_t now_ms) const
>+      RTC_EXCLUSIVE_LOCKS_REQUIRED(lock_);
>+  void Reset() RTC_EXCLUSIVE_LOCKS_REQUIRED(lock_);
>+  void CullOldPackets(int64_t now_ms) RTC_EXCLUSIVE_LOCKS_REQUIRED(lock_);
>+  // Removes the packet from the history, and context/mapping that has been
>+  // stored. Returns the RTP packet instance contained within the StoredPacket.
>+  std::unique_ptr<RtpPacketToSend> RemovePacket(StoredPacketIterator packet)
>+      RTC_EXCLUSIVE_LOCKS_REQUIRED(lock_);
>+  static PacketState StoredPacketToPacketState(
>+      const StoredPacket& stored_packet);
>+
>+  Clock* const clock_;
>+  rtc::CriticalSection lock_;
>+  size_t number_to_store_ RTC_GUARDED_BY(lock_);
>+  StorageMode mode_ RTC_GUARDED_BY(lock_);
>+  int64_t rtt_ms_ RTC_GUARDED_BY(lock_);
>+
>+  // Map from rtp sequence numbers to stored packet.
>+  std::map<uint16_t, StoredPacket> packet_history_ RTC_GUARDED_BY(lock_);
>+
>+  // The earliest packet in the history. This might not be the lowest sequence
>+  // number, in case there is a wraparound.
>+  absl::optional<uint16_t> start_seqno_ RTC_GUARDED_BY(lock_);
>+
>   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RtpPacketHistory);
> };
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
>index cdfd3a0a25e..ab5aeb0be62 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.cc
>@@ -17,14 +17,22 @@
> #include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
> #include "system_wrappers/include/clock.h"
> #include "test/gtest.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
>+namespace {
>+// Set a high sequence number so we'll suffer a wrap-around.
>+constexpr uint16_t kStartSeqNum = 65534u;
>+
>+// Utility method for truncating sequence numbers to uint16.
>+uint16_t To16u(size_t sequence_number) {
>+  return static_cast<uint16_t>(sequence_number & 0xFFFF);
>+}
>+}  // namespace
>+
>+using StorageMode = RtpPacketHistory::StorageMode;
> 
> class RtpPacketHistoryTest : public ::testing::Test {
>  protected:
>-  static constexpr uint16_t kSeqNum = 88;
>-
>   RtpPacketHistoryTest() : fake_clock_(123456), hist_(&fake_clock_) {}
> 
>   SimulatedClock fake_clock_;
>@@ -40,256 +48,444 @@ class RtpPacketHistoryTest : public ::testing::Test {
> };
> 
> TEST_F(RtpPacketHistoryTest, SetStoreStatus) {
>-  EXPECT_FALSE(hist_.StorePackets());
>-  hist_.SetStorePacketsStatus(true, 10);
>-  EXPECT_TRUE(hist_.StorePackets());
>-  hist_.SetStorePacketsStatus(false, 0);
>-  EXPECT_FALSE(hist_.StorePackets());
>+  EXPECT_EQ(StorageMode::kDisabled, hist_.GetStorageMode());
>+  hist_.SetStorePacketsStatus(StorageMode::kStore, 10);
>+  EXPECT_EQ(StorageMode::kStore, hist_.GetStorageMode());
>+  hist_.SetStorePacketsStatus(StorageMode::kStoreAndCull, 10);
>+  EXPECT_EQ(StorageMode::kStoreAndCull, hist_.GetStorageMode());
>+  hist_.SetStorePacketsStatus(StorageMode::kDisabled, 0);
>+  EXPECT_EQ(StorageMode::kDisabled, hist_.GetStorageMode());
>+}
>+
>+TEST_F(RtpPacketHistoryTest, ClearsHistoryAfterSetStoreStatus) {
>+  hist_.SetStorePacketsStatus(StorageMode::kStore, 10);
>+  // Store a packet, but with send-time. It should then not be removed.
>+  hist_.PutRtpPacket(CreateRtpPacket(kStartSeqNum), kAllowRetransmission,
>+                     absl::nullopt);
>+  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum, false));
>+
>+  // Changing store status, even to the current one, will clear the history.
>+  hist_.SetStorePacketsStatus(StorageMode::kStore, 10);
>+  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum, false));
>+}
>+
>+TEST_F(RtpPacketHistoryTest, StartSeqResetAfterReset) {
>+  hist_.SetStorePacketsStatus(StorageMode::kStoreAndCull, 10);
>+  // Store a packet, but with send-time. It should then not be removed.
>+  hist_.PutRtpPacket(CreateRtpPacket(kStartSeqNum), kAllowRetransmission,
>+                     absl::nullopt);
>+  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum, false));
>+
>+  // Changing store status, to clear the history.
>+  hist_.SetStorePacketsStatus(StorageMode::kStoreAndCull, 10);
>+  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum, false));
>+
>+  // Add a new packet.
>+  hist_.PutRtpPacket(CreateRtpPacket(kStartSeqNum + 1), kAllowRetransmission,
>+                     absl::nullopt);
>+  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum + 1, false));
>+
>+  // Advance time past where packet expires.
>+  fake_clock_.AdvanceTimeMilliseconds(
>+      RtpPacketHistory::kPacketCullingDelayFactor *
>+      RtpPacketHistory::kMinPacketDurationMs);
>+
>+  // Add one more packet and verify no state left from packet before reset.
>+  hist_.PutRtpPacket(CreateRtpPacket(To16u(kStartSeqNum + 2)),
>+                     kAllowRetransmission, absl::nullopt);
>+  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum, false));
>+  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum + 1, false));
>+  EXPECT_TRUE(hist_.GetPacketState(To16u(kStartSeqNum + 2), false));
> }
> 
> TEST_F(RtpPacketHistoryTest, NoStoreStatus) {
>-  EXPECT_FALSE(hist_.StorePackets());
>-  std::unique_ptr<RtpPacketToSend> packet = CreateRtpPacket(kSeqNum);
>-  hist_.PutRtpPacket(std::move(packet), kAllowRetransmission, false);
>+  EXPECT_EQ(StorageMode::kDisabled, hist_.GetStorageMode());
>+  std::unique_ptr<RtpPacketToSend> packet = CreateRtpPacket(kStartSeqNum);
>+  hist_.PutRtpPacket(std::move(packet), kAllowRetransmission, absl::nullopt);
>   // Packet should not be stored.
>-  EXPECT_FALSE(hist_.GetPacketAndSetSendTime(kSeqNum, 0, false));
>+  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum, false));
> }
> 
> TEST_F(RtpPacketHistoryTest, GetRtpPacket_NotStored) {
>-  hist_.SetStorePacketsStatus(true, 10);
>-  EXPECT_FALSE(hist_.GetPacketAndSetSendTime(0, 0, false));
>+  hist_.SetStorePacketsStatus(StorageMode::kStore, 10);
>+  EXPECT_FALSE(hist_.GetPacketState(0, false));
> }
> 
> TEST_F(RtpPacketHistoryTest, PutRtpPacket) {
>-  hist_.SetStorePacketsStatus(true, 10);
>-  std::unique_ptr<RtpPacketToSend> packet = CreateRtpPacket(kSeqNum);
>+  hist_.SetStorePacketsStatus(StorageMode::kStore, 10);
>+  std::unique_ptr<RtpPacketToSend> packet = CreateRtpPacket(kStartSeqNum);
> 
>-  EXPECT_FALSE(hist_.HasRtpPacket(kSeqNum));
>-  hist_.PutRtpPacket(std::move(packet), kAllowRetransmission, false);
>-  EXPECT_TRUE(hist_.HasRtpPacket(kSeqNum));
>+  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum, false));
>+  hist_.PutRtpPacket(std::move(packet), kAllowRetransmission, absl::nullopt);
>+  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum, false));
> }
> 
> TEST_F(RtpPacketHistoryTest, GetRtpPacket) {
>-  hist_.SetStorePacketsStatus(true, 10);
>+  hist_.SetStorePacketsStatus(StorageMode::kStore, 10);
>   int64_t capture_time_ms = 1;
>-  std::unique_ptr<RtpPacketToSend> packet = CreateRtpPacket(kSeqNum);
>+  std::unique_ptr<RtpPacketToSend> packet = CreateRtpPacket(kStartSeqNum);
>   packet->set_capture_time_ms(capture_time_ms);
>   rtc::CopyOnWriteBuffer buffer = packet->Buffer();
>-  hist_.PutRtpPacket(std::move(packet), kAllowRetransmission, false);
>+  hist_.PutRtpPacket(std::move(packet), kAllowRetransmission, absl::nullopt);
> 
>   std::unique_ptr<RtpPacketToSend> packet_out =
>-      hist_.GetPacketAndSetSendTime(kSeqNum, 0, false);
>+      hist_.GetPacketAndSetSendTime(kStartSeqNum, false);
>   EXPECT_TRUE(packet_out);
>   EXPECT_EQ(buffer, packet_out->Buffer());
>   EXPECT_EQ(capture_time_ms, packet_out->capture_time_ms());
> }
> 
> TEST_F(RtpPacketHistoryTest, NoCaptureTime) {
>-  hist_.SetStorePacketsStatus(true, 10);
>+  hist_.SetStorePacketsStatus(StorageMode::kStore, 10);
>   fake_clock_.AdvanceTimeMilliseconds(1);
>   int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
>-  std::unique_ptr<RtpPacketToSend> packet = CreateRtpPacket(kSeqNum);
>+  std::unique_ptr<RtpPacketToSend> packet = CreateRtpPacket(kStartSeqNum);
>   packet->set_capture_time_ms(-1);
>   rtc::CopyOnWriteBuffer buffer = packet->Buffer();
>-  hist_.PutRtpPacket(std::move(packet), kAllowRetransmission, false);
>+  hist_.PutRtpPacket(std::move(packet), kAllowRetransmission, absl::nullopt);
> 
>   std::unique_ptr<RtpPacketToSend> packet_out =
>-      hist_.GetPacketAndSetSendTime(kSeqNum, 0, false);
>+      hist_.GetPacketAndSetSendTime(kStartSeqNum, false);
>   EXPECT_TRUE(packet_out);
>   EXPECT_EQ(buffer, packet_out->Buffer());
>   EXPECT_EQ(capture_time_ms, packet_out->capture_time_ms());
> }
> 
> TEST_F(RtpPacketHistoryTest, DontRetransmit) {
>-  hist_.SetStorePacketsStatus(true, 10);
>+  hist_.SetStorePacketsStatus(StorageMode::kStore, 10);
>   int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
>-  std::unique_ptr<RtpPacketToSend> packet = CreateRtpPacket(kSeqNum);
>+  std::unique_ptr<RtpPacketToSend> packet = CreateRtpPacket(kStartSeqNum);
>   rtc::CopyOnWriteBuffer buffer = packet->Buffer();
>-  hist_.PutRtpPacket(std::move(packet), kDontRetransmit, false);
>+  hist_.PutRtpPacket(std::move(packet), kDontRetransmit, absl::nullopt);
> 
>+  // Get the packet and verify data.
>   std::unique_ptr<RtpPacketToSend> packet_out;
>-  packet_out = hist_.GetPacketAndSetSendTime(kSeqNum, 0, true);
>-  EXPECT_FALSE(packet_out);
>-
>-  packet_out = hist_.GetPacketAndSetSendTime(kSeqNum, 0, false);
>-  EXPECT_TRUE(packet_out);
>-
>+  packet_out = hist_.GetPacketAndSetSendTime(kStartSeqNum, false);
>+  ASSERT_TRUE(packet_out);
>   EXPECT_EQ(buffer.size(), packet_out->size());
>   EXPECT_EQ(capture_time_ms, packet_out->capture_time_ms());
>+
>+  // Non-retransmittable packets are immediately removed, so getting in again
>+  // should fail.
>+  EXPECT_FALSE(hist_.GetPacketAndSetSendTime(kStartSeqNum, false));
>+}
>+
>+TEST_F(RtpPacketHistoryTest, PacketStateIsCorrect) {
>+  const uint32_t kSsrc = 92384762;
>+  hist_.SetStorePacketsStatus(StorageMode::kStoreAndCull, 10);
>+  std::unique_ptr<RtpPacketToSend> packet = CreateRtpPacket(kStartSeqNum);
>+  packet->SetSsrc(kSsrc);
>+  packet->SetPayloadSize(1234);
>+  const size_t packet_size = packet->size();
>+
>+  hist_.PutRtpPacket(std::move(packet), StorageType::kAllowRetransmission,
>+                     fake_clock_.TimeInMilliseconds());
>+
>+  absl::optional<RtpPacketHistory::PacketState> state =
>+      hist_.GetPacketState(kStartSeqNum, false);
>+  ASSERT_TRUE(state);
>+  EXPECT_EQ(state->rtp_sequence_number, kStartSeqNum);
>+  EXPECT_EQ(state->send_time_ms, fake_clock_.TimeInMilliseconds());
>+  EXPECT_EQ(state->capture_time_ms, fake_clock_.TimeInMilliseconds());
>+  EXPECT_EQ(state->ssrc, kSsrc);
>+  EXPECT_EQ(state->payload_size, packet_size);
>+  EXPECT_EQ(state->times_retransmitted, 0u);
>+
>+  fake_clock_.AdvanceTimeMilliseconds(1);
>+  EXPECT_TRUE(hist_.GetPacketAndSetSendTime(kStartSeqNum, false));
>+
>+  state = hist_.GetPacketState(kStartSeqNum, false);
>+  ASSERT_TRUE(state);
>+  EXPECT_EQ(state->times_retransmitted, 1u);
> }
> 
>-TEST_F(RtpPacketHistoryTest, MinResendTime) {
>+TEST_F(RtpPacketHistoryTest, MinResendTimeWithPacer) {
>   static const int64_t kMinRetransmitIntervalMs = 100;
> 
>-  hist_.SetStorePacketsStatus(true, 10);
>+  hist_.SetStorePacketsStatus(StorageMode::kStore, 10);
>+  hist_.SetRtt(kMinRetransmitIntervalMs);
>   int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
>-  std::unique_ptr<RtpPacketToSend> packet = CreateRtpPacket(kSeqNum);
>+  std::unique_ptr<RtpPacketToSend> packet = CreateRtpPacket(kStartSeqNum);
>   size_t len = packet->size();
>-  hist_.PutRtpPacket(std::move(packet), kAllowRetransmission, false);
>+  hist_.PutRtpPacket(std::move(packet), kAllowRetransmission, absl::nullopt);
> 
>   // First transmission: TimeToSendPacket() call from pacer.
>-  EXPECT_TRUE(hist_.GetPacketAndSetSendTime(kSeqNum, 0, false));
>+  EXPECT_TRUE(hist_.GetPacketAndSetSendTime(kStartSeqNum, false));
> 
>-  fake_clock_.AdvanceTimeMilliseconds(kMinRetransmitIntervalMs);
>-  // Time has elapsed.
>-  std::unique_ptr<RtpPacketToSend> packet_out =
>-      hist_.GetPacketAndSetSendTime(kSeqNum, kMinRetransmitIntervalMs, true);
>-  EXPECT_TRUE(packet_out);
>-  EXPECT_EQ(len, packet_out->size());
>-  EXPECT_EQ(capture_time_ms, packet_out->capture_time_ms());
>+  // First retransmission - allow early retransmission.
>+  fake_clock_.AdvanceTimeMilliseconds(1);
> 
>+  // With pacer there's two calls to history:
>+  // 1) When the NACK request arrived, use GetPacketState() to see if the
>+  //    packet is there and verify RTT constraints. Then we use the ssrc
>+  //    and sequence number to enqueue the retransmission in the pacer
>+  // 2) When the pacer determines that it is time to send the packet, it calls
>+  //    GetPacketAndSetSendTime(). This time we do not need to verify RTT as
>+  //    has that has already been done.
>+  absl::optional<RtpPacketHistory::PacketState> packet_state =
>+      hist_.GetPacketState(kStartSeqNum, /*verify_rtt=*/true);
>+  EXPECT_TRUE(packet_state);
>+  EXPECT_EQ(len, packet_state->payload_size);
>+  EXPECT_EQ(capture_time_ms, packet_state->capture_time_ms);
>+
>+  // Retransmission was allowed, next send it from pacer.
>+  EXPECT_TRUE(hist_.GetPacketAndSetSendTime(kStartSeqNum,
>+                                            /*verify_rtt=*/false));
>+
>+  // Second retransmission - advance time to just before retransmission OK.
>   fake_clock_.AdvanceTimeMilliseconds(kMinRetransmitIntervalMs - 1);
>-  // Time has not elapsed. Packet should be found, but no bytes copied.
>-  EXPECT_TRUE(hist_.HasRtpPacket(kSeqNum));
>-  EXPECT_FALSE(
>-      hist_.GetPacketAndSetSendTime(kSeqNum, kMinRetransmitIntervalMs, true));
>+  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum, /*verify_rtt=*/true));
>+
>+  // Advance time to just after retransmission OK.
>+  fake_clock_.AdvanceTimeMilliseconds(1);
>+  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum, /*verify_rtt=*/true));
>+  EXPECT_TRUE(hist_.GetPacketAndSetSendTime(kStartSeqNum, false));
> }
> 
>-TEST_F(RtpPacketHistoryTest, EarlyFirstResend) {
>+TEST_F(RtpPacketHistoryTest, MinResendTimeWithoutPacer) {
>   static const int64_t kMinRetransmitIntervalMs = 100;
> 
>-  hist_.SetStorePacketsStatus(true, 10);
>+  hist_.SetStorePacketsStatus(StorageMode::kStore, 10);
>+  hist_.SetRtt(kMinRetransmitIntervalMs);
>   int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
>-  std::unique_ptr<RtpPacketToSend> packet = CreateRtpPacket(kSeqNum);
>-  rtc::CopyOnWriteBuffer buffer = packet->Buffer();
>-  hist_.PutRtpPacket(std::move(packet), kAllowRetransmission, false);
>+  std::unique_ptr<RtpPacketToSend> packet = CreateRtpPacket(kStartSeqNum);
>+  size_t len = packet->size();
>+  hist_.PutRtpPacket(std::move(packet), kAllowRetransmission,
>+                     fake_clock_.TimeInMilliseconds());
> 
>-  // First transmission: TimeToSendPacket() call from pacer.
>-  EXPECT_TRUE(hist_.GetPacketAndSetSendTime(kSeqNum, 0, false));
>+  // First retransmission - allow early retransmission.
>+  fake_clock_.AdvanceTimeMilliseconds(1);
>+  packet = hist_.GetPacketAndSetSendTime(kStartSeqNum, true);
>+  EXPECT_TRUE(packet);
>+  EXPECT_EQ(len, packet->size());
>+  EXPECT_EQ(capture_time_ms, packet->capture_time_ms());
> 
>+  // Second retransmission - advance time to just before retransmission OK.
>   fake_clock_.AdvanceTimeMilliseconds(kMinRetransmitIntervalMs - 1);
>-  // Time has not elapsed, but this is the first retransmission request so
>-  // allow anyway.
>-  std::unique_ptr<RtpPacketToSend> packet_out =
>-      hist_.GetPacketAndSetSendTime(kSeqNum, kMinRetransmitIntervalMs, true);
>-  EXPECT_TRUE(packet_out);
>-  EXPECT_EQ(buffer, packet_out->Buffer());
>-  EXPECT_EQ(capture_time_ms, packet_out->capture_time_ms());
>+  EXPECT_FALSE(hist_.GetPacketAndSetSendTime(kStartSeqNum, true));
> 
>-  fake_clock_.AdvanceTimeMilliseconds(kMinRetransmitIntervalMs - 1);
>-  // Time has not elapsed. Packet should be found, but no bytes copied.
>-  EXPECT_TRUE(hist_.HasRtpPacket(kSeqNum));
>-  EXPECT_FALSE(
>-      hist_.GetPacketAndSetSendTime(kSeqNum, kMinRetransmitIntervalMs, true));
>+  // Advance time to just after retransmission OK.
>+  fake_clock_.AdvanceTimeMilliseconds(1);
>+  EXPECT_TRUE(hist_.GetPacketAndSetSendTime(kStartSeqNum, true));
> }
> 
>-TEST_F(RtpPacketHistoryTest, DynamicExpansion) {
>-  hist_.SetStorePacketsStatus(true, 10);
>-
>-  // Add 4 packets, and then send them.
>-  for (int i = 0; i < 4; ++i) {
>-    std::unique_ptr<RtpPacketToSend> packet = CreateRtpPacket(kSeqNum + i);
>-    hist_.PutRtpPacket(std::move(packet), kAllowRetransmission, false);
>+TEST_F(RtpPacketHistoryTest, RemovesOldestSentPacketWhenAtMaxSize) {
>+  const size_t kMaxNumPackets = 10;
>+  hist_.SetStorePacketsStatus(StorageMode::kStore, kMaxNumPackets);
>+
>+  // History does not allow removing packets within kMinPacketDurationMs,
>+  // so in order to test capacity, make sure insertion spans this time.
>+  const int64_t kPacketIntervalMs =
>+      RtpPacketHistory::kMinPacketDurationMs / kMaxNumPackets;
>+
>+  // Add packets until the buffer is full.
>+  for (size_t i = 0; i < kMaxNumPackets; ++i) {
>+    std::unique_ptr<RtpPacketToSend> packet = CreateRtpPacket(kStartSeqNum + i);
>+    // Immediate mark packet as sent.
>+    hist_.PutRtpPacket(std::move(packet), kAllowRetransmission,
>+                       fake_clock_.TimeInMilliseconds());
>+    fake_clock_.AdvanceTimeMilliseconds(kPacketIntervalMs);
>   }
>-  for (int i = 0; i < 4; ++i) {
>-    EXPECT_TRUE(hist_.GetPacketAndSetSendTime(kSeqNum + i, 100, false));
>-  }
>-  fake_clock_.AdvanceTimeMilliseconds(33);
> 
>-  // Add 16 packets, and then send them. History should expand to make this
>-  // work.
>-  for (int i = 4; i < 20; ++i) {
>-    std::unique_ptr<RtpPacketToSend> packet = CreateRtpPacket(kSeqNum + i);
>-    hist_.PutRtpPacket(std::move(packet), kAllowRetransmission, false);
>-  }
>-  for (int i = 4; i < 20; ++i) {
>-    EXPECT_TRUE(hist_.GetPacketAndSetSendTime(kSeqNum + i, 100, false));
>-  }
>+  // First packet should still be there.
>+  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum, false));
> 
>-  fake_clock_.AdvanceTimeMilliseconds(100);
>+  // History is full, oldest one should be overwritten.
>+  std::unique_ptr<RtpPacketToSend> packet =
>+      CreateRtpPacket(To16u(kStartSeqNum + kMaxNumPackets));
>+  hist_.PutRtpPacket(std::move(packet), kAllowRetransmission,
>+                     fake_clock_.TimeInMilliseconds());
> 
>-  // Retransmit last 16 packets.
>-  for (int i = 4; i < 20; ++i) {
>-    EXPECT_TRUE(hist_.GetPacketAndSetSendTime(kSeqNum + i, 100, false));
>+  // Oldest packet should be gone, but packet after than one still present.
>+  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum, false));
>+  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum + 1, false));
>+}
>+
>+TEST_F(RtpPacketHistoryTest, RemovesOldestPacketWhenAtMaxCapacity) {
>+  // Tests the absolute upper bound on number of stored packets. Don't allow
>+  // storing more than this, even if packets have not yet been sent.
>+  const size_t kMaxNumPackets = RtpPacketHistory::kMaxCapacity;
>+  hist_.SetStorePacketsStatus(StorageMode::kStore,
>+                              RtpPacketHistory::kMaxCapacity);
>+
>+  // Add packets until the buffer is full.
>+  for (size_t i = 0; i < kMaxNumPackets; ++i) {
>+    std::unique_ptr<RtpPacketToSend> packet = CreateRtpPacket(kStartSeqNum + i);
>+    // Don't mark packets as sent, preventing them from being removed.
>+    hist_.PutRtpPacket(std::move(packet), kAllowRetransmission, absl::nullopt);
>   }
>+
>+  // First packet should still be there.
>+  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum, false));
>+
>+  // History is full, oldest one should be overwritten.
>+  std::unique_ptr<RtpPacketToSend> packet =
>+      CreateRtpPacket(To16u(kStartSeqNum + kMaxNumPackets));
>+  hist_.PutRtpPacket(std::move(packet), kAllowRetransmission,
>+                     fake_clock_.TimeInMilliseconds());
>+
>+  // Oldest packet should be gone, but packet after than one still present.
>+  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum, false));
>+  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum + 1, false));
> }
> 
>-TEST_F(RtpPacketHistoryTest, FullExpansion) {
>-  static const int kSendSidePacketHistorySize = 600;
>-  hist_.SetStorePacketsStatus(true, kSendSidePacketHistorySize);
>-  for (size_t i = 0; i < RtpPacketHistory::kMaxCapacity + 1; ++i) {
>-    std::unique_ptr<RtpPacketToSend> packet = CreateRtpPacket(kSeqNum + i);
>-    hist_.PutRtpPacket(std::move(packet), kAllowRetransmission, false);
>+TEST_F(RtpPacketHistoryTest, DontRemoveUnsentPackets) {
>+  const size_t kMaxNumPackets = 10;
>+  hist_.SetStorePacketsStatus(StorageMode::kStore, kMaxNumPackets);
>+
>+  // Add packets until the buffer is full.
>+  for (size_t i = 0; i < kMaxNumPackets; ++i) {
>+    // Mark packets as unsent.
>+    hist_.PutRtpPacket(CreateRtpPacket(To16u(kStartSeqNum + i)),
>+                       kAllowRetransmission, absl::nullopt);
>   }
>+  fake_clock_.AdvanceTimeMilliseconds(RtpPacketHistory::kMinPacketDurationMs);
>+
>+  // First packet should still be there.
>+  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum, false));
> 
>-  fake_clock_.AdvanceTimeMilliseconds(100);
>+  // History is full, but old packets not sent, so allow expansion.
>+  hist_.PutRtpPacket(CreateRtpPacket(To16u(kStartSeqNum + kMaxNumPackets)),
>+                     kAllowRetransmission, fake_clock_.TimeInMilliseconds());
>+  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum, false));
> 
>-  // Retransmit all packets currently in buffer.
>-  for (size_t i = 1; i < RtpPacketHistory::kMaxCapacity + 1; ++i) {
>-    EXPECT_TRUE(hist_.GetPacketAndSetSendTime(kSeqNum + i, 100, false));
>+  // Set all packet as sent and advance time past min packet duration time,
>+  // otherwise packets till still be prevented from being removed.
>+  for (size_t i = 0; i <= kMaxNumPackets; ++i) {
>+    EXPECT_TRUE(hist_.GetPacketAndSetSendTime(To16u(kStartSeqNum + i), false));
>   }
>+  fake_clock_.AdvanceTimeMilliseconds(RtpPacketHistory::kMinPacketDurationMs);
>+  // Add a new packet, this means the two oldest ones will be culled.
>+  hist_.PutRtpPacket(CreateRtpPacket(To16u(kStartSeqNum + kMaxNumPackets + 1)),
>+                     kAllowRetransmission, fake_clock_.TimeInMilliseconds());
>+  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum, false));
>+  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum + 1, false));
>+  EXPECT_TRUE(hist_.GetPacketState(To16u(kStartSeqNum + 2), false));
> }
> 
>-TEST_F(RtpPacketHistoryTest, DontExpandIfPacketIsOldEnough) {
>-  const size_t kSendSidePacketHistorySize = 600;
>-  const int64_t kRttMs = 334;
>-  hist_.SetStorePacketsStatus(true, kSendSidePacketHistorySize);
>-  hist_.SetRtt(kRttMs);
>+TEST_F(RtpPacketHistoryTest, DontRemoveTooRecentlyTransmittedPackets) {
>+  // Set size to remove old packets as soon as possible.
>+  hist_.SetStorePacketsStatus(StorageMode::kStore, 1);
> 
>-  // Fill up the buffer with packets.
>-  for (size_t i = 0; i < kSendSidePacketHistorySize; ++i) {
>-    std::unique_ptr<RtpPacketToSend> packet = CreateRtpPacket(kSeqNum + i);
>-    hist_.PutRtpPacket(std::move(packet), kAllowRetransmission, false);
>-    EXPECT_TRUE(hist_.GetPacketAndSetSendTime(kSeqNum + i, 100, false));
>-  }
>+  // Add a packet, marked as send, and advance time to just before removal time.
>+  hist_.PutRtpPacket(CreateRtpPacket(kStartSeqNum), kAllowRetransmission,
>+                     fake_clock_.TimeInMilliseconds());
>+  fake_clock_.AdvanceTimeMilliseconds(RtpPacketHistory::kMinPacketDurationMs -
>+                                      1);
> 
>-  // Move clock forward past expiration time.
>-  fake_clock_.AdvanceTimeMilliseconds(kRttMs * 3 + 1);
>+  // Add a new packet to trigger culling.
>+  hist_.PutRtpPacket(CreateRtpPacket(kStartSeqNum + 1), kAllowRetransmission,
>+                     fake_clock_.TimeInMilliseconds());
>+  // First packet should still be there.
>+  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum, false));
> 
>-  // Insert a new packet and check that the old one for this index has been
>-  // overwritten.
>-  std::unique_ptr<RtpPacketToSend> packet =
>-      CreateRtpPacket(kSeqNum + kSendSidePacketHistorySize);
>-  hist_.PutRtpPacket(std::move(packet), kAllowRetransmission, true);
>-  EXPECT_FALSE(hist_.HasRtpPacket(kSeqNum));
>+  // Advance time to where packet will be eligible for removal and try again.
>+  fake_clock_.AdvanceTimeMilliseconds(1);
>+  hist_.PutRtpPacket(CreateRtpPacket(To16u(kStartSeqNum + 2)),
>+                     kAllowRetransmission, fake_clock_.TimeInMilliseconds());
>+  // First packet should no be gone, but next one still there.
>+  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum, false));
>+  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum + 1, false));
> }
> 
>-TEST_F(RtpPacketHistoryTest, ExpandIfPacketTooRecentlyTransmitted) {
>-  const size_t kSendSidePacketHistorySize = 600;
>-  const int64_t kRttMs = 334;
>-  hist_.SetStorePacketsStatus(true, kSendSidePacketHistorySize);
>+TEST_F(RtpPacketHistoryTest, DontRemoveTooRecentlyTransmittedPacketsHighRtt) {
>+  const int64_t kRttMs = RtpPacketHistory::kMinPacketDurationMs * 2;
>+  const int64_t kPacketTimeoutMs =
>+      kRttMs * RtpPacketHistory::kMinPacketDurationRtt;
>+
>+  // Set size to remove old packets as soon as possible.
>+  hist_.SetStorePacketsStatus(StorageMode::kStore, 1);
>   hist_.SetRtt(kRttMs);
> 
>-  // Fill up the buffer with packets.
>-  for (size_t i = 0; i < kSendSidePacketHistorySize; ++i) {
>-    std::unique_ptr<RtpPacketToSend> packet = CreateRtpPacket(kSeqNum + i);
>-    hist_.PutRtpPacket(std::move(packet), kAllowRetransmission, false);
>-    EXPECT_TRUE(hist_.GetPacketAndSetSendTime(kSeqNum + i, kRttMs, false));
>-  }
>+  // Add a packet, marked as send, and advance time to just before removal time.
>+  hist_.PutRtpPacket(CreateRtpPacket(kStartSeqNum), kAllowRetransmission,
>+                     fake_clock_.TimeInMilliseconds());
>+  fake_clock_.AdvanceTimeMilliseconds(kPacketTimeoutMs - 1);
> 
>-  // Move clock forward to just before expiration time.
>-  fake_clock_.AdvanceTimeMilliseconds(kRttMs * 3);
>+  // Add a new packet to trigger culling.
>+  hist_.PutRtpPacket(CreateRtpPacket(kStartSeqNum + 1), kAllowRetransmission,
>+                     fake_clock_.TimeInMilliseconds());
>+  // First packet should still be there.
>+  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum, false));
> 
>-  // Insert a new packet and verify that the old one for this index still
>-  // exists - ie the buffer has been expanded.
>-  std::unique_ptr<RtpPacketToSend> packet =
>-      CreateRtpPacket(kSeqNum + kSendSidePacketHistorySize);
>-  hist_.PutRtpPacket(std::move(packet), kAllowRetransmission, true);
>-  EXPECT_TRUE(hist_.HasRtpPacket(kSeqNum));
>+  // Advance time to where packet will be eligible for removal and try again.
>+  fake_clock_.AdvanceTimeMilliseconds(1);
>+  hist_.PutRtpPacket(CreateRtpPacket(To16u(kStartSeqNum + 2)),
>+                     kAllowRetransmission, fake_clock_.TimeInMilliseconds());
>+  // First packet should no be gone, but next one still there.
>+  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum, false));
>+  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum + 1, false));
> }
> 
>-TEST_F(RtpPacketHistoryTest, ExpandIfPacketTooRecentlyTransmittedOnFastLink) {
>-  const size_t kSendSidePacketHistorySize = 600;
>-  const int64_t kRttMs = 5;
>-  hist_.SetStorePacketsStatus(true, kSendSidePacketHistorySize);
>+TEST_F(RtpPacketHistoryTest, RemovesOldWithCulling) {
>+  const size_t kMaxNumPackets = 10;
>+  // Enable culling. Even without feedback, this can trigger early removal.
>+  hist_.SetStorePacketsStatus(StorageMode::kStoreAndCull, kMaxNumPackets);
>+
>+  hist_.PutRtpPacket(CreateRtpPacket(kStartSeqNum), kAllowRetransmission,
>+                     fake_clock_.TimeInMilliseconds());
>+
>+  int64_t kMaxPacketDurationMs = RtpPacketHistory::kMinPacketDurationMs *
>+                                 RtpPacketHistory::kPacketCullingDelayFactor;
>+  fake_clock_.AdvanceTimeMilliseconds(kMaxPacketDurationMs - 1);
>+
>+  // First packet should still be there.
>+  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum, false));
>+
>+  // Advance to where packet can be culled, even if buffer is not full.
>+  fake_clock_.AdvanceTimeMilliseconds(1);
>+  hist_.PutRtpPacket(CreateRtpPacket(kStartSeqNum + 1), kAllowRetransmission,
>+                     fake_clock_.TimeInMilliseconds());
>+
>+  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum, false));
>+}
>+
>+TEST_F(RtpPacketHistoryTest, RemovesOldWithCullingHighRtt) {
>+  const size_t kMaxNumPackets = 10;
>+  const int64_t kRttMs = RtpPacketHistory::kMinPacketDurationMs * 2;
>+  // Enable culling. Even without feedback, this can trigger early removal.
>+  hist_.SetStorePacketsStatus(StorageMode::kStoreAndCull, kMaxNumPackets);
>   hist_.SetRtt(kRttMs);
> 
>-  // Fill up the buffer with packets.
>-  for (size_t i = 0; i < kSendSidePacketHistorySize; ++i) {
>-    std::unique_ptr<RtpPacketToSend> packet = CreateRtpPacket(kSeqNum + i);
>-    hist_.PutRtpPacket(std::move(packet), kAllowRetransmission, false);
>-    EXPECT_TRUE(hist_.GetPacketAndSetSendTime(kSeqNum + i, kRttMs, false));
>-  }
>+  hist_.PutRtpPacket(CreateRtpPacket(kStartSeqNum), kAllowRetransmission,
>+                     fake_clock_.TimeInMilliseconds());
> 
>-  // Move clock forward after expiration time based on RTT, but before
>-  // expiration time based on absolute time.
>-  fake_clock_.AdvanceTimeMilliseconds(999);
>+  int64_t kMaxPacketDurationMs = kRttMs *
>+                                 RtpPacketHistory::kMinPacketDurationRtt *
>+                                 RtpPacketHistory::kPacketCullingDelayFactor;
>+  fake_clock_.AdvanceTimeMilliseconds(kMaxPacketDurationMs - 1);
> 
>-  // Insert a new packet and verify that the old one for this index still
>-  // exists - ie the buffer has been expanded.
>-  std::unique_ptr<RtpPacketToSend> packet =
>-      CreateRtpPacket(kSeqNum + kSendSidePacketHistorySize);
>-  hist_.PutRtpPacket(std::move(packet), kAllowRetransmission, true);
>-  EXPECT_TRUE(hist_.HasRtpPacket(kSeqNum));
>+  // First packet should still be there.
>+  EXPECT_TRUE(hist_.GetPacketState(kStartSeqNum, false));
>+
>+  // Advance to where packet can be culled, even if buffer is not full.
>+  fake_clock_.AdvanceTimeMilliseconds(1);
>+  hist_.PutRtpPacket(CreateRtpPacket(kStartSeqNum + 1), kAllowRetransmission,
>+                     fake_clock_.TimeInMilliseconds());
>+
>+  EXPECT_FALSE(hist_.GetPacketState(kStartSeqNum, false));
>+}
>+
>+TEST_F(RtpPacketHistoryTest, GetBestFittingPacket) {
>+  const size_t kTargetSize = 500;
>+  hist_.SetStorePacketsStatus(StorageMode::kStore, 10);
>+
>+  // Add three packets of various sizes.
>+  std::unique_ptr<RtpPacketToSend> packet = CreateRtpPacket(kStartSeqNum);
>+  packet->SetPayloadSize(kTargetSize);
>+  const size_t target_packet_size = packet->size();
>+  hist_.PutRtpPacket(std::move(packet), kAllowRetransmission,
>+                     fake_clock_.TimeInMilliseconds());
>+  packet = CreateRtpPacket(kStartSeqNum + 1);
>+  packet->SetPayloadSize(kTargetSize - 1);
>+  hist_.PutRtpPacket(std::move(packet), kAllowRetransmission,
>+                     fake_clock_.TimeInMilliseconds());
>+  packet = CreateRtpPacket(To16u(kStartSeqNum + 2));
>+  packet->SetPayloadSize(kTargetSize + 1);
>+  hist_.PutRtpPacket(std::move(packet), kAllowRetransmission,
>+                     fake_clock_.TimeInMilliseconds());
>+
>+  EXPECT_EQ(target_packet_size,
>+            hist_.GetBestFittingPacket(target_packet_size)->size());
> }
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_received.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_received.cc
>index 1106e22a24e..6acc9b55d43 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_received.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_received.cc
>@@ -20,6 +20,13 @@ namespace webrtc {
> RtpPacketReceived::RtpPacketReceived() = default;
> RtpPacketReceived::RtpPacketReceived(const ExtensionManager* extensions)
>     : RtpPacket(extensions) {}
>+RtpPacketReceived::RtpPacketReceived(const RtpPacketReceived& packet) = default;
>+RtpPacketReceived::RtpPacketReceived(RtpPacketReceived&& packet) = default;
>+
>+RtpPacketReceived& RtpPacketReceived::operator=(
>+    const RtpPacketReceived& packet) = default;
>+RtpPacketReceived& RtpPacketReceived::operator=(RtpPacketReceived&& packet) =
>+    default;
> 
> RtpPacketReceived::~RtpPacketReceived() {}
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_received.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_received.h
>index 4cf6b23bd7b..86900310e49 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_received.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_received.h
>@@ -22,6 +22,11 @@ class RtpPacketReceived : public RtpPacket {
>  public:
>   RtpPacketReceived();
>   explicit RtpPacketReceived(const ExtensionManager* extensions);
>+  RtpPacketReceived(const RtpPacketReceived& packet);
>+  RtpPacketReceived(RtpPacketReceived&& packet);
>+
>+  RtpPacketReceived& operator=(const RtpPacketReceived& packet);
>+  RtpPacketReceived& operator=(RtpPacketReceived&& packet);
> 
>   ~RtpPacketReceived();
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.cc
>new file mode 100644
>index 00000000000..0153bbe73ac
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.cc
>@@ -0,0 +1,29 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
>+
>+namespace webrtc {
>+
>+RtpPacketToSend::RtpPacketToSend(const ExtensionManager* extensions)
>+    : RtpPacket(extensions) {}
>+RtpPacketToSend::RtpPacketToSend(const ExtensionManager* extensions,
>+                                 size_t capacity)
>+    : RtpPacket(extensions, capacity) {}
>+RtpPacketToSend::RtpPacketToSend(const RtpPacketToSend& packet) = default;
>+RtpPacketToSend::RtpPacketToSend(RtpPacketToSend&& packet) = default;
>+
>+RtpPacketToSend& RtpPacketToSend::operator=(const RtpPacketToSend& packet) =
>+    default;
>+RtpPacketToSend& RtpPacketToSend::operator=(RtpPacketToSend&& packet) = default;
>+
>+RtpPacketToSend::~RtpPacketToSend() = default;
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h
>index c287f0ca295..1ed7dda4782 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h
>@@ -10,6 +10,9 @@
> #ifndef MODULES_RTP_RTCP_SOURCE_RTP_PACKET_TO_SEND_H_
> #define MODULES_RTP_RTCP_SOURCE_RTP_PACKET_TO_SEND_H_
> 
>+#include <vector>
>+
>+#include "api/array_view.h"
> #include "modules/rtp_rtcp/source/rtp_header_extensions.h"
> #include "modules/rtp_rtcp/source/rtp_packet.h"
> 
>@@ -17,19 +20,31 @@ namespace webrtc {
> // Class to hold rtp packet with metadata for sender side.
> class RtpPacketToSend : public RtpPacket {
>  public:
>-  explicit RtpPacketToSend(const ExtensionManager* extensions)
>-      : RtpPacket(extensions) {}
>-  RtpPacketToSend(const RtpPacketToSend& packet) = default;
>-  RtpPacketToSend(const ExtensionManager* extensions, size_t capacity)
>-      : RtpPacket(extensions, capacity) {}
>+  explicit RtpPacketToSend(const ExtensionManager* extensions);
>+  RtpPacketToSend(const ExtensionManager* extensions, size_t capacity);
>+  RtpPacketToSend(const RtpPacketToSend& packet);
>+  RtpPacketToSend(RtpPacketToSend&& packet);
>+
>+  RtpPacketToSend& operator=(const RtpPacketToSend& packet);
>+  RtpPacketToSend& operator=(RtpPacketToSend&& packet);
> 
>-  RtpPacketToSend& operator=(const RtpPacketToSend& packet) = default;
>+  ~RtpPacketToSend();
> 
>   // Time in local time base as close as it can to frame capture time.
>   int64_t capture_time_ms() const { return capture_time_ms_; }
> 
>   void set_capture_time_ms(int64_t time) { capture_time_ms_ = time; }
> 
>+  // Additional data bound to the RTP packet for use in application code,
>+  // outside of WebRTC.
>+  rtc::ArrayView<const uint8_t> application_data() const {
>+    return application_data_;
>+  }
>+
>+  void set_application_data(rtc::ArrayView<const uint8_t> data) {
>+    application_data_.assign(data.begin(), data.end());
>+  }
>+
>   void set_packetization_finish_time_ms(int64_t time) {
>     SetExtension<VideoTimingExtension>(
>         VideoSendTiming::GetDeltaCappedMs(capture_time_ms_, time),
>@@ -56,6 +71,7 @@ class RtpPacketToSend : public RtpPacket {
> 
>  private:
>   int64_t capture_time_ms_ = 0;
>+  std::vector<uint8_t> application_data_;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_unittest.cc
>index 253ce32db00..e19f1914398 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_packet_unittest.cc
>@@ -196,50 +196,27 @@ TEST(RtpPacketTest, TryToCreateWithLongMid) {
>   EXPECT_FALSE(packet.SetExtension<RtpMid>(kLongMid));
> }
> 
>-TEST(RtpPacketTest, CreateWithExtensionsWithoutManager) {
>-  RtpPacketToSend packet(nullptr);
>-  packet.SetPayloadType(kPayloadType);
>-  packet.SetSequenceNumber(kSeqNum);
>-  packet.SetTimestamp(kTimestamp);
>-  packet.SetSsrc(kSsrc);
>-
>-  auto raw = packet.AllocateRawExtension(kTransmissionOffsetExtensionId,
>-                                         TransmissionOffset::kValueSizeBytes);
>-  EXPECT_EQ(raw.size(), TransmissionOffset::kValueSizeBytes);
>-  TransmissionOffset::Write(raw.data(), kTimeOffset);
>-
>-  raw = packet.AllocateRawExtension(kAudioLevelExtensionId,
>-                                    AudioLevel::kValueSizeBytes);
>-  EXPECT_EQ(raw.size(), AudioLevel::kValueSizeBytes);
>-  AudioLevel::Write(raw.data(), kVoiceActive, kAudioLevel);
>-
>-  EXPECT_THAT(kPacketWithTOAndAL,
>-              ElementsAreArray(packet.data(), packet.size()));
>-}
>-
> TEST(RtpPacketTest, CreateWithMaxSizeHeaderExtension) {
>-  const size_t kMaxExtensionSize = 16;
>-  const int kId = 1;
>-  const uint8_t kValue[16] = "123456789abcdef";
>+  const std::string kValue = "123456789abcdef";
>+  RtpPacket::ExtensionManager extensions;
>+  extensions.Register<RtpMid>(1);
>+  extensions.Register<RtpStreamId>(2);
> 
>-  // Write packet with a custom extension.
>-  RtpPacketToSend packet(nullptr);
>-  packet.SetRawExtension(kId, kValue);
>-  // Using different size for same id is not allowed.
>-  EXPECT_TRUE(packet.AllocateRawExtension(kId, kMaxExtensionSize - 1).empty());
>+  RtpPacket packet(&extensions);
>+  EXPECT_TRUE(packet.SetExtension<RtpMid>(kValue));
> 
>   packet.SetPayloadSize(42);
>   // Rewriting allocated extension is allowed.
>-  EXPECT_EQ(packet.AllocateRawExtension(kId, kMaxExtensionSize).size(),
>-            kMaxExtensionSize);
>+  EXPECT_TRUE(packet.SetExtension<RtpMid>(kValue));
>   // Adding another extension after payload is set is not allowed.
>-  EXPECT_TRUE(packet.AllocateRawExtension(kId + 1, kMaxExtensionSize).empty());
>+  EXPECT_FALSE(packet.SetExtension<RtpStreamId>(kValue));
> 
>-  // Read packet with the custom extension.
>-  RtpPacketReceived parsed;
>+  // Read packet with the extension.
>+  RtpPacketReceived parsed(&extensions);
>   EXPECT_TRUE(parsed.Parse(packet.Buffer()));
>-  auto read_raw = parsed.GetRawExtension(kId);
>-  EXPECT_THAT(read_raw, ElementsAreArray(kValue, kMaxExtensionSize));
>+  std::string read;
>+  EXPECT_TRUE(parsed.GetExtension<RtpMid>(&read));
>+  EXPECT_EQ(read, kValue);
> }
> 
> TEST(RtpPacketTest, SetReservedExtensionsAfterPayload) {
>@@ -426,23 +403,6 @@ TEST(RtpPacketTest, ParseWithExtensionDelayed) {
>   EXPECT_EQ(0u, packet.padding_size());
> }
> 
>-TEST(RtpPacketTest, ParseWithoutExtensionManager) {
>-  RtpPacketReceived packet;
>-  EXPECT_TRUE(packet.Parse(kPacketWithTO, sizeof(kPacketWithTO)));
>-
>-  EXPECT_FALSE(packet.HasRawExtension(kAudioLevelExtensionId));
>-  EXPECT_TRUE(packet.GetRawExtension(kAudioLevelExtensionId).empty());
>-
>-  EXPECT_TRUE(packet.HasRawExtension(kTransmissionOffsetExtensionId));
>-
>-  int32_t time_offset = 0;
>-  auto raw_extension = packet.GetRawExtension(kTransmissionOffsetExtensionId);
>-  EXPECT_EQ(raw_extension.size(), TransmissionOffset::kValueSizeBytes);
>-  EXPECT_TRUE(TransmissionOffset::Parse(raw_extension, &time_offset));
>-
>-  EXPECT_EQ(time_offset, kTimeOffset);
>-}
>-
> TEST(RtpPacketTest, ParseDynamicSizeExtension) {
>   // clang-format off
>   const uint8_t kPacket1[] = {
>@@ -493,23 +453,6 @@ TEST(RtpPacketTest, ParseWithMid) {
>   EXPECT_EQ(mid, kMid);
> }
> 
>-TEST(RtpPacketTest, RawExtensionFunctionsAcceptZeroIdAndReturnFalse) {
>-  RtpPacketReceived::ExtensionManager extensions;
>-  RtpPacketReceived packet(&extensions);
>-  // Use ExtensionManager to set kInvalidId to 0 to demonstrate natural way for
>-  // using zero value as a parameter to Packet::*RawExtension functions.
>-  const int kInvalidId = extensions.GetId(TransmissionOffset::kId);
>-  ASSERT_EQ(kInvalidId, 0);
>-
>-  ASSERT_TRUE(packet.Parse(kPacket, sizeof(kPacket)));
>-
>-  EXPECT_FALSE(packet.HasRawExtension(kInvalidId));
>-  EXPECT_THAT(packet.GetRawExtension(kInvalidId), IsEmpty());
>-  const uint8_t kExtension[] = {'e', 'x', 't'};
>-  EXPECT_FALSE(packet.SetRawExtension(kInvalidId, kExtension));
>-  EXPECT_THAT(packet.AllocateRawExtension(kInvalidId, 3), IsEmpty());
>-}
>-
> TEST(RtpPacketTest, CreateAndParseTimingFrameExtension) {
>   // Create a packet with video frame timing extension populated.
>   RtpPacketToSend::ExtensionManager send_extensions;
>@@ -526,7 +469,7 @@ TEST(RtpPacketTest, CreateAndParseTimingFrameExtension) {
>   timing.packetization_finish_delta_ms = 3;
>   timing.pacer_exit_delta_ms = 4;
>   timing.flags =
>-      TimingFrameFlags::kTriggeredByTimer + TimingFrameFlags::kTriggeredBySize;
>+      VideoSendTiming::kTriggeredByTimer | VideoSendTiming::kTriggeredBySize;
> 
>   send_packet.SetExtension<VideoTimingExtension>(timing);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc
>index 60c8c75af75..74a77882616 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_payload_registry.cc
>@@ -12,7 +12,6 @@
> 
> #include <algorithm>
> 
>-#include "common_types.h"  // NOLINT(build/include)
> #include "modules/audio_coding/codecs/audio_format_conversion.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>@@ -31,7 +30,8 @@ bool PayloadIsCompatible(const RtpUtility::Payload& payload,
> bool PayloadIsCompatible(const RtpUtility::Payload& payload,
>                          const VideoCodec& video_codec) {
>   if (!payload.typeSpecific.is_video() ||
>-      _stricmp(payload.name, video_codec.plName) != 0)
>+      _stricmp(payload.name, CodecTypeToPayloadString(video_codec.codecType)) !=
>+          0)
>     return false;
>   // For H264, profiles must match as well.
>   if (video_codec.codecType == kVideoCodecH264) {
>@@ -50,18 +50,11 @@ RtpUtility::Payload CreatePayloadType(const SdpAudioFormat& audio_format) {
> RtpVideoCodecTypes ConvertToRtpVideoCodecType(VideoCodecType type) {
>   switch (type) {
>     case kVideoCodecVP8:
>-      return kRtpVideoVp8;
>     case kVideoCodecVP9:
>-      return kRtpVideoVp9;
>     case kVideoCodecH264:
>-      return kRtpVideoH264;
>-    case kVideoCodecRED:
>-    case kVideoCodecULPFEC:
>-      return kRtpVideoNone;
>-    case kVideoCodecStereo:
>-      return kRtpVideoStereo;
>+      return type;
>     default:
>-      return kRtpVideoGeneric;
>+      return kVideoCodecGeneric;
>   }
> }
> 
>@@ -70,7 +63,7 @@ RtpUtility::Payload CreatePayloadType(const VideoCodec& video_codec) {
>   p.videoCodecType = ConvertToRtpVideoCodecType(video_codec.codecType);
>   if (video_codec.codecType == kVideoCodecH264)
>     p.h264_profile = video_codec.H264().profile;
>-  return {video_codec.plName, PayloadUnion(p)};
>+  return {CodecTypeToPayloadString(video_codec.codecType), PayloadUnion(p)};
> }
> 
> bool IsPayloadTypeValid(int8_t payload_type) {
>@@ -79,15 +72,15 @@ bool IsPayloadTypeValid(int8_t payload_type) {
>   // Sanity check.
>   switch (payload_type) {
>     // Reserved payload types to avoid RTCP conflicts when marker bit is set.
>-    case 64:        //  192 Full INTRA-frame request.
>-    case 72:        //  200 Sender report.
>-    case 73:        //  201 Receiver report.
>-    case 74:        //  202 Source description.
>-    case 75:        //  203 Goodbye.
>-    case 76:        //  204 Application-defined.
>-    case 77:        //  205 Transport layer FB message.
>-    case 78:        //  206 Payload-specific FB message.
>-    case 79:        //  207 Extended report.
>+    case 64:  //  192 Full INTRA-frame request.
>+    case 72:  //  200 Sender report.
>+    case 73:  //  201 Receiver report.
>+    case 74:  //  202 Source description.
>+    case 75:  //  203 Goodbye.
>+    case 76:  //  204 Application-defined.
>+    case 77:  //  205 Transport layer FB message.
>+    case 78:  //  206 Payload-specific FB message.
>+    case 79:  //  207 Extended report.
>       RTC_LOG(LS_ERROR) << "Can't register invalid receiver payload type: "
>                         << payload_type;
>       return false;
>@@ -98,12 +91,7 @@ bool IsPayloadTypeValid(int8_t payload_type) {
> 
> }  // namespace
> 
>-RTPPayloadRegistry::RTPPayloadRegistry()
>-    : incoming_payload_type_(-1),
>-      last_received_payload_type_(-1),
>-      last_received_media_payload_type_(-1),
>-      rtx_(false),
>-      ssrc_rtx_(0) {}
>+RTPPayloadRegistry::RTPPayloadRegistry() = default;
> 
> RTPPayloadRegistry::~RTPPayloadRegistry() = default;
> 
>@@ -124,11 +112,6 @@ void RTPPayloadRegistry::SetAudioReceivePayloads(
>     payload_type_map_.emplace(rtp_payload_type,
>                               CreatePayloadType(audio_format));
>   }
>-
>-  // Clear the value of last received payload type since it might mean
>-  // something else now.
>-  last_received_payload_type_ = -1;
>-  last_received_media_payload_type_ = -1;
> }
> 
> int32_t RTPPayloadRegistry::RegisterReceivePayload(
>@@ -166,10 +149,7 @@ int32_t RTPPayloadRegistry::RegisterReceivePayload(
>   RTC_DCHECK(insert_status.second);  // Insertion succeeded.
>   *created_new_payload = true;
> 
>-  // Successful set of payload type, clear the value of last received payload
>-  // type since it might mean something else.
>-  last_received_payload_type_ = -1;
>-  last_received_media_payload_type_ = -1;
>+  // Successful set of payload type.
>   return 0;
> }
> 
>@@ -200,10 +180,7 @@ int32_t RTPPayloadRegistry::RegisterReceivePayload(
>       video_codec.plType, CreatePayloadType(video_codec));
>   RTC_DCHECK(insert_status.second);  // Insertion succeeded.
> 
>-  // Successful set of payload type, clear the value of last received payload
>-  // type since it might mean something else.
>-  last_received_payload_type_ = -1;
>-  last_received_media_payload_type_ = -1;
>+  // Successful set of payload type.
>   return 0;
> }
> 
>@@ -229,76 +206,7 @@ void RTPPayloadRegistry::DeregisterAudioCodecOrRedTypeRegardlessOfPayloadType(
>   }
> }
> 
>-int32_t RTPPayloadRegistry::ReceivePayloadType(
>-    const SdpAudioFormat& audio_format,
>-    int8_t* payload_type) const {
>-  assert(payload_type);
>-  rtc::CritScope cs(&crit_sect_);
>-
>-  for (const auto& it : payload_type_map_) {
>-    if (PayloadIsCompatible(it.second, audio_format)) {
>-      *payload_type = it.first;
>-      return 0;
>-    }
>-  }
>-  return -1;
>-}
>-
>-int32_t RTPPayloadRegistry::ReceivePayloadType(const VideoCodec& video_codec,
>-                                               int8_t* payload_type) const {
>-  assert(payload_type);
>-  rtc::CritScope cs(&crit_sect_);
>-
>-  for (const auto& it : payload_type_map_) {
>-    if (PayloadIsCompatible(it.second, video_codec)) {
>-      *payload_type = it.first;
>-      return 0;
>-    }
>-  }
>-  return -1;
>-}
>-
>-bool RTPPayloadRegistry::RtxEnabled() const {
>-  rtc::CritScope cs(&crit_sect_);
>-  return rtx_;
>-}
>-
>-bool RTPPayloadRegistry::IsRtxInternal(const RTPHeader& header) const {
>-  return rtx_ && ssrc_rtx_ == header.ssrc;
>-}
>-
>-void RTPPayloadRegistry::SetRtxSsrc(uint32_t ssrc) {
>-  rtc::CritScope cs(&crit_sect_);
>-  ssrc_rtx_ = ssrc;
>-  rtx_ = true;
>-}
>-
>-bool RTPPayloadRegistry::GetRtxSsrc(uint32_t* ssrc) const {
>-  rtc::CritScope cs(&crit_sect_);
>-  *ssrc = ssrc_rtx_;
>-  return rtx_;
>-}
>-
>-void RTPPayloadRegistry::SetRtxPayloadType(int payload_type,
>-                                           int associated_payload_type) {
>-  rtc::CritScope cs(&crit_sect_);
>-  if (payload_type < 0) {
>-    RTC_LOG(LS_ERROR) << "Invalid RTX payload type: " << payload_type;
>-    return;
>-  }
>-
>-  rtx_payload_type_map_[payload_type] = associated_payload_type;
>-  rtx_ = true;
>-}
>-
>-bool RTPPayloadRegistry::IsRed(const RTPHeader& header) const {
>-  rtc::CritScope cs(&crit_sect_);
>-  auto it = payload_type_map_.find(header.payloadType);
>-  return it != payload_type_map_.end() && _stricmp(it->second.name, "red") == 0;
>-}
>-
>-int RTPPayloadRegistry::GetPayloadTypeFrequency(
>-    uint8_t payload_type) const {
>+int RTPPayloadRegistry::GetPayloadTypeFrequency(uint8_t payload_type) const {
>   const auto payload = PayloadTypeToPayload(payload_type);
>   if (!payload) {
>     return -1;
>@@ -309,40 +217,13 @@ int RTPPayloadRegistry::GetPayloadTypeFrequency(
>              : kVideoPayloadTypeFrequency;
> }
> 
>-rtc::Optional<RtpUtility::Payload> RTPPayloadRegistry::PayloadTypeToPayload(
>+absl::optional<RtpUtility::Payload> RTPPayloadRegistry::PayloadTypeToPayload(
>     uint8_t payload_type) const {
>   rtc::CritScope cs(&crit_sect_);
>   const auto it = payload_type_map_.find(payload_type);
>   return it == payload_type_map_.end()
>-             ? rtc::nullopt
>-             : rtc::Optional<RtpUtility::Payload>(it->second);
>-}
>-
>-void RTPPayloadRegistry::SetIncomingPayloadType(const RTPHeader& header) {
>-  rtc::CritScope cs(&crit_sect_);
>-  if (!IsRtxInternal(header))
>-    incoming_payload_type_ = header.payloadType;
>-}
>-
>-bool RTPPayloadRegistry::ReportMediaPayloadType(uint8_t media_payload_type) {
>-  rtc::CritScope cs(&crit_sect_);
>-  if (last_received_media_payload_type_ == media_payload_type) {
>-    // Media type unchanged.
>-    return true;
>-  }
>-  last_received_media_payload_type_ = media_payload_type;
>-  return false;
>-}
>-
>-// Returns -1 if a payload with name |payload_name| is not registered.
>-int8_t RTPPayloadRegistry::GetPayloadTypeWithName(
>-    const char* payload_name) const {
>-  rtc::CritScope cs(&crit_sect_);
>-  for (const auto& it : payload_type_map_) {
>-    if (_stricmp(it.second.name, payload_name) == 0)
>-      return it.first;
>-  }
>-  return -1;
>+             ? absl::nullopt
>+             : absl::optional<RtpUtility::Payload>(it->second);
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc
>index ee2c016d267..9ae6aef0274 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_payload_registry_unittest.cc
>@@ -30,7 +30,6 @@ TEST(RtpPayloadRegistryTest,
>   const uint8_t payload_type = 97;
>   VideoCodec video_codec;
>   video_codec.codecType = kVideoCodecVP8;
>-  strncpy(video_codec.plName, "VP8", RTP_PAYLOAD_NAME_SIZE);
>   video_codec.plType = payload_type;
> 
>   EXPECT_EQ(0, rtp_payload_registry.RegisterReceivePayload(video_codec));
>@@ -42,7 +41,7 @@ TEST(RtpPayloadRegistryTest,
>   // We should get back the corresponding payload that we registered.
>   EXPECT_STREQ("VP8", retrieved_payload->name);
>   EXPECT_TRUE(retrieved_payload->typeSpecific.is_video());
>-  EXPECT_EQ(kRtpVideoVp8,
>+  EXPECT_EQ(kVideoCodecVP8,
>             retrieved_payload->typeSpecific.video_payload().videoCodecType);
> 
>   // Now forget about it and verify it's gone.
>@@ -76,23 +75,6 @@ TEST(RtpPayloadRegistryTest,
>   EXPECT_FALSE(rtp_payload_registry.PayloadTypeToPayload(payload_type));
> }
> 
>-TEST(RtpPayloadRegistryTest, AudioRedWorkProperly) {
>-  RTPPayloadRegistry rtp_payload_registry;
>-  bool new_payload_created = false;
>-  const SdpAudioFormat red_format("red", 8000, 1);
>-
>-  EXPECT_EQ(0, rtp_payload_registry.RegisterReceivePayload(
>-                   127, red_format, &new_payload_created));
>-  EXPECT_TRUE(new_payload_created);
>-
>-  EXPECT_EQ(127, rtp_payload_registry.red_payload_type());
>-
>-  const auto retrieved_payload = rtp_payload_registry.PayloadTypeToPayload(127);
>-  EXPECT_TRUE(retrieved_payload);
>-  EXPECT_TRUE(retrieved_payload->typeSpecific.is_audio());
>-  EXPECT_EQ(red_format, retrieved_payload->typeSpecific.audio_payload().format);
>-}
>-
> TEST(RtpPayloadRegistryTest,
>      DoesNotAcceptSamePayloadTypeTwiceExceptIfPayloadIsCompatible) {
>   constexpr int payload_type = 97;
>@@ -167,28 +149,6 @@ TEST(RtpPayloadRegistryTest,
>       << "Not compatible; both payloads should be kept.";
> }
> 
>-TEST(RtpPayloadRegistryTest,
>-     LastReceivedCodecTypesAreResetWhenRegisteringNewPayloadTypes) {
>-  RTPPayloadRegistry rtp_payload_registry;
>-  rtp_payload_registry.set_last_received_payload_type(17);
>-  EXPECT_EQ(17, rtp_payload_registry.last_received_payload_type());
>-
>-  bool media_type_unchanged = rtp_payload_registry.ReportMediaPayloadType(18);
>-  EXPECT_FALSE(media_type_unchanged);
>-  media_type_unchanged = rtp_payload_registry.ReportMediaPayloadType(18);
>-  EXPECT_TRUE(media_type_unchanged);
>-
>-  bool ignored;
>-  constexpr int payload_type = 34;
>-  const SdpAudioFormat audio_format("name", 44000, 1);
>-  EXPECT_EQ(0, rtp_payload_registry.RegisterReceivePayload(
>-                   payload_type, audio_format, &ignored));
>-
>-  EXPECT_EQ(-1, rtp_payload_registry.last_received_payload_type());
>-  media_type_unchanged = rtp_payload_registry.ReportMediaPayloadType(18);
>-  EXPECT_FALSE(media_type_unchanged);
>-}
>-
> class ParameterizedRtpPayloadRegistryTest
>     : public ::testing::TestWithParam<int> {};
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc
>index 270c00d48ae..030c79f2ca3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.cc
>@@ -11,7 +11,7 @@
> #include "modules/rtp_rtcp/source/rtp_receiver_audio.h"
> 
> #include <assert.h>  // assert
>-#include <math.h>   // pow()
>+#include <math.h>    // pow()
> #include <string.h>  // memcpy()
> 
> #include "common_types.h"  // NOLINT(build/include)
>@@ -25,50 +25,9 @@ RTPReceiverStrategy* RTPReceiverStrategy::CreateAudioStrategy(
> }
> 
> RTPReceiverAudio::RTPReceiverAudio(RtpData* data_callback)
>-    : RTPReceiverStrategy(data_callback),
>-      TelephoneEventHandler(),
>-      telephone_event_forward_to_decoder_(false),
>-      telephone_event_payload_type_(-1),
>-      cng_nb_payload_type_(-1),
>-      cng_wb_payload_type_(-1),
>-      cng_swb_payload_type_(-1),
>-      cng_fb_payload_type_(-1),
>-      num_energy_(0),
>-      current_remote_energy_() {
>-  memset(current_remote_energy_, 0, sizeof(current_remote_energy_));
>-}
>-
>-// Outband TelephoneEvent(DTMF) detection
>-void RTPReceiverAudio::SetTelephoneEventForwardToDecoder(
>-    bool forward_to_decoder) {
>-  rtc::CritScope lock(&crit_sect_);
>-  telephone_event_forward_to_decoder_ = forward_to_decoder;
>-}
>+    : RTPReceiverStrategy(data_callback) {}
> 
>-// Is forwarding of outband telephone events turned on/off?
>-bool RTPReceiverAudio::TelephoneEventForwardToDecoder() const {
>-  rtc::CritScope lock(&crit_sect_);
>-  return telephone_event_forward_to_decoder_;
>-}
>-
>-bool RTPReceiverAudio::TelephoneEventPayloadType(
>-    int8_t payload_type) const {
>-  rtc::CritScope lock(&crit_sect_);
>-  return telephone_event_payload_type_ == payload_type;
>-}
>-
>-bool RTPReceiverAudio::CNGPayloadType(int8_t payload_type) {
>-  rtc::CritScope lock(&crit_sect_);
>-  return payload_type == cng_nb_payload_type_ ||
>-         payload_type == cng_wb_payload_type_ ||
>-         payload_type == cng_swb_payload_type_ ||
>-         payload_type == cng_fb_payload_type_;
>-}
>-
>-bool RTPReceiverAudio::ShouldReportCsrcChanges(uint8_t payload_type) const {
>-  // Don't do this for DTMF packets, otherwise it's fine.
>-  return !TelephoneEventPayloadType(payload_type);
>-}
>+RTPReceiverAudio::~RTPReceiverAudio() = default;
> 
> // -   Sample based or frame based codecs based on RFC 3551
> // -
>@@ -102,102 +61,18 @@ bool RTPReceiverAudio::ShouldReportCsrcChanges(uint8_t payload_type) const {
> // -   MPA       frame         N/A              var.      var.
> // -
> // -   G7221     frame         N/A
>-int32_t RTPReceiverAudio::OnNewPayloadTypeCreated(
>-    int payload_type,
>-    const SdpAudioFormat& audio_format) {
>-  rtc::CritScope lock(&crit_sect_);
>-
>-  if (RtpUtility::StringCompare(audio_format.name.c_str(), "telephone-event",
>-                                15)) {
>-    telephone_event_payload_type_ = payload_type;
>-  }
>-  if (RtpUtility::StringCompare(audio_format.name.c_str(), "cn", 2)) {
>-    // We support comfort noise at four different frequencies.
>-    if (audio_format.clockrate_hz == 8000) {
>-      cng_nb_payload_type_ = payload_type;
>-    } else if (audio_format.clockrate_hz == 16000) {
>-      cng_wb_payload_type_ = payload_type;
>-    } else if (audio_format.clockrate_hz == 32000) {
>-      cng_swb_payload_type_ = payload_type;
>-    } else if (audio_format.clockrate_hz == 48000) {
>-      cng_fb_payload_type_ = payload_type;
>-    } else {
>-      assert(false);
>-      return -1;
>-    }
>-  }
>-  return 0;
>-}
> 
> int32_t RTPReceiverAudio::ParseRtpPacket(WebRtcRTPHeader* rtp_header,
>                                          const PayloadUnion& specific_payload,
>-                                         bool is_red,
>                                          const uint8_t* payload,
>                                          size_t payload_length,
>                                          int64_t timestamp_ms) {
>-  TRACE_EVENT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "Audio::ParseRtp",
>-               "seqnum", rtp_header->header.sequenceNumber, "timestamp",
>-               rtp_header->header.timestamp);
>-  rtp_header->type.Audio.numEnergy = rtp_header->header.numCSRCs;
>-  num_energy_ = rtp_header->type.Audio.numEnergy;
>-  if (rtp_header->type.Audio.numEnergy > 0 &&
>-      rtp_header->type.Audio.numEnergy <= kRtpCsrcSize) {
>-    memcpy(current_remote_energy_,
>-           rtp_header->type.Audio.arrOfEnergy,
>-           rtp_header->type.Audio.numEnergy);
>-  }
>-
>   if (first_packet_received_()) {
>     RTC_LOG(LS_INFO) << "Received first audio RTP packet";
>   }
> 
>   return ParseAudioCodecSpecific(rtp_header, payload, payload_length,
>-                                 specific_payload.audio_payload(), is_red);
>-}
>-
>-RTPAliveType RTPReceiverAudio::ProcessDeadOrAlive(
>-    uint16_t last_payload_length) const {
>-
>-  // Our CNG is 9 bytes; if it's a likely CNG the receiver needs to check
>-  // kRtpNoRtp against NetEq speech_type kOutputPLCtoCNG.
>-  if (last_payload_length < 10) {  // our CNG is 9 bytes
>-    return kRtpNoRtp;
>-  } else {
>-    return kRtpDead;
>-  }
>-}
>-
>-void RTPReceiverAudio::CheckPayloadChanged(int8_t payload_type,
>-                                           PayloadUnion* /* specific_payload */,
>-                                           bool* should_discard_changes) {
>-  *should_discard_changes =
>-      TelephoneEventPayloadType(payload_type) || CNGPayloadType(payload_type);
>-}
>-
>-int RTPReceiverAudio::Energy(uint8_t array_of_energy[kRtpCsrcSize]) const {
>-  rtc::CritScope cs(&crit_sect_);
>-
>-  assert(num_energy_ <= kRtpCsrcSize);
>-
>-  if (num_energy_ > 0) {
>-    memcpy(array_of_energy, current_remote_energy_,
>-           sizeof(uint8_t) * num_energy_);
>-  }
>-  return num_energy_;
>-}
>-
>-int32_t RTPReceiverAudio::InvokeOnInitializeDecoder(
>-    RtpFeedback* callback,
>-    int8_t payload_type,
>-    const char payload_name[RTP_PAYLOAD_NAME_SIZE],
>-    const PayloadUnion& specific_payload) const {
>-  const auto& ap = specific_payload.audio_payload();
>-  if (callback->OnInitializeDecoder(payload_type, ap.format, ap.rate) == -1) {
>-    RTC_LOG(LS_ERROR) << "Failed to create decoder for payload type: "
>-                      << payload_name << "/" << static_cast<int>(payload_type);
>-    return -1;
>-  }
>-  return 0;
>+                                 specific_payload.audio_payload());
> }
> 
> // We are not allowed to have any critsects when calling data_callback.
>@@ -205,103 +80,15 @@ int32_t RTPReceiverAudio::ParseAudioCodecSpecific(
>     WebRtcRTPHeader* rtp_header,
>     const uint8_t* payload_data,
>     size_t payload_length,
>-    const AudioPayload& audio_specific,
>-    bool is_red) {
>+    const AudioPayload& audio_specific) {
>   RTC_DCHECK_GE(payload_length, rtp_header->header.paddingLength);
>   const size_t payload_data_length =
>       payload_length - rtp_header->header.paddingLength;
>   if (payload_data_length == 0) {
>-    rtp_header->type.Audio.isCNG = false;
>     rtp_header->frameType = kEmptyFrame;
>     return data_callback_->OnReceivedPayloadData(nullptr, 0, rtp_header);
>   }
> 
>-  bool telephone_event_packet =
>-      TelephoneEventPayloadType(rtp_header->header.payloadType);
>-  if (telephone_event_packet) {
>-    rtc::CritScope lock(&crit_sect_);
>-
>-    // RFC 4733 2.3
>-    // 0                   1                   2                   3
>-    // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
>-    // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>-    // |     event     |E|R| volume    |          duration             |
>-    // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
>-    //
>-    if (payload_data_length % 4 != 0) {
>-      return -1;
>-    }
>-    size_t number_of_events = payload_data_length / 4;
>-
>-    // sanity
>-    if (number_of_events >= MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS) {
>-      number_of_events = MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS;
>-    }
>-    for (size_t n = 0; n < number_of_events; ++n) {
>-      RTC_DCHECK_GE(payload_data_length, (4 * n) + 2);
>-      bool end = (payload_data[(4 * n) + 1] & 0x80) ? true : false;
>-
>-      std::set<uint8_t>::iterator event =
>-          telephone_event_reported_.find(payload_data[4 * n]);
>-
>-      if (event != telephone_event_reported_.end()) {
>-        // we have already seen this event
>-        if (end) {
>-          telephone_event_reported_.erase(payload_data[4 * n]);
>-        }
>-      } else {
>-        if (end) {
>-          // don't add if it's a end of a tone
>-        } else {
>-          telephone_event_reported_.insert(payload_data[4 * n]);
>-        }
>-      }
>-    }
>-
>-    // RFC 4733 2.5.1.3 & 2.5.2.3 Long-Duration Events
>-    // should not be a problem since we don't care about the duration
>-
>-    // RFC 4733 See 2.5.1.5. & 2.5.2.4.  Multiple Events in a Packet
>-  }
>-
>-  {
>-    rtc::CritScope lock(&crit_sect_);
>-
>-    // Check if this is a CNG packet, receiver might want to know
>-    if (CNGPayloadType(rtp_header->header.payloadType)) {
>-      rtp_header->type.Audio.isCNG = true;
>-      rtp_header->frameType = kAudioFrameCN;
>-    } else {
>-      rtp_header->frameType = kAudioFrameSpeech;
>-      rtp_header->type.Audio.isCNG = false;
>-    }
>-
>-    // check if it's a DTMF event, hence something we can playout
>-    if (telephone_event_packet) {
>-      if (!telephone_event_forward_to_decoder_) {
>-        // don't forward event to decoder
>-        return 0;
>-      }
>-      std::set<uint8_t>::iterator first =
>-          telephone_event_reported_.begin();
>-      if (first != telephone_event_reported_.end() && *first > 15) {
>-        // don't forward non DTMF events
>-        return 0;
>-      }
>-    }
>-  }
>-  // TODO(holmer): Break this out to have RED parsing handled generically.
>-  RTC_DCHECK_GT(payload_data_length, 0);
>-  if (is_red && !(payload_data[0] & 0x80)) {
>-    // we recive only one frame packed in a RED packet remove the RED wrapper
>-    rtp_header->header.payloadType = payload_data[0];
>-
>-    // only one frame in the RED strip the one byte to help NetEq
>-    return data_callback_->OnReceivedPayloadData(
>-        payload_data + 1, payload_data_length - 1, rtp_header);
>-  }
>-
>-  rtp_header->type.Audio.channel = audio_specific.format.num_channels;
>   return data_callback_->OnReceivedPayloadData(payload_data,
>                                                payload_data_length, rtp_header);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h
>index b031002709d..5d97a1f3a94 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_receiver_audio.h
>@@ -18,78 +18,26 @@
> #include "modules/rtp_rtcp/source/rtp_receiver_strategy.h"
> #include "modules/rtp_rtcp/source/rtp_utility.h"
> #include "rtc_base/onetimeevent.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
> // Handles audio RTP packets. This class is thread-safe.
>-class RTPReceiverAudio : public RTPReceiverStrategy,
>-                         public TelephoneEventHandler {
>+class RTPReceiverAudio : public RTPReceiverStrategy {
>  public:
>   explicit RTPReceiverAudio(RtpData* data_callback);
>-  virtual ~RTPReceiverAudio() {}
>-
>-  // The following three methods implement the TelephoneEventHandler interface.
>-  // Forward DTMFs to decoder for playout.
>-  void SetTelephoneEventForwardToDecoder(bool forward_to_decoder) override;
>-
>-  // Is forwarding of outband telephone events turned on/off?
>-  bool TelephoneEventForwardToDecoder() const override;
>-
>-  // Is TelephoneEvent configured with |payload_type|.
>-  bool TelephoneEventPayloadType(const int8_t payload_type) const override;
>-
>-  TelephoneEventHandler* GetTelephoneEventHandler() override { return this; }
>-
>-  // Returns true if CNG is configured with |payload_type|.
>-  bool CNGPayloadType(const int8_t payload_type);
>+  ~RTPReceiverAudio() override;
> 
>   int32_t ParseRtpPacket(WebRtcRTPHeader* rtp_header,
>                          const PayloadUnion& specific_payload,
>-                         bool is_red,
>                          const uint8_t* packet,
>                          size_t payload_length,
>                          int64_t timestamp_ms) override;
> 
>-  RTPAliveType ProcessDeadOrAlive(uint16_t last_payload_length) const override;
>-
>-  bool ShouldReportCsrcChanges(uint8_t payload_type) const override;
>-
>-  int32_t OnNewPayloadTypeCreated(int payload_type,
>-                                  const SdpAudioFormat& audio_format) override;
>-
>-  int32_t InvokeOnInitializeDecoder(
>-      RtpFeedback* callback,
>-      int8_t payload_type,
>-      const char payload_name[RTP_PAYLOAD_NAME_SIZE],
>-      const PayloadUnion& specific_payload) const override;
>-
>-  // We need to look out for special payload types here and sometimes reset
>-  // statistics. In addition we sometimes need to tweak the frequency.
>-  void CheckPayloadChanged(int8_t payload_type,
>-                           PayloadUnion* specific_payload,
>-                           bool* should_discard_changes) override;
>-
>-  int Energy(uint8_t array_of_energy[kRtpCsrcSize]) const override;
>-
>  private:
>   int32_t ParseAudioCodecSpecific(WebRtcRTPHeader* rtp_header,
>                                   const uint8_t* payload_data,
>                                   size_t payload_length,
>-                                  const AudioPayload& audio_specific,
>-                                  bool is_red);
>-
>-  bool telephone_event_forward_to_decoder_;
>-  int8_t telephone_event_payload_type_;
>-  std::set<uint8_t> telephone_event_reported_;
>-
>-  int8_t cng_nb_payload_type_;
>-  int8_t cng_wb_payload_type_;
>-  int8_t cng_swb_payload_type_;
>-  int8_t cng_fb_payload_type_;
>-
>-  uint8_t num_energy_;
>-  uint8_t current_remote_energy_[kRtpCsrcSize];
>+                                  const AudioPayload& audio_specific);
> 
>   ThreadUnsafeOneTimeEvent first_packet_received_;
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
>index 6ed4d7ce7da..ea63e5ee193 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.cc
>@@ -29,7 +29,7 @@
> namespace webrtc {
> 
> namespace {
>-bool InOrderPacket(rtc::Optional<uint16_t> latest_sequence_number,
>+bool InOrderPacket(absl::optional<uint16_t> latest_sequence_number,
>                    uint16_t current_sequence_number) {
>   if (!latest_sequence_number)
>     return true;
>@@ -56,26 +56,20 @@ const int64_t kGetSourcesTimeoutMs = 10000;
> RtpReceiver* RtpReceiver::CreateVideoReceiver(
>     Clock* clock,
>     RtpData* incoming_payload_callback,
>-    RtpFeedback* incoming_messages_callback,
>     RTPPayloadRegistry* rtp_payload_registry) {
>   RTC_DCHECK(incoming_payload_callback != nullptr);
>-  if (!incoming_messages_callback)
>-    incoming_messages_callback = NullObjectRtpFeedback();
>   return new RtpReceiverImpl(
>-      clock, incoming_messages_callback, rtp_payload_registry,
>+      clock, rtp_payload_registry,
>       RTPReceiverStrategy::CreateVideoStrategy(incoming_payload_callback));
> }
> 
> RtpReceiver* RtpReceiver::CreateAudioReceiver(
>     Clock* clock,
>     RtpData* incoming_payload_callback,
>-    RtpFeedback* incoming_messages_callback,
>     RTPPayloadRegistry* rtp_payload_registry) {
>   RTC_DCHECK(incoming_payload_callback != nullptr);
>-  if (!incoming_messages_callback)
>-    incoming_messages_callback = NullObjectRtpFeedback();
>   return new RtpReceiverImpl(
>-      clock, incoming_messages_callback, rtp_payload_registry,
>+      clock, rtp_payload_registry,
>       RTPReceiverStrategy::CreateAudioStrategy(incoming_payload_callback));
> }
> 
>@@ -85,28 +79,16 @@ int32_t RtpReceiver::RegisterReceivePayload(const CodecInst& audio_codec) {
> }
> 
> RtpReceiverImpl::RtpReceiverImpl(Clock* clock,
>-                                 RtpFeedback* incoming_messages_callback,
>                                  RTPPayloadRegistry* rtp_payload_registry,
>                                  RTPReceiverStrategy* rtp_media_receiver)
>     : clock_(clock),
>       rtp_payload_registry_(rtp_payload_registry),
>       rtp_media_receiver_(rtp_media_receiver),
>-      cb_rtp_feedback_(incoming_messages_callback),
>       ssrc_(0),
>-      num_csrcs_(0),
>-      current_remote_csrc_(),
>       last_received_timestamp_(0),
>-      last_received_frame_time_ms_(-1) {
>-  assert(incoming_messages_callback);
>+      last_received_frame_time_ms_(-1) {}
> 
>-  memset(current_remote_csrc_, 0, sizeof(current_remote_csrc_));
>-}
>-
>-RtpReceiverImpl::~RtpReceiverImpl() {
>-  for (int i = 0; i < num_csrcs_; ++i) {
>-    cb_rtp_feedback_->OnIncomingCSRCChanged(current_remote_csrc_[i], false);
>-  }
>-}
>+RtpReceiverImpl::~RtpReceiverImpl() {}
> 
> int32_t RtpReceiverImpl::RegisterReceivePayload(
>     int payload_type,
>@@ -119,14 +101,6 @@ int32_t RtpReceiverImpl::RegisterReceivePayload(
>   bool created_new_payload = false;
>   int32_t result = rtp_payload_registry_->RegisterReceivePayload(
>       payload_type, audio_format, &created_new_payload);
>-  if (created_new_payload) {
>-    if (rtp_media_receiver_->OnNewPayloadTypeCreated(payload_type,
>-                                                     audio_format) != 0) {
>-      RTC_LOG(LS_ERROR) << "Failed to register payload: " << audio_format.name
>-                        << "/" << payload_type;
>-      return -1;
>-    }
>-  }
>   return result;
> }
> 
>@@ -135,8 +109,7 @@ int32_t RtpReceiverImpl::RegisterReceivePayload(const VideoCodec& video_codec) {
>   return rtp_payload_registry_->RegisterReceivePayload(video_codec);
> }
> 
>-int32_t RtpReceiverImpl::DeRegisterReceivePayload(
>-    const int8_t payload_type) {
>+int32_t RtpReceiverImpl::DeRegisterReceivePayload(const int8_t payload_type) {
>   rtc::CritScope lock(&critical_section_rtp_receiver_);
>   return rtp_payload_registry_->DeRegisterReceivePayload(payload_type);
> }
>@@ -146,23 +119,6 @@ uint32_t RtpReceiverImpl::SSRC() const {
>   return ssrc_;
> }
> 
>-// Get remote CSRC.
>-int32_t RtpReceiverImpl::CSRCs(uint32_t array_of_csrcs[kRtpCsrcSize]) const {
>-  rtc::CritScope lock(&critical_section_rtp_receiver_);
>-
>-  assert(num_csrcs_ <= kRtpCsrcSize);
>-
>-  if (num_csrcs_ > 0) {
>-    memcpy(array_of_csrcs, current_remote_csrc_, sizeof(uint32_t)*num_csrcs_);
>-  }
>-  return num_csrcs_;
>-}
>-
>-int32_t RtpReceiverImpl::Energy(
>-    uint8_t array_of_energy[kRtpCsrcSize]) const {
>-  return rtp_media_receiver_->Energy(array_of_energy);
>-}
>-
> bool RtpReceiverImpl::IncomingRtpPacket(const RTPHeader& rtp_header,
>                                         const uint8_t* payload,
>                                         size_t payload_length,
>@@ -170,33 +126,30 @@ bool RtpReceiverImpl::IncomingRtpPacket(const RTPHeader& rtp_header,
>   // Trigger our callbacks.
>   CheckSSRCChanged(rtp_header);
> 
>-  int8_t first_payload_byte = payload_length > 0 ? payload[0] : 0;
>-  bool is_red = false;
>+  if (payload_length == 0) {
>+    // OK, keep-alive packet.
>+    return true;
>+  }
>+  int64_t now_ms = clock_->TimeInMilliseconds();
> 
>-  if (CheckPayloadChanged(rtp_header, first_payload_byte, &is_red,
>-                          &payload_specific) == -1) {
>-    if (payload_length == 0) {
>-      // OK, keep-alive packet.
>-      return true;
>-    }
>-    RTC_LOG(LS_WARNING) << "Receiving invalid payload type.";
>-    return false;
>+  {
>+    rtc::CritScope lock(&critical_section_rtp_receiver_);
>+
>+    csrcs_.Update(
>+        now_ms, rtc::MakeArrayView(rtp_header.arrOfCSRCs, rtp_header.numCSRCs));
>   }
> 
>-  WebRtcRTPHeader webrtc_rtp_header;
>-  memset(&webrtc_rtp_header, 0, sizeof(webrtc_rtp_header));
>+  WebRtcRTPHeader webrtc_rtp_header{};
>   webrtc_rtp_header.header = rtp_header;
>-  CheckCSRC(webrtc_rtp_header);
> 
>   auto audio_level =
>       rtp_header.extension.hasAudioLevel
>-          ? rtc::Optional<uint8_t>(rtp_header.extension.audioLevel)
>-          : rtc::nullopt;
>+          ? absl::optional<uint8_t>(rtp_header.extension.audioLevel)
>+          : absl::nullopt;
>   UpdateSources(audio_level);
> 
>   int32_t ret_val = rtp_media_receiver_->ParseRtpPacket(
>-      &webrtc_rtp_header, payload_specific, is_red, payload, payload_length,
>-      clock_->TimeInMilliseconds());
>+      &webrtc_rtp_header, payload_specific, payload, payload_length, now_ms);
> 
>   if (ret_val < 0) {
>     return false;
>@@ -220,24 +173,15 @@ bool RtpReceiverImpl::IncomingRtpPacket(const RTPHeader& rtp_header,
>   return true;
> }
> 
>-TelephoneEventHandler* RtpReceiverImpl::GetTelephoneEventHandler() {
>-  return rtp_media_receiver_->GetTelephoneEventHandler();
>-}
>-
> std::vector<RtpSource> RtpReceiverImpl::GetSources() const {
>   rtc::CritScope lock(&critical_section_rtp_receiver_);
> 
>   int64_t now_ms = clock_->TimeInMilliseconds();
>-  std::vector<RtpSource> sources;
>-
>   RTC_DCHECK(std::is_sorted(ssrc_sources_.begin(), ssrc_sources_.end(),
>                             [](const RtpSource& lhs, const RtpSource& rhs) {
>                               return lhs.timestamp_ms() < rhs.timestamp_ms();
>                             }));
>-  RTC_DCHECK(std::is_sorted(csrc_sources_.begin(), csrc_sources_.end(),
>-                            [](const RtpSource& lhs, const RtpSource& rhs) {
>-                              return lhs.timestamp_ms() < rhs.timestamp_ms();
>-                            }));
>+  std::vector<RtpSource> sources = csrcs_.GetSources(now_ms);
> 
>   std::set<uint32_t> selected_ssrcs;
>   for (auto rit = ssrc_sources_.rbegin(); rit != ssrc_sources_.rend(); ++rit) {
>@@ -248,13 +192,6 @@ std::vector<RtpSource> RtpReceiverImpl::GetSources() const {
>       sources.push_back(*rit);
>     }
>   }
>-
>-  for (auto rit = csrc_sources_.rbegin(); rit != csrc_sources_.rend(); ++rit) {
>-    if ((now_ms - rit->timestamp_ms()) > kGetSourcesTimeoutMs) {
>-      break;
>-    }
>-    sources.push_back(*rit);
>-  }
>   return sources;
> }
> 
>@@ -270,257 +207,18 @@ bool RtpReceiverImpl::GetLatestTimestamps(uint32_t* timestamp,
>   return true;
> }
> 
>+// TODO(nisse): Delete.
> // Implementation note: must not hold critsect when called.
> void RtpReceiverImpl::CheckSSRCChanged(const RTPHeader& rtp_header) {
>-  bool new_ssrc = false;
>-  rtc::Optional<AudioPayload> reinitialize_audio_payload;
>-
>-  {
>-    rtc::CritScope lock(&critical_section_rtp_receiver_);
>-
>-    int8_t last_received_payload_type =
>-        rtp_payload_registry_->last_received_payload_type();
>-    if (ssrc_ != rtp_header.ssrc ||
>-        (last_received_payload_type == -1 && ssrc_ == 0)) {
>-      // We need the payload_type_ to make the call if the remote SSRC is 0.
>-      new_ssrc = true;
>-
>-      last_received_timestamp_ = 0;
>-      last_received_frame_time_ms_ = -1;
>-
>-      // Do we have a SSRC? Then the stream is restarted.
>-      if (ssrc_ != 0) {
>-        // Do we have the same codec? Then re-initialize coder.
>-        if (rtp_header.payloadType == last_received_payload_type) {
>-          const auto payload = rtp_payload_registry_->PayloadTypeToPayload(
>-              rtp_header.payloadType);
>-          if (!payload) {
>-            return;
>-          }
>-          if (payload->typeSpecific.is_audio()) {
>-            reinitialize_audio_payload.emplace(
>-                payload->typeSpecific.audio_payload());
>-          } else {
>-            // OnInitializeDecoder() is only used for audio.
>-          }
>-        }
>-      }
>-      ssrc_ = rtp_header.ssrc;
>-    }
>-  }
>-
>-  if (new_ssrc) {
>-    // We need to get this to our RTCP sender and receiver.
>-    // We need to do this outside critical section.
>-    cb_rtp_feedback_->OnIncomingSSRCChanged(rtp_header.ssrc);
>-  }
>-
>-  if (reinitialize_audio_payload) {
>-    if (-1 == cb_rtp_feedback_->OnInitializeDecoder(
>-                  rtp_header.payloadType, reinitialize_audio_payload->format,
>-                  reinitialize_audio_payload->rate)) {
>-      // New stream, same codec.
>-      RTC_LOG(LS_ERROR) << "Failed to create decoder for payload type: "
>-                        << static_cast<int>(rtp_header.payloadType);
>-    }
>-  }
>-}
>-
>-// Implementation note: must not hold critsect when called.
>-// TODO(phoglund): Move as much as possible of this code path into the media
>-// specific receivers. Basically this method goes through a lot of trouble to
>-// compute something which is only used by the media specific parts later. If
>-// this code path moves we can get rid of some of the rtp_receiver ->
>-// media_specific interface (such as CheckPayloadChange, possibly get/set
>-// last known payload).
>-int32_t RtpReceiverImpl::CheckPayloadChanged(const RTPHeader& rtp_header,
>-                                             const int8_t first_payload_byte,
>-                                             bool* is_red,
>-                                             PayloadUnion* specific_payload) {
>-  bool re_initialize_decoder = false;
>-
>-  char payload_name[RTP_PAYLOAD_NAME_SIZE];
>-  int8_t payload_type = rtp_header.payloadType;
>-
>-  {
>-    rtc::CritScope lock(&critical_section_rtp_receiver_);
>-
>-    int8_t last_received_payload_type =
>-        rtp_payload_registry_->last_received_payload_type();
>-    // TODO(holmer): Remove this code when RED parsing has been broken out from
>-    // RtpReceiverAudio.
>-    if (payload_type != last_received_payload_type) {
>-      if (rtp_payload_registry_->red_payload_type() == payload_type) {
>-        // Get the real codec payload type.
>-        payload_type = first_payload_byte & 0x7f;
>-        *is_red = true;
>-
>-        if (rtp_payload_registry_->red_payload_type() == payload_type) {
>-          // Invalid payload type, traced by caller. If we proceeded here,
>-          // this would be set as |_last_received_payload_type|, and we would no
>-          // longer catch corrupt packets at this level.
>-          return -1;
>-        }
>-
>-        // When we receive RED we need to check the real payload type.
>-        if (payload_type == last_received_payload_type) {
>-          rtp_media_receiver_->GetLastMediaSpecificPayload(specific_payload);
>-          return 0;
>-        }
>-      }
>-      bool should_discard_changes = false;
>-
>-      rtp_media_receiver_->CheckPayloadChanged(
>-        payload_type, specific_payload,
>-        &should_discard_changes);
>-
>-      if (should_discard_changes) {
>-        *is_red = false;
>-        return 0;
>-      }
>-
>-      const auto payload =
>-          rtp_payload_registry_->PayloadTypeToPayload(payload_type);
>-      if (!payload) {
>-        // Not a registered payload type.
>-        return -1;
>-      }
>-      payload_name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
>-      strncpy(payload_name, payload->name, RTP_PAYLOAD_NAME_SIZE - 1);
>-
>-      rtp_payload_registry_->set_last_received_payload_type(payload_type);
>-
>-      re_initialize_decoder = true;
>-
>-      rtp_media_receiver_->SetLastMediaSpecificPayload(payload->typeSpecific);
>-      rtp_media_receiver_->GetLastMediaSpecificPayload(specific_payload);
>-
>-      if (!payload->typeSpecific.is_audio()) {
>-        bool media_type_unchanged =
>-            rtp_payload_registry_->ReportMediaPayloadType(payload_type);
>-        if (media_type_unchanged) {
>-          // Only reset the decoder if the media codec type has changed.
>-          re_initialize_decoder = false;
>-        }
>-      }
>-    } else {
>-      rtp_media_receiver_->GetLastMediaSpecificPayload(specific_payload);
>-      *is_red = false;
>-    }
>-  }  // End critsect.
>-
>-  if (re_initialize_decoder) {
>-    if (-1 ==
>-        rtp_media_receiver_->InvokeOnInitializeDecoder(
>-            cb_rtp_feedback_, payload_type, payload_name, *specific_payload)) {
>-      return -1;  // Wrong payload type.
>-    }
>-  }
>-  return 0;
>-}
>-
>-// Implementation note: must not hold critsect when called.
>-void RtpReceiverImpl::CheckCSRC(const WebRtcRTPHeader& rtp_header) {
>-  int32_t num_csrcs_diff = 0;
>-  uint32_t old_remote_csrc[kRtpCsrcSize];
>-  uint8_t old_num_csrcs = 0;
>-
>-  {
>-    rtc::CritScope lock(&critical_section_rtp_receiver_);
>-
>-    if (!rtp_media_receiver_->ShouldReportCsrcChanges(
>-        rtp_header.header.payloadType)) {
>-      return;
>-    }
>-    old_num_csrcs  = num_csrcs_;
>-    if (old_num_csrcs > 0) {
>-      // Make a copy of old.
>-      memcpy(old_remote_csrc, current_remote_csrc_,
>-             num_csrcs_ * sizeof(uint32_t));
>-    }
>-    const uint8_t num_csrcs = rtp_header.header.numCSRCs;
>-    if ((num_csrcs > 0) && (num_csrcs <= kRtpCsrcSize)) {
>-      // Copy new.
>-      memcpy(current_remote_csrc_,
>-             rtp_header.header.arrOfCSRCs,
>-             num_csrcs * sizeof(uint32_t));
>-    }
>-    if (num_csrcs > 0 || old_num_csrcs > 0) {
>-      num_csrcs_diff = num_csrcs - old_num_csrcs;
>-      num_csrcs_ = num_csrcs;  // Update stored CSRCs.
>-    } else {
>-      // No change.
>-      return;
>-    }
>-  }  // End critsect.
>-
>-  bool have_called_callback = false;
>-  // Search for new CSRC in old array.
>-  for (uint8_t i = 0; i < rtp_header.header.numCSRCs; ++i) {
>-    const uint32_t csrc = rtp_header.header.arrOfCSRCs[i];
>-
>-    bool found_match = false;
>-    for (uint8_t j = 0; j < old_num_csrcs; ++j) {
>-      if (csrc == old_remote_csrc[j]) {  // old list
>-        found_match = true;
>-        break;
>-      }
>-    }
>-    if (!found_match && csrc) {
>-      // Didn't find it, report it as new.
>-      have_called_callback = true;
>-      cb_rtp_feedback_->OnIncomingCSRCChanged(csrc, true);
>-    }
>-  }
>-  // Search for old CSRC in new array.
>-  for (uint8_t i = 0; i < old_num_csrcs; ++i) {
>-    const uint32_t csrc = old_remote_csrc[i];
>-
>-    bool found_match = false;
>-    for (uint8_t j = 0; j < rtp_header.header.numCSRCs; ++j) {
>-      if (csrc == rtp_header.header.arrOfCSRCs[j]) {
>-        found_match = true;
>-        break;
>-      }
>-    }
>-    if (!found_match && csrc) {
>-      // Did not find it, report as removed.
>-      have_called_callback = true;
>-      cb_rtp_feedback_->OnIncomingCSRCChanged(csrc, false);
>-    }
>-  }
>-  if (!have_called_callback) {
>-    // If the CSRC list contain non-unique entries we will end up here.
>-    // Using CSRC 0 to signal this event, not interop safe, other
>-    // implementations might have CSRC 0 as a valid value.
>-    if (num_csrcs_diff > 0) {
>-      cb_rtp_feedback_->OnIncomingCSRCChanged(0, true);
>-    } else if (num_csrcs_diff < 0) {
>-      cb_rtp_feedback_->OnIncomingCSRCChanged(0, false);
>-    }
>-  }
>+  rtc::CritScope lock(&critical_section_rtp_receiver_);
>+  ssrc_ = rtp_header.ssrc;
> }
> 
> void RtpReceiverImpl::UpdateSources(
>-    const rtc::Optional<uint8_t>& ssrc_audio_level) {
>+    const absl::optional<uint8_t>& ssrc_audio_level) {
>   rtc::CritScope lock(&critical_section_rtp_receiver_);
>   int64_t now_ms = clock_->TimeInMilliseconds();
> 
>-  for (size_t i = 0; i < num_csrcs_; ++i) {
>-    auto map_it = iterator_by_csrc_.find(current_remote_csrc_[i]);
>-    if (map_it == iterator_by_csrc_.end()) {
>-      // If it is a new CSRC, append a new object to the end of the list.
>-      csrc_sources_.emplace_back(now_ms, current_remote_csrc_[i],
>-                                 RtpSourceType::CSRC);
>-    } else {
>-      // If it is an existing CSRC, move the object to the end of the list.
>-      map_it->second->update_timestamp_ms(now_ms);
>-      csrc_sources_.splice(csrc_sources_.end(), csrc_sources_, map_it->second);
>-    }
>-    // Update the unordered_map.
>-    iterator_by_csrc_[current_remote_csrc_[i]] = std::prev(csrc_sources_.end());
>-  }
>-
>   // If this is the first packet or the SSRC is changed, insert a new
>   // contributing source that uses the SSRC.
>   if (ssrc_sources_.empty() || ssrc_sources_.rbegin()->source_id() != ssrc_) {
>@@ -535,15 +233,6 @@ void RtpReceiverImpl::UpdateSources(
> }
> 
> void RtpReceiverImpl::RemoveOutdatedSources(int64_t now_ms) {
>-  std::list<RtpSource>::iterator it;
>-  for (it = csrc_sources_.begin(); it != csrc_sources_.end(); ++it) {
>-    if ((now_ms - it->timestamp_ms()) <= kGetSourcesTimeoutMs) {
>-      break;
>-    }
>-    iterator_by_csrc_.erase(it->source_id());
>-  }
>-  csrc_sources_.erase(csrc_sources_.begin(), it);
>-
>   std::vector<RtpSource>::iterator vec_it;
>   for (vec_it = ssrc_sources_.begin(); vec_it != ssrc_sources_.end();
>        ++vec_it) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h
>index cd6b619ff01..66265f5fb10 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_receiver_impl.h
>@@ -11,17 +11,16 @@
> #ifndef MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_
> #define MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_IMPL_H_
> 
>-#include <list>
> #include <memory>
> #include <unordered_map>
> #include <vector>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> #include "modules/rtp_rtcp/include/rtp_receiver.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>+#include "modules/rtp_rtcp/source/contributing_sources.h"
> #include "modules/rtp_rtcp/source/rtp_receiver_strategy.h"
> #include "rtc_base/criticalsection.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -31,11 +30,10 @@ class RtpReceiverImpl : public RtpReceiver {
>   // want callbacks to do nothing). This class takes ownership of the media
>   // receiver but nothing else.
>   RtpReceiverImpl(Clock* clock,
>-                  RtpFeedback* incoming_messages_callback,
>                   RTPPayloadRegistry* rtp_payload_registry,
>                   RTPReceiverStrategy* rtp_media_receiver);
> 
>-  virtual ~RtpReceiverImpl();
>+  ~RtpReceiverImpl() override;
> 
>   int32_t RegisterReceivePayload(int payload_type,
>                                  const SdpAudioFormat& audio_format) override;
>@@ -53,31 +51,12 @@ class RtpReceiverImpl : public RtpReceiver {
> 
>   uint32_t SSRC() const override;
> 
>-  int32_t CSRCs(uint32_t array_of_csrc[kRtpCsrcSize]) const override;
>-
>-  int32_t Energy(uint8_t array_of_energy[kRtpCsrcSize]) const override;
>-
>-  TelephoneEventHandler* GetTelephoneEventHandler() override;
>-
>   std::vector<RtpSource> GetSources() const override;
> 
>-  const std::vector<RtpSource>& ssrc_sources_for_testing() const {
>-    return ssrc_sources_;
>-  }
>-
>-  const std::list<RtpSource>& csrc_sources_for_testing() const {
>-    return csrc_sources_;
>-  }
>-
>  private:
>   void CheckSSRCChanged(const RTPHeader& rtp_header);
>-  void CheckCSRC(const WebRtcRTPHeader& rtp_header);
>-  int32_t CheckPayloadChanged(const RTPHeader& rtp_header,
>-                              const int8_t first_payload_byte,
>-                              bool* is_red,
>-                              PayloadUnion* payload);
> 
>-  void UpdateSources(const rtc::Optional<uint8_t>& ssrc_audio_level);
>+  void UpdateSources(const absl::optional<uint8_t>& ssrc_audio_level);
>   void RemoveOutdatedSources(int64_t now_ms);
> 
>   Clock* clock_;
>@@ -87,26 +66,20 @@ class RtpReceiverImpl : public RtpReceiver {
>       RTC_PT_GUARDED_BY(critical_section_rtp_receiver_);
>   const std::unique_ptr<RTPReceiverStrategy> rtp_media_receiver_;
> 
>-  RtpFeedback* const cb_rtp_feedback_;
>-
>   // SSRCs.
>   uint32_t ssrc_ RTC_GUARDED_BY(critical_section_rtp_receiver_);
>-  uint8_t num_csrcs_ RTC_GUARDED_BY(critical_section_rtp_receiver_);
>-  uint32_t current_remote_csrc_[kRtpCsrcSize] RTC_GUARDED_BY(
>-      critical_section_rtp_receiver_);
>+
>+  ContributingSources csrcs_ RTC_GUARDED_BY(critical_section_rtp_receiver_);
> 
>   // Sequence number and timestamps for the latest in-order packet.
>-  rtc::Optional<uint16_t> last_received_sequence_number_
>+  absl::optional<uint16_t> last_received_sequence_number_
>       RTC_GUARDED_BY(critical_section_rtp_receiver_);
>   uint32_t last_received_timestamp_
>       RTC_GUARDED_BY(critical_section_rtp_receiver_);
>   int64_t last_received_frame_time_ms_
>       RTC_GUARDED_BY(critical_section_rtp_receiver_);
> 
>-  std::unordered_map<uint32_t, std::list<RtpSource>::iterator>
>-      iterator_by_csrc_;
>   // The RtpSource objects are sorted chronologically.
>-  std::list<RtpSource> csrc_sources_;
>   std::vector<RtpSource> ssrc_sources_;
> };
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.cc
>index 6db24c9a021..647eceaf227 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.cc
>@@ -17,29 +17,6 @@ namespace webrtc {
> RTPReceiverStrategy::RTPReceiverStrategy(RtpData* data_callback)
>     : data_callback_(data_callback) {}
> 
>-void RTPReceiverStrategy::GetLastMediaSpecificPayload(
>-    PayloadUnion* payload) const {
>-  rtc::CritScope cs(&crit_sect_);
>-  if (last_payload_) {
>-    *payload = *last_payload_;
>-  }
>-}
>-
>-void RTPReceiverStrategy::SetLastMediaSpecificPayload(
>-    const PayloadUnion& payload) {
>-  rtc::CritScope cs(&crit_sect_);
>-  last_payload_.emplace(payload);
>-}
>-
>-void RTPReceiverStrategy::CheckPayloadChanged(int8_t payload_type,
>-                                              PayloadUnion* specific_payload,
>-                                              bool* should_discard_changes) {
>-  // Default: Keep changes.
>-  *should_discard_changes = false;
>-}
>-
>-int RTPReceiverStrategy::Energy(uint8_t array_of_energy[kRtpCsrcSize]) const {
>-  return -1;
>-}
>+RTPReceiverStrategy::~RTPReceiverStrategy() = default;
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h
>index 3288329977f..987af7c41aa 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h
>@@ -15,14 +15,11 @@
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> #include "modules/rtp_rtcp/source/rtp_utility.h"
> #include "rtc_base/criticalsection.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
> struct CodecInst;
> 
>-class TelephoneEventHandler;
>-
> // This strategy deals with media-specific RTP packet processing.
> // This class is not thread-safe and must be protected by its caller.
> class RTPReceiverStrategy {
>@@ -30,7 +27,7 @@ class RTPReceiverStrategy {
>   static RTPReceiverStrategy* CreateVideoStrategy(RtpData* data_callback);
>   static RTPReceiverStrategy* CreateAudioStrategy(RtpData* data_callback);
> 
>-  virtual ~RTPReceiverStrategy() {}
>+  virtual ~RTPReceiverStrategy();
> 
>   // Parses the RTP packet and calls the data callback with the payload data.
>   // Implementations are encouraged to use the provided packet buffer and RTP
>@@ -39,46 +36,10 @@ class RTPReceiverStrategy {
>   // provides audio or video-specific data.
>   virtual int32_t ParseRtpPacket(WebRtcRTPHeader* rtp_header,
>                                  const PayloadUnion& specific_payload,
>-                                 bool is_red,
>                                  const uint8_t* payload,
>                                  size_t payload_length,
>                                  int64_t timestamp_ms) = 0;
> 
>-  virtual TelephoneEventHandler* GetTelephoneEventHandler() = 0;
>-
>-  // Computes the current dead-or-alive state.
>-  virtual RTPAliveType ProcessDeadOrAlive(
>-      uint16_t last_payload_length) const = 0;
>-
>-  // Returns true if we should report CSRC changes for this payload type.
>-  // TODO(phoglund): should move out of here along with other payload stuff.
>-  virtual bool ShouldReportCsrcChanges(uint8_t payload_type) const = 0;
>-
>-  // Notifies the strategy that we have created a new non-RED audio payload type
>-  // in the payload registry.
>-  virtual int32_t OnNewPayloadTypeCreated(
>-      int payload_type,
>-      const SdpAudioFormat& audio_format) = 0;
>-
>-  // Invokes the OnInitializeDecoder callback in a media-specific way.
>-  virtual int32_t InvokeOnInitializeDecoder(
>-      RtpFeedback* callback,
>-      int8_t payload_type,
>-      const char payload_name[RTP_PAYLOAD_NAME_SIZE],
>-      const PayloadUnion& specific_payload) const = 0;
>-
>-  // Checks if the payload type has changed, and returns whether we should
>-  // reset statistics and/or discard this packet.
>-  virtual void CheckPayloadChanged(int8_t payload_type,
>-                                   PayloadUnion* specific_payload,
>-                                   bool* should_discard_changes);
>-
>-  virtual int Energy(uint8_t array_of_energy[kRtpCsrcSize]) const;
>-
>-  // Stores / retrieves the last media specific payload for later reference.
>-  void GetLastMediaSpecificPayload(PayloadUnion* payload) const;
>-  void SetLastMediaSpecificPayload(const PayloadUnion& payload);
>-
>  protected:
>   // The data callback is where we should send received payload data.
>   // See ParseRtpPacket. This class does not claim ownership of the callback.
>@@ -91,7 +52,6 @@ class RTPReceiverStrategy {
>   explicit RTPReceiverStrategy(RtpData* data_callback);
> 
>   rtc::CriticalSection crit_sect_;
>-  rtc::Optional<PayloadUnion> last_payload_;
>   RtpData* data_callback_;
> };
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_receiver_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_receiver_unittest.cc
>index 3d23b50c50a..1400288ee95 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_receiver_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_receiver_unittest.cc
>@@ -16,7 +16,6 @@
> #include "modules/rtp_rtcp/include/rtp_receiver.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> #include "modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
>-#include "modules/rtp_rtcp/source/rtp_receiver_impl.h"
> #include "test/gmock.h"
> #include "test/gtest.h"
> 
>@@ -48,7 +47,6 @@ class RtpReceiverTest : public ::testing::Test {
>         rtp_receiver_(
>             RtpReceiver::CreateAudioReceiver(&fake_clock_,
>                                              &mock_rtp_data_,
>-                                             nullptr,
>                                              &rtp_payload_registry_)) {
>     rtp_receiver_->RegisterReceivePayload(kPcmuPayloadType,
>                                           SdpAudioFormat("PCMU", 8000, 1));
>@@ -236,20 +234,11 @@ TEST_F(RtpReceiverTest, GetSourcesRemoveOutdatedSource) {
>   header.arrOfCSRCs[0] = kCsrc1;
>   EXPECT_TRUE(rtp_receiver_->IncomingRtpPacket(
>       header, kTestPayload, sizeof(kTestPayload), payload_specific));
>-  auto rtp_receiver_impl = static_cast<RtpReceiverImpl*>(rtp_receiver_.get());
>-  auto ssrc_sources = rtp_receiver_impl->ssrc_sources_for_testing();
>-  ASSERT_EQ(1u, ssrc_sources.size());
>-  EXPECT_EQ(kSsrc1, ssrc_sources.begin()->source_id());
>-  EXPECT_EQ(RtpSourceType::SSRC, ssrc_sources.begin()->source_type());
>-  EXPECT_EQ(fake_clock_.TimeInMilliseconds(),
>-            ssrc_sources.begin()->timestamp_ms());
>-
>-  auto csrc_sources = rtp_receiver_impl->csrc_sources_for_testing();
>-  ASSERT_EQ(1u, csrc_sources.size());
>-  EXPECT_EQ(kCsrc1, csrc_sources.begin()->source_id());
>-  EXPECT_EQ(RtpSourceType::CSRC, csrc_sources.begin()->source_type());
>-  EXPECT_EQ(fake_clock_.TimeInMilliseconds(),
>-            csrc_sources.begin()->timestamp_ms());
>+  now_ms = fake_clock_.TimeInMilliseconds();
>+  sources = rtp_receiver_->GetSources();
>+  EXPECT_THAT(sources, UnorderedElementsAre(
>+                           RtpSource(now_ms, kSsrc1, RtpSourceType::SSRC),
>+                           RtpSource(now_ms, kCsrc1, RtpSourceType::CSRC)));
> }
> 
> // The audio level from the RTPHeader extension should be stored in the
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
>index 65d1831e8d2..1101becfb1a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
>@@ -32,34 +32,16 @@ RTPReceiverStrategy* RTPReceiverStrategy::CreateVideoStrategy(
> }
> 
> RTPReceiverVideo::RTPReceiverVideo(RtpData* data_callback)
>-    : RTPReceiverStrategy(data_callback) {
>-}
>-
>-RTPReceiverVideo::~RTPReceiverVideo() {
>-}
>-
>-bool RTPReceiverVideo::ShouldReportCsrcChanges(uint8_t payload_type) const {
>-  // Always do this for video packets.
>-  return true;
>-}
>+    : RTPReceiverStrategy(data_callback) {}
> 
>-int32_t RTPReceiverVideo::OnNewPayloadTypeCreated(
>-    int payload_type,
>-    const SdpAudioFormat& audio_format) {
>-  RTC_NOTREACHED();
>-  return 0;
>-}
>+RTPReceiverVideo::~RTPReceiverVideo() {}
> 
> int32_t RTPReceiverVideo::ParseRtpPacket(WebRtcRTPHeader* rtp_header,
>                                          const PayloadUnion& specific_payload,
>-                                         bool is_red,
>                                          const uint8_t* payload,
>                                          size_t payload_length,
>                                          int64_t timestamp_ms) {
>-  TRACE_EVENT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "Video::ParseRtp",
>-               "seqnum", rtp_header->header.sequenceNumber, "timestamp",
>-               rtp_header->header.timestamp);
>-  rtp_header->type.Video.codec =
>+  rtp_header->video_header().codec =
>       specific_payload.video_payload().videoCodecType;
> 
>   RTC_DCHECK_GE(payload_length, rtp_header->header.paddingLength);
>@@ -77,7 +59,7 @@ int32_t RTPReceiverVideo::ParseRtpPacket(WebRtcRTPHeader* rtp_header,
> 
>   // We are not allowed to hold a critical section when calling below functions.
>   std::unique_ptr<RtpDepacketizer> depacketizer(
>-      RtpDepacketizer::Create(rtp_header->type.Video.codec));
>+      RtpDepacketizer::Create(rtp_header->video_header().codec));
>   if (depacketizer.get() == NULL) {
>     RTC_LOG(LS_ERROR) << "Failed to create depacketizer.";
>     return -1;
>@@ -88,28 +70,28 @@ int32_t RTPReceiverVideo::ParseRtpPacket(WebRtcRTPHeader* rtp_header,
>     return -1;
> 
>   rtp_header->frameType = parsed_payload.frame_type;
>-  rtp_header->type = parsed_payload.type;
>-  rtp_header->type.Video.rotation = kVideoRotation_0;
>-  rtp_header->type.Video.content_type = VideoContentType::UNSPECIFIED;
>-  rtp_header->type.Video.video_timing.flags = TimingFrameFlags::kInvalid;
>+  rtp_header->video_header() = parsed_payload.video_header();
>+  rtp_header->video_header().rotation = kVideoRotation_0;
>+  rtp_header->video_header().content_type = VideoContentType::UNSPECIFIED;
>+  rtp_header->video_header().video_timing.flags = VideoSendTiming::kInvalid;
> 
>   // Retrieve the video rotation information.
>   if (rtp_header->header.extension.hasVideoRotation) {
>-    rtp_header->type.Video.rotation =
>+    rtp_header->video_header().rotation =
>         rtp_header->header.extension.videoRotation;
>   }
> 
>   if (rtp_header->header.extension.hasVideoContentType) {
>-    rtp_header->type.Video.content_type =
>+    rtp_header->video_header().content_type =
>         rtp_header->header.extension.videoContentType;
>   }
> 
>   if (rtp_header->header.extension.has_video_timing) {
>-    rtp_header->type.Video.video_timing =
>+    rtp_header->video_header().video_timing =
>         rtp_header->header.extension.video_timing;
>   }
> 
>-  rtp_header->type.Video.playout_delay =
>+  rtp_header->video_header().playout_delay =
>       rtp_header->header.extension.playout_delay;
> 
>   return data_callback_->OnReceivedPayloadData(parsed_payload.payload,
>@@ -119,19 +101,4 @@ int32_t RTPReceiverVideo::ParseRtpPacket(WebRtcRTPHeader* rtp_header,
>              : -1;
> }
> 
>-RTPAliveType RTPReceiverVideo::ProcessDeadOrAlive(
>-    uint16_t last_payload_length) const {
>-  return kRtpDead;
>-}
>-
>-int32_t RTPReceiverVideo::InvokeOnInitializeDecoder(
>-    RtpFeedback* callback,
>-    int8_t payload_type,
>-    const char payload_name[RTP_PAYLOAD_NAME_SIZE],
>-    const PayloadUnion& specific_payload) const {
>-  // TODO(pbos): Remove as soon as audio can handle a changing payload type
>-  // without this callback.
>-  return 0;
>-}
>-
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h
>index f1e14f4d407..46b97f5033e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h
>@@ -15,7 +15,6 @@
> #include "modules/rtp_rtcp/source/rtp_receiver_strategy.h"
> #include "modules/rtp_rtcp/source/rtp_utility.h"
> #include "rtc_base/onetimeevent.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -23,30 +22,14 @@ class RTPReceiverVideo : public RTPReceiverStrategy {
>  public:
>   explicit RTPReceiverVideo(RtpData* data_callback);
> 
>-  virtual ~RTPReceiverVideo();
>+  ~RTPReceiverVideo() override;
> 
>   int32_t ParseRtpPacket(WebRtcRTPHeader* rtp_header,
>                          const PayloadUnion& specific_payload,
>-                         bool is_red,
>                          const uint8_t* packet,
>                          size_t packet_length,
>                          int64_t timestamp) override;
> 
>-  TelephoneEventHandler* GetTelephoneEventHandler() override { return NULL; }
>-
>-  RTPAliveType ProcessDeadOrAlive(uint16_t last_payload_length) const override;
>-
>-  bool ShouldReportCsrcChanges(uint8_t payload_type) const override;
>-
>-  int32_t OnNewPayloadTypeCreated(int payload_type,
>-                                  const SdpAudioFormat& audio_format) override;
>-
>-  int32_t InvokeOnInitializeDecoder(
>-      RtpFeedback* callback,
>-      int8_t payload_type,
>-      const char payload_name[RTP_PAYLOAD_NAME_SIZE],
>-      const PayloadUnion& specific_payload) const override;
>-
>   void SetPacketOverHead(uint16_t packet_over_head);
> 
>  private:
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h
>index 09deef35cd1..6863c4c353e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h
>@@ -13,33 +13,11 @@
> 
> // Configuration file for RTP utilities (RTPSender, RTPReceiver ...)
> namespace webrtc {
>-enum { NACK_BYTECOUNT_SIZE = 60 };  // size of our NACK history
>-// A sanity for the NACK list parsing at the send-side.
>-enum { kSendSideNackListSizeSanity = 20000 };
> enum { kDefaultMaxReorderingThreshold = 50 };  // In sequence numbers.
> enum { kRtcpMaxNackFields = 253 };
> 
>-enum { RTCP_INTERVAL_VIDEO_MS = 1000 };
>-enum { RTCP_INTERVAL_AUDIO_MS = 5000 };
> enum { RTCP_SEND_BEFORE_KEY_FRAME_MS = 100 };
> enum { RTCP_MAX_REPORT_BLOCKS = 31 };  // RFC 3550 page 37
>-enum {
>-  kRtcpAppCode_DATA_SIZE = 32 * 4
>-};  // multiple of 4, this is not a limitation of the size
>-enum { RTCP_NUMBER_OF_SR = 60 };
>-
>-enum { MAX_NUMBER_OF_TEMPORAL_ID = 8 };              // RFC
>-enum { MAX_NUMBER_OF_DEPENDENCY_QUALITY_ID = 128 };  // RFC
>-enum { MAX_NUMBER_OF_REMB_FEEDBACK_SSRCS = 255 };
>-
>-enum { BW_HISTORY_SIZE = 35 };
>-
>-#define MIN_AUDIO_BW_MANAGEMENT_BITRATE 6
>-#define MIN_VIDEO_BW_MANAGEMENT_BITRATE 30
>-
>-enum { RTP_MAX_BURST_SLEEP_TIME = 500 };
>-enum { RTP_AUDIO_LEVEL_UNIQUE_ID = 0xbede };
>-enum { RTP_MAX_PACKETS_PER_FRAME = 512 };  // must be multiple of 32
> }  // namespace webrtc
> 
> #endif  // MODULES_RTP_RTCP_SOURCE_RTP_RTCP_CONFIG_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
>index 629bf8a4305..fe8dbf385c8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
>@@ -17,7 +17,6 @@
> #include <string>
> 
> #include "api/rtpparameters.h"
>-#include "common_types.h"  // NOLINT(build/include)
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> 
>@@ -51,6 +50,8 @@ RTPExtensionType StringToRtpExtensionType(const std::string& extension) {
>     return kRtpExtensionVideoContentType;
>   if (extension == RtpExtension::kVideoTimingUri)
>     return kRtpExtensionVideoTiming;
>+  if (extension == RtpExtension::kMidUri)
>+    return kRtpExtensionMid;
>   RTC_NOTREACHED() << "Looking up unsupported RTP extension.";
>   return kRtpExtensionNone;
> }
>@@ -63,8 +64,7 @@ RtpRtcp* RtpRtcp::CreateRtpRtcp(const RtpRtcp::Configuration& configuration) {
>   } else {
>     // No clock implementation provided, use default clock.
>     RtpRtcp::Configuration configuration_copy;
>-    memcpy(&configuration_copy, &configuration,
>-           sizeof(RtpRtcp::Configuration));
>+    memcpy(&configuration_copy, &configuration, sizeof(RtpRtcp::Configuration));
>     configuration_copy.clock = Clock::GetRealTimeClock();
>     return new ModuleRtpRtcpImpl(configuration_copy);
>   }
>@@ -84,7 +84,8 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
>                    configuration.receive_statistics,
>                    configuration.rtcp_packet_type_counter_observer,
>                    configuration.event_log,
>-                   configuration.outgoing_transport),
>+                   configuration.outgoing_transport,
>+                   configuration.rtcp_interval_config),
>       rtcp_receiver_(configuration.clock,
>                      configuration.receiver_only,
>                      configuration.rtcp_packet_type_counter_observer,
>@@ -111,20 +112,18 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
>       rtt_ms_(0) {
>   if (!configuration.receiver_only) {
>     rtp_sender_.reset(new RTPSender(
>-        configuration.audio,
>-        configuration.clock,
>-        configuration.outgoing_transport,
>-        configuration.paced_sender,
>+        configuration.audio, configuration.clock,
>+        configuration.outgoing_transport, configuration.paced_sender,
>         configuration.flexfec_sender,
>         configuration.transport_sequence_number_allocator,
>         configuration.transport_feedback_callback,
>         configuration.send_bitrate_observer,
>         configuration.send_frame_count_observer,
>-        configuration.send_side_delay_observer,
>-        configuration.event_log,
>+        configuration.send_side_delay_observer, configuration.event_log,
>         configuration.send_packet_observer,
>         configuration.retransmission_rate_limiter,
>-        configuration.overhead_observer));
>+        configuration.overhead_observer,
>+        configuration.populate_network2_timestamp));
>     // Make sure rtcp sender use same timestamp offset as rtp sender.
>     rtcp_sender_.SetTimestampOffset(rtp_sender_->TimestampOffset());
> 
>@@ -141,6 +140,8 @@ ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
>   SetMaxRtpPacketSize(IP_PACKET_SIZE - kTcpOverIpv4HeaderSize);
> }
> 
>+ModuleRtpRtcpImpl::~ModuleRtpRtcpImpl() = default;
>+
> // Returns the number of milliseconds until the module want a worker thread
> // to call Process.
> int64_t ModuleRtpRtcpImpl::TimeUntilNextProcess() {
>@@ -219,10 +220,10 @@ void ModuleRtpRtcpImpl::Process() {
>   } else {
>     // Report rtt from receiver.
>     if (process_rtt) {
>-       int64_t rtt_ms;
>-       if (rtt_stats_ && rtcp_receiver_.GetAndResetXrRrRtt(&rtt_ms)) {
>-         rtt_stats_->OnRttUpdate(rtt_ms);
>-       }
>+      int64_t rtt_ms;
>+      if (rtt_stats_ && rtcp_receiver_.GetAndResetXrRrRtt(&rtt_ms)) {
>+        rtt_stats_->OnRttUpdate(rtt_ms);
>+      }
>     }
>   }
> 
>@@ -264,10 +265,10 @@ void ModuleRtpRtcpImpl::SetRtxSendPayloadType(int payload_type,
>   rtp_sender_->SetRtxPayloadType(payload_type, associated_payload_type);
> }
> 
>-rtc::Optional<uint32_t> ModuleRtpRtcpImpl::FlexfecSsrc() const {
>+absl::optional<uint32_t> ModuleRtpRtcpImpl::FlexfecSsrc() const {
>   if (rtp_sender_)
>     return rtp_sender_->FlexfecSsrc();
>-  return rtc::nullopt;
>+  return absl::nullopt;
> }
> 
> void ModuleRtpRtcpImpl::IncomingRtcpPacket(const uint8_t* rtcp_packet,
>@@ -275,18 +276,12 @@ void ModuleRtpRtcpImpl::IncomingRtcpPacket(const uint8_t* rtcp_packet,
>   rtcp_receiver_.IncomingPacket(rtcp_packet, length);
> }
> 
>-int32_t ModuleRtpRtcpImpl::RegisterSendPayload(
>-    const CodecInst& voice_codec) {
>+int32_t ModuleRtpRtcpImpl::RegisterSendPayload(const CodecInst& voice_codec) {
>   return rtp_sender_->RegisterPayload(
>       voice_codec.plname, voice_codec.pltype, voice_codec.plfreq,
>       voice_codec.channels, (voice_codec.rate < 0) ? 0 : voice_codec.rate);
> }
> 
>-int32_t ModuleRtpRtcpImpl::RegisterSendPayload(const VideoCodec& video_codec) {
>-  return rtp_sender_->RegisterPayload(video_codec.plName, video_codec.plType,
>-                                     90000, 0, 0);
>-}
>-
> void ModuleRtpRtcpImpl::RegisterVideoSendPayload(int payload_type,
>                                                  const char* payload_name) {
>   RTC_CHECK_EQ(
>@@ -345,6 +340,14 @@ void ModuleRtpRtcpImpl::SetSSRC(const uint32_t ssrc) {
>   SetRtcpReceiverSsrcs(ssrc);
> }
> 
>+void ModuleRtpRtcpImpl::SetMid(const std::string& mid) {
>+  if (rtp_sender_) {
>+    rtp_sender_->SetMid(mid);
>+  }
>+  // TODO(bugs.webrtc.org/4050): If we end up supporting the MID SDES item for
>+  // RTCP, this will need to be passed down to the RTCPSender also.
>+}
>+
> void ModuleRtpRtcpImpl::SetCsrcs(const std::vector<uint32_t>& csrcs) {
>   rtcp_sender_.SetCsrcs(csrcs);
>   rtp_sender_->SetCsrcs(csrcs);
>@@ -360,20 +363,18 @@ RTCPSender::FeedbackState ModuleRtpRtcpImpl::GetFeedbackState() {
>     StreamDataCounters rtp_stats;
>     StreamDataCounters rtx_stats;
>     rtp_sender_->GetDataCounters(&rtp_stats, &rtx_stats);
>-    state.packets_sent = rtp_stats.transmitted.packets +
>-                         rtx_stats.transmitted.packets;
>+    state.packets_sent =
>+        rtp_stats.transmitted.packets + rtx_stats.transmitted.packets;
>     state.media_bytes_sent = rtp_stats.transmitted.payload_bytes +
>                              rtx_stats.transmitted.payload_bytes;
>     state.send_bitrate = rtp_sender_->BitrateSent();
>   }
>   state.module = this;
> 
>-  LastReceivedNTP(&state.last_rr_ntp_secs,
>-                  &state.last_rr_ntp_frac,
>+  LastReceivedNTP(&state.last_rr_ntp_secs, &state.last_rr_ntp_frac,
>                   &state.remote_sr);
> 
>-  state.has_last_xr_rr =
>-      rtcp_receiver_.LastReceivedXrReferenceTimeInfo(&state.last_xr_rr);
>+  state.last_xr_rtis = rtcp_receiver_.ConsumeReceivedXrReferenceTimeInfo();
> 
>   return state;
> }
>@@ -428,7 +429,7 @@ bool ModuleRtpRtcpImpl::SendOutgoingData(
>   rtcp_sender_.SetLastRtpTime(time_stamp, capture_time_ms);
>   // Make sure an RTCP report isn't queued behind a key frame.
>   if (rtcp_sender_.TimeToSendRTCPReport(kVideoFrameKey == frame_type)) {
>-      rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpReport);
>+    rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpReport);
>   }
>   int64_t expected_retransmission_time_ms = rtt_ms();
>   if (expected_retransmission_time_ms == 0) {
>@@ -452,7 +453,7 @@ bool ModuleRtpRtcpImpl::TimeToSendPacket(uint32_t ssrc,
>                                          bool retransmission,
>                                          const PacedPacketInfo& pacing_info) {
>   return rtp_sender_->TimeToSendPacket(ssrc, sequence_number, capture_time_ms,
>-                                      retransmission, pacing_info);
>+                                       retransmission, pacing_info);
> }
> 
> size_t ModuleRtpRtcpImpl::TimeToSendPadding(
>@@ -497,22 +498,18 @@ int32_t ModuleRtpRtcpImpl::RemoveMixedCNAME(const uint32_t ssrc) {
>   return rtcp_sender_.RemoveMixedCNAME(ssrc);
> }
> 
>-int32_t ModuleRtpRtcpImpl::RemoteCNAME(
>-    const uint32_t remote_ssrc,
>-    char c_name[RTCP_CNAME_SIZE]) const {
>+int32_t ModuleRtpRtcpImpl::RemoteCNAME(const uint32_t remote_ssrc,
>+                                       char c_name[RTCP_CNAME_SIZE]) const {
>   return rtcp_receiver_.CNAME(remote_ssrc, c_name);
> }
> 
>-int32_t ModuleRtpRtcpImpl::RemoteNTP(
>-    uint32_t* received_ntpsecs,
>-    uint32_t* received_ntpfrac,
>-    uint32_t* rtcp_arrival_time_secs,
>-    uint32_t* rtcp_arrival_time_frac,
>-    uint32_t* rtcp_timestamp) const {
>-  return rtcp_receiver_.NTP(received_ntpsecs,
>-                            received_ntpfrac,
>-                            rtcp_arrival_time_secs,
>-                            rtcp_arrival_time_frac,
>+int32_t ModuleRtpRtcpImpl::RemoteNTP(uint32_t* received_ntpsecs,
>+                                     uint32_t* received_ntpfrac,
>+                                     uint32_t* rtcp_arrival_time_secs,
>+                                     uint32_t* rtcp_arrival_time_frac,
>+                                     uint32_t* rtcp_timestamp) const {
>+  return rtcp_receiver_.NTP(received_ntpsecs, received_ntpfrac,
>+                            rtcp_arrival_time_secs, rtcp_arrival_time_frac,
>                             rtcp_timestamp)
>              ? 0
>              : -1;
>@@ -550,13 +547,13 @@ int32_t ModuleRtpRtcpImpl::SetRTCPApplicationSpecificData(
>     const uint32_t name,
>     const uint8_t* data,
>     const uint16_t length) {
>-  return  rtcp_sender_.SetApplicationSpecificData(sub_type, name, data, length);
>+  return rtcp_sender_.SetApplicationSpecificData(sub_type, name, data, length);
> }
> 
> // (XR) VOIP metric.
> int32_t ModuleRtpRtcpImpl::SetRTCPVoIPMetrics(
>-  const RTCPVoIPMetric* voip_metric) {
>-  return  rtcp_sender_.SetRTCPVoIPMetrics(voip_metric);
>+    const RTCPVoIPMetric* voip_metric) {
>+  return rtcp_sender_.SetRTCPVoIPMetrics(voip_metric);
> }
> 
> void ModuleRtpRtcpImpl::SetRtcpXrRrtrStatus(bool enable) {
>@@ -569,9 +566,8 @@ bool ModuleRtpRtcpImpl::RtcpXrRrtrStatus() const {
> }
> 
> // TODO(asapersson): Replace this method with the one below.
>-int32_t ModuleRtpRtcpImpl::DataCountersRTP(
>-    size_t* bytes_sent,
>-    uint32_t* packets_sent) const {
>+int32_t ModuleRtpRtcpImpl::DataCountersRTP(size_t* bytes_sent,
>+                                           uint32_t* packets_sent) const {
>   StreamDataCounters rtp_stats;
>   StreamDataCounters rtx_stats;
>   rtp_sender_->GetDataCounters(&rtp_stats, &rtx_stats);
>@@ -585,8 +581,8 @@ int32_t ModuleRtpRtcpImpl::DataCountersRTP(
>                   rtx_stats.transmitted.header_bytes;
>   }
>   if (packets_sent) {
>-    *packets_sent = rtp_stats.transmitted.packets +
>-                    rtx_stats.transmitted.packets;
>+    *packets_sent =
>+        rtp_stats.transmitted.packets + rtx_stats.transmitted.packets;
>   }
>   return 0;
> }
>@@ -601,7 +597,8 @@ void ModuleRtpRtcpImpl::GetRtpPacketLossStats(
>     bool outgoing,
>     uint32_t ssrc,
>     struct RtpPacketLossStats* loss_stats) const {
>-  if (!loss_stats) return;
>+  if (!loss_stats)
>+    return;
>   const PacketLossStats* stats_source = NULL;
>   if (outgoing) {
>     if (SSRC() == ssrc) {
>@@ -613,8 +610,7 @@ void ModuleRtpRtcpImpl::GetRtpPacketLossStats(
>     }
>   }
>   if (stats_source) {
>-    loss_stats->single_packet_loss_count =
>-        stats_source->GetSingleLossCount();
>+    loss_stats->single_packet_loss_count = stats_source->GetSingleLossCount();
>     loss_stats->multiple_packet_loss_event_count =
>         stats_source->GetMultipleLossEventCount();
>     loss_stats->multiple_packet_loss_packet_count =
>@@ -772,15 +768,13 @@ bool ModuleRtpRtcpImpl::SendFeedbackPacket(
> }
> 
> // Send a TelephoneEvent tone using RFC 2833 (4733).
>-int32_t ModuleRtpRtcpImpl::SendTelephoneEventOutband(
>-    const uint8_t key,
>-    const uint16_t time_ms,
>-    const uint8_t level) {
>+int32_t ModuleRtpRtcpImpl::SendTelephoneEventOutband(const uint8_t key,
>+                                                     const uint16_t time_ms,
>+                                                     const uint8_t level) {
>   return rtp_sender_->SendTelephoneEvent(key, time_ms, level);
> }
> 
>-int32_t ModuleRtpRtcpImpl::SetAudioLevel(
>-    const uint8_t level_d_bov) {
>+int32_t ModuleRtpRtcpImpl::SetAudioLevel(const uint8_t level_d_bov) {
>   return rtp_sender_->SetAudioLevel(level_d_bov);
> }
> 
>@@ -839,8 +833,7 @@ void ModuleRtpRtcpImpl::OnReceivedNack(
>   for (uint16_t nack_sequence_number : nack_sequence_numbers) {
>     send_loss_stats_.AddLostPacket(nack_sequence_number);
>   }
>-  if (!rtp_sender_->StorePackets() ||
>-      nack_sequence_numbers.size() == 0) {
>+  if (!rtp_sender_->StorePackets() || nack_sequence_numbers.size() == 0) {
>     return;
>   }
>   // Use RTT from RtcpRttStats class if provided.
>@@ -865,11 +858,8 @@ bool ModuleRtpRtcpImpl::LastReceivedNTP(
>   uint32_t ntp_secs = 0;
>   uint32_t ntp_frac = 0;
> 
>-  if (!rtcp_receiver_.NTP(&ntp_secs,
>-                          &ntp_frac,
>-                          rtcp_arrival_time_secs,
>-                          rtcp_arrival_time_frac,
>-                          NULL)) {
>+  if (!rtcp_receiver_.NTP(&ntp_secs, &ntp_frac, rtcp_arrival_time_secs,
>+                          rtcp_arrival_time_frac, NULL)) {
>     return false;
>   }
>   *remote_sr =
>@@ -884,9 +874,9 @@ std::vector<rtcp::TmmbItem> ModuleRtpRtcpImpl::BoundingSet(bool* tmmbr_owner) {
> 
> int64_t ModuleRtpRtcpImpl::RtcpReportInterval() {
>   if (audio_)
>-    return RTCP_INTERVAL_AUDIO_MS;
>+    return rtcp_sender_.RtcpAudioReportInverval();
>   else
>-    return RTCP_INTERVAL_VIDEO_MS;
>+    return rtcp_sender_.RtcpVideoReportInverval();
> }
> 
> void ModuleRtpRtcpImpl::SetRtcpReceiverSsrcs(uint32_t main_ssrc) {
>@@ -894,7 +884,7 @@ void ModuleRtpRtcpImpl::SetRtcpReceiverSsrcs(uint32_t main_ssrc) {
>   ssrcs.insert(main_ssrc);
>   if (RtxSendStatus() != kRtxOff)
>     ssrcs.insert(rtp_sender_->RtxSsrc());
>-  rtc::Optional<uint32_t> flexfec_ssrc = FlexfecSsrc();
>+  absl::optional<uint32_t> flexfec_ssrc = FlexfecSsrc();
>   if (flexfec_ssrc)
>     ssrcs.insert(*flexfec_ssrc);
>   rtcp_receiver_.SetSsrcs(main_ssrc, ssrcs);
>@@ -918,12 +908,12 @@ void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback(
> }
> 
> StreamDataCountersCallback*
>-    ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
>+ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
>   return rtp_sender_->GetRtpStatisticsCallback();
> }
> 
> void ModuleRtpRtcpImpl::SetVideoBitrateAllocation(
>-    const BitrateAllocation& bitrate) {
>+    const VideoBitrateAllocation& bitrate) {
>   rtcp_sender_.SetVideoBitrateAllocation(bitrate);
> }
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
>index 0bb9edee5bb..9f53416d2f3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h
>@@ -13,10 +13,12 @@
> 
> #include <memory>
> #include <set>
>+#include <string>
> #include <utility>
> #include <vector>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
>+#include "api/video/video_bitrate_allocation.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> #include "modules/rtp_rtcp/source/packet_loss_stats.h"
>@@ -31,6 +33,7 @@ namespace webrtc {
> class ModuleRtpRtcpImpl : public RtpRtcp, public RTCPReceiver::ModuleRtpRtcp {
>  public:
>   explicit ModuleRtpRtcpImpl(const RtpRtcp::Configuration& configuration);
>+  ~ModuleRtpRtcpImpl() override;
> 
>   // Returns the number of milliseconds until the module want a worker thread to
>   // call Process.
>@@ -51,8 +54,6 @@ class ModuleRtpRtcpImpl : public RtpRtcp, public RTCPReceiver::ModuleRtpRtcp {
> 
>   int32_t RegisterSendPayload(const CodecInst& voice_codec) override;
> 
>-  int32_t RegisterSendPayload(const VideoCodec& video_codec) override;
>-
>   void RegisterVideoSendPayload(int payload_type,
>                                 const char* payload_name) override;
> 
>@@ -87,6 +88,8 @@ class ModuleRtpRtcpImpl : public RtpRtcp, public RTCPReceiver::ModuleRtpRtcp {
>   // Configure SSRC, default is a random number.
>   void SetSSRC(uint32_t ssrc) override;
> 
>+  void SetMid(const std::string& mid) override;
>+
>   void SetCsrcs(const std::vector<uint32_t>& csrcs) override;
> 
>   RTCPSender::FeedbackState GetFeedbackState();
>@@ -99,7 +102,7 @@ class ModuleRtpRtcpImpl : public RtpRtcp, public RTCPReceiver::ModuleRtpRtcp {
>   void SetRtxSendPayloadType(int payload_type,
>                              int associated_payload_type) override;
> 
>-  rtc::Optional<uint32_t> FlexfecSsrc() const override;
>+  absl::optional<uint32_t> FlexfecSsrc() const override;
> 
>   // Sends kRtcpByeCode when going from true to false.
>   int32_t SetSendingStatus(bool sending) override;
>@@ -290,7 +293,8 @@ class ModuleRtpRtcpImpl : public RtpRtcp, public RTCPReceiver::ModuleRtpRtcp {
>       const ReportBlockList& report_blocks) override;
>   void OnRequestSendReport() override;
> 
>-  void SetVideoBitrateAllocation(const BitrateAllocation& bitrate) override;
>+  void SetVideoBitrateAllocation(
>+      const VideoBitrateAllocation& bitrate) override;
> 
>  protected:
>   bool UpdateRTCPReceiveInformationTimers();
>@@ -341,7 +345,7 @@ class ModuleRtpRtcpImpl : public RtpRtcp, public RTCPReceiver::ModuleRtpRtcp {
> 
>   RemoteBitrateEstimator* remote_bitrate_;
> 
>-  RtcpRttStats* rtt_stats_;
>+  RtcpRttStats* const rtt_stats_;
> 
>   PacketLossStats send_loss_stats_;
>   PacketLossStats receive_loss_stats_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
>index ad3ec0bbede..5071eeeecf1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc
>@@ -12,7 +12,7 @@
> #include <memory>
> #include <set>
> 
>-#include "common_types.h"  // NOLINT(build/include)
>+#include "api/video_codecs/video_codec.h"
> #include "modules/rtp_rtcp/include/rtp_header_parser.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> #include "modules/rtp_rtcp/source/rtcp_packet.h"
>@@ -42,27 +42,25 @@ const int64_t kMaxRttMs = 1000;
> class RtcpRttStatsTestImpl : public RtcpRttStats {
>  public:
>   RtcpRttStatsTestImpl() : rtt_ms_(0) {}
>-  virtual ~RtcpRttStatsTestImpl() {}
>+  ~RtcpRttStatsTestImpl() override = default;
> 
>   void OnRttUpdate(int64_t rtt_ms) override { rtt_ms_ = rtt_ms; }
>   int64_t LastProcessedRtt() const override { return rtt_ms_; }
>   int64_t rtt_ms_;
> };
> 
>-class SendTransport : public Transport,
>-                      public RtpData {
>+class SendTransport : public Transport, public RtpData {
>  public:
>   SendTransport()
>       : receiver_(nullptr),
>         clock_(nullptr),
>         delay_ms_(0),
>         rtp_packets_sent_(0),
>+        rtcp_packets_sent_(0),
>         keepalive_payload_type_(0),
>         num_keepalive_sent_(0) {}
> 
>-  void SetRtpRtcpModule(ModuleRtpRtcpImpl* receiver) {
>-    receiver_ = receiver;
>-  }
>+  void SetRtpRtcpModule(ModuleRtpRtcpImpl* receiver) { receiver_ = receiver; }
>   void SimulateNetworkDelay(int64_t delay_ms, SimulatedClock* clock) {
>     clock_ = clock;
>     delay_ms_ = delay_ms;
>@@ -89,6 +87,7 @@ class SendTransport : public Transport,
>     }
>     EXPECT_TRUE(receiver_);
>     receiver_->IncomingRtcpPacket(data, len);
>+    ++rtcp_packets_sent_;
>     return true;
>   }
>   int32_t OnReceivedPayloadData(const uint8_t* payload_data,
>@@ -100,10 +99,12 @@ class SendTransport : public Transport,
>     keepalive_payload_type_ = payload_type;
>   }
>   size_t NumKeepaliveSent() { return num_keepalive_sent_; }
>+  size_t NumRtcpSent() { return rtcp_packets_sent_; }
>   ModuleRtpRtcpImpl* receiver_;
>   SimulatedClock* clock_;
>   int64_t delay_ms_;
>   int rtp_packets_sent_;
>+  size_t rtcp_packets_sent_;
>   RTPHeader last_rtp_header_;
>   std::vector<uint16_t> last_nack_list_;
>   uint8_t keepalive_payload_type_;
>@@ -130,6 +131,7 @@ class RtpRtcpModule : public RtcpPacketTypeCounterObserver {
>   uint32_t remote_ssrc_;
>   RateLimiter retransmission_rate_limiter_;
>   RtpKeepAliveConfig keepalive_config_;
>+  RtcpIntervalConfig rtcp_interval_config_;
> 
>   void SetRemoteSsrc(uint32_t ssrc) {
>     remote_ssrc_ = ssrc;
>@@ -151,9 +153,7 @@ class RtpRtcpModule : public RtcpPacketTypeCounterObserver {
>     // Received RTCP stats for (own) local SSRC.
>     return counter_map_[impl_->SSRC()];
>   }
>-  int RtpSent() {
>-    return transport_.rtp_packets_sent_;
>-  }
>+  int RtpSent() { return transport_.rtp_packets_sent_; }
>   uint16_t LastRtpSequenceNumber() {
>     return transport_.last_rtp_header_.sequenceNumber;
>   }
>@@ -166,6 +166,10 @@ class RtpRtcpModule : public RtcpPacketTypeCounterObserver {
>     CreateModuleImpl();
>     transport_.SetKeepalivePayloadType(config.payload_type);
>   }
>+  void SetRtcpIntervalConfigAndReset(const RtcpIntervalConfig& config) {
>+    rtcp_interval_config_ = config;
>+    CreateModuleImpl();
>+  }
> 
>  private:
>   void CreateModuleImpl() {
>@@ -178,6 +182,7 @@ class RtpRtcpModule : public RtcpPacketTypeCounterObserver {
>     config.rtt_stats = &rtt_stats_;
>     config.retransmission_rate_limiter = &retransmission_rate_limiter_;
>     config.keepalive_config = keepalive_config_;
>+    config.rtcp_interval_config = rtcp_interval_config_;
> 
>     impl_.reset(new ModuleRtpRtcpImpl(config));
>     impl_->SetRTCPStatus(RtcpMode::kCompound);
>@@ -204,10 +209,9 @@ class RtpRtcpImplTest : public ::testing::Test {
> 
>     memset(&codec_, 0, sizeof(VideoCodec));
>     codec_.plType = 100;
>-    strncpy(codec_.plName, "VP8", 3);
>     codec_.width = 320;
>     codec_.height = 180;
>-    EXPECT_EQ(0, sender_.impl_->RegisterSendPayload(codec_));
>+    sender_.impl_->RegisterVideoSendPayload(codec_.plType, "VP8");
> 
>     // Receive module.
>     EXPECT_EQ(0, receiver_.impl_->SetSendingStatus(false));
>@@ -235,14 +239,14 @@ class RtpRtcpImplTest : public ::testing::Test {
>     rtp_video_header.playout_delay = {-1, -1};
>     rtp_video_header.is_first_packet_in_frame = true;
>     rtp_video_header.simulcastIdx = 0;
>-    rtp_video_header.codec = kRtpVideoVp8;
>-    rtp_video_header.codecHeader = {vp8_header};
>+    rtp_video_header.codec = kVideoCodecVP8;
>+    rtp_video_header.vp8() = vp8_header;
>     rtp_video_header.video_timing = {0u, 0u, 0u, 0u, 0u, 0u, false};
> 
>     const uint8_t payload[100] = {0};
>     EXPECT_EQ(true, module->impl_->SendOutgoingData(
>-                     kVideoFrameKey, codec_.plType, 0, 0, payload,
>-                     sizeof(payload), nullptr, &rtp_video_header, nullptr));
>+                        kVideoFrameKey, codec_.plType, 0, 0, payload,
>+                        sizeof(payload), nullptr, &rtp_video_header, nullptr));
>   }
> 
>   void IncomingRtcpNack(const RtpRtcpModule* module, uint16_t sequence_number) {
>@@ -339,16 +343,16 @@ TEST_F(RtpRtcpImplTest, Rtt) {
>   int64_t avg_rtt;
>   int64_t min_rtt;
>   int64_t max_rtt;
>-  EXPECT_EQ(0,
>-      sender_.impl_->RTT(kReceiverSsrc, &rtt, &avg_rtt, &min_rtt, &max_rtt));
>+  EXPECT_EQ(
>+      0, sender_.impl_->RTT(kReceiverSsrc, &rtt, &avg_rtt, &min_rtt, &max_rtt));
>   EXPECT_NEAR(2 * kOneWayNetworkDelayMs, rtt, 1);
>   EXPECT_NEAR(2 * kOneWayNetworkDelayMs, avg_rtt, 1);
>   EXPECT_NEAR(2 * kOneWayNetworkDelayMs, min_rtt, 1);
>   EXPECT_NEAR(2 * kOneWayNetworkDelayMs, max_rtt, 1);
> 
>   // No RTT from other ssrc.
>-  EXPECT_EQ(-1,
>-      sender_.impl_->RTT(kReceiverSsrc+1, &rtt, &avg_rtt, &min_rtt, &max_rtt));
>+  EXPECT_EQ(-1, sender_.impl_->RTT(kReceiverSsrc + 1, &rtt, &avg_rtt, &min_rtt,
>+                                   &max_rtt));
> 
>   // Verify RTT from rtt_stats config.
>   EXPECT_EQ(0, sender_.rtt_stats_.LastProcessedRtt());
>@@ -455,8 +459,8 @@ TEST_F(RtpRtcpImplTest, AddStreamDataCounters) {
>   rtp.transmitted.header_bytes = 2;
>   rtp.transmitted.padding_bytes = 3;
>   EXPECT_EQ(rtp.transmitted.TotalBytes(), rtp.transmitted.payload_bytes +
>-                                          rtp.transmitted.header_bytes +
>-                                          rtp.transmitted.padding_bytes);
>+                                              rtp.transmitted.header_bytes +
>+                                              rtp.transmitted.padding_bytes);
> 
>   StreamDataCounters rtp2;
>   rtp2.first_packet_time_ms = -1;
>@@ -643,4 +647,56 @@ TEST_F(RtpRtcpImplTest, SendsKeepaliveAfterTimout) {
>   sender_.impl_->Process();
>   EXPECT_EQ(3U, sender_.transport_.NumKeepaliveSent());
> }
>+
>+TEST_F(RtpRtcpImplTest, ConfigurableRtcpReportInterval) {
>+  const int kVideoReportInterval = 3000;
>+
>+  RtcpIntervalConfig config;
>+  config.video_interval_ms = kVideoReportInterval;
>+
>+  // Recreate sender impl with new configuration, and redo setup.
>+  sender_.SetRtcpIntervalConfigAndReset(config);
>+  SetUp();
>+
>+  SendFrame(&sender_, kBaseLayerTid);
>+
>+  // Initial state
>+  sender_.impl_->Process();
>+  EXPECT_EQ(sender_.RtcpSent().first_packet_time_ms, -1);
>+  EXPECT_EQ(0u, sender_.transport_.NumRtcpSent());
>+
>+  // Move ahead to the last ms before a rtcp is expected, no action.
>+  clock_.AdvanceTimeMilliseconds(kVideoReportInterval / 2 - 1);
>+  sender_.impl_->Process();
>+  EXPECT_EQ(sender_.RtcpSent().first_packet_time_ms, -1);
>+  EXPECT_EQ(sender_.transport_.NumRtcpSent(), 0u);
>+
>+  // Move ahead to the first rtcp. Send RTCP.
>+  clock_.AdvanceTimeMilliseconds(1);
>+  sender_.impl_->Process();
>+  EXPECT_GT(sender_.RtcpSent().first_packet_time_ms, -1);
>+  EXPECT_EQ(sender_.transport_.NumRtcpSent(), 1u);
>+
>+  SendFrame(&sender_, kBaseLayerTid);
>+
>+  // Move ahead to the last possible second before second rtcp is expected.
>+  clock_.AdvanceTimeMilliseconds(kVideoReportInterval * 1 / 2 - 1);
>+  sender_.impl_->Process();
>+  EXPECT_EQ(sender_.transport_.NumRtcpSent(), 1u);
>+
>+  // Move ahead into the range of second rtcp, the second rtcp may be sent.
>+  clock_.AdvanceTimeMilliseconds(1);
>+  sender_.impl_->Process();
>+  EXPECT_GE(sender_.transport_.NumRtcpSent(), 1u);
>+
>+  clock_.AdvanceTimeMilliseconds(kVideoReportInterval / 2);
>+  sender_.impl_->Process();
>+  EXPECT_GE(sender_.transport_.NumRtcpSent(), 1u);
>+
>+  // Move out the range of second rtcp, the second rtcp must have been sent.
>+  clock_.AdvanceTimeMilliseconds(kVideoReportInterval / 2);
>+  sender_.impl_->Process();
>+  EXPECT_EQ(sender_.transport_.NumRtcpSent(), 2u);
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
>index b5f57b120d6..e0347f043ff 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
>@@ -11,8 +11,10 @@
> #include "modules/rtp_rtcp/source/rtp_sender.h"
> 
> #include <algorithm>
>+#include <string>
> #include <utility>
> 
>+#include "absl/memory/memory.h"
> #include "logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.h"
> #include "logging/rtc_event_log/rtc_event_log.h"
> #include "modules/remote_bitrate_estimator/test/bwe_test_logging.h"
>@@ -28,7 +30,6 @@
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/numerics/safe_minmax.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/rate_limiter.h"
> #include "rtc_base/timeutils.h"
> #include "rtc_base/trace_event.h"
>@@ -54,21 +55,38 @@ constexpr RtpExtensionSize CreateExtensionSize() {
> }
> 
> // Size info for header extensions that might be used in padding or FEC packets.
>-constexpr RtpExtensionSize kExtensionSizes[] = {
>+constexpr RtpExtensionSize kFecOrPaddingExtensionSizes[] = {
>     CreateExtensionSize<AbsoluteSendTime>(),
>     CreateExtensionSize<TransmissionOffset>(),
>     CreateExtensionSize<TransportSequenceNumber>(),
>     CreateExtensionSize<PlayoutDelayLimits>(),
>+    {RtpMid::kId, RtpMid::kMaxValueSizeBytes},
>+};
>+
>+// Size info for header extensions that might be used in video packets.
>+constexpr RtpExtensionSize kVideoExtensionSizes[] = {
>+    CreateExtensionSize<AbsoluteSendTime>(),
>+    CreateExtensionSize<TransmissionOffset>(),
>+    CreateExtensionSize<TransportSequenceNumber>(),
>+    CreateExtensionSize<PlayoutDelayLimits>(),
>+    CreateExtensionSize<VideoOrientation>(),
>+    CreateExtensionSize<VideoContentTypeExtension>(),
>+    CreateExtensionSize<VideoTimingExtension>(),
>+    {RtpMid::kId, RtpMid::kMaxValueSizeBytes},
> };
> 
> const char* FrameTypeToString(FrameType frame_type) {
>   switch (frame_type) {
>     case kEmptyFrame:
>       return "empty";
>-    case kAudioFrameSpeech: return "audio_speech";
>-    case kAudioFrameCN: return "audio_cn";
>-    case kVideoFrameKey: return "video_key";
>-    case kVideoFrameDelta: return "video_delta";
>+    case kAudioFrameSpeech:
>+      return "audio_speech";
>+    case kAudioFrameCN:
>+      return "audio_cn";
>+    case kVideoFrameKey:
>+      return "video_key";
>+    case kVideoFrameDelta:
>+      return "video_delta";
>   }
>   return "";
> }
>@@ -96,7 +114,8 @@ RTPSender::RTPSender(
>     RtcEventLog* event_log,
>     SendPacketObserver* send_packet_observer,
>     RateLimiter* retransmission_rate_limiter,
>-    OverheadObserver* overhead_observer)
>+    OverheadObserver* overhead_observer,
>+    bool populate_network2_timestamp)
>     : clock_(clock),
>       // TODO(holmer): Remove this conversion?
>       clock_delta_ms_(clock_->TimeInMilliseconds() - rtc::TimeMillis()),
>@@ -111,7 +130,7 @@ RTPSender::RTPSender(
>       transport_(transport),
>       sending_media_(true),                   // Default to sending media.
>       max_packet_size_(IP_PACKET_SIZE - 28),  // Default is IP-v4/UDP.
>-      payload_type_(-1),
>+      last_payload_type_(-1),
>       payload_type_map_(),
>       rtp_header_extension_map_(),
>       packet_history_(clock),
>@@ -139,6 +158,7 @@ RTPSender::RTPSender(
>       rtp_overhead_bytes_per_packet_(0),
>       retransmission_rate_limiter_(retransmission_rate_limiter),
>       overhead_observer_(overhead_observer),
>+      populate_network2_timestamp_(populate_network2_timestamp),
>       send_side_bwe_with_overhead_(
>           webrtc::field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")) {
>   // This random initialization is not intended to be cryptographic strong.
>@@ -151,7 +171,8 @@ RTPSender::RTPSender(
>   // be found when paced.
>   if (flexfec_sender) {
>     flexfec_packet_history_.SetStorePacketsStatus(
>-        true, kMinFlexfecPacketsToStoreForPacing);
>+        RtpPacketHistory::StorageMode::kStore,
>+        kMinFlexfecPacketsToStoreForPacing);
>   }
> }
> 
>@@ -174,7 +195,13 @@ RTPSender::~RTPSender() {
> }
> 
> rtc::ArrayView<const RtpExtensionSize> RTPSender::FecExtensionSizes() {
>-  return rtc::MakeArrayView(kExtensionSizes, arraysize(kExtensionSizes));
>+  return rtc::MakeArrayView(kFecOrPaddingExtensionSizes,
>+                            arraysize(kFecOrPaddingExtensionSizes));
>+}
>+
>+rtc::ArrayView<const RtpExtensionSize> RTPSender::VideoExtensionSizes() {
>+  return rtc::MakeArrayView(kVideoExtensionSizes,
>+                            arraysize(kVideoExtensionSizes));
> }
> 
> uint16_t RTPSender::ActualSendBitrateKbit() const {
>@@ -237,8 +264,8 @@ int32_t RTPSender::RegisterPayload(
>     RTC_DCHECK(payload);
> 
>     // Check if it's the same as we already have.
>-    if (RtpUtility::StringCompare(
>-            payload->name, payload_name, RTP_PAYLOAD_NAME_SIZE - 1)) {
>+    if (RtpUtility::StringCompare(payload->name, payload_name,
>+                                  RTP_PAYLOAD_NAME_SIZE - 1)) {
>       if (audio_configured_ && payload->typeSpecific.is_audio()) {
>         auto& p = payload->typeSpecific.audio_payload();
>         if (rtc::SafeEq(p.format.clockrate_hz, frequency) &&
>@@ -284,12 +311,6 @@ int32_t RTPSender::DeRegisterSendPayload(int8_t payload_type) {
>   return 0;
> }
> 
>-// TODO(nisse): Delete this method, only used internally and by test code.
>-void RTPSender::SetSendPayloadType(int8_t payload_type) {
>-  rtc::CritScope lock(&send_critsect_);
>-  payload_type_ = payload_type;
>-}
>-
> void RTPSender::SetMaxRtpPacketSize(size_t max_packet_size) {
>   RTC_DCHECK_GE(max_packet_size, 100);
>   RTC_DCHECK_LE(max_packet_size, IP_PACKET_SIZE);
>@@ -336,14 +357,14 @@ void RTPSender::SetRtxPayloadType(int payload_type,
> }
> 
> int32_t RTPSender::CheckPayloadType(int8_t payload_type,
>-                                    RtpVideoCodecTypes* video_type) {
>+                                    VideoCodecType* video_type) {
>   rtc::CritScope lock(&send_critsect_);
> 
>   if (payload_type < 0) {
>     RTC_LOG(LS_ERROR) << "Invalid payload_type " << payload_type << ".";
>     return -1;
>   }
>-  if (payload_type_ == payload_type) {
>+  if (last_payload_type_ == payload_type) {
>     if (!audio_configured_) {
>       *video_type = video_->VideoCodecType();
>     }
>@@ -356,7 +377,6 @@ int32_t RTPSender::CheckPayloadType(int8_t payload_type,
>                         << " not registered.";
>     return -1;
>   }
>-  SetSendPayloadType(payload_type);
>   RtpUtility::Payload* payload = it->second;
>   RTC_DCHECK(payload);
>   if (payload->typeSpecific.is_video() && !audio_configured_) {
>@@ -393,7 +413,7 @@ bool RTPSender::SendOutgoingData(FrameType frame_type,
>     if (!sending_media_)
>       return true;
>   }
>-  RtpVideoCodecTypes video_type = kRtpVideoGeneric;
>+  VideoCodecType video_type = kVideoCodecGeneric;
>   if (CheckPayloadType(payload_type, &video_type) != 0) {
>     RTC_LOG(LS_ERROR) << "Don't send data with unknown payload type: "
>                       << static_cast<int>(payload_type) << ".";
>@@ -423,8 +443,8 @@ bool RTPSender::SendOutgoingData(FrameType frame_type,
>     result = audio_->SendAudio(frame_type, payload_type, rtp_timestamp,
>                                payload_data, payload_size);
>   } else {
>-    TRACE_EVENT_ASYNC_STEP1("webrtc", "Video", capture_time_ms,
>-                            "Send", "type", FrameTypeToString(frame_type));
>+    TRACE_EVENT_ASYNC_STEP1("webrtc", "Video", capture_time_ms, "Send", "type",
>+                            FrameTypeToString(frame_type));
>     if (frame_type == kEmptyFrame)
>       return true;
> 
>@@ -511,7 +531,7 @@ size_t RTPSender::SendPadData(size_t bytes,
>       timestamp = last_rtp_timestamp_;
>       capture_time_ms = capture_time_ms_;
>       if (rtx_ == kRtxOff) {
>-        if (payload_type_ == -1)
>+        if (last_payload_type_ == -1)
>           break;
>         // Without RTX we can't send padding in the middle of frames.
>         // For audio marker bits doesn't mark the end of a frame and frames
>@@ -530,7 +550,7 @@ size_t RTPSender::SendPadData(size_t bytes,
> 
>         sequence_number = sequence_number_;
>         ++sequence_number_;
>-        payload_type = payload_type_;
>+        payload_type = last_payload_type_;
>         over_rtx = false;
>       } else {
>         // Without abs-send-time or transport sequence number a media packet
>@@ -578,6 +598,8 @@ size_t RTPSender::SendPadData(size_t bytes,
>     padding_packet.SetExtension<AbsoluteSendTime>(
>         AbsoluteSendTime::MsTo24Bits(now_ms));
>     PacketOptions options;
>+    // Padding packets are never retransmissions.
>+    options.is_retransmit = false;
>     bool has_transport_seq_num =
>         UpdateTransportSequenceNumber(&padding_packet, &options.packet_id);
>     padding_packet.SetPadding(padding_bytes_in_packet, &random_);
>@@ -598,43 +620,60 @@ size_t RTPSender::SendPadData(size_t bytes,
> }
> 
> void RTPSender::SetStorePacketsStatus(bool enable, uint16_t number_to_store) {
>-  packet_history_.SetStorePacketsStatus(enable, number_to_store);
>+  RtpPacketHistory::StorageMode mode =
>+      enable ? RtpPacketHistory::StorageMode::kStore
>+             : RtpPacketHistory::StorageMode::kDisabled;
>+  packet_history_.SetStorePacketsStatus(mode, number_to_store);
> }
> 
> bool RTPSender::StorePackets() const {
>-  return packet_history_.StorePackets();
>+  return packet_history_.GetStorageMode() !=
>+         RtpPacketHistory::StorageMode::kDisabled;
> }
> 
>-int32_t RTPSender::ReSendPacket(uint16_t packet_id, int64_t min_resend_time) {
>-  std::unique_ptr<RtpPacketToSend> packet =
>-      packet_history_.GetPacketAndSetSendTime(packet_id, min_resend_time, true);
>-  if (!packet) {
>+int32_t RTPSender::ReSendPacket(uint16_t packet_id) {
>+  // Try to find packet in RTP packet history. Also verify RTT here, so that we
>+  // don't retransmit too often.
>+  absl::optional<RtpPacketHistory::PacketState> stored_packet =
>+      packet_history_.GetPacketState(packet_id, true);
>+  if (!stored_packet) {
>     // Packet not found.
>     return 0;
>   }
> 
>+  const int32_t packet_size = static_cast<int32_t>(stored_packet->payload_size);
>+
>+  RTC_DCHECK(retransmission_rate_limiter_);
>   // Check if we're overusing retransmission bitrate.
>   // TODO(sprang): Add histograms for nack success or failure reasons.
>-  RTC_DCHECK(retransmission_rate_limiter_);
>-  if (!retransmission_rate_limiter_->TryUseRate(packet->size()))
>+  if (!retransmission_rate_limiter_->TryUseRate(packet_size)) {
>     return -1;
>+  }
> 
>   if (paced_sender_) {
>     // Convert from TickTime to Clock since capture_time_ms is based on
>     // TickTime.
>     int64_t corrected_capture_tims_ms =
>-        packet->capture_time_ms() + clock_delta_ms_;
>-    paced_sender_->InsertPacket(RtpPacketSender::kNormalPriority,
>-                                packet->Ssrc(), packet->SequenceNumber(),
>-                                corrected_capture_tims_ms,
>-                                packet->payload_size(), true);
>+        stored_packet->capture_time_ms + clock_delta_ms_;
>+    paced_sender_->InsertPacket(
>+        RtpPacketSender::kNormalPriority, stored_packet->ssrc,
>+        stored_packet->rtp_sequence_number, corrected_capture_tims_ms,
>+        stored_packet->payload_size, true);
> 
>-    return packet->size();
>+    return packet_size;
>   }
>-  bool rtx = (RtxStatus() & kRtxRetransmitted) > 0;
>-  int32_t packet_size = static_cast<int32_t>(packet->size());
>+
>+  std::unique_ptr<RtpPacketToSend> packet =
>+      packet_history_.GetPacketAndSetSendTime(packet_id, true);
>+  if (!packet) {
>+    // Packet could theoretically time out between the first check and this one.
>+    return 0;
>+  }
>+
>+  const bool rtx = (RtxStatus() & kRtxRetransmitted) > 0;
>   if (!PrepareAndSendPacket(std::move(packet), rtx, true, PacedPacketInfo()))
>     return -1;
>+
>   return packet_size;
> }
> 
>@@ -648,13 +687,10 @@ bool RTPSender::SendPacketToNetwork(const RtpPacketToSend& packet,
>                      ? static_cast<int>(packet.size())
>                      : -1;
>     if (event_log_ && bytes_sent > 0) {
>-      event_log_->Log(rtc::MakeUnique<RtcEventRtpPacketOutgoing>(
>+      event_log_->Log(absl::make_unique<RtcEventRtpPacketOutgoing>(
>           packet, pacing_info.probe_cluster_id));
>     }
>   }
>-  TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
>-                       "RTPSender::SendPacketToNetwork", "size", packet.size(),
>-                       "sent", bytes_sent);
>   // TODO(pwestin): Add a separate bitrate for sent bitrate after pacer.
>   if (bytes_sent <= 0) {
>     RTC_LOG(LS_WARNING) << "Transport failed to send packet.";
>@@ -679,11 +715,9 @@ int RTPSender::SetSelectiveRetransmissions(uint8_t settings) {
> void RTPSender::OnReceivedNack(
>     const std::vector<uint16_t>& nack_sequence_numbers,
>     int64_t avg_rtt) {
>-  TRACE_EVENT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
>-               "RTPSender::OnReceivedNACK", "num_seqnum",
>-               nack_sequence_numbers.size(), "avg_rtt", avg_rtt);
>+  packet_history_.SetRtt(5 + avg_rtt);
>   for (uint16_t seq_no : nack_sequence_numbers) {
>-    const int32_t bytes_sent = ReSendPacket(seq_no, 5 + avg_rtt);
>+    const int32_t bytes_sent = ReSendPacket(seq_no);
>     if (bytes_sent < 0) {
>       // Failed to send one Sequence number. Give up the rest in this nack.
>       RTC_LOG(LS_WARNING) << "Failed resending RTP packet " << seq_no
>@@ -708,12 +742,13 @@ bool RTPSender::TimeToSendPacket(uint32_t ssrc,
>     return true;
> 
>   std::unique_ptr<RtpPacketToSend> packet;
>+  // No need to verify RTT here, it has already been checked before putting the
>+  // packet into the pacer. But _do_ update the send time.
>   if (ssrc == SSRC()) {
>-    packet = packet_history_.GetPacketAndSetSendTime(sequence_number, 0,
>-                                                     retransmission);
>+    packet = packet_history_.GetPacketAndSetSendTime(sequence_number, false);
>   } else if (ssrc == FlexfecSsrc()) {
>-    packet = flexfec_packet_history_.GetPacketAndSetSendTime(sequence_number, 0,
>-                                                             retransmission);
>+    packet =
>+        flexfec_packet_history_.GetPacketAndSetSendTime(sequence_number, false);
>   }
> 
>   if (!packet) {
>@@ -735,15 +770,6 @@ bool RTPSender::PrepareAndSendPacket(std::unique_ptr<RtpPacketToSend> packet,
>   int64_t capture_time_ms = packet->capture_time_ms();
>   RtpPacketToSend* packet_to_send = packet.get();
> 
>-  if (!is_retransmit && packet->Marker()) {
>-    TRACE_EVENT_ASYNC_END0(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "PacedSend",
>-                           capture_time_ms);
>-  }
>-
>-  TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
>-                       "PrepareAndSendPacket", "timestamp", packet->Timestamp(),
>-                       "seqnum", packet->SequenceNumber());
>-
>   std::unique_ptr<RtpPacketToSend> packet_rtx;
>   if (send_over_rtx) {
>     packet_rtx = BuildRtxPacket(*packet);
>@@ -767,14 +793,25 @@ bool RTPSender::PrepareAndSendPacket(std::unique_ptr<RtpPacketToSend> packet,
>   packet_to_send->SetExtension<AbsoluteSendTime>(
>       AbsoluteSendTime::MsTo24Bits(now_ms));
> 
>-  if (packet_to_send->HasExtension<VideoTimingExtension>())
>-    packet_to_send->set_pacer_exit_time_ms(now_ms);
>+  if (packet_to_send->HasExtension<VideoTimingExtension>()) {
>+    if (populate_network2_timestamp_) {
>+      packet_to_send->set_network2_time_ms(now_ms);
>+    } else {
>+      packet_to_send->set_pacer_exit_time_ms(now_ms);
>+    }
>+  }
> 
>   PacketOptions options;
>+  // If we are sending over RTX, it also means this is a retransmission.
>+  // E.g. RTPSender::TrySendRedundantPayloads calls PrepareAndSendPacket with
>+  // send_over_rtx = true but is_retransmit = false.
>+  options.is_retransmit = is_retransmit || send_over_rtx;
>   if (UpdateTransportSequenceNumber(packet_to_send, &options.packet_id)) {
>     AddPacketToTransportFeedback(options.packet_id, *packet_to_send,
>                                  pacing_info);
>   }
>+  options.application_data.assign(packet_to_send->application_data().begin(),
>+                                  packet_to_send->application_data().end());
> 
>   if (!is_retransmit && !send_over_rtx) {
>     UpdateDelayStatistics(packet->capture_time_ms(), now_ms);
>@@ -857,8 +894,6 @@ bool RTPSender::SendToNetwork(std::unique_ptr<RtpPacketToSend> packet,
>   if (packet->capture_time_ms() > 0) {
>     packet->SetExtension<TransmissionOffset>(
>         kTimestampTicksPerMs * (now_ms - packet->capture_time_ms()));
>-    if (packet->HasExtension<VideoTimingExtension>())
>-      packet->set_pacer_exit_time_ms(now_ms);
>   }
>   packet->SetExtension<AbsoluteSendTime>(AbsoluteSendTime::MsTo24Bits(now_ms));
> 
>@@ -877,7 +912,7 @@ bool RTPSender::SendToNetwork(std::unique_ptr<RtpPacketToSend> packet,
>   }
> 
>   uint32_t ssrc = packet->Ssrc();
>-  rtc::Optional<uint32_t> flexfec_ssrc = FlexfecSsrc();
>+  absl::optional<uint32_t> flexfec_ssrc = FlexfecSsrc();
>   if (paced_sender_) {
>     uint16_t seq_no = packet->SequenceNumber();
>     // Correct offset between implementations of millisecond time stamps in
>@@ -887,9 +922,10 @@ bool RTPSender::SendToNetwork(std::unique_ptr<RtpPacketToSend> packet,
>     if (ssrc == flexfec_ssrc) {
>       // Store FlexFEC packets in the history here, so they can be found
>       // when the pacer calls TimeToSendPacket.
>-      flexfec_packet_history_.PutRtpPacket(std::move(packet), storage, false);
>+      flexfec_packet_history_.PutRtpPacket(std::move(packet), storage,
>+                                           absl::nullopt);
>     } else {
>-      packet_history_.PutRtpPacket(std::move(packet), storage, false);
>+      packet_history_.PutRtpPacket(std::move(packet), storage, absl::nullopt);
>     }
> 
>     paced_sender_->InsertPacket(priority, ssrc, seq_no, corrected_time_ms,
>@@ -897,18 +933,18 @@ bool RTPSender::SendToNetwork(std::unique_ptr<RtpPacketToSend> packet,
>     if (last_capture_time_ms_sent_ == 0 ||
>         corrected_time_ms > last_capture_time_ms_sent_) {
>       last_capture_time_ms_sent_ = corrected_time_ms;
>-      TRACE_EVENT_ASYNC_BEGIN1(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
>-                               "PacedSend", corrected_time_ms,
>-                               "capture_time_ms", corrected_time_ms);
>     }
>     return true;
>   }
> 
>   PacketOptions options;
>+  options.is_retransmit = false;
>   if (UpdateTransportSequenceNumber(packet.get(), &options.packet_id)) {
>     AddPacketToTransportFeedback(options.packet_id, *packet.get(),
>                                  PacedPacketInfo());
>   }
>+  options.application_data.assign(packet->application_data().begin(),
>+                                  packet->application_data().end());
> 
>   UpdateDelayStatistics(packet->capture_time_ms(), now_ms);
>   UpdateOnSendPacket(options.packet_id, packet->capture_time_ms(),
>@@ -928,7 +964,7 @@ bool RTPSender::SendToNetwork(std::unique_ptr<RtpPacketToSend> packet,
>   // packet history (even if send failed).
>   if (storage == kAllowRetransmission) {
>     RTC_DCHECK_EQ(ssrc, SSRC());
>-    packet_history_.PutRtpPacket(std::move(packet), storage, true);
>+    packet_history_.PutRtpPacket(std::move(packet), storage, now_ms);
>   }
> 
>   return sent;
>@@ -951,9 +987,9 @@ void RTPSender::UpdateDelayStatistics(int64_t capture_time_ms, int64_t now_ms) {
>     rtc::CritScope cs(&statistics_crit_);
>     // TODO(holmer): Compute this iteratively instead.
>     send_delays_[now_ms] = now_ms - capture_time_ms;
>-    send_delays_.erase(send_delays_.begin(),
>-                       send_delays_.lower_bound(now_ms -
>-                       kSendSideDelayWindowMs));
>+    send_delays_.erase(
>+        send_delays_.begin(),
>+        send_delays_.lower_bound(now_ms - kSendSideDelayWindowMs));
>     int num_delays = 0;
>     for (auto it = send_delays_.upper_bound(now_ms - kSendSideDelayWindowMs);
>          it != send_delays_.end(); ++it) {
>@@ -999,8 +1035,8 @@ size_t RTPSender::RtpHeaderLength() const {
>   rtc::CritScope lock(&send_critsect_);
>   size_t rtp_header_length = kRtpHeaderLength;
>   rtp_header_length += sizeof(uint32_t) * csrcs_.size();
>-  rtp_header_length +=
>-      rtp_header_extension_map_.GetTotalLengthInBytes(kExtensionSizes);
>+  rtp_header_length += rtp_header_extension_map_.GetTotalLengthInBytes(
>+      kFecOrPaddingExtensionSizes);
>   return rtp_header_length;
> }
> 
>@@ -1033,6 +1069,10 @@ std::unique_ptr<RtpPacketToSend> RTPSender::AllocatePacket() const {
>     packet->SetExtension<PlayoutDelayLimits>(
>         playout_delay_oracle_.playout_delay());
>   }
>+  if (!mid_.empty()) {
>+    // This is a no-op if the MID header extension is not registered.
>+    packet->SetExtension<RtpMid>(mid_);
>+  }
>   return packet;
> }
> 
>@@ -1046,6 +1086,8 @@ bool RTPSender::AssignSequenceNumber(RtpPacketToSend* packet) {
>   // Remember marker bit to determine if padding can be inserted with
>   // sequence number following |packet|.
>   last_packet_marker_bit_ = packet->Marker();
>+  // Remember payload type to use in the padding packet if rtx is disabled.
>+  last_payload_type_ = packet->PayloadType();
>   // Save timestamps to generate timestamp field and extensions for the padding.
>   last_rtp_timestamp_ = packet->Timestamp();
>   last_timestamp_time_ms_ = clock_->TimeInMilliseconds();
>@@ -1111,11 +1153,17 @@ uint32_t RTPSender::SSRC() const {
>   return *ssrc_;
> }
> 
>-rtc::Optional<uint32_t> RTPSender::FlexfecSsrc() const {
>+void RTPSender::SetMid(const std::string& mid) {
>+  // This is configured via the API.
>+  rtc::CritScope lock(&send_critsect_);
>+  mid_ = mid;
>+}
>+
>+absl::optional<uint32_t> RTPSender::FlexfecSsrc() const {
>   if (video_) {
>     return video_->FlexfecSsrc();
>   }
>-  return rtc::nullopt;
>+  return absl::nullopt;
> }
> 
> void RTPSender::SetCsrcs(const std::vector<uint32_t>& csrcs) {
>@@ -1149,11 +1197,6 @@ int32_t RTPSender::SetAudioLevel(uint8_t level_d_bov) {
>   return audio_->SetAudioLevel(level_d_bov);
> }
> 
>-RtpVideoCodecTypes RTPSender::VideoCodecType() const {
>-  RTC_DCHECK(!audio_configured_) << "Sender is an audio stream!";
>-  return video_->VideoCodecType();
>-}
>-
> void RTPSender::SetUlpfecConfig(int red_payload_type, int ulpfec_payload_type) {
>   RTC_DCHECK(!audio_configured_);
>   video_->SetUlpfecConfig(red_payload_type, ulpfec_payload_type);
>@@ -1194,6 +1237,12 @@ std::unique_ptr<RtpPacketToSend> RTPSender::BuildRtxPacket(
> 
>     // Replace SSRC.
>     rtx_packet->SetSsrc(*ssrc_rtx_);
>+
>+    // Possibly include the MID header extension.
>+    if (!mid_.empty()) {
>+      // This is a no-op if the MID header extension is not registered.
>+      rtx_packet->SetExtension<RtpMid>(mid_);
>+    }
>   }
> 
>   uint8_t* rtx_payload =
>@@ -1206,6 +1255,9 @@ std::unique_ptr<RtpPacketToSend> RTPSender::BuildRtxPacket(
>   auto payload = packet.payload();
>   memcpy(rtx_payload + kRtxHeaderSize, payload.data(), payload.size());
> 
>+  // Add original application data.
>+  rtx_packet->set_application_data(packet.application_data());
>+
>   return rtx_packet;
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender.h
>index de265522b0d..5ff4a0f195a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender.h
>@@ -13,12 +13,13 @@
> 
> #include <map>
> #include <memory>
>+#include <string>
> #include <utility>
> #include <vector>
> 
>+#include "absl/types/optional.h"
> #include "api/array_view.h"
> #include "api/call/transport.h"
>-#include "api/optional.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/rtp_rtcp/include/flexfec_sender.h"
> #include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
>@@ -60,7 +61,8 @@ class RTPSender {
>             RtcEventLog* event_log,
>             SendPacketObserver* send_packet_observer,
>             RateLimiter* nack_rate_limiter,
>-            OverheadObserver* overhead_observer);
>+            OverheadObserver* overhead_observer,
>+            bool populate_network2_timestamp);
> 
>   ~RTPSender();
> 
>@@ -80,8 +82,6 @@ class RTPSender {
> 
>   int32_t DeRegisterSendPayload(const int8_t payload_type);
> 
>-  void SetSendPayloadType(int8_t payload_type);
>-
>   void SetSendingMediaStatus(bool enabled);
>   bool SendingMedia() const;
> 
>@@ -93,6 +93,8 @@ class RTPSender {
> 
>   void SetSSRC(uint32_t ssrc);
> 
>+  void SetMid(const std::string& mid);
>+
>   uint16_t SequenceNumber() const;
>   void SetSequenceNumber(uint16_t seq);
> 
>@@ -133,9 +135,10 @@ class RTPSender {
> 
>   bool StorePackets() const;
> 
>-  int32_t ReSendPacket(uint16_t packet_id, int64_t min_resend_time = 0);
>+  int32_t ReSendPacket(uint16_t packet_id);
> 
>-  // Feedback to decide when to stop sending playout delay.
>+  // Feedback to decide when to stop sending the playout delay and MID header
>+  // extensions.
>   void OnReceivedRtcpReportBlocks(const ReportBlockList& report_blocks);
> 
>   // RTX.
>@@ -150,6 +153,9 @@ class RTPSender {
>   // Size info for header extensions used by FEC packets.
>   static rtc::ArrayView<const RtpExtensionSize> FecExtensionSizes();
> 
>+  // Size info for header extensions used by video packets.
>+  static rtc::ArrayView<const RtpExtensionSize> VideoExtensionSizes();
>+
>   // Create empty packet, fills ssrc, csrcs and reserve place for header
>   // extensions RtpSender updates before sending.
>   std::unique_ptr<RtpPacketToSend> AllocatePacket() const;
>@@ -166,7 +172,7 @@ class RTPSender {
> 
>   uint32_t SSRC() const;
> 
>-  rtc::Optional<uint32_t> FlexfecSsrc() const;
>+  absl::optional<uint32_t> FlexfecSsrc() const;
> 
>   bool SendToNetwork(std::unique_ptr<RtpPacketToSend> packet,
>                      StorageType storage,
>@@ -181,8 +187,6 @@ class RTPSender {
>   // header-extension-for-audio-level-indication.
>   int32_t SetAudioLevel(uint8_t level_d_bov);
> 
>-  RtpVideoCodecTypes VideoCodecType() const;
>-
>   uint32_t MaxConfiguredBitrateVideo() const;
> 
>   // ULPFEC.
>@@ -208,7 +212,7 @@ class RTPSender {
>   void SetRtt(int64_t rtt_ms);
> 
>  protected:
>-  int32_t CheckPayloadType(int8_t payload_type, RtpVideoCodecTypes* video_type);
>+  int32_t CheckPayloadType(int8_t payload_type, VideoCodecType* video_type);
> 
>  private:
>   // Maps capture time in milliseconds to send-side delay in milliseconds.
>@@ -273,7 +277,7 @@ class RTPSender {
> 
>   size_t max_packet_size_;
> 
>-  int8_t payload_type_ RTC_GUARDED_BY(send_critsect_);
>+  int8_t last_payload_type_ RTC_GUARDED_BY(send_critsect_);
>   std::map<int8_t, RtpUtility::Payload*> payload_type_map_;
> 
>   RtpHeaderExtensionMap rtp_header_extension_map_
>@@ -311,9 +315,11 @@ class RTPSender {
>   bool sequence_number_forced_ RTC_GUARDED_BY(send_critsect_);
>   uint16_t sequence_number_ RTC_GUARDED_BY(send_critsect_);
>   uint16_t sequence_number_rtx_ RTC_GUARDED_BY(send_critsect_);
>-  // Must be explicitly set by the application, use of rtc::Optional
>+  // Must be explicitly set by the application, use of absl::optional
>   // only to keep track of correct use.
>-  rtc::Optional<uint32_t> ssrc_ RTC_GUARDED_BY(send_critsect_);
>+  absl::optional<uint32_t> ssrc_ RTC_GUARDED_BY(send_critsect_);
>+  // MID value to send in the MID header extension.
>+  std::string mid_ RTC_GUARDED_BY(send_critsect_);
>   uint32_t last_rtp_timestamp_ RTC_GUARDED_BY(send_critsect_);
>   int64_t capture_time_ms_ RTC_GUARDED_BY(send_critsect_);
>   int64_t last_timestamp_time_ms_ RTC_GUARDED_BY(send_critsect_);
>@@ -321,13 +327,14 @@ class RTPSender {
>   bool last_packet_marker_bit_ RTC_GUARDED_BY(send_critsect_);
>   std::vector<uint32_t> csrcs_ RTC_GUARDED_BY(send_critsect_);
>   int rtx_ RTC_GUARDED_BY(send_critsect_);
>-  rtc::Optional<uint32_t> ssrc_rtx_ RTC_GUARDED_BY(send_critsect_);
>+  absl::optional<uint32_t> ssrc_rtx_ RTC_GUARDED_BY(send_critsect_);
>   // Mapping rtx_payload_type_map_[associated] = rtx.
>   std::map<int8_t, int8_t> rtx_payload_type_map_ RTC_GUARDED_BY(send_critsect_);
>   size_t rtp_overhead_bytes_per_packet_ RTC_GUARDED_BY(send_critsect_);
> 
>   RateLimiter* const retransmission_rate_limiter_;
>   OverheadObserver* overhead_observer_;
>+  const bool populate_network2_timestamp_;
> 
>   const bool send_side_bwe_with_overhead_;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
>index 93f2b9fa334..bf86a39339b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.cc
>@@ -26,8 +26,7 @@
> namespace webrtc {
> 
> RTPSenderAudio::RTPSenderAudio(Clock* clock, RTPSender* rtp_sender)
>-    : clock_(clock),
>-      rtp_sender_(rtp_sender) {}
>+    : clock_(clock), rtp_sender_(rtp_sender) {}
> 
> RTPSenderAudio::~RTPSenderAudio() {}
> 
>@@ -190,7 +189,8 @@ bool RTPSenderAudio::SendAudio(FrameType frame_type,
>         dtmf_duration_samples -= 0xffff;
>         dtmf_length_samples_ -= 0xffff;
> 
>-        return SendTelephoneEventPacket(ended, dtmf_timestamp_,
>+        return SendTelephoneEventPacket(
>+            ended, dtmf_timestamp_,
>             static_cast<uint16_t>(dtmf_duration_samples), false);
>       } else {
>         if (!SendTelephoneEventPacket(ended, dtmf_timestamp_,
>@@ -321,9 +321,6 @@ bool RTPSenderAudio::SendTelephoneEventPacket(bool ended,
>     dtmfbuffer[1] = E | R | volume;
>     ByteWriter<uint16_t>::WriteBigEndian(dtmfbuffer + 2, duration);
> 
>-    TRACE_EVENT_INSTANT2(
>-        TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "Audio::SendTelephoneEvent",
>-        "timestamp", packet->Timestamp(), "seqnum", packet->SequenceNumber());
>     result = rtp_sender_->SendToNetwork(std::move(packet), kAllowRetransmission,
>                                         RtpPacketSender::kHighPriority);
>     send_count--;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
>index 16648cd136a..63dfc2b99c3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender_audio.h
>@@ -19,7 +19,6 @@
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/onetimeevent.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
>index db760ecf7c8..d143c8700e5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
>@@ -11,6 +11,7 @@
> #include <memory>
> #include <vector>
> 
>+#include "absl/memory/memory.h"
> #include "api/video/video_timing.h"
> #include "logging/rtc_event_log/events/rtc_event.h"
> #include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
>@@ -28,13 +29,11 @@
> #include "modules/rtp_rtcp/source/rtp_utility.h"
> #include "rtc_base/arraysize.h"
> #include "rtc_base/buffer.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/rate_limiter.h"
> #include "test/field_trial.h"
> #include "test/gmock.h"
> #include "test/gtest.h"
> #include "test/mock_transport.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -43,6 +42,7 @@ const int kTransmissionTimeOffsetExtensionId = 1;
> const int kAbsoluteSendTimeExtensionId = 14;
> const int kTransportSequenceNumberExtensionId = 13;
> const int kVideoTimingExtensionId = 12;
>+const int kMidExtensionId = 11;
> const int kPayload = 100;
> const int kRtxPayload = 98;
> const uint32_t kTimestamp = 10;
>@@ -70,7 +70,7 @@ uint64_t ConvertMsToAbsSendTime(int64_t time_ms) {
> 
> class LoopbackTransportTest : public webrtc::Transport {
>  public:
>-  LoopbackTransportTest() : total_bytes_sent_(0), last_packet_id_(-1) {
>+  LoopbackTransportTest() : total_bytes_sent_(0) {
>     receivers_extensions_.Register(kRtpExtensionTransmissionTimeOffset,
>                                    kTransmissionTimeOffsetExtensionId);
>     receivers_extensions_.Register(kRtpExtensionAbsoluteSendTime,
>@@ -83,12 +83,13 @@ class LoopbackTransportTest : public webrtc::Transport {
>                                    kAudioLevelExtensionId);
>     receivers_extensions_.Register(kRtpExtensionVideoTiming,
>                                    kVideoTimingExtensionId);
>+    receivers_extensions_.Register(kRtpExtensionMid, kMidExtensionId);
>   }
> 
>   bool SendRtp(const uint8_t* data,
>                size_t len,
>                const PacketOptions& options) override {
>-    last_packet_id_ = options.packet_id;
>+    last_options_ = options;
>     total_bytes_sent_ += len;
>     sent_packets_.push_back(RtpPacketReceived(&receivers_extensions_));
>     EXPECT_TRUE(sent_packets_.back().Parse(data, len));
>@@ -99,7 +100,7 @@ class LoopbackTransportTest : public webrtc::Transport {
>   int packets_sent() { return sent_packets_.size(); }
> 
>   size_t total_bytes_sent_;
>-  int last_packet_id_;
>+  PacketOptions last_options_;
>   std::vector<RtpPacketReceived> sent_packets_;
> 
>  private:
>@@ -164,15 +165,14 @@ class RtpSenderTest : public ::testing::TestWithParam<bool> {
>         field_trials_(GetParam() ? "WebRTC-SendSideBwe-WithOverhead/Enabled/"
>                                  : "") {}
> 
>-  void SetUp() override { SetUpRtpSender(true); }
>+  void SetUp() override { SetUpRtpSender(true, false); }
> 
>-  void SetUpRtpSender(bool pacer) {
>+  void SetUpRtpSender(bool pacer, bool populate_network2) {
>     rtp_sender_.reset(new RTPSender(
>         false, &fake_clock_, &transport_, pacer ? &mock_paced_sender_ : nullptr,
>         nullptr, &seq_num_allocator_, nullptr, nullptr, nullptr, nullptr,
>         &mock_rtc_event_log_, &send_packet_observer_,
>-        &retransmission_rate_limiter_, nullptr));
>-    rtp_sender_->SetSendPayloadType(kPayload);
>+        &retransmission_rate_limiter_, nullptr, populate_network2));
>     rtp_sender_->SetSequenceNumber(kSeqNum);
>     rtp_sender_->SetTimestampOffset(0);
>     rtp_sender_->SetSSRC(kSsrc);
>@@ -244,9 +244,10 @@ class RtpSenderTest : public ::testing::TestWithParam<bool> {
>     EXPECT_EQ(0, rtp_sender_->RegisterPayload(payload_name, kPayloadType, 90000,
>                                               0, 1500));
> 
>+    RTPVideoHeader video_header;
>     EXPECT_TRUE(rtp_sender_->SendOutgoingData(
>         kVideoFrameKey, kPayloadType, kTimestamp, kCaptureTimeMs, kPayloadData,
>-        sizeof(kPayloadData), nullptr, nullptr, nullptr,
>+        sizeof(kPayloadData), nullptr, &video_header, nullptr,
>         kDefaultExpectedRetransmissionTimeMs));
>   }
> };
>@@ -255,7 +256,7 @@ class RtpSenderTest : public ::testing::TestWithParam<bool> {
> // default code path.
> class RtpSenderTestWithoutPacer : public RtpSenderTest {
>  public:
>-  void SetUp() override { SetUpRtpSender(false); }
>+  void SetUp() override { SetUpRtpSender(false, false); }
> };
> 
> class TestRtpSenderVideo : public RTPSenderVideo {
>@@ -279,7 +280,7 @@ class RtpSenderVideoTest : public RtpSenderTest {
>  protected:
>   void SetUp() override {
>     // TODO(pbos): Set up to use pacer.
>-    SetUpRtpSender(false);
>+    SetUpRtpSender(false, false);
>     rtp_sender_video_.reset(
>         new TestRtpSenderVideo(&fake_clock_, rtp_sender_.get(), nullptr));
>   }
>@@ -346,7 +347,7 @@ TEST_P(RtpSenderTestWithoutPacer, AssignSequenceNumberFailsOnNotSending) {
>   EXPECT_FALSE(rtp_sender_->AssignSequenceNumber(packet.get()));
> }
> 
>-TEST_P(RtpSenderTestWithoutPacer, AssignSequenceNumberMayAllowPadding) {
>+TEST_P(RtpSenderTestWithoutPacer, AssignSequenceNumberMayAllowPaddingOnVideo) {
>   constexpr size_t kPaddingSize = 100;
>   auto packet = rtp_sender_->AllocatePacket();
>   ASSERT_TRUE(packet);
>@@ -354,7 +355,7 @@ TEST_P(RtpSenderTestWithoutPacer, AssignSequenceNumberMayAllowPadding) {
>   ASSERT_FALSE(rtp_sender_->TimeToSendPadding(kPaddingSize, PacedPacketInfo()));
>   packet->SetMarker(false);
>   ASSERT_TRUE(rtp_sender_->AssignSequenceNumber(packet.get()));
>-  // Packet without marker bit doesn't allow padding.
>+  // Packet without marker bit doesn't allow padding on video stream.
>   EXPECT_FALSE(rtp_sender_->TimeToSendPadding(kPaddingSize, PacedPacketInfo()));
> 
>   packet->SetMarker(true);
>@@ -363,6 +364,36 @@ TEST_P(RtpSenderTestWithoutPacer, AssignSequenceNumberMayAllowPadding) {
>   EXPECT_TRUE(rtp_sender_->TimeToSendPadding(kPaddingSize, PacedPacketInfo()));
> }
> 
>+TEST_P(RtpSenderTest, AssignSequenceNumberAllowsPaddingOnAudio) {
>+  MockTransport transport;
>+  const bool kEnableAudio = true;
>+  rtp_sender_.reset(new RTPSender(
>+      kEnableAudio, &fake_clock_, &transport, &mock_paced_sender_, nullptr,
>+      nullptr, nullptr, nullptr, nullptr, nullptr, &mock_rtc_event_log_,
>+      nullptr, &retransmission_rate_limiter_, nullptr, false));
>+  rtp_sender_->SetTimestampOffset(0);
>+  rtp_sender_->SetSSRC(kSsrc);
>+
>+  std::unique_ptr<RtpPacketToSend> audio_packet = rtp_sender_->AllocatePacket();
>+  // Padding on audio stream allowed regardless of marker in the last packet.
>+  audio_packet->SetMarker(false);
>+  audio_packet->SetPayloadType(kPayload);
>+  rtp_sender_->AssignSequenceNumber(audio_packet.get());
>+
>+  const size_t kPaddingSize = 59;
>+  EXPECT_CALL(transport, SendRtp(_, kPaddingSize + kRtpHeaderSize, _))
>+      .WillOnce(testing::Return(true));
>+  EXPECT_EQ(kPaddingSize,
>+            rtp_sender_->TimeToSendPadding(kPaddingSize, PacedPacketInfo()));
>+
>+  // Requested padding size is too small, will send a larger one.
>+  const size_t kMinPaddingSize = 50;
>+  EXPECT_CALL(transport, SendRtp(_, kMinPaddingSize + kRtpHeaderSize, _))
>+      .WillOnce(testing::Return(true));
>+  EXPECT_EQ(kMinPaddingSize, rtp_sender_->TimeToSendPadding(kMinPaddingSize - 5,
>+                                                            PacedPacketInfo()));
>+}
>+
> TEST_P(RtpSenderTestWithoutPacer, AssignSequenceNumberSetPaddingTimestamps) {
>   constexpr size_t kPaddingSize = 100;
>   auto packet = rtp_sender_->AllocatePacket();
>@@ -385,7 +416,7 @@ TEST_P(RtpSenderTestWithoutPacer,
>   rtp_sender_.reset(new RTPSender(
>       false, &fake_clock_, &transport_, nullptr, nullptr, &seq_num_allocator_,
>       &feedback_observer_, nullptr, nullptr, nullptr, &mock_rtc_event_log_,
>-      nullptr, &retransmission_rate_limiter_, &mock_overhead_observer));
>+      nullptr, &retransmission_rate_limiter_, &mock_overhead_observer, false));
>   rtp_sender_->SetSSRC(kSsrc);
>   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
>                    kRtpExtensionTransportSequenceNumber,
>@@ -412,7 +443,7 @@ TEST_P(RtpSenderTestWithoutPacer, SendsPacketsWithTransportSequenceNumber) {
>   rtp_sender_.reset(new RTPSender(
>       false, &fake_clock_, &transport_, nullptr, nullptr, &seq_num_allocator_,
>       &feedback_observer_, nullptr, nullptr, nullptr, &mock_rtc_event_log_,
>-      &send_packet_observer_, &retransmission_rate_limiter_, nullptr));
>+      &send_packet_observer_, &retransmission_rate_limiter_, nullptr, false));
>   rtp_sender_->SetSSRC(kSsrc);
>   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
>                    kRtpExtensionTransportSequenceNumber,
>@@ -435,7 +466,19 @@ TEST_P(RtpSenderTestWithoutPacer, SendsPacketsWithTransportSequenceNumber) {
>   uint16_t transport_seq_no;
>   ASSERT_TRUE(packet.GetExtension<TransportSequenceNumber>(&transport_seq_no));
>   EXPECT_EQ(kTransportSequenceNumber, transport_seq_no);
>-  EXPECT_EQ(transport_.last_packet_id_, transport_seq_no);
>+  EXPECT_EQ(transport_.last_options_.packet_id, transport_seq_no);
>+}
>+
>+TEST_P(RtpSenderTestWithoutPacer, PacketOptionsNoRetransmission) {
>+  rtp_sender_.reset(new RTPSender(
>+      false, &fake_clock_, &transport_, nullptr, nullptr, &seq_num_allocator_,
>+      &feedback_observer_, nullptr, nullptr, nullptr, &mock_rtc_event_log_,
>+      &send_packet_observer_, &retransmission_rate_limiter_, nullptr, false));
>+  rtp_sender_->SetSSRC(kSsrc);
>+
>+  SendGenericPayload();
>+
>+  EXPECT_FALSE(transport_.last_options_.is_retransmit);
> }
> 
> TEST_P(RtpSenderTestWithoutPacer, NoAllocationIfNotRegistered) {
>@@ -460,7 +503,7 @@ TEST_P(RtpSenderTest, SendsPacketsWithTransportSequenceNumber) {
>       false, &fake_clock_, &transport_, &mock_paced_sender_, nullptr,
>       &seq_num_allocator_, &feedback_observer_, nullptr, nullptr, nullptr,
>       &mock_rtc_event_log_, &send_packet_observer_,
>-      &retransmission_rate_limiter_, nullptr));
>+      &retransmission_rate_limiter_, nullptr, false));
>   rtp_sender_->SetSequenceNumber(kSeqNum);
>   rtp_sender_->SetSSRC(kSsrc);
>   rtp_sender_->SetStorePacketsStatus(true, 10);
>@@ -488,10 +531,10 @@ TEST_P(RtpSenderTest, SendsPacketsWithTransportSequenceNumber) {
>   uint16_t transport_seq_no;
>   EXPECT_TRUE(packet.GetExtension<TransportSequenceNumber>(&transport_seq_no));
>   EXPECT_EQ(kTransportSequenceNumber, transport_seq_no);
>-  EXPECT_EQ(transport_.last_packet_id_, transport_seq_no);
>+  EXPECT_EQ(transport_.last_options_.packet_id, transport_seq_no);
> }
> 
>-TEST_P(RtpSenderTestWithoutPacer, WritesTimestampToTimingExtension) {
>+TEST_P(RtpSenderTest, WritesPacerExitToTimingExtension) {
>   rtp_sender_->SetStorePacketsStatus(true, 10);
>   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
>                    kRtpExtensionVideoTiming, kVideoTimingExtensionId));
>@@ -507,11 +550,17 @@ TEST_P(RtpSenderTestWithoutPacer, WritesTimestampToTimingExtension) {
>   size_t packet_size = packet->size();
> 
>   const int kStoredTimeInMs = 100;
>+  {
>+    EXPECT_CALL(
>+        mock_paced_sender_,
>+        InsertPacket(RtpPacketSender::kNormalPriority, kSsrc, _, _, _, _));
>+    EXPECT_TRUE(rtp_sender_->SendToNetwork(std::move(packet),
>+                                           kAllowRetransmission,
>+                                           RtpPacketSender::kNormalPriority));
>+  }
>   fake_clock_.AdvanceTimeMilliseconds(kStoredTimeInMs);
>-
>-  EXPECT_TRUE(rtp_sender_->SendToNetwork(std::move(packet),
>-                                         kAllowRetransmission,
>-                                         RtpPacketSender::kNormalPriority));
>+  rtp_sender_->TimeToSendPacket(kSsrc, kSeqNum, capture_time_ms, false,
>+                                PacedPacketInfo());
>   EXPECT_EQ(1, transport_.packets_sent());
>   EXPECT_EQ(packet_size, transport_.last_sent_packet().size());
> 
>@@ -519,17 +568,45 @@ TEST_P(RtpSenderTestWithoutPacer, WritesTimestampToTimingExtension) {
>   EXPECT_TRUE(transport_.last_sent_packet().GetExtension<VideoTimingExtension>(
>       &video_timing));
>   EXPECT_EQ(kStoredTimeInMs, video_timing.pacer_exit_delta_ms);
>+}
>+
>+TEST_P(RtpSenderTest, WritesNetwork2ToTimingExtension) {
>+  SetUpRtpSender(true, true);
>+  rtp_sender_->SetStorePacketsStatus(true, 10);
>+  EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
>+                   kRtpExtensionVideoTiming, kVideoTimingExtensionId));
>+  int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
>+  auto packet = rtp_sender_->AllocatePacket();
>+  packet->SetPayloadType(kPayload);
>+  packet->SetMarker(true);
>+  packet->SetTimestamp(kTimestamp);
>+  packet->set_capture_time_ms(capture_time_ms);
>+  const uint16_t kPacerExitMs = 1234u;
>+  const VideoSendTiming kVideoTiming = {0u, 0u, 0u, kPacerExitMs, 0u, 0u, true};
>+  packet->SetExtension<VideoTimingExtension>(kVideoTiming);
>+  EXPECT_TRUE(rtp_sender_->AssignSequenceNumber(packet.get()));
>+  size_t packet_size = packet->size();
> 
>+  const int kStoredTimeInMs = 100;
>+  {
>+    EXPECT_CALL(
>+        mock_paced_sender_,
>+        InsertPacket(RtpPacketSender::kNormalPriority, kSsrc, _, _, _, _));
>+    EXPECT_TRUE(rtp_sender_->SendToNetwork(std::move(packet),
>+                                           kAllowRetransmission,
>+                                           RtpPacketSender::kNormalPriority));
>+  }
>   fake_clock_.AdvanceTimeMilliseconds(kStoredTimeInMs);
>   rtp_sender_->TimeToSendPacket(kSsrc, kSeqNum, capture_time_ms, false,
>                                 PacedPacketInfo());
>-
>-  EXPECT_EQ(2, transport_.packets_sent());
>+  EXPECT_EQ(1, transport_.packets_sent());
>   EXPECT_EQ(packet_size, transport_.last_sent_packet().size());
> 
>+  VideoSendTiming video_timing;
>   EXPECT_TRUE(transport_.last_sent_packet().GetExtension<VideoTimingExtension>(
>       &video_timing));
>-  EXPECT_EQ(kStoredTimeInMs * 2, video_timing.pacer_exit_delta_ms);
>+  EXPECT_EQ(kStoredTimeInMs, video_timing.network2_timestamp_delta_ms);
>+  EXPECT_EQ(kPacerExitMs, video_timing.pacer_exit_delta_ms);
> }
> 
> TEST_P(RtpSenderTest, TrafficSmoothingWithExtensions) {
>@@ -778,6 +855,7 @@ TEST_P(RtpSenderTest, OnSendPacketNotUpdatedForRetransmits) {
>                                 fake_clock_.TimeInMilliseconds(), kIsRetransmit,
>                                 PacedPacketInfo());
>   EXPECT_EQ(1, transport_.packets_sent());
>+  EXPECT_TRUE(transport_.last_options_.is_retransmit);
> }
> 
> TEST_P(RtpSenderTest, OnSendPacketNotUpdatedWithoutSeqNumAllocator) {
>@@ -785,7 +863,7 @@ TEST_P(RtpSenderTest, OnSendPacketNotUpdatedWithoutSeqNumAllocator) {
>       false, &fake_clock_, &transport_, &mock_paced_sender_, nullptr,
>       nullptr /* TransportSequenceNumberAllocator */, nullptr, nullptr, nullptr,
>       nullptr, nullptr, &send_packet_observer_, &retransmission_rate_limiter_,
>-      nullptr));
>+      nullptr, false));
>   rtp_sender_->SetSequenceNumber(kSeqNum);
>   rtp_sender_->SetSSRC(kSsrc);
>   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
>@@ -811,7 +889,7 @@ TEST_P(RtpSenderTest, SendRedundantPayloads) {
>   rtp_sender_.reset(new RTPSender(
>       false, &fake_clock_, &transport, &mock_paced_sender_, nullptr, nullptr,
>       nullptr, nullptr, nullptr, nullptr, &mock_rtc_event_log_, nullptr,
>-      &retransmission_rate_limiter_, nullptr));
>+      &retransmission_rate_limiter_, nullptr, false));
>   rtp_sender_->SetSequenceNumber(kSeqNum);
>   rtp_sender_->SetSSRC(kSsrc);
>   rtp_sender_->SetRtxPayloadType(kRtxPayload, kPayload);
>@@ -859,20 +937,28 @@ TEST_P(RtpSenderTest, SendRedundantPayloads) {
>   EXPECT_EQ(kMaxPaddingSize,
>             rtp_sender_->TimeToSendPadding(49, PacedPacketInfo()));
> 
>+  PacketOptions options;
>   EXPECT_CALL(transport,
>               SendRtp(_, kPayloadSizes[0] + rtp_header_len + kRtxHeaderSize, _))
>-      .WillOnce(testing::Return(true));
>+      .WillOnce(
>+          testing::DoAll(testing::SaveArg<2>(&options), testing::Return(true)));
>   EXPECT_EQ(kPayloadSizes[0],
>             rtp_sender_->TimeToSendPadding(500, PacedPacketInfo()));
>+  EXPECT_TRUE(options.is_retransmit);
> 
>-  EXPECT_CALL(transport, SendRtp(_, kPayloadSizes[kNumPayloadSizes - 1] +
>-                                        rtp_header_len + kRtxHeaderSize,
>+  EXPECT_CALL(transport, SendRtp(_,
>+                                 kPayloadSizes[kNumPayloadSizes - 1] +
>+                                     rtp_header_len + kRtxHeaderSize,
>                                  _))
>       .WillOnce(testing::Return(true));
>+
>+  options.is_retransmit = false;
>   EXPECT_CALL(transport, SendRtp(_, kMaxPaddingSize + rtp_header_len, _))
>-      .WillOnce(testing::Return(true));
>+      .WillOnce(
>+          testing::DoAll(testing::SaveArg<2>(&options), testing::Return(true)));
>   EXPECT_EQ(kPayloadSizes[kNumPayloadSizes - 1] + kMaxPaddingSize,
>             rtp_sender_->TimeToSendPadding(999, PacedPacketInfo()));
>+  EXPECT_FALSE(options.is_retransmit);
> }
> 
> TEST_P(RtpSenderTestWithoutPacer, SendGenericVideo) {
>@@ -883,9 +969,10 @@ TEST_P(RtpSenderTestWithoutPacer, SendGenericVideo) {
>   uint8_t payload[] = {47, 11, 32, 93, 89};
> 
>   // Send keyframe
>+  RTPVideoHeader video_header;
>   ASSERT_TRUE(rtp_sender_->SendOutgoingData(
>       kVideoFrameKey, payload_type, 1234, 4321, payload, sizeof(payload),
>-      nullptr, nullptr, nullptr, kDefaultExpectedRetransmissionTimeMs));
>+      nullptr, &video_header, nullptr, kDefaultExpectedRetransmissionTimeMs));
> 
>   auto sent_payload = transport_.last_sent_packet().payload();
>   uint8_t generic_header = sent_payload[0];
>@@ -900,7 +987,7 @@ TEST_P(RtpSenderTestWithoutPacer, SendGenericVideo) {
> 
>   ASSERT_TRUE(rtp_sender_->SendOutgoingData(
>       kVideoFrameDelta, payload_type, 1234, 4321, payload, sizeof(payload),
>-      nullptr, nullptr, nullptr, kDefaultExpectedRetransmissionTimeMs));
>+      nullptr, &video_header, nullptr, kDefaultExpectedRetransmissionTimeMs));
> 
>   sent_payload = transport_.last_sent_packet().payload();
>   generic_header = sent_payload[0];
>@@ -914,10 +1001,11 @@ TEST_P(RtpSenderTest, SendFlexfecPackets) {
>   constexpr int kFlexfecPayloadType = 118;
>   constexpr uint32_t kMediaSsrc = 1234;
>   constexpr uint32_t kFlexfecSsrc = 5678;
>+  const char kNoMid[] = "";
>   const std::vector<RtpExtension> kNoRtpExtensions;
>   const std::vector<RtpExtensionSize> kNoRtpExtensionSizes;
>   FlexfecSender flexfec_sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc,
>-                               kNoRtpExtensions, kNoRtpExtensionSizes,
>+                               kNoMid, kNoRtpExtensions, kNoRtpExtensionSizes,
>                                nullptr /* rtp_state */, &fake_clock_);
> 
>   // Reset |rtp_sender_| to use FlexFEC.
>@@ -925,10 +1013,9 @@ TEST_P(RtpSenderTest, SendFlexfecPackets) {
>       false, &fake_clock_, &transport_, &mock_paced_sender_, &flexfec_sender,
>       &seq_num_allocator_, nullptr, nullptr, nullptr, nullptr,
>       &mock_rtc_event_log_, &send_packet_observer_,
>-      &retransmission_rate_limiter_, nullptr));
>+      &retransmission_rate_limiter_, nullptr, false));
>   rtp_sender_->SetSSRC(kMediaSsrc);
>   rtp_sender_->SetSequenceNumber(kSeqNum);
>-  rtp_sender_->SetSendPayloadType(kMediaPayloadType);
>   rtp_sender_->SetStorePacketsStatus(true, 10);
> 
>   // Parameters selected to generate a single FEC packet per media packet.
>@@ -973,11 +1060,12 @@ TEST_P(RtpSenderTest, NoFlexfecForTimingFrames) {
>   constexpr int kFlexfecPayloadType = 118;
>   constexpr uint32_t kMediaSsrc = 1234;
>   constexpr uint32_t kFlexfecSsrc = 5678;
>+  const char kNoMid[] = "";
>   const std::vector<RtpExtension> kNoRtpExtensions;
>   const std::vector<RtpExtensionSize> kNoRtpExtensionSizes;
> 
>   FlexfecSender flexfec_sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc,
>-                               kNoRtpExtensions, kNoRtpExtensionSizes,
>+                               kNoMid, kNoRtpExtensions, kNoRtpExtensionSizes,
>                                nullptr /* rtp_state */, &fake_clock_);
> 
>   // Reset |rtp_sender_| to use FlexFEC.
>@@ -985,10 +1073,9 @@ TEST_P(RtpSenderTest, NoFlexfecForTimingFrames) {
>       false, &fake_clock_, &transport_, &mock_paced_sender_, &flexfec_sender,
>       &seq_num_allocator_, nullptr, nullptr, nullptr, nullptr,
>       &mock_rtc_event_log_, &send_packet_observer_,
>-      &retransmission_rate_limiter_, nullptr));
>+      &retransmission_rate_limiter_, nullptr, false));
>   rtp_sender_->SetSSRC(kMediaSsrc);
>   rtp_sender_->SetSequenceNumber(kSeqNum);
>-  rtp_sender_->SetSendPayloadType(kMediaPayloadType);
>   rtp_sender_->SetStorePacketsStatus(true, 10);
> 
>   // Need extension to be registered for timing frames to be sent.
>@@ -1016,8 +1103,7 @@ TEST_P(RtpSenderTest, NoFlexfecForTimingFrames) {
>   EXPECT_EQ(0, rtp_sender_->RegisterPayload(payload_name, kPayloadType, 90000,
>                                             0, 1500));
>   RTPVideoHeader video_header;
>-  memset(&video_header, 0, sizeof(RTPVideoHeader));
>-  video_header.video_timing.flags = TimingFrameFlags::kTriggeredByTimer;
>+  video_header.video_timing.flags = VideoSendTiming::kTriggeredByTimer;
>   EXPECT_TRUE(rtp_sender_->SendOutgoingData(
>       kVideoFrameKey, kPayloadType, kTimestamp, kCaptureTimeMs, kPayloadData,
>       sizeof(kPayloadData), nullptr, &video_header, nullptr,
>@@ -1043,7 +1129,7 @@ TEST_P(RtpSenderTest, NoFlexfecForTimingFrames) {
>   EXPECT_CALL(mock_paced_sender_,
>               InsertPacket(RtpPacketSender::kLowPriority, kMediaSsrc,
>                            kSeqNum + 1, _, _, false));
>-  video_header.video_timing.flags = TimingFrameFlags::kInvalid;
>+  video_header.video_timing.flags = VideoSendTiming::kInvalid;
>   EXPECT_TRUE(rtp_sender_->SendOutgoingData(
>       kVideoFrameKey, kPayloadType, kTimestamp + 1, kCaptureTimeMs + 1,
>       kPayloadData, sizeof(kPayloadData), nullptr, &video_header, nullptr,
>@@ -1074,21 +1160,21 @@ TEST_P(RtpSenderTestWithoutPacer, SendFlexfecPackets) {
>   constexpr int kFlexfecPayloadType = 118;
>   constexpr uint32_t kMediaSsrc = 1234;
>   constexpr uint32_t kFlexfecSsrc = 5678;
>+  const char kNoMid[] = "";
>   const std::vector<RtpExtension> kNoRtpExtensions;
>   const std::vector<RtpExtensionSize> kNoRtpExtensionSizes;
>   FlexfecSender flexfec_sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc,
>-                               kNoRtpExtensions, kNoRtpExtensionSizes,
>+                               kNoMid, kNoRtpExtensions, kNoRtpExtensionSizes,
>                                nullptr /* rtp_state */, &fake_clock_);
> 
>   // Reset |rtp_sender_| to use FlexFEC.
>-  rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport_, nullptr,
>-                                  &flexfec_sender, &seq_num_allocator_, nullptr,
>-                                  nullptr, nullptr, nullptr,
>-                                  &mock_rtc_event_log_, &send_packet_observer_,
>-                                  &retransmission_rate_limiter_, nullptr));
>+  rtp_sender_.reset(
>+      new RTPSender(false, &fake_clock_, &transport_, nullptr, &flexfec_sender,
>+                    &seq_num_allocator_, nullptr, nullptr, nullptr, nullptr,
>+                    &mock_rtc_event_log_, &send_packet_observer_,
>+                    &retransmission_rate_limiter_, nullptr, false));
>   rtp_sender_->SetSSRC(kMediaSsrc);
>   rtp_sender_->SetSequenceNumber(kSeqNum);
>-  rtp_sender_->SetSendPayloadType(kMediaPayloadType);
> 
>   // Parameters selected to generate a single FEC packet per media packet.
>   FecProtectionParams params;
>@@ -1110,15 +1196,39 @@ TEST_P(RtpSenderTestWithoutPacer, SendFlexfecPackets) {
>   EXPECT_EQ(kFlexfecSsrc, flexfec_packet.Ssrc());
> }
> 
>+// Test that the MID header extension is included on sent packets when
>+// configured.
>+TEST_P(RtpSenderTestWithoutPacer, MidIncludedOnSentPackets) {
>+  const char kMid[] = "mid";
>+
>+  // Register MID header extension and set the MID for the RTPSender.
>+  rtp_sender_->SetSendingMediaStatus(false);
>+  rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionMid, kMidExtensionId);
>+  rtp_sender_->SetMid(kMid);
>+  rtp_sender_->SetSendingMediaStatus(true);
>+
>+  // Send a couple packets.
>+  SendGenericPayload();
>+  SendGenericPayload();
>+
>+  // Expect both packets to have the MID set.
>+  ASSERT_EQ(2u, transport_.sent_packets_.size());
>+  for (const RtpPacketReceived& packet : transport_.sent_packets_) {
>+    std::string mid;
>+    ASSERT_TRUE(packet.GetExtension<RtpMid>(&mid));
>+    EXPECT_EQ(kMid, mid);
>+  }
>+}
>+
> TEST_P(RtpSenderTest, FecOverheadRate) {
>-  constexpr int kMediaPayloadType = 127;
>   constexpr int kFlexfecPayloadType = 118;
>   constexpr uint32_t kMediaSsrc = 1234;
>   constexpr uint32_t kFlexfecSsrc = 5678;
>+  const char kNoMid[] = "";
>   const std::vector<RtpExtension> kNoRtpExtensions;
>   const std::vector<RtpExtensionSize> kNoRtpExtensionSizes;
>   FlexfecSender flexfec_sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc,
>-                               kNoRtpExtensions, kNoRtpExtensionSizes,
>+                               kNoMid, kNoRtpExtensions, kNoRtpExtensionSizes,
>                                nullptr /* rtp_state */, &fake_clock_);
> 
>   // Reset |rtp_sender_| to use FlexFEC.
>@@ -1126,10 +1236,9 @@ TEST_P(RtpSenderTest, FecOverheadRate) {
>       false, &fake_clock_, &transport_, &mock_paced_sender_, &flexfec_sender,
>       &seq_num_allocator_, nullptr, nullptr, nullptr, nullptr,
>       &mock_rtc_event_log_, &send_packet_observer_,
>-      &retransmission_rate_limiter_, nullptr));
>+      &retransmission_rate_limiter_, nullptr, false));
>   rtp_sender_->SetSSRC(kMediaSsrc);
>   rtp_sender_->SetSequenceNumber(kSeqNum);
>-  rtp_sender_->SetSendPayloadType(kMediaPayloadType);
> 
>   // Parameters selected to generate a single FEC packet per media packet.
>   FecProtectionParams params;
>@@ -1162,7 +1271,7 @@ TEST_P(RtpSenderTest, FrameCountCallbacks) {
>   class TestCallback : public FrameCountObserver {
>    public:
>     TestCallback() : FrameCountObserver(), num_calls_(0), ssrc_(0) {}
>-    virtual ~TestCallback() {}
>+    ~TestCallback() override = default;
> 
>     void FrameCountUpdated(const FrameCounts& frame_counts,
>                            uint32_t ssrc) override {
>@@ -1176,10 +1285,10 @@ TEST_P(RtpSenderTest, FrameCountCallbacks) {
>     FrameCounts frame_counts_;
>   } callback;
> 
>-  rtp_sender_.reset(
>-      new RTPSender(false, &fake_clock_, &transport_, &mock_paced_sender_,
>-                    nullptr, nullptr, nullptr, nullptr, &callback, nullptr,
>-                    nullptr, nullptr, &retransmission_rate_limiter_, nullptr));
>+  rtp_sender_.reset(new RTPSender(
>+      false, &fake_clock_, &transport_, &mock_paced_sender_, nullptr, nullptr,
>+      nullptr, nullptr, &callback, nullptr, nullptr, nullptr,
>+      &retransmission_rate_limiter_, nullptr, false));
>   rtp_sender_->SetSSRC(kSsrc);
>   char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC";
>   const uint8_t payload_type = 127;
>@@ -1192,9 +1301,10 @@ TEST_P(RtpSenderTest, FrameCountCallbacks) {
>   EXPECT_CALL(mock_paced_sender_, InsertPacket(_, _, _, _, _, _))
>       .Times(::testing::AtLeast(2));
> 
>+  RTPVideoHeader video_header;
>   ASSERT_TRUE(rtp_sender_->SendOutgoingData(
>       kVideoFrameKey, payload_type, 1234, 4321, payload, sizeof(payload),
>-      nullptr, nullptr, nullptr, kDefaultExpectedRetransmissionTimeMs));
>+      nullptr, &video_header, nullptr, kDefaultExpectedRetransmissionTimeMs));
> 
>   EXPECT_EQ(1U, callback.num_calls_);
>   EXPECT_EQ(ssrc, callback.ssrc_);
>@@ -1203,7 +1313,7 @@ TEST_P(RtpSenderTest, FrameCountCallbacks) {
> 
>   ASSERT_TRUE(rtp_sender_->SendOutgoingData(
>       kVideoFrameDelta, payload_type, 1234, 4321, payload, sizeof(payload),
>-      nullptr, nullptr, nullptr, kDefaultExpectedRetransmissionTimeMs));
>+      nullptr, &video_header, nullptr, kDefaultExpectedRetransmissionTimeMs));
> 
>   EXPECT_EQ(2U, callback.num_calls_);
>   EXPECT_EQ(ssrc, callback.ssrc_);
>@@ -1222,7 +1332,7 @@ TEST_P(RtpSenderTest, BitrateCallbacks) {
>           ssrc_(0),
>           total_bitrate_(0),
>           retransmit_bitrate_(0) {}
>-    virtual ~TestCallback() {}
>+    ~TestCallback() override = default;
> 
>     void Notify(uint32_t total_bitrate,
>                 uint32_t retransmit_bitrate,
>@@ -1238,10 +1348,10 @@ TEST_P(RtpSenderTest, BitrateCallbacks) {
>     uint32_t total_bitrate_;
>     uint32_t retransmit_bitrate_;
>   } callback;
>-  rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport_, nullptr,
>-                                  nullptr, nullptr, nullptr, &callback, nullptr,
>-                                  nullptr, nullptr, nullptr,
>-                                  &retransmission_rate_limiter_, nullptr));
>+  rtp_sender_.reset(
>+      new RTPSender(false, &fake_clock_, &transport_, nullptr, nullptr, nullptr,
>+                    nullptr, &callback, nullptr, nullptr, nullptr, nullptr,
>+                    &retransmission_rate_limiter_, nullptr, false));
>   rtp_sender_->SetSSRC(kSsrc);
> 
>   // Simulate kNumPackets sent with kPacketInterval ms intervals, with the
>@@ -1266,10 +1376,11 @@ TEST_P(RtpSenderTest, BitrateCallbacks) {
>   rtp_sender_->ProcessBitrate();
> 
>   // Send a few frames.
>+  RTPVideoHeader video_header;
>   for (uint32_t i = 0; i < kNumPackets; ++i) {
>     ASSERT_TRUE(rtp_sender_->SendOutgoingData(
>         kVideoFrameKey, payload_type, 1234, 4321, payload, sizeof(payload),
>-        nullptr, nullptr, nullptr, kDefaultExpectedRetransmissionTimeMs));
>+        nullptr, &video_header, nullptr, kDefaultExpectedRetransmissionTimeMs));
>     fake_clock_.AdvanceTimeMilliseconds(kPacketInterval);
>   }
> 
>@@ -1297,10 +1408,10 @@ class RtpSenderAudioTest : public RtpSenderTest {
> 
>   void SetUp() override {
>     payload_ = kAudioPayload;
>-    rtp_sender_.reset(new RTPSender(true, &fake_clock_, &transport_, nullptr,
>-                                    nullptr, nullptr, nullptr, nullptr, nullptr,
>-                                    nullptr, nullptr, nullptr,
>-                                    &retransmission_rate_limiter_, nullptr));
>+    rtp_sender_.reset(
>+        new RTPSender(true, &fake_clock_, &transport_, nullptr, nullptr,
>+                      nullptr, nullptr, nullptr, nullptr, nullptr, nullptr,
>+                      nullptr, &retransmission_rate_limiter_, nullptr, false));
>     rtp_sender_->SetSSRC(kSsrc);
>     rtp_sender_->SetSequenceNumber(kSeqNum);
>   }
>@@ -1310,7 +1421,7 @@ TEST_P(RtpSenderTestWithoutPacer, StreamDataCountersCallbacks) {
>   class TestCallback : public StreamDataCountersCallback {
>    public:
>     TestCallback() : StreamDataCountersCallback(), ssrc_(0), counters_() {}
>-    virtual ~TestCallback() {}
>+    ~TestCallback() override = default;
> 
>     void DataCountersUpdated(const StreamDataCounters& counters,
>                              uint32_t ssrc) override {
>@@ -1350,9 +1461,10 @@ TEST_P(RtpSenderTestWithoutPacer, StreamDataCountersCallbacks) {
>   rtp_sender_->RegisterRtpStatisticsCallback(&callback);
> 
>   // Send a frame.
>+  RTPVideoHeader video_header;
>   ASSERT_TRUE(rtp_sender_->SendOutgoingData(
>       kVideoFrameKey, payload_type, 1234, 4321, payload, sizeof(payload),
>-      nullptr, nullptr, nullptr, kDefaultExpectedRetransmissionTimeMs));
>+      nullptr, &video_header, nullptr, kDefaultExpectedRetransmissionTimeMs));
>   StreamDataCounters expected;
>   expected.transmitted.payload_bytes = 6;
>   expected.transmitted.header_bytes = 12;
>@@ -1367,7 +1479,7 @@ TEST_P(RtpSenderTestWithoutPacer, StreamDataCountersCallbacks) {
> 
>   // Retransmit a frame.
>   uint16_t seqno = rtp_sender_->SequenceNumber() - 1;
>-  rtp_sender_->ReSendPacket(seqno, 0);
>+  rtp_sender_->ReSendPacket(seqno);
>   expected.transmitted.payload_bytes = 12;
>   expected.transmitted.header_bytes = 24;
>   expected.transmitted.packets = 2;
>@@ -1394,7 +1506,7 @@ TEST_P(RtpSenderTestWithoutPacer, StreamDataCountersCallbacks) {
>   rtp_sender_->SetFecParameters(fec_params, fec_params);
>   ASSERT_TRUE(rtp_sender_->SendOutgoingData(
>       kVideoFrameDelta, payload_type, 1234, 4321, payload, sizeof(payload),
>-      nullptr, nullptr, nullptr, kDefaultExpectedRetransmissionTimeMs));
>+      nullptr, &video_header, nullptr, kDefaultExpectedRetransmissionTimeMs));
>   expected.transmitted.payload_bytes = 40;
>   expected.transmitted.header_bytes = 60;
>   expected.transmitted.packets = 5;
>@@ -1411,9 +1523,10 @@ TEST_P(RtpSenderAudioTest, SendAudio) {
>                                             0, 1500));
>   uint8_t payload[] = {47, 11, 32, 93, 89};
> 
>+  RTPVideoHeader video_header;
>   ASSERT_TRUE(rtp_sender_->SendOutgoingData(
>       kAudioFrameCN, payload_type, 1234, 4321, payload, sizeof(payload),
>-      nullptr, nullptr, nullptr, kDefaultExpectedRetransmissionTimeMs));
>+      nullptr, &video_header, nullptr, kDefaultExpectedRetransmissionTimeMs));
> 
>   auto sent_payload = transport_.last_sent_packet().payload();
>   EXPECT_THAT(sent_payload, ElementsAreArray(payload));
>@@ -1430,9 +1543,10 @@ TEST_P(RtpSenderAudioTest, SendAudioWithAudioLevelExtension) {
>                                             0, 1500));
>   uint8_t payload[] = {47, 11, 32, 93, 89};
> 
>+  RTPVideoHeader video_header;
>   ASSERT_TRUE(rtp_sender_->SendOutgoingData(
>       kAudioFrameCN, payload_type, 1234, 4321, payload, sizeof(payload),
>-      nullptr, nullptr, nullptr, kDefaultExpectedRetransmissionTimeMs));
>+      nullptr, &video_header, nullptr, kDefaultExpectedRetransmissionTimeMs));
> 
>   auto sent_payload = transport_.last_sent_packet().payload();
>   EXPECT_THAT(sent_payload, ElementsAreArray(payload));
>@@ -1469,22 +1583,23 @@ TEST_P(RtpSenderAudioTest, CheckMarkerBitForTelephoneEvents) {
>   // During start, it takes the starting timestamp as last sent timestamp.
>   // The duration is calculated as the difference of current and last sent
>   // timestamp. So for first call it will skip since the duration is zero.
>+  RTPVideoHeader video_header;
>   ASSERT_TRUE(rtp_sender_->SendOutgoingData(
>       kEmptyFrame, kPayloadType, capture_time_ms, 0, nullptr, 0, nullptr,
>-      nullptr, nullptr, kDefaultExpectedRetransmissionTimeMs));
>+      &video_header, nullptr, kDefaultExpectedRetransmissionTimeMs));
>   // DTMF Sample Length is (Frequency/1000) * Duration.
>   // So in this case, it is (8000/1000) * 500 = 4000.
>   // Sending it as two packets.
>   ASSERT_TRUE(rtp_sender_->SendOutgoingData(
>       kEmptyFrame, kPayloadType, capture_time_ms + 2000, 0, nullptr, 0, nullptr,
>-      nullptr, nullptr, kDefaultExpectedRetransmissionTimeMs));
>+      &video_header, nullptr, kDefaultExpectedRetransmissionTimeMs));
> 
>   // Marker Bit should be set to 1 for first packet.
>   EXPECT_TRUE(transport_.last_sent_packet().Marker());
> 
>   ASSERT_TRUE(rtp_sender_->SendOutgoingData(
>       kEmptyFrame, kPayloadType, capture_time_ms + 4000, 0, nullptr, 0, nullptr,
>-      nullptr, nullptr, kDefaultExpectedRetransmissionTimeMs));
>+      &video_header, nullptr, kDefaultExpectedRetransmissionTimeMs));
>   // Marker Bit should be set to 0 for rest of the packets.
>   EXPECT_FALSE(transport_.last_sent_packet().Marker());
> }
>@@ -1501,9 +1616,10 @@ TEST_P(RtpSenderTestWithoutPacer, BytesReportedCorrectly) {
>                                             0, 1500));
>   uint8_t payload[] = {47, 11, 32, 93, 89};
> 
>+  RTPVideoHeader video_header;
>   ASSERT_TRUE(rtp_sender_->SendOutgoingData(
>       kVideoFrameKey, kPayloadType, 1234, 4321, payload, sizeof(payload),
>-      nullptr, nullptr, nullptr, kDefaultExpectedRetransmissionTimeMs));
>+      nullptr, &video_header, nullptr, kDefaultExpectedRetransmissionTimeMs));
> 
>   // Will send 2 full-size padding packets.
>   rtp_sender_->TimeToSendPadding(1, PacedPacketInfo());
>@@ -1573,9 +1689,9 @@ TEST_P(RtpSenderVideoTest, KeyFrameHasCVO) {
>   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
>                    kRtpExtensionVideoRotation, kVideoRotationExtensionId));
> 
>-  RTPVideoHeader hdr = {0};
>+  RTPVideoHeader hdr;
>   hdr.rotation = kVideoRotation_0;
>-  rtp_sender_video_->SendVideo(kRtpVideoGeneric, kVideoFrameKey, kPayload,
>+  rtp_sender_video_->SendVideo(kVideoCodecGeneric, kVideoFrameKey, kPayload,
>                                kTimestamp, 0, kFrame, sizeof(kFrame), nullptr,
>                                &hdr, kDefaultExpectedRetransmissionTimeMs);
> 
>@@ -1595,13 +1711,13 @@ TEST_P(RtpSenderVideoTest, TimingFrameHasPacketizationTimstampSet) {
> 
>   const int64_t kCaptureTimestamp = fake_clock_.TimeInMilliseconds();
> 
>-  RTPVideoHeader hdr = {0};
>-  hdr.video_timing.flags = TimingFrameFlags::kTriggeredByTimer;
>+  RTPVideoHeader hdr;
>+  hdr.video_timing.flags = VideoSendTiming::kTriggeredByTimer;
>   hdr.video_timing.encode_start_delta_ms = kEncodeStartDeltaMs;
>   hdr.video_timing.encode_finish_delta_ms = kEncodeFinishDeltaMs;
> 
>   fake_clock_.AdvanceTimeMilliseconds(kPacketizationTimeMs);
>-  rtp_sender_video_->SendVideo(kRtpVideoGeneric, kVideoFrameKey, kPayload,
>+  rtp_sender_video_->SendVideo(kVideoCodecGeneric, kVideoFrameKey, kPayload,
>                                kTimestamp, kCaptureTimestamp, kFrame,
>                                sizeof(kFrame), nullptr, &hdr,
>                                kDefaultExpectedRetransmissionTimeMs);
>@@ -1618,15 +1734,15 @@ TEST_P(RtpSenderVideoTest, DeltaFrameHasCVOWhenChanged) {
>   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
>                    kRtpExtensionVideoRotation, kVideoRotationExtensionId));
> 
>-  RTPVideoHeader hdr = {0};
>+  RTPVideoHeader hdr;
>   hdr.rotation = kVideoRotation_90;
>   EXPECT_TRUE(rtp_sender_video_->SendVideo(
>-      kRtpVideoGeneric, kVideoFrameKey, kPayload, kTimestamp, 0, kFrame,
>+      kVideoCodecGeneric, kVideoFrameKey, kPayload, kTimestamp, 0, kFrame,
>       sizeof(kFrame), nullptr, &hdr, kDefaultExpectedRetransmissionTimeMs));
> 
>   hdr.rotation = kVideoRotation_0;
>   EXPECT_TRUE(rtp_sender_video_->SendVideo(
>-      kRtpVideoGeneric, kVideoFrameDelta, kPayload, kTimestamp + 1, 0, kFrame,
>+      kVideoCodecGeneric, kVideoFrameDelta, kPayload, kTimestamp + 1, 0, kFrame,
>       sizeof(kFrame), nullptr, &hdr, kDefaultExpectedRetransmissionTimeMs));
> 
>   VideoRotation rotation;
>@@ -1640,14 +1756,14 @@ TEST_P(RtpSenderVideoTest, DeltaFrameHasCVOWhenNonZero) {
>   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
>                    kRtpExtensionVideoRotation, kVideoRotationExtensionId));
> 
>-  RTPVideoHeader hdr = {0};
>+  RTPVideoHeader hdr;
>   hdr.rotation = kVideoRotation_90;
>   EXPECT_TRUE(rtp_sender_video_->SendVideo(
>-      kRtpVideoGeneric, kVideoFrameKey, kPayload, kTimestamp, 0, kFrame,
>+      kVideoCodecGeneric, kVideoFrameKey, kPayload, kTimestamp, 0, kFrame,
>       sizeof(kFrame), nullptr, &hdr, kDefaultExpectedRetransmissionTimeMs));
> 
>   EXPECT_TRUE(rtp_sender_video_->SendVideo(
>-      kRtpVideoGeneric, kVideoFrameDelta, kPayload, kTimestamp + 1, 0, kFrame,
>+      kVideoCodecGeneric, kVideoFrameDelta, kPayload, kTimestamp + 1, 0, kFrame,
>       sizeof(kFrame), nullptr, &hdr, kDefaultExpectedRetransmissionTimeMs));
> 
>   VideoRotation rotation;
>@@ -1679,7 +1795,7 @@ TEST_P(RtpSenderVideoTest, SendVideoWithCameraAndFlipCVO) {
> 
> TEST_P(RtpSenderVideoTest, RetransmissionTypesGeneric) {
>   RTPVideoHeader header;
>-  header.codec = kRtpVideoGeneric;
>+  header.codec = kVideoCodecGeneric;
> 
>   EXPECT_EQ(kDontRetransmit,
>             rtp_sender_video_->GetStorageType(
>@@ -1701,9 +1817,9 @@ TEST_P(RtpSenderVideoTest, RetransmissionTypesGeneric) {
> 
> TEST_P(RtpSenderVideoTest, RetransmissionTypesH264) {
>   RTPVideoHeader header;
>-  header.codec = kRtpVideoH264;
>-  header.codecHeader.H264.packetization_mode =
>+  header.video_type_header.emplace<RTPVideoHeaderH264>().packetization_mode =
>       H264PacketizationMode::NonInterleaved;
>+  header.codec = kVideoCodecH264;
> 
>   EXPECT_EQ(kDontRetransmit,
>             rtp_sender_video_->GetStorageType(
>@@ -1725,8 +1841,8 @@ TEST_P(RtpSenderVideoTest, RetransmissionTypesH264) {
> 
> TEST_P(RtpSenderVideoTest, RetransmissionTypesVP8BaseLayer) {
>   RTPVideoHeader header;
>-  header.codec = kRtpVideoVp8;
>-  header.codecHeader.VP8.temporalIdx = 0;
>+  header.codec = kVideoCodecVP8;
>+  header.vp8().temporalIdx = 0;
> 
>   EXPECT_EQ(kDontRetransmit,
>             rtp_sender_video_->GetStorageType(
>@@ -1756,10 +1872,10 @@ TEST_P(RtpSenderVideoTest, RetransmissionTypesVP8BaseLayer) {
> 
> TEST_P(RtpSenderVideoTest, RetransmissionTypesVP8HigherLayers) {
>   RTPVideoHeader header;
>-  header.codec = kRtpVideoVp8;
>+  header.codec = kVideoCodecVP8;
> 
>   for (int tid = 1; tid <= kMaxTemporalStreams; ++tid) {
>-    header.codecHeader.VP8.temporalIdx = tid;
>+    header.vp8().temporalIdx = tid;
> 
>     EXPECT_EQ(kDontRetransmit, rtp_sender_video_->GetStorageType(
>                                    header, kRetransmitOff,
>@@ -1782,10 +1898,11 @@ TEST_P(RtpSenderVideoTest, RetransmissionTypesVP8HigherLayers) {
> 
> TEST_P(RtpSenderVideoTest, RetransmissionTypesVP9) {
>   RTPVideoHeader header;
>-  header.codec = kRtpVideoVp9;
>+  header.codec = kVideoCodecVP9;
> 
>+  auto& vp9_header = header.video_type_header.emplace<RTPVideoHeaderVP9>();
>   for (int tid = 1; tid <= kMaxTemporalStreams; ++tid) {
>-    header.codecHeader.VP9.temporal_idx = tid;
>+    vp9_header.temporal_idx = tid;
> 
>     EXPECT_EQ(kDontRetransmit, rtp_sender_video_->GetStorageType(
>                                    header, kRetransmitOff,
>@@ -1814,7 +1931,7 @@ TEST_P(RtpSenderVideoTest, ConditionalRetransmit) {
> 
>   // Insert VP8 frames for all temporal layers, but stop before the final index.
>   RTPVideoHeader header;
>-  header.codec = kRtpVideoVp8;
>+  header.codec = kVideoCodecVP8;
> 
>   // Fill averaging window to prevent rounding errors.
>   constexpr int kNumRepetitions =
>@@ -1822,7 +1939,7 @@ TEST_P(RtpSenderVideoTest, ConditionalRetransmit) {
>       kFrameIntervalMs;
>   constexpr int kPattern[] = {0, 2, 1, 2};
>   for (size_t i = 0; i < arraysize(kPattern) * kNumRepetitions; ++i) {
>-    header.codecHeader.VP8.temporalIdx = kPattern[i % arraysize(kPattern)];
>+    header.vp8().temporalIdx = kPattern[i % arraysize(kPattern)];
>     rtp_sender_video_->GetStorageType(header, kSettings, kRttMs);
>     fake_clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
>   }
>@@ -1831,7 +1948,7 @@ TEST_P(RtpSenderVideoTest, ConditionalRetransmit) {
>   // right now. We will wait at most one expected retransmission time before
>   // acknowledging that it did not arrive, which means this frame and the next
>   // will not be retransmitted.
>-  header.codecHeader.VP8.temporalIdx = 1;
>+  header.vp8().temporalIdx = 1;
>   EXPECT_EQ(StorageType::kDontRetransmit,
>             rtp_sender_video_->GetStorageType(header, kSettings, kRttMs));
>   fake_clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
>@@ -1847,7 +1964,7 @@ TEST_P(RtpSenderVideoTest, ConditionalRetransmit) {
>   // Insert a frame for TL2. We just had frame in TL1, so the next one there is
>   // in three frames away. TL0 is still too far in the past. So, allow
>   // retransmission.
>-  header.codecHeader.VP8.temporalIdx = 2;
>+  header.vp8().temporalIdx = 2;
>   EXPECT_EQ(StorageType::kAllowRetransmission,
>             rtp_sender_video_->GetStorageType(header, kSettings, kRttMs));
>   fake_clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
>@@ -1871,7 +1988,7 @@ TEST_P(RtpSenderVideoTest, ConditionalRetransmitLimit) {
> 
>   // Insert VP8 frames for all temporal layers, but stop before the final index.
>   RTPVideoHeader header;
>-  header.codec = kRtpVideoVp8;
>+  header.codec = kVideoCodecVP8;
> 
>   // Fill averaging window to prevent rounding errors.
>   constexpr int kNumRepetitions =
>@@ -1879,7 +1996,7 @@ TEST_P(RtpSenderVideoTest, ConditionalRetransmitLimit) {
>       kFrameIntervalMs;
>   constexpr int kPattern[] = {0, 2, 2, 2};
>   for (size_t i = 0; i < arraysize(kPattern) * kNumRepetitions; ++i) {
>-    header.codecHeader.VP8.temporalIdx = kPattern[i % arraysize(kPattern)];
>+    header.vp8().temporalIdx = kPattern[i % arraysize(kPattern)];
> 
>     rtp_sender_video_->GetStorageType(header, kSettings, kRttMs);
>     fake_clock_.AdvanceTimeMilliseconds(kFrameIntervalMs);
>@@ -1890,17 +2007,17 @@ TEST_P(RtpSenderVideoTest, ConditionalRetransmitLimit) {
>   // we don't store for retransmission because we expect a frame in a lower
>   // layer, but that last frame in TL1 was a long time ago in absolute terms,
>   // so allow retransmission anyway.
>-  header.codecHeader.VP8.temporalIdx = 1;
>+  header.vp8().temporalIdx = 1;
>   EXPECT_EQ(StorageType::kAllowRetransmission,
>             rtp_sender_video_->GetStorageType(header, kSettings, kRttMs));
> }
> 
> TEST_P(RtpSenderTest, OnOverheadChanged) {
>   MockOverheadObserver mock_overhead_observer;
>-  rtp_sender_.reset(
>-      new RTPSender(false, &fake_clock_, &transport_, nullptr, nullptr, nullptr,
>-                    nullptr, nullptr, nullptr, nullptr, nullptr, nullptr,
>-                    &retransmission_rate_limiter_, &mock_overhead_observer));
>+  rtp_sender_.reset(new RTPSender(
>+      false, &fake_clock_, &transport_, nullptr, nullptr, nullptr, nullptr,
>+      nullptr, nullptr, nullptr, nullptr, nullptr,
>+      &retransmission_rate_limiter_, &mock_overhead_observer, false));
>   rtp_sender_->SetSSRC(kSsrc);
> 
>   // RTP overhead is 12B.
>@@ -1918,10 +2035,10 @@ TEST_P(RtpSenderTest, OnOverheadChanged) {
> 
> TEST_P(RtpSenderTest, DoesNotUpdateOverheadOnEqualSize) {
>   MockOverheadObserver mock_overhead_observer;
>-  rtp_sender_.reset(
>-      new RTPSender(false, &fake_clock_, &transport_, nullptr, nullptr, nullptr,
>-                    nullptr, nullptr, nullptr, nullptr, nullptr, nullptr,
>-                    &retransmission_rate_limiter_, &mock_overhead_observer));
>+  rtp_sender_.reset(new RTPSender(
>+      false, &fake_clock_, &transport_, nullptr, nullptr, nullptr, nullptr,
>+      nullptr, nullptr, nullptr, nullptr, nullptr,
>+      &retransmission_rate_limiter_, &mock_overhead_observer, false));
>   rtp_sender_->SetSSRC(kSsrc);
> 
>   EXPECT_CALL(mock_overhead_observer, OnOverheadChanged(_)).Times(1);
>@@ -1929,40 +2046,12 @@ TEST_P(RtpSenderTest, DoesNotUpdateOverheadOnEqualSize) {
>   SendGenericPayload();
> }
> 
>-TEST_P(RtpSenderTest, SendAudioPadding) {
>-  MockTransport transport;
>-  const bool kEnableAudio = true;
>-  rtp_sender_.reset(new RTPSender(
>-      kEnableAudio, &fake_clock_, &transport, &mock_paced_sender_, nullptr,
>-      nullptr, nullptr, nullptr, nullptr, nullptr, &mock_rtc_event_log_,
>-      nullptr, &retransmission_rate_limiter_, nullptr));
>-  rtp_sender_->SetSendPayloadType(kPayload);
>-  rtp_sender_->SetSequenceNumber(kSeqNum);
>-  rtp_sender_->SetTimestampOffset(0);
>-  rtp_sender_->SetSSRC(kSsrc);
>-
>-  const size_t kPaddingSize = 59;
>-  EXPECT_CALL(transport, SendRtp(_, kPaddingSize + kRtpHeaderSize, _))
>-      .WillOnce(testing::Return(true));
>-  EXPECT_EQ(kPaddingSize,
>-            rtp_sender_->TimeToSendPadding(kPaddingSize, PacedPacketInfo()));
>-
>-  // Requested padding size is too small, will send a larger one.
>-  const size_t kMinPaddingSize = 50;
>-  EXPECT_CALL(transport, SendRtp(_, kMinPaddingSize + kRtpHeaderSize, _))
>-      .WillOnce(testing::Return(true));
>-  EXPECT_EQ(
>-      kMinPaddingSize,
>-      rtp_sender_->TimeToSendPadding(kMinPaddingSize - 5, PacedPacketInfo()));
>-}
>-
> TEST_P(RtpSenderTest, SendsKeepAlive) {
>   MockTransport transport;
>-  rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport, nullptr,
>-                                  nullptr, nullptr, nullptr, nullptr, nullptr,
>-                                  nullptr, &mock_rtc_event_log_, nullptr,
>-                                  &retransmission_rate_limiter_, nullptr));
>-  rtp_sender_->SetSendPayloadType(kPayload);
>+  rtp_sender_.reset(
>+      new RTPSender(false, &fake_clock_, &transport, nullptr, nullptr, nullptr,
>+                    nullptr, nullptr, nullptr, nullptr, &mock_rtc_event_log_,
>+                    nullptr, &retransmission_rate_limiter_, nullptr, false));
>   rtp_sender_->SetSequenceNumber(kSeqNum);
>   rtp_sender_->SetTimestampOffset(0);
>   rtp_sender_->SetSSRC(kSsrc);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
>index da756889c02..43450dc21e9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc
>@@ -18,6 +18,7 @@
> #include <utility>
> #include <vector>
> 
>+#include "absl/memory/memory.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> #include "modules/rtp_rtcp/source/byte_io.h"
> #include "modules/rtp_rtcp/source/rtp_format_video_generic.h"
>@@ -27,7 +28,6 @@
> #include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/trace_event.h"
> 
> namespace webrtc {
>@@ -54,7 +54,7 @@ RTPSenderVideo::RTPSenderVideo(Clock* clock,
>                                FlexfecSender* flexfec_sender)
>     : rtp_sender_(rtp_sender),
>       clock_(clock),
>-      video_type_(kRtpVideoGeneric),
>+      video_type_(kVideoCodecGeneric),
>       retransmission_settings_(kRetransmitBaseLayer |
>                                kConditionallyRetransmitHigherLayers),
>       last_rotation_(kVideoRotation_0),
>@@ -68,11 +68,11 @@ RTPSenderVideo::RTPSenderVideo(Clock* clock,
> 
> RTPSenderVideo::~RTPSenderVideo() {}
> 
>-void RTPSenderVideo::SetVideoCodecType(RtpVideoCodecTypes video_type) {
>+void RTPSenderVideo::SetVideoCodecType(enum VideoCodecType video_type) {
>   video_type_ = video_type;
> }
> 
>-RtpVideoCodecTypes RTPSenderVideo::VideoCodecType() const {
>+VideoCodecType RTPSenderVideo::VideoCodecType() const {
>   return video_type_;
> }
> 
>@@ -80,19 +80,19 @@ RtpVideoCodecTypes RTPSenderVideo::VideoCodecType() const {
> RtpUtility::Payload* RTPSenderVideo::CreateVideoPayload(
>     const char payload_name[RTP_PAYLOAD_NAME_SIZE],
>     int8_t payload_type) {
>-  RtpVideoCodecTypes video_type = kRtpVideoGeneric;
>+  enum VideoCodecType video_type = kVideoCodecGeneric;
>   if (RtpUtility::StringCompare(payload_name, "VP8", 3)) {
>-    video_type = kRtpVideoVp8;
>+    video_type = kVideoCodecVP8;
>   } else if (RtpUtility::StringCompare(payload_name, "VP9", 3)) {
>-    video_type = kRtpVideoVp9;
>+    video_type = kVideoCodecVP9;
>   } else if (RtpUtility::StringCompare(payload_name, "H264", 4)) {
>-    video_type = kRtpVideoH264;
>+    video_type = kVideoCodecH264;
>   } else if (RtpUtility::StringCompare(payload_name, "I420", 4)) {
>-    video_type = kRtpVideoGeneric;
>+    video_type = kVideoCodecGeneric;
>   } else if (RtpUtility::StringCompare(payload_name, "stereo", 6)) {
>-    video_type = kRtpVideoStereo;
>+    video_type = kVideoCodecGeneric;
>   } else {
>-    video_type = kRtpVideoGeneric;
>+    video_type = kVideoCodecGeneric;
>   }
>   VideoPayload vp;
>   vp.videoCodecType = video_type;
>@@ -104,7 +104,6 @@ void RTPSenderVideo::SendVideoPacket(std::unique_ptr<RtpPacketToSend> packet,
>   // Remember some values about the packet before sending it away.
>   size_t packet_size = packet->size();
>   uint16_t seq_num = packet->SequenceNumber();
>-  uint32_t rtp_timestamp = packet->Timestamp();
>   if (!rtp_sender_->SendToNetwork(std::move(packet), storage,
>                                   RtpPacketSender::kLowPriority)) {
>     RTC_LOG(LS_WARNING) << "Failed to send video packet " << seq_num;
>@@ -112,16 +111,12 @@ void RTPSenderVideo::SendVideoPacket(std::unique_ptr<RtpPacketToSend> packet,
>   }
>   rtc::CritScope cs(&stats_crit_);
>   video_bitrate_.Update(packet_size, clock_->TimeInMilliseconds());
>-  TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
>-                       "Video::PacketNormal", "timestamp", rtp_timestamp,
>-                       "seqnum", seq_num);
> }
> 
> void RTPSenderVideo::SendVideoPacketAsRedMaybeWithUlpfec(
>     std::unique_ptr<RtpPacketToSend> media_packet,
>     StorageType media_packet_storage,
>     bool protect_media_packet) {
>-  uint32_t rtp_timestamp = media_packet->Timestamp();
>   uint16_t media_seq_num = media_packet->SequenceNumber();
> 
>   std::unique_ptr<RtpPacketToSend> red_packet(
>@@ -158,9 +153,6 @@ void RTPSenderVideo::SendVideoPacketAsRedMaybeWithUlpfec(
>                                  RtpPacketSender::kLowPriority)) {
>     rtc::CritScope cs(&stats_crit_);
>     video_bitrate_.Update(red_packet_size, clock_->TimeInMilliseconds());
>-    TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
>-                         "Video::PacketRed", "timestamp", rtp_timestamp,
>-                         "seqnum", media_seq_num);
>   } else {
>     RTC_LOG(LS_WARNING) << "Failed to send RED packet " << media_seq_num;
>   }
>@@ -176,9 +168,6 @@ void RTPSenderVideo::SendVideoPacketAsRedMaybeWithUlpfec(
>                                    RtpPacketSender::kLowPriority)) {
>       rtc::CritScope cs(&stats_crit_);
>       fec_bitrate_.Update(fec_packet->length(), clock_->TimeInMilliseconds());
>-      TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
>-                           "Video::PacketUlpfec", "timestamp", rtp_timestamp,
>-                           "seqnum", fec_sequence_number);
>     } else {
>       RTC_LOG(LS_WARNING) << "Failed to send ULPFEC packet "
>                           << fec_sequence_number;
>@@ -202,15 +191,11 @@ void RTPSenderVideo::SendVideoPacketWithFlexfec(
>         flexfec_sender_->GetFecPackets();
>     for (auto& fec_packet : fec_packets) {
>       size_t packet_length = fec_packet->size();
>-      uint32_t timestamp = fec_packet->Timestamp();
>       uint16_t seq_num = fec_packet->SequenceNumber();
>       if (rtp_sender_->SendToNetwork(std::move(fec_packet), kDontRetransmit,
>                                      RtpPacketSender::kLowPriority)) {
>         rtc::CritScope cs(&stats_crit_);
>         fec_bitrate_.Update(packet_length, clock_->TimeInMilliseconds());
>-        TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
>-                             "Video::PacketFlexfec", "timestamp", timestamp,
>-                             "seqnum", seq_num);
>       } else {
>         RTC_LOG(LS_WARNING) << "Failed to send FlexFEC packet " << seq_num;
>       }
>@@ -233,10 +218,7 @@ void RTPSenderVideo::SetUlpfecConfig(int red_payload_type,
>   ulpfec_payload_type_ = ulpfec_payload_type;
> 
>   // Must not enable ULPFEC without RED.
>-  // TODO(brandtr): We currently support enabling RED without ULPFEC. Change
>-  // this when we have removed the RED/RTX send-side workaround, so that we
>-  // ensure that RED and ULPFEC are only enabled together.
>-  RTC_DCHECK(red_enabled() || !ulpfec_enabled());
>+  RTC_DCHECK(!(red_enabled() ^ ulpfec_enabled()));
> 
>   // Reset FEC parameters.
>   delta_fec_params_ = FecProtectionParams{0, 1, kFecMaskRandom};
>@@ -279,14 +261,14 @@ void RTPSenderVideo::SetFecParameters(const FecProtectionParams& delta_params,
>   key_fec_params_ = key_params;
> }
> 
>-rtc::Optional<uint32_t> RTPSenderVideo::FlexfecSsrc() const {
>+absl::optional<uint32_t> RTPSenderVideo::FlexfecSsrc() const {
>   if (flexfec_sender_) {
>     return flexfec_sender_->ssrc();
>   }
>-  return rtc::nullopt;
>+  return absl::nullopt;
> }
> 
>-bool RTPSenderVideo::SendVideo(RtpVideoCodecTypes video_type,
>+bool RTPSenderVideo::SendVideo(enum VideoCodecType video_type,
>                                FrameType frame_type,
>                                int8_t payload_type,
>                                uint32_t rtp_timestamp,
>@@ -304,7 +286,7 @@ bool RTPSenderVideo::SendVideo(RtpVideoCodecTypes video_type,
>   rtp_header->SetPayloadType(payload_type);
>   rtp_header->SetTimestamp(rtp_timestamp);
>   rtp_header->set_capture_time_ms(capture_time_ms);
>-  auto last_packet = rtc::MakeUnique<RtpPacketToSend>(*rtp_header);
>+  auto last_packet = absl::make_unique<RtpPacketToSend>(*rtp_header);
> 
>   size_t fec_packet_overhead;
>   bool red_enabled;
>@@ -335,7 +317,7 @@ bool RTPSenderVideo::SendVideo(RtpVideoCodecTypes video_type,
>         last_packet->SetExtension<VideoContentTypeExtension>(
>             video_header->content_type);
>       }
>-      if (video_header->video_timing.flags != TimingFrameFlags::kInvalid) {
>+      if (video_header->video_timing.flags != VideoSendTiming::kInvalid) {
>         last_packet->SetExtension<VideoTimingExtension>(
>             video_header->video_timing);
>       }
>@@ -367,13 +349,12 @@ bool RTPSenderVideo::SendVideo(RtpVideoCodecTypes video_type,
> 
>   std::unique_ptr<RtpPacketizer> packetizer(RtpPacketizer::Create(
>       video_type, max_data_payload_length, last_packet_reduction_len,
>-      video_header ? &(video_header->codecHeader) : nullptr, frame_type));
>+      video_header, frame_type));
> 
>   const uint8_t temporal_id =
>       video_header ? GetTemporalId(*video_header) : kNoTemporalIdx;
>   StorageType storage = GetStorageType(temporal_id, retransmission_settings,
>                                        expected_retransmission_time_ms);
>-
>   size_t num_packets =
>       packetizer->SetPayloadData(payload_data, payload_size, fragmentation);
> 
>@@ -384,7 +365,7 @@ bool RTPSenderVideo::SendVideo(RtpVideoCodecTypes video_type,
>   for (size_t i = 0; i < num_packets; ++i) {
>     bool last = (i + 1) == num_packets;
>     auto packet = last ? std::move(last_packet)
>-                       : rtc::MakeUnique<RtpPacketToSend>(*rtp_header);
>+                       : absl::make_unique<RtpPacketToSend>(*rtp_header);
>     if (!packetizer->NextPacket(packet.get()))
>       return false;
>     RTC_DCHECK_LE(packet->payload_size(),
>@@ -487,10 +468,11 @@ StorageType RTPSenderVideo::GetStorageType(
> 
> uint8_t RTPSenderVideo::GetTemporalId(const RTPVideoHeader& header) {
>   switch (header.codec) {
>-    case kRtpVideoVp8:
>-      return header.codecHeader.VP8.temporalIdx;
>-    case kRtpVideoVp9:
>-      return header.codecHeader.VP9.temporal_idx;
>+    case kVideoCodecVP8:
>+      return header.vp8().temporalIdx;
>+    case kVideoCodecVP9:
>+      return absl::get<RTPVideoHeaderVP9>(header.video_type_header)
>+          .temporal_idx;
>     default:
>       return kNoTemporalIdx;
>   }
>@@ -519,7 +501,7 @@ bool RTPSenderVideo::UpdateConditionalRetransmit(
>       int64_t expected_next_frame_time = kUndefined;
>       for (int i = temporal_id - 1; i >= 0; --i) {
>         TemporalLayerStats* stats = &frame_stats_by_temporal_layer_[i];
>-        rtc::Optional<uint32_t> rate = stats->frame_rate_fp1000s.Rate(now_ms);
>+        absl::optional<uint32_t> rate = stats->frame_rate_fp1000s.Rate(now_ms);
>         if (rate) {
>           int64_t tl_next = stats->last_frame_time_ms + 1000000 / *rate;
>           if (tl_next - now_ms > -expected_retransmission_time_ms &&
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h
>index 33bccb74071..ce7be16149c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h
>@@ -14,7 +14,7 @@
> #include <map>
> #include <memory>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/rtp_rtcp/include/flexfec_sender.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>@@ -22,13 +22,11 @@
> #include "modules/rtp_rtcp/source/rtp_sender.h"
> #include "modules/rtp_rtcp/source/rtp_utility.h"
> #include "modules/rtp_rtcp/source/ulpfec_generator.h"
>-#include "modules/rtp_rtcp/source/video_codec_information.h"
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/onetimeevent.h"
> #include "rtc_base/rate_statistics.h"
> #include "rtc_base/sequenced_task_checker.h"
> #include "rtc_base/thread_annotations.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> class RtpPacketizer;
>@@ -43,13 +41,13 @@ class RTPSenderVideo {
>                  FlexfecSender* flexfec_sender);
>   virtual ~RTPSenderVideo();
> 
>-  virtual RtpVideoCodecTypes VideoCodecType() const;
>+  virtual enum VideoCodecType VideoCodecType() const;
> 
>   static RtpUtility::Payload* CreateVideoPayload(
>       const char payload_name[RTP_PAYLOAD_NAME_SIZE],
>       int8_t payload_type);
> 
>-  bool SendVideo(RtpVideoCodecTypes video_type,
>+  bool SendVideo(enum VideoCodecType video_type,
>                  FrameType frame_type,
>                  int8_t payload_type,
>                  uint32_t capture_timestamp,
>@@ -60,7 +58,7 @@ class RTPSenderVideo {
>                  const RTPVideoHeader* video_header,
>                  int64_t expected_retransmission_time_ms);
> 
>-  void SetVideoCodecType(RtpVideoCodecTypes type);
>+  void SetVideoCodecType(enum VideoCodecType type);
> 
>   // ULPFEC.
>   void SetUlpfecConfig(int red_payload_type, int ulpfec_payload_type);
>@@ -71,7 +69,7 @@ class RTPSenderVideo {
>                         const FecProtectionParams& key_params);
> 
>   // FlexFEC.
>-  rtc::Optional<uint32_t> FlexfecSsrc() const;
>+  absl::optional<uint32_t> FlexfecSsrc() const;
> 
>   uint32_t VideoBitrateSent() const;
>   uint32_t FecOverheadRate() const;
>@@ -133,7 +131,7 @@ class RTPSenderVideo {
>   // Should never be held when calling out of this class.
>   rtc::CriticalSection crit_;
> 
>-  RtpVideoCodecTypes video_type_;
>+  enum VideoCodecType video_type_;
>   int32_t retransmission_settings_ RTC_GUARDED_BY(crit_);
>   VideoRotation last_rotation_ RTC_GUARDED_BY(crit_);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_utility.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_utility.cc
>index 204b7ed0adb..d150ce2eb35 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_utility.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_utility.cc
>@@ -18,12 +18,6 @@
> 
> namespace webrtc {
> 
>-RtpFeedback* NullObjectRtpFeedback() {
>-  static std::aligned_storage<sizeof(NullRtpFeedback), std::alignment_of<NullRtpFeedback>::value>::type null_rtp_feedback_storage;
>-  static NullRtpFeedback* null_rtp_feedback = new (&null_rtp_feedback_storage) NullRtpFeedback;
>-  return null_rtp_feedback;
>-}
>-
> namespace RtpUtility {
> 
> enum {
>@@ -39,8 +33,7 @@ enum {
>  * Misc utility routines
>  */
> 
>-bool StringCompare(const char* str1, const char* str2,
>-                   const uint32_t length) {
>+bool StringCompare(const char* str1, const char* str2, const uint32_t length) {
>   return _strnicmp(str1, str2, length) == 0;
> }
> 
>@@ -54,11 +47,9 @@ size_t Word32Align(size_t size) {
> RtpHeaderParser::RtpHeaderParser(const uint8_t* rtpData,
>                                  const size_t rtpDataLength)
>     : _ptrRTPDataBegin(rtpData),
>-      _ptrRTPDataEnd(rtpData ? (rtpData + rtpDataLength) : NULL) {
>-}
>+      _ptrRTPDataEnd(rtpData ? (rtpData + rtpDataLength) : NULL) {}
> 
>-RtpHeaderParser::~RtpHeaderParser() {
>-}
>+RtpHeaderParser::~RtpHeaderParser() {}
> 
> bool RtpHeaderParser::RTCP() const {
>   // 72 to 76 is reserved for RTP
>@@ -66,21 +57,21 @@ bool RtpHeaderParser::RTCP() const {
>   // for RTCP 200 SR  == marker bit + 72
>   // for RTCP 204 APP == marker bit + 76
>   /*
>-  *       RTCP
>-  *
>-  * FIR      full INTRA-frame request             192     [RFC2032]   supported
>-  * NACK     negative acknowledgement             193     [RFC2032]
>-  * IJ       Extended inter-arrival jitter report 195     [RFC-ietf-avt-rtp-toff
>-  * set-07.txt] http://tools.ietf.org/html/draft-ietf-avt-rtp-toffset-07
>-  * SR       sender report                        200     [RFC3551]   supported
>-  * RR       receiver report                      201     [RFC3551]   supported
>-  * SDES     source description                   202     [RFC3551]   supported
>-  * BYE      goodbye                              203     [RFC3551]   supported
>-  * APP      application-defined                  204     [RFC3551]   ignored
>-  * RTPFB    Transport layer FB message           205     [RFC4585]   supported
>-  * PSFB     Payload-specific FB message          206     [RFC4585]   supported
>-  * XR       extended report                      207     [RFC3611]   supported
>-  */
>+   *       RTCP
>+   *
>+   * FIR      full INTRA-frame request             192     [RFC2032]   supported
>+   * NACK     negative acknowledgement             193     [RFC2032]
>+   * IJ       Extended inter-arrival jitter report 195 [RFC-ietf-avt-rtp-toff
>+   * set-07.txt] http://tools.ietf.org/html/draft-ietf-avt-rtp-toffset-07
>+   * SR       sender report                        200     [RFC3551]   supported
>+   * RR       receiver report                      201     [RFC3551]   supported
>+   * SDES     source description                   202     [RFC3551]   supported
>+   * BYE      goodbye                              203     [RFC3551]   supported
>+   * APP      application-defined                  204     [RFC3551]   ignored
>+   * RTPFB    Transport layer FB message           205     [RFC4585]   supported
>+   * PSFB     Payload-specific FB message          206     [RFC4585]   supported
>+   * XR       extended report                      207     [RFC3611]   supported
>+   */
> 
>   /* 205       RFC 5104
>    * FMT 1      NACK       supported
>@@ -90,15 +81,15 @@ bool RtpHeaderParser::RTCP() const {
>    */
> 
>   /* 206      RFC 5104
>-  * FMT 1:     Picture Loss Indication (PLI)                      supported
>-  * FMT 2:     Slice Lost Indication (SLI)
>-  * FMT 3:     Reference Picture Selection Indication (RPSI)
>-  * FMT 4:     Full Intra Request (FIR) Command                   supported
>-  * FMT 5:     Temporal-Spatial Trade-off Request (TSTR)
>-  * FMT 6:     Temporal-Spatial Trade-off Notification (TSTN)
>-  * FMT 7:     Video Back Channel Message (VBCM)
>-  * FMT 15:    Application layer FB message
>-  */
>+   * FMT 1:     Picture Loss Indication (PLI)                      supported
>+   * FMT 2:     Slice Lost Indication (SLI)
>+   * FMT 3:     Reference Picture Selection Indication (RPSI)
>+   * FMT 4:     Full Intra Request (FIR) Command                   supported
>+   * FMT 5:     Temporal-Spatial Trade-off Request (TSTR)
>+   * FMT 6:     Temporal-Spatial Trade-off Notification (TSTN)
>+   * FMT 7:     Video Back Channel Message (VBCM)
>+   * FMT 15:    Application layer FB message
>+   */
> 
>   const ptrdiff_t length = _ptrRTPDataEnd - _ptrRTPDataBegin;
>   if (length < kRtcpMinHeaderLength) {
>@@ -153,33 +144,34 @@ bool RtpHeaderParser::ParseRtcp(RTPHeader* header) const {
>   uint32_t SSRC = ByteReader<uint32_t>::ReadBigEndian(ptr);
>   ptr += 4;
> 
>-  header->payloadType  = PT;
>-  header->ssrc         = SSRC;
>+  header->payloadType = PT;
>+  header->ssrc = SSRC;
>   header->headerLength = 4 + (len << 2);
> 
>   return true;
> }
> 
>-bool RtpHeaderParser::Parse(RTPHeader* header,
>-                            RtpHeaderExtensionMap* ptrExtensionMap) const {
>+bool RtpHeaderParser::Parse(
>+    RTPHeader* header,
>+    const RtpHeaderExtensionMap* ptrExtensionMap) const {
>   const ptrdiff_t length = _ptrRTPDataEnd - _ptrRTPDataBegin;
>   if (length < kRtpMinParseLength) {
>     return false;
>   }
> 
>   // Version
>-  const uint8_t V  = _ptrRTPDataBegin[0] >> 6;
>+  const uint8_t V = _ptrRTPDataBegin[0] >> 6;
>   // Padding
>-  const bool          P  = ((_ptrRTPDataBegin[0] & 0x20) == 0) ? false : true;
>+  const bool P = ((_ptrRTPDataBegin[0] & 0x20) == 0) ? false : true;
>   // eXtension
>-  const bool          X  = ((_ptrRTPDataBegin[0] & 0x10) == 0) ? false : true;
>+  const bool X = ((_ptrRTPDataBegin[0] & 0x10) == 0) ? false : true;
>   const uint8_t CC = _ptrRTPDataBegin[0] & 0x0f;
>-  const bool          M  = ((_ptrRTPDataBegin[1] & 0x80) == 0) ? false : true;
>+  const bool M = ((_ptrRTPDataBegin[1] & 0x80) == 0) ? false : true;
> 
>   const uint8_t PT = _ptrRTPDataBegin[1] & 0x7f;
> 
>-  const uint16_t sequenceNumber = (_ptrRTPDataBegin[2] << 8) +
>-      _ptrRTPDataBegin[3];
>+  const uint16_t sequenceNumber =
>+      (_ptrRTPDataBegin[2] << 8) + _ptrRTPDataBegin[3];
> 
>   const uint8_t* ptr = &_ptrRTPDataBegin[4];
> 
>@@ -199,13 +191,13 @@ bool RtpHeaderParser::Parse(RTPHeader* header,
>     return false;
>   }
> 
>-  header->markerBit      = M;
>-  header->payloadType    = PT;
>+  header->markerBit = M;
>+  header->payloadType = PT;
>   header->sequenceNumber = sequenceNumber;
>-  header->timestamp      = RTPTimestamp;
>-  header->ssrc           = SSRC;
>-  header->numCSRCs       = CC;
>-  header->paddingLength  = P ? *(_ptrRTPDataEnd - 1) : 0;
>+  header->timestamp = RTPTimestamp;
>+  header->ssrc = SSRC;
>+  header->numCSRCs = CC;
>+  header->paddingLength = P ? *(_ptrRTPDataEnd - 1) : 0;
> 
>   for (uint8_t i = 0; i < CC; ++i) {
>     uint32_t CSRC = ByteReader<uint32_t>::ReadBigEndian(ptr);
>@@ -213,7 +205,7 @@ bool RtpHeaderParser::Parse(RTPHeader* header,
>     header->arrOfCSRCs[i] = CSRC;
>   }
> 
>-  header->headerLength   = 12 + CSRCocts;
>+  header->headerLength = 12 + CSRCocts;
> 
>   // If in effect, MAY be omitted for those packets for which the offset
>   // is zero.
>@@ -275,10 +267,8 @@ bool RtpHeaderParser::Parse(RTPHeader* header,
>     static constexpr uint16_t kRtpOneByteHeaderExtensionId = 0xBEDE;
>     if (definedByProfile == kRtpOneByteHeaderExtensionId) {
>       const uint8_t* ptrRTPDataExtensionEnd = ptr + XLen;
>-      ParseOneByteExtensionHeader(header,
>-                                  ptrExtensionMap,
>-                                  ptrRTPDataExtensionEnd,
>-                                  ptr);
>+      ParseOneByteExtensionHeader(header, ptrExtensionMap,
>+                                  ptrRTPDataExtensionEnd, ptr);
>     }
>     header->headerLength += XLen;
>   }
>@@ -477,6 +467,10 @@ void RtpHeaderParser::ParseOneByteExtensionHeader(
>           header->extension.mid.Set(rtc::MakeArrayView(ptr, len + 1));
>           break;
>         }
>+        case kRtpExtensionGenericFrameDescriptor:
>+          RTC_LOG(WARNING)
>+              << "RtpGenericFrameDescriptor unsupported by rtp header parser.";
>+          break;
>         case kRtpExtensionNone:
>         case kRtpExtensionNumberOfExtensions: {
>           RTC_NOTREACHED() << "Invalid extension type: " << type;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_utility.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_utility.h
>index b71b9dca161..762f964a2e2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_utility.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_utility.h
>@@ -19,14 +19,11 @@
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> #include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
> #include "rtc_base/deprecation.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
> const uint8_t kRtpMarkerBitMask = 0x80;
> 
>-RtpFeedback* NullObjectRtpFeedback();
>-
> namespace RtpUtility {
> 
> struct Payload {
>@@ -51,7 +48,7 @@ class RtpHeaderParser {
>   bool RTCP() const;
>   bool ParseRtcp(RTPHeader* header) const;
>   bool Parse(RTPHeader* parsedPacket,
>-             RtpHeaderExtensionMap* ptrExtensionMap = nullptr) const;
>+             const RtpHeaderExtensionMap* ptrExtensionMap = nullptr) const;
> 
>  private:
>   void ParseOneByteExtensionHeader(RTPHeader* parsedPacket,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_video_header.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_video_header.cc
>new file mode 100644
>index 00000000000..cab0d2e320a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_video_header.cc
>@@ -0,0 +1,19 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/rtp_rtcp/source/rtp_video_header.h"
>+
>+namespace webrtc {
>+
>+RTPVideoHeader::RTPVideoHeader() : playout_delay(), video_timing() {}
>+RTPVideoHeader::RTPVideoHeader(const RTPVideoHeader& other) = default;
>+RTPVideoHeader::~RTPVideoHeader() = default;
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_video_header.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_video_header.h
>new file mode 100644
>index 00000000000..edeb48d5d37
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/rtp_video_header.h
>@@ -0,0 +1,71 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#ifndef MODULES_RTP_RTCP_SOURCE_RTP_VIDEO_HEADER_H_
>+#define MODULES_RTP_RTCP_SOURCE_RTP_VIDEO_HEADER_H_
>+
>+#include "absl/container/inlined_vector.h"
>+#include "absl/types/variant.h"
>+#include "api/video/video_content_type.h"
>+#include "api/video/video_rotation.h"
>+#include "api/video/video_timing.h"
>+#include "common_types.h"  // NOLINT(build/include)
>+#include "modules/video_coding/codecs/h264/include/h264_globals.h"
>+#include "modules/video_coding/codecs/vp8/include/vp8_globals.h"
>+#include "modules/video_coding/codecs/vp9/include/vp9_globals.h"
>+
>+namespace webrtc {
>+using RTPVideoTypeHeader =
>+    absl::variant<RTPVideoHeaderVP8, RTPVideoHeaderVP9, RTPVideoHeaderH264>;
>+
>+struct RTPVideoHeader {
>+  RTPVideoHeader();
>+  RTPVideoHeader(const RTPVideoHeader& other);
>+
>+  ~RTPVideoHeader();
>+
>+  // TODO(philipel): Remove when downstream projects have been updated.
>+  RTPVideoHeaderVP8& vp8() {
>+    if (!absl::holds_alternative<RTPVideoHeaderVP8>(video_type_header))
>+      video_type_header.emplace<RTPVideoHeaderVP8>();
>+
>+    return absl::get<RTPVideoHeaderVP8>(video_type_header);
>+  }
>+  // TODO(philipel): Remove when downstream projects have been updated.
>+  const RTPVideoHeaderVP8& vp8() const {
>+    if (!absl::holds_alternative<RTPVideoHeaderVP8>(video_type_header))
>+      video_type_header.emplace<RTPVideoHeaderVP8>();
>+
>+    return absl::get<RTPVideoHeaderVP8>(video_type_header);
>+  }
>+
>+  // Information for generic codec descriptor.
>+  int64_t frame_id = kNoPictureId;
>+  int spatial_index = 0;
>+  int temporal_index = 0;
>+  absl::InlinedVector<int64_t, 5> dependencies;
>+  absl::InlinedVector<int, 5> higher_spatial_layers;
>+
>+  uint16_t width = 0;
>+  uint16_t height = 0;
>+  VideoRotation rotation = VideoRotation::kVideoRotation_0;
>+  VideoContentType content_type = VideoContentType::UNSPECIFIED;
>+  bool is_first_packet_in_frame = false;
>+  uint8_t simulcastIdx = 0;
>+  VideoCodecType codec = VideoCodecType::kVideoCodecUnknown;
>+
>+  PlayoutDelay playout_delay;
>+  VideoSendTiming video_timing;
>+  // TODO(philipel): remove mutable when downstream projects have been updated.
>+  mutable RTPVideoTypeHeader video_type_header;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_RTP_RTCP_SOURCE_RTP_VIDEO_HEADER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/time_util.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/time_util.h
>index 3e7d1e29f12..672722c07f9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/time_util.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/time_util.h
>@@ -20,8 +20,8 @@ namespace webrtc {
> // Converts time obtained using rtc::TimeMicros to ntp format.
> // TimeMicrosToNtp guarantees difference of the returned values matches
> // difference of the passed values.
>-// As a result TimeMicrosToNtp(rtc::TimeMicros()) doesn't guarantte to match
>-// system time after first call.
>+// As a result TimeMicrosToNtp(rtc::TimeMicros()) doesn't guarantee to match
>+// system time.
> NtpTime TimeMicrosToNtp(int64_t time_us);
> 
> // Converts NTP timestamp to RTP timestamp.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/time_util_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/time_util_unittest.cc
>index 25cda491f9d..0ff4f7b0797 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/time_util_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/time_util_unittest.cc
>@@ -16,29 +16,6 @@
> 
> namespace webrtc {
> 
>-// Disabled as flake: http://bugs.webrtc.org/8610
>-#ifdef WEBRTC_IOS
>-#define MAYBE_TimeMicrosToNtpMatchRealTimeClockInitially \
>-  DISABLED_TimeMicrosToNtpMatchRealTimeClockInitially
>-#else
>-#define MAYBE_TimeMicrosToNtpMatchRealTimeClockInitially \
>-  TimeMicrosToNtpMatchRealTimeClockInitially
>-#endif
>-TEST(TimeUtilTest, MAYBE_TimeMicrosToNtpMatchRealTimeClockInitially) {
>-  Clock* legacy_clock = Clock::GetRealTimeClock();
>-  NtpTime before_legacy_time = TimeMicrosToNtp(rtc::TimeMicros());
>-  NtpTime legacy_time = legacy_clock->CurrentNtpTime();
>-  NtpTime after_legacy_time = TimeMicrosToNtp(rtc::TimeMicros());
>-
>-  // This test will fail once every 136 years, when NtpTime wraparound.
>-  // More often than that, it will fail if system adjust ntp time while test
>-  // is running.
>-  // To mitigate ntp time adjustment and potentional different precisions of
>-  // Clock and TimeMicrosToNtp, relax expectation by a millisecond.
>-  EXPECT_GE(legacy_time.ToMs(), before_legacy_time.ToMs() - 1);
>-  EXPECT_LE(legacy_time.ToMs(), after_legacy_time.ToMs() + 1);
>-}
>-
> TEST(TimeUtilTest, TimeMicrosToNtpDoesntChangeBetweenRuns) {
>   rtc::ScopedFakeClock clock;
>   // TimeMicrosToNtp is not pure: it behave differently between different
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/tmmbr_help.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/tmmbr_help.h
>index 46ce845507a..91aeaf46410 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/tmmbr_help.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/tmmbr_help.h
>@@ -13,7 +13,6 @@
> 
> #include <vector>
> #include "modules/rtp_rtcp/source/rtcp_packet/tmmb_item.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_generator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_generator.cc
>index 3432bdd11d1..e5777edab85 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_generator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_generator.cc
>@@ -17,7 +17,6 @@
> #include "modules/rtp_rtcp/source/byte_io.h"
> #include "modules/rtp_rtcp/source/forward_error_correction.h"
> #include "modules/rtp_rtcp/source/rtp_utility.h"
>-#include "rtc_base/basictypes.h"
> #include "rtc_base/checks.h"
> 
> namespace webrtc {
>@@ -62,6 +61,8 @@ constexpr uint32_t kUnknownSsrc = 0;
> RedPacket::RedPacket(size_t length)
>     : data_(new uint8_t[length]), length_(length), header_length_(0) {}
> 
>+RedPacket::~RedPacket() = default;
>+
> void RedPacket::CreateHeader(const uint8_t* rtp_header,
>                              size_t header_length,
>                              int red_payload_type,
>@@ -213,7 +214,7 @@ std::vector<std::unique_ptr<RedPacket>> UlpfecGenerator::GetUlpfecPacketsAsRed(
>   ForwardErrorCorrection::Packet* last_media_packet =
>       media_packets_.back().get();
>   uint16_t seq_num = first_seq_num;
>-  for (const auto& fec_packet : generated_fec_packets_) {
>+  for (const auto* fec_packet : generated_fec_packets_) {
>     // Wrap FEC packet (including FEC headers) in a RED packet. Since the
>     // FEC packets in |generated_fec_packets_| don't have RTP headers, we
>     // reuse the header from the last media packet.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_generator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_generator.h
>index 639c15be253..efc753f4c74 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_generator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_generator.h
>@@ -24,6 +24,7 @@ class FlexfecSender;
> class RedPacket {
>  public:
>   explicit RedPacket(size_t length);
>+  ~RedPacket();
> 
>   void CreateHeader(const uint8_t* rtp_header,
>                     size_t header_length,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_generator_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_generator_unittest.cc
>index b05775443ce..39ead62ca52 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_generator_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_generator_unittest.cc
>@@ -17,7 +17,6 @@
> #include "modules/rtp_rtcp/source/fec_test_helper.h"
> #include "modules/rtp_rtcp/source/forward_error_correction.h"
> #include "modules/rtp_rtcp/source/ulpfec_generator.h"
>-#include "rtc_base/basictypes.h"
> #include "test/gtest.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_header_reader_writer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_header_reader_writer.h
>index ac9fcc171bc..fc83afdf0cc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_header_reader_writer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_header_reader_writer.h
>@@ -12,7 +12,6 @@
> #define MODULES_RTP_RTCP_SOURCE_ULPFEC_HEADER_READER_WRITER_H_
> 
> #include "modules/rtp_rtcp/source/forward_error_correction.h"
>-#include "rtc_base/basictypes.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_header_reader_writer_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_header_reader_writer_unittest.cc
>index 9eaaf89b445..a1d1bd5fb2f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_header_reader_writer_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_header_reader_writer_unittest.cc
>@@ -17,7 +17,6 @@
> #include "modules/rtp_rtcp/source/forward_error_correction.h"
> #include "modules/rtp_rtcp/source/forward_error_correction_internal.h"
> #include "modules/rtp_rtcp/source/ulpfec_header_reader_writer.h"
>-#include "rtc_base/basictypes.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/random.h"
> #include "rtc_base/scoped_ref_ptr.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_receiver_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_receiver_impl.cc
>index a5512e139ff..3afd61228f9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_receiver_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_receiver_impl.cc
>@@ -80,7 +80,11 @@ int32_t UlpfecReceiverImpl::AddReceivedRedPacket(
>         << "Received RED packet with different SSRC than expected; dropping.";
>     return -1;
>   }
>-
>+  if (packet_length > IP_PACKET_SIZE) {
>+    RTC_LOG(LS_WARNING) << "Received RED packet with length exceeds maximum IP "
>+                           "packet size; dropping.";
>+    return -1;
>+  }
>   rtc::CritScope cs(&crit_sect_);
> 
>   uint8_t red_header_length = 1;
>@@ -180,6 +184,7 @@ int32_t UlpfecReceiverImpl::AddReceivedRedPacket(
> 
>   } else if (received_packet->is_fec) {
>     ++packet_counter_.num_fec_packets;
>+
>     // everything behind the RED header
>     memcpy(received_packet->pkt->data,
>            incoming_rtp_packet + header.headerLength + red_header_length,
>@@ -219,7 +224,19 @@ int32_t UlpfecReceiverImpl::AddReceivedRedPacket(
> // TODO(nisse): Drop always-zero return value.
> int32_t UlpfecReceiverImpl::ProcessReceivedFec() {
>   crit_sect_.Enter();
>-  for (const auto& received_packet : received_packets_) {
>+
>+  // If we iterate over |received_packets_| and it contains a packet that cause
>+  // us to recurse back to this function (for example a RED packet encapsulating
>+  // a RED packet), then we will recurse forever. To avoid this we swap
>+  // |received_packets_| with an empty vector so that the next recursive call
>+  // wont iterate over the same packet again. This also solves the problem of
>+  // not modifying the vector we are currently iterating over (packets are added
>+  // in AddReceivedRedPacket).
>+  std::vector<std::unique_ptr<ForwardErrorCorrection::ReceivedPacket>>
>+      received_packets;
>+  received_packets.swap(received_packets_);
>+
>+  for (const auto& received_packet : received_packets) {
>     // Send received media packet to VCM.
>     if (!received_packet->is_fec) {
>       ForwardErrorCorrection::Packet* packet = received_packet->pkt;
>@@ -230,7 +247,6 @@ int32_t UlpfecReceiverImpl::ProcessReceivedFec() {
>     }
>     fec_->DecodeFec(*received_packet, &recovered_packets_);
>   }
>-  received_packets_.clear();
> 
>   // Send any recovered media packets to VCM.
>   for (const auto& recovered_packet : recovered_packets_) {
>@@ -244,10 +260,10 @@ int32_t UlpfecReceiverImpl::ProcessReceivedFec() {
>     // header, OnRecoveredPacket will recurse back here.
>     recovered_packet->returned = true;
>     crit_sect_.Leave();
>-    recovered_packet_callback_->OnRecoveredPacket(packet->data,
>-                                                  packet->length);
>+    recovered_packet_callback_->OnRecoveredPacket(packet->data, packet->length);
>     crit_sect_.Enter();
>   }
>+
>   crit_sect_.Leave();
>   return 0;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_receiver_impl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_receiver_impl.h
>index edc3d31269b..96367dc7aa8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_receiver_impl.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_receiver_impl.h
>@@ -18,14 +18,13 @@
> #include "modules/rtp_rtcp/include/ulpfec_receiver.h"
> #include "modules/rtp_rtcp/source/forward_error_correction.h"
> #include "rtc_base/criticalsection.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
> class UlpfecReceiverImpl : public UlpfecReceiver {
>  public:
>   explicit UlpfecReceiverImpl(uint32_t ssrc, RecoveredPacketReceiver* callback);
>-  virtual ~UlpfecReceiverImpl();
>+  ~UlpfecReceiverImpl() override;
> 
>   int32_t AddReceivedRedPacket(const RTPHeader& rtp_header,
>                                const uint8_t* incoming_rtp_packet,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_receiver_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_receiver_unittest.cc
>index c3bd5a73045..e050f807d0d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_receiver_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/ulpfec_receiver_unittest.cc
>@@ -15,8 +15,8 @@
> 
> #include "modules/rtp_rtcp/include/rtp_header_parser.h"
> #include "modules/rtp_rtcp/include/ulpfec_receiver.h"
>-#include "modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
> #include "modules/rtp_rtcp/mocks/mock_recovered_packet_receiver.h"
>+#include "modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
> #include "modules/rtp_rtcp/source/byte_io.h"
> #include "modules/rtp_rtcp/source/fec_test_helper.h"
> #include "modules/rtp_rtcp/source/forward_error_correction.h"
>@@ -351,8 +351,7 @@ TEST_F(UlpfecReceiverTest, PacketNotDroppedTooEarly) {
>   EncodeFec(media_packets_batch1, kNumFecPacketsBatch1, &fec_packets);
> 
>   BuildAndAddRedMediaPacket(augmented_media_packets_batch1.front());
>-  EXPECT_CALL(recovered_packet_receiver_, OnRecoveredPacket(_, _))
>-      .Times(1);
>+  EXPECT_CALL(recovered_packet_receiver_, OnRecoveredPacket(_, _)).Times(1);
>   EXPECT_EQ(0, receiver_fec_->ProcessReceivedFec());
>   delayed_fec = fec_packets.front();
> 
>@@ -367,15 +366,13 @@ TEST_F(UlpfecReceiverTest, PacketNotDroppedTooEarly) {
>   for (auto it = augmented_media_packets_batch2.begin();
>        it != augmented_media_packets_batch2.end(); ++it) {
>     BuildAndAddRedMediaPacket(*it);
>-    EXPECT_CALL(recovered_packet_receiver_, OnRecoveredPacket(_, _))
>-        .Times(1);
>+    EXPECT_CALL(recovered_packet_receiver_, OnRecoveredPacket(_, _)).Times(1);
>     EXPECT_EQ(0, receiver_fec_->ProcessReceivedFec());
>   }
> 
>   // Add the delayed FEC packet. One packet should be reconstructed.
>   BuildAndAddRedFecPacket(delayed_fec);
>-  EXPECT_CALL(recovered_packet_receiver_, OnRecoveredPacket(_, _))
>-      .Times(1);
>+  EXPECT_CALL(recovered_packet_receiver_, OnRecoveredPacket(_, _)).Times(1);
>   EXPECT_EQ(0, receiver_fec_->ProcessReceivedFec());
> }
> 
>@@ -393,8 +390,7 @@ TEST_F(UlpfecReceiverTest, PacketDroppedWhenTooOld) {
>   EncodeFec(media_packets_batch1, kNumFecPacketsBatch1, &fec_packets);
> 
>   BuildAndAddRedMediaPacket(augmented_media_packets_batch1.front());
>-  EXPECT_CALL(recovered_packet_receiver_, OnRecoveredPacket(_, _))
>-      .Times(1);
>+  EXPECT_CALL(recovered_packet_receiver_, OnRecoveredPacket(_, _)).Times(1);
>   EXPECT_EQ(0, receiver_fec_->ProcessReceivedFec());
>   delayed_fec = fec_packets.front();
> 
>@@ -409,8 +405,7 @@ TEST_F(UlpfecReceiverTest, PacketDroppedWhenTooOld) {
>   for (auto it = augmented_media_packets_batch2.begin();
>        it != augmented_media_packets_batch2.end(); ++it) {
>     BuildAndAddRedMediaPacket(*it);
>-    EXPECT_CALL(recovered_packet_receiver_, OnRecoveredPacket(_, _))
>-        .Times(1);
>+    EXPECT_CALL(recovered_packet_receiver_, OnRecoveredPacket(_, _)).Times(1);
>     EXPECT_EQ(0, receiver_fec_->ProcessReceivedFec());
>   }
> 
>@@ -451,8 +446,7 @@ TEST_F(UlpfecReceiverTest, OldFecPacketDropped) {
>   // and should have been dropped. Only the media packet we inserted will be
>   // returned.
>   BuildAndAddRedMediaPacket(augmented_media_packets.front());
>-  EXPECT_CALL(recovered_packet_receiver_, OnRecoveredPacket(_, _))
>-      .Times(1);
>+  EXPECT_CALL(recovered_packet_receiver_, OnRecoveredPacket(_, _)).Times(1);
>   EXPECT_EQ(0, receiver_fec_->ProcessReceivedFec());
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/video_codec_information.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/video_codec_information.h
>deleted file mode 100644
>index 593d87d01c9..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/source/video_codec_information.h
>+++ /dev/null
>@@ -1,27 +0,0 @@
>-/*
>- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MODULES_RTP_RTCP_SOURCE_VIDEO_CODEC_INFORMATION_H_
>-#define MODULES_RTP_RTCP_SOURCE_VIDEO_CODEC_INFORMATION_H_
>-
>-#include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
>-#include "modules/rtp_rtcp/source/rtp_utility.h"
>-
>-namespace webrtc {
>-class VideoCodecInformation {
>- public:
>-  virtual void Reset() = 0;
>-
>-  virtual RtpVideoCodecTypes Type() = 0;
>-  virtual ~VideoCodecInformation() {}
>-};
>-}  // namespace webrtc
>-
>-#endif  // MODULES_RTP_RTCP_SOURCE_VIDEO_CODEC_INFORMATION_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc
>index e2afa428568..5fa1de07fb0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/test/testAPI/test_api.cc
>@@ -12,6 +12,7 @@
> 
> #include <algorithm>
> #include <memory>
>+#include <string>
> #include <vector>
> 
> #include "rtc_base/checks.h"
>@@ -81,44 +82,34 @@ class RtpRtcpAPITest : public ::testing::Test {
>  protected:
>   RtpRtcpAPITest()
>       : fake_clock_(123456), retransmission_rate_limiter_(&fake_clock_, 1000) {
>-    test_csrcs_.push_back(1234);
>-    test_csrcs_.push_back(2345);
>-    test_ssrc_ = 3456;
>-    test_timestamp_ = 4567;
>-    test_sequence_number_ = 2345;
>   }
>-  ~RtpRtcpAPITest() {}
>-
>-  const uint32_t initial_ssrc = 8888;
>+  ~RtpRtcpAPITest() override = default;
> 
>   void SetUp() override {
>+    const uint32_t kInitialSsrc = 8888;
>     RtpRtcp::Configuration configuration;
>     configuration.audio = true;
>     configuration.clock = &fake_clock_;
>     configuration.outgoing_transport = &null_transport_;
>     configuration.retransmission_rate_limiter = &retransmission_rate_limiter_;
>     module_.reset(RtpRtcp::CreateRtpRtcp(configuration));
>-    module_->SetSSRC(initial_ssrc);
>-    rtp_payload_registry_.reset(new RTPPayloadRegistry());
>+    module_->SetSSRC(kInitialSsrc);
>   }
> 
>-  std::unique_ptr<RTPPayloadRegistry> rtp_payload_registry_;
>   std::unique_ptr<RtpRtcp> module_;
>-  uint32_t test_ssrc_;
>-  uint32_t test_timestamp_;
>-  uint16_t test_sequence_number_;
>-  std::vector<uint32_t> test_csrcs_;
>   SimulatedClock fake_clock_;
>   test::NullTransport null_transport_;
>   RateLimiter retransmission_rate_limiter_;
> };
> 
> TEST_F(RtpRtcpAPITest, Basic) {
>-  module_->SetSequenceNumber(test_sequence_number_);
>-  EXPECT_EQ(test_sequence_number_, module_->SequenceNumber());
>+  const uint16_t kSequenceNumber = 2345;
>+  module_->SetSequenceNumber(kSequenceNumber);
>+  EXPECT_EQ(kSequenceNumber, module_->SequenceNumber());
> 
>-  module_->SetStartTimestamp(test_timestamp_);
>-  EXPECT_EQ(test_timestamp_, module_->StartTimestamp());
>+  const uint32_t kTimestamp = 4567;
>+  module_->SetStartTimestamp(kTimestamp);
>+  EXPECT_EQ(kTimestamp, module_->StartTimestamp());
> 
>   EXPECT_FALSE(module_->Sending());
>   EXPECT_EQ(0, module_->SetSendingStatus(true));
>@@ -131,8 +122,9 @@ TEST_F(RtpRtcpAPITest, PacketSize) {
> }
> 
> TEST_F(RtpRtcpAPITest, SSRC) {
>-  module_->SetSSRC(test_ssrc_);
>-  EXPECT_EQ(test_ssrc_, module_->SSRC());
>+  const uint32_t kSsrc = 3456;
>+  module_->SetSSRC(kSsrc);
>+  EXPECT_EQ(kSsrc, module_->SSRC());
> }
> 
> TEST_F(RtpRtcpAPITest, RTCP) {
>@@ -160,4 +152,54 @@ TEST_F(RtpRtcpAPITest, RtxSender) {
>   EXPECT_EQ(kRtxRetransmitted, module_->RtxSendStatus());
> }
> 
>+TEST_F(RtpRtcpAPITest, LegalMidName) {
>+  static const std::string kLegalMidNames[] = {
>+      // clang-format off
>+      "audio",
>+      "audio0",
>+      "audio_0",
>+      // clang-format on
>+  };
>+  for (const auto& name : kLegalMidNames) {
>+    EXPECT_TRUE(StreamId::IsLegalMidName(name))
>+        << "Mid should be legal: " << name;
>+  }
>+
>+  static const std::string kNonLegalMidNames[] = {
>+      // clang-format off
>+      "",
>+      "(audio0)",
>+      // clang-format on
>+  };
>+  for (const auto& name : kNonLegalMidNames) {
>+    EXPECT_FALSE(StreamId::IsLegalMidName(name))
>+        << "Mid should not be legal: " << name;
>+  }
>+}
>+
>+TEST_F(RtpRtcpAPITest, LegalRsidName) {
>+  static const std::string kLegalRsidNames[] = {
>+      // clang-format off
>+      "audio",
>+      "audio0",
>+      // clang-format on
>+  };
>+  for (const auto& name : kLegalRsidNames) {
>+    EXPECT_TRUE(StreamId::IsLegalRsidName(name))
>+        << "Rsid should be legal: " << name;
>+  }
>+
>+  static const std::string kNonLegalRsidNames[] = {
>+      // clang-format off
>+      "",
>+      "audio_0",
>+      "(audio0)",
>+      // clang-format on
>+  };
>+  for (const auto& name : kNonLegalRsidNames) {
>+    EXPECT_FALSE(StreamId::IsLegalRsidName(name))
>+        << "Rsid should not be legal: " << name;
>+  }
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
>index a668425dcf0..2618433dd5d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
>@@ -25,9 +25,11 @@ namespace webrtc {
> namespace {
> 
> const uint32_t kTestRate = 64000u;
>-const uint8_t kTestPayload[] = { 't', 'e', 's', 't' };
>+const uint8_t kTestPayload[] = {'t', 'e', 's', 't'};
> const uint8_t kPcmuPayloadType = 96;
> const uint8_t kDtmfPayloadType = 97;
>+const uint32_t kSsrc = 3456;
>+const uint32_t kTimestamp = 4567;
> 
> struct CngCodecSpec {
>   int payload_type;
>@@ -39,13 +41,26 @@ const CngCodecSpec kCngCodecs[] = {{13, 8000},
>                                    {104, 32000},
>                                    {105, 48000}};
> 
>-bool IsComfortNoisePayload(uint8_t payload_type) {
>-  for (const auto& c : kCngCodecs) {
>-    if (c.payload_type == payload_type)
>-      return true;
>-  }
>-
>-  return false;
>+// Rough sanity check of DTMF payload.
>+void VerifyDtmf(const uint8_t* payloadData,
>+                size_t payloadSize) {
>+  EXPECT_EQ(payloadSize, 4u);
>+  uint8_t p0 = (payloadSize > 0) ? payloadData[0] : 0xff;
>+  uint8_t p1 = (payloadSize > 1) ? payloadData[1] : 0xff;
>+  uint8_t p2 = (payloadSize > 2) ? payloadData[2] : 0xff;
>+  uint8_t p3 = (payloadSize > 3) ? payloadData[3] : 0xff;
>+  uint8_t event = p0;
>+  bool reserved = (p1 >> 6) & 1;
>+  uint8_t volume = p1 & 63;
>+  uint16_t duration = (p2 << 8) | p3;
>+
>+  // 0-15 are digits, #, *, A-D, 32 is answer tone (see rfc 4734)
>+  EXPECT_LE(event, 32u);
>+  EXPECT_TRUE(event < 16u || event == 32u);
>+  EXPECT_FALSE(reserved);
>+  EXPECT_EQ(volume, 10u);
>+  // Long duration for answer tone events only
>+  EXPECT_TRUE(duration <= 1280 || event == 32u);
> }
> 
> class VerifyingAudioReceiver : public RtpData {
>@@ -55,26 +70,15 @@ class VerifyingAudioReceiver : public RtpData {
>       size_t payloadSize,
>       const webrtc::WebRtcRTPHeader* rtpHeader) override {
>     const uint8_t payload_type = rtpHeader->header.payloadType;
>-    if (payload_type == kPcmuPayloadType || payload_type == kDtmfPayloadType) {
>+    if (payload_type == kPcmuPayloadType) {
>       EXPECT_EQ(sizeof(kTestPayload), payloadSize);
>-      // All our test vectors for PCMU and DTMF are equal to |kTestPayload|.
>+      // All our test vectors for PCMU are equal to |kTestPayload|.
>       const size_t min_size = std::min(sizeof(kTestPayload), payloadSize);
>       EXPECT_EQ(0, memcmp(payloadData, kTestPayload, min_size));
>-    } else if (IsComfortNoisePayload(payload_type)) {
>-      // CNG types should be recognized properly.
>-      EXPECT_EQ(kAudioFrameCN, rtpHeader->frameType);
>-      EXPECT_TRUE(rtpHeader->type.Audio.isCNG);
>+    } else if (payload_type == kDtmfPayloadType) {
>+      VerifyDtmf(payloadData, payloadSize);
>     }
>-    return 0;
>-  }
>-};
> 
>-class RTPCallback : public NullRtpFeedback {
>- public:
>-  int32_t OnInitializeDecoder(int payload_type,
>-                              const SdpAudioFormat& audio_format,
>-                              uint32_t rate) override {
>-    EXPECT_EQ(0u, rate) << "The rate should be zero";
>     return 0;
>   }
> };
>@@ -84,45 +88,34 @@ class RTPCallback : public NullRtpFeedback {
> class RtpRtcpAudioTest : public ::testing::Test {
>  protected:
>   RtpRtcpAudioTest()
>-      : fake_clock(123456), retransmission_rate_limiter_(&fake_clock, 1000) {
>-    test_CSRC[0] = 1234;
>-    test_CSRC[2] = 2345;
>-    test_ssrc = 3456;
>-    test_timestamp = 4567;
>-    test_sequence_number = 2345;
>-  }
>-  ~RtpRtcpAudioTest() {}
>+      : fake_clock_(123456), retransmission_rate_limiter_(&fake_clock_, 1000) {}
>+  ~RtpRtcpAudioTest() override = default;
> 
>   void SetUp() override {
>-    receive_statistics1_.reset(ReceiveStatistics::Create(&fake_clock));
>-    receive_statistics2_.reset(ReceiveStatistics::Create(&fake_clock));
>-
>-    rtp_payload_registry1_.reset(new RTPPayloadRegistry());
>-    rtp_payload_registry2_.reset(new RTPPayloadRegistry());
>+    receive_statistics1_.reset(ReceiveStatistics::Create(&fake_clock_));
>+    receive_statistics2_.reset(ReceiveStatistics::Create(&fake_clock_));
> 
>     RtpRtcp::Configuration configuration;
>     configuration.audio = true;
>-    configuration.clock = &fake_clock;
>+    configuration.clock = &fake_clock_;
>     configuration.receive_statistics = receive_statistics1_.get();
>     configuration.outgoing_transport = &transport1;
>     configuration.retransmission_rate_limiter = &retransmission_rate_limiter_;
> 
>     module1.reset(RtpRtcp::CreateRtpRtcp(configuration));
>     rtp_receiver1_.reset(RtpReceiver::CreateAudioReceiver(
>-        &fake_clock, &data_receiver1, &rtp_callback,
>-        rtp_payload_registry1_.get()));
>+        &fake_clock_, &data_receiver1, &rtp_payload_registry1_));
> 
>     configuration.receive_statistics = receive_statistics2_.get();
>     configuration.outgoing_transport = &transport2;
> 
>     module2.reset(RtpRtcp::CreateRtpRtcp(configuration));
>     rtp_receiver2_.reset(RtpReceiver::CreateAudioReceiver(
>-        &fake_clock, &data_receiver2, &rtp_callback,
>-        rtp_payload_registry2_.get()));
>+        &fake_clock_, &data_receiver2, &rtp_payload_registry2_));
> 
>-    transport1.SetSendModule(module2.get(), rtp_payload_registry2_.get(),
>+    transport1.SetSendModule(module2.get(), &rtp_payload_registry2_,
>                              rtp_receiver2_.get(), receive_statistics2_.get());
>-    transport2.SetSendModule(module1.get(), rtp_payload_registry1_.get(),
>+    transport2.SetSendModule(module1.get(), &rtp_payload_registry1_,
>                              rtp_receiver1_.get(), receive_statistics1_.get());
>   }
> 
>@@ -137,28 +130,23 @@ class RtpRtcpAudioTest : public ::testing::Test {
> 
>   VerifyingAudioReceiver data_receiver1;
>   VerifyingAudioReceiver data_receiver2;
>-  RTPCallback rtp_callback;
>   std::unique_ptr<ReceiveStatistics> receive_statistics1_;
>   std::unique_ptr<ReceiveStatistics> receive_statistics2_;
>-  std::unique_ptr<RTPPayloadRegistry> rtp_payload_registry1_;
>-  std::unique_ptr<RTPPayloadRegistry> rtp_payload_registry2_;
>+  RTPPayloadRegistry rtp_payload_registry1_;
>+  RTPPayloadRegistry rtp_payload_registry2_;
>   std::unique_ptr<RtpReceiver> rtp_receiver1_;
>   std::unique_ptr<RtpReceiver> rtp_receiver2_;
>   std::unique_ptr<RtpRtcp> module1;
>   std::unique_ptr<RtpRtcp> module2;
>   LoopBackTransport transport1;
>   LoopBackTransport transport2;
>-  uint32_t test_ssrc;
>-  uint32_t test_timestamp;
>-  uint16_t test_sequence_number;
>-  uint32_t test_CSRC[webrtc::kRtpCsrcSize];
>-  SimulatedClock fake_clock;
>+  SimulatedClock fake_clock_;
>   RateLimiter retransmission_rate_limiter_;
> };
> 
> TEST_F(RtpRtcpAudioTest, Basic) {
>-  module1->SetSSRC(test_ssrc);
>-  module1->SetStartTimestamp(test_timestamp);
>+  module1->SetSSRC(kSsrc);
>+  module1->SetStartTimestamp(kTimestamp);
> 
>   // Test detection at the end of a DTMF tone.
>   // EXPECT_EQ(0, module2->SetTelephoneEventForwardToDecoder(true));
>@@ -184,13 +172,13 @@ TEST_F(RtpRtcpAudioTest, Basic) {
>                                         kPcmuPayloadType, 0, -1, kTestPayload,
>                                         4, nullptr, nullptr, nullptr));
> 
>-  EXPECT_EQ(test_ssrc, rtp_receiver2_->SSRC());
>+  EXPECT_EQ(kSsrc, rtp_receiver2_->SSRC());
>   uint32_t timestamp;
>   int64_t receive_time_ms;
>   EXPECT_TRUE(
>       rtp_receiver2_->GetLatestTimestamps(&timestamp, &receive_time_ms));
>-  EXPECT_EQ(test_timestamp, timestamp);
>-  EXPECT_EQ(fake_clock.TimeInMilliseconds(), receive_time_ms);
>+  EXPECT_EQ(kTimestamp, timestamp);
>+  EXPECT_EQ(fake_clock_.TimeInMilliseconds(), receive_time_ms);
> }
> 
> TEST_F(RtpRtcpAudioTest, DTMF) {
>@@ -201,8 +189,8 @@ TEST_F(RtpRtcpAudioTest, DTMF) {
>   memcpy(voice_codec.plname, "PCMU", 5);
>   RegisterPayload(voice_codec);
> 
>-  module1->SetSSRC(test_ssrc);
>-  module1->SetStartTimestamp(test_timestamp);
>+  module1->SetSSRC(kSsrc);
>+  module1->SetStartTimestamp(kTimestamp);
>   EXPECT_EQ(0, module1->SetSendingStatus(true));
> 
>   // Prepare for DTMF.
>@@ -229,7 +217,7 @@ TEST_F(RtpRtcpAudioTest, DTMF) {
>     EXPECT_TRUE(module1->SendOutgoingData(
>         webrtc::kAudioFrameSpeech, kPcmuPayloadType, timeStamp, -1,
>         kTestPayload, 4, nullptr, nullptr, nullptr));
>-    fake_clock.AdvanceTimeMilliseconds(20);
>+    fake_clock_.AdvanceTimeMilliseconds(20);
>     module1->Process();
>   }
>   EXPECT_EQ(0, module1->SendTelephoneEventOutband(32, 9000, 10));
>@@ -238,14 +226,14 @@ TEST_F(RtpRtcpAudioTest, DTMF) {
>     EXPECT_TRUE(module1->SendOutgoingData(
>         webrtc::kAudioFrameSpeech, kPcmuPayloadType, timeStamp, -1,
>         kTestPayload, 4, nullptr, nullptr, nullptr));
>-    fake_clock.AdvanceTimeMilliseconds(20);
>+    fake_clock_.AdvanceTimeMilliseconds(20);
>     module1->Process();
>   }
> }
> 
> TEST_F(RtpRtcpAudioTest, ComfortNoise) {
>-  module1->SetSSRC(test_ssrc);
>-  module1->SetStartTimestamp(test_timestamp);
>+  module1->SetSSRC(kSsrc);
>+  module1->SetStartTimestamp(kTimestamp);
> 
>   EXPECT_EQ(0, module1->SetSendingStatus(true));
> 
>@@ -274,25 +262,25 @@ TEST_F(RtpRtcpAudioTest, ComfortNoise) {
>         webrtc::kAudioFrameSpeech, kPcmuPayloadType, in_timestamp, -1,
>         kTestPayload, 4, nullptr, nullptr, nullptr));
> 
>-    EXPECT_EQ(test_ssrc, rtp_receiver2_->SSRC());
>+    EXPECT_EQ(kSsrc, rtp_receiver2_->SSRC());
>     EXPECT_TRUE(
>         rtp_receiver2_->GetLatestTimestamps(&timestamp, &receive_time_ms));
>-    EXPECT_EQ(test_timestamp + in_timestamp, timestamp);
>-    EXPECT_EQ(fake_clock.TimeInMilliseconds(), receive_time_ms);
>+    EXPECT_EQ(kTimestamp + in_timestamp, timestamp);
>+    EXPECT_EQ(fake_clock_.TimeInMilliseconds(), receive_time_ms);
>     in_timestamp += 10;
>-    fake_clock.AdvanceTimeMilliseconds(20);
>+    fake_clock_.AdvanceTimeMilliseconds(20);
> 
>     EXPECT_TRUE(module1->SendOutgoingData(webrtc::kAudioFrameCN, c.payload_type,
>                                           in_timestamp, -1, kTestPayload, 1,
>                                           nullptr, nullptr, nullptr));
> 
>-    EXPECT_EQ(test_ssrc, rtp_receiver2_->SSRC());
>+    EXPECT_EQ(kSsrc, rtp_receiver2_->SSRC());
>     EXPECT_TRUE(
>         rtp_receiver2_->GetLatestTimestamps(&timestamp, &receive_time_ms));
>-    EXPECT_EQ(test_timestamp + in_timestamp, timestamp);
>-    EXPECT_EQ(fake_clock.TimeInMilliseconds(), receive_time_ms);
>+    EXPECT_EQ(kTimestamp + in_timestamp, timestamp);
>+    EXPECT_EQ(fake_clock_.TimeInMilliseconds(), receive_time_ms);
>     in_timestamp += 10;
>-    fake_clock.AdvanceTimeMilliseconds(20);
>+    fake_clock_.AdvanceTimeMilliseconds(20);
>   }
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc
>index 4b5b50b8b61..7551e71c18f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc
>@@ -25,104 +25,66 @@
> 
> namespace webrtc {
> namespace {
>+const uint16_t kSequenceNumber = 2345;
>+const uint32_t kSsrc = 3456;
>+const uint32_t kTimestamp = 4567;
> 
> class RtcpCallback : public RtcpIntraFrameObserver {
>  public:
>-  void SetModule(RtpRtcp* module) {
>-    _rtpRtcpModule = module;
>-  }
>-  virtual void OnRTCPPacketTimeout(const int32_t id) {
>-  }
>-  virtual void OnLipSyncUpdate(const int32_t id,
>-                               const int32_t audioVideoOffset) {}
>-  virtual void OnReceivedIntraFrameRequest(uint32_t ssrc) {}
>-
>- private:
>-  RtpRtcp* _rtpRtcpModule;
>-};
>-
>-class TestRtpFeedback : public NullRtpFeedback {
>- public:
>-  explicit TestRtpFeedback(RtpRtcp* rtp_rtcp) : rtp_rtcp_(rtp_rtcp) {}
>-  virtual ~TestRtpFeedback() {}
>-
>-  void OnIncomingSSRCChanged(uint32_t ssrc) override {
>-    rtp_rtcp_->SetRemoteSSRC(ssrc);
>-  }
>-
>- private:
>-  RtpRtcp* rtp_rtcp_;
>+  void OnReceivedIntraFrameRequest(uint32_t ssrc) override {}
> };
> 
> class RtpRtcpRtcpTest : public ::testing::Test {
>  protected:
>   RtpRtcpRtcpTest()
>-      : fake_clock(123456), retransmission_rate_limiter_(&fake_clock, 1000) {
>-    test_csrcs.push_back(1234);
>-    test_csrcs.push_back(2345);
>-    test_ssrc = 3456;
>-    test_timestamp = 4567;
>-    test_sequence_number = 2345;
>-  }
>-  ~RtpRtcpRtcpTest() {}
>+      : fake_clock_(123456), retransmission_rate_limiter_(&fake_clock_, 1000) {}
>+  ~RtpRtcpRtcpTest() override = default;
> 
>-  virtual void SetUp() {
>+  void SetUp() override {
>     receiver = new TestRtpReceiver();
>     transport1 = new LoopBackTransport();
>     transport2 = new LoopBackTransport();
>-    myRTCPFeedback1 = new RtcpCallback();
>-    myRTCPFeedback2 = new RtcpCallback();
> 
>-    receive_statistics1_.reset(ReceiveStatistics::Create(&fake_clock));
>-    receive_statistics2_.reset(ReceiveStatistics::Create(&fake_clock));
>+    receive_statistics1_.reset(ReceiveStatistics::Create(&fake_clock_));
>+    receive_statistics2_.reset(ReceiveStatistics::Create(&fake_clock_));
> 
>     RtpRtcp::Configuration configuration;
>     configuration.audio = true;
>-    configuration.clock = &fake_clock;
>+    configuration.clock = &fake_clock_;
>     configuration.receive_statistics = receive_statistics1_.get();
>     configuration.outgoing_transport = transport1;
>-    configuration.intra_frame_callback = myRTCPFeedback1;
>+    configuration.intra_frame_callback = &rtcp_callback1_;
>     configuration.retransmission_rate_limiter = &retransmission_rate_limiter_;
> 
>-    rtp_payload_registry1_.reset(new RTPPayloadRegistry());
>-    rtp_payload_registry2_.reset(new RTPPayloadRegistry());
>-
>     module1 = RtpRtcp::CreateRtpRtcp(configuration);
> 
>-    rtp_feedback1_.reset(new TestRtpFeedback(module1));
>-
>     rtp_receiver1_.reset(RtpReceiver::CreateAudioReceiver(
>-        &fake_clock, receiver, rtp_feedback1_.get(),
>-        rtp_payload_registry1_.get()));
>+        &fake_clock_, receiver, &rtp_payload_registry1_));
> 
>     configuration.receive_statistics = receive_statistics2_.get();
>     configuration.outgoing_transport = transport2;
>-    configuration.intra_frame_callback = myRTCPFeedback2;
>+    configuration.intra_frame_callback = &rtcp_callback2_;
> 
>     module2 = RtpRtcp::CreateRtpRtcp(configuration);
> 
>-    rtp_feedback2_.reset(new TestRtpFeedback(module2));
>-
>     rtp_receiver2_.reset(RtpReceiver::CreateAudioReceiver(
>-        &fake_clock, receiver, rtp_feedback2_.get(),
>-        rtp_payload_registry2_.get()));
>+        &fake_clock_, receiver, &rtp_payload_registry2_));
> 
>-    transport1->SetSendModule(module2, rtp_payload_registry2_.get(),
>+    transport1->SetSendModule(module2, &rtp_payload_registry2_,
>                               rtp_receiver2_.get(), receive_statistics2_.get());
>-    transport2->SetSendModule(module1, rtp_payload_registry1_.get(),
>+    transport2->SetSendModule(module1, &rtp_payload_registry1_,
>                               rtp_receiver1_.get(), receive_statistics1_.get());
>-    myRTCPFeedback1->SetModule(module1);
>-    myRTCPFeedback2->SetModule(module2);
> 
>     module1->SetRTCPStatus(RtcpMode::kCompound);
>     module2->SetRTCPStatus(RtcpMode::kCompound);
> 
>-    module2->SetSSRC(test_ssrc + 1);
>-    module1->SetSSRC(test_ssrc);
>-    module1->SetSequenceNumber(test_sequence_number);
>-    module1->SetStartTimestamp(test_timestamp);
>+    module2->SetSSRC(kSsrc + 1);
>+    module2->SetRemoteSSRC(kSsrc);
>+    module1->SetSSRC(kSsrc);
>+    module1->SetSequenceNumber(kSequenceNumber);
>+    module1->SetStartTimestamp(kTimestamp);
> 
>-    module1->SetCsrcs(test_csrcs);
>+    module1->SetCsrcs(kCsrcs);
>     EXPECT_EQ(0, module1->SetCNAME("john.doe@test.test"));
> 
>     EXPECT_EQ(0, module1->SetSendingStatus(true));
>@@ -142,29 +104,27 @@ class RtpRtcpRtcpTest : public ::testing::Test {
> 
>     // We need to send one RTP packet to get the RTCP packet to be accepted by
>     // the receiving module.
>-    // send RTP packet with the data "testtest"
>+    // Send RTP packet with the data "testtest".
>     const uint8_t test[9] = "testtest";
>     EXPECT_EQ(true,
>               module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96, 0, -1,
>                                         test, 8, nullptr, nullptr, nullptr));
>   }
> 
>-  virtual void TearDown() {
>+  void TearDown() override {
>     delete module1;
>     delete module2;
>-    delete myRTCPFeedback1;
>-    delete myRTCPFeedback2;
>     delete transport1;
>     delete transport2;
>     delete receiver;
>   }
> 
>-  std::unique_ptr<TestRtpFeedback> rtp_feedback1_;
>-  std::unique_ptr<TestRtpFeedback> rtp_feedback2_;
>+  RtcpCallback rtcp_callback1_;
>+  RtcpCallback rtcp_callback2_;
>+  RTPPayloadRegistry rtp_payload_registry1_;
>+  RTPPayloadRegistry rtp_payload_registry2_;
>   std::unique_ptr<ReceiveStatistics> receive_statistics1_;
>   std::unique_ptr<ReceiveStatistics> receive_statistics2_;
>-  std::unique_ptr<RTPPayloadRegistry> rtp_payload_registry1_;
>-  std::unique_ptr<RTPPayloadRegistry> rtp_payload_registry2_;
>   std::unique_ptr<RtpReceiver> rtp_receiver1_;
>   std::unique_ptr<RtpReceiver> rtp_receiver2_;
>   RtpRtcp* module1;
>@@ -172,35 +132,25 @@ class RtpRtcpRtcpTest : public ::testing::Test {
>   TestRtpReceiver* receiver;
>   LoopBackTransport* transport1;
>   LoopBackTransport* transport2;
>-  RtcpCallback* myRTCPFeedback1;
>-  RtcpCallback* myRTCPFeedback2;
>-
>-  uint32_t test_ssrc;
>-  uint32_t test_timestamp;
>-  uint16_t test_sequence_number;
>-  std::vector<uint32_t> test_csrcs;
>-  SimulatedClock fake_clock;
>+
>+  const std::vector<uint32_t> kCsrcs = {1234, 2345};
>+  SimulatedClock fake_clock_;
>   RateLimiter retransmission_rate_limiter_;
> };
> 
> TEST_F(RtpRtcpRtcpTest, RTCP_CNAME) {
>-  uint32_t testOfCSRC[webrtc::kRtpCsrcSize];
>-  EXPECT_EQ(2, rtp_receiver2_->CSRCs(testOfCSRC));
>-  EXPECT_EQ(test_csrcs[0], testOfCSRC[0]);
>-  EXPECT_EQ(test_csrcs[1], testOfCSRC[1]);
>-
>   // Set cname of mixed.
>-  EXPECT_EQ(0, module1->AddMixedCNAME(test_csrcs[0], "john@192.168.0.1"));
>-  EXPECT_EQ(0, module1->AddMixedCNAME(test_csrcs[1], "jane@192.168.0.2"));
>+  EXPECT_EQ(0, module1->AddMixedCNAME(kCsrcs[0], "john@192.168.0.1"));
>+  EXPECT_EQ(0, module1->AddMixedCNAME(kCsrcs[1], "jane@192.168.0.2"));
> 
>-  EXPECT_EQ(-1, module1->RemoveMixedCNAME(test_csrcs[0] + 1));
>-  EXPECT_EQ(0, module1->RemoveMixedCNAME(test_csrcs[1]));
>-  EXPECT_EQ(0, module1->AddMixedCNAME(test_csrcs[1], "jane@192.168.0.2"));
>+  EXPECT_EQ(-1, module1->RemoveMixedCNAME(kCsrcs[0] + 1));
>+  EXPECT_EQ(0, module1->RemoveMixedCNAME(kCsrcs[1]));
>+  EXPECT_EQ(0, module1->AddMixedCNAME(kCsrcs[1], "jane@192.168.0.2"));
> 
>-  // send RTCP packet, triggered by timer
>-  fake_clock.AdvanceTimeMilliseconds(7500);
>+  // Send RTCP packet, triggered by timer.
>+  fake_clock_.AdvanceTimeMilliseconds(7500);
>   module1->Process();
>-  fake_clock.AdvanceTimeMilliseconds(100);
>+  fake_clock_.AdvanceTimeMilliseconds(100);
>   module2->Process();
> 
>   char cName[RTCP_CNAME_SIZE];
>@@ -210,15 +160,15 @@ TEST_F(RtpRtcpRtcpTest, RTCP_CNAME) {
>   EXPECT_EQ(0, module2->RemoteCNAME(rtp_receiver2_->SSRC(), cName));
>   EXPECT_EQ(0, strncmp(cName, "john.doe@test.test", RTCP_CNAME_SIZE));
> 
>-  EXPECT_EQ(0, module2->RemoteCNAME(test_csrcs[0], cName));
>+  EXPECT_EQ(0, module2->RemoteCNAME(kCsrcs[0], cName));
>   EXPECT_EQ(0, strncmp(cName, "john@192.168.0.1", RTCP_CNAME_SIZE));
> 
>-  EXPECT_EQ(0, module2->RemoteCNAME(test_csrcs[1], cName));
>+  EXPECT_EQ(0, module2->RemoteCNAME(kCsrcs[1], cName));
>   EXPECT_EQ(0, strncmp(cName, "jane@192.168.0.2", RTCP_CNAME_SIZE));
> 
>   EXPECT_EQ(0, module1->SetSendingStatus(false));
> 
>-  // Test that BYE clears the CNAME
>+  // Test that BYE clears the CNAME.
>   EXPECT_EQ(-1, module2->RemoteCNAME(rtp_receiver2_->SSRC(), cName));
> }
> 
>@@ -228,23 +178,22 @@ TEST_F(RtpRtcpRtcpTest, RemoteRTCPStatRemote) {
>   EXPECT_EQ(0, module1->RemoteRTCPStat(&report_blocks));
>   EXPECT_EQ(0u, report_blocks.size());
> 
>-  // send RTCP packet, triggered by timer
>-  fake_clock.AdvanceTimeMilliseconds(7500);
>+  // Send RTCP packet, triggered by timer.
>+  fake_clock_.AdvanceTimeMilliseconds(7500);
>   module1->Process();
>-  fake_clock.AdvanceTimeMilliseconds(100);
>+  fake_clock_.AdvanceTimeMilliseconds(100);
>   module2->Process();
> 
>   EXPECT_EQ(0, module1->RemoteRTCPStat(&report_blocks));
>   ASSERT_EQ(1u, report_blocks.size());
> 
>-  // |test_ssrc+1| is the SSRC of module2 that send the report.
>-  EXPECT_EQ(test_ssrc + 1, report_blocks[0].sender_ssrc);
>-  EXPECT_EQ(test_ssrc, report_blocks[0].source_ssrc);
>+  // |kSsrc+1| is the SSRC of module2 that send the report.
>+  EXPECT_EQ(kSsrc + 1, report_blocks[0].sender_ssrc);
>+  EXPECT_EQ(kSsrc, report_blocks[0].source_ssrc);
> 
>-  EXPECT_EQ(0u, report_blocks[0].packets_lost);
>+  EXPECT_EQ(0, report_blocks[0].packets_lost);
>   EXPECT_LT(0u, report_blocks[0].delay_since_last_sender_report);
>-  EXPECT_EQ(test_sequence_number,
>-            report_blocks[0].extended_highest_sequence_number);
>+  EXPECT_EQ(kSequenceNumber, report_blocks[0].extended_highest_sequence_number);
>   EXPECT_EQ(0u, report_blocks[0].fraction_lost);
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc
>index 63b78514dc0..4d8190bdad9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/test/testAPI/test_api_video.cc
>@@ -12,9 +12,8 @@
> 
> #include <algorithm>
> #include <memory>
>-#include <vector>
> 
>-#include "common_types.h"  // NOLINT(build/include)
>+#include "api/video_codecs/video_codec.h"
> #include "modules/rtp_rtcp/include/rtp_payload_registry.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>@@ -25,9 +24,8 @@
> #include "test/gtest.h"
> 
> namespace {
>-
>+const uint32_t kSsrc = 3456;
> const unsigned char kPayloadType = 100;
>-
> };
> 
> namespace webrtc {
>@@ -35,29 +33,25 @@ namespace webrtc {
> class RtpRtcpVideoTest : public ::testing::Test {
>  protected:
>   RtpRtcpVideoTest()
>-      : test_ssrc_(3456),
>-        test_timestamp_(4567),
>-        test_sequence_number_(2345),
>-        fake_clock(123456),
>-        retransmission_rate_limiter_(&fake_clock, 1000) {}
>-  ~RtpRtcpVideoTest() {}
>-
>-  virtual void SetUp() {
>+      : fake_clock_(123456), retransmission_rate_limiter_(&fake_clock_, 1000) {}
>+  ~RtpRtcpVideoTest() override = default;
>+
>+  void SetUp() override {
>     transport_ = new LoopBackTransport();
>     receiver_ = new TestRtpReceiver();
>-    receive_statistics_.reset(ReceiveStatistics::Create(&fake_clock));
>+    receive_statistics_.reset(ReceiveStatistics::Create(&fake_clock_));
>     RtpRtcp::Configuration configuration;
>     configuration.audio = false;
>-    configuration.clock = &fake_clock;
>+    configuration.clock = &fake_clock_;
>     configuration.outgoing_transport = transport_;
>     configuration.retransmission_rate_limiter = &retransmission_rate_limiter_;
> 
>     video_module_ = RtpRtcp::CreateRtpRtcp(configuration);
>     rtp_receiver_.reset(RtpReceiver::CreateVideoReceiver(
>-        &fake_clock, receiver_, NULL, &rtp_payload_registry_));
>+        &fake_clock_, receiver_, &rtp_payload_registry_));
> 
>     video_module_->SetRTCPStatus(RtcpMode::kCompound);
>-    video_module_->SetSSRC(test_ssrc_);
>+    video_module_->SetSSRC(kSsrc);
>     video_module_->SetStorePacketsStatus(true, 600);
>     EXPECT_EQ(0, video_module_->SetSendingStatus(true));
> 
>@@ -67,26 +61,26 @@ class RtpRtcpVideoTest : public ::testing::Test {
>     VideoCodec video_codec;
>     memset(&video_codec, 0, sizeof(video_codec));
>     video_codec.plType = 123;
>-    memcpy(video_codec.plName, "I420", 5);
>+    video_codec.codecType = kVideoCodecI420;
> 
>-    EXPECT_EQ(0, video_module_->RegisterSendPayload(video_codec));
>+    video_module_->RegisterVideoSendPayload(123, "I420");
>     EXPECT_EQ(0, rtp_payload_registry_.RegisterReceivePayload(video_codec));
> 
>     payload_data_length_ = sizeof(video_frame_);
> 
>     for (size_t n = 0; n < payload_data_length_; n++) {
>-      video_frame_[n] = n%10;
>+      video_frame_[n] = n % 10;
>     }
>   }
> 
>-  size_t BuildRTPheader(uint8_t* dataBuffer,
>-                         uint32_t timestamp,
>-                         uint32_t sequence_number) {
>-    dataBuffer[0] = static_cast<uint8_t>(0x80);  // version 2
>-    dataBuffer[1] = static_cast<uint8_t>(kPayloadType);
>-    ByteWriter<uint16_t>::WriteBigEndian(dataBuffer + 2, sequence_number);
>-    ByteWriter<uint32_t>::WriteBigEndian(dataBuffer + 4, timestamp);
>-    ByteWriter<uint32_t>::WriteBigEndian(dataBuffer + 8, 0x1234);  // SSRC.
>+  size_t BuildRTPheader(uint8_t* buffer,
>+                        uint32_t timestamp,
>+                        uint32_t sequence_number) {
>+    buffer[0] = static_cast<uint8_t>(0x80);  // version 2
>+    buffer[1] = static_cast<uint8_t>(kPayloadType);
>+    ByteWriter<uint16_t>::WriteBigEndian(buffer + 2, sequence_number);
>+    ByteWriter<uint32_t>::WriteBigEndian(buffer + 4, timestamp);
>+    ByteWriter<uint32_t>::WriteBigEndian(buffer + 8, 0x1234);  // SSRC.
>     size_t rtpHeaderLength = 12;
>     return rtpHeaderLength;
>   }
>@@ -105,8 +99,7 @@ class RtpRtcpVideoTest : public ::testing::Test {
>     // Correct seq num, timestamp and payload type.
>     size_t header_length = BuildRTPheader(buffer, timestamp, sequence_number);
>     buffer[0] |= 0x20;  // Set padding bit.
>-    int32_t* data =
>-        reinterpret_cast<int32_t*>(&(buffer[header_length]));
>+    int32_t* data = reinterpret_cast<int32_t*>(&(buffer[header_length]));
> 
>     // Fill data buffer with random data.
>     for (size_t j = 0; j < (padding_bytes_in_packet >> 2); j++) {
>@@ -118,33 +111,30 @@ class RtpRtcpVideoTest : public ::testing::Test {
>     return padding_bytes_in_packet + header_length;
>   }
> 
>-  virtual void TearDown() {
>+  void TearDown() override {
>     delete video_module_;
>     delete transport_;
>     delete receiver_;
>   }
> 
>-  int test_id_;
>   std::unique_ptr<ReceiveStatistics> receive_statistics_;
>   RTPPayloadRegistry rtp_payload_registry_;
>   std::unique_ptr<RtpReceiver> rtp_receiver_;
>   RtpRtcp* video_module_;
>   LoopBackTransport* transport_;
>   TestRtpReceiver* receiver_;
>-  uint32_t test_ssrc_;
>-  uint32_t test_timestamp_;
>-  uint16_t test_sequence_number_;
>-  uint8_t  video_frame_[65000];
>+  uint8_t video_frame_[65000];
>   size_t payload_data_length_;
>-  SimulatedClock fake_clock;
>+  SimulatedClock fake_clock_;
>   RateLimiter retransmission_rate_limiter_;
> };
> 
> TEST_F(RtpRtcpVideoTest, BasicVideo) {
>   uint32_t timestamp = 3000;
>+  RTPVideoHeader video_header;
>   EXPECT_TRUE(video_module_->SendOutgoingData(
>       kVideoFrameDelta, 123, timestamp, timestamp / 90, video_frame_,
>-      payload_data_length_, nullptr, nullptr, nullptr));
>+      payload_data_length_, nullptr, &video_header, nullptr));
> }
> 
> TEST_F(RtpRtcpVideoTest, PaddingOnlyFrames) {
>@@ -155,12 +145,11 @@ TEST_F(RtpRtcpVideoTest, PaddingOnlyFrames) {
>   VideoCodec codec;
>   codec.codecType = kVideoCodecVP8;
>   codec.plType = kPayloadType;
>-  strncpy(codec.plName, "VP8", 4);
>   EXPECT_EQ(0, rtp_payload_registry_.RegisterReceivePayload(codec));
>   for (int frame_idx = 0; frame_idx < 10; ++frame_idx) {
>     for (int packet_idx = 0; packet_idx < 5; ++packet_idx) {
>-      size_t packet_size = PaddingPacket(padding_packet, timestamp, seq_num,
>-                                         kPadSize);
>+      size_t packet_size =
>+          PaddingPacket(padding_packet, timestamp, seq_num, kPadSize);
>       ++seq_num;
>       RTPHeader header;
>       std::unique_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
>@@ -176,7 +165,7 @@ TEST_F(RtpRtcpVideoTest, PaddingOnlyFrames) {
>       EXPECT_EQ(payload_length, receiver_->rtp_header().header.paddingLength);
>     }
>     timestamp += 3000;
>-    fake_clock.AdvanceTimeMilliseconds(33);
>+    fake_clock_.AdvanceTimeMilliseconds(33);
>   }
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/test/testFec/average_residual_loss_xor_codes.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/test/testFec/average_residual_loss_xor_codes.h
>index ccf9752ee9f..0bc2f569179 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/test/testFec/average_residual_loss_xor_codes.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/test/testFec/average_residual_loss_xor_codes.h
>@@ -23,169 +23,35 @@ const int kNumberCodes = kMaxMediaPacketsTest * (kMaxMediaPacketsTest + 1) / 2;
> // loss expected for each code size up to:
> // (kMaxMediaPacketsTest, kMaxMediaPacketsTest).
> const float kMaxResidualLossRandomMask[kNumberCodes] = {
>-  0.009463f,
>-  0.022436f,
>-  0.007376f,
>-  0.033895f,
>-  0.012423f,
>-  0.004644f,
>-  0.043438f,
>-  0.019937f,
>-  0.008820f,
>-  0.003438f,
>-  0.051282f,
>-  0.025795f,
>-  0.012872f,
>-  0.006458f,
>-  0.003195f,
>-  0.057728f,
>-  0.032146f,
>-  0.016708f,
>-  0.009242f,
>-  0.005054f,
>-  0.003078f,
>-  0.063050f,
>-  0.037261f,
>-  0.021767f,
>-  0.012447f,
>-  0.007099f,
>-  0.003826f,
>-  0.002504f,
>-  0.067476f,
>-  0.042348f,
>-  0.026169f,
>-  0.015695f,
>-  0.009478f,
>-  0.005887f,
>-  0.003568f,
>-  0.001689f,
>-  0.071187f,
>-  0.046575f,
>-  0.031697f,
>-  0.019797f,
>-  0.012433f,
>-  0.007027f,
>-  0.004845f,
>-  0.002777f,
>-  0.001753f,
>-  0.074326f,
>-  0.050628f,
>-  0.034978f,
>-  0.021955f,
>-  0.014821f,
>-  0.009462f,
>-  0.006393f,
>-  0.004181f,
>-  0.003105f,
>-  0.001231f,
>-  0.077008f,
>-  0.054226f,
>-  0.038407f,
>-  0.026251f,
>-  0.018634f,
>-  0.011568f,
>-  0.008130f,
>-  0.004957f,
>-  0.003334f,
>-  0.002069f,
>-  0.001304f,
>-  0.079318f,
>-  0.057180f,
>-  0.041268f,
>-  0.028842f,
>-  0.020033f,
>-  0.014061f,
>-  0.009636f,
>-  0.006411f,
>-  0.004583f,
>-  0.002817f,
>-  0.001770f,
>-  0.001258f
>-};
>+    0.009463f, 0.022436f, 0.007376f, 0.033895f, 0.012423f, 0.004644f, 0.043438f,
>+    0.019937f, 0.008820f, 0.003438f, 0.051282f, 0.025795f, 0.012872f, 0.006458f,
>+    0.003195f, 0.057728f, 0.032146f, 0.016708f, 0.009242f, 0.005054f, 0.003078f,
>+    0.063050f, 0.037261f, 0.021767f, 0.012447f, 0.007099f, 0.003826f, 0.002504f,
>+    0.067476f, 0.042348f, 0.026169f, 0.015695f, 0.009478f, 0.005887f, 0.003568f,
>+    0.001689f, 0.071187f, 0.046575f, 0.031697f, 0.019797f, 0.012433f, 0.007027f,
>+    0.004845f, 0.002777f, 0.001753f, 0.074326f, 0.050628f, 0.034978f, 0.021955f,
>+    0.014821f, 0.009462f, 0.006393f, 0.004181f, 0.003105f, 0.001231f, 0.077008f,
>+    0.054226f, 0.038407f, 0.026251f, 0.018634f, 0.011568f, 0.008130f, 0.004957f,
>+    0.003334f, 0.002069f, 0.001304f, 0.079318f, 0.057180f, 0.041268f, 0.028842f,
>+    0.020033f, 0.014061f, 0.009636f, 0.006411f, 0.004583f, 0.002817f, 0.001770f,
>+    0.001258f};
> 
> // For the bursty mask type: reference level for the maximum average residual
> // loss expected for each code size up to:
> // (kMaxMediaPacketsTestf, kMaxMediaPacketsTest).
> const float kMaxResidualLossBurstyMask[kNumberCodes] = {
>-  0.033236f,
>-  0.053344f,
>-  0.026616f,
>-  0.064129f,
>-  0.036589f,
>-  0.021892f,
>-  0.071055f,
>-  0.043890f,
>-  0.028009f,
>-  0.018524f,
>-  0.075968f,
>-  0.049828f,
>-  0.033288f,
>-  0.022791f,
>-  0.016088f,
>-  0.079672f,
>-  0.054586f,
>-  0.037872f,
>-  0.026679f,
>-  0.019326f,
>-  0.014293f,
>-  0.082582f,
>-  0.058719f,
>-  0.042045f,
>-  0.030504f,
>-  0.022391f,
>-  0.016894f,
>-  0.012946f,
>-  0.084935f,
>-  0.062169f,
>-  0.045620f,
>-  0.033713f,
>-  0.025570f,
>-  0.019439f,
>-  0.015121f,
>-  0.011920f,
>-  0.086881f,
>-  0.065267f,
>-  0.048721f,
>-  0.037613f,
>-  0.028278f,
>-  0.022152f,
>-  0.017314f,
>-  0.013791f,
>-  0.011130f,
>-  0.088516f,
>-  0.067911f,
>-  0.051709f,
>-  0.040819f,
>-  0.030777f,
>-  0.024547f,
>-  0.019689f,
>-  0.015877f,
>-  0.012773f,
>-  0.010516f,
>-  0.089909f,
>-  0.070332f,
>-  0.054402f,
>-  0.043210f,
>-  0.034096f,
>-  0.026625f,
>-  0.021823f,
>-  0.017648f,
>-  0.014649f,
>-  0.011982f,
>-  0.010035f,
>-  0.091109f,
>-  0.072428f,
>-  0.056775f,
>-  0.045418f,
>-  0.036679f,
>-  0.028599f,
>-  0.023693f,
>-  0.019966f,
>-  0.016603f,
>-  0.013690f,
>-  0.011359f,
>-  0.009657f
>-};
>+    0.033236f, 0.053344f, 0.026616f, 0.064129f, 0.036589f, 0.021892f, 0.071055f,
>+    0.043890f, 0.028009f, 0.018524f, 0.075968f, 0.049828f, 0.033288f, 0.022791f,
>+    0.016088f, 0.079672f, 0.054586f, 0.037872f, 0.026679f, 0.019326f, 0.014293f,
>+    0.082582f, 0.058719f, 0.042045f, 0.030504f, 0.022391f, 0.016894f, 0.012946f,
>+    0.084935f, 0.062169f, 0.045620f, 0.033713f, 0.025570f, 0.019439f, 0.015121f,
>+    0.011920f, 0.086881f, 0.065267f, 0.048721f, 0.037613f, 0.028278f, 0.022152f,
>+    0.017314f, 0.013791f, 0.011130f, 0.088516f, 0.067911f, 0.051709f, 0.040819f,
>+    0.030777f, 0.024547f, 0.019689f, 0.015877f, 0.012773f, 0.010516f, 0.089909f,
>+    0.070332f, 0.054402f, 0.043210f, 0.034096f, 0.026625f, 0.021823f, 0.017648f,
>+    0.014649f, 0.011982f, 0.010035f, 0.091109f, 0.072428f, 0.056775f, 0.045418f,
>+    0.036679f, 0.028599f, 0.023693f, 0.019966f, 0.016603f, 0.013690f, 0.011359f,
>+    0.009657f};
> 
> }  // namespace webrtc
> #endif  // MODULES_RTP_RTCP_TEST_TESTFEC_AVERAGE_RESIDUAL_LOSS_XOR_CODES_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/test/testFec/test_fec.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/test/testFec/test_fec.cc
>index 32dc374e4fc..7c65d3dca8c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/test/testFec/test_fec.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/test/testFec/test_fec.cc
>@@ -192,10 +192,9 @@ void RunTest(bool use_flexfec) {
>                    num_media_packets * mask_bytes_per_fec_packet);
> 
>             // Transfer packet masks from bit-mask to byte-mask.
>-            internal::GeneratePacketMasks(num_media_packets, num_fec_packets,
>-                                          num_imp_packets,
>-                                          kUseUnequalProtection,
>-                                          mask_table, packet_mask.get());
>+            internal::GeneratePacketMasks(
>+                num_media_packets, num_fec_packets, num_imp_packets,
>+                kUseUnequalProtection, &mask_table, packet_mask.get());
> 
> #ifdef VERBOSE_OUTPUT
>             printf(
>@@ -253,8 +252,8 @@ void RunTest(bool use_flexfec) {
>               const uint32_t kMinPacketSize = 12;
>               const uint32_t kMaxPacketSize = static_cast<uint32_t>(
>                   IP_PACKET_SIZE - 12 - 28 - fec->MaxPacketOverhead());
>-              media_packet->length = random.Rand(kMinPacketSize,
>-                                                 kMaxPacketSize);
>+              media_packet->length =
>+                  random.Rand(kMinPacketSize, kMaxPacketSize);
> 
>               // Generate random values for the first 2 bytes.
>               media_packet->data[0] = random.Rand<uint8_t>();
>@@ -457,8 +456,8 @@ void RunTest(bool use_flexfec) {
>           }  // loop over num_imp_packets
>         }    // loop over FecPackets
>       }      // loop over num_media_packets
>-    }  // loop over loss rates
>-  }    // loop over mask types
>+    }        // loop over loss rates
>+  }          // loop over mask types
> 
>   // Have DecodeFec clear the recovered packet list.
>   fec->ResetState(&recovered_packet_list);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/test/testFec/test_packet_masks_metrics.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/test/testFec/test_packet_masks_metrics.cc
>index f40a8e8ed65..f6d2dea242a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/test/testFec/test_packet_masks_metrics.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/rtp_rtcp/test/testFec/test_packet_masks_metrics.cc
>@@ -41,7 +41,7 @@
>  * (1) Random loss: Bernoulli process, characterized by the average loss rate.
>  * (2) Bursty loss: Markov chain (Gilbert-Elliot model), characterized by two
>  * parameters: average loss rate and average burst length.
>-*/
>+ */
> 
> #include <math.h>
> 
>@@ -71,9 +71,9 @@ const int kNumStatesDistribution = 2 * kMaxMediaPacketsTest * kMaxGapSize + 1;
> 
> // The code type.
> enum CodeType {
>-  xor_random_code,    // XOR with random mask type.
>-  xor_bursty_code,    // XOR with bursty mask type.
>-  rs_code             // Reed_solomon.
>+  xor_random_code,  // XOR with random mask type.
>+  xor_bursty_code,  // XOR with bursty mask type.
>+  rs_code           // Reed_solomon.
> };
> 
> // The code size parameters.
>@@ -89,10 +89,7 @@ struct CodeSizeParams {
> };
> 
> // The type of loss models.
>-enum LossModelType {
>-  kRandomLossModel,
>-  kBurstyLossModel
>-};
>+enum LossModelType { kRandomLossModel, kBurstyLossModel };
> 
> struct LossModel {
>   LossModelType loss_type;
>@@ -101,19 +98,19 @@ struct LossModel {
> };
> 
> // Average loss rates.
>-const float kAverageLossRate[] = { 0.025f, 0.05f, 0.1f, 0.25f };
>+const float kAverageLossRate[] = {0.025f, 0.05f, 0.1f, 0.25f};
> 
> // Average burst lengths. The case of |kAverageBurstLength = 1.0| refers to
> // the random model. Note that for the random (Bernoulli) model, the average
> // burst length is determined by the average loss rate, i.e.,
> // AverageBurstLength = 1 / (1 - AverageLossRate) for random model.
>-const float kAverageBurstLength[] = { 1.0f, 2.0f, 4.0f };
>+const float kAverageBurstLength[] = {1.0f, 2.0f, 4.0f};
> 
> // Total number of loss models: For each burst length case, there are
> // a number of models corresponding to the loss rates.
>-const int kNumLossModels =  (sizeof(kAverageBurstLength) /
>-    sizeof(*kAverageBurstLength)) * (sizeof(kAverageLossRate) /
>-        sizeof(*kAverageLossRate));
>+const int kNumLossModels =
>+    (sizeof(kAverageBurstLength) / sizeof(*kAverageBurstLength)) *
>+    (sizeof(kAverageLossRate) / sizeof(*kAverageLossRate));
> 
> // Thresholds on the average loss rate of the packet loss model, below which
> // certain properties of the codes are expected.
>@@ -123,8 +120,8 @@ float loss_rate_lower_threshold = 0.025f;
> // Set of thresholds on the expected average recovery rate, for each code type.
> // These are global thresholds for now; in future version we may condition them
> // on the code length/size and protection level.
>-const float kRecoveryRateXorRandom[3] = { 0.94f, 0.50f, 0.19f };
>-const float kRecoveryRateXorBursty[3] = { 0.90f, 0.54f, 0.22f };
>+const float kRecoveryRateXorRandom[3] = {0.94f, 0.50f, 0.19f};
>+const float kRecoveryRateXorBursty[3] = {0.90f, 0.54f, 0.22f};
> 
> // Metrics for a given FEC code; each code is defined by the code type
> // (RS, XOR-random/bursty), and the code size parameters (k,m), where
>@@ -150,7 +147,7 @@ MetricsFecCode kMetricsReedSolomon[kNumberCodes];
> 
> class FecPacketMaskMetricsTest : public ::testing::Test {
>  protected:
>-  FecPacketMaskMetricsTest() { }
>+  FecPacketMaskMetricsTest() {}
> 
>   int max_num_codes_;
>   LossModel loss_model_[kNumLossModels];
>@@ -251,10 +248,9 @@ class FecPacketMaskMetricsTest : public ::testing::Test {
>                                 int tot_num_packets) {
>     // Loop over the loss models.
>     for (int k = 0; k < kNumLossModels; k++) {
>-      double loss_rate = static_cast<double>(
>-          loss_model_[k].average_loss_rate);
>-      double burst_length = static_cast<double>(
>-          loss_model_[k].average_burst_length);
>+      double loss_rate = static_cast<double>(loss_model_[k].average_loss_rate);
>+      double burst_length =
>+          static_cast<double>(loss_model_[k].average_burst_length);
>       double result = 1.0;
>       if (loss_model_[k].loss_type == kRandomLossModel) {
>         for (int i = 0; i < tot_num_packets; i++) {
>@@ -279,7 +275,7 @@ class FecPacketMaskMetricsTest : public ::testing::Test {
>         // Use stationary probability for first state/packet.
>         if (state[0] == 0) {  // Received
>           result = (1.0 - loss_rate);
>-        } else {   // Lost
>+        } else {  // Lost
>           result = loss_rate;
>         }
> 
>@@ -287,13 +283,13 @@ class FecPacketMaskMetricsTest : public ::testing::Test {
>         for (int i = 1; i < tot_num_packets; i++) {
>           // Current state is received
>           if (state[i] == 0) {
>-            if (state[i-1] == 0) {
>-              result *= prob00;   // Previous received, current received.
>-              } else {
>-                result *= prob10;  // Previous lost, current received.
>-              }
>+            if (state[i - 1] == 0) {
>+              result *= prob00;  // Previous received, current received.
>+            } else {
>+              result *= prob10;  // Previous lost, current received.
>+            }
>           } else {  // Current state is lost
>-            if (state[i-1] == 0) {
>+            if (state[i - 1] == 0) {
>               result *= prob01;  // Previous received, current lost.
>             } else {
>               result *= prob11;  // Previous lost, current lost.
>@@ -333,13 +329,12 @@ class FecPacketMaskMetricsTest : public ::testing::Test {
>     for (int loss = 1; loss <= num_fec_packets; loss++) {
>       int index = gap_number * (2 * kMaxMediaPacketsTest) + loss;
>       residual_loss_gap += metrics.residual_loss_per_loss_gap[index];
>-      tot_num_configs +=
>-          code_params_[code_index].configuration_density[index];
>+      tot_num_configs += code_params_[code_index].configuration_density[index];
>     }
>     // Normalize, to compare across code sizes.
>     if (tot_num_configs > 0) {
>-      residual_loss_gap = residual_loss_gap /
>-          static_cast<double>(tot_num_configs);
>+      residual_loss_gap =
>+          residual_loss_gap / static_cast<double>(tot_num_configs);
>     }
>     return residual_loss_gap;
>   }
>@@ -354,7 +349,7 @@ class FecPacketMaskMetricsTest : public ::testing::Test {
>       metrics->recovery_rate_per_loss[loss] = 0.0;
>       int tot_num_configs = 0;
>       double arl = 0.0;
>-      for (int gap = 0; gap < kMaxGapSize; gap ++) {
>+      for (int gap = 0; gap < kMaxGapSize; gap++) {
>         int index = gap * (2 * kMaxMediaPacketsTest) + loss;
>         arl += metrics->residual_loss_per_loss_gap[index];
>         tot_num_configs +=
>@@ -366,7 +361,8 @@ class FecPacketMaskMetricsTest : public ::testing::Test {
>       }
>       // Recovery rate for a given loss |loss| is 1 minus the scaled |arl|,
>       // where the scale factor is relative to code size/parameters.
>-      double scaled_loss = static_cast<double>(loss * num_media_packets) /
>+      double scaled_loss =
>+          static_cast<double>(loss * num_media_packets) /
>           static_cast<double>(num_media_packets + num_fec_packets);
>       metrics->recovery_rate_per_loss[loss] = 1.0 - arl / scaled_loss;
>     }
>@@ -384,10 +380,9 @@ class FecPacketMaskMetricsTest : public ::testing::Test {
>   // Compute the metrics for an FEC code, given by the code type |code_type|
>   // (XOR-random/ bursty or RS), and by the code index |code_index|
>   // (which containes the code size parameters/protection length).
>-  void ComputeMetricsForCode(CodeType code_type,
>-                             int code_index) {
>+  void ComputeMetricsForCode(CodeType code_type, int code_index) {
>     std::unique_ptr<double[]> prob_weight(new double[kNumLossModels]);
>-    memset(prob_weight.get() , 0, sizeof(double) * kNumLossModels);
>+    memset(prob_weight.get(), 0, sizeof(double) * kNumLossModels);
>     MetricsFecCode metrics_code;
>     SetMetricsZero(&metrics_code);
> 
>@@ -395,7 +390,7 @@ class FecPacketMaskMetricsTest : public ::testing::Test {
>     int num_fec_packets = code_params_[code_index].num_fec_packets;
>     int tot_num_packets = num_media_packets + num_fec_packets;
>     std::unique_ptr<uint8_t[]> state(new uint8_t[tot_num_packets]);
>-    memset(state.get() , 0, tot_num_packets);
>+    memset(state.get(), 0, tot_num_packets);
> 
>     int num_loss_configurations = static_cast<int>(pow(2.0f, tot_num_packets));
>     // Loop over all loss configurations for the symbol sequence of length
>@@ -426,9 +421,9 @@ class FecPacketMaskMetricsTest : public ::testing::Test {
>         if (bit_value == 1) {
>           state[j] = 1;  // Lost state.
>           num_packets_lost++;
>-           if (j < num_media_packets) {
>-             num_media_packets_lost++;
>-           }
>+          if (j < num_media_packets) {
>+            num_media_packets_lost++;
>+          }
>         }
>       }  // Done with loop over total number of packets.
>       assert(num_media_packets_lost <= num_media_packets);
>@@ -440,9 +435,8 @@ class FecPacketMaskMetricsTest : public ::testing::Test {
>         // Compute the number of recovered packets.
>         int num_recovered_packets = 0;
>         if (code_type == xor_random_code || code_type == xor_bursty_code) {
>-          num_recovered_packets = RecoveredMediaPackets(num_media_packets,
>-                                                        num_fec_packets,
>-                                                        state.get());
>+          num_recovered_packets = RecoveredMediaPackets(
>+              num_media_packets, num_fec_packets, state.get());
>         } else {
>           // For the RS code, we can either completely recover all the packets
>           // if the loss is less than or equal to the number of FEC packets,
>@@ -455,18 +449,17 @@ class FecPacketMaskMetricsTest : public ::testing::Test {
>         assert(num_recovered_packets <= num_media_packets);
>         // Compute the residual loss. We only care about recovering media/source
>         // packets, so residual loss is based on lost/recovered media packets.
>-        residual_loss = static_cast<double>(num_media_packets_lost -
>-                                            num_recovered_packets);
>+        residual_loss =
>+            static_cast<double>(num_media_packets_lost - num_recovered_packets);
>         // Compute the probability weights for this configuration.
>-        ComputeProbabilityWeight(prob_weight.get(),
>-                                 state.get(),
>+        ComputeProbabilityWeight(prob_weight.get(), state.get(),
>                                  tot_num_packets);
>         // Update the average and variance of the residual loss.
>         for (int k = 0; k < kNumLossModels; k++) {
>-          metrics_code.average_residual_loss[k] += residual_loss *
>-              prob_weight[k];
>-          metrics_code.variance_residual_loss[k] += residual_loss *
>+          metrics_code.average_residual_loss[k] +=
>               residual_loss * prob_weight[k];
>+          metrics_code.variance_residual_loss[k] +=
>+              residual_loss * residual_loss * prob_weight[k];
>         }
>       }  // Done with processing for num_media_packets_lost >= 1.
>       // Update the distribution statistics.
>@@ -494,12 +487,12 @@ class FecPacketMaskMetricsTest : public ::testing::Test {
>           metrics_code.average_residual_loss[k] /
>           static_cast<double>(tot_num_packets);
>       metrics_code.variance_residual_loss[k] =
>-               metrics_code.variance_residual_loss[k] /
>-               static_cast<double>(num_media_packets * num_media_packets);
>+          metrics_code.variance_residual_loss[k] /
>+          static_cast<double>(num_media_packets * num_media_packets);
>       metrics_code.variance_residual_loss[k] =
>           metrics_code.variance_residual_loss[k] -
>           (metrics_code.average_residual_loss[k] *
>-              metrics_code.average_residual_loss[k]);
>+           metrics_code.average_residual_loss[k]);
>       assert(metrics_code.variance_residual_loss[k] >= 0.0);
>       assert(metrics_code.average_residual_loss[k] > 0.0);
>       metrics_code.variance_residual_loss[k] =
>@@ -508,10 +501,8 @@ class FecPacketMaskMetricsTest : public ::testing::Test {
>     }
> 
>     // Compute marginal distribution as a function of loss parameter.
>-    ComputeRecoveryRatePerLoss(&metrics_code,
>-                               num_media_packets,
>-                               num_fec_packets,
>-                               code_index);
>+    ComputeRecoveryRatePerLoss(&metrics_code, num_media_packets,
>+                               num_fec_packets, code_index);
>     if (code_type == rs_code) {
>       CopyMetrics(&kMetricsReedSolomon[code_index], metrics_code);
>     } else if (code_type == xor_random_code) {
>@@ -523,24 +514,25 @@ class FecPacketMaskMetricsTest : public ::testing::Test {
>     }
>   }
> 
>-  void WriteOutMetricsAllFecCodes()  {
>+  void WriteOutMetricsAllFecCodes() {
>     std::string filename = test::OutputPath() + "data_metrics_all_codes";
>     FILE* fp = fopen(filename.c_str(), "wb");
>     // Loop through codes up to |kMaxMediaPacketsTest|.
>     int code_index = 0;
>     for (int num_media_packets = 1; num_media_packets <= kMaxMediaPacketsTest;
>-        num_media_packets++) {
>+         num_media_packets++) {
>       for (int num_fec_packets = 1; num_fec_packets <= num_media_packets;
>-          num_fec_packets++) {
>+           num_fec_packets++) {
>         fprintf(fp, "FOR CODE: (%d, %d) \n", num_media_packets,
>                 num_fec_packets);
>         for (int k = 0; k < kNumLossModels; k++) {
>           float loss_rate = loss_model_[k].average_loss_rate;
>           float burst_length = loss_model_[k].average_burst_length;
>-          fprintf(fp, "Loss rate = %.2f, Burst length = %.2f:  %.4f  %.4f  %.4f"
>+          fprintf(
>+              fp,
>+              "Loss rate = %.2f, Burst length = %.2f:  %.4f  %.4f  %.4f"
>               " **** %.4f %.4f %.4f \n",
>-              loss_rate,
>-              burst_length,
>+              loss_rate, burst_length,
>               100 * kMetricsReedSolomon[code_index].average_residual_loss[k],
>               100 * kMetricsXorRandom[code_index].average_residual_loss[k],
>               100 * kMetricsXorBursty[code_index].average_residual_loss[k],
>@@ -548,40 +540,27 @@ class FecPacketMaskMetricsTest : public ::testing::Test {
>               kMetricsXorRandom[code_index].variance_residual_loss[k],
>               kMetricsXorBursty[code_index].variance_residual_loss[k]);
>         }
>-        for (int gap = 0; gap < kGapSizeOutput; gap ++) {
>-          double rs_residual_loss = ComputeResidualLossPerGap(
>-              kMetricsReedSolomon[code_index],
>-              gap,
>-              num_fec_packets,
>-              code_index);
>+        for (int gap = 0; gap < kGapSizeOutput; gap++) {
>+          double rs_residual_loss =
>+              ComputeResidualLossPerGap(kMetricsReedSolomon[code_index], gap,
>+                                        num_fec_packets, code_index);
>           double xor_random_residual_loss = ComputeResidualLossPerGap(
>-              kMetricsXorRandom[code_index],
>-              gap,
>-              num_fec_packets,
>-              code_index);
>+              kMetricsXorRandom[code_index], gap, num_fec_packets, code_index);
>           double xor_bursty_residual_loss = ComputeResidualLossPerGap(
>-              kMetricsXorBursty[code_index],
>-              gap,
>-              num_fec_packets,
>-              code_index);
>-          fprintf(fp, "Residual loss as a function of gap "
>-              "%d: %.4f %.4f %.4f \n",
>-              gap,
>-              rs_residual_loss,
>-              xor_random_residual_loss,
>-              xor_bursty_residual_loss);
>+              kMetricsXorBursty[code_index], gap, num_fec_packets, code_index);
>+          fprintf(fp,
>+                  "Residual loss as a function of gap "
>+                  "%d: %.4f %.4f %.4f \n",
>+                  gap, rs_residual_loss, xor_random_residual_loss,
>+                  xor_bursty_residual_loss);
>         }
>         fprintf(fp, "Recovery rate as a function of loss number \n");
>         for (int loss = 1; loss <= num_media_packets + num_fec_packets;
>-                     loss ++) {
>-          fprintf(fp, "For loss number %d: %.4f %.4f %.4f \n",
>-                  loss,
>-                  kMetricsReedSolomon[code_index].
>-                  recovery_rate_per_loss[loss],
>-                  kMetricsXorRandom[code_index].
>-                  recovery_rate_per_loss[loss],
>-                  kMetricsXorBursty[code_index].
>-                  recovery_rate_per_loss[loss]);
>+             loss++) {
>+          fprintf(fp, "For loss number %d: %.4f %.4f %.4f \n", loss,
>+                  kMetricsReedSolomon[code_index].recovery_rate_per_loss[loss],
>+                  kMetricsXorRandom[code_index].recovery_rate_per_loss[loss],
>+                  kMetricsXorBursty[code_index].recovery_rate_per_loss[loss]);
>         }
>         fprintf(fp, "******************\n");
>         fprintf(fp, "\n");
>@@ -592,10 +571,9 @@ class FecPacketMaskMetricsTest : public ::testing::Test {
>   }
> 
>   void SetLossModels() {
>-    int num_loss_rates = sizeof(kAverageLossRate) /
>-        sizeof(*kAverageLossRate);
>-    int num_burst_lengths = sizeof(kAverageBurstLength) /
>-        sizeof(*kAverageBurstLength);
>+    int num_loss_rates = sizeof(kAverageLossRate) / sizeof(*kAverageLossRate);
>+    int num_burst_lengths =
>+        sizeof(kAverageBurstLength) / sizeof(*kAverageBurstLength);
>     int num_loss_models = 0;
>     for (int k = 0; k < num_burst_lengths; k++) {
>       for (int k2 = 0; k2 < num_loss_rates; k2++) {
>@@ -617,9 +595,9 @@ class FecPacketMaskMetricsTest : public ::testing::Test {
>   void SetCodeParams() {
>     int code_index = 0;
>     for (int num_media_packets = 1; num_media_packets <= kMaxMediaPacketsTest;
>-        num_media_packets++) {
>+         num_media_packets++) {
>       for (int num_fec_packets = 1; num_fec_packets <= num_media_packets;
>-          num_fec_packets++) {
>+           num_fec_packets++) {
>         code_params_[code_index].num_media_packets = num_media_packets;
>         code_params_[code_index].num_fec_packets = num_fec_packets;
>         code_params_[code_index].protection_level =
>@@ -646,8 +624,10 @@ class FecPacketMaskMetricsTest : public ::testing::Test {
>         }
>       }
>       if (row_degree == 0) {
>-        printf("Invalid mask: FEC packet has empty mask (does not protect "
>-            "anything) %d %d %d \n", i, num_media_packets, num_fec_packets);
>+        printf(
>+            "Invalid mask: FEC packet has empty mask (does not protect "
>+            "anything) %d %d %d \n",
>+            i, num_media_packets, num_fec_packets);
>         return -1;
>       }
>     }
>@@ -660,8 +640,10 @@ class FecPacketMaskMetricsTest : public ::testing::Test {
>         }
>       }
>       if (column_degree == 0) {
>-        printf("Invalid mask: Media packet has no protection at all %d %d %d "
>-            "\n", j, num_media_packets, num_fec_packets);
>+        printf(
>+            "Invalid mask: Media packet has no protection at all %d %d %d "
>+            "\n",
>+            j, num_media_packets, num_fec_packets);
>         return -1;
>       }
>     }
>@@ -696,8 +678,10 @@ class FecPacketMaskMetricsTest : public ::testing::Test {
>             }
>           }
>           if (degree == num_fec_packets) {
>-            printf("Invalid mask: Two media packets are have full degree "
>-                "%d %d %d %d \n", j, j2, num_media_packets, num_fec_packets);
>+            printf(
>+                "Invalid mask: Two media packets are have full degree "
>+                "%d %d %d %d \n",
>+                j, j2, num_media_packets, num_fec_packets);
>             return -1;
>           }
>         }
>@@ -725,8 +709,7 @@ class FecPacketMaskMetricsTest : public ::testing::Test {
>     fprintf(fp_mask_, "\n");
>   }
> 
>-  int ProcessXORPacketMasks(CodeType code_type,
>-                          FecMaskType fec_mask_type) {
>+  int ProcessXORPacketMasks(CodeType code_type, FecMaskType fec_mask_type) {
>     int code_index = 0;
>     // Maximum number of media packets allowed for the mask type.
>     const int packet_mask_max = kMaxMediaPackets[fec_mask_type];
>@@ -734,17 +717,16 @@ class FecPacketMaskMetricsTest : public ::testing::Test {
>         new uint8_t[packet_mask_max * kUlpfecMaxPacketMaskSize]);
>     // Loop through codes up to |kMaxMediaPacketsTest|.
>     for (int num_media_packets = 1; num_media_packets <= kMaxMediaPacketsTest;
>-        num_media_packets++) {
>+         ++num_media_packets) {
>       const int mask_bytes_fec_packet =
>           static_cast<int>(internal::PacketMaskSize(num_media_packets));
>       internal::PacketMaskTable mask_table(fec_mask_type, num_media_packets);
>       for (int num_fec_packets = 1; num_fec_packets <= num_media_packets;
>-          num_fec_packets++) {
>+           num_fec_packets++) {
>         memset(packet_mask.get(), 0, num_media_packets * mask_bytes_fec_packet);
>-        memcpy(packet_mask.get(),
>-               mask_table.fec_packet_mask_table()[num_media_packets - 1]
>-                                                 [num_fec_packets - 1],
>-               num_fec_packets * mask_bytes_fec_packet);
>+        rtc::ArrayView<const uint8_t> mask =
>+            mask_table.LookUp(num_media_packets, num_fec_packets);
>+        memcpy(packet_mask.get(), &mask[0], mask.size());
>         // Convert to bit mask.
>         GetPacketMaskConvertToBitMask(packet_mask.get(), num_media_packets,
>                                       num_fec_packets, mask_bytes_fec_packet,
>@@ -753,8 +735,7 @@ class FecPacketMaskMetricsTest : public ::testing::Test {
>           return -1;
>         }
>         // Compute the metrics for this code/mask.
>-        ComputeMetricsForCode(code_type,
>-                              code_index);
>+        ComputeMetricsForCode(code_type, code_index);
>         code_index++;
>       }
>     }
>@@ -765,12 +746,11 @@ class FecPacketMaskMetricsTest : public ::testing::Test {
>   void ProcessRS(CodeType code_type) {
>     int code_index = 0;
>     for (int num_media_packets = 1; num_media_packets <= kMaxMediaPacketsTest;
>-        num_media_packets++) {
>+         num_media_packets++) {
>       for (int num_fec_packets = 1; num_fec_packets <= num_media_packets;
>-          num_fec_packets++) {
>+           num_fec_packets++) {
>         // Compute the metrics for this code type.
>-        ComputeMetricsForCode(code_type,
>-                              code_index);
>+        ComputeMetricsForCode(code_type, code_index);
>         code_index++;
>       }
>     }
>@@ -803,10 +783,9 @@ TEST_F(FecPacketMaskMetricsTest, FecXorMaxResidualLoss) {
>   SetCodeParams();
>   ComputeMetricsAllCodes();
>   WriteOutMetricsAllFecCodes();
>-  int num_loss_rates = sizeof(kAverageLossRate) /
>-      sizeof(*kAverageLossRate);
>-  int num_burst_lengths = sizeof(kAverageBurstLength) /
>-      sizeof(*kAverageBurstLength);
>+  int num_loss_rates = sizeof(kAverageLossRate) / sizeof(*kAverageLossRate);
>+  int num_burst_lengths =
>+      sizeof(kAverageBurstLength) / sizeof(*kAverageBurstLength);
>   for (int code_index = 0; code_index < max_num_codes_; code_index++) {
>     double sum_residual_loss_random_mask_random_loss = 0.0;
>     double sum_residual_loss_bursty_mask_bursty_loss = 0.0;
>@@ -824,7 +803,7 @@ TEST_F(FecPacketMaskMetricsTest, FecXorMaxResidualLoss) {
>         sum_residual_loss_random_mask_random_loss / num_loss_rates;
>     float average_residual_loss_bursty_mask_bursty_loss =
>         sum_residual_loss_bursty_mask_bursty_loss /
>-        (num_loss_rates * (num_burst_lengths  - 1));
>+        (num_loss_rates * (num_burst_lengths - 1));
>     const float ref_random_mask = kMaxResidualLossRandomMask[code_index];
>     const float ref_bursty_mask = kMaxResidualLossBurstyMask[code_index];
>     EXPECT_LE(average_residual_loss_random_mask_random_loss, ref_random_mask);
>@@ -846,16 +825,16 @@ TEST_F(FecPacketMaskMetricsTest, FecXorVsRS) {
>       float loss_rate = loss_model_[k].average_loss_rate;
>       float protection_level = code_params_[code_index].protection_level;
>       // Under these conditions we expect XOR to not be better than RS.
>-       if (loss_model_[k].loss_type == kRandomLossModel &&
>-           loss_rate <= protection_level) {
>+      if (loss_model_[k].loss_type == kRandomLossModel &&
>+          loss_rate <= protection_level) {
>         EXPECT_GE(kMetricsXorRandom[code_index].average_residual_loss[k],
>                   kMetricsReedSolomon[code_index].average_residual_loss[k]);
>         EXPECT_GE(kMetricsXorBursty[code_index].average_residual_loss[k],
>                   kMetricsReedSolomon[code_index].average_residual_loss[k]);
>-       }
>-       // TODO(marpan): There are some cases (for high loss rates and/or
>-       // burst loss models) where XOR is better than RS. Is there some pattern
>-       // we can identify and enforce as a constraint?
>+      }
>+      // TODO(marpan): There are some cases (for high loss rates and/or
>+      // burst loss models) where XOR is better than RS. Is there some pattern
>+      // we can identify and enforce as a constraint?
>     }
>   }
> }
>@@ -870,25 +849,24 @@ TEST_F(FecPacketMaskMetricsTest, FecTrendXorVsRsLossRate) {
>   // TODO(marpan): Examine this further to see if the condition can be strictly
>   // satisfied (i.e., scale = 1.0) for all codes with different/better masks.
>   double scale = 0.90;
>-  int num_loss_rates = sizeof(kAverageLossRate) /
>-      sizeof(*kAverageLossRate);
>-  int num_burst_lengths = sizeof(kAverageBurstLength) /
>-      sizeof(*kAverageBurstLength);
>+  int num_loss_rates = sizeof(kAverageLossRate) / sizeof(*kAverageLossRate);
>+  int num_burst_lengths =
>+      sizeof(kAverageBurstLength) / sizeof(*kAverageBurstLength);
>   for (int code_index = 0; code_index < max_num_codes_; code_index++) {
>     for (int i = 0; i < num_burst_lengths; i++) {
>       for (int j = 0; j < num_loss_rates - 1; j++) {
>         int k = num_loss_rates * i + j;
>         // For XOR random.
>         if (kMetricsXorRandom[code_index].average_residual_loss[k] >
>-        kMetricsReedSolomon[code_index].average_residual_loss[k]) {
>+            kMetricsReedSolomon[code_index].average_residual_loss[k]) {
>           double diff_rs_xor_random_loss1 =
>               (kMetricsXorRandom[code_index].average_residual_loss[k] -
>                kMetricsReedSolomon[code_index].average_residual_loss[k]) /
>-               kMetricsXorRandom[code_index].average_residual_loss[k];
>+              kMetricsXorRandom[code_index].average_residual_loss[k];
>           double diff_rs_xor_random_loss2 =
>-              (kMetricsXorRandom[code_index].average_residual_loss[k+1] -
>-               kMetricsReedSolomon[code_index].average_residual_loss[k+1]) /
>-               kMetricsXorRandom[code_index].average_residual_loss[k+1];
>+              (kMetricsXorRandom[code_index].average_residual_loss[k + 1] -
>+               kMetricsReedSolomon[code_index].average_residual_loss[k + 1]) /
>+              kMetricsXorRandom[code_index].average_residual_loss[k + 1];
>           EXPECT_GE(diff_rs_xor_random_loss1, scale * diff_rs_xor_random_loss2);
>         }
>         // TODO(marpan): Investigate the cases for the bursty mask where
>@@ -911,11 +889,11 @@ TEST_F(FecPacketMaskMetricsTest, FecBehaviorViaProtectionLevelAndLength) {
>   for (int code_index1 = 0; code_index1 < max_num_codes_; code_index1++) {
>     float protection_level1 = code_params_[code_index1].protection_level;
>     int length1 = code_params_[code_index1].num_media_packets +
>-        code_params_[code_index1].num_fec_packets;
>+                  code_params_[code_index1].num_fec_packets;
>     for (int code_index2 = 0; code_index2 < max_num_codes_; code_index2++) {
>       float protection_level2 = code_params_[code_index2].protection_level;
>       int length2 = code_params_[code_index2].num_media_packets +
>-          code_params_[code_index2].num_fec_packets;
>+                    code_params_[code_index2].num_fec_packets;
>       // Codes with higher protection are more efficient, conditioned on the
>       // length of the code (higher protection but shorter length codes are
>       // generally not more efficient). For two codes with equal protection,
>@@ -961,11 +939,9 @@ TEST_F(FecPacketMaskMetricsTest, FecVarianceBehaviorXorVsRs) {
>   double scale = 0.95;
>   for (int code_index = 0; code_index < max_num_codes_; code_index++) {
>     for (int k = 0; k < kNumLossModels; k++) {
>-      EXPECT_LE(scale *
>-                kMetricsXorRandom[code_index].variance_residual_loss[k],
>+      EXPECT_LE(scale * kMetricsXorRandom[code_index].variance_residual_loss[k],
>                 kMetricsReedSolomon[code_index].variance_residual_loss[k]);
>-      EXPECT_LE(scale *
>-                kMetricsXorBursty[code_index].variance_residual_loss[k],
>+      EXPECT_LE(scale * kMetricsXorBursty[code_index].variance_residual_loss[k],
>                 kMetricsReedSolomon[code_index].variance_residual_loss[k]);
>     }
>   }
>@@ -981,8 +957,8 @@ TEST_F(FecPacketMaskMetricsTest, FecXorBurstyPerfectRecoveryConsecutiveLoss) {
>     int num_fec_packets = code_params_[code_index].num_fec_packets;
>     for (int loss = 1; loss <= num_fec_packets; loss++) {
>       int index = loss;  // |gap| is zero.
>-      EXPECT_EQ(kMetricsXorBursty[code_index].
>-                residual_loss_per_loss_gap[index], 0.0);
>+      EXPECT_EQ(kMetricsXorBursty[code_index].residual_loss_per_loss_gap[index],
>+                0.0);
>     }
>   }
> }
>@@ -1038,42 +1014,46 @@ TEST_F(FecPacketMaskMetricsTest, FecRecoveryRateUnderLossConditions) {
>     // Perfect recovery (|recovery_rate_per_loss| == 1) is expected for
>     // |loss_number| = 1, for all codes.
>     int loss_number = 1;
>-    EXPECT_EQ(kMetricsReedSolomon[code_index].
>-              recovery_rate_per_loss[loss_number], 1.0);
>-    EXPECT_EQ(kMetricsXorRandom[code_index].
>-              recovery_rate_per_loss[loss_number], 1.0);
>-    EXPECT_EQ(kMetricsXorBursty[code_index].
>-              recovery_rate_per_loss[loss_number], 1.0);
>+    EXPECT_EQ(
>+        kMetricsReedSolomon[code_index].recovery_rate_per_loss[loss_number],
>+        1.0);
>+    EXPECT_EQ(kMetricsXorRandom[code_index].recovery_rate_per_loss[loss_number],
>+              1.0);
>+    EXPECT_EQ(kMetricsXorBursty[code_index].recovery_rate_per_loss[loss_number],
>+              1.0);
>     // For |loss_number| = |num_fec_packets| / 2, we expect the following:
>     // Perfect recovery for RS, and recovery for XOR above the threshold.
>     loss_number = num_fec_packets / 2 > 0 ? num_fec_packets / 2 : 1;
>-    EXPECT_EQ(kMetricsReedSolomon[code_index].
>-              recovery_rate_per_loss[loss_number], 1.0);
>-    EXPECT_GE(kMetricsXorRandom[code_index].
>-              recovery_rate_per_loss[loss_number], kRecoveryRateXorRandom[0]);
>-    EXPECT_GE(kMetricsXorBursty[code_index].
>-              recovery_rate_per_loss[loss_number], kRecoveryRateXorBursty[0]);
>+    EXPECT_EQ(
>+        kMetricsReedSolomon[code_index].recovery_rate_per_loss[loss_number],
>+        1.0);
>+    EXPECT_GE(kMetricsXorRandom[code_index].recovery_rate_per_loss[loss_number],
>+              kRecoveryRateXorRandom[0]);
>+    EXPECT_GE(kMetricsXorBursty[code_index].recovery_rate_per_loss[loss_number],
>+              kRecoveryRateXorBursty[0]);
>     // For |loss_number| = |num_fec_packets|, we expect the following:
>     // Perfect recovery for RS, and recovery for XOR above the threshold.
>     loss_number = num_fec_packets;
>-    EXPECT_EQ(kMetricsReedSolomon[code_index].
>-              recovery_rate_per_loss[loss_number], 1.0);
>-    EXPECT_GE(kMetricsXorRandom[code_index].
>-              recovery_rate_per_loss[loss_number], kRecoveryRateXorRandom[1]);
>-    EXPECT_GE(kMetricsXorBursty[code_index].
>-              recovery_rate_per_loss[loss_number], kRecoveryRateXorBursty[1]);
>+    EXPECT_EQ(
>+        kMetricsReedSolomon[code_index].recovery_rate_per_loss[loss_number],
>+        1.0);
>+    EXPECT_GE(kMetricsXorRandom[code_index].recovery_rate_per_loss[loss_number],
>+              kRecoveryRateXorRandom[1]);
>+    EXPECT_GE(kMetricsXorBursty[code_index].recovery_rate_per_loss[loss_number],
>+              kRecoveryRateXorBursty[1]);
>     // For |loss_number| = |num_fec_packets| + 1, we expect the following:
>     // Zero recovery for RS, but non-zero recovery for XOR.
>     if (num_fec_packets > 1 && num_media_packets > 2) {
>-      loss_number =  num_fec_packets + 1;
>-      EXPECT_EQ(kMetricsReedSolomon[code_index].
>-                recovery_rate_per_loss[loss_number], 0.0);
>-      EXPECT_GE(kMetricsXorRandom[code_index].
>-                recovery_rate_per_loss[loss_number],
>-                kRecoveryRateXorRandom[2]);
>-      EXPECT_GE(kMetricsXorBursty[code_index].
>-                recovery_rate_per_loss[loss_number],
>-                kRecoveryRateXorBursty[2]);
>+      loss_number = num_fec_packets + 1;
>+      EXPECT_EQ(
>+          kMetricsReedSolomon[code_index].recovery_rate_per_loss[loss_number],
>+          0.0);
>+      EXPECT_GE(
>+          kMetricsXorRandom[code_index].recovery_rate_per_loss[loss_number],
>+          kRecoveryRateXorRandom[2]);
>+      EXPECT_GE(
>+          kMetricsXorBursty[code_index].recovery_rate_per_loss[loss_number],
>+          kRecoveryRateXorBursty[2]);
>     }
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/fft/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/fft/BUILD.gn
>new file mode 100644
>index 00000000000..ad513410c77
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/fft/BUILD.gn
>@@ -0,0 +1,16 @@
>+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+#
>+# Use of this source code is governed by a BSD-style license
>+# that can be found in the ../../../LICENSE file in the root of the source
>+# tree. An additional intellectual property rights grant can be found
>+# in the file PATENTS.  All contributing project authors may
>+# be found in the AUTHORS file in the root of the source tree.
>+
>+import("../../../webrtc.gni")
>+
>+rtc_source_set("fft") {
>+  sources = [
>+    "fft.c",
>+    "fft.h",
>+  ]
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/fft/LICENSE b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/fft/LICENSE
>new file mode 100644
>index 00000000000..c0a78058cff
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/fft/LICENSE
>@@ -0,0 +1,25 @@
>+/*
>+ * Copyright(c)1995,97 Mark Olesen <olesen@me.QueensU.CA>
>+ *    Queen's Univ at Kingston (Canada)
>+ *
>+ * Permission to use, copy, modify, and distribute this software for
>+ * any purpose without fee is hereby granted, provided that this
>+ * entire notice is included in all copies of any software which is
>+ * or includes a copy or modification of this software and in all
>+ * copies of the supporting documentation for such software.
>+ *
>+ * THIS SOFTWARE IS BEING PROVIDED "AS IS", WITHOUT ANY EXPRESS OR
>+ * IMPLIED WARRANTY.  IN PARTICULAR, NEITHER THE AUTHOR NOR QUEEN'S
>+ * UNIVERSITY AT KINGSTON MAKES ANY REPRESENTATION OR WARRANTY OF ANY
>+ * KIND CONCERNING THE MERCHANTABILITY OF THIS SOFTWARE OR ITS
>+ * FITNESS FOR ANY PARTICULAR PURPOSE.
>+ *
>+ * All of which is to say that you can do what you like with this
>+ * source code provided you don't try to sell it as your own and you
>+ * include an unaltered copy of this message (including the
>+ * copyright).
>+ *
>+ * It is also implicitly understood that bug fixes and improvements
>+ * should make their way back to the general Internet community so
>+ * that everyone benefits.
>+ */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/fft/README.chromium b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/fft/README.chromium
>new file mode 100644
>index 00000000000..94d20d422f2
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/fft/README.chromium
>@@ -0,0 +1,12 @@
>+Name: fft
>+Short Name: fft
>+URL:
>+Version: 0
>+Date: 2018-07-26
>+License: Custom license
>+License File: LICENSE
>+Security Critical: yes
>+
>+Description:
>+Multivariate complex Fourier transform, computed in place
>+using mixed-radix Fast Fourier Transform algorithm.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/fft.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/fft/fft.c
>similarity index 99%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/fft.c
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/fft/fft.c
>index c854d8c27c4..72604626d97 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/fft.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/fft/fft.c
>@@ -123,12 +123,11 @@
>  * following two constants should agree with the array dimensions.
>  *
>  *----------------------------------------------------------------------*/
>-#include "fft.h"
> 
> #include <stdlib.h>
> #include <math.h>
> 
>-
>+#include "modules/third_party/fft/fft.h"
> 
> /* double precision routine */
> static int
>@@ -212,7 +211,7 @@ int  WebRtcIsac_Fftns(unsigned int ndim, const int dims[],
>     {
>       max_factors = (int)nSpan;
>     }
>-    if ((int)nSpan > max_perm) 
>+    if ((int)nSpan > max_perm)
>     {
>       max_perm = (int)nSpan;
>     }
>@@ -416,7 +415,7 @@ static int   FFTRADIX (REAL Re[],
>   }
> 
>   /* test that mfactors is in range */
>-  if (mfactor > NFACTOR)
>+  if (mfactor > FFT_NFACTOR)
>   {
>     return -1;
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/fft.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/fft/fft.h
>similarity index 61%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/fft.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/fft/fft.h
>index b583b50268d..f8f8b6f17ba 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/isac/main/source/fft.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/fft/fft.h
>@@ -2,7 +2,7 @@
>  *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
>  *
>  *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>+ *  that can be found in the ../../../LICENSE file in the root of the source
>  *  tree. An additional intellectual property rights grant can be found
>  *  in the file PATENTS.  All contributing project authors may
>  *  be found in the AUTHORS file in the root of the source tree.
>@@ -27,19 +27,32 @@
>  * See the comments in the code for correct usage!
>  */
> 
>-#ifndef MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_FFT_H_
>-#define MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_FFT_H_
>+#ifndef MODULES_THIRD_PARTY_FFT_FFT_H_
>+#define MODULES_THIRD_PARTY_FFT_FFT_H_
> 
>+#define FFT_MAXFFTSIZE 2048
>+#define FFT_NFACTOR 11
> 
>-#include "structs.h"
>+typedef struct {
>+  unsigned int SpaceAlloced;
>+  unsigned int MaxPermAlloced;
>+  double Tmp0[FFT_MAXFFTSIZE];
>+  double Tmp1[FFT_MAXFFTSIZE];
>+  double Tmp2[FFT_MAXFFTSIZE];
>+  double Tmp3[FFT_MAXFFTSIZE];
>+  int Perm[FFT_MAXFFTSIZE];
>+  int factor[FFT_NFACTOR];
> 
>+} FFTstr;
> 
> /* double precision routine */
> 
>+int WebRtcIsac_Fftns(unsigned int ndim,
>+                     const int dims[],
>+                     double Re[],
>+                     double Im[],
>+                     int isign,
>+                     double scaling,
>+                     FFTstr* fftstate);
> 
>-int WebRtcIsac_Fftns (unsigned int ndim, const int dims[], double Re[], double Im[],
>-                     int isign, double scaling, FFTstr *fftstate);
>-
>-
>-
>-#endif /* MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_FFT_H_ */
>+#endif /* MODULES_THIRD_PARTY_FFT_FFT_H_ */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/g711/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/g711/BUILD.gn
>new file mode 100644
>index 00000000000..ca1ccf3b233
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/g711/BUILD.gn
>@@ -0,0 +1,17 @@
>+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+#
>+# Use of this source code is governed by a BSD-style license
>+# that can be found in the ../../../LICENSE file in the root of the source
>+# tree. An additional intellectual property rights grant can be found
>+# in the file PATENTS.  All contributing project authors may
>+# be found in the AUTHORS file in the root of the source tree.
>+
>+import("../../../webrtc.gni")
>+
>+rtc_source_set("g711_3p") {
>+  poisonous = [ "audio_codecs" ]
>+  sources = [
>+    "g711.c",
>+    "g711.h",
>+  ]
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/g711/LICENSE b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/g711/LICENSE
>new file mode 100644
>index 00000000000..3cdf910304d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/g711/LICENSE
>@@ -0,0 +1,14 @@
>+/*
>+ * SpanDSP - a series of DSP components for telephony
>+ *
>+ * g711.h - In line A-law and u-law conversion routines
>+ *
>+ * Written by Steve Underwood <steveu@coppice.org>
>+ *
>+ * Copyright (C) 2001 Steve Underwood
>+ *
>+ *  Despite my general liking of the GPL, I place this code in the
>+ *  public domain for the benefit of all mankind - even the slimy
>+ *  ones who might try to proprietize my work and use it to my
>+ *  detriment.
>+ */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/g711/README.chromium b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/g711/README.chromium
>new file mode 100644
>index 00000000000..1baa2637a92
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/g711/README.chromium
>@@ -0,0 +1,11 @@
>+Name: In line A-law and u-law conversion routines
>+Short Name: g711
>+URL:
>+Version: 0
>+Date: 2018-06-25
>+License: Custom license
>+License File: LICENSE
>+Security Critical: yes
>+
>+Description:
>+In line A-law and u-law conversion routines
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g711/g711.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/g711/g711.c
>similarity index 98%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g711/g711.c
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/g711/g711.c
>index 2b1a428a3a9..b7ae6bb109e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g711/g711.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/g711/g711.c
>@@ -20,8 +20,7 @@
>  * -Added option to run encoder bitexact with ITU-T reference implementation
>  */
> 
>-#include "g711.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include "modules/third_party/g711/g711.h"
> 
> /* Copied from the CCITT G.711 specification */
> static const uint8_t ulaw_to_alaw_table[256] = {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g711/g711.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/g711/g711.h
>similarity index 89%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g711/g711.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/g711/g711.h
>index 8b1fc81ad15..4eef42c0bf1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g711/g711.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/g711/g711.h
>@@ -17,7 +17,8 @@
>  * Modifications for WebRtc, 2011/04/28, by tlegrand:
>  * -Changed to use WebRtc types
>  * -Changed __inline__ to __inline
>- * -Two changes to make implementation bitexact with ITU-T reference implementation
>+ * -Two changes to make implementation bitexact with ITU-T reference
>+ * implementation
>  */
> 
> /*! \page g711_page A-law and mu-law handling
>@@ -42,14 +43,14 @@ difficult to achieve the precise transcoding procedure laid down in the G.711
> specification by other means.
> */
> 
>-#if !defined(_G711_H_)
>-#define _G711_H_
>+#ifndef MODULES_THIRD_PARTY_G711_G711_H_
>+#define MODULES_THIRD_PARTY_G711_G711_H_
> 
> #ifdef __cplusplus
> extern "C" {
> #endif
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include <stdint.h>
> 
> #if defined(__i386__)
> /*! \brief Find the bit position of the highest set bit in a word
>@@ -58,10 +59,11 @@ extern "C" {
> static __inline__ int top_bit(unsigned int bits) {
>   int res;
> 
>-  __asm__ __volatile__(" movl $-1,%%edx;\n"
>-                       " bsrl %%eax,%%edx;\n"
>-                       : "=d" (res)
>-                       : "a" (bits));
>+  __asm__ __volatile__(
>+      " movl $-1,%%edx;\n"
>+      " bsrl %%eax,%%edx;\n"
>+      : "=d"(res)
>+      : "a"(bits));
>   return res;
> }
> 
>@@ -71,30 +73,33 @@ static __inline__ int top_bit(unsigned int bits) {
> static __inline__ int bottom_bit(unsigned int bits) {
>   int res;
> 
>-  __asm__ __volatile__(" movl $-1,%%edx;\n"
>-                       " bsfl %%eax,%%edx;\n"
>-                       : "=d" (res)
>-                       : "a" (bits));
>+  __asm__ __volatile__(
>+      " movl $-1,%%edx;\n"
>+      " bsfl %%eax,%%edx;\n"
>+      : "=d"(res)
>+      : "a"(bits));
>   return res;
> }
> #elif defined(__x86_64__)
> static __inline__ int top_bit(unsigned int bits) {
>   int res;
> 
>-  __asm__ __volatile__(" movq $-1,%%rdx;\n"
>-                       " bsrq %%rax,%%rdx;\n"
>-                       : "=d" (res)
>-                       : "a" (bits));
>+  __asm__ __volatile__(
>+      " movq $-1,%%rdx;\n"
>+      " bsrq %%rax,%%rdx;\n"
>+      : "=d"(res)
>+      : "a"(bits));
>   return res;
> }
> 
> static __inline__ int bottom_bit(unsigned int bits) {
>   int res;
> 
>-  __asm__ __volatile__(" movq $-1,%%rdx;\n"
>-                       " bsfq %%rax,%%rdx;\n"
>-                       : "=d" (res)
>-                       : "a" (bits));
>+  __asm__ __volatile__(
>+      " movq $-1,%%rdx;\n"
>+      " bsfq %%rax,%%rdx;\n"
>+      : "=d"(res)
>+      : "a"(bits));
>   return res;
> }
> #else
>@@ -166,8 +171,8 @@ static __inline int bottom_bit(unsigned int bits) {
>  *      linear sound like peanuts these days, and shouldn't an array lookup be
>  *      real fast? No! When the cache sloshes as badly as this one will, a tight
>  *      calculation may be better. The messiest part is normally finding the
>- *      segment, but a little inline assembly can fix that on an i386, x86_64 and
>- *      many other modern processors.
>+ *      segment, but a little inline assembly can fix that on an i386, x86_64
>+ * and many other modern processors.
>  */
> 
> /*
>@@ -196,8 +201,9 @@ static __inline int bottom_bit(unsigned int bits) {
>  * John Wiley & Sons, pps 98-111 and 472-476.
>  */
> 
>-//#define ULAW_ZEROTRAP                 /* turn on the trap as per the MIL-STD */
>-#define ULAW_BIAS 0x84  /* Bias for linear code. */
>+//#define ULAW_ZEROTRAP                 /* turn on the trap as per the MIL-STD
>+//*/
>+#define ULAW_BIAS 0x84 /* Bias for linear code. */
> 
> /*! \brief Encode a linear sample to u-law
>     \param linear The sample to encode.
>@@ -249,7 +255,7 @@ static __inline int16_t ulaw_to_linear(uint8_t ulaw) {
>    * Extract and bias the quantization bits. Then
>    * shift up by the segment number and subtract out the bias.
>    */
>-  t = (((ulaw & 0x0F) << 3) + ULAW_BIAS) << (((int) ulaw & 0x70) >> 4);
>+  t = (((ulaw & 0x0F) << 3) + ULAW_BIAS) << (((int)ulaw & 0x70) >> 4);
>   return (int16_t)((ulaw & 0x80) ? (ULAW_BIAS - t) : (t - ULAW_BIAS));
> }
> 
>@@ -317,7 +323,7 @@ static __inline int16_t alaw_to_linear(uint8_t alaw) {
> 
>   alaw ^= ALAW_AMI_MASK;
>   i = ((alaw & 0x0F) << 4);
>-  seg = (((int) alaw & 0x70) >> 4);
>+  seg = (((int)alaw & 0x70) >> 4);
>   if (seg)
>     i = (i + 0x108) << (seg - 1);
>   else
>@@ -341,4 +347,4 @@ uint8_t ulaw_to_alaw(uint8_t ulaw);
> }
> #endif
> 
>-#endif
>+#endif /* MODULES_THIRD_PARTY_G711_G711_H_ */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/g722/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/g722/BUILD.gn
>new file mode 100644
>index 00000000000..a1bfe89257c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/g722/BUILD.gn
>@@ -0,0 +1,18 @@
>+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+#
>+# Use of this source code is governed by a BSD-style license
>+# that can be found in the ../../../LICENSE file in the root of the source
>+# tree. An additional intellectual property rights grant can be found
>+# in the file PATENTS.  All contributing project authors may
>+# be found in the AUTHORS file in the root of the source tree.
>+
>+import("../../../webrtc.gni")
>+
>+rtc_source_set("g722_3p") {
>+  poisonous = [ "audio_codecs" ]
>+  sources = [
>+    "g722_decode.c",
>+    "g722_enc_dec.h",
>+    "g722_encode.c",
>+  ]
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/g722/LICENSE b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/g722/LICENSE
>new file mode 100644
>index 00000000000..ea6308f0db7
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/g722/LICENSE
>@@ -0,0 +1,20 @@
>+/*
>+ * SpanDSP - a series of DSP components for telephony
>+ *
>+ * g722_decode.c - The ITU G.722 codec, decode part.
>+ *
>+ * Written by Steve Underwood <steveu@coppice.org>
>+ *
>+ * Copyright (C) 2005 Steve Underwood
>+ *
>+ *  Despite my general liking of the GPL, I place my own contributions
>+ *  to this code in the public domain for the benefit of all mankind -
>+ *  even the slimy ones who might try to proprietize my work and use it
>+ *  to my detriment.
>+ *
>+ * Based in part on a single channel G.722 codec which is:
>+ *
>+ * Copyright (c) CMU 1993
>+ * Computer Science, Speech Group
>+ * Chengxiang Lu and Alex Hauptmann
>+ */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/g722/README.chromium b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/g722/README.chromium
>new file mode 100644
>index 00000000000..c427ed8cf25
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/g722/README.chromium
>@@ -0,0 +1,11 @@
>+Name: The ITU G.722 codec, encode and decode part.
>+Short Name: g722
>+URL:
>+Version: 0
>+Date: 2018-06-25
>+License: Custom license
>+License File: LICENSE
>+Security Critical: yes
>+
>+Description:
>+The ITU G.722 codec, encode and decode part.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/g722_decode.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/g722/g722_decode.c
>similarity index 99%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/g722_decode.c
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/g722/g722_decode.c
>index 84af1550974..012aeb50284 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/g722_decode.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/g722/g722_decode.c
>@@ -34,8 +34,7 @@
> #include <stdio.h>
> #include <stdlib.h>
> 
>-#include "g722_enc_dec.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include "modules/third_party/g722/g722_enc_dec.h"
> 
> #if !defined(FALSE)
> #define FALSE 0
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/g722/g722_enc_dec.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/g722/g722_enc_dec.h
>new file mode 100644
>index 00000000000..898fa279cca
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/g722/g722_enc_dec.h
>@@ -0,0 +1,153 @@
>+/*
>+ * SpanDSP - a series of DSP components for telephony
>+ *
>+ * g722.h - The ITU G.722 codec.
>+ *
>+ * Written by Steve Underwood <steveu@coppice.org>
>+ *
>+ * Copyright (C) 2005 Steve Underwood
>+ *
>+ *  Despite my general liking of the GPL, I place my own contributions
>+ *  to this code in the public domain for the benefit of all mankind -
>+ *  even the slimy ones who might try to proprietize my work and use it
>+ *  to my detriment.
>+ *
>+ * Based on a single channel G.722 codec which is:
>+ *
>+ *****    Copyright (c) CMU    1993      *****
>+ * Computer Science, Speech Group
>+ * Chengxiang Lu and Alex Hauptmann
>+ *
>+ * $Id: g722.h,v 1.10 2006/06/16 12:45:53 steveu Exp $
>+ *
>+ * Modifications for WebRtc, 2011/04/28, by tlegrand:
>+ * -Changed to use WebRtc types
>+ * -Added new defines for minimum and maximum values of short int
>+ */
>+
>+/*! \file */
>+
>+#ifndef MODULES_THIRD_PARTY_G722_G722_H_
>+#define MODULES_THIRD_PARTY_G722_G722_H_
>+
>+#include <stdint.h>
>+
>+/*! \page g722_page G.722 encoding and decoding
>+\section g722_page_sec_1 What does it do?
>+The G.722 module is a bit exact implementation of the ITU G.722 specification
>+for all three specified bit rates - 64000bps, 56000bps and 48000bps. It passes
>+the ITU tests.
>+
>+To allow fast and flexible interworking with narrow band telephony, the encoder
>+and decoder support an option for the linear audio to be an 8k samples/second
>+stream. In this mode the codec is considerably faster, and still fully
>+compatible with wideband terminals using G.722.
>+
>+\section g722_page_sec_2 How does it work?
>+???.
>+*/
>+
>+#define WEBRTC_INT16_MAX 32767
>+#define WEBRTC_INT16_MIN -32768
>+
>+enum { G722_SAMPLE_RATE_8000 = 0x0001, G722_PACKED = 0x0002 };
>+
>+typedef struct {
>+  /*! TRUE if the operating in the special ITU test mode, with the band split
>+     filters disabled. */
>+  int itu_test_mode;
>+  /*! TRUE if the G.722 data is packed */
>+  int packed;
>+  /*! TRUE if encode from 8k samples/second */
>+  int eight_k;
>+  /*! 6 for 48000kbps, 7 for 56000kbps, or 8 for 64000kbps. */
>+  int bits_per_sample;
>+
>+  /*! Signal history for the QMF */
>+  int x[24];
>+
>+  struct {
>+    int s;
>+    int sp;
>+    int sz;
>+    int r[3];
>+    int a[3];
>+    int ap[3];
>+    int p[3];
>+    int d[7];
>+    int b[7];
>+    int bp[7];
>+    int sg[7];
>+    int nb;
>+    int det;
>+  } band[2];
>+
>+  unsigned int in_buffer;
>+  int in_bits;
>+  unsigned int out_buffer;
>+  int out_bits;
>+} G722EncoderState;
>+
>+typedef struct {
>+  /*! TRUE if the operating in the special ITU test mode, with the band split
>+     filters disabled. */
>+  int itu_test_mode;
>+  /*! TRUE if the G.722 data is packed */
>+  int packed;
>+  /*! TRUE if decode to 8k samples/second */
>+  int eight_k;
>+  /*! 6 for 48000kbps, 7 for 56000kbps, or 8 for 64000kbps. */
>+  int bits_per_sample;
>+
>+  /*! Signal history for the QMF */
>+  int x[24];
>+
>+  struct {
>+    int s;
>+    int sp;
>+    int sz;
>+    int r[3];
>+    int a[3];
>+    int ap[3];
>+    int p[3];
>+    int d[7];
>+    int b[7];
>+    int bp[7];
>+    int sg[7];
>+    int nb;
>+    int det;
>+  } band[2];
>+
>+  unsigned int in_buffer;
>+  int in_bits;
>+  unsigned int out_buffer;
>+  int out_bits;
>+} G722DecoderState;
>+
>+#ifdef __cplusplus
>+extern "C" {
>+#endif
>+
>+G722EncoderState* WebRtc_g722_encode_init(G722EncoderState* s,
>+                                          int rate,
>+                                          int options);
>+int WebRtc_g722_encode_release(G722EncoderState* s);
>+size_t WebRtc_g722_encode(G722EncoderState* s,
>+                          uint8_t g722_data[],
>+                          const int16_t amp[],
>+                          size_t len);
>+
>+G722DecoderState* WebRtc_g722_decode_init(G722DecoderState* s,
>+                                          int rate,
>+                                          int options);
>+int WebRtc_g722_decode_release(G722DecoderState* s);
>+size_t WebRtc_g722_decode(G722DecoderState* s,
>+                          int16_t amp[],
>+                          const uint8_t g722_data[],
>+                          size_t len);
>+
>+#ifdef __cplusplus
>+}
>+#endif
>+
>+#endif /* MODULES_THIRD_PARTY_G722_G722_H_ */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/g722_encode.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/g722/g722_encode.c
>similarity index 99%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/g722_encode.c
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/g722/g722_encode.c
>index ed4b1c85eeb..10a5bcfe7c0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/g722/g722_encode.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/g722/g722_encode.c
>@@ -34,8 +34,7 @@
> #include <stdio.h>
> #include <stdlib.h>
> 
>-#include "g722_enc_dec.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include "modules/third_party/g722/g722_enc_dec.h"
> 
> #if !defined(FALSE)
> #define FALSE 0
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/portaudio/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/portaudio/BUILD.gn
>new file mode 100644
>index 00000000000..1f701dbcd87
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/portaudio/BUILD.gn
>@@ -0,0 +1,18 @@
>+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+#
>+# Use of this source code is governed by a BSD-style license
>+# that can be found in the ../../../LICENSE file in the root of the source
>+# tree. An additional intellectual property rights grant can be found
>+# in the file PATENTS.  All contributing project authors may
>+# be found in the AUTHORS file in the root of the source tree.
>+
>+import("../../../webrtc.gni")
>+
>+rtc_source_set("mac_portaudio") {
>+  visibility = [ "../../audio_device:*" ]
>+  sources = [
>+    "pa_memorybarrier.h",
>+    "pa_ringbuffer.c",
>+    "pa_ringbuffer.h",
>+  ]
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/portaudio/LICENSE b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/portaudio/LICENSE
>new file mode 100644
>index 00000000000..6ccaca29107
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/portaudio/LICENSE
>@@ -0,0 +1,91 @@
>+/*
>+ * $Id: pa_memorybarrier.h 1240 2007-07-17 13:05:07Z bjornroche $
>+ * Portable Audio I/O Library
>+ * Memory barrier utilities
>+ *
>+ * Author: Bjorn Roche, XO Audio, LLC
>+ *
>+ * This program uses the PortAudio Portable Audio Library.
>+ * For more information see: http://www.portaudio.com
>+ * Copyright (c) 1999-2000 Ross Bencina and Phil Burk
>+ *
>+ * Permission is hereby granted, free of charge, to any person obtaining
>+ * a copy of this software and associated documentation files
>+ * (the "Software"), to deal in the Software without restriction,
>+ * including without limitation the rights to use, copy, modify, merge,
>+ * publish, distribute, sublicense, and/or sell copies of the Software,
>+ * and to permit persons to whom the Software is furnished to do so,
>+ * subject to the following conditions:
>+ *
>+ * The above copyright notice and this permission notice shall be
>+ * included in all copies or substantial portions of the Software.
>+ *
>+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
>+ * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
>+ * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
>+ * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
>+ * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
>+ * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
>+ * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
>+ */
>+
>+/*
>+ * The text above constitutes the entire PortAudio license; however,
>+ * the PortAudio community also makes the following non-binding requests:
>+ *
>+ * Any person wishing to distribute modifications to the Software is
>+ * requested to send the modifications to the original developer so that
>+ * they can be incorporated into the canonical version. It is also
>+ * requested that these non-binding requests be included along with the
>+ * license above.
>+ */
>+
>+/*
>+ * $Id: pa_ringbuffer.c 1421 2009-11-18 16:09:05Z bjornroche $
>+ * Portable Audio I/O Library
>+ * Ring Buffer utility.
>+ *
>+ * Author: Phil Burk, http://www.softsynth.com
>+ * modified for SMP safety on Mac OS X by Bjorn Roche
>+ * modified for SMP safety on Linux by Leland Lucius
>+ * also, allowed for const where possible
>+ * modified for multiple-byte-sized data elements by Sven Fischer
>+ *
>+ * Note that this is safe only for a single-thread reader and a
>+ * single-thread writer.
>+ *
>+ * This program uses the PortAudio Portable Audio Library.
>+ * For more information see: http://www.portaudio.com
>+ * Copyright (c) 1999-2000 Ross Bencina and Phil Burk
>+ *
>+ * Permission is hereby granted, free of charge, to any person obtaining
>+ * a copy of this software and associated documentation files
>+ * (the "Software"), to deal in the Software without restriction,
>+ * including without limitation the rights to use, copy, modify, merge,
>+ * publish, distribute, sublicense, and/or sell copies of the Software,
>+ * and to permit persons to whom the Software is furnished to do so,
>+ * subject to the following conditions:
>+ *
>+ * The above copyright notice and this permission notice shall be
>+ * included in all copies or substantial portions of the Software.
>+ *
>+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
>+ * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
>+ * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
>+ * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
>+ * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
>+ * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
>+ * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
>+ */
>+
>+/*
>+ * The text above constitutes the entire PortAudio license; however,
>+ * the PortAudio community also makes the following non-binding requests:
>+ *
>+ * Any person wishing to distribute modifications to the Software is
>+ * requested to send the modifications to the original developer so that
>+ * they can be incorporated into the canonical version. It is also
>+ * requested that these non-binding requests be included along with the
>+ * license above.
>+ */
>+
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/portaudio/README.chromium b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/portaudio/README.chromium
>new file mode 100644
>index 00000000000..7a15b2e1fca
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/portaudio/README.chromium
>@@ -0,0 +1,12 @@
>+Name: Portaudio library for mac
>+Short Name: portaudio
>+URL: https://app.assembla.com/spaces/portaudio/git/source/master/src/common
>+Version: 0
>+Date: 2018-02-01
>+License: Custom license
>+License File: LICENSE
>+Security Critical: yes
>+
>+Description:
>+Part of portaudio library to operate with memory barriers and ring buffer.
>+
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/portaudio/pa_memorybarrier.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/portaudio/pa_memorybarrier.h
>new file mode 100644
>index 00000000000..c1040d1027d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/portaudio/pa_memorybarrier.h
>@@ -0,0 +1,132 @@
>+/*
>+ * $Id: pa_memorybarrier.h 1240 2007-07-17 13:05:07Z bjornroche $
>+ * Portable Audio I/O Library
>+ * Memory barrier utilities
>+ *
>+ * Author: Bjorn Roche, XO Audio, LLC
>+ *
>+ * This program uses the PortAudio Portable Audio Library.
>+ * For more information see: http://www.portaudio.com
>+ * Copyright (c) 1999-2000 Ross Bencina and Phil Burk
>+ *
>+ * Permission is hereby granted, free of charge, to any person obtaining
>+ * a copy of this software and associated documentation files
>+ * (the "Software"), to deal in the Software without restriction,
>+ * including without limitation the rights to use, copy, modify, merge,
>+ * publish, distribute, sublicense, and/or sell copies of the Software,
>+ * and to permit persons to whom the Software is furnished to do so,
>+ * subject to the following conditions:
>+ *
>+ * The above copyright notice and this permission notice shall be
>+ * included in all copies or substantial portions of the Software.
>+ *
>+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
>+ * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
>+ * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
>+ * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
>+ * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
>+ * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
>+ * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
>+ */
>+
>+/*
>+ * The text above constitutes the entire PortAudio license; however,
>+ * the PortAudio community also makes the following non-binding requests:
>+ *
>+ * Any person wishing to distribute modifications to the Software is
>+ * requested to send the modifications to the original developer so that
>+ * they can be incorporated into the canonical version. It is also
>+ * requested that these non-binding requests be included along with the
>+ * license above.
>+ */
>+
>+/**
>+ @file pa_memorybarrier.h
>+ @ingroup common_src
>+*/
>+
>+/****************
>+ * Some memory barrier primitives based on the system.
>+ * right now only OS X, FreeBSD, and Linux are supported. In addition to
>+ *providing memory barriers, these functions should ensure that data cached in
>+ *registers is written out to cache where it can be snooped by other CPUs. (ie,
>+ *the volatile keyword should not be required)
>+ *
>+ * the primitives that must be defined are:
>+ *
>+ * PaUtil_FullMemoryBarrier()
>+ * PaUtil_ReadMemoryBarrier()
>+ * PaUtil_WriteMemoryBarrier()
>+ *
>+ ****************/
>+
>+#ifndef MODULES_THIRD_PARTY_PORTAUDIO_PA_MEMORYBARRIER_H_
>+#define MODULES_THIRD_PARTY_PORTAUDIO_PA_MEMORYBARRIER_H_
>+
>+#if defined(__APPLE__)
>+#include <libkern/OSAtomic.h>
>+/* Here are the memory barrier functions. Mac OS X only provides
>+   full memory barriers, so the three types of barriers are the same,
>+   however, these barriers are superior to compiler-based ones. */
>+#define PaUtil_FullMemoryBarrier() OSMemoryBarrier()
>+#define PaUtil_ReadMemoryBarrier() OSMemoryBarrier()
>+#define PaUtil_WriteMemoryBarrier() OSMemoryBarrier()
>+#elif defined(__GNUC__)
>+/* GCC >= 4.1 has built-in intrinsics. We'll use those */
>+#if (__GNUC__ > 4) || (__GNUC__ == 4 && __GNUC_MINOR__ >= 1)
>+#define PaUtil_FullMemoryBarrier() __sync_synchronize()
>+#define PaUtil_ReadMemoryBarrier() __sync_synchronize()
>+#define PaUtil_WriteMemoryBarrier() __sync_synchronize()
>+/* as a fallback, GCC understands volatile asm and "memory" to mean it
>+ * should not reorder memory read/writes */
>+/* Note that it is not clear that any compiler actually defines __PPC__,
>+ * it can probably removed safely. */
>+#elif defined(__ppc__) || defined(__powerpc__) || defined(__PPC__)
>+#define PaUtil_FullMemoryBarrier() asm volatile("sync" ::: "memory")
>+#define PaUtil_ReadMemoryBarrier() asm volatile("sync" ::: "memory")
>+#define PaUtil_WriteMemoryBarrier() asm volatile("sync" ::: "memory")
>+#elif defined(__i386__) || defined(__i486__) || defined(__i586__) || \
>+    defined(__i686__) || defined(__x86_64__)
>+#define PaUtil_FullMemoryBarrier() asm volatile("mfence" ::: "memory")
>+#define PaUtil_ReadMemoryBarrier() asm volatile("lfence" ::: "memory")
>+#define PaUtil_WriteMemoryBarrier() asm volatile("sfence" ::: "memory")
>+#else
>+#ifdef ALLOW_SMP_DANGERS
>+#warning Memory barriers not defined on this system or system unknown
>+#warning For SMP safety, you should fix this.
>+#define PaUtil_FullMemoryBarrier()
>+#define PaUtil_ReadMemoryBarrier()
>+#define PaUtil_WriteMemoryBarrier()
>+#else
>+#         error Memory barriers are not defined on this system. You can still compile by defining ALLOW_SMP_DANGERS, but SMP safety will not be guaranteed.
>+#endif
>+#endif
>+#elif (_MSC_VER >= 1400) && !defined(_WIN32_WCE)
>+#include <intrin.h>
>+#pragma intrinsic(_ReadWriteBarrier)
>+#pragma intrinsic(_ReadBarrier)
>+#pragma intrinsic(_WriteBarrier)
>+#define PaUtil_FullMemoryBarrier() _ReadWriteBarrier()
>+#define PaUtil_ReadMemoryBarrier() _ReadBarrier()
>+#define PaUtil_WriteMemoryBarrier() _WriteBarrier()
>+#elif defined(_WIN32_WCE)
>+#define PaUtil_FullMemoryBarrier()
>+#define PaUtil_ReadMemoryBarrier()
>+#define PaUtil_WriteMemoryBarrier()
>+#elif defined(_MSC_VER) || defined(__BORLANDC__)
>+#define PaUtil_FullMemoryBarrier() _asm { lock add    [esp], 0}
>+#define PaUtil_ReadMemoryBarrier() _asm { lock add    [esp], 0}
>+#define PaUtil_WriteMemoryBarrier() _asm { lock add    [esp], 0}
>+#else
>+#ifdef ALLOW_SMP_DANGERS
>+#warning Memory barriers not defined on this system or system unknown
>+#warning For SMP safety, you should fix this.
>+#define PaUtil_FullMemoryBarrier()
>+#define PaUtil_ReadMemoryBarrier()
>+#define PaUtil_WriteMemoryBarrier()
>+#else
>+#      error Memory barriers are not defined on this system. You can still compile by defining ALLOW_SMP_DANGERS, but SMP safety will not be guaranteed.
>+#endif
>+#endif
>+
>+#endif /* MODULES_THIRD_PARTY_PORTAUDIO_PA_MEMORYBARRIER_H_ */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/mac/portaudio/pa_ringbuffer.c b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/portaudio/pa_ringbuffer.c
>similarity index 98%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/mac/portaudio/pa_ringbuffer.c
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/portaudio/pa_ringbuffer.c
>index 7a95024757b..fc1053c5ffc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/mac/portaudio/pa_ringbuffer.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/portaudio/pa_ringbuffer.c
>@@ -56,8 +56,8 @@
> #include <stdio.h>
> #include <stdlib.h>
> #include <string.h>
>-#include "modules/audio_device/mac/portaudio/pa_memorybarrier.h"
>-#include "modules/audio_device/mac/portaudio/pa_ringbuffer.h"
>+#include "modules/third_party/portaudio/pa_memorybarrier.h"
>+#include "modules/third_party/portaudio/pa_ringbuffer.h"
> 
> /***************************************************************************
>  * Initialize FIFO.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/mac/portaudio/pa_ringbuffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/portaudio/pa_ringbuffer.h
>similarity index 66%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/mac/portaudio/pa_ringbuffer.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/portaudio/pa_ringbuffer.h
>index 210a65e4d87..3ad821664c6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/mac/portaudio/pa_ringbuffer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/third_party/portaudio/pa_ringbuffer.h
>@@ -1,5 +1,5 @@
>-#ifndef AUDIO_DEVICE_PA_RINGBUFFER_H_
>-#define AUDIO_DEVICE_PA_RINGBUFFER_H_
>+#ifndef MODULES_THIRD_PARTY_PORTAUDIO_PA_RINGBUFFER_H_
>+#define MODULES_THIRD_PARTY_PORTAUDIO_PA_RINGBUFFER_H_
> /*
>  * $Id: pa_ringbuffer.h 1421 2009-11-18 16:09:05Z bjornroche $
>  * Portable Audio I/O Library
>@@ -8,7 +8,7 @@
>  * Author: Phil Burk, http://www.softsynth.com
>  * modified for SMP safety on OS X by Bjorn Roche.
>  * also allowed for const where possible.
>- * modified for multiple-byte-sized data elements by Sven Fischer 
>+ * modified for multiple-byte-sized data elements by Sven Fischer
>  *
>  * Note that this is safe only for a single-thread reader
>  * and a single-thread writer.
>@@ -38,13 +38,13 @@
>  */
> 
> /*
>- * The text above constitutes the entire PortAudio license; however, 
>+ * The text above constitutes the entire PortAudio license; however,
>  * the PortAudio community also makes the following non-binding requests:
>  *
>  * Any person wishing to distribute modifications to the Software is
>  * requested to send the modifications to the original developer so that
>- * they can be incorporated into the canonical version. It is also 
>- * requested that these non-binding requests be included along with the 
>+ * they can be incorporated into the canonical version. It is also
>+ * requested that these non-binding requests be included along with the
>  * license above.
>  */
> 
>@@ -58,11 +58,11 @@
>  a single reader and a single writer (ie. one thread or callback writes
>  to the ring buffer, another thread or callback reads from it).
> 
>- The PaUtilRingBuffer structure manages a ring buffer containing N 
>- elements, where N must be a power of two. An element may be any size 
>+ The PaUtilRingBuffer structure manages a ring buffer containing N
>+ elements, where N must be a power of two. An element may be any size
>  (specified in bytes).
> 
>- The memory area used to store the buffer elements must be allocated by 
>+ The memory area used to store the buffer elements must be allocated by
>  the client prior to calling PaUtil_InitializeRingBuffer() and must outlive
>  the use of the ring buffer.
> */
>@@ -70,7 +70,7 @@
> #if defined(__APPLE__)
> #include <sys/types.h>
> typedef int32_t PaRingBufferSize;
>-#elif defined( __GNUC__ )
>+#elif defined(__GNUC__)
> typedef long PaRingBufferSize;
> #elif (_MSC_VER >= 1400)
> typedef long PaRingBufferSize;
>@@ -80,23 +80,24 @@ typedef long PaRingBufferSize;
> typedef long PaRingBufferSize;
> #endif
> 
>-
>-
> #ifdef __cplusplus
>-extern "C"
>-{
>+extern "C" {
> #endif /* __cplusplus */
> 
>-typedef struct PaUtilRingBuffer
>-{
>-    PaRingBufferSize  bufferSize; /**< Number of elements in FIFO. Power of 2. Set by PaUtil_InitRingBuffer. */
>-    PaRingBufferSize  writeIndex; /**< Index of next writable element. Set by PaUtil_AdvanceRingBufferWriteIndex. */
>-    PaRingBufferSize  readIndex;  /**< Index of next readable element. Set by PaUtil_AdvanceRingBufferReadIndex. */
>-    PaRingBufferSize  bigMask;    /**< Used for wrapping indices with extra bit to distinguish full/empty. */
>-    PaRingBufferSize  smallMask;  /**< Used for fitting indices to buffer. */
>-    PaRingBufferSize  elementSizeBytes; /**< Number of bytes per element. */
>-    char  *buffer;    /**< Pointer to the buffer containing the actual data. */
>-}PaUtilRingBuffer;
>+typedef struct PaUtilRingBuffer {
>+  PaRingBufferSize
>+      bufferSize;                    /**< Number of elements in FIFO. Power of 2. Set by
>+                                        PaUtil_InitRingBuffer. */
>+  PaRingBufferSize writeIndex;       /**< Index of next writable element. Set by
>+                                        PaUtil_AdvanceRingBufferWriteIndex. */
>+  PaRingBufferSize readIndex;        /**< Index of next readable element. Set by
>+                                        PaUtil_AdvanceRingBufferReadIndex. */
>+  PaRingBufferSize bigMask;          /**< Used for wrapping indices with extra bit to
>+                                        distinguish full/empty. */
>+  PaRingBufferSize smallMask;        /**< Used for fitting indices to buffer. */
>+  PaRingBufferSize elementSizeBytes; /**< Number of bytes per element. */
>+  char* buffer; /**< Pointer to the buffer containing the actual data. */
>+} PaUtilRingBuffer;
> 
> /** Initialize Ring Buffer.
> 
>@@ -111,13 +112,16 @@ typedef struct PaUtilRingBuffer
> 
>  @return -1 if elementCount is not a power of 2, otherwise 0.
> */
>-PaRingBufferSize PaUtil_InitializeRingBuffer( PaUtilRingBuffer *rbuf, PaRingBufferSize elementSizeBytes, PaRingBufferSize elementCount, void *dataPtr );
>+PaRingBufferSize PaUtil_InitializeRingBuffer(PaUtilRingBuffer* rbuf,
>+                                             PaRingBufferSize elementSizeBytes,
>+                                             PaRingBufferSize elementCount,
>+                                             void* dataPtr);
> 
> /** Clear buffer. Should only be called when buffer is NOT being read.
> 
>  @param rbuf The ring buffer.
> */
>-void PaUtil_FlushRingBuffer( PaUtilRingBuffer *rbuf );
>+void PaUtil_FlushRingBuffer(PaUtilRingBuffer* rbuf);
> 
> /** Retrieve the number of elements available in the ring buffer for writing.
> 
>@@ -125,7 +129,7 @@ void PaUtil_FlushRingBuffer( PaUtilRingBuffer *rbuf );
> 
>  @return The number of elements available for writing.
> */
>-PaRingBufferSize PaUtil_GetRingBufferWriteAvailable( PaUtilRingBuffer *rbuf );
>+PaRingBufferSize PaUtil_GetRingBufferWriteAvailable(PaUtilRingBuffer* rbuf);
> 
> /** Retrieve the number of elements available in the ring buffer for reading.
> 
>@@ -133,7 +137,7 @@ PaRingBufferSize PaUtil_GetRingBufferWriteAvailable( PaUtilRingBuffer *rbuf );
> 
>  @return The number of elements available for reading.
> */
>-PaRingBufferSize PaUtil_GetRingBufferReadAvailable( PaUtilRingBuffer *rbuf );
>+PaRingBufferSize PaUtil_GetRingBufferReadAvailable(PaUtilRingBuffer* rbuf);
> 
> /** Write data to the ring buffer.
> 
>@@ -145,7 +149,9 @@ PaRingBufferSize PaUtil_GetRingBufferReadAvailable( PaUtilRingBuffer *rbuf );
> 
>  @return The number of elements written.
> */
>-PaRingBufferSize PaUtil_WriteRingBuffer( PaUtilRingBuffer *rbuf, const void *data, PaRingBufferSize elementCount );
>+PaRingBufferSize PaUtil_WriteRingBuffer(PaUtilRingBuffer* rbuf,
>+                                        const void* data,
>+                                        PaRingBufferSize elementCount);
> 
> /** Read data from the ring buffer.
> 
>@@ -157,7 +163,9 @@ PaRingBufferSize PaUtil_WriteRingBuffer( PaUtilRingBuffer *rbuf, const void *dat
> 
>  @return The number of elements read.
> */
>-PaRingBufferSize PaUtil_ReadRingBuffer( PaUtilRingBuffer *rbuf, void *data, PaRingBufferSize elementCount );
>+PaRingBufferSize PaUtil_ReadRingBuffer(PaUtilRingBuffer* rbuf,
>+                                       void* data,
>+                                       PaRingBufferSize elementCount);
> 
> /** Get address of region(s) to which we can write data.
> 
>@@ -179,9 +187,12 @@ PaRingBufferSize PaUtil_ReadRingBuffer( PaUtilRingBuffer *rbuf, void *data, PaRi
> 
>  @return The room available to be written or elementCount, whichever is smaller.
> */
>-PaRingBufferSize PaUtil_GetRingBufferWriteRegions( PaUtilRingBuffer *rbuf, PaRingBufferSize elementCount,
>-                                       void **dataPtr1, PaRingBufferSize *sizePtr1,
>-                                       void **dataPtr2, PaRingBufferSize *sizePtr2 );
>+PaRingBufferSize PaUtil_GetRingBufferWriteRegions(PaUtilRingBuffer* rbuf,
>+                                                  PaRingBufferSize elementCount,
>+                                                  void** dataPtr1,
>+                                                  PaRingBufferSize* sizePtr1,
>+                                                  void** dataPtr2,
>+                                                  PaRingBufferSize* sizePtr2);
> 
> /** Advance the write index to the next location to be written.
> 
>@@ -191,7 +202,9 @@ PaRingBufferSize PaUtil_GetRingBufferWriteRegions( PaUtilRingBuffer *rbuf, PaRin
> 
>  @return The new position.
> */
>-PaRingBufferSize PaUtil_AdvanceRingBufferWriteIndex( PaUtilRingBuffer *rbuf, PaRingBufferSize elementCount );
>+PaRingBufferSize PaUtil_AdvanceRingBufferWriteIndex(
>+    PaUtilRingBuffer* rbuf,
>+    PaRingBufferSize elementCount);
> 
> /** Get address of region(s) from which we can write data.
> 
>@@ -213,9 +226,12 @@ PaRingBufferSize PaUtil_AdvanceRingBufferWriteIndex( PaUtilRingBuffer *rbuf, PaR
> 
>  @return The number of elements available for reading.
> */
>-PaRingBufferSize PaUtil_GetRingBufferReadRegions( PaUtilRingBuffer *rbuf, PaRingBufferSize elementCount,
>-                                      void **dataPtr1, PaRingBufferSize *sizePtr1,
>-                                      void **dataPtr2, PaRingBufferSize *sizePtr2 );
>+PaRingBufferSize PaUtil_GetRingBufferReadRegions(PaUtilRingBuffer* rbuf,
>+                                                 PaRingBufferSize elementCount,
>+                                                 void** dataPtr1,
>+                                                 PaRingBufferSize* sizePtr1,
>+                                                 void** dataPtr2,
>+                                                 PaRingBufferSize* sizePtr2);
> 
> /** Advance the read index to the next location to be read.
> 
>@@ -225,9 +241,11 @@ PaRingBufferSize PaUtil_GetRingBufferReadRegions( PaUtilRingBuffer *rbuf, PaRing
> 
>  @return The new position.
> */
>-PaRingBufferSize PaUtil_AdvanceRingBufferReadIndex( PaUtilRingBuffer *rbuf, PaRingBufferSize elementCount );
>+PaRingBufferSize PaUtil_AdvanceRingBufferReadIndex(
>+    PaUtilRingBuffer* rbuf,
>+    PaRingBufferSize elementCount);
> 
> #ifdef __cplusplus
> }
> #endif /* __cplusplus */
>-#endif /* MODULES_AUDIO_DEVICE_MAIN_SOURCE_MAC_PORTAUDIO_PA_RINGBUFFER_H_ */
>+#endif /* MODULES_THIRD_PARTY_PORTAUDIO_PA_RINGBUFFER_H_ */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/BUILD.gn
>index fbfa61ec168..32da10950f2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/BUILD.gn
>@@ -11,7 +11,6 @@ import("../../webrtc.gni")
> rtc_static_library("utility") {
>   visibility = [ "*" ]
>   sources = [
>-    "include/audio_frame_operations.h",
>     "include/helpers_android.h",
>     "include/jvm_android.h",
>     "include/process_thread.h",
>@@ -21,11 +20,6 @@ rtc_static_library("utility") {
>     "source/process_thread_impl.h",
>   ]
> 
>-  if (!build_with_chromium && is_clang) {
>-    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>-    suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-  }
>-
>   if (is_ios) {
>     libs = [ "AVFoundation.framework" ]
>   }
>@@ -33,14 +27,12 @@ rtc_static_library("utility") {
>   deps = [
>     "..:module_api",
>     "../..:webrtc_common",
>-    "../../:typedefs",
>-    "../../audio/utility:audio_frame_operations",
>     "../../common_audio",
>     "../../rtc_base:checks",
>     "../../rtc_base:rtc_base_approved",
>     "../../rtc_base:rtc_task_queue",
>+    "../../rtc_base/system:arch",
>     "../../system_wrappers",
>-    "../media_file",
>   ]
> }
> 
>@@ -70,7 +62,6 @@ if (rtc_include_tests) {
>       "../../rtc_base:rtc_base_approved",
>       "../../rtc_base:rtc_task_queue",
>       "../../test:test_support",
>-      "//testing/gmock",
>     ]
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/DEPS b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/DEPS
>index d693ea6221a..e9017d6b275 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/DEPS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/DEPS
>@@ -1,7 +1,4 @@
> include_rules = [
>-  # TODO(aleloi): remove when clients update. See
>-  # bugs.webrtc.org/6548.
>-  "+audio/utility/audio_frame_operations.h",
>   "+common_audio",
>   "+common_video",
>   "+system_wrappers",
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/include/audio_frame_operations.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/include/audio_frame_operations.h
>deleted file mode 100644
>index adaadfeca7e..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/include/audio_frame_operations.h
>+++ /dev/null
>@@ -1,20 +0,0 @@
>-/*
>- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_
>-#define MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_
>-// The contents of this file have moved to
>-// //audio/utility. This file is deprecated.
>-
>-// TODO(aleloi): Remove this file when clients have updated their
>-// includes. See bugs.webrtc.org/6548.
>-#include "audio/utility/audio_frame_operations.h"
>-
>-#endif  // #ifndef MODULES_UTILITY_INCLUDE_AUDIO_FRAME_OPERATIONS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/include/helpers_android.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/include/helpers_android.h
>index 2ee577ceeae..763a0c37cfd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/include/helpers_android.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/include/helpers_android.h
>@@ -14,12 +14,23 @@
> #include <jni.h>
> #include <string>
> 
>+#include "rtc_base/system/arch.h"
>+
> // Abort the process if |jni| has a Java exception pending.
> // TODO(henrika): merge with CHECK_JNI_EXCEPTION() in jni_helpers.h.
> #define CHECK_EXCEPTION(jni)        \
>   RTC_CHECK(!jni->ExceptionCheck()) \
>       << (jni->ExceptionDescribe(), jni->ExceptionClear(), "")
> 
>+#if defined(WEBRTC_ARCH_X86)
>+// Dalvik JIT generated code doesn't guarantee 16-byte stack alignment on
>+// x86 - use force_align_arg_pointer to realign the stack at the JNI
>+// boundary. bugs.webrtc.org/9050
>+#define JNI_FUNCTION_ALIGN __attribute__((force_align_arg_pointer))
>+#else
>+#define JNI_FUNCTION_ALIGN
>+#endif
>+
> namespace webrtc {
> 
> // Return a |JNIEnv*| usable on this thread or NULL if this thread is detached.
>@@ -33,11 +44,15 @@ jlong PointerTojlong(void* ptr);
> // JNIEnv-helper methods that wraps the API which uses the JNI interface
> // pointer (JNIEnv*). It allows us to RTC_CHECK success and that no Java
> // exception is thrown while calling the method.
>-jmethodID GetMethodID(
>-    JNIEnv* jni, jclass c, const char* name, const char* signature);
>+jmethodID GetMethodID(JNIEnv* jni,
>+                      jclass c,
>+                      const char* name,
>+                      const char* signature);
> 
>-jmethodID GetStaticMethodID(
>-    JNIEnv* jni, jclass c, const char* name, const char* signature);
>+jmethodID GetStaticMethodID(JNIEnv* jni,
>+                            jclass c,
>+                            const char* name,
>+                            const char* signature);
> 
> jclass FindClass(JNIEnv* jni, const char* name);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/include/jvm_android.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/include/jvm_android.h
>index f899263f76a..eb18cc27f3c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/include/jvm_android.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/include/jvm_android.h
>@@ -78,8 +78,9 @@ class NativeRegistration : public JavaClass {
>   NativeRegistration(JNIEnv* jni, jclass clazz);
>   ~NativeRegistration();
> 
>-  std::unique_ptr<GlobalRef> NewObject(
>-      const char* name, const char* signature, ...);
>+  std::unique_ptr<GlobalRef> NewObject(const char* name,
>+                                       const char* signature,
>+                                       ...);
> 
>  private:
>   JNIEnv* const jni_;
>@@ -99,7 +100,9 @@ class JNIEnvironment {
>   // |loaded_classes| array defined in jvm_android.cc.
>   // This method must be called on the construction thread.
>   std::unique_ptr<NativeRegistration> RegisterNatives(
>-      const char* name, const JNINativeMethod *methods, int num_methods);
>+      const char* name,
>+      const JNINativeMethod* methods,
>+      int num_methods);
> 
>   // Converts from Java string to std::string.
>   // This method must be called on the construction thread.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/include/process_thread.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/include/process_thread.h
>index 28e11910850..8c6919660aa 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/include/process_thread.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/include/process_thread.h
>@@ -13,8 +13,6 @@
> 
> #include <memory>
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>-
> #if defined(WEBRTC_WIN)
> // Due to a bug in the std::unique_ptr implementation that ships with MSVS,
> // we need the full definition of QueuedTask, on Windows.
>@@ -74,4 +72,4 @@ class ProcessThread {
> 
> }  // namespace webrtc
> 
>-#endif // MODULES_UTILITY_INCLUDE_PROCESS_THREAD_H_
>+#endif  // MODULES_UTILITY_INCLUDE_PROCESS_THREAD_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/source/helpers_android.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/source/helpers_android.cc
>index 5e760b4c03c..0206d152744 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/source/helpers_android.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/source/helpers_android.cc
>@@ -47,8 +47,10 @@ jlong PointerTojlong(void* ptr) {
>   return ret;
> }
> 
>-jmethodID GetMethodID (
>-    JNIEnv* jni, jclass c, const char* name, const char* signature) {
>+jmethodID GetMethodID(JNIEnv* jni,
>+                      jclass c,
>+                      const char* name,
>+                      const char* signature) {
>   jmethodID m = jni->GetMethodID(c, name, signature);
>   CHECK_EXCEPTION(jni) << "Error during GetMethodID: " << name << ", "
>                        << signature;
>@@ -56,8 +58,10 @@ jmethodID GetMethodID (
>   return m;
> }
> 
>-jmethodID GetStaticMethodID (
>-    JNIEnv* jni, jclass c, const char* name, const char* signature) {
>+jmethodID GetStaticMethodID(JNIEnv* jni,
>+                            jclass c,
>+                            const char* name,
>+                            const char* signature) {
>   jmethodID m = jni->GetStaticMethodID(c, name, signature);
>   CHECK_EXCEPTION(jni) << "Error during GetStaticMethodID: " << name << ", "
>                        << signature;
>@@ -107,6 +111,8 @@ AttachThreadScoped::~AttachThreadScoped() {
>   }
> }
> 
>-JNIEnv* AttachThreadScoped::env() { return env_; }
>+JNIEnv* AttachThreadScoped::env() {
>+  return env_;
>+}
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/source/jvm_android.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/source/jvm_android.cc
>index fc0739caf86..2d48c2d7be9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/source/jvm_android.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/source/jvm_android.cc
>@@ -15,12 +15,9 @@
> #include "modules/utility/include/jvm_android.h"
> 
> #include "rtc_base/checks.h"
>+#include "rtc_base/logging.h"
> #include "rtc_base/platform_thread.h"
> 
>-#define TAG "JVM"
>-#define ALOGD(...) __android_log_print(ANDROID_LOG_DEBUG, TAG, __VA_ARGS__)
>-#define ALOGE(...) __android_log_print(ANDROID_LOG_ERROR, TAG, __VA_ARGS__)
>-
> namespace webrtc {
> 
> JVM* g_jvm;
>@@ -30,10 +27,10 @@ struct {
>   const char* name;
>   jclass clazz;
> } loaded_classes[] = {
>-  {"org/webrtc/voiceengine/BuildInfo", nullptr},
>-  {"org/webrtc/voiceengine/WebRtcAudioManager", nullptr},
>-  {"org/webrtc/voiceengine/WebRtcAudioRecord", nullptr},
>-  {"org/webrtc/voiceengine/WebRtcAudioTrack", nullptr},
>+    {"org/webrtc/voiceengine/BuildInfo", nullptr},
>+    {"org/webrtc/voiceengine/WebRtcAudioManager", nullptr},
>+    {"org/webrtc/voiceengine/WebRtcAudioRecord", nullptr},
>+    {"org/webrtc/voiceengine/WebRtcAudioTrack", nullptr},
> };
> 
> // Android's FindClass() is trickier than usual because the app-specific
>@@ -41,10 +38,10 @@ struct {
> // stack.  Consequently, we only look up all classes once in native WebRTC.
> // http://developer.android.com/training/articles/perf-jni.html#faq_FindClass
> void LoadClasses(JNIEnv* jni) {
>-  ALOGD("LoadClasses");
>+  RTC_LOG(INFO) << "LoadClasses:";
>   for (auto& c : loaded_classes) {
>     jclass localRef = FindClass(jni, c.name);
>-    ALOGD("name: %s", c.name);
>+    RTC_LOG(INFO) << "name: " << c.name;
>     CHECK_EXCEPTION(jni) << "Error during FindClass: " << c.name;
>     RTC_CHECK(localRef) << c.name;
>     jclass globalRef = reinterpret_cast<jclass>(jni->NewGlobalRef(localRef));
>@@ -71,14 +68,13 @@ jclass LookUpClass(const char* name) {
> }
> 
> // AttachCurrentThreadIfNeeded implementation.
>-AttachCurrentThreadIfNeeded::AttachCurrentThreadIfNeeded()
>-    : attached_(false) {
>-  ALOGD("AttachCurrentThreadIfNeeded::ctor[tid=%d]", rtc::CurrentThreadId());
>+AttachCurrentThreadIfNeeded::AttachCurrentThreadIfNeeded() : attached_(false) {
>+  RTC_LOG(INFO) << "AttachCurrentThreadIfNeeded::ctor";
>   JavaVM* jvm = JVM::GetInstance()->jvm();
>   RTC_CHECK(jvm);
>   JNIEnv* jni = GetEnv(jvm);
>   if (!jni) {
>-    ALOGD("Attaching thread to JVM");
>+    RTC_LOG(INFO) << "Attaching thread to JVM";
>     JNIEnv* env = nullptr;
>     jint ret = jvm->AttachCurrentThread(&env, nullptr);
>     attached_ = (ret == JNI_OK);
>@@ -86,10 +82,10 @@ AttachCurrentThreadIfNeeded::AttachCurrentThreadIfNeeded()
> }
> 
> AttachCurrentThreadIfNeeded::~AttachCurrentThreadIfNeeded() {
>-  ALOGD("AttachCurrentThreadIfNeeded::dtor[tid=%d]", rtc::CurrentThreadId());
>+  RTC_LOG(INFO) << "AttachCurrentThreadIfNeeded::dtor";
>   RTC_DCHECK(thread_checker_.CalledOnValidThread());
>   if (attached_) {
>-    ALOGD("Detaching thread from JVM");
>+    RTC_LOG(INFO) << "Detaching thread from JVM";
>     jint res = JVM::GetInstance()->jvm()->DetachCurrentThread();
>     RTC_CHECK(res == JNI_OK) << "DetachCurrentThread failed: " << res;
>   }
>@@ -98,11 +94,11 @@ AttachCurrentThreadIfNeeded::~AttachCurrentThreadIfNeeded() {
> // GlobalRef implementation.
> GlobalRef::GlobalRef(JNIEnv* jni, jobject object)
>     : jni_(jni), j_object_(NewGlobalRef(jni, object)) {
>-  ALOGD("GlobalRef::ctor[tid=%d]", rtc::CurrentThreadId());
>+  RTC_LOG(INFO) << "GlobalRef::ctor";
> }
> 
> GlobalRef::~GlobalRef() {
>-  ALOGD("GlobalRef::dtor[tid=%d]", rtc::CurrentThreadId());
>+  RTC_LOG(INFO) << "GlobalRef::dtor";
>   DeleteGlobalRef(jni_, j_object_);
> }
> 
>@@ -135,36 +131,35 @@ void GlobalRef::CallVoidMethod(jmethodID methodID, ...) {
> // NativeRegistration implementation.
> NativeRegistration::NativeRegistration(JNIEnv* jni, jclass clazz)
>     : JavaClass(jni, clazz), jni_(jni) {
>-  ALOGD("NativeRegistration::ctor[tid=%d]", rtc::CurrentThreadId());
>+  RTC_LOG(INFO) << "NativeRegistration::ctor";
> }
> 
> NativeRegistration::~NativeRegistration() {
>-  ALOGD("NativeRegistration::dtor[tid=%d]", rtc::CurrentThreadId());
>+  RTC_LOG(INFO) << "NativeRegistration::dtor";
>   jni_->UnregisterNatives(j_class_);
>   CHECK_EXCEPTION(jni_) << "Error during UnregisterNatives";
> }
> 
>-std::unique_ptr<GlobalRef> NativeRegistration::NewObject(
>-    const char* name, const char* signature, ...) {
>-  ALOGD("NativeRegistration::NewObject[tid=%d]", rtc::CurrentThreadId());
>+std::unique_ptr<GlobalRef> NativeRegistration::NewObject(const char* name,
>+                                                         const char* signature,
>+                                                         ...) {
>+  RTC_LOG(INFO) << "NativeRegistration::NewObject";
>   va_list args;
>   va_start(args, signature);
>-  jobject obj = jni_->NewObjectV(j_class_,
>-                                 GetMethodID(jni_, j_class_, name, signature),
>-                                 args);
>+  jobject obj = jni_->NewObjectV(
>+      j_class_, GetMethodID(jni_, j_class_, name, signature), args);
>   CHECK_EXCEPTION(jni_) << "Error during NewObjectV";
>   va_end(args);
>   return std::unique_ptr<GlobalRef>(new GlobalRef(jni_, obj));
> }
> 
> // JavaClass implementation.
>-jmethodID JavaClass::GetMethodId(
>-    const char* name, const char* signature) {
>+jmethodID JavaClass::GetMethodId(const char* name, const char* signature) {
>   return GetMethodID(jni_, j_class_, name, signature);
> }
> 
>-jmethodID JavaClass::GetStaticMethodId(
>-    const char* name, const char* signature) {
>+jmethodID JavaClass::GetStaticMethodId(const char* name,
>+                                       const char* signature) {
>   return GetStaticMethodID(jni_, j_class_, name, signature);
> }
> 
>@@ -186,17 +181,19 @@ jint JavaClass::CallStaticIntMethod(jmethodID methodID, ...) {
> 
> // JNIEnvironment implementation.
> JNIEnvironment::JNIEnvironment(JNIEnv* jni) : jni_(jni) {
>-  ALOGD("JNIEnvironment::ctor[tid=%d]", rtc::CurrentThreadId());
>+  RTC_LOG(INFO) << "JNIEnvironment::ctor";
> }
> 
> JNIEnvironment::~JNIEnvironment() {
>-  ALOGD("JNIEnvironment::dtor[tid=%d]", rtc::CurrentThreadId());
>+  RTC_LOG(INFO) << "JNIEnvironment::dtor";
>   RTC_DCHECK(thread_checker_.CalledOnValidThread());
> }
> 
> std::unique_ptr<NativeRegistration> JNIEnvironment::RegisterNatives(
>-    const char* name, const JNINativeMethod *methods, int num_methods) {
>-  ALOGD("JNIEnvironment::RegisterNatives(%s)", name);
>+    const char* name,
>+    const JNINativeMethod* methods,
>+    int num_methods) {
>+  RTC_LOG(INFO) << "JNIEnvironment::RegisterNatives: " << name;
>   RTC_DCHECK(thread_checker_.CalledOnValidThread());
>   jclass clazz = LookUpClass(name);
>   jni_->RegisterNatives(clazz, methods, num_methods);
>@@ -219,7 +216,7 @@ std::string JNIEnvironment::JavaToStdString(const jstring& j_string) {
> 
> // static
> void JVM::Initialize(JavaVM* jvm) {
>-  ALOGD("JVM::Initialize[tid=%d]", rtc::CurrentThreadId());
>+  RTC_LOG(INFO) << "JVM::Initialize";
>   RTC_CHECK(!g_jvm);
>   g_jvm = new JVM(jvm);
> }
>@@ -237,7 +234,7 @@ void JVM::Initialize(JavaVM* jvm, jobject context) {
> 
> // static
> void JVM::Uninitialize() {
>-  ALOGD("JVM::Uninitialize[tid=%d]", rtc::CurrentThreadId());
>+  RTC_LOG(INFO) << "JVM::Uninitialize";
>   RTC_DCHECK(g_jvm);
>   delete g_jvm;
>   g_jvm = nullptr;
>@@ -250,19 +247,20 @@ JVM* JVM::GetInstance() {
> }
> 
> JVM::JVM(JavaVM* jvm) : jvm_(jvm) {
>-  ALOGD("JVM::JVM[tid=%d]", rtc::CurrentThreadId());
>+  RTC_LOG(INFO) << "JVM::JVM";
>   RTC_CHECK(jni()) << "AttachCurrentThread() must be called on this thread.";
>   LoadClasses(jni());
> }
> 
> JVM::~JVM() {
>-  ALOGD("JVM::~JVM[tid=%d]", rtc::CurrentThreadId());
>+  RTC_LOG(INFO) << "JVM::~JVM";
>   RTC_DCHECK(thread_checker_.CalledOnValidThread());
>   FreeClassReferences(jni());
> }
> 
> std::unique_ptr<JNIEnvironment> JVM::environment() {
>-  ALOGD("JVM::environment[tid=%d]", rtc::CurrentThreadId());
>+  RTC_LOG(INFO) << "JVM::environment";
>+  ;
>   // The JNIEnv is used for thread-local storage. For this reason, we cannot
>   // share a JNIEnv between threads. If a piece of code has no other way to get
>   // its JNIEnv, we should share the JavaVM, and use GetEnv to discover the
>@@ -270,14 +268,15 @@ std::unique_ptr<JNIEnvironment> JVM::environment() {
>   // See // http://developer.android.com/training/articles/perf-jni.html.
>   JNIEnv* jni = GetEnv(jvm_);
>   if (!jni) {
>-    ALOGE("AttachCurrentThread() has not been called on this thread.");
>+    RTC_LOG(LS_ERROR)
>+        << "AttachCurrentThread() has not been called on this thread";
>     return std::unique_ptr<JNIEnvironment>();
>   }
>   return std::unique_ptr<JNIEnvironment>(new JNIEnvironment(jni));
> }
> 
> JavaClass JVM::GetClass(const char* name) {
>-  ALOGD("JVM::GetClass(%s)[tid=%d]", name, rtc::CurrentThreadId());
>+  RTC_LOG(INFO) << "JVM::GetClass: " << name;
>   RTC_DCHECK(thread_checker_.CalledOnValidThread());
>   return JavaClass(jni(), LookUpClass(name));
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/source/process_thread_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/source/process_thread_impl.cc
>index 014624e48cb..97272da2bf1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/source/process_thread_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/source/process_thread_impl.cc
>@@ -32,13 +32,12 @@ int64_t GetNextCallbackTime(Module* module, int64_t time_now) {
>   }
>   return time_now + interval;
> }
>-}
>+}  // namespace
> 
> ProcessThread::~ProcessThread() {}
> 
> // static
>-std::unique_ptr<ProcessThread> ProcessThread::Create(
>-    const char* thread_name) {
>+std::unique_ptr<ProcessThread> ProcessThread::Create(const char* thread_name) {
>   return std::unique_ptr<ProcessThread>(new ProcessThreadImpl(thread_name));
> }
> 
>@@ -76,7 +75,7 @@ void ProcessThreadImpl::Start() {
> 
> void ProcessThreadImpl::Stop() {
>   RTC_DCHECK(thread_checker_.CalledOnValidThread());
>-  if(!thread_.get())
>+  if (!thread_.get())
>     return;
> 
>   {
>@@ -155,9 +154,8 @@ void ProcessThreadImpl::DeRegisterModule(Module* module) {
> 
>   {
>     rtc::CritScope lock(&lock_);
>-    modules_.remove_if([&module](const ModuleCallback& m) {
>-        return m.module == module;
>-      });
>+    modules_.remove_if(
>+        [&module](const ModuleCallback& m) { return m.module == module; });
>   }
> 
>   // Notify the module that it's been detached.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/source/process_thread_impl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/source/process_thread_impl.h
>index 9856fba5766..c1f6ed44475 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/source/process_thread_impl.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/source/process_thread_impl.h
>@@ -21,7 +21,6 @@
> #include "rtc_base/platform_thread.h"
> #include "rtc_base/thread_checker.h"
> #include "system_wrappers/include/event_wrapper.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -85,4 +84,4 @@ class ProcessThreadImpl : public ProcessThread {
> 
> }  // namespace webrtc
> 
>-#endif // MODULES_UTILITY_SOURCE_PROCESS_THREAD_IMPL_H_
>+#endif  // MODULES_UTILITY_SOURCE_PROCESS_THREAD_IMPL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/source/process_thread_impl_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/source/process_thread_impl_unittest.cc
>index d27896410ed..d5926b26c3e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/source/process_thread_impl_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/utility/source/process_thread_impl_unittest.cc
>@@ -137,9 +137,8 @@ TEST(ProcessThreadImpl, Deregister) {
>       .WillOnce(Return(0))
>       .WillRepeatedly(Return(1));
>   EXPECT_CALL(module, Process())
>-      .WillOnce(DoAll(SetEvent(event.get()),
>-                      Increment(&process_count),
>-                      Return()))
>+      .WillOnce(
>+          DoAll(SetEvent(event.get()), Increment(&process_count), Return()))
>       .WillRepeatedly(DoAll(Increment(&process_count), Return()));
> 
>   thread.RegisterModule(&module, RTC_FROM_HERE);
>@@ -174,13 +173,11 @@ void ProcessCallAfterAFewMs(int64_t milliseconds) {
>   int64_t start_time = 0;
>   int64_t called_time = 0;
>   EXPECT_CALL(module, TimeUntilNextProcess())
>-      .WillOnce(DoAll(SetTimestamp(&start_time),
>-                      Return(milliseconds)))
>+      .WillOnce(DoAll(SetTimestamp(&start_time), Return(milliseconds)))
>       .WillRepeatedly(Return(milliseconds));
>   EXPECT_CALL(module, Process())
>-      .WillOnce(DoAll(SetTimestamp(&called_time),
>-                      SetEvent(event.get()),
>-                      Return()))
>+      .WillOnce(
>+          DoAll(SetTimestamp(&called_time), SetEvent(event.get()), Return()))
>       .WillRepeatedly(Return());
> 
>   EXPECT_CALL(module, ProcessThreadAttached(&thread)).Times(1);
>@@ -238,11 +235,9 @@ TEST(ProcessThreadImpl, DISABLED_Process50Times) {
>   MockModule module;
>   int callback_count = 0;
>   // Ask for a callback after 20ms.
>-  EXPECT_CALL(module, TimeUntilNextProcess())
>-      .WillRepeatedly(Return(20));
>+  EXPECT_CALL(module, TimeUntilNextProcess()).WillRepeatedly(Return(20));
>   EXPECT_CALL(module, Process())
>-      .WillRepeatedly(DoAll(Increment(&callback_count),
>-                            Return()));
>+      .WillRepeatedly(DoAll(Increment(&callback_count), Return()));
> 
>   EXPECT_CALL(module, ProcessThreadAttached(&thread)).Times(1);
>   thread.RegisterModule(&module, RTC_FROM_HERE);
>@@ -281,8 +276,7 @@ TEST(ProcessThreadImpl, WakeUp) {
>   // The second time TimeUntilNextProcess is then called, is after Process
>   // has been called and we don't expect any more calls.
>   EXPECT_CALL(module, TimeUntilNextProcess())
>-      .WillOnce(DoAll(SetTimestamp(&start_time),
>-                      SetEvent(started.get()),
>+      .WillOnce(DoAll(SetTimestamp(&start_time), SetEvent(started.get()),
>                       Return(1000)))
>       .WillOnce(Return(1000));
>   EXPECT_CALL(module, Process())
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/BUILD.gn
>index d0ee6f88fae..df5139ef586 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/BUILD.gn
>@@ -29,14 +29,14 @@ rtc_static_library("video_capture_module") {
>   deps = [
>     "..:module_api",
>     "../..:webrtc_common",
>-    "../../:typedefs",
>     "../../api:libjingle_peerconnection_api",
>-    "../../api:video_frame_api",
>-    "../../api:video_frame_api_i420",
>+    "../../api/video:video_frame",
>+    "../../api/video:video_frame_i420",
>     "../../common_video",
>     "../../media:rtc_media_base",
>     "../../rtc_base:rtc_base_approved",
>     "../../rtc_base:stringutils",
>+    "../../rtc_base/synchronization:rw_lock_wrapper",
>     "../../system_wrappers",
>     "//third_party/libyuv",
>   ]
>@@ -77,28 +77,9 @@ if (!build_with_chromium) {
>     }
>   }
> 
>-  config("video_capture_internal_impl_warnings_config") {
>-    if (is_win && is_clang) {
>-      cflags = [
>-        "-Wno-comment",
>-        "-Wno-ignored-attributes",
>-
>-        # See https://bugs.chromium.org/p/webrtc/issues/detail?id=6269
>-        # for -Wno-ignored-qualifiers
>-        "-Wno-ignored-qualifiers",
>-        "-Wno-microsoft-extra-qualification",
>-        "-Wno-missing-braces",
>-        "-Wno-overloaded-virtual",
>-        "-Wno-reorder",
>-        "-Wno-writable-strings",
>-      ]
>-    }
>-  }
>-
>   if (is_ios || is_mac) {
>     rtc_source_set("video_capture_internal_impl_objc") {
>       visibility = [ ":video_capture_internal_impl" ]
>-      configs += [ ":video_capture_internal_impl_warnings_config" ]
> 
>       deps = [
>         ":video_capture_module",
>@@ -127,8 +108,6 @@ if (!build_with_chromium) {
>   }
> 
>   rtc_source_set("video_capture_internal_impl") {
>-    configs += [ ":video_capture_internal_impl_warnings_config" ]
>-
>     deps = [
>       ":video_capture_module",
>       "../../rtc_base:checks",
>@@ -165,7 +144,7 @@ if (!build_with_chromium) {
>         "windows/video_capture_mf.h",
>       ]
> 
>-      libs = [ "Strmiids.lib" ]
>+      libs = [ "strmiids.lib" ]
> 
>       if (build_with_mozilla) {
>         sources += [
>@@ -208,12 +187,6 @@ if (!build_with_chromium) {
>       sources = [
>         "test/video_capture_unittest.cc",
>       ]
>-
>-      cflags = []
>-      if (is_linux || is_mac) {
>-        cflags += [ "-Wno-write-strings" ]
>-      }
>-
>       ldflags = []
>       if (is_linux || is_mac) {
>         ldflags += [
>@@ -232,8 +205,8 @@ if (!build_with_chromium) {
>       deps = [
>         ":video_capture_internal_impl",
>         ":video_capture_module",
>-        "../../api:video_frame_api",
>-        "../../api:video_frame_api_i420",
>+        "../../api/video:video_frame",
>+        "../../api/video:video_frame_i420",
>         "../../common_video:common_video",
>         "../../rtc_base:rtc_base_approved",
>         "../../system_wrappers:system_wrappers",
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/device_info_impl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/device_info_impl.h
>index 296adfbbab7..f6e197007b2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/device_info_impl.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/device_info_impl.h
>@@ -14,47 +14,44 @@
> #include <vector>
> 
> #include "modules/video_capture/video_capture.h"
>-#include "system_wrappers/include/rw_lock_wrapper.h"
>-
>-namespace webrtc
>-{
>-namespace videocapturemodule
>-{
>-class DeviceInfoImpl: public VideoCaptureModule::DeviceInfo
>-{
>-public:
>-    DeviceInfoImpl();
>-    virtual ~DeviceInfoImpl(void);
>-    virtual int32_t NumberOfCapabilities(const char* deviceUniqueIdUTF8);
>-    virtual int32_t GetCapability(
>-        const char* deviceUniqueIdUTF8,
>-        const uint32_t deviceCapabilityNumber,
>-        VideoCaptureCapability& capability);
>-
>-    virtual int32_t GetBestMatchedCapability(
>-        const char* deviceUniqueIdUTF8,
>-        const VideoCaptureCapability& requested,
>-        VideoCaptureCapability& resulting);
>-    virtual int32_t GetOrientation(const char* deviceUniqueIdUTF8,
>-                                   VideoRotation& orientation);
>-
>-protected:
>-    /* Initialize this object*/
>-
>-    virtual int32_t Init()=0;
>-    /*
>-     * Fills the member variable _captureCapabilities with capabilities for the given device name.
>-     */
>-    virtual int32_t CreateCapabilityMap(const char* deviceUniqueIdUTF8)=0;
>-
>-protected:
>-    // Data members
>-    typedef std::vector<VideoCaptureCapability> VideoCaptureCapabilities;
>-    VideoCaptureCapabilities _captureCapabilities;
>-    RWLockWrapper& _apiLock;
>-    char* _lastUsedDeviceName;
>-    uint32_t _lastUsedDeviceNameLength;
>+#include "rtc_base/synchronization/rw_lock_wrapper.h"
>+
>+namespace webrtc {
>+namespace videocapturemodule {
>+class DeviceInfoImpl : public VideoCaptureModule::DeviceInfo {
>+ public:
>+  DeviceInfoImpl();
>+  virtual ~DeviceInfoImpl(void);
>+  virtual int32_t NumberOfCapabilities(const char* deviceUniqueIdUTF8);
>+  virtual int32_t GetCapability(const char* deviceUniqueIdUTF8,
>+                                const uint32_t deviceCapabilityNumber,
>+                                VideoCaptureCapability& capability);
>+
>+  virtual int32_t GetBestMatchedCapability(
>+      const char* deviceUniqueIdUTF8,
>+      const VideoCaptureCapability& requested,
>+      VideoCaptureCapability& resulting);
>+  virtual int32_t GetOrientation(const char* deviceUniqueIdUTF8,
>+                                 VideoRotation& orientation);
>+
>+ protected:
>+  /* Initialize this object*/
>+
>+  virtual int32_t Init() = 0;
>+  /*
>+   * Fills the member variable _captureCapabilities with capabilities for the
>+   * given device name.
>+   */
>+  virtual int32_t CreateCapabilityMap(const char* deviceUniqueIdUTF8) = 0;
>+
>+ protected:
>+  // Data members
>+  typedef std::vector<VideoCaptureCapability> VideoCaptureCapabilities;
>+  VideoCaptureCapabilities _captureCapabilities;
>+  RWLockWrapper& _apiLock;
>+  char* _lastUsedDeviceName;
>+  uint32_t _lastUsedDeviceNameLength;
> };
> }  // namespace videocapturemodule
> }  // namespace webrtc
>-#endif // MODULES_VIDEO_CAPTURE_MAIN_SOURCE_DEVICE_INFO_IMPL_H_
>+#endif  // MODULES_VIDEO_CAPTURE_MAIN_SOURCE_DEVICE_INFO_IMPL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/external/device_info_external.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/external/device_info_external.cc
>index 4951dc8927d..cd660e2e0ba 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/external/device_info_external.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/external/device_info_external.cc
>@@ -25,19 +25,21 @@ class ExternalDeviceInfo : public DeviceInfoImpl {
>       const char* /*dialogTitleUTF8*/,
>       void* /*parentWindow*/,
>       uint32_t /*positionX*/,
>-      uint32_t /*positionY*/) { return -1; }
>-  virtual int32_t GetDeviceName(
>-      uint32_t deviceNumber,
>-      char* deviceNameUTF8,
>-      uint32_t deviceNameLength,
>-      char* deviceUniqueIdUTF8,
>-      uint32_t deviceUniqueIdUTF8Length,
>-      char* productUniqueIdUTF8=0,
>-      uint32_t productUniqueIdUTF8Length=0) {
>+      uint32_t /*positionY*/) {
>     return -1;
>   }
>-  virtual int32_t CreateCapabilityMap(
>-      const char* deviceUniqueIdUTF8) { return 0; }
>+  virtual int32_t GetDeviceName(uint32_t deviceNumber,
>+                                char* deviceNameUTF8,
>+                                uint32_t deviceNameLength,
>+                                char* deviceUniqueIdUTF8,
>+                                uint32_t deviceUniqueIdUTF8Length,
>+                                char* productUniqueIdUTF8 = 0,
>+                                uint32_t productUniqueIdUTF8Length = 0) {
>+    return -1;
>+  }
>+  virtual int32_t CreateCapabilityMap(const char* deviceUniqueIdUTF8) {
>+    return 0;
>+  }
>   virtual int32_t Init() { return 0; }
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/linux/device_info_linux.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/linux/device_info_linux.h
>index 86eb61efa1e..fc7b4d8d03f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/linux/device_info_linux.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/linux/device_info_linux.h
>@@ -14,40 +14,39 @@
> #include "modules/video_capture/device_info_impl.h"
> #include "modules/video_capture/video_capture_impl.h"
> 
>-namespace webrtc
>-{
>-namespace videocapturemodule
>-{
>-class DeviceInfoLinux: public DeviceInfoImpl
>-{
>-public:
>-    DeviceInfoLinux();
>-    virtual ~DeviceInfoLinux();
>-    virtual uint32_t NumberOfDevices();
>-    virtual int32_t GetDeviceName(
>-        uint32_t deviceNumber,
>-        char* deviceNameUTF8,
>-        uint32_t deviceNameLength,
>-        char* deviceUniqueIdUTF8,
>-        uint32_t deviceUniqueIdUTF8Length,
>-        char* productUniqueIdUTF8=0,
>-        uint32_t productUniqueIdUTF8Length=0);
>-    /*
>-    * Fills the membervariable _captureCapabilities with capabilites for the given device name.
>-    */
>-    virtual int32_t CreateCapabilityMap (const char* deviceUniqueIdUTF8);
>-    virtual int32_t DisplayCaptureSettingsDialogBox(
>-        const char* /*deviceUniqueIdUTF8*/,
>-        const char* /*dialogTitleUTF8*/,
>-        void* /*parentWindow*/,
>-        uint32_t /*positionX*/,
>-        uint32_t /*positionY*/) { return -1;}
>-    int32_t FillCapabilities(int fd);
>-    int32_t Init();
>-private:
>+namespace webrtc {
>+namespace videocapturemodule {
>+class DeviceInfoLinux : public DeviceInfoImpl {
>+ public:
>+  DeviceInfoLinux();
>+  virtual ~DeviceInfoLinux();
>+  virtual uint32_t NumberOfDevices();
>+  virtual int32_t GetDeviceName(uint32_t deviceNumber,
>+                                char* deviceNameUTF8,
>+                                uint32_t deviceNameLength,
>+                                char* deviceUniqueIdUTF8,
>+                                uint32_t deviceUniqueIdUTF8Length,
>+                                char* productUniqueIdUTF8 = 0,
>+                                uint32_t productUniqueIdUTF8Length = 0);
>+  /*
>+   * Fills the membervariable _captureCapabilities with capabilites for the
>+   * given device name.
>+   */
>+  virtual int32_t CreateCapabilityMap(const char* deviceUniqueIdUTF8);
>+  virtual int32_t DisplayCaptureSettingsDialogBox(
>+      const char* /*deviceUniqueIdUTF8*/,
>+      const char* /*dialogTitleUTF8*/,
>+      void* /*parentWindow*/,
>+      uint32_t /*positionX*/,
>+      uint32_t /*positionY*/) {
>+    return -1;
>+  }
>+  int32_t FillCapabilities(int fd);
>+  int32_t Init();
> 
>-    bool IsDeviceNameMatches(const char* name, const char* deviceUniqueIdUTF8);
>+ private:
>+  bool IsDeviceNameMatches(const char* name, const char* deviceUniqueIdUTF8);
> };
> }  // namespace videocapturemodule
> }  // namespace webrtc
>-#endif // MODULES_VIDEO_CAPTURE_MAIN_SOURCE_LINUX_DEVICE_INFO_LINUX_H_
>+#endif  // MODULES_VIDEO_CAPTURE_MAIN_SOURCE_LINUX_DEVICE_INFO_LINUX_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/linux/video_capture_linux.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/linux/video_capture_linux.cc
>index 5be4272b0b3..70a4a80fcfc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/linux/video_capture_linux.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/linux/video_capture_linux.cc
>@@ -20,7 +20,6 @@
> #include <sys/stat.h>
> #include <unistd.h>
> 
>-#include <iostream>
> #include <new>
> 
> #include "media/base/videocommon.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/linux/video_capture_linux.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/linux/video_capture_linux.h
>index c0649a5170b..88455f9b9b4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/linux/video_capture_linux.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/linux/video_capture_linux.h
>@@ -18,50 +18,46 @@
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/platform_thread.h"
> 
>-namespace webrtc
>-{
>-namespace videocapturemodule
>-{
>-class VideoCaptureModuleV4L2: public VideoCaptureImpl
>-{
>-public:
>-    VideoCaptureModuleV4L2();
>-    virtual ~VideoCaptureModuleV4L2();
>-    virtual int32_t Init(const char* deviceUniqueId);
>-    virtual int32_t StartCapture(const VideoCaptureCapability& capability);
>-    virtual int32_t StopCapture();
>-    virtual bool CaptureStarted();
>-    virtual int32_t CaptureSettings(VideoCaptureCapability& settings);
>-
>-private:
>-    enum {kNoOfV4L2Bufffers=4};
>-
>-    static bool CaptureThread(void*);
>-    bool CaptureProcess();
>-    bool AllocateVideoBuffers();
>-    bool DeAllocateVideoBuffers();
>-
>-    // TODO(pbos): Stop using unique_ptr and resetting the thread.
>-    std::unique_ptr<rtc::PlatformThread> _captureThread;
>-    rtc::CriticalSection _captureCritSect;
>-
>-    int32_t _deviceId;
>-    int32_t _deviceFd;
>-
>-    int32_t _buffersAllocatedByDevice;
>-    int32_t _currentWidth;
>-    int32_t _currentHeight;
>-    int32_t _currentFrameRate;
>-    bool _captureStarted;
>-    VideoType _captureVideoType;
>-    struct Buffer
>-    {
>-        void *start;
>-        size_t length;
>-    };
>-    Buffer *_pool;
>+namespace webrtc {
>+namespace videocapturemodule {
>+class VideoCaptureModuleV4L2 : public VideoCaptureImpl {
>+ public:
>+  VideoCaptureModuleV4L2();
>+  virtual ~VideoCaptureModuleV4L2();
>+  virtual int32_t Init(const char* deviceUniqueId);
>+  virtual int32_t StartCapture(const VideoCaptureCapability& capability);
>+  virtual int32_t StopCapture();
>+  virtual bool CaptureStarted();
>+  virtual int32_t CaptureSettings(VideoCaptureCapability& settings);
>+
>+ private:
>+  enum { kNoOfV4L2Bufffers = 4 };
>+
>+  static bool CaptureThread(void*);
>+  bool CaptureProcess();
>+  bool AllocateVideoBuffers();
>+  bool DeAllocateVideoBuffers();
>+
>+  // TODO(pbos): Stop using unique_ptr and resetting the thread.
>+  std::unique_ptr<rtc::PlatformThread> _captureThread;
>+  rtc::CriticalSection _captureCritSect;
>+
>+  int32_t _deviceId;
>+  int32_t _deviceFd;
>+
>+  int32_t _buffersAllocatedByDevice;
>+  int32_t _currentWidth;
>+  int32_t _currentHeight;
>+  int32_t _currentFrameRate;
>+  bool _captureStarted;
>+  VideoType _captureVideoType;
>+  struct Buffer {
>+    void* start;
>+    size_t length;
>+  };
>+  Buffer* _pool;
> };
> }  // namespace videocapturemodule
> }  // namespace webrtc
> 
>-#endif // MODULES_VIDEO_CAPTURE_MAIN_SOURCE_LINUX_VIDEO_CAPTURE_LINUX_H_
>+#endif  // MODULES_VIDEO_CAPTURE_MAIN_SOURCE_LINUX_VIDEO_CAPTURE_LINUX_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/objc/rtc_video_capture_objc.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/objc/rtc_video_capture_objc.mm
>index 92bfd64fa3c..61ef8dbc5ad 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/objc/rtc_video_capture_objc.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/objc/rtc_video_capture_objc.mm
>@@ -59,7 +59,7 @@ - (id)initWithOwner:(VideoCaptureIos*)owner {
>     AVCaptureVideoDataOutput* captureOutput = [[AVCaptureVideoDataOutput alloc] init];
>     NSString* key = (NSString*)kCVPixelBufferPixelFormatTypeKey;
> 
>-    NSNumber* val = [NSNumber numberWithUnsignedInt:kCVPixelFormatType_420YpCbCr8BiPlanarFullRange];
>+    NSNumber* val = [NSNumber numberWithUnsignedInt:kCVPixelFormatType_422YpCbCr8];
>     NSDictionary* videoSettings = [NSDictionary dictionaryWithObject:val forKey:key];
>     captureOutput.videoSettings = videoSettings;
> 
>@@ -319,21 +319,16 @@ - (void)captureOutput:(AVCaptureOutput*)captureOutput
>     return;
>   }
> 
>-  const int kYPlaneIndex = 0;
>-  const int kUVPlaneIndex = 1;
>-
>-  uint8_t* baseAddress = (uint8_t*)CVPixelBufferGetBaseAddressOfPlane(videoFrame, kYPlaneIndex);
>-  size_t yPlaneBytesPerRow = CVPixelBufferGetBytesPerRowOfPlane(videoFrame, kYPlaneIndex);
>-  size_t yPlaneHeight = CVPixelBufferGetHeightOfPlane(videoFrame, kYPlaneIndex);
>-  size_t uvPlaneBytesPerRow = CVPixelBufferGetBytesPerRowOfPlane(videoFrame, kUVPlaneIndex);
>-  size_t uvPlaneHeight = CVPixelBufferGetHeightOfPlane(videoFrame, kUVPlaneIndex);
>-  size_t frameSize = yPlaneBytesPerRow * yPlaneHeight + uvPlaneBytesPerRow * uvPlaneHeight;
>+  uint8_t* baseAddress = (uint8_t*)CVPixelBufferGetBaseAddress(videoFrame);
>+  const size_t width = CVPixelBufferGetWidth(videoFrame);
>+  const size_t height = CVPixelBufferGetHeight(videoFrame);
>+  const size_t frameSize = width * height * 2;
> 
>   VideoCaptureCapability tempCaptureCapability;
>-  tempCaptureCapability.width = CVPixelBufferGetWidth(videoFrame);
>-  tempCaptureCapability.height = CVPixelBufferGetHeight(videoFrame);
>+  tempCaptureCapability.width = width;
>+  tempCaptureCapability.height = height;
>   tempCaptureCapability.maxFPS = _capability.maxFPS;
>-  tempCaptureCapability.videoType = VideoType::kNV12;
>+  tempCaptureCapability.videoType = VideoType::kUYVY;
> 
>   _owner->IncomingFrame(baseAddress, frameSize, tempCaptureCapability, 0);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/test/video_capture_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/test/video_capture_unittest.cc
>index 16330b616f4..aed299c012c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/test/video_capture_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/test/video_capture_unittest.cc
>@@ -32,25 +32,24 @@ using webrtc::VideoCaptureCapability;
> using webrtc::VideoCaptureFactory;
> using webrtc::VideoCaptureModule;
> 
>-
>-#define WAIT_(ex, timeout, res) \
>-  do { \
>-    res = (ex); \
>-    int64_t start = rtc::TimeMillis(); \
>+#define WAIT_(ex, timeout, res)                           \
>+  do {                                                    \
>+    res = (ex);                                           \
>+    int64_t start = rtc::TimeMillis();                    \
>     while (!res && rtc::TimeMillis() < start + timeout) { \
>-      SleepMs(5); \
>-      res = (ex); \
>-    } \
>+      SleepMs(5);                                         \
>+      res = (ex);                                         \
>+    }                                                     \
>   } while (0)
> 
> #define EXPECT_TRUE_WAIT(ex, timeout) \
>-  do { \
>-    bool res; \
>-    WAIT_(ex, timeout, res); \
>-    if (!res) EXPECT_TRUE(ex); \
>+  do {                                \
>+    bool res;                         \
>+    WAIT_(ex, timeout, res);          \
>+    if (!res)                         \
>+      EXPECT_TRUE(ex);                \
>   } while (0)
> 
>-
> static const int kTimeOut = 5000;
> static const int kTestHeight = 288;
> static const int kTestWidth = 352;
>@@ -59,8 +58,8 @@ static const int kTestFramerate = 30;
> class TestVideoCaptureCallback
>     : public rtc::VideoSinkInterface<webrtc::VideoFrame> {
>  public:
>-  TestVideoCaptureCallback() :
>-        last_render_time_ms_(0),
>+  TestVideoCaptureCallback()
>+      : last_render_time_ms_(0),
>         incoming_frames_(0),
>         timing_warnings_(0),
>         rotate_frame_(webrtc::kVideoRotation_0) {}
>@@ -85,16 +84,15 @@ class TestVideoCaptureCallback
>     EXPECT_EQ(rotate_frame_, videoFrame.rotation());
> #endif
>     // RenderTimstamp should be the time now.
>-    EXPECT_TRUE(
>-        videoFrame.render_time_ms() >= rtc::TimeMillis()-30 &&
>-        videoFrame.render_time_ms() <= rtc::TimeMillis());
>+    EXPECT_TRUE(videoFrame.render_time_ms() >= rtc::TimeMillis() - 30 &&
>+                videoFrame.render_time_ms() <= rtc::TimeMillis());
> 
>     if ((videoFrame.render_time_ms() >
>-            last_render_time_ms_ + (1000 * 1.1) / capability_.maxFPS &&
>-            last_render_time_ms_ > 0) ||
>+             last_render_time_ms_ + (1000 * 1.1) / capability_.maxFPS &&
>+         last_render_time_ms_ > 0) ||
>         (videoFrame.render_time_ms() <
>-            last_render_time_ms_ + (1000 * 0.9) / capability_.maxFPS &&
>-            last_render_time_ms_ > 0)) {
>+             last_render_time_ms_ + (1000 * 0.9) / capability_.maxFPS &&
>+         last_render_time_ms_ > 0)) {
>       timing_warnings_++;
>     }
> 
>@@ -105,7 +103,7 @@ class TestVideoCaptureCallback
> 
>   void SetExpectedCapability(VideoCaptureCapability capability) {
>     rtc::CritScope cs(&capture_cs_);
>-    capability_= capability;
>+    capability_ = capability;
>     incoming_frames_ = 0;
>     last_render_time_ms_ = 0;
>   }
>@@ -161,8 +159,8 @@ class VideoCaptureTest : public testing::Test {
>     char device_name[256];
>     char unique_name[256];
> 
>-    EXPECT_EQ(0, device_info_->GetDeviceName(
>-        device, device_name, 256, unique_name, 256));
>+    EXPECT_EQ(0, device_info_->GetDeviceName(device, device_name, 256,
>+                                             unique_name, 256));
> 
>     rtc::scoped_refptr<VideoCaptureModule> module(
>         VideoCaptureFactory::Create(unique_name));
>@@ -251,8 +249,8 @@ TEST_F(VideoCaptureTest, MAYBE_Capabilities) {
>       OpenVideoCaptureDevice(0, &capture_observer));
>   ASSERT_TRUE(module.get() != NULL);
> 
>-  int number_of_capabilities = device_info_->NumberOfCapabilities(
>-      module->CurrentDeviceName());
>+  int number_of_capabilities =
>+      device_info_->NumberOfCapabilities(module->CurrentDeviceName());
>   EXPECT_GT(number_of_capabilities, 0);
>   // Key is <width>x<height>, value is vector of maxFPS values at that
>   // resolution.
>@@ -293,8 +291,7 @@ TEST_F(VideoCaptureTest, MAYBE_Capabilities) {
>   EXPECT_GT(frame_rates_by_resolution.size(), 1U);
>   for (FrameRatesByResolution::const_iterator it =
>            frame_rates_by_resolution.begin();
>-       it != frame_rates_by_resolution.end();
>-       ++it) {
>+       it != frame_rates_by_resolution.end(); ++it) {
>     EXPECT_GT(it->second.size(), 1U) << it->first;
>   }
> #endif  // WEBRTC_ANDROID
>@@ -328,7 +325,6 @@ TEST_F(VideoCaptureTest, DISABLED_TestTwoCameras) {
>       OpenVideoCaptureDevice(1, &capture_observer2));
>   ASSERT_TRUE(module1.get() != NULL);
> 
>-
>   VideoCaptureCapability capability2;
> #ifndef WEBRTC_MAC
>   device_info_->GetCapability(module2->CurrentDeviceName(), 0, capability2);
>@@ -370,16 +366,15 @@ class VideoCaptureExternalTest : public testing::Test {
>            buffer->ChromaHeight() * buffer->StrideU());
>     memset(buffer->MutableDataV(), 127,
>            buffer->ChromaHeight() * buffer->StrideV());
>-    test_frame_.reset(
>-        new webrtc::VideoFrame(buffer, 0, 0, webrtc::kVideoRotation_0));
>+    test_frame_.reset(new webrtc::VideoFrame(buffer, webrtc::kVideoRotation_0,
>+                                             0 /* timestamp_us */));
> 
>     SleepMs(1);  // Wait 1ms so that two tests can't have the same timestamp.
> 
>     capture_module_->RegisterCaptureDataCallback(&capture_callback_);
>   }
> 
>-  void TearDown() {
>-  }
>+  void TearDown() {}
> 
>   webrtc::VideoCaptureExternal* capture_input_interface_;
>   rtc::scoped_refptr<VideoCaptureModule> capture_module_;
>@@ -393,8 +388,9 @@ TEST_F(VideoCaptureExternalTest, TestExternalCapture) {
>       webrtc::VideoType::kI420, test_frame_->width(), test_frame_->height());
>   std::unique_ptr<uint8_t[]> test_buffer(new uint8_t[length]);
>   webrtc::ExtractBuffer(*test_frame_, length, test_buffer.get());
>-  EXPECT_EQ(0, capture_input_interface_->IncomingFrame(test_buffer.get(),
>-      length, capture_callback_.capability(), 0));
>+  EXPECT_EQ(0,
>+            capture_input_interface_->IncomingFrame(
>+                test_buffer.get(), length, capture_callback_.capability(), 0));
>   EXPECT_TRUE(capture_callback_.CompareLastFrame(*test_frame_));
> }
> 
>@@ -404,18 +400,22 @@ TEST_F(VideoCaptureExternalTest, Rotation) {
>       webrtc::VideoType::kI420, test_frame_->width(), test_frame_->height());
>   std::unique_ptr<uint8_t[]> test_buffer(new uint8_t[length]);
>   webrtc::ExtractBuffer(*test_frame_, length, test_buffer.get());
>-  EXPECT_EQ(0, capture_input_interface_->IncomingFrame(test_buffer.get(),
>-    length, capture_callback_.capability(), 0));
>+  EXPECT_EQ(0,
>+            capture_input_interface_->IncomingFrame(
>+                test_buffer.get(), length, capture_callback_.capability(), 0));
>   EXPECT_EQ(0, capture_module_->SetCaptureRotation(webrtc::kVideoRotation_90));
>   capture_callback_.SetExpectedCaptureRotation(webrtc::kVideoRotation_90);
>-  EXPECT_EQ(0, capture_input_interface_->IncomingFrame(test_buffer.get(),
>-    length, capture_callback_.capability(), 0));
>+  EXPECT_EQ(0,
>+            capture_input_interface_->IncomingFrame(
>+                test_buffer.get(), length, capture_callback_.capability(), 0));
>   EXPECT_EQ(0, capture_module_->SetCaptureRotation(webrtc::kVideoRotation_180));
>   capture_callback_.SetExpectedCaptureRotation(webrtc::kVideoRotation_180);
>-  EXPECT_EQ(0, capture_input_interface_->IncomingFrame(test_buffer.get(),
>-    length, capture_callback_.capability(), 0));
>+  EXPECT_EQ(0,
>+            capture_input_interface_->IncomingFrame(
>+                test_buffer.get(), length, capture_callback_.capability(), 0));
>   EXPECT_EQ(0, capture_module_->SetCaptureRotation(webrtc::kVideoRotation_270));
>   capture_callback_.SetExpectedCaptureRotation(webrtc::kVideoRotation_270);
>-  EXPECT_EQ(0, capture_input_interface_->IncomingFrame(test_buffer.get(),
>-    length, capture_callback_.capability(), 0));
>+  EXPECT_EQ(0,
>+            capture_input_interface_->IncomingFrame(
>+                test_buffer.get(), length, capture_callback_.capability(), 0));
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/video_capture.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/video_capture.h
>index 28340a5c135..0231f28ef7a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/video_capture.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/video_capture.h
>@@ -12,13 +12,13 @@
> #define MODULES_VIDEO_CAPTURE_VIDEO_CAPTURE_H_
> 
> #include "api/video/video_rotation.h"
>-#include "api/videosinkinterface.h"
>+#include "api/video/video_sink_interface.h"
> #include "modules/include/module.h"
> #include "modules/video_capture/video_capture_defines.h"
> 
> namespace webrtc {
> 
>-class VideoCaptureModule: public rtc::RefCountInterface {
>+class VideoCaptureModule : public rtc::RefCountInterface {
>  public:
>   // Interface for receiving information about available camera devices.
>   class DeviceInfo {
>@@ -32,25 +32,21 @@ class VideoCaptureModule: public rtc::RefCountInterface {
>     //                      Otherwise same as deviceNameUTF8.
>     // productUniqueIdUTF8 - Unique product id if it exist.
>     //                       Null terminated otherwise.
>-    virtual int32_t GetDeviceName(
>-        uint32_t deviceNumber,
>-        char* deviceNameUTF8,
>-        uint32_t deviceNameLength,
>-        char* deviceUniqueIdUTF8,
>-        uint32_t deviceUniqueIdUTF8Length,
>-        char* productUniqueIdUTF8 = 0,
>-        uint32_t productUniqueIdUTF8Length = 0) = 0;
>-
>+    virtual int32_t GetDeviceName(uint32_t deviceNumber,
>+                                  char* deviceNameUTF8,
>+                                  uint32_t deviceNameLength,
>+                                  char* deviceUniqueIdUTF8,
>+                                  uint32_t deviceUniqueIdUTF8Length,
>+                                  char* productUniqueIdUTF8 = 0,
>+                                  uint32_t productUniqueIdUTF8Length = 0) = 0;
> 
>     // Returns the number of capabilities this device.
>-    virtual int32_t NumberOfCapabilities(
>-        const char* deviceUniqueIdUTF8) = 0;
>+    virtual int32_t NumberOfCapabilities(const char* deviceUniqueIdUTF8) = 0;
> 
>     // Gets the capabilities of the named device.
>-    virtual int32_t GetCapability(
>-        const char* deviceUniqueIdUTF8,
>-        const uint32_t deviceCapabilityNumber,
>-        VideoCaptureCapability& capability) = 0;
>+    virtual int32_t GetCapability(const char* deviceUniqueIdUTF8,
>+                                  const uint32_t deviceCapabilityNumber,
>+                                  VideoCaptureCapability& capability) = 0;
> 
>     // Gets clockwise angle the captured frames should be rotated in order
>     // to be displayed correctly on a normally rotated display.
>@@ -65,7 +61,7 @@ class VideoCaptureModule: public rtc::RefCountInterface {
>         const VideoCaptureCapability& requested,
>         VideoCaptureCapability& resulting) = 0;
> 
>-     // Display OS /capture device specific settings dialog
>+    // Display OS /capture device specific settings dialog
>     virtual int32_t DisplayCaptureSettingsDialogBox(
>         const char* deviceUniqueIdUTF8,
>         const char* dialogTitleUTF8,
>@@ -78,14 +74,13 @@ class VideoCaptureModule: public rtc::RefCountInterface {
> 
>   //   Register capture data callback
>   virtual void RegisterCaptureDataCallback(
>-      rtc::VideoSinkInterface<VideoFrame> *dataCallback) = 0;
>+      rtc::VideoSinkInterface<VideoFrame>* dataCallback) = 0;
> 
>   //  Remove capture data callback
>   virtual void DeRegisterCaptureDataCallback() = 0;
> 
>   // Start capture device
>-  virtual int32_t StartCapture(
>-      const VideoCaptureCapability& capability) = 0;
>+  virtual int32_t StartCapture(const VideoCaptureCapability& capability) = 0;
> 
>   virtual int32_t StopCapture() = 0;
> 
>@@ -113,8 +108,8 @@ class VideoCaptureModule: public rtc::RefCountInterface {
>   // Return whether the rotation is applied or left pending.
>   virtual bool GetApplyRotation() = 0;
> 
>-protected:
>-  virtual ~VideoCaptureModule() {};
>+ protected:
>+  virtual ~VideoCaptureModule(){};
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/video_capture_config.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/video_capture_config.h
>index 4cfff6ee715..f285b9eeb14 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/video_capture_config.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/video_capture_config.h
>@@ -11,23 +11,23 @@
> #ifndef MODULES_VIDEO_CAPTURE_MAIN_SOURCE_VIDEO_CAPTURE_CONFIG_H_
> #define MODULES_VIDEO_CAPTURE_MAIN_SOURCE_VIDEO_CAPTURE_CONFIG_H_
> 
>-namespace webrtc
>-{
>-namespace videocapturemodule
>-{
>-enum {kDefaultWidth = 640};  // Start width
>-enum {kDefaultHeight = 480}; // Start heigt
>-enum {kDefaultFrameRate = 30}; // Start frame rate
>+namespace webrtc {
>+namespace videocapturemodule {
>+enum { kDefaultWidth = 640 };     // Start width
>+enum { kDefaultHeight = 480 };    // Start heigt
>+enum { kDefaultFrameRate = 30 };  // Start frame rate
> 
>-enum {kMaxFrameRate =60}; // Max allowed frame rate of the start image 
>+enum { kMaxFrameRate = 60 };  // Max allowed frame rate of the start image
> 
>-enum {kDefaultCaptureDelay = 120}; 
>-enum {kMaxCaptureDelay = 270}; // Max capture delay allowed in the precompiled capture delay values.  
>+enum { kDefaultCaptureDelay = 120 };
>+enum {
>+  kMaxCaptureDelay = 270
>+};  // Max capture delay allowed in the precompiled capture delay values.
> 
>-enum {kFrameRateCallbackInterval = 1000}; 
>-enum {kFrameRateCountHistorySize = 90};
>-enum {kFrameRateHistoryWindowMs = 2000};
>+enum { kFrameRateCallbackInterval = 1000 };
>+enum { kFrameRateCountHistorySize = 90 };
>+enum { kFrameRateHistoryWindowMs = 2000 };
> }  // namespace videocapturemodule
> }  // namespace webrtc
> 
>-#endif // MODULES_VIDEO_CAPTURE_MAIN_SOURCE_VIDEO_CAPTURE_CONFIG_H_
>+#endif  // MODULES_VIDEO_CAPTURE_MAIN_SOURCE_VIDEO_CAPTURE_CONFIG_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/video_capture_defines.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/video_capture_defines.h
>index a87f5ee2f9b..3987c03e100 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/video_capture_defines.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/video_capture_defines.h
>@@ -13,69 +13,60 @@
> 
> #include "api/video/video_frame.h"
> #include "modules/include/module_common_types.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
>-namespace webrtc
>-{
>-// Defines
>-#ifndef NULL
>-    #define NULL    0
>-#endif
>+namespace webrtc {
> 
>-enum {kVideoCaptureUniqueNameLength =1024}; //Max unique capture device name lenght
>-enum {kVideoCaptureDeviceNameLength =256}; //Max capture device name lenght
>-enum {kVideoCaptureProductIdLength =128}; //Max product id length
>+enum {
>+  kVideoCaptureUniqueNameLength = 1024
>+};  // Max unique capture device name lenght
>+enum { kVideoCaptureDeviceNameLength = 256 };  // Max capture device name lenght
>+enum { kVideoCaptureProductIdLength = 128 };   // Max product id length
> 
>-struct VideoCaptureCapability
>-{
>-    int32_t width;
>-    int32_t height;
>-    int32_t maxFPS;
>-    VideoType videoType;
>-    bool interlaced;
>+struct VideoCaptureCapability {
>+  int32_t width;
>+  int32_t height;
>+  int32_t maxFPS;
>+  VideoType videoType;
>+  bool interlaced;
> 
>-    VideoCaptureCapability()
>-    {
>-        width = 0;
>-        height = 0;
>-        maxFPS = 0;
>-        videoType = VideoType::kUnknown;
>-        interlaced = false;
>-    }
>-    ;
>-    bool operator!=(const VideoCaptureCapability &other) const
>-    {
>-        if (width != other.width)
>-            return true;
>-        if (height != other.height)
>-            return true;
>-        if (maxFPS != other.maxFPS)
>-            return true;
>-        if (videoType != other.videoType)
>-          return true;
>-        if (interlaced != other.interlaced)
>-            return true;
>-        return false;
>-    }
>-    bool operator==(const VideoCaptureCapability &other) const
>-    {
>-        return !operator!=(other);
>-    }
>+  VideoCaptureCapability() {
>+    width = 0;
>+    height = 0;
>+    maxFPS = 0;
>+    videoType = VideoType::kUnknown;
>+    interlaced = false;
>+  };
>+  bool operator!=(const VideoCaptureCapability& other) const {
>+    if (width != other.width)
>+      return true;
>+    if (height != other.height)
>+      return true;
>+    if (maxFPS != other.maxFPS)
>+      return true;
>+    if (videoType != other.videoType)
>+      return true;
>+    if (interlaced != other.interlaced)
>+      return true;
>+    return false;
>+  }
>+  bool operator==(const VideoCaptureCapability& other) const {
>+    return !operator!=(other);
>+  }
> };
> 
> /* External Capture interface. Returned by Create
>  and implemented by the capture module.
>  */
>-class VideoCaptureExternal
>-{
>-public:
>-    // |capture_time| must be specified in the NTP time format in milliseconds.
>-    virtual int32_t IncomingFrame(uint8_t* videoFrame,
>-                                  size_t videoFrameLength,
>-                                  const VideoCaptureCapability& frameInfo,
>-                                  int64_t captureTime = 0) = 0;
>-protected:
>-    ~VideoCaptureExternal() {}
>+class VideoCaptureExternal {
>+ public:
>+  // |capture_time| must be specified in the NTP time format in milliseconds.
>+  virtual int32_t IncomingFrame(uint8_t* videoFrame,
>+                                size_t videoFrameLength,
>+                                const VideoCaptureCapability& frameInfo,
>+                                int64_t captureTime = 0) = 0;
>+
>+ protected:
>+  ~VideoCaptureExternal() {}
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/video_capture_impl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/video_capture_impl.h
>index d2525976435..7826427f398 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/video_capture_impl.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/video_capture_impl.h
>@@ -22,98 +22,97 @@
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/scoped_ref_ptr.h"
> 
>-namespace webrtc
>-{
>+namespace webrtc {
> 
> namespace videocapturemodule {
> // Class definitions
>-class VideoCaptureImpl: public VideoCaptureModule, public VideoCaptureExternal
>-{
>-public:
>-
>-    /*
>-     *   Create a video capture module object
>-     *
>-     *   id              - unique identifier of this video capture module object
>-     *   deviceUniqueIdUTF8 -  name of the device. Available names can be found by using GetDeviceName
>-     */
>-   static rtc::scoped_refptr<VideoCaptureModule> Create(
>-       const char* deviceUniqueIdUTF8);
>-
>-    /*
>-     *   Create a video capture module object used for external capture.
>-     *
>-     *   id              - unique identifier of this video capture module object
>-     *   externalCapture - [out] interface to call when a new frame is captured.
>-     */
>-   static rtc::scoped_refptr<VideoCaptureModule> Create(
>-       VideoCaptureExternal*& externalCapture);
>-
>-    static DeviceInfo* CreateDeviceInfo();
>-
>-    // Helpers for converting between (integral) degrees and
>-    // VideoRotation values.  Return 0 on success.
>-    static int32_t RotationFromDegrees(int degrees, VideoRotation* rotation);
>-    static int32_t RotationInDegrees(VideoRotation rotation, int* degrees);
>-
>-    //Call backs
>-    void RegisterCaptureDataCallback(
>-        rtc::VideoSinkInterface<VideoFrame>* dataCallback) override;
>-    void DeRegisterCaptureDataCallback() override;
>-
>-    int32_t SetCaptureRotation(VideoRotation rotation) override;
>-    bool SetApplyRotation(bool enable) override;
>-    bool GetApplyRotation() override {
>-      return apply_rotation_;
>-    }
>-
>-    const char* CurrentDeviceName() const override;
>-
>-    // Implement VideoCaptureExternal
>-    // |capture_time| must be specified in NTP time format in milliseconds.
>-    int32_t IncomingFrame(uint8_t* videoFrame,
>-                          size_t videoFrameLength,
>-                          const VideoCaptureCapability& frameInfo,
>-                          int64_t captureTime = 0) override;
>-
>-    // Platform dependent
>-    int32_t StartCapture(const VideoCaptureCapability& capability) override
>-    {
>-        _requestedCapability = capability;
>-        return -1;
>-    }
>-    int32_t StopCapture() override { return -1; }
>-    bool CaptureStarted() override {return false; }
>-    int32_t CaptureSettings(VideoCaptureCapability& /*settings*/) override
>-    { return -1; }
>-
>-protected:
>-    VideoCaptureImpl();
>-    virtual ~VideoCaptureImpl();
>-    int32_t DeliverCapturedFrame(VideoFrame& captureFrame);
>-
>-    char* _deviceUniqueId; // current Device unique name;
>-    rtc::CriticalSection _apiCs;
>-    VideoCaptureCapability _requestedCapability; // Should be set by platform dependent code in StartCapture.
>-private:
>-    void UpdateFrameCount();
>-    uint32_t CalculateFrameRate(int64_t now_ns);
>-
>-    // last time the module process function was called.
>-    int64_t _lastProcessTimeNanos;
>-    // last time the frame rate callback function was called.
>-    int64_t _lastFrameRateCallbackTimeNanos;
>-
>-    rtc::VideoSinkInterface<VideoFrame>* _dataCallBack;
>-
>-    int64_t _lastProcessFrameTimeNanos;
>-    // timestamp for local captured frames
>-    int64_t _incomingFrameTimesNanos[kFrameRateCountHistorySize];
>-    VideoRotation _rotateFrame;  // Set if the frame should be rotated by the
>-                                 // capture module.
>-
>-    // Indicate whether rotation should be applied before delivered externally.
>-    bool apply_rotation_;
>+class VideoCaptureImpl : public VideoCaptureModule,
>+                         public VideoCaptureExternal {
>+ public:
>+  /*
>+   *   Create a video capture module object
>+   *
>+   *   id              - unique identifier of this video capture module object
>+   *   deviceUniqueIdUTF8 -  name of the device. Available names can be found by
>+   * using GetDeviceName
>+   */
>+  static rtc::scoped_refptr<VideoCaptureModule> Create(
>+      const char* deviceUniqueIdUTF8);
>+
>+  /*
>+   *   Create a video capture module object used for external capture.
>+   *
>+   *   id              - unique identifier of this video capture module object
>+   *   externalCapture - [out] interface to call when a new frame is captured.
>+   */
>+  static rtc::scoped_refptr<VideoCaptureModule> Create(
>+      VideoCaptureExternal*& externalCapture);
>+
>+  static DeviceInfo* CreateDeviceInfo();
>+
>+  // Helpers for converting between (integral) degrees and
>+  // VideoRotation values.  Return 0 on success.
>+  static int32_t RotationFromDegrees(int degrees, VideoRotation* rotation);
>+  static int32_t RotationInDegrees(VideoRotation rotation, int* degrees);
>+
>+  // Call backs
>+  void RegisterCaptureDataCallback(
>+      rtc::VideoSinkInterface<VideoFrame>* dataCallback) override;
>+  void DeRegisterCaptureDataCallback() override;
>+
>+  int32_t SetCaptureRotation(VideoRotation rotation) override;
>+  bool SetApplyRotation(bool enable) override;
>+  bool GetApplyRotation() override { return apply_rotation_; }
>+
>+  const char* CurrentDeviceName() const override;
>+
>+  // Implement VideoCaptureExternal
>+  // |capture_time| must be specified in NTP time format in milliseconds.
>+  int32_t IncomingFrame(uint8_t* videoFrame,
>+                        size_t videoFrameLength,
>+                        const VideoCaptureCapability& frameInfo,
>+                        int64_t captureTime = 0) override;
>+
>+  // Platform dependent
>+  int32_t StartCapture(const VideoCaptureCapability& capability) override {
>+    _requestedCapability = capability;
>+    return -1;
>+  }
>+  int32_t StopCapture() override { return -1; }
>+  bool CaptureStarted() override { return false; }
>+  int32_t CaptureSettings(VideoCaptureCapability& /*settings*/) override {
>+    return -1;
>+  }
>+
>+ protected:
>+  VideoCaptureImpl();
>+  virtual ~VideoCaptureImpl();
>+  int32_t DeliverCapturedFrame(VideoFrame& captureFrame);
>+
>+  char* _deviceUniqueId;  // current Device unique name;
>+  rtc::CriticalSection _apiCs;
>+  VideoCaptureCapability _requestedCapability;  // Should be set by platform
>+                                                // dependent code in
>+                                                // StartCapture.
>+ private:
>+  void UpdateFrameCount();
>+  uint32_t CalculateFrameRate(int64_t now_ns);
>+
>+  // last time the module process function was called.
>+  int64_t _lastProcessTimeNanos;
>+  // last time the frame rate callback function was called.
>+  int64_t _lastFrameRateCallbackTimeNanos;
>+
>+  rtc::VideoSinkInterface<VideoFrame>* _dataCallBack;
>+
>+  int64_t _lastProcessFrameTimeNanos;
>+  // timestamp for local captured frames
>+  int64_t _incomingFrameTimesNanos[kFrameRateCountHistorySize];
>+  VideoRotation _rotateFrame;  // Set if the frame should be rotated by the
>+                               // capture module.
>+
>+  // Indicate whether rotation should be applied before delivered externally.
>+  bool apply_rotation_;
> };
> }  // namespace videocapturemodule
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/windows/device_info_ds.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/windows/device_info_ds.cc
>index d788ae13b2e..7cba0d40d49 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/windows/device_info_ds.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/windows/device_info_ds.cc
>@@ -10,14 +10,13 @@
> 
> #include "modules/video_capture/windows/device_info_ds.h"
> 
>-#include <ios>  // std::hex
>-
> #include "modules/video_capture/video_capture_config.h"
> #include "modules/video_capture/windows/help_functions_ds.h"
> #include "rtc_base/logging.h"
>+#include "rtc_base/stringutils.h"
> 
>-#include <Dvdmedia.h>
>-#include <Streams.h>
>+#include <dvdmedia.h>
>+#include <streams.h>
> 
> namespace webrtc {
> namespace videocapturemodule {
>@@ -75,7 +74,7 @@ DeviceInfoDS::DeviceInfoDS()
>       //
>       RTC_LOG(LS_INFO) << __FUNCTION__
>                        << ": CoInitializeEx(NULL, COINIT_APARTMENTTHREADED)"
>-                       << " => RPC_E_CHANGED_MODE, error 0x" << std::hex << hr;
>+                       << " => RPC_E_CHANGED_MODE, error 0x" << rtc::ToHex(hr);
>     }
>   }
> }
>@@ -93,7 +92,7 @@ int32_t DeviceInfoDS::Init() {
>                                 IID_ICreateDevEnum, (void**)&_dsDevEnum);
>   if (hr != NOERROR) {
>     RTC_LOG(LS_INFO) << "Failed to create CLSID_SystemDeviceEnum, error 0x"
>-                     << std::hex << hr;
>+                     << rtc::ToHex(hr);
>     return -1;
>   }
>   return 0;
>@@ -132,7 +131,7 @@ int32_t DeviceInfoDS::GetDeviceInfo(uint32_t deviceNumber,
>                                                  &_dsMonikerDevEnum, 0);
>   if (hr != NOERROR) {
>     RTC_LOG(LS_INFO) << "Failed to enumerate CLSID_SystemDeviceEnum, error 0x"
>-                     << std::hex << hr << ". No webcam exist?";
>+                     << rtc::ToHex(hr) << ". No webcam exist?";
>     return 0;
>   }
> 
>@@ -223,7 +222,7 @@ IBaseFilter* DeviceInfoDS::GetDeviceFilter(const char* deviceUniqueIdUTF8,
>                                                  &_dsMonikerDevEnum, 0);
>   if (hr != NOERROR) {
>     RTC_LOG(LS_INFO) << "Failed to enumerate CLSID_SystemDeviceEnum, error 0x"
>-                     << std::hex << hr << ". No webcam exist?";
>+                     << rtc::ToHex(hr) << ". No webcam exist?";
>     return 0;
>   }
>   _dsMonikerDevEnum->Reset();
>@@ -539,13 +538,12 @@ int32_t DeviceInfoDS::CreateCapabilityMap(const char* deviceUniqueIdUTF8)
>   return static_cast<int32_t>(_captureCapabilities.size());
> }
> 
>-/* Constructs a product ID from the Windows DevicePath. on a USB device the
>- devicePath contains product id and vendor id. This seems to work for firewire
>- as well
>- /* Example of device path
>- "\\?\usb#vid_0408&pid_2010&mi_00#7&258e7aaf&0&0000#{65e8773d-8f56-11d0-a3b9-00a0c9223196}\global"
>- "\\?\avc#sony&dv-vcr&camcorder&dv#65b2d50301460008#{65e8773d-8f56-11d0-a3b9-00a0c9223196}\global"
>- */
>+// Constructs a product ID from the Windows DevicePath. on a USB device the
>+// devicePath contains product id and vendor id. This seems to work for firewire
>+// as well.
>+// Example of device path:
>+// "\\?\usb#vid_0408&pid_2010&mi_00#7&258e7aaf&0&0000#{65e8773d-8f56-11d0-a3b9-00a0c9223196}\global"
>+// "\\?\avc#sony&dv-vcr&camcorder&dv#65b2d50301460008#{65e8773d-8f56-11d0-a3b9-00a0c9223196}\global"
> void DeviceInfoDS::GetProductId(const char* devicePath,
>                                 char* productUniqueIdUTF8,
>                                 uint32_t productUniqueIdUTF8Length) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/windows/device_info_ds.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/windows/device_info_ds.h
>index 9da349f42d1..68988557c78 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/windows/device_info_ds.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/windows/device_info_ds.h
>@@ -14,93 +14,86 @@
> #include "modules/video_capture/device_info_impl.h"
> #include "modules/video_capture/video_capture_impl.h"
> 
>-#include <Dshow.h>
>-
>-namespace webrtc
>-{
>-namespace videocapturemodule
>-{
>-struct VideoCaptureCapabilityWindows: public VideoCaptureCapability
>-{
>-    uint32_t directShowCapabilityIndex;
>-    bool supportFrameRateControl;
>-    VideoCaptureCapabilityWindows()
>-    {
>-        directShowCapabilityIndex = 0;
>-        supportFrameRateControl = false;
>-    }
>+#include <dshow.h>
>+
>+namespace webrtc {
>+namespace videocapturemodule {
>+struct VideoCaptureCapabilityWindows : public VideoCaptureCapability {
>+  uint32_t directShowCapabilityIndex;
>+  bool supportFrameRateControl;
>+  VideoCaptureCapabilityWindows() {
>+    directShowCapabilityIndex = 0;
>+    supportFrameRateControl = false;
>+  }
> };
> 
>-class DeviceInfoDS: public DeviceInfoImpl
>-{
>-public:
>-    // Factory function.
>-    static DeviceInfoDS* Create();
>-
>-    DeviceInfoDS();
>-    virtual ~DeviceInfoDS();
>-
>-    int32_t Init();
>-    virtual uint32_t NumberOfDevices();
>-
>-    /*
>-     * Returns the available capture devices.
>-     */
>-    virtual int32_t
>-        GetDeviceName(uint32_t deviceNumber,
>-                      char* deviceNameUTF8,
>-                      uint32_t deviceNameLength,
>-                      char* deviceUniqueIdUTF8,
>-                      uint32_t deviceUniqueIdUTF8Length,
>-                      char* productUniqueIdUTF8,
>-                      uint32_t productUniqueIdUTF8Length);
>-
>-    /*
>-     * Display OS /capture device specific settings dialog
>-     */
>-    virtual int32_t
>-        DisplayCaptureSettingsDialogBox(
>-                                        const char* deviceUniqueIdUTF8,
>-                                        const char* dialogTitleUTF8,
>-                                        void* parentWindow,
>-                                        uint32_t positionX,
>-                                        uint32_t positionY);
>-
>-    // Windows specific
>-
>-    /* Gets a capture device filter
>-     The user of this API is responsible for releasing the filter when it not needed.
>-     */
>-    IBaseFilter * GetDeviceFilter(const char* deviceUniqueIdUTF8,
>-                                  char* productUniqueIdUTF8 = NULL,
>-                                  uint32_t productUniqueIdUTF8Length = 0);
>-
>-    int32_t
>-        GetWindowsCapability(const int32_t capabilityIndex,
>-                             VideoCaptureCapabilityWindows& windowsCapability);
>-
>-    static void GetProductId(const char* devicePath,
>-                             char* productUniqueIdUTF8,
>-                             uint32_t productUniqueIdUTF8Length);
>-
>-protected:
>-    int32_t GetDeviceInfo(uint32_t deviceNumber,
>-                          char* deviceNameUTF8,
>-                          uint32_t deviceNameLength,
>-                          char* deviceUniqueIdUTF8,
>-                          uint32_t deviceUniqueIdUTF8Length,
>-                          char* productUniqueIdUTF8,
>-                          uint32_t productUniqueIdUTF8Length);
>-
>-    virtual int32_t
>-        CreateCapabilityMap(const char* deviceUniqueIdUTF8);
>-
>-private:
>-    ICreateDevEnum* _dsDevEnum;
>-    IEnumMoniker* _dsMonikerDevEnum;
>-    bool _CoUninitializeIsRequired;
>-    std::vector<VideoCaptureCapabilityWindows> _captureCapabilitiesWindows;
>+class DeviceInfoDS : public DeviceInfoImpl {
>+ public:
>+  // Factory function.
>+  static DeviceInfoDS* Create();
>+
>+  DeviceInfoDS();
>+  virtual ~DeviceInfoDS();
>+
>+  int32_t Init();
>+  virtual uint32_t NumberOfDevices();
>+
>+  /*
>+   * Returns the available capture devices.
>+   */
>+  virtual int32_t GetDeviceName(uint32_t deviceNumber,
>+                                char* deviceNameUTF8,
>+                                uint32_t deviceNameLength,
>+                                char* deviceUniqueIdUTF8,
>+                                uint32_t deviceUniqueIdUTF8Length,
>+                                char* productUniqueIdUTF8,
>+                                uint32_t productUniqueIdUTF8Length);
>+
>+  /*
>+   * Display OS /capture device specific settings dialog
>+   */
>+  virtual int32_t DisplayCaptureSettingsDialogBox(
>+      const char* deviceUniqueIdUTF8,
>+      const char* dialogTitleUTF8,
>+      void* parentWindow,
>+      uint32_t positionX,
>+      uint32_t positionY);
>+
>+  // Windows specific
>+
>+  /* Gets a capture device filter
>+   The user of this API is responsible for releasing the filter when it not
>+   needed.
>+   */
>+  IBaseFilter* GetDeviceFilter(const char* deviceUniqueIdUTF8,
>+                               char* productUniqueIdUTF8 = NULL,
>+                               uint32_t productUniqueIdUTF8Length = 0);
>+
>+  int32_t GetWindowsCapability(
>+      const int32_t capabilityIndex,
>+      VideoCaptureCapabilityWindows& windowsCapability);
>+
>+  static void GetProductId(const char* devicePath,
>+                           char* productUniqueIdUTF8,
>+                           uint32_t productUniqueIdUTF8Length);
>+
>+ protected:
>+  int32_t GetDeviceInfo(uint32_t deviceNumber,
>+                        char* deviceNameUTF8,
>+                        uint32_t deviceNameLength,
>+                        char* deviceUniqueIdUTF8,
>+                        uint32_t deviceUniqueIdUTF8Length,
>+                        char* productUniqueIdUTF8,
>+                        uint32_t productUniqueIdUTF8Length);
>+
>+  virtual int32_t CreateCapabilityMap(const char* deviceUniqueIdUTF8);
>+
>+ private:
>+  ICreateDevEnum* _dsDevEnum;
>+  IEnumMoniker* _dsMonikerDevEnum;
>+  bool _CoUninitializeIsRequired;
>+  std::vector<VideoCaptureCapabilityWindows> _captureCapabilitiesWindows;
> };
> }  // namespace videocapturemodule
> }  // namespace webrtc
>-#endif // MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_DEVICE_INFO_DS_H_
>+#endif  // MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_DEVICE_INFO_DS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/windows/device_info_mf.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/windows/device_info_mf.cc
>index 3bdd7553a5c..4002f25ac06 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/windows/device_info_mf.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/windows/device_info_mf.cc
>@@ -13,11 +13,9 @@
> namespace webrtc {
> namespace videocapturemodule {
> 
>-DeviceInfoMF::DeviceInfoMF() {
>-}
>+DeviceInfoMF::DeviceInfoMF() {}
> 
>-DeviceInfoMF::~DeviceInfoMF() {
>-}
>+DeviceInfoMF::~DeviceInfoMF() {}
> 
> int32_t DeviceInfoMF::Init() {
>   return -1;
>@@ -27,14 +25,13 @@ uint32_t DeviceInfoMF::NumberOfDevices() {
>   return 0;
> }
> 
>-int32_t DeviceInfoMF::GetDeviceName(
>-    uint32_t deviceNumber,
>-    char* deviceNameUTF8,
>-    uint32_t deviceNameLength,
>-    char* deviceUniqueIdUTF8,
>-    uint32_t deviceUniqueIdUTF8Length,
>-    char* productUniqueIdUTF8,
>-    uint32_t productUniqueIdUTF8Length) {
>+int32_t DeviceInfoMF::GetDeviceName(uint32_t deviceNumber,
>+                                    char* deviceNameUTF8,
>+                                    uint32_t deviceNameLength,
>+                                    char* deviceUniqueIdUTF8,
>+                                    uint32_t deviceUniqueIdUTF8Length,
>+                                    char* productUniqueIdUTF8,
>+                                    uint32_t productUniqueIdUTF8Length) {
>   return -1;
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/windows/device_info_mf.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/windows/device_info_mf.h
>index 027f7c44010..984d20674a2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/windows/device_info_mf.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/windows/device_info_mf.h
>@@ -25,7 +25,8 @@ class DeviceInfoMF : public DeviceInfoImpl {
>   int32_t Init();
>   virtual uint32_t NumberOfDevices();
> 
>-  virtual int32_t GetDeviceName(uint32_t deviceNumber, char* deviceNameUTF8,
>+  virtual int32_t GetDeviceName(uint32_t deviceNumber,
>+                                char* deviceNameUTF8,
>                                 uint32_t deviceNameLength,
>                                 char* deviceUniqueIdUTF8,
>                                 uint32_t deviceUniqueIdUTF8Length,
>@@ -33,8 +34,11 @@ class DeviceInfoMF : public DeviceInfoImpl {
>                                 uint32_t productUniqueIdUTF8Length);
> 
>   virtual int32_t DisplayCaptureSettingsDialogBox(
>-      const char* deviceUniqueIdUTF8, const char* dialogTitleUTF8,
>-      void* parentWindow, uint32_t positionX, uint32_t positionY);
>+      const char* deviceUniqueIdUTF8,
>+      const char* dialogTitleUTF8,
>+      void* parentWindow,
>+      uint32_t positionX,
>+      uint32_t positionY);
> };
> 
> }  // namespace videocapturemodule
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/windows/help_functions_ds.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/windows/help_functions_ds.cc
>index 12deab01281..ab85da94888 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/windows/help_functions_ds.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/windows/help_functions_ds.cc
>@@ -16,104 +16,90 @@
> 
> #include <cguid.h>
> 
>-namespace webrtc
>-{
>-namespace videocapturemodule
>-{
>+namespace webrtc {
>+namespace videocapturemodule {
> // This returns minimum :), which will give max frame rate...
>-LONGLONG GetMaxOfFrameArray(LONGLONG *maxFps, long size)
>-{
>-    LONGLONG maxFPS = maxFps[0];
>-    for (int i = 0; i < size; i++)
>-    {
>-        if (maxFPS > maxFps[i])
>-            maxFPS = maxFps[i];
>-    }
>-    return maxFPS;
>+LONGLONG GetMaxOfFrameArray(LONGLONG* maxFps, long size) {
>+  LONGLONG maxFPS = maxFps[0];
>+  for (int i = 0; i < size; i++) {
>+    if (maxFPS > maxFps[i])
>+      maxFPS = maxFps[i];
>+  }
>+  return maxFPS;
> }
> 
>-IPin* GetInputPin(IBaseFilter* filter)
>-{
>-    HRESULT hr;
>-    IPin* pin = NULL;
>-    IEnumPins* pPinEnum = NULL;
>-    filter->EnumPins(&pPinEnum);
>-    if (pPinEnum == NULL)
>-    {
>-        return NULL;
>-    }
>+IPin* GetInputPin(IBaseFilter* filter) {
>+  HRESULT hr;
>+  IPin* pin = NULL;
>+  IEnumPins* pPinEnum = NULL;
>+  filter->EnumPins(&pPinEnum);
>+  if (pPinEnum == NULL) {
>+    return NULL;
>+  }
> 
>-    // get first unconnected pin
>-    hr = pPinEnum->Reset(); // set to first pin
>+  // get first unconnected pin
>+  hr = pPinEnum->Reset();  // set to first pin
> 
>-    while (S_OK == pPinEnum->Next(1, &pin, NULL))
>+  while (S_OK == pPinEnum->Next(1, &pin, NULL)) {
>+    PIN_DIRECTION pPinDir;
>+    pin->QueryDirection(&pPinDir);
>+    if (PINDIR_INPUT == pPinDir)  // This is an input pin
>     {
>-        PIN_DIRECTION pPinDir;
>-        pin->QueryDirection(&pPinDir);
>-        if (PINDIR_INPUT == pPinDir) // This is an input pin
>-        {
>-            IPin* tempPin = NULL;
>-            if (S_OK != pin->ConnectedTo(&tempPin)) // The pint is not connected
>-            {
>-                pPinEnum->Release();
>-                return pin;
>-            }
>-        }
>-        pin->Release();
>+      IPin* tempPin = NULL;
>+      if (S_OK != pin->ConnectedTo(&tempPin))  // The pint is not connected
>+      {
>+        pPinEnum->Release();
>+        return pin;
>+      }
>     }
>-    pPinEnum->Release();
>-    return NULL;
>+    pin->Release();
>+  }
>+  pPinEnum->Release();
>+  return NULL;
> }
> 
>-IPin* GetOutputPin(IBaseFilter* filter, REFGUID Category)
>-{
>-    HRESULT hr;
>-    IPin* pin = NULL;
>-    IEnumPins* pPinEnum = NULL;
>-    filter->EnumPins(&pPinEnum);
>-    if (pPinEnum == NULL)
>-    {
>-        return NULL;
>-    }
>-    // get first unconnected pin
>-    hr = pPinEnum->Reset(); // set to first pin
>-    while (S_OK == pPinEnum->Next(1, &pin, NULL))
>+IPin* GetOutputPin(IBaseFilter* filter, REFGUID Category) {
>+  HRESULT hr;
>+  IPin* pin = NULL;
>+  IEnumPins* pPinEnum = NULL;
>+  filter->EnumPins(&pPinEnum);
>+  if (pPinEnum == NULL) {
>+    return NULL;
>+  }
>+  // get first unconnected pin
>+  hr = pPinEnum->Reset();  // set to first pin
>+  while (S_OK == pPinEnum->Next(1, &pin, NULL)) {
>+    PIN_DIRECTION pPinDir;
>+    pin->QueryDirection(&pPinDir);
>+    if (PINDIR_OUTPUT == pPinDir)  // This is an output pin
>     {
>-        PIN_DIRECTION pPinDir;
>-        pin->QueryDirection(&pPinDir);
>-        if (PINDIR_OUTPUT == pPinDir) // This is an output pin
>-        {
>-            if (Category == GUID_NULL || PinMatchesCategory(pin, Category))
>-            {
>-                pPinEnum->Release();
>-                return pin;
>-            }
>-        }
>-        pin->Release();
>-        pin = NULL;
>+      if (Category == GUID_NULL || PinMatchesCategory(pin, Category)) {
>+        pPinEnum->Release();
>+        return pin;
>+      }
>     }
>-    pPinEnum->Release();
>-    return NULL;
>+    pin->Release();
>+    pin = NULL;
>+  }
>+  pPinEnum->Release();
>+  return NULL;
> }
> 
>-BOOL PinMatchesCategory(IPin *pPin, REFGUID Category)
>-{
>-    BOOL bFound = FALSE;
>-    IKsPropertySet *pKs = NULL;
>-    HRESULT hr = pPin->QueryInterface(IID_PPV_ARGS(&pKs));
>-    if (SUCCEEDED(hr))
>-    {
>-        GUID PinCategory;
>-        DWORD cbReturned;
>-        hr = pKs->Get(AMPROPSETID_Pin, AMPROPERTY_PIN_CATEGORY, NULL, 0, &PinCategory,
>-                      sizeof(GUID), &cbReturned);
>-        if (SUCCEEDED(hr) && (cbReturned == sizeof(GUID)))
>-        {
>-            bFound = (PinCategory == Category);
>-        }
>-        pKs->Release();
>+BOOL PinMatchesCategory(IPin* pPin, REFGUID Category) {
>+  BOOL bFound = FALSE;
>+  IKsPropertySet* pKs = NULL;
>+  HRESULT hr = pPin->QueryInterface(IID_PPV_ARGS(&pKs));
>+  if (SUCCEEDED(hr)) {
>+    GUID PinCategory;
>+    DWORD cbReturned;
>+    hr = pKs->Get(AMPROPSETID_Pin, AMPROPERTY_PIN_CATEGORY, NULL, 0,
>+                  &PinCategory, sizeof(GUID), &cbReturned);
>+    if (SUCCEEDED(hr) && (cbReturned == sizeof(GUID))) {
>+      bFound = (PinCategory == Category);
>     }
>-    return bFound;
>+    pKs->Release();
>+  }
>+  return bFound;
> }
> }  // namespace videocapturemodule
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/windows/help_functions_ds.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/windows/help_functions_ds.h
>index 990f4183cc9..e77efd12371 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/windows/help_functions_ds.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/windows/help_functions_ds.h
>@@ -13,23 +13,45 @@
> 
> #include <dshow.h>
> 
>-DEFINE_GUID(MEDIASUBTYPE_I420, 0x30323449, 0x0000, 0x0010, 0x80, 0x00, 0x00,
>-            0xAA, 0x00, 0x38, 0x9B, 0x71);
>-DEFINE_GUID(MEDIASUBTYPE_HDYC, 0x43594448, 0x0000, 0x0010, 0x80, 0x00, 0x00,
>-            0xAA, 0x00, 0x38, 0x9B, 0x71);
>+DEFINE_GUID(MEDIASUBTYPE_I420,
>+            0x30323449,
>+            0x0000,
>+            0x0010,
>+            0x80,
>+            0x00,
>+            0x00,
>+            0xAA,
>+            0x00,
>+            0x38,
>+            0x9B,
>+            0x71);
>+DEFINE_GUID(MEDIASUBTYPE_HDYC,
>+            0x43594448,
>+            0x0000,
>+            0x0010,
>+            0x80,
>+            0x00,
>+            0x00,
>+            0xAA,
>+            0x00,
>+            0x38,
>+            0x9B,
>+            0x71);
> 
>-#define RELEASE_AND_CLEAR(p) if (p) { (p) -> Release () ; (p) = NULL ; }
>+#define RELEASE_AND_CLEAR(p) \
>+  if (p) {                   \
>+    (p)->Release();          \
>+    (p) = NULL;              \
>+  }
> 
>-namespace webrtc
>-{
>-namespace videocapturemodule
>-{
>-LONGLONG GetMaxOfFrameArray(LONGLONG *maxFps, long size);
>+namespace webrtc {
>+namespace videocapturemodule {
>+LONGLONG GetMaxOfFrameArray(LONGLONG* maxFps, long size);
> 
> IPin* GetInputPin(IBaseFilter* filter);
> IPin* GetOutputPin(IBaseFilter* filter, REFGUID Category);
>-BOOL PinMatchesCategory(IPin *pPin, REFGUID Category);
>+BOOL PinMatchesCategory(IPin* pPin, REFGUID Category);
> 
> }  // namespace videocapturemodule
> }  // namespace webrtc
>-#endif // MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_HELP_FUNCTIONS_DS_H_
>+#endif  // MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_HELP_FUNCTIONS_DS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/windows/sink_filter_ds.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/windows/sink_filter_ds.cc
>index 6efd0232657..c2db44b3f27 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/windows/sink_filter_ds.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/windows/sink_filter_ds.cc
>@@ -10,14 +10,13 @@
> 
> #include "modules/video_capture/windows/sink_filter_ds.h"
> 
>-#include <ios>  // std::hex
>-
> #include "modules/video_capture/windows/help_functions_ds.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/platform_thread.h"
>+#include "rtc_base/stringutils.h"
> 
>-#include <Dvdmedia.h>  // VIDEOINFOHEADER2
>+#include <dvdmedia.h>  // VIDEOINFOHEADER2
> #include <initguid.h>
> 
> #define DELETE_RESET(p) \
>@@ -148,7 +147,7 @@ CaptureInputPin::GetMediaType(IN int iPosition, OUT CMediaType* pmt) {
>   RTC_LOG(LS_INFO) << "GetMediaType position " << iPosition << ", width "
>                    << _requestedCapability.width << ", height "
>                    << _requestedCapability.height << ", biCompression 0x"
>-                   << std::hex << pvi->bmiHeader.biCompression;
>+                   << rtc::ToHex(pvi->bmiHeader.biCompression);
>   return NOERROR;
> }
> 
>@@ -188,8 +187,8 @@ CaptureInputPin::CheckMediaType(IN const CMediaType* pMediaType) {
> 
>     RTC_LOG(LS_INFO) << "CheckMediaType width:" << pvi->bmiHeader.biWidth
>                      << " height:" << pvi->bmiHeader.biHeight
>-                     << " Compression:0x" << std::hex
>-                     << pvi->bmiHeader.biCompression;
>+                     << " Compression:0x"
>+                     << rtc::ToHex(pvi->bmiHeader.biCompression);
> 
>     if (*SubType == MEDIASUBTYPE_MJPG &&
>         pvi->bmiHeader.biCompression == MAKEFOURCC('M', 'J', 'P', 'G')) {
>@@ -233,8 +232,8 @@ CaptureInputPin::CheckMediaType(IN const CMediaType* pMediaType) {
> 
>     RTC_LOG(LS_INFO) << "CheckMediaType width:" << pvi->bmiHeader.biWidth
>                      << " height:" << pvi->bmiHeader.biHeight
>-                     << " Compression:0x" << std::hex
>-                     << pvi->bmiHeader.biCompression;
>+                     << " Compression:0x"
>+                     << rtc::ToHex(pvi->bmiHeader.biCompression);
> 
>     _resultingCapability.width = pvi->bmiHeader.biWidth;
> 
>@@ -327,7 +326,7 @@ HRESULT CaptureInputPin::SetMatchingMediaType(
>   return S_OK;
> }
> //  ----------------------------------------------------------------------------
>-CaptureSinkFilter::CaptureSinkFilter(IN TCHAR* tszName,
>+CaptureSinkFilter::CaptureSinkFilter(const IN TCHAR* tszName,
>                                      IN LPUNKNOWN punk,
>                                      OUT HRESULT* phr,
>                                      VideoCaptureExternal& captureObserver)
>@@ -335,8 +334,9 @@ CaptureSinkFilter::CaptureSinkFilter(IN TCHAR* tszName,
>       m_pInput(NULL),
>       _captureObserver(captureObserver) {
>   (*phr) = S_OK;
>-  m_pInput = new CaptureInputPin(NAME("VideoCaptureInputPin"), this,
>-                                 &m_crtFilter, phr, L"VideoCapture");
>+  TCHAR inputPinName[] = L"VideoCaptureInputPin";
>+  m_pInput = new CaptureInputPin(inputPinName, this, &m_crtFilter, phr,
>+                                 L"VideoCapture");
>   if (m_pInput == NULL || FAILED(*phr)) {
>     (*phr) = FAILED(*phr) ? (*phr) : E_OUTOFMEMORY;
>     goto cleanup;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/windows/sink_filter_ds.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/windows/sink_filter_ds.h
>index af0afe93578..a73022db468 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/windows/sink_filter_ds.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/windows/sink_filter_ds.h
>@@ -11,85 +11,82 @@
> #ifndef MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_SINK_FILTER_DS_H_
> #define MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_SINK_FILTER_DS_H_
> 
>-#include <Streams.h> // Include base DS filter header files
>+#include <streams.h>  // Include base DS filter header files
> 
> #include "modules/video_capture/video_capture_defines.h"
> 
>-namespace webrtc
>-{
>-namespace videocapturemodule
>-{
>-//forward declaration
>+namespace webrtc {
>+namespace videocapturemodule {
>+// forward declaration
> 
> class CaptureSinkFilter;
> /**
>- *	input pin for camera input
>+ *     input pin for camera input
>  *
>  */
>-class CaptureInputPin: public CBaseInputPin
>-{
>-public:
>-    VideoCaptureCapability _requestedCapability;
>-    VideoCaptureCapability _resultingCapability;
>-    HANDLE _threadHandle;
>+class CaptureInputPin : public CBaseInputPin {
>+ public:
>+  VideoCaptureCapability _requestedCapability;
>+  VideoCaptureCapability _resultingCapability;
>+  HANDLE _threadHandle;
> 
>-    CaptureInputPin(IN TCHAR* szName,
>-                    IN CaptureSinkFilter* pFilter,
>-                    IN CCritSec * pLock,
>-                    OUT HRESULT * pHr,
>-                    IN LPCWSTR pszName);
>-    virtual ~CaptureInputPin();
>+  CaptureInputPin(IN TCHAR* szName,
>+                  IN CaptureSinkFilter* pFilter,
>+                  IN CCritSec* pLock,
>+                  OUT HRESULT* pHr,
>+                  IN LPCWSTR pszName);
>+  virtual ~CaptureInputPin();
> 
>-    HRESULT GetMediaType (IN int iPos, OUT CMediaType * pmt);
>-    HRESULT CheckMediaType (IN const CMediaType * pmt);
>-    STDMETHODIMP Receive (IN IMediaSample *);
>-    HRESULT SetMatchingMediaType(const VideoCaptureCapability& capability);
>+  HRESULT GetMediaType(IN int iPos, OUT CMediaType* pmt);
>+  HRESULT CheckMediaType(IN const CMediaType* pmt);
>+  STDMETHODIMP Receive(IN IMediaSample*);
>+  HRESULT SetMatchingMediaType(const VideoCaptureCapability& capability);
> };
> 
>-class CaptureSinkFilter: public CBaseFilter
>-{
>+class CaptureSinkFilter : public CBaseFilter {
>+ public:
>+  CaptureSinkFilter(const IN TCHAR* tszName,
>+                    IN LPUNKNOWN punk,
>+                    OUT HRESULT* phr,
>+                    VideoCaptureExternal& captureObserver);
>+  virtual ~CaptureSinkFilter();
> 
>-public:
>-    CaptureSinkFilter(IN TCHAR * tszName,
>-                      IN LPUNKNOWN punk,
>-                      OUT HRESULT * phr,
>-                      VideoCaptureExternal& captureObserver);
>-    virtual ~CaptureSinkFilter();
>+  //  --------------------------------------------------------------------
>+  //  class methods
> 
>-    //  --------------------------------------------------------------------
>-    //  class methods
>+  void ProcessCapturedFrame(unsigned char* pBuffer,
>+                            size_t length,
>+                            const VideoCaptureCapability& frameInfo);
>+  //  explicit receiver lock aquisition and release
>+  void LockReceive() { m_crtRecv.Lock(); }
>+  void UnlockReceive() { m_crtRecv.Unlock(); }
>+  //  explicit filter lock aquisition and release
>+  void LockFilter() { m_crtFilter.Lock(); }
>+  void UnlockFilter() { m_crtFilter.Unlock(); }
>+  void SetFilterGraph(IGraphBuilder* graph);  // Used if EVR
> 
>-    void ProcessCapturedFrame(unsigned char* pBuffer, size_t length,
>-                              const VideoCaptureCapability& frameInfo);
>-    //  explicit receiver lock aquisition and release
>-    void LockReceive()  { m_crtRecv.Lock();}
>-    void UnlockReceive() {m_crtRecv.Unlock();}
>-    //  explicit filter lock aquisition and release
>-    void LockFilter() {m_crtFilter.Lock();}
>-    void UnlockFilter() { m_crtFilter.Unlock(); }
>-    void SetFilterGraph(IGraphBuilder* graph); // Used if EVR
>+  //  --------------------------------------------------------------------
>+  //  COM interfaces
>+  DECLARE_IUNKNOWN;
>+  STDMETHODIMP SetMatchingMediaType(const VideoCaptureCapability& capability);
> 
>-    //  --------------------------------------------------------------------
>-    //  COM interfaces
>-DECLARE_IUNKNOWN    ;
>-    STDMETHODIMP SetMatchingMediaType(const VideoCaptureCapability& capability);
>+  //  --------------------------------------------------------------------
>+  //  CBaseFilter methods
>+  int GetPinCount();
>+  CBasePin* GetPin(IN int Index);
>+  STDMETHODIMP Pause();
>+  STDMETHODIMP Stop();
>+  STDMETHODIMP GetClassID(OUT CLSID* pCLSID);
>+  //  --------------------------------------------------------------------
>+  //  class factory calls this
>+  static CUnknown* CreateInstance(IN LPUNKNOWN punk, OUT HRESULT* phr);
> 
>-    //  --------------------------------------------------------------------
>-    //  CBaseFilter methods
>-    int GetPinCount ();
>-    CBasePin * GetPin ( IN int Index);
>-    STDMETHODIMP Pause ();
>-    STDMETHODIMP Stop ();
>-    STDMETHODIMP GetClassID ( OUT CLSID * pCLSID);
>-    //  --------------------------------------------------------------------
>-    //  class factory calls this
>-    static CUnknown * CreateInstance (IN LPUNKNOWN punk, OUT HRESULT * phr);
>-private:
>-    CCritSec m_crtFilter; //  filter lock
>-    CCritSec m_crtRecv;  //  receiver lock; always acquire before filter lock
>-    CaptureInputPin * m_pInput;
>-    VideoCaptureExternal& _captureObserver;
>+ private:
>+  CCritSec m_crtFilter;  //  filter lock
>+  CCritSec m_crtRecv;    //  receiver lock; always acquire before filter lock
>+  CaptureInputPin* m_pInput;
>+  VideoCaptureExternal& _captureObserver;
> };
> }  // namespace videocapturemodule
> }  // namespace webrtc
>-#endif // MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_SINK_FILTER_DS_H_
>+#endif  // MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_SINK_FILTER_DS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/windows/video_capture_ds.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/windows/video_capture_ds.cc
>index 4f4ae90d4a5..778fd8761ac 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/windows/video_capture_ds.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/windows/video_capture_ds.cc
>@@ -15,7 +15,7 @@
> #include "modules/video_capture/windows/sink_filter_ds.h"
> #include "rtc_base/logging.h"
> 
>-#include <Dvdmedia.h>  // VIDEOINFOHEADER2
>+#include <dvdmedia.h>  // VIDEOINFOHEADER2
> 
> namespace webrtc {
> namespace videocapturemodule {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/windows/video_capture_ds.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/windows/video_capture_ds.h
>index 54c5f00c655..db3d34a0847 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/windows/video_capture_ds.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/windows/video_capture_ds.h
>@@ -17,63 +17,57 @@
> #define CAPTURE_FILTER_NAME L"VideoCaptureFilter"
> #define SINK_FILTER_NAME L"SinkFilter"
> 
>-namespace webrtc
>-{
>-namespace videocapturemodule
>-{
>+namespace webrtc {
>+namespace videocapturemodule {
> // Forward declaraion
> class CaptureSinkFilter;
> 
>-class VideoCaptureDS: public VideoCaptureImpl
>-{
>-public:
>-    VideoCaptureDS();
>+class VideoCaptureDS : public VideoCaptureImpl {
>+ public:
>+  VideoCaptureDS();
> 
>-    virtual int32_t Init(const char* deviceUniqueIdUTF8);
>+  virtual int32_t Init(const char* deviceUniqueIdUTF8);
> 
>-    /*************************************************************************
>-     *
>-     *   Start/Stop
>-     *
>-     *************************************************************************/
>-    virtual int32_t
>-        StartCapture(const VideoCaptureCapability& capability);
>-    virtual int32_t StopCapture();
>+  /*************************************************************************
>+   *
>+   *   Start/Stop
>+   *
>+   *************************************************************************/
>+  virtual int32_t StartCapture(const VideoCaptureCapability& capability);
>+  virtual int32_t StopCapture();
> 
>-    /**************************************************************************
>-     *
>-     *   Properties of the set device
>-     *
>-     **************************************************************************/
>+  /**************************************************************************
>+   *
>+   *   Properties of the set device
>+   *
>+   **************************************************************************/
> 
>-    virtual bool CaptureStarted();
>-    virtual int32_t CaptureSettings(VideoCaptureCapability& settings);
>+  virtual bool CaptureStarted();
>+  virtual int32_t CaptureSettings(VideoCaptureCapability& settings);
> 
>-protected:
>-    virtual ~VideoCaptureDS();
>+ protected:
>+  virtual ~VideoCaptureDS();
> 
>-    // Help functions
>+  // Help functions
> 
>-    int32_t
>-        SetCameraOutput(const VideoCaptureCapability& requestedCapability);
>-    int32_t DisconnectGraph();
>-    HRESULT VideoCaptureDS::ConnectDVCamera();
>+  int32_t SetCameraOutput(const VideoCaptureCapability& requestedCapability);
>+  int32_t DisconnectGraph();
>+  HRESULT ConnectDVCamera();
> 
>-    DeviceInfoDS _dsInfo;
>+  DeviceInfoDS _dsInfo;
> 
>-    IBaseFilter* _captureFilter;
>-    IGraphBuilder* _graphBuilder;
>-    IMediaControl* _mediaControl;
>-    CaptureSinkFilter* _sinkFilter;
>-    IPin* _inputSendPin;
>-    IPin* _outputCapturePin;
>-
>-    // Microsoft DV interface (external DV cameras)
>-    IBaseFilter* _dvFilter;
>-    IPin* _inputDvPin;
>-    IPin* _outputDvPin;
>+  IBaseFilter* _captureFilter;
>+  IGraphBuilder* _graphBuilder;
>+  IMediaControl* _mediaControl;
>+  CaptureSinkFilter* _sinkFilter;
>+  IPin* _inputSendPin;
>+  IPin* _outputCapturePin;
> 
>+  // Microsoft DV interface (external DV cameras)
>+  IBaseFilter* _dvFilter;
>+  IPin* _inputDvPin;
>+  IPin* _outputDvPin;
> };
> }  // namespace videocapturemodule
> }  // namespace webrtc
>-#endif // MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_VIDEO_CAPTURE_DS_H_
>+#endif  // MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_VIDEO_CAPTURE_DS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/windows/video_capture_mf.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/windows/video_capture_mf.cc
>index 086b7083d9b..4c72ee96a30 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/windows/video_capture_mf.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/windows/video_capture_mf.cc
>@@ -20,8 +20,7 @@ int32_t VideoCaptureMF::Init(const char* device_id) {
>   return 0;
> }
> 
>-int32_t VideoCaptureMF::StartCapture(
>-    const VideoCaptureCapability& capability) {
>+int32_t VideoCaptureMF::StartCapture(const VideoCaptureCapability& capability) {
>   return -1;
> }
> 
>@@ -33,8 +32,7 @@ bool VideoCaptureMF::CaptureStarted() {
>   return false;
> }
> 
>-int32_t VideoCaptureMF::CaptureSettings(
>-    VideoCaptureCapability& settings) {
>+int32_t VideoCaptureMF::CaptureSettings(VideoCaptureCapability& settings) {
>   return -1;
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/windows/video_capture_mf.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/windows/video_capture_mf.h
>index 896be4e2603..ed9c382f1ed 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/windows/video_capture_mf.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_capture/windows/video_capture_mf.h
>@@ -30,8 +30,7 @@ class VideoCaptureMF : public VideoCaptureImpl {
>   virtual int32_t StartCapture(const VideoCaptureCapability& capability);
>   virtual int32_t StopCapture();
>   virtual bool CaptureStarted();
>-  virtual int32_t CaptureSettings(
>-      VideoCaptureCapability& settings);  // NOLINT
>+  virtual int32_t CaptureSettings(VideoCaptureCapability& settings);  // NOLINT
> 
>  protected:
>   virtual ~VideoCaptureMF();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/BUILD.gn
>index 5b59df7dac6..a81dd7c0f32 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/BUILD.gn
>@@ -8,17 +8,85 @@
> 
> import("../../webrtc.gni")
> 
>+rtc_static_library("encoded_frame") {
>+  visibility = [ "*" ]
>+  sources = [
>+    "encoded_frame.cc",
>+    "encoded_frame.h",
>+  ]
>+  deps = [
>+    ":video_codec_interface",
>+    "../../:webrtc_common",
>+    "../../api/video:video_frame_i420",
>+    "../../common_video:common_video",
>+    "../../modules:module_api",
>+    "../../modules:module_api_public",
>+    "../../modules/video_coding:video_coding_utility",
>+    "../../rtc_base:checks",
>+    "../../rtc_base:rtc_base_approved",
>+    "../../rtc_base/experiments:alr_experiment",
>+    "../../system_wrappers:field_trial_api",
>+    "../../system_wrappers:system_wrappers",
>+    "//third_party/abseil-cpp/absl/types:optional",
>+  ]
>+
>+  if (!build_with_chromium && is_clang) {
>+    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>+    suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>+  }
>+}
>+
>+rtc_static_library("nack_module") {
>+  visibility = [ "*" ]
>+  sources = [
>+    "histogram.cc",
>+    "histogram.h",
>+    "nack_module.cc",
>+    "nack_module.h",
>+  ]
>+
>+  if (!build_with_chromium && is_clang) {
>+    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>+    suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>+  }
>+
>+  deps = [
>+    ":packet",
>+    "..:module_api",
>+    "../../rtc_base:checks",
>+    "../../rtc_base:rtc_base_approved",
>+    "../../rtc_base:rtc_numerics",
>+    "../../system_wrappers",
>+    "../utility:utility",
>+  ]
>+}
>+
>+rtc_static_library("packet") {
>+  visibility = [ "*" ]
>+  sources = [
>+    "packet.cc",
>+    "packet.h",
>+  ]
>+  deps = [
>+    "..:module_api",
>+  ]
>+}
>+
> rtc_static_library("video_coding") {
>   visibility = [ "*" ]
>+  deps = []
>+
>   sources = [
>-    "codec_database.cc",
>-    "codec_database.h",
>     "codec_timer.cc",
>     "codec_timer.h",
>+    "decoder_database.cc",
>+    "decoder_database.h",
>     "decoding_state.cc",
>     "decoding_state.h",
>-    "encoded_frame.cc",
>-    "encoded_frame.h",
>+    "encoder_database.cc",
>+    "encoder_database.h",
>+    "fec_controller_default.cc",
>+    "fec_controller_default.h",
>     "fec_rate_table.h",
>     "frame_buffer.cc",
>     "frame_buffer.h",
>@@ -34,9 +102,8 @@ rtc_static_library("video_coding") {
>     "h264_sprop_parameter_sets.h",
>     "h264_sps_pps_tracker.cc",
>     "h264_sps_pps_tracker.h",
>-    "histogram.cc",
>-    "histogram.h",
>     "include/video_codec_initializer.h",
>+    "include/video_coding.h",
>     "inter_frame_delay.cc",
>     "inter_frame_delay.h",
>     "internal_defines.h",
>@@ -50,14 +117,8 @@ rtc_static_library("video_coding") {
>     "media_optimization.cc",
>     "media_optimization.h",
>     "nack_fec_tables.h",
>-    "nack_module.cc",
>-    "nack_module.h",
>-    "packet.cc",
>-    "packet.h",
>     "packet_buffer.cc",
>     "packet_buffer.h",
>-    "protection_bitrate_calculator.cc",
>-    "protection_bitrate_calculator.h",
>     "qp_parser.cc",
>     "qp_parser.h",
>     "receiver.cc",
>@@ -79,33 +140,27 @@ rtc_static_library("video_coding") {
>     "video_sender.cc",
>   ]
> 
>-  # TODO(jschuh): Bug 1348: fix this warning.
>-  configs += [ "//build/config/compiler:no_size_t_to_int_warning" ]
>-
>   if (!build_with_chromium && is_clang) {
>     # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>     suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>   }
> 
>-  deps = [
>+  deps += [
>     ":codec_globals_headers",
>+    ":encoded_frame",
>+    ":packet",
>     ":video_codec_interface",
>     ":video_coding_utility",
>-    ":webrtc_h264",
>-    ":webrtc_i420",
>-    ":webrtc_stereo",
>-    ":webrtc_vp8",
>-    ":webrtc_vp8_helpers",
>-    ":webrtc_vp9",
>+    ":webrtc_vp9_helpers",
>     "..:module_api",
>     "..:module_api_public",
>     "../..:webrtc_common",
>-    "../../:typedefs",
>-    "../../api:optional",
>-    "../../api:video_frame_api",
>-    "../../api:video_frame_api_i420",
>+    "../../api:fec_controller_api",
>+    "../../api/video:encoded_frame",
>+    "../../api/video:video_bitrate_allocator",
>+    "../../api/video:video_frame",
>+    "../../api/video:video_frame_i420",
>     "../../api/video_codecs:video_codecs_api",
>-    "../../call:video_stream_api",
>     "../../common_video",
>     "../../rtc_base:checks",
>     "../../rtc_base:rtc_base",
>@@ -114,11 +169,16 @@ rtc_static_library("video_coding") {
>     "../../rtc_base:rtc_task_queue",
>     "../../rtc_base:sequenced_task_checker",
>     "../../rtc_base/experiments:alr_experiment",
>+    "../../rtc_base/system:fallthrough",
>+    "../../rtc_base/third_party/base64",
>+    "../../rtc_base/time:timestamp_extrapolator",
>     "../../system_wrappers",
>     "../../system_wrappers:field_trial_api",
>     "../../system_wrappers:metrics_api",
>     "../rtp_rtcp:rtp_rtcp_format",
>     "../utility:utility",
>+    "//third_party/abseil-cpp/absl/types:optional",
>+    "//third_party/abseil-cpp/absl/types:variant",
>   ]
> }
> 
>@@ -128,12 +188,12 @@ rtc_source_set("video_codec_interface") {
>     "include/video_codec_interface.h",
>     "include/video_coding_defines.h",
>     "include/video_error_codes.h",
>+    "video_coding_defines.cc",
>   ]
>   deps = [
>     "..:module_api",
>-    "../..:typedefs",
>     "../..:webrtc_common",
>-    "../../api:video_frame_api",
>+    "../../api/video:video_frame",
>     "../../api/video_codecs:video_codecs_api",
>     "../../common_video:common_video",
>   ]
>@@ -142,14 +202,12 @@ rtc_source_set("video_codec_interface") {
> rtc_source_set("mock_headers") {
>   testonly = true
>   sources = [
>-    "codecs/interface/mock/mock_video_codec_interface.h",
>     "include/mock/mock_video_codec_interface.h",
>   ]
>   deps = [
>     ":video_codec_interface",
>     ":video_coding_utility",
>     "../..:webrtc_common",
>-    "../../:typedefs",
>     "../../test:test_support",
>   ]
> }
>@@ -158,17 +216,18 @@ rtc_source_set("codec_globals_headers") {
>   sources = [
>     "codecs/h264/include/h264_globals.h",
>     "codecs/interface/common_constants.h",
>-    "codecs/stereo/include/stereo_globals.h",
>     "codecs/vp8/include/vp8_globals.h",
>     "codecs/vp9/include/vp9_globals.h",
>   ]
>+
>+  deps = [
>+    "../../rtc_base:checks",
>+  ]
> }
> 
> rtc_source_set("video_coding_utility") {
>   visibility = [ "*" ]
>   sources = [
>-    "codecs/interface/video_codec_interface.h",
>-    "codecs/interface/video_error_codes.h",
>     "utility/default_video_bitrate_allocator.cc",
>     "utility/default_video_bitrate_allocator.h",
>     "utility/frame_dropper.cc",
>@@ -179,6 +238,10 @@ rtc_source_set("video_coding_utility") {
>     "utility/moving_average.h",
>     "utility/quality_scaler.cc",
>     "utility/quality_scaler.h",
>+    "utility/simulcast_rate_allocator.cc",
>+    "utility/simulcast_rate_allocator.h",
>+    "utility/simulcast_utility.cc",
>+    "utility/simulcast_utility.h",
>     "utility/vp8_header_parser.cc",
>     "utility/vp8_header_parser.h",
>     "utility/vp9_uncompressed_header_parser.cc",
>@@ -194,8 +257,7 @@ rtc_source_set("video_coding_utility") {
>     ":video_codec_interface",
>     "..:module_api",
>     "../..:webrtc_common",
>-    "../../:typedefs",
>-    "../../api:optional",
>+    "../../api/video:video_bitrate_allocator",
>     "../../api/video_codecs:video_codecs_api",
>     "../../common_video",
>     "../../modules/rtp_rtcp",
>@@ -204,8 +266,13 @@ rtc_source_set("video_coding_utility") {
>     "../../rtc_base:rtc_numerics",
>     "../../rtc_base:rtc_task_queue",
>     "../../rtc_base:sequenced_task_checker",
>+    "../../rtc_base/experiments:quality_scaling_experiment",
>+    "../../rtc_base/system:arch",
>     "../../system_wrappers",
>+    "../../system_wrappers:field_trial_api",
>+    "../../system_wrappers:metrics_api",
>     "../rtp_rtcp:rtp_rtcp_format",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
> 
>@@ -225,21 +292,23 @@ rtc_static_library("webrtc_h264") {
>   deps = [
>     ":video_codec_interface",
>     ":video_coding_utility",
>-    "../../api:video_frame_api_i420",
>+    "../../api/video:video_frame",
>+    "../../api/video:video_frame_i420",
>     "../../api/video_codecs:video_codecs_api",
>     "../../media:rtc_h264_profile_id",
>     "../../media:rtc_media_base",
>     "../../rtc_base:checks",
>     "../../rtc_base:rtc_base",
>     "../../system_wrappers:metrics_api",
>+    "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/libyuv",
>   ]
> 
>   if (rtc_use_h264) {
>     defines += [ "WEBRTC_USE_H264" ]
>-    if (rtc_initialize_ffmpeg) {
>-      defines += [ "WEBRTC_INITIALIZE_FFMPEG" ]
>-    }
>     sources += [
>+      "codecs/h264/h264_color_space.cc",
>+      "codecs/h264/h264_color_space.h",
>       "codecs/h264/h264_decoder_impl.cc",
>       "codecs/h264/h264_decoder_impl.h",
>       "codecs/h264/h264_encoder_impl.cc",
>@@ -251,11 +320,6 @@ rtc_static_library("webrtc_h264") {
>       "//third_party/openh264:encoder",
>     ]
> 
>-    # TODO(bugs.webrtc.org/8745): stop using av_lockmgr_register
>-    if (!is_win) {
>-      cflags = [ "-Wno-deprecated-declarations" ]
>-    }
>-
>     if (!build_with_mozilla) {
>       deps += [ "../../media:rtc_media_base" ]
>     }
>@@ -278,8 +342,7 @@ rtc_static_library("webrtc_i420") {
>     ":video_codec_interface",
>     ":video_coding_utility",
>     "../..:webrtc_common",
>-    "../../:typedefs",
>-    "../../api:video_frame_api_i420",
>+    "../../api/video:video_frame_i420",
>     "../../common_video:common_video",
>     "../../rtc_base:checks",
>     "../../rtc_base:rtc_base_approved",
>@@ -288,12 +351,16 @@ rtc_static_library("webrtc_i420") {
>   ]
> }
> 
>-rtc_static_library("webrtc_stereo") {
>+rtc_static_library("webrtc_multiplex") {
>   sources = [
>-    "codecs/stereo/include/stereo_decoder_adapter.h",
>-    "codecs/stereo/include/stereo_encoder_adapter.h",
>-    "codecs/stereo/stereo_decoder_adapter.cc",
>-    "codecs/stereo/stereo_encoder_adapter.cc",
>+    "codecs/multiplex/augmented_video_frame_buffer.cc",
>+    "codecs/multiplex/include/augmented_video_frame_buffer.h",
>+    "codecs/multiplex/include/multiplex_decoder_adapter.h",
>+    "codecs/multiplex/include/multiplex_encoded_image_packer.h",
>+    "codecs/multiplex/include/multiplex_encoder_adapter.h",
>+    "codecs/multiplex/multiplex_decoder_adapter.cc",
>+    "codecs/multiplex/multiplex_encoded_image_packer.cc",
>+    "codecs/multiplex/multiplex_encoder_adapter.cc",
>   ]
> 
>   if (!build_with_chromium && is_clang) {
>@@ -306,57 +373,34 @@ rtc_static_library("webrtc_stereo") {
>     ":video_coding_utility",
>     "..:module_api",
>     "../..:webrtc_common",
>-    "../../api:video_frame_api",
>-    "../../api:video_frame_api_i420",
>+    "../../api/video:video_frame",
>+    "../../api/video:video_frame_i420",
>     "../../api/video_codecs:video_codecs_api",
>     "../../common_video:common_video",
>     "../../rtc_base:rtc_base",
>     "../../system_wrappers",
>-  ]
>-}
>-
>-# This target includes VP8 files that may be used for any VP8 codec, internal SW or external HW.
>-rtc_source_set("webrtc_vp8_helpers") {
>-  sources = [
>-    "codecs/vp8/include/vp8_common_types.h",
>-    "codecs/vp8/screenshare_layers.h",
>-    "codecs/vp8/simulcast_rate_allocator.h",
>-    "codecs/vp8/temporal_layers.h",
>-  ]
>-
>-  deps = [
>-    ":video_coding_utility",
>-    "..:module_api",
>-    "../..:webrtc_common",
>-    "../../:typedefs",
>-    "../../api/video_codecs:video_codecs_api",
>-    "../../common_video",
>-    "../../rtc_base:checks",
>-    "../../rtc_base:rtc_base_approved",
>-    "../../system_wrappers",
>-    "../../system_wrappers:field_trial_api",
>-    "../../system_wrappers:metrics_api",
>+    "../rtp_rtcp:rtp_rtcp_format",
>   ]
> }
> 
> # This target includes the internal SW codec.
> rtc_static_library("webrtc_vp8") {
>   visibility = [ "*" ]
>+  poisonous = [ "software_video_codecs" ]
>   sources = [
>     "codecs/vp8/default_temporal_layers.cc",
>     "codecs/vp8/default_temporal_layers.h",
>     "codecs/vp8/include/vp8.h",
>-    "codecs/vp8/include/vp8_common_types.h",
>+    "codecs/vp8/libvpx_vp8_decoder.cc",
>+    "codecs/vp8/libvpx_vp8_decoder.h",
>+    "codecs/vp8/libvpx_vp8_encoder.cc",
>+    "codecs/vp8/libvpx_vp8_encoder.h",
>     "codecs/vp8/screenshare_layers.cc",
>-    "codecs/vp8/simulcast_rate_allocator.cc",
>+    "codecs/vp8/screenshare_layers.h",
>     "codecs/vp8/temporal_layers.cc",
>-    "codecs/vp8/vp8_impl.cc",
>-    "codecs/vp8/vp8_impl.h",
>+    "codecs/vp8/temporal_layers.h",
>   ]
> 
>-  # TODO(jschuh): Bug 1348: fix this warning.
>-  configs += [ "//build/config/compiler:no_size_t_to_int_warning" ]
>-
>   if (!build_with_chromium && is_clang) {
>     # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>     suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>@@ -365,12 +409,10 @@ rtc_static_library("webrtc_vp8") {
>   deps = [
>     ":video_codec_interface",
>     ":video_coding_utility",
>-    ":webrtc_vp8_helpers",
>     "..:module_api",
>     "../..:webrtc_common",
>-    "../../:typedefs",
>-    "../../api:optional",
>-    "../../api:video_frame_api",
>+    "../../api/video:video_frame",
>+    "../../api/video_codecs:video_codecs_api",
>     "../../common_video",
>     "../../rtc_base:checks",
>     "../../rtc_base:rtc_base_approved",
>@@ -378,6 +420,8 @@ rtc_static_library("webrtc_vp8") {
>     "../../system_wrappers",
>     "../../system_wrappers:field_trial_api",
>     "../../system_wrappers:metrics_api",
>+    "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/types:optional",
>     "//third_party/libyuv",
>   ]
>   if (rtc_build_libvpx) {
>@@ -385,13 +429,36 @@ rtc_static_library("webrtc_vp8") {
>   }
> }
> 
>+# This target includes VP9 files that may be used for any VP9 codec, internal SW or external HW.
>+rtc_static_library("webrtc_vp9_helpers") {
>+  sources = [
>+    "codecs/vp9/svc_config.cc",
>+    "codecs/vp9/svc_config.h",
>+    "codecs/vp9/svc_rate_allocator.cc",
>+    "codecs/vp9/svc_rate_allocator.h",
>+  ]
>+
>+  if (!build_with_chromium && is_clang) {
>+    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>+    suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>+  }
>+
>+  deps = [
>+    ":video_codec_interface",
>+    "../..:webrtc_common",
>+    "../../api/video:video_bitrate_allocator",
>+    "../../api/video_codecs:video_codecs_api",
>+    "../../common_video",
>+    "../../rtc_base:checks",
>+  ]
>+}
>+
> rtc_static_library("webrtc_vp9") {
>   visibility = [ "*" ]
>+  poisonous = [ "software_video_codecs" ]
>   if (rtc_libvpx_build_vp9) {
>     sources = [
>       "codecs/vp9/include/vp9.h",
>-      "codecs/vp9/screenshare_layers.cc",
>-      "codecs/vp9/screenshare_layers.h",
>       "codecs/vp9/vp9_frame_buffer_pool.cc",
>       "codecs/vp9/vp9_frame_buffer_pool.h",
>       "codecs/vp9/vp9_impl.cc",
>@@ -411,11 +478,20 @@ rtc_static_library("webrtc_vp9") {
>   deps = [
>     ":video_codec_interface",
>     ":video_coding_utility",
>+    ":webrtc_vp9_helpers",
>     "..:module_api",
>+    "../..:webrtc_common",
>+    "../../api/video:video_frame",
>+    "../../api/video:video_frame_i010",
>+    "../../api/video_codecs:video_codecs_api",
>     "../../common_video",
>+    "../../media:rtc_media_base",
>+    "../../media:rtc_vp9_profile",
>     "../../rtc_base:checks",
>     "../../rtc_base:rtc_base",
>     "../../system_wrappers",
>+    "../rtp_rtcp:rtp_rtcp_format",
>+    "//third_party/abseil-cpp/absl/memory",
>   ]
>   if (rtc_build_libvpx) {
>     deps += [ rtc_libvpx_dir ]
>@@ -423,23 +499,45 @@ rtc_static_library("webrtc_vp9") {
> }
> 
> if (rtc_include_tests) {
>+  if (is_android) {
>+    rtc_static_library("android_codec_factory_helper") {
>+      sources = [
>+        "codecs/test/android_codec_factory_helper.cc",
>+        "codecs/test/android_codec_factory_helper.h",
>+      ]
>+
>+      deps = [
>+        "../../api/video_codecs:video_codecs_api",
>+        "../../rtc_base:checks",
>+        "../../rtc_base:rtc_base_approved",
>+        "../../sdk/android:native_api_base",
>+        "../../sdk/android:native_api_codecs",
>+        "../../sdk/android:native_api_jni",
>+        "//base",
>+      ]
>+    }
>+  }
>+
>   if (is_ios || is_mac) {
>-    rtc_static_library("codec_test_objc") {
>+    rtc_static_library("objc_codec_factory_helper") {
>       sources = [
>-        "codecs/test/objc_codec_h264_test.h",
>-        "codecs/test/objc_codec_h264_test.mm",
>+        "codecs/test/objc_codec_factory_helper.h",
>+        "codecs/test/objc_codec_factory_helper.mm",
>       ]
>+
>       deps = [
>-        "../../api:video_frame_api",
>+        "../../api/video:video_frame",
>         "../../api/video_codecs:video_codecs_api",
>         "../../media:rtc_audio_video",
>+        "../../media:rtc_media_base",
>         "../../modules:module_api",
>         "../../rtc_base:rtc_base_approved",
>         "../../sdk:common_objc",
>-        "../../sdk:peerconnection_objc",
>-        "../../sdk:peerconnectionfactory_objc",
>+        "../../sdk:native_api",
>+        "../../sdk:peerconnectionfactory_base_objc",
>+        "../../sdk:videocodec_objc",
>+        "../../sdk:videosource_objc",
>         "../../sdk:videotoolbox_objc",
>-        "../../sdk:videotracksource_objc",
>       ]
> 
>       if (!build_with_chromium && is_clang) {
>@@ -449,10 +547,11 @@ if (rtc_include_tests) {
>     }
>   }
> 
>-  rtc_source_set("simulcast_test_utility") {
>+  rtc_source_set("simulcast_test_fixture_impl") {
>     testonly = true
>     sources = [
>-      "codecs/vp8/simulcast_test_utility.h",
>+      "utility/simulcast_test_fixture_impl.cc",
>+      "utility/simulcast_test_fixture_impl.h",
>     ]
> 
>     if (!build_with_chromium && is_clang) {
>@@ -464,10 +563,12 @@ if (rtc_include_tests) {
>       ":mock_headers",
>       ":video_codec_interface",
>       ":video_coding",
>-      ":webrtc_vp8",
>-      ":webrtc_vp8_helpers",
>-      "../../api:video_frame_api",
>-      "../../api:video_frame_api_i420",
>+      ":video_coding_utility",
>+      "../../:webrtc_common",
>+      "../../api:simulcast_test_fixture_api",
>+      "../../api/video:video_frame",
>+      "../../api/video:video_frame_i420",
>+      "../../api/video_codecs:video_codecs_api",
>       "../../common_video:common_video",
>       "../../rtc_base:checks",
>       "../../rtc_base:rtc_base_approved",
>@@ -478,15 +579,8 @@ if (rtc_include_tests) {
>   rtc_source_set("video_codecs_test_framework") {
>     testonly = true
>     sources = [
>-      "codecs/test/mock/mock_packet_manipulator.h",
>-      "codecs/test/packet_manipulator.cc",
>-      "codecs/test/packet_manipulator.h",
>-      "codecs/test/stats.cc",
>-      "codecs/test/stats.h",
>-      "codecs/test/test_config.cc",
>-      "codecs/test/test_config.h",
>-      "codecs/test/video_codec_test.cc",
>-      "codecs/test/video_codec_test.h",
>+      "codecs/test/video_codec_unittest.cc",
>+      "codecs/test/video_codec_unittest.h",
>       "codecs/test/videoprocessor.cc",
>       "codecs/test/videoprocessor.h",
>     ]
>@@ -497,43 +591,104 @@ if (rtc_include_tests) {
>     }
> 
>     deps = [
>-      ":codec_globals_headers",
>       ":video_codec_interface",
>       ":video_coding",
>       ":video_coding_utility",
>-      ":webrtc_vp8",
>-      ":webrtc_vp8_helpers",
>+      ":webrtc_vp9_helpers",
>       "../..:webrtc_common",
>-      "../../:typedefs",
>-      "../../api:video_frame_api",
>-      "../../api:video_frame_api_i420",
>+      "../../api:videocodec_test_fixture_api",
>+      "../../api/video:video_bitrate_allocator",
>+      "../../api/video:video_frame",
>+      "../../api/video:video_frame_i420",
>       "../../api/video_codecs:video_codecs_api",
>       "../../common_video:common_video",
>       "../../rtc_base:checks",
>       "../../rtc_base:rtc_base_approved",
>       "../../rtc_base:rtc_task_queue",
>+      "../../rtc_base:rtc_task_queue_for_test",
>       "../../rtc_base:sequenced_task_checker",
>-      "../../system_wrappers",
>       "../../test:test_support",
>       "../../test:video_test_common",
>       "../../test:video_test_support",
>+      "../rtp_rtcp:rtp_rtcp_format",
>+      "//third_party/libyuv",
>     ]
>   }
> 
>-  video_coding_modules_tests_resources = [
>-    "../../resources/foreman_128x96.yuv",
>-    "../../resources/foreman_160x120.yuv",
>-    "../../resources/foreman_176x144.yuv",
>-    "../../resources/foreman_320x240.yuv",
>-  ]
>+  video_coding_modules_tests_resources = []
>+  if (is_android) {
>+    video_coding_modules_tests_resources += [
>+      "../../resources/foreman_128x96.yuv",
>+      "../../resources/foreman_160x120.yuv",
>+      "../../resources/foreman_176x144.yuv",
>+      "../../resources/foreman_240x136.yuv",
>+      "../../resources/foreman_320x240.yuv",
>+      "../../resources/foreman_480x272.yuv",
>+    ]
>+  }
>+  if (!is_android) {
>+    video_coding_modules_tests_resources += [
>+      "../../resources/FourPeople_1280x720_30.yuv",
>+      "../../resources/ConferenceMotion_1280_720_50.yuv",
>+    ]
>+  }
> 
>-  if (is_ios || is_mac) {
>-    bundle_data("video_coding_modules_tests_resources_bundle_data") {
>-      testonly = true
>-      sources = video_coding_modules_tests_resources
>-      outputs = [
>-        "{{bundle_resources_dir}}/{{source_file_part}}",
>-      ]
>+  num_video_coding_modules_tests_resources = 0
>+  foreach(i, video_coding_modules_tests_resources) {
>+    num_video_coding_modules_tests_resources += 1
>+  }
>+
>+  if (num_video_coding_modules_tests_resources > 0) {
>+    if (is_ios || is_mac) {
>+      bundle_data("video_coding_modules_tests_resources_bundle_data") {
>+        testonly = true
>+        sources = video_coding_modules_tests_resources
>+        outputs = [
>+          "{{bundle_resources_dir}}/{{source_file_part}}",
>+        ]
>+      }
>+    }
>+  }
>+
>+  rtc_source_set("videocodec_test_impl") {
>+    testonly = true
>+    sources = [
>+      "codecs/test/videocodec_test_fixture_impl.cc",
>+      "codecs/test/videocodec_test_fixture_impl.h",
>+      "codecs/test/videocodec_test_stats_impl.cc",
>+      "codecs/test/videocodec_test_stats_impl.h",
>+    ]
>+    deps = [
>+      ":video_codec_interface",
>+      ":video_codecs_test_framework",
>+      ":video_coding_utility",
>+      ":webrtc_vp9_helpers",
>+      "../..:webrtc_common",
>+      "../../api:videocodec_test_fixture_api",
>+      "../../api/video_codecs:video_codecs_api",
>+      "../../call:video_stream_api",
>+      "../../common_video",
>+      "../../media:rtc_audio_video",
>+      "../../media:rtc_h264_profile_id",
>+      "../../media:rtc_internal_video_codecs",
>+      "../../media:rtc_media_base",
>+      "../../rtc_base:checks",
>+      "../../rtc_base:rtc_base_approved",
>+      "../../rtc_base:rtc_base_tests_utils",
>+      "../../rtc_base:rtc_event",
>+      "../../rtc_base:rtc_task_queue_for_test",
>+      "../../system_wrappers",
>+      "../../test:fileutils",
>+      "../../test:test_common",
>+      "../../test:test_support",
>+      "../../test:video_test_common",
>+      "../../test:video_test_support",
>+      "../rtp_rtcp:rtp_rtcp_format",
>+      "//third_party/abseil-cpp/absl/memory",
>+    ]
>+    if (!build_with_chromium && is_clang) {
>+      # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>+      suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>     }
>   }
> 
>@@ -542,77 +697,66 @@ if (rtc_include_tests) {
> 
>     sources = [
>       "codecs/h264/test/h264_impl_unittest.cc",
>-      "codecs/stereo/test/stereo_adapter_unittest.cc",
>-      "codecs/test/videoprocessor_integrationtest.cc",
>-      "codecs/test/videoprocessor_integrationtest.h",
>-      "codecs/test/videoprocessor_integrationtest_libvpx.cc",
>-      "codecs/test/videoprocessor_integrationtest_openh264.cc",
>-      "codecs/test/videoprocessor_integrationtest_parameterized.cc",
>+      "codecs/multiplex/test/multiplex_adapter_unittest.cc",
>+      "codecs/test/video_encoder_decoder_instantiation_tests.cc",
>+      "codecs/test/videocodec_test_libvpx.cc",
>+      "codecs/test/videocodec_test_parameterized.cc",
>       "codecs/vp8/test/vp8_impl_unittest.cc",
>       "codecs/vp9/test/vp9_impl_unittest.cc",
>     ]
>+    if (rtc_use_h264) {
>+      sources += [ "codecs/test/videocodec_test_openh264.cc" ]
>+    }
> 
>     deps = [
>-      ":video_codec_interface",
>       ":video_codecs_test_framework",
>-      ":video_coding",
>       ":video_coding_utility",
>+      ":videocodec_test_impl",
>       ":webrtc_h264",
>-      ":webrtc_stereo",
>+      ":webrtc_multiplex",
>       ":webrtc_vp8",
>-      ":webrtc_vp8_helpers",
>       ":webrtc_vp9",
>+      ":webrtc_vp9_helpers",
>       "../..:webrtc_common",
>+      "../../api:create_videocodec_test_fixture_api",
>       "../../api:mock_video_codec_factory",
>-      "../../api:optional",
>-      "../../api:video_frame_api",
>-      "../../api:video_frame_api_i420",
>+      "../../api:videocodec_test_fixture_api",
>+      "../../api/video:video_frame",
>+      "../../api/video:video_frame_i420",
>+      "../../api/video_codecs:rtc_software_fallback_wrappers",
>       "../../api/video_codecs:video_codecs_api",
>       "../../common_video",
>-      "../../media:rtc_audio_video",
>       "../../media:rtc_h264_profile_id",
>       "../../media:rtc_internal_video_codecs",
>       "../../media:rtc_media_base",
>-      "../../rtc_base:checks",
>+      "../../media:rtc_vp9_profile",
>       "../../rtc_base:rtc_base",
>-      "../../rtc_base:rtc_base_tests_utils",
>-      "../../system_wrappers",
>-      "../../test:field_trial",
>+      "../../test:fileutils",
>+      "../../test:test_common",
>       "../../test:test_support",
>       "../../test:video_test_common",
>-      "../../test:video_test_support",
>-      "../video_capture",
>+      "../rtp_rtcp:rtp_rtcp_format",
>+      "//third_party/abseil-cpp/absl/memory",
>+      "//third_party/abseil-cpp/absl/types:optional",
>+      "//third_party/libyuv",
>     ]
> 
>     data = video_coding_modules_tests_resources
> 
>     if (is_android) {
>-      sources += [
>-        "codecs/test/android_test_initializer.cc",
>-        "codecs/test/android_test_initializer.h",
>-        "codecs/test/videoprocessor_integrationtest_mediacodec.cc",
>-      ]
>+      sources += [ "codecs/test/videocodec_test_mediacodec.cc" ]
> 
>-      deps += [
>-        "../../sdk/android:libjingle_peerconnection_jni",
>-        "//base",
>-      ]
>+      deps += [ ":android_codec_factory_helper" ]
>     }
> 
>     if (is_ios || is_mac) {
>-      sources +=
>-          [ "codecs/test/videoprocessor_integrationtest_videotoolbox.cc" ]
>+      sources += [ "codecs/test/videocodec_test_videotoolbox.cc" ]
> 
>-      deps += [
>-        ":codec_test_objc",
>-        ":video_coding_modules_tests_resources_bundle_data",
>-        "../../media:rtc_media_base",
>-        "../../sdk:videotoolbox_objc",
>-      ]
>-    }
>+      deps += [ ":objc_codec_factory_helper" ]
> 
>-    if (rtc_use_h264) {
>-      defines = [ "WEBRTC_USE_H264" ]
>+      if (num_video_coding_modules_tests_resources > 0) {
>+        deps += [ ":video_coding_modules_tests_resources_bundle_data" ]
>+      }
>     }
> 
>     if (!build_with_chromium && is_clang) {
>@@ -625,14 +769,16 @@ if (rtc_include_tests) {
>     testonly = true
> 
>     sources = [
>-      "codecs/test/packet_manipulator_unittest.cc",
>-      "codecs/test/stats_unittest.cc",
>-      "codecs/test/test_config_unittest.cc",
>+      "codecs/test/videocodec_test_fixture_config_unittest.cc",
>+      "codecs/test/videocodec_test_stats_impl_unittest.cc",
>       "codecs/test/videoprocessor_unittest.cc",
>       "codecs/vp8/default_temporal_layers_unittest.cc",
>+      "codecs/vp8/libvpx_vp8_simulcast_test.cc",
>       "codecs/vp8/screenshare_layers_unittest.cc",
>-      "codecs/vp8/simulcast_unittest.cc",
>+      "codecs/vp9/svc_config_unittest.cc",
>+      "codecs/vp9/svc_rate_allocator_unittest.cc",
>       "decoding_state_unittest.cc",
>+      "fec_controller_unittest.cc",
>       "frame_buffer2_unittest.cc",
>       "generic_encoder_unittest.cc",
>       "h264_sprop_parameter_sets_unittest.cc",
>@@ -642,7 +788,6 @@ if (rtc_include_tests) {
>       "jitter_buffer_unittest.cc",
>       "jitter_estimator_tests.cc",
>       "nack_module_unittest.cc",
>-      "protection_bitrate_calculator_unittest.cc",
>       "receiver_unittest.cc",
>       "rtp_frame_reference_finder_unittest.cc",
>       "session_info_unittest.cc",
>@@ -662,58 +807,63 @@ if (rtc_include_tests) {
>       "video_receiver_unittest.cc",
>       "video_sender_unittest.cc",
>     ]
>-    if (rtc_libvpx_build_vp9) {
>-      sources += [ "codecs/vp9/vp9_screenshare_layers_unittest.cc" ]
>-    }
>     if (rtc_use_h264) {
>-      sources += [ "codecs/h264/h264_encoder_impl_unittest.cc" ]
>+      sources += [
>+        "codecs/h264/h264_encoder_impl_unittest.cc",
>+        "codecs/h264/h264_simulcast_unittest.cc",
>+      ]
>     }
>+
>     deps = [
>+      ":codec_globals_headers",
>+      ":encoded_frame",
>       ":mock_headers",
>-      ":simulcast_test_utility",
>+      ":nack_module",
>+      ":packet",
>+      ":simulcast_test_fixture_impl",
>       ":video_codec_interface",
>       ":video_codecs_test_framework",
>       ":video_coding",
>       ":video_coding_utility",
>+      ":videocodec_test_impl",
>       ":webrtc_h264",
>       ":webrtc_vp8",
>-      ":webrtc_vp8_helpers",
>       ":webrtc_vp9",
>+      ":webrtc_vp9_helpers",
>       "..:module_api",
>       "../..:webrtc_common",
>-      "../../:typedefs",
>-      "../../api:video_frame_api",
>-      "../../api:video_frame_api_i420",
>+      "../../api:create_simulcast_test_fixture_api",
>+      "../../api:simulcast_test_fixture_api",
>+      "../../api:videocodec_test_fixture_api",
>+      "../../api/video:video_bitrate_allocator",
>+      "../../api/video:video_frame",
>+      "../../api/video:video_frame_i420",
>       "../../api/video_codecs:video_codecs_api",
>       "../../common_video:common_video",
>+      "../../media:rtc_media_base",
>       "../../rtc_base:checks",
>       "../../rtc_base:rtc_base",
>       "../../rtc_base:rtc_base_approved",
>+      "../../rtc_base:rtc_base_tests_utils",
>       "../../rtc_base:rtc_numerics",
>       "../../rtc_base:rtc_task_queue",
>+      "../../rtc_base:rtc_task_queue_for_test",
>       "../../system_wrappers",
>       "../../system_wrappers:field_trial_api",
>       "../../system_wrappers:metrics_api",
>       "../../system_wrappers:metrics_default",
>       "../../test:field_trial",
>+      "../../test:fileutils",
>+      "../../test:test_common",
>       "../../test:test_support",
>       "../../test:video_test_common",
>       "../../test:video_test_support",
>       "../rtp_rtcp:rtp_rtcp_format",
>-      "//testing/gmock",
>+      "//third_party/abseil-cpp/absl/memory",
>     ]
>     if (rtc_build_libvpx) {
>       deps += [ rtc_libvpx_dir ]
>     }
>-    if (is_win) {
>-      cflags = [
>-        # TODO(kjellander): bugs.webrtc.org/261: Fix this warning.
>-        "/wd4373",  # virtual function override.
>-      ]
>-    }
>-
>-    # TODO(jschuh): bugs.webrtc.org/1348: fix this warning.
>-    configs += [ "//build/config/compiler:no_size_t_to_int_warning" ]
>     if (!build_with_chromium && is_clang) {
>       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/DEPS b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/DEPS
>index 9a6afbf318f..98242bed416 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/DEPS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/DEPS
>@@ -2,20 +2,18 @@ include_rules = [
>   "+vpx",
>   "+call",
>   "+common_video",
>-  "+experiments",
>+  "+sdk",
>   "+system_wrappers",
>   "+rtc_tools",
>   "+third_party/libyuv",
> ]
> 
> specific_include_rules = {
>-  "android_test_initializer\.cc": [
>+  "android_codec_factory_helper\.cc": [
>     "+base/android",
>-    "+sdk",
>   ],
>-  "(.*test\.cc|.*test\.h|.*test\.mm)": [
>+  ".*test.*\.cc": [
>      "+media/base",
>      "+media/engine",
>-     "+sdk",
>   ],
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/OWNERS
>index 0ebe326c6e0..24c38e379ca 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/OWNERS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/OWNERS
>@@ -1,7 +1,8 @@
> asapersson@webrtc.org
> brandtr@webrtc.org
>-sprang@webrtc.org
> marpan@webrtc.org
>+philipel@webrtc.org
>+sprang@webrtc.org
> stefan@webrtc.org
> 
> # These are for the common case of adding or renaming files. If you're doing
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codec_database.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codec_database.cc
>deleted file mode 100644
>index e300ad3d079..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codec_database.cc
>+++ /dev/null
>@@ -1,463 +0,0 @@
>-/*
>- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "modules/video_coding/codec_database.h"
>-
>-#include "modules/video_coding/codecs/h264/include/h264.h"
>-#include "modules/video_coding/codecs/i420/include/i420.h"
>-#include "modules/video_coding/codecs/vp8/include/vp8.h"
>-#include "modules/video_coding/codecs/vp9/include/vp9.h"
>-#include "modules/video_coding/internal_defines.h"
>-#include "rtc_base/checks.h"
>-#include "rtc_base/logging.h"
>-
>-namespace {
>-const size_t kDefaultPayloadSize = 1440;
>-}
>-
>-namespace webrtc {
>-
>-// Create an internal Decoder given a codec type
>-static std::unique_ptr<VCMGenericDecoder> CreateDecoder(VideoCodecType type) {
>-  switch (type) {
>-    case kVideoCodecVP8:
>-      return std::unique_ptr<VCMGenericDecoder>(
>-          new VCMGenericDecoder(VP8Decoder::Create()));
>-    case kVideoCodecVP9:
>-      return std::unique_ptr<VCMGenericDecoder>(
>-          new VCMGenericDecoder(VP9Decoder::Create()));
>-    case kVideoCodecI420:
>-      return std::unique_ptr<VCMGenericDecoder>(
>-          new VCMGenericDecoder(new I420Decoder()));
>-    case kVideoCodecH264:
>-      if (H264Decoder::IsSupported()) {
>-        return std::unique_ptr<VCMGenericDecoder>(
>-            new VCMGenericDecoder(H264Decoder::Create()));
>-      }
>-      break;
>-    default:
>-      break;
>-  }
>-  RTC_LOG(LS_WARNING) << "No internal decoder of this type exists.";
>-  return std::unique_ptr<VCMGenericDecoder>();
>-}
>-
>-VCMDecoderMapItem::VCMDecoderMapItem(VideoCodec* settings,
>-                                     int number_of_cores,
>-                                     bool require_key_frame)
>-    : settings(settings),
>-      number_of_cores(number_of_cores),
>-      require_key_frame(require_key_frame) {
>-  RTC_DCHECK_GE(number_of_cores, 0);
>-}
>-
>-VCMExtDecoderMapItem::VCMExtDecoderMapItem(
>-    VideoDecoder* external_decoder_instance,
>-    uint8_t payload_type)
>-    : payload_type(payload_type),
>-      external_decoder_instance(external_decoder_instance) {}
>-
>-VCMCodecDataBase::VCMCodecDataBase(
>-    VCMEncodedFrameCallback* encoded_frame_callback)
>-    : number_of_cores_(0),
>-      max_payload_size_(kDefaultPayloadSize),
>-      periodic_key_frames_(false),
>-      pending_encoder_reset_(true),
>-      send_codec_(),
>-      receive_codec_(),
>-      encoder_payload_type_(0),
>-      external_encoder_(nullptr),
>-      internal_source_(false),
>-      encoded_frame_callback_(encoded_frame_callback),
>-      dec_map_(),
>-      dec_external_map_() {}
>-
>-VCMCodecDataBase::~VCMCodecDataBase() {
>-  DeleteEncoder();
>-  ptr_decoder_.reset();
>-  for (auto& kv : dec_map_)
>-    delete kv.second;
>-  for (auto& kv : dec_external_map_)
>-    delete kv.second;
>-}
>-
>-// Assuming only one registered encoder - since only one used, no need for more.
>-bool VCMCodecDataBase::SetSendCodec(const VideoCodec* send_codec,
>-                                    int number_of_cores,
>-                                    size_t max_payload_size) {
>-  RTC_DCHECK(send_codec);
>-  if (max_payload_size == 0) {
>-    max_payload_size = kDefaultPayloadSize;
>-  }
>-  RTC_DCHECK_GE(number_of_cores, 1);
>-  RTC_DCHECK_GE(send_codec->plType, 1);
>-  // Make sure the start bit rate is sane...
>-  RTC_DCHECK_LE(send_codec->startBitrate, 1000000);
>-  RTC_DCHECK(send_codec->codecType != kVideoCodecUnknown);
>-  bool reset_required = pending_encoder_reset_;
>-  if (number_of_cores_ != number_of_cores) {
>-    number_of_cores_ = number_of_cores;
>-    reset_required = true;
>-  }
>-  if (max_payload_size_ != max_payload_size) {
>-    max_payload_size_ = max_payload_size;
>-    reset_required = true;
>-  }
>-
>-  VideoCodec new_send_codec;
>-  memcpy(&new_send_codec, send_codec, sizeof(new_send_codec));
>-
>-  if (new_send_codec.maxBitrate == 0) {
>-    // max is one bit per pixel
>-    new_send_codec.maxBitrate = (static_cast<int>(send_codec->height) *
>-                                 static_cast<int>(send_codec->width) *
>-                                 static_cast<int>(send_codec->maxFramerate)) /
>-                                1000;
>-    if (send_codec->startBitrate > new_send_codec.maxBitrate) {
>-      // But if the user tries to set a higher start bit rate we will
>-      // increase the max accordingly.
>-      new_send_codec.maxBitrate = send_codec->startBitrate;
>-    }
>-  }
>-
>-  if (new_send_codec.startBitrate > new_send_codec.maxBitrate)
>-    new_send_codec.startBitrate = new_send_codec.maxBitrate;
>-
>-  if (!reset_required) {
>-    reset_required = RequiresEncoderReset(new_send_codec);
>-  }
>-
>-  memcpy(&send_codec_, &new_send_codec, sizeof(send_codec_));
>-
>-  if (!reset_required) {
>-    return true;
>-  }
>-
>-  // If encoder exists, will destroy it and create new one.
>-  DeleteEncoder();
>-  RTC_DCHECK_EQ(encoder_payload_type_, send_codec_.plType)
>-      << "Encoder not registered for payload type " << send_codec_.plType;
>-  ptr_encoder_.reset(new VCMGenericEncoder(
>-      external_encoder_, encoded_frame_callback_, internal_source_));
>-  encoded_frame_callback_->SetInternalSource(internal_source_);
>-  if (ptr_encoder_->InitEncode(&send_codec_, number_of_cores_,
>-                               max_payload_size_) < 0) {
>-    RTC_LOG(LS_ERROR) << "Failed to initialize video encoder.";
>-    DeleteEncoder();
>-    return false;
>-  }
>-
>-  // Intentionally don't check return value since the encoder registration
>-  // shouldn't fail because the codec doesn't support changing the periodic key
>-  // frame setting.
>-  ptr_encoder_->SetPeriodicKeyFrames(periodic_key_frames_);
>-
>-  pending_encoder_reset_ = false;
>-
>-  return true;
>-}
>-
>-bool VCMCodecDataBase::SendCodec(VideoCodec* current_send_codec) const {
>-  if (!ptr_encoder_) {
>-    return false;
>-  }
>-  memcpy(current_send_codec, &send_codec_, sizeof(VideoCodec));
>-  return true;
>-}
>-
>-VideoCodecType VCMCodecDataBase::SendCodec() const {
>-  if (!ptr_encoder_) {
>-    return kVideoCodecUnknown;
>-  }
>-  return send_codec_.codecType;
>-}
>-
>-bool VCMCodecDataBase::DeregisterExternalEncoder(uint8_t payload_type,
>-                                                 bool* was_send_codec) {
>-  RTC_DCHECK(was_send_codec);
>-  *was_send_codec = false;
>-  if (encoder_payload_type_ != payload_type) {
>-    return false;
>-  }
>-  if (send_codec_.plType == payload_type) {
>-    // De-register as send codec if needed.
>-    DeleteEncoder();
>-    memset(&send_codec_, 0, sizeof(VideoCodec));
>-    *was_send_codec = true;
>-  }
>-  encoder_payload_type_ = 0;
>-  external_encoder_ = nullptr;
>-  internal_source_ = false;
>-  return true;
>-}
>-
>-void VCMCodecDataBase::RegisterExternalEncoder(VideoEncoder* external_encoder,
>-                                               uint8_t payload_type,
>-                                               bool internal_source) {
>-  // Since only one encoder can be used at a given time, only one external
>-  // encoder can be registered/used.
>-  external_encoder_ = external_encoder;
>-  encoder_payload_type_ = payload_type;
>-  internal_source_ = internal_source;
>-  pending_encoder_reset_ = true;
>-}
>-
>-bool VCMCodecDataBase::RequiresEncoderReset(const VideoCodec& new_send_codec) {
>-  if (!ptr_encoder_)
>-    return true;
>-
>-  // Does not check startBitrate or maxFramerate
>-  if (new_send_codec.codecType != send_codec_.codecType ||
>-      strcmp(new_send_codec.plName, send_codec_.plName) != 0 ||
>-      new_send_codec.plType != send_codec_.plType ||
>-      new_send_codec.width != send_codec_.width ||
>-      new_send_codec.height != send_codec_.height ||
>-      new_send_codec.maxBitrate != send_codec_.maxBitrate ||
>-      new_send_codec.minBitrate != send_codec_.minBitrate ||
>-      new_send_codec.qpMax != send_codec_.qpMax ||
>-      new_send_codec.numberOfSimulcastStreams !=
>-          send_codec_.numberOfSimulcastStreams ||
>-      new_send_codec.mode != send_codec_.mode) {
>-    return true;
>-  }
>-
>-  switch (new_send_codec.codecType) {
>-    case kVideoCodecVP8:
>-      if (memcmp(&new_send_codec.VP8(), send_codec_.VP8(),
>-                 sizeof(new_send_codec.VP8())) != 0) {
>-        return true;
>-      }
>-      break;
>-    case kVideoCodecVP9:
>-      if (memcmp(&new_send_codec.VP9(), send_codec_.VP9(),
>-                 sizeof(new_send_codec.VP9())) != 0) {
>-        return true;
>-      }
>-      break;
>-    case kVideoCodecH264:
>-      if (memcmp(&new_send_codec.H264(), send_codec_.H264(),
>-                 sizeof(new_send_codec.H264())) != 0) {
>-        return true;
>-      }
>-      break;
>-    case kVideoCodecGeneric:
>-      break;
>-    // Known codecs without payload-specifics
>-    case kVideoCodecI420:
>-    case kVideoCodecRED:
>-    case kVideoCodecULPFEC:
>-    case kVideoCodecFlexfec:
>-    case kVideoCodecStereo:
>-      break;
>-    // Unknown codec type, reset just to be sure.
>-    case kVideoCodecUnknown:
>-      return true;
>-  }
>-
>-  if (new_send_codec.numberOfSimulcastStreams > 0) {
>-    for (unsigned char i = 0; i < new_send_codec.numberOfSimulcastStreams;
>-         ++i) {
>-      if (memcmp(&new_send_codec.simulcastStream[i],
>-                 &send_codec_.simulcastStream[i],
>-                 sizeof(new_send_codec.simulcastStream[i])) != 0) {
>-        return true;
>-      }
>-    }
>-  }
>-  return false;
>-}
>-
>-VCMGenericEncoder* VCMCodecDataBase::GetEncoder() {
>-  return ptr_encoder_.get();
>-}
>-
>-bool VCMCodecDataBase::SetPeriodicKeyFrames(bool enable) {
>-  periodic_key_frames_ = enable;
>-  if (ptr_encoder_) {
>-    return (ptr_encoder_->SetPeriodicKeyFrames(periodic_key_frames_) == 0);
>-  }
>-  return true;
>-}
>-
>-bool VCMCodecDataBase::DeregisterExternalDecoder(uint8_t payload_type) {
>-  ExternalDecoderMap::iterator it = dec_external_map_.find(payload_type);
>-  if (it == dec_external_map_.end()) {
>-    // Not found
>-    return false;
>-  }
>-  // We can't use payload_type to check if the decoder is currently in use,
>-  // because payload type may be out of date (e.g. before we decode the first
>-  // frame after RegisterReceiveCodec)
>-  if (ptr_decoder_ &&
>-      ptr_decoder_->IsSameDecoder((*it).second->external_decoder_instance)) {
>-    // Release it if it was registered and in use.
>-    ptr_decoder_.reset();
>-  }
>-  DeregisterReceiveCodec(payload_type);
>-  delete it->second;
>-  dec_external_map_.erase(it);
>-  return true;
>-}
>-
>-// Add the external encoder object to the list of external decoders.
>-// Won't be registered as a receive codec until RegisterReceiveCodec is called.
>-void VCMCodecDataBase::RegisterExternalDecoder(VideoDecoder* external_decoder,
>-                                               uint8_t payload_type) {
>-  // Check if payload value already exists, if so  - erase old and insert new.
>-  VCMExtDecoderMapItem* ext_decoder =
>-      new VCMExtDecoderMapItem(external_decoder, payload_type);
>-  DeregisterExternalDecoder(payload_type);
>-  dec_external_map_[payload_type] = ext_decoder;
>-}
>-
>-bool VCMCodecDataBase::DecoderRegistered() const {
>-  return !dec_map_.empty();
>-}
>-
>-bool VCMCodecDataBase::RegisterReceiveCodec(const VideoCodec* receive_codec,
>-                                            int number_of_cores,
>-                                            bool require_key_frame) {
>-  if (number_of_cores < 0) {
>-    return false;
>-  }
>-  // Check if payload value already exists, if so  - erase old and insert new.
>-  DeregisterReceiveCodec(receive_codec->plType);
>-  if (receive_codec->codecType == kVideoCodecUnknown) {
>-    return false;
>-  }
>-  VideoCodec* new_receive_codec = new VideoCodec(*receive_codec);
>-  dec_map_[receive_codec->plType] = new VCMDecoderMapItem(
>-      new_receive_codec, number_of_cores, require_key_frame);
>-  return true;
>-}
>-
>-bool VCMCodecDataBase::DeregisterReceiveCodec(uint8_t payload_type) {
>-  DecoderMap::iterator it = dec_map_.find(payload_type);
>-  if (it == dec_map_.end()) {
>-    return false;
>-  }
>-  delete it->second;
>-  dec_map_.erase(it);
>-  if (receive_codec_.plType == payload_type) {
>-    // This codec is currently in use.
>-    memset(&receive_codec_, 0, sizeof(VideoCodec));
>-  }
>-  return true;
>-}
>-
>-VCMGenericDecoder* VCMCodecDataBase::GetDecoder(
>-    const VCMEncodedFrame& frame,
>-    VCMDecodedFrameCallback* decoded_frame_callback) {
>-  RTC_DCHECK(decoded_frame_callback->UserReceiveCallback());
>-  uint8_t payload_type = frame.PayloadType();
>-  if (payload_type == receive_codec_.plType || payload_type == 0) {
>-    return ptr_decoder_.get();
>-  }
>-  // Check for exisitng decoder, if exists - delete.
>-  if (ptr_decoder_) {
>-    ptr_decoder_.reset();
>-    memset(&receive_codec_, 0, sizeof(VideoCodec));
>-  }
>-  ptr_decoder_ = CreateAndInitDecoder(frame, &receive_codec_);
>-  if (!ptr_decoder_) {
>-    return nullptr;
>-  }
>-  VCMReceiveCallback* callback = decoded_frame_callback->UserReceiveCallback();
>-  callback->OnIncomingPayloadType(receive_codec_.plType);
>-  if (ptr_decoder_->RegisterDecodeCompleteCallback(decoded_frame_callback) <
>-      0) {
>-    ptr_decoder_.reset();
>-    memset(&receive_codec_, 0, sizeof(VideoCodec));
>-    return nullptr;
>-  }
>-  return ptr_decoder_.get();
>-}
>-
>-VCMGenericDecoder* VCMCodecDataBase::GetCurrentDecoder() {
>-  return ptr_decoder_.get();
>-}
>-
>-bool VCMCodecDataBase::PrefersLateDecoding() const {
>-  return ptr_decoder_ ? ptr_decoder_->PrefersLateDecoding() : true;
>-}
>-
>-bool VCMCodecDataBase::MatchesCurrentResolution(int width, int height) const {
>-  return send_codec_.width == width && send_codec_.height == height;
>-}
>-
>-std::unique_ptr<VCMGenericDecoder> VCMCodecDataBase::CreateAndInitDecoder(
>-    const VCMEncodedFrame& frame,
>-    VideoCodec* new_codec) const {
>-  uint8_t payload_type = frame.PayloadType();
>-  RTC_LOG(LS_INFO) << "Initializing decoder with payload type '"
>-                   << static_cast<int>(payload_type) << "'.";
>-  RTC_DCHECK(new_codec);
>-  const VCMDecoderMapItem* decoder_item = FindDecoderItem(payload_type);
>-  if (!decoder_item) {
>-    RTC_LOG(LS_ERROR) << "Can't find a decoder associated with payload type: "
>-                      << static_cast<int>(payload_type);
>-    return nullptr;
>-  }
>-  std::unique_ptr<VCMGenericDecoder> ptr_decoder;
>-  const VCMExtDecoderMapItem* external_dec_item =
>-      FindExternalDecoderItem(payload_type);
>-  if (external_dec_item) {
>-    // External codec.
>-    ptr_decoder.reset(new VCMGenericDecoder(
>-        external_dec_item->external_decoder_instance, true));
>-  } else {
>-    // Create decoder.
>-    ptr_decoder = CreateDecoder(decoder_item->settings->codecType);
>-  }
>-  if (!ptr_decoder)
>-    return nullptr;
>-
>-  // Copy over input resolutions to prevent codec reinitialization due to
>-  // the first frame being of a different resolution than the database values.
>-  // This is best effort, since there's no guarantee that width/height have been
>-  // parsed yet (and may be zero).
>-  if (frame.EncodedImage()._encodedWidth > 0 &&
>-      frame.EncodedImage()._encodedHeight > 0) {
>-    decoder_item->settings->width = frame.EncodedImage()._encodedWidth;
>-    decoder_item->settings->height = frame.EncodedImage()._encodedHeight;
>-  }
>-  if (ptr_decoder->InitDecode(decoder_item->settings.get(),
>-                              decoder_item->number_of_cores) < 0) {
>-    return nullptr;
>-  }
>-  memcpy(new_codec, decoder_item->settings.get(), sizeof(VideoCodec));
>-  return ptr_decoder;
>-}
>-
>-void VCMCodecDataBase::DeleteEncoder() {
>-  if (!ptr_encoder_)
>-    return;
>-  ptr_encoder_->Release();
>-  ptr_encoder_.reset();
>-}
>-
>-const VCMDecoderMapItem* VCMCodecDataBase::FindDecoderItem(
>-    uint8_t payload_type) const {
>-  DecoderMap::const_iterator it = dec_map_.find(payload_type);
>-  if (it != dec_map_.end()) {
>-    return (*it).second;
>-  }
>-  return nullptr;
>-}
>-
>-const VCMExtDecoderMapItem* VCMCodecDataBase::FindExternalDecoderItem(
>-    uint8_t payload_type) const {
>-  ExternalDecoderMap::const_iterator it = dec_external_map_.find(payload_type);
>-  if (it != dec_external_map_.end()) {
>-    return (*it).second;
>-  }
>-  return nullptr;
>-}
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codec_database.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codec_database.h
>deleted file mode 100644
>index bfab4c2e535..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codec_database.h
>+++ /dev/null
>@@ -1,153 +0,0 @@
>-/*
>- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MODULES_VIDEO_CODING_CODEC_DATABASE_H_
>-#define MODULES_VIDEO_CODING_CODEC_DATABASE_H_
>-
>-#include <map>
>-#include <memory>
>-
>-#include "modules/video_coding/include/video_codec_interface.h"
>-#include "modules/video_coding/include/video_coding.h"
>-#include "modules/video_coding/generic_decoder.h"
>-#include "modules/video_coding/generic_encoder.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>-
>-namespace webrtc {
>-
>-struct VCMDecoderMapItem {
>- public:
>-  VCMDecoderMapItem(VideoCodec* settings,
>-                    int number_of_cores,
>-                    bool require_key_frame);
>-
>-  std::unique_ptr<VideoCodec> settings;
>-  int number_of_cores;
>-  bool require_key_frame;
>-};
>-
>-struct VCMExtDecoderMapItem {
>- public:
>-  VCMExtDecoderMapItem(VideoDecoder* external_decoder_instance,
>-                       uint8_t payload_type);
>-
>-  uint8_t payload_type;
>-  VideoDecoder* external_decoder_instance;
>-};
>-
>-class VCMCodecDataBase {
>- public:
>-  explicit VCMCodecDataBase(VCMEncodedFrameCallback* encoded_frame_callback);
>-  ~VCMCodecDataBase();
>-
>-  // Sets the sender side codec and initiates the desired codec given the
>-  // VideoCodec struct.
>-  // Returns true if the codec was successfully registered, false otherwise.
>-  bool SetSendCodec(const VideoCodec* send_codec,
>-                    int number_of_cores,
>-                    size_t max_payload_size);
>-
>-  // Gets the current send codec. Relevant for internal codecs only.
>-  // Returns true if there is a send codec, false otherwise.
>-  bool SendCodec(VideoCodec* current_send_codec) const;
>-
>-  // Gets current send side codec type. Relevant for internal codecs only.
>-  // Returns kVideoCodecUnknown if there is no send codec.
>-  VideoCodecType SendCodec() const;
>-
>-  // Registers and initializes an external encoder object.
>-  // |internal_source| should be set to true if the codec has an internal
>-  // video source and doesn't need the user to provide it with frames via
>-  // the Encode() method.
>-  void RegisterExternalEncoder(VideoEncoder* external_encoder,
>-                               uint8_t payload_type,
>-                               bool internal_source);
>-
>-  // Deregisters an external encoder. Returns true if the encoder was
>-  // found and deregistered, false otherwise. |was_send_codec| is set to true
>-  // if the external encoder was the send codec before being deregistered.
>-  bool DeregisterExternalEncoder(uint8_t payload_type, bool* was_send_codec);
>-
>-  VCMGenericEncoder* GetEncoder();
>-
>-  bool SetPeriodicKeyFrames(bool enable);
>-
>-  // Deregisters an external decoder object specified by |payload_type|.
>-  bool DeregisterExternalDecoder(uint8_t payload_type);
>-
>-  // Registers an external decoder object to the payload type |payload_type|.
>-  void RegisterExternalDecoder(VideoDecoder* external_decoder,
>-                               uint8_t payload_type);
>-
>-  bool DecoderRegistered() const;
>-
>-  bool RegisterReceiveCodec(const VideoCodec* receive_codec,
>-                            int number_of_cores,
>-                            bool require_key_frame);
>-
>-  bool DeregisterReceiveCodec(uint8_t payload_type);
>-
>-  // Returns a decoder specified by |payload_type|. The decoded frame callback
>-  // of the encoder is set to |decoded_frame_callback|. If no such decoder
>-  // already exists an instance will be created and initialized.
>-  // NULL is returned if no encoder with the specified payload type was found
>-  // and the function failed to create one.
>-  VCMGenericDecoder* GetDecoder(
>-      const VCMEncodedFrame& frame,
>-      VCMDecodedFrameCallback* decoded_frame_callback);
>-
>-  // Returns the current decoder (i.e. the same value as was last returned from
>-  // GetDecoder();
>-  VCMGenericDecoder* GetCurrentDecoder();
>-
>-  // Returns true if the currently active decoder prefer to decode frames late.
>-  // That means that frames must be decoded near the render times stamp.
>-  bool PrefersLateDecoding() const;
>-
>-  bool MatchesCurrentResolution(int width, int height) const;
>-
>- private:
>-  typedef std::map<uint8_t, VCMDecoderMapItem*> DecoderMap;
>-  typedef std::map<uint8_t, VCMExtDecoderMapItem*> ExternalDecoderMap;
>-
>-  std::unique_ptr<VCMGenericDecoder> CreateAndInitDecoder(
>-      const VCMEncodedFrame& frame,
>-      VideoCodec* new_codec) const;
>-
>-  // Determines whether a new codec has to be created or not.
>-  // Checks every setting apart from maxFramerate and startBitrate.
>-  bool RequiresEncoderReset(const VideoCodec& send_codec);
>-
>-  void DeleteEncoder();
>-
>-  const VCMDecoderMapItem* FindDecoderItem(uint8_t payload_type) const;
>-
>-  const VCMExtDecoderMapItem* FindExternalDecoderItem(
>-      uint8_t payload_type) const;
>-
>-  int number_of_cores_;
>-  size_t max_payload_size_;
>-  bool periodic_key_frames_;
>-  bool pending_encoder_reset_;
>-  VideoCodec send_codec_;
>-  VideoCodec receive_codec_;
>-  uint8_t encoder_payload_type_;
>-  VideoEncoder* external_encoder_;
>-  bool internal_source_;
>-  VCMEncodedFrameCallback* const encoded_frame_callback_;
>-  std::unique_ptr<VCMGenericEncoder> ptr_encoder_;
>-  std::unique_ptr<VCMGenericDecoder> ptr_decoder_;
>-  DecoderMap dec_map_;
>-  ExternalDecoderMap dec_external_map_;
>-};  // VCMCodecDataBase
>-
>-}  // namespace webrtc
>-
>-#endif  // MODULES_VIDEO_CODING_CODEC_DATABASE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codec_timer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codec_timer.cc
>index 5a9b6e4e786..725f02501d2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codec_timer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codec_timer.cc
>@@ -25,6 +25,7 @@ const int64_t kTimeLimitMs = 10000;
> 
> VCMCodecTimer::VCMCodecTimer()
>     : ignored_sample_count_(0), filter_(kPercentile) {}
>+VCMCodecTimer::~VCMCodecTimer() = default;
> 
> void VCMCodecTimer::AddTiming(int64_t decode_time_ms, int64_t now_ms) {
>   // Ignore the first |kIgnoredSampleCount| samples.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codec_timer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codec_timer.h
>index d112959660e..e5d68395044 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codec_timer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codec_timer.h
>@@ -15,13 +15,13 @@
> 
> #include "modules/include/module_common_types.h"
> #include "rtc_base/numerics/percentile_filter.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
> class VCMCodecTimer {
>  public:
>   VCMCodecTimer();
>+  ~VCMCodecTimer();
> 
>   // Add a new decode time to the filter.
>   void AddTiming(int64_t new_decode_time_ms, int64_t now_ms);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/h264.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/h264.cc
>index 7771b13a2f9..aff9f7dee61 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/h264.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/h264.cc
>@@ -19,9 +19,9 @@
> #include "modules/video_coding/codecs/h264/h264_encoder_impl.h"
> #endif
> 
>+#include "absl/memory/memory.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/ptr_util.h"
> 
> namespace webrtc {
> 
>@@ -40,14 +40,17 @@ bool IsH264CodecSupported() {
> #endif
> }
> 
>-SdpVideoFormat CreateH264Format(H264::Profile profile, H264::Level level) {
>-  const rtc::Optional<std::string> profile_string =
>+SdpVideoFormat CreateH264Format(H264::Profile profile,
>+                                H264::Level level,
>+                                const std::string& packetization_mode) {
>+  const absl::optional<std::string> profile_string =
>       H264::ProfileLevelIdToString(H264::ProfileLevelId(profile, level));
>   RTC_CHECK(profile_string);
>-  return SdpVideoFormat(cricket::kH264CodecName,
>-                        {{cricket::kH264FmtpProfileLevelId, *profile_string},
>-                         {cricket::kH264FmtpLevelAsymmetryAllowed, "1"},
>-                         {cricket::kH264FmtpPacketizationMode, "1"}});
>+  return SdpVideoFormat(
>+      cricket::kH264CodecName,
>+      {{cricket::kH264FmtpProfileLevelId, *profile_string},
>+       {cricket::kH264FmtpLevelAsymmetryAllowed, "1"},
>+       {cricket::kH264FmtpPacketizationMode, packetization_mode}});
> }
> 
> }  // namespace
>@@ -67,8 +70,15 @@ std::vector<SdpVideoFormat> SupportedH264Codecs() {
>   // decoder for that profile is required to be able to decode CBP. This means
>   // we can encode and send CBP even though we negotiated a potentially
>   // higher profile. See the H264 spec for more information.
>-  return {CreateH264Format(H264::kProfileBaseline, H264::kLevel3_1),
>-          CreateH264Format(H264::kProfileConstrainedBaseline, H264::kLevel3_1)};
>+  //
>+  // We support both packetization modes 0 (mandatory) and 1 (optional,
>+  // preferred).
>+  return {
>+      CreateH264Format(H264::kProfileBaseline, H264::kLevel3_1, "1"),
>+      CreateH264Format(H264::kProfileBaseline, H264::kLevel3_1, "0"),
>+      CreateH264Format(H264::kProfileConstrainedBaseline, H264::kLevel3_1, "1"),
>+      CreateH264Format(H264::kProfileConstrainedBaseline, H264::kLevel3_1,
>+                       "0")};
> }
> 
> std::unique_ptr<H264Encoder> H264Encoder::Create(
>@@ -77,7 +87,7 @@ std::unique_ptr<H264Encoder> H264Encoder::Create(
> #if defined(WEBRTC_USE_H264)
>   RTC_CHECK(g_rtc_use_h264);
>   RTC_LOG(LS_INFO) << "Creating H264EncoderImpl.";
>-  return rtc::MakeUnique<H264EncoderImpl>(codec);
>+  return absl::make_unique<H264EncoderImpl>(codec);
> #else
>   RTC_NOTREACHED();
>   return nullptr;
>@@ -93,7 +103,7 @@ std::unique_ptr<H264Decoder> H264Decoder::Create() {
> #if defined(WEBRTC_USE_H264)
>   RTC_CHECK(g_rtc_use_h264);
>   RTC_LOG(LS_INFO) << "Creating H264DecoderImpl.";
>-  return rtc::MakeUnique<H264DecoderImpl>();
>+  return absl::make_unique<H264DecoderImpl>();
> #else
>   RTC_NOTREACHED();
>   return nullptr;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/h264_color_space.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/h264_color_space.cc
>new file mode 100644
>index 00000000000..b48ccfb4b5f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/h264_color_space.cc
>@@ -0,0 +1,171 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/video_coding/codecs/h264/h264_color_space.h"
>+
>+namespace webrtc {
>+
>+ColorSpace ExtractH264ColorSpace(AVCodecContext* codec) {
>+  ColorSpace::PrimaryID primaries = ColorSpace::PrimaryID::kInvalid;
>+  switch (codec->color_primaries) {
>+    case AVCOL_PRI_BT709:
>+      primaries = ColorSpace::PrimaryID::kBT709;
>+      break;
>+    case AVCOL_PRI_BT470M:
>+      primaries = ColorSpace::PrimaryID::kBT470M;
>+      break;
>+    case AVCOL_PRI_BT470BG:
>+      primaries = ColorSpace::PrimaryID::kBT470BG;
>+      break;
>+    case AVCOL_PRI_SMPTE170M:
>+      primaries = ColorSpace::PrimaryID::kSMPTE170M;
>+      break;
>+    case AVCOL_PRI_SMPTE240M:
>+      primaries = ColorSpace::PrimaryID::kSMPTE240M;
>+      break;
>+    case AVCOL_PRI_FILM:
>+      primaries = ColorSpace::PrimaryID::kFILM;
>+      break;
>+    case AVCOL_PRI_BT2020:
>+      primaries = ColorSpace::PrimaryID::kBT2020;
>+      break;
>+    case AVCOL_PRI_SMPTE428:
>+      primaries = ColorSpace::PrimaryID::kSMPTEST428;
>+      break;
>+    case AVCOL_PRI_SMPTE431:
>+      primaries = ColorSpace::PrimaryID::kSMPTEST431;
>+      break;
>+    case AVCOL_PRI_SMPTE432:
>+      primaries = ColorSpace::PrimaryID::kSMPTEST432;
>+      break;
>+    case AVCOL_PRI_JEDEC_P22:
>+      primaries = ColorSpace::PrimaryID::kJEDECP22;
>+      break;
>+    case AVCOL_PRI_RESERVED0:
>+    case AVCOL_PRI_UNSPECIFIED:
>+    case AVCOL_PRI_RESERVED:
>+    default:
>+      break;
>+  }
>+
>+  ColorSpace::TransferID transfer = ColorSpace::TransferID::kInvalid;
>+  switch (codec->color_trc) {
>+    case AVCOL_TRC_BT709:
>+      transfer = ColorSpace::TransferID::kBT709;
>+      break;
>+    case AVCOL_TRC_GAMMA22:
>+      transfer = ColorSpace::TransferID::kGAMMA22;
>+      break;
>+    case AVCOL_TRC_GAMMA28:
>+      transfer = ColorSpace::TransferID::kGAMMA28;
>+      break;
>+    case AVCOL_TRC_SMPTE170M:
>+      transfer = ColorSpace::TransferID::kSMPTE170M;
>+      break;
>+    case AVCOL_TRC_SMPTE240M:
>+      transfer = ColorSpace::TransferID::kSMPTE240M;
>+      break;
>+    case AVCOL_TRC_LINEAR:
>+      transfer = ColorSpace::TransferID::kLINEAR;
>+      break;
>+    case AVCOL_TRC_LOG:
>+      transfer = ColorSpace::TransferID::kLOG;
>+      break;
>+    case AVCOL_TRC_LOG_SQRT:
>+      transfer = ColorSpace::TransferID::kLOG_SQRT;
>+      break;
>+    case AVCOL_TRC_IEC61966_2_4:
>+      transfer = ColorSpace::TransferID::kIEC61966_2_4;
>+      break;
>+    case AVCOL_TRC_BT1361_ECG:
>+      transfer = ColorSpace::TransferID::kBT1361_ECG;
>+      break;
>+    case AVCOL_TRC_IEC61966_2_1:
>+      transfer = ColorSpace::TransferID::kIEC61966_2_1;
>+      break;
>+    case AVCOL_TRC_BT2020_10:
>+      transfer = ColorSpace::TransferID::kBT2020_10;
>+      break;
>+    case AVCOL_TRC_BT2020_12:
>+      transfer = ColorSpace::TransferID::kBT2020_12;
>+      break;
>+    case AVCOL_TRC_SMPTE2084:
>+      transfer = ColorSpace::TransferID::kSMPTEST2084;
>+      break;
>+    case AVCOL_TRC_SMPTE428:
>+      transfer = ColorSpace::TransferID::kSMPTEST428;
>+      break;
>+    case AVCOL_TRC_ARIB_STD_B67:
>+      transfer = ColorSpace::TransferID::kARIB_STD_B67;
>+      break;
>+    case AVCOL_TRC_RESERVED0:
>+    case AVCOL_TRC_UNSPECIFIED:
>+    case AVCOL_TRC_RESERVED:
>+    default:
>+      break;
>+  }
>+
>+  ColorSpace::MatrixID matrix = ColorSpace::MatrixID::kInvalid;
>+  switch (codec->colorspace) {
>+    case AVCOL_SPC_RGB:
>+      matrix = ColorSpace::MatrixID::kRGB;
>+      break;
>+    case AVCOL_SPC_BT709:
>+      matrix = ColorSpace::MatrixID::kBT709;
>+      break;
>+    case AVCOL_SPC_FCC:
>+      matrix = ColorSpace::MatrixID::kFCC;
>+      break;
>+    case AVCOL_SPC_BT470BG:
>+      matrix = ColorSpace::MatrixID::kBT470BG;
>+      break;
>+    case AVCOL_SPC_SMPTE170M:
>+      matrix = ColorSpace::MatrixID::kSMPTE170M;
>+      break;
>+    case AVCOL_SPC_SMPTE240M:
>+      matrix = ColorSpace::MatrixID::kSMPTE240M;
>+      break;
>+    case AVCOL_SPC_YCGCO:
>+      matrix = ColorSpace::MatrixID::kYCOCG;
>+      break;
>+    case AVCOL_SPC_BT2020_NCL:
>+      matrix = ColorSpace::MatrixID::kBT2020_NCL;
>+      break;
>+    case AVCOL_SPC_BT2020_CL:
>+      matrix = ColorSpace::MatrixID::kBT2020_CL;
>+      break;
>+    case AVCOL_SPC_SMPTE2085:
>+      matrix = ColorSpace::MatrixID::kSMPTE2085;
>+      break;
>+    case AVCOL_SPC_CHROMA_DERIVED_NCL:
>+    case AVCOL_SPC_CHROMA_DERIVED_CL:
>+    case AVCOL_SPC_ICTCP:
>+    case AVCOL_SPC_UNSPECIFIED:
>+    case AVCOL_SPC_RESERVED:
>+    default:
>+      break;
>+  }
>+
>+  ColorSpace::RangeID range = ColorSpace::RangeID::kInvalid;
>+  switch (codec->color_range) {
>+    case AVCOL_RANGE_MPEG:
>+      range = ColorSpace::RangeID::kLimited;
>+      break;
>+    case AVCOL_RANGE_JPEG:
>+      range = ColorSpace::RangeID::kFull;
>+      break;
>+    case AVCOL_RANGE_UNSPECIFIED:
>+    default:
>+      break;
>+  }
>+  return ColorSpace(primaries, transfer, matrix, range);
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/h264_color_space.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/h264_color_space.h
>new file mode 100644
>index 00000000000..5608ab1f055
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/h264_color_space.h
>@@ -0,0 +1,27 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_VIDEO_CODING_CODECS_H264_H264_COLOR_SPACE_H_
>+#define MODULES_VIDEO_CODING_CODECS_H264_H264_COLOR_SPACE_H_
>+
>+#include "api/video/color_space.h"
>+
>+extern "C" {
>+#include "third_party/ffmpeg/libavcodec/avcodec.h"
>+}  // extern "C"
>+
>+namespace webrtc {
>+
>+// Helper class for extracting color space information from H264 stream.
>+ColorSpace ExtractH264ColorSpace(AVCodecContext* codec);
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_VIDEO_CODING_CODECS_H264_H264_COLOR_SPACE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/h264_decoder_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/h264_decoder_impl.cc
>index 2544fb737f9..ae3de069c5d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/h264_decoder_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/h264_decoder_impl.cc
>@@ -20,8 +20,10 @@ extern "C" {
> #include "third_party/ffmpeg/libavutil/imgutils.h"
> }  // extern "C"
> 
>+#include "api/video/color_space.h"
> #include "api/video/i420_buffer.h"
> #include "common_video/include/video_frame_buffer.h"
>+#include "modules/video_coding/codecs/h264/h264_color_space.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/keep_ref_until_done.h"
>@@ -45,49 +47,6 @@ enum H264DecoderImplEvent {
>   kH264DecoderEventMax = 16,
> };
> 
>-#if defined(WEBRTC_INITIALIZE_FFMPEG)
>-
>-rtc::CriticalSection ffmpeg_init_lock;
>-bool ffmpeg_initialized = false;
>-
>-// Called by FFmpeg to do mutex operations if initialized using
>-// |InitializeFFmpeg|. Disabling thread safety analysis because void** does not
>-// play nicely with thread_annotations.h macros.
>-int LockManagerOperation(void** lock,
>-                         AVLockOp op) RTC_NO_THREAD_SAFETY_ANALYSIS {
>-  switch (op) {
>-    case AV_LOCK_CREATE:
>-      *lock = new rtc::CriticalSection();
>-      return 0;
>-    case AV_LOCK_OBTAIN:
>-      static_cast<rtc::CriticalSection*>(*lock)->Enter();
>-      return 0;
>-    case AV_LOCK_RELEASE:
>-      static_cast<rtc::CriticalSection*>(*lock)->Leave();
>-      return 0;
>-    case AV_LOCK_DESTROY:
>-      delete static_cast<rtc::CriticalSection*>(*lock);
>-      *lock = nullptr;
>-      return 0;
>-  }
>-  RTC_NOTREACHED() << "Unrecognized AVLockOp.";
>-  return -1;
>-}
>-
>-void InitializeFFmpeg() {
>-  rtc::CritScope cs(&ffmpeg_init_lock);
>-  if (!ffmpeg_initialized) {
>-    if (av_lockmgr_register(LockManagerOperation) < 0) {
>-      RTC_NOTREACHED() << "av_lockmgr_register failed.";
>-      return;
>-    }
>-    av_register_all();
>-    ffmpeg_initialized = true;
>-  }
>-}
>-
>-#endif  // defined(WEBRTC_INITIALIZE_FFMPEG)
>-
> }  // namespace
> 
> int H264DecoderImpl::AVGetBuffer2(
>@@ -165,9 +124,8 @@ int H264DecoderImpl::AVGetBuffer2(
>       total_size,
>       AVFreeBuffer2,
>       static_cast<void*>(new VideoFrame(frame_buffer,
>-                                        0 /* timestamp */,
>-                                        0 /* render_time_ms */,
>-                                        kVideoRotation_0)),
>+                                        kVideoRotation_0,
>+                                        0 /* timestamp_us */)),
>       0);
>   RTC_CHECK(av_frame->buf[0]);
>   return 0;
>@@ -200,18 +158,6 @@ int32_t H264DecoderImpl::InitDecode(const VideoCodec* codec_settings,
>     return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
>   }
> 
>-  // FFmpeg must have been initialized (with |av_lockmgr_register| and
>-  // |av_register_all|) before we proceed. |InitializeFFmpeg| does this, which
>-  // makes sense for WebRTC standalone. In other cases, such as Chromium, FFmpeg
>-  // is initialized externally and calling |InitializeFFmpeg| would be
>-  // thread-unsafe and result in FFmpeg being initialized twice, which could
>-  // break other FFmpeg usage. See the |rtc_initialize_ffmpeg| flag.
>-#if defined(WEBRTC_INITIALIZE_FFMPEG)
>-  // Make sure FFmpeg has been initialized. Subsequent |InitializeFFmpeg| calls
>-  // do nothing.
>-  InitializeFFmpeg();
>-#endif
>-
>   // Release necessary in case of re-initializing.
>   int32_t ret = Release();
>   if (ret != WEBRTC_VIDEO_CODEC_OK) {
>@@ -279,7 +225,6 @@ int32_t H264DecoderImpl::RegisterDecodeCompleteCallback(
> 
> int32_t H264DecoderImpl::Decode(const EncodedImage& input_image,
>                                 bool /*missing_frames*/,
>-                                const RTPFragmentationHeader* /*fragmentation*/,
>                                 const CodecSpecificInfo* codec_specific_info,
>                                 int64_t /*render_time_ms*/) {
>   if (!IsInitialized()) {
>@@ -345,17 +290,26 @@ int32_t H264DecoderImpl::Decode(const EncodedImage& input_image,
>   RTC_DCHECK_EQ(av_frame_->reordered_opaque, frame_timestamp_us);
> 
>   // Obtain the |video_frame| containing the decoded image.
>-  VideoFrame* video_frame = static_cast<VideoFrame*>(
>-      av_buffer_get_opaque(av_frame_->buf[0]));
>-  RTC_DCHECK(video_frame);
>+  VideoFrame* input_frame =
>+      static_cast<VideoFrame*>(av_buffer_get_opaque(av_frame_->buf[0]));
>+  RTC_DCHECK(input_frame);
>   rtc::scoped_refptr<webrtc::I420BufferInterface> i420_buffer =
>-      video_frame->video_frame_buffer()->GetI420();
>+      input_frame->video_frame_buffer()->GetI420();
>   RTC_CHECK_EQ(av_frame_->data[kYPlaneIndex], i420_buffer->DataY());
>   RTC_CHECK_EQ(av_frame_->data[kUPlaneIndex], i420_buffer->DataU());
>   RTC_CHECK_EQ(av_frame_->data[kVPlaneIndex], i420_buffer->DataV());
>-  video_frame->set_timestamp(input_image._timeStamp);
> 
>-  rtc::Optional<uint8_t> qp;
>+  const ColorSpace& color_space = ExtractH264ColorSpace(av_context_.get());
>+  VideoFrame decoded_frame =
>+      VideoFrame::Builder()
>+          .set_video_frame_buffer(input_frame->video_frame_buffer())
>+          .set_timestamp_us(input_frame->timestamp_us())
>+          .set_timestamp_rtp(input_image._timeStamp)
>+          .set_rotation(input_frame->rotation())
>+          .set_color_space(color_space)
>+          .build();
>+
>+  absl::optional<uint8_t> qp;
>   // TODO(sakal): Maybe it is possible to get QP directly from FFmpeg.
>   h264_bitstream_parser_.ParseBitstream(input_image._buffer,
>                                         input_image._length);
>@@ -376,19 +330,24 @@ int32_t H264DecoderImpl::Decode(const EncodedImage& input_image,
>             i420_buffer->DataU(), i420_buffer->StrideU(),
>             i420_buffer->DataV(), i420_buffer->StrideV(),
>             rtc::KeepRefUntilDone(i420_buffer)));
>-    VideoFrame cropped_frame(
>-        cropped_buf, video_frame->timestamp(), video_frame->render_time_ms(),
>-        video_frame->rotation());
>+    VideoFrame cropped_frame =
>+        VideoFrame::Builder()
>+            .set_video_frame_buffer(cropped_buf)
>+            .set_timestamp_ms(decoded_frame.render_time_ms())
>+            .set_timestamp_rtp(decoded_frame.timestamp())
>+            .set_rotation(decoded_frame.rotation())
>+            .set_color_space(color_space)
>+            .build();
>     // TODO(nisse): Timestamp and rotation are all zero here. Change decoder
>     // interface to pass a VideoFrameBuffer instead of a VideoFrame?
>-    decoded_image_callback_->Decoded(cropped_frame, rtc::nullopt, qp);
>+    decoded_image_callback_->Decoded(cropped_frame, absl::nullopt, qp);
>   } else {
>     // Return decoded frame.
>-    decoded_image_callback_->Decoded(*video_frame, rtc::nullopt, qp);
>+    decoded_image_callback_->Decoded(decoded_frame, absl::nullopt, qp);
>   }
>-  // Stop referencing it, possibly freeing |video_frame|.
>+  // Stop referencing it, possibly freeing |input_frame|.
>   av_frame_unref(av_frame_.get());
>-  video_frame = nullptr;
>+  input_frame = nullptr;
> 
>   return WEBRTC_VIDEO_CODEC_OK;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/h264_decoder_impl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/h264_decoder_impl.h
>index 74633e35843..a709177946c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/h264_decoder_impl.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/h264_decoder_impl.h
>@@ -49,7 +49,6 @@ class H264DecoderImpl : public H264Decoder {
>   // |missing_frames|, |fragmentation| and |render_time_ms| are ignored.
>   int32_t Decode(const EncodedImage& input_image,
>                  bool /*missing_frames*/,
>-                 const RTPFragmentationHeader* /*fragmentation*/,
>                  const CodecSpecificInfo* codec_specific_info = nullptr,
>                  int64_t render_time_ms = -1) override;
> 
>@@ -59,8 +58,9 @@ class H264DecoderImpl : public H264Decoder {
>   // Called by FFmpeg when it needs a frame buffer to store decoded frames in.
>   // The |VideoFrame| returned by FFmpeg at |Decode| originate from here. Their
>   // buffers are reference counted and freed by FFmpeg using |AVFreeBuffer2|.
>-  static int AVGetBuffer2(
>-      AVCodecContext* context, AVFrame* av_frame, int flags);
>+  static int AVGetBuffer2(AVCodecContext* context,
>+                          AVFrame* av_frame,
>+                          int flags);
>   // Called by FFmpeg when it is done with a video frame, see |AVGetBuffer2|.
>   static void AVFreeBuffer2(void* opaque, uint8_t* data);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/h264_encoder_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/h264_encoder_impl.cc
>index 15ad1f6700e..0b84e3ae8cf 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/h264_encoder_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/h264_encoder_impl.cc
>@@ -20,10 +20,14 @@
> #include "third_party/openh264/src/codec/api/svc/codec_ver.h"
> 
> #include "common_video/libyuv/include/webrtc_libyuv.h"
>+#include "modules/video_coding/utility/simulcast_rate_allocator.h"
>+#include "modules/video_coding/utility/simulcast_utility.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/timeutils.h"
> #include "system_wrappers/include/metrics.h"
>+#include "third_party/libyuv/include/libyuv/convert.h"
>+#include "third_party/libyuv/include/libyuv/scale.h"
> 
> namespace webrtc {
> 
>@@ -31,6 +35,10 @@ namespace {
> 
> const bool kOpenH264EncoderDetailedLogging = false;
> 
>+// QP scaling thresholds.
>+static const int kLowH264QpThreshold = 24;
>+static const int kHighH264QpThreshold = 37;
>+
> // Used by histograms. Values of entries should not be changed.
> enum H264EncoderImplEvent {
>   kH264EncoderEventInit = 0,
>@@ -41,17 +49,17 @@ enum H264EncoderImplEvent {
> int NumberOfThreads(int width, int height, int number_of_cores) {
>   // TODO(hbos): In Chromium, multiple threads do not work with sandbox on Mac,
>   // see crbug.com/583348. Until further investigated, only use one thread.
>-//  if (width * height >= 1920 * 1080 && number_of_cores > 8) {
>-//    return 8;  // 8 threads for 1080p on high perf machines.
>-//  } else if (width * height > 1280 * 960 && number_of_cores >= 6) {
>-//    return 3;  // 3 threads for 1080p.
>-//  } else if (width * height > 640 * 480 && number_of_cores >= 3) {
>-//    return 2;  // 2 threads for qHD/HD.
>-//  } else {
>-//    return 1;  // 1 thread for VGA or less.
>-//  }
>-// TODO(sprang): Also check sSliceArgument.uiSliceNum om GetEncoderPrams(),
>-//               before enabling multithreading here.
>+  //  if (width * height >= 1920 * 1080 && number_of_cores > 8) {
>+  //    return 8;  // 8 threads for 1080p on high perf machines.
>+  //  } else if (width * height > 1280 * 960 && number_of_cores >= 6) {
>+  //    return 3;  // 3 threads for 1080p.
>+  //  } else if (width * height > 640 * 480 && number_of_cores >= 3) {
>+  //    return 2;  // 2 threads for qHD/HD.
>+  //  } else {
>+  //    return 1;  // 1 thread for VGA or less.
>+  //  }
>+  // TODO(sprang): Also check sSliceArgument.uiSliceNum om GetEncoderPrams(),
>+  //               before enabling multithreading here.
>   return 1;
> }
> 
>@@ -135,10 +143,10 @@ static void RtpFragmentize(EncodedImage* encoded_image,
>       // Because the sum of all layer lengths, |required_size|, fits in a
>       // |size_t|, we know that any indices in-between will not overflow.
>       RTC_DCHECK_GE(layerInfo.pNalLengthInByte[nal], 4);
>-      RTC_DCHECK_EQ(layerInfo.pBsBuf[layer_len+0], start_code[0]);
>-      RTC_DCHECK_EQ(layerInfo.pBsBuf[layer_len+1], start_code[1]);
>-      RTC_DCHECK_EQ(layerInfo.pBsBuf[layer_len+2], start_code[2]);
>-      RTC_DCHECK_EQ(layerInfo.pBsBuf[layer_len+3], start_code[3]);
>+      RTC_DCHECK_EQ(layerInfo.pBsBuf[layer_len + 0], start_code[0]);
>+      RTC_DCHECK_EQ(layerInfo.pBsBuf[layer_len + 1], start_code[1]);
>+      RTC_DCHECK_EQ(layerInfo.pBsBuf[layer_len + 2], start_code[2]);
>+      RTC_DCHECK_EQ(layerInfo.pBsBuf[layer_len + 3], start_code[3]);
>       frag_header->fragmentationOffset[frag] =
>           encoded_image->_length + layer_len + sizeof(start_code);
>       frag_header->fragmentationLength[frag] =
>@@ -146,24 +154,14 @@ static void RtpFragmentize(EncodedImage* encoded_image,
>       layer_len += layerInfo.pNalLengthInByte[nal];
>     }
>     // Copy the entire layer's data (including start codes).
>-    memcpy(encoded_image->_buffer + encoded_image->_length,
>-           layerInfo.pBsBuf,
>+    memcpy(encoded_image->_buffer + encoded_image->_length, layerInfo.pBsBuf,
>            layer_len);
>     encoded_image->_length += layer_len;
>   }
> }
> 
> H264EncoderImpl::H264EncoderImpl(const cricket::VideoCodec& codec)
>-    : openh264_encoder_(nullptr),
>-      width_(0),
>-      height_(0),
>-      max_frame_rate_(0.0f),
>-      target_bps_(0),
>-      max_bps_(0),
>-      mode_(kRealtimeVideo),
>-      frame_dropping_on_(false),
>-      key_frame_interval_(0),
>-      packetization_mode_(H264PacketizationMode::SingleNalUnit),
>+    : packetization_mode_(H264PacketizationMode::SingleNalUnit),
>       max_payload_size_(0),
>       number_of_cores_(0),
>       encoded_image_callback_(nullptr),
>@@ -176,26 +174,30 @@ H264EncoderImpl::H264EncoderImpl(const cricket::VideoCodec& codec)
>       packetization_mode_string == "1") {
>     packetization_mode_ = H264PacketizationMode::NonInterleaved;
>   }
>+  downscaled_buffers_.reserve(kMaxSimulcastStreams - 1);
>+  encoded_images_.reserve(kMaxSimulcastStreams);
>+  encoded_image_buffers_.reserve(kMaxSimulcastStreams);
>+  encoders_.reserve(kMaxSimulcastStreams);
>+  configurations_.reserve(kMaxSimulcastStreams);
> }
> 
> H264EncoderImpl::~H264EncoderImpl() {
>   Release();
> }
> 
>-int32_t H264EncoderImpl::InitEncode(const VideoCodec* codec_settings,
>+int32_t H264EncoderImpl::InitEncode(const VideoCodec* inst,
>                                     int32_t number_of_cores,
>                                     size_t max_payload_size) {
>   ReportInit();
>-  if (!codec_settings ||
>-      codec_settings->codecType != kVideoCodecH264) {
>+  if (!inst || inst->codecType != kVideoCodecH264) {
>     ReportError();
>     return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
>   }
>-  if (codec_settings->maxFramerate == 0) {
>+  if (inst->maxFramerate == 0) {
>     ReportError();
>     return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
>   }
>-  if (codec_settings->width < 1 || codec_settings->height < 1) {
>+  if (inst->width < 1 || inst->height < 1) {
>     ReportError();
>     return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
>   }
>@@ -205,75 +207,128 @@ int32_t H264EncoderImpl::InitEncode(const VideoCodec* codec_settings,
>     ReportError();
>     return release_ret;
>   }
>-  RTC_DCHECK(!openh264_encoder_);
> 
>-  // Create encoder.
>-  if (WelsCreateSVCEncoder(&openh264_encoder_) != 0) {
>-    // Failed to create encoder.
>-    RTC_LOG(LS_ERROR) << "Failed to create OpenH264 encoder";
>-    RTC_DCHECK(!openh264_encoder_);
>-    ReportError();
>-    return WEBRTC_VIDEO_CODEC_ERROR;
>-  }
>-  RTC_DCHECK(openh264_encoder_);
>-  if (kOpenH264EncoderDetailedLogging) {
>-    int trace_level = WELS_LOG_DETAIL;
>-    openh264_encoder_->SetOption(ENCODER_OPTION_TRACE_LEVEL,
>-                                 &trace_level);
>+  int number_of_streams = SimulcastUtility::NumberOfSimulcastStreams(*inst);
>+  bool doing_simulcast = (number_of_streams > 1);
>+
>+  if (doing_simulcast && (!SimulcastUtility::ValidSimulcastResolutions(
>+                              *inst, number_of_streams) ||
>+                          !SimulcastUtility::ValidSimulcastTemporalLayers(
>+                              *inst, number_of_streams))) {
>+    return WEBRTC_VIDEO_CODEC_ERR_SIMULCAST_PARAMETERS_NOT_SUPPORTED;
>   }
>-  // else WELS_LOG_DEFAULT is used by default.
>+  downscaled_buffers_.resize(number_of_streams - 1);
>+  encoded_images_.resize(number_of_streams);
>+  encoded_image_buffers_.resize(number_of_streams);
>+  encoders_.resize(number_of_streams);
>+  pictures_.resize(number_of_streams);
>+  configurations_.resize(number_of_streams);
> 
>   number_of_cores_ = number_of_cores;
>-  // Set internal settings from codec_settings
>-  width_ = codec_settings->width;
>-  height_ = codec_settings->height;
>-  max_frame_rate_ = static_cast<float>(codec_settings->maxFramerate);
>-  mode_ = codec_settings->mode;
>-  frame_dropping_on_ = codec_settings->H264().frameDroppingOn;
>-  key_frame_interval_ = codec_settings->H264().keyFrameInterval;
>   max_payload_size_ = max_payload_size;
>+  codec_ = *inst;
>+
>+  // Code expects simulcastStream resolutions to be correct, make sure they are
>+  // filled even when there are no simulcast layers.
>+  if (codec_.numberOfSimulcastStreams == 0) {
>+    codec_.simulcastStream[0].width = codec_.width;
>+    codec_.simulcastStream[0].height = codec_.height;
>+  }
> 
>-  // Codec_settings uses kbits/second; encoder uses bits/second.
>-  max_bps_ = codec_settings->maxBitrate * 1000;
>-  if (codec_settings->targetBitrate == 0)
>-    target_bps_ = codec_settings->startBitrate * 1000;
>-  else
>-    target_bps_ = codec_settings->targetBitrate * 1000;
>+  for (int i = 0, idx = number_of_streams - 1; i < number_of_streams;
>+       ++i, --idx) {
>+    // Temporal layers still not supported.
>+    if (inst->simulcastStream[i].numberOfTemporalLayers > 1) {
>+      Release();
>+      return WEBRTC_VIDEO_CODEC_ERR_SIMULCAST_PARAMETERS_NOT_SUPPORTED;
>+    }
>+    ISVCEncoder* openh264_encoder;
>+    // Create encoder.
>+    if (WelsCreateSVCEncoder(&openh264_encoder) != 0) {
>+      // Failed to create encoder.
>+      RTC_LOG(LS_ERROR) << "Failed to create OpenH264 encoder";
>+      RTC_DCHECK(!openh264_encoder);
>+      Release();
>+      ReportError();
>+      return WEBRTC_VIDEO_CODEC_ERROR;
>+    }
>+    RTC_DCHECK(openh264_encoder);
>+    if (kOpenH264EncoderDetailedLogging) {
>+      int trace_level = WELS_LOG_DETAIL;
>+      openh264_encoder->SetOption(ENCODER_OPTION_TRACE_LEVEL, &trace_level);
>+    }
>+    // else WELS_LOG_DEFAULT is used by default.
>+
>+    // Store h264 encoder.
>+    encoders_[i] = openh264_encoder;
>+
>+    // Set internal settings from codec_settings
>+    configurations_[i].simulcast_idx = idx;
>+    configurations_[i].sending = false;
>+    configurations_[i].width = codec_.simulcastStream[idx].width;
>+    configurations_[i].height = codec_.simulcastStream[idx].height;
>+    configurations_[i].max_frame_rate = static_cast<float>(codec_.maxFramerate);
>+    configurations_[i].frame_dropping_on = codec_.H264()->frameDroppingOn;
>+    configurations_[i].key_frame_interval = codec_.H264()->keyFrameInterval;
>+
>+    // Create downscaled image buffers.
>+    if (i > 0) {
>+      downscaled_buffers_[i - 1] = I420Buffer::Create(
>+          configurations_[i].width, configurations_[i].height,
>+          configurations_[i].width, configurations_[i].width / 2,
>+          configurations_[i].width / 2);
>+    }
> 
>-  SEncParamExt encoder_params = CreateEncoderParams();
>+    // Codec_settings uses kbits/second; encoder uses bits/second.
>+    configurations_[i].max_bps = codec_.maxBitrate * 1000;
>+    configurations_[i].target_bps = codec_.startBitrate * 1000;
> 
>-  // Initialize.
>-  if (openh264_encoder_->InitializeExt(&encoder_params) != 0) {
>-    RTC_LOG(LS_ERROR) << "Failed to initialize OpenH264 encoder";
>-    Release();
>-    ReportError();
>-    return WEBRTC_VIDEO_CODEC_ERROR;
>+    // Create encoder parameters based on the layer configuration.
>+    SEncParamExt encoder_params = CreateEncoderParams(i);
>+
>+    // Initialize.
>+    if (openh264_encoder->InitializeExt(&encoder_params) != 0) {
>+      RTC_LOG(LS_ERROR) << "Failed to initialize OpenH264 encoder";
>+      Release();
>+      ReportError();
>+      return WEBRTC_VIDEO_CODEC_ERROR;
>+    }
>+    // TODO(pbos): Base init params on these values before submitting.
>+    int video_format = EVideoFormatType::videoFormatI420;
>+    openh264_encoder->SetOption(ENCODER_OPTION_DATAFORMAT, &video_format);
>+
>+    // Initialize encoded image. Default buffer size: size of unencoded data.
>+    encoded_images_[i]._size =
>+        CalcBufferSize(VideoType::kI420, codec_.simulcastStream[idx].width,
>+                       codec_.simulcastStream[idx].height);
>+    encoded_images_[i]._buffer = new uint8_t[encoded_images_[i]._size];
>+    encoded_image_buffers_[i].reset(encoded_images_[i]._buffer);
>+    encoded_images_[i]._completeFrame = true;
>+    encoded_images_[i]._encodedWidth = codec_.simulcastStream[idx].width;
>+    encoded_images_[i]._encodedHeight = codec_.simulcastStream[idx].height;
>+    encoded_images_[i]._length = 0;
>   }
>-  // TODO(pbos): Base init params on these values before submitting.
>-  int video_format = EVideoFormatType::videoFormatI420;
>-  openh264_encoder_->SetOption(ENCODER_OPTION_DATAFORMAT,
>-                               &video_format);
>-
>-  // Initialize encoded image. Default buffer size: size of unencoded data.
>-  encoded_image_._size = CalcBufferSize(VideoType::kI420, codec_settings->width,
>-                                        codec_settings->height);
>-  encoded_image_._buffer = new uint8_t[encoded_image_._size];
>-  encoded_image_buffer_.reset(encoded_image_._buffer);
>-  encoded_image_._completeFrame = true;
>-  encoded_image_._encodedWidth = 0;
>-  encoded_image_._encodedHeight = 0;
>-  encoded_image_._length = 0;
>-  return WEBRTC_VIDEO_CODEC_OK;
>+
>+  SimulcastRateAllocator init_allocator(codec_);
>+  BitrateAllocation allocation = init_allocator.GetAllocation(
>+      codec_.startBitrate * 1000, codec_.maxFramerate);
>+  return SetRateAllocation(allocation, codec_.maxFramerate);
> }
> 
> int32_t H264EncoderImpl::Release() {
>-  if (openh264_encoder_) {
>-    RTC_CHECK_EQ(0, openh264_encoder_->Uninitialize());
>-    WelsDestroySVCEncoder(openh264_encoder_);
>-    openh264_encoder_ = nullptr;
>+  while (!encoders_.empty()) {
>+    ISVCEncoder* openh264_encoder = encoders_.back();
>+    if (openh264_encoder) {
>+      RTC_CHECK_EQ(0, openh264_encoder->Uninitialize());
>+      WelsDestroySVCEncoder(openh264_encoder);
>+    }
>+    encoders_.pop_back();
>   }
>-  encoded_image_._buffer = nullptr;
>-  encoded_image_buffer_.reset();
>+  downscaled_buffers_.clear();
>+  configurations_.clear();
>+  encoded_images_.clear();
>+  encoded_image_buffers_.clear();
>+  pictures_.clear();
>   return WEBRTC_VIDEO_CODEC_OK;
> }
> 
>@@ -284,28 +339,59 @@ int32_t H264EncoderImpl::RegisterEncodeCompleteCallback(
> }
> 
> int32_t H264EncoderImpl::SetRateAllocation(
>-    const BitrateAllocation& bitrate_allocation,
>-    uint32_t framerate) {
>-  if (bitrate_allocation.get_sum_bps() <= 0 || framerate <= 0)
>+    const BitrateAllocation& bitrate,
>+    uint32_t new_framerate) {
>+  if (encoders_.empty())
>+    return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
>+
>+  if (new_framerate < 1)
>     return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
> 
>-  target_bps_ = bitrate_allocation.get_sum_bps();
>-  max_frame_rate_ = static_cast<float>(framerate);
>+  if (bitrate.get_sum_bps() == 0) {
>+    // Encoder paused, turn off all encoding.
>+    for (size_t i = 0; i < configurations_.size(); ++i)
>+      configurations_[i].SetStreamState(false);
>+    return WEBRTC_VIDEO_CODEC_OK;
>+  }
>+
>+  // At this point, bitrate allocation should already match codec settings.
>+  if (codec_.maxBitrate > 0)
>+    RTC_DCHECK_LE(bitrate.get_sum_kbps(), codec_.maxBitrate);
>+  RTC_DCHECK_GE(bitrate.get_sum_kbps(), codec_.minBitrate);
>+  if (codec_.numberOfSimulcastStreams > 0)
>+    RTC_DCHECK_GE(bitrate.get_sum_kbps(), codec_.simulcastStream[0].minBitrate);
>+
>+  codec_.maxFramerate = new_framerate;
>+
>+  size_t stream_idx = encoders_.size() - 1;
>+  for (size_t i = 0; i < encoders_.size(); ++i, --stream_idx) {
>+    // Update layer config.
>+    configurations_[i].target_bps = bitrate.GetSpatialLayerSum(stream_idx);
>+    configurations_[i].max_frame_rate = static_cast<float>(new_framerate);
>+
>+    if (configurations_[i].target_bps) {
>+      configurations_[i].SetStreamState(true);
>+
>+      // Update h264 encoder.
>+      SBitrateInfo target_bitrate;
>+      memset(&target_bitrate, 0, sizeof(SBitrateInfo));
>+      target_bitrate.iLayer = SPATIAL_LAYER_ALL,
>+      target_bitrate.iBitrate = configurations_[i].target_bps;
>+      encoders_[i]->SetOption(ENCODER_OPTION_BITRATE, &target_bitrate);
>+      encoders_[i]->SetOption(ENCODER_OPTION_FRAME_RATE,
>+                              &configurations_[i].max_frame_rate);
>+    } else {
>+      configurations_[i].SetStreamState(false);
>+    }
>+  }
> 
>-  SBitrateInfo target_bitrate;
>-  memset(&target_bitrate, 0, sizeof(SBitrateInfo));
>-  target_bitrate.iLayer = SPATIAL_LAYER_ALL,
>-  target_bitrate.iBitrate = target_bps_;
>-  openh264_encoder_->SetOption(ENCODER_OPTION_BITRATE,
>-                               &target_bitrate);
>-  openh264_encoder_->SetOption(ENCODER_OPTION_FRAME_RATE, &max_frame_rate_);
>   return WEBRTC_VIDEO_CODEC_OK;
> }
> 
> int32_t H264EncoderImpl::Encode(const VideoFrame& input_frame,
>                                 const CodecSpecificInfo* codec_specific_info,
>                                 const std::vector<FrameType>* frame_types) {
>-  if (!IsInitialized()) {
>+  if (encoders_.empty()) {
>     ReportError();
>     return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
>   }
>@@ -317,83 +403,134 @@ int32_t H264EncoderImpl::Encode(const VideoFrame& input_frame,
>     return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
>   }
> 
>-  bool force_key_frame = false;
>-  if (frame_types != nullptr) {
>-    // We only support a single stream.
>-    RTC_DCHECK_EQ(frame_types->size(), 1);
>-    // Skip frame?
>-    if ((*frame_types)[0] == kEmptyFrame) {
>-      return WEBRTC_VIDEO_CODEC_OK;
>-    }
>-    // Force key frame?
>-    force_key_frame = (*frame_types)[0] == kVideoFrameKey;
>-  }
>-  if (force_key_frame) {
>-    // API doc says ForceIntraFrame(false) does nothing, but calling this
>-    // function forces a key frame regardless of the |bIDR| argument's value.
>-    // (If every frame is a key frame we get lag/delays.)
>-    openh264_encoder_->ForceIntraFrame(true);
>-  }
>   rtc::scoped_refptr<const I420BufferInterface> frame_buffer =
>       input_frame.video_frame_buffer()->ToI420();
>-  // EncodeFrame input.
>-  SSourcePicture picture;
>-  memset(&picture, 0, sizeof(SSourcePicture));
>-  picture.iPicWidth = frame_buffer->width();
>-  picture.iPicHeight = frame_buffer->height();
>-  picture.iColorFormat = EVideoFormatType::videoFormatI420;
>-  picture.uiTimeStamp = input_frame.ntp_time_ms();
>-  picture.iStride[0] = frame_buffer->StrideY();
>-  picture.iStride[1] = frame_buffer->StrideU();
>-  picture.iStride[2] = frame_buffer->StrideV();
>-  picture.pData[0] = const_cast<uint8_t*>(frame_buffer->DataY());
>-  picture.pData[1] = const_cast<uint8_t*>(frame_buffer->DataU());
>-  picture.pData[2] = const_cast<uint8_t*>(frame_buffer->DataV());
>-
>-  // EncodeFrame output.
>-  SFrameBSInfo info;
>-  memset(&info, 0, sizeof(SFrameBSInfo));
>-
>-  // Encode!
>-  int enc_ret = openh264_encoder_->EncodeFrame(&picture, &info);
>-  if (enc_ret != 0) {
>-    RTC_LOG(LS_ERROR) << "OpenH264 frame encoding failed, EncodeFrame returned "
>-                      << enc_ret << ".";
>-    ReportError();
>-    return WEBRTC_VIDEO_CODEC_ERROR;
>+
>+  bool send_key_frame = false;
>+  for (size_t i = 0; i < configurations_.size(); ++i) {
>+    if (configurations_[i].key_frame_request && configurations_[i].sending) {
>+      send_key_frame = true;
>+      break;
>+    }
>   }
>+  if (!send_key_frame && frame_types) {
>+    for (size_t i = 0; i < frame_types->size() && i < configurations_.size();
>+         ++i) {
>+      if ((*frame_types)[i] == kVideoFrameKey && configurations_[i].sending) {
>+        send_key_frame = true;
>+        break;
>+      }
>+    }
>+  }
>+
>+  RTC_DCHECK_EQ(configurations_[0].width, frame_buffer->width());
>+  RTC_DCHECK_EQ(configurations_[0].height, frame_buffer->height());
>+
>+  // Encode image for each layer.
>+  for (size_t i = 0; i < encoders_.size(); ++i) {
>+    // EncodeFrame input.
>+    pictures_[i] = {0};
>+    pictures_[i].iPicWidth = configurations_[i].width;
>+    pictures_[i].iPicHeight = configurations_[i].height;
>+    pictures_[i].iColorFormat = EVideoFormatType::videoFormatI420;
>+    pictures_[i].uiTimeStamp = input_frame.ntp_time_ms();
>+    // Downscale images on second and ongoing layers.
>+    if (i == 0) {
>+      pictures_[i].iStride[0] = frame_buffer->StrideY();
>+      pictures_[i].iStride[1] = frame_buffer->StrideU();
>+      pictures_[i].iStride[2] = frame_buffer->StrideV();
>+      pictures_[i].pData[0] = const_cast<uint8_t*>(frame_buffer->DataY());
>+      pictures_[i].pData[1] = const_cast<uint8_t*>(frame_buffer->DataU());
>+      pictures_[i].pData[2] = const_cast<uint8_t*>(frame_buffer->DataV());
>+    } else {
>+      pictures_[i].iStride[0] = downscaled_buffers_[i - 1]->StrideY();
>+      pictures_[i].iStride[1] = downscaled_buffers_[i - 1]->StrideU();
>+      pictures_[i].iStride[2] = downscaled_buffers_[i - 1]->StrideV();
>+      pictures_[i].pData[0] =
>+          const_cast<uint8_t*>(downscaled_buffers_[i - 1]->DataY());
>+      pictures_[i].pData[1] =
>+          const_cast<uint8_t*>(downscaled_buffers_[i - 1]->DataU());
>+      pictures_[i].pData[2] =
>+          const_cast<uint8_t*>(downscaled_buffers_[i - 1]->DataV());
>+      // Scale the image down a number of times by downsampling factor.
>+      libyuv::I420Scale(pictures_[i - 1].pData[0], pictures_[i - 1].iStride[0],
>+                        pictures_[i - 1].pData[1], pictures_[i - 1].iStride[1],
>+                        pictures_[i - 1].pData[2], pictures_[i - 1].iStride[2],
>+                        configurations_[i - 1].width,
>+                        configurations_[i - 1].height, pictures_[i].pData[0],
>+                        pictures_[i].iStride[0], pictures_[i].pData[1],
>+                        pictures_[i].iStride[1], pictures_[i].pData[2],
>+                        pictures_[i].iStride[2], configurations_[i].width,
>+                        configurations_[i].height, libyuv::kFilterBilinear);
>+    }
>+
>+    if (!configurations_[i].sending) {
>+      continue;
>+    }
>+    if (frame_types != nullptr) {
>+      // Skip frame?
>+      if ((*frame_types)[i] == kEmptyFrame) {
>+        continue;
>+      }
>+    }
>+    if (send_key_frame) {
>+      // API doc says ForceIntraFrame(false) does nothing, but calling this
>+      // function forces a key frame regardless of the |bIDR| argument's value.
>+      // (If every frame is a key frame we get lag/delays.)
>+      encoders_[i]->ForceIntraFrame(true);
>+      configurations_[i].key_frame_request = false;
>+    }
>+    // EncodeFrame output.
>+    SFrameBSInfo info;
>+    memset(&info, 0, sizeof(SFrameBSInfo));
>+
>+    // Encode!
>+    int enc_ret = encoders_[i]->EncodeFrame(&pictures_[i], &info);
>+    if (enc_ret != 0) {
>+      RTC_LOG(LS_ERROR)
>+          << "OpenH264 frame encoding failed, EncodeFrame returned " << enc_ret
>+          << ".";
>+      ReportError();
>+      return WEBRTC_VIDEO_CODEC_ERROR;
>+    }
> 
>-  encoded_image_._encodedWidth = frame_buffer->width();
>-  encoded_image_._encodedHeight = frame_buffer->height();
>-  encoded_image_._timeStamp = input_frame.timestamp();
>-  encoded_image_.ntp_time_ms_ = input_frame.ntp_time_ms();
>-  encoded_image_.capture_time_ms_ = input_frame.render_time_ms();
>-  encoded_image_.rotation_ = input_frame.rotation();
>-  encoded_image_.content_type_ = (mode_ == kScreensharing)
>-                                     ? VideoContentType::SCREENSHARE
>-                                     : VideoContentType::UNSPECIFIED;
>-  encoded_image_.timing_.flags = TimingFrameFlags::kInvalid;
>-  encoded_image_._frameType = ConvertToVideoFrameType(info.eFrameType);
>-
>-  // Split encoded image up into fragments. This also updates |encoded_image_|.
>-  RTPFragmentationHeader frag_header;
>-  RtpFragmentize(&encoded_image_, &encoded_image_buffer_, *frame_buffer, &info,
>-                 &frag_header);
>-
>-  // Encoder can skip frames to save bandwidth in which case
>-  // |encoded_image_._length| == 0.
>-  if (encoded_image_._length > 0) {
>-    // Parse QP.
>-    h264_bitstream_parser_.ParseBitstream(encoded_image_._buffer,
>-                                          encoded_image_._length);
>-    h264_bitstream_parser_.GetLastSliceQp(&encoded_image_.qp_);
>-
>-    // Deliver encoded image.
>-    CodecSpecificInfo codec_specific;
>-    codec_specific.codecType = kVideoCodecH264;
>-    codec_specific.codecSpecific.H264.packetization_mode = packetization_mode_;
>-    encoded_image_callback_->OnEncodedImage(encoded_image_, &codec_specific,
>-                                            &frag_header);
>+    encoded_images_[i]._encodedWidth = configurations_[i].width;
>+    encoded_images_[i]._encodedHeight = configurations_[i].height;
>+    encoded_images_[i]._timeStamp = input_frame.timestamp();
>+    encoded_images_[i].ntp_time_ms_ = input_frame.ntp_time_ms();
>+    encoded_images_[i].capture_time_ms_ = input_frame.render_time_ms();
>+    encoded_images_[i].rotation_ = input_frame.rotation();
>+    encoded_images_[i].content_type_ =
>+            (codec_.mode == VideoCodecMode::kScreensharing)
>+            ? VideoContentType::SCREENSHARE
>+            : VideoContentType::UNSPECIFIED;
>+    encoded_images_[i].timing_.flags = VideoSendTiming::kInvalid;
>+    encoded_images_[i]._frameType = ConvertToVideoFrameType(info.eFrameType);
>+
>+    // Split encoded image up into fragments. This also updates
>+    // |encoded_image_|.
>+    RTPFragmentationHeader frag_header;
>+    RtpFragmentize(&encoded_images_[i], &encoded_image_buffers_[i],
>+                   *frame_buffer, &info, &frag_header);
>+
>+    // Encoder can skip frames to save bandwidth in which case
>+    // |encoded_images_[i]._length| == 0.
>+    if (encoded_images_[i]._length > 0) {
>+      // Parse QP.
>+      h264_bitstream_parser_.ParseBitstream(encoded_images_[i]._buffer,
>+                                            encoded_images_[i]._length);
>+      h264_bitstream_parser_.GetLastSliceQp(&encoded_images_[i].qp_);
>+
>+      // Deliver encoded image.
>+      CodecSpecificInfo codec_specific;
>+      codec_specific.codecType = kVideoCodecH264;
>+      codec_specific.codecSpecific.H264.packetization_mode =
>+          packetization_mode_;
>+      codec_specific.codecSpecific.H264.simulcast_idx =
>+          configurations_[i].simulcast_idx;
>+      encoded_image_callback_->OnEncodedImage(encoded_images_[i],
>+                                              &codec_specific, &frag_header);
>+    }
>   }
>   return WEBRTC_VIDEO_CODEC_OK;
> }
>@@ -402,40 +539,35 @@ const char* H264EncoderImpl::ImplementationName() const {
>   return "OpenH264";
> }
> 
>-bool H264EncoderImpl::IsInitialized() const {
>-  return openh264_encoder_ != nullptr;
>-}
>-
> // Initialization parameters.
> // There are two ways to initialize. There is SEncParamBase (cleared with
> // memset(&p, 0, sizeof(SEncParamBase)) used in Initialize, and SEncParamExt
> // which is a superset of SEncParamBase (cleared with GetDefaultParams) used
> // in InitializeExt.
>-SEncParamExt H264EncoderImpl::CreateEncoderParams() const {
>-  RTC_DCHECK(openh264_encoder_);
>+SEncParamExt H264EncoderImpl::CreateEncoderParams(size_t i) const {
>   SEncParamExt encoder_params;
>-  openh264_encoder_->GetDefaultParams(&encoder_params);
>-  if (mode_ == kRealtimeVideo) {
>+  encoders_[i]->GetDefaultParams(&encoder_params);
>+  if (codec_.mode == VideoCodecMode::kRealtimeVideo) {
>     encoder_params.iUsageType = CAMERA_VIDEO_REAL_TIME;
>-  } else if (mode_ == kScreensharing) {
>+  } else if (codec_.mode == VideoCodecMode::kScreensharing) {
>     encoder_params.iUsageType = SCREEN_CONTENT_REAL_TIME;
>   } else {
>     RTC_NOTREACHED();
>   }
>-  encoder_params.iPicWidth = width_;
>-  encoder_params.iPicHeight = height_;
>-  encoder_params.iTargetBitrate = target_bps_;
>-  encoder_params.iMaxBitrate = max_bps_;
>+  encoder_params.iPicWidth = configurations_[i].width;
>+  encoder_params.iPicHeight = configurations_[i].height;
>+  encoder_params.iTargetBitrate = configurations_[i].target_bps;
>+  encoder_params.iMaxBitrate = configurations_[i].max_bps;
>   // Rate Control mode
>   encoder_params.iRCMode = RC_BITRATE_MODE;
>-  encoder_params.fMaxFrameRate = max_frame_rate_;
>+  encoder_params.fMaxFrameRate = configurations_[i].max_frame_rate;
> 
>   // The following parameters are extension parameters (they're in SEncParamExt,
>   // not in SEncParamBase).
>-  encoder_params.bEnableFrameSkip = frame_dropping_on_;
>+  encoder_params.bEnableFrameSkip = configurations_[i].frame_dropping_on;
>   // |uiIntraPeriod|    - multiple of GOP size
>   // |keyFrameInterval| - number of frames
>-  encoder_params.uiIntraPeriod = key_frame_interval_;
>+  encoder_params.uiIntraPeriod = configurations_[i].key_frame_interval;
>   encoder_params.uiMaxNalSize = 0;
>   // Threading model: use auto.
>   //  0: auto (dynamic imp. internal encoder)
>@@ -461,6 +593,8 @@ SEncParamExt H264EncoderImpl::CreateEncoderParams() const {
>           SM_SIZELIMITED_SLICE;
>       encoder_params.sSpatialLayers[0].sSliceArgument.uiSliceSizeConstraint =
>           static_cast<unsigned int>(max_payload_size_);
>+      RTC_LOG(INFO) << "Encoder is configured with NALU constraint: "
>+                    << max_payload_size_ << " bytes";
>       break;
>     case H264PacketizationMode::NonInterleaved:
>       // When uiSliceMode = SM_FIXEDSLCNUM_SLICE, uiSliceNum = 0 means auto
>@@ -479,8 +613,7 @@ void H264EncoderImpl::ReportInit() {
>   if (has_reported_init_)
>     return;
>   RTC_HISTOGRAM_ENUMERATION("WebRTC.Video.H264EncoderImpl.Event",
>-                            kH264EncoderEventInit,
>-                            kH264EncoderEventMax);
>+                            kH264EncoderEventInit, kH264EncoderEventMax);
>   has_reported_init_ = true;
> }
> 
>@@ -488,22 +621,26 @@ void H264EncoderImpl::ReportError() {
>   if (has_reported_error_)
>     return;
>   RTC_HISTOGRAM_ENUMERATION("WebRTC.Video.H264EncoderImpl.Event",
>-                            kH264EncoderEventError,
>-                            kH264EncoderEventMax);
>+                            kH264EncoderEventError, kH264EncoderEventMax);
>   has_reported_error_ = true;
> }
> 
>-int32_t H264EncoderImpl::SetChannelParameters(
>-    uint32_t packet_loss, int64_t rtt) {
>+int32_t H264EncoderImpl::SetChannelParameters(uint32_t packet_loss,
>+                                              int64_t rtt) {
>   return WEBRTC_VIDEO_CODEC_OK;
> }
> 
>-int32_t H264EncoderImpl::SetPeriodicKeyFrames(bool enable) {
>-  return WEBRTC_VIDEO_CODEC_OK;
>+VideoEncoder::ScalingSettings H264EncoderImpl::GetScalingSettings() const {
>+  return VideoEncoder::ScalingSettings(kLowH264QpThreshold,
>+                                       kHighH264QpThreshold);
> }
> 
>-VideoEncoder::ScalingSettings H264EncoderImpl::GetScalingSettings() const {
>-  return VideoEncoder::ScalingSettings(true);
>+void H264EncoderImpl::LayerConfig::SetStreamState(bool send_stream) {
>+  if (send_stream && !sending) {
>+    // Need a key frame if we have not sent this stream before.
>+    key_frame_request = true;
>+  }
>+  sending = send_stream;
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/h264_encoder_impl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/h264_encoder_impl.h
>index c9b45d0a623..0d259660f69 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/h264_encoder_impl.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/h264_encoder_impl.h
>@@ -15,6 +15,7 @@
> #include <memory>
> #include <vector>
> 
>+#include "api/video/i420_buffer.h"
> #include "common_video/h264/h264_bitstream_parser.h"
> #include "modules/video_coding/codecs/h264/include/h264.h"
> #include "modules/video_coding/utility/quality_scaler.h"
>@@ -26,6 +27,22 @@ class ISVCEncoder;
> namespace webrtc {
> 
> class H264EncoderImpl : public H264Encoder {
>+ public:
>+  struct LayerConfig {
>+    int simulcast_idx = 0;
>+    int width = -1;
>+    int height = -1;
>+    bool sending = true;
>+    bool key_frame_request = false;
>+    float max_frame_rate = 0;
>+    uint32_t target_bps = 0;
>+    uint32_t max_bps = 0;
>+    bool frame_dropping_on = false;
>+    int key_frame_interval = 0;
>+
>+    void SetStreamState(bool send_stream);
>+  };
>+
>  public:
>   explicit H264EncoderImpl(const cricket::VideoCodec& codec);
>   ~H264EncoderImpl() override;
>@@ -44,7 +61,7 @@ class H264EncoderImpl : public H264Encoder {
> 
>   int32_t RegisterEncodeCompleteCallback(
>       EncodedImageCallback* callback) override;
>-  int32_t SetRateAllocation(const BitrateAllocation& bitrate_allocation,
>+  int32_t SetRateAllocation(const VideoBitrateAllocation& bitrate_allocation,
>                             uint32_t framerate) override;
> 
>   // The result of encoding - an EncodedImage and RTPFragmentationHeader - are
>@@ -59,7 +76,6 @@ class H264EncoderImpl : public H264Encoder {
> 
>   // Unsupported / Do nothing.
>   int32_t SetChannelParameters(uint32_t packet_loss, int64_t rtt) override;
>-  int32_t SetPeriodicKeyFrames(bool enable) override;
> 
>   // Exposed for testing.
>   H264PacketizationMode PacketizationModeForTesting() const {
>@@ -67,32 +83,24 @@ class H264EncoderImpl : public H264Encoder {
>   }
> 
>  private:
>-  bool IsInitialized() const;
>-  SEncParamExt CreateEncoderParams() const;
>+  SEncParamExt CreateEncoderParams(size_t i) const;
> 
>   webrtc::H264BitstreamParser h264_bitstream_parser_;
>   // Reports statistics with histograms.
>   void ReportInit();
>   void ReportError();
> 
>-  ISVCEncoder* openh264_encoder_;
>-  // Settings that are used by this encoder.
>-  int width_;
>-  int height_;
>-  float max_frame_rate_;
>-  uint32_t target_bps_;
>-  uint32_t max_bps_;
>-  VideoCodecMode mode_;
>-  // H.264 specifc parameters
>-  bool frame_dropping_on_;
>-  int key_frame_interval_;
>-  H264PacketizationMode packetization_mode_;
>+  std::vector<ISVCEncoder*> encoders_;
>+  std::vector<SSourcePicture> pictures_;
>+  std::vector<rtc::scoped_refptr<I420Buffer>> downscaled_buffers_;
>+  std::vector<LayerConfig> configurations_;
>+  std::vector<EncodedImage> encoded_images_;
>+  std::vector<std::unique_ptr<uint8_t[]>> encoded_image_buffers_;
> 
>+  VideoCodec codec_;
>+  H264PacketizationMode packetization_mode_;
>   size_t max_payload_size_;
>   int32_t number_of_cores_;
>-
>-  EncodedImage encoded_image_;
>-  std::unique_ptr<uint8_t[]> encoded_image_buffer_;
>   EncodedImageCallback* encoded_image_callback_;
> 
>   bool has_reported_init_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/h264_encoder_impl_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/h264_encoder_impl_unittest.cc
>index f143c9e8490..b8b10aef2ac 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/h264_encoder_impl_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/h264_encoder_impl_unittest.cc
>@@ -28,7 +28,7 @@ void SetDefaultSettings(VideoCodec* codec_settings) {
>   // If frame dropping is false, we get a warning that bitrate can't
>   // be controlled for RC_QUALITY_MODE; RC_BITRATE_MODE and RC_TIMESTAMP_MODE
>   codec_settings->H264()->frameDroppingOn = true;
>-  codec_settings->targetBitrate = 2000;
>+  codec_settings->startBitrate = 2000;
>   codec_settings->maxBitrate = 4000;
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/h264_simulcast_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/h264_simulcast_unittest.cc
>new file mode 100644
>index 00000000000..6c6fae8c1a0
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/h264_simulcast_unittest.cc
>@@ -0,0 +1,99 @@
>+/*
>+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <memory>
>+
>+#include "absl/memory/memory.h"
>+#include "api/test/create_simulcast_test_fixture.h"
>+#include "api/test/simulcast_test_fixture.h"
>+#include "modules/video_coding/codecs/h264/include/h264.h"
>+#include "test/function_video_decoder_factory.h"
>+#include "test/function_video_encoder_factory.h"
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+namespace {
>+std::unique_ptr<SimulcastTestFixture> CreateSpecificSimulcastTestFixture() {
>+  std::unique_ptr<VideoEncoderFactory> encoder_factory =
>+      absl::make_unique<FunctionVideoEncoderFactory>(
>+          []() { return H264Encoder::Create(cricket::VideoCodec("H264")); });
>+  std::unique_ptr<VideoDecoderFactory> decoder_factory =
>+      absl::make_unique<FunctionVideoDecoderFactory>(
>+          []() { return H264Decoder::Create(); });
>+  return CreateSimulcastTestFixture(std::move(encoder_factory),
>+                                    std::move(decoder_factory),
>+                                    SdpVideoFormat("H264"));
>+}
>+}  // namespace
>+
>+TEST(TestH264Simulcast, TestKeyFrameRequestsOnAllStreams) {
>+  auto fixture = CreateSpecificSimulcastTestFixture();
>+  fixture->TestKeyFrameRequestsOnAllStreams();
>+}
>+
>+TEST(TestH264Simulcast, TestPaddingAllStreams) {
>+  auto fixture = CreateSpecificSimulcastTestFixture();
>+  fixture->TestPaddingAllStreams();
>+}
>+
>+TEST(TestH264Simulcast, TestPaddingTwoStreams) {
>+  auto fixture = CreateSpecificSimulcastTestFixture();
>+  fixture->TestPaddingTwoStreams();
>+}
>+
>+TEST(TestH264Simulcast, TestPaddingTwoStreamsOneMaxedOut) {
>+  auto fixture = CreateSpecificSimulcastTestFixture();
>+  fixture->TestPaddingTwoStreamsOneMaxedOut();
>+}
>+
>+TEST(TestH264Simulcast, TestPaddingOneStream) {
>+  auto fixture = CreateSpecificSimulcastTestFixture();
>+  fixture->TestPaddingOneStream();
>+}
>+
>+TEST(TestH264Simulcast, TestPaddingOneStreamTwoMaxedOut) {
>+  auto fixture = CreateSpecificSimulcastTestFixture();
>+  fixture->TestPaddingOneStreamTwoMaxedOut();
>+}
>+
>+TEST(TestH264Simulcast, TestSendAllStreams) {
>+  auto fixture = CreateSpecificSimulcastTestFixture();
>+  fixture->TestSendAllStreams();
>+}
>+
>+TEST(TestH264Simulcast, TestDisablingStreams) {
>+  auto fixture = CreateSpecificSimulcastTestFixture();
>+  fixture->TestDisablingStreams();
>+}
>+
>+TEST(TestH264Simulcast, TestActiveStreams) {
>+  auto fixture = CreateSpecificSimulcastTestFixture();
>+  fixture->TestActiveStreams();
>+}
>+
>+TEST(TestH264Simulcast, TestSwitchingToOneStream) {
>+  auto fixture = CreateSpecificSimulcastTestFixture();
>+  fixture->TestSwitchingToOneStream();
>+}
>+
>+TEST(TestH264Simulcast, TestSwitchingToOneOddStream) {
>+  auto fixture = CreateSpecificSimulcastTestFixture();
>+  fixture->TestSwitchingToOneOddStream();
>+}
>+
>+TEST(TestH264Simulcast, TestStrideEncodeDecode) {
>+  auto fixture = CreateSpecificSimulcastTestFixture();
>+  fixture->TestStrideEncodeDecode();
>+}
>+
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/include/h264_globals.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/include/h264_globals.h
>index cae270cc399..e3215002d5c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/include/h264_globals.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/include/h264_globals.h
>@@ -14,6 +14,10 @@
> #ifndef MODULES_VIDEO_CODING_CODECS_H264_INCLUDE_H264_GLOBALS_H_
> #define MODULES_VIDEO_CODING_CODECS_H264_INCLUDE_H264_GLOBALS_H_
> 
>+#include <string>
>+
>+#include "rtc_base/checks.h"
>+
> namespace webrtc {
> 
> // The packetization types that we support: single, aggregated, and fragmented.
>@@ -40,17 +44,15 @@ enum class H264PacketizationMode {
> // This function is declared inline because it is not clear which
> // .cc file it should belong to.
> // TODO(hta): Refactor. https://bugs.webrtc.org/6842
>-inline std::ostream& operator<<(std::ostream& stream,
>-                                H264PacketizationMode mode) {
>-  switch (mode) {
>-    case H264PacketizationMode::NonInterleaved:
>-      stream << "NonInterleaved";
>-      break;
>-    case H264PacketizationMode::SingleNalUnit:
>-      stream << "SingleNalUnit";
>-      break;
>+// TODO(jonasolsson): Use absl::string_view instead when that's available.
>+inline std::string ToString(H264PacketizationMode mode) {
>+  if (mode == H264PacketizationMode::NonInterleaved) {
>+    return "NonInterleaved";
>+  } else if (mode == H264PacketizationMode::SingleNalUnit) {
>+    return "SingleNalUnit";
>   }
>-  return stream;
>+  RTC_NOTREACHED();
>+  return "";
> }
> 
> struct NaluInfo {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/test/h264_impl_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/test/h264_impl_unittest.cc
>index 8c32df4e9f5..14bb6bcc47e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/test/h264_impl_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/h264/test/h264_impl_unittest.cc
>@@ -8,13 +8,15 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>+#include "api/video/color_space.h"
> #include "common_video/libyuv/include/webrtc_libyuv.h"
> #include "modules/video_coding/codecs/h264/include/h264.h"
>-#include "modules/video_coding/codecs/test/video_codec_test.h"
>+#include "modules/video_coding/codecs/test/video_codec_unittest.h"
>+#include "test/video_codec_settings.h"
> 
> namespace webrtc {
> 
>-class TestH264Impl : public VideoCodecTest {
>+class TestH264Impl : public VideoCodecUnitTest {
>  protected:
>   std::unique_ptr<VideoEncoder> CreateEncoder() override {
>     return H264Encoder::Create(cricket::VideoCodec(cricket::kH264CodecName));
>@@ -24,13 +26,8 @@ class TestH264Impl : public VideoCodecTest {
>     return H264Decoder::Create();
>   }
> 
>-  VideoCodec codec_settings() override {
>-    VideoCodec codec_inst;
>-    codec_inst.codecType = webrtc::kVideoCodecH264;
>-    // If frame dropping is false, we get a warning that bitrate can't
>-    // be controlled for RC_QUALITY_MODE; RC_BITRATE_MODE and RC_TIMESTAMP_MODE
>-    codec_inst.H264()->frameDroppingOn = true;
>-    return codec_inst;
>+  void ModifyCodecSettings(VideoCodec* codec_settings) override {
>+    webrtc::test::CodecSettings(kVideoCodecH264, codec_settings);
>   }
> };
> 
>@@ -43,34 +40,41 @@ class TestH264Impl : public VideoCodecTest {
> #endif
> 
> TEST_F(TestH264Impl, MAYBE_EncodeDecode) {
>+  VideoFrame* input_frame = NextInputFrame();
>   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>-            encoder_->Encode(*input_frame_, nullptr, nullptr));
>+            encoder_->Encode(*input_frame, nullptr, nullptr));
>   EncodedImage encoded_frame;
>   CodecSpecificInfo codec_specific_info;
>   ASSERT_TRUE(WaitForEncodedFrame(&encoded_frame, &codec_specific_info));
>   // First frame should be a key frame.
>   encoded_frame._frameType = kVideoFrameKey;
>   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>-            decoder_->Decode(encoded_frame, false, nullptr));
>+            decoder_->Decode(encoded_frame, false, nullptr, 0));
>   std::unique_ptr<VideoFrame> decoded_frame;
>-  rtc::Optional<uint8_t> decoded_qp;
>+  absl::optional<uint8_t> decoded_qp;
>   ASSERT_TRUE(WaitForDecodedFrame(&decoded_frame, &decoded_qp));
>   ASSERT_TRUE(decoded_frame);
>-  EXPECT_GT(I420PSNR(input_frame_.get(), decoded_frame.get()), 36);
>+  EXPECT_GT(I420PSNR(input_frame, decoded_frame.get()), 36);
>+
>+  const ColorSpace color_space = decoded_frame->color_space().value();
>+  EXPECT_EQ(ColorSpace::PrimaryID::kInvalid, color_space.primaries());
>+  EXPECT_EQ(ColorSpace::TransferID::kInvalid, color_space.transfer());
>+  EXPECT_EQ(ColorSpace::MatrixID::kInvalid, color_space.matrix());
>+  EXPECT_EQ(ColorSpace::RangeID::kLimited, color_space.range());
> }
> 
> TEST_F(TestH264Impl, MAYBE_DecodedQpEqualsEncodedQp) {
>   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>-            encoder_->Encode(*input_frame_, nullptr, nullptr));
>+            encoder_->Encode(*NextInputFrame(), nullptr, nullptr));
>   EncodedImage encoded_frame;
>   CodecSpecificInfo codec_specific_info;
>   ASSERT_TRUE(WaitForEncodedFrame(&encoded_frame, &codec_specific_info));
>   // First frame should be a key frame.
>   encoded_frame._frameType = kVideoFrameKey;
>   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>-            decoder_->Decode(encoded_frame, false, nullptr));
>+            decoder_->Decode(encoded_frame, false, nullptr, 0));
>   std::unique_ptr<VideoFrame> decoded_frame;
>-  rtc::Optional<uint8_t> decoded_qp;
>+  absl::optional<uint8_t> decoded_qp;
>   ASSERT_TRUE(WaitForDecodedFrame(&decoded_frame, &decoded_qp));
>   ASSERT_TRUE(decoded_frame);
>   ASSERT_TRUE(decoded_qp);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/i420/i420.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/i420/i420.cc
>index 58ef32e2fcd..565a39e553c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/i420/i420.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/i420/i420.cc
>@@ -139,11 +139,7 @@ int I420Encoder::RegisterEncodeCompleteCallback(
>   return WEBRTC_VIDEO_CODEC_OK;
> }
> 
>-I420Decoder::I420Decoder()
>-    : _width(0),
>-      _height(0),
>-      _inited(false),
>-      _decodeCompleteCallback(NULL) {}
>+I420Decoder::I420Decoder() : _decodeCompleteCallback(NULL) {}
> 
> I420Decoder::~I420Decoder() {
>   Release();
>@@ -151,20 +147,11 @@ I420Decoder::~I420Decoder() {
> 
> int I420Decoder::InitDecode(const VideoCodec* codecSettings,
>                             int /*numberOfCores */) {
>-  if (codecSettings == NULL) {
>-    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
>-  } else if (codecSettings->width < 1 || codecSettings->height < 1) {
>-    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
>-  }
>-  _width = codecSettings->width;
>-  _height = codecSettings->height;
>-  _inited = true;
>   return WEBRTC_VIDEO_CODEC_OK;
> }
> 
> int I420Decoder::Decode(const EncodedImage& inputImage,
>                         bool /*missingFrames*/,
>-                        const RTPFragmentationHeader* /*fragmentation*/,
>                         const CodecSpecificInfo* /*codecSpecificInfo*/,
>                         int64_t /*renderTimeMs*/) {
>   if (inputImage._buffer == NULL) {
>@@ -179,34 +166,28 @@ int I420Decoder::Decode(const EncodedImage& inputImage,
>   if (inputImage._completeFrame == false) {
>     return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
>   }
>-  if (!_inited) {
>-    return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
>-  }
>   if (inputImage._length < kI420HeaderSize) {
>     return WEBRTC_VIDEO_CODEC_ERROR;
>   }
> 
>   const uint8_t* buffer = inputImage._buffer;
>   uint16_t width, height;
>-
>   buffer = ExtractHeader(buffer, &width, &height);
>-  _width = width;
>-  _height = height;
> 
>   // Verify that the available length is sufficient:
>   size_t req_length =
>-      CalcBufferSize(VideoType::kI420, _width, _height) + kI420HeaderSize;
>+      CalcBufferSize(VideoType::kI420, width, height) + kI420HeaderSize;
> 
>   if (req_length > inputImage._length) {
>     return WEBRTC_VIDEO_CODEC_ERROR;
>   }
>   // Set decoded image parameters.
>   rtc::scoped_refptr<webrtc::I420Buffer> frame_buffer =
>-      I420Buffer::Create(_width, _height);
>+      I420Buffer::Create(width, height);
> 
>   // Converting from raw buffer I420Buffer.
>-  int y_stride = 16 * ((_width + 15) / 16);
>-  int uv_stride = 16 * ((_width + 31) / 32);
>+  int y_stride = 16 * ((width + 15) / 16);
>+  int uv_stride = 16 * ((width + 31) / 32);
>   int y_size = y_stride * height;
>   int u_size = uv_stride * frame_buffer->ChromaHeight();
>   int ret = libyuv::I420Copy(
>@@ -214,7 +195,7 @@ int I420Decoder::Decode(const EncodedImage& inputImage,
>       uv_stride, frame_buffer.get()->MutableDataY(),
>       frame_buffer.get()->StrideY(), frame_buffer.get()->MutableDataU(),
>       frame_buffer.get()->StrideU(), frame_buffer.get()->MutableDataV(),
>-      frame_buffer.get()->StrideV(), _width, _height);
>+      frame_buffer.get()->StrideV(), width, height);
>   if (ret < 0) {
>     return WEBRTC_VIDEO_CODEC_MEMORY;
>   }
>@@ -243,7 +224,6 @@ int I420Decoder::RegisterDecodeCompleteCallback(
> }
> 
> int I420Decoder::Release() {
>-  _inited = false;
>   return WEBRTC_VIDEO_CODEC_OK;
> }
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/i420/include/i420.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/i420/include/i420.h
>index f3e8a3d9b86..580bc16c2f8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/i420/include/i420.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/i420/include/i420.h
>@@ -15,7 +15,6 @@
> 
> #include "modules/video_coding/include/video_codec_interface.h"
> #include "rtc_base/checks.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -108,7 +107,6 @@ class I420Decoder : public VideoDecoder {
>   //                                 <0 - Error
>   int Decode(const EncodedImage& inputImage,
>              bool missingFrames,
>-             const RTPFragmentationHeader* /*fragmentation*/,
>              const CodecSpecificInfo* /*codecSpecificInfo*/,
>              int64_t /*renderTimeMs*/) override;
> 
>@@ -131,9 +129,6 @@ class I420Decoder : public VideoDecoder {
>                                       uint16_t* width,
>                                       uint16_t* height);
> 
>-  int _width;
>-  int _height;
>-  bool _inited;
>   DecodedImageCallback* _decodeCompleteCallback;
> };  // class I420Decoder
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/interface/mock/mock_video_codec_interface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/interface/mock/mock_video_codec_interface.h
>deleted file mode 100644
>index 5f5880ac668..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/interface/mock/mock_video_codec_interface.h
>+++ /dev/null
>@@ -1,81 +0,0 @@
>-/*
>- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MODULES_VIDEO_CODING_CODECS_INTERFACE_MOCK_MOCK_VIDEO_CODEC_INTERFACE_H_
>-#define MODULES_VIDEO_CODING_CODECS_INTERFACE_MOCK_MOCK_VIDEO_CODEC_INTERFACE_H_
>-
>-#pragma message("WARNING: video_coding/codecs/interface is DEPRECATED; "
>-    "use video_coding/include")
>-#include <string>
>-#include <vector>
>-
>-#include "modules/video_coding/include/video_codec_interface.h"
>-#include "test/gmock.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>-
>-namespace webrtc {
>-
>-class MockEncodedImageCallback : public EncodedImageCallback {
>- public:
>-  MOCK_METHOD3(Encoded,
>-               int32_t(const EncodedImage& encodedImage,
>-                       const CodecSpecificInfo* codecSpecificInfo,
>-                       const RTPFragmentationHeader* fragmentation));
>-};
>-
>-class MockVideoEncoder : public VideoEncoder {
>- public:
>-  MOCK_CONST_METHOD2(Version, int32_t(int8_t* version, int32_t length));
>-  MOCK_METHOD3(InitEncode,
>-               int32_t(const VideoCodec* codecSettings,
>-                       int32_t numberOfCores,
>-                       size_t maxPayloadSize));
>-  MOCK_METHOD3(Encode,
>-               int32_t(const VideoFrame& inputImage,
>-                       const CodecSpecificInfo* codecSpecificInfo,
>-                       const std::vector<FrameType>* frame_types));
>-  MOCK_METHOD1(RegisterEncodeCompleteCallback,
>-               int32_t(EncodedImageCallback* callback));
>-  MOCK_METHOD0(Release, int32_t());
>-  MOCK_METHOD0(Reset, int32_t());
>-  MOCK_METHOD2(SetChannelParameters, int32_t(uint32_t packetLoss, int64_t rtt));
>-  MOCK_METHOD2(SetRates, int32_t(uint32_t newBitRate, uint32_t frameRate));
>-  MOCK_METHOD1(SetPeriodicKeyFrames, int32_t(bool enable));
>-};
>-
>-class MockDecodedImageCallback : public DecodedImageCallback {
>- public:
>-  MOCK_METHOD1(Decoded, int32_t(const VideoFrame& decodedImage));
>-  MOCK_METHOD2(Decoded,
>-               int32_t(const VideoFrame& decodedImage, int64_t decode_time_ms));
>-  MOCK_METHOD1(ReceivedDecodedReferenceFrame,
>-               int32_t(const uint64_t pictureId));
>-  MOCK_METHOD1(ReceivedDecodedFrame, int32_t(const uint64_t pictureId));
>-};
>-
>-class MockVideoDecoder : public VideoDecoder {
>- public:
>-  MOCK_METHOD2(InitDecode,
>-               int32_t(const VideoCodec* codecSettings, int32_t numberOfCores));
>-  MOCK_METHOD5(Decode,
>-               int32_t(const EncodedImage& inputImage,
>-                       bool missingFrames,
>-                       const RTPFragmentationHeader* fragmentation,
>-                       const CodecSpecificInfo* codecSpecificInfo,
>-                       int64_t renderTimeMs));
>-  MOCK_METHOD1(RegisterDecodeCompleteCallback,
>-               int32_t(DecodedImageCallback* callback));
>-  MOCK_METHOD0(Release, int32_t());
>-  MOCK_METHOD0(Copy, VideoDecoder*());
>-};
>-
>-}  // namespace webrtc
>-
>-#endif  // MODULES_VIDEO_CODING_CODECS_INTERFACE_MOCK_MOCK_VIDEO_CODEC_INTERFACE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/interface/video_codec_interface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/interface/video_codec_interface.h
>deleted file mode 100644
>index 682fde69626..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/interface/video_codec_interface.h
>+++ /dev/null
>@@ -1,19 +0,0 @@
>-/*
>- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MODULES_VIDEO_CODING_CODECS_INTERFACE_VIDEO_CODEC_INTERFACE_H_
>-#define MODULES_VIDEO_CODING_CODECS_INTERFACE_VIDEO_CODEC_INTERFACE_H_
>-
>-#pragma message("WARNING: video_coding/codecs/interface is DEPRECATED; "
>-    "use video_coding/include")
>-
>-#include "modules/video_coding/include/video_codec_interface.h"
>-
>-#endif  // MODULES_VIDEO_CODING_CODECS_INTERFACE_VIDEO_CODEC_INTERFACE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/interface/video_error_codes.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/interface/video_error_codes.h
>deleted file mode 100644
>index 582d3091a55..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/interface/video_error_codes.h
>+++ /dev/null
>@@ -1,35 +0,0 @@
>-/*
>- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MODULES_VIDEO_CODING_CODECS_INTERFACE_VIDEO_ERROR_CODES_H_
>-#define MODULES_VIDEO_CODING_CODECS_INTERFACE_VIDEO_ERROR_CODES_H_
>-
>-#pragma message("WARNING: video_coding/codecs/interface is DEPRECATED; "
>-    "use video_coding/include")
>-
>-// NOTE: in sync with video_coding_module_defines.h
>-
>-// Define return values
>-
>-#define WEBRTC_VIDEO_CODEC_REQUEST_SLI 2
>-#define WEBRTC_VIDEO_CODEC_NO_OUTPUT 1
>-#define WEBRTC_VIDEO_CODEC_OK 0
>-#define WEBRTC_VIDEO_CODEC_ERROR -1
>-#define WEBRTC_VIDEO_CODEC_LEVEL_EXCEEDED -2
>-#define WEBRTC_VIDEO_CODEC_MEMORY -3
>-#define WEBRTC_VIDEO_CODEC_ERR_PARAMETER -4
>-#define WEBRTC_VIDEO_CODEC_ERR_SIZE -5
>-#define WEBRTC_VIDEO_CODEC_TIMEOUT -6
>-#define WEBRTC_VIDEO_CODEC_UNINITIALIZED -7
>-#define WEBRTC_VIDEO_CODEC_ERR_REQUEST_SLI -12
>-#define WEBRTC_VIDEO_CODEC_FALLBACK_SOFTWARE -13
>-#define WEBRTC_VIDEO_CODEC_TARGET_BITRATE_OVERSHOOT -14
>-
>-#endif  // MODULES_VIDEO_CODING_CODECS_INTERFACE_VIDEO_ERROR_CODES_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/stereo/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/multiplex/OWNERS
>similarity index 100%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/stereo/OWNERS
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/multiplex/OWNERS
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/multiplex/augmented_video_frame_buffer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/multiplex/augmented_video_frame_buffer.cc
>new file mode 100644
>index 00000000000..b61a4a6e532
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/multiplex/augmented_video_frame_buffer.cc
>@@ -0,0 +1,52 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/video_coding/codecs/multiplex/include/augmented_video_frame_buffer.h"
>+#include "api/video/video_frame_buffer.h"
>+
>+namespace webrtc {
>+
>+AugmentedVideoFrameBuffer::AugmentedVideoFrameBuffer(
>+    const rtc::scoped_refptr<VideoFrameBuffer>& video_frame_buffer,
>+    std::unique_ptr<uint8_t[]> augmenting_data,
>+    uint16_t augmenting_data_size)
>+    : augmenting_data_size_(augmenting_data_size),
>+      augmenting_data_(std::move(augmenting_data)),
>+      video_frame_buffer_(video_frame_buffer) {}
>+
>+rtc::scoped_refptr<VideoFrameBuffer>
>+AugmentedVideoFrameBuffer::GetVideoFrameBuffer() const {
>+  return video_frame_buffer_;
>+}
>+
>+uint8_t* AugmentedVideoFrameBuffer::GetAugmentingData() const {
>+  return augmenting_data_.get();
>+}
>+
>+uint16_t AugmentedVideoFrameBuffer::GetAugmentingDataSize() const {
>+  return augmenting_data_size_;
>+}
>+
>+VideoFrameBuffer::Type AugmentedVideoFrameBuffer::type() const {
>+  return video_frame_buffer_->type();
>+}
>+
>+int AugmentedVideoFrameBuffer::width() const {
>+  return video_frame_buffer_->width();
>+}
>+
>+int AugmentedVideoFrameBuffer::height() const {
>+  return video_frame_buffer_->height();
>+}
>+
>+rtc::scoped_refptr<I420BufferInterface> AugmentedVideoFrameBuffer::ToI420() {
>+  return video_frame_buffer_->ToI420();
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/multiplex/include/augmented_video_frame_buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/multiplex/include/augmented_video_frame_buffer.h
>new file mode 100644
>index 00000000000..96496ed7530
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/multiplex/include/augmented_video_frame_buffer.h
>@@ -0,0 +1,54 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_VIDEO_CODING_CODECS_MULTIPLEX_INCLUDE_AUGMENTED_VIDEO_FRAME_BUFFER_H_
>+#define MODULES_VIDEO_CODING_CODECS_MULTIPLEX_INCLUDE_AUGMENTED_VIDEO_FRAME_BUFFER_H_
>+
>+#include <memory>
>+
>+#include "api/video/video_frame_buffer.h"
>+
>+namespace webrtc {
>+class AugmentedVideoFrameBuffer : public VideoFrameBuffer {
>+ public:
>+  AugmentedVideoFrameBuffer(
>+      const rtc::scoped_refptr<VideoFrameBuffer>& video_frame_buffer,
>+      std::unique_ptr<uint8_t[]> augmenting_data,
>+      uint16_t augmenting_data_size);
>+
>+  // Retrieves the underlying VideoFrameBuffer without the augmented data
>+  rtc::scoped_refptr<VideoFrameBuffer> GetVideoFrameBuffer() const;
>+
>+  // Gets a pointer to the augmenting data and moves ownership to the caller
>+  uint8_t* GetAugmentingData() const;
>+
>+  // Get the size of the augmenting data
>+  uint16_t GetAugmentingDataSize() const;
>+
>+  // Returns the type of the underlying VideoFrameBuffer
>+  Type type() const final;
>+
>+  // Returns the width of the underlying VideoFrameBuffer
>+  int width() const final;
>+
>+  // Returns the height of the underlying VideoFrameBuffer
>+  int height() const final;
>+
>+  // Get the I140 Buffer from the underlying frame buffer
>+  rtc::scoped_refptr<I420BufferInterface> ToI420() final;
>+
>+ private:
>+  uint16_t augmenting_data_size_;
>+  std::unique_ptr<uint8_t[]> augmenting_data_;
>+  rtc::scoped_refptr<webrtc::VideoFrameBuffer> video_frame_buffer_;
>+};
>+}  // namespace webrtc
>+
>+#endif  // MODULES_VIDEO_CODING_CODECS_MULTIPLEX_INCLUDE_AUGMENTED_VIDEO_FRAME_BUFFER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/stereo/include/stereo_decoder_adapter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/multiplex/include/multiplex_decoder_adapter.h
>similarity index 58%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/stereo/include/stereo_decoder_adapter.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/multiplex/include/multiplex_decoder_adapter.h
>index c6237eece4f..b8a90b4e0f8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/stereo/include/stereo_decoder_adapter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/multiplex/include/multiplex_decoder_adapter.h
>@@ -8,8 +8,8 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#ifndef MODULES_VIDEO_CODING_CODECS_STEREO_INCLUDE_STEREO_DECODER_ADAPTER_H_
>-#define MODULES_VIDEO_CODING_CODECS_STEREO_INCLUDE_STEREO_DECODER_ADAPTER_H_
>+#ifndef MODULES_VIDEO_CODING_CODECS_MULTIPLEX_INCLUDE_MULTIPLEX_DECODER_ADAPTER_H_
>+#define MODULES_VIDEO_CODING_CODECS_MULTIPLEX_INCLUDE_MULTIPLEX_DECODER_ADAPTER_H_
> 
> #include <map>
> #include <memory>
>@@ -18,23 +18,23 @@
> #include "api/video_codecs/sdp_video_format.h"
> #include "api/video_codecs/video_decoder.h"
> #include "api/video_codecs/video_decoder_factory.h"
>-#include "modules/video_coding/codecs/stereo/include/stereo_encoder_adapter.h"
>+#include "modules/video_coding/codecs/multiplex/include/multiplex_encoder_adapter.h"
> 
> namespace webrtc {
> 
>-class StereoDecoderAdapter : public VideoDecoder {
>+class MultiplexDecoderAdapter : public VideoDecoder {
>  public:
>   // |factory| is not owned and expected to outlive this class' lifetime.
>-  explicit StereoDecoderAdapter(VideoDecoderFactory* factory,
>-                                const SdpVideoFormat& associated_format);
>-  virtual ~StereoDecoderAdapter();
>+  MultiplexDecoderAdapter(VideoDecoderFactory* factory,
>+                          const SdpVideoFormat& associated_format,
>+                          bool supports_augmenting_data = false);
>+  virtual ~MultiplexDecoderAdapter();
> 
>   // Implements VideoDecoder
>   int32_t InitDecode(const VideoCodec* codec_settings,
>                      int32_t number_of_cores) override;
>   int32_t Decode(const EncodedImage& input_image,
>                  bool missing_frames,
>-                 const RTPFragmentationHeader* fragmentation,
>                  const CodecSpecificInfo* codec_specific_info,
>                  int64_t render_time_ms) override;
>   int32_t RegisterDecodeCompleteCallback(
>@@ -43,8 +43,8 @@ class StereoDecoderAdapter : public VideoDecoder {
> 
>   void Decoded(AlphaCodecStream stream_idx,
>                VideoFrame* decoded_image,
>-               rtc::Optional<int32_t> decode_time_ms,
>-               rtc::Optional<uint8_t> qp);
>+               absl::optional<int32_t> decode_time_ms,
>+               absl::optional<uint8_t> qp);
> 
>  private:
>   // Wrapper class that redirects Decoded() calls.
>@@ -53,12 +53,17 @@ class StereoDecoderAdapter : public VideoDecoder {
>   // Holds the decoded image output of a frame.
>   struct DecodedImageData;
> 
>+  // Holds the augmenting data of an image
>+  struct AugmentingData;
>+
>   void MergeAlphaImages(VideoFrame* decoded_image,
>-                        const rtc::Optional<int32_t>& decode_time_ms,
>-                        const rtc::Optional<uint8_t>& qp,
>-                        VideoFrame* stereo_decoded_image,
>-                        const rtc::Optional<int32_t>& stereo_decode_time_ms,
>-                        const rtc::Optional<uint8_t>& stereo_qp);
>+                        const absl::optional<int32_t>& decode_time_ms,
>+                        const absl::optional<uint8_t>& qp,
>+                        VideoFrame* multiplex_decoded_image,
>+                        const absl::optional<int32_t>& multiplex_decode_time_ms,
>+                        const absl::optional<uint8_t>& multiplex_qp,
>+                        std::unique_ptr<uint8_t[]> augmenting_data,
>+                        uint16_t augmenting_data_length);
> 
>   VideoDecoderFactory* const factory_;
>   const SdpVideoFormat associated_format_;
>@@ -68,8 +73,10 @@ class StereoDecoderAdapter : public VideoDecoder {
> 
>   // Holds YUV or AXX decode output of a frame that is identified by timestamp.
>   std::map<uint32_t /* timestamp */, DecodedImageData> decoded_data_;
>+  std::map<uint32_t /* timestamp */, AugmentingData> decoded_augmenting_data_;
>+  const bool supports_augmenting_data_;
> };
> 
> }  // namespace webrtc
> 
>-#endif  // MODULES_VIDEO_CODING_CODECS_STEREO_INCLUDE_STEREO_DECODER_ADAPTER_H_
>+#endif  // MODULES_VIDEO_CODING_CODECS_MULTIPLEX_INCLUDE_MULTIPLEX_DECODER_ADAPTER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/multiplex/include/multiplex_encoded_image_packer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/multiplex/include/multiplex_encoded_image_packer.h
>new file mode 100644
>index 00000000000..220221a8492
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/multiplex/include/multiplex_encoded_image_packer.h
>@@ -0,0 +1,119 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_VIDEO_CODING_CODECS_MULTIPLEX_INCLUDE_MULTIPLEX_ENCODED_IMAGE_PACKER_H_
>+#define MODULES_VIDEO_CODING_CODECS_MULTIPLEX_INCLUDE_MULTIPLEX_ENCODED_IMAGE_PACKER_H_
>+
>+#include <memory>
>+#include <vector>
>+
>+#include "common_types.h"  // NOLINT(build/include)
>+#include "common_video/include/video_frame.h"
>+
>+namespace webrtc {
>+
>+// Struct describing the whole bundle of multiple frames of an image.
>+// This struct is expected to be the set in the beginning of a picture's
>+// bitstream data.
>+struct MultiplexImageHeader {
>+  // The number of frame components making up the complete picture data.
>+  // For example, |frame_count| = 2 for the case of YUV frame with Alpha frame.
>+  uint8_t component_count;
>+
>+  // The increasing image ID given by the encoder. For different components
>+  // of a single picture, they have the same |picture_index|.
>+  uint16_t image_index;
>+
>+  // The location of the first MultiplexImageComponentHeader in the bitstream,
>+  // in terms of byte from the beginning of the bitstream.
>+  uint32_t first_component_header_offset;
>+
>+  // The location of the augmenting data in the bitstream, in terms of bytes
>+  // from the beginning of the bitstream
>+  uint32_t augmenting_data_offset;
>+
>+  // The size of the augmenting data in the bitstream it terms of byte
>+  uint16_t augmenting_data_size;
>+};
>+const int kMultiplexImageHeaderSize =
>+    sizeof(uint8_t) + 2 * sizeof(uint16_t) + 2 * sizeof(uint32_t);
>+
>+// Struct describing the individual image component's content.
>+struct MultiplexImageComponentHeader {
>+  // The location of the next MultiplexImageComponentHeader in the bitstream,
>+  // in terms of the byte from the beginning of the bitstream;
>+  uint32_t next_component_header_offset;
>+
>+  // Identifies which component this frame represent, i.e. YUV frame vs Alpha
>+  // frame.
>+  uint8_t component_index;
>+
>+  // The location of the real encoded image data of the frame in the bitstream,
>+  // in terms of byte from the beginning of the bitstream.
>+  uint32_t bitstream_offset;
>+
>+  // Indicates the number of bytes of the encoded image data.
>+  uint32_t bitstream_length;
>+
>+  // Indicated the underlying VideoCodecType of the frame, i.e. VP9 or VP8 etc.
>+  VideoCodecType codec_type;
>+
>+  // Indicated the underlying frame is a key frame or delta frame.
>+  FrameType frame_type;
>+};
>+const int kMultiplexImageComponentHeaderSize =
>+    sizeof(uint32_t) + sizeof(uint8_t) + sizeof(uint32_t) + sizeof(uint32_t) +
>+    sizeof(uint8_t) + sizeof(uint8_t);
>+
>+// Struct holding the encoded image for one component.
>+struct MultiplexImageComponent {
>+  // Indicated the underlying VideoCodecType of the frame, i.e. VP9 or VP8 etc.
>+  VideoCodecType codec_type;
>+
>+  // Identifies which component this frame represent, i.e. YUV frame vs Alpha
>+  // frame.
>+  uint8_t component_index;
>+
>+  // Stores the actual frame data of the encoded image.
>+  EncodedImage encoded_image;
>+};
>+
>+// Struct holding the whole frame bundle of components of an image.
>+struct MultiplexImage {
>+  uint16_t image_index;
>+  uint8_t component_count;
>+  uint16_t augmenting_data_size;
>+  std::unique_ptr<uint8_t[]> augmenting_data;
>+  std::vector<MultiplexImageComponent> image_components;
>+
>+  MultiplexImage(uint16_t picture_index,
>+                 uint8_t component_count,
>+                 std::unique_ptr<uint8_t[]> augmenting_data,
>+                 uint16_t augmenting_data_size);
>+};
>+
>+// A utility class providing conversion between two representations of a
>+// multiplex image frame:
>+// 1. Packed version is just one encoded image, we pack all necessary metadata
>+//    in the bitstream as headers.
>+// 2. Unpacked version is essentially a list of encoded images, one for one
>+//    component.
>+class MultiplexEncodedImagePacker {
>+ public:
>+  // Note: It is caller responsibility to release the buffer of the result.
>+  static EncodedImage PackAndRelease(const MultiplexImage& image);
>+
>+  // Note: The image components just share the memory with |combined_image|.
>+  static MultiplexImage Unpack(const EncodedImage& combined_image);
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_VIDEO_CODING_CODECS_MULTIPLEX_INCLUDE_MULTIPLEX_ENCODED_IMAGE_PACKER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/stereo/include/stereo_encoder_adapter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/multiplex/include/multiplex_encoder_adapter.h
>similarity index 67%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/stereo/include/stereo_encoder_adapter.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/multiplex/include/multiplex_encoder_adapter.h
>index 95283f19a8a..a24932302f3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/stereo/include/stereo_encoder_adapter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/multiplex/include/multiplex_encoder_adapter.h
>@@ -8,8 +8,8 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#ifndef MODULES_VIDEO_CODING_CODECS_STEREO_INCLUDE_STEREO_ENCODER_ADAPTER_H_
>-#define MODULES_VIDEO_CODING_CODECS_STEREO_INCLUDE_STEREO_ENCODER_ADAPTER_H_
>+#ifndef MODULES_VIDEO_CODING_CODECS_MULTIPLEX_INCLUDE_MULTIPLEX_ENCODER_ADAPTER_H_
>+#define MODULES_VIDEO_CODING_CODECS_MULTIPLEX_INCLUDE_MULTIPLEX_ENCODER_ADAPTER_H_
> 
> #include <map>
> #include <memory>
>@@ -18,7 +18,9 @@
> #include "api/video_codecs/sdp_video_format.h"
> #include "api/video_codecs/video_encoder.h"
> #include "api/video_codecs/video_encoder_factory.h"
>+#include "modules/video_coding/codecs/multiplex/include/multiplex_encoded_image_packer.h"
> #include "modules/video_coding/include/video_codec_interface.h"
>+#include "rtc_base/criticalsection.h"
> 
> namespace webrtc {
> 
>@@ -28,12 +30,13 @@ enum AlphaCodecStream {
>   kAlphaCodecStreams = 2,
> };
> 
>-class StereoEncoderAdapter : public VideoEncoder {
>+class MultiplexEncoderAdapter : public VideoEncoder {
>  public:
>   // |factory| is not owned and expected to outlive this class' lifetime.
>-  explicit StereoEncoderAdapter(VideoEncoderFactory* factory,
>-                                const SdpVideoFormat& associated_format);
>-  virtual ~StereoEncoderAdapter();
>+  MultiplexEncoderAdapter(VideoEncoderFactory* factory,
>+                          const SdpVideoFormat& associated_format,
>+                          bool supports_augmenting_data = false);
>+  virtual ~MultiplexEncoderAdapter();
> 
>   // Implements VideoEncoder
>   int InitEncode(const VideoCodec* inst,
>@@ -44,7 +47,7 @@ class StereoEncoderAdapter : public VideoEncoder {
>              const std::vector<FrameType>* frame_types) override;
>   int RegisterEncodeCompleteCallback(EncodedImageCallback* callback) override;
>   int SetChannelParameters(uint32_t packet_loss, int64_t rtt) override;
>-  int SetRateAllocation(const BitrateAllocation& bitrate,
>+  int SetRateAllocation(const VideoBitrateAllocation& bitrate,
>                         uint32_t new_framerate) override;
>   int Release() override;
>   const char* ImplementationName() const override;
>@@ -65,14 +68,21 @@ class StereoEncoderAdapter : public VideoEncoder {
>   std::vector<std::unique_ptr<AdapterEncodedImageCallback>> adapter_callbacks_;
>   EncodedImageCallback* encoded_complete_callback_;
> 
>-  // Holds the encoded image info.
>-  struct ImageStereoInfo;
>-  std::map<uint32_t /* timestamp */, ImageStereoInfo> image_stereo_info_;
>+  std::map<uint32_t /* timestamp */, MultiplexImage> stashed_images_
>+      RTC_GUARDED_BY(crit_);
> 
>   uint16_t picture_index_ = 0;
>-  std::vector<uint8_t> stereo_dummy_planes_;
>+  std::vector<uint8_t> multiplex_dummy_planes_;
>+
>+  int key_frame_interval_;
>+  EncodedImage combined_image_;
>+
>+  rtc::CriticalSection crit_;
>+
>+  const bool supports_augmented_data_;
>+  int augmenting_data_size_ = 0;
> };
> 
> }  // namespace webrtc
> 
>-#endif  // MODULES_VIDEO_CODING_CODECS_STEREO_INCLUDE_STEREO_ENCODER_ADAPTER_H_
>+#endif  // MODULES_VIDEO_CODING_CODECS_MULTIPLEX_INCLUDE_MULTIPLEX_ENCODER_ADAPTER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/stereo/stereo_decoder_adapter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/multiplex/multiplex_decoder_adapter.cc
>similarity index 50%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/stereo/stereo_decoder_adapter.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/multiplex/multiplex_decoder_adapter.cc
>index 96d55524d45..e0d06182b99 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/stereo/stereo_decoder_adapter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/multiplex/multiplex_decoder_adapter.cc
>@@ -8,13 +8,14 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "modules/video_coding/codecs/stereo/include/stereo_decoder_adapter.h"
>+#include "modules/video_coding/codecs/multiplex/include/multiplex_decoder_adapter.h"
> 
> #include "api/video/i420_buffer.h"
> #include "api/video/video_frame_buffer.h"
> #include "common_video/include/video_frame.h"
> #include "common_video/include/video_frame_buffer.h"
> #include "common_video/libyuv/include/webrtc_libyuv.h"
>+#include "modules/video_coding/codecs/multiplex/include/augmented_video_frame_buffer.h"
> #include "rtc_base/keep_ref_until_done.h"
> #include "rtc_base/logging.h"
> 
>@@ -25,16 +26,16 @@ void KeepBufferRefs(rtc::scoped_refptr<webrtc::VideoFrameBuffer>,
> 
> namespace webrtc {
> 
>-class StereoDecoderAdapter::AdapterDecodedImageCallback
>+class MultiplexDecoderAdapter::AdapterDecodedImageCallback
>     : public webrtc::DecodedImageCallback {
>  public:
>-  AdapterDecodedImageCallback(webrtc::StereoDecoderAdapter* adapter,
>+  AdapterDecodedImageCallback(webrtc::MultiplexDecoderAdapter* adapter,
>                               AlphaCodecStream stream_idx)
>       : adapter_(adapter), stream_idx_(stream_idx) {}
> 
>   void Decoded(VideoFrame& decoded_image,
>-               rtc::Optional<int32_t> decode_time_ms,
>-               rtc::Optional<uint8_t> qp) override {
>+               absl::optional<int32_t> decode_time_ms,
>+               absl::optional<uint8_t> qp) override {
>     if (!adapter_)
>       return;
>     adapter_->Decoded(stream_idx_, &decoded_image, decode_time_ms, qp);
>@@ -49,11 +50,11 @@ class StereoDecoderAdapter::AdapterDecodedImageCallback
>   }
> 
>  private:
>-  StereoDecoderAdapter* adapter_;
>+  MultiplexDecoderAdapter* adapter_;
>   const AlphaCodecStream stream_idx_;
> };
> 
>-struct StereoDecoderAdapter::DecodedImageData {
>+struct MultiplexDecoderAdapter::DecodedImageData {
>   explicit DecodedImageData(AlphaCodecStream stream_idx)
>       : stream_idx_(stream_idx),
>         decoded_image_(I420Buffer::Create(1 /* width */, 1 /* height */),
>@@ -64,33 +65,46 @@ struct StereoDecoderAdapter::DecodedImageData {
>   }
>   DecodedImageData(AlphaCodecStream stream_idx,
>                    const VideoFrame& decoded_image,
>-                   const rtc::Optional<int32_t>& decode_time_ms,
>-                   const rtc::Optional<uint8_t>& qp)
>+                   const absl::optional<int32_t>& decode_time_ms,
>+                   const absl::optional<uint8_t>& qp)
>       : stream_idx_(stream_idx),
>         decoded_image_(decoded_image),
>         decode_time_ms_(decode_time_ms),
>         qp_(qp) {}
>   const AlphaCodecStream stream_idx_;
>   VideoFrame decoded_image_;
>-  const rtc::Optional<int32_t> decode_time_ms_;
>-  const rtc::Optional<uint8_t> qp_;
>+  const absl::optional<int32_t> decode_time_ms_;
>+  const absl::optional<uint8_t> qp_;
> 
>  private:
>   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(DecodedImageData);
> };
> 
>-StereoDecoderAdapter::StereoDecoderAdapter(
>+struct MultiplexDecoderAdapter::AugmentingData {
>+  AugmentingData(std::unique_ptr<uint8_t[]> augmenting_data, uint16_t data_size)
>+      : data_(std::move(augmenting_data)), size_(data_size) {}
>+  std::unique_ptr<uint8_t[]> data_;
>+  const uint16_t size_;
>+
>+ private:
>+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AugmentingData);
>+};
>+
>+MultiplexDecoderAdapter::MultiplexDecoderAdapter(
>     VideoDecoderFactory* factory,
>-    const SdpVideoFormat& associated_format)
>-    : factory_(factory), associated_format_(associated_format) {}
>+    const SdpVideoFormat& associated_format,
>+    bool supports_augmenting_data)
>+    : factory_(factory),
>+      associated_format_(associated_format),
>+      supports_augmenting_data_(supports_augmenting_data) {}
> 
>-StereoDecoderAdapter::~StereoDecoderAdapter() {
>+MultiplexDecoderAdapter::~MultiplexDecoderAdapter() {
>   Release();
> }
> 
>-int32_t StereoDecoderAdapter::InitDecode(const VideoCodec* codec_settings,
>-                                         int32_t number_of_cores) {
>-  RTC_DCHECK_EQ(kVideoCodecStereo, codec_settings->codecType);
>+int32_t MultiplexDecoderAdapter::InitDecode(const VideoCodec* codec_settings,
>+                                            int32_t number_of_cores) {
>+  RTC_DCHECK_EQ(kVideoCodecMultiplex, codec_settings->codecType);
>   VideoCodec settings = *codec_settings;
>   settings.codecType = PayloadStringToCodecType(associated_format_.name);
>   for (size_t i = 0; i < kAlphaCodecStreams; ++i) {
>@@ -100,7 +114,7 @@ int32_t StereoDecoderAdapter::InitDecode(const VideoCodec* codec_settings,
>     if (rv)
>       return rv;
>     adapter_callbacks_.emplace_back(
>-        new StereoDecoderAdapter::AdapterDecodedImageCallback(
>+        new MultiplexDecoderAdapter::AdapterDecodedImageCallback(
>             this, static_cast<AlphaCodecStream>(i)));
>     decoder->RegisterDecodeCompleteCallback(adapter_callbacks_.back().get());
>     decoders_.emplace_back(std::move(decoder));
>@@ -108,37 +122,47 @@ int32_t StereoDecoderAdapter::InitDecode(const VideoCodec* codec_settings,
>   return WEBRTC_VIDEO_CODEC_OK;
> }
> 
>-int32_t StereoDecoderAdapter::Decode(
>+int32_t MultiplexDecoderAdapter::Decode(
>     const EncodedImage& input_image,
>     bool missing_frames,
>-    const RTPFragmentationHeader* /*fragmentation*/,
>     const CodecSpecificInfo* codec_specific_info,
>     int64_t render_time_ms) {
>-  const CodecSpecificInfoStereo& stereo_info =
>-      codec_specific_info->codecSpecific.stereo;
>-  RTC_DCHECK_LT(static_cast<size_t>(stereo_info.indices.frame_index),
>-                decoders_.size());
>-  if (stereo_info.indices.frame_count == 1) {
>-    RTC_DCHECK_EQ(static_cast<int>(stereo_info.indices.frame_index), 0);
>+  MultiplexImage image = MultiplexEncodedImagePacker::Unpack(input_image);
>+
>+  if (supports_augmenting_data_) {
>+    RTC_DCHECK(decoded_augmenting_data_.find(input_image._timeStamp) ==
>+               decoded_augmenting_data_.end());
>+    decoded_augmenting_data_.emplace(
>+        std::piecewise_construct, std::forward_as_tuple(input_image._timeStamp),
>+        std::forward_as_tuple(std::move(image.augmenting_data),
>+                              image.augmenting_data_size));
>+  }
>+
>+  if (image.component_count == 1) {
>     RTC_DCHECK(decoded_data_.find(input_image._timeStamp) ==
>                decoded_data_.end());
>     decoded_data_.emplace(std::piecewise_construct,
>                           std::forward_as_tuple(input_image._timeStamp),
>                           std::forward_as_tuple(kAXXStream));
>   }
>-
>-  int32_t rv = decoders_[stereo_info.indices.frame_index]->Decode(
>-      input_image, missing_frames, nullptr, nullptr, render_time_ms);
>+  int32_t rv = 0;
>+  for (size_t i = 0; i < image.image_components.size(); i++) {
>+    rv = decoders_[image.image_components[i].component_index]->Decode(
>+        image.image_components[i].encoded_image, missing_frames, nullptr,
>+        render_time_ms);
>+    if (rv != WEBRTC_VIDEO_CODEC_OK)
>+      return rv;
>+  }
>   return rv;
> }
> 
>-int32_t StereoDecoderAdapter::RegisterDecodeCompleteCallback(
>+int32_t MultiplexDecoderAdapter::RegisterDecodeCompleteCallback(
>     DecodedImageCallback* callback) {
>   decoded_complete_callback_ = callback;
>   return WEBRTC_VIDEO_CODEC_OK;
> }
> 
>-int32_t StereoDecoderAdapter::Release() {
>+int32_t MultiplexDecoderAdapter::Release() {
>   for (auto& decoder : decoders_) {
>     const int32_t rv = decoder->Release();
>     if (rv)
>@@ -149,63 +173,83 @@ int32_t StereoDecoderAdapter::Release() {
>   return WEBRTC_VIDEO_CODEC_OK;
> }
> 
>-void StereoDecoderAdapter::Decoded(AlphaCodecStream stream_idx,
>-                                   VideoFrame* decoded_image,
>-                                   rtc::Optional<int32_t> decode_time_ms,
>-                                   rtc::Optional<uint8_t> qp) {
>+void MultiplexDecoderAdapter::Decoded(AlphaCodecStream stream_idx,
>+                                      VideoFrame* decoded_image,
>+                                      absl::optional<int32_t> decode_time_ms,
>+                                      absl::optional<uint8_t> qp) {
>   const auto& other_decoded_data_it =
>       decoded_data_.find(decoded_image->timestamp());
>+  const auto& augmenting_data_it =
>+      decoded_augmenting_data_.find(decoded_image->timestamp());
>   if (other_decoded_data_it != decoded_data_.end()) {
>+    uint16_t augmenting_data_size =
>+        augmenting_data_it == decoded_augmenting_data_.end()
>+            ? 0
>+            : augmenting_data_it->second.size_;
>+    std::unique_ptr<uint8_t[]> augmenting_data =
>+        augmenting_data_size == 0 ? NULL
>+                                  : std::move(augmenting_data_it->second.data_);
>     auto& other_image_data = other_decoded_data_it->second;
>     if (stream_idx == kYUVStream) {
>       RTC_DCHECK_EQ(kAXXStream, other_image_data.stream_idx_);
>       MergeAlphaImages(decoded_image, decode_time_ms, qp,
>                        &other_image_data.decoded_image_,
>-                       other_image_data.decode_time_ms_, other_image_data.qp_);
>+                       other_image_data.decode_time_ms_, other_image_data.qp_,
>+                       std::move(augmenting_data), augmenting_data_size);
>     } else {
>       RTC_DCHECK_EQ(kYUVStream, other_image_data.stream_idx_);
>       RTC_DCHECK_EQ(kAXXStream, stream_idx);
>       MergeAlphaImages(&other_image_data.decoded_image_,
>                        other_image_data.decode_time_ms_, other_image_data.qp_,
>-                       decoded_image, decode_time_ms, qp);
>+                       decoded_image, decode_time_ms, qp,
>+                       std::move(augmenting_data), augmenting_data_size);
>     }
>     decoded_data_.erase(decoded_data_.begin(), other_decoded_data_it);
>+    if (supports_augmenting_data_) {
>+      decoded_augmenting_data_.erase(decoded_augmenting_data_.begin(),
>+                                     augmenting_data_it);
>+    }
>     return;
>   }
>   RTC_DCHECK(decoded_data_.find(decoded_image->timestamp()) ==
>              decoded_data_.end());
>-  // decoded_data_[decoded_image->timestamp()] =
>-  //     DecodedImageData(stream_idx, *decoded_image, decode_time_ms, qp);
>   decoded_data_.emplace(
>       std::piecewise_construct,
>       std::forward_as_tuple(decoded_image->timestamp()),
>       std::forward_as_tuple(stream_idx, *decoded_image, decode_time_ms, qp));
> }
> 
>-void StereoDecoderAdapter::MergeAlphaImages(
>+void MultiplexDecoderAdapter::MergeAlphaImages(
>     VideoFrame* decoded_image,
>-    const rtc::Optional<int32_t>& decode_time_ms,
>-    const rtc::Optional<uint8_t>& qp,
>+    const absl::optional<int32_t>& decode_time_ms,
>+    const absl::optional<uint8_t>& qp,
>     VideoFrame* alpha_decoded_image,
>-    const rtc::Optional<int32_t>& alpha_decode_time_ms,
>-    const rtc::Optional<uint8_t>& alpha_qp) {
>+    const absl::optional<int32_t>& alpha_decode_time_ms,
>+    const absl::optional<uint8_t>& alpha_qp,
>+    std::unique_ptr<uint8_t[]> augmenting_data,
>+    uint16_t augmenting_data_length) {
>+  rtc::scoped_refptr<VideoFrameBuffer> merged_buffer;
>   if (!alpha_decoded_image->timestamp()) {
>-    decoded_complete_callback_->Decoded(*decoded_image, decode_time_ms, qp);
>-    return;
>+    merged_buffer = decoded_image->video_frame_buffer();
>+  } else {
>+    rtc::scoped_refptr<webrtc::I420BufferInterface> yuv_buffer =
>+        decoded_image->video_frame_buffer()->ToI420();
>+    rtc::scoped_refptr<webrtc::I420BufferInterface> alpha_buffer =
>+        alpha_decoded_image->video_frame_buffer()->ToI420();
>+    RTC_DCHECK_EQ(yuv_buffer->width(), alpha_buffer->width());
>+    RTC_DCHECK_EQ(yuv_buffer->height(), alpha_buffer->height());
>+    merged_buffer = WrapI420ABuffer(
>+        yuv_buffer->width(), yuv_buffer->height(), yuv_buffer->DataY(),
>+        yuv_buffer->StrideY(), yuv_buffer->DataU(), yuv_buffer->StrideU(),
>+        yuv_buffer->DataV(), yuv_buffer->StrideV(), alpha_buffer->DataY(),
>+        alpha_buffer->StrideY(),
>+        rtc::Bind(&KeepBufferRefs, yuv_buffer, alpha_buffer));
>+  }
>+  if (supports_augmenting_data_) {
>+    merged_buffer = rtc::scoped_refptr<webrtc::AugmentedVideoFrameBuffer>(
>+        new rtc::RefCountedObject<AugmentedVideoFrameBuffer>(
>+            merged_buffer, std::move(augmenting_data), augmenting_data_length));
>   }
>-
>-  rtc::scoped_refptr<webrtc::I420BufferInterface> yuv_buffer =
>-      decoded_image->video_frame_buffer()->ToI420();
>-  rtc::scoped_refptr<webrtc::I420BufferInterface> alpha_buffer =
>-      alpha_decoded_image->video_frame_buffer()->ToI420();
>-  RTC_DCHECK_EQ(yuv_buffer->width(), alpha_buffer->width());
>-  RTC_DCHECK_EQ(yuv_buffer->height(), alpha_buffer->height());
>-  rtc::scoped_refptr<I420ABufferInterface> merged_buffer = WrapI420ABuffer(
>-      yuv_buffer->width(), yuv_buffer->height(), yuv_buffer->DataY(),
>-      yuv_buffer->StrideY(), yuv_buffer->DataU(), yuv_buffer->StrideU(),
>-      yuv_buffer->DataV(), yuv_buffer->StrideV(), alpha_buffer->DataY(),
>-      alpha_buffer->StrideY(),
>-      rtc::Bind(&KeepBufferRefs, yuv_buffer, alpha_buffer));
> 
>   VideoFrame merged_image(merged_buffer, decoded_image->timestamp(),
>                           0 /* render_time_ms */, decoded_image->rotation());
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/multiplex/multiplex_encoded_image_packer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/multiplex/multiplex_encoded_image_packer.cc
>new file mode 100644
>index 00000000000..63cd6dab476
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/multiplex/multiplex_encoded_image_packer.cc
>@@ -0,0 +1,279 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/video_coding/codecs/multiplex/include/multiplex_encoded_image_packer.h"
>+
>+#include <cstring>
>+
>+#include "modules/rtp_rtcp/source/byte_io.h"
>+
>+namespace webrtc {
>+int PackHeader(uint8_t* buffer, MultiplexImageHeader header) {
>+  int offset = 0;
>+  ByteWriter<uint8_t>::WriteBigEndian(buffer + offset, header.component_count);
>+  offset += sizeof(uint8_t);
>+
>+  ByteWriter<uint16_t>::WriteBigEndian(buffer + offset, header.image_index);
>+  offset += sizeof(uint16_t);
>+
>+  ByteWriter<uint16_t>::WriteBigEndian(buffer + offset,
>+                                       header.augmenting_data_size);
>+  offset += sizeof(uint16_t);
>+
>+  ByteWriter<uint32_t>::WriteBigEndian(buffer + offset,
>+                                       header.augmenting_data_offset);
>+  offset += sizeof(uint32_t);
>+
>+  ByteWriter<uint32_t>::WriteBigEndian(buffer + offset,
>+                                       header.first_component_header_offset);
>+  offset += sizeof(uint32_t);
>+
>+  RTC_DCHECK_EQ(offset, kMultiplexImageHeaderSize);
>+  return offset;
>+}
>+
>+MultiplexImageHeader UnpackHeader(uint8_t* buffer) {
>+  MultiplexImageHeader header;
>+  int offset = 0;
>+  header.component_count = ByteReader<uint8_t>::ReadBigEndian(buffer + offset);
>+  offset += sizeof(uint8_t);
>+
>+  header.image_index = ByteReader<uint16_t>::ReadBigEndian(buffer + offset);
>+  offset += sizeof(uint16_t);
>+
>+  header.augmenting_data_size =
>+      ByteReader<uint16_t>::ReadBigEndian(buffer + offset);
>+  offset += sizeof(uint16_t);
>+
>+  header.augmenting_data_offset =
>+      ByteReader<uint32_t>::ReadBigEndian(buffer + offset);
>+  offset += sizeof(uint32_t);
>+
>+  header.first_component_header_offset =
>+      ByteReader<uint32_t>::ReadBigEndian(buffer + offset);
>+  offset += sizeof(uint32_t);
>+
>+  RTC_DCHECK_EQ(offset, kMultiplexImageHeaderSize);
>+  return header;
>+}
>+
>+int PackFrameHeader(uint8_t* buffer,
>+                    MultiplexImageComponentHeader frame_header) {
>+  int offset = 0;
>+  ByteWriter<uint32_t>::WriteBigEndian(
>+      buffer + offset, frame_header.next_component_header_offset);
>+  offset += sizeof(uint32_t);
>+
>+  ByteWriter<uint8_t>::WriteBigEndian(buffer + offset,
>+                                      frame_header.component_index);
>+  offset += sizeof(uint8_t);
>+
>+  ByteWriter<uint32_t>::WriteBigEndian(buffer + offset,
>+                                       frame_header.bitstream_offset);
>+  offset += sizeof(uint32_t);
>+
>+  ByteWriter<uint32_t>::WriteBigEndian(buffer + offset,
>+                                       frame_header.bitstream_length);
>+  offset += sizeof(uint32_t);
>+
>+  ByteWriter<uint8_t>::WriteBigEndian(buffer + offset, frame_header.codec_type);
>+  offset += sizeof(uint8_t);
>+
>+  ByteWriter<uint8_t>::WriteBigEndian(buffer + offset, frame_header.frame_type);
>+  offset += sizeof(uint8_t);
>+
>+  RTC_DCHECK_EQ(offset, kMultiplexImageComponentHeaderSize);
>+  return offset;
>+}
>+
>+MultiplexImageComponentHeader UnpackFrameHeader(uint8_t* buffer) {
>+  MultiplexImageComponentHeader frame_header;
>+  int offset = 0;
>+
>+  frame_header.next_component_header_offset =
>+      ByteReader<uint32_t>::ReadBigEndian(buffer + offset);
>+  offset += sizeof(uint32_t);
>+
>+  frame_header.component_index =
>+      ByteReader<uint8_t>::ReadBigEndian(buffer + offset);
>+  offset += sizeof(uint8_t);
>+
>+  frame_header.bitstream_offset =
>+      ByteReader<uint32_t>::ReadBigEndian(buffer + offset);
>+  offset += sizeof(uint32_t);
>+
>+  frame_header.bitstream_length =
>+      ByteReader<uint32_t>::ReadBigEndian(buffer + offset);
>+  offset += sizeof(uint32_t);
>+
>+  frame_header.codec_type = static_cast<VideoCodecType>(
>+      ByteReader<uint8_t>::ReadBigEndian(buffer + offset));
>+  offset += sizeof(uint8_t);
>+
>+  frame_header.frame_type = static_cast<FrameType>(
>+      ByteReader<uint8_t>::ReadBigEndian(buffer + offset));
>+  offset += sizeof(uint8_t);
>+
>+  RTC_DCHECK_EQ(offset, kMultiplexImageComponentHeaderSize);
>+  return frame_header;
>+}
>+
>+void PackBitstream(uint8_t* buffer, MultiplexImageComponent image) {
>+  memcpy(buffer, image.encoded_image._buffer, image.encoded_image._length);
>+}
>+
>+MultiplexImage::MultiplexImage(uint16_t picture_index,
>+                               uint8_t frame_count,
>+                               std::unique_ptr<uint8_t[]> augmenting_data,
>+                               uint16_t augmenting_data_size)
>+    : image_index(picture_index),
>+      component_count(frame_count),
>+      augmenting_data_size(augmenting_data_size),
>+      augmenting_data(std::move(augmenting_data)) {}
>+
>+EncodedImage MultiplexEncodedImagePacker::PackAndRelease(
>+    const MultiplexImage& multiplex_image) {
>+  MultiplexImageHeader header;
>+  std::vector<MultiplexImageComponentHeader> frame_headers;
>+
>+  header.component_count = multiplex_image.component_count;
>+  header.image_index = multiplex_image.image_index;
>+  int header_offset = kMultiplexImageHeaderSize;
>+  header.first_component_header_offset = header_offset;
>+  header.augmenting_data_offset =
>+      header_offset +
>+      kMultiplexImageComponentHeaderSize * header.component_count;
>+  header.augmenting_data_size = multiplex_image.augmenting_data_size;
>+  int bitstream_offset =
>+      header.augmenting_data_offset + header.augmenting_data_size;
>+
>+  const std::vector<MultiplexImageComponent>& images =
>+      multiplex_image.image_components;
>+  EncodedImage combined_image = images[0].encoded_image;
>+  for (size_t i = 0; i < images.size(); i++) {
>+    MultiplexImageComponentHeader frame_header;
>+    header_offset += kMultiplexImageComponentHeaderSize;
>+    frame_header.next_component_header_offset =
>+        (i == images.size() - 1) ? 0 : header_offset;
>+    frame_header.component_index = images[i].component_index;
>+
>+    frame_header.bitstream_offset = bitstream_offset;
>+    const size_t padding =
>+        EncodedImage::GetBufferPaddingBytes(images[i].codec_type);
>+    frame_header.bitstream_length =
>+        static_cast<uint32_t>(images[i].encoded_image._length + padding);
>+    bitstream_offset += frame_header.bitstream_length;
>+
>+    frame_header.codec_type = images[i].codec_type;
>+    frame_header.frame_type = images[i].encoded_image._frameType;
>+
>+    // As long as one component is delta frame, we have to mark the combined
>+    // frame as delta frame, because it is necessary for all components to be
>+    // key frame so as to decode the whole image without previous frame data.
>+    // Thus only when all components are key frames, we can mark the combined
>+    // frame as key frame.
>+    if (frame_header.frame_type == FrameType::kVideoFrameDelta) {
>+      combined_image._frameType = FrameType::kVideoFrameDelta;
>+    }
>+
>+    frame_headers.push_back(frame_header);
>+  }
>+
>+  combined_image._length = combined_image._size = bitstream_offset;
>+  combined_image._buffer = new uint8_t[combined_image._length];
>+
>+  // header
>+  header_offset = PackHeader(combined_image._buffer, header);
>+  RTC_DCHECK_EQ(header.first_component_header_offset,
>+                kMultiplexImageHeaderSize);
>+
>+  // Frame Header
>+  for (size_t i = 0; i < images.size(); i++) {
>+    int relative_offset = PackFrameHeader(
>+        combined_image._buffer + header_offset, frame_headers[i]);
>+    RTC_DCHECK_EQ(relative_offset, kMultiplexImageComponentHeaderSize);
>+
>+    header_offset = frame_headers[i].next_component_header_offset;
>+    RTC_DCHECK_EQ(header_offset,
>+                  (i == images.size() - 1)
>+                      ? 0
>+                      : (kMultiplexImageHeaderSize +
>+                         kMultiplexImageComponentHeaderSize * (i + 1)));
>+  }
>+
>+  // Augmenting Data
>+  if (multiplex_image.augmenting_data_size != 0) {
>+    memcpy(combined_image._buffer + header.augmenting_data_offset,
>+           multiplex_image.augmenting_data.get(),
>+           multiplex_image.augmenting_data_size);
>+  }
>+
>+  // Bitstreams
>+  for (size_t i = 0; i < images.size(); i++) {
>+    PackBitstream(combined_image._buffer + frame_headers[i].bitstream_offset,
>+                  images[i]);
>+    delete[] images[i].encoded_image._buffer;
>+  }
>+
>+  return combined_image;
>+}
>+
>+MultiplexImage MultiplexEncodedImagePacker::Unpack(
>+    const EncodedImage& combined_image) {
>+  const MultiplexImageHeader& header = UnpackHeader(combined_image._buffer);
>+
>+  std::vector<MultiplexImageComponentHeader> frame_headers;
>+  int header_offset = header.first_component_header_offset;
>+
>+  while (header_offset > 0) {
>+    frame_headers.push_back(
>+        UnpackFrameHeader(combined_image._buffer + header_offset));
>+    header_offset = frame_headers.back().next_component_header_offset;
>+  }
>+
>+  RTC_DCHECK_LE(frame_headers.size(), header.component_count);
>+  std::unique_ptr<uint8_t[]> augmenting_data = nullptr;
>+  if (header.augmenting_data_size != 0) {
>+    augmenting_data =
>+        std::unique_ptr<uint8_t[]>(new uint8_t[header.augmenting_data_size]);
>+    memcpy(augmenting_data.get(),
>+           combined_image._buffer + header.augmenting_data_offset,
>+           header.augmenting_data_size);
>+  }
>+
>+  MultiplexImage multiplex_image(header.image_index, header.component_count,
>+                                 std::move(augmenting_data),
>+                                 header.augmenting_data_size);
>+
>+  for (size_t i = 0; i < frame_headers.size(); i++) {
>+    MultiplexImageComponent image_component;
>+    image_component.component_index = frame_headers[i].component_index;
>+    image_component.codec_type = frame_headers[i].codec_type;
>+
>+    EncodedImage encoded_image = combined_image;
>+    encoded_image._timeStamp = combined_image._timeStamp;
>+    encoded_image._frameType = frame_headers[i].frame_type;
>+    encoded_image._size =
>+        static_cast<size_t>(frame_headers[i].bitstream_length);
>+    const size_t padding =
>+        EncodedImage::GetBufferPaddingBytes(image_component.codec_type);
>+    encoded_image._length = encoded_image._size - padding;
>+    encoded_image._buffer =
>+        combined_image._buffer + frame_headers[i].bitstream_offset;
>+
>+    image_component.encoded_image = encoded_image;
>+
>+    multiplex_image.image_components.push_back(image_component);
>+  }
>+
>+  return multiplex_image;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/multiplex/multiplex_encoder_adapter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/multiplex/multiplex_encoder_adapter.cc
>new file mode 100644
>index 00000000000..6f921e371ee
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/multiplex/multiplex_encoder_adapter.cc
>@@ -0,0 +1,297 @@
>+/*
>+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/video_coding/codecs/multiplex/include/multiplex_encoder_adapter.h"
>+
>+#include <cstring>
>+
>+#include "common_video/include/video_frame.h"
>+#include "common_video/include/video_frame_buffer.h"
>+#include "common_video/libyuv/include/webrtc_libyuv.h"
>+#include "modules/include/module_common_types.h"
>+#include "modules/video_coding/codecs/multiplex/include/augmented_video_frame_buffer.h"
>+#include "rtc_base/keep_ref_until_done.h"
>+#include "rtc_base/logging.h"
>+
>+namespace webrtc {
>+
>+// Callback wrapper that helps distinguish returned results from |encoders_|
>+// instances.
>+class MultiplexEncoderAdapter::AdapterEncodedImageCallback
>+    : public webrtc::EncodedImageCallback {
>+ public:
>+  AdapterEncodedImageCallback(webrtc::MultiplexEncoderAdapter* adapter,
>+                              AlphaCodecStream stream_idx)
>+      : adapter_(adapter), stream_idx_(stream_idx) {}
>+
>+  EncodedImageCallback::Result OnEncodedImage(
>+      const EncodedImage& encoded_image,
>+      const CodecSpecificInfo* codec_specific_info,
>+      const RTPFragmentationHeader* fragmentation) override {
>+    if (!adapter_)
>+      return Result(Result::OK);
>+    return adapter_->OnEncodedImage(stream_idx_, encoded_image,
>+                                    codec_specific_info, fragmentation);
>+  }
>+
>+ private:
>+  MultiplexEncoderAdapter* adapter_;
>+  const AlphaCodecStream stream_idx_;
>+};
>+
>+MultiplexEncoderAdapter::MultiplexEncoderAdapter(
>+    VideoEncoderFactory* factory,
>+    const SdpVideoFormat& associated_format,
>+    bool supports_augmented_data)
>+    : factory_(factory),
>+      associated_format_(associated_format),
>+      encoded_complete_callback_(nullptr),
>+      supports_augmented_data_(supports_augmented_data) {}
>+
>+MultiplexEncoderAdapter::~MultiplexEncoderAdapter() {
>+  Release();
>+}
>+
>+int MultiplexEncoderAdapter::InitEncode(const VideoCodec* inst,
>+                                        int number_of_cores,
>+                                        size_t max_payload_size) {
>+  const size_t buffer_size =
>+      CalcBufferSize(VideoType::kI420, inst->width, inst->height);
>+  multiplex_dummy_planes_.resize(buffer_size);
>+  // It is more expensive to encode 0x00, so use 0x80 instead.
>+  std::fill(multiplex_dummy_planes_.begin(), multiplex_dummy_planes_.end(),
>+            0x80);
>+
>+  RTC_DCHECK_EQ(kVideoCodecMultiplex, inst->codecType);
>+  VideoCodec settings = *inst;
>+  settings.codecType = PayloadStringToCodecType(associated_format_.name);
>+
>+  // Take over the key frame interval at adapter level, because we have to
>+  // sync the key frames for both sub-encoders.
>+  switch (settings.codecType) {
>+    case kVideoCodecVP8:
>+      key_frame_interval_ = settings.VP8()->keyFrameInterval;
>+      settings.VP8()->keyFrameInterval = 0;
>+      break;
>+    case kVideoCodecVP9:
>+      key_frame_interval_ = settings.VP9()->keyFrameInterval;
>+      settings.VP9()->keyFrameInterval = 0;
>+      break;
>+    case kVideoCodecH264:
>+      key_frame_interval_ = settings.H264()->keyFrameInterval;
>+      settings.H264()->keyFrameInterval = 0;
>+      break;
>+    default:
>+      break;
>+  }
>+
>+  for (size_t i = 0; i < kAlphaCodecStreams; ++i) {
>+    std::unique_ptr<VideoEncoder> encoder =
>+        factory_->CreateVideoEncoder(associated_format_);
>+    const int rv =
>+        encoder->InitEncode(&settings, number_of_cores, max_payload_size);
>+    if (rv) {
>+      RTC_LOG(LS_ERROR) << "Failed to create multiplex codec index " << i;
>+      return rv;
>+    }
>+    adapter_callbacks_.emplace_back(new AdapterEncodedImageCallback(
>+        this, static_cast<AlphaCodecStream>(i)));
>+    encoder->RegisterEncodeCompleteCallback(adapter_callbacks_.back().get());
>+    encoders_.emplace_back(std::move(encoder));
>+  }
>+  return WEBRTC_VIDEO_CODEC_OK;
>+}
>+
>+int MultiplexEncoderAdapter::Encode(
>+    const VideoFrame& input_image,
>+    const CodecSpecificInfo* codec_specific_info,
>+    const std::vector<FrameType>* frame_types) {
>+  if (!encoded_complete_callback_) {
>+    return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
>+  }
>+
>+  std::vector<FrameType> adjusted_frame_types;
>+  if (key_frame_interval_ > 0 && picture_index_ % key_frame_interval_ == 0) {
>+    adjusted_frame_types.push_back(kVideoFrameKey);
>+  } else {
>+    adjusted_frame_types.push_back(kVideoFrameDelta);
>+  }
>+  const bool has_alpha = input_image.video_frame_buffer()->type() ==
>+                         VideoFrameBuffer::Type::kI420A;
>+  std::unique_ptr<uint8_t[]> augmenting_data = nullptr;
>+  uint16_t augmenting_data_length = 0;
>+  AugmentedVideoFrameBuffer* augmented_video_frame_buffer = nullptr;
>+  if (supports_augmented_data_) {
>+    augmented_video_frame_buffer = static_cast<AugmentedVideoFrameBuffer*>(
>+        input_image.video_frame_buffer().get());
>+    augmenting_data_length =
>+        augmented_video_frame_buffer->GetAugmentingDataSize();
>+    augmenting_data =
>+        std::unique_ptr<uint8_t[]>(new uint8_t[augmenting_data_length]);
>+    memcpy(augmenting_data.get(),
>+           augmented_video_frame_buffer->GetAugmentingData(),
>+           augmenting_data_length);
>+    augmenting_data_size_ = augmenting_data_length;
>+  }
>+
>+  {
>+    rtc::CritScope cs(&crit_);
>+    stashed_images_.emplace(
>+        std::piecewise_construct,
>+        std::forward_as_tuple(input_image.timestamp()),
>+        std::forward_as_tuple(
>+            picture_index_, has_alpha ? kAlphaCodecStreams : 1,
>+            std::move(augmenting_data), augmenting_data_length));
>+  }
>+
>+  ++picture_index_;
>+
>+  // Encode YUV
>+  int rv = encoders_[kYUVStream]->Encode(input_image, codec_specific_info,
>+                                         &adjusted_frame_types);
>+
>+  // If we do not receive an alpha frame, we send a single frame for this
>+  // |picture_index_|. The receiver will receive |frame_count| as 1 which
>+  // specifies this case.
>+  if (rv || !has_alpha)
>+    return rv;
>+
>+  // Encode AXX
>+  const I420ABufferInterface* yuva_buffer =
>+      supports_augmented_data_
>+          ? augmented_video_frame_buffer->GetVideoFrameBuffer()->GetI420A()
>+          : input_image.video_frame_buffer()->GetI420A();
>+  rtc::scoped_refptr<I420BufferInterface> alpha_buffer =
>+      WrapI420Buffer(input_image.width(), input_image.height(),
>+                     yuva_buffer->DataA(), yuva_buffer->StrideA(),
>+                     multiplex_dummy_planes_.data(), yuva_buffer->StrideU(),
>+                     multiplex_dummy_planes_.data(), yuva_buffer->StrideV(),
>+                     rtc::KeepRefUntilDone(input_image.video_frame_buffer()));
>+  VideoFrame alpha_image(alpha_buffer, input_image.timestamp(),
>+                         input_image.render_time_ms(), input_image.rotation());
>+  rv = encoders_[kAXXStream]->Encode(alpha_image, codec_specific_info,
>+                                     &adjusted_frame_types);
>+  return rv;
>+}
>+
>+int MultiplexEncoderAdapter::RegisterEncodeCompleteCallback(
>+    EncodedImageCallback* callback) {
>+  encoded_complete_callback_ = callback;
>+  return WEBRTC_VIDEO_CODEC_OK;
>+}
>+
>+int MultiplexEncoderAdapter::SetChannelParameters(uint32_t packet_loss,
>+                                                  int64_t rtt) {
>+  for (auto& encoder : encoders_) {
>+    const int rv = encoder->SetChannelParameters(packet_loss, rtt);
>+    if (rv)
>+      return rv;
>+  }
>+  return WEBRTC_VIDEO_CODEC_OK;
>+}
>+
>+int MultiplexEncoderAdapter::SetRateAllocation(
>+    const VideoBitrateAllocation& bitrate,
>+    uint32_t framerate) {
>+  VideoBitrateAllocation bitrate_allocation(bitrate);
>+  bitrate_allocation.SetBitrate(
>+      0, 0, bitrate.GetBitrate(0, 0) - augmenting_data_size_);
>+  for (auto& encoder : encoders_) {
>+    // TODO(emircan): |framerate| is used to calculate duration in encoder
>+    // instances. We report the total frame rate to keep real time for now.
>+    // Remove this after refactoring duration logic.
>+    const int rv = encoder->SetRateAllocation(
>+        bitrate_allocation,
>+        static_cast<uint32_t>(encoders_.size()) * framerate);
>+    if (rv)
>+      return rv;
>+  }
>+  return WEBRTC_VIDEO_CODEC_OK;
>+}
>+
>+int MultiplexEncoderAdapter::Release() {
>+  for (auto& encoder : encoders_) {
>+    const int rv = encoder->Release();
>+    if (rv)
>+      return rv;
>+  }
>+  encoders_.clear();
>+  adapter_callbacks_.clear();
>+  rtc::CritScope cs(&crit_);
>+  for (auto& stashed_image : stashed_images_) {
>+    for (auto& image_component : stashed_image.second.image_components) {
>+      delete[] image_component.encoded_image._buffer;
>+    }
>+  }
>+  stashed_images_.clear();
>+  if (combined_image_._buffer) {
>+    delete[] combined_image_._buffer;
>+    combined_image_._buffer = nullptr;
>+  }
>+  return WEBRTC_VIDEO_CODEC_OK;
>+}
>+
>+const char* MultiplexEncoderAdapter::ImplementationName() const {
>+  return "MultiplexEncoderAdapter";
>+}
>+
>+EncodedImageCallback::Result MultiplexEncoderAdapter::OnEncodedImage(
>+    AlphaCodecStream stream_idx,
>+    const EncodedImage& encodedImage,
>+    const CodecSpecificInfo* codecSpecificInfo,
>+    const RTPFragmentationHeader* fragmentation) {
>+  // Save the image
>+  MultiplexImageComponent image_component;
>+  image_component.component_index = stream_idx;
>+  image_component.codec_type =
>+      PayloadStringToCodecType(associated_format_.name);
>+  image_component.encoded_image = encodedImage;
>+  image_component.encoded_image._buffer = new uint8_t[encodedImage._length];
>+  std::memcpy(image_component.encoded_image._buffer, encodedImage._buffer,
>+              encodedImage._length);
>+
>+  rtc::CritScope cs(&crit_);
>+  const auto& stashed_image_itr = stashed_images_.find(encodedImage._timeStamp);
>+  const auto& stashed_image_next_itr = std::next(stashed_image_itr, 1);
>+  RTC_DCHECK(stashed_image_itr != stashed_images_.end());
>+  MultiplexImage& stashed_image = stashed_image_itr->second;
>+  const uint8_t frame_count = stashed_image.component_count;
>+
>+  stashed_image.image_components.push_back(image_component);
>+
>+  if (stashed_image.image_components.size() == frame_count) {
>+    // Complete case
>+    for (auto iter = stashed_images_.begin();
>+         iter != stashed_images_.end() && iter != stashed_image_next_itr;
>+         iter++) {
>+      // No image at all, skip.
>+      if (iter->second.image_components.size() == 0)
>+        continue;
>+
>+      // We have to send out those stashed frames, otherwise the delta frame
>+      // dependency chain is broken.
>+      if (combined_image_._buffer)
>+        delete[] combined_image_._buffer;
>+      combined_image_ =
>+          MultiplexEncodedImagePacker::PackAndRelease(iter->second);
>+
>+      CodecSpecificInfo codec_info = *codecSpecificInfo;
>+      codec_info.codecType = kVideoCodecMultiplex;
>+      codec_info.codecSpecific.generic.simulcast_idx = 0;
>+      encoded_complete_callback_->OnEncodedImage(combined_image_, &codec_info,
>+                                                 fragmentation);
>+    }
>+
>+    stashed_images_.erase(stashed_images_.begin(), stashed_image_next_itr);
>+  }
>+  return EncodedImageCallback::Result(EncodedImageCallback::Result::OK);
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/multiplex/test/multiplex_adapter_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/multiplex/test/multiplex_adapter_unittest.cc
>new file mode 100644
>index 00000000000..303914d95ea
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/multiplex/test/multiplex_adapter_unittest.cc
>@@ -0,0 +1,292 @@
>+/*
>+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "absl/memory/memory.h"
>+#include "api/test/mock_video_decoder_factory.h"
>+#include "api/test/mock_video_encoder_factory.h"
>+#include "api/video_codecs/sdp_video_format.h"
>+#include "common_video/include/video_frame_buffer.h"
>+#include "common_video/libyuv/include/webrtc_libyuv.h"
>+#include "media/base/mediaconstants.h"
>+#include "modules/video_coding/codecs/multiplex/include/augmented_video_frame_buffer.h"
>+#include "modules/video_coding/codecs/multiplex/include/multiplex_decoder_adapter.h"
>+#include "modules/video_coding/codecs/multiplex/include/multiplex_encoded_image_packer.h"
>+#include "modules/video_coding/codecs/multiplex/include/multiplex_encoder_adapter.h"
>+#include "modules/video_coding/codecs/test/video_codec_unittest.h"
>+#include "modules/video_coding/codecs/vp9/include/vp9.h"
>+#include "rtc_base/keep_ref_until_done.h"
>+#include "test/video_codec_settings.h"
>+
>+using testing::_;
>+using testing::Return;
>+
>+namespace webrtc {
>+
>+constexpr const char* kMultiplexAssociatedCodecName = cricket::kVp9CodecName;
>+const VideoCodecType kMultiplexAssociatedCodecType =
>+    PayloadStringToCodecType(kMultiplexAssociatedCodecName);
>+
>+class TestMultiplexAdapter
>+    : public VideoCodecUnitTest,
>+      public testing::WithParamInterface<bool /* supports_augmenting_data */> {
>+ public:
>+  TestMultiplexAdapter()
>+      : decoder_factory_(new webrtc::MockVideoDecoderFactory),
>+        encoder_factory_(new webrtc::MockVideoEncoderFactory),
>+        supports_augmenting_data_(GetParam()) {}
>+
>+ protected:
>+  std::unique_ptr<VideoDecoder> CreateDecoder() override {
>+    return absl::make_unique<MultiplexDecoderAdapter>(
>+        decoder_factory_.get(), SdpVideoFormat(kMultiplexAssociatedCodecName),
>+        supports_augmenting_data_);
>+  }
>+
>+  std::unique_ptr<VideoEncoder> CreateEncoder() override {
>+    return absl::make_unique<MultiplexEncoderAdapter>(
>+        encoder_factory_.get(), SdpVideoFormat(kMultiplexAssociatedCodecName),
>+        supports_augmenting_data_);
>+  }
>+
>+  void ModifyCodecSettings(VideoCodec* codec_settings) override {
>+    webrtc::test::CodecSettings(kMultiplexAssociatedCodecType, codec_settings);
>+    codec_settings->VP9()->numberOfTemporalLayers = 1;
>+    codec_settings->VP9()->numberOfSpatialLayers = 1;
>+    codec_settings->codecType = webrtc::kVideoCodecMultiplex;
>+  }
>+
>+  std::unique_ptr<VideoFrame> CreateDataAugmentedInputFrame(
>+      VideoFrame* video_frame) {
>+    rtc::scoped_refptr<VideoFrameBuffer> video_buffer =
>+        video_frame->video_frame_buffer();
>+    std::unique_ptr<uint8_t[]> data =
>+        std::unique_ptr<uint8_t[]>(new uint8_t[16]);
>+    for (int i = 0; i < 16; i++) {
>+      data[i] = i;
>+    }
>+    rtc::scoped_refptr<AugmentedVideoFrameBuffer> augmented_video_frame_buffer =
>+        new rtc::RefCountedObject<AugmentedVideoFrameBuffer>(
>+            video_buffer, std::move(data), 16);
>+    return absl::WrapUnique<VideoFrame>(
>+        new VideoFrame(augmented_video_frame_buffer, video_frame->timestamp(),
>+                       video_frame->render_time_ms(), video_frame->rotation()));
>+  }
>+
>+  std::unique_ptr<VideoFrame> CreateI420AInputFrame() {
>+    VideoFrame* input_frame = NextInputFrame();
>+    rtc::scoped_refptr<webrtc::I420BufferInterface> yuv_buffer =
>+        input_frame->video_frame_buffer()->ToI420();
>+    rtc::scoped_refptr<I420ABufferInterface> yuva_buffer = WrapI420ABuffer(
>+        yuv_buffer->width(), yuv_buffer->height(), yuv_buffer->DataY(),
>+        yuv_buffer->StrideY(), yuv_buffer->DataU(), yuv_buffer->StrideU(),
>+        yuv_buffer->DataV(), yuv_buffer->StrideV(), yuv_buffer->DataY(),
>+        yuv_buffer->StrideY(), rtc::KeepRefUntilDone(yuv_buffer));
>+    return absl::WrapUnique<VideoFrame>(
>+        new VideoFrame(yuva_buffer, 123 /* RTP timestamp */,
>+                       345 /* render_time_ms */, kVideoRotation_0));
>+  }
>+
>+  std::unique_ptr<VideoFrame> CreateInputFrame(bool contains_alpha) {
>+    std::unique_ptr<VideoFrame> video_frame;
>+    if (contains_alpha) {
>+      video_frame = CreateI420AInputFrame();
>+    } else {
>+      VideoFrame* next_frame = NextInputFrame();
>+      video_frame = absl::WrapUnique<VideoFrame>(new VideoFrame(
>+          next_frame->video_frame_buffer(), next_frame->timestamp(),
>+          next_frame->render_time_ms(), next_frame->rotation()));
>+    }
>+    if (supports_augmenting_data_) {
>+      video_frame = CreateDataAugmentedInputFrame(video_frame.get());
>+    }
>+
>+    return video_frame;
>+  }
>+
>+  void CheckData(rtc::scoped_refptr<VideoFrameBuffer> video_frame_buffer) {
>+    if (!supports_augmenting_data_) {
>+      return;
>+    }
>+    AugmentedVideoFrameBuffer* augmented_buffer =
>+        static_cast<AugmentedVideoFrameBuffer*>(video_frame_buffer.get());
>+    EXPECT_EQ(augmented_buffer->GetAugmentingDataSize(), 16);
>+    uint8_t* data = augmented_buffer->GetAugmentingData();
>+    for (int i = 0; i < 16; i++) {
>+      EXPECT_EQ(data[i], i);
>+    }
>+  }
>+
>+  std::unique_ptr<VideoFrame> ExtractAXXFrame(const VideoFrame& video_frame) {
>+    rtc::scoped_refptr<VideoFrameBuffer> video_frame_buffer =
>+        video_frame.video_frame_buffer();
>+    if (supports_augmenting_data_) {
>+      AugmentedVideoFrameBuffer* augmentedBuffer =
>+          static_cast<AugmentedVideoFrameBuffer*>(video_frame_buffer.get());
>+      video_frame_buffer = augmentedBuffer->GetVideoFrameBuffer();
>+    }
>+    const I420ABufferInterface* yuva_buffer = video_frame_buffer->GetI420A();
>+    rtc::scoped_refptr<I420BufferInterface> axx_buffer = WrapI420Buffer(
>+        yuva_buffer->width(), yuva_buffer->height(), yuva_buffer->DataA(),
>+        yuva_buffer->StrideA(), yuva_buffer->DataU(), yuva_buffer->StrideU(),
>+        yuva_buffer->DataV(), yuva_buffer->StrideV(),
>+        rtc::KeepRefUntilDone(video_frame_buffer));
>+    return absl::WrapUnique<VideoFrame>(
>+        new VideoFrame(axx_buffer, 123 /* RTP timestamp */,
>+                       345 /* render_time_ms */, kVideoRotation_0));
>+  }
>+
>+ private:
>+  void SetUp() override {
>+    EXPECT_CALL(*decoder_factory_, Die());
>+    // The decoders/encoders will be owned by the caller of
>+    // CreateVideoDecoder()/CreateVideoEncoder().
>+    VideoDecoder* decoder1 = VP9Decoder::Create().release();
>+    VideoDecoder* decoder2 = VP9Decoder::Create().release();
>+    EXPECT_CALL(*decoder_factory_, CreateVideoDecoderProxy(_))
>+        .WillOnce(Return(decoder1))
>+        .WillOnce(Return(decoder2));
>+
>+    EXPECT_CALL(*encoder_factory_, Die());
>+    VideoEncoder* encoder1 = VP9Encoder::Create().release();
>+    VideoEncoder* encoder2 = VP9Encoder::Create().release();
>+    EXPECT_CALL(*encoder_factory_, CreateVideoEncoderProxy(_))
>+        .WillOnce(Return(encoder1))
>+        .WillOnce(Return(encoder2));
>+
>+    VideoCodecUnitTest::SetUp();
>+  }
>+
>+  const std::unique_ptr<webrtc::MockVideoDecoderFactory> decoder_factory_;
>+  const std::unique_ptr<webrtc::MockVideoEncoderFactory> encoder_factory_;
>+  const bool supports_augmenting_data_;
>+};
>+
>+// TODO(emircan): Currently VideoCodecUnitTest tests do a complete setup
>+// step that goes beyond constructing |decoder_|. Simplify these tests to do
>+// less.
>+TEST_P(TestMultiplexAdapter, ConstructAndDestructDecoder) {
>+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, decoder_->Release());
>+}
>+
>+TEST_P(TestMultiplexAdapter, ConstructAndDestructEncoder) {
>+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Release());
>+}
>+
>+TEST_P(TestMultiplexAdapter, EncodeDecodeI420Frame) {
>+  std::unique_ptr<VideoFrame> input_frame = CreateInputFrame(false);
>+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>+            encoder_->Encode(*input_frame, nullptr, nullptr));
>+  EncodedImage encoded_frame;
>+  CodecSpecificInfo codec_specific_info;
>+  ASSERT_TRUE(WaitForEncodedFrame(&encoded_frame, &codec_specific_info));
>+  EXPECT_EQ(kVideoCodecMultiplex, codec_specific_info.codecType);
>+
>+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>+            decoder_->Decode(encoded_frame, false, &codec_specific_info, -1));
>+  std::unique_ptr<VideoFrame> decoded_frame;
>+  absl::optional<uint8_t> decoded_qp;
>+  ASSERT_TRUE(WaitForDecodedFrame(&decoded_frame, &decoded_qp));
>+  ASSERT_TRUE(decoded_frame);
>+  EXPECT_GT(I420PSNR(input_frame.get(), decoded_frame.get()), 36);
>+  CheckData(decoded_frame->video_frame_buffer());
>+}
>+
>+TEST_P(TestMultiplexAdapter, EncodeDecodeI420AFrame) {
>+  std::unique_ptr<VideoFrame> yuva_frame = CreateInputFrame(true);
>+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>+            encoder_->Encode(*yuva_frame, nullptr, nullptr));
>+  EncodedImage encoded_frame;
>+  CodecSpecificInfo codec_specific_info;
>+  ASSERT_TRUE(WaitForEncodedFrame(&encoded_frame, &codec_specific_info));
>+  EXPECT_EQ(kVideoCodecMultiplex, codec_specific_info.codecType);
>+
>+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>+            decoder_->Decode(encoded_frame, false, nullptr, 0));
>+  std::unique_ptr<VideoFrame> decoded_frame;
>+  absl::optional<uint8_t> decoded_qp;
>+  ASSERT_TRUE(WaitForDecodedFrame(&decoded_frame, &decoded_qp));
>+  ASSERT_TRUE(decoded_frame);
>+  EXPECT_GT(I420PSNR(yuva_frame.get(), decoded_frame.get()), 36);
>+
>+  // Find PSNR for AXX bits.
>+  std::unique_ptr<VideoFrame> input_axx_frame = ExtractAXXFrame(*yuva_frame);
>+  std::unique_ptr<VideoFrame> output_axx_frame =
>+      ExtractAXXFrame(*decoded_frame);
>+  EXPECT_GT(I420PSNR(input_axx_frame.get(), output_axx_frame.get()), 47);
>+
>+  CheckData(decoded_frame->video_frame_buffer());
>+}
>+
>+TEST_P(TestMultiplexAdapter, CheckSingleFrameEncodedBitstream) {
>+  std::unique_ptr<VideoFrame> input_frame = CreateInputFrame(false);
>+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>+            encoder_->Encode(*input_frame, nullptr, nullptr));
>+  EncodedImage encoded_frame;
>+  CodecSpecificInfo codec_specific_info;
>+  ASSERT_TRUE(WaitForEncodedFrame(&encoded_frame, &codec_specific_info));
>+  EXPECT_EQ(kVideoCodecMultiplex, codec_specific_info.codecType);
>+  EXPECT_EQ(0, codec_specific_info.codecSpecific.generic.simulcast_idx);
>+
>+  const MultiplexImage& unpacked_frame =
>+      MultiplexEncodedImagePacker::Unpack(encoded_frame);
>+  EXPECT_EQ(0, unpacked_frame.image_index);
>+  EXPECT_EQ(1, unpacked_frame.component_count);
>+  const MultiplexImageComponent& component = unpacked_frame.image_components[0];
>+  EXPECT_EQ(0, component.component_index);
>+  EXPECT_NE(nullptr, component.encoded_image._buffer);
>+  EXPECT_EQ(kVideoFrameKey, component.encoded_image._frameType);
>+}
>+
>+TEST_P(TestMultiplexAdapter, CheckDoubleFramesEncodedBitstream) {
>+  std::unique_ptr<VideoFrame> yuva_frame = CreateInputFrame(true);
>+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>+            encoder_->Encode(*yuva_frame, nullptr, nullptr));
>+  EncodedImage encoded_frame;
>+  CodecSpecificInfo codec_specific_info;
>+  ASSERT_TRUE(WaitForEncodedFrame(&encoded_frame, &codec_specific_info));
>+  EXPECT_EQ(kVideoCodecMultiplex, codec_specific_info.codecType);
>+  EXPECT_EQ(0, codec_specific_info.codecSpecific.generic.simulcast_idx);
>+
>+  const MultiplexImage& unpacked_frame =
>+      MultiplexEncodedImagePacker::Unpack(encoded_frame);
>+  EXPECT_EQ(0, unpacked_frame.image_index);
>+  EXPECT_EQ(2, unpacked_frame.component_count);
>+  EXPECT_EQ(unpacked_frame.image_components.size(),
>+            unpacked_frame.component_count);
>+  for (int i = 0; i < unpacked_frame.component_count; ++i) {
>+    const MultiplexImageComponent& component =
>+        unpacked_frame.image_components[i];
>+    EXPECT_EQ(i, component.component_index);
>+    EXPECT_NE(nullptr, component.encoded_image._buffer);
>+    EXPECT_EQ(kVideoFrameKey, component.encoded_image._frameType);
>+  }
>+}
>+
>+TEST_P(TestMultiplexAdapter, ImageIndexIncreases) {
>+  std::unique_ptr<VideoFrame> yuva_frame = CreateInputFrame(true);
>+  const size_t expected_num_encoded_frames = 3;
>+  for (size_t i = 0; i < expected_num_encoded_frames; ++i) {
>+    EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>+              encoder_->Encode(*yuva_frame, nullptr, nullptr));
>+    EncodedImage encoded_frame;
>+    CodecSpecificInfo codec_specific_info;
>+    ASSERT_TRUE(WaitForEncodedFrame(&encoded_frame, &codec_specific_info));
>+    const MultiplexImage& unpacked_frame =
>+        MultiplexEncodedImagePacker::Unpack(encoded_frame);
>+    EXPECT_EQ(i, unpacked_frame.image_index);
>+    EXPECT_EQ(i ? kVideoFrameDelta : kVideoFrameKey, encoded_frame._frameType);
>+  }
>+}
>+
>+INSTANTIATE_TEST_CASE_P(TestMultiplexAdapter,
>+                        TestMultiplexAdapter,
>+                        ::testing::Bool());
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/stereo/stereo_encoder_adapter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/stereo/stereo_encoder_adapter.cc
>deleted file mode 100644
>index 60e3ea8ceec..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/stereo/stereo_encoder_adapter.cc
>+++ /dev/null
>@@ -1,212 +0,0 @@
>-/*
>- *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "modules/video_coding/codecs/stereo/include/stereo_encoder_adapter.h"
>-
>-#include "common_video/include/video_frame.h"
>-#include "common_video/include/video_frame_buffer.h"
>-#include "common_video/libyuv/include/webrtc_libyuv.h"
>-#include "modules/include/module_common_types.h"
>-#include "rtc_base/keep_ref_until_done.h"
>-#include "rtc_base/logging.h"
>-
>-namespace webrtc {
>-
>-// Callback wrapper that helps distinguish returned results from |encoders_|
>-// instances.
>-class StereoEncoderAdapter::AdapterEncodedImageCallback
>-    : public webrtc::EncodedImageCallback {
>- public:
>-  AdapterEncodedImageCallback(webrtc::StereoEncoderAdapter* adapter,
>-                              AlphaCodecStream stream_idx)
>-      : adapter_(adapter), stream_idx_(stream_idx) {}
>-
>-  EncodedImageCallback::Result OnEncodedImage(
>-      const EncodedImage& encoded_image,
>-      const CodecSpecificInfo* codec_specific_info,
>-      const RTPFragmentationHeader* fragmentation) override {
>-    if (!adapter_)
>-      return Result(Result::OK);
>-    return adapter_->OnEncodedImage(stream_idx_, encoded_image,
>-                                    codec_specific_info, fragmentation);
>-  }
>-
>- private:
>-  StereoEncoderAdapter* adapter_;
>-  const AlphaCodecStream stream_idx_;
>-};
>-
>-// Holds the encoded image info.
>-struct StereoEncoderAdapter::ImageStereoInfo {
>-  ImageStereoInfo(uint16_t picture_index, uint8_t frame_count)
>-      : picture_index(picture_index),
>-        frame_count(frame_count),
>-        encoded_count(0) {}
>-  uint16_t picture_index;
>-  uint8_t frame_count;
>-  uint8_t encoded_count;
>-
>- private:
>-  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(ImageStereoInfo);
>-};
>-
>-StereoEncoderAdapter::StereoEncoderAdapter(
>-    VideoEncoderFactory* factory,
>-    const SdpVideoFormat& associated_format)
>-    : factory_(factory),
>-      associated_format_(associated_format),
>-      encoded_complete_callback_(nullptr) {}
>-
>-StereoEncoderAdapter::~StereoEncoderAdapter() {
>-  Release();
>-}
>-
>-int StereoEncoderAdapter::InitEncode(const VideoCodec* inst,
>-                                     int number_of_cores,
>-                                     size_t max_payload_size) {
>-  const size_t buffer_size =
>-      CalcBufferSize(VideoType::kI420, inst->width, inst->height);
>-  stereo_dummy_planes_.resize(buffer_size);
>-  // It is more expensive to encode 0x00, so use 0x80 instead.
>-  std::fill(stereo_dummy_planes_.begin(), stereo_dummy_planes_.end(), 0x80);
>-
>-  RTC_DCHECK_EQ(kVideoCodecStereo, inst->codecType);
>-  VideoCodec settings = *inst;
>-  settings.codecType = PayloadStringToCodecType(associated_format_.name);
>-  for (size_t i = 0; i < kAlphaCodecStreams; ++i) {
>-    std::unique_ptr<VideoEncoder> encoder =
>-        factory_->CreateVideoEncoder(associated_format_);
>-    const int rv =
>-        encoder->InitEncode(&settings, number_of_cores, max_payload_size);
>-    if (rv) {
>-      RTC_LOG(LS_ERROR) << "Failed to create stereo codec index " << i;
>-      return rv;
>-    }
>-    adapter_callbacks_.emplace_back(new AdapterEncodedImageCallback(
>-        this, static_cast<AlphaCodecStream>(i)));
>-    encoder->RegisterEncodeCompleteCallback(adapter_callbacks_.back().get());
>-    encoders_.emplace_back(std::move(encoder));
>-  }
>-  return WEBRTC_VIDEO_CODEC_OK;
>-}
>-
>-int StereoEncoderAdapter::Encode(const VideoFrame& input_image,
>-                                 const CodecSpecificInfo* codec_specific_info,
>-                                 const std::vector<FrameType>* frame_types) {
>-  if (!encoded_complete_callback_) {
>-    return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
>-  }
>-
>-  const bool has_alpha = input_image.video_frame_buffer()->type() ==
>-                         VideoFrameBuffer::Type::kI420A;
>-  image_stereo_info_.emplace(
>-      std::piecewise_construct, std::forward_as_tuple(input_image.timestamp()),
>-      std::forward_as_tuple(picture_index_++,
>-                            has_alpha ? kAlphaCodecStreams : 1));
>-
>-  // Encode YUV
>-  int rv = encoders_[kYUVStream]->Encode(input_image, codec_specific_info,
>-                                         frame_types);
>-  // If we do not receive an alpha frame, we send a single frame for this
>-  // |picture_index_|. The receiver will receive |frame_count| as 1 which
>-  // soecifies this case.
>-  if (rv || !has_alpha)
>-    return rv;
>-
>-  // Encode AXX
>-  const I420ABufferInterface* yuva_buffer =
>-      input_image.video_frame_buffer()->GetI420A();
>-  rtc::scoped_refptr<I420BufferInterface> alpha_buffer =
>-      WrapI420Buffer(input_image.width(), input_image.height(),
>-                     yuva_buffer->DataA(), yuva_buffer->StrideA(),
>-                     stereo_dummy_planes_.data(), yuva_buffer->StrideU(),
>-                     stereo_dummy_planes_.data(), yuva_buffer->StrideV(),
>-                     rtc::KeepRefUntilDone(input_image.video_frame_buffer()));
>-  VideoFrame alpha_image(alpha_buffer, input_image.timestamp(),
>-                         input_image.render_time_ms(), input_image.rotation());
>-  rv = encoders_[kAXXStream]->Encode(alpha_image, codec_specific_info,
>-                                     frame_types);
>-  return rv;
>-}
>-
>-int StereoEncoderAdapter::RegisterEncodeCompleteCallback(
>-    EncodedImageCallback* callback) {
>-  encoded_complete_callback_ = callback;
>-  return WEBRTC_VIDEO_CODEC_OK;
>-}
>-
>-int StereoEncoderAdapter::SetChannelParameters(uint32_t packet_loss,
>-                                               int64_t rtt) {
>-  for (auto& encoder : encoders_) {
>-    const int rv = encoder->SetChannelParameters(packet_loss, rtt);
>-    if (rv)
>-      return rv;
>-  }
>-  return WEBRTC_VIDEO_CODEC_OK;
>-}
>-
>-int StereoEncoderAdapter::SetRateAllocation(const BitrateAllocation& bitrate,
>-                                            uint32_t framerate) {
>-  for (auto& encoder : encoders_) {
>-    // TODO(emircan): |framerate| is used to calculate duration in encoder
>-    // instances. We report the total frame rate to keep real time for now.
>-    // Remove this after refactoring duration logic.
>-    const int rv = encoder->SetRateAllocation(
>-        bitrate, static_cast<uint32_t>(encoders_.size()) * framerate);
>-    if (rv)
>-      return rv;
>-  }
>-  return WEBRTC_VIDEO_CODEC_OK;
>-}
>-
>-int StereoEncoderAdapter::Release() {
>-  for (auto& encoder : encoders_) {
>-    const int rv = encoder->Release();
>-    if (rv)
>-      return rv;
>-  }
>-  encoders_.clear();
>-  adapter_callbacks_.clear();
>-  return WEBRTC_VIDEO_CODEC_OK;
>-}
>-
>-const char* StereoEncoderAdapter::ImplementationName() const {
>-  return "StereoEncoderAdapter";
>-}
>-
>-EncodedImageCallback::Result StereoEncoderAdapter::OnEncodedImage(
>-    AlphaCodecStream stream_idx,
>-    const EncodedImage& encodedImage,
>-    const CodecSpecificInfo* codecSpecificInfo,
>-    const RTPFragmentationHeader* fragmentation) {
>-  const VideoCodecType associated_codec_type = codecSpecificInfo->codecType;
>-  const auto& image_stereo_info_itr =
>-      image_stereo_info_.find(encodedImage._timeStamp);
>-  RTC_DCHECK(image_stereo_info_itr != image_stereo_info_.end());
>-  ImageStereoInfo& image_stereo_info = image_stereo_info_itr->second;
>-  const uint8_t frame_count = image_stereo_info.frame_count;
>-  const uint16_t picture_index = image_stereo_info.picture_index;
>-  if (++image_stereo_info.encoded_count == frame_count)
>-    image_stereo_info_.erase(image_stereo_info_itr);
>-
>-  CodecSpecificInfo codec_info = *codecSpecificInfo;
>-  codec_info.codecType = kVideoCodecStereo;
>-  codec_info.codec_name = "stereo";
>-  codec_info.codecSpecific.stereo.associated_codec_type = associated_codec_type;
>-  codec_info.codecSpecific.stereo.indices.frame_index = stream_idx;
>-  codec_info.codecSpecific.stereo.indices.frame_count = frame_count;
>-  codec_info.codecSpecific.stereo.indices.picture_index = picture_index;
>-
>-  encoded_complete_callback_->OnEncodedImage(encodedImage, &codec_info,
>-                                             fragmentation);
>-  return EncodedImageCallback::Result(EncodedImageCallback::Result::OK);
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/stereo/test/stereo_adapter_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/stereo/test/stereo_adapter_unittest.cc
>deleted file mode 100644
>index 976a5bb8a25..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/stereo/test/stereo_adapter_unittest.cc
>+++ /dev/null
>@@ -1,173 +0,0 @@
>-/*
>- *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "api/test/mock_video_decoder_factory.h"
>-#include "api/test/mock_video_encoder_factory.h"
>-#include "api/video_codecs/sdp_video_format.h"
>-#include "common_video/include/video_frame_buffer.h"
>-#include "common_video/libyuv/include/webrtc_libyuv.h"
>-#include "media/base/mediaconstants.h"
>-#include "modules/video_coding/codecs/stereo/include/stereo_decoder_adapter.h"
>-#include "modules/video_coding/codecs/stereo/include/stereo_encoder_adapter.h"
>-#include "modules/video_coding/codecs/test/video_codec_test.h"
>-#include "modules/video_coding/codecs/vp9/include/vp9.h"
>-#include "rtc_base/keep_ref_until_done.h"
>-#include "rtc_base/ptr_util.h"
>-
>-using testing::_;
>-using testing::Return;
>-
>-namespace webrtc {
>-
>-constexpr const char* kStereoAssociatedCodecName = cricket::kVp9CodecName;
>-const VideoCodecType kStereoAssociatedCodecType =
>-    PayloadStringToCodecType(kStereoAssociatedCodecName);
>-
>-class TestStereoAdapter : public VideoCodecTest {
>- public:
>-  TestStereoAdapter()
>-      : decoder_factory_(new webrtc::MockVideoDecoderFactory),
>-        encoder_factory_(new webrtc::MockVideoEncoderFactory) {}
>-
>- protected:
>-  std::unique_ptr<VideoDecoder> CreateDecoder() override {
>-    return rtc::MakeUnique<StereoDecoderAdapter>(
>-        decoder_factory_.get(), SdpVideoFormat(kStereoAssociatedCodecName));
>-  }
>-
>-  std::unique_ptr<VideoEncoder> CreateEncoder() override {
>-    return rtc::MakeUnique<StereoEncoderAdapter>(
>-        encoder_factory_.get(), SdpVideoFormat(kStereoAssociatedCodecName));
>-  }
>-
>-  VideoCodec codec_settings() override {
>-    VideoCodec codec_settings;
>-    codec_settings.codecType = kStereoAssociatedCodecType;
>-    codec_settings.VP9()->numberOfTemporalLayers = 1;
>-    codec_settings.VP9()->numberOfSpatialLayers = 1;
>-    codec_settings.codecType = webrtc::kVideoCodecStereo;
>-    return codec_settings;
>-  }
>-
>-  std::unique_ptr<VideoFrame> CreateI420AInputFrame() {
>-    rtc::scoped_refptr<webrtc::I420BufferInterface> yuv_buffer =
>-        input_frame_->video_frame_buffer()->ToI420();
>-    rtc::scoped_refptr<I420ABufferInterface> yuva_buffer = WrapI420ABuffer(
>-        yuv_buffer->width(), yuv_buffer->height(), yuv_buffer->DataY(),
>-        yuv_buffer->StrideY(), yuv_buffer->DataU(), yuv_buffer->StrideU(),
>-        yuv_buffer->DataV(), yuv_buffer->StrideV(), yuv_buffer->DataY(),
>-        yuv_buffer->StrideY(), rtc::KeepRefUntilDone(yuv_buffer));
>-    return rtc::WrapUnique<VideoFrame>(
>-        new VideoFrame(yuva_buffer, kVideoRotation_0, 0));
>-  }
>-
>- private:
>-  void SetUp() override {
>-    EXPECT_CALL(*decoder_factory_, Die());
>-    // The decoders/encoders will be owned by the caller of
>-    // CreateVideoDecoder()/CreateVideoEncoder().
>-    VideoDecoder* decoder1 = VP9Decoder::Create().release();
>-    VideoDecoder* decoder2 = VP9Decoder::Create().release();
>-    EXPECT_CALL(*decoder_factory_, CreateVideoDecoderProxy(_))
>-        .WillOnce(Return(decoder1))
>-        .WillOnce(Return(decoder2));
>-
>-    EXPECT_CALL(*encoder_factory_, Die());
>-    VideoEncoder* encoder1 = VP9Encoder::Create().release();
>-    VideoEncoder* encoder2 = VP9Encoder::Create().release();
>-    EXPECT_CALL(*encoder_factory_, CreateVideoEncoderProxy(_))
>-        .WillOnce(Return(encoder1))
>-        .WillOnce(Return(encoder2));
>-
>-    VideoCodecTest::SetUp();
>-  }
>-
>-  const std::unique_ptr<webrtc::MockVideoDecoderFactory> decoder_factory_;
>-  const std::unique_ptr<webrtc::MockVideoEncoderFactory> encoder_factory_;
>-};
>-
>-// TODO(emircan): Currently VideoCodecTest tests do a complete setup
>-// step that goes beyond constructing |decoder_|. Simplify these tests to do
>-// less.
>-TEST_F(TestStereoAdapter, ConstructAndDestructDecoder) {
>-  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, decoder_->Release());
>-}
>-
>-TEST_F(TestStereoAdapter, ConstructAndDestructEncoder) {
>-  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Release());
>-}
>-
>-TEST_F(TestStereoAdapter, EncodeDecodeI420Frame) {
>-  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>-            encoder_->Encode(*input_frame_, nullptr, nullptr));
>-  EncodedImage encoded_frame;
>-  CodecSpecificInfo codec_specific_info;
>-  ASSERT_TRUE(WaitForEncodedFrame(&encoded_frame, &codec_specific_info));
>-
>-  EXPECT_EQ(kVideoCodecStereo, codec_specific_info.codecType);
>-  EXPECT_EQ(kStereoAssociatedCodecType,
>-            codec_specific_info.codecSpecific.stereo.associated_codec_type);
>-  EXPECT_EQ(0, codec_specific_info.codecSpecific.stereo.indices.frame_index);
>-  EXPECT_EQ(1, codec_specific_info.codecSpecific.stereo.indices.frame_count);
>-  EXPECT_EQ(0ull,
>-            codec_specific_info.codecSpecific.stereo.indices.picture_index);
>-
>-  EXPECT_EQ(
>-      WEBRTC_VIDEO_CODEC_OK,
>-      decoder_->Decode(encoded_frame, false, nullptr, &codec_specific_info));
>-  std::unique_ptr<VideoFrame> decoded_frame;
>-  rtc::Optional<uint8_t> decoded_qp;
>-  ASSERT_TRUE(WaitForDecodedFrame(&decoded_frame, &decoded_qp));
>-  ASSERT_TRUE(decoded_frame);
>-  EXPECT_GT(I420PSNR(input_frame_.get(), decoded_frame.get()), 36);
>-}
>-
>-TEST_F(TestStereoAdapter, EncodeDecodeI420AFrame) {
>-  std::unique_ptr<VideoFrame> yuva_frame = CreateI420AInputFrame();
>-  const size_t expected_num_encoded_frames = 2;
>-  SetWaitForEncodedFramesThreshold(expected_num_encoded_frames);
>-  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>-            encoder_->Encode(*yuva_frame, nullptr, nullptr));
>-  std::vector<EncodedImage> encoded_frames;
>-  std::vector<CodecSpecificInfo> codec_specific_infos;
>-  ASSERT_TRUE(WaitForEncodedFrames(&encoded_frames, &codec_specific_infos));
>-  EXPECT_EQ(expected_num_encoded_frames, encoded_frames.size());
>-  EXPECT_EQ(expected_num_encoded_frames, codec_specific_infos.size());
>-
>-  const CodecSpecificInfo& yuv_info = codec_specific_infos[kYUVStream];
>-  EXPECT_EQ(kVideoCodecStereo, yuv_info.codecType);
>-  EXPECT_EQ(kStereoAssociatedCodecType,
>-            yuv_info.codecSpecific.stereo.associated_codec_type);
>-  EXPECT_EQ(kYUVStream, yuv_info.codecSpecific.stereo.indices.frame_index);
>-  EXPECT_EQ(kAlphaCodecStreams,
>-            yuv_info.codecSpecific.stereo.indices.frame_count);
>-  EXPECT_EQ(0ull, yuv_info.codecSpecific.stereo.indices.picture_index);
>-
>-  const CodecSpecificInfo& axx_info = codec_specific_infos[kAXXStream];
>-  EXPECT_EQ(kVideoCodecStereo, axx_info.codecType);
>-  EXPECT_EQ(kStereoAssociatedCodecType,
>-            axx_info.codecSpecific.stereo.associated_codec_type);
>-  EXPECT_EQ(kAXXStream, axx_info.codecSpecific.stereo.indices.frame_index);
>-  EXPECT_EQ(kAlphaCodecStreams,
>-            axx_info.codecSpecific.stereo.indices.frame_count);
>-  EXPECT_EQ(0ull, axx_info.codecSpecific.stereo.indices.picture_index);
>-
>-  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, decoder_->Decode(encoded_frames[kYUVStream],
>-                                                    false, nullptr, &yuv_info));
>-  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, decoder_->Decode(encoded_frames[kAXXStream],
>-                                                    false, nullptr, &axx_info));
>-  std::unique_ptr<VideoFrame> decoded_frame;
>-  rtc::Optional<uint8_t> decoded_qp;
>-  ASSERT_TRUE(WaitForDecodedFrame(&decoded_frame, &decoded_qp));
>-  ASSERT_TRUE(decoded_frame);
>-  EXPECT_GT(I420PSNR(yuva_frame.get(), decoded_frame.get()), 36);
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/android_codec_factory_helper.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/android_codec_factory_helper.cc
>new file mode 100644
>index 00000000000..3ebc6b7ca46
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/android_codec_factory_helper.cc
>@@ -0,0 +1,84 @@
>+/*
>+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <pthread.h>
>+
>+#include <memory>
>+
>+#include "modules/video_coding/codecs/test/android_codec_factory_helper.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/ignore_wundef.h"
>+#include "sdk/android/native_api/base/init.h"
>+#include "sdk/android/native_api/codecs/wrapper.h"
>+#include "sdk/android/native_api/jni/class_loader.h"
>+#include "sdk/android/native_api/jni/jvm.h"
>+#include "sdk/android/native_api/jni/scoped_java_ref.h"
>+
>+// Note: this dependency is dangerous since it reaches into Chromium's base.
>+// There's a risk of e.g. macro clashes. This file may only be used in tests.
>+// Since we use Chrome's build system for creating the gtest binary, this should
>+// be fine.
>+RTC_PUSH_IGNORING_WUNDEF()
>+#include "base/android/jni_android.h"
>+RTC_POP_IGNORING_WUNDEF()
>+
>+namespace webrtc {
>+namespace test {
>+
>+namespace {
>+
>+static pthread_once_t g_initialize_once = PTHREAD_ONCE_INIT;
>+
>+// There can only be one JNI_OnLoad in each binary. So since this is a GTEST
>+// C++ runner binary, we want to initialize the same global objects we normally
>+// do if this had been a Java binary.
>+void EnsureInitializedOnce() {
>+  RTC_CHECK(::base::android::IsVMInitialized());
>+  JNIEnv* env = ::base::android::AttachCurrentThread();
>+  JavaVM* jvm = nullptr;
>+  RTC_CHECK_EQ(0, env->GetJavaVM(&jvm));
>+
>+  InitAndroid(jvm);
>+}
>+
>+}  // namespace
>+
>+void InitializeAndroidObjects() {
>+  RTC_CHECK_EQ(0, pthread_once(&g_initialize_once, &EnsureInitializedOnce));
>+}
>+
>+std::unique_ptr<VideoEncoderFactory> CreateAndroidEncoderFactory() {
>+  JNIEnv* env = AttachCurrentThreadIfNeeded();
>+  ScopedJavaLocalRef<jclass> factory_class =
>+      GetClass(env, "org/webrtc/HardwareVideoEncoderFactory");
>+  jmethodID factory_constructor = env->GetMethodID(
>+      factory_class.obj(), "<init>", "(Lorg/webrtc/EglBase$Context;ZZ)V");
>+  ScopedJavaLocalRef<jobject> factory_object(
>+      env, env->NewObject(factory_class.obj(), factory_constructor,
>+                          nullptr /* shared_context */,
>+                          false /* enable_intel_vp8_encoder */,
>+                          true /* enable_h264_high_profile */));
>+  return JavaToNativeVideoEncoderFactory(env, factory_object.obj());
>+}
>+
>+std::unique_ptr<VideoDecoderFactory> CreateAndroidDecoderFactory() {
>+  JNIEnv* env = AttachCurrentThreadIfNeeded();
>+  ScopedJavaLocalRef<jclass> factory_class =
>+      GetClass(env, "org/webrtc/HardwareVideoDecoderFactory");
>+  jmethodID factory_constructor = env->GetMethodID(
>+      factory_class.obj(), "<init>", "(Lorg/webrtc/EglBase$Context;)V");
>+  ScopedJavaLocalRef<jobject> factory_object(
>+      env, env->NewObject(factory_class.obj(), factory_constructor,
>+                          nullptr /* shared_context */));
>+  return JavaToNativeVideoDecoderFactory(env, factory_object.obj());
>+}
>+
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/android_codec_factory_helper.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/android_codec_factory_helper.h
>new file mode 100644
>index 00000000000..ad9cf35162e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/android_codec_factory_helper.h
>@@ -0,0 +1,30 @@
>+/*
>+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_VIDEO_CODING_CODECS_TEST_ANDROID_CODEC_FACTORY_HELPER_H_
>+#define MODULES_VIDEO_CODING_CODECS_TEST_ANDROID_CODEC_FACTORY_HELPER_H_
>+
>+#include <memory>
>+
>+#include "api/video_codecs/video_decoder_factory.h"
>+#include "api/video_codecs/video_encoder_factory.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+void InitializeAndroidObjects();
>+
>+std::unique_ptr<VideoEncoderFactory> CreateAndroidEncoderFactory();
>+std::unique_ptr<VideoDecoderFactory> CreateAndroidDecoderFactory();
>+
>+}  // namespace test
>+}  // namespace webrtc
>+
>+#endif  // MODULES_VIDEO_CODING_CODECS_TEST_ANDROID_CODEC_FACTORY_HELPER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/android_test_initializer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/android_test_initializer.cc
>deleted file mode 100644
>index 43cf22aceb4..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/android_test_initializer.cc
>+++ /dev/null
>@@ -1,55 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include <pthread.h>
>-
>-#include "modules/video_coding/codecs/test/android_test_initializer.h"
>-#include "rtc_base/ignore_wundef.h"
>-#include "sdk/android/src/jni/classreferenceholder.h"
>-#include "sdk/android/src/jni/jni_helpers.h"
>-
>-// Note: this dependency is dangerous since it reaches into Chromium's base.
>-// There's a risk of e.g. macro clashes. This file may only be used in tests.
>-// Since we use Chrome's build system for creating the gtest binary, this should
>-// be fine.
>-RTC_PUSH_IGNORING_WUNDEF()
>-#include "base/android/jni_android.h"
>-RTC_POP_IGNORING_WUNDEF()
>-
>-#include "rtc_base/checks.h"
>-
>-namespace webrtc {
>-
>-namespace {
>-
>-static pthread_once_t g_initialize_once = PTHREAD_ONCE_INIT;
>-
>-// There can only be one JNI_OnLoad in each binary. So since this is a GTEST
>-// C++ runner binary, we want to initialize the same global objects we normally
>-// do if this had been a Java binary.
>-void EnsureInitializedOnce() {
>-  RTC_CHECK(::base::android::IsVMInitialized());
>-  JNIEnv* jni = ::base::android::AttachCurrentThread();
>-  JavaVM* jvm = NULL;
>-  RTC_CHECK_EQ(0, jni->GetJavaVM(&jvm));
>-
>-  jint ret = jni::InitGlobalJniVariables(jvm);
>-  RTC_DCHECK_GE(ret, 0);
>-
>-  jni::LoadGlobalClassReferenceHolder();
>-}
>-
>-}  // namespace
>-
>-void InitializeAndroidObjects() {
>-  RTC_CHECK_EQ(0, pthread_once(&g_initialize_once, &EnsureInitializedOnce));
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/batch/empty-runtime-deps b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/batch/empty-runtime-deps
>new file mode 100644
>index 00000000000..6702195ca94
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/batch/empty-runtime-deps
>@@ -0,0 +1 @@
>+does-not-exist
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/batch/run-instantiation-tests.sh b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/batch/run-instantiation-tests.sh
>new file mode 100755
>index 00000000000..28083b1808f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/batch/run-instantiation-tests.sh
>@@ -0,0 +1,56 @@
>+#!/bin/bash
>+
>+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+#
>+# Use of this source code is governed by a BSD-style license
>+# that can be found in the LICENSE file in the root of the source
>+# tree. An additional intellectual property rights grant can be found
>+# in the file PATENTS.  All contributing project authors may
>+# be found in the AUTHORS file in the root of the source tree.
>+
>+if [ $# -ne 1 ]; then
>+    echo "Usage: run-instantiation-tests.sh ADB-DEVICE-ID"
>+    exit 1
>+fi
>+
>+# Paths: update these based on your git checkout and gn output folder names.
>+WEBRTC_DIR=$HOME/src/webrtc/src
>+BUILD_DIR=$WEBRTC_DIR/out/Android_Release
>+
>+# Other settings.
>+ADB=`which adb`
>+SERIAL=$1
>+TIMEOUT=7200
>+
>+# Ensure we are using the latest version.
>+ninja -C $BUILD_DIR modules_tests
>+
>+# Transfer the required files by trying to run a test that doesn't exist.
>+echo "===> Transferring required resources to device $1."
>+$WEBRTC_DIR/build/android/test_runner.py gtest \
>+  --output-directory $BUILD_DIR \
>+  --suite modules_tests \
>+  --gtest_filter "DoesNotExist" \
>+  --shard-timeout $TIMEOUT \
>+  --runtime-deps-path $BUILD_DIR/gen.runtime/modules/modules_tests__test_runner_script.runtime_deps \
>+  --adb-path $ADB \
>+  --device $SERIAL \
>+  --verbose
>+
>+# Run all tests as separate test invocations.
>+mkdir $SERIAL
>+pushd $SERIAL
>+$WEBRTC_DIR/build/android/test_runner.py gtest \
>+  --output-directory $BUILD_DIR \
>+  --suite modules_tests \
>+  --gtest_filter "*InstantiationTest*" \
>+  --gtest_also_run_disabled_tests \
>+  --shard-timeout $TIMEOUT \
>+  --runtime-deps-path ../empty-runtime-deps \
>+  --test-launcher-retry-limit 0 \
>+  --adb-path $ADB \
>+  --device $SERIAL \
>+  --verbose \
>+  --num-retries 0 \
>+  2>&1 | tee -a instantiation-tests.log
>+popd
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/batch/run-videoprocessor-tests.sh b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/batch/run-videoprocessor-tests.sh
>new file mode 100755
>index 00000000000..25c971ba61c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/batch/run-videoprocessor-tests.sh
>@@ -0,0 +1,70 @@
>+#!/bin/bash
>+
>+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+#
>+# Use of this source code is governed by a BSD-style license
>+# that can be found in the LICENSE file in the root of the source
>+# tree. An additional intellectual property rights grant can be found
>+# in the file PATENTS.  All contributing project authors may
>+# be found in the AUTHORS file in the root of the source tree.
>+
>+if [ $# -ne 1 ]; then
>+    echo "Usage: run.sh ADB-DEVICE-ID"
>+    exit 1
>+fi
>+
>+# Paths: update these based on your git checkout and gn output folder names.
>+WEBRTC_DIR=$HOME/src/webrtc/src
>+BUILD_DIR=$WEBRTC_DIR/out/Android_Release
>+
>+# Clips: update these to encode/decode other content.
>+CLIPS=('Foreman')
>+RESOLUTIONS=('128x96' '160x120' '176x144' '320x240' '352x288')
>+FRAMERATES=(30)
>+
>+# Other settings.
>+ADB=`which adb`
>+SERIAL=$1
>+TIMEOUT=7200
>+
>+# Ensure we are using the latest version.
>+ninja -C $BUILD_DIR modules_tests
>+
>+# Transfer the required files by trying to run a test that doesn't exist.
>+echo "===> Transferring required resources to device $1."
>+$WEBRTC_DIR/build/android/test_runner.py gtest \
>+  --output-directory $BUILD_DIR \
>+  --suite modules_tests \
>+  --gtest_filter "DoesNotExist" \
>+  --shard-timeout $TIMEOUT \
>+  --runtime-deps-path $BUILD_DIR/gen.runtime/modules/modules_tests__test_runner_script.runtime_deps \
>+  --adb-path $ADB \
>+  --device $SERIAL \
>+  --verbose
>+
>+# Run all tests as separate test invocations.
>+mkdir $SERIAL
>+pushd $SERIAL
>+for clip in "${CLIPS[@]}"; do
>+  for resolution in "${RESOLUTIONS[@]}"; do
>+    for framerate in "${FRAMERATES[@]}"; do
>+      test_name="${clip}_${resolution}_${framerate}"
>+      log_name="${test_name}.log"
>+
>+      echo "===> Running ${test_name} on device $1."
>+
>+      $WEBRTC_DIR/build/android/test_runner.py gtest \
>+        --output-directory $BUILD_DIR \
>+        --suite modules_tests \
>+        --gtest_filter "CodecSettings/*${test_name}*" \
>+        --shard-timeout $TIMEOUT \
>+        --runtime-deps-path ../empty-runtime-deps \
>+        --test-launcher-retry-limit 0 \
>+        --adb-path $ADB \
>+        --device $SERIAL \
>+        --verbose \
>+        2>&1 | tee -a ${log_name}
>+    done
>+  done
>+done
>+popd
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/mock/mock_packet_manipulator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/mock/mock_packet_manipulator.h
>deleted file mode 100644
>index ed7605550b0..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/mock/mock_packet_manipulator.h
>+++ /dev/null
>@@ -1,32 +0,0 @@
>-/*
>- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MODULES_VIDEO_CODING_CODECS_TEST_MOCK_MOCK_PACKET_MANIPULATOR_H_
>-#define MODULES_VIDEO_CODING_CODECS_TEST_MOCK_MOCK_PACKET_MANIPULATOR_H_
>-
>-#include <string>
>-
>-#include "modules/video_coding/codecs/test/packet_manipulator.h"
>-#include "test/gmock.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>-#include "common_video/include/video_frame.h"
>-
>-namespace webrtc {
>-namespace test {
>-
>-class MockPacketManipulator : public PacketManipulator {
>- public:
>-  MOCK_METHOD1(ManipulatePackets, int(webrtc::EncodedImage* encoded_image));
>-};
>-
>-}  // namespace test
>-}  // namespace webrtc
>-
>-#endif  // MODULES_VIDEO_CODING_CODECS_TEST_MOCK_MOCK_PACKET_MANIPULATOR_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/objc_codec_h264_test.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/objc_codec_factory_helper.h
>similarity index 73%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/objc_codec_h264_test.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/objc_codec_factory_helper.h
>index 0b18f8eb938..475d0fdd08d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/objc_codec_h264_test.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/objc_codec_factory_helper.h
>@@ -8,8 +8,8 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#ifndef MODULES_VIDEO_CODING_CODECS_TEST_OBJC_CODEC_H264_TEST_H_
>-#define MODULES_VIDEO_CODING_CODECS_TEST_OBJC_CODEC_H264_TEST_H_
>+#ifndef MODULES_VIDEO_CODING_CODECS_TEST_OBJC_CODEC_FACTORY_HELPER_H_
>+#define MODULES_VIDEO_CODING_CODECS_TEST_OBJC_CODEC_FACTORY_HELPER_H_
> 
> #include <memory>
> 
>@@ -17,10 +17,12 @@
> #include "api/video_codecs/video_encoder_factory.h"
> 
> namespace webrtc {
>+namespace test {
> 
> std::unique_ptr<VideoEncoderFactory> CreateObjCEncoderFactory();
> std::unique_ptr<VideoDecoderFactory> CreateObjCDecoderFactory();
> 
>+}  // namespace test
> }  // namespace webrtc
> 
>-#endif  // MODULES_VIDEO_CODING_CODECS_TEST_OBJC_CODEC_H264_TEST_H_
>+#endif  // MODULES_VIDEO_CODING_CODECS_TEST_OBJC_CODEC_FACTORY_HELPER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/objc_codec_h264_test.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/objc_codec_factory_helper.mm
>similarity index 58%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/objc_codec_h264_test.mm
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/objc_codec_factory_helper.mm
>index 74ba84a677d..6fb14f346e0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/objc_codec_h264_test.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/objc_codec_factory_helper.mm
>@@ -8,21 +8,22 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "modules/video_coding/codecs/test/objc_codec_h264_test.h"
>+#include "modules/video_coding/codecs/test/objc_codec_factory_helper.h"
> 
> #import "WebRTC/RTCVideoCodecH264.h"
>-#include "rtc_base/ptr_util.h"
>-#include "sdk/objc/Framework/Classes/VideoToolbox/objc_video_decoder_factory.h"
>-#include "sdk/objc/Framework/Classes/VideoToolbox/objc_video_encoder_factory.h"
>+#include "sdk/objc/Framework/Native/api/video_decoder_factory.h"
>+#include "sdk/objc/Framework/Native/api/video_encoder_factory.h"
> 
> namespace webrtc {
>+namespace test {
> 
> std::unique_ptr<VideoEncoderFactory> CreateObjCEncoderFactory() {
>-  return rtc::MakeUnique<ObjCVideoEncoderFactory>([[RTCVideoEncoderFactoryH264 alloc] init]);
>+  return ObjCToNativeVideoEncoderFactory([[RTCVideoEncoderFactoryH264 alloc] init]);
> }
> 
> std::unique_ptr<VideoDecoderFactory> CreateObjCDecoderFactory() {
>-  return rtc::MakeUnique<ObjCVideoDecoderFactory>([[RTCVideoDecoderFactoryH264 alloc] init]);
>+  return ObjCToNativeVideoDecoderFactory([[RTCVideoDecoderFactoryH264 alloc] init]);
> }
> 
>+}  // namespace test
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/packet_manipulator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/packet_manipulator.cc
>deleted file mode 100644
>index 779faf68c43..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/packet_manipulator.cc
>+++ /dev/null
>@@ -1,95 +0,0 @@
>-/*
>- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "modules/video_coding/codecs/test/packet_manipulator.h"
>-
>-#include <stdio.h>
>-
>-#include "rtc_base/checks.h"
>-#include "rtc_base/format_macros.h"
>-
>-namespace webrtc {
>-namespace test {
>-
>-PacketManipulatorImpl::PacketManipulatorImpl(PacketReader* packet_reader,
>-                                             const NetworkingConfig& config,
>-                                             bool verbose)
>-    : packet_reader_(packet_reader),
>-      config_(config),
>-      verbose_(verbose),
>-      active_burst_packets_(0),
>-      random_seed_(1) {
>-  RTC_DCHECK(packet_reader);
>-}
>-
>-int PacketManipulatorImpl::ManipulatePackets(
>-    webrtc::EncodedImage* encoded_image) {
>-  int nbr_packets_dropped = 0;
>-  // There's no need to build a copy of the image data since viewing an
>-  // EncodedImage object, setting the length to a new lower value represents
>-  // that everything is dropped after that position in the byte array.
>-  // EncodedImage._size is the allocated bytes.
>-  // EncodedImage._length is how many that are filled with data.
>-  int new_length = 0;
>-  packet_reader_->InitializeReading(encoded_image->_buffer,
>-                                    encoded_image->_length,
>-                                    config_.packet_size_in_bytes);
>-  uint8_t* packet = nullptr;
>-  int nbr_bytes_to_read;
>-  // keep track of if we've lost any packets, since then we shall loose
>-  // the remains of the current frame:
>-  bool packet_loss_has_occurred = false;
>-  while ((nbr_bytes_to_read = packet_reader_->NextPacket(&packet)) > 0) {
>-    // Check if we're currently in a packet loss burst that is not completed:
>-    if (active_burst_packets_ > 0) {
>-      active_burst_packets_--;
>-      nbr_packets_dropped++;
>-    } else if (RandomUniform() < config_.packet_loss_probability ||
>-               packet_loss_has_occurred) {
>-      packet_loss_has_occurred = true;
>-      nbr_packets_dropped++;
>-      if (config_.packet_loss_mode == kBurst) {
>-        // Initiate a new burst
>-        active_burst_packets_ = config_.packet_loss_burst_length - 1;
>-      }
>-    } else {
>-      new_length += nbr_bytes_to_read;
>-    }
>-  }
>-  encoded_image->_length = new_length;
>-  if (nbr_packets_dropped > 0) {
>-    // Must set completeFrame to false to inform the decoder about this:
>-    encoded_image->_completeFrame = false;
>-    if (verbose_) {
>-      printf("Dropped %d packets for frame %d (frame length: %" PRIuS ")\n",
>-             nbr_packets_dropped, encoded_image->_timeStamp,
>-             encoded_image->_length);
>-    }
>-  }
>-  return nbr_packets_dropped;
>-}
>-
>-void PacketManipulatorImpl::InitializeRandomSeed(unsigned int seed) {
>-  random_seed_ = seed;
>-}
>-
>-inline double PacketManipulatorImpl::RandomUniform() {
>-  // Use the previous result as new seed before each rand() call. Doing this
>-  // it doesn't matter if other threads are calling rand() since we'll always
>-  // get the same behavior as long as we're using a fixed initial seed.
>-  critsect_.Enter();
>-  srand(random_seed_);
>-  random_seed_ = rand();  // NOLINT (rand_r instead of rand)
>-  critsect_.Leave();
>-  return (random_seed_ + 1.0) / (RAND_MAX + 1.0);
>-}
>-
>-}  // namespace test
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/packet_manipulator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/packet_manipulator.h
>deleted file mode 100644
>index 5a2c9cc56af..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/packet_manipulator.h
>+++ /dev/null
>@@ -1,112 +0,0 @@
>-/*
>- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MODULES_VIDEO_CODING_CODECS_TEST_PACKET_MANIPULATOR_H_
>-#define MODULES_VIDEO_CODING_CODECS_TEST_PACKET_MANIPULATOR_H_
>-
>-#include <stdlib.h>
>-
>-#include "modules/video_coding/include/video_codec_interface.h"
>-#include "rtc_base/criticalsection.h"
>-#include "test/testsupport/packet_reader.h"
>-
>-namespace webrtc {
>-namespace test {
>-
>-// Which mode the packet loss shall be performed according to.
>-enum PacketLossMode {
>-  // Drops packets with a configured probability independently for each packet
>-  kUniform,
>-  // Drops packets similar to uniform but when a packet is being dropped, the
>-  // number of lost packets in a row is equal to the configured burst length.
>-  kBurst
>-};
>-
>-// Contains configurations related to networking and simulation of
>-// scenarios caused by network interference.
>-struct NetworkingConfig {
>-  NetworkingConfig()
>-      : packet_size_in_bytes(1500),
>-        max_payload_size_in_bytes(1440),
>-        packet_loss_mode(kUniform),
>-        packet_loss_probability(0.0),
>-        packet_loss_burst_length(1) {}
>-
>-  // Packet size in bytes. Default: 1500 bytes.
>-  size_t packet_size_in_bytes;
>-
>-  // Encoder specific setting of maximum size in bytes of each payload.
>-  // Default: 1440 bytes.
>-  size_t max_payload_size_in_bytes;
>-
>-  // Packet loss mode. Two different packet loss models are supported:
>-  // uniform or burst. This setting has no effect unless
>-  // packet_loss_probability is >0.
>-  // Default: uniform.
>-  PacketLossMode packet_loss_mode;
>-
>-  // Packet loss probability. A value between 0.0 and 1.0 that defines the
>-  // probability of a packet being lost. 0.1 means 10% and so on.
>-  // Default: 0 (no loss).
>-  double packet_loss_probability;
>-
>-  // Packet loss burst length. Defines how many packets will be lost in a burst
>-  // when a packet has been decided to be lost. Must be >=1. Default: 1.
>-  int packet_loss_burst_length;
>-};
>-
>-// Class for simulating packet loss on the encoded frame data.
>-// When a packet loss has occurred in a frame, the remaining data in that
>-// frame is lost (even if burst length is only a single packet).
>-// TODO(kjellander): Support discarding only individual packets in the frame
>-// when CL 172001 has been submitted. This also requires a correct
>-// fragmentation header to be passed to the decoder.
>-//
>-// To get a repeatable packet drop pattern, re-initialize the random seed
>-// using InitializeRandomSeed before each test run.
>-class PacketManipulator {
>- public:
>-  virtual ~PacketManipulator() {}
>-
>-  // Manipulates the data of the encoded_image to simulate parts being lost
>-  // during transport.
>-  // If packets are dropped from frame data, the completedFrame field will be
>-  // set to false.
>-  // Returns the number of packets being dropped.
>-  virtual int ManipulatePackets(webrtc::EncodedImage* encoded_image) = 0;
>-};
>-
>-class PacketManipulatorImpl : public PacketManipulator {
>- public:
>-  PacketManipulatorImpl(PacketReader* packet_reader,
>-                        const NetworkingConfig& config,
>-                        bool verbose);
>-  ~PacketManipulatorImpl() = default;
>-  int ManipulatePackets(webrtc::EncodedImage* encoded_image) override;
>-  virtual void InitializeRandomSeed(unsigned int seed);
>-
>- protected:
>-  // Returns a uniformly distributed random value between 0.0 and 1.0
>-  virtual double RandomUniform();
>-
>- private:
>-  PacketReader* const packet_reader_;
>-  const NetworkingConfig& config_;
>-  const bool verbose_;
>-  // Used to simulate a burst over several frames.
>-  int active_burst_packets_;
>-  rtc::CriticalSection critsect_;
>-  unsigned int random_seed_;
>-};
>-
>-}  // namespace test
>-}  // namespace webrtc
>-
>-#endif  // MODULES_VIDEO_CODING_CODECS_TEST_PACKET_MANIPULATOR_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/packet_manipulator_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/packet_manipulator_unittest.cc
>deleted file mode 100644
>index 3a226d64b62..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/packet_manipulator_unittest.cc
>+++ /dev/null
>@@ -1,96 +0,0 @@
>-/*
>- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "modules/video_coding/codecs/test/packet_manipulator.h"
>-
>-#include <queue>
>-
>-#include "modules/video_coding/include/video_codec_interface.h"
>-#include "test/gtest.h"
>-#include "test/testsupport/unittest_utils.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>-
>-namespace webrtc {
>-namespace test {
>-
>-const double kNeverDropProbability = 0.0;
>-const double kAlwaysDropProbability = 1.0;
>-const int kBurstLength = 1;
>-
>-class PacketManipulatorTest : public PacketRelatedTest {
>- protected:
>-  PacketReader packet_reader_;
>-  EncodedImage image_;
>-  NetworkingConfig drop_config_;
>-  NetworkingConfig no_drop_config_;
>-
>-  PacketManipulatorTest() {
>-    image_._buffer = packet_data_;
>-    image_._length = kPacketDataLength;
>-    image_._size = kPacketDataLength;
>-
>-    drop_config_.packet_size_in_bytes = kPacketSizeInBytes;
>-    drop_config_.packet_loss_probability = kAlwaysDropProbability;
>-    drop_config_.packet_loss_burst_length = kBurstLength;
>-    drop_config_.packet_loss_mode = kUniform;
>-
>-    no_drop_config_.packet_size_in_bytes = kPacketSizeInBytes;
>-    no_drop_config_.packet_loss_probability = kNeverDropProbability;
>-    no_drop_config_.packet_loss_burst_length = kBurstLength;
>-    no_drop_config_.packet_loss_mode = kUniform;
>-  }
>-
>-  virtual ~PacketManipulatorTest() {}
>-
>-  void SetUp() { PacketRelatedTest::SetUp(); }
>-
>-  void TearDown() { PacketRelatedTest::TearDown(); }
>-
>-  void VerifyPacketLoss(int expected_nbr_packets_dropped,
>-                        int actual_nbr_packets_dropped,
>-                        size_t expected_packet_data_length,
>-                        uint8_t* expected_packet_data,
>-                        const EncodedImage& actual_image) {
>-    EXPECT_EQ(expected_nbr_packets_dropped, actual_nbr_packets_dropped);
>-    EXPECT_EQ(expected_packet_data_length, image_._length);
>-    EXPECT_EQ(0, memcmp(expected_packet_data, actual_image._buffer,
>-                        expected_packet_data_length));
>-  }
>-};
>-
>-TEST_F(PacketManipulatorTest, Constructor) {
>-  PacketManipulatorImpl manipulator(&packet_reader_, no_drop_config_, false);
>-}
>-
>-TEST_F(PacketManipulatorTest, DropNone) {
>-  PacketManipulatorImpl manipulator(&packet_reader_, no_drop_config_, false);
>-  int nbr_packets_dropped = manipulator.ManipulatePackets(&image_);
>-  VerifyPacketLoss(0, nbr_packets_dropped, kPacketDataLength, packet_data_,
>-                   image_);
>-}
>-
>-TEST_F(PacketManipulatorTest, UniformDropNoneSmallFrame) {
>-  size_t data_length = 400;  // smaller than the packet size
>-  image_._length = data_length;
>-  PacketManipulatorImpl manipulator(&packet_reader_, no_drop_config_, false);
>-  int nbr_packets_dropped = manipulator.ManipulatePackets(&image_);
>-
>-  VerifyPacketLoss(0, nbr_packets_dropped, data_length, packet_data_, image_);
>-}
>-
>-TEST_F(PacketManipulatorTest, UniformDropAll) {
>-  PacketManipulatorImpl manipulator(&packet_reader_, drop_config_, false);
>-  int nbr_packets_dropped = manipulator.ManipulatePackets(&image_);
>-  VerifyPacketLoss(kPacketDataNumberOfPackets, nbr_packets_dropped, 0,
>-                   packet_data_, image_);
>-}
>-
>-}  // namespace test
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/plot_webrtc_test_logs.py b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/plot_webrtc_test_logs.py
>index ae5621f44a3..a0cdc0637fd 100755
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/plot_webrtc_test_logs.py
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/plot_webrtc_test_logs.py
>@@ -18,42 +18,37 @@ import re
> import matplotlib.pyplot as plt
> 
> # Log events.
>-EVENT_START = \
>-  'RUN      ] CodecSettings/VideoProcessorIntegrationTestParameterized.'
>-EVENT_END = 'OK ] CodecSettings/VideoProcessorIntegrationTestParameterized.'
>+EVENT_START = 'RUN      ] CodecSettings/VideoCodecTestParameterized.'
>+EVENT_END = 'OK ] CodecSettings/VideoCodecTestParameterized.'
> 
> # Metrics to plot, tuple: (name to parse in file, label to use when plotting).
>-BITRATE = ('Target bitrate', 'target bitrate (kbps)')
>-WIDTH = ('Width', 'width')
>-HEIGHT = ('Height', 'height')
>-FILENAME = ('Filename', 'clip')
>-CODEC_TYPE = ('Codec type', 'Codec')
>-ENCODER_IMPLEMENTATION_NAME = ('Encoder implementation name', 'enc name')
>-DECODER_IMPLEMENTATION_NAME = ('Decoder implementation name', 'dec name')
>-CODEC_IMPLEMENTATION_NAME = ('Codec implementation name', 'codec name')
>-CORES = ('# CPU cores used', 'CPU cores used')
>-DENOISING = ('Denoising', 'denoising')
>-RESILIENCE = ('Resilience', 'resilience')
>-ERROR_CONCEALMENT = ('Error concealment', 'error concealment')
>-QP = ('Average QP', 'avg QP')
>-CPU_USAGE = ('CPU usage %', 'CPU usage (%)')
>-PSNR = ('PSNR avg', 'PSNR (dB)')
>-SSIM = ('SSIM avg', 'SSIM')
>-ENC_BITRATE = ('Encoded bitrate', 'encoded bitrate (kbps)')
>-FRAMERATE = ('Frame rate', 'fps')
>-NUM_FRAMES = ('# processed frames', 'num frames')
>-NUM_DROPPED_FRAMES = ('# dropped frames', 'num dropped frames')
>-NUM_FRAMES_TO_TARGET = ('# frames to convergence',
>-                        'frames to reach target rate')
>-ENCODE_TIME = ('Encoding time', 'encode time (us)')
>-ENCODE_TIME_AVG = ('Encoding time', 'encode time (us) avg')
>-DECODE_TIME = ('Decoding time', 'decode time (us)')
>-DECODE_TIME_AVG = ('Decoding time', 'decode time (us) avg')
>-FRAME_SIZE = ('Frame sizes', 'frame size (bytes)')
>-FRAME_SIZE_AVG = ('Frame sizes', 'frame size (bytes) avg')
>-AVG_KEY_FRAME_SIZE = ('Average key frame size', 'avg key frame size (bytes)')
>-AVG_NON_KEY_FRAME_SIZE = ('Average non-key frame size',
>-                          'avg non-key frame size (bytes)')
>+WIDTH = ('width', 'width')
>+HEIGHT = ('height', 'height')
>+FILENAME = ('filename', 'clip')
>+CODEC_TYPE = ('codec_type', 'Codec')
>+ENCODER_IMPLEMENTATION_NAME = ('enc_impl_name', 'enc name')
>+DECODER_IMPLEMENTATION_NAME = ('dec_impl_name', 'dec name')
>+CODEC_IMPLEMENTATION_NAME = ('codec_impl_name', 'codec name')
>+CORES = ('num_cores', 'CPU cores used')
>+DENOISING = ('denoising', 'denoising')
>+RESILIENCE = ('resilience', 'resilience')
>+ERROR_CONCEALMENT = ('error_concealment', 'error concealment')
>+CPU_USAGE = ('cpu_usage_percent', 'CPU usage (%)')
>+BITRATE = ('target_bitrate_kbps', 'target bitrate (kbps)')
>+FRAMERATE = ('input_framerate_fps', 'fps')
>+QP = ('avg_qp', 'QP avg')
>+PSNR = ('avg_psnr', 'PSNR (dB)')
>+SSIM = ('avg_ssim', 'SSIM')
>+ENC_BITRATE = ('bitrate_kbps', 'encoded bitrate (kbps)')
>+NUM_FRAMES = ('num_input_frames', 'num frames')
>+NUM_DROPPED_FRAMES = ('num_dropped_frames', 'num dropped frames')
>+TIME_TO_TARGET = ('time_to_reach_target_bitrate_sec',
>+                  'time to reach target rate (sec)')
>+ENCODE_SPEED_FPS = ('enc_speed_fps', 'encode speed (fps)')
>+DECODE_SPEED_FPS = ('dec_speed_fps', 'decode speed (fps)')
>+AVG_KEY_FRAME_SIZE = ('avg_key_frame_size_bytes', 'avg key frame size (bytes)')
>+AVG_DELTA_FRAME_SIZE = ('avg_delta_frame_size_bytes',
>+                        'avg delta frame size (bytes)')
> 
> # Settings.
> SETTINGS = [
>@@ -61,9 +56,6 @@ SETTINGS = [
>   HEIGHT,
>   FILENAME,
>   NUM_FRAMES,
>-  ENCODE_TIME,
>-  DECODE_TIME,
>-  FRAME_SIZE,
> ]
> 
> # Settings, options for x-axis.
>@@ -90,13 +82,13 @@ RESULTS = [
>   SSIM,
>   ENC_BITRATE,
>   NUM_DROPPED_FRAMES,
>-  NUM_FRAMES_TO_TARGET,
>-  ENCODE_TIME_AVG,
>-  DECODE_TIME_AVG,
>+  TIME_TO_TARGET,
>+  ENCODE_SPEED_FPS,
>+  DECODE_SPEED_FPS,
>   QP,
>   CPU_USAGE,
>   AVG_KEY_FRAME_SIZE,
>-  AVG_NON_KEY_FRAME_SIZE,
>+  AVG_DELTA_FRAME_SIZE,
> ]
> 
> METRICS_TO_PARSE = SETTINGS + SUBPLOT_SETTINGS + RESULTS
>@@ -141,7 +133,7 @@ def ParseSetting(filename, setting):
>               settings.append(s)
>           break
> 
>-        TryFindMetric(parsed, line, settings_file)
>+        TryFindMetric(parsed, line)
> 
>   settings_file.close()
>   return settings
>@@ -214,36 +206,20 @@ def ParseMetrics(filename, setting1, setting2):
> 
>           break
> 
>-        TryFindMetric(parsed, line, settings_file)
>+        TryFindMetric(parsed, line)
> 
>   settings_file.close()
>   return metrics
> 
> 
>-def TryFindMetric(parsed, line, settings_file):
>+def TryFindMetric(parsed, line):
>   for metric in METRICS_TO_PARSE:
>     name = metric[0]
>     label = metric[1]
>     if re.search(r'%s' % name, line):
>       found, value = GetMetric(name, line)
>-      if not found:
>-        # TODO(asapersson): Change format.
>-        # Try find min, max, average stats.
>-        found, minimum = GetMetric("Min", settings_file.readline())
>-        if not found:
>-          return
>-        found, maximum = GetMetric("Max", settings_file.readline())
>-        if not found:
>-          return
>-        found, average = GetMetric("Average", settings_file.readline())
>-        if not found:
>-          return
>-
>-        parsed[label + ' min'] = minimum
>-        parsed[label + ' max'] = maximum
>-        parsed[label + ' avg'] = average
>-
>-      parsed[label] = value
>+      if found:
>+        parsed[label] = value
>       return
> 
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/stats.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/stats.cc
>deleted file mode 100644
>index a3199ab6fac..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/stats.cc
>+++ /dev/null
>@@ -1,222 +0,0 @@
>-/*
>- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "modules/video_coding/codecs/test/stats.h"
>-
>-#include <stdio.h>
>-
>-#include <algorithm>
>-
>-#include "rtc_base/checks.h"
>-#include "rtc_base/format_macros.h"
>-
>-namespace webrtc {
>-namespace test {
>-
>-namespace {
>-
>-bool LessForEncodeTime(const FrameStatistic& s1, const FrameStatistic& s2) {
>-  RTC_DCHECK_NE(s1.frame_number, s2.frame_number);
>-  return s1.encode_time_us < s2.encode_time_us;
>-}
>-
>-bool LessForDecodeTime(const FrameStatistic& s1, const FrameStatistic& s2) {
>-  RTC_DCHECK_NE(s1.frame_number, s2.frame_number);
>-  return s1.decode_time_us < s2.decode_time_us;
>-}
>-
>-bool LessForEncodedSize(const FrameStatistic& s1, const FrameStatistic& s2) {
>-  RTC_DCHECK_NE(s1.frame_number, s2.frame_number);
>-  return s1.encoded_frame_size_bytes < s2.encoded_frame_size_bytes;
>-}
>-
>-bool LessForBitRate(const FrameStatistic& s1, const FrameStatistic& s2) {
>-  RTC_DCHECK_NE(s1.frame_number, s2.frame_number);
>-  return s1.bitrate_kbps < s2.bitrate_kbps;
>-}
>-
>-bool LessForPsnr(const FrameStatistic& s1, const FrameStatistic& s2) {
>-  RTC_DCHECK_NE(s1.frame_number, s2.frame_number);
>-  return s1.psnr < s2.psnr;
>-}
>-
>-bool LessForSsim(const FrameStatistic& s1, const FrameStatistic& s2) {
>-  RTC_DCHECK_NE(s1.frame_number, s2.frame_number);
>-  return s1.ssim < s2.ssim;
>-}
>-
>-}  // namespace
>-
>-FrameStatistic* Stats::AddFrame() {
>-  // We don't expect more frames than what can be stored in an int.
>-  stats_.emplace_back(static_cast<int>(stats_.size()));
>-  return &stats_.back();
>-}
>-
>-FrameStatistic* Stats::GetFrame(int frame_number) {
>-  RTC_CHECK_GE(frame_number, 0);
>-  RTC_CHECK_LT(frame_number, stats_.size());
>-  return &stats_[frame_number];
>-}
>-
>-size_t Stats::size() const {
>-  return stats_.size();
>-}
>-
>-void Stats::PrintSummary() const {
>-  if (stats_.empty()) {
>-    printf("No frame statistics have been logged yet.\n");
>-    return;
>-  }
>-
>-  printf("Encode/decode statistics\n==\n");
>-
>-  // Calculate min, max, average and total encoding time.
>-  int total_encoding_time_us = 0;
>-  int total_decoding_time_us = 0;
>-  size_t total_encoded_frame_size_bytes = 0;
>-  size_t total_encoded_key_frame_size_bytes = 0;
>-  size_t total_encoded_delta_frame_size_bytes = 0;
>-  size_t num_key_frames = 0;
>-  size_t num_delta_frames = 0;
>-  int num_encode_failures = 0;
>-  double total_psnr = 0.0;
>-  double total_ssim = 0.0;
>-
>-  for (const FrameStatistic& stat : stats_) {
>-    total_encoding_time_us += stat.encode_time_us;
>-    total_decoding_time_us += stat.decode_time_us;
>-    total_encoded_frame_size_bytes += stat.encoded_frame_size_bytes;
>-    if (stat.frame_type == webrtc::kVideoFrameKey) {
>-      total_encoded_key_frame_size_bytes += stat.encoded_frame_size_bytes;
>-      ++num_key_frames;
>-    } else {
>-      total_encoded_delta_frame_size_bytes += stat.encoded_frame_size_bytes;
>-      ++num_delta_frames;
>-    }
>-    if (stat.encode_return_code != 0) {
>-      ++num_encode_failures;
>-    }
>-    if (stat.decoding_successful) {
>-      total_psnr += stat.psnr;
>-      total_ssim += stat.ssim;
>-    }
>-  }
>-
>-  // Encoding stats.
>-  printf("# Encoded frame failures: %d\n", num_encode_failures);
>-  printf("Encoding time:\n");
>-  auto frame_it =
>-      std::min_element(stats_.begin(), stats_.end(), LessForEncodeTime);
>-  printf("  Min     : %7d us (frame %d)\n", frame_it->encode_time_us,
>-         frame_it->frame_number);
>-  frame_it = std::max_element(stats_.begin(), stats_.end(), LessForEncodeTime);
>-  printf("  Max     : %7d us (frame %d)\n", frame_it->encode_time_us,
>-         frame_it->frame_number);
>-  printf("  Average : %7d us\n",
>-         static_cast<int>(total_encoding_time_us / stats_.size()));
>-
>-  // Decoding stats.
>-  printf("Decoding time:\n");
>-  // Only consider successfully decoded frames (packet loss may cause failures).
>-  std::vector<FrameStatistic> decoded_frames;
>-  for (const FrameStatistic& stat : stats_) {
>-    if (stat.decoding_successful) {
>-      decoded_frames.push_back(stat);
>-    }
>-  }
>-  if (decoded_frames.empty()) {
>-    printf("No successfully decoded frames exist in this statistics.\n");
>-  } else {
>-    frame_it = std::min_element(decoded_frames.begin(), decoded_frames.end(),
>-                                LessForDecodeTime);
>-    printf("  Min     : %7d us (frame %d)\n", frame_it->decode_time_us,
>-           frame_it->frame_number);
>-    frame_it = std::max_element(decoded_frames.begin(), decoded_frames.end(),
>-                                LessForDecodeTime);
>-    printf("  Max     : %7d us (frame %d)\n", frame_it->decode_time_us,
>-           frame_it->frame_number);
>-    printf("  Average : %7d us\n",
>-           static_cast<int>(total_decoding_time_us / decoded_frames.size()));
>-    printf("  Failures: %d frames failed to decode.\n",
>-           static_cast<int>(stats_.size() - decoded_frames.size()));
>-  }
>-
>-  // Frame size stats.
>-  printf("Frame sizes:\n");
>-  frame_it = std::min_element(stats_.begin(), stats_.end(), LessForEncodedSize);
>-  printf("  Min     : %7" PRIuS " bytes (frame %d)\n",
>-         frame_it->encoded_frame_size_bytes, frame_it->frame_number);
>-  frame_it = std::max_element(stats_.begin(), stats_.end(), LessForEncodedSize);
>-  printf("  Max     : %7" PRIuS " bytes (frame %d)\n",
>-         frame_it->encoded_frame_size_bytes, frame_it->frame_number);
>-  printf("  Average : %7" PRIuS " bytes\n",
>-         total_encoded_frame_size_bytes / stats_.size());
>-  if (num_key_frames > 0) {
>-    printf("  Average key frame size    : %7" PRIuS " bytes (%" PRIuS
>-           " keyframes)\n",
>-           total_encoded_key_frame_size_bytes / num_key_frames, num_key_frames);
>-  }
>-  if (num_delta_frames > 0) {
>-    printf("  Average non-key frame size: %7" PRIuS " bytes (%" PRIuS
>-           " frames)\n",
>-           total_encoded_delta_frame_size_bytes / num_delta_frames,
>-           num_delta_frames);
>-  }
>-
>-  // Bitrate stats.
>-  printf("Bitrates:\n");
>-  frame_it = std::min_element(stats_.begin(), stats_.end(), LessForBitRate);
>-  printf("  Min bitrate: %7d kbps (frame %d)\n", frame_it->bitrate_kbps,
>-         frame_it->frame_number);
>-  frame_it = std::max_element(stats_.begin(), stats_.end(), LessForBitRate);
>-  printf("  Max bitrate: %7d kbps (frame %d)\n", frame_it->bitrate_kbps,
>-         frame_it->frame_number);
>-
>-  // Quality.
>-  printf("Quality:\n");
>-  if (decoded_frames.empty()) {
>-    printf("No successfully decoded frames exist in this statistics.\n");
>-  } else {
>-    frame_it = std::min_element(decoded_frames.begin(), decoded_frames.end(),
>-                                LessForPsnr);
>-    printf("  PSNR min: %f (frame %d)\n", frame_it->psnr,
>-           frame_it->frame_number);
>-    printf("  PSNR avg: %f\n", total_psnr / decoded_frames.size());
>-
>-    frame_it = std::min_element(decoded_frames.begin(), decoded_frames.end(),
>-                                LessForSsim);
>-    printf("  SSIM min: %f (frame %d)\n", frame_it->ssim,
>-           frame_it->frame_number);
>-    printf("  SSIM avg: %f\n", total_ssim / decoded_frames.size());
>-  }
>-
>-  printf("\n");
>-  printf("Total encoding time  : %7d ms.\n", total_encoding_time_us / 1000);
>-  printf("Total decoding time  : %7d ms.\n", total_decoding_time_us / 1000);
>-  printf("Total processing time: %7d ms.\n",
>-         (total_encoding_time_us + total_decoding_time_us) / 1000);
>-
>-  // QP stats.
>-  int total_qp = 0;
>-  int total_qp_count = 0;
>-  for (const FrameStatistic& stat : stats_) {
>-    if (stat.qp >= 0) {
>-      total_qp += stat.qp;
>-      ++total_qp_count;
>-    }
>-  }
>-  int avg_qp = (total_qp_count > 0) ? (total_qp / total_qp_count) : -1;
>-  printf("Average QP: %d\n", avg_qp);
>-  printf("\n");
>-}
>-
>-}  // namespace test
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/stats.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/stats.h
>deleted file mode 100644
>index 02ca6410410..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/stats.h
>+++ /dev/null
>@@ -1,83 +0,0 @@
>-/*
>- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MODULES_VIDEO_CODING_CODECS_TEST_STATS_H_
>-#define MODULES_VIDEO_CODING_CODECS_TEST_STATS_H_
>-
>-#include <vector>
>-
>-#include "common_types.h"  // NOLINT(build/include)
>-
>-namespace webrtc {
>-namespace test {
>-
>-// Statistics for one processed frame.
>-struct FrameStatistic {
>-  explicit FrameStatistic(int frame_number) : frame_number(frame_number) {}
>-  const int frame_number = 0;
>-
>-  // Encoding.
>-  int64_t encode_start_ns = 0;
>-  int encode_return_code = 0;
>-  bool encoding_successful = false;
>-  int encode_time_us = 0;
>-  int bitrate_kbps = 0;
>-  size_t encoded_frame_size_bytes = 0;
>-  webrtc::FrameType frame_type = kVideoFrameDelta;
>-
>-  // H264 specific.
>-  rtc::Optional<size_t> max_nalu_length;
>-
>-  // Decoding.
>-  int64_t decode_start_ns = 0;
>-  int decode_return_code = 0;
>-  bool decoding_successful = false;
>-  int decode_time_us = 0;
>-  int decoded_width = 0;
>-  int decoded_height = 0;
>-
>-  // Quantization.
>-  int qp = -1;
>-
>-  // How many packets were discarded of the encoded frame data (if any).
>-  int packets_dropped = 0;
>-  size_t total_packets = 0;
>-  size_t manipulated_length = 0;
>-
>-  // Quality.
>-  float psnr = 0.0;
>-  float ssim = 0.0;
>-};
>-
>-// Statistics for a sequence of processed frames. This class is not thread safe.
>-class Stats {
>- public:
>-  Stats() = default;
>-  ~Stats() = default;
>-
>-  // Creates a FrameStatistic for the next frame to be processed.
>-  FrameStatistic* AddFrame();
>-
>-  // Returns the FrameStatistic corresponding to |frame_number|.
>-  FrameStatistic* GetFrame(int frame_number);
>-
>-  size_t size() const;
>-
>-  // TODO(brandtr): Add output as CSV.
>-  void PrintSummary() const;
>-
>- private:
>-  std::vector<FrameStatistic> stats_;
>-};
>-
>-}  // namespace test
>-}  // namespace webrtc
>-
>-#endif  // MODULES_VIDEO_CODING_CODECS_TEST_STATS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/stats_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/stats_unittest.cc
>deleted file mode 100644
>index 87727e2889f..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/stats_unittest.cc
>+++ /dev/null
>@@ -1,43 +0,0 @@
>-/*
>- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "modules/video_coding/codecs/test/stats.h"
>-
>-#include "test/gtest.h"
>-
>-namespace webrtc {
>-namespace test {
>-
>-TEST(StatsTest, TestEmptyObject) {
>-  Stats stats;
>-  stats.PrintSummary();  // Should not crash.
>-}
>-
>-TEST(StatsTest, AddSingleFrame) {
>-  Stats stats;
>-  FrameStatistic* frame_stat = stats.AddFrame();
>-  EXPECT_EQ(0, frame_stat->frame_number);
>-  EXPECT_EQ(1u, stats.size());
>-}
>-
>-TEST(StatsTest, AddMultipleFrames) {
>-  Stats stats;
>-  const int kNumFrames = 1000;
>-  for (int i = 0; i < kNumFrames; ++i) {
>-    FrameStatistic* frame_stat = stats.AddFrame();
>-    EXPECT_EQ(i, frame_stat->frame_number);
>-  }
>-  EXPECT_EQ(kNumFrames, static_cast<int>(stats.size()));
>-
>-  stats.PrintSummary();  // Should not crash.
>-}
>-
>-}  // namespace test
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/test_config.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/test_config.cc
>deleted file mode 100644
>index a289c619a5f..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/test_config.cc
>+++ /dev/null
>@@ -1,203 +0,0 @@
>-/*
>- *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "modules/video_coding/codecs/test/test_config.h"
>-
>-#include <sstream>
>-
>-#include "modules/video_coding/include/video_codec_interface.h"
>-#include "rtc_base/checks.h"
>-#include "system_wrappers/include/cpu_info.h"
>-#include "test/video_codec_settings.h"
>-
>-namespace webrtc {
>-namespace test {
>-
>-namespace {
>-const int kBaseKeyFrameInterval = 3000;
>-
>-std::string CodecSpecificToString(const webrtc::VideoCodec& codec) {
>-  std::stringstream ss;
>-  switch (codec.codecType) {
>-    case kVideoCodecVP8:
>-      ss << "\n  Complexity        : " << codec.VP8().complexity;
>-      ss << "\n  Resilience        : " << codec.VP8().resilience;
>-      ss << "\n  # temporal layers : "
>-         << static_cast<int>(codec.VP8().numberOfTemporalLayers);
>-      ss << "\n  Denoising         : " << codec.VP8().denoisingOn;
>-      ss << "\n  Error concealment : " << codec.VP8().errorConcealmentOn;
>-      ss << "\n  Automatic resize  : " << codec.VP8().automaticResizeOn;
>-      ss << "\n  Frame dropping    : " << codec.VP8().frameDroppingOn;
>-      ss << "\n  Key frame interval: " << codec.VP8().keyFrameInterval;
>-      break;
>-    case kVideoCodecVP9:
>-      ss << "\n  Complexity        : " << codec.VP9().complexity;
>-      ss << "\n  Resilience        : " << codec.VP9().resilienceOn;
>-      ss << "\n  # temporal layers : "
>-         << static_cast<int>(codec.VP9().numberOfTemporalLayers);
>-      ss << "\n  Denoising         : " << codec.VP9().denoisingOn;
>-      ss << "\n  Frame dropping    : " << codec.VP9().frameDroppingOn;
>-      ss << "\n  Key frame interval: " << codec.VP9().keyFrameInterval;
>-      ss << "\n  Adaptive QP mode  : " << codec.VP9().adaptiveQpMode;
>-      ss << "\n  Automatic resize  : " << codec.VP9().automaticResizeOn;
>-      ss << "\n  # spatial layers  : "
>-         << static_cast<int>(codec.VP9().numberOfSpatialLayers);
>-      ss << "\n  Flexible mode     : " << codec.VP9().flexibleMode;
>-      break;
>-    case kVideoCodecH264:
>-      ss << "\n  Frame dropping    : " << codec.H264().frameDroppingOn;
>-      ss << "\n  Key frame interval: " << codec.H264().keyFrameInterval;
>-      ss << "\n  Profile           : " << codec.H264().profile;
>-      break;
>-    default:
>-      break;
>-  }
>-  ss << "\n";
>-  return ss.str();
>-}
>-}  // namespace
>-
>-void TestConfig::SetCodecSettings(VideoCodecType codec_type,
>-                                  int num_temporal_layers,
>-                                  bool error_concealment_on,
>-                                  bool denoising_on,
>-                                  bool frame_dropper_on,
>-                                  bool spatial_resize_on,
>-                                  bool resilience_on,
>-                                  int width,
>-                                  int height) {
>-  webrtc::test::CodecSettings(codec_type, &codec_settings);
>-
>-  // TODO(brandtr): Move the setting of |width| and |height| to the tests, and
>-  // DCHECK that they are set before initializing the codec instead.
>-  codec_settings.width = width;
>-  codec_settings.height = height;
>-
>-  switch (codec_settings.codecType) {
>-    case kVideoCodecVP8:
>-      codec_settings.VP8()->resilience =
>-          resilience_on ? kResilientStream : kResilienceOff;
>-      codec_settings.VP8()->numberOfTemporalLayers = num_temporal_layers;
>-      codec_settings.VP8()->denoisingOn = denoising_on;
>-      codec_settings.VP8()->errorConcealmentOn = error_concealment_on;
>-      codec_settings.VP8()->automaticResizeOn = spatial_resize_on;
>-      codec_settings.VP8()->frameDroppingOn = frame_dropper_on;
>-      codec_settings.VP8()->keyFrameInterval = kBaseKeyFrameInterval;
>-      break;
>-    case kVideoCodecVP9:
>-      codec_settings.VP9()->resilienceOn = resilience_on;
>-      codec_settings.VP9()->numberOfTemporalLayers = num_temporal_layers;
>-      codec_settings.VP9()->denoisingOn = denoising_on;
>-      codec_settings.VP9()->frameDroppingOn = frame_dropper_on;
>-      codec_settings.VP9()->keyFrameInterval = kBaseKeyFrameInterval;
>-      codec_settings.VP9()->automaticResizeOn = spatial_resize_on;
>-      break;
>-    case kVideoCodecH264:
>-      codec_settings.H264()->frameDroppingOn = frame_dropper_on;
>-      codec_settings.H264()->keyFrameInterval = kBaseKeyFrameInterval;
>-      break;
>-    default:
>-      RTC_NOTREACHED();
>-      break;
>-  }
>-}
>-
>-int TestConfig::NumberOfCores() const {
>-  return use_single_core ? 1 : CpuInfo::DetectNumberOfCores();
>-}
>-
>-int TestConfig::NumberOfTemporalLayers() const {
>-  if (codec_settings.codecType == kVideoCodecVP8) {
>-    return codec_settings.VP8().numberOfTemporalLayers;
>-  } else if (codec_settings.codecType == kVideoCodecVP9) {
>-    return codec_settings.VP9().numberOfTemporalLayers;
>-  } else {
>-    return 1;
>-  }
>-}
>-
>-int TestConfig::TemporalLayerForFrame(int frame_idx) const {
>-  int tl = -1;
>-  switch (NumberOfTemporalLayers()) {
>-    case 1:
>-      tl = 0;
>-      break;
>-    case 2:
>-      // temporal layer 1:     1     3
>-      // temporal layer 0:  0     2     4 ...
>-      tl = (frame_idx % 2 == 0) ? 0 : 1;
>-      break;
>-    case 3:
>-      // temporal layer 2:     1     3     5     7
>-      // temporal layer 1:        2           6
>-      // temporal layer 0:  0           4           8 ...
>-      if (frame_idx % 4 == 0) {
>-        tl = 0;
>-      } else if ((frame_idx + 2) % 4 == 0) {
>-        tl = 1;
>-      } else if ((frame_idx + 1) % 2 == 0) {
>-        tl = 2;
>-      }
>-      break;
>-    default:
>-      RTC_NOTREACHED();
>-      break;
>-  }
>-  return tl;
>-}
>-
>-std::vector<FrameType> TestConfig::FrameTypeForFrame(int frame_idx) const {
>-  if (keyframe_interval > 0 && (frame_idx % keyframe_interval == 0)) {
>-    return {kVideoFrameKey};
>-  }
>-  return {kVideoFrameDelta};
>-}
>-
>-std::string TestConfig::ToString() const {
>-  std::string codec_type = CodecTypeToPayloadString(codec_settings.codecType);
>-  std::stringstream ss;
>-  ss << "\n Filename         : " << filename;
>-  ss << "\n # CPU cores used : " << NumberOfCores();
>-  ss << "\n General:";
>-  ss << "\n  Codec type        : " << codec_type;
>-  ss << "\n  Start bitrate     : " << codec_settings.startBitrate << " kbps";
>-  ss << "\n  Max bitrate       : " << codec_settings.maxBitrate << " kbps";
>-  ss << "\n  Min bitrate       : " << codec_settings.minBitrate << " kbps";
>-  ss << "\n  Width             : " << codec_settings.width;
>-  ss << "\n  Height            : " << codec_settings.height;
>-  ss << "\n  Max frame rate    : " << codec_settings.maxFramerate;
>-  ss << "\n  QPmax             : " << codec_settings.qpMax;
>-  ss << "\n " << codec_type << " specific: ";
>-  ss << CodecSpecificToString(codec_settings);
>-  return ss.str();
>-}
>-
>-std::string TestConfig::CodecName() const {
>-  std::string codec_name = CodecTypeToPayloadString(codec_settings.codecType);
>-  if (codec_settings.codecType == kVideoCodecH264) {
>-    if (h264_codec_settings.profile == H264::kProfileConstrainedHigh) {
>-      codec_name += "-CHP";
>-    } else {
>-      RTC_DCHECK_EQ(h264_codec_settings.profile,
>-                    H264::kProfileConstrainedBaseline);
>-      codec_name += "-CBP";
>-    }
>-  }
>-  return codec_name;
>-}
>-
>-std::string TestConfig::FilenameWithParams() const {
>-  std::string implementation_type = hw_encoder ? "hw" : "sw";
>-  return filename + "_" + CodecName() + "_" + implementation_type + "_" +
>-         std::to_string(codec_settings.startBitrate);
>-}
>-
>-}  // namespace test
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/test_config.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/test_config.h
>deleted file mode 100644
>index 04a687878c5..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/test_config.h
>+++ /dev/null
>@@ -1,126 +0,0 @@
>-/*
>- *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MODULES_VIDEO_CODING_CODECS_TEST_TEST_CONFIG_H_
>-#define MODULES_VIDEO_CODING_CODECS_TEST_TEST_CONFIG_H_
>-
>-#include <string>
>-#include <vector>
>-
>-#include "common_types.h"  // NOLINT(build/include)
>-#include "modules/video_coding/codecs/h264/include/h264_globals.h"
>-#include "modules/video_coding/codecs/test/packet_manipulator.h"
>-
>-namespace webrtc {
>-namespace test {
>-
>-// Defines which frame types shall be excluded from packet loss and when.
>-enum ExcludeFrameTypes {
>-  // Will exclude the first keyframe in the video sequence from packet loss.
>-  // Following keyframes will be targeted for packet loss.
>-  kExcludeOnlyFirstKeyFrame,
>-  // Exclude all keyframes from packet loss, no matter where in the video
>-  // sequence they occur.
>-  kExcludeAllKeyFrames
>-};
>-
>-// Test configuration for a test run.
>-struct TestConfig {
>-  class EncodedFrameChecker {
>-   public:
>-    virtual ~EncodedFrameChecker() = default;
>-
>-    virtual void CheckEncodedFrame(webrtc::VideoCodecType codec,
>-                                   const EncodedImage& encoded_frame) const = 0;
>-  };
>-
>-  void SetCodecSettings(VideoCodecType codec_type,
>-                        int num_temporal_layers,
>-                        bool error_concealment_on,
>-                        bool denoising_on,
>-                        bool frame_dropper_on,
>-                        bool spatial_resize_on,
>-                        bool resilience_on,
>-                        int width,
>-                        int height);
>-
>-  int NumberOfCores() const;
>-  int NumberOfTemporalLayers() const;
>-  int TemporalLayerForFrame(int frame_idx) const;
>-  std::vector<FrameType> FrameTypeForFrame(int frame_idx) const;
>-  std::string ToString() const;
>-  std::string CodecName() const;
>-  std::string FilenameWithParams() const;
>-
>-  // Plain name of YUV file to process without file extension.
>-  std::string filename;
>-
>-  // File to process. This must be a video file in the YUV format.
>-  std::string input_filename;
>-
>-  // File to write to during processing for the test. Will be a video file in
>-  // the YUV format.
>-  std::string output_filename;
>-
>-  // Number of frames to process.
>-  int num_frames = 0;
>-
>-  // Configurations related to networking.
>-  NetworkingConfig networking_config;
>-
>-  // Decides how the packet loss simulations shall exclude certain frames from
>-  // packet loss.
>-  ExcludeFrameTypes exclude_frame_types = kExcludeOnlyFirstKeyFrame;
>-
>-  // Force the encoder and decoder to use a single core for processing.
>-  // Using a single core is necessary to get a deterministic behavior for the
>-  // encoded frames - using multiple cores will produce different encoded frames
>-  // since multiple cores are competing to consume the byte budget for each
>-  // frame in parallel.
>-  // If set to false, the maximum number of available cores will be used.
>-  bool use_single_core = false;
>-
>-  // Should cpu usage be measured?
>-  // If set to true, the encoding will run in real-time.
>-  bool measure_cpu = false;
>-
>-  // If > 0: forces the encoder to create a keyframe every Nth frame.
>-  // Note that the encoder may create a keyframe in other locations in addition
>-  // to this setting. Forcing key frames may also affect encoder planning
>-  // optimizations in a negative way, since it will suddenly be forced to
>-  // produce an expensive key frame.
>-  int keyframe_interval = 0;
>-
>-  // Codec settings to use.
>-  webrtc::VideoCodec codec_settings;
>-
>-  // H.264 specific settings.
>-  struct H264CodecSettings {
>-    H264::Profile profile = H264::kProfileConstrainedBaseline;
>-    H264PacketizationMode packetization_mode =
>-        webrtc::H264PacketizationMode::NonInterleaved;
>-  } h264_codec_settings;
>-
>-  // Should hardware accelerated codecs be used?
>-  bool hw_encoder = false;
>-  bool hw_decoder = false;
>-
>-  // Should the hardware codecs be wrapped in software fallbacks?
>-  bool sw_fallback_encoder = false;
>-  bool sw_fallback_decoder = false;
>-
>-  // Custom checker that will be called for each frame.
>-  const EncodedFrameChecker* encoded_frame_checker = nullptr;
>-};
>-
>-}  // namespace test
>-}  // namespace webrtc
>-
>-#endif  // MODULES_VIDEO_CODING_CODECS_TEST_TEST_CONFIG_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/test_config_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/test_config_unittest.cc
>deleted file mode 100644
>index 968e1f39932..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/test_config_unittest.cc
>+++ /dev/null
>@@ -1,162 +0,0 @@
>-/*
>- *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "modules/video_coding/codecs/test/test_config.h"
>-#include "test/gmock.h"
>-#include "test/gtest.h"
>-#include "test/video_codec_settings.h"
>-
>-using ::testing::ElementsAre;
>-
>-namespace webrtc {
>-namespace test {
>-
>-namespace {
>-const int kNumTemporalLayers = 2;
>-}  // namespace
>-
>-TEST(TestConfig, NumberOfCoresWithUseSingleCore) {
>-  TestConfig config;
>-  config.use_single_core = true;
>-  EXPECT_EQ(1, config.NumberOfCores());
>-}
>-
>-TEST(TestConfig, NumberOfCoresWithoutUseSingleCore) {
>-  TestConfig config;
>-  config.use_single_core = false;
>-  EXPECT_GE(config.NumberOfCores(), 1);
>-}
>-
>-TEST(TestConfig, NumberOfTemporalLayersIsOne) {
>-  TestConfig config;
>-  webrtc::test::CodecSettings(kVideoCodecH264, &config.codec_settings);
>-  EXPECT_EQ(1, config.NumberOfTemporalLayers());
>-}
>-
>-TEST(TestConfig, NumberOfTemporalLayers_Vp8) {
>-  TestConfig config;
>-  webrtc::test::CodecSettings(kVideoCodecVP8, &config.codec_settings);
>-  config.codec_settings.VP8()->numberOfTemporalLayers = kNumTemporalLayers;
>-  EXPECT_EQ(kNumTemporalLayers, config.NumberOfTemporalLayers());
>-}
>-
>-TEST(TestConfig, NumberOfTemporalLayers_Vp9) {
>-  TestConfig config;
>-  webrtc::test::CodecSettings(kVideoCodecVP9, &config.codec_settings);
>-  config.codec_settings.VP9()->numberOfTemporalLayers = kNumTemporalLayers;
>-  EXPECT_EQ(kNumTemporalLayers, config.NumberOfTemporalLayers());
>-}
>-
>-TEST(TestConfig, TemporalLayersForFrame_OneLayer) {
>-  TestConfig config;
>-  webrtc::test::CodecSettings(kVideoCodecVP8, &config.codec_settings);
>-  config.codec_settings.VP8()->numberOfTemporalLayers = 1;
>-  EXPECT_EQ(0, config.TemporalLayerForFrame(0));
>-  EXPECT_EQ(0, config.TemporalLayerForFrame(1));
>-  EXPECT_EQ(0, config.TemporalLayerForFrame(2));
>-}
>-
>-TEST(TestConfig, TemporalLayersForFrame_TwoLayers) {
>-  TestConfig config;
>-  webrtc::test::CodecSettings(kVideoCodecVP8, &config.codec_settings);
>-  config.codec_settings.VP8()->numberOfTemporalLayers = 2;
>-  EXPECT_EQ(0, config.TemporalLayerForFrame(0));
>-  EXPECT_EQ(1, config.TemporalLayerForFrame(1));
>-  EXPECT_EQ(0, config.TemporalLayerForFrame(2));
>-  EXPECT_EQ(1, config.TemporalLayerForFrame(3));
>-}
>-
>-TEST(TestConfig, TemporalLayersForFrame_ThreeLayers) {
>-  TestConfig config;
>-  webrtc::test::CodecSettings(kVideoCodecVP8, &config.codec_settings);
>-  config.codec_settings.VP8()->numberOfTemporalLayers = 3;
>-  EXPECT_EQ(0, config.TemporalLayerForFrame(0));
>-  EXPECT_EQ(2, config.TemporalLayerForFrame(1));
>-  EXPECT_EQ(1, config.TemporalLayerForFrame(2));
>-  EXPECT_EQ(2, config.TemporalLayerForFrame(3));
>-  EXPECT_EQ(0, config.TemporalLayerForFrame(4));
>-  EXPECT_EQ(2, config.TemporalLayerForFrame(5));
>-  EXPECT_EQ(1, config.TemporalLayerForFrame(6));
>-  EXPECT_EQ(2, config.TemporalLayerForFrame(7));
>-}
>-
>-TEST(TestConfig, ForcedKeyFrameIntervalOff) {
>-  TestConfig config;
>-  config.keyframe_interval = 0;
>-  EXPECT_THAT(config.FrameTypeForFrame(0), ElementsAre(kVideoFrameDelta));
>-  EXPECT_THAT(config.FrameTypeForFrame(1), ElementsAre(kVideoFrameDelta));
>-  EXPECT_THAT(config.FrameTypeForFrame(2), ElementsAre(kVideoFrameDelta));
>-}
>-
>-TEST(TestConfig, ForcedKeyFrameIntervalOn) {
>-  TestConfig config;
>-  config.keyframe_interval = 3;
>-  EXPECT_THAT(config.FrameTypeForFrame(0), ElementsAre(kVideoFrameKey));
>-  EXPECT_THAT(config.FrameTypeForFrame(1), ElementsAre(kVideoFrameDelta));
>-  EXPECT_THAT(config.FrameTypeForFrame(2), ElementsAre(kVideoFrameDelta));
>-  EXPECT_THAT(config.FrameTypeForFrame(3), ElementsAre(kVideoFrameKey));
>-  EXPECT_THAT(config.FrameTypeForFrame(4), ElementsAre(kVideoFrameDelta));
>-  EXPECT_THAT(config.FrameTypeForFrame(5), ElementsAre(kVideoFrameDelta));
>-}
>-
>-TEST(TestConfig, ToString_Vp8) {
>-  TestConfig config;
>-  config.filename = "yuvfile";
>-  config.use_single_core = true;
>-
>-  config.SetCodecSettings(kVideoCodecVP8, 2, true,  // error_concealment_on,
>-                          false,                    // denoising_on,
>-                          false,                    // frame_dropper_on,
>-                          true,                     // spatial_resize_on,
>-                          false,                    // resilience_on,
>-                          320, 180);
>-  config.codec_settings.startBitrate = 400;
>-  config.codec_settings.maxBitrate = 500;
>-  config.codec_settings.minBitrate = 70;
>-  config.codec_settings.maxFramerate = 35;
>-  config.codec_settings.qpMax = 66;
>-  config.codec_settings.VP8()->complexity = kComplexityNormal;
>-  config.codec_settings.VP8()->keyFrameInterval = 999;
>-
>-  EXPECT_EQ(
>-      "\n Filename         : yuvfile"
>-      "\n # CPU cores used : 1"
>-      "\n General:"
>-      "\n  Codec type        : VP8"
>-      "\n  Start bitrate     : 400 kbps"
>-      "\n  Max bitrate       : 500 kbps"
>-      "\n  Min bitrate       : 70 kbps"
>-      "\n  Width             : 320"
>-      "\n  Height            : 180"
>-      "\n  Max frame rate    : 35"
>-      "\n  QPmax             : 66"
>-      "\n VP8 specific: "
>-      "\n  Complexity        : 0"
>-      "\n  Resilience        : 0"
>-      "\n  # temporal layers : 2"
>-      "\n  Denoising         : 0"
>-      "\n  Error concealment : 1"
>-      "\n  Automatic resize  : 1"
>-      "\n  Frame dropping    : 0"
>-      "\n  Key frame interval: 999\n",
>-      config.ToString());
>-}
>-
>-TEST(TestConfig, FilenameWithParams) {
>-  TestConfig config;
>-  config.filename = "filename";
>-  webrtc::test::CodecSettings(kVideoCodecVP8, &config.codec_settings);
>-  config.hw_encoder = true;
>-  config.codec_settings.startBitrate = 400;
>-  EXPECT_EQ("filename_VP8_hw_400", config.FilenameWithParams());
>-}
>-
>-}  // namespace test
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/video_codec_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/video_codec_unittest.cc
>similarity index 71%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/video_codec_test.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/video_codec_unittest.cc
>index 88e9b39ba17..12ffdd06411 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/video_codec_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/video_codec_unittest.cc
>@@ -8,27 +8,25 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "modules/video_coding/codecs/test/video_codec_test.h"
>+#include "modules/video_coding/codecs/test/video_codec_unittest.h"
> 
>-#include "api/video/i420_buffer.h"
>+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> #include "modules/video_coding/include/video_error_codes.h"
>-#include "test/frame_utils.h"
>-#include "test/testsupport/fileutils.h"
>+#include "test/video_codec_settings.h"
> 
> static const int kEncodeTimeoutMs = 100;
> static const int kDecodeTimeoutMs = 25;
> // Set bitrate to get higher quality.
> static const int kStartBitrate = 300;
>-static const int kTargetBitrate = 2000;
> static const int kMaxBitrate = 4000;
>-static const int kWidth = 172;        // Width of the input image.
>+static const int kWidth = 176;        // Width of the input image.
> static const int kHeight = 144;       // Height of the input image.
> static const int kMaxFramerate = 30;  // Arbitrary value.
> 
> namespace webrtc {
> 
> EncodedImageCallback::Result
>-VideoCodecTest::FakeEncodeCompleteCallback::OnEncodedImage(
>+VideoCodecUnitTest::FakeEncodeCompleteCallback::OnEncodedImage(
>     const EncodedImage& frame,
>     const CodecSpecificInfo* codec_specific_info,
>     const RTPFragmentationHeader* fragmentation) {
>@@ -49,35 +47,57 @@ VideoCodecTest::FakeEncodeCompleteCallback::OnEncodedImage(
>   return Result(Result::OK);
> }
> 
>-void VideoCodecTest::FakeDecodeCompleteCallback::Decoded(
>+void VideoCodecUnitTest::FakeDecodeCompleteCallback::Decoded(
>     VideoFrame& frame,
>-    rtc::Optional<int32_t> decode_time_ms,
>-    rtc::Optional<uint8_t> qp) {
>+    absl::optional<int32_t> decode_time_ms,
>+    absl::optional<uint8_t> qp) {
>   rtc::CritScope lock(&test_->decoded_frame_section_);
>   test_->decoded_frame_.emplace(frame);
>   test_->decoded_qp_ = qp;
>   test_->decoded_frame_event_.Set();
> }
> 
>-void VideoCodecTest::SetUp() {
>-  // Using a QCIF image. Processing only one frame.
>-  FILE* source_file_ =
>-      fopen(test::ResourcePath("paris_qcif", "yuv").c_str(), "rb");
>-  ASSERT_TRUE(source_file_ != NULL);
>-  rtc::scoped_refptr<VideoFrameBuffer> video_frame_buffer(
>-      test::ReadI420Buffer(kWidth, kHeight, source_file_));
>-  input_frame_.reset(new VideoFrame(video_frame_buffer, kVideoRotation_0, 0));
>-  fclose(source_file_);
>+void VideoCodecUnitTest::SetUp() {
>+  webrtc::test::CodecSettings(kVideoCodecVP8, &codec_settings_);
>+  codec_settings_.startBitrate = kStartBitrate;
>+  codec_settings_.maxBitrate = kMaxBitrate;
>+  codec_settings_.maxFramerate = kMaxFramerate;
>+  codec_settings_.width = kWidth;
>+  codec_settings_.height = kHeight;
>+
>+  ModifyCodecSettings(&codec_settings_);
>+
>+  input_frame_generator_ = test::FrameGenerator::CreateSquareGenerator(
>+      codec_settings_.width, codec_settings_.height,
>+      test::FrameGenerator::OutputType::I420, absl::optional<int>());
> 
>   encoder_ = CreateEncoder();
>   decoder_ = CreateDecoder();
>   encoder_->RegisterEncodeCompleteCallback(&encode_complete_callback_);
>   decoder_->RegisterDecodeCompleteCallback(&decode_complete_callback_);
> 
>-  InitCodecs();
>+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>+            encoder_->InitEncode(&codec_settings_, 1 /* number of cores */,
>+                                 0 /* max payload size (unused) */));
>+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>+            decoder_->InitDecode(&codec_settings_, 1 /* number of cores */));
>+}
>+
>+void VideoCodecUnitTest::ModifyCodecSettings(VideoCodec* codec_settings) {}
>+
>+VideoFrame* VideoCodecUnitTest::NextInputFrame() {
>+  VideoFrame* input_frame = input_frame_generator_->NextFrame();
>+
>+  const uint32_t timestamp =
>+      last_input_frame_timestamp_ +
>+      kVideoPayloadTypeFrequency / codec_settings_.maxFramerate;
>+  input_frame->set_timestamp(timestamp);
>+
>+  last_input_frame_timestamp_ = timestamp;
>+  return input_frame;
> }
> 
>-bool VideoCodecTest::WaitForEncodedFrame(
>+bool VideoCodecUnitTest::WaitForEncodedFrame(
>     EncodedImage* frame,
>     CodecSpecificInfo* codec_specific_info) {
>   std::vector<EncodedImage> frames;
>@@ -91,12 +111,12 @@ bool VideoCodecTest::WaitForEncodedFrame(
>   return true;
> }
> 
>-void VideoCodecTest::SetWaitForEncodedFramesThreshold(size_t num_frames) {
>+void VideoCodecUnitTest::SetWaitForEncodedFramesThreshold(size_t num_frames) {
>   rtc::CritScope lock(&encoded_frame_section_);
>   wait_for_encoded_frames_threshold_ = num_frames;
> }
> 
>-bool VideoCodecTest::WaitForEncodedFrames(
>+bool VideoCodecUnitTest::WaitForEncodedFrames(
>     std::vector<EncodedImage>* frames,
>     std::vector<CodecSpecificInfo>* codec_specific_info) {
>   EXPECT_TRUE(encoded_frame_event_.Wait(kEncodeTimeoutMs))
>@@ -118,8 +138,8 @@ bool VideoCodecTest::WaitForEncodedFrames(
>   }
> }
> 
>-bool VideoCodecTest::WaitForDecodedFrame(std::unique_ptr<VideoFrame>* frame,
>-                                         rtc::Optional<uint8_t>* qp) {
>+bool VideoCodecUnitTest::WaitForDecodedFrame(std::unique_ptr<VideoFrame>* frame,
>+                                             absl::optional<uint8_t>* qp) {
>   bool ret = decoded_frame_event_.Wait(kDecodeTimeoutMs);
>   EXPECT_TRUE(ret) << "Timed out while waiting for a decoded frame.";
>   // This becomes unsafe if there are multiple threads waiting for frames.
>@@ -135,19 +155,9 @@ bool VideoCodecTest::WaitForDecodedFrame(std::unique_ptr<VideoFrame>* frame,
>   }
> }
> 
>-void VideoCodecTest::InitCodecs() {
>-  codec_settings_ = codec_settings();
>-  codec_settings_.startBitrate = kStartBitrate;
>-  codec_settings_.targetBitrate = kTargetBitrate;
>-  codec_settings_.maxBitrate = kMaxBitrate;
>-  codec_settings_.maxFramerate = kMaxFramerate;
>-  codec_settings_.width = kWidth;
>-  codec_settings_.height = kHeight;
>-  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>-            encoder_->InitEncode(&codec_settings_, 1 /* number of cores */,
>-                                 0 /* max payload size (unused) */));
>-  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>-            decoder_->InitDecode(&codec_settings_, 1 /* number of cores */));
>+size_t VideoCodecUnitTest::GetNumEncodedFrames() {
>+  rtc::CritScope lock(&encoded_frame_section_);
>+  return encoded_frames_.size();
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/video_codec_test.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/video_codec_unittest.h
>similarity index 74%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/video_codec_test.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/video_codec_unittest.h
>index 6c67a26c7db..48860c35a6f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/video_codec_test.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/video_codec_unittest.h
>@@ -8,8 +8,8 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#ifndef MODULES_VIDEO_CODING_CODECS_TEST_VIDEO_CODEC_TEST_H_
>-#define MODULES_VIDEO_CODING_CODECS_TEST_VIDEO_CODEC_TEST_H_
>+#ifndef MODULES_VIDEO_CODING_CODECS_TEST_VIDEO_CODEC_UNITTEST_H_
>+#define MODULES_VIDEO_CODING_CODECS_TEST_VIDEO_CODEC_UNITTEST_H_
> 
> #include <memory>
> #include <vector>
>@@ -22,37 +22,41 @@
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/event.h"
> #include "rtc_base/thread_annotations.h"
>+#include "test/frame_generator.h"
> #include "test/gtest.h"
> 
> namespace webrtc {
> 
>-class VideoCodecTest : public ::testing::Test {
>+class VideoCodecUnitTest : public ::testing::Test {
>  public:
>-  VideoCodecTest()
>+  VideoCodecUnitTest()
>       : encode_complete_callback_(this),
>         decode_complete_callback_(this),
>         encoded_frame_event_(false /* manual reset */,
>                              false /* initially signaled */),
>         wait_for_encoded_frames_threshold_(1),
>         decoded_frame_event_(false /* manual reset */,
>-                             false /* initially signaled */) {}
>+                             false /* initially signaled */),
>+        last_input_frame_timestamp_(0) {}
> 
>  protected:
>   class FakeEncodeCompleteCallback : public webrtc::EncodedImageCallback {
>    public:
>-    explicit FakeEncodeCompleteCallback(VideoCodecTest* test) : test_(test) {}
>+    explicit FakeEncodeCompleteCallback(VideoCodecUnitTest* test)
>+        : test_(test) {}
> 
>     Result OnEncodedImage(const EncodedImage& frame,
>                           const CodecSpecificInfo* codec_specific_info,
>                           const RTPFragmentationHeader* fragmentation);
> 
>    private:
>-    VideoCodecTest* const test_;
>+    VideoCodecUnitTest* const test_;
>   };
> 
>   class FakeDecodeCompleteCallback : public webrtc::DecodedImageCallback {
>    public:
>-    explicit FakeDecodeCompleteCallback(VideoCodecTest* test) : test_(test) {}
>+    explicit FakeDecodeCompleteCallback(VideoCodecUnitTest* test)
>+        : test_(test) {}
> 
>     int32_t Decoded(VideoFrame& frame) override {
>       RTC_NOTREACHED();
>@@ -63,19 +67,22 @@ class VideoCodecTest : public ::testing::Test {
>       return -1;
>     }
>     void Decoded(VideoFrame& frame,
>-                 rtc::Optional<int32_t> decode_time_ms,
>-                 rtc::Optional<uint8_t> qp) override;
>+                 absl::optional<int32_t> decode_time_ms,
>+                 absl::optional<uint8_t> qp) override;
> 
>    private:
>-    VideoCodecTest* const test_;
>+    VideoCodecUnitTest* const test_;
>   };
> 
>   virtual std::unique_ptr<VideoEncoder> CreateEncoder() = 0;
>   virtual std::unique_ptr<VideoDecoder> CreateDecoder() = 0;
>-  virtual VideoCodec codec_settings() = 0;
> 
>   void SetUp() override;
> 
>+  virtual void ModifyCodecSettings(VideoCodec* codec_settings);
>+
>+  VideoFrame* NextInputFrame();
>+
>   // Helper method for waiting a single encoded frame.
>   bool WaitForEncodedFrame(EncodedImage* frame,
>                            CodecSpecificInfo* codec_specific_info);
>@@ -90,19 +97,17 @@ class VideoCodecTest : public ::testing::Test {
> 
>   // Helper method for waiting a single decoded frame.
>   bool WaitForDecodedFrame(std::unique_ptr<VideoFrame>* frame,
>-                           rtc::Optional<uint8_t>* qp);
>+                           absl::optional<uint8_t>* qp);
> 
>-  // Populated by InitCodecs().
>-  VideoCodec codec_settings_;
>+  size_t GetNumEncodedFrames();
> 
>-  std::unique_ptr<VideoFrame> input_frame_;
>+  VideoCodec codec_settings_;
> 
>   std::unique_ptr<VideoEncoder> encoder_;
>   std::unique_ptr<VideoDecoder> decoder_;
>+  std::unique_ptr<test::FrameGenerator> input_frame_generator_;
> 
>  private:
>-  void InitCodecs();
>-
>   FakeEncodeCompleteCallback encode_complete_callback_;
>   FakeDecodeCompleteCallback decode_complete_callback_;
> 
>@@ -116,11 +121,13 @@ class VideoCodecTest : public ::testing::Test {
> 
>   rtc::Event decoded_frame_event_;
>   rtc::CriticalSection decoded_frame_section_;
>-  rtc::Optional<VideoFrame> decoded_frame_
>+  absl::optional<VideoFrame> decoded_frame_
>       RTC_GUARDED_BY(decoded_frame_section_);
>-  rtc::Optional<uint8_t> decoded_qp_ RTC_GUARDED_BY(decoded_frame_section_);
>+  absl::optional<uint8_t> decoded_qp_ RTC_GUARDED_BY(decoded_frame_section_);
>+
>+  uint32_t last_input_frame_timestamp_;
> };
> 
> }  // namespace webrtc
> 
>-#endif  // MODULES_VIDEO_CODING_CODECS_TEST_VIDEO_CODEC_TEST_H_
>+#endif  // MODULES_VIDEO_CODING_CODECS_TEST_VIDEO_CODEC_UNITTEST_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/video_encoder_decoder_instantiation_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/video_encoder_decoder_instantiation_tests.cc
>new file mode 100644
>index 00000000000..b4fa6a10879
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/video_encoder_decoder_instantiation_tests.cc
>@@ -0,0 +1,151 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <memory>
>+#include <vector>
>+
>+#include "api/video_codecs/sdp_video_format.h"
>+#include "api/video_codecs/video_decoder.h"
>+#include "api/video_codecs/video_decoder_factory.h"
>+#include "api/video_codecs/video_encoder.h"
>+#include "api/video_codecs/video_encoder_factory.h"
>+#include "common_types.h"  // NOLINT(build/include)
>+#if defined(WEBRTC_ANDROID)
>+#include "modules/video_coding/codecs/test/android_codec_factory_helper.h"
>+#elif defined(WEBRTC_IOS)
>+#include "modules/video_coding/codecs/test/objc_codec_factory_helper.h"
>+#endif
>+#include "test/gtest.h"
>+#include "test/video_codec_settings.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+namespace {
>+
>+int32_t InitEncoder(VideoCodecType codec_type, VideoEncoder* encoder) {
>+  VideoCodec codec;
>+  CodecSettings(codec_type, &codec);
>+  codec.width = 640;
>+  codec.height = 480;
>+  codec.maxFramerate = 30;
>+  RTC_CHECK(encoder);
>+  return encoder->InitEncode(&codec, 1 /* number_of_cores */,
>+                             1200 /* max_payload_size */);
>+}
>+
>+int32_t InitDecoder(VideoCodecType codec_type, VideoDecoder* decoder) {
>+  VideoCodec codec;
>+  CodecSettings(codec_type, &codec);
>+  codec.width = 640;
>+  codec.height = 480;
>+  codec.maxFramerate = 30;
>+  RTC_CHECK(decoder);
>+  return decoder->InitDecode(&codec, 1 /* number_of_cores */);
>+}
>+
>+}  // namespace
>+
>+class VideoEncoderDecoderInstantiationTest
>+    : public ::testing::Test,
>+      public ::testing::WithParamInterface<::testing::tuple<int, int>> {
>+ protected:
>+  VideoEncoderDecoderInstantiationTest()
>+      : vp8_format_("VP8"),
>+        vp9_format_("VP9"),
>+        h264cbp_format_("H264"),
>+        num_encoders_(::testing::get<0>(GetParam())),
>+        num_decoders_(::testing::get<1>(GetParam())) {
>+#if defined(WEBRTC_ANDROID)
>+    InitializeAndroidObjects();
>+    encoder_factory_ = CreateAndroidEncoderFactory();
>+    decoder_factory_ = CreateAndroidDecoderFactory();
>+#elif defined(WEBRTC_IOS)
>+    encoder_factory_ = CreateObjCEncoderFactory();
>+    decoder_factory_ = CreateObjCDecoderFactory();
>+#else
>+    RTC_NOTREACHED() << "Only support Android and iOS.";
>+#endif
>+  }
>+
>+  ~VideoEncoderDecoderInstantiationTest() {
>+    for (auto& encoder : encoders_) {
>+      encoder->Release();
>+    }
>+    for (auto& decoder : decoders_) {
>+      decoder->Release();
>+    }
>+  }
>+
>+  const SdpVideoFormat vp8_format_;
>+  const SdpVideoFormat vp9_format_;
>+  const SdpVideoFormat h264cbp_format_;
>+  std::unique_ptr<VideoEncoderFactory> encoder_factory_;
>+  std::unique_ptr<VideoDecoderFactory> decoder_factory_;
>+
>+  const int num_encoders_;
>+  const int num_decoders_;
>+  std::vector<std::unique_ptr<VideoEncoder>> encoders_;
>+  std::vector<std::unique_ptr<VideoDecoder>> decoders_;
>+};
>+
>+INSTANTIATE_TEST_CASE_P(MultipleEncoders,
>+                        VideoEncoderDecoderInstantiationTest,
>+                        ::testing::Combine(::testing::Range(1, 4),
>+                                           ::testing::Range(1, 2)));
>+
>+INSTANTIATE_TEST_CASE_P(MultipleDecoders,
>+                        VideoEncoderDecoderInstantiationTest,
>+                        ::testing::Combine(::testing::Range(1, 2),
>+                                           ::testing::Range(1, 9)));
>+
>+INSTANTIATE_TEST_CASE_P(MultipleEncodersDecoders,
>+                        VideoEncoderDecoderInstantiationTest,
>+                        ::testing::Combine(::testing::Range(1, 4),
>+                                           ::testing::Range(1, 9)));
>+
>+// TODO(brandtr): Check that the factories actually support the codecs before
>+// trying to instantiate. Currently, we will just crash with a Java exception
>+// if the factory does not support the codec.
>+TEST_P(VideoEncoderDecoderInstantiationTest, DISABLED_InstantiateVp8Codecs) {
>+  for (int i = 0; i < num_encoders_; ++i) {
>+    std::unique_ptr<VideoEncoder> encoder =
>+        encoder_factory_->CreateVideoEncoder(vp8_format_);
>+    EXPECT_EQ(0, InitEncoder(kVideoCodecVP8, encoder.get()));
>+    encoders_.emplace_back(std::move(encoder));
>+  }
>+
>+  for (int i = 0; i < num_decoders_; ++i) {
>+    std::unique_ptr<VideoDecoder> decoder =
>+        decoder_factory_->CreateVideoDecoder(vp8_format_);
>+    EXPECT_EQ(0, InitDecoder(kVideoCodecVP8, decoder.get()));
>+    decoders_.emplace_back(std::move(decoder));
>+  }
>+}
>+
>+TEST_P(VideoEncoderDecoderInstantiationTest,
>+       DISABLED_InstantiateH264CBPCodecs) {
>+  for (int i = 0; i < num_encoders_; ++i) {
>+    std::unique_ptr<VideoEncoder> encoder =
>+        encoder_factory_->CreateVideoEncoder(h264cbp_format_);
>+    EXPECT_EQ(0, InitEncoder(kVideoCodecH264, encoder.get()));
>+    encoders_.emplace_back(std::move(encoder));
>+  }
>+
>+  for (int i = 0; i < num_decoders_; ++i) {
>+    std::unique_ptr<VideoDecoder> decoder =
>+        decoder_factory_->CreateVideoDecoder(h264cbp_format_);
>+    EXPECT_EQ(0, InitDecoder(kVideoCodecH264, decoder.get()));
>+    decoders_.emplace_back(std::move(decoder));
>+  }
>+}
>+
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_fixture_config_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_fixture_config_unittest.cc
>new file mode 100644
>index 00000000000..d01a842e972
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_fixture_config_unittest.cc
>@@ -0,0 +1,60 @@
>+/*
>+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/test/videocodec_test_fixture.h"
>+#include "test/gmock.h"
>+#include "test/gtest.h"
>+#include "test/video_codec_settings.h"
>+
>+using ::testing::ElementsAre;
>+
>+namespace webrtc {
>+namespace test {
>+
>+using Config = VideoCodecTestFixture::Config;
>+
>+namespace {
>+const size_t kNumTemporalLayers = 2;
>+}  // namespace
>+
>+TEST(Config, NumberOfCoresWithUseSingleCore) {
>+  Config config;
>+  config.use_single_core = true;
>+  EXPECT_EQ(1u, config.NumberOfCores());
>+}
>+
>+TEST(Config, NumberOfCoresWithoutUseSingleCore) {
>+  Config config;
>+  config.use_single_core = false;
>+  EXPECT_GE(config.NumberOfCores(), 1u);
>+}
>+
>+TEST(Config, NumberOfTemporalLayersIsOne) {
>+  Config config;
>+  webrtc::test::CodecSettings(kVideoCodecH264, &config.codec_settings);
>+  EXPECT_EQ(1u, config.NumberOfTemporalLayers());
>+}
>+
>+TEST(Config, NumberOfTemporalLayers_Vp8) {
>+  Config config;
>+  webrtc::test::CodecSettings(kVideoCodecVP8, &config.codec_settings);
>+  config.codec_settings.VP8()->numberOfTemporalLayers = kNumTemporalLayers;
>+  EXPECT_EQ(kNumTemporalLayers, config.NumberOfTemporalLayers());
>+}
>+
>+TEST(Config, NumberOfTemporalLayers_Vp9) {
>+  Config config;
>+  webrtc::test::CodecSettings(kVideoCodecVP9, &config.codec_settings);
>+  config.codec_settings.VP9()->numberOfTemporalLayers = kNumTemporalLayers;
>+  EXPECT_EQ(kNumTemporalLayers, config.NumberOfTemporalLayers());
>+}
>+
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_fixture_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_fixture_impl.cc
>new file mode 100644
>index 00000000000..a5c76ad5eb4
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_fixture_impl.cc
>@@ -0,0 +1,700 @@
>+/*
>+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/video_coding/codecs/test/videocodec_test_fixture_impl.h"
>+
>+#include <algorithm>
>+#include <memory>
>+#include <utility>
>+
>+#include "absl/memory/memory.h"
>+#include "api/video_codecs/sdp_video_format.h"
>+#include "common_types.h"  // NOLINT(build/include)
>+#include "media/base/h264_profile_level_id.h"
>+#include "media/base/mediaconstants.h"
>+#include "media/engine/internaldecoderfactory.h"
>+#include "media/engine/internalencoderfactory.h"
>+#include "media/engine/simulcast.h"
>+#include "modules/video_coding/codecs/vp9/svc_config.h"
>+#include "modules/video_coding/include/video_codec_interface.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/cpu_time.h"
>+#include "rtc_base/event.h"
>+#include "rtc_base/file.h"
>+#include "rtc_base/strings/string_builder.h"
>+#include "system_wrappers/include/cpu_info.h"
>+#include "system_wrappers/include/sleep.h"
>+#include "test/gtest.h"
>+#include "test/testsupport/fileutils.h"
>+#include "test/video_codec_settings.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+using VideoStatistics = VideoCodecTestStats::VideoStatistics;
>+
>+namespace {
>+const int kBaseKeyFrameInterval = 3000;
>+const int kMaxBitrateBps = 5000 * 1000;  // From kSimulcastFormats.
>+const double kBitratePriority = 1.0;
>+const int kMaxFramerateFps = 30;
>+const int kMaxQp = 56;
>+
>+void ConfigureSimulcast(VideoCodec* codec_settings) {
>+  const std::vector<webrtc::VideoStream> streams = cricket::GetSimulcastConfig(
>+      codec_settings->numberOfSimulcastStreams, codec_settings->width,
>+      codec_settings->height, kMaxBitrateBps, kBitratePriority, kMaxQp,
>+      kMaxFramerateFps, /* is_screenshare = */ false, true);
>+
>+  for (size_t i = 0; i < streams.size(); ++i) {
>+    SimulcastStream* ss = &codec_settings->simulcastStream[i];
>+    ss->width = static_cast<uint16_t>(streams[i].width);
>+    ss->height = static_cast<uint16_t>(streams[i].height);
>+    ss->numberOfTemporalLayers =
>+        static_cast<unsigned char>(*streams[i].num_temporal_layers);
>+    ss->maxBitrate = streams[i].max_bitrate_bps / 1000;
>+    ss->targetBitrate = streams[i].target_bitrate_bps / 1000;
>+    ss->minBitrate = streams[i].min_bitrate_bps / 1000;
>+    ss->qpMax = streams[i].max_qp;
>+    ss->active = true;
>+  }
>+}
>+
>+void ConfigureSvc(VideoCodec* codec_settings) {
>+  RTC_CHECK_EQ(kVideoCodecVP9, codec_settings->codecType);
>+
>+  const std::vector<SpatialLayer> layers =
>+      GetSvcConfig(codec_settings->width, codec_settings->height,
>+                   codec_settings->VP9()->numberOfSpatialLayers,
>+                   codec_settings->VP9()->numberOfTemporalLayers,
>+                   /* is_screen_sharing = */ false);
>+  ASSERT_EQ(codec_settings->VP9()->numberOfSpatialLayers, layers.size())
>+      << "GetSvcConfig returned fewer spatial layers than configured.";
>+
>+  for (size_t i = 0; i < layers.size(); ++i) {
>+    codec_settings->spatialLayers[i] = layers[i];
>+  }
>+}
>+
>+std::string CodecSpecificToString(const VideoCodec& codec) {
>+  char buf[1024];
>+  rtc::SimpleStringBuilder ss(buf);
>+  switch (codec.codecType) {
>+    case kVideoCodecVP8:
>+      ss << "complexity: " << static_cast<int>(codec.VP8().complexity);
>+      ss << "\nnum_temporal_layers: "
>+         << static_cast<int>(codec.VP8().numberOfTemporalLayers);
>+      ss << "\ndenoising: " << codec.VP8().denoisingOn;
>+      ss << "\nautomatic_resize: " << codec.VP8().automaticResizeOn;
>+      ss << "\nframe_dropping: " << codec.VP8().frameDroppingOn;
>+      ss << "\nkey_frame_interval: " << codec.VP8().keyFrameInterval;
>+      break;
>+    case kVideoCodecVP9:
>+      ss << "complexity: " << static_cast<int>(codec.VP9().complexity);
>+      ss << "\nnum_temporal_layers: "
>+         << static_cast<int>(codec.VP9().numberOfTemporalLayers);
>+      ss << "\nnum_spatial_layers: "
>+         << static_cast<int>(codec.VP9().numberOfSpatialLayers);
>+      ss << "\ndenoising: " << codec.VP9().denoisingOn;
>+      ss << "\nframe_dropping: " << codec.VP9().frameDroppingOn;
>+      ss << "\nkey_frame_interval: " << codec.VP9().keyFrameInterval;
>+      ss << "\nadaptive_qp_mode: " << codec.VP9().adaptiveQpMode;
>+      ss << "\nautomatic_resize: " << codec.VP9().automaticResizeOn;
>+      ss << "\nflexible_mode: " << codec.VP9().flexibleMode;
>+      break;
>+    case kVideoCodecH264:
>+      ss << "frame_dropping: " << codec.H264().frameDroppingOn;
>+      ss << "\nkey_frame_interval: " << codec.H264().keyFrameInterval;
>+      ss << "\nprofile: " << codec.H264().profile;
>+      break;
>+    default:
>+      break;
>+  }
>+  return ss.str();
>+}
>+
>+bool RunEncodeInRealTime(const VideoCodecTestFixtureImpl::Config& config) {
>+  if (config.measure_cpu) {
>+    return true;
>+  }
>+#if defined(WEBRTC_ANDROID)
>+  // In order to not overwhelm the OpenMAX buffers in the Android MediaCodec.
>+  return (config.hw_encoder || config.hw_decoder);
>+#else
>+  return false;
>+#endif
>+}
>+
>+std::string FilenameWithParams(
>+    const VideoCodecTestFixtureImpl::Config& config) {
>+  std::string implementation_type = config.hw_encoder ? "hw" : "sw";
>+  return config.filename + "_" + config.CodecName() + "_" +
>+         implementation_type + "_" +
>+         std::to_string(config.codec_settings.startBitrate);
>+}
>+
>+}  // namespace
>+
>+VideoCodecTestFixtureImpl::Config::Config() = default;
>+
>+void VideoCodecTestFixtureImpl::Config::SetCodecSettings(
>+    std::string codec_name,
>+    size_t num_simulcast_streams,
>+    size_t num_spatial_layers,
>+    size_t num_temporal_layers,
>+    bool denoising_on,
>+    bool frame_dropper_on,
>+    bool spatial_resize_on,
>+    size_t width,
>+    size_t height) {
>+  this->codec_name = codec_name;
>+  VideoCodecType codec_type = PayloadStringToCodecType(codec_name);
>+  webrtc::test::CodecSettings(codec_type, &codec_settings);
>+
>+  // TODO(brandtr): Move the setting of |width| and |height| to the tests, and
>+  // DCHECK that they are set before initializing the codec instead.
>+  codec_settings.width = static_cast<uint16_t>(width);
>+  codec_settings.height = static_cast<uint16_t>(height);
>+
>+  RTC_CHECK(num_simulcast_streams >= 1 &&
>+            num_simulcast_streams <= kMaxSimulcastStreams);
>+  RTC_CHECK(num_spatial_layers >= 1 && num_spatial_layers <= kMaxSpatialLayers);
>+  RTC_CHECK(num_temporal_layers >= 1 &&
>+            num_temporal_layers <= kMaxTemporalStreams);
>+
>+  // Simulcast is only available with VP8.
>+  RTC_CHECK(num_simulcast_streams < 2 || codec_type == kVideoCodecVP8);
>+
>+  // Spatial scalability is only available with VP9.
>+  RTC_CHECK(num_spatial_layers < 2 || codec_type == kVideoCodecVP9);
>+
>+  // Some base code requires numberOfSimulcastStreams to be set to zero
>+  // when simulcast is not used.
>+  codec_settings.numberOfSimulcastStreams =
>+      num_simulcast_streams <= 1 ? 0
>+                                 : static_cast<uint8_t>(num_simulcast_streams);
>+
>+  switch (codec_settings.codecType) {
>+    case kVideoCodecVP8:
>+      codec_settings.VP8()->numberOfTemporalLayers =
>+          static_cast<uint8_t>(num_temporal_layers);
>+      codec_settings.VP8()->denoisingOn = denoising_on;
>+      codec_settings.VP8()->automaticResizeOn = spatial_resize_on;
>+      codec_settings.VP8()->frameDroppingOn = frame_dropper_on;
>+      codec_settings.VP8()->keyFrameInterval = kBaseKeyFrameInterval;
>+      break;
>+    case kVideoCodecVP9:
>+      codec_settings.VP9()->numberOfTemporalLayers =
>+          static_cast<uint8_t>(num_temporal_layers);
>+      codec_settings.VP9()->denoisingOn = denoising_on;
>+      codec_settings.VP9()->frameDroppingOn = frame_dropper_on;
>+      codec_settings.VP9()->keyFrameInterval = kBaseKeyFrameInterval;
>+      codec_settings.VP9()->automaticResizeOn = spatial_resize_on;
>+      codec_settings.VP9()->numberOfSpatialLayers =
>+          static_cast<uint8_t>(num_spatial_layers);
>+      break;
>+    case kVideoCodecH264:
>+      codec_settings.H264()->frameDroppingOn = frame_dropper_on;
>+      codec_settings.H264()->keyFrameInterval = kBaseKeyFrameInterval;
>+      break;
>+    default:
>+      break;
>+  }
>+
>+  if (codec_settings.numberOfSimulcastStreams > 1) {
>+    ConfigureSimulcast(&codec_settings);
>+  } else if (codec_settings.codecType == kVideoCodecVP9 &&
>+             codec_settings.VP9()->numberOfSpatialLayers > 1) {
>+    ConfigureSvc(&codec_settings);
>+  }
>+}
>+
>+size_t VideoCodecTestFixtureImpl::Config::NumberOfCores() const {
>+  return use_single_core ? 1 : CpuInfo::DetectNumberOfCores();
>+}
>+
>+size_t VideoCodecTestFixtureImpl::Config::NumberOfTemporalLayers() const {
>+  if (codec_settings.codecType == kVideoCodecVP8) {
>+    return codec_settings.VP8().numberOfTemporalLayers;
>+  } else if (codec_settings.codecType == kVideoCodecVP9) {
>+    return codec_settings.VP9().numberOfTemporalLayers;
>+  } else {
>+    return 1;
>+  }
>+}
>+
>+size_t VideoCodecTestFixtureImpl::Config::NumberOfSpatialLayers() const {
>+  if (codec_settings.codecType == kVideoCodecVP9) {
>+    return codec_settings.VP9().numberOfSpatialLayers;
>+  } else {
>+    return 1;
>+  }
>+}
>+
>+size_t VideoCodecTestFixtureImpl::Config::NumberOfSimulcastStreams() const {
>+  return codec_settings.numberOfSimulcastStreams;
>+}
>+
>+std::string VideoCodecTestFixtureImpl::Config::ToString() const {
>+  std::string codec_type = CodecTypeToPayloadString(codec_settings.codecType);
>+  std::stringstream ss;
>+  ss << "filename: " << filename;
>+  ss << "\nnum_frames: " << num_frames;
>+  ss << "\nmax_payload_size_bytes: " << max_payload_size_bytes;
>+  ss << "\ndecode: " << decode;
>+  ss << "\nuse_single_core: " << use_single_core;
>+  ss << "\nmeasure_cpu: " << measure_cpu;
>+  ss << "\nnum_cores: " << NumberOfCores();
>+  ss << "\nkeyframe_interval: " << keyframe_interval;
>+  ss << "\ncodec_type: " << codec_type;
>+  ss << "\n\n--> codec_settings";
>+  ss << "\nwidth: " << codec_settings.width;
>+  ss << "\nheight: " << codec_settings.height;
>+  ss << "\nmax_framerate_fps: " << codec_settings.maxFramerate;
>+  ss << "\nstart_bitrate_kbps: " << codec_settings.startBitrate;
>+  ss << "\nmax_bitrate_kbps: " << codec_settings.maxBitrate;
>+  ss << "\nmin_bitrate_kbps: " << codec_settings.minBitrate;
>+  ss << "\nmax_qp: " << codec_settings.qpMax;
>+  ss << "\nnum_simulcast_streams: "
>+     << static_cast<int>(codec_settings.numberOfSimulcastStreams);
>+  ss << "\n\n--> codec_settings." << codec_type;
>+  ss << "\n" << CodecSpecificToString(codec_settings);
>+  if (codec_settings.numberOfSimulcastStreams > 1) {
>+    for (int i = 0; i < codec_settings.numberOfSimulcastStreams; ++i) {
>+      ss << "\n\n--> codec_settings.simulcastStream[" << i << "]";
>+      const SimulcastStream& simulcast_stream =
>+          codec_settings.simulcastStream[i];
>+      ss << "\nwidth: " << simulcast_stream.width;
>+      ss << "\nheight: " << simulcast_stream.height;
>+      ss << "\nnum_temporal_layers: "
>+         << static_cast<int>(simulcast_stream.numberOfTemporalLayers);
>+      ss << "\nmin_bitrate_kbps: " << simulcast_stream.minBitrate;
>+      ss << "\ntarget_bitrate_kbps: " << simulcast_stream.targetBitrate;
>+      ss << "\nmax_bitrate_kbps: " << simulcast_stream.maxBitrate;
>+      ss << "\nmax_qp: " << simulcast_stream.qpMax;
>+      ss << "\nactive: " << simulcast_stream.active;
>+    }
>+  }
>+  ss << "\n";
>+  return ss.str();
>+}
>+
>+std::string VideoCodecTestFixtureImpl::Config::CodecName() const {
>+  std::string name = codec_name;
>+  if (name.empty()) {
>+    name = CodecTypeToPayloadString(codec_settings.codecType);
>+  }
>+  if (codec_settings.codecType == kVideoCodecH264) {
>+    if (h264_codec_settings.profile == H264::kProfileConstrainedHigh) {
>+      return name + "-CHP";
>+    } else {
>+      RTC_DCHECK_EQ(h264_codec_settings.profile,
>+                    H264::kProfileConstrainedBaseline);
>+      return name + "-CBP";
>+    }
>+  }
>+  return name;
>+}
>+
>+bool VideoCodecTestFixtureImpl::Config::IsAsyncCodec() const {
>+  return hw_encoder || hw_decoder;
>+}
>+
>+// TODO(kthelgason): Move this out of the test fixture impl and
>+// make available as a shared utility class.
>+void VideoCodecTestFixtureImpl::H264KeyframeChecker::CheckEncodedFrame(
>+    webrtc::VideoCodecType codec,
>+    const EncodedImage& encoded_frame) const {
>+  EXPECT_EQ(kVideoCodecH264, codec);
>+  bool contains_sps = false;
>+  bool contains_pps = false;
>+  bool contains_idr = false;
>+  const std::vector<webrtc::H264::NaluIndex> nalu_indices =
>+      webrtc::H264::FindNaluIndices(encoded_frame._buffer,
>+                                    encoded_frame._length);
>+  for (const webrtc::H264::NaluIndex& index : nalu_indices) {
>+    webrtc::H264::NaluType nalu_type = webrtc::H264::ParseNaluType(
>+        encoded_frame._buffer[index.payload_start_offset]);
>+    if (nalu_type == webrtc::H264::NaluType::kSps) {
>+      contains_sps = true;
>+    } else if (nalu_type == webrtc::H264::NaluType::kPps) {
>+      contains_pps = true;
>+    } else if (nalu_type == webrtc::H264::NaluType::kIdr) {
>+      contains_idr = true;
>+    }
>+  }
>+  if (encoded_frame._frameType == kVideoFrameKey) {
>+    EXPECT_TRUE(contains_sps) << "Keyframe should contain SPS.";
>+    EXPECT_TRUE(contains_pps) << "Keyframe should contain PPS.";
>+    EXPECT_TRUE(contains_idr) << "Keyframe should contain IDR.";
>+  } else if (encoded_frame._frameType == kVideoFrameDelta) {
>+    EXPECT_FALSE(contains_sps) << "Delta frame should not contain SPS.";
>+    EXPECT_FALSE(contains_pps) << "Delta frame should not contain PPS.";
>+    EXPECT_FALSE(contains_idr) << "Delta frame should not contain IDR.";
>+  } else {
>+    RTC_NOTREACHED();
>+  }
>+}
>+
>+class VideoCodecTestFixtureImpl::CpuProcessTime final {
>+ public:
>+  explicit CpuProcessTime(const Config& config) : config_(config) {}
>+  ~CpuProcessTime() {}
>+
>+  void Start() {
>+    if (config_.measure_cpu) {
>+      cpu_time_ -= rtc::GetProcessCpuTimeNanos();
>+      wallclock_time_ -= rtc::SystemTimeNanos();
>+    }
>+  }
>+  void Stop() {
>+    if (config_.measure_cpu) {
>+      cpu_time_ += rtc::GetProcessCpuTimeNanos();
>+      wallclock_time_ += rtc::SystemTimeNanos();
>+    }
>+  }
>+  void Print() const {
>+    if (config_.measure_cpu) {
>+      printf("cpu_usage_percent: %f\n",
>+             GetUsagePercent() / config_.NumberOfCores());
>+      printf("\n");
>+    }
>+  }
>+
>+ private:
>+  double GetUsagePercent() const {
>+    return static_cast<double>(cpu_time_) / wallclock_time_ * 100.0;
>+  }
>+
>+  const Config config_;
>+  int64_t cpu_time_ = 0;
>+  int64_t wallclock_time_ = 0;
>+};
>+
>+VideoCodecTestFixtureImpl::VideoCodecTestFixtureImpl(Config config)
>+    : encoder_factory_(absl::make_unique<InternalEncoderFactory>()),
>+      decoder_factory_(absl::make_unique<InternalDecoderFactory>()),
>+      config_(config) {}
>+
>+VideoCodecTestFixtureImpl::VideoCodecTestFixtureImpl(
>+    Config config,
>+    std::unique_ptr<VideoDecoderFactory> decoder_factory,
>+    std::unique_ptr<VideoEncoderFactory> encoder_factory)
>+    : encoder_factory_(std::move(encoder_factory)),
>+      decoder_factory_(std::move(decoder_factory)),
>+      config_(config) {}
>+
>+VideoCodecTestFixtureImpl::~VideoCodecTestFixtureImpl() = default;
>+
>+// Processes all frames in the clip and verifies the result.
>+void VideoCodecTestFixtureImpl::RunTest(
>+    const std::vector<RateProfile>& rate_profiles,
>+    const std::vector<RateControlThresholds>* rc_thresholds,
>+    const std::vector<QualityThresholds>* quality_thresholds,
>+    const BitstreamThresholds* bs_thresholds) {
>+  RTC_DCHECK(!rate_profiles.empty());
>+
>+  // To emulate operation on a production VideoStreamEncoder, we call the
>+  // codecs on a task queue.
>+  rtc::test::TaskQueueForTest task_queue("VidProc TQ");
>+
>+  SetUpAndInitObjects(&task_queue,
>+                      static_cast<const int>(rate_profiles[0].target_kbps),
>+                      static_cast<const int>(rate_profiles[0].input_fps));
>+  PrintSettings(&task_queue);
>+  ProcessAllFrames(&task_queue, rate_profiles);
>+  ReleaseAndCloseObjects(&task_queue);
>+
>+  AnalyzeAllFrames(rate_profiles, rc_thresholds, quality_thresholds,
>+                   bs_thresholds);
>+}
>+
>+void VideoCodecTestFixtureImpl::ProcessAllFrames(
>+    rtc::TaskQueue* task_queue,
>+    const std::vector<RateProfile>& rate_profiles) {
>+  // Process all frames.
>+  size_t rate_update_index = 0;
>+
>+  // Set initial rates.
>+  task_queue->PostTask([this, &rate_profiles, rate_update_index] {
>+    processor_->SetRates(rate_profiles[rate_update_index].target_kbps,
>+                         rate_profiles[rate_update_index].input_fps);
>+  });
>+
>+  cpu_process_time_->Start();
>+
>+  for (size_t frame_number = 0; frame_number < config_.num_frames;
>+       ++frame_number) {
>+    if (frame_number ==
>+        rate_profiles[rate_update_index].frame_index_rate_update) {
>+      ++rate_update_index;
>+      RTC_DCHECK_GT(rate_profiles.size(), rate_update_index);
>+
>+      task_queue->PostTask([this, &rate_profiles, rate_update_index] {
>+        processor_->SetRates(rate_profiles[rate_update_index].target_kbps,
>+                             rate_profiles[rate_update_index].input_fps);
>+      });
>+    }
>+
>+    task_queue->PostTask([this] { processor_->ProcessFrame(); });
>+
>+    if (RunEncodeInRealTime(config_)) {
>+      // Roughly pace the frames.
>+      const size_t frame_duration_ms =
>+          rtc::kNumMillisecsPerSec / rate_profiles[rate_update_index].input_fps;
>+      SleepMs(static_cast<int>(frame_duration_ms));
>+    }
>+  }
>+
>+  // Wait until we know that the last frame has been sent for encode.
>+  rtc::Event sync_event(false, false);
>+  task_queue->PostTask([&sync_event] { sync_event.Set(); });
>+  sync_event.Wait(rtc::Event::kForever);
>+
>+  // Give the VideoProcessor pipeline some time to process the last frame,
>+  // and then release the codecs.
>+  if (config_.IsAsyncCodec()) {
>+    SleepMs(1 * rtc::kNumMillisecsPerSec);
>+  }
>+
>+  cpu_process_time_->Stop();
>+}
>+
>+void VideoCodecTestFixtureImpl::AnalyzeAllFrames(
>+    const std::vector<RateProfile>& rate_profiles,
>+    const std::vector<RateControlThresholds>* rc_thresholds,
>+    const std::vector<QualityThresholds>* quality_thresholds,
>+    const BitstreamThresholds* bs_thresholds) {
>+  for (size_t rate_update_idx = 0; rate_update_idx < rate_profiles.size();
>+       ++rate_update_idx) {
>+    const size_t first_frame_num =
>+        (rate_update_idx == 0)
>+            ? 0
>+            : rate_profiles[rate_update_idx - 1].frame_index_rate_update;
>+    const size_t last_frame_num =
>+        rate_profiles[rate_update_idx].frame_index_rate_update - 1;
>+    RTC_CHECK(last_frame_num >= first_frame_num);
>+
>+    std::vector<VideoStatistics> layer_stats =
>+        stats_.SliceAndCalcLayerVideoStatistic(first_frame_num, last_frame_num);
>+    printf("==> Receive stats\n");
>+    for (const auto& layer_stat : layer_stats) {
>+      printf("%s\n\n", layer_stat.ToString("recv_").c_str());
>+    }
>+
>+    VideoStatistics send_stat = stats_.SliceAndCalcAggregatedVideoStatistic(
>+        first_frame_num, last_frame_num);
>+    printf("==> Send stats\n");
>+    printf("%s\n", send_stat.ToString("send_").c_str());
>+
>+    const RateControlThresholds* rc_threshold =
>+        rc_thresholds ? &(*rc_thresholds)[rate_update_idx] : nullptr;
>+    const QualityThresholds* quality_threshold =
>+        quality_thresholds ? &(*quality_thresholds)[rate_update_idx] : nullptr;
>+
>+    VerifyVideoStatistic(send_stat, rc_threshold, quality_threshold,
>+                         bs_thresholds,
>+                         rate_profiles[rate_update_idx].target_kbps,
>+                         rate_profiles[rate_update_idx].input_fps);
>+  }
>+
>+  if (config_.print_frame_level_stats) {
>+    stats_.PrintFrameStatistics();
>+  }
>+
>+  cpu_process_time_->Print();
>+  printf("\n");
>+}
>+
>+void VideoCodecTestFixtureImpl::VerifyVideoStatistic(
>+    const VideoStatistics& video_stat,
>+    const RateControlThresholds* rc_thresholds,
>+    const QualityThresholds* quality_thresholds,
>+    const BitstreamThresholds* bs_thresholds,
>+    size_t target_bitrate_kbps,
>+    float input_framerate_fps) {
>+  if (rc_thresholds) {
>+    const float bitrate_mismatch_percent =
>+        100 * std::fabs(1.0f * video_stat.bitrate_kbps - target_bitrate_kbps) /
>+        target_bitrate_kbps;
>+    const float framerate_mismatch_percent =
>+        100 * std::fabs(video_stat.framerate_fps - input_framerate_fps) /
>+        input_framerate_fps;
>+    EXPECT_LE(bitrate_mismatch_percent,
>+              rc_thresholds->max_avg_bitrate_mismatch_percent);
>+    EXPECT_LE(video_stat.time_to_reach_target_bitrate_sec,
>+              rc_thresholds->max_time_to_reach_target_bitrate_sec);
>+    EXPECT_LE(framerate_mismatch_percent,
>+              rc_thresholds->max_avg_framerate_mismatch_percent);
>+    EXPECT_LE(video_stat.avg_delay_sec,
>+              rc_thresholds->max_avg_buffer_level_sec);
>+    EXPECT_LE(video_stat.max_key_frame_delay_sec,
>+              rc_thresholds->max_max_key_frame_delay_sec);
>+    EXPECT_LE(video_stat.max_delta_frame_delay_sec,
>+              rc_thresholds->max_max_delta_frame_delay_sec);
>+    EXPECT_LE(video_stat.num_spatial_resizes,
>+              rc_thresholds->max_num_spatial_resizes);
>+    EXPECT_LE(video_stat.num_key_frames, rc_thresholds->max_num_key_frames);
>+  }
>+
>+  if (quality_thresholds) {
>+    EXPECT_GT(video_stat.avg_psnr, quality_thresholds->min_avg_psnr);
>+    EXPECT_GT(video_stat.min_psnr, quality_thresholds->min_min_psnr);
>+    EXPECT_GT(video_stat.avg_ssim, quality_thresholds->min_avg_ssim);
>+    EXPECT_GT(video_stat.min_ssim, quality_thresholds->min_min_ssim);
>+  }
>+
>+  if (bs_thresholds) {
>+    EXPECT_LE(video_stat.max_nalu_size_bytes,
>+              bs_thresholds->max_max_nalu_size_bytes);
>+  }
>+}
>+
>+void VideoCodecTestFixtureImpl::CreateEncoderAndDecoder() {
>+  SdpVideoFormat::Parameters params;
>+  if (config_.codec_settings.codecType == kVideoCodecH264) {
>+    const char* packetization_mode =
>+        config_.h264_codec_settings.packetization_mode ==
>+                H264PacketizationMode::NonInterleaved
>+            ? "1"
>+            : "0";
>+    params = {{cricket::kH264FmtpProfileLevelId,
>+               *H264::ProfileLevelIdToString(H264::ProfileLevelId(
>+                   config_.h264_codec_settings.profile, H264::kLevel3_1))},
>+              {cricket::kH264FmtpPacketizationMode, packetization_mode}};
>+  } else {
>+    params = {};
>+  }
>+  SdpVideoFormat format(config_.codec_name);
>+
>+  encoder_ = encoder_factory_->CreateVideoEncoder(format);
>+  EXPECT_TRUE(encoder_) << "Encoder not successfully created.";
>+
>+  const size_t num_simulcast_or_spatial_layers = std::max(
>+      config_.NumberOfSimulcastStreams(), config_.NumberOfSpatialLayers());
>+  for (size_t i = 0; i < num_simulcast_or_spatial_layers; ++i) {
>+    decoders_.push_back(std::unique_ptr<VideoDecoder>(
>+        decoder_factory_->CreateVideoDecoder(format)));
>+  }
>+
>+  for (const auto& decoder : decoders_) {
>+    EXPECT_TRUE(decoder) << "Decoder not successfully created.";
>+  }
>+}
>+
>+void VideoCodecTestFixtureImpl::DestroyEncoderAndDecoder() {
>+  decoders_.clear();
>+  encoder_.reset();
>+}
>+
>+VideoCodecTestStats& VideoCodecTestFixtureImpl::GetStats() {
>+  return stats_;
>+}
>+
>+void VideoCodecTestFixtureImpl::SetUpAndInitObjects(
>+    rtc::test::TaskQueueForTest* task_queue,
>+    int initial_bitrate_kbps,
>+    int initial_framerate_fps) {
>+  config_.codec_settings.minBitrate = 0;
>+  config_.codec_settings.startBitrate = initial_bitrate_kbps;
>+  config_.codec_settings.maxFramerate = initial_framerate_fps;
>+
>+  // Create file objects for quality analysis.
>+  source_frame_reader_.reset(
>+      new YuvFrameReaderImpl(config_.filepath, config_.codec_settings.width,
>+                             config_.codec_settings.height));
>+  EXPECT_TRUE(source_frame_reader_->Init());
>+
>+  RTC_DCHECK(encoded_frame_writers_.empty());
>+  RTC_DCHECK(decoded_frame_writers_.empty());
>+  const size_t num_simulcast_or_spatial_layers = std::max(
>+      config_.NumberOfSimulcastStreams(), config_.NumberOfSpatialLayers());
>+  for (size_t simulcast_svc_idx = 0;
>+       simulcast_svc_idx < num_simulcast_or_spatial_layers;
>+       ++simulcast_svc_idx) {
>+    const std::string output_filename_base = OutputPath() +
>+                                             FilenameWithParams(config_) + "_" +
>+                                             std::to_string(simulcast_svc_idx);
>+
>+    if (config_.visualization_params.save_encoded_ivf) {
>+      rtc::File post_encode_file =
>+          rtc::File::Create(output_filename_base + ".ivf");
>+      encoded_frame_writers_.push_back(
>+          IvfFileWriter::Wrap(std::move(post_encode_file), 0));
>+    }
>+
>+    if (config_.visualization_params.save_decoded_y4m) {
>+      FrameWriter* decoded_frame_writer = new Y4mFrameWriterImpl(
>+          output_filename_base + ".y4m", config_.codec_settings.width,
>+          config_.codec_settings.height, initial_framerate_fps);
>+      EXPECT_TRUE(decoded_frame_writer->Init());
>+      decoded_frame_writers_.push_back(
>+          std::unique_ptr<FrameWriter>(decoded_frame_writer));
>+    }
>+  }
>+
>+  stats_.Clear();
>+
>+  cpu_process_time_.reset(new CpuProcessTime(config_));
>+
>+  task_queue->SendTask([this]() {
>+    CreateEncoderAndDecoder();
>+    processor_ = absl::make_unique<VideoProcessor>(
>+        encoder_.get(), &decoders_, source_frame_reader_.get(), config_,
>+        &stats_,
>+        encoded_frame_writers_.empty() ? nullptr : &encoded_frame_writers_,
>+        decoded_frame_writers_.empty() ? nullptr : &decoded_frame_writers_);
>+  });
>+}
>+
>+void VideoCodecTestFixtureImpl::ReleaseAndCloseObjects(
>+    rtc::test::TaskQueueForTest* task_queue) {
>+  task_queue->SendTask([this]() {
>+    processor_.reset();
>+    // The VideoProcessor must be destroyed before the codecs.
>+    DestroyEncoderAndDecoder();
>+  });
>+
>+  source_frame_reader_->Close();
>+
>+  // Close visualization files.
>+  for (auto& encoded_frame_writer : encoded_frame_writers_) {
>+    EXPECT_TRUE(encoded_frame_writer->Close());
>+  }
>+  encoded_frame_writers_.clear();
>+  for (auto& decoded_frame_writer : decoded_frame_writers_) {
>+    decoded_frame_writer->Close();
>+  }
>+  decoded_frame_writers_.clear();
>+}
>+
>+void VideoCodecTestFixtureImpl::PrintSettings(
>+    rtc::test::TaskQueueForTest* task_queue) const {
>+  printf("==> Config\n");
>+  printf("%s\n", config_.ToString().c_str());
>+
>+  printf("==> Codec names\n");
>+  std::string encoder_name;
>+  std::string decoder_name;
>+  task_queue->SendTask([this, &encoder_name, &decoder_name] {
>+    encoder_name = encoder_->ImplementationName();
>+    decoder_name = decoders_.at(0)->ImplementationName();
>+  });
>+  printf("enc_impl_name: %s\n", encoder_name.c_str());
>+  printf("dec_impl_name: %s\n", decoder_name.c_str());
>+  if (encoder_name == decoder_name) {
>+    printf("codec_impl_name: %s_%s\n", config_.CodecName().c_str(),
>+           encoder_name.c_str());
>+  }
>+  printf("\n");
>+}
>+
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_fixture_impl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_fixture_impl.h
>new file mode 100644
>index 00000000000..a54f6bd7b87
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_fixture_impl.h
>@@ -0,0 +1,107 @@
>+/*
>+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_VIDEO_CODING_CODECS_TEST_VIDEOCODEC_TEST_FIXTURE_IMPL_H_
>+#define MODULES_VIDEO_CODING_CODECS_TEST_VIDEOCODEC_TEST_FIXTURE_IMPL_H_
>+
>+#include <memory>
>+#include <string>
>+#include <vector>
>+
>+#include "api/test/videocodec_test_fixture.h"
>+#include "api/video_codecs/video_decoder_factory.h"
>+#include "api/video_codecs/video_encoder_factory.h"
>+#include "common_types.h"  // NOLINT(build/include)
>+#include "common_video/h264/h264_common.h"
>+#include "modules/video_coding/codecs/test/videocodec_test_stats_impl.h"
>+#include "modules/video_coding/codecs/test/videoprocessor.h"
>+#include "modules/video_coding/utility/ivf_file_writer.h"
>+#include "rtc_base/task_queue_for_test.h"
>+#include "test/testsupport/frame_reader.h"
>+#include "test/testsupport/frame_writer.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+// Integration test for video processor. It does rate control and frame quality
>+// analysis using frame statistics collected by video processor and logs the
>+// results. If thresholds are specified it checks that corresponding metrics
>+// are in desirable range.
>+class VideoCodecTestFixtureImpl : public VideoCodecTestFixture {
>+  // Verifies that all H.264 keyframes contain SPS/PPS/IDR NALUs.
>+ public:
>+  class H264KeyframeChecker : public EncodedFrameChecker {
>+   public:
>+    void CheckEncodedFrame(webrtc::VideoCodecType codec,
>+                           const EncodedImage& encoded_frame) const override;
>+  };
>+
>+  explicit VideoCodecTestFixtureImpl(Config config);
>+  VideoCodecTestFixtureImpl(
>+      Config config,
>+      std::unique_ptr<VideoDecoderFactory> decoder_factory,
>+      std::unique_ptr<VideoEncoderFactory> encoder_factory);
>+  ~VideoCodecTestFixtureImpl() override;
>+
>+  void RunTest(const std::vector<RateProfile>& rate_profiles,
>+               const std::vector<RateControlThresholds>* rc_thresholds,
>+               const std::vector<QualityThresholds>* quality_thresholds,
>+               const BitstreamThresholds* bs_thresholds) override;
>+
>+  VideoCodecTestStats& GetStats() override;
>+
>+ private:
>+  class CpuProcessTime;
>+
>+  void CreateEncoderAndDecoder();
>+  void DestroyEncoderAndDecoder();
>+  void SetUpAndInitObjects(rtc::test::TaskQueueForTest* task_queue,
>+                           int initial_bitrate_kbps,
>+                           int initial_framerate_fps);
>+  void ReleaseAndCloseObjects(rtc::test::TaskQueueForTest* task_queue);
>+
>+  void ProcessAllFrames(rtc::TaskQueue* task_queue,
>+                        const std::vector<RateProfile>& rate_profiles);
>+  void AnalyzeAllFrames(
>+      const std::vector<RateProfile>& rate_profiles,
>+      const std::vector<RateControlThresholds>* rc_thresholds,
>+      const std::vector<QualityThresholds>* quality_thresholds,
>+      const BitstreamThresholds* bs_thresholds);
>+
>+  void VerifyVideoStatistic(
>+      const VideoCodecTestStats::VideoStatistics& video_stat,
>+      const RateControlThresholds* rc_thresholds,
>+      const QualityThresholds* quality_thresholds,
>+      const BitstreamThresholds* bs_thresholds,
>+      size_t target_bitrate_kbps,
>+      float input_framerate_fps);
>+
>+  void PrintSettings(rtc::test::TaskQueueForTest* task_queue) const;
>+
>+  // Codecs.
>+  const std::unique_ptr<VideoEncoderFactory> encoder_factory_;
>+  std::unique_ptr<VideoEncoder> encoder_;
>+  const std::unique_ptr<VideoDecoderFactory> decoder_factory_;
>+  VideoProcessor::VideoDecoderList decoders_;
>+
>+  // Helper objects.
>+  Config config_;
>+  VideoCodecTestStatsImpl stats_;
>+  std::unique_ptr<FrameReader> source_frame_reader_;
>+  VideoProcessor::IvfFileWriterList encoded_frame_writers_;
>+  VideoProcessor::FrameWriterList decoded_frame_writers_;
>+  std::unique_ptr<VideoProcessor> processor_;
>+  std::unique_ptr<CpuProcessTime> cpu_process_time_;
>+};
>+
>+}  // namespace test
>+}  // namespace webrtc
>+
>+#endif  // MODULES_VIDEO_CODING_CODECS_TEST_VIDEOCODEC_TEST_FIXTURE_IMPL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_libvpx.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_libvpx.cc
>new file mode 100644
>index 00000000000..3bd0fe4b29a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_libvpx.cc
>@@ -0,0 +1,489 @@
>+/*
>+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <vector>
>+
>+#include "absl/memory/memory.h"
>+#include "api/test/create_videocodec_test_fixture.h"
>+#include "api/video_codecs/sdp_video_format.h"
>+#include "media/base/mediaconstants.h"
>+#include "media/engine/internaldecoderfactory.h"
>+#include "media/engine/internalencoderfactory.h"
>+#include "media/engine/simulcast_encoder_adapter.h"
>+#include "modules/video_coding/utility/vp8_header_parser.h"
>+#include "modules/video_coding/utility/vp9_uncompressed_header_parser.h"
>+#include "test/function_video_encoder_factory.h"
>+#include "test/gtest.h"
>+#include "test/testsupport/fileutils.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+using VideoStatistics = VideoCodecTestStats::VideoStatistics;
>+
>+namespace {
>+// Codec settings.
>+const int kCifWidth = 352;
>+const int kCifHeight = 288;
>+const int kNumFramesShort = 100;
>+const int kNumFramesLong = 300;
>+const size_t kBitrateRdPerfKbps[] = {100,  200,  300,  400,  500,  600,
>+                                     700,  800,  1000, 1250, 1400, 1600,
>+                                     1800, 2000, 2200, 2500};
>+const size_t kNumFirstFramesToSkipAtRdPerfAnalysis = 60;
>+
>+class QpFrameChecker : public VideoCodecTestFixture::EncodedFrameChecker {
>+ public:
>+  void CheckEncodedFrame(webrtc::VideoCodecType codec,
>+                         const EncodedImage& encoded_frame) const override {
>+    int qp;
>+    if (codec == kVideoCodecVP8) {
>+      EXPECT_TRUE(
>+          vp8::GetQp(encoded_frame._buffer, encoded_frame._length, &qp));
>+    } else if (codec == kVideoCodecVP9) {
>+      EXPECT_TRUE(
>+          vp9::GetQp(encoded_frame._buffer, encoded_frame._length, &qp));
>+    } else {
>+      RTC_NOTREACHED();
>+    }
>+    EXPECT_EQ(encoded_frame.qp_, qp) << "Encoder QP != parsed bitstream QP.";
>+  }
>+};
>+
>+VideoCodecTestFixture::Config CreateConfig() {
>+  VideoCodecTestFixture::Config config;
>+  config.filename = "foreman_cif";
>+  config.filepath = ResourcePath(config.filename, "yuv");
>+  config.num_frames = kNumFramesLong;
>+  config.use_single_core = true;
>+  config.hw_encoder = false;
>+  config.hw_decoder = false;
>+  return config;
>+}
>+
>+void PrintRdPerf(std::map<size_t, std::vector<VideoStatistics>> rd_stats) {
>+  printf("--> Summary\n");
>+  printf("%11s %5s %6s %11s %12s %11s %13s %13s %5s %7s %7s %7s %13s %13s\n",
>+         "uplink_kbps", "width", "height", "spatial_idx", "temporal_idx",
>+         "target_kbps", "downlink_kbps", "framerate_fps", "psnr", "psnr_y",
>+         "psnr_u", "psnr_v", "enc_speed_fps", "dec_speed_fps");
>+  for (const auto& rd_stat : rd_stats) {
>+    const size_t bitrate_kbps = rd_stat.first;
>+    for (const auto& layer_stat : rd_stat.second) {
>+      printf(
>+          "%11zu %5zu %6zu %11zu %12zu %11zu %13zu %13.2f %5.2f %7.2f %7.2f "
>+          "%7.2f"
>+          "%13.2f %13.2f\n",
>+          bitrate_kbps, layer_stat.width, layer_stat.height,
>+          layer_stat.spatial_idx, layer_stat.temporal_idx,
>+          layer_stat.target_bitrate_kbps, layer_stat.bitrate_kbps,
>+          layer_stat.framerate_fps, layer_stat.avg_psnr, layer_stat.avg_psnr_y,
>+          layer_stat.avg_psnr_u, layer_stat.avg_psnr_v,
>+          layer_stat.enc_speed_fps, layer_stat.dec_speed_fps);
>+    }
>+  }
>+}
>+}  // namespace
>+
>+#if !defined(RTC_DISABLE_VP9)
>+TEST(VideoCodecTestLibvpx, HighBitrateVP9) {
>+  auto config = CreateConfig();
>+  config.SetCodecSettings(cricket::kVp9CodecName, 1, 1, 1, false, true, false,
>+                          kCifWidth, kCifHeight);
>+  config.num_frames = kNumFramesShort;
>+  const auto frame_checker = absl::make_unique<QpFrameChecker>();
>+  config.encoded_frame_checker = frame_checker.get();
>+  auto fixture = CreateVideoCodecTestFixture(config);
>+
>+  std::vector<RateProfile> rate_profiles = {{500, 30, kNumFramesShort}};
>+
>+  std::vector<RateControlThresholds> rc_thresholds = {
>+      {5, 1, 0, 0.11, 0.3, 0.1, 0, 1}};
>+
>+  std::vector<QualityThresholds> quality_thresholds = {{37, 36, 0.94, 0.92}};
>+
>+  fixture->RunTest(rate_profiles, &rc_thresholds, &quality_thresholds, nullptr);
>+}
>+
>+TEST(VideoCodecTestLibvpx, ChangeBitrateVP9) {
>+  auto config = CreateConfig();
>+  config.SetCodecSettings(cricket::kVp9CodecName, 1, 1, 1, false, true, false,
>+                          kCifWidth, kCifHeight);
>+  const auto frame_checker = absl::make_unique<QpFrameChecker>();
>+  config.encoded_frame_checker = frame_checker.get();
>+  auto fixture = CreateVideoCodecTestFixture(config);
>+
>+  std::vector<RateProfile> rate_profiles = {
>+      {200, 30, 100},  // target_kbps, input_fps, frame_index_rate_update
>+      {700, 30, 200},
>+      {500, 30, kNumFramesLong}};
>+
>+  std::vector<RateControlThresholds> rc_thresholds = {
>+      {5, 1, 0, 0.15, 0.5, 0.1, 0, 1},
>+      {15, 2, 0, 0.2, 0.5, 0.1, 0, 0},
>+      {10, 1, 0, 0.3, 0.5, 0.1, 0, 0}};
>+
>+  std::vector<QualityThresholds> quality_thresholds = {
>+      {34, 33, 0.90, 0.88}, {38, 35, 0.95, 0.91}, {35, 34, 0.93, 0.90}};
>+
>+  fixture->RunTest(rate_profiles, &rc_thresholds, &quality_thresholds, nullptr);
>+}
>+
>+TEST(VideoCodecTestLibvpx, ChangeFramerateVP9) {
>+  auto config = CreateConfig();
>+  config.SetCodecSettings(cricket::kVp9CodecName, 1, 1, 1, false, true, false,
>+                          kCifWidth, kCifHeight);
>+  const auto frame_checker = absl::make_unique<QpFrameChecker>();
>+  config.encoded_frame_checker = frame_checker.get();
>+  auto fixture = CreateVideoCodecTestFixture(config);
>+
>+  std::vector<RateProfile> rate_profiles = {
>+      {100, 24, 100},  // target_kbps, input_fps, frame_index_rate_update
>+      {100, 15, 200},
>+      {100, 10, kNumFramesLong}};
>+
>+  // Framerate mismatch should be lower for lower framerate.
>+  std::vector<RateControlThresholds> rc_thresholds = {
>+      {10, 2, 40, 0.4, 0.5, 0.2, 0, 1},
>+      {8, 2, 5, 0.2, 0.5, 0.2, 0, 0},
>+      {5, 2, 0, 0.21, 0.5, 0.3, 0, 0}};
>+
>+  // Quality should be higher for lower framerates for the same content.
>+  std::vector<QualityThresholds> quality_thresholds = {
>+      {33, 32, 0.89, 0.87}, {33.5, 32, 0.90, 0.86}, {33.5, 31.5, 0.90, 0.85}};
>+
>+  fixture->RunTest(rate_profiles, &rc_thresholds, &quality_thresholds, nullptr);
>+}
>+
>+TEST(VideoCodecTestLibvpx, DenoiserOnVP9) {
>+  auto config = CreateConfig();
>+  config.SetCodecSettings(cricket::kVp9CodecName, 1, 1, 1, true, true, false,
>+                          kCifWidth, kCifHeight);
>+  config.num_frames = kNumFramesShort;
>+  const auto frame_checker = absl::make_unique<QpFrameChecker>();
>+  config.encoded_frame_checker = frame_checker.get();
>+  auto fixture = CreateVideoCodecTestFixture(config);
>+
>+  std::vector<RateProfile> rate_profiles = {{500, 30, kNumFramesShort}};
>+
>+  std::vector<RateControlThresholds> rc_thresholds = {
>+      {5, 1, 0, 0.11, 0.3, 0.1, 0, 1}};
>+
>+  std::vector<QualityThresholds> quality_thresholds = {{37.5, 36, 0.94, 0.93}};
>+
>+  fixture->RunTest(rate_profiles, &rc_thresholds, &quality_thresholds, nullptr);
>+}
>+
>+TEST(VideoCodecTestLibvpx, VeryLowBitrateVP9) {
>+  auto config = CreateConfig();
>+  config.SetCodecSettings(cricket::kVp9CodecName, 1, 1, 1, false, true, true,
>+                          kCifWidth, kCifHeight);
>+  const auto frame_checker = absl::make_unique<QpFrameChecker>();
>+  config.encoded_frame_checker = frame_checker.get();
>+  auto fixture = CreateVideoCodecTestFixture(config);
>+
>+  std::vector<RateProfile> rate_profiles = {{50, 30, kNumFramesLong}};
>+
>+  std::vector<RateControlThresholds> rc_thresholds = {
>+      {15, 3, 75, 1.0, 0.5, 0.4, 1, 1}};
>+
>+  std::vector<QualityThresholds> quality_thresholds = {{28, 25, 0.80, 0.65}};
>+
>+  fixture->RunTest(rate_profiles, &rc_thresholds, &quality_thresholds, nullptr);
>+}
>+
>+// TODO(marpan): Add temporal layer test for VP9, once changes are in
>+// vp9 wrapper for this.
>+
>+#endif  // !defined(RTC_DISABLE_VP9)
>+
>+TEST(VideoCodecTestLibvpx, HighBitrateVP8) {
>+  auto config = CreateConfig();
>+  config.SetCodecSettings(cricket::kVp8CodecName, 1, 1, 1, true, true, false,
>+                          kCifWidth, kCifHeight);
>+  config.num_frames = kNumFramesShort;
>+  const auto frame_checker = absl::make_unique<QpFrameChecker>();
>+  config.encoded_frame_checker = frame_checker.get();
>+  auto fixture = CreateVideoCodecTestFixture(config);
>+
>+  std::vector<RateProfile> rate_profiles = {{500, 30, kNumFramesShort}};
>+
>+  std::vector<RateControlThresholds> rc_thresholds = {
>+      {5, 1, 0, 0.1, 0.2, 0.1, 0, 1}};
>+
>+  // std::vector<QualityThresholds> quality_thresholds = {{37, 35, 0.93, 0.91}};
>+  // TODO(webrtc:8757): ARM VP8 encoder's quality is significantly worse
>+  // than quality of x86 version. Use lower thresholds for now.
>+  std::vector<QualityThresholds> quality_thresholds = {{35, 33, 0.91, 0.89}};
>+
>+  fixture->RunTest(rate_profiles, &rc_thresholds, &quality_thresholds, nullptr);
>+}
>+
>+// The tests below are currently disabled for Android. For ARM, the encoder
>+// uses |cpu_speed| = 12, as opposed to default |cpu_speed| <= 6 for x86,
>+// which leads to significantly different quality. The quality and rate control
>+// settings in the tests below are defined for encoder speed setting
>+// |cpu_speed| <= ~6. A number of settings would need to be significantly
>+// modified for the |cpu_speed| = 12 case. For now, keep the tests below
>+// disabled on Android. Some quality parameter in the above test has been
>+// adjusted to also pass for |cpu_speed| <= 12.
>+
>+// TODO(webrtc:9267): Fails on iOS
>+#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
>+#define MAYBE_ChangeBitrateVP8 DISABLED_ChangeBitrateVP8
>+#else
>+#define MAYBE_ChangeBitrateVP8 ChangeBitrateVP8
>+#endif
>+TEST(VideoCodecTestLibvpx, MAYBE_ChangeBitrateVP8) {
>+  auto config = CreateConfig();
>+  config.SetCodecSettings(cricket::kVp8CodecName, 1, 1, 1, true, true, false,
>+                          kCifWidth, kCifHeight);
>+  const auto frame_checker = absl::make_unique<QpFrameChecker>();
>+  config.encoded_frame_checker = frame_checker.get();
>+  auto fixture = CreateVideoCodecTestFixture(config);
>+
>+  std::vector<RateProfile> rate_profiles = {
>+      {200, 30, 100},  // target_kbps, input_fps, frame_index_rate_update
>+      {800, 30, 200},
>+      {500, 30, kNumFramesLong}};
>+
>+  std::vector<RateControlThresholds> rc_thresholds = {
>+      {5, 1, 0, 0.1, 0.2, 0.1, 0, 1},
>+      {15.5, 1, 0, 0.1, 0.2, 0.1, 0, 0},
>+      {15, 1, 0, 0.3, 0.2, 0.1, 0, 0}};
>+
>+  // std::vector<QualityThresholds> quality_thresholds = {
>+  //     {33, 32, 0.89, 0.88}, {38, 36, 0.94, 0.93}, {35, 34, 0.92, 0.91}};
>+  // TODO(webrtc:8757): ARM VP8 encoder's quality is significantly worse
>+  // than quality of x86 version. Use lower thresholds for now.
>+  std::vector<QualityThresholds> quality_thresholds = {
>+      {31.8, 31, 0.86, 0.85}, {36, 34.8, 0.92, 0.90}, {33.5, 32, 0.90, 0.88}};
>+
>+  fixture->RunTest(rate_profiles, &rc_thresholds, &quality_thresholds, nullptr);
>+}
>+
>+// TODO(webrtc:9267): Fails on iOS
>+#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
>+#define MAYBE_ChangeFramerateVP8 DISABLED_ChangeFramerateVP8
>+#else
>+#define MAYBE_ChangeFramerateVP8 ChangeFramerateVP8
>+#endif
>+TEST(VideoCodecTestLibvpx, MAYBE_ChangeFramerateVP8) {
>+  auto config = CreateConfig();
>+  config.SetCodecSettings(cricket::kVp8CodecName, 1, 1, 1, true, true, false,
>+                          kCifWidth, kCifHeight);
>+  const auto frame_checker = absl::make_unique<QpFrameChecker>();
>+  config.encoded_frame_checker = frame_checker.get();
>+  auto fixture = CreateVideoCodecTestFixture(config);
>+
>+  std::vector<RateProfile> rate_profiles = {
>+      {80, 24, 100},  // target_kbps, input_fps, frame_index_rate_update
>+      {80, 15, 200},
>+      {80, 10, kNumFramesLong}};
>+
>+  // std::vector<RateControlThresholds> rc_thresholds = {
>+  //     {10, 2, 20, 0.4, 0.3, 0.1, 0, 1},
>+  //     {5, 2, 5, 0.3, 0.3, 0.1, 0, 0},
>+  //     {4, 2, 1, 0.2, 0.3, 0.2, 0, 0}};
>+  // TODO(webrtc:8757): ARM VP8 drops more frames than x86 version. Use lower
>+  // thresholds for now.
>+  std::vector<RateControlThresholds> rc_thresholds = {
>+      {10, 2, 60, 0.5, 0.3, 0.3, 0, 1},
>+      {10, 2, 30, 0.3, 0.3, 0.3, 0, 0},
>+      {10, 2, 10, 0.2, 0.3, 0.2, 0, 0}};
>+
>+  // std::vector<QualityThresholds> quality_thresholds = {
>+  //     {31, 30, 0.87, 0.86}, {32, 31, 0.89, 0.86}, {32, 30, 0.87, 0.82}};
>+  // TODO(webrtc:8757): ARM VP8 encoder's quality is significantly worse
>+  // than quality of x86 version. Use lower thresholds for now.
>+  std::vector<QualityThresholds> quality_thresholds = {
>+      {31, 30, 0.85, 0.84}, {31.5, 30.5, 0.86, 0.84}, {30.5, 29, 0.83, 0.78}};
>+
>+  fixture->RunTest(rate_profiles, &rc_thresholds, &quality_thresholds, nullptr);
>+}
>+
>+#if defined(WEBRTC_ANDROID)
>+#define MAYBE_TemporalLayersVP8 DISABLED_TemporalLayersVP8
>+#else
>+#define MAYBE_TemporalLayersVP8 TemporalLayersVP8
>+#endif
>+TEST(VideoCodecTestLibvpx, MAYBE_TemporalLayersVP8) {
>+  auto config = CreateConfig();
>+  config.SetCodecSettings(cricket::kVp8CodecName, 1, 1, 3, true, true, false,
>+                          kCifWidth, kCifHeight);
>+  const auto frame_checker = absl::make_unique<QpFrameChecker>();
>+  config.encoded_frame_checker = frame_checker.get();
>+  auto fixture = CreateVideoCodecTestFixture(config);
>+
>+  std::vector<RateProfile> rate_profiles = {{200, 30, 150},
>+                                            {400, 30, kNumFramesLong}};
>+
>+  // std::vector<RateControlThresholds> rc_thresholds = {
>+  //     {5, 1, 0, 0.1, 0.2, 0.1, 0, 1}, {10, 2, 0, 0.1, 0.2, 0.1, 0, 1}};
>+  // TODO(webrtc:8757): ARM VP8 drops more frames than x86 version. Use lower
>+  // thresholds for now.
>+  std::vector<RateControlThresholds> rc_thresholds = {
>+      {10, 1, 2, 0.3, 0.2, 0.1, 0, 1}, {12, 2, 3, 0.1, 0.2, 0.1, 0, 1}};
>+
>+  // Min SSIM drops because of high motion scene with complex backgound (trees).
>+  // std::vector<QualityThresholds> quality_thresholds = {{32, 30, 0.88, 0.85},
>+  //                                                     {33, 30, 0.89, 0.83}};
>+  // TODO(webrtc:8757): ARM VP8 encoder's quality is significantly worse
>+  // than quality of x86 version. Use lower thresholds for now.
>+  std::vector<QualityThresholds> quality_thresholds = {{31, 30, 0.85, 0.84},
>+                                                       {31, 28, 0.85, 0.75}};
>+
>+  fixture->RunTest(rate_profiles, &rc_thresholds, &quality_thresholds, nullptr);
>+}
>+
>+// TODO(webrtc:9267): Fails on iOS
>+#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
>+#define MAYBE_MultiresVP8 DISABLED_MultiresVP8
>+#else
>+#define MAYBE_MultiresVP8 MultiresVP8
>+#endif
>+TEST(VideoCodecTestLibvpx, MAYBE_MultiresVP8) {
>+  auto config = CreateConfig();
>+  config.filename = "ConferenceMotion_1280_720_50";
>+  config.filepath = ResourcePath(config.filename, "yuv");
>+  config.num_frames = 100;
>+  config.SetCodecSettings(cricket::kVp8CodecName, 3, 1, 3, true, true, false,
>+                          1280, 720);
>+  const auto frame_checker = absl::make_unique<QpFrameChecker>();
>+  config.encoded_frame_checker = frame_checker.get();
>+  auto fixture = CreateVideoCodecTestFixture(config);
>+
>+  std::vector<RateProfile> rate_profiles = {{1500, 30, config.num_frames}};
>+
>+  std::vector<RateControlThresholds> rc_thresholds = {
>+      {5, 1, 5, 0.2, 0.3, 0.1, 0, 1}};
>+  std::vector<QualityThresholds> quality_thresholds = {{34, 32, 0.90, 0.88}};
>+
>+  fixture->RunTest(rate_profiles, &rc_thresholds, &quality_thresholds, nullptr);
>+}
>+
>+#if defined(WEBRTC_ANDROID)
>+#define MAYBE_SimulcastVP8 DISABLED_SimulcastVP8
>+#else
>+#define MAYBE_SimulcastVP8 SimulcastVP8
>+#endif
>+TEST(VideoCodecTestLibvpx, MAYBE_SimulcastVP8) {
>+  auto config = CreateConfig();
>+  config.filename = "ConferenceMotion_1280_720_50";
>+  config.filepath = ResourcePath(config.filename, "yuv");
>+  config.num_frames = 100;
>+  config.SetCodecSettings(cricket::kVp8CodecName, 3, 1, 3, true, true, false,
>+                          1280, 720);
>+  const auto frame_checker = absl::make_unique<QpFrameChecker>();
>+  config.encoded_frame_checker = frame_checker.get();
>+
>+  InternalEncoderFactory internal_encoder_factory;
>+  std::unique_ptr<VideoEncoderFactory> adapted_encoder_factory =
>+      absl::make_unique<FunctionVideoEncoderFactory>([&]() {
>+        return absl::make_unique<SimulcastEncoderAdapter>(
>+            &internal_encoder_factory, SdpVideoFormat(cricket::kVp8CodecName));
>+      });
>+  std::unique_ptr<InternalDecoderFactory> internal_decoder_factory(
>+      new InternalDecoderFactory());
>+
>+  auto fixture =
>+      CreateVideoCodecTestFixture(config, std::move(internal_decoder_factory),
>+                                  std::move(adapted_encoder_factory));
>+
>+  std::vector<RateProfile> rate_profiles = {{1500, 30, config.num_frames}};
>+
>+  std::vector<RateControlThresholds> rc_thresholds = {
>+      {20, 5, 90, 0.8, 0.5, 0.3, 0, 1}};
>+  std::vector<QualityThresholds> quality_thresholds = {{34, 32, 0.90, 0.88}};
>+
>+  fixture->RunTest(rate_profiles, &rc_thresholds, &quality_thresholds, nullptr);
>+}
>+
>+#if defined(WEBRTC_ANDROID)
>+#define MAYBE_SvcVP9 DISABLED_SvcVP9
>+#else
>+#define MAYBE_SvcVP9 SvcVP9
>+#endif
>+TEST(VideoCodecTestLibvpx, MAYBE_SvcVP9) {
>+  auto config = CreateConfig();
>+  config.filename = "ConferenceMotion_1280_720_50";
>+  config.filepath = ResourcePath(config.filename, "yuv");
>+  config.num_frames = 100;
>+  config.SetCodecSettings(cricket::kVp9CodecName, 1, 3, 3, true, true, false,
>+                          1280, 720);
>+  const auto frame_checker = absl::make_unique<QpFrameChecker>();
>+  config.encoded_frame_checker = frame_checker.get();
>+  auto fixture = CreateVideoCodecTestFixture(config);
>+
>+  std::vector<RateProfile> rate_profiles = {{1500, 30, config.num_frames}};
>+
>+  std::vector<RateControlThresholds> rc_thresholds = {
>+      {5, 1, 5, 0.2, 0.3, 0.1, 0, 1}};
>+  std::vector<QualityThresholds> quality_thresholds = {{36, 34, 0.93, 0.90}};
>+
>+  fixture->RunTest(rate_profiles, &rc_thresholds, &quality_thresholds, nullptr);
>+}
>+
>+TEST(VideoCodecTestLibvpx, DISABLED_MultiresVP8RdPerf) {
>+  auto config = CreateConfig();
>+  config.filename = "FourPeople_1280x720_30";
>+  config.filepath = ResourcePath(config.filename, "yuv");
>+  config.num_frames = 300;
>+  config.print_frame_level_stats = true;
>+  config.SetCodecSettings(cricket::kVp8CodecName, 3, 1, 3, true, true, false,
>+                          1280, 720);
>+  const auto frame_checker = absl::make_unique<QpFrameChecker>();
>+  config.encoded_frame_checker = frame_checker.get();
>+  auto fixture = CreateVideoCodecTestFixture(config);
>+
>+  std::map<size_t, std::vector<VideoStatistics>> rd_stats;
>+  for (size_t bitrate_kbps : kBitrateRdPerfKbps) {
>+    std::vector<RateProfile> rate_profiles = {
>+        {bitrate_kbps, 30, config.num_frames}};
>+
>+    fixture->RunTest(rate_profiles, nullptr, nullptr, nullptr);
>+
>+    rd_stats[bitrate_kbps] =
>+        fixture->GetStats().SliceAndCalcLayerVideoStatistic(
>+            kNumFirstFramesToSkipAtRdPerfAnalysis, config.num_frames - 1);
>+  }
>+
>+  PrintRdPerf(rd_stats);
>+}
>+
>+TEST(VideoCodecTestLibvpx, DISABLED_SvcVP9RdPerf) {
>+  auto config = CreateConfig();
>+  config.filename = "FourPeople_1280x720_30";
>+  config.filepath = ResourcePath(config.filename, "yuv");
>+  config.num_frames = 300;
>+  config.print_frame_level_stats = true;
>+  config.SetCodecSettings(cricket::kVp9CodecName, 1, 3, 3, true, true, false,
>+                          1280, 720);
>+  const auto frame_checker = absl::make_unique<QpFrameChecker>();
>+  config.encoded_frame_checker = frame_checker.get();
>+  auto fixture = CreateVideoCodecTestFixture(config);
>+
>+  std::map<size_t, std::vector<VideoStatistics>> rd_stats;
>+  for (size_t bitrate_kbps : kBitrateRdPerfKbps) {
>+    std::vector<RateProfile> rate_profiles = {
>+        {bitrate_kbps, 30, config.num_frames}};
>+
>+    fixture->RunTest(rate_profiles, nullptr, nullptr, nullptr);
>+
>+    rd_stats[bitrate_kbps] =
>+        fixture->GetStats().SliceAndCalcLayerVideoStatistic(
>+            kNumFirstFramesToSkipAtRdPerfAnalysis, config.num_frames - 1);
>+  }
>+
>+  PrintRdPerf(rd_stats);
>+}
>+
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_mediacodec.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_mediacodec.cc
>new file mode 100644
>index 00000000000..4fc4c3a8deb
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_mediacodec.cc
>@@ -0,0 +1,152 @@
>+/*
>+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <string>
>+#include <tuple>
>+#include <vector>
>+
>+#include "absl/memory/memory.h"
>+#include "api/test/create_videocodec_test_fixture.h"
>+#include "common_types.h"  // NOLINT(build/include)
>+#include "media/base/mediaconstants.h"
>+#include "modules/video_coding/codecs/test/android_codec_factory_helper.h"
>+#include "modules/video_coding/codecs/test/videocodec_test_fixture_impl.h"
>+#include "test/gtest.h"
>+#include "test/testsupport/fileutils.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+namespace {
>+const int kForemanNumFrames = 300;
>+const int kForemanFramerateFps = 30;
>+
>+VideoCodecTestFixture::Config CreateConfig() {
>+  VideoCodecTestFixture::Config config;
>+  config.filename = "foreman_cif";
>+  config.filepath = ResourcePath(config.filename, "yuv");
>+  config.num_frames = kForemanNumFrames;
>+  config.hw_encoder = true;
>+  config.hw_decoder = true;
>+  return config;
>+}
>+
>+std::unique_ptr<VideoCodecTestFixture> CreateTestFixtureWithConfig(
>+    VideoCodecTestFixture::Config config) {
>+  InitializeAndroidObjects();  // Idempotent.
>+  auto encoder_factory = CreateAndroidEncoderFactory();
>+  auto decoder_factory = CreateAndroidDecoderFactory();
>+  return CreateVideoCodecTestFixture(config, std::move(decoder_factory),
>+                                     std::move(encoder_factory));
>+}
>+}  // namespace
>+
>+TEST(VideoCodecTestMediaCodec, ForemanCif500kbpsVp8) {
>+  auto config = CreateConfig();
>+  config.SetCodecSettings(cricket::kVp8CodecName, 1, 1, 1, false, false, false,
>+                          352, 288);
>+  auto fixture = CreateTestFixtureWithConfig(config);
>+
>+  std::vector<RateProfile> rate_profiles = {
>+      {500, kForemanFramerateFps, kForemanNumFrames}};
>+
>+  // The thresholds below may have to be tweaked to let even poor MediaCodec
>+  // implementations pass. If this test fails on the bots, disable it and
>+  // ping brandtr@.
>+  std::vector<RateControlThresholds> rc_thresholds = {
>+      {10, 1, 1, 0.1, 0.2, 0.1, 0, 1}};
>+
>+  std::vector<QualityThresholds> quality_thresholds = {{36, 31, 0.92, 0.86}};
>+
>+  fixture->RunTest(rate_profiles, &rc_thresholds, &quality_thresholds, nullptr);
>+}
>+
>+TEST(VideoCodecTestMediaCodec, ForemanCif500kbpsH264CBP) {
>+  auto config = CreateConfig();
>+  const auto frame_checker =
>+      absl::make_unique<VideoCodecTestFixtureImpl::H264KeyframeChecker>();
>+  config.encoded_frame_checker = frame_checker.get();
>+  config.SetCodecSettings(cricket::kH264CodecName, 1, 1, 1, false, false, false,
>+                          352, 288);
>+  auto fixture = CreateTestFixtureWithConfig(config);
>+
>+  std::vector<RateProfile> rate_profiles = {
>+      {500, kForemanFramerateFps, kForemanNumFrames}};
>+
>+  // The thresholds below may have to be tweaked to let even poor MediaCodec
>+  // implementations pass. If this test fails on the bots, disable it and
>+  // ping brandtr@.
>+  std::vector<RateControlThresholds> rc_thresholds = {
>+      {10, 1, 1, 0.1, 0.2, 0.1, 0, 1}};
>+
>+  std::vector<QualityThresholds> quality_thresholds = {{36, 31, 0.92, 0.86}};
>+
>+  fixture->RunTest(rate_profiles, &rc_thresholds, &quality_thresholds, nullptr);
>+}
>+
>+// TODO(brandtr): Enable this test when we have trybots/buildbots with
>+// HW encoders that support CHP.
>+TEST(VideoCodecTestMediaCodec, DISABLED_ForemanCif500kbpsH264CHP) {
>+  auto config = CreateConfig();
>+  const auto frame_checker =
>+      absl::make_unique<VideoCodecTestFixtureImpl::H264KeyframeChecker>();
>+
>+  config.h264_codec_settings.profile = H264::kProfileConstrainedHigh;
>+  config.encoded_frame_checker = frame_checker.get();
>+  config.SetCodecSettings(cricket::kH264CodecName, 1, 1, 1, false, false, false,
>+                          352, 288);
>+  auto fixture = CreateTestFixtureWithConfig(config);
>+
>+  std::vector<RateProfile> rate_profiles = {
>+      {500, kForemanFramerateFps, kForemanNumFrames}};
>+
>+  // The thresholds below may have to be tweaked to let even poor MediaCodec
>+  // implementations pass. If this test fails on the bots, disable it and
>+  // ping brandtr@.
>+  std::vector<RateControlThresholds> rc_thresholds = {
>+      {5, 1, 0, 0.1, 0.2, 0.1, 0, 1}};
>+
>+  std::vector<QualityThresholds> quality_thresholds = {{37, 35, 0.93, 0.91}};
>+
>+  fixture->RunTest(rate_profiles, &rc_thresholds, &quality_thresholds, nullptr);
>+}
>+
>+TEST(VideoCodecTestMediaCodec, ForemanMixedRes100kbpsVp8H264) {
>+  auto config = CreateConfig();
>+  const int kNumFrames = 30;
>+  const std::vector<std::string> codecs = {cricket::kVp8CodecName,
>+                                           cricket::kH264CodecName};
>+  const std::vector<std::tuple<int, int>> resolutions = {
>+      {128, 96}, {160, 120}, {176, 144}, {240, 136}, {320, 240}, {480, 272}};
>+  const std::vector<RateProfile> rate_profiles = {
>+      {100, kForemanFramerateFps, kNumFrames}};
>+  const std::vector<QualityThresholds> quality_thresholds = {
>+      {29, 26, 0.8, 0.75}};
>+
>+  for (const auto& codec : codecs) {
>+    for (const auto& resolution : resolutions) {
>+      const int width = std::get<0>(resolution);
>+      const int height = std::get<1>(resolution);
>+      config.filename = std::string("foreman_") + std::to_string(width) + "x" +
>+                        std::to_string(height);
>+      config.filepath = ResourcePath(config.filename, "yuv");
>+      config.num_frames = kNumFrames;
>+      config.SetCodecSettings(codec, 1, 1, 1, false, false, false, width,
>+                              height);
>+
>+      auto fixture = CreateTestFixtureWithConfig(config);
>+      fixture->RunTest(rate_profiles, nullptr /* rc_thresholds */,
>+                       &quality_thresholds, nullptr /* bs_thresholds */);
>+    }
>+  }
>+}
>+
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_openh264.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_openh264.cc
>new file mode 100644
>index 00000000000..e5cf71ba90f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_openh264.cc
>@@ -0,0 +1,88 @@
>+/*
>+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <vector>
>+
>+#include "api/test/create_videocodec_test_fixture.h"
>+#include "media/base/mediaconstants.h"
>+#include "modules/video_coding/codecs/test/videocodec_test_fixture_impl.h"
>+#include "test/gtest.h"
>+#include "test/testsupport/fileutils.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+namespace {
>+// Codec settings.
>+const int kCifWidth = 352;
>+const int kCifHeight = 288;
>+const int kNumFrames = 100;
>+
>+VideoCodecTestFixture::Config CreateConfig() {
>+  VideoCodecTestFixture::Config config;
>+  config.filename = "foreman_cif";
>+  config.filepath = ResourcePath(config.filename, "yuv");
>+  config.num_frames = kNumFrames;
>+  // Only allow encoder/decoder to use single core, for predictability.
>+  config.use_single_core = true;
>+  config.hw_encoder = false;
>+  config.hw_decoder = false;
>+  return config;
>+}
>+}  // namespace
>+
>+TEST(VideoCodecTestOpenH264, ConstantHighBitrate) {
>+  auto frame_checker =
>+      absl::make_unique<VideoCodecTestFixtureImpl::H264KeyframeChecker>();
>+  auto config = CreateConfig();
>+  config.SetCodecSettings(cricket::kH264CodecName, 1, 1, 1, false, true, false,
>+                          kCifWidth, kCifHeight);
>+  config.encoded_frame_checker = frame_checker.get();
>+  auto fixture = CreateVideoCodecTestFixture(config);
>+
>+  std::vector<RateProfile> rate_profiles = {{500, 30, kNumFrames}};
>+
>+  std::vector<RateControlThresholds> rc_thresholds = {
>+      {5, 1, 0, 0.1, 0.2, 0.1, 0, 1}};
>+
>+  std::vector<QualityThresholds> quality_thresholds = {{37, 35, 0.93, 0.91}};
>+
>+  fixture->RunTest(rate_profiles, &rc_thresholds, &quality_thresholds, nullptr);
>+}
>+
>+// H264: Enable SingleNalUnit packetization mode. Encoder should split
>+// large frames into multiple slices and limit length of NAL units.
>+TEST(VideoCodecTestOpenH264, SingleNalUnit) {
>+  auto frame_checker =
>+      absl::make_unique<VideoCodecTestFixtureImpl::H264KeyframeChecker>();
>+  auto config = CreateConfig();
>+  config.h264_codec_settings.packetization_mode =
>+      H264PacketizationMode::SingleNalUnit;
>+  config.max_payload_size_bytes = 500;
>+  config.SetCodecSettings(cricket::kH264CodecName, 1, 1, 1, false, true, false,
>+                          kCifWidth, kCifHeight);
>+  config.encoded_frame_checker = frame_checker.get();
>+  auto fixture = CreateVideoCodecTestFixture(config);
>+
>+  std::vector<RateProfile> rate_profiles = {{500, 30, kNumFrames}};
>+
>+  std::vector<RateControlThresholds> rc_thresholds = {
>+      {5, 1, 0, 0.1, 0.2, 0.1, 0, 1}};
>+
>+  std::vector<QualityThresholds> quality_thresholds = {{37, 35, 0.93, 0.91}};
>+
>+  BitstreamThresholds bs_thresholds = {config.max_payload_size_bytes};
>+
>+  fixture->RunTest(rate_profiles, &rc_thresholds, &quality_thresholds,
>+                   &bs_thresholds);
>+}
>+
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_parameterized.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_parameterized.cc
>new file mode 100644
>index 00000000000..0d49b80d95a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_parameterized.cc
>@@ -0,0 +1,101 @@
>+/*
>+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/test/create_videocodec_test_fixture.h"
>+#include "test/gtest.h"
>+#include "test/testsupport/fileutils.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+namespace {
>+
>+// Loop variables.
>+const size_t kBitrates[] = {500};
>+const VideoCodecType kVideoCodecType[] = {kVideoCodecVP8};
>+const bool kHwCodec[] = {false};
>+
>+// Codec settings.
>+const int kNumSpatialLayers = 1;
>+const int kNumTemporalLayers = 1;
>+const bool kDenoisingOn = false;
>+const bool kSpatialResizeOn = false;
>+const bool kFrameDropperOn = false;
>+
>+// Test settings.
>+const bool kUseSingleCore = false;
>+const bool kMeasureCpu = false;
>+const int kNumFrames = 30;
>+}  // namespace
>+
>+// Tests for plotting statistics from logs.
>+class VideoCodecTestParameterized
>+    : public ::testing::Test,
>+      public ::testing::WithParamInterface<
>+          ::testing::tuple<size_t, VideoCodecType, bool>> {
>+ protected:
>+  VideoCodecTestParameterized()
>+      : bitrate_(::testing::get<0>(GetParam())),
>+        codec_type_(::testing::get<1>(GetParam())),
>+        hw_codec_(::testing::get<2>(GetParam())) {}
>+  ~VideoCodecTestParameterized() override = default;
>+
>+  void RunTest(size_t width,
>+               size_t height,
>+               size_t framerate,
>+               const std::string& filename) {
>+    VideoCodecTestFixture::Config config;
>+    config.filename = filename;
>+    config.filepath = ResourcePath(filename, "yuv");
>+    config.use_single_core = kUseSingleCore;
>+    config.measure_cpu = kMeasureCpu;
>+    config.hw_encoder = hw_codec_;
>+    config.hw_decoder = hw_codec_;
>+    config.num_frames = kNumFrames;
>+
>+    const size_t num_simulcast_streams =
>+        codec_type_ == kVideoCodecVP8 ? kNumSpatialLayers : 1;
>+    const size_t num_spatial_layers =
>+        codec_type_ == kVideoCodecVP9 ? kNumSpatialLayers : 1;
>+
>+    const std::string codec_name = CodecTypeToPayloadString(codec_type_);
>+    config.SetCodecSettings(codec_name, num_simulcast_streams,
>+                            num_spatial_layers, kNumTemporalLayers,
>+                            kDenoisingOn, kFrameDropperOn, kSpatialResizeOn,
>+                            width, height);
>+
>+    std::vector<RateProfile> rate_profiles = {
>+        {bitrate_, framerate, kNumFrames}};
>+
>+    fixture_ = CreateVideoCodecTestFixture(config);
>+    fixture_->RunTest(rate_profiles, nullptr, nullptr, nullptr);
>+  }
>+  std::unique_ptr<VideoCodecTestFixture> fixture_;
>+  const size_t bitrate_;
>+  const VideoCodecType codec_type_;
>+  const bool hw_codec_;
>+};
>+
>+INSTANTIATE_TEST_CASE_P(CodecSettings,
>+                        VideoCodecTestParameterized,
>+                        ::testing::Combine(::testing::ValuesIn(kBitrates),
>+                                           ::testing::ValuesIn(kVideoCodecType),
>+                                           ::testing::ValuesIn(kHwCodec)));
>+
>+TEST_P(VideoCodecTestParameterized, Foreman_352x288_30) {
>+  RunTest(352, 288, 30, "foreman_cif");
>+}
>+
>+TEST_P(VideoCodecTestParameterized, DISABLED_FourPeople_1280x720_30) {
>+  RunTest(1280, 720, 30, "FourPeople_1280x720_30");
>+}
>+
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_stats_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_stats_impl.cc
>new file mode 100644
>index 00000000000..37c38e8b425
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_stats_impl.cc
>@@ -0,0 +1,354 @@
>+/*
>+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/video_coding/codecs/test/videocodec_test_stats_impl.h"
>+
>+#include <algorithm>
>+#include <cmath>
>+#include <numeric>
>+
>+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>+#include "rtc_base/checks.h"
>+#include "test/statistics.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+using FrameStatistics = VideoCodecTestStats::FrameStatistics;
>+using VideoStatistics = VideoCodecTestStats::VideoStatistics;
>+
>+namespace {
>+const int kMaxBitrateMismatchPercent = 20;
>+}
>+
>+VideoCodecTestStatsImpl::VideoCodecTestStatsImpl() = default;
>+VideoCodecTestStatsImpl::~VideoCodecTestStatsImpl() = default;
>+
>+FrameStatistics* VideoCodecTestStatsImpl::AddFrame(size_t timestamp,
>+                                                   size_t layer_idx) {
>+  RTC_DCHECK(rtp_timestamp_to_frame_num_[layer_idx].find(timestamp) ==
>+             rtp_timestamp_to_frame_num_[layer_idx].end());
>+  const size_t frame_num = layer_stats_[layer_idx].size();
>+  rtp_timestamp_to_frame_num_[layer_idx][timestamp] = frame_num;
>+  layer_stats_[layer_idx].emplace_back(frame_num, timestamp);
>+  return &layer_stats_[layer_idx].back();
>+}
>+
>+FrameStatistics* VideoCodecTestStatsImpl::GetFrame(size_t frame_num,
>+                                                   size_t layer_idx) {
>+  RTC_CHECK_LT(frame_num, layer_stats_[layer_idx].size());
>+  return &layer_stats_[layer_idx][frame_num];
>+}
>+
>+FrameStatistics* VideoCodecTestStatsImpl::GetFrameWithTimestamp(
>+    size_t timestamp,
>+    size_t layer_idx) {
>+  RTC_DCHECK(rtp_timestamp_to_frame_num_[layer_idx].find(timestamp) !=
>+             rtp_timestamp_to_frame_num_[layer_idx].end());
>+
>+  return GetFrame(rtp_timestamp_to_frame_num_[layer_idx][timestamp], layer_idx);
>+}
>+
>+std::vector<VideoStatistics>
>+VideoCodecTestStatsImpl::SliceAndCalcLayerVideoStatistic(
>+    size_t first_frame_num,
>+    size_t last_frame_num) {
>+  std::vector<VideoStatistics> layer_stats;
>+
>+  size_t num_spatial_layers = 0;
>+  size_t num_temporal_layers = 0;
>+  GetNumberOfEncodedLayers(first_frame_num, last_frame_num, &num_spatial_layers,
>+                           &num_temporal_layers);
>+  RTC_CHECK_GT(num_spatial_layers, 0);
>+  RTC_CHECK_GT(num_temporal_layers, 0);
>+
>+  for (size_t spatial_idx = 0; spatial_idx < num_spatial_layers;
>+       ++spatial_idx) {
>+    for (size_t temporal_idx = 0; temporal_idx < num_temporal_layers;
>+         ++temporal_idx) {
>+      VideoStatistics layer_stat = SliceAndCalcVideoStatistic(
>+          first_frame_num, last_frame_num, spatial_idx, temporal_idx, false);
>+      layer_stats.push_back(layer_stat);
>+    }
>+  }
>+
>+  return layer_stats;
>+}
>+
>+VideoStatistics VideoCodecTestStatsImpl::SliceAndCalcAggregatedVideoStatistic(
>+    size_t first_frame_num,
>+    size_t last_frame_num) {
>+  size_t num_spatial_layers = 0;
>+  size_t num_temporal_layers = 0;
>+  GetNumberOfEncodedLayers(first_frame_num, last_frame_num, &num_spatial_layers,
>+                           &num_temporal_layers);
>+  RTC_CHECK_GT(num_spatial_layers, 0);
>+  RTC_CHECK_GT(num_temporal_layers, 0);
>+
>+  return SliceAndCalcVideoStatistic(first_frame_num, last_frame_num,
>+                                    num_spatial_layers - 1,
>+                                    num_temporal_layers - 1, true);
>+}
>+
>+void VideoCodecTestStatsImpl::PrintFrameStatistics() {
>+  for (size_t frame_num = 0; frame_num < layer_stats_[0].size(); ++frame_num) {
>+    for (const auto& it : layer_stats_) {
>+      const FrameStatistics& frame_stat = it.second[frame_num];
>+      printf("\n%s", frame_stat.ToString().c_str());
>+    }
>+  }
>+}
>+
>+size_t VideoCodecTestStatsImpl::Size(size_t spatial_idx) {
>+  return layer_stats_[spatial_idx].size();
>+}
>+
>+void VideoCodecTestStatsImpl::Clear() {
>+  layer_stats_.clear();
>+  rtp_timestamp_to_frame_num_.clear();
>+}
>+
>+FrameStatistics VideoCodecTestStatsImpl::AggregateFrameStatistic(
>+    size_t frame_num,
>+    size_t spatial_idx,
>+    bool aggregate_independent_layers) {
>+  FrameStatistics frame_stat = *GetFrame(frame_num, spatial_idx);
>+  bool inter_layer_predicted = frame_stat.inter_layer_predicted;
>+  while (spatial_idx-- > 0) {
>+    if (aggregate_independent_layers || inter_layer_predicted) {
>+      FrameStatistics* base_frame_stat = GetFrame(frame_num, spatial_idx);
>+      frame_stat.length_bytes += base_frame_stat->length_bytes;
>+      frame_stat.target_bitrate_kbps += base_frame_stat->target_bitrate_kbps;
>+
>+      inter_layer_predicted = base_frame_stat->inter_layer_predicted;
>+    }
>+  }
>+
>+  return frame_stat;
>+}
>+
>+size_t VideoCodecTestStatsImpl::CalcLayerTargetBitrateKbps(
>+    size_t first_frame_num,
>+    size_t last_frame_num,
>+    size_t spatial_idx,
>+    size_t temporal_idx,
>+    bool aggregate_independent_layers) {
>+  size_t target_bitrate_kbps = 0;
>+
>+  // We don't know if superframe includes all required spatial layers because
>+  // of possible frame drops. Run through all frames in specified range, find
>+  // and return maximum target bitrate. Assume that target bitrate in frame
>+  // statistic is specified per temporal layer.
>+  for (size_t frame_num = first_frame_num; frame_num <= last_frame_num;
>+       ++frame_num) {
>+    FrameStatistics superframe = AggregateFrameStatistic(
>+        frame_num, spatial_idx, aggregate_independent_layers);
>+
>+    if (superframe.temporal_idx <= temporal_idx) {
>+      target_bitrate_kbps =
>+          std::max(target_bitrate_kbps, superframe.target_bitrate_kbps);
>+    }
>+  }
>+
>+  RTC_DCHECK_GT(target_bitrate_kbps, 0);
>+  return target_bitrate_kbps;
>+}
>+
>+VideoStatistics VideoCodecTestStatsImpl::SliceAndCalcVideoStatistic(
>+    size_t first_frame_num,
>+    size_t last_frame_num,
>+    size_t spatial_idx,
>+    size_t temporal_idx,
>+    bool aggregate_independent_layers) {
>+  VideoStatistics video_stat;
>+
>+  float buffer_level_bits = 0.0f;
>+  Statistics buffer_level_sec;
>+
>+  Statistics key_frame_size_bytes;
>+  Statistics delta_frame_size_bytes;
>+
>+  Statistics frame_encoding_time_us;
>+  Statistics frame_decoding_time_us;
>+
>+  Statistics psnr_y;
>+  Statistics psnr_u;
>+  Statistics psnr_v;
>+  Statistics psnr;
>+  Statistics ssim;
>+  Statistics qp;
>+
>+  size_t rtp_timestamp_first_frame = 0;
>+  size_t rtp_timestamp_prev_frame = 0;
>+
>+  FrameStatistics last_successfully_decoded_frame(0, 0);
>+
>+  const size_t target_bitrate_kbps =
>+      CalcLayerTargetBitrateKbps(first_frame_num, last_frame_num, spatial_idx,
>+                                 temporal_idx, aggregate_independent_layers);
>+
>+  for (size_t frame_num = first_frame_num; frame_num <= last_frame_num;
>+       ++frame_num) {
>+    FrameStatistics frame_stat = AggregateFrameStatistic(
>+        frame_num, spatial_idx, aggregate_independent_layers);
>+
>+    float time_since_first_frame_sec =
>+        1.0f * (frame_stat.rtp_timestamp - rtp_timestamp_first_frame) /
>+        kVideoPayloadTypeFrequency;
>+    float time_since_prev_frame_sec =
>+        1.0f * (frame_stat.rtp_timestamp - rtp_timestamp_prev_frame) /
>+        kVideoPayloadTypeFrequency;
>+
>+    if (frame_stat.temporal_idx > temporal_idx) {
>+      continue;
>+    }
>+
>+    buffer_level_bits -= time_since_prev_frame_sec * 1000 * target_bitrate_kbps;
>+    buffer_level_bits = std::max(0.0f, buffer_level_bits);
>+    buffer_level_bits += 8.0 * frame_stat.length_bytes;
>+    buffer_level_sec.AddSample(buffer_level_bits /
>+                               (1000 * target_bitrate_kbps));
>+
>+    video_stat.length_bytes += frame_stat.length_bytes;
>+
>+    if (frame_stat.encoding_successful) {
>+      ++video_stat.num_encoded_frames;
>+
>+      if (frame_stat.frame_type == kVideoFrameKey) {
>+        key_frame_size_bytes.AddSample(frame_stat.length_bytes);
>+        ++video_stat.num_key_frames;
>+      } else {
>+        delta_frame_size_bytes.AddSample(frame_stat.length_bytes);
>+      }
>+
>+      frame_encoding_time_us.AddSample(frame_stat.encode_time_us);
>+      qp.AddSample(frame_stat.qp);
>+
>+      video_stat.max_nalu_size_bytes = std::max(video_stat.max_nalu_size_bytes,
>+                                                frame_stat.max_nalu_size_bytes);
>+    }
>+
>+    if (frame_stat.decoding_successful) {
>+      ++video_stat.num_decoded_frames;
>+
>+      video_stat.width = std::max(video_stat.width, frame_stat.decoded_width);
>+      video_stat.height =
>+          std::max(video_stat.height, frame_stat.decoded_height);
>+
>+      psnr_y.AddSample(frame_stat.psnr_y);
>+      psnr_u.AddSample(frame_stat.psnr_u);
>+      psnr_v.AddSample(frame_stat.psnr_v);
>+      psnr.AddSample(frame_stat.psnr);
>+      ssim.AddSample(frame_stat.ssim);
>+
>+      if (video_stat.num_decoded_frames > 1) {
>+        if (last_successfully_decoded_frame.decoded_width !=
>+                frame_stat.decoded_width ||
>+            last_successfully_decoded_frame.decoded_height !=
>+                frame_stat.decoded_height) {
>+          ++video_stat.num_spatial_resizes;
>+        }
>+      }
>+
>+      frame_decoding_time_us.AddSample(frame_stat.decode_time_us);
>+      last_successfully_decoded_frame = frame_stat;
>+    }
>+
>+    if (video_stat.num_input_frames > 0) {
>+      if (video_stat.time_to_reach_target_bitrate_sec == 0.0f) {
>+        const float curr_kbps =
>+            8.0 * video_stat.length_bytes / 1000 / time_since_first_frame_sec;
>+        const float bitrate_mismatch_percent =
>+            100 * std::fabs(curr_kbps - target_bitrate_kbps) /
>+            target_bitrate_kbps;
>+        if (bitrate_mismatch_percent < kMaxBitrateMismatchPercent) {
>+          video_stat.time_to_reach_target_bitrate_sec =
>+              time_since_first_frame_sec;
>+        }
>+      }
>+    }
>+
>+    rtp_timestamp_prev_frame = frame_stat.rtp_timestamp;
>+    if (video_stat.num_input_frames == 0) {
>+      rtp_timestamp_first_frame = frame_stat.rtp_timestamp;
>+    }
>+
>+    ++video_stat.num_input_frames;
>+  }
>+
>+  const size_t num_frames = last_frame_num - first_frame_num + 1;
>+  const size_t timestamp_delta =
>+      GetFrame(first_frame_num + 1, spatial_idx)->rtp_timestamp -
>+      GetFrame(first_frame_num, spatial_idx)->rtp_timestamp;
>+  const float input_framerate_fps =
>+      1.0 * kVideoPayloadTypeFrequency / timestamp_delta;
>+  const float duration_sec = num_frames / input_framerate_fps;
>+
>+  video_stat.target_bitrate_kbps = target_bitrate_kbps;
>+  video_stat.input_framerate_fps = input_framerate_fps;
>+
>+  video_stat.spatial_idx = spatial_idx;
>+  video_stat.temporal_idx = temporal_idx;
>+
>+  video_stat.bitrate_kbps =
>+      static_cast<size_t>(8 * video_stat.length_bytes / 1000 / duration_sec);
>+  video_stat.framerate_fps = video_stat.num_encoded_frames / duration_sec;
>+
>+  video_stat.enc_speed_fps = 1000000 / frame_encoding_time_us.Mean();
>+  video_stat.dec_speed_fps = 1000000 / frame_decoding_time_us.Mean();
>+
>+  video_stat.avg_delay_sec = buffer_level_sec.Mean();
>+  video_stat.max_key_frame_delay_sec =
>+      8 * key_frame_size_bytes.Max() / 1000 / target_bitrate_kbps;
>+  video_stat.max_delta_frame_delay_sec =
>+      8 * delta_frame_size_bytes.Max() / 1000 / target_bitrate_kbps;
>+
>+  video_stat.avg_key_frame_size_bytes = key_frame_size_bytes.Mean();
>+  video_stat.avg_delta_frame_size_bytes = delta_frame_size_bytes.Mean();
>+  video_stat.avg_qp = qp.Mean();
>+
>+  video_stat.avg_psnr_y = psnr_y.Mean();
>+  video_stat.avg_psnr_u = psnr_u.Mean();
>+  video_stat.avg_psnr_v = psnr_v.Mean();
>+  video_stat.avg_psnr = psnr.Mean();
>+  video_stat.min_psnr = psnr.Min();
>+  video_stat.avg_ssim = ssim.Mean();
>+  video_stat.min_ssim = ssim.Min();
>+
>+  return video_stat;
>+}
>+
>+void VideoCodecTestStatsImpl::GetNumberOfEncodedLayers(
>+    size_t first_frame_num,
>+    size_t last_frame_num,
>+    size_t* num_encoded_spatial_layers,
>+    size_t* num_encoded_temporal_layers) {
>+  *num_encoded_spatial_layers = 0;
>+  *num_encoded_temporal_layers = 0;
>+
>+  const size_t num_spatial_layers = layer_stats_.size();
>+
>+  for (size_t frame_num = first_frame_num; frame_num <= last_frame_num;
>+       ++frame_num) {
>+    for (size_t spatial_idx = 0; spatial_idx < num_spatial_layers;
>+         ++spatial_idx) {
>+      FrameStatistics* frame_stat = GetFrame(frame_num, spatial_idx);
>+      if (frame_stat->encoding_successful) {
>+        *num_encoded_spatial_layers =
>+            std::max(*num_encoded_spatial_layers, frame_stat->spatial_idx + 1);
>+        *num_encoded_temporal_layers = std::max(*num_encoded_temporal_layers,
>+                                                frame_stat->temporal_idx + 1);
>+      }
>+    }
>+  }
>+}
>+
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_stats_impl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_stats_impl.h
>new file mode 100644
>index 00000000000..a2b2a0a79bc
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_stats_impl.h
>@@ -0,0 +1,85 @@
>+/*
>+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_VIDEO_CODING_CODECS_TEST_VIDEOCODEC_TEST_STATS_IMPL_H_
>+#define MODULES_VIDEO_CODING_CODECS_TEST_VIDEOCODEC_TEST_STATS_IMPL_H_
>+
>+#include <map>
>+#include <string>
>+#include <vector>
>+
>+#include "api/test/videocodec_test_stats.h"  // NOLINT(build/include)
>+#include "common_types.h"                    // NOLINT(build/include)
>+
>+namespace webrtc {
>+namespace test {
>+
>+// Statistics for a sequence of processed frames. This class is not thread safe.
>+class VideoCodecTestStatsImpl : public VideoCodecTestStats {
>+ public:
>+  VideoCodecTestStatsImpl();
>+  ~VideoCodecTestStatsImpl() override;
>+
>+  // Creates a FrameStatistics for the next frame to be processed.
>+  FrameStatistics* AddFrame(size_t timestamp, size_t spatial_idx) override;
>+
>+  // Returns the FrameStatistics corresponding to |frame_number| or |timestamp|.
>+  FrameStatistics* GetFrame(size_t frame_number, size_t spatial_idx) override;
>+  FrameStatistics* GetFrameWithTimestamp(size_t timestamp,
>+                                         size_t spatial_idx) override;
>+
>+  std::vector<VideoStatistics> SliceAndCalcLayerVideoStatistic(
>+      size_t first_frame_num,
>+      size_t last_frame_num) override;
>+
>+  VideoStatistics SliceAndCalcAggregatedVideoStatistic(
>+      size_t first_frame_num,
>+      size_t last_frame_num) override;
>+
>+  void PrintFrameStatistics() override;
>+
>+  size_t Size(size_t spatial_idx) override;
>+
>+  void Clear() override;
>+
>+ private:
>+  VideoCodecTestStats::FrameStatistics AggregateFrameStatistic(
>+      size_t frame_num,
>+      size_t spatial_idx,
>+      bool aggregate_independent_layers);
>+
>+  size_t CalcLayerTargetBitrateKbps(size_t first_frame_num,
>+                                    size_t last_frame_num,
>+                                    size_t spatial_idx,
>+                                    size_t temporal_idx,
>+                                    bool aggregate_independent_layers);
>+
>+  VideoCodecTestStats::VideoStatistics SliceAndCalcVideoStatistic(
>+      size_t first_frame_num,
>+      size_t last_frame_num,
>+      size_t spatial_idx,
>+      size_t temporal_idx,
>+      bool aggregate_independent_layers);
>+
>+  void GetNumberOfEncodedLayers(size_t first_frame_num,
>+                                size_t last_frame_num,
>+                                size_t* num_encoded_spatial_layers,
>+                                size_t* num_encoded_temporal_layers);
>+
>+  // layer_idx -> stats.
>+  std::map<size_t, std::vector<FrameStatistics>> layer_stats_;
>+  // layer_idx -> rtp_timestamp -> frame_num.
>+  std::map<size_t, std::map<size_t, size_t>> rtp_timestamp_to_frame_num_;
>+};
>+
>+}  // namespace test
>+}  // namespace webrtc
>+
>+#endif  // MODULES_VIDEO_CODING_CODECS_TEST_VIDEOCODEC_TEST_STATS_IMPL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_stats_impl_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_stats_impl_unittest.cc
>new file mode 100644
>index 00000000000..145b2514cec
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_stats_impl_unittest.cc
>@@ -0,0 +1,66 @@
>+/*
>+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/video_coding/codecs/test/videocodec_test_stats_impl.h"
>+
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+namespace test {
>+using FrameStatistics = VideoCodecTestStatsImpl::FrameStatistics;
>+namespace {
>+const size_t kTimestamp = 12345;
>+}  // namespace
>+
>+TEST(StatsTest, AddFrame) {
>+  VideoCodecTestStatsImpl stats;
>+  FrameStatistics* frame_stat = stats.AddFrame(kTimestamp, 0);
>+  EXPECT_EQ(0ull, frame_stat->frame_number);
>+  EXPECT_EQ(kTimestamp, frame_stat->rtp_timestamp);
>+  EXPECT_EQ(1u, stats.Size(0));
>+}
>+
>+TEST(StatsTest, GetFrame) {
>+  VideoCodecTestStatsImpl stats;
>+  stats.AddFrame(kTimestamp, 0);
>+  FrameStatistics* frame_stat = stats.GetFrame(0u, 0);
>+  EXPECT_EQ(0u, frame_stat->frame_number);
>+  EXPECT_EQ(kTimestamp, frame_stat->rtp_timestamp);
>+}
>+
>+TEST(StatsTest, AddFrames) {
>+  VideoCodecTestStatsImpl stats;
>+  const size_t kNumFrames = 1000;
>+  for (size_t i = 0; i < kNumFrames; ++i) {
>+    FrameStatistics* frame_stat = stats.AddFrame(kTimestamp + i, 0);
>+    EXPECT_EQ(i, frame_stat->frame_number);
>+    EXPECT_EQ(kTimestamp + i, frame_stat->rtp_timestamp);
>+  }
>+  EXPECT_EQ(kNumFrames, stats.Size(0));
>+  // Get frame.
>+  size_t i = 22;
>+  FrameStatistics* frame_stat = stats.GetFrameWithTimestamp(kTimestamp + i, 0);
>+  EXPECT_EQ(i, frame_stat->frame_number);
>+  EXPECT_EQ(kTimestamp + i, frame_stat->rtp_timestamp);
>+}
>+
>+TEST(StatsTest, AddFrameLayering) {
>+  VideoCodecTestStatsImpl stats;
>+  for (size_t i = 0; i < 3; ++i) {
>+    stats.AddFrame(kTimestamp + i, i);
>+    FrameStatistics* frame_stat = stats.GetFrame(0u, i);
>+    EXPECT_EQ(0u, frame_stat->frame_number);
>+    EXPECT_EQ(kTimestamp, frame_stat->rtp_timestamp - i);
>+    EXPECT_EQ(1u, stats.Size(i));
>+  }
>+}
>+
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_videotoolbox.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_videotoolbox.cc
>new file mode 100644
>index 00000000000..f31ff548a81
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videocodec_test_videotoolbox.cc
>@@ -0,0 +1,89 @@
>+/*
>+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <vector>
>+
>+#include "api/test/create_videocodec_test_fixture.h"
>+#include "media/base/mediaconstants.h"
>+#include "modules/video_coding/codecs/test/objc_codec_factory_helper.h"
>+#include "modules/video_coding/codecs/test/videocodec_test_fixture_impl.h"
>+#include "test/gtest.h"
>+#include "test/testsupport/fileutils.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+namespace {
>+const int kForemanNumFrames = 300;
>+
>+VideoCodecTestFixture::Config CreateConfig() {
>+  VideoCodecTestFixture::Config config;
>+  config.filename = "foreman_cif";
>+  config.filepath = ResourcePath(config.filename, "yuv");
>+  config.num_frames = kForemanNumFrames;
>+  config.hw_encoder = true;
>+  config.hw_decoder = true;
>+  return config;
>+}
>+
>+std::unique_ptr<VideoCodecTestFixture> CreateTestFixtureWithConfig(
>+    VideoCodecTestFixture::Config config) {
>+  auto decoder_factory = CreateObjCDecoderFactory();
>+  auto encoder_factory = CreateObjCEncoderFactory();
>+  return CreateVideoCodecTestFixture(config, std::move(decoder_factory),
>+                                     std::move(encoder_factory));
>+}
>+}  // namespace
>+
>+// TODO(webrtc:9099): Disabled until the issue is fixed.
>+// HW codecs don't work on simulators. Only run these tests on device.
>+// #if TARGET_OS_IPHONE && !TARGET_IPHONE_SIMULATOR
>+// #define MAYBE_TEST TEST
>+// #else
>+#define MAYBE_TEST(s, name) TEST(s, DISABLED_##name)
>+// #endif
>+
>+// TODO(kthelgason): Use RC Thresholds when the internal bitrateAdjuster is no
>+// longer in use.
>+MAYBE_TEST(VideoCodecTestVideoToolbox, ForemanCif500kbpsH264CBP) {
>+  const auto frame_checker =
>+      absl::make_unique<VideoCodecTestFixtureImpl::H264KeyframeChecker>();
>+  auto config = CreateConfig();
>+  config.SetCodecSettings(cricket::kH264CodecName, 1, 1, 1, false, false, false,
>+                          352, 288);
>+  config.encoded_frame_checker = frame_checker.get();
>+  auto fixture = CreateTestFixtureWithConfig(config);
>+
>+  std::vector<RateProfile> rate_profiles = {{500, 30, kForemanNumFrames}};
>+
>+  std::vector<QualityThresholds> quality_thresholds = {{33, 29, 0.9, 0.82}};
>+
>+  fixture->RunTest(rate_profiles, nullptr, &quality_thresholds, nullptr);
>+}
>+
>+MAYBE_TEST(VideoCodecTestVideoToolbox, ForemanCif500kbpsH264CHP) {
>+  const auto frame_checker =
>+      absl::make_unique<VideoCodecTestFixtureImpl::H264KeyframeChecker>();
>+  auto config = CreateConfig();
>+  config.h264_codec_settings.profile = H264::kProfileConstrainedHigh;
>+  config.SetCodecSettings(cricket::kH264CodecName, 1, 1, 1, false, false, false,
>+                          352, 288);
>+  config.encoded_frame_checker = frame_checker.get();
>+  auto fixture = CreateTestFixtureWithConfig(config);
>+
>+  std::vector<RateProfile> rate_profiles = {{500, 30, kForemanNumFrames}};
>+
>+  std::vector<QualityThresholds> quality_thresholds = {{33, 30, 0.91, 0.83}};
>+
>+  fixture->RunTest(rate_profiles, nullptr, &quality_thresholds, nullptr);
>+}
>+
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videoprocessor.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videoprocessor.cc
>index 06475e196db..5a1269c056d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videoprocessor.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videoprocessor.cc
>@@ -17,36 +17,31 @@
> #include "api/video/i420_buffer.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "common_video/h264/h264_common.h"
>-#include "modules/video_coding/codecs/vp8/simulcast_rate_allocator.h"
>+#include "common_video/libyuv/include/webrtc_libyuv.h"
>+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> #include "modules/video_coding/include/video_codec_initializer.h"
>+#include "modules/video_coding/include/video_error_codes.h"
> #include "modules/video_coding/utility/default_video_bitrate_allocator.h"
>+#include "modules/video_coding/utility/simulcast_rate_allocator.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/timeutils.h"
> #include "test/gtest.h"
>+#include "third_party/libyuv/include/libyuv/compare.h"
>+#include "third_party/libyuv/include/libyuv/scale.h"
> 
> namespace webrtc {
> namespace test {
> 
>-namespace {
>-
>-const int kRtpClockRateHz = 90000;
>+using FrameStatistics = VideoCodecTestStats::FrameStatistics;
> 
>-std::unique_ptr<VideoBitrateAllocator> CreateBitrateAllocator(
>-    TestConfig* config) {
>-  std::unique_ptr<TemporalLayersFactory> tl_factory;
>-  if (config->codec_settings.codecType == VideoCodecType::kVideoCodecVP8) {
>-    tl_factory.reset(new TemporalLayersFactory());
>-    config->codec_settings.VP8()->tl_factory = tl_factory.get();
>-  }
>-  return std::unique_ptr<VideoBitrateAllocator>(
>-      VideoCodecInitializer::CreateBitrateAllocator(config->codec_settings,
>-                                                    std::move(tl_factory)));
>-}
>+namespace {
>+const int kMsToRtpTimestamp = kVideoPayloadTypeFrequency / 1000;
>+const int kMaxBufferedInputFrames = 20;
> 
>-rtc::Optional<size_t> GetMaxNaluLength(const EncodedImage& encoded_frame,
>-                                       const TestConfig& config) {
>+size_t GetMaxNaluSizeBytes(const EncodedImage& encoded_frame,
>+                           const VideoCodecTestFixture::Config& config) {
>   if (config.codec_settings.codecType != kVideoCodecH264)
>-    return rtc::nullopt;
>+    return 0;
> 
>   std::vector<webrtc::H264::NaluIndex> nalu_indices =
>       webrtc::H264::FindNaluIndices(encoded_frame._buffer,
>@@ -54,11 +49,29 @@ rtc::Optional<size_t> GetMaxNaluLength(const EncodedImage& encoded_frame,
> 
>   RTC_CHECK(!nalu_indices.empty());
> 
>-  size_t max_length = 0;
>+  size_t max_size = 0;
>   for (const webrtc::H264::NaluIndex& index : nalu_indices)
>-    max_length = std::max(max_length, index.payload_size);
>+    max_size = std::max(max_size, index.payload_size);
> 
>-  return max_length;
>+  return max_size;
>+}
>+
>+void GetLayerIndices(const CodecSpecificInfo& codec_specific,
>+                     size_t* spatial_idx,
>+                     size_t* temporal_idx) {
>+  if (codec_specific.codecType == kVideoCodecVP8) {
>+    *spatial_idx = codec_specific.codecSpecific.VP8.simulcastIdx;
>+    *temporal_idx = codec_specific.codecSpecific.VP8.temporalIdx;
>+  } else if (codec_specific.codecType == kVideoCodecVP9) {
>+    *spatial_idx = codec_specific.codecSpecific.VP9.spatial_idx;
>+    *temporal_idx = codec_specific.codecSpecific.VP9.temporal_idx;
>+  }
>+  if (*spatial_idx == kNoSpatialIdx) {
>+    *spatial_idx = 0;
>+  }
>+  if (*temporal_idx == kNoTemporalIdx) {
>+    *temporal_idx = 0;
>+  }
> }
> 
> int GetElapsedTimeMicroseconds(int64_t start_ns, int64_t stop_ns) {
>@@ -68,10 +81,10 @@ int GetElapsedTimeMicroseconds(int64_t start_ns, int64_t stop_ns) {
>   return static_cast<int>(diff_us);
> }
> 
>-void ExtractBufferWithSize(const VideoFrame& image,
>-                           int width,
>-                           int height,
>-                           rtc::Buffer* buffer) {
>+void ExtractI420BufferWithSize(const VideoFrame& image,
>+                               int width,
>+                               int height,
>+                               rtc::Buffer* buffer) {
>   if (image.width() != width || image.height() != height) {
>     EXPECT_DOUBLE_EQ(static_cast<double>(width) / height,
>                      static_cast<double>(image.width()) / image.height());
>@@ -93,284 +106,485 @@ void ExtractBufferWithSize(const VideoFrame& image,
>   RTC_CHECK_NE(ExtractBuffer(image, length, buffer->data()), -1);
> }
> 
>+void CalculateFrameQuality(const I420BufferInterface& ref_buffer,
>+                           const I420BufferInterface& dec_buffer,
>+                           FrameStatistics* frame_stat) {
>+  if (ref_buffer.width() != dec_buffer.width() ||
>+      ref_buffer.height() != dec_buffer.height()) {
>+    RTC_CHECK_GE(ref_buffer.width(), dec_buffer.width());
>+    RTC_CHECK_GE(ref_buffer.height(), dec_buffer.height());
>+    // Downscale reference frame.
>+    rtc::scoped_refptr<I420Buffer> scaled_buffer =
>+        I420Buffer::Create(dec_buffer.width(), dec_buffer.height());
>+    I420Scale(ref_buffer.DataY(), ref_buffer.StrideY(), ref_buffer.DataU(),
>+              ref_buffer.StrideU(), ref_buffer.DataV(), ref_buffer.StrideV(),
>+              ref_buffer.width(), ref_buffer.height(),
>+              scaled_buffer->MutableDataY(), scaled_buffer->StrideY(),
>+              scaled_buffer->MutableDataU(), scaled_buffer->StrideU(),
>+              scaled_buffer->MutableDataV(), scaled_buffer->StrideV(),
>+              scaled_buffer->width(), scaled_buffer->height(),
>+              libyuv::kFilterBox);
>+
>+    CalculateFrameQuality(*scaled_buffer, dec_buffer, frame_stat);
>+  } else {
>+    const uint64_t sse_y = libyuv::ComputeSumSquareErrorPlane(
>+        dec_buffer.DataY(), dec_buffer.StrideY(), ref_buffer.DataY(),
>+        ref_buffer.StrideY(), dec_buffer.width(), dec_buffer.height());
>+
>+    const uint64_t sse_u = libyuv::ComputeSumSquareErrorPlane(
>+        dec_buffer.DataU(), dec_buffer.StrideU(), ref_buffer.DataU(),
>+        ref_buffer.StrideU(), dec_buffer.width() / 2, dec_buffer.height() / 2);
>+
>+    const uint64_t sse_v = libyuv::ComputeSumSquareErrorPlane(
>+        dec_buffer.DataV(), dec_buffer.StrideV(), ref_buffer.DataV(),
>+        ref_buffer.StrideV(), dec_buffer.width() / 2, dec_buffer.height() / 2);
>+
>+    const size_t num_y_samples = dec_buffer.width() * dec_buffer.height();
>+    const size_t num_u_samples =
>+        dec_buffer.width() / 2 * dec_buffer.height() / 2;
>+
>+    frame_stat->psnr_y = libyuv::SumSquareErrorToPsnr(sse_y, num_y_samples);
>+    frame_stat->psnr_u = libyuv::SumSquareErrorToPsnr(sse_u, num_u_samples);
>+    frame_stat->psnr_v = libyuv::SumSquareErrorToPsnr(sse_v, num_u_samples);
>+    frame_stat->psnr = libyuv::SumSquareErrorToPsnr(
>+        sse_y + sse_u + sse_v, num_y_samples + 2 * num_u_samples);
>+    frame_stat->ssim = I420SSIM(ref_buffer, dec_buffer);
>+  }
>+}
>+
>+std::vector<FrameType> FrameTypeForFrame(
>+    const VideoCodecTestFixture::Config& config,
>+    size_t frame_idx) {
>+  if (config.keyframe_interval > 0 &&
>+      (frame_idx % config.keyframe_interval == 0)) {
>+    return {kVideoFrameKey};
>+  }
>+  return {kVideoFrameDelta};
>+}
>+
> }  // namespace
> 
> VideoProcessor::VideoProcessor(webrtc::VideoEncoder* encoder,
>-                               webrtc::VideoDecoder* decoder,
>-                               FrameReader* analysis_frame_reader,
>-                               PacketManipulator* packet_manipulator,
>-                               const TestConfig& config,
>-                               Stats* stats,
>-                               IvfFileWriter* encoded_frame_writer,
>-                               FrameWriter* decoded_frame_writer)
>+                               VideoDecoderList* decoders,
>+                               FrameReader* input_frame_reader,
>+                               const VideoCodecTestFixture::Config& config,
>+                               VideoCodecTestStats* stats,
>+                               IvfFileWriterList* encoded_frame_writers,
>+                               FrameWriterList* decoded_frame_writers)
>     : config_(config),
>+      num_simulcast_or_spatial_layers_(
>+          std::max(config_.NumberOfSimulcastStreams(),
>+                   config_.NumberOfSpatialLayers())),
>+      stats_(stats),
>       encoder_(encoder),
>-      decoder_(decoder),
>-      bitrate_allocator_(CreateBitrateAllocator(&config_)),
>+      decoders_(decoders),
>+      bitrate_allocator_(VideoCodecInitializer::CreateBitrateAllocator(
>+          config_.codec_settings)),
>+      framerate_fps_(0),
>       encode_callback_(this),
>-      decode_callback_(this),
>-      packet_manipulator_(packet_manipulator),
>-      analysis_frame_reader_(analysis_frame_reader),
>-      encoded_frame_writer_(encoded_frame_writer),
>-      decoded_frame_writer_(decoded_frame_writer),
>-      last_inputed_frame_num_(-1),
>-      last_encoded_frame_num_(-1),
>-      last_decoded_frame_num_(-1),
>-      first_key_frame_has_been_excluded_(false),
>-      last_decoded_frame_buffer_(analysis_frame_reader->FrameLength()),
>-      stats_(stats),
>-      rate_update_index_(-1) {
>-  RTC_DCHECK(encoder);
>-  RTC_DCHECK(decoder);
>-  RTC_DCHECK(packet_manipulator);
>-  RTC_DCHECK(analysis_frame_reader);
>-  RTC_DCHECK(stats);
>-
>-  // Setup required callbacks for the encoder and decoder.
>+      input_frame_reader_(input_frame_reader),
>+      merged_encoded_frames_(num_simulcast_or_spatial_layers_),
>+      encoded_frame_writers_(encoded_frame_writers),
>+      decoded_frame_writers_(decoded_frame_writers),
>+      last_inputed_frame_num_(0),
>+      last_inputed_timestamp_(0),
>+      first_encoded_frame_(num_simulcast_or_spatial_layers_, true),
>+      last_encoded_frame_num_(num_simulcast_or_spatial_layers_),
>+      first_decoded_frame_(num_simulcast_or_spatial_layers_, true),
>+      last_decoded_frame_num_(num_simulcast_or_spatial_layers_),
>+      decoded_frame_buffer_(num_simulcast_or_spatial_layers_),
>+      post_encode_time_ns_(0) {
>+  // Sanity checks.
>+  RTC_CHECK(rtc::TaskQueue::Current())
>+      << "VideoProcessor must be run on a task queue.";
>+  RTC_CHECK(encoder);
>+  RTC_CHECK(decoders);
>+  RTC_CHECK_EQ(decoders->size(), num_simulcast_or_spatial_layers_);
>+  RTC_CHECK(input_frame_reader);
>+  RTC_CHECK(stats);
>+  RTC_CHECK(!encoded_frame_writers ||
>+            encoded_frame_writers->size() == num_simulcast_or_spatial_layers_);
>+  RTC_CHECK(!decoded_frame_writers ||
>+            decoded_frame_writers->size() == num_simulcast_or_spatial_layers_);
>+
>+  // Setup required callbacks for the encoder and decoder and initialize them.
>   RTC_CHECK_EQ(encoder_->RegisterEncodeCompleteCallback(&encode_callback_),
>                WEBRTC_VIDEO_CODEC_OK);
>-  RTC_CHECK_EQ(decoder_->RegisterDecodeCompleteCallback(&decode_callback_),
>+
>+  // Initialize codecs so that they are ready to receive frames.
>+  RTC_CHECK_EQ(encoder_->InitEncode(&config_.codec_settings,
>+                                    static_cast<int>(config_.NumberOfCores()),
>+                                    config_.max_payload_size_bytes),
>                WEBRTC_VIDEO_CODEC_OK);
> 
>-  // Initialize the encoder and decoder.
>-  RTC_CHECK_EQ(
>-      encoder_->InitEncode(&config_.codec_settings, config_.NumberOfCores(),
>-                           config_.networking_config.max_payload_size_in_bytes),
>-      WEBRTC_VIDEO_CODEC_OK);
>-  RTC_CHECK_EQ(
>-      decoder_->InitDecode(&config_.codec_settings, config_.NumberOfCores()),
>-      WEBRTC_VIDEO_CODEC_OK);
>+  for (size_t i = 0; i < num_simulcast_or_spatial_layers_; ++i) {
>+    decode_callback_.push_back(
>+        absl::make_unique<VideoProcessorDecodeCompleteCallback>(this, i));
>+    RTC_CHECK_EQ(
>+        decoders_->at(i)->InitDecode(&config_.codec_settings,
>+                                     static_cast<int>(config_.NumberOfCores())),
>+        WEBRTC_VIDEO_CODEC_OK);
>+    RTC_CHECK_EQ(decoders_->at(i)->RegisterDecodeCompleteCallback(
>+                     decode_callback_.at(i).get()),
>+                 WEBRTC_VIDEO_CODEC_OK);
>+  }
> }
> 
> VideoProcessor::~VideoProcessor() {
>   RTC_DCHECK_CALLED_SEQUENTIALLY(&sequence_checker_);
> 
>+  // Explicitly reset codecs, in case they don't do that themselves when they
>+  // go out of scope.
>   RTC_CHECK_EQ(encoder_->Release(), WEBRTC_VIDEO_CODEC_OK);
>-  RTC_CHECK_EQ(decoder_->Release(), WEBRTC_VIDEO_CODEC_OK);
>-
>   encoder_->RegisterEncodeCompleteCallback(nullptr);
>-  decoder_->RegisterDecodeCompleteCallback(nullptr);
>+  for (auto& decoder : *decoders_) {
>+    RTC_CHECK_EQ(decoder->Release(), WEBRTC_VIDEO_CODEC_OK);
>+    decoder->RegisterDecodeCompleteCallback(nullptr);
>+  }
>+
>+  // Sanity check.
>+  RTC_CHECK_LE(input_frames_.size(), kMaxBufferedInputFrames);
>+
>+  // Deal with manual memory management of EncodedImage's.
>+  for (size_t i = 0; i < num_simulcast_or_spatial_layers_; ++i) {
>+    uint8_t* buffer = merged_encoded_frames_.at(i)._buffer;
>+    if (buffer) {
>+      delete[] buffer;
>+    }
>+  }
> }
> 
> void VideoProcessor::ProcessFrame() {
>   RTC_DCHECK_CALLED_SEQUENTIALLY(&sequence_checker_);
>-  const int frame_number = ++last_inputed_frame_num_;
>+  const size_t frame_number = last_inputed_frame_num_++;
> 
>-  // Get frame from file.
>-  rtc::scoped_refptr<I420BufferInterface> buffer(
>-      analysis_frame_reader_->ReadFrame());
>+  // Get input frame and store for future quality calculation.
>+  rtc::scoped_refptr<I420BufferInterface> buffer =
>+      input_frame_reader_->ReadFrame();
>   RTC_CHECK(buffer) << "Tried to read too many frames from the file.";
>-  // Use the frame number as the basis for timestamp to identify frames. Let the
>-  // first timestamp be non-zero, to not make the IvfFileWriter believe that we
>-  // want to use capture timestamps in the IVF files.
>-  const uint32_t rtp_timestamp = (frame_number + 1) * kRtpClockRateHz /
>-                                 config_.codec_settings.maxFramerate;
>-  const int64_t render_time_ms = (frame_number + 1) * rtc::kNumMillisecsPerSec /
>-                                 config_.codec_settings.maxFramerate;
>-  rtp_timestamp_to_frame_num_[rtp_timestamp] = frame_number;
>-  input_frames_[frame_number] = rtc::MakeUnique<VideoFrame>(
>-      buffer, rtp_timestamp, render_time_ms, webrtc::kVideoRotation_0);
>-
>-  std::vector<FrameType> frame_types = config_.FrameTypeForFrame(frame_number);
>-
>-  // Create frame statistics object used for aggregation at end of test run.
>-  FrameStatistic* frame_stat = stats_->AddFrame();
>+  const size_t timestamp =
>+      last_inputed_timestamp_ + kVideoPayloadTypeFrequency / framerate_fps_;
>+  VideoFrame input_frame(buffer, static_cast<uint32_t>(timestamp),
>+                         static_cast<int64_t>(timestamp / kMsToRtpTimestamp),
>+                         webrtc::kVideoRotation_0);
>+  // Store input frame as a reference for quality calculations.
>+  if (config_.decode && !config_.measure_cpu) {
>+    if (input_frames_.size() == kMaxBufferedInputFrames) {
>+      input_frames_.erase(input_frames_.begin());
>+    }
>+    input_frames_.emplace(frame_number, input_frame);
>+  }
>+  last_inputed_timestamp_ = timestamp;
>+
>+  post_encode_time_ns_ = 0;
>+
>+  // Create frame statistics object for all simulcast/spatial layers.
>+  for (size_t i = 0; i < num_simulcast_or_spatial_layers_; ++i) {
>+    stats_->AddFrame(timestamp, i);
>+  }
> 
>   // For the highest measurement accuracy of the encode time, the start/stop
>   // time recordings should wrap the Encode call as tightly as possible.
>-  frame_stat->encode_start_ns = rtc::TimeNanos();
>-  frame_stat->encode_return_code =
>-      encoder_->Encode(*input_frames_[frame_number], nullptr, &frame_types);
>+  const int64_t encode_start_ns = rtc::TimeNanos();
>+  for (size_t i = 0; i < num_simulcast_or_spatial_layers_; ++i) {
>+    FrameStatistics* frame_stat = stats_->GetFrame(frame_number, i);
>+    frame_stat->encode_start_ns = encode_start_ns;
>+  }
>+
>+  // Encode.
>+  const std::vector<FrameType> frame_types =
>+      FrameTypeForFrame(config_, frame_number);
>+  const int encode_return_code =
>+      encoder_->Encode(input_frame, nullptr, &frame_types);
>+  for (size_t i = 0; i < num_simulcast_or_spatial_layers_; ++i) {
>+    FrameStatistics* frame_stat = stats_->GetFrame(frame_number, i);
>+    frame_stat->encode_return_code = encode_return_code;
>+  }
> }
> 
>-void VideoProcessor::SetRates(int bitrate_kbps, int framerate_fps) {
>+void VideoProcessor::SetRates(size_t bitrate_kbps, size_t framerate_fps) {
>   RTC_DCHECK_CALLED_SEQUENTIALLY(&sequence_checker_);
>-  config_.codec_settings.maxFramerate = framerate_fps;
>-  int set_rates_result = encoder_->SetRateAllocation(
>-      bitrate_allocator_->GetAllocation(bitrate_kbps * 1000, framerate_fps),
>-      framerate_fps);
>+  framerate_fps_ = static_cast<uint32_t>(framerate_fps);
>+  bitrate_allocation_ = bitrate_allocator_->GetAllocation(
>+      static_cast<uint32_t>(bitrate_kbps * 1000), framerate_fps_);
>+  const int set_rates_result =
>+      encoder_->SetRateAllocation(bitrate_allocation_, framerate_fps_);
>   RTC_DCHECK_GE(set_rates_result, 0)
>       << "Failed to update encoder with new rate " << bitrate_kbps << ".";
>-  ++rate_update_index_;
>-  num_dropped_frames_.push_back(0);
>-  num_spatial_resizes_.push_back(0);
>-}
>-
>-std::vector<int> VideoProcessor::NumberDroppedFramesPerRateUpdate() const {
>-  RTC_DCHECK_CALLED_SEQUENTIALLY(&sequence_checker_);
>-  return num_dropped_frames_;
> }
> 
>-std::vector<int> VideoProcessor::NumberSpatialResizesPerRateUpdate() const {
>-  RTC_DCHECK_CALLED_SEQUENTIALLY(&sequence_checker_);
>-  return num_spatial_resizes_;
>+int32_t VideoProcessor::VideoProcessorDecodeCompleteCallback::Decoded(
>+    VideoFrame& image) {
>+  // Post the callback to the right task queue, if needed.
>+  if (!task_queue_->IsCurrent()) {
>+    // There might be a limited amount of output buffers, make a copy to make
>+    // sure we don't block the decoder.
>+    VideoFrame copy(I420Buffer::Copy(*image.video_frame_buffer()->ToI420()),
>+                    image.rotation(), image.timestamp_us());
>+    copy.set_timestamp(image.timestamp());
>+
>+    task_queue_->PostTask([this, copy]() {
>+      video_processor_->FrameDecoded(copy, simulcast_svc_idx_);
>+    });
>+    return 0;
>+  }
>+  video_processor_->FrameDecoded(image, simulcast_svc_idx_);
>+  return 0;
> }
> 
>-void VideoProcessor::FrameEncoded(webrtc::VideoCodecType codec,
>-                                  const EncodedImage& encoded_image) {
>+void VideoProcessor::FrameEncoded(
>+    const webrtc::EncodedImage& encoded_image,
>+    const webrtc::CodecSpecificInfo& codec_specific) {
>   RTC_DCHECK_CALLED_SEQUENTIALLY(&sequence_checker_);
> 
>   // For the highest measurement accuracy of the encode time, the start/stop
>   // time recordings should wrap the Encode call as tightly as possible.
>-  int64_t encode_stop_ns = rtc::TimeNanos();
>+  const int64_t encode_stop_ns = rtc::TimeNanos();
> 
>+  const VideoCodecType codec_type = codec_specific.codecType;
>   if (config_.encoded_frame_checker) {
>-    config_.encoded_frame_checker->CheckEncodedFrame(codec, encoded_image);
>+    config_.encoded_frame_checker->CheckEncodedFrame(codec_type, encoded_image);
>   }
> 
>-  const int frame_number =
>-      rtp_timestamp_to_frame_num_[encoded_image._timeStamp];
>-
>-  // Ensure strict monotonicity.
>-  RTC_CHECK_GT(frame_number, last_encoded_frame_num_);
>-
>-  // Check for dropped frames.
>-  bool last_frame_missing = false;
>-  if (frame_number > 0) {
>-    int num_dropped_from_last_encode =
>-        frame_number - last_encoded_frame_num_ - 1;
>-    RTC_DCHECK_GE(num_dropped_from_last_encode, 0);
>-    RTC_CHECK_GE(rate_update_index_, 0);
>-    num_dropped_frames_[rate_update_index_] += num_dropped_from_last_encode;
>-    const FrameStatistic* last_encoded_frame_stat =
>-        stats_->GetFrame(last_encoded_frame_num_);
>-    last_frame_missing = (last_encoded_frame_stat->manipulated_length == 0);
>+  // Layer metadata.
>+  size_t spatial_idx = 0;
>+  size_t temporal_idx = 0;
>+  GetLayerIndices(codec_specific, &spatial_idx, &temporal_idx);
>+
>+  FrameStatistics* frame_stat =
>+      stats_->GetFrameWithTimestamp(encoded_image._timeStamp, spatial_idx);
>+  const size_t frame_number = frame_stat->frame_number;
>+
>+  // Ensure that the encode order is monotonically increasing, within this
>+  // simulcast/spatial layer.
>+  RTC_CHECK(first_encoded_frame_[spatial_idx] ||
>+            last_encoded_frame_num_[spatial_idx] < frame_number);
>+
>+  // Ensure SVC spatial layers are delivered in ascending order.
>+  if (!first_encoded_frame_[spatial_idx] &&
>+      config_.NumberOfSpatialLayers() > 1) {
>+    for (size_t i = 0; i < spatial_idx; ++i) {
>+      RTC_CHECK_LE(last_encoded_frame_num_[i], frame_number);
>+    }
>+    for (size_t i = spatial_idx + 1; i < num_simulcast_or_spatial_layers_;
>+         ++i) {
>+      RTC_CHECK_GT(frame_number, last_encoded_frame_num_[i]);
>+    }
>   }
>-  last_encoded_frame_num_ = frame_number;
>+  first_encoded_frame_[spatial_idx] = false;
>+  last_encoded_frame_num_[spatial_idx] = frame_number;
> 
>   // Update frame statistics.
>-  FrameStatistic* frame_stat = stats_->GetFrame(frame_number);
>-  frame_stat->encode_time_us =
>-      GetElapsedTimeMicroseconds(frame_stat->encode_start_ns, encode_stop_ns);
>   frame_stat->encoding_successful = true;
>-  frame_stat->encoded_frame_size_bytes = encoded_image._length;
>+  frame_stat->encode_time_us = GetElapsedTimeMicroseconds(
>+      frame_stat->encode_start_ns, encode_stop_ns - post_encode_time_ns_);
>+  frame_stat->target_bitrate_kbps =
>+      bitrate_allocation_.GetTemporalLayerSum(spatial_idx, temporal_idx) / 1000;
>+  frame_stat->length_bytes = encoded_image._length;
>   frame_stat->frame_type = encoded_image._frameType;
>+  frame_stat->temporal_idx = temporal_idx;
>+  frame_stat->spatial_idx = spatial_idx;
>+  frame_stat->max_nalu_size_bytes = GetMaxNaluSizeBytes(encoded_image, config_);
>   frame_stat->qp = encoded_image.qp_;
>-  frame_stat->bitrate_kbps = static_cast<int>(
>-      encoded_image._length * config_.codec_settings.maxFramerate * 8 / 1000);
>-  frame_stat->total_packets =
>-      encoded_image._length / config_.networking_config.packet_size_in_bytes +
>-      1;
>-  frame_stat->max_nalu_length = GetMaxNaluLength(encoded_image, config_);
>-
>-  // Make a raw copy of |encoded_image| to feed to the decoder.
>-  size_t copied_buffer_size = encoded_image._length +
>-                              EncodedImage::GetBufferPaddingBytes(codec);
>-  std::unique_ptr<uint8_t[]> copied_buffer(new uint8_t[copied_buffer_size]);
>-  memcpy(copied_buffer.get(), encoded_image._buffer, encoded_image._length);
>-  EncodedImage copied_image = encoded_image;
>-  copied_image._size = copied_buffer_size;
>-  copied_image._buffer = copied_buffer.get();
> 
>-  // Simulate packet loss.
>-  if (!ExcludeFrame(copied_image)) {
>-    frame_stat->packets_dropped =
>-        packet_manipulator_->ManipulatePackets(&copied_image);
>+  const size_t num_spatial_layers = config_.NumberOfSpatialLayers();
>+  bool end_of_picture = false;
>+  if (codec_type == kVideoCodecVP9) {
>+    const CodecSpecificInfoVP9& vp9_info = codec_specific.codecSpecific.VP9;
>+    frame_stat->inter_layer_predicted = vp9_info.inter_layer_predicted;
>+    frame_stat->non_ref_for_inter_layer_pred =
>+        vp9_info.non_ref_for_inter_layer_pred;
>+    end_of_picture = vp9_info.end_of_picture;
>+  } else {
>+    frame_stat->inter_layer_predicted = false;
>+    frame_stat->non_ref_for_inter_layer_pred = true;
>   }
>-  frame_stat->manipulated_length = copied_image._length;
> 
>-  // For the highest measurement accuracy of the decode time, the start/stop
>-  // time recordings should wrap the Decode call as tightly as possible.
>-  frame_stat->decode_start_ns = rtc::TimeNanos();
>-  frame_stat->decode_return_code =
>-      decoder_->Decode(copied_image, last_frame_missing, nullptr);
>+  const webrtc::EncodedImage* encoded_image_for_decode = &encoded_image;
>+  if (config_.decode || encoded_frame_writers_) {
>+    if (num_spatial_layers > 1) {
>+      encoded_image_for_decode = BuildAndStoreSuperframe(
>+          encoded_image, codec_type, frame_number, spatial_idx,
>+          frame_stat->inter_layer_predicted);
>+    }
>+  }
> 
>-  if (encoded_frame_writer_) {
>-    RTC_CHECK(encoded_frame_writer_->WriteFrame(encoded_image, codec));
>+  if (config_.decode) {
>+    DecodeFrame(*encoded_image_for_decode, spatial_idx);
>+
>+    if (end_of_picture && num_spatial_layers > 1) {
>+      // If inter-layer prediction is enabled and upper layer was dropped then
>+      // base layer should be passed to upper layer decoder. Otherwise decoder
>+      // won't be able to decode next superframe.
>+      const EncodedImage* base_image = nullptr;
>+      const FrameStatistics* base_stat = nullptr;
>+      for (size_t i = 0; i < num_spatial_layers; ++i) {
>+        const bool layer_dropped = (first_decoded_frame_[i] ||
>+                                    last_decoded_frame_num_[i] < frame_number);
>+
>+        // Ensure current layer was decoded.
>+        RTC_CHECK(layer_dropped == false || i != spatial_idx);
>+
>+        if (!layer_dropped) {
>+          base_image = &merged_encoded_frames_[i];
>+          base_stat =
>+              stats_->GetFrameWithTimestamp(encoded_image._timeStamp, i);
>+        } else if (base_image && !base_stat->non_ref_for_inter_layer_pred) {
>+          DecodeFrame(*base_image, i);
>+        }
>+      }
>+    }
>+  } else {
>+    frame_stat->decode_return_code = WEBRTC_VIDEO_CODEC_NO_OUTPUT;
>+  }
>+
>+  if (encoded_frame_writers_) {
>+    RTC_CHECK(encoded_frame_writers_->at(spatial_idx)
>+                  ->WriteFrame(*encoded_image_for_decode,
>+                               config_.codec_settings.codecType));
>+  }
>+
>+  if (!config_.IsAsyncCodec()) {
>+    // To get pure encode time for next layers, measure time spent in encode
>+    // callback and subtract it from encode time of next layers.
>+    post_encode_time_ns_ += rtc::TimeNanos() - encode_stop_ns;
>   }
> }
> 
>-void VideoProcessor::FrameDecoded(const VideoFrame& decoded_frame) {
>+void VideoProcessor::FrameDecoded(const VideoFrame& decoded_frame,
>+                                  size_t spatial_idx) {
>   RTC_DCHECK_CALLED_SEQUENTIALLY(&sequence_checker_);
> 
>   // For the highest measurement accuracy of the decode time, the start/stop
>   // time recordings should wrap the Decode call as tightly as possible.
>-  int64_t decode_stop_ns = rtc::TimeNanos();
>+  const int64_t decode_stop_ns = rtc::TimeNanos();
>+
>+  FrameStatistics* frame_stat =
>+      stats_->GetFrameWithTimestamp(decoded_frame.timestamp(), spatial_idx);
>+  const size_t frame_number = frame_stat->frame_number;
>+
>+  if (decoded_frame_writers_ && !first_decoded_frame_[spatial_idx]) {
>+    // Fill drops with last decoded frame to make them look like freeze at
>+    // playback and to keep decoded layers in sync.
>+    for (size_t i = last_decoded_frame_num_[spatial_idx] + 1; i < frame_number;
>+         ++i) {
>+      RTC_CHECK(decoded_frame_writers_->at(spatial_idx)
>+                    ->WriteFrame(decoded_frame_buffer_[spatial_idx].data()));
>+    }
>+  }
>+
>+  // Ensure that the decode order is monotonically increasing, within this
>+  // simulcast/spatial layer.
>+  RTC_CHECK(first_decoded_frame_[spatial_idx] ||
>+            last_decoded_frame_num_[spatial_idx] < frame_number);
>+  first_decoded_frame_[spatial_idx] = false;
>+  last_decoded_frame_num_[spatial_idx] = frame_number;
> 
>   // Update frame statistics.
>-  const int frame_number =
>-      rtp_timestamp_to_frame_num_[decoded_frame.timestamp()];
>-  FrameStatistic* frame_stat = stats_->GetFrame(frame_number);
>-  frame_stat->decoded_width = decoded_frame.width();
>-  frame_stat->decoded_height = decoded_frame.height();
>+  frame_stat->decoding_successful = true;
>   frame_stat->decode_time_us =
>       GetElapsedTimeMicroseconds(frame_stat->decode_start_ns, decode_stop_ns);
>-  frame_stat->decoding_successful = true;
>-
>-  // Ensure strict monotonicity.
>-  RTC_CHECK_GT(frame_number, last_decoded_frame_num_);
>-
>-  // Check if the codecs have resized the frame since previously decoded frame.
>-  if (frame_number > 0) {
>-    if (decoded_frame_writer_ && last_decoded_frame_num_ >= 0) {
>-      // For dropped/lost frames, write out the last decoded frame to make it
>-      // look like a freeze at playback.
>-      const int num_dropped_frames = frame_number - last_decoded_frame_num_;
>-      for (int i = 0; i < num_dropped_frames; i++) {
>-        WriteDecodedFrameToFile(&last_decoded_frame_buffer_);
>-      }
>-    }
>-    // TODO(ssilkin): move to FrameEncoded when webm:1474 is implemented.
>-    const FrameStatistic* last_decoded_frame_stat =
>-        stats_->GetFrame(last_decoded_frame_num_);
>-    if (decoded_frame.width() != last_decoded_frame_stat->decoded_width ||
>-        decoded_frame.height() != last_decoded_frame_stat->decoded_height) {
>-      RTC_CHECK_GE(rate_update_index_, 0);
>-      ++num_spatial_resizes_[rate_update_index_];
>-    }
>-  }
>-  last_decoded_frame_num_ = frame_number;
>+  frame_stat->decoded_width = decoded_frame.width();
>+  frame_stat->decoded_height = decoded_frame.height();
> 
>   // Skip quality metrics calculation to not affect CPU usage.
>   if (!config_.measure_cpu) {
>-    frame_stat->psnr =
>-        I420PSNR(input_frames_[frame_number].get(), &decoded_frame);
>-    frame_stat->ssim =
>-        I420SSIM(input_frames_[frame_number].get(), &decoded_frame);
>+    const auto reference_frame = input_frames_.find(frame_number);
>+    RTC_CHECK(reference_frame != input_frames_.cend())
>+        << "The codecs are either buffering too much, dropping too much, or "
>+           "being too slow relative the input frame rate.";
>+    CalculateFrameQuality(
>+        *reference_frame->second.video_frame_buffer()->ToI420(),
>+        *decoded_frame.video_frame_buffer()->ToI420(), frame_stat);
>+
>+    // Erase all buffered input frames that we have moved past for all
>+    // simulcast/spatial layers. Never buffer more than
>+    // |kMaxBufferedInputFrames| frames, to protect against long runs of
>+    // consecutive frame drops for a particular layer.
>+    const auto min_last_decoded_frame_num = std::min_element(
>+        last_decoded_frame_num_.cbegin(), last_decoded_frame_num_.cend());
>+    const size_t min_buffered_frame_num = std::max(
>+        0, static_cast<int>(frame_number) - kMaxBufferedInputFrames + 1);
>+    RTC_CHECK(min_last_decoded_frame_num != last_decoded_frame_num_.cend());
>+    const auto input_frames_erase_before = input_frames_.lower_bound(
>+        std::max(*min_last_decoded_frame_num, min_buffered_frame_num));
>+    input_frames_.erase(input_frames_.cbegin(), input_frames_erase_before);
>   }
> 
>-  // Delay erasing of input frames by one frame. The current frame might
>-  // still be needed for other simulcast stream or spatial layer.
>-  const int frame_number_to_erase = frame_number - 1;
>-  if (frame_number_to_erase >= 0) {
>-    auto input_frame_erase_to =
>-        input_frames_.lower_bound(frame_number_to_erase);
>-    input_frames_.erase(input_frames_.begin(), input_frame_erase_to);
>-  }
>-
>-  if (decoded_frame_writer_) {
>-    ExtractBufferWithSize(decoded_frame, config_.codec_settings.width,
>-                          config_.codec_settings.height,
>-                          &last_decoded_frame_buffer_);
>-    WriteDecodedFrameToFile(&last_decoded_frame_buffer_);
>+  if (decoded_frame_writers_) {
>+    ExtractI420BufferWithSize(decoded_frame, config_.codec_settings.width,
>+                              config_.codec_settings.height,
>+                              &decoded_frame_buffer_[spatial_idx]);
>+    RTC_CHECK_EQ(decoded_frame_buffer_[spatial_idx].size(),
>+                 decoded_frame_writers_->at(spatial_idx)->FrameLength());
>+    RTC_CHECK(decoded_frame_writers_->at(spatial_idx)
>+                  ->WriteFrame(decoded_frame_buffer_[spatial_idx].data()));
>   }
> }
> 
>-void VideoProcessor::WriteDecodedFrameToFile(rtc::Buffer* buffer) {
>-  RTC_DCHECK_EQ(buffer->size(), decoded_frame_writer_->FrameLength());
>-  RTC_CHECK(decoded_frame_writer_->WriteFrame(buffer->data()));
>+void VideoProcessor::DecodeFrame(const EncodedImage& encoded_image,
>+                                 size_t spatial_idx) {
>+  RTC_DCHECK_CALLED_SEQUENTIALLY(&sequence_checker_);
>+  FrameStatistics* frame_stat =
>+      stats_->GetFrameWithTimestamp(encoded_image._timeStamp, spatial_idx);
>+
>+  frame_stat->decode_start_ns = rtc::TimeNanos();
>+  frame_stat->decode_return_code =
>+      decoders_->at(spatial_idx)->Decode(encoded_image, false, nullptr, 0);
> }
> 
>-bool VideoProcessor::ExcludeFrame(const EncodedImage& encoded_image) {
>-  RTC_DCHECK_CALLED_SEQUENTIALLY(&sequence_checker_);
>-  if (encoded_image._frameType != kVideoFrameKey) {
>-    return false;
>-  }
>-  bool exclude_frame = false;
>-  switch (config_.exclude_frame_types) {
>-    case kExcludeOnlyFirstKeyFrame:
>-      if (!first_key_frame_has_been_excluded_) {
>-        first_key_frame_has_been_excluded_ = true;
>-        exclude_frame = true;
>+const webrtc::EncodedImage* VideoProcessor::BuildAndStoreSuperframe(
>+    const EncodedImage& encoded_image,
>+    const VideoCodecType codec,
>+    size_t frame_number,
>+    size_t spatial_idx,
>+    bool inter_layer_predicted) {
>+  // Should only be called for SVC.
>+  RTC_CHECK_GT(config_.NumberOfSpatialLayers(), 1);
>+
>+  EncodedImage base_image;
>+  RTC_CHECK_EQ(base_image._length, 0);
>+
>+  // Each SVC layer is decoded with dedicated decoder. Find the nearest
>+  // non-dropped base frame and merge it and current frame into superframe.
>+  if (inter_layer_predicted) {
>+    for (int base_idx = static_cast<int>(spatial_idx) - 1; base_idx >= 0;
>+         --base_idx) {
>+      EncodedImage lower_layer = merged_encoded_frames_.at(base_idx);
>+      if (lower_layer._timeStamp == encoded_image._timeStamp) {
>+        base_image = lower_layer;
>+        break;
>       }
>-      break;
>-    case kExcludeAllKeyFrames:
>-      exclude_frame = true;
>-      break;
>-    default:
>-      RTC_NOTREACHED();
>+    }
>+  }
>+  const size_t payload_size_bytes = base_image._length + encoded_image._length;
>+  const size_t buffer_size_bytes =
>+      payload_size_bytes + EncodedImage::GetBufferPaddingBytes(codec);
>+
>+  uint8_t* copied_buffer = new uint8_t[buffer_size_bytes];
>+  RTC_CHECK(copied_buffer);
>+
>+  if (base_image._length) {
>+    RTC_CHECK(base_image._buffer);
>+    memcpy(copied_buffer, base_image._buffer, base_image._length);
>   }
>-  return exclude_frame;
>+  memcpy(copied_buffer + base_image._length, encoded_image._buffer,
>+         encoded_image._length);
>+
>+  EncodedImage copied_image = encoded_image;
>+  copied_image = encoded_image;
>+  copied_image._buffer = copied_buffer;
>+  copied_image._length = payload_size_bytes;
>+  copied_image._size = buffer_size_bytes;
>+
>+  // Replace previous EncodedImage for this spatial layer.
>+  uint8_t* old_buffer = merged_encoded_frames_.at(spatial_idx)._buffer;
>+  if (old_buffer) {
>+    delete[] old_buffer;
>+  }
>+  merged_encoded_frames_.at(spatial_idx) = copied_image;
>+
>+  return &merged_encoded_frames_.at(spatial_idx);
> }
> 
> }  // namespace test
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videoprocessor.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videoprocessor.h
>index 62a12ef871d..9f1fa133c52 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videoprocessor.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videoprocessor.h
>@@ -16,18 +16,13 @@
> #include <string>
> #include <vector>
> 
>+#include "api/test/videocodec_test_fixture.h"
>+#include "api/test/videocodec_test_stats.h"
>+#include "api/video/video_bitrate_allocator.h"
> #include "api/video/video_frame.h"
>-#include "common_video/libyuv/include/webrtc_libyuv.h"
>-#include "modules/video_coding/codecs/h264/include/h264_globals.h"
>-#include "modules/video_coding/codecs/test/packet_manipulator.h"
>-#include "modules/video_coding/codecs/test/stats.h"
>-#include "modules/video_coding/codecs/test/test_config.h"
> #include "modules/video_coding/include/video_codec_interface.h"
> #include "modules/video_coding/utility/ivf_file_writer.h"
>-#include "modules/video_coding/utility/vp8_header_parser.h"
>-#include "modules/video_coding/utility/vp9_uncompressed_header_parser.h"
> #include "rtc_base/buffer.h"
>-#include "rtc_base/checks.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/sequenced_task_checker.h"
> #include "rtc_base/task_queue.h"
>@@ -35,9 +30,6 @@
> #include "test/testsupport/frame_writer.h"
> 
> namespace webrtc {
>-
>-class VideoBitrateAllocator;
>-
> namespace test {
> 
> // Handles encoding/decoding of video using the VideoEncoder/VideoDecoder
>@@ -45,44 +37,29 @@ namespace test {
> // measure times properly.
> // The class processes a frame at the time for the configured input file.
> // It maintains state of where in the source input file the processing is at.
>-//
>-// Regarding packet loss: Note that keyframes are excluded (first or all
>-// depending on the ExcludeFrameTypes setting). This is because if key frames
>-// would be altered, all the following delta frames would be pretty much
>-// worthless. VP8 has an error-resilience feature that makes it able to handle
>-// packet loss in key non-first keyframes, which is why only the first is
>-// excluded by default.
>-// Packet loss in such important frames is handled on a higher level in the
>-// Video Engine, where signaling would request a retransmit of the lost packets,
>-// since they're so important.
>-//
>-// Note this class is not thread safe and is meant for simple testing purposes.
> class VideoProcessor {
>  public:
>+  using VideoDecoderList = std::vector<std::unique_ptr<VideoDecoder>>;
>+  using IvfFileWriterList = std::vector<std::unique_ptr<IvfFileWriter>>;
>+  using FrameWriterList = std::vector<std::unique_ptr<FrameWriter>>;
>+
>   VideoProcessor(webrtc::VideoEncoder* encoder,
>-                 webrtc::VideoDecoder* decoder,
>-                 FrameReader* analysis_frame_reader,
>-                 PacketManipulator* packet_manipulator,
>-                 const TestConfig& config,
>-                 Stats* stats,
>-                 IvfFileWriter* encoded_frame_writer,
>-                 FrameWriter* decoded_frame_writer);
>+                 VideoDecoderList* decoders,
>+                 FrameReader* input_frame_reader,
>+                 const VideoCodecTestFixture::Config& config,
>+                 VideoCodecTestStats* stats,
>+                 IvfFileWriterList* encoded_frame_writers,
>+                 FrameWriterList* decoded_frame_writers);
>   ~VideoProcessor();
> 
>-  // Reads a frame from the analysis frame reader and sends it to the encoder.
>-  // When the encode callback is received, the encoded frame is sent to the
>-  // decoder. The decoded frame is written to disk by the analysis frame writer.
>-  // Objective video quality metrics can thus be calculated after the fact.
>+  // Reads a frame and sends it to the encoder. When the encode callback
>+  // is received, the encoded frame is buffered. After encoding is finished
>+  // buffered frame is sent to decoder. Quality evaluation is done in
>+  // the decode callback.
>   void ProcessFrame();
> 
>   // Updates the encoder with target rates. Must be called at least once.
>-  void SetRates(int bitrate_kbps, int framerate_fps);
>-
>-  // Returns the number of dropped frames.
>-  std::vector<int> NumberDroppedFramesPerRateUpdate() const;
>-
>-  // Returns the number of spatial resizes.
>-  std::vector<int> NumberSpatialResizesPerRateUpdate() const;
>+  void SetRates(size_t bitrate_kbps, size_t framerate_fps);
> 
>  private:
>   class VideoProcessorEncodeCompleteCallback
>@@ -91,7 +68,10 @@ class VideoProcessor {
>     explicit VideoProcessorEncodeCompleteCallback(
>         VideoProcessor* video_processor)
>         : video_processor_(video_processor),
>-          task_queue_(rtc::TaskQueue::Current()) {}
>+          task_queue_(rtc::TaskQueue::Current()) {
>+      RTC_DCHECK(video_processor_);
>+      RTC_DCHECK(task_queue_);
>+    }
> 
>     Result OnEncodedImage(
>         const webrtc::EncodedImage& encoded_image,
>@@ -99,15 +79,15 @@ class VideoProcessor {
>         const webrtc::RTPFragmentationHeader* fragmentation) override {
>       RTC_CHECK(codec_specific_info);
> 
>-      if (task_queue_ && !task_queue_->IsCurrent()) {
>+      // Post the callback to the right task queue, if needed.
>+      if (!task_queue_->IsCurrent()) {
>         task_queue_->PostTask(
>             std::unique_ptr<rtc::QueuedTask>(new EncodeCallbackTask(
>                 video_processor_, encoded_image, codec_specific_info)));
>         return Result(Result::OK, 0);
>       }
> 
>-      video_processor_->FrameEncoded(codec_specific_info->codecType,
>-                                     encoded_image);
>+      video_processor_->FrameEncoded(encoded_image, *codec_specific_info);
>       return Result(Result::OK, 0);
>     }
> 
>@@ -125,8 +105,7 @@ class VideoProcessor {
>       }
> 
>       bool Run() override {
>-        video_processor_->FrameEncoded(codec_specific_info_.codecType,
>-                                       encoded_image_);
>+        video_processor_->FrameEncoded(encoded_image_, codec_specific_info_);
>         return true;
>       }
> 
>@@ -145,100 +124,119 @@ class VideoProcessor {
>       : public webrtc::DecodedImageCallback {
>    public:
>     explicit VideoProcessorDecodeCompleteCallback(
>-        VideoProcessor* video_processor)
>+        VideoProcessor* video_processor,
>+        size_t simulcast_svc_idx)
>         : video_processor_(video_processor),
>-          task_queue_(rtc::TaskQueue::Current()) {}
>-
>-    int32_t Decoded(webrtc::VideoFrame& image) override {
>-      if (task_queue_ && !task_queue_->IsCurrent()) {
>-        task_queue_->PostTask(
>-            [this, image]() { video_processor_->FrameDecoded(image); });
>-        return 0;
>-      }
>-      video_processor_->FrameDecoded(image);
>-      return 0;
>+          simulcast_svc_idx_(simulcast_svc_idx),
>+          task_queue_(rtc::TaskQueue::Current()) {
>+      RTC_DCHECK(video_processor_);
>+      RTC_DCHECK(task_queue_);
>     }
> 
>+    int32_t Decoded(webrtc::VideoFrame& image) override;
>+
>     int32_t Decoded(webrtc::VideoFrame& image,
>                     int64_t decode_time_ms) override {
>       return Decoded(image);
>     }
> 
>     void Decoded(webrtc::VideoFrame& image,
>-                 rtc::Optional<int32_t> decode_time_ms,
>-                 rtc::Optional<uint8_t> qp) override {
>+                 absl::optional<int32_t> decode_time_ms,
>+                 absl::optional<uint8_t> qp) override {
>       Decoded(image);
>     }
> 
>    private:
>     VideoProcessor* const video_processor_;
>+    const size_t simulcast_svc_idx_;
>     rtc::TaskQueue* const task_queue_;
>   };
> 
>   // Invoked by the callback adapter when a frame has completed encoding.
>-  void FrameEncoded(webrtc::VideoCodecType codec,
>-                    const webrtc::EncodedImage& encodedImage);
>+  void FrameEncoded(const webrtc::EncodedImage& encoded_image,
>+                    const webrtc::CodecSpecificInfo& codec_specific);
> 
>   // Invoked by the callback adapter when a frame has completed decoding.
>-  void FrameDecoded(const webrtc::VideoFrame& image);
>+  void FrameDecoded(const webrtc::VideoFrame& image, size_t simulcast_svc_idx);
>+
>+  void DecodeFrame(const EncodedImage& encoded_image, size_t simulcast_svc_idx);
> 
>-  void WriteDecodedFrameToFile(rtc::Buffer* buffer);
>-  bool ExcludeFrame(const EncodedImage& encoded_image);
>+  // In order to supply the SVC decoders with super frames containing all
>+  // lower layer frames, we merge and store the layer frames in this method.
>+  const webrtc::EncodedImage* BuildAndStoreSuperframe(
>+      const EncodedImage& encoded_image,
>+      const VideoCodecType codec,
>+      size_t frame_number,
>+      size_t simulcast_svc_idx,
>+      bool inter_layer_predicted) RTC_RUN_ON(sequence_checker_);
> 
>-  TestConfig config_ RTC_GUARDED_BY(sequence_checker_);
>+  // Test input/output.
>+  VideoCodecTestFixture::Config config_ RTC_GUARDED_BY(sequence_checker_);
>+  const size_t num_simulcast_or_spatial_layers_;
>+  VideoCodecTestStats* const stats_;
> 
>+  // Codecs.
>   webrtc::VideoEncoder* const encoder_;
>-  webrtc::VideoDecoder* const decoder_;
>+  VideoDecoderList* const decoders_;
>   const std::unique_ptr<VideoBitrateAllocator> bitrate_allocator_;
>+  VideoBitrateAllocation bitrate_allocation_ RTC_GUARDED_BY(sequence_checker_);
>+  uint32_t framerate_fps_ RTC_GUARDED_BY(sequence_checker_);
> 
>   // Adapters for the codec callbacks.
>   VideoProcessorEncodeCompleteCallback encode_callback_;
>-  VideoProcessorDecodeCompleteCallback decode_callback_;
>+  // Assign separate callback object to each decoder. This allows us to identify
>+  // decoded layer in frame decode callback.
>+  // simulcast_svc_idx -> decode callback.
>+  std::vector<std::unique_ptr<VideoProcessorDecodeCompleteCallback>>
>+      decode_callback_;
> 
>-  // Fake network.
>-  PacketManipulator* const packet_manipulator_;
>+  // Each call to ProcessFrame() will read one frame from |input_frame_reader_|.
>+  FrameReader* const input_frame_reader_;
> 
>-  // Input frames. Used as reference at frame quality evaluation.
>+  // Input frames are used as reference for frame quality evaluations.
>   // Async codecs might queue frames. To handle that we keep input frame
>   // and release it after corresponding coded frame is decoded and quality
>   // measurement is done.
>-  std::map<int, std::unique_ptr<VideoFrame>> input_frames_
>+  // frame_number -> frame.
>+  std::map<size_t, VideoFrame> input_frames_ RTC_GUARDED_BY(sequence_checker_);
>+
>+  // Encoder delivers coded frame layer-by-layer. We store coded frames and
>+  // then, after all layers are encoded, decode them. Such separation of
>+  // frame processing on superframe level simplifies encoding/decoding time
>+  // measurement.
>+  // simulcast_svc_idx -> merged SVC encoded frame.
>+  std::vector<EncodedImage> merged_encoded_frames_
>       RTC_GUARDED_BY(sequence_checker_);
> 
>-  // These (mandatory) file manipulators are used for, e.g., objective PSNR and
>-  // SSIM calculations at the end of a test run.
>-  FrameReader* const analysis_frame_reader_;
>-
>   // These (optional) file writers are used to persistently store the encoded
>-  // and decoded bitstreams. The purpose is to give the experimenter an option
>-  // to subjectively evaluate the quality of the processing. Each frame writer
>-  // is enabled by being non-null.
>-  IvfFileWriter* const encoded_frame_writer_;
>-  FrameWriter* const decoded_frame_writer_;
>-
>-  // Keep track of inputed/encoded/decoded frames, so we can detect frame drops.
>-  int last_inputed_frame_num_ RTC_GUARDED_BY(sequence_checker_);
>-  int last_encoded_frame_num_ RTC_GUARDED_BY(sequence_checker_);
>-  int last_decoded_frame_num_ RTC_GUARDED_BY(sequence_checker_);
>-
>-  // Store an RTP timestamp -> frame number map, since the timestamps are
>-  // based off of the frame rate, which can change mid-test.
>-  std::map<uint32_t, int> rtp_timestamp_to_frame_num_
>+  // and decoded bitstreams. Each frame writer is enabled by being non-null.
>+  IvfFileWriterList* const encoded_frame_writers_;
>+  FrameWriterList* const decoded_frame_writers_;
>+
>+  // Metadata for inputed/encoded/decoded frames. Used for frame identification,
>+  // frame drop detection, etc. We assume that encoded/decoded frames are
>+  // ordered within each simulcast/spatial layer, but we do not make any
>+  // assumptions of frame ordering between layers.
>+  size_t last_inputed_frame_num_ RTC_GUARDED_BY(sequence_checker_);
>+  size_t last_inputed_timestamp_ RTC_GUARDED_BY(sequence_checker_);
>+  // simulcast_svc_idx -> encode status.
>+  std::vector<bool> first_encoded_frame_ RTC_GUARDED_BY(sequence_checker_);
>+  // simulcast_svc_idx -> frame_number.
>+  std::vector<size_t> last_encoded_frame_num_ RTC_GUARDED_BY(sequence_checker_);
>+  // simulcast_svc_idx -> decode status.
>+  std::vector<bool> first_decoded_frame_ RTC_GUARDED_BY(sequence_checker_);
>+  // simulcast_svc_idx -> frame_number.
>+  std::vector<size_t> last_decoded_frame_num_ RTC_GUARDED_BY(sequence_checker_);
>+  // simulcast_svc_idx -> buffer.
>+  std::vector<rtc::Buffer> decoded_frame_buffer_
>       RTC_GUARDED_BY(sequence_checker_);
> 
>-  // Keep track of if we have excluded the first key frame from packet loss.
>-  bool first_key_frame_has_been_excluded_ RTC_GUARDED_BY(sequence_checker_);
>-
>-  // Keep track of the last successfully decoded frame, since we write that
>-  // frame to disk when decoding fails.
>-  rtc::Buffer last_decoded_frame_buffer_ RTC_GUARDED_BY(sequence_checker_);
>-
>-  // Statistics.
>-  Stats* stats_;
>-  std::vector<int> num_dropped_frames_ RTC_GUARDED_BY(sequence_checker_);
>-  std::vector<int> num_spatial_resizes_ RTC_GUARDED_BY(sequence_checker_);
>-  int rate_update_index_ RTC_GUARDED_BY(sequence_checker_);
>+  // Time spent in frame encode callback. It is accumulated for layers and
>+  // reset when frame encode starts. When next layer is encoded post-encode time
>+  // is substracted from measured encode time. Thus we get pure encode time.
>+  int64_t post_encode_time_ns_ RTC_GUARDED_BY(sequence_checker_);
> 
>+  // This class must be operated on a TaskQueue.
>   rtc::SequencedTaskChecker sequence_checker_;
> 
>   RTC_DISALLOW_COPY_AND_ASSIGN(VideoProcessor);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest.cc
>deleted file mode 100644
>index dc9b146560d..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest.cc
>+++ /dev/null
>@@ -1,665 +0,0 @@
>-/*
>- *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "modules/video_coding/codecs/test/videoprocessor_integrationtest.h"
>-
>-#include <algorithm>
>-#include <utility>
>-
>-#if defined(WEBRTC_ANDROID)
>-#include "modules/video_coding/codecs/test/android_test_initializer.h"
>-#include "sdk/android/src/jni/class_loader.h"
>-#include "sdk/android/src/jni/videodecoderfactorywrapper.h"
>-#include "sdk/android/src/jni/videoencoderfactorywrapper.h"
>-#elif defined(WEBRTC_IOS)
>-#include "modules/video_coding/codecs/test/objc_codec_h264_test.h"
>-#endif
>-
>-#include "common_types.h"  // NOLINT(build/include)
>-#include "media/base/h264_profile_level_id.h"
>-#include "media/engine/internaldecoderfactory.h"
>-#include "media/engine/internalencoderfactory.h"
>-#include "media/engine/videodecodersoftwarefallbackwrapper.h"
>-#include "media/engine/videoencodersoftwarefallbackwrapper.h"
>-#include "modules/video_coding/codecs/vp8/include/vp8_common_types.h"
>-#include "modules/video_coding/include/video_codec_interface.h"
>-#include "modules/video_coding/include/video_coding.h"
>-#include "rtc_base/checks.h"
>-#include "rtc_base/cpu_time.h"
>-#include "rtc_base/event.h"
>-#include "rtc_base/file.h"
>-#include "rtc_base/ptr_util.h"
>-#include "system_wrappers/include/sleep.h"
>-#include "test/testsupport/fileutils.h"
>-#include "test/testsupport/metrics/video_metrics.h"
>-
>-namespace webrtc {
>-namespace test {
>-
>-namespace {
>-
>-const int kMaxBitrateMismatchPercent = 20;
>-
>-// Parameters from VP8 wrapper, which control target size of key frames.
>-const float kInitialBufferSize = 0.5f;
>-const float kOptimalBufferSize = 0.6f;
>-const float kScaleKeyFrameSize = 0.5f;
>-
>-bool RunEncodeInRealTime(const TestConfig& config) {
>-  if (config.measure_cpu) {
>-    return true;
>-  }
>-#if defined(WEBRTC_ANDROID)
>-  // In order to not overwhelm the OpenMAX buffers in the Android MediaCodec.
>-  return (config.hw_encoder || config.hw_decoder);
>-#else
>-  return false;
>-#endif
>-}
>-
>-SdpVideoFormat CreateSdpVideoFormat(const TestConfig& config) {
>-  switch (config.codec_settings.codecType) {
>-    case kVideoCodecVP8:
>-      return SdpVideoFormat(cricket::kVp8CodecName);
>-
>-    case kVideoCodecVP9:
>-      return SdpVideoFormat(cricket::kVp9CodecName);
>-
>-    case kVideoCodecH264: {
>-      const char* packetization_mode =
>-          config.h264_codec_settings.packetization_mode ==
>-                  H264PacketizationMode::NonInterleaved
>-              ? "1"
>-              : "0";
>-      return SdpVideoFormat(
>-          cricket::kH264CodecName,
>-          {{cricket::kH264FmtpProfileLevelId,
>-            *H264::ProfileLevelIdToString(H264::ProfileLevelId(
>-                config.h264_codec_settings.profile, H264::kLevel3_1))},
>-           {cricket::kH264FmtpPacketizationMode, packetization_mode}});
>-    }
>-    default:
>-      RTC_NOTREACHED();
>-      return SdpVideoFormat("");
>-  }
>-}
>-
>-}  // namespace
>-
>-void VideoProcessorIntegrationTest::H264KeyframeChecker::CheckEncodedFrame(
>-    webrtc::VideoCodecType codec,
>-    const EncodedImage& encoded_frame) const {
>-  EXPECT_EQ(kVideoCodecH264, codec);
>-  bool contains_sps = false;
>-  bool contains_pps = false;
>-  bool contains_idr = false;
>-  const std::vector<webrtc::H264::NaluIndex> nalu_indices =
>-      webrtc::H264::FindNaluIndices(encoded_frame._buffer,
>-                                    encoded_frame._length);
>-  for (const webrtc::H264::NaluIndex& index : nalu_indices) {
>-    webrtc::H264::NaluType nalu_type = webrtc::H264::ParseNaluType(
>-        encoded_frame._buffer[index.payload_start_offset]);
>-    if (nalu_type == webrtc::H264::NaluType::kSps) {
>-      contains_sps = true;
>-    } else if (nalu_type == webrtc::H264::NaluType::kPps) {
>-      contains_pps = true;
>-    } else if (nalu_type == webrtc::H264::NaluType::kIdr) {
>-      contains_idr = true;
>-    }
>-  }
>-  if (encoded_frame._frameType == kVideoFrameKey) {
>-    EXPECT_TRUE(contains_sps) << "Keyframe should contain SPS.";
>-    EXPECT_TRUE(contains_pps) << "Keyframe should contain PPS.";
>-    EXPECT_TRUE(contains_idr) << "Keyframe should contain IDR.";
>-  } else if (encoded_frame._frameType == kVideoFrameDelta) {
>-    EXPECT_FALSE(contains_sps) << "Delta frame should not contain SPS.";
>-    EXPECT_FALSE(contains_pps) << "Delta frame should not contain PPS.";
>-    EXPECT_FALSE(contains_idr) << "Delta frame should not contain IDR.";
>-  } else {
>-    RTC_NOTREACHED();
>-  }
>-}
>-
>-class VideoProcessorIntegrationTest::CpuProcessTime final {
>- public:
>-  explicit CpuProcessTime(const TestConfig& config) : config_(config) {}
>-  ~CpuProcessTime() {}
>-
>-  void Start() {
>-    if (config_.measure_cpu) {
>-      cpu_time_ -= rtc::GetProcessCpuTimeNanos();
>-      wallclock_time_ -= rtc::SystemTimeNanos();
>-    }
>-  }
>-  void Stop() {
>-    if (config_.measure_cpu) {
>-      cpu_time_ += rtc::GetProcessCpuTimeNanos();
>-      wallclock_time_ += rtc::SystemTimeNanos();
>-    }
>-  }
>-  void Print() const {
>-    if (config_.measure_cpu) {
>-      printf("CPU usage %%: %f\n", GetUsagePercent() / config_.NumberOfCores());
>-      printf("\n");
>-    }
>-  }
>-
>- private:
>-  double GetUsagePercent() const {
>-    return static_cast<double>(cpu_time_) / wallclock_time_ * 100.0;
>-  }
>-
>-  const TestConfig config_;
>-  int64_t cpu_time_ = 0;
>-  int64_t wallclock_time_ = 0;
>-};
>-
>-VideoProcessorIntegrationTest::VideoProcessorIntegrationTest() {
>-#if defined(WEBRTC_ANDROID)
>-  InitializeAndroidObjects();
>-#endif
>-}
>-
>-VideoProcessorIntegrationTest::~VideoProcessorIntegrationTest() = default;
>-
>-// Processes all frames in the clip and verifies the result.
>-void VideoProcessorIntegrationTest::ProcessFramesAndMaybeVerify(
>-    const std::vector<RateProfile>& rate_profiles,
>-    const std::vector<RateControlThresholds>* rc_thresholds,
>-    const QualityThresholds* quality_thresholds,
>-    const BitstreamThresholds* bs_thresholds,
>-    const VisualizationParams* visualization_params) {
>-  RTC_DCHECK(!rate_profiles.empty());
>-  // The Android HW codec needs to be run on a task queue, so we simply always
>-  // run the test on a task queue.
>-  rtc::TaskQueue task_queue("VidProc TQ");
>-  rtc::Event sync_event(false, false);
>-
>-  SetUpAndInitObjects(&task_queue, rate_profiles[0].target_kbps,
>-                      rate_profiles[0].input_fps, visualization_params);
>-  PrintSettings();
>-
>-  // Set initial rates.
>-  int rate_update_index = 0;
>-  task_queue.PostTask([this, &rate_profiles, rate_update_index] {
>-    processor_->SetRates(rate_profiles[rate_update_index].target_kbps,
>-                         rate_profiles[rate_update_index].input_fps);
>-  });
>-
>-  cpu_process_time_->Start();
>-
>-  // Process all frames.
>-  int frame_number = 0;
>-  const int num_frames = config_.num_frames;
>-  RTC_DCHECK_GE(num_frames, 1);
>-  while (frame_number < num_frames) {
>-    if (RunEncodeInRealTime(config_)) {
>-      // Roughly pace the frames.
>-      SleepMs(rtc::kNumMillisecsPerSec /
>-              rate_profiles[rate_update_index].input_fps);
>-    }
>-
>-    task_queue.PostTask([this] { processor_->ProcessFrame(); });
>-    ++frame_number;
>-
>-    if (frame_number ==
>-        rate_profiles[rate_update_index].frame_index_rate_update) {
>-      ++rate_update_index;
>-      RTC_DCHECK_GT(rate_profiles.size(), rate_update_index);
>-
>-      task_queue.PostTask([this, &rate_profiles, rate_update_index] {
>-        processor_->SetRates(rate_profiles[rate_update_index].target_kbps,
>-                             rate_profiles[rate_update_index].input_fps);
>-      });
>-    }
>-  }
>-
>-  // Give the VideoProcessor pipeline some time to process the last frame,
>-  // and then release the codecs.
>-  if (config_.hw_encoder || config_.hw_decoder) {
>-    SleepMs(1 * rtc::kNumMillisecsPerSec);
>-  }
>-  cpu_process_time_->Stop();
>-
>-  std::vector<int> num_dropped_frames;
>-  std::vector<int> num_spatial_resizes;
>-  sync_event.Reset();
>-  task_queue.PostTask(
>-      [this, &num_dropped_frames, &num_spatial_resizes, &sync_event]() {
>-        num_dropped_frames = processor_->NumberDroppedFramesPerRateUpdate();
>-        num_spatial_resizes = processor_->NumberSpatialResizesPerRateUpdate();
>-        sync_event.Set();
>-      });
>-  sync_event.Wait(rtc::Event::kForever);
>-
>-  ReleaseAndCloseObjects(&task_queue);
>-
>-  // Calculate and print rate control statistics.
>-  rate_update_index = 0;
>-  frame_number = 0;
>-  quality_ = QualityMetrics();
>-  ResetRateControlMetrics(rate_update_index, rate_profiles);
>-  while (frame_number < num_frames) {
>-    UpdateRateControlMetrics(frame_number);
>-
>-    if (quality_thresholds) {
>-      UpdateQualityMetrics(frame_number);
>-    }
>-
>-    if (bs_thresholds) {
>-      VerifyBitstream(frame_number, *bs_thresholds);
>-    }
>-
>-    ++frame_number;
>-
>-    if (frame_number ==
>-        rate_profiles[rate_update_index].frame_index_rate_update) {
>-      PrintRateControlMetrics(rate_update_index, num_dropped_frames,
>-                              num_spatial_resizes);
>-      VerifyRateControlMetrics(rate_update_index, rc_thresholds,
>-                               num_dropped_frames, num_spatial_resizes);
>-      ++rate_update_index;
>-      ResetRateControlMetrics(rate_update_index, rate_profiles);
>-    }
>-  }
>-
>-  PrintRateControlMetrics(rate_update_index, num_dropped_frames,
>-                          num_spatial_resizes);
>-  VerifyRateControlMetrics(rate_update_index, rc_thresholds, num_dropped_frames,
>-                           num_spatial_resizes);
>-
>-  if (quality_thresholds) {
>-    VerifyQualityMetrics(*quality_thresholds);
>-  }
>-
>-  // Calculate and print other statistics.
>-  EXPECT_EQ(num_frames, static_cast<int>(stats_.size()));
>-  stats_.PrintSummary();
>-  cpu_process_time_->Print();
>-}
>-
>-void VideoProcessorIntegrationTest::CreateEncoderAndDecoder() {
>-  std::unique_ptr<VideoEncoderFactory> encoder_factory;
>-  if (config_.hw_encoder) {
>-#if defined(WEBRTC_ANDROID)
>-    JNIEnv* env = jni::AttachCurrentThreadIfNeeded();
>-    jni::ScopedJavaLocalRef<jclass> factory_class =
>-        jni::GetClass(env, "org/webrtc/HardwareVideoEncoderFactory");
>-    jmethodID factory_constructor = env->GetMethodID(
>-        factory_class.obj(), "<init>", "(Lorg/webrtc/EglBase$Context;ZZ)V");
>-    jni::ScopedJavaLocalRef<jobject> factory_object(
>-        env, env->NewObject(factory_class.obj(), factory_constructor,
>-                            nullptr /* shared_context */,
>-                            false /* enable_intel_vp8_encoder */,
>-                            true /* enable_h264_high_profile */));
>-    encoder_factory = rtc::MakeUnique<webrtc::jni::VideoEncoderFactoryWrapper>(
>-        env, factory_object);
>-#elif defined(WEBRTC_IOS)
>-    EXPECT_EQ(kVideoCodecH264, config_.codec_settings.codecType)
>-        << "iOS HW codecs only support H264.";
>-    encoder_factory = CreateObjCEncoderFactory();
>-#else
>-    RTC_NOTREACHED() << "Only support HW encoder on Android and iOS.";
>-#endif
>-  } else {
>-    encoder_factory = rtc::MakeUnique<InternalEncoderFactory>();
>-  }
>-
>-  std::unique_ptr<VideoDecoderFactory> decoder_factory;
>-  if (config_.hw_decoder) {
>-#if defined(WEBRTC_ANDROID)
>-    JNIEnv* env = jni::AttachCurrentThreadIfNeeded();
>-    jni::ScopedJavaLocalRef<jclass> factory_class =
>-        jni::GetClass(env, "org/webrtc/HardwareVideoDecoderFactory");
>-    jmethodID factory_constructor = env->GetMethodID(
>-        factory_class.obj(), "<init>", "(Lorg/webrtc/EglBase$Context;)V");
>-    jni::ScopedJavaLocalRef<jobject> factory_object(
>-        env, env->NewObject(factory_class.obj(), factory_constructor,
>-                            nullptr /* shared_context */));
>-    decoder_factory = rtc::MakeUnique<webrtc::jni::VideoDecoderFactoryWrapper>(
>-        env, factory_object);
>-#elif defined(WEBRTC_IOS)
>-    EXPECT_EQ(kVideoCodecH264, config_.codec_settings.codecType)
>-        << "iOS HW codecs only support H264.";
>-    decoder_factory = CreateObjCDecoderFactory();
>-#else
>-    RTC_NOTREACHED() << "Only support HW decoder on Android and iOS.";
>-#endif
>-  } else {
>-    decoder_factory = rtc::MakeUnique<InternalDecoderFactory>();
>-  }
>-
>-  const SdpVideoFormat format = CreateSdpVideoFormat(config_);
>-  encoder_ = encoder_factory->CreateVideoEncoder(format);
>-  decoder_ = decoder_factory->CreateVideoDecoder(format);
>-
>-  if (config_.sw_fallback_encoder) {
>-    encoder_ = rtc::MakeUnique<VideoEncoderSoftwareFallbackWrapper>(
>-        InternalEncoderFactory().CreateVideoEncoder(format),
>-        std::move(encoder_));
>-  }
>-  if (config_.sw_fallback_decoder) {
>-    decoder_ = rtc::MakeUnique<VideoDecoderSoftwareFallbackWrapper>(
>-        InternalDecoderFactory().CreateVideoDecoder(format),
>-        std::move(decoder_));
>-  }
>-
>-  EXPECT_TRUE(encoder_) << "Encoder not successfully created.";
>-  EXPECT_TRUE(decoder_) << "Decoder not successfully created.";
>-}
>-
>-void VideoProcessorIntegrationTest::DestroyEncoderAndDecoder() {
>-  encoder_.reset();
>-  decoder_.reset();
>-}
>-
>-void VideoProcessorIntegrationTest::SetUpAndInitObjects(
>-    rtc::TaskQueue* task_queue,
>-    const int initial_bitrate_kbps,
>-    const int initial_framerate_fps,
>-    const VisualizationParams* visualization_params) {
>-  CreateEncoderAndDecoder();
>-
>-  config_.codec_settings.minBitrate = 0;
>-  config_.codec_settings.startBitrate = initial_bitrate_kbps;
>-  config_.codec_settings.maxFramerate = initial_framerate_fps;
>-
>-  // Create file objects for quality analysis.
>-  analysis_frame_reader_.reset(new YuvFrameReaderImpl(
>-      config_.input_filename, config_.codec_settings.width,
>-      config_.codec_settings.height));
>-  analysis_frame_writer_.reset(new YuvFrameWriterImpl(
>-      config_.output_filename, config_.codec_settings.width,
>-      config_.codec_settings.height));
>-  EXPECT_TRUE(analysis_frame_reader_->Init());
>-  EXPECT_TRUE(analysis_frame_writer_->Init());
>-
>-  if (visualization_params) {
>-    const std::string output_filename_base =
>-        OutputPath() + config_.FilenameWithParams();
>-    if (visualization_params->save_encoded_ivf) {
>-      rtc::File post_encode_file =
>-          rtc::File::Create(output_filename_base + ".ivf");
>-      encoded_frame_writer_ =
>-          IvfFileWriter::Wrap(std::move(post_encode_file), 0);
>-    }
>-    if (visualization_params->save_decoded_y4m) {
>-      decoded_frame_writer_.reset(new Y4mFrameWriterImpl(
>-          output_filename_base + ".y4m", config_.codec_settings.width,
>-          config_.codec_settings.height, initial_framerate_fps));
>-      EXPECT_TRUE(decoded_frame_writer_->Init());
>-    }
>-  }
>-
>-  cpu_process_time_.reset(new CpuProcessTime(config_));
>-  packet_manipulator_.reset(new PacketManipulatorImpl(
>-      &packet_reader_, config_.networking_config, false));
>-
>-  rtc::Event sync_event(false, false);
>-  task_queue->PostTask([this, &sync_event]() {
>-    processor_ = rtc::MakeUnique<VideoProcessor>(
>-        encoder_.get(), decoder_.get(), analysis_frame_reader_.get(),
>-        packet_manipulator_.get(), config_, &stats_,
>-        encoded_frame_writer_.get(), decoded_frame_writer_.get());
>-    sync_event.Set();
>-  });
>-  sync_event.Wait(rtc::Event::kForever);
>-}
>-
>-void VideoProcessorIntegrationTest::ReleaseAndCloseObjects(
>-    rtc::TaskQueue* task_queue) {
>-  rtc::Event sync_event(false, false);
>-  task_queue->PostTask([this, &sync_event]() {
>-    processor_.reset();
>-    sync_event.Set();
>-  });
>-  sync_event.Wait(rtc::Event::kForever);
>-
>-  // The VideoProcessor must be destroyed before the codecs.
>-  DestroyEncoderAndDecoder();
>-
>-  analysis_frame_reader_->Close();
>-
>-  // Close visualization files.
>-  if (encoded_frame_writer_) {
>-    EXPECT_TRUE(encoded_frame_writer_->Close());
>-  }
>-  if (decoded_frame_writer_) {
>-    decoded_frame_writer_->Close();
>-  }
>-}
>-
>-// For every encoded frame, update the rate control metrics.
>-void VideoProcessorIntegrationTest::UpdateRateControlMetrics(int frame_number) {
>-  RTC_CHECK_GE(frame_number, 0);
>-
>-  const int tl_idx = config_.TemporalLayerForFrame(frame_number);
>-  ++actual_.num_frames_layer[tl_idx];
>-  ++actual_.num_frames;
>-
>-  const FrameStatistic* frame_stat = stats_.GetFrame(frame_number);
>-  FrameType frame_type = frame_stat->frame_type;
>-  float framesize_kbits = frame_stat->encoded_frame_size_bytes * 8.0f / 1000.0f;
>-
>-  // Update rate mismatch relative to per-frame bandwidth.
>-  if (frame_type == kVideoFrameDelta) {
>-    // TODO(marpan): Should we count dropped (zero size) frames in mismatch?
>-    actual_.sum_delta_framesize_mismatch_layer[tl_idx] +=
>-        fabs(framesize_kbits - target_.framesize_kbits_layer[tl_idx]) /
>-        target_.framesize_kbits_layer[tl_idx];
>-  } else {
>-    float key_framesize_kbits = (frame_number == 0)
>-                                    ? target_.key_framesize_kbits_initial
>-                                    : target_.key_framesize_kbits;
>-    actual_.sum_key_framesize_mismatch +=
>-        fabs(framesize_kbits - key_framesize_kbits) / key_framesize_kbits;
>-    ++actual_.num_key_frames;
>-  }
>-  actual_.sum_framesize_kbits += framesize_kbits;
>-  actual_.sum_framesize_kbits_layer[tl_idx] += framesize_kbits;
>-
>-  // Encoded bitrate: from the start of the update/run to current frame.
>-  actual_.kbps = actual_.sum_framesize_kbits * target_.fps / actual_.num_frames;
>-  actual_.kbps_layer[tl_idx] = actual_.sum_framesize_kbits_layer[tl_idx] *
>-                               target_.fps_layer[tl_idx] /
>-                               actual_.num_frames_layer[tl_idx];
>-
>-  // Number of frames to hit target bitrate.
>-  if (actual_.BitrateMismatchPercent(target_.kbps) <
>-      kMaxBitrateMismatchPercent) {
>-    actual_.num_frames_to_hit_target =
>-        std::min(actual_.num_frames, actual_.num_frames_to_hit_target);
>-  }
>-}
>-
>-// Verify expected behavior of rate control.
>-void VideoProcessorIntegrationTest::VerifyRateControlMetrics(
>-    int rate_update_index,
>-    const std::vector<RateControlThresholds>* rc_thresholds,
>-    const std::vector<int>& num_dropped_frames,
>-    const std::vector<int>& num_spatial_resizes) const {
>-  if (!rc_thresholds)
>-    return;
>-
>-  const RateControlThresholds& rc_threshold =
>-      (*rc_thresholds)[rate_update_index];
>-
>-  EXPECT_LE(num_dropped_frames[rate_update_index],
>-            rc_threshold.max_num_dropped_frames);
>-  EXPECT_EQ(rc_threshold.num_spatial_resizes,
>-            num_spatial_resizes[rate_update_index]);
>-
>-  EXPECT_LE(actual_.num_frames_to_hit_target,
>-            rc_threshold.max_num_frames_to_hit_target);
>-  EXPECT_EQ(rc_threshold.num_key_frames, actual_.num_key_frames);
>-  EXPECT_LE(actual_.KeyFrameSizeMismatchPercent(),
>-            rc_threshold.max_key_framesize_mismatch_percent);
>-  EXPECT_LE(actual_.BitrateMismatchPercent(target_.kbps),
>-            rc_threshold.max_bitrate_mismatch_percent);
>-
>-  const int num_temporal_layers = config_.NumberOfTemporalLayers();
>-  for (int i = 0; i < num_temporal_layers; ++i) {
>-    EXPECT_LE(actual_.DeltaFrameSizeMismatchPercent(i),
>-              rc_threshold.max_delta_framesize_mismatch_percent);
>-    EXPECT_LE(actual_.BitrateMismatchPercent(i, target_.kbps_layer[i]),
>-              rc_threshold.max_bitrate_mismatch_percent);
>-  }
>-}
>-
>-void VideoProcessorIntegrationTest::UpdateQualityMetrics(int frame_number) {
>-  FrameStatistic* frame_stat = stats_.GetFrame(frame_number);
>-  if (frame_stat->decoding_successful) {
>-    ++quality_.num_decoded_frames;
>-    quality_.total_psnr += frame_stat->psnr;
>-    quality_.total_ssim += frame_stat->ssim;
>-    if (frame_stat->psnr < quality_.min_psnr)
>-      quality_.min_psnr = frame_stat->psnr;
>-    if (frame_stat->ssim < quality_.min_ssim)
>-      quality_.min_ssim = frame_stat->ssim;
>-  }
>-}
>-
>-void VideoProcessorIntegrationTest::PrintRateControlMetrics(
>-    int rate_update_index,
>-    const std::vector<int>& num_dropped_frames,
>-    const std::vector<int>& num_spatial_resizes) const {
>-  if (rate_update_index == 0) {
>-    printf("Rate control statistics\n==\n");
>-  }
>-
>-  printf("Rate update #%d:\n", rate_update_index);
>-  printf(" Target bitrate          : %d\n", target_.kbps);
>-  printf(" Encoded bitrate         : %f\n", actual_.kbps);
>-  printf(" Frame rate              : %d\n", target_.fps);
>-  printf(" # processed frames      : %d\n", actual_.num_frames);
>-  printf(" # frames to convergence : %d\n", actual_.num_frames_to_hit_target);
>-  printf(" # dropped frames        : %d\n",
>-         num_dropped_frames[rate_update_index]);
>-  printf(" # spatial resizes       : %d\n",
>-         num_spatial_resizes[rate_update_index]);
>-  printf(" # key frames            : %d\n", actual_.num_key_frames);
>-  printf(" Key frame rate mismatch : %d\n",
>-         actual_.KeyFrameSizeMismatchPercent());
>-
>-  const int num_temporal_layers = config_.NumberOfTemporalLayers();
>-  for (int i = 0; i < num_temporal_layers; ++i) {
>-    printf(" Temporal layer #%d:\n", i);
>-    printf("  TL%d target bitrate        : %f\n", i, target_.kbps_layer[i]);
>-    printf("  TL%d encoded bitrate       : %f\n", i, actual_.kbps_layer[i]);
>-    printf("  TL%d frame rate            : %f\n", i, target_.fps_layer[i]);
>-    printf("  TL%d # processed frames    : %d\n", i,
>-           actual_.num_frames_layer[i]);
>-    printf("  TL%d frame size %% mismatch : %d\n", i,
>-           actual_.DeltaFrameSizeMismatchPercent(i));
>-    printf("  TL%d bitrate %% mismatch    : %d\n", i,
>-           actual_.BitrateMismatchPercent(i, target_.kbps_layer[i]));
>-    printf("  TL%d per-frame bitrate     : %f\n", i,
>-           target_.framesize_kbits_layer[i]);
>-  }
>-  printf("\n");
>-}
>-
>-void VideoProcessorIntegrationTest::PrintSettings() const {
>-  printf("VideoProcessor settings\n==\n");
>-  printf(" Total # of frames: %d", analysis_frame_reader_->NumberOfFrames());
>-  printf("%s\n", config_.ToString().c_str());
>-
>-  printf("VideoProcessorIntegrationTest settings\n==\n");
>-  const char* encoder_name = encoder_->ImplementationName();
>-  printf(" Encoder implementation name: %s\n", encoder_name);
>-  const char* decoder_name = decoder_->ImplementationName();
>-  printf(" Decoder implementation name: %s\n", decoder_name);
>-  if (strcmp(encoder_name, decoder_name) == 0) {
>-    printf(" Codec implementation name : %s_%s\n", config_.CodecName().c_str(),
>-           encoder_name);
>-  }
>-  printf("\n");
>-}
>-
>-void VideoProcessorIntegrationTest::VerifyBitstream(
>-    int frame_number,
>-    const BitstreamThresholds& bs_thresholds) {
>-  RTC_CHECK_GE(frame_number, 0);
>-  const FrameStatistic* frame_stat = stats_.GetFrame(frame_number);
>-  EXPECT_LE(*(frame_stat->max_nalu_length), bs_thresholds.max_nalu_length);
>-}
>-
>-void VideoProcessorIntegrationTest::VerifyQualityMetrics(
>-    const QualityThresholds& quality_thresholds) {
>-  EXPECT_GT(quality_.num_decoded_frames, 0);
>-  EXPECT_GT(quality_.total_psnr / quality_.num_decoded_frames,
>-            quality_thresholds.min_avg_psnr);
>-  EXPECT_GT(quality_.min_psnr, quality_thresholds.min_min_psnr);
>-  EXPECT_GT(quality_.total_ssim / quality_.num_decoded_frames,
>-            quality_thresholds.min_avg_ssim);
>-  EXPECT_GT(quality_.min_ssim, quality_thresholds.min_min_ssim);
>-}
>-
>-// Reset quantities before each encoder rate update.
>-void VideoProcessorIntegrationTest::ResetRateControlMetrics(
>-    int rate_update_index,
>-    const std::vector<RateProfile>& rate_profiles) {
>-  RTC_DCHECK_GT(rate_profiles.size(), rate_update_index);
>-  // Set new rates.
>-  target_.kbps = rate_profiles[rate_update_index].target_kbps;
>-  target_.fps = rate_profiles[rate_update_index].input_fps;
>-  SetRatesPerTemporalLayer();
>-
>-  // Set key frame target sizes.
>-  if (rate_update_index == 0) {
>-    target_.key_framesize_kbits_initial =
>-        0.5 * kInitialBufferSize * target_.kbps_layer[0];
>-  }
>-
>-  // Set maximum size of key frames, following setting in the VP8 wrapper.
>-  float max_key_size = kScaleKeyFrameSize * kOptimalBufferSize * target_.fps;
>-  // We don't know exact target size of the key frames (except for first one),
>-  // but the minimum in libvpx is ~|3 * per_frame_bandwidth| and maximum is
>-  // set by |max_key_size_ * per_frame_bandwidth|. Take middle point/average
>-  // as reference for mismatch. Note key frames always correspond to base
>-  // layer frame in this test.
>-  target_.key_framesize_kbits =
>-      0.5 * (3 + max_key_size) * target_.framesize_kbits_layer[0];
>-
>-  // Reset rate control metrics.
>-  actual_ = TestResults();
>-  actual_.num_frames_to_hit_target =  // Set to max number of frames.
>-      rate_profiles[rate_update_index].frame_index_rate_update;
>-}
>-
>-void VideoProcessorIntegrationTest::SetRatesPerTemporalLayer() {
>-  const int num_temporal_layers = config_.NumberOfTemporalLayers();
>-  RTC_DCHECK_LE(num_temporal_layers, kMaxNumTemporalLayers);
>-
>-  for (int i = 0; i < num_temporal_layers; ++i) {
>-    float bitrate_ratio;
>-    if (i > 0) {
>-      bitrate_ratio = kVp8LayerRateAlloction[num_temporal_layers - 1][i] -
>-                      kVp8LayerRateAlloction[num_temporal_layers - 1][i - 1];
>-    } else {
>-      bitrate_ratio = kVp8LayerRateAlloction[num_temporal_layers - 1][i];
>-    }
>-    target_.kbps_layer[i] = target_.kbps * bitrate_ratio;
>-    target_.fps_layer[i] =
>-        target_.fps / static_cast<float>(1 << (num_temporal_layers - 1));
>-  }
>-  if (num_temporal_layers == 3) {
>-    target_.fps_layer[2] = target_.fps / 2.0f;
>-  }
>-
>-  // Update layer per-frame-bandwidth.
>-  for (int i = 0; i < num_temporal_layers; ++i) {
>-    target_.framesize_kbits_layer[i] =
>-        target_.kbps_layer[i] / target_.fps_layer[i];
>-  }
>-}
>-
>-}  // namespace test
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest.h
>deleted file mode 100644
>index 10677da2bc9..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest.h
>+++ /dev/null
>@@ -1,229 +0,0 @@
>-/*
>- *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MODULES_VIDEO_CODING_CODECS_TEST_VIDEOPROCESSOR_INTEGRATIONTEST_H_
>-#define MODULES_VIDEO_CODING_CODECS_TEST_VIDEOPROCESSOR_INTEGRATIONTEST_H_
>-
>-#include <cmath>
>-#include <limits>
>-#include <memory>
>-#include <string>
>-#include <vector>
>-
>-#include "common_types.h"  // NOLINT(build/include)
>-#include "common_video/h264/h264_common.h"
>-#include "media/engine/webrtcvideodecoderfactory.h"
>-#include "media/engine/webrtcvideoencoderfactory.h"
>-#include "modules/video_coding/codecs/test/packet_manipulator.h"
>-#include "modules/video_coding/codecs/test/stats.h"
>-#include "modules/video_coding/codecs/test/test_config.h"
>-#include "modules/video_coding/codecs/test/videoprocessor.h"
>-#include "modules/video_coding/utility/ivf_file_writer.h"
>-#include "test/gtest.h"
>-#include "test/testsupport/frame_reader.h"
>-#include "test/testsupport/frame_writer.h"
>-#include "test/testsupport/packet_reader.h"
>-
>-namespace webrtc {
>-namespace test {
>-
>-// Rates for the encoder and the frame number when to change profile.
>-struct RateProfile {
>-  int target_kbps;
>-  int input_fps;
>-  int frame_index_rate_update;
>-};
>-
>-// Thresholds for the rate control metrics. The thresholds are defined for each
>-// rate update sequence. |max_num_frames_to_hit_target| is defined as number of
>-// frames, after a rate update is made to the encoder, for the encoder to reach
>-// |kMaxBitrateMismatchPercent| of new target rate.
>-struct RateControlThresholds {
>-  int max_num_dropped_frames;
>-  int max_key_framesize_mismatch_percent;
>-  int max_delta_framesize_mismatch_percent;
>-  int max_bitrate_mismatch_percent;
>-  int max_num_frames_to_hit_target;
>-  int num_spatial_resizes;
>-  int num_key_frames;
>-};
>-
>-// Thresholds for the quality metrics.
>-struct QualityThresholds {
>-  QualityThresholds(double min_avg_psnr,
>-                    double min_min_psnr,
>-                    double min_avg_ssim,
>-                    double min_min_ssim)
>-      : min_avg_psnr(min_avg_psnr),
>-        min_min_psnr(min_min_psnr),
>-        min_avg_ssim(min_avg_ssim),
>-        min_min_ssim(min_min_ssim) {}
>-  double min_avg_psnr;
>-  double min_min_psnr;
>-  double min_avg_ssim;
>-  double min_min_ssim;
>-};
>-
>-struct BitstreamThresholds {
>-  explicit BitstreamThresholds(size_t max_nalu_length)
>-      : max_nalu_length(max_nalu_length) {}
>-  size_t max_nalu_length;
>-};
>-
>-// Should video files be saved persistently to disk for post-run visualization?
>-struct VisualizationParams {
>-  bool save_encoded_ivf;
>-  bool save_decoded_y4m;
>-};
>-
>-// Integration test for video processor. Encodes+decodes a clip and
>-// writes it to the output directory. After completion, quality metrics
>-// (PSNR and SSIM) and rate control metrics are computed and compared to given
>-// thresholds, to verify that the quality and encoder response is acceptable.
>-// The rate control tests allow us to verify the behavior for changing bit rate,
>-// changing frame rate, frame dropping/spatial resize, and temporal layers.
>-// The thresholds for the rate control metrics are set to be fairly
>-// conservative, so failure should only happen when some significant regression
>-// or breakdown occurs.
>-class VideoProcessorIntegrationTest : public testing::Test {
>- protected:
>-  // Verifies that all H.264 keyframes contain SPS/PPS/IDR NALUs.
>-  class H264KeyframeChecker : public TestConfig::EncodedFrameChecker {
>-   public:
>-    void CheckEncodedFrame(webrtc::VideoCodecType codec,
>-                           const EncodedImage& encoded_frame) const override;
>-  };
>-
>-  VideoProcessorIntegrationTest();
>-  ~VideoProcessorIntegrationTest() override;
>-
>-  void ProcessFramesAndMaybeVerify(
>-      const std::vector<RateProfile>& rate_profiles,
>-      const std::vector<RateControlThresholds>* rc_thresholds,
>-      const QualityThresholds* quality_thresholds,
>-      const BitstreamThresholds* bs_thresholds,
>-      const VisualizationParams* visualization_params);
>-
>-  // Config.
>-  TestConfig config_;
>-
>-  // Can be used by all H.264 tests.
>-  const H264KeyframeChecker h264_keyframe_checker_;
>-
>- private:
>-  class CpuProcessTime;
>-  static const int kMaxNumTemporalLayers = 3;
>-
>-  struct TestResults {
>-    int KeyFrameSizeMismatchPercent() const {
>-      if (num_key_frames == 0) {
>-        return -1;
>-      }
>-      return 100 * sum_key_framesize_mismatch / num_key_frames;
>-    }
>-    int DeltaFrameSizeMismatchPercent(int i) const {
>-      return 100 * sum_delta_framesize_mismatch_layer[i] / num_frames_layer[i];
>-    }
>-    int BitrateMismatchPercent(float target_kbps) const {
>-      return 100 * std::fabs(kbps - target_kbps) / target_kbps;
>-    }
>-    int BitrateMismatchPercent(int i, float target_kbps_layer) const {
>-      return 100 * std::fabs(kbps_layer[i] - target_kbps_layer) /
>-          target_kbps_layer;
>-    }
>-    int num_frames = 0;
>-    int num_frames_layer[kMaxNumTemporalLayers] = {0};
>-    int num_key_frames = 0;
>-    int num_frames_to_hit_target = 0;
>-    float sum_framesize_kbits = 0.0f;
>-    float sum_framesize_kbits_layer[kMaxNumTemporalLayers] = {0};
>-    float kbps = 0.0f;
>-    float kbps_layer[kMaxNumTemporalLayers] = {0};
>-    float sum_key_framesize_mismatch = 0.0f;
>-    float sum_delta_framesize_mismatch_layer[kMaxNumTemporalLayers] = {0};
>-  };
>-
>-  struct TargetRates {
>-    int kbps;
>-    int fps;
>-    float kbps_layer[kMaxNumTemporalLayers];
>-    float fps_layer[kMaxNumTemporalLayers];
>-    float framesize_kbits_layer[kMaxNumTemporalLayers];
>-    float key_framesize_kbits_initial;
>-    float key_framesize_kbits;
>-  };
>-
>-  struct QualityMetrics {
>-    int num_decoded_frames = 0;
>-    double total_psnr = 0.0;
>-    double total_ssim = 0.0;
>-    double min_psnr = std::numeric_limits<double>::max();
>-    double min_ssim = std::numeric_limits<double>::max();
>-  };
>-
>-  void CreateEncoderAndDecoder();
>-  void DestroyEncoderAndDecoder();
>-  void SetUpAndInitObjects(rtc::TaskQueue* task_queue,
>-                           const int initial_bitrate_kbps,
>-                           const int initial_framerate_fps,
>-                           const VisualizationParams* visualization_params);
>-  void ReleaseAndCloseObjects(rtc::TaskQueue* task_queue);
>-
>-  // Rate control metrics.
>-  void ResetRateControlMetrics(int rate_update_index,
>-                               const std::vector<RateProfile>& rate_profiles);
>-  void SetRatesPerTemporalLayer();
>-  void UpdateRateControlMetrics(int frame_number);
>-  void PrintRateControlMetrics(
>-      int rate_update_index,
>-      const std::vector<int>& num_dropped_frames,
>-      const std::vector<int>& num_spatial_resizes) const;
>-  void VerifyRateControlMetrics(
>-      int rate_update_index,
>-      const std::vector<RateControlThresholds>* rc_thresholds,
>-      const std::vector<int>& num_dropped_frames,
>-      const std::vector<int>& num_spatial_resizes) const;
>-
>-  void VerifyBitstream(int frame_number,
>-                       const BitstreamThresholds& bs_thresholds);
>-
>-  void UpdateQualityMetrics(int frame_number);
>-  void VerifyQualityMetrics(const QualityThresholds& quality_thresholds);
>-
>-  void PrintSettings() const;
>-
>-  // Codecs.
>-  std::unique_ptr<VideoEncoder> encoder_;
>-  std::unique_ptr<VideoDecoder> decoder_;
>-
>-  // Helper objects.
>-  std::unique_ptr<FrameReader> analysis_frame_reader_;
>-  std::unique_ptr<FrameWriter> analysis_frame_writer_;
>-  std::unique_ptr<IvfFileWriter> encoded_frame_writer_;
>-  std::unique_ptr<FrameWriter> decoded_frame_writer_;
>-  PacketReader packet_reader_;
>-  std::unique_ptr<PacketManipulator> packet_manipulator_;
>-  Stats stats_;
>-  std::unique_ptr<VideoProcessor> processor_;
>-  std::unique_ptr<CpuProcessTime> cpu_process_time_;
>-
>-  // Quantities updated for every encoded frame.
>-  TestResults actual_;
>-
>-  // Rates set for every encoder rate update.
>-  TargetRates target_;
>-
>-  QualityMetrics quality_;
>-};
>-
>-}  // namespace test
>-}  // namespace webrtc
>-
>-#endif  // MODULES_VIDEO_CODING_CODECS_TEST_VIDEOPROCESSOR_INTEGRATIONTEST_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest_libvpx.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest_libvpx.cc
>deleted file mode 100644
>index ee4a8bb6caa..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest_libvpx.cc
>+++ /dev/null
>@@ -1,324 +0,0 @@
>-/*
>- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "modules/video_coding/codecs/test/videoprocessor_integrationtest.h"
>-
>-#include <vector>
>-
>-#include "modules/video_coding/codecs/test/test_config.h"
>-#include "rtc_base/ptr_util.h"
>-#include "test/testsupport/fileutils.h"
>-
>-namespace webrtc {
>-namespace test {
>-
>-namespace {
>-
>-// Codec settings.
>-const bool kResilienceOn = true;
>-const int kCifWidth = 352;
>-const int kCifHeight = 288;
>-#if !defined(WEBRTC_IOS)
>-const int kNumFramesShort = 100;
>-#endif
>-const int kNumFramesLong = 300;
>-
>-const std::nullptr_t kNoVisualizationParams = nullptr;
>-
>-}  // namespace
>-
>-class VideoProcessorIntegrationTestLibvpx
>-    : public VideoProcessorIntegrationTest {
>- protected:
>-  VideoProcessorIntegrationTestLibvpx() {
>-    config_.filename = "foreman_cif";
>-    config_.input_filename = ResourcePath(config_.filename, "yuv");
>-    config_.output_filename =
>-        TempFilename(OutputPath(), "videoprocessor_integrationtest_libvpx");
>-    config_.num_frames = kNumFramesLong;
>-    // Only allow encoder/decoder to use single core, for predictability.
>-    config_.use_single_core = true;
>-    config_.hw_encoder = false;
>-    config_.hw_decoder = false;
>-    config_.encoded_frame_checker = &qp_frame_checker_;
>-  }
>-
>- private:
>-  // Verify that the QP parser returns the same QP as the encoder does.
>-  const class QpFrameChecker : public TestConfig::EncodedFrameChecker {
>-   public:
>-    void CheckEncodedFrame(webrtc::VideoCodecType codec,
>-                           const EncodedImage& encoded_frame) const override {
>-      int qp;
>-      if (codec == kVideoCodecVP8) {
>-        EXPECT_TRUE(
>-            vp8::GetQp(encoded_frame._buffer, encoded_frame._length, &qp));
>-      } else if (codec == kVideoCodecVP9) {
>-        EXPECT_TRUE(
>-            vp9::GetQp(encoded_frame._buffer, encoded_frame._length, &qp));
>-      } else {
>-        RTC_NOTREACHED();
>-      }
>-      EXPECT_EQ(encoded_frame.qp_, qp) << "Encoder QP != parsed bitstream QP.";
>-    }
>-  } qp_frame_checker_;
>-};
>-
>-// Fails on iOS. See webrtc:4755.
>-#if !defined(WEBRTC_IOS)
>-
>-#if !defined(RTC_DISABLE_VP9)
>-// VP9: Run with no packet loss and fixed bitrate. Quality should be very high.
>-// One key frame (first frame only) in sequence.
>-TEST_F(VideoProcessorIntegrationTestLibvpx, Process0PercentPacketLossVP9) {
>-  config_.SetCodecSettings(kVideoCodecVP9, 1, false, false, true, false,
>-                           kResilienceOn, kCifWidth, kCifHeight);
>-  config_.num_frames = kNumFramesShort;
>-
>-  std::vector<RateProfile> rate_profiles = {{500, 30, kNumFramesShort + 1}};
>-
>-  std::vector<RateControlThresholds> rc_thresholds = {
>-      {0, 40, 20, 10, 20, 0, 1}};
>-
>-  QualityThresholds quality_thresholds(37.0, 36.0, 0.93, 0.92);
>-
>-  ProcessFramesAndMaybeVerify(rate_profiles, &rc_thresholds,
>-                              &quality_thresholds, nullptr,
>-                              kNoVisualizationParams);
>-}
>-
>-// VP9: Run with no packet loss, with varying bitrate (3 rate updates):
>-// low to high to medium. Check that quality and encoder response to the new
>-// target rate/per-frame bandwidth (for each rate update) is within limits.
>-// One key frame (first frame only) in sequence.
>-TEST_F(VideoProcessorIntegrationTestLibvpx, ProcessNoLossChangeBitRateVP9) {
>-  config_.SetCodecSettings(kVideoCodecVP9, 1, false, false, true, false,
>-                           kResilienceOn, kCifWidth, kCifHeight);
>-
>-  std::vector<RateProfile> rate_profiles = {
>-      {200, 30, 100},  // target_kbps, input_fps, frame_index_rate_update
>-      {700, 30, 200},
>-      {500, 30, kNumFramesLong + 1}};
>-
>-  std::vector<RateControlThresholds> rc_thresholds = {{0, 35, 20, 20, 35, 0, 1},
>-                                                      {2, 0, 20, 20, 60, 0, 0},
>-                                                      {0, 0, 25, 20, 40, 0, 0}};
>-
>-  QualityThresholds quality_thresholds(35.5, 30.0, 0.90, 0.85);
>-
>-  ProcessFramesAndMaybeVerify(rate_profiles, &rc_thresholds,
>-                              &quality_thresholds, nullptr,
>-                              kNoVisualizationParams);
>-}
>-
>-// VP9: Run with no packet loss, with an update (decrease) in frame rate.
>-// Lower frame rate means higher per-frame-bandwidth, so easier to encode.
>-// At the low bitrate in this test, this means better rate control after the
>-// update(s) to lower frame rate. So expect less frame drops, and max values
>-// for the rate control metrics can be lower. One key frame (first frame only).
>-// Note: quality after update should be higher but we currently compute quality
>-// metrics averaged over whole sequence run.
>-TEST_F(VideoProcessorIntegrationTestLibvpx,
>-       ProcessNoLossChangeFrameRateFrameDropVP9) {
>-  config_.SetCodecSettings(kVideoCodecVP9, 1, false, false, true, false,
>-                           kResilienceOn, kCifWidth, kCifHeight);
>-
>-  std::vector<RateProfile> rate_profiles = {
>-      {100, 24, 100},  // target_kbps, input_fps, frame_index_rate_update
>-      {100, 15, 200},
>-      {100, 10, kNumFramesLong + 1}};
>-
>-  std::vector<RateControlThresholds> rc_thresholds = {
>-      {45, 50, 95, 15, 45, 0, 1},
>-      {20, 0, 50, 10, 30, 0, 0},
>-      {5, 0, 30, 5, 25, 0, 0}};
>-
>-  QualityThresholds quality_thresholds(31.5, 18.0, 0.80, 0.43);
>-
>-  ProcessFramesAndMaybeVerify(rate_profiles, &rc_thresholds,
>-                              &quality_thresholds, nullptr,
>-                              kNoVisualizationParams);
>-}
>-
>-// VP9: Run with no packet loss and denoiser on. One key frame (first frame).
>-TEST_F(VideoProcessorIntegrationTestLibvpx, ProcessNoLossDenoiserOnVP9) {
>-  config_.SetCodecSettings(kVideoCodecVP9, 1, false, true, true, false,
>-                           kResilienceOn, kCifWidth, kCifHeight);
>-  config_.num_frames = kNumFramesShort;
>-
>-  std::vector<RateProfile> rate_profiles = {{500, 30, kNumFramesShort + 1}};
>-
>-  std::vector<RateControlThresholds> rc_thresholds = {
>-      {0, 40, 20, 10, 20, 0, 1}};
>-
>-  QualityThresholds quality_thresholds(36.8, 35.8, 0.92, 0.91);
>-
>-  ProcessFramesAndMaybeVerify(rate_profiles, &rc_thresholds,
>-                              &quality_thresholds, nullptr,
>-                              kNoVisualizationParams);
>-}
>-
>-// Run with no packet loss, at low bitrate.
>-// spatial_resize is on, for this low bitrate expect one resize in sequence.
>-// Resize happens on delta frame. Expect only one key frame (first frame).
>-TEST_F(VideoProcessorIntegrationTestLibvpx,
>-       DISABLED_ProcessNoLossSpatialResizeFrameDropVP9) {
>-  config_.SetCodecSettings(kVideoCodecVP9, 1, false, false, true, true,
>-                           kResilienceOn, kCifWidth, kCifHeight);
>-
>-  std::vector<RateProfile> rate_profiles = {{50, 30, kNumFramesLong + 1}};
>-
>-  std::vector<RateControlThresholds> rc_thresholds = {
>-      {228, 70, 160, 15, 80, 1, 1}};
>-
>-  QualityThresholds quality_thresholds(24.0, 13.0, 0.65, 0.37);
>-
>-  ProcessFramesAndMaybeVerify(rate_profiles, &rc_thresholds,
>-                              &quality_thresholds, nullptr,
>-                              kNoVisualizationParams);
>-}
>-
>-// TODO(marpan): Add temporal layer test for VP9, once changes are in
>-// vp9 wrapper for this.
>-
>-#endif  // !defined(RTC_DISABLE_VP9)
>-
>-// VP8: Run with no packet loss and fixed bitrate. Quality should be very high.
>-// One key frame (first frame only) in sequence. Setting |key_frame_interval|
>-// to -1 below means no periodic key frames in test.
>-TEST_F(VideoProcessorIntegrationTestLibvpx, ProcessZeroPacketLoss) {
>-  config_.SetCodecSettings(kVideoCodecVP8, 1, false, true, true, false,
>-                           kResilienceOn, kCifWidth, kCifHeight);
>-  config_.num_frames = kNumFramesShort;
>-
>-  std::vector<RateProfile> rate_profiles = {{500, 30, kNumFramesShort + 1}};
>-
>-  std::vector<RateControlThresholds> rc_thresholds = {
>-      {0, 40, 20, 10, 15, 0, 1}};
>-
>-  QualityThresholds quality_thresholds(34.95, 33.0, 0.90, 0.89);
>-
>-  ProcessFramesAndMaybeVerify(rate_profiles, &rc_thresholds,
>-                              &quality_thresholds, nullptr,
>-                              kNoVisualizationParams);
>-}
>-
>-#endif  // !defined(WEBRTC_IOS)
>-
>-// The tests below are currently disabled for Android. For ARM, the encoder
>-// uses |cpu_speed| = 12, as opposed to default |cpu_speed| <= 6 for x86,
>-// which leads to significantly different quality. The quality and rate control
>-// settings in the tests below are defined for encoder speed setting
>-// |cpu_speed| <= ~6. A number of settings would need to be significantly
>-// modified for the |cpu_speed| = 12 case. For now, keep the tests below
>-// disabled on Android. Some quality parameter in the above test has been
>-// adjusted to also pass for |cpu_speed| <= 12.
>-
>-// VP8: Run with no packet loss, with varying bitrate (3 rate updates):
>-// low to high to medium. Check that quality and encoder response to the new
>-// target rate/per-frame bandwidth (for each rate update) is within limits.
>-// One key frame (first frame only) in sequence.
>-// Too slow to finish before timeout on iOS. See webrtc:4755.
>-#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
>-#define MAYBE_ProcessNoLossChangeBitRateVP8 \
>-  DISABLED_ProcessNoLossChangeBitRateVP8
>-#else
>-#define MAYBE_ProcessNoLossChangeBitRateVP8 ProcessNoLossChangeBitRateVP8
>-#endif
>-TEST_F(VideoProcessorIntegrationTestLibvpx,
>-       MAYBE_ProcessNoLossChangeBitRateVP8) {
>-  config_.SetCodecSettings(kVideoCodecVP8, 1, false, true, true, false,
>-                           kResilienceOn, kCifWidth, kCifHeight);
>-
>-  std::vector<RateProfile> rate_profiles = {
>-      {200, 30, 100},  // target_kbps, input_fps, frame_index_rate_update
>-      {800, 30, 200},
>-      {500, 30, kNumFramesLong + 1}};
>-
>-  std::vector<RateControlThresholds> rc_thresholds = {{0, 45, 20, 10, 15, 0, 1},
>-                                                      {0, 0, 25, 20, 10, 0, 0},
>-                                                      {0, 0, 25, 15, 10, 0, 0}};
>-
>-  QualityThresholds quality_thresholds(34.0, 32.0, 0.85, 0.80);
>-
>-  ProcessFramesAndMaybeVerify(rate_profiles, &rc_thresholds,
>-                              &quality_thresholds, nullptr,
>-                              kNoVisualizationParams);
>-}
>-
>-// VP8: Run with no packet loss, with an update (decrease) in frame rate.
>-// Lower frame rate means higher per-frame-bandwidth, so easier to encode.
>-// At the bitrate in this test, this means better rate control after the
>-// update(s) to lower frame rate. So expect less frame drops, and max values
>-// for the rate control metrics can be lower. One key frame (first frame only).
>-// Note: quality after update should be higher but we currently compute quality
>-// metrics averaged over whole sequence run.
>-// Too slow to finish before timeout on iOS. See webrtc:4755.
>-#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
>-#define MAYBE_ProcessNoLossChangeFrameRateFrameDropVP8 \
>-  DISABLED_ProcessNoLossChangeFrameRateFrameDropVP8
>-#else
>-#define MAYBE_ProcessNoLossChangeFrameRateFrameDropVP8 \
>-  ProcessNoLossChangeFrameRateFrameDropVP8
>-#endif
>-TEST_F(VideoProcessorIntegrationTestLibvpx,
>-       MAYBE_ProcessNoLossChangeFrameRateFrameDropVP8) {
>-  config_.SetCodecSettings(kVideoCodecVP8, 1, false, true, true, false,
>-                           kResilienceOn, kCifWidth, kCifHeight);
>-
>-  std::vector<RateProfile> rate_profiles = {
>-      {80, 24, 100},  // target_kbps, input_fps, frame_index_rate_update
>-      {80, 15, 200},
>-      {80, 10, kNumFramesLong + 1}};
>-
>-  std::vector<RateControlThresholds> rc_thresholds = {
>-      {40, 20, 75, 15, 60, 0, 1},
>-      {10, 0, 25, 10, 35, 0, 0},
>-      {0, 0, 20, 10, 15, 0, 0}};
>-
>-  QualityThresholds quality_thresholds(31.0, 22.0, 0.80, 0.65);
>-
>-  ProcessFramesAndMaybeVerify(rate_profiles, &rc_thresholds,
>-                              &quality_thresholds, nullptr,
>-                              kNoVisualizationParams);
>-}
>-
>-// VP8: Run with no packet loss, with 3 temporal layers, with a rate update in
>-// the middle of the sequence. The max values for the frame size mismatch and
>-// encoding rate mismatch are applied to each layer.
>-// No dropped frames in this test, and internal spatial resizer is off.
>-// One key frame (first frame only) in sequence, so no spatial resizing.
>-// Too slow to finish before timeout on iOS. See webrtc:4755.
>-#if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
>-#define MAYBE_ProcessNoLossTemporalLayersVP8 \
>-  DISABLED_ProcessNoLossTemporalLayersVP8
>-#else
>-#define MAYBE_ProcessNoLossTemporalLayersVP8 ProcessNoLossTemporalLayersVP8
>-#endif
>-TEST_F(VideoProcessorIntegrationTestLibvpx,
>-       MAYBE_ProcessNoLossTemporalLayersVP8) {
>-  config_.SetCodecSettings(kVideoCodecVP8, 3, false, true, true, false,
>-                           kResilienceOn, kCifWidth, kCifHeight);
>-
>-  std::vector<RateProfile> rate_profiles = {{200, 30, 150},
>-                                            {400, 30, kNumFramesLong + 1}};
>-
>-  std::vector<RateControlThresholds> rc_thresholds = {{0, 20, 30, 10, 10, 0, 1},
>-                                                      {0, 0, 30, 15, 10, 0, 0}};
>-
>-  QualityThresholds quality_thresholds(32.5, 30.0, 0.85, 0.80);
>-
>-  ProcessFramesAndMaybeVerify(rate_profiles, &rc_thresholds,
>-                              &quality_thresholds, nullptr,
>-                              kNoVisualizationParams);
>-}
>-
>-}  // namespace test
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest_mediacodec.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest_mediacodec.cc
>deleted file mode 100644
>index 7f6b40b0b9f..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest_mediacodec.cc
>+++ /dev/null
>@@ -1,110 +0,0 @@
>-/*
>- *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "modules/video_coding/codecs/test/videoprocessor_integrationtest.h"
>-
>-#include <vector>
>-
>-#include "common_types.h"  // NOLINT(build/include)
>-#include "test/field_trial.h"
>-#include "test/testsupport/fileutils.h"
>-
>-namespace webrtc {
>-namespace test {
>-
>-#if defined(WEBRTC_ANDROID)
>-
>-namespace {
>-const int kForemanNumFrames = 300;
>-const std::nullptr_t kNoVisualizationParams = nullptr;
>-}  // namespace
>-
>-class VideoProcessorIntegrationTestMediaCodec
>-    : public VideoProcessorIntegrationTest {
>- protected:
>-  VideoProcessorIntegrationTestMediaCodec() {
>-    config_.filename = "foreman_cif";
>-    config_.input_filename = ResourcePath(config_.filename, "yuv");
>-    config_.output_filename =
>-        TempFilename(OutputPath(), "videoprocessor_integrationtest_mediacodec");
>-    config_.num_frames = kForemanNumFrames;
>-    config_.hw_encoder = true;
>-    config_.hw_decoder = true;
>-  }
>-};
>-
>-TEST_F(VideoProcessorIntegrationTestMediaCodec, ForemanCif500kbpsVp8) {
>-  config_.SetCodecSettings(kVideoCodecVP8, 1, false, false, false, false, false,
>-                           352, 288);
>-
>-  std::vector<RateProfile> rate_profiles = {{500, 30, kForemanNumFrames + 1}};
>-
>-  // The thresholds below may have to be tweaked to let even poor MediaCodec
>-  // implementations pass. If this test fails on the bots, disable it and
>-  // ping brandtr@.
>-  std::vector<RateControlThresholds> rc_thresholds = {
>-      {20, 95, 22, 11, 50, 0, 1}};
>-
>-  QualityThresholds quality_thresholds(30.0, 14.0, 0.86, 0.39);
>-
>-  ProcessFramesAndMaybeVerify(rate_profiles, &rc_thresholds,
>-                              &quality_thresholds, nullptr,
>-                              kNoVisualizationParams);
>-}
>-
>-TEST_F(VideoProcessorIntegrationTestMediaCodec, ForemanCif500kbpsH264CBP) {
>-  config_.encoded_frame_checker = &h264_keyframe_checker_;
>-  config_.SetCodecSettings(kVideoCodecH264, 1, false, false, false, false,
>-                           false, 352, 288);
>-
>-  std::vector<RateProfile> rate_profiles = {{500, 30, kForemanNumFrames + 1}};
>-
>-  // The thresholds below may have to be tweaked to let even poor MediaCodec
>-  // implementations pass. If this test fails on the bots, disable it and
>-  // ping brandtr@.
>-  std::vector<RateControlThresholds> rc_thresholds = {
>-      {20, 95, 22, 11, 20, 0, 1}};
>-
>-  QualityThresholds quality_thresholds(30.0, 14.0, 0.86, 0.39);
>-
>-  ProcessFramesAndMaybeVerify(rate_profiles, &rc_thresholds,
>-                              &quality_thresholds, nullptr,
>-                              kNoVisualizationParams);
>-}
>-
>-// TODO(brandtr): Enable this test when we have trybots/buildbots with
>-// HW encoders that support CHP.
>-TEST_F(VideoProcessorIntegrationTestMediaCodec,
>-       DISABLED_ForemanCif500kbpsH264CHP) {
>-  ScopedFieldTrials override_field_trials("WebRTC-H264HighProfile/Enabled/");
>-
>-  config_.h264_codec_settings.profile = H264::kProfileConstrainedHigh;
>-  config_.encoded_frame_checker = &h264_keyframe_checker_;
>-  config_.SetCodecSettings(kVideoCodecH264, 1, false, false, false, false,
>-                           false, 352, 288);
>-
>-  std::vector<RateProfile> rate_profiles = {{500, 30, kForemanNumFrames + 1}};
>-
>-  // The thresholds below may have to be tweaked to let even poor MediaCodec
>-  // implementations pass. If this test fails on the bots, disable it and
>-  // ping brandtr@.
>-  std::vector<RateControlThresholds> rc_thresholds = {{5, 60, 20, 5, 15, 0, 1}};
>-
>-  QualityThresholds quality_thresholds(33.0, 30.0, 0.90, 0.85);
>-
>-  ProcessFramesAndMaybeVerify(rate_profiles, &rc_thresholds,
>-                              &quality_thresholds, nullptr,
>-                              kNoVisualizationParams);
>-}
>-
>-#endif  // defined(WEBRTC_ANDROID)
>-
>-}  // namespace test
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest_openh264.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest_openh264.cc
>deleted file mode 100644
>index 9967fbddfaa..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest_openh264.cc
>+++ /dev/null
>@@ -1,100 +0,0 @@
>-/*
>- *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "modules/video_coding/codecs/test/videoprocessor_integrationtest.h"
>-
>-#include <vector>
>-
>-#include "test/testsupport/fileutils.h"
>-
>-namespace webrtc {
>-namespace test {
>-
>-#if defined(WEBRTC_USE_H264)
>-
>-namespace {
>-
>-// Codec settings.
>-const bool kResilienceOn = true;
>-const int kCifWidth = 352;
>-const int kCifHeight = 288;
>-const int kNumFrames = 100;
>-
>-const std::nullptr_t kNoVisualizationParams = nullptr;
>-
>-}  // namespace
>-
>-class VideoProcessorIntegrationTestOpenH264
>-    : public VideoProcessorIntegrationTest {
>- protected:
>-  VideoProcessorIntegrationTestOpenH264() {
>-    config_.filename = "foreman_cif";
>-    config_.input_filename = ResourcePath(config_.filename, "yuv");
>-    config_.output_filename =
>-        TempFilename(OutputPath(), "videoprocessor_integrationtest_libvpx");
>-    config_.num_frames = kNumFrames;
>-    config_.networking_config.packet_loss_probability = 0.0;
>-    // Only allow encoder/decoder to use single core, for predictability.
>-    config_.use_single_core = true;
>-    config_.hw_encoder = false;
>-    config_.hw_decoder = false;
>-    config_.encoded_frame_checker = &h264_keyframe_checker_;
>-  }
>-};
>-
>-// H264: Run with no packet loss and fixed bitrate. Quality should be very high.
>-// Note(hbos): The PacketManipulatorImpl code used to simulate packet loss in
>-// these unittests appears to drop "packets" in a way that is not compatible
>-// with H264. Therefore ProcessXPercentPacketLossH264, X != 0, unittests have
>-// not been added.
>-TEST_F(VideoProcessorIntegrationTestOpenH264, Process0PercentPacketLoss) {
>-  config_.SetCodecSettings(kVideoCodecH264, 1, false, false, true, false,
>-                           kResilienceOn, kCifWidth, kCifHeight);
>-
>-  std::vector<RateProfile> rate_profiles = {{500, 30, kNumFrames + 1}};
>-
>-  std::vector<RateControlThresholds> rc_thresholds = {
>-      {2, 60, 20, 10, 20, 0, 1}};
>-
>-  QualityThresholds quality_thresholds(35.0, 25.0, 0.93, 0.70);
>-
>-  ProcessFramesAndMaybeVerify(rate_profiles, &rc_thresholds,
>-                              &quality_thresholds, nullptr,
>-                              kNoVisualizationParams);
>-}
>-
>-// H264: Enable SingleNalUnit packetization mode. Encoder should split
>-// large frames into multiple slices and limit length of NAL units.
>-TEST_F(VideoProcessorIntegrationTestOpenH264, ProcessNoLossSingleNalUnit) {
>-  config_.h264_codec_settings.packetization_mode =
>-      H264PacketizationMode::SingleNalUnit;
>-  config_.networking_config.max_payload_size_in_bytes = 500;
>-  config_.SetCodecSettings(kVideoCodecH264, 1, false, false, true, false,
>-                           kResilienceOn, kCifWidth, kCifHeight);
>-
>-  std::vector<RateProfile> rate_profiles = {{500, 30, kNumFrames + 1}};
>-
>-  std::vector<RateControlThresholds> rc_thresholds = {
>-      {2, 60, 30, 10, 20, 0, 1}};
>-
>-  QualityThresholds quality_thresholds(35.0, 25.0, 0.93, 0.70);
>-
>-  BitstreamThresholds bs_thresholds(
>-      config_.networking_config.max_payload_size_in_bytes);
>-
>-  ProcessFramesAndMaybeVerify(rate_profiles, &rc_thresholds,
>-                              &quality_thresholds, &bs_thresholds,
>-                              kNoVisualizationParams);
>-}
>-
>-#endif  // defined(WEBRTC_USE_H264)
>-
>-}  // namespace test
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest_parameterized.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest_parameterized.cc
>deleted file mode 100644
>index b058e2992ad..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest_parameterized.cc
>+++ /dev/null
>@@ -1,113 +0,0 @@
>-/*
>- *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "modules/video_coding/codecs/test/videoprocessor_integrationtest.h"
>-
>-#include "test/testsupport/fileutils.h"
>-
>-namespace webrtc {
>-namespace test {
>-
>-namespace {
>-
>-// Loop variables.
>-const int kBitrates[] = {500};
>-const VideoCodecType kVideoCodecType[] = {kVideoCodecVP8};
>-const bool kHwCodec[] = {false};
>-
>-// Codec settings.
>-const bool kResilienceOn = false;
>-const int kNumTemporalLayers = 1;
>-const bool kDenoisingOn = false;
>-const bool kErrorConcealmentOn = false;
>-const bool kSpatialResizeOn = false;
>-const bool kFrameDropperOn = false;
>-
>-// Test settings.
>-const bool kUseSingleCore = false;
>-const bool kMeasureCpu = false;
>-const VisualizationParams kVisualizationParams = {
>-    false,  // save_encoded_ivf
>-    false,  // save_decoded_y4m
>-};
>-
>-const int kNumFrames = 30;
>-
>-}  // namespace
>-
>-// Tests for plotting statistics from logs.
>-class VideoProcessorIntegrationTestParameterized
>-    : public VideoProcessorIntegrationTest,
>-      public ::testing::WithParamInterface<
>-          ::testing::tuple<int, VideoCodecType, bool>> {
>- protected:
>-  VideoProcessorIntegrationTestParameterized()
>-      : bitrate_(::testing::get<0>(GetParam())),
>-        codec_type_(::testing::get<1>(GetParam())),
>-        hw_codec_(::testing::get<2>(GetParam())) {}
>-  ~VideoProcessorIntegrationTestParameterized() override = default;
>-
>-  void RunTest(int width,
>-               int height,
>-               int framerate,
>-               const std::string& filename) {
>-    config_.filename = filename;
>-    config_.input_filename = ResourcePath(filename, "yuv");
>-    config_.output_filename =
>-        TempFilename(OutputPath(), "plot_videoprocessor_integrationtest");
>-    config_.use_single_core = kUseSingleCore;
>-    config_.measure_cpu = kMeasureCpu;
>-    config_.hw_encoder = hw_codec_;
>-    config_.hw_decoder = hw_codec_;
>-    config_.num_frames = kNumFrames;
>-    config_.SetCodecSettings(codec_type_, kNumTemporalLayers,
>-                             kErrorConcealmentOn, kDenoisingOn, kFrameDropperOn,
>-                             kSpatialResizeOn, kResilienceOn, width, height);
>-
>-    std::vector<RateProfile> rate_profiles = {
>-        {bitrate_, framerate, kNumFrames + 1}};
>-
>-    ProcessFramesAndMaybeVerify(rate_profiles, nullptr, nullptr, nullptr,
>-                                &kVisualizationParams);
>-  }
>-
>-  const int bitrate_;
>-  const VideoCodecType codec_type_;
>-  const bool hw_codec_;
>-};
>-
>-INSTANTIATE_TEST_CASE_P(CodecSettings,
>-                        VideoProcessorIntegrationTestParameterized,
>-                        ::testing::Combine(::testing::ValuesIn(kBitrates),
>-                                           ::testing::ValuesIn(kVideoCodecType),
>-                                           ::testing::ValuesIn(kHwCodec)));
>-
>-TEST_P(VideoProcessorIntegrationTestParameterized, Process_128x96_30fps) {
>-  RunTest(128, 96, 30, "foreman_128x96");
>-}
>-
>-TEST_P(VideoProcessorIntegrationTestParameterized, Process_160x120_30fps) {
>-  RunTest(160, 120, 30, "foreman_160x120");
>-}
>-
>-TEST_P(VideoProcessorIntegrationTestParameterized, Process_176x144_30fps) {
>-  RunTest(176, 144, 30, "foreman_176x144");
>-}
>-
>-TEST_P(VideoProcessorIntegrationTestParameterized, Process_320x240_30fps) {
>-  RunTest(320, 240, 30, "foreman_320x240");
>-}
>-
>-TEST_P(VideoProcessorIntegrationTestParameterized, Process_352x288_30fps) {
>-  RunTest(352, 288, 30, "foreman_cif");
>-}
>-
>-}  // namespace test
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest_videotoolbox.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest_videotoolbox.cc
>deleted file mode 100644
>index 2e040a49645..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videoprocessor_integrationtest_videotoolbox.cc
>+++ /dev/null
>@@ -1,85 +0,0 @@
>-/*
>- *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "modules/video_coding/codecs/test/videoprocessor_integrationtest.h"
>-
>-#include <vector>
>-
>-#include "test/field_trial.h"
>-#include "test/testsupport/fileutils.h"
>-
>-namespace webrtc {
>-namespace test {
>-
>-#if defined(WEBRTC_IOS)
>-
>-namespace {
>-const int kForemanNumFrames = 300;
>-const std::nullptr_t kNoVisualizationParams = nullptr;
>-}  // namespace
>-
>-class VideoProcessorIntegrationTestVideoToolbox
>-    : public VideoProcessorIntegrationTest {
>- protected:
>-  VideoProcessorIntegrationTestVideoToolbox() {
>-    config_.filename = "foreman_cif";
>-    config_.input_filename = ResourcePath(config_.filename, "yuv");
>-    config_.output_filename = TempFilename(
>-        OutputPath(), "videoprocessor_integrationtest_videotoolbox");
>-    config_.num_frames = kForemanNumFrames;
>-    config_.hw_encoder = true;
>-    config_.hw_decoder = true;
>-    config_.encoded_frame_checker = &h264_keyframe_checker_;
>-  }
>-};
>-
>-// Since we don't currently run the iOS tests on physical devices on the bots,
>-// the tests are disabled.
>-
>-TEST_F(VideoProcessorIntegrationTestVideoToolbox,
>-       DISABLED_ForemanCif500kbpsH264CBP) {
>-  config_.SetCodecSettings(kVideoCodecH264, 1, false, false, false, false,
>-                           false, 352, 288);
>-
>-  std::vector<RateProfile> rate_profiles = {{500, 30, kForemanNumFrames + 1}};
>-
>-  std::vector<RateControlThresholds> rc_thresholds = {
>-      {20, 95, 60, 60, 10, 0, 1}};
>-
>-  QualityThresholds quality_thresholds(30.0, 14.0, 0.86, 0.39);
>-
>-  ProcessFramesAndMaybeVerify(rate_profiles, &rc_thresholds,
>-                              &quality_thresholds, nullptr,
>-                              kNoVisualizationParams);
>-}
>-
>-TEST_F(VideoProcessorIntegrationTestVideoToolbox,
>-       DISABLED_ForemanCif500kbpsH264CHP) {
>-  ScopedFieldTrials override_field_trials("WebRTC-H264HighProfile/Enabled/");
>-
>-  config_.h264_codec_settings.profile = H264::kProfileConstrainedHigh;
>-  config_.SetCodecSettings(kVideoCodecH264, 1, false, false, false, false,
>-                           false, 352, 288);
>-
>-  std::vector<RateProfile> rate_profiles = {{500, 30, kForemanNumFrames + 1}};
>-
>-  std::vector<RateControlThresholds> rc_thresholds = {{5, 75, 65, 60, 6, 0, 1}};
>-
>-  QualityThresholds quality_thresholds(30.0, 14.0, 0.86, 0.39);
>-
>-  ProcessFramesAndMaybeVerify(rate_profiles, &rc_thresholds,
>-                              &quality_thresholds, nullptr,
>-                              kNoVisualizationParams);
>-}
>-
>-#endif  // defined(WEBRTC_IOS)
>-
>-}  // namespace test
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videoprocessor_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videoprocessor_unittest.cc
>index 1a51df223de..589a3d898a8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videoprocessor_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/videoprocessor_unittest.cc
>@@ -10,23 +10,22 @@
> 
> #include <memory>
> 
>+#include "absl/memory/memory.h"
>+#include "api/test/videocodec_test_fixture.h"
> #include "api/video/i420_buffer.h"
> #include "common_types.h"  // NOLINT(build/include)
>-#include "modules/video_coding/codecs/test/mock/mock_packet_manipulator.h"
>+#include "media/base/mediaconstants.h"
>+#include "modules/video_coding/codecs/test/videocodec_test_stats_impl.h"
> #include "modules/video_coding/codecs/test/videoprocessor.h"
> #include "modules/video_coding/include/mock/mock_video_codec_interface.h"
> #include "modules/video_coding/include/video_coding.h"
>-#include "rtc_base/ptr_util.h"
>+#include "rtc_base/event.h"
>+#include "rtc_base/task_queue_for_test.h"
> #include "test/gmock.h"
> #include "test/gtest.h"
> #include "test/testsupport/mock/mock_frame_reader.h"
>-#include "test/testsupport/packet_reader.h"
>-#include "test/testsupport/unittest_utils.h"
>-#include "test/video_codec_settings.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> using ::testing::_;
>-using ::testing::ElementsAre;
> using ::testing::Property;
> using ::testing::Return;
> 
>@@ -43,42 +42,51 @@ const int kFrameSize = kWidth * kHeight * 3 / 2;  // I420.
> 
> class VideoProcessorTest : public testing::Test {
>  protected:
>-  VideoProcessorTest() {
>-    // Get a codec configuration struct and configure it.
>-    webrtc::test::CodecSettings(kVideoCodecVP8, &config_.codec_settings);
>-    config_.codec_settings.width = kWidth;
>-    config_.codec_settings.height = kHeight;
>+  VideoProcessorTest() : q_("VP queue") {
>+    config_.SetCodecSettings(cricket::kVp8CodecName, 1, 1, 1, false, false,
>+                             false, kWidth, kHeight);
>+
>+    decoder_mock_ = new MockVideoDecoder();
>+    decoders_.push_back(std::unique_ptr<VideoDecoder>(decoder_mock_));
> 
>     ExpectInit();
>     EXPECT_CALL(frame_reader_mock_, FrameLength())
>         .WillRepeatedly(Return(kFrameSize));
>-    video_processor_ = rtc::MakeUnique<VideoProcessor>(
>-        &encoder_mock_, &decoder_mock_, &frame_reader_mock_,
>-        &packet_manipulator_mock_, config_, &stats_,
>-        nullptr /* encoded_frame_writer */, nullptr /* decoded_frame_writer */);
>+    q_.SendTask([this] {
>+      video_processor_ = absl::make_unique<VideoProcessor>(
>+          &encoder_mock_, &decoders_, &frame_reader_mock_, config_, &stats_,
>+          nullptr /* encoded_frame_writer */,
>+          nullptr /* decoded_frame_writer */);
>+    });
>+  }
>+
>+  ~VideoProcessorTest() {
>+    q_.SendTask([this] { video_processor_.reset(); });
>   }
> 
>   void ExpectInit() {
>     EXPECT_CALL(encoder_mock_, InitEncode(_, _, _)).Times(1);
>     EXPECT_CALL(encoder_mock_, RegisterEncodeCompleteCallback(_)).Times(1);
>-    EXPECT_CALL(decoder_mock_, InitDecode(_, _)).Times(1);
>-    EXPECT_CALL(decoder_mock_, RegisterDecodeCompleteCallback(_)).Times(1);
>+    EXPECT_CALL(*decoder_mock_, InitDecode(_, _)).Times(1);
>+    EXPECT_CALL(*decoder_mock_, RegisterDecodeCompleteCallback(_)).Times(1);
>   }
> 
>   void ExpectRelease() {
>     EXPECT_CALL(encoder_mock_, Release()).Times(1);
>     EXPECT_CALL(encoder_mock_, RegisterEncodeCompleteCallback(_)).Times(1);
>-    EXPECT_CALL(decoder_mock_, Release()).Times(1);
>-    EXPECT_CALL(decoder_mock_, RegisterDecodeCompleteCallback(_)).Times(1);
>+    EXPECT_CALL(*decoder_mock_, Release()).Times(1);
>+    EXPECT_CALL(*decoder_mock_, RegisterDecodeCompleteCallback(_)).Times(1);
>   }
> 
>-  TestConfig config_;
>+  rtc::test::TaskQueueForTest q_;
>+
>+  VideoCodecTestFixture::Config config_;
> 
>   MockVideoEncoder encoder_mock_;
>-  MockVideoDecoder decoder_mock_;
>+  MockVideoDecoder* decoder_mock_;
>+  std::vector<std::unique_ptr<VideoDecoder>> decoders_;
>   MockFrameReader frame_reader_mock_;
>-  MockPacketManipulator packet_manipulator_mock_;
>-  Stats stats_;
>+  VideoCodecTestStatsImpl stats_;
>   std::unique_ptr<VideoProcessor> video_processor_;
> };
> 
>@@ -92,7 +100,7 @@ TEST_F(VideoProcessorTest, ProcessFrames_FixedFramerate) {
>   EXPECT_CALL(encoder_mock_, SetRateAllocation(_, kFramerateFps))
>       .Times(1)
>       .WillOnce(Return(0));
>-  video_processor_->SetRates(kBitrateKbps, kFramerateFps);
>+  q_.SendTask([=] { video_processor_->SetRates(kBitrateKbps, kFramerateFps); });
> 
>   EXPECT_CALL(frame_reader_mock_, ReadFrame())
>       .WillRepeatedly(Return(I420Buffer::Create(kWidth, kHeight)));
>@@ -100,13 +108,13 @@ TEST_F(VideoProcessorTest, ProcessFrames_FixedFramerate) {
>       encoder_mock_,
>       Encode(Property(&VideoFrame::timestamp, 1 * 90000 / kFramerateFps), _, _))
>       .Times(1);
>-  video_processor_->ProcessFrame();
>+  q_.SendTask([this] { video_processor_->ProcessFrame(); });
> 
>   EXPECT_CALL(
>       encoder_mock_,
>       Encode(Property(&VideoFrame::timestamp, 2 * 90000 / kFramerateFps), _, _))
>       .Times(1);
>-  video_processor_->ProcessFrame();
>+  q_.SendTask([this] { video_processor_->ProcessFrame(); });
> 
>   ExpectRelease();
> }
>@@ -114,30 +122,33 @@ TEST_F(VideoProcessorTest, ProcessFrames_FixedFramerate) {
> TEST_F(VideoProcessorTest, ProcessFrames_VariableFramerate) {
>   const int kBitrateKbps = 456;
>   const int kStartFramerateFps = 27;
>+  const int kStartTimestamp = 90000 / kStartFramerateFps;
>   EXPECT_CALL(encoder_mock_, SetRateAllocation(_, kStartFramerateFps))
>       .Times(1)
>       .WillOnce(Return(0));
>-  video_processor_->SetRates(kBitrateKbps, kStartFramerateFps);
>+  q_.SendTask(
>+      [=] { video_processor_->SetRates(kBitrateKbps, kStartFramerateFps); });
> 
>   EXPECT_CALL(frame_reader_mock_, ReadFrame())
>       .WillRepeatedly(Return(I420Buffer::Create(kWidth, kHeight)));
>-  EXPECT_CALL(encoder_mock_, Encode(Property(&VideoFrame::timestamp,
>-                                             1 * 90000 / kStartFramerateFps),
>-                                    _, _))
>+  EXPECT_CALL(encoder_mock_,
>+              Encode(Property(&VideoFrame::timestamp, kStartTimestamp), _, _))
>       .Times(1);
>-  video_processor_->ProcessFrame();
>+  q_.SendTask([this] { video_processor_->ProcessFrame(); });
> 
>   const int kNewFramerateFps = 13;
>   EXPECT_CALL(encoder_mock_, SetRateAllocation(_, kNewFramerateFps))
>       .Times(1)
>       .WillOnce(Return(0));
>-  video_processor_->SetRates(kBitrateKbps, kNewFramerateFps);
>+  q_.SendTask(
>+      [=] { video_processor_->SetRates(kBitrateKbps, kNewFramerateFps); });
> 
>-  EXPECT_CALL(encoder_mock_, Encode(Property(&VideoFrame::timestamp,
>-                                             2 * 90000 / kNewFramerateFps),
>-                                    _, _))
>+  EXPECT_CALL(encoder_mock_,
>+              Encode(Property(&VideoFrame::timestamp,
>+                              kStartTimestamp + 90000 / kNewFramerateFps),
>+                     _, _))
>       .Times(1);
>-  video_processor_->ProcessFrame();
>+  q_.SendTask([this] { video_processor_->ProcessFrame(); });
> 
>   ExpectRelease();
> }
>@@ -147,27 +158,20 @@ TEST_F(VideoProcessorTest, SetRates) {
>   const int kFramerateFps = 17;
>   EXPECT_CALL(encoder_mock_,
>               SetRateAllocation(
>-                  Property(&BitrateAllocation::get_sum_kbps, kBitrateKbps),
>+                  Property(&VideoBitrateAllocation::get_sum_kbps, kBitrateKbps),
>                   kFramerateFps))
>       .Times(1);
>-  video_processor_->SetRates(kBitrateKbps, kFramerateFps);
>-  EXPECT_THAT(video_processor_->NumberDroppedFramesPerRateUpdate(),
>-              ElementsAre(0));
>-  EXPECT_THAT(video_processor_->NumberSpatialResizesPerRateUpdate(),
>-              ElementsAre(0));
>+  q_.SendTask([=] { video_processor_->SetRates(kBitrateKbps, kFramerateFps); });
> 
>   const int kNewBitrateKbps = 456;
>   const int kNewFramerateFps = 34;
>   EXPECT_CALL(encoder_mock_,
>-              SetRateAllocation(
>-                  Property(&BitrateAllocation::get_sum_kbps, kNewBitrateKbps),
>-                  kNewFramerateFps))
>+              SetRateAllocation(Property(&VideoBitrateAllocation::get_sum_kbps,
>+                                         kNewBitrateKbps),
>+                                kNewFramerateFps))
>       .Times(1);
>-  video_processor_->SetRates(kNewBitrateKbps, kNewFramerateFps);
>-  EXPECT_THAT(video_processor_->NumberDroppedFramesPerRateUpdate(),
>-              ElementsAre(0, 0));
>-  EXPECT_THAT(video_processor_->NumberSpatialResizesPerRateUpdate(),
>-              ElementsAre(0, 0));
>+  q_.SendTask(
>+      [=] { video_processor_->SetRates(kNewBitrateKbps, kNewFramerateFps); });
> 
>   ExpectRelease();
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.cc
>index f070750e7bc..8389eff16e8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.cc
>@@ -13,20 +13,18 @@
> #include <string.h>
> 
> #include <algorithm>
>+#include <array>
> #include <memory>
> #include <set>
>+#include <utility>
> #include <vector>
> 
> #include "modules/include/module_common_types.h"
>-#include "modules/video_coding/codecs/vp8/include/vp8_common_types.h"
> #include "modules/video_coding/include/video_codec_interface.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> #include "system_wrappers/include/field_trial.h"
> 
>-#include "vpx/vp8cx.h"
>-#include "vpx/vpx_encoder.h"
>-
> namespace webrtc {
> 
> TemporalLayers::FrameConfig::FrameConfig()
>@@ -56,9 +54,16 @@ TemporalLayers::FrameConfig::FrameConfig(TemporalLayers::BufferFlags last,
>       encoder_layer_id(0),
>       packetizer_temporal_idx(kNoTemporalIdx),
>       layer_sync(false),
>-      freeze_entropy(freeze_entropy) {}
>+      freeze_entropy(freeze_entropy),
>+      first_reference(Vp8BufferReference::kNone),
>+      second_reference(Vp8BufferReference::kNone) {}
> 
> namespace {
>+static constexpr uint8_t kUninitializedPatternIndex =
>+    std::numeric_limits<uint8_t>::max();
>+static constexpr std::array<Vp8BufferReference, 3> kAllBuffers = {
>+    {Vp8BufferReference::kLast, Vp8BufferReference::kGolden,
>+     Vp8BufferReference::kAltref}};
> 
> std::vector<unsigned int> GetTemporalIds(size_t num_layers) {
>   switch (num_layers) {
>@@ -92,29 +97,40 @@ std::vector<unsigned int> GetTemporalIds(size_t num_layers) {
>   return {0};
> }
> 
>-std::vector<bool> GetTemporalLayerSync(size_t num_layers) {
>-  switch (num_layers) {
>-    case 1:
>-      return {false};
>-    case 2:
>-      return {false, true, false, false, false, false, false, false};
>-    case 3:
>-      if (field_trial::IsEnabled("WebRTC-UseShortVP8TL3Pattern")) {
>-        return {false, true, true, false};
>-      } else {
>-        return {false, true, true, false, false, false, false, false};
>+uint8_t GetUpdatedBuffers(const TemporalLayers::FrameConfig& config) {
>+  uint8_t flags = 0;
>+  if (config.last_buffer_flags & TemporalLayers::BufferFlags::kUpdate) {
>+    flags |= static_cast<uint8_t>(Vp8BufferReference::kLast);
>+  }
>+  if (config.golden_buffer_flags & TemporalLayers::BufferFlags::kUpdate) {
>+    flags |= static_cast<uint8_t>(Vp8BufferReference::kGolden);
>+  }
>+  if (config.arf_buffer_flags & TemporalLayers::BufferFlags::kUpdate) {
>+    flags |= static_cast<uint8_t>(Vp8BufferReference::kAltref);
>+  }
>+  return flags;
>+}
>+
>+// Find the set of buffers that are never updated by the given pattern.
>+std::set<Vp8BufferReference> FindKfBuffers(
>+    const std::vector<TemporalLayers::FrameConfig>& frame_configs) {
>+  std::set<Vp8BufferReference> kf_buffers(kAllBuffers.begin(),
>+                                          kAllBuffers.end());
>+  for (TemporalLayers::FrameConfig config : frame_configs) {
>+    // Get bit-masked set of update buffers for this frame config.
>+    uint8_t updated_buffers = GetUpdatedBuffers(config);
>+    for (Vp8BufferReference buffer : kAllBuffers) {
>+      if (static_cast<uint8_t>(buffer) & updated_buffers) {
>+        kf_buffers.erase(buffer);
>       }
>-    case 4:
>-      return {false, true,  true,  false, true,  false, false, false,
>-              false, false, false, false, false, false, false, false};
>-    default:
>-      break;
>+    }
>   }
>-  RTC_NOTREACHED() << num_layers;
>-  return {};
>+  return kf_buffers;
> }
>+}  // namespace
> 
>-std::vector<TemporalLayers::FrameConfig> GetTemporalPattern(size_t num_layers) {
>+std::vector<TemporalLayers::FrameConfig>
>+DefaultTemporalLayers::GetTemporalPattern(size_t num_layers) {
>   // For indexing in the patterns described below (which temporal layers they
>   // belong to), see the diagram above.
>   // Layer sync is done similarly for all patterns (except single stream) and
>@@ -131,38 +147,37 @@ std::vector<TemporalLayers::FrameConfig> GetTemporalPattern(size_t num_layers) {
>   switch (num_layers) {
>     case 1:
>       // All frames reference all buffers and the 'last' buffer is updated.
>-      return {TemporalLayers::FrameConfig(TemporalLayers::kReferenceAndUpdate,
>-                                          TemporalLayers::kReference,
>-                                          TemporalLayers::kReference)};
>+      return {FrameConfig(kReferenceAndUpdate, kReference, kReference)};
>     case 2:
>       // All layers can reference but not update the 'alt' buffer, this means
>       // that the 'alt' buffer reference is effectively the last keyframe.
>       // TL0 also references and updates the 'last' buffer.
>       // TL1 also references 'last' and references and updates 'golden'.
>-      return {TemporalLayers::FrameConfig(TemporalLayers::kReferenceAndUpdate,
>-                                          TemporalLayers::kNone,
>-                                          TemporalLayers::kReference),
>-              TemporalLayers::FrameConfig(TemporalLayers::kReference,
>-                                          TemporalLayers::kUpdate,
>-                                          TemporalLayers::kReference),
>-              TemporalLayers::FrameConfig(TemporalLayers::kReferenceAndUpdate,
>-                                          TemporalLayers::kNone,
>-                                          TemporalLayers::kReference),
>-              TemporalLayers::FrameConfig(TemporalLayers::kReference,
>-                                          TemporalLayers::kReferenceAndUpdate,
>-                                          TemporalLayers::kReference),
>-              TemporalLayers::FrameConfig(TemporalLayers::kReferenceAndUpdate,
>-                                          TemporalLayers::kNone,
>-                                          TemporalLayers::kReference),
>-              TemporalLayers::FrameConfig(TemporalLayers::kReference,
>-                                          TemporalLayers::kReferenceAndUpdate,
>-                                          TemporalLayers::kReference),
>-              TemporalLayers::FrameConfig(TemporalLayers::kReferenceAndUpdate,
>-                                          TemporalLayers::kNone,
>-                                          TemporalLayers::kReference),
>-              TemporalLayers::FrameConfig(
>-                  TemporalLayers::kReference, TemporalLayers::kReference,
>-                  TemporalLayers::kReference, TemporalLayers::kFreezeEntropy)};
>+      if (!field_trial::IsDisabled("WebRTC-UseShortVP8TL2Pattern")) {
>+        // Shortened 4-frame pattern:
>+        //   1---1   1---1 ...
>+        //  /   /   /   /
>+        // 0---0---0---0 ...
>+        return {
>+            FrameConfig(kReferenceAndUpdate, kNone, kReference),
>+            FrameConfig(kReference, kUpdate, kReference),
>+            FrameConfig(kReferenceAndUpdate, kNone, kReference),
>+            FrameConfig(kReference, kReference, kReference, kFreezeEntropy)};
>+      } else {
>+        // "Default" 8-frame pattern:
>+        //   1---1---1---1   1---1---1---1 ...
>+        //  /   /   /   /   /   /   /   /
>+        // 0---0---0---0---0---0---0---0 ...
>+        return {
>+            FrameConfig(kReferenceAndUpdate, kNone, kReference),
>+            FrameConfig(kReference, kUpdate, kReference),
>+            FrameConfig(kReferenceAndUpdate, kNone, kReference),
>+            FrameConfig(kReference, kReferenceAndUpdate, kReference),
>+            FrameConfig(kReferenceAndUpdate, kNone, kReference),
>+            FrameConfig(kReference, kReferenceAndUpdate, kReference),
>+            FrameConfig(kReferenceAndUpdate, kNone, kReference),
>+            FrameConfig(kReference, kReference, kReference, kFreezeEntropy)};
>+      }
>     case 3:
>       if (field_trial::IsEnabled("WebRTC-UseShortVP8TL3Pattern")) {
>         // This field trial is intended to check if it is worth using a shorter
>@@ -181,136 +196,62 @@ std::vector<TemporalLayers::FrameConfig> GetTemporalPattern(size_t num_layers) {
>         // TL1  references 'last' and references and updates 'golden'.
>         // TL2 references both 'last' & 'golden' and references and updates
>         // 'arf'.
>-        return {TemporalLayers::FrameConfig(TemporalLayers::kReferenceAndUpdate,
>-                                            TemporalLayers::kNone,
>-                                            TemporalLayers::kNone),
>-                TemporalLayers::FrameConfig(TemporalLayers::kReference,
>-                                            TemporalLayers::kNone,
>-                                            TemporalLayers::kUpdate),
>-                TemporalLayers::FrameConfig(TemporalLayers::kReference,
>-                                            TemporalLayers::kUpdate,
>-                                            TemporalLayers::kNone),
>-                TemporalLayers::FrameConfig(TemporalLayers::kReference,
>-                                            TemporalLayers::kReference,
>-                                            TemporalLayers::kReference,
>-                                            TemporalLayers::kFreezeEntropy)};
>+        return {
>+            FrameConfig(kReferenceAndUpdate, kNone, kNone),
>+            FrameConfig(kReference, kNone, kUpdate),
>+            FrameConfig(kReference, kUpdate, kNone),
>+            FrameConfig(kReference, kReference, kReference, kFreezeEntropy)};
>       } else {
>         // All layers can reference but not update the 'alt' buffer, this means
>         // that the 'alt' buffer reference is effectively the last keyframe.
>         // TL0 also references and updates the 'last' buffer.
>         // TL1 also references 'last' and references and updates 'golden'.
>         // TL2 references both 'last' and 'golden' but updates no buffer.
>-        return {TemporalLayers::FrameConfig(TemporalLayers::kReferenceAndUpdate,
>-                                            TemporalLayers::kNone,
>-                                            TemporalLayers::kReference),
>-                TemporalLayers::FrameConfig(
>-                    TemporalLayers::kReference, TemporalLayers::kNone,
>-                    TemporalLayers::kReference, TemporalLayers::kFreezeEntropy),
>-                TemporalLayers::FrameConfig(TemporalLayers::kReference,
>-                                            TemporalLayers::kUpdate,
>-                                            TemporalLayers::kReference),
>-                TemporalLayers::FrameConfig(
>-                    TemporalLayers::kReference, TemporalLayers::kReference,
>-                    TemporalLayers::kReference, TemporalLayers::kFreezeEntropy),
>-                TemporalLayers::FrameConfig(TemporalLayers::kReferenceAndUpdate,
>-                                            TemporalLayers::kNone,
>-                                            TemporalLayers::kReference),
>-                TemporalLayers::FrameConfig(
>-                    TemporalLayers::kReference, TemporalLayers::kReference,
>-                    TemporalLayers::kReference, TemporalLayers::kFreezeEntropy),
>-                TemporalLayers::FrameConfig(TemporalLayers::kReference,
>-                                            TemporalLayers::kReferenceAndUpdate,
>-                                            TemporalLayers::kReference),
>-                TemporalLayers::FrameConfig(TemporalLayers::kReference,
>-                                            TemporalLayers::kReference,
>-                                            TemporalLayers::kReference,
>-                                            TemporalLayers::kFreezeEntropy)};
>+        return {
>+            FrameConfig(kReferenceAndUpdate, kNone, kReference),
>+            FrameConfig(kReference, kNone, kReference, kFreezeEntropy),
>+            FrameConfig(kReference, kUpdate, kReference),
>+            FrameConfig(kReference, kReference, kReference, kFreezeEntropy),
>+            FrameConfig(kReferenceAndUpdate, kNone, kReference),
>+            FrameConfig(kReference, kReference, kReference, kFreezeEntropy),
>+            FrameConfig(kReference, kReferenceAndUpdate, kReference),
>+            FrameConfig(kReference, kReference, kReference, kFreezeEntropy)};
>       }
>     case 4:
>       // TL0 references and updates only the 'last' buffer.
>       // TL1 references 'last' and updates and references 'golden'.
>       // TL2 references 'last' and 'golden', and references and updates 'arf'.
>       // TL3 references all buffers but update none of them.
>-      return {TemporalLayers::FrameConfig(TemporalLayers::kReferenceAndUpdate,
>-                                          TemporalLayers::kNone,
>-                                          TemporalLayers::kNone),
>-              TemporalLayers::FrameConfig(
>-                  TemporalLayers::kReference, TemporalLayers::kNone,
>-                  TemporalLayers::kNone, TemporalLayers::kFreezeEntropy),
>-              TemporalLayers::FrameConfig(TemporalLayers::kReference,
>-                                          TemporalLayers::kNone,
>-                                          TemporalLayers::kUpdate),
>-              TemporalLayers::FrameConfig(
>-                  TemporalLayers::kReference, TemporalLayers::kNone,
>-                  TemporalLayers::kReference, TemporalLayers::kFreezeEntropy),
>-              TemporalLayers::FrameConfig(TemporalLayers::kReference,
>-                                          TemporalLayers::kUpdate,
>-                                          TemporalLayers::kNone),
>-              TemporalLayers::FrameConfig(
>-                  TemporalLayers::kReference, TemporalLayers::kReference,
>-                  TemporalLayers::kReference, TemporalLayers::kFreezeEntropy),
>-              TemporalLayers::FrameConfig(TemporalLayers::kReference,
>-                                          TemporalLayers::kReference,
>-                                          TemporalLayers::kReferenceAndUpdate),
>-              TemporalLayers::FrameConfig(
>-                  TemporalLayers::kReference, TemporalLayers::kReference,
>-                  TemporalLayers::kReference, TemporalLayers::kFreezeEntropy),
>-              TemporalLayers::FrameConfig(TemporalLayers::kReferenceAndUpdate,
>-                                          TemporalLayers::kNone,
>-                                          TemporalLayers::kNone),
>-              TemporalLayers::FrameConfig(
>-                  TemporalLayers::kReference, TemporalLayers::kReference,
>-                  TemporalLayers::kReference, TemporalLayers::kFreezeEntropy),
>-              TemporalLayers::FrameConfig(TemporalLayers::kReference,
>-                                          TemporalLayers::kReference,
>-                                          TemporalLayers::kReferenceAndUpdate),
>-              TemporalLayers::FrameConfig(
>-                  TemporalLayers::kReference, TemporalLayers::kReference,
>-                  TemporalLayers::kReference, TemporalLayers::kFreezeEntropy),
>-              TemporalLayers::FrameConfig(TemporalLayers::kReference,
>-                                          TemporalLayers::kReferenceAndUpdate,
>-                                          TemporalLayers::kNone),
>-              TemporalLayers::FrameConfig(
>-                  TemporalLayers::kReference, TemporalLayers::kReference,
>-                  TemporalLayers::kReference, TemporalLayers::kFreezeEntropy),
>-              TemporalLayers::FrameConfig(TemporalLayers::kReference,
>-                                          TemporalLayers::kReference,
>-                                          TemporalLayers::kReferenceAndUpdate),
>-              TemporalLayers::FrameConfig(
>-                  TemporalLayers::kReference, TemporalLayers::kReference,
>-                  TemporalLayers::kReference, TemporalLayers::kFreezeEntropy)};
>+      return {FrameConfig(kReferenceAndUpdate, kNone, kNone),
>+              FrameConfig(kReference, kNone, kNone, kFreezeEntropy),
>+              FrameConfig(kReference, kNone, kUpdate),
>+              FrameConfig(kReference, kNone, kReference, kFreezeEntropy),
>+              FrameConfig(kReference, kUpdate, kNone),
>+              FrameConfig(kReference, kReference, kReference, kFreezeEntropy),
>+              FrameConfig(kReference, kReference, kReferenceAndUpdate),
>+              FrameConfig(kReference, kReference, kReference, kFreezeEntropy),
>+              FrameConfig(kReferenceAndUpdate, kNone, kNone),
>+              FrameConfig(kReference, kReference, kReference, kFreezeEntropy),
>+              FrameConfig(kReference, kReference, kReferenceAndUpdate),
>+              FrameConfig(kReference, kReference, kReference, kFreezeEntropy),
>+              FrameConfig(kReference, kReferenceAndUpdate, kNone),
>+              FrameConfig(kReference, kReference, kReference, kFreezeEntropy),
>+              FrameConfig(kReference, kReference, kReferenceAndUpdate),
>+              FrameConfig(kReference, kReference, kReference, kFreezeEntropy)};
>     default:
>       RTC_NOTREACHED();
>       break;
>   }
>   RTC_NOTREACHED();
>-  return {TemporalLayers::FrameConfig(
>-      TemporalLayers::kNone, TemporalLayers::kNone, TemporalLayers::kNone)};
>+  return {FrameConfig(kNone, kNone, kNone)};
> }
> 
>-// Temporary fix for forced SW fallback.
>-// For VP8 SW codec, |TemporalLayers| is created and reported to
>-// SimulcastRateAllocator::OnTemporalLayersCreated but not for VP8 HW.
>-// Causes an issue when going from forced SW -> HW as |TemporalLayers| is not
>-// deregistred when deleted by SW codec (tl factory might not exist, owned by
>-// SimulcastRateAllocator).
>-bool ExcludeOnTemporalLayersCreated(int num_temporal_layers) {
>-  return webrtc::field_trial::IsEnabled(
>-             "WebRTC-VP8-Forced-Fallback-Encoder-v2") &&
>-         num_temporal_layers == 1;
>-}
>-}  // namespace
>-
>-DefaultTemporalLayers::DefaultTemporalLayers(int number_of_temporal_layers,
>-                                             uint8_t initial_tl0_pic_idx)
>+DefaultTemporalLayers::DefaultTemporalLayers(int number_of_temporal_layers)
>     : num_layers_(std::max(1, number_of_temporal_layers)),
>       temporal_ids_(GetTemporalIds(num_layers_)),
>-      temporal_layer_sync_(GetTemporalLayerSync(num_layers_)),
>       temporal_pattern_(GetTemporalPattern(num_layers_)),
>-      tl0_pic_idx_(initial_tl0_pic_idx),
>-      pattern_idx_(255),
>-      last_base_layer_sync_(false) {
>-  RTC_DCHECK_EQ(temporal_pattern_.size(), temporal_layer_sync_.size());
>+      kf_buffers_(FindKfBuffers(temporal_pattern_)),
>+      pattern_idx_(kUninitializedPatternIndex) {
>   RTC_CHECK_GE(kMaxTemporalStreams, number_of_temporal_layers);
>   RTC_CHECK_GE(number_of_temporal_layers, 0);
>   RTC_CHECK_LE(number_of_temporal_layers, 4);
>@@ -318,48 +259,34 @@ DefaultTemporalLayers::DefaultTemporalLayers(int number_of_temporal_layers,
>   // temporal_ids_ are ever longer. If this is no longer correct it needs to
>   // wrap at max(temporal_ids_.size(), temporal_pattern_.size()).
>   RTC_DCHECK_LE(temporal_ids_.size(), temporal_pattern_.size());
>-}
> 
>-uint8_t DefaultTemporalLayers::Tl0PicIdx() const {
>-  return tl0_pic_idx_;
>+  // Always need to start with a keyframe, so pre-populate all frame counters.
>+  for (Vp8BufferReference buffer : kAllBuffers) {
>+    frames_since_buffer_refresh_[buffer] = 0;
>+  }
> }
> 
>-std::vector<uint32_t> DefaultTemporalLayers::OnRatesUpdated(
>-    int bitrate_kbps,
>-    int max_bitrate_kbps,
>-    int framerate) {
>-  std::vector<uint32_t> bitrates;
>-  for (size_t i = 0; i < num_layers_; ++i) {
>-    float layer_bitrate =
>-        bitrate_kbps * kVp8LayerRateAlloction[num_layers_ - 1][i];
>-    bitrates.push_back(static_cast<uint32_t>(layer_bitrate + 0.5));
>+void DefaultTemporalLayers::OnRatesUpdated(
>+    const std::vector<uint32_t>& bitrates_bps,
>+    int framerate_fps) {
>+  RTC_DCHECK_GT(bitrates_bps.size(), 0);
>+  RTC_DCHECK_LE(bitrates_bps.size(), num_layers_);
>+  // |bitrates_bps| uses individual rate per layer, but Vp8EncoderConfig wants
>+  // the accumulated rate, so sum them up.
>+  new_bitrates_bps_ = bitrates_bps;
>+  new_bitrates_bps_->resize(num_layers_);
>+  for (size_t i = 1; i < num_layers_; ++i) {
>+    (*new_bitrates_bps_)[i] += (*new_bitrates_bps_)[i - 1];
>   }
>-  new_bitrates_kbps_ = rtc::Optional<std::vector<uint32_t>>(bitrates);
>-
>-  // Allocation table is of aggregates, transform to individual rates.
>-  uint32_t sum = 0;
>-  for (size_t i = 0; i < num_layers_; ++i) {
>-    uint32_t layer_bitrate = bitrates[i];
>-    RTC_DCHECK_LE(sum, bitrates[i]);
>-    bitrates[i] -= sum;
>-    sum = layer_bitrate;
>-
>-    if (sum >= static_cast<uint32_t>(bitrate_kbps)) {
>-      // Sum adds up; any subsequent layers will be 0.
>-      bitrates.resize(i + 1);
>-      break;
>-    }
>-  }
>-
>-  return bitrates;
> }
> 
>-bool DefaultTemporalLayers::UpdateConfiguration(vpx_codec_enc_cfg_t* cfg) {
>-  if (!new_bitrates_kbps_)
>+bool DefaultTemporalLayers::UpdateConfiguration(Vp8EncoderConfig* cfg) {
>+  if (!new_bitrates_bps_) {
>     return false;
>+  }
> 
>   for (size_t i = 0; i < num_layers_; ++i) {
>-    cfg->ts_target_bitrate[i] = (*new_bitrates_kbps_)[i];
>+    cfg->ts_target_bitrate[i] = (*new_bitrates_bps_)[i] / 1000;
>     // ..., 4, 2, 1
>     cfg->ts_rate_decimator[i] = 1 << (num_layers_ - i - 1);
>   }
>@@ -369,7 +296,35 @@ bool DefaultTemporalLayers::UpdateConfiguration(vpx_codec_enc_cfg_t* cfg) {
>   memcpy(cfg->ts_layer_id, &temporal_ids_[0],
>          sizeof(unsigned int) * temporal_ids_.size());
> 
>-  new_bitrates_kbps_ = rtc::Optional<std::vector<uint32_t>>();
>+  new_bitrates_bps_.reset();
>+
>+  return true;
>+}
>+
>+bool DefaultTemporalLayers::IsSyncFrame(const FrameConfig& config) const {
>+  // Since we always assign TL0 to 'last' in these patterns, we can infer layer
>+  // sync by checking if temporal id > 0 and we only reference TL0 or buffers
>+  // containing the last key-frame.
>+  if (config.packetizer_temporal_idx == 0) {
>+    // TL0 frames are per definition not sync frames.
>+    return false;
>+  }
>+
>+  if ((config.last_buffer_flags & BufferFlags::kReference) == 0) {
>+    // Sync frames must reference TL0.
>+    return false;
>+  }
>+
>+  if ((config.golden_buffer_flags & BufferFlags::kReference) &&
>+      kf_buffers_.find(Vp8BufferReference::kGolden) == kf_buffers_.end()) {
>+    // Referencing a golden frame that contains a non-(base layer|key frame).
>+    return false;
>+  }
>+  if ((config.arf_buffer_flags & BufferFlags::kReference) &&
>+      kf_buffers_.find(Vp8BufferReference::kAltref) == kf_buffers_.end()) {
>+    // Referencing an altref frame that contains a non-(base layer|key frame).
>+    return false;
>+  }
> 
>   return true;
> }
>@@ -378,15 +333,105 @@ TemporalLayers::FrameConfig DefaultTemporalLayers::UpdateLayerConfig(
>     uint32_t timestamp) {
>   RTC_DCHECK_GT(num_layers_, 0);
>   RTC_DCHECK_LT(0, temporal_pattern_.size());
>+
>   pattern_idx_ = (pattern_idx_ + 1) % temporal_pattern_.size();
>   TemporalLayers::FrameConfig tl_config = temporal_pattern_[pattern_idx_];
>-  tl_config.layer_sync =
>-      temporal_layer_sync_[pattern_idx_ % temporal_layer_sync_.size()];
>   tl_config.encoder_layer_id = tl_config.packetizer_temporal_idx =
>       temporal_ids_[pattern_idx_ % temporal_ids_.size()];
>+
>+  if (pattern_idx_ == 0) {
>+    // Start of new pattern iteration, set up clear state by invalidating any
>+    // pending frames, so that we don't make an invalid reference to a buffer
>+    // containing data from a previous iteration.
>+    pending_frames_.clear();
>+  }
>+
>+  // Last is always ok to reference as it contains the base layer. For other
>+  // buffers though, we need to check if the buffer has actually been refreshed
>+  // this cycle of the temporal pattern. If the encoder dropped a frame, it
>+  // might not have.
>+  ValidateReferences(&tl_config.golden_buffer_flags,
>+                     Vp8BufferReference::kGolden);
>+  ValidateReferences(&tl_config.arf_buffer_flags, Vp8BufferReference::kAltref);
>+  // Update search order to let the encoder know which buffers contains the most
>+  // recent data.
>+  UpdateSearchOrder(&tl_config);
>+  // Figure out if this a sync frame (non-base-layer frame with only base-layer
>+  // references).
>+  tl_config.layer_sync = IsSyncFrame(tl_config);
>+
>+  // Increment frame age, this needs to be in sync with |pattern_idx_|, so must
>+  // update it here. Resetting age to 0 must be done when encoding is complete
>+  // though, and so in the case of pipelining encoder it might lag. To prevent
>+  // this data spill over into the next iteration, the |pedning_frames_| map
>+  // is reset in loops. If delay is constant, the relative age should still be
>+  // OK for the search order.
>+  for (Vp8BufferReference buffer : kAllBuffers) {
>+    ++frames_since_buffer_refresh_[buffer];
>+  }
>+
>+  // Add frame to set of pending frames, awaiting completion.
>+  pending_frames_[timestamp] = GetUpdatedBuffers(tl_config);
>+
>   return tl_config;
> }
> 
>+void DefaultTemporalLayers::ValidateReferences(BufferFlags* flags,
>+                                               Vp8BufferReference ref) const {
>+  // Check if the buffer specified by |ref| is actually referenced, and if so
>+  // if it also a dynamically updating one (buffers always just containing
>+  // keyframes are always safe to reference).
>+  if ((*flags & BufferFlags::kReference) &&
>+      kf_buffers_.find(ref) == kf_buffers_.end()) {
>+    auto it = frames_since_buffer_refresh_.find(ref);
>+    if (it == frames_since_buffer_refresh_.end() ||
>+        it->second >= pattern_idx_) {
>+      // No valid buffer state, or buffer contains frame that is older than the
>+      // current pattern. This reference is not valid, so remove it.
>+      *flags = static_cast<BufferFlags>(*flags & ~BufferFlags::kReference);
>+    }
>+  }
>+}
>+
>+void DefaultTemporalLayers::UpdateSearchOrder(FrameConfig* config) {
>+  // Figure out which of the buffers we can reference, and order them so that
>+  // the most recently refreshed is first. Otherwise prioritize last first,
>+  // golden second, and altref third.
>+  using BufferRefAge = std::pair<Vp8BufferReference, size_t>;
>+  std::vector<BufferRefAge> eligible_buffers;
>+  if (config->last_buffer_flags & BufferFlags::kReference) {
>+    eligible_buffers.emplace_back(
>+        Vp8BufferReference::kLast,
>+        frames_since_buffer_refresh_[Vp8BufferReference::kLast]);
>+  }
>+  if (config->golden_buffer_flags & BufferFlags::kReference) {
>+    eligible_buffers.emplace_back(
>+        Vp8BufferReference::kGolden,
>+        frames_since_buffer_refresh_[Vp8BufferReference::kGolden]);
>+  }
>+  if (config->arf_buffer_flags & BufferFlags::kReference) {
>+    eligible_buffers.emplace_back(
>+        Vp8BufferReference::kAltref,
>+        frames_since_buffer_refresh_[Vp8BufferReference::kAltref]);
>+  }
>+
>+  std::sort(eligible_buffers.begin(), eligible_buffers.end(),
>+            [](const BufferRefAge& lhs, const BufferRefAge& rhs) {
>+              if (lhs.second != rhs.second) {
>+                // Lower count has highest precedence.
>+                return lhs.second < rhs.second;
>+              }
>+              return lhs.first < rhs.first;
>+            });
>+
>+  // Populate the search order fields where possible.
>+  if (!eligible_buffers.empty()) {
>+    config->first_reference = eligible_buffers.front().first;
>+    if (eligible_buffers.size() > 1)
>+      config->second_reference = eligible_buffers[1].first;
>+  }
>+}
>+
> void DefaultTemporalLayers::PopulateCodecSpecific(
>     bool frame_is_keyframe,
>     const TemporalLayers::FrameConfig& tl_config,
>@@ -397,48 +442,53 @@ void DefaultTemporalLayers::PopulateCodecSpecific(
>   if (num_layers_ == 1) {
>     vp8_info->temporalIdx = kNoTemporalIdx;
>     vp8_info->layerSync = false;
>-    vp8_info->tl0PicIdx = kNoTl0PicIdx;
>   } else {
>-    vp8_info->temporalIdx = tl_config.packetizer_temporal_idx;
>-    vp8_info->layerSync = tl_config.layer_sync;
>     if (frame_is_keyframe) {
>+      // Restart the temporal pattern on keyframes.
>+      pattern_idx_ = 0;
>       vp8_info->temporalIdx = 0;
>-      vp8_info->layerSync = true;
>-    }
>-    if (last_base_layer_sync_ && vp8_info->temporalIdx != 0) {
>-      // Regardless of pattern the frame after a base layer sync will always
>-      // be a layer sync.
>-      vp8_info->layerSync = true;
>+      vp8_info->layerSync = true;  // Keyframes are always sync frames.
>+      // Update frame count of all kf-only buffers, regardless of state of
>+      // |pending_frames_|.
>+      for (auto it : kf_buffers_) {
>+        frames_since_buffer_refresh_[it] = 0;
>+      }
>+      auto pending_frames = pending_frames_.find(timestamp);
>+      if (pending_frames != pending_frames_.end()) {
>+        for (Vp8BufferReference buffer : kAllBuffers) {
>+          if (kf_buffers_.find(buffer) == kf_buffers_.end()) {
>+            // Key-frames update all buffers, this should be reflected if when
>+            // updating state in FrameEncoded().
>+            pending_frames->second |= static_cast<uint8_t>(buffer);
>+          }
>+        }
>+      }
>+    } else {
>+      // Delta frame, update codec specifics with temporal id and sync flag.
>+      vp8_info->temporalIdx = tl_config.packetizer_temporal_idx;
>+      vp8_info->layerSync = tl_config.layer_sync;
>     }
>-    if (vp8_info->temporalIdx == 0)
>-      tl0_pic_idx_++;
>-    last_base_layer_sync_ = frame_is_keyframe;
>-    vp8_info->tl0PicIdx = tl0_pic_idx_;
>   }
> }
> 
>-TemporalLayers* TemporalLayersFactory::Create(
>-    int simulcast_id,
>-    int temporal_layers,
>-    uint8_t initial_tl0_pic_idx) const {
>-  TemporalLayers* tl =
>-      new DefaultTemporalLayers(temporal_layers, initial_tl0_pic_idx);
>-  if (listener_ && !ExcludeOnTemporalLayersCreated(temporal_layers))
>-    listener_->OnTemporalLayersCreated(simulcast_id, tl);
>-  return tl;
>-}
>-
>-std::unique_ptr<TemporalLayersChecker> TemporalLayersFactory::CreateChecker(
>-    int /*simulcast_id*/,
>-    int temporal_layers,
>-    uint8_t initial_tl0_pic_idx) const {
>-  TemporalLayersChecker* tlc =
>-      new DefaultTemporalLayersChecker(temporal_layers, initial_tl0_pic_idx);
>-  return std::unique_ptr<TemporalLayersChecker>(tlc);
>-}
>-
>-void TemporalLayersFactory::SetListener(TemporalLayersListener* listener) {
>-  listener_ = listener;
>+void DefaultTemporalLayers::FrameEncoded(uint32_t rtp_timestamp,
>+                                         size_t size,
>+                                         int qp) {
>+  auto pending_frame = pending_frames_.find(rtp_timestamp);
>+  if (pending_frame == pending_frames_.end()) {
>+    // Might happen if pipelined encoder delayed encoding until after pattern
>+    // looped.
>+    return;
>+  }
>+  if (size == 0) {
>+    pending_frames_.erase(pending_frame);
>+    return;
>+  }
>+  for (Vp8BufferReference buffer : kAllBuffers) {
>+    if (pending_frame->second & static_cast<uint8_t>(buffer)) {
>+      frames_since_buffer_refresh_[buffer] = 0;
>+    }
>+  }
> }
> 
> // Returns list of temporal dependencies for each frame in the temporal pattern.
>@@ -468,9 +518,8 @@ std::vector<std::set<uint8_t>> GetTemporalDependencies(
> }
> 
> DefaultTemporalLayersChecker::DefaultTemporalLayersChecker(
>-    int num_temporal_layers,
>-    uint8_t initial_tl0_pic_idx)
>-    : TemporalLayersChecker(num_temporal_layers, initial_tl0_pic_idx),
>+    int num_temporal_layers)
>+    : TemporalLayersChecker(num_temporal_layers),
>       num_layers_(std::max(1, num_temporal_layers)),
>       temporal_ids_(GetTemporalIds(num_layers_)),
>       temporal_dependencies_(GetTemporalDependencies(num_layers_)),
>@@ -491,6 +540,15 @@ bool DefaultTemporalLayersChecker::CheckTemporalConfig(
>   if (frame_config.drop_frame) {
>     return true;
>   }
>+
>+  if (frame_is_keyframe) {
>+    pattern_idx_ = 0;
>+    last_ = BufferState();
>+    golden_ = BufferState();
>+    arf_ = BufferState();
>+    return true;
>+  }
>+
>   ++pattern_idx_;
>   if (pattern_idx_ == temporal_ids_.size()) {
>     // All non key-frame buffers should be updated each pattern cycle.
>@@ -533,6 +591,11 @@ bool DefaultTemporalLayersChecker::CheckTemporalConfig(
>     if (!last_.is_keyframe) {
>       dependencies.push_back(last_.pattern_idx);
>     }
>+  } else if (frame_config.first_reference == Vp8BufferReference::kLast ||
>+             frame_config.second_reference == Vp8BufferReference::kLast) {
>+    RTC_LOG(LS_ERROR)
>+        << "Last buffer not referenced, but present in search order.";
>+    return false;
>   }
> 
>   if (frame_config.arf_buffer_flags & TemporalLayers::BufferFlags::kReference) {
>@@ -543,6 +606,11 @@ bool DefaultTemporalLayersChecker::CheckTemporalConfig(
>     if (!arf_.is_keyframe) {
>       dependencies.push_back(arf_.pattern_idx);
>     }
>+  } else if (frame_config.first_reference == Vp8BufferReference::kAltref ||
>+             frame_config.second_reference == Vp8BufferReference::kAltref) {
>+    RTC_LOG(LS_ERROR)
>+        << "Altret buffer not referenced, but present in search order.";
>+    return false;
>   }
> 
>   if (frame_config.golden_buffer_flags &
>@@ -554,6 +622,11 @@ bool DefaultTemporalLayersChecker::CheckTemporalConfig(
>     if (!golden_.is_keyframe) {
>       dependencies.push_back(golden_.pattern_idx);
>     }
>+  } else if (frame_config.first_reference == Vp8BufferReference::kGolden ||
>+             frame_config.second_reference == Vp8BufferReference::kGolden) {
>+    RTC_LOG(LS_ERROR)
>+        << "Golden buffer not referenced, but present in search order.";
>+    return false;
>   }
> 
>   if (need_sync != frame_config.layer_sync) {
>@@ -592,11 +665,6 @@ bool DefaultTemporalLayersChecker::CheckTemporalConfig(
>     golden_.pattern_idx = pattern_idx_;
>     golden_.is_keyframe = false;
>   }
>-  if (frame_is_keyframe) {
>-    last_.is_keyframe = true;
>-    arf_.is_keyframe = true;
>-    golden_.is_keyframe = true;
>-  }
>   return true;
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.h
>index f1925569209..9978a9ae240 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.h
>@@ -12,58 +12,70 @@
> #ifndef MODULES_VIDEO_CODING_CODECS_VP8_DEFAULT_TEMPORAL_LAYERS_H_
> #define MODULES_VIDEO_CODING_CODECS_VP8_DEFAULT_TEMPORAL_LAYERS_H_
> 
>+#include <limits>
>+#include <map>
> #include <set>
> #include <vector>
> 
> #include "modules/video_coding/codecs/vp8/temporal_layers.h"
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> 
> namespace webrtc {
> 
> class DefaultTemporalLayers : public TemporalLayers {
>  public:
>-  DefaultTemporalLayers(int number_of_temporal_layers,
>-                        uint8_t initial_tl0_pic_idx);
>+  explicit DefaultTemporalLayers(int number_of_temporal_layers);
>   virtual ~DefaultTemporalLayers() {}
> 
>   // Returns the recommended VP8 encode flags needed. May refresh the decoder
>   // and/or update the reference buffers.
>   TemporalLayers::FrameConfig UpdateLayerConfig(uint32_t timestamp) override;
> 
>-  // Update state based on new bitrate target and incoming framerate.
>-  // Returns the bitrate allocation for the active temporal layers.
>-  std::vector<uint32_t> OnRatesUpdated(int bitrate_kbps,
>-                                       int max_bitrate_kbps,
>-                                       int framerate) override;
>+  // New target bitrate, per temporal layer.
>+  void OnRatesUpdated(const std::vector<uint32_t>& bitrates_bps,
>+                      int framerate_fps) override;
> 
>-  bool UpdateConfiguration(vpx_codec_enc_cfg_t* cfg) override;
>+  bool UpdateConfiguration(Vp8EncoderConfig* cfg) override;
> 
>   void PopulateCodecSpecific(bool frame_is_keyframe,
>                              const TemporalLayers::FrameConfig& tl_config,
>                              CodecSpecificInfoVP8* vp8_info,
>                              uint32_t timestamp) override;
> 
>-  void FrameEncoded(unsigned int size, int qp) override {}
>-
>-  uint8_t Tl0PicIdx() const override;
>+  void FrameEncoded(uint32_t rtp_timestamp, size_t size, int qp) override;
> 
>  private:
>+  static constexpr size_t kKeyframeBuffer = std::numeric_limits<size_t>::max();
>+  static std::vector<TemporalLayers::FrameConfig> GetTemporalPattern(
>+      size_t num_layers);
>+  bool IsSyncFrame(const FrameConfig& config) const;
>+  void ValidateReferences(BufferFlags* flags, Vp8BufferReference ref) const;
>+  void UpdateSearchOrder(FrameConfig* config);
>+
>   const size_t num_layers_;
>   const std::vector<unsigned int> temporal_ids_;
>-  const std::vector<bool> temporal_layer_sync_;
>   const std::vector<TemporalLayers::FrameConfig> temporal_pattern_;
>+  // Set of buffers that are never updated except by keyframes.
>+  const std::set<Vp8BufferReference> kf_buffers_;
> 
>-  uint8_t tl0_pic_idx_;
>   uint8_t pattern_idx_;
>-  bool last_base_layer_sync_;
>-  rtc::Optional<std::vector<uint32_t>> new_bitrates_kbps_;
>+  // Updated cumulative bitrates, per temporal layer.
>+  absl::optional<std::vector<uint32_t>> new_bitrates_bps_;
>+
>+  // Map from rtp timestamp to a bitmask of Vp8BufferReference indicating which
>+  // buffers this frame should update. Reset on pattern loop.
>+  std::map<uint32_t, uint8_t> pending_frames_;
>+
>+  // One counter per Vp8BufferReference, indicating number of frames since last
>+  // refresh. For non-base-layer frames (ie golden, altref buffers), this is
>+  // reset when the pattern loops.
>+  std::map<Vp8BufferReference, size_t> frames_since_buffer_refresh_;
> };
> 
> class DefaultTemporalLayersChecker : public TemporalLayersChecker {
>  public:
>-  DefaultTemporalLayersChecker(int number_of_temporal_layers,
>-                               uint8_t initial_tl0_pic_idx);
>+  explicit DefaultTemporalLayersChecker(int number_of_temporal_layers);
>   bool CheckTemporalConfig(
>       bool frame_is_keyframe,
>       const TemporalLayers::FrameConfig& frame_config) override;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers_unittest.cc
>index 1c1d63f1f8d..6021c65bcee 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers_unittest.cc
>@@ -9,16 +9,15 @@
>  */
> 
> #include "modules/video_coding/codecs/vp8/default_temporal_layers.h"
>-#include "modules/video_coding/codecs/vp8/vp8_impl.h"
>+#include "modules/video_coding/codecs/vp8/libvpx_vp8_encoder.h"
> #include "modules/video_coding/include/video_codec_interface.h"
>+#include "modules/video_coding/utility/simulcast_rate_allocator.h"
> #include "test/field_trial.h"
> #include "test/gtest.h"
>-#include "vpx/vp8cx.h"
>-#include "vpx/vpx_encoder.h"
> 
> namespace webrtc {
> namespace test {
>-
>+namespace {
> enum {
>   kTemporalUpdateLast = VP8_EFLAG_NO_UPD_GF | VP8_EFLAG_NO_UPD_ARF |
>                         VP8_EFLAG_NO_REF_GF | VP8_EFLAG_NO_REF_ARF,
>@@ -51,44 +50,74 @@ enum {
>       VP8_EFLAG_NO_UPD_ARF | VP8_EFLAG_NO_REF_GF,
> };
> 
>+std::vector<uint32_t> GetTemporalLayerRates(int target_bitrate_kbps,
>+                                            int framerate_fps,
>+                                            int num_temporal_layers) {
>+  VideoCodec codec;
>+  codec.codecType = VideoCodecType::kVideoCodecVP8;
>+  codec.numberOfSimulcastStreams = 1;
>+  codec.maxBitrate = target_bitrate_kbps;
>+  codec.maxFramerate = framerate_fps;
>+  codec.simulcastStream[0].targetBitrate = target_bitrate_kbps;
>+  codec.simulcastStream[0].maxBitrate = target_bitrate_kbps;
>+  codec.simulcastStream[0].numberOfTemporalLayers = num_temporal_layers;
>+  codec.simulcastStream[0].active = true;
>+  SimulcastRateAllocator allocator(codec);
>+  return allocator.GetAllocation(target_bitrate_kbps, framerate_fps)
>+      .GetTemporalLayerAllocation(0);
>+}
>+
>+constexpr int kDefaultBitrateBps = 500;
>+constexpr int kDefaultFramerate = 30;
>+constexpr int kDefaultBytesPerFrame =
>+    (kDefaultBitrateBps / 8) / kDefaultFramerate;
>+constexpr int kDefaultQp = 2;
>+}  // namespace
>+
>+using BufferFlags = TemporalLayers::BufferFlags;
>+
> TEST(TemporalLayersTest, 2Layers) {
>-  DefaultTemporalLayers tl(2, 0);
>-  DefaultTemporalLayersChecker checker(2, 0);
>-  vpx_codec_enc_cfg_t cfg;
>+  constexpr int kNumLayers = 2;
>+  DefaultTemporalLayers tl(kNumLayers);
>+  DefaultTemporalLayersChecker checker(kNumLayers);
>+  Vp8EncoderConfig cfg;
>   CodecSpecificInfoVP8 vp8_info;
>-  tl.OnRatesUpdated(500, 500, 30);
>+  tl.OnRatesUpdated(GetTemporalLayerRates(kDefaultBytesPerFrame,
>+                                          kDefaultFramerate, kNumLayers),
>+                    kDefaultFramerate);
>   tl.UpdateConfiguration(&cfg);
> 
>   int expected_flags[16] = {
>       kTemporalUpdateLastRefAltRef,
>       kTemporalUpdateGoldenWithoutDependencyRefAltRef,
>       kTemporalUpdateLastRefAltRef,
>-      kTemporalUpdateGoldenRefAltRef,
>+      kTemporalUpdateNone,
>       kTemporalUpdateLastRefAltRef,
>-      kTemporalUpdateGoldenRefAltRef,
>+      kTemporalUpdateGoldenWithoutDependencyRefAltRef,
>       kTemporalUpdateLastRefAltRef,
>       kTemporalUpdateNone,
>       kTemporalUpdateLastRefAltRef,
>       kTemporalUpdateGoldenWithoutDependencyRefAltRef,
>       kTemporalUpdateLastRefAltRef,
>-      kTemporalUpdateGoldenRefAltRef,
>+      kTemporalUpdateNone,
>       kTemporalUpdateLastRefAltRef,
>-      kTemporalUpdateGoldenRefAltRef,
>+      kTemporalUpdateGoldenWithoutDependencyRefAltRef,
>       kTemporalUpdateLastRefAltRef,
>       kTemporalUpdateNone,
>   };
>   int expected_temporal_idx[16] = {0, 1, 0, 1, 0, 1, 0, 1,
>                                    0, 1, 0, 1, 0, 1, 0, 1};
> 
>-  bool expected_layer_sync[16] = {false, true,  false, false, false, false,
>+  bool expected_layer_sync[16] = {false, true,  false, false, false, true,
>                                   false, false, false, true,  false, false,
>-                                  false, false, false, false};
>+                                  false, true,  false, false};
> 
>   uint32_t timestamp = 0;
>   for (int i = 0; i < 16; ++i) {
>     TemporalLayers::FrameConfig tl_config = tl.UpdateLayerConfig(timestamp);
>-    EXPECT_EQ(expected_flags[i], VP8EncoderImpl::EncodeFlags(tl_config)) << i;
>-    tl.PopulateCodecSpecific(i == 0, tl_config, &vp8_info, 0);
>+    EXPECT_EQ(expected_flags[i], LibvpxVp8Encoder::EncodeFlags(tl_config)) << i;
>+    tl.PopulateCodecSpecific(i == 0, tl_config, &vp8_info, timestamp);
>+    tl.FrameEncoded(timestamp, kDefaultBytesPerFrame, kDefaultQp);
>     EXPECT_TRUE(checker.CheckTemporalConfig(i == 0, tl_config));
>     EXPECT_EQ(expected_temporal_idx[i], vp8_info.temporalIdx);
>     EXPECT_EQ(expected_temporal_idx[i], tl_config.packetizer_temporal_idx);
>@@ -100,11 +129,14 @@ TEST(TemporalLayersTest, 2Layers) {
> }
> 
> TEST(TemporalLayersTest, 3Layers) {
>-  DefaultTemporalLayers tl(3, 0);
>-  DefaultTemporalLayersChecker checker(3, 0);
>-  vpx_codec_enc_cfg_t cfg;
>+  constexpr int kNumLayers = 3;
>+  DefaultTemporalLayers tl(kNumLayers);
>+  DefaultTemporalLayersChecker checker(kNumLayers);
>+  Vp8EncoderConfig cfg;
>   CodecSpecificInfoVP8 vp8_info;
>-  tl.OnRatesUpdated(500, 500, 30);
>+  tl.OnRatesUpdated(GetTemporalLayerRates(kDefaultBytesPerFrame,
>+                                          kDefaultFramerate, kNumLayers),
>+                    kDefaultFramerate);
>   tl.UpdateConfiguration(&cfg);
> 
>   int expected_flags[16] = {
>@@ -135,8 +167,9 @@ TEST(TemporalLayersTest, 3Layers) {
>   unsigned int timestamp = 0;
>   for (int i = 0; i < 16; ++i) {
>     TemporalLayers::FrameConfig tl_config = tl.UpdateLayerConfig(timestamp);
>-    EXPECT_EQ(expected_flags[i], VP8EncoderImpl::EncodeFlags(tl_config)) << i;
>-    tl.PopulateCodecSpecific(i == 0, tl_config, &vp8_info, 0);
>+    EXPECT_EQ(expected_flags[i], LibvpxVp8Encoder::EncodeFlags(tl_config)) << i;
>+    tl.PopulateCodecSpecific(i == 0, tl_config, &vp8_info, timestamp);
>+    tl.FrameEncoded(timestamp, kDefaultBytesPerFrame, kDefaultQp);
>     EXPECT_TRUE(checker.CheckTemporalConfig(i == 0, tl_config));
>     EXPECT_EQ(expected_temporal_idx[i], vp8_info.temporalIdx);
>     EXPECT_EQ(expected_temporal_idx[i], tl_config.packetizer_temporal_idx);
>@@ -148,12 +181,15 @@ TEST(TemporalLayersTest, 3Layers) {
> }
> 
> TEST(TemporalLayersTest, Alternative3Layers) {
>+  constexpr int kNumLayers = 3;
>   ScopedFieldTrials field_trial("WebRTC-UseShortVP8TL3Pattern/Enabled/");
>-  DefaultTemporalLayers tl(3, 0);
>-  DefaultTemporalLayersChecker checker(3, 0);
>-  vpx_codec_enc_cfg_t cfg;
>+  DefaultTemporalLayers tl(kNumLayers);
>+  DefaultTemporalLayersChecker checker(kNumLayers);
>+  Vp8EncoderConfig cfg;
>   CodecSpecificInfoVP8 vp8_info;
>-  tl.OnRatesUpdated(500, 500, 30);
>+  tl.OnRatesUpdated(GetTemporalLayerRates(kDefaultBytesPerFrame,
>+                                          kDefaultFramerate, kNumLayers),
>+                    kDefaultFramerate);
>   tl.UpdateConfiguration(&cfg);
> 
>   int expected_flags[8] = {kTemporalUpdateLast,
>@@ -172,8 +208,9 @@ TEST(TemporalLayersTest, Alternative3Layers) {
>   unsigned int timestamp = 0;
>   for (int i = 0; i < 8; ++i) {
>     TemporalLayers::FrameConfig tl_config = tl.UpdateLayerConfig(timestamp);
>-    EXPECT_EQ(expected_flags[i], VP8EncoderImpl::EncodeFlags(tl_config)) << i;
>-    tl.PopulateCodecSpecific(i == 0, tl_config, &vp8_info, 0);
>+    EXPECT_EQ(expected_flags[i], LibvpxVp8Encoder::EncodeFlags(tl_config)) << i;
>+    tl.PopulateCodecSpecific(i == 0, tl_config, &vp8_info, timestamp);
>+    tl.FrameEncoded(timestamp, kDefaultBytesPerFrame, kDefaultQp);
>     EXPECT_TRUE(checker.CheckTemporalConfig(i == 0, tl_config));
>     EXPECT_EQ(expected_temporal_idx[i], vp8_info.temporalIdx);
>     EXPECT_EQ(expected_temporal_idx[i], tl_config.packetizer_temporal_idx);
>@@ -184,12 +221,100 @@ TEST(TemporalLayersTest, Alternative3Layers) {
>   }
> }
> 
>+TEST(TemporalLayersTest, SearchOrder) {
>+  constexpr int kNumLayers = 3;
>+  ScopedFieldTrials field_trial("WebRTC-UseShortVP8TL3Pattern/Enabled/");
>+  DefaultTemporalLayers tl(kNumLayers);
>+  DefaultTemporalLayersChecker checker(kNumLayers);
>+  Vp8EncoderConfig cfg;
>+  CodecSpecificInfoVP8 vp8_info;
>+  tl.OnRatesUpdated(GetTemporalLayerRates(kDefaultBytesPerFrame,
>+                                          kDefaultFramerate, kNumLayers),
>+                    kDefaultFramerate);
>+  tl.UpdateConfiguration(&cfg);
>+
>+  // Use a repeating pattern of tl 0, 2, 1, 2.
>+  // Tl 0, 1, 2 update last, golden, altref respectively.
>+
>+  // Start with a key-frame. tl_config flags can be ignored.
>+  uint32_t timestamp = 0;
>+  TemporalLayers::FrameConfig tl_config = tl.UpdateLayerConfig(timestamp);
>+  tl.PopulateCodecSpecific(true, tl_config, &vp8_info, timestamp);
>+  tl.FrameEncoded(timestamp, kDefaultBytesPerFrame, kDefaultQp);
>+
>+  // TL2 frame. First one only references TL0. Updates altref.
>+  tl_config = tl.UpdateLayerConfig(++timestamp);
>+  tl.PopulateCodecSpecific(false, tl_config, &vp8_info, timestamp);
>+  tl.FrameEncoded(timestamp, kDefaultBytesPerFrame, kDefaultQp);
>+  EXPECT_EQ(tl_config.first_reference, Vp8BufferReference::kLast);
>+  EXPECT_EQ(tl_config.second_reference, Vp8BufferReference::kNone);
>+
>+  // TL1 frame. Can only reference TL0. Updated golden.
>+  tl_config = tl.UpdateLayerConfig(++timestamp);
>+  tl.PopulateCodecSpecific(false, tl_config, &vp8_info, timestamp);
>+  tl.FrameEncoded(timestamp, kDefaultBytesPerFrame, kDefaultQp);
>+  EXPECT_EQ(tl_config.first_reference, Vp8BufferReference::kLast);
>+  EXPECT_EQ(tl_config.second_reference, Vp8BufferReference::kNone);
>+
>+  // TL2 frame. Can reference all three buffers. Golden was the last to be
>+  // updated, the next to last was altref.
>+  tl_config = tl.UpdateLayerConfig(++timestamp);
>+  tl.PopulateCodecSpecific(false, tl_config, &vp8_info, timestamp);
>+  tl.FrameEncoded(timestamp, kDefaultBytesPerFrame, kDefaultQp);
>+  EXPECT_EQ(tl_config.first_reference, Vp8BufferReference::kGolden);
>+  EXPECT_EQ(tl_config.second_reference, Vp8BufferReference::kAltref);
>+}
>+
>+TEST(TemporalLayersTest, SearchOrderWithDrop) {
>+  constexpr int kNumLayers = 3;
>+  ScopedFieldTrials field_trial("WebRTC-UseShortVP8TL3Pattern/Enabled/");
>+  DefaultTemporalLayers tl(kNumLayers);
>+  DefaultTemporalLayersChecker checker(kNumLayers);
>+  Vp8EncoderConfig cfg;
>+  CodecSpecificInfoVP8 vp8_info;
>+  tl.OnRatesUpdated(GetTemporalLayerRates(kDefaultBytesPerFrame,
>+                                          kDefaultFramerate, kNumLayers),
>+                    kDefaultFramerate);
>+  tl.UpdateConfiguration(&cfg);
>+
>+  // Use a repeating pattern of tl 0, 2, 1, 2.
>+  // Tl 0, 1, 2 update last, golden, altref respectively.
>+
>+  // Start with a key-frame. tl_config flags can be ignored.
>+  uint32_t timestamp = 0;
>+  TemporalLayers::FrameConfig tl_config = tl.UpdateLayerConfig(timestamp);
>+  tl.PopulateCodecSpecific(true, tl_config, &vp8_info, timestamp);
>+  tl.FrameEncoded(timestamp, kDefaultBytesPerFrame, kDefaultQp);
>+
>+  // TL2 frame. First one only references TL0. Updates altref.
>+  tl_config = tl.UpdateLayerConfig(++timestamp);
>+  tl.PopulateCodecSpecific(false, tl_config, &vp8_info, timestamp);
>+  tl.FrameEncoded(timestamp, kDefaultBytesPerFrame, kDefaultQp);
>+  EXPECT_EQ(tl_config.first_reference, Vp8BufferReference::kLast);
>+  EXPECT_EQ(tl_config.second_reference, Vp8BufferReference::kNone);
>+
>+  // Dropped TL1 frame. Can only reference TL0. Should have updated golden.
>+  tl_config = tl.UpdateLayerConfig(++timestamp);
>+  tl.FrameEncoded(timestamp, 0, 0);
>+
>+  // TL2 frame. Can normally reference all three buffers, but golden has not
>+  // been populated this cycle. Altref was last to be updated, before that last.
>+  tl_config = tl.UpdateLayerConfig(++timestamp);
>+  tl.PopulateCodecSpecific(false, tl_config, &vp8_info, timestamp);
>+  tl.FrameEncoded(timestamp, kDefaultBytesPerFrame, kDefaultQp);
>+  EXPECT_EQ(tl_config.first_reference, Vp8BufferReference::kAltref);
>+  EXPECT_EQ(tl_config.second_reference, Vp8BufferReference::kLast);
>+}
>+
> TEST(TemporalLayersTest, 4Layers) {
>-  DefaultTemporalLayers tl(4, 0);
>-  DefaultTemporalLayersChecker checker(4, 0);
>-  vpx_codec_enc_cfg_t cfg;
>+  constexpr int kNumLayers = 4;
>+  DefaultTemporalLayers tl(kNumLayers);
>+  DefaultTemporalLayersChecker checker(kNumLayers);
>+  Vp8EncoderConfig cfg;
>   CodecSpecificInfoVP8 vp8_info;
>-  tl.OnRatesUpdated(500, 500, 30);
>+  tl.OnRatesUpdated(GetTemporalLayerRates(kDefaultBytesPerFrame,
>+                                          kDefaultFramerate, kNumLayers),
>+                    kDefaultFramerate);
>   tl.UpdateConfiguration(&cfg);
>   int expected_flags[16] = {
>       kTemporalUpdateLast,
>@@ -219,8 +344,9 @@ TEST(TemporalLayersTest, 4Layers) {
>   uint32_t timestamp = 0;
>   for (int i = 0; i < 16; ++i) {
>     TemporalLayers::FrameConfig tl_config = tl.UpdateLayerConfig(timestamp);
>-    EXPECT_EQ(expected_flags[i], VP8EncoderImpl::EncodeFlags(tl_config)) << i;
>-    tl.PopulateCodecSpecific(i == 0, tl_config, &vp8_info, 0);
>+    EXPECT_EQ(expected_flags[i], LibvpxVp8Encoder::EncodeFlags(tl_config)) << i;
>+    tl.PopulateCodecSpecific(i == 0, tl_config, &vp8_info, timestamp);
>+    tl.FrameEncoded(timestamp, kDefaultBytesPerFrame, kDefaultQp);
>     EXPECT_TRUE(checker.CheckTemporalConfig(i == 0, tl_config));
>     EXPECT_EQ(expected_temporal_idx[i], vp8_info.temporalIdx);
>     EXPECT_EQ(expected_temporal_idx[i], tl_config.packetizer_temporal_idx);
>@@ -231,12 +357,236 @@ TEST(TemporalLayersTest, 4Layers) {
>   }
> }
> 
>+TEST(TemporalLayersTest, DoesNotReferenceDroppedFrames) {
>+  constexpr int kNumLayers = 3;
>+  // Use a repeating pattern of tl 0, 2, 1, 2.
>+  // Tl 0, 1, 2 update last, golden, altref respectively.
>+  ScopedFieldTrials field_trial("WebRTC-UseShortVP8TL3Pattern/Enabled/");
>+  DefaultTemporalLayers tl(kNumLayers);
>+  DefaultTemporalLayersChecker checker(kNumLayers);
>+  Vp8EncoderConfig cfg;
>+  CodecSpecificInfoVP8 vp8_info;
>+  tl.OnRatesUpdated(GetTemporalLayerRates(kDefaultBytesPerFrame,
>+                                          kDefaultFramerate, kNumLayers),
>+                    kDefaultFramerate);
>+  tl.UpdateConfiguration(&cfg);
>+
>+  // Start with a keyframe.
>+  uint32_t timestamp = 0;
>+  TemporalLayers::FrameConfig tl_config = tl.UpdateLayerConfig(timestamp);
>+  tl.PopulateCodecSpecific(true, tl_config, &vp8_info, timestamp);
>+  tl.FrameEncoded(timestamp, kDefaultBytesPerFrame, kDefaultQp);
>+
>+  // Dropped TL2 frame.
>+  tl_config = tl.UpdateLayerConfig(++timestamp);
>+  tl.FrameEncoded(timestamp, 0, 0);
>+
>+  // Dropped TL1 frame.
>+  tl_config = tl.UpdateLayerConfig(++timestamp);
>+  tl.FrameEncoded(timestamp, 0, 0);
>+
>+  // TL2 frame. Can reference all three buffers, valid since golden and altref
>+  // both contain the last keyframe.
>+  tl_config = tl.UpdateLayerConfig(++timestamp);
>+  tl.PopulateCodecSpecific(false, tl_config, &vp8_info, timestamp);
>+  tl.FrameEncoded(timestamp, kDefaultBytesPerFrame, kDefaultQp);
>+  EXPECT_TRUE(tl_config.last_buffer_flags & BufferFlags::kReference);
>+  EXPECT_TRUE(tl_config.golden_buffer_flags & BufferFlags::kReference);
>+  EXPECT_TRUE(tl_config.arf_buffer_flags & BufferFlags::kReference);
>+
>+  // Restart of cycle!
>+
>+  // TL0 base layer frame, updating and referencing last.
>+  tl_config = tl.UpdateLayerConfig(++timestamp);
>+  tl.PopulateCodecSpecific(false, tl_config, &vp8_info, timestamp);
>+  tl.FrameEncoded(timestamp, kDefaultBytesPerFrame, kDefaultQp);
>+
>+  // TL2 frame, updating altref.
>+  tl_config = tl.UpdateLayerConfig(++timestamp);
>+  tl.PopulateCodecSpecific(false, tl_config, &vp8_info, timestamp);
>+  tl.FrameEncoded(timestamp, kDefaultBytesPerFrame, kDefaultQp);
>+
>+  // TL1 frame, updating golden.
>+  tl_config = tl.UpdateLayerConfig(++timestamp);
>+  tl.PopulateCodecSpecific(false, tl_config, &vp8_info, timestamp);
>+  tl.FrameEncoded(timestamp, kDefaultBytesPerFrame, kDefaultQp);
>+
>+  // TL2 frame. Can still reference all buffer since they have been update this
>+  // cycle.
>+  tl_config = tl.UpdateLayerConfig(++timestamp);
>+  tl.PopulateCodecSpecific(false, tl_config, &vp8_info, timestamp);
>+  tl.FrameEncoded(timestamp, kDefaultBytesPerFrame, kDefaultQp);
>+  EXPECT_TRUE(tl_config.last_buffer_flags & BufferFlags::kReference);
>+  EXPECT_TRUE(tl_config.golden_buffer_flags & BufferFlags::kReference);
>+  EXPECT_TRUE(tl_config.arf_buffer_flags & BufferFlags::kReference);
>+
>+  // Restart of cycle!
>+
>+  // TL0 base layer frame, updating and referencing last.
>+  tl_config = tl.UpdateLayerConfig(++timestamp);
>+  tl.PopulateCodecSpecific(false, tl_config, &vp8_info, timestamp);
>+  tl.FrameEncoded(timestamp, kDefaultBytesPerFrame, kDefaultQp);
>+
>+  // Dropped TL2 frame.
>+  tl_config = tl.UpdateLayerConfig(++timestamp);
>+  tl.FrameEncoded(timestamp, 0, 0);
>+
>+  // Dropped TL1 frame.
>+  tl_config = tl.UpdateLayerConfig(++timestamp);
>+  tl.FrameEncoded(timestamp, 0, 0);
>+
>+  // TL2 frame. This time golden and altref contain data from the previous cycle
>+  // and cannot be referenced.
>+  tl_config = tl.UpdateLayerConfig(++timestamp);
>+  tl.PopulateCodecSpecific(false, tl_config, &vp8_info, timestamp);
>+  tl.FrameEncoded(timestamp, kDefaultBytesPerFrame, kDefaultQp);
>+  EXPECT_TRUE(tl_config.last_buffer_flags & BufferFlags::kReference);
>+  EXPECT_FALSE(tl_config.golden_buffer_flags & BufferFlags::kReference);
>+  EXPECT_FALSE(tl_config.arf_buffer_flags & BufferFlags::kReference);
>+}
>+
>+TEST(TemporalLayersTest, DoesNotReferenceUnlessGuaranteedToExist) {
>+  constexpr int kNumLayers = 3;
>+  // Use a repeating pattern of tl 0, 2, 1, 2.
>+  // Tl 0, 1 updates last, golden respectively. Altref is always last keyframe.
>+  DefaultTemporalLayers tl(kNumLayers);
>+  DefaultTemporalLayersChecker checker(kNumLayers);
>+  Vp8EncoderConfig cfg;
>+  CodecSpecificInfoVP8 vp8_info;
>+  tl.OnRatesUpdated(GetTemporalLayerRates(kDefaultBytesPerFrame,
>+                                          kDefaultFramerate, kNumLayers),
>+                    kDefaultFramerate);
>+  tl.UpdateConfiguration(&cfg);
>+
>+  // Start with a keyframe.
>+  uint32_t timestamp = 0;
>+  TemporalLayers::FrameConfig tl_config = tl.UpdateLayerConfig(timestamp);
>+  tl.PopulateCodecSpecific(true, tl_config, &vp8_info, timestamp);
>+  tl.FrameEncoded(timestamp, kDefaultBytesPerFrame, kDefaultQp);
>+
>+  // Do a full cycle of the pattern.
>+  for (int i = 0; i < 7; ++i) {
>+    tl_config = tl.UpdateLayerConfig(++timestamp);
>+    tl.PopulateCodecSpecific(false, tl_config, &vp8_info, timestamp);
>+    tl.FrameEncoded(timestamp, kDefaultBytesPerFrame, kDefaultQp);
>+  }
>+
>+  // TL0 base layer frame, starting the cycle over.
>+  tl_config = tl.UpdateLayerConfig(++timestamp);
>+  tl.PopulateCodecSpecific(false, tl_config, &vp8_info, timestamp);
>+  tl.FrameEncoded(timestamp, kDefaultBytesPerFrame, kDefaultQp);
>+
>+  // TL2 frame.
>+  tl_config = tl.UpdateLayerConfig(++timestamp);
>+  tl.PopulateCodecSpecific(false, tl_config, &vp8_info, timestamp);
>+  tl.FrameEncoded(timestamp, kDefaultBytesPerFrame, kDefaultQp);
>+
>+  // Encoder has a hiccup and builds a queue, so frame encoding is delayed.
>+  // TL1 frame, updating golden.
>+  tl_config = tl.UpdateLayerConfig(++timestamp);
>+
>+  // TL2 frame, that should be referencing golden, but we can't be certain it's
>+  // not going to be dropped, so that is not allowed.
>+  tl_config = tl.UpdateLayerConfig(timestamp + 1);
>+  EXPECT_TRUE(tl_config.last_buffer_flags & BufferFlags::kReference);
>+  EXPECT_FALSE(tl_config.golden_buffer_flags & BufferFlags::kReference);
>+  EXPECT_TRUE(tl_config.arf_buffer_flags & BufferFlags::kReference);
>+
>+  // TL0 base layer frame.
>+  tl_config = tl.UpdateLayerConfig(timestamp + 2);
>+
>+  // The previous four enqueued frames finally get encoded, and the updated
>+  // buffers are now OK to reference.
>+  // Enqueued TL1 frame ready.
>+  tl.PopulateCodecSpecific(false, tl_config, &vp8_info, timestamp);
>+  tl.FrameEncoded(timestamp, kDefaultBytesPerFrame, kDefaultQp);
>+  // Enqueued TL2 frame.
>+  tl.PopulateCodecSpecific(false, tl_config, &vp8_info, ++timestamp);
>+  tl.FrameEncoded(timestamp, kDefaultBytesPerFrame, kDefaultQp);
>+  // Enqueued TL0 frame.
>+  tl.PopulateCodecSpecific(false, tl_config, &vp8_info, ++timestamp);
>+  tl.FrameEncoded(timestamp, kDefaultBytesPerFrame, kDefaultQp);
>+
>+  // TL2 frame, all buffers are now in a known good state, OK to reference.
>+  tl_config = tl.UpdateLayerConfig(++timestamp + 1);
>+  EXPECT_TRUE(tl_config.last_buffer_flags & BufferFlags::kReference);
>+  EXPECT_TRUE(tl_config.golden_buffer_flags & BufferFlags::kReference);
>+  EXPECT_TRUE(tl_config.arf_buffer_flags & BufferFlags::kReference);
>+}
>+
>+TEST(TemporalLayersTest, DoesNotReferenceUnlessGuaranteedToExistLongDelay) {
>+  constexpr int kNumLayers = 3;
>+  // Use a repeating pattern of tl 0, 2, 1, 2.
>+  // Tl 0, 1 updates last, golden, altref respectively.
>+  ScopedFieldTrials field_trial("WebRTC-UseShortVP8TL3Pattern/Enabled/");
>+  DefaultTemporalLayers tl(kNumLayers);
>+  DefaultTemporalLayersChecker checker(kNumLayers);
>+  Vp8EncoderConfig cfg;
>+  CodecSpecificInfoVP8 vp8_info;
>+  tl.OnRatesUpdated(GetTemporalLayerRates(kDefaultBytesPerFrame,
>+                                          kDefaultFramerate, kNumLayers),
>+                    kDefaultFramerate);
>+  tl.UpdateConfiguration(&cfg);
>+
>+  // Start with a keyframe.
>+  uint32_t timestamp = 0;
>+  TemporalLayers::FrameConfig tl_config = tl.UpdateLayerConfig(timestamp);
>+  tl.PopulateCodecSpecific(true, tl_config, &vp8_info, timestamp);
>+  tl.FrameEncoded(timestamp, kDefaultBytesPerFrame, kDefaultQp);
>+
>+  // Do a full cycle of the pattern.
>+  for (int i = 0; i < 3; ++i) {
>+    tl_config = tl.UpdateLayerConfig(++timestamp);
>+    tl.PopulateCodecSpecific(false, tl_config, &vp8_info, timestamp);
>+    tl.FrameEncoded(timestamp, kDefaultBytesPerFrame, kDefaultQp);
>+  }
>+
>+  // TL0 base layer frame, starting the cycle over.
>+  tl_config = tl.UpdateLayerConfig(++timestamp);
>+  tl.PopulateCodecSpecific(false, tl_config, &vp8_info, timestamp);
>+  tl.FrameEncoded(timestamp, kDefaultBytesPerFrame, kDefaultQp);
>+
>+  // TL2 frame.
>+  tl_config = tl.UpdateLayerConfig(++timestamp);
>+  tl.PopulateCodecSpecific(false, tl_config, &vp8_info, timestamp);
>+  tl.FrameEncoded(timestamp, kDefaultBytesPerFrame, kDefaultQp);
>+
>+  // Encoder has a hiccup and builds a queue, so frame encoding is delayed.
>+  // Encoded, but delayed frames in TL 1, 2.
>+  tl_config = tl.UpdateLayerConfig(timestamp + 1);
>+  tl_config = tl.UpdateLayerConfig(timestamp + 2);
>+
>+  // Restart of the pattern!
>+
>+  // Encoded, but delayed frames in TL 2, 1.
>+  tl_config = tl.UpdateLayerConfig(timestamp + 3);
>+  tl_config = tl.UpdateLayerConfig(timestamp + 4);
>+
>+  // TL1 frame from last cycle is ready.
>+  tl.PopulateCodecSpecific(false, tl_config, &vp8_info, timestamp + 1);
>+  tl.FrameEncoded(timestamp, kDefaultBytesPerFrame, kDefaultQp);
>+  // TL2 frame from last cycle is ready.
>+  tl.PopulateCodecSpecific(false, tl_config, &vp8_info, timestamp + 2);
>+  tl.FrameEncoded(timestamp, kDefaultBytesPerFrame, kDefaultQp);
>+
>+  // TL2 frame, that should be referencing all buffers, but altref and golden
>+  // haven not been updated this cycle. (Don't be fooled by the late frames from
>+  // the last cycle!)
>+  tl_config = tl.UpdateLayerConfig(timestamp + 5);
>+  EXPECT_TRUE(tl_config.last_buffer_flags & BufferFlags::kReference);
>+  EXPECT_FALSE(tl_config.golden_buffer_flags & BufferFlags::kReference);
>+  EXPECT_FALSE(tl_config.arf_buffer_flags & BufferFlags::kReference);
>+}
>+
> TEST(TemporalLayersTest, KeyFrame) {
>-  DefaultTemporalLayers tl(3, 0);
>-  DefaultTemporalLayersChecker checker(3, 0);
>-  vpx_codec_enc_cfg_t cfg;
>+  constexpr int kNumLayers = 3;
>+  DefaultTemporalLayers tl(kNumLayers);
>+  DefaultTemporalLayersChecker checker(kNumLayers);
>+  Vp8EncoderConfig cfg;
>   CodecSpecificInfoVP8 vp8_info;
>-  tl.OnRatesUpdated(500, 500, 30);
>+  tl.OnRatesUpdated(GetTemporalLayerRates(kDefaultBytesPerFrame,
>+                                          kDefaultFramerate, kNumLayers),
>+                    kDefaultFramerate);
>   tl.UpdateConfiguration(&cfg);
> 
>   int expected_flags[8] = {
>@@ -250,37 +600,36 @@ TEST(TemporalLayersTest, KeyFrame) {
>       kTemporalUpdateNone,
>   };
>   int expected_temporal_idx[8] = {0, 2, 1, 2, 0, 2, 1, 2};
>-  bool expected_layer_sync[8] = {false, true,  true,  false,
>+  bool expected_layer_sync[8] = {true,  true,  true,  false,
>                                  false, false, false, false};
> 
>   uint32_t timestamp = 0;
>   for (int i = 0; i < 7; ++i) {
>+    // Temporal pattern starts from 0 after key frame. Let the first |i| - 1
>+    // frames be delta frames, and the |i|th one key frame.
>+    for (int j = 1; j <= i; ++j) {
>+      // Since last frame was always a keyframe and thus index 0 in the pattern,
>+      // this loop starts at index 1.
>+      TemporalLayers::FrameConfig tl_config = tl.UpdateLayerConfig(timestamp);
>+      EXPECT_EQ(expected_flags[j], LibvpxVp8Encoder::EncodeFlags(tl_config))
>+          << j;
>+      tl.PopulateCodecSpecific(false, tl_config, &vp8_info, timestamp);
>+      tl.FrameEncoded(timestamp, kDefaultBytesPerFrame, kDefaultQp);
>+      EXPECT_TRUE(checker.CheckTemporalConfig(false, tl_config));
>+      EXPECT_EQ(expected_temporal_idx[j], tl_config.packetizer_temporal_idx);
>+      EXPECT_EQ(expected_temporal_idx[j], tl_config.encoder_layer_id);
>+      EXPECT_EQ(expected_layer_sync[j], tl_config.layer_sync);
>+      timestamp += 3000;
>+    }
>+
>     TemporalLayers::FrameConfig tl_config = tl.UpdateLayerConfig(timestamp);
>-    EXPECT_EQ(expected_flags[i], VP8EncoderImpl::EncodeFlags(tl_config)) << i;
>-    tl.PopulateCodecSpecific(true, tl_config, &vp8_info, 0);
>-    EXPECT_TRUE(checker.CheckTemporalConfig(true, tl_config));
>-    EXPECT_EQ(expected_temporal_idx[i], tl_config.packetizer_temporal_idx);
>-    EXPECT_EQ(expected_temporal_idx[i], tl_config.encoder_layer_id);
>+    tl.PopulateCodecSpecific(true, tl_config, &vp8_info, timestamp);
>+    tl.FrameEncoded(timestamp, kDefaultBytesPerFrame, kDefaultQp);
>+    EXPECT_TRUE(vp8_info.layerSync) << "Key frame should be marked layer sync.";
>     EXPECT_EQ(0, vp8_info.temporalIdx)
>         << "Key frame should always be packetized as layer 0";
>-    EXPECT_EQ(expected_layer_sync[i], tl_config.layer_sync);
>-    EXPECT_TRUE(vp8_info.layerSync) << "Key frame should be marked layer sync.";
>-    timestamp += 3000;
>+    EXPECT_TRUE(checker.CheckTemporalConfig(true, tl_config));
>   }
>-  TemporalLayers::FrameConfig tl_config = tl.UpdateLayerConfig(timestamp);
>-  EXPECT_EQ(expected_flags[7], VP8EncoderImpl::EncodeFlags(tl_config));
>-  tl.PopulateCodecSpecific(false, tl_config, &vp8_info, 0);
>-  EXPECT_TRUE(checker.CheckTemporalConfig(false, tl_config));
>-  EXPECT_NE(0, vp8_info.temporalIdx)
>-      << "To test something useful, this frame should not use layer 0.";
>-  EXPECT_EQ(expected_temporal_idx[7], vp8_info.temporalIdx)
>-      << "Non-keyframe, should use frame temporal index.";
>-  EXPECT_EQ(expected_temporal_idx[7], tl_config.packetizer_temporal_idx);
>-  EXPECT_EQ(expected_temporal_idx[7], tl_config.encoder_layer_id);
>-  EXPECT_FALSE(tl_config.layer_sync);
>-  EXPECT_TRUE(vp8_info.layerSync) << "Frame after keyframe should always be "
>-                                     "marked layer sync since it only depends "
>-                                     "on the base layer.";
> }
> 
> class TemporalLayersReferenceTest : public ::testing::TestWithParam<int> {
>@@ -347,9 +696,12 @@ INSTANTIATE_TEST_CASE_P(DefaultTemporalLayersTest,
> 
> TEST_P(TemporalLayersReferenceTest, ValidFrameConfigs) {
>   const int num_layers = GetParam();
>-  DefaultTemporalLayers tl(num_layers, 0);
>-  vpx_codec_enc_cfg_t cfg;
>-  tl.OnRatesUpdated(500, 500, 30);
>+  DefaultTemporalLayers tl(num_layers);
>+  Vp8EncoderConfig cfg;
>+  CodecSpecificInfoVP8 vp8_specifics;
>+  tl.OnRatesUpdated(
>+      GetTemporalLayerRates(kDefaultBytesPerFrame, kDefaultFramerate, 1),
>+      kDefaultFramerate);
>   tl.UpdateConfiguration(&cfg);
> 
>   // Run through the pattern and store the frame dependencies, plus keep track
>@@ -359,7 +711,9 @@ TEST_P(TemporalLayersReferenceTest, ValidFrameConfigs) {
>   // updates |last|.
>   std::vector<TemporalLayers::FrameConfig> tl_configs(kMaxPatternLength);
>   for (int i = 0; i < kMaxPatternLength; ++i) {
>-    TemporalLayers::FrameConfig tl_config = tl.UpdateLayerConfig(timestamp_++);
>+    TemporalLayers::FrameConfig tl_config = tl.UpdateLayerConfig(timestamp_);
>+    tl.PopulateCodecSpecific(i == 0, tl_config, &vp8_specifics, timestamp_);
>+    ++timestamp_;
>     EXPECT_FALSE(tl_config.drop_frame);
>     tl_configs.push_back(tl_config);
>     int temporal_idx = tl_config.encoder_layer_id;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/include/vp8.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/include/vp8.h
>index 9199405e346..580755f08de 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/include/vp8.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/include/vp8.h
>@@ -21,18 +21,18 @@ namespace webrtc {
> 
> class VP8Encoder : public VideoEncoder {
>  public:
>-  static bool IsSupported();
>   static std::unique_ptr<VP8Encoder> Create();
>+  static bool IsSupported();
> 
>-  virtual ~VP8Encoder() {}
>+  ~VP8Encoder() override {}
> };  // end of VP8Encoder class
> 
> class VP8Decoder : public VideoDecoder {
>  public:
>-  static bool IsSupported();
>   static std::unique_ptr<VP8Decoder> Create();
>+  static bool IsSupported();
> 
>-  virtual ~VP8Decoder() {}
>+  ~VP8Decoder() override {}
> };  // end of VP8Decoder class
> }  // namespace webrtc
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/include/vp8_common_types.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/include/vp8_common_types.h
>deleted file mode 100644
>index dff70ac3328..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/include/vp8_common_types.h
>+++ /dev/null
>@@ -1,29 +0,0 @@
>-/*
>- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_COMMON_TYPES_H_
>-#define MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_COMMON_TYPES_H_
>-
>-#include "common_types.h"  // NOLINT(build/include)
>-
>-namespace webrtc {
>-
>-// Ratio allocation between temporal streams:
>-// Values as required for the VP8 codec (accumulating).
>-static const float
>-    kVp8LayerRateAlloction[kMaxSimulcastStreams][kMaxTemporalStreams] = {
>-        {1.0f, 1.0f, 1.0f, 1.0f},  // 1 layer
>-        {0.6f, 1.0f, 1.0f, 1.0f},  // 2 layers {60%, 40%}
>-        {0.4f, 0.6f, 1.0f, 1.0f},  // 3 layers {40%, 20%, 40%}
>-        {0.25f, 0.4f, 0.6f, 1.0f}  // 4 layers {25%, 15%, 20%, 40%}
>-};
>-
>-}  // namespace webrtc
>-#endif  // MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_COMMON_TYPES_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/libvpx_vp8_decoder.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/libvpx_vp8_decoder.cc
>new file mode 100644
>index 00000000000..49959ef54cd
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/libvpx_vp8_decoder.cc
>@@ -0,0 +1,341 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <algorithm>
>+#include <string>
>+
>+#include "absl/memory/memory.h"
>+#include "common_video/libyuv/include/webrtc_libyuv.h"
>+#include "modules/video_coding/codecs/vp8/libvpx_vp8_decoder.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/numerics/exp_filter.h"
>+#include "rtc_base/timeutils.h"
>+#include "system_wrappers/include/field_trial.h"
>+#include "system_wrappers/include/metrics.h"
>+#include "third_party/libyuv/include/libyuv/convert.h"
>+#include "third_party/libyuv/include/libyuv/scale.h"
>+
>+namespace webrtc {
>+namespace {
>+constexpr int kVp8ErrorPropagationTh = 30;
>+// vpx_decoder.h documentation indicates decode deadline is time in us, with
>+// "Set to zero for unlimited.", but actual implementation requires this to be
>+// a mode with 0 meaning allow delay and 1 not allowing it.
>+constexpr long kDecodeDeadlineRealtime = 1;  // NOLINT
>+
>+const char kVp8PostProcArmFieldTrial[] = "WebRTC-VP8-Postproc-Config-Arm";
>+
>+void GetPostProcParamsFromFieldTrialGroup(
>+    LibvpxVp8Decoder::DeblockParams* deblock_params) {
>+  std::string group =
>+      webrtc::field_trial::FindFullName(kVp8PostProcArmFieldTrial);
>+  if (group.empty())
>+    return;
>+
>+  LibvpxVp8Decoder::DeblockParams params;
>+  if (sscanf(group.c_str(), "Enabled-%d,%d,%d", &params.max_level,
>+             &params.min_qp, &params.degrade_qp) != 3)
>+    return;
>+
>+  if (params.max_level < 0 || params.max_level > 16)
>+    return;
>+
>+  if (params.min_qp < 0 || params.degrade_qp <= params.min_qp)
>+    return;
>+
>+  *deblock_params = params;
>+}
>+
>+}  // namespace
>+
>+std::unique_ptr<VP8Decoder> VP8Decoder::Create() {
>+  return absl::make_unique<LibvpxVp8Decoder>();
>+}
>+
>+class LibvpxVp8Decoder::QpSmoother {
>+ public:
>+  QpSmoother() : last_sample_ms_(rtc::TimeMillis()), smoother_(kAlpha) {}
>+
>+  int GetAvg() const {
>+    float value = smoother_.filtered();
>+    return (value == rtc::ExpFilter::kValueUndefined) ? 0
>+                                                      : static_cast<int>(value);
>+  }
>+
>+  void Add(float sample) {
>+    int64_t now_ms = rtc::TimeMillis();
>+    smoother_.Apply(static_cast<float>(now_ms - last_sample_ms_), sample);
>+    last_sample_ms_ = now_ms;
>+  }
>+
>+  void Reset() { smoother_.Reset(kAlpha); }
>+
>+ private:
>+  const float kAlpha = 0.95f;
>+  int64_t last_sample_ms_;
>+  rtc::ExpFilter smoother_;
>+};
>+
>+LibvpxVp8Decoder::LibvpxVp8Decoder()
>+    : use_postproc_arm_(
>+          webrtc::field_trial::IsEnabled(kVp8PostProcArmFieldTrial)),
>+      buffer_pool_(false, 300 /* max_number_of_buffers*/),
>+      decode_complete_callback_(NULL),
>+      inited_(false),
>+      decoder_(NULL),
>+      propagation_cnt_(-1),
>+      last_frame_width_(0),
>+      last_frame_height_(0),
>+      key_frame_required_(true),
>+      qp_smoother_(use_postproc_arm_ ? new QpSmoother() : nullptr) {
>+  if (use_postproc_arm_)
>+    GetPostProcParamsFromFieldTrialGroup(&deblock_);
>+}
>+
>+LibvpxVp8Decoder::~LibvpxVp8Decoder() {
>+  inited_ = true;  // in order to do the actual release
>+  Release();
>+}
>+
>+int LibvpxVp8Decoder::InitDecode(const VideoCodec* inst, int number_of_cores) {
>+  int ret_val = Release();
>+  if (ret_val < 0) {
>+    return ret_val;
>+  }
>+  if (decoder_ == NULL) {
>+    decoder_ = new vpx_codec_ctx_t;
>+    memset(decoder_, 0, sizeof(*decoder_));
>+  }
>+  vpx_codec_dec_cfg_t cfg;
>+  // Setting number of threads to a constant value (1)
>+  cfg.threads = 1;
>+  cfg.h = cfg.w = 0;  // set after decode
>+
>+#if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64) || \
>+    defined(WEBRTC_ANDROID)
>+  vpx_codec_flags_t flags = use_postproc_arm_ ? VPX_CODEC_USE_POSTPROC : 0;
>+#else
>+  vpx_codec_flags_t flags = VPX_CODEC_USE_POSTPROC;
>+#endif
>+
>+  if (vpx_codec_dec_init(decoder_, vpx_codec_vp8_dx(), &cfg, flags)) {
>+    delete decoder_;
>+    decoder_ = nullptr;
>+    return WEBRTC_VIDEO_CODEC_MEMORY;
>+  }
>+
>+  propagation_cnt_ = -1;
>+  inited_ = true;
>+
>+  // Always start with a complete key frame.
>+  key_frame_required_ = true;
>+  return WEBRTC_VIDEO_CODEC_OK;
>+}
>+
>+int LibvpxVp8Decoder::Decode(const EncodedImage& input_image,
>+                             bool missing_frames,
>+                             const CodecSpecificInfo* codec_specific_info,
>+                             int64_t /*render_time_ms*/) {
>+  if (!inited_) {
>+    return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
>+  }
>+  if (decode_complete_callback_ == NULL) {
>+    return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
>+  }
>+  if (input_image._buffer == NULL && input_image._length > 0) {
>+    // Reset to avoid requesting key frames too often.
>+    if (propagation_cnt_ > 0)
>+      propagation_cnt_ = 0;
>+    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
>+  }
>+
>+// Post process configurations.
>+#if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64) || \
>+    defined(WEBRTC_ANDROID)
>+  if (use_postproc_arm_) {
>+    vp8_postproc_cfg_t ppcfg;
>+    ppcfg.post_proc_flag = VP8_MFQE;
>+    // For low resolutions, use stronger deblocking filter.
>+    int last_width_x_height = last_frame_width_ * last_frame_height_;
>+    if (last_width_x_height > 0 && last_width_x_height <= 320 * 240) {
>+      // Enable the deblock and demacroblocker based on qp thresholds.
>+      RTC_DCHECK(qp_smoother_);
>+      int qp = qp_smoother_->GetAvg();
>+      if (qp > deblock_.min_qp) {
>+        int level = deblock_.max_level;
>+        if (qp < deblock_.degrade_qp) {
>+          // Use lower level.
>+          level = deblock_.max_level * (qp - deblock_.min_qp) /
>+                  (deblock_.degrade_qp - deblock_.min_qp);
>+        }
>+        // Deblocking level only affects VP8_DEMACROBLOCK.
>+        ppcfg.deblocking_level = std::max(level, 1);
>+        ppcfg.post_proc_flag |= VP8_DEBLOCK | VP8_DEMACROBLOCK;
>+      }
>+    }
>+    vpx_codec_control(decoder_, VP8_SET_POSTPROC, &ppcfg);
>+  }
>+#else
>+  vp8_postproc_cfg_t ppcfg;
>+  // MFQE enabled to reduce key frame popping.
>+  ppcfg.post_proc_flag = VP8_MFQE | VP8_DEBLOCK;
>+  // For VGA resolutions and lower, enable the demacroblocker postproc.
>+  if (last_frame_width_ * last_frame_height_ <= 640 * 360) {
>+    ppcfg.post_proc_flag |= VP8_DEMACROBLOCK;
>+  }
>+  // Strength of deblocking filter. Valid range:[0,16]
>+  ppcfg.deblocking_level = 3;
>+  vpx_codec_control(decoder_, VP8_SET_POSTPROC, &ppcfg);
>+#endif
>+
>+  // Always start with a complete key frame.
>+  if (key_frame_required_) {
>+    if (input_image._frameType != kVideoFrameKey)
>+      return WEBRTC_VIDEO_CODEC_ERROR;
>+    // We have a key frame - is it complete?
>+    if (input_image._completeFrame) {
>+      key_frame_required_ = false;
>+    } else {
>+      return WEBRTC_VIDEO_CODEC_ERROR;
>+    }
>+  }
>+  // Restrict error propagation using key frame requests.
>+  // Reset on a key frame refresh.
>+  if (input_image._frameType == kVideoFrameKey && input_image._completeFrame) {
>+    propagation_cnt_ = -1;
>+    // Start count on first loss.
>+  } else if ((!input_image._completeFrame || missing_frames) &&
>+             propagation_cnt_ == -1) {
>+    propagation_cnt_ = 0;
>+  }
>+  if (propagation_cnt_ >= 0) {
>+    propagation_cnt_++;
>+  }
>+
>+  vpx_codec_iter_t iter = NULL;
>+  vpx_image_t* img;
>+  int ret;
>+
>+  // Check for missing frames.
>+  if (missing_frames) {
>+    // Call decoder with zero data length to signal missing frames.
>+    if (vpx_codec_decode(decoder_, NULL, 0, 0, kDecodeDeadlineRealtime)) {
>+      // Reset to avoid requesting key frames too often.
>+      if (propagation_cnt_ > 0)
>+        propagation_cnt_ = 0;
>+      return WEBRTC_VIDEO_CODEC_ERROR;
>+    }
>+    img = vpx_codec_get_frame(decoder_, &iter);
>+    iter = NULL;
>+  }
>+
>+  uint8_t* buffer = input_image._buffer;
>+  if (input_image._length == 0) {
>+    buffer = NULL;  // Triggers full frame concealment.
>+  }
>+  if (vpx_codec_decode(decoder_, buffer, input_image._length, 0,
>+                       kDecodeDeadlineRealtime)) {
>+    // Reset to avoid requesting key frames too often.
>+    if (propagation_cnt_ > 0) {
>+      propagation_cnt_ = 0;
>+    }
>+    return WEBRTC_VIDEO_CODEC_ERROR;
>+  }
>+
>+  img = vpx_codec_get_frame(decoder_, &iter);
>+  int qp;
>+  vpx_codec_err_t vpx_ret =
>+      vpx_codec_control(decoder_, VPXD_GET_LAST_QUANTIZER, &qp);
>+  RTC_DCHECK_EQ(vpx_ret, VPX_CODEC_OK);
>+  ret = ReturnFrame(img, input_image._timeStamp, input_image.ntp_time_ms_, qp);
>+  if (ret != 0) {
>+    // Reset to avoid requesting key frames too often.
>+    if (ret < 0 && propagation_cnt_ > 0)
>+      propagation_cnt_ = 0;
>+    return ret;
>+  }
>+  // Check Vs. threshold
>+  if (propagation_cnt_ > kVp8ErrorPropagationTh) {
>+    // Reset to avoid requesting key frames too often.
>+    propagation_cnt_ = 0;
>+    return WEBRTC_VIDEO_CODEC_ERROR;
>+  }
>+  return WEBRTC_VIDEO_CODEC_OK;
>+}
>+
>+int LibvpxVp8Decoder::ReturnFrame(const vpx_image_t* img,
>+                                  uint32_t timestamp,
>+                                  int64_t ntp_time_ms,
>+                                  int qp) {
>+  if (img == NULL) {
>+    // Decoder OK and NULL image => No show frame
>+    return WEBRTC_VIDEO_CODEC_NO_OUTPUT;
>+  }
>+  if (qp_smoother_) {
>+    if (last_frame_width_ != static_cast<int>(img->d_w) ||
>+        last_frame_height_ != static_cast<int>(img->d_h)) {
>+      qp_smoother_->Reset();
>+    }
>+    qp_smoother_->Add(qp);
>+  }
>+  last_frame_width_ = img->d_w;
>+  last_frame_height_ = img->d_h;
>+  // Allocate memory for decoded image.
>+  rtc::scoped_refptr<I420Buffer> buffer =
>+      buffer_pool_.CreateBuffer(img->d_w, img->d_h);
>+  if (!buffer.get()) {
>+    // Pool has too many pending frames.
>+    RTC_HISTOGRAM_BOOLEAN("WebRTC.Video.LibvpxVp8Decoder.TooManyPendingFrames",
>+                          1);
>+    return WEBRTC_VIDEO_CODEC_NO_OUTPUT;
>+  }
>+
>+  libyuv::I420Copy(img->planes[VPX_PLANE_Y], img->stride[VPX_PLANE_Y],
>+                   img->planes[VPX_PLANE_U], img->stride[VPX_PLANE_U],
>+                   img->planes[VPX_PLANE_V], img->stride[VPX_PLANE_V],
>+                   buffer->MutableDataY(), buffer->StrideY(),
>+                   buffer->MutableDataU(), buffer->StrideU(),
>+                   buffer->MutableDataV(), buffer->StrideV(), img->d_w,
>+                   img->d_h);
>+
>+  VideoFrame decoded_image(buffer, timestamp, 0, kVideoRotation_0);
>+  decoded_image.set_ntp_time_ms(ntp_time_ms);
>+  decode_complete_callback_->Decoded(decoded_image, absl::nullopt, qp);
>+
>+  return WEBRTC_VIDEO_CODEC_OK;
>+}
>+
>+int LibvpxVp8Decoder::RegisterDecodeCompleteCallback(
>+    DecodedImageCallback* callback) {
>+  decode_complete_callback_ = callback;
>+  return WEBRTC_VIDEO_CODEC_OK;
>+}
>+
>+int LibvpxVp8Decoder::Release() {
>+  int ret_val = WEBRTC_VIDEO_CODEC_OK;
>+
>+  if (decoder_ != NULL) {
>+    if (inited_) {
>+      if (vpx_codec_destroy(decoder_)) {
>+        ret_val = WEBRTC_VIDEO_CODEC_MEMORY;
>+      }
>+    }
>+    delete decoder_;
>+    decoder_ = NULL;
>+  }
>+  buffer_pool_.Release();
>+  inited_ = false;
>+  return ret_val;
>+}
>+
>+const char* LibvpxVp8Decoder::ImplementationName() const {
>+  return "libvpx";
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/libvpx_vp8_decoder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/libvpx_vp8_decoder.h
>new file mode 100644
>index 00000000000..9771b9e9788
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/libvpx_vp8_decoder.h
>@@ -0,0 +1,74 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_VIDEO_CODING_CODECS_VP8_LIBVPX_VP8_DECODER_H_
>+#define MODULES_VIDEO_CODING_CODECS_VP8_LIBVPX_VP8_DECODER_H_
>+
>+#include <memory>
>+
>+#include "api/video_codecs/video_decoder.h"
>+#include "common_types.h"  // NOLINT(build/include)
>+#include "common_video/include/i420_buffer_pool.h"
>+#include "common_video/include/video_frame.h"
>+#include "modules/include/module_common_types.h"
>+#include "modules/video_coding/codecs/vp8/include/vp8.h"
>+#include "modules/video_coding/include/video_codec_interface.h"
>+#include "vpx/vp8dx.h"
>+#include "vpx/vpx_decoder.h"
>+
>+namespace webrtc {
>+
>+class LibvpxVp8Decoder : public VP8Decoder {
>+ public:
>+  LibvpxVp8Decoder();
>+  ~LibvpxVp8Decoder() override;
>+
>+  int InitDecode(const VideoCodec* inst, int number_of_cores) override;
>+
>+  int Decode(const EncodedImage& input_image,
>+             bool missing_frames,
>+             const CodecSpecificInfo* codec_specific_info,
>+             int64_t /*render_time_ms*/) override;
>+
>+  int RegisterDecodeCompleteCallback(DecodedImageCallback* callback) override;
>+  int Release() override;
>+
>+  const char* ImplementationName() const override;
>+
>+  struct DeblockParams {
>+    int max_level = 6;   // Deblocking strength: [0, 16].
>+    int degrade_qp = 1;  // If QP value is below, start lowering |max_level|.
>+    int min_qp = 0;      // If QP value is below, turn off deblocking.
>+  };
>+
>+ private:
>+  class QpSmoother;
>+  int ReturnFrame(const vpx_image_t* img,
>+                  uint32_t timeStamp,
>+                  int64_t ntp_time_ms,
>+                  int qp);
>+
>+  const bool use_postproc_arm_;
>+
>+  I420BufferPool buffer_pool_;
>+  DecodedImageCallback* decode_complete_callback_;
>+  bool inited_;
>+  vpx_codec_ctx_t* decoder_;
>+  int propagation_cnt_;
>+  int last_frame_width_;
>+  int last_frame_height_;
>+  bool key_frame_required_;
>+  DeblockParams deblock_;
>+  const std::unique_ptr<QpSmoother> qp_smoother_;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_VIDEO_CODING_CODECS_VP8_LIBVPX_VP8_DECODER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc
>similarity index 62%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc
>index 13820e31753..9e9b47c4edb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.cc
>@@ -8,43 +8,33 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "modules/video_coding/codecs/vp8/vp8_impl.h"
>-
>-#include <stdlib.h>
>-#include <string.h>
>-#include <time.h>
> #include <algorithm>
> #include <string>
>+#include <vector>
> 
>-#include "common_types.h"  // NOLINT(build/include)
>+#include "absl/memory/memory.h"
> #include "common_video/libyuv/include/webrtc_libyuv.h"
>-#include "modules/include/module_common_types.h"
>-#include "modules/video_coding/codecs/vp8/include/vp8_common_types.h"
>-#include "modules/video_coding/codecs/vp8/screenshare_layers.h"
>-#include "modules/video_coding/codecs/vp8/simulcast_rate_allocator.h"
>-#include "modules/video_coding/codecs/vp8/temporal_layers.h"
>-#include "modules/video_coding/include/video_codec_interface.h"
>+#include "modules/video_coding/codecs/vp8/libvpx_vp8_encoder.h"
>+#include "modules/video_coding/utility/simulcast_rate_allocator.h"
>+#include "modules/video_coding/utility/simulcast_utility.h"
> #include "rtc_base/checks.h"
>-#include "rtc_base/numerics/exp_filter.h"
>-#include "rtc_base/ptr_util.h"
>-#include "rtc_base/random.h"
> #include "rtc_base/timeutils.h"
> #include "rtc_base/trace_event.h"
>-#include "system_wrappers/include/clock.h"
> #include "system_wrappers/include/field_trial.h"
>-#include "system_wrappers/include/metrics.h"
> #include "third_party/libyuv/include/libyuv/convert.h"
> #include "third_party/libyuv/include/libyuv/scale.h"
> 
> namespace webrtc {
> namespace {
>-
>-const char kVp8PostProcArmFieldTrial[] = "WebRTC-VP8-Postproc-Config-Arm";
> const char kVp8GfBoostFieldTrial[] = "WebRTC-VP8-GfBoost";
> 
>-const int kTokenPartitions = VP8_ONE_TOKENPARTITION;
>-enum { kVp8ErrorPropagationTh = 30 };
>-enum { kVp832ByteAlign = 32 };
>+// QP is obtained from VP8-bitstream for HW, so the QP corresponds to the
>+// bitstream range of [0, 127] and not the user-level range of [0,63].
>+constexpr int kLowVp8QpThreshold = 29;
>+constexpr int kHighVp8QpThreshold = 95;
>+
>+constexpr int kTokenPartitions = VP8_ONE_TOKENPARTITION;
>+constexpr uint32_t kVp832ByteAlign = 32u;
> 
> // VP8 denoiser states.
> enum denoiserState {
>@@ -58,7 +48,7 @@ enum denoiserState {
> };
> 
> // Greatest common divisior
>-int GCD(int a, int b) {
>+static int GCD(int a, int b) {
>   int c = a % b;
>   while (c != 0) {
>     a = b;
>@@ -68,53 +58,6 @@ int GCD(int a, int b) {
>   return b;
> }
> 
>-uint32_t SumStreamMaxBitrate(int streams, const VideoCodec& codec) {
>-  uint32_t bitrate_sum = 0;
>-  for (int i = 0; i < streams; ++i) {
>-    bitrate_sum += codec.simulcastStream[i].maxBitrate;
>-  }
>-  return bitrate_sum;
>-}
>-
>-int NumberOfStreams(const VideoCodec& codec) {
>-  int streams =
>-      codec.numberOfSimulcastStreams < 1 ? 1 : codec.numberOfSimulcastStreams;
>-  uint32_t simulcast_max_bitrate = SumStreamMaxBitrate(streams, codec);
>-  if (simulcast_max_bitrate == 0) {
>-    streams = 1;
>-  }
>-  return streams;
>-}
>-
>-bool ValidSimulcastResolutions(const VideoCodec& codec, int num_streams) {
>-  if (codec.width != codec.simulcastStream[num_streams - 1].width ||
>-      codec.height != codec.simulcastStream[num_streams - 1].height) {
>-    return false;
>-  }
>-  for (int i = 0; i < num_streams; ++i) {
>-    if (codec.width * codec.simulcastStream[i].height !=
>-        codec.height * codec.simulcastStream[i].width) {
>-      return false;
>-    }
>-  }
>-  for (int i = 1; i < num_streams; ++i) {
>-    if (codec.simulcastStream[i].width !=
>-        codec.simulcastStream[i - 1].width * 2) {
>-      return false;
>-    }
>-  }
>-  return true;
>-}
>-
>-bool ValidSimulcastTemporalLayers(const VideoCodec& codec, int num_streams) {
>-  for (int i = 0; i < num_streams - 1; ++i) {
>-    if (codec.simulcastStream[i].numberOfTemporalLayers !=
>-        codec.simulcastStream[i + 1].numberOfTemporalLayers)
>-      return false;
>-  }
>-  return true;
>-}
>-
> bool GetGfBoostPercentageFromFieldTrialGroup(int* boost_percentage) {
>   std::string group = webrtc::field_trial::FindFullName(kVp8GfBoostFieldTrial);
>   if (group.empty())
>@@ -129,46 +72,61 @@ bool GetGfBoostPercentageFromFieldTrialGroup(int* boost_percentage) {
>   return true;
> }
> 
>-void GetPostProcParamsFromFieldTrialGroup(
>-    VP8DecoderImpl::DeblockParams* deblock_params) {
>-  std::string group =
>-      webrtc::field_trial::FindFullName(kVp8PostProcArmFieldTrial);
>-  if (group.empty())
>-    return;
>-
>-  VP8DecoderImpl::DeblockParams params;
>-  if (sscanf(group.c_str(), "Enabled-%d,%d,%d", &params.max_level,
>-             &params.min_qp, &params.degrade_qp) != 3)
>-    return;
>-
>-  if (params.max_level < 0 || params.max_level > 16)
>-    return;
>-
>-  if (params.min_qp < 0 || params.degrade_qp <= params.min_qp)
>-    return;
>-
>-  *deblock_params = params;
>+static_assert(
>+    VP8_TS_MAX_PERIODICITY == VPX_TS_MAX_PERIODICITY,
>+    "VP8_TS_MAX_PERIODICITY must be kept in sync with the constant in libvpx.");
>+static_assert(
>+    VP8_TS_MAX_LAYERS == VPX_TS_MAX_LAYERS,
>+    "VP8_TS_MAX_LAYERS must be kept in sync with the constant in libvpx.");
>+
>+static Vp8EncoderConfig GetEncoderConfig(vpx_codec_enc_cfg* vpx_config) {
>+  Vp8EncoderConfig config;
>+
>+  config.ts_number_layers = vpx_config->ts_number_layers;
>+  memcpy(config.ts_target_bitrate, vpx_config->ts_target_bitrate,
>+         sizeof(unsigned int) * VP8_TS_MAX_LAYERS);
>+  memcpy(config.ts_rate_decimator, vpx_config->ts_rate_decimator,
>+         sizeof(unsigned int) * VP8_TS_MAX_LAYERS);
>+  config.ts_periodicity = vpx_config->ts_periodicity;
>+  memcpy(config.ts_layer_id, vpx_config->ts_layer_id,
>+         sizeof(unsigned int) * VP8_TS_MAX_PERIODICITY);
>+  config.rc_target_bitrate = vpx_config->rc_target_bitrate;
>+  config.rc_min_quantizer = vpx_config->rc_min_quantizer;
>+  config.rc_max_quantizer = vpx_config->rc_max_quantizer;
>+
>+  return config;
>+}
>+
>+static void FillInEncoderConfig(vpx_codec_enc_cfg* vpx_config,
>+                                const Vp8EncoderConfig& config) {
>+  vpx_config->ts_number_layers = config.ts_number_layers;
>+  memcpy(vpx_config->ts_target_bitrate, config.ts_target_bitrate,
>+         sizeof(unsigned int) * VP8_TS_MAX_LAYERS);
>+  memcpy(vpx_config->ts_rate_decimator, config.ts_rate_decimator,
>+         sizeof(unsigned int) * VP8_TS_MAX_LAYERS);
>+  vpx_config->ts_periodicity = config.ts_periodicity;
>+  memcpy(vpx_config->ts_layer_id, config.ts_layer_id,
>+         sizeof(unsigned int) * VP8_TS_MAX_PERIODICITY);
>+  vpx_config->rc_target_bitrate = config.rc_target_bitrate;
>+  vpx_config->rc_min_quantizer = config.rc_min_quantizer;
>+  vpx_config->rc_max_quantizer = config.rc_max_quantizer;
>+}
>+
>+bool UpdateVpxConfiguration(TemporalLayers* temporal_layers,
>+                            vpx_codec_enc_cfg_t* cfg) {
>+  Vp8EncoderConfig config = GetEncoderConfig(cfg);
>+  const bool res = temporal_layers->UpdateConfiguration(&config);
>+  if (res)
>+    FillInEncoderConfig(cfg, config);
>+  return res;
> }
>-
> }  // namespace
> 
>-bool VP8Encoder::IsSupported() {
>-  return true;
>-}
>-
> std::unique_ptr<VP8Encoder> VP8Encoder::Create() {
>-  return rtc::MakeUnique<VP8EncoderImpl>();
>+  return absl::make_unique<LibvpxVp8Encoder>();
> }
> 
>-bool VP8Decoder::IsSupported() {
>-  return true;
>-}
>-
>-std::unique_ptr<VP8Decoder> VP8Decoder::Create() {
>-  return rtc::MakeUnique<VP8DecoderImpl>();
>-}
>-
>-vpx_enc_frame_flags_t VP8EncoderImpl::EncodeFlags(
>+vpx_enc_frame_flags_t LibvpxVp8Encoder::EncodeFlags(
>     const TemporalLayers::FrameConfig& references) {
>   RTC_DCHECK(!references.drop_frame);
> 
>@@ -192,7 +150,7 @@ vpx_enc_frame_flags_t VP8EncoderImpl::EncodeFlags(
>   return flags;
> }
> 
>-VP8EncoderImpl::VP8EncoderImpl()
>+LibvpxVp8Encoder::LibvpxVp8Encoder()
>     : use_gf_boost_(webrtc::field_trial::IsEnabled(kVp8GfBoostFieldTrial)),
>       encoded_complete_callback_(nullptr),
>       inited_(false),
>@@ -202,12 +160,6 @@ VP8EncoderImpl::VP8EncoderImpl()
>       number_of_cores_(0),
>       rc_max_intra_target_(0),
>       key_frame_request_(kMaxSimulcastStreams, false) {
>-  Random random(rtc::TimeMicros());
>-  picture_id_.reserve(kMaxSimulcastStreams);
>-  for (int i = 0; i < kMaxSimulcastStreams; ++i) {
>-    picture_id_.push_back(random.Rand<uint16_t>() & 0x7FFF);
>-    tl0_pic_idx_.push_back(random.Rand<uint8_t>());
>-  }
>   temporal_layers_.reserve(kMaxSimulcastStreams);
>   temporal_layers_checkers_.reserve(kMaxSimulcastStreams);
>   raw_images_.reserve(kMaxSimulcastStreams);
>@@ -219,11 +171,11 @@ VP8EncoderImpl::VP8EncoderImpl()
>   downsampling_factors_.reserve(kMaxSimulcastStreams);
> }
> 
>-VP8EncoderImpl::~VP8EncoderImpl() {
>+LibvpxVp8Encoder::~LibvpxVp8Encoder() {
>   Release();
> }
> 
>-int VP8EncoderImpl::Release() {
>+int LibvpxVp8Encoder::Release() {
>   int ret_val = WEBRTC_VIDEO_CODEC_OK;
> 
>   while (!encoded_images_.empty()) {
>@@ -233,8 +185,10 @@ int VP8EncoderImpl::Release() {
>   }
>   while (!encoders_.empty()) {
>     vpx_codec_ctx_t& encoder = encoders_.back();
>-    if (vpx_codec_destroy(&encoder)) {
>-      ret_val = WEBRTC_VIDEO_CODEC_MEMORY;
>+    if (inited_) {
>+      if (vpx_codec_destroy(&encoder)) {
>+        ret_val = WEBRTC_VIDEO_CODEC_MEMORY;
>+      }
>     }
>     encoders_.pop_back();
>   }
>@@ -245,17 +199,14 @@ int VP8EncoderImpl::Release() {
>     vpx_img_free(&raw_images_.back());
>     raw_images_.pop_back();
>   }
>-  for (size_t i = 0; i < temporal_layers_.size(); ++i) {
>-    tl0_pic_idx_[i] = temporal_layers_[i]->Tl0PicIdx();
>-  }
>   temporal_layers_.clear();
>   temporal_layers_checkers_.clear();
>   inited_ = false;
>   return ret_val;
> }
> 
>-int VP8EncoderImpl::SetRateAllocation(const BitrateAllocation& bitrate,
>-                                      uint32_t new_framerate) {
>+int LibvpxVp8Encoder::SetRateAllocation(const VideoBitrateAllocation& bitrate,
>+                                        uint32_t new_framerate) {
>   if (!inited_)
>     return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
> 
>@@ -306,7 +257,13 @@ int VP8EncoderImpl::SetRateAllocation(const BitrateAllocation& bitrate,
>       SetStreamState(send_stream, stream_idx);
> 
>     configurations_[i].rc_target_bitrate = target_bitrate_kbps;
>-    temporal_layers_[stream_idx]->UpdateConfiguration(&configurations_[i]);
>+    if (send_stream) {
>+      temporal_layers_[stream_idx]->OnRatesUpdated(
>+          bitrate.GetTemporalLayerAllocation(stream_idx), new_framerate);
>+    }
>+
>+    UpdateVpxConfiguration(temporal_layers_[stream_idx].get(),
>+                           &configurations_[i]);
> 
>     if (vpx_codec_enc_config_set(&encoders_[i], &configurations_[i])) {
>       return WEBRTC_VIDEO_CODEC_ERROR;
>@@ -315,11 +272,11 @@ int VP8EncoderImpl::SetRateAllocation(const BitrateAllocation& bitrate,
>   return WEBRTC_VIDEO_CODEC_OK;
> }
> 
>-const char* VP8EncoderImpl::ImplementationName() const {
>+const char* LibvpxVp8Encoder::ImplementationName() const {
>   return "libvpx";
> }
> 
>-void VP8EncoderImpl::SetStreamState(bool send_stream, int stream_idx) {
>+void LibvpxVp8Encoder::SetStreamState(bool send_stream, int stream_idx) {
>   if (send_stream && !send_stream_[stream_idx]) {
>     // Need a key frame if we have not sent this stream before.
>     key_frame_request_[stream_idx] = true;
>@@ -327,32 +284,21 @@ void VP8EncoderImpl::SetStreamState(bool send_stream, int stream_idx) {
>   send_stream_[stream_idx] = send_stream;
> }
> 
>-void VP8EncoderImpl::SetupTemporalLayers(int num_streams,
>-                                         int num_temporal_layers,
>-                                         const VideoCodec& codec) {
>-  RTC_DCHECK(codec.VP8().tl_factory != nullptr);
>-  const TemporalLayersFactory* tl_factory = codec.VP8().tl_factory;
>-  if (num_streams == 1) {
>+void LibvpxVp8Encoder::SetupTemporalLayers(int num_streams,
>+                                           int num_temporal_layers,
>+                                           const VideoCodec& codec) {
>+  RTC_DCHECK(temporal_layers_.empty());
>+  for (int i = 0; i < num_streams; ++i) {
>     temporal_layers_.emplace_back(
>-        tl_factory->Create(0, num_temporal_layers, tl0_pic_idx_[0]));
>+        TemporalLayers::CreateTemporalLayers(codec, i));
>     temporal_layers_checkers_.emplace_back(
>-        tl_factory->CreateChecker(0, num_temporal_layers, tl0_pic_idx_[0]));
>-  } else {
>-    for (int i = 0; i < num_streams; ++i) {
>-      RTC_CHECK_GT(num_temporal_layers, 0);
>-      int layers = std::max(static_cast<uint8_t>(1),
>-                            codec.simulcastStream[i].numberOfTemporalLayers);
>-      temporal_layers_.emplace_back(
>-          tl_factory->Create(i, layers, tl0_pic_idx_[i]));
>-      temporal_layers_checkers_.emplace_back(
>-          tl_factory->CreateChecker(i, layers, tl0_pic_idx_[i]));
>-    }
>+        TemporalLayers::CreateTemporalLayersChecker(codec, i));
>   }
> }
> 
>-int VP8EncoderImpl::InitEncode(const VideoCodec* inst,
>-                               int number_of_cores,
>-                               size_t /*maxPayloadSize */) {
>+int LibvpxVp8Encoder::InitEncode(const VideoCodec* inst,
>+                                 int number_of_cores,
>+                                 size_t /*maxPayloadSize */) {
>   if (inst == NULL) {
>     return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
>   }
>@@ -377,12 +323,13 @@ int VP8EncoderImpl::InitEncode(const VideoCodec* inst,
>     return retVal;
>   }
> 
>-  int number_of_streams = NumberOfStreams(*inst);
>+  int number_of_streams = SimulcastUtility::NumberOfSimulcastStreams(*inst);
>   bool doing_simulcast = (number_of_streams > 1);
> 
>-  if (doing_simulcast &&
>-      (!ValidSimulcastResolutions(*inst, number_of_streams) ||
>-       !ValidSimulcastTemporalLayers(*inst, number_of_streams))) {
>+  if (doing_simulcast && (!SimulcastUtility::ValidSimulcastResolutions(
>+                              *inst, number_of_streams) ||
>+                          !SimulcastUtility::ValidSimulcastTemporalLayers(
>+                              *inst, number_of_streams))) {
>     return WEBRTC_VIDEO_CODEC_ERR_SIMULCAST_PARAMETERS_NOT_SUPPORTED;
>   }
> 
>@@ -447,17 +394,9 @@ int VP8EncoderImpl::InitEncode(const VideoCodec* inst,
>   configurations_[0].g_timebase.den = 90000;
>   configurations_[0].g_lag_in_frames = 0;  // 0- no frame lagging
> 
>-  // Set the error resilience mode according to user settings.
>-  switch (inst->VP8().resilience) {
>-    case kResilienceOff:
>-      configurations_[0].g_error_resilient = 0;
>-      break;
>-    case kResilientStream:
>-      configurations_[0].g_error_resilient = VPX_ERROR_RESILIENT_DEFAULT;
>-      break;
>-    case kResilientFrames:
>-      return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;  // Not supported
>-  }
>+  // Set the error resilience mode for temporal layers (but not simulcast).
>+  configurations_[0].g_error_resilient =
>+      (num_temporal_layers > 1) ? VPX_ERROR_RESILIENT_DEFAULT : 0;
> 
>   // rate control settings
>   configurations_[0].rc_dropframe_thresh = inst->VP8().frameDroppingOn ? 30 : 0;
>@@ -488,13 +427,13 @@ int VP8EncoderImpl::InitEncode(const VideoCodec* inst,
> 
>   // Allow the user to set the complexity for the base stream.
>   switch (inst->VP8().complexity) {
>-    case kComplexityHigh:
>+    case VideoCodecComplexity::kComplexityHigh:
>       cpu_speed_[0] = -5;
>       break;
>-    case kComplexityHigher:
>+    case VideoCodecComplexity::kComplexityHigher:
>       cpu_speed_[0] = -4;
>       break;
>-    case kComplexityMax:
>+    case VideoCodecComplexity::kComplexityMax:
>       cpu_speed_[0] = -3;
>       break;
>     default:
>@@ -526,8 +465,8 @@ int VP8EncoderImpl::InitEncode(const VideoCodec* inst,
>   // Note the order we use is different from webm, we have lowest resolution
>   // at position 0 and they have highest resolution at position 0.
>   int stream_idx = encoders_.size() - 1;
>-  SimulcastRateAllocator init_allocator(codec_, nullptr);
>-  BitrateAllocation allocation = init_allocator.GetAllocation(
>+  SimulcastRateAllocator init_allocator(codec_);
>+  VideoBitrateAllocation allocation = init_allocator.GetAllocation(
>       inst->startBitrate * 1000, inst->maxFramerate);
>   std::vector<uint32_t> stream_bitrates;
>   for (int i = 0; i == 0 || i < inst->numberOfSimulcastStreams; ++i) {
>@@ -536,9 +475,13 @@ int VP8EncoderImpl::InitEncode(const VideoCodec* inst,
>   }
> 
>   configurations_[0].rc_target_bitrate = stream_bitrates[stream_idx];
>-  temporal_layers_[stream_idx]->OnRatesUpdated(
>-      stream_bitrates[stream_idx], inst->maxBitrate, inst->maxFramerate);
>-  temporal_layers_[stream_idx]->UpdateConfiguration(&configurations_[0]);
>+  if (stream_bitrates[stream_idx] > 0) {
>+    temporal_layers_[stream_idx]->OnRatesUpdated(
>+        allocation.GetTemporalLayerAllocation(stream_idx), inst->maxFramerate);
>+  }
>+  UpdateVpxConfiguration(temporal_layers_[stream_idx].get(),
>+                         &configurations_[0]);
>+
>   --stream_idx;
>   for (size_t i = 1; i < encoders_.size(); ++i, --stream_idx) {
>     memcpy(&configurations_[i], &configurations_[0],
>@@ -558,17 +501,21 @@ int VP8EncoderImpl::InitEncode(const VideoCodec* inst,
>                   inst->simulcastStream[stream_idx].height, kVp832ByteAlign);
>     SetStreamState(stream_bitrates[stream_idx] > 0, stream_idx);
>     configurations_[i].rc_target_bitrate = stream_bitrates[stream_idx];
>-    temporal_layers_[stream_idx]->OnRatesUpdated(
>-        stream_bitrates[stream_idx], inst->maxBitrate, inst->maxFramerate);
>-    temporal_layers_[stream_idx]->UpdateConfiguration(&configurations_[i]);
>+    if (stream_bitrates[stream_idx] > 0) {
>+      temporal_layers_[stream_idx]->OnRatesUpdated(
>+          allocation.GetTemporalLayerAllocation(stream_idx),
>+          inst->maxFramerate);
>+    }
>+    UpdateVpxConfiguration(temporal_layers_[stream_idx].get(),
>+                           &configurations_[i]);
>   }
> 
>   return InitAndSetControlSettings();
> }
> 
>-int VP8EncoderImpl::SetCpuSpeed(int width, int height) {
>-#if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64) \
>-  || defined(WEBRTC_ANDROID)
>+int LibvpxVp8Encoder::SetCpuSpeed(int width, int height) {
>+#if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64) || \
>+    defined(WEBRTC_ANDROID)
>   // On mobile platform, use a lower speed setting for lower resolutions for
>   // CPUs with 4 or more cores.
>   RTC_DCHECK_GT(number_of_cores_, 0);
>@@ -592,7 +539,7 @@ int VP8EncoderImpl::SetCpuSpeed(int width, int height) {
> #endif
> }
> 
>-int VP8EncoderImpl::NumberOfThreads(int width, int height, int cpus) {
>+int LibvpxVp8Encoder::NumberOfThreads(int width, int height, int cpus) {
> #if defined(WEBRTC_ANDROID)
>   if (width * height >= 320 * 180) {
>     if (cpus >= 4) {
>@@ -613,7 +560,12 @@ int VP8EncoderImpl::NumberOfThreads(int width, int height, int cpus) {
>     // 3 threads for 1080p.
>     return 3;
>   } else if (width * height > 640 * 480 && cpus >= 3) {
>-    // 2 threads for qHD/HD.
>+    // Default 2 threads for qHD/HD, but allow 3 if core count is high enough,
>+    // as this will allow more margin for high-core/low clock machines or if
>+    // not built with highest optimization.
>+    if (cpus >= 6) {
>+      return 3;
>+    }
>     return 2;
>   } else {
>     // 1 thread for VGA or less.
>@@ -622,7 +574,7 @@ int VP8EncoderImpl::NumberOfThreads(int width, int height, int cpus) {
> #endif
> }
> 
>-int VP8EncoderImpl::InitAndSetControlSettings() {
>+int LibvpxVp8Encoder::InitAndSetControlSettings() {
>   vpx_codec_flags_t flags = 0;
>   flags |= VPX_CODEC_USE_OUTPUT_PARTITION;
> 
>@@ -647,8 +599,8 @@ int VP8EncoderImpl::InitAndSetControlSettings() {
>   // when encoding lower resolution streams. Would it work with the
>   // multi-res encoding feature?
>   denoiserState denoiser_state = kDenoiserOnYOnly;
>-#if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64) \
>-  || defined(WEBRTC_ANDROID)
>+#if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64) || \
>+    defined(WEBRTC_ANDROID)
>   denoiser_state = kDenoiserOnYOnly;
> #else
>   denoiser_state = kDenoiserOnAdaptive;
>@@ -663,7 +615,7 @@ int VP8EncoderImpl::InitAndSetControlSettings() {
>   for (size_t i = 0; i < encoders_.size(); ++i) {
>     // Allow more screen content to be detected as static.
>     vpx_codec_control(&(encoders_[i]), VP8E_SET_STATIC_THRESHOLD,
>-                      codec_.mode == kScreensharing ? 300 : 1);
>+                      codec_.mode == VideoCodecMode::kScreensharing ? 300 : 1);
>     vpx_codec_control(&(encoders_[i]), VP8E_SET_CPUUSED, cpu_speed_[i]);
>     vpx_codec_control(&(encoders_[i]), VP8E_SET_TOKEN_PARTITIONS,
>                       static_cast<vp8e_token_partitions>(kTokenPartitions));
>@@ -672,9 +624,9 @@ int VP8EncoderImpl::InitAndSetControlSettings() {
>     // VP8E_SET_SCREEN_CONTENT_MODE 2 = screen content with more aggressive
>     // rate control (drop frames on large target bitrate overshoot)
>     vpx_codec_control(&(encoders_[i]), VP8E_SET_SCREEN_CONTENT_MODE,
>-                      codec_.mode == kScreensharing ? 2 : 0);
>+                      codec_.mode == VideoCodecMode::kScreensharing ? 2 : 0);
>     // Apply boost on golden frames (has only effect when resilience is off).
>-    if (use_gf_boost_ && codec_.VP8()->resilience == kResilienceOff) {
>+    if (use_gf_boost_ && configurations_[0].g_error_resilient == 0) {
>       int gf_boost_percent;
>       if (GetGfBoostPercentageFromFieldTrialGroup(&gf_boost_percent)) {
>         vpx_codec_control(&(encoders_[i]), VP8E_SET_GF_CBR_BOOST_PCT,
>@@ -686,7 +638,7 @@ int VP8EncoderImpl::InitAndSetControlSettings() {
>   return WEBRTC_VIDEO_CODEC_OK;
> }
> 
>-uint32_t VP8EncoderImpl::MaxIntraTarget(uint32_t optimalBuffersize) {
>+uint32_t LibvpxVp8Encoder::MaxIntraTarget(uint32_t optimalBuffersize) {
>   // Set max to the optimal buffer level (normalized by target BR),
>   // and scaled by a scalePar.
>   // Max target size = scalePar * optimalBufferSize * targetBR[Kbps].
>@@ -702,9 +654,9 @@ uint32_t VP8EncoderImpl::MaxIntraTarget(uint32_t optimalBuffersize) {
>   return (targetPct < minIntraTh) ? minIntraTh : targetPct;
> }
> 
>-int VP8EncoderImpl::Encode(const VideoFrame& frame,
>-                           const CodecSpecificInfo* codec_specific_info,
>-                           const std::vector<FrameType>* frame_types) {
>+int LibvpxVp8Encoder::Encode(const VideoFrame& frame,
>+                             const CodecSpecificInfo* codec_specific_info,
>+                             const std::vector<FrameType>* frame_types) {
>   RTC_DCHECK_EQ(frame.width(), codec_.width);
>   RTC_DCHECK_EQ(frame.height(), codec_.height);
> 
>@@ -769,8 +721,6 @@ int VP8EncoderImpl::Encode(const VideoFrame& frame,
>   TemporalLayers::FrameConfig tl_configs[kMaxSimulcastStreams];
>   for (size_t i = 0; i < encoders_.size(); ++i) {
>     tl_configs[i] = temporal_layers_[i]->UpdateLayerConfig(frame.timestamp());
>-    RTC_DCHECK(temporal_layers_checkers_[i]->CheckTemporalConfig(
>-        send_key_frame, tl_configs[i]));
>     if (tl_configs[i].drop_frame) {
>       // Drop this frame.
>       return WEBRTC_VIDEO_CODEC_OK;
>@@ -780,7 +730,8 @@ int VP8EncoderImpl::Encode(const VideoFrame& frame,
>   if (send_key_frame) {
>     // Adapt the size of the key frame when in screenshare with 1 temporal
>     // layer.
>-    if (encoders_.size() == 1 && codec_.mode == kScreensharing &&
>+    if (encoders_.size() == 1 &&
>+        codec_.mode == VideoCodecMode::kScreensharing &&
>         codec_.VP8()->numberOfTemporalLayers <= 1) {
>       const uint32_t forceKeyFrameIntraTh = 100;
>       vpx_codec_control(&(encoders_[0]), VP8E_SET_MAX_INTRA_BITRATE_PCT,
>@@ -805,7 +756,8 @@ int VP8EncoderImpl::Encode(const VideoFrame& frame,
>     // the next update.
>     vpx_codec_enc_cfg_t temp_config;
>     memcpy(&temp_config, &configurations_[i], sizeof(vpx_codec_enc_cfg_t));
>-    if (temporal_layers_[stream_idx]->UpdateConfiguration(&temp_config)) {
>+    if (UpdateVpxConfiguration(temporal_layers_[stream_idx].get(),
>+                               &temp_config)) {
>       if (vpx_codec_enc_config_set(&encoders_[i], &temp_config))
>         return WEBRTC_VIDEO_CODEC_ERROR;
>     }
>@@ -849,7 +801,7 @@ int VP8EncoderImpl::Encode(const VideoFrame& frame,
>   return error;
> }
> 
>-void VP8EncoderImpl::PopulateCodecSpecific(
>+void LibvpxVp8Encoder::PopulateCodecSpecific(
>     CodecSpecificInfo* codec_specific,
>     const TemporalLayers::FrameConfig& tl_config,
>     const vpx_codec_cx_pkt_t& pkt,
>@@ -859,18 +811,15 @@ void VP8EncoderImpl::PopulateCodecSpecific(
>   codec_specific->codecType = kVideoCodecVP8;
>   codec_specific->codec_name = ImplementationName();
>   CodecSpecificInfoVP8* vp8Info = &(codec_specific->codecSpecific.VP8);
>-  vp8Info->pictureId = picture_id_[stream_idx];
>   vp8Info->simulcastIdx = stream_idx;
>   vp8Info->keyIdx = kNoKeyIdx;  // TODO(hlundin) populate this
>   vp8Info->nonReference = (pkt.data.frame.flags & VPX_FRAME_IS_DROPPABLE) != 0;
>   temporal_layers_[stream_idx]->PopulateCodecSpecific(
>       (pkt.data.frame.flags & VPX_FRAME_IS_KEY) != 0, tl_config, vp8Info,
>       timestamp);
>-  // Prepare next.
>-  picture_id_[stream_idx] = (picture_id_[stream_idx] + 1) & 0x7FFF;
> }
> 
>-int VP8EncoderImpl::GetEncodedPartitions(
>+int LibvpxVp8Encoder::GetEncodedPartitions(
>     const TemporalLayers::FrameConfig tl_configs[],
>     const VideoFrame& input_image) {
>   int stream_idx = static_cast<int>(encoders_.size()) - 1;
>@@ -885,6 +834,7 @@ int VP8EncoderImpl::GetEncodedPartitions(
>     // kTokenPartitions is number of bits used.
>     frag_info.VerifyAndAllocateFragmentationHeader((1 << kTokenPartitions) + 1);
>     CodecSpecificInfo codec_specific;
>+    bool is_keyframe = false;
>     const vpx_codec_cx_pkt_t* pkt = NULL;
>     while ((pkt = vpx_codec_get_cx_data(&encoders_[encoder_idx], &iter)) !=
>            NULL) {
>@@ -918,6 +868,7 @@ int VP8EncoderImpl::GetEncodedPartitions(
>         // check if encoded frame is a key frame
>         if (pkt->data.frame.flags & VPX_FRAME_IS_KEY) {
>           encoded_images_[encoder_idx]._frameType = kVideoFrameKey;
>+          is_keyframe = true;
>         }
>         PopulateCodecSpecific(&codec_specific, tl_configs[stream_idx], *pkt,
>                               stream_idx, input_image.timestamp());
>@@ -929,14 +880,15 @@ int VP8EncoderImpl::GetEncodedPartitions(
>         input_image.render_time_ms();
>     encoded_images_[encoder_idx].rotation_ = input_image.rotation();
>     encoded_images_[encoder_idx].content_type_ =
>-        (codec_.mode == kScreensharing) ? VideoContentType::SCREENSHARE
>-                                        : VideoContentType::UNSPECIFIED;
>-    encoded_images_[encoder_idx].timing_.flags = TimingFrameFlags::kInvalid;
>+        (codec_.mode == VideoCodecMode::kScreensharing)
>+            ? VideoContentType::SCREENSHARE
>+            : VideoContentType::UNSPECIFIED;
>+    encoded_images_[encoder_idx].timing_.flags = VideoSendTiming::kInvalid;
> 
>     int qp = -1;
>     vpx_codec_control(&encoders_[encoder_idx], VP8E_GET_LAST_QUANTIZER_64, &qp);
>     temporal_layers_[stream_idx]->FrameEncoded(
>-        encoded_images_[encoder_idx]._length, qp);
>+        input_image.timestamp(), encoded_images_[encoder_idx]._length, qp);
>     if (send_stream_[stream_idx]) {
>       if (encoded_images_[encoder_idx]._length > 0) {
>         TRACE_COUNTER_ID1("webrtc", "EncodedFrameSize", encoder_idx,
>@@ -951,306 +903,37 @@ int VP8EncoderImpl::GetEncodedPartitions(
>         encoded_images_[encoder_idx].qp_ = qp_128;
>         encoded_complete_callback_->OnEncodedImage(encoded_images_[encoder_idx],
>                                                    &codec_specific, &frag_info);
>-      } else if (codec_.mode == kScreensharing) {
>+      } else if (codec_.mode == VideoCodecMode::kScreensharing) {
>         result = WEBRTC_VIDEO_CODEC_TARGET_BITRATE_OVERSHOOT;
>       }
>     }
>+    if (result != WEBRTC_VIDEO_CODEC_TARGET_BITRATE_OVERSHOOT) {
>+      // Don't run checker on drop before reencode as that will incorrectly
>+      // increase the pattern index twice.
>+      RTC_DCHECK(temporal_layers_checkers_[stream_idx]->CheckTemporalConfig(
>+          is_keyframe, tl_configs[stream_idx]));
>+    }
>   }
>   return result;
> }
> 
>-VideoEncoder::ScalingSettings VP8EncoderImpl::GetScalingSettings() const {
>+VideoEncoder::ScalingSettings LibvpxVp8Encoder::GetScalingSettings() const {
>   const bool enable_scaling = encoders_.size() == 1 &&
>                               configurations_[0].rc_dropframe_thresh > 0 &&
>                               codec_.VP8().automaticResizeOn;
>-  return VideoEncoder::ScalingSettings(enable_scaling);
>+  return enable_scaling ? VideoEncoder::ScalingSettings(kLowVp8QpThreshold,
>+                                                        kHighVp8QpThreshold)
>+                        : VideoEncoder::ScalingSettings::kOff;
> }
> 
>-int VP8EncoderImpl::SetChannelParameters(uint32_t packetLoss, int64_t rtt) {
>+int LibvpxVp8Encoder::SetChannelParameters(uint32_t packetLoss, int64_t rtt) {
>   return WEBRTC_VIDEO_CODEC_OK;
> }
> 
>-int VP8EncoderImpl::RegisterEncodeCompleteCallback(
>+int LibvpxVp8Encoder::RegisterEncodeCompleteCallback(
>     EncodedImageCallback* callback) {
>   encoded_complete_callback_ = callback;
>   return WEBRTC_VIDEO_CODEC_OK;
> }
> 
>-class VP8DecoderImpl::QpSmoother {
>- public:
>-  QpSmoother() : last_sample_ms_(rtc::TimeMillis()), smoother_(kAlpha) {}
>-
>-  int GetAvg() const {
>-    float value = smoother_.filtered();
>-    return (value == rtc::ExpFilter::kValueUndefined) ? 0
>-                                                      : static_cast<int>(value);
>-  }
>-
>-  void Add(float sample) {
>-    int64_t now_ms = rtc::TimeMillis();
>-    smoother_.Apply(static_cast<float>(now_ms - last_sample_ms_), sample);
>-    last_sample_ms_ = now_ms;
>-  }
>-
>-  void Reset() { smoother_.Reset(kAlpha); }
>-
>- private:
>-  const float kAlpha = 0.95f;
>-  int64_t last_sample_ms_;
>-  rtc::ExpFilter smoother_;
>-};
>-
>-VP8DecoderImpl::VP8DecoderImpl()
>-    : use_postproc_arm_(
>-          webrtc::field_trial::IsEnabled(kVp8PostProcArmFieldTrial)),
>-      buffer_pool_(false, 300 /* max_number_of_buffers*/),
>-      decode_complete_callback_(NULL),
>-      inited_(false),
>-      decoder_(NULL),
>-      propagation_cnt_(-1),
>-      last_frame_width_(0),
>-      last_frame_height_(0),
>-      key_frame_required_(true),
>-      qp_smoother_(use_postproc_arm_ ? new QpSmoother() : nullptr) {
>-  if (use_postproc_arm_)
>-    GetPostProcParamsFromFieldTrialGroup(&deblock_);
>-}
>-
>-VP8DecoderImpl::~VP8DecoderImpl() {
>-  inited_ = true;  // in order to do the actual release
>-  Release();
>-}
>-
>-int VP8DecoderImpl::InitDecode(const VideoCodec* inst, int number_of_cores) {
>-  int ret_val = Release();
>-  if (ret_val < 0) {
>-    return ret_val;
>-  }
>-  if (decoder_ == NULL) {
>-    decoder_ = new vpx_codec_ctx_t;
>-    memset(decoder_, 0, sizeof(*decoder_));
>-  }
>-  vpx_codec_dec_cfg_t cfg;
>-  // Setting number of threads to a constant value (1)
>-  cfg.threads = 1;
>-  cfg.h = cfg.w = 0;  // set after decode
>-
>-#if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64) \
>-  || defined(WEBRTC_ANDROID)
>-  vpx_codec_flags_t flags = use_postproc_arm_ ? VPX_CODEC_USE_POSTPROC : 0;
>-#else
>-  vpx_codec_flags_t flags = VPX_CODEC_USE_POSTPROC;
>-#endif
>-
>-  if (vpx_codec_dec_init(decoder_, vpx_codec_vp8_dx(), &cfg, flags)) {
>-    delete decoder_;
>-    decoder_ = nullptr;
>-    return WEBRTC_VIDEO_CODEC_MEMORY;
>-  }
>-
>-  propagation_cnt_ = -1;
>-  inited_ = true;
>-
>-  // Always start with a complete key frame.
>-  key_frame_required_ = true;
>-  return WEBRTC_VIDEO_CODEC_OK;
>-}
>-
>-int VP8DecoderImpl::Decode(const EncodedImage& input_image,
>-                           bool missing_frames,
>-                           const RTPFragmentationHeader* fragmentation,
>-                           const CodecSpecificInfo* codec_specific_info,
>-                           int64_t /*render_time_ms*/) {
>-  if (!inited_) {
>-    return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
>-  }
>-  if (decode_complete_callback_ == NULL) {
>-    return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
>-  }
>-  if (input_image._buffer == NULL && input_image._length > 0) {
>-    // Reset to avoid requesting key frames too often.
>-    if (propagation_cnt_ > 0)
>-      propagation_cnt_ = 0;
>-    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
>-  }
>-
>-// Post process configurations.
>-#if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64) \
>-  || defined(WEBRTC_ANDROID)
>-  if (use_postproc_arm_) {
>-    vp8_postproc_cfg_t ppcfg;
>-    ppcfg.post_proc_flag = VP8_MFQE;
>-    // For low resolutions, use stronger deblocking filter.
>-    int last_width_x_height = last_frame_width_ * last_frame_height_;
>-    if (last_width_x_height > 0 && last_width_x_height <= 320 * 240) {
>-      // Enable the deblock and demacroblocker based on qp thresholds.
>-      RTC_DCHECK(qp_smoother_);
>-      int qp = qp_smoother_->GetAvg();
>-      if (qp > deblock_.min_qp) {
>-        int level = deblock_.max_level;
>-        if (qp < deblock_.degrade_qp) {
>-          // Use lower level.
>-          level = deblock_.max_level * (qp - deblock_.min_qp) /
>-                  (deblock_.degrade_qp - deblock_.min_qp);
>-        }
>-        // Deblocking level only affects VP8_DEMACROBLOCK.
>-        ppcfg.deblocking_level = std::max(level, 1);
>-        ppcfg.post_proc_flag |= VP8_DEBLOCK | VP8_DEMACROBLOCK;
>-      }
>-    }
>-    vpx_codec_control(decoder_, VP8_SET_POSTPROC, &ppcfg);
>-  }
>-#else
>-  vp8_postproc_cfg_t ppcfg;
>-  // MFQE enabled to reduce key frame popping.
>-  ppcfg.post_proc_flag = VP8_MFQE | VP8_DEBLOCK;
>-  // For VGA resolutions and lower, enable the demacroblocker postproc.
>-  if (last_frame_width_ * last_frame_height_ <= 640 * 360) {
>-    ppcfg.post_proc_flag |= VP8_DEMACROBLOCK;
>-  }
>-  // Strength of deblocking filter. Valid range:[0,16]
>-  ppcfg.deblocking_level = 3;
>-  vpx_codec_control(decoder_, VP8_SET_POSTPROC, &ppcfg);
>-#endif
>-
>-  // Always start with a complete key frame.
>-  if (key_frame_required_) {
>-    if (input_image._frameType != kVideoFrameKey)
>-      return WEBRTC_VIDEO_CODEC_ERROR;
>-    // We have a key frame - is it complete?
>-    if (input_image._completeFrame) {
>-      key_frame_required_ = false;
>-    } else {
>-      return WEBRTC_VIDEO_CODEC_ERROR;
>-    }
>-  }
>-  // Restrict error propagation using key frame requests.
>-  // Reset on a key frame refresh.
>-  if (input_image._frameType == kVideoFrameKey && input_image._completeFrame) {
>-    propagation_cnt_ = -1;
>-    // Start count on first loss.
>-  } else if ((!input_image._completeFrame || missing_frames) &&
>-             propagation_cnt_ == -1) {
>-    propagation_cnt_ = 0;
>-  }
>-  if (propagation_cnt_ >= 0) {
>-    propagation_cnt_++;
>-  }
>-
>-  vpx_codec_iter_t iter = NULL;
>-  vpx_image_t* img;
>-  int ret;
>-
>-  // Check for missing frames.
>-  if (missing_frames) {
>-    // Call decoder with zero data length to signal missing frames.
>-    if (vpx_codec_decode(decoder_, NULL, 0, 0, VPX_DL_REALTIME)) {
>-      // Reset to avoid requesting key frames too often.
>-      if (propagation_cnt_ > 0)
>-        propagation_cnt_ = 0;
>-      return WEBRTC_VIDEO_CODEC_ERROR;
>-    }
>-    img = vpx_codec_get_frame(decoder_, &iter);
>-    iter = NULL;
>-  }
>-
>-  uint8_t* buffer = input_image._buffer;
>-  if (input_image._length == 0) {
>-    buffer = NULL;  // Triggers full frame concealment.
>-  }
>-  if (vpx_codec_decode(decoder_, buffer, input_image._length, 0,
>-                       VPX_DL_REALTIME)) {
>-    // Reset to avoid requesting key frames too often.
>-    if (propagation_cnt_ > 0) {
>-      propagation_cnt_ = 0;
>-    }
>-    return WEBRTC_VIDEO_CODEC_ERROR;
>-  }
>-
>-  img = vpx_codec_get_frame(decoder_, &iter);
>-  int qp;
>-  vpx_codec_err_t vpx_ret =
>-      vpx_codec_control(decoder_, VPXD_GET_LAST_QUANTIZER, &qp);
>-  RTC_DCHECK_EQ(vpx_ret, VPX_CODEC_OK);
>-  ret = ReturnFrame(img, input_image._timeStamp, input_image.ntp_time_ms_, qp);
>-  if (ret != 0) {
>-    // Reset to avoid requesting key frames too often.
>-    if (ret < 0 && propagation_cnt_ > 0)
>-      propagation_cnt_ = 0;
>-    return ret;
>-  }
>-  // Check Vs. threshold
>-  if (propagation_cnt_ > kVp8ErrorPropagationTh) {
>-    // Reset to avoid requesting key frames too often.
>-    propagation_cnt_ = 0;
>-    return WEBRTC_VIDEO_CODEC_ERROR;
>-  }
>-  return WEBRTC_VIDEO_CODEC_OK;
>-}
>-
>-int VP8DecoderImpl::ReturnFrame(const vpx_image_t* img,
>-                                uint32_t timestamp,
>-                                int64_t ntp_time_ms,
>-                                int qp) {
>-  if (img == NULL) {
>-    // Decoder OK and NULL image => No show frame
>-    return WEBRTC_VIDEO_CODEC_NO_OUTPUT;
>-  }
>-  if (qp_smoother_) {
>-    if (last_frame_width_ != static_cast<int>(img->d_w) ||
>-        last_frame_height_ != static_cast<int>(img->d_h)) {
>-      qp_smoother_->Reset();
>-    }
>-    qp_smoother_->Add(qp);
>-  }
>-  last_frame_width_ = img->d_w;
>-  last_frame_height_ = img->d_h;
>-  // Allocate memory for decoded image.
>-  rtc::scoped_refptr<I420Buffer> buffer =
>-      buffer_pool_.CreateBuffer(img->d_w, img->d_h);
>-  if (!buffer.get()) {
>-    // Pool has too many pending frames.
>-    RTC_HISTOGRAM_BOOLEAN("WebRTC.Video.VP8DecoderImpl.TooManyPendingFrames",
>-                          1);
>-    return WEBRTC_VIDEO_CODEC_NO_OUTPUT;
>-  }
>-
>-  libyuv::I420Copy(img->planes[VPX_PLANE_Y], img->stride[VPX_PLANE_Y],
>-                   img->planes[VPX_PLANE_U], img->stride[VPX_PLANE_U],
>-                   img->planes[VPX_PLANE_V], img->stride[VPX_PLANE_V],
>-                   buffer->MutableDataY(), buffer->StrideY(),
>-                   buffer->MutableDataU(), buffer->StrideU(),
>-                   buffer->MutableDataV(), buffer->StrideV(), img->d_w,
>-                   img->d_h);
>-
>-  VideoFrame decoded_image(buffer, timestamp, 0, kVideoRotation_0);
>-  decoded_image.set_ntp_time_ms(ntp_time_ms);
>-  decode_complete_callback_->Decoded(decoded_image, rtc::nullopt, qp);
>-
>-  return WEBRTC_VIDEO_CODEC_OK;
>-}
>-
>-int VP8DecoderImpl::RegisterDecodeCompleteCallback(
>-    DecodedImageCallback* callback) {
>-  decode_complete_callback_ = callback;
>-  return WEBRTC_VIDEO_CODEC_OK;
>-}
>-
>-int VP8DecoderImpl::Release() {
>-  if (decoder_ != NULL) {
>-    if (vpx_codec_destroy(decoder_)) {
>-      return WEBRTC_VIDEO_CODEC_MEMORY;
>-    }
>-    delete decoder_;
>-    decoder_ = NULL;
>-  }
>-  buffer_pool_.Release();
>-  inited_ = false;
>-  return WEBRTC_VIDEO_CODEC_OK;
>-}
>-
>-const char* VP8DecoderImpl::ImplementationName() const {
>-  return "libvpx";
>-}
>-
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/vp8_impl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.h
>similarity index 61%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/vp8_impl.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.h
>index e5d5d81e35c..4e9d374c4cc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/vp8_impl.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/libvpx_vp8_encoder.h
>@@ -1,45 +1,36 @@
> /*
>- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>  *
>  *  Use of this source code is governed by a BSD-style license
>  *  that can be found in the LICENSE file in the root of the source
>  *  tree. An additional intellectual property rights grant can be found
>  *  in the file PATENTS.  All contributing project authors may
>  *  be found in the AUTHORS file in the root of the source tree.
>- *
>- * WEBRTC VP8 wrapper interface
>  */
> 
>-#ifndef MODULES_VIDEO_CODING_CODECS_VP8_VP8_IMPL_H_
>-#define MODULES_VIDEO_CODING_CODECS_VP8_VP8_IMPL_H_
>+#ifndef MODULES_VIDEO_CODING_CODECS_VP8_LIBVPX_VP8_ENCODER_H_
>+#define MODULES_VIDEO_CODING_CODECS_VP8_LIBVPX_VP8_ENCODER_H_
> 
> #include <memory>
> #include <vector>
> 
>-// NOTE: This include order must remain to avoid compile errors, even though
>-//       it breaks the style guide.
>-#include "vpx/vp8cx.h"
>-#include "vpx/vp8dx.h"
>-#include "vpx/vpx_decoder.h"
>-#include "vpx/vpx_encoder.h"
>-
> #include "api/video/video_frame.h"
>-#include "common_video/include/i420_buffer_pool.h"
>+#include "api/video_codecs/video_encoder.h"
>+#include "common_types.h"  // NOLINT(build/include)
> #include "common_video/include/video_frame.h"
> #include "modules/video_coding/codecs/vp8/include/vp8.h"
> #include "modules/video_coding/codecs/vp8/temporal_layers.h"
> #include "modules/video_coding/include/video_codec_interface.h"
>-#include "modules/video_coding/utility/quality_scaler.h"
> 
>-namespace webrtc {
>+#include "vpx/vp8cx.h"
>+#include "vpx/vpx_encoder.h"
> 
>-class TemporalLayers;
>+namespace webrtc {
> 
>-class VP8EncoderImpl : public VP8Encoder {
>+class LibvpxVp8Encoder : public VP8Encoder {
>  public:
>-  VP8EncoderImpl();
>-
>-  virtual ~VP8EncoderImpl();
>+  LibvpxVp8Encoder();
>+  ~LibvpxVp8Encoder() override;
> 
>   int Release() override;
> 
>@@ -55,7 +46,7 @@ class VP8EncoderImpl : public VP8Encoder {
> 
>   int SetChannelParameters(uint32_t packet_loss, int64_t rtt) override;
> 
>-  int SetRateAllocation(const BitrateAllocation& bitrate,
>+  int SetRateAllocation(const VideoBitrateAllocation& bitrate,
>                         uint32_t new_framerate) override;
> 
>   ScalingSettings GetScalingSettings() const override;
>@@ -105,8 +96,6 @@ class VP8EncoderImpl : public VP8Encoder {
>   uint32_t rc_max_intra_target_;
>   std::vector<std::unique_ptr<TemporalLayers>> temporal_layers_;
>   std::vector<std::unique_ptr<TemporalLayersChecker>> temporal_layers_checkers_;
>-  std::vector<uint16_t> picture_id_;
>-  std::vector<uint8_t> tl0_pic_idx_;
>   std::vector<bool> key_frame_request_;
>   std::vector<bool> send_stream_;
>   std::vector<int> cpu_speed_;
>@@ -117,51 +106,6 @@ class VP8EncoderImpl : public VP8Encoder {
>   std::vector<vpx_rational_t> downsampling_factors_;
> };
> 
>-class VP8DecoderImpl : public VP8Decoder {
>- public:
>-  VP8DecoderImpl();
>-
>-  virtual ~VP8DecoderImpl();
>-
>-  int InitDecode(const VideoCodec* inst, int number_of_cores) override;
>-
>-  int Decode(const EncodedImage& input_image,
>-             bool missing_frames,
>-             const RTPFragmentationHeader* fragmentation,
>-             const CodecSpecificInfo* codec_specific_info,
>-             int64_t /*render_time_ms*/) override;
>-
>-  int RegisterDecodeCompleteCallback(DecodedImageCallback* callback) override;
>-  int Release() override;
>-
>-  const char* ImplementationName() const override;
>-
>-  struct DeblockParams {
>-    int max_level = 6;   // Deblocking strength: [0, 16].
>-    int degrade_qp = 1;  // If QP value is below, start lowering |max_level|.
>-    int min_qp = 0;      // If QP value is below, turn off deblocking.
>-  };
>-
>- private:
>-  class QpSmoother;
>-  int ReturnFrame(const vpx_image_t* img,
>-                  uint32_t timeStamp,
>-                  int64_t ntp_time_ms,
>-                  int qp);
>-
>-  const bool use_postproc_arm_;
>-
>-  I420BufferPool buffer_pool_;
>-  DecodedImageCallback* decode_complete_callback_;
>-  bool inited_;
>-  vpx_codec_ctx_t* decoder_;
>-  int propagation_cnt_;
>-  int last_frame_width_;
>-  int last_frame_height_;
>-  bool key_frame_required_;
>-  DeblockParams deblock_;
>-  const std::unique_ptr<QpSmoother> qp_smoother_;
>-};
> }  // namespace webrtc
> 
>-#endif  // MODULES_VIDEO_CODING_CODECS_VP8_VP8_IMPL_H_
>+#endif  // MODULES_VIDEO_CODING_CODECS_VP8_LIBVPX_VP8_ENCODER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/libvpx_vp8_simulcast_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/libvpx_vp8_simulcast_test.cc
>new file mode 100644
>index 00000000000..5f974d643bb
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/libvpx_vp8_simulcast_test.cc
>@@ -0,0 +1,109 @@
>+/*
>+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <memory>
>+
>+#include "absl/memory/memory.h"
>+#include "api/test/create_simulcast_test_fixture.h"
>+#include "api/test/simulcast_test_fixture.h"
>+#include "modules/video_coding/codecs/vp8/include/vp8.h"
>+#include "test/function_video_decoder_factory.h"
>+#include "test/function_video_encoder_factory.h"
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+namespace {
>+std::unique_ptr<SimulcastTestFixture> CreateSpecificSimulcastTestFixture() {
>+  std::unique_ptr<VideoEncoderFactory> encoder_factory =
>+      absl::make_unique<FunctionVideoEncoderFactory>(
>+          []() { return VP8Encoder::Create(); });
>+  std::unique_ptr<VideoDecoderFactory> decoder_factory =
>+      absl::make_unique<FunctionVideoDecoderFactory>(
>+          []() { return VP8Decoder::Create(); });
>+  return CreateSimulcastTestFixture(std::move(encoder_factory),
>+                                    std::move(decoder_factory),
>+                                    SdpVideoFormat("VP8"));
>+}
>+}  // namespace
>+
>+TEST(LibvpxVp8SimulcastTest, TestKeyFrameRequestsOnAllStreams) {
>+  auto fixture = CreateSpecificSimulcastTestFixture();
>+  fixture->TestKeyFrameRequestsOnAllStreams();
>+}
>+
>+TEST(LibvpxVp8SimulcastTest, TestPaddingAllStreams) {
>+  auto fixture = CreateSpecificSimulcastTestFixture();
>+  fixture->TestPaddingAllStreams();
>+}
>+
>+TEST(LibvpxVp8SimulcastTest, TestPaddingTwoStreams) {
>+  auto fixture = CreateSpecificSimulcastTestFixture();
>+  fixture->TestPaddingTwoStreams();
>+}
>+
>+TEST(LibvpxVp8SimulcastTest, TestPaddingTwoStreamsOneMaxedOut) {
>+  auto fixture = CreateSpecificSimulcastTestFixture();
>+  fixture->TestPaddingTwoStreamsOneMaxedOut();
>+}
>+
>+TEST(LibvpxVp8SimulcastTest, TestPaddingOneStream) {
>+  auto fixture = CreateSpecificSimulcastTestFixture();
>+  fixture->TestPaddingOneStream();
>+}
>+
>+TEST(LibvpxVp8SimulcastTest, TestPaddingOneStreamTwoMaxedOut) {
>+  auto fixture = CreateSpecificSimulcastTestFixture();
>+  fixture->TestPaddingOneStreamTwoMaxedOut();
>+}
>+
>+TEST(LibvpxVp8SimulcastTest, TestSendAllStreams) {
>+  auto fixture = CreateSpecificSimulcastTestFixture();
>+  fixture->TestSendAllStreams();
>+}
>+
>+TEST(LibvpxVp8SimulcastTest, TestDisablingStreams) {
>+  auto fixture = CreateSpecificSimulcastTestFixture();
>+  fixture->TestDisablingStreams();
>+}
>+
>+TEST(LibvpxVp8SimulcastTest, TestActiveStreams) {
>+  auto fixture = CreateSpecificSimulcastTestFixture();
>+  fixture->TestActiveStreams();
>+}
>+
>+TEST(LibvpxVp8SimulcastTest, TestSwitchingToOneStream) {
>+  auto fixture = CreateSpecificSimulcastTestFixture();
>+  fixture->TestSwitchingToOneStream();
>+}
>+
>+TEST(LibvpxVp8SimulcastTest, TestSwitchingToOneOddStream) {
>+  auto fixture = CreateSpecificSimulcastTestFixture();
>+  fixture->TestSwitchingToOneOddStream();
>+}
>+
>+TEST(LibvpxVp8SimulcastTest, TestSwitchingToOneSmallStream) {
>+  auto fixture = CreateSpecificSimulcastTestFixture();
>+  fixture->TestSwitchingToOneSmallStream();
>+}
>+
>+TEST(LibvpxVp8SimulcastTest, TestSpatioTemporalLayers333PatternEncoder) {
>+  auto fixture = CreateSpecificSimulcastTestFixture();
>+  fixture->TestSpatioTemporalLayers333PatternEncoder();
>+}
>+
>+TEST(LibvpxVp8SimulcastTest, TestStrideEncodeDecode) {
>+  auto fixture = CreateSpecificSimulcastTestFixture();
>+  fixture->TestStrideEncodeDecode();
>+}
>+
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/screenshare_layers.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/screenshare_layers.cc
>index 8c63cf1a080..ebee39bd3f0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/screenshare_layers.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/screenshare_layers.cc
>@@ -19,8 +19,6 @@
> #include "rtc_base/logging.h"
> #include "system_wrappers/include/clock.h"
> #include "system_wrappers/include/metrics.h"
>-#include "vpx/vp8cx.h"
>-#include "vpx/vpx_encoder.h"
> 
> namespace webrtc {
> 
>@@ -39,48 +37,11 @@ constexpr int ScreenshareLayers::kMaxNumTemporalLayers;
> // been exceeded. This prevents needless keyframe requests.
> const int ScreenshareLayers::kMaxFrameIntervalMs = 2750;
> 
>-webrtc::TemporalLayers* ScreenshareTemporalLayersFactory::Create(
>-    int simulcast_id,
>-    int num_temporal_layers,
>-    uint8_t initial_tl0_pic_idx) const {
>-  webrtc::TemporalLayers* tl;
>-  if (simulcast_id == 0) {
>-    tl = new webrtc::ScreenshareLayers(num_temporal_layers, rand(),
>-                                       webrtc::Clock::GetRealTimeClock());
>-  } else {
>-    TemporalLayersFactory rt_tl_factory;
>-    tl = rt_tl_factory.Create(simulcast_id, num_temporal_layers, rand());
>-  }
>-  if (listener_)
>-    listener_->OnTemporalLayersCreated(simulcast_id, tl);
>-  return tl;
>-}
>-
>-std::unique_ptr<webrtc::TemporalLayersChecker>
>-ScreenshareTemporalLayersFactory::CreateChecker(
>-    int simulcast_id,
>-    int temporal_layers,
>-    uint8_t initial_tl0_pic_idx) const {
>-  webrtc::TemporalLayersChecker* tlc;
>-  if (simulcast_id == 0) {
>-    tlc =
>-        new webrtc::TemporalLayersChecker(temporal_layers, initial_tl0_pic_idx);
>-  } else {
>-    TemporalLayersFactory rt_tl_factory;
>-    return rt_tl_factory.CreateChecker(simulcast_id, temporal_layers,
>-                                       initial_tl0_pic_idx);
>-  }
>-  return std::unique_ptr<webrtc::TemporalLayersChecker>(tlc);
>-}
>-
> ScreenshareLayers::ScreenshareLayers(int num_temporal_layers,
>-                                     uint8_t initial_tl0_pic_idx,
>                                      Clock* clock)
>     : clock_(clock),
>       number_of_temporal_layers_(
>           std::min(kMaxNumTemporalLayers, num_temporal_layers)),
>-      last_base_layer_sync_(false),
>-      tl0_pic_idx_(initial_tl0_pic_idx),
>       active_layer_(-1),
>       last_timestamp_(-1),
>       last_sync_timestamp_(-1),
>@@ -99,10 +60,6 @@ ScreenshareLayers::~ScreenshareLayers() {
>   UpdateHistograms();
> }
> 
>-uint8_t ScreenshareLayers::Tl0PicIdx() const {
>-  return tl0_pic_idx_;
>-}
>-
> TemporalLayers::FrameConfig ScreenshareLayers::UpdateLayerConfig(
>     uint32_t timestamp) {
>   if (number_of_temporal_layers_ <= 1) {
>@@ -189,7 +146,8 @@ TemporalLayers::FrameConfig ScreenshareLayers::UpdateLayerConfig(
>       break;
>     case 1:
>       if (layers_[1].state != TemporalLayer::State::kDropped) {
>-        if (TimeToSync(unwrapped_timestamp)) {
>+        if (TimeToSync(unwrapped_timestamp) ||
>+            layers_[1].state == TemporalLayer::State::kKeyFrame) {
>           last_sync_timestamp_ = unwrapped_timestamp;
>           layer_state = TemporalLayerState::kTl1Sync;
>         } else {
>@@ -241,40 +199,47 @@ TemporalLayers::FrameConfig ScreenshareLayers::UpdateLayerConfig(
>   return tl_config;
> }
> 
>-std::vector<uint32_t> ScreenshareLayers::OnRatesUpdated(int bitrate_kbps,
>-                                                        int max_bitrate_kbps,
>-                                                        int framerate) {
>-  RTC_DCHECK_GT(framerate, 0);
>+void ScreenshareLayers::OnRatesUpdated(
>+    const std::vector<uint32_t>& bitrates_bps,
>+    int framerate_fps) {
>+  RTC_DCHECK_GT(framerate_fps, 0);
>+  RTC_DCHECK_GE(bitrates_bps.size(), 1);
>+  RTC_DCHECK_LE(bitrates_bps.size(), 2);
>+
>+  // |bitrates_bps| uses individual rates per layer, but we want to use the
>+  // accumulated rate here.
>+  uint32_t tl0_kbps = bitrates_bps[0] / 1000;
>+  uint32_t tl1_kbps = tl0_kbps;
>+  if (bitrates_bps.size() > 1) {
>+    tl1_kbps += bitrates_bps[1] / 1000;
>+  }
>+
>   if (!target_framerate_) {
>-    // First OnRatesUpdated() is called during construction, with the configured
>-    // targets as parameters.
>-    target_framerate_.emplace(framerate);
>+    // First OnRatesUpdated() is called during construction, with the
>+    // configured targets as parameters.
>+    target_framerate_ = framerate_fps;
>     capture_framerate_ = target_framerate_;
>     bitrate_updated_ = true;
>   } else {
>-    bitrate_updated_ =
>-        bitrate_kbps != static_cast<int>(layers_[0].target_rate_kbps_) ||
>-        max_bitrate_kbps != static_cast<int>(layers_[1].target_rate_kbps_) ||
>-        (capture_framerate_ &&
>-         framerate != static_cast<int>(*capture_framerate_));
>-    if (framerate < 0) {
>+    if ((capture_framerate_ &&
>+         framerate_fps != static_cast<int>(*capture_framerate_)) ||
>+        (tl0_kbps != layers_[0].target_rate_kbps_) ||
>+        (tl1_kbps != layers_[1].target_rate_kbps_)) {
>+      bitrate_updated_ = true;
>+    }
>+
>+    if (framerate_fps < 0) {
>       capture_framerate_.reset();
>     } else {
>-      capture_framerate_.emplace(framerate);
>+      capture_framerate_ = framerate_fps;
>     }
>   }
> 
>-  layers_[0].target_rate_kbps_ = bitrate_kbps;
>-  layers_[1].target_rate_kbps_ = max_bitrate_kbps;
>-
>-  std::vector<uint32_t> allocation;
>-  allocation.push_back(bitrate_kbps);
>-  if (max_bitrate_kbps > bitrate_kbps)
>-    allocation.push_back(max_bitrate_kbps - bitrate_kbps);
>-  return allocation;
>+  layers_[0].target_rate_kbps_ = tl0_kbps;
>+  layers_[1].target_rate_kbps_ = tl1_kbps;
> }
> 
>-void ScreenshareLayers::FrameEncoded(unsigned int size, int qp) {
>+void ScreenshareLayers::FrameEncoded(uint32_t timestamp, size_t size, int qp) {
>   if (size > 0)
>     encode_framerate_.Update(1, clock_->TimeInMilliseconds());
> 
>@@ -317,7 +282,6 @@ void ScreenshareLayers::PopulateCodecSpecific(
>   if (number_of_temporal_layers_ == 1) {
>     vp8_info->temporalIdx = kNoTemporalIdx;
>     vp8_info->layerSync = false;
>-    vp8_info->tl0PicIdx = kNoTl0PicIdx;
>   } else {
>     int64_t unwrapped_timestamp = time_wrap_handler_.Unwrap(timestamp);
>     vp8_info->temporalIdx = tl_config.packetizer_temporal_idx;
>@@ -326,17 +290,9 @@ void ScreenshareLayers::PopulateCodecSpecific(
>       vp8_info->temporalIdx = 0;
>       last_sync_timestamp_ = unwrapped_timestamp;
>       vp8_info->layerSync = true;
>-    } else if (last_base_layer_sync_ && vp8_info->temporalIdx != 0) {
>-      // Regardless of pattern the frame after a base layer sync will always
>-      // be a layer sync.
>-      last_sync_timestamp_ = unwrapped_timestamp;
>-      vp8_info->layerSync = true;
>-    }
>-    if (vp8_info->temporalIdx == 0) {
>-      tl0_pic_idx_++;
>+      layers_[0].state = TemporalLayer::State::kKeyFrame;
>+      layers_[1].state = TemporalLayer::State::kKeyFrame;
>     }
>-    last_base_layer_sync_ = frame_is_keyframe;
>-    vp8_info->tl0PicIdx = tl0_pic_idx_;
>   }
> }
> 
>@@ -381,7 +337,7 @@ uint32_t ScreenshareLayers::GetCodecTargetBitrateKbps() const {
>   return std::max(layers_[0].target_rate_kbps_, target_bitrate_kbps);
> }
> 
>-bool ScreenshareLayers::UpdateConfiguration(vpx_codec_enc_cfg_t* cfg) {
>+bool ScreenshareLayers::UpdateConfiguration(Vp8EncoderConfig* cfg) {
>   bool cfg_updated = false;
>   uint32_t target_bitrate_kbps = GetCodecTargetBitrateKbps();
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/screenshare_layers.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/screenshare_layers.h
>index 71433514c4f..873d5259388 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/screenshare_layers.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/screenshare_layers.h
>@@ -15,7 +15,6 @@
> #include "modules/video_coding/utility/frame_dropper.h"
> #include "rtc_base/rate_statistics.h"
> #include "rtc_base/timeutils.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -29,32 +28,28 @@ class ScreenshareLayers : public TemporalLayers {
>   static const int kMaxFrameIntervalMs;
> 
>   ScreenshareLayers(int num_temporal_layers,
>-                    uint8_t initial_tl0_pic_idx,
>                     Clock* clock);
>   virtual ~ScreenshareLayers();
> 
>   // Returns the recommended VP8 encode flags needed. May refresh the decoder
>   // and/or update the reference buffers.
>-  TemporalLayers::FrameConfig UpdateLayerConfig(uint32_t timestamp) override;
>+  TemporalLayers::FrameConfig UpdateLayerConfig(
>+      uint32_t rtp_timestamp) override;
> 
>-  // Update state based on new bitrate target and incoming framerate.
>-  // Returns the bitrate allocation for the active temporal layers.
>-  std::vector<uint32_t> OnRatesUpdated(int bitrate_kbps,
>-                                       int max_bitrate_kbps,
>-                                       int framerate) override;
>+  // New target bitrate, per temporal layer.
>+  void OnRatesUpdated(const std::vector<uint32_t>& bitrates_bps,
>+                      int framerate_fps) override;
> 
>   // Update the encoder configuration with target bitrates or other parameters.
>   // Returns true iff the configuration was actually modified.
>-  bool UpdateConfiguration(vpx_codec_enc_cfg_t* cfg) override;
>+  bool UpdateConfiguration(Vp8EncoderConfig* cfg) override;
> 
>   void PopulateCodecSpecific(bool base_layer_sync,
>                              const TemporalLayers::FrameConfig& tl_config,
>                              CodecSpecificInfoVP8* vp8_info,
>-                             uint32_t timestamp) override;
>+                             uint32_t rtp_timestamp) override;
> 
>-  void FrameEncoded(unsigned int size, int qp) override;
>-
>-  uint8_t Tl0PicIdx() const override;
>+  void FrameEncoded(uint32_t rtp_timestamp, size_t size, int qp) override;
> 
>  private:
>   enum class TemporalLayerState : int { kDrop, kTl0, kTl1, kTl1Sync };
>@@ -65,8 +60,6 @@ class ScreenshareLayers : public TemporalLayers {
>   Clock* const clock_;
> 
>   int number_of_temporal_layers_;
>-  bool last_base_layer_sync_;
>-  uint8_t tl0_pic_idx_;
>   int active_layer_;
>   int64_t last_timestamp_;
>   int64_t last_sync_timestamp_;
>@@ -78,9 +71,9 @@ class ScreenshareLayers : public TemporalLayers {
>   uint32_t max_debt_bytes_;
> 
>   // Configured max framerate.
>-  rtc::Optional<uint32_t> target_framerate_;
>+  absl::optional<uint32_t> target_framerate_;
>   // Incoming framerate from capturer.
>-  rtc::Optional<uint32_t> capture_framerate_;
>+  absl::optional<uint32_t> capture_framerate_;
>   // Tracks what framerate we actually encode, and drops frames on overshoot.
>   RateStatistics encode_framerate_;
>   bool bitrate_updated_;
>@@ -99,6 +92,7 @@ class ScreenshareLayers : public TemporalLayers {
>       kDropped,
>       kReencoded,
>       kQualityBoost,
>+      kKeyFrame
>     } state;
> 
>     int enhanced_max_qp;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/screenshare_layers_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/screenshare_layers_unittest.cc
>index 818308f3b01..547fc62cdb1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/screenshare_layers_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/screenshare_layers_unittest.cc
>@@ -11,10 +11,8 @@
> #include <memory>
> #include <vector>
> 
>-#include "vpx/vp8cx.h"
>-#include "vpx/vpx_encoder.h"
>+#include "modules/video_coding/codecs/vp8/libvpx_vp8_encoder.h"
> #include "modules/video_coding/codecs/vp8/screenshare_layers.h"
>-#include "modules/video_coding/codecs/vp8/vp8_impl.h"
> #include "modules/video_coding/include/video_codec_interface.h"
> #include "modules/video_coding/utility/mock/mock_frame_dropper.h"
> #include "system_wrappers/include/clock.h"
>@@ -28,7 +26,7 @@ using ::testing::NiceMock;
> using ::testing::Return;
> 
> namespace webrtc {
>-
>+namespace {
> // 5 frames per second at 90 kHz.
> const uint32_t kTimestampDelta5Fps = 90000 / 5;
> const int kDefaultQp = 54;
>@@ -45,6 +43,11 @@ const int kTl1Flags =
>     VP8_EFLAG_NO_REF_ARF | VP8_EFLAG_NO_UPD_ARF | VP8_EFLAG_NO_UPD_LAST;
> const int kTl1SyncFlags = VP8_EFLAG_NO_REF_ARF | VP8_EFLAG_NO_REF_GF |
>                           VP8_EFLAG_NO_UPD_ARF | VP8_EFLAG_NO_UPD_LAST;
>+const std::vector<uint32_t> kDefault2TlBitratesBps = {
>+    kDefaultTl0BitrateKbps * 1000,
>+    (kDefaultTl1BitrateKbps - kDefaultTl0BitrateKbps) * 1000};
>+
>+}  // namespace
> 
> class ScreenshareLayerTest : public ::testing::Test {
>  protected:
>@@ -58,14 +61,14 @@ class ScreenshareLayerTest : public ::testing::Test {
>   virtual ~ScreenshareLayerTest() {}
> 
>   void SetUp() override {
>-    layers_.reset(new ScreenshareLayers(2, 0, &clock_));
>+    layers_.reset(new ScreenshareLayers(2, &clock_));
>     cfg_ = ConfigureBitrates();
>   }
> 
>   int EncodeFrame(bool base_sync) {
>     int flags = ConfigureFrame(base_sync);
>     if (flags != -1)
>-      layers_->FrameEncoded(frame_size_, kDefaultQp);
>+      layers_->FrameEncoded(timestamp_, frame_size_, kDefaultQp);
>     return flags;
>   }
> 
>@@ -79,7 +82,7 @@ class ScreenshareLayerTest : public ::testing::Test {
>       return -1;
>     }
>     config_updated_ = layers_->UpdateConfiguration(&cfg_);
>-    int flags = VP8EncoderImpl::EncodeFlags(tl_config_);
>+    int flags = LibvpxVp8Encoder::EncodeFlags(tl_config_);
>     layers_->PopulateCodecSpecific(key_frame, tl_config_, &vp8_info_,
>                                    timestamp_);
>     EXPECT_NE(-1, frame_size_);
>@@ -96,18 +99,15 @@ class ScreenshareLayerTest : public ::testing::Test {
>     return ((bitrate_kbps * 1000) / 8) / kFrameRate;
>   }
> 
>-  vpx_codec_enc_cfg_t ConfigureBitrates() {
>-    vpx_codec_enc_cfg_t vpx_cfg;
>-    memset(&vpx_cfg, 0, sizeof(vpx_codec_enc_cfg_t));
>-    vpx_cfg.rc_min_quantizer = min_qp_;
>-    vpx_cfg.rc_max_quantizer = max_qp_;
>-    EXPECT_THAT(layers_->OnRatesUpdated(kDefaultTl0BitrateKbps,
>-                                        kDefaultTl1BitrateKbps, kFrameRate),
>-                ElementsAre(kDefaultTl0BitrateKbps,
>-                            kDefaultTl1BitrateKbps - kDefaultTl0BitrateKbps));
>-    EXPECT_TRUE(layers_->UpdateConfiguration(&vpx_cfg));
>-    frame_size_ = FrameSizeForBitrate(vpx_cfg.rc_target_bitrate);
>-    return vpx_cfg;
>+  Vp8EncoderConfig ConfigureBitrates() {
>+    Vp8EncoderConfig vp8_cfg;
>+    memset(&vp8_cfg, 0, sizeof(Vp8EncoderConfig));
>+    vp8_cfg.rc_min_quantizer = min_qp_;
>+    vp8_cfg.rc_max_quantizer = max_qp_;
>+    layers_->OnRatesUpdated(kDefault2TlBitratesBps, kFrameRate);
>+    EXPECT_TRUE(layers_->UpdateConfiguration(&vp8_cfg));
>+    frame_size_ = FrameSizeForBitrate(vp8_cfg.rc_target_bitrate);
>+    return vp8_cfg;
>   }
> 
>   void WithQpLimits(int min_qp, int max_qp) {
>@@ -138,11 +138,11 @@ class ScreenshareLayerTest : public ::testing::Test {
>   // FrameEncoded() call will be omitted and needs to be done by the caller.
>   // Returns the flags for the last frame.
>   int SkipUntilTl(int layer) {
>-    return SkipUntilTlAndSync(layer, rtc::nullopt);
>+    return SkipUntilTlAndSync(layer, absl::nullopt);
>   }
> 
>   // Same as SkipUntilTl, but also waits until the sync bit condition is met.
>-  int SkipUntilTlAndSync(int layer, rtc::Optional<bool> sync) {
>+  int SkipUntilTlAndSync(int layer, absl::optional<bool> sync) {
>     int flags = 0;
>     const int kMaxFramesToSkip =
>         1 + (sync.value_or(false) ? kMaxSyncPeriodSeconds : 1) * kFrameRate;
>@@ -151,7 +151,7 @@ class ScreenshareLayerTest : public ::testing::Test {
>       timestamp_ += kTimestampDelta5Fps;
>       if (vp8_info_.temporalIdx != layer ||
>           (sync && *sync != vp8_info_.layerSync)) {
>-        layers_->FrameEncoded(frame_size_, kDefaultQp);
>+        layers_->FrameEncoded(timestamp_, frame_size_, kDefaultQp);
>       } else {
>         // Found frame from sought after layer.
>         return flags;
>@@ -169,13 +169,13 @@ class ScreenshareLayerTest : public ::testing::Test {
> 
>   uint32_t timestamp_;
>   TemporalLayers::FrameConfig tl_config_;
>-  vpx_codec_enc_cfg_t cfg_;
>+  Vp8EncoderConfig cfg_;
>   bool config_updated_;
>   CodecSpecificInfoVP8 vp8_info_;
> };
> 
> TEST_F(ScreenshareLayerTest, 1Layer) {
>-  layers_.reset(new ScreenshareLayers(1, 0, &clock_));
>+  layers_.reset(new ScreenshareLayers(1, &clock_));
>   ConfigureBitrates();
>   // One layer screenshare should not use the frame dropper as all frames will
>   // belong to the base layer.
>@@ -184,13 +184,11 @@ TEST_F(ScreenshareLayerTest, 1Layer) {
>   timestamp_ += kTimestampDelta5Fps;
>   EXPECT_EQ(static_cast<uint8_t>(kNoTemporalIdx), vp8_info_.temporalIdx);
>   EXPECT_FALSE(vp8_info_.layerSync);
>-  EXPECT_EQ(kNoTl0PicIdx, vp8_info_.tl0PicIdx);
> 
>   flags = EncodeFrame(false);
>   EXPECT_EQ(kSingleLayerFlags, flags);
>   EXPECT_EQ(static_cast<uint8_t>(kNoTemporalIdx), vp8_info_.temporalIdx);
>   EXPECT_FALSE(vp8_info_.layerSync);
>-  EXPECT_EQ(kNoTl0PicIdx, vp8_info_.tl0PicIdx);
> }
> 
> TEST_F(ScreenshareLayerTest, 2LayersPeriodicSync) {
>@@ -218,9 +216,9 @@ TEST_F(ScreenshareLayerTest, 2LayersSyncAfterTimeout) {
> 
>     // Simulate TL1 being at least 8 qp steps better.
>     if (vp8_info_.temporalIdx == 0) {
>-      layers_->FrameEncoded(frame_size_, kDefaultQp);
>+      layers_->FrameEncoded(timestamp_, frame_size_, kDefaultQp);
>     } else {
>-      layers_->FrameEncoded(frame_size_, kDefaultQp - 8);
>+      layers_->FrameEncoded(timestamp_, frame_size_, kDefaultQp - 8);
>     }
> 
>     if (vp8_info_.temporalIdx == 1 && vp8_info_.layerSync)
>@@ -244,9 +242,9 @@ TEST_F(ScreenshareLayerTest, 2LayersSyncAfterSimilarQP) {
> 
>     // Simulate TL1 being at least 8 qp steps better.
>     if (vp8_info_.temporalIdx == 0) {
>-      layers_->FrameEncoded(frame_size_, kDefaultQp);
>+      layers_->FrameEncoded(timestamp_, frame_size_, kDefaultQp);
>     } else {
>-      layers_->FrameEncoded(frame_size_, kDefaultQp - 8);
>+      layers_->FrameEncoded(timestamp_, frame_size_, kDefaultQp - 8);
>     }
> 
>     if (vp8_info_.temporalIdx == 1 && vp8_info_.layerSync)
>@@ -262,10 +260,10 @@ TEST_F(ScreenshareLayerTest, 2LayersSyncAfterSimilarQP) {
>     int flags = ConfigureFrame(false);
>     if (vp8_info_.temporalIdx == 0) {
>       // Bump TL0 to same quality as TL1.
>-      layers_->FrameEncoded(frame_size_, kDefaultQp - 8);
>+      layers_->FrameEncoded(timestamp_, frame_size_, kDefaultQp - 8);
>       bumped_tl0_quality = true;
>     } else {
>-      layers_->FrameEncoded(frame_size_, kDefaultQp - 8);
>+      layers_->FrameEncoded(timestamp_, frame_size_, kDefaultQp - 8);
>       if (bumped_tl0_quality) {
>         EXPECT_TRUE(vp8_info_.layerSync);
>         EXPECT_EQ(kTl1SyncFlags, flags);
>@@ -346,10 +344,9 @@ TEST_F(ScreenshareLayerTest, TooHighBitrate) {
> TEST_F(ScreenshareLayerTest, TargetBitrateCappedByTL0) {
>   const int kTl0_kbps = 100;
>   const int kTl1_kbps = 1000;
>-  layers_->OnRatesUpdated(kTl0_kbps, kTl1_kbps, 5);
>-
>-  EXPECT_THAT(layers_->OnRatesUpdated(kTl0_kbps, kTl1_kbps, 5),
>-              ElementsAre(kTl0_kbps, kTl1_kbps - kTl0_kbps));
>+  const std::vector<uint32_t> layer_rates = {kTl0_kbps * 1000,
>+                                             (kTl1_kbps - kTl0_kbps) * 1000};
>+  layers_->OnRatesUpdated(layer_rates, kFrameRate);
>   EXPECT_TRUE(layers_->UpdateConfiguration(&cfg_));
> 
>   EXPECT_EQ(static_cast<unsigned int>(
>@@ -360,8 +357,9 @@ TEST_F(ScreenshareLayerTest, TargetBitrateCappedByTL0) {
> TEST_F(ScreenshareLayerTest, TargetBitrateCappedByTL1) {
>   const int kTl0_kbps = 100;
>   const int kTl1_kbps = 450;
>-  EXPECT_THAT(layers_->OnRatesUpdated(kTl0_kbps, kTl1_kbps, 5),
>-              ElementsAre(kTl0_kbps, kTl1_kbps - kTl0_kbps));
>+  const std::vector<uint32_t> layer_rates = {kTl0_kbps * 1000,
>+                                             (kTl1_kbps - kTl0_kbps) * 1000};
>+  layers_->OnRatesUpdated(layer_rates, kFrameRate);
>   EXPECT_TRUE(layers_->UpdateConfiguration(&cfg_));
> 
>   EXPECT_EQ(static_cast<unsigned int>(
>@@ -371,12 +369,11 @@ TEST_F(ScreenshareLayerTest, TargetBitrateCappedByTL1) {
> 
> TEST_F(ScreenshareLayerTest, TargetBitrateBelowTL0) {
>   const int kTl0_kbps = 100;
>-  const int kTl1_kbps = 100;
>-  EXPECT_THAT(layers_->OnRatesUpdated(kTl0_kbps, kTl1_kbps, 5),
>-              ElementsAre(kTl0_kbps));
>+  const std::vector<uint32_t> layer_rates = {kTl0_kbps * 1000};
>+  layers_->OnRatesUpdated(layer_rates, kFrameRate);
>   EXPECT_TRUE(layers_->UpdateConfiguration(&cfg_));
> 
>-  EXPECT_EQ(static_cast<uint32_t>(kTl1_kbps), cfg_.rc_target_bitrate);
>+  EXPECT_EQ(static_cast<uint32_t>(kTl0_kbps), cfg_.rc_target_bitrate);
> }
> 
> TEST_F(ScreenshareLayerTest, EncoderDrop) {
>@@ -384,7 +381,7 @@ TEST_F(ScreenshareLayerTest, EncoderDrop) {
>   SkipUntilTl(0);
> 
>   // Size 0 indicates dropped frame.
>-  layers_->FrameEncoded(0, kDefaultQp);
>+  layers_->FrameEncoded(timestamp_, 0, kDefaultQp);
> 
>   // Re-encode frame (so don't advance timestamp).
>   int flags = EncodeFrame(false);
>@@ -396,18 +393,18 @@ TEST_F(ScreenshareLayerTest, EncoderDrop) {
>   SkipUntilTl(0);
>   EXPECT_TRUE(config_updated_);
>   EXPECT_LT(cfg_.rc_max_quantizer, static_cast<unsigned int>(kDefaultQp));
>-  layers_->FrameEncoded(frame_size_, kDefaultQp);
>+  layers_->FrameEncoded(timestamp_, frame_size_, kDefaultQp);
>   timestamp_ += kTimestampDelta5Fps;
> 
>   // ...then back to standard setup.
>   SkipUntilTl(0);
>-  layers_->FrameEncoded(frame_size_, kDefaultQp);
>+  layers_->FrameEncoded(timestamp_, frame_size_, kDefaultQp);
>   timestamp_ += kTimestampDelta5Fps;
>   EXPECT_EQ(cfg_.rc_max_quantizer, static_cast<unsigned int>(kDefaultQp));
> 
>   // Next drop in TL1.
>   SkipUntilTl(1);
>-  layers_->FrameEncoded(0, kDefaultQp);
>+  layers_->FrameEncoded(timestamp_, 0, kDefaultQp);
> 
>   // Re-encode frame (so don't advance timestamp).
>   flags = EncodeFrame(false);
>@@ -419,13 +416,13 @@ TEST_F(ScreenshareLayerTest, EncoderDrop) {
>   SkipUntilTl(1);
>   EXPECT_TRUE(config_updated_);
>   EXPECT_LT(cfg_.rc_max_quantizer, static_cast<unsigned int>(kDefaultQp));
>-  layers_->FrameEncoded(frame_size_, kDefaultQp);
>+  layers_->FrameEncoded(timestamp_, frame_size_, kDefaultQp);
>   timestamp_ += kTimestampDelta5Fps;
> 
>   // ...and back to normal.
>   SkipUntilTl(1);
>   EXPECT_EQ(cfg_.rc_max_quantizer, static_cast<unsigned int>(kDefaultQp));
>-  layers_->FrameEncoded(frame_size_, kDefaultQp);
>+  layers_->FrameEncoded(timestamp_, frame_size_, kDefaultQp);
>   timestamp_ += kTimestampDelta5Fps;
> }
> 
>@@ -434,12 +431,13 @@ TEST_F(ScreenshareLayerTest, RespectsMaxIntervalBetweenFrames) {
>   const int kLargeFrameSizeBytes = 100000;
>   const uint32_t kStartTimestamp = 1234;
> 
>-  layers_->OnRatesUpdated(kLowBitrateKbps, kLowBitrateKbps, 5);
>+  const std::vector<uint32_t> layer_rates = {kLowBitrateKbps * 1000};
>+  layers_->OnRatesUpdated(layer_rates, kFrameRate);
>   layers_->UpdateConfiguration(&cfg_);
> 
>   EXPECT_EQ(kTl0Flags,
>-            VP8EncoderImpl::EncodeFlags(UpdateLayerConfig(kStartTimestamp)));
>-  layers_->FrameEncoded(kLargeFrameSizeBytes, kDefaultQp);
>+            LibvpxVp8Encoder::EncodeFlags(UpdateLayerConfig(kStartTimestamp)));
>+  layers_->FrameEncoded(timestamp_, kLargeFrameSizeBytes, kDefaultQp);
> 
>   const uint32_t kTwoSecondsLater =
>       kStartTimestamp + (ScreenshareLayers::kMaxFrameIntervalMs * 90);
>@@ -456,7 +454,7 @@ TEST_F(ScreenshareLayerTest, RespectsMaxIntervalBetweenFrames) {
> 
>   // More than two seconds has passed since last frame, one should be emitted
>   // even if bitrate target is then exceeded.
>-  EXPECT_EQ(kTl0Flags, VP8EncoderImpl::EncodeFlags(
>+  EXPECT_EQ(kTl0Flags, LibvpxVp8Encoder::EncodeFlags(
>                            UpdateLayerConfig(kTwoSecondsLater + 90)));
> }
> 
>@@ -475,26 +473,26 @@ TEST_F(ScreenshareLayerTest, UpdatesHistograms) {
>       dropped_frame = true;
>       continue;
>     }
>-    int flags = VP8EncoderImpl::EncodeFlags(tl_config_);
>+    int flags = LibvpxVp8Encoder::EncodeFlags(tl_config_);
>     if (flags != -1)
>       layers_->UpdateConfiguration(&cfg_);
> 
>     if (timestamp >= kTimestampDelta5Fps * 5 && !overshoot && flags != -1) {
>       // Simulate one overshoot.
>-      layers_->FrameEncoded(0, 0);
>+      layers_->FrameEncoded(timestamp_, 0, 0);
>       overshoot = true;
>     }
> 
>     if (flags == kTl0Flags) {
>       if (timestamp >= kTimestampDelta5Fps * 20 && !trigger_drop) {
>         // Simulate a too large frame, to cause frame drop.
>-        layers_->FrameEncoded(frame_size_ * 10, kTl0Qp);
>+        layers_->FrameEncoded(timestamp_, frame_size_ * 10, kTl0Qp);
>         trigger_drop = true;
>       } else {
>-        layers_->FrameEncoded(frame_size_, kTl0Qp);
>+        layers_->FrameEncoded(timestamp_, frame_size_, kTl0Qp);
>       }
>     } else if (flags == kTl1Flags || flags == kTl1SyncFlags) {
>-      layers_->FrameEncoded(frame_size_, kTl1Qp);
>+      layers_->FrameEncoded(timestamp_, frame_size_, kTl1Qp);
>     } else if (flags == -1) {
>       dropped_frame = true;
>     } else {
>@@ -540,7 +538,7 @@ TEST_F(ScreenshareLayerTest, UpdatesHistograms) {
> }
> 
> TEST_F(ScreenshareLayerTest, AllowsUpdateConfigBeforeSetRates) {
>-  layers_.reset(new ScreenshareLayers(2, 0, &clock_));
>+  layers_.reset(new ScreenshareLayers(2, &clock_));
>   // New layer instance, OnRatesUpdated() never called.
>   // UpdateConfiguration() call should not cause crash.
>   layers_->UpdateConfiguration(&cfg_);
>@@ -560,7 +558,7 @@ TEST_F(ScreenshareLayerTest, RespectsConfiguredFramerate) {
>       ++num_discarded_frames;
>     } else {
>       size_t frame_size_bytes = kDefaultTl0BitrateKbps * kFrameIntervalsMs / 8;
>-      layers_->FrameEncoded(frame_size_bytes, kDefaultQp);
>+      layers_->FrameEncoded(timestamp_, frame_size_bytes, kDefaultQp);
>     }
>     timestamp += kFrameIntervalsMs * 90;
>     clock_.AdvanceTimeMilliseconds(kFrameIntervalsMs);
>@@ -576,7 +574,7 @@ TEST_F(ScreenshareLayerTest, RespectsConfiguredFramerate) {
>       ++num_discarded_frames;
>     } else {
>       size_t frame_size_bytes = kDefaultTl0BitrateKbps * kFrameIntervalsMs / 8;
>-      layers_->FrameEncoded(frame_size_bytes, kDefaultQp);
>+      layers_->FrameEncoded(timestamp_, frame_size_bytes, kDefaultQp);
>     }
>     timestamp += kFrameIntervalsMs * 90 / 2;
>     clock_.AdvanceTimeMilliseconds(kFrameIntervalsMs / 2);
>@@ -596,10 +594,10 @@ TEST_F(ScreenshareLayerTest, 2LayersSyncAtOvershootDrop) {
>   ASSERT_TRUE(vp8_info_.layerSync);
> 
>   // Simulate overshoot of this frame.
>-  layers_->FrameEncoded(0, -1);
>+  layers_->FrameEncoded(timestamp_, 0, -1);
> 
>   config_updated_ = layers_->UpdateConfiguration(&cfg_);
>-  EXPECT_EQ(kTl1SyncFlags, VP8EncoderImpl::EncodeFlags(tl_config_));
>+  EXPECT_EQ(kTl1SyncFlags, LibvpxVp8Encoder::EncodeFlags(tl_config_));
> 
>   CodecSpecificInfoVP8 new_vp8_info;
>   layers_->PopulateCodecSpecific(false, tl_config_, &new_vp8_info, timestamp_);
>@@ -613,7 +611,7 @@ TEST_F(ScreenshareLayerTest, DropOnTooShortFrameInterval) {
>   // Add a large gap, so there's plenty of room in the rate tracker.
>   timestamp_ += kTimestampDelta5Fps * 3;
>   EXPECT_FALSE(UpdateLayerConfig(timestamp_).drop_frame);
>-  layers_->FrameEncoded(frame_size_, kDefaultQp);
>+  layers_->FrameEncoded(timestamp_, frame_size_, kDefaultQp);
> 
>   // Frame interval below 90% if desired time is not allowed, try inserting
>   // frame just before this limit.
>@@ -630,7 +628,7 @@ TEST_F(ScreenshareLayerTest, AdjustsBitrateWhenDroppingFrames) {
>   const uint32_t kTimestampDelta10Fps = kTimestampDelta5Fps / 2;
>   const int kNumFrames = 30;
>   uint32_t default_bitrate = cfg_.rc_target_bitrate;
>-  layers_->OnRatesUpdated(kDefaultTl0BitrateKbps, kDefaultTl1BitrateKbps, 10);
>+  layers_->OnRatesUpdated(kDefault2TlBitratesBps, 10);
> 
>   int num_dropped_frames = 0;
>   for (int i = 0; i < kNumFrames; ++i) {
>@@ -644,4 +642,23 @@ TEST_F(ScreenshareLayerTest, AdjustsBitrateWhenDroppingFrames) {
>   EXPECT_EQ(cfg_.rc_target_bitrate, default_bitrate * 2);
> }
> 
>+TEST_F(ScreenshareLayerTest, UpdatesConfigurationAfterRateChange) {
>+  // Set inital rate again, no need to update configuration.
>+  layers_->OnRatesUpdated(kDefault2TlBitratesBps, kFrameRate);
>+  EXPECT_FALSE(layers_->UpdateConfiguration(&cfg_));
>+
>+  // Rate changed, now update config.
>+  std::vector<uint32_t> bitrates = kDefault2TlBitratesBps;
>+  bitrates[1] -= 100000;
>+  layers_->OnRatesUpdated(bitrates, 5);
>+  EXPECT_TRUE(layers_->UpdateConfiguration(&cfg_));
>+
>+  // Changed rate, but then set changed rate again before trying to update
>+  // configuration, update should still apply.
>+  bitrates[1] -= 100000;
>+  layers_->OnRatesUpdated(bitrates, 5);
>+  layers_->OnRatesUpdated(bitrates, 5);
>+  EXPECT_TRUE(layers_->UpdateConfiguration(&cfg_));
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/simulcast_rate_allocator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/simulcast_rate_allocator.cc
>deleted file mode 100644
>index b2b3334815f..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/simulcast_rate_allocator.cc
>+++ /dev/null
>@@ -1,163 +0,0 @@
>-/*
>- *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "modules/video_coding/codecs/vp8/simulcast_rate_allocator.h"
>-
>-#include <algorithm>
>-#include <memory>
>-#include <vector>
>-#include <utility>
>-
>-#include "rtc_base/checks.h"
>-
>-namespace webrtc {
>-
>-SimulcastRateAllocator::SimulcastRateAllocator(
>-    const VideoCodec& codec,
>-    std::unique_ptr<TemporalLayersFactory> tl_factory)
>-    : codec_(codec), tl_factory_(std::move(tl_factory)) {
>-  if (tl_factory_.get())
>-    tl_factory_->SetListener(this);
>-}
>-
>-void SimulcastRateAllocator::OnTemporalLayersCreated(int simulcast_id,
>-                                                     TemporalLayers* layers) {
>-  RTC_DCHECK(temporal_layers_.find(simulcast_id) == temporal_layers_.end());
>-  RTC_DCHECK(layers);
>-  temporal_layers_[simulcast_id] = layers;
>-}
>-
>-BitrateAllocation SimulcastRateAllocator::GetAllocation(
>-    uint32_t total_bitrate_bps,
>-    uint32_t framerate) {
>-  uint32_t left_to_allocate = total_bitrate_bps;
>-  if (codec_.maxBitrate && codec_.maxBitrate * 1000 < left_to_allocate)
>-    left_to_allocate = codec_.maxBitrate * 1000;
>-
>-  BitrateAllocation allocated_bitrates_bps;
>-  if (codec_.numberOfSimulcastStreams == 0) {
>-    // No simulcast, just set the target as this has been capped already.
>-    allocated_bitrates_bps.SetBitrate(
>-        0, 0, std::max(codec_.minBitrate * 1000, left_to_allocate));
>-  } else {
>-    // Always allocate enough bitrate for the minimum bitrate of the first
>-    // layer. Suspending below min bitrate is controlled outside the codec
>-    // implementation and is not overridden by this.
>-    left_to_allocate =
>-        std::max(codec_.simulcastStream[0].minBitrate * 1000, left_to_allocate);
>-
>-    // Begin by allocating bitrate to simulcast streams, putting all bitrate in
>-    // temporal layer 0. We'll then distribute this bitrate, across potential
>-    // temporal layers, when stream allocation is done.
>-
>-    // Allocate up to the target bitrate for each simulcast layer.
>-    size_t layer = 0;
>-    for (; layer < codec_.numberOfSimulcastStreams; ++layer) {
>-      const SimulcastStream& stream = codec_.simulcastStream[layer];
>-      if (left_to_allocate < stream.minBitrate * 1000)
>-        break;
>-      uint32_t allocation =
>-          std::min(left_to_allocate, stream.targetBitrate * 1000);
>-      allocated_bitrates_bps.SetBitrate(layer, 0, allocation);
>-      RTC_DCHECK_LE(allocation, left_to_allocate);
>-      left_to_allocate -= allocation;
>-    }
>-
>-    // Next, try allocate remaining bitrate, up to max bitrate, in top stream.
>-    // TODO(sprang): Allocate up to max bitrate for all layers once we have a
>-    //               better idea of possible performance implications.
>-    if (left_to_allocate > 0) {
>-      size_t active_layer = layer - 1;
>-      const SimulcastStream& stream = codec_.simulcastStream[active_layer];
>-      uint32_t bitrate_bps =
>-          allocated_bitrates_bps.GetSpatialLayerSum(active_layer);
>-      uint32_t allocation =
>-          std::min(left_to_allocate, stream.maxBitrate * 1000 - bitrate_bps);
>-      bitrate_bps += allocation;
>-      RTC_DCHECK_LE(allocation, left_to_allocate);
>-      left_to_allocate -= allocation;
>-      allocated_bitrates_bps.SetBitrate(active_layer, 0, bitrate_bps);
>-    }
>-  }
>-
>-  const int num_spatial_streams =
>-      std::max(1, static_cast<int>(codec_.numberOfSimulcastStreams));
>-
>-  // Finally, distribute the bitrate for the simulcast streams across the
>-  // available temporal layers.
>-  for (int simulcast_id = 0; simulcast_id < num_spatial_streams;
>-       ++simulcast_id) {
>-    auto tl_it = temporal_layers_.find(simulcast_id);
>-    if (tl_it == temporal_layers_.end())
>-      continue;  // TODO(sprang): If > 1 SS, assume default TL alloc?
>-
>-    uint32_t target_bitrate_kbps =
>-        allocated_bitrates_bps.GetBitrate(simulcast_id, 0) / 1000;
>-    const uint32_t expected_allocated_bitrate_kbps = target_bitrate_kbps;
>-    RTC_DCHECK_EQ(
>-        target_bitrate_kbps,
>-        allocated_bitrates_bps.GetSpatialLayerSum(simulcast_id) / 1000);
>-    const int num_temporal_streams = std::max<uint8_t>(
>-        1, codec_.numberOfSimulcastStreams == 0
>-               ? codec_.VP8().numberOfTemporalLayers
>-               : codec_.simulcastStream[simulcast_id].numberOfTemporalLayers);
>-
>-    uint32_t max_bitrate_kbps;
>-    // Legacy temporal-layered only screenshare, or simulcast screenshare
>-    // with legacy mode for simulcast stream 0.
>-    if (codec_.mode == kScreensharing && codec_.targetBitrate > 0 &&
>-        ((num_spatial_streams == 1 && num_temporal_streams == 2) ||  // Legacy.
>-         (num_spatial_streams > 1 && simulcast_id == 0))) {  // Simulcast.
>-      // TODO(holmer): This is a "temporary" hack for screensharing, where we
>-      // interpret the startBitrate as the encoder target bitrate. This is
>-      // to allow for a different max bitrate, so if the codec can't meet
>-      // the target we still allow it to overshoot up to the max before dropping
>-      // frames. This hack should be improved.
>-      int tl0_bitrate = std::min(codec_.targetBitrate, target_bitrate_kbps);
>-      max_bitrate_kbps = std::min(codec_.maxBitrate, target_bitrate_kbps);
>-      target_bitrate_kbps = tl0_bitrate;
>-    } else if (num_spatial_streams == 1) {
>-      max_bitrate_kbps = codec_.maxBitrate;
>-    } else {
>-      max_bitrate_kbps = codec_.simulcastStream[simulcast_id].maxBitrate;
>-    }
>-
>-    std::vector<uint32_t> tl_allocation = tl_it->second->OnRatesUpdated(
>-        target_bitrate_kbps, max_bitrate_kbps, framerate);
>-    RTC_DCHECK_GT(tl_allocation.size(), 0);
>-    RTC_DCHECK_LE(tl_allocation.size(), num_temporal_streams);
>-
>-    uint64_t tl_allocation_sum_kbps = 0;
>-    for (size_t tl_index = 0; tl_index < tl_allocation.size(); ++tl_index) {
>-      uint32_t layer_rate_kbps = tl_allocation[tl_index];
>-      allocated_bitrates_bps.SetBitrate(simulcast_id, tl_index,
>-                                        layer_rate_kbps * 1000);
>-      tl_allocation_sum_kbps += layer_rate_kbps;
>-    }
>-    RTC_DCHECK_LE(tl_allocation_sum_kbps, expected_allocated_bitrate_kbps);
>-  }
>-
>-  return allocated_bitrates_bps;
>-}
>-
>-uint32_t SimulcastRateAllocator::GetPreferredBitrateBps(uint32_t framerate) {
>-  // Create a temporary instance without temporal layers, as they may be
>-  // stateful, and updating the bitrate to max here can cause side effects.
>-  SimulcastRateAllocator temp_allocator(codec_, nullptr);
>-  BitrateAllocation allocation =
>-      temp_allocator.GetAllocation(codec_.maxBitrate * 1000, framerate);
>-  return allocation.get_sum_bps();
>-}
>-
>-const VideoCodec& webrtc::SimulcastRateAllocator::GetCodec() const {
>-  return codec_;
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/simulcast_rate_allocator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/simulcast_rate_allocator.h
>deleted file mode 100644
>index 929abba4132..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/simulcast_rate_allocator.h
>+++ /dev/null
>@@ -1,51 +0,0 @@
>-/*
>- *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MODULES_VIDEO_CODING_CODECS_VP8_SIMULCAST_RATE_ALLOCATOR_H_
>-#define MODULES_VIDEO_CODING_CODECS_VP8_SIMULCAST_RATE_ALLOCATOR_H_
>-
>-#include <stdint.h>
>-
>-#include <map>
>-#include <memory>
>-
>-#include "api/video_codecs/video_encoder.h"
>-#include "common_video/include/video_bitrate_allocator.h"
>-#include "modules/video_coding/codecs/vp8/temporal_layers.h"
>-#include "rtc_base/constructormagic.h"
>-
>-namespace webrtc {
>-
>-class SimulcastRateAllocator : public VideoBitrateAllocator,
>-                               public TemporalLayersListener {
>- public:
>-  explicit SimulcastRateAllocator(
>-      const VideoCodec& codec,
>-      std::unique_ptr<TemporalLayersFactory> tl_factory);
>-
>-  void OnTemporalLayersCreated(int simulcast_id,
>-                               TemporalLayers* layers) override;
>-
>-  BitrateAllocation GetAllocation(uint32_t total_bitrate_bps,
>-                                  uint32_t framerate) override;
>-  uint32_t GetPreferredBitrateBps(uint32_t framerate) override;
>-  const VideoCodec& GetCodec() const;
>-
>- private:
>-  const VideoCodec codec_;
>-  std::map<uint32_t, TemporalLayers*> temporal_layers_;
>-  std::unique_ptr<TemporalLayersFactory> tl_factory_;
>-
>-  RTC_DISALLOW_COPY_AND_ASSIGN(SimulcastRateAllocator);
>-};
>-
>-}  // namespace webrtc
>-
>-#endif  // MODULES_VIDEO_CODING_CODECS_VP8_SIMULCAST_RATE_ALLOCATOR_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/simulcast_test_utility.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/simulcast_test_utility.h
>deleted file mode 100644
>index 7f32ad2f8e5..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/simulcast_test_utility.h
>+++ /dev/null
>@@ -1,755 +0,0 @@
>-/*
>- *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MODULES_VIDEO_CODING_CODECS_VP8_SIMULCAST_TEST_UTILITY_H_
>-#define MODULES_VIDEO_CODING_CODECS_VP8_SIMULCAST_TEST_UTILITY_H_
>-
>-#include <algorithm>
>-#include <map>
>-#include <memory>
>-#include <vector>
>-
>-#include "api/video/i420_buffer.h"
>-#include "api/video/video_frame.h"
>-#include "common_video/include/video_frame.h"
>-#include "common_video/libyuv/include/webrtc_libyuv.h"
>-#include "modules/video_coding/codecs/vp8/include/vp8.h"
>-#include "modules/video_coding/codecs/vp8/simulcast_rate_allocator.h"
>-#include "modules/video_coding/codecs/vp8/temporal_layers.h"
>-#include "modules/video_coding/include/mock/mock_video_codec_interface.h"
>-#include "modules/video_coding/include/video_coding_defines.h"
>-#include "rtc_base/checks.h"
>-#include "test/gtest.h"
>-
>-using ::testing::_;
>-using ::testing::AllOf;
>-using ::testing::Field;
>-using ::testing::Return;
>-
>-namespace webrtc {
>-namespace testing {
>-
>-const int kDefaultWidth = 1280;
>-const int kDefaultHeight = 720;
>-const int kNumberOfSimulcastStreams = 3;
>-const int kColorY = 66;
>-const int kColorU = 22;
>-const int kColorV = 33;
>-const int kMaxBitrates[kNumberOfSimulcastStreams] = {150, 600, 1200};
>-const int kMinBitrates[kNumberOfSimulcastStreams] = {50, 150, 600};
>-const int kTargetBitrates[kNumberOfSimulcastStreams] = {100, 450, 1000};
>-const int kDefaultTemporalLayerProfile[3] = {3, 3, 3};
>-
>-template <typename T>
>-void SetExpectedValues3(T value0, T value1, T value2, T* expected_values) {
>-  expected_values[0] = value0;
>-  expected_values[1] = value1;
>-  expected_values[2] = value2;
>-}
>-
>-enum PlaneType {
>-  kYPlane = 0,
>-  kUPlane = 1,
>-  kVPlane = 2,
>-  kNumOfPlanes = 3,
>-};
>-
>-class Vp8TestEncodedImageCallback : public EncodedImageCallback {
>- public:
>-  Vp8TestEncodedImageCallback() : picture_id_(-1) {
>-    memset(temporal_layer_, -1, sizeof(temporal_layer_));
>-    memset(layer_sync_, false, sizeof(layer_sync_));
>-  }
>-
>-  ~Vp8TestEncodedImageCallback() {
>-    delete[] encoded_key_frame_._buffer;
>-    delete[] encoded_frame_._buffer;
>-  }
>-
>-  virtual Result OnEncodedImage(const EncodedImage& encoded_image,
>-                                const CodecSpecificInfo* codec_specific_info,
>-                                const RTPFragmentationHeader* fragmentation) {
>-    // Only store the base layer.
>-    if (codec_specific_info->codecSpecific.VP8.simulcastIdx == 0) {
>-      if (encoded_image._frameType == kVideoFrameKey) {
>-        delete[] encoded_key_frame_._buffer;
>-        encoded_key_frame_._buffer = new uint8_t[encoded_image._size];
>-        encoded_key_frame_._size = encoded_image._size;
>-        encoded_key_frame_._length = encoded_image._length;
>-        encoded_key_frame_._frameType = kVideoFrameKey;
>-        encoded_key_frame_._completeFrame = encoded_image._completeFrame;
>-        memcpy(encoded_key_frame_._buffer, encoded_image._buffer,
>-               encoded_image._length);
>-      } else {
>-        delete[] encoded_frame_._buffer;
>-        encoded_frame_._buffer = new uint8_t[encoded_image._size];
>-        encoded_frame_._size = encoded_image._size;
>-        encoded_frame_._length = encoded_image._length;
>-        memcpy(encoded_frame_._buffer, encoded_image._buffer,
>-               encoded_image._length);
>-      }
>-    }
>-    picture_id_ = codec_specific_info->codecSpecific.VP8.pictureId;
>-    layer_sync_[codec_specific_info->codecSpecific.VP8.simulcastIdx] =
>-        codec_specific_info->codecSpecific.VP8.layerSync;
>-    temporal_layer_[codec_specific_info->codecSpecific.VP8.simulcastIdx] =
>-        codec_specific_info->codecSpecific.VP8.temporalIdx;
>-    return Result(Result::OK, encoded_image._timeStamp);
>-  }
>-  void GetLastEncodedFrameInfo(int* picture_id,
>-                               int* temporal_layer,
>-                               bool* layer_sync,
>-                               int stream) {
>-    *picture_id = picture_id_;
>-    *temporal_layer = temporal_layer_[stream];
>-    *layer_sync = layer_sync_[stream];
>-  }
>-  void GetLastEncodedKeyFrame(EncodedImage* encoded_key_frame) {
>-    *encoded_key_frame = encoded_key_frame_;
>-  }
>-  void GetLastEncodedFrame(EncodedImage* encoded_frame) {
>-    *encoded_frame = encoded_frame_;
>-  }
>-
>- private:
>-  EncodedImage encoded_key_frame_;
>-  EncodedImage encoded_frame_;
>-  int picture_id_;
>-  int temporal_layer_[kNumberOfSimulcastStreams];
>-  bool layer_sync_[kNumberOfSimulcastStreams];
>-};
>-
>-class Vp8TestDecodedImageCallback : public DecodedImageCallback {
>- public:
>-  Vp8TestDecodedImageCallback() : decoded_frames_(0) {}
>-  int32_t Decoded(VideoFrame& decoded_image) override {
>-    rtc::scoped_refptr<I420BufferInterface> i420_buffer =
>-        decoded_image.video_frame_buffer()->ToI420();
>-    for (int i = 0; i < decoded_image.width(); ++i) {
>-      EXPECT_NEAR(kColorY, i420_buffer->DataY()[i], 1);
>-    }
>-
>-    // TODO(mikhal): Verify the difference between U,V and the original.
>-    for (int i = 0; i < i420_buffer->ChromaWidth(); ++i) {
>-      EXPECT_NEAR(kColorU, i420_buffer->DataU()[i], 4);
>-      EXPECT_NEAR(kColorV, i420_buffer->DataV()[i], 4);
>-    }
>-    decoded_frames_++;
>-    return 0;
>-  }
>-  int32_t Decoded(VideoFrame& decoded_image, int64_t decode_time_ms) override {
>-    RTC_NOTREACHED();
>-    return -1;
>-  }
>-  void Decoded(VideoFrame& decoded_image,
>-               rtc::Optional<int32_t> decode_time_ms,
>-               rtc::Optional<uint8_t> qp) override {
>-    Decoded(decoded_image);
>-  }
>-  int DecodedFrames() { return decoded_frames_; }
>-
>- private:
>-  int decoded_frames_;
>-};
>-
>-class TestVp8Simulcast : public ::testing::Test {
>- public:
>-  static void SetPlane(uint8_t* data,
>-                       uint8_t value,
>-                       int width,
>-                       int height,
>-                       int stride) {
>-    for (int i = 0; i < height; i++, data += stride) {
>-      // Setting allocated area to zero - setting only image size to
>-      // requested values - will make it easier to distinguish between image
>-      // size and frame size (accounting for stride).
>-      memset(data, value, width);
>-      memset(data + width, 0, stride - width);
>-    }
>-  }
>-
>-  // Fills in an I420Buffer from |plane_colors|.
>-  static void CreateImage(const rtc::scoped_refptr<I420Buffer>& buffer,
>-                          int plane_colors[kNumOfPlanes]) {
>-    SetPlane(buffer->MutableDataY(), plane_colors[0], buffer->width(),
>-             buffer->height(), buffer->StrideY());
>-
>-    SetPlane(buffer->MutableDataU(), plane_colors[1], buffer->ChromaWidth(),
>-             buffer->ChromaHeight(), buffer->StrideU());
>-
>-    SetPlane(buffer->MutableDataV(), plane_colors[2], buffer->ChromaWidth(),
>-             buffer->ChromaHeight(), buffer->StrideV());
>-  }
>-
>-  static void DefaultSettings(VideoCodec* settings,
>-                              const int* temporal_layer_profile) {
>-    RTC_CHECK(settings);
>-    memset(settings, 0, sizeof(VideoCodec));
>-    strncpy(settings->plName, "VP8", 4);
>-    settings->codecType = kVideoCodecVP8;
>-    // 96 to 127 dynamic payload types for video codecs
>-    settings->plType = 120;
>-    settings->startBitrate = 300;
>-    settings->minBitrate = 30;
>-    settings->maxBitrate = 0;
>-    settings->maxFramerate = 30;
>-    settings->width = kDefaultWidth;
>-    settings->height = kDefaultHeight;
>-    settings->numberOfSimulcastStreams = kNumberOfSimulcastStreams;
>-    ASSERT_EQ(3, kNumberOfSimulcastStreams);
>-    settings->timing_frame_thresholds = {kDefaultTimingFramesDelayMs,
>-                                         kDefaultOutlierFrameSizePercent};
>-    ConfigureStream(kDefaultWidth / 4, kDefaultHeight / 4, kMaxBitrates[0],
>-                    kMinBitrates[0], kTargetBitrates[0],
>-                    &settings->simulcastStream[0], temporal_layer_profile[0]);
>-    ConfigureStream(kDefaultWidth / 2, kDefaultHeight / 2, kMaxBitrates[1],
>-                    kMinBitrates[1], kTargetBitrates[1],
>-                    &settings->simulcastStream[1], temporal_layer_profile[1]);
>-    ConfigureStream(kDefaultWidth, kDefaultHeight, kMaxBitrates[2],
>-                    kMinBitrates[2], kTargetBitrates[2],
>-                    &settings->simulcastStream[2], temporal_layer_profile[2]);
>-    settings->VP8()->resilience = kResilientStream;
>-    settings->VP8()->denoisingOn = true;
>-    settings->VP8()->errorConcealmentOn = false;
>-    settings->VP8()->automaticResizeOn = false;
>-    settings->VP8()->frameDroppingOn = true;
>-    settings->VP8()->keyFrameInterval = 3000;
>-  }
>-
>-  static void ConfigureStream(int width,
>-                              int height,
>-                              int max_bitrate,
>-                              int min_bitrate,
>-                              int target_bitrate,
>-                              SimulcastStream* stream,
>-                              int num_temporal_layers) {
>-    assert(stream);
>-    stream->width = width;
>-    stream->height = height;
>-    stream->maxBitrate = max_bitrate;
>-    stream->minBitrate = min_bitrate;
>-    stream->targetBitrate = target_bitrate;
>-    stream->numberOfTemporalLayers = num_temporal_layers;
>-    stream->qpMax = 45;
>-  }
>-
>- protected:
>-  virtual std::unique_ptr<VP8Encoder> CreateEncoder() = 0;
>-  virtual std::unique_ptr<VP8Decoder> CreateDecoder() = 0;
>-
>-  void SetUp() override {
>-    encoder_ = CreateEncoder();
>-    decoder_ = CreateDecoder();
>-    SetUpCodec(kDefaultTemporalLayerProfile);
>-  }
>-
>-  void TearDown() override {
>-    encoder_->Release();
>-    decoder_->Release();
>-    encoder_.reset();
>-    decoder_.reset();
>-  }
>-
>-  void SetUpCodec(const int* temporal_layer_profile) {
>-    encoder_->RegisterEncodeCompleteCallback(&encoder_callback_);
>-    decoder_->RegisterDecodeCompleteCallback(&decoder_callback_);
>-    DefaultSettings(&settings_, temporal_layer_profile);
>-    SetUpRateAllocator();
>-    EXPECT_EQ(0, encoder_->InitEncode(&settings_, 1, 1200));
>-    EXPECT_EQ(0, decoder_->InitDecode(&settings_, 1));
>-    input_buffer_ = I420Buffer::Create(kDefaultWidth, kDefaultHeight);
>-    input_buffer_->InitializeData();
>-    input_frame_.reset(
>-        new VideoFrame(input_buffer_, 0, 0, webrtc::kVideoRotation_0));
>-  }
>-
>-  void SetUpRateAllocator() {
>-    TemporalLayersFactory* tl_factory = new TemporalLayersFactory();
>-    rate_allocator_.reset(new SimulcastRateAllocator(
>-        settings_, std::unique_ptr<TemporalLayersFactory>(tl_factory)));
>-    settings_.VP8()->tl_factory = tl_factory;
>-  }
>-
>-  void SetRates(uint32_t bitrate_kbps, uint32_t fps) {
>-    encoder_->SetRateAllocation(
>-        rate_allocator_->GetAllocation(bitrate_kbps * 1000, fps), fps);
>-  }
>-
>-  void ExpectStreams(FrameType frame_type, int expected_video_streams) {
>-    ASSERT_GE(expected_video_streams, 0);
>-    ASSERT_LE(expected_video_streams, kNumberOfSimulcastStreams);
>-    if (expected_video_streams >= 1) {
>-      EXPECT_CALL(
>-          encoder_callback_,
>-          OnEncodedImage(
>-              AllOf(Field(&EncodedImage::_frameType, frame_type),
>-                    Field(&EncodedImage::_encodedWidth, kDefaultWidth / 4),
>-                    Field(&EncodedImage::_encodedHeight, kDefaultHeight / 4)),
>-              _, _))
>-          .Times(1)
>-          .WillRepeatedly(Return(EncodedImageCallback::Result(
>-              EncodedImageCallback::Result::OK, 0)));
>-    }
>-    if (expected_video_streams >= 2) {
>-      EXPECT_CALL(
>-          encoder_callback_,
>-          OnEncodedImage(
>-              AllOf(Field(&EncodedImage::_frameType, frame_type),
>-                    Field(&EncodedImage::_encodedWidth, kDefaultWidth / 2),
>-                    Field(&EncodedImage::_encodedHeight, kDefaultHeight / 2)),
>-              _, _))
>-          .Times(1)
>-          .WillRepeatedly(Return(EncodedImageCallback::Result(
>-              EncodedImageCallback::Result::OK, 0)));
>-    }
>-    if (expected_video_streams >= 3) {
>-      EXPECT_CALL(
>-          encoder_callback_,
>-          OnEncodedImage(
>-              AllOf(Field(&EncodedImage::_frameType, frame_type),
>-                    Field(&EncodedImage::_encodedWidth, kDefaultWidth),
>-                    Field(&EncodedImage::_encodedHeight, kDefaultHeight)),
>-              _, _))
>-          .Times(1)
>-          .WillRepeatedly(Return(EncodedImageCallback::Result(
>-              EncodedImageCallback::Result::OK, 0)));
>-    }
>-  }
>-
>-  void VerifyTemporalIdxAndSyncForAllSpatialLayers(
>-      Vp8TestEncodedImageCallback* encoder_callback,
>-      const int* expected_temporal_idx,
>-      const bool* expected_layer_sync,
>-      int num_spatial_layers) {
>-    int picture_id = -1;
>-    int temporal_layer = -1;
>-    bool layer_sync = false;
>-    for (int i = 0; i < num_spatial_layers; i++) {
>-      encoder_callback->GetLastEncodedFrameInfo(&picture_id, &temporal_layer,
>-                                                &layer_sync, i);
>-      EXPECT_EQ(expected_temporal_idx[i], temporal_layer);
>-      EXPECT_EQ(expected_layer_sync[i], layer_sync);
>-    }
>-  }
>-
>-  // We currently expect all active streams to generate a key frame even though
>-  // a key frame was only requested for some of them.
>-  void TestKeyFrameRequestsOnAllStreams() {
>-    SetRates(kMaxBitrates[2], 30);  // To get all three streams.
>-    std::vector<FrameType> frame_types(kNumberOfSimulcastStreams,
>-                                       kVideoFrameDelta);
>-    ExpectStreams(kVideoFrameKey, kNumberOfSimulcastStreams);
>-    EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>-
>-    ExpectStreams(kVideoFrameDelta, kNumberOfSimulcastStreams);
>-    input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>-    EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>-
>-    frame_types[0] = kVideoFrameKey;
>-    ExpectStreams(kVideoFrameKey, kNumberOfSimulcastStreams);
>-    input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>-    EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>-
>-    std::fill(frame_types.begin(), frame_types.end(), kVideoFrameDelta);
>-    frame_types[1] = kVideoFrameKey;
>-    ExpectStreams(kVideoFrameKey, kNumberOfSimulcastStreams);
>-    input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>-    EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>-
>-    std::fill(frame_types.begin(), frame_types.end(), kVideoFrameDelta);
>-    frame_types[2] = kVideoFrameKey;
>-    ExpectStreams(kVideoFrameKey, kNumberOfSimulcastStreams);
>-    input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>-    EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>-
>-    std::fill(frame_types.begin(), frame_types.end(), kVideoFrameDelta);
>-    ExpectStreams(kVideoFrameDelta, kNumberOfSimulcastStreams);
>-    input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>-    EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>-  }
>-
>-  void TestPaddingAllStreams() {
>-    // We should always encode the base layer.
>-    SetRates(kMinBitrates[0] - 1, 30);
>-    std::vector<FrameType> frame_types(kNumberOfSimulcastStreams,
>-                                       kVideoFrameDelta);
>-    ExpectStreams(kVideoFrameKey, 1);
>-    EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>-
>-    ExpectStreams(kVideoFrameDelta, 1);
>-    input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>-    EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>-  }
>-
>-  void TestPaddingTwoStreams() {
>-    // We have just enough to get only the first stream and padding for two.
>-    SetRates(kMinBitrates[0], 30);
>-    std::vector<FrameType> frame_types(kNumberOfSimulcastStreams,
>-                                       kVideoFrameDelta);
>-    ExpectStreams(kVideoFrameKey, 1);
>-    EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>-
>-    ExpectStreams(kVideoFrameDelta, 1);
>-    input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>-    EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>-  }
>-
>-  void TestPaddingTwoStreamsOneMaxedOut() {
>-    // We are just below limit of sending second stream, so we should get
>-    // the first stream maxed out (at |maxBitrate|), and padding for two.
>-    SetRates(kTargetBitrates[0] + kMinBitrates[1] - 1, 30);
>-    std::vector<FrameType> frame_types(kNumberOfSimulcastStreams,
>-                                       kVideoFrameDelta);
>-    ExpectStreams(kVideoFrameKey, 1);
>-    EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>-
>-    ExpectStreams(kVideoFrameDelta, 1);
>-    input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>-    EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>-  }
>-
>-  void TestPaddingOneStream() {
>-    // We have just enough to send two streams, so padding for one stream.
>-    SetRates(kTargetBitrates[0] + kMinBitrates[1], 30);
>-    std::vector<FrameType> frame_types(kNumberOfSimulcastStreams,
>-                                       kVideoFrameDelta);
>-    ExpectStreams(kVideoFrameKey, 2);
>-    EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>-
>-    ExpectStreams(kVideoFrameDelta, 2);
>-    input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>-    EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>-  }
>-
>-  void TestPaddingOneStreamTwoMaxedOut() {
>-    // We are just below limit of sending third stream, so we should get
>-    // first stream's rate maxed out at |targetBitrate|, second at |maxBitrate|.
>-    SetRates(kTargetBitrates[0] + kTargetBitrates[1] + kMinBitrates[2] - 1, 30);
>-    std::vector<FrameType> frame_types(kNumberOfSimulcastStreams,
>-                                       kVideoFrameDelta);
>-    ExpectStreams(kVideoFrameKey, 2);
>-    EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>-
>-    ExpectStreams(kVideoFrameDelta, 2);
>-    input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>-    EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>-  }
>-
>-  void TestSendAllStreams() {
>-    // We have just enough to send all streams.
>-    SetRates(kTargetBitrates[0] + kTargetBitrates[1] + kMinBitrates[2], 30);
>-    std::vector<FrameType> frame_types(kNumberOfSimulcastStreams,
>-                                       kVideoFrameDelta);
>-    ExpectStreams(kVideoFrameKey, 3);
>-    EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>-
>-    ExpectStreams(kVideoFrameDelta, 3);
>-    input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>-    EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>-  }
>-
>-  void TestDisablingStreams() {
>-    // We should get three media streams.
>-    SetRates(kMaxBitrates[0] + kMaxBitrates[1] + kMaxBitrates[2], 30);
>-    std::vector<FrameType> frame_types(kNumberOfSimulcastStreams,
>-                                       kVideoFrameDelta);
>-    ExpectStreams(kVideoFrameKey, 3);
>-    EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>-
>-    ExpectStreams(kVideoFrameDelta, 3);
>-    input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>-    EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>-
>-    // We should only get two streams and padding for one.
>-    SetRates(kTargetBitrates[0] + kTargetBitrates[1] + kMinBitrates[2] / 2, 30);
>-    ExpectStreams(kVideoFrameDelta, 2);
>-    input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>-    EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>-
>-    // We should only get the first stream and padding for two.
>-    SetRates(kTargetBitrates[0] + kMinBitrates[1] / 2, 30);
>-    ExpectStreams(kVideoFrameDelta, 1);
>-    input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>-    EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>-
>-    // We don't have enough bitrate for the thumbnail stream, but we should get
>-    // it anyway with current configuration.
>-    SetRates(kTargetBitrates[0] - 1, 30);
>-    ExpectStreams(kVideoFrameDelta, 1);
>-    input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>-    EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>-
>-    // We should only get two streams and padding for one.
>-    SetRates(kTargetBitrates[0] + kTargetBitrates[1] + kMinBitrates[2] / 2, 30);
>-    // We get a key frame because a new stream is being enabled.
>-    ExpectStreams(kVideoFrameKey, 2);
>-    input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>-    EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>-
>-    // We should get all three streams.
>-    SetRates(kTargetBitrates[0] + kTargetBitrates[1] + kTargetBitrates[2], 30);
>-    // We get a key frame because a new stream is being enabled.
>-    ExpectStreams(kVideoFrameKey, 3);
>-    input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>-    EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>-  }
>-
>-  void SwitchingToOneStream(int width, int height) {
>-    // Disable all streams except the last and set the bitrate of the last to
>-    // 100 kbps. This verifies the way GTP switches to screenshare mode.
>-    settings_.VP8()->numberOfTemporalLayers = 1;
>-    settings_.maxBitrate = 100;
>-    settings_.startBitrate = 100;
>-    settings_.width = width;
>-    settings_.height = height;
>-    for (int i = 0; i < settings_.numberOfSimulcastStreams - 1; ++i) {
>-      settings_.simulcastStream[i].maxBitrate = 0;
>-      settings_.simulcastStream[i].width = settings_.width;
>-      settings_.simulcastStream[i].height = settings_.height;
>-    }
>-    // Setting input image to new resolution.
>-    input_buffer_ = I420Buffer::Create(settings_.width, settings_.height);
>-    input_buffer_->InitializeData();
>-
>-    input_frame_.reset(
>-        new VideoFrame(input_buffer_, 0, 0, webrtc::kVideoRotation_0));
>-
>-    // The for loop above did not set the bitrate of the highest layer.
>-    settings_.simulcastStream[settings_.numberOfSimulcastStreams - 1]
>-        .maxBitrate = 0;
>-    // The highest layer has to correspond to the non-simulcast resolution.
>-    settings_.simulcastStream[settings_.numberOfSimulcastStreams - 1].width =
>-        settings_.width;
>-    settings_.simulcastStream[settings_.numberOfSimulcastStreams - 1].height =
>-        settings_.height;
>-    SetUpRateAllocator();
>-    EXPECT_EQ(0, encoder_->InitEncode(&settings_, 1, 1200));
>-
>-    // Encode one frame and verify.
>-    SetRates(kMaxBitrates[0] + kMaxBitrates[1], 30);
>-    std::vector<FrameType> frame_types(kNumberOfSimulcastStreams,
>-                                       kVideoFrameDelta);
>-    EXPECT_CALL(
>-        encoder_callback_,
>-        OnEncodedImage(AllOf(Field(&EncodedImage::_frameType, kVideoFrameKey),
>-                             Field(&EncodedImage::_encodedWidth, width),
>-                             Field(&EncodedImage::_encodedHeight, height)),
>-                       _, _))
>-        .Times(1)
>-        .WillRepeatedly(Return(
>-            EncodedImageCallback::Result(EncodedImageCallback::Result::OK, 0)));
>-    EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>-
>-    // Switch back.
>-    DefaultSettings(&settings_, kDefaultTemporalLayerProfile);
>-    // Start at the lowest bitrate for enabling base stream.
>-    settings_.startBitrate = kMinBitrates[0];
>-    SetUpRateAllocator();
>-    EXPECT_EQ(0, encoder_->InitEncode(&settings_, 1, 1200));
>-    SetRates(settings_.startBitrate, 30);
>-    ExpectStreams(kVideoFrameKey, 1);
>-    // Resize |input_frame_| to the new resolution.
>-    input_buffer_ = I420Buffer::Create(settings_.width, settings_.height);
>-    input_buffer_->InitializeData();
>-    input_frame_.reset(
>-        new VideoFrame(input_buffer_, 0, 0, webrtc::kVideoRotation_0));
>-    EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>-  }
>-
>-  void TestSwitchingToOneStream() { SwitchingToOneStream(1024, 768); }
>-
>-  void TestSwitchingToOneOddStream() { SwitchingToOneStream(1023, 769); }
>-
>-  void TestSwitchingToOneSmallStream() { SwitchingToOneStream(4, 4); }
>-
>-  // Test the layer pattern and sync flag for various spatial-temporal patterns.
>-  // 3-3-3 pattern: 3 temporal layers for all spatial streams, so same
>-  // temporal_layer id and layer_sync is expected for all streams.
>-  void TestSaptioTemporalLayers333PatternEncoder() {
>-    Vp8TestEncodedImageCallback encoder_callback;
>-    encoder_->RegisterEncodeCompleteCallback(&encoder_callback);
>-    SetRates(kMaxBitrates[2], 30);  // To get all three streams.
>-
>-    int expected_temporal_idx[3] = {-1, -1, -1};
>-    bool expected_layer_sync[3] = {false, false, false};
>-
>-    // First frame: #0.
>-    EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, NULL));
>-    SetExpectedValues3<int>(0, 0, 0, expected_temporal_idx);
>-    SetExpectedValues3<bool>(true, true, true, expected_layer_sync);
>-    VerifyTemporalIdxAndSyncForAllSpatialLayers(
>-        &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
>-
>-    // Next frame: #1.
>-    input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>-    EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, NULL));
>-    SetExpectedValues3<int>(2, 2, 2, expected_temporal_idx);
>-    SetExpectedValues3<bool>(true, true, true, expected_layer_sync);
>-    VerifyTemporalIdxAndSyncForAllSpatialLayers(
>-        &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
>-
>-    // Next frame: #2.
>-    input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>-    EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, NULL));
>-    SetExpectedValues3<int>(1, 1, 1, expected_temporal_idx);
>-    SetExpectedValues3<bool>(true, true, true, expected_layer_sync);
>-    VerifyTemporalIdxAndSyncForAllSpatialLayers(
>-        &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
>-
>-    // Next frame: #3.
>-    input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>-    EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, NULL));
>-    SetExpectedValues3<int>(2, 2, 2, expected_temporal_idx);
>-    SetExpectedValues3<bool>(false, false, false, expected_layer_sync);
>-    VerifyTemporalIdxAndSyncForAllSpatialLayers(
>-        &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
>-
>-    // Next frame: #4.
>-    input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>-    EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, NULL));
>-    SetExpectedValues3<int>(0, 0, 0, expected_temporal_idx);
>-    SetExpectedValues3<bool>(false, false, false, expected_layer_sync);
>-    VerifyTemporalIdxAndSyncForAllSpatialLayers(
>-        &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
>-
>-    // Next frame: #5.
>-    input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>-    EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, NULL));
>-    SetExpectedValues3<int>(2, 2, 2, expected_temporal_idx);
>-    SetExpectedValues3<bool>(false, false, false, expected_layer_sync);
>-    VerifyTemporalIdxAndSyncForAllSpatialLayers(
>-        &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
>-  }
>-
>-  // Test the layer pattern and sync flag for various spatial-temporal patterns.
>-  // 3-2-1 pattern: 3 temporal layers for lowest resolution, 2 for middle, and
>-  // 1 temporal layer for highest resolution.
>-  // For this profile, we expect the temporal index pattern to be:
>-  // 1st stream: 0, 2, 1, 2, ....
>-  // 2nd stream: 0, 1, 0, 1, ...
>-  // 3rd stream: -1, -1, -1, -1, ....
>-  // Regarding the 3rd stream, note that a stream/encoder with 1 temporal layer
>-  // should always have temporal layer idx set to kNoTemporalIdx = -1.
>-  // Since CodecSpecificInfoVP8.temporalIdx is uint8_t, this will wrap to 255.
>-  // TODO(marpan): Although this seems safe for now, we should fix this.
>-  void TestSpatioTemporalLayers321PatternEncoder() {
>-    int temporal_layer_profile[3] = {3, 2, 1};
>-    SetUpCodec(temporal_layer_profile);
>-    Vp8TestEncodedImageCallback encoder_callback;
>-    encoder_->RegisterEncodeCompleteCallback(&encoder_callback);
>-    SetRates(kMaxBitrates[2], 30);  // To get all three streams.
>-
>-    int expected_temporal_idx[3] = {-1, -1, -1};
>-    bool expected_layer_sync[3] = {false, false, false};
>-
>-    // First frame: #0.
>-    EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, NULL));
>-    SetExpectedValues3<int>(0, 0, 255, expected_temporal_idx);
>-    SetExpectedValues3<bool>(true, true, false, expected_layer_sync);
>-    VerifyTemporalIdxAndSyncForAllSpatialLayers(
>-        &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
>-
>-    // Next frame: #1.
>-    input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>-    EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, NULL));
>-    SetExpectedValues3<int>(2, 1, 255, expected_temporal_idx);
>-    SetExpectedValues3<bool>(true, true, false, expected_layer_sync);
>-    VerifyTemporalIdxAndSyncForAllSpatialLayers(
>-        &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
>-
>-    // Next frame: #2.
>-    input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>-    EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, NULL));
>-    SetExpectedValues3<int>(1, 0, 255, expected_temporal_idx);
>-    SetExpectedValues3<bool>(true, false, false, expected_layer_sync);
>-    VerifyTemporalIdxAndSyncForAllSpatialLayers(
>-        &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
>-
>-    // Next frame: #3.
>-    input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>-    EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, NULL));
>-    SetExpectedValues3<int>(2, 1, 255, expected_temporal_idx);
>-    SetExpectedValues3<bool>(false, false, false, expected_layer_sync);
>-    VerifyTemporalIdxAndSyncForAllSpatialLayers(
>-        &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
>-
>-    // Next frame: #4.
>-    input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>-    EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, NULL));
>-    SetExpectedValues3<int>(0, 0, 255, expected_temporal_idx);
>-    SetExpectedValues3<bool>(false, false, false, expected_layer_sync);
>-    VerifyTemporalIdxAndSyncForAllSpatialLayers(
>-        &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
>-
>-    // Next frame: #5.
>-    input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>-    EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, NULL));
>-    SetExpectedValues3<int>(2, 1, 255, expected_temporal_idx);
>-    SetExpectedValues3<bool>(false, false, false, expected_layer_sync);
>-    VerifyTemporalIdxAndSyncForAllSpatialLayers(
>-        &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
>-  }
>-
>-  void TestStrideEncodeDecode() {
>-    Vp8TestEncodedImageCallback encoder_callback;
>-    Vp8TestDecodedImageCallback decoder_callback;
>-    encoder_->RegisterEncodeCompleteCallback(&encoder_callback);
>-    decoder_->RegisterDecodeCompleteCallback(&decoder_callback);
>-
>-    SetRates(kMaxBitrates[2], 30);  // To get all three streams.
>-    // Setting two (possibly) problematic use cases for stride:
>-    // 1. stride > width 2. stride_y != stride_uv/2
>-    int stride_y = kDefaultWidth + 20;
>-    int stride_uv = ((kDefaultWidth + 1) / 2) + 5;
>-    input_buffer_ = I420Buffer::Create(kDefaultWidth, kDefaultHeight, stride_y,
>-                                       stride_uv, stride_uv);
>-    input_frame_.reset(
>-        new VideoFrame(input_buffer_, 0, 0, webrtc::kVideoRotation_0));
>-
>-    // Set color.
>-    int plane_offset[kNumOfPlanes];
>-    plane_offset[kYPlane] = kColorY;
>-    plane_offset[kUPlane] = kColorU;
>-    plane_offset[kVPlane] = kColorV;
>-    CreateImage(input_buffer_, plane_offset);
>-
>-    EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, NULL));
>-
>-    // Change color.
>-    plane_offset[kYPlane] += 1;
>-    plane_offset[kUPlane] += 1;
>-    plane_offset[kVPlane] += 1;
>-    CreateImage(input_buffer_, plane_offset);
>-    input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>-    EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, NULL));
>-
>-    EncodedImage encoded_frame;
>-    // Only encoding one frame - so will be a key frame.
>-    encoder_callback.GetLastEncodedKeyFrame(&encoded_frame);
>-    EXPECT_EQ(0, decoder_->Decode(encoded_frame, false, NULL));
>-    encoder_callback.GetLastEncodedFrame(&encoded_frame);
>-    decoder_->Decode(encoded_frame, false, NULL);
>-    EXPECT_EQ(2, decoder_callback.DecodedFrames());
>-  }
>-
>-  std::unique_ptr<VP8Encoder> encoder_;
>-  MockEncodedImageCallback encoder_callback_;
>-  std::unique_ptr<VP8Decoder> decoder_;
>-  MockDecodedImageCallback decoder_callback_;
>-  VideoCodec settings_;
>-  rtc::scoped_refptr<I420Buffer> input_buffer_;
>-  std::unique_ptr<VideoFrame> input_frame_;
>-  std::unique_ptr<SimulcastRateAllocator> rate_allocator_;
>-};
>-
>-}  // namespace testing
>-}  // namespace webrtc
>-
>-#endif  // MODULES_VIDEO_CODING_CODECS_VP8_SIMULCAST_TEST_UTILITY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/simulcast_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/simulcast_unittest.cc
>deleted file mode 100644
>index b1dd794f291..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/simulcast_unittest.cc
>+++ /dev/null
>@@ -1,78 +0,0 @@
>-/*
>- *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "modules/video_coding/codecs/vp8/simulcast_test_utility.h"
>-
>-namespace webrtc {
>-namespace testing {
>-
>-class TestVp8Impl : public TestVp8Simulcast {
>- protected:
>-  std::unique_ptr<VP8Encoder> CreateEncoder() override {
>-    return VP8Encoder::Create();
>-  }
>-  std::unique_ptr<VP8Decoder> CreateDecoder() override {
>-    return VP8Decoder::Create();
>-  }
>-};
>-
>-TEST_F(TestVp8Impl, TestKeyFrameRequestsOnAllStreams) {
>-  TestVp8Simulcast::TestKeyFrameRequestsOnAllStreams();
>-}
>-
>-TEST_F(TestVp8Impl, TestPaddingAllStreams) {
>-  TestVp8Simulcast::TestPaddingAllStreams();
>-}
>-
>-TEST_F(TestVp8Impl, TestPaddingTwoStreams) {
>-  TestVp8Simulcast::TestPaddingTwoStreams();
>-}
>-
>-TEST_F(TestVp8Impl, TestPaddingTwoStreamsOneMaxedOut) {
>-  TestVp8Simulcast::TestPaddingTwoStreamsOneMaxedOut();
>-}
>-
>-TEST_F(TestVp8Impl, TestPaddingOneStream) {
>-  TestVp8Simulcast::TestPaddingOneStream();
>-}
>-
>-TEST_F(TestVp8Impl, TestPaddingOneStreamTwoMaxedOut) {
>-  TestVp8Simulcast::TestPaddingOneStreamTwoMaxedOut();
>-}
>-
>-TEST_F(TestVp8Impl, TestSendAllStreams) {
>-  TestVp8Simulcast::TestSendAllStreams();
>-}
>-
>-TEST_F(TestVp8Impl, TestDisablingStreams) {
>-  TestVp8Simulcast::TestDisablingStreams();
>-}
>-
>-TEST_F(TestVp8Impl, TestSwitchingToOneStream) {
>-  TestVp8Simulcast::TestSwitchingToOneStream();
>-}
>-
>-TEST_F(TestVp8Impl, TestSwitchingToOneOddStream) {
>-  TestVp8Simulcast::TestSwitchingToOneOddStream();
>-}
>-
>-TEST_F(TestVp8Impl, TestSwitchingToOneSmallStream) {
>-  TestVp8Simulcast::TestSwitchingToOneSmallStream();
>-}
>-
>-TEST_F(TestVp8Impl, TestSaptioTemporalLayers333PatternEncoder) {
>-  TestVp8Simulcast::TestSaptioTemporalLayers333PatternEncoder();
>-}
>-
>-TEST_F(TestVp8Impl, TestStrideEncodeDecode) {
>-  TestVp8Simulcast::TestStrideEncodeDecode();
>-}
>-}  // namespace testing
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/temporal_layers.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/temporal_layers.cc
>index 0a9b9931291..236950ad3d4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/temporal_layers.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/temporal_layers.cc
>@@ -14,17 +14,79 @@
> #include <set>
> #include <vector>
> 
>+#include "absl/memory/memory.h"
> #include "modules/include/module_common_types.h"
>-#include "modules/video_coding/codecs/vp8/include/vp8_common_types.h"
>+#include "modules/video_coding/codecs/vp8/default_temporal_layers.h"
>+#include "modules/video_coding/codecs/vp8/screenshare_layers.h"
> #include "modules/video_coding/include/video_codec_interface.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>+#include "system_wrappers/include/clock.h"
> #include "system_wrappers/include/field_trial.h"
> 
> namespace webrtc {
>+namespace {
>+uint8_t NumTemporalLayers(const VideoCodec& codec, int spatial_id) {
>+  uint8_t num_temporal_layers =
>+      std::max<uint8_t>(1, codec.VP8().numberOfTemporalLayers);
>+  if (codec.numberOfSimulcastStreams > 0) {
>+    RTC_DCHECK_LT(spatial_id, codec.numberOfSimulcastStreams);
>+    num_temporal_layers =
>+        std::max(num_temporal_layers,
>+                 codec.simulcastStream[spatial_id].numberOfTemporalLayers);
>+  }
>+  return num_temporal_layers;
>+}
>+
>+bool IsConferenceModeScreenshare(const VideoCodec& codec) {
>+  if (codec.mode != VideoCodecMode::kScreensharing ||
>+      NumTemporalLayers(codec, 0) != 2) {
>+    return false;
>+  }
>+  // Fixed default bitrates for legacy screenshare layers mode.
>+  return (codec.numberOfSimulcastStreams == 0 && codec.maxBitrate == 1000) ||
>+         (codec.numberOfSimulcastStreams >= 1 &&
>+          codec.simulcastStream[0].maxBitrate == 1000 &&
>+          codec.simulcastStream[0].targetBitrate == 200);
>+}
>+}  // namespace
>+
>+bool TemporalLayers::FrameConfig::operator==(const FrameConfig& o) const {
>+  return drop_frame == o.drop_frame &&
>+         last_buffer_flags == o.last_buffer_flags &&
>+         golden_buffer_flags == o.golden_buffer_flags &&
>+         arf_buffer_flags == o.arf_buffer_flags && layer_sync == o.layer_sync &&
>+         freeze_entropy == o.freeze_entropy &&
>+         encoder_layer_id == o.encoder_layer_id &&
>+         packetizer_temporal_idx == o.packetizer_temporal_idx;
>+}
>+
>+std::unique_ptr<TemporalLayers> TemporalLayers::CreateTemporalLayers(
>+    const VideoCodec& codec,
>+    size_t spatial_id) {
>+  if (IsConferenceModeScreenshare(codec) && spatial_id == 0) {
>+    // Conference mode temporal layering for screen content in base stream.
>+    return absl::make_unique<ScreenshareLayers>(2, Clock::GetRealTimeClock());
>+  }
>+
>+  return absl::make_unique<DefaultTemporalLayers>(
>+      NumTemporalLayers(codec, spatial_id));
>+}
>+
>+std::unique_ptr<TemporalLayersChecker>
>+TemporalLayers::CreateTemporalLayersChecker(const VideoCodec& codec,
>+                                            size_t spatial_id) {
>+  if (IsConferenceModeScreenshare(codec) && spatial_id == 0) {
>+    // Conference mode temporal layering for screen content in base stream,
>+    // use generic checker.
>+    return absl::make_unique<TemporalLayersChecker>(2);
>+  }
>+
>+  return absl::make_unique<DefaultTemporalLayersChecker>(
>+      NumTemporalLayers(codec, spatial_id));
>+}
> 
>-TemporalLayersChecker::TemporalLayersChecker(int num_temporal_layers,
>-                                             uint8_t /*initial_tl0_pic_idx*/)
>+TemporalLayersChecker::TemporalLayersChecker(int num_temporal_layers)
>     : num_temporal_layers_(num_temporal_layers),
>       sequence_number_(0),
>       last_sync_sequence_number_(0),
>@@ -39,7 +101,7 @@ bool TemporalLayersChecker::CheckAndUpdateBufferState(
>     uint32_t sequence_number,
>     uint32_t* lowest_sequence_referenced) {
>   if (flags & TemporalLayers::BufferFlags::kReference) {
>-    if (state->temporal_layer > 0) {
>+    if (state->temporal_layer > 0 && !state->is_keyframe) {
>       *need_sync = false;
>     }
>     if (!state->is_keyframe && !frame_is_keyframe &&
>@@ -125,7 +187,8 @@ bool TemporalLayersChecker::CheckTemporalConfig(
>     last_sync_sequence_number_ = last_tl0_sequence_number_;
>   }
> 
>-  if (need_sync != frame_config.layer_sync) {
>+  // Ignore sync flag on key-frames as it really doesn't matter.
>+  if (need_sync != frame_config.layer_sync && !frame_is_keyframe) {
>     RTC_LOG(LS_ERROR) << "Sync bit is set incorrectly on a frame. Expected: "
>                       << need_sync << " Actual: " << frame_config.layer_sync;
>     return false;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/temporal_layers.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/temporal_layers.h
>index dad718cd0f3..20da1a44fad 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/temporal_layers.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/temporal_layers.h
>@@ -15,17 +15,70 @@
> #include <vector>
> #include <memory>
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>-struct vpx_codec_enc_cfg;
>-typedef struct vpx_codec_enc_cfg vpx_codec_enc_cfg_t;
>+#include "api/video_codecs/video_codec.h"
>+
>+#define VP8_TS_MAX_PERIODICITY 16
>+#define VP8_TS_MAX_LAYERS 5
> 
> namespace webrtc {
> 
>+// Some notes on the prerequisites of the TemporalLayers interface.
>+// * Implementations of TemporalLayers may not contain internal synchronization
>+//   so caller must make sure doing so thread safe.
>+// * The encoder is assumed to encode all frames in order, and callbacks to
>+//   PopulateCodecSpecific() / FrameEncoded() must happen in the same order.
>+//
>+// This means that in the case of pipelining encoders, it is OK to have a chain
>+// of calls such as this:
>+// - UpdateLayerConfig(timestampA)
>+// - UpdateLayerConfig(timestampB)
>+// - PopulateCodecSpecific(timestampA, ...)
>+// - UpdateLayerConfig(timestampC)
>+// - FrameEncoded(timestampA, 1234, ...)
>+// - FrameEncoded(timestampB, 0, ...)
>+// - PopulateCodecSpecific(timestampC, ...)
>+// - FrameEncoded(timestampC, 1234, ...)
>+// Note that UpdateLayerConfig() for a new frame can happen before
>+// FrameEncoded() for a previous one, but calls themselves must be both
>+// synchronized (e.g. run on a task queue) and in order (per type).
>+
> struct CodecSpecificInfoVP8;
>+enum class Vp8BufferReference : uint8_t {
>+  kNone = 0,
>+  kLast = 1,
>+  kGolden = 2,
>+  kAltref = 4
>+};
> 
>+struct Vp8EncoderConfig {
>+  // Number of active temporal layers. Set to 0 if not used.
>+  unsigned int ts_number_layers;
>+  // Arrays of length |ts_number_layers|, indicating (cumulative) target bitrate
>+  // and rate decimator (e.g. 4 if every 4th frame is in the given layer) for
>+  // each active temporal layer, starting with temporal id 0.
>+  unsigned int ts_target_bitrate[VP8_TS_MAX_LAYERS];
>+  unsigned int ts_rate_decimator[VP8_TS_MAX_LAYERS];
>+
>+  // The periodicity of the temporal pattern. Set to 0 if not used.
>+  unsigned int ts_periodicity;
>+  // Array of length |ts_periodicity| indicating the sequence of temporal id's
>+  // to assign to incoming frames.
>+  unsigned int ts_layer_id[VP8_TS_MAX_PERIODICITY];
>+
>+  // Target bitrate, in bps.
>+  unsigned int rc_target_bitrate;
>+
>+  // Clamp QP to min/max. Use 0 to disable clamping.
>+  unsigned int rc_min_quantizer;
>+  unsigned int rc_max_quantizer;
>+};
>+
>+// This interface defines a way of getting the encoder settings needed to
>+// realize a temporal layer structure of predefined size.
>+class TemporalLayersChecker;
> class TemporalLayers {
>  public:
>-  enum BufferFlags {
>+  enum BufferFlags : int {
>     kNone = 0,
>     kReference = 1,
>     kUpdate = 2,
>@@ -65,15 +118,15 @@ class TemporalLayers {
> 
>     bool freeze_entropy;
> 
>-    bool operator==(const FrameConfig& o) const {
>-      return drop_frame == o.drop_frame &&
>-             last_buffer_flags == o.last_buffer_flags &&
>-             golden_buffer_flags == o.golden_buffer_flags &&
>-             arf_buffer_flags == o.arf_buffer_flags &&
>-             layer_sync == o.layer_sync && freeze_entropy == o.freeze_entropy &&
>-             encoder_layer_id == o.encoder_layer_id &&
>-             packetizer_temporal_idx == o.packetizer_temporal_idx;
>-    }
>+    // Indicates in which order the encoder should search the reference buffers
>+    // when doing motion prediction. Set to kNone to use unspecified order. Any
>+    // buffer indicated here must not have the corresponding no_ref bit set.
>+    // If all three buffers can be reference, the one not listed here should be
>+    // searched last.
>+    Vp8BufferReference first_reference;
>+    Vp8BufferReference second_reference;
>+
>+    bool operator==(const FrameConfig& o) const;
>     bool operator!=(const FrameConfig& o) const { return !(*this == o); }
> 
>    private:
>@@ -85,85 +138,52 @@ class TemporalLayers {
> 
>   // Factory for TemporalLayer strategy. Default behavior is a fixed pattern
>   // of temporal layers. See default_temporal_layers.cc
>-  virtual ~TemporalLayers() {}
>+  static std::unique_ptr<TemporalLayers> CreateTemporalLayers(
>+      const VideoCodec& codec,
>+      size_t spatial_id);
>+  static std::unique_ptr<TemporalLayersChecker> CreateTemporalLayersChecker(
>+      const VideoCodec& codec,
>+      size_t spatial_id);
> 
>-  // Returns the recommended VP8 encode flags needed. May refresh the decoder
>-  // and/or update the reference buffers.
>-  virtual FrameConfig UpdateLayerConfig(uint32_t timestamp) = 0;
>+  virtual ~TemporalLayers() = default;
> 
>-  // Update state based on new bitrate target and incoming framerate.
>-  // Returns the bitrate allocation for the active temporal layers.
>-  virtual std::vector<uint32_t> OnRatesUpdated(int bitrate_kbps,
>-                                               int max_bitrate_kbps,
>-                                               int framerate) = 0;
>+  // New target bitrate, per temporal layer.
>+  virtual void OnRatesUpdated(const std::vector<uint32_t>& bitrates_bps,
>+                              int framerate_fps) = 0;
> 
>   // Update the encoder configuration with target bitrates or other parameters.
>   // Returns true iff the configuration was actually modified.
>-  virtual bool UpdateConfiguration(vpx_codec_enc_cfg_t* cfg) = 0;
>-
>+  virtual bool UpdateConfiguration(Vp8EncoderConfig* cfg) = 0;
>+
>+  // Returns the recommended VP8 encode flags needed, and moves the temporal
>+  // pattern to the next frame.
>+  // The timestamp may be used as both a time and a unique identifier, and so
>+  // the caller must make sure no two frames use the same timestamp.
>+  // The timestamp uses a 90kHz RTP clock.
>+  // After calling this method, the actual encoder should be called with the
>+  // provided frame configuration, after which:
>+  // * On success, call PopulateCodecSpecific() and then FrameEncoded();
>+  // * On failure/ frame drop: Call FrameEncoded() with size = 0.
>+  virtual FrameConfig UpdateLayerConfig(uint32_t rtp_timestamp) = 0;
>+
>+  // Called after successful encoding of a frame. The rtp timestamp must match
>+  // the one using in UpdateLayerConfig(). Some fields in |vp8_info| may have
>+  // already been populated by the encoder, check before overwriting.
>+  // |tl_config| is the frame config returned by UpdateLayerConfig() for this
>+  // rtp_timestamp;
>+  // If |is_keyframe| is true, the flags in |tl_config| will be ignored.
>   virtual void PopulateCodecSpecific(
>       bool is_keyframe,
>       const TemporalLayers::FrameConfig& tl_config,
>       CodecSpecificInfoVP8* vp8_info,
>-      uint32_t timestamp) = 0;
>-
>-  virtual void FrameEncoded(unsigned int size, int qp) = 0;
>-
>-  // Returns the current tl0_pic_idx, so it can be reused in future
>-  // instantiations.
>-  virtual uint8_t Tl0PicIdx() const = 0;
>-};
>-
>-class TemporalLayersListener;
>-class TemporalLayersChecker;
>-
>-class TemporalLayersFactory {
>- public:
>-  TemporalLayersFactory() : listener_(nullptr) {}
>-  virtual ~TemporalLayersFactory() {}
>-  virtual TemporalLayers* Create(int simulcast_id,
>-                                 int temporal_layers,
>-                                 uint8_t initial_tl0_pic_idx) const;
>-
>-  // Creates helper class which performs online checks of a correctness of
>-  // temporal layers dependencies returned by TemporalLayers class created in
>-  // the same factory.
>-  virtual std::unique_ptr<TemporalLayersChecker> CreateChecker(
>-      int simulcast_id,
>-      int temporal_layers,
>-      uint8_t initial_tl0_pic_idx) const;
>-
>-  void SetListener(TemporalLayersListener* listener);
>-
>- protected:
>-  TemporalLayersListener* listener_;
>-};
>-
>-class ScreenshareTemporalLayersFactory : public webrtc::TemporalLayersFactory {
>- public:
>-  ScreenshareTemporalLayersFactory() {}
>-  virtual ~ScreenshareTemporalLayersFactory() {}
>-
>-  webrtc::TemporalLayers* Create(int simulcast_id,
>-                                 int num_temporal_layers,
>-                                 uint8_t initial_tl0_pic_idx) const override;
>-
>-  // Creates helper class which performs online checks of a correctness of
>-  // temporal layers dependencies returned by TemporalLayers class created in
>-  // the same factory.
>-  std::unique_ptr<webrtc::TemporalLayersChecker> CreateChecker(
>-      int simulcast_id,
>-      int temporal_layers,
>-      uint8_t initial_tl0_pic_idx) const override;
>-};
>-
>-class TemporalLayersListener {
>- public:
>-  TemporalLayersListener() {}
>-  virtual ~TemporalLayersListener() {}
>-
>-  virtual void OnTemporalLayersCreated(int simulcast_id,
>-                                       TemporalLayers* layers) = 0;
>+      uint32_t rtp_timestamp) = 0;
>+
>+  // Called after an encode event. If the frame was dropped, |size_bytes| must
>+  // be set to 0. The rtp timestamp must match the one using in
>+  // UpdateLayerConfig()
>+  virtual void FrameEncoded(uint32_t rtp_timestamp,
>+                            size_t size_bytes,
>+                            int qp) = 0;
> };
> 
> // Used only inside RTC_DCHECK(). It checks correctness of temporal layers
>@@ -171,7 +191,7 @@ class TemporalLayersListener {
> // each UpdateLayersConfig() of a corresponding TemporalLayers class.
> class TemporalLayersChecker {
>  public:
>-  TemporalLayersChecker(int num_temporal_layers, uint8_t initial_tl0_pic_idx);
>+  explicit TemporalLayersChecker(int num_temporal_layers);
>   virtual ~TemporalLayersChecker() {}
> 
>   virtual bool CheckTemporalConfig(
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc
>index ad1003095af..a3d7cd985a7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc
>@@ -12,18 +12,12 @@
> 
> #include <memory>
> 
>-#include "api/optional.h"
>-#include "api/video/i420_buffer.h"
> #include "common_video/libyuv/include/webrtc_libyuv.h"
>-#include "modules/video_coding/codecs/test/video_codec_test.h"
>+#include "modules/video_coding/codecs/test/video_codec_unittest.h"
> #include "modules/video_coding/codecs/vp8/include/vp8.h"
> #include "modules/video_coding/codecs/vp8/temporal_layers.h"
>-#include "rtc_base/checks.h"
>+#include "modules/video_coding/utility/vp8_header_parser.h"
> #include "rtc_base/timeutils.h"
>-#include "test/field_trial.h"
>-#include "test/frame_utils.h"
>-#include "test/gtest.h"
>-#include "test/testsupport/fileutils.h"
> #include "test/video_codec_settings.h"
> 
> namespace webrtc {
>@@ -32,199 +26,72 @@ namespace {
> constexpr uint32_t kInitialTimestampRtp = 123;
> constexpr int64_t kTestNtpTimeMs = 456;
> constexpr int64_t kInitialTimestampMs = 789;
>-constexpr uint32_t kTimestampIncrement = 3000;
> constexpr int kNumCores = 1;
> constexpr size_t kMaxPayloadSize = 1440;
> constexpr int kDefaultMinPixelsPerFrame = 320 * 180;
> constexpr int kWidth = 172;
> constexpr int kHeight = 144;
>-
>-void Calc16ByteAlignedStride(int width, int* stride_y, int* stride_uv) {
>-  *stride_y = 16 * ((width + 15) / 16);
>-  *stride_uv = 16 * ((width + 31) / 32);
>-}
> }  // namespace
> 
>-class EncodedImageCallbackTestImpl : public webrtc::EncodedImageCallback {
>- public:
>-  Result OnEncodedImage(const EncodedImage& encoded_frame,
>-                        const CodecSpecificInfo* codec_specific_info,
>-                        const RTPFragmentationHeader* fragmentation) override {
>-    EXPECT_GT(encoded_frame._length, 0u);
>-    VerifyQpParser(encoded_frame);
>-
>-    if (encoded_frame_._size != encoded_frame._size) {
>-      delete[] encoded_frame_._buffer;
>-      frame_buffer_.reset(new uint8_t[encoded_frame._size]);
>-    }
>-    RTC_DCHECK(frame_buffer_);
>-    memcpy(frame_buffer_.get(), encoded_frame._buffer, encoded_frame._length);
>-    encoded_frame_ = encoded_frame;
>-    encoded_frame_._buffer = frame_buffer_.get();
>-
>-    // Skip |codec_name|, to avoid allocating.
>-    EXPECT_STREQ("libvpx", codec_specific_info->codec_name);
>-    EXPECT_EQ(kVideoCodecVP8, codec_specific_info->codecType);
>-    EXPECT_EQ(0u, codec_specific_info->codecSpecific.VP8.simulcastIdx);
>-    codec_specific_info_.codecType = codec_specific_info->codecType;
>-    codec_specific_info_.codecSpecific = codec_specific_info->codecSpecific;
>-    complete_ = true;
>-    return Result(Result::OK, 0);
>-  }
>-
>-  void VerifyQpParser(const EncodedImage& encoded_frame) const {
>-    int qp;
>-    ASSERT_TRUE(vp8::GetQp(encoded_frame._buffer, encoded_frame._length, &qp));
>-    EXPECT_EQ(encoded_frame.qp_, qp) << "Encoder QP != parsed bitstream QP.";
>-  }
>-
>-  bool EncodeComplete() {
>-    if (complete_) {
>-      complete_ = false;
>-      return true;
>-    }
>-    return false;
>-  }
>-
>-  EncodedImage encoded_frame_;
>-  CodecSpecificInfo codec_specific_info_;
>-  std::unique_ptr<uint8_t[]> frame_buffer_;
>-  bool complete_ = false;
>-};
>-
>-class DecodedImageCallbackTestImpl : public webrtc::DecodedImageCallback {
>- public:
>-  int32_t Decoded(VideoFrame& frame) override {
>-    RTC_NOTREACHED();
>-    return -1;
>-  }
>-  int32_t Decoded(VideoFrame& frame, int64_t decode_time_ms) override {
>-    RTC_NOTREACHED();
>-    return -1;
>-  }
>-  void Decoded(VideoFrame& frame,
>-               rtc::Optional<int32_t> decode_time_ms,
>-               rtc::Optional<uint8_t> qp) override {
>-    EXPECT_GT(frame.width(), 0);
>-    EXPECT_GT(frame.height(), 0);
>-    EXPECT_TRUE(qp);
>-    frame_ = frame;
>-    qp_ = qp;
>-    complete_ = true;
>-  }
>-
>-  bool DecodeComplete() {
>-    if (complete_) {
>-      complete_ = false;
>-      return true;
>-    }
>-    return false;
>-  }
>-
>-  rtc::Optional<VideoFrame> frame_;
>-  rtc::Optional<uint8_t> qp_;
>-  bool complete_ = false;
>-};
>-
>-class TestVp8Impl : public ::testing::Test {
>- public:
>-  TestVp8Impl() : TestVp8Impl("") {}
>-  explicit TestVp8Impl(const std::string& field_trials)
>-      : override_field_trials_(field_trials),
>-        encoder_(VP8Encoder::Create()),
>-        decoder_(VP8Decoder::Create()) {}
>-  virtual ~TestVp8Impl() {}
>-
>+class TestVp8Impl : public VideoCodecUnitTest {
>  protected:
>-  virtual void SetUp() {
>-    encoder_->RegisterEncodeCompleteCallback(&encoded_cb_);
>-    decoder_->RegisterDecodeCompleteCallback(&decoded_cb_);
>-    SetupCodecSettings();
>-    SetupInputFrame();
>+  std::unique_ptr<VideoEncoder> CreateEncoder() override {
>+    return VP8Encoder::Create();
>   }
> 
>-  void SetupInputFrame() {
>-    // Using a QCIF image (aligned stride (u,v planes) > width).
>-    // Processing only one frame.
>-    FILE* file = fopen(test::ResourcePath("paris_qcif", "yuv").c_str(), "rb");
>-    ASSERT_TRUE(file != nullptr);
>-    rtc::scoped_refptr<I420BufferInterface> compact_buffer(
>-        test::ReadI420Buffer(kWidth, kHeight, file));
>-    ASSERT_TRUE(compact_buffer);
>-
>-    // Setting aligned stride values.
>-    int stride_uv;
>-    int stride_y;
>-    Calc16ByteAlignedStride(kWidth, &stride_y, &stride_uv);
>-    EXPECT_EQ(stride_y, 176);
>-    EXPECT_EQ(stride_uv, 96);
>-    rtc::scoped_refptr<I420Buffer> stride_buffer(
>-        I420Buffer::Create(kWidth, kHeight, stride_y, stride_uv, stride_uv));
>-
>-    // No scaling in our case, just a copy, to add stride to the image.
>-    stride_buffer->ScaleFrom(*compact_buffer);
>-
>-    input_frame_.reset(new VideoFrame(stride_buffer, kInitialTimestampRtp,
>-                                      kInitialTimestampMs, kVideoRotation_0));
>-    fclose(file);
>+  std::unique_ptr<VideoDecoder> CreateDecoder() override {
>+    return VP8Decoder::Create();
>   }
> 
>-  void SetupCodecSettings() {
>-    webrtc::test::CodecSettings(kVideoCodecVP8, &codec_settings_);
>-    codec_settings_.maxBitrate = 4000;
>-    codec_settings_.width = kWidth;
>-    codec_settings_.height = kHeight;
>-    codec_settings_.VP8()->denoisingOn = true;
>-    codec_settings_.VP8()->frameDroppingOn = false;
>-    codec_settings_.VP8()->automaticResizeOn = false;
>-    codec_settings_.VP8()->complexity = kComplexityNormal;
>-    codec_settings_.VP8()->tl_factory = &tl_factory_;
>+  void ModifyCodecSettings(VideoCodec* codec_settings) override {
>+    webrtc::test::CodecSettings(kVideoCodecVP8, codec_settings);
>+    codec_settings->width = kWidth;
>+    codec_settings->height = kHeight;
>+    codec_settings->VP8()->denoisingOn = true;
>+    codec_settings->VP8()->frameDroppingOn = false;
>+    codec_settings->VP8()->automaticResizeOn = false;
>+    codec_settings->VP8()->complexity = VideoCodecComplexity::kComplexityNormal;
>   }
> 
>-  void InitEncodeDecode() {
>-    EXPECT_EQ(
>-        WEBRTC_VIDEO_CODEC_OK,
>-        encoder_->InitEncode(&codec_settings_, kNumCores, kMaxPayloadSize));
>+  void EncodeAndWaitForFrame(const VideoFrame& input_frame,
>+                             EncodedImage* encoded_frame,
>+                             CodecSpecificInfo* codec_specific_info) {
>     EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>-              decoder_->InitDecode(&codec_settings_, kNumCores));
>+              encoder_->Encode(input_frame, nullptr, nullptr));
>+    ASSERT_TRUE(WaitForEncodedFrame(encoded_frame, codec_specific_info));
>+    VerifyQpParser(*encoded_frame);
>+    EXPECT_STREQ("libvpx", codec_specific_info->codec_name);
>+    EXPECT_EQ(kVideoCodecVP8, codec_specific_info->codecType);
>+    EXPECT_EQ(0u, codec_specific_info->codecSpecific.VP8.simulcastIdx);
>   }
> 
>-  void EncodeFrame() {
>-    EXPECT_FALSE(encoded_cb_.EncodeComplete());
>-    EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>-              encoder_->Encode(*input_frame_, nullptr, nullptr));
>-    EXPECT_TRUE(encoded_cb_.EncodeComplete());
>+  void EncodeAndExpectFrameWith(const VideoFrame& input_frame,
>+                                uint8_t temporal_idx) {
>+    EncodedImage encoded_frame;
>+    CodecSpecificInfo codec_specific_info;
>+    EncodeAndWaitForFrame(input_frame, &encoded_frame, &codec_specific_info);
>+    EXPECT_EQ(temporal_idx, codec_specific_info.codecSpecific.VP8.temporalIdx);
>   }
> 
>-  void ExpectFrameWith(int16_t picture_id,
>-                       int tl0_pic_idx,
>-                       uint8_t temporal_idx) {
>-    EXPECT_EQ(picture_id % (1 << 15),
>-              encoded_cb_.codec_specific_info_.codecSpecific.VP8.pictureId);
>-    EXPECT_EQ(tl0_pic_idx % (1 << 8),
>-              encoded_cb_.codec_specific_info_.codecSpecific.VP8.tl0PicIdx);
>-    EXPECT_EQ(temporal_idx,
>-              encoded_cb_.codec_specific_info_.codecSpecific.VP8.temporalIdx);
>+  void VerifyQpParser(const EncodedImage& encoded_frame) const {
>+    int qp;
>+    EXPECT_GT(encoded_frame._length, 0u);
>+    ASSERT_TRUE(vp8::GetQp(encoded_frame._buffer, encoded_frame._length, &qp));
>+    EXPECT_EQ(encoded_frame.qp_, qp) << "Encoder QP != parsed bitstream QP.";
>   }
>-
>-  test::ScopedFieldTrials override_field_trials_;
>-  EncodedImageCallbackTestImpl encoded_cb_;
>-  DecodedImageCallbackTestImpl decoded_cb_;
>-  std::unique_ptr<VideoFrame> input_frame_;
>-  const std::unique_ptr<VideoEncoder> encoder_;
>-  const std::unique_ptr<VideoDecoder> decoder_;
>-  VideoCodec codec_settings_;
>-  TemporalLayersFactory tl_factory_;
> };
> 
> TEST_F(TestVp8Impl, SetRateAllocation) {
>+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Release());
>+
>   const int kBitrateBps = 300000;
>-  BitrateAllocation bitrate_allocation;
>+  VideoBitrateAllocation bitrate_allocation;
>   bitrate_allocation.SetBitrate(0, 0, kBitrateBps);
>   EXPECT_EQ(WEBRTC_VIDEO_CODEC_UNINITIALIZED,
>             encoder_->SetRateAllocation(bitrate_allocation,
>                                         codec_settings_.maxFramerate));
>-  InitEncodeDecode();
>+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>+            encoder_->InitEncode(&codec_settings_, kNumCores, kMaxPayloadSize));
>   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>             encoder_->SetRateAllocation(bitrate_allocation,
>                                         codec_settings_.maxFramerate));
>@@ -234,10 +101,15 @@ TEST_F(TestVp8Impl, EncodeFrameAndRelease) {
>   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Release());
>   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>             encoder_->InitEncode(&codec_settings_, kNumCores, kMaxPayloadSize));
>-  EncodeFrame();
>+
>+  EncodedImage encoded_frame;
>+  CodecSpecificInfo codec_specific_info;
>+  EncodeAndWaitForFrame(*NextInputFrame(), &encoded_frame,
>+                        &codec_specific_info);
>+
>   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Release());
>   EXPECT_EQ(WEBRTC_VIDEO_CODEC_UNINITIALIZED,
>-            encoder_->Encode(*input_frame_, nullptr, nullptr));
>+            encoder_->Encode(*NextInputFrame(), nullptr, nullptr));
> }
> 
> TEST_F(TestVp8Impl, InitDecode) {
>@@ -247,13 +119,18 @@ TEST_F(TestVp8Impl, InitDecode) {
> }
> 
> TEST_F(TestVp8Impl, OnEncodedImageReportsInfo) {
>-  InitEncodeDecode();
>-  EncodeFrame();
>-  EXPECT_EQ(kInitialTimestampRtp, encoded_cb_.encoded_frame_._timeStamp);
>-  EXPECT_EQ(kInitialTimestampMs, encoded_cb_.encoded_frame_.capture_time_ms_);
>-  EXPECT_EQ(kWidth, static_cast<int>(encoded_cb_.encoded_frame_._encodedWidth));
>-  EXPECT_EQ(kHeight,
>-            static_cast<int>(encoded_cb_.encoded_frame_._encodedHeight));
>+  VideoFrame* input_frame = NextInputFrame();
>+  input_frame->set_timestamp(kInitialTimestampRtp);
>+  input_frame->set_timestamp_us(kInitialTimestampMs *
>+                                rtc::kNumMicrosecsPerMillisec);
>+  EncodedImage encoded_frame;
>+  CodecSpecificInfo codec_specific_info;
>+  EncodeAndWaitForFrame(*input_frame, &encoded_frame, &codec_specific_info);
>+
>+  EXPECT_EQ(kInitialTimestampRtp, encoded_frame._timeStamp);
>+  EXPECT_EQ(kInitialTimestampMs, encoded_frame.capture_time_ms_);
>+  EXPECT_EQ(kWidth, static_cast<int>(encoded_frame._encodedWidth));
>+  EXPECT_EQ(kHeight, static_cast<int>(encoded_frame._encodedHeight));
> }
> 
> // We only test the encoder here, since the decoded frame rotation is set based
>@@ -261,26 +138,36 @@ TEST_F(TestVp8Impl, OnEncodedImageReportsInfo) {
> // TODO(brandtr): Consider passing through the rotation flag through the decoder
> // in the same way as done in the encoder.
> TEST_F(TestVp8Impl, EncodedRotationEqualsInputRotation) {
>-  InitEncodeDecode();
>-  input_frame_->set_rotation(kVideoRotation_0);
>-  EncodeFrame();
>-  EXPECT_EQ(kVideoRotation_0, encoded_cb_.encoded_frame_.rotation_);
>-
>-  input_frame_->set_rotation(kVideoRotation_90);
>-  EncodeFrame();
>-  EXPECT_EQ(kVideoRotation_90, encoded_cb_.encoded_frame_.rotation_);
>+  VideoFrame* input_frame = NextInputFrame();
>+  input_frame->set_rotation(kVideoRotation_0);
>+
>+  EncodedImage encoded_frame;
>+  CodecSpecificInfo codec_specific_info;
>+  EncodeAndWaitForFrame(*input_frame, &encoded_frame, &codec_specific_info);
>+  EXPECT_EQ(kVideoRotation_0, encoded_frame.rotation_);
>+
>+  input_frame->set_rotation(kVideoRotation_90);
>+  EncodeAndWaitForFrame(*input_frame, &encoded_frame, &codec_specific_info);
>+  EXPECT_EQ(kVideoRotation_90, encoded_frame.rotation_);
> }
> 
> TEST_F(TestVp8Impl, DecodedQpEqualsEncodedQp) {
>-  InitEncodeDecode();
>-  EncodeFrame();
>+  VideoFrame* input_frame = NextInputFrame();
>+  EncodedImage encoded_frame;
>+  CodecSpecificInfo codec_specific_info;
>+  EncodeAndWaitForFrame(*input_frame, &encoded_frame, &codec_specific_info);
>+
>   // First frame should be a key frame.
>-  encoded_cb_.encoded_frame_._frameType = kVideoFrameKey;
>+  encoded_frame._frameType = kVideoFrameKey;
>   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>-            decoder_->Decode(encoded_cb_.encoded_frame_, false, nullptr));
>-  EXPECT_TRUE(decoded_cb_.DecodeComplete());
>-  EXPECT_GT(I420PSNR(input_frame_.get(), &*decoded_cb_.frame_), 36);
>-  EXPECT_EQ(encoded_cb_.encoded_frame_.qp_, *decoded_cb_.qp_);
>+            decoder_->Decode(encoded_frame, false, nullptr, -1));
>+  std::unique_ptr<VideoFrame> decoded_frame;
>+  absl::optional<uint8_t> decoded_qp;
>+  ASSERT_TRUE(WaitForDecodedFrame(&decoded_frame, &decoded_qp));
>+  ASSERT_TRUE(decoded_frame);
>+  ASSERT_TRUE(decoded_qp);
>+  EXPECT_GT(I420PSNR(input_frame, decoded_frame.get()), 36);
>+  EXPECT_EQ(encoded_frame.qp_, *decoded_qp);
> }
> 
> TEST_F(TestVp8Impl, ChecksSimulcastSettings) {
>@@ -347,18 +234,28 @@ TEST_F(TestVp8Impl, ChecksSimulcastSettings) {
> #define MAYBE_AlignedStrideEncodeDecode AlignedStrideEncodeDecode
> #endif
> TEST_F(TestVp8Impl, MAYBE_AlignedStrideEncodeDecode) {
>-  InitEncodeDecode();
>-  EncodeFrame();
>+  VideoFrame* input_frame = NextInputFrame();
>+  input_frame->set_timestamp(kInitialTimestampRtp);
>+  input_frame->set_timestamp_us(kInitialTimestampMs *
>+                                rtc::kNumMicrosecsPerMillisec);
>+  EncodedImage encoded_frame;
>+  CodecSpecificInfo codec_specific_info;
>+  EncodeAndWaitForFrame(*input_frame, &encoded_frame, &codec_specific_info);
>+
>   // First frame should be a key frame.
>-  encoded_cb_.encoded_frame_._frameType = kVideoFrameKey;
>-  encoded_cb_.encoded_frame_.ntp_time_ms_ = kTestNtpTimeMs;
>+  encoded_frame._frameType = kVideoFrameKey;
>+  encoded_frame.ntp_time_ms_ = kTestNtpTimeMs;
>   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>-            decoder_->Decode(encoded_cb_.encoded_frame_, false, nullptr));
>-  EXPECT_TRUE(decoded_cb_.DecodeComplete());
>+            decoder_->Decode(encoded_frame, false, nullptr, -1));
>+
>+  std::unique_ptr<VideoFrame> decoded_frame;
>+  absl::optional<uint8_t> decoded_qp;
>+  ASSERT_TRUE(WaitForDecodedFrame(&decoded_frame, &decoded_qp));
>+  ASSERT_TRUE(decoded_frame);
>   // Compute PSNR on all planes (faster than SSIM).
>-  EXPECT_GT(I420PSNR(input_frame_.get(), &*decoded_cb_.frame_), 36);
>-  EXPECT_EQ(kInitialTimestampRtp, decoded_cb_.frame_->timestamp());
>-  EXPECT_EQ(kTestNtpTimeMs, decoded_cb_.frame_->ntp_time_ms());
>+  EXPECT_GT(I420PSNR(input_frame, decoded_frame.get()), 36);
>+  EXPECT_EQ(kInitialTimestampRtp, decoded_frame->timestamp());
>+  EXPECT_EQ(kTestNtpTimeMs, decoded_frame->ntp_time_ms());
> }
> 
> #if defined(WEBRTC_ANDROID)
>@@ -367,117 +264,49 @@ TEST_F(TestVp8Impl, MAYBE_AlignedStrideEncodeDecode) {
> #define MAYBE_DecodeWithACompleteKeyFrame DecodeWithACompleteKeyFrame
> #endif
> TEST_F(TestVp8Impl, MAYBE_DecodeWithACompleteKeyFrame) {
>-  InitEncodeDecode();
>-  EncodeFrame();
>+  VideoFrame* input_frame = NextInputFrame();
>+  EncodedImage encoded_frame;
>+  CodecSpecificInfo codec_specific_info;
>+  EncodeAndWaitForFrame(*input_frame, &encoded_frame, &codec_specific_info);
>+
>   // Setting complete to false -> should return an error.
>-  encoded_cb_.encoded_frame_._completeFrame = false;
>+  encoded_frame._completeFrame = false;
>   EXPECT_EQ(WEBRTC_VIDEO_CODEC_ERROR,
>-            decoder_->Decode(encoded_cb_.encoded_frame_, false, nullptr));
>+            decoder_->Decode(encoded_frame, false, nullptr, -1));
>   // Setting complete back to true. Forcing a delta frame.
>-  encoded_cb_.encoded_frame_._frameType = kVideoFrameDelta;
>-  encoded_cb_.encoded_frame_._completeFrame = true;
>+  encoded_frame._frameType = kVideoFrameDelta;
>+  encoded_frame._completeFrame = true;
>   EXPECT_EQ(WEBRTC_VIDEO_CODEC_ERROR,
>-            decoder_->Decode(encoded_cb_.encoded_frame_, false, nullptr));
>+            decoder_->Decode(encoded_frame, false, nullptr, -1));
>   // Now setting a key frame.
>-  encoded_cb_.encoded_frame_._frameType = kVideoFrameKey;
>+  encoded_frame._frameType = kVideoFrameKey;
>   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>-            decoder_->Decode(encoded_cb_.encoded_frame_, false, nullptr));
>-  ASSERT_TRUE(decoded_cb_.frame_);
>-  EXPECT_GT(I420PSNR(input_frame_.get(), &*decoded_cb_.frame_), 36);
>+            decoder_->Decode(encoded_frame, false, nullptr, -1));
>+  std::unique_ptr<VideoFrame> decoded_frame;
>+  absl::optional<uint8_t> decoded_qp;
>+  ASSERT_TRUE(WaitForDecodedFrame(&decoded_frame, &decoded_qp));
>+  ASSERT_TRUE(decoded_frame);
>+  EXPECT_GT(I420PSNR(input_frame, decoded_frame.get()), 36);
> }
> 
>-TEST_F(TestVp8Impl, EncoderWith2TemporalLayersRetainsRtpStateAfterRelease) {
>+TEST_F(TestVp8Impl, EncoderWith2TemporalLayers) {
>   codec_settings_.VP8()->numberOfTemporalLayers = 2;
>-  InitEncodeDecode();
>-
>-  // Temporal layer 0.
>-  EncodeFrame();
>-  EXPECT_EQ(0, encoded_cb_.codec_specific_info_.codecSpecific.VP8.temporalIdx);
>-  int16_t picture_id =
>-      encoded_cb_.codec_specific_info_.codecSpecific.VP8.pictureId;
>-  int tl0_pic_idx =
>-      encoded_cb_.codec_specific_info_.codecSpecific.VP8.tl0PicIdx;
>-  // Temporal layer 1.
>-  input_frame_->set_timestamp(input_frame_->timestamp() + kTimestampIncrement);
>-  EncodeFrame();
>-  ExpectFrameWith(picture_id + 1, tl0_pic_idx + 0, 1);
>-  // Temporal layer 0.
>-  input_frame_->set_timestamp(input_frame_->timestamp() + kTimestampIncrement);
>-  EncodeFrame();
>-  ExpectFrameWith(picture_id + 2, tl0_pic_idx + 1, 0);
>-  // Temporal layer 1.
>-  input_frame_->set_timestamp(input_frame_->timestamp() + kTimestampIncrement);
>-  EncodeFrame();
>-  ExpectFrameWith(picture_id + 3, tl0_pic_idx + 1, 1);
>-
>-  // Reinit.
>-  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Release());
>   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>             encoder_->InitEncode(&codec_settings_, kNumCores, kMaxPayloadSize));
> 
>   // Temporal layer 0.
>-  input_frame_->set_timestamp(input_frame_->timestamp() + kTimestampIncrement);
>-  EncodeFrame();
>-  ExpectFrameWith(picture_id + 4, tl0_pic_idx + 2, 0);
>-  // Temporal layer 1.
>-  input_frame_->set_timestamp(input_frame_->timestamp() + kTimestampIncrement);
>-  EncodeFrame();
>-  ExpectFrameWith(picture_id + 5, tl0_pic_idx + 2, 1);
>-  // Temporal layer 0.
>-  input_frame_->set_timestamp(input_frame_->timestamp() + kTimestampIncrement);
>-  EncodeFrame();
>-  ExpectFrameWith(picture_id + 6, tl0_pic_idx + 3, 0);
>-  // Temporal layer 1.
>-  input_frame_->set_timestamp(input_frame_->timestamp() + kTimestampIncrement);
>-  EncodeFrame();
>-  ExpectFrameWith(picture_id + 7, tl0_pic_idx + 3, 1);
>-}
>+  EncodedImage encoded_frame;
>+  CodecSpecificInfo codec_specific_info;
>+  EncodeAndWaitForFrame(*NextInputFrame(), &encoded_frame,
>+                        &codec_specific_info);
> 
>-TEST_F(TestVp8Impl, EncoderWith3TemporalLayersRetainsRtpStateAfterRelease) {
>-  codec_settings_.VP8()->numberOfTemporalLayers = 3;
>-  InitEncodeDecode();
>-
>-  // Temporal layer 0.
>-  EncodeFrame();
>-  EXPECT_EQ(0, encoded_cb_.codec_specific_info_.codecSpecific.VP8.temporalIdx);
>-  int16_t picture_id =
>-      encoded_cb_.codec_specific_info_.codecSpecific.VP8.pictureId;
>-  int tl0_pic_idx =
>-      encoded_cb_.codec_specific_info_.codecSpecific.VP8.tl0PicIdx;
>-  // Temporal layer 2.
>-  input_frame_->set_timestamp(input_frame_->timestamp() + kTimestampIncrement);
>-  EncodeFrame();
>-  ExpectFrameWith(picture_id + 1, tl0_pic_idx + 0, 2);
>+  EXPECT_EQ(0, codec_specific_info.codecSpecific.VP8.temporalIdx);
>   // Temporal layer 1.
>-  input_frame_->set_timestamp(input_frame_->timestamp() + kTimestampIncrement);
>-  EncodeFrame();
>-  ExpectFrameWith(picture_id + 2, tl0_pic_idx + 0, 1);
>-  // Temporal layer 2.
>-  input_frame_->set_timestamp(input_frame_->timestamp() + kTimestampIncrement);
>-  EncodeFrame();
>-  ExpectFrameWith(picture_id + 3, tl0_pic_idx + 0, 2);
>-
>-  // Reinit.
>-  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Release());
>-  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>-            encoder_->InitEncode(&codec_settings_, kNumCores, kMaxPayloadSize));
>-
>+  EncodeAndExpectFrameWith(*NextInputFrame(), 1);
>   // Temporal layer 0.
>-  input_frame_->set_timestamp(input_frame_->timestamp() + kTimestampIncrement);
>-  EncodeFrame();
>-  ExpectFrameWith(picture_id + 4, tl0_pic_idx + 1, 0);
>-  // Temporal layer 2.
>-  input_frame_->set_timestamp(input_frame_->timestamp() + kTimestampIncrement);
>-  EncodeFrame();
>-  ExpectFrameWith(picture_id + 5, tl0_pic_idx + 1, 2);
>+  EncodeAndExpectFrameWith(*NextInputFrame(), 0);
>   // Temporal layer 1.
>-  input_frame_->set_timestamp(input_frame_->timestamp() + kTimestampIncrement);
>-  EncodeFrame();
>-  ExpectFrameWith(picture_id + 6, tl0_pic_idx + 1, 1);
>-  // Temporal layer 2.
>-  input_frame_->set_timestamp(input_frame_->timestamp() + kTimestampIncrement);
>-  EncodeFrame();
>-  ExpectFrameWith(picture_id + 7, tl0_pic_idx + 1, 2);
>+  EncodeAndExpectFrameWith(*NextInputFrame(), 1);
> }
> 
> TEST_F(TestVp8Impl, ScalingDisabledIfAutomaticResizeOff) {
>@@ -487,7 +316,7 @@ TEST_F(TestVp8Impl, ScalingDisabledIfAutomaticResizeOff) {
>             encoder_->InitEncode(&codec_settings_, kNumCores, kMaxPayloadSize));
> 
>   VideoEncoder::ScalingSettings settings = encoder_->GetScalingSettings();
>-  EXPECT_FALSE(settings.enabled);
>+  EXPECT_FALSE(settings.thresholds.has_value());
> }
> 
> TEST_F(TestVp8Impl, ScalingEnabledIfAutomaticResizeOn) {
>@@ -497,7 +326,7 @@ TEST_F(TestVp8Impl, ScalingEnabledIfAutomaticResizeOn) {
>             encoder_->InitEncode(&codec_settings_, kNumCores, kMaxPayloadSize));
> 
>   VideoEncoder::ScalingSettings settings = encoder_->GetScalingSettings();
>-  EXPECT_TRUE(settings.enabled);
>+  EXPECT_TRUE(settings.thresholds.has_value());
>   EXPECT_EQ(kDefaultMinPixelsPerFrame, settings.min_pixels_per_frame);
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/DEPS b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/DEPS
>new file mode 100644
>index 00000000000..cc5cd701423
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/DEPS
>@@ -0,0 +1,3 @@
>+include_rules = [
>+  "+media/base",
>+]
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/include/vp9.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/include/vp9.h
>index 172e69ee787..8091cacec90 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/include/vp9.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/include/vp9.h
>@@ -13,25 +13,34 @@
> #define MODULES_VIDEO_CODING_CODECS_VP9_INCLUDE_VP9_H_
> 
> #include <memory>
>+#include <vector>
> 
>+#include "api/video_codecs/sdp_video_format.h"
>+#include "media/base/codec.h"
> #include "modules/video_coding/include/video_codec_interface.h"
> 
> namespace webrtc {
> 
>+// Returns a vector with all supported internal VP9 profiles that we can
>+// negotiate in SDP, in order of preference.
>+std::vector<SdpVideoFormat> SupportedVP9Codecs();
>+
> class VP9Encoder : public VideoEncoder {
>  public:
>-  static bool IsSupported();
>+  // Deprecated. Returns default implementation using VP9 Profile 0.
>+  // TODO(emircan): Remove once this is no longer used.
>   static std::unique_ptr<VP9Encoder> Create();
>+  // Parses VP9 Profile from |codec| and returns the appropriate implementation.
>+  static std::unique_ptr<VP9Encoder> Create(const cricket::VideoCodec& codec);
> 
>-  virtual ~VP9Encoder() {}
>+  ~VP9Encoder() override {}
> };
> 
> class VP9Decoder : public VideoDecoder {
>  public:
>-  static bool IsSupported();
>   static std::unique_ptr<VP9Decoder> Create();
> 
>-  virtual ~VP9Decoder() {}
>+  ~VP9Decoder() override {}
> };
> }  // namespace webrtc
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/include/vp9_globals.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/include/vp9_globals.h
>index 945eb0cfcd1..aa532a68764 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/include/vp9_globals.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/include/vp9_globals.h
>@@ -14,6 +14,8 @@
> #ifndef MODULES_VIDEO_CODING_CODECS_VP9_INCLUDE_VP9_GLOBALS_H_
> #define MODULES_VIDEO_CODING_CODECS_VP9_INCLUDE_VP9_GLOBALS_H_
> 
>+#include <assert.h>
>+
> #include "modules/video_coding/codecs/interface/common_constants.h"
> 
> namespace webrtc {
>@@ -27,6 +29,9 @@ const size_t kMaxVp9RefPics = 3;
> const size_t kMaxVp9FramesInGof = 0xFF;  // 8 bits
> const size_t kMaxVp9NumberOfSpatialLayers = 8;
> 
>+const size_t kMinVp9SpatialLayerWidth = 320;
>+const size_t kMinVp9SpatialLayerHeight = 180;
>+
> enum TemporalStructureMode {
>   kTemporalStructureMode1,  // 1 temporal layer structure - i.e., IPPP...
>   kTemporalStructureMode2,  // 2 temporal layers 01...
>@@ -74,10 +79,9 @@ struct GofInfoVP9 {
>         pid_diff[2][0] = 2;
> 
>         temporal_idx[3] = 2;
>-        temporal_up_switch[3] = false;
>-        num_ref_pics[3] = 2;
>+        temporal_up_switch[3] = true;
>+        num_ref_pics[3] = 1;
>         pid_diff[3][0] = 1;
>-        pid_diff[3][1] = 2;
>         break;
>       case kTemporalStructureMode4:
>         num_frames_in_gof = 8;
>@@ -157,6 +161,7 @@ struct RTPVideoHeaderVP9 {
>     beginning_of_frame = false;
>     end_of_frame = false;
>     ss_data_available = false;
>+    non_ref_for_inter_layer_pred = false;
>     picture_id = kNoPictureId;
>     max_picture_id = kMaxTwoBytePictureId;
>     tl0_pic_idx = kNoTl0PicIdx;
>@@ -167,6 +172,7 @@ struct RTPVideoHeaderVP9 {
>     gof_idx = kNoGofIdx;
>     num_ref_pics = 0;
>     num_spatial_layers = 1;
>+    end_of_picture = true;
>   }
> 
>   bool inter_pic_predicted;  // This layer frame is dependent on previously
>@@ -175,10 +181,12 @@ struct RTPVideoHeaderVP9 {
>   bool beginning_of_frame;   // True if this packet is the first in a VP9 layer
>                              // frame.
>   bool end_of_frame;  // True if this packet is the last in a VP9 layer frame.
>-  bool ss_data_available;   // True if SS data is available in this payload
>-                            // descriptor.
>-  int16_t picture_id;       // PictureID index, 15 bits;
>-                            // kNoPictureId if PictureID does not exist.
>+  bool ss_data_available;  // True if SS data is available in this payload
>+                           // descriptor.
>+  bool non_ref_for_inter_layer_pred;  // True for frame which is not used as
>+                                      // reference for inter-layer prediction.
>+  int16_t picture_id;                 // PictureID index, 15 bits;
>+                       // kNoPictureId if PictureID does not exist.
>   int16_t max_picture_id;   // Maximum picture ID index; either 0x7F or 0x7FFF;
>   int16_t tl0_pic_idx;      // TL0PIC_IDX, 8 bits;
>                             // kNoTl0PicIdx means no value provided.
>@@ -203,6 +211,8 @@ struct RTPVideoHeaderVP9 {
>   uint16_t width[kMaxVp9NumberOfSpatialLayers];
>   uint16_t height[kMaxVp9NumberOfSpatialLayers];
>   GofInfoVP9 gof;
>+
>+  bool end_of_picture;  // This frame is the last frame in picture.
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/screenshare_layers.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/screenshare_layers.cc
>deleted file mode 100644
>index 158b0e52dc2..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/screenshare_layers.cc
>+++ /dev/null
>@@ -1,93 +0,0 @@
>-/* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
>-*
>-*  Use of this source code is governed by a BSD-style license
>-*  that can be found in the LICENSE file in the root of the source
>-*  tree. An additional intellectual property rights grant can be found
>-*  in the file PATENTS.  All contributing project authors may
>-*  be found in the AUTHORS file in the root of the source tree.
>-*/
>-
>-#include "modules/video_coding/codecs/vp9/screenshare_layers.h"
>-#include <algorithm>
>-#include "rtc_base/checks.h"
>-
>-namespace webrtc {
>-
>-ScreenshareLayersVP9::ScreenshareLayersVP9(uint8_t num_layers)
>-    : num_layers_(num_layers),
>-      start_layer_(0),
>-      last_timestamp_(0),
>-      timestamp_initialized_(false) {
>-  RTC_DCHECK_GT(num_layers, 0);
>-  RTC_DCHECK_LE(num_layers, kMaxVp9NumberOfSpatialLayers);
>-  memset(bits_used_, 0, sizeof(bits_used_));
>-  memset(threshold_kbps_, 0, sizeof(threshold_kbps_));
>-}
>-
>-uint8_t ScreenshareLayersVP9::GetStartLayer() const {
>-  return start_layer_;
>-}
>-
>-void ScreenshareLayersVP9::ConfigureBitrate(int threshold_kbps,
>-                                            uint8_t layer_id) {
>-  // The upper layer is always the layer we spill frames
>-  // to when the bitrate becomes to high, therefore setting
>-  // a max limit is not allowed. The top layer bitrate is
>-  // never used either so configuring it makes no difference.
>-  RTC_DCHECK_LT(layer_id, num_layers_ - 1);
>-  threshold_kbps_[layer_id] = threshold_kbps;
>-}
>-
>-void ScreenshareLayersVP9::LayerFrameEncoded(unsigned int size_bytes,
>-                                             uint8_t layer_id) {
>-  RTC_DCHECK_LT(layer_id, num_layers_);
>-  bits_used_[layer_id] += size_bytes * 8;
>-}
>-
>-VP9EncoderImpl::SuperFrameRefSettings
>-ScreenshareLayersVP9::GetSuperFrameSettings(uint32_t timestamp,
>-                                            bool is_keyframe) {
>-  VP9EncoderImpl::SuperFrameRefSettings settings;
>-  if (!timestamp_initialized_) {
>-    last_timestamp_ = timestamp;
>-    timestamp_initialized_ = true;
>-  }
>-  float time_diff = (timestamp - last_timestamp_) / 90.f;
>-  float total_bits_used = 0;
>-  float total_threshold_kbps = 0;
>-  start_layer_ = 0;
>-
>-  // Up to (num_layers - 1) because we only have
>-  // (num_layers - 1) thresholds to check.
>-  for (int layer_id = 0; layer_id < num_layers_ - 1; ++layer_id) {
>-    bits_used_[layer_id] = std::max(
>-        0.f, bits_used_[layer_id] - time_diff * threshold_kbps_[layer_id]);
>-    total_bits_used += bits_used_[layer_id];
>-    total_threshold_kbps += threshold_kbps_[layer_id];
>-
>-    // If this is a keyframe then there should be no
>-    // references to any previous frames.
>-    if (!is_keyframe) {
>-      settings.layer[layer_id].ref_buf1 = layer_id;
>-      if (total_bits_used > total_threshold_kbps * 1000)
>-        start_layer_ = layer_id + 1;
>-    }
>-
>-    settings.layer[layer_id].upd_buf = layer_id;
>-  }
>-  // Since the above loop does not iterate over the last layer
>-  // the reference of the last layer has to be set after the loop,
>-  // and if this is a keyframe there should be no references to
>-  // any previous frames.
>-  if (!is_keyframe)
>-    settings.layer[num_layers_ - 1].ref_buf1 = num_layers_ - 1;
>-
>-  settings.layer[num_layers_ - 1].upd_buf = num_layers_ - 1;
>-  settings.is_keyframe = is_keyframe;
>-  settings.start_layer = start_layer_;
>-  settings.stop_layer = num_layers_ - 1;
>-  last_timestamp_ = timestamp;
>-  return settings;
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/screenshare_layers.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/screenshare_layers.h
>deleted file mode 100644
>index 6725ab76f90..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/screenshare_layers.h
>+++ /dev/null
>@@ -1,66 +0,0 @@
>-/* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
>-*
>-*  Use of this source code is governed by a BSD-style license
>-*  that can be found in the LICENSE file in the root of the source
>-*  tree. An additional intellectual property rights grant can be found
>-*  in the file PATENTS.  All contributing project authors may
>-*  be found in the AUTHORS file in the root of the source tree.
>-*/
>-
>-#ifndef MODULES_VIDEO_CODING_CODECS_VP9_SCREENSHARE_LAYERS_H_
>-#define MODULES_VIDEO_CODING_CODECS_VP9_SCREENSHARE_LAYERS_H_
>-
>-#include "modules/video_coding/codecs/vp9/vp9_impl.h"
>-
>-namespace webrtc {
>-
>-class ScreenshareLayersVP9 {
>- public:
>-  explicit ScreenshareLayersVP9(uint8_t num_layers);
>-
>-  // The target bitrate for layer with id layer_id.
>-  void ConfigureBitrate(int threshold_kbps, uint8_t layer_id);
>-
>-  // The current start layer.
>-  uint8_t GetStartLayer() const;
>-
>-  // Update the layer with the size of the layer frame.
>-  void LayerFrameEncoded(unsigned int size_bytes, uint8_t layer_id);
>-
>-  // Get the layer settings for the next superframe.
>-  //
>-  // In short, each time the GetSuperFrameSettings is called the
>-  // bitrate of every layer is calculated and if the cummulative
>-  // bitrate exceeds the configured cummulative bitrates
>-  // (ConfigureBitrate to configure) up to and including that
>-  // layer then the resulting encoding settings for the
>-  // superframe will only encode layers above that layer.
>-  VP9EncoderImpl::SuperFrameRefSettings GetSuperFrameSettings(
>-      uint32_t timestamp,
>-      bool is_keyframe);
>-
>- private:
>-  // How many layers that are used.
>-  uint8_t num_layers_;
>-
>-  // The index of the first layer to encode.
>-  uint8_t start_layer_;
>-
>-  // Cummulative target kbps for the different layers.
>-  float threshold_kbps_[kMaxVp9NumberOfSpatialLayers - 1];
>-
>-  // How many bits that has been used for a certain layer. Increased in
>-  // FrameEncoded() by the size of the encoded frame and decreased in
>-  // GetSuperFrameSettings() depending on the time between frames.
>-  float bits_used_[kMaxVp9NumberOfSpatialLayers];
>-
>-  // Timestamp of last frame.
>-  uint32_t last_timestamp_;
>-
>-  // If the last_timestamp_ has been set.
>-  bool timestamp_initialized_;
>-};
>-
>-}  // namespace webrtc
>-
>-#endif  // MODULES_VIDEO_CODING_CODECS_VP9_SCREENSHARE_LAYERS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/svc_config.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/svc_config.cc
>new file mode 100644
>index 00000000000..22902feceae
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/svc_config.cc
>@@ -0,0 +1,122 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/video_coding/codecs/vp9/svc_config.h"
>+
>+#include <algorithm>
>+#include <cmath>
>+#include <vector>
>+
>+#include "modules/video_coding/codecs/vp9/svc_rate_allocator.h"
>+#include "modules/video_coding/include/video_codec_interface.h"
>+
>+namespace webrtc {
>+
>+namespace {
>+const size_t kMinVp9SvcBitrateKbps = 30;
>+
>+const size_t kMaxNumLayersForScreenSharing = 2;
>+const size_t kMaxScreenSharingLayerBitrateKbps[] = {200, 500};
>+}  // namespace
>+
>+std::vector<SpatialLayer> ConfigureSvcScreenSharing(size_t input_width,
>+                                                    size_t input_height,
>+                                                    size_t num_spatial_layers) {
>+  num_spatial_layers =
>+      std::min(num_spatial_layers, kMaxNumLayersForScreenSharing);
>+  std::vector<SpatialLayer> spatial_layers;
>+
>+  for (size_t sl_idx = 0; sl_idx < num_spatial_layers; ++sl_idx) {
>+    SpatialLayer spatial_layer = {0};
>+    spatial_layer.width = input_width;
>+    spatial_layer.height = input_height;
>+    spatial_layer.numberOfTemporalLayers = 1;
>+    spatial_layer.minBitrate = static_cast<int>(kMinVp9SvcBitrateKbps);
>+    spatial_layer.maxBitrate =
>+        static_cast<int>(kMaxScreenSharingLayerBitrateKbps[sl_idx]);
>+    spatial_layer.targetBitrate = spatial_layer.maxBitrate;
>+    spatial_layers.push_back(spatial_layer);
>+  }
>+
>+  return spatial_layers;
>+}
>+
>+std::vector<SpatialLayer> ConfigureSvcNormalVideo(size_t input_width,
>+                                                  size_t input_height,
>+                                                  size_t num_spatial_layers,
>+                                                  size_t num_temporal_layers) {
>+  std::vector<SpatialLayer> spatial_layers;
>+
>+  // Limit number of layers for given resolution.
>+  const size_t num_layers_fit_horz = static_cast<size_t>(std::floor(
>+      1 + std::max(0.0f,
>+                   std::log2(1.0f * input_width / kMinVp9SpatialLayerWidth))));
>+  const size_t num_layers_fit_vert = static_cast<size_t>(
>+      std::floor(1 + std::max(0.0f, std::log2(1.0f * input_height /
>+                                              kMinVp9SpatialLayerHeight))));
>+  num_spatial_layers =
>+      std::min({num_spatial_layers, num_layers_fit_horz, num_layers_fit_vert});
>+
>+  float top_fraction = 0.;
>+  for (size_t sl_idx = 0; sl_idx < num_spatial_layers; ++sl_idx) {
>+    SpatialLayer spatial_layer = {0};
>+    spatial_layer.width = input_width >> (num_spatial_layers - sl_idx - 1);
>+    spatial_layer.height = input_height >> (num_spatial_layers - sl_idx - 1);
>+    spatial_layer.numberOfTemporalLayers = num_temporal_layers;
>+
>+    // minBitrate and maxBitrate formulas were derived from
>+    // subjective-quality data to determing bit rates below which video
>+    // quality is unacceptable and above which additional bits do not provide
>+    // benefit. The formulas express rate in units of kbps.
>+
>+    // TODO(ssilkin): Add to the comment PSNR/SSIM we get at encoding certain
>+    // video to min/max bitrate specified by those formulas.
>+    const size_t num_pixels = spatial_layer.width * spatial_layer.height;
>+    int min_bitrate =
>+        static_cast<int>((600. * std::sqrt(num_pixels) - 95000.) / 1000.);
>+    min_bitrate = std::max(min_bitrate, 0);
>+    spatial_layer.minBitrate =
>+        std::max(static_cast<size_t>(min_bitrate), kMinVp9SvcBitrateKbps);
>+    spatial_layer.maxBitrate =
>+        static_cast<int>((1.6 * num_pixels + 50 * 1000) / 1000);
>+    spatial_layer.targetBitrate =
>+        (spatial_layer.minBitrate + spatial_layer.maxBitrate) / 2;
>+    spatial_layers.push_back(spatial_layer);
>+    top_fraction += std::pow(kSpatialLayeringRateScalingFactor, sl_idx);
>+  }
>+  // Compute spatial_layers[num_spatial_layers - 1].targetBitrate, which is
>+  // used to set max_padding_bitrate_. Set max_padding_bitrate_ equal to the
>+  // minimum total bit rate required to support all spatial layers.
>+  spatial_layers[num_spatial_layers - 1].targetBitrate =
>+      static_cast<unsigned int>(
>+          spatial_layers[num_spatial_layers - 1].minBitrate * top_fraction);
>+  return spatial_layers;
>+}
>+
>+std::vector<SpatialLayer> GetSvcConfig(size_t input_width,
>+                                       size_t input_height,
>+                                       size_t num_spatial_layers,
>+                                       size_t num_temporal_layers,
>+                                       bool is_screen_sharing) {
>+  RTC_DCHECK_GT(input_width, 0);
>+  RTC_DCHECK_GT(input_height, 0);
>+  RTC_DCHECK_GT(num_spatial_layers, 0);
>+  RTC_DCHECK_GT(num_temporal_layers, 0);
>+
>+  if (is_screen_sharing) {
>+    return ConfigureSvcScreenSharing(input_width, input_height,
>+                                     num_spatial_layers);
>+  } else {
>+    return ConfigureSvcNormalVideo(input_width, input_height,
>+                                   num_spatial_layers, num_temporal_layers);
>+  }
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/svc_config.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/svc_config.h
>new file mode 100644
>index 00000000000..c8561a4c9d3
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/svc_config.h
>@@ -0,0 +1,27 @@
>+/* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_VIDEO_CODING_CODECS_VP9_SVC_CONFIG_H_
>+#define MODULES_VIDEO_CODING_CODECS_VP9_SVC_CONFIG_H_
>+
>+#include <vector>
>+
>+#include "common_types.h"  // NOLINT(build/include)
>+
>+namespace webrtc {
>+
>+std::vector<SpatialLayer> GetSvcConfig(size_t input_width,
>+                                       size_t input_height,
>+                                       size_t num_spatial_layers,
>+                                       size_t num_temporal_layers,
>+                                       bool is_screen_sharing);
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_VIDEO_CODING_CODECS_VP9_SVC_CONFIG_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/svc_config_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/svc_config_unittest.cc
>new file mode 100644
>index 00000000000..ebefddab13f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/svc_config_unittest.cc
>@@ -0,0 +1,63 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <cstddef>
>+#include <cstdint>
>+#include <vector>
>+
>+#include "modules/video_coding/codecs/vp9/include/vp9_globals.h"
>+#include "modules/video_coding/codecs/vp9/svc_config.h"
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+TEST(SvcConfig, NumSpatialLayers) {
>+  const size_t max_num_spatial_layers = 6;
>+  const size_t num_spatial_layers = 2;
>+
>+  std::vector<SpatialLayer> spatial_layers =
>+      GetSvcConfig(kMinVp9SpatialLayerWidth << (num_spatial_layers - 1),
>+                   kMinVp9SpatialLayerHeight << (num_spatial_layers - 1),
>+                   max_num_spatial_layers, 1, false);
>+
>+  EXPECT_EQ(spatial_layers.size(), num_spatial_layers);
>+}
>+
>+TEST(SvcConfig, BitrateThresholds) {
>+  const size_t num_spatial_layers = 3;
>+  std::vector<SpatialLayer> spatial_layers =
>+      GetSvcConfig(kMinVp9SpatialLayerWidth << (num_spatial_layers - 1),
>+                   kMinVp9SpatialLayerHeight << (num_spatial_layers - 1),
>+                   num_spatial_layers, 1, false);
>+
>+  EXPECT_EQ(spatial_layers.size(), num_spatial_layers);
>+
>+  for (const SpatialLayer& layer : spatial_layers) {
>+    EXPECT_LE(layer.minBitrate, layer.maxBitrate);
>+    EXPECT_LE(layer.minBitrate, layer.targetBitrate);
>+    EXPECT_LE(layer.targetBitrate, layer.maxBitrate);
>+  }
>+}
>+
>+TEST(SvcConfig, ScreenSharing) {
>+  std::vector<SpatialLayer> spatial_layers =
>+      GetSvcConfig(1920, 1080, 3, 3, true);
>+
>+  EXPECT_EQ(spatial_layers.size(), 2UL);
>+
>+  for (const SpatialLayer& layer : spatial_layers) {
>+    EXPECT_EQ(layer.width, 1920);
>+    EXPECT_EQ(layer.height, 1080);
>+    EXPECT_EQ(layer.numberOfTemporalLayers, 1);
>+    EXPECT_LE(layer.minBitrate, layer.maxBitrate);
>+    EXPECT_LE(layer.minBitrate, layer.targetBitrate);
>+    EXPECT_LE(layer.targetBitrate, layer.maxBitrate);
>+  }
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/svc_rate_allocator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/svc_rate_allocator.cc
>new file mode 100644
>index 00000000000..9b846ade47d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/svc_rate_allocator.cc
>@@ -0,0 +1,189 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/video_coding/codecs/vp9/svc_rate_allocator.h"
>+
>+#include <algorithm>
>+#include <cmath>
>+#include <numeric>
>+
>+#include "rtc_base/checks.h"
>+
>+namespace webrtc {
>+
>+const float kSpatialLayeringRateScalingFactor = 0.55f;
>+const float kTemporalLayeringRateScalingFactor = 0.55f;
>+
>+SvcRateAllocator::SvcRateAllocator(const VideoCodec& codec) : codec_(codec) {
>+  RTC_DCHECK_EQ(codec.codecType, kVideoCodecVP9);
>+}
>+
>+VideoBitrateAllocation SvcRateAllocator::GetAllocation(
>+    uint32_t total_bitrate_bps,
>+    uint32_t framerate_fps) {
>+  if (codec_.maxBitrate != 0) {
>+    total_bitrate_bps = std::min(total_bitrate_bps, codec_.maxBitrate * 1000);
>+  }
>+
>+  if (codec_.spatialLayers[0].targetBitrate == 0) {
>+    // Delegate rate distribution to VP9 encoder wrapper if bitrate thresholds
>+    // are not initialized.
>+    VideoBitrateAllocation bitrate_allocation;
>+    bitrate_allocation.SetBitrate(0, 0, total_bitrate_bps);
>+    return bitrate_allocation;
>+  } else if (codec_.mode == VideoCodecMode::kRealtimeVideo) {
>+    return GetAllocationNormalVideo(total_bitrate_bps);
>+  } else {
>+    return GetAllocationScreenSharing(total_bitrate_bps);
>+  }
>+}
>+
>+VideoBitrateAllocation SvcRateAllocator::GetAllocationNormalVideo(
>+    uint32_t total_bitrate_bps) const {
>+  size_t num_spatial_layers = codec_.VP9().numberOfSpatialLayers;
>+  RTC_CHECK(num_spatial_layers > 0);
>+  size_t num_temporal_layers = codec_.VP9().numberOfTemporalLayers;
>+  RTC_CHECK(num_temporal_layers > 0);
>+
>+  std::vector<size_t> spatial_layer_bitrate_bps;
>+
>+  // Distribute total bitrate across spatial layers. If there is not enough
>+  // bitrate to provide all layers with at least minimum required bitrate
>+  // then number of layers is reduced by one and distribution is repeated
>+  // until that condition is met or if number of layers is reduced to one.
>+  for (;; --num_spatial_layers) {
>+    spatial_layer_bitrate_bps =
>+        SplitBitrate(num_spatial_layers, total_bitrate_bps,
>+                     kSpatialLayeringRateScalingFactor);
>+
>+    const bool enough_bitrate = AdjustAndVerify(&spatial_layer_bitrate_bps);
>+    if (enough_bitrate || num_spatial_layers == 1) {
>+      break;
>+    }
>+  }
>+
>+  VideoBitrateAllocation bitrate_allocation;
>+
>+  for (size_t sl_idx = 0; sl_idx < num_spatial_layers; ++sl_idx) {
>+    std::vector<size_t> temporal_layer_bitrate_bps =
>+        SplitBitrate(num_temporal_layers, spatial_layer_bitrate_bps[sl_idx],
>+                     kTemporalLayeringRateScalingFactor);
>+
>+    // Distribute rate across temporal layers. Allocate more bits to lower
>+    // layers since they are used for prediction of higher layers and their
>+    // references are far apart.
>+    if (num_temporal_layers == 1) {
>+      bitrate_allocation.SetBitrate(sl_idx, 0, temporal_layer_bitrate_bps[0]);
>+    } else if (num_temporal_layers == 2) {
>+      bitrate_allocation.SetBitrate(sl_idx, 0, temporal_layer_bitrate_bps[1]);
>+      bitrate_allocation.SetBitrate(sl_idx, 1, temporal_layer_bitrate_bps[0]);
>+    } else {
>+      RTC_CHECK_EQ(num_temporal_layers, 3);
>+      // In case of three temporal layers the high layer has two frames and the
>+      // middle layer has one frame within GOP (in between two consecutive low
>+      // layer frames). Thus high layer requires more bits (comparing pure
>+      // bitrate of layer, excluding bitrate of base layers) to keep quality on
>+      // par with lower layers.
>+      bitrate_allocation.SetBitrate(sl_idx, 0, temporal_layer_bitrate_bps[2]);
>+      bitrate_allocation.SetBitrate(sl_idx, 1, temporal_layer_bitrate_bps[0]);
>+      bitrate_allocation.SetBitrate(sl_idx, 2, temporal_layer_bitrate_bps[1]);
>+    }
>+  }
>+
>+  return bitrate_allocation;
>+}
>+
>+bool SvcRateAllocator::AdjustAndVerify(
>+    std::vector<size_t>* spatial_layer_bitrate_bps) const {
>+  bool enough_bitrate = true;
>+  size_t excess_rate = 0;
>+  for (size_t sl_idx = 0;
>+       sl_idx < spatial_layer_bitrate_bps->size() && enough_bitrate; ++sl_idx) {
>+    RTC_DCHECK_GT(codec_.spatialLayers[sl_idx].maxBitrate, 0);
>+    RTC_DCHECK_GE(codec_.spatialLayers[sl_idx].maxBitrate,
>+                  codec_.spatialLayers[sl_idx].minBitrate);
>+
>+    const size_t min_bitrate_bps =
>+        codec_.spatialLayers[sl_idx].minBitrate * 1000;
>+    const size_t max_bitrate_bps =
>+        codec_.spatialLayers[sl_idx].maxBitrate * 1000;
>+
>+    spatial_layer_bitrate_bps->at(sl_idx) += excess_rate;
>+    if (spatial_layer_bitrate_bps->at(sl_idx) < max_bitrate_bps) {
>+      excess_rate = 0;
>+    } else {
>+      excess_rate = spatial_layer_bitrate_bps->at(sl_idx) - max_bitrate_bps;
>+      spatial_layer_bitrate_bps->at(sl_idx) = max_bitrate_bps;
>+    }
>+
>+    enough_bitrate = (spatial_layer_bitrate_bps->at(sl_idx) >= min_bitrate_bps);
>+  }
>+
>+  return enough_bitrate;
>+}
>+
>+VideoBitrateAllocation SvcRateAllocator::GetAllocationScreenSharing(
>+    uint32_t total_bitrate_bps) const {
>+  const size_t num_spatial_layers = codec_.VP9().numberOfSpatialLayers;
>+  RTC_CHECK(num_spatial_layers > 0);
>+  RTC_CHECK_EQ(codec_.VP9().numberOfTemporalLayers, 1U);
>+
>+  VideoBitrateAllocation bitrate_allocation;
>+
>+  // Add next layer after bitrate of previous layer has reached its maximum.
>+  size_t left_bitrate_bps = total_bitrate_bps;
>+  for (size_t sl_idx = 0; sl_idx < num_spatial_layers; ++sl_idx) {
>+    const size_t min_bitrate_bps =
>+        codec_.spatialLayers[sl_idx].minBitrate * 1000;
>+    const size_t max_bitrate_bps =
>+        codec_.spatialLayers[sl_idx].maxBitrate * 1000;
>+
>+    const size_t bitrate_bps = std::min(left_bitrate_bps, max_bitrate_bps);
>+    if (bitrate_bps >= min_bitrate_bps) {
>+      bitrate_allocation.SetBitrate(sl_idx, 0, bitrate_bps);
>+    } else {
>+      break;
>+    }
>+
>+    left_bitrate_bps -= bitrate_bps;
>+  }
>+
>+  return bitrate_allocation;
>+}
>+
>+std::vector<size_t> SvcRateAllocator::SplitBitrate(
>+    size_t num_layers,
>+    size_t total_bitrate,
>+    float rate_scaling_factor) const {
>+  std::vector<size_t> bitrates;
>+
>+  double denominator = 0.0;
>+  for (size_t layer_idx = 0; layer_idx < num_layers; ++layer_idx) {
>+    denominator += std::pow(rate_scaling_factor, layer_idx);
>+  }
>+
>+  double numerator = std::pow(rate_scaling_factor, num_layers - 1);
>+  for (size_t layer_idx = 0; layer_idx < num_layers; ++layer_idx) {
>+    bitrates.push_back(numerator * total_bitrate / denominator);
>+    numerator /= rate_scaling_factor;
>+  }
>+
>+  const size_t sum = std::accumulate(bitrates.begin(), bitrates.end(), 0);
>+  // Ensure the sum of split bitrates doesn't exceed the total bitrate.
>+  RTC_DCHECK_LE(sum, total_bitrate);
>+
>+  // Keep the sum of split bitrates equal to the total bitrate by adding bits,
>+  // which were lost due to rounding, to the latest layer.
>+  bitrates.back() += total_bitrate - sum;
>+
>+  return bitrates;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/svc_rate_allocator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/svc_rate_allocator.h
>new file mode 100644
>index 00000000000..10e22edc603
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/svc_rate_allocator.h
>@@ -0,0 +1,48 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_VIDEO_CODING_CODECS_VP9_SVC_RATE_ALLOCATOR_H_
>+#define MODULES_VIDEO_CODING_CODECS_VP9_SVC_RATE_ALLOCATOR_H_
>+
>+#include <stdint.h>
>+
>+#include <vector>
>+
>+#include "api/video/video_bitrate_allocator.h"
>+#include "api/video_codecs/video_codec.h"
>+
>+namespace webrtc {
>+
>+extern const float kSpatialLayeringRateScalingFactor;
>+extern const float kTemporalLayeringRateScalingFactor;
>+
>+class SvcRateAllocator : public VideoBitrateAllocator {
>+ public:
>+  explicit SvcRateAllocator(const VideoCodec& codec);
>+
>+  VideoBitrateAllocation GetAllocation(uint32_t total_bitrate_bps,
>+                                       uint32_t framerate_fps) override;
>+
>+ private:
>+  VideoBitrateAllocation GetAllocationNormalVideo(
>+      uint32_t total_bitrate_bps) const;
>+  VideoBitrateAllocation GetAllocationScreenSharing(
>+      uint32_t total_bitrate_bps) const;
>+  std::vector<size_t> SplitBitrate(size_t num_layers,
>+                                   size_t total_bitrate,
>+                                   float rate_scaling_factor) const;
>+  bool AdjustAndVerify(std::vector<size_t>* spatial_layer_bitrate_bps) const;
>+
>+  const VideoCodec codec_;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_VIDEO_CODING_CODECS_VP9_SVC_RATE_ALLOCATOR_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/svc_rate_allocator_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/svc_rate_allocator_unittest.cc
>new file mode 100644
>index 00000000000..1b6aa10f4c2
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/svc_rate_allocator_unittest.cc
>@@ -0,0 +1,165 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <algorithm>
>+
>+#include "modules/video_coding/codecs/vp9/svc_config.h"
>+#include "modules/video_coding/codecs/vp9/svc_rate_allocator.h"
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+namespace {
>+static VideoCodec Configure(size_t width,
>+                            size_t height,
>+                            size_t num_spatial_layers,
>+                            size_t num_temporal_layers,
>+                            bool is_screen_sharing) {
>+  VideoCodec codec;
>+  codec.width = width;
>+  codec.height = height;
>+  codec.codecType = kVideoCodecVP9;
>+  codec.mode = is_screen_sharing ? VideoCodecMode::kScreensharing
>+                                 : VideoCodecMode::kRealtimeVideo;
>+
>+  std::vector<SpatialLayer> spatial_layers =
>+      GetSvcConfig(width, height, num_spatial_layers, num_temporal_layers,
>+                   is_screen_sharing);
>+  RTC_CHECK_LE(spatial_layers.size(), kMaxSpatialLayers);
>+
>+  codec.VP9()->numberOfSpatialLayers =
>+      std::min<unsigned char>(num_spatial_layers, spatial_layers.size());
>+  codec.VP9()->numberOfTemporalLayers = std::min<unsigned char>(
>+      num_temporal_layers, spatial_layers.back().numberOfTemporalLayers);
>+
>+  for (size_t sl_idx = 0; sl_idx < spatial_layers.size(); ++sl_idx) {
>+    codec.spatialLayers[sl_idx] = spatial_layers[sl_idx];
>+  }
>+
>+  return codec;
>+}
>+}  // namespace
>+
>+TEST(SvcRateAllocatorTest, SingleLayerFor320x180Input) {
>+  VideoCodec codec = Configure(320, 180, 3, 3, false);
>+  SvcRateAllocator allocator = SvcRateAllocator(codec);
>+
>+  VideoBitrateAllocation allocation = allocator.GetAllocation(1000 * 1000, 30);
>+
>+  EXPECT_GT(allocation.GetSpatialLayerSum(0), 0u);
>+  EXPECT_EQ(allocation.GetSpatialLayerSum(1), 0u);
>+}
>+
>+TEST(SvcRateAllocatorTest, TwoLayersFor640x360Input) {
>+  VideoCodec codec = Configure(640, 360, 3, 3, false);
>+  SvcRateAllocator allocator = SvcRateAllocator(codec);
>+
>+  VideoBitrateAllocation allocation = allocator.GetAllocation(1000 * 1000, 30);
>+
>+  EXPECT_GT(allocation.GetSpatialLayerSum(0), 0u);
>+  EXPECT_GT(allocation.GetSpatialLayerSum(1), 0u);
>+  EXPECT_EQ(allocation.GetSpatialLayerSum(2), 0u);
>+}
>+
>+TEST(SvcRateAllocatorTest, ThreeLayersFor1280x720Input) {
>+  VideoCodec codec = Configure(1280, 720, 3, 3, false);
>+  SvcRateAllocator allocator = SvcRateAllocator(codec);
>+
>+  VideoBitrateAllocation allocation = allocator.GetAllocation(1000 * 1000, 30);
>+
>+  EXPECT_GT(allocation.GetSpatialLayerSum(0), 0u);
>+  EXPECT_GT(allocation.GetSpatialLayerSum(1), 0u);
>+  EXPECT_GT(allocation.GetSpatialLayerSum(2), 0u);
>+}
>+
>+TEST(SvcRateAllocatorTest,
>+     BaseLayerNonZeroBitrateEvenIfTotalIfLessThanMinimum) {
>+  VideoCodec codec = Configure(1280, 720, 3, 3, false);
>+  SvcRateAllocator allocator = SvcRateAllocator(codec);
>+
>+  const SpatialLayer* layers = codec.spatialLayers;
>+
>+  VideoBitrateAllocation allocation =
>+      allocator.GetAllocation(layers[0].minBitrate * 1000 / 2, 30);
>+
>+  EXPECT_GT(allocation.GetSpatialLayerSum(0), 0u);
>+  EXPECT_LT(allocation.GetSpatialLayerSum(0), layers[0].minBitrate * 1000);
>+  EXPECT_EQ(allocation.GetSpatialLayerSum(1), 0u);
>+}
>+
>+TEST(SvcRateAllocatorTest, Disable640x360Layer) {
>+  VideoCodec codec = Configure(1280, 720, 3, 3, false);
>+  SvcRateAllocator allocator = SvcRateAllocator(codec);
>+
>+  const SpatialLayer* layers = codec.spatialLayers;
>+
>+  size_t min_bitrate_for_640x360_layer_kbps =
>+      layers[0].minBitrate + layers[1].minBitrate;
>+
>+  VideoBitrateAllocation allocation = allocator.GetAllocation(
>+      min_bitrate_for_640x360_layer_kbps * 1000 - 1, 30);
>+
>+  EXPECT_GT(allocation.GetSpatialLayerSum(0), 0u);
>+  EXPECT_EQ(allocation.GetSpatialLayerSum(1), 0u);
>+}
>+
>+TEST(SvcRateAllocatorTest, Disable1280x720Layer) {
>+  VideoCodec codec = Configure(1280, 720, 3, 3, false);
>+  SvcRateAllocator allocator = SvcRateAllocator(codec);
>+
>+  const SpatialLayer* layers = codec.spatialLayers;
>+
>+  size_t min_bitrate_for_1280x720_layer_kbps =
>+      layers[0].minBitrate + layers[1].minBitrate + layers[2].minBitrate;
>+
>+  VideoBitrateAllocation allocation = allocator.GetAllocation(
>+      min_bitrate_for_1280x720_layer_kbps * 1000 - 1, 30);
>+
>+  EXPECT_GT(allocation.GetSpatialLayerSum(0), 0u);
>+  EXPECT_GT(allocation.GetSpatialLayerSum(1), 0u);
>+  EXPECT_EQ(allocation.GetSpatialLayerSum(2), 0u);
>+}
>+
>+TEST(SvcRateAllocatorTest, BitrateIsCapped) {
>+  VideoCodec codec = Configure(1280, 720, 3, 3, false);
>+  SvcRateAllocator allocator = SvcRateAllocator(codec);
>+
>+  const SpatialLayer* layers = codec.spatialLayers;
>+
>+  const uint32_t link_mbps = 100;
>+  VideoBitrateAllocation allocation =
>+      allocator.GetAllocation(link_mbps * 1000000, 30);
>+
>+  EXPECT_EQ(allocation.get_sum_kbps(),
>+            layers[0].maxBitrate + layers[1].maxBitrate + layers[2].maxBitrate);
>+  EXPECT_EQ(allocation.GetSpatialLayerSum(0) / 1000, layers[0].maxBitrate);
>+  EXPECT_EQ(allocation.GetSpatialLayerSum(1) / 1000, layers[1].maxBitrate);
>+  EXPECT_EQ(allocation.GetSpatialLayerSum(2) / 1000, layers[2].maxBitrate);
>+}
>+
>+TEST(SvcRateAllocatorTest, MinBitrateToGetQualityLayer) {
>+  VideoCodec codec = Configure(1280, 720, 3, 1, true);
>+  SvcRateAllocator allocator = SvcRateAllocator(codec);
>+
>+  const SpatialLayer* layers = codec.spatialLayers;
>+
>+  EXPECT_LE(codec.VP9()->numberOfSpatialLayers, 2U);
>+
>+  VideoBitrateAllocation allocation =
>+      allocator.GetAllocation(layers[0].minBitrate * 1000, 30);
>+  EXPECT_EQ(allocation.GetSpatialLayerSum(0) / 1000, layers[0].minBitrate);
>+  EXPECT_EQ(allocation.GetSpatialLayerSum(1), 0UL);
>+
>+  allocation = allocator.GetAllocation(
>+      (layers[0].maxBitrate + layers[1].minBitrate) * 1000, 30);
>+  EXPECT_EQ(allocation.GetSpatialLayerSum(0) / 1000, layers[0].maxBitrate);
>+  EXPECT_EQ(allocation.GetSpatialLayerSum(1) / 1000, layers[1].minBitrate);
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/test/vp9_impl_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/test/vp9_impl_unittest.cc
>index 3718707a10a..ec4efe86da5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/test/vp9_impl_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/test/vp9_impl_unittest.cc
>@@ -8,17 +8,24 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>+#include "api/video/color_space.h"
>+#include "api/video/i420_buffer.h"
> #include "common_video/libyuv/include/webrtc_libyuv.h"
>+#include "media/base/vp9_profile.h"
>+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>+#include "modules/video_coding/codecs/test/video_codec_unittest.h"
> #include "modules/video_coding/codecs/vp9/include/vp9.h"
>-#include "modules/video_coding/codecs/test/video_codec_test.h"
>+#include "modules/video_coding/codecs/vp9/svc_config.h"
>+#include "test/video_codec_settings.h"
> 
> namespace webrtc {
> 
> namespace {
>-constexpr uint32_t kTimestampIncrementPerFrame = 3000;
>+const size_t kWidth = 1280;
>+const size_t kHeight = 720;
> }  // namespace
> 
>-class TestVp9Impl : public VideoCodecTest {
>+class TestVp9Impl : public VideoCodecUnitTest {
>  protected:
>   std::unique_ptr<VideoEncoder> CreateEncoder() override {
>     return VP9Encoder::Create();
>@@ -28,24 +35,34 @@ class TestVp9Impl : public VideoCodecTest {
>     return VP9Decoder::Create();
>   }
> 
>-  VideoCodec codec_settings() override {
>-    VideoCodec codec_settings;
>-    codec_settings.codecType = webrtc::kVideoCodecVP9;
>-    codec_settings.VP9()->numberOfTemporalLayers = 1;
>-    codec_settings.VP9()->numberOfSpatialLayers = 1;
>-    return codec_settings;
>+  void ModifyCodecSettings(VideoCodec* codec_settings) override {
>+    webrtc::test::CodecSettings(kVideoCodecVP9, codec_settings);
>+    codec_settings->width = kWidth;
>+    codec_settings->height = kHeight;
>+    codec_settings->VP9()->numberOfTemporalLayers = 1;
>+    codec_settings->VP9()->numberOfSpatialLayers = 1;
>   }
> 
>-  void ExpectFrameWith(int16_t picture_id,
>-                       int tl0_pic_idx,
>-                       uint8_t temporal_idx) {
>+  void ExpectFrameWith(uint8_t temporal_idx) {
>     EncodedImage encoded_frame;
>     CodecSpecificInfo codec_specific_info;
>     ASSERT_TRUE(WaitForEncodedFrame(&encoded_frame, &codec_specific_info));
>-    EXPECT_EQ(picture_id, codec_specific_info.codecSpecific.VP9.picture_id);
>-    EXPECT_EQ(tl0_pic_idx, codec_specific_info.codecSpecific.VP9.tl0_pic_idx);
>     EXPECT_EQ(temporal_idx, codec_specific_info.codecSpecific.VP9.temporal_idx);
>   }
>+
>+  void ConfigureSvc(size_t num_spatial_layers) {
>+    codec_settings_.VP9()->numberOfSpatialLayers =
>+        static_cast<unsigned char>(num_spatial_layers);
>+    codec_settings_.VP9()->numberOfTemporalLayers = 1;
>+    codec_settings_.VP9()->frameDroppingOn = false;
>+
>+    std::vector<SpatialLayer> layers =
>+        GetSvcConfig(codec_settings_.width, codec_settings_.height,
>+                     num_spatial_layers, 1, false);
>+    for (size_t i = 0; i < layers.size(); ++i) {
>+      codec_settings_.spatialLayers[i] = layers[i];
>+    }
>+  }
> };
> 
> // Disabled on ios as flake, see https://crbug.com/webrtc/7057
>@@ -54,20 +71,27 @@ TEST_F(TestVp9Impl, DISABLED_EncodeDecode) {
> #else
> TEST_F(TestVp9Impl, EncodeDecode) {
> #endif
>+  VideoFrame* input_frame = NextInputFrame();
>   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>-            encoder_->Encode(*input_frame_, nullptr, nullptr));
>+            encoder_->Encode(*input_frame, nullptr, nullptr));
>   EncodedImage encoded_frame;
>   CodecSpecificInfo codec_specific_info;
>   ASSERT_TRUE(WaitForEncodedFrame(&encoded_frame, &codec_specific_info));
>   // First frame should be a key frame.
>   encoded_frame._frameType = kVideoFrameKey;
>   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>-            decoder_->Decode(encoded_frame, false, nullptr));
>+            decoder_->Decode(encoded_frame, false, nullptr, 0));
>   std::unique_ptr<VideoFrame> decoded_frame;
>-  rtc::Optional<uint8_t> decoded_qp;
>+  absl::optional<uint8_t> decoded_qp;
>   ASSERT_TRUE(WaitForDecodedFrame(&decoded_frame, &decoded_qp));
>   ASSERT_TRUE(decoded_frame);
>-  EXPECT_GT(I420PSNR(input_frame_.get(), decoded_frame.get()), 36);
>+  EXPECT_GT(I420PSNR(input_frame, decoded_frame.get()), 36);
>+
>+  const ColorSpace color_space = decoded_frame->color_space().value();
>+  EXPECT_EQ(ColorSpace::PrimaryID::kInvalid, color_space.primaries());
>+  EXPECT_EQ(ColorSpace::TransferID::kInvalid, color_space.transfer());
>+  EXPECT_EQ(ColorSpace::MatrixID::kInvalid, color_space.matrix());
>+  EXPECT_EQ(ColorSpace::RangeID::kLimited, color_space.range());
> }
> 
> // We only test the encoder here, since the decoded frame rotation is set based
>@@ -75,33 +99,34 @@ TEST_F(TestVp9Impl, EncodeDecode) {
> // TODO(brandtr): Consider passing through the rotation flag through the decoder
> // in the same way as done in the encoder.
> TEST_F(TestVp9Impl, EncodedRotationEqualsInputRotation) {
>-  input_frame_->set_rotation(kVideoRotation_0);
>+  VideoFrame* input_frame = NextInputFrame();
>+  input_frame->set_rotation(kVideoRotation_0);
>   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>-            encoder_->Encode(*input_frame_, nullptr, nullptr));
>+            encoder_->Encode(*input_frame, nullptr, nullptr));
>   EncodedImage encoded_frame;
>   CodecSpecificInfo codec_specific_info;
>   ASSERT_TRUE(WaitForEncodedFrame(&encoded_frame, &codec_specific_info));
>   EXPECT_EQ(kVideoRotation_0, encoded_frame.rotation_);
> 
>-  input_frame_->set_rotation(kVideoRotation_90);
>+  input_frame->set_rotation(kVideoRotation_90);
>   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>-            encoder_->Encode(*input_frame_, nullptr, nullptr));
>+            encoder_->Encode(*input_frame, nullptr, nullptr));
>   ASSERT_TRUE(WaitForEncodedFrame(&encoded_frame, &codec_specific_info));
>   EXPECT_EQ(kVideoRotation_90, encoded_frame.rotation_);
> }
> 
> TEST_F(TestVp9Impl, DecodedQpEqualsEncodedQp) {
>   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>-            encoder_->Encode(*input_frame_, nullptr, nullptr));
>+            encoder_->Encode(*NextInputFrame(), nullptr, nullptr));
>   EncodedImage encoded_frame;
>   CodecSpecificInfo codec_specific_info;
>   ASSERT_TRUE(WaitForEncodedFrame(&encoded_frame, &codec_specific_info));
>   // First frame should be a key frame.
>   encoded_frame._frameType = kVideoFrameKey;
>   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>-            decoder_->Decode(encoded_frame, false, nullptr));
>+            decoder_->Decode(encoded_frame, false, nullptr, 0));
>   std::unique_ptr<VideoFrame> decoded_frame;
>-  rtc::Optional<uint8_t> decoded_qp;
>+  absl::optional<uint8_t> decoded_qp;
>   ASSERT_TRUE(WaitForDecodedFrame(&decoded_frame, &decoded_qp));
>   ASSERT_TRUE(decoded_frame);
>   ASSERT_TRUE(decoded_qp);
>@@ -110,18 +135,17 @@ TEST_F(TestVp9Impl, DecodedQpEqualsEncodedQp) {
> 
> TEST_F(TestVp9Impl, ParserQpEqualsEncodedQp) {
>   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>-            encoder_->Encode(*input_frame_, nullptr, nullptr));
>+            encoder_->Encode(*NextInputFrame(), nullptr, nullptr));
>   EncodedImage encoded_frame;
>   CodecSpecificInfo codec_specific_info;
>   ASSERT_TRUE(WaitForEncodedFrame(&encoded_frame, &codec_specific_info));
> 
>   int qp = 0;
>   ASSERT_TRUE(vp9::GetQp(encoded_frame._buffer, encoded_frame._length, &qp));
>-
>   EXPECT_EQ(encoded_frame.qp_, qp);
> }
> 
>-TEST_F(TestVp9Impl, EncoderRetainsRtpStateAfterRelease) {
>+TEST_F(TestVp9Impl, EncoderWith2TemporalLayers) {
>   // Override default settings.
>   codec_settings_.VP9()->numberOfTemporalLayers = 2;
>   // Tl0PidIdx is only used in non-flexible mode.
>@@ -132,74 +156,379 @@ TEST_F(TestVp9Impl, EncoderRetainsRtpStateAfterRelease) {
> 
>   // Temporal layer 0.
>   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>-            encoder_->Encode(*input_frame_, nullptr, nullptr));
>+            encoder_->Encode(*NextInputFrame(), nullptr, nullptr));
>   EncodedImage encoded_frame;
>   CodecSpecificInfo codec_specific_info;
>   ASSERT_TRUE(WaitForEncodedFrame(&encoded_frame, &codec_specific_info));
>-  int16_t picture_id = codec_specific_info.codecSpecific.VP9.picture_id;
>-  int tl0_pic_idx = codec_specific_info.codecSpecific.VP9.tl0_pic_idx;
>   EXPECT_EQ(0, codec_specific_info.codecSpecific.VP9.temporal_idx);
> 
>   // Temporal layer 1.
>-  input_frame_->set_timestamp(input_frame_->timestamp() +
>-                              kTimestampIncrementPerFrame);
>   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>-            encoder_->Encode(*input_frame_, nullptr, nullptr));
>-  ExpectFrameWith((picture_id + 1) % (1 << 15), tl0_pic_idx, 1);
>+            encoder_->Encode(*NextInputFrame(), nullptr, nullptr));
>+  ExpectFrameWith(1);
> 
>   // Temporal layer 0.
>-  input_frame_->set_timestamp(input_frame_->timestamp() +
>-                              kTimestampIncrementPerFrame);
>   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>-            encoder_->Encode(*input_frame_, nullptr, nullptr));
>-  ExpectFrameWith((picture_id + 2) % (1 << 15), (tl0_pic_idx + 1) % (1 << 8),
>-                  0);
>+            encoder_->Encode(*NextInputFrame(), nullptr, nullptr));
>+  ExpectFrameWith(0);
> 
>   // Temporal layer 1.
>-  input_frame_->set_timestamp(input_frame_->timestamp() +
>-                              kTimestampIncrementPerFrame);
>   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>-            encoder_->Encode(*input_frame_, nullptr, nullptr));
>-  ExpectFrameWith((picture_id + 3) % (1 << 15), (tl0_pic_idx + 1) % (1 << 8),
>-                  1);
>+            encoder_->Encode(*NextInputFrame(), nullptr, nullptr));
>+  ExpectFrameWith(1);
>+}
> 
>-  // Reinit.
>-  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Release());
>+TEST_F(TestVp9Impl, EncoderExplicitLayering) {
>+  // Override default settings.
>+  codec_settings_.VP9()->numberOfTemporalLayers = 1;
>+  codec_settings_.VP9()->numberOfSpatialLayers = 2;
>+
>+  codec_settings_.width = 960;
>+  codec_settings_.height = 540;
>+  codec_settings_.spatialLayers[0].minBitrate = 200;
>+  codec_settings_.spatialLayers[0].maxBitrate = 500;
>+  codec_settings_.spatialLayers[0].targetBitrate =
>+      (codec_settings_.spatialLayers[0].minBitrate +
>+       codec_settings_.spatialLayers[0].maxBitrate) /
>+      2;
>+  codec_settings_.spatialLayers[1].minBitrate = 400;
>+  codec_settings_.spatialLayers[1].maxBitrate = 1500;
>+  codec_settings_.spatialLayers[1].targetBitrate =
>+      (codec_settings_.spatialLayers[1].minBitrate +
>+       codec_settings_.spatialLayers[1].maxBitrate) /
>+      2;
>+
>+  codec_settings_.spatialLayers[0].width = codec_settings_.width / 2;
>+  codec_settings_.spatialLayers[0].height = codec_settings_.height / 2;
>+  codec_settings_.spatialLayers[1].width = codec_settings_.width;
>+  codec_settings_.spatialLayers[1].height = codec_settings_.height;
>   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>             encoder_->InitEncode(&codec_settings_, 1 /* number of cores */,
>                                  0 /* max payload size (unused) */));
> 
>-  // Temporal layer 0.
>-  input_frame_->set_timestamp(input_frame_->timestamp() +
>-                              kTimestampIncrementPerFrame);
>+  // Ensure it fails if scaling factors in horz/vert dimentions are different.
>+  codec_settings_.spatialLayers[0].width = codec_settings_.width;
>+  codec_settings_.spatialLayers[0].height = codec_settings_.height / 2;
>+  codec_settings_.spatialLayers[1].width = codec_settings_.width;
>+  codec_settings_.spatialLayers[1].height = codec_settings_.height;
>+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_ERR_PARAMETER,
>+            encoder_->InitEncode(&codec_settings_, 1 /* number of cores */,
>+                                 0 /* max payload size (unused) */));
>+
>+  // Ensure it fails if scaling factor is not power of two.
>+  codec_settings_.spatialLayers[0].width = codec_settings_.width / 3;
>+  codec_settings_.spatialLayers[0].height = codec_settings_.height / 3;
>+  codec_settings_.spatialLayers[1].width = codec_settings_.width;
>+  codec_settings_.spatialLayers[1].height = codec_settings_.height;
>+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_ERR_PARAMETER,
>+            encoder_->InitEncode(&codec_settings_, 1 /* number of cores */,
>+                                 0 /* max payload size (unused) */));
>+}
>+
>+TEST_F(TestVp9Impl, EnableDisableSpatialLayers) {
>+  // Configure encoder to produce N spatial layers. Encode frames of layer 0
>+  // then enable layer 1 and encode more frames and so on until layer N-1.
>+  // Then disable layers one by one in the same way.
>+  // Note: bit rate allocation is high to avoid frame dropping due to rate
>+  // control, the encoder should always produce a frame. A dropped
>+  // frame indicates a problem and the test will fail.
>+  const size_t num_spatial_layers = 3;
>+  const size_t num_frames_to_encode = 5;
>+
>+  ConfigureSvc(num_spatial_layers);
>+  codec_settings_.VP9()->frameDroppingOn = true;
>+
>   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>-            encoder_->Encode(*input_frame_, nullptr, nullptr));
>-  ExpectFrameWith((picture_id + 4) % (1 << 15), (tl0_pic_idx + 2) % (1 << 8),
>-                  0);
>+            encoder_->InitEncode(&codec_settings_, 1 /* number of cores */,
>+                                 0 /* max payload size (unused) */));
>+
>+  VideoBitrateAllocation bitrate_allocation;
>+  for (size_t sl_idx = 0; sl_idx < num_spatial_layers; ++sl_idx) {
>+    // Allocate high bit rate to avoid frame dropping due to rate control.
>+    bitrate_allocation.SetBitrate(
>+        sl_idx, 0,
>+        codec_settings_.spatialLayers[sl_idx].targetBitrate * 1000 * 2);
>+    EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>+              encoder_->SetRateAllocation(bitrate_allocation,
>+                                          codec_settings_.maxFramerate));
>+
>+    for (size_t frame_num = 0; frame_num < num_frames_to_encode; ++frame_num) {
>+      SetWaitForEncodedFramesThreshold(sl_idx + 1);
>+      EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>+                encoder_->Encode(*NextInputFrame(), nullptr, nullptr));
>+      std::vector<EncodedImage> encoded_frame;
>+      std::vector<CodecSpecificInfo> codec_specific_info;
>+      ASSERT_TRUE(WaitForEncodedFrames(&encoded_frame, &codec_specific_info));
>+    }
>+  }
>+
>+  for (size_t i = 0; i < num_spatial_layers - 1; ++i) {
>+    const size_t sl_idx = num_spatial_layers - i - 1;
>+    bitrate_allocation.SetBitrate(sl_idx, 0, 0);
>+    EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>+              encoder_->SetRateAllocation(bitrate_allocation,
>+                                          codec_settings_.maxFramerate));
>+
>+    for (size_t frame_num = 0; frame_num < num_frames_to_encode; ++frame_num) {
>+      SetWaitForEncodedFramesThreshold(sl_idx);
>+      EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>+                encoder_->Encode(*NextInputFrame(), nullptr, nullptr));
>+      std::vector<EncodedImage> encoded_frame;
>+      std::vector<CodecSpecificInfo> codec_specific_info;
>+      ASSERT_TRUE(WaitForEncodedFrames(&encoded_frame, &codec_specific_info));
>+    }
>+  }
>+}
>+
>+TEST_F(TestVp9Impl, EndOfPicture) {
>+  const size_t num_spatial_layers = 2;
>+  ConfigureSvc(num_spatial_layers);
> 
>-  // Temporal layer 1.
>-  input_frame_->set_timestamp(input_frame_->timestamp() +
>-                              kTimestampIncrementPerFrame);
>   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>-            encoder_->Encode(*input_frame_, nullptr, nullptr));
>-  ExpectFrameWith((picture_id + 5) % (1 << 15), (tl0_pic_idx + 2) % (1 << 8),
>-                  1);
>+            encoder_->InitEncode(&codec_settings_, 1 /* number of cores */,
>+                                 0 /* max payload size (unused) */));
> 
>-  // Temporal layer 0.
>-  input_frame_->set_timestamp(input_frame_->timestamp() +
>-                              kTimestampIncrementPerFrame);
>+  // Encode both base and upper layers. Check that end-of-superframe flag is
>+  // set on upper layer frame but not on base layer frame.
>+  VideoBitrateAllocation bitrate_allocation;
>+  bitrate_allocation.SetBitrate(
>+      0, 0, codec_settings_.spatialLayers[0].targetBitrate * 1000);
>+  bitrate_allocation.SetBitrate(
>+      1, 0, codec_settings_.spatialLayers[1].targetBitrate * 1000);
>+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>+            encoder_->SetRateAllocation(bitrate_allocation,
>+                                        codec_settings_.maxFramerate));
>+  SetWaitForEncodedFramesThreshold(2);
>   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>-            encoder_->Encode(*input_frame_, nullptr, nullptr));
>-  ExpectFrameWith((picture_id + 6) % (1 << 15), (tl0_pic_idx + 3) % (1 << 8),
>-                  0);
>+            encoder_->Encode(*NextInputFrame(), nullptr, nullptr));
> 
>-  // Temporal layer 1.
>-  input_frame_->set_timestamp(input_frame_->timestamp() +
>-                              kTimestampIncrementPerFrame);
>+  std::vector<EncodedImage> frames;
>+  std::vector<CodecSpecificInfo> codec_specific;
>+  ASSERT_TRUE(WaitForEncodedFrames(&frames, &codec_specific));
>+  EXPECT_FALSE(codec_specific[0].codecSpecific.VP9.end_of_picture);
>+  EXPECT_TRUE(codec_specific[1].codecSpecific.VP9.end_of_picture);
>+
>+  // Encode only base layer. Check that end-of-superframe flag is
>+  // set on base layer frame.
>+  bitrate_allocation.SetBitrate(1, 0, 0);
>+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>+            encoder_->SetRateAllocation(bitrate_allocation,
>+                                        codec_settings_.maxFramerate));
>   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>-            encoder_->Encode(*input_frame_, nullptr, nullptr));
>-  ExpectFrameWith((picture_id + 7) % (1 << 15), (tl0_pic_idx + 3) % (1 << 8),
>-                  1);
>+            encoder_->InitEncode(&codec_settings_, 1 /* number of cores */,
>+                                 0 /* max payload size (unused) */));
>+
>+  SetWaitForEncodedFramesThreshold(1);
>+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>+            encoder_->Encode(*NextInputFrame(), nullptr, nullptr));
>+
>+  ASSERT_TRUE(WaitForEncodedFrames(&frames, &codec_specific));
>+  EXPECT_EQ(codec_specific[0].codecSpecific.VP9.spatial_idx, kNoSpatialIdx);
>+  EXPECT_TRUE(codec_specific[0].codecSpecific.VP9.end_of_picture);
>+}
>+
>+TEST_F(TestVp9Impl, InterLayerPred) {
>+  const size_t num_spatial_layers = 2;
>+  ConfigureSvc(num_spatial_layers);
>+  codec_settings_.VP9()->frameDroppingOn = false;
>+
>+  BitrateAllocation bitrate_allocation;
>+  for (size_t i = 0; i < num_spatial_layers; ++i) {
>+    bitrate_allocation.SetBitrate(
>+        i, 0, codec_settings_.spatialLayers[i].targetBitrate * 1000);
>+  }
>+
>+  const std::vector<InterLayerPredMode> inter_layer_pred_modes = {
>+      InterLayerPredMode::kOff, InterLayerPredMode::kOn,
>+      InterLayerPredMode::kOnKeyPic};
>+
>+  for (const InterLayerPredMode inter_layer_pred : inter_layer_pred_modes) {
>+    codec_settings_.VP9()->interLayerPred = inter_layer_pred;
>+    EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>+              encoder_->InitEncode(&codec_settings_, 1 /* number of cores */,
>+                                   0 /* max payload size (unused) */));
>+
>+    EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>+              encoder_->SetRateAllocation(bitrate_allocation,
>+                                          codec_settings_.maxFramerate));
>+
>+    SetWaitForEncodedFramesThreshold(2);
>+    EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>+              encoder_->Encode(*NextInputFrame(), nullptr, nullptr));
>+
>+    std::vector<EncodedImage> frames;
>+    std::vector<CodecSpecificInfo> codec_specific;
>+    ASSERT_TRUE(WaitForEncodedFrames(&frames, &codec_specific));
>+
>+    // Key frame.
>+    EXPECT_FALSE(codec_specific[0].codecSpecific.VP9.inter_pic_predicted);
>+    EXPECT_EQ(codec_specific[0].codecSpecific.VP9.spatial_idx, 0);
>+    EXPECT_EQ(codec_specific[0].codecSpecific.VP9.non_ref_for_inter_layer_pred,
>+              inter_layer_pred == InterLayerPredMode::kOff);
>+    EXPECT_TRUE(
>+        codec_specific[1].codecSpecific.VP9.non_ref_for_inter_layer_pred);
>+
>+    SetWaitForEncodedFramesThreshold(2);
>+    EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>+              encoder_->Encode(*NextInputFrame(), nullptr, nullptr));
>+    ASSERT_TRUE(WaitForEncodedFrames(&frames, &codec_specific));
>+
>+    // Delta frame.
>+    EXPECT_TRUE(codec_specific[0].codecSpecific.VP9.inter_pic_predicted);
>+    EXPECT_EQ(codec_specific[0].codecSpecific.VP9.spatial_idx, 0);
>+    EXPECT_EQ(codec_specific[0].codecSpecific.VP9.non_ref_for_inter_layer_pred,
>+              inter_layer_pred == InterLayerPredMode::kOff ||
>+                  inter_layer_pred == InterLayerPredMode::kOnKeyPic);
>+    EXPECT_TRUE(
>+        codec_specific[1].codecSpecific.VP9.non_ref_for_inter_layer_pred);
>+  }
>+}
>+
>+TEST_F(TestVp9Impl,
>+       EnablingUpperLayerTriggersKeyFrameIfInterLayerPredIsDisabled) {
>+  const size_t num_spatial_layers = 3;
>+  const size_t num_frames_to_encode = 2;
>+
>+  ConfigureSvc(num_spatial_layers);
>+  codec_settings_.VP9()->frameDroppingOn = false;
>+
>+  const std::vector<InterLayerPredMode> inter_layer_pred_modes = {
>+      InterLayerPredMode::kOff, InterLayerPredMode::kOn,
>+      InterLayerPredMode::kOnKeyPic};
>+
>+  for (const InterLayerPredMode inter_layer_pred : inter_layer_pred_modes) {
>+    codec_settings_.VP9()->interLayerPred = inter_layer_pred;
>+    EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>+              encoder_->InitEncode(&codec_settings_, 1 /* number of cores */,
>+                                   0 /* max payload size (unused) */));
>+
>+    VideoBitrateAllocation bitrate_allocation;
>+    for (size_t sl_idx = 0; sl_idx < num_spatial_layers; ++sl_idx) {
>+      bitrate_allocation.SetBitrate(
>+          sl_idx, 0,
>+          codec_settings_.spatialLayers[sl_idx].targetBitrate * 1000);
>+      EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>+                encoder_->SetRateAllocation(bitrate_allocation,
>+                                            codec_settings_.maxFramerate));
>+
>+      for (size_t frame_num = 0; frame_num < num_frames_to_encode;
>+           ++frame_num) {
>+        SetWaitForEncodedFramesThreshold(sl_idx + 1);
>+        EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>+                  encoder_->Encode(*NextInputFrame(), nullptr, nullptr));
>+        std::vector<EncodedImage> encoded_frame;
>+        std::vector<CodecSpecificInfo> codec_specific_info;
>+        ASSERT_TRUE(WaitForEncodedFrames(&encoded_frame, &codec_specific_info));
>+
>+        const bool is_first_upper_layer_frame = (sl_idx > 0 && frame_num == 0);
>+        if (is_first_upper_layer_frame) {
>+          if (inter_layer_pred == InterLayerPredMode::kOn) {
>+            EXPECT_EQ(encoded_frame[0]._frameType, kVideoFrameDelta);
>+          } else {
>+            EXPECT_EQ(encoded_frame[0]._frameType, kVideoFrameKey);
>+          }
>+        } else if (sl_idx == 0 && frame_num == 0) {
>+          EXPECT_EQ(encoded_frame[0]._frameType, kVideoFrameKey);
>+        } else {
>+          for (size_t i = 0; i <= sl_idx; ++i) {
>+            EXPECT_EQ(encoded_frame[i]._frameType, kVideoFrameDelta);
>+          }
>+        }
>+      }
>+    }
>+  }
>+}
>+
>+class TestVp9ImplFrameDropping : public TestVp9Impl {
>+ protected:
>+  void ModifyCodecSettings(VideoCodec* codec_settings) override {
>+    webrtc::test::CodecSettings(kVideoCodecVP9, codec_settings);
>+    // We need to encode quite a lot of frames in this test. Use low resolution
>+    // to reduce execution time.
>+    codec_settings->width = 64;
>+    codec_settings->height = 64;
>+    codec_settings->mode = VideoCodecMode::kScreensharing;
>+  }
>+};
>+
>+TEST_F(TestVp9ImplFrameDropping, PreEncodeFrameDropping) {
>+  const size_t num_frames_to_encode = 100;
>+  const float input_framerate_fps = 30.0;
>+  const float video_duration_secs = num_frames_to_encode / input_framerate_fps;
>+  const float expected_framerate_fps = 5.0f;
>+  const float max_abs_framerate_error_fps = expected_framerate_fps * 0.1f;
>+
>+  VideoFrame* input_frame = NextInputFrame();
>+  for (size_t frame_num = 0; frame_num < num_frames_to_encode; ++frame_num) {
>+    EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>+              encoder_->Encode(*input_frame, nullptr, nullptr));
>+    const size_t timestamp = input_frame->timestamp() +
>+                             kVideoPayloadTypeFrequency / input_framerate_fps;
>+    input_frame->set_timestamp(static_cast<uint32_t>(timestamp));
>+  }
>+
>+  const size_t num_encoded_frames = GetNumEncodedFrames();
>+  const float encoded_framerate_fps = num_encoded_frames / video_duration_secs;
>+  EXPECT_NEAR(encoded_framerate_fps, expected_framerate_fps,
>+              max_abs_framerate_error_fps);
>+}
>+
>+class TestVp9ImplProfile2 : public TestVp9Impl {
>+ protected:
>+  void SetUp() override {
>+    // Profile 2 might not be available on some platforms until
>+    // https://bugs.chromium.org/p/webm/issues/detail?id=1544 is solved.
>+    bool profile_2_is_supported = false;
>+    for (const auto& codec : SupportedVP9Codecs()) {
>+      if (ParseSdpForVP9Profile(codec.parameters)
>+              .value_or(VP9Profile::kProfile0) == VP9Profile::kProfile2) {
>+        profile_2_is_supported = true;
>+      }
>+    }
>+    if (!profile_2_is_supported)
>+      return;
>+
>+    TestVp9Impl::SetUp();
>+    input_frame_generator_ = test::FrameGenerator::CreateSquareGenerator(
>+        codec_settings_.width, codec_settings_.height,
>+        test::FrameGenerator::OutputType::I010, absl::optional<int>());
>+  }
>+
>+  std::unique_ptr<VideoEncoder> CreateEncoder() override {
>+    cricket::VideoCodec profile2_codec;
>+    profile2_codec.SetParam(kVP9FmtpProfileId,
>+                            VP9ProfileToString(VP9Profile::kProfile2));
>+    return VP9Encoder::Create(profile2_codec);
>+  }
>+
>+  std::unique_ptr<VideoDecoder> CreateDecoder() override {
>+    return VP9Decoder::Create();
>+  }
>+};
>+
>+TEST_F(TestVp9ImplProfile2, EncodeDecode) {
>+  if (!encoder_)
>+    return;
>+
>+  VideoFrame* input_frame = NextInputFrame();
>+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>+            encoder_->Encode(*input_frame, nullptr, nullptr));
>+  EncodedImage encoded_frame;
>+  CodecSpecificInfo codec_specific_info;
>+  ASSERT_TRUE(WaitForEncodedFrame(&encoded_frame, &codec_specific_info));
>+  // First frame should be a key frame.
>+  encoded_frame._frameType = kVideoFrameKey;
>+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>+            decoder_->Decode(encoded_frame, false, nullptr, 0));
>+  std::unique_ptr<VideoFrame> decoded_frame;
>+  absl::optional<uint8_t> decoded_qp;
>+  ASSERT_TRUE(WaitForDecodedFrame(&decoded_frame, &decoded_qp));
>+  ASSERT_TRUE(decoded_frame);
>+
>+  // TODO(emircan): Add PSNR for different color depths.
>+  EXPECT_GT(I420PSNR(*input_frame->video_frame_buffer()->ToI420(),
>+                     *decoded_frame->video_frame_buffer()->ToI420()),
>+            31);
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/vp9_frame_buffer_pool.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/vp9_frame_buffer_pool.h
>index fcb30af560d..9b035eb94ad 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/vp9_frame_buffer_pool.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/vp9_frame_buffer_pool.h
>@@ -14,7 +14,6 @@
> 
> #include <vector>
> 
>-#include "rtc_base/basictypes.h"
> #include "rtc_base/buffer.h"
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/refcount.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/vp9_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/vp9_impl.cc
>index 22993eb66aa..0cbc34e3728 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/vp9_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/vp9_impl.cc
>@@ -11,29 +11,33 @@
> 
> #include "modules/video_coding/codecs/vp9/vp9_impl.h"
> 
>-#include <stdlib.h>
>-#include <string.h>
>-#include <time.h>
>+#include <algorithm>
>+#include <limits>
> #include <vector>
> 
>-#include "vpx/vpx_encoder.h"
>-#include "vpx/vpx_decoder.h"
> #include "vpx/vp8cx.h"
> #include "vpx/vp8dx.h"
>+#include "vpx/vpx_decoder.h"
>+#include "vpx/vpx_encoder.h"
> 
>+#include "absl/memory/memory.h"
>+#include "api/video/color_space.h"
>+#include "api/video/i010_buffer.h"
> #include "common_video/include/video_frame_buffer.h"
> #include "common_video/libyuv/include/webrtc_libyuv.h"
>-#include "modules/video_coding/codecs/vp9/screenshare_layers.h"
>+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>+#include "modules/video_coding/codecs/vp9/svc_rate_allocator.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/keep_ref_until_done.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/ptr_util.h"
>-#include "rtc_base/random.h"
> #include "rtc_base/timeutils.h"
> #include "rtc_base/trace_event.h"
> 
> namespace webrtc {
> 
>+namespace {
>+const float kMaxScreenSharingFramerateFps = 5.0f;
>+
> // Only positive speeds, range for real-time coding currently is: 5 - 8.
> // Lower means slower/better quality, higher means fastest/lower quality.
> int GetCpuSpeed(int width, int height) {
>@@ -48,13 +52,84 @@ int GetCpuSpeed(int width, int height) {
>     return 7;
> #endif
> }
>-
>-bool VP9Encoder::IsSupported() {
>-  return true;
>+// Helper class for extracting VP9 colorspace.
>+ColorSpace ExtractVP9ColorSpace(vpx_color_space_t space_t,
>+                                vpx_color_range_t range_t,
>+                                unsigned int bit_depth) {
>+  ColorSpace::PrimaryID primaries = ColorSpace::PrimaryID::kInvalid;
>+  ColorSpace::TransferID transfer = ColorSpace::TransferID::kInvalid;
>+  ColorSpace::MatrixID matrix = ColorSpace::MatrixID::kInvalid;
>+  switch (space_t) {
>+    case VPX_CS_BT_601:
>+    case VPX_CS_SMPTE_170:
>+      primaries = ColorSpace::PrimaryID::kSMPTE170M;
>+      transfer = ColorSpace::TransferID::kSMPTE170M;
>+      matrix = ColorSpace::MatrixID::kSMPTE170M;
>+      break;
>+    case VPX_CS_SMPTE_240:
>+      primaries = ColorSpace::PrimaryID::kSMPTE240M;
>+      transfer = ColorSpace::TransferID::kSMPTE240M;
>+      matrix = ColorSpace::MatrixID::kSMPTE240M;
>+      break;
>+    case VPX_CS_BT_709:
>+      primaries = ColorSpace::PrimaryID::kBT709;
>+      transfer = ColorSpace::TransferID::kBT709;
>+      matrix = ColorSpace::MatrixID::kBT709;
>+      break;
>+    case VPX_CS_BT_2020:
>+      primaries = ColorSpace::PrimaryID::kBT2020;
>+      switch (bit_depth) {
>+        case 8:
>+          transfer = ColorSpace::TransferID::kBT709;
>+          break;
>+        case 10:
>+          transfer = ColorSpace::TransferID::kBT2020_10;
>+          break;
>+        default:
>+          RTC_NOTREACHED();
>+          break;
>+      }
>+      matrix = ColorSpace::MatrixID::kBT2020_NCL;
>+      break;
>+    case VPX_CS_SRGB:
>+      primaries = ColorSpace::PrimaryID::kBT709;
>+      transfer = ColorSpace::TransferID::kIEC61966_2_1;
>+      matrix = ColorSpace::MatrixID::kBT709;
>+      break;
>+    default:
>+      break;
>+  }
>+
>+  ColorSpace::RangeID range = ColorSpace::RangeID::kInvalid;
>+  switch (range_t) {
>+    case VPX_CR_STUDIO_RANGE:
>+      range = ColorSpace::RangeID::kLimited;
>+      break;
>+    case VPX_CR_FULL_RANGE:
>+      range = ColorSpace::RangeID::kFull;
>+      break;
>+    default:
>+      break;
>+  }
>+  return ColorSpace(primaries, transfer, matrix, range);
>+}
>+}  // namespace
>+
>+std::vector<SdpVideoFormat> SupportedVP9Codecs() {
>+  // TODO(emircan): Add Profile 2 support after fixing browser_tests.
>+  std::vector<SdpVideoFormat> supported_formats{SdpVideoFormat(
>+      cricket::kVp9CodecName,
>+      {{kVP9FmtpProfileId, VP9ProfileToString(VP9Profile::kProfile0)}})};
>+  return supported_formats;
> }
> 
> std::unique_ptr<VP9Encoder> VP9Encoder::Create() {
>-  return rtc::MakeUnique<VP9EncoderImpl>();
>+  return absl::make_unique<VP9EncoderImpl>(cricket::VideoCodec());
>+}
>+
>+std::unique_ptr<VP9Encoder> VP9Encoder::Create(
>+    const cricket::VideoCodec& codec) {
>+  return absl::make_unique<VP9EncoderImpl>(codec);
> }
> 
> void VP9EncoderImpl::EncoderOutputCodedPacketCallback(vpx_codec_cx_pkt* pkt,
>@@ -63,9 +138,11 @@ void VP9EncoderImpl::EncoderOutputCodedPacketCallback(vpx_codec_cx_pkt* pkt,
>   enc->GetEncodedLayerFrame(pkt);
> }
> 
>-VP9EncoderImpl::VP9EncoderImpl()
>+VP9EncoderImpl::VP9EncoderImpl(const cricket::VideoCodec& codec)
>     : encoded_image_(),
>       encoded_complete_callback_(nullptr),
>+      profile_(
>+          ParseSdpForVP9Profile(codec.params).value_or(VP9Profile::kProfile0)),
>       inited_(false),
>       timestamp_(0),
>       cpu_speed_(3),
>@@ -74,19 +151,17 @@ VP9EncoderImpl::VP9EncoderImpl()
>       config_(nullptr),
>       raw_(nullptr),
>       input_image_(nullptr),
>-      frames_since_kf_(0),
>+      force_key_frame_(true),
>+      pics_since_key_(0),
>       num_temporal_layers_(0),
>       num_spatial_layers_(0),
>-      is_flexible_mode_(false),
>-      frames_encoded_(0),
>-      // Use two spatial when screensharing with flexible mode.
>-      spatial_layer_(new ScreenshareLayersVP9(2)) {
>+      is_svc_(false),
>+      inter_layer_pred_(InterLayerPredMode::kOn),
>+      output_framerate_(1000.0, 1000.0),
>+      last_encoded_frame_rtp_timestamp_(0),
>+      is_flexible_mode_(false) {
>   memset(&codec_, 0, sizeof(codec_));
>   memset(&svc_params_, 0, sizeof(vpx_svc_extra_cfg_t));
>-
>-  Random random(rtc::TimeMicros());
>-  picture_id_ = random.Rand<uint16_t>() & 0x7FFF;
>-  tl0_pic_idx_ = random.Rand<uint8_t>();
> }
> 
> VP9EncoderImpl::~VP9EncoderImpl() {
>@@ -94,13 +169,17 @@ VP9EncoderImpl::~VP9EncoderImpl() {
> }
> 
> int VP9EncoderImpl::Release() {
>+  int ret_val = WEBRTC_VIDEO_CODEC_OK;
>+
>   if (encoded_image_._buffer != nullptr) {
>     delete[] encoded_image_._buffer;
>     encoded_image_._buffer = nullptr;
>   }
>   if (encoder_ != nullptr) {
>-    if (vpx_codec_destroy(encoder_)) {
>-      return WEBRTC_VIDEO_CODEC_MEMORY;
>+    if (inited_) {
>+      if (vpx_codec_destroy(encoder_)) {
>+        ret_val = WEBRTC_VIDEO_CODEC_MEMORY;
>+      }
>     }
>     delete encoder_;
>     encoder_ = nullptr;
>@@ -114,53 +193,63 @@ int VP9EncoderImpl::Release() {
>     raw_ = nullptr;
>   }
>   inited_ = false;
>-  return WEBRTC_VIDEO_CODEC_OK;
>+  return ret_val;
> }
> 
> bool VP9EncoderImpl::ExplicitlyConfiguredSpatialLayers() const {
>   // We check target_bitrate_bps of the 0th layer to see if the spatial layers
>   // (i.e. bitrates) were explicitly configured.
>-  return num_spatial_layers_ > 1 &&
>-         codec_.spatialLayers[0].target_bitrate_bps > 0;
>+  return num_spatial_layers_ > 1 && codec_.spatialLayers[0].targetBitrate > 0;
> }
> 
>-bool VP9EncoderImpl::SetSvcRates() {
>+bool VP9EncoderImpl::SetSvcRates(
>+    const VideoBitrateAllocation& bitrate_allocation) {
>   uint8_t i = 0;
> 
>+  config_->rc_target_bitrate = bitrate_allocation.get_sum_kbps();
>+
>+  num_active_spatial_layers_ = 0;
>+  for (i = 0; i < num_spatial_layers_; ++i)
>+    num_active_spatial_layers_ += bitrate_allocation.IsSpatialLayerUsed(i);
>+  RTC_DCHECK_GT(num_active_spatial_layers_, 0);
>+  RTC_DCHECK_LE(num_active_spatial_layers_, num_spatial_layers_);
>+
>   if (ExplicitlyConfiguredSpatialLayers()) {
>-    if (num_temporal_layers_ > 1) {
>-      RTC_LOG(LS_ERROR) << "Multiple temporal layers when manually specifying "
>-                           "spatial layers not implemented yet!";
>-      return false;
>-    }
>-    int total_bitrate_bps = 0;
>-    for (i = 0; i < num_spatial_layers_; ++i)
>-      total_bitrate_bps += codec_.spatialLayers[i].target_bitrate_bps;
>-    // If total bitrate differs now from what has been specified at the
>-    // beginning, update the bitrates in the same ratio as before.
>-    for (i = 0; i < num_spatial_layers_; ++i) {
>-      config_->ss_target_bitrate[i] = config_->layer_target_bitrate[i] =
>-          static_cast<int>(static_cast<int64_t>(config_->rc_target_bitrate) *
>-                           codec_.spatialLayers[i].target_bitrate_bps /
>-                           total_bitrate_bps);
>+    for (size_t sl_idx = 0; sl_idx < num_spatial_layers_; ++sl_idx) {
>+      const bool was_layer_enabled = (config_->ss_target_bitrate[sl_idx] > 0);
>+      config_->ss_target_bitrate[sl_idx] =
>+          bitrate_allocation.GetSpatialLayerSum(sl_idx) / 1000;
>+
>+      for (size_t tl_idx = 0; tl_idx < num_temporal_layers_; ++tl_idx) {
>+        config_->layer_target_bitrate[sl_idx * num_temporal_layers_ + tl_idx] =
>+            bitrate_allocation.GetTemporalLayerSum(sl_idx, tl_idx) / 1000;
>+      }
>+
>+      const bool is_layer_enabled = (config_->ss_target_bitrate[sl_idx] > 0);
>+      if (is_layer_enabled && !was_layer_enabled) {
>+        if (inter_layer_pred_ == InterLayerPredMode::kOff ||
>+            inter_layer_pred_ == InterLayerPredMode::kOnKeyPic) {
>+          // TODO(wemb:1526): remove key frame request when issue is fixed.
>+          force_key_frame_ = true;
>+        }
>+      }
>     }
>   } else {
>     float rate_ratio[VPX_MAX_LAYERS] = {0};
>     float total = 0;
>-
>     for (i = 0; i < num_spatial_layers_; ++i) {
>       if (svc_params_.scaling_factor_num[i] <= 0 ||
>           svc_params_.scaling_factor_den[i] <= 0) {
>         RTC_LOG(LS_ERROR) << "Scaling factors not specified!";
>         return false;
>       }
>-      rate_ratio[i] =
>-          static_cast<float>(svc_params_.scaling_factor_num[i]) /
>-          svc_params_.scaling_factor_den[i];
>+      rate_ratio[i] = static_cast<float>(svc_params_.scaling_factor_num[i]) /
>+                      svc_params_.scaling_factor_den[i];
>       total += rate_ratio[i];
>     }
> 
>     for (i = 0; i < num_spatial_layers_; ++i) {
>+      RTC_CHECK_GT(total, 0);
>       config_->ss_target_bitrate[i] = static_cast<unsigned int>(
>           config_->rc_target_bitrate * rate_ratio[i] / total);
>       if (num_temporal_layers_ == 1) {
>@@ -185,19 +274,11 @@ bool VP9EncoderImpl::SetSvcRates() {
>       }
>     }
>   }
>-
>-  // For now, temporal layers only supported when having one spatial layer.
>-  if (num_spatial_layers_ == 1) {
>-    for (i = 0; i < num_temporal_layers_; ++i) {
>-      config_->ts_target_bitrate[i] = config_->layer_target_bitrate[i];
>-    }
>-  }
>-
>   return true;
> }
> 
> int VP9EncoderImpl::SetRateAllocation(
>-    const BitrateAllocation& bitrate_allocation,
>+    const VideoBitrateAllocation& bitrate_allocation,
>     uint32_t frame_rate) {
>   if (!inited_) {
>     return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
>@@ -214,12 +295,9 @@ int VP9EncoderImpl::SetRateAllocation(
>     return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
>   }
> 
>-  // TODO(sprang): Actually use BitrateAllocation layer info.
>-  config_->rc_target_bitrate = bitrate_allocation.get_sum_kbps();
>   codec_.maxFramerate = frame_rate;
>-  spatial_layer_->ConfigureBitrate(bitrate_allocation.get_sum_kbps(), 0);
> 
>-  if (!SetSvcRates()) {
>+  if (!SetSvcRates(bitrate_allocation)) {
>     return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
>   }
> 
>@@ -273,10 +351,24 @@ int VP9EncoderImpl::InitEncode(const VideoCodec* inst,
>   }
> 
>   num_spatial_layers_ = inst->VP9().numberOfSpatialLayers;
>+  RTC_DCHECK_GT(num_spatial_layers_, 0);
>   num_temporal_layers_ = inst->VP9().numberOfTemporalLayers;
>   if (num_temporal_layers_ == 0)
>     num_temporal_layers_ = 1;
> 
>+  // Init framerate controller.
>+  output_framerate_.Reset();
>+  if (codec_.mode == VideoCodecMode::kScreensharing) {
>+    target_framerate_fps_ = kMaxScreenSharingFramerateFps;
>+  } else {
>+    target_framerate_fps_.reset();
>+  }
>+
>+  is_svc_ = (num_spatial_layers_ > 1 || num_temporal_layers_ > 1);
>+  // Flexible mode requires SVC to be enabled since libvpx API only allows
>+  // to get reference list in SVC mode.
>+  RTC_DCHECK(!inst->VP9().flexibleMode || is_svc_);
>+
>   // Allocate memory for encoded image
>   if (encoded_image_._buffer != nullptr) {
>     delete[] encoded_image_._buffer;
>@@ -285,19 +377,41 @@ int VP9EncoderImpl::InitEncode(const VideoCodec* inst,
>       CalcBufferSize(VideoType::kI420, codec_.width, codec_.height);
>   encoded_image_._buffer = new uint8_t[encoded_image_._size];
>   encoded_image_._completeFrame = true;
>-  // Creating a wrapper to the image - setting image data to nullptr. Actual
>-  // pointer will be set in encode. Setting align to 1, as it is meaningless
>-  // (actual memory is not allocated).
>-  raw_ = vpx_img_wrap(nullptr, VPX_IMG_FMT_I420, codec_.width, codec_.height, 1,
>-                      nullptr);
>   // Populate encoder configuration with default values.
>   if (vpx_codec_enc_config_default(vpx_codec_vp9_cx(), config_, 0)) {
>     return WEBRTC_VIDEO_CODEC_ERROR;
>   }
>+
>+  vpx_img_fmt img_fmt = VPX_IMG_FMT_NONE;
>+  unsigned int bits_for_storage = 8;
>+  switch (profile_) {
>+    case VP9Profile::kProfile0:
>+      img_fmt = VPX_IMG_FMT_I420;
>+      bits_for_storage = 8;
>+      config_->g_bit_depth = VPX_BITS_8;
>+      config_->g_profile = 0;
>+      config_->g_input_bit_depth = 8;
>+      break;
>+    case VP9Profile::kProfile2:
>+      img_fmt = VPX_IMG_FMT_I42016;
>+      bits_for_storage = 16;
>+      config_->g_bit_depth = VPX_BITS_10;
>+      config_->g_profile = 2;
>+      config_->g_input_bit_depth = 10;
>+      break;
>+  }
>+
>+  // Creating a wrapper to the image - setting image data to nullptr. Actual
>+  // pointer will be set in encode. Setting align to 1, as it is meaningless
>+  // (actual memory is not allocated).
>+  raw_ =
>+      vpx_img_wrap(nullptr, img_fmt, codec_.width, codec_.height, 1, nullptr);
>+  raw_->bit_depth = bits_for_storage;
>+
>   config_->g_w = codec_.width;
>   config_->g_h = codec_.height;
>   config_->rc_target_bitrate = inst->startBitrate;  // in kbit/s
>-  config_->g_error_resilient = inst->VP9().resilienceOn ? 1 : 0;
>+  config_->g_error_resilient = is_svc_ ? VPX_ERROR_RESILIENT_DEFAULT : 0;
>   // Setting the time base of the codec.
>   config_->g_timebase.num = 1;
>   config_->g_timebase.den = 90000;
>@@ -335,12 +449,11 @@ int VP9EncoderImpl::InitEncode(const VideoCodec* inst,
>   // TODO(asapersson): Check configuration of temporal switch up and increase
>   // pattern length.
>   is_flexible_mode_ = inst->VP9().flexibleMode;
>-  if (is_flexible_mode_) {
>-    config_->temporal_layering_mode = VP9E_TEMPORAL_LAYERING_MODE_BYPASS;
>-    config_->ts_number_layers = num_temporal_layers_;
>-    if (codec_.mode == kScreensharing)
>-      spatial_layer_->ConfigureBitrate(inst->startBitrate, 0);
>-  } else if (num_temporal_layers_ == 1) {
>+
>+  // TODO(ssilkin): Only non-flexible mode is supported for now.
>+  RTC_DCHECK(!is_flexible_mode_);
>+
>+  if (num_temporal_layers_ == 1) {
>     gof_.SetGofInfoVP9(kTemporalStructureMode1);
>     config_->temporal_layering_mode = VP9E_TEMPORAL_LAYERING_MODE_NOLAYERING;
>     config_->ts_number_layers = 1;
>@@ -372,6 +485,10 @@ int VP9EncoderImpl::InitEncode(const VideoCodec* inst,
>     return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
>   }
> 
>+  inter_layer_pred_ = inst->VP9().interLayerPred;
>+
>+  ref_buf_.clear();
>+
>   return InitAndSetControlSettings(inst);
> }
> 
>@@ -385,7 +502,7 @@ int VP9EncoderImpl::NumberOfThreads(int width,
>   } else if (width * height >= 640 * 360 && number_of_cores > 2) {
>     return 2;
>   } else {
>-    // Use 2 threads for low res on ARM.
>+// Use 2 threads for low res on ARM.
> #if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64) || \
>     defined(WEBRTC_ANDROID)
>     if (width * height >= 320 * 180 && number_of_cores > 2) {
>@@ -408,8 +525,28 @@ int VP9EncoderImpl::InitAndSetControlSettings(const VideoCodec* inst) {
>   if (ExplicitlyConfiguredSpatialLayers()) {
>     for (int i = 0; i < num_spatial_layers_; ++i) {
>       const auto& layer = codec_.spatialLayers[i];
>-      svc_params_.scaling_factor_num[i] = layer.scaling_factor_num;
>-      svc_params_.scaling_factor_den[i] = layer.scaling_factor_den;
>+      RTC_CHECK_GT(layer.width, 0);
>+      const int scale_factor = codec_.width / layer.width;
>+      RTC_DCHECK_GT(scale_factor, 0);
>+
>+      // Ensure scaler factor is integer.
>+      if (scale_factor * layer.width != codec_.width) {
>+        return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
>+      }
>+
>+      // Ensure scale factor is the same in both dimensions.
>+      if (scale_factor * layer.height != codec_.height) {
>+        return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
>+      }
>+
>+      // Ensure scale factor is power of two.
>+      const bool is_pow_of_two = (scale_factor & (scale_factor - 1)) == 0;
>+      if (!is_pow_of_two) {
>+        return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
>+      }
>+
>+      svc_params_.scaling_factor_num[i] = 1;
>+      svc_params_.scaling_factor_den[i] = scale_factor;
>     }
>   } else {
>     int scaling_factor_num = 256;
>@@ -417,16 +554,21 @@ int VP9EncoderImpl::InitAndSetControlSettings(const VideoCodec* inst) {
>       // 1:2 scaling in each dimension.
>       svc_params_.scaling_factor_num[i] = scaling_factor_num;
>       svc_params_.scaling_factor_den[i] = 256;
>-      if (codec_.mode != kScreensharing)
>-        scaling_factor_num /= 2;
>     }
>   }
> 
>-  if (!SetSvcRates()) {
>+  SvcRateAllocator init_allocator(codec_);
>+  VideoBitrateAllocation allocation = init_allocator.GetAllocation(
>+      inst->startBitrate * 1000, inst->maxFramerate);
>+  if (!SetSvcRates(allocation)) {
>     return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
>   }
> 
>-  if (vpx_codec_enc_init(encoder_, vpx_codec_vp9_cx(), config_, 0)) {
>+  const vpx_codec_err_t rv = vpx_codec_enc_init(
>+      encoder_, vpx_codec_vp9_cx(), config_,
>+      config_->g_bit_depth == VPX_BITS_8 ? 0 : VPX_CODEC_USE_HIGHBITDEPTH);
>+  if (rv != VPX_CODEC_OK) {
>+    RTC_LOG(LS_ERROR) << "Init error: " << vpx_codec_err_to_string(rv);
>     return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
>   }
>   vpx_codec_control(encoder_, VP8E_SET_CPUUSED, cpu_speed_);
>@@ -436,13 +578,40 @@ int VP9EncoderImpl::InitAndSetControlSettings(const VideoCodec* inst) {
>                     inst->VP9().adaptiveQpMode ? 3 : 0);
> 
>   vpx_codec_control(encoder_, VP9E_SET_FRAME_PARALLEL_DECODING, 0);
>-  vpx_codec_control(
>-      encoder_, VP9E_SET_SVC,
>-      (num_temporal_layers_ > 1 || num_spatial_layers_ > 1) ? 1 : 0);
>-  if (num_temporal_layers_ > 1 || num_spatial_layers_ > 1) {
>-    vpx_codec_control(encoder_, VP9E_SET_SVC_PARAMETERS,
>-                      &svc_params_);
>+
>+  if (is_svc_) {
>+    vpx_codec_control(encoder_, VP9E_SET_SVC, 1);
>+    vpx_codec_control(encoder_, VP9E_SET_SVC_PARAMETERS, &svc_params_);
>+  }
>+
>+  if (num_spatial_layers_ > 1) {
>+    switch (inter_layer_pred_) {
>+      case InterLayerPredMode::kOn:
>+        vpx_codec_control(encoder_, VP9E_SET_SVC_INTER_LAYER_PRED, 0);
>+        break;
>+      case InterLayerPredMode::kOff:
>+        vpx_codec_control(encoder_, VP9E_SET_SVC_INTER_LAYER_PRED, 1);
>+        break;
>+      case InterLayerPredMode::kOnKeyPic:
>+        vpx_codec_control(encoder_, VP9E_SET_SVC_INTER_LAYER_PRED, 2);
>+        break;
>+      default:
>+        RTC_NOTREACHED();
>+    }
>+
>+    // Configure encoder to drop entire superframe whenever it needs to drop
>+    // a layer. This mode is prefered over per-layer dropping which causes
>+    // quality flickering and is not compatible with RTP non-flexible mode.
>+    vpx_svc_frame_drop_t svc_drop_frame;
>+    memset(&svc_drop_frame, 0, sizeof(svc_drop_frame));
>+    svc_drop_frame.framedrop_mode = FULL_SUPERFRAME_DROP;
>+    svc_drop_frame.max_consec_drop = std::numeric_limits<int>::max();
>+    for (size_t i = 0; i < num_spatial_layers_; ++i) {
>+      svc_drop_frame.framedrop_thresh[i] = config_->rc_dropframe_thresh;
>+    }
>+    vpx_codec_control(encoder_, VP9E_SET_SVC_FRAME_DROP_LAYER, &svc_drop_frame);
>   }
>+
>   // Register callback for getting each spatial layer.
>   vpx_codec_priv_output_cx_pkt_cb_pair_t cbp = {
>       VP9EncoderImpl::EncoderOutputCodedPacketCallback,
>@@ -460,14 +629,14 @@ int VP9EncoderImpl::InitAndSetControlSettings(const VideoCodec* inst) {
>   vpx_codec_control(encoder_, VP9E_SET_ROW_MT, 1);
> 
> #if !defined(WEBRTC_ARCH_ARM) && !defined(WEBRTC_ARCH_ARM64) && \
>-  !defined(ANDROID)
>+    !defined(ANDROID)
>   // Do not enable the denoiser on ARM since optimization is pending.
>   // Denoiser is on by default on other platforms.
>   vpx_codec_control(encoder_, VP9E_SET_NOISE_SENSITIVITY,
>                     inst->VP9().denoisingOn ? 1 : 0);
> #endif
> 
>-  if (codec_.mode == kScreensharing) {
>+  if (codec_.mode == VideoCodecMode::kScreensharing) {
>     // Adjust internal parameters to screen content.
>     vpx_codec_control(encoder_, VP9E_SET_TUNE_CONTENT, 1);
>   }
>@@ -501,11 +670,20 @@ int VP9EncoderImpl::Encode(const VideoFrame& input_image,
>   if (encoded_complete_callback_ == nullptr) {
>     return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
>   }
>-  FrameType frame_type = kVideoFrameDelta;
>+
>   // We only support one stream at the moment.
>-  if (frame_types && frame_types->size() > 0) {
>-    frame_type = (*frame_types)[0];
>+  if (frame_types && !frame_types->empty()) {
>+    if ((*frame_types)[0] == kVideoFrameKey) {
>+      force_key_frame_ = true;
>+    }
>   }
>+
>+  if (VideoCodecMode::kScreensharing == codec_.mode && !force_key_frame_) {
>+    if (DropFrame(input_image.timestamp())) {
>+      return WEBRTC_VIDEO_CODEC_OK;
>+    }
>+  }
>+
>   RTC_DCHECK_EQ(input_image.width(), raw_->d_w);
>   RTC_DCHECK_EQ(input_image.height(), raw_->d_h);
> 
>@@ -515,67 +693,83 @@ int VP9EncoderImpl::Encode(const VideoFrame& input_image,
>   // doing this.
>   input_image_ = &input_image;
> 
>-  rtc::scoped_refptr<I420BufferInterface> i420_buffer =
>-      input_image.video_frame_buffer()->ToI420();
>-  // Image in vpx_image_t format.
>-  // Input image is const. VPX's raw image is not defined as const.
>-  raw_->planes[VPX_PLANE_Y] = const_cast<uint8_t*>(i420_buffer->DataY());
>-  raw_->planes[VPX_PLANE_U] = const_cast<uint8_t*>(i420_buffer->DataU());
>-  raw_->planes[VPX_PLANE_V] = const_cast<uint8_t*>(i420_buffer->DataV());
>-  raw_->stride[VPX_PLANE_Y] = i420_buffer->StrideY();
>-  raw_->stride[VPX_PLANE_U] = i420_buffer->StrideU();
>-  raw_->stride[VPX_PLANE_V] = i420_buffer->StrideV();
>+  // Keep reference to buffer until encode completes.
>+  rtc::scoped_refptr<I420BufferInterface> i420_buffer;
>+  rtc::scoped_refptr<I010BufferInterface> i010_buffer;
>+  switch (profile_) {
>+    case VP9Profile::kProfile0: {
>+      i420_buffer = input_image.video_frame_buffer()->ToI420();
>+      // Image in vpx_image_t format.
>+      // Input image is const. VPX's raw image is not defined as const.
>+      raw_->planes[VPX_PLANE_Y] = const_cast<uint8_t*>(i420_buffer->DataY());
>+      raw_->planes[VPX_PLANE_U] = const_cast<uint8_t*>(i420_buffer->DataU());
>+      raw_->planes[VPX_PLANE_V] = const_cast<uint8_t*>(i420_buffer->DataV());
>+      raw_->stride[VPX_PLANE_Y] = i420_buffer->StrideY();
>+      raw_->stride[VPX_PLANE_U] = i420_buffer->StrideU();
>+      raw_->stride[VPX_PLANE_V] = i420_buffer->StrideV();
>+      break;
>+    }
>+    case VP9Profile::kProfile2: {
>+      // We can inject kI010 frames directly for encode. All other formats
>+      // should be converted to it.
>+      switch (input_image.video_frame_buffer()->type()) {
>+        case VideoFrameBuffer::Type::kI010: {
>+          i010_buffer = input_image.video_frame_buffer()->GetI010();
>+          break;
>+        }
>+        default: {
>+          i010_buffer =
>+              I010Buffer::Copy(*input_image.video_frame_buffer()->ToI420());
>+        }
>+      }
>+      raw_->planes[VPX_PLANE_Y] = const_cast<uint8_t*>(
>+          reinterpret_cast<const uint8_t*>(i010_buffer->DataY()));
>+      raw_->planes[VPX_PLANE_U] = const_cast<uint8_t*>(
>+          reinterpret_cast<const uint8_t*>(i010_buffer->DataU()));
>+      raw_->planes[VPX_PLANE_V] = const_cast<uint8_t*>(
>+          reinterpret_cast<const uint8_t*>(i010_buffer->DataV()));
>+      raw_->stride[VPX_PLANE_Y] = i010_buffer->StrideY() * 2;
>+      raw_->stride[VPX_PLANE_U] = i010_buffer->StrideU() * 2;
>+      raw_->stride[VPX_PLANE_V] = i010_buffer->StrideV() * 2;
>+      break;
>+    }
>+  }
> 
>   vpx_enc_frame_flags_t flags = 0;
>-  bool send_keyframe = (frame_type == kVideoFrameKey);
>-  if (send_keyframe) {
>-    // Key frame request from caller.
>+  if (force_key_frame_) {
>     flags = VPX_EFLAG_FORCE_KF;
>   }
> 
>-  if (is_flexible_mode_) {
>-    SuperFrameRefSettings settings;
>-
>-    // These structs are copied when calling vpx_codec_control,
>-    // therefore it is ok for them to go out of scope.
>-    vpx_svc_ref_frame_config enc_layer_conf;
>-    vpx_svc_layer_id layer_id;
>-
>-    if (codec_.mode == kRealtimeVideo) {
>-      // Real time video not yet implemented in flexible mode.
>-      RTC_NOTREACHED();
>-    } else {
>-      settings = spatial_layer_->GetSuperFrameSettings(input_image.timestamp(),
>-                                                       send_keyframe);
>-    }
>-    enc_layer_conf = GenerateRefsAndFlags(settings);
>-    layer_id.temporal_layer_id = 0;
>-    layer_id.spatial_layer_id = settings.start_layer;
>-    vpx_codec_control(encoder_, VP9E_SET_SVC_LAYER_ID, &layer_id);
>-    vpx_codec_control(encoder_, VP9E_SET_SVC_REF_FRAME_CONFIG, &enc_layer_conf);
>-  }
>-
>   RTC_CHECK_GT(codec_.maxFramerate, 0);
>-  uint32_t duration = 90000 / codec_.maxFramerate;
>-  if (vpx_codec_encode(encoder_, raw_, timestamp_, duration, flags,
>-                       VPX_DL_REALTIME)) {
>+  uint32_t duration =
>+      90000 / target_framerate_fps_.value_or(codec_.maxFramerate);
>+  const vpx_codec_err_t rv = vpx_codec_encode(encoder_, raw_, timestamp_,
>+                                              duration, flags, VPX_DL_REALTIME);
>+  if (rv != VPX_CODEC_OK) {
>+    RTC_LOG(LS_ERROR) << "Encoding error: " << vpx_codec_err_to_string(rv)
>+                      << "\n"
>+                      << "Details: " << vpx_codec_error(encoder_) << "\n"
>+                      << vpx_codec_error_detail(encoder_);
>     return WEBRTC_VIDEO_CODEC_ERROR;
>   }
>   timestamp_ += duration;
> 
>+  const bool end_of_picture = true;
>+  DeliverBufferedFrame(end_of_picture);
>+
>   return WEBRTC_VIDEO_CODEC_OK;
> }
> 
> void VP9EncoderImpl::PopulateCodecSpecific(CodecSpecificInfo* codec_specific,
>                                            const vpx_codec_cx_pkt& pkt,
>-                                           uint32_t timestamp) {
>+                                           uint32_t timestamp,
>+                                           bool first_frame_in_picture) {
>   RTC_CHECK(codec_specific != nullptr);
>   codec_specific->codecType = kVideoCodecVP9;
>   codec_specific->codec_name = ImplementationName();
>   CodecSpecificInfoVP9* vp9_info = &(codec_specific->codecSpecific.VP9);
>-  // TODO(asapersson): Set correct value.
>-  vp9_info->inter_pic_predicted =
>-      (pkt.data.frame.flags & VPX_FRAME_IS_KEY) ? false : true;
>+
>+  vp9_info->first_frame_in_picture = first_frame_in_picture;
>   vp9_info->flexible_mode = codec_.VP9()->flexibleMode;
>   vp9_info->ss_data_available =
>       ((pkt.data.frame.flags & VPX_FRAME_IS_KEY) && !codec_.VP9()->flexibleMode)
>@@ -586,14 +780,14 @@ void VP9EncoderImpl::PopulateCodecSpecific(CodecSpecificInfo* codec_specific,
>   vpx_codec_control(encoder_, VP9E_GET_SVC_LAYER_ID, &layer_id);
> 
>   RTC_CHECK_GT(num_temporal_layers_, 0);
>-  RTC_CHECK_GT(num_spatial_layers_, 0);
>+  RTC_CHECK_GT(num_active_spatial_layers_, 0);
>   if (num_temporal_layers_ == 1) {
>     RTC_CHECK_EQ(layer_id.temporal_layer_id, 0);
>     vp9_info->temporal_idx = kNoTemporalIdx;
>   } else {
>     vp9_info->temporal_idx = layer_id.temporal_layer_id;
>   }
>-  if (num_spatial_layers_ == 1) {
>+  if (num_active_spatial_layers_ == 1) {
>     RTC_CHECK_EQ(layer_id.spatial_layer_id, 0);
>     vp9_info->spatial_idx = kNoSpatialIdx;
>   } else {
>@@ -606,62 +800,57 @@ void VP9EncoderImpl::PopulateCodecSpecific(CodecSpecificInfo* codec_specific,
>   // TODO(asapersson): this info has to be obtained from the encoder.
>   vp9_info->temporal_up_switch = false;
> 
>-  bool is_first_frame = false;
>-  if (is_flexible_mode_) {
>-    is_first_frame =
>-        layer_id.spatial_layer_id == spatial_layer_->GetStartLayer();
>-  } else {
>-    is_first_frame = layer_id.spatial_layer_id == 0;
>-  }
>-
>-  if (is_first_frame) {
>-    picture_id_ = (picture_id_ + 1) & 0x7FFF;
>-    // TODO(asapersson): this info has to be obtained from the encoder.
>-    vp9_info->inter_layer_predicted = false;
>-    ++frames_since_kf_;
>-  } else {
>-    // TODO(asapersson): this info has to be obtained from the encoder.
>-    vp9_info->inter_layer_predicted = true;
>-  }
>-
>   if (pkt.data.frame.flags & VPX_FRAME_IS_KEY) {
>-    frames_since_kf_ = 0;
>-  }
>-
>-  vp9_info->picture_id = picture_id_;
>-
>-  if (!vp9_info->flexible_mode) {
>-    if (layer_id.temporal_layer_id == 0 && layer_id.spatial_layer_id == 0) {
>-      tl0_pic_idx_++;
>-    }
>-    vp9_info->tl0_pic_idx = tl0_pic_idx_;
>-  }
>+    pics_since_key_ = 0;
>+  } else if (first_frame_in_picture) {
>+    ++pics_since_key_;
>+  }
>+
>+  const bool is_key_pic = (pics_since_key_ == 0);
>+  const bool is_inter_layer_pred_allowed =
>+      (inter_layer_pred_ == InterLayerPredMode::kOn ||
>+       (inter_layer_pred_ == InterLayerPredMode::kOnKeyPic && is_key_pic));
>+
>+  // Always set inter_layer_predicted to true on high layer frame if inter-layer
>+  // prediction (ILP) is allowed even if encoder didn't actually use it.
>+  // Setting inter_layer_predicted to false would allow receiver to decode high
>+  // layer frame without decoding low layer frame. If that would happen (e.g.
>+  // if low layer frame is lost) then receiver won't be able to decode next high
>+  // layer frame which uses ILP.
>+  vp9_info->inter_layer_predicted =
>+      first_frame_in_picture ? false : is_inter_layer_pred_allowed;
>+
>+  const bool is_last_layer =
>+      (layer_id.spatial_layer_id + 1 == num_active_spatial_layers_);
>+  vp9_info->non_ref_for_inter_layer_pred =
>+      is_last_layer ? true : !is_inter_layer_pred_allowed;
> 
>   // Always populate this, so that the packetizer can properly set the marker
>   // bit.
>-  vp9_info->num_spatial_layers = num_spatial_layers_;
>+  vp9_info->num_spatial_layers = num_active_spatial_layers_;
>+
>+  RTC_DCHECK(!vp9_info->flexible_mode);
> 
>   vp9_info->num_ref_pics = 0;
>   if (vp9_info->flexible_mode) {
>     vp9_info->gof_idx = kNoGofIdx;
>-    vp9_info->num_ref_pics = num_ref_pics_[layer_id.spatial_layer_id];
>-    for (int i = 0; i < num_ref_pics_[layer_id.spatial_layer_id]; ++i) {
>-      vp9_info->p_diff[i] = p_diff_[layer_id.spatial_layer_id][i];
>-    }
>+    FillReferenceIndices(pkt, pics_since_key_, vp9_info->inter_layer_predicted,
>+                         vp9_info);
>   } else {
>     vp9_info->gof_idx =
>-        static_cast<uint8_t>(frames_since_kf_ % gof_.num_frames_in_gof);
>+        static_cast<uint8_t>(pics_since_key_ % gof_.num_frames_in_gof);
>     vp9_info->temporal_up_switch = gof_.temporal_up_switch[vp9_info->gof_idx];
>+    vp9_info->num_ref_pics = gof_.num_ref_pics[vp9_info->gof_idx];
>   }
> 
>+  vp9_info->inter_pic_predicted = (!is_key_pic && vp9_info->num_ref_pics > 0);
>+
>   if (vp9_info->ss_data_available) {
>     vp9_info->spatial_layer_resolution_present = true;
>-    for (size_t i = 0; i < vp9_info->num_spatial_layers; ++i) {
>-      vp9_info->width[i] = codec_.width *
>-                           svc_params_.scaling_factor_num[i] /
>+    for (size_t i = 0; i < num_active_spatial_layers_; ++i) {
>+      vp9_info->width[i] = codec_.width * svc_params_.scaling_factor_num[i] /
>                            svc_params_.scaling_factor_den[i];
>-      vp9_info->height[i] = codec_.height *
>-                            svc_params_.scaling_factor_num[i] /
>+      vp9_info->height[i] = codec_.height * svc_params_.scaling_factor_num[i] /
>                             svc_params_.scaling_factor_den[i];
>     }
>     if (!vp9_info->flexible_mode) {
>@@ -670,9 +859,125 @@ void VP9EncoderImpl::PopulateCodecSpecific(CodecSpecificInfo* codec_specific,
>   }
> }
> 
>+void VP9EncoderImpl::FillReferenceIndices(const vpx_codec_cx_pkt& pkt,
>+                                          const size_t pic_num,
>+                                          const bool inter_layer_predicted,
>+                                          CodecSpecificInfoVP9* vp9_info) {
>+  vpx_svc_layer_id_t layer_id = {0};
>+  vpx_codec_control(encoder_, VP9E_GET_SVC_LAYER_ID, &layer_id);
>+
>+  vpx_svc_ref_frame_config_t enc_layer_conf = {{0}};
>+  vpx_codec_control(encoder_, VP9E_GET_SVC_REF_FRAME_CONFIG, &enc_layer_conf);
>+
>+  std::vector<RefFrameBuffer> ref_buf_list;
>+  if (enc_layer_conf.reference_last[layer_id.spatial_layer_id]) {
>+    const size_t fb_idx = enc_layer_conf.lst_fb_idx[layer_id.spatial_layer_id];
>+    RTC_DCHECK(ref_buf_.find(fb_idx) != ref_buf_.end());
>+    ref_buf_list.push_back(ref_buf_.at(fb_idx));
>+  }
>+
>+  if (enc_layer_conf.reference_alt_ref[layer_id.spatial_layer_id]) {
>+    const size_t fb_idx = enc_layer_conf.alt_fb_idx[layer_id.spatial_layer_id];
>+    RTC_DCHECK(ref_buf_.find(fb_idx) != ref_buf_.end());
>+    ref_buf_list.push_back(ref_buf_.at(fb_idx));
>+  }
>+
>+  if (enc_layer_conf.reference_golden[layer_id.spatial_layer_id]) {
>+    const size_t fb_idx = enc_layer_conf.gld_fb_idx[layer_id.spatial_layer_id];
>+    RTC_DCHECK(ref_buf_.find(fb_idx) != ref_buf_.end());
>+    ref_buf_list.push_back(ref_buf_.at(fb_idx));
>+  }
>+
>+  size_t max_ref_temporal_layer_id = 0;
>+
>+  vp9_info->num_ref_pics = 0;
>+  for (const RefFrameBuffer& ref_buf : ref_buf_list) {
>+    RTC_DCHECK_LE(ref_buf.pic_num, pic_num);
>+    if (ref_buf.pic_num < pic_num) {
>+      if (inter_layer_pred_ != InterLayerPredMode::kOn) {
>+        // RTP spec limits temporal prediction to the same spatial layer.
>+        // It is safe to ignore this requirement if inter-layer prediction is
>+        // enabled for all frames when all base frames are relayed to receiver.
>+        RTC_DCHECK_EQ(ref_buf.spatial_layer_id, layer_id.spatial_layer_id);
>+      }
>+      RTC_DCHECK_LE(ref_buf.temporal_layer_id, layer_id.temporal_layer_id);
>+
>+      const size_t p_diff = pic_num - ref_buf.pic_num;
>+      RTC_DCHECK_LE(p_diff, 127UL);
>+
>+      vp9_info->p_diff[vp9_info->num_ref_pics] = static_cast<uint8_t>(p_diff);
>+      ++vp9_info->num_ref_pics;
>+
>+      max_ref_temporal_layer_id =
>+          std::max(max_ref_temporal_layer_id, ref_buf.temporal_layer_id);
>+    } else {
>+      RTC_DCHECK(inter_layer_predicted);
>+      // RTP spec only allows to use previous spatial layer for inter-layer
>+      // prediction.
>+      RTC_DCHECK_EQ(ref_buf.spatial_layer_id + 1, layer_id.spatial_layer_id);
>+    }
>+  }
>+
>+  vp9_info->temporal_up_switch =
>+      (max_ref_temporal_layer_id <
>+       static_cast<size_t>(layer_id.temporal_layer_id));
>+}
>+
>+void VP9EncoderImpl::UpdateReferenceBuffers(const vpx_codec_cx_pkt& pkt,
>+                                            const size_t pic_num) {
>+  vpx_svc_layer_id_t layer_id = {0};
>+  vpx_codec_control(encoder_, VP9E_GET_SVC_LAYER_ID, &layer_id);
>+
>+  vpx_svc_ref_frame_config_t enc_layer_conf = {{0}};
>+  vpx_codec_control(encoder_, VP9E_GET_SVC_REF_FRAME_CONFIG, &enc_layer_conf);
>+
>+  const bool is_key_frame =
>+      (pkt.data.frame.flags & VPX_FRAME_IS_KEY) ? true : false;
>+
>+  RefFrameBuffer frame_buf(pic_num, layer_id.spatial_layer_id,
>+                           layer_id.temporal_layer_id);
>+
>+  if (is_key_frame && layer_id.spatial_layer_id == 0) {
>+    // Key frame updates all ref buffers.
>+    for (size_t i = 0; i < kNumVp9Buffers; ++i) {
>+      ref_buf_[i] = frame_buf;
>+    }
>+  } else {
>+    if (enc_layer_conf.update_last[layer_id.spatial_layer_id]) {
>+      ref_buf_[enc_layer_conf.lst_fb_idx[layer_id.spatial_layer_id]] =
>+          frame_buf;
>+    }
>+
>+    if (enc_layer_conf.update_alt_ref[layer_id.spatial_layer_id]) {
>+      ref_buf_[enc_layer_conf.alt_fb_idx[layer_id.spatial_layer_id]] =
>+          frame_buf;
>+    }
>+
>+    if (enc_layer_conf.update_golden[layer_id.spatial_layer_id]) {
>+      ref_buf_[enc_layer_conf.gld_fb_idx[layer_id.spatial_layer_id]] =
>+          frame_buf;
>+    }
>+  }
>+}
>+
> int VP9EncoderImpl::GetEncodedLayerFrame(const vpx_codec_cx_pkt* pkt) {
>   RTC_DCHECK_EQ(pkt->kind, VPX_CODEC_CX_FRAME_PKT);
> 
>+  if (pkt->data.frame.sz == 0) {
>+    // Ignore dropped frame.
>+    return WEBRTC_VIDEO_CODEC_OK;
>+  }
>+
>+  vpx_svc_layer_id_t layer_id = {0};
>+  vpx_codec_control(encoder_, VP9E_GET_SVC_LAYER_ID, &layer_id);
>+
>+  const bool first_frame_in_picture = encoded_image_._length == 0;
>+  // Ensure we don't buffer layers of previous picture (superframe).
>+  RTC_DCHECK(first_frame_in_picture || layer_id.spatial_layer_id > 0);
>+
>+  const bool end_of_picture = false;
>+  DeliverBufferedFrame(end_of_picture);
>+
>   if (pkt->data.frame.sz > encoded_image_._size) {
>     delete[] encoded_image_._buffer;
>     encoded_image_._size = pkt->data.frame.sz;
>@@ -681,154 +986,100 @@ int VP9EncoderImpl::GetEncodedLayerFrame(const vpx_codec_cx_pkt* pkt) {
>   memcpy(encoded_image_._buffer, pkt->data.frame.buf, pkt->data.frame.sz);
>   encoded_image_._length = pkt->data.frame.sz;
> 
>-  // No data partitioning in VP9, so 1 partition only.
>-  int part_idx = 0;
>-  RTPFragmentationHeader frag_info;
>-  frag_info.VerifyAndAllocateFragmentationHeader(1);
>-  frag_info.fragmentationOffset[part_idx] = 0;
>-  frag_info.fragmentationLength[part_idx] = pkt->data.frame.sz;
>-  frag_info.fragmentationPlType[part_idx] = 0;
>-  frag_info.fragmentationTimeDiff[part_idx] = 0;
>-
>-  vpx_svc_layer_id_t layer_id = {0};
>-  vpx_codec_control(encoder_, VP9E_GET_SVC_LAYER_ID, &layer_id);
>-  if (is_flexible_mode_ && codec_.mode == kScreensharing)
>-    spatial_layer_->LayerFrameEncoded(
>-        static_cast<unsigned int>(encoded_image_._length),
>-        layer_id.spatial_layer_id);
>+  const bool is_key_frame =
>+      (pkt->data.frame.flags & VPX_FRAME_IS_KEY) ? true : false;
>+  // Ensure encoder issued key frame on request.
>+  RTC_DCHECK(is_key_frame || !force_key_frame_);
> 
>-  // End of frame.
>   // Check if encoded frame is a key frame.
>   encoded_image_._frameType = kVideoFrameDelta;
>-  if (pkt->data.frame.flags & VPX_FRAME_IS_KEY) {
>+  if (is_key_frame) {
>     encoded_image_._frameType = kVideoFrameKey;
>+    force_key_frame_ = false;
>   }
>   RTC_DCHECK_LE(encoded_image_._length, encoded_image_._size);
> 
>-  CodecSpecificInfo codec_specific;
>-  PopulateCodecSpecific(&codec_specific, *pkt, input_image_->timestamp());
>+  memset(&codec_specific_, 0, sizeof(codec_specific_));
>+  PopulateCodecSpecific(&codec_specific_, *pkt, input_image_->timestamp(),
>+                        first_frame_in_picture);
> 
>+  if (is_flexible_mode_) {
>+    UpdateReferenceBuffers(*pkt, pics_since_key_);
>+  }
>+
>+  TRACE_COUNTER1("webrtc", "EncodedFrameSize", encoded_image_._length);
>+  encoded_image_._timeStamp = input_image_->timestamp();
>+  encoded_image_.capture_time_ms_ = input_image_->render_time_ms();
>+  encoded_image_.rotation_ = input_image_->rotation();
>+  encoded_image_.content_type_ = (codec_.mode == VideoCodecMode::kScreensharing)
>+                                     ? VideoContentType::SCREENSHARE
>+                                     : VideoContentType::UNSPECIFIED;
>+  encoded_image_._encodedHeight =
>+      pkt->data.frame.height[layer_id.spatial_layer_id];
>+  encoded_image_._encodedWidth =
>+      pkt->data.frame.width[layer_id.spatial_layer_id];
>+  encoded_image_.timing_.flags = VideoSendTiming::kInvalid;
>+  int qp = -1;
>+  vpx_codec_control(encoder_, VP8E_GET_LAST_QUANTIZER, &qp);
>+  encoded_image_.qp_ = qp;
>+
>+  return WEBRTC_VIDEO_CODEC_OK;
>+}
>+
>+void VP9EncoderImpl::DeliverBufferedFrame(bool end_of_picture) {
>   if (encoded_image_._length > 0) {
>-    TRACE_COUNTER1("webrtc", "EncodedFrameSize", encoded_image_._length);
>-    encoded_image_._timeStamp = input_image_->timestamp();
>-    encoded_image_.capture_time_ms_ = input_image_->render_time_ms();
>-    encoded_image_.rotation_ = input_image_->rotation();
>-    encoded_image_.content_type_ = (codec_.mode == kScreensharing)
>-                                       ? VideoContentType::SCREENSHARE
>-                                       : VideoContentType::UNSPECIFIED;
>-    encoded_image_._encodedHeight = raw_->d_h;
>-    encoded_image_._encodedWidth = raw_->d_w;
>-    encoded_image_.timing_.flags = TimingFrameFlags::kInvalid;
>-    int qp = -1;
>-    vpx_codec_control(encoder_, VP8E_GET_LAST_QUANTIZER, &qp);
>-    encoded_image_.qp_ = qp;
>-
>-    encoded_complete_callback_->OnEncodedImage(encoded_image_, &codec_specific,
>+    codec_specific_.codecSpecific.VP9.end_of_picture = end_of_picture;
>+
>+    // No data partitioning in VP9, so 1 partition only.
>+    int part_idx = 0;
>+    RTPFragmentationHeader frag_info;
>+    frag_info.VerifyAndAllocateFragmentationHeader(1);
>+    frag_info.fragmentationOffset[part_idx] = 0;
>+    frag_info.fragmentationLength[part_idx] = encoded_image_._length;
>+    frag_info.fragmentationPlType[part_idx] = 0;
>+    frag_info.fragmentationTimeDiff[part_idx] = 0;
>+
>+    encoded_complete_callback_->OnEncodedImage(encoded_image_, &codec_specific_,
>                                                &frag_info);
>+    encoded_image_._length = 0;
>+
>+    if (end_of_picture) {
>+      const uint32_t timestamp_ms =
>+          1000 * encoded_image_._timeStamp / kVideoPayloadTypeFrequency;
>+      output_framerate_.Update(1, timestamp_ms);
>+      last_encoded_frame_rtp_timestamp_ = encoded_image_._timeStamp;
>+    }
>   }
>-  return WEBRTC_VIDEO_CODEC_OK;
> }
> 
>-vpx_svc_ref_frame_config VP9EncoderImpl::GenerateRefsAndFlags(
>-    const SuperFrameRefSettings& settings) {
>-  static const vpx_enc_frame_flags_t kAllFlags =
>-      VP8_EFLAG_NO_REF_ARF | VP8_EFLAG_NO_REF_GF | VP8_EFLAG_NO_REF_LAST |
>-      VP8_EFLAG_NO_UPD_LAST | VP8_EFLAG_NO_UPD_ARF | VP8_EFLAG_NO_UPD_GF;
>-  vpx_svc_ref_frame_config sf_conf = {};
>-  if (settings.is_keyframe) {
>-    // Used later on to make sure we don't make any invalid references.
>-    memset(buffer_updated_at_frame_, -1, sizeof(buffer_updated_at_frame_));
>-    for (int layer = settings.start_layer; layer <= settings.stop_layer;
>-         ++layer) {
>-      num_ref_pics_[layer] = 0;
>-      buffer_updated_at_frame_[settings.layer[layer].upd_buf] = frames_encoded_;
>-      // When encoding a keyframe only the alt_fb_idx is used
>-      // to specify which layer ends up in which buffer.
>-      sf_conf.alt_fb_idx[layer] = settings.layer[layer].upd_buf;
>+bool VP9EncoderImpl::DropFrame(uint32_t rtp_timestamp) {
>+  if (target_framerate_fps_) {
>+    if (rtp_timestamp < last_encoded_frame_rtp_timestamp_) {
>+      // Timestamp has wrapped around. Reset framerate statistic.
>+      output_framerate_.Reset();
>+      return false;
>     }
>-  } else {
>-    for (int layer_idx = settings.start_layer; layer_idx <= settings.stop_layer;
>-         ++layer_idx) {
>-      vpx_enc_frame_flags_t layer_flags = kAllFlags;
>-      num_ref_pics_[layer_idx] = 0;
>-      int8_t refs[3] = {settings.layer[layer_idx].ref_buf1,
>-                        settings.layer[layer_idx].ref_buf2,
>-                        settings.layer[layer_idx].ref_buf3};
>-
>-      for (unsigned int ref_idx = 0; ref_idx < kMaxVp9RefPics; ++ref_idx) {
>-        if (refs[ref_idx] == -1)
>-          continue;
>-
>-        RTC_DCHECK_GE(refs[ref_idx], 0);
>-        RTC_DCHECK_LE(refs[ref_idx], 7);
>-        // Easier to remove flags from all flags rather than having to
>-        // build the flags from 0.
>-        switch (num_ref_pics_[layer_idx]) {
>-          case 0: {
>-            sf_conf.lst_fb_idx[layer_idx] = refs[ref_idx];
>-            layer_flags &= ~VP8_EFLAG_NO_REF_LAST;
>-            break;
>-          }
>-          case 1: {
>-            sf_conf.gld_fb_idx[layer_idx] = refs[ref_idx];
>-            layer_flags &= ~VP8_EFLAG_NO_REF_GF;
>-            break;
>-          }
>-          case 2: {
>-            sf_conf.alt_fb_idx[layer_idx] = refs[ref_idx];
>-            layer_flags &= ~VP8_EFLAG_NO_REF_ARF;
>-            break;
>-          }
>-        }
>-        // Make sure we don't reference a buffer that hasn't been
>-        // used at all or hasn't been used since a keyframe.
>-        RTC_DCHECK_NE(buffer_updated_at_frame_[refs[ref_idx]], -1);
> 
>-        p_diff_[layer_idx][num_ref_pics_[layer_idx]] =
>-            frames_encoded_ - buffer_updated_at_frame_[refs[ref_idx]];
>-        num_ref_pics_[layer_idx]++;
>-      }
>+    const uint32_t timestamp_ms =
>+        1000 * rtp_timestamp / kVideoPayloadTypeFrequency;
>+    const uint32_t framerate_fps =
>+        output_framerate_.Rate(timestamp_ms).value_or(0);
>+    if (framerate_fps > *target_framerate_fps_) {
>+      return true;
>+    }
> 
>-      bool upd_buf_same_as_a_ref = false;
>-      if (settings.layer[layer_idx].upd_buf != -1) {
>-        for (unsigned int ref_idx = 0; ref_idx < kMaxVp9RefPics; ++ref_idx) {
>-          if (settings.layer[layer_idx].upd_buf == refs[ref_idx]) {
>-            switch (ref_idx) {
>-              case 0: {
>-                layer_flags &= ~VP8_EFLAG_NO_UPD_LAST;
>-                break;
>-              }
>-              case 1: {
>-                layer_flags &= ~VP8_EFLAG_NO_UPD_GF;
>-                break;
>-              }
>-              case 2: {
>-                layer_flags &= ~VP8_EFLAG_NO_UPD_ARF;
>-                break;
>-              }
>-            }
>-            upd_buf_same_as_a_ref = true;
>-            break;
>-          }
>-        }
>-        if (!upd_buf_same_as_a_ref) {
>-          // If we have three references and a buffer is specified to be
>-          // updated, then that buffer must be the same as one of the
>-          // three references.
>-          RTC_CHECK_LT(num_ref_pics_[layer_idx], kMaxVp9RefPics);
>-
>-          sf_conf.alt_fb_idx[layer_idx] = settings.layer[layer_idx].upd_buf;
>-          layer_flags ^= VP8_EFLAG_NO_UPD_ARF;
>-        }
>+    // Primarily check if frame interval is too short using frame timestamps,
>+    // as if they are correct they won't be affected by queuing in webrtc.
>+    const uint32_t expected_frame_interval =
>+        kVideoPayloadTypeFrequency / *target_framerate_fps_;
> 
>-        int updated_buffer = settings.layer[layer_idx].upd_buf;
>-        buffer_updated_at_frame_[updated_buffer] = frames_encoded_;
>-        sf_conf.frame_flags[layer_idx] = layer_flags;
>-      }
>+    const uint32_t ts_diff = rtp_timestamp - last_encoded_frame_rtp_timestamp_;
>+    if (ts_diff < 85 * expected_frame_interval / 100) {
>+      return true;
>     }
>   }
>-  ++frames_encoded_;
>-  return sf_conf;
>+
>+  return false;
> }
> 
> int VP9EncoderImpl::SetChannelParameters(uint32_t packet_loss, int64_t rtt) {
>@@ -845,21 +1096,15 @@ const char* VP9EncoderImpl::ImplementationName() const {
>   return "libvpx";
> }
> 
>-bool VP9Decoder::IsSupported() {
>-  return true;
>-}
>-
> std::unique_ptr<VP9Decoder> VP9Decoder::Create() {
>-  return rtc::MakeUnique<VP9DecoderImpl>();
>+  return absl::make_unique<VP9DecoderImpl>();
> }
> 
> VP9DecoderImpl::VP9DecoderImpl()
>     : decode_complete_callback_(nullptr),
>       inited_(false),
>       decoder_(nullptr),
>-      key_frame_required_(true) {
>-  memset(&codec_, 0, sizeof(codec_));
>-}
>+      key_frame_required_(true) {}
> 
> VP9DecoderImpl::~VP9DecoderImpl() {
>   inited_ = true;  // in order to do the actual release
>@@ -875,9 +1120,6 @@ VP9DecoderImpl::~VP9DecoderImpl() {
> }
> 
> int VP9DecoderImpl::InitDecode(const VideoCodec* inst, int number_of_cores) {
>-  if (inst == nullptr) {
>-    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
>-  }
>   int ret_val = Release();
>   if (ret_val < 0) {
>     return ret_val;
>@@ -893,10 +1135,6 @@ int VP9DecoderImpl::InitDecode(const VideoCodec* inst, int number_of_cores) {
>   if (vpx_codec_dec_init(decoder_, vpx_codec_vp9_dx(), &cfg, flags)) {
>     return WEBRTC_VIDEO_CODEC_MEMORY;
>   }
>-  if (&codec_ != inst) {
>-    // Save VideoCodec instance for later; mainly for duplicating the decoder.
>-    codec_ = *inst;
>-  }
> 
>   if (!frame_buffer_pool_.InitializeVpxUsePool(decoder_)) {
>     return WEBRTC_VIDEO_CODEC_MEMORY;
>@@ -910,7 +1148,6 @@ int VP9DecoderImpl::InitDecode(const VideoCodec* inst, int number_of_cores) {
> 
> int VP9DecoderImpl::Decode(const EncodedImage& input_image,
>                            bool missing_frames,
>-                           const RTPFragmentationHeader* fragmentation,
>                            const CodecSpecificInfo* codec_specific_info,
>                            int64_t /*render_time_ms*/) {
>   if (!inited_) {
>@@ -973,10 +1210,13 @@ int VP9DecoderImpl::ReturnFrame(const vpx_image_t* img,
>   // vpx_codec_decode calls or vpx_codec_destroy).
>   Vp9FrameBufferPool::Vp9FrameBuffer* img_buffer =
>       static_cast<Vp9FrameBufferPool::Vp9FrameBuffer*>(img->fb_priv);
>+
>   // The buffer can be used directly by the VideoFrame (without copy) by
>-  // using a WrappedI420Buffer.
>-  rtc::scoped_refptr<WrappedI420Buffer> img_wrapped_buffer(
>-      new rtc::RefCountedObject<webrtc::WrappedI420Buffer>(
>+  // using a Wrapped*Buffer.
>+  rtc::scoped_refptr<VideoFrameBuffer> img_wrapped_buffer;
>+  switch (img->bit_depth) {
>+    case 8:
>+      img_wrapped_buffer = WrapI420Buffer(
>           img->d_w, img->d_h, img->planes[VPX_PLANE_Y],
>           img->stride[VPX_PLANE_Y], img->planes[VPX_PLANE_U],
>           img->stride[VPX_PLANE_U], img->planes[VPX_PLANE_V],
>@@ -984,13 +1224,33 @@ int VP9DecoderImpl::ReturnFrame(const vpx_image_t* img,
>           // WrappedI420Buffer's mechanism for allowing the release of its frame
>           // buffer is through a callback function. This is where we should
>           // release |img_buffer|.
>-          rtc::KeepRefUntilDone(img_buffer)));
>-
>-  VideoFrame decoded_image(img_wrapped_buffer, timestamp,
>-                           0 /* render_time_ms */, webrtc::kVideoRotation_0);
>-  decoded_image.set_ntp_time_ms(ntp_time_ms);
>-
>-  decode_complete_callback_->Decoded(decoded_image, rtc::nullopt, qp);
>+          rtc::KeepRefUntilDone(img_buffer));
>+      break;
>+    case 10:
>+      img_wrapped_buffer = WrapI010Buffer(
>+          img->d_w, img->d_h,
>+          reinterpret_cast<const uint16_t*>(img->planes[VPX_PLANE_Y]),
>+          img->stride[VPX_PLANE_Y] / 2,
>+          reinterpret_cast<const uint16_t*>(img->planes[VPX_PLANE_U]),
>+          img->stride[VPX_PLANE_U] / 2,
>+          reinterpret_cast<const uint16_t*>(img->planes[VPX_PLANE_V]),
>+          img->stride[VPX_PLANE_V] / 2, rtc::KeepRefUntilDone(img_buffer));
>+      break;
>+    default:
>+      RTC_NOTREACHED();
>+      return WEBRTC_VIDEO_CODEC_NO_OUTPUT;
>+  }
>+
>+  VideoFrame decoded_image = VideoFrame::Builder()
>+                                 .set_video_frame_buffer(img_wrapped_buffer)
>+                                 .set_timestamp_ms(0)
>+                                 .set_timestamp_rtp(timestamp)
>+                                 .set_ntp_time_ms(ntp_time_ms)
>+                                 .set_rotation(webrtc::kVideoRotation_0)
>+                                 .set_color_space(ExtractVP9ColorSpace(
>+                                     img->cs, img->range, img->bit_depth))
>+                                 .build();
>+  decode_complete_callback_->Decoded(decoded_image, absl::nullopt, qp);
>   return WEBRTC_VIDEO_CODEC_OK;
> }
> 
>@@ -1001,11 +1261,15 @@ int VP9DecoderImpl::RegisterDecodeCompleteCallback(
> }
> 
> int VP9DecoderImpl::Release() {
>+  int ret_val = WEBRTC_VIDEO_CODEC_OK;
>+
>   if (decoder_ != nullptr) {
>-    // When a codec is destroyed libvpx will release any buffers of
>-    // |frame_buffer_pool_| it is currently using.
>-    if (vpx_codec_destroy(decoder_)) {
>-      return WEBRTC_VIDEO_CODEC_MEMORY;
>+    if (inited_) {
>+      // When a codec is destroyed libvpx will release any buffers of
>+      // |frame_buffer_pool_| it is currently using.
>+      if (vpx_codec_destroy(decoder_)) {
>+        ret_val = WEBRTC_VIDEO_CODEC_MEMORY;
>+      }
>     }
>     delete decoder_;
>     decoder_ = nullptr;
>@@ -1015,7 +1279,7 @@ int VP9DecoderImpl::Release() {
>   // to the pool.
>   frame_buffer_pool_.ClearPool();
>   inited_ = false;
>-  return WEBRTC_VIDEO_CODEC_OK;
>+  return ret_val;
> }
> 
> const char* VP9DecoderImpl::ImplementationName() const {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/vp9_impl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/vp9_impl.h
>index d2f7822713e..7009311072f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/vp9_impl.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/vp9_impl.h
>@@ -12,11 +12,15 @@
> #ifndef MODULES_VIDEO_CODING_CODECS_VP9_VP9_IMPL_H_
> #define MODULES_VIDEO_CODING_CODECS_VP9_VP9_IMPL_H_
> 
>+#include <map>
> #include <memory>
> #include <vector>
> 
> #include "modules/video_coding/codecs/vp9/include/vp9.h"
>+
>+#include "media/base/vp9_profile.h"
> #include "modules/video_coding/codecs/vp9/vp9_frame_buffer_pool.h"
>+#include "rtc_base/rate_statistics.h"
> 
> #include "vpx/vp8cx.h"
> #include "vpx/vpx_decoder.h"
>@@ -24,11 +28,9 @@
> 
> namespace webrtc {
> 
>-class ScreenshareLayersVP9;
>-
> class VP9EncoderImpl : public VP9Encoder {
>  public:
>-  VP9EncoderImpl();
>+  explicit VP9EncoderImpl(const cricket::VideoCodec& codec);
> 
>   virtual ~VP9EncoderImpl();
> 
>@@ -46,25 +48,11 @@ class VP9EncoderImpl : public VP9Encoder {
> 
>   int SetChannelParameters(uint32_t packet_loss, int64_t rtt) override;
> 
>-  int SetRateAllocation(const BitrateAllocation& bitrate_allocation,
>+  int SetRateAllocation(const VideoBitrateAllocation& bitrate_allocation,
>                         uint32_t frame_rate) override;
> 
>   const char* ImplementationName() const override;
> 
>-  struct LayerFrameRefSettings {
>-    int8_t upd_buf = -1;   // -1 - no update,    0..7 - update buffer 0..7
>-    int8_t ref_buf1 = -1;  // -1 - no reference, 0..7 - reference buffer 0..7
>-    int8_t ref_buf2 = -1;  // -1 - no reference, 0..7 - reference buffer 0..7
>-    int8_t ref_buf3 = -1;  // -1 - no reference, 0..7 - reference buffer 0..7
>-  };
>-
>-  struct SuperFrameRefSettings {
>-    LayerFrameRefSettings layer[kMaxVp9NumberOfSpatialLayers];
>-    uint8_t start_layer = 0;  // The first spatial layer to be encoded.
>-    uint8_t stop_layer = 0;   // The last spatial layer to be encoded.
>-    bool is_keyframe = false;
>-  };
>-
>  private:
>   // Determine number of encoder threads to use.
>   int NumberOfThreads(int width, int height, int number_of_cores);
>@@ -74,19 +62,17 @@ class VP9EncoderImpl : public VP9Encoder {
> 
>   void PopulateCodecSpecific(CodecSpecificInfo* codec_specific,
>                              const vpx_codec_cx_pkt& pkt,
>-                             uint32_t timestamp);
>+                             uint32_t timestamp,
>+                             bool first_frame_in_picture);
>+  void FillReferenceIndices(const vpx_codec_cx_pkt& pkt,
>+                            const size_t pic_num,
>+                            const bool inter_layer_predicted,
>+                            CodecSpecificInfoVP9* vp9_info);
>+  void UpdateReferenceBuffers(const vpx_codec_cx_pkt& pkt,
>+                              const size_t pic_num);
> 
>   bool ExplicitlyConfiguredSpatialLayers() const;
>-  bool SetSvcRates();
>-
>-  // Used for flexible mode to set the flags and buffer references used
>-  // by the encoder. Also calculates the references used by the RTP
>-  // packetizer.
>-  //
>-  // Has to be called for every frame (keyframes included) to update the
>-  // state used to calculate references.
>-  vpx_svc_ref_frame_config GenerateRefsAndFlags(
>-      const SuperFrameRefSettings& settings);
>+  bool SetSvcRates(const VideoBitrateAllocation& bitrate_allocation);
> 
>   virtual int GetEncodedLayerFrame(const vpx_codec_cx_pkt* pkt);
> 
>@@ -94,6 +80,10 @@ class VP9EncoderImpl : public VP9Encoder {
>   static void EncoderOutputCodedPacketCallback(vpx_codec_cx_pkt* pkt,
>                                                void* user_data);
> 
>+  void DeliverBufferedFrame(bool end_of_picture);
>+
>+  bool DropFrame(uint32_t rtp_timestamp);
>+
>   // Determine maximum target for Intra frames
>   //
>   // Input:
>@@ -103,8 +93,10 @@ class VP9EncoderImpl : public VP9Encoder {
>   uint32_t MaxIntraTarget(uint32_t optimal_buffer_size);
> 
>   EncodedImage encoded_image_;
>+  CodecSpecificInfo codec_specific_;
>   EncodedImageCallback* encoded_complete_callback_;
>   VideoCodec codec_;
>+  const VP9Profile profile_;
>   bool inited_;
>   int64_t timestamp_;
>   int cpu_speed_;
>@@ -114,23 +106,36 @@ class VP9EncoderImpl : public VP9Encoder {
>   vpx_image_t* raw_;
>   vpx_svc_extra_cfg_t svc_params_;
>   const VideoFrame* input_image_;
>-  GofInfoVP9 gof_;       // Contains each frame's temporal information for
>-                         // non-flexible mode.
>-  size_t frames_since_kf_;
>+  GofInfoVP9 gof_;  // Contains each frame's temporal information for
>+                    // non-flexible mode.
>+  bool force_key_frame_;
>+  size_t pics_since_key_;
>   uint8_t num_temporal_layers_;
>-  uint8_t num_spatial_layers_;
>+  uint8_t num_spatial_layers_;         // Number of configured SLs
>+  uint8_t num_active_spatial_layers_;  // Number of actively encoded SLs
>+  bool is_svc_;
>+  InterLayerPredMode inter_layer_pred_;
>+
>+  // Framerate controller.
>+  absl::optional<float> target_framerate_fps_;
>+  RateStatistics output_framerate_;
>+  uint32_t last_encoded_frame_rtp_timestamp_;
> 
>   // Used for flexible mode.
>   bool is_flexible_mode_;
>-  int64_t buffer_updated_at_frame_[kNumVp9Buffers];
>-  int64_t frames_encoded_;
>-  uint8_t num_ref_pics_[kMaxVp9NumberOfSpatialLayers];
>-  uint8_t p_diff_[kMaxVp9NumberOfSpatialLayers][kMaxVp9RefPics];
>-  std::unique_ptr<ScreenshareLayersVP9> spatial_layer_;
>-
>-  // RTP state.
>-  uint16_t picture_id_;
>-  uint8_t tl0_pic_idx_;  // Only used in non-flexible mode.
>+  struct RefFrameBuffer {
>+    RefFrameBuffer(size_t pic_num,
>+                   size_t spatial_layer_id,
>+                   size_t temporal_layer_id)
>+        : pic_num(pic_num),
>+          spatial_layer_id(spatial_layer_id),
>+          temporal_layer_id(temporal_layer_id) {}
>+    RefFrameBuffer() {}
>+    size_t pic_num = 0;
>+    size_t spatial_layer_id = 0;
>+    size_t temporal_layer_id = 0;
>+  };
>+  std::map<size_t, RefFrameBuffer> ref_buf_;
> };
> 
> class VP9DecoderImpl : public VP9Decoder {
>@@ -143,7 +148,6 @@ class VP9DecoderImpl : public VP9Decoder {
> 
>   int Decode(const EncodedImage& input_image,
>              bool missing_frames,
>-             const RTPFragmentationHeader* fragmentation,
>              const CodecSpecificInfo* codec_specific_info,
>              int64_t /*render_time_ms*/) override;
> 
>@@ -164,7 +168,6 @@ class VP9DecoderImpl : public VP9Decoder {
>   DecodedImageCallback* decode_complete_callback_;
>   bool inited_;
>   vpx_codec_ctx_t* decoder_;
>-  VideoCodec codec_;
>   bool key_frame_required_;
> };
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/vp9_noop.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/vp9_noop.cc
>index baa04862c3c..b058c2d6728 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/vp9_noop.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/vp9_noop.cc
>@@ -14,12 +14,14 @@
> #endif  // !defined(RTC_DISABLE_VP9)
> 
> #include "modules/video_coding/codecs/vp9/include/vp9.h"
>+
>+#include "api/video_codecs/sdp_video_format.h"
> #include "rtc_base/checks.h"
> 
> namespace webrtc {
> 
>-bool VP9Encoder::IsSupported() {
>-  return false;
>+std::vector<SdpVideoFormat> SupportedVP9Codecs() {
>+  return std::vector<SdpVideoFormat>();
> }
> 
> std::unique_ptr<VP9Encoder> VP9Encoder::Create() {
>@@ -27,8 +29,10 @@ std::unique_ptr<VP9Encoder> VP9Encoder::Create() {
>   return nullptr;
> }
> 
>-bool VP9Decoder::IsSupported() {
>-  return false;
>+std::unique_ptr<VP9Encoder> VP9Encoder::Create(
>+    const cricket::VideoCodec& codec) {
>+  RTC_NOTREACHED();
>+  return nullptr;
> }
> 
> std::unique_ptr<VP9Decoder> VP9Decoder::Create() {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/vp9_screenshare_layers_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/vp9_screenshare_layers_unittest.cc
>deleted file mode 100644
>index 5ef7470c3b0..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp9/vp9_screenshare_layers_unittest.cc
>+++ /dev/null
>@@ -1,324 +0,0 @@
>-/*
>- *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include <limits>
>-#include <memory>
>-
>-#include "vpx/vp8cx.h"
>-#include "modules/video_coding/codecs/vp9/screenshare_layers.h"
>-#include "modules/video_coding/codecs/vp9/vp9_impl.h"
>-#include "rtc_base/logging.h"
>-#include "system_wrappers/include/clock.h"
>-#include "test/gtest.h"
>-
>-namespace webrtc {
>-
>-typedef VP9EncoderImpl::SuperFrameRefSettings Settings;
>-
>-const uint32_t kTickFrequency = 90000;
>-
>-class ScreenshareLayerTestVP9 : public ::testing::Test {
>- protected:
>-  ScreenshareLayerTestVP9() : clock_(0) {}
>-  virtual ~ScreenshareLayerTestVP9() {}
>-
>-  void InitScreenshareLayers(int layers) {
>-    layers_.reset(new ScreenshareLayersVP9(layers));
>-  }
>-
>-  void ConfigureBitrateForLayer(int kbps, uint8_t layer_id) {
>-    layers_->ConfigureBitrate(kbps, layer_id);
>-  }
>-
>-  void AdvanceTime(int64_t milliseconds) {
>-    clock_.AdvanceTimeMilliseconds(milliseconds);
>-  }
>-
>-  void AddKilobitsToLayer(int kilobits, uint8_t layer_id) {
>-    layers_->LayerFrameEncoded(kilobits * 1000 / 8, layer_id);
>-  }
>-
>-  void EqualRefsForLayer(const Settings& actual, uint8_t layer_id) {
>-    EXPECT_EQ(expected_.layer[layer_id].upd_buf,
>-              actual.layer[layer_id].upd_buf);
>-    EXPECT_EQ(expected_.layer[layer_id].ref_buf1,
>-              actual.layer[layer_id].ref_buf1);
>-    EXPECT_EQ(expected_.layer[layer_id].ref_buf2,
>-              actual.layer[layer_id].ref_buf2);
>-    EXPECT_EQ(expected_.layer[layer_id].ref_buf3,
>-              actual.layer[layer_id].ref_buf3);
>-  }
>-
>-  void EqualRefs(const Settings& actual) {
>-    for (unsigned int layer_id = 0; layer_id < kMaxVp9NumberOfSpatialLayers;
>-         ++layer_id) {
>-      EqualRefsForLayer(actual, layer_id);
>-    }
>-  }
>-
>-  void EqualStartStopKeyframe(const Settings& actual) {
>-    EXPECT_EQ(expected_.start_layer, actual.start_layer);
>-    EXPECT_EQ(expected_.stop_layer, actual.stop_layer);
>-    EXPECT_EQ(expected_.is_keyframe, actual.is_keyframe);
>-  }
>-
>-  // Check that the settings returned by GetSuperFrameSettings() is
>-  // equal to the expected_ settings.
>-  void EqualToExpected() {
>-    uint32_t frame_timestamp_ =
>-        clock_.TimeInMilliseconds() * (kTickFrequency / 1000);
>-    Settings actual =
>-        layers_->GetSuperFrameSettings(frame_timestamp_, expected_.is_keyframe);
>-    EqualRefs(actual);
>-    EqualStartStopKeyframe(actual);
>-  }
>-
>-  Settings expected_;
>-  SimulatedClock clock_;
>-  std::unique_ptr<ScreenshareLayersVP9> layers_;
>-};
>-
>-TEST_F(ScreenshareLayerTestVP9, NoRefsOnKeyFrame) {
>-  const int kNumLayers = kMaxVp9NumberOfSpatialLayers;
>-  InitScreenshareLayers(kNumLayers);
>-  expected_.start_layer = 0;
>-  expected_.stop_layer = kNumLayers - 1;
>-
>-  for (int l = 0; l < kNumLayers; ++l) {
>-    expected_.layer[l].upd_buf = l;
>-  }
>-  expected_.is_keyframe = true;
>-  EqualToExpected();
>-
>-  for (int l = 0; l < kNumLayers; ++l) {
>-    expected_.layer[l].ref_buf1 = l;
>-  }
>-  expected_.is_keyframe = false;
>-  EqualToExpected();
>-}
>-
>-// Test if it is possible to send at a high bitrate (over the threshold)
>-// after a longer period of low bitrate. This should not be possible.
>-TEST_F(ScreenshareLayerTestVP9, DontAccumelateAvailableBitsOverTime) {
>-  InitScreenshareLayers(2);
>-  ConfigureBitrateForLayer(100, 0);
>-
>-  expected_.layer[0].upd_buf = 0;
>-  expected_.layer[0].ref_buf1 = 0;
>-  expected_.layer[1].upd_buf = 1;
>-  expected_.layer[1].ref_buf1 = 1;
>-  expected_.start_layer = 0;
>-  expected_.stop_layer = 1;
>-
>-  // Send 10 frames at a low bitrate (50 kbps)
>-  for (int i = 0; i < 10; ++i) {
>-    AdvanceTime(200);
>-    EqualToExpected();
>-    AddKilobitsToLayer(10, 0);
>-  }
>-
>-  AdvanceTime(200);
>-  EqualToExpected();
>-  AddKilobitsToLayer(301, 0);
>-
>-  // Send 10 frames at a high bitrate (200 kbps)
>-  expected_.start_layer = 1;
>-  for (int i = 0; i < 10; ++i) {
>-    AdvanceTime(200);
>-    EqualToExpected();
>-    AddKilobitsToLayer(40, 1);
>-  }
>-}
>-
>-// Test if used bits are accumelated over layers, as they should;
>-TEST_F(ScreenshareLayerTestVP9, AccumelateUsedBitsOverLayers) {
>-  const int kNumLayers = kMaxVp9NumberOfSpatialLayers;
>-  InitScreenshareLayers(kNumLayers);
>-  for (int l = 0; l < kNumLayers - 1; ++l)
>-    ConfigureBitrateForLayer(100, l);
>-  for (int l = 0; l < kNumLayers; ++l) {
>-    expected_.layer[l].upd_buf = l;
>-    expected_.layer[l].ref_buf1 = l;
>-  }
>-
>-  expected_.start_layer = 0;
>-  expected_.stop_layer = kNumLayers - 1;
>-  EqualToExpected();
>-
>-  for (int layer = 0; layer < kNumLayers - 1; ++layer) {
>-    expected_.start_layer = layer;
>-    EqualToExpected();
>-    AddKilobitsToLayer(101, layer);
>-  }
>-}
>-
>-// General testing of the bitrate controller.
>-TEST_F(ScreenshareLayerTestVP9, 2LayerBitrate) {
>-  InitScreenshareLayers(2);
>-  ConfigureBitrateForLayer(100, 0);
>-
>-  expected_.layer[0].upd_buf = 0;
>-  expected_.layer[1].upd_buf = 1;
>-  expected_.layer[0].ref_buf1 = -1;
>-  expected_.layer[1].ref_buf1 = -1;
>-  expected_.start_layer = 0;
>-  expected_.stop_layer = 1;
>-
>-  expected_.is_keyframe = true;
>-  EqualToExpected();
>-  AddKilobitsToLayer(100, 0);
>-
>-  expected_.layer[0].ref_buf1 = 0;
>-  expected_.layer[1].ref_buf1 = 1;
>-  expected_.is_keyframe = false;
>-  AdvanceTime(199);
>-  EqualToExpected();
>-  AddKilobitsToLayer(100, 0);
>-
>-  expected_.start_layer = 1;
>-  for (int frame = 0; frame < 3; ++frame) {
>-    AdvanceTime(200);
>-    EqualToExpected();
>-    AddKilobitsToLayer(100, 1);
>-  }
>-
>-  // Just before enough bits become available for L0 @0.999 seconds.
>-  AdvanceTime(199);
>-  EqualToExpected();
>-  AddKilobitsToLayer(100, 1);
>-
>-  // Just after enough bits become available for L0 @1.0001 seconds.
>-  expected_.start_layer = 0;
>-  AdvanceTime(2);
>-  EqualToExpected();
>-  AddKilobitsToLayer(100, 0);
>-
>-  // Keyframes always encode all layers, even if it is over budget.
>-  expected_.layer[0].ref_buf1 = -1;
>-  expected_.layer[1].ref_buf1 = -1;
>-  expected_.is_keyframe = true;
>-  AdvanceTime(499);
>-  EqualToExpected();
>-  expected_.layer[0].ref_buf1 = 0;
>-  expected_.layer[1].ref_buf1 = 1;
>-  expected_.start_layer = 1;
>-  expected_.is_keyframe = false;
>-  EqualToExpected();
>-  AddKilobitsToLayer(100, 0);
>-
>-  // 400 kb in L0 --> @3 second mark to fall below the threshold..
>-  // just before @2.999 seconds.
>-  expected_.is_keyframe = false;
>-  AdvanceTime(1499);
>-  EqualToExpected();
>-  AddKilobitsToLayer(100, 1);
>-
>-  // just after @3.001 seconds.
>-  expected_.start_layer = 0;
>-  AdvanceTime(2);
>-  EqualToExpected();
>-  AddKilobitsToLayer(100, 0);
>-}
>-
>-// General testing of the bitrate controller.
>-TEST_F(ScreenshareLayerTestVP9, 3LayerBitrate) {
>-  InitScreenshareLayers(3);
>-  ConfigureBitrateForLayer(100, 0);
>-  ConfigureBitrateForLayer(100, 1);
>-
>-  for (int l = 0; l < 3; ++l) {
>-    expected_.layer[l].upd_buf = l;
>-    expected_.layer[l].ref_buf1 = l;
>-  }
>-  expected_.start_layer = 0;
>-  expected_.stop_layer = 2;
>-
>-  EqualToExpected();
>-  AddKilobitsToLayer(105, 0);
>-  AddKilobitsToLayer(30, 1);
>-
>-  AdvanceTime(199);
>-  EqualToExpected();
>-  AddKilobitsToLayer(105, 0);
>-  AddKilobitsToLayer(30, 1);
>-
>-  expected_.start_layer = 1;
>-  AdvanceTime(200);
>-  EqualToExpected();
>-  AddKilobitsToLayer(130, 1);
>-
>-  expected_.start_layer = 2;
>-  AdvanceTime(200);
>-  EqualToExpected();
>-
>-  // 400 kb in L1 --> @1.0 second mark to fall below threshold.
>-  // 210 kb in L0 --> @1.1 second mark to fall below threshold.
>-  // Just before L1 @0.999 seconds.
>-  AdvanceTime(399);
>-  EqualToExpected();
>-
>-  // Just after L1 @1.001 seconds.
>-  expected_.start_layer = 1;
>-  AdvanceTime(2);
>-  EqualToExpected();
>-
>-  // Just before L0 @1.099 seconds.
>-  AdvanceTime(99);
>-  EqualToExpected();
>-
>-  // Just after L0 @1.101 seconds.
>-  expected_.start_layer = 0;
>-  AdvanceTime(2);
>-  EqualToExpected();
>-
>-  // @1.1 seconds
>-  AdvanceTime(99);
>-  EqualToExpected();
>-  AddKilobitsToLayer(200, 1);
>-
>-  expected_.is_keyframe = true;
>-  for (int l = 0; l < 3; ++l)
>-    expected_.layer[l].ref_buf1 = -1;
>-  AdvanceTime(200);
>-  EqualToExpected();
>-
>-  expected_.is_keyframe = false;
>-  expected_.start_layer = 2;
>-  for (int l = 0; l < 3; ++l)
>-    expected_.layer[l].ref_buf1 = l;
>-  AdvanceTime(200);
>-  EqualToExpected();
>-}
>-
>-// Test that the bitrate calculations are
>-// correct when the timestamp wrap.
>-TEST_F(ScreenshareLayerTestVP9, TimestampWrap) {
>-  InitScreenshareLayers(2);
>-  ConfigureBitrateForLayer(100, 0);
>-
>-  expected_.layer[0].upd_buf = 0;
>-  expected_.layer[0].ref_buf1 = 0;
>-  expected_.layer[1].upd_buf = 1;
>-  expected_.layer[1].ref_buf1 = 1;
>-  expected_.start_layer = 0;
>-  expected_.stop_layer = 1;
>-
>-  // Advance time to just before the timestamp wraps.
>-  AdvanceTime(std::numeric_limits<uint32_t>::max() / (kTickFrequency / 1000));
>-  EqualToExpected();
>-  AddKilobitsToLayer(200, 0);
>-
>-  // Wrap
>-  expected_.start_layer = 1;
>-  AdvanceTime(1);
>-  EqualToExpected();
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/decoder_database.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/decoder_database.cc
>new file mode 100644
>index 00000000000..c113239fbf6
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/decoder_database.cc
>@@ -0,0 +1,197 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/video_coding/decoder_database.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/logging.h"
>+
>+namespace webrtc {
>+
>+VCMDecoderMapItem::VCMDecoderMapItem(VideoCodec* settings,
>+                                     int number_of_cores,
>+                                     bool require_key_frame)
>+    : settings(settings),
>+      number_of_cores(number_of_cores),
>+      require_key_frame(require_key_frame) {
>+  RTC_DCHECK_GE(number_of_cores, 0);
>+}
>+
>+VCMExtDecoderMapItem::VCMExtDecoderMapItem(
>+    VideoDecoder* external_decoder_instance,
>+    uint8_t payload_type)
>+    : payload_type(payload_type),
>+      external_decoder_instance(external_decoder_instance) {}
>+
>+VCMDecoderMapItem::~VCMDecoderMapItem() {}
>+
>+VCMDecoderDataBase::VCMDecoderDataBase()
>+    : receive_codec_(), dec_map_(), dec_external_map_() {}
>+
>+VCMDecoderDataBase::~VCMDecoderDataBase() {
>+  ptr_decoder_.reset();
>+  for (auto& kv : dec_map_)
>+    delete kv.second;
>+  for (auto& kv : dec_external_map_)
>+    delete kv.second;
>+}
>+
>+bool VCMDecoderDataBase::DeregisterExternalDecoder(uint8_t payload_type) {
>+  ExternalDecoderMap::iterator it = dec_external_map_.find(payload_type);
>+  if (it == dec_external_map_.end()) {
>+    // Not found.
>+    return false;
>+  }
>+  // We can't use payload_type to check if the decoder is currently in use,
>+  // because payload type may be out of date (e.g. before we decode the first
>+  // frame after RegisterReceiveCodec).
>+  if (ptr_decoder_ &&
>+      ptr_decoder_->IsSameDecoder((*it).second->external_decoder_instance)) {
>+    // Release it if it was registered and in use.
>+    ptr_decoder_.reset();
>+  }
>+  DeregisterReceiveCodec(payload_type);
>+  delete it->second;
>+  dec_external_map_.erase(it);
>+  return true;
>+}
>+
>+// Add the external decoder object to the list of external decoders.
>+// Won't be registered as a receive codec until RegisterReceiveCodec is called.
>+void VCMDecoderDataBase::RegisterExternalDecoder(VideoDecoder* external_decoder,
>+                                                 uint8_t payload_type) {
>+  // If payload value already exists, erase old and insert new.
>+  VCMExtDecoderMapItem* ext_decoder =
>+      new VCMExtDecoderMapItem(external_decoder, payload_type);
>+  DeregisterExternalDecoder(payload_type);
>+  dec_external_map_[payload_type] = ext_decoder;
>+}
>+
>+bool VCMDecoderDataBase::RegisterReceiveCodec(const VideoCodec* receive_codec,
>+                                              int number_of_cores,
>+                                              bool require_key_frame) {
>+  if (number_of_cores < 0) {
>+    return false;
>+  }
>+  // If payload value already exists, erase old and insert new.
>+  DeregisterReceiveCodec(receive_codec->plType);
>+  VideoCodec* new_receive_codec = new VideoCodec(*receive_codec);
>+  dec_map_[receive_codec->plType] = new VCMDecoderMapItem(
>+      new_receive_codec, number_of_cores, require_key_frame);
>+  return true;
>+}
>+
>+bool VCMDecoderDataBase::DeregisterReceiveCodec(uint8_t payload_type) {
>+  DecoderMap::iterator it = dec_map_.find(payload_type);
>+  if (it == dec_map_.end()) {
>+    return false;
>+  }
>+  delete it->second;
>+  dec_map_.erase(it);
>+  if (receive_codec_.plType == payload_type) {
>+    // This codec is currently in use.
>+    memset(&receive_codec_, 0, sizeof(VideoCodec));
>+  }
>+  return true;
>+}
>+
>+VCMGenericDecoder* VCMDecoderDataBase::GetDecoder(
>+    const VCMEncodedFrame& frame,
>+    VCMDecodedFrameCallback* decoded_frame_callback) {
>+  RTC_DCHECK(decoded_frame_callback->UserReceiveCallback());
>+  uint8_t payload_type = frame.PayloadType();
>+  if (payload_type == receive_codec_.plType || payload_type == 0) {
>+    return ptr_decoder_.get();
>+  }
>+  // If decoder exists - delete.
>+  if (ptr_decoder_) {
>+    ptr_decoder_.reset();
>+    memset(&receive_codec_, 0, sizeof(VideoCodec));
>+  }
>+  ptr_decoder_ = CreateAndInitDecoder(frame, &receive_codec_);
>+  if (!ptr_decoder_) {
>+    return nullptr;
>+  }
>+  VCMReceiveCallback* callback = decoded_frame_callback->UserReceiveCallback();
>+  callback->OnIncomingPayloadType(receive_codec_.plType);
>+  if (ptr_decoder_->RegisterDecodeCompleteCallback(decoded_frame_callback) <
>+      0) {
>+    ptr_decoder_.reset();
>+    memset(&receive_codec_, 0, sizeof(VideoCodec));
>+    return nullptr;
>+  }
>+  return ptr_decoder_.get();
>+}
>+
>+bool VCMDecoderDataBase::PrefersLateDecoding() const {
>+  return ptr_decoder_ ? ptr_decoder_->PrefersLateDecoding() : true;
>+}
>+
>+std::unique_ptr<VCMGenericDecoder> VCMDecoderDataBase::CreateAndInitDecoder(
>+    const VCMEncodedFrame& frame,
>+    VideoCodec* new_codec) const {
>+  uint8_t payload_type = frame.PayloadType();
>+  RTC_LOG(LS_INFO) << "Initializing decoder with payload type '"
>+                   << static_cast<int>(payload_type) << "'.";
>+  RTC_DCHECK(new_codec);
>+  const VCMDecoderMapItem* decoder_item = FindDecoderItem(payload_type);
>+  if (!decoder_item) {
>+    RTC_LOG(LS_ERROR) << "Can't find a decoder associated with payload type: "
>+                      << static_cast<int>(payload_type);
>+    return nullptr;
>+  }
>+  std::unique_ptr<VCMGenericDecoder> ptr_decoder;
>+  const VCMExtDecoderMapItem* external_dec_item =
>+      FindExternalDecoderItem(payload_type);
>+  if (external_dec_item) {
>+    // External codec.
>+    ptr_decoder.reset(new VCMGenericDecoder(
>+        external_dec_item->external_decoder_instance, true));
>+  } else {
>+    RTC_LOG(LS_ERROR) << "No decoder of this type exists.";
>+  }
>+  if (!ptr_decoder)
>+    return nullptr;
>+
>+  // Copy over input resolutions to prevent codec reinitialization due to
>+  // the first frame being of a different resolution than the database values.
>+  // This is best effort, since there's no guarantee that width/height have been
>+  // parsed yet (and may be zero).
>+  if (frame.EncodedImage()._encodedWidth > 0 &&
>+      frame.EncodedImage()._encodedHeight > 0) {
>+    decoder_item->settings->width = frame.EncodedImage()._encodedWidth;
>+    decoder_item->settings->height = frame.EncodedImage()._encodedHeight;
>+  }
>+  if (ptr_decoder->InitDecode(decoder_item->settings.get(),
>+                              decoder_item->number_of_cores) < 0) {
>+    return nullptr;
>+  }
>+  memcpy(new_codec, decoder_item->settings.get(), sizeof(VideoCodec));
>+  return ptr_decoder;
>+}
>+
>+const VCMDecoderMapItem* VCMDecoderDataBase::FindDecoderItem(
>+    uint8_t payload_type) const {
>+  DecoderMap::const_iterator it = dec_map_.find(payload_type);
>+  if (it != dec_map_.end()) {
>+    return (*it).second;
>+  }
>+  return nullptr;
>+}
>+
>+const VCMExtDecoderMapItem* VCMDecoderDataBase::FindExternalDecoderItem(
>+    uint8_t payload_type) const {
>+  ExternalDecoderMap::const_iterator it = dec_external_map_.find(payload_type);
>+  if (it != dec_external_map_.end()) {
>+    return (*it).second;
>+  }
>+  return nullptr;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/decoder_database.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/decoder_database.h
>new file mode 100644
>index 00000000000..8c96b41efdd
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/decoder_database.h
>@@ -0,0 +1,90 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_VIDEO_CODING_DECODER_DATABASE_H_
>+#define MODULES_VIDEO_CODING_DECODER_DATABASE_H_
>+
>+#include <map>
>+#include <memory>
>+
>+#include "modules/video_coding/generic_decoder.h"
>+
>+namespace webrtc {
>+
>+struct VCMDecoderMapItem {
>+ public:
>+  VCMDecoderMapItem(VideoCodec* settings,
>+                    int number_of_cores,
>+                    bool require_key_frame);
>+  ~VCMDecoderMapItem();
>+
>+  std::unique_ptr<VideoCodec> settings;
>+  int number_of_cores;
>+  bool require_key_frame;
>+};
>+
>+struct VCMExtDecoderMapItem {
>+ public:
>+  VCMExtDecoderMapItem(VideoDecoder* external_decoder_instance,
>+                       uint8_t payload_type);
>+
>+  uint8_t payload_type;
>+  VideoDecoder* external_decoder_instance;
>+};
>+
>+class VCMDecoderDataBase {
>+ public:
>+  VCMDecoderDataBase();
>+  ~VCMDecoderDataBase();
>+
>+  bool DeregisterExternalDecoder(uint8_t payload_type);
>+  void RegisterExternalDecoder(VideoDecoder* external_decoder,
>+                               uint8_t payload_type);
>+
>+  bool RegisterReceiveCodec(const VideoCodec* receive_codec,
>+                            int number_of_cores,
>+                            bool require_key_frame);
>+  bool DeregisterReceiveCodec(uint8_t payload_type);
>+
>+  // Returns a decoder specified by frame.PayloadType. The decoded frame
>+  // callback of the decoder is set to |decoded_frame_callback|. If no such
>+  // decoder already exists an instance will be created and initialized.
>+  // nullptr is returned if no decoder with the specified payload type was found
>+  // and the function failed to create one.
>+  VCMGenericDecoder* GetDecoder(
>+      const VCMEncodedFrame& frame,
>+      VCMDecodedFrameCallback* decoded_frame_callback);
>+
>+  // Returns true if the currently active decoder prefer to decode frames late.
>+  // That means that frames must be decoded near the render times stamp.
>+  bool PrefersLateDecoding() const;
>+
>+ private:
>+  typedef std::map<uint8_t, VCMDecoderMapItem*> DecoderMap;
>+  typedef std::map<uint8_t, VCMExtDecoderMapItem*> ExternalDecoderMap;
>+
>+  std::unique_ptr<VCMGenericDecoder> CreateAndInitDecoder(
>+      const VCMEncodedFrame& frame,
>+      VideoCodec* new_codec) const;
>+
>+  const VCMDecoderMapItem* FindDecoderItem(uint8_t payload_type) const;
>+
>+  const VCMExtDecoderMapItem* FindExternalDecoderItem(
>+      uint8_t payload_type) const;
>+
>+  VideoCodec receive_codec_;
>+  std::unique_ptr<VCMGenericDecoder> ptr_decoder_;
>+  DecoderMap dec_map_;
>+  ExternalDecoderMap dec_external_map_;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_VIDEO_CODING_DECODER_DATABASE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/decoding_state.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/decoding_state.h
>index 02a932b80b3..b87fb2d0345 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/decoding_state.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/decoding_state.h
>@@ -15,8 +15,6 @@
> #include <set>
> #include <vector>
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>-
> namespace webrtc {
> 
> // Forward declarations
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/decoding_state_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/decoding_state_unittest.cc
>index 96ba32fb360..0049ca7e1db 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/decoding_state_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/decoding_state_unittest.cc
>@@ -36,8 +36,8 @@ TEST(TestDecodingState, FrameContinuity) {
>   packet.timestamp = 1;
>   packet.seqNum = 0xffff;
>   packet.frameType = kVideoFrameDelta;
>-  packet.video_header.codec = kRtpVideoVp8;
>-  packet.video_header.codecHeader.VP8.pictureId = 0x007F;
>+  packet.video_header.codec = kVideoCodecVP8;
>+  packet.video_header.vp8().pictureId = 0x007F;
>   FrameData frame_data;
>   frame_data.rtt_ms = 0;
>   frame_data.rolling_average_packets_per_frame = -1;
>@@ -53,17 +53,17 @@ TEST(TestDecodingState, FrameContinuity) {
>   packet.frameType = kVideoFrameDelta;
>   // Use pictureId
>   packet.is_first_packet_in_frame = false;
>-  packet.video_header.codecHeader.VP8.pictureId = 0x0002;
>+  packet.video_header.vp8().pictureId = 0x0002;
>   EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
>   EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
>   frame.Reset();
>-  packet.video_header.codecHeader.VP8.pictureId = 0;
>+  packet.video_header.vp8().pictureId = 0;
>   packet.seqNum = 10;
>   EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
>   EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
> 
>   // Use sequence numbers.
>-  packet.video_header.codecHeader.VP8.pictureId = kNoPictureId;
>+  packet.video_header.vp8().pictureId = kNoPictureId;
>   frame.Reset();
>   packet.seqNum = dec_state.sequence_num() - 1u;
>   EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
>@@ -82,9 +82,9 @@ TEST(TestDecodingState, FrameContinuity) {
>   // Insert packet with temporal info.
>   dec_state.Reset();
>   frame.Reset();
>-  packet.video_header.codecHeader.VP8.tl0PicIdx = 0;
>-  packet.video_header.codecHeader.VP8.temporalIdx = 0;
>-  packet.video_header.codecHeader.VP8.pictureId = 0;
>+  packet.video_header.vp8().tl0PicIdx = 0;
>+  packet.video_header.vp8().temporalIdx = 0;
>+  packet.video_header.vp8().pictureId = 0;
>   packet.seqNum = 1;
>   packet.timestamp = 1;
>   EXPECT_TRUE(dec_state.full_sync());
>@@ -93,9 +93,9 @@ TEST(TestDecodingState, FrameContinuity) {
>   EXPECT_TRUE(dec_state.full_sync());
>   frame.Reset();
>   // 1 layer up - still good.
>-  packet.video_header.codecHeader.VP8.tl0PicIdx = 0;
>-  packet.video_header.codecHeader.VP8.temporalIdx = 1;
>-  packet.video_header.codecHeader.VP8.pictureId = 1;
>+  packet.video_header.vp8().tl0PicIdx = 0;
>+  packet.video_header.vp8().temporalIdx = 1;
>+  packet.video_header.vp8().pictureId = 1;
>   packet.seqNum = 2;
>   packet.timestamp = 2;
>   EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
>@@ -104,18 +104,18 @@ TEST(TestDecodingState, FrameContinuity) {
>   EXPECT_TRUE(dec_state.full_sync());
>   frame.Reset();
>   // Lost non-base layer packet => should update sync parameter.
>-  packet.video_header.codecHeader.VP8.tl0PicIdx = 0;
>-  packet.video_header.codecHeader.VP8.temporalIdx = 3;
>-  packet.video_header.codecHeader.VP8.pictureId = 3;
>+  packet.video_header.vp8().tl0PicIdx = 0;
>+  packet.video_header.vp8().temporalIdx = 3;
>+  packet.video_header.vp8().pictureId = 3;
>   packet.seqNum = 4;
>   packet.timestamp = 4;
>   EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
>   EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
>   // Now insert the next non-base layer (belonging to a next tl0PicId).
>   frame.Reset();
>-  packet.video_header.codecHeader.VP8.tl0PicIdx = 1;
>-  packet.video_header.codecHeader.VP8.temporalIdx = 2;
>-  packet.video_header.codecHeader.VP8.pictureId = 4;
>+  packet.video_header.vp8().tl0PicIdx = 1;
>+  packet.video_header.vp8().temporalIdx = 2;
>+  packet.video_header.vp8().pictureId = 4;
>   packet.seqNum = 5;
>   packet.timestamp = 5;
>   EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
>@@ -125,9 +125,9 @@ TEST(TestDecodingState, FrameContinuity) {
>   EXPECT_TRUE(dec_state.full_sync());
>   // Next base layer (dropped interim non-base layers) - should update sync.
>   frame.Reset();
>-  packet.video_header.codecHeader.VP8.tl0PicIdx = 1;
>-  packet.video_header.codecHeader.VP8.temporalIdx = 0;
>-  packet.video_header.codecHeader.VP8.pictureId = 5;
>+  packet.video_header.vp8().tl0PicIdx = 1;
>+  packet.video_header.vp8().temporalIdx = 0;
>+  packet.video_header.vp8().pictureId = 5;
>   packet.seqNum = 6;
>   packet.timestamp = 6;
>   EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
>@@ -137,18 +137,18 @@ TEST(TestDecodingState, FrameContinuity) {
> 
>   // Check wrap for temporal layers.
>   frame.Reset();
>-  packet.video_header.codecHeader.VP8.tl0PicIdx = 0x00FF;
>-  packet.video_header.codecHeader.VP8.temporalIdx = 0;
>-  packet.video_header.codecHeader.VP8.pictureId = 6;
>+  packet.video_header.vp8().tl0PicIdx = 0x00FF;
>+  packet.video_header.vp8().temporalIdx = 0;
>+  packet.video_header.vp8().pictureId = 6;
>   packet.seqNum = 7;
>   packet.timestamp = 7;
>   EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
>   dec_state.SetState(&frame);
>   EXPECT_FALSE(dec_state.full_sync());
>   frame.Reset();
>-  packet.video_header.codecHeader.VP8.tl0PicIdx = 0x0000;
>-  packet.video_header.codecHeader.VP8.temporalIdx = 0;
>-  packet.video_header.codecHeader.VP8.pictureId = 7;
>+  packet.video_header.vp8().tl0PicIdx = 0x0000;
>+  packet.video_header.vp8().temporalIdx = 0;
>+  packet.video_header.vp8().pictureId = 7;
>   packet.seqNum = 8;
>   packet.timestamp = 8;
>   EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
>@@ -211,12 +211,12 @@ TEST(TestDecodingState, MultiLayerBehavior) {
>   VCMFrameBuffer frame;
>   VCMPacket packet;
>   packet.frameType = kVideoFrameDelta;
>-  packet.video_header.codec = kRtpVideoVp8;
>+  packet.video_header.codec = kVideoCodecVP8;
>   packet.timestamp = 0;
>   packet.seqNum = 0;
>-  packet.video_header.codecHeader.VP8.tl0PicIdx = 0;
>-  packet.video_header.codecHeader.VP8.temporalIdx = 0;
>-  packet.video_header.codecHeader.VP8.pictureId = 0;
>+  packet.video_header.vp8().tl0PicIdx = 0;
>+  packet.video_header.vp8().temporalIdx = 0;
>+  packet.video_header.vp8().pictureId = 0;
>   FrameData frame_data;
>   frame_data.rtt_ms = 0;
>   frame_data.rolling_average_packets_per_frame = -1;
>@@ -226,9 +226,9 @@ TEST(TestDecodingState, MultiLayerBehavior) {
>   frame.Reset();
>   packet.timestamp = 1;
>   packet.seqNum = 1;
>-  packet.video_header.codecHeader.VP8.tl0PicIdx = 0;
>-  packet.video_header.codecHeader.VP8.temporalIdx = 1;
>-  packet.video_header.codecHeader.VP8.pictureId = 1;
>+  packet.video_header.vp8().tl0PicIdx = 0;
>+  packet.video_header.vp8().temporalIdx = 1;
>+  packet.video_header.vp8().pictureId = 1;
>   EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
>   EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
>   dec_state.SetState(&frame);
>@@ -238,9 +238,9 @@ TEST(TestDecodingState, MultiLayerBehavior) {
>   frame.Reset();
>   packet.timestamp = 3;
>   packet.seqNum = 3;
>-  packet.video_header.codecHeader.VP8.tl0PicIdx = 0;
>-  packet.video_header.codecHeader.VP8.temporalIdx = 3;
>-  packet.video_header.codecHeader.VP8.pictureId = 3;
>+  packet.video_header.vp8().tl0PicIdx = 0;
>+  packet.video_header.vp8().temporalIdx = 3;
>+  packet.video_header.vp8().pictureId = 3;
>   EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
>   EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
>   dec_state.SetState(&frame);
>@@ -249,9 +249,9 @@ TEST(TestDecodingState, MultiLayerBehavior) {
>   frame.Reset();
>   packet.timestamp = 4;
>   packet.seqNum = 4;
>-  packet.video_header.codecHeader.VP8.tl0PicIdx = 1;
>-  packet.video_header.codecHeader.VP8.temporalIdx = 0;
>-  packet.video_header.codecHeader.VP8.pictureId = 4;
>+  packet.video_header.vp8().tl0PicIdx = 1;
>+  packet.video_header.vp8().temporalIdx = 0;
>+  packet.video_header.vp8().pictureId = 4;
>   EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
>   EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
>   dec_state.SetState(&frame);
>@@ -263,9 +263,9 @@ TEST(TestDecodingState, MultiLayerBehavior) {
>   packet.is_first_packet_in_frame = 1;
>   packet.timestamp = 5;
>   packet.seqNum = 5;
>-  packet.video_header.codecHeader.VP8.tl0PicIdx = 2;
>-  packet.video_header.codecHeader.VP8.temporalIdx = 0;
>-  packet.video_header.codecHeader.VP8.pictureId = 5;
>+  packet.video_header.vp8().tl0PicIdx = 2;
>+  packet.video_header.vp8().temporalIdx = 0;
>+  packet.video_header.vp8().pictureId = 5;
>   EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
>   EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
>   dec_state.SetState(&frame);
>@@ -276,9 +276,9 @@ TEST(TestDecodingState, MultiLayerBehavior) {
>   packet.frameType = kVideoFrameDelta;
>   packet.timestamp = 6;
>   packet.seqNum = 6;
>-  packet.video_header.codecHeader.VP8.tl0PicIdx = 3;
>-  packet.video_header.codecHeader.VP8.temporalIdx = 0;
>-  packet.video_header.codecHeader.VP8.pictureId = 6;
>+  packet.video_header.vp8().tl0PicIdx = 3;
>+  packet.video_header.vp8().temporalIdx = 0;
>+  packet.video_header.vp8().pictureId = 6;
>   EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
>   EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
>   EXPECT_TRUE(dec_state.full_sync());
>@@ -287,9 +287,9 @@ TEST(TestDecodingState, MultiLayerBehavior) {
>   packet.is_first_packet_in_frame = 1;
>   packet.timestamp = 8;
>   packet.seqNum = 8;
>-  packet.video_header.codecHeader.VP8.tl0PicIdx = 4;
>-  packet.video_header.codecHeader.VP8.temporalIdx = 0;
>-  packet.video_header.codecHeader.VP8.pictureId = 8;
>+  packet.video_header.vp8().tl0PicIdx = 4;
>+  packet.video_header.vp8().temporalIdx = 0;
>+  packet.video_header.vp8().pictureId = 8;
>   EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
>   EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
>   EXPECT_TRUE(dec_state.full_sync());
>@@ -302,10 +302,10 @@ TEST(TestDecodingState, MultiLayerBehavior) {
>   packet.is_first_packet_in_frame = 1;
>   packet.timestamp = 9;
>   packet.seqNum = 9;
>-  packet.video_header.codecHeader.VP8.tl0PicIdx = 4;
>-  packet.video_header.codecHeader.VP8.temporalIdx = 2;
>-  packet.video_header.codecHeader.VP8.pictureId = 9;
>-  packet.video_header.codecHeader.VP8.layerSync = true;
>+  packet.video_header.vp8().tl0PicIdx = 4;
>+  packet.video_header.vp8().temporalIdx = 2;
>+  packet.video_header.vp8().pictureId = 9;
>+  packet.video_header.vp8().layerSync = true;
>   EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
>   dec_state.SetState(&frame);
>   EXPECT_TRUE(dec_state.full_sync());
>@@ -323,10 +323,10 @@ TEST(TestDecodingState, MultiLayerBehavior) {
>   packet.markerBit = 1;
>   packet.timestamp = 0;
>   packet.seqNum = 0;
>-  packet.video_header.codecHeader.VP8.tl0PicIdx = 0;
>-  packet.video_header.codecHeader.VP8.temporalIdx = 0;
>-  packet.video_header.codecHeader.VP8.pictureId = 0;
>-  packet.video_header.codecHeader.VP8.layerSync = false;
>+  packet.video_header.vp8().tl0PicIdx = 0;
>+  packet.video_header.vp8().temporalIdx = 0;
>+  packet.video_header.vp8().pictureId = 0;
>+  packet.video_header.vp8().layerSync = false;
>   EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
>   dec_state.SetState(&frame);
>   EXPECT_TRUE(dec_state.full_sync());
>@@ -337,10 +337,10 @@ TEST(TestDecodingState, MultiLayerBehavior) {
>   packet.markerBit = 0;
>   packet.timestamp = 1;
>   packet.seqNum = 1;
>-  packet.video_header.codecHeader.VP8.tl0PicIdx = 0;
>-  packet.video_header.codecHeader.VP8.temporalIdx = 2;
>-  packet.video_header.codecHeader.VP8.pictureId = 1;
>-  packet.video_header.codecHeader.VP8.layerSync = true;
>+  packet.video_header.vp8().tl0PicIdx = 0;
>+  packet.video_header.vp8().temporalIdx = 2;
>+  packet.video_header.vp8().pictureId = 1;
>+  packet.video_header.vp8().layerSync = true;
>   EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
>   EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
>   // Layer 1
>@@ -350,10 +350,10 @@ TEST(TestDecodingState, MultiLayerBehavior) {
>   packet.markerBit = 1;
>   packet.timestamp = 2;
>   packet.seqNum = 3;
>-  packet.video_header.codecHeader.VP8.tl0PicIdx = 0;
>-  packet.video_header.codecHeader.VP8.temporalIdx = 1;
>-  packet.video_header.codecHeader.VP8.pictureId = 2;
>-  packet.video_header.codecHeader.VP8.layerSync = true;
>+  packet.video_header.vp8().tl0PicIdx = 0;
>+  packet.video_header.vp8().temporalIdx = 1;
>+  packet.video_header.vp8().pictureId = 2;
>+  packet.video_header.vp8().layerSync = true;
>   EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
>   EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
>   EXPECT_TRUE(dec_state.full_sync());
>@@ -365,12 +365,12 @@ TEST(TestDecodingState, DiscontinuousPicIdContinuousSeqNum) {
>   VCMPacket packet;
>   frame.Reset();
>   packet.frameType = kVideoFrameKey;
>-  packet.video_header.codec = kRtpVideoVp8;
>+  packet.video_header.codec = kVideoCodecVP8;
>   packet.timestamp = 0;
>   packet.seqNum = 0;
>-  packet.video_header.codecHeader.VP8.tl0PicIdx = 0;
>-  packet.video_header.codecHeader.VP8.temporalIdx = 0;
>-  packet.video_header.codecHeader.VP8.pictureId = 0;
>+  packet.video_header.vp8().tl0PicIdx = 0;
>+  packet.video_header.vp8().temporalIdx = 0;
>+  packet.video_header.vp8().pictureId = 0;
>   FrameData frame_data;
>   frame_data.rtt_ms = 0;
>   frame_data.rolling_average_packets_per_frame = -1;
>@@ -384,8 +384,8 @@ TEST(TestDecodingState, DiscontinuousPicIdContinuousSeqNum) {
>   packet.frameType = kVideoFrameDelta;
>   packet.timestamp += 3000;
>   ++packet.seqNum;
>-  packet.video_header.codecHeader.VP8.temporalIdx = 1;
>-  packet.video_header.codecHeader.VP8.pictureId = 2;
>+  packet.video_header.vp8().temporalIdx = 1;
>+  packet.video_header.vp8().pictureId = 2;
>   EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
>   EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
>   dec_state.SetState(&frame);
>@@ -418,12 +418,12 @@ TEST(TestDecodingState, PictureIdRepeat) {
>   VCMFrameBuffer frame;
>   VCMPacket packet;
>   packet.frameType = kVideoFrameDelta;
>-  packet.video_header.codec = kRtpVideoVp8;
>+  packet.video_header.codec = kVideoCodecVP8;
>   packet.timestamp = 0;
>   packet.seqNum = 0;
>-  packet.video_header.codecHeader.VP8.tl0PicIdx = 0;
>-  packet.video_header.codecHeader.VP8.temporalIdx = 0;
>-  packet.video_header.codecHeader.VP8.pictureId = 0;
>+  packet.video_header.vp8().tl0PicIdx = 0;
>+  packet.video_header.vp8().temporalIdx = 0;
>+  packet.video_header.vp8().pictureId = 0;
>   FrameData frame_data;
>   frame_data.rtt_ms = 0;
>   frame_data.rolling_average_packets_per_frame = -1;
>@@ -433,15 +433,15 @@ TEST(TestDecodingState, PictureIdRepeat) {
>   frame.Reset();
>   ++packet.timestamp;
>   ++packet.seqNum;
>-  packet.video_header.codecHeader.VP8.temporalIdx++;
>-  packet.video_header.codecHeader.VP8.pictureId++;
>+  packet.video_header.vp8().temporalIdx++;
>+  packet.video_header.vp8().pictureId++;
>   EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
>   EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
>   frame.Reset();
>   // Testing only gap in tl0PicIdx when tl0PicIdx in continuous.
>-  packet.video_header.codecHeader.VP8.tl0PicIdx += 3;
>-  packet.video_header.codecHeader.VP8.temporalIdx++;
>-  packet.video_header.codecHeader.VP8.tl0PicIdx = 1;
>+  packet.video_header.vp8().tl0PicIdx += 3;
>+  packet.video_header.vp8().temporalIdx++;
>+  packet.video_header.vp8().tl0PicIdx = 1;
>   EXPECT_LE(0, frame.InsertPacket(packet, 0, kNoErrors, frame_data));
>   EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
> }
>@@ -456,9 +456,10 @@ TEST(TestDecodingState, FrameContinuityFlexibleModeKeyFrame) {
>   uint8_t data[] = "I need a data pointer for this test!";
>   packet.sizeBytes = sizeof(data);
>   packet.dataPtr = data;
>-  packet.video_header.codec = kRtpVideoVp9;
>+  packet.video_header.codec = kVideoCodecVP9;
> 
>-  RTPVideoHeaderVP9& vp9_hdr = packet.video_header.codecHeader.VP9;
>+  auto& vp9_hdr =
>+      packet.video_header.video_type_header.emplace<RTPVideoHeaderVP9>();
>   vp9_hdr.picture_id = 10;
>   vp9_hdr.flexible_mode = true;
> 
>@@ -499,9 +500,10 @@ TEST(TestDecodingState, FrameContinuityFlexibleModeOutOfOrderFrames) {
>   uint8_t data[] = "I need a data pointer for this test!";
>   packet.sizeBytes = sizeof(data);
>   packet.dataPtr = data;
>-  packet.video_header.codec = kRtpVideoVp9;
>+  packet.video_header.codec = kVideoCodecVP9;
> 
>-  RTPVideoHeaderVP9& vp9_hdr = packet.video_header.codecHeader.VP9;
>+  auto& vp9_hdr =
>+      packet.video_header.video_type_header.emplace<RTPVideoHeaderVP9>();
>   vp9_hdr.picture_id = 10;
>   vp9_hdr.flexible_mode = true;
> 
>@@ -554,9 +556,10 @@ TEST(TestDecodingState, FrameContinuityFlexibleModeGeneral) {
>   uint8_t data[] = "I need a data pointer for this test!";
>   packet.sizeBytes = sizeof(data);
>   packet.dataPtr = data;
>-  packet.video_header.codec = kRtpVideoVp9;
>+  packet.video_header.codec = kVideoCodecVP9;
> 
>-  RTPVideoHeaderVP9& vp9_hdr = packet.video_header.codecHeader.VP9;
>+  auto& vp9_hdr =
>+      packet.video_header.video_type_header.emplace<RTPVideoHeaderVP9>();
>   vp9_hdr.picture_id = 10;
>   vp9_hdr.flexible_mode = true;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/encoded_frame.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/encoded_frame.cc
>index 2d8716cb8c2..91ee8713226 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/encoded_frame.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/encoded_frame.cc
>@@ -8,10 +8,7 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "modules/video_coding/include/video_coding_defines.h"
> #include "modules/video_coding/encoded_frame.h"
>-#include "modules/video_coding/generic_encoder.h"
>-#include "modules/video_coding/jitter_buffer_common.h"
> 
> namespace webrtc {
> 
>@@ -20,46 +17,9 @@ VCMEncodedFrame::VCMEncodedFrame()
>       _renderTimeMs(-1),
>       _payloadType(0),
>       _missingFrame(false),
>-      _codec(kVideoCodecUnknown),
>+      _codec(kVideoCodecGeneric),
>       _rotation_set(false) {
>-  _codecSpecificInfo.codecType = kVideoCodecUnknown;
>-}
>-
>-VCMEncodedFrame::VCMEncodedFrame(const webrtc::EncodedImage& rhs)
>-    : webrtc::EncodedImage(rhs),
>-      _renderTimeMs(-1),
>-      _payloadType(0),
>-      _missingFrame(false),
>-      _codec(kVideoCodecUnknown),
>-      _rotation_set(false) {
>-  _codecSpecificInfo.codecType = kVideoCodecUnknown;
>-  _buffer = NULL;
>-  _size = 0;
>-  _length = 0;
>-  if (rhs._buffer != NULL) {
>-    VerifyAndAllocate(rhs._length +
>-                      EncodedImage::GetBufferPaddingBytes(_codec));
>-    memcpy(_buffer, rhs._buffer, rhs._length);
>-  }
>-}
>-
>-VCMEncodedFrame::VCMEncodedFrame(const VCMEncodedFrame& rhs)
>-    : webrtc::EncodedImage(rhs),
>-      _renderTimeMs(rhs._renderTimeMs),
>-      _payloadType(rhs._payloadType),
>-      _missingFrame(rhs._missingFrame),
>-      _codecSpecificInfo(rhs._codecSpecificInfo),
>-      _codec(rhs._codec),
>-      _rotation_set(rhs._rotation_set) {
>-  _buffer = NULL;
>-  _size = 0;
>-  _length = 0;
>-  if (rhs._buffer != NULL) {
>-    VerifyAndAllocate(rhs._length +
>-                      EncodedImage::GetBufferPaddingBytes(_codec));
>-    memcpy(_buffer, rhs._buffer, rhs._length);
>-    _length = rhs._length;
>-  }
>+  _codecSpecificInfo.codecType = kVideoCodecGeneric;
> }
> 
> VCMEncodedFrame::~VCMEncodedFrame() {
>@@ -84,139 +44,101 @@ void VCMEncodedFrame::Reset() {
>   _completeFrame = false;
>   _missingFrame = false;
>   _length = 0;
>-  _codecSpecificInfo.codecType = kVideoCodecUnknown;
>-  _codec = kVideoCodecUnknown;
>+  _codecSpecificInfo.codecType = kVideoCodecGeneric;
>+  _codec = kVideoCodecGeneric;
>   rotation_ = kVideoRotation_0;
>   content_type_ = VideoContentType::UNSPECIFIED;
>-  timing_.flags = TimingFrameFlags::kInvalid;
>+  timing_.flags = VideoSendTiming::kInvalid;
>   _rotation_set = false;
> }
> 
> void VCMEncodedFrame::CopyCodecSpecific(const RTPVideoHeader* header) {
>   if (header) {
>     switch (header->codec) {
>-      case kRtpVideoVp8: {
>+      case kVideoCodecVP8: {
>         if (_codecSpecificInfo.codecType != kVideoCodecVP8) {
>           // This is the first packet for this frame.
>-          _codecSpecificInfo.codecSpecific.VP8.pictureId = -1;
>           _codecSpecificInfo.codecSpecific.VP8.temporalIdx = 0;
>           _codecSpecificInfo.codecSpecific.VP8.layerSync = false;
>           _codecSpecificInfo.codecSpecific.VP8.keyIdx = -1;
>           _codecSpecificInfo.codecType = kVideoCodecVP8;
>         }
>         _codecSpecificInfo.codecSpecific.VP8.nonReference =
>-            header->codecHeader.VP8.nonReference;
>-        if (header->codecHeader.VP8.pictureId != kNoPictureId) {
>-          _codecSpecificInfo.codecSpecific.VP8.pictureId =
>-              header->codecHeader.VP8.pictureId;
>-        }
>-        if (header->codecHeader.VP8.temporalIdx != kNoTemporalIdx) {
>+            header->vp8().nonReference;
>+        if (header->vp8().temporalIdx != kNoTemporalIdx) {
>           _codecSpecificInfo.codecSpecific.VP8.temporalIdx =
>-              header->codecHeader.VP8.temporalIdx;
>+              header->vp8().temporalIdx;
>           _codecSpecificInfo.codecSpecific.VP8.layerSync =
>-              header->codecHeader.VP8.layerSync;
>+              header->vp8().layerSync;
>         }
>-        if (header->codecHeader.VP8.keyIdx != kNoKeyIdx) {
>-          _codecSpecificInfo.codecSpecific.VP8.keyIdx =
>-              header->codecHeader.VP8.keyIdx;
>+        if (header->vp8().keyIdx != kNoKeyIdx) {
>+          _codecSpecificInfo.codecSpecific.VP8.keyIdx = header->vp8().keyIdx;
>         }
>         break;
>       }
>-      case kRtpVideoVp9: {
>+      case kVideoCodecVP9: {
>+        const auto& vp9_header =
>+            absl::get<RTPVideoHeaderVP9>(header->video_type_header);
>         if (_codecSpecificInfo.codecType != kVideoCodecVP9) {
>           // This is the first packet for this frame.
>-          _codecSpecificInfo.codecSpecific.VP9.picture_id = -1;
>           _codecSpecificInfo.codecSpecific.VP9.temporal_idx = 0;
>           _codecSpecificInfo.codecSpecific.VP9.spatial_idx = 0;
>           _codecSpecificInfo.codecSpecific.VP9.gof_idx = 0;
>           _codecSpecificInfo.codecSpecific.VP9.inter_layer_predicted = false;
>-          _codecSpecificInfo.codecSpecific.VP9.tl0_pic_idx = -1;
>           _codecSpecificInfo.codecType = kVideoCodecVP9;
>         }
>         _codecSpecificInfo.codecSpecific.VP9.inter_pic_predicted =
>-            header->codecHeader.VP9.inter_pic_predicted;
>+            vp9_header.inter_pic_predicted;
>         _codecSpecificInfo.codecSpecific.VP9.flexible_mode =
>-            header->codecHeader.VP9.flexible_mode;
>+            vp9_header.flexible_mode;
>         _codecSpecificInfo.codecSpecific.VP9.num_ref_pics =
>-            header->codecHeader.VP9.num_ref_pics;
>-        for (uint8_t r = 0; r < header->codecHeader.VP9.num_ref_pics; ++r) {
>+            vp9_header.num_ref_pics;
>+        for (uint8_t r = 0; r < vp9_header.num_ref_pics; ++r) {
>           _codecSpecificInfo.codecSpecific.VP9.p_diff[r] =
>-              header->codecHeader.VP9.pid_diff[r];
>+              vp9_header.pid_diff[r];
>         }
>         _codecSpecificInfo.codecSpecific.VP9.ss_data_available =
>-            header->codecHeader.VP9.ss_data_available;
>-        if (header->codecHeader.VP9.picture_id != kNoPictureId) {
>-          _codecSpecificInfo.codecSpecific.VP9.picture_id =
>-              header->codecHeader.VP9.picture_id;
>-        }
>-        if (header->codecHeader.VP9.tl0_pic_idx != kNoTl0PicIdx) {
>-          _codecSpecificInfo.codecSpecific.VP9.tl0_pic_idx =
>-              header->codecHeader.VP9.tl0_pic_idx;
>-        }
>-        if (header->codecHeader.VP9.temporal_idx != kNoTemporalIdx) {
>+            vp9_header.ss_data_available;
>+        if (vp9_header.temporal_idx != kNoTemporalIdx) {
>           _codecSpecificInfo.codecSpecific.VP9.temporal_idx =
>-              header->codecHeader.VP9.temporal_idx;
>+              vp9_header.temporal_idx;
>           _codecSpecificInfo.codecSpecific.VP9.temporal_up_switch =
>-              header->codecHeader.VP9.temporal_up_switch;
>+              vp9_header.temporal_up_switch;
>         }
>-        if (header->codecHeader.VP9.spatial_idx != kNoSpatialIdx) {
>+        if (vp9_header.spatial_idx != kNoSpatialIdx) {
>           _codecSpecificInfo.codecSpecific.VP9.spatial_idx =
>-              header->codecHeader.VP9.spatial_idx;
>+              vp9_header.spatial_idx;
>           _codecSpecificInfo.codecSpecific.VP9.inter_layer_predicted =
>-              header->codecHeader.VP9.inter_layer_predicted;
>+              vp9_header.inter_layer_predicted;
>         }
>-        if (header->codecHeader.VP9.gof_idx != kNoGofIdx) {
>-          _codecSpecificInfo.codecSpecific.VP9.gof_idx =
>-              header->codecHeader.VP9.gof_idx;
>+        if (vp9_header.gof_idx != kNoGofIdx) {
>+          _codecSpecificInfo.codecSpecific.VP9.gof_idx = vp9_header.gof_idx;
>         }
>-        if (header->codecHeader.VP9.ss_data_available) {
>+        if (vp9_header.ss_data_available) {
>           _codecSpecificInfo.codecSpecific.VP9.num_spatial_layers =
>-              header->codecHeader.VP9.num_spatial_layers;
>+              vp9_header.num_spatial_layers;
>           _codecSpecificInfo.codecSpecific.VP9
>               .spatial_layer_resolution_present =
>-              header->codecHeader.VP9.spatial_layer_resolution_present;
>-          if (header->codecHeader.VP9.spatial_layer_resolution_present) {
>-            for (size_t i = 0; i < header->codecHeader.VP9.num_spatial_layers;
>-                 ++i) {
>+              vp9_header.spatial_layer_resolution_present;
>+          if (vp9_header.spatial_layer_resolution_present) {
>+            for (size_t i = 0; i < vp9_header.num_spatial_layers; ++i) {
>               _codecSpecificInfo.codecSpecific.VP9.width[i] =
>-                  header->codecHeader.VP9.width[i];
>+                  vp9_header.width[i];
>               _codecSpecificInfo.codecSpecific.VP9.height[i] =
>-                  header->codecHeader.VP9.height[i];
>+                  vp9_header.height[i];
>             }
>           }
>           _codecSpecificInfo.codecSpecific.VP9.gof.CopyGofInfoVP9(
>-              header->codecHeader.VP9.gof);
>+              vp9_header.gof);
>         }
>         break;
>       }
>-      case kRtpVideoH264: {
>+      case kVideoCodecH264: {
>         _codecSpecificInfo.codecType = kVideoCodecH264;
>         break;
>       }
>-      case kRtpVideoStereo: {
>-        _codecSpecificInfo.codecType = kVideoCodecStereo;
>-        VideoCodecType associated_codec_type = kVideoCodecUnknown;
>-        switch (header->codecHeader.stereo.associated_codec_type) {
>-          case kRtpVideoVp8:
>-            associated_codec_type = kVideoCodecVP8;
>-            break;
>-          case kRtpVideoVp9:
>-            associated_codec_type = kVideoCodecVP9;
>-            break;
>-          case kRtpVideoH264:
>-            associated_codec_type = kVideoCodecH264;
>-            break;
>-          default:
>-            RTC_NOTREACHED();
>-        }
>-        _codecSpecificInfo.codecSpecific.stereo.associated_codec_type =
>-            associated_codec_type;
>-        _codecSpecificInfo.codecSpecific.stereo.indices =
>-            header->codecHeader.stereo.indices;
>-        break;
>-      }
>       default: {
>-        _codecSpecificInfo.codecType = kVideoCodecUnknown;
>+        _codecSpecificInfo.codecType = kVideoCodecGeneric;
>         break;
>       }
>     }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/encoded_frame.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/encoded_frame.h
>index 80f0dcc86bc..252dea483f8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/encoded_frame.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/encoded_frame.h
>@@ -24,24 +24,23 @@ namespace webrtc {
> class VCMEncodedFrame : protected EncodedImage {
>  public:
>   VCMEncodedFrame();
>-  explicit VCMEncodedFrame(const webrtc::EncodedImage& rhs);
>-  VCMEncodedFrame(const VCMEncodedFrame& rhs);
>+  VCMEncodedFrame(const VCMEncodedFrame&) = delete;
> 
>   ~VCMEncodedFrame();
>   /**
>-  *   Delete VideoFrame and resets members to zero
>-  */
>+   *   Delete VideoFrame and resets members to zero
>+   */
>   void Free();
>   /**
>-  *   Set render time in milliseconds
>-  */
>+   *   Set render time in milliseconds
>+   */
>   void SetRenderTime(const int64_t renderTimeMs) {
>     _renderTimeMs = renderTimeMs;
>   }
> 
>   /**
>-  *   Set the encoded frame size
>-  */
>+   *   Set the encoded frame size
>+   */
>   void SetEncodedSize(uint32_t width, uint32_t height) {
>     _encodedWidth = width;
>     _encodedHeight = height;
>@@ -52,34 +51,34 @@ class VCMEncodedFrame : protected EncodedImage {
>   }
> 
>   /**
>-  *   Get the encoded image
>-  */
>+   *   Get the encoded image
>+   */
>   const webrtc::EncodedImage& EncodedImage() const {
>     return static_cast<const webrtc::EncodedImage&>(*this);
>   }
>   /**
>-  *   Get pointer to frame buffer
>-  */
>+   *   Get pointer to frame buffer
>+   */
>   const uint8_t* Buffer() const { return _buffer; }
>   /**
>-  *   Get frame length
>-  */
>+   *   Get frame length
>+   */
>   size_t Length() const { return _length; }
>   /**
>-  *   Get frame timestamp (90kHz)
>-  */
>+   *   Get frame timestamp (90kHz)
>+   */
>   uint32_t TimeStamp() const { return _timeStamp; }
>   /**
>-  *   Get render time in milliseconds
>-  */
>+   *   Get render time in milliseconds
>+   */
>   int64_t RenderTimeMs() const { return _renderTimeMs; }
>   /**
>-  *   Get frame type
>-  */
>+   *   Get frame type
>+   */
>   webrtc::FrameType FrameType() const { return _frameType; }
>   /**
>-  *   Get frame rotation
>-  */
>+   *   Get frame rotation
>+   */
>   VideoRotation rotation() const { return rotation_; }
>   /**
>    *  Get video content type
>@@ -94,30 +93,30 @@ class VCMEncodedFrame : protected EncodedImage {
>    */
>   bool Complete() const { return _completeFrame; }
>   /**
>-  *   True if there's a frame missing before this frame
>-  */
>+   *   True if there's a frame missing before this frame
>+   */
>   bool MissingFrame() const { return _missingFrame; }
>   /**
>-  *   Payload type of the encoded payload
>-  */
>+   *   Payload type of the encoded payload
>+   */
>   uint8_t PayloadType() const { return _payloadType; }
>   /**
>-  *   Get codec specific info.
>-  *   The returned pointer is only valid as long as the VCMEncodedFrame
>-  *   is valid. Also, VCMEncodedFrame owns the pointer and will delete
>-  *   the object.
>-  */
>+   *   Get codec specific info.
>+   *   The returned pointer is only valid as long as the VCMEncodedFrame
>+   *   is valid. Also, VCMEncodedFrame owns the pointer and will delete
>+   *   the object.
>+   */
>   const CodecSpecificInfo* CodecSpecific() const { return &_codecSpecificInfo; }
> 
>  protected:
>   /**
>-  * Verifies that current allocated buffer size is larger than or equal to the
>-  * input size.
>-  * If the current buffer size is smaller, a new allocation is made and the old
>-  * buffer data
>-  * is copied to the new buffer.
>-  * Buffer size is updated to minimumSize.
>-  */
>+   * Verifies that current allocated buffer size is larger than or equal to the
>+   * input size.
>+   * If the current buffer size is smaller, a new allocation is made and the old
>+   * buffer data
>+   * is copied to the new buffer.
>+   * Buffer size is updated to minimumSize.
>+   */
>   void VerifyAndAllocate(size_t minimumSize);
> 
>   void Reset();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/encoder_database.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/encoder_database.cc
>new file mode 100644
>index 00000000000..5a89590c0ea
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/encoder_database.cc
>@@ -0,0 +1,183 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/video_coding/encoder_database.h"
>+
>+#include "rtc_base/checks.h"
>+#include "rtc_base/logging.h"
>+
>+namespace webrtc {
>+
>+namespace {
>+const size_t kDefaultPayloadSize = 1440;
>+}
>+
>+VCMEncoderDataBase::VCMEncoderDataBase(
>+    VCMEncodedFrameCallback* encoded_frame_callback)
>+    : number_of_cores_(0),
>+      max_payload_size_(kDefaultPayloadSize),
>+      pending_encoder_reset_(true),
>+      send_codec_(),
>+      external_encoder_(nullptr),
>+      internal_source_(false),
>+      encoded_frame_callback_(encoded_frame_callback) {}
>+
>+VCMEncoderDataBase::~VCMEncoderDataBase() {
>+  DeleteEncoder();
>+}
>+
>+// Assuming only one registered encoder - since only one used, no need for more.
>+bool VCMEncoderDataBase::SetSendCodec(const VideoCodec* send_codec,
>+                                      int number_of_cores,
>+                                      size_t max_payload_size) {
>+  RTC_DCHECK(send_codec);
>+  if (max_payload_size == 0) {
>+    max_payload_size = kDefaultPayloadSize;
>+  }
>+  RTC_DCHECK_GE(number_of_cores, 1);
>+  // Make sure the start bit rate is sane...
>+  RTC_DCHECK_LE(send_codec->startBitrate, 1000000);
>+  bool reset_required = pending_encoder_reset_;
>+  if (number_of_cores_ != number_of_cores) {
>+    number_of_cores_ = number_of_cores;
>+    reset_required = true;
>+  }
>+  if (max_payload_size_ != max_payload_size) {
>+    max_payload_size_ = max_payload_size;
>+    reset_required = true;
>+  }
>+
>+  VideoCodec new_send_codec;
>+  memcpy(&new_send_codec, send_codec, sizeof(new_send_codec));
>+
>+  if (new_send_codec.maxBitrate == 0) {
>+    // max is one bit per pixel
>+    new_send_codec.maxBitrate = (static_cast<int>(send_codec->height) *
>+                                 static_cast<int>(send_codec->width) *
>+                                 static_cast<int>(send_codec->maxFramerate)) /
>+                                1000;
>+    if (send_codec->startBitrate > new_send_codec.maxBitrate) {
>+      // But if the user tries to set a higher start bit rate we will
>+      // increase the max accordingly.
>+      new_send_codec.maxBitrate = send_codec->startBitrate;
>+    }
>+  }
>+
>+  if (new_send_codec.startBitrate > new_send_codec.maxBitrate)
>+    new_send_codec.startBitrate = new_send_codec.maxBitrate;
>+
>+  if (!reset_required) {
>+    reset_required = RequiresEncoderReset(new_send_codec);
>+  }
>+
>+  memcpy(&send_codec_, &new_send_codec, sizeof(send_codec_));
>+
>+  if (!reset_required) {
>+    return true;
>+  }
>+
>+  // If encoder exists, will destroy it and create new one.
>+  DeleteEncoder();
>+  ptr_encoder_.reset(new VCMGenericEncoder(
>+      external_encoder_, encoded_frame_callback_, internal_source_));
>+  encoded_frame_callback_->SetInternalSource(internal_source_);
>+  if (ptr_encoder_->InitEncode(&send_codec_, number_of_cores_,
>+                               max_payload_size_) < 0) {
>+    RTC_LOG(LS_ERROR) << "Failed to initialize video encoder.";
>+    DeleteEncoder();
>+    return false;
>+  }
>+
>+  pending_encoder_reset_ = false;
>+
>+  return true;
>+}
>+
>+void VCMEncoderDataBase::DeregisterExternalEncoder() {
>+  DeleteEncoder();
>+  memset(&send_codec_, 0, sizeof(VideoCodec));
>+  external_encoder_ = nullptr;
>+  internal_source_ = false;
>+}
>+
>+void VCMEncoderDataBase::RegisterExternalEncoder(VideoEncoder* external_encoder,
>+                                                 bool internal_source) {
>+  // Since only one encoder can be used at a given time, only one external
>+  // encoder can be registered/used.
>+  RTC_CHECK(external_encoder_ == nullptr);
>+  external_encoder_ = external_encoder;
>+  internal_source_ = internal_source;
>+  pending_encoder_reset_ = true;
>+}
>+
>+bool VCMEncoderDataBase::RequiresEncoderReset(
>+    const VideoCodec& new_send_codec) {
>+  if (!ptr_encoder_)
>+    return true;
>+
>+  // Does not check startBitrate, maxFramerate or plType
>+  if (new_send_codec.codecType != send_codec_.codecType ||
>+      new_send_codec.width != send_codec_.width ||
>+      new_send_codec.height != send_codec_.height ||
>+      new_send_codec.maxBitrate != send_codec_.maxBitrate ||
>+      new_send_codec.minBitrate != send_codec_.minBitrate ||
>+      new_send_codec.qpMax != send_codec_.qpMax ||
>+      new_send_codec.numberOfSimulcastStreams !=
>+          send_codec_.numberOfSimulcastStreams ||
>+      new_send_codec.mode != send_codec_.mode) {
>+    return true;
>+  }
>+
>+  switch (new_send_codec.codecType) {
>+    case kVideoCodecVP8:
>+      if (new_send_codec.VP8() != *send_codec_.VP8()) {
>+        return true;
>+      }
>+      break;
>+
>+    case kVideoCodecVP9:
>+      if (new_send_codec.VP9() != *send_codec_.VP9()) {
>+        return true;
>+      }
>+      break;
>+
>+    case kVideoCodecH264:
>+      if (new_send_codec.H264() != *send_codec_.H264()) {
>+        return true;
>+      }
>+      break;
>+
>+    default:
>+      break;
>+  }
>+
>+  for (unsigned char i = 0; i < new_send_codec.numberOfSimulcastStreams; ++i) {
>+    if (new_send_codec.simulcastStream[i] != send_codec_.simulcastStream[i])
>+      return true;
>+  }
>+  return false;
>+}
>+
>+VCMGenericEncoder* VCMEncoderDataBase::GetEncoder() {
>+  return ptr_encoder_.get();
>+}
>+
>+void VCMEncoderDataBase::DeleteEncoder() {
>+  if (!ptr_encoder_)
>+    return;
>+  ptr_encoder_->Release();
>+  ptr_encoder_.reset();
>+}
>+
>+bool VCMEncoderDataBase::MatchesCurrentResolution(int width, int height) const {
>+  return send_codec_.width == width && send_codec_.height == height;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/encoder_database.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/encoder_database.h
>new file mode 100644
>index 00000000000..a9db4e15c80
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/encoder_database.h
>@@ -0,0 +1,65 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_VIDEO_CODING_ENCODER_DATABASE_H_
>+#define MODULES_VIDEO_CODING_ENCODER_DATABASE_H_
>+
>+#include <memory>
>+
>+#include "modules/video_coding/generic_encoder.h"
>+
>+namespace webrtc {
>+
>+class VCMEncoderDataBase {
>+ public:
>+  explicit VCMEncoderDataBase(VCMEncodedFrameCallback* encoded_frame_callback);
>+  ~VCMEncoderDataBase();
>+
>+  // Sets the sender side codec and initiates the desired codec given the
>+  // VideoCodec struct.
>+  // Returns true if the codec was successfully registered, false otherwise.
>+  bool SetSendCodec(const VideoCodec* send_codec,
>+                    int number_of_cores,
>+                    size_t max_payload_size);
>+
>+  // Registers and initializes an external encoder object.
>+  // |internal_source| should be set to true if the codec has an internal
>+  // video source and doesn't need the user to provide it with frames via
>+  // the Encode() method.
>+  void RegisterExternalEncoder(VideoEncoder* external_encoder,
>+                               bool internal_source);
>+
>+  // Deregisters any external encoder.
>+  void DeregisterExternalEncoder();
>+
>+  VCMGenericEncoder* GetEncoder();
>+
>+  bool MatchesCurrentResolution(int width, int height) const;
>+
>+ private:
>+  // Determines whether a new codec has to be created or not.
>+  // Checks every setting apart from maxFramerate and startBitrate.
>+  bool RequiresEncoderReset(const VideoCodec& send_codec);
>+
>+  void DeleteEncoder();
>+
>+  int number_of_cores_;
>+  size_t max_payload_size_;
>+  bool pending_encoder_reset_;
>+  VideoCodec send_codec_;
>+  VideoEncoder* external_encoder_;
>+  bool internal_source_;
>+  VCMEncodedFrameCallback* const encoded_frame_callback_;
>+  std::unique_ptr<VCMGenericEncoder> ptr_encoder_;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_VIDEO_CODING_ENCODER_DATABASE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/protection_bitrate_calculator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/fec_controller_default.cc
>similarity index 82%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/protection_bitrate_calculator.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/fec_controller_default.cc
>index 3c1da234d9c..84e8a7bfcc5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/protection_bitrate_calculator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/fec_controller_default.cc
>@@ -8,13 +8,11 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "modules/video_coding/protection_bitrate_calculator.h"
>+#include "modules/video_coding/fec_controller_default.h"
> 
> namespace webrtc {
>-
> using rtc::CritScope;
>-
>-ProtectionBitrateCalculator::ProtectionBitrateCalculator(
>+FecControllerDefault::FecControllerDefault(
>     Clock* clock,
>     VCMProtectionCallback* protection_callback)
>     : clock_(clock),
>@@ -23,24 +21,36 @@ ProtectionBitrateCalculator::ProtectionBitrateCalculator(
>           clock_->TimeInMilliseconds())),
>       max_payload_size_(1460) {}
> 
>-ProtectionBitrateCalculator::~ProtectionBitrateCalculator(void) {
>+FecControllerDefault::FecControllerDefault(Clock* clock)
>+    : clock_(clock),
>+      loss_prot_logic_(new media_optimization::VCMLossProtectionLogic(
>+          clock_->TimeInMilliseconds())),
>+      max_payload_size_(1460) {}
>+
>+FecControllerDefault::~FecControllerDefault(void) {
>   loss_prot_logic_->Release();
> }
> 
>-void ProtectionBitrateCalculator::SetEncodingData(size_t width,
>-                                                  size_t height,
>-                                                  size_t num_temporal_layers,
>-                                                  size_t max_payload_size) {
>+void FecControllerDefault::SetProtectionCallback(
>+    VCMProtectionCallback* protection_callback) {
>+  protection_callback_ = protection_callback;
>+}
>+
>+void FecControllerDefault::SetEncodingData(size_t width,
>+                                           size_t height,
>+                                           size_t num_temporal_layers,
>+                                           size_t max_payload_size) {
>   CritScope lock(&crit_sect_);
>   loss_prot_logic_->UpdateFrameSize(width, height);
>   loss_prot_logic_->UpdateNumLayers(num_temporal_layers);
>   max_payload_size_ = max_payload_size;
> }
> 
>-uint32_t ProtectionBitrateCalculator::SetTargetRates(
>+uint32_t FecControllerDefault::UpdateFecRates(
>     uint32_t estimated_bitrate_bps,
>     int actual_framerate_fps,
>     uint8_t fraction_lost,
>+    std::vector<bool> loss_mask_vector,
>     int64_t round_trip_time_ms) {
>   float target_bitrate_kbps =
>       static_cast<float>(estimated_bitrate_bps) / 1000.0f;
>@@ -48,19 +58,15 @@ uint32_t ProtectionBitrateCalculator::SetTargetRates(
>   if (actual_framerate_fps < 1.0) {
>     actual_framerate_fps = 1.0;
>   }
>-
>   FecProtectionParams delta_fec_params;
>   FecProtectionParams key_fec_params;
>   {
>     CritScope lock(&crit_sect_);
>-
>     loss_prot_logic_->UpdateBitRate(target_bitrate_kbps);
>     loss_prot_logic_->UpdateRtt(round_trip_time_ms);
>-
>     // Update frame rate for the loss protection logic class: frame rate should
>     // be the actual/sent rate.
>     loss_prot_logic_->UpdateFrameRate(actual_framerate_fps);
>-
>     // Returns the filtered packet loss, used for the protection setting.
>     // The filtered loss may be the received loss (no filter), or some
>     // filtered value (average or max window filter).
>@@ -69,31 +75,24 @@ uint32_t ProtectionBitrateCalculator::SetTargetRates(
>         media_optimization::kMaxFilter;
>     uint8_t packet_loss_enc = loss_prot_logic_->FilteredLoss(
>         clock_->TimeInMilliseconds(), filter_mode, fraction_lost);
>-
>     // For now use the filtered loss for computing the robustness settings.
>     loss_prot_logic_->UpdateFilteredLossPr(packet_loss_enc);
>-
>     if (loss_prot_logic_->SelectedType() == media_optimization::kNone) {
>       return estimated_bitrate_bps;
>     }
>-
>     // Update method will compute the robustness settings for the given
>     // protection method and the overhead cost
>     // the protection method is set by the user via SetVideoProtection.
>     loss_prot_logic_->UpdateMethod();
>-
>     // Get the bit cost of protection method, based on the amount of
>     // overhead data actually transmitted (including headers) the last
>     // second.
>-
>     // Get the FEC code rate for Key frames (set to 0 when NA).
>     key_fec_params.fec_rate =
>         loss_prot_logic_->SelectedMethod()->RequiredProtectionFactorK();
>-
>     // Get the FEC code rate for Delta frames (set to 0 when NA).
>     delta_fec_params.fec_rate =
>         loss_prot_logic_->SelectedMethod()->RequiredProtectionFactorD();
>-
>     // The RTP module currently requires the same |max_fec_frames| for both
>     // key and delta frames.
>     delta_fec_params.max_fec_frames =
>@@ -101,26 +100,22 @@ uint32_t ProtectionBitrateCalculator::SetTargetRates(
>     key_fec_params.max_fec_frames =
>         loss_prot_logic_->SelectedMethod()->MaxFramesFec();
>   }
>-
>   // Set the FEC packet mask type. |kFecMaskBursty| is more effective for
>   // consecutive losses and little/no packet re-ordering. As we currently
>   // do not have feedback data on the degree of correlated losses and packet
>   // re-ordering, we keep default setting to |kFecMaskRandom| for now.
>   delta_fec_params.fec_mask_type = kFecMaskRandom;
>   key_fec_params.fec_mask_type = kFecMaskRandom;
>-
>   // Update protection callback with protection settings.
>   uint32_t sent_video_rate_bps = 0;
>   uint32_t sent_nack_rate_bps = 0;
>   uint32_t sent_fec_rate_bps = 0;
>   // Rate cost of the protection methods.
>   float protection_overhead_rate = 0.0f;
>-
>   // TODO(Marco): Pass FEC protection values per layer.
>   protection_callback_->ProtectionRequest(
>       &delta_fec_params, &key_fec_params, &sent_video_rate_bps,
>       &sent_nack_rate_bps, &sent_fec_rate_bps);
>-
>   uint32_t sent_total_rate_bps =
>       sent_video_rate_bps + sent_nack_rate_bps + sent_fec_rate_bps;
>   // Estimate the overhead costs of the next second as staying the same
>@@ -133,13 +128,11 @@ uint32_t ProtectionBitrateCalculator::SetTargetRates(
>   // Cap the overhead estimate to 50%.
>   if (protection_overhead_rate > 0.5)
>     protection_overhead_rate = 0.5;
>-
>   // Source coding rate: total rate - protection overhead.
>   return estimated_bitrate_bps * (1.0 - protection_overhead_rate);
> }
>-
>-void ProtectionBitrateCalculator::SetProtectionMethod(bool enable_fec,
>-                                                      bool enable_nack) {
>+void FecControllerDefault::SetProtectionMethod(bool enable_fec,
>+                                               bool enable_nack) {
>   media_optimization::VCMProtectionMethodEnum method(media_optimization::kNone);
>   if (enable_fec && enable_nack) {
>     method = media_optimization::kNackFec;
>@@ -151,14 +144,13 @@ void ProtectionBitrateCalculator::SetProtectionMethod(bool enable_fec,
>   CritScope lock(&crit_sect_);
>   loss_prot_logic_->SetMethod(method);
> }
>-
>-void ProtectionBitrateCalculator::UpdateWithEncodedData(
>-    const EncodedImage& encoded_image) {
>-  const size_t encoded_length = encoded_image._length;
>+void FecControllerDefault::UpdateWithEncodedData(
>+    const size_t encoded_image_length,
>+    const FrameType encoded_image_frametype) {
>+  const size_t encoded_length = encoded_image_length;
>   CritScope lock(&crit_sect_);
>   if (encoded_length > 0) {
>-    const bool delta_frame = encoded_image._frameType != kVideoFrameKey;
>-
>+    const bool delta_frame = encoded_image_frametype != kVideoFrameKey;
>     if (max_payload_size_ > 0 && encoded_length > 0) {
>       const float min_packets_per_frame =
>           encoded_length / static_cast<float>(max_payload_size_);
>@@ -175,5 +167,4 @@ void ProtectionBitrateCalculator::UpdateWithEncodedData(
>     }
>   }
> }
>-
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/fec_controller_default.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/fec_controller_default.h
>new file mode 100644
>index 00000000000..776977a27d4
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/fec_controller_default.h
>@@ -0,0 +1,58 @@
>+/*
>+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_VIDEO_CODING_FEC_CONTROLLER_DEFAULT_H_
>+#define MODULES_VIDEO_CODING_FEC_CONTROLLER_DEFAULT_H_
>+
>+#include <memory>
>+#include <vector>
>+#include "api/fec_controller.h"
>+#include "modules/include/module_common_types.h"
>+#include "modules/video_coding/media_opt_util.h"
>+#include "rtc_base/criticalsection.h"
>+#include "system_wrappers/include/clock.h"
>+
>+namespace webrtc {
>+
>+class FecControllerDefault : public FecController {
>+ public:
>+  FecControllerDefault(Clock* clock,
>+                       VCMProtectionCallback* protection_callback);
>+  explicit FecControllerDefault(Clock* clock);
>+  ~FecControllerDefault();
>+  void SetProtectionCallback(
>+      VCMProtectionCallback* protection_callback) override;
>+  void SetProtectionMethod(bool enable_fec, bool enable_nack) override;
>+  void SetEncodingData(size_t width,
>+                       size_t height,
>+                       size_t num_temporal_layers,
>+                       size_t max_payload_size) override;
>+  uint32_t UpdateFecRates(uint32_t estimated_bitrate_bps,
>+                          int actual_framerate_fps,
>+                          uint8_t fraction_lost,
>+                          std::vector<bool> loss_mask_vector,
>+                          int64_t round_trip_time_ms) override;
>+  void UpdateWithEncodedData(const size_t encoded_image_length,
>+                             const FrameType encoded_image_frametype) override;
>+  bool UseLossVectorMask() override { return false; }
>+
>+ private:
>+  enum { kBitrateAverageWinMs = 1000 };
>+  Clock* const clock_;
>+  VCMProtectionCallback* protection_callback_;
>+  rtc::CriticalSection crit_sect_;
>+  std::unique_ptr<media_optimization::VCMLossProtectionLogic> loss_prot_logic_
>+      RTC_GUARDED_BY(crit_sect_);
>+  size_t max_payload_size_ RTC_GUARDED_BY(crit_sect_);
>+  RTC_DISALLOW_COPY_AND_ASSIGN(FecControllerDefault);
>+};
>+
>+}  // namespace webrtc
>+#endif  // MODULES_VIDEO_CODING_FEC_CONTROLLER_DEFAULT_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/protection_bitrate_calculator_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/fec_controller_unittest.cc
>similarity index 67%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/protection_bitrate_calculator_unittest.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/fec_controller_unittest.cc
>index 66e2fa6f848..51ae15808f2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/protection_bitrate_calculator_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/fec_controller_unittest.cc
>@@ -8,7 +8,7 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "modules/video_coding/protection_bitrate_calculator.h"
>+#include "modules/video_coding/fec_controller_default.h"
> #include "system_wrappers/include/clock.h"
> #include "test/gtest.h"
> 
>@@ -42,22 +42,24 @@ class ProtectionBitrateCalculatorTest : public ::testing::Test {
>   // Note: simulated clock starts at 1 seconds, since parts of webrtc use 0 as
>   // a special case (e.g. frame rate in media optimization).
>   ProtectionBitrateCalculatorTest()
>-      : clock_(1000), media_opt_(&clock_, &protection_callback_) {}
>+      : clock_(1000), fec_controller_(&clock_, &protection_callback_) {}
> 
>   SimulatedClock clock_;
>   ProtectionCallback protection_callback_;
>-  ProtectionBitrateCalculator media_opt_;
>+  FecControllerDefault fec_controller_;
> };
> 
> TEST_F(ProtectionBitrateCalculatorTest, ProtectsUsingFecBitrate) {
>   static const uint32_t kMaxBitrateBps = 130000;
> 
>-  media_opt_.SetProtectionMethod(true /*enable_fec*/, false /* enable_nack */);
>-  media_opt_.SetEncodingData(640, 480, 1, 1000);
>+  fec_controller_.SetProtectionMethod(true /*enable_fec*/,
>+                                      false /* enable_nack */);
>+  fec_controller_.SetEncodingData(640, 480, 1, 1000);
> 
>   // Using 10% of codec bitrate for FEC.
>   protection_callback_.fec_rate_bps_ = kCodecBitrateBps / 10;
>-  uint32_t target_bitrate = media_opt_.SetTargetRates(kMaxBitrateBps, 30, 0, 0);
>+  uint32_t target_bitrate = fec_controller_.UpdateFecRates(
>+      kMaxBitrateBps, 30, 0, std::vector<bool>(1, false), 0);
> 
>   EXPECT_GT(target_bitrate, 0u);
>   EXPECT_GT(kMaxBitrateBps, target_bitrate);
>@@ -65,35 +67,40 @@ TEST_F(ProtectionBitrateCalculatorTest, ProtectsUsingFecBitrate) {
>   // Using as much for codec bitrate as fec rate, new target rate should share
>   // both equally, but only be half of max (since that ceiling should be hit).
>   protection_callback_.fec_rate_bps_ = kCodecBitrateBps;
>-  target_bitrate = media_opt_.SetTargetRates(kMaxBitrateBps, 30, 128, 100);
>+  target_bitrate = fec_controller_.UpdateFecRates(
>+      kMaxBitrateBps, 30, 128, std::vector<bool>(1, false), 100);
>   EXPECT_EQ(kMaxBitrateBps / 2, target_bitrate);
> }
> 
> TEST_F(ProtectionBitrateCalculatorTest, ProtectsUsingNackBitrate) {
>   static const uint32_t kMaxBitrateBps = 130000;
> 
>-  media_opt_.SetProtectionMethod(false /*enable_fec*/, true /* enable_nack */);
>-  media_opt_.SetEncodingData(640, 480, 1, 1000);
>+  fec_controller_.SetProtectionMethod(false /*enable_fec*/,
>+                                      true /* enable_nack */);
>+  fec_controller_.SetEncodingData(640, 480, 1, 1000);
> 
>-  uint32_t target_bitrate = media_opt_.SetTargetRates(kMaxBitrateBps, 30, 0, 0);
>+  uint32_t target_bitrate = fec_controller_.UpdateFecRates(
>+      kMaxBitrateBps, 30, 0, std::vector<bool>(1, false), 0);
> 
>   EXPECT_EQ(kMaxBitrateBps, target_bitrate);
> 
>   // Using as much for codec bitrate as nack rate, new target rate should share
>   // both equally, but only be half of max (since that ceiling should be hit).
>   protection_callback_.nack_rate_bps_ = kMaxBitrateBps;
>-  target_bitrate = media_opt_.SetTargetRates(kMaxBitrateBps, 30, 128, 100);
>+  target_bitrate = fec_controller_.UpdateFecRates(
>+      kMaxBitrateBps, 30, 128, std::vector<bool>(1, false), 100);
>   EXPECT_EQ(kMaxBitrateBps / 2, target_bitrate);
> }
> 
> TEST_F(ProtectionBitrateCalculatorTest, NoProtection) {
>   static const uint32_t kMaxBitrateBps = 130000;
> 
>-  media_opt_.SetProtectionMethod(false /*enable_fec*/, false /* enable_nack */);
>-  media_opt_.SetEncodingData(640, 480, 1, 1000);
>+  fec_controller_.SetProtectionMethod(false /*enable_fec*/,
>+                                      false /* enable_nack */);
>+  fec_controller_.SetEncodingData(640, 480, 1, 1000);
> 
>-  uint32_t target_bitrate =
>-      media_opt_.SetTargetRates(kMaxBitrateBps, 30, 128, 100);
>+  uint32_t target_bitrate = fec_controller_.UpdateFecRates(
>+      kMaxBitrateBps, 30, 128, std::vector<bool>(1, false), 100);
>   EXPECT_EQ(kMaxBitrateBps, target_bitrate);
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/frame_buffer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/frame_buffer.cc
>index f06e40f69ad..b9241b93d6f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/frame_buffer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/frame_buffer.cc
>@@ -25,16 +25,6 @@ VCMFrameBuffer::VCMFrameBuffer()
> 
> VCMFrameBuffer::~VCMFrameBuffer() {}
> 
>-VCMFrameBuffer::VCMFrameBuffer(const VCMFrameBuffer& rhs)
>-    : VCMEncodedFrame(rhs),
>-      _state(rhs._state),
>-      _sessionInfo(),
>-      _nackCount(rhs._nackCount),
>-      _latestPacketTimeMs(rhs._latestPacketTimeMs) {
>-  _sessionInfo = rhs._sessionInfo;
>-  _sessionInfo.UpdateDataPointers(rhs._buffer, _buffer);
>-}
>-
> webrtc::FrameType VCMFrameBuffer::FrameType() const {
>   return _sessionInfo.FrameType();
> }
>@@ -63,10 +53,6 @@ int VCMFrameBuffer::Tl0PicId() const {
>   return _sessionInfo.Tl0PicId();
> }
> 
>-bool VCMFrameBuffer::NonReference() const {
>-  return _sessionInfo.NonReference();
>-}
>-
> std::vector<NaluInfo> VCMFrameBuffer::GetNaluInfos() const {
>   return _sessionInfo.GetNaluInfos();
> }
>@@ -164,7 +150,7 @@ VCMFrameBufferEnum VCMFrameBuffer::InsertPacket(
>     rotation_ = packet.video_header.rotation;
>     _rotation_set = true;
>     content_type_ = packet.video_header.content_type;
>-    if (packet.video_header.video_timing.flags != TimingFrameFlags::kInvalid) {
>+    if (packet.video_header.video_timing.flags != VideoSendTiming::kInvalid) {
>       timing_.encode_start_ms =
>           ntp_time_ms_ + packet.video_header.video_timing.encode_start_delta_ms;
>       timing_.encode_finish_ms =
>@@ -219,11 +205,6 @@ bool VCMFrameBuffer::HaveFirstPacket() const {
>   return _sessionInfo.HaveFirstPacket();
> }
> 
>-bool VCMFrameBuffer::HaveLastPacket() const {
>-  TRACE_EVENT0("webrtc", "VCMFrameBuffer::HaveLastPacket");
>-  return _sessionInfo.HaveLastPacket();
>-}
>-
> int VCMFrameBuffer::NumPackets() const {
>   TRACE_EVENT0("webrtc", "VCMFrameBuffer::NumPackets");
>   return _sessionInfo.NumPackets();
>@@ -278,17 +259,6 @@ VCMFrameBufferStateEnum VCMFrameBuffer::GetState() const {
>   return _state;
> }
> 
>-// Get current state of frame
>-VCMFrameBufferStateEnum VCMFrameBuffer::GetState(uint32_t& timeStamp) const {
>-  TRACE_EVENT0("webrtc", "VCMFrameBuffer::GetState");
>-  timeStamp = TimeStamp();
>-  return GetState();
>-}
>-
>-bool VCMFrameBuffer::IsRetransmitted() const {
>-  return _sessionInfo.session_nack();
>-}
>-
> void VCMFrameBuffer::PrepareForDecode(bool continuous) {
>   TRACE_EVENT0("webrtc", "VCMFrameBuffer::PrepareForDecode");
>   size_t bytes_removed = _sessionInfo.MakeDecodable();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/frame_buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/frame_buffer.h
>index 72c1197af25..0b8c220a60e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/frame_buffer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/frame_buffer.h
>@@ -14,11 +14,10 @@
> #include <vector>
> 
> #include "modules/include/module_common_types.h"
>-#include "modules/video_coding/include/video_coding.h"
> #include "modules/video_coding/encoded_frame.h"
>+#include "modules/video_coding/include/video_coding.h"
> #include "modules/video_coding/jitter_buffer_common.h"
> #include "modules/video_coding/session_info.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -27,8 +26,6 @@ class VCMFrameBuffer : public VCMEncodedFrame {
>   VCMFrameBuffer();
>   virtual ~VCMFrameBuffer();
> 
>-  VCMFrameBuffer(const VCMFrameBuffer& rhs);
>-
>   virtual void Reset();
> 
>   VCMFrameBufferEnum InsertPacket(const VCMPacket& packet,
>@@ -39,17 +36,11 @@ class VCMFrameBuffer : public VCMEncodedFrame {
>   // State
>   // Get current state of frame
>   VCMFrameBufferStateEnum GetState() const;
>-  // Get current state and timestamp of frame
>-  VCMFrameBufferStateEnum GetState(uint32_t& timeStamp) const;
>   void PrepareForDecode(bool continuous);
> 
>-  bool IsRetransmitted() const;
>   bool IsSessionComplete() const;
>   bool HaveFirstPacket() const;
>-  bool HaveLastPacket() const;
>   int NumPackets() const;
>-  // Makes sure the session contain a decodable stream.
>-  void MakeSessionDecodable();
> 
>   // Sequence numbers
>   // Get lowest packet sequence number in frame
>@@ -61,7 +52,6 @@ class VCMFrameBuffer : public VCMEncodedFrame {
>   int TemporalId() const;
>   bool LayerSync() const;
>   int Tl0PicId() const;
>-  bool NonReference() const;
> 
>   std::vector<NaluInfo> GetNaluInfos() const;
> 
>@@ -77,10 +67,6 @@ class VCMFrameBuffer : public VCMEncodedFrame {
>   int64_t LatestPacketTimeMs() const;
> 
>   webrtc::FrameType FrameType() const;
>-  void SetPreviousFrameLoss();
>-
>-  // The number of packets discarded because the decoder can't make use of them.
>-  int NotDecodablePackets() const;
> 
>  private:
>   void SetState(VCMFrameBufferStateEnum state);  // Set state of frame
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/frame_buffer2.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/frame_buffer2.cc
>index 87fc498a81b..f13ed35aef3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/frame_buffer2.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/frame_buffer2.cc
>@@ -13,6 +13,7 @@
> #include <algorithm>
> #include <cstring>
> #include <queue>
>+#include <vector>
> 
> #include "modules/video_coding/include/video_coding_defines.h"
> #include "modules/video_coding/jitter_estimator.h"
>@@ -22,7 +23,6 @@
> #include "rtc_base/trace_event.h"
> #include "system_wrappers/include/clock.h"
> #include "system_wrappers/include/field_trial.h"
>-#include "system_wrappers/include/metrics.h"
> 
> namespace webrtc {
> namespace video_coding {
>@@ -34,6 +34,10 @@ constexpr int kMaxFramesBuffered = 600;
> // Max number of decoded frame info that will be saved.
> constexpr int kMaxFramesHistory = 50;
> 
>+// The time it's allowed for a frame to be late to its rendering prediction and
>+// still be rendered.
>+constexpr int kMaxAllowedFrameDelayMs = 5;
>+
> constexpr int64_t kLogNonDecodedIntervalMs = 5000;
> }  // namespace
> 
>@@ -60,7 +64,7 @@ FrameBuffer::~FrameBuffer() {}
> 
> FrameBuffer::ReturnReason FrameBuffer::NextFrame(
>     int64_t max_wait_time_ms,
>-    std::unique_ptr<FrameObject>* frame_out,
>+    std::unique_ptr<EncodedFrame>* frame_out,
>     bool keyframe_required) {
>   TRACE_EVENT0("webrtc", "FrameBuffer::NextFrame");
>   int64_t latest_return_time_ms =
>@@ -106,7 +110,7 @@ FrameBuffer::ReturnReason FrameBuffer::NextFrame(
>           continue;
>         }
> 
>-        FrameObject* frame = frame_it->second.frame.get();
>+        EncodedFrame* frame = frame_it->second.frame.get();
> 
>         if (keyframe_required && !frame->is_keyframe())
>           continue;
>@@ -119,7 +123,9 @@ FrameBuffer::ReturnReason FrameBuffer::NextFrame(
>         // This will cause the frame buffer to prefer high framerate rather
>         // than high resolution in the case of the decoder not decoding fast
>         // enough and the stream has multiple spatial and temporal layers.
>-        if (wait_ms == 0)
>+        // For multiple temporal layers it may cause non-base layer frames to be
>+        // skipped if they are late.
>+        if (wait_ms < -kMaxAllowedFrameDelayMs)
>           continue;
> 
>         break;
>@@ -134,7 +140,7 @@ FrameBuffer::ReturnReason FrameBuffer::NextFrame(
>     rtc::CritScope lock(&crit_);
>     now_ms = clock_->TimeInMilliseconds();
>     if (next_frame_it_ != frames_.end()) {
>-      std::unique_ptr<FrameObject> frame =
>+      std::unique_ptr<EncodedFrame> frame =
>           std::move(next_frame_it_->second.frame);
> 
>       if (!frame->delayed_by_retransmission()) {
>@@ -166,8 +172,9 @@ FrameBuffer::ReturnReason FrameBuffer::NextFrame(
> 
>       // Sanity check for RTP timestamp monotonicity.
>       if (last_decoded_frame_it_ != frames_.end()) {
>-        const FrameKey& last_decoded_frame_key = last_decoded_frame_it_->first;
>-        const FrameKey& frame_key = next_frame_it_->first;
>+        const VideoLayerFrameId& last_decoded_frame_key =
>+            last_decoded_frame_it_->first;
>+        const VideoLayerFrameId& frame_key = next_frame_it_->first;
> 
>         const bool frame_is_higher_spatial_layer_of_last_decoded_frame =
>             last_decoded_frame_timestamp_ == frame->timestamp &&
>@@ -180,8 +187,8 @@ FrameBuffer::ReturnReason FrameBuffer::NextFrame(
>           // these conditions.
>           RTC_LOG(LS_WARNING)
>               << "Frame with (timestamp:picture_id:spatial_id) ("
>-              << frame->timestamp << ":" << frame->picture_id << ":"
>-              << static_cast<int>(frame->spatial_layer) << ")"
>+              << frame->timestamp << ":" << frame->id.picture_id << ":"
>+              << static_cast<int>(frame->id.spatial_layer) << ")"
>               << " sent to decoder after frame with"
>               << " (timestamp:picture_id:spatial_id) ("
>               << last_decoded_frame_timestamp_ << ":"
>@@ -208,13 +215,18 @@ FrameBuffer::ReturnReason FrameBuffer::NextFrame(
>   return kTimeout;
> }
> 
>-bool FrameBuffer::HasBadRenderTiming(const FrameObject& frame, int64_t now_ms) {
>+bool FrameBuffer::HasBadRenderTiming(const EncodedFrame& frame,
>+                                     int64_t now_ms) {
>   // Assume that render timing errors are due to changes in the video stream.
>   int64_t render_time_ms = frame.RenderTimeMs();
>-  const int64_t kMaxVideoDelayMs = 10000;
>+  // Zero render time means render immediately.
>+  if (render_time_ms == 0) {
>+    return false;
>+  }
>   if (render_time_ms < 0) {
>     return true;
>   }
>+  const int64_t kMaxVideoDelayMs = 10000;
>   if (std::abs(render_time_ms - now_ms) > kMaxVideoDelayMs) {
>     int frame_delay = static_cast<int>(std::abs(render_time_ms - now_ms));
>     RTC_LOG(LS_WARNING)
>@@ -255,12 +267,12 @@ void FrameBuffer::UpdateRtt(int64_t rtt_ms) {
>   jitter_estimator_->UpdateRtt(rtt_ms);
> }
> 
>-bool FrameBuffer::ValidReferences(const FrameObject& frame) const {
>-  if (frame.picture_id < 0)
>+bool FrameBuffer::ValidReferences(const EncodedFrame& frame) const {
>+  if (frame.id.picture_id < 0)
>     return false;
> 
>   for (size_t i = 0; i < frame.num_references; ++i) {
>-    if (frame.references[i] < 0 || frame.references[i] >= frame.picture_id)
>+    if (frame.references[i] < 0 || frame.references[i] >= frame.id.picture_id)
>       return false;
> 
>     for (size_t j = i + 1; j < frame.num_references; ++j) {
>@@ -269,13 +281,13 @@ bool FrameBuffer::ValidReferences(const FrameObject& frame) const {
>     }
>   }
> 
>-  if (frame.inter_layer_predicted && frame.spatial_layer == 0)
>+  if (frame.inter_layer_predicted && frame.id.spatial_layer == 0)
>     return false;
> 
>   return true;
> }
> 
>-void FrameBuffer::UpdatePlayoutDelays(const FrameObject& frame) {
>+void FrameBuffer::UpdatePlayoutDelays(const EncodedFrame& frame) {
>   TRACE_EVENT0("webrtc", "FrameBuffer::UpdatePlayoutDelays");
>   PlayoutDelay playout_delay = frame.EncodedImage().playout_delay_;
>   if (playout_delay.min_ms >= 0)
>@@ -283,15 +295,18 @@ void FrameBuffer::UpdatePlayoutDelays(const FrameObject& frame) {
> 
>   if (playout_delay.max_ms >= 0)
>     timing_->set_max_playout_delay(playout_delay.max_ms);
>+
>+  if (!frame.delayed_by_retransmission())
>+    timing_->IncomingTimestamp(frame.timestamp, frame.ReceivedTime());
> }
> 
>-int64_t FrameBuffer::InsertFrame(std::unique_ptr<FrameObject> frame) {
>+int64_t FrameBuffer::InsertFrame(std::unique_ptr<EncodedFrame> frame) {
>   TRACE_EVENT0("webrtc", "FrameBuffer::InsertFrame");
>   RTC_DCHECK(frame);
>   if (stats_callback_)
>     stats_callback_->OnCompleteFrame(frame->is_keyframe(), frame->size(),
>                                      frame->contentType());
>-  FrameKey key(frame->picture_id, frame->spatial_layer);
>+  const VideoLayerFrameId& id = frame->id;
> 
>   rtc::CritScope lock(&crit_);
> 
>@@ -302,23 +317,32 @@ int64_t FrameBuffer::InsertFrame(std::unique_ptr<FrameObject> frame) {
> 
>   if (!ValidReferences(*frame)) {
>     RTC_LOG(LS_WARNING) << "Frame with (picture_id:spatial_id) ("
>-                        << key.picture_id << ":"
>-                        << static_cast<int>(key.spatial_layer)
>+                        << id.picture_id << ":"
>+                        << static_cast<int>(id.spatial_layer)
>                         << ") has invalid frame references, dropping frame.";
>     return last_continuous_picture_id;
>   }
> 
>   if (num_frames_buffered_ >= kMaxFramesBuffered) {
>-    RTC_LOG(LS_WARNING) << "Frame with (picture_id:spatial_id) ("
>-                        << key.picture_id << ":"
>-                        << static_cast<int>(key.spatial_layer)
>-                        << ") could not be inserted due to the frame "
>-                        << "buffer being full, dropping frame.";
>-    return last_continuous_picture_id;
>+    if (frame->is_keyframe()) {
>+      RTC_LOG(LS_WARNING) << "Inserting keyframe (picture_id:spatial_id) ("
>+                          << id.picture_id << ":"
>+                          << static_cast<int>(id.spatial_layer)
>+                          << ") but buffer is full, clearing"
>+                          << " buffer and inserting the frame.";
>+      ClearFramesAndHistory();
>+    } else {
>+      RTC_LOG(LS_WARNING) << "Frame with (picture_id:spatial_id) ("
>+                          << id.picture_id << ":"
>+                          << static_cast<int>(id.spatial_layer)
>+                          << ") could not be inserted due to the frame "
>+                          << "buffer being full, dropping frame.";
>+      return last_continuous_picture_id;
>+    }
>   }
> 
>   if (last_decoded_frame_it_ != frames_.end() &&
>-      key <= last_decoded_frame_it_->first) {
>+      id <= last_decoded_frame_it_->first) {
>     if (AheadOf(frame->timestamp, last_decoded_frame_timestamp_) &&
>         frame->is_keyframe()) {
>       // If this frame has a newer timestamp but an earlier picture id then we
>@@ -332,8 +356,8 @@ int64_t FrameBuffer::InsertFrame(std::unique_ptr<FrameObject> frame) {
>       last_continuous_picture_id = -1;
>     } else {
>       RTC_LOG(LS_WARNING) << "Frame with (picture_id:spatial_id) ("
>-                          << key.picture_id << ":"
>-                          << static_cast<int>(key.spatial_layer)
>+                          << id.picture_id << ":"
>+                          << static_cast<int>(id.spatial_layer)
>                           << ") inserted after frame ("
>                           << last_decoded_frame_it_->first.picture_id << ":"
>                           << static_cast<int>(
>@@ -346,21 +370,20 @@ int64_t FrameBuffer::InsertFrame(std::unique_ptr<FrameObject> frame) {
>   // Test if inserting this frame would cause the order of the frames to become
>   // ambiguous (covering more than half the interval of 2^16). This can happen
>   // when the picture id make large jumps mid stream.
>-  if (!frames_.empty() &&
>-      key < frames_.begin()->first &&
>-      frames_.rbegin()->first < key) {
>+  if (!frames_.empty() && id < frames_.begin()->first &&
>+      frames_.rbegin()->first < id) {
>     RTC_LOG(LS_WARNING)
>         << "A jump in picture id was detected, clearing buffer.";
>     ClearFramesAndHistory();
>     last_continuous_picture_id = -1;
>   }
> 
>-  auto info = frames_.insert(std::make_pair(key, FrameInfo())).first;
>+  auto info = frames_.emplace(id, FrameInfo()).first;
> 
>   if (info->second.frame) {
>     RTC_LOG(LS_WARNING) << "Frame with (picture_id:spatial_id) ("
>-                        << key.picture_id << ":"
>-                        << static_cast<int>(key.spatial_layer)
>+                        << id.picture_id << ":"
>+                        << static_cast<int>(id.spatial_layer)
>                         << ") already inserted, dropping frame.";
>     return last_continuous_picture_id;
>   }
>@@ -368,6 +391,7 @@ int64_t FrameBuffer::InsertFrame(std::unique_ptr<FrameObject> frame) {
>   if (!UpdateFrameInfoWithIncomingFrame(*frame, info))
>     return last_continuous_picture_id;
>   UpdatePlayoutDelays(*frame);
>+
>   info->second.frame = std::move(frame);
>   ++num_frames_buffered_;
> 
>@@ -459,88 +483,96 @@ void FrameBuffer::AdvanceLastDecodedFrame(FrameMap::iterator decoded) {
>   }
> }
> 
>-bool FrameBuffer::UpdateFrameInfoWithIncomingFrame(const FrameObject& frame,
>+bool FrameBuffer::UpdateFrameInfoWithIncomingFrame(const EncodedFrame& frame,
>                                                    FrameMap::iterator info) {
>   TRACE_EVENT0("webrtc", "FrameBuffer::UpdateFrameInfoWithIncomingFrame");
>-  FrameKey key(frame.picture_id, frame.spatial_layer);
>-  info->second.num_missing_continuous = frame.num_references;
>-  info->second.num_missing_decodable = frame.num_references;
>+  const VideoLayerFrameId& id = frame.id;
> 
>   RTC_DCHECK(last_decoded_frame_it_ == frames_.end() ||
>              last_decoded_frame_it_->first < info->first);
> 
>-  // Check how many dependencies that have already been fulfilled.
>+  // In this function we determine how many missing dependencies this |frame|
>+  // has to become continuous/decodable. If a frame that this |frame| depend
>+  // on has already been decoded then we can ignore that dependency since it has
>+  // already been fulfilled.
>+  //
>+  // For all other frames we will register a backwards reference to this |frame|
>+  // so that |num_missing_continuous| and |num_missing_decodable| can be
>+  // decremented as frames become continuous/are decoded.
>+  struct Dependency {
>+    VideoLayerFrameId id;
>+    bool continuous;
>+  };
>+  std::vector<Dependency> not_yet_fulfilled_dependencies;
>+
>+  // Find all dependencies that have not yet been fulfilled.
>   for (size_t i = 0; i < frame.num_references; ++i) {
>-    FrameKey ref_key(frame.references[i], frame.spatial_layer);
>+    VideoLayerFrameId ref_key(frame.references[i], frame.id.spatial_layer);
>     auto ref_info = frames_.find(ref_key);
> 
>-    // Does |frame| depend on a frame earlier than the last decoded frame?
>+    // Does |frame| depend on a frame earlier than the last decoded one?
>     if (last_decoded_frame_it_ != frames_.end() &&
>         ref_key <= last_decoded_frame_it_->first) {
>+      // Was that frame decoded? If not, this |frame| will never become
>+      // decodable.
>       if (ref_info == frames_.end()) {
>         int64_t now_ms = clock_->TimeInMilliseconds();
>         if (last_log_non_decoded_ms_ + kLogNonDecodedIntervalMs < now_ms) {
>           RTC_LOG(LS_WARNING)
>-              << "Frame with (picture_id:spatial_id) (" << key.picture_id << ":"
>-              << static_cast<int>(key.spatial_layer)
>+              << "Frame with (picture_id:spatial_id) (" << id.picture_id << ":"
>+              << static_cast<int>(id.spatial_layer)
>               << ") depends on a non-decoded frame more previous than"
>               << " the last decoded frame, dropping frame.";
>           last_log_non_decoded_ms_ = now_ms;
>         }
>         return false;
>       }
>-
>-      --info->second.num_missing_continuous;
>-      --info->second.num_missing_decodable;
>     } else {
>-      if (ref_info == frames_.end())
>-        ref_info = frames_.insert(std::make_pair(ref_key, FrameInfo())).first;
>-
>-      if (ref_info->second.continuous)
>-        --info->second.num_missing_continuous;
>-
>-      // Add backwards reference so |frame| can be updated when new
>-      // frames are inserted or decoded.
>-      ref_info->second.dependent_frames[ref_info->second.num_dependent_frames] =
>-          key;
>-      RTC_DCHECK_LT(ref_info->second.num_dependent_frames,
>-                    (FrameInfo::kMaxNumDependentFrames - 1));
>-      // TODO(philipel): Look into why this could happen and handle
>-      // appropriately.
>-      if (ref_info->second.num_dependent_frames <
>-          (FrameInfo::kMaxNumDependentFrames - 1)) {
>-        ++ref_info->second.num_dependent_frames;
>-      }
>+      bool ref_continuous =
>+          ref_info != frames_.end() && ref_info->second.continuous;
>+      not_yet_fulfilled_dependencies.push_back({ref_key, ref_continuous});
>     }
>-    RTC_DCHECK_LE(ref_info->second.num_missing_continuous,
>-                  ref_info->second.num_missing_decodable);
>   }
> 
>-  // Check if we have the lower spatial layer frame.
>+  // Does |frame| depend on the lower spatial layer?
>   if (frame.inter_layer_predicted) {
>-    ++info->second.num_missing_continuous;
>-    ++info->second.num_missing_decodable;
>+    VideoLayerFrameId ref_key(frame.id.picture_id, frame.id.spatial_layer - 1);
>+    auto ref_info = frames_.find(ref_key);
> 
>-    FrameKey ref_key(frame.picture_id, frame.spatial_layer - 1);
>-    // Gets or create the FrameInfo for the referenced frame.
>-    auto ref_info = frames_.insert(std::make_pair(ref_key, FrameInfo())).first;
>-    if (ref_info->second.continuous)
>+    bool lower_layer_continuous =
>+        ref_info != frames_.end() && ref_info->second.continuous;
>+    bool lower_layer_decoded = last_decoded_frame_it_ != frames_.end() &&
>+                               last_decoded_frame_it_->first == ref_key;
>+
>+    if (!lower_layer_continuous || !lower_layer_decoded) {
>+      not_yet_fulfilled_dependencies.push_back(
>+          {ref_key, lower_layer_continuous});
>+    }
>+  }
>+
>+  info->second.num_missing_continuous = not_yet_fulfilled_dependencies.size();
>+  info->second.num_missing_decodable = not_yet_fulfilled_dependencies.size();
>+
>+  for (const Dependency& dep : not_yet_fulfilled_dependencies) {
>+    if (dep.continuous)
>       --info->second.num_missing_continuous;
> 
>-    if (ref_info == last_decoded_frame_it_) {
>-      --info->second.num_missing_decodable;
>+    // At this point we know we want to insert this frame, so here we
>+    // intentionally get or create the FrameInfo for this dependency.
>+    FrameInfo* dep_info = &frames_[dep.id];
>+
>+    if (dep_info->num_dependent_frames <
>+        (FrameInfo::kMaxNumDependentFrames - 1)) {
>+      dep_info->dependent_frames[dep_info->num_dependent_frames] = id;
>+      ++dep_info->num_dependent_frames;
>     } else {
>-      ref_info->second.dependent_frames[ref_info->second.num_dependent_frames] =
>-          key;
>-      ++ref_info->second.num_dependent_frames;
>+      RTC_LOG(LS_WARNING) << "Frame with (picture_id:spatial_id) ("
>+                          << dep.id.picture_id << ":"
>+                          << static_cast<int>(dep.id.spatial_layer)
>+                          << ") is referenced by too many frames.";
>     }
>-    RTC_DCHECK_LE(ref_info->second.num_missing_continuous,
>-                  ref_info->second.num_missing_decodable);
>   }
> 
>-  RTC_DCHECK_LE(info->second.num_missing_continuous,
>-                info->second.num_missing_decodable);
>-
>   return true;
> }
> 
>@@ -567,8 +599,8 @@ void FrameBuffer::UpdateJitterDelay() {
> 
> void FrameBuffer::UpdateTimingFrameInfo() {
>   TRACE_EVENT0("webrtc", "FrameBuffer::UpdateTimingFrameInfo");
>-  rtc::Optional<TimingFrameInfo> info = timing_->GetTimingFrameInfo();
>-  if (info)
>+  absl::optional<TimingFrameInfo> info = timing_->GetTimingFrameInfo();
>+  if (info && stats_callback_)
>     stats_callback_->OnTimingFrameInfoUpdated(*info);
> }
> 
>@@ -582,5 +614,9 @@ void FrameBuffer::ClearFramesAndHistory() {
>   num_frames_buffered_ = 0;
> }
> 
>+FrameBuffer::FrameInfo::FrameInfo() = default;
>+FrameBuffer::FrameInfo::FrameInfo(FrameInfo&&) = default;
>+FrameBuffer::FrameInfo::~FrameInfo() = default;
>+
> }  // namespace video_coding
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/frame_buffer2.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/frame_buffer2.h
>index 19fd94ec5cd..bb57b83feb3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/frame_buffer2.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/frame_buffer2.h
>@@ -16,7 +16,7 @@
> #include <memory>
> #include <utility>
> 
>-#include "modules/video_coding/frame_object.h"
>+#include "api/video/encoded_frame.h"
> #include "modules/video_coding/include/video_coding_defines.h"
> #include "modules/video_coding/inter_frame_delay.h"
> #include "rtc_base/constructormagic.h"
>@@ -47,7 +47,8 @@ class FrameBuffer {
> 
>   // Insert a frame into the frame buffer. Returns the picture id
>   // of the last continuous frame or -1 if there is no continuous frame.
>-  int64_t InsertFrame(std::unique_ptr<FrameObject> frame);
>+  // TODO(philipel): Return a VideoLayerFrameId and not only the picture id.
>+  int64_t InsertFrame(std::unique_ptr<EncodedFrame> frame);
> 
>   // Get the next frame for decoding. Will return at latest after
>   // |max_wait_time_ms|.
>@@ -57,7 +58,7 @@ class FrameBuffer {
>   //    kTimeout.
>   //  - If the FrameBuffer is stopped then it will return kStopped.
>   ReturnReason NextFrame(int64_t max_wait_time_ms,
>-                         std::unique_ptr<FrameObject>* frame_out,
>+                         std::unique_ptr<EncodedFrame>* frame_out,
>                          bool keyframe_required = false);
> 
>   // Tells the FrameBuffer which protection mode that is in use. Affects
>@@ -78,24 +79,11 @@ class FrameBuffer {
>   void UpdateRtt(int64_t rtt_ms);
> 
>  private:
>-  struct FrameKey {
>-    FrameKey() : picture_id(-1), spatial_layer(0) {}
>-    FrameKey(int64_t picture_id, uint8_t spatial_layer)
>-        : picture_id(picture_id), spatial_layer(spatial_layer) {}
>-
>-    bool operator<(const FrameKey& rhs) const {
>-      if (picture_id == rhs.picture_id)
>-        return spatial_layer < rhs.spatial_layer;
>-      return picture_id < rhs.picture_id;
>-    }
>-
>-    bool operator<=(const FrameKey& rhs) const { return !(rhs < *this); }
>-
>-    int64_t picture_id;
>-    uint8_t spatial_layer;
>-  };
>-
>   struct FrameInfo {
>+    FrameInfo();
>+    FrameInfo(FrameInfo&&);
>+    ~FrameInfo();
>+
>     // The maximum number of frames that can depend on this frame.
>     static constexpr size_t kMaxNumDependentFrames = 8;
> 
>@@ -103,7 +91,7 @@ class FrameBuffer {
>     // on this frame.
>     // TODO(philipel): Add simple modify/access functions to prevent adding too
>     // many |dependent_frames|.
>-    FrameKey dependent_frames[kMaxNumDependentFrames];
>+    VideoLayerFrameId dependent_frames[kMaxNumDependentFrames];
>     size_t num_dependent_frames = 0;
> 
>     // A frame is continiuous if it has all its referenced/indirectly
>@@ -120,18 +108,18 @@ class FrameBuffer {
>     // If this frame is continuous or not.
>     bool continuous = false;
> 
>-    // The actual FrameObject.
>-    std::unique_ptr<FrameObject> frame;
>+    // The actual EncodedFrame.
>+    std::unique_ptr<EncodedFrame> frame;
>   };
> 
>-  using FrameMap = std::map<FrameKey, FrameInfo>;
>+  using FrameMap = std::map<VideoLayerFrameId, FrameInfo>;
> 
>   // Check that the references of |frame| are valid.
>-  bool ValidReferences(const FrameObject& frame) const;
>+  bool ValidReferences(const EncodedFrame& frame) const;
> 
>   // Updates the minimal and maximal playout delays
>   // depending on the frame.
>-  void UpdatePlayoutDelays(const FrameObject& frame)
>+  void UpdatePlayoutDelays(const EncodedFrame& frame)
>       RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
> 
>   // Update all directly dependent and indirectly dependent frames and mark
>@@ -151,7 +139,7 @@ class FrameBuffer {
>   // Update the corresponding FrameInfo of |frame| and all FrameInfos that
>   // |frame| references.
>   // Return false if |frame| will never be decodable, true otherwise.
>-  bool UpdateFrameInfoWithIncomingFrame(const FrameObject& frame,
>+  bool UpdateFrameInfoWithIncomingFrame(const EncodedFrame& frame,
>                                         FrameMap::iterator info)
>       RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
> 
>@@ -161,7 +149,7 @@ class FrameBuffer {
> 
>   void ClearFramesAndHistory() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
> 
>-  bool HasBadRenderTiming(const FrameObject& frame, int64_t now_ms)
>+  bool HasBadRenderTiming(const EncodedFrame& frame, int64_t now_ms)
>       RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
> 
>   FrameMap frames_ RTC_GUARDED_BY(crit_);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/frame_buffer2_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/frame_buffer2_unittest.cc
>index 9cdd81bd3ac..1378be5a842 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/frame_buffer2_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/frame_buffer2_unittest.cc
>@@ -52,9 +52,9 @@ class VCMTimingFake : public VCMTiming {
>     return last_ms_;
>   }
> 
>-  uint32_t MaxWaitingTime(int64_t render_time_ms,
>-                          int64_t now_ms) const override {
>-    return std::max<int>(0, render_time_ms - now_ms - kDecodeTime);
>+  int64_t MaxWaitingTime(int64_t render_time_ms,
>+                         int64_t now_ms) const override {
>+    return render_time_ms - now_ms - kDecodeTime;
>   }
> 
>   bool GetTimings(int* decode_ms,
>@@ -86,7 +86,7 @@ class VCMJitterEstimatorMock : public VCMJitterEstimator {
>   MOCK_METHOD1(GetJitterEstimate, int(double rttMultiplier));
> };
> 
>-class FrameObjectFake : public FrameObject {
>+class FrameObjectFake : public EncodedFrame {
>  public:
>   bool GetBitstream(uint8_t* destination) const override { return true; }
> 
>@@ -133,7 +133,10 @@ class TestFrameBuffer2 : public ::testing::Test {
>       : clock_(0),
>         timing_(&clock_),
>         jitter_estimator_(&clock_),
>-        buffer_(&clock_, &jitter_estimator_, &timing_, &stats_callback_),
>+        buffer_(new FrameBuffer(&clock_,
>+                                &jitter_estimator_,
>+                                &timing_,
>+                                &stats_callback_)),
>         rand_(0x34678213),
>         tear_down_(false),
>         extract_thread_(&ExtractLoop, this, "Extract Thread"),
>@@ -155,28 +158,28 @@ class TestFrameBuffer2 : public ::testing::Test {
>                   bool inter_layer_predicted,
>                   T... refs) {
>     static_assert(sizeof...(refs) <= kMaxReferences,
>-                  "To many references specified for FrameObject.");
>+                  "To many references specified for EncodedFrame.");
>     std::array<uint16_t, sizeof...(refs)> references = {
>         {rtc::checked_cast<uint16_t>(refs)...}};
> 
>     std::unique_ptr<FrameObjectFake> frame(new FrameObjectFake());
>-    frame->picture_id = picture_id;
>-    frame->spatial_layer = spatial_layer;
>+    frame->id.picture_id = picture_id;
>+    frame->id.spatial_layer = spatial_layer;
>     frame->timestamp = ts_ms * 90;
>     frame->num_references = references.size();
>     frame->inter_layer_predicted = inter_layer_predicted;
>     for (size_t r = 0; r < references.size(); ++r)
>       frame->references[r] = references[r];
> 
>-    return buffer_.InsertFrame(std::move(frame));
>+    return buffer_->InsertFrame(std::move(frame));
>   }
> 
>   void ExtractFrame(int64_t max_wait_time = 0, bool keyframe_required = false) {
>     crit_.Enter();
>     if (max_wait_time == 0) {
>-      std::unique_ptr<FrameObject> frame;
>+      std::unique_ptr<EncodedFrame> frame;
>       FrameBuffer::ReturnReason res =
>-          buffer_.NextFrame(0, &frame, keyframe_required);
>+          buffer_->NextFrame(0, &frame, keyframe_required);
>       if (res != FrameBuffer::ReturnReason::kStopped)
>         frames_.emplace_back(std::move(frame));
>       crit_.Leave();
>@@ -193,8 +196,8 @@ class TestFrameBuffer2 : public ::testing::Test {
>     rtc::CritScope lock(&crit_);
>     ASSERT_LT(index, frames_.size());
>     ASSERT_TRUE(frames_[index]);
>-    ASSERT_EQ(picture_id, frames_[index]->picture_id);
>-    ASSERT_EQ(spatial_layer, frames_[index]->spatial_layer);
>+    ASSERT_EQ(picture_id, frames_[index]->id.picture_id);
>+    ASSERT_EQ(spatial_layer, frames_[index]->id.spatial_layer);
>   }
> 
>   void CheckNoFrame(size_t index) {
>@@ -213,9 +216,9 @@ class TestFrameBuffer2 : public ::testing::Test {
>         if (tfb->tear_down_)
>           return;
> 
>-        std::unique_ptr<FrameObject> frame;
>+        std::unique_ptr<EncodedFrame> frame;
>         FrameBuffer::ReturnReason res =
>-            tfb->buffer_.NextFrame(tfb->max_wait_time_, &frame);
>+            tfb->buffer_->NextFrame(tfb->max_wait_time_, &frame);
>         if (res != FrameBuffer::ReturnReason::kStopped)
>           tfb->frames_.emplace_back(std::move(frame));
>       }
>@@ -227,8 +230,8 @@ class TestFrameBuffer2 : public ::testing::Test {
>   SimulatedClock clock_;
>   VCMTimingFake timing_;
>   ::testing::NiceMock<VCMJitterEstimatorMock> jitter_estimator_;
>-  FrameBuffer buffer_;
>-  std::vector<std::unique_ptr<FrameObject>> frames_;
>+  std::unique_ptr<FrameBuffer> buffer_;
>+  std::vector<std::unique_ptr<EncodedFrame>> frames_;
>   Random rand_;
>   ::testing::NiceMock<VCMReceiveStatisticsCallbackMock> stats_callback_;
> 
>@@ -269,12 +272,27 @@ TEST_F(TestFrameBuffer2, OneSuperFrame) {
> TEST_F(TestFrameBuffer2, SetPlayoutDelay) {
>   const PlayoutDelay kPlayoutDelayMs = {123, 321};
>   std::unique_ptr<FrameObjectFake> test_frame(new FrameObjectFake());
>+  test_frame->id.picture_id = 0;
>   test_frame->SetPlayoutDelay(kPlayoutDelayMs);
>-  buffer_.InsertFrame(std::move(test_frame));
>+  buffer_->InsertFrame(std::move(test_frame));
>   EXPECT_EQ(kPlayoutDelayMs.min_ms, timing_.min_playout_delay());
>   EXPECT_EQ(kPlayoutDelayMs.max_ms, timing_.max_playout_delay());
> }
> 
>+TEST_F(TestFrameBuffer2, ZeroPlayoutDelay) {
>+  VCMTiming timing(&clock_);
>+  buffer_.reset(
>+      new FrameBuffer(&clock_, &jitter_estimator_, &timing, &stats_callback_));
>+  const PlayoutDelay kPlayoutDelayMs = {0, 0};
>+  std::unique_ptr<FrameObjectFake> test_frame(new FrameObjectFake());
>+  test_frame->id.picture_id = 0;
>+  test_frame->SetPlayoutDelay(kPlayoutDelayMs);
>+  buffer_->InsertFrame(std::move(test_frame));
>+  ExtractFrame(0, false);
>+  CheckFrame(0, 0, 0);
>+  EXPECT_EQ(0, frames_[0]->RenderTimeMs());
>+}
>+
> // Flaky test, see bugs.webrtc.org/7068.
> TEST_F(TestFrameBuffer2, DISABLED_OneUnorderedSuperFrame) {
>   uint16_t pid = Rand();
>@@ -359,7 +377,7 @@ TEST_F(TestFrameBuffer2, DropTemporalLayerSlowDecoder) {
> 
>   for (int i = 0; i < 10; ++i) {
>     ExtractFrame();
>-    clock_.AdvanceTimeMilliseconds(60);
>+    clock_.AdvanceTimeMilliseconds(70);
>   }
> 
>   CheckFrame(0, pid, 0);
>@@ -388,10 +406,10 @@ TEST_F(TestFrameBuffer2, DropSpatialLayerSlowDecoder) {
> 
>   ExtractFrame();
>   ExtractFrame();
>-  clock_.AdvanceTimeMilliseconds(55);
>+  clock_.AdvanceTimeMilliseconds(57);
>   for (int i = 2; i < 12; ++i) {
>     ExtractFrame();
>-    clock_.AdvanceTimeMilliseconds(55);
>+    clock_.AdvanceTimeMilliseconds(57);
>   }
> 
>   CheckFrame(0, pid, 0);
>@@ -432,7 +450,7 @@ TEST_F(TestFrameBuffer2, ProtectionMode) {
>   InsertFrame(pid, 0, ts, false);
>   ExtractFrame();
> 
>-  buffer_.SetProtectionMode(kProtectionNackFEC);
>+  buffer_->SetProtectionMode(kProtectionNackFEC);
>   EXPECT_CALL(jitter_estimator_, GetJitterEstimate(0.0));
>   InsertFrame(pid + 1, 0, ts, false);
>   ExtractFrame();
>@@ -500,13 +518,13 @@ TEST_F(TestFrameBuffer2, StatsCallback) {
>   {
>     std::unique_ptr<FrameObjectFake> frame(new FrameObjectFake());
>     frame->SetSize(kFrameSize);
>-    frame->picture_id = pid;
>-    frame->spatial_layer = 0;
>+    frame->id.picture_id = pid;
>+    frame->id.spatial_layer = 0;
>     frame->timestamp = ts;
>     frame->num_references = 0;
>     frame->inter_layer_predicted = false;
> 
>-    EXPECT_EQ(buffer_.InsertFrame(std::move(frame)), pid);
>+    EXPECT_EQ(buffer_->InsertFrame(std::move(frame)), pid);
>   }
> 
>   ExtractFrame();
>@@ -559,5 +577,30 @@ TEST_F(TestFrameBuffer2, KeyframeRequired) {
>   CheckNoFrame(2);
> }
> 
>+TEST_F(TestFrameBuffer2, KeyframeClearsFullBuffer) {
>+  const int kMaxBufferSize = 600;
>+
>+  for (int i = 1; i <= kMaxBufferSize; ++i)
>+    EXPECT_EQ(-1, InsertFrame(i, 0, i * 1000, false, i - 1));
>+  ExtractFrame();
>+  CheckNoFrame(0);
>+
>+  EXPECT_EQ(
>+      kMaxBufferSize + 1,
>+      InsertFrame(kMaxBufferSize + 1, 0, (kMaxBufferSize + 1) * 1000, false));
>+  ExtractFrame();
>+  CheckFrame(1, kMaxBufferSize + 1, 0);
>+}
>+
>+TEST_F(TestFrameBuffer2, DontUpdateOnUndecodableFrame) {
>+  InsertFrame(1, 0, 0, false);
>+  ExtractFrame(0, true);
>+  InsertFrame(3, 0, 0, false, 2, 0);
>+  InsertFrame(3, 0, 0, false, 0);
>+  InsertFrame(2, 0, 0, false);
>+  ExtractFrame(0, true);
>+  ExtractFrame(0, true);
>+}
>+
> }  // namespace video_coding
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/frame_object.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/frame_object.cc
>index 6eb28de1cd7..b990cdd78b6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/frame_object.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/frame_object.cc
>@@ -17,13 +17,6 @@
> namespace webrtc {
> namespace video_coding {
> 
>-FrameObject::FrameObject()
>-    : picture_id(0),
>-      spatial_layer(0),
>-      timestamp(0),
>-      num_references(0),
>-      inter_layer_predicted(false) {}
>-
> RtpFrameObject::RtpFrameObject(PacketBuffer* packet_buffer,
>                                uint16_t first_seq_num,
>                                uint16_t last_seq_num,
>@@ -39,18 +32,13 @@ RtpFrameObject::RtpFrameObject(PacketBuffer* packet_buffer,
>   VCMPacket* first_packet = packet_buffer_->GetPacket(first_seq_num);
>   RTC_CHECK(first_packet);
> 
>-  // RtpFrameObject members
>+  // EncodedFrame members
>   frame_type_ = first_packet->frameType;
>   codec_type_ = first_packet->codec;
> 
>-  // Stereo codec appends CopyCodecSpecific to last packet to avoid copy.
>-  VCMPacket* packet_with_codec_specific =
>-      codec_type_ == kVideoCodecStereo ? packet_buffer_->GetPacket(last_seq_num)
>-                                       : first_packet;
>-
>-  // TODO(philipel): Remove when encoded image is replaced by FrameObject.
>+  // TODO(philipel): Remove when encoded image is replaced by EncodedFrame.
>   // VCMEncodedFrame members
>-  CopyCodecSpecific(&packet_with_codec_specific->video_header);
>+  CopyCodecSpecific(&first_packet->video_header);
>   _completeFrame = true;
>   _payloadType = first_packet->payloadType;
>   _timeStamp = first_packet->timestamp;
>@@ -80,7 +68,7 @@ RtpFrameObject::RtpFrameObject(PacketBuffer* packet_buffer,
>   _encodedWidth = first_packet->width;
>   _encodedHeight = first_packet->height;
> 
>-  // FrameObject members
>+  // EncodedFrame members
>   timestamp = first_packet->timestamp;
> 
>   VCMPacket* last_packet = packet_buffer_->GetPacket(last_seq_num);
>@@ -96,7 +84,7 @@ RtpFrameObject::RtpFrameObject(PacketBuffer* packet_buffer,
>   _rotation_set = true;
>   content_type_ = last_packet->video_header.content_type;
>   if (last_packet->video_header.video_timing.flags !=
>-      TimingFrameFlags::kInvalid) {
>+      VideoSendTiming::kInvalid) {
>     // ntp_time_ms_ may be -1 if not estimated yet. This is not a problem,
>     // as this will be dealt with at the time of reporting.
>     timing_.encode_start_ms =
>@@ -168,12 +156,12 @@ bool RtpFrameObject::delayed_by_retransmission() const {
>   return times_nacked() > 0;
> }
> 
>-rtc::Optional<RTPVideoTypeHeader> RtpFrameObject::GetCodecHeader() const {
>+absl::optional<RTPVideoTypeHeader> RtpFrameObject::GetCodecHeader() const {
>   rtc::CritScope lock(&packet_buffer_->crit_);
>   VCMPacket* packet = packet_buffer_->GetPacket(first_seq_num_);
>   if (!packet)
>-    return rtc::nullopt;
>-  return packet->video_header.codecHeader;
>+    return absl::nullopt;
>+  return packet->video_header.video_type_header;
> }
> 
> }  // namespace video_coding
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/frame_object.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/frame_object.h
>index 6f31c1af712..898098451dc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/frame_object.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/frame_object.h
>@@ -11,58 +11,17 @@
> #ifndef MODULES_VIDEO_CODING_FRAME_OBJECT_H_
> #define MODULES_VIDEO_CODING_FRAME_OBJECT_H_
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
>+#include "api/video/encoded_frame.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/include/module_common_types.h"
>-#include "modules/video_coding/encoded_frame.h"
> 
> namespace webrtc {
> namespace video_coding {
> 
>-class FrameObject : public webrtc::VCMEncodedFrame {
>- public:
>-  static const uint8_t kMaxFrameReferences = 5;
>-
>-  FrameObject();
>-  virtual ~FrameObject() {}
>-
>-  virtual bool GetBitstream(uint8_t* destination) const = 0;
>-
>-  // The capture timestamp of this frame.
>-  virtual uint32_t Timestamp() const = 0;
>-
>-  // When this frame was received.
>-  virtual int64_t ReceivedTime() const = 0;
>-
>-  // When this frame should be rendered.
>-  virtual int64_t RenderTime() const = 0;
>-
>-  // This information is currently needed by the timing calculation class.
>-  // TODO(philipel): Remove this function when a new timing class has
>-  //                 been implemented.
>-  virtual bool delayed_by_retransmission() const { return 0; }
>-
>-  size_t size() const { return _length; }
>-
>-  bool is_keyframe() const { return num_references == 0; }
>-
>-  // The tuple (|picture_id|, |spatial_layer|) uniquely identifies a frame
>-  // object. For codec types that don't necessarily have picture ids they
>-  // have to be constructed from the header data relevant to that codec.
>-  int64_t picture_id;
>-  uint8_t spatial_layer;
>-  uint32_t timestamp;
>-
>-  // TODO(philipel): Add simple modify/access functions to prevent adding too
>-  // many |references|.
>-  size_t num_references;
>-  int64_t references[kMaxFrameReferences];
>-  bool inter_layer_predicted;
>-};
>-
> class PacketBuffer;
> 
>-class RtpFrameObject : public FrameObject {
>+class RtpFrameObject : public EncodedFrame {
>  public:
>   RtpFrameObject(PacketBuffer* packet_buffer,
>                  uint16_t first_seq_num,
>@@ -82,7 +41,7 @@ class RtpFrameObject : public FrameObject {
>   int64_t ReceivedTime() const override;
>   int64_t RenderTime() const override;
>   bool delayed_by_retransmission() const override;
>-  rtc::Optional<RTPVideoTypeHeader> GetCodecHeader() const;
>+  absl::optional<RTPVideoTypeHeader> GetCodecHeader() const;
> 
>  private:
>   rtc::scoped_refptr<PacketBuffer> packet_buffer_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/generic_decoder.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/generic_decoder.cc
>index a7d6a85b902..eb16400f4bb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/generic_decoder.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/generic_decoder.cc
>@@ -32,8 +32,7 @@ VCMDecodedFrameCallback::VCMDecodedFrameCallback(VCMTiming* timing,
>       _clock->CurrentNtpInMilliseconds() - _clock->TimeInMilliseconds();
> }
> 
>-VCMDecodedFrameCallback::~VCMDecodedFrameCallback() {
>-}
>+VCMDecodedFrameCallback::~VCMDecodedFrameCallback() {}
> 
> void VCMDecodedFrameCallback::SetUserReceiveCallback(
>     VCMReceiveCallback* receiveCallback) {
>@@ -57,15 +56,15 @@ int32_t VCMDecodedFrameCallback::Decoded(VideoFrame& decodedImage) {
> int32_t VCMDecodedFrameCallback::Decoded(VideoFrame& decodedImage,
>                                          int64_t decode_time_ms) {
>   Decoded(decodedImage,
>-          decode_time_ms >= 0 ? rtc::Optional<int32_t>(decode_time_ms)
>-                              : rtc::nullopt,
>-          rtc::nullopt);
>+          decode_time_ms >= 0 ? absl::optional<int32_t>(decode_time_ms)
>+                              : absl::nullopt,
>+          absl::nullopt);
>   return WEBRTC_VIDEO_CODEC_OK;
> }
> 
> void VCMDecodedFrameCallback::Decoded(VideoFrame& decodedImage,
>-                                      rtc::Optional<int32_t> decode_time_ms,
>-                                      rtc::Optional<uint8_t> qp) {
>+                                      absl::optional<int32_t> decode_time_ms,
>+                                      absl::optional<uint8_t> qp) {
>   RTC_DCHECK(_receiveCallback) << "Callback must not be null at this point";
>   TRACE_EVENT_INSTANT1("webrtc", "VCMDecodedFrameCallback::Decoded",
>                        "timestamp", decodedImage.timestamp());
>@@ -91,7 +90,7 @@ void VCMDecodedFrameCallback::Decoded(VideoFrame& decodedImage,
>                            frameInfo->renderTimeMs);
> 
>   // Report timing information.
>-  if (frameInfo->timing.flags != TimingFrameFlags::kInvalid) {
>+  if (frameInfo->timing.flags != VideoSendTiming::kInvalid) {
>     int64_t capture_time_ms = decodedImage.ntp_time_ms() - ntp_offset_;
>     // Convert remote timestamps to local time from ntp timestamps.
>     frameInfo->timing.encode_start_ms -= ntp_offset_;
>@@ -141,8 +140,8 @@ void VCMDecodedFrameCallback::Decoded(VideoFrame& decodedImage,
>     _timing->SetTimingFrameInfo(timing_frame_info);
>   }
> 
>-  decodedImage.set_timestamp_us(
>-      frameInfo->renderTimeMs * rtc::kNumMicrosecsPerMillisec);
>+  decodedImage.set_timestamp_us(frameInfo->renderTimeMs *
>+                                rtc::kNumMicrosecsPerMillisec);
>   decodedImage.set_rotation(frameInfo->rotation);
>   _receiveCallback->FrameToRender(decodedImage, qp, frameInfo->content_type);
> }
>@@ -189,7 +188,7 @@ VCMGenericDecoder::VCMGenericDecoder(VideoDecoder* decoder, bool isExternal)
>       _frameInfos(),
>       _nextFrameInfoIdx(0),
>       decoder_(decoder),
>-      _codecType(kVideoCodecUnknown),
>+      _codecType(kVideoCodecGeneric),
>       _isExternal(isExternal),
>       _last_keyframe_content_type(VideoContentType::UNSPECIFIED) {
>   RTC_DCHECK(decoder_);
>@@ -211,41 +210,39 @@ int32_t VCMGenericDecoder::InitDecode(const VideoCodec* settings,
> }
> 
> int32_t VCMGenericDecoder::Decode(const VCMEncodedFrame& frame, int64_t nowMs) {
>-    TRACE_EVENT1("webrtc", "VCMGenericDecoder::Decode", "timestamp",
>-                 frame.EncodedImage()._timeStamp);
>-    _frameInfos[_nextFrameInfoIdx].decodeStartTimeMs = nowMs;
>-    _frameInfos[_nextFrameInfoIdx].renderTimeMs = frame.RenderTimeMs();
>-    _frameInfos[_nextFrameInfoIdx].rotation = frame.rotation();
>-    _frameInfos[_nextFrameInfoIdx].timing = frame.video_timing();
>-    // Set correctly only for key frames. Thus, use latest key frame
>-    // content type. If the corresponding key frame was lost, decode will fail
>-    // and content type will be ignored.
>-    if (frame.FrameType() == kVideoFrameKey) {
>-      _frameInfos[_nextFrameInfoIdx].content_type = frame.contentType();
>-      _last_keyframe_content_type = frame.contentType();
>-    } else {
>-      _frameInfos[_nextFrameInfoIdx].content_type = _last_keyframe_content_type;
>-    }
>-    _callback->Map(frame.TimeStamp(), &_frameInfos[_nextFrameInfoIdx]);
>-
>-    _nextFrameInfoIdx = (_nextFrameInfoIdx + 1) % kDecoderFrameMemoryLength;
>-    const RTPFragmentationHeader dummy_header;
>-    int32_t ret = decoder_->Decode(frame.EncodedImage(), frame.MissingFrame(),
>-                                   &dummy_header,
>-                                   frame.CodecSpecific(), frame.RenderTimeMs());
>-
>-    _callback->OnDecoderImplementationName(decoder_->ImplementationName());
>-    if (ret < WEBRTC_VIDEO_CODEC_OK) {
>-      RTC_LOG(LS_WARNING) << "Failed to decode frame with timestamp "
>-                          << frame.TimeStamp() << ", error code: " << ret;
>-      _callback->Pop(frame.TimeStamp());
>-      return ret;
>-    } else if (ret == WEBRTC_VIDEO_CODEC_NO_OUTPUT ||
>-               ret == WEBRTC_VIDEO_CODEC_REQUEST_SLI) {
>-        // No output
>-        _callback->Pop(frame.TimeStamp());
>-    }
>+  TRACE_EVENT1("webrtc", "VCMGenericDecoder::Decode", "timestamp",
>+               frame.EncodedImage()._timeStamp);
>+  _frameInfos[_nextFrameInfoIdx].decodeStartTimeMs = nowMs;
>+  _frameInfos[_nextFrameInfoIdx].renderTimeMs = frame.RenderTimeMs();
>+  _frameInfos[_nextFrameInfoIdx].rotation = frame.rotation();
>+  _frameInfos[_nextFrameInfoIdx].timing = frame.video_timing();
>+  // Set correctly only for key frames. Thus, use latest key frame
>+  // content type. If the corresponding key frame was lost, decode will fail
>+  // and content type will be ignored.
>+  if (frame.FrameType() == kVideoFrameKey) {
>+    _frameInfos[_nextFrameInfoIdx].content_type = frame.contentType();
>+    _last_keyframe_content_type = frame.contentType();
>+  } else {
>+    _frameInfos[_nextFrameInfoIdx].content_type = _last_keyframe_content_type;
>+  }
>+  _callback->Map(frame.TimeStamp(), &_frameInfos[_nextFrameInfoIdx]);
>+
>+  _nextFrameInfoIdx = (_nextFrameInfoIdx + 1) % kDecoderFrameMemoryLength;
>+  int32_t ret = decoder_->Decode(frame.EncodedImage(), frame.MissingFrame(),
>+                                 frame.CodecSpecific(), frame.RenderTimeMs());
>+
>+  _callback->OnDecoderImplementationName(decoder_->ImplementationName());
>+  if (ret < WEBRTC_VIDEO_CODEC_OK) {
>+    RTC_LOG(LS_WARNING) << "Failed to decode frame with timestamp "
>+                        << frame.TimeStamp() << ", error code: " << ret;
>+    _callback->Pop(frame.TimeStamp());
>     return ret;
>+  } else if (ret == WEBRTC_VIDEO_CODEC_NO_OUTPUT ||
>+             ret == WEBRTC_VIDEO_CODEC_REQUEST_SLI) {
>+    // No output
>+    _callback->Pop(frame.TimeStamp());
>+  }
>+  return ret;
> }
> 
> int32_t VCMGenericDecoder::RegisterDecodeCompleteCallback(
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/generic_decoder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/generic_decoder.h
>index 2d940d883a0..17d2897d634 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/generic_decoder.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/generic_decoder.h
>@@ -46,8 +46,8 @@ class VCMDecodedFrameCallback : public DecodedImageCallback {
>   int32_t Decoded(VideoFrame& decodedImage) override;
>   int32_t Decoded(VideoFrame& decodedImage, int64_t decode_time_ms) override;
>   void Decoded(VideoFrame& decodedImage,
>-               rtc::Optional<int32_t> decode_time_ms,
>-               rtc::Optional<uint8_t> qp) override;
>+               absl::optional<int32_t> decode_time_ms,
>+               absl::optional<uint8_t> qp) override;
>   int32_t ReceivedDecodedReferenceFrame(const uint64_t pictureId) override;
>   int32_t ReceivedDecodedFrame(const uint64_t pictureId) override;
> 
>@@ -81,20 +81,20 @@ class VCMGenericDecoder {
>   ~VCMGenericDecoder();
> 
>   /**
>-  * Initialize the decoder with the information from the VideoCodec
>-  */
>+   * Initialize the decoder with the information from the VideoCodec
>+   */
>   int32_t InitDecode(const VideoCodec* settings, int32_t numberOfCores);
> 
>   /**
>-  * Decode to a raw I420 frame,
>-  *
>-  * inputVideoBuffer reference to encoded video frame
>-  */
>+   * Decode to a raw I420 frame,
>+   *
>+   * inputVideoBuffer reference to encoded video frame
>+   */
>   int32_t Decode(const VCMEncodedFrame& inputFrame, int64_t nowMs);
> 
>   /**
>-  * Set decode callback. Deregistering while decoding is illegal.
>-  */
>+   * Set decode callback. Deregistering while decoding is illegal.
>+   */
>   int32_t RegisterDecodeCompleteCallback(VCMDecodedFrameCallback* callback);
> 
>   bool External() const;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/generic_encoder.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/generic_encoder.cc
>index bb5f75e11bf..5cea0a5a652 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/generic_encoder.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/generic_encoder.cc
>@@ -1,18 +1,18 @@
> /*
>-*  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>-*
>-*  Use of this source code is governed by a BSD-style license
>-*  that can be found in the LICENSE file in the root of the source
>-*  tree. An additional intellectual property rights grant can be found
>-*  in the file PATENTS.  All contributing project authors may
>-*  be found in the AUTHORS file in the root of the source tree.
>-*/
>+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
> 
> #include "modules/video_coding/generic_encoder.h"
> 
> #include <vector>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> #include "api/video/i420_buffer.h"
> #include "modules/include/module_common_types_public.h"
> #include "modules/video_coding/encoded_frame.h"
>@@ -31,6 +31,9 @@ const int kMessagesThrottlingThreshold = 2;
> const int kThrottleRatio = 100000;
> }  // namespace
> 
>+VCMEncodedFrameCallback::TimingFramesLayerInfo::TimingFramesLayerInfo() {}
>+VCMEncodedFrameCallback::TimingFramesLayerInfo::~TimingFramesLayerInfo() {}
>+
> VCMGenericEncoder::VCMGenericEncoder(
>     VideoEncoder* encoder,
>     VCMEncodedFrameCallback* encoded_frame_callback,
>@@ -38,8 +41,9 @@ VCMGenericEncoder::VCMGenericEncoder(
>     : encoder_(encoder),
>       vcm_encoded_frame_callback_(encoded_frame_callback),
>       internal_source_(internal_source),
>-      encoder_params_({BitrateAllocation(), 0, 0, 0}),
>-      streams_or_svc_num_(0) {}
>+      encoder_params_({VideoBitrateAllocation(), 0, 0, 0}),
>+      streams_or_svc_num_(0),
>+      codec_type_(VideoCodecType::kVideoCodecGeneric) {}
> 
> VCMGenericEncoder::~VCMGenericEncoder() {}
> 
>@@ -68,8 +72,9 @@ int32_t VCMGenericEncoder::InitEncode(const VideoCodec* settings,
> 
>   if (encoder_->InitEncode(settings, number_of_cores, max_payload_size) != 0) {
>     RTC_LOG(LS_ERROR) << "Failed to initialize the encoder associated with "
>-                         "payload name: "
>-                      << settings->plName;
>+                         "codec type: "
>+                      << CodecTypeToPayloadString(settings->codecType) << " ("
>+                      << settings->codecType << ")";
>     return -1;
>   }
>   vcm_encoded_frame_callback_->Reset();
>@@ -127,21 +132,8 @@ void VCMGenericEncoder::SetEncoderParameters(const EncoderParameters& params) {
>     }
>     vcm_encoded_frame_callback_->OnFrameRateChanged(params.input_frame_rate);
>     for (size_t i = 0; i < streams_or_svc_num_; ++i) {
>-      size_t layer_bitrate_bytes_per_sec =
>-          params.target_bitrate.GetSpatialLayerSum(i) / 8;
>-      // VP9 rate control is not yet moved out of VP9Impl. Due to that rates
>-      // are not split among spatial layers. Use default 1:2:4 bitrate
>-      // distribution.
>-      // TODO(ilnik): move bitrate per spatial layer calculations out of
>-      // vp9_impl.cc and drop the check below.
>-      if (codec_type_ == kVideoCodecVP9) {
>-        int scaling_factor_num = 1 << i;                          // 1, 2 or 4
>-        int scaling_factor_den = (1 << streams_or_svc_num_) - 1;  // 1 + 2 + 4
>-        layer_bitrate_bytes_per_sec = params.target_bitrate.get_sum_bps() / 8 *
>-                                      scaling_factor_num / scaling_factor_den;
>-      }
>       vcm_encoded_frame_callback_->OnTargetBitrateChanged(
>-          layer_bitrate_bytes_per_sec, i);
>+          params.target_bitrate.GetSpatialLayerSum(i) / 8, i);
>     }
>   }
> }
>@@ -151,11 +143,6 @@ EncoderParameters VCMGenericEncoder::GetEncoderParameters() const {
>   return encoder_params_;
> }
> 
>-int32_t VCMGenericEncoder::SetPeriodicKeyFrames(bool enable) {
>-  RTC_DCHECK_RUNS_SERIALIZED(&race_checker_);
>-  return encoder_->SetPeriodicKeyFrames(enable);
>-}
>-
> int32_t VCMGenericEncoder::RequestFrame(
>     const std::vector<FrameType>& frame_types) {
>   RTC_DCHECK_RUNS_SERIALIZED(&race_checker_);
>@@ -167,9 +154,9 @@ int32_t VCMGenericEncoder::RequestFrame(
>   // VideoSendStreamTest.VideoSendStreamStopSetEncoderRateToZero, set
>   // internal_source to true and use FakeEncoder. And the latter will
>   // happily encode this 1x1 frame and pass it on down the pipeline.
>-  return encoder_->Encode(VideoFrame(I420Buffer::Create(1, 1),
>-                                     kVideoRotation_0, 0),
>-                          NULL, &frame_types);
>+  return encoder_->Encode(
>+      VideoFrame(I420Buffer::Create(1, 1), kVideoRotation_0, 0), NULL,
>+      &frame_types);
>   return 0;
> }
> 
>@@ -194,7 +181,7 @@ VCMEncodedFrameCallback::VCMEncodedFrameCallback(
>       incorrect_capture_time_logged_messages_(0),
>       reordered_frames_logged_messages_(0),
>       stalled_encoder_logged_messages_(0) {
>-  rtc::Optional<AlrExperimentSettings> experiment_settings =
>+  absl::optional<AlrExperimentSettings> experiment_settings =
>       AlrExperimentSettings::CreateFromFieldTrial(
>           AlrExperimentSettings::kStrictPacingAndProbingExperimentName);
>   if (experiment_settings) {
>@@ -265,88 +252,79 @@ void VCMEncodedFrameCallback::OnEncodeStarted(uint32_t rtp_timestamp,
>       rtp_timestamp, capture_time_ms, rtc::TimeMillis());
> }
> 
>-EncodedImageCallback::Result VCMEncodedFrameCallback::OnEncodedImage(
>-    const EncodedImage& encoded_image,
>-    const CodecSpecificInfo* codec_specific,
>-    const RTPFragmentationHeader* fragmentation_header) {
>-  TRACE_EVENT_INSTANT1("webrtc", "VCMEncodedFrameCallback::Encoded",
>-                       "timestamp", encoded_image._timeStamp);
>-  size_t simulcast_svc_idx = 0;
>-  if (codec_specific->codecType == kVideoCodecVP9) {
>-    if (codec_specific->codecSpecific.VP9.num_spatial_layers > 1)
>-      simulcast_svc_idx = codec_specific->codecSpecific.VP9.spatial_idx;
>-  } else if (codec_specific->codecType == kVideoCodecVP8) {
>-    simulcast_svc_idx = codec_specific->codecSpecific.VP8.simulcastIdx;
>-  } else if (codec_specific->codecType == kVideoCodecGeneric) {
>-    simulcast_svc_idx = codec_specific->codecSpecific.generic.simulcast_idx;
>-  } else if (codec_specific->codecType == kVideoCodecH264) {
>-    // TODO(ilnik): When h264 simulcast is landed, extract simulcast idx here.
>-  }
>-
>-  EncodedImage image_copy(encoded_image);
>-
>-  rtc::Optional<size_t> outlier_frame_size;
>-  rtc::Optional<int64_t> encode_start_ms;
>-  size_t num_simulcast_svc_streams = 1;
>-  uint8_t timing_flags = TimingFrameFlags::kNotTriggered;
>-  if (!internal_source_) {
>-    rtc::CritScope crit(&timing_params_lock_);
>-
>-    // Encoders with internal sources do not call OnEncodeStarted and
>-    // OnFrameRateChanged. |timing_frames_info_| may be not filled here.
>-    num_simulcast_svc_streams = timing_frames_info_.size();
>-    if (simulcast_svc_idx < num_simulcast_svc_streams) {
>-      auto encode_start_list =
>-          &timing_frames_info_[simulcast_svc_idx].encode_start_list;
>-      // Skip frames for which there was OnEncodeStarted but no OnEncodedImage
>-      // call. These are dropped by encoder internally.
>-      // Because some hardware encoders don't preserve capture timestamp we use
>-      // RTP timestamps here.
>-      while (!encode_start_list->empty() &&
>-             IsNewerTimestamp(image_copy._timeStamp,
>-                              encode_start_list->front().rtp_timestamp)) {
>-        post_encode_callback_->OnDroppedFrame(DropReason::kDroppedByEncoder);
>-        encode_start_list->pop_front();
>-      }
>-      if (encode_start_list->size() > 0 &&
>-          encode_start_list->front().rtp_timestamp == image_copy._timeStamp) {
>-        encode_start_ms.emplace(
>-            encode_start_list->front().encode_start_time_ms);
>-        if (image_copy.capture_time_ms_ !=
>-            encode_start_list->front().capture_time_ms) {
>-          // Force correct capture timestamp.
>-          image_copy.capture_time_ms_ =
>-              encode_start_list->front().capture_time_ms;
>-          ++incorrect_capture_time_logged_messages_;
>-          if (incorrect_capture_time_logged_messages_ <=
>-                  kMessagesThrottlingThreshold ||
>-              incorrect_capture_time_logged_messages_ % kThrottleRatio == 0) {
>-            RTC_LOG(LS_WARNING)
>-                << "Encoder is not preserving capture timestamps.";
>-            if (incorrect_capture_time_logged_messages_ ==
>-                kMessagesThrottlingThreshold) {
>-              RTC_LOG(LS_WARNING) << "Too many log messages. Further incorrect "
>-                                     "timestamps warnings will be throttled.";
>-            }
>-          }
>-        }
>-        encode_start_list->pop_front();
>-      } else {
>-        ++reordered_frames_logged_messages_;
>-        if (reordered_frames_logged_messages_ <= kMessagesThrottlingThreshold ||
>-            reordered_frames_logged_messages_ % kThrottleRatio == 0) {
>+absl::optional<int64_t> VCMEncodedFrameCallback::ExtractEncodeStartTime(
>+    size_t simulcast_svc_idx,
>+    EncodedImage* encoded_image) {
>+  absl::optional<int64_t> result;
>+  size_t num_simulcast_svc_streams = timing_frames_info_.size();
>+  if (simulcast_svc_idx < num_simulcast_svc_streams) {
>+    auto encode_start_list =
>+        &timing_frames_info_[simulcast_svc_idx].encode_start_list;
>+    // Skip frames for which there was OnEncodeStarted but no OnEncodedImage
>+    // call. These are dropped by encoder internally.
>+    // Because some hardware encoders don't preserve capture timestamp we
>+    // use RTP timestamps here.
>+    while (!encode_start_list->empty() &&
>+           IsNewerTimestamp(encoded_image->_timeStamp,
>+                            encode_start_list->front().rtp_timestamp)) {
>+      post_encode_callback_->OnDroppedFrame(DropReason::kDroppedByEncoder);
>+      encode_start_list->pop_front();
>+    }
>+    if (encode_start_list->size() > 0 &&
>+        encode_start_list->front().rtp_timestamp == encoded_image->_timeStamp) {
>+      result.emplace(encode_start_list->front().encode_start_time_ms);
>+      if (encoded_image->capture_time_ms_ !=
>+          encode_start_list->front().capture_time_ms) {
>+        // Force correct capture timestamp.
>+        encoded_image->capture_time_ms_ =
>+            encode_start_list->front().capture_time_ms;
>+        ++incorrect_capture_time_logged_messages_;
>+        if (incorrect_capture_time_logged_messages_ <=
>+                kMessagesThrottlingThreshold ||
>+            incorrect_capture_time_logged_messages_ % kThrottleRatio == 0) {
>           RTC_LOG(LS_WARNING)
>-              << "Frame with no encode started time recordings. "
>-                 "Encoder may be reordering frames "
>-                 "or not preserving RTP timestamps.";
>-          if (reordered_frames_logged_messages_ ==
>+              << "Encoder is not preserving capture timestamps.";
>+          if (incorrect_capture_time_logged_messages_ ==
>               kMessagesThrottlingThreshold) {
>-            RTC_LOG(LS_WARNING) << "Too many log messages. Further frames "
>-                                   "reordering warnings will be throttled.";
>+            RTC_LOG(LS_WARNING) << "Too many log messages. Further incorrect "
>+                                   "timestamps warnings will be throttled.";
>           }
>         }
>       }
>+      encode_start_list->pop_front();
>+    } else {
>+      ++reordered_frames_logged_messages_;
>+      if (reordered_frames_logged_messages_ <= kMessagesThrottlingThreshold ||
>+          reordered_frames_logged_messages_ % kThrottleRatio == 0) {
>+        RTC_LOG(LS_WARNING) << "Frame with no encode started time recordings. "
>+                               "Encoder may be reordering frames "
>+                               "or not preserving RTP timestamps.";
>+        if (reordered_frames_logged_messages_ == kMessagesThrottlingThreshold) {
>+          RTC_LOG(LS_WARNING) << "Too many log messages. Further frames "
>+                                 "reordering warnings will be throttled.";
>+        }
>+      }
>+    }
>+  }
>+  return result;
>+}
> 
>+void VCMEncodedFrameCallback::FillTimingInfo(size_t simulcast_svc_idx,
>+                                             EncodedImage* encoded_image) {
>+  absl::optional<size_t> outlier_frame_size;
>+  absl::optional<int64_t> encode_start_ms;
>+  uint8_t timing_flags = VideoSendTiming::kNotTriggered;
>+  {
>+    rtc::CritScope crit(&timing_params_lock_);
>+
>+    // Encoders with internal sources do not call OnEncodeStarted
>+    // |timing_frames_info_| may be not filled here.
>+    if (!internal_source_) {
>+      encode_start_ms =
>+          ExtractEncodeStartTime(simulcast_svc_idx, encoded_image);
>+    }
>+
>+    if (timing_frames_info_.size() > simulcast_svc_idx) {
>       size_t target_bitrate =
>           timing_frames_info_[simulcast_svc_idx].target_bitrate_bytes_per_sec;
>       if (framerate_ > 0 && target_bitrate > 0) {
>@@ -358,24 +336,39 @@ EncodedImageCallback::Result VCMEncodedFrameCallback::OnEncodedImage(
>       }
>     }
> 
>+    // Outliers trigger timing frames, but do not affect scheduled timing
>+    // frames.
>+    if (outlier_frame_size && encoded_image->_length >= *outlier_frame_size) {
>+      timing_flags |= VideoSendTiming::kTriggeredBySize;
>+    }
>+
>     // Check if it's time to send a timing frame.
>     int64_t timing_frame_delay_ms =
>-        image_copy.capture_time_ms_ - last_timing_frame_time_ms_;
>+        encoded_image->capture_time_ms_ - last_timing_frame_time_ms_;
>     // Trigger threshold if it's a first frame, too long passed since the last
>     // timing frame, or we already sent timing frame on a different simulcast
>     // stream with the same capture time.
>     if (last_timing_frame_time_ms_ == -1 ||
>         timing_frame_delay_ms >= timing_frames_thresholds_.delay_ms ||
>         timing_frame_delay_ms == 0) {
>-      timing_flags = TimingFrameFlags::kTriggeredByTimer;
>-      last_timing_frame_time_ms_ = image_copy.capture_time_ms_;
>-    }
>-
>-    // Outliers trigger timing frames, but do not affect scheduled timing
>-    // frames.
>-    if (outlier_frame_size && image_copy._length >= *outlier_frame_size) {
>-      timing_flags |= TimingFrameFlags::kTriggeredBySize;
>+      timing_flags |= VideoSendTiming::kTriggeredByTimer;
>+      last_timing_frame_time_ms_ = encoded_image->capture_time_ms_;
>     }
>+  }  // rtc::CritScope crit(&timing_params_lock_);
>+
>+  int64_t now_ms = rtc::TimeMillis();
>+  // Workaround for chromoting encoder: it passes encode start and finished
>+  // timestamps in |timing_| field, but they (together with capture timestamp)
>+  // are not in the WebRTC clock.
>+  if (internal_source_ && encoded_image->timing_.encode_finish_ms > 0 &&
>+      encoded_image->timing_.encode_start_ms > 0) {
>+    int64_t clock_offset_ms = now_ms - encoded_image->timing_.encode_finish_ms;
>+    // Translate capture timestamp to local WebRTC clock.
>+    encoded_image->capture_time_ms_ += clock_offset_ms;
>+    encoded_image->_timeStamp =
>+        static_cast<uint32_t>(encoded_image->capture_time_ms_ * 90);
>+    encode_start_ms.emplace(encoded_image->timing_.encode_start_ms +
>+                            clock_offset_ms);
>   }
> 
>   // If encode start is not available that means that encoder uses internal
>@@ -384,12 +377,35 @@ EncodedImageCallback::Result VCMEncodedFrameCallback::OnEncodedImage(
>   // because to being sent in the network capture time required to be less than
>   // all the other timestamps.
>   if (encode_start_ms) {
>-    image_copy.SetEncodeTime(*encode_start_ms, rtc::TimeMillis());
>-    image_copy.timing_.flags = timing_flags;
>+    encoded_image->SetEncodeTime(*encode_start_ms, now_ms);
>+    encoded_image->timing_.flags = timing_flags;
>   } else {
>-    image_copy.timing_.flags = TimingFrameFlags::kInvalid;
>+    encoded_image->timing_.flags = VideoSendTiming::kInvalid;
>+  }
>+}
>+
>+EncodedImageCallback::Result VCMEncodedFrameCallback::OnEncodedImage(
>+    const EncodedImage& encoded_image,
>+    const CodecSpecificInfo* codec_specific,
>+    const RTPFragmentationHeader* fragmentation_header) {
>+  TRACE_EVENT_INSTANT1("webrtc", "VCMEncodedFrameCallback::Encoded",
>+                       "timestamp", encoded_image._timeStamp);
>+  size_t simulcast_svc_idx = 0;
>+  if (codec_specific->codecType == kVideoCodecVP9) {
>+    if (codec_specific->codecSpecific.VP9.num_spatial_layers > 1)
>+      simulcast_svc_idx = codec_specific->codecSpecific.VP9.spatial_idx;
>+  } else if (codec_specific->codecType == kVideoCodecVP8) {
>+    simulcast_svc_idx = codec_specific->codecSpecific.VP8.simulcastIdx;
>+  } else if (codec_specific->codecType == kVideoCodecGeneric) {
>+    simulcast_svc_idx = codec_specific->codecSpecific.generic.simulcast_idx;
>+  } else if (codec_specific->codecType == kVideoCodecH264) {
>+    // TODO(ilnik): When h264 simulcast is landed, extract simulcast idx here.
>   }
> 
>+  EncodedImage image_copy(encoded_image);
>+
>+  FillTimingInfo(simulcast_svc_idx, &image_copy);
>+
>   // Piggyback ALR experiment group id and simulcast id into the content type.
>   uint8_t experiment_id =
>       experiment_groups_[videocontenttypehelpers::IsScreenshare(
>@@ -412,7 +428,8 @@ EncodedImageCallback::Result VCMEncodedFrameCallback::OnEncodedImage(
>     return result;
> 
>   if (media_opt_) {
>-    media_opt_->UpdateWithEncodedData(image_copy);
>+    media_opt_->UpdateWithEncodedData(image_copy._length,
>+                                      image_copy._frameType);
>     if (internal_source_) {
>       // Signal to encoder to drop next frame.
>       result.drop_next_frame = media_opt_->DropFrame();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/generic_encoder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/generic_encoder.h
>index 244c61444ee..151e93e4975 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/generic_encoder.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/generic_encoder.h
>@@ -28,7 +28,7 @@ class MediaOptimization;
> }  // namespace media_optimization
> 
> struct EncoderParameters {
>-  BitrateAllocation target_bitrate;
>+  VideoBitrateAllocation target_bitrate;
>   uint8_t loss_rate;
>   int64_t rtt;
>   uint32_t input_frame_rate;
>@@ -38,7 +38,7 @@ class VCMEncodedFrameCallback : public EncodedImageCallback {
>  public:
>   VCMEncodedFrameCallback(EncodedImageCallback* post_encode_callback,
>                           media_optimization::MediaOptimization* media_opt);
>-  virtual ~VCMEncodedFrameCallback();
>+  ~VCMEncodedFrameCallback() override;
> 
>   // Implements EncodedImageCallback.
>   EncodedImageCallback::Result OnEncodedImage(
>@@ -77,6 +77,14 @@ class VCMEncodedFrameCallback : public EncodedImageCallback {
>   }
> 
>  private:
>+  // For non-internal-source encoders, returns encode started time and fixes
>+  // capture timestamp for the frame, if corrupted by the encoder.
>+  absl::optional<int64_t> ExtractEncodeStartTime(size_t simulcast_svc_idx,
>+                                                 EncodedImage* encoded_image)
>+      RTC_EXCLUSIVE_LOCKS_REQUIRED(timing_params_lock_);
>+
>+  void FillTimingInfo(size_t simulcast_svc_idx, EncodedImage* encoded_image);
>+
>   rtc::CriticalSection timing_params_lock_;
>   bool internal_source_;
>   EncodedImageCallback* const post_encode_callback_;
>@@ -94,6 +102,8 @@ class VCMEncodedFrameCallback : public EncodedImageCallback {
>     int64_t encode_start_time_ms;
>   };
>   struct TimingFramesLayerInfo {
>+    TimingFramesLayerInfo();
>+    ~TimingFramesLayerInfo();
>     size_t target_bitrate_bytes_per_sec = 0;
>     std::list<EncodeStartTimeRecord> encode_start_list;
>   };
>@@ -134,7 +144,6 @@ class VCMGenericEncoder {
>   void SetEncoderParameters(const EncoderParameters& params);
>   EncoderParameters GetEncoderParameters() const;
> 
>-  int32_t SetPeriodicKeyFrames(bool enable);
>   int32_t RequestFrame(const std::vector<FrameType>& frame_types);
>   bool InternalSource() const;
>   bool SupportsNativeHandle() const;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/generic_encoder_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/generic_encoder_unittest.cc
>index 2b788c55710..603738176fd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/generic_encoder_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/generic_encoder_unittest.cc
>@@ -13,6 +13,7 @@
> #include "modules/video_coding/encoded_frame.h"
> #include "modules/video_coding/generic_encoder.h"
> #include "modules/video_coding/include/video_coding_defines.h"
>+#include "rtc_base/fakeclock.h"
> #include "test/gtest.h"
> 
> namespace webrtc {
>@@ -35,8 +36,8 @@ class FakeEncodedImageCallback : public EncodedImageCallback {
>                         const CodecSpecificInfo* codec_specific_info,
>                         const RTPFragmentationHeader* fragmentation) override {
>     last_frame_was_timing_ =
>-        encoded_image.timing_.flags != TimingFrameFlags::kInvalid &&
>-        encoded_image.timing_.flags != TimingFrameFlags::kNotTriggered;
>+        encoded_image.timing_.flags != VideoSendTiming::kInvalid &&
>+        encoded_image.timing_.flags != VideoSendTiming::kNotTriggered;
>     last_capture_timestamp_ = encoded_image.capture_time_ms_;
>     return Result(Result::OK);
>   };
>@@ -208,6 +209,46 @@ TEST(TestVCMEncodedFrameCallback, NoTimingFrameIfNoEncodeStartTime) {
>   EXPECT_FALSE(sink.WasTimingFrame());
> }
> 
>+TEST(TestVCMEncodedFrameCallback, AdjustsCaptureTimeForInternalSourceEncoder) {
>+  rtc::ScopedFakeClock clock;
>+  clock.SetTimeMicros(1234567);
>+  EncodedImage image;
>+  CodecSpecificInfo codec_specific;
>+  const int64_t kEncodeStartDelayMs = 2;
>+  const int64_t kEncodeFinishDelayMs = 10;
>+  int64_t timestamp = 1;
>+  image._length = 500;
>+  image.capture_time_ms_ = timestamp;
>+  image._timeStamp = static_cast<uint32_t>(timestamp * 90);
>+  codec_specific.codecType = kVideoCodecGeneric;
>+  codec_specific.codecSpecific.generic.simulcast_idx = 0;
>+  FakeEncodedImageCallback sink;
>+  VCMEncodedFrameCallback callback(&sink, nullptr);
>+  callback.SetInternalSource(true);
>+  VideoCodec::TimingFrameTriggerThresholds thresholds;
>+  thresholds.delay_ms = 1;  // Make all frames timing frames.
>+  callback.SetTimingFramesThresholds(thresholds);
>+  callback.OnTargetBitrateChanged(500, 0);
>+
>+  // Verify a single frame without encode timestamps isn't a timing frame.
>+  callback.OnEncodedImage(image, &codec_specific, nullptr);
>+  EXPECT_FALSE(sink.WasTimingFrame());
>+
>+  // New frame, but this time with encode timestamps set in timing_.
>+  // This should be a timing frame.
>+  image.capture_time_ms_ = ++timestamp;
>+  image._timeStamp = static_cast<uint32_t>(timestamp * 90);
>+  image.timing_.encode_start_ms = timestamp + kEncodeStartDelayMs;
>+  image.timing_.encode_finish_ms = timestamp + kEncodeFinishDelayMs;
>+  callback.OnEncodedImage(image, &codec_specific, nullptr);
>+  EXPECT_TRUE(sink.WasTimingFrame());
>+  // Frame is captured kEncodeFinishDelayMs before it's encoded, so restored
>+  // capture timestamp should be kEncodeFinishDelayMs in the past.
>+  EXPECT_EQ(
>+      sink.GetLastCaptureTimestamp(),
>+      clock.TimeNanos() / rtc::kNumNanosecsPerMillisec - kEncodeFinishDelayMs);
>+}
>+
> TEST(TestVCMEncodedFrameCallback, NotifiesAboutDroppedFrames) {
>   EncodedImage image;
>   CodecSpecificInfo codec_specific;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/h264_sprop_parameter_sets.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/h264_sprop_parameter_sets.cc
>index be0d17ceff8..e8585456820 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/h264_sprop_parameter_sets.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/h264_sprop_parameter_sets.cc
>@@ -13,9 +13,8 @@
> #include <string>
> #include <vector>
> 
>-#include "rtc_base/base64.h"
>-#include "rtc_base/basictypes.h"
> #include "rtc_base/logging.h"
>+#include "rtc_base/third_party/base64/base64.h"
> 
> namespace {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/h264_sps_pps_tracker.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/h264_sps_pps_tracker.cc
>index e8864e28918..d67d827b081 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/h264_sps_pps_tracker.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/h264_sps_pps_tracker.cc
>@@ -36,14 +36,15 @@ H264SpsPpsTracker::PacketAction H264SpsPpsTracker::CopyAndFixBitstream(
>   const uint8_t* data = packet->dataPtr;
>   const size_t data_size = packet->sizeBytes;
>   const RTPVideoHeader& video_header = packet->video_header;
>-  RTPVideoHeaderH264* codec_header = &packet->video_header.codecHeader.H264;
>+  auto& h264_header =
>+      absl::get<RTPVideoHeaderH264>(packet->video_header.video_type_header);
> 
>   bool append_sps_pps = false;
>   auto sps = sps_data_.end();
>   auto pps = pps_data_.end();
> 
>-  for (size_t i = 0; i < codec_header->nalus_length; ++i) {
>-    const NaluInfo& nalu = codec_header->nalus[i];
>+  for (size_t i = 0; i < h264_header.nalus_length; ++i) {
>+    const NaluInfo& nalu = h264_header.nalus[i];
>     switch (nalu.type) {
>       case H264::NaluType::kSps: {
>         sps_data_[nalu.sps_id].width = packet->width;
>@@ -110,7 +111,7 @@ H264SpsPpsTracker::PacketAction H264SpsPpsTracker::CopyAndFixBitstream(
>     required_size += pps->second.size + sizeof(start_code_h264);
>   }
> 
>-  if (codec_header->packetization_type == kH264StapA) {
>+  if (h264_header.packetization_type == kH264StapA) {
>     const uint8_t* nalu_ptr = data + 1;
>     while (nalu_ptr < data + data_size) {
>       RTC_DCHECK(video_header.is_first_packet_in_frame);
>@@ -155,9 +156,9 @@ H264SpsPpsTracker::PacketAction H264SpsPpsTracker::CopyAndFixBitstream(
>     pps_info.type = H264::NaluType::kPps;
>     pps_info.sps_id = sps->first;
>     pps_info.pps_id = pps->first;
>-    if (codec_header->nalus_length + 2 <= kMaxNalusPerPacket) {
>-      codec_header->nalus[codec_header->nalus_length++] = sps_info;
>-      codec_header->nalus[codec_header->nalus_length++] = pps_info;
>+    if (h264_header.nalus_length + 2 <= kMaxNalusPerPacket) {
>+      h264_header.nalus[h264_header.nalus_length++] = sps_info;
>+      h264_header.nalus[h264_header.nalus_length++] = pps_info;
>     } else {
>       RTC_LOG(LS_WARNING) << "Not enough space in H.264 codec header to insert "
>                              "SPS/PPS provided out-of-band.";
>@@ -165,7 +166,7 @@ H264SpsPpsTracker::PacketAction H264SpsPpsTracker::CopyAndFixBitstream(
>   }
> 
>   // Copy the rest of the bitstream and insert start codes.
>-  if (codec_header->packetization_type == kH264StapA) {
>+  if (h264_header.packetization_type == kH264StapA) {
>     const uint8_t* nalu_ptr = data + 1;
>     while (nalu_ptr < data + data_size) {
>       memcpy(insert_at, start_code_h264, sizeof(start_code_h264));
>@@ -219,9 +220,9 @@ void H264SpsPpsTracker::InsertSpsPpsNalus(const std::vector<uint8_t>& sps,
>     RTC_LOG(LS_WARNING) << "SPS Nalu header missing";
>     return;
>   }
>-  rtc::Optional<SpsParser::SpsState> parsed_sps = SpsParser::ParseSps(
>+  absl::optional<SpsParser::SpsState> parsed_sps = SpsParser::ParseSps(
>       sps.data() + kNaluHeaderOffset, sps.size() - kNaluHeaderOffset);
>-  rtc::Optional<PpsParser::PpsState> parsed_pps = PpsParser::ParsePps(
>+  absl::optional<PpsParser::PpsState> parsed_pps = PpsParser::ParsePps(
>       pps.data() + kNaluHeaderOffset, pps.size() - kNaluHeaderOffset);
> 
>   if (!parsed_sps) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/h264_sps_pps_tracker_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/h264_sps_pps_tracker_unittest.cc
>index f88992f28bb..dad82943dcd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/h264_sps_pps_tracker_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/h264_sps_pps_tracker_unittest.cc
>@@ -12,9 +12,9 @@
> 
> #include <vector>
> 
>+#include "common_video/h264/h264_common.h"
> #include "modules/video_coding/packet.h"
> #include "test/gtest.h"
>-#include "common_video/h264/h264_common.h"
> 
> namespace webrtc {
> namespace video_coding {
>@@ -46,21 +46,29 @@ void ExpectSpsPpsIdr(const RTPVideoHeaderH264& codec_header,
>   EXPECT_TRUE(contains_idr);
> }
> 
>+class H264VcmPacket : public VCMPacket {
>+ public:
>+  H264VcmPacket() {
>+    codec = kVideoCodecH264;
>+    video_header.is_first_packet_in_frame = false;
>+    auto& type_header =
>+        video_header.video_type_header.emplace<RTPVideoHeaderH264>();
>+    type_header.nalus_length = 0;
>+    type_header.packetization_type = kH264SingleNalu;
>+  }
>+
>+  RTPVideoHeaderH264& h264() {
>+    return absl::get<RTPVideoHeaderH264>(video_header.video_type_header);
>+  }
>+};
>+
> }  // namespace
> 
> class TestH264SpsPpsTracker : public ::testing::Test {
>  public:
>-  VCMPacket GetDefaultPacket() {
>-    VCMPacket packet;
>-    packet.codec = kVideoCodecH264;
>-    packet.video_header.codecHeader.H264.nalus_length = 0;
>-    packet.video_header.is_first_packet_in_frame = false;
>-    packet.video_header.codecHeader.H264.packetization_type = kH264SingleNalu;
>-
>-    return packet;
>-  }
>-
>-  void AddSps(VCMPacket* packet, uint8_t sps_id, std::vector<uint8_t>* data) {
>+  void AddSps(H264VcmPacket* packet,
>+              uint8_t sps_id,
>+              std::vector<uint8_t>* data) {
>     NaluInfo info;
>     info.type = H264::NaluType::kSps;
>     info.sps_id = sps_id;
>@@ -68,11 +76,10 @@ class TestH264SpsPpsTracker : public ::testing::Test {
>     data->push_back(H264::NaluType::kSps);
>     data->push_back(sps_id);  // The sps data, just a single byte.
> 
>-    packet->video_header.codecHeader.H264
>-        .nalus[packet->video_header.codecHeader.H264.nalus_length++] = info;
>+    packet->h264().nalus[packet->h264().nalus_length++] = info;
>   }
> 
>-  void AddPps(VCMPacket* packet,
>+  void AddPps(H264VcmPacket* packet,
>               uint8_t sps_id,
>               uint8_t pps_id,
>               std::vector<uint8_t>* data) {
>@@ -83,18 +90,16 @@ class TestH264SpsPpsTracker : public ::testing::Test {
>     data->push_back(H264::NaluType::kPps);
>     data->push_back(pps_id);  // The pps data, just a single byte.
> 
>-    packet->video_header.codecHeader.H264
>-        .nalus[packet->video_header.codecHeader.H264.nalus_length++] = info;
>+    packet->h264().nalus[packet->h264().nalus_length++] = info;
>   }
> 
>-  void AddIdr(VCMPacket* packet, int pps_id) {
>+  void AddIdr(H264VcmPacket* packet, int pps_id) {
>     NaluInfo info;
>     info.type = H264::NaluType::kIdr;
>     info.sps_id = -1;
>     info.pps_id = pps_id;
> 
>-    packet->video_header.codecHeader.H264
>-        .nalus[packet->video_header.codecHeader.H264.nalus_length++] = info;
>+    packet->h264().nalus[packet->h264().nalus_length++] = info;
>   }
> 
>  protected:
>@@ -103,8 +108,8 @@ class TestH264SpsPpsTracker : public ::testing::Test {
> 
> TEST_F(TestH264SpsPpsTracker, NoNalus) {
>   uint8_t data[] = {1, 2, 3};
>-  VCMPacket packet = GetDefaultPacket();
>-  packet.video_header.codecHeader.H264.packetization_type = kH264FuA;
>+  H264VcmPacket packet;
>+  packet.h264().packetization_type = kH264FuA;
>   packet.dataPtr = data;
>   packet.sizeBytes = sizeof(data);
> 
>@@ -115,8 +120,8 @@ TEST_F(TestH264SpsPpsTracker, NoNalus) {
> 
> TEST_F(TestH264SpsPpsTracker, FuAFirstPacket) {
>   uint8_t data[] = {1, 2, 3};
>-  VCMPacket packet = GetDefaultPacket();
>-  packet.video_header.codecHeader.H264.packetization_type = kH264FuA;
>+  H264VcmPacket packet;
>+  packet.h264().packetization_type = kH264FuA;
>   packet.video_header.is_first_packet_in_frame = true;
>   packet.dataPtr = data;
>   packet.sizeBytes = sizeof(data);
>@@ -131,8 +136,8 @@ TEST_F(TestH264SpsPpsTracker, FuAFirstPacket) {
> 
> TEST_F(TestH264SpsPpsTracker, StapAIncorrectSegmentLength) {
>   uint8_t data[] = {0, 0, 2, 0};
>-  VCMPacket packet = GetDefaultPacket();
>-  packet.video_header.codecHeader.H264.packetization_type = kH264StapA;
>+  H264VcmPacket packet;
>+  packet.h264().packetization_type = kH264StapA;
>   packet.video_header.is_first_packet_in_frame = true;
>   packet.dataPtr = data;
>   packet.sizeBytes = sizeof(data);
>@@ -142,7 +147,7 @@ TEST_F(TestH264SpsPpsTracker, StapAIncorrectSegmentLength) {
> 
> TEST_F(TestH264SpsPpsTracker, NoNalusFirstPacket) {
>   uint8_t data[] = {1, 2, 3};
>-  VCMPacket packet = GetDefaultPacket();
>+  H264VcmPacket packet;
>   packet.video_header.is_first_packet_in_frame = true;
>   packet.dataPtr = data;
>   packet.sizeBytes = sizeof(data);
>@@ -157,8 +162,8 @@ TEST_F(TestH264SpsPpsTracker, NoNalusFirstPacket) {
> 
> TEST_F(TestH264SpsPpsTracker, IdrNoSpsPpsInserted) {
>   std::vector<uint8_t> data = {1, 2, 3};
>-  VCMPacket packet = GetDefaultPacket();
>-  packet.video_header.codecHeader.H264.packetization_type = kH264FuA;
>+  H264VcmPacket packet;
>+  packet.h264().packetization_type = kH264FuA;
> 
>   AddIdr(&packet, 0);
>   packet.dataPtr = data.data();
>@@ -171,7 +176,7 @@ TEST_F(TestH264SpsPpsTracker, IdrNoSpsPpsInserted) {
> 
> TEST_F(TestH264SpsPpsTracker, IdrFirstPacketNoSpsPpsInserted) {
>   std::vector<uint8_t> data = {1, 2, 3};
>-  VCMPacket packet = GetDefaultPacket();
>+  H264VcmPacket packet;
>   packet.video_header.is_first_packet_in_frame = true;
> 
>   AddIdr(&packet, 0);
>@@ -184,7 +189,7 @@ TEST_F(TestH264SpsPpsTracker, IdrFirstPacketNoSpsPpsInserted) {
> 
> TEST_F(TestH264SpsPpsTracker, IdrFirstPacketNoPpsInserted) {
>   std::vector<uint8_t> data = {1, 2, 3};
>-  VCMPacket packet = GetDefaultPacket();
>+  H264VcmPacket packet;
>   packet.video_header.is_first_packet_in_frame = true;
> 
>   AddSps(&packet, 0, &data);
>@@ -198,7 +203,7 @@ TEST_F(TestH264SpsPpsTracker, IdrFirstPacketNoPpsInserted) {
> 
> TEST_F(TestH264SpsPpsTracker, IdrFirstPacketNoSpsInserted) {
>   std::vector<uint8_t> data = {1, 2, 3};
>-  VCMPacket packet = GetDefaultPacket();
>+  H264VcmPacket packet;
>   packet.video_header.is_first_packet_in_frame = true;
> 
>   AddPps(&packet, 0, 0, &data);
>@@ -212,7 +217,7 @@ TEST_F(TestH264SpsPpsTracker, IdrFirstPacketNoSpsInserted) {
> 
> TEST_F(TestH264SpsPpsTracker, SpsPpsPacketThenIdrFirstPacket) {
>   std::vector<uint8_t> data;
>-  VCMPacket sps_pps_packet = GetDefaultPacket();
>+  H264VcmPacket sps_pps_packet;
> 
>   // Insert SPS/PPS
>   AddSps(&sps_pps_packet, 0, &data);
>@@ -225,7 +230,7 @@ TEST_F(TestH264SpsPpsTracker, SpsPpsPacketThenIdrFirstPacket) {
>   data.clear();
> 
>   // Insert first packet of the IDR
>-  VCMPacket idr_packet = GetDefaultPacket();
>+  H264VcmPacket idr_packet;
>   idr_packet.video_header.is_first_packet_in_frame = true;
>   AddIdr(&idr_packet, 1);
>   data.insert(data.end(), {1, 2, 3});
>@@ -243,8 +248,8 @@ TEST_F(TestH264SpsPpsTracker, SpsPpsPacketThenIdrFirstPacket) {
> 
> TEST_F(TestH264SpsPpsTracker, SpsPpsIdrInStapA) {
>   std::vector<uint8_t> data;
>-  VCMPacket packet = GetDefaultPacket();
>-  packet.video_header.codecHeader.H264.packetization_type = kH264StapA;
>+  H264VcmPacket packet;
>+  packet.h264().packetization_type = kH264StapA;
>   packet.video_header.is_first_packet_in_frame = true;  // Always true for StapA
> 
>   data.insert(data.end(), {0});     // First byte is ignored
>@@ -284,18 +289,18 @@ TEST_F(TestH264SpsPpsTracker, SpsPpsOutOfBand) {
>   tracker_.InsertSpsPpsNalus(sps, pps);
> 
>   // Insert first packet of the IDR.
>-  VCMPacket idr_packet = GetDefaultPacket();
>+  H264VcmPacket idr_packet;
>   idr_packet.video_header.is_first_packet_in_frame = true;
>   AddIdr(&idr_packet, 0);
>   idr_packet.dataPtr = kData;
>   idr_packet.sizeBytes = sizeof(kData);
>-  EXPECT_EQ(1u, idr_packet.video_header.codecHeader.H264.nalus_length);
>+  EXPECT_EQ(1u, idr_packet.h264().nalus_length);
>   EXPECT_EQ(H264SpsPpsTracker::kInsert,
>             tracker_.CopyAndFixBitstream(&idr_packet));
>-  EXPECT_EQ(3u, idr_packet.video_header.codecHeader.H264.nalus_length);
>+  EXPECT_EQ(3u, idr_packet.h264().nalus_length);
>   EXPECT_EQ(320, idr_packet.width);
>   EXPECT_EQ(240, idr_packet.height);
>-  ExpectSpsPpsIdr(idr_packet.video_header.codecHeader.H264, 0, 0);
>+  ExpectSpsPpsIdr(idr_packet.h264(), 0, 0);
> 
>   if (idr_packet.dataPtr != kData) {
>     // In case CopyAndFixBitStream() prepends SPS/PPS nalus to the packet, it
>@@ -317,7 +322,7 @@ TEST_F(TestH264SpsPpsTracker, SpsPpsOutOfBandWrongNaluHeader) {
>   tracker_.InsertSpsPpsNalus(sps, pps);
> 
>   // Insert first packet of the IDR.
>-  VCMPacket idr_packet = GetDefaultPacket();
>+  H264VcmPacket idr_packet;
>   idr_packet.video_header.is_first_packet_in_frame = true;
>   AddIdr(&idr_packet, 0);
>   idr_packet.dataPtr = kData;
>@@ -336,7 +341,7 @@ TEST_F(TestH264SpsPpsTracker, SpsPpsOutOfBandIncompleteNalu) {
>   tracker_.InsertSpsPpsNalus(sps, pps);
> 
>   // Insert first packet of the IDR.
>-  VCMPacket idr_packet = GetDefaultPacket();
>+  H264VcmPacket idr_packet;
>   idr_packet.video_header.is_first_packet_in_frame = true;
>   AddIdr(&idr_packet, 0);
>   idr_packet.dataPtr = kData;
>@@ -350,7 +355,7 @@ TEST_F(TestH264SpsPpsTracker, SaveRestoreWidthHeight) {
> 
>   // Insert an SPS/PPS packet with width/height and make sure
>   // that information is set on the first IDR packet.
>-  VCMPacket sps_pps_packet = GetDefaultPacket();
>+  H264VcmPacket sps_pps_packet;
>   AddSps(&sps_pps_packet, 0, &data);
>   AddPps(&sps_pps_packet, 0, 1, &data);
>   sps_pps_packet.dataPtr = data.data();
>@@ -361,7 +366,7 @@ TEST_F(TestH264SpsPpsTracker, SaveRestoreWidthHeight) {
>             tracker_.CopyAndFixBitstream(&sps_pps_packet));
>   delete[] sps_pps_packet.dataPtr;
> 
>-  VCMPacket idr_packet = GetDefaultPacket();
>+  H264VcmPacket idr_packet;
>   idr_packet.video_header.is_first_packet_in_frame = true;
>   AddIdr(&idr_packet, 1);
>   data.insert(data.end(), {1, 2, 3});
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/include/mock/mock_vcm_callbacks.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/include/mock/mock_vcm_callbacks.h
>index ab67f39cfe7..e604cab5b0f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/include/mock/mock_vcm_callbacks.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/include/mock/mock_vcm_callbacks.h
>@@ -13,7 +13,6 @@
> 
> #include "modules/video_coding/include/video_coding_defines.h"
> #include "test/gmock.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -34,7 +33,7 @@ class MockVCMReceiveCallback : public VCMReceiveCallback {
>   virtual ~MockVCMReceiveCallback() {}
> 
>   MOCK_METHOD3(FrameToRender,
>-               int32_t(VideoFrame&, rtc::Optional<uint8_t>, VideoContentType));
>+               int32_t(VideoFrame&, absl::optional<uint8_t>, VideoContentType));
>   MOCK_METHOD1(ReceivedDecodedReferenceFrame, int32_t(const uint64_t));
>   MOCK_METHOD1(OnIncomingPayloadType, void(int));
>   MOCK_METHOD1(OnDecoderImplementationName, void(const char*));
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/include/mock/mock_video_codec_interface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/include/mock/mock_video_codec_interface.h
>index 2fc3bc47499..c29efeead68 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/include/mock/mock_video_codec_interface.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/include/mock/mock_video_codec_interface.h
>@@ -16,7 +16,6 @@
> 
> #include "modules/video_coding/include/video_codec_interface.h"
> #include "test/gmock.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -46,9 +45,8 @@ class MockVideoEncoder : public VideoEncoder {
>   MOCK_METHOD2(SetChannelParameters, int32_t(uint32_t packetLoss, int64_t rtt));
>   MOCK_METHOD2(SetRates, int32_t(uint32_t newBitRate, uint32_t frameRate));
>   MOCK_METHOD2(SetRateAllocation,
>-               int32_t(const BitrateAllocation& newBitRate,
>+               int32_t(const VideoBitrateAllocation& newBitRate,
>                        uint32_t frameRate));
>-  MOCK_METHOD1(SetPeriodicKeyFrames, int32_t(bool enable));
> };
> 
> class MockDecodedImageCallback : public DecodedImageCallback {
>@@ -59,8 +57,8 @@ class MockDecodedImageCallback : public DecodedImageCallback {
>                        int64_t decode_time_ms));
>   MOCK_METHOD3(Decoded,
>                void(VideoFrame& decodedImage,  // NOLINT
>-                    rtc::Optional<int32_t> decode_time_ms,
>-                    rtc::Optional<uint8_t> qp));
>+                    absl::optional<int32_t> decode_time_ms,
>+                    absl::optional<uint8_t> qp));
>   MOCK_METHOD1(ReceivedDecodedReferenceFrame,
>                int32_t(const uint64_t pictureId));
>   MOCK_METHOD1(ReceivedDecodedFrame, int32_t(const uint64_t pictureId));
>@@ -70,10 +68,9 @@ class MockVideoDecoder : public VideoDecoder {
>  public:
>   MOCK_METHOD2(InitDecode,
>                int32_t(const VideoCodec* codecSettings, int32_t numberOfCores));
>-  MOCK_METHOD5(Decode,
>+  MOCK_METHOD4(Decode,
>                int32_t(const EncodedImage& inputImage,
>                        bool missingFrames,
>-                       const RTPFragmentationHeader* fragmentation,
>                        const CodecSpecificInfo* codecSpecificInfo,
>                        int64_t renderTimeMs));
>   MOCK_METHOD1(RegisterDecodeCompleteCallback,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/include/video_codec_initializer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/include/video_codec_initializer.h
>index 078d7ab3277..ee70810f62e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/include/video_codec_initializer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/include/video_codec_initializer.h
>@@ -15,45 +15,35 @@
> #include <string>
> #include <vector>
> 
>-#include "call/video_send_stream.h"
>+#include "api/video_codecs/video_encoder_config.h"
> 
> namespace webrtc {
> 
>-class TemporalLayersFactory;
> class VideoBitrateAllocator;
> class VideoCodec;
>-class VideoEncoderConfig;
> 
> class VideoCodecInitializer {
>  public:
>-  // Takes an EncoderSettings, a VideoEncoderConfig and the VideoStream
>-  // configuration and translated them into the old school VideoCodec type.
>+  // Takes a VideoEncoderConfig and the VideoStream configuration and
>+  // translates them into the old school VideoCodec type.
>   // It also creates a VideoBitrateAllocator instance, suitable for the codec
>   // type used. For instance, VP8 will create an allocator than can handle
>   // simulcast and temporal layering.
>   // GetBitrateAllocator is called implicitly from here, no need to call again.
>   static bool SetupCodec(
>       const VideoEncoderConfig& config,
>-      const VideoSendStream::Config::EncoderSettings settings,
>       const std::vector<VideoStream>& streams,
>-      bool nack_enabled,
>       VideoCodec* codec,
>       std::unique_ptr<VideoBitrateAllocator>* bitrate_allocator);
> 
>-  // Create a bitrate allocator for the specified codec. |tl_factory| is
>-  // optional, if it is populated, ownership of that instance will be
>-  // transferred to the VideoBitrateAllocator instance.
>+  // Create a bitrate allocator for the specified codec.
>   static std::unique_ptr<VideoBitrateAllocator> CreateBitrateAllocator(
>-      const VideoCodec& codec,
>-      std::unique_ptr<TemporalLayersFactory> tl_factory);
>+      const VideoCodec& codec);
> 
>  private:
>   static VideoCodec VideoEncoderConfigToVideoCodec(
>       const VideoEncoderConfig& config,
>-      const std::vector<VideoStream>& streams,
>-      const std::string& payload_name,
>-      int payload_type,
>-      bool nack_enabled);
>+      const std::vector<VideoStream>& streams);
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/include/video_codec_interface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/include/video_codec_interface.h
>index ef52d8bffae..9108625a546 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/include/video_codec_interface.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/include/video_codec_interface.h
>@@ -19,7 +19,6 @@
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/include/module_common_types.h"
> #include "modules/video_coding/include/video_error_codes.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -28,24 +27,21 @@ class RTPFragmentationHeader;  // forward declaration
> // Note: if any pointers are added to this struct, it must be fitted
> // with a copy-constructor. See below.
> struct CodecSpecificInfoVP8 {
>-  int16_t pictureId;  // Negative value to skip pictureId.
>   bool nonReference;
>   uint8_t simulcastIdx;
>   uint8_t temporalIdx;
>   bool layerSync;
>-  int tl0PicIdx;  // Negative value to skip tl0PicIdx.
>   int8_t keyIdx;  // Negative value to skip keyIdx.
> };
> 
> struct CodecSpecificInfoVP9 {
>-  int16_t picture_id;  // Negative value to skip pictureId.
>-
>-  bool inter_pic_predicted;  // This layer frame is dependent on previously
>-                             // coded frame(s).
>+  bool first_frame_in_picture;  // First frame, increment picture_id.
>+  bool inter_pic_predicted;     // This layer frame is dependent on previously
>+                                // coded frame(s).
>   bool flexible_mode;
>   bool ss_data_available;
>+  bool non_ref_for_inter_layer_pred;
> 
>-  int tl0_pic_idx;  // Negative value to skip tl0PicIdx.
>   uint8_t temporal_idx;
>   uint8_t spatial_idx;
>   bool temporal_up_switch;
>@@ -63,6 +59,8 @@ struct CodecSpecificInfoVP9 {
>   // Frame reference data.
>   uint8_t num_ref_pics;
>   uint8_t p_diff[kMaxVp9RefPics];
>+
>+  bool end_of_picture;
> };
> 
> struct CodecSpecificInfoGeneric {
>@@ -71,11 +69,7 @@ struct CodecSpecificInfoGeneric {
> 
> struct CodecSpecificInfoH264 {
>   H264PacketizationMode packetization_mode;
>-};
>-
>-struct CodecSpecificInfoStereo {
>-  VideoCodecType associated_codec_type;
>-  StereoIndices indices;
>+  uint8_t simulcast_idx;
> };
> 
> union CodecSpecificInfoUnion {
>@@ -83,14 +77,15 @@ union CodecSpecificInfoUnion {
>   CodecSpecificInfoVP8 VP8;
>   CodecSpecificInfoVP9 VP9;
>   CodecSpecificInfoH264 H264;
>-  CodecSpecificInfoStereo stereo;
> };
> 
> // Note: if any pointers are added to this struct or its sub-structs, it
> // must be fitted with a copy-constructor. This is because it is copied
> // in the copy-constructor of VCMEncodedFrame.
> struct CodecSpecificInfo {
>-  CodecSpecificInfo() : codecType(kVideoCodecUnknown), codec_name(nullptr) {}
>+  CodecSpecificInfo() : codecType(kVideoCodecGeneric), codec_name(nullptr) {
>+    memset(&codecSpecific, 0, sizeof(codecSpecific));
>+  }
>   VideoCodecType codecType;
>   const char* codec_name;
>   CodecSpecificInfoUnion codecSpecific;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/include/video_coding.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/include/video_coding.h
>index d204f192fed..8ef046aa231 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/include/video_coding.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/include/video_coding.h
>@@ -21,7 +21,9 @@
> #include <windows.h>
> #endif
> 
>+#include "api/fec_controller.h"
> #include "api/video/video_frame.h"
>+#include "api/video_codecs/video_codec.h"
> #include "modules/include/module.h"
> #include "modules/include/module_common_types.h"
> #include "modules/video_coding/include/video_coding_defines.h"
>@@ -31,13 +33,8 @@ namespace webrtc {
> 
> class Clock;
> class EncodedImageCallback;
>-// TODO(pbos): Remove VCMQMSettingsCallback completely. This might be done by
>-// removing the VCM and use VideoSender/VideoReceiver as a public interface
>-// directly.
>-class VCMQMSettingsCallback;
>-class VideoBitrateAllocator;
>-class VideoEncoder;
> class VideoDecoder;
>+class VideoEncoder;
> struct CodecSpecificInfo;
> 
> class EventFactory {
>@@ -49,9 +46,9 @@ class EventFactory {
> 
> class EventFactoryImpl : public EventFactory {
>  public:
>-  virtual ~EventFactoryImpl() {}
>+  ~EventFactoryImpl() override {}
> 
>-  virtual EventWrapper* CreateEvent() { return EventWrapper::Create(); }
>+  EventWrapper* CreateEvent() override;
> };
> 
> // Used to indicate which decode with errors mode should be used.
>@@ -75,8 +72,8 @@ class VideoCodingModule : public Module {
>   static VideoCodingModule* Create(Clock* clock, EventFactory* event_factory);
> 
>   /*
>-  *   Sender
>-  */
>+   *   Sender
>+   */
> 
>   // Registers a codec to be used for encoding. Calling this
>   // API multiple times overwrites any previously registered codecs.
>@@ -113,18 +110,6 @@ class VideoCodingModule : public Module {
>                                           uint8_t payloadType,
>                                           bool internalSource = false) = 0;
> 
>-  // API to get currently configured encoder target bitrate in bits/s.
>-  //
>-  // Return value      : 0,   on success.
>-  //                     < 0, on error.
>-  virtual int Bitrate(unsigned int* bitrate) const = 0;
>-
>-  // API to get currently configured encoder target frame rate.
>-  //
>-  // Return value      : 0,   on success.
>-  //                     < 0, on error.
>-  virtual int FrameRate(unsigned int* framerate) const = 0;
>-
>   // Sets the parameters describing the send channel. These parameters are
>   // inputs to the
>   // Media Optimization inside the VCM and also specifies the target bit rate
>@@ -145,27 +130,6 @@ class VideoCodingModule : public Module {
>                                        uint8_t lossRate,
>                                        int64_t rtt) = 0;
> 
>-  // Sets the parameters describing the receive channel. These parameters are
>-  // inputs to the
>-  // Media Optimization inside the VCM.
>-  //
>-  // Input:
>-  //      - rtt                   : Current round-trip time in ms.
>-  //                                with the most amount available bandwidth in
>-  //                                a conference
>-  //                                scenario
>-  //
>-  // Return value      : VCM_OK, on success.
>-  //                     < 0,    on error.
>-  virtual int32_t SetReceiveChannelParameters(int64_t rtt) = 0;
>-
>-  // Deprecated: This method currently does not have any effect.
>-  // Register a video protection callback which will be called to deliver
>-  // the requested FEC rate and NACK status (on/off).
>-  // TODO(perkj): Remove once no projects use it.
>-  virtual int32_t RegisterProtectionCallback(
>-      VCMProtectionCallback* protection) = 0;
>-
>   // Enable or disable a video protection method.
>   //
>   // Input:
>@@ -218,8 +182,8 @@ class VideoCodingModule : public Module {
>   virtual int32_t EnableFrameDropper(bool enable) = 0;
> 
>   /*
>-  *   Receiver
>-  */
>+   *   Receiver
>+   */
> 
>   // Register possible receive codecs, can be called multiple times for
>   // different codecs.
>@@ -243,18 +207,11 @@ class VideoCodingModule : public Module {
>                                        int32_t numberOfCores,
>                                        bool requireKeyFrame = false) = 0;
> 
>-  // Register an externally defined decoder/renderer object. Can be a decoder
>-  // only or a
>-  // decoder coupled with a renderer. Note that RegisterReceiveCodec must be
>-  // called to
>-  // be used for decoding incoming streams.
>+  // Register an external decoder object.
>   //
>   // Input:
>-  //      - externalDecoder        : The external decoder/renderer object.
>-  //      - payloadType            : The payload type which this decoder should
>-  //      be
>-  //                                 registered to.
>-  //
>+  //      - externalDecoder : Decoder object to be used for decoding frames.
>+  //      - payloadType     : The payload type which this decoder is bound to.
>   virtual void RegisterExternalDecoder(VideoDecoder* externalDecoder,
>                                        uint8_t payloadType) = 0;
> 
>@@ -273,20 +230,6 @@ class VideoCodingModule : public Module {
>   virtual int32_t RegisterReceiveCallback(
>       VCMReceiveCallback* receiveCallback) = 0;
> 
>-  // Register a receive statistics callback which will be called to deliver
>-  // information
>-  // about the video stream received by the receiving side of the VCM, for
>-  // instance the
>-  // average frame rate and bit rate.
>-  //
>-  // Input:
>-  //      - receiveStats  : The callback object to register.
>-  //
>-  // Return value      : VCM_OK, on success.
>-  //                     < 0,    on error.
>-  virtual int32_t RegisterReceiveStatisticsCallback(
>-      VCMReceiveStatisticsCallback* receiveStats) = 0;
>-
>   // Register a frame type request callback. This callback will be called when
>   // the
>   // module needs to request specific frame types from the send side.
>@@ -341,35 +284,6 @@ class VideoCodingModule : public Module {
>                                  size_t payloadLength,
>                                  const WebRtcRTPHeader& rtpInfo) = 0;
> 
>-  // Minimum playout delay (Used for lip-sync). This is the minimum delay
>-  // required
>-  // to sync with audio. Not included in  VideoCodingModule::Delay()
>-  // Defaults to 0 ms.
>-  //
>-  // Input:
>-  //      - minPlayoutDelayMs   : Additional delay in ms.
>-  //
>-  // Return value      : VCM_OK, on success.
>-  //                     < 0,    on error.
>-  virtual int32_t SetMinimumPlayoutDelay(uint32_t minPlayoutDelayMs) = 0;
>-
>-  // Set the time required by the renderer to render a frame.
>-  //
>-  // Input:
>-  //      - timeMS        : The time in ms required by the renderer to render a
>-  //      frame.
>-  //
>-  // Return value      : VCM_OK, on success.
>-  //                     < 0,    on error.
>-  virtual int32_t SetRenderDelay(uint32_t timeMS) = 0;
>-
>-  // The total delay desired by the VCM. Can be less than the minimum
>-  // delay set with SetMinimumPlayoutDelay.
>-  //
>-  // Return value      : Total delay in ms, on success.
>-  //                     < 0,               on error.
>-  virtual int32_t Delay() const = 0;
>-
>   // Robustness APIs
> 
>   // DEPRECATED.
>@@ -389,13 +303,6 @@ class VideoCodingModule : public Module {
>   virtual int SetReceiverRobustnessMode(ReceiverRobustness robustnessMode,
>                                         VCMDecodeErrorMode errorMode) = 0;
> 
>-  // Set the decode error mode. The mode decides which errors (if any) are
>-  // allowed in decodable frames. Note that setting decode_error_mode to
>-  // anything other than kWithErrors without enabling nack will cause
>-  // long-term freezes (resulting from frequent key frame requests) if
>-  // packet loss occurs.
>-  virtual void SetDecodeErrorMode(VCMDecodeErrorMode decode_error_mode) = 0;
>-
>   // Sets the maximum number of sequence numbers that we are allowed to NACK
>   // and the oldest sequence number that we will consider to NACK. If a
>   // sequence number older than |max_packet_age_to_nack| is missing
>@@ -406,14 +313,8 @@ class VideoCodingModule : public Module {
>                                int max_packet_age_to_nack,
>                                int max_incomplete_time_ms) = 0;
> 
>-  // Setting a desired delay to the VCM receiver. Video rendering will be
>-  // delayed by at least desired_delay_ms.
>-  virtual int SetMinReceiverDelay(int desired_delay_ms) = 0;
>-
>   virtual void RegisterPostEncodeImageCallback(
>       EncodedImageCallback* post_encode_callback) = 0;
>-  // Releases pending decode calls, permitting faster thread shutdown.
>-  virtual void TriggerDecoderShutdown() = 0;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/include/video_coding_defines.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/include/video_coding_defines.h
>index 08497b08cb9..7fb6f583647 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/include/video_coding_defines.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/include/video_coding_defines.h
>@@ -18,7 +18,6 @@
> // For EncodedImage
> #include "common_video/include/video_frame.h"
> #include "modules/include/module_common_types.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -70,15 +69,13 @@ struct VCMFrameCount {
> class VCMReceiveCallback {
>  public:
>   virtual int32_t FrameToRender(VideoFrame& videoFrame,  // NOLINT
>-                                rtc::Optional<uint8_t> qp,
>+                                absl::optional<uint8_t> qp,
>                                 VideoContentType content_type) = 0;
> 
>-  virtual int32_t ReceivedDecodedReferenceFrame(const uint64_t pictureId) {
>-    return -1;
>-  }
>+  virtual int32_t ReceivedDecodedReferenceFrame(const uint64_t pictureId);
>   // Called when the current receive codec changes.
>-  virtual void OnIncomingPayloadType(int payload_type) {}
>-  virtual void OnDecoderImplementationName(const char* implementation_name) {}
>+  virtual void OnIncomingPayloadType(int payload_type);
>+  virtual void OnDecoderImplementationName(const char* implementation_name);
> 
>  protected:
>   virtual ~VCMReceiveCallback() {}
>@@ -108,20 +105,6 @@ class VCMReceiveStatisticsCallback {
>   virtual ~VCMReceiveStatisticsCallback() {}
> };
> 
>-// Callback class used for telling the user about how to configure the FEC,
>-// and the rates sent the last second is returned to the VCM.
>-class VCMProtectionCallback {
>- public:
>-  virtual int ProtectionRequest(const FecProtectionParams* delta_params,
>-                                const FecProtectionParams* key_params,
>-                                uint32_t* sent_video_rate_bps,
>-                                uint32_t* sent_nack_rate_bps,
>-                                uint32_t* sent_fec_rate_bps) = 0;
>-
>- protected:
>-  virtual ~VCMProtectionCallback() {}
>-};
>-
> // Callback class used for telling the user about what frame type needed to
> // continue decoding.
> // Typically a key frame when the stream has been corrupted in some way.
>@@ -147,22 +130,6 @@ class VCMPacketRequestCallback {
>   virtual ~VCMPacketRequestCallback() {}
> };
> 
>-class NackSender {
>- public:
>-  virtual void SendNack(const std::vector<uint16_t>& sequence_numbers) = 0;
>-
>- protected:
>-  virtual ~NackSender() {}
>-};
>-
>-class KeyFrameRequestSender {
>- public:
>-  virtual void RequestKeyFrame() = 0;
>-
>- protected:
>-  virtual ~KeyFrameRequestSender() {}
>-};
>-
> }  // namespace webrtc
> 
> #endif  // MODULES_VIDEO_CODING_INCLUDE_VIDEO_CODING_DEFINES_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/inter_frame_delay.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/inter_frame_delay.h
>index 01cfd2e36bb..c66b1c5177f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/inter_frame_delay.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/inter_frame_delay.h
>@@ -11,7 +11,7 @@
> #ifndef MODULES_VIDEO_CODING_INTER_FRAME_DELAY_H_
> #define MODULES_VIDEO_CODING_INTER_FRAME_DELAY_H_
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include <stdint.h>
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/internal_defines.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/internal_defines.h
>index 07c972ac65c..0c06e9a47f9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/internal_defines.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/internal_defines.h
>@@ -11,8 +11,6 @@
> #ifndef MODULES_VIDEO_CODING_INTERNAL_DEFINES_H_
> #define MODULES_VIDEO_CODING_INTERNAL_DEFINES_H_
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>-
> namespace webrtc {
> 
> #define MASK_32_BITS(x) (0xFFFFFFFF & (x))
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/jitter_buffer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/jitter_buffer.cc
>index aefe6aee5f9..03a9845d510 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/jitter_buffer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/jitter_buffer.cc
>@@ -25,6 +25,7 @@
> #include "modules/video_coding/packet.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>+#include "rtc_base/system/fallthrough.h"
> #include "rtc_base/trace_event.h"
> #include "system_wrappers/include/clock.h"
> #include "system_wrappers/include/event_wrapper.h"
>@@ -122,11 +123,16 @@ void FrameList::Reset(UnorderedFrameList* free_frames) {
>   }
> }
> 
>+Vp9SsMap::Vp9SsMap() {}
>+Vp9SsMap::~Vp9SsMap() {}
>+
> bool Vp9SsMap::Insert(const VCMPacket& packet) {
>-  if (!packet.video_header.codecHeader.VP9.ss_data_available)
>+  const auto& vp9_header =
>+      absl::get<RTPVideoHeaderVP9>(packet.video_header.video_type_header);
>+  if (!vp9_header.ss_data_available)
>     return false;
> 
>-  ss_map_[packet.timestamp] = packet.video_header.codecHeader.VP9.gof;
>+  ss_map_[packet.timestamp] = vp9_header.gof;
>   return true;
> }
> 
>@@ -174,7 +180,9 @@ void Vp9SsMap::AdvanceFront(uint32_t timestamp) {
> 
> // TODO(asapersson): Update according to updates in RTP payload profile.
> bool Vp9SsMap::UpdatePacket(VCMPacket* packet) {
>-  uint8_t gof_idx = packet->video_header.codecHeader.VP9.gof_idx;
>+  auto& vp9_header =
>+      absl::get<RTPVideoHeaderVP9>(packet->video_header.video_type_header);
>+  uint8_t gof_idx = vp9_header.gof_idx;
>   if (gof_idx == kNoGofIdx)
>     return false;  // No update needed.
> 
>@@ -185,14 +193,13 @@ bool Vp9SsMap::UpdatePacket(VCMPacket* packet) {
>   if (gof_idx >= it->second.num_frames_in_gof)
>     return false;  // Assume corresponding SS not yet received.
> 
>-  RTPVideoHeaderVP9* vp9 = &packet->video_header.codecHeader.VP9;
>-  vp9->temporal_idx = it->second.temporal_idx[gof_idx];
>-  vp9->temporal_up_switch = it->second.temporal_up_switch[gof_idx];
>+  vp9_header.temporal_idx = it->second.temporal_idx[gof_idx];
>+  vp9_header.temporal_up_switch = it->second.temporal_up_switch[gof_idx];
> 
>   // TODO(asapersson): Set vp9.ref_picture_id[i] and add usage.
>-  vp9->num_ref_pics = it->second.num_ref_pics[gof_idx];
>+  vp9_header.num_ref_pics = it->second.num_ref_pics[gof_idx];
>   for (uint8_t i = 0; i < it->second.num_ref_pics[gof_idx]; ++i) {
>-    vp9->pid_diff[i] = it->second.pid_diff[gof_idx][i];
>+    vp9_header.pid_diff[i] = it->second.pid_diff[gof_idx][i];
>   }
>   return true;
> }
>@@ -749,7 +756,7 @@ VCMFrameBufferEnum VCMJitterBuffer::InsertPacket(const VCMPacket& packet,
>           frame_event_->Set();
>         }
>       }
>-      FALLTHROUGH();
>+      RTC_FALLTHROUGH();
>     }
>     // Note: There is no break here - continuing to kDecodableSession.
>     case kDecodableSession: {
>@@ -760,9 +767,8 @@ VCMFrameBufferEnum VCMJitterBuffer::InsertPacket(const VCMPacket& packet,
>       } else {
>         incomplete_frames_.InsertFrame(frame);
>         // If NACKs are enabled, keyframes are triggered by |GetNackList|.
>-        if (nack_mode_ == kNoNack &&
>-            NonContinuousOrIncompleteDuration() >
>-                90 * kMaxDiscontinuousFramesTime) {
>+        if (nack_mode_ == kNoNack && NonContinuousOrIncompleteDuration() >
>+                                         90 * kMaxDiscontinuousFramesTime) {
>           return kFlushIndicator;
>         }
>       }
>@@ -776,9 +782,8 @@ VCMFrameBufferEnum VCMJitterBuffer::InsertPacket(const VCMPacket& packet,
>       } else {
>         incomplete_frames_.InsertFrame(frame);
>         // If NACKs are enabled, keyframes are triggered by |GetNackList|.
>-        if (nack_mode_ == kNoNack &&
>-            NonContinuousOrIncompleteDuration() >
>-                90 * kMaxDiscontinuousFramesTime) {
>+        if (nack_mode_ == kNoNack && NonContinuousOrIncompleteDuration() >
>+                                         90 * kMaxDiscontinuousFramesTime) {
>           return kFlushIndicator;
>         }
>       }
>@@ -1048,8 +1053,6 @@ bool VCMJitterBuffer::UpdateNackList(uint16_t sequence_number) {
>     for (uint16_t i = latest_received_sequence_number_ + 1;
>          IsNewerSequenceNumber(sequence_number, i); ++i) {
>       missing_sequence_numbers_.insert(missing_sequence_numbers_.end(), i);
>-      TRACE_EVENT_INSTANT1(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "AddNack",
>-                           "seqnum", i);
>     }
>     if (TooLargeNackList() && !HandleTooLargeNackList()) {
>       RTC_LOG(LS_WARNING) << "Requesting key frame due to too large NACK list.";
>@@ -1063,8 +1066,6 @@ bool VCMJitterBuffer::UpdateNackList(uint16_t sequence_number) {
>     }
>   } else {
>     missing_sequence_numbers_.erase(sequence_number);
>-    TRACE_EVENT_INSTANT1(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"), "RemoveNack",
>-                         "seqnum", sequence_number);
>   }
>   return true;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/jitter_buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/jitter_buffer.h
>index 2732af8c0bd..f649adc8b3a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/jitter_buffer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/jitter_buffer.h
>@@ -25,11 +25,9 @@
> #include "modules/video_coding/inter_frame_delay.h"
> #include "modules/video_coding/jitter_buffer_common.h"
> #include "modules/video_coding/jitter_estimator.h"
>-#include "modules/video_coding/nack_module.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/thread_annotations.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -76,6 +74,9 @@ class FrameList
> class Vp9SsMap {
>  public:
>   typedef std::map<uint32_t, GofInfoVP9, TimestampLessThan> SsMap;
>+  Vp9SsMap();
>+  ~Vp9SsMap();
>+
>   bool Insert(const VCMPacket& packet);
>   void Reset();
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/jitter_buffer_common.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/jitter_buffer_common.h
>index 257ebc0eacb..202b4a386dc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/jitter_buffer_common.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/jitter_buffer_common.h
>@@ -11,8 +11,6 @@
> #ifndef MODULES_VIDEO_CODING_JITTER_BUFFER_COMMON_H_
> #define MODULES_VIDEO_CODING_JITTER_BUFFER_COMMON_H_
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>-
> namespace webrtc {
> 
> // Used to estimate rolling average of packets per frame.
>@@ -58,15 +56,6 @@ enum VCMFrameBufferStateEnum {
> };
> 
> enum { kH264StartCodeLengthBytes = 4 };
>-
>-// Used to indicate if a received packet contain a complete NALU (or equivalent)
>-enum VCMNaluCompleteness {
>-  kNaluUnset = 0,     // Packet has not been filled.
>-  kNaluComplete = 1,  // Packet can be decoded as is.
>-  kNaluStart,         // Packet contain beginning of NALU
>-  kNaluIncomplete,    // Packet is not beginning or end of NALU
>-  kNaluEnd,           // Packet is the end of a NALU
>-};
> }  // namespace webrtc
> 
> #endif  // MODULES_VIDEO_CODING_JITTER_BUFFER_COMMON_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/jitter_buffer_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/jitter_buffer_unittest.cc
>index f9dffc03221..d1744d549aa 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/jitter_buffer_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/jitter_buffer_unittest.cc
>@@ -41,6 +41,8 @@ class Vp9SsMapTest : public ::testing::Test {
>   Vp9SsMapTest() : packet_() {}
> 
>   virtual void SetUp() {
>+    auto& vp9_header =
>+        packet_.video_header.video_type_header.emplace<RTPVideoHeaderVP9>();
>     packet_.is_first_packet_in_frame = true;
>     packet_.dataPtr = data_;
>     packet_.sizeBytes = 1400;
>@@ -49,13 +51,13 @@ class Vp9SsMapTest : public ::testing::Test {
>     packet_.markerBit = true;
>     packet_.frameType = kVideoFrameKey;
>     packet_.codec = kVideoCodecVP9;
>-    packet_.video_header.codec = kRtpVideoVp9;
>-    packet_.video_header.codecHeader.VP9.flexible_mode = false;
>-    packet_.video_header.codecHeader.VP9.gof_idx = 0;
>-    packet_.video_header.codecHeader.VP9.temporal_idx = kNoTemporalIdx;
>-    packet_.video_header.codecHeader.VP9.temporal_up_switch = false;
>-    packet_.video_header.codecHeader.VP9.ss_data_available = true;
>-    packet_.video_header.codecHeader.VP9.gof.SetGofInfoVP9(
>+    packet_.video_header.codec = kVideoCodecVP9;
>+    vp9_header.flexible_mode = false;
>+    vp9_header.gof_idx = 0;
>+    vp9_header.temporal_idx = kNoTemporalIdx;
>+    vp9_header.temporal_up_switch = false;
>+    vp9_header.ss_data_available = true;
>+    vp9_header.gof.SetGofInfoVP9(
>         kTemporalStructureMode3);  // kTemporalStructureMode3: 0-2-1-2..
>   }
> 
>@@ -69,7 +71,8 @@ TEST_F(Vp9SsMapTest, Insert) {
> }
> 
> TEST_F(Vp9SsMapTest, Insert_NoSsData) {
>-  packet_.video_header.codecHeader.VP9.ss_data_available = false;
>+  absl::get<RTPVideoHeaderVP9>(packet_.video_header.video_type_header)
>+      .ss_data_available = false;
>   EXPECT_FALSE(map_.Insert(packet_));
> }
> 
>@@ -146,53 +149,57 @@ TEST_F(Vp9SsMapTest, RemoveOld_WithWrap) {
> }
> 
> TEST_F(Vp9SsMapTest, UpdatePacket_NoSsData) {
>-  packet_.video_header.codecHeader.VP9.gof_idx = 0;
>+  absl::get<RTPVideoHeaderVP9>(packet_.video_header.video_type_header).gof_idx =
>+      0;
>   EXPECT_FALSE(map_.UpdatePacket(&packet_));
> }
> 
> TEST_F(Vp9SsMapTest, UpdatePacket_NoGofIdx) {
>   EXPECT_TRUE(map_.Insert(packet_));
>-  packet_.video_header.codecHeader.VP9.gof_idx = kNoGofIdx;
>+  absl::get<RTPVideoHeaderVP9>(packet_.video_header.video_type_header).gof_idx =
>+      kNoGofIdx;
>   EXPECT_FALSE(map_.UpdatePacket(&packet_));
> }
> 
> TEST_F(Vp9SsMapTest, UpdatePacket_InvalidGofIdx) {
>   EXPECT_TRUE(map_.Insert(packet_));
>-  packet_.video_header.codecHeader.VP9.gof_idx = 4;
>+  absl::get<RTPVideoHeaderVP9>(packet_.video_header.video_type_header).gof_idx =
>+      4;
>   EXPECT_FALSE(map_.UpdatePacket(&packet_));
> }
> 
> TEST_F(Vp9SsMapTest, UpdatePacket) {
>+  auto& vp9_header =
>+      absl::get<RTPVideoHeaderVP9>(packet_.video_header.video_type_header);
>   EXPECT_TRUE(map_.Insert(packet_));  // kTemporalStructureMode3: 0-2-1-2..
> 
>-  packet_.video_header.codecHeader.VP9.gof_idx = 0;
>+  vp9_header.gof_idx = 0;
>   EXPECT_TRUE(map_.UpdatePacket(&packet_));
>-  EXPECT_EQ(0, packet_.video_header.codecHeader.VP9.temporal_idx);
>-  EXPECT_FALSE(packet_.video_header.codecHeader.VP9.temporal_up_switch);
>-  EXPECT_EQ(1U, packet_.video_header.codecHeader.VP9.num_ref_pics);
>-  EXPECT_EQ(4, packet_.video_header.codecHeader.VP9.pid_diff[0]);
>+  EXPECT_EQ(0, vp9_header.temporal_idx);
>+  EXPECT_FALSE(vp9_header.temporal_up_switch);
>+  EXPECT_EQ(1U, vp9_header.num_ref_pics);
>+  EXPECT_EQ(4, vp9_header.pid_diff[0]);
> 
>-  packet_.video_header.codecHeader.VP9.gof_idx = 1;
>+  vp9_header.gof_idx = 1;
>   EXPECT_TRUE(map_.UpdatePacket(&packet_));
>-  EXPECT_EQ(2, packet_.video_header.codecHeader.VP9.temporal_idx);
>-  EXPECT_TRUE(packet_.video_header.codecHeader.VP9.temporal_up_switch);
>-  EXPECT_EQ(1U, packet_.video_header.codecHeader.VP9.num_ref_pics);
>-  EXPECT_EQ(1, packet_.video_header.codecHeader.VP9.pid_diff[0]);
>+  EXPECT_EQ(2, vp9_header.temporal_idx);
>+  EXPECT_TRUE(vp9_header.temporal_up_switch);
>+  EXPECT_EQ(1U, vp9_header.num_ref_pics);
>+  EXPECT_EQ(1, vp9_header.pid_diff[0]);
> 
>-  packet_.video_header.codecHeader.VP9.gof_idx = 2;
>+  vp9_header.gof_idx = 2;
>   EXPECT_TRUE(map_.UpdatePacket(&packet_));
>-  EXPECT_EQ(1, packet_.video_header.codecHeader.VP9.temporal_idx);
>-  EXPECT_TRUE(packet_.video_header.codecHeader.VP9.temporal_up_switch);
>-  EXPECT_EQ(1U, packet_.video_header.codecHeader.VP9.num_ref_pics);
>-  EXPECT_EQ(2, packet_.video_header.codecHeader.VP9.pid_diff[0]);
>+  EXPECT_EQ(1, vp9_header.temporal_idx);
>+  EXPECT_TRUE(vp9_header.temporal_up_switch);
>+  EXPECT_EQ(1U, vp9_header.num_ref_pics);
>+  EXPECT_EQ(2, vp9_header.pid_diff[0]);
> 
>-  packet_.video_header.codecHeader.VP9.gof_idx = 3;
>+  vp9_header.gof_idx = 3;
>   EXPECT_TRUE(map_.UpdatePacket(&packet_));
>-  EXPECT_EQ(2, packet_.video_header.codecHeader.VP9.temporal_idx);
>-  EXPECT_FALSE(packet_.video_header.codecHeader.VP9.temporal_up_switch);
>-  EXPECT_EQ(2U, packet_.video_header.codecHeader.VP9.num_ref_pics);
>-  EXPECT_EQ(1, packet_.video_header.codecHeader.VP9.pid_diff[0]);
>-  EXPECT_EQ(2, packet_.video_header.codecHeader.VP9.pid_diff[1]);
>+  EXPECT_EQ(2, vp9_header.temporal_idx);
>+  EXPECT_TRUE(vp9_header.temporal_up_switch);
>+  EXPECT_EQ(1U, vp9_header.num_ref_pics);
>+  EXPECT_EQ(1, vp9_header.pid_diff[0]);
> }
> 
> class TestBasicJitterBuffer : public ::testing::TestWithParam<std::string>,
>@@ -215,8 +222,7 @@ class TestBasicJitterBuffer : public ::testing::TestWithParam<std::string>,
>     clock_.reset(new SimulatedClock(0));
>     jitter_buffer_.reset(new VCMJitterBuffer(
>         clock_.get(),
>-        std::unique_ptr<EventWrapper>(event_factory_.CreateEvent()),
>-        this,
>+        std::unique_ptr<EventWrapper>(event_factory_.CreateEvent()), this,
>         this));
>     jitter_buffer_->Start();
>     seq_num_ = 1234;
>@@ -244,6 +250,8 @@ class TestBasicJitterBuffer : public ::testing::TestWithParam<std::string>,
>     rtpHeader.header.timestamp = timestamp_;
>     rtpHeader.header.markerBit = true;
>     rtpHeader.frameType = kVideoFrameDelta;
>+    rtpHeader.video_header().codec = kVideoCodecGeneric;
>+    rtpHeader.video_header().is_first_packet_in_frame = true;
>     packet_.reset(new VCMPacket(data_, size_, rtpHeader));
>   }
> 
>@@ -333,8 +341,8 @@ class TestRunningJitterBuffer : public ::testing::TestWithParam<std::string>,
>     oldest_packet_to_nack_ = 250;
>     jitter_buffer_ = new VCMJitterBuffer(
>         clock_.get(),
>-        std::unique_ptr<EventWrapper>(event_factory_.CreateEvent()),
>-        this, this);
>+        std::unique_ptr<EventWrapper>(event_factory_.CreateEvent()), this,
>+        this);
>     stream_generator_ = new StreamGenerator(0, clock_->TimeInMilliseconds());
>     jitter_buffer_->Start();
>     jitter_buffer_->SetNackSettings(max_nack_list_size_, oldest_packet_to_nack_,
>@@ -801,6 +809,7 @@ TEST_F(TestBasicJitterBuffer, TestReorderingWithPadding) {
>   rtpHeader.header.sequenceNumber = seq_num_ + 2;
>   rtpHeader.header.timestamp = timestamp_ + (33 * 90);
>   rtpHeader.header.markerBit = false;
>+  rtpHeader.video_header().codec = kVideoCodecGeneric;
>   VCMPacket empty_packet(data_, 0, rtpHeader);
>   EXPECT_EQ(kOldPacket,
>             jitter_buffer_->InsertPacket(empty_packet, &retransmitted));
>@@ -919,25 +928,28 @@ TEST_F(TestBasicJitterBuffer, TestSkipForwardVp9) {
>   //  -------------------------------------------------
>   // |<----------tl0idx:200--------->|<---tl0idx:201---
> 
>+  auto& vp9_header =
>+      packet_->video_header.video_type_header.emplace<RTPVideoHeaderVP9>();
>+
>   bool re = false;
>   packet_->codec = kVideoCodecVP9;
>-  packet_->video_header.codec = kRtpVideoVp9;
>+  packet_->video_header.codec = kVideoCodecVP9;
>   packet_->is_first_packet_in_frame = true;
>   packet_->markerBit = true;
>-  packet_->video_header.codecHeader.VP9.flexible_mode = false;
>-  packet_->video_header.codecHeader.VP9.spatial_idx = 0;
>-  packet_->video_header.codecHeader.VP9.beginning_of_frame = true;
>-  packet_->video_header.codecHeader.VP9.end_of_frame = true;
>-  packet_->video_header.codecHeader.VP9.temporal_up_switch = false;
>+  vp9_header.flexible_mode = false;
>+  vp9_header.spatial_idx = 0;
>+  vp9_header.beginning_of_frame = true;
>+  vp9_header.end_of_frame = true;
>+  vp9_header.temporal_up_switch = false;
> 
>   packet_->seqNum = 65485;
>   packet_->timestamp = 1000;
>   packet_->frameType = kVideoFrameKey;
>-  packet_->video_header.codecHeader.VP9.picture_id = 5;
>-  packet_->video_header.codecHeader.VP9.tl0_pic_idx = 200;
>-  packet_->video_header.codecHeader.VP9.temporal_idx = 0;
>-  packet_->video_header.codecHeader.VP9.ss_data_available = true;
>-  packet_->video_header.codecHeader.VP9.gof.SetGofInfoVP9(
>+  vp9_header.picture_id = 5;
>+  vp9_header.tl0_pic_idx = 200;
>+  vp9_header.temporal_idx = 0;
>+  vp9_header.ss_data_available = true;
>+  vp9_header.gof.SetGofInfoVP9(
>       kTemporalStructureMode3);  // kTemporalStructureMode3: 0-2-1-2..
>   EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_, &re));
> 
>@@ -945,10 +957,10 @@ TEST_F(TestBasicJitterBuffer, TestSkipForwardVp9) {
>   packet_->seqNum = 65489;
>   packet_->timestamp = 13000;
>   packet_->frameType = kVideoFrameDelta;
>-  packet_->video_header.codecHeader.VP9.picture_id = 9;
>-  packet_->video_header.codecHeader.VP9.tl0_pic_idx = 201;
>-  packet_->video_header.codecHeader.VP9.temporal_idx = 0;
>-  packet_->video_header.codecHeader.VP9.ss_data_available = false;
>+  vp9_header.picture_id = 9;
>+  vp9_header.tl0_pic_idx = 201;
>+  vp9_header.temporal_idx = 0;
>+  vp9_header.ss_data_available = false;
>   EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_, &re));
> 
>   VCMEncodedFrame* frame_out = DecodeCompleteFrame();
>@@ -972,31 +984,34 @@ TEST_F(TestBasicJitterBuffer, ReorderedVp9SsData_3TlLayers) {
>   //  --------------------------------
>   // |<--------tl0idx:200--------->|
> 
>+  auto& vp9_header =
>+      packet_->video_header.video_type_header.emplace<RTPVideoHeaderVP9>();
>+
>   bool re = false;
>   packet_->codec = kVideoCodecVP9;
>-  packet_->video_header.codec = kRtpVideoVp9;
>+  packet_->video_header.codec = kVideoCodecVP9;
>   packet_->is_first_packet_in_frame = true;
>   packet_->markerBit = true;
>-  packet_->video_header.codecHeader.VP9.flexible_mode = false;
>-  packet_->video_header.codecHeader.VP9.spatial_idx = 0;
>-  packet_->video_header.codecHeader.VP9.beginning_of_frame = true;
>-  packet_->video_header.codecHeader.VP9.end_of_frame = true;
>-  packet_->video_header.codecHeader.VP9.tl0_pic_idx = 200;
>+  vp9_header.flexible_mode = false;
>+  vp9_header.spatial_idx = 0;
>+  vp9_header.beginning_of_frame = true;
>+  vp9_header.end_of_frame = true;
>+  vp9_header.tl0_pic_idx = 200;
> 
>   packet_->seqNum = 65486;
>   packet_->timestamp = 6000;
>   packet_->frameType = kVideoFrameDelta;
>-  packet_->video_header.codecHeader.VP9.picture_id = 6;
>-  packet_->video_header.codecHeader.VP9.temporal_idx = 2;
>-  packet_->video_header.codecHeader.VP9.temporal_up_switch = true;
>+  vp9_header.picture_id = 6;
>+  vp9_header.temporal_idx = 2;
>+  vp9_header.temporal_up_switch = true;
>   EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_, &re));
> 
>   packet_->seqNum = 65487;
>   packet_->timestamp = 9000;
>   packet_->frameType = kVideoFrameDelta;
>-  packet_->video_header.codecHeader.VP9.picture_id = 7;
>-  packet_->video_header.codecHeader.VP9.temporal_idx = 1;
>-  packet_->video_header.codecHeader.VP9.temporal_up_switch = true;
>+  vp9_header.picture_id = 7;
>+  vp9_header.temporal_idx = 1;
>+  vp9_header.temporal_up_switch = true;
>   EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_, &re));
> 
>   // Insert first frame with SS data.
>@@ -1005,11 +1020,11 @@ TEST_F(TestBasicJitterBuffer, ReorderedVp9SsData_3TlLayers) {
>   packet_->frameType = kVideoFrameKey;
>   packet_->width = 352;
>   packet_->height = 288;
>-  packet_->video_header.codecHeader.VP9.picture_id = 5;
>-  packet_->video_header.codecHeader.VP9.temporal_idx = 0;
>-  packet_->video_header.codecHeader.VP9.temporal_up_switch = false;
>-  packet_->video_header.codecHeader.VP9.ss_data_available = true;
>-  packet_->video_header.codecHeader.VP9.gof.SetGofInfoVP9(
>+  vp9_header.picture_id = 5;
>+  vp9_header.temporal_idx = 0;
>+  vp9_header.temporal_up_switch = false;
>+  vp9_header.ss_data_available = true;
>+  vp9_header.gof.SetGofInfoVP9(
>       kTemporalStructureMode3);  // kTemporalStructureMode3: 0-2-1-2..
>   EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_, &re));
> 
>@@ -1048,33 +1063,36 @@ TEST_F(TestBasicJitterBuffer, ReorderedVp9SsData_2Tl2SLayers) {
>   //  -----------------------------------------
>   // |<-----------tl0idx:200------------>|
> 
>+  auto& vp9_header =
>+      packet_->video_header.video_type_header.emplace<RTPVideoHeaderVP9>();
>+
>   bool re = false;
>   packet_->codec = kVideoCodecVP9;
>-  packet_->video_header.codec = kRtpVideoVp9;
>-  packet_->video_header.codecHeader.VP9.flexible_mode = false;
>-  packet_->video_header.codecHeader.VP9.beginning_of_frame = true;
>-  packet_->video_header.codecHeader.VP9.end_of_frame = true;
>-  packet_->video_header.codecHeader.VP9.tl0_pic_idx = 200;
>+  packet_->video_header.codec = kVideoCodecVP9;
>+  vp9_header.flexible_mode = false;
>+  vp9_header.beginning_of_frame = true;
>+  vp9_header.end_of_frame = true;
>+  vp9_header.tl0_pic_idx = 200;
> 
>   packet_->is_first_packet_in_frame = true;
>   packet_->markerBit = false;
>   packet_->seqNum = 65486;
>   packet_->timestamp = 6000;
>   packet_->frameType = kVideoFrameDelta;
>-  packet_->video_header.codecHeader.VP9.spatial_idx = 0;
>-  packet_->video_header.codecHeader.VP9.picture_id = 6;
>-  packet_->video_header.codecHeader.VP9.temporal_idx = 1;
>-  packet_->video_header.codecHeader.VP9.temporal_up_switch = true;
>+  vp9_header.spatial_idx = 0;
>+  vp9_header.picture_id = 6;
>+  vp9_header.temporal_idx = 1;
>+  vp9_header.temporal_up_switch = true;
>   EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_, &re));
> 
>   packet_->is_first_packet_in_frame = false;
>   packet_->markerBit = true;
>   packet_->seqNum = 65487;
>   packet_->frameType = kVideoFrameDelta;
>-  packet_->video_header.codecHeader.VP9.spatial_idx = 1;
>-  packet_->video_header.codecHeader.VP9.picture_id = 6;
>-  packet_->video_header.codecHeader.VP9.temporal_idx = 1;
>-  packet_->video_header.codecHeader.VP9.temporal_up_switch = true;
>+  vp9_header.spatial_idx = 1;
>+  vp9_header.picture_id = 6;
>+  vp9_header.temporal_idx = 1;
>+  vp9_header.temporal_up_switch = true;
>   EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_, &re));
> 
>   packet_->is_first_packet_in_frame = false;
>@@ -1082,10 +1100,10 @@ TEST_F(TestBasicJitterBuffer, ReorderedVp9SsData_2Tl2SLayers) {
>   packet_->seqNum = 65485;
>   packet_->timestamp = 3000;
>   packet_->frameType = kVideoFrameKey;
>-  packet_->video_header.codecHeader.VP9.spatial_idx = 1;
>-  packet_->video_header.codecHeader.VP9.picture_id = 5;
>-  packet_->video_header.codecHeader.VP9.temporal_idx = 0;
>-  packet_->video_header.codecHeader.VP9.temporal_up_switch = false;
>+  vp9_header.spatial_idx = 1;
>+  vp9_header.picture_id = 5;
>+  vp9_header.temporal_idx = 0;
>+  vp9_header.temporal_up_switch = false;
>   EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_, &re));
> 
>   // Insert first frame with SS data.
>@@ -1095,12 +1113,12 @@ TEST_F(TestBasicJitterBuffer, ReorderedVp9SsData_2Tl2SLayers) {
>   packet_->frameType = kVideoFrameKey;
>   packet_->width = 352;
>   packet_->height = 288;
>-  packet_->video_header.codecHeader.VP9.spatial_idx = 0;
>-  packet_->video_header.codecHeader.VP9.picture_id = 5;
>-  packet_->video_header.codecHeader.VP9.temporal_idx = 0;
>-  packet_->video_header.codecHeader.VP9.temporal_up_switch = false;
>-  packet_->video_header.codecHeader.VP9.ss_data_available = true;
>-  packet_->video_header.codecHeader.VP9.gof.SetGofInfoVP9(
>+  vp9_header.spatial_idx = 0;
>+  vp9_header.picture_id = 5;
>+  vp9_header.temporal_idx = 0;
>+  vp9_header.temporal_up_switch = false;
>+  vp9_header.ss_data_available = true;
>+  vp9_header.gof.SetGofInfoVP9(
>       kTemporalStructureMode2);  // kTemporalStructureMode3: 0-1-0-1..
>   EXPECT_EQ(kCompleteSession, jitter_buffer_->InsertPacket(*packet_, &re));
> 
>@@ -1154,17 +1172,19 @@ TEST_F(TestBasicJitterBuffer, H264InsertStartCode) {
> TEST_F(TestBasicJitterBuffer, SpsAndPpsHandling) {
>   jitter_buffer_->SetDecodeErrorMode(kNoErrors);
> 
>+  auto& h264_header =
>+      packet_->video_header.video_type_header.emplace<RTPVideoHeaderH264>();
>   packet_->timestamp = timestamp_;
>   packet_->frameType = kVideoFrameKey;
>   packet_->is_first_packet_in_frame = true;
>   packet_->markerBit = true;
>   packet_->codec = kVideoCodecH264;
>-  packet_->video_header.codec = kRtpVideoH264;
>-  packet_->video_header.codecHeader.H264.nalu_type = H264::NaluType::kIdr;
>-  packet_->video_header.codecHeader.H264.nalus[0].type = H264::NaluType::kIdr;
>-  packet_->video_header.codecHeader.H264.nalus[0].sps_id = -1;
>-  packet_->video_header.codecHeader.H264.nalus[0].pps_id = 0;
>-  packet_->video_header.codecHeader.H264.nalus_length = 1;
>+  packet_->video_header.codec = kVideoCodecH264;
>+  h264_header.nalu_type = H264::NaluType::kIdr;
>+  h264_header.nalus[0].type = H264::NaluType::kIdr;
>+  h264_header.nalus[0].sps_id = -1;
>+  h264_header.nalus[0].pps_id = 0;
>+  h264_header.nalus_length = 1;
>   bool retransmitted = false;
>   EXPECT_EQ(kCompleteSession,
>             jitter_buffer_->InsertPacket(*packet_, &retransmitted));
>@@ -1179,15 +1199,15 @@ TEST_F(TestBasicJitterBuffer, SpsAndPpsHandling) {
>   packet_->is_first_packet_in_frame = true;
>   packet_->markerBit = false;
>   packet_->codec = kVideoCodecH264;
>-  packet_->video_header.codec = kRtpVideoH264;
>-  packet_->video_header.codecHeader.H264.nalu_type = H264::NaluType::kStapA;
>-  packet_->video_header.codecHeader.H264.nalus[0].type = H264::NaluType::kSps;
>-  packet_->video_header.codecHeader.H264.nalus[0].sps_id = 0;
>-  packet_->video_header.codecHeader.H264.nalus[0].pps_id = -1;
>-  packet_->video_header.codecHeader.H264.nalus[1].type = H264::NaluType::kPps;
>-  packet_->video_header.codecHeader.H264.nalus[1].sps_id = 0;
>-  packet_->video_header.codecHeader.H264.nalus[1].pps_id = 0;
>-  packet_->video_header.codecHeader.H264.nalus_length = 2;
>+  packet_->video_header.codec = kVideoCodecH264;
>+  h264_header.nalu_type = H264::NaluType::kStapA;
>+  h264_header.nalus[0].type = H264::NaluType::kSps;
>+  h264_header.nalus[0].sps_id = 0;
>+  h264_header.nalus[0].pps_id = -1;
>+  h264_header.nalus[1].type = H264::NaluType::kPps;
>+  h264_header.nalus[1].sps_id = 0;
>+  h264_header.nalus[1].pps_id = 0;
>+  h264_header.nalus_length = 2;
>   // Not complete since the marker bit hasn't been received.
>   EXPECT_EQ(kIncomplete,
>             jitter_buffer_->InsertPacket(*packet_, &retransmitted));
>@@ -1198,12 +1218,12 @@ TEST_F(TestBasicJitterBuffer, SpsAndPpsHandling) {
>   packet_->is_first_packet_in_frame = false;
>   packet_->markerBit = true;
>   packet_->codec = kVideoCodecH264;
>-  packet_->video_header.codec = kRtpVideoH264;
>-  packet_->video_header.codecHeader.H264.nalu_type = H264::NaluType::kIdr;
>-  packet_->video_header.codecHeader.H264.nalus[0].type = H264::NaluType::kIdr;
>-  packet_->video_header.codecHeader.H264.nalus[0].sps_id = -1;
>-  packet_->video_header.codecHeader.H264.nalus[0].pps_id = 0;
>-  packet_->video_header.codecHeader.H264.nalus_length = 1;
>+  packet_->video_header.codec = kVideoCodecH264;
>+  h264_header.nalu_type = H264::NaluType::kIdr;
>+  h264_header.nalus[0].type = H264::NaluType::kIdr;
>+  h264_header.nalus[0].sps_id = -1;
>+  h264_header.nalus[0].pps_id = 0;
>+  h264_header.nalus_length = 1;
>   // Complete and decodable since the pps and sps are received in the first
>   // packet of this frame.
>   EXPECT_EQ(kCompleteSession,
>@@ -1220,12 +1240,12 @@ TEST_F(TestBasicJitterBuffer, SpsAndPpsHandling) {
>   packet_->is_first_packet_in_frame = true;
>   packet_->markerBit = true;
>   packet_->codec = kVideoCodecH264;
>-  packet_->video_header.codec = kRtpVideoH264;
>-  packet_->video_header.codecHeader.H264.nalu_type = H264::NaluType::kSlice;
>-  packet_->video_header.codecHeader.H264.nalus[0].type = H264::NaluType::kSlice;
>-  packet_->video_header.codecHeader.H264.nalus[0].sps_id = -1;
>-  packet_->video_header.codecHeader.H264.nalus[0].pps_id = 0;
>-  packet_->video_header.codecHeader.H264.nalus_length = 1;
>+  packet_->video_header.codec = kVideoCodecH264;
>+  h264_header.nalu_type = H264::NaluType::kSlice;
>+  h264_header.nalus[0].type = H264::NaluType::kSlice;
>+  h264_header.nalus[0].sps_id = -1;
>+  h264_header.nalus[0].pps_id = 0;
>+  h264_header.nalus_length = 1;
>   // Complete and decodable since sps, pps and key frame has been received.
>   EXPECT_EQ(kCompleteSession,
>             jitter_buffer_->InsertPacket(*packet_, &retransmitted));
>@@ -2164,6 +2184,7 @@ TEST_F(TestBasicJitterBuffer, H264IncompleteNalu) {
>   timestamp_ += 33 * 90;
>   WebRtcRTPHeader rtpHeader;
>   memset(&rtpHeader, 0, sizeof(rtpHeader));
>+  rtpHeader.video_header().codec = kVideoCodecGeneric;
>   VCMPacket emptypacket(data_, 0, rtpHeader);
>   emptypacket.seqNum = seq_num_;
>   emptypacket.timestamp = timestamp_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/jitter_estimator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/jitter_estimator.cc
>index 41db158cca7..5e754f10885 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/jitter_estimator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/jitter_estimator.cc
>@@ -139,10 +139,10 @@ void VCMJitterEstimator::UpdateEstimate(int64_t frameDelayMS,
>     // Update the variance anyway since we want to capture cases where we only
>     // get
>     // key frames.
>-    _varFrameSize = VCM_MAX(_phi * _varFrameSize +
>-                                (1 - _phi) * (frameSizeBytes - avgFrameSize) *
>-                                    (frameSizeBytes - avgFrameSize),
>-                            1.0);
>+    _varFrameSize = VCM_MAX(
>+        _phi * _varFrameSize + (1 - _phi) * (frameSizeBytes - avgFrameSize) *
>+                                   (frameSizeBytes - avgFrameSize),
>+        1.0);
>   }
> 
>   // Update max frameSize estimate
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/jitter_estimator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/jitter_estimator.h
>index ff11eeb6dd5..aeba3eded33 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/jitter_estimator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/jitter_estimator.h
>@@ -13,7 +13,6 @@
> 
> #include "modules/video_coding/rtt_filter.h"
> #include "rtc_base/rollingaccumulator.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/media_opt_util.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/media_opt_util.cc
>index e3083b0504d..125e6db81d2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/media_opt_util.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/media_opt_util.cc
>@@ -18,10 +18,11 @@
> #include <limits>
> 
> #include "modules/include/module_common_types.h"
>-#include "modules/video_coding/codecs/vp8/include/vp8_common_types.h"
>-#include "modules/video_coding/include/video_coding_defines.h"
> #include "modules/video_coding/fec_rate_table.h"
>+#include "modules/video_coding/include/video_coding_defines.h"
> #include "modules/video_coding/nack_fec_tables.h"
>+#include "modules/video_coding/utility/simulcast_rate_allocator.h"
>+#include "rtc_base/numerics/safe_conversions.h"
> 
> namespace webrtc {
> // Max value of loss rates in off-line model
>@@ -29,6 +30,20 @@ static const int kPacketLossMax = 129;
> 
> namespace media_optimization {
> 
>+VCMProtectionParameters::VCMProtectionParameters()
>+    : rtt(0),
>+      lossPr(0.0f),
>+      bitRate(0.0f),
>+      packetsPerFrame(0.0f),
>+      packetsPerFrameKey(0.0f),
>+      frameRate(0.0f),
>+      keyFrameSize(0.0f),
>+      fecRateDelta(0),
>+      fecRateKey(0),
>+      codecWidth(0),
>+      codecHeight(0),
>+      numLayers(1) {}
>+
> VCMProtectionMethod::VCMProtectionMethod()
>     : _effectivePacketLoss(0),
>       _protectionFactorK(0),
>@@ -40,6 +55,34 @@ VCMProtectionMethod::VCMProtectionMethod()
> 
> VCMProtectionMethod::~VCMProtectionMethod() {}
> 
>+enum VCMProtectionMethodEnum VCMProtectionMethod::Type() const {
>+  return _type;
>+}
>+
>+uint8_t VCMProtectionMethod::RequiredPacketLossER() {
>+  return _effectivePacketLoss;
>+}
>+
>+uint8_t VCMProtectionMethod::RequiredProtectionFactorK() {
>+  return _protectionFactorK;
>+}
>+
>+uint8_t VCMProtectionMethod::RequiredProtectionFactorD() {
>+  return _protectionFactorD;
>+}
>+
>+bool VCMProtectionMethod::RequiredUepProtectionK() {
>+  return _useUepProtectionK;
>+}
>+
>+bool VCMProtectionMethod::RequiredUepProtectionD() {
>+  return _useUepProtectionD;
>+}
>+
>+int VCMProtectionMethod::MaxFramesFec() const {
>+  return 1;
>+}
>+
> VCMNackFecMethod::VCMNackFecMethod(int64_t lowRttNackThresholdMs,
>                                    int64_t highRttNackThresholdMs)
>     : VCMFecMethod(),
>@@ -85,8 +128,8 @@ bool VCMNackFecMethod::ProtectionFactor(
> 
>     // Adjust FEC with NACK on (for delta frame only)
>     // table depends on RTT relative to rttMax (NACK Threshold)
>-    _protectionFactorD = static_cast<uint8_t>(
>-        adjustRtt * static_cast<float>(_protectionFactorD));
>+    _protectionFactorD = rtc::saturated_cast<uint8_t>(
>+        adjustRtt * rtc::saturated_cast<float>(_protectionFactorD));
>     // update FEC rates after applying adjustment
>     VCMFecMethod::UpdateProtectionFactorD(_protectionFactorD);
>   }
>@@ -108,10 +151,10 @@ int VCMNackFecMethod::ComputeMaxFramesFec(
>   // RTP module based on the actual number of packets and the protection factor.
>   float base_layer_framerate =
>       parameters->frameRate /
>-      static_cast<float>(1 << (parameters->numLayers - 1));
>+      rtc::saturated_cast<float>(1 << (parameters->numLayers - 1));
>   int max_frames_fec = std::max(
>-      static_cast<int>(2.0f * base_layer_framerate * parameters->rtt / 1000.0f +
>-                       0.5f),
>+      rtc::saturated_cast<int>(
>+          2.0f * base_layer_framerate * parameters->rtt / 1000.0f + 0.5f),
>       1);
>   // |kUpperLimitFramesFec| is the upper limit on how many frames we
>   // allow any FEC to be based on.
>@@ -226,9 +269,9 @@ uint8_t VCMFecMethod::BoostCodeRateKey(uint8_t packetFrameDelta,
> }
> 
> uint8_t VCMFecMethod::ConvertFECRate(uint8_t codeRateRTP) const {
>-  return static_cast<uint8_t>(VCM_MIN(
>-      255,
>-      (0.5 + 255.0 * codeRateRTP / static_cast<float>(255 - codeRateRTP))));
>+  return rtc::saturated_cast<uint8_t>(
>+      VCM_MIN(255, (0.5 + 255.0 * codeRateRTP /
>+                              rtc::saturated_cast<float>(255 - codeRateRTP))));
> }
> 
> // Update FEC with protectionFactorD
>@@ -245,7 +288,7 @@ bool VCMFecMethod::ProtectionFactor(const VCMProtectionParameters* parameters) {
>   // FEC PROTECTION SETTINGS: varies with packet loss and bitrate
> 
>   // No protection if (filtered) packetLoss is 0
>-  uint8_t packetLoss = static_cast<uint8_t>(255 * parameters->lossPr);
>+  uint8_t packetLoss = rtc::saturated_cast<uint8_t>(255 * parameters->lossPr);
>   if (packetLoss == 0) {
>     _protectionFactorK = 0;
>     _protectionFactorD = 0;
>@@ -256,7 +299,7 @@ bool VCMFecMethod::ProtectionFactor(const VCMProtectionParameters* parameters) {
>   // first partition size, thresholds, table pars, spatial resoln fac.
> 
>   // First partition protection: ~ 20%
>-  uint8_t firstPartitionProt = static_cast<uint8_t>(255 * 0.20);
>+  uint8_t firstPartitionProt = rtc::saturated_cast<uint8_t>(255 * 0.20);
> 
>   // Minimum protection level needed to generate one FEC packet for one
>   // source packet/frame (in RTP sender)
>@@ -272,9 +315,9 @@ bool VCMFecMethod::ProtectionFactor(const VCMProtectionParameters* parameters) {
>   const uint8_t ratePar2 = 49;
> 
>   // Spatial resolution size, relative to a reference size.
>-  float spatialSizeToRef =
>-      static_cast<float>(parameters->codecWidth * parameters->codecHeight) /
>-      (static_cast<float>(704 * 576));
>+  float spatialSizeToRef = rtc::saturated_cast<float>(parameters->codecWidth *
>+                                                      parameters->codecHeight) /
>+                           (rtc::saturated_cast<float>(704 * 576));
>   // resolnFac: This parameter will generally increase/decrease the FEC rate
>   // (for fixed bitRate and packetLoss) based on system size.
>   // Use a smaller exponent (< 1) to control/soften system size effect.
>@@ -283,11 +326,9 @@ bool VCMFecMethod::ProtectionFactor(const VCMProtectionParameters* parameters) {
>   const int bitRatePerFrame = BitsPerFrame(parameters);
> 
>   // Average number of packets per frame (source and fec):
>-  const uint8_t avgTotPackets = static_cast<uint8_t>(
>-      std::min(static_cast<float>(std::numeric_limits<uint8_t>::max()),
>-               1.5f +
>-                   static_cast<float>(bitRatePerFrame) * 1000.0f /
>-                       static_cast<float>(8.0 * _maxPayloadSize)));
>+  const uint8_t avgTotPackets = rtc::saturated_cast<uint8_t>(
>+      1.5f + rtc::saturated_cast<float>(bitRatePerFrame) * 1000.0f /
>+                 rtc::saturated_cast<float>(8.0 * _maxPayloadSize));
> 
>   // FEC rate parameters: for P and I frame
>   uint8_t codeRateDelta = 0;
>@@ -297,8 +338,8 @@ bool VCMFecMethod::ProtectionFactor(const VCMProtectionParameters* parameters) {
>   // The range on the rate index corresponds to rates (bps)
>   // from ~200k to ~8000k, for 30fps
>   const uint16_t effRateFecTable =
>-      static_cast<uint16_t>(resolnFac * bitRatePerFrame);
>-  uint8_t rateIndexTable = static_cast<uint8_t>(
>+      rtc::saturated_cast<uint16_t>(resolnFac * bitRatePerFrame);
>+  uint8_t rateIndexTable = rtc::saturated_cast<uint8_t>(
>       VCM_MAX(VCM_MIN((effRateFecTable - ratePar1) / ratePar1, ratePar2), 0));
> 
>   // Restrict packet loss range to 50:
>@@ -331,12 +372,12 @@ bool VCMFecMethod::ProtectionFactor(const VCMProtectionParameters* parameters) {
>   // The boost factor may depend on several factors: ratio of packet
>   // number of I to P frames, how much protection placed on P frames, etc.
>   const uint8_t packetFrameDelta =
>-      static_cast<uint8_t>(0.5 + parameters->packetsPerFrame);
>+      rtc::saturated_cast<uint8_t>(0.5 + parameters->packetsPerFrame);
>   const uint8_t packetFrameKey =
>-      static_cast<uint8_t>(0.5 + parameters->packetsPerFrameKey);
>+      rtc::saturated_cast<uint8_t>(0.5 + parameters->packetsPerFrameKey);
>   const uint8_t boostKey = BoostCodeRateKey(packetFrameDelta, packetFrameKey);
> 
>-  rateIndexTable = static_cast<uint8_t>(VCM_MAX(
>+  rateIndexTable = rtc::saturated_cast<uint8_t>(VCM_MAX(
>       VCM_MIN(1 + (boostKey * effRateFecTable - ratePar1) / ratePar1, ratePar2),
>       0));
>   uint16_t indexTableKey = rateIndexTable * kPacketLossMax + packetLoss;
>@@ -357,7 +398,7 @@ bool VCMFecMethod::ProtectionFactor(const VCMProtectionParameters* parameters) {
> 
>   // Make sure I frame protection is at least larger than P frame protection,
>   // and at least as high as filtered packet loss.
>-  codeRateKey = static_cast<uint8_t>(
>+  codeRateKey = rtc::saturated_cast<uint8_t>(
>       VCM_MAX(packetLoss, VCM_MAX(boostKeyProt, codeRateKey)));
> 
>   // Check limit on amount of protection for I frame: 50% is max.
>@@ -377,12 +418,14 @@ bool VCMFecMethod::ProtectionFactor(const VCMProtectionParameters* parameters) {
>   // The correction factor (_corrFecCost) attempts to corrects this, at least
>   // for cases of low rates (small #packets) and low protection levels.
> 
>-  float numPacketsFl = 1.0f + (static_cast<float>(bitRatePerFrame) * 1000.0 /
>-                                   static_cast<float>(8.0 * _maxPayloadSize) +
>-                               0.5);
>+  float numPacketsFl =
>+      1.0f + (rtc::saturated_cast<float>(bitRatePerFrame) * 1000.0 /
>+                  rtc::saturated_cast<float>(8.0 * _maxPayloadSize) +
>+              0.5);
> 
>   const float estNumFecGen =
>-      0.5f + static_cast<float>(_protectionFactorD * numPacketsFl / 255.0f);
>+      0.5f +
>+      rtc::saturated_cast<float>(_protectionFactorD * numPacketsFl / 255.0f);
> 
>   // We reduce cost factor (which will reduce overhead for FEC and
>   // hybrid method) and not the protectionFactor.
>@@ -402,7 +445,8 @@ int VCMFecMethod::BitsPerFrame(const VCMProtectionParameters* parameters) {
>   // When temporal layers are available FEC will only be applied on the base
>   // layer.
>   const float bitRateRatio =
>-      kVp8LayerRateAlloction[parameters->numLayers - 1][0];
>+      webrtc::SimulcastRateAllocator::GetTemporalRateAllocation(
>+          parameters->numLayers, 0);
>   float frameRateRatio = powf(1 / 2.0, parameters->numLayers - 1);
>   float bitRate = parameters->bitRate * bitRateRatio;
>   float frameRate = parameters->frameRate * frameRateRatio;
>@@ -413,7 +457,7 @@ int VCMFecMethod::BitsPerFrame(const VCMProtectionParameters* parameters) {
>   if (frameRate < 1.0f)
>     frameRate = 1.0f;
>   // Average bits per frame (units of kbits)
>-  return static_cast<int>(adjustmentFactor * bitRate / frameRate);
>+  return rtc::saturated_cast<int>(adjustmentFactor * bitRate / frameRate);
> }
> 
> bool VCMFecMethod::EffectivePacketLoss(
>@@ -555,8 +599,8 @@ uint8_t VCMLossProtectionLogic::FilteredLoss(int64_t nowMs,
>   UpdateMaxLossHistory(lossPr255, nowMs);
> 
>   // Update the recursive average filter.
>-  _lossPr255.Apply(static_cast<float>(nowMs - _lastPrUpdateT),
>-                   static_cast<float>(lossPr255));
>+  _lossPr255.Apply(rtc::saturated_cast<float>(nowMs - _lastPrUpdateT),
>+                   rtc::saturated_cast<float>(lossPr255));
>   _lastPrUpdateT = nowMs;
> 
>   // Filtered loss: default is received loss (no filtering).
>@@ -566,7 +610,7 @@ uint8_t VCMLossProtectionLogic::FilteredLoss(int64_t nowMs,
>     case kNoFilter:
>       break;
>     case kAvgFilter:
>-      filtered_loss = static_cast<uint8_t>(_lossPr255.filtered() + 0.5);
>+      filtered_loss = rtc::saturated_cast<uint8_t>(_lossPr255.filtered() + 0.5);
>       break;
>     case kMaxFilter:
>       filtered_loss = MaxFilteredLossPr(nowMs);
>@@ -577,7 +621,7 @@ uint8_t VCMLossProtectionLogic::FilteredLoss(int64_t nowMs,
> }
> 
> void VCMLossProtectionLogic::UpdateFilteredLossPr(uint8_t packetLossEnc) {
>-  _lossPr = static_cast<float>(packetLossEnc) / 255.0;
>+  _lossPr = rtc::saturated_cast<float>(packetLossEnc) / 255.0;
> }
> 
> void VCMLossProtectionLogic::UpdateBitRate(float bitRate) {
>@@ -586,15 +630,16 @@ void VCMLossProtectionLogic::UpdateBitRate(float bitRate) {
> 
> void VCMLossProtectionLogic::UpdatePacketsPerFrame(float nPackets,
>                                                    int64_t nowMs) {
>-  _packetsPerFrame.Apply(static_cast<float>(nowMs - _lastPacketPerFrameUpdateT),
>-                         nPackets);
>+  _packetsPerFrame.Apply(
>+      rtc::saturated_cast<float>(nowMs - _lastPacketPerFrameUpdateT), nPackets);
>   _lastPacketPerFrameUpdateT = nowMs;
> }
> 
> void VCMLossProtectionLogic::UpdatePacketsPerFrameKey(float nPackets,
>                                                       int64_t nowMs) {
>   _packetsPerFrameKey.Apply(
>-      static_cast<float>(nowMs - _lastPacketPerFrameUpdateTKey), nPackets);
>+      rtc::saturated_cast<float>(nowMs - _lastPacketPerFrameUpdateTKey),
>+      nPackets);
>   _lastPacketPerFrameUpdateTKey = nowMs;
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/media_opt_util.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/media_opt_util.h
>index c91ab2bbe7c..1e2c197673e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/media_opt_util.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/media_opt_util.h
>@@ -18,7 +18,6 @@
> 
> #include "modules/video_coding/internal_defines.h"
> #include "rtc_base/numerics/exp_filter.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> namespace media_optimization {
>@@ -48,19 +47,7 @@ const int64_t kLowRttNackMs = 20;
> const int kMaxRttDelayThreshold = 500;
> 
> struct VCMProtectionParameters {
>-  VCMProtectionParameters()
>-      : rtt(0),
>-        lossPr(0.0f),
>-        bitRate(0.0f),
>-        packetsPerFrame(0.0f),
>-        packetsPerFrameKey(0.0f),
>-        frameRate(0.0f),
>-        keyFrameSize(0.0f),
>-        fecRateDelta(0),
>-        fecRateKey(0),
>-        codecWidth(0),
>-        codecHeight(0),
>-        numLayers(1) {}
>+  VCMProtectionParameters();
> 
>   int64_t rtt;
>   float lossPr;
>@@ -107,38 +94,38 @@ class VCMProtectionMethod {
>   // Returns the protection type
>   //
>   // Return value                 : The protection type
>-  enum VCMProtectionMethodEnum Type() const { return _type; }
>+  VCMProtectionMethodEnum Type() const;
> 
>   // Returns the effective packet loss for ER, required by this protection
>   // method
>   //
>   // Return value                 : Required effective packet loss
>-  virtual uint8_t RequiredPacketLossER() { return _effectivePacketLoss; }
>+  virtual uint8_t RequiredPacketLossER();
> 
>   // Extracts the FEC protection factor for Key frame, required by this
>   // protection method
>   //
>   // Return value                 : Required protectionFactor for Key frame
>-  virtual uint8_t RequiredProtectionFactorK() { return _protectionFactorK; }
>+  virtual uint8_t RequiredProtectionFactorK();
> 
>   // Extracts the FEC protection factor for Delta frame, required by this
>   // protection method
>   //
>   // Return value                 : Required protectionFactor for delta frame
>-  virtual uint8_t RequiredProtectionFactorD() { return _protectionFactorD; }
>+  virtual uint8_t RequiredProtectionFactorD();
> 
>   // Extracts whether the FEC Unequal protection (UEP) is used for Key frame.
>   //
>   // Return value                 : Required Unequal protection on/off state.
>-  virtual bool RequiredUepProtectionK() { return _useUepProtectionK; }
>+  virtual bool RequiredUepProtectionK();
> 
>   // Extracts whether the the FEC Unequal protection (UEP) is used for Delta
>   // frame.
>   //
>   // Return value                 : Required Unequal protection on/off state.
>-  virtual bool RequiredUepProtectionD() { return _useUepProtectionD; }
>+  virtual bool RequiredUepProtectionD();
> 
>-  virtual int MaxFramesFec() const { return 1; }
>+  virtual int MaxFramesFec() const;
> 
>  protected:
>   uint8_t _effectivePacketLoss;
>@@ -151,14 +138,14 @@ class VCMProtectionMethod {
>   bool _useUepProtectionK;
>   bool _useUepProtectionD;
>   float _corrFecCost;
>-  enum VCMProtectionMethodEnum _type;
>+  VCMProtectionMethodEnum _type;
> };
> 
> class VCMNackMethod : public VCMProtectionMethod {
>  public:
>   VCMNackMethod();
>-  virtual ~VCMNackMethod();
>-  virtual bool UpdateParameters(const VCMProtectionParameters* parameters);
>+  ~VCMNackMethod() override;
>+  bool UpdateParameters(const VCMProtectionParameters* parameters) override;
>   // Get the effective packet loss
>   bool EffectivePacketLoss(const VCMProtectionParameters* parameter);
> };
>@@ -166,8 +153,8 @@ class VCMNackMethod : public VCMProtectionMethod {
> class VCMFecMethod : public VCMProtectionMethod {
>  public:
>   VCMFecMethod();
>-  virtual ~VCMFecMethod();
>-  virtual bool UpdateParameters(const VCMProtectionParameters* parameters);
>+  ~VCMFecMethod() override;
>+  bool UpdateParameters(const VCMProtectionParameters* parameters) override;
>   // Get the effective packet loss for ER
>   bool EffectivePacketLoss(const VCMProtectionParameters* parameters);
>   // Get the FEC protection factors
>@@ -202,14 +189,14 @@ class VCMNackFecMethod : public VCMFecMethod {
>  public:
>   VCMNackFecMethod(int64_t lowRttNackThresholdMs,
>                    int64_t highRttNackThresholdMs);
>-  virtual ~VCMNackFecMethod();
>-  virtual bool UpdateParameters(const VCMProtectionParameters* parameters);
>+  ~VCMNackFecMethod() override;
>+  bool UpdateParameters(const VCMProtectionParameters* parameters) override;
>   // Get the effective packet loss for ER
>   bool EffectivePacketLoss(const VCMProtectionParameters* parameters);
>   // Get the protection factors
>   bool ProtectionFactor(const VCMProtectionParameters* parameters);
>   // Get the max number of frames the FEC is allowed to be based on.
>-  int MaxFramesFec() const;
>+  int MaxFramesFec() const override;
>   // Turn off the FEC based on low bitrate and other factors.
>   bool BitRateTooLowForFec(const VCMProtectionParameters* parameters);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/media_optimization.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/media_optimization.cc
>index 51c5bcb6a6a..5433edbab77 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/media_optimization.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/media_optimization.cc
>@@ -30,8 +30,7 @@ MediaOptimization::MediaOptimization(Clock* clock)
>   memset(incoming_frame_times_, -1, sizeof(incoming_frame_times_));
> }
> 
>-MediaOptimization::~MediaOptimization(void) {
>-}
>+MediaOptimization::~MediaOptimization(void) {}
> 
> void MediaOptimization::Reset() {
>   rtc::CritScope lock(&crit_sect_);
>@@ -90,11 +89,12 @@ uint32_t MediaOptimization::InputFrameRateInternal() {
> }
> 
> void MediaOptimization::UpdateWithEncodedData(
>-    const EncodedImage& encoded_image) {
>-  size_t encoded_length = encoded_image._length;
>+    const size_t encoded_image_length,
>+    const FrameType encoded_image_frametype) {
>+  size_t encoded_length = encoded_image_length;
>   rtc::CritScope lock(&crit_sect_);
>   if (encoded_length > 0) {
>-    const bool delta_frame = encoded_image._frameType != kVideoFrameKey;
>+    const bool delta_frame = encoded_image_frametype != kVideoFrameKey;
>     frame_dropper_->Fill(encoded_length, delta_frame);
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/media_optimization.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/media_optimization.h
>index 5fa025453cd..411f86ce512 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/media_optimization.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/media_optimization.h
>@@ -51,7 +51,8 @@ class MediaOptimization {
>   // Informs Media Optimization of encoded output.
>   // TODO(perkj): Deprecate SetEncodingData once its not used for stats in
>   // VideoStreamEncoder.
>-  void UpdateWithEncodedData(const EncodedImage& encoded_image);
>+  void UpdateWithEncodedData(const size_t encoded_image_length,
>+                             const FrameType encoded_image_frametype);
> 
>   // InputFrameRate 0 = no frame rate estimate available.
>   uint32_t InputFrameRate();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/nack_module.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/nack_module.cc
>index ed2e82a6e19..ab66b2b562f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/nack_module.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/nack_module.cc
>@@ -55,16 +55,13 @@ NackModule::NackModule(Clock* clock,
>   RTC_DCHECK(keyframe_request_sender_);
> }
> 
>-int NackModule::OnReceivedPacket(const VCMPacket& packet) {
>+int NackModule::OnReceivedPacket(uint16_t seq_num, bool is_keyframe) {
>   rtc::CritScope lock(&crit_);
>-  uint16_t seq_num = packet.seqNum;
>   // TODO(philipel): When the packet includes information whether it is
>   //                 retransmitted or not, use that value instead. For
>   //                 now set it to true, which will cause the reordering
>   //                 statistics to never be updated.
>   bool is_retransmitted = true;
>-  bool is_keyframe =
>-      packet.is_first_packet_in_frame && packet.frameType == kVideoFrameKey;
> 
>   if (!initialized_) {
>     newest_seq_num_ = seq_num;
>@@ -111,6 +108,12 @@ int NackModule::OnReceivedPacket(const VCMPacket& packet) {
>   return 0;
> }
> 
>+int NackModule::OnReceivedPacket(const VCMPacket& packet) {
>+  return OnReceivedPacket(
>+      packet.seqNum,
>+      packet.is_first_packet_in_frame && packet.frameType == kVideoFrameKey);
>+}
>+
> void NackModule::ClearUpTo(uint16_t seq_num) {
>   rtc::CritScope lock(&crit_);
>   nack_list_.erase(nack_list_.begin(), nack_list_.lower_bound(seq_num));
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/nack_module.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/nack_module.h
>index 0b91f0b2040..fc2f2a79779 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/nack_module.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/nack_module.h
>@@ -12,12 +12,12 @@
> #define MODULES_VIDEO_CODING_NACK_MODULE_H_
> 
> #include <map>
>-#include <vector>
> #include <set>
>+#include <vector>
> 
> #include "modules/include/module.h"
>+#include "modules/include/module_common_types.h"
> #include "modules/video_coding/histogram.h"
>-#include "modules/video_coding/include/video_coding_defines.h"
> #include "modules/video_coding/packet.h"
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/numerics/sequence_number_util.h"
>@@ -32,6 +32,7 @@ class NackModule : public Module {
>              NackSender* nack_sender,
>              KeyFrameRequestSender* keyframe_request_sender);
> 
>+  int OnReceivedPacket(uint16_t seq_num, bool is_keyframe);
>   int OnReceivedPacket(const VCMPacket& packet);
>   void ClearUpTo(uint16_t seq_num);
>   void UpdateRtt(int64_t rtt_ms);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/packet.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/packet.cc
>index 9ae5ba0ba6c..691ec62011d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/packet.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/packet.cc
>@@ -26,7 +26,7 @@ VCMPacket::VCMPacket()
>       markerBit(false),
>       timesNacked(-1),
>       frameType(kEmptyFrame),
>-      codec(kVideoCodecUnknown),
>+      codec(kVideoCodecGeneric),
>       is_first_packet_in_frame(false),
>       completeNALU(kNaluUnset),
>       insertStartCode(false),
>@@ -49,100 +49,34 @@ VCMPacket::VCMPacket(const uint8_t* ptr,
>       markerBit(rtpHeader.header.markerBit),
>       timesNacked(-1),
>       frameType(rtpHeader.frameType),
>-      codec(kVideoCodecUnknown),
>-      is_first_packet_in_frame(rtpHeader.type.Video.is_first_packet_in_frame),
>-      completeNALU(kNaluComplete),
>-      insertStartCode(false),
>-      width(rtpHeader.type.Video.width),
>-      height(rtpHeader.type.Video.height),
>-      video_header(rtpHeader.type.Video) {
>-  CopyCodecSpecifics(rtpHeader.type.Video);
>+      codec(rtpHeader.video_header().codec),
>+      is_first_packet_in_frame(
>+          rtpHeader.video_header().is_first_packet_in_frame),
>+      completeNALU(kNaluIncomplete),
>+      insertStartCode(rtpHeader.video_header().codec == kVideoCodecH264 &&
>+                      rtpHeader.video_header().is_first_packet_in_frame),
>+      width(rtpHeader.video_header().width),
>+      height(rtpHeader.video_header().height),
>+      video_header(rtpHeader.video_header()) {
>+  if (is_first_packet_in_frame && markerBit) {
>+    completeNALU = kNaluComplete;
>+  } else if (is_first_packet_in_frame) {
>+    completeNALU = kNaluStart;
>+  } else if (markerBit) {
>+    completeNALU = kNaluEnd;
>+  } else {
>+    completeNALU = kNaluIncomplete;
>+  }
> 
>   if (markerBit) {
>-    video_header.rotation = rtpHeader.type.Video.rotation;
>+    video_header.rotation = rtpHeader.video_header().rotation;
>   }
>   // Playout decisions are made entirely based on first packet in a frame.
>   if (is_first_packet_in_frame) {
>-    video_header.playout_delay = rtpHeader.type.Video.playout_delay;
>+    video_header.playout_delay = rtpHeader.video_header().playout_delay;
>   } else {
>     video_header.playout_delay = {-1, -1};
>   }
> }
> 
>-void VCMPacket::Reset() {
>-  payloadType = 0;
>-  timestamp = 0;
>-  ntp_time_ms_ = 0;
>-  seqNum = 0;
>-  dataPtr = NULL;
>-  sizeBytes = 0;
>-  markerBit = false;
>-  timesNacked = -1;
>-  frameType = kEmptyFrame;
>-  codec = kVideoCodecUnknown;
>-  is_first_packet_in_frame = false;
>-  completeNALU = kNaluUnset;
>-  insertStartCode = false;
>-  width = 0;
>-  height = 0;
>-  memset(&video_header, 0, sizeof(RTPVideoHeader));
>-}
>-
>-void VCMPacket::CopyCodecSpecifics(const RTPVideoHeader& videoHeader) {
>-  switch (videoHeader.codec) {
>-    case kRtpVideoVp8:
>-      // Handle all packets within a frame as depending on the previous packet
>-      // TODO(holmer): This should be changed to make fragments independent
>-      // when the VP8 RTP receiver supports fragments.
>-      if (is_first_packet_in_frame && markerBit)
>-        completeNALU = kNaluComplete;
>-      else if (is_first_packet_in_frame)
>-        completeNALU = kNaluStart;
>-      else if (markerBit)
>-        completeNALU = kNaluEnd;
>-      else
>-        completeNALU = kNaluIncomplete;
>-
>-      codec = kVideoCodecVP8;
>-      return;
>-    case kRtpVideoVp9:
>-      if (is_first_packet_in_frame && markerBit)
>-        completeNALU = kNaluComplete;
>-      else if (is_first_packet_in_frame)
>-        completeNALU = kNaluStart;
>-      else if (markerBit)
>-        completeNALU = kNaluEnd;
>-      else
>-        completeNALU = kNaluIncomplete;
>-
>-      codec = kVideoCodecVP9;
>-      return;
>-    case kRtpVideoH264:
>-      is_first_packet_in_frame = videoHeader.is_first_packet_in_frame;
>-      if (is_first_packet_in_frame)
>-        insertStartCode = true;
>-
>-      if (is_first_packet_in_frame && markerBit) {
>-        completeNALU = kNaluComplete;
>-      } else if (is_first_packet_in_frame) {
>-        completeNALU = kNaluStart;
>-      } else if (markerBit) {
>-        completeNALU = kNaluEnd;
>-      } else {
>-        completeNALU = kNaluIncomplete;
>-      }
>-      codec = kVideoCodecH264;
>-      return;
>-    case kRtpVideoStereo:
>-      codec = kVideoCodecStereo;
>-      return;
>-    case kRtpVideoGeneric:
>-      codec = kVideoCodecGeneric;
>-      return;
>-    case kRtpVideoNone:
>-      codec = kVideoCodecUnknown;
>-      return;
>-  }
>-}
>-
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/packet.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/packet.h
>index e377fb975eb..80a39cabbe0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/packet.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/packet.h
>@@ -12,8 +12,6 @@
> #define MODULES_VIDEO_CODING_PACKET_H_
> 
> #include "modules/include/module_common_types.h"
>-#include "modules/video_coding/jitter_buffer_common.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -24,8 +22,6 @@ class VCMPacket {
>             const size_t size,
>             const WebRtcRTPHeader& rtpHeader);
> 
>-  void Reset();
>-
>   uint8_t payloadType;
>   uint32_t timestamp;
>   // NTP time of the capture time in local timebase in milliseconds.
>@@ -48,9 +44,6 @@ class VCMPacket {
>   RTPVideoHeader video_header;
> 
>   int64_t receive_time_ms;
>-
>- protected:
>-  void CopyCodecSpecifics(const RTPVideoHeader& videoHeader);
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/packet_buffer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/packet_buffer.cc
>index 741c11e2aac..e1172b37496 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/packet_buffer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/packet_buffer.cc
>@@ -12,7 +12,6 @@
> 
> #include <algorithm>
> #include <limits>
>-#include <sstream>
> #include <utility>
> 
> #include "common_video/h264/h264_common.h"
>@@ -48,6 +47,7 @@ PacketBuffer::PacketBuffer(Clock* clock,
>       data_buffer_(start_buffer_size),
>       sequence_buffer_(start_buffer_size),
>       received_frame_callback_(received_frame_callback),
>+      unique_frames_seen_(0),
>       sps_pps_idr_is_h264_keyframe_(
>           field_trial::IsEnabled("WebRTC-SpsPpsIdrIsH264Keyframe")) {
>   RTC_DCHECK_LE(start_buffer_size, max_buffer_size);
>@@ -65,6 +65,8 @@ bool PacketBuffer::InsertPacket(VCMPacket* packet) {
>   {
>     rtc::CritScope lock(&crit_);
> 
>+    OnTimestampReceived(packet->timestamp);
>+
>     uint16_t seq_num = packet->seqNum;
>     size_t index = seq_num % size_;
> 
>@@ -197,16 +199,21 @@ void PacketBuffer::PaddingReceived(uint16_t seq_num) {
>     received_frame_callback_->OnReceivedFrame(std::move(frame));
> }
> 
>-rtc::Optional<int64_t> PacketBuffer::LastReceivedPacketMs() const {
>+absl::optional<int64_t> PacketBuffer::LastReceivedPacketMs() const {
>   rtc::CritScope lock(&crit_);
>   return last_received_packet_ms_;
> }
> 
>-rtc::Optional<int64_t> PacketBuffer::LastReceivedKeyframePacketMs() const {
>+absl::optional<int64_t> PacketBuffer::LastReceivedKeyframePacketMs() const {
>   rtc::CritScope lock(&crit_);
>   return last_received_keyframe_packet_ms_;
> }
> 
>+int PacketBuffer::GetUniqueFramesSeen() const {
>+  rtc::CritScope lock(&crit_);
>+  return unique_frames_seen_;
>+}
>+
> bool PacketBuffer::ExpandBufferSize() {
>   if (size_ == max_size_) {
>     RTC_LOG(LS_WARNING) << "PacketBuffer is already at max size (" << max_size_
>@@ -296,14 +303,17 @@ std::vector<std::unique_ptr<RtpFrameObject>> PacketBuffer::FindFrames(
>           break;
> 
>         if (is_h264 && !is_h264_keyframe) {
>-          const RTPVideoHeaderH264& header =
>-              data_buffer_[start_index].video_header.codecHeader.H264;
>-          for (size_t j = 0; j < header.nalus_length; ++j) {
>-            if (header.nalus[j].type == H264::NaluType::kSps) {
>+          const auto* h264_header = absl::get_if<RTPVideoHeaderH264>(
>+              &data_buffer_[start_index].video_header.video_type_header);
>+          if (!h264_header || h264_header->nalus_length >= kMaxNalusPerPacket)
>+            return found_frames;
>+
>+          for (size_t j = 0; j < h264_header->nalus_length; ++j) {
>+            if (h264_header->nalus[j].type == H264::NaluType::kSps) {
>               has_h264_sps = true;
>-            } else if (header.nalus[j].type == H264::NaluType::kPps) {
>+            } else if (h264_header->nalus[j].type == H264::NaluType::kPps) {
>               has_h264_pps = true;
>-            } else if (header.nalus[j].type == H264::NaluType::kIdr) {
>+            } else if (h264_header->nalus[j].type == H264::NaluType::kIdr) {
>               has_h264_idr = true;
>             }
>           }
>@@ -337,17 +347,13 @@ std::vector<std::unique_ptr<RtpFrameObject>> PacketBuffer::FindFrames(
>       if (is_h264) {
>         // Warn if this is an unsafe frame.
>         if (has_h264_idr && (!has_h264_sps || !has_h264_pps)) {
>-          std::stringstream ss;
>-          ss << "Received H.264-IDR frame "
>-             << "(SPS: " << has_h264_sps << ", PPS: " << has_h264_pps << "). ";
>-          if (sps_pps_idr_is_h264_keyframe_) {
>-            ss << "Treating as delta frame since "
>-                  "WebRTC-SpsPpsIdrIsH264Keyframe is enabled.";
>-          } else {
>-            ss << "Treating as key frame since "
>-                  "WebRTC-SpsPpsIdrIsH264Keyframe is disabled.";
>-          }
>-          RTC_LOG(LS_WARNING) << ss.str();
>+          RTC_LOG(LS_WARNING)
>+              << "Received H.264-IDR frame "
>+              << "(SPS: " << has_h264_sps << ", PPS: " << has_h264_pps
>+              << "). Treating as "
>+              << (sps_pps_idr_is_h264_keyframe_ ? "delta" : "key")
>+              << " frame since WebRTC-SpsPpsIdrIsH264Keyframe is "
>+              << (sps_pps_idr_is_h264_keyframe_ ? "enabled." : "disabled");
>         }
> 
>         // Now that we have decided whether to treat this frame as a key frame
>@@ -422,8 +428,8 @@ bool PacketBuffer::GetBitstream(const RtpFrameObject& frame,
>     RTC_DCHECK_EQ(data_buffer_[index].seqNum, sequence_buffer_[index].seq_num);
>     size_t length = data_buffer_[index].sizeBytes;
>     if (destination + length > destination_end) {
>-      RTC_LOG(LS_WARNING) << "Frame (" << frame.picture_id << ":"
>-                          << static_cast<int>(frame.spatial_layer) << ")"
>+      RTC_LOG(LS_WARNING) << "Frame (" << frame.id.picture_id << ":"
>+                          << static_cast<int>(frame.id.spatial_layer) << ")"
>                           << " bitstream buffer is not large enough.";
>       return false;
>     }
>@@ -484,5 +490,18 @@ void PacketBuffer::UpdateMissingPackets(uint16_t seq_num) {
>   }
> }
> 
>+void PacketBuffer::OnTimestampReceived(uint32_t rtp_timestamp) {
>+  const size_t kMaxTimestampsHistory = 1000;
>+  if (rtp_timestamps_history_set_.insert(rtp_timestamp).second) {
>+    rtp_timestamps_history_queue_.push(rtp_timestamp);
>+    ++unique_frames_seen_;
>+    if (rtp_timestamps_history_set_.size() > kMaxTimestampsHistory) {
>+      uint32_t discarded_timestamp = rtp_timestamps_history_queue_.front();
>+      rtp_timestamps_history_set_.erase(discarded_timestamp);
>+      rtp_timestamps_history_queue_.pop();
>+    }
>+  }
>+}
>+
> }  // namespace video_coding
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/packet_buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/packet_buffer.h
>index c1ca7b8d393..51d27828373 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/packet_buffer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/packet_buffer.h
>@@ -12,6 +12,7 @@
> #define MODULES_VIDEO_CODING_PACKET_BUFFER_H_
> 
> #include <memory>
>+#include <queue>
> #include <set>
> #include <vector>
> 
>@@ -29,7 +30,6 @@ class Clock;
> 
> namespace video_coding {
> 
>-class FrameObject;
> class RtpFrameObject;
> 
> // A received frame is a frame which has received all its packets.
>@@ -58,8 +58,11 @@ class PacketBuffer {
>   void PaddingReceived(uint16_t seq_num);
> 
>   // Timestamp (not RTP timestamp) of the last received packet/keyframe packet.
>-  rtc::Optional<int64_t> LastReceivedPacketMs() const;
>-  rtc::Optional<int64_t> LastReceivedKeyframePacketMs() const;
>+  absl::optional<int64_t> LastReceivedPacketMs() const;
>+  absl::optional<int64_t> LastReceivedKeyframePacketMs() const;
>+
>+  // Returns number of different frames seen in the packet buffer
>+  int GetUniqueFramesSeen() const;
> 
>   int AddRef() const;
>   int Release() const;
>@@ -126,6 +129,10 @@ class PacketBuffer {
>   void UpdateMissingPackets(uint16_t seq_num)
>       RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
> 
>+  // Counts unique received timestamps and updates |unique_frames_seen_|.
>+  void OnTimestampReceived(uint32_t rtp_timestamp)
>+      RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
>+
>   rtc::CriticalSection crit_;
> 
>   // Buffer size_ and max_size_ must always be a power of two.
>@@ -152,11 +159,13 @@ class PacketBuffer {
>   OnReceivedFrameCallback* const received_frame_callback_;
> 
>   // Timestamp (not RTP timestamp) of the last received packet/keyframe packet.
>-  rtc::Optional<int64_t> last_received_packet_ms_ RTC_GUARDED_BY(crit_);
>-  rtc::Optional<int64_t> last_received_keyframe_packet_ms_
>+  absl::optional<int64_t> last_received_packet_ms_ RTC_GUARDED_BY(crit_);
>+  absl::optional<int64_t> last_received_keyframe_packet_ms_
>       RTC_GUARDED_BY(crit_);
> 
>-  rtc::Optional<uint16_t> newest_inserted_seq_num_ RTC_GUARDED_BY(crit_);
>+  int unique_frames_seen_ RTC_GUARDED_BY(crit_);
>+
>+  absl::optional<uint16_t> newest_inserted_seq_num_ RTC_GUARDED_BY(crit_);
>   std::set<uint16_t, DescendingSeqNumComp<uint16_t>> missing_packets_
>       RTC_GUARDED_BY(crit_);
> 
>@@ -164,6 +173,11 @@ class PacketBuffer {
>   // RTP timestamp to treat the corresponding frame as a keyframe.
>   const bool sps_pps_idr_is_h264_keyframe_;
> 
>+  // Stores several last seen unique timestamps for quick search.
>+  std::set<uint32_t> rtp_timestamps_history_set_ RTC_GUARDED_BY(crit_);
>+  // Stores the same unique timestamps in the order of insertion.
>+  std::queue<uint32_t> rtp_timestamps_history_queue_ RTC_GUARDED_BY(crit_);
>+
>   mutable volatile int ref_count_ = 0;
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/protection_bitrate_calculator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/protection_bitrate_calculator.h
>deleted file mode 100644
>index bf50589fe8f..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/protection_bitrate_calculator.h
>+++ /dev/null
>@@ -1,78 +0,0 @@
>-/*
>- *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef MODULES_VIDEO_CODING_PROTECTION_BITRATE_CALCULATOR_H_
>-#define MODULES_VIDEO_CODING_PROTECTION_BITRATE_CALCULATOR_H_
>-
>-#include <list>
>-#include <memory>
>-
>-#include "modules/include/module_common_types.h"
>-#include "modules/video_coding/include/video_coding.h"
>-#include "modules/video_coding/media_opt_util.h"
>-#include "rtc_base/criticalsection.h"
>-#include "system_wrappers/include/clock.h"
>-
>-namespace webrtc {
>-
>-// ProtectionBitrateCalculator calculates how much of the allocated network
>-// capacity that can be used by an encoder and how much that
>-// is needed for redundant packets such as FEC and NACK. It uses an
>-// implementation of |VCMProtectionCallback| to set new FEC parameters and get
>-// the bitrate currently used for FEC and NACK.
>-// Usage:
>-// Setup by calling SetProtectionMethod and SetEncodingData.
>-// For each encoded image, call UpdateWithEncodedData.
>-// Each time the bandwidth estimate change, call SetTargetRates. SetTargetRates
>-// will return the bitrate that can be used by an encoder.
>-// A lock is used to protect internal states, so methods can be called on an
>-// arbitrary thread.
>-class ProtectionBitrateCalculator {
>- public:
>-  ProtectionBitrateCalculator(Clock* clock,
>-                              VCMProtectionCallback* protection_callback);
>-  ~ProtectionBitrateCalculator();
>-
>-  void SetProtectionMethod(bool enable_fec, bool enable_nack);
>-
>-  // Informs media optimization of initial encoding state.
>-  void SetEncodingData(size_t width,
>-                       size_t height,
>-                       size_t num_temporal_layers,
>-                       size_t max_payload_size);
>-
>-  // Returns target rate for the encoder given the channel parameters.
>-  // Inputs:  estimated_bitrate_bps - the estimated network bitrate in bits/s.
>-  //          actual_framerate - encoder frame rate.
>-  //          fraction_lost - packet loss rate in % in the network.
>-  //          round_trip_time_ms - round trip time in milliseconds.
>-  uint32_t SetTargetRates(uint32_t estimated_bitrate_bps,
>-                          int actual_framerate,
>-                          uint8_t fraction_lost,
>-                          int64_t round_trip_time_ms);
>-  // Informs of encoded output.
>-  void UpdateWithEncodedData(const EncodedImage& encoded_image);
>-
>- private:
>-  enum { kBitrateAverageWinMs = 1000 };
>-
>-  Clock* const clock_;
>-  VCMProtectionCallback* const protection_callback_;
>-
>-  rtc::CriticalSection crit_sect_;
>-  std::unique_ptr<media_optimization::VCMLossProtectionLogic> loss_prot_logic_
>-      RTC_GUARDED_BY(crit_sect_);
>-  size_t max_payload_size_ RTC_GUARDED_BY(crit_sect_);
>-
>-  RTC_DISALLOW_COPY_AND_ASSIGN(ProtectionBitrateCalculator);
>-};
>-
>-}  // namespace webrtc
>-#endif  // MODULES_VIDEO_CODING_PROTECTION_BITRATE_CALCULATOR_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/qp_parser.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/qp_parser.cc
>index 418b8b7e5ff..2db0d9cbf53 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/qp_parser.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/qp_parser.cc
>@@ -17,13 +17,11 @@
> namespace webrtc {
> 
> bool QpParser::GetQp(const VCMEncodedFrame& frame, int* qp) {
>-    switch (frame.CodecSpecific()->codecType) {
>-#if !RTC_DISABLE_VP8
>+  switch (frame.CodecSpecific()->codecType) {
>     case kVideoCodecVP8:
>       // QP range: [0, 127].
>       return vp8::GetQp(frame.Buffer(), frame.Length(), qp);
>-#endif
>-#if !RTC_DISABLE_VP9
>+#if !defined(RTC_DISABLE_VP9)
>     case kVideoCodecVP9:
>       // QP range: [0, 255].
>       return vp9::GetQp(frame.Buffer(), frame.Length(), qp);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/receiver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/receiver.cc
>index 157cb88c4f9..4c56d12043e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/receiver.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/receiver.cc
>@@ -20,6 +20,7 @@
> #include "modules/video_coding/internal_defines.h"
> #include "modules/video_coding/media_opt_util.h"
> #include "rtc_base/logging.h"
>+#include "rtc_base/numerics/safe_conversions.h"
> #include "rtc_base/trace_event.h"
> #include "system_wrappers/include/clock.h"
> 
>@@ -192,8 +193,8 @@ VCMEncodedFrame* VCMReceiver::FrameForDecoding(uint16_t max_wait_time_ms,
>         static_cast<int32_t>(clock_->TimeInMilliseconds() - start_time_ms);
>     uint16_t new_max_wait_time =
>         static_cast<uint16_t>(VCM_MAX(available_wait_time, 0));
>-    uint32_t wait_time_ms =
>-        timing_->MaxWaitingTime(render_time_ms, clock_->TimeInMilliseconds());
>+    uint32_t wait_time_ms = rtc::saturated_cast<uint32_t>(
>+        timing_->MaxWaitingTime(render_time_ms, clock_->TimeInMilliseconds()));
>     if (new_max_wait_time < wait_time_ms) {
>       // We're not allowed to wait until the frame is supposed to be rendered,
>       // waiting as long as we're allowed to avoid busy looping, and then return
>@@ -271,17 +272,6 @@ VCMDecodeErrorMode VCMReceiver::DecodeErrorMode() const {
>   return jitter_buffer_.decode_error_mode();
> }
> 
>-int VCMReceiver::SetMinReceiverDelay(int desired_delay_ms) {
>-  rtc::CritScope cs(&crit_sect_);
>-  if (desired_delay_ms < 0 || desired_delay_ms > kMaxReceiverDelayMs) {
>-    return -1;
>-  }
>-  max_video_delay_ms_ = desired_delay_ms + kMaxVideoDelayMs;
>-  // Initializing timing to the desired delay.
>-  timing_->set_min_playout_delay(desired_delay_ms);
>-  return 0;
>-}
>-
> void VCMReceiver::RegisterStatsCallback(
>     VCMReceiveStatisticsCallback* callback) {
>   jitter_buffer_.RegisterStatsCallback(callback);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/receiver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/receiver.h
>index 91e264533ea..4f6590d2748 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/receiver.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/receiver.h
>@@ -79,9 +79,6 @@ class VCMReceiver {
>   VCMNackMode NackMode() const;
>   std::vector<uint16_t> NackList(bool* request_key_frame);
> 
>-  // Receiver video delay.
>-  int SetMinReceiverDelay(int desired_delay_ms);
>-
>   // Decoding with errors.
>   void SetDecodeErrorMode(VCMDecodeErrorMode decode_error_mode);
>   VCMDecodeErrorMode DecodeErrorMode() const;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/receiver_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/receiver_unittest.cc
>index 5920f800a5a..1ff866298e2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/receiver_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/receiver_unittest.cc
>@@ -132,7 +132,7 @@ TEST_F(TestVCMReceiver, NonDecodableDuration_OneIncomplete) {
>   const int kMinDelayMs = 500;
>   receiver_.SetNackSettings(kMaxNackListSize, kMaxPacketAgeToNack,
>                             kMaxNonDecodableDuration);
>-  receiver_.SetMinReceiverDelay(kMinDelayMs);
>+  timing_.set_min_playout_delay(kMinDelayMs);
>   int64_t key_frame_inserted = clock_->TimeInMilliseconds();
>   EXPECT_GE(InsertFrame(kVideoFrameKey, true), kNoError);
>   // Insert an incomplete frame.
>@@ -162,7 +162,7 @@ TEST_F(TestVCMReceiver, NonDecodableDuration_NoTrigger) {
>   const int kMinDelayMs = 500;
>   receiver_.SetNackSettings(kMaxNackListSize, kMaxPacketAgeToNack,
>                             kMaxNonDecodableDuration);
>-  receiver_.SetMinReceiverDelay(kMinDelayMs);
>+  timing_.set_min_playout_delay(kMinDelayMs);
>   int64_t key_frame_inserted = clock_->TimeInMilliseconds();
>   EXPECT_GE(InsertFrame(kVideoFrameKey, true), kNoError);
>   // Insert an incomplete frame.
>@@ -194,7 +194,7 @@ TEST_F(TestVCMReceiver, NonDecodableDuration_NoTrigger2) {
>   const int kMinDelayMs = 500;
>   receiver_.SetNackSettings(kMaxNackListSize, kMaxPacketAgeToNack,
>                             kMaxNonDecodableDuration);
>-  receiver_.SetMinReceiverDelay(kMinDelayMs);
>+  timing_.set_min_playout_delay(kMinDelayMs);
>   int64_t key_frame_inserted = clock_->TimeInMilliseconds();
>   EXPECT_GE(InsertFrame(kVideoFrameKey, true), kNoError);
>   // Insert enough frames to have too long non-decodable sequence, except that
>@@ -226,7 +226,7 @@ TEST_F(TestVCMReceiver, NonDecodableDuration_KeyFrameAfterIncompleteFrames) {
>   const int kMinDelayMs = 500;
>   receiver_.SetNackSettings(kMaxNackListSize, kMaxPacketAgeToNack,
>                             kMaxNonDecodableDuration);
>-  receiver_.SetMinReceiverDelay(kMinDelayMs);
>+  timing_.set_min_playout_delay(kMinDelayMs);
>   int64_t key_frame_inserted = clock_->TimeInMilliseconds();
>   EXPECT_GE(InsertFrame(kVideoFrameKey, true), kNoError);
>   // Insert an incomplete frame.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/rtp_frame_reference_finder.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/rtp_frame_reference_finder.cc
>index 0aaeaec79fc..217ec047d9c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/rtp_frame_reference_finder.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/rtp_frame_reference_finder.cc
>@@ -13,10 +13,12 @@
> #include <algorithm>
> #include <limits>
> 
>+#include "absl/types/variant.h"
> #include "modules/video_coding/frame_object.h"
> #include "modules/video_coding/packet_buffer.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>+#include "rtc_base/system/fallthrough.h"
> 
> namespace webrtc {
> namespace video_coding {
>@@ -71,7 +73,7 @@ void RtpFrameReferenceFinder::RetryStashedFrames() {
>         case kHandOff:
>           complete_frame = true;
>           frame_callback_->OnCompleteFrame(std::move(*frame_it));
>-          FALLTHROUGH();
>+          RTC_FALLTHROUGH();
>         case kDrop:
>           frame_it = stashed_frames_.erase(frame_it);
>       }
>@@ -82,30 +84,13 @@ void RtpFrameReferenceFinder::RetryStashedFrames() {
> RtpFrameReferenceFinder::FrameDecision
> RtpFrameReferenceFinder::ManageFrameInternal(RtpFrameObject* frame) {
>   switch (frame->codec_type()) {
>-    case kVideoCodecFlexfec:
>-    case kVideoCodecULPFEC:
>-    case kVideoCodecRED:
>-      RTC_NOTREACHED();
>-      break;
>     case kVideoCodecVP8:
>       return ManageFrameVp8(frame);
>     case kVideoCodecVP9:
>       return ManageFrameVp9(frame);
>-    // Since the EndToEndTests use kVicdeoCodecUnknow we treat it the same as
>-    // kVideoCodecGeneric.
>-    // TODO(philipel): Take a look at the EndToEndTests and see if maybe they
>-    //                 should be changed to use kVideoCodecGeneric instead.
>-    case kVideoCodecUnknown:
>-    case kVideoCodecH264:
>-    case kVideoCodecI420:
>-    case kVideoCodecStereo:
>-    case kVideoCodecGeneric:
>+    default:
>       return ManageFrameGeneric(frame, kNoPictureId);
>   }
>-
>-  // If not all code paths return a value it makes the win compiler sad.
>-  RTC_NOTREACHED();
>-  return kDrop;
> }
> 
> void RtpFrameReferenceFinder::PaddingReceived(uint16_t seq_num) {
>@@ -177,9 +162,9 @@ RtpFrameReferenceFinder::ManageFrameGeneric(RtpFrameObject* frame,
>     if (last_unwrap_ == -1)
>       last_unwrap_ = picture_id;
> 
>-    frame->picture_id = unwrapper_.Unwrap(picture_id);
>+    frame->id.picture_id = unwrapper_.Unwrap(picture_id);
>     frame->num_references = frame->frame_type() == kVideoFrameKey ? 0 : 1;
>-    frame->references[0] = frame->picture_id - 1;
>+    frame->references[0] = frame->id.picture_id - 1;
>     return kHandOff;
>   }
> 
>@@ -228,30 +213,31 @@ RtpFrameReferenceFinder::ManageFrameGeneric(RtpFrameObject* frame,
> 
>   // Since keyframes can cause reordering we can't simply assign the
>   // picture id according to some incrementing counter.
>-  frame->picture_id = frame->last_seq_num();
>+  frame->id.picture_id = frame->last_seq_num();
>   frame->num_references = frame->frame_type() == kVideoFrameDelta;
>   frame->references[0] = generic_unwrapper_.Unwrap(last_picture_id_gop);
>-  if (AheadOf<uint16_t>(frame->picture_id, last_picture_id_gop)) {
>-    seq_num_it->second.first = frame->picture_id;
>-    seq_num_it->second.second = frame->picture_id;
>+  if (AheadOf<uint16_t>(frame->id.picture_id, last_picture_id_gop)) {
>+    seq_num_it->second.first = frame->id.picture_id;
>+    seq_num_it->second.second = frame->id.picture_id;
>   }
> 
>-  last_picture_id_ = frame->picture_id;
>-  UpdateLastPictureIdWithPadding(frame->picture_id);
>-  frame->picture_id = generic_unwrapper_.Unwrap(frame->picture_id);
>+  last_picture_id_ = frame->id.picture_id;
>+  UpdateLastPictureIdWithPadding(frame->id.picture_id);
>+  frame->id.picture_id = generic_unwrapper_.Unwrap(frame->id.picture_id);
>   return kHandOff;
> }
> 
> RtpFrameReferenceFinder::FrameDecision RtpFrameReferenceFinder::ManageFrameVp8(
>     RtpFrameObject* frame) {
>-  rtc::Optional<RTPVideoTypeHeader> rtp_codec_header = frame->GetCodecHeader();
>+  absl::optional<RTPVideoTypeHeader> rtp_codec_header = frame->GetCodecHeader();
>   if (!rtp_codec_header) {
>     RTC_LOG(LS_WARNING)
>         << "Failed to get codec header from frame, dropping frame.";
>     return kDrop;
>   }
> 
>-  const RTPVideoHeaderVP8& codec_header = rtp_codec_header->VP8;
>+  const RTPVideoHeaderVP8& codec_header =
>+      absl::get<RTPVideoHeaderVP8>(*rtp_codec_header);
> 
>   if (codec_header.pictureId == kNoPictureId ||
>       codec_header.temporalIdx == kNoTemporalIdx ||
>@@ -259,45 +245,46 @@ RtpFrameReferenceFinder::FrameDecision RtpFrameReferenceFinder::ManageFrameVp8(
>     return ManageFrameGeneric(std::move(frame), codec_header.pictureId);
>   }
> 
>-  frame->picture_id = codec_header.pictureId % kPicIdLength;
>+  frame->id.picture_id = codec_header.pictureId % kPicIdLength;
> 
>   if (last_unwrap_ == -1)
>     last_unwrap_ = codec_header.pictureId;
> 
>   if (last_picture_id_ == -1)
>-    last_picture_id_ = frame->picture_id;
>+    last_picture_id_ = frame->id.picture_id;
> 
>   // Find if there has been a gap in fully received frames and save the picture
>   // id of those frames in |not_yet_received_frames_|.
>-  if (AheadOf<uint16_t, kPicIdLength>(frame->picture_id, last_picture_id_)) {
>+  if (AheadOf<uint16_t, kPicIdLength>(frame->id.picture_id, last_picture_id_)) {
>     do {
>       last_picture_id_ = Add<kPicIdLength>(last_picture_id_, 1);
>       not_yet_received_frames_.insert(last_picture_id_);
>-    } while (last_picture_id_ != frame->picture_id);
>+    } while (last_picture_id_ != frame->id.picture_id);
>   }
> 
>+  int64_t unwrapped_tl0 = tl0_unwrapper_.Unwrap(codec_header.tl0PicIdx);
>+
>   // Clean up info for base layers that are too old.
>-  uint8_t old_tl0_pic_idx = codec_header.tl0PicIdx - kMaxLayerInfo;
>+  int64_t old_tl0_pic_idx = unwrapped_tl0 - kMaxLayerInfo;
>   auto clean_layer_info_to = layer_info_.lower_bound(old_tl0_pic_idx);
>   layer_info_.erase(layer_info_.begin(), clean_layer_info_to);
> 
>   // Clean up info about not yet received frames that are too old.
>   uint16_t old_picture_id =
>-      Subtract<kPicIdLength>(frame->picture_id, kMaxNotYetReceivedFrames);
>+      Subtract<kPicIdLength>(frame->id.picture_id, kMaxNotYetReceivedFrames);
>   auto clean_frames_to = not_yet_received_frames_.lower_bound(old_picture_id);
>   not_yet_received_frames_.erase(not_yet_received_frames_.begin(),
>                                  clean_frames_to);
> 
>   if (frame->frame_type() == kVideoFrameKey) {
>     frame->num_references = 0;
>-    layer_info_[codec_header.tl0PicIdx].fill(-1);
>-    UpdateLayerInfoVp8(frame, codec_header);
>+    layer_info_[unwrapped_tl0].fill(-1);
>+    UpdateLayerInfoVp8(frame, unwrapped_tl0, codec_header.temporalIdx);
>     return kHandOff;
>   }
> 
>-  auto layer_info_it = layer_info_.find(codec_header.temporalIdx == 0
>-                                            ? codec_header.tl0PicIdx - 1
>-                                            : codec_header.tl0PicIdx);
>+  auto layer_info_it = layer_info_.find(
>+      codec_header.temporalIdx == 0 ? unwrapped_tl0 - 1 : unwrapped_tl0);
> 
>   // If we don't have the base layer frame yet, stash this frame.
>   if (layer_info_it == layer_info_.end())
>@@ -308,12 +295,10 @@ RtpFrameReferenceFinder::FrameDecision RtpFrameReferenceFinder::ManageFrameVp8(
>   // base layer frame.
>   if (codec_header.temporalIdx == 0) {
>     layer_info_it =
>-        layer_info_
>-            .insert(make_pair(codec_header.tl0PicIdx, layer_info_it->second))
>-            .first;
>+        layer_info_.emplace(unwrapped_tl0, layer_info_it->second).first;
>     frame->num_references = 1;
>     frame->references[0] = layer_info_it->second[0];
>-    UpdateLayerInfoVp8(frame, codec_header);
>+    UpdateLayerInfoVp8(frame, unwrapped_tl0, codec_header.temporalIdx);
>     return kHandOff;
>   }
> 
>@@ -322,7 +307,7 @@ RtpFrameReferenceFinder::FrameDecision RtpFrameReferenceFinder::ManageFrameVp8(
>     frame->num_references = 1;
>     frame->references[0] = layer_info_it->second[0];
> 
>-    UpdateLayerInfoVp8(frame, codec_header);
>+    UpdateLayerInfoVp8(frame, unwrapped_tl0, codec_header.temporalIdx);
>     return kHandOff;
>   }
> 
>@@ -338,7 +323,7 @@ RtpFrameReferenceFinder::FrameDecision RtpFrameReferenceFinder::ManageFrameVp8(
>     // a layer sync frame has been received after this frame for the same
>     // base layer frame, drop this frame.
>     if (AheadOf<uint16_t, kPicIdLength>(layer_info_it->second[layer],
>-                                        frame->picture_id)) {
>+                                        frame->id.picture_id)) {
>       return kDrop;
>     }
> 
>@@ -347,14 +332,14 @@ RtpFrameReferenceFinder::FrameDecision RtpFrameReferenceFinder::ManageFrameVp8(
>     auto not_received_frame_it =
>         not_yet_received_frames_.upper_bound(layer_info_it->second[layer]);
>     if (not_received_frame_it != not_yet_received_frames_.end() &&
>-        AheadOf<uint16_t, kPicIdLength>(frame->picture_id,
>+        AheadOf<uint16_t, kPicIdLength>(frame->id.picture_id,
>                                         *not_received_frame_it)) {
>       return kStash;
>     }
> 
>-    if (!(AheadOf<uint16_t, kPicIdLength>(frame->picture_id,
>+    if (!(AheadOf<uint16_t, kPicIdLength>(frame->id.picture_id,
>                                           layer_info_it->second[layer]))) {
>-      RTC_LOG(LS_WARNING) << "Frame with picture id " << frame->picture_id
>+      RTC_LOG(LS_WARNING) << "Frame with picture id " << frame->id.picture_id
>                           << " and packet range [" << frame->first_seq_num()
>                           << ", " << frame->last_seq_num()
>                           << "] already received, "
>@@ -366,156 +351,161 @@ RtpFrameReferenceFinder::FrameDecision RtpFrameReferenceFinder::ManageFrameVp8(
>     frame->references[layer] = layer_info_it->second[layer];
>   }
> 
>-  UpdateLayerInfoVp8(frame, codec_header);
>+  UpdateLayerInfoVp8(frame, unwrapped_tl0, codec_header.temporalIdx);
>   return kHandOff;
> }
> 
>-void RtpFrameReferenceFinder::UpdateLayerInfoVp8(
>-    RtpFrameObject* frame,
>-    const RTPVideoHeaderVP8& codec_header) {
>-  uint8_t tl0_pic_idx = codec_header.tl0PicIdx;
>-  uint8_t temporal_index = codec_header.temporalIdx;
>-  auto layer_info_it = layer_info_.find(tl0_pic_idx);
>+void RtpFrameReferenceFinder::UpdateLayerInfoVp8(RtpFrameObject* frame,
>+                                                 int64_t unwrapped_tl0,
>+                                                 uint8_t temporal_idx) {
>+  auto layer_info_it = layer_info_.find(unwrapped_tl0);
> 
>   // Update this layer info and newer.
>   while (layer_info_it != layer_info_.end()) {
>-    if (layer_info_it->second[temporal_index] != -1 &&
>-        AheadOf<uint16_t, kPicIdLength>(layer_info_it->second[temporal_index],
>-                                        frame->picture_id)) {
>+    if (layer_info_it->second[temporal_idx] != -1 &&
>+        AheadOf<uint16_t, kPicIdLength>(layer_info_it->second[temporal_idx],
>+                                        frame->id.picture_id)) {
>       // The frame was not newer, then no subsequent layer info have to be
>       // update.
>       break;
>     }
> 
>-    layer_info_it->second[codec_header.temporalIdx] = frame->picture_id;
>-    ++tl0_pic_idx;
>-    layer_info_it = layer_info_.find(tl0_pic_idx);
>+    layer_info_it->second[temporal_idx] = frame->id.picture_id;
>+    ++unwrapped_tl0;
>+    layer_info_it = layer_info_.find(unwrapped_tl0);
>   }
>-  not_yet_received_frames_.erase(frame->picture_id);
>+  not_yet_received_frames_.erase(frame->id.picture_id);
> 
>   UnwrapPictureIds(frame);
> }
> 
> RtpFrameReferenceFinder::FrameDecision RtpFrameReferenceFinder::ManageFrameVp9(
>     RtpFrameObject* frame) {
>-  rtc::Optional<RTPVideoTypeHeader> rtp_codec_header = frame->GetCodecHeader();
>+  absl::optional<RTPVideoTypeHeader> rtp_codec_header = frame->GetCodecHeader();
>   if (!rtp_codec_header) {
>     RTC_LOG(LS_WARNING)
>         << "Failed to get codec header from frame, dropping frame.";
>     return kDrop;
>   }
> 
>-  const RTPVideoHeaderVP9& codec_header = rtp_codec_header->VP9;
>+  const RTPVideoHeaderVP9& codec_header =
>+      absl::get<RTPVideoHeaderVP9>(*rtp_codec_header);
> 
>   if (codec_header.picture_id == kNoPictureId ||
>-      codec_header.temporal_idx == kNoTemporalIdx) {
>+      codec_header.temporal_idx == kNoTemporalIdx ||
>+      codec_header.tl0_pic_idx == kNoTl0PicIdx) {
>     return ManageFrameGeneric(std::move(frame), codec_header.picture_id);
>   }
> 
>-  frame->spatial_layer = codec_header.spatial_idx;
>+  frame->id.spatial_layer = codec_header.spatial_idx;
>   frame->inter_layer_predicted = codec_header.inter_layer_predicted;
>-  frame->picture_id = codec_header.picture_id % kPicIdLength;
>+  frame->id.picture_id = codec_header.picture_id % kPicIdLength;
> 
>   if (last_unwrap_ == -1)
>     last_unwrap_ = codec_header.picture_id;
> 
>   if (last_picture_id_ == -1)
>-    last_picture_id_ = frame->picture_id;
>+    last_picture_id_ = frame->id.picture_id;
> 
>   if (codec_header.flexible_mode) {
>     frame->num_references = codec_header.num_ref_pics;
>     for (size_t i = 0; i < frame->num_references; ++i) {
>-      frame->references[i] =
>-          Subtract<kPicIdLength>(frame->picture_id, codec_header.pid_diff[i]);
>+      frame->references[i] = Subtract<kPicIdLength>(frame->id.picture_id,
>+                                                    codec_header.pid_diff[i]);
>     }
> 
>     UnwrapPictureIds(frame);
>     return kHandOff;
>   }
> 
>+  GofInfo* info;
>+  int64_t unwrapped_tl0 = tl0_unwrapper_.Unwrap(codec_header.tl0_pic_idx);
>   if (codec_header.ss_data_available) {
>-    // Scalability structures can only be sent with tl0 frames.
>     if (codec_header.temporal_idx != 0) {
>-      RTC_LOG(LS_WARNING)
>-          << "Received scalability structure on a non base layer"
>-             " frame. Scalability structure ignored.";
>+      RTC_LOG(LS_WARNING) << "Received scalability structure on a non base "
>+                             "layer frame. Scalability structure ignored.";
>     } else {
>       current_ss_idx_ = Add<kMaxGofSaved>(current_ss_idx_, 1);
>-      scalability_structures_[current_ss_idx_] = codec_header.gof;
>-      scalability_structures_[current_ss_idx_].pid_start = frame->picture_id;
>+      if (codec_header.gof.num_frames_in_gof == 0 ||
>+          codec_header.gof.num_frames_in_gof > kMaxVp9FramesInGof) {
>+        return kDrop;
>+      }
> 
>-      GofInfo info(&scalability_structures_[current_ss_idx_],
>-                   frame->picture_id);
>-      gof_info_.insert(std::make_pair(codec_header.tl0_pic_idx, info));
>+      scalability_structures_[current_ss_idx_] = codec_header.gof;
>+      scalability_structures_[current_ss_idx_].pid_start = frame->id.picture_id;
>+      gof_info_.emplace(unwrapped_tl0,
>+                        GofInfo(&scalability_structures_[current_ss_idx_],
>+                                frame->id.picture_id));
>     }
>-  }
> 
>-  // Clean up info for base layers that are too old.
>-  uint8_t old_tl0_pic_idx = codec_header.tl0_pic_idx - kMaxGofSaved;
>-  auto clean_gof_info_to = gof_info_.lower_bound(old_tl0_pic_idx);
>-  gof_info_.erase(gof_info_.begin(), clean_gof_info_to);
>+    const auto gof_info_it = gof_info_.find(unwrapped_tl0);
>+    if (gof_info_it == gof_info_.end())
>+      return kStash;
> 
>-  if (frame->frame_type() == kVideoFrameKey) {
>-    // When using GOF all keyframes must include the scalability structure.
>-    if (!codec_header.ss_data_available)
>+    info = &gof_info_it->second;
>+
>+    if (frame->frame_type() == kVideoFrameKey) {
>+      frame->num_references = 0;
>+      FrameReceivedVp9(frame->id.picture_id, info);
>+      UnwrapPictureIds(frame);
>+      return kHandOff;
>+    }
>+  } else {
>+    if (frame->frame_type() == kVideoFrameKey) {
>       RTC_LOG(LS_WARNING) << "Received keyframe without scalability structure";
>+      return kDrop;
>+    }
> 
>-    frame->num_references = 0;
>-    GofInfo info = gof_info_.find(codec_header.tl0_pic_idx)->second;
>-    FrameReceivedVp9(frame->picture_id, &info);
>-    UnwrapPictureIds(frame);
>-    return kHandOff;
>-  }
>+    auto gof_info_it = gof_info_.find(
>+        (codec_header.temporal_idx == 0) ? unwrapped_tl0 - 1 : unwrapped_tl0);
> 
>-  auto gof_info_it = gof_info_.find(
>-      (codec_header.temporal_idx == 0 && !codec_header.ss_data_available)
>-          ? codec_header.tl0_pic_idx - 1
>-          : codec_header.tl0_pic_idx);
>+    // Gof info for this frame is not available yet, stash this frame.
>+    if (gof_info_it == gof_info_.end())
>+      return kStash;
> 
>-  // Gof info for this frame is not available yet, stash this frame.
>-  if (gof_info_it == gof_info_.end())
>-    return kStash;
>+    if (codec_header.temporal_idx == 0) {
>+      gof_info_it = gof_info_
>+                        .emplace(unwrapped_tl0, GofInfo(gof_info_it->second.gof,
>+                                                        frame->id.picture_id))
>+                        .first;
>+    }
> 
>-  GofInfo* info = &gof_info_it->second;
>-  FrameReceivedVp9(frame->picture_id, info);
>+    info = &gof_info_it->second;
>+  }
>+
>+  // Clean up info for base layers that are too old.
>+  int64_t old_tl0_pic_idx = unwrapped_tl0 - kMaxGofSaved;
>+  auto clean_gof_info_to = gof_info_.lower_bound(old_tl0_pic_idx);
>+  gof_info_.erase(gof_info_.begin(), clean_gof_info_to);
>+
>+  FrameReceivedVp9(frame->id.picture_id, info);
> 
>   // Make sure we don't miss any frame that could potentially have the
>   // up switch flag set.
>-  if (MissingRequiredFrameVp9(frame->picture_id, *info))
>+  if (MissingRequiredFrameVp9(frame->id.picture_id, *info))
>     return kStash;
> 
>-  if (codec_header.temporal_up_switch) {
>-    auto pid_tidx =
>-        std::make_pair(frame->picture_id, codec_header.temporal_idx);
>-    up_switch_.insert(pid_tidx);
>-  }
>-
>-  // If this is a base layer frame that contains a scalability structure
>-  // then gof info has already been inserted earlier, so we only want to
>-  // insert if we haven't done so already.
>-  if (codec_header.temporal_idx == 0 && !codec_header.ss_data_available) {
>-    GofInfo new_info(info->gof, frame->picture_id);
>-    gof_info_.insert(std::make_pair(codec_header.tl0_pic_idx, new_info));
>-  }
>+  if (codec_header.temporal_up_switch)
>+    up_switch_.emplace(frame->id.picture_id, codec_header.temporal_idx);
> 
>   // Clean out old info about up switch frames.
>-  uint16_t old_picture_id = Subtract<kPicIdLength>(frame->picture_id, 50);
>+  uint16_t old_picture_id = Subtract<kPicIdLength>(frame->id.picture_id, 50);
>   auto up_switch_erase_to = up_switch_.lower_bound(old_picture_id);
>   up_switch_.erase(up_switch_.begin(), up_switch_erase_to);
> 
>   size_t diff = ForwardDiff<uint16_t, kPicIdLength>(info->gof->pid_start,
>-                                                    frame->picture_id);
>+                                                    frame->id.picture_id);
>   size_t gof_idx = diff % info->gof->num_frames_in_gof;
> 
>   // Populate references according to the scalability structure.
>   frame->num_references = info->gof->num_ref_pics[gof_idx];
>   for (size_t i = 0; i < frame->num_references; ++i) {
>     frame->references[i] = Subtract<kPicIdLength>(
>-        frame->picture_id, info->gof->pid_diff[gof_idx][i]);
>+        frame->id.picture_id, info->gof->pid_diff[gof_idx][i]);
> 
>     // If this is a reference to a frame earlier than the last up switch point,
>     // then ignore this reference.
>-    if (UpSwitchInIntervalVp9(frame->picture_id, codec_header.temporal_idx,
>+    if (UpSwitchInIntervalVp9(frame->id.picture_id, codec_header.temporal_idx,
>                               frame->references[i])) {
>       --frame->num_references;
>     }
>@@ -532,6 +522,12 @@ bool RtpFrameReferenceFinder::MissingRequiredFrameVp9(uint16_t picture_id,
>   size_t gof_idx = diff % info.gof->num_frames_in_gof;
>   size_t temporal_idx = info.gof->temporal_idx[gof_idx];
> 
>+  if (temporal_idx >= kMaxTemporalLayers) {
>+    RTC_LOG(LS_WARNING) << "At most " << kMaxTemporalLayers << " temporal "
>+                        << "layers are supported.";
>+    return true;
>+  }
>+
>   // For every reference this frame has, check if there is a frame missing in
>   // the interval (|ref_pid|, |picture_id|) in any of the lower temporal
>   // layers. If so, we are missing a required frame.
>@@ -553,6 +549,7 @@ bool RtpFrameReferenceFinder::MissingRequiredFrameVp9(uint16_t picture_id,
> void RtpFrameReferenceFinder::FrameReceivedVp9(uint16_t picture_id,
>                                                GofInfo* info) {
>   int last_picture_id = info->last_picture_id;
>+  size_t gof_size = std::min(info->gof->num_frames_in_gof, kMaxVp9FramesInGof);
> 
>   // If there is a gap, find which temporal layer the missing frames
>   // belong to and add the frame as missing for that temporal layer.
>@@ -560,22 +557,38 @@ void RtpFrameReferenceFinder::FrameReceivedVp9(uint16_t picture_id,
>   if (AheadOf<uint16_t, kPicIdLength>(picture_id, last_picture_id)) {
>     size_t diff = ForwardDiff<uint16_t, kPicIdLength>(info->gof->pid_start,
>                                                       last_picture_id);
>-    size_t gof_idx = diff % info->gof->num_frames_in_gof;
>+    size_t gof_idx = diff % gof_size;
> 
>     last_picture_id = Add<kPicIdLength>(last_picture_id, 1);
>     while (last_picture_id != picture_id) {
>-      ++gof_idx;
>-      RTC_DCHECK_NE(0ul, gof_idx % info->gof->num_frames_in_gof);
>+      gof_idx = (gof_idx + 1) % gof_size;
>+      RTC_CHECK(gof_idx < kMaxVp9FramesInGof);
>+
>       size_t temporal_idx = info->gof->temporal_idx[gof_idx];
>+      if (temporal_idx >= kMaxTemporalLayers) {
>+        RTC_LOG(LS_WARNING) << "At most " << kMaxTemporalLayers << " temporal "
>+                            << "layers are supported.";
>+        return;
>+      }
>+
>       missing_frames_for_layer_[temporal_idx].insert(last_picture_id);
>       last_picture_id = Add<kPicIdLength>(last_picture_id, 1);
>     }
>+
>     info->last_picture_id = last_picture_id;
>   } else {
>     size_t diff =
>         ForwardDiff<uint16_t, kPicIdLength>(info->gof->pid_start, picture_id);
>-    size_t gof_idx = diff % info->gof->num_frames_in_gof;
>+    size_t gof_idx = diff % gof_size;
>+    RTC_CHECK(gof_idx < kMaxVp9FramesInGof);
>+
>     size_t temporal_idx = info->gof->temporal_idx[gof_idx];
>+    if (temporal_idx >= kMaxTemporalLayers) {
>+      RTC_LOG(LS_WARNING) << "At most " << kMaxTemporalLayers << " temporal "
>+                          << "layers are supported.";
>+      return;
>+    }
>+
>     missing_frames_for_layer_[temporal_idx].erase(picture_id);
>   }
> }
>@@ -597,7 +610,7 @@ bool RtpFrameReferenceFinder::UpSwitchInIntervalVp9(uint16_t picture_id,
> void RtpFrameReferenceFinder::UnwrapPictureIds(RtpFrameObject* frame) {
>   for (size_t i = 0; i < frame->num_references; ++i)
>     frame->references[i] = unwrapper_.Unwrap(frame->references[i]);
>-  frame->picture_id = unwrapper_.Unwrap(frame->picture_id);
>+  frame->id.picture_id = unwrapper_.Unwrap(frame->id.picture_id);
> }
> 
> }  // namespace video_coding
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/rtp_frame_reference_finder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/rtp_frame_reference_finder.h
>index 95da4f327e6..f7e2720e7d2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/rtp_frame_reference_finder.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/rtp_frame_reference_finder.h
>@@ -12,9 +12,9 @@
> #define MODULES_VIDEO_CODING_RTP_FRAME_REFERENCE_FINDER_H_
> 
> #include <array>
>+#include <deque>
> #include <map>
> #include <memory>
>-#include <deque>
> #include <set>
> #include <utility>
> 
>@@ -26,7 +26,7 @@
> namespace webrtc {
> namespace video_coding {
> 
>-class FrameObject;
>+class EncodedFrame;
> class RtpFrameObject;
> 
> // A complete frame is a frame which has received all its packets and all its
>@@ -34,7 +34,7 @@ class RtpFrameObject;
> class OnCompleteFrameCallback {
>  public:
>   virtual ~OnCompleteFrameCallback() {}
>-  virtual void OnCompleteFrame(std::unique_ptr<FrameObject> frame) = 0;
>+  virtual void OnCompleteFrame(std::unique_ptr<EncodedFrame> frame) = 0;
> };
> 
> class RtpFrameReferenceFinder {
>@@ -60,7 +60,7 @@ class RtpFrameReferenceFinder {
>   static const uint16_t kPicIdLength = 1 << 15;
>   static const uint8_t kMaxTemporalLayers = 5;
>   static const int kMaxLayerInfo = 50;
>-  static const int kMaxStashedFrames = 50;
>+  static const int kMaxStashedFrames = 100;
>   static const int kMaxNotYetReceivedFrames = 100;
>   static const int kMaxGofSaved = 50;
>   static const int kMaxPaddingAge = 100;
>@@ -100,7 +100,8 @@ class RtpFrameReferenceFinder {
>   // Updates necessary layer info state used to determine frame references for
>   // Vp8.
>   void UpdateLayerInfoVp8(RtpFrameObject* frame,
>-                          const RTPVideoHeaderVP8& codec_header)
>+                          int64_t unwrapped_tl0,
>+                          uint8_t temporal_idx)
>       RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
> 
>   // Find references for Vp9 frames
>@@ -129,25 +130,6 @@ class RtpFrameReferenceFinder {
>   void UnwrapPictureIds(RtpFrameObject* frame)
>       RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
> 
>-  // Returns true if the frame is old and should be dropped.
>-  // TODO(philipel): Remove when VP9 PID/TL0 does not jump mid-stream (should be
>-  //                 around M59).
>-  bool Vp9PidTl0Fix(const RtpFrameObject& frame,
>-                    int16_t* picture_id,
>-                    int16_t* tl0_pic_idx) RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
>-
>-  // TODO(philipel): Remove when VP9 PID/TL0 does not jump mid-stream (should be
>-  //                 around M59).
>-  bool DetectVp9PicIdJump(int fixed_pid,
>-                          int fixed_tl0,
>-                          uint32_t timestamp) const
>-      RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
>-
>-  // TODO(philipel): Remove when VP9 PID/TL0 does not jump mid-stream (should be
>-  //                 around M59).
>-  bool DetectVp9Tl0PicIdxJump(int fixed_tl0, uint32_t timestamp) const
>-      RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
>-
>   // For every group of pictures, hold two sequence numbers. The first being
>   // the sequence number of the last packet of the last completed frame, and
>   // the second being the sequence number of the last packet of the last
>@@ -181,11 +163,9 @@ class RtpFrameReferenceFinder {
>       RTC_GUARDED_BY(crit_);
> 
>   // Holds the information about the last completed frame for a given temporal
>-  // layer given a Tl0 picture index.
>-  std::map<uint8_t,
>-           std::array<int16_t, kMaxTemporalLayers>,
>-           DescendingSeqNumComp<uint8_t>>
>-      layer_info_ RTC_GUARDED_BY(crit_);
>+  // layer given an unwrapped Tl0 picture index.
>+  std::map<int64_t, std::array<int16_t, kMaxTemporalLayers>> layer_info_
>+      RTC_GUARDED_BY(crit_);
> 
>   // Where the current scalability structure is in the
>   // |scalability_structures_| array.
>@@ -195,9 +175,8 @@ class RtpFrameReferenceFinder {
>   std::array<GofInfoVP9, kMaxGofSaved> scalability_structures_
>       RTC_GUARDED_BY(crit_);
> 
>-  // Holds the the Gof information for a given TL0 picture index.
>-  std::map<uint8_t, GofInfo, DescendingSeqNumComp<uint8_t>> gof_info_
>-      RTC_GUARDED_BY(crit_);
>+  // Holds the the Gof information for a given unwrapped TL0 picture index.
>+  std::map<int64_t, GofInfo> gof_info_ RTC_GUARDED_BY(crit_);
> 
>   // Keep track of which picture id and which temporal layer that had the
>   // up switch flag set.
>@@ -223,6 +202,8 @@ class RtpFrameReferenceFinder {
>   // Unwrapper used to unwrap VP8/VP9 streams which have their picture id
>   // specified.
>   SeqNumUnwrapper<uint16_t, kPicIdLength> unwrapper_ RTC_GUARDED_BY(crit_);
>+
>+  SeqNumUnwrapper<uint8_t> tl0_unwrapper_ RTC_GUARDED_BY(crit_);
> };
> 
> }  // namespace video_coding
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/rtp_frame_reference_finder_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/rtp_frame_reference_finder_unittest.cc
>index 5a8198787d3..9d26a2c8cfe 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/rtp_frame_reference_finder_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/rtp_frame_reference_finder_unittest.cc
>@@ -64,9 +64,9 @@ class TestRtpFrameReferenceFinder : public ::testing::Test,
> 
>   uint16_t Rand() { return rand_.Rand<uint16_t>(); }
> 
>-  void OnCompleteFrame(std::unique_ptr<FrameObject> frame) override {
>-    int64_t pid = frame->picture_id;
>-    uint16_t sidx = frame->spatial_layer;
>+  void OnCompleteFrame(std::unique_ptr<EncodedFrame> frame) override {
>+    int64_t pid = frame->id.picture_id;
>+    uint16_t sidx = frame->id.spatial_layer;
>     auto frame_it = frames_from_callback_.find(std::make_pair(pid, sidx));
>     if (frame_it != frames_from_callback_.end()) {
>       ADD_FAILURE() << "Already received frame with (pid:sidx): (" << pid << ":"
>@@ -108,10 +108,10 @@ class TestRtpFrameReferenceFinder : public ::testing::Test,
>     packet.seqNum = seq_num_start;
>     packet.markerBit = (seq_num_start == seq_num_end);
>     packet.frameType = keyframe ? kVideoFrameKey : kVideoFrameDelta;
>-    packet.video_header.codecHeader.VP8.pictureId = pid % (1 << 15);
>-    packet.video_header.codecHeader.VP8.temporalIdx = tid;
>-    packet.video_header.codecHeader.VP8.tl0PicIdx = tl0;
>-    packet.video_header.codecHeader.VP8.layerSync = sync;
>+    packet.video_header.vp8().pictureId = pid % (1 << 15);
>+    packet.video_header.vp8().temporalIdx = tid;
>+    packet.video_header.vp8().tl0PicIdx = tl0;
>+    packet.video_header.vp8().layerSync = sync;
>     ref_packet_buffer_->InsertPacket(&packet);
> 
>     if (seq_num_start != seq_num_end) {
>@@ -135,26 +135,28 @@ class TestRtpFrameReferenceFinder : public ::testing::Test,
>                     bool up_switch = false,
>                     GofInfoVP9* ss = nullptr) {
>     VCMPacket packet;
>+    auto& vp9_header =
>+        packet.video_header.video_type_header.emplace<RTPVideoHeaderVP9>();
>     packet.timestamp = pid;
>     packet.codec = kVideoCodecVP9;
>     packet.seqNum = seq_num_start;
>     packet.markerBit = (seq_num_start == seq_num_end);
>     packet.frameType = keyframe ? kVideoFrameKey : kVideoFrameDelta;
>-    packet.video_header.codecHeader.VP9.flexible_mode = false;
>-    packet.video_header.codecHeader.VP9.picture_id = pid % (1 << 15);
>-    packet.video_header.codecHeader.VP9.temporal_idx = tid;
>-    packet.video_header.codecHeader.VP9.spatial_idx = sid;
>-    packet.video_header.codecHeader.VP9.tl0_pic_idx = tl0;
>-    packet.video_header.codecHeader.VP9.temporal_up_switch = up_switch;
>+    vp9_header.flexible_mode = false;
>+    vp9_header.picture_id = pid % (1 << 15);
>+    vp9_header.temporal_idx = tid;
>+    vp9_header.spatial_idx = sid;
>+    vp9_header.tl0_pic_idx = tl0;
>+    vp9_header.temporal_up_switch = up_switch;
>     if (ss != nullptr) {
>-      packet.video_header.codecHeader.VP9.ss_data_available = true;
>-      packet.video_header.codecHeader.VP9.gof = *ss;
>+      vp9_header.ss_data_available = true;
>+      vp9_header.gof = *ss;
>     }
>     ref_packet_buffer_->InsertPacket(&packet);
> 
>     if (seq_num_start != seq_num_end) {
>       packet.markerBit = true;
>-      packet.video_header.codecHeader.VP9.ss_data_available = false;
>+      vp9_header.ss_data_available = false;
>       packet.seqNum = seq_num_end;
>       ref_packet_buffer_->InsertPacket(&packet);
>     }
>@@ -174,20 +176,22 @@ class TestRtpFrameReferenceFinder : public ::testing::Test,
>                      bool inter = false,
>                      std::vector<uint8_t> refs = std::vector<uint8_t>()) {
>     VCMPacket packet;
>+    auto& vp9_header =
>+        packet.video_header.video_type_header.emplace<RTPVideoHeaderVP9>();
>     packet.timestamp = pid;
>     packet.codec = kVideoCodecVP9;
>     packet.seqNum = seq_num_start;
>     packet.markerBit = (seq_num_start == seq_num_end);
>     packet.frameType = keyframe ? kVideoFrameKey : kVideoFrameDelta;
>-    packet.video_header.codecHeader.VP9.inter_layer_predicted = inter;
>-    packet.video_header.codecHeader.VP9.flexible_mode = true;
>-    packet.video_header.codecHeader.VP9.picture_id = pid % (1 << 15);
>-    packet.video_header.codecHeader.VP9.temporal_idx = tid;
>-    packet.video_header.codecHeader.VP9.spatial_idx = sid;
>-    packet.video_header.codecHeader.VP9.tl0_pic_idx = tl0;
>-    packet.video_header.codecHeader.VP9.num_ref_pics = refs.size();
>+    vp9_header.inter_layer_predicted = inter;
>+    vp9_header.flexible_mode = true;
>+    vp9_header.picture_id = pid % (1 << 15);
>+    vp9_header.temporal_idx = tid;
>+    vp9_header.spatial_idx = sid;
>+    vp9_header.tl0_pic_idx = tl0;
>+    vp9_header.num_ref_pics = refs.size();
>     for (size_t i = 0; i < refs.size(); ++i)
>-      packet.video_header.codecHeader.VP9.pid_diff[i] = refs[i];
>+      vp9_header.pid_diff[i] = refs[i];
>     ref_packet_buffer_->InsertPacket(&packet);
> 
>     if (seq_num_start != seq_num_end) {
>@@ -260,8 +264,9 @@ class TestRtpFrameReferenceFinder : public ::testing::Test,
>       return f1.first < f2.first;
>     }
>   };
>-  std::map<std::pair<int64_t, uint8_t>, std::unique_ptr<FrameObject>, FrameComp>
>-      frames_from_callback_;
>+  std::
>+      map<std::pair<int64_t, uint8_t>, std::unique_ptr<EncodedFrame>, FrameComp>
>+          frames_from_callback_;
> };
> 
> TEST_F(TestRtpFrameReferenceFinder, PaddingPackets) {
>@@ -879,7 +884,7 @@ TEST_F(TestRtpFrameReferenceFinder, Vp9GofSkipFramesTemporalLayers_0212) {
>   CheckReferencesVp9(0, 0);
>   CheckReferencesVp9(1, 0, 0);
>   CheckReferencesVp9(2, 0, 0);
>-  CheckReferencesVp9(3, 0, 1, 2);
>+  CheckReferencesVp9(3, 0, 2);
> 
>   // Skip frames with tl0 = 1
> 
>@@ -892,7 +897,7 @@ TEST_F(TestRtpFrameReferenceFinder, Vp9GofSkipFramesTemporalLayers_0212) {
>   CheckReferencesVp9(8, 0);
>   CheckReferencesVp9(9, 0, 8);
>   CheckReferencesVp9(10, 0, 8);
>-  CheckReferencesVp9(11, 0, 9, 10);
>+  CheckReferencesVp9(11, 0, 10);
> 
>   // Now insert frames with tl0 = 1
>   InsertVp9Gof(sn + 4, sn + 4, true, pid + 4, 0, 0, 1, false, &ss);
>@@ -1042,23 +1047,23 @@ TEST_F(TestRtpFrameReferenceFinder, Vp9GofTemporalLayers_0212) {
>   CheckReferencesVp9(0, 0);
>   CheckReferencesVp9(1, 0, 0);
>   CheckReferencesVp9(2, 0, 0);
>-  CheckReferencesVp9(3, 0, 1, 2);
>+  CheckReferencesVp9(3, 0, 2);
>   CheckReferencesVp9(4, 0, 0);
>   CheckReferencesVp9(5, 0, 4);
>   CheckReferencesVp9(6, 0, 4);
>-  CheckReferencesVp9(7, 0, 5, 6);
>+  CheckReferencesVp9(7, 0, 6);
>   CheckReferencesVp9(8, 0, 4);
>   CheckReferencesVp9(9, 0, 8);
>   CheckReferencesVp9(10, 0, 8);
>-  CheckReferencesVp9(11, 0, 9, 10);
>+  CheckReferencesVp9(11, 0, 10);
>   CheckReferencesVp9(12, 0, 8);
>   CheckReferencesVp9(13, 0, 12);
>   CheckReferencesVp9(14, 0, 12);
>-  CheckReferencesVp9(15, 0, 13, 14);
>+  CheckReferencesVp9(15, 0, 14);
>   CheckReferencesVp9(16, 0, 12);
>   CheckReferencesVp9(17, 0, 16);
>   CheckReferencesVp9(18, 0, 16);
>-  CheckReferencesVp9(19, 0, 17, 18);
>+  CheckReferencesVp9(19, 0, 18);
> }
> 
> TEST_F(TestRtpFrameReferenceFinder, Vp9GofTemporalLayersReordered_0212) {
>@@ -1092,23 +1097,23 @@ TEST_F(TestRtpFrameReferenceFinder, Vp9GofTemporalLayersReordered_0212) {
>   CheckReferencesVp9(0, 0);
>   CheckReferencesVp9(1, 0, 0);
>   CheckReferencesVp9(2, 0, 0);
>-  CheckReferencesVp9(3, 0, 1, 2);
>+  CheckReferencesVp9(3, 0, 2);
>   CheckReferencesVp9(4, 0, 0);
>   CheckReferencesVp9(5, 0, 4);
>   CheckReferencesVp9(6, 0, 4);
>-  CheckReferencesVp9(7, 0, 5, 6);
>+  CheckReferencesVp9(7, 0, 6);
>   CheckReferencesVp9(8, 0, 4);
>   CheckReferencesVp9(9, 0, 8);
>   CheckReferencesVp9(10, 0, 8);
>-  CheckReferencesVp9(11, 0, 9, 10);
>+  CheckReferencesVp9(11, 0, 10);
>   CheckReferencesVp9(12, 0, 8);
>   CheckReferencesVp9(13, 0, 12);
>   CheckReferencesVp9(14, 0, 12);
>-  CheckReferencesVp9(15, 0, 13, 14);
>+  CheckReferencesVp9(15, 0, 14);
>   CheckReferencesVp9(16, 0, 12);
>   CheckReferencesVp9(17, 0, 16);
>   CheckReferencesVp9(18, 0, 16);
>-  CheckReferencesVp9(19, 0, 17, 18);
>+  CheckReferencesVp9(19, 0, 18);
> }
> 
> TEST_F(TestRtpFrameReferenceFinder, Vp9GofTemporalLayersUpSwitch_02120212) {
>@@ -1224,11 +1229,11 @@ TEST_F(TestRtpFrameReferenceFinder, Vp9GofTemporalLayersReordered_01_0212) {
>   CheckReferencesVp9(4, 0, 0);
>   CheckReferencesVp9(5, 0, 4);
>   CheckReferencesVp9(6, 0, 4);
>-  CheckReferencesVp9(7, 0, 5, 6);
>+  CheckReferencesVp9(7, 0, 6);
>   CheckReferencesVp9(8, 0, 4);
>   CheckReferencesVp9(9, 0, 8);
>   CheckReferencesVp9(10, 0, 8);
>-  CheckReferencesVp9(11, 0, 9, 10);
>+  CheckReferencesVp9(11, 0, 10);
> }
> 
> TEST_F(TestRtpFrameReferenceFinder, Vp9FlexibleModeOneFrame) {
>@@ -1320,5 +1325,41 @@ TEST_F(TestRtpFrameReferenceFinder, WrappingFlexReference) {
>   CheckReferencesVp9(1, 0, 0);
> }
> 
>+TEST_F(TestRtpFrameReferenceFinder, Vp9GofPidJump) {
>+  uint16_t pid = Rand();
>+  uint16_t sn = Rand();
>+  GofInfoVP9 ss;
>+  ss.SetGofInfoVP9(kTemporalStructureMode3);
>+
>+  InsertVp9Gof(sn, sn, true, pid, 0, 0, 0, false, &ss);
>+  InsertVp9Gof(sn + 1, sn + 1, false, pid + 1000, 0, 0, 1);
>+}
>+
>+TEST_F(TestRtpFrameReferenceFinder, Vp9GofTl0Jump) {
>+  uint16_t pid = Rand();
>+  uint16_t sn = Rand();
>+  GofInfoVP9 ss;
>+  ss.SetGofInfoVP9(kTemporalStructureMode3);
>+
>+  InsertVp9Gof(sn, sn, true, pid + 0, 0, 0, 125, true, &ss);
>+  InsertVp9Gof(sn + 1, sn + 1, false, pid + 1, 0, 0, 0, false, &ss);
>+}
>+
>+TEST_F(TestRtpFrameReferenceFinder, Vp9GofTidTooHigh) {
>+  // Same as RtpFrameReferenceFinder::kMaxTemporalLayers.
>+  const int kMaxTemporalLayers = 5;
>+  uint16_t pid = Rand();
>+  uint16_t sn = Rand();
>+  GofInfoVP9 ss;
>+  ss.SetGofInfoVP9(kTemporalStructureMode2);
>+  ss.temporal_idx[1] = kMaxTemporalLayers;
>+
>+  InsertVp9Gof(sn, sn, true, pid, 0, 0, 0, false, &ss);
>+  InsertVp9Gof(sn + 1, sn + 1, false, pid + 1, 0, 0, 1);
>+
>+  ASSERT_EQ(1UL, frames_from_callback_.size());
>+  CheckReferencesVp9(0, 0);
>+}
>+
> }  // namespace video_coding
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/rtt_filter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/rtt_filter.cc
>index 9760929a2f3..75813a46ad9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/rtt_filter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/rtt_filter.cc
>@@ -49,8 +49,8 @@ void VCMRttFilter::Reset() {
>   _filtFactCount = 1;
>   _jumpCount = 0;
>   _driftCount = 0;
>-  memset(_jumpBuf, 0, kMaxDriftJumpCount);
>-  memset(_driftBuf, 0, kMaxDriftJumpCount);
>+  memset(_jumpBuf, 0, sizeof(_jumpBuf));
>+  memset(_driftBuf, 0, sizeof(_driftBuf));
> }
> 
> void VCMRttFilter::Update(int64_t rttMs) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/rtt_filter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/rtt_filter.h
>index 95de363eb95..073fabb85b0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/rtt_filter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/rtt_filter.h
>@@ -11,7 +11,7 @@
> #ifndef MODULES_VIDEO_CODING_RTT_FILTER_H_
> #define MODULES_VIDEO_CODING_RTT_FILTER_H_
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include <stdint.h>
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/session_info.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/session_info.cc
>index 74d7bfa17e7..a80c7b2898b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/session_info.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/session_info.cc
>@@ -9,7 +9,7 @@
>  */
> 
> #include "modules/video_coding/session_info.h"
>-
>+#include "modules/video_coding/jitter_buffer_common.h"
> #include "modules/video_coding/packet.h"
> #include "rtc_base/logging.h"
> 
>@@ -24,8 +24,7 @@ uint16_t BufferToUWord16(const uint8_t* dataBuffer) {
> }  // namespace
> 
> VCMSessionInfo::VCMSessionInfo()
>-    : session_nack_(false),
>-      complete_(false),
>+    : complete_(false),
>       decodable_(false),
>       frame_type_(kVideoFrameDelta),
>       packets_(),
>@@ -34,6 +33,8 @@ VCMSessionInfo::VCMSessionInfo()
>       first_packet_seq_num_(-1),
>       last_packet_seq_num_(-1) {}
> 
>+VCMSessionInfo::~VCMSessionInfo() {}
>+
> void VCMSessionInfo::UpdateDataPointers(const uint8_t* old_base_ptr,
>                                         const uint8_t* new_base_ptr) {
>   for (PacketIterator it = packets_.begin(); it != packets_.end(); ++it)
>@@ -60,10 +61,12 @@ int VCMSessionInfo::HighSequenceNumber() const {
> int VCMSessionInfo::PictureId() const {
>   if (packets_.empty())
>     return kNoPictureId;
>-  if (packets_.front().video_header.codec == kRtpVideoVp8) {
>-    return packets_.front().video_header.codecHeader.VP8.pictureId;
>-  } else if (packets_.front().video_header.codec == kRtpVideoVp9) {
>-    return packets_.front().video_header.codecHeader.VP9.picture_id;
>+  if (packets_.front().video_header.codec == kVideoCodecVP8) {
>+    return packets_.front().video_header.vp8().pictureId;
>+  } else if (packets_.front().video_header.codec == kVideoCodecVP9) {
>+    return absl::get<RTPVideoHeaderVP9>(
>+               packets_.front().video_header.video_type_header)
>+        .picture_id;
>   } else {
>     return kNoPictureId;
>   }
>@@ -72,10 +75,12 @@ int VCMSessionInfo::PictureId() const {
> int VCMSessionInfo::TemporalId() const {
>   if (packets_.empty())
>     return kNoTemporalIdx;
>-  if (packets_.front().video_header.codec == kRtpVideoVp8) {
>-    return packets_.front().video_header.codecHeader.VP8.temporalIdx;
>-  } else if (packets_.front().video_header.codec == kRtpVideoVp9) {
>-    return packets_.front().video_header.codecHeader.VP9.temporal_idx;
>+  if (packets_.front().video_header.codec == kVideoCodecVP8) {
>+    return packets_.front().video_header.vp8().temporalIdx;
>+  } else if (packets_.front().video_header.codec == kVideoCodecVP9) {
>+    return absl::get<RTPVideoHeaderVP9>(
>+               packets_.front().video_header.video_type_header)
>+        .temporal_idx;
>   } else {
>     return kNoTemporalIdx;
>   }
>@@ -84,10 +89,12 @@ int VCMSessionInfo::TemporalId() const {
> bool VCMSessionInfo::LayerSync() const {
>   if (packets_.empty())
>     return false;
>-  if (packets_.front().video_header.codec == kRtpVideoVp8) {
>-    return packets_.front().video_header.codecHeader.VP8.layerSync;
>-  } else if (packets_.front().video_header.codec == kRtpVideoVp9) {
>-    return packets_.front().video_header.codecHeader.VP9.temporal_up_switch;
>+  if (packets_.front().video_header.codec == kVideoCodecVP8) {
>+    return packets_.front().video_header.vp8().layerSync;
>+  } else if (packets_.front().video_header.codec == kVideoCodecVP9) {
>+    return absl::get<RTPVideoHeaderVP9>(
>+               packets_.front().video_header.video_type_header)
>+        .temporal_up_switch;
>   } else {
>     return false;
>   }
>@@ -96,53 +103,50 @@ bool VCMSessionInfo::LayerSync() const {
> int VCMSessionInfo::Tl0PicId() const {
>   if (packets_.empty())
>     return kNoTl0PicIdx;
>-  if (packets_.front().video_header.codec == kRtpVideoVp8) {
>-    return packets_.front().video_header.codecHeader.VP8.tl0PicIdx;
>-  } else if (packets_.front().video_header.codec == kRtpVideoVp9) {
>-    return packets_.front().video_header.codecHeader.VP9.tl0_pic_idx;
>+  if (packets_.front().video_header.codec == kVideoCodecVP8) {
>+    return packets_.front().video_header.vp8().tl0PicIdx;
>+  } else if (packets_.front().video_header.codec == kVideoCodecVP9) {
>+    return absl::get<RTPVideoHeaderVP9>(
>+               packets_.front().video_header.video_type_header)
>+        .tl0_pic_idx;
>   } else {
>     return kNoTl0PicIdx;
>   }
> }
> 
>-bool VCMSessionInfo::NonReference() const {
>-  if (packets_.empty() || packets_.front().video_header.codec != kRtpVideoVp8)
>-    return false;
>-  return packets_.front().video_header.codecHeader.VP8.nonReference;
>-}
>-
> std::vector<NaluInfo> VCMSessionInfo::GetNaluInfos() const {
>-  if (packets_.empty() || packets_.front().video_header.codec != kRtpVideoH264)
>+  if (packets_.empty() ||
>+      packets_.front().video_header.codec != kVideoCodecH264)
>     return std::vector<NaluInfo>();
>   std::vector<NaluInfo> nalu_infos;
>   for (const VCMPacket& packet : packets_) {
>-    for (size_t i = 0; i < packet.video_header.codecHeader.H264.nalus_length;
>-         ++i) {
>-      nalu_infos.push_back(packet.video_header.codecHeader.H264.nalus[i]);
>+    const auto& h264 =
>+        absl::get<RTPVideoHeaderH264>(packet.video_header.video_type_header);
>+    for (size_t i = 0; i < h264.nalus_length; ++i) {
>+      nalu_infos.push_back(h264.nalus[i]);
>     }
>   }
>   return nalu_infos;
> }
> 
> void VCMSessionInfo::SetGofInfo(const GofInfoVP9& gof_info, size_t idx) {
>-  if (packets_.empty() || packets_.front().video_header.codec != kRtpVideoVp9 ||
>-      packets_.front().video_header.codecHeader.VP9.flexible_mode) {
>+  if (packets_.empty())
>     return;
>-  }
>-  packets_.front().video_header.codecHeader.VP9.temporal_idx =
>-      gof_info.temporal_idx[idx];
>-  packets_.front().video_header.codecHeader.VP9.temporal_up_switch =
>-      gof_info.temporal_up_switch[idx];
>-  packets_.front().video_header.codecHeader.VP9.num_ref_pics =
>-      gof_info.num_ref_pics[idx];
>+
>+  auto* vp9_header = absl::get_if<RTPVideoHeaderVP9>(
>+      &packets_.front().video_header.video_type_header);
>+  if (!vp9_header || vp9_header->flexible_mode)
>+    return;
>+
>+  vp9_header->temporal_idx = gof_info.temporal_idx[idx];
>+  vp9_header->temporal_up_switch = gof_info.temporal_up_switch[idx];
>+  vp9_header->num_ref_pics = gof_info.num_ref_pics[idx];
>   for (uint8_t i = 0; i < gof_info.num_ref_pics[idx]; ++i) {
>-    packets_.front().video_header.codecHeader.VP9.pid_diff[i] =
>-        gof_info.pid_diff[idx][i];
>+    vp9_header->pid_diff[i] = gof_info.pid_diff[idx][i];
>   }
> }
> 
> void VCMSessionInfo::Reset() {
>-  session_nack_ = false;
>   complete_ = false;
>   decodable_ = false;
>   frame_type_ = kVideoFrameDelta;
>@@ -185,8 +189,9 @@ size_t VCMSessionInfo::InsertBuffer(uint8_t* frame_buffer,
>   // header supplied by the H264 depacketizer.
>   const size_t kH264NALHeaderLengthInBytes = 1;
>   const size_t kLengthFieldLength = 2;
>-  if (packet.video_header.codec == kRtpVideoH264 &&
>-      packet.video_header.codecHeader.H264.packetization_type == kH264StapA) {
>+  const auto* h264 =
>+      absl::get_if<RTPVideoHeaderH264>(&packet.video_header.video_type_header);
>+  if (h264 && h264->packetization_type == kH264StapA) {
>     size_t required_length = 0;
>     const uint8_t* nalu_ptr = packet_buffer + kH264NALHeaderLengthInBytes;
>     while (nalu_ptr < packet_buffer + packet.sizeBytes) {
>@@ -342,7 +347,7 @@ size_t VCMSessionInfo::DeletePacketData(PacketIterator start,
> VCMSessionInfo::PacketIterator VCMSessionInfo::FindNextPartitionBeginning(
>     PacketIterator it) const {
>   while (it != packets_.end()) {
>-    if ((*it).video_header.codecHeader.VP8.beginningOfPartition) {
>+    if ((*it).video_header.vp8().beginningOfPartition) {
>       return it;
>     }
>     ++it;
>@@ -354,10 +359,10 @@ VCMSessionInfo::PacketIterator VCMSessionInfo::FindPartitionEnd(
>     PacketIterator it) const {
>   assert((*it).codec == kVideoCodecVP8);
>   PacketIterator prev_it = it;
>-  const int partition_id = (*it).video_header.codecHeader.VP8.partitionId;
>+  const int partition_id = (*it).video_header.vp8().partitionId;
>   while (it != packets_.end()) {
>-    bool beginning = (*it).video_header.codecHeader.VP8.beginningOfPartition;
>-    int current_partition_id = (*it).video_header.codecHeader.VP8.partitionId;
>+    bool beginning = (*it).video_header.vp8().beginningOfPartition;
>+    int current_partition_id = (*it).video_header.vp8().partitionId;
>     bool packet_loss_found = (!beginning && !InSequence(it, prev_it));
>     if (packet_loss_found ||
>         (beginning && current_partition_id != partition_id)) {
>@@ -407,12 +412,6 @@ size_t VCMSessionInfo::MakeDecodable() {
>   return return_length;
> }
> 
>-void VCMSessionInfo::SetNotDecodableIfIncomplete() {
>-  // We don't need to check for completeness first because the two are
>-  // orthogonal. If complete_ is true, decodable_ is irrelevant.
>-  decodable_ = false;
>-}
>-
> bool VCMSessionInfo::HaveFirstPacket() const {
>   return !packets_.empty() && (first_packet_seq_num_ != -1);
> }
>@@ -421,10 +420,6 @@ bool VCMSessionInfo::HaveLastPacket() const {
>   return !packets_.empty() && (last_packet_seq_num_ != -1);
> }
> 
>-bool VCMSessionInfo::session_nack() const {
>-  return session_nack_;
>-}
>-
> int VCMSessionInfo::InsertPacket(const VCMPacket& packet,
>                                  uint8_t* frame_buffer,
>                                  VCMDecodeErrorMode decode_error_mode,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/session_info.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/session_info.h
>index 2999e9ad84d..8d577fe0ccc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/session_info.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/session_info.h
>@@ -17,7 +17,6 @@
> #include "modules/include/module_common_types.h"
> #include "modules/video_coding/include/video_coding.h"
> #include "modules/video_coding/packet.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> // Used to pass data from jitter buffer to session info.
>@@ -30,22 +29,10 @@ struct FrameData {
> class VCMSessionInfo {
>  public:
>   VCMSessionInfo();
>+  ~VCMSessionInfo();
> 
>   void UpdateDataPointers(const uint8_t* old_base_ptr,
>                           const uint8_t* new_base_ptr);
>-  // NACK - Building the NACK lists.
>-  // Build hard NACK list: Zero out all entries in list up to and including
>-  // _lowSeqNum.
>-  int BuildHardNackList(int* seq_num_list,
>-                        int seq_num_list_length,
>-                        int nack_seq_nums_index);
>-
>-  // Build soft NACK list:  Zero out only a subset of the packets, discard
>-  // empty packets.
>-  int BuildSoftNackList(int* seq_num_list,
>-                        int seq_num_list_length,
>-                        int nack_seq_nums_index,
>-                        int rtt_ms);
>   void Reset();
>   int InsertPacket(const VCMPacket& packet,
>                    uint8_t* frame_buffer,
>@@ -60,17 +47,11 @@ class VCMSessionInfo {
>   // Returns the number of bytes deleted from the session.
>   size_t MakeDecodable();
> 
>-  // Sets decodable_ to false.
>-  // Used by the dual decoder. After the mode is changed to kNoErrors from
>-  // kWithErrors or kSelective errors, any states that have been marked
>-  // decodable and are not complete are marked as non-decodable.
>-  void SetNotDecodableIfIncomplete();
>-
>+  // TODO(nisse): Used by tests only.
>   size_t SessionLength() const;
>   int NumPackets() const;
>   bool HaveFirstPacket() const;
>   bool HaveLastPacket() const;
>-  bool session_nack() const;
>   webrtc::FrameType FrameType() const { return frame_type_; }
>   int LowSequenceNumber() const;
> 
>@@ -80,16 +61,11 @@ class VCMSessionInfo {
>   int TemporalId() const;
>   bool LayerSync() const;
>   int Tl0PicId() const;
>-  bool NonReference() const;
> 
>   std::vector<NaluInfo> GetNaluInfos() const;
> 
>   void SetGofInfo(const GofInfoVP9& gof_info, size_t idx);
> 
>-  // The number of packets discarded because the decoder can't make use of
>-  // them.
>-  int packets_not_decodable() const;
>-
>  private:
>   enum { kMaxVP8Partitions = 9 };
> 
>@@ -142,8 +118,6 @@ class VCMSessionInfo {
>   //        frame, we know that the frame is medium or large-sized.
>   void UpdateDecodableSession(const FrameData& frame_data);
> 
>-  // If this session has been NACKed by the jitter buffer.
>-  bool session_nack_;
>   bool complete_;
>   bool decodable_;
>   webrtc::FrameType frame_type_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/session_info_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/session_info_unittest.cc
>index 1863ad7eef6..2846653b3fc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/session_info_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/session_info_unittest.cc
>@@ -23,7 +23,6 @@ class TestSessionInfo : public ::testing::Test {
>     memset(packet_buffer_, 0, sizeof(packet_buffer_));
>     memset(frame_buffer_, 0, sizeof(frame_buffer_));
>     session_.Reset();
>-    packet_.Reset();
>     packet_.frameType = kVideoFrameDelta;
>     packet_.sizeBytes = packet_buffer_size();
>     packet_.dataPtr = packet_buffer_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/test/stream_generator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/test/stream_generator.h
>index 894df5e6155..38ca92d7634 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/test/stream_generator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/test/stream_generator.h
>@@ -14,7 +14,6 @@
> 
> #include "modules/video_coding/packet.h"
> #include "rtc_base/constructormagic.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/test/test_util.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/test/test_util.h
>index 404e49ebcfe..a38fc58b128 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/test/test_util.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/test/test_util.h
>@@ -18,6 +18,7 @@ class NullEventFactory : public webrtc::EventFactory {
>   virtual ~NullEventFactory() {}
> 
>   webrtc::EventWrapper* CreateEvent() override { return new NullEvent; }
>+
>  private:
>   // Private class to avoid more dependencies on it in tests.
>   class NullEvent : public webrtc::EventWrapper {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/timestamp_map.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/timestamp_map.h
>index 15f6e27222d..6657787ac21 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/timestamp_map.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/timestamp_map.h
>@@ -13,8 +13,6 @@
> 
> #include <memory>
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>-
> namespace webrtc {
> 
> struct VCMFrameInformation;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/timing.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/timing.cc
>index f991a7f496b..310467b553d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/timing.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/timing.cc
>@@ -12,11 +12,8 @@
> 
> #include <algorithm>
> 
>-#include "modules/video_coding/internal_defines.h"
>-#include "modules/video_coding/jitter_buffer_common.h"
>+#include "rtc_base/time/timestamp_extrapolator.h"
> #include "system_wrappers/include/clock.h"
>-#include "system_wrappers/include/metrics.h"
>-#include "system_wrappers/include/timestamp_extrapolator.h"
> 
> namespace webrtc {
> 
>@@ -33,10 +30,7 @@ VCMTiming::VCMTiming(Clock* clock, VCMTiming* master_timing)
>       last_decode_ms_(0),
>       prev_frame_timestamp_(0),
>       timing_frame_info_(),
>-      num_decoded_frames_(0),
>-      num_delayed_decoded_frames_(0),
>-      first_decoded_frame_ms_(-1),
>-      sum_missed_render_deadline_ms_(0) {
>+      num_decoded_frames_(0) {
>   if (master_timing == NULL) {
>     master_ = true;
>     ts_extrapolator_ = new TimestampExtrapolator(clock_->TimeInMilliseconds());
>@@ -46,35 +40,11 @@ VCMTiming::VCMTiming(Clock* clock, VCMTiming* master_timing)
> }
> 
> VCMTiming::~VCMTiming() {
>-  UpdateHistograms();
>   if (master_) {
>     delete ts_extrapolator_;
>   }
> }
> 
>-void VCMTiming::UpdateHistograms() const {
>-  rtc::CritScope cs(&crit_sect_);
>-  if (num_decoded_frames_ == 0) {
>-    return;
>-  }
>-  int64_t elapsed_sec =
>-      (clock_->TimeInMilliseconds() - first_decoded_frame_ms_) / 1000;
>-  if (elapsed_sec < metrics::kMinRunTimeInSeconds) {
>-    return;
>-  }
>-  RTC_HISTOGRAM_COUNTS_100(
>-      "WebRTC.Video.DecodedFramesPerSecond",
>-      static_cast<int>((num_decoded_frames_ / elapsed_sec) + 0.5f));
>-  RTC_HISTOGRAM_PERCENTAGE(
>-      "WebRTC.Video.DelayedFramesToRenderer",
>-      num_delayed_decoded_frames_ * 100 / num_decoded_frames_);
>-  if (num_delayed_decoded_frames_ > 0) {
>-    RTC_HISTOGRAM_COUNTS_1000(
>-        "WebRTC.Video.DelayedFramesToRenderer_AvgDelayInMs",
>-        sum_missed_render_deadline_ms_ / num_delayed_decoded_frames_);
>-  }
>-}
>-
> void VCMTiming::Reset() {
>   rtc::CritScope cs(&crit_sect_);
>   ts_extrapolator_->Reset(clock_->TimeInMilliseconds());
>@@ -86,11 +56,6 @@ void VCMTiming::Reset() {
>   prev_frame_timestamp_ = 0;
> }
> 
>-void VCMTiming::ResetDecodeTime() {
>-  rtc::CritScope cs(&crit_sect_);
>-  codec_timer_.reset(new VCMCodecTimer());
>-}
>-
> void VCMTiming::set_render_delay(int render_delay_ms) {
>   rtc::CritScope cs(&crit_sect_);
>   render_delay_ms_ = render_delay_ms;
>@@ -155,9 +120,8 @@ void VCMTiming::UpdateCurrentDelay(uint32_t frame_timestamp) {
>     }
> 
>     if (max_change_ms <= 0) {
>-      // Any changes less than 1 ms are truncated and
>-      // will be postponed. Negative change will be due
>-      // to reordering and should be ignored.
>+      // Any changes less than 1 ms are truncated and will be postponed.
>+      // Negative change will be due to reordering and should be ignored.
>       return;
>     }
>     delay_diff_ms = std::max(delay_diff_ms, -max_change_ms);
>@@ -185,26 +149,15 @@ void VCMTiming::UpdateCurrentDelay(int64_t render_time_ms,
>   }
> }
> 
>-int32_t VCMTiming::StopDecodeTimer(uint32_t time_stamp,
>-                                   int32_t decode_time_ms,
>-                                   int64_t now_ms,
>-                                   int64_t render_time_ms) {
>+void VCMTiming::StopDecodeTimer(uint32_t time_stamp,
>+                                int32_t decode_time_ms,
>+                                int64_t now_ms,
>+                                int64_t render_time_ms) {
>   rtc::CritScope cs(&crit_sect_);
>   codec_timer_->AddTiming(decode_time_ms, now_ms);
>   assert(decode_time_ms >= 0);
>   last_decode_ms_ = decode_time_ms;
>-
>-  // Update stats.
>   ++num_decoded_frames_;
>-  if (num_decoded_frames_ == 1) {
>-    first_decoded_frame_ms_ = now_ms;
>-  }
>-  int time_until_rendering_ms = render_time_ms - render_delay_ms_ - now_ms;
>-  if (time_until_rendering_ms < 0) {
>-    sum_missed_render_deadline_ms_ += -time_until_rendering_ms;
>-    ++num_delayed_decoded_frames_;
>-  }
>-  return 0;
> }
> 
> void VCMTiming::IncomingTimestamp(uint32_t time_stamp, int64_t now_ms) {
>@@ -215,23 +168,21 @@ void VCMTiming::IncomingTimestamp(uint32_t time_stamp, int64_t now_ms) {
> int64_t VCMTiming::RenderTimeMs(uint32_t frame_timestamp,
>                                 int64_t now_ms) const {
>   rtc::CritScope cs(&crit_sect_);
>-  const int64_t render_time_ms = RenderTimeMsInternal(frame_timestamp, now_ms);
>-  return render_time_ms;
>+  return RenderTimeMsInternal(frame_timestamp, now_ms);
> }
> 
> int64_t VCMTiming::RenderTimeMsInternal(uint32_t frame_timestamp,
>                                         int64_t now_ms) const {
>+  if (min_playout_delay_ms_ == 0 && max_playout_delay_ms_ == 0) {
>+    // Render as soon as possible.
>+    return 0;
>+  }
>   int64_t estimated_complete_time_ms =
>       ts_extrapolator_->ExtrapolateLocalTime(frame_timestamp);
>   if (estimated_complete_time_ms == -1) {
>     estimated_complete_time_ms = now_ms;
>   }
> 
>-  if (min_playout_delay_ms_ == 0 && max_playout_delay_ms_ == 0) {
>-    // Render as soon as possible
>-    return now_ms;
>-  }
>-
>   // Make sure the actual delay stays in the range of |min_playout_delay_ms_|
>   // and |max_playout_delay_ms_|.
>   int actual_delay = std::max(current_delay_ms_, min_playout_delay_ms_);
>@@ -239,42 +190,20 @@ int64_t VCMTiming::RenderTimeMsInternal(uint32_t frame_timestamp,
>   return estimated_complete_time_ms + actual_delay;
> }
> 
>-// Must be called from inside a critical section.
> int VCMTiming::RequiredDecodeTimeMs() const {
>   const int decode_time_ms = codec_timer_->RequiredDecodeTimeMs();
>   assert(decode_time_ms >= 0);
>   return decode_time_ms;
> }
> 
>-uint32_t VCMTiming::MaxWaitingTime(int64_t render_time_ms,
>-                                   int64_t now_ms) const {
>+int64_t VCMTiming::MaxWaitingTime(int64_t render_time_ms,
>+                                  int64_t now_ms) const {
>   rtc::CritScope cs(&crit_sect_);
> 
>   const int64_t max_wait_time_ms =
>       render_time_ms - now_ms - RequiredDecodeTimeMs() - render_delay_ms_;
> 
>-  if (max_wait_time_ms < 0) {
>-    return 0;
>-  }
>-  return static_cast<uint32_t>(max_wait_time_ms);
>-}
>-
>-bool VCMTiming::EnoughTimeToDecode(
>-    uint32_t available_processing_time_ms) const {
>-  rtc::CritScope cs(&crit_sect_);
>-  int64_t required_decode_time_ms = RequiredDecodeTimeMs();
>-  if (required_decode_time_ms < 0) {
>-    // Haven't decoded any frames yet, try decoding one to get an estimate
>-    // of the decode time.
>-    return true;
>-  } else if (required_decode_time_ms == 0) {
>-    // Decode time is less than 1, set to 1 for now since
>-    // we don't have any better precision. Count ticks later?
>-    required_decode_time_ms = 1;
>-  }
>-  return static_cast<int64_t>(available_processing_time_ms) -
>-             required_decode_time_ms >
>-         0;
>+  return max_wait_time_ms;
> }
> 
> int VCMTiming::TargetVideoDelay() const {
>@@ -310,7 +239,7 @@ void VCMTiming::SetTimingFrameInfo(const TimingFrameInfo& info) {
>   timing_frame_info_.emplace(info);
> }
> 
>-rtc::Optional<TimingFrameInfo> VCMTiming::GetTimingFrameInfo() {
>+absl::optional<TimingFrameInfo> VCMTiming::GetTimingFrameInfo() {
>   rtc::CritScope cs(&crit_sect_);
>   return timing_frame_info_;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/timing.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/timing.h
>index 12fa0c27980..2b5e21d6495 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/timing.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/timing.h
>@@ -16,7 +16,6 @@
> #include "modules/video_coding/codec_timer.h"
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/thread_annotations.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -32,7 +31,6 @@ class VCMTiming {
> 
>   // Resets the timing to the initial state.
>   void Reset();
>-  void ResetDecodeTime();
> 
>   // Set the amount of time needed to render an image. Defaults to 10 ms.
>   void set_render_delay(int render_delay_ms);
>@@ -41,16 +39,12 @@ class VCMTiming {
>   // get the desired jitter buffer level.
>   void SetJitterDelay(int required_delay_ms);
> 
>-  // Set the minimum playout delay from capture to render in ms.
>+  // Set/get the minimum playout delay from capture to render in ms.
>   void set_min_playout_delay(int min_playout_delay_ms);
>-
>-  // Returns the minimum playout delay from capture to render in ms.
>   int min_playout_delay();
> 
>-  // Set the maximum playout delay from capture to render in ms.
>+  // Set/get the maximum playout delay from capture to render in ms.
>   void set_max_playout_delay(int max_playout_delay_ms);
>-
>-  // Returns the maximum playout delay from capture to render in ms.
>   int max_playout_delay();
> 
>   // Increases or decreases the current delay to get closer to the target delay.
>@@ -67,31 +61,28 @@ class VCMTiming {
> 
>   // Stops the decoder timer, should be called when the decoder returns a frame
>   // or when the decoded frame callback is called.
>-  int32_t StopDecodeTimer(uint32_t time_stamp,
>-                          int32_t decode_time_ms,
>-                          int64_t now_ms,
>-                          int64_t render_time_ms);
>+  void StopDecodeTimer(uint32_t time_stamp,
>+                       int32_t decode_time_ms,
>+                       int64_t now_ms,
>+                       int64_t render_time_ms);
> 
>   // Used to report that a frame is passed to decoding. Updates the timestamp
>   // filter which is used to map between timestamps and receiver system time.
>   void IncomingTimestamp(uint32_t time_stamp, int64_t last_packet_time_ms);
>+
>   // Returns the receiver system time when the frame with timestamp
>-  // frame_timestamp should be rendered, assuming that the system time currently
>-  // is now_ms.
>+  // |frame_timestamp| should be rendered, assuming that the system time
>+  // currently is |now_ms|.
>   virtual int64_t RenderTimeMs(uint32_t frame_timestamp, int64_t now_ms) const;
> 
>   // Returns the maximum time in ms that we can wait for a frame to become
>   // complete before we must pass it to the decoder.
>-  virtual uint32_t MaxWaitingTime(int64_t render_time_ms, int64_t now_ms) const;
>+  virtual int64_t MaxWaitingTime(int64_t render_time_ms, int64_t now_ms) const;
> 
>   // Returns the current target delay which is required delay + decode time +
>   // render delay.
>   int TargetVideoDelay() const;
> 
>-  // Calculates whether or not there is enough time to decode a frame given a
>-  // certain amount of processing time.
>-  bool EnoughTimeToDecode(uint32_t available_processing_time_ms) const;
>-
>   // Return current timing information. Returns true if the first frame has been
>   // decoded, false otherwise.
>   virtual bool GetTimings(int* decode_ms,
>@@ -103,7 +94,7 @@ class VCMTiming {
>                           int* render_delay_ms) const;
> 
>   void SetTimingFrameInfo(const TimingFrameInfo& info);
>-  rtc::Optional<TimingFrameInfo> GetTimingFrameInfo();
>+  absl::optional<TimingFrameInfo> GetTimingFrameInfo();
> 
>   enum { kDefaultRenderDelayMs = 10 };
>   enum { kDelayMaxChangeMsPerS = 100 };
>@@ -115,8 +106,6 @@ class VCMTiming {
>   int TargetDelayInternal() const RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
> 
>  private:
>-  void UpdateHistograms() const;
>-
>   rtc::CriticalSection crit_sect_;
>   Clock* const clock_;
>   bool master_ RTC_GUARDED_BY(crit_sect_);
>@@ -134,13 +123,8 @@ class VCMTiming {
>   int current_delay_ms_ RTC_GUARDED_BY(crit_sect_);
>   int last_decode_ms_ RTC_GUARDED_BY(crit_sect_);
>   uint32_t prev_frame_timestamp_ RTC_GUARDED_BY(crit_sect_);
>-  rtc::Optional<TimingFrameInfo> timing_frame_info_ RTC_GUARDED_BY(crit_sect_);
>-
>-  // Statistics.
>+  absl::optional<TimingFrameInfo> timing_frame_info_ RTC_GUARDED_BY(crit_sect_);
>   size_t num_decoded_frames_ RTC_GUARDED_BY(crit_sect_);
>-  size_t num_delayed_decoded_frames_ RTC_GUARDED_BY(crit_sect_);
>-  int64_t first_decoded_frame_ms_ RTC_GUARDED_BY(crit_sect_);
>-  uint64_t sum_missed_render_deadline_ms_ RTC_GUARDED_BY(crit_sect_);
> };
> }  // namespace webrtc
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/timing_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/timing_unittest.cc
>index 40c9f1f38d1..b9397ea005c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/timing_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/timing_unittest.cc
>@@ -8,138 +8,123 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include <math.h>
>-#include <stdio.h>
>-#include <stdlib.h>
>-
>-#include "modules/video_coding/include/video_coding.h"
>-#include "modules/video_coding/internal_defines.h"
> #include "modules/video_coding/timing.h"
> #include "system_wrappers/include/clock.h"
> #include "test/gtest.h"
>-#include "test/testsupport/fileutils.h"
> 
> namespace webrtc {
>+namespace {
>+const int kFps = 25;
>+}  // namespace
> 
> TEST(ReceiverTiming, Tests) {
>   SimulatedClock clock(0);
>   VCMTiming timing(&clock);
>-  uint32_t waitTime = 0;
>-  uint32_t jitterDelayMs = 0;
>-  uint32_t requiredDecodeTimeMs = 0;
>-  uint32_t timeStamp = 0;
>-
>   timing.Reset();
> 
>-  timing.UpdateCurrentDelay(timeStamp);
>+  uint32_t timestamp = 0;
>+  timing.UpdateCurrentDelay(timestamp);
> 
>   timing.Reset();
> 
>-  timing.IncomingTimestamp(timeStamp, clock.TimeInMilliseconds());
>-  jitterDelayMs = 20;
>-  timing.SetJitterDelay(jitterDelayMs);
>-  timing.UpdateCurrentDelay(timeStamp);
>+  timing.IncomingTimestamp(timestamp, clock.TimeInMilliseconds());
>+  uint32_t jitter_delay_ms = 20;
>+  timing.SetJitterDelay(jitter_delay_ms);
>+  timing.UpdateCurrentDelay(timestamp);
>   timing.set_render_delay(0);
>-  waitTime = timing.MaxWaitingTime(
>-      timing.RenderTimeMs(timeStamp, clock.TimeInMilliseconds()),
>+  uint32_t wait_time_ms = timing.MaxWaitingTime(
>+      timing.RenderTimeMs(timestamp, clock.TimeInMilliseconds()),
>       clock.TimeInMilliseconds());
>   // First update initializes the render time. Since we have no decode delay
>-  // we get waitTime = renderTime - now - renderDelay = jitter.
>-  EXPECT_EQ(jitterDelayMs, waitTime);
>+  // we get wait_time_ms = renderTime - now - renderDelay = jitter.
>+  EXPECT_EQ(jitter_delay_ms, wait_time_ms);
> 
>-  jitterDelayMs += VCMTiming::kDelayMaxChangeMsPerS + 10;
>-  timeStamp += 90000;
>+  jitter_delay_ms += VCMTiming::kDelayMaxChangeMsPerS + 10;
>+  timestamp += 90000;
>   clock.AdvanceTimeMilliseconds(1000);
>-  timing.SetJitterDelay(jitterDelayMs);
>-  timing.UpdateCurrentDelay(timeStamp);
>-  waitTime = timing.MaxWaitingTime(
>-      timing.RenderTimeMs(timeStamp, clock.TimeInMilliseconds()),
>+  timing.SetJitterDelay(jitter_delay_ms);
>+  timing.UpdateCurrentDelay(timestamp);
>+  wait_time_ms = timing.MaxWaitingTime(
>+      timing.RenderTimeMs(timestamp, clock.TimeInMilliseconds()),
>       clock.TimeInMilliseconds());
>   // Since we gradually increase the delay we only get 100 ms every second.
>-  EXPECT_EQ(jitterDelayMs - 10, waitTime);
>+  EXPECT_EQ(jitter_delay_ms - 10, wait_time_ms);
> 
>-  timeStamp += 90000;
>+  timestamp += 90000;
>   clock.AdvanceTimeMilliseconds(1000);
>-  timing.UpdateCurrentDelay(timeStamp);
>-  waitTime = timing.MaxWaitingTime(
>-      timing.RenderTimeMs(timeStamp, clock.TimeInMilliseconds()),
>+  timing.UpdateCurrentDelay(timestamp);
>+  wait_time_ms = timing.MaxWaitingTime(
>+      timing.RenderTimeMs(timestamp, clock.TimeInMilliseconds()),
>       clock.TimeInMilliseconds());
>-  EXPECT_EQ(waitTime, jitterDelayMs);
>+  EXPECT_EQ(jitter_delay_ms, wait_time_ms);
> 
>-  // 300 incoming frames without jitter, verify that this gives the exact wait
>-  // time.
>-  for (int i = 0; i < 300; i++) {
>-    clock.AdvanceTimeMilliseconds(1000 / 25);
>-    timeStamp += 90000 / 25;
>-    timing.IncomingTimestamp(timeStamp, clock.TimeInMilliseconds());
>+  // Insert frames without jitter, verify that this gives the exact wait time.
>+  const int kNumFrames = 300;
>+  for (int i = 0; i < kNumFrames; i++) {
>+    clock.AdvanceTimeMilliseconds(1000 / kFps);
>+    timestamp += 90000 / kFps;
>+    timing.IncomingTimestamp(timestamp, clock.TimeInMilliseconds());
>   }
>-  timing.UpdateCurrentDelay(timeStamp);
>-  waitTime = timing.MaxWaitingTime(
>-      timing.RenderTimeMs(timeStamp, clock.TimeInMilliseconds()),
>+  timing.UpdateCurrentDelay(timestamp);
>+  wait_time_ms = timing.MaxWaitingTime(
>+      timing.RenderTimeMs(timestamp, clock.TimeInMilliseconds()),
>       clock.TimeInMilliseconds());
>-  EXPECT_EQ(waitTime, jitterDelayMs);
>+  EXPECT_EQ(jitter_delay_ms, wait_time_ms);
> 
>-  // Add decode time estimates.
>-  for (int i = 0; i < 10; i++) {
>-    int64_t startTimeMs = clock.TimeInMilliseconds();
>-    clock.AdvanceTimeMilliseconds(10);
>+  // Add decode time estimates for 1 second.
>+  const uint32_t kDecodeTimeMs = 10;
>+  for (int i = 0; i < kFps; i++) {
>+    clock.AdvanceTimeMilliseconds(kDecodeTimeMs);
>     timing.StopDecodeTimer(
>-        timeStamp, clock.TimeInMilliseconds() - startTimeMs,
>-        clock.TimeInMilliseconds(),
>-        timing.RenderTimeMs(timeStamp, clock.TimeInMilliseconds()));
>-    timeStamp += 90000 / 25;
>-    clock.AdvanceTimeMilliseconds(1000 / 25 - 10);
>-    timing.IncomingTimestamp(timeStamp, clock.TimeInMilliseconds());
>+        timestamp, kDecodeTimeMs, clock.TimeInMilliseconds(),
>+        timing.RenderTimeMs(timestamp, clock.TimeInMilliseconds()));
>+    timestamp += 90000 / kFps;
>+    clock.AdvanceTimeMilliseconds(1000 / kFps - kDecodeTimeMs);
>+    timing.IncomingTimestamp(timestamp, clock.TimeInMilliseconds());
>   }
>-  requiredDecodeTimeMs = 10;
>-  timing.SetJitterDelay(jitterDelayMs);
>-  clock.AdvanceTimeMilliseconds(1000);
>-  timeStamp += 90000;
>-  timing.UpdateCurrentDelay(timeStamp);
>-  waitTime = timing.MaxWaitingTime(
>-      timing.RenderTimeMs(timeStamp, clock.TimeInMilliseconds()),
>+  timing.UpdateCurrentDelay(timestamp);
>+  wait_time_ms = timing.MaxWaitingTime(
>+      timing.RenderTimeMs(timestamp, clock.TimeInMilliseconds()),
>       clock.TimeInMilliseconds());
>-  EXPECT_EQ(waitTime, jitterDelayMs);
>+  EXPECT_EQ(jitter_delay_ms, wait_time_ms);
> 
>-  int minTotalDelayMs = 200;
>-  timing.set_min_playout_delay(minTotalDelayMs);
>+  const int kMinTotalDelayMs = 200;
>+  timing.set_min_playout_delay(kMinTotalDelayMs);
>   clock.AdvanceTimeMilliseconds(5000);
>-  timeStamp += 5 * 90000;
>-  timing.UpdateCurrentDelay(timeStamp);
>+  timestamp += 5 * 90000;
>+  timing.UpdateCurrentDelay(timestamp);
>   const int kRenderDelayMs = 10;
>   timing.set_render_delay(kRenderDelayMs);
>-  waitTime = timing.MaxWaitingTime(
>-      timing.RenderTimeMs(timeStamp, clock.TimeInMilliseconds()),
>+  wait_time_ms = timing.MaxWaitingTime(
>+      timing.RenderTimeMs(timestamp, clock.TimeInMilliseconds()),
>       clock.TimeInMilliseconds());
>-  // We should at least have minTotalDelayMs - decodeTime (10) - renderTime
>+  // We should at least have kMinTotalDelayMs - decodeTime (10) - renderTime
>   // (10) to wait.
>-  EXPECT_EQ(waitTime, minTotalDelayMs - requiredDecodeTimeMs - kRenderDelayMs);
>+  EXPECT_EQ(kMinTotalDelayMs - kDecodeTimeMs - kRenderDelayMs, wait_time_ms);
>   // The total video delay should be equal to the min total delay.
>-  EXPECT_EQ(minTotalDelayMs, timing.TargetVideoDelay());
>+  EXPECT_EQ(kMinTotalDelayMs, timing.TargetVideoDelay());
> 
>   // Reset playout delay.
>   timing.set_min_playout_delay(0);
>   clock.AdvanceTimeMilliseconds(5000);
>-  timeStamp += 5 * 90000;
>-  timing.UpdateCurrentDelay(timeStamp);
>+  timestamp += 5 * 90000;
>+  timing.UpdateCurrentDelay(timestamp);
> }
> 
> TEST(ReceiverTiming, WrapAround) {
>-  const int kFramerate = 25;
>   SimulatedClock clock(0);
>   VCMTiming timing(&clock);
>-  // Provoke a wrap-around. The forth frame will have wrapped at 25 fps.
>-  uint32_t timestamp = 0xFFFFFFFFu - 3 * 90000 / kFramerate;
>-  for (int i = 0; i < 4; ++i) {
>+  // Provoke a wrap-around. The fifth frame will have wrapped at 25 fps.
>+  uint32_t timestamp = 0xFFFFFFFFu - 3 * 90000 / kFps;
>+  for (int i = 0; i < 5; ++i) {
>     timing.IncomingTimestamp(timestamp, clock.TimeInMilliseconds());
>-    clock.AdvanceTimeMilliseconds(1000 / kFramerate);
>-    timestamp += 90000 / kFramerate;
>-    int64_t render_time =
>-        timing.RenderTimeMs(0xFFFFFFFFu, clock.TimeInMilliseconds());
>-    EXPECT_EQ(3 * 1000 / kFramerate, render_time);
>-    render_time = timing.RenderTimeMs(89u,  // One second later in 90 kHz.
>-                                      clock.TimeInMilliseconds());
>-    EXPECT_EQ(3 * 1000 / kFramerate + 1, render_time);
>+    clock.AdvanceTimeMilliseconds(1000 / kFps);
>+    timestamp += 90000 / kFps;
>+    EXPECT_EQ(3 * 1000 / kFps,
>+              timing.RenderTimeMs(0xFFFFFFFFu, clock.TimeInMilliseconds()));
>+    EXPECT_EQ(3 * 1000 / kFps + 1,
>+              timing.RenderTimeMs(89u,  // One ms later in 90 kHz.
>+                                  clock.TimeInMilliseconds()));
>   }
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/default_video_bitrate_allocator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/default_video_bitrate_allocator.cc
>index 5e3b3a2c5d5..a4a40dd08bc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/default_video_bitrate_allocator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/default_video_bitrate_allocator.cc
>@@ -20,11 +20,11 @@ DefaultVideoBitrateAllocator::DefaultVideoBitrateAllocator(
> 
> DefaultVideoBitrateAllocator::~DefaultVideoBitrateAllocator() {}
> 
>-BitrateAllocation DefaultVideoBitrateAllocator::GetAllocation(
>+VideoBitrateAllocation DefaultVideoBitrateAllocator::GetAllocation(
>     uint32_t total_bitrate_bps,
>     uint32_t framerate) {
>-  BitrateAllocation allocation;
>-  if (total_bitrate_bps == 0)
>+  VideoBitrateAllocation allocation;
>+  if (total_bitrate_bps == 0 || !codec_.active)
>     return allocation;
> 
>   if (total_bitrate_bps < codec_.minBitrate * 1000) {
>@@ -38,9 +38,4 @@ BitrateAllocation DefaultVideoBitrateAllocator::GetAllocation(
>   return allocation;
> }
> 
>-uint32_t DefaultVideoBitrateAllocator::GetPreferredBitrateBps(
>-    uint32_t framerate) {
>-  return GetAllocation(codec_.maxBitrate * 1000, framerate).get_sum_bps();
>-}
>-
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/default_video_bitrate_allocator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/default_video_bitrate_allocator.h
>index 0ef709efa79..6a9d040de24 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/default_video_bitrate_allocator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/default_video_bitrate_allocator.h
>@@ -11,7 +11,8 @@
> #ifndef MODULES_VIDEO_CODING_UTILITY_DEFAULT_VIDEO_BITRATE_ALLOCATOR_H_
> #define MODULES_VIDEO_CODING_UTILITY_DEFAULT_VIDEO_BITRATE_ALLOCATOR_H_
> 
>-#include "common_video/include/video_bitrate_allocator.h"
>+#include "api/video/video_bitrate_allocator.h"
>+#include "api/video_codecs/video_codec.h"
> 
> namespace webrtc {
> 
>@@ -20,9 +21,8 @@ class DefaultVideoBitrateAllocator : public VideoBitrateAllocator {
>   explicit DefaultVideoBitrateAllocator(const VideoCodec& codec);
>   ~DefaultVideoBitrateAllocator() override;
> 
>-  BitrateAllocation GetAllocation(uint32_t total_bitrate,
>-                                  uint32_t framerate) override;
>-  uint32_t GetPreferredBitrateBps(uint32_t framerate) override;
>+  VideoBitrateAllocation GetAllocation(uint32_t total_bitrate,
>+                                       uint32_t framerate) override;
> 
>  private:
>   const VideoCodec codec_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/default_video_bitrate_allocator_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/default_video_bitrate_allocator_unittest.cc
>index 695ac689e88..78fcdaada09 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/default_video_bitrate_allocator_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/default_video_bitrate_allocator_unittest.cc
>@@ -30,7 +30,6 @@ class DefaultVideoBitrateAllocatorTest : public ::testing::Test {
>     codec_.codecType = kVideoCodecVP8;
>     codec_.minBitrate = kMinBitrateBps / 1000;
>     codec_.maxBitrate = kMaxBitrateBps / 1000;
>-    codec_.targetBitrate = (kMinBitrateBps + kMaxBitrateBps) / 2000;
>     codec_.maxFramerate = kMaxFramerate;
>     allocator_.reset(new DefaultVideoBitrateAllocator(codec_));
>   }
>@@ -41,12 +40,22 @@ class DefaultVideoBitrateAllocatorTest : public ::testing::Test {
> };
> 
> TEST_F(DefaultVideoBitrateAllocatorTest, ZeroIsOff) {
>-  BitrateAllocation allocation = allocator_->GetAllocation(0, kMaxFramerate);
>+  VideoBitrateAllocation allocation =
>+      allocator_->GetAllocation(0, kMaxFramerate);
>+  EXPECT_EQ(0u, allocation.get_sum_bps());
>+}
>+
>+TEST_F(DefaultVideoBitrateAllocatorTest, Inactive) {
>+  codec_.active = false;
>+  allocator_.reset(new DefaultVideoBitrateAllocator(codec_));
>+  VideoBitrateAllocation allocation =
>+      allocator_->GetAllocation(1, kMaxFramerate);
>   EXPECT_EQ(0u, allocation.get_sum_bps());
> }
> 
> TEST_F(DefaultVideoBitrateAllocatorTest, CapsToMin) {
>-  BitrateAllocation allocation = allocator_->GetAllocation(1, kMaxFramerate);
>+  VideoBitrateAllocation allocation =
>+      allocator_->GetAllocation(1, kMaxFramerate);
>   EXPECT_EQ(kMinBitrateBps, allocation.get_sum_bps());
> 
>   allocation = allocator_->GetAllocation(kMinBitrateBps - 1, kMaxFramerate);
>@@ -57,7 +66,7 @@ TEST_F(DefaultVideoBitrateAllocatorTest, CapsToMin) {
> }
> 
> TEST_F(DefaultVideoBitrateAllocatorTest, CapsToMax) {
>-  BitrateAllocation allocation =
>+  VideoBitrateAllocation allocation =
>       allocator_->GetAllocation(kMaxBitrateBps, kMaxFramerate);
>   EXPECT_EQ(kMaxBitrateBps, allocation.get_sum_bps());
> 
>@@ -70,7 +79,7 @@ TEST_F(DefaultVideoBitrateAllocatorTest, CapsToMax) {
> }
> 
> TEST_F(DefaultVideoBitrateAllocatorTest, GoodInBetween) {
>-  BitrateAllocation allocation =
>+  VideoBitrateAllocation allocation =
>       allocator_->GetAllocation(kMinBitrateBps + 1, kMaxFramerate);
>   EXPECT_EQ(kMinBitrateBps + 1, allocation.get_sum_bps());
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/frame_dropper.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/frame_dropper.h
>index 224f31b77fd..2764994b25e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/frame_dropper.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/frame_dropper.h
>@@ -11,10 +11,10 @@
> #ifndef MODULES_VIDEO_CODING_UTILITY_FRAME_DROPPER_H_
> #define MODULES_VIDEO_CODING_UTILITY_FRAME_DROPPER_H_
> 
>-#include <cstddef>
>+#include <stddef.h>
>+#include <stdint.h>
> 
> #include "rtc_base/numerics/exp_filter.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/ivf_file_writer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/ivf_file_writer.cc
>index 4b2cf3dda5f..454aeb08934 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/ivf_file_writer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/ivf_file_writer.cc
>@@ -25,7 +25,7 @@ namespace webrtc {
> const size_t kIvfHeaderSize = 32;
> 
> IvfFileWriter::IvfFileWriter(rtc::File file, size_t byte_limit)
>-    : codec_type_(kVideoCodecUnknown),
>+    : codec_type_(kVideoCodecGeneric),
>       bytes_written_(0),
>       byte_limit_(byte_limit),
>       num_frames_(0),
>@@ -122,8 +122,7 @@ bool IvfFileWriter::InitFromFirstFrame(const EncodedImage& encoded_image,
>   if (!WriteHeader())
>     return false;
> 
>-  const char* codec_name =
>-      CodecTypeToPayloadString(codec_type_);
>+  const char* codec_name = CodecTypeToPayloadString(codec_type_);
>   RTC_LOG(LS_WARNING) << "Created IVF file for codec data of type "
>                       << codec_name << " at resolution " << width_ << " x "
>                       << height_ << ", using "
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/mock/mock_frame_dropper.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/mock/mock_frame_dropper.h
>index 91ab221eed1..a0f22200808 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/mock/mock_frame_dropper.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/mock/mock_frame_dropper.h
>@@ -14,7 +14,6 @@
> 
> #include "modules/video_coding/utility/frame_dropper.h"
> #include "test/gmock.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/moving_average.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/moving_average.cc
>index 8076f934766..eb23e3d92f3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/moving_average.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/moving_average.cc
>@@ -15,6 +15,7 @@
> namespace webrtc {
> 
> MovingAverage::MovingAverage(size_t s) : sum_history_(s + 1, 0) {}
>+MovingAverage::~MovingAverage() = default;
> 
> void MovingAverage::AddSample(int sample) {
>   count_++;
>@@ -22,13 +23,13 @@ void MovingAverage::AddSample(int sample) {
>   sum_history_[count_ % sum_history_.size()] = sum_;
> }
> 
>-rtc::Optional<int> MovingAverage::GetAverage() const {
>+absl::optional<int> MovingAverage::GetAverage() const {
>   return GetAverage(size());
> }
> 
>-rtc::Optional<int> MovingAverage::GetAverage(size_t num_samples) const {
>+absl::optional<int> MovingAverage::GetAverage(size_t num_samples) const {
>   if (num_samples > size() || num_samples == 0)
>-    return rtc::nullopt;
>+    return absl::nullopt;
>   int sum = sum_ - sum_history_[(count_ - num_samples) % sum_history_.size()];
>   return sum / static_cast<int>(num_samples);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/moving_average.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/moving_average.h
>index 7df3d3dad60..77ce633b18f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/moving_average.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/moving_average.h
>@@ -13,15 +13,16 @@
> 
> #include <vector>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> 
> namespace webrtc {
> class MovingAverage {
>  public:
>   explicit MovingAverage(size_t s);
>+  ~MovingAverage();
>   void AddSample(int sample);
>-  rtc::Optional<int> GetAverage() const;
>-  rtc::Optional<int> GetAverage(size_t num_samples) const;
>+  absl::optional<int> GetAverage() const;
>+  absl::optional<int> GetAverage(size_t num_samples) const;
>   void Reset();
>   size_t size() const;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/quality_scaler.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/quality_scaler.cc
>index 447506e0fb9..ea19e418c9d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/quality_scaler.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/quality_scaler.cc
>@@ -17,7 +17,9 @@
> 
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>+#include "rtc_base/numerics/exp_filter.h"
> #include "rtc_base/task_queue.h"
>+#include "rtc_base/timeutils.h"
> 
> // TODO(kthelgason): Some versions of Android have issues with log2.
> // See https://code.google.com/p/android/issues/detail?id=212634 for details
>@@ -28,52 +30,46 @@
> namespace webrtc {
> 
> namespace {
>+// TODO(nisse): Delete, delegate to encoders.
> // Threshold constant used until first downscale (to permit fast rampup).
> static const int kMeasureMs = 2000;
> static const float kSamplePeriodScaleFactor = 2.5;
> static const int kFramedropPercentThreshold = 60;
>-// QP scaling threshold defaults:
>-static const int kLowH264QpThreshold = 24;
>-static const int kHighH264QpThreshold = 37;
>-// QP is obtained from VP8-bitstream for HW, so the QP corresponds to the
>-// bitstream range of [0, 127] and not the user-level range of [0,63].
>-static const int kLowVp8QpThreshold = 29;
>-static const int kHighVp8QpThreshold = 95;
>-// QP is obtained from VP9-bitstream for HW, so the QP corresponds to the
>-// bitstream range of [0, 255] and not the user-level range of [0,63].
>-// Current VP9 settings are mapped from VP8 thresholds above.
>-static const int kLowVp9QpThreshold = 96;
>-static const int kHighVp9QpThreshold = 185;
> static const int kMinFramesNeededToScale = 2 * 30;
> 
>-static VideoEncoder::QpThresholds CodecTypeToDefaultThresholds(
>-    VideoCodecType codec_type) {
>-  int low = -1;
>-  int high = -1;
>-  switch (codec_type) {
>-    case kVideoCodecH264:
>-      low = kLowH264QpThreshold;
>-      high = kHighH264QpThreshold;
>-      break;
>-    case kVideoCodecVP8:
>-      low = kLowVp8QpThreshold;
>-      high = kHighVp8QpThreshold;
>-      break;
>-    case kVideoCodecVP9:
>-      low = kLowVp9QpThreshold;
>-      high = kHighVp9QpThreshold;
>-      break;
>-    default:
>-      RTC_NOTREACHED() << "Invalid codec type for QualityScaler.";
>-  }
>-  return VideoEncoder::QpThresholds(low, high);
>-}
> }  // namespace
> 
>-class QualityScaler::CheckQPTask : public rtc::QueuedTask {
>+class QualityScaler::QpSmoother {
>  public:
>-  explicit CheckQPTask(QualityScaler* scaler) : scaler_(scaler) {
>-    RTC_LOG(LS_INFO) << "Created CheckQPTask. Scheduling on queue...";
>+  explicit QpSmoother(float alpha)
>+      : alpha_(alpha), last_sample_ms_(rtc::TimeMillis()), smoother_(alpha) {}
>+
>+  absl::optional<int> GetAvg() const {
>+    float value = smoother_.filtered();
>+    if (value == rtc::ExpFilter::kValueUndefined) {
>+      return absl::nullopt;
>+    }
>+    return static_cast<int>(value);
>+  }
>+
>+  void Add(float sample) {
>+    int64_t now_ms = rtc::TimeMillis();
>+    smoother_.Apply(static_cast<float>(now_ms - last_sample_ms_), sample);
>+    last_sample_ms_ = now_ms;
>+  }
>+
>+  void Reset() { smoother_.Reset(alpha_); }
>+
>+ private:
>+  const float alpha_;
>+  int64_t last_sample_ms_;
>+  rtc::ExpFilter smoother_;
>+};
>+
>+class QualityScaler::CheckQpTask : public rtc::QueuedTask {
>+ public:
>+  explicit CheckQpTask(QualityScaler* scaler) : scaler_(scaler) {
>+    RTC_LOG(LS_INFO) << "Created CheckQpTask. Scheduling on queue...";
>     rtc::TaskQueue::Current()->PostDelayedTask(
>         std::unique_ptr<rtc::QueuedTask>(this), scaler_->GetSamplingPeriodMs());
>   }
>@@ -88,7 +84,7 @@ class QualityScaler::CheckQPTask : public rtc::QueuedTask {
>     RTC_DCHECK_CALLED_SEQUENTIALLY(&task_checker_);
>     if (stop_)
>       return true;  // TaskQueue will free this task.
>-    scaler_->CheckQP();
>+    scaler_->CheckQp();
>     rtc::TaskQueue::Current()->PostDelayedTask(
>         std::unique_ptr<rtc::QueuedTask>(this), scaler_->GetSamplingPeriodMs());
>     return false;  // Retain the task in order to reuse it.
>@@ -99,10 +95,6 @@ class QualityScaler::CheckQPTask : public rtc::QueuedTask {
>   rtc::SequencedTaskChecker task_checker_;
> };
> 
>-QualityScaler::QualityScaler(AdaptationObserverInterface* observer,
>-                             VideoCodecType codec_type)
>-    : QualityScaler(observer, CodecTypeToDefaultThresholds(codec_type)) {}
>-
> QualityScaler::QualityScaler(AdaptationObserverInterface* observer,
>                              VideoEncoder::QpThresholds thresholds)
>     : QualityScaler(observer, thresholds, kMeasureMs) {}
>@@ -110,18 +102,26 @@ QualityScaler::QualityScaler(AdaptationObserverInterface* observer,
> // Protected ctor, should not be called directly.
> QualityScaler::QualityScaler(AdaptationObserverInterface* observer,
>                              VideoEncoder::QpThresholds thresholds,
>-                             int64_t sampling_period)
>+                             int64_t sampling_period_ms)
>     : check_qp_task_(nullptr),
>       observer_(observer),
>-      sampling_period_ms_(sampling_period),
>+      thresholds_(thresholds),
>+      sampling_period_ms_(sampling_period_ms),
>       fast_rampup_(true),
>       // Arbitrarily choose size based on 30 fps for 5 seconds.
>       average_qp_(5 * 30),
>-      framedrop_percent_(5 * 30),
>-      thresholds_(thresholds) {
>+      framedrop_percent_media_opt_(5 * 30),
>+      framedrop_percent_all_(5 * 30),
>+      experiment_enabled_(QualityScalingExperiment::Enabled()),
>+      observed_enough_frames_(false) {
>   RTC_DCHECK_CALLED_SEQUENTIALLY(&task_checker_);
>+  if (experiment_enabled_) {
>+    config_ = QualityScalingExperiment::GetConfig();
>+    qp_smoother_high_.reset(new QpSmoother(config_.alpha_high));
>+    qp_smoother_low_.reset(new QpSmoother(config_.alpha_low));
>+  }
>   RTC_DCHECK(observer_ != nullptr);
>-  check_qp_task_ = new CheckQPTask(this);
>+  check_qp_task_ = new CheckQpTask(this);
>   RTC_LOG(LS_INFO) << "QP thresholds: low: " << thresholds_.low
>                    << ", high: " << thresholds_.high;
> }
>@@ -133,61 +133,92 @@ QualityScaler::~QualityScaler() {
> 
> int64_t QualityScaler::GetSamplingPeriodMs() const {
>   RTC_DCHECK_CALLED_SEQUENTIALLY(&task_checker_);
>-  return fast_rampup_ ? sampling_period_ms_
>-                      : (sampling_period_ms_ * kSamplePeriodScaleFactor);
>+  if (fast_rampup_) {
>+    return sampling_period_ms_;
>+  }
>+  if (experiment_enabled_ && !observed_enough_frames_) {
>+    // Use half the interval while waiting for enough frames.
>+    return sampling_period_ms_ / 2;
>+  }
>+  return sampling_period_ms_ * kSamplePeriodScaleFactor;
> }
> 
>-void QualityScaler::ReportDroppedFrame() {
>+void QualityScaler::ReportDroppedFrameByMediaOpt() {
>   RTC_DCHECK_CALLED_SEQUENTIALLY(&task_checker_);
>-  framedrop_percent_.AddSample(100);
>+  framedrop_percent_media_opt_.AddSample(100);
>+  framedrop_percent_all_.AddSample(100);
> }
> 
>-void QualityScaler::ReportQP(int qp) {
>+void QualityScaler::ReportDroppedFrameByEncoder() {
>   RTC_DCHECK_CALLED_SEQUENTIALLY(&task_checker_);
>-  framedrop_percent_.AddSample(0);
>+  framedrop_percent_all_.AddSample(100);
>+}
>+
>+void QualityScaler::ReportQp(int qp) {
>+  RTC_DCHECK_CALLED_SEQUENTIALLY(&task_checker_);
>+  framedrop_percent_media_opt_.AddSample(0);
>+  framedrop_percent_all_.AddSample(0);
>   average_qp_.AddSample(qp);
>+  if (qp_smoother_high_)
>+    qp_smoother_high_->Add(qp);
>+  if (qp_smoother_low_)
>+    qp_smoother_low_->Add(qp);
> }
> 
>-void QualityScaler::CheckQP() {
>+void QualityScaler::CheckQp() {
>   RTC_DCHECK_CALLED_SEQUENTIALLY(&task_checker_);
>   // Should be set through InitEncode -> Should be set by now.
>   RTC_DCHECK_GE(thresholds_.low, 0);
> 
>-  // If we have not observed at least this many frames we can't
>-  // make a good scaling decision.
>-  if (framedrop_percent_.size() < kMinFramesNeededToScale)
>+  // If we have not observed at least this many frames we can't make a good
>+  // scaling decision.
>+  const size_t frames = config_.use_all_drop_reasons
>+                            ? framedrop_percent_all_.size()
>+                            : framedrop_percent_media_opt_.size();
>+  if (frames < kMinFramesNeededToScale) {
>+    observed_enough_frames_ = false;
>     return;
>+  }
>+  observed_enough_frames_ = true;
> 
>   // Check if we should scale down due to high frame drop.
>-  const rtc::Optional<int> drop_rate = framedrop_percent_.GetAverage();
>+  const absl::optional<int> drop_rate =
>+      config_.use_all_drop_reasons ? framedrop_percent_all_.GetAverage()
>+                                   : framedrop_percent_media_opt_.GetAverage();
>   if (drop_rate && *drop_rate >= kFramedropPercentThreshold) {
>-    ReportQPHigh();
>+    RTC_LOG(LS_INFO) << "Reporting high QP, framedrop percent " << *drop_rate;
>+    ReportQpHigh();
>     return;
>   }
> 
>   // Check if we should scale up or down based on QP.
>-  const rtc::Optional<int> avg_qp = average_qp_.GetAverage();
>-  if (avg_qp) {
>-    RTC_LOG(LS_INFO) << "Checking average QP " << *avg_qp;
>-    if (*avg_qp > thresholds_.high) {
>-      ReportQPHigh();
>+  const absl::optional<int> avg_qp_high = qp_smoother_high_
>+                                              ? qp_smoother_high_->GetAvg()
>+                                              : average_qp_.GetAverage();
>+  const absl::optional<int> avg_qp_low =
>+      qp_smoother_low_ ? qp_smoother_low_->GetAvg() : average_qp_.GetAverage();
>+  if (avg_qp_high && avg_qp_low) {
>+    RTC_LOG(LS_INFO) << "Checking average QP " << *avg_qp_high << " ("
>+                     << *avg_qp_low << ").";
>+    if (*avg_qp_high > thresholds_.high) {
>+      ReportQpHigh();
>       return;
>     }
>-    if (*avg_qp <= thresholds_.low) {
>+    if (*avg_qp_low <= thresholds_.low) {
>       // QP has been low. We want to try a higher resolution.
>-      ReportQPLow();
>+      ReportQpLow();
>       return;
>     }
>   }
> }
> 
>-void QualityScaler::ReportQPLow() {
>+void QualityScaler::ReportQpLow() {
>   RTC_DCHECK_CALLED_SEQUENTIALLY(&task_checker_);
>   ClearSamples();
>   observer_->AdaptUp(AdaptationObserverInterface::AdaptReason::kQuality);
> }
> 
>-void QualityScaler::ReportQPHigh() {
>+void QualityScaler::ReportQpHigh() {
>   RTC_DCHECK_CALLED_SEQUENTIALLY(&task_checker_);
>   ClearSamples();
>   observer_->AdaptDown(AdaptationObserverInterface::AdaptReason::kQuality);
>@@ -199,7 +230,12 @@ void QualityScaler::ReportQPHigh() {
> 
> void QualityScaler::ClearSamples() {
>   RTC_DCHECK_CALLED_SEQUENTIALLY(&task_checker_);
>-  framedrop_percent_.Reset();
>+  framedrop_percent_media_opt_.Reset();
>+  framedrop_percent_all_.Reset();
>   average_qp_.Reset();
>+  if (qp_smoother_high_)
>+    qp_smoother_high_->Reset();
>+  if (qp_smoother_low_)
>+    qp_smoother_low_->Reset();
> }
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/quality_scaler.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/quality_scaler.h
>index ad998a295de..3091ebf36c4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/quality_scaler.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/quality_scaler.h
>@@ -11,12 +11,14 @@
> #ifndef MODULES_VIDEO_CODING_UTILITY_QUALITY_SCALER_H_
> #define MODULES_VIDEO_CODING_UTILITY_QUALITY_SCALER_H_
> 
>+#include <memory>
> #include <utility>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> #include "api/video_codecs/video_encoder.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/video_coding/utility/moving_average.h"
>+#include "rtc_base/experiments/quality_scaling_experiment.h"
> #include "rtc_base/sequenced_task_checker.h"
> 
> namespace webrtc {
>@@ -43,44 +45,50 @@ class AdaptationObserverInterface {
> // signal an intent to scale up or down.
> class QualityScaler {
>  public:
>-  // Construct a QualityScaler with a given |observer|.
>+  // Construct a QualityScaler with given |thresholds| and |observer|.
>   // This starts the quality scaler periodically checking what the average QP
>   // has been recently.
>-  QualityScaler(AdaptationObserverInterface* observer,
>-                VideoCodecType codec_type);
>-  // If specific thresholds are desired these can be supplied as |thresholds|.
>   QualityScaler(AdaptationObserverInterface* observer,
>                 VideoEncoder::QpThresholds thresholds);
>   virtual ~QualityScaler();
>-  // Should be called each time the encoder drops a frame
>-  void ReportDroppedFrame();
>+  // Should be called each time a frame is dropped at encoding.
>+  void ReportDroppedFrameByMediaOpt();
>+  void ReportDroppedFrameByEncoder();
>   // Inform the QualityScaler of the last seen QP.
>-  void ReportQP(int qp);
>+  void ReportQp(int qp);
> 
>-  // The following members declared protected for testing purposes
>+  // The following members declared protected for testing purposes.
>  protected:
>   QualityScaler(AdaptationObserverInterface* observer,
>                 VideoEncoder::QpThresholds thresholds,
>-                int64_t sampling_period);
>+                int64_t sampling_period_ms);
> 
>  private:
>-  class CheckQPTask;
>-  void CheckQP();
>+  class CheckQpTask;
>+  class QpSmoother;
>+  void CheckQp();
>   void ClearSamples();
>-  void ReportQPLow();
>-  void ReportQPHigh();
>+  void ReportQpLow();
>+  void ReportQpHigh();
>   int64_t GetSamplingPeriodMs() const;
> 
>-  CheckQPTask* check_qp_task_ RTC_GUARDED_BY(&task_checker_);
>+  CheckQpTask* check_qp_task_ RTC_GUARDED_BY(&task_checker_);
>   AdaptationObserverInterface* const observer_ RTC_GUARDED_BY(&task_checker_);
>   rtc::SequencedTaskChecker task_checker_;
> 
>+  const VideoEncoder::QpThresholds thresholds_;
>   const int64_t sampling_period_ms_;
>   bool fast_rampup_ RTC_GUARDED_BY(&task_checker_);
>   MovingAverage average_qp_ RTC_GUARDED_BY(&task_checker_);
>-  MovingAverage framedrop_percent_ RTC_GUARDED_BY(&task_checker_);
>+  MovingAverage framedrop_percent_media_opt_ RTC_GUARDED_BY(&task_checker_);
>+  MovingAverage framedrop_percent_all_ RTC_GUARDED_BY(&task_checker_);
> 
>-  VideoEncoder::QpThresholds thresholds_ RTC_GUARDED_BY(&task_checker_);
>+  // Used by QualityScalingExperiment.
>+  const bool experiment_enabled_;
>+  QualityScalingExperiment::Config config_ RTC_GUARDED_BY(&task_checker_);
>+  std::unique_ptr<QpSmoother> qp_smoother_high_ RTC_GUARDED_BY(&task_checker_);
>+  std::unique_ptr<QpSmoother> qp_smoother_low_ RTC_GUARDED_BY(&task_checker_);
>+  bool observed_enough_frames_ RTC_GUARDED_BY(&task_checker_);
> };
> }  // namespace webrtc
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/quality_scaler_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/quality_scaler_unittest.cc
>index d9da1b4b330..b17062df5c1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/quality_scaler_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/quality_scaler_unittest.cc
>@@ -11,9 +11,11 @@
> #include "modules/video_coding/utility/quality_scaler.h"
> 
> #include <memory>
>+#include <string>
> 
> #include "rtc_base/event.h"
> #include "rtc_base/task_queue.h"
>+#include "test/field_trial.h"
> #include "test/gmock.h"
> #include "test/gtest.h"
> 
>@@ -21,21 +23,21 @@ namespace webrtc {
> namespace {
> static const int kFramerate = 30;
> static const int kLowQp = 15;
>-static const int kLowQpThreshold = 18;
> static const int kHighQp = 40;
>+static const int kMinFramesNeededToScale = 60;  // From quality_scaler.cc.
> static const size_t kDefaultTimeoutMs = 150;
> }  // namespace
> 
>-#define DO_SYNC(q, block) do {    \
>-  rtc::Event event(false, false); \
>-  q->PostTask([this, &event] {    \
>-    block;                        \
>-    event.Set();                  \
>-  });                             \
>-  RTC_CHECK(event.Wait(1000));    \
>+#define DO_SYNC(q, block)           \
>+  do {                              \
>+    rtc::Event event(false, false); \
>+    q->PostTask([this, &event] {    \
>+      block;                        \
>+      event.Set();                  \
>+    });                             \
>+    RTC_CHECK(event.Wait(1000));    \
>   } while (0)
> 
>-
> class MockAdaptationObserver : public AdaptationObserverInterface {
>  public:
>   MockAdaptationObserver() : event(false, false) {}
>@@ -63,9 +65,11 @@ class QualityScalerUnderTest : public QualityScaler {
>       : QualityScaler(observer, thresholds, 5) {}
> };
> 
>-class QualityScalerTest : public ::testing::Test {
>+class QualityScalerTest : public ::testing::Test,
>+                          public ::testing::WithParamInterface<std::string> {
>  protected:
>   enum ScaleDirection {
>+    kKeepScaleAboveLowQp,
>     kKeepScaleAtHighQp,
>     kScaleDown,
>     kScaleDownAboveHighQp,
>@@ -73,68 +77,81 @@ class QualityScalerTest : public ::testing::Test {
>   };
> 
>   QualityScalerTest()
>-      : q_(new rtc::TaskQueue("QualityScalerTestQueue")),
>+      : scoped_field_trial_(GetParam()),
>+        q_(new rtc::TaskQueue("QualityScalerTestQueue")),
>         observer_(new MockAdaptationObserver()) {
>     DO_SYNC(q_, {
>       qs_ = std::unique_ptr<QualityScaler>(new QualityScalerUnderTest(
>-          observer_.get(),
>-          VideoEncoder::QpThresholds(kLowQpThreshold, kHighQp)));});
>+          observer_.get(), VideoEncoder::QpThresholds(kLowQp, kHighQp)));
>+    });
>   }
> 
>   ~QualityScalerTest() {
>-    DO_SYNC(q_, {qs_.reset(nullptr);});
>+    DO_SYNC(q_, { qs_.reset(nullptr); });
>   }
> 
>   void TriggerScale(ScaleDirection scale_direction) {
>     for (int i = 0; i < kFramerate * 5; ++i) {
>       switch (scale_direction) {
>+        case kKeepScaleAboveLowQp:
>+          qs_->ReportQp(kLowQp + 1);
>+          break;
>         case kScaleUp:
>-          qs_->ReportQP(kLowQp);
>+          qs_->ReportQp(kLowQp);
>           break;
>         case kScaleDown:
>-          qs_->ReportDroppedFrame();
>+          qs_->ReportDroppedFrameByMediaOpt();
>           break;
>         case kKeepScaleAtHighQp:
>-          qs_->ReportQP(kHighQp);
>+          qs_->ReportQp(kHighQp);
>           break;
>         case kScaleDownAboveHighQp:
>-          qs_->ReportQP(kHighQp + 1);
>+          qs_->ReportQp(kHighQp + 1);
>           break;
>       }
>     }
>   }
> 
>+  test::ScopedFieldTrials scoped_field_trial_;
>   std::unique_ptr<rtc::TaskQueue> q_;
>   std::unique_ptr<QualityScaler> qs_;
>   std::unique_ptr<MockAdaptationObserver> observer_;
> };
> 
>-TEST_F(QualityScalerTest, DownscalesAfterContinuousFramedrop) {
>+INSTANTIATE_TEST_CASE_P(
>+    FieldTrials,
>+    QualityScalerTest,
>+    ::testing::Values(
>+        "WebRTC-Video-QualityScaling/Enabled-1,2,3,4,5,6,7,8,0.9,0.99,1/",
>+        ""));
>+
>+TEST_P(QualityScalerTest, DownscalesAfterContinuousFramedrop) {
>   DO_SYNC(q_, { TriggerScale(kScaleDown); });
>   EXPECT_TRUE(observer_->event.Wait(kDefaultTimeoutMs));
>   EXPECT_EQ(1, observer_->adapt_down_events_);
>+  EXPECT_EQ(0, observer_->adapt_up_events_);
> }
> 
>-TEST_F(QualityScalerTest, KeepsScaleAtHighQp) {
>+TEST_P(QualityScalerTest, KeepsScaleAtHighQp) {
>   DO_SYNC(q_, { TriggerScale(kKeepScaleAtHighQp); });
>   EXPECT_FALSE(observer_->event.Wait(kDefaultTimeoutMs));
>   EXPECT_EQ(0, observer_->adapt_down_events_);
>   EXPECT_EQ(0, observer_->adapt_up_events_);
> }
> 
>-TEST_F(QualityScalerTest, DownscalesAboveHighQp) {
>+TEST_P(QualityScalerTest, DownscalesAboveHighQp) {
>   DO_SYNC(q_, { TriggerScale(kScaleDownAboveHighQp); });
>   EXPECT_TRUE(observer_->event.Wait(kDefaultTimeoutMs));
>   EXPECT_EQ(1, observer_->adapt_down_events_);
>   EXPECT_EQ(0, observer_->adapt_up_events_);
> }
> 
>-TEST_F(QualityScalerTest, DownscalesAfterTwoThirdsFramedrop) {
>+TEST_P(QualityScalerTest, DownscalesAfterTwoThirdsFramedrop) {
>   DO_SYNC(q_, {
>     for (int i = 0; i < kFramerate * 5; ++i) {
>-      qs_->ReportDroppedFrame();
>-      qs_->ReportDroppedFrame();
>-      qs_->ReportQP(kHighQp);
>+      qs_->ReportDroppedFrameByMediaOpt();
>+      qs_->ReportDroppedFrameByMediaOpt();
>+      qs_->ReportQp(kHighQp);
>     }
>   });
>   EXPECT_TRUE(observer_->event.Wait(kDefaultTimeoutMs));
>@@ -142,33 +159,47 @@ TEST_F(QualityScalerTest, DownscalesAfterTwoThirdsFramedrop) {
>   EXPECT_EQ(0, observer_->adapt_up_events_);
> }
> 
>-TEST_F(QualityScalerTest, DoesNotDownscaleOnNormalQp) {
>-  DO_SYNC(q_, { TriggerScale(kScaleDownAboveHighQp); });
>-  EXPECT_TRUE(observer_->event.Wait(kDefaultTimeoutMs));
>-  EXPECT_EQ(1, observer_->adapt_down_events_);
>+TEST_P(QualityScalerTest, DoesNotDownscaleAfterHalfFramedrop) {
>+  DO_SYNC(q_, {
>+    for (int i = 0; i < kFramerate * 5; ++i) {
>+      qs_->ReportDroppedFrameByMediaOpt();
>+      qs_->ReportQp(kHighQp);
>+    }
>+  });
>+  EXPECT_FALSE(observer_->event.Wait(kDefaultTimeoutMs));
>+  EXPECT_EQ(0, observer_->adapt_down_events_);
>   EXPECT_EQ(0, observer_->adapt_up_events_);
> }
> 
>-TEST_F(QualityScalerTest, DoesNotDownscaleAfterHalfFramedrop) {
>+TEST_P(QualityScalerTest, DownscalesAfterTwoThirdsIfFieldTrialEnabled) {
>+  const bool kDownScaleExpected = !GetParam().empty();
>   DO_SYNC(q_, {
>     for (int i = 0; i < kFramerate * 5; ++i) {
>-      qs_->ReportDroppedFrame();
>-      qs_->ReportQP(kHighQp);
>+      qs_->ReportDroppedFrameByMediaOpt();
>+      qs_->ReportDroppedFrameByEncoder();
>+      qs_->ReportQp(kHighQp);
>     }
>   });
>+  EXPECT_EQ(kDownScaleExpected, observer_->event.Wait(kDefaultTimeoutMs));
>+  EXPECT_EQ(kDownScaleExpected ? 1 : 0, observer_->adapt_down_events_);
>+  EXPECT_EQ(0, observer_->adapt_up_events_);
>+}
>+
>+TEST_P(QualityScalerTest, KeepsScaleOnNormalQp) {
>+  DO_SYNC(q_, { TriggerScale(kKeepScaleAboveLowQp); });
>   EXPECT_FALSE(observer_->event.Wait(kDefaultTimeoutMs));
>   EXPECT_EQ(0, observer_->adapt_down_events_);
>   EXPECT_EQ(0, observer_->adapt_up_events_);
> }
> 
>-TEST_F(QualityScalerTest, UpscalesAfterLowQp) {
>+TEST_P(QualityScalerTest, UpscalesAfterLowQp) {
>   DO_SYNC(q_, { TriggerScale(kScaleUp); });
>   EXPECT_TRUE(observer_->event.Wait(kDefaultTimeoutMs));
>   EXPECT_EQ(0, observer_->adapt_down_events_);
>   EXPECT_EQ(1, observer_->adapt_up_events_);
> }
> 
>-TEST_F(QualityScalerTest, ScalesDownAndBackUp) {
>+TEST_P(QualityScalerTest, ScalesDownAndBackUp) {
>   DO_SYNC(q_, { TriggerScale(kScaleDown); });
>   EXPECT_TRUE(observer_->event.Wait(kDefaultTimeoutMs));
>   EXPECT_EQ(1, observer_->adapt_down_events_);
>@@ -179,24 +210,51 @@ TEST_F(QualityScalerTest, ScalesDownAndBackUp) {
>   EXPECT_EQ(1, observer_->adapt_up_events_);
> }
> 
>-TEST_F(QualityScalerTest, DoesNotScaleUntilEnoughFramesObserved) {
>+TEST_P(QualityScalerTest, DoesNotScaleUntilEnoughFramesObserved) {
>   DO_SYNC(q_, {
>-      // Send 30 frames. This should not be enough to make a decision.
>-      for (int i = 0; i < kFramerate; ++i) {
>-        qs_->ReportQP(kLowQp);
>-      }
>-    });
>+    // Not enough frames to make a decision.
>+    for (int i = 0; i < kMinFramesNeededToScale - 1; ++i) {
>+      qs_->ReportQp(kLowQp);
>+    }
>+  });
>   EXPECT_FALSE(observer_->event.Wait(kDefaultTimeoutMs));
>   DO_SYNC(q_, {
>-      // Send 30 more. This should result in an adapt request as
>-      // enough frames have now been observed.
>-      for (int i = 0; i < kFramerate; ++i) {
>-        qs_->ReportQP(kLowQp);
>-      }
>-    });
>+    // Send 1 more. Enough frames observed, should result in an adapt request.
>+    qs_->ReportQp(kLowQp);
>+  });
>   EXPECT_TRUE(observer_->event.Wait(kDefaultTimeoutMs));
>   EXPECT_EQ(0, observer_->adapt_down_events_);
>   EXPECT_EQ(1, observer_->adapt_up_events_);
>+
>+  // Samples should be cleared after an adapt request.
>+  DO_SYNC(q_, {
>+    // Not enough frames to make a decision.
>+    qs_->ReportQp(kLowQp);
>+  });
>+  EXPECT_FALSE(observer_->event.Wait(kDefaultTimeoutMs));
>+  EXPECT_EQ(0, observer_->adapt_down_events_);
>+  EXPECT_EQ(1, observer_->adapt_up_events_);
>+}
>+
>+TEST_P(QualityScalerTest, ScalesDownAndBackUpWithMinFramesNeeded) {
>+  DO_SYNC(q_, {
>+    for (int i = 0; i < kMinFramesNeededToScale; ++i) {
>+      qs_->ReportQp(kHighQp + 1);
>+    }
>+  });
>+  EXPECT_TRUE(observer_->event.Wait(kDefaultTimeoutMs));
>+  EXPECT_EQ(1, observer_->adapt_down_events_);
>+  EXPECT_EQ(0, observer_->adapt_up_events_);
>+  // Samples cleared.
>+  DO_SYNC(q_, {
>+    for (int i = 0; i < kMinFramesNeededToScale; ++i) {
>+      qs_->ReportQp(kLowQp);
>+    }
>+  });
>+  EXPECT_TRUE(observer_->event.Wait(kDefaultTimeoutMs));
>+  EXPECT_EQ(1, observer_->adapt_down_events_);
>+  EXPECT_EQ(1, observer_->adapt_up_events_);
> }
>+
> }  // namespace webrtc
> #undef DO_SYNC
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/simulcast_rate_allocator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/simulcast_rate_allocator.cc
>new file mode 100644
>index 00000000000..f2cbe0b6049
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/simulcast_rate_allocator.cc
>@@ -0,0 +1,276 @@
>+/*
>+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/video_coding/utility/simulcast_rate_allocator.h"
>+
>+#include <algorithm>
>+#include <memory>
>+#include <utility>
>+#include <vector>
>+
>+#include "common_types.h"  // NOLINT(build/include)
>+#include "rtc_base/checks.h"
>+#include "system_wrappers/include/field_trial.h"
>+
>+namespace webrtc {
>+namespace {
>+// Ratio allocation between temporal streams:
>+// Values as required for the VP8 codec (accumulating).
>+static const float
>+    kLayerRateAllocation[kMaxTemporalStreams][kMaxTemporalStreams] = {
>+        {1.0f, 1.0f, 1.0f, 1.0f},  // 1 layer
>+        {0.6f, 1.0f, 1.0f, 1.0f},  // 2 layers {60%, 40%}
>+        {0.4f, 0.6f, 1.0f, 1.0f},  // 3 layers {40%, 20%, 40%}
>+        {0.25f, 0.4f, 0.6f, 1.0f}  // 4 layers {25%, 15%, 20%, 40%}
>+};
>+
>+static const float kShort3TlRateAllocation[kMaxTemporalStreams] = {
>+    0.6f, 0.8f, 1.0f, 1.0f  // 3 layers {60%, 20%, 20%}
>+};
>+
>+const uint32_t kLegacyScreenshareTl0BitrateKbps = 200;
>+const uint32_t kLegacyScreenshareTl1BitrateKbps = 1000;
>+}  // namespace
>+
>+float SimulcastRateAllocator::GetTemporalRateAllocation(int num_layers,
>+                                                        int temporal_id) {
>+  RTC_CHECK_GT(num_layers, 0);
>+  RTC_CHECK_LE(num_layers, kMaxTemporalStreams);
>+  RTC_CHECK_GE(temporal_id, 0);
>+  RTC_CHECK_LT(temporal_id, num_layers);
>+  if (num_layers == 3 &&
>+      field_trial::IsEnabled("WebRTC-UseShortVP8TL3Pattern")) {
>+    return kShort3TlRateAllocation[temporal_id];
>+  }
>+  return kLayerRateAllocation[num_layers - 1][temporal_id];
>+}
>+
>+SimulcastRateAllocator::SimulcastRateAllocator(const VideoCodec& codec)
>+    : codec_(codec) {}
>+
>+VideoBitrateAllocation SimulcastRateAllocator::GetAllocation(
>+    uint32_t total_bitrate_bps,
>+    uint32_t framerate) {
>+  VideoBitrateAllocation allocated_bitrates_bps;
>+  DistributeAllocationToSimulcastLayers(total_bitrate_bps,
>+                                        &allocated_bitrates_bps);
>+  DistributeAllocationToTemporalLayers(framerate, &allocated_bitrates_bps);
>+  return allocated_bitrates_bps;
>+}
>+
>+void SimulcastRateAllocator::DistributeAllocationToSimulcastLayers(
>+    uint32_t total_bitrate_bps,
>+    VideoBitrateAllocation* allocated_bitrates_bps) const {
>+  uint32_t left_to_allocate = total_bitrate_bps;
>+  if (codec_.maxBitrate && codec_.maxBitrate * 1000 < left_to_allocate)
>+    left_to_allocate = codec_.maxBitrate * 1000;
>+
>+  if (codec_.numberOfSimulcastStreams == 0) {
>+    // No simulcast, just set the target as this has been capped already.
>+    if (codec_.active) {
>+      allocated_bitrates_bps->SetBitrate(
>+          0, 0, std::max(codec_.minBitrate * 1000, left_to_allocate));
>+    }
>+    return;
>+  }
>+  // Find the first active layer. We don't allocate to inactive layers.
>+  size_t active_layer = 0;
>+  for (; active_layer < codec_.numberOfSimulcastStreams; ++active_layer) {
>+    if (codec_.simulcastStream[active_layer].active) {
>+      // Found the first active layer.
>+      break;
>+    }
>+  }
>+  // All streams could be inactive, and nothing more to do.
>+  if (active_layer == codec_.numberOfSimulcastStreams) {
>+    return;
>+  }
>+
>+  // Always allocate enough bitrate for the minimum bitrate of the first
>+  // active layer. Suspending below min bitrate is controlled outside the
>+  // codec implementation and is not overridden by this.
>+  left_to_allocate = std::max(
>+      codec_.simulcastStream[active_layer].minBitrate * 1000, left_to_allocate);
>+
>+  // Begin by allocating bitrate to simulcast streams, putting all bitrate in
>+  // temporal layer 0. We'll then distribute this bitrate, across potential
>+  // temporal layers, when stream allocation is done.
>+
>+  size_t top_active_layer = active_layer;
>+  // Allocate up to the target bitrate for each active simulcast layer.
>+  for (; active_layer < codec_.numberOfSimulcastStreams; ++active_layer) {
>+    const SimulcastStream& stream = codec_.simulcastStream[active_layer];
>+    if (!stream.active) {
>+      continue;
>+    }
>+    // If we can't allocate to the current layer we can't allocate to higher
>+    // layers because they require a higher minimum bitrate.
>+    if (left_to_allocate < stream.minBitrate * 1000) {
>+      break;
>+    }
>+    // We are allocating to this layer so it is the current active allocation.
>+    top_active_layer = active_layer;
>+    uint32_t allocation =
>+        std::min(left_to_allocate, stream.targetBitrate * 1000);
>+    allocated_bitrates_bps->SetBitrate(active_layer, 0, allocation);
>+    RTC_DCHECK_LE(allocation, left_to_allocate);
>+    left_to_allocate -= allocation;
>+  }
>+
>+  // Next, try allocate remaining bitrate, up to max bitrate, in top active
>+  // stream.
>+  // TODO(sprang): Allocate up to max bitrate for all layers once we have a
>+  //               better idea of possible performance implications.
>+  if (left_to_allocate > 0) {
>+    const SimulcastStream& stream = codec_.simulcastStream[top_active_layer];
>+    uint32_t bitrate_bps =
>+        allocated_bitrates_bps->GetSpatialLayerSum(top_active_layer);
>+    uint32_t allocation =
>+        std::min(left_to_allocate, stream.maxBitrate * 1000 - bitrate_bps);
>+    bitrate_bps += allocation;
>+    RTC_DCHECK_LE(allocation, left_to_allocate);
>+    left_to_allocate -= allocation;
>+    allocated_bitrates_bps->SetBitrate(top_active_layer, 0, bitrate_bps);
>+  }
>+}
>+
>+void SimulcastRateAllocator::DistributeAllocationToTemporalLayers(
>+    uint32_t framerate,
>+    VideoBitrateAllocation* allocated_bitrates_bps) const {
>+  const int num_spatial_streams =
>+      std::max(1, static_cast<int>(codec_.numberOfSimulcastStreams));
>+
>+  // Finally, distribute the bitrate for the simulcast streams across the
>+  // available temporal layers.
>+  for (int simulcast_id = 0; simulcast_id < num_spatial_streams;
>+       ++simulcast_id) {
>+    uint32_t target_bitrate_kbps =
>+        allocated_bitrates_bps->GetBitrate(simulcast_id, 0) / 1000;
>+    if (target_bitrate_kbps == 0) {
>+      continue;
>+    }
>+
>+    const uint32_t expected_allocated_bitrate_kbps = target_bitrate_kbps;
>+    RTC_DCHECK_EQ(
>+        target_bitrate_kbps,
>+        allocated_bitrates_bps->GetSpatialLayerSum(simulcast_id) / 1000);
>+    const int num_temporal_streams = NumTemporalStreams(simulcast_id);
>+    uint32_t max_bitrate_kbps;
>+    // Legacy temporal-layered only screenshare, or simulcast screenshare
>+    // with legacy mode for simulcast stream 0.
>+    const bool conference_screenshare_mode =
>+        codec_.mode == VideoCodecMode::kScreensharing &&
>+        ((num_spatial_streams == 1 && num_temporal_streams == 2) ||  // Legacy.
>+         (num_spatial_streams > 1 && simulcast_id == 0));  // Simulcast.
>+    if (conference_screenshare_mode) {
>+      // TODO(holmer): This is a "temporary" hack for screensharing, where we
>+      // interpret the startBitrate as the encoder target bitrate. This is
>+      // to allow for a different max bitrate, so if the codec can't meet
>+      // the target we still allow it to overshoot up to the max before dropping
>+      // frames. This hack should be improved.
>+      max_bitrate_kbps =
>+          std::min(kLegacyScreenshareTl1BitrateKbps, target_bitrate_kbps);
>+      target_bitrate_kbps =
>+          std::min(kLegacyScreenshareTl0BitrateKbps, target_bitrate_kbps);
>+    } else if (num_spatial_streams == 1) {
>+      max_bitrate_kbps = codec_.maxBitrate;
>+    } else {
>+      max_bitrate_kbps = codec_.simulcastStream[simulcast_id].maxBitrate;
>+    }
>+
>+    std::vector<uint32_t> tl_allocation;
>+    if (num_temporal_streams == 1) {
>+      tl_allocation.push_back(target_bitrate_kbps);
>+    } else {
>+      if (conference_screenshare_mode) {
>+        tl_allocation = ScreenshareTemporalLayerAllocation(
>+            target_bitrate_kbps, max_bitrate_kbps, framerate, simulcast_id);
>+      } else {
>+        tl_allocation = DefaultTemporalLayerAllocation(
>+            target_bitrate_kbps, max_bitrate_kbps, framerate, simulcast_id);
>+      }
>+    }
>+    RTC_DCHECK_GT(tl_allocation.size(), 0);
>+    RTC_DCHECK_LE(tl_allocation.size(), num_temporal_streams);
>+
>+    uint64_t tl_allocation_sum_kbps = 0;
>+    for (size_t tl_index = 0; tl_index < tl_allocation.size(); ++tl_index) {
>+      uint32_t layer_rate_kbps = tl_allocation[tl_index];
>+      if (layer_rate_kbps > 0) {
>+        allocated_bitrates_bps->SetBitrate(simulcast_id, tl_index,
>+                                           layer_rate_kbps * 1000);
>+      }
>+      tl_allocation_sum_kbps += layer_rate_kbps;
>+    }
>+    RTC_DCHECK_LE(tl_allocation_sum_kbps, expected_allocated_bitrate_kbps);
>+  }
>+}
>+
>+std::vector<uint32_t> SimulcastRateAllocator::DefaultTemporalLayerAllocation(
>+    int bitrate_kbps,
>+    int max_bitrate_kbps,
>+    int framerate,
>+    int simulcast_id) const {
>+  const size_t num_temporal_layers = NumTemporalStreams(simulcast_id);
>+  std::vector<uint32_t> bitrates;
>+  for (size_t i = 0; i < num_temporal_layers; ++i) {
>+    float layer_bitrate =
>+        bitrate_kbps * GetTemporalRateAllocation(num_temporal_layers, i);
>+    bitrates.push_back(static_cast<uint32_t>(layer_bitrate + 0.5));
>+  }
>+
>+  // Allocation table is of aggregates, transform to individual rates.
>+  uint32_t sum = 0;
>+  for (size_t i = 0; i < num_temporal_layers; ++i) {
>+    uint32_t layer_bitrate = bitrates[i];
>+    RTC_DCHECK_LE(sum, bitrates[i]);
>+    bitrates[i] -= sum;
>+    sum = layer_bitrate;
>+
>+    if (sum >= static_cast<uint32_t>(bitrate_kbps)) {
>+      // Sum adds up; any subsequent layers will be 0.
>+      bitrates.resize(i + 1);
>+      break;
>+    }
>+  }
>+
>+  return bitrates;
>+}
>+
>+std::vector<uint32_t>
>+SimulcastRateAllocator::ScreenshareTemporalLayerAllocation(
>+    int bitrate_kbps,
>+    int max_bitrate_kbps,
>+    int framerate,
>+    int simulcast_id) const {
>+  if (simulcast_id > 0) {
>+    return DefaultTemporalLayerAllocation(bitrate_kbps, max_bitrate_kbps,
>+                                          framerate, simulcast_id);
>+  }
>+  std::vector<uint32_t> allocation;
>+  allocation.push_back(bitrate_kbps);
>+  if (max_bitrate_kbps > bitrate_kbps)
>+    allocation.push_back(max_bitrate_kbps - bitrate_kbps);
>+  return allocation;
>+}
>+
>+const VideoCodec& webrtc::SimulcastRateAllocator::GetCodec() const {
>+  return codec_;
>+}
>+
>+int SimulcastRateAllocator::NumTemporalStreams(size_t simulcast_id) const {
>+  return std::max<uint8_t>(
>+      1,
>+      codec_.codecType == kVideoCodecVP8 && codec_.numberOfSimulcastStreams == 0
>+          ? codec_.VP8().numberOfTemporalLayers
>+          : codec_.simulcastStream[simulcast_id].numberOfTemporalLayers);
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/simulcast_rate_allocator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/simulcast_rate_allocator.h
>new file mode 100644
>index 00000000000..594e88b5096
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/simulcast_rate_allocator.h
>@@ -0,0 +1,62 @@
>+/*
>+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_VIDEO_CODING_UTILITY_SIMULCAST_RATE_ALLOCATOR_H_
>+#define MODULES_VIDEO_CODING_UTILITY_SIMULCAST_RATE_ALLOCATOR_H_
>+
>+#include <stdint.h>
>+
>+#include <map>
>+#include <memory>
>+#include <vector>
>+
>+#include "api/video/video_bitrate_allocator.h"
>+#include "api/video_codecs/video_encoder.h"
>+#include "common_types.h"  // NOLINT(build/include)
>+#include "rtc_base/constructormagic.h"
>+
>+namespace webrtc {
>+
>+class SimulcastRateAllocator : public VideoBitrateAllocator {
>+ public:
>+  explicit SimulcastRateAllocator(const VideoCodec& codec);
>+
>+  VideoBitrateAllocation GetAllocation(uint32_t total_bitrate_bps,
>+                                       uint32_t framerate) override;
>+  const VideoCodec& GetCodec() const;
>+
>+  static float GetTemporalRateAllocation(int num_layers, int temporal_id);
>+
>+ private:
>+  void DistributeAllocationToSimulcastLayers(
>+      uint32_t total_bitrate_bps,
>+      VideoBitrateAllocation* allocated_bitrates_bps) const;
>+  void DistributeAllocationToTemporalLayers(
>+      uint32_t framerate,
>+      VideoBitrateAllocation* allocated_bitrates_bps) const;
>+  std::vector<uint32_t> DefaultTemporalLayerAllocation(int bitrate_kbps,
>+                                                       int max_bitrate_kbps,
>+                                                       int framerate,
>+                                                       int simulcast_id) const;
>+  std::vector<uint32_t> ScreenshareTemporalLayerAllocation(
>+      int bitrate_kbps,
>+      int max_bitrate_kbps,
>+      int framerate,
>+      int simulcast_id) const;
>+  int NumTemporalStreams(size_t simulcast_id) const;
>+
>+  const VideoCodec codec_;
>+
>+  RTC_DISALLOW_COPY_AND_ASSIGN(SimulcastRateAllocator);
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_VIDEO_CODING_UTILITY_SIMULCAST_RATE_ALLOCATOR_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/simulcast_rate_allocator_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/simulcast_rate_allocator_unittest.cc
>index 15ac938ae53..b1e6927e251 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/simulcast_rate_allocator_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/simulcast_rate_allocator_unittest.cc
>@@ -8,13 +8,15 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "modules/video_coding/codecs/vp8/simulcast_rate_allocator.h"
>+#include "modules/video_coding/utility/simulcast_rate_allocator.h"
> 
> #include <limits>
> #include <memory>
> #include <utility>
> #include <vector>
> 
>+#include "modules/video_coding/codecs/vp8/temporal_layers.h"
>+
> #include "test/gmock.h"
> #include "test/gtest.h"
> 
>@@ -22,22 +24,23 @@ namespace webrtc {
> namespace {
> using ::testing::_;
> 
>+constexpr uint32_t kFramerateFps = 5;
> constexpr uint32_t kMinBitrateKbps = 50;
>-constexpr uint32_t kTargetBitrateKbps = 100;
>+// These correspond to kLegacyScreenshareTl(0|1)BitrateKbps in cc.
>+constexpr uint32_t kTargetBitrateKbps = 200;
> constexpr uint32_t kMaxBitrateKbps = 1000;
>-constexpr uint32_t kFramerateFps = 5;
> 
> class MockTemporalLayers : public TemporalLayers {
>  public:
>   MOCK_METHOD1(UpdateLayerConfig, TemporalLayers::FrameConfig(uint32_t));
>-  MOCK_METHOD3(OnRatesUpdated, std::vector<uint32_t>(int, int, int));
>-  MOCK_METHOD1(UpdateConfiguration, bool(vpx_codec_enc_cfg_t*));
>+  MOCK_METHOD2(OnRatesUpdated, void(const std::vector<uint32_t>&, int));
>+  MOCK_METHOD1(UpdateConfiguration, bool(Vp8EncoderConfig*));
>   MOCK_METHOD4(PopulateCodecSpecific,
>                void(bool,
>                     const TemporalLayers::FrameConfig&,
>                     CodecSpecificInfoVP8*,
>                     uint32_t));
>-  MOCK_METHOD2(FrameEncoded, void(unsigned int, int));
>+  MOCK_METHOD3(FrameEncoded, void(uint32_t, size_t, int));
>   MOCK_CONST_METHOD0(Tl0PicIdx, uint8_t());
>   MOCK_CONST_METHOD1(GetTemporalLayerId,
>                      int(const TemporalLayers::FrameConfig&));
>@@ -48,9 +51,10 @@ class SimulcastRateAllocatorTest : public ::testing::TestWithParam<bool> {
>  public:
>   SimulcastRateAllocatorTest() {
>     memset(&codec_, 0, sizeof(VideoCodec));
>+    codec_.codecType = kVideoCodecVP8;
>     codec_.minBitrate = kMinBitrateKbps;
>-    codec_.targetBitrate = kTargetBitrateKbps;
>     codec_.maxBitrate = kMaxBitrateKbps;
>+    codec_.active = true;
>     CreateAllocator();
>   }
>   virtual ~SimulcastRateAllocatorTest() {}
>@@ -64,43 +68,55 @@ class SimulcastRateAllocatorTest : public ::testing::TestWithParam<bool> {
>   }
> 
>   template <size_t S>
>-  void ExpectEqual(uint32_t (&expected)[S], const BitrateAllocation& actual) {
>+  void ExpectEqual(uint32_t (&expected)[S],
>+                   const VideoBitrateAllocation& actual) {
>     // EXPECT_EQ(S, actual.size());
>     uint32_t sum = 0;
>     for (size_t i = 0; i < S; ++i) {
>       uint32_t layer_bitrate = actual.GetSpatialLayerSum(i);
>-      EXPECT_EQ(expected[i] * 1000U, layer_bitrate) << "Mismatch at index "
>-                                                    << i;
>+      if (layer_bitrate == 0) {
>+        EXPECT_FALSE(actual.IsSpatialLayerUsed(i));
>+      }
>+      EXPECT_EQ(expected[i] * 1000U, layer_bitrate)
>+          << "Mismatch at index " << i;
>       sum += layer_bitrate;
>     }
>     EXPECT_EQ(sum, actual.get_sum_bps());
>   }
> 
>   void CreateAllocator() {
>-    std::unique_ptr<TemporalLayersFactory> tl_factory(GetTlFactory());
>-    codec_.VP8()->tl_factory = tl_factory.get();
>-    allocator_.reset(new SimulcastRateAllocator(codec_, std::move(tl_factory)));
>-
>-    // Simulate InitEncode().
>-    tl_factories_.clear();
>-    if (codec_.numberOfSimulcastStreams == 0) {
>-      tl_factories_.push_back(
>-          std::unique_ptr<TemporalLayers>(codec_.VP8()->tl_factory->Create(
>-              0, codec_.VP8()->numberOfTemporalLayers, 0)));
>-    } else {
>-      for (uint32_t i = 0; i < codec_.numberOfSimulcastStreams; ++i) {
>-        tl_factories_.push_back(
>-            std::unique_ptr<TemporalLayers>(codec_.VP8()->tl_factory->Create(
>-                i, codec_.simulcastStream[i].numberOfTemporalLayers, 0)));
>-      }
>-    }
>+    allocator_.reset(new SimulcastRateAllocator(codec_));
>   }
> 
>-  virtual std::unique_ptr<TemporalLayersFactory> GetTlFactory() {
>-    return std::unique_ptr<TemporalLayersFactory>(new TemporalLayersFactory());
>+  void SetupCodecThreeSimulcastStreams(
>+      const std::vector<bool>& active_streams) {
>+    size_t num_streams = 3;
>+    RTC_DCHECK_GE(active_streams.size(), num_streams);
>+    SetupCodecTwoSimulcastStreams(active_streams);
>+    codec_.numberOfSimulcastStreams = num_streams;
>+    codec_.simulcastStream[2].minBitrate = 2000;
>+    codec_.simulcastStream[2].targetBitrate = 3000;
>+    codec_.simulcastStream[2].maxBitrate = 4000;
>+    codec_.simulcastStream[2].active = active_streams[2];
>+  }
>+
>+  void SetupCodecTwoSimulcastStreams(const std::vector<bool>& active_streams) {
>+    size_t num_streams = 2;
>+    RTC_DCHECK_GE(active_streams.size(), num_streams);
>+    codec_.numberOfSimulcastStreams = num_streams;
>+    codec_.maxBitrate = 0;
>+    codec_.simulcastStream[0].minBitrate = 10;
>+    codec_.simulcastStream[0].targetBitrate = 100;
>+    codec_.simulcastStream[0].maxBitrate = 500;
>+    codec_.simulcastStream[1].minBitrate = 50;
>+    codec_.simulcastStream[1].targetBitrate = 500;
>+    codec_.simulcastStream[1].maxBitrate = 1000;
>+    for (size_t i = 0; i < num_streams; ++i) {
>+      codec_.simulcastStream[i].active = active_streams[i];
>+    }
>   }
> 
>-  BitrateAllocation GetAllocation(uint32_t target_bitrate) {
>+  VideoBitrateAllocation GetAllocation(uint32_t target_bitrate) {
>     return allocator_->GetAllocation(target_bitrate * 1000U, kDefaultFrameRate);
>   }
> 
>@@ -108,11 +124,11 @@ class SimulcastRateAllocatorTest : public ::testing::TestWithParam<bool> {
>   static const int kDefaultFrameRate = 30;
>   VideoCodec codec_;
>   std::unique_ptr<SimulcastRateAllocator> allocator_;
>-  std::vector<std::unique_ptr<TemporalLayers>> tl_factories_;
> };
> 
> TEST_F(SimulcastRateAllocatorTest, NoSimulcastBelowMin) {
>   uint32_t expected[] = {codec_.minBitrate};
>+  codec_.active = true;
>   ExpectEqual(expected, GetAllocation(codec_.minBitrate - 1));
>   ExpectEqual(expected, GetAllocation(1));
>   ExpectEqual(expected, GetAllocation(0));
>@@ -120,12 +136,14 @@ TEST_F(SimulcastRateAllocatorTest, NoSimulcastBelowMin) {
> 
> TEST_F(SimulcastRateAllocatorTest, NoSimulcastAboveMax) {
>   uint32_t expected[] = {codec_.maxBitrate};
>+  codec_.active = true;
>   ExpectEqual(expected, GetAllocation(codec_.maxBitrate + 1));
>   ExpectEqual(expected, GetAllocation(std::numeric_limits<uint32_t>::max()));
> }
> 
> TEST_F(SimulcastRateAllocatorTest, NoSimulcastNoMax) {
>-  const uint32_t kMax = BitrateAllocation::kMaxBitrateBps / 1000;
>+  const uint32_t kMax = VideoBitrateAllocation::kMaxBitrateBps / 1000;
>+  codec_.active = true;
>   codec_.maxBitrate = 0;
>   CreateAllocator();
> 
>@@ -134,6 +152,7 @@ TEST_F(SimulcastRateAllocatorTest, NoSimulcastNoMax) {
> }
> 
> TEST_F(SimulcastRateAllocatorTest, NoSimulcastWithinLimits) {
>+  codec_.active = true;
>   for (uint32_t bitrate = codec_.minBitrate; bitrate <= codec_.maxBitrate;
>        ++bitrate) {
>     uint32_t expected[] = {bitrate};
>@@ -141,12 +160,25 @@ TEST_F(SimulcastRateAllocatorTest, NoSimulcastWithinLimits) {
>   }
> }
> 
>+// Tests that when we aren't using simulcast and the codec is marked inactive no
>+// bitrate will be allocated.
>+TEST_F(SimulcastRateAllocatorTest, NoSimulcastInactive) {
>+  codec_.active = false;
>+  uint32_t expected[] = {0};
>+  CreateAllocator();
>+
>+  ExpectEqual(expected, GetAllocation(kMinBitrateKbps - 10));
>+  ExpectEqual(expected, GetAllocation(kTargetBitrateKbps));
>+  ExpectEqual(expected, GetAllocation(kMaxBitrateKbps + 10));
>+}
>+
> TEST_F(SimulcastRateAllocatorTest, SingleSimulcastBelowMin) {
>   // With simulcast, use the min bitrate from the ss spec instead of the global.
>   codec_.numberOfSimulcastStreams = 1;
>   const uint32_t kMin = codec_.minBitrate - 10;
>   codec_.simulcastStream[0].minBitrate = kMin;
>   codec_.simulcastStream[0].targetBitrate = kTargetBitrateKbps;
>+  codec_.simulcastStream[0].active = true;
>   CreateAllocator();
> 
>   uint32_t expected[] = {kMin};
>@@ -160,6 +192,7 @@ TEST_F(SimulcastRateAllocatorTest, SingleSimulcastAboveMax) {
>   codec_.simulcastStream[0].minBitrate = kMinBitrateKbps;
>   const uint32_t kMax = codec_.simulcastStream[0].maxBitrate + 1000;
>   codec_.simulcastStream[0].maxBitrate = kMax;
>+  codec_.simulcastStream[0].active = true;
>   CreateAllocator();
> 
>   uint32_t expected[] = {kMax};
>@@ -173,6 +206,7 @@ TEST_F(SimulcastRateAllocatorTest, SingleSimulcastWithinLimits) {
>   codec_.simulcastStream[0].minBitrate = kMinBitrateKbps;
>   codec_.simulcastStream[0].targetBitrate = kTargetBitrateKbps;
>   codec_.simulcastStream[0].maxBitrate = kMaxBitrateKbps;
>+  codec_.simulcastStream[0].active = true;
>   CreateAllocator();
> 
>   for (uint32_t bitrate = kMinBitrateKbps; bitrate <= kMaxBitrateKbps;
>@@ -182,18 +216,23 @@ TEST_F(SimulcastRateAllocatorTest, SingleSimulcastWithinLimits) {
>   }
> }
> 
>+TEST_F(SimulcastRateAllocatorTest, SingleSimulcastInactive) {
>+  codec_.numberOfSimulcastStreams = 1;
>+  codec_.simulcastStream[0].minBitrate = kMinBitrateKbps;
>+  codec_.simulcastStream[0].targetBitrate = kTargetBitrateKbps;
>+  codec_.simulcastStream[0].maxBitrate = kMaxBitrateKbps;
>+  codec_.simulcastStream[0].active = false;
>+  CreateAllocator();
>+
>+  uint32_t expected[] = {0};
>+  ExpectEqual(expected, GetAllocation(kMinBitrateKbps - 10));
>+  ExpectEqual(expected, GetAllocation(kTargetBitrateKbps));
>+  ExpectEqual(expected, GetAllocation(kMaxBitrateKbps + 10));
>+}
>+
> TEST_F(SimulcastRateAllocatorTest, OneToThreeStreams) {
>-  codec_.numberOfSimulcastStreams = 3;
>-  codec_.maxBitrate = 0;
>-  codec_.simulcastStream[0].minBitrate = 10;
>-  codec_.simulcastStream[0].targetBitrate = 100;
>-  codec_.simulcastStream[0].maxBitrate = 500;
>-  codec_.simulcastStream[1].minBitrate = 50;
>-  codec_.simulcastStream[1].targetBitrate = 500;
>-  codec_.simulcastStream[1].maxBitrate = 1000;
>-  codec_.simulcastStream[2].minBitrate = 2000;
>-  codec_.simulcastStream[2].targetBitrate = 3000;
>-  codec_.simulcastStream[2].maxBitrate = 4000;
>+  const std::vector<bool> active_streams(3, true);
>+  SetupCodecThreeSimulcastStreams(active_streams);
>   CreateAllocator();
> 
>   {
>@@ -267,45 +306,169 @@ TEST_F(SimulcastRateAllocatorTest, OneToThreeStreams) {
>                            codec_.simulcastStream[2].maxBitrate};
>     ExpectEqual(expected, GetAllocation(bitrate));
>   }
>+
>+  {
>+    // Enough to max out all streams which will allocate the target amount to
>+    // the lower streams.
>+    const uint32_t bitrate = codec_.simulcastStream[0].maxBitrate +
>+                             codec_.simulcastStream[1].maxBitrate +
>+                             codec_.simulcastStream[2].maxBitrate;
>+    uint32_t expected[] = {codec_.simulcastStream[0].targetBitrate,
>+                           codec_.simulcastStream[1].targetBitrate,
>+                           codec_.simulcastStream[2].maxBitrate};
>+    ExpectEqual(expected, GetAllocation(bitrate));
>+  }
> }
> 
>-TEST_F(SimulcastRateAllocatorTest, GetPreferredBitrateBps) {
>-  MockTemporalLayers mock_layers;
>-  allocator_.reset(new SimulcastRateAllocator(codec_, nullptr));
>-  allocator_->OnTemporalLayersCreated(0, &mock_layers);
>-  EXPECT_CALL(mock_layers, OnRatesUpdated(_, _, _)).Times(0);
>-  EXPECT_EQ(codec_.maxBitrate * 1000,
>-            allocator_->GetPreferredBitrateBps(codec_.maxFramerate));
>+// If three simulcast streams that are all inactive, none of them should be
>+// allocated bitrate.
>+TEST_F(SimulcastRateAllocatorTest, ThreeStreamsInactive) {
>+  const std::vector<bool> active_streams(3, false);
>+  SetupCodecThreeSimulcastStreams(active_streams);
>+  CreateAllocator();
>+
>+  // Just enough to allocate the min.
>+  const uint32_t min_bitrate = codec_.simulcastStream[0].minBitrate +
>+                               codec_.simulcastStream[1].minBitrate +
>+                               codec_.simulcastStream[2].minBitrate;
>+  // Enough bitrate to allocate target to all streams.
>+  const uint32_t target_bitrate = codec_.simulcastStream[0].targetBitrate +
>+                                  codec_.simulcastStream[1].targetBitrate +
>+                                  codec_.simulcastStream[2].targetBitrate;
>+  // Enough bitrate to allocate max to all streams.
>+  const uint32_t max_bitrate = codec_.simulcastStream[0].maxBitrate +
>+                               codec_.simulcastStream[1].maxBitrate +
>+                               codec_.simulcastStream[2].maxBitrate;
>+  uint32_t expected[] = {0, 0, 0};
>+  ExpectEqual(expected, GetAllocation(0));
>+  ExpectEqual(expected, GetAllocation(min_bitrate));
>+  ExpectEqual(expected, GetAllocation(target_bitrate));
>+  ExpectEqual(expected, GetAllocation(max_bitrate));
> }
> 
>-TEST_F(SimulcastRateAllocatorTest, GetPreferredBitrateSimulcast) {
>-  codec_.numberOfSimulcastStreams = 3;
>-  codec_.maxBitrate = 999999;
>-  codec_.simulcastStream[0].minBitrate = 10;
>-  codec_.simulcastStream[0].targetBitrate = 100;
>+// If there are two simulcast streams, we expect the high active stream to be
>+// allocated as if it is a single active stream.
>+TEST_F(SimulcastRateAllocatorTest, TwoStreamsLowInactive) {
>+  const std::vector<bool> active_streams({false, true});
>+  SetupCodecTwoSimulcastStreams(active_streams);
>+  CreateAllocator();
>+
>+  const uint32_t kActiveStreamMinBitrate = codec_.simulcastStream[1].minBitrate;
>+  const uint32_t kActiveStreamTargetBitrate =
>+      codec_.simulcastStream[1].targetBitrate;
>+  const uint32_t kActiveStreamMaxBitrate = codec_.simulcastStream[1].maxBitrate;
>+  {
>+    // Expect that the stream is always allocated its min bitrate.
>+    uint32_t expected[] = {0, kActiveStreamMinBitrate};
>+    ExpectEqual(expected, GetAllocation(0));
>+    ExpectEqual(expected, GetAllocation(kActiveStreamMinBitrate - 10));
>+    ExpectEqual(expected, GetAllocation(kActiveStreamMinBitrate));
>+  }
> 
>-  codec_.simulcastStream[0].maxBitrate = 500;
>-  codec_.simulcastStream[1].minBitrate = 50;
>-  codec_.simulcastStream[1].targetBitrate = 500;
>-  codec_.simulcastStream[1].maxBitrate = 1000;
>+  {
>+    // The stream should be allocated its target bitrate.
>+    uint32_t expected[] = {0, kActiveStreamTargetBitrate};
>+    ExpectEqual(expected, GetAllocation(kActiveStreamTargetBitrate));
>+  }
> 
>-  codec_.simulcastStream[2].minBitrate = 2000;
>-  codec_.simulcastStream[2].targetBitrate = 3000;
>-  codec_.simulcastStream[2].maxBitrate = 4000;
>+  {
>+    // The stream should be allocated its max if the target input is sufficient.
>+    uint32_t expected[] = {0, kActiveStreamMaxBitrate};
>+    ExpectEqual(expected, GetAllocation(kActiveStreamMaxBitrate));
>+    ExpectEqual(expected, GetAllocation(std::numeric_limits<uint32_t>::max()));
>+  }
>+}
>+
>+// If there are two simulcast streams, we expect the low active stream to be
>+// allocated as if it is a single active stream.
>+TEST_F(SimulcastRateAllocatorTest, TwoStreamsHighInactive) {
>+  const std::vector<bool> active_streams({true, false});
>+  SetupCodecTwoSimulcastStreams(active_streams);
>   CreateAllocator();
> 
>-  uint32_t preferred_bitrate_kbps;
>-  preferred_bitrate_kbps = codec_.simulcastStream[0].targetBitrate;
>-  preferred_bitrate_kbps += codec_.simulcastStream[1].targetBitrate;
>-  preferred_bitrate_kbps += codec_.simulcastStream[2].maxBitrate;
>+  const uint32_t kActiveStreamMinBitrate = codec_.simulcastStream[0].minBitrate;
>+  const uint32_t kActiveStreamTargetBitrate =
>+      codec_.simulcastStream[0].targetBitrate;
>+  const uint32_t kActiveStreamMaxBitrate = codec_.simulcastStream[0].maxBitrate;
>+  {
>+    // Expect that the stream is always allocated its min bitrate.
>+    uint32_t expected[] = {kActiveStreamMinBitrate, 0};
>+    ExpectEqual(expected, GetAllocation(0));
>+    ExpectEqual(expected, GetAllocation(kActiveStreamMinBitrate - 10));
>+    ExpectEqual(expected, GetAllocation(kActiveStreamMinBitrate));
>+  }
>+
>+  {
>+    // The stream should be allocated its target bitrate.
>+    uint32_t expected[] = {kActiveStreamTargetBitrate, 0};
>+    ExpectEqual(expected, GetAllocation(kActiveStreamTargetBitrate));
>+  }
>+
>+  {
>+    // The stream should be allocated its max if the target input is sufficent.
>+    uint32_t expected[] = {kActiveStreamMaxBitrate, 0};
>+    ExpectEqual(expected, GetAllocation(kActiveStreamMaxBitrate));
>+    ExpectEqual(expected, GetAllocation(std::numeric_limits<uint32_t>::max()));
>+  }
>+}
>+
>+// If there are three simulcast streams and the middle stream is inactive, the
>+// other two streams should be allocated bitrate the same as if they are two
>+// active simulcast streams.
>+TEST_F(SimulcastRateAllocatorTest, ThreeStreamsMiddleInactive) {
>+  const std::vector<bool> active_streams({true, false, true});
>+  SetupCodecThreeSimulcastStreams(active_streams);
>+  CreateAllocator();
>+
>+  {
>+    const uint32_t kLowStreamMinBitrate = codec_.simulcastStream[0].minBitrate;
>+    // The lowest stream should always be allocated its minimum bitrate.
>+    uint32_t expected[] = {kLowStreamMinBitrate, 0, 0};
>+    ExpectEqual(expected, GetAllocation(0));
>+    ExpectEqual(expected, GetAllocation(kLowStreamMinBitrate - 10));
>+    ExpectEqual(expected, GetAllocation(kLowStreamMinBitrate));
>+  }
>+
>+  {
>+    // The lowest stream gets its target bitrate.
>+    uint32_t expected[] = {codec_.simulcastStream[0].targetBitrate, 0, 0};
>+    ExpectEqual(expected,
>+                GetAllocation(codec_.simulcastStream[0].targetBitrate));
>+  }
>+
>+  {
>+    // The lowest stream gets its max bitrate, but not enough for the high
>+    // stream.
>+    const uint32_t bitrate = codec_.simulcastStream[0].targetBitrate +
>+                             codec_.simulcastStream[2].minBitrate - 1;
>+    uint32_t expected[] = {codec_.simulcastStream[0].maxBitrate, 0, 0};
>+    ExpectEqual(expected, GetAllocation(bitrate));
>+  }
> 
>-  EXPECT_EQ(preferred_bitrate_kbps * 1000,
>-            allocator_->GetPreferredBitrateBps(codec_.maxFramerate));
>+  {
>+    // Both active streams get allocated target bitrate.
>+    const uint32_t bitrate = codec_.simulcastStream[0].targetBitrate +
>+                             codec_.simulcastStream[2].targetBitrate;
>+    uint32_t expected[] = {codec_.simulcastStream[0].targetBitrate, 0,
>+                           codec_.simulcastStream[2].targetBitrate};
>+    ExpectEqual(expected, GetAllocation(bitrate));
>+  }
>+
>+  {
>+    // Lowest stream gets its target bitrate, high stream gets its max bitrate.
>+    uint32_t bitrate = codec_.simulcastStream[0].targetBitrate +
>+                       codec_.simulcastStream[2].maxBitrate;
>+    uint32_t expected[] = {codec_.simulcastStream[0].targetBitrate, 0,
>+                           codec_.simulcastStream[2].maxBitrate};
>+    ExpectEqual(expected, GetAllocation(bitrate));
>+    ExpectEqual(expected, GetAllocation(bitrate + 10));
>+    ExpectEqual(expected, GetAllocation(std::numeric_limits<uint32_t>::max()));
>+  }
> }
> 
> class ScreenshareRateAllocationTest : public SimulcastRateAllocatorTest {
>  public:
>-  void SetupConferenceScreenshare(bool use_simulcast) {
>+  void SetupConferenceScreenshare(bool use_simulcast, bool active = true) {
>     codec_.mode = VideoCodecMode::kScreensharing;
>     codec_.minBitrate = kMinBitrateKbps;
>     codec_.maxBitrate = kMaxBitrateKbps;
>@@ -315,17 +478,13 @@ class ScreenshareRateAllocationTest : public SimulcastRateAllocatorTest {
>       codec_.simulcastStream[0].targetBitrate = kTargetBitrateKbps;
>       codec_.simulcastStream[0].maxBitrate = kMaxBitrateKbps;
>       codec_.simulcastStream[0].numberOfTemporalLayers = 2;
>+      codec_.simulcastStream[0].active = active;
>     } else {
>       codec_.numberOfSimulcastStreams = 0;
>-      codec_.targetBitrate = kTargetBitrateKbps;
>       codec_.VP8()->numberOfTemporalLayers = 2;
>+      codec_.active = active;
>     }
>   }
>-
>-  std::unique_ptr<TemporalLayersFactory> GetTlFactory() override {
>-    return std::unique_ptr<TemporalLayersFactory>(
>-        new ScreenshareTemporalLayersFactory());
>-  }
> };
> 
> INSTANTIATE_TEST_CASE_P(ScreenshareTest,
>@@ -336,7 +495,7 @@ TEST_P(ScreenshareRateAllocationTest, BitrateBelowTl0) {
>   SetupConferenceScreenshare(GetParam());
>   CreateAllocator();
> 
>-  BitrateAllocation allocation =
>+  VideoBitrateAllocation allocation =
>       allocator_->GetAllocation(kTargetBitrateKbps * 1000, kFramerateFps);
> 
>   // All allocation should go in TL0.
>@@ -349,7 +508,7 @@ TEST_P(ScreenshareRateAllocationTest, BitrateAboveTl0) {
>   CreateAllocator();
> 
>   uint32_t target_bitrate_kbps = (kTargetBitrateKbps + kMaxBitrateKbps) / 2;
>-  BitrateAllocation allocation =
>+  VideoBitrateAllocation allocation =
>       allocator_->GetAllocation(target_bitrate_kbps * 1000, kFramerateFps);
> 
>   // Fill TL0, then put the rest in TL1.
>@@ -363,7 +522,7 @@ TEST_P(ScreenshareRateAllocationTest, BitrateAboveTl1) {
>   SetupConferenceScreenshare(GetParam());
>   CreateAllocator();
> 
>-  BitrateAllocation allocation =
>+  VideoBitrateAllocation allocation =
>       allocator_->GetAllocation(kMaxBitrateKbps * 2000, kFramerateFps);
> 
>   // Fill both TL0 and TL1, but no more.
>@@ -373,4 +532,17 @@ TEST_P(ScreenshareRateAllocationTest, BitrateAboveTl1) {
>             allocation.GetBitrate(0, 1) / 1000);
> }
> 
>+// This tests when the screenshare is inactive it should be allocated 0 bitrate
>+// for all layers.
>+TEST_P(ScreenshareRateAllocationTest, InactiveScreenshare) {
>+  SetupConferenceScreenshare(GetParam(), false);
>+  CreateAllocator();
>+
>+  // Enough bitrate for TL0 and TL1.
>+  uint32_t target_bitrate_kbps = (kTargetBitrateKbps + kMaxBitrateKbps) / 2;
>+  VideoBitrateAllocation allocation =
>+      allocator_->GetAllocation(target_bitrate_kbps * 1000, kFramerateFps);
>+
>+  EXPECT_EQ(0U, allocation.get_sum_kbps());
>+}
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/simulcast_test_fixture_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/simulcast_test_fixture_impl.cc
>new file mode 100644
>index 00000000000..e81c2f4679e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/simulcast_test_fixture_impl.cc
>@@ -0,0 +1,829 @@
>+/*
>+ *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/video_coding/utility/simulcast_test_fixture_impl.h"
>+
>+#include <algorithm>
>+#include <map>
>+#include <memory>
>+#include <vector>
>+
>+#include "api/video_codecs/sdp_video_format.h"
>+#include "common_video/include/video_frame.h"
>+#include "common_video/libyuv/include/webrtc_libyuv.h"
>+#include "modules/video_coding/include/video_coding_defines.h"
>+#include "rtc_base/checks.h"
>+#include "test/gtest.h"
>+
>+using ::testing::_;
>+using ::testing::AllOf;
>+using ::testing::Field;
>+using ::testing::Return;
>+
>+namespace webrtc {
>+namespace test {
>+
>+namespace {
>+
>+const int kDefaultWidth = 1280;
>+const int kDefaultHeight = 720;
>+const int kNumberOfSimulcastStreams = 3;
>+const int kColorY = 66;
>+const int kColorU = 22;
>+const int kColorV = 33;
>+const int kMaxBitrates[kNumberOfSimulcastStreams] = {150, 600, 1200};
>+const int kMinBitrates[kNumberOfSimulcastStreams] = {50, 150, 600};
>+const int kTargetBitrates[kNumberOfSimulcastStreams] = {100, 450, 1000};
>+const int kDefaultTemporalLayerProfile[3] = {3, 3, 3};
>+const int kNoTemporalLayerProfile[3] = {0, 0, 0};
>+
>+template <typename T>
>+void SetExpectedValues3(T value0, T value1, T value2, T* expected_values) {
>+  expected_values[0] = value0;
>+  expected_values[1] = value1;
>+  expected_values[2] = value2;
>+}
>+
>+enum PlaneType {
>+  kYPlane = 0,
>+  kUPlane = 1,
>+  kVPlane = 2,
>+  kNumOfPlanes = 3,
>+};
>+
>+}  // namespace
>+
>+class SimulcastTestFixtureImpl::TestEncodedImageCallback
>+    : public EncodedImageCallback {
>+ public:
>+  TestEncodedImageCallback() {
>+    memset(temporal_layer_, -1, sizeof(temporal_layer_));
>+    memset(layer_sync_, false, sizeof(layer_sync_));
>+  }
>+
>+  ~TestEncodedImageCallback() {
>+    delete[] encoded_key_frame_._buffer;
>+    delete[] encoded_frame_._buffer;
>+  }
>+
>+  virtual Result OnEncodedImage(const EncodedImage& encoded_image,
>+                                const CodecSpecificInfo* codec_specific_info,
>+                                const RTPFragmentationHeader* fragmentation) {
>+    uint16_t simulcast_idx = 0;
>+    bool is_vp8 = (codec_specific_info->codecType == kVideoCodecVP8);
>+    if (is_vp8) {
>+      simulcast_idx = codec_specific_info->codecSpecific.VP8.simulcastIdx;
>+    } else {
>+      simulcast_idx = codec_specific_info->codecSpecific.H264.simulcast_idx;
>+    }
>+    // Only store the base layer.
>+    if (simulcast_idx) {
>+      if (encoded_image._frameType == kVideoFrameKey) {
>+        delete[] encoded_key_frame_._buffer;
>+        encoded_key_frame_._buffer = new uint8_t[encoded_image._size];
>+        encoded_key_frame_._size = encoded_image._size;
>+        encoded_key_frame_._length = encoded_image._length;
>+        encoded_key_frame_._frameType = kVideoFrameKey;
>+        encoded_key_frame_._completeFrame = encoded_image._completeFrame;
>+        memcpy(encoded_key_frame_._buffer, encoded_image._buffer,
>+               encoded_image._length);
>+      } else {
>+        delete[] encoded_frame_._buffer;
>+        encoded_frame_._buffer = new uint8_t[encoded_image._size];
>+        encoded_frame_._size = encoded_image._size;
>+        encoded_frame_._length = encoded_image._length;
>+        memcpy(encoded_frame_._buffer, encoded_image._buffer,
>+               encoded_image._length);
>+      }
>+    }
>+    if (is_vp8) {
>+      layer_sync_[codec_specific_info->codecSpecific.VP8.simulcastIdx] =
>+          codec_specific_info->codecSpecific.VP8.layerSync;
>+      temporal_layer_[codec_specific_info->codecSpecific.VP8.simulcastIdx] =
>+          codec_specific_info->codecSpecific.VP8.temporalIdx;
>+    }
>+    return Result(Result::OK, encoded_image._timeStamp);
>+  }
>+  // This method only makes sense for VP8.
>+  void GetLastEncodedFrameInfo(int* temporal_layer,
>+                               bool* layer_sync,
>+                               int stream) {
>+    *temporal_layer = temporal_layer_[stream];
>+    *layer_sync = layer_sync_[stream];
>+  }
>+  void GetLastEncodedKeyFrame(EncodedImage* encoded_key_frame) {
>+    *encoded_key_frame = encoded_key_frame_;
>+  }
>+  void GetLastEncodedFrame(EncodedImage* encoded_frame) {
>+    *encoded_frame = encoded_frame_;
>+  }
>+
>+ private:
>+  EncodedImage encoded_key_frame_;
>+  EncodedImage encoded_frame_;
>+  int temporal_layer_[kNumberOfSimulcastStreams];
>+  bool layer_sync_[kNumberOfSimulcastStreams];
>+};
>+
>+class SimulcastTestFixtureImpl::TestDecodedImageCallback
>+    : public DecodedImageCallback {
>+ public:
>+  TestDecodedImageCallback() : decoded_frames_(0) {}
>+  int32_t Decoded(VideoFrame& decoded_image) override {
>+    rtc::scoped_refptr<I420BufferInterface> i420_buffer =
>+        decoded_image.video_frame_buffer()->ToI420();
>+    for (int i = 0; i < decoded_image.width(); ++i) {
>+      EXPECT_NEAR(kColorY, i420_buffer->DataY()[i], 1);
>+    }
>+
>+    // TODO(mikhal): Verify the difference between U,V and the original.
>+    for (int i = 0; i < i420_buffer->ChromaWidth(); ++i) {
>+      EXPECT_NEAR(kColorU, i420_buffer->DataU()[i], 4);
>+      EXPECT_NEAR(kColorV, i420_buffer->DataV()[i], 4);
>+    }
>+    decoded_frames_++;
>+    return 0;
>+  }
>+  int32_t Decoded(VideoFrame& decoded_image, int64_t decode_time_ms) override {
>+    RTC_NOTREACHED();
>+    return -1;
>+  }
>+  void Decoded(VideoFrame& decoded_image,
>+               absl::optional<int32_t> decode_time_ms,
>+               absl::optional<uint8_t> qp) override {
>+    Decoded(decoded_image);
>+  }
>+  int DecodedFrames() { return decoded_frames_; }
>+
>+ private:
>+  int decoded_frames_;
>+};
>+
>+namespace {
>+
>+void SetPlane(uint8_t* data, uint8_t value, int width, int height, int stride) {
>+  for (int i = 0; i < height; i++, data += stride) {
>+    // Setting allocated area to zero - setting only image size to
>+    // requested values - will make it easier to distinguish between image
>+    // size and frame size (accounting for stride).
>+    memset(data, value, width);
>+    memset(data + width, 0, stride - width);
>+  }
>+}
>+
>+// Fills in an I420Buffer from |plane_colors|.
>+void CreateImage(const rtc::scoped_refptr<I420Buffer>& buffer,
>+                 int plane_colors[kNumOfPlanes]) {
>+  SetPlane(buffer->MutableDataY(), plane_colors[0], buffer->width(),
>+           buffer->height(), buffer->StrideY());
>+
>+  SetPlane(buffer->MutableDataU(), plane_colors[1], buffer->ChromaWidth(),
>+           buffer->ChromaHeight(), buffer->StrideU());
>+
>+  SetPlane(buffer->MutableDataV(), plane_colors[2], buffer->ChromaWidth(),
>+           buffer->ChromaHeight(), buffer->StrideV());
>+}
>+
>+void ConfigureStream(int width,
>+                     int height,
>+                     int max_bitrate,
>+                     int min_bitrate,
>+                     int target_bitrate,
>+                     SimulcastStream* stream,
>+                     int num_temporal_layers) {
>+  assert(stream);
>+  stream->width = width;
>+  stream->height = height;
>+  stream->maxBitrate = max_bitrate;
>+  stream->minBitrate = min_bitrate;
>+  stream->targetBitrate = target_bitrate;
>+  if (num_temporal_layers >= 0) {
>+    stream->numberOfTemporalLayers = num_temporal_layers;
>+  }
>+  stream->qpMax = 45;
>+  stream->active = true;
>+}
>+
>+}  // namespace
>+
>+void SimulcastTestFixtureImpl::DefaultSettings(
>+    VideoCodec* settings,
>+    const int* temporal_layer_profile,
>+    VideoCodecType codec_type) {
>+  RTC_CHECK(settings);
>+  memset(settings, 0, sizeof(VideoCodec));
>+  settings->codecType = codec_type;
>+  // 96 to 127 dynamic payload types for video codecs
>+  settings->plType = 120;
>+  settings->startBitrate = 300;
>+  settings->minBitrate = 30;
>+  settings->maxBitrate = 0;
>+  settings->maxFramerate = 30;
>+  settings->width = kDefaultWidth;
>+  settings->height = kDefaultHeight;
>+  settings->numberOfSimulcastStreams = kNumberOfSimulcastStreams;
>+  settings->active = true;
>+  ASSERT_EQ(3, kNumberOfSimulcastStreams);
>+  settings->timing_frame_thresholds = {kDefaultTimingFramesDelayMs,
>+                                       kDefaultOutlierFrameSizePercent};
>+  ConfigureStream(kDefaultWidth / 4, kDefaultHeight / 4, kMaxBitrates[0],
>+                  kMinBitrates[0], kTargetBitrates[0],
>+                  &settings->simulcastStream[0], temporal_layer_profile[0]);
>+  ConfigureStream(kDefaultWidth / 2, kDefaultHeight / 2, kMaxBitrates[1],
>+                  kMinBitrates[1], kTargetBitrates[1],
>+                  &settings->simulcastStream[1], temporal_layer_profile[1]);
>+  ConfigureStream(kDefaultWidth, kDefaultHeight, kMaxBitrates[2],
>+                  kMinBitrates[2], kTargetBitrates[2],
>+                  &settings->simulcastStream[2], temporal_layer_profile[2]);
>+    if (codec_type == kVideoCodecVP8) {
>+      settings->VP8()->denoisingOn = true;
>+      settings->VP8()->automaticResizeOn = false;
>+      settings->VP8()->frameDroppingOn = true;
>+      settings->VP8()->keyFrameInterval = 3000;
>+    } else {
>+      settings->H264()->frameDroppingOn = true;
>+      settings->H264()->keyFrameInterval = 3000;
>+    }
>+}
>+
>+SimulcastTestFixtureImpl::SimulcastTestFixtureImpl(
>+    std::unique_ptr<VideoEncoderFactory> encoder_factory,
>+    std::unique_ptr<VideoDecoderFactory> decoder_factory,
>+    SdpVideoFormat video_format)
>+    : codec_type_(PayloadStringToCodecType(video_format.name)) {
>+  encoder_ = encoder_factory->CreateVideoEncoder(video_format);
>+  decoder_ = decoder_factory->CreateVideoDecoder(video_format);
>+  SetUpCodec(codec_type_ == kVideoCodecVP8 ? kDefaultTemporalLayerProfile
>+    : kNoTemporalLayerProfile);
>+}
>+
>+SimulcastTestFixtureImpl::~SimulcastTestFixtureImpl() {
>+  encoder_->Release();
>+  decoder_->Release();
>+}
>+
>+void SimulcastTestFixtureImpl::SetUpCodec(const int* temporal_layer_profile) {
>+  encoder_->RegisterEncodeCompleteCallback(&encoder_callback_);
>+  decoder_->RegisterDecodeCompleteCallback(&decoder_callback_);
>+  DefaultSettings(&settings_, temporal_layer_profile, codec_type_);
>+  SetUpRateAllocator();
>+  EXPECT_EQ(0, encoder_->InitEncode(&settings_, 1, 1200));
>+  EXPECT_EQ(0, decoder_->InitDecode(&settings_, 1));
>+  input_buffer_ = I420Buffer::Create(kDefaultWidth, kDefaultHeight);
>+  input_buffer_->InitializeData();
>+  input_frame_.reset(new VideoFrame(input_buffer_, webrtc::kVideoRotation_0,
>+                                    0 /* timestamp_us */));
>+}
>+
>+void SimulcastTestFixtureImpl::SetUpRateAllocator() {
>+  rate_allocator_.reset(new SimulcastRateAllocator(settings_));
>+}
>+
>+void SimulcastTestFixtureImpl::SetRates(uint32_t bitrate_kbps, uint32_t fps) {
>+  encoder_->SetRateAllocation(
>+      rate_allocator_->GetAllocation(bitrate_kbps * 1000, fps), fps);
>+}
>+
>+void SimulcastTestFixtureImpl::RunActiveStreamsTest(
>+    const std::vector<bool> active_streams) {
>+  std::vector<FrameType> frame_types(kNumberOfSimulcastStreams,
>+                                     kVideoFrameDelta);
>+  UpdateActiveStreams(active_streams);
>+  // Set sufficient bitrate for all streams so we can test active without
>+  // bitrate being an issue.
>+  SetRates(kMaxBitrates[0] + kMaxBitrates[1] + kMaxBitrates[2], 30);
>+
>+  ExpectStreams(kVideoFrameKey, active_streams);
>+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>+
>+  ExpectStreams(kVideoFrameDelta, active_streams);
>+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>+}
>+
>+void SimulcastTestFixtureImpl::UpdateActiveStreams(
>+    const std::vector<bool> active_streams) {
>+  ASSERT_EQ(static_cast<int>(active_streams.size()), kNumberOfSimulcastStreams);
>+  for (size_t i = 0; i < active_streams.size(); ++i) {
>+    settings_.simulcastStream[i].active = active_streams[i];
>+  }
>+  // Re initialize the allocator and encoder with the new settings.
>+  // TODO(bugs.webrtc.org/8807): Currently, we do a full "hard"
>+  // reconfiguration of the allocator and encoder. When the video bitrate
>+  // allocator has support for updating active streams without a
>+  // reinitialization, we can just call that here instead.
>+  SetUpRateAllocator();
>+  EXPECT_EQ(0, encoder_->InitEncode(&settings_, 1, 1200));
>+}
>+
>+void SimulcastTestFixtureImpl::ExpectStreams(
>+    FrameType frame_type,
>+    const std::vector<bool> expected_streams_active) {
>+  ASSERT_EQ(static_cast<int>(expected_streams_active.size()),
>+            kNumberOfSimulcastStreams);
>+  if (expected_streams_active[0]) {
>+    EXPECT_CALL(
>+        encoder_callback_,
>+        OnEncodedImage(
>+            AllOf(Field(&EncodedImage::_frameType, frame_type),
>+                  Field(&EncodedImage::_encodedWidth, kDefaultWidth / 4),
>+                  Field(&EncodedImage::_encodedHeight, kDefaultHeight / 4)),
>+            _, _))
>+        .Times(1)
>+        .WillRepeatedly(Return(
>+            EncodedImageCallback::Result(EncodedImageCallback::Result::OK, 0)));
>+  }
>+  if (expected_streams_active[1]) {
>+    EXPECT_CALL(
>+        encoder_callback_,
>+        OnEncodedImage(
>+            AllOf(Field(&EncodedImage::_frameType, frame_type),
>+                  Field(&EncodedImage::_encodedWidth, kDefaultWidth / 2),
>+                  Field(&EncodedImage::_encodedHeight, kDefaultHeight / 2)),
>+            _, _))
>+        .Times(1)
>+        .WillRepeatedly(Return(
>+            EncodedImageCallback::Result(EncodedImageCallback::Result::OK, 0)));
>+  }
>+  if (expected_streams_active[2]) {
>+    EXPECT_CALL(encoder_callback_,
>+                OnEncodedImage(
>+                    AllOf(Field(&EncodedImage::_frameType, frame_type),
>+                          Field(&EncodedImage::_encodedWidth, kDefaultWidth),
>+                          Field(&EncodedImage::_encodedHeight, kDefaultHeight)),
>+                    _, _))
>+        .Times(1)
>+        .WillRepeatedly(Return(
>+            EncodedImageCallback::Result(EncodedImageCallback::Result::OK, 0)));
>+  }
>+}
>+
>+void SimulcastTestFixtureImpl::ExpectStreams(FrameType frame_type,
>+                                             int expected_video_streams) {
>+  ASSERT_GE(expected_video_streams, 0);
>+  ASSERT_LE(expected_video_streams, kNumberOfSimulcastStreams);
>+  std::vector<bool> expected_streams_active(kNumberOfSimulcastStreams, false);
>+  for (int i = 0; i < expected_video_streams; ++i) {
>+    expected_streams_active[i] = true;
>+  }
>+  ExpectStreams(frame_type, expected_streams_active);
>+}
>+
>+void SimulcastTestFixtureImpl::VerifyTemporalIdxAndSyncForAllSpatialLayers(
>+    TestEncodedImageCallback* encoder_callback,
>+    const int* expected_temporal_idx,
>+    const bool* expected_layer_sync,
>+    int num_spatial_layers) {
>+  int temporal_layer = -1;
>+  bool layer_sync = false;
>+  for (int i = 0; i < num_spatial_layers; i++) {
>+    encoder_callback->GetLastEncodedFrameInfo(&temporal_layer, &layer_sync, i);
>+    EXPECT_EQ(expected_temporal_idx[i], temporal_layer);
>+    EXPECT_EQ(expected_layer_sync[i], layer_sync);
>+  }
>+}
>+
>+// We currently expect all active streams to generate a key frame even though
>+// a key frame was only requested for some of them.
>+void SimulcastTestFixtureImpl::TestKeyFrameRequestsOnAllStreams() {
>+  SetRates(kMaxBitrates[2], 30);  // To get all three streams.
>+  std::vector<FrameType> frame_types(kNumberOfSimulcastStreams,
>+                                     kVideoFrameDelta);
>+  ExpectStreams(kVideoFrameKey, kNumberOfSimulcastStreams);
>+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>+
>+  ExpectStreams(kVideoFrameDelta, kNumberOfSimulcastStreams);
>+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>+
>+  frame_types[0] = kVideoFrameKey;
>+  ExpectStreams(kVideoFrameKey, kNumberOfSimulcastStreams);
>+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>+
>+  std::fill(frame_types.begin(), frame_types.end(), kVideoFrameDelta);
>+  frame_types[1] = kVideoFrameKey;
>+  ExpectStreams(kVideoFrameKey, kNumberOfSimulcastStreams);
>+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>+
>+  std::fill(frame_types.begin(), frame_types.end(), kVideoFrameDelta);
>+  frame_types[2] = kVideoFrameKey;
>+  ExpectStreams(kVideoFrameKey, kNumberOfSimulcastStreams);
>+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>+
>+  std::fill(frame_types.begin(), frame_types.end(), kVideoFrameDelta);
>+  ExpectStreams(kVideoFrameDelta, kNumberOfSimulcastStreams);
>+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>+}
>+
>+void SimulcastTestFixtureImpl::TestPaddingAllStreams() {
>+  // We should always encode the base layer.
>+  SetRates(kMinBitrates[0] - 1, 30);
>+  std::vector<FrameType> frame_types(kNumberOfSimulcastStreams,
>+                                     kVideoFrameDelta);
>+  ExpectStreams(kVideoFrameKey, 1);
>+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>+
>+  ExpectStreams(kVideoFrameDelta, 1);
>+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>+}
>+
>+void SimulcastTestFixtureImpl::TestPaddingTwoStreams() {
>+  // We have just enough to get only the first stream and padding for two.
>+  SetRates(kMinBitrates[0], 30);
>+  std::vector<FrameType> frame_types(kNumberOfSimulcastStreams,
>+                                     kVideoFrameDelta);
>+  ExpectStreams(kVideoFrameKey, 1);
>+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>+
>+  ExpectStreams(kVideoFrameDelta, 1);
>+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>+}
>+
>+void SimulcastTestFixtureImpl::TestPaddingTwoStreamsOneMaxedOut() {
>+  // We are just below limit of sending second stream, so we should get
>+  // the first stream maxed out (at |maxBitrate|), and padding for two.
>+  SetRates(kTargetBitrates[0] + kMinBitrates[1] - 1, 30);
>+  std::vector<FrameType> frame_types(kNumberOfSimulcastStreams,
>+                                     kVideoFrameDelta);
>+  ExpectStreams(kVideoFrameKey, 1);
>+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>+
>+  ExpectStreams(kVideoFrameDelta, 1);
>+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>+}
>+
>+void SimulcastTestFixtureImpl::TestPaddingOneStream() {
>+  // We have just enough to send two streams, so padding for one stream.
>+  SetRates(kTargetBitrates[0] + kMinBitrates[1], 30);
>+  std::vector<FrameType> frame_types(kNumberOfSimulcastStreams,
>+                                     kVideoFrameDelta);
>+  ExpectStreams(kVideoFrameKey, 2);
>+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>+
>+  ExpectStreams(kVideoFrameDelta, 2);
>+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>+}
>+
>+void SimulcastTestFixtureImpl::TestPaddingOneStreamTwoMaxedOut() {
>+  // We are just below limit of sending third stream, so we should get
>+  // first stream's rate maxed out at |targetBitrate|, second at |maxBitrate|.
>+  SetRates(kTargetBitrates[0] + kTargetBitrates[1] + kMinBitrates[2] - 1, 30);
>+  std::vector<FrameType> frame_types(kNumberOfSimulcastStreams,
>+                                     kVideoFrameDelta);
>+  ExpectStreams(kVideoFrameKey, 2);
>+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>+
>+  ExpectStreams(kVideoFrameDelta, 2);
>+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>+}
>+
>+void SimulcastTestFixtureImpl::TestSendAllStreams() {
>+  // We have just enough to send all streams.
>+  SetRates(kTargetBitrates[0] + kTargetBitrates[1] + kMinBitrates[2], 30);
>+  std::vector<FrameType> frame_types(kNumberOfSimulcastStreams,
>+                                     kVideoFrameDelta);
>+  ExpectStreams(kVideoFrameKey, 3);
>+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>+
>+  ExpectStreams(kVideoFrameDelta, 3);
>+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>+}
>+
>+void SimulcastTestFixtureImpl::TestDisablingStreams() {
>+  // We should get three media streams.
>+  SetRates(kMaxBitrates[0] + kMaxBitrates[1] + kMaxBitrates[2], 30);
>+  std::vector<FrameType> frame_types(kNumberOfSimulcastStreams,
>+                                     kVideoFrameDelta);
>+  ExpectStreams(kVideoFrameKey, 3);
>+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>+
>+  ExpectStreams(kVideoFrameDelta, 3);
>+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>+
>+  // We should only get two streams and padding for one.
>+  SetRates(kTargetBitrates[0] + kTargetBitrates[1] + kMinBitrates[2] / 2, 30);
>+  ExpectStreams(kVideoFrameDelta, 2);
>+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>+
>+  // We should only get the first stream and padding for two.
>+  SetRates(kTargetBitrates[0] + kMinBitrates[1] / 2, 30);
>+  ExpectStreams(kVideoFrameDelta, 1);
>+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>+
>+  // We don't have enough bitrate for the thumbnail stream, but we should get
>+  // it anyway with current configuration.
>+  SetRates(kTargetBitrates[0] - 1, 30);
>+  ExpectStreams(kVideoFrameDelta, 1);
>+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>+
>+  // We should only get two streams and padding for one.
>+  SetRates(kTargetBitrates[0] + kTargetBitrates[1] + kMinBitrates[2] / 2, 30);
>+  // We get a key frame because a new stream is being enabled.
>+  ExpectStreams(kVideoFrameKey, 2);
>+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>+
>+  // We should get all three streams.
>+  SetRates(kTargetBitrates[0] + kTargetBitrates[1] + kTargetBitrates[2], 30);
>+  // We get a key frame because a new stream is being enabled.
>+  ExpectStreams(kVideoFrameKey, 3);
>+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>+}
>+
>+void SimulcastTestFixtureImpl::TestActiveStreams() {
>+  // All streams on.
>+  RunActiveStreamsTest({true, true, true});
>+  // All streams off.
>+  RunActiveStreamsTest({false, false, false});
>+  // Low stream off.
>+  RunActiveStreamsTest({false, true, true});
>+  // Middle stream off.
>+  RunActiveStreamsTest({true, false, true});
>+  // High stream off.
>+  RunActiveStreamsTest({true, true, false});
>+  // Only low stream turned on.
>+  RunActiveStreamsTest({true, false, false});
>+  // Only middle stream turned on.
>+  RunActiveStreamsTest({false, true, false});
>+  // Only high stream turned on.
>+  RunActiveStreamsTest({false, false, true});
>+}
>+
>+void SimulcastTestFixtureImpl::SwitchingToOneStream(int width, int height) {
>+  const int* temporal_layer_profile = nullptr;
>+  // Disable all streams except the last and set the bitrate of the last to
>+  // 100 kbps. This verifies the way GTP switches to screenshare mode.
>+  if (codec_type_ == kVideoCodecVP8) {
>+    settings_.VP8()->numberOfTemporalLayers = 1;
>+    temporal_layer_profile = kDefaultTemporalLayerProfile;
>+  } else {
>+    temporal_layer_profile = kNoTemporalLayerProfile;
>+  }
>+  settings_.maxBitrate = 100;
>+  settings_.startBitrate = 100;
>+  settings_.width = width;
>+  settings_.height = height;
>+  for (int i = 0; i < settings_.numberOfSimulcastStreams - 1; ++i) {
>+    settings_.simulcastStream[i].maxBitrate = 0;
>+    settings_.simulcastStream[i].width = settings_.width;
>+    settings_.simulcastStream[i].height = settings_.height;
>+    settings_.simulcastStream[i].numberOfTemporalLayers = 1;
>+  }
>+  // Setting input image to new resolution.
>+  input_buffer_ = I420Buffer::Create(settings_.width, settings_.height);
>+  input_buffer_->InitializeData();
>+
>+  input_frame_.reset(new VideoFrame(input_buffer_, webrtc::kVideoRotation_0,
>+                                    0 /* timestamp_us */));
>+
>+  // The for loop above did not set the bitrate of the highest layer.
>+  settings_.simulcastStream[settings_.numberOfSimulcastStreams - 1].maxBitrate =
>+      0;
>+  // The highest layer has to correspond to the non-simulcast resolution.
>+  settings_.simulcastStream[settings_.numberOfSimulcastStreams - 1].width =
>+      settings_.width;
>+  settings_.simulcastStream[settings_.numberOfSimulcastStreams - 1].height =
>+      settings_.height;
>+  SetUpRateAllocator();
>+  EXPECT_EQ(0, encoder_->InitEncode(&settings_, 1, 1200));
>+
>+  // Encode one frame and verify.
>+  SetRates(kMaxBitrates[0] + kMaxBitrates[1], 30);
>+  std::vector<FrameType> frame_types(kNumberOfSimulcastStreams,
>+                                     kVideoFrameDelta);
>+  EXPECT_CALL(
>+      encoder_callback_,
>+      OnEncodedImage(AllOf(Field(&EncodedImage::_frameType, kVideoFrameKey),
>+                           Field(&EncodedImage::_encodedWidth, width),
>+                           Field(&EncodedImage::_encodedHeight, height)),
>+                     _, _))
>+      .Times(1)
>+      .WillRepeatedly(Return(
>+          EncodedImageCallback::Result(EncodedImageCallback::Result::OK, 0)));
>+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>+
>+  // Switch back.
>+  DefaultSettings(&settings_, temporal_layer_profile, codec_type_);
>+  // Start at the lowest bitrate for enabling base stream.
>+  settings_.startBitrate = kMinBitrates[0];
>+  SetUpRateAllocator();
>+  EXPECT_EQ(0, encoder_->InitEncode(&settings_, 1, 1200));
>+  SetRates(settings_.startBitrate, 30);
>+  ExpectStreams(kVideoFrameKey, 1);
>+  // Resize |input_frame_| to the new resolution.
>+  input_buffer_ = I420Buffer::Create(settings_.width, settings_.height);
>+  input_buffer_->InitializeData();
>+  input_frame_.reset(new VideoFrame(input_buffer_, webrtc::kVideoRotation_0,
>+                                    0 /* timestamp_us */));
>+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, &frame_types));
>+}
>+
>+void SimulcastTestFixtureImpl::TestSwitchingToOneStream() {
>+  SwitchingToOneStream(1024, 768);
>+}
>+
>+void SimulcastTestFixtureImpl::TestSwitchingToOneOddStream() {
>+  SwitchingToOneStream(1023, 769);
>+}
>+
>+void SimulcastTestFixtureImpl::TestSwitchingToOneSmallStream() {
>+  SwitchingToOneStream(4, 4);
>+}
>+
>+// Test the layer pattern and sync flag for various spatial-temporal patterns.
>+// 3-3-3 pattern: 3 temporal layers for all spatial streams, so same
>+// temporal_layer id and layer_sync is expected for all streams.
>+void SimulcastTestFixtureImpl::TestSpatioTemporalLayers333PatternEncoder() {
>+  EXPECT_EQ(codec_type_, kVideoCodecVP8);
>+  TestEncodedImageCallback encoder_callback;
>+  encoder_->RegisterEncodeCompleteCallback(&encoder_callback);
>+  SetRates(kMaxBitrates[2], 30);  // To get all three streams.
>+
>+  int expected_temporal_idx[3] = {-1, -1, -1};
>+  bool expected_layer_sync[3] = {false, false, false};
>+
>+  // First frame: #0.
>+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, NULL));
>+  SetExpectedValues3<int>(0, 0, 0, expected_temporal_idx);
>+  SetExpectedValues3<bool>(true, true, true, expected_layer_sync);
>+  VerifyTemporalIdxAndSyncForAllSpatialLayers(
>+      &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
>+
>+  // Next frame: #1.
>+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, NULL));
>+  SetExpectedValues3<int>(2, 2, 2, expected_temporal_idx);
>+  SetExpectedValues3<bool>(true, true, true, expected_layer_sync);
>+  VerifyTemporalIdxAndSyncForAllSpatialLayers(
>+      &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
>+
>+  // Next frame: #2.
>+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, NULL));
>+  SetExpectedValues3<int>(1, 1, 1, expected_temporal_idx);
>+  SetExpectedValues3<bool>(true, true, true, expected_layer_sync);
>+  VerifyTemporalIdxAndSyncForAllSpatialLayers(
>+      &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
>+
>+  // Next frame: #3.
>+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, NULL));
>+  SetExpectedValues3<int>(2, 2, 2, expected_temporal_idx);
>+  SetExpectedValues3<bool>(false, false, false, expected_layer_sync);
>+  VerifyTemporalIdxAndSyncForAllSpatialLayers(
>+      &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
>+
>+  // Next frame: #4.
>+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, NULL));
>+  SetExpectedValues3<int>(0, 0, 0, expected_temporal_idx);
>+  SetExpectedValues3<bool>(false, false, false, expected_layer_sync);
>+  VerifyTemporalIdxAndSyncForAllSpatialLayers(
>+      &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
>+
>+  // Next frame: #5.
>+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, NULL));
>+  SetExpectedValues3<int>(2, 2, 2, expected_temporal_idx);
>+  SetExpectedValues3<bool>(false, false, false, expected_layer_sync);
>+  VerifyTemporalIdxAndSyncForAllSpatialLayers(
>+      &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
>+}
>+
>+// Test the layer pattern and sync flag for various spatial-temporal patterns.
>+// 3-2-1 pattern: 3 temporal layers for lowest resolution, 2 for middle, and
>+// 1 temporal layer for highest resolution.
>+// For this profile, we expect the temporal index pattern to be:
>+// 1st stream: 0, 2, 1, 2, ....
>+// 2nd stream: 0, 1, 0, 1, ...
>+// 3rd stream: -1, -1, -1, -1, ....
>+// Regarding the 3rd stream, note that a stream/encoder with 1 temporal layer
>+// should always have temporal layer idx set to kNoTemporalIdx = -1.
>+// Since CodecSpecificInfoVP8.temporalIdx is uint8_t, this will wrap to 255.
>+// TODO(marpan): Although this seems safe for now, we should fix this.
>+void SimulcastTestFixtureImpl::TestSpatioTemporalLayers321PatternEncoder() {
>+  EXPECT_EQ(codec_type_, kVideoCodecVP8);
>+  int temporal_layer_profile[3] = {3, 2, 1};
>+  SetUpCodec(temporal_layer_profile);
>+  TestEncodedImageCallback encoder_callback;
>+  encoder_->RegisterEncodeCompleteCallback(&encoder_callback);
>+  SetRates(kMaxBitrates[2], 30);  // To get all three streams.
>+
>+  int expected_temporal_idx[3] = {-1, -1, -1};
>+  bool expected_layer_sync[3] = {false, false, false};
>+
>+  // First frame: #0.
>+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, NULL));
>+  SetExpectedValues3<int>(0, 0, 255, expected_temporal_idx);
>+  SetExpectedValues3<bool>(true, true, false, expected_layer_sync);
>+  VerifyTemporalIdxAndSyncForAllSpatialLayers(
>+      &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
>+
>+  // Next frame: #1.
>+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, NULL));
>+  SetExpectedValues3<int>(2, 1, 255, expected_temporal_idx);
>+  SetExpectedValues3<bool>(true, true, false, expected_layer_sync);
>+  VerifyTemporalIdxAndSyncForAllSpatialLayers(
>+      &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
>+
>+  // Next frame: #2.
>+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, NULL));
>+  SetExpectedValues3<int>(1, 0, 255, expected_temporal_idx);
>+  SetExpectedValues3<bool>(true, false, false, expected_layer_sync);
>+  VerifyTemporalIdxAndSyncForAllSpatialLayers(
>+      &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
>+
>+  // Next frame: #3.
>+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, NULL));
>+  SetExpectedValues3<int>(2, 1, 255, expected_temporal_idx);
>+  SetExpectedValues3<bool>(false, false, false, expected_layer_sync);
>+  VerifyTemporalIdxAndSyncForAllSpatialLayers(
>+      &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
>+
>+  // Next frame: #4.
>+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, NULL));
>+  SetExpectedValues3<int>(0, 0, 255, expected_temporal_idx);
>+  SetExpectedValues3<bool>(false, false, false, expected_layer_sync);
>+  VerifyTemporalIdxAndSyncForAllSpatialLayers(
>+      &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
>+
>+  // Next frame: #5.
>+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, NULL));
>+  SetExpectedValues3<int>(2, 1, 255, expected_temporal_idx);
>+  SetExpectedValues3<bool>(false, true, false, expected_layer_sync);
>+  VerifyTemporalIdxAndSyncForAllSpatialLayers(
>+      &encoder_callback, expected_temporal_idx, expected_layer_sync, 3);
>+}
>+
>+void SimulcastTestFixtureImpl::TestStrideEncodeDecode() {
>+  TestEncodedImageCallback encoder_callback;
>+  TestDecodedImageCallback decoder_callback;
>+  encoder_->RegisterEncodeCompleteCallback(&encoder_callback);
>+  decoder_->RegisterDecodeCompleteCallback(&decoder_callback);
>+
>+  SetRates(kMaxBitrates[2], 30);  // To get all three streams.
>+  // Setting two (possibly) problematic use cases for stride:
>+  // 1. stride > width 2. stride_y != stride_uv/2
>+  int stride_y = kDefaultWidth + 20;
>+  int stride_uv = ((kDefaultWidth + 1) / 2) + 5;
>+  input_buffer_ = I420Buffer::Create(kDefaultWidth, kDefaultHeight, stride_y,
>+                                     stride_uv, stride_uv);
>+  input_frame_.reset(new VideoFrame(input_buffer_, webrtc::kVideoRotation_0,
>+                                    0 /* timestamp_us */));
>+
>+  // Set color.
>+  int plane_offset[kNumOfPlanes];
>+  plane_offset[kYPlane] = kColorY;
>+  plane_offset[kUPlane] = kColorU;
>+  plane_offset[kVPlane] = kColorV;
>+  CreateImage(input_buffer_, plane_offset);
>+
>+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, NULL));
>+
>+  // Change color.
>+  plane_offset[kYPlane] += 1;
>+  plane_offset[kUPlane] += 1;
>+  plane_offset[kVPlane] += 1;
>+  CreateImage(input_buffer_, plane_offset);
>+  input_frame_->set_timestamp(input_frame_->timestamp() + 3000);
>+  EXPECT_EQ(0, encoder_->Encode(*input_frame_, NULL, NULL));
>+
>+  EncodedImage encoded_frame;
>+  // Only encoding one frame - so will be a key frame.
>+  encoder_callback.GetLastEncodedKeyFrame(&encoded_frame);
>+  EXPECT_EQ(0, decoder_->Decode(encoded_frame, false, NULL, 0));
>+  encoder_callback.GetLastEncodedFrame(&encoded_frame);
>+  decoder_->Decode(encoded_frame, false, NULL, 0);
>+  EXPECT_EQ(2, decoder_callback.DecodedFrames());
>+}
>+
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/simulcast_test_fixture_impl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/simulcast_test_fixture_impl.h
>new file mode 100644
>index 00000000000..6634a69ad1c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/simulcast_test_fixture_impl.h
>@@ -0,0 +1,91 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_VIDEO_CODING_UTILITY_SIMULCAST_TEST_FIXTURE_IMPL_H_
>+#define MODULES_VIDEO_CODING_UTILITY_SIMULCAST_TEST_FIXTURE_IMPL_H_
>+
>+#include <memory>
>+#include <vector>
>+
>+#include "api/test/simulcast_test_fixture.h"
>+#include "api/video/i420_buffer.h"
>+#include "api/video/video_frame.h"
>+#include "api/video_codecs/video_decoder_factory.h"
>+#include "api/video_codecs/video_encoder_factory.h"
>+#include "common_types.h"  // NOLINT(build/include)
>+#include "modules/video_coding/utility/simulcast_rate_allocator.h"
>+#include "modules/video_coding/include/mock/mock_video_codec_interface.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+class SimulcastTestFixtureImpl final : public SimulcastTestFixture {
>+ public:
>+  SimulcastTestFixtureImpl(
>+      std::unique_ptr<VideoEncoderFactory> encoder_factory,
>+      std::unique_ptr<VideoDecoderFactory> decoder_factory,
>+      SdpVideoFormat video_format);
>+  ~SimulcastTestFixtureImpl() final;
>+
>+  // Implements SimulcastTestFixture.
>+  void TestKeyFrameRequestsOnAllStreams() override;
>+  void TestPaddingAllStreams() override;
>+  void TestPaddingTwoStreams() override;
>+  void TestPaddingTwoStreamsOneMaxedOut() override;
>+  void TestPaddingOneStream() override;
>+  void TestPaddingOneStreamTwoMaxedOut() override;
>+  void TestSendAllStreams() override;
>+  void TestDisablingStreams() override;
>+  void TestActiveStreams() override;
>+  void TestSwitchingToOneStream() override;
>+  void TestSwitchingToOneOddStream() override;
>+  void TestSwitchingToOneSmallStream() override;
>+  void TestSpatioTemporalLayers333PatternEncoder() override;
>+  void TestSpatioTemporalLayers321PatternEncoder() override;
>+  void TestStrideEncodeDecode() override;
>+
>+  static void DefaultSettings(VideoCodec* settings,
>+                              const int* temporal_layer_profile,
>+                              VideoCodecType codec_type);
>+
>+ private:
>+  class TestEncodedImageCallback;
>+  class TestDecodedImageCallback;
>+
>+  void SetUpCodec(const int* temporal_layer_profile);
>+  void SetUpRateAllocator();
>+  void SetRates(uint32_t bitrate_kbps, uint32_t fps);
>+  void RunActiveStreamsTest(const std::vector<bool> active_streams);
>+  void UpdateActiveStreams(const std::vector<bool> active_streams);
>+  void ExpectStreams(FrameType frame_type,
>+                     const std::vector<bool> expected_streams_active);
>+  void ExpectStreams(FrameType frame_type, int expected_video_streams);
>+  void VerifyTemporalIdxAndSyncForAllSpatialLayers(
>+      TestEncodedImageCallback* encoder_callback,
>+      const int* expected_temporal_idx,
>+      const bool* expected_layer_sync,
>+      int num_spatial_layers);
>+  void SwitchingToOneStream(int width, int height);
>+
>+  std::unique_ptr<VideoEncoder> encoder_;
>+  MockEncodedImageCallback encoder_callback_;
>+  std::unique_ptr<VideoDecoder> decoder_;
>+  MockDecodedImageCallback decoder_callback_;
>+  VideoCodec settings_;
>+  rtc::scoped_refptr<I420Buffer> input_buffer_;
>+  std::unique_ptr<VideoFrame> input_frame_;
>+  std::unique_ptr<SimulcastRateAllocator> rate_allocator_;
>+  VideoCodecType codec_type_;
>+};
>+
>+}  // namespace test
>+}  // namespace webrtc
>+
>+#endif  // MODULES_VIDEO_CODING_UTILITY_SIMULCAST_TEST_FIXTURE_IMPL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/simulcast_utility.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/simulcast_utility.cc
>new file mode 100644
>index 00000000000..60cf0627da9
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/simulcast_utility.cc
>@@ -0,0 +1,65 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/video_coding/utility/simulcast_utility.h"
>+
>+namespace webrtc {
>+
>+uint32_t SimulcastUtility::SumStreamMaxBitrate(int streams,
>+                                               const VideoCodec& codec) {
>+  uint32_t bitrate_sum = 0;
>+  for (int i = 0; i < streams; ++i) {
>+    bitrate_sum += codec.simulcastStream[i].maxBitrate;
>+  }
>+  return bitrate_sum;
>+}
>+
>+int SimulcastUtility::NumberOfSimulcastStreams(const VideoCodec& codec) {
>+  int streams =
>+      codec.numberOfSimulcastStreams < 1 ? 1 : codec.numberOfSimulcastStreams;
>+  uint32_t simulcast_max_bitrate = SumStreamMaxBitrate(streams, codec);
>+  if (simulcast_max_bitrate == 0) {
>+    streams = 1;
>+  }
>+  return streams;
>+}
>+
>+bool SimulcastUtility::ValidSimulcastResolutions(const VideoCodec& codec,
>+                                                 int num_streams) {
>+  if (codec.width != codec.simulcastStream[num_streams - 1].width ||
>+      codec.height != codec.simulcastStream[num_streams - 1].height) {
>+    return false;
>+  }
>+  for (int i = 0; i < num_streams; ++i) {
>+    if (codec.width * codec.simulcastStream[i].height !=
>+        codec.height * codec.simulcastStream[i].width) {
>+      return false;
>+    }
>+  }
>+  for (int i = 1; i < num_streams; ++i) {
>+    if (codec.simulcastStream[i].width !=
>+        codec.simulcastStream[i - 1].width * 2) {
>+      return false;
>+    }
>+  }
>+  return true;
>+}
>+
>+bool SimulcastUtility::ValidSimulcastTemporalLayers(const VideoCodec& codec,
>+                                                    int num_streams) {
>+  for (int i = 0; i < num_streams - 1; ++i) {
>+    if (codec.simulcastStream[i].numberOfTemporalLayers !=
>+        codec.simulcastStream[i + 1].numberOfTemporalLayers)
>+      return false;
>+  }
>+  return true;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/simulcast_utility.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/simulcast_utility.h
>new file mode 100644
>index 00000000000..cf690f29a81
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/simulcast_utility.h
>@@ -0,0 +1,30 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_VIDEO_CODING_UTILITY_SIMULCAST_UTILITY_H_
>+#define MODULES_VIDEO_CODING_UTILITY_SIMULCAST_UTILITY_H_
>+
>+#include "api/video_codecs/video_encoder.h"
>+
>+namespace webrtc {
>+
>+class SimulcastUtility {
>+ public:
>+  static uint32_t SumStreamMaxBitrate(int streams, const VideoCodec& codec);
>+  static int NumberOfSimulcastStreams(const VideoCodec& codec);
>+  static bool ValidSimulcastResolutions(const VideoCodec& codec,
>+                                        int num_streams);
>+  static bool ValidSimulcastTemporalLayers(const VideoCodec& codec,
>+                                           int num_streams);
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // MODULES_VIDEO_CODING_UTILITY_SIMULCAST_UTILITY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/vp8_header_parser.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/vp8_header_parser.cc
>index d1cfb08b44f..6b65fad4007 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/vp8_header_parser.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/utility/vp8_header_parser.cc
>@@ -10,6 +10,7 @@
> #include "modules/video_coding/utility/vp8_header_parser.h"
> 
> #include "rtc_base/logging.h"
>+#include "rtc_base/system/arch.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_codec_initializer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_codec_initializer.cc
>index 87b49e4912b..cd73551891a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_codec_initializer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_codec_initializer.cc
>@@ -10,98 +10,57 @@
> 
> #include "modules/video_coding/include/video_codec_initializer.h"
> 
>+#include "api/video/video_bitrate_allocator.h"
>+#include "api/video_codecs/video_encoder.h"
> #include "common_types.h"  // NOLINT(build/include)
>-#include "common_video/include/video_bitrate_allocator.h"
>-#include "modules/video_coding/codecs/vp8/screenshare_layers.h"
>-#include "modules/video_coding/codecs/vp8/simulcast_rate_allocator.h"
>-#include "modules/video_coding/codecs/vp8/temporal_layers.h"
>+#include "modules/video_coding/codecs/vp9/svc_config.h"
>+#include "modules/video_coding/codecs/vp9/svc_rate_allocator.h"
> #include "modules/video_coding/include/video_coding_defines.h"
> #include "modules/video_coding/utility/default_video_bitrate_allocator.h"
>-#include "rtc_base/basictypes.h"
>+#include "modules/video_coding/utility/simulcast_rate_allocator.h"
> #include "rtc_base/logging.h"
>+#include "rtc_base/system/fallthrough.h"
> #include "system_wrappers/include/clock.h"
> 
> namespace webrtc {
>-namespace {
>-bool TemporalLayersConfigured(const std::vector<VideoStream>& streams) {
>-  for (const VideoStream& stream : streams) {
>-    if (stream.temporal_layer_thresholds_bps.size() > 0)
>-      return true;
>-  }
>-  return false;
>-}
>-}  // namespace
> 
> bool VideoCodecInitializer::SetupCodec(
>     const VideoEncoderConfig& config,
>-    const VideoSendStream::Config::EncoderSettings settings,
>     const std::vector<VideoStream>& streams,
>-    bool nack_enabled,
>     VideoCodec* codec,
>     std::unique_ptr<VideoBitrateAllocator>* bitrate_allocator) {
>-  if (PayloadStringToCodecType(settings.payload_name) == kVideoCodecStereo) {
>-    VideoSendStream::Config::EncoderSettings associated_codec_settings =
>-        settings;
>-    associated_codec_settings.payload_name =
>-        CodecTypeToPayloadString(kVideoCodecVP9);
>-    if (!SetupCodec(config, associated_codec_settings, streams, nack_enabled,
>-                    codec, bitrate_allocator)) {
>+  if (config.codec_type == kVideoCodecMultiplex) {
>+    VideoEncoderConfig associated_config = config.Copy();
>+    associated_config.codec_type = kVideoCodecVP9;
>+    if (!SetupCodec(associated_config, streams, codec, bitrate_allocator)) {
>       RTC_LOG(LS_ERROR) << "Failed to create stereo encoder configuration.";
>       return false;
>     }
>-    codec->codecType = kVideoCodecStereo;
>-    strncpy(codec->plName, settings.payload_name.c_str(),
>-            sizeof(codec->plName));
>+    codec->codecType = kVideoCodecMultiplex;
>     return true;
>   }
> 
>-  *codec =
>-      VideoEncoderConfigToVideoCodec(config, streams, settings.payload_name,
>-                                     settings.payload_type, nack_enabled);
>-#if !defined(RTC_DISABLE_VP8)
>-  std::unique_ptr<TemporalLayersFactory> tl_factory;
>-  switch (codec->codecType) {
>-    case kVideoCodecVP8: {
>-      if (!codec->VP8()->tl_factory) {
>-        if (codec->mode == kScreensharing &&
>-            (codec->numberOfSimulcastStreams > 1 ||
>-             (codec->numberOfSimulcastStreams == 1 &&
>-              codec->VP8()->numberOfTemporalLayers == 2))) {
>-          // Conference mode temporal layering for screen content.
>-          tl_factory.reset(new ScreenshareTemporalLayersFactory());
>-        } else {
>-          // Standard video temporal layers.
>-          tl_factory.reset(new TemporalLayersFactory());
>-        }
>-        codec->VP8()->tl_factory = tl_factory.get();
>-      }
>-      break;
>-    }
>-    default: {
>-      // TODO(sprang): Warn, once we have specific allocators for all supported
>-      //               codec types.
>-      break;
>-    }
>-  }
>-  *bitrate_allocator = CreateBitrateAllocator(*codec, std::move(tl_factory));
>-#endif
>+  *codec = VideoEncoderConfigToVideoCodec(config, streams);
>+  *bitrate_allocator = CreateBitrateAllocator(*codec);
>+
>   return true;
> }
> 
> std::unique_ptr<VideoBitrateAllocator>
>-VideoCodecInitializer::CreateBitrateAllocator(
>-    const VideoCodec& codec,
>-    std::unique_ptr<TemporalLayersFactory> tl_factory) {
>+VideoCodecInitializer::CreateBitrateAllocator(const VideoCodec& codec) {
>   std::unique_ptr<VideoBitrateAllocator> rate_allocator;
> 
>   switch (codec.codecType) {
>-    case kVideoCodecVP8: {
>-#if !defined(RTC_DISABLE_VP8)
>-      // Set up default VP8 temporal layer factory, if not provided.
>-      rate_allocator.reset(
>-          new SimulcastRateAllocator(codec, std::move(tl_factory)));
>+    case kVideoCodecVP8:
>+      RTC_FALLTHROUGH();
>+    case kVideoCodecH264:
>+      rate_allocator.reset(new SimulcastRateAllocator(codec));
>+      break;
>+#if !defined(RTC_DISABLE_VP9)
>+    case kVideoCodecVP9:
>+      rate_allocator.reset(new SvcRateAllocator(codec));
>+      break;
> #endif
>-    } break;
>     default:
>       rate_allocator.reset(new DefaultVideoBitrateAllocator(codec));
>   }
>@@ -112,103 +71,45 @@ VideoCodecInitializer::CreateBitrateAllocator(
> // TODO(sprang): Split this up and separate the codec specific parts.
> VideoCodec VideoCodecInitializer::VideoEncoderConfigToVideoCodec(
>     const VideoEncoderConfig& config,
>-    const std::vector<VideoStream>& streams,
>-    const std::string& payload_name,
>-    int payload_type,
>-    bool nack_enabled) {
>+    const std::vector<VideoStream>& streams) {
>   static const int kEncoderMinBitrateKbps = 30;
>   RTC_DCHECK(!streams.empty());
>   RTC_DCHECK_GE(config.min_transmit_bitrate_bps, 0);
> 
>   VideoCodec video_codec;
>   memset(&video_codec, 0, sizeof(video_codec));
>-  video_codec.codecType = PayloadStringToCodecType(payload_name);
>+  video_codec.codecType = config.codec_type;
> 
>   switch (config.content_type) {
>     case VideoEncoderConfig::ContentType::kRealtimeVideo:
>-      video_codec.mode = kRealtimeVideo;
>+      video_codec.mode = VideoCodecMode::kRealtimeVideo;
>       break;
>     case VideoEncoderConfig::ContentType::kScreen:
>-      video_codec.mode = kScreensharing;
>-      if (!streams.empty() &&
>-          streams[0].temporal_layer_thresholds_bps.size() == 1) {
>-        video_codec.targetBitrate =
>-            streams[0].temporal_layer_thresholds_bps[0] / 1000;
>-      }
>-      break;
>-  }
>-
>-  if (config.encoder_specific_settings)
>-    config.encoder_specific_settings->FillEncoderSpecificSettings(&video_codec);
>-
>-  switch (video_codec.codecType) {
>-    case kVideoCodecVP8: {
>-      if (!config.encoder_specific_settings)
>-        *video_codec.VP8() = VideoEncoder::GetDefaultVp8Settings();
>-      video_codec.VP8()->numberOfTemporalLayers = static_cast<unsigned char>(
>-          streams.back().temporal_layer_thresholds_bps.size() + 1);
>-
>-      if (nack_enabled && !TemporalLayersConfigured(streams)) {
>-        RTC_LOG(LS_INFO)
>-            << "No temporal layers and nack enabled -> resilience off";
>-        video_codec.VP8()->resilience = kResilienceOff;
>-      }
>-      break;
>-    }
>-    case kVideoCodecVP9: {
>-      if (!config.encoder_specific_settings)
>-        *video_codec.VP9() = VideoEncoder::GetDefaultVp9Settings();
>-      if (video_codec.mode == kScreensharing &&
>-          config.encoder_specific_settings) {
>-        video_codec.VP9()->flexibleMode = true;
>-        // For now VP9 screensharing use 1 temporal and 2 spatial layers.
>-        RTC_DCHECK_EQ(1, video_codec.VP9()->numberOfTemporalLayers);
>-        RTC_DCHECK_EQ(2, video_codec.VP9()->numberOfSpatialLayers);
>-      }
>-      video_codec.VP9()->numberOfTemporalLayers = static_cast<unsigned char>(
>-          streams.back().temporal_layer_thresholds_bps.size() + 1);
>-
>-      if (nack_enabled && !TemporalLayersConfigured(streams) &&
>-          video_codec.VP9()->numberOfSpatialLayers == 1) {
>-        RTC_LOG(LS_INFO) << "No temporal or spatial layers and nack enabled -> "
>-                         << "resilience off";
>-        video_codec.VP9()->resilienceOn = false;
>-      }
>-      break;
>-    }
>-    case kVideoCodecH264: {
>-      if (!config.encoder_specific_settings)
>-        *video_codec.H264() = VideoEncoder::GetDefaultH264Settings();
>-      break;
>-    }
>-    default:
>-      // TODO(pbos): Support encoder_settings codec-agnostically.
>-      RTC_DCHECK(!config.encoder_specific_settings)
>-          << "Encoder-specific settings for codec type not wired up.";
>+      video_codec.mode = VideoCodecMode::kScreensharing;
>       break;
>   }
> 
>-  strncpy(video_codec.plName, payload_name.c_str(), kPayloadNameSize - 1);
>-  video_codec.plName[kPayloadNameSize - 1] = '\0';
>-  video_codec.plType = payload_type;
>+  // TODO(nisse): The plType field should be deleted. Luckily, our
>+  // callers don't need it.
>+  video_codec.plType = 0;
>   video_codec.numberOfSimulcastStreams =
>       static_cast<unsigned char>(streams.size());
>   video_codec.minBitrate = streams[0].min_bitrate_bps / 1000;
>+  bool codec_active = false;
>+  for (const VideoStream& stream : streams) {
>+    if (stream.active) {
>+      codec_active = true;
>+      break;
>+    }
>+  }
>+  // Set active for the entire video codec for the non simulcast case.
>+  video_codec.active = codec_active;
>   if (video_codec.minBitrate < kEncoderMinBitrateKbps)
>     video_codec.minBitrate = kEncoderMinBitrateKbps;
>   video_codec.timing_frame_thresholds = {kDefaultTimingFramesDelayMs,
>                                          kDefaultOutlierFrameSizePercent};
>   RTC_DCHECK_LE(streams.size(), kMaxSimulcastStreams);
>-  if (video_codec.codecType == kVideoCodecVP9) {
>-    // If the vector is empty, bitrates will be configured automatically.
>-    RTC_DCHECK(config.spatial_layers.empty() ||
>-               config.spatial_layers.size() ==
>-                   video_codec.VP9()->numberOfSpatialLayers);
>-    RTC_DCHECK_LE(video_codec.VP9()->numberOfSpatialLayers,
>-                  kMaxSimulcastStreams);
>-    for (size_t i = 0; i < config.spatial_layers.size(); ++i)
>-      video_codec.spatialLayers[i] = config.spatial_layers[i];
>-  }
>+
>   for (size_t i = 0; i < streams.size(); ++i) {
>     SimulcastStream* sim_stream = &video_codec.simulcastStream[i];
>     RTC_DCHECK_GT(streams[i].width, 0);
>@@ -231,8 +132,9 @@ VideoCodec VideoCodecInitializer::VideoEncoderConfigToVideoCodec(
>     sim_stream->targetBitrate = streams[i].target_bitrate_bps / 1000;
>     sim_stream->maxBitrate = streams[i].max_bitrate_bps / 1000;
>     sim_stream->qpMax = streams[i].max_qp;
>-    sim_stream->numberOfTemporalLayers = static_cast<unsigned char>(
>-        streams[i].temporal_layer_thresholds_bps.size() + 1);
>+    sim_stream->numberOfTemporalLayers =
>+        static_cast<unsigned char>(streams[i].num_temporal_layers.value_or(1));
>+    sim_stream->active = streams[i].active;
> 
>     video_codec.width =
>         std::max(video_codec.width, static_cast<uint16_t>(streams[i].width));
>@@ -257,6 +159,95 @@ VideoCodec VideoCodecInitializer::VideoEncoderConfigToVideoCodec(
> 
>   RTC_DCHECK_GT(streams[0].max_framerate, 0);
>   video_codec.maxFramerate = streams[0].max_framerate;
>+
>+  // Set codec specific options
>+  if (config.encoder_specific_settings)
>+    config.encoder_specific_settings->FillEncoderSpecificSettings(&video_codec);
>+
>+  switch (video_codec.codecType) {
>+    case kVideoCodecVP8: {
>+      if (!config.encoder_specific_settings) {
>+        *video_codec.VP8() = VideoEncoder::GetDefaultVp8Settings();
>+      }
>+
>+      video_codec.VP8()->numberOfTemporalLayers = static_cast<unsigned char>(
>+          streams.back().num_temporal_layers.value_or(
>+              video_codec.VP8()->numberOfTemporalLayers));
>+      RTC_DCHECK_GE(video_codec.VP8()->numberOfTemporalLayers, 1);
>+      RTC_DCHECK_LE(video_codec.VP8()->numberOfTemporalLayers,
>+                    kMaxTemporalStreams);
>+
>+      break;
>+    }
>+#if !defined(RTC_DISABLE_VP9)
>+    case kVideoCodecVP9: {
>+      if (!config.encoder_specific_settings) {
>+        *video_codec.VP9() = VideoEncoder::GetDefaultVp9Settings();
>+      }
>+
>+      video_codec.VP9()->numberOfTemporalLayers = static_cast<unsigned char>(
>+          streams.back().num_temporal_layers.value_or(
>+              video_codec.VP9()->numberOfTemporalLayers));
>+      RTC_DCHECK_GE(video_codec.VP9()->numberOfTemporalLayers, 1);
>+      RTC_DCHECK_LE(video_codec.VP9()->numberOfTemporalLayers,
>+                    kMaxTemporalStreams);
>+
>+      RTC_DCHECK(config.spatial_layers.empty() ||
>+                 config.spatial_layers.size() ==
>+                     video_codec.VP9()->numberOfSpatialLayers);
>+
>+      std::vector<SpatialLayer> spatial_layers;
>+      if (!config.spatial_layers.empty()) {
>+        // Layering is set explicitly.
>+        spatial_layers = config.spatial_layers;
>+      } else {
>+        spatial_layers =
>+            GetSvcConfig(video_codec.width, video_codec.height,
>+                         video_codec.VP9()->numberOfSpatialLayers,
>+                         video_codec.VP9()->numberOfTemporalLayers,
>+                         video_codec.mode == VideoCodecMode::kScreensharing);
>+
>+        const bool no_spatial_layering = (spatial_layers.size() == 1);
>+        if (no_spatial_layering) {
>+          // Use codec's bitrate limits.
>+          spatial_layers.back().minBitrate = video_codec.minBitrate;
>+          spatial_layers.back().maxBitrate = video_codec.maxBitrate;
>+        }
>+      }
>+
>+      RTC_DCHECK(!spatial_layers.empty());
>+      for (size_t i = 0; i < spatial_layers.size(); ++i) {
>+        video_codec.spatialLayers[i] = spatial_layers[i];
>+      }
>+
>+      // Update layering settings.
>+      video_codec.VP9()->numberOfSpatialLayers =
>+          static_cast<unsigned char>(spatial_layers.size());
>+      RTC_DCHECK_GE(video_codec.VP9()->numberOfSpatialLayers, 1);
>+      RTC_DCHECK_LE(video_codec.VP9()->numberOfSpatialLayers,
>+                    kMaxSpatialLayers);
>+
>+      video_codec.VP9()->numberOfTemporalLayers = static_cast<unsigned char>(
>+          spatial_layers.back().numberOfTemporalLayers);
>+      RTC_DCHECK_GE(video_codec.VP9()->numberOfTemporalLayers, 1);
>+      RTC_DCHECK_LE(video_codec.VP9()->numberOfTemporalLayers,
>+                    kMaxTemporalStreams);
>+
>+      break;
>+    }
>+#endif
>+    case kVideoCodecH264: {
>+      if (!config.encoder_specific_settings)
>+        *video_codec.H264() = VideoEncoder::GetDefaultH264Settings();
>+      break;
>+    }
>+    default:
>+      // TODO(pbos): Support encoder_settings codec-agnostically.
>+      RTC_DCHECK(!config.encoder_specific_settings)
>+          << "Encoder-specific settings for codec type not wired up.";
>+      break;
>+  }
>+
>   return video_codec;
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_codec_initializer_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_codec_initializer_unittest.cc
>index 0862b7282e6..80a30160f46 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_codec_initializer_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_codec_initializer_unittest.cc
>@@ -8,19 +8,18 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>+#include "modules/video_coding/include/video_codec_initializer.h"
>+#include "api/video/video_bitrate_allocator.h"
> #include "api/video_codecs/video_encoder.h"
>-#include "common_video/include/video_bitrate_allocator.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/video_coding/codecs/vp8/temporal_layers.h"
>-#include "modules/video_coding/include/video_codec_initializer.h"
>+#include "modules/video_coding/codecs/vp9/include/vp9_globals.h"
> #include "rtc_base/refcountedobject.h"
> #include "test/gtest.h"
> 
> namespace webrtc {
> 
> namespace {
>-static const char* kVp8PayloadName = "VP8";
>-static const int kVp8PayloadType = 100;
> static const int kDefaultWidth = 1280;
> static const int kDefaultHeight = 720;
> static const int kDefaultFrameRate = 30;
>@@ -29,7 +28,7 @@ static const uint32_t kDefaultTargetBitrateBps = 2000000;
> static const uint32_t kDefaultMaxBitrateBps = 2000000;
> static const uint32_t kDefaultMinTransmitBitrateBps = 400000;
> static const int kDefaultMaxQp = 48;
>-static const uint32_t kScreenshareTl0BitrateBps = 100000;
>+static const uint32_t kScreenshareTl0BitrateBps = 200000;
> static const uint32_t kScreenshareCodecTargetBitrateBps = 200000;
> static const uint32_t kScreenshareDefaultFramerate = 5;
> // Bitrates for the temporal layers of the higher screenshare simulcast stream.
>@@ -37,27 +36,10 @@ static const uint32_t kHighScreenshareTl0Bps = 800000;
> static const uint32_t kHighScreenshareTl1Bps = 1200000;
> }  // namespace
> 
>-/*
>- *   static bool SetupCodec(
>-      const VideoEncoderConfig& config,
>-      const VideoSendStream::Config::EncoderSettings settings,
>-      const std::vector<VideoStream>& streams,
>-      bool nack_enabled,
>-      VideoCodec* codec,
>-      std::unique_ptr<VideoBitrateAllocator>* bitrate_allocator);
>-
>-  // Create a bitrate allocator for the specified codec. |tl_factory| is
>-  // optional, if it is populated, ownership of that instance will be
>-  // transferred to the VideoBitrateAllocator instance.
>-  static std::unique_ptr<VideoBitrateAllocator> CreateBitrateAllocator(
>-      const VideoCodec& codec,
>-      std::unique_ptr<TemporalLayersFactory> tl_factory);
>- */
>-
> // TODO(sprang): Extend coverage to handle the rest of the codec initializer.
> class VideoCodecInitializerTest : public ::testing::Test {
>  public:
>-  VideoCodecInitializerTest() : nack_enabled_(false) {}
>+  VideoCodecInitializerTest() {}
>   virtual ~VideoCodecInitializerTest() {}
> 
>  protected:
>@@ -66,6 +48,8 @@ class VideoCodecInitializerTest : public ::testing::Test {
>                 int num_temporal_streams,
>                 bool screenshare) {
>     config_ = VideoEncoderConfig();
>+    config_.codec_type = type;
>+
>     if (screenshare) {
>       config_.min_transmit_bitrate_bps = kDefaultMinTransmitBitrateBps;
>       config_.content_type = VideoEncoderConfig::ContentType::kScreen;
>@@ -77,10 +61,13 @@ class VideoCodecInitializerTest : public ::testing::Test {
>       vp8_settings.numberOfTemporalLayers = num_temporal_streams;
>       config_.encoder_specific_settings = new rtc::RefCountedObject<
>           webrtc::VideoEncoderConfig::Vp8EncoderSpecificSettings>(vp8_settings);
>-      settings_.payload_name = kVp8PayloadName;
>-      settings_.payload_type = kVp8PayloadType;
>-    } else if (type == VideoCodecType::kVideoCodecStereo) {
>-    } else {
>+    } else if (type == VideoCodecType::kVideoCodecVP9) {
>+      VideoCodecVP9 vp9_settings = VideoEncoder::GetDefaultVp9Settings();
>+      vp9_settings.numberOfSpatialLayers = num_spatial_streams;
>+      vp9_settings.numberOfTemporalLayers = num_temporal_streams;
>+      config_.encoder_specific_settings = new rtc::RefCountedObject<
>+          webrtc::VideoEncoderConfig::Vp9EncoderSpecificSettings>(vp9_settings);
>+    } else if (type != VideoCodecType::kVideoCodecMultiplex) {
>       ADD_FAILURE() << "Unexpected codec type: " << type;
>     }
>   }
>@@ -89,21 +76,18 @@ class VideoCodecInitializerTest : public ::testing::Test {
>     codec_out_ = VideoCodec();
>     bitrate_allocator_out_.reset();
>     temporal_layers_.clear();
>-    if (!VideoCodecInitializer::SetupCodec(config_, settings_, streams_,
>-                                           nack_enabled_, &codec_out_,
>+    if (!VideoCodecInitializer::SetupCodec(config_, streams_, &codec_out_,
>                                            &bitrate_allocator_out_)) {
>       return false;
>     }
>-    if (codec_out_.codecType == VideoCodecType::kVideoCodecStereo)
>+    if (codec_out_.codecType == VideoCodecType::kVideoCodecMultiplex)
>       return true;
>+
>     // Make sure temporal layers instances have been created.
>     if (codec_out_.codecType == VideoCodecType::kVideoCodecVP8) {
>-      if (!codec_out_.VP8()->tl_factory)
>-        return false;
>-
>       for (int i = 0; i < codec_out_.numberOfSimulcastStreams; ++i) {
>-        temporal_layers_.emplace_back(codec_out_.VP8()->tl_factory->Create(
>-            i, streams_[i].temporal_layer_thresholds_bps.size() + 1, 0));
>+        temporal_layers_.emplace_back(
>+            TemporalLayers::CreateTemporalLayers(codec_out_, i));
>       }
>     }
>     return true;
>@@ -118,6 +102,8 @@ class VideoCodecInitializerTest : public ::testing::Test {
>     stream.target_bitrate_bps = kDefaultTargetBitrateBps;
>     stream.max_bitrate_bps = kDefaultMaxBitrateBps;
>     stream.max_qp = kDefaultMaxQp;
>+    stream.num_temporal_layers = 1;
>+    stream.active = true;
>     return stream;
>   }
> 
>@@ -127,15 +113,14 @@ class VideoCodecInitializerTest : public ::testing::Test {
>     stream.target_bitrate_bps = kScreenshareTl0BitrateBps;
>     stream.max_bitrate_bps = 1000000;
>     stream.max_framerate = kScreenshareDefaultFramerate;
>-    stream.temporal_layer_thresholds_bps.push_back(kScreenshareTl0BitrateBps);
>+    stream.num_temporal_layers = 2;
>+    stream.active = true;
>     return stream;
>   }
> 
>   // Input settings.
>   VideoEncoderConfig config_;
>-  VideoSendStream::Config::EncoderSettings settings_;
>   std::vector<VideoStream> streams_;
>-  bool nack_enabled_;
> 
>   // Output.
>   VideoCodec codec_out_;
>@@ -148,13 +133,29 @@ TEST_F(VideoCodecInitializerTest, SingleStreamVp8Screenshare) {
>   streams_.push_back(DefaultStream());
>   EXPECT_TRUE(InitializeCodec());
> 
>-  BitrateAllocation bitrate_allocation = bitrate_allocator_out_->GetAllocation(
>-      kDefaultTargetBitrateBps, kDefaultFrameRate);
>+  VideoBitrateAllocation bitrate_allocation =
>+      bitrate_allocator_out_->GetAllocation(kDefaultTargetBitrateBps,
>+                                            kDefaultFrameRate);
>   EXPECT_EQ(1u, codec_out_.numberOfSimulcastStreams);
>   EXPECT_EQ(1u, codec_out_.VP8()->numberOfTemporalLayers);
>   EXPECT_EQ(kDefaultTargetBitrateBps, bitrate_allocation.get_sum_bps());
> }
> 
>+TEST_F(VideoCodecInitializerTest, SingleStreamVp8ScreenshareInactive) {
>+  SetUpFor(VideoCodecType::kVideoCodecVP8, 1, 1, true);
>+  VideoStream inactive_stream = DefaultStream();
>+  inactive_stream.active = false;
>+  streams_.push_back(inactive_stream);
>+  EXPECT_TRUE(InitializeCodec());
>+
>+  VideoBitrateAllocation bitrate_allocation =
>+      bitrate_allocator_out_->GetAllocation(kDefaultTargetBitrateBps,
>+                                            kDefaultFrameRate);
>+  EXPECT_EQ(1u, codec_out_.numberOfSimulcastStreams);
>+  EXPECT_EQ(1u, codec_out_.VP8()->numberOfTemporalLayers);
>+  EXPECT_EQ(0U, bitrate_allocation.get_sum_bps());
>+}
>+
> TEST_F(VideoCodecInitializerTest, TemporalLayeredVp8Screenshare) {
>   SetUpFor(VideoCodecType::kVideoCodecVP8, 1, 2, true);
>   streams_.push_back(DefaultScreenshareStream());
>@@ -162,14 +163,15 @@ TEST_F(VideoCodecInitializerTest, TemporalLayeredVp8Screenshare) {
> 
>   EXPECT_EQ(1u, codec_out_.numberOfSimulcastStreams);
>   EXPECT_EQ(2u, codec_out_.VP8()->numberOfTemporalLayers);
>-  BitrateAllocation bitrate_allocation = bitrate_allocator_out_->GetAllocation(
>-      kScreenshareCodecTargetBitrateBps, kScreenshareDefaultFramerate);
>+  VideoBitrateAllocation bitrate_allocation =
>+      bitrate_allocator_out_->GetAllocation(kScreenshareCodecTargetBitrateBps,
>+                                            kScreenshareDefaultFramerate);
>   EXPECT_EQ(kScreenshareCodecTargetBitrateBps,
>             bitrate_allocation.get_sum_bps());
>   EXPECT_EQ(kScreenshareTl0BitrateBps, bitrate_allocation.GetBitrate(0, 0));
> }
> 
>-TEST_F(VideoCodecInitializerTest, SimlucastVp8Screenshare) {
>+TEST_F(VideoCodecInitializerTest, SimulcastVp8Screenshare) {
>   SetUpFor(VideoCodecType::kVideoCodecVP8, 2, 1, true);
>   streams_.push_back(DefaultScreenshareStream());
>   VideoStream video_stream = DefaultStream();
>@@ -181,8 +183,9 @@ TEST_F(VideoCodecInitializerTest, SimlucastVp8Screenshare) {
>   EXPECT_EQ(1u, codec_out_.VP8()->numberOfTemporalLayers);
>   const uint32_t max_bitrate_bps =
>       streams_[0].target_bitrate_bps + streams_[1].max_bitrate_bps;
>-  BitrateAllocation bitrate_allocation = bitrate_allocator_out_->GetAllocation(
>-      max_bitrate_bps, kScreenshareDefaultFramerate);
>+  VideoBitrateAllocation bitrate_allocation =
>+      bitrate_allocator_out_->GetAllocation(max_bitrate_bps,
>+                                            kScreenshareDefaultFramerate);
>   EXPECT_EQ(max_bitrate_bps, bitrate_allocation.get_sum_bps());
>   EXPECT_EQ(static_cast<uint32_t>(streams_[0].target_bitrate_bps),
>             bitrate_allocation.GetSpatialLayerSum(0));
>@@ -190,14 +193,38 @@ TEST_F(VideoCodecInitializerTest, SimlucastVp8Screenshare) {
>             bitrate_allocation.GetSpatialLayerSum(1));
> }
> 
>-TEST_F(VideoCodecInitializerTest, HighFpsSimlucastVp8Screenshare) {
>+// Tests that when a video stream is inactive, then the bitrate allocation will
>+// be 0 for that stream.
>+TEST_F(VideoCodecInitializerTest, SimulcastVp8ScreenshareInactive) {
>+  SetUpFor(VideoCodecType::kVideoCodecVP8, 2, 1, true);
>+  streams_.push_back(DefaultScreenshareStream());
>+  VideoStream inactive_video_stream = DefaultStream();
>+  inactive_video_stream.active = false;
>+  inactive_video_stream.max_framerate = kScreenshareDefaultFramerate;
>+  streams_.push_back(inactive_video_stream);
>+  EXPECT_TRUE(InitializeCodec());
>+
>+  EXPECT_EQ(2u, codec_out_.numberOfSimulcastStreams);
>+  EXPECT_EQ(1u, codec_out_.VP8()->numberOfTemporalLayers);
>+  const uint32_t target_bitrate =
>+      streams_[0].target_bitrate_bps + streams_[1].target_bitrate_bps;
>+  VideoBitrateAllocation bitrate_allocation =
>+      bitrate_allocator_out_->GetAllocation(target_bitrate,
>+                                            kScreenshareDefaultFramerate);
>+  EXPECT_EQ(static_cast<uint32_t>(streams_[0].max_bitrate_bps),
>+            bitrate_allocation.get_sum_bps());
>+  EXPECT_EQ(static_cast<uint32_t>(streams_[0].max_bitrate_bps),
>+            bitrate_allocation.GetSpatialLayerSum(0));
>+  EXPECT_EQ(0U, bitrate_allocation.GetSpatialLayerSum(1));
>+}
>+
>+TEST_F(VideoCodecInitializerTest, HighFpsSimulcastVp8Screenshare) {
>   // Two simulcast streams, the lower one using legacy settings (two temporal
>   // streams, 5fps), the higher one using 3 temporal streams and 30fps.
>   SetUpFor(VideoCodecType::kVideoCodecVP8, 2, 3, true);
>   streams_.push_back(DefaultScreenshareStream());
>   VideoStream video_stream = DefaultStream();
>-  video_stream.temporal_layer_thresholds_bps.push_back(kHighScreenshareTl0Bps);
>-  video_stream.temporal_layer_thresholds_bps.push_back(kHighScreenshareTl1Bps);
>+  video_stream.num_temporal_layers = 3;
>   streams_.push_back(video_stream);
>   EXPECT_TRUE(InitializeCodec());
> 
>@@ -205,7 +232,7 @@ TEST_F(VideoCodecInitializerTest, HighFpsSimlucastVp8Screenshare) {
>   EXPECT_EQ(3u, codec_out_.VP8()->numberOfTemporalLayers);
>   const uint32_t max_bitrate_bps =
>       streams_[0].target_bitrate_bps + streams_[1].max_bitrate_bps;
>-  BitrateAllocation bitrate_allocation =
>+  VideoBitrateAllocation bitrate_allocation =
>       bitrate_allocator_out_->GetAllocation(max_bitrate_bps, kDefaultFrameRate);
>   EXPECT_EQ(max_bitrate_bps, bitrate_allocation.get_sum_bps());
>   EXPECT_EQ(static_cast<uint32_t>(streams_[0].target_bitrate_bps),
>@@ -217,10 +244,47 @@ TEST_F(VideoCodecInitializerTest, HighFpsSimlucastVp8Screenshare) {
>             bitrate_allocation.GetBitrate(1, 1));
> }
> 
>-TEST_F(VideoCodecInitializerTest, SingleStreamStereoCodec) {
>-  SetUpFor(VideoCodecType::kVideoCodecStereo, 1, 1, true);
>+TEST_F(VideoCodecInitializerTest, SingleStreamMultiplexCodec) {
>+  SetUpFor(VideoCodecType::kVideoCodecMultiplex, 1, 1, true);
>   streams_.push_back(DefaultStream());
>   EXPECT_TRUE(InitializeCodec());
> }
> 
>+TEST_F(VideoCodecInitializerTest, Vp9SvcDefaultLayering) {
>+  SetUpFor(VideoCodecType::kVideoCodecVP9, 3, 3, false);
>+  VideoStream stream = DefaultStream();
>+  stream.num_temporal_layers = 3;
>+  streams_.push_back(stream);
>+
>+  EXPECT_TRUE(InitializeCodec());
>+  EXPECT_EQ(codec_out_.VP9()->numberOfSpatialLayers, 3u);
>+  EXPECT_EQ(codec_out_.VP9()->numberOfTemporalLayers, 3u);
>+}
>+
>+TEST_F(VideoCodecInitializerTest, Vp9SvcAdjustedLayering) {
>+  SetUpFor(VideoCodecType::kVideoCodecVP9, 3, 3, false);
>+  VideoStream stream = DefaultStream();
>+  stream.num_temporal_layers = 3;
>+  // Set resolution which is only enough to produce 2 spatial layers.
>+  stream.width = kMinVp9SpatialLayerWidth * 2;
>+  stream.height = kMinVp9SpatialLayerHeight * 2;
>+
>+  streams_.push_back(stream);
>+
>+  EXPECT_TRUE(InitializeCodec());
>+  EXPECT_EQ(codec_out_.VP9()->numberOfSpatialLayers, 2u);
>+}
>+
>+TEST_F(VideoCodecInitializerTest,
>+       Vp9SingleSpatialLayerMaxBitrateIsEqualToCodecMaxBitrate) {
>+  SetUpFor(VideoCodecType::kVideoCodecVP9, 1, 3, false);
>+  VideoStream stream = DefaultStream();
>+  stream.num_temporal_layers = 3;
>+  streams_.push_back(stream);
>+
>+  EXPECT_TRUE(InitializeCodec());
>+  EXPECT_EQ(codec_out_.spatialLayers[0].maxBitrate,
>+            kDefaultMaxBitrateBps / 1000);
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_coding_defines.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_coding_defines.cc
>new file mode 100644
>index 00000000000..0a9cb3a3310
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_coding_defines.cc
>@@ -0,0 +1,23 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/video_coding/include/video_coding_defines.h"
>+
>+namespace webrtc {
>+
>+int32_t VCMReceiveCallback::ReceivedDecodedReferenceFrame(
>+    const uint64_t pictureId) {
>+  return -1;
>+}
>+void VCMReceiveCallback::OnIncomingPayloadType(int payload_type) {}
>+void VCMReceiveCallback::OnDecoderImplementationName(
>+    const char* implementation_name) {}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_coding_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_coding_impl.cc
>index a4c6f4db003..e02d4c0e144 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_coding_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_coding_impl.cc
>@@ -13,10 +13,9 @@
> #include <algorithm>
> #include <utility>
> 
>+#include "api/video/video_bitrate_allocator.h"
> #include "common_types.h"  // NOLINT(build/include)
>-#include "common_video/include/video_bitrate_allocator.h"
> #include "common_video/libyuv/include/webrtc_libyuv.h"
>-#include "modules/video_coding/codecs/vp8/temporal_layers.h"
> #include "modules/video_coding/encoded_frame.h"
> #include "modules/video_coding/include/video_codec_initializer.h"
> #include "modules/video_coding/include/video_codec_interface.h"
>@@ -28,6 +27,10 @@
> #include "system_wrappers/include/clock.h"
> 
> namespace webrtc {
>+EventWrapper* EventFactoryImpl::CreateEvent() {
>+  return EventWrapper::Create();
>+}
>+
> namespace vcm {
> 
> int64_t VCMProcessTimer::Period() const {
>@@ -100,47 +103,32 @@ class VideoCodingModuleImpl : public VideoCodingModule {
>     return receiver_time;
>   }
> 
>-  void Process() override {
>-    receiver_.Process();
>-  }
>+  void Process() override { receiver_.Process(); }
> 
>   int32_t RegisterSendCodec(const VideoCodec* sendCodec,
>                             uint32_t numberOfCores,
>                             uint32_t maxPayloadSize) override {
>-    if (sendCodec != nullptr && sendCodec->codecType == kVideoCodecVP8) {
>-      // Set up a rate allocator and temporal layers factory for this vp8
>+    if (sendCodec != nullptr && ((sendCodec->codecType == kVideoCodecVP8) ||
>+                                 (sendCodec->codecType == kVideoCodecH264))) {
>+      // Set up a rate allocator and temporal layers factory for this codec
>       // instance. The codec impl will have a raw pointer to the TL factory,
>       // and will call it when initializing. Since this can happen
>       // asynchronously keep the instance alive until destruction or until a
>       // new send codec is registered.
>-      VideoCodec vp8_codec = *sendCodec;
>-      std::unique_ptr<TemporalLayersFactory> tl_factory(
>-          new TemporalLayersFactory());
>-      vp8_codec.VP8()->tl_factory = tl_factory.get();
>-      rate_allocator_ = VideoCodecInitializer::CreateBitrateAllocator(
>-          vp8_codec, std::move(tl_factory));
>-      return sender_.RegisterSendCodec(&vp8_codec, numberOfCores,
>-                                       maxPayloadSize);
>+      VideoCodec codec = *sendCodec;
>+      rate_allocator_ = VideoCodecInitializer::CreateBitrateAllocator(codec);
>+      return sender_.RegisterSendCodec(&codec, numberOfCores, maxPayloadSize);
>     }
>     return sender_.RegisterSendCodec(sendCodec, numberOfCores, maxPayloadSize);
>   }
> 
>   int32_t RegisterExternalEncoder(VideoEncoder* externalEncoder,
>-                                  uint8_t payloadType,
>+                                  uint8_t /* payloadType */,
>                                   bool internalSource) override {
>-    sender_.RegisterExternalEncoder(externalEncoder, payloadType,
>-                                    internalSource);
>+    sender_.RegisterExternalEncoder(externalEncoder, internalSource);
>     return 0;
>   }
> 
>-  int Bitrate(unsigned int* bitrate) const override {
>-    return sender_.Bitrate(bitrate);
>-  }
>-
>-  int FrameRate(unsigned int* framerate) const override {
>-    return sender_.FrameRate(framerate);
>-  }
>-
>   int32_t SetChannelParameters(uint32_t target_bitrate,  // bits/s.
>                                uint8_t lossRate,
>                                int64_t rtt) override {
>@@ -148,11 +136,6 @@ class VideoCodingModuleImpl : public VideoCodingModule {
>                                         rate_allocator_.get(), nullptr);
>   }
> 
>-  int32_t RegisterProtectionCallback(
>-      VCMProtectionCallback* protection) override {
>-    return sender_.RegisterProtectionCallback(protection);
>-  }
>-
>   int32_t SetVideoProtection(VCMVideoProtection videoProtection,
>                              bool enable) override {
>     // TODO(pbos): Remove enable from receive-side protection modes as well.
>@@ -190,11 +173,6 @@ class VideoCodingModuleImpl : public VideoCodingModule {
>     return receiver_.RegisterReceiveCallback(receiveCallback);
>   }
> 
>-  int32_t RegisterReceiveStatisticsCallback(
>-      VCMReceiveStatisticsCallback* receiveStats) override {
>-    return receiver_.RegisterReceiveStatisticsCallback(receiveStats);
>-  }
>-
>   int32_t RegisterFrameTypeCallback(
>       VCMFrameTypeCallback* frameTypeCallback) override {
>     return receiver_.RegisterFrameTypeCallback(frameTypeCallback);
>@@ -216,16 +194,6 @@ class VideoCodingModuleImpl : public VideoCodingModule {
>     return receiver_.IncomingPacket(incomingPayload, payloadLength, rtpInfo);
>   }
> 
>-  int32_t SetMinimumPlayoutDelay(uint32_t minPlayoutDelayMs) override {
>-    return receiver_.SetMinimumPlayoutDelay(minPlayoutDelayMs);
>-  }
>-
>-  int32_t SetRenderDelay(uint32_t timeMS) override {
>-    return receiver_.SetRenderDelay(timeMS);
>-  }
>-
>-  int32_t Delay() const override { return receiver_.Delay(); }
>-
>   int SetReceiverRobustnessMode(ReceiverRobustness robustnessMode,
>                                 VCMDecodeErrorMode errorMode) override {
>     return receiver_.SetReceiverRobustnessMode(robustnessMode, errorMode);
>@@ -238,25 +206,11 @@ class VideoCodingModuleImpl : public VideoCodingModule {
>                                      max_incomplete_time_ms);
>   }
> 
>-  void SetDecodeErrorMode(VCMDecodeErrorMode decode_error_mode) override {
>-    return receiver_.SetDecodeErrorMode(decode_error_mode);
>-  }
>-
>-  int SetMinReceiverDelay(int desired_delay_ms) override {
>-    return receiver_.SetMinReceiverDelay(desired_delay_ms);
>-  }
>-
>-  int32_t SetReceiveChannelParameters(int64_t rtt) override {
>-    return receiver_.SetReceiveChannelParameters(rtt);
>-  }
>-
>   void RegisterPostEncodeImageCallback(
>       EncodedImageCallback* observer) override {
>     post_encode_callback_.Register(observer);
>   }
> 
>-  void TriggerDecoderShutdown() override { receiver_.TriggerDecoderShutdown(); }
>-
>  private:
>   rtc::ThreadChecker construction_thread_;
>   EncodedImageCallbackWrapper post_encode_callback_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_coding_impl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_coding_impl.h
>index 0bd5ab7ca01..2e28b0fbe99 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_coding_impl.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_coding_impl.h
>@@ -18,7 +18,8 @@
> #include <vector>
> 
> #include "common_video/include/frame_callback.h"
>-#include "modules/video_coding/codec_database.h"
>+#include "modules/video_coding/decoder_database.h"
>+#include "modules/video_coding/encoder_database.h"
> #include "modules/video_coding/frame_buffer.h"
> #include "modules/video_coding/generic_decoder.h"
> #include "modules/video_coding/generic_encoder.h"
>@@ -62,8 +63,7 @@ class VideoSender {
>  public:
>   typedef VideoCodingModule::SenderNackMode SenderNackMode;
> 
>-  VideoSender(Clock* clock,
>-              EncodedImageCallback* post_encode_callback);
>+  VideoSender(Clock* clock, EncodedImageCallback* post_encode_callback);
> 
>   ~VideoSender();
> 
>@@ -74,12 +74,8 @@ class VideoSender {
>                             uint32_t maxPayloadSize);
> 
>   void RegisterExternalEncoder(VideoEncoder* externalEncoder,
>-                               uint8_t payloadType,
>                                bool internalSource);
> 
>-  int Bitrate(unsigned int* bitrate) const;
>-  int FrameRate(unsigned int* framerate) const;
>-
>   // Update the channel parameters based on new rates and rtt. This will also
>   // cause an immediate call to VideoEncoder::SetRateAllocation().
>   int32_t SetChannelParameters(
>@@ -94,14 +90,10 @@ class VideoSender {
>   // caps may be updated to a change to a new VideoCodec or allocation mode.
>   // The new parameters will be stored as pending EncoderParameters, and the
>   // encoder will only be updated on the next frame.
>-  void UpdateChannelParemeters(
>+  void UpdateChannelParameters(
>       VideoBitrateAllocator* bitrate_allocator,
>       VideoBitrateAllocationObserver* bitrate_updated_callback);
> 
>-  // Deprecated:
>-  // TODO(perkj): Remove once no projects use it.
>-  int32_t RegisterProtectionCallback(VCMProtectionCallback* protection);
>-
>   int32_t AddVideoFrame(const VideoFrame& videoFrame,
>                         const CodecSpecificInfo* codecSpecificInfo);
> 
>@@ -121,7 +113,7 @@ class VideoSender {
>   media_optimization::MediaOptimization _mediaOpt;
>   VCMEncodedFrameCallback _encodedFrameCallback RTC_GUARDED_BY(encoder_crit_);
>   EncodedImageCallback* const post_encode_callback_;
>-  VCMCodecDataBase _codecDataBase RTC_GUARDED_BY(encoder_crit_);
>+  VCMEncoderDataBase _codecDataBase RTC_GUARDED_BY(encoder_crit_);
>   bool frame_dropper_enabled_ RTC_GUARDED_BY(encoder_crit_);
> 
>   // Must be accessed on the construction thread of VideoSender.
>@@ -142,7 +134,7 @@ class VideoReceiver : public Module {
>                 VCMTiming* timing,
>                 NackSender* nack_sender = nullptr,
>                 KeyFrameRequestSender* keyframe_request_sender = nullptr);
>-  ~VideoReceiver();
>+  ~VideoReceiver() override;
> 
>   int32_t RegisterReceiveCodec(const VideoCodec* receiveCodec,
>                                int32_t numberOfCores,
>@@ -160,9 +152,6 @@ class VideoReceiver : public Module {
> 
>   int32_t Decode(const webrtc::VCMEncodedFrame* frame);
> 
>-  // Called on the decoder thread when thread is exiting.
>-  void DecodingStopped();
>-
>   int32_t IncomingPacket(const uint8_t* incomingPayload,
>                          size_t payloadLength,
>                          const WebRtcRTPHeader& rtpInfo);
>@@ -180,48 +169,78 @@ class VideoReceiver : public Module {
>                        int max_incomplete_time_ms);
> 
>   void SetDecodeErrorMode(VCMDecodeErrorMode decode_error_mode);
>-  int SetMinReceiverDelay(int desired_delay_ms);
> 
>   int32_t SetReceiveChannelParameters(int64_t rtt);
>   int32_t SetVideoProtection(VCMVideoProtection videoProtection, bool enable);
> 
>   int64_t TimeUntilNextProcess() override;
>   void Process() override;
>+  void ProcessThreadAttached(ProcessThread* process_thread) override;
> 
>   void TriggerDecoderShutdown();
> 
>+  // Notification methods that are used to check our internal state and validate
>+  // threading assumptions. These are called by VideoReceiveStream.
>+  // See |IsDecoderThreadRunning()| for more details.
>+  void DecoderThreadStarting();
>+  void DecoderThreadStopped();
>+
>  protected:
>-  int32_t Decode(const webrtc::VCMEncodedFrame& frame)
>-      RTC_EXCLUSIVE_LOCKS_REQUIRED(receive_crit_);
>+  int32_t Decode(const webrtc::VCMEncodedFrame& frame);
>   int32_t RequestKeyFrame();
> 
>  private:
>-  rtc::ThreadChecker construction_thread_;
>+  // Used for DCHECKing thread correctness.
>+  // In build where DCHECKs are enabled, will return false before
>+  // DecoderThreadStarting is called, then true until DecoderThreadStopped
>+  // is called.
>+  // In builds where DCHECKs aren't enabled, it will return true.
>+  bool IsDecoderThreadRunning();
>+
>+  rtc::ThreadChecker construction_thread_checker_;
>+  rtc::ThreadChecker decoder_thread_checker_;
>+  rtc::ThreadChecker module_thread_checker_;
>   Clock* const clock_;
>   rtc::CriticalSection process_crit_;
>-  rtc::CriticalSection receive_crit_;
>   VCMTiming* _timing;
>   VCMReceiver _receiver;
>   VCMDecodedFrameCallback _decodedFrameCallback;
>-  VCMFrameTypeCallback* _frameTypeCallback RTC_GUARDED_BY(process_crit_);
>-  VCMReceiveStatisticsCallback* _receiveStatsCallback
>-      RTC_GUARDED_BY(process_crit_);
>-  VCMPacketRequestCallback* _packetRequestCallback
>-      RTC_GUARDED_BY(process_crit_);
> 
>-  VCMFrameBuffer _frameFromFile;
>+  // These callbacks are set on the construction thread before being attached
>+  // to the module thread or decoding started, so a lock is not required.
>+  VCMFrameTypeCallback* _frameTypeCallback;
>+  VCMReceiveStatisticsCallback* _receiveStatsCallback;
>+  VCMPacketRequestCallback* _packetRequestCallback;
>+
>+  // Used on both the module and decoder thread.
>   bool _scheduleKeyRequest RTC_GUARDED_BY(process_crit_);
>   bool drop_frames_until_keyframe_ RTC_GUARDED_BY(process_crit_);
>-  size_t max_nack_list_size_ RTC_GUARDED_BY(process_crit_);
>-
>-  VCMCodecDataBase _codecDataBase RTC_GUARDED_BY(receive_crit_);
>-  EncodedImageCallback* pre_decode_image_callback_;
> 
>-  VCMProcessTimer _receiveStatsTimer;
>-  VCMProcessTimer _retransmissionTimer;
>-  VCMProcessTimer _keyRequestTimer;
>-  QpParser qp_parser_;
>-  ThreadUnsafeOneTimeEvent first_frame_received_;
>+  // Modified on the construction thread while not attached to the process
>+  // thread.  Once attached to the process thread, its value is only read
>+  // so a lock is not required.
>+  size_t max_nack_list_size_;
>+
>+  // Callbacks are set before the decoder thread starts.
>+  // Once the decoder thread has been started, usage of |_codecDataBase| moves
>+  // over to the decoder thread.
>+  VCMDecoderDataBase _codecDataBase;
>+  EncodedImageCallback* const pre_decode_image_callback_;
>+
>+  VCMProcessTimer _receiveStatsTimer RTC_GUARDED_BY(module_thread_checker_);
>+  VCMProcessTimer _retransmissionTimer RTC_GUARDED_BY(module_thread_checker_);
>+  VCMProcessTimer _keyRequestTimer RTC_GUARDED_BY(module_thread_checker_);
>+  QpParser qp_parser_ RTC_GUARDED_BY(decoder_thread_checker_);
>+  ThreadUnsafeOneTimeEvent first_frame_received_
>+      RTC_GUARDED_BY(decoder_thread_checker_);
>+  // Modified on the construction thread. Can be read without a lock and assumed
>+  // to be non-null on the module and decoder threads.
>+  ProcessThread* process_thread_ = nullptr;
>+  bool is_attached_to_process_thread_
>+      RTC_GUARDED_BY(construction_thread_checker_) = false;
>+#if RTC_DCHECK_IS_ON
>+  bool decoder_thread_is_running_ = false;
>+#endif
> };
> 
> }  // namespace vcm
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_packet_buffer_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_packet_buffer_unittest.cc
>index 893649364a7..f588026ec91 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_packet_buffer_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_packet_buffer_unittest.cc
>@@ -54,14 +54,16 @@ class TestPacketBuffer : public ::testing::Test,
>   enum IsFirst { kFirst, kNotFirst };
>   enum IsLast { kLast, kNotLast };
> 
>-  bool Insert(uint16_t seq_num,           // packet sequence number
>-              IsKeyFrame keyframe,        // is keyframe
>-              IsFirst first,              // is first packet of frame
>-              IsLast last,                // is last packet of frame
>-              int data_size = 0,          // size of data
>-              uint8_t* data = nullptr) {  // data pointer
>+  bool Insert(uint16_t seq_num,             // packet sequence number
>+              IsKeyFrame keyframe,          // is keyframe
>+              IsFirst first,                // is first packet of frame
>+              IsLast last,                  // is last packet of frame
>+              int data_size = 0,            // size of data
>+              uint8_t* data = nullptr,      // data pointer
>+              uint32_t timestamp = 123u) {  // rtp timestamp
>     VCMPacket packet;
>     packet.codec = kVideoCodecGeneric;
>+    packet.timestamp = timestamp;
>     packet.seqNum = seq_num;
>     packet.frameType =
>         keyframe == kKeyFrame ? kVideoFrameKey : kVideoFrameDelta;
>@@ -195,6 +197,64 @@ TEST_F(TestPacketBuffer, FrameSize) {
>   EXPECT_EQ(20UL, frames_from_callback_.begin()->second->size());
> }
> 
>+TEST_F(TestPacketBuffer, CountsUniqueFrames) {
>+  const uint16_t seq_num = Rand();
>+
>+  ASSERT_EQ(0, packet_buffer_->GetUniqueFramesSeen());
>+
>+  EXPECT_TRUE(Insert(seq_num, kKeyFrame, kFirst, kNotLast, 0, nullptr, 100));
>+  ASSERT_EQ(1, packet_buffer_->GetUniqueFramesSeen());
>+  // Still the same frame.
>+  EXPECT_TRUE(
>+      Insert(seq_num + 1, kKeyFrame, kNotFirst, kLast, 0, nullptr, 100));
>+  ASSERT_EQ(1, packet_buffer_->GetUniqueFramesSeen());
>+
>+  // Second frame.
>+  EXPECT_TRUE(
>+      Insert(seq_num + 2, kKeyFrame, kFirst, kNotLast, 0, nullptr, 200));
>+  ASSERT_EQ(2, packet_buffer_->GetUniqueFramesSeen());
>+  EXPECT_TRUE(
>+      Insert(seq_num + 3, kKeyFrame, kNotFirst, kLast, 0, nullptr, 200));
>+  ASSERT_EQ(2, packet_buffer_->GetUniqueFramesSeen());
>+
>+  // Old packet.
>+  EXPECT_TRUE(
>+      Insert(seq_num + 1, kKeyFrame, kNotFirst, kLast, 0, nullptr, 100));
>+  ASSERT_EQ(2, packet_buffer_->GetUniqueFramesSeen());
>+
>+  // Missing middle packet.
>+  EXPECT_TRUE(
>+      Insert(seq_num + 4, kKeyFrame, kFirst, kNotLast, 0, nullptr, 300));
>+  EXPECT_TRUE(
>+      Insert(seq_num + 6, kKeyFrame, kNotFirst, kLast, 0, nullptr, 300));
>+  ASSERT_EQ(3, packet_buffer_->GetUniqueFramesSeen());
>+}
>+
>+TEST_F(TestPacketBuffer, HasHistoryOfUniqueFrames) {
>+  const int kNumFrames = 1500;
>+  const int kRequiredHistoryLength = 1000;
>+  const uint16_t seq_num = Rand();
>+  const uint32_t timestamp = 0xFFFFFFF0;  // Large enough to cause wrap-around.
>+
>+  for (int i = 0; i < kNumFrames; ++i) {
>+    EXPECT_TRUE(Insert(seq_num + i, kKeyFrame, kFirst, kNotLast, 0, nullptr,
>+                       timestamp + 10 * i));
>+  }
>+  ASSERT_EQ(kNumFrames, packet_buffer_->GetUniqueFramesSeen());
>+
>+  // Old packets within history should not affect number of seen unique frames.
>+  for (int i = kNumFrames - kRequiredHistoryLength; i < kNumFrames; ++i) {
>+    EXPECT_TRUE(Insert(seq_num + i, kKeyFrame, kFirst, kNotLast, 0, nullptr,
>+                       timestamp + 10 * i));
>+  }
>+  ASSERT_EQ(kNumFrames, packet_buffer_->GetUniqueFramesSeen());
>+
>+  // Very old packets should be treated as unique.
>+  EXPECT_TRUE(
>+      Insert(seq_num, kKeyFrame, kFirst, kNotLast, 0, nullptr, timestamp));
>+  ASSERT_EQ(kNumFrames + 1, packet_buffer_->GetUniqueFramesSeen());
>+}
>+
> TEST_F(TestPacketBuffer, ExpandBuffer) {
>   const uint16_t seq_num = Rand();
> 
>@@ -448,21 +508,19 @@ class TestPacketBufferH264 : public TestPacketBuffer {
>                   uint8_t* data = nullptr) {  // data pointer
>     VCMPacket packet;
>     packet.codec = kVideoCodecH264;
>+    auto& h264_header =
>+        packet.video_header.video_type_header.emplace<RTPVideoHeaderH264>();
>     packet.seqNum = seq_num;
>     packet.timestamp = timestamp;
>     if (keyframe == kKeyFrame) {
>       if (sps_pps_idr_is_keyframe_) {
>-        packet.video_header.codecHeader.H264.nalus[0].type =
>-            H264::NaluType::kSps;
>-        packet.video_header.codecHeader.H264.nalus[1].type =
>-            H264::NaluType::kPps;
>-        packet.video_header.codecHeader.H264.nalus[2].type =
>-            H264::NaluType::kIdr;
>-        packet.video_header.codecHeader.H264.nalus_length = 3;
>+        h264_header.nalus[0].type = H264::NaluType::kSps;
>+        h264_header.nalus[1].type = H264::NaluType::kPps;
>+        h264_header.nalus[2].type = H264::NaluType::kIdr;
>+        h264_header.nalus_length = 3;
>       } else {
>-        packet.video_header.codecHeader.H264.nalus[0].type =
>-            H264::NaluType::kIdr;
>-        packet.video_header.codecHeader.H264.nalus_length = 1;
>+        h264_header.nalus[0].type = H264::NaluType::kIdr;
>+        h264_header.nalus_length = 1;
>       }
>     }
>     packet.is_first_packet_in_frame = first == kFirst;
>@@ -536,12 +594,14 @@ TEST_P(TestPacketBufferH264Parameterized, GetBitstreamBufferPadding) {
>       new uint8_t[sizeof(data_data) + EncodedImage::kBufferPaddingBytesH264]);
> 
>   VCMPacket packet;
>-  packet.video_header.codecHeader.H264.nalus_length = 1;
>-  packet.video_header.codecHeader.H264.nalus[0].type = H264::NaluType::kIdr;
>+  auto& h264_header =
>+      packet.video_header.video_type_header.emplace<RTPVideoHeaderH264>();
>+  h264_header.nalus_length = 1;
>+  h264_header.nalus[0].type = H264::NaluType::kIdr;
>+  h264_header.packetization_type = kH264SingleNalu;
>   packet.seqNum = seq_num;
>   packet.codec = kVideoCodecH264;
>   packet.insertStartCode = true;
>-  packet.video_header.codecHeader.H264.packetization_type = kH264SingleNalu;
>   packet.dataPtr = data;
>   packet.sizeBytes = sizeof(data_data);
>   packet.is_first_packet_in_frame = true;
>@@ -651,8 +711,8 @@ TEST_F(TestPacketBuffer, ContinuousSeqNumDoubleMarkerBit) {
> }
> 
> TEST_F(TestPacketBuffer, PacketTimestamps) {
>-  rtc::Optional<int64_t> packet_ms;
>-  rtc::Optional<int64_t> packet_keyframe_ms;
>+  absl::optional<int64_t> packet_ms;
>+  absl::optional<int64_t> packet_keyframe_ms;
> 
>   packet_ms = packet_buffer_->LastReceivedPacketMs();
>   packet_keyframe_ms = packet_buffer_->LastReceivedKeyframePacketMs();
>@@ -685,6 +745,55 @@ TEST_F(TestPacketBuffer, PacketTimestamps) {
>   EXPECT_FALSE(packet_keyframe_ms);
> }
> 
>+TEST_F(TestPacketBuffer, IncomingCodecChange) {
>+  VCMPacket packet;
>+  packet.is_first_packet_in_frame = true;
>+  packet.markerBit = true;
>+  packet.sizeBytes = 0;
>+  packet.dataPtr = nullptr;
>+
>+  packet.codec = kVideoCodecVP8;
>+  packet.timestamp = 1;
>+  packet.seqNum = 1;
>+  packet.frameType = kVideoFrameKey;
>+  EXPECT_TRUE(packet_buffer_->InsertPacket(&packet));
>+
>+  packet.codec = kVideoCodecH264;
>+  auto& h264_header =
>+      packet.video_header.video_type_header.emplace<RTPVideoHeaderH264>();
>+  h264_header.nalus_length = 1;
>+  packet.timestamp = 3;
>+  packet.seqNum = 3;
>+  EXPECT_TRUE(packet_buffer_->InsertPacket(&packet));
>+
>+  packet.codec = kVideoCodecVP8;
>+  packet.timestamp = 2;
>+  packet.seqNum = 2;
>+  packet.frameType = kVideoFrameDelta;
>+
>+  EXPECT_TRUE(packet_buffer_->InsertPacket(&packet));
>+
>+  EXPECT_EQ(3UL, frames_from_callback_.size());
>+}
>+
>+TEST_F(TestPacketBuffer, TooManyNalusInPacket) {
>+  VCMPacket packet;
>+  packet.codec = kVideoCodecH264;
>+  packet.timestamp = 1;
>+  packet.seqNum = 1;
>+  packet.frameType = kVideoFrameKey;
>+  packet.is_first_packet_in_frame = true;
>+  packet.markerBit = true;
>+  auto& h264_header =
>+      packet.video_header.video_type_header.emplace<RTPVideoHeaderH264>();
>+  h264_header.nalus_length = kMaxNalusPerPacket;
>+  packet.sizeBytes = 0;
>+  packet.dataPtr = nullptr;
>+  EXPECT_TRUE(packet_buffer_->InsertPacket(&packet));
>+
>+  EXPECT_EQ(0UL, frames_from_callback_.size());
>+}
>+
> TEST_P(TestPacketBufferH264Parameterized, OneFrameFillBuffer) {
>   InsertH264(0, kKeyFrame, kFirst, kNotLast, 1000);
>   for (int i = 1; i < kStartSize - 1; ++i)
>@@ -772,9 +881,10 @@ class TestPacketBufferH264IdrIsKeyframe
> };
> 
> TEST_F(TestPacketBufferH264IdrIsKeyframe, IdrIsKeyframe) {
>-  packet_.video_header.codecHeader.H264.nalus[0].type = H264::NaluType::kIdr;
>-  packet_.video_header.codecHeader.H264.nalus_length = 1;
>-
>+  auto& h264_header =
>+      packet_.video_header.video_type_header.emplace<RTPVideoHeaderH264>();
>+  h264_header.nalus[0].type = H264::NaluType::kIdr;
>+  h264_header.nalus_length = 1;
>   packet_buffer_->InsertPacket(&packet_);
> 
>   ASSERT_EQ(1u, frames_from_callback_.size());
>@@ -782,10 +892,12 @@ TEST_F(TestPacketBufferH264IdrIsKeyframe, IdrIsKeyframe) {
> }
> 
> TEST_F(TestPacketBufferH264IdrIsKeyframe, SpsPpsIdrIsKeyframe) {
>-  packet_.video_header.codecHeader.H264.nalus[0].type = H264::NaluType::kSps;
>-  packet_.video_header.codecHeader.H264.nalus[1].type = H264::NaluType::kPps;
>-  packet_.video_header.codecHeader.H264.nalus[2].type = H264::NaluType::kIdr;
>-  packet_.video_header.codecHeader.H264.nalus_length = 3;
>+  auto& h264_header =
>+      packet_.video_header.video_type_header.emplace<RTPVideoHeaderH264>();
>+  h264_header.nalus[0].type = H264::NaluType::kSps;
>+  h264_header.nalus[1].type = H264::NaluType::kPps;
>+  h264_header.nalus[2].type = H264::NaluType::kIdr;
>+  h264_header.nalus_length = 3;
> 
>   packet_buffer_->InsertPacket(&packet_);
> 
>@@ -801,8 +913,10 @@ class TestPacketBufferH264SpsPpsIdrIsKeyframe
> };
> 
> TEST_F(TestPacketBufferH264SpsPpsIdrIsKeyframe, IdrIsNotKeyframe) {
>-  packet_.video_header.codecHeader.H264.nalus[0].type = H264::NaluType::kIdr;
>-  packet_.video_header.codecHeader.H264.nalus_length = 1;
>+  auto& h264_header =
>+      packet_.video_header.video_type_header.emplace<RTPVideoHeaderH264>();
>+  h264_header.nalus[0].type = H264::NaluType::kIdr;
>+  h264_header.nalus_length = 1;
> 
>   packet_buffer_->InsertPacket(&packet_);
> 
>@@ -811,9 +925,11 @@ TEST_F(TestPacketBufferH264SpsPpsIdrIsKeyframe, IdrIsNotKeyframe) {
> }
> 
> TEST_F(TestPacketBufferH264SpsPpsIdrIsKeyframe, SpsPpsIsNotKeyframe) {
>-  packet_.video_header.codecHeader.H264.nalus[0].type = H264::NaluType::kSps;
>-  packet_.video_header.codecHeader.H264.nalus[1].type = H264::NaluType::kPps;
>-  packet_.video_header.codecHeader.H264.nalus_length = 2;
>+  auto& h264_header =
>+      packet_.video_header.video_type_header.emplace<RTPVideoHeaderH264>();
>+  h264_header.nalus[0].type = H264::NaluType::kSps;
>+  h264_header.nalus[1].type = H264::NaluType::kPps;
>+  h264_header.nalus_length = 2;
> 
>   packet_buffer_->InsertPacket(&packet_);
> 
>@@ -822,10 +938,12 @@ TEST_F(TestPacketBufferH264SpsPpsIdrIsKeyframe, SpsPpsIsNotKeyframe) {
> }
> 
> TEST_F(TestPacketBufferH264SpsPpsIdrIsKeyframe, SpsPpsIdrIsKeyframe) {
>-  packet_.video_header.codecHeader.H264.nalus[0].type = H264::NaluType::kSps;
>-  packet_.video_header.codecHeader.H264.nalus[1].type = H264::NaluType::kPps;
>-  packet_.video_header.codecHeader.H264.nalus[2].type = H264::NaluType::kIdr;
>-  packet_.video_header.codecHeader.H264.nalus_length = 3;
>+  auto& h264_header =
>+      packet_.video_header.video_type_header.emplace<RTPVideoHeaderH264>();
>+  h264_header.nalus[0].type = H264::NaluType::kSps;
>+  h264_header.nalus[1].type = H264::NaluType::kPps;
>+  h264_header.nalus[2].type = H264::NaluType::kIdr;
>+  h264_header.nalus_length = 3;
> 
>   packet_buffer_->InsertPacket(&packet_);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_receiver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_receiver.cc
>index 112edda84f4..5baf430b5d9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_receiver.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_receiver.cc
>@@ -10,12 +10,14 @@
> 
> #include "common_types.h"  // NOLINT(build/include)
> #include "common_video/libyuv/include/webrtc_libyuv.h"
>+#include "modules/utility/include/process_thread.h"
> #include "modules/video_coding/encoded_frame.h"
> #include "modules/video_coding/include/video_codec_interface.h"
> #include "modules/video_coding/jitter_buffer.h"
> #include "modules/video_coding/packet.h"
> #include "modules/video_coding/video_coding_impl.h"
> #include "rtc_base/checks.h"
>+#include "rtc_base/location.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/trace_event.h"
> #include "system_wrappers/include/clock.h"
>@@ -40,26 +42,30 @@ VideoReceiver::VideoReceiver(Clock* clock,
>       _frameTypeCallback(nullptr),
>       _receiveStatsCallback(nullptr),
>       _packetRequestCallback(nullptr),
>-      _frameFromFile(),
>       _scheduleKeyRequest(false),
>       drop_frames_until_keyframe_(false),
>       max_nack_list_size_(0),
>-      _codecDataBase(nullptr),
>+      _codecDataBase(),
>       pre_decode_image_callback_(pre_decode_image_callback),
>       _receiveStatsTimer(1000, clock_),
>       _retransmissionTimer(10, clock_),
>-      _keyRequestTimer(500, clock_) {}
>+      _keyRequestTimer(500, clock_) {
>+  decoder_thread_checker_.DetachFromThread();
>+  module_thread_checker_.DetachFromThread();
>+}
> 
>-VideoReceiver::~VideoReceiver() {}
>+VideoReceiver::~VideoReceiver() {
>+  RTC_DCHECK_RUN_ON(&construction_thread_checker_);
>+}
> 
> void VideoReceiver::Process() {
>+  RTC_DCHECK_RUN_ON(&module_thread_checker_);
>   // Receive-side statistics
> 
>   // TODO(philipel): Remove this if block when we know what to do with
>   //                 ReceiveStatisticsProxy::QualitySample.
>   if (_receiveStatsTimer.TimeUntilProcess() == 0) {
>     _receiveStatsTimer.Processed();
>-    rtc::CritScope cs(&process_crit_);
>     if (_receiveStatsCallback != nullptr) {
>       _receiveStatsCallback->OnReceiveRatesUpdated(0, 0);
>     }
>@@ -68,10 +74,10 @@ void VideoReceiver::Process() {
>   // Key frame requests
>   if (_keyRequestTimer.TimeUntilProcess() == 0) {
>     _keyRequestTimer.Processed();
>-    bool request_key_frame = false;
>-    {
>+    bool request_key_frame = _frameTypeCallback != nullptr;
>+    if (request_key_frame) {
>       rtc::CritScope cs(&process_crit_);
>-      request_key_frame = _scheduleKeyRequest && _frameTypeCallback != nullptr;
>+      request_key_frame = _scheduleKeyRequest;
>     }
>     if (request_key_frame)
>       RequestKeyFrame();
>@@ -82,13 +88,8 @@ void VideoReceiver::Process() {
>   // disabled when NACK is off.
>   if (_retransmissionTimer.TimeUntilProcess() == 0) {
>     _retransmissionTimer.Processed();
>-    bool callback_registered = false;
>-    uint16_t length;
>-    {
>-      rtc::CritScope cs(&process_crit_);
>-      length = max_nack_list_size_;
>-      callback_registered = _packetRequestCallback != nullptr;
>-    }
>+    bool callback_registered = _packetRequestCallback != nullptr;
>+    uint16_t length = max_nack_list_size_;
>     if (callback_registered && length > 0) {
>       // Collect sequence numbers from the default receiver.
>       bool request_key_frame = false;
>@@ -107,7 +108,19 @@ void VideoReceiver::Process() {
>   }
> }
> 
>+void VideoReceiver::ProcessThreadAttached(ProcessThread* process_thread) {
>+  RTC_DCHECK_RUN_ON(&construction_thread_checker_);
>+  if (process_thread) {
>+    is_attached_to_process_thread_ = true;
>+    RTC_DCHECK(!process_thread_ || process_thread_ == process_thread);
>+    process_thread_ = process_thread;
>+  } else {
>+    is_attached_to_process_thread_ = false;
>+  }
>+}
>+
> int64_t VideoReceiver::TimeUntilNextProcess() {
>+  RTC_DCHECK_RUN_ON(&module_thread_checker_);
>   int64_t timeUntilNextProcess = _receiveStatsTimer.TimeUntilProcess();
>   if (_receiver.NackMode() != kNoNack) {
>     // We need a Process call more often if we are relying on
>@@ -122,7 +135,7 @@ int64_t VideoReceiver::TimeUntilNextProcess() {
> }
> 
> int32_t VideoReceiver::SetReceiveChannelParameters(int64_t rtt) {
>-  rtc::CritScope cs(&receive_crit_);
>+  RTC_DCHECK_RUN_ON(&module_thread_checker_);
>   _receiver.UpdateRtt(rtt);
>   return 0;
> }
>@@ -142,10 +155,8 @@ int32_t VideoReceiver::SetVideoProtection(VCMVideoProtection videoProtection,
>     }
> 
>     case kProtectionNackFEC: {
>-      rtc::CritScope cs(&receive_crit_);
>       RTC_DCHECK(enable);
>-      _receiver.SetNackMode(kNack,
>-                            media_optimization::kLowRttNackMs,
>+      _receiver.SetNackMode(kNack, media_optimization::kLowRttNackMs,
>                             media_optimization::kMaxRttDelayThreshold);
>       _receiver.SetDecodeErrorMode(kNoErrors);
>       break;
>@@ -165,20 +176,22 @@ int32_t VideoReceiver::SetVideoProtection(VCMVideoProtection videoProtection,
> // ready for rendering.
> int32_t VideoReceiver::RegisterReceiveCallback(
>     VCMReceiveCallback* receiveCallback) {
>-  RTC_DCHECK(construction_thread_.CalledOnValidThread());
>-  // TODO(tommi): Callback may be null, but only after the decoder thread has
>-  // been stopped. Use the signal we now get that tells us when the decoder
>-  // thread isn't running, to DCHECK that the method is never called while it
>-  // is. Once we're confident, we can remove the lock.
>-  rtc::CritScope cs(&receive_crit_);
>+  RTC_DCHECK_RUN_ON(&construction_thread_checker_);
>+  RTC_DCHECK(!IsDecoderThreadRunning());
>+  // This value is set before the decoder thread starts and unset after
>+  // the decoder thread has been stopped.
>   _decodedFrameCallback.SetUserReceiveCallback(receiveCallback);
>   return VCM_OK;
> }
> 
> int32_t VideoReceiver::RegisterReceiveStatisticsCallback(
>     VCMReceiveStatisticsCallback* receiveStats) {
>-  RTC_DCHECK(construction_thread_.CalledOnValidThread());
>-  rtc::CritScope cs(&process_crit_);
>+  RTC_DCHECK_RUN_ON(&construction_thread_checker_);
>+  RTC_DCHECK(!IsDecoderThreadRunning() && !is_attached_to_process_thread_);
>+  // |_receiver| is used on both the decoder and module threads.
>+  // However, since we make sure that we never do anything on the module thread
>+  // when the decoder thread is not running, we don't need a lock for the
>+  // |_receiver| or |_receiveStatsCallback| here.
>   _receiver.RegisterStatsCallback(receiveStats);
>   _receiveStatsCallback = receiveStats;
>   return VCM_OK;
>@@ -187,10 +200,8 @@ int32_t VideoReceiver::RegisterReceiveStatisticsCallback(
> // Register an externally defined decoder object.
> void VideoReceiver::RegisterExternalDecoder(VideoDecoder* externalDecoder,
>                                             uint8_t payloadType) {
>-  RTC_DCHECK(construction_thread_.CalledOnValidThread());
>-  // TODO(tommi): This method must be called when the decoder thread is not
>-  // running.  Do we need a lock in that case?
>-  rtc::CritScope cs(&receive_crit_);
>+  RTC_DCHECK_RUN_ON(&construction_thread_checker_);
>+  RTC_DCHECK(!IsDecoderThreadRunning());
>   if (externalDecoder == nullptr) {
>     RTC_CHECK(_codecDataBase.DeregisterExternalDecoder(payloadType));
>     return;
>@@ -201,53 +212,91 @@ void VideoReceiver::RegisterExternalDecoder(VideoDecoder* externalDecoder,
> // Register a frame type request callback.
> int32_t VideoReceiver::RegisterFrameTypeCallback(
>     VCMFrameTypeCallback* frameTypeCallback) {
>-  rtc::CritScope cs(&process_crit_);
>+  RTC_DCHECK_RUN_ON(&construction_thread_checker_);
>+  RTC_DCHECK(!IsDecoderThreadRunning() && !is_attached_to_process_thread_);
>+  // This callback is used on the module thread, but since we don't get
>+  // callbacks on the module thread while the decoder thread isn't running
>+  // (and this function must not be called when the decoder is running),
>+  // we don't need a lock here.
>   _frameTypeCallback = frameTypeCallback;
>   return VCM_OK;
> }
> 
> int32_t VideoReceiver::RegisterPacketRequestCallback(
>     VCMPacketRequestCallback* callback) {
>-  rtc::CritScope cs(&process_crit_);
>+  RTC_DCHECK_RUN_ON(&construction_thread_checker_);
>+  RTC_DCHECK(!IsDecoderThreadRunning() && !is_attached_to_process_thread_);
>+  // This callback is used on the module thread, but since we don't get
>+  // callbacks on the module thread while the decoder thread isn't running
>+  // (and this function must not be called when the decoder is running),
>+  // we don't need a lock here.
>   _packetRequestCallback = callback;
>   return VCM_OK;
> }
> 
> void VideoReceiver::TriggerDecoderShutdown() {
>-  RTC_DCHECK(construction_thread_.CalledOnValidThread());
>+  RTC_DCHECK_RUN_ON(&construction_thread_checker_);
>+  RTC_DCHECK(IsDecoderThreadRunning());
>   _receiver.TriggerDecoderShutdown();
> }
> 
>+void VideoReceiver::DecoderThreadStarting() {
>+  RTC_DCHECK_RUN_ON(&construction_thread_checker_);
>+  RTC_DCHECK(!IsDecoderThreadRunning());
>+  if (process_thread_ && !is_attached_to_process_thread_) {
>+    process_thread_->RegisterModule(this, RTC_FROM_HERE);
>+  }
>+#if RTC_DCHECK_IS_ON
>+  decoder_thread_is_running_ = true;
>+#endif
>+}
>+
>+void VideoReceiver::DecoderThreadStopped() {
>+  RTC_DCHECK_RUN_ON(&construction_thread_checker_);
>+  RTC_DCHECK(IsDecoderThreadRunning());
>+  if (process_thread_ && is_attached_to_process_thread_) {
>+    process_thread_->DeRegisterModule(this);
>+  }
>+#if RTC_DCHECK_IS_ON
>+  decoder_thread_is_running_ = false;
>+  decoder_thread_checker_.DetachFromThread();
>+#endif
>+}
>+
> // Decode next frame, blocking.
> // Should be called as often as possible to get the most out of the decoder.
> int32_t VideoReceiver::Decode(uint16_t maxWaitTimeMs) {
>-  bool prefer_late_decoding = false;
>-  {
>-    // TODO(tommi): Chances are that this lock isn't required.
>-    rtc::CritScope cs(&receive_crit_);
>-    prefer_late_decoding = _codecDataBase.PrefersLateDecoding();
>-  }
>-
>-  VCMEncodedFrame* frame =
>-      _receiver.FrameForDecoding(maxWaitTimeMs, prefer_late_decoding);
>+  RTC_DCHECK_RUN_ON(&decoder_thread_checker_);
>+  VCMEncodedFrame* frame = _receiver.FrameForDecoding(
>+      maxWaitTimeMs, _codecDataBase.PrefersLateDecoding());
> 
>   if (!frame)
>     return VCM_FRAME_NOT_READY;
> 
>+  bool drop_frame = false;
>   {
>     rtc::CritScope cs(&process_crit_);
>     if (drop_frames_until_keyframe_) {
>       // Still getting delta frames, schedule another keyframe request as if
>       // decode failed.
>       if (frame->FrameType() != kVideoFrameKey) {
>+        drop_frame = true;
>         _scheduleKeyRequest = true;
>-        _receiver.ReleaseFrame(frame);
>-        return VCM_FRAME_NOT_READY;
>+        // TODO(tommi): Consider if we could instead post a task to the module
>+        // thread and call RequestKeyFrame directly. Here we call WakeUp so that
>+        // TimeUntilNextProcess() gets called straight away.
>+        process_thread_->WakeUp(this);
>+      } else {
>+        drop_frames_until_keyframe_ = false;
>       }
>-      drop_frames_until_keyframe_ = false;
>     }
>   }
> 
>+  if (drop_frame) {
>+    _receiver.ReleaseFrame(frame);
>+    return VCM_FRAME_NOT_READY;
>+  }
>+
>   if (pre_decode_image_callback_) {
>     EncodedImage encoded_image(frame->EncodedImage());
>     int qp = -1;
>@@ -258,7 +307,6 @@ int32_t VideoReceiver::Decode(uint16_t maxWaitTimeMs) {
>                                                frame->CodecSpecific(), nullptr);
>   }
> 
>-  rtc::CritScope cs(&receive_crit_);
>   // If this frame was too late, we should adjust the delay accordingly
>   _timing->UpdateCurrentDelay(frame->RenderTimeMs(),
>                               clock_->TimeInMilliseconds());
>@@ -278,7 +326,7 @@ int32_t VideoReceiver::Decode(uint16_t maxWaitTimeMs) {
> // TODO(philipel): Clean up among the Decode functions as we replace
> //                 VCMEncodedFrame with FrameObject.
> int32_t VideoReceiver::Decode(const webrtc::VCMEncodedFrame* frame) {
>-  rtc::CritScope lock(&receive_crit_);
>+  RTC_DCHECK_RUN_ON(&decoder_thread_checker_);
>   if (pre_decode_image_callback_) {
>     EncodedImage encoded_image(frame->EncodedImage());
>     int qp = -1;
>@@ -291,19 +339,20 @@ int32_t VideoReceiver::Decode(const webrtc::VCMEncodedFrame* frame) {
>   return Decode(*frame);
> }
> 
>-void VideoReceiver::DecodingStopped() {
>-  // No further calls to Decode() will be made after this point.
>-  // TODO(tommi): Make use of this to clarify and check threading model.
>-}
>-
> int32_t VideoReceiver::RequestKeyFrame() {
>+  RTC_DCHECK_RUN_ON(&module_thread_checker_);
>+
>+  // Since we deregister from the module thread when the decoder thread isn't
>+  // running, we should get no calls here if decoding isn't being done.
>+  RTC_DCHECK(IsDecoderThreadRunning());
>+
>   TRACE_EVENT0("webrtc", "RequestKeyFrame");
>-  rtc::CritScope cs(&process_crit_);
>   if (_frameTypeCallback != nullptr) {
>     const int32_t ret = _frameTypeCallback->RequestKeyFrame();
>     if (ret < 0) {
>       return ret;
>     }
>+    rtc::CritScope cs(&process_crit_);
>     _scheduleKeyRequest = false;
>   } else {
>     return VCM_MISSING_CALLBACK;
>@@ -313,6 +362,7 @@ int32_t VideoReceiver::RequestKeyFrame() {
> 
> // Must be called from inside the receive side critical section.
> int32_t VideoReceiver::Decode(const VCMEncodedFrame& frame) {
>+  RTC_DCHECK_RUN_ON(&decoder_thread_checker_);
>   TRACE_EVENT0("webrtc", "VideoReceiver::Decode");
>   // Change decoder if payload type has changed
>   VCMGenericDecoder* decoder =
>@@ -327,11 +377,8 @@ int32_t VideoReceiver::Decode(const VCMEncodedFrame& frame) {
> int32_t VideoReceiver::RegisterReceiveCodec(const VideoCodec* receiveCodec,
>                                             int32_t numberOfCores,
>                                             bool requireKeyFrame) {
>-  RTC_DCHECK(construction_thread_.CalledOnValidThread());
>-  // TODO(tommi): This method must only be called when the decoder thread
>-  // is not running. Do we need a lock? If not, it looks like we might not need
>-  // a lock at all for |_codecDataBase|.
>-  rtc::CritScope cs(&receive_crit_);
>+  RTC_DCHECK_RUN_ON(&construction_thread_checker_);
>+  RTC_DCHECK(!IsDecoderThreadRunning());
>   if (receiveCodec == nullptr) {
>     return VCM_PARAMETER_ERROR;
>   }
>@@ -346,6 +393,7 @@ int32_t VideoReceiver::RegisterReceiveCodec(const VideoCodec* receiveCodec,
> int32_t VideoReceiver::IncomingPacket(const uint8_t* incomingPayload,
>                                       size_t payloadLength,
>                                       const WebRtcRTPHeader& rtpInfo) {
>+  RTC_DCHECK_RUN_ON(&module_thread_checker_);
>   if (rtpInfo.frameType == kVideoFrameKey) {
>     TRACE_EVENT1("webrtc", "VCM::PacketKeyFrame", "seqnum",
>                  rtpInfo.header.sequenceNumber);
>@@ -377,6 +425,7 @@ int32_t VideoReceiver::IncomingPacket(const uint8_t* incomingPayload,
> // to sync with audio. Not included in  VideoCodingModule::Delay()
> // Defaults to 0 ms.
> int32_t VideoReceiver::SetMinimumPlayoutDelay(uint32_t minPlayoutDelayMs) {
>+  RTC_DCHECK_RUN_ON(&module_thread_checker_);
>   _timing->set_min_playout_delay(minPlayoutDelayMs);
>   return VCM_OK;
> }
>@@ -384,22 +433,24 @@ int32_t VideoReceiver::SetMinimumPlayoutDelay(uint32_t minPlayoutDelayMs) {
> // The estimated delay caused by rendering, defaults to
> // kDefaultRenderDelayMs = 10 ms
> int32_t VideoReceiver::SetRenderDelay(uint32_t timeMS) {
>+  RTC_DCHECK_RUN_ON(&construction_thread_checker_);
>+  RTC_DCHECK(!IsDecoderThreadRunning());
>   _timing->set_render_delay(timeMS);
>   return VCM_OK;
> }
> 
> // Current video delay
> int32_t VideoReceiver::Delay() const {
>+  RTC_DCHECK_RUN_ON(&module_thread_checker_);
>   return _timing->TargetVideoDelay();
> }
> 
>+// Only used by VCMRobustnessTest.
> int VideoReceiver::SetReceiverRobustnessMode(
>     VideoCodingModule::ReceiverRobustness robustnessMode,
>     VCMDecodeErrorMode decode_error_mode) {
>-  RTC_DCHECK(construction_thread_.CalledOnValidThread());
>-  // TODO(tommi): This method must only be called when the decoder thread
>-  // is not running and we don't need to hold this lock.
>-  rtc::CritScope cs(&receive_crit_);
>+  RTC_DCHECK_RUN_ON(&construction_thread_checker_);
>+  RTC_DCHECK(!IsDecoderThreadRunning());
>   switch (robustnessMode) {
>     case VideoCodingModule::kNone:
>       _receiver.SetNackMode(kNoNack, -1, -1);
>@@ -417,23 +468,29 @@ int VideoReceiver::SetReceiverRobustnessMode(
> }
> 
> void VideoReceiver::SetDecodeErrorMode(VCMDecodeErrorMode decode_error_mode) {
>-  rtc::CritScope cs(&receive_crit_);
>+  RTC_DCHECK_RUN_ON(&construction_thread_checker_);
>+  RTC_DCHECK(!IsDecoderThreadRunning());
>   _receiver.SetDecodeErrorMode(decode_error_mode);
> }
> 
> void VideoReceiver::SetNackSettings(size_t max_nack_list_size,
>                                     int max_packet_age_to_nack,
>                                     int max_incomplete_time_ms) {
>+  RTC_DCHECK_RUN_ON(&construction_thread_checker_);
>+  RTC_DCHECK(!IsDecoderThreadRunning());
>   if (max_nack_list_size != 0) {
>-    rtc::CritScope cs(&process_crit_);
>     max_nack_list_size_ = max_nack_list_size;
>   }
>   _receiver.SetNackSettings(max_nack_list_size, max_packet_age_to_nack,
>                             max_incomplete_time_ms);
> }
> 
>-int VideoReceiver::SetMinReceiverDelay(int desired_delay_ms) {
>-  return _receiver.SetMinReceiverDelay(desired_delay_ms);
>+bool VideoReceiver::IsDecoderThreadRunning() {
>+#if RTC_DCHECK_IS_ON
>+  return decoder_thread_is_running_;
>+#else
>+  return true;
>+#endif
> }
> 
> }  // namespace vcm
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_receiver_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_receiver_unittest.cc
>index 917cdc11134..bec1e96bb83 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_receiver_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_receiver_unittest.cc
>@@ -65,7 +65,7 @@ class TestVideoReceiver : public ::testing::Test {
>       ++header->header.sequenceNumber;
>     }
>     receiver_->Process();
>-    EXPECT_CALL(decoder_, Decode(_, _, _, _, _)).Times(0);
>+    EXPECT_CALL(decoder_, Decode(_, _, _, _)).Times(0);
>     EXPECT_EQ(VCM_FRAME_NOT_READY, receiver_->Decode(100));
>   }
> 
>@@ -76,8 +76,8 @@ class TestVideoReceiver : public ::testing::Test {
>     EXPECT_EQ(0, receiver_->IncomingPacket(payload, length, *header));
>     ++header->header.sequenceNumber;
>     EXPECT_CALL(packet_request_callback_, ResendPackets(_, _)).Times(0);
>-    receiver_->Process();;
>-    EXPECT_CALL(decoder_, Decode(_, _, _, _, _)).Times(1);
>+    receiver_->Process();
>+    EXPECT_CALL(decoder_, Decode(_, _, _, _)).Times(1);
>     EXPECT_EQ(0, receiver_->Decode(100));
>   }
> 
>@@ -98,15 +98,14 @@ TEST_F(TestVideoReceiver, PaddingOnlyFrames) {
>       0, receiver_->RegisterPacketRequestCallback(&packet_request_callback_));
>   const size_t kPaddingSize = 220;
>   const uint8_t payload[kPaddingSize] = {0};
>-  WebRtcRTPHeader header;
>-  memset(&header, 0, sizeof(header));
>+  WebRtcRTPHeader header = {};
>   header.frameType = kEmptyFrame;
>   header.header.markerBit = false;
>   header.header.paddingLength = kPaddingSize;
>   header.header.payloadType = kUnusedPayloadType;
>   header.header.ssrc = 1;
>   header.header.headerLength = 12;
>-  header.type.Video.codec = kRtpVideoVp8;
>+  header.video_header().codec = kVideoCodecVP8;
>   for (int i = 0; i < 10; ++i) {
>     EXPECT_CALL(packet_request_callback_, ResendPackets(_, _)).Times(0);
>     InsertAndVerifyPaddingFrame(payload, &header);
>@@ -122,25 +121,24 @@ TEST_F(TestVideoReceiver, PaddingOnlyFramesWithLosses) {
>   const size_t kFrameSize = 1200;
>   const size_t kPaddingSize = 220;
>   const uint8_t payload[kFrameSize] = {0};
>-  WebRtcRTPHeader header;
>-  memset(&header, 0, sizeof(header));
>+  WebRtcRTPHeader header = {};
>   header.frameType = kEmptyFrame;
>   header.header.markerBit = false;
>   header.header.paddingLength = kPaddingSize;
>   header.header.payloadType = kUnusedPayloadType;
>   header.header.ssrc = 1;
>   header.header.headerLength = 12;
>-  header.type.Video.codec = kRtpVideoVp8;
>+  header.video_header().codec = kVideoCodecVP8;
>   // Insert one video frame to get one frame decoded.
>   header.frameType = kVideoFrameKey;
>-  header.type.Video.is_first_packet_in_frame = true;
>+  header.video_header().is_first_packet_in_frame = true;
>   header.header.markerBit = true;
>   InsertAndVerifyDecodableFrame(payload, kFrameSize, &header);
>   clock_.AdvanceTimeMilliseconds(33);
>   header.header.timestamp += 3000;
> 
>   header.frameType = kEmptyFrame;
>-  header.type.Video.is_first_packet_in_frame = false;
>+  header.video_header().is_first_packet_in_frame = false;
>   header.header.markerBit = false;
>   // Insert padding frames.
>   for (int i = 0; i < 10; ++i) {
>@@ -173,18 +171,17 @@ TEST_F(TestVideoReceiver, PaddingOnlyAndVideo) {
>   const size_t kFrameSize = 1200;
>   const size_t kPaddingSize = 220;
>   const uint8_t payload[kFrameSize] = {0};
>-  WebRtcRTPHeader header;
>-  memset(&header, 0, sizeof(header));
>+  WebRtcRTPHeader header = {};
>   header.frameType = kEmptyFrame;
>-  header.type.Video.is_first_packet_in_frame = false;
>+  header.video_header().is_first_packet_in_frame = false;
>   header.header.markerBit = false;
>   header.header.paddingLength = kPaddingSize;
>   header.header.payloadType = kUnusedPayloadType;
>   header.header.ssrc = 1;
>   header.header.headerLength = 12;
>-  header.type.Video.codec = kRtpVideoVp8;
>-  header.type.Video.codecHeader.VP8.pictureId = -1;
>-  header.type.Video.codecHeader.VP8.tl0PicIdx = -1;
>+  header.video_header().codec = kVideoCodecVP8;
>+  header.video_header().vp8().pictureId = -1;
>+  header.video_header().vp8().tl0PicIdx = -1;
>   for (int i = 0; i < 3; ++i) {
>     // Insert 2 video frames.
>     for (int j = 0; j < 2; ++j) {
>@@ -192,7 +189,7 @@ TEST_F(TestVideoReceiver, PaddingOnlyAndVideo) {
>         header.frameType = kVideoFrameKey;
>       else
>         header.frameType = kVideoFrameDelta;
>-      header.type.Video.is_first_packet_in_frame = true;
>+      header.video_header().is_first_packet_in_frame = true;
>       header.header.markerBit = true;
>       InsertAndVerifyDecodableFrame(payload, kFrameSize, &header);
>       clock_.AdvanceTimeMilliseconds(33);
>@@ -201,7 +198,7 @@ TEST_F(TestVideoReceiver, PaddingOnlyAndVideo) {
> 
>     // Insert 2 padding only frames.
>     header.frameType = kEmptyFrame;
>-    header.type.Video.is_first_packet_in_frame = false;
>+    header.video_header().is_first_packet_in_frame = false;
>     header.header.markerBit = false;
>     for (int j = 0; j < 2; ++j) {
>       // InsertAndVerifyPaddingFrame(payload, &header);
>@@ -211,13 +208,6 @@ TEST_F(TestVideoReceiver, PaddingOnlyAndVideo) {
>   }
> }
> 
>-TEST_F(TestVideoReceiver, ReceiverDelay) {
>-  EXPECT_EQ(0, receiver_->SetMinReceiverDelay(0));
>-  EXPECT_EQ(0, receiver_->SetMinReceiverDelay(5000));
>-  EXPECT_EQ(-1, receiver_->SetMinReceiverDelay(-100));
>-  EXPECT_EQ(-1, receiver_->SetMinReceiverDelay(10010));
>-}
>-
> }  // namespace
> }  // namespace vcm
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_sender.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_sender.cc
>index 4d211655124..bae60edf622 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_sender.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_sender.cc
>@@ -11,10 +11,9 @@
> 
> #include <algorithm>  // std::max
> 
>+#include "api/video/video_bitrate_allocator.h"
> #include "common_types.h"  // NOLINT(build/include)
>-#include "common_video/include/video_bitrate_allocator.h"
> #include "common_video/libyuv/include/webrtc_libyuv.h"
>-#include "modules/video_coding/codecs/vp8/temporal_layers.h"
> #include "modules/video_coding/encoded_frame.h"
> #include "modules/video_coding/include/video_codec_interface.h"
> #include "modules/video_coding/utility/default_video_bitrate_allocator.h"
>@@ -36,7 +35,7 @@ VideoSender::VideoSender(Clock* clock,
>       _codecDataBase(&_encodedFrameCallback),
>       frame_dropper_enabled_(true),
>       current_codec_(),
>-      encoder_params_({BitrateAllocation(), 0, 0, 0}),
>+      encoder_params_({VideoBitrateAllocation(), 0, 0, 0}),
>       encoder_has_internal_source_(false),
>       next_frame_types_(1, kVideoFrameDelta) {
>   _mediaOpt.Reset();
>@@ -69,8 +68,8 @@ int32_t VideoSender::RegisterSendCodec(const VideoCodec* sendCodec,
>   current_codec_ = *sendCodec;
> 
>   if (!ret) {
>-    RTC_LOG(LS_ERROR) << "Failed to initialize set encoder with payload name '"
>-                      << sendCodec->plName << "'.";
>+    RTC_LOG(LS_ERROR) << "Failed to initialize set encoder with codec type '"
>+                      << sendCodec->codecType << "'.";
>     return VCM_CODEC_ERROR;
>   }
> 
>@@ -93,7 +92,7 @@ int32_t VideoSender::RegisterSendCodec(const VideoCodec* sendCodec,
> 
>   // If we have screensharing and we have layers, we disable frame dropper.
>   bool disable_frame_dropper =
>-      numLayers > 1 && sendCodec->mode == kScreensharing;
>+      numLayers > 1 && sendCodec->mode == VideoCodecMode::kScreensharing;
>   if (disable_frame_dropper) {
>     _mediaOpt.EnableFrameDropper(false);
>   } else if (frame_dropper_enabled_) {
>@@ -123,17 +122,14 @@ int32_t VideoSender::RegisterSendCodec(const VideoCodec* sendCodec,
> // Register an external decoder object.
> // This can not be used together with external decoder callbacks.
> void VideoSender::RegisterExternalEncoder(VideoEncoder* externalEncoder,
>-                                          uint8_t payloadType,
>                                           bool internalSource /*= false*/) {
>   RTC_DCHECK(sequenced_checker_.CalledSequentially());
> 
>   rtc::CritScope lock(&encoder_crit_);
> 
>   if (externalEncoder == nullptr) {
>-    bool wasSendCodec = false;
>-    RTC_CHECK(
>-        _codecDataBase.DeregisterExternalEncoder(payloadType, &wasSendCodec));
>-    if (wasSendCodec) {
>+    _codecDataBase.DeregisterExternalEncoder();
>+    {
>       // Make sure the VCM doesn't use the de-registered codec
>       rtc::CritScope params_lock(&params_crit_);
>       _encoder = nullptr;
>@@ -141,35 +137,10 @@ void VideoSender::RegisterExternalEncoder(VideoEncoder* externalEncoder,
>     }
>     return;
>   }
>-  _codecDataBase.RegisterExternalEncoder(externalEncoder, payloadType,
>+  _codecDataBase.RegisterExternalEncoder(externalEncoder,
>                                          internalSource);
> }
> 
>-// Get encode bitrate
>-int VideoSender::Bitrate(unsigned int* bitrate) const {
>-  RTC_DCHECK(sequenced_checker_.CalledSequentially());
>-  // Since we're running on the thread that's the only thread known to modify
>-  // the value of _encoder, we don't need to grab the lock here.
>-
>-  if (!_encoder)
>-    return VCM_UNINITIALIZED;
>-  *bitrate = _encoder->GetEncoderParameters().target_bitrate.get_sum_bps();
>-  return 0;
>-}
>-
>-// Get encode frame rate
>-int VideoSender::FrameRate(unsigned int* framerate) const {
>-  RTC_DCHECK(sequenced_checker_.CalledSequentially());
>-  // Since we're running on the thread that's the only thread known to modify
>-  // the value of _encoder, we don't need to grab the lock here.
>-
>-  if (!_encoder)
>-    return VCM_UNINITIALIZED;
>-
>-  *framerate = _encoder->GetEncoderParameters().input_frame_rate;
>-  return 0;
>-}
>-
> EncoderParameters VideoSender::UpdateEncoderParameters(
>     const EncoderParameters& params,
>     VideoBitrateAllocator* bitrate_allocator,
>@@ -179,7 +150,7 @@ EncoderParameters VideoSender::UpdateEncoderParameters(
>   if (input_frame_rate == 0)
>     input_frame_rate = current_codec_.maxFramerate;
> 
>-  BitrateAllocation bitrate_allocation;
>+  VideoBitrateAllocation bitrate_allocation;
>   // Only call allocators if bitrate > 0 (ie, not suspended), otherwise they
>   // might cap the bitrate to the min bitrate configured.
>   if (target_bitrate_bps > 0) {
>@@ -197,10 +168,10 @@ EncoderParameters VideoSender::UpdateEncoderParameters(
>   return new_encoder_params;
> }
> 
>-void VideoSender::UpdateChannelParemeters(
>+void VideoSender::UpdateChannelParameters(
>     VideoBitrateAllocator* bitrate_allocator,
>     VideoBitrateAllocationObserver* bitrate_updated_callback) {
>-  BitrateAllocation target_rate;
>+  VideoBitrateAllocation target_rate;
>   {
>     rtc::CritScope cs(&params_crit_);
>     encoder_params_ =
>@@ -268,17 +239,6 @@ void VideoSender::SetEncoderParameters(EncoderParameters params,
>     _encoder->SetEncoderParameters(params);
> }
> 
>-// Deprecated:
>-// TODO(perkj): Remove once no projects call this method. It currently do
>-// nothing.
>-int32_t VideoSender::RegisterProtectionCallback(
>-    VCMProtectionCallback* protection_callback) {
>-  // Deprecated:
>-  // TODO(perkj): Remove once no projects call this method. It currently do
>-  // nothing.
>-  return VCM_OK;
>-}
>-
> // Add one raw video frame to the encoder, blocking.
> int32_t VideoSender::AddVideoFrame(const VideoFrame& videoFrame,
>                                    const CodecSpecificInfo* codecSpecificInfo) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_sender_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_sender_unittest.cc
>index 0719b1677a1..c391510ec13 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_sender_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_sender_unittest.cc
>@@ -13,14 +13,13 @@
> 
> #include "api/video/i420_buffer.h"
> #include "modules/video_coding/codecs/vp8/include/vp8.h"
>-#include "modules/video_coding/codecs/vp8/include/vp8_common_types.h"
>-#include "modules/video_coding/codecs/vp8/simulcast_rate_allocator.h"
> #include "modules/video_coding/codecs/vp8/temporal_layers.h"
> #include "modules/video_coding/include/mock/mock_vcm_callbacks.h"
> #include "modules/video_coding/include/mock/mock_video_codec_interface.h"
> #include "modules/video_coding/include/video_coding.h"
>-#include "modules/video_coding/video_coding_impl.h"
> #include "modules/video_coding/utility/default_video_bitrate_allocator.h"
>+#include "modules/video_coding/utility/simulcast_rate_allocator.h"
>+#include "modules/video_coding/video_coding_impl.h"
> #include "system_wrappers/include/clock.h"
> #include "test/frame_generator.h"
> #include "test/gtest.h"
>@@ -206,7 +205,7 @@ class TestVideoSenderWithMockEncoder : public TestVideoSender {
>  protected:
>   void SetUp() override {
>     TestVideoSender::SetUp();
>-    sender_->RegisterExternalEncoder(&encoder_, kUnusedPayloadType, false);
>+    sender_->RegisterExternalEncoder(&encoder_, false);
>     webrtc::test::CodecSettings(kVideoCodecVP8, &settings_);
>     settings_.numberOfSimulcastStreams = kNumberOfStreams;
>     ConfigureStream(kDefaultWidth / 4, kDefaultHeight / 4, 100,
>@@ -228,17 +227,16 @@ class TestVideoSenderWithMockEncoder : public TestVideoSender {
>     ExpectEncodeWithFrameTypes(stream, false);
>   }
> 
>-  void ExpectInitialKeyFrames() {
>-    ExpectEncodeWithFrameTypes(-1, true);
>-  }
>+  void ExpectInitialKeyFrames() { ExpectEncodeWithFrameTypes(-1, true); }
> 
>   void ExpectEncodeWithFrameTypes(int intra_request_stream, bool first_frame) {
>     if (intra_request_stream == -1) {
>       // No intra request expected, keyframes on first frame.
>       FrameType frame_type = first_frame ? kVideoFrameKey : kVideoFrameDelta;
>-      EXPECT_CALL(encoder_,
>-                  Encode(_, _, Pointee(ElementsAre(frame_type, frame_type,
>-                                                   frame_type))))
>+      EXPECT_CALL(
>+          encoder_,
>+          Encode(_, _,
>+                 Pointee(ElementsAre(frame_type, frame_type, frame_type))))
>           .Times(1)
>           .WillRepeatedly(Return(0));
>       return;
>@@ -248,9 +246,10 @@ class TestVideoSenderWithMockEncoder : public TestVideoSender {
>     ASSERT_LT(intra_request_stream, kNumberOfStreams);
>     std::vector<FrameType> frame_types(kNumberOfStreams, kVideoFrameDelta);
>     frame_types[intra_request_stream] = kVideoFrameKey;
>-    EXPECT_CALL(encoder_,
>-                Encode(_, _, Pointee(ElementsAreArray(&frame_types[0],
>-                                                      frame_types.size()))))
>+    EXPECT_CALL(
>+        encoder_,
>+        Encode(_, _,
>+               Pointee(ElementsAreArray(&frame_types[0], frame_types.size()))))
>         .Times(1)
>         .WillRepeatedly(Return(0));
>   }
>@@ -306,7 +305,7 @@ TEST_F(TestVideoSenderWithMockEncoder, TestSetRate) {
>   const uint32_t new_bitrate_kbps = settings_.startBitrate + 300;
> 
>   // Initial frame rate is taken from config, as we have no data yet.
>-  BitrateAllocation new_rate_allocation = rate_allocator_->GetAllocation(
>+  VideoBitrateAllocation new_rate_allocation = rate_allocator_->GetAllocation(
>       new_bitrate_kbps * 1000, settings_.maxFramerate);
>   EXPECT_CALL(encoder_,
>               SetRateAllocation(new_rate_allocation, settings_.maxFramerate))
>@@ -325,9 +324,9 @@ TEST_F(TestVideoSenderWithMockEncoder, TestSetRate) {
> 
> TEST_F(TestVideoSenderWithMockEncoder, TestIntraRequestsInternalCapture) {
>   // De-register current external encoder.
>-  sender_->RegisterExternalEncoder(nullptr, kUnusedPayloadType, false);
>+  sender_->RegisterExternalEncoder(nullptr, false);
>   // Register encoder with internal capture.
>-  sender_->RegisterExternalEncoder(&encoder_, kUnusedPayloadType, true);
>+  sender_->RegisterExternalEncoder(&encoder_, true);
>   EXPECT_EQ(0, sender_->RegisterSendCodec(&settings_, 1, 1200));
>   // Initial request should be all keyframes.
>   ExpectInitialKeyFrames();
>@@ -344,14 +343,14 @@ TEST_F(TestVideoSenderWithMockEncoder, TestIntraRequestsInternalCapture) {
> 
> TEST_F(TestVideoSenderWithMockEncoder, TestEncoderParametersForInternalSource) {
>   // De-register current external encoder.
>-  sender_->RegisterExternalEncoder(nullptr, kUnusedPayloadType, false);
>+  sender_->RegisterExternalEncoder(nullptr, false);
>   // Register encoder with internal capture.
>-  sender_->RegisterExternalEncoder(&encoder_, kUnusedPayloadType, true);
>+  sender_->RegisterExternalEncoder(&encoder_, true);
>   EXPECT_EQ(0, sender_->RegisterSendCodec(&settings_, 1, 1200));
>   // Update encoder bitrate parameters. We expect that to immediately call
>   // SetRates on the encoder without waiting for AddFrame processing.
>   const uint32_t new_bitrate_kbps = settings_.startBitrate + 300;
>-  BitrateAllocation new_rate_allocation = rate_allocator_->GetAllocation(
>+  VideoBitrateAllocation new_rate_allocation = rate_allocator_->GetAllocation(
>       new_bitrate_kbps * 1000, settings_.maxFramerate);
>   EXPECT_CALL(encoder_, SetRateAllocation(new_rate_allocation, _))
>       .Times(1)
>@@ -383,7 +382,7 @@ TEST_F(TestVideoSenderWithMockEncoder,
>   // Call to SetChannelParameters with changed bitrate should call encoder
>   // SetRates but not encoder SetChannelParameters (that are unchanged).
>   uint32_t new_bitrate_bps = 2 * settings_.startBitrate * 1000;
>-  BitrateAllocation new_rate_allocation =
>+  VideoBitrateAllocation new_rate_allocation =
>       rate_allocator_->GetAllocation(new_bitrate_bps, kInputFps);
>   EXPECT_CALL(encoder_, SetRateAllocation(new_rate_allocation, kInputFps))
>       .Times(1)
>@@ -413,13 +412,10 @@ class TestVideoSenderWithVp8 : public TestVideoSender {
>     codec_.startBitrate = codec_bitrate_kbps_;
>     codec_.maxBitrate = codec_bitrate_kbps_;
> 
>-    TemporalLayersFactory* tl_factory = new TemporalLayersFactory();
>-    rate_allocator_.reset(new SimulcastRateAllocator(
>-        codec_, std::unique_ptr<TemporalLayersFactory>(tl_factory)));
>-    codec_.VP8()->tl_factory = tl_factory;
>+    rate_allocator_.reset(new SimulcastRateAllocator(codec_));
> 
>     encoder_ = VP8Encoder::Create();
>-    sender_->RegisterExternalEncoder(encoder_.get(), codec_.plType, false);
>+    sender_->RegisterExternalEncoder(encoder_.get(), false);
>     EXPECT_EQ(0, sender_->RegisterSendCodec(&codec_, 1, 1200));
>   }
> 
>@@ -475,9 +471,15 @@ class TestVideoSenderWithVp8 : public TestVideoSender {
> #define MAYBE_FixedTemporalLayersStrategy FixedTemporalLayersStrategy
> #endif
> TEST_F(TestVideoSenderWithVp8, MAYBE_FixedTemporalLayersStrategy) {
>-  const int low_b = codec_bitrate_kbps_ * kVp8LayerRateAlloction[2][0];
>-  const int mid_b = codec_bitrate_kbps_ * kVp8LayerRateAlloction[2][1];
>-  const int high_b = codec_bitrate_kbps_ * kVp8LayerRateAlloction[2][2];
>+  const int low_b =
>+      codec_bitrate_kbps_ *
>+      webrtc::SimulcastRateAllocator::GetTemporalRateAllocation(3, 0);
>+  const int mid_b =
>+      codec_bitrate_kbps_ *
>+      webrtc::SimulcastRateAllocator::GetTemporalRateAllocation(3, 1);
>+  const int high_b =
>+      codec_bitrate_kbps_ *
>+      webrtc::SimulcastRateAllocator::GetTemporalRateAllocation(3, 2);
>   {
>     Vp8StreamInfo expected = {{7.5, 15.0, 30.0}, {low_b, mid_b, high_b}};
>     EXPECT_THAT(SimulateWithFramerate(30.0), MatchesVp8StreamInfo(expected));
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_processing/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_processing/BUILD.gn
>index 351e5792f8e..d3724994e63 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_processing/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_processing/BUILD.gn
>@@ -6,7 +6,6 @@
> # in the file PATENTS.  All contributing project authors may
> # be found in the AUTHORS file in the root of the source tree.
> 
>-import("//build/config/arm.gni")
> import("../../webrtc.gni")
> 
> build_video_processing_sse2 = current_cpu == "x86" || current_cpu == "x64"
>@@ -33,6 +32,7 @@ rtc_static_library("video_processing") {
>     "../../modules/utility",
>     "../../rtc_base:checks",
>     "../../rtc_base:rtc_base_approved",
>+    "../../rtc_base/system:arch",
>     "../../system_wrappers:cpu_features_api",
>     "//third_party/libyuv",
>   ]
>@@ -78,7 +78,7 @@ if (build_video_processing_sse2) {
>       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>     }
> 
>-    if (is_posix) {
>+    if (is_posix || is_fuchsia) {
>       cflags = [ "-msse2" ]
>     }
>   }
>@@ -112,6 +112,7 @@ if (rtc_include_tests) {
>     deps = [
>       ":video_processing",
>       "../../common_video:common_video",
>+      "../../test:fileutils",
>       "../../test:test_support",
>       "../../test:video_test_common",
>     ]
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_processing/test/denoiser_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_processing/test/denoiser_test.cc
>index 601ae0ed4a6..327b16ed40e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_processing/test/denoiser_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_processing/test/denoiser_test.cc
>@@ -14,8 +14,8 @@
> 
> #include "common_video/include/i420_buffer_pool.h"
> #include "modules/video_processing/video_denoiser.h"
>-#include "test/gtest.h"
> #include "test/frame_utils.h"
>+#include "test/gtest.h"
> #include "test/testsupport/fileutils.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_processing/util/denoiser_filter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_processing/util/denoiser_filter.cc
>index cf666332a48..a88865b5090 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_processing/util/denoiser_filter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_processing/util/denoiser_filter.cc
>@@ -13,6 +13,7 @@
> #include "modules/video_processing/util/denoiser_filter_neon.h"
> #include "modules/video_processing/util/denoiser_filter_sse2.h"
> #include "rtc_base/checks.h"
>+#include "rtc_base/system/arch.h"
> #include "system_wrappers/include/cpu_features_wrapper.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_processing/util/denoiser_filter_sse2.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_processing/util/denoiser_filter_sse2.cc
>index 83b7e54feb9..49d2548854c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_processing/util/denoiser_filter_sse2.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_processing/util/denoiser_filter_sse2.cc
>@@ -8,8 +8,8 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include <emmintrin.h>
> #include "modules/video_processing/util/denoiser_filter_sse2.h"
>+#include <emmintrin.h>
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_processing/util/skin_detection.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_processing/util/skin_detection.cc
>index 8eaa2c5a9c1..c711a90385a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_processing/util/skin_detection.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_processing/util/skin_detection.cc
>@@ -16,11 +16,14 @@
> namespace webrtc {
> 
> // Fixed-point skin color model parameters.
>-static const int skin_mean[5][2] = {
>-    {7463, 9614}, {6400, 10240}, {7040, 10240}, {8320, 9280}, {6800, 9614}};
>+static const int skin_mean[5][2] = {{7463, 9614},
>+                                    {6400, 10240},
>+                                    {7040, 10240},
>+                                    {8320, 9280},
>+                                    {6800, 9614}};
> static const int skin_inv_cov[4] = {4107, 1663, 1663, 2157};  // q16
>-static const int skin_threshold[6] = {1570636, 1400000, 800000, 800000, 800000,
>-    800000};  // q18
>+static const int skin_threshold[6] = {1570636, 1400000, 800000,
>+                                      800000,  800000,  800000};  // q18
> 
> // Thresholds on luminance.
> static const int y_low = 40;
>@@ -39,10 +42,9 @@ static int EvaluateSkinColorDifference(int cb, int cr, int idx) {
>   const int cb_diff_q2 = (cb_diff_q12 + (1 << 9)) >> 10;
>   const int cbcr_diff_q2 = (cbcr_diff_q12 + (1 << 9)) >> 10;
>   const int cr_diff_q2 = (cr_diff_q12 + (1 << 9)) >> 10;
>-  const int skin_diff = skin_inv_cov[0] * cb_diff_q2 +
>-      skin_inv_cov[1] * cbcr_diff_q2 +
>-      skin_inv_cov[2] * cbcr_diff_q2 +
>-      skin_inv_cov[3] * cr_diff_q2;
>+  const int skin_diff =
>+      skin_inv_cov[0] * cb_diff_q2 + skin_inv_cov[1] * cbcr_diff_q2 +
>+      skin_inv_cov[2] * cbcr_diff_q2 + skin_inv_cov[3] * cr_diff_q2;
>   return skin_diff;
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_processing/video_denoiser.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_processing/video_denoiser.cc
>index a87f2496c8c..cf262f01261 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_processing/video_denoiser.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_processing/video_denoiser.cc
>@@ -31,10 +31,14 @@ static void ShowRect(const std::unique_ptr<DenoiserFilter>& filter,
>                      const std::unique_ptr<uint8_t[]>& moving_edge_red,
>                      const std::unique_ptr<uint8_t[]>& x_density,
>                      const std::unique_ptr<uint8_t[]>& y_density,
>-                     const uint8_t* u_src, int stride_u_src,
>-                     const uint8_t* v_src, int stride_v_src,
>-                     uint8_t* u_dst, int stride_u_dst,
>-                     uint8_t* v_dst, int stride_v_dst,
>+                     const uint8_t* u_src,
>+                     int stride_u_src,
>+                     const uint8_t* v_src,
>+                     int stride_v_src,
>+                     uint8_t* u_dst,
>+                     int stride_u_dst,
>+                     uint8_t* v_dst,
>+                     int stride_v_dst,
>                      int mb_rows_,
>                      int mb_cols_) {
>   for (int mb_row = 0; mb_row < mb_rows_; ++mb_row) {
>@@ -323,20 +327,17 @@ rtc::scoped_refptr<I420BufferInterface> VideoDenoiser::DenoiseFrame(
>     CopyLumaOnMargin(y_src, stride_y_src, y_dst, stride_y_dst);
> 
>   // Copy u/v planes.
>-  libyuv::CopyPlane(frame->DataU(), frame->StrideU(),
>-                    dst->MutableDataU(), dst->StrideU(),
>-                    (width_ + 1) >> 1, (height_ + 1) >> 1);
>-  libyuv::CopyPlane(frame->DataV(), frame->StrideV(),
>-                    dst->MutableDataV(), dst->StrideV(),
>-                    (width_ + 1) >> 1, (height_ + 1) >> 1);
>+  libyuv::CopyPlane(frame->DataU(), frame->StrideU(), dst->MutableDataU(),
>+                    dst->StrideU(), (width_ + 1) >> 1, (height_ + 1) >> 1);
>+  libyuv::CopyPlane(frame->DataV(), frame->StrideV(), dst->MutableDataV(),
>+                    dst->StrideV(), (width_ + 1) >> 1, (height_ + 1) >> 1);
> 
> #if DISPLAY || DISPLAYNEON
>   // Show rectangular region
>   ShowRect(filter_, moving_edge_, moving_object_, x_density_, y_density_,
>            frame->DataU(), frame->StrideU(), frame->DataV(), frame->StrideV(),
>-           dst->MutableDataU(), dst->StrideU(),
>-           dst->MutableDataV(), dst->StrideV(),
>-           mb_rows_, mb_cols_);
>+           dst->MutableDataU(), dst->StrideU(), dst->MutableDataV(),
>+           dst->StrideV(), mb_rows_, mb_cols_);
> #endif
>   prev_buffer_ = dst;
>   return dst;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/BUILD.gn
>index 20055568b6b..d0fb00f5ca6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/BUILD.gn
>@@ -21,8 +21,6 @@ rtc_static_library("ortc") {
>     "ortcrtpreceiveradapter.h",
>     "ortcrtpsenderadapter.cc",
>     "ortcrtpsenderadapter.h",
>-    "rtpparametersconversion.cc",
>-    "rtpparametersconversion.h",
>     "rtptransportadapter.cc",
>     "rtptransportadapter.h",
>     "rtptransportcontrolleradapter.cc",
>@@ -34,11 +32,13 @@ rtc_static_library("ortc") {
>   # libjingle_peerconnection.
>   deps = [
>     "../api:libjingle_peerconnection_api",
>-    "../api:optional",
>     "../api:ortc_api",
>+    "../api/video_codecs:builtin_video_decoder_factory",
>+    "../api/video_codecs:builtin_video_encoder_factory",
>     "../call:call_interfaces",
>     "../call:rtp_sender",
>     "../logging:rtc_event_log_api",
>+    "../logging:rtc_event_log_impl_base",
>     "../media:rtc_audio_video",
>     "../media:rtc_media",
>     "../media:rtc_media_base",
>@@ -51,6 +51,9 @@ rtc_static_library("ortc") {
>     "../rtc_base:checks",
>     "../rtc_base:rtc_base",
>     "../rtc_base:rtc_base_approved",
>+    "../rtc_base/third_party/sigslot",
>+    "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> 
>   if (!build_with_chromium && is_clang) {
>@@ -68,7 +71,6 @@ if (rtc_include_tests) {
>       "ortcfactory_unittest.cc",
>       "ortcrtpreceiver_unittest.cc",
>       "ortcrtpsender_unittest.cc",
>-      "rtpparametersconversion_unittest.cc",
>       "rtptransport_unittest.cc",
>       "rtptransportcontroller_unittest.cc",
>       "srtptransport_unittest.cc",
>@@ -86,10 +88,12 @@ if (rtc_include_tests) {
>       "../p2p:p2p_test_utils",
>       "../p2p:rtc_p2p",
>       "../pc:pc_test_utils",
>+      "../pc:peerconnection",
>       "../rtc_base:rtc_base",
>       "../rtc_base:rtc_base_approved",
>       "../rtc_base:rtc_base_tests_main",
>       "../rtc_base:rtc_base_tests_utils",
>+      "../rtc_base/system:arch",
>       "../system_wrappers:metrics_default",
>       "../system_wrappers:runtime_enabled_features_default",
>     ]
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/OWNERS
>index 6c85dbc0c74..4c08fff50b8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/OWNERS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/OWNERS
>@@ -1,5 +1,5 @@
>-deadbeef@webrtc.org
> pthatcher@webrtc.org
>+steveanton@webrtc.org
> 
> # These are for the common case of adding or renaming files. If you're doing
> # structural changes, please get a review from a reviewer in this file.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcfactory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcfactory.cc
>index c64898516b6..66ae77d1810 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcfactory.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcfactory.cc
>@@ -14,9 +14,12 @@
> #include <utility>  // For std::move.
> #include <vector>
> 
>+#include "absl/memory/memory.h"
> #include "api/mediastreamtrackproxy.h"
> #include "api/proxy.h"
> #include "api/rtcerror.h"
>+#include "api/video_codecs/builtin_video_decoder_factory.h"
>+#include "api/video_codecs/builtin_video_encoder_factory.h"
> #include "api/videosourceproxy.h"
> #include "logging/rtc_event_log/rtc_event_log.h"
> #include "media/base/mediaconstants.h"
>@@ -24,7 +27,6 @@
> #include "modules/audio_processing/include/audio_processing.h"
> #include "ortc/ortcrtpreceiveradapter.h"
> #include "ortc/ortcrtpsenderadapter.h"
>-#include "ortc/rtpparametersconversion.h"
> #include "ortc/rtptransportadapter.h"
> #include "ortc/rtptransportcontrolleradapter.h"
> #include "p2p/base/basicpacketsocketfactory.h"
>@@ -32,14 +34,13 @@
> #include "pc/audiotrack.h"
> #include "pc/channelmanager.h"
> #include "pc/localaudiosource.h"
>-#include "pc/videocapturertracksource.h"
>+#include "pc/rtpparametersconversion.h"
> #include "pc/videotrack.h"
> #include "rtc_base/asyncpacketsocket.h"
> #include "rtc_base/bind.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/helpers.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/ptr_util.h"
> 
> namespace {
> 
>@@ -114,10 +115,6 @@ PROXY_WORKER_METHOD3(RTCErrorOr<std::unique_ptr<UdpTransportInterface>>,
> PROXY_METHOD1(rtc::scoped_refptr<AudioSourceInterface>,
>               CreateAudioSource,
>               const cricket::AudioOptions&)
>-PROXY_METHOD2(rtc::scoped_refptr<VideoTrackSourceInterface>,
>-              CreateVideoSource,
>-              std::unique_ptr<cricket::VideoCapturer>,
>-              const MediaConstraintsInterface*)
> PROXY_METHOD2(rtc::scoped_refptr<VideoTrackInterface>,
>               CreateVideoTrack,
>               const std::string&,
>@@ -195,6 +192,7 @@ OrtcFactory::OrtcFactory(
>   // The worker thread is created internally because it's an implementation
>   // detail, and consumers of the API don't need to really know about it.
>   worker_thread_ = rtc::Thread::Create();
>+  worker_thread_->SetName("ORTC-worker", this);
>   worker_thread_->Start();
> 
>   if (signaling_thread_) {
>@@ -208,6 +206,11 @@ OrtcFactory::OrtcFactory(
>       wraps_signaling_thread_ = true;
>     }
>   }
>+
>+  if (signaling_thread_->name().empty()) {
>+    signaling_thread_->SetName("ORTC-signaling", this);
>+  }
>+
>   if (!network_manager_) {
>     owned_network_manager_.reset(new rtc::BasicNetworkManager());
>     network_manager_ = owned_network_manager_.get();
>@@ -231,7 +234,7 @@ OrtcFactory::CreateRtpTransportController() {
>   RTC_DCHECK_RUN_ON(signaling_thread_);
>   return RtpTransportControllerAdapter::CreateProxied(
>       cricket::MediaConfig(), channel_manager_.get(), null_event_log_.get(),
>-      signaling_thread_, worker_thread_.get());
>+      signaling_thread_, worker_thread_.get(), network_thread_);
> }
> 
> RTCErrorOr<std::unique_ptr<RtpTransportInterface>>
>@@ -474,17 +477,6 @@ rtc::scoped_refptr<AudioSourceInterface> OrtcFactory::CreateAudioSource(
>       LocalAudioSource::Create(&options));
> }
> 
>-rtc::scoped_refptr<VideoTrackSourceInterface> OrtcFactory::CreateVideoSource(
>-    std::unique_ptr<cricket::VideoCapturer> capturer,
>-    const MediaConstraintsInterface* constraints) {
>-  RTC_DCHECK_RUN_ON(signaling_thread_);
>-  rtc::scoped_refptr<VideoTrackSourceInterface> source(
>-      VideoCapturerTrackSource::Create(
>-          worker_thread_.get(), std::move(capturer), constraints, false));
>-  return VideoTrackSourceProxy::Create(signaling_thread_, worker_thread_.get(),
>-                                       source);
>-}
>-
> rtc::scoped_refptr<VideoTrackInterface> OrtcFactory::CreateVideoTrack(
>     const std::string& id,
>     VideoTrackSourceInterface* source) {
>@@ -541,7 +533,7 @@ RTCError OrtcFactory::Initialize(
>   }
> 
>   channel_manager_.reset(new cricket::ChannelManager(
>-      std::move(media_engine), rtc::MakeUnique<cricket::RtpDataEngine>(),
>+      std::move(media_engine), absl::make_unique<cricket::RtpDataEngine>(),
>       worker_thread_.get(), network_thread_));
>   channel_manager_->SetVideoRtxEnabled(true);
>   if (!channel_manager_->Init()) {
>@@ -561,8 +553,11 @@ OrtcFactory::CreateMediaEngine_w() {
>   // AudioDeviceModule will be used.
>   return std::unique_ptr<cricket::MediaEngineInterface>(
>       cricket::WebRtcMediaEngineFactory::Create(
>-          adm_, audio_encoder_factory_, audio_decoder_factory_, nullptr,
>-          nullptr, nullptr, webrtc::AudioProcessingBuilder().Create()));
>+          rtc::scoped_refptr<webrtc::AudioDeviceModule>(adm_),
>+          audio_encoder_factory_, audio_decoder_factory_,
>+          webrtc::CreateBuiltinVideoEncoderFactory(),
>+          webrtc::CreateBuiltinVideoDecoderFactory(), nullptr,
>+          webrtc::AudioProcessingBuilder().Create()));
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcfactory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcfactory.h
>index 750fcf75863..d72f6121dbb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcfactory.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcfactory.h
>@@ -81,10 +81,6 @@ class OrtcFactory : public OrtcFactoryInterface {
>   rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource(
>       const cricket::AudioOptions& options) override;
> 
>-  rtc::scoped_refptr<VideoTrackSourceInterface> CreateVideoSource(
>-      std::unique_ptr<cricket::VideoCapturer> capturer,
>-      const MediaConstraintsInterface* constraints) override;
>-
>   rtc::scoped_refptr<VideoTrackInterface> CreateVideoTrack(
>       const std::string& id,
>       VideoTrackSourceInterface* source) override;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcfactory_integrationtest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcfactory_integrationtest.cc
>index 6385dee9162..681dda948e2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcfactory_integrationtest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcfactory_integrationtest.cc
>@@ -17,17 +17,20 @@
> #include "ortc/testrtpparameters.h"
> #include "p2p/base/udptransport.h"
> #include "pc/test/fakeaudiocapturemodule.h"
>-#include "pc/test/fakeperiodicvideocapturer.h"
>+#include "pc/test/fakeperiodicvideotracksource.h"
> #include "pc/test/fakevideotrackrenderer.h"
>+#include "pc/videotracksource.h"
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/fakenetwork.h"
> #include "rtc_base/gunit.h"
>+#include "rtc_base/system/arch.h"
> #include "rtc_base/virtualsocketserver.h"
> 
> namespace {
> 
> const int kDefaultTimeout = 10000;    // 10 seconds.
> const int kReceivingDuration = 1000;  // 1 second.
>+
> // Default number of audio/video frames to wait for before considering a test a
> // success.
> const int kDefaultNumFrames = 3;
>@@ -77,6 +80,7 @@ class OrtcFactoryIntegrationTest : public testing::Test {
>     // Sockets are bound to the ANY address, so this is needed to tell the
>     // virtual network which address to use in this case.
>     virtual_socket_server_.SetDefaultRoute(kIPv4LocalHostAddress);
>+    network_thread_.SetName("TestNetworkThread", this);
>     network_thread_.Start();
>     // Need to create after network thread is started.
>     ortc_factory1_ =
>@@ -216,18 +220,15 @@ class OrtcFactoryIntegrationTest : public testing::Test {
>     return ortc_factory->CreateAudioTrack(id, source);
>   }
> 
>-  // Stores created capturer in |fake_video_capturers_|.
>+  // Stores created video source in |fake_video_track_sources_|.
>   rtc::scoped_refptr<webrtc::VideoTrackInterface>
>-  CreateLocalVideoTrackAndFakeCapturer(const std::string& id,
>-                                       OrtcFactoryInterface* ortc_factory) {
>-    cricket::FakeVideoCapturer* fake_capturer =
>-        new webrtc::FakePeriodicVideoCapturer();
>-    fake_video_capturers_.push_back(fake_capturer);
>-    rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> source =
>-        ortc_factory->CreateVideoSource(
>-            std::unique_ptr<cricket::VideoCapturer>(fake_capturer));
>-    return rtc::scoped_refptr<webrtc::VideoTrackInterface>(
>-        ortc_factory->CreateVideoTrack(id, source));
>+  CreateLocalVideoTrackAndFakeSource(const std::string& id,
>+                                     OrtcFactoryInterface* ortc_factory) {
>+    fake_video_track_sources_.emplace_back(
>+        new rtc::RefCountedObject<FakePeriodicVideoTrackSource>(
>+            false /* remote */));
>+    return rtc::scoped_refptr<VideoTrackInterface>(
>+        ortc_factory->CreateVideoTrack(id, fake_video_track_sources_.back()));
>   }
> 
>   // Helper function used to test two way RTP senders and receivers with basic
>@@ -282,13 +283,13 @@ class OrtcFactoryIntegrationTest : public testing::Test {
>         CreateLocalAudioTrack("audio", ortc_factory1_.get()));
>     EXPECT_TRUE(error.ok());
>     error = video_sender1->SetTrack(
>-        CreateLocalVideoTrackAndFakeCapturer("video", ortc_factory1_.get()));
>+        CreateLocalVideoTrackAndFakeSource("video", ortc_factory1_.get()));
>     EXPECT_TRUE(error.ok());
>     error = audio_sender2->SetTrack(
>         CreateLocalAudioTrack("audio", ortc_factory2_.get()));
>     EXPECT_TRUE(error.ok());
>     error = video_sender2->SetTrack(
>-        CreateLocalVideoTrackAndFakeCapturer("video", ortc_factory2_.get()));
>+        CreateLocalVideoTrackAndFakeSource("video", ortc_factory2_.get()));
>     EXPECT_TRUE(error.ok());
> 
>     // "sent_X_parameters1" are the parameters that endpoint 1 sends with and
>@@ -324,15 +325,15 @@ class OrtcFactoryIntegrationTest : public testing::Test {
>               fake_audio_capture_module2_->frames_received() >
>                   kDefaultNumFrames &&
>               fake_video_renderer2.num_rendered_frames() > kDefaultNumFrames,
>-          kDefaultTimeout) << "Audio capture module 1 received "
>-                           << fake_audio_capture_module1_->frames_received()
>-                           << " frames, Video renderer 1 rendered "
>-                           << fake_video_renderer1.num_rendered_frames()
>-                           << " frames, Audio capture module 2 received "
>-                           << fake_audio_capture_module2_->frames_received()
>-                           << " frames, Video renderer 2 rendered "
>-                           << fake_video_renderer2.num_rendered_frames()
>-                           << " frames.";
>+          kDefaultTimeout)
>+          << "Audio capture module 1 received "
>+          << fake_audio_capture_module1_->frames_received()
>+          << " frames, Video renderer 1 rendered "
>+          << fake_video_renderer1.num_rendered_frames()
>+          << " frames, Audio capture module 2 received "
>+          << fake_audio_capture_module2_->frames_received()
>+          << " frames, Video renderer 2 rendered "
>+          << fake_video_renderer2.num_rendered_frames() << " frames.";
>     } else {
>       WAIT(false, kReceivingDuration);
>       rendered_video_frames1_ = fake_video_renderer1.num_rendered_frames();
>@@ -349,8 +350,7 @@ class OrtcFactoryIntegrationTest : public testing::Test {
>   rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module2_;
>   std::unique_ptr<OrtcFactoryInterface> ortc_factory1_;
>   std::unique_ptr<OrtcFactoryInterface> ortc_factory2_;
>-  // Actually owned by video tracks.
>-  std::vector<cricket::FakeVideoCapturer*> fake_video_capturers_;
>+  std::vector<rtc::scoped_refptr<VideoTrackSource>> fake_video_track_sources_;
>   int received_audio_frames1_ = 0;
>   int received_audio_frames2_ = 0;
>   int rendered_video_frames1_ = 0;
>@@ -414,7 +414,7 @@ TEST_F(OrtcFactoryIntegrationTest, BasicOneWayVideoRtpSenderAndReceiver) {
>   auto receiver = receiver_result.MoveValue();
> 
>   RTCError error = sender->SetTrack(
>-      CreateLocalVideoTrackAndFakeCapturer("video", ortc_factory1_.get()));
>+      CreateLocalVideoTrackAndFakeSource("video", ortc_factory1_.get()));
>   EXPECT_TRUE(error.ok());
> 
>   RtpParameters vp8_parameters = MakeMinimalVp8Parameters();
>@@ -449,7 +449,7 @@ TEST_F(OrtcFactoryIntegrationTest, SetTrackWhileSending) {
>   auto receiver = receiver_result.MoveValue();
> 
>   RTCError error = sender->SetTrack(
>-      CreateLocalVideoTrackAndFakeCapturer("video_1", ortc_factory1_.get()));
>+      CreateLocalVideoTrackAndFakeSource("video_1", ortc_factory1_.get()));
>   EXPECT_TRUE(error.ok());
>   RtpParameters vp8_parameters = MakeMinimalVp8Parameters();
>   EXPECT_TRUE(receiver->Receive(vp8_parameters).ok());
>@@ -459,27 +459,50 @@ TEST_F(OrtcFactoryIntegrationTest, SetTrackWhileSending) {
>   // Expect for some initial number of frames to be received.
>   EXPECT_TRUE_WAIT(fake_renderer.num_rendered_frames() > kDefaultNumFrames,
>                    kDefaultTimeout);
>-  // Stop the old capturer, set a new track, and verify new frames are received
>-  // from the new track. Stopping the old capturer ensures that we aren't
>-  // actually still getting frames from it.
>-  fake_video_capturers_[0]->Stop();
>+  // Destroy old source, set a new track, and verify new frames are received
>+  // from the new track. The VideoTrackSource is reference counted and may live
>+  // a little longer, so tell it that its source is going away now.
>+  fake_video_track_sources_[0] = nullptr;
>   int prev_num_frames = fake_renderer.num_rendered_frames();
>   error = sender->SetTrack(
>-      CreateLocalVideoTrackAndFakeCapturer("video_2", ortc_factory1_.get()));
>+      CreateLocalVideoTrackAndFakeSource("video_2", ortc_factory1_.get()));
>   EXPECT_TRUE(error.ok());
>   EXPECT_TRUE_WAIT(
>       fake_renderer.num_rendered_frames() > kDefaultNumFrames + prev_num_frames,
>       kDefaultTimeout);
> }
> 
>+// TODO(webrtc:7915, webrtc:9184): Tests below are disabled for iOS 64 on debug
>+// builds because of flakiness.
>+#if !(defined(WEBRTC_IOS) && defined(WEBRTC_ARCH_64_BITS) && !defined(NDEBUG))
>+#define MAYBE_BasicTwoWayAudioVideoRtpSendersAndReceivers \
>+  BasicTwoWayAudioVideoRtpSendersAndReceivers
>+#define MAYBE_BasicTwoWayAudioVideoSrtpSendersAndReceivers \
>+  BasicTwoWayAudioVideoSrtpSendersAndReceivers
>+#define MAYBE_SrtpSendersAndReceiversWithMismatchingKeys \
>+  SrtpSendersAndReceiversWithMismatchingKeys
>+#define MAYBE_OneSideSrtpSenderAndReceiver OneSideSrtpSenderAndReceiver
>+#define MAYBE_FullTwoWayAudioVideoSrtpSendersAndReceivers \
>+  FullTwoWayAudioVideoSrtpSendersAndReceivers
>+#else
>+#define MAYBE_BasicTwoWayAudioVideoRtpSendersAndReceivers \
>+  DISABLED_BasicTwoWayAudioVideoRtpSendersAndReceivers
>+#define MAYBE_BasicTwoWayAudioVideoSrtpSendersAndReceivers \
>+  DISABLED_BasicTwoWayAudioVideoSrtpSendersAndReceivers
>+#define MAYBE_SrtpSendersAndReceiversWithMismatchingKeys \
>+  DISABLED_SrtpSendersAndReceiversWithMismatchingKeys
>+#define MAYBE_OneSideSrtpSenderAndReceiver DISABLED_OneSideSrtpSenderAndReceiver
>+#define MAYBE_FullTwoWayAudioVideoSrtpSendersAndReceivers \
>+  DISABLED_FullTwoWayAudioVideoSrtpSendersAndReceivers
>+#endif
>+
> // End-to-end test with two pairs of RTP senders and receivers, for audio and
> // video.
> //
> // Uses muxed RTCP, and minimal parameters with hard-coded configs that are
> // known to work.
>-#if !(defined(WEBRTC_IOS) && defined(WEBRTC_ARCH_64_BITS) && !defined(NDEBUG))
> TEST_F(OrtcFactoryIntegrationTest,
>-       BasicTwoWayAudioVideoRtpSendersAndReceivers) {
>+       MAYBE_BasicTwoWayAudioVideoRtpSendersAndReceivers) {
>   auto udp_transports = CreateAndConnectUdpTransportPair();
>   auto rtp_transports =
>       CreateRtpTransportPair(MakeRtcpMuxParameters(), udp_transports);
>@@ -489,7 +512,7 @@ TEST_F(OrtcFactoryIntegrationTest,
> }
> 
> TEST_F(OrtcFactoryIntegrationTest,
>-       BasicTwoWayAudioVideoSrtpSendersAndReceivers) {
>+       MAYBE_BasicTwoWayAudioVideoSrtpSendersAndReceivers) {
>   auto udp_transports = CreateAndConnectUdpTransportPair();
>   auto srtp_transports = CreateSrtpTransportPairAndSetKeys(
>       MakeRtcpMuxParameters(), udp_transports);
>@@ -497,10 +520,11 @@ TEST_F(OrtcFactoryIntegrationTest,
>   BasicTwoWayRtpSendersAndReceiversTest(std::move(srtp_transports),
>                                         expect_success);
> }
>-#endif
> 
> // Tests that the packets cannot be decoded if the keys are mismatched.
>-TEST_F(OrtcFactoryIntegrationTest, SrtpSendersAndReceiversWithMismatchingKeys) {
>+// TODO(webrtc:9184): Disabled because this test is flaky.
>+TEST_F(OrtcFactoryIntegrationTest,
>+       MAYBE_SrtpSendersAndReceiversWithMismatchingKeys) {
>   auto udp_transports = CreateAndConnectUdpTransportPair();
>   auto srtp_transports = CreateSrtpTransportPairAndSetMismatchingKeys(
>       MakeRtcpMuxParameters(), udp_transports);
>@@ -513,7 +537,7 @@ TEST_F(OrtcFactoryIntegrationTest, SrtpSendersAndReceiversWithMismatchingKeys) {
> }
> 
> // Tests that the frames cannot be decoded if only one side uses SRTP.
>-TEST_F(OrtcFactoryIntegrationTest, OneSideSrtpSenderAndReceiver) {
>+TEST_F(OrtcFactoryIntegrationTest, MAYBE_OneSideSrtpSenderAndReceiver) {
>   auto rtcp_parameters = MakeRtcpMuxParameters();
>   auto udp_transports = CreateAndConnectUdpTransportPair();
>   auto rtcp_udp_transports = UdpTransportPair();
>@@ -552,7 +576,7 @@ TEST_F(OrtcFactoryIntegrationTest, OneSideSrtpSenderAndReceiver) {
> // TODO(deadbeef): Update this test as more audio/video features become
> // supported.
> TEST_F(OrtcFactoryIntegrationTest,
>-       FullTwoWayAudioVideoSrtpSendersAndReceivers) {
>+       MAYBE_FullTwoWayAudioVideoSrtpSendersAndReceivers) {
>   // We want four pairs of UDP transports for this test, for audio/video and
>   // RTP/RTCP.
>   auto audio_rtp_udp_transports = CreateAndConnectUdpTransportPair();
>@@ -627,13 +651,13 @@ TEST_F(OrtcFactoryIntegrationTest,
>       CreateLocalAudioTrack("audio", ortc_factory1_.get()));
>   EXPECT_TRUE(error.ok());
>   error = video_sender1->SetTrack(
>-      CreateLocalVideoTrackAndFakeCapturer("video", ortc_factory1_.get()));
>+      CreateLocalVideoTrackAndFakeSource("video", ortc_factory1_.get()));
>   EXPECT_TRUE(error.ok());
>   error = audio_sender2->SetTrack(
>       CreateLocalAudioTrack("audio", ortc_factory2_.get()));
>   EXPECT_TRUE(error.ok());
>   error = video_sender2->SetTrack(
>-      CreateLocalVideoTrackAndFakeCapturer("video", ortc_factory2_.get()));
>+      CreateLocalVideoTrackAndFakeSource("video", ortc_factory2_.get()));
>   EXPECT_TRUE(error.ok());
> 
>   // Use different codecs in different directions for extra challenge.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcrtpreceiveradapter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcrtpreceiveradapter.cc
>index 67f4de642ea..689412217ac 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcrtpreceiveradapter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcrtpreceiveradapter.cc
>@@ -10,7 +10,9 @@
> 
> #include "ortc/ortcrtpreceiveradapter.h"
> 
>+#include <string>
> #include <utility>
>+#include <vector>
> 
> #include "media/base/mediaconstants.h"
> #include "ortc/rtptransportadapter.h"
>@@ -151,22 +153,29 @@ void OrtcRtpReceiverAdapter::MaybeRecreateInternalReceiver() {
>   internal_receiver_ = nullptr;
>   switch (kind_) {
>     case cricket::MEDIA_TYPE_AUDIO: {
>-      auto* voice_channel = rtp_transport_controller_->voice_channel();
>-      internal_receiver_ = new AudioRtpReceiver(
>+      auto* audio_receiver = new AudioRtpReceiver(
>           rtp_transport_controller_->worker_thread(), rtc::CreateRandomUuid(),
>-          {}, ssrc, (voice_channel ? voice_channel->media_channel() : nullptr));
>+          std::vector<std::string>({}));
>+      auto* voice_channel = rtp_transport_controller_->voice_channel();
>+      RTC_DCHECK(voice_channel);
>+      audio_receiver->SetVoiceMediaChannel(voice_channel->media_channel());
>+      internal_receiver_ = audio_receiver;
>       break;
>     }
>     case cricket::MEDIA_TYPE_VIDEO: {
>-      auto* video_channel = rtp_transport_controller_->video_channel();
>-      internal_receiver_ = new VideoRtpReceiver(
>+      auto* video_receiver = new VideoRtpReceiver(
>           rtp_transport_controller_->worker_thread(), rtc::CreateRandomUuid(),
>-          {}, ssrc, (video_channel ? video_channel->media_channel() : nullptr));
>+          std::vector<std::string>({}));
>+      auto* video_channel = rtp_transport_controller_->video_channel();
>+      RTC_DCHECK(video_channel);
>+      video_receiver->SetVideoMediaChannel(video_channel->media_channel());
>+      internal_receiver_ = video_receiver;
>       break;
>     }
>     case cricket::MEDIA_TYPE_DATA:
>       RTC_NOTREACHED();
>   }
>+  internal_receiver_->SetupMediaChannel(ssrc);
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcrtpreceiveradapter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcrtpreceiveradapter.h
>index 4e0a40672cc..4647b043145 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcrtpreceiveradapter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcrtpreceiveradapter.h
>@@ -19,7 +19,7 @@
> #include "ortc/rtptransportcontrolleradapter.h"
> #include "pc/rtpreceiver.h"
> #include "rtc_base/constructormagic.h"
>-#include "rtc_base/sigslot.h"
>+#include "rtc_base/third_party/sigslot/sigslot.h"
> #include "rtc_base/thread.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcrtpsenderadapter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcrtpsenderadapter.cc
>index 2b25d46eebc..f2541cb8f85 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcrtpsenderadapter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcrtpsenderadapter.cc
>@@ -157,19 +157,19 @@ void OrtcRtpSenderAdapter::CreateInternalSender() {
>   switch (kind_) {
>     case cricket::MEDIA_TYPE_AUDIO: {
>       auto* audio_sender = new AudioRtpSender(
>-          rtp_transport_controller_->worker_thread(), nullptr);
>+          rtp_transport_controller_->worker_thread(), /*id=*/"", nullptr);
>       auto* voice_channel = rtp_transport_controller_->voice_channel();
>       RTC_DCHECK(voice_channel);
>-      audio_sender->SetMediaChannel(voice_channel->media_channel());
>+      audio_sender->SetVoiceMediaChannel(voice_channel->media_channel());
>       internal_sender_ = audio_sender;
>       break;
>     }
>     case cricket::MEDIA_TYPE_VIDEO: {
>-      auto* video_sender =
>-          new VideoRtpSender(rtp_transport_controller_->worker_thread());
>+      auto* video_sender = new VideoRtpSender(
>+          rtp_transport_controller_->worker_thread(), /*id=*/"");
>       auto* video_channel = rtp_transport_controller_->video_channel();
>       RTC_DCHECK(video_channel);
>-      video_sender->SetMediaChannel(video_channel->media_channel());
>+      video_sender->SetVideoMediaChannel(video_channel->media_channel());
>       internal_sender_ = video_sender;
>       break;
>     }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcrtpsenderadapter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcrtpsenderadapter.h
>index e1f096f6d34..b078da985c9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcrtpsenderadapter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcrtpsenderadapter.h
>@@ -19,7 +19,7 @@
> #include "ortc/rtptransportcontrolleradapter.h"
> #include "pc/rtpsender.h"
> #include "rtc_base/constructormagic.h"
>-#include "rtc_base/sigslot.h"
>+#include "rtc_base/third_party/sigslot/sigslot.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtptransportadapter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtptransportadapter.cc
>index 0cf9217c6c4..2829427c6c4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtptransportadapter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtptransportadapter.cc
>@@ -15,6 +15,7 @@
> #include <sstream>
> #include <utility>  // For std::move.
> 
>+#include "absl/memory/memory.h"
> #include "api/proxy.h"
> #include "rtc_base/logging.h"
> 
>@@ -115,10 +116,11 @@ RtpTransportAdapter::CreateSrtpProxied(
>     LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
>                          "Must provide an RTP transport controller.");
>   }
>-  std::unique_ptr<RtpTransportAdapter> transport_adapter(
>-      new RtpTransportAdapter(parameters.rtcp, rtp, rtcp,
>-                              rtp_transport_controller,
>-                              true /*is_srtp_transport*/));
>+
>+  std::unique_ptr<RtpTransportAdapter> transport_adapter;
>+  transport_adapter.reset(new RtpTransportAdapter(parameters.rtcp, rtp, rtcp,
>+                                                  rtp_transport_controller,
>+                                                  true /*is_srtp_transport*/));
>   RTCError params_result = transport_adapter->SetParameters(parameters);
>   if (!params_result.ok()) {
>     return std::move(params_result);
>@@ -145,25 +147,41 @@ RtpTransportAdapter::RtpTransportAdapter(
>     : rtp_packet_transport_(rtp),
>       rtcp_packet_transport_(rtcp),
>       rtp_transport_controller_(rtp_transport_controller),
>-      is_srtp_transport_(is_srtp_transport) {
>+      network_thread_(rtp_transport_controller_->network_thread()) {
>   parameters_.rtcp = rtcp_params;
>   // CNAME should have been filled by OrtcFactory if empty.
>   RTC_DCHECK(!parameters_.rtcp.cname.empty());
>   RTC_DCHECK(rtp_transport_controller);
>+
>+  if (is_srtp_transport) {
>+    srtp_transport_ = absl::make_unique<SrtpTransport>(rtcp == nullptr);
>+    transport_ = srtp_transport_.get();
>+  } else {
>+    unencrypted_rtp_transport_ =
>+        absl::make_unique<RtpTransport>(rtcp == nullptr);
>+    transport_ = unencrypted_rtp_transport_.get();
>+  }
>+  RTC_DCHECK(transport_);
>+
>+  network_thread_->Invoke<void>(RTC_FROM_HERE, [=] {
>+    SetRtpPacketTransport(rtp->GetInternal());
>+    if (rtcp) {
>+      SetRtcpPacketTransport(rtcp->GetInternal());
>+    }
>+  });
>+
>+  transport_->SignalReadyToSend.connect(this,
>+                                        &RtpTransportAdapter::OnReadyToSend);
>+  transport_->SignalRtcpPacketReceived.connect(
>+      this, &RtpTransportAdapter::OnRtcpPacketReceived);
>+  transport_->SignalWritableState.connect(
>+      this, &RtpTransportAdapter::OnWritableState);
> }
> 
> RtpTransportAdapter::~RtpTransportAdapter() {
>   SignalDestroyed(this);
> }
> 
>-PacketTransportInterface* RtpTransportAdapter::GetRtpPacketTransport() const {
>-  return rtp_packet_transport_;
>-}
>-
>-PacketTransportInterface* RtpTransportAdapter::GetRtcpPacketTransport() const {
>-  return rtcp_packet_transport_;
>-}
>-
> RTCError RtpTransportAdapter::SetParameters(
>     const RtpTransportParameters& parameters) {
>   if (!parameters.rtcp.mux && parameters_.rtcp.mux) {
>@@ -194,34 +212,20 @@ RTCError RtpTransportAdapter::SetParameters(
> 
> RTCError RtpTransportAdapter::SetSrtpSendKey(
>     const cricket::CryptoParams& params) {
>-  if (send_key_) {
>-    LOG_AND_RETURN_ERROR(
>-        webrtc::RTCErrorType::UNSUPPORTED_OPERATION,
>-        "Setting the SRTP send key twice is currently unsupported.");
>-  }
>-  if (receive_key_ && receive_key_->cipher_suite != params.cipher_suite) {
>-    LOG_AND_RETURN_ERROR(
>-        webrtc::RTCErrorType::UNSUPPORTED_OPERATION,
>-        "The send key and receive key must have the same cipher suite.");
>+  if (!network_thread_->IsCurrent()) {
>+    return network_thread_->Invoke<RTCError>(
>+        RTC_FROM_HERE, [&] { return SetSrtpSendKey(params); });
>   }
>-  send_key_ = params;
>-  return RTCError::OK();
>+  return transport_->SetSrtpSendKey(params);
> }
> 
> RTCError RtpTransportAdapter::SetSrtpReceiveKey(
>     const cricket::CryptoParams& params) {
>-  if (receive_key_) {
>-    LOG_AND_RETURN_ERROR(
>-        webrtc::RTCErrorType::UNSUPPORTED_OPERATION,
>-        "Setting the SRTP receive key twice is currently unsupported.");
>-  }
>-  if (send_key_ && send_key_->cipher_suite != params.cipher_suite) {
>-    LOG_AND_RETURN_ERROR(
>-        webrtc::RTCErrorType::UNSUPPORTED_OPERATION,
>-        "The send key and receive key must have the same cipher suite.");
>+  if (!network_thread_->IsCurrent()) {
>+    return network_thread_->Invoke<RTCError>(
>+        RTC_FROM_HERE, [&] { return SetSrtpReceiveKey(params); });
>   }
>-  receive_key_ = params;
>-  return RTCError::OK();
>+  return transport_->SetSrtpReceiveKey(params);
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtptransportadapter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtptransportadapter.h
>index 2c98f5df873..81914f6b8d1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtptransportadapter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtptransportadapter.h
>@@ -14,24 +14,28 @@
> #include <memory>
> #include <vector>
> 
>-#include "api/ortc/srtptransportinterface.h"
> #include "api/rtcerror.h"
> #include "media/base/streamparams.h"
> #include "ortc/rtptransportcontrolleradapter.h"
> #include "pc/channel.h"
>+#include "pc/rtptransportinternaladapter.h"
>+#include "pc/srtptransport.h"
> #include "rtc_base/constructormagic.h"
>-#include "rtc_base/sigslot.h"
>+#include "rtc_base/third_party/sigslot/sigslot.h"
> 
> namespace webrtc {
>-
>-// Implementation of SrtpTransportInterface to be used with RtpSenderAdapter,
>-// RtpReceiverAdapter, and RtpTransportControllerAdapter classes. This class
>-// is used to implement both a secure and insecure RTP transport.
>+// This class is a wrapper over an RtpTransport or an SrtpTransport. The base
>+// class RtpTransportInternalAdapter keeps a raw pointer, |transport_|, of the
>+// transport object and implements both the public SrtpTransportInterface and
>+// RtpTransport internal interface by calling the |transport_| underneath.
>+//
>+// This adapter can be used as an unencrypted RTP transport or an SrtpTransport
>+// with RtpSenderAdapter, RtpReceiverAdapter, and RtpTransportControllerAdapter.
> //
> // TODO(deadbeef): When BaseChannel is split apart into separate
> // "RtpTransport"/"RtpTransceiver"/"RtpSender"/"RtpReceiver" objects, this
> // adapter object can be removed.
>-class RtpTransportAdapter : public SrtpTransportInterface {
>+class RtpTransportAdapter : public RtpTransportInternalAdapter {
>  public:
>   // |rtp| can't be null. |rtcp| can if RTCP muxing is used immediately (meaning
>   // |rtcp_parameters.mux| is also true).
>@@ -50,8 +54,12 @@ class RtpTransportAdapter : public SrtpTransportInterface {
>   ~RtpTransportAdapter() override;
> 
>   // RtpTransportInterface implementation.
>-  PacketTransportInterface* GetRtpPacketTransport() const override;
>-  PacketTransportInterface* GetRtcpPacketTransport() const override;
>+  PacketTransportInterface* GetRtpPacketTransport() const override {
>+    return rtp_packet_transport_;
>+  }
>+  PacketTransportInterface* GetRtcpPacketTransport() const override {
>+    return rtcp_packet_transport_;
>+  }
>   RTCError SetParameters(const RtpTransportParameters& parameters) override;
>   RtpTransportParameters GetParameters() const override { return parameters_; }
> 
>@@ -70,13 +78,7 @@ class RtpTransportAdapter : public SrtpTransportInterface {
>   // returning this transport from GetTransports().
>   sigslot::signal1<RtpTransportAdapter*> SignalDestroyed;
> 
>-  // Used by the RtpTransportControllerAdapter to tell if an rtp sender or
>-  // receiver can be created.
>-  bool is_srtp_transport() { return is_srtp_transport_; }
>-  // Used by the RtpTransportControllerAdapter to set keys for senders and
>-  // receivers.
>-  rtc::Optional<cricket::CryptoParams> send_key() { return send_key_; }
>-  rtc::Optional<cricket::CryptoParams> receive_key() { return receive_key_; }
>+  bool IsSrtpActive() const override { return transport_->IsSrtpActive(); }
> 
>  protected:
>   RtpTransportAdapter* GetInternal() override { return this; }
>@@ -88,18 +90,28 @@ class RtpTransportAdapter : public SrtpTransportInterface {
>                       RtpTransportControllerAdapter* rtp_transport_controller,
>                       bool is_srtp_transport);
> 
>-  PacketTransportInterface* rtp_packet_transport_;
>-  PacketTransportInterface* rtcp_packet_transport_;
>-  RtpTransportControllerAdapter* const rtp_transport_controller_;
>+  void OnReadyToSend(bool ready) { SignalReadyToSend(ready); }
>+
>+  void OnRtcpPacketReceived(rtc::CopyOnWriteBuffer* packet,
>+                            const rtc::PacketTime& time) {
>+    SignalRtcpPacketReceived(packet, time);
>+  }
>+
>+  void OnWritableState(bool writable) { SignalWritableState(writable); }
>+
>+  PacketTransportInterface* rtp_packet_transport_ = nullptr;
>+  PacketTransportInterface* rtcp_packet_transport_ = nullptr;
>+  RtpTransportControllerAdapter* const rtp_transport_controller_ = nullptr;
>   // Non-null if this class owns the transport controller.
>   std::unique_ptr<RtpTransportControllerInterface>
>       owned_rtp_transport_controller_;
>   RtpTransportParameters parameters_;
> 
>-  // SRTP specific members.
>-  rtc::Optional<cricket::CryptoParams> send_key_;
>-  rtc::Optional<cricket::CryptoParams> receive_key_;
>-  bool is_srtp_transport_;
>+  // Only one of them is non-null;
>+  std::unique_ptr<RtpTransport> unencrypted_rtp_transport_;
>+  std::unique_ptr<SrtpTransport> srtp_transport_;
>+
>+  rtc::Thread* network_thread_ = nullptr;
> 
>   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RtpTransportAdapter);
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtptransportcontroller_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtptransportcontroller_unittest.cc
>index a663b82e7fc..49548417ee7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtptransportcontroller_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtptransportcontroller_unittest.cc
>@@ -185,6 +185,7 @@ TEST_F(RtpTransportControllerTest,
>   auto video_sender_result = ortc_factory_->CreateRtpSender(
>       cricket::MEDIA_TYPE_VIDEO, rtp_transport2.get());
>   EXPECT_TRUE(video_sender_result.ok());
>+
>   auto video_receiver_result = ortc_factory_->CreateRtpReceiver(
>       cricket::MEDIA_TYPE_VIDEO, rtp_transport1.get());
>   EXPECT_EQ(RTCErrorType::UNSUPPORTED_OPERATION,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtptransportcontrolleradapter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtptransportcontrolleradapter.cc
>index 9d07be651b1..e4c9ea24d2c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtptransportcontrolleradapter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtptransportcontrolleradapter.cc
>@@ -16,13 +16,14 @@
> #include <unordered_map>
> #include <utility>  // For std::move.
> 
>+#include "absl/memory/memory.h"
> #include "api/proxy.h"
> #include "media/base/mediaconstants.h"
> #include "ortc/ortcrtpreceiveradapter.h"
> #include "ortc/ortcrtpsenderadapter.h"
>-#include "ortc/rtpparametersconversion.h"
> #include "ortc/rtptransportadapter.h"
> #include "pc/rtpmediautils.h"
>+#include "pc/rtpparametersconversion.h"
> #include "rtc_base/checks.h"
> 
> namespace webrtc {
>@@ -96,10 +97,12 @@ RtpTransportControllerAdapter::CreateProxied(
>     cricket::ChannelManager* channel_manager,
>     webrtc::RtcEventLog* event_log,
>     rtc::Thread* signaling_thread,
>-    rtc::Thread* worker_thread) {
>+    rtc::Thread* worker_thread,
>+    rtc::Thread* network_thread) {
>   std::unique_ptr<RtpTransportControllerAdapter> wrapped(
>       new RtpTransportControllerAdapter(config, channel_manager, event_log,
>-                                        signaling_thread, worker_thread));
>+                                        signaling_thread, worker_thread,
>+                                        network_thread));
>   return RtpTransportControllerProxyWithInternal<
>       RtpTransportControllerAdapter>::Create(signaling_thread, worker_thread,
>                                              std::move(wrapped));
>@@ -124,8 +127,7 @@ RtpTransportControllerAdapter::~RtpTransportControllerAdapter() {
>   }
>   // Call must be destroyed on the worker thread.
>   worker_thread_->Invoke<void>(
>-      RTC_FROM_HERE,
>-      rtc::Bind(&RtpTransportControllerAdapter::Close_w, this));
>+      RTC_FROM_HERE, rtc::Bind(&RtpTransportControllerAdapter::Close_w, this));
> }
> 
> RTCErrorOr<std::unique_ptr<RtpTransportInterface>>
>@@ -609,9 +611,11 @@ RtpTransportControllerAdapter::RtpTransportControllerAdapter(
>     cricket::ChannelManager* channel_manager,
>     webrtc::RtcEventLog* event_log,
>     rtc::Thread* signaling_thread,
>-    rtc::Thread* worker_thread)
>+    rtc::Thread* worker_thread,
>+    rtc::Thread* network_thread)
>     : signaling_thread_(signaling_thread),
>       worker_thread_(worker_thread),
>+      network_thread_(network_thread),
>       media_config_(config),
>       channel_manager_(channel_manager),
>       event_log_(event_log),
>@@ -631,8 +635,7 @@ RtpTransportControllerAdapter::RtpTransportControllerAdapter(
>   remote_video_description_.AddCodec(dummy_video);
> 
>   worker_thread_->Invoke<void>(
>-      RTC_FROM_HERE,
>-      rtc::Bind(&RtpTransportControllerAdapter::Init_w, this));
>+      RTC_FROM_HERE, rtc::Bind(&RtpTransportControllerAdapter::Init_w, this));
> }
> 
> // TODO(nisse): Duplicates corresponding method in PeerConnection (used
>@@ -650,12 +653,12 @@ void RtpTransportControllerAdapter::Init_w() {
>   call_config.bitrate_config.min_bitrate_bps = kMinBandwidthBps;
>   call_config.bitrate_config.start_bitrate_bps = kStartBandwidthBps;
>   call_config.bitrate_config.max_bitrate_bps = kMaxBandwidthBps;
>-
>-  call_send_rtp_transport_controller_ =
>-      new RtpTransportControllerSend(Clock::GetRealTimeClock(), event_log_);
>-  call_.reset(webrtc::Call::Create(
>-      call_config, std::unique_ptr<RtpTransportControllerSendInterface>(
>-                       call_send_rtp_transport_controller_)));
>+  std::unique_ptr<RtpTransportControllerSend> controller_send =
>+      absl::make_unique<RtpTransportControllerSend>(
>+          Clock::GetRealTimeClock(), event_log_,
>+          call_config.network_controller_factory, call_config.bitrate_config);
>+  call_send_rtp_transport_controller_ = controller_send.get();
>+  call_.reset(webrtc::Call::Create(call_config, std::move(controller_send)));
> }
> 
> void RtpTransportControllerAdapter::Close_w() {
>@@ -678,11 +681,7 @@ RTCError RtpTransportControllerAdapter::AttachAudioSender(
>                          "RtpSender and RtpReceiver is not currently "
>                          "supported.");
>   }
>-  RTCError err = MaybeSetCryptos(inner_transport, &local_audio_description_,
>-                                 &remote_audio_description_);
>-  if (!err.ok()) {
>-    return err;
>-  }
>+
>   // If setting new transport, extract its RTCP parameters and create voice
>   // channel.
>   if (!inner_audio_transport_) {
>@@ -713,11 +712,7 @@ RTCError RtpTransportControllerAdapter::AttachVideoSender(
>                          "RtpSender and RtpReceiver is not currently "
>                          "supported.");
>   }
>-  RTCError err = MaybeSetCryptos(inner_transport, &local_video_description_,
>-                                 &remote_video_description_);
>-  if (!err.ok()) {
>-    return err;
>-  }
>+
>   // If setting new transport, extract its RTCP parameters and create video
>   // channel.
>   if (!inner_video_transport_) {
>@@ -748,11 +743,7 @@ RTCError RtpTransportControllerAdapter::AttachAudioReceiver(
>                          "RtpReceiver and RtpReceiver is not currently "
>                          "supported.");
>   }
>-  RTCError err = MaybeSetCryptos(inner_transport, &local_audio_description_,
>-                                 &remote_audio_description_);
>-  if (!err.ok()) {
>-    return err;
>-  }
>+
>   // If setting new transport, extract its RTCP parameters and create voice
>   // channel.
>   if (!inner_audio_transport_) {
>@@ -783,11 +774,6 @@ RTCError RtpTransportControllerAdapter::AttachVideoReceiver(
>                          "RtpReceiver and RtpReceiver is not currently "
>                          "supported.");
>   }
>-  RTCError err = MaybeSetCryptos(inner_transport, &local_video_description_,
>-                                 &remote_video_description_);
>-  if (!err.ok()) {
>-    return err;
>-  }
>   // If setting new transport, extract its RTCP parameters and create video
>   // channel.
>   if (!inner_video_transport_) {
>@@ -877,24 +863,18 @@ void RtpTransportControllerAdapter::OnVideoReceiverDestroyed() {
> 
> void RtpTransportControllerAdapter::CreateVoiceChannel() {
>   voice_channel_ = channel_manager_->CreateVoiceChannel(
>-      call_.get(), media_config_,
>-      inner_audio_transport_->GetRtpPacketTransport()->GetInternal(),
>-      inner_audio_transport_->GetRtcpPacketTransport()
>-          ? inner_audio_transport_->GetRtcpPacketTransport()->GetInternal()
>-          : nullptr,
>-      signaling_thread_, "audio", false, cricket::AudioOptions());
>+      call_.get(), media_config_, inner_audio_transport_->GetInternal(),
>+      signaling_thread_, "audio", false, rtc::CryptoOptions(),
>+      cricket::AudioOptions());
>   RTC_DCHECK(voice_channel_);
>   voice_channel_->Enable(true);
> }
> 
> void RtpTransportControllerAdapter::CreateVideoChannel() {
>   video_channel_ = channel_manager_->CreateVideoChannel(
>-      call_.get(), media_config_,
>-      inner_video_transport_->GetRtpPacketTransport()->GetInternal(),
>-      inner_video_transport_->GetRtcpPacketTransport()
>-          ? inner_video_transport_->GetRtcpPacketTransport()->GetInternal()
>-          : nullptr,
>-      signaling_thread_, "video", false, cricket::VideoOptions());
>+      call_.get(), media_config_, inner_video_transport_->GetInternal(),
>+      signaling_thread_, "video", false, rtc::CryptoOptions(),
>+      cricket::VideoOptions());
>   RTC_DCHECK(video_channel_);
>   video_channel_->Enable(true);
> }
>@@ -989,25 +969,4 @@ RtpTransportControllerAdapter::MakeSendStreamParamsVec(
>   return result;
> }
> 
>-RTCError RtpTransportControllerAdapter::MaybeSetCryptos(
>-    RtpTransportInterface* rtp_transport,
>-    cricket::MediaContentDescription* local_description,
>-    cricket::MediaContentDescription* remote_description) {
>-  if (rtp_transport->GetInternal()->is_srtp_transport()) {
>-    if (!rtp_transport->GetInternal()->send_key() ||
>-        !rtp_transport->GetInternal()->receive_key()) {
>-      LOG_AND_RETURN_ERROR(webrtc::RTCErrorType::UNSUPPORTED_PARAMETER,
>-                           "The SRTP send key or receive key is not set.")
>-    }
>-    std::vector<cricket::CryptoParams> cryptos;
>-    cryptos.push_back(*(rtp_transport->GetInternal()->receive_key()));
>-    local_description->set_cryptos(cryptos);
>-
>-    cryptos.clear();
>-    cryptos.push_back(*(rtp_transport->GetInternal()->send_key()));
>-    remote_description->set_cryptos(cryptos);
>-  }
>-  return RTCError::OK();
>-}
>-
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtptransportcontrolleradapter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtptransportcontrolleradapter.h
>index ec306879799..14c578aa46f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtptransportcontrolleradapter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtptransportcontrolleradapter.h
>@@ -26,7 +26,7 @@
> #include "media/base/mediachannel.h"  // For MediaConfig.
> #include "pc/channelmanager.h"
> #include "rtc_base/constructormagic.h"
>-#include "rtc_base/sigslot.h"
>+#include "rtc_base/third_party/sigslot/sigslot.h"
> #include "rtc_base/thread.h"
> 
> namespace webrtc {
>@@ -66,7 +66,8 @@ class RtpTransportControllerAdapter : public RtpTransportControllerInterface,
>       cricket::ChannelManager* channel_manager,
>       webrtc::RtcEventLog* event_log,
>       rtc::Thread* signaling_thread,
>-      rtc::Thread* worker_thread);
>+      rtc::Thread* worker_thread,
>+      rtc::Thread* network_thread);
> 
>   ~RtpTransportControllerAdapter() override;
> 
>@@ -100,6 +101,7 @@ class RtpTransportControllerAdapter : public RtpTransportControllerInterface,
>   // Methods used internally by other "adapter" classes.
>   rtc::Thread* signaling_thread() const { return signaling_thread_; }
>   rtc::Thread* worker_thread() const { return worker_thread_; }
>+  rtc::Thread* network_thread() const { return network_thread_; }
> 
>   // |parameters.keepalive| will be set for ALL RTP transports in the call.
>   RTCError SetRtpTransportParameters(const RtpTransportParameters& parameters,
>@@ -131,7 +133,8 @@ class RtpTransportControllerAdapter : public RtpTransportControllerInterface,
>                                 cricket::ChannelManager* channel_manager,
>                                 webrtc::RtcEventLog* event_log,
>                                 rtc::Thread* signaling_thread,
>-                                rtc::Thread* worker_thread);
>+                                rtc::Thread* worker_thread,
>+                                rtc::Thread* network_thread);
>   void Init_w();
>   void Close_w();
> 
>@@ -180,15 +183,9 @@ class RtpTransportControllerAdapter : public RtpTransportControllerInterface,
>       const std::string& cname,
>       const cricket::MediaContentDescription& description) const;
> 
>-  // If the |rtp_transport| is a SrtpTransport, set the cryptos of the
>-  // audio/video content descriptions.
>-  RTCError MaybeSetCryptos(
>-      RtpTransportInterface* rtp_transport,
>-      cricket::MediaContentDescription* local_description,
>-      cricket::MediaContentDescription* remote_description);
>-
>   rtc::Thread* signaling_thread_;
>   rtc::Thread* worker_thread_;
>+  rtc::Thread* network_thread_;
>   // |transport_proxies_| and |inner_audio_transport_|/|inner_audio_transport_|
>   // are somewhat redundant, but the latter are only set when
>   // RtpSenders/RtpReceivers are attached to the transport.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/srtptransport_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/srtptransport_unittest.cc
>index 8323603af03..1506aa70414 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/srtptransport_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/srtptransport_unittest.cc
>@@ -122,49 +122,4 @@ TEST_F(SrtpTransportTest, SetSrtpSendAndReceiveKeyDifferentCipherSuite) {
>   EXPECT_TRUE(receiver_result.ok());
> }
> 
>-class SrtpTransportTestWithMediaType
>-    : public SrtpTransportTest,
>-      public ::testing::WithParamInterface<cricket::MediaType> {};
>-
>-// Tests that the senders cannot be created before setting the keys.
>-TEST_P(SrtpTransportTestWithMediaType, CreateSenderBeforeSettingKeys) {
>-  auto sender_result =
>-      ortc_factory_->CreateRtpSender(GetParam(), srtp_transport_.get());
>-  EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER, sender_result.error().type());
>-  EXPECT_TRUE(srtp_transport_->SetSrtpSendKey(kTestSha1CryptoParams1).ok());
>-  sender_result =
>-      ortc_factory_->CreateRtpSender(GetParam(), srtp_transport_.get());
>-  EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER, sender_result.error().type());
>-  EXPECT_TRUE(srtp_transport_->SetSrtpReceiveKey(kTestSha1CryptoParams2).ok());
>-  // Ensure that after the keys are set, a sender can be created, despite the
>-  // previous errors.
>-  sender_result =
>-      ortc_factory_->CreateRtpSender(GetParam(), srtp_transport_.get());
>-  EXPECT_TRUE(sender_result.ok());
>-}
>-
>-// Tests that the receivers cannot be created before setting the keys.
>-TEST_P(SrtpTransportTestWithMediaType, CreateReceiverBeforeSettingKeys) {
>-  auto receiver_result =
>-      ortc_factory_->CreateRtpReceiver(GetParam(), srtp_transport_.get());
>-  EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
>-            receiver_result.error().type());
>-  EXPECT_TRUE(srtp_transport_->SetSrtpSendKey(kTestSha1CryptoParams1).ok());
>-  receiver_result =
>-      ortc_factory_->CreateRtpReceiver(GetParam(), srtp_transport_.get());
>-  EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
>-            receiver_result.error().type());
>-  EXPECT_TRUE(srtp_transport_->SetSrtpReceiveKey(kTestSha1CryptoParams2).ok());
>-  // Ensure that after the keys are set, a receiver can be created, despite the
>-  // previous errors.
>-  receiver_result =
>-      ortc_factory_->CreateRtpReceiver(GetParam(), srtp_transport_.get());
>-  EXPECT_TRUE(receiver_result.ok());
>-}
>-
>-INSTANTIATE_TEST_CASE_P(SenderCreatationTest,
>-                        SrtpTransportTestWithMediaType,
>-                        ::testing::Values(cricket::MEDIA_TYPE_AUDIO,
>-                                          cricket::MEDIA_TYPE_VIDEO));
>-
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/BUILD.gn
>index 9ab79e470bc..5efdb0b4c73 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/BUILD.gn
>@@ -15,19 +15,16 @@ group("p2p") {
>   ]
> }
> 
>-config("rtc_p2p_inherited_config") {
>-  defines = [ "FEATURE_ENABLE_VOICEMAIL" ]
>-}
>-
> rtc_static_library("rtc_p2p") {
>   visibility = [ "*" ]
>   sources = [
>     "base/asyncstuntcpsocket.cc",
>     "base/asyncstuntcpsocket.h",
>+    "base/basicasyncresolverfactory.cc",
>+    "base/basicasyncresolverfactory.h",
>     "base/basicpacketsocketfactory.cc",
>     "base/basicpacketsocketfactory.h",
>     "base/candidatepairinterface.h",
>-    "base/common.h",
>     "base/dtlstransport.cc",
>     "base/dtlstransport.h",
>     "base/dtlstransportinternal.cc",
>@@ -53,10 +50,10 @@ rtc_static_library("rtc_p2p") {
>     "base/portinterface.h",
>     "base/pseudotcp.cc",
>     "base/pseudotcp.h",
>+    "base/regatheringcontroller.cc",
>+    "base/regatheringcontroller.h",
>     "base/relayport.cc",
>     "base/relayport.h",
>-    "base/session.cc",
>-    "base/session.h",
>     "base/stun.cc",
>     "base/stun.h",
>     "base/stunport.cc",
>@@ -70,6 +67,7 @@ rtc_static_library("rtc_p2p") {
>     "base/transportdescription.h",
>     "base/transportdescriptionfactory.cc",
>     "base/transportdescriptionfactory.h",
>+    "base/transportfactoryinterface.h",
>     "base/transportinfo.h",
>     "base/turnport.cc",
>     "base/turnport.h",
>@@ -79,26 +77,26 @@ rtc_static_library("rtc_p2p") {
>     "client/basicportallocator.cc",
>     "client/basicportallocator.h",
>     "client/relayportfactoryinterface.h",
>-    "client/socketmonitor.cc",
>-    "client/socketmonitor.h",
>     "client/turnportfactory.cc",
>     "client/turnportfactory.h",
>   ]
> 
>-  defines = []
>-
>   deps = [
>     "../api:libjingle_peerconnection_api",
>-    "../api:optional",
>     "../api:ortc_api",
>+    "../logging:ice_log",
>     "../rtc_base:checks",
>     "../rtc_base:rtc_base",
>+    "../rtc_base:safe_minmax",
>     "../rtc_base:stringutils",
>+    "../rtc_base/third_party/base64",
>+    "../rtc_base/third_party/sigslot",
>     "../system_wrappers:field_trial_api",
>+    "../system_wrappers:metrics_api",
>+    "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> 
>-  public_configs = [ ":rtc_p2p_inherited_config" ]
>-
>   if (build_with_chromium) {
>     if (is_nacl) {
>       deps += [ "//native_client_sdk/src/libraries/nacl_io" ]
>@@ -112,10 +110,6 @@ rtc_static_library("rtc_p2p") {
>       "base/turnserver.cc",
>       "base/turnserver.h",
>     ]
>-    defines += [
>-      "FEATURE_ENABLE_VOICEMAIL",
>-      "FEATURE_ENABLE_PSTN",
>-    ]
>   }
> }
> 
>@@ -142,8 +136,9 @@ if (rtc_include_tests) {
>       "../rtc_base:rtc_base",
>       "../rtc_base:rtc_base_approved",
>       "../rtc_base:rtc_base_tests_utils",
>+      "../rtc_base/third_party/sigslot",
>       "../test:test_support",
>-      "//testing/gmock",
>+      "//third_party/abseil-cpp/absl/memory",
>     ]
>   }
> 
>@@ -152,12 +147,14 @@ if (rtc_include_tests) {
> 
>     sources = [
>       "base/asyncstuntcpsocket_unittest.cc",
>+      "base/basicasyncresolverfactory_unittest.cc",
>       "base/dtlstransport_unittest.cc",
>       "base/p2ptransportchannel_unittest.cc",
>       "base/packetlossestimator_unittest.cc",
>       "base/port_unittest.cc",
>       "base/portallocator_unittest.cc",
>       "base/pseudotcp_unittest.cc",
>+      "base/regatheringcontroller_unittest.cc",
>       "base/relayport_unittest.cc",
>       "base/relayserver_unittest.cc",
>       "base/stun_unittest.cc",
>@@ -174,22 +171,21 @@ if (rtc_include_tests) {
>     deps = [
>       ":p2p_test_utils",
>       ":rtc_p2p",
>-      "../api:fakemetricsobserver",
>       "../api:ortc_api",
>       "../rtc_base:checks",
>       "../rtc_base:rtc_base",
>       "../rtc_base:rtc_base_approved",
>       "../rtc_base:rtc_base_tests_utils",
>       "../rtc_base:stringutils",
>+      "../system_wrappers:metrics_default",
>       "../test:test_support",
>-      "//testing/gmock",
>       "//testing/gtest",
>+      "//third_party/abseil-cpp/absl/memory",
>     ]
>     if (!build_with_chromium && is_clang) {
>       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>     }
>-    defines = [ "GTEST_RELATIVE_PATH" ]
>   }
> }
> 
>@@ -203,7 +199,6 @@ rtc_static_library("libstunprober") {
>   deps = [
>     ":rtc_p2p",
>     "..:webrtc_common",
>-    "../:typedefs",
>     "../rtc_base:checks",
>     "../rtc_base:rtc_base",
>   ]
>@@ -223,9 +218,7 @@ if (rtc_include_tests) {
>       "../rtc_base:checks",
>       "../rtc_base:rtc_base",
>       "../rtc_base:rtc_base_tests_utils",
>-      "//testing/gmock",
>       "//testing/gtest",
>     ]
>-    defines = [ "GTEST_RELATIVE_PATH" ]
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/DEPS b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/DEPS
>index 94f4bcb2add..8243179d409 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/DEPS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/DEPS
>@@ -1,4 +1,5 @@
> include_rules = [
>+  "+logging",
>   "+net",
>   "+system_wrappers",
> ]
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/OWNERS
>index 26bedddc005..18f6044b163 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/OWNERS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/OWNERS
>@@ -1,15 +1,13 @@
>-henrika@webrtc.org
>-henrikg@webrtc.org
>+honghaiz@webrtc.org
> hta@webrtc.org
> jiayl@webrtc.org
> juberti@webrtc.org
> mflodman@webrtc.org
> perkj@webrtc.org
> pthatcher@webrtc.org
>+qingsi@webrtc.org
> sergeyu@chromium.org
> tommi@webrtc.org
>-deadbeef@webrtc.org
>-honghaiz@webrtc.org
> 
> # These are for the common case of adding or renaming files. If you're doing
> # structural changes, please get a review from a reviewer in this file.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/asyncstuntcpsocket.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/asyncstuntcpsocket.cc
>index ef9d3c5f13a..8bb796fd1d5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/asyncstuntcpsocket.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/asyncstuntcpsocket.cc
>@@ -39,16 +39,16 @@ AsyncStunTCPSocket* AsyncStunTCPSocket::Create(
>     rtc::AsyncSocket* socket,
>     const rtc::SocketAddress& bind_address,
>     const rtc::SocketAddress& remote_address) {
>-  return new AsyncStunTCPSocket(AsyncTCPSocketBase::ConnectSocket(
>-      socket, bind_address, remote_address), false);
>+  return new AsyncStunTCPSocket(
>+      AsyncTCPSocketBase::ConnectSocket(socket, bind_address, remote_address),
>+      false);
> }
> 
>-AsyncStunTCPSocket::AsyncStunTCPSocket(
>-    rtc::AsyncSocket* socket, bool listen)
>-    : rtc::AsyncTCPSocketBase(socket, listen, kBufSize) {
>-}
>+AsyncStunTCPSocket::AsyncStunTCPSocket(rtc::AsyncSocket* socket, bool listen)
>+    : rtc::AsyncTCPSocketBase(socket, listen, kBufSize) {}
> 
>-int AsyncStunTCPSocket::Send(const void *pv, size_t cb,
>+int AsyncStunTCPSocket::Send(const void* pv,
>+                             size_t cb,
>                              const rtc::PacketOptions& options) {
>   if (cb > kBufSize || cb < kPacketLenSize + kPacketLenOffset) {
>     SetError(EMSGSIZE);
>@@ -121,12 +121,12 @@ void AsyncStunTCPSocket::ProcessInput(char* data, size_t* len) {
>   }
> }
> 
>-void AsyncStunTCPSocket::HandleIncomingConnection(
>-    rtc::AsyncSocket* socket) {
>+void AsyncStunTCPSocket::HandleIncomingConnection(rtc::AsyncSocket* socket) {
>   SignalNewConnection(this, new AsyncStunTCPSocket(socket, false));
> }
> 
>-size_t AsyncStunTCPSocket::GetExpectedLength(const void* data, size_t len,
>+size_t AsyncStunTCPSocket::GetExpectedLength(const void* data,
>+                                             size_t len,
>                                              int* pad_bytes) {
>   *pad_bytes = 0;
>   PacketLength pkt_len =
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/asyncstuntcpsocket.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/asyncstuntcpsocket.h
>index 15ca911b715..a299e611450 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/asyncstuntcpsocket.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/asyncstuntcpsocket.h
>@@ -22,10 +22,9 @@ class AsyncStunTCPSocket : public rtc::AsyncTCPSocketBase {
>   // Binds and connects |socket| and creates AsyncTCPSocket for
>   // it. Takes ownership of |socket|. Returns NULL if bind() or
>   // connect() fail (|socket| is destroyed in that case).
>-  static AsyncStunTCPSocket* Create(
>-      rtc::AsyncSocket* socket,
>-      const rtc::SocketAddress& bind_address,
>-      const rtc::SocketAddress& remote_address);
>+  static AsyncStunTCPSocket* Create(rtc::AsyncSocket* socket,
>+                                    const rtc::SocketAddress& bind_address,
>+                                    const rtc::SocketAddress& remote_address);
> 
>   AsyncStunTCPSocket(rtc::AsyncSocket* socket, bool listen);
> 
>@@ -39,8 +38,7 @@ class AsyncStunTCPSocket : public rtc::AsyncTCPSocketBase {
>   // This method returns the message hdr + length written in the header.
>   // This method also returns the number of padding bytes needed/added to the
>   // turn message. |pad_bytes| should be used only when |is_turn| is true.
>-  size_t GetExpectedLength(const void* data, size_t len,
>-                           int* pad_bytes);
>+  size_t GetExpectedLength(const void* data, size_t len, int* pad_bytes);
> 
>   RTC_DISALLOW_COPY_AND_ASSIGN(AsyncStunTCPSocket);
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/asyncstuntcpsocket_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/asyncstuntcpsocket_unittest.cc
>index c0f7fc0994c..de9541941f4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/asyncstuntcpsocket_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/asyncstuntcpsocket_unittest.cc
>@@ -20,49 +20,34 @@
> namespace cricket {
> 
> static unsigned char kStunMessageWithZeroLength[] = {
>-  0x00, 0x01, 0x00, 0x00,  // length of 0 (last 2 bytes)
>-  0x21, 0x12, 0xA4, 0x42,
>-  '0', '1', '2', '3',
>-  '4', '5', '6', '7',
>-  '8', '9', 'a', 'b',
>+    0x00, 0x01, 0x00, 0x00,  // length of 0 (last 2 bytes)
>+    0x21, 0x12, 0xA4, 0x42, '0', '1', '2', '3',
>+    '4',  '5',  '6',  '7',  '8', '9', 'a', 'b',
> };
> 
>-
> static unsigned char kTurnChannelDataMessageWithZeroLength[] = {
>-  0x40, 0x00, 0x00, 0x00,  // length of 0 (last 2 bytes)
>+    0x40, 0x00, 0x00, 0x00,  // length of 0 (last 2 bytes)
> };
> 
> static unsigned char kTurnChannelDataMessage[] = {
>-  0x40, 0x00, 0x00, 0x10,
>-  0x21, 0x12, 0xA4, 0x42,
>-  '0', '1', '2', '3',
>-  '4', '5', '6', '7',
>-  '8', '9', 'a', 'b',
>+    0x40, 0x00, 0x00, 0x10, 0x21, 0x12, 0xA4, 0x42, '0', '1',
>+    '2',  '3',  '4',  '5',  '6',  '7',  '8',  '9',  'a', 'b',
> };
> 
> static unsigned char kStunMessageWithInvalidLength[] = {
>-  0x00, 0x01, 0x00, 0x10,
>-  0x21, 0x12, 0xA4, 0x42,
>-  '0', '1', '2', '3',
>-  '4', '5', '6', '7',
>-  '8', '9', 'a', 'b',
>+    0x00, 0x01, 0x00, 0x10, 0x21, 0x12, 0xA4, 0x42, '0', '1',
>+    '2',  '3',  '4',  '5',  '6',  '7',  '8',  '9',  'a', 'b',
> };
> 
> static unsigned char kTurnChannelDataMessageWithInvalidLength[] = {
>-  0x80, 0x00, 0x00, 0x20,
>-  0x21, 0x12, 0xA4, 0x42,
>-  '0', '1', '2', '3',
>-  '4', '5', '6', '7',
>-  '8', '9', 'a', 'b',
>+    0x80, 0x00, 0x00, 0x20, 0x21, 0x12, 0xA4, 0x42, '0', '1',
>+    '2',  '3',  '4',  '5',  '6',  '7',  '8',  '9',  'a', 'b',
> };
> 
> static unsigned char kTurnChannelDataMessageWithOddLength[] = {
>-  0x40, 0x00, 0x00, 0x05,
>-  0x21, 0x12, 0xA4, 0x42,
>-  '0',
>+    0x40, 0x00, 0x00, 0x05, 0x21, 0x12, 0xA4, 0x42, '0',
> };
> 
>-
> static const rtc::SocketAddress kClientAddr("11.11.11.11", 0);
> static const rtc::SocketAddress kServerAddr("22.22.22.22", 0);
> 
>@@ -72,20 +57,18 @@ class AsyncStunTCPSocketTest : public testing::Test,
>   AsyncStunTCPSocketTest()
>       : vss_(new rtc::VirtualSocketServer()), thread_(vss_.get()) {}
> 
>-  virtual void SetUp() {
>-    CreateSockets();
>-  }
>+  virtual void SetUp() { CreateSockets(); }
> 
>   void CreateSockets() {
>-    rtc::AsyncSocket* server = vss_->CreateAsyncSocket(
>-        kServerAddr.family(), SOCK_STREAM);
>+    rtc::AsyncSocket* server =
>+        vss_->CreateAsyncSocket(kServerAddr.family(), SOCK_STREAM);
>     server->Bind(kServerAddr);
>     recv_socket_.reset(new AsyncStunTCPSocket(server, true));
>     recv_socket_->SignalNewConnection.connect(
>         this, &AsyncStunTCPSocketTest::OnNewConnection);
> 
>-    rtc::AsyncSocket* client = vss_->CreateAsyncSocket(
>-        kClientAddr.family(), SOCK_STREAM);
>+    rtc::AsyncSocket* client =
>+        vss_->CreateAsyncSocket(kClientAddr.family(), SOCK_STREAM);
>     send_socket_.reset(AsyncStunTCPSocket::Create(
>         client, kClientAddr, recv_socket_->GetLocalAddress()));
>     send_socket_->SignalSentPacket.connect(
>@@ -94,8 +77,10 @@ class AsyncStunTCPSocketTest : public testing::Test,
>     vss_->ProcessMessagesUntilIdle();
>   }
> 
>-  void OnReadPacket(rtc::AsyncPacketSocket* socket, const char* data,
>-                    size_t len, const rtc::SocketAddress& remote_addr,
>+  void OnReadPacket(rtc::AsyncPacketSocket* socket,
>+                    const char* data,
>+                    size_t len,
>+                    const rtc::SocketAddress& remote_addr,
>                     const rtc::PacketTime& packet_time) {
>     recv_packets_.push_back(std::string(data, len));
>   }
>@@ -108,14 +93,14 @@ class AsyncStunTCPSocketTest : public testing::Test,
>   void OnNewConnection(rtc::AsyncPacketSocket* server,
>                        rtc::AsyncPacketSocket* new_socket) {
>     listen_socket_.reset(new_socket);
>-    new_socket->SignalReadPacket.connect(
>-        this, &AsyncStunTCPSocketTest::OnReadPacket);
>+    new_socket->SignalReadPacket.connect(this,
>+                                         &AsyncStunTCPSocketTest::OnReadPacket);
>   }
> 
>   bool Send(const void* data, size_t len) {
>     rtc::PacketOptions options;
>-    size_t ret = send_socket_->Send(
>-        reinterpret_cast<const char*>(data), len, options);
>+    size_t ret =
>+        send_socket_->Send(reinterpret_cast<const char*>(data), len, options);
>     vss_->ProcessMessagesUntilIdle();
>     return (ret == len);
>   }
>@@ -123,7 +108,7 @@ class AsyncStunTCPSocketTest : public testing::Test,
>   bool CheckData(const void* data, int len) {
>     bool ret = false;
>     if (recv_packets_.size()) {
>-      std::string packet =  recv_packets_.front();
>+      std::string packet = recv_packets_.front();
>       recv_packets_.pop_front();
>       ret = (memcmp(data, packet.c_str(), len) == 0);
>     }
>@@ -141,8 +126,8 @@ class AsyncStunTCPSocketTest : public testing::Test,
> 
> // Testing a stun packet sent/recv properly.
> TEST_F(AsyncStunTCPSocketTest, TestSingleStunPacket) {
>-  EXPECT_TRUE(Send(kStunMessageWithZeroLength,
>-                   sizeof(kStunMessageWithZeroLength)));
>+  EXPECT_TRUE(
>+      Send(kStunMessageWithZeroLength, sizeof(kStunMessageWithZeroLength)));
>   EXPECT_EQ(1u, recv_packets_.size());
>   EXPECT_TRUE(CheckData(kStunMessageWithZeroLength,
>                         sizeof(kStunMessageWithZeroLength)));
>@@ -150,14 +135,14 @@ TEST_F(AsyncStunTCPSocketTest, TestSingleStunPacket) {
> 
> // Verify sending multiple packets.
> TEST_F(AsyncStunTCPSocketTest, TestMultipleStunPackets) {
>-  EXPECT_TRUE(Send(kStunMessageWithZeroLength,
>-                   sizeof(kStunMessageWithZeroLength)));
>-  EXPECT_TRUE(Send(kStunMessageWithZeroLength,
>-                   sizeof(kStunMessageWithZeroLength)));
>-  EXPECT_TRUE(Send(kStunMessageWithZeroLength,
>-                   sizeof(kStunMessageWithZeroLength)));
>-  EXPECT_TRUE(Send(kStunMessageWithZeroLength,
>-                   sizeof(kStunMessageWithZeroLength)));
>+  EXPECT_TRUE(
>+      Send(kStunMessageWithZeroLength, sizeof(kStunMessageWithZeroLength)));
>+  EXPECT_TRUE(
>+      Send(kStunMessageWithZeroLength, sizeof(kStunMessageWithZeroLength)));
>+  EXPECT_TRUE(
>+      Send(kStunMessageWithZeroLength, sizeof(kStunMessageWithZeroLength)));
>+  EXPECT_TRUE(
>+      Send(kStunMessageWithZeroLength, sizeof(kStunMessageWithZeroLength)));
>   EXPECT_EQ(4u, recv_packets_.size());
> }
> 
>@@ -172,11 +157,10 @@ TEST_F(AsyncStunTCPSocketTest, TestTurnChannelDataWithZeroLength) {
> 
> // Verifying TURN channel data message.
> TEST_F(AsyncStunTCPSocketTest, TestTurnChannelData) {
>-  EXPECT_TRUE(Send(kTurnChannelDataMessage,
>-                   sizeof(kTurnChannelDataMessage)));
>+  EXPECT_TRUE(Send(kTurnChannelDataMessage, sizeof(kTurnChannelDataMessage)));
>   EXPECT_EQ(1u, recv_packets_.size());
>-  EXPECT_TRUE(CheckData(kTurnChannelDataMessage,
>-                        sizeof(kTurnChannelDataMessage)));
>+  EXPECT_TRUE(
>+      CheckData(kTurnChannelDataMessage, sizeof(kTurnChannelDataMessage)));
> }
> 
> // Verifying TURN channel messages which needs padding handled properly.
>@@ -210,18 +194,18 @@ TEST_F(AsyncStunTCPSocketTest, TestStunInvalidLength) {
> // Verifying TURN channel data message with invalid length.
> TEST_F(AsyncStunTCPSocketTest, TestTurnChannelDataWithInvalidLength) {
>   EXPECT_FALSE(Send(kTurnChannelDataMessageWithInvalidLength,
>-                   sizeof(kTurnChannelDataMessageWithInvalidLength)));
>+                    sizeof(kTurnChannelDataMessageWithInvalidLength)));
>   // Modify the length to larger value.
>   kTurnChannelDataMessageWithInvalidLength[2] = 0xFF;
>   kTurnChannelDataMessageWithInvalidLength[3] = 0xF0;
>   EXPECT_FALSE(Send(kTurnChannelDataMessageWithInvalidLength,
>-                   sizeof(kTurnChannelDataMessageWithInvalidLength)));
>+                    sizeof(kTurnChannelDataMessageWithInvalidLength)));
> 
>   // Modify the length to smaller value.
>   kTurnChannelDataMessageWithInvalidLength[2] = 0x00;
>   kTurnChannelDataMessageWithInvalidLength[3] = 0x00;
>   EXPECT_FALSE(Send(kTurnChannelDataMessageWithInvalidLength,
>-                   sizeof(kTurnChannelDataMessageWithInvalidLength)));
>+                    sizeof(kTurnChannelDataMessageWithInvalidLength)));
> }
> 
> // Verifying a small buffer handled (dropped) properly. This will be
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/umametrics.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/basicasyncresolverfactory.cc
>similarity index 59%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/api/umametrics.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/basicasyncresolverfactory.cc
>index d5f2bb62b27..22dac685104 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/umametrics.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/basicasyncresolverfactory.cc
>@@ -1,5 +1,5 @@
> /*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>  *
>  *  Use of this source code is governed by a BSD-style license
>  *  that can be found in the LICENSE file in the root of the source
>@@ -8,14 +8,14 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "api/umametrics.h"
>+#include "p2p/base/basicasyncresolverfactory.h"
>+
>+#include "rtc_base/nethelpers.h"
> 
> namespace webrtc {
> 
>-void MetricsObserverInterface::IncrementSparseEnumCounter(
>-    PeerConnectionEnumCounterType type,
>-    int counter) {
>-  IncrementEnumCounter(type, counter, 0 /* Ignored */);
>+rtc::AsyncResolverInterface* BasicAsyncResolverFactory::Create() {
>+  return new rtc::AsyncResolver();
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/basicasyncresolverfactory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/basicasyncresolverfactory.h
>new file mode 100644
>index 00000000000..29fe6d005d1
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/basicasyncresolverfactory.h
>@@ -0,0 +1,25 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef P2P_BASE_BASICASYNCRESOLVERFACTORY_H_
>+#define P2P_BASE_BASICASYNCRESOLVERFACTORY_H_
>+
>+#include "api/asyncresolverfactory.h"
>+
>+namespace webrtc {
>+
>+class BasicAsyncResolverFactory : public AsyncResolverFactory {
>+ public:
>+  rtc::AsyncResolverInterface* Create() override;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // P2P_BASE_BASICASYNCRESOLVERFACTORY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/basicasyncresolverfactory_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/basicasyncresolverfactory_unittest.cc
>new file mode 100644
>index 00000000000..4529d5eb198
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/basicasyncresolverfactory_unittest.cc
>@@ -0,0 +1,45 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "p2p/base/basicasyncresolverfactory.h"
>+#include "rtc_base/gunit.h"
>+
>+namespace webrtc {
>+
>+class BasicAsyncResolverFactoryTest : public testing::Test,
>+                                      public sigslot::has_slots<> {
>+ public:
>+  void TestCreate() {
>+    BasicAsyncResolverFactory factory;
>+    rtc::AsyncResolverInterface* resolver = factory.Create();
>+    ASSERT_TRUE(resolver);
>+    resolver->SignalDone.connect(
>+        this, &BasicAsyncResolverFactoryTest::SetAddressResolved);
>+
>+    rtc::SocketAddress address("", 0);
>+    resolver->Start(address);
>+    ASSERT_TRUE_WAIT(address_resolved_, 10000 /*ms*/);
>+  }
>+
>+  void SetAddressResolved(rtc::AsyncResolverInterface* resolver) {
>+    address_resolved_ = true;
>+  }
>+
>+ private:
>+  bool address_resolved_ = false;
>+};
>+
>+// This test is primarily intended to let tools check that the created resolver
>+// doesn't leak.
>+TEST_F(BasicAsyncResolverFactoryTest, TestCreate) {
>+  TestCreate();
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/basicpacketsocketfactory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/basicpacketsocketfactory.cc
>index 86b9e8541ce..b7eb8ed1c6d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/basicpacketsocketfactory.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/basicpacketsocketfactory.cc
>@@ -27,23 +27,16 @@
> namespace rtc {
> 
> BasicPacketSocketFactory::BasicPacketSocketFactory()
>-    : thread_(Thread::Current()),
>-      socket_factory_(NULL) {
>-}
>+    : thread_(Thread::Current()), socket_factory_(NULL) {}
> 
> BasicPacketSocketFactory::BasicPacketSocketFactory(Thread* thread)
>-    : thread_(thread),
>-      socket_factory_(NULL) {
>-}
>+    : thread_(thread), socket_factory_(NULL) {}
> 
> BasicPacketSocketFactory::BasicPacketSocketFactory(
>     SocketFactory* socket_factory)
>-    : thread_(NULL),
>-      socket_factory_(socket_factory) {
>-}
>+    : thread_(NULL), socket_factory_(socket_factory) {}
> 
>-BasicPacketSocketFactory::~BasicPacketSocketFactory() {
>-}
>+BasicPacketSocketFactory::~BasicPacketSocketFactory() {}
> 
> AsyncPacketSocket* BasicPacketSocketFactory::CreateUdpSocket(
>     const SocketAddress& address,
>@@ -170,6 +163,7 @@ AsyncPacketSocket* BasicPacketSocketFactory::CreateClientTcpSocket(
> 
>     ssl_adapter->SetAlpnProtocols(tcp_options.tls_alpn_protocols);
>     ssl_adapter->SetEllipticCurves(tcp_options.tls_elliptic_curves);
>+    ssl_adapter->SetCertVerifier(tcp_options.tls_cert_verifier);
> 
>     socket = ssl_adapter;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/common.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/common.h
>deleted file mode 100644
>index aa5da42d8e2..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/common.h
>+++ /dev/null
>@@ -1,20 +0,0 @@
>-/*
>- *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef P2P_BASE_COMMON_H_
>-#define P2P_BASE_COMMON_H_
>-
>-#include "rtc_base/logging.h"
>-
>-// Common log description format for jingle messages
>-#define LOG_J(sev, obj) RTC_LOG(sev) << "Jingle:" << obj->ToString() << ": "
>-#define LOG_JV(sev, obj) RTC_LOG_V(sev) << "Jingle:" << obj->ToString() << ": "
>-
>-#endif  // P2P_BASE_COMMON_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/dtlstransport.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/dtlstransport.cc
>index 179b032adcb..314f63ce74f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/dtlstransport.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/dtlstransport.cc
>@@ -14,11 +14,11 @@
> 
> #include "p2p/base/dtlstransport.h"
> 
>-#include "p2p/base/common.h"
> #include "p2p/base/packettransportinternal.h"
> #include "rtc_base/buffer.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/dscp.h"
>+#include "rtc_base/logging.h"
> #include "rtc_base/messagequeue.h"
> #include "rtc_base/sslstreamadapter.h"
> #include "rtc_base/stream.h"
>@@ -117,24 +117,28 @@ DtlsTransport::DtlsTransport(IceTransportInternal* ice_transport,
>                              const rtc::CryptoOptions& crypto_options)
>     : transport_name_(ice_transport->transport_name()),
>       component_(ice_transport->component()),
>-      network_thread_(rtc::Thread::Current()),
>       ice_transport_(ice_transport),
>       downward_(NULL),
>       srtp_ciphers_(GetSupportedDtlsSrtpCryptoSuites(crypto_options)),
>-      ssl_role_(rtc::SSL_CLIENT),
>       ssl_max_version_(rtc::SSL_PROTOCOL_DTLS_12),
>       crypto_options_(crypto_options) {
>   RTC_DCHECK(ice_transport_);
>-  ice_transport_->SignalWritableState.connect(this,
>-                                              &DtlsTransport::OnWritableState);
>-  ice_transport_->SignalReadPacket.connect(this, &DtlsTransport::OnReadPacket);
>-  ice_transport_->SignalSentPacket.connect(this, &DtlsTransport::OnSentPacket);
>-  ice_transport_->SignalReadyToSend.connect(this,
>-                                            &DtlsTransport::OnReadyToSend);
>-  ice_transport_->SignalReceivingState.connect(
>-      this, &DtlsTransport::OnReceivingState);
>-  ice_transport_->SignalNetworkRouteChanged.connect(
>-      this, &DtlsTransport::OnNetworkRouteChanged);
>+  ConnectToIceTransport();
>+}
>+
>+DtlsTransport::DtlsTransport(
>+    std::unique_ptr<IceTransportInternal> ice_transport,
>+    const rtc::CryptoOptions& crypto_options)
>+    : transport_name_(ice_transport->transport_name()),
>+      component_(ice_transport->component()),
>+      ice_transport_(ice_transport.get()),
>+      owned_ice_transport_(std::move(ice_transport)),
>+      downward_(NULL),
>+      srtp_ciphers_(GetSupportedDtlsSrtpCryptoSuites(crypto_options)),
>+      ssl_max_version_(rtc::SSL_PROTOCOL_DTLS_12),
>+      crypto_options_(crypto_options) {
>+  RTC_DCHECK(owned_ice_transport_);
>+  ConnectToIceTransport();
> }
> 
> DtlsTransport::~DtlsTransport() = default;
>@@ -164,10 +168,11 @@ bool DtlsTransport::SetLocalCertificate(
>   if (dtls_active_) {
>     if (certificate == local_certificate_) {
>       // This may happen during renegotiation.
>-      LOG_J(LS_INFO, this) << "Ignoring identical DTLS identity";
>+      RTC_LOG(LS_INFO) << ToString() << ": Ignoring identical DTLS identity";
>       return true;
>     } else {
>-      LOG_J(LS_ERROR, this) << "Can't change DTLS local identity in this state";
>+      RTC_LOG(LS_ERROR) << ToString()
>+                        << ": Can't change DTLS local identity in this state";
>       return false;
>     }
>   }
>@@ -176,7 +181,8 @@ bool DtlsTransport::SetLocalCertificate(
>     local_certificate_ = certificate;
>     dtls_active_ = true;
>   } else {
>-    LOG_J(LS_INFO, this) << "NULL DTLS identity supplied. Not doing DTLS";
>+    RTC_LOG(LS_INFO) << ToString()
>+                     << ": NULL DTLS identity supplied. Not doing DTLS";
>   }
> 
>   return true;
>@@ -190,7 +196,7 @@ rtc::scoped_refptr<rtc::RTCCertificate> DtlsTransport::GetLocalCertificate()
> bool DtlsTransport::SetSslMaxProtocolVersion(rtc::SSLProtocolVersion version) {
>   if (dtls_active_) {
>     RTC_LOG(LS_ERROR) << "Not changing max. protocol version "
>-                      << "while DTLS is negotiating";
>+                         "while DTLS is negotiating";
>     return false;
>   }
> 
>@@ -198,9 +204,10 @@ bool DtlsTransport::SetSslMaxProtocolVersion(rtc::SSLProtocolVersion version) {
>   return true;
> }
> 
>-bool DtlsTransport::SetSslRole(rtc::SSLRole role) {
>+bool DtlsTransport::SetDtlsRole(rtc::SSLRole role) {
>   if (dtls_) {
>-    if (ssl_role_ != role) {
>+    RTC_DCHECK(dtls_role_);
>+    if (*dtls_role_ != role) {
>       RTC_LOG(LS_ERROR)
>           << "SSL Role can't be reversed after the session is setup.";
>       return false;
>@@ -208,12 +215,15 @@ bool DtlsTransport::SetSslRole(rtc::SSLRole role) {
>     return true;
>   }
> 
>-  ssl_role_ = role;
>+  dtls_role_ = std::move(role);
>   return true;
> }
> 
>-bool DtlsTransport::GetSslRole(rtc::SSLRole* role) const {
>-  *role = ssl_role_;
>+bool DtlsTransport::GetDtlsRole(rtc::SSLRole* role) const {
>+  if (!dtls_role_) {
>+    return false;
>+  }
>+  *role = *dtls_role_;
>   return true;
> }
> 
>@@ -235,7 +245,8 @@ bool DtlsTransport::SetRemoteFingerprint(const std::string& digest_alg,
>   if (dtls_active_ && remote_fingerprint_value_ == remote_fingerprint_value &&
>       !digest_alg.empty()) {
>     // This may happen during renegotiation.
>-    LOG_J(LS_INFO, this) << "Ignoring identical remote DTLS fingerprint";
>+    RTC_LOG(LS_INFO) << ToString()
>+                     << ": Ignoring identical remote DTLS fingerprint";
>     return true;
>   }
> 
>@@ -246,7 +257,7 @@ bool DtlsTransport::SetRemoteFingerprint(const std::string& digest_alg,
>   // SetRemoteFingerprint shouldn't even be called though.
>   if (digest_alg.empty()) {
>     RTC_DCHECK(!digest_len);
>-    LOG_J(LS_INFO, this) << "Other side didn't support DTLS.";
>+    RTC_LOG(LS_INFO) << ToString() << ": Other side didn't support DTLS.";
>     dtls_active_ = false;
>     return true;
>   }
>@@ -254,7 +265,8 @@ bool DtlsTransport::SetRemoteFingerprint(const std::string& digest_alg,
>   // Otherwise, we must have a local certificate before setting remote
>   // fingerprint.
>   if (!dtls_active_) {
>-    LOG_J(LS_ERROR, this) << "Can't set DTLS remote settings in this state.";
>+    RTC_LOG(LS_ERROR) << ToString()
>+                      << ": Can't set DTLS remote settings in this state.";
>     return false;
>   }
> 
>@@ -272,7 +284,8 @@ bool DtlsTransport::SetRemoteFingerprint(const std::string& digest_alg,
>             remote_fingerprint_algorithm_,
>             reinterpret_cast<unsigned char*>(remote_fingerprint_value_.data()),
>             remote_fingerprint_value_.size(), &err)) {
>-      LOG_J(LS_ERROR, this) << "Couldn't set DTLS certificate digest.";
>+      RTC_LOG(LS_ERROR) << ToString()
>+                        << ": Couldn't set DTLS certificate digest.";
>       set_dtls_state(DTLS_TRANSPORT_FAILED);
>       // If the error is "verification failed", don't return false, because
>       // this means the fingerprint was formatted correctly but didn't match
>@@ -299,13 +312,13 @@ bool DtlsTransport::SetRemoteFingerprint(const std::string& digest_alg,
>   return true;
> }
> 
>-std::unique_ptr<rtc::SSLCertificate> DtlsTransport::GetRemoteSSLCertificate()
>+std::unique_ptr<rtc::SSLCertChain> DtlsTransport::GetRemoteSSLCertChain()
>     const {
>   if (!dtls_) {
>     return nullptr;
>   }
> 
>-  return dtls_->GetPeerCertificate();
>+  return dtls_->GetPeerSSLCertChain();
> }
> 
> bool DtlsTransport::ExportKeyingMaterial(const std::string& label,
>@@ -321,11 +334,12 @@ bool DtlsTransport::ExportKeyingMaterial(const std::string& label,
> }
> 
> bool DtlsTransport::SetupDtls() {
>+  RTC_DCHECK(dtls_role_);
>   StreamInterfaceChannel* downward = new StreamInterfaceChannel(ice_transport_);
> 
>   dtls_.reset(rtc::SSLStreamAdapter::Create(downward));
>   if (!dtls_) {
>-    LOG_J(LS_ERROR, this) << "Failed to create DTLS adapter.";
>+    RTC_LOG(LS_ERROR) << ToString() << ": Failed to create DTLS adapter.";
>     delete downward;
>     return false;
>   }
>@@ -335,7 +349,7 @@ bool DtlsTransport::SetupDtls() {
>   dtls_->SetIdentity(local_certificate_->identity()->GetReference());
>   dtls_->SetMode(rtc::SSL_MODE_DTLS);
>   dtls_->SetMaxProtocolVersion(ssl_max_version_);
>-  dtls_->SetServerRole(ssl_role_);
>+  dtls_->SetServerRole(*dtls_role_);
>   dtls_->SignalEvent.connect(this, &DtlsTransport::OnDtlsEvent);
>   dtls_->SignalSSLHandshakeError.connect(this,
>                                          &DtlsTransport::OnDtlsHandshakeError);
>@@ -344,21 +358,22 @@ bool DtlsTransport::SetupDtls() {
>           remote_fingerprint_algorithm_,
>           reinterpret_cast<unsigned char*>(remote_fingerprint_value_.data()),
>           remote_fingerprint_value_.size())) {
>-    LOG_J(LS_ERROR, this) << "Couldn't set DTLS certificate digest.";
>+    RTC_LOG(LS_ERROR) << ToString()
>+                      << ": Couldn't set DTLS certificate digest.";
>     return false;
>   }
> 
>   // Set up DTLS-SRTP, if it's been enabled.
>   if (!srtp_ciphers_.empty()) {
>     if (!dtls_->SetDtlsSrtpCryptoSuites(srtp_ciphers_)) {
>-      LOG_J(LS_ERROR, this) << "Couldn't set DTLS-SRTP ciphers.";
>+      RTC_LOG(LS_ERROR) << ToString() << ": Couldn't set DTLS-SRTP ciphers.";
>       return false;
>     }
>   } else {
>-    LOG_J(LS_INFO, this) << "Not using DTLS-SRTP.";
>+    RTC_LOG(LS_INFO) << ToString() << ": Not using DTLS-SRTP.";
>   }
> 
>-  LOG_J(LS_INFO, this) << "DTLS setup complete.";
>+  RTC_LOG(LS_INFO) << ToString() << ": DTLS setup complete.";
> 
>   // If the underlying ice_transport is already writable at this point, we may
>   // be able to start DTLS right away.
>@@ -435,7 +450,7 @@ int DtlsTransport::GetError() {
>   return ice_transport_->GetError();
> }
> 
>-rtc::Optional<rtc::NetworkRoute> DtlsTransport::network_route() const {
>+absl::optional<rtc::NetworkRoute> DtlsTransport::network_route() const {
>   return ice_transport_->network_route();
> }
> 
>@@ -447,6 +462,20 @@ int DtlsTransport::SetOption(rtc::Socket::Option opt, int value) {
>   return ice_transport_->SetOption(opt, value);
> }
> 
>+void DtlsTransport::ConnectToIceTransport() {
>+  RTC_DCHECK(ice_transport_);
>+  ice_transport_->SignalWritableState.connect(this,
>+                                              &DtlsTransport::OnWritableState);
>+  ice_transport_->SignalReadPacket.connect(this, &DtlsTransport::OnReadPacket);
>+  ice_transport_->SignalSentPacket.connect(this, &DtlsTransport::OnSentPacket);
>+  ice_transport_->SignalReadyToSend.connect(this,
>+                                            &DtlsTransport::OnReadyToSend);
>+  ice_transport_->SignalReceivingState.connect(
>+      this, &DtlsTransport::OnReceivingState);
>+  ice_transport_->SignalNetworkRouteChanged.connect(
>+      this, &DtlsTransport::OnNetworkRouteChanged);
>+}
>+
> // The state transition logic here is as follows:
> // (1) If we're not doing DTLS-SRTP, then the state is just the
> //     state of the underlying impl()
>@@ -458,11 +487,11 @@ int DtlsTransport::SetOption(rtc::Socket::Option opt, int value) {
> //     - Once the DTLS handshake completes, the state is that of the
> //       impl again
> void DtlsTransport::OnWritableState(rtc::PacketTransportInternal* transport) {
>-  RTC_DCHECK(rtc::Thread::Current() == network_thread_);
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>   RTC_DCHECK(transport == ice_transport_);
>-  LOG_J(LS_VERBOSE, this)
>-      << "DTLSTransportChannelWrapper: ice_transport writable state changed to "
>-      << ice_transport_->writable();
>+  RTC_LOG(LS_VERBOSE) << ToString()
>+                      << ": ice_transport writable state changed to "
>+                      << ice_transport_->writable();
> 
>   if (!dtls_active_) {
>     // Not doing DTLS.
>@@ -490,11 +519,12 @@ void DtlsTransport::OnWritableState(rtc::PacketTransportInternal* transport) {
> }
> 
> void DtlsTransport::OnReceivingState(rtc::PacketTransportInternal* transport) {
>-  RTC_DCHECK(rtc::Thread::Current() == network_thread_);
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>   RTC_DCHECK(transport == ice_transport_);
>-  LOG_J(LS_VERBOSE, this) << "DTLSTransportChannelWrapper: ice_transport "
>-                             "receiving state changed to "
>-                          << ice_transport_->receiving();
>+  RTC_LOG(LS_VERBOSE) << ToString()
>+                      << ": ice_transport "
>+                         "receiving state changed to "
>+                      << ice_transport_->receiving();
>   if (!dtls_active_ || dtls_state() == DTLS_TRANSPORT_CONNECTED) {
>     // Note: SignalReceivingState fired by set_receiving.
>     set_receiving(ice_transport_->receiving());
>@@ -506,7 +536,7 @@ void DtlsTransport::OnReadPacket(rtc::PacketTransportInternal* transport,
>                                  size_t size,
>                                  const rtc::PacketTime& packet_time,
>                                  int flags) {
>-  RTC_DCHECK(rtc::Thread::Current() == network_thread_);
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>   RTC_DCHECK(transport == ice_transport_);
>   RTC_DCHECK(flags == 0);
> 
>@@ -519,26 +549,30 @@ void DtlsTransport::OnReadPacket(rtc::PacketTransportInternal* transport,
>   switch (dtls_state()) {
>     case DTLS_TRANSPORT_NEW:
>       if (dtls_) {
>-        LOG_J(LS_INFO, this) << "Packet received before DTLS started.";
>+        RTC_LOG(LS_INFO) << ToString()
>+                         << ": Packet received before DTLS started.";
>       } else {
>-        LOG_J(LS_WARNING, this) << "Packet received before we know if we are "
>-                                << "doing DTLS or not.";
>+        RTC_LOG(LS_WARNING) << ToString()
>+                            << ": Packet received before we know if we are "
>+                               "doing DTLS or not.";
>       }
>       // Cache a client hello packet received before DTLS has actually started.
>       if (IsDtlsClientHelloPacket(data, size)) {
>-        LOG_J(LS_INFO, this) << "Caching DTLS ClientHello packet until DTLS is "
>-                             << "started.";
>+        RTC_LOG(LS_INFO) << ToString()
>+                         << ": Caching DTLS ClientHello packet until DTLS is "
>+                            "started.";
>         cached_client_hello_.SetData(data, size);
>         // If we haven't started setting up DTLS yet (because we don't have a
>         // remote fingerprint/role), we can use the client hello as a clue that
>         // the peer has chosen the client role, and proceed with the handshake.
>         // The fingerprint will be verified when it's set.
>         if (!dtls_ && local_certificate_) {
>-          SetSslRole(rtc::SSL_SERVER);
>+          SetDtlsRole(rtc::SSL_SERVER);
>           SetupDtls();
>         }
>       } else {
>-        LOG_J(LS_INFO, this) << "Not a DTLS ClientHello packet; dropping.";
>+        RTC_LOG(LS_INFO) << ToString()
>+                         << ": Not a DTLS ClientHello packet; dropping.";
>       }
>       break;
> 
>@@ -548,20 +582,22 @@ void DtlsTransport::OnReadPacket(rtc::PacketTransportInternal* transport,
>       // Is this potentially a DTLS packet?
>       if (IsDtlsPacket(data, size)) {
>         if (!HandleDtlsPacket(data, size)) {
>-          LOG_J(LS_ERROR, this) << "Failed to handle DTLS packet.";
>+          RTC_LOG(LS_ERROR) << ToString() << ": Failed to handle DTLS packet.";
>           return;
>         }
>       } else {
>         // Not a DTLS packet; our handshake should be complete by now.
>         if (dtls_state() != DTLS_TRANSPORT_CONNECTED) {
>-          LOG_J(LS_ERROR, this) << "Received non-DTLS packet before DTLS "
>-                                << "complete.";
>+          RTC_LOG(LS_ERROR) << ToString()
>+                            << ": Received non-DTLS packet before DTLS "
>+                               "complete.";
>           return;
>         }
> 
>         // And it had better be a SRTP packet.
>         if (!IsRtpPacket(data, size)) {
>-          LOG_J(LS_ERROR, this) << "Received unexpected non-DTLS packet.";
>+          RTC_LOG(LS_ERROR)
>+              << ToString() << ": Received unexpected non-DTLS packet.";
>           return;
>         }
> 
>@@ -581,23 +617,23 @@ void DtlsTransport::OnReadPacket(rtc::PacketTransportInternal* transport,
> 
> void DtlsTransport::OnSentPacket(rtc::PacketTransportInternal* transport,
>                                  const rtc::SentPacket& sent_packet) {
>-  RTC_DCHECK(rtc::Thread::Current() == network_thread_);
>-
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>   SignalSentPacket(this, sent_packet);
> }
> 
> void DtlsTransport::OnReadyToSend(rtc::PacketTransportInternal* transport) {
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>   if (writable()) {
>     SignalReadyToSend(this);
>   }
> }
> 
> void DtlsTransport::OnDtlsEvent(rtc::StreamInterface* dtls, int sig, int err) {
>-  RTC_DCHECK(rtc::Thread::Current() == network_thread_);
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>   RTC_DCHECK(dtls == dtls_.get());
>   if (sig & rtc::SE_OPEN) {
>     // This is the first time.
>-    LOG_J(LS_INFO, this) << "DTLS handshake complete.";
>+    RTC_LOG(LS_INFO) << ToString() << ": DTLS handshake complete.";
>     if (dtls_->GetState() == rtc::SS_OPEN) {
>       // The check for OPEN shouldn't be necessary but let's make
>       // sure we don't accidentally frob the state if it's closed.
>@@ -618,12 +654,13 @@ void DtlsTransport::OnDtlsEvent(rtc::StreamInterface* dtls, int sig, int err) {
>         SignalReadPacket(this, buf, read, rtc::CreatePacketTime(0), 0);
>       } else if (ret == rtc::SR_EOS) {
>         // Remote peer shut down the association with no error.
>-        LOG_J(LS_INFO, this) << "DTLS transport closed";
>+        RTC_LOG(LS_INFO) << ToString() << ": DTLS transport closed";
>         set_writable(false);
>         set_dtls_state(DTLS_TRANSPORT_CLOSED);
>       } else if (ret == rtc::SR_ERROR) {
>         // Remote peer shut down the association with an error.
>-        LOG_J(LS_INFO, this) << "DTLS transport error, code=" << read_error;
>+        RTC_LOG(LS_INFO) << ToString()
>+                         << ": DTLS transport error, code=" << read_error;
>         set_writable(false);
>         set_dtls_state(DTLS_TRANSPORT_FAILED);
>       }
>@@ -633,17 +670,18 @@ void DtlsTransport::OnDtlsEvent(rtc::StreamInterface* dtls, int sig, int err) {
>     RTC_DCHECK(sig == rtc::SE_CLOSE);  // SE_CLOSE should be by itself.
>     set_writable(false);
>     if (!err) {
>-      LOG_J(LS_INFO, this) << "DTLS transport closed";
>+      RTC_LOG(LS_INFO) << ToString() << ": DTLS transport closed";
>       set_dtls_state(DTLS_TRANSPORT_CLOSED);
>     } else {
>-      LOG_J(LS_INFO, this) << "DTLS transport error, code=" << err;
>+      RTC_LOG(LS_INFO) << ToString() << ": DTLS transport error, code=" << err;
>       set_dtls_state(DTLS_TRANSPORT_FAILED);
>     }
>   }
> }
> 
> void DtlsTransport::OnNetworkRouteChanged(
>-    rtc::Optional<rtc::NetworkRoute> network_route) {
>+    absl::optional<rtc::NetworkRoute> network_route) {
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>   SignalNetworkRouteChanged(network_route);
> }
> 
>@@ -659,24 +697,26 @@ void DtlsTransport::MaybeStartDtls() {
>       // ignore write errors, thus any errors must be because of
>       // configuration and therefore are our fault.
>       RTC_NOTREACHED() << "StartSSL failed.";
>-      LOG_J(LS_ERROR, this) << "Couldn't start DTLS handshake";
>+      RTC_LOG(LS_ERROR) << ToString() << ": Couldn't start DTLS handshake";
>       set_dtls_state(DTLS_TRANSPORT_FAILED);
>       return;
>     }
>-    LOG_J(LS_INFO, this) << "DtlsTransport: Started DTLS handshake";
>+    RTC_LOG(LS_INFO) << ToString() << ": DtlsTransport: Started DTLS handshake";
>     set_dtls_state(DTLS_TRANSPORT_CONNECTING);
>     // Now that the handshake has started, we can process a cached ClientHello
>     // (if one exists).
>     if (cached_client_hello_.size()) {
>-      if (ssl_role_ == rtc::SSL_SERVER) {
>-        LOG_J(LS_INFO, this) << "Handling cached DTLS ClientHello packet.";
>+      if (*dtls_role_ == rtc::SSL_SERVER) {
>+        RTC_LOG(LS_INFO) << ToString()
>+                         << ": Handling cached DTLS ClientHello packet.";
>         if (!HandleDtlsPacket(cached_client_hello_.data<char>(),
>                               cached_client_hello_.size())) {
>-          LOG_J(LS_ERROR, this) << "Failed to handle DTLS packet.";
>+          RTC_LOG(LS_ERROR) << ToString() << ": Failed to handle DTLS packet.";
>         }
>       } else {
>-        LOG_J(LS_WARNING, this) << "Discarding cached DTLS ClientHello packet "
>-                                << "because we don't have the server role.";
>+        RTC_LOG(LS_WARNING) << ToString()
>+                            << ": Discarding cached DTLS ClientHello packet "
>+                               "because we don't have the server role.";
>       }
>       cached_client_hello_.Clear();
>     }
>@@ -718,8 +758,7 @@ void DtlsTransport::set_writable(bool writable) {
>   if (writable_ == writable) {
>     return;
>   }
>-  LOG_J(LS_VERBOSE, this) << "set_writable from:" << writable_ << " to "
>-                          << writable;
>+  RTC_LOG(LS_VERBOSE) << ToString() << ": set_writable to: " << writable;
>   writable_ = writable;
>   if (writable_) {
>     SignalReadyToSend(this);
>@@ -731,8 +770,8 @@ void DtlsTransport::set_dtls_state(DtlsTransportState state) {
>   if (dtls_state_ == state) {
>     return;
>   }
>-  LOG_J(LS_VERBOSE, this) << "set_dtls_state from:" << dtls_state_ << " to "
>-                          << state;
>+  RTC_LOG(LS_VERBOSE) << ToString() << ": set_dtls_state from:" << dtls_state_
>+                      << " to " << state;
>   dtls_state_ = state;
>   SignalDtlsState(this, state);
> }
>@@ -743,19 +782,20 @@ void DtlsTransport::OnDtlsHandshakeError(rtc::SSLHandshakeError error) {
> 
> void DtlsTransport::ConfigureHandshakeTimeout() {
>   RTC_DCHECK(dtls_);
>-  rtc::Optional<int> rtt = ice_transport_->GetRttEstimate();
>+  absl::optional<int> rtt = ice_transport_->GetRttEstimate();
>   if (rtt) {
>     // Limit the timeout to a reasonable range in case the ICE RTT takes
>     // extreme values.
>     int initial_timeout = std::max(kMinHandshakeTimeout,
>                                    std::min(kMaxHandshakeTimeout, 2 * (*rtt)));
>-    LOG_J(LS_INFO, this) << "configuring DTLS handshake timeout "
>-                         << initial_timeout << " based on ICE RTT " << *rtt;
>+    RTC_LOG(LS_INFO) << ToString() << ": configuring DTLS handshake timeout "
>+                     << initial_timeout << " based on ICE RTT " << *rtt;
> 
>     dtls_->SetInitialRetransmissionTimeout(initial_timeout);
>   } else {
>-    LOG_J(LS_INFO, this)
>-        << "no RTT estimate - using default DTLS handshake timeout";
>+    RTC_LOG(LS_INFO)
>+        << ToString()
>+        << ": no RTT estimate - using default DTLS handshake timeout";
>   }
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/dtlstransport.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/dtlstransport.h
>index e7b0c5654a2..609f3d72150 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/dtlstransport.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/dtlstransport.h
>@@ -22,6 +22,7 @@
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/sslstreamadapter.h"
> #include "rtc_base/stream.h"
>+#include "rtc_base/thread_checker.h"
> 
> namespace rtc {
> class PacketTransportInternal;
>@@ -83,14 +84,21 @@ class StreamInterfaceChannel : public rtc::StreamInterface {
> //
> //   - The SSLStreamAdapter writes to downward_->Write() which translates it
> //     into packet writes on ice_transport_.
>+//
>+// This class is not thread safe; all methods must be called on the same thread
>+// as the constructor.
> class DtlsTransport : public DtlsTransportInternal {
>  public:
>   // |ice_transport| is the ICE transport this DTLS transport is wrapping.
>   //
>   // |crypto_options| are the options used for the DTLS handshake. This affects
>   // whether GCM crypto suites are negotiated.
>+  // TODO(zhihuang): Remove this once we switch to JsepTransportController.
>   explicit DtlsTransport(IceTransportInternal* ice_transport,
>                          const rtc::CryptoOptions& crypto_options);
>+  explicit DtlsTransport(std::unique_ptr<IceTransportInternal> ice_transport,
>+                         const rtc::CryptoOptions& crypto_options);
>+
>   ~DtlsTransport() override;
> 
>   const rtc::CryptoOptions& crypto_options() const override;
>@@ -114,7 +122,7 @@ class DtlsTransport : public DtlsTransportInternal {
>   rtc::scoped_refptr<rtc::RTCCertificate> GetLocalCertificate() const override;
> 
>   // SetRemoteFingerprint must be called after SetLocalCertificate, and any
>-  // other methods like SetSslRole. It's what triggers the actual DTLS setup.
>+  // other methods like SetDtlsRole. It's what triggers the actual DTLS setup.
>   // TODO(deadbeef): Rename to "Start" like in ORTC?
>   bool SetRemoteFingerprint(const std::string& digest_alg,
>                             const uint8_t* digest,
>@@ -128,20 +136,21 @@ class DtlsTransport : public DtlsTransportInternal {
> 
>   bool GetOption(rtc::Socket::Option opt, int* value) override;
> 
>-  virtual bool SetSslMaxProtocolVersion(rtc::SSLProtocolVersion version);
>+  bool SetSslMaxProtocolVersion(rtc::SSLProtocolVersion version) override;
> 
>   // Find out which DTLS-SRTP cipher was negotiated
>   bool GetSrtpCryptoSuite(int* cipher) override;
> 
>-  bool GetSslRole(rtc::SSLRole* role) const override;
>-  bool SetSslRole(rtc::SSLRole role) override;
>+  bool GetDtlsRole(rtc::SSLRole* role) const override;
>+  bool SetDtlsRole(rtc::SSLRole role) override;
> 
>   // Find out which DTLS cipher was negotiated
>   bool GetSslCipherSuite(int* cipher) override;
> 
>-  // Once DTLS has been established, this method retrieves the certificate in
>-  // use by the remote peer, for use in external identity verification.
>-  std::unique_ptr<rtc::SSLCertificate> GetRemoteSSLCertificate() const override;
>+  // Once DTLS has been established, this method retrieves the certificate
>+  // chain in use by the remote peer, for use in external identity
>+  // verification.
>+  std::unique_ptr<rtc::SSLCertChain> GetRemoteSSLCertChain() const override;
> 
>   // Once DTLS has established (i.e., this ice_transport is writable), this
>   // method extracts the keys negotiated during the DTLS handshake, for use in
>@@ -166,7 +175,7 @@ class DtlsTransport : public DtlsTransportInternal {
> 
>   int GetError() override;
> 
>-  rtc::Optional<rtc::NetworkRoute> network_route() const override;
>+  absl::optional<rtc::NetworkRoute> network_route() const override;
> 
>   int SetOption(rtc::Socket::Option opt, int value) override;
> 
>@@ -180,6 +189,8 @@ class DtlsTransport : public DtlsTransportInternal {
>   }
> 
>  private:
>+  void ConnectToIceTransport();
>+
>   void OnWritableState(rtc::PacketTransportInternal* transport);
>   void OnReadPacket(rtc::PacketTransportInternal* transport,
>                     const char* data,
>@@ -191,7 +202,7 @@ class DtlsTransport : public DtlsTransportInternal {
>   void OnReadyToSend(rtc::PacketTransportInternal* transport);
>   void OnReceivingState(rtc::PacketTransportInternal* transport);
>   void OnDtlsEvent(rtc::StreamInterface* stream_, int sig, int err);
>-  void OnNetworkRouteChanged(rtc::Optional<rtc::NetworkRoute> network_route);
>+  void OnNetworkRouteChanged(absl::optional<rtc::NetworkRoute> network_route);
>   bool SetupDtls();
>   void MaybeStartDtls();
>   bool HandleDtlsPacket(const char* data, size_t size);
>@@ -203,19 +214,21 @@ class DtlsTransport : public DtlsTransportInternal {
>   // Sets the DTLS state, signaling if necessary.
>   void set_dtls_state(DtlsTransportState state);
> 
>+  rtc::ThreadChecker thread_checker_;
>+
>   std::string transport_name_;
>   int component_;
>   DtlsTransportState dtls_state_ = DTLS_TRANSPORT_NEW;
>-  rtc::Thread* network_thread_;  // Everything should occur on this thread.
>   // Underlying ice_transport, not owned by this class.
>   IceTransportInternal* const ice_transport_;
>+  std::unique_ptr<IceTransportInternal> owned_ice_transport_;
>   std::unique_ptr<rtc::SSLStreamAdapter> dtls_;  // The DTLS stream
>   StreamInterfaceChannel*
>       downward_;  // Wrapper for ice_transport_, owned by dtls_.
>   std::vector<int> srtp_ciphers_;  // SRTP ciphers to use with DTLS.
>   bool dtls_active_ = false;
>   rtc::scoped_refptr<rtc::RTCCertificate> local_certificate_;
>-  rtc::SSLRole ssl_role_;
>+  absl::optional<rtc::SSLRole> dtls_role_;
>   rtc::SSLProtocolVersion ssl_max_version_;
>   rtc::CryptoOptions crypto_options_;
>   rtc::Buffer remote_fingerprint_value_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/dtlstransport_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/dtlstransport_unittest.cc
>index cac78647d6c..3cf423e8513 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/dtlstransport_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/dtlstransport_unittest.cc
>@@ -256,7 +256,7 @@ class DtlsTestClient : public sigslot::has_slots<> {
>                                     size_t size,
>                                     const rtc::PacketTime& time,
>                                     int flags) {
>-    // Flags shouldn't be set on the underlying TransportChannel packets.
>+    // Flags shouldn't be set on the underlying Transport packets.
>     ASSERT_EQ(0, flags);
> 
>     // Look at the handshake packets to see what role we played.
>@@ -291,15 +291,14 @@ class DtlsTestClient : public sigslot::has_slots<> {
>   rtc::SentPacket sent_packet_;
> };
> 
>-// Base class for DtlsTransportChannelTest and DtlsEventOrderingTest, which
>+// Base class for DtlsTransportTest and DtlsEventOrderingTest, which
> // inherit from different variants of testing::Test.
> //
> // Note that this test always uses a FakeClock, due to the |fake_clock_| member
> // variable.
>-class DtlsTransportChannelTestBase {
>+class DtlsTransportTestBase {
>  public:
>-  DtlsTransportChannelTestBase()
>-      : client1_("P1"), client2_("P2"), use_dtls_(false) {}
>+  DtlsTransportTestBase() : client1_("P1"), client2_("P2"), use_dtls_(false) {}
> 
>   void SetMaxProtocolVersions(rtc::SSLProtocolVersion c1,
>                               rtc::SSLProtocolVersion c2) {
>@@ -335,9 +334,9 @@ class DtlsTransportChannelTestBase {
> 
>     if (use_dtls_) {
>       // Check that we negotiated the right ciphers. Since GCM ciphers are not
>-      // negotiated by default, we should end up with SRTP_AES128_CM_SHA1_32.
>-      client1_.CheckSrtp(rtc::SRTP_AES128_CM_SHA1_32);
>-      client2_.CheckSrtp(rtc::SRTP_AES128_CM_SHA1_32);
>+      // negotiated by default, we should end up with SRTP_AES128_CM_SHA1_80.
>+      client1_.CheckSrtp(rtc::SRTP_AES128_CM_SHA1_80);
>+      client2_.CheckSrtp(rtc::SRTP_AES128_CM_SHA1_80);
>     } else {
>       // If DTLS isn't actually being used, GetSrtpCryptoSuite should return
>       // false.
>@@ -354,10 +353,10 @@ class DtlsTransportChannelTestBase {
>   void Negotiate(bool client1_server = true) {
>     client1_.SetupTransports(ICEROLE_CONTROLLING);
>     client2_.SetupTransports(ICEROLE_CONTROLLED);
>-    client1_.dtls_transport()->SetSslRole(client1_server ? rtc::SSL_SERVER
>-                                                         : rtc::SSL_CLIENT);
>-    client2_.dtls_transport()->SetSslRole(client1_server ? rtc::SSL_CLIENT
>-                                                         : rtc::SSL_SERVER);
>+    client1_.dtls_transport()->SetDtlsRole(client1_server ? rtc::SSL_SERVER
>+                                                          : rtc::SSL_CLIENT);
>+    client2_.dtls_transport()->SetDtlsRole(client1_server ? rtc::SSL_CLIENT
>+                                                          : rtc::SSL_SERVER);
>     if (client2_.certificate()) {
>       SetRemoteFingerprintFromCert(client1_.dtls_transport(),
>                                    client2_.certificate());
>@@ -380,22 +379,21 @@ class DtlsTransportChannelTestBase {
>   rtc::ScopedFakeClock fake_clock_;
>   DtlsTestClient client1_;
>   DtlsTestClient client2_;
>-  int channel_ct_;
>   bool use_dtls_;
>   rtc::SSLProtocolVersion ssl_expected_version_;
> };
> 
>-class DtlsTransportChannelTest : public DtlsTransportChannelTestBase,
>-                                 public ::testing::Test {};
>+class DtlsTransportTest : public DtlsTransportTestBase,
>+                          public ::testing::Test {};
> 
> // Connect without DTLS, and transfer RTP data.
>-TEST_F(DtlsTransportChannelTest, TestTransferRtp) {
>+TEST_F(DtlsTransportTest, TestTransferRtp) {
>   ASSERT_TRUE(Connect());
>   TestTransfer(1000, 100, /*srtp=*/false);
> }
> 
> // Test that the SignalSentPacket signal is wired up.
>-TEST_F(DtlsTransportChannelTest, TestSignalSentPacket) {
>+TEST_F(DtlsTransportTest, TestSignalSentPacket) {
>   ASSERT_TRUE(Connect());
>   // Sanity check default value (-1).
>   ASSERT_EQ(client1_.sent_packet().send_time_ms, -1);
>@@ -407,13 +405,13 @@ TEST_F(DtlsTransportChannelTest, TestSignalSentPacket) {
> }
> 
> // Connect without DTLS, and transfer SRTP data.
>-TEST_F(DtlsTransportChannelTest, TestTransferSrtp) {
>+TEST_F(DtlsTransportTest, TestTransferSrtp) {
>   ASSERT_TRUE(Connect());
>   TestTransfer(1000, 100, /*srtp=*/true);
> }
> 
> // Connect with DTLS, and transfer data over DTLS.
>-TEST_F(DtlsTransportChannelTest, TestTransferDtls) {
>+TEST_F(DtlsTransportTest, TestTransferDtls) {
>   PrepareDtls(rtc::KT_DEFAULT);
>   ASSERT_TRUE(Connect());
>   TestTransfer(1000, 100, /*srtp=*/false);
>@@ -423,7 +421,7 @@ TEST_F(DtlsTransportChannelTest, TestTransferDtls) {
> // Our DTLS implementation doesn't do this, but other implementations may;
> // see https://tools.ietf.org/html/rfc6347#section-4.1.1.
> // This has caused interoperability problems with ORTCLib in the past.
>-TEST_F(DtlsTransportChannelTest, TestTransferDtlsCombineRecords) {
>+TEST_F(DtlsTransportTest, TestTransferDtlsCombineRecords) {
>   PrepareDtls(rtc::KT_DEFAULT);
>   ASSERT_TRUE(Connect());
>   // Our DTLS implementation always sends one record per packet, so to simulate
>@@ -435,8 +433,8 @@ TEST_F(DtlsTransportChannelTest, TestTransferDtlsCombineRecords) {
>   TestTransfer(500, 100, /*srtp=*/false);
> }
> 
>-class DtlsTransportChannelVersionTest
>-    : public DtlsTransportChannelTestBase,
>+class DtlsTransportVersionTest
>+    : public DtlsTransportTestBase,
>       public ::testing::TestWithParam<
>           ::testing::tuple<rtc::SSLProtocolVersion, rtc::SSLProtocolVersion>> {
> };
>@@ -444,7 +442,7 @@ class DtlsTransportChannelVersionTest
> // Test that an acceptable cipher suite is negotiated when different versions
> // of DTLS are supported. Note that it's IsAcceptableCipher that does the actual
> // work.
>-TEST_P(DtlsTransportChannelVersionTest, TestCipherSuiteNegotiation) {
>+TEST_P(DtlsTransportVersionTest, TestCipherSuiteNegotiation) {
>   PrepareDtls(rtc::KT_DEFAULT);
>   SetMaxProtocolVersions(::testing::get<0>(GetParam()),
>                          ::testing::get<1>(GetParam()));
>@@ -454,14 +452,14 @@ TEST_P(DtlsTransportChannelVersionTest, TestCipherSuiteNegotiation) {
> // Will test every combination of 1.0/1.2 on the client and server.
> INSTANTIATE_TEST_CASE_P(
>     TestCipherSuiteNegotiation,
>-    DtlsTransportChannelVersionTest,
>+    DtlsTransportVersionTest,
>     ::testing::Combine(::testing::Values(rtc::SSL_PROTOCOL_DTLS_10,
>                                          rtc::SSL_PROTOCOL_DTLS_12),
>                        ::testing::Values(rtc::SSL_PROTOCOL_DTLS_10,
>                                          rtc::SSL_PROTOCOL_DTLS_12)));
> 
> // Connect with DTLS, negotiating DTLS-SRTP, and transfer SRTP using bypass.
>-TEST_F(DtlsTransportChannelTest, TestTransferDtlsSrtp) {
>+TEST_F(DtlsTransportTest, TestTransferDtlsSrtp) {
>   PrepareDtls(rtc::KT_DEFAULT);
>   ASSERT_TRUE(Connect());
>   TestTransfer(1000, 100, /*srtp=*/true);
>@@ -469,14 +467,15 @@ TEST_F(DtlsTransportChannelTest, TestTransferDtlsSrtp) {
> 
> // Connect with DTLS-SRTP, transfer an invalid SRTP packet, and expects -1
> // returned.
>-TEST_F(DtlsTransportChannelTest, TestTransferDtlsInvalidSrtpPacket) {
>+TEST_F(DtlsTransportTest, TestTransferDtlsInvalidSrtpPacket) {
>   PrepareDtls(rtc::KT_DEFAULT);
>   ASSERT_TRUE(Connect());
>   EXPECT_EQ(-1, client1_.SendInvalidSrtpPacket(100));
> }
> 
>-// Create a single channel with DTLS, and send normal data and SRTP data on it.
>-TEST_F(DtlsTransportChannelTest, TestTransferDtlsSrtpDemux) {
>+// Create a single transport with DTLS, and send normal data and SRTP data on
>+// it.
>+TEST_F(DtlsTransportTest, TestTransferDtlsSrtpDemux) {
>   PrepareDtls(rtc::KT_DEFAULT);
>   ASSERT_TRUE(Connect());
>   TestTransfer(1000, 100, /*srtp=*/false);
>@@ -484,7 +483,7 @@ TEST_F(DtlsTransportChannelTest, TestTransferDtlsSrtpDemux) {
> }
> 
> // Test transferring when the "answerer" has the server role.
>-TEST_F(DtlsTransportChannelTest, TestTransferDtlsSrtpAnswererIsPassive) {
>+TEST_F(DtlsTransportTest, TestTransferDtlsSrtpAnswererIsPassive) {
>   PrepareDtls(rtc::KT_DEFAULT);
>   ASSERT_TRUE(Connect(/*client1_server=*/false));
>   TestTransfer(1000, 100, /*srtp=*/true);
>@@ -492,7 +491,7 @@ TEST_F(DtlsTransportChannelTest, TestTransferDtlsSrtpAnswererIsPassive) {
> 
> // Test that renegotiation (setting same role and fingerprint again) can be
> // started before the clients become connected in the first negotiation.
>-TEST_F(DtlsTransportChannelTest, TestRenegotiateBeforeConnect) {
>+TEST_F(DtlsTransportTest, TestRenegotiateBeforeConnect) {
>   PrepareDtls(rtc::KT_DEFAULT);
>   // Note: This is doing the same thing Connect normally does, minus some
>   // additional checks not relevant for this test.
>@@ -506,7 +505,7 @@ TEST_F(DtlsTransportChannelTest, TestRenegotiateBeforeConnect) {
> }
> 
> // Test Certificates state after negotiation but before connection.
>-TEST_F(DtlsTransportChannelTest, TestCertificatesBeforeConnect) {
>+TEST_F(DtlsTransportTest, TestCertificatesBeforeConnect) {
>   PrepareDtls(rtc::KT_DEFAULT);
>   Negotiate();
> 
>@@ -516,12 +515,12 @@ TEST_F(DtlsTransportChannelTest, TestCertificatesBeforeConnect) {
>   auto certificate2 = client2_.dtls_transport()->GetLocalCertificate();
>   ASSERT_NE(certificate1->ssl_certificate().ToPEMString(),
>             certificate2->ssl_certificate().ToPEMString());
>-  ASSERT_FALSE(client1_.dtls_transport()->GetRemoteSSLCertificate());
>-  ASSERT_FALSE(client2_.dtls_transport()->GetRemoteSSLCertificate());
>+  ASSERT_FALSE(client1_.dtls_transport()->GetRemoteSSLCertChain());
>+  ASSERT_FALSE(client2_.dtls_transport()->GetRemoteSSLCertChain());
> }
> 
> // Test Certificates state after connection.
>-TEST_F(DtlsTransportChannelTest, TestCertificatesAfterConnect) {
>+TEST_F(DtlsTransportTest, TestCertificatesAfterConnect) {
>   PrepareDtls(rtc::KT_DEFAULT);
>   ASSERT_TRUE(Connect());
> 
>@@ -532,15 +531,17 @@ TEST_F(DtlsTransportChannelTest, TestCertificatesAfterConnect) {
>             certificate2->ssl_certificate().ToPEMString());
> 
>   // Each side's remote certificate is the other side's local certificate.
>-  std::unique_ptr<rtc::SSLCertificate> remote_cert1 =
>-      client1_.dtls_transport()->GetRemoteSSLCertificate();
>+  std::unique_ptr<rtc::SSLCertChain> remote_cert1 =
>+      client1_.dtls_transport()->GetRemoteSSLCertChain();
>   ASSERT_TRUE(remote_cert1);
>-  ASSERT_EQ(remote_cert1->ToPEMString(),
>+  ASSERT_EQ(1u, remote_cert1->GetSize());
>+  ASSERT_EQ(remote_cert1->Get(0).ToPEMString(),
>             certificate2->ssl_certificate().ToPEMString());
>-  std::unique_ptr<rtc::SSLCertificate> remote_cert2 =
>-      client2_.dtls_transport()->GetRemoteSSLCertificate();
>+  std::unique_ptr<rtc::SSLCertChain> remote_cert2 =
>+      client2_.dtls_transport()->GetRemoteSSLCertChain();
>   ASSERT_TRUE(remote_cert2);
>-  ASSERT_EQ(remote_cert2->ToPEMString(),
>+  ASSERT_EQ(1u, remote_cert2->GetSize());
>+  ASSERT_EQ(remote_cert2->Get(0).ToPEMString(),
>             certificate1->ssl_certificate().ToPEMString());
> }
> 
>@@ -548,7 +549,7 @@ TEST_F(DtlsTransportChannelTest, TestCertificatesAfterConnect) {
> // Each time a timeout occurs, the retransmission timer should be doubled up to
> // 60 seconds. The timer defaults to 1 second, but for WebRTC we should be
> // initializing it to 50ms.
>-TEST_F(DtlsTransportChannelTest, TestRetransmissionSchedule) {
>+TEST_F(DtlsTransportTest, TestRetransmissionSchedule) {
>   // We can only change the retransmission schedule with a recently-added
>   // BoringSSL API. Skip the test if not built with BoringSSL.
>   MAYBE_SKIP_TEST(IsBoringSsl);
>@@ -578,10 +579,9 @@ TEST_F(DtlsTransportChannelTest, TestRetransmissionSchedule) {
>     // millisecond before the expected time and verify that no unexpected
>     // retransmissions were sent. Then advance it the final millisecond and
>     // verify that the expected retransmission was sent.
>-    fake_clock_.AdvanceTime(
>-        rtc::TimeDelta::FromMilliseconds(timeout_schedule_ms[i] - 1));
>+    fake_clock_.AdvanceTime(webrtc::TimeDelta::ms(timeout_schedule_ms[i] - 1));
>     EXPECT_EQ(expected_hellos, client1_.received_dtls_client_hellos());
>-    fake_clock_.AdvanceTime(rtc::TimeDelta::FromMilliseconds(1));
>+    fake_clock_.AdvanceTime(webrtc::TimeDelta::ms(1));
>     EXPECT_EQ(++expected_hellos, client1_.received_dtls_client_hellos());
>   }
> }
>@@ -609,7 +609,7 @@ enum DtlsTransportEvent {
> };
> 
> class DtlsEventOrderingTest
>-    : public DtlsTransportChannelTestBase,
>+    : public DtlsTransportTestBase,
>       public ::testing::TestWithParam<
>           ::testing::tuple<std::vector<DtlsTransportEvent>, bool>> {
>  protected:
>@@ -628,8 +628,8 @@ class DtlsEventOrderingTest
>     client1_.SetupTransports(ICEROLE_CONTROLLING, simulated_delay_ms);
>     client2_.SetupTransports(ICEROLE_CONTROLLED, simulated_delay_ms);
>     // Similar to how NegotiateOrdering works.
>-    client1_.dtls_transport()->SetSslRole(rtc::SSL_SERVER);
>-    client2_.dtls_transport()->SetSslRole(rtc::SSL_CLIENT);
>+    client1_.dtls_transport()->SetDtlsRole(rtc::SSL_SERVER);
>+    client2_.dtls_transport()->SetDtlsRole(rtc::SSL_CLIENT);
>     SetRemoteFingerprintFromCert(client2_.dtls_transport(),
>                                  client1_.certificate());
> 
>@@ -683,7 +683,7 @@ class DtlsEventOrderingTest
>                              client2_.dtls_transport()->dtls_state(), kTimeout,
>                              fake_clock_);
> 
>-    // Channel should be writable iff there was a valid fingerprint.
>+    // Transports should be writable iff there was a valid fingerprint.
>     EXPECT_EQ(valid_fingerprint, client1_.dtls_transport()->writable());
>     EXPECT_EQ(valid_fingerprint, client2_.dtls_transport()->writable());
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/dtlstransportinternal.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/dtlstransportinternal.h
>index 2acbda64faa..58d36fa08db 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/dtlstransportinternal.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/dtlstransportinternal.h
>@@ -59,9 +59,9 @@ class DtlsTransportInternal : public rtc::PacketTransportInternal {
> 
>   virtual bool IsDtlsActive() const = 0;
> 
>-  virtual bool GetSslRole(rtc::SSLRole* role) const = 0;
>+  virtual bool GetDtlsRole(rtc::SSLRole* role) const = 0;
> 
>-  virtual bool SetSslRole(rtc::SSLRole role) = 0;
>+  virtual bool SetDtlsRole(rtc::SSLRole role) = 0;
> 
>   // Finds out which DTLS-SRTP cipher was negotiated.
>   // TODO(zhihuang): Remove this once all dependencies implement this.
>@@ -78,9 +78,8 @@ class DtlsTransportInternal : public rtc::PacketTransportInternal {
>   virtual bool SetLocalCertificate(
>       const rtc::scoped_refptr<rtc::RTCCertificate>& certificate) = 0;
> 
>-  // Gets a copy of the remote side's SSL certificate.
>-  virtual std::unique_ptr<rtc::SSLCertificate> GetRemoteSSLCertificate()
>-      const = 0;
>+  // Gets a copy of the remote side's SSL certificate chain.
>+  virtual std::unique_ptr<rtc::SSLCertChain> GetRemoteSSLCertChain() const = 0;
> 
>   // Allows key material to be extracted for external encryption.
>   virtual bool ExportKeyingMaterial(const std::string& label,
>@@ -95,6 +94,8 @@ class DtlsTransportInternal : public rtc::PacketTransportInternal {
>                                     const uint8_t* digest,
>                                     size_t digest_len) = 0;
> 
>+  virtual bool SetSslMaxProtocolVersion(rtc::SSLProtocolVersion version) = 0;
>+
>   // Expose the underneath IceTransport.
>   virtual IceTransportInternal* ice_transport() = 0;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/fakedtlstransport.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/fakedtlstransport.h
>index 54b70180d14..cb944d3db5b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/fakedtlstransport.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/fakedtlstransport.h
>@@ -13,8 +13,10 @@
> 
> #include <memory>
> #include <string>
>+#include <utility>
> #include <vector>
> 
>+#include "absl/memory/memory.h"
> #include "p2p/base/dtlstransportinternal.h"
> #include "p2p/base/fakeicetransport.h"
> #include "rtc_base/fakesslidentity.h"
>@@ -38,10 +40,8 @@ class FakeDtlsTransport : public DtlsTransportInternal {
>         this, &FakeDtlsTransport::OnNetworkRouteChanged);
>   }
> 
>-  // If this constructor is called, a new fake ICE transport will be created,
>-  // and this FakeDtlsTransport will take the ownership.
>-  explicit FakeDtlsTransport(const std::string& name, int component)
>-      : owned_ice_transport_(new FakeIceTransport(name, component)),
>+  explicit FakeDtlsTransport(std::unique_ptr<FakeIceTransport> ice)
>+      : owned_ice_transport_(std::move(ice)),
>         transport_name_(owned_ice_transport_->transport_name()),
>         component_(owned_ice_transport_->component()),
>         dtls_fingerprint_("", nullptr, 0) {
>@@ -52,6 +52,12 @@ class FakeDtlsTransport : public DtlsTransportInternal {
>         this, &FakeDtlsTransport::OnNetworkRouteChanged);
>   }
> 
>+  // If this constructor is called, a new fake ICE transport will be created,
>+  // and this FakeDtlsTransport will take the ownership.
>+  explicit FakeDtlsTransport(const std::string& name, int component)
>+      : FakeDtlsTransport(
>+            absl::make_unique<FakeIceTransport>(name, component)) {}
>+
>   ~FakeDtlsTransport() override {
>     if (dest_ && dest_->dest_ == this) {
>       dest_->dest_ = nullptr;
>@@ -101,6 +107,10 @@ class FakeDtlsTransport : public DtlsTransportInternal {
>         dest->SetDestination(this, true);
>       }
>       dtls_state_ = DTLS_TRANSPORT_CONNECTED;
>+      // If the |dtls_role_| is unset, set it to SSL_CLIENT by default.
>+      if (!dtls_role_) {
>+        dtls_role_ = std::move(rtc::SSL_CLIENT);
>+      }
>       SignalDtlsState(this, dtls_state_);
>       ice_transport_->SetDestination(
>           static_cast<FakeIceTransport*>(dest->ice_transport()), asymmetric);
>@@ -125,12 +135,18 @@ class FakeDtlsTransport : public DtlsTransportInternal {
>     dtls_fingerprint_ = rtc::SSLFingerprint(alg, digest, digest_len);
>     return true;
>   }
>-  bool SetSslRole(rtc::SSLRole role) override {
>-    ssl_role_ = role;
>+  bool SetSslMaxProtocolVersion(rtc::SSLProtocolVersion version) override {
>+    return true;
>+  }
>+  bool SetDtlsRole(rtc::SSLRole role) override {
>+    dtls_role_ = std::move(role);
>     return true;
>   }
>-  bool GetSslRole(rtc::SSLRole* role) const override {
>-    *role = ssl_role_;
>+  bool GetDtlsRole(rtc::SSLRole* role) const override {
>+    if (!dtls_role_) {
>+      return false;
>+    }
>+    *role = *dtls_role_;
>     return true;
>   }
>   const rtc::CryptoOptions& crypto_options() const override {
>@@ -156,17 +172,13 @@ class FakeDtlsTransport : public DtlsTransportInternal {
>     *crypto_suite = crypto_suite_;
>     return true;
>   }
>-  void SetSrtpCryptoSuite(int crypto_suite) {
>-    crypto_suite_ = crypto_suite;
>-  }
>+  void SetSrtpCryptoSuite(int crypto_suite) { crypto_suite_ = crypto_suite; }
>   bool GetSslCipherSuite(int* cipher_suite) override { return false; }
>   rtc::scoped_refptr<rtc::RTCCertificate> GetLocalCertificate() const override {
>     return local_cert_;
>   }
>-  std::unique_ptr<rtc::SSLCertificate> GetRemoteSSLCertificate()
>-      const override {
>-    return remote_cert_ ? std::unique_ptr<rtc::SSLCertificate>(
>-                              remote_cert_->GetReference())
>+  std::unique_ptr<rtc::SSLCertChain> GetRemoteSSLCertChain() const override {
>+    return remote_cert_ ? absl::make_unique<rtc::SSLCertChain>(remote_cert_)
>                         : nullptr;
>   }
>   bool ExportKeyingMaterial(const std::string& label,
>@@ -212,7 +224,7 @@ class FakeDtlsTransport : public DtlsTransportInternal {
>   }
>   int GetError() override { return ice_transport_->GetError(); }
> 
>-  rtc::Optional<rtc::NetworkRoute> network_route() const override {
>+  absl::optional<rtc::NetworkRoute> network_route() const override {
>     return ice_transport_->network_route();
>   }
> 
>@@ -244,7 +256,7 @@ class FakeDtlsTransport : public DtlsTransportInternal {
>     SignalWritableState(this);
>   }
> 
>-  void OnNetworkRouteChanged(rtc::Optional<rtc::NetworkRoute> network_route) {
>+  void OnNetworkRouteChanged(absl::optional<rtc::NetworkRoute> network_route) {
>     SignalNetworkRouteChanged(network_route);
>   }
> 
>@@ -258,7 +270,7 @@ class FakeDtlsTransport : public DtlsTransportInternal {
>   bool do_dtls_ = false;
>   rtc::SSLProtocolVersion ssl_max_version_ = rtc::SSL_PROTOCOL_DTLS_12;
>   rtc::SSLFingerprint dtls_fingerprint_;
>-  rtc::SSLRole ssl_role_ = rtc::SSL_CLIENT;
>+  absl::optional<rtc::SSLRole> dtls_role_;
>   int crypto_suite_ = rtc::SRTP_AES128_CM_SHA1_80;
>   rtc::CryptoOptions crypto_options_;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/fakeicetransport.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/fakeicetransport.h
>index f3a8313792d..48157451ea6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/fakeicetransport.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/fakeicetransport.h
>@@ -85,7 +85,9 @@ class FakeIceTransport : public IceTransportInternal {
>   }
> 
>   // Convenience functions for accessing ICE config and other things.
>-  int receiving_timeout() const { return ice_config_.receiving_timeout; }
>+  int receiving_timeout() const {
>+    return ice_config_.receiving_timeout_or_default();
>+  }
>   bool gather_continually() const { return ice_config_.gather_continually(); }
>   const Candidates& remote_candidates() const { return remote_candidates_; }
> 
>@@ -144,20 +146,30 @@ class FakeIceTransport : public IceTransportInternal {
>   void AddRemoteCandidate(const Candidate& candidate) override {
>     remote_candidates_.push_back(candidate);
>   }
>-  void RemoveRemoteCandidate(const Candidate& candidate) override {}
>+  void RemoveRemoteCandidate(const Candidate& candidate) override {
>+    auto it = std::find(remote_candidates_.begin(), remote_candidates_.end(),
>+                        candidate);
>+    if (it == remote_candidates_.end()) {
>+      RTC_LOG(LS_INFO) << "Trying to remove a candidate which doesn't exist.";
>+      return;
>+    }
> 
>-  bool GetStats(ConnectionInfos* infos) override {
>-    ConnectionInfo info;
>-    infos->clear();
>-    infos->push_back(info);
>-    return true;
>+    remote_candidates_.erase(it);
>   }
> 
>-  rtc::Optional<int> GetRttEstimate() override { return rtc::nullopt; }
>-
>-  void SetMetricsObserver(webrtc::MetricsObserverInterface* observer) override {
>+  bool GetStats(ConnectionInfos* candidate_pair_stats_list,
>+                CandidateStatsList* candidate_stats_list) override {
>+    CandidateStats candidate_stats;
>+    ConnectionInfo candidate_pair_stats;
>+    candidate_stats_list->clear();
>+    candidate_stats_list->push_back(candidate_stats);
>+    candidate_pair_stats_list->clear();
>+    candidate_pair_stats_list->push_back(candidate_pair_stats);
>+    return true;
>   }
> 
>+  absl::optional<int> GetRttEstimate() override { return absl::nullopt; }
>+
>   // Fake PacketTransportInternal implementation.
>   bool writable() const override { return writable_; }
>   bool receiving() const override { return receiving_; }
>@@ -209,10 +221,10 @@ class FakeIceTransport : public IceTransportInternal {
> 
>   rtc::CopyOnWriteBuffer last_sent_packet() { return last_sent_packet_; }
> 
>-  rtc::Optional<rtc::NetworkRoute> network_route() const override {
>+  absl::optional<rtc::NetworkRoute> network_route() const override {
>     return network_route_;
>   }
>-  void SetNetworkRoute(rtc::Optional<rtc::NetworkRoute> network_route) {
>+  void SetNetworkRoute(absl::optional<rtc::NetworkRoute> network_route) {
>     network_route_ = network_route;
>   }
> 
>@@ -221,7 +233,7 @@ class FakeIceTransport : public IceTransportInternal {
>     if (writable_ == writable) {
>       return;
>     }
>-    RTC_LOG(INFO) << "set_writable from:" << writable_ << " to " << writable;
>+    RTC_LOG(INFO) << "Change writable_ to " << writable;
>     writable_ = writable;
>     if (writable_) {
>       SignalReadyToSend(this);
>@@ -267,7 +279,7 @@ class FakeIceTransport : public IceTransportInternal {
>   bool receiving_ = false;
>   bool combine_outgoing_packets_ = false;
>   rtc::CopyOnWriteBuffer send_packet_;
>-  rtc::Optional<rtc::NetworkRoute> network_route_;
>+  absl::optional<rtc::NetworkRoute> network_route_;
>   std::map<rtc::Socket::Option, int> socket_options_;
>   rtc::CopyOnWriteBuffer last_sent_packet_;
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/fakepackettransport.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/fakepackettransport.h
>index cbff5110ed0..e57bc179a79 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/fakepackettransport.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/fakepackettransport.h
>@@ -88,10 +88,10 @@ class FakePacketTransport : public PacketTransportInternal {
> 
>   const CopyOnWriteBuffer* last_sent_packet() { return &last_sent_packet_; }
> 
>-  Optional<NetworkRoute> network_route() const override {
>+  absl::optional<NetworkRoute> network_route() const override {
>     return network_route_;
>   }
>-  void SetNetworkRoute(Optional<NetworkRoute> network_route) {
>+  void SetNetworkRoute(absl::optional<NetworkRoute> network_route) {
>     network_route_ = network_route;
>   }
> 
>@@ -132,7 +132,7 @@ class FakePacketTransport : public PacketTransportInternal {
>   bool writable_ = false;
>   bool receiving_ = false;
> 
>-  Optional<NetworkRoute> network_route_;
>+  absl::optional<NetworkRoute> network_route_;
> };
> 
> }  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/fakeportallocator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/fakeportallocator.h
>index 3f0a1d4f8c5..fc5f23722f5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/fakeportallocator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/fakeportallocator.h
>@@ -18,11 +18,12 @@
> #include "p2p/base/basicpacketsocketfactory.h"
> #include "p2p/base/portallocator.h"
> #include "p2p/base/udpport.h"
>+#include "rtc_base/bind.h"
> #include "rtc_base/nethelpers.h"
>+#include "rtc_base/thread.h"
> 
> namespace rtc {
> class SocketFactory;
>-class Thread;
> }
> 
> namespace cricket {
>@@ -128,6 +129,7 @@ class FakePortAllocatorSession : public PortAllocatorSession {
>       port_.reset(TestUDPPort::Create(network_thread_, factory_, &network, 0, 0,
>                                       username(), password(), std::string(),
>                                       false));
>+      RTC_DCHECK(port_);
>       port_->SignalDestroyed.connect(
>           this, &FakePortAllocatorSession::OnPortDestroyed);
>       AddPort(port_.get());
>@@ -138,7 +140,16 @@ class FakePortAllocatorSession : public PortAllocatorSession {
> 
>   void StopGettingPorts() override { running_ = false; }
>   bool IsGettingPorts() override { return running_; }
>-  void ClearGettingPorts() override {}
>+  void ClearGettingPorts() override { is_cleared = true; }
>+  bool IsCleared() const override { return is_cleared; }
>+
>+  void RegatherOnAllNetworks() override {
>+    SignalIceRegathering(this, IceRegatheringReason::OCCASIONAL_REFRESH);
>+  }
>+
>+  void RegatherOnFailedNetworks() override {
>+    SignalIceRegathering(this, IceRegatheringReason::NETWORK_FAILURE);
>+  }
> 
>   std::vector<PortInterface*> ReadyPorts() const override {
>     return ready_ports_;
>@@ -203,6 +214,7 @@ class FakePortAllocatorSession : public PortAllocatorSession {
>   std::vector<Candidate> candidates_;
>   std::vector<PortInterface*> ready_ports_;
>   bool allocation_done_ = false;
>+  bool is_cleared = false;
>   ServerAddresses stun_servers_;
>   std::vector<RelayServerConfig> turn_servers_;
>   uint32_t candidate_filter_ = CF_ALL;
>@@ -219,12 +231,15 @@ class FakePortAllocator : public cricket::PortAllocator {
>       owned_factory_.reset(new rtc::BasicPacketSocketFactory(network_thread_));
>       factory_ = owned_factory_.get();
>     }
>-  }
> 
>-  void Initialize() override {
>-    // Port allocator should be initialized on the network thread.
>-    RTC_CHECK(network_thread_->IsCurrent());
>-    initialized_ = true;
>+    if (network_thread_ == nullptr) {
>+      network_thread_ = rtc::Thread::Current();
>+      Initialize();
>+      return;
>+    }
>+    network_thread_->Invoke<void>(RTC_FROM_HERE,
>+                                  rtc::Bind(&PortAllocator::Initialize,
>+                                            static_cast<PortAllocator*>(this)));
>   }
> 
>   void SetNetworkIgnoreMask(int network_ignore_mask) override {}
>@@ -245,7 +260,6 @@ class FakePortAllocator : public cricket::PortAllocator {
>   rtc::Thread* network_thread_;
>   rtc::PacketSocketFactory* factory_;
>   std::unique_ptr<rtc::BasicPacketSocketFactory> owned_factory_;
>-  bool initialized_ = false;
> };
> 
> }  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/icetransportinternal.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/icetransportinternal.cc
>index e4e56fe39f1..261e0f0e607 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/icetransportinternal.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/icetransportinternal.cc
>@@ -10,6 +10,8 @@
> 
> #include "p2p/base/icetransportinternal.h"
> 
>+#include "p2p/base/p2pconstants.h"
>+
> namespace cricket {
> 
> IceConfig::IceConfig() = default;
>@@ -36,6 +38,43 @@ IceConfig::IceConfig(int receiving_timeout_ms,
> 
> IceConfig::~IceConfig() = default;
> 
>+int IceConfig::receiving_timeout_or_default() const {
>+  return receiving_timeout.value_or(RECEIVING_TIMEOUT);
>+}
>+int IceConfig::backup_connection_ping_interval_or_default() const {
>+  return backup_connection_ping_interval.value_or(
>+      BACKUP_CONNECTION_PING_INTERVAL);
>+}
>+int IceConfig::stable_writable_connection_ping_interval_or_default() const {
>+  return stable_writable_connection_ping_interval.value_or(
>+      STRONG_AND_STABLE_WRITABLE_CONNECTION_PING_INTERVAL);
>+}
>+int IceConfig::regather_on_failed_networks_interval_or_default() const {
>+  return regather_on_failed_networks_interval.value_or(
>+      REGATHER_ON_FAILED_NETWORKS_INTERVAL);
>+}
>+int IceConfig::receiving_switching_delay_or_default() const {
>+  return receiving_switching_delay.value_or(RECEIVING_SWITCHING_DELAY);
>+}
>+int IceConfig::ice_check_interval_strong_connectivity_or_default() const {
>+  return ice_check_interval_strong_connectivity.value_or(STRONG_PING_INTERVAL);
>+}
>+int IceConfig::ice_check_interval_weak_connectivity_or_default() const {
>+  return ice_check_interval_weak_connectivity.value_or(WEAK_PING_INTERVAL);
>+}
>+int IceConfig::ice_check_min_interval_or_default() const {
>+  return ice_check_min_interval.value_or(-1);
>+}
>+int IceConfig::ice_unwritable_timeout_or_default() const {
>+  return ice_unwritable_timeout.value_or(CONNECTION_WRITE_CONNECT_TIMEOUT);
>+}
>+int IceConfig::ice_unwritable_min_checks_or_default() const {
>+  return ice_unwritable_min_checks.value_or(CONNECTION_WRITE_CONNECT_FAILURES);
>+}
>+int IceConfig::stun_keepalive_interval_or_default() const {
>+  return stun_keepalive_interval.value_or(STUN_KEEPALIVE_INTERVAL);
>+}
>+
> IceTransportInternal::IceTransportInternal() = default;
> 
> IceTransportInternal::~IceTransportInternal() = default;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/icetransportinternal.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/icetransportinternal.h
>index c0284ab1e1c..7483c3babc2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/icetransportinternal.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/icetransportinternal.h
>@@ -21,10 +21,6 @@
> #include "p2p/base/transportdescription.h"
> #include "rtc_base/stringencode.h"
> 
>-namespace webrtc {
>-class MetricsObserverInterface;
>-}
>-
> namespace cricket {
> 
> typedef std::vector<Candidate> Candidates;
>@@ -51,9 +47,6 @@ enum ContinualGatheringPolicy {
>   GATHER_ONCE = 0,
>   // The most recent port allocator session will keep on running.
>   GATHER_CONTINUALLY,
>-  // The most recent port allocator session will keep on running, and it will
>-  // try to recover connectivity if the channel becomes disconnected.
>-  GATHER_CONTINUALLY_AND_RECOVER,
> };
> 
> // ICE Nomination mode.
>@@ -66,20 +59,19 @@ enum class NominationMode {
> };
> 
> // Information about ICE configuration.
>-// TODO(deadbeef): Use rtc::Optional to represent unset values, instead of
>+// TODO(deadbeef): Use absl::optional to represent unset values, instead of
> // -1.
> struct IceConfig {
>   // The ICE connection receiving timeout value in milliseconds.
>-  int receiving_timeout = -1;
>+  absl::optional<int> receiving_timeout;
>   // Time interval in milliseconds to ping a backup connection when the ICE
>   // channel is strongly connected.
>-  int backup_connection_ping_interval = -1;
>+  absl::optional<int> backup_connection_ping_interval;
> 
>   ContinualGatheringPolicy continual_gathering_policy = GATHER_ONCE;
> 
>   bool gather_continually() const {
>-    return continual_gathering_policy == GATHER_CONTINUALLY ||
>-           continual_gathering_policy == GATHER_CONTINUALLY_AND_RECOVER;
>+    return continual_gathering_policy == GATHER_CONTINUALLY;
>   }
> 
>   // Whether we should prioritize Relay/Relay candidate when nothing
>@@ -87,7 +79,7 @@ struct IceConfig {
>   bool prioritize_most_likely_candidate_pairs = false;
> 
>   // Writable connections are pinged at a slower rate once stablized.
>-  int stable_writable_connection_ping_interval = -1;
>+  absl::optional<int> stable_writable_connection_ping_interval;
> 
>   // If set to true, this means the ICE transport should presume TURN-to-TURN
>   // candidate pairs will succeed, even before a binding response is received.
>@@ -95,25 +87,55 @@ struct IceConfig {
> 
>   // Interval to check on all networks and to perform ICE regathering on any
>   // active network having no connection on it.
>-  rtc::Optional<int> regather_on_failed_networks_interval;
>+  absl::optional<int> regather_on_failed_networks_interval;
> 
>   // Interval to perform ICE regathering on all networks
>   // The delay in milliseconds is sampled from the uniform distribution [a, b]
>-  rtc::Optional<rtc::IntervalRange> regather_all_networks_interval_range;
>+  absl::optional<rtc::IntervalRange> regather_all_networks_interval_range;
> 
>   // The time period in which we will not switch the selected connection
>   // when a new connection becomes receiving but the selected connection is not
>   // in case that the selected connection may become receiving soon.
>-  rtc::Optional<int> receiving_switching_delay;
>+  absl::optional<int> receiving_switching_delay;
> 
>   // TODO(honghaiz): Change the default to regular nomination.
>   // Default nomination mode if the remote does not support renomination.
>   NominationMode default_nomination_mode = NominationMode::SEMI_AGGRESSIVE;
> 
>+  // The interval in milliseconds at which ICE checks (STUN pings) will be sent
>+  // for a candidate pair when it is both writable and receiving (strong
>+  // connectivity). This parameter overrides the default value given by
>+  // |STRONG_PING_INTERVAL| in p2ptransport.h if set.
>+  absl::optional<int> ice_check_interval_strong_connectivity;
>+  // The interval in milliseconds at which ICE checks (STUN pings) will be sent
>+  // for a candidate pair when it is either not writable or not receiving (weak
>+  // connectivity). This parameter overrides the default value given by
>+  // |WEAK_PING_INTERVAL| in p2ptransport.h if set.
>+  absl::optional<int> ice_check_interval_weak_connectivity;
>   // ICE checks (STUN pings) will not be sent at higher rate (lower interval)
>   // than this, no matter what other settings there are.
>   // Measure in milliseconds.
>-  rtc::Optional<int> ice_check_min_interval;
>+  //
>+  // Note that this parameter overrides both the above check intervals for
>+  // candidate pairs with strong or weak connectivity, if either of the above
>+  // interval is shorter than the min interval.
>+  absl::optional<int> ice_check_min_interval;
>+  // The min time period for which a candidate pair must wait for response to
>+  // connectivity checks before it becomes unwritable. This parameter
>+  // overrides the default value given by |CONNECTION_WRITE_CONNECT_TIMEOUT|
>+  // in port.h if set, when determining the writability of a candidate pair.
>+  absl::optional<int> ice_unwritable_timeout;
>+
>+  // The min number of connectivity checks that a candidate pair must sent
>+  // without receiving response before it becomes unwritable. This parameter
>+  // overrides the default value given by |CONNECTION_WRITE_CONNECT_FAILURES| in
>+  // port.h if set, when determining the writability of a candidate pair.
>+  absl::optional<int> ice_unwritable_min_checks;
>+  // The interval in milliseconds at which STUN candidates will resend STUN
>+  // binding requests to keep NAT bindings open.
>+  absl::optional<int> stun_keepalive_interval;
>+
>+  absl::optional<rtc::AdapterType> network_preference;
> 
>   IceConfig();
>   IceConfig(int receiving_timeout_ms,
>@@ -125,6 +147,21 @@ struct IceConfig {
>             int regather_on_failed_networks_interval_ms,
>             int receiving_switching_delay_ms);
>   ~IceConfig();
>+
>+  // Helper getters for parameters with implementation-specific default value.
>+  // By convention, parameters with default value are represented by
>+  // absl::optional and setting a parameter to null restores its default value.
>+  int receiving_timeout_or_default() const;
>+  int backup_connection_ping_interval_or_default() const;
>+  int stable_writable_connection_ping_interval_or_default() const;
>+  int regather_on_failed_networks_interval_or_default() const;
>+  int receiving_switching_delay_or_default() const;
>+  int ice_check_interval_strong_connectivity_or_default() const;
>+  int ice_check_interval_weak_connectivity_or_default() const;
>+  int ice_check_min_interval_or_default() const;
>+  int ice_unwritable_timeout_or_default() const;
>+  int ice_unwritable_min_checks_or_default() const;
>+  int stun_keepalive_interval_or_default() const;
> };
> 
> // TODO(zhihuang): Replace this with
>@@ -163,7 +200,7 @@ class IceTransportInternal : public rtc::PacketTransportInternal {
> 
>   // TODO(zhihuang): Remove this once it's no longer called in
>   // remoting/protocol/libjingle_transport_factory.cc
>-  virtual void SetIceProtocolType(IceProtocolType) {}
>+  virtual void SetIceProtocolType(IceProtocolType type) {}
> 
>   virtual void SetIceCredentials(const std::string& ice_ufrag,
>                                  const std::string& ice_pwd);
>@@ -185,9 +222,6 @@ class IceTransportInternal : public rtc::PacketTransportInternal {
>   // occurred.
>   virtual void MaybeStartGathering() = 0;
> 
>-  virtual void SetMetricsObserver(
>-      webrtc::MetricsObserverInterface* observer) = 0;
>-
>   virtual void AddRemoteCandidate(const Candidate& candidate) = 0;
> 
>   virtual void RemoveRemoteCandidate(const Candidate& candidate) = 0;
>@@ -195,11 +229,12 @@ class IceTransportInternal : public rtc::PacketTransportInternal {
>   virtual IceGatheringState gathering_state() const = 0;
> 
>   // Returns the current stats for this connection.
>-  virtual bool GetStats(ConnectionInfos* infos) = 0;
>+  virtual bool GetStats(ConnectionInfos* candidate_pair_stats_list,
>+                        CandidateStatsList* candidate_stats_list) = 0;
> 
>   // Returns RTT estimate over the currently active connection, or an empty
>-  // rtc::Optional if there is none.
>-  virtual rtc::Optional<int> GetRttEstimate() = 0;
>+  // absl::optional if there is none.
>+  virtual absl::optional<int> GetRttEstimate() = 0;
> 
>   sigslot::signal1<IceTransportInternal*> SignalGatheringState;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/mockicetransport.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/mockicetransport.h
>index 3e7aed7297b..d1863c52935 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/mockicetransport.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/mockicetransport.h
>@@ -40,7 +40,9 @@ class MockIceTransport : public IceTransportInternal {
>   MOCK_METHOD2(SetOption, int(rtc::Socket::Option opt, int value));
>   MOCK_METHOD0(GetError, int());
>   MOCK_CONST_METHOD0(GetIceRole, cricket::IceRole());
>-  MOCK_METHOD1(GetStats, bool(cricket::ConnectionInfos* infos));
>+  MOCK_METHOD2(GetStats,
>+               bool(cricket::ConnectionInfos* candidate_pair_stats_list,
>+                    cricket::CandidateStatsList* candidate_stats_list));
> 
>   IceTransportState GetState() const override {
>     return IceTransportState::STATE_INIT;
>@@ -55,10 +57,8 @@ class MockIceTransport : public IceTransportInternal {
>   void SetRemoteIceParameters(const IceParameters& ice_params) override {}
>   void SetRemoteIceMode(IceMode mode) override {}
>   void SetIceConfig(const IceConfig& config) override {}
>-  rtc::Optional<int> GetRttEstimate() override { return rtc::nullopt; }
>+  absl::optional<int> GetRttEstimate() override { return absl::nullopt; }
>   void MaybeStartGathering() override {}
>-  void SetMetricsObserver(webrtc::MetricsObserverInterface* observer) override {
>-  }
>   void AddRemoteCandidate(const Candidate& candidate) override {}
>   void RemoveRemoteCandidate(const Candidate& candidate) override {}
>   IceGatheringState gathering_state() const override {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/p2pconstants.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/p2pconstants.cc
>index dc6db636e58..5df28618778 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/p2pconstants.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/p2pconstants.cc
>@@ -43,4 +43,37 @@ const char CONNECTIONROLE_PASSIVE_STR[] = "passive";
> const char CONNECTIONROLE_ACTPASS_STR[] = "actpass";
> const char CONNECTIONROLE_HOLDCONN_STR[] = "holdconn";
> 
>+const int MIN_CHECK_RECEIVING_INTERVAL = 50;
>+const int RECEIVING_TIMEOUT = MIN_CHECK_RECEIVING_INTERVAL * 50;
>+const int RECEIVING_SWITCHING_DELAY = 1000;
>+const int BACKUP_CONNECTION_PING_INTERVAL = 25 * 1000;
>+const int REGATHER_ON_FAILED_NETWORKS_INTERVAL = 5 * 60 * 1000;
>+
>+// When the socket is unwritable, we will use 10 Kbps (ignoring IP+UDP headers)
>+// for pinging. When the socket is writable, we will use only 1 Kbps because we
>+// don't want to degrade the quality on a modem.  These numbers should work well
>+// on a 28.8K modem, which is the slowest connection on which the voice quality
>+// is reasonable at all.
>+const int STUN_PING_PACKET_SIZE = 60 * 8;
>+const int STRONG_PING_INTERVAL = 1000 * STUN_PING_PACKET_SIZE / 1000;  // 480ms.
>+const int WEAK_PING_INTERVAL = 1000 * STUN_PING_PACKET_SIZE / 10000;   // 48ms.
>+const int WEAK_OR_STABILIZING_WRITABLE_CONNECTION_PING_INTERVAL = 900;
>+const int STRONG_AND_STABLE_WRITABLE_CONNECTION_PING_INTERVAL = 2500;
>+const int CONNECTION_WRITE_CONNECT_TIMEOUT = 5 * 1000;  // 5 seconds
>+const uint32_t CONNECTION_WRITE_CONNECT_FAILURES = 5;   // 5 pings
>+
>+const int STUN_KEEPALIVE_INTERVAL = 10 * 1000;  // 10 seconds
>+
>+const int MIN_CONNECTION_LIFETIME = 10 * 1000;          // 10 seconds.
>+const int DEAD_CONNECTION_RECEIVE_TIMEOUT = 30 * 1000;  // 30 seconds.
>+const int WEAK_CONNECTION_RECEIVE_TIMEOUT = 2500;       // 2.5 seconds
>+const int CONNECTION_WRITE_TIMEOUT = 15 * 1000;         // 15 seconds
>+// There is no harm to keep this value high other than a small amount
>+// of increased memory, but in some networks (2G), we observe up to 60s RTTs.
>+const int CONNECTION_RESPONSE_TIMEOUT = 60 * 1000;  // 60 seconds
>+
>+// TODO(qingsi): Review and calibrate the value (bugs.webrtc.org/9427).
>+const int kMaxWaitMsBeforeSignalingAnyAddressPortsAndCandidates =
>+    2.5 * 1000;  // 2.5 seconds
>+
> }  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/p2pconstants.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/p2pconstants.h
>index 584eac2fd39..3378335bb58 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/p2pconstants.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/p2pconstants.h
>@@ -45,6 +45,69 @@ extern const char CONNECTIONROLE_PASSIVE_STR[];
> extern const char CONNECTIONROLE_ACTPASS_STR[];
> extern const char CONNECTIONROLE_HOLDCONN_STR[];
> 
>+// Constants for time intervals are in milliseconds unless otherwise stated.
>+//
>+// Most of the following constants are the default values of IceConfig
>+// paramters. See IceConfig for detailed definition.
>+//
>+// Default value of IceConfig.receiving_timeout.
>+extern const int RECEIVING_TIMEOUT;
>+// Default value IceConfig.ice_check_min_interval.
>+extern const int MIN_CHECK_RECEIVING_INTERVAL;
>+// The next two ping intervals are at the ICE transport level.
>+//
>+// STRONG_PING_INTERVAL is applied when the selected connection is both
>+// writable and receiving.
>+//
>+// Default value of IceConfig.ice_check_interval_strong_connectivity.
>+extern const int STRONG_PING_INTERVAL;
>+// WEAK_PING_INTERVAL is applied when the selected connection is either
>+// not writable or not receiving.
>+//
>+// Defaul value of IceConfig.ice_check_interval_weak_connectivity.
>+extern const int WEAK_PING_INTERVAL;
>+// The next two ping intervals are at the candidate pair level.
>+//
>+// Writable candidate pairs are pinged at a slower rate once they are stabilized
>+// and the channel is strongly connected.
>+extern const int STRONG_AND_STABLE_WRITABLE_CONNECTION_PING_INTERVAL;
>+// Writable candidate pairs are pinged at a faster rate while the connections
>+// are stabilizing or the channel is weak.
>+extern const int WEAK_OR_STABILIZING_WRITABLE_CONNECTION_PING_INTERVAL;
>+// Default value of IceConfig.backup_connection_ping_interval
>+extern const int BACKUP_CONNECTION_PING_INTERVAL;
>+// Defualt value of IceConfig.receiving_switching_delay.
>+extern const int RECEIVING_SWITCHING_DELAY;
>+// Default value of IceConfig.regather_on_failed_networks_interval.
>+extern const int REGATHER_ON_FAILED_NETWORKS_INTERVAL;
>+// Default vaule of IceConfig.ice_unwritable_timeout.
>+extern const int CONNECTION_WRITE_CONNECT_TIMEOUT;
>+// Default vaule of IceConfig.ice_unwritable_min_checks.
>+extern const uint32_t CONNECTION_WRITE_CONNECT_FAILURES;
>+// Default value of IceConfig.stun_keepalive_interval;
>+extern const int STUN_KEEPALIVE_INTERVAL;
>+
>+// The following constants are used at the candidate pair level to determine the
>+// state of a candidate pair.
>+//
>+// The timeout duration when a connection does not receive anything.
>+extern const int WEAK_CONNECTION_RECEIVE_TIMEOUT;
>+// A connection will be declared dead if it has not received anything for this
>+// long.
>+extern const int DEAD_CONNECTION_RECEIVE_TIMEOUT;
>+// The length of time we wait before timing out writability on a connection.
>+extern const int CONNECTION_WRITE_TIMEOUT;
>+// This is the length of time that we wait for a ping response to come back.
>+extern const int CONNECTION_RESPONSE_TIMEOUT;
>+// The minimum time we will wait before destroying a connection after creating
>+// it.
>+extern const int MIN_CONNECTION_LIFETIME;
>+// TODO(qingsi): Rename all constants to kConstant style.
>+//
>+// The maximum time in milliseconds we will wait before signaling any address
>+// ports and candidates gathered from these ports, if the candidate allocation
>+// is not done yet.
>+extern const int kMaxWaitMsBeforeSignalingAnyAddressPortsAndCandidates;
> }  // namespace cricket
> 
> #endif  // P2P_BASE_P2PCONSTANTS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/p2ptransportchannel.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/p2ptransportchannel.cc
>index ddd240322e6..3fbc90c7b39 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/p2ptransportchannel.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/p2ptransportchannel.cc
>@@ -16,9 +16,8 @@
> #include <utility>
> 
> #include "api/candidate.h"
>-#include "api/umametrics.h"
>+#include "logging/rtc_event_log/icelogger.h"
> #include "p2p/base/candidatepairinterface.h"
>-#include "p2p/base/common.h"
> #include "p2p/base/relayport.h"  // For RELAY_PORT_TYPE.
> #include "p2p/base/stunport.h"   // For STUN_PORT_TYPE.
> #include "rtc_base/checks.h"
>@@ -28,17 +27,10 @@
> #include "rtc_base/stringencode.h"
> #include "rtc_base/timeutils.h"
> #include "system_wrappers/include/field_trial.h"
>+#include "system_wrappers/include/metrics.h"
> 
> namespace {
> 
>-// messages for queuing up work for ourselves
>-enum {
>-  MSG_SORT_AND_UPDATE_STATE = 1,
>-  MSG_CHECK_AND_PING,
>-  MSG_REGATHER_ON_FAILED_NETWORKS,
>-  MSG_REGATHER_ON_ALL_NETWORKS
>-};
>-
> // The minimum improvement in RTT that justifies a switch.
> const int kMinImprovement = 10;
> 
>@@ -61,45 +53,51 @@ cricket::PortInterface::CandidateOrigin GetOrigin(cricket::PortInterface* port,
>     return cricket::PortInterface::ORIGIN_OTHER_PORT;
> }
> 
>-}  // unnamed namespace
>-
>-namespace cricket {
>-
>-// When the socket is unwritable, we will use 10 Kbps (ignoring IP+UDP headers)
>-// for pinging.  When the socket is writable, we will use only 1 Kbps because
>-// we don't want to degrade the quality on a modem.  These numbers should work
>-// well on a 28.8K modem, which is the slowest connection on which the voice
>-// quality is reasonable at all.
>-static const int PING_PACKET_SIZE = 60 * 8;
>-
>-// The next two ping intervals are at the channel level.
>-// STRONG_PING_INTERVAL (480ms) is applied when the selected connection is both
>-// writable and receiving.
>-const int STRONG_PING_INTERVAL = 1000 * PING_PACKET_SIZE / 1000;
>-// WEAK_PING_INTERVAL (48ms) is applied when the selected connection is either
>-// not writable or not receiving.
>-const int WEAK_PING_INTERVAL = 1000 * PING_PACKET_SIZE / 10000;
>-
>-// The next two ping intervals are at the connection level.
>-// Writable connections are pinged at a faster rate while the connections are
>-// stabilizing or the channel is weak.
>-const int WEAK_OR_STABILIZING_WRITABLE_CONNECTION_PING_INTERVAL = 900;  // ms
>-// Writable connections are pinged at a slower rate once they are stabilized and
>-// the channel is strongly connected.
>-const int STRONG_AND_STABLE_WRITABLE_CONNECTION_PING_INTERVAL = 2500;  // ms
>-
>-static const int MIN_CHECK_RECEIVING_INTERVAL = 50;  // ms
>+// TODO(qingsi) Use an enum to replace the following constants for all
>+// comparision results.
>+static constexpr int a_is_better = 1;
>+static constexpr int b_is_better = -1;
>+static constexpr int a_and_b_equal = 0;
>+
>+bool LocalCandidateUsesPreferredNetwork(
>+    const cricket::Connection* conn,
>+    absl::optional<rtc::AdapterType> network_preference) {
>+  rtc::AdapterType network_type = conn->port()->Network()->type();
>+  return network_preference.has_value() && (network_type == network_preference);
>+}
>+
>+int CompareCandidatePairsByNetworkPreference(
>+    const cricket::Connection* a,
>+    const cricket::Connection* b,
>+    absl::optional<rtc::AdapterType> network_preference) {
>+  bool a_uses_preferred_network =
>+      LocalCandidateUsesPreferredNetwork(a, network_preference);
>+  bool b_uses_preferred_network =
>+      LocalCandidateUsesPreferredNetwork(b, network_preference);
>+  if (a_uses_preferred_network && !b_uses_preferred_network) {
>+    return a_is_better;
>+  } else if (!a_uses_preferred_network && b_uses_preferred_network) {
>+    return b_is_better;
>+  }
>+  return a_and_b_equal;
>+}
> 
>-static const int RECEIVING_SWITCHING_DELAY = 1000;  // ms
>+uint32_t GetWeakPingIntervalInFieldTrial() {
>+  uint32_t weak_ping_interval = ::strtoul(
>+      webrtc::field_trial::FindFullName("WebRTC-StunInterPacketDelay").c_str(),
>+      nullptr, 10);
>+  if (weak_ping_interval) {
>+    return static_cast<int>(weak_ping_interval);
>+  }
>+  return cricket::WEAK_PING_INTERVAL;
>+}
> 
>-// We periodically check if any existing networks do not have any connection
>-// and regather on those networks.
>-static const int DEFAULT_REGATHER_ON_FAILED_NETWORKS_INTERVAL = 5 * 60 * 1000;
>+}  // unnamed namespace
> 
>-static constexpr int DEFAULT_BACKUP_CONNECTION_PING_INTERVAL = 25 * 1000;
>+namespace cricket {
> 
>-static constexpr int a_is_better = 1;
>-static constexpr int b_is_better = -1;
>+using webrtc::RTCErrorType;
>+using webrtc::RTCError;
> 
> bool IceCredentialsChanged(const std::string& old_ufrag,
>                            const std::string& old_pwd,
>@@ -115,9 +113,18 @@ bool IceCredentialsChanged(const std::string& old_ufrag,
> P2PTransportChannel::P2PTransportChannel(const std::string& transport_name,
>                                          int component,
>                                          PortAllocator* allocator)
>+    : P2PTransportChannel(transport_name, component, allocator, nullptr) {}
>+
>+P2PTransportChannel::P2PTransportChannel(
>+    const std::string& transport_name,
>+    int component,
>+    PortAllocator* allocator,
>+    webrtc::AsyncResolverFactory* async_resolver_factory,
>+    webrtc::RtcEventLog* event_log)
>     : transport_name_(transport_name),
>       component_(component),
>       allocator_(allocator),
>+      async_resolver_factory_(async_resolver_factory),
>       network_thread_(rtc::Thread::Current()),
>       incoming_only_(false),
>       error_(0),
>@@ -126,22 +133,24 @@ P2PTransportChannel::P2PTransportChannel(const std::string& transport_name,
>       ice_role_(ICEROLE_UNKNOWN),
>       tiebreaker_(0),
>       gathering_state_(kIceGatheringNew),
>-      rand_(rtc::SystemTimeNanos()),
>-      check_receiving_interval_(MIN_CHECK_RECEIVING_INTERVAL * 5),
>-      config_(MIN_CHECK_RECEIVING_INTERVAL * 50 /* receiving_timeout */,
>-              DEFAULT_BACKUP_CONNECTION_PING_INTERVAL,
>+      config_(RECEIVING_TIMEOUT,
>+              BACKUP_CONNECTION_PING_INTERVAL,
>               GATHER_ONCE /* continual_gathering_policy */,
>               false /* prioritize_most_likely_candidate_pairs */,
>               STRONG_AND_STABLE_WRITABLE_CONNECTION_PING_INTERVAL,
>               true /* presume_writable_when_fully_relayed */,
>-              DEFAULT_REGATHER_ON_FAILED_NETWORKS_INTERVAL,
>+              REGATHER_ON_FAILED_NETWORKS_INTERVAL,
>               RECEIVING_SWITCHING_DELAY) {
>-  uint32_t weak_ping_interval = ::strtoul(
>-      webrtc::field_trial::FindFullName("WebRTC-StunInterPacketDelay").c_str(),
>-      nullptr, 10);
>-  if (weak_ping_interval) {
>-    weak_ping_interval_ = static_cast<int>(weak_ping_interval);
>-  }
>+  weak_ping_interval_ = GetWeakPingIntervalInFieldTrial();
>+  // Validate IceConfig even for mostly built-in constant default values in case
>+  // we change them.
>+  RTC_DCHECK(ValidateIceConfig(config_).ok());
>+  webrtc::BasicRegatheringController::Config regathering_config(
>+      config_.regather_all_networks_interval_range,
>+      config_.regather_on_failed_networks_interval_or_default());
>+  regathering_controller_.reset(new webrtc::BasicRegatheringController(
>+      regathering_config, this, network_thread_));
>+  ice_event_log_.set_event_log(event_log);
> }
> 
> P2PTransportChannel::~P2PTransportChannel() {
>@@ -167,6 +176,7 @@ void P2PTransportChannel::AddAllocatorSession(
>     allocator_session()->PruneAllPorts();
>   }
>   allocator_sessions_.push_back(std::move(session));
>+  regathering_controller_->set_allocator_session(allocator_session());
> 
>   // We now only want to apply new candidates that we receive to the ports
>   // created by this new session because these are replacing those of the
>@@ -179,6 +189,8 @@ void P2PTransportChannel::AddConnection(Connection* connection) {
>   unpinged_connections_.insert(connection);
>   connection->set_remote_ice_mode(remote_ice_mode_);
>   connection->set_receiving_timeout(config_.receiving_timeout);
>+  connection->set_unwritable_timeout(config_.ice_unwritable_timeout);
>+  connection->set_unwritable_min_checks(config_.ice_unwritable_min_checks);
>   connection->SignalReadPacket.connect(
>       this, &P2PTransportChannel::OnReadPacket);
>   connection->SignalReadyToSend.connect(
>@@ -188,7 +200,12 @@ void P2PTransportChannel::AddConnection(Connection* connection) {
>   connection->SignalDestroyed.connect(
>       this, &P2PTransportChannel::OnConnectionDestroyed);
>   connection->SignalNominated.connect(this, &P2PTransportChannel::OnNominated);
>+
>   had_connection_ = true;
>+
>+  connection->set_ice_event_log(&ice_event_log_);
>+  LogCandidatePairConfig(connection,
>+                         webrtc::IceCandidatePairConfigType::kAdded);
> }
> 
> // Determines whether we should switch the selected connection to
>@@ -218,16 +235,17 @@ bool P2PTransportChannel::ShouldSwitchSelectedConnection(
>     return true;
>   }
> 
>-  // Do not switch to a connection that is not receiving if it has higher cost
>-  // because it may be just spuriously better.
>-  if (new_connection->ComputeNetworkCost() >
>-          selected_connection_->ComputeNetworkCost() &&
>-      !new_connection->receiving()) {
>+  // Do not switch to a connection that is not receiving if it is not on a
>+  // preferred network or it has higher cost because it may be just spuriously
>+  // better.
>+  int compare_a_b_by_networks = CompareCandidatePairNetworks(
>+      new_connection, selected_connection_, config_.network_preference);
>+  if (compare_a_b_by_networks == b_is_better && !new_connection->receiving()) {
>     return false;
>   }
> 
>-  rtc::Optional<int64_t> receiving_unchanged_threshold(
>-      rtc::TimeMillis() - config_.receiving_switching_delay.value_or(0));
>+  absl::optional<int64_t> receiving_unchanged_threshold(
>+      rtc::TimeMillis() - config_.receiving_switching_delay_or_default());
>   int cmp = CompareConnections(selected_connection_, new_connection,
>                                receiving_unchanged_threshold,
>                                missed_receiving_unchanged_threshold);
>@@ -246,18 +264,23 @@ bool P2PTransportChannel::MaybeSwitchSelectedConnection(
>   bool missed_receiving_unchanged_threshold = false;
>   if (ShouldSwitchSelectedConnection(new_connection,
>                                      &missed_receiving_unchanged_threshold)) {
>-    RTC_LOG(LS_INFO) << "Switching selected connection due to " << reason;
>+    RTC_LOG(LS_INFO) << "Switching selected connection due to: " << reason;
>     SwitchSelectedConnection(new_connection);
>     return true;
>   }
>   if (missed_receiving_unchanged_threshold &&
>-      config_.receiving_switching_delay) {
>+      config_.receiving_switching_delay_or_default()) {
>     // If we do not switch to the connection because it missed the receiving
>     // threshold, the new connection is in a better receiving state than the
>     // currently selected connection. So we need to re-check whether it needs
>     // to be switched at a later time.
>-    thread()->PostDelayed(RTC_FROM_HERE, *config_.receiving_switching_delay,
>-                          this, MSG_SORT_AND_UPDATE_STATE);
>+    const std::string reason_to_sort =
>+        reason + " (after switching dampening interval)";
>+    invoker_.AsyncInvokeDelayed<void>(
>+        RTC_FROM_HERE, thread(),
>+        rtc::Bind(&P2PTransportChannel::SortConnectionsAndUpdateState, this,
>+                  reason_to_sort),
>+        config_.receiving_switching_delay_or_default());
>   }
>   return false;
> }
>@@ -316,12 +339,12 @@ IceGatheringState P2PTransportChannel::gathering_state() const {
>   return gathering_state_;
> }
> 
>-rtc::Optional<int> P2PTransportChannel::GetRttEstimate() {
>+absl::optional<int> P2PTransportChannel::GetRttEstimate() {
>   if (selected_connection_ != nullptr
>       && selected_connection_->rtt_samples() > 0) {
>     return selected_connection_->rtt();
>   } else {
>-    return rtc::nullopt;
>+    return absl::nullopt;
>   }
> }
> 
>@@ -348,13 +371,15 @@ IceTransportState P2PTransportChannel::ComputeState() const {
>     if (networks.find(network) == networks.end()) {
>       networks.insert(network);
>     } else {
>-      LOG_J(LS_VERBOSE, this) << "Ice not completed yet for this channel as "
>-                              << network->ToString()
>-                              << " has more than 1 connection.";
>+      RTC_LOG(LS_VERBOSE) << ToString()
>+                          << ": Ice not completed yet for this channel as "
>+                          << network->ToString()
>+                          << " has more than 1 connection.";
>       return IceTransportState::STATE_CONNECTING;
>     }
>   }
> 
>+  ice_event_log_.DumpCandidatePairDescriptionToMemoryAsConfigEvents();
>   return IceTransportState::STATE_COMPLETED;
> }
> 
>@@ -395,18 +420,23 @@ void P2PTransportChannel::SetRemoteIceParameters(
>         ice_params, static_cast<int>(remote_ice_parameters_.size() - 1));
>   }
>   // Updating the remote ICE candidate generation could change the sort order.
>-  RequestSortAndStateUpdate();
>+  RequestSortAndStateUpdate("remote candidate generation maybe changed");
> }
> 
> void P2PTransportChannel::SetRemoteIceMode(IceMode mode) {
>   remote_ice_mode_ = mode;
> }
> 
>+// TODO(qingsi): We apply the convention that setting a absl::optional parameter
>+// to null restores its default value in the implementation. However, some
>+// absl::optional parameters are only processed below if non-null, e.g.,
>+// regather_on_failed_networks_interval, and thus there is no way to restore the
>+// defaults. Fix this issue later for consistency.
> void P2PTransportChannel::SetIceConfig(const IceConfig& config) {
>   if (config_.continual_gathering_policy != config.continual_gathering_policy) {
>     if (!allocator_sessions_.empty()) {
>       RTC_LOG(LS_ERROR) << "Trying to change continual gathering policy "
>-                        << "when gathering has already started!";
>+                           "when gathering has already started!";
>     } else {
>       config_.continual_gathering_policy = config.continual_gathering_policy;
>       RTC_LOG(LS_INFO) << "Set continual_gathering_policy to "
>@@ -414,27 +444,22 @@ void P2PTransportChannel::SetIceConfig(const IceConfig& config) {
>     }
>   }
> 
>-  if (config.backup_connection_ping_interval >= 0 &&
>-      config_.backup_connection_ping_interval !=
>-          config.backup_connection_ping_interval) {
>+  if (config_.backup_connection_ping_interval !=
>+      config.backup_connection_ping_interval) {
>     config_.backup_connection_ping_interval =
>         config.backup_connection_ping_interval;
>     RTC_LOG(LS_INFO) << "Set backup connection ping interval to "
>-                     << config_.backup_connection_ping_interval
>+                     << config_.backup_connection_ping_interval_or_default()
>                      << " milliseconds.";
>   }
>-
>-  if (config.receiving_timeout >= 0 &&
>-      config_.receiving_timeout != config.receiving_timeout) {
>+  if (config_.receiving_timeout != config.receiving_timeout) {
>     config_.receiving_timeout = config.receiving_timeout;
>-    check_receiving_interval_ =
>-        std::max(MIN_CHECK_RECEIVING_INTERVAL, config_.receiving_timeout / 10);
>-
>     for (Connection* connection : connections_) {
>       connection->set_receiving_timeout(config_.receiving_timeout);
>     }
>     RTC_LOG(LS_INFO) << "Set ICE receiving timeout to "
>-                     << config_.receiving_timeout << " milliseconds";
>+                     << config_.receiving_timeout_or_default()
>+                     << " milliseconds";
>   }
> 
>   config_.prioritize_most_likely_candidate_pairs =
>@@ -442,20 +467,20 @@ void P2PTransportChannel::SetIceConfig(const IceConfig& config) {
>   RTC_LOG(LS_INFO) << "Set ping most likely connection to "
>                    << config_.prioritize_most_likely_candidate_pairs;
> 
>-  if (config.stable_writable_connection_ping_interval >= 0 &&
>-      config_.stable_writable_connection_ping_interval !=
>-          config.stable_writable_connection_ping_interval) {
>+  if (config_.stable_writable_connection_ping_interval !=
>+      config.stable_writable_connection_ping_interval) {
>     config_.stable_writable_connection_ping_interval =
>         config.stable_writable_connection_ping_interval;
>-    RTC_LOG(LS_INFO) << "Set stable_writable_connection_ping_interval to "
>-                     << config_.stable_writable_connection_ping_interval;
>+    RTC_LOG(LS_INFO)
>+        << "Set stable_writable_connection_ping_interval to "
>+        << config_.stable_writable_connection_ping_interval_or_default();
>   }
> 
>-  if (config.presume_writable_when_fully_relayed !=
>-      config_.presume_writable_when_fully_relayed) {
>+  if (config_.presume_writable_when_fully_relayed !=
>+      config.presume_writable_when_fully_relayed) {
>     if (!connections_.empty()) {
>       RTC_LOG(LS_ERROR) << "Trying to change 'presume writable' "
>-                        << "while connections already exist!";
>+                           "while connections already exist!";
>     } else {
>       config_.presume_writable_when_fully_relayed =
>           config.presume_writable_when_fully_relayed;
>@@ -464,26 +489,31 @@ void P2PTransportChannel::SetIceConfig(const IceConfig& config) {
>     }
>   }
> 
>-  if (config.regather_on_failed_networks_interval) {
>+  if (config_.regather_on_failed_networks_interval !=
>+      config.regather_on_failed_networks_interval) {
>     config_.regather_on_failed_networks_interval =
>         config.regather_on_failed_networks_interval;
>-    RTC_LOG(LS_INFO) << "Set regather_on_failed_networks_interval to "
>-                     << *config_.regather_on_failed_networks_interval;
>+    RTC_LOG(LS_INFO)
>+        << "Set regather_on_failed_networks_interval to "
>+        << config_.regather_on_failed_networks_interval_or_default();
>   }
> 
>-  if (config.regather_all_networks_interval_range) {
>+  if (config_.regather_all_networks_interval_range !=
>+      config.regather_all_networks_interval_range) {
>     // Config validation is assumed to have already happened at the API layer.
>     RTC_DCHECK(config.continual_gathering_policy != GATHER_ONCE);
>     config_.regather_all_networks_interval_range =
>         config.regather_all_networks_interval_range;
>     RTC_LOG(LS_INFO) << "Set regather_all_networks_interval_range to "
>-                     << config.regather_all_networks_interval_range->ToString();
>+                     << config.regather_all_networks_interval_range
>+                            .value_or(rtc::IntervalRange(-1, 0))
>+                            .ToString();
>   }
> 
>-  if (config.receiving_switching_delay) {
>+  if (config_.receiving_switching_delay != config.receiving_switching_delay) {
>     config_.receiving_switching_delay = config.receiving_switching_delay;
>-    RTC_LOG(LS_INFO) << "Set receiving_switching_delay to"
>-                     << *config_.receiving_switching_delay;
>+    RTC_LOG(LS_INFO) << "Set receiving_switching_delay to "
>+                     << config_.receiving_switching_delay_or_default();
>   }
> 
>   if (config_.default_nomination_mode != config.default_nomination_mode) {
>@@ -492,27 +522,149 @@ void P2PTransportChannel::SetIceConfig(const IceConfig& config) {
>                      << static_cast<int>(config_.default_nomination_mode);
>   }
> 
>+  if (config_.ice_check_interval_strong_connectivity !=
>+      config.ice_check_interval_strong_connectivity) {
>+    config_.ice_check_interval_strong_connectivity =
>+        config.ice_check_interval_strong_connectivity;
>+    RTC_LOG(LS_INFO)
>+        << "Set strong ping interval to "
>+        << config_.ice_check_interval_strong_connectivity_or_default();
>+  }
>+
>+  if (config_.ice_check_interval_weak_connectivity !=
>+      config.ice_check_interval_weak_connectivity) {
>+    config_.ice_check_interval_weak_connectivity =
>+        config.ice_check_interval_weak_connectivity;
>+    RTC_LOG(LS_INFO)
>+        << "Set weak ping interval to "
>+        << config_.ice_check_interval_weak_connectivity_or_default();
>+  }
>+
>   if (config_.ice_check_min_interval != config.ice_check_min_interval) {
>     config_.ice_check_min_interval = config.ice_check_min_interval;
>     RTC_LOG(LS_INFO) << "Set min ping interval to "
>-                     << *config_.ice_check_min_interval;
>+                     << config_.ice_check_min_interval_or_default();
>+  }
>+
>+  if (config_.ice_unwritable_timeout != config.ice_unwritable_timeout) {
>+    config_.ice_unwritable_timeout = config.ice_unwritable_timeout;
>+    for (Connection* conn : connections_) {
>+      conn->set_unwritable_timeout(config_.ice_unwritable_timeout);
>+    }
>+    RTC_LOG(LS_INFO) << "Set unwritable timeout to "
>+                     << config_.ice_unwritable_timeout_or_default();
>+  }
>+
>+  if (config_.ice_unwritable_min_checks != config.ice_unwritable_min_checks) {
>+    config_.ice_unwritable_min_checks = config.ice_unwritable_min_checks;
>+    for (Connection* conn : connections_) {
>+      conn->set_unwritable_min_checks(config_.ice_unwritable_min_checks);
>+    }
>+    RTC_LOG(LS_INFO) << "Set unwritable min checks to "
>+                     << config_.ice_unwritable_min_checks_or_default();
>+  }
>+
>+  if (config_.network_preference != config.network_preference) {
>+    config_.network_preference = config.network_preference;
>+    RequestSortAndStateUpdate("network preference changed");
>+    RTC_LOG(LS_INFO) << "Set network preference to "
>+                     << (config_.network_preference.has_value()
>+                             ? config_.network_preference.value()
>+                             : -1);  // network_preference cannot be bound to
>+                                     // int with value_or.
>   }
>+
>+  // TODO(qingsi): Resolve the naming conflict of stun_keepalive_delay in
>+  // UDPPort and stun_keepalive_interval.
>+  if (config_.stun_keepalive_interval != config.stun_keepalive_interval) {
>+    config_.stun_keepalive_interval = config.stun_keepalive_interval;
>+    allocator_session()->SetStunKeepaliveIntervalForReadyPorts(
>+        config_.stun_keepalive_interval);
>+    RTC_LOG(LS_INFO) << "Set STUN keepalive interval to "
>+                     << config.stun_keepalive_interval_or_default();
>+  }
>+
>+  webrtc::BasicRegatheringController::Config regathering_config(
>+      config_.regather_all_networks_interval_range,
>+      config_.regather_on_failed_networks_interval_or_default());
>+  regathering_controller_->SetConfig(regathering_config);
>+
>+  RTC_DCHECK(ValidateIceConfig(config_).ok());
> }
> 
> const IceConfig& P2PTransportChannel::config() const {
>   return config_;
> }
> 
>-void P2PTransportChannel::SetMetricsObserver(
>-    webrtc::MetricsObserverInterface* observer) {
>-  metrics_observer_ = observer;
>+// TODO(qingsi): Add tests for the config validation starting from
>+// PeerConnection::SetConfiguration.
>+RTCError P2PTransportChannel::ValidateIceConfig(const IceConfig& config) {
>+  if (config.regather_all_networks_interval_range &&
>+      config.continual_gathering_policy == GATHER_ONCE) {
>+    return RTCError(RTCErrorType::INVALID_PARAMETER,
>+                    "regather_all_networks_interval_range specified but "
>+                    "continual gathering policy is GATHER_ONCE");
>+  }
>+
>+  if (config.ice_check_interval_strong_connectivity_or_default() <
>+      config.ice_check_interval_weak_connectivity.value_or(
>+          GetWeakPingIntervalInFieldTrial())) {
>+    return RTCError(RTCErrorType::INVALID_PARAMETER,
>+                    "Ping interval of candidate pairs is shorter when ICE is "
>+                    "strongly connected than that when ICE is weakly "
>+                    "connected");
>+  }
>+
>+  if (config.receiving_timeout_or_default() <
>+      std::max(config.ice_check_interval_strong_connectivity_or_default(),
>+               config.ice_check_min_interval_or_default())) {
>+    return RTCError(
>+        RTCErrorType::INVALID_PARAMETER,
>+        "Receiving timeout is shorter than the minimal ping interval.");
>+  }
>+
>+  if (config.backup_connection_ping_interval_or_default() <
>+      config.ice_check_interval_strong_connectivity_or_default()) {
>+    return RTCError(RTCErrorType::INVALID_PARAMETER,
>+                    "Ping interval of backup candidate pairs is shorter than "
>+                    "that of general candidate pairs when ICE is strongly "
>+                    "connected");
>+  }
>+
>+  if (config.stable_writable_connection_ping_interval_or_default() <
>+      config.ice_check_interval_strong_connectivity_or_default()) {
>+    return RTCError(RTCErrorType::INVALID_PARAMETER,
>+                    "Ping interval of stable and writable candidate pairs is "
>+                    "shorter than that of general candidate pairs when ICE is "
>+                    "strongly connected");
>+  }
>+
>+  if (config.ice_unwritable_timeout_or_default() > CONNECTION_WRITE_TIMEOUT) {
>+    return RTCError(RTCErrorType::INVALID_PARAMETER,
>+                    "The timeout period for the writability state to become "
>+                    "UNRELIABLE is longer than that to become TIMEOUT.");
>+  }
>+
>+  if (config.regather_all_networks_interval_range &&
>+      config.regather_all_networks_interval_range.value().min() < 0) {
>+    return RTCError(
>+        RTCErrorType::INVALID_RANGE,
>+        "The minimum regathering interval for all networks is negative.");
>+  }
>+
>+  return RTCError::OK();
>+}
>+
>+int P2PTransportChannel::check_receiving_interval() const {
>+  return std::max(MIN_CHECK_RECEIVING_INTERVAL,
>+                  config_.receiving_timeout_or_default() / 10);
> }
> 
> void P2PTransportChannel::MaybeStartGathering() {
>   if (ice_parameters_.ufrag.empty() || ice_parameters_.pwd.empty()) {
>     RTC_LOG(LS_ERROR)
>         << "Cannot gather candidates because ICE parameters are empty"
>-        << " ufrag: " << ice_parameters_.ufrag
>+           " ufrag: " << ice_parameters_.ufrag
>         << " pwd: " << ice_parameters_.pwd;
>     return;
>   }
>@@ -526,7 +678,7 @@ void P2PTransportChannel::MaybeStartGathering() {
>       SignalGatheringState(this);
>     }
> 
>-    if (metrics_observer_ && !allocator_sessions_.empty()) {
>+    if (!allocator_sessions_.empty()) {
>       IceRestartState state;
>       if (writable()) {
>         state = IceRestartState::CONNECTED;
>@@ -535,9 +687,9 @@ void P2PTransportChannel::MaybeStartGathering() {
>       } else {
>         state = IceRestartState::DISCONNECTED;
>       }
>-      metrics_observer_->IncrementEnumCounter(
>-          webrtc::kEnumCounterIceRestart, static_cast<int>(state),
>-          static_cast<int>(IceRestartState::MAX_VALUE));
>+      RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.IceRestartState",
>+                                static_cast<int>(state),
>+                                static_cast<int>(IceRestartState::MAX_VALUE));
>     }
> 
>     // Time for a new allocator.
>@@ -578,9 +730,10 @@ void P2PTransportChannel::OnPortReady(PortAllocatorSession *session,
>        ++it) {
>     int val = port->SetOption(it->first, it->second);
>     if (val < 0) {
>-      LOG_J(LS_WARNING, port) << "SetOption(" << it->first
>-                              << ", " << it->second
>-                              << ") failed: " << port->GetError();
>+      // Errors are frequent, so use LS_INFO. bugs.webrtc.org/9221
>+      RTC_LOG(LS_INFO) << port->ToString() << ": SetOption(" << it->first
>+                       << ", " << it->second
>+                       << ") failed: " << port->GetError();
>     }
>   }
> 
>@@ -608,7 +761,8 @@ void P2PTransportChannel::OnPortReady(PortAllocatorSession *session,
>     CreateConnection(port, *iter, iter->origin_port());
>   }
> 
>-  SortConnectionsAndUpdateState();
>+  SortConnectionsAndUpdateState(
>+      "new candidate pairs created from a new local candidate");
> }
> 
> // A new candidate is available, let listeners know
>@@ -628,7 +782,7 @@ void P2PTransportChannel::OnCandidatesAllocationDone(
>     RTC_LOG(LS_INFO) << "P2PTransportChannel: " << transport_name()
>                      << ", component " << component()
>                      << " gathering complete, but using continual "
>-                     << "gathering so not changing gathering state.";
>+                        "gathering so not changing gathering state.";
>     return;
>   }
>   gathering_state_ = kIceGatheringComplete;
>@@ -682,8 +836,8 @@ void P2PTransportChannel::OnUnknownAddress(
>         stun_msg->GetUInt32(STUN_ATTR_PRIORITY);
>     if (!priority_attr) {
>       RTC_LOG(LS_WARNING) << "P2PTransportChannel::OnUnknownAddress - "
>-                          << "No STUN_ATTR_PRIORITY found in the "
>-                          << "stun request message";
>+                             "No STUN_ATTR_PRIORITY found in the "
>+                             "stun request message";
>       port->SendBindingErrorResponse(stun_msg, address, STUN_ERROR_BAD_REQUEST,
>                                      STUN_ERROR_REASON_BAD_REQUEST);
>       return;
>@@ -713,7 +867,7 @@ void P2PTransportChannel::OnUnknownAddress(
>     // The foundation of the candidate is set to an arbitrary value, different
>     // from the foundation for all other remote candidates.
>     remote_candidate.set_foundation(
>-        rtc::ToString<uint32_t>(rtc::ComputeCrc32(remote_candidate.id())));
>+        rtc::ToString(rtc::ComputeCrc32(remote_candidate.id())));
>   }
> 
>   // RFC5245, the agent constructs a pair whose local candidate is equal to
>@@ -728,7 +882,7 @@ void P2PTransportChannel::OnUnknownAddress(
>   if (port->GetConnection(remote_candidate.address())) {
>     if (port_muxed) {
>       RTC_LOG(LS_INFO) << "Connection already exists for peer reflexive "
>-                       << "candidate: " << remote_candidate.ToString();
>+                          "candidate: " << remote_candidate.ToString();
>       return;
>     } else {
>       RTC_NOTREACHED();
>@@ -759,7 +913,8 @@ void P2PTransportChannel::OnUnknownAddress(
>   // Update the list of connections since we just added another.  We do this
>   // after sending the response since it could (in principle) delete the
>   // connection in question.
>-  SortConnectionsAndUpdateState();
>+  SortConnectionsAndUpdateState(
>+      "a new candidate pair created from an unknown remote address");
> }
> 
> void P2PTransportChannel::OnRoleConflict(PortInterface* port) {
>@@ -790,11 +945,13 @@ void P2PTransportChannel::OnNominated(Connection* conn) {
>     return;
>   }
> 
>+  // TODO(qingsi): RequestSortAndStateUpdate will eventually call
>+  // MaybeSwitchSelectedConnection again. Rewrite this logic.
>   if (MaybeSwitchSelectedConnection(conn,
>                                     "nomination on the controlled side")) {
>     // Now that we have selected a connection, it is time to prune other
>     // connections and update the read/write state of the channel.
>-    RequestSortAndStateUpdate();
>+    RequestSortAndStateUpdate("nomination on the controlled side");
>   } else {
>     RTC_LOG(LS_INFO)
>         << "Not switching the selected connection on controlled side yet: "
>@@ -847,7 +1004,8 @@ void P2PTransportChannel::AddRemoteCandidate(const Candidate& candidate) {
>   CreateConnections(new_remote_candidate, NULL);
> 
>   // Resort the connections list, which may have new elements.
>-  SortConnectionsAndUpdateState();
>+  SortConnectionsAndUpdateState(
>+      "new candidate pairs created from a new remote candidate");
> }
> 
> void P2PTransportChannel::RemoveRemoteCandidate(
>@@ -936,8 +1094,9 @@ bool P2PTransportChannel::CreateConnection(PortInterface* port,
>       return false;
>     }
>     AddConnection(connection);
>-    LOG_J(LS_INFO, this) << "Created connection with origin=" << origin << ", ("
>-                         << connections_.size() << " total)";
>+    RTC_LOG(LS_INFO) << ToString()
>+                     << ": Created connection with origin: " << origin
>+                     << ", total: " << connections_.size();
>     return true;
>   }
> 
>@@ -946,7 +1105,7 @@ bool P2PTransportChannel::CreateConnection(PortInterface* port,
>   // connection; however, the other side can send a duplicate candidate.
>   if (!remote_candidate.IsEquivalent(connection->remote_candidate())) {
>     RTC_LOG(INFO) << "Attempt to change a remote candidate."
>-                  << " Existing remote candidate: "
>+                     " Existing remote candidate: "
>                   << connection->remote_candidate().ToString()
>                   << "New remote candidate: " << remote_candidate.ToString();
>   }
>@@ -1072,7 +1231,10 @@ int P2PTransportChannel::SendPacket(const char *data, size_t len,
>   }
> 
>   last_sent_packet_id_ = options.packet_id;
>-  int sent = selected_connection_->Send(data, len, options);
>+  rtc::PacketOptions modified_options(options);
>+  modified_options.info_signaled_after_sent.packet_type =
>+      rtc::PacketType::kData;
>+  int sent = selected_connection_->Send(data, len, modified_options);
>   if (sent <= 0) {
>     RTC_DCHECK(sent < 0);
>     error_ = selected_connection_->GetError();
>@@ -1080,22 +1242,29 @@ int P2PTransportChannel::SendPacket(const char *data, size_t len,
>   return sent;
> }
> 
>-bool P2PTransportChannel::GetStats(ConnectionInfos *infos) {
>+bool P2PTransportChannel::GetStats(ConnectionInfos* candidate_pair_stats_list,
>+                                   CandidateStatsList* candidate_stats_list) {
>   RTC_DCHECK(network_thread_ == rtc::Thread::Current());
>-  // Gather connection infos.
>-  infos->clear();
>+  // Gather candidate and candidate pair stats.
>+  candidate_stats_list->clear();
>+  candidate_pair_stats_list->clear();
> 
>+  if (!allocator_sessions_.empty()) {
>+    allocator_session()->GetCandidateStatsFromReadyPorts(candidate_stats_list);
>+  }
>+
>+  // TODO(qingsi): Remove naming inconsistency for candidate pair/connection.
>   for (Connection* connection : connections_) {
>-    ConnectionInfo info = connection->stats();
>-    info.best_connection = (selected_connection_ == connection);
>-    infos->push_back(std::move(info));
>+    ConnectionInfo candidate_pair_stats = connection->stats();
>+    candidate_pair_stats.best_connection = (selected_connection_ == connection);
>+    candidate_pair_stats_list->push_back(std::move(candidate_pair_stats));
>     connection->set_reported(true);
>   }
> 
>   return true;
> }
> 
>-rtc::Optional<rtc::NetworkRoute> P2PTransportChannel::network_route() const {
>+absl::optional<rtc::NetworkRoute> P2PTransportChannel::network_route() const {
>   return network_route_;
> }
> 
>@@ -1119,9 +1288,13 @@ void P2PTransportChannel::UpdateConnectionStates() {
> }
> 
> // Prepare for best candidate sorting.
>-void P2PTransportChannel::RequestSortAndStateUpdate() {
>+void P2PTransportChannel::RequestSortAndStateUpdate(
>+    const std::string& reason_to_sort) {
>   if (!sort_dirty_) {
>-    network_thread_->Post(RTC_FROM_HERE, this, MSG_SORT_AND_UPDATE_STATE);
>+    invoker_.AsyncInvoke<void>(
>+        RTC_FROM_HERE, thread(),
>+        rtc::Bind(&P2PTransportChannel::SortConnectionsAndUpdateState, this,
>+                  reason_to_sort));
>     sort_dirty_ = true;
>   }
> }
>@@ -1135,27 +1308,47 @@ void P2PTransportChannel::MaybeStartPinging() {
>   if (std::any_of(
>           connections_.begin(), connections_.end(),
>           [this, now](const Connection* c) { return IsPingable(c, now); })) {
>-    LOG_J(LS_INFO, this) << "Have a pingable connection for the first time; "
>-                         << "starting to ping.";
>-    thread()->Post(RTC_FROM_HERE, this, MSG_CHECK_AND_PING);
>-    thread()->PostDelayed(RTC_FROM_HERE,
>-                          *config_.regather_on_failed_networks_interval, this,
>-                          MSG_REGATHER_ON_FAILED_NETWORKS);
>-    if (config_.regather_all_networks_interval_range) {
>-      thread()->PostDelayed(RTC_FROM_HERE,
>-                            SampleRegatherAllNetworksInterval(), this,
>-                            MSG_REGATHER_ON_ALL_NETWORKS);
>-    }
>+    RTC_LOG(LS_INFO) << ToString()
>+                     << ": Have a pingable connection for the first time; "
>+                        "starting to ping.";
>+    invoker_.AsyncInvoke<void>(
>+        RTC_FROM_HERE, thread(),
>+        rtc::Bind(&P2PTransportChannel::CheckAndPing, this));
>+    regathering_controller_->Start();
>     started_pinging_ = true;
>   }
> }
> 
>+int P2PTransportChannel::CompareCandidatePairNetworks(
>+    const Connection* a,
>+    const Connection* b,
>+    absl::optional<rtc::AdapterType> network_preference) const {
>+  int compare_a_b_by_network_preference =
>+      CompareCandidatePairsByNetworkPreference(a, b,
>+                                               config_.network_preference);
>+  // The network preference has a higher precedence than the network cost.
>+  if (compare_a_b_by_network_preference != a_and_b_equal) {
>+    return compare_a_b_by_network_preference;
>+  }
>+
>+  uint32_t a_cost = a->ComputeNetworkCost();
>+  uint32_t b_cost = b->ComputeNetworkCost();
>+  // Prefer lower network cost.
>+  if (a_cost < b_cost) {
>+    return a_is_better;
>+  }
>+  if (a_cost > b_cost) {
>+    return b_is_better;
>+  }
>+  return a_and_b_equal;
>+}
>+
> // Compare two connections based on their writing, receiving, and connected
> // states.
> int P2PTransportChannel::CompareConnectionStates(
>     const Connection* a,
>     const Connection* b,
>-    rtc::Optional<int64_t> receiving_unchanged_threshold,
>+    absl::optional<int64_t> receiving_unchanged_threshold,
>     bool* missed_receiving_unchanged_threshold) const {
>   // First, prefer a connection that's writable or presumed writable over
>   // one that's not writable.
>@@ -1230,15 +1423,10 @@ int P2PTransportChannel::CompareConnectionStates(
> int P2PTransportChannel::CompareConnectionCandidates(
>     const Connection* a,
>     const Connection* b) const {
>-  // Prefer lower network cost.
>-  uint32_t a_cost = a->ComputeNetworkCost();
>-  uint32_t b_cost = b->ComputeNetworkCost();
>-  // Smaller cost is better.
>-  if (a_cost < b_cost) {
>-    return a_is_better;
>-  }
>-  if (a_cost > b_cost) {
>-    return b_is_better;
>+  int compare_a_b_by_networks =
>+      CompareCandidatePairNetworks(a, b, config_.network_preference);
>+  if (compare_a_b_by_networks != a_and_b_equal) {
>+    return compare_a_b_by_networks;
>   }
> 
>   // Compare connection priority. Lower values get sorted last.
>@@ -1289,7 +1477,7 @@ bool P2PTransportChannel::IsRemoteCandidatePruned(const Candidate& cand) const {
> int P2PTransportChannel::CompareConnections(
>     const Connection* a,
>     const Connection* b,
>-    rtc::Optional<int64_t> receiving_unchanged_threshold,
>+    absl::optional<int64_t> receiving_unchanged_threshold,
>     bool* missed_receiving_unchanged_threshold) const {
>   RTC_CHECK(a != nullptr);
>   RTC_CHECK(b != nullptr);
>@@ -1335,7 +1523,8 @@ bool P2PTransportChannel::PresumedWritable(const Connection* conn) const {
> 
> // Sort the available connections to find the best one.  We also monitor
> // the number of available connections and the current state.
>-void P2PTransportChannel::SortConnectionsAndUpdateState() {
>+void P2PTransportChannel::SortConnectionsAndUpdateState(
>+    const std::string& reason_to_sort) {
>   RTC_DCHECK(network_thread_ == rtc::Thread::Current());
> 
>   // Make sure the connection states are up-to-date since this affects how they
>@@ -1352,7 +1541,7 @@ void P2PTransportChannel::SortConnectionsAndUpdateState() {
>   // TODO(honghaiz): Don't sort;  Just use std::max_element in the right places.
>   std::stable_sort(connections_.begin(), connections_.end(),
>                    [this](const Connection* a, const Connection* b) {
>-                     int cmp = CompareConnections(a, b, rtc::nullopt, nullptr);
>+                     int cmp = CompareConnections(a, b, absl::nullopt, nullptr);
>                      if (cmp != 0) {
>                        return cmp > 0;
>                      }
>@@ -1361,7 +1550,7 @@ void P2PTransportChannel::SortConnectionsAndUpdateState() {
>                    });
> 
>   RTC_LOG(LS_VERBOSE) << "Sorting " << connections_.size()
>-                      << " available connections:";
>+                      << " available connections";
>   for (size_t i = 0; i < connections_.size(); ++i) {
>     RTC_LOG(LS_VERBOSE) << connections_[i]->ToString();
>   }
>@@ -1372,7 +1561,7 @@ void P2PTransportChannel::SortConnectionsAndUpdateState() {
>   // If necessary, switch to the new choice. Note that |top_connection| doesn't
>   // have to be writable to become the selected connection although it will
>   // have higher priority if it is writable.
>-  MaybeSwitchSelectedConnection(top_connection, "sorting");
>+  MaybeSwitchSelectedConnection(top_connection, reason_to_sort);
> 
>   // The controlled side can prune only if the selected connection has been
>   // nominated because otherwise it may prune the connection that will be
>@@ -1483,15 +1672,22 @@ void P2PTransportChannel::SwitchSelectedConnection(Connection* conn) {
>   // destroyed, so don't use it.
>   Connection* old_selected_connection = selected_connection_;
>   selected_connection_ = conn;
>+  LogCandidatePairConfig(conn, webrtc::IceCandidatePairConfigType::kSelected);
>   network_route_.reset();
>+  if (old_selected_connection) {
>+    old_selected_connection->set_selected(false);
>+  }
>   if (selected_connection_) {
>     ++nomination_;
>+    selected_connection_->set_selected(true);
>     if (old_selected_connection) {
>-      LOG_J(LS_INFO, this) << "Previous selected connection: "
>-                           << old_selected_connection->ToString();
>+      RTC_LOG(LS_INFO) << ToString()
>+                       << ": Previous selected connection: "
>+                       << old_selected_connection->ToString();
>     }
>-    LOG_J(LS_INFO, this) << "New selected connection: "
>-                         << selected_connection_->ToString();
>+    RTC_LOG(LS_INFO) << ToString()
>+                     << ": New selected connection: "
>+                     << selected_connection_->ToString();
>     SignalRouteChange(this, selected_connection_->remote_candidate());
>     // This is a temporary, but safe fix to webrtc issue 5705.
>     // TODO(honghaiz): Make all ENOTCONN error routed through the transport
>@@ -1515,7 +1711,8 @@ void P2PTransportChannel::SwitchSelectedConnection(Connection* conn) {
>             selected_connection_->local_candidate().address().family()) +
>         GetProtocolOverhead(selected_connection_->local_candidate().protocol());
>   } else {
>-    LOG_J(LS_INFO, this) << "No selected connection";
>+    RTC_LOG(LS_INFO) << ToString()
>+                     << ": No selected connection";
>   }
> 
>   SignalNetworkRouteChanged(network_route_);
>@@ -1530,9 +1727,10 @@ void P2PTransportChannel::SwitchSelectedConnection(Connection* conn) {
> void P2PTransportChannel::UpdateState() {
>   IceTransportState state = ComputeState();
>   if (state_ != state) {
>-    LOG_J(LS_INFO, this) << "Transport channel state changed from "
>-                         << static_cast<int>(state_) << " to "
>-                         << static_cast<int>(state);
>+    RTC_LOG(LS_INFO) << ToString()
>+                     << ": Transport channel state changed from "
>+                     << static_cast<int>(state_) << " to "
>+                     << static_cast<int>(state);
>     // Check that the requested transition is allowed. Note that
>     // P2PTransportChannel does not (yet) implement a direct mapping of the ICE
>     // states from the standard; the difference is covered by
>@@ -1627,29 +1825,8 @@ bool P2PTransportChannel::ReadyToSend(Connection* connection) const {
>           PresumedWritable(connection));
> }
> 
>-// Handle any queued up requests
>-void P2PTransportChannel::OnMessage(rtc::Message *pmsg) {
>-  switch (pmsg->message_id) {
>-    case MSG_SORT_AND_UPDATE_STATE:
>-      SortConnectionsAndUpdateState();
>-      break;
>-    case MSG_CHECK_AND_PING:
>-      OnCheckAndPing();
>-      break;
>-    case MSG_REGATHER_ON_FAILED_NETWORKS:
>-      OnRegatherOnFailedNetworks();
>-      break;
>-    case MSG_REGATHER_ON_ALL_NETWORKS:
>-      OnRegatherOnAllNetworks();
>-      break;
>-    default:
>-      RTC_NOTREACHED();
>-      break;
>-  }
>-}
>-
> // Handle queued up check-and-ping request
>-void P2PTransportChannel::OnCheckAndPing() {
>+void P2PTransportChannel::CheckAndPing() {
>   // Make sure the states of the connections are up-to-date (since this affects
>   // which ones are pingable).
>   UpdateConnectionStates();
>@@ -1671,8 +1848,10 @@ void P2PTransportChannel::OnCheckAndPing() {
>       MarkConnectionPinged(conn);
>     }
>   }
>-  int delay = std::min(ping_interval, check_receiving_interval_);
>-  thread()->PostDelayed(RTC_FROM_HERE, delay, this, MSG_CHECK_AND_PING);
>+  int delay = std::min(ping_interval, check_receiving_interval());
>+  invoker_.AsyncInvokeDelayed<void>(
>+      RTC_FROM_HERE, thread(),
>+      rtc::Bind(&P2PTransportChannel::CheckAndPing, this), delay);
> }
> 
> // A connection is considered a backup connection if the channel state
>@@ -1717,7 +1896,7 @@ bool P2PTransportChannel::IsPingable(const Connection* conn,
>   if (IsBackupConnection(conn)) {
>     return conn->rtt_samples() == 0 ||
>            (now >= conn->last_ping_response_received() +
>-                       config_.backup_connection_ping_interval);
>+                       config_.backup_connection_ping_interval_or_default());
>   }
>   // Don't ping inactive non-backup connections.
>   if (!conn->active()) {
>@@ -1750,7 +1929,8 @@ int P2PTransportChannel::CalculateActiveWritablePingInterval(
>     return weak_ping_interval();
>   }
> 
>-  int stable_interval = config_.stable_writable_connection_ping_interval;
>+  int stable_interval =
>+      config_.stable_writable_connection_ping_interval_or_default();
>   int weak_or_stablizing_interval = std::min(
>       stable_interval, WEAK_OR_STABILIZING_WRITABLE_CONNECTION_PING_INTERVAL);
>   // If the channel is weak or the connection is not stable yet, use the
>@@ -1831,6 +2011,11 @@ Connection* P2PTransportChannel::FindNextPingableConnection() {
>   auto iter =
>       std::max_element(pingable_connections.begin(), pingable_connections.end(),
>                        [this](Connection* conn1, Connection* conn2) {
>+                         // Some implementations of max_element compare an
>+                         // element with itself.
>+                         if (conn1 == conn2) {
>+                           return false;
>+                         }
>                          return MorePingable(conn1, conn2) == conn2;
>                        });
>   if (iter != pingable_connections.end()) {
>@@ -1929,7 +2114,7 @@ void P2PTransportChannel::OnConnectionStateChange(Connection* connection) {
> 
>   // We have to unroll the stack before doing this because we may be changing
>   // the state of connections while sorting.
>-  RequestSortAndStateUpdate();
>+  RequestSortAndStateUpdate("candidate pair state changed");
> }
> 
> // When a connection is removed, edit it out, and then update our best
>@@ -1948,9 +2133,8 @@ void P2PTransportChannel::OnConnectionDestroyed(Connection* connection) {
>   unpinged_connections_.erase(*iter);
>   connections_.erase(iter);
> 
>-  LOG_J(LS_INFO, this) << "Removed connection " << std::hex << connection
>-                       << std::dec << " (" << connections_.size()
>-                       << " remaining)";
>+  RTC_LOG(LS_INFO) << ToString() << ": Removed connection " << connection
>+                   << " (" << connections_.size() << " remaining)";
> 
>   // If this is currently the selected connection, then we need to pick a new
>   // one. The call to SortConnectionsAndUpdateState will pick a new one. It
>@@ -1961,7 +2145,7 @@ void P2PTransportChannel::OnConnectionDestroyed(Connection* connection) {
>   if (selected_connection_ == connection) {
>     RTC_LOG(LS_INFO) << "Selected connection destroyed. Will choose a new one.";
>     SwitchSelectedConnection(nullptr);
>-    RequestSortAndStateUpdate();
>+    RequestSortAndStateUpdate("selected candidate pair destroyed");
>   } else {
>     // If a non-selected connection was destroyed, we don't need to re-sort but
>     // we do need to update state, because we could be switching to "failed" or
>@@ -2014,29 +2198,6 @@ void P2PTransportChannel::OnCandidatesRemoved(
>   SignalCandidatesRemoved(this, candidates_to_remove);
> }
> 
>-void P2PTransportChannel::OnRegatherOnFailedNetworks() {
>-  // Only re-gather when the current session is in the CLEARED state (i.e., not
>-  // running or stopped). It is only possible to enter this state when we gather
>-  // continually, so there is an implicit check on continual gathering here.
>-  if (!allocator_sessions_.empty() && allocator_session()->IsCleared()) {
>-    allocator_session()->RegatherOnFailedNetworks();
>-  }
>-
>-  thread()->PostDelayed(RTC_FROM_HERE,
>-                        *config_.regather_on_failed_networks_interval, this,
>-                        MSG_REGATHER_ON_FAILED_NETWORKS);
>-}
>-
>-void P2PTransportChannel::OnRegatherOnAllNetworks() {
>-  if (!allocator_sessions_.empty() && allocator_session()->IsCleared()) {
>-    allocator_session()->RegatherOnAllNetworks();
>-  }
>-
>-  thread()->PostDelayed(RTC_FROM_HERE,
>-                        SampleRegatherAllNetworksInterval(), this,
>-                        MSG_REGATHER_ON_ALL_NETWORKS);
>-}
>-
> void P2PTransportChannel::PruneAllPorts() {
>   pruned_ports_.insert(pruned_ports_.end(), ports_.begin(), ports_.end());
>   ports_.clear();
>@@ -2173,8 +2334,8 @@ void P2PTransportChannel::set_writable(bool writable) {
>   if (writable_ == writable) {
>     return;
>   }
>-  LOG_J(LS_VERBOSE, this) << "set_writable from:" << writable_ << " to "
>-                          << writable;
>+  RTC_LOG(LS_VERBOSE) << ToString()
>+                      << ": Changed writable_ to " << writable;
>   writable_ = writable;
>   if (writable_) {
>     SignalReadyToSend(this);
>@@ -2190,10 +2351,14 @@ void P2PTransportChannel::set_receiving(bool receiving) {
>   SignalReceivingState(this);
> }
> 
>-int P2PTransportChannel::SampleRegatherAllNetworksInterval() {
>-  auto interval = config_.regather_all_networks_interval_range;
>-  RTC_DCHECK(interval);
>-  return rand_.Rand(interval->min(), interval->max());
>+void P2PTransportChannel::LogCandidatePairConfig(
>+    Connection* conn,
>+    webrtc::IceCandidatePairConfigType type) {
>+  if (conn == nullptr) {
>+    return;
>+  }
>+  ice_event_log_.LogCandidatePairConfig(type, conn->id(),
>+                                        conn->ToLogDescription());
> }
> 
> }  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/p2ptransportchannel.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/p2ptransportchannel.h
>index c2febbf1d3c..1ac898e4902 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/p2ptransportchannel.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/p2ptransportchannel.h
>@@ -20,21 +20,32 @@
> #ifndef P2P_BASE_P2PTRANSPORTCHANNEL_H_
> #define P2P_BASE_P2PTRANSPORTCHANNEL_H_
> 
>+#include <algorithm>
> #include <map>
> #include <memory>
> #include <set>
> #include <string>
> #include <vector>
> 
>+#include "api/asyncresolverfactory.h"
> #include "api/candidate.h"
>+#include "api/rtcerror.h"
>+#include "logging/rtc_event_log/events/rtc_event_ice_candidate_pair_config.h"
>+#include "logging/rtc_event_log/icelogger.h"
> #include "p2p/base/candidatepairinterface.h"
> #include "p2p/base/icetransportinternal.h"
>+#include "p2p/base/p2pconstants.h"
> #include "p2p/base/portallocator.h"
> #include "p2p/base/portinterface.h"
>+#include "p2p/base/regatheringcontroller.h"
>+#include "rtc_base/asyncinvoker.h"
> #include "rtc_base/asyncpacketsocket.h"
> #include "rtc_base/constructormagic.h"
>-#include "rtc_base/random.h"
>-#include "rtc_base/sigslot.h"
>+#include "rtc_base/third_party/sigslot/sigslot.h"
>+
>+namespace webrtc {
>+class RtcEventLog;
>+}  // namespace webrtc
> 
> namespace cricket {
> 
>@@ -42,10 +53,6 @@ namespace cricket {
> // connected/connecting/disconnected when ICE restart happens.
> enum class IceRestartState { CONNECTING, CONNECTED, DISCONNECTED, MAX_VALUE };
> 
>-extern const int WEAK_PING_INTERVAL;
>-extern const int STRONG_PING_INTERVAL;
>-extern const int WEAK_OR_STABILIZING_WRITABLE_CONNECTION_PING_INTERVAL;
>-extern const int STRONG_AND_STABLE_WRITABLE_CONNECTION_PING_INTERVAL;
> static const int MIN_PINGS_AT_WEAK_PING_INTERVAL = 3;
> 
> bool IceCredentialsChanged(const std::string& old_ufrag,
>@@ -67,12 +74,18 @@ class RemoteCandidate : public Candidate {
> 
> // P2PTransportChannel manages the candidates and connection process to keep
> // two P2P clients connected to each other.
>-class P2PTransportChannel : public IceTransportInternal,
>-                            public rtc::MessageHandler {
>+class P2PTransportChannel : public IceTransportInternal {
>  public:
>+  // For testing only.
>+  // TODO(zstein): Remove once AsyncResolverFactory is required.
>   P2PTransportChannel(const std::string& transport_name,
>                       int component,
>                       PortAllocator* allocator);
>+  P2PTransportChannel(const std::string& transport_name,
>+                      int component,
>+                      PortAllocator* allocator,
>+                      webrtc::AsyncResolverFactory* async_resolver_factory,
>+                      webrtc::RtcEventLog* event_log = nullptr);
>   ~P2PTransportChannel() override;
> 
>   // From TransportChannelImpl:
>@@ -98,10 +111,10 @@ class P2PTransportChannel : public IceTransportInternal,
>   // only update the parameter if it is considered set in |config|. For example,
>   // a negative value of receiving_timeout will be considered "not set" and we
>   // will not use it to update the respective parameter in |config_|.
>-  // TODO(deadbeef): Use rtc::Optional instead of negative values.
>+  // TODO(deadbeef): Use absl::optional instead of negative values.
>   void SetIceConfig(const IceConfig& config) override;
>   const IceConfig& config() const;
>-  void SetMetricsObserver(webrtc::MetricsObserverInterface* observer) override;
>+  static webrtc::RTCError ValidateIceConfig(const IceConfig& config);
> 
>   // From TransportChannel:
>   int SendPacket(const char* data,
>@@ -111,8 +124,9 @@ class P2PTransportChannel : public IceTransportInternal,
>   int SetOption(rtc::Socket::Option opt, int value) override;
>   bool GetOption(rtc::Socket::Option opt, int* value) override;
>   int GetError() override;
>-  bool GetStats(std::vector<ConnectionInfo>* stats) override;
>-  rtc::Optional<int> GetRttEstimate() override;
>+  bool GetStats(std::vector<ConnectionInfo>* candidate_pair_stats_list,
>+                std::vector<CandidateStats>* candidate_stats_list) override;
>+  absl::optional<int> GetRttEstimate() override;
> 
>   // TODO(honghaiz): Remove this method once the reference of it in
>   // Chromoting is removed.
>@@ -129,10 +143,8 @@ class P2PTransportChannel : public IceTransportInternal,
>   IceMode remote_ice_mode() const { return remote_ice_mode_; }
> 
>   void PruneAllPorts();
>-  int receiving_timeout() const { return config_.receiving_timeout; }
>-  int check_receiving_interval() const { return check_receiving_interval_; }
>-
>-  rtc::Optional<rtc::NetworkRoute> network_route() const override;
>+  int check_receiving_interval() const;
>+  absl::optional<rtc::NetworkRoute> network_route() const override;
> 
>   // Helper method used only in unittest.
>   rtc::DiffServCodePoint DefaultDscpValue() const;
>@@ -145,7 +157,10 @@ class P2PTransportChannel : public IceTransportInternal,
>   const std::vector<Connection*>& connections() const { return connections_; }
> 
>   // Public for unit tests.
>-  PortAllocatorSession* allocator_session() {
>+  PortAllocatorSession* allocator_session() const {
>+    if (allocator_sessions_.empty()) {
>+      return nullptr;
>+    }
>     return allocator_sessions_.back().get();
>   }
> 
>@@ -165,6 +180,7 @@ class P2PTransportChannel : public IceTransportInternal,
> 
>  private:
>   rtc::Thread* thread() const { return network_thread_; }
>+
>   bool IsGettingPorts() { return allocator_session()->IsGettingPorts(); }
> 
>   // A transport channel is weak if the current best connection is either
>@@ -172,29 +188,28 @@ class P2PTransportChannel : public IceTransportInternal,
>   bool weak() const;
> 
>   int weak_ping_interval() const {
>-    if (config_.ice_check_min_interval &&
>-        weak_ping_interval_ < *config_.ice_check_min_interval) {
>-      return *config_.ice_check_min_interval;
>-    }
>-    return weak_ping_interval_;
>+    return std::max(config_.ice_check_interval_weak_connectivity_or_default(),
>+                    config_.ice_check_min_interval_or_default());
>   }
> 
>   int strong_ping_interval() const {
>-    if (config_.ice_check_min_interval &&
>-        STRONG_PING_INTERVAL < *config_.ice_check_min_interval) {
>-      return *config_.ice_check_min_interval;
>-    }
>-    return STRONG_PING_INTERVAL;
>+    return std::max(config_.ice_check_interval_strong_connectivity_or_default(),
>+                    config_.ice_check_min_interval_or_default());
>   }
> 
>   // Returns true if it's possible to send packets on |connection|.
>   bool ReadyToSend(Connection* connection) const;
>   void UpdateConnectionStates();
>-  void RequestSortAndStateUpdate();
>+  void RequestSortAndStateUpdate(const std::string& reason_to_sort);
>   // Start pinging if we haven't already started, and we now have a connection
>   // that's pingable.
>   void MaybeStartPinging();
> 
>+  int CompareCandidatePairNetworks(
>+      const Connection* a,
>+      const Connection* b,
>+      absl::optional<rtc::AdapterType> network_preference) const;
>+
>   // The methods below return a positive value if |a| is preferable to |b|,
>   // a negative value if |b| is preferable, and 0 if they're equally preferable.
>   // If |receiving_unchanged_threshold| is set, then when |b| is receiving and
>@@ -205,7 +220,7 @@ class P2PTransportChannel : public IceTransportInternal,
>   int CompareConnectionStates(
>       const cricket::Connection* a,
>       const cricket::Connection* b,
>-      rtc::Optional<int64_t> receiving_unchanged_threshold,
>+      absl::optional<int64_t> receiving_unchanged_threshold,
>       bool* missed_receiving_unchanged_threshold) const;
>   int CompareConnectionCandidates(const cricket::Connection* a,
>                                   const cricket::Connection* b) const;
>@@ -216,12 +231,12 @@ class P2PTransportChannel : public IceTransportInternal,
>   // Returns a positive value if |a| is better than |b|.
>   int CompareConnections(const cricket::Connection* a,
>                          const cricket::Connection* b,
>-                         rtc::Optional<int64_t> receiving_unchanged_threshold,
>+                         absl::optional<int64_t> receiving_unchanged_threshold,
>                          bool* missed_receiving_unchanged_threshold) const;
> 
>   bool PresumedWritable(const cricket::Connection* conn) const;
> 
>-  void SortConnectionsAndUpdateState();
>+  void SortConnectionsAndUpdateState(const std::string& reason_to_sort);
>   void SwitchSelectedConnection(Connection* conn);
>   void UpdateState();
>   void HandleAllTimedOut();
>@@ -251,10 +266,10 @@ class P2PTransportChannel : public IceTransportInternal,
>   void AddAllocatorSession(std::unique_ptr<PortAllocatorSession> session);
>   void AddConnection(Connection* connection);
> 
>-  void OnPortReady(PortAllocatorSession *session, PortInterface* port);
>+  void OnPortReady(PortAllocatorSession* session, PortInterface* port);
>   void OnPortsPruned(PortAllocatorSession* session,
>                      const std::vector<PortInterface*>& ports);
>-  void OnCandidatesReady(PortAllocatorSession *session,
>+  void OnCandidatesReady(PortAllocatorSession* session,
>                          const std::vector<Candidate>& candidates);
>   void OnCandidatesRemoved(PortAllocatorSession* session,
>                            const std::vector<Candidate>& candidates);
>@@ -275,18 +290,20 @@ class P2PTransportChannel : public IceTransportInternal,
>   void OnRoleConflict(PortInterface* port);
> 
>   void OnConnectionStateChange(Connection* connection);
>-  void OnReadPacket(Connection *connection, const char *data, size_t len,
>+  void OnReadPacket(Connection* connection,
>+                    const char* data,
>+                    size_t len,
>                     const rtc::PacketTime& packet_time);
>   void OnSentPacket(const rtc::SentPacket& sent_packet);
>   void OnReadyToSend(Connection* connection);
>-  void OnConnectionDestroyed(Connection *connection);
>+  void OnConnectionDestroyed(Connection* connection);
> 
>   void OnNominated(Connection* conn);
> 
>-  void OnMessage(rtc::Message* pmsg) override;
>-  void OnCheckAndPing();
>-  void OnRegatherOnFailedNetworks();
>-  void OnRegatherOnAllNetworks();
>+  void CheckAndPing();
>+
>+  void LogCandidatePairConfig(Connection* conn,
>+                              webrtc::IceCandidatePairConfigType type);
> 
>   uint32_t GetNominationAttr(Connection* conn) const;
>   bool GetUseCandidateAttr(Connection* conn, NominationMode mode) const;
>@@ -335,10 +352,6 @@ class P2PTransportChannel : public IceTransportInternal,
>                : static_cast<uint32_t>(remote_ice_parameters_.size() - 1);
>   }
> 
>-  // Samples a delay from the uniform distribution defined by the
>-  // regather_on_all_networks_interval ICE configuration pair.
>-  int SampleRegatherAllNetworksInterval();
>-
>   // Indicates if the given local port has been pruned.
>   bool IsPortPruned(const Port* port) const;
> 
>@@ -353,6 +366,7 @@ class P2PTransportChannel : public IceTransportInternal,
>   std::string transport_name_;
>   int component_;
>   PortAllocator* allocator_;
>+  webrtc::AsyncResolverFactory* async_resolver_factory_;
>   rtc::Thread* network_thread_;
>   bool incoming_only_;
>   int error_;
>@@ -371,7 +385,7 @@ class P2PTransportChannel : public IceTransportInternal,
>   // |pinged_connections_| and |unpinged_connections_| has the same
>   // connections as |connections_|. These 2 sets maintain whether a
>   // connection should be pinged next or not.
>-  std::vector<Connection *> connections_;
>+  std::vector<Connection*> connections_;
>   std::set<Connection*> pinged_connections_;
>   std::set<Connection*> unpinged_connections_;
> 
>@@ -388,11 +402,7 @@ class P2PTransportChannel : public IceTransportInternal,
>   IceRole ice_role_;
>   uint64_t tiebreaker_;
>   IceGatheringState gathering_state_;
>-
>-  // Used to generate random intervals for regather_all_networks_interval_range.
>-  webrtc::Random rand_;
>-
>-  int check_receiving_interval_;
>+  std::unique_ptr<webrtc::BasicRegatheringController> regathering_controller_;
>   int64_t last_ping_sent_ms_ = 0;
>   int weak_ping_interval_ = WEAK_PING_INTERVAL;
>   IceTransportState state_ = IceTransportState::STATE_INIT;
>@@ -405,9 +415,9 @@ class P2PTransportChannel : public IceTransportInternal,
>   bool receiving_ = false;
>   bool writable_ = false;
> 
>-  webrtc::MetricsObserverInterface* metrics_observer_ = nullptr;
>-
>-  rtc::Optional<rtc::NetworkRoute> network_route_;
>+  rtc::AsyncInvoker invoker_;
>+  absl::optional<rtc::NetworkRoute> network_route_;
>+  webrtc::IceEventLog ice_event_log_;
> 
>   RTC_DISALLOW_COPY_AND_ASSIGN(P2PTransportChannel);
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/p2ptransportchannel_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/p2ptransportchannel_unittest.cc
>index fc2b65c8b3d..1c3ae9ab778 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/p2ptransportchannel_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/p2ptransportchannel_unittest.cc
>@@ -12,7 +12,7 @@
> #include <list>
> #include <memory>
> 
>-#include "api/fakemetricsobserver.h"
>+#include "absl/memory/memory.h"
> #include "p2p/base/fakeportallocator.h"
> #include "p2p/base/icetransportinternal.h"
> #include "p2p/base/p2ptransportchannel.h"
>@@ -32,11 +32,11 @@
> #include "rtc_base/natserver.h"
> #include "rtc_base/natsocketfactory.h"
> #include "rtc_base/proxyserver.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/socketaddress.h"
> #include "rtc_base/ssladapter.h"
> #include "rtc_base/thread.h"
> #include "rtc_base/virtualsocketserver.h"
>+#include "system_wrappers/include/metrics_default.h"
> 
> namespace {
> 
>@@ -52,9 +52,10 @@ static const int kOnlyLocalPorts = cricket::PORTALLOCATOR_DISABLE_STUN |
>                                    cricket::PORTALLOCATOR_DISABLE_RELAY |
>                                    cricket::PORTALLOCATOR_DISABLE_TCP;
> static const int LOW_RTT = 20;
>+static const SocketAddress kAnyAddr("0.0.0.0", 0);
> // Addresses on the public internet.
>-static const SocketAddress kPublicAddrs[2] =
>-    { SocketAddress("11.11.11.11", 0), SocketAddress("22.22.22.22", 0) };
>+static const SocketAddress kPublicAddrs[2] = {SocketAddress("11.11.11.11", 0),
>+                                              SocketAddress("22.22.22.22", 0)};
> // IPv6 Addresses on the public internet.
> static const SocketAddress kIPv6PublicAddrs[2] = {
>     SocketAddress("2400:4030:1:2c00:be30:abcd:efab:cdef", 0),
>@@ -66,23 +67,23 @@ static const SocketAddress kIPv6AlternateAddrs[2] = {
>     SocketAddress("2401:4030:1:2c00:be30:abcd:efab:cdef", 0),
>     SocketAddress("2601:0:1000:1b03:2e41:38ff:fea6:f2a4", 0)};
> // Addresses for HTTP proxy servers.
>-static const SocketAddress kHttpsProxyAddrs[2] =
>-    { SocketAddress("11.11.11.1", 443), SocketAddress("22.22.22.1", 443) };
>+static const SocketAddress kHttpsProxyAddrs[2] = {
>+    SocketAddress("11.11.11.1", 443), SocketAddress("22.22.22.1", 443)};
> // Addresses for SOCKS proxy servers.
>-static const SocketAddress kSocksProxyAddrs[2] =
>-    { SocketAddress("11.11.11.1", 1080), SocketAddress("22.22.22.1", 1080) };
>+static const SocketAddress kSocksProxyAddrs[2] = {
>+    SocketAddress("11.11.11.1", 1080), SocketAddress("22.22.22.1", 1080)};
> // Internal addresses for NAT boxes.
>-static const SocketAddress kNatAddrs[2] =
>-    { SocketAddress("192.168.1.1", 0), SocketAddress("192.168.2.1", 0) };
>+static const SocketAddress kNatAddrs[2] = {SocketAddress("192.168.1.1", 0),
>+                                           SocketAddress("192.168.2.1", 0)};
> // Private addresses inside the NAT private networks.
>-static const SocketAddress kPrivateAddrs[2] =
>-    { SocketAddress("192.168.1.11", 0), SocketAddress("192.168.2.22", 0) };
>+static const SocketAddress kPrivateAddrs[2] = {
>+    SocketAddress("192.168.1.11", 0), SocketAddress("192.168.2.22", 0)};
> // For cascaded NATs, the internal addresses of the inner NAT boxes.
>-static const SocketAddress kCascadedNatAddrs[2] =
>-    { SocketAddress("192.168.10.1", 0), SocketAddress("192.168.20.1", 0) };
>+static const SocketAddress kCascadedNatAddrs[2] = {
>+    SocketAddress("192.168.10.1", 0), SocketAddress("192.168.20.1", 0)};
> // For cascaded NATs, private addresses inside the inner private networks.
>-static const SocketAddress kCascadedPrivateAddrs[2] =
>-    { SocketAddress("192.168.10.11", 0), SocketAddress("192.168.20.22", 0) };
>+static const SocketAddress kCascadedPrivateAddrs[2] = {
>+    SocketAddress("192.168.10.11", 0), SocketAddress("192.168.20.22", 0)};
> // The address of the public STUN server.
> static const SocketAddress kStunAddr("99.99.99.1", cricket::STUN_SERVER_PORT);
> // The addresses for the public turn server.
>@@ -113,7 +114,7 @@ enum { MSG_ADD_CANDIDATES, MSG_REMOVE_CANDIDATES };
> cricket::IceConfig CreateIceConfig(
>     int receiving_timeout,
>     cricket::ContinualGatheringPolicy continual_gathering_policy,
>-    int backup_ping_interval = -1) {
>+    absl::optional<int> backup_ping_interval = absl::nullopt) {
>   cricket::IceConfig config;
>   config.receiving_timeout = receiving_timeout;
>   config.continual_gathering_policy = continual_gathering_policy;
>@@ -155,6 +156,7 @@ cricket::BasicPortAllocator* CreateBasicPortAllocator(
> 
>   cricket::BasicPortAllocator* allocator =
>       new cricket::BasicPortAllocator(network_manager);
>+  allocator->Initialize();
>   allocator->SetConfiguration(stun_servers, turn_servers, 0, false);
>   return allocator;
> }
>@@ -205,31 +207,26 @@ class P2PTransportChannelTestBase : public testing::Test,
>     ep1_.allocator_.reset(
>         CreateBasicPortAllocator(&ep1_.network_manager_, stun_servers,
>                                  kTurnUdpIntAddr, rtc::SocketAddress()));
>-    ep1_.metrics_observer_ =
>-        new rtc::RefCountedObject<webrtc::FakeMetricsObserver>();
>-    ep1_.allocator_->SetMetricsObserver(ep1_.metrics_observer_);
>     ep2_.allocator_.reset(
>         CreateBasicPortAllocator(&ep2_.network_manager_, stun_servers,
>                                  kTurnUdpIntAddr, rtc::SocketAddress()));
>-    ep2_.metrics_observer_ =
>-        new rtc::RefCountedObject<webrtc::FakeMetricsObserver>();
>-    ep2_.allocator_->SetMetricsObserver(ep2_.metrics_observer_);
>+    webrtc::metrics::Reset();
>   }
> 
>  protected:
>   enum Config {
>-    OPEN,                           // Open to the Internet
>-    NAT_FULL_CONE,                  // NAT, no filtering
>-    NAT_ADDR_RESTRICTED,            // NAT, must send to an addr to recv
>-    NAT_PORT_RESTRICTED,            // NAT, must send to an addr+port to recv
>-    NAT_SYMMETRIC,                  // NAT, endpoint-dependent bindings
>-    NAT_DOUBLE_CONE,                // Double NAT, both cone
>-    NAT_SYMMETRIC_THEN_CONE,        // Double NAT, symmetric outer, cone inner
>-    BLOCK_UDP,                      // Firewall, UDP in/out blocked
>-    BLOCK_UDP_AND_INCOMING_TCP,     // Firewall, UDP in/out and TCP in blocked
>-    BLOCK_ALL_BUT_OUTGOING_HTTP,    // Firewall, only TCP out on 80/443
>-    PROXY_HTTPS,                    // All traffic through HTTPS proxy
>-    PROXY_SOCKS,                    // All traffic through SOCKS proxy
>+    OPEN,                         // Open to the Internet
>+    NAT_FULL_CONE,                // NAT, no filtering
>+    NAT_ADDR_RESTRICTED,          // NAT, must send to an addr to recv
>+    NAT_PORT_RESTRICTED,          // NAT, must send to an addr+port to recv
>+    NAT_SYMMETRIC,                // NAT, endpoint-dependent bindings
>+    NAT_DOUBLE_CONE,              // Double NAT, both cone
>+    NAT_SYMMETRIC_THEN_CONE,      // Double NAT, symmetric outer, cone inner
>+    BLOCK_UDP,                    // Firewall, UDP in/out blocked
>+    BLOCK_UDP_AND_INCOMING_TCP,   // Firewall, UDP in/out and TCP in blocked
>+    BLOCK_ALL_BUT_OUTGOING_HTTP,  // Firewall, only TCP out on 80/443
>+    PROXY_HTTPS,                  // All traffic through HTTPS proxy
>+    PROXY_SOCKS,                  // All traffic through SOCKS proxy
>     NUM_CONFIGS
>   };
> 
>@@ -259,7 +256,7 @@ class P2PTransportChannelTestBase : public testing::Test,
>     bool CheckData(const char* data, int len) {
>       bool ret = false;
>       if (!ch_packets_.empty()) {
>-        std::string packet =  ch_packets_.front();
>+        std::string packet = ch_packets_.front();
>         ret = (packet == std::string(data, len));
>         ch_packets_.pop_front();
>       }
>@@ -280,7 +277,7 @@ class P2PTransportChannelTestBase : public testing::Test,
>     Candidates candidates;
>   };
> 
>-  struct Endpoint {
>+  struct Endpoint : public sigslot::has_slots<> {
>     Endpoint()
>         : role_(ICEROLE_UNKNOWN),
>           tiebreaker_(0),
>@@ -311,10 +308,15 @@ class P2PTransportChannelTestBase : public testing::Test,
>       allocator_->set_allow_tcp_listen(allow_tcp_listen);
>     }
> 
>+    void OnIceRegathering(PortAllocatorSession*, IceRegatheringReason reason) {
>+      ++ice_regathering_counter_[reason];
>+    }
>+
>+    int GetIceRegatheringCountForReason(IceRegatheringReason reason) {
>+      return ice_regathering_counter_[reason];
>+    }
>+
>     rtc::FakeNetworkManager network_manager_;
>-    // |metrics_observer_| should outlive |allocator_| as the former may be
>-    // used by the latter.
>-    rtc::scoped_refptr<webrtc::FakeMetricsObserver> metrics_observer_;
>     std::unique_ptr<BasicPortAllocator> allocator_;
>     ChannelData cd1_;
>     ChannelData cd2_;
>@@ -324,6 +326,7 @@ class P2PTransportChannelTestBase : public testing::Test,
>     bool save_candidates_;
>     std::vector<std::unique_ptr<CandidatesData>> saved_candidates_;
>     bool ready_to_send_ = false;
>+    std::map<IceRegatheringReason, int> ice_regathering_counter_;
>   };
> 
>   ChannelData* GetChannelData(rtc::PacketTransportInternal* transport) {
>@@ -351,12 +354,14 @@ class P2PTransportChannelTestBase : public testing::Test,
>                                       ice_ep1_cd1_ch, ice_ep2_cd1_ch));
>     ep2_.cd1_.ch_.reset(CreateChannel(1, ICE_CANDIDATE_COMPONENT_DEFAULT,
>                                       ice_ep2_cd1_ch, ice_ep1_cd1_ch));
>-    ep1_.cd1_.ch_->SetMetricsObserver(ep1_.metrics_observer_);
>-    ep2_.cd1_.ch_->SetMetricsObserver(ep2_.metrics_observer_);
>     ep1_.cd1_.ch_->SetIceConfig(ep1_config);
>     ep2_.cd1_.ch_->SetIceConfig(ep2_config);
>     ep1_.cd1_.ch_->MaybeStartGathering();
>     ep2_.cd1_.ch_->MaybeStartGathering();
>+    ep1_.cd1_.ch_->allocator_session()->SignalIceRegathering.connect(
>+        &ep1_, &Endpoint::OnIceRegathering);
>+    ep2_.cd1_.ch_->allocator_session()->SignalIceRegathering.connect(
>+        &ep2_, &Endpoint::OnIceRegathering);
>   }
> 
>   void CreateChannels() {
>@@ -382,6 +387,8 @@ class P2PTransportChannelTestBase : public testing::Test,
>         this, &P2PTransportChannelTestBase::OnRoleConflict);
>     channel->SignalNetworkRouteChanged.connect(
>         this, &P2PTransportChannelTestBase::OnNetworkRouteChanged);
>+    channel->SignalSentPacket.connect(
>+        this, &P2PTransportChannelTestBase::OnSentPacket);
>     channel->SetIceParameters(local_ice);
>     if (remote_ice_parameter_source_ == FROM_SETICEPARAMETERS) {
>       channel->SetRemoteIceParameters(remote_ice);
>@@ -391,6 +398,7 @@ class P2PTransportChannelTestBase : public testing::Test,
>     return channel;
>   }
>   void DestroyChannels() {
>+    main_.Clear(this);
>     ep1_.cd1_.ch_.reset();
>     ep2_.cd1_.ch_.reset();
>     ep1_.cd2_.ch_.reset();
>@@ -422,6 +430,7 @@ class P2PTransportChannelTestBase : public testing::Test,
> 
>   rtc::NATSocketServer* nat() { return nss_.get(); }
>   rtc::FirewallSocketServer* fw() { return ss_.get(); }
>+  rtc::VirtualSocketServer* vss() { return vss_.get(); }
> 
>   Endpoint* GetEndpoint(int endpoint) {
>     if (endpoint == 0) {
>@@ -435,9 +444,6 @@ class P2PTransportChannelTestBase : public testing::Test,
>   BasicPortAllocator* GetAllocator(int endpoint) {
>     return GetEndpoint(endpoint)->allocator_.get();
>   }
>-  webrtc::FakeMetricsObserver* GetMetricsObserver(int endpoint) {
>-    return GetEndpoint(endpoint)->metrics_observer_;
>-  }
>   void AddAddress(int endpoint, const SocketAddress& addr) {
>     GetEndpoint(endpoint)->network_manager_.AddInterface(addr);
>   }
>@@ -455,8 +461,8 @@ class P2PTransportChannelTestBase : public testing::Test,
>   void SetProxy(int endpoint, rtc::ProxyType type) {
>     rtc::ProxyInfo info;
>     info.type = type;
>-    info.address = (type == rtc::PROXY_HTTPS) ?
>-        kHttpsProxyAddrs[endpoint] : kSocksProxyAddrs[endpoint];
>+    info.address = (type == rtc::PROXY_HTTPS) ? kHttpsProxyAddrs[endpoint]
>+                                              : kSocksProxyAddrs[endpoint];
>     GetAllocator(endpoint)->set_proxy("unittest/1.0", info);
>   }
>   void SetAllocatorFlags(int endpoint, int flags) {
>@@ -560,8 +566,8 @@ class P2PTransportChannelTestBase : public testing::Test,
>     if (connect_time < expected.connect_wait) {
>       RTC_LOG(LS_INFO) << "Connect time: " << connect_time << " ms";
>     } else {
>-      RTC_LOG(LS_INFO) << "Connect time: "
>-                       << "TIMEOUT (" << expected.connect_wait << " ms)";
>+      RTC_LOG(LS_INFO) << "Connect time: TIMEOUT (" << expected.connect_wait
>+                       << " ms)";
>     }
> 
>     // Allow a few turns of the crank for the selected connections to emerge.
>@@ -585,8 +591,8 @@ class P2PTransportChannelTestBase : public testing::Test,
>       if (converge_time < converge_wait) {
>         RTC_LOG(LS_INFO) << "Converge time: " << converge_time << " ms";
>       } else {
>-        RTC_LOG(LS_INFO) << "Converge time: "
>-                         << "TIMEOUT (" << converge_wait << " ms)";
>+        RTC_LOG(LS_INFO) << "Converge time: TIMEOUT (" << converge_wait
>+                         << " ms)";
>       }
>     }
>     // Try sending some data to other end.
>@@ -681,6 +687,13 @@ class P2PTransportChannelTestBase : public testing::Test,
>     TestSendRecv(&clock);
>   }
> 
>+  void TestPacketInfoIsSet(rtc::PacketInfo info) {
>+    EXPECT_NE(info.packet_type, rtc::PacketType::kUnknown);
>+    EXPECT_NE(info.protocol, rtc::PacketInfoProtocolType::kUnknown);
>+    EXPECT_TRUE(info.network_id.has_value());
>+    EXPECT_FALSE(info.local_socket_address.IsNil());
>+  }
>+
>   void OnReadyToSend(rtc::PacketTransportInternal* transport) {
>     GetEndpoint(transport)->ready_to_send_ = true;
>   }
>@@ -699,7 +712,7 @@ class P2PTransportChannelTestBase : public testing::Test,
>     }
>   }
> 
>-  void OnNetworkRouteChanged(rtc::Optional<rtc::NetworkRoute> network_route) {
>+  void OnNetworkRouteChanged(absl::optional<rtc::NetworkRoute> network_route) {
>     // If the |network_route| is unset, don't count. This is used in the case
>     // when the network on remote side is down, the signal will be fired with an
>     // unset network route and it shouldn't trigger a connection switch.
>@@ -804,6 +817,11 @@ class P2PTransportChannelTestBase : public testing::Test,
>     channel->SetIceRole(new_role);
>   }
> 
>+  void OnSentPacket(rtc::PacketTransportInternal* transport,
>+                    const rtc::SentPacket& packet) {
>+    TestPacketInfoIsSet(packet.info);
>+  }
>+
>   int SendData(IceTransportInternal* channel, const char* data, size_t len) {
>     rtc::PacketOptions options;
>     return channel->SendPacket(data, len, options, 0);
>@@ -858,9 +876,7 @@ class P2PTransportChannelTestBase : public testing::Test,
>     remote_ice_parameter_source_ = source;
>   }
> 
>-  void set_force_relay(bool relay) {
>-    force_relay_ = relay;
>-  }
>+  void set_force_relay(bool relay) { force_relay_ = relay; }
> 
>   void ConnectSignalNominated(Connection* conn) {
>     conn->SignalNominated.connect(this,
>@@ -1003,20 +1019,22 @@ class P2PTransportChannelTest : public P2PTransportChannelTestBase {
>       case NAT_SYMMETRIC:
>         AddAddress(endpoint, kPrivateAddrs[endpoint]);
>         // Add a single NAT of the desired type
>-        nat()->AddTranslator(kPublicAddrs[endpoint], kNatAddrs[endpoint],
>-            static_cast<rtc::NATType>(config - NAT_FULL_CONE))->
>-            AddClient(kPrivateAddrs[endpoint]);
>+        nat()
>+            ->AddTranslator(kPublicAddrs[endpoint], kNatAddrs[endpoint],
>+                            static_cast<rtc::NATType>(config - NAT_FULL_CONE))
>+            ->AddClient(kPrivateAddrs[endpoint]);
>         break;
>       case NAT_DOUBLE_CONE:
>       case NAT_SYMMETRIC_THEN_CONE:
>         AddAddress(endpoint, kCascadedPrivateAddrs[endpoint]);
>         // Add a two cascaded NATs of the desired types
>-        nat()->AddTranslator(kPublicAddrs[endpoint], kNatAddrs[endpoint],
>-            (config == NAT_DOUBLE_CONE) ?
>-                rtc::NAT_OPEN_CONE : rtc::NAT_SYMMETRIC)->
>-            AddTranslator(kPrivateAddrs[endpoint], kCascadedNatAddrs[endpoint],
>-                rtc::NAT_OPEN_CONE)->
>-                AddClient(kCascadedPrivateAddrs[endpoint]);
>+        nat()
>+            ->AddTranslator(kPublicAddrs[endpoint], kNatAddrs[endpoint],
>+                            (config == NAT_DOUBLE_CONE) ? rtc::NAT_OPEN_CONE
>+                                                        : rtc::NAT_SYMMETRIC)
>+            ->AddTranslator(kPrivateAddrs[endpoint],
>+                            kCascadedNatAddrs[endpoint], rtc::NAT_OPEN_CONE)
>+            ->AddClient(kCascadedPrivateAddrs[endpoint]);
>         break;
>       case BLOCK_UDP:
>       case BLOCK_UDP_AND_INCOMING_TCP:
>@@ -1025,12 +1043,13 @@ class P2PTransportChannelTest : public P2PTransportChannelTestBase {
>       case PROXY_SOCKS:
>         AddAddress(endpoint, kPublicAddrs[endpoint]);
>         // Block all UDP
>-        fw()->AddRule(false, rtc::FP_UDP, rtc::FD_ANY,
>-                      kPublicAddrs[endpoint]);
>+        fw()->AddRule(false, rtc::FP_UDP, rtc::FD_ANY, kPublicAddrs[endpoint]);
>+        fw()->AddRule(false, rtc::FP_UDP, rtc::FD_ANY, kAnyAddr);
>         if (config == BLOCK_UDP_AND_INCOMING_TCP) {
>           // Block TCP inbound to the endpoint
>           fw()->AddRule(false, rtc::FP_TCP, SocketAddress(),
>                         kPublicAddrs[endpoint]);
>+          fw()->AddRule(false, rtc::FP_TCP, SocketAddress(), kAnyAddr);
>         } else if (config == BLOCK_ALL_BUT_OUTGOING_HTTP) {
>           // Block all TCP to/from the endpoint except 80/443 out
>           fw()->AddRule(true, rtc::FP_TCP, kPublicAddrs[endpoint],
>@@ -1039,12 +1058,14 @@ class P2PTransportChannelTest : public P2PTransportChannelTestBase {
>                         SocketAddress(rtc::IPAddress(INADDR_ANY), 443));
>           fw()->AddRule(false, rtc::FP_TCP, rtc::FD_ANY,
>                         kPublicAddrs[endpoint]);
>+          fw()->AddRule(false, rtc::FP_TCP, rtc::FD_ANY, kAnyAddr);
>         } else if (config == PROXY_HTTPS) {
>           // Block all TCP to/from the endpoint except to the proxy server
>           fw()->AddRule(true, rtc::FP_TCP, kPublicAddrs[endpoint],
>                         kHttpsProxyAddrs[endpoint]);
>           fw()->AddRule(false, rtc::FP_TCP, rtc::FD_ANY,
>                         kPublicAddrs[endpoint]);
>+          fw()->AddRule(false, rtc::FP_TCP, rtc::FD_ANY, kAnyAddr);
>           SetProxy(endpoint, rtc::PROXY_HTTPS);
>         } else if (config == PROXY_SOCKS) {
>           // Block all TCP to/from the endpoint except to the proxy server
>@@ -1052,6 +1073,7 @@ class P2PTransportChannelTest : public P2PTransportChannelTestBase {
>                         kSocksProxyAddrs[endpoint]);
>           fw()->AddRule(false, rtc::FP_TCP, rtc::FD_ANY,
>                         kPublicAddrs[endpoint]);
>+          fw()->AddRule(false, rtc::FP_TCP, rtc::FD_ANY, kAnyAddr);
>           SetProxy(endpoint, rtc::PROXY_SOCKS5);
>         }
>         break;
>@@ -1088,19 +1110,43 @@ class P2PTransportChannelTest : public P2PTransportChannelTestBase {
> // TODO(?): Rearrange rows/columns from best to worst.
> const P2PTransportChannelTest::Result*
>     P2PTransportChannelTest::kMatrix[NUM_CONFIGS][NUM_CONFIGS] = {
>-//      OPEN  CONE  ADDR  PORT  SYMM  2CON  SCON  !UDP  !TCP  HTTP  PRXH  PRXS
>-/*OP*/ {LULU, LUSU, LUSU, LUSU, LUPU, LUSU, LUPU, LTPT, LTPT, LSRS, NULL, LTPT},
>-/*CO*/ {SULU, SUSU, SUSU, SUSU, SUPU, SUSU, SUPU, NULL, NULL, LSRS, NULL, LTRT},
>-/*AD*/ {SULU, SUSU, SUSU, SUSU, SUPU, SUSU, SUPU, NULL, NULL, LSRS, NULL, LTRT},
>-/*PO*/ {SULU, SUSU, SUSU, SUSU, RUPU, SUSU, RUPU, NULL, NULL, LSRS, NULL, LTRT},
>-/*SY*/ {PULU, PUSU, PUSU, PURU, PURU, PUSU, PURU, NULL, NULL, LSRS, NULL, LTRT},
>-/*2C*/ {SULU, SUSU, SUSU, SUSU, SUPU, SUSU, SUPU, NULL, NULL, LSRS, NULL, LTRT},
>-/*SC*/ {PULU, PUSU, PUSU, PURU, PURU, PUSU, PURU, NULL, NULL, LSRS, NULL, LTRT},
>-/*!U*/ {LTPT, NULL, NULL, NULL, NULL, NULL, NULL, LTPT, LTPT, LSRS, NULL, LTRT},
>-/*!T*/ {PTLT, NULL, NULL, NULL, NULL, NULL, NULL, PTLT, LTRT, LSRS, NULL, LTRT},
>-/*HT*/ {LSRS, LSRS, LSRS, LSRS, LSRS, LSRS, LSRS, LSRS, LSRS, LSRS, NULL, LSRS},
>-/*PR*/ {NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL},
>-/*PR*/ {LTRT, LTRT, LTRT, LTRT, LTRT, LTRT, LTRT, LTRT, LTRT, LSRS, NULL, LTRT},
>+        //      OPEN  CONE  ADDR  PORT  SYMM  2CON  SCON  !UDP  !TCP  HTTP  PRXH
>+        //      PRXS
>+        /*OP*/ {LULU, LUSU, LUSU, LUSU, LUPU, LUSU, LUPU, LTPT, LTPT, LSRS,
>+                NULL, LTPT},
>+        /*CO*/
>+        {SULU, SUSU, SUSU, SUSU, SUPU, SUSU, SUPU, NULL, NULL, LSRS, NULL,
>+         LTRT},
>+        /*AD*/
>+        {SULU, SUSU, SUSU, SUSU, SUPU, SUSU, SUPU, NULL, NULL, LSRS, NULL,
>+         LTRT},
>+        /*PO*/
>+        {SULU, SUSU, SUSU, SUSU, RUPU, SUSU, RUPU, NULL, NULL, LSRS, NULL,
>+         LTRT},
>+        /*SY*/
>+        {PULU, PUSU, PUSU, PURU, PURU, PUSU, PURU, NULL, NULL, LSRS, NULL,
>+         LTRT},
>+        /*2C*/
>+        {SULU, SUSU, SUSU, SUSU, SUPU, SUSU, SUPU, NULL, NULL, LSRS, NULL,
>+         LTRT},
>+        /*SC*/
>+        {PULU, PUSU, PUSU, PURU, PURU, PUSU, PURU, NULL, NULL, LSRS, NULL,
>+         LTRT},
>+        /*!U*/
>+        {LTPT, NULL, NULL, NULL, NULL, NULL, NULL, LTPT, LTPT, LSRS, NULL,
>+         LTRT},
>+        /*!T*/
>+        {PTLT, NULL, NULL, NULL, NULL, NULL, NULL, PTLT, LTRT, LSRS, NULL,
>+         LTRT},
>+        /*HT*/
>+        {LSRS, LSRS, LSRS, LSRS, LSRS, LSRS, LSRS, LSRS, LSRS, LSRS, NULL,
>+         LSRS},
>+        /*PR*/
>+        {NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL, NULL,
>+         NULL},
>+        /*PR*/
>+        {LTRT, LTRT, LTRT, LTRT, LTRT, LTRT, LTRT, LTRT, LTRT, LSRS, NULL,
>+         LTRT},
> };
> 
> // The actual tests that exercise all the various configurations.
>@@ -1176,8 +1222,10 @@ TEST_F(P2PTransportChannelTest, GetStats) {
>                              kMediumTimeout, clock);
>   TestSendRecv(&clock);
>   ConnectionInfos infos;
>-  ASSERT_TRUE(ep1_ch1()->GetStats(&infos));
>+  CandidateStatsList candidate_stats_list;
>+  ASSERT_TRUE(ep1_ch1()->GetStats(&infos, &candidate_stats_list));
>   ASSERT_GE(infos.size(), 1u);
>+  ASSERT_GE(candidate_stats_list.size(), 1u);
>   ConnectionInfo* best_conn_info = nullptr;
>   for (ConnectionInfo& info : infos) {
>     if (info.best_connection) {
>@@ -1218,15 +1266,15 @@ TEST_F(P2PTransportChannelTest, TestUMAIceRestartWhileDisconnected) {
>   ep1_ch1()->SetIceParameters(kIceParams[2]);
>   ep1_ch1()->SetRemoteIceParameters(kIceParams[3]);
>   ep1_ch1()->MaybeStartGathering();
>-  EXPECT_EQ(1, GetMetricsObserver(0)->GetEnumCounter(
>-                   webrtc::kEnumCounterIceRestart,
>+  EXPECT_EQ(1, webrtc::metrics::NumEvents(
>+                   "WebRTC.PeerConnection.IceRestartState",
>                    static_cast<int>(IceRestartState::DISCONNECTED)));
> 
>   ep2_ch1()->SetIceParameters(kIceParams[3]);
>   ep2_ch1()->SetRemoteIceParameters(kIceParams[2]);
>   ep2_ch1()->MaybeStartGathering();
>-  EXPECT_EQ(1, GetMetricsObserver(1)->GetEnumCounter(
>-                   webrtc::kEnumCounterIceRestart,
>+  EXPECT_EQ(2, webrtc::metrics::NumEvents(
>+                   "WebRTC.PeerConnection.IceRestartState",
>                    static_cast<int>(IceRestartState::DISCONNECTED)));
> 
>   DestroyChannels();
>@@ -1247,15 +1295,15 @@ TEST_F(P2PTransportChannelTest, TestUMAIceRestartWhileConnected) {
>   ep1_ch1()->SetIceParameters(kIceParams[2]);
>   ep1_ch1()->SetRemoteIceParameters(kIceParams[3]);
>   ep1_ch1()->MaybeStartGathering();
>-  EXPECT_EQ(1, GetMetricsObserver(0)->GetEnumCounter(
>-                   webrtc::kEnumCounterIceRestart,
>+  EXPECT_EQ(1, webrtc::metrics::NumEvents(
>+                   "WebRTC.PeerConnection.IceRestartState",
>                    static_cast<int>(IceRestartState::CONNECTED)));
> 
>   ep2_ch1()->SetIceParameters(kIceParams[3]);
>   ep2_ch1()->SetRemoteIceParameters(kIceParams[2]);
>   ep2_ch1()->MaybeStartGathering();
>-  EXPECT_EQ(1, GetMetricsObserver(1)->GetEnumCounter(
>-                   webrtc::kEnumCounterIceRestart,
>+  EXPECT_EQ(2, webrtc::metrics::NumEvents(
>+                   "WebRTC.PeerConnection.IceRestartState",
>                    static_cast<int>(IceRestartState::CONNECTED)));
> 
>   DestroyChannels();
>@@ -1273,15 +1321,15 @@ TEST_F(P2PTransportChannelTest, TestUMAIceRestartWhileConnecting) {
>   ep1_ch1()->SetIceParameters(kIceParams[2]);
>   ep1_ch1()->SetRemoteIceParameters(kIceParams[3]);
>   ep1_ch1()->MaybeStartGathering();
>-  EXPECT_EQ(1, GetMetricsObserver(0)->GetEnumCounter(
>-                   webrtc::kEnumCounterIceRestart,
>+  EXPECT_EQ(1, webrtc::metrics::NumEvents(
>+                   "WebRTC.PeerConnection.IceRestartState",
>                    static_cast<int>(IceRestartState::CONNECTING)));
> 
>   ep2_ch1()->SetIceParameters(kIceParams[3]);
>   ep2_ch1()->SetRemoteIceParameters(kIceParams[2]);
>   ep2_ch1()->MaybeStartGathering();
>-  EXPECT_EQ(1, GetMetricsObserver(1)->GetEnumCounter(
>-                   webrtc::kEnumCounterIceRestart,
>+  EXPECT_EQ(2, webrtc::metrics::NumEvents(
>+                   "WebRTC.PeerConnection.IceRestartState",
>                    static_cast<int>(IceRestartState::CONNECTING)));
> 
>   DestroyChannels();
>@@ -1307,11 +1355,10 @@ TEST_F(P2PTransportChannelTest,
> 
>   // Adding address in ep1 will trigger continual gathering.
>   AddAddress(0, kAlternateAddrs[0]);
>-  EXPECT_EQ_SIMULATED_WAIT(
>-      1, GetMetricsObserver(0)->GetEnumCounter(
>-             webrtc::kEnumCounterIceRegathering,
>-             static_cast<int>(IceRegatheringReason::NETWORK_CHANGE)),
>-      kDefaultTimeout, clock);
>+  EXPECT_EQ_SIMULATED_WAIT(1,
>+                           GetEndpoint(0)->GetIceRegatheringCountForReason(
>+                               IceRegatheringReason::NETWORK_CHANGE),
>+                           kDefaultTimeout, clock);
> 
>   ep2_ch1()->SetIceParameters(kIceParams[3]);
>   ep2_ch1()->SetRemoteIceParameters(kIceParams[2]);
>@@ -1320,9 +1367,8 @@ TEST_F(P2PTransportChannelTest,
>   AddAddress(1, kAlternateAddrs[1]);
>   SIMULATED_WAIT(false, kDefaultTimeout, clock);
>   // ep2 has not enabled continual gathering.
>-  EXPECT_EQ(0, GetMetricsObserver(1)->GetEnumCounter(
>-                   webrtc::kEnumCounterIceRegathering,
>-                   static_cast<int>(IceRegatheringReason::NETWORK_CHANGE)));
>+  EXPECT_EQ(0, GetEndpoint(1)->GetIceRegatheringCountForReason(
>+                   IceRegatheringReason::NETWORK_CHANGE));
> 
>   DestroyChannels();
> }
>@@ -1350,12 +1396,13 @@ TEST_F(P2PTransportChannelTest,
>   // Timeout value such that all connections are deleted.
>   const int kNetworkFailureTimeout = 35000;
>   SIMULATED_WAIT(false, kNetworkFailureTimeout, clock);
>-  EXPECT_LE(1, GetMetricsObserver(0)->GetEnumCounter(
>-                   webrtc::kEnumCounterIceRegathering,
>-                   static_cast<int>(IceRegatheringReason::NETWORK_FAILURE)));
>-  EXPECT_EQ(0, GetMetricsObserver(1)->GetEnumCounter(
>-                   webrtc::kEnumCounterIceRegathering,
>+  EXPECT_LE(1, GetEndpoint(0)->GetIceRegatheringCountForReason(
>+                   IceRegatheringReason::NETWORK_FAILURE));
>+  EXPECT_LE(1, webrtc::metrics::NumEvents(
>+                   "WebRTC.PeerConnection.IceRegatheringReason",
>                    static_cast<int>(IceRegatheringReason::NETWORK_FAILURE)));
>+  EXPECT_EQ(0, GetEndpoint(1)->GetIceRegatheringCountForReason(
>+                   IceRegatheringReason::NETWORK_FAILURE));
> 
>   DestroyChannels();
> }
>@@ -1369,8 +1416,8 @@ TEST_F(P2PTransportChannelTest, TestIceRegatherOnAllNetworksContinual) {
>   // ep1 gathers continually but ep2 does not.
>   const int kRegatherInterval = 2000;
>   IceConfig config1 = CreateIceConfig(1000, GATHER_CONTINUALLY);
>-  config1.regather_all_networks_interval_range.emplace(
>-      kRegatherInterval, kRegatherInterval);
>+  config1.regather_all_networks_interval_range.emplace(kRegatherInterval,
>+                                                       kRegatherInterval);
>   IceConfig config2;
>   CreateChannels(config1, config2);
> 
>@@ -1384,13 +1431,14 @@ TEST_F(P2PTransportChannelTest, TestIceRegatherOnAllNetworksContinual) {
>   const int kNetworkGatherDuration = 11000;
>   SIMULATED_WAIT(false, kNetworkGatherDuration, clock);
>   // Expect regathering to happen 5 times in 11s with 2s interval.
>-  EXPECT_LE(5, GetMetricsObserver(0)->GetEnumCounter(
>-                   webrtc::kEnumCounterIceRegathering,
>+  EXPECT_LE(5, GetEndpoint(0)->GetIceRegatheringCountForReason(
>+                   IceRegatheringReason::OCCASIONAL_REFRESH));
>+  EXPECT_LE(5, webrtc::metrics::NumEvents(
>+                   "WebRTC.PeerConnection.IceRegatheringReason",
>                    static_cast<int>(IceRegatheringReason::OCCASIONAL_REFRESH)));
>   // Expect no regathering if continual gathering not configured.
>-  EXPECT_EQ(0, GetMetricsObserver(1)->GetEnumCounter(
>-                   webrtc::kEnumCounterIceRegathering,
>-                   static_cast<int>(IceRegatheringReason::OCCASIONAL_REFRESH)));
>+  EXPECT_EQ(0, GetEndpoint(1)->GetIceRegatheringCountForReason(
>+                   IceRegatheringReason::OCCASIONAL_REFRESH));
> 
>   DestroyChannels();
> }
>@@ -1404,7 +1452,7 @@ class P2PTransportRegatherAllNetworksTest : public P2PTransportChannelTest {
>   void TestWithRoles(IceRole p1_role, IceRole p2_role) {
>     rtc::ScopedFakeClock clock;
>     ConfigureEndpoints(NAT_SYMMETRIC, NAT_SYMMETRIC, kDefaultPortAllocatorFlags,
>-        kDefaultPortAllocatorFlags);
>+                       kDefaultPortAllocatorFlags);
>     set_force_relay(true);
> 
>     const int kRegatherInterval = 2000;
>@@ -1412,8 +1460,8 @@ class P2PTransportRegatherAllNetworksTest : public P2PTransportChannelTest {
> 
>     // Set up peer 1 to auto regather every 2s.
>     IceConfig config1 = CreateIceConfig(1000, GATHER_CONTINUALLY);
>-    config1.regather_all_networks_interval_range.emplace(
>-        kRegatherInterval, kRegatherInterval);
>+    config1.regather_all_networks_interval_range.emplace(kRegatherInterval,
>+                                                         kRegatherInterval);
>     IceConfig config2 = CreateIceConfig(1000, GATHER_CONTINUALLY);
> 
>     // Set peer roles.
>@@ -1423,11 +1471,10 @@ class P2PTransportRegatherAllNetworksTest : public P2PTransportChannelTest {
>     CreateChannels(config1, config2);
> 
>     // Wait for initial connection to be made.
>-    EXPECT_TRUE_SIMULATED_WAIT(ep1_ch1()->receiving() &&
>-                                   ep1_ch1()->writable() &&
>-                                   ep2_ch1()->receiving() &&
>-                                   ep2_ch1()->writable(),
>-                               kMediumTimeout, clock);
>+    EXPECT_TRUE_SIMULATED_WAIT(
>+        ep1_ch1()->receiving() && ep1_ch1()->writable() &&
>+            ep2_ch1()->receiving() && ep2_ch1()->writable(),
>+        kMediumTimeout, clock);
> 
>     const Connection* initial_selected = ep1_ch1()->selected_connection();
> 
>@@ -1436,9 +1483,8 @@ class P2PTransportRegatherAllNetworksTest : public P2PTransportChannelTest {
>     const int kWaitRegather =
>         kRegatherInterval * kNumRegathers + kRegatherInterval / 2;
>     SIMULATED_WAIT(false, kWaitRegather, clock);
>-    EXPECT_EQ(kNumRegathers, GetMetricsObserver(0)->GetEnumCounter(
>-        webrtc::kEnumCounterIceRegathering,
>-        static_cast<int>(IceRegatheringReason::OCCASIONAL_REFRESH)));
>+    EXPECT_EQ(kNumRegathers, GetEndpoint(0)->GetIceRegatheringCountForReason(
>+                                 IceRegatheringReason::OCCASIONAL_REFRESH));
> 
>     const Connection* new_selected = ep1_ch1()->selected_connection();
> 
>@@ -1486,14 +1532,14 @@ TEST_F(P2PTransportChannelTest, PeerReflexiveCandidateBeforeSignaling) {
>   ASSERT_TRUE_WAIT(
>       (selected_connection = ep1_ch1()->selected_connection()) != nullptr,
>       kMediumTimeout);
>-  EXPECT_EQ("prflx", selected_connection->remote_candidate().type());
>+  EXPECT_EQ(PRFLX_PORT_TYPE, selected_connection->remote_candidate().type());
>   EXPECT_EQ(kIceUfrag[1], selected_connection->remote_candidate().username());
>   EXPECT_EQ(kIcePwd[1], selected_connection->remote_candidate().password());
>   EXPECT_EQ(1u, selected_connection->remote_candidate().generation());
> 
>   ResumeCandidates(1);
>   // Verify ep1's selected connection is updated to use the 'local' candidate.
>-  EXPECT_EQ_WAIT("local",
>+  EXPECT_EQ_WAIT(LOCAL_PORT_TYPE,
>                  ep1_ch1()->selected_connection()->remote_candidate().type(),
>                  kMediumTimeout);
>   EXPECT_EQ(selected_connection, ep1_ch1()->selected_connection());
>@@ -1528,14 +1574,14 @@ TEST_F(P2PTransportChannelTest, PeerReflexiveCandidateBeforeSignalingWithNAT) {
>   ASSERT_TRUE_WAIT(
>       (selected_connection = ep1_ch1()->selected_connection()) != nullptr,
>       kMediumTimeout);
>-  EXPECT_EQ("prflx", selected_connection->remote_candidate().type());
>+  EXPECT_EQ(PRFLX_PORT_TYPE, selected_connection->remote_candidate().type());
>   EXPECT_EQ(kIceUfrag[1], selected_connection->remote_candidate().username());
>   EXPECT_EQ(kIcePwd[1], selected_connection->remote_candidate().password());
>   EXPECT_EQ(1u, selected_connection->remote_candidate().generation());
> 
>   ResumeCandidates(1);
> 
>-  EXPECT_EQ_WAIT("prflx",
>+  EXPECT_EQ_WAIT(PRFLX_PORT_TYPE,
>                  ep1_ch1()->selected_connection()->remote_candidate().type(),
>                  kMediumTimeout);
>   EXPECT_EQ(selected_connection, ep1_ch1()->selected_connection());
>@@ -1579,7 +1625,7 @@ TEST_F(P2PTransportChannelTest,
> 
>   // The caller should have the selected connection connected to the peer
>   // reflexive candidate.
>-  EXPECT_EQ_WAIT("prflx",
>+  EXPECT_EQ_WAIT(PRFLX_PORT_TYPE,
>                  ep1_ch1()->selected_connection()->remote_candidate().type(),
>                  kDefaultTimeout);
>   const Connection* prflx_selected_connection =
>@@ -1595,7 +1641,7 @@ TEST_F(P2PTransportChannelTest,
>   // their information to update the peer reflexive candidate.
>   ResumeCandidates(1);
> 
>-  EXPECT_EQ_WAIT("relay",
>+  EXPECT_EQ_WAIT(RELAY_PORT_TYPE,
>                  ep1_ch1()->selected_connection()->remote_candidate().type(),
>                  kDefaultTimeout);
>   EXPECT_EQ(prflx_selected_connection, ep1_ch1()->selected_connection());
>@@ -1667,16 +1713,8 @@ TEST_F(P2PTransportChannelTest, IncomingOnlyOpen) {
> // connections. This has been observed in some scenarios involving
> // VPNs/firewalls.
> TEST_F(P2PTransportChannelTest, CanOnlyMakeOutgoingTcpConnections) {
>-  // The PORTALLOCATOR_ENABLE_ANY_ADDRESS_PORTS flag is required if the
>-  // application needs this use case to work, since the application must accept
>-  // the tradeoff that more candidates need to be allocated.
>-  //
>-  // TODO(deadbeef): Later, make this flag the default, and do more elegant
>-  // things to ensure extra candidates don't waste resources?
>-  ConfigureEndpoints(
>-      OPEN, OPEN,
>-      kDefaultPortAllocatorFlags | PORTALLOCATOR_ENABLE_ANY_ADDRESS_PORTS,
>-      kDefaultPortAllocatorFlags);
>+  ConfigureEndpoints(OPEN, OPEN, kDefaultPortAllocatorFlags,
>+                     kDefaultPortAllocatorFlags);
>   // In order to simulate nothing working but outgoing TCP connections, prevent
>   // the endpoint from binding to its interface's address as well as the
>   // "any" addresses. It can then only make a connection by using "Connect()".
>@@ -1797,26 +1835,16 @@ TEST_F(P2PTransportChannelTest, TestDefaultDscpValue) {
>   AddAddress(1, kPublicAddrs[1]);
> 
>   CreateChannels();
>-  EXPECT_EQ(rtc::DSCP_NO_CHANGE,
>-            GetEndpoint(0)->cd1_.ch_->DefaultDscpValue());
>-  EXPECT_EQ(rtc::DSCP_NO_CHANGE,
>-            GetEndpoint(1)->cd1_.ch_->DefaultDscpValue());
>-  GetEndpoint(0)->cd1_.ch_->SetOption(
>-      rtc::Socket::OPT_DSCP, rtc::DSCP_CS6);
>-  GetEndpoint(1)->cd1_.ch_->SetOption(
>-      rtc::Socket::OPT_DSCP, rtc::DSCP_CS6);
>-  EXPECT_EQ(rtc::DSCP_CS6,
>-            GetEndpoint(0)->cd1_.ch_->DefaultDscpValue());
>-  EXPECT_EQ(rtc::DSCP_CS6,
>-            GetEndpoint(1)->cd1_.ch_->DefaultDscpValue());
>-  GetEndpoint(0)->cd1_.ch_->SetOption(
>-      rtc::Socket::OPT_DSCP, rtc::DSCP_AF41);
>-  GetEndpoint(1)->cd1_.ch_->SetOption(
>-      rtc::Socket::OPT_DSCP, rtc::DSCP_AF41);
>-  EXPECT_EQ(rtc::DSCP_AF41,
>-            GetEndpoint(0)->cd1_.ch_->DefaultDscpValue());
>-  EXPECT_EQ(rtc::DSCP_AF41,
>-            GetEndpoint(1)->cd1_.ch_->DefaultDscpValue());
>+  EXPECT_EQ(rtc::DSCP_NO_CHANGE, GetEndpoint(0)->cd1_.ch_->DefaultDscpValue());
>+  EXPECT_EQ(rtc::DSCP_NO_CHANGE, GetEndpoint(1)->cd1_.ch_->DefaultDscpValue());
>+  GetEndpoint(0)->cd1_.ch_->SetOption(rtc::Socket::OPT_DSCP, rtc::DSCP_CS6);
>+  GetEndpoint(1)->cd1_.ch_->SetOption(rtc::Socket::OPT_DSCP, rtc::DSCP_CS6);
>+  EXPECT_EQ(rtc::DSCP_CS6, GetEndpoint(0)->cd1_.ch_->DefaultDscpValue());
>+  EXPECT_EQ(rtc::DSCP_CS6, GetEndpoint(1)->cd1_.ch_->DefaultDscpValue());
>+  GetEndpoint(0)->cd1_.ch_->SetOption(rtc::Socket::OPT_DSCP, rtc::DSCP_AF41);
>+  GetEndpoint(1)->cd1_.ch_->SetOption(rtc::Socket::OPT_DSCP, rtc::DSCP_AF41);
>+  EXPECT_EQ(rtc::DSCP_AF41, GetEndpoint(0)->cd1_.ch_->DefaultDscpValue());
>+  EXPECT_EQ(rtc::DSCP_AF41, GetEndpoint(1)->cd1_.ch_->DefaultDscpValue());
> }
> 
> // Verify IPv6 connection is preferred over IPv4.
>@@ -1873,10 +1901,10 @@ TEST_F(P2PTransportChannelTest, TestForceTurn) {
>   EXPECT_TRUE(ep1_ch1()->selected_connection() &&
>               ep2_ch1()->selected_connection());
> 
>-  EXPECT_EQ("relay", RemoteCandidate(ep1_ch1())->type());
>-  EXPECT_EQ("relay", LocalCandidate(ep1_ch1())->type());
>-  EXPECT_EQ("relay", RemoteCandidate(ep2_ch1())->type());
>-  EXPECT_EQ("relay", LocalCandidate(ep2_ch1())->type());
>+  EXPECT_EQ(RELAY_PORT_TYPE, RemoteCandidate(ep1_ch1())->type());
>+  EXPECT_EQ(RELAY_PORT_TYPE, LocalCandidate(ep1_ch1())->type());
>+  EXPECT_EQ(RELAY_PORT_TYPE, RemoteCandidate(ep2_ch1())->type());
>+  EXPECT_EQ(RELAY_PORT_TYPE, LocalCandidate(ep2_ch1())->type());
> 
>   TestSendRecv(&clock);
>   DestroyChannels();
>@@ -2171,6 +2199,50 @@ TEST_F(P2PTransportChannelTest, SignalReadyToSendWithPresumedWritable) {
>   EXPECT_EQ(len, SendData(ep1_ch1(), data, len));
> }
> 
>+// Test that role conflict error responses are sent as expected when receiving a
>+// ping from an unknown address over a TURN connection. Regression test for
>+// crbug.com/webrtc/9034.
>+TEST_F(P2PTransportChannelTest,
>+       TurnToPrflxSelectedAfterResolvingIceControllingRoleConflict) {
>+  rtc::ScopedFakeClock clock;
>+  // Gather only relay candidates.
>+  ConfigureEndpoints(NAT_SYMMETRIC, NAT_SYMMETRIC,
>+                     kDefaultPortAllocatorFlags | PORTALLOCATOR_DISABLE_UDP |
>+                         PORTALLOCATOR_DISABLE_STUN | PORTALLOCATOR_DISABLE_TCP,
>+                     kDefaultPortAllocatorFlags | PORTALLOCATOR_DISABLE_UDP |
>+                         PORTALLOCATOR_DISABLE_STUN |
>+                         PORTALLOCATOR_DISABLE_TCP);
>+  // With conflicting ICE roles, endpoint 1 has the higher tie breaker and will
>+  // send a binding error response.
>+  SetIceRole(0, ICEROLE_CONTROLLING);
>+  SetIceTiebreaker(0, kHighTiebreaker);
>+  SetIceRole(1, ICEROLE_CONTROLLING);
>+  SetIceTiebreaker(1, kLowTiebreaker);
>+  // We want the remote TURN candidate to show up as prflx. To do this we need
>+  // to configure the server to accept packets from an address we haven't
>+  // explicitly installed permission for.
>+  test_turn_server()->set_enable_permission_checks(false);
>+  GetEndpoint(0)->cd1_.ch_.reset(CreateChannel(
>+      0, ICE_CANDIDATE_COMPONENT_DEFAULT, kIceParams[0], kIceParams[1]));
>+  GetEndpoint(1)->cd1_.ch_.reset(CreateChannel(
>+      1, ICE_CANDIDATE_COMPONENT_DEFAULT, kIceParams[1], kIceParams[0]));
>+  // Don't signal candidates from channel 2, so that channel 1 sees the TURN
>+  // candidate as peer reflexive.
>+  PauseCandidates(1);
>+  ep1_ch1()->MaybeStartGathering();
>+  ep2_ch1()->MaybeStartGathering();
>+
>+  EXPECT_TRUE_SIMULATED_WAIT(ep1_ch1()->receiving() && ep1_ch1()->writable(),
>+                             kMediumTimeout, clock);
>+
>+  ASSERT_NE(nullptr, ep1_ch1()->selected_connection());
>+
>+  EXPECT_EQ(RELAY_PORT_TYPE, LocalCandidate(ep1_ch1())->type());
>+  EXPECT_EQ(PRFLX_PORT_TYPE, RemoteCandidate(ep1_ch1())->type());
>+
>+  DestroyChannels();
>+}
>+
> // Test what happens when we have 2 users behind the same NAT. This can lead
> // to interesting behavior because the STUN server will only give out the
> // address of the outermost NAT.
>@@ -2179,15 +2251,16 @@ class P2PTransportChannelSameNatTest : public P2PTransportChannelTestBase {
>   void ConfigureEndpoints(Config nat_type, Config config1, Config config2) {
>     RTC_CHECK_GE(nat_type, NAT_FULL_CONE);
>     RTC_CHECK_LE(nat_type, NAT_SYMMETRIC);
>-    rtc::NATSocketServer::Translator* outer_nat =
>-        nat()->AddTranslator(kPublicAddrs[0], kNatAddrs[0],
>-            static_cast<rtc::NATType>(nat_type - NAT_FULL_CONE));
>+    rtc::NATSocketServer::Translator* outer_nat = nat()->AddTranslator(
>+        kPublicAddrs[0], kNatAddrs[0],
>+        static_cast<rtc::NATType>(nat_type - NAT_FULL_CONE));
>     ConfigureEndpoint(outer_nat, 0, config1);
>     ConfigureEndpoint(outer_nat, 1, config2);
>     set_remote_ice_parameter_source(FROM_SETICEPARAMETERS);
>   }
>   void ConfigureEndpoint(rtc::NATSocketServer::Translator* nat,
>-                         int endpoint, Config config) {
>+                         int endpoint,
>+                         Config config) {
>     RTC_CHECK(config <= NAT_SYMMETRIC);
>     if (config == OPEN) {
>       AddAddress(endpoint, kPrivateAddrs[endpoint]);
>@@ -2195,8 +2268,8 @@ class P2PTransportChannelSameNatTest : public P2PTransportChannelTestBase {
>     } else {
>       AddAddress(endpoint, kCascadedPrivateAddrs[endpoint]);
>       nat->AddTranslator(kPrivateAddrs[endpoint], kCascadedNatAddrs[endpoint],
>-          static_cast<rtc::NATType>(config - NAT_FULL_CONE))->AddClient(
>-              kCascadedPrivateAddrs[endpoint]);
>+                         static_cast<rtc::NATType>(config - NAT_FULL_CONE))
>+          ->AddClient(kCascadedPrivateAddrs[endpoint]);
>     }
>   }
> };
>@@ -2359,7 +2432,7 @@ TEST_F(P2PTransportChannelMultihomedTest, TestFailoverControllingSide) {
>                                  ep2_ch1()->selected_connection()->receiving(),
>                              kMediumTimeout, clock);
>   EXPECT_TRUE(
>-    LocalCandidate(ep1_ch1())->address().EqualIPs(kAlternateAddrs[0]));
>+      LocalCandidate(ep1_ch1())->address().EqualIPs(kAlternateAddrs[0]));
>   EXPECT_TRUE(RemoteCandidate(ep1_ch1())->address().EqualIPs(kPublicAddrs[1]));
>   EXPECT_TRUE(
>       RemoteCandidate(ep2_ch1())->address().EqualIPs(kAlternateAddrs[0]));
>@@ -2921,10 +2994,10 @@ TEST_F(P2PTransportChannelMultihomedTest,
>                                  ep2_ch1()->receiving() &&
>                                  ep2_ch1()->writable(),
>                              kMediumTimeout, clock);
>-  EXPECT_TRUE(
>-      ep1_ch1()->selected_connection() && ep2_ch1()->selected_connection() &&
>-      LocalCandidate(ep1_ch1())->address().EqualIPs(kPublicAddrs[0]) &&
>-      RemoteCandidate(ep1_ch1())->address().EqualIPs(kPublicAddrs[1]));
>+  EXPECT_TRUE(ep1_ch1()->selected_connection() &&
>+              ep2_ch1()->selected_connection() &&
>+              LocalCandidate(ep1_ch1())->address().EqualIPs(kPublicAddrs[0]) &&
>+              RemoteCandidate(ep1_ch1())->address().EqualIPs(kPublicAddrs[1]));
> 
>   // Add the new address first and then remove the other one.
>   RTC_LOG(LS_INFO) << "Draining...";
>@@ -2950,54 +3023,6 @@ TEST_F(P2PTransportChannelMultihomedTest,
>   DestroyChannels();
> }
> 
>-/*
>-TODO(honghaiz) Once continual gathering fully supports
>-GATHER_CONTINUALLY_AND_RECOVER, put this test back.
>-
>-// Tests that if the backup connections are lost and then the interface with the
>-// selected connection is gone, continual gathering will restore the
>-// connectivity.
>-TEST_F(P2PTransportChannelMultihomedTest,
>-       TestBackupConnectionLostThenInterfaceGone) {
>-  rtc::ScopedFakeClock clock;
>-  auto& wifi = kAlternateAddrs;
>-  auto& cellular = kPublicAddrs;
>-  AddAddress(0, wifi[0], "test_wifi0", rtc::ADAPTER_TYPE_WIFI);
>-  AddAddress(0, cellular[0], "test_cell0", rtc::ADAPTER_TYPE_CELLULAR);
>-  AddAddress(1, wifi[1], "test_wifi1", rtc::ADAPTER_TYPE_WIFI);
>-  AddAddress(1, cellular[1], "test_cell1", rtc::ADAPTER_TYPE_CELLULAR);
>-  // Use only local ports for simplicity.
>-  SetAllocatorFlags(0, kOnlyLocalPorts);
>-  SetAllocatorFlags(1, kOnlyLocalPorts);
>-
>-  // Set continual gathering policy.
>-  IceConfig config = CreateIceConfig(1000, GATHER_CONTINUALLY_AND_RECOVER);
>-  // Create channels and let them go writable, as usual.
>-  CreateChannels(config, config);
>-  EXPECT_TRUE_SIMULATED_WAIT(ep1_ch1()->receiving() && ep1_ch1()->writable() &&
>-                              ep2_ch1()->receiving() && ep2_ch1()->writable(),
>-                          kMediumTimeout, clock);
>-  EXPECT_TRUE(ep1_ch1()->selected_connection() &&
>-              ep2_ch1()->selected_connection() &&
>-              LocalCandidate(ep1_ch1())->address().EqualIPs(wifi[0]) &&
>-              RemoteCandidate(ep1_ch1())->address().EqualIPs(wifi[1]));
>-
>-  // First destroy all backup connection.
>-  DestroyAllButBestConnection(ep1_ch1());
>-
>-  SIMULATED_WAIT(false, 10, clock);
>-  // Then the interface of the best connection goes away.
>-  RemoveAddress(0, wifi[0]);
>-  EXPECT_TRUE_SIMULATED_WAIT(
>-      ep1_ch1()->selected_connection() && ep2_ch1()->selected_connection() &&
>-          LocalCandidate(ep1_ch1())->address().EqualIPs(cellular[0]) &&
>-          RemoteCandidate(ep1_ch1())->address().EqualIPs(wifi[1]),
>-      kMediumTimeout, clock);
>-
>-  DestroyChannels();
>-}
>-*/
>-
> // Tests that the backup connection will be restored after it is destroyed.
> TEST_F(P2PTransportChannelMultihomedTest, TestRestoreBackupConnection) {
>   rtc::ScopedFakeClock clock;
>@@ -3040,6 +3065,33 @@ TEST_F(P2PTransportChannelMultihomedTest, TestRestoreBackupConnection) {
>   DestroyChannels();
> }
> 
>+// Test that when explicit binding to network interfaces is disallowed, we may
>+// still establish a connection by using the any address fallback.
>+TEST_F(P2PTransportChannelMultihomedTest,
>+       BindingToAnyAddressRevealsViableRouteWhenExplicitBindingFails) {
>+  rtc::ScopedFakeClock clock;
>+  AddAddress(0, kPublicAddrs[0]);
>+  AddAddress(1, kPublicAddrs[1]);
>+  fw()->SetUnbindableIps({kPublicAddrs[0].ipaddr()});
>+  // When bound to the any address, the port allocator should discover the
>+  // alternative local address.
>+  vss()->SetAlternativeLocalAddress(kAnyAddr.ipaddr(),
>+                                    kAlternateAddrs[0].ipaddr());
>+  SetAllocatorFlags(0, kOnlyLocalPorts);
>+  SetAllocatorFlags(1, kOnlyLocalPorts);
>+  IceConfig default_config;
>+  CreateChannels(default_config, default_config);
>+  EXPECT_TRUE_SIMULATED_WAIT(ep1_ch1()->receiving() && ep1_ch1()->writable() &&
>+                                 ep2_ch1()->receiving() &&
>+                                 ep2_ch1()->writable(),
>+                             kMediumTimeout, clock);
>+  EXPECT_TRUE(
>+      ep1_ch1()->selected_connection() && ep2_ch1()->selected_connection() &&
>+      LocalCandidate(ep1_ch1())->address().EqualIPs(kAlternateAddrs[0]));
>+
>+  DestroyChannels();
>+}
>+
> // A collection of tests which tests a single P2PTransportChannel by sending
> // pings.
> class P2PTransportChannelPingTest : public testing::Test,
>@@ -3140,7 +3192,7 @@ class P2PTransportChannelPingTest : public testing::Test,
>     conn->SignalNominated(conn);
>   }
> 
>-  void OnNetworkRouteChanged(rtc::Optional<rtc::NetworkRoute> network_route) {
>+  void OnNetworkRouteChanged(absl::optional<rtc::NetworkRoute> network_route) {
>     last_network_route_ = network_route;
>     if (last_network_route_) {
>       last_sent_packet_id_ = last_network_route_->last_sent_packet_id;
>@@ -3154,13 +3206,13 @@ class P2PTransportChannelPingTest : public testing::Test,
>                                uint32_t nomination = 0) {
>     IceMessage msg;
>     msg.SetType(STUN_BINDING_REQUEST);
>-    msg.AddAttribute(rtc::MakeUnique<StunByteStringAttribute>(
>+    msg.AddAttribute(absl::make_unique<StunByteStringAttribute>(
>         STUN_ATTR_USERNAME,
>         conn->local_candidate().username() + ":" + remote_ufrag));
>     msg.AddAttribute(
>-        rtc::MakeUnique<StunUInt32Attribute>(STUN_ATTR_PRIORITY, priority));
>+        absl::make_unique<StunUInt32Attribute>(STUN_ATTR_PRIORITY, priority));
>     if (nomination != 0) {
>-      msg.AddAttribute(rtc::MakeUnique<StunUInt32Attribute>(
>+      msg.AddAttribute(absl::make_unique<StunUInt32Attribute>(
>           STUN_ATTR_NOMINATION, nomination));
>     }
>     msg.SetTransactionID(rtc::CreateRandomString(kStunTransactionIdLength));
>@@ -3207,7 +3259,7 @@ class P2PTransportChannelPingTest : public testing::Test,
>   int last_sent_packet_id_ = -1;
>   bool channel_ready_to_send_ = false;
>   IceTransportState channel_state_ = IceTransportState::STATE_INIT;
>-  rtc::Optional<rtc::NetworkRoute> last_network_route_;
>+  absl::optional<rtc::NetworkRoute> last_network_route_;
> };
> 
> TEST_F(P2PTransportChannelPingTest, TestTriggeredChecks) {
>@@ -3366,11 +3418,11 @@ TEST_F(P2PTransportChannelPingTest, PingingStartedAsSoonAsPossible) {
>   // candidate pair while we still don't have remote ICE parameters.
>   IceMessage request;
>   request.SetType(STUN_BINDING_REQUEST);
>-  request.AddAttribute(rtc::MakeUnique<StunByteStringAttribute>(
>+  request.AddAttribute(absl::make_unique<StunByteStringAttribute>(
>       STUN_ATTR_USERNAME, kIceUfrag[1]));
>   uint32_t prflx_priority = ICE_TYPE_PREFERENCE_PRFLX << 24;
>-  request.AddAttribute(
>-      rtc::MakeUnique<StunUInt32Attribute>(STUN_ATTR_PRIORITY, prflx_priority));
>+  request.AddAttribute(absl::make_unique<StunUInt32Attribute>(
>+      STUN_ATTR_PRIORITY, prflx_priority));
>   Port* port = GetPort(&ch);
>   ASSERT_NE(nullptr, port);
>   port->SignalUnknownAddress(port, rtc::SocketAddress("1.1.1.1", 1), PROTO_UDP,
>@@ -3535,11 +3587,11 @@ TEST_F(P2PTransportChannelPingTest, ConnectionResurrection) {
>   // Create a minimal STUN message with prflx priority.
>   IceMessage request;
>   request.SetType(STUN_BINDING_REQUEST);
>-  request.AddAttribute(rtc::MakeUnique<StunByteStringAttribute>(
>+  request.AddAttribute(absl::make_unique<StunByteStringAttribute>(
>       STUN_ATTR_USERNAME, kIceUfrag[1]));
>   uint32_t prflx_priority = ICE_TYPE_PREFERENCE_PRFLX << 24;
>-  request.AddAttribute(
>-      rtc::MakeUnique<StunUInt32Attribute>(STUN_ATTR_PRIORITY, prflx_priority));
>+  request.AddAttribute(absl::make_unique<StunUInt32Attribute>(
>+      STUN_ATTR_PRIORITY, prflx_priority));
>   EXPECT_NE(prflx_priority, remote_priority);
> 
>   Port* port = GetPort(&ch);
>@@ -3565,17 +3617,17 @@ TEST_F(P2PTransportChannelPingTest, TestReceivingStateChange) {
>   PrepareChannel(&ch);
>   // Default receiving timeout and checking receiving interval should not be too
>   // small.
>-  EXPECT_LE(1000, ch.receiving_timeout());
>+  EXPECT_LE(1000, ch.config().receiving_timeout_or_default());
>   EXPECT_LE(200, ch.check_receiving_interval());
>   ch.SetIceConfig(CreateIceConfig(500, GATHER_ONCE));
>-  EXPECT_EQ(500, ch.receiving_timeout());
>+  EXPECT_EQ(500, ch.config().receiving_timeout_or_default());
>   EXPECT_EQ(50, ch.check_receiving_interval());
>   ch.MaybeStartGathering();
>   ch.AddRemoteCandidate(CreateUdpCandidate(LOCAL_PORT_TYPE, "1.1.1.1", 1, 1));
>   Connection* conn1 = WaitForConnectionTo(&ch, "1.1.1.1", 1, &clock);
>   ASSERT_TRUE(conn1 != nullptr);
> 
>-  clock.AdvanceTime(rtc::TimeDelta::FromSeconds(1));
>+  clock.AdvanceTime(webrtc::TimeDelta::seconds(1));
>   conn1->ReceivedPing();
>   conn1->OnReadPacket("ABC", 3, rtc::CreatePacketTime(0));
>   EXPECT_TRUE_SIMULATED_WAIT(ch.receiving(), kShortTimeout, clock);
>@@ -3678,11 +3730,11 @@ TEST_F(P2PTransportChannelPingTest, TestSelectConnectionFromUnknownAddress) {
>   // A minimal STUN message with prflx priority.
>   IceMessage request;
>   request.SetType(STUN_BINDING_REQUEST);
>-  request.AddAttribute(rtc::MakeUnique<StunByteStringAttribute>(
>+  request.AddAttribute(absl::make_unique<StunByteStringAttribute>(
>       STUN_ATTR_USERNAME, kIceUfrag[1]));
>   uint32_t prflx_priority = ICE_TYPE_PREFERENCE_PRFLX << 24;
>-  request.AddAttribute(
>-      rtc::MakeUnique<StunUInt32Attribute>(STUN_ATTR_PRIORITY, prflx_priority));
>+  request.AddAttribute(absl::make_unique<StunUInt32Attribute>(
>+      STUN_ATTR_PRIORITY, prflx_priority));
>   TestUDPPort* port = static_cast<TestUDPPort*>(GetPort(&ch));
>   port->SignalUnknownAddress(port, rtc::SocketAddress("1.1.1.1", 1), PROTO_UDP,
>                              &request, kIceUfrag[1], false);
>@@ -3721,7 +3773,7 @@ TEST_F(P2PTransportChannelPingTest, TestSelectConnectionFromUnknownAddress) {
>   // However if the request contains use_candidate attribute, it will be
>   // selected as the selected connection.
>   request.AddAttribute(
>-      rtc::MakeUnique<StunByteStringAttribute>(STUN_ATTR_USE_CANDIDATE));
>+      absl::make_unique<StunByteStringAttribute>(STUN_ATTR_USE_CANDIDATE));
>   port->SignalUnknownAddress(port, rtc::SocketAddress("4.4.4.4", 4), PROTO_UDP,
>                              &request, kIceUfrag[1], false);
>   Connection* conn4 = WaitForConnectionTo(&ch, "4.4.4.4", 4);
>@@ -3776,13 +3828,13 @@ TEST_F(P2PTransportChannelPingTest, TestSelectConnectionBasedOnMediaReceived) {
>   // nominate the selected connection.
>   IceMessage request;
>   request.SetType(STUN_BINDING_REQUEST);
>-  request.AddAttribute(rtc::MakeUnique<StunByteStringAttribute>(
>+  request.AddAttribute(absl::make_unique<StunByteStringAttribute>(
>       STUN_ATTR_USERNAME, kIceUfrag[1]));
>   uint32_t prflx_priority = ICE_TYPE_PREFERENCE_PRFLX << 24;
>+  request.AddAttribute(absl::make_unique<StunUInt32Attribute>(
>+      STUN_ATTR_PRIORITY, prflx_priority));
>   request.AddAttribute(
>-      rtc::MakeUnique<StunUInt32Attribute>(STUN_ATTR_PRIORITY, prflx_priority));
>-  request.AddAttribute(
>-      rtc::MakeUnique<StunByteStringAttribute>(STUN_ATTR_USE_CANDIDATE));
>+      absl::make_unique<StunByteStringAttribute>(STUN_ATTR_USE_CANDIDATE));
>   Port* port = GetPort(&ch);
>   port->SignalUnknownAddress(port, rtc::SocketAddress("3.3.3.3", 3), PROTO_UDP,
>                              &request, kIceUfrag[1], false);
>@@ -3803,7 +3855,7 @@ TEST_F(P2PTransportChannelPingTest, TestSelectConnectionBasedOnMediaReceived) {
> TEST_F(P2PTransportChannelPingTest,
>        TestControlledAgentDataReceivingTakesHigherPrecedenceThanPriority) {
>   rtc::ScopedFakeClock clock;
>-  clock.AdvanceTime(rtc::TimeDelta::FromSeconds(1));
>+  clock.AdvanceTime(webrtc::TimeDelta::seconds(1));
>   FakePortAllocator pa(rtc::Thread::Current(), nullptr);
>   P2PTransportChannel ch("SwitchSelectedConnection", 1, &pa);
>   PrepareChannel(&ch);
>@@ -3851,7 +3903,7 @@ TEST_F(P2PTransportChannelPingTest,
> TEST_F(P2PTransportChannelPingTest,
>        TestControlledAgentNominationTakesHigherPrecedenceThanDataReceiving) {
>   rtc::ScopedFakeClock clock;
>-  clock.AdvanceTime(rtc::TimeDelta::FromSeconds(1));
>+  clock.AdvanceTime(webrtc::TimeDelta::seconds(1));
> 
>   FakePortAllocator pa(rtc::Thread::Current(), nullptr);
>   P2PTransportChannel ch("SwitchSelectedConnection", 1, &pa);
>@@ -3891,7 +3943,7 @@ TEST_F(P2PTransportChannelPingTest,
> TEST_F(P2PTransportChannelPingTest,
>        TestControlledAgentSelectsConnectionWithHigherNomination) {
>   rtc::ScopedFakeClock clock;
>-  clock.AdvanceTime(rtc::TimeDelta::FromSeconds(1));
>+  clock.AdvanceTime(webrtc::TimeDelta::seconds(1));
> 
>   FakePortAllocator pa(rtc::Thread::Current(), nullptr);
>   P2PTransportChannel ch("test", 1, &pa);
>@@ -3938,7 +3990,7 @@ TEST_F(P2PTransportChannelPingTest,
> TEST_F(P2PTransportChannelPingTest,
>        TestControlledAgentIgnoresSmallerNomination) {
>   rtc::ScopedFakeClock clock;
>-  clock.AdvanceTime(rtc::TimeDelta::FromSeconds(1));
>+  clock.AdvanceTime(webrtc::TimeDelta::seconds(1));
> 
>   FakePortAllocator pa(rtc::Thread::Current(), nullptr);
>   P2PTransportChannel ch("test", 1, &pa);
>@@ -4038,7 +4090,7 @@ TEST_F(P2PTransportChannelPingTest, TestAddRemoteCandidateWithAddressReuse) {
> // will be pruned. Otherwise, lower-priority connections are kept.
> TEST_F(P2PTransportChannelPingTest, TestDontPruneWhenWeak) {
>   rtc::ScopedFakeClock clock;
>-  clock.AdvanceTime(rtc::TimeDelta::FromSeconds(1));
>+  clock.AdvanceTime(webrtc::TimeDelta::seconds(1));
>   FakePortAllocator pa(rtc::Thread::Current(), nullptr);
>   P2PTransportChannel ch("test channel", 1, &pa);
>   PrepareChannel(&ch);
>@@ -4098,7 +4150,7 @@ TEST_F(P2PTransportChannelPingTest, TestDontPruneHighPriorityConnections) {
> // Test that GetState returns the state correctly.
> TEST_F(P2PTransportChannelPingTest, TestGetState) {
>   rtc::ScopedFakeClock clock;
>-  clock.AdvanceTime(rtc::TimeDelta::FromSeconds(1));
>+  clock.AdvanceTime(webrtc::TimeDelta::seconds(1));
>   FakePortAllocator pa(rtc::Thread::Current(), nullptr);
>   P2PTransportChannel ch("test channel", 1, &pa);
>   PrepareChannel(&ch);
>@@ -4126,7 +4178,7 @@ TEST_F(P2PTransportChannelPingTest, TestGetState) {
> // right away, and it can become active and be pruned again.
> TEST_F(P2PTransportChannelPingTest, TestConnectionPrunedAgain) {
>   rtc::ScopedFakeClock clock;
>-  clock.AdvanceTime(rtc::TimeDelta::FromSeconds(1));
>+  clock.AdvanceTime(webrtc::TimeDelta::seconds(1));
> 
>   FakePortAllocator pa(rtc::Thread::Current(), nullptr);
>   P2PTransportChannel ch("test channel", 1, &pa);
>@@ -4305,7 +4357,7 @@ TEST_F(P2PTransportChannelPingTest, TestPortDestroyedAfterTimeoutAndPruned) {
>   // Simulate 2 minutes going by. This should be enough time for the port to
>   // time out.
>   for (int second = 0; second < 120; ++second) {
>-    fake_clock.AdvanceTime(rtc::TimeDelta::FromSeconds(1));
>+    fake_clock.AdvanceTime(webrtc::TimeDelta::seconds(1));
>   }
>   EXPECT_EQ(nullptr, GetConnectionTo(&ch, "1.1.1.1", 1));
>   // Port will not be removed because it is not pruned yet.
>@@ -4377,7 +4429,7 @@ class P2PTransportChannelMostLikelyToWorkFirstTest
> // we have a selected connection.
> TEST_F(P2PTransportChannelMostLikelyToWorkFirstTest,
>        TestRelayRelayFirstWhenNothingPingedYet) {
>-  const int max_strong_interval = 100;
>+  const int max_strong_interval = 500;
>   P2PTransportChannel& ch = StartTransportChannel(true, max_strong_interval);
>   EXPECT_TRUE_WAIT(ch.ports().size() == 2, kDefaultTimeout);
>   EXPECT_EQ(ch.ports()[0]->Type(), LOCAL_PORT_TYPE);
>@@ -4434,7 +4486,7 @@ TEST_F(P2PTransportChannelMostLikelyToWorkFirstTest,
> // in the first round.
> TEST_F(P2PTransportChannelMostLikelyToWorkFirstTest,
>        TestRelayRelayFirstWhenEverythingPinged) {
>-  P2PTransportChannel& ch = StartTransportChannel(true, 100);
>+  P2PTransportChannel& ch = StartTransportChannel(true, 500);
>   EXPECT_TRUE_WAIT(ch.ports().size() == 2, kDefaultTimeout);
>   EXPECT_EQ(ch.ports()[0]->Type(), LOCAL_PORT_TYPE);
>   EXPECT_EQ(ch.ports()[1]->Type(), RELAY_PORT_TYPE);
>@@ -4465,7 +4517,7 @@ TEST_F(P2PTransportChannelMostLikelyToWorkFirstTest,
> // before we re-ping Relay/Relay connections again.
> TEST_F(P2PTransportChannelMostLikelyToWorkFirstTest,
>        TestNoStarvationOnNonRelayConnection) {
>-  P2PTransportChannel& ch = StartTransportChannel(true, 100);
>+  P2PTransportChannel& ch = StartTransportChannel(true, 500);
>   EXPECT_TRUE_WAIT(ch.ports().size() == 2, kDefaultTimeout);
>   EXPECT_EQ(ch.ports()[0]->Type(), LOCAL_PORT_TYPE);
>   EXPECT_EQ(ch.ports()[1]->Type(), RELAY_PORT_TYPE);
>@@ -4504,7 +4556,7 @@ TEST_F(P2PTransportChannelMostLikelyToWorkFirstTest, TestTcpTurn) {
>   config.ports.push_back(ProtocolAddress(kTurnTcpIntAddr, PROTO_TCP));
>   allocator()->AddTurnServer(config);
> 
>-  P2PTransportChannel& ch = StartTransportChannel(true, 100);
>+  P2PTransportChannel& ch = StartTransportChannel(true, 500);
>   EXPECT_TRUE_WAIT(ch.ports().size() == 3, kDefaultTimeout);
>   EXPECT_EQ(ch.ports()[0]->Type(), LOCAL_PORT_TYPE);
>   EXPECT_EQ(ch.ports()[1]->Type(), RELAY_PORT_TYPE);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/packetlossestimator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/packetlossestimator.cc
>index ae05ff5107c..be326f423d1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/packetlossestimator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/packetlossestimator.cc
>@@ -8,8 +8,6 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include <sstream>
>-
> #include "p2p/base/packetlossestimator.h"
> 
> #include "rtc_base/checks.h"
>@@ -106,23 +104,4 @@ std::size_t PacketLossEstimator::tracked_packet_count_for_testing() const {
>   return tracked_packets_.size();
> }
> 
>-std::string PacketLossEstimator::TrackedPacketsStringForTesting(
>-    std::size_t max) const {
>-  std::ostringstream oss;
>-
>-  size_t count = 0;
>-  for (const auto& p : tracked_packets_) {
>-    const auto& id = p.first;
>-    const auto& packet_info = p.second;
>-    oss << "{ " << id << ", " << packet_info.sent_time << "}, ";
>-    count += 1;
>-    if (count == max) {
>-      oss << "...";
>-      break;
>-    }
>-  }
>-
>-  return oss.str();
>-}
>-
> }  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/packetlossestimator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/packetlossestimator.h
>index 2256bdd2fee..9e334bbc8a1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/packetlossestimator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/packetlossestimator.h
>@@ -56,9 +56,6 @@ class PacketLossEstimator {
> 
>   std::size_t tracked_packet_count_for_testing() const;
> 
>-  // Output tracked packet state as a string.
>-  std::string TrackedPacketsStringForTesting(std::size_t max) const;
>-
>  private:
>   struct PacketInfo {
>     int64_t sent_time;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/packetsocketfactory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/packetsocketfactory.h
>index c4a60eae765..4667bb1fd70 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/packetsocketfactory.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/packetsocketfactory.h
>@@ -16,6 +16,7 @@
> 
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/proxyinfo.h"
>+#include "rtc_base/sslcertificate.h"
> 
> namespace rtc {
> 
>@@ -27,6 +28,9 @@ struct PacketSocketTcpOptions {
>   int opts = 0;
>   std::vector<std::string> tls_alpn_protocols;
>   std::vector<std::string> tls_elliptic_curves;
>+  // An optional custom SSL certificate verifier that an API user can provide to
>+  // inject their own certificate verification logic.
>+  SSLCertificateVerifier* tls_cert_verifier = nullptr;
> };
> 
> class AsyncPacketSocket;
>@@ -46,7 +50,7 @@ class PacketSocketFactory {
>     OPT_SSLTCP = OPT_TLS_FAKE,
>   };
> 
>-  PacketSocketFactory() { }
>+  PacketSocketFactory() {}
>   virtual ~PacketSocketFactory() = default;
> 
>   virtual AsyncPacketSocket* CreateUdpSocket(const SocketAddress& address,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/packettransportinternal.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/packettransportinternal.cc
>index 2d28ffbd1ec..ba5f1e90687 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/packettransportinternal.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/packettransportinternal.cc
>@@ -24,8 +24,8 @@ bool PacketTransportInternal::GetOption(rtc::Socket::Option opt, int* value) {
>   return false;
> }
> 
>-rtc::Optional<NetworkRoute> PacketTransportInternal::network_route() const {
>-  return rtc::Optional<NetworkRoute>();
>+absl::optional<NetworkRoute> PacketTransportInternal::network_route() const {
>+  return absl::optional<NetworkRoute>();
> }
> 
> }  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/packettransportinternal.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/packettransportinternal.h
>index a5b9772c278..ca7f0036114 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/packettransportinternal.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/packettransportinternal.h
>@@ -14,14 +14,14 @@
> #include <string>
> #include <vector>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> // This is included for PacketOptions.
> #include "api/ortc/packettransportinterface.h"
> #include "p2p/base/port.h"
> #include "rtc_base/asyncpacketsocket.h"
> #include "rtc_base/networkroute.h"
>-#include "rtc_base/sigslot.h"
> #include "rtc_base/socket.h"
>+#include "rtc_base/third_party/sigslot/sigslot.h"
> 
> namespace rtc {
> struct PacketOptions;
>@@ -66,7 +66,7 @@ class PacketTransportInternal : public virtual webrtc::PacketTransportInterface,
> 
>   // Returns the current network route with transport overhead.
>   // TODO(zhihuang): Make it pure virtual once the Chrome/remoting is updated.
>-  virtual rtc::Optional<NetworkRoute> network_route() const;
>+  virtual absl::optional<NetworkRoute> network_route() const;
> 
>   // Emitted when the writable state, represented by |writable()|, changes.
>   sigslot::signal1<PacketTransportInternal*> SignalWritableState;
>@@ -94,7 +94,7 @@ class PacketTransportInternal : public virtual webrtc::PacketTransportInterface,
>       SignalSentPacket;
> 
>   // Signalled when the current network route has changed.
>-  sigslot::signal1<rtc::Optional<rtc::NetworkRoute>> SignalNetworkRouteChanged;
>+  sigslot::signal1<absl::optional<rtc::NetworkRoute>> SignalNetworkRouteChanged;
> 
>  protected:
>   PacketTransportInternal();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/port.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/port.cc
>index 21b817ae83b..8ab33a31185 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/port.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/port.cc
>@@ -16,9 +16,8 @@
> #include <utility>
> #include <vector>
> 
>-#include "p2p/base/common.h"
>+#include "absl/memory/memory.h"
> #include "p2p/base/portallocator.h"
>-#include "rtc_base/base64.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/crc32.h"
> #include "rtc_base/helpers.h"
>@@ -26,9 +25,9 @@
> #include "rtc_base/messagedigest.h"
> #include "rtc_base/network.h"
> #include "rtc_base/numerics/safe_minmax.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/stringencode.h"
> #include "rtc_base/stringutils.h"
>+#include "rtc_base/third_party/base64/base64.h"
> 
> namespace {
> 
>@@ -62,9 +61,79 @@ inline bool TooLongWithoutResponse(
>   return now > (first.sent_time + maximum_time);
> }
> 
>+// Helper methods for converting string values of log description fields to
>+// enum.
>+webrtc::IceCandidateType GetCandidateTypeByString(const std::string& type) {
>+  if (type == cricket::LOCAL_PORT_TYPE) {
>+    return webrtc::IceCandidateType::kLocal;
>+  } else if (type == cricket::STUN_PORT_TYPE) {
>+    return webrtc::IceCandidateType::kStun;
>+  } else if (type == cricket::PRFLX_PORT_TYPE) {
>+    return webrtc::IceCandidateType::kPrflx;
>+  } else if (type == cricket::RELAY_PORT_TYPE) {
>+    return webrtc::IceCandidateType::kRelay;
>+  }
>+  return webrtc::IceCandidateType::kUnknown;
>+}
>+
>+webrtc::IceCandidatePairProtocol GetProtocolByString(
>+    const std::string& protocol) {
>+  if (protocol == cricket::UDP_PROTOCOL_NAME) {
>+    return webrtc::IceCandidatePairProtocol::kUdp;
>+  } else if (protocol == cricket::TCP_PROTOCOL_NAME) {
>+    return webrtc::IceCandidatePairProtocol::kTcp;
>+  } else if (protocol == cricket::SSLTCP_PROTOCOL_NAME) {
>+    return webrtc::IceCandidatePairProtocol::kSsltcp;
>+  } else if (protocol == cricket::TLS_PROTOCOL_NAME) {
>+    return webrtc::IceCandidatePairProtocol::kTls;
>+  }
>+  return webrtc::IceCandidatePairProtocol::kUnknown;
>+}
>+
>+webrtc::IceCandidatePairAddressFamily GetAddressFamilyByInt(
>+    int address_family) {
>+  if (address_family == AF_INET) {
>+    return webrtc::IceCandidatePairAddressFamily::kIpv4;
>+  } else if (address_family == AF_INET6) {
>+    return webrtc::IceCandidatePairAddressFamily::kIpv6;
>+  }
>+  return webrtc::IceCandidatePairAddressFamily::kUnknown;
>+}
>+
>+webrtc::IceCandidateNetworkType ConvertNetworkType(rtc::AdapterType type) {
>+  if (type == rtc::ADAPTER_TYPE_ETHERNET) {
>+    return webrtc::IceCandidateNetworkType::kEthernet;
>+  } else if (type == rtc::ADAPTER_TYPE_LOOPBACK) {
>+    return webrtc::IceCandidateNetworkType::kLoopback;
>+  } else if (type == rtc::ADAPTER_TYPE_WIFI) {
>+    return webrtc::IceCandidateNetworkType::kWifi;
>+  } else if (type == rtc::ADAPTER_TYPE_VPN) {
>+    return webrtc::IceCandidateNetworkType::kVpn;
>+  } else if (type == rtc::ADAPTER_TYPE_CELLULAR) {
>+    return webrtc::IceCandidateNetworkType::kCellular;
>+  }
>+  return webrtc::IceCandidateNetworkType::kUnknown;
>+}
>+
>+rtc::PacketInfoProtocolType ConvertProtocolTypeToPacketInfoProtocolType(
>+    cricket::ProtocolType type) {
>+  switch (type) {
>+    case cricket::ProtocolType::PROTO_UDP:
>+      return rtc::PacketInfoProtocolType::kUdp;
>+    case cricket::ProtocolType::PROTO_TCP:
>+      return rtc::PacketInfoProtocolType::kTcp;
>+    case cricket::ProtocolType::PROTO_SSLTCP:
>+      return rtc::PacketInfoProtocolType::kSsltcp;
>+    case cricket::ProtocolType::PROTO_TLS:
>+      return rtc::PacketInfoProtocolType::kTls;
>+    default:
>+      return rtc::PacketInfoProtocolType::kUnknown;
>+  }
>+}
>+
> // We will restrict RTT estimates (when used for determining state) to be
> // within a reasonable range.
>-const int MINIMUM_RTT = 100;   // 0.1 seconds
>+const int MINIMUM_RTT = 100;    // 0.1 seconds
> const int MAXIMUM_RTT = 60000;  // 60 seconds
> 
> // When we don't have any RTT data, we have to pick something reasonable.  We
>@@ -93,6 +162,9 @@ const int64_t kForgetPacketAfter = 30000;  // 30 seconds
> 
> namespace cricket {
> 
>+using webrtc::RTCErrorType;
>+using webrtc::RTCError;
>+
> // TODO(ronghuawu): Use "local", "srflx", "prflx" and "relay". But this requires
> // the signaling part be updated correspondingly as well.
> const char LOCAL_PORT_TYPE[] = "local";
>@@ -136,9 +208,19 @@ static std::string ComputeFoundation(const std::string& type,
>                                      const rtc::SocketAddress& base_address) {
>   std::ostringstream ost;
>   ost << type << base_address.ipaddr().ToString() << protocol << relay_protocol;
>-  return rtc::ToString<uint32_t>(rtc::ComputeCrc32(ost.str()));
>+  return rtc::ToString(rtc::ComputeCrc32(ost.str()));
>+}
>+
>+CandidateStats::CandidateStats() = default;
>+
>+CandidateStats::CandidateStats(const CandidateStats&) = default;
>+
>+CandidateStats::CandidateStats(Candidate candidate) {
>+  this->candidate = candidate;
> }
> 
>+CandidateStats::~CandidateStats() = default;
>+
> ConnectionInfo::ConnectionInfo()
>     : best_connection(false),
>       writable(false),
>@@ -192,15 +274,6 @@ Port::Port(rtc::Thread* thread,
>   Construct();
> }
> 
>-Port::Port(rtc::Thread* thread,
>-           const std::string& type,
>-           rtc::PacketSocketFactory* factory,
>-           rtc::Network* network,
>-           const rtc::IPAddress& ip,
>-           const std::string& username_fragment,
>-           const std::string& password)
>-    : Port(thread, type, factory, network, username_fragment, password) {}
>-
> Port::Port(rtc::Thread* thread,
>            const std::string& type,
>            rtc::PacketSocketFactory* factory,
>@@ -243,7 +316,8 @@ void Port::Construct() {
> 
>   thread_->PostDelayed(RTC_FROM_HERE, timeout_delay_, this,
>                        MSG_DESTROY_IF_DEAD);
>-  LOG_J(LS_INFO, this) << "Port created with network cost " << network_cost_;
>+  RTC_LOG(LS_INFO) << ToString() << ": Port created with network cost "
>+                   << network_cost_;
> }
> 
> Port::~Port() {
>@@ -368,9 +442,11 @@ void Port::AddOrReplaceConnection(Connection* conn) {
>   // If there is a different connection on the same remote address, replace
>   // it with the new one and destroy the old one.
>   if (ret.second == false && ret.first->second != conn) {
>-    LOG_J(LS_WARNING, this)
>-        << "A new connection was created on an existing remote address. "
>-        << "New remote candidate: " << conn->remote_candidate().ToString();
>+    RTC_LOG(LS_WARNING)
>+        << ToString()
>+        << ": A new connection was created on an existing remote address. "
>+           "New remote candidate: "
>+        << conn->remote_candidate().ToString();
>     ret.first->second->SignalDestroyed.disconnect(this);
>     ret.first->second->Destroy();
>     ret.first->second = conn;
>@@ -379,9 +455,10 @@ void Port::AddOrReplaceConnection(Connection* conn) {
>   SignalConnectionCreated(this, conn);
> }
> 
>-void Port::OnReadPacket(
>-    const char* data, size_t size, const rtc::SocketAddress& addr,
>-    ProtocolType proto) {
>+void Port::OnReadPacket(const char* data,
>+                        size_t size,
>+                        const rtc::SocketAddress& addr,
>+                        ProtocolType proto) {
>   // If the user has enabled port packets, just hand this over.
>   if (enable_port_packets_) {
>     SignalReadPacket(this, data, size, addr);
>@@ -393,31 +470,34 @@ void Port::OnReadPacket(
>   std::unique_ptr<IceMessage> msg;
>   std::string remote_username;
>   if (!GetStunMessage(data, size, addr, &msg, &remote_username)) {
>-    LOG_J(LS_ERROR, this) << "Received non-STUN packet from unknown address ("
>-                          << addr.ToSensitiveString() << ")";
>+    RTC_LOG(LS_ERROR) << ToString()
>+                      << ": Received non-STUN packet from unknown address: "
>+                      << addr.ToSensitiveString();
>   } else if (!msg) {
>     // STUN message handled already
>   } else if (msg->type() == STUN_BINDING_REQUEST) {
>-    RTC_LOG(LS_INFO) << "Received STUN ping "
>-                     << " id=" << rtc::hex_encode(msg->transaction_id())
>+    RTC_LOG(LS_INFO) << "Received STUN ping id="
>+                     << rtc::hex_encode(msg->transaction_id())
>                      << " from unknown address " << addr.ToSensitiveString();
>-
>+    // We need to signal an unknown address before we handle any role conflict
>+    // below. Otherwise there would be no candidate pair and TURN entry created
>+    // to send the error response in case of a role conflict.
>+    SignalUnknownAddress(this, addr, proto, msg.get(), remote_username, false);
>     // Check for role conflicts.
>     if (!MaybeIceRoleConflict(addr, msg.get(), remote_username)) {
>       RTC_LOG(LS_INFO) << "Received conflicting role from the peer.";
>       return;
>     }
>-
>-    SignalUnknownAddress(this, addr, proto, msg.get(), remote_username, false);
>   } else {
>     // NOTE(tschmelcher): STUN_BINDING_RESPONSE is benign. It occurs if we
>     // pruned a connection for this port while it had STUN requests in flight,
>     // because we then get back responses for them, which this code correctly
>     // does not handle.
>     if (msg->type() != STUN_BINDING_RESPONSE) {
>-      LOG_J(LS_ERROR, this) << "Received unexpected STUN message type ("
>-                            << msg->type() << ") from unknown address ("
>-                            << addr.ToSensitiveString() << ")";
>+      RTC_LOG(LS_ERROR) << ToString()
>+                        << ": Received unexpected STUN message type: "
>+                        << msg->type() << " from unknown address: "
>+                        << addr.ToSensitiveString();
>     }
>   }
> }
>@@ -465,8 +545,9 @@ bool Port::GetStunMessage(const char* data,
>     // If not present, fail with a 400 Bad Request.
>     if (!stun_msg->GetByteString(STUN_ATTR_USERNAME) ||
>         !stun_msg->GetByteString(STUN_ATTR_MESSAGE_INTEGRITY)) {
>-      LOG_J(LS_ERROR, this) << "Received STUN request without username/M-I "
>-                            << "from " << addr.ToSensitiveString();
>+      RTC_LOG(LS_ERROR) << ToString()
>+                        << ": Received STUN request without username/M-I from: "
>+                        << addr.ToSensitiveString();
>       SendBindingErrorResponse(stun_msg.get(), addr, STUN_ERROR_BAD_REQUEST,
>                                STUN_ERROR_REASON_BAD_REQUEST);
>       return true;
>@@ -477,9 +558,9 @@ bool Port::GetStunMessage(const char* data,
>     std::string remote_ufrag;
>     if (!ParseStunUsername(stun_msg.get(), &local_ufrag, &remote_ufrag) ||
>         local_ufrag != username_fragment()) {
>-      LOG_J(LS_ERROR, this) << "Received STUN request with bad local username "
>-                            << local_ufrag << " from "
>-                            << addr.ToSensitiveString();
>+      RTC_LOG(LS_ERROR) << ToString()
>+                        << ": Received STUN request with bad local username "
>+                        << local_ufrag << " from " << addr.ToSensitiveString();
>       SendBindingErrorResponse(stun_msg.get(), addr, STUN_ERROR_UNAUTHORIZED,
>                                STUN_ERROR_REASON_UNAUTHORIZED);
>       return true;
>@@ -487,9 +568,10 @@ bool Port::GetStunMessage(const char* data,
> 
>     // If ICE, and the MESSAGE-INTEGRITY is bad, fail with a 401 Unauthorized
>     if (!stun_msg->ValidateMessageIntegrity(data, size, password_)) {
>-      LOG_J(LS_ERROR, this) << "Received STUN request with bad M-I "
>-                            << "from " << addr.ToSensitiveString()
>-                            << ", password_=" << password_;
>+      RTC_LOG(LS_ERROR) << ToString()
>+                        << ": Received STUN request with bad M-I from "
>+                        << addr.ToSensitiveString()
>+                        << ", password_=" << password_;
>       SendBindingErrorResponse(stun_msg.get(), addr, STUN_ERROR_UNAUTHORIZED,
>                                STUN_ERROR_REASON_UNAUTHORIZED);
>       return true;
>@@ -499,30 +581,35 @@ bool Port::GetStunMessage(const char* data,
>              (stun_msg->type() == STUN_BINDING_ERROR_RESPONSE)) {
>     if (stun_msg->type() == STUN_BINDING_ERROR_RESPONSE) {
>       if (const StunErrorCodeAttribute* error_code = stun_msg->GetErrorCode()) {
>-        LOG_J(LS_ERROR, this) << "Received STUN binding error:"
>-                              << " class=" << error_code->eclass()
>-                              << " number=" << error_code->number()
>-                              << " reason='" << error_code->reason() << "'"
>-                              << " from " << addr.ToSensitiveString();
>+        RTC_LOG(LS_ERROR) << ToString()
>+                          << ": Received STUN binding error: class="
>+                          << error_code->eclass()
>+                          << " number=" << error_code->number() << " reason='"
>+                          << error_code->reason() << "' from "
>+                          << addr.ToSensitiveString();
>         // Return message to allow error-specific processing
>       } else {
>-        LOG_J(LS_ERROR, this) << "Received STUN binding error without a error "
>-                              << "code from " << addr.ToSensitiveString();
>+        RTC_LOG(LS_ERROR)
>+            << ToString()
>+            << ": Received STUN binding error without a error code from "
>+            << addr.ToSensitiveString();
>         return true;
>       }
>     }
>     // NOTE: Username should not be used in verifying response messages.
>     out_username->clear();
>   } else if (stun_msg->type() == STUN_BINDING_INDICATION) {
>-    LOG_J(LS_VERBOSE, this) << "Received STUN binding indication:"
>-                            << " from " << addr.ToSensitiveString();
>+    RTC_LOG(LS_VERBOSE) << ToString()
>+                        << ": Received STUN binding indication: from "
>+                        << addr.ToSensitiveString();
>     out_username->clear();
>     // No stun attributes will be verified, if it's stun indication message.
>     // Returning from end of the this method.
>   } else {
>-    LOG_J(LS_ERROR, this) << "Received STUN packet with invalid type ("
>-                          << stun_msg->type() << ") from "
>-                          << addr.ToSensitiveString();
>+    RTC_LOG(LS_ERROR) << ToString()
>+                      << ": Received STUN packet with invalid type ("
>+                      << stun_msg->type() << ") from "
>+                      << addr.ToSensitiveString();
>     return true;
>   }
> 
>@@ -555,7 +642,7 @@ bool Port::ParseStunUsername(const StunMessage* stun_msg,
>   local_ufrag->clear();
>   remote_ufrag->clear();
>   const StunByteStringAttribute* username_attr =
>-        stun_msg->GetByteString(STUN_ATTR_USERNAME);
>+      stun_msg->GetByteString(STUN_ATTR_USERNAME);
>   if (username_attr == NULL)
>     return false;
> 
>@@ -571,9 +658,9 @@ bool Port::ParseStunUsername(const StunMessage* stun_msg,
>   return true;
> }
> 
>-bool Port::MaybeIceRoleConflict(
>-    const rtc::SocketAddress& addr, IceMessage* stun_msg,
>-    const std::string& remote_ufrag) {
>+bool Port::MaybeIceRoleConflict(const rtc::SocketAddress& addr,
>+                                IceMessage* stun_msg,
>+                                const std::string& remote_ufrag) {
>   // Validate ICE_CONTROLLING or ICE_CONTROLLED attributes.
>   bool ret = true;
>   IceRole remote_ice_role = ICEROLE_UNKNOWN;
>@@ -608,8 +695,8 @@ bool Port::MaybeIceRoleConflict(
>           SignalRoleConflict(this);
>         } else {
>           // Send Role Conflict (487) error response.
>-          SendBindingErrorResponse(stun_msg, addr,
>-              STUN_ERROR_ROLE_CONFLICT, STUN_ERROR_REASON_ROLE_CONFLICT);
>+          SendBindingErrorResponse(stun_msg, addr, STUN_ERROR_ROLE_CONFLICT,
>+                                   STUN_ERROR_REASON_ROLE_CONFLICT);
>           ret = false;
>         }
>       }
>@@ -620,8 +707,8 @@ bool Port::MaybeIceRoleConflict(
>           SignalRoleConflict(this);
>         } else {
>           // Send Role Conflict (487) error response.
>-          SendBindingErrorResponse(stun_msg, addr,
>-              STUN_ERROR_ROLE_CONFLICT, STUN_ERROR_REASON_ROLE_CONFLICT);
>+          SendBindingErrorResponse(stun_msg, addr, STUN_ERROR_ROLE_CONFLICT,
>+                                   STUN_ERROR_REASON_ROLE_CONFLICT);
>           ret = false;
>         }
>       }
>@@ -675,17 +762,18 @@ void Port::SendBindingResponse(StunMessage* request,
>   if (retransmit_attr) {
>     // Inherit the incoming retransmit value in the response so the other side
>     // can see our view of lost pings.
>-    response.AddAttribute(rtc::MakeUnique<StunUInt32Attribute>(
>+    response.AddAttribute(absl::make_unique<StunUInt32Attribute>(
>         STUN_ATTR_RETRANSMIT_COUNT, retransmit_attr->value()));
> 
>     if (retransmit_attr->value() > CONNECTION_WRITE_CONNECT_FAILURES) {
>-      LOG_J(LS_INFO, this)
>-          << "Received a remote ping with high retransmit count: "
>+      RTC_LOG(LS_INFO)
>+          << ToString()
>+          << ": Received a remote ping with high retransmit count: "
>           << retransmit_attr->value();
>     }
>   }
> 
>-  response.AddAttribute(rtc::MakeUnique<StunXorAddressAttribute>(
>+  response.AddAttribute(absl::make_unique<StunXorAddressAttribute>(
>       STUN_ATTR_XOR_MAPPED_ADDRESS, addr));
>   response.AddMessageIntegrity(password_);
>   response.AddFingerprint();
>@@ -694,31 +782,34 @@ void Port::SendBindingResponse(StunMessage* request,
>   rtc::ByteBufferWriter buf;
>   response.Write(&buf);
>   rtc::PacketOptions options(DefaultDscpValue());
>+  options.info_signaled_after_sent.packet_type =
>+      rtc::PacketType::kIceConnectivityCheckResponse;
>   auto err = SendTo(buf.Data(), buf.Length(), addr, options, false);
>   if (err < 0) {
>-    LOG_J(LS_ERROR, this)
>-        << "Failed to send STUN ping response"
>-        << ", to=" << addr.ToSensitiveString()
>-        << ", err=" << err
>-        << ", id=" << rtc::hex_encode(response.transaction_id());
>+    RTC_LOG(LS_ERROR) << ToString()
>+                      << ": Failed to send STUN ping response, to="
>+                      << addr.ToSensitiveString() << ", err=" << err
>+                      << ", id=" << rtc::hex_encode(response.transaction_id());
>   } else {
>     // Log at LS_INFO if we send a stun ping response on an unwritable
>     // connection.
>     Connection* conn = GetConnection(addr);
>-    rtc::LoggingSeverity sev = (conn && !conn->writable()) ?
>-        rtc::LS_INFO : rtc::LS_VERBOSE;
>-    LOG_JV(sev, this)
>-        << "Sent STUN ping response"
>-        << ", to=" << addr.ToSensitiveString()
>-        << ", id=" << rtc::hex_encode(response.transaction_id());
>+    rtc::LoggingSeverity sev =
>+        (conn && !conn->writable()) ? rtc::LS_INFO : rtc::LS_VERBOSE;
>+    RTC_LOG_V(sev) << ToString() << ": Sent STUN ping response, to="
>+                   << addr.ToSensitiveString()
>+                   << ", id=" << rtc::hex_encode(response.transaction_id());
> 
>     conn->stats_.sent_ping_responses++;
>+    conn->LogCandidatePairEvent(
>+        webrtc::IceCandidatePairEventType::kCheckResponseSent);
>   }
> }
> 
> void Port::SendBindingErrorResponse(StunMessage* request,
>                                     const rtc::SocketAddress& addr,
>-                                    int error_code, const std::string& reason) {
>+                                    int error_code,
>+                                    const std::string& reason) {
>   RTC_DCHECK(request->type() == STUN_BINDING_REQUEST);
> 
>   // Fill in the response message.
>@@ -744,9 +835,12 @@ void Port::SendBindingErrorResponse(StunMessage* request,
>   rtc::ByteBufferWriter buf;
>   response.Write(&buf);
>   rtc::PacketOptions options(DefaultDscpValue());
>+  options.info_signaled_after_sent.packet_type =
>+      rtc::PacketType::kIceConnectivityCheckResponse;
>   SendTo(buf.Data(), buf.Length(), addr, options, false);
>-  LOG_J(LS_INFO, this) << "Sending STUN binding error: reason=" << reason
>-                       << " to " << addr.ToSensitiveString();
>+  RTC_LOG(LS_INFO) << ToString()
>+                   << ": Sending STUN binding error: reason=" << reason
>+                   << " to " << addr.ToSensitiveString();
> }
> 
> void Port::KeepAliveUntilPruned() {
>@@ -761,7 +855,7 @@ void Port::Prune() {
>   thread_->Post(RTC_FROM_HERE, this, MSG_DESTROY_IF_DEAD);
> }
> 
>-void Port::OnMessage(rtc::Message *pmsg) {
>+void Port::OnMessage(rtc::Message* pmsg) {
>   RTC_DCHECK(pmsg->message_id == MSG_DESTROY_IF_DEAD);
>   bool dead =
>       (state_ == State::INIT || state_ == State::PRUNED) &&
>@@ -835,7 +929,7 @@ void Port::OnConnectionDestroyed(Connection* conn) {
> 
> void Port::Destroy() {
>   RTC_DCHECK(connections_.empty());
>-  LOG_J(LS_INFO, this) << "Port deleted";
>+  RTC_LOG(LS_INFO) << ToString() << ": Port deleted";
>   SignalDestroyed(this);
>   delete this;
> }
>@@ -844,51 +938,54 @@ const std::string Port::username_fragment() const {
>   return ice_username_fragment_;
> }
> 
>+void Port::CopyPortInformationToPacketInfo(rtc::PacketInfo* info) const {
>+  info->protocol = ConvertProtocolTypeToPacketInfoProtocolType(GetProtocol());
>+  info->network_id = Network()->id();
>+}
>+
> // A ConnectionRequest is a simple STUN ping used to determine writability.
> class ConnectionRequest : public StunRequest {
>  public:
>   explicit ConnectionRequest(Connection* connection)
>-      : StunRequest(new IceMessage()),
>-        connection_(connection) {
>-  }
>+      : StunRequest(new IceMessage()), connection_(connection) {}
> 
>   void Prepare(StunMessage* request) override {
>     request->SetType(STUN_BINDING_REQUEST);
>     std::string username;
>     connection_->port()->CreateStunUsername(
>         connection_->remote_candidate().username(), &username);
>-    request->AddAttribute(
>-        rtc::MakeUnique<StunByteStringAttribute>(STUN_ATTR_USERNAME, username));
>+    request->AddAttribute(absl::make_unique<StunByteStringAttribute>(
>+        STUN_ATTR_USERNAME, username));
> 
>     // connection_ already holds this ping, so subtract one from count.
>     if (connection_->port()->send_retransmit_count_attribute()) {
>-      request->AddAttribute(rtc::MakeUnique<StunUInt32Attribute>(
>+      request->AddAttribute(absl::make_unique<StunUInt32Attribute>(
>           STUN_ATTR_RETRANSMIT_COUNT,
>           static_cast<uint32_t>(connection_->pings_since_last_response_.size() -
>                                 1)));
>     }
>     uint32_t network_info = connection_->port()->Network()->id();
>     network_info = (network_info << 16) | connection_->port()->network_cost();
>-    request->AddAttribute(rtc::MakeUnique<StunUInt32Attribute>(
>+    request->AddAttribute(absl::make_unique<StunUInt32Attribute>(
>         STUN_ATTR_NETWORK_INFO, network_info));
> 
>     // Adding ICE_CONTROLLED or ICE_CONTROLLING attribute based on the role.
>     if (connection_->port()->GetIceRole() == ICEROLE_CONTROLLING) {
>-      request->AddAttribute(rtc::MakeUnique<StunUInt64Attribute>(
>+      request->AddAttribute(absl::make_unique<StunUInt64Attribute>(
>           STUN_ATTR_ICE_CONTROLLING, connection_->port()->IceTiebreaker()));
>       // We should have either USE_CANDIDATE attribute or ICE_NOMINATION
>       // attribute but not both. That was enforced in p2ptransportchannel.
>       if (connection_->use_candidate_attr()) {
>-        request->AddAttribute(
>-            rtc::MakeUnique<StunByteStringAttribute>(STUN_ATTR_USE_CANDIDATE));
>+        request->AddAttribute(absl::make_unique<StunByteStringAttribute>(
>+            STUN_ATTR_USE_CANDIDATE));
>       }
>       if (connection_->nomination() &&
>           connection_->nomination() != connection_->acked_nomination()) {
>-        request->AddAttribute(rtc::MakeUnique<StunUInt32Attribute>(
>+        request->AddAttribute(absl::make_unique<StunUInt32Attribute>(
>             STUN_ATTR_NOMINATION, connection_->nomination()));
>       }
>     } else if (connection_->port()->GetIceRole() == ICEROLE_CONTROLLED) {
>-      request->AddAttribute(rtc::MakeUnique<StunUInt64Attribute>(
>+      request->AddAttribute(absl::make_unique<StunUInt64Attribute>(
>           STUN_ATTR_ICE_CONTROLLED, connection_->port()->IceTiebreaker()));
>     } else {
>       RTC_NOTREACHED();
>@@ -907,7 +1004,7 @@ class ConnectionRequest : public StunRequest {
>     uint32_t prflx_priority =
>         type_preference << 24 |
>         (connection_->local_candidate().priority() & 0x00FFFFFF);
>-    request->AddAttribute(rtc::MakeUnique<StunUInt32Attribute>(
>+    request->AddAttribute(absl::make_unique<StunUInt32Attribute>(
>         STUN_ATTR_PRIORITY, prflx_priority));
> 
>     // Adding Message Integrity attribute.
>@@ -924,9 +1021,7 @@ class ConnectionRequest : public StunRequest {
>     connection_->OnConnectionRequestErrorResponse(this, response);
>   }
> 
>-  void OnTimeout() override {
>-    connection_->OnConnectionRequestTimeout(this);
>-  }
>+  void OnTimeout() override { connection_->OnConnectionRequestTimeout(this); }
> 
>   void OnSent() override {
>     connection_->OnConnectionRequestSent(this);
>@@ -935,9 +1030,7 @@ class ConnectionRequest : public StunRequest {
>     timeout_ = true;
>   }
> 
>-  int resend_delay() override {
>-    return CONNECTION_RESPONSE_TIMEOUT;
>-  }
>+  int resend_delay() override { return CONNECTION_RESPONSE_TIMEOUT; }
> 
>  private:
>   Connection* connection_;
>@@ -950,7 +1043,8 @@ class ConnectionRequest : public StunRequest {
> Connection::Connection(Port* port,
>                        size_t index,
>                        const Candidate& remote_candidate)
>-    : port_(port),
>+    : id_(rtc::CreateRandomId()),
>+      port_(port),
>       local_candidate_index_(index),
>       remote_candidate_(remote_candidate),
>       recv_rate_tracker_(100, 10u),
>@@ -970,17 +1064,15 @@ Connection::Connection(Port* port,
>       packet_loss_estimator_(kConsiderPacketLostAfter, kForgetPacketAfter),
>       reported_(false),
>       state_(IceCandidatePairState::WAITING),
>-      receiving_timeout_(WEAK_CONNECTION_RECEIVE_TIMEOUT),
>       time_created_ms_(rtc::TimeMillis()) {
>   // All of our connections start in WAITING state.
>   // TODO(mallinath) - Start connections from STATE_FROZEN.
>   // Wire up to send stun packets
>   requests_.SignalSendPacket.connect(this, &Connection::OnSendStunPacket);
>-  LOG_J(LS_INFO, this) << "Connection created";
>+  RTC_LOG(LS_INFO) << ToString() << ": Connection created";
> }
> 
>-Connection::~Connection() {
>-}
>+Connection::~Connection() {}
> 
> const Candidate& Connection::local_candidate() const {
>   RTC_DCHECK(local_candidate_index_ < port_->Candidates().size());
>@@ -1020,19 +1112,32 @@ void Connection::set_write_state(WriteState value) {
>   WriteState old_value = write_state_;
>   write_state_ = value;
>   if (value != old_value) {
>-    LOG_J(LS_VERBOSE, this) << "set_write_state from: " << old_value << " to "
>-                            << value;
>+    RTC_LOG(LS_VERBOSE) << ToString() << ": set_write_state from: " << old_value
>+                        << " to " << value;
>     SignalStateChange(this);
>   }
> }
> 
> void Connection::UpdateReceiving(int64_t now) {
>-  bool receiving =
>-      last_received() > 0 && now <= last_received() + receiving_timeout_;
>+  bool receiving;
>+  if (last_ping_sent() < last_ping_response_received()) {
>+    // We consider any candidate pair that has its last connectivity check
>+    // acknowledged by a response as receiving, particularly for backup
>+    // candidate pairs that send checks at a much slower pace than the selected
>+    // one. Otherwise, a backup candidate pair constantly becomes not receiving
>+    // as a side effect of a long ping interval, since we do not have a separate
>+    // receiving timeout for backup candidate pairs. See
>+    // IceConfig.ice_backup_candidate_pair_ping_interval,
>+    // IceConfig.ice_connection_receiving_timeout and their default value.
>+    receiving = true;
>+  } else {
>+    receiving =
>+        last_received() > 0 && now <= last_received() + receiving_timeout();
>+  }
>   if (receiving_ == receiving) {
>     return;
>   }
>-  LOG_J(LS_VERBOSE, this) << "set_receiving to " << receiving;
>+  RTC_LOG(LS_VERBOSE) << ToString() << ": set_receiving to " << receiving;
>   receiving_ = receiving;
>   receiving_unchanged_since_ = now;
>   SignalStateChange(this);
>@@ -1042,7 +1147,7 @@ void Connection::set_state(IceCandidatePairState state) {
>   IceCandidatePairState old_state = state_;
>   state_ = state;
>   if (state != old_state) {
>-    LOG_J(LS_VERBOSE, this) << "set_state";
>+    RTC_LOG(LS_VERBOSE) << ToString() << ": set_state";
>   }
> }
> 
>@@ -1050,8 +1155,7 @@ void Connection::set_connected(bool value) {
>   bool old_value = connected_;
>   connected_ = value;
>   if (value != old_value) {
>-    LOG_J(LS_VERBOSE, this) << "set_connected from: " << old_value << " to "
>-                            << value;
>+    RTC_LOG(LS_VERBOSE) << ToString() << ": Change connected_ to " << value;
>     SignalStateChange(this);
>   }
> }
>@@ -1060,20 +1164,37 @@ void Connection::set_use_candidate_attr(bool enable) {
>   use_candidate_attr_ = enable;
> }
> 
>-void Connection::OnSendStunPacket(const void* data, size_t size,
>+int Connection::unwritable_timeout() const {
>+  return unwritable_timeout_.value_or(CONNECTION_WRITE_CONNECT_TIMEOUT);
>+}
>+
>+int Connection::unwritable_min_checks() const {
>+  return unwritable_min_checks_.value_or(CONNECTION_WRITE_CONNECT_FAILURES);
>+}
>+
>+int Connection::receiving_timeout() const {
>+  return receiving_timeout_.value_or(WEAK_CONNECTION_RECEIVE_TIMEOUT);
>+}
>+
>+void Connection::OnSendStunPacket(const void* data,
>+                                  size_t size,
>                                   StunRequest* req) {
>   rtc::PacketOptions options(port_->DefaultDscpValue());
>-  auto err = port_->SendTo(
>-      data, size, remote_candidate_.address(), options, false);
>+  options.info_signaled_after_sent.packet_type =
>+      rtc::PacketType::kIceConnectivityCheck;
>+  auto err =
>+      port_->SendTo(data, size, remote_candidate_.address(), options, false);
>   if (err < 0) {
>-    LOG_J(LS_WARNING, this) << "Failed to send STUN ping "
>-                            << " err=" << err
>-                            << " id=" << rtc::hex_encode(req->id());
>+    RTC_LOG(LS_WARNING) << ToString()
>+                        << ": Failed to send STUN ping "
>+                           " err="
>+                        << err << " id=" << rtc::hex_encode(req->id());
>   }
> }
> 
>-void Connection::OnReadPacket(
>-  const char* data, size_t size, const rtc::PacketTime& packet_time) {
>+void Connection::OnReadPacket(const char* data,
>+                              size_t size,
>+                              const rtc::PacketTime& packet_time) {
>   std::unique_ptr<IceMessage> msg;
>   std::string remote_ufrag;
>   const rtc::SocketAddress& addr(remote_candidate_.address());
>@@ -1089,7 +1210,7 @@ void Connection::OnReadPacket(
>     if (!pruned_ && (write_state_ == STATE_WRITE_TIMEOUT)) {
>       RTC_LOG(LS_WARNING)
>           << "Received a data packet on a timed-out Connection. "
>-          << "Resetting state to STATE_WRITE_INIT.";
>+             "Resetting state to STATE_WRITE_INIT.";
>       set_write_state(STATE_WRITE_INIT);
>     }
>   } else if (!msg) {
>@@ -1103,17 +1224,18 @@ void Connection::OnReadPacket(
>     rtc::LoggingSeverity sev = (!writable() ? rtc::LS_INFO : rtc::LS_VERBOSE);
>     switch (msg->type()) {
>       case STUN_BINDING_REQUEST:
>-        LOG_JV(sev, this) << "Received STUN ping"
>-                          << ", id=" << rtc::hex_encode(msg->transaction_id());
>+        RTC_LOG_V(sev) << ToString() << ": Received STUN ping, id="
>+                       << rtc::hex_encode(msg->transaction_id());
> 
>         if (remote_ufrag == remote_candidate_.username()) {
>           HandleBindingRequest(msg.get());
>         } else {
>           // The packet had the right local username, but the remote username
>           // was not the right one for the remote address.
>-          LOG_J(LS_ERROR, this)
>-            << "Received STUN request with bad remote username "
>-            << remote_ufrag;
>+          RTC_LOG(LS_ERROR)
>+              << ToString()
>+              << ": Received STUN request with bad remote username "
>+              << remote_ufrag;
>           port_->SendBindingErrorResponse(msg.get(), addr,
>                                           STUN_ERROR_UNAUTHORIZED,
>                                           STUN_ERROR_REASON_UNAUTHORIZED);
>@@ -1125,8 +1247,8 @@ void Connection::OnReadPacket(
>       // id's match.
>       case STUN_BINDING_RESPONSE:
>       case STUN_BINDING_ERROR_RESPONSE:
>-        if (msg->ValidateMessageIntegrity(
>-                data, size, remote_candidate().password())) {
>+        if (msg->ValidateMessageIntegrity(data, size,
>+                                          remote_candidate().password())) {
>           requests_.CheckResponse(msg.get());
>         }
>         // Otherwise silently discard the response message.
>@@ -1160,6 +1282,7 @@ void Connection::HandleBindingRequest(IceMessage* msg) {
>   }
> 
>   stats_.recv_ping_requests++;
>+  LogCandidatePairEvent(webrtc::IceCandidatePairEventType::kCheckReceived);
> 
>   // This is a validated stun request from remote peer.
>   port_->SendBindingResponse(msg, remote_addr);
>@@ -1214,7 +1337,7 @@ void Connection::OnReadyToSend() {
> 
> void Connection::Prune() {
>   if (!pruned_ || active()) {
>-    LOG_J(LS_INFO, this) << "Connection pruned";
>+    RTC_LOG(LS_INFO) << ToString() << ": Connection pruned";
>     pruned_ = true;
>     requests_.Clear();
>     set_write_state(STATE_WRITE_TIMEOUT);
>@@ -1227,8 +1350,9 @@ void Connection::Destroy() {
>   // with the networking thread on which this message is posted). Also affects
>   // tests, with a workaround in
>   // AutoSocketServerThread::~AutoSocketServerThread.
>-  LOG_J(LS_VERBOSE, this) << "Connection destroyed";
>+  RTC_LOG(LS_VERBOSE) << ToString() << ": Connection destroyed";
>   port_->thread()->Post(RTC_FROM_HERE, this, MSG_DELETE);
>+  LogCandidatePairConfig(webrtc::IceCandidatePairConfigType::kDestroyed);
> }
> 
> void Connection::FailAndDestroy() {
>@@ -1264,13 +1388,13 @@ void Connection::UpdateState(int64_t now) {
>   if (RTC_LOG_CHECK_LEVEL(LS_VERBOSE)) {
>     std::string pings;
>     PrintPingsSinceLastResponse(&pings, 5);
>-    LOG_J(LS_VERBOSE, this) << "UpdateState()"
>-                            << ", ms since last received response="
>-                            << now - last_ping_response_received_
>-                            << ", ms since last received data="
>-                            << now - last_data_received_
>-                            << ", rtt=" << rtt
>-                            << ", pings_since_last_response=" << pings;
>+    RTC_LOG(LS_VERBOSE) << ToString()
>+                        << ": UpdateState()"
>+                           ", ms since last received response="
>+                        << now - last_ping_response_received_
>+                        << ", ms since last received data="
>+                        << now - last_data_received_ << ", rtt=" << rtt
>+                        << ", pings_since_last_response=" << pings;
>   }
> 
>   // Check the writable state.  (The order of these checks is important.)
>@@ -1283,34 +1407,28 @@ void Connection::UpdateState(int64_t now) {
>   // allow for changes in network conditions.
> 
>   if ((write_state_ == STATE_WRITABLE) &&
>-      TooManyFailures(pings_since_last_response_,
>-                      CONNECTION_WRITE_CONNECT_FAILURES,
>-                      rtt,
>+      TooManyFailures(pings_since_last_response_, unwritable_min_checks(), rtt,
>                       now) &&
>-      TooLongWithoutResponse(pings_since_last_response_,
>-                             CONNECTION_WRITE_CONNECT_TIMEOUT,
>+      TooLongWithoutResponse(pings_since_last_response_, unwritable_timeout(),
>                              now)) {
>-    uint32_t max_pings = CONNECTION_WRITE_CONNECT_FAILURES;
>-    LOG_J(LS_INFO, this) << "Unwritable after " << max_pings
>-                         << " ping failures and "
>-                         << now - pings_since_last_response_[0].sent_time
>-                         << " ms without a response,"
>-                         << " ms since last received ping="
>-                         << now - last_ping_received_
>-                         << " ms since last received data="
>-                         << now - last_data_received_
>-                         << " rtt=" << rtt;
>+    uint32_t max_pings = unwritable_min_checks();
>+    RTC_LOG(LS_INFO) << ToString() << ": Unwritable after " << max_pings
>+                     << " ping failures and "
>+                     << now - pings_since_last_response_[0].sent_time
>+                     << " ms without a response,"
>+                        " ms since last received ping="
>+                     << now - last_ping_received_
>+                     << " ms since last received data="
>+                     << now - last_data_received_ << " rtt=" << rtt;
>     set_write_state(STATE_WRITE_UNRELIABLE);
>   }
>   if ((write_state_ == STATE_WRITE_UNRELIABLE ||
>        write_state_ == STATE_WRITE_INIT) &&
>       TooLongWithoutResponse(pings_since_last_response_,
>-                             CONNECTION_WRITE_TIMEOUT,
>-                             now)) {
>-    LOG_J(LS_INFO, this) << "Timed out after "
>-                         << now - pings_since_last_response_[0].sent_time
>-                         << " ms without a response"
>-                         << ", rtt=" << rtt;
>+                             CONNECTION_WRITE_TIMEOUT, now)) {
>+    RTC_LOG(LS_INFO) << ToString() << ": Timed out after "
>+                     << now - pings_since_last_response_[0].sent_time
>+                     << " ms without a response, rtt=" << rtt;
>     set_write_state(STATE_WRITE_TIMEOUT);
>   }
> 
>@@ -1323,7 +1441,7 @@ void Connection::UpdateState(int64_t now) {
> 
> void Connection::Ping(int64_t now) {
>   last_ping_sent_ = now;
>-  ConnectionRequest *req = new ConnectionRequest(this);
>+  ConnectionRequest* req = new ConnectionRequest(this);
>   // If not using renomination, we use "1" to mean "nominated" and "0" to mean
>   // "not nominated". If using renomination, values greater than 1 are used for
>   // re-nominated pairs.
>@@ -1333,9 +1451,9 @@ void Connection::Ping(int64_t now) {
>   }
>   pings_since_last_response_.push_back(SentPing(req->id(), now, nomination));
>   packet_loss_estimator_.ExpectResponse(req->id(), now);
>-  LOG_J(LS_VERBOSE, this) << "Sending STUN ping "
>-                          << ", id=" << rtc::hex_encode(req->id())
>-                          << ", nomination=" << nomination_;
>+  RTC_LOG(LS_VERBOSE) << ToString() << ": Sending STUN ping, id="
>+                      << rtc::hex_encode(req->id())
>+                      << ", nomination=" << nomination_;
>   requests_.Send(req);
>   state_ = IceCandidatePairState::IN_PROGRESS;
>   num_pings_sent_++;
>@@ -1370,7 +1488,7 @@ void Connection::ReceivedPingResponse(int rtt, const std::string& request_id) {
>   set_write_state(STATE_WRITABLE);
>   set_state(IceCandidatePairState::SUCCEEDED);
>   if (rtt_samples_ > 0) {
>-    rtt_ = (RTT_RATIO * rtt_ + rtt) / (RTT_RATIO + 1);
>+    rtt_ = rtc::GetNextMovingAverage(rtt_, rtt, RTT_RATIO);
>   } else {
>     rtt_ = rtt;
>   }
>@@ -1423,38 +1541,43 @@ uint32_t Connection::ComputeNetworkCost() const {
> 
> std::string Connection::ToString() const {
>   const char CONNECT_STATE_ABBREV[2] = {
>-    '-',  // not connected (false)
>-    'C',  // connected (true)
>+      '-',  // not connected (false)
>+      'C',  // connected (true)
>   };
>   const char RECEIVE_STATE_ABBREV[2] = {
>-    '-',  // not receiving (false)
>-    'R',  // receiving (true)
>+      '-',  // not receiving (false)
>+      'R',  // receiving (true)
>   };
>   const char WRITE_STATE_ABBREV[4] = {
>-    'W',  // STATE_WRITABLE
>-    'w',  // STATE_WRITE_UNRELIABLE
>-    '-',  // STATE_WRITE_INIT
>-    'x',  // STATE_WRITE_TIMEOUT
>+      'W',  // STATE_WRITABLE
>+      'w',  // STATE_WRITE_UNRELIABLE
>+      '-',  // STATE_WRITE_INIT
>+      'x',  // STATE_WRITE_TIMEOUT
>   };
>   const std::string ICESTATE[4] = {
>-    "W",  // STATE_WAITING
>-    "I",  // STATE_INPROGRESS
>-    "S",  // STATE_SUCCEEDED
>-    "F"   // STATE_FAILED
>+      "W",  // STATE_WAITING
>+      "I",  // STATE_INPROGRESS
>+      "S",  // STATE_SUCCEEDED
>+      "F"   // STATE_FAILED
>+  };
>+  const std::string SELECTED_STATE_ABBREV[2] = {
>+      "-",  // candidate pair not selected (false)
>+      "S",  // selected (true)
>   };
>   const Candidate& local = local_candidate();
>   const Candidate& remote = remote_candidate();
>   std::stringstream ss;
>   ss << "Conn[" << ToDebugId() << ":" << port_->content_name() << ":"
>-     << local.id() << ":" << local.component() << ":" << local.generation()
>-     << ":" << local.type() << ":" << local.protocol() << ":"
>-     << local.address().ToSensitiveString() << "->" << remote.id() << ":"
>-     << remote.component() << ":" << remote.priority() << ":" << remote.type()
>-     << ":" << remote.protocol() << ":" << remote.address().ToSensitiveString()
>-     << "|" << CONNECT_STATE_ABBREV[connected()]
>-     << RECEIVE_STATE_ABBREV[receiving()] << WRITE_STATE_ABBREV[write_state()]
>-     << ICESTATE[static_cast<int>(state())] << "|" << remote_nomination() << "|"
>-     << nomination() << "|" << priority() << "|";
>+     << port_->Network()->ToString() << ":" << local.id() << ":"
>+     << local.component() << ":" << local.generation() << ":" << local.type()
>+     << ":" << local.protocol() << ":" << local.address().ToSensitiveString()
>+     << "->" << remote.id() << ":" << remote.component() << ":"
>+     << remote.priority() << ":" << remote.type() << ":" << remote.protocol()
>+     << ":" << remote.address().ToSensitiveString() << "|"
>+     << CONNECT_STATE_ABBREV[connected()] << RECEIVE_STATE_ABBREV[receiving()]
>+     << WRITE_STATE_ABBREV[write_state()] << ICESTATE[static_cast<int>(state())]
>+     << "|" << SELECTED_STATE_ABBREV[selected()] << "|" << remote_nomination()
>+     << "|" << nomination() << "|" << priority() << "|";
>   if (rtt_ < DEFAULT_RTT) {
>     ss << rtt_ << "]";
>   } else {
>@@ -1467,6 +1590,45 @@ std::string Connection::ToSensitiveString() const {
>   return ToString();
> }
> 
>+const webrtc::IceCandidatePairDescription& Connection::ToLogDescription() {
>+  if (log_description_.has_value()) {
>+    return log_description_.value();
>+  }
>+  const Candidate& local = local_candidate();
>+  const Candidate& remote = remote_candidate();
>+  const rtc::Network* network = port()->Network();
>+  log_description_ = webrtc::IceCandidatePairDescription();
>+  log_description_->local_candidate_type =
>+      GetCandidateTypeByString(local.type());
>+  log_description_->local_relay_protocol =
>+      GetProtocolByString(local.relay_protocol());
>+  log_description_->local_network_type = ConvertNetworkType(network->type());
>+  log_description_->local_address_family =
>+      GetAddressFamilyByInt(local.address().family());
>+  log_description_->remote_candidate_type =
>+      GetCandidateTypeByString(remote.type());
>+  log_description_->remote_address_family =
>+      GetAddressFamilyByInt(remote.address().family());
>+  log_description_->candidate_pair_protocol =
>+      GetProtocolByString(local.protocol());
>+  return log_description_.value();
>+}
>+
>+void Connection::LogCandidatePairConfig(
>+    webrtc::IceCandidatePairConfigType type) {
>+  if (ice_event_log_ == nullptr) {
>+    return;
>+  }
>+  ice_event_log_->LogCandidatePairConfig(type, id(), ToLogDescription());
>+}
>+
>+void Connection::LogCandidatePairEvent(webrtc::IceCandidatePairEventType type) {
>+  if (ice_event_log_ == nullptr) {
>+    return;
>+  }
>+  ice_event_log_->LogCandidatePairEvent(type, id());
>+}
>+
> void Connection::OnConnectionRequestResponse(ConnectionRequest* request,
>                                              StunMessage* response) {
>   // Log at LS_INFO if we receive a ping response on an unwritable
>@@ -1478,11 +1640,11 @@ void Connection::OnConnectionRequestResponse(ConnectionRequest* request,
>   if (RTC_LOG_CHECK_LEVEL_V(sev)) {
>     std::string pings;
>     PrintPingsSinceLastResponse(&pings, 5);
>-    LOG_JV(sev, this) << "Received STUN ping response"
>-                      << ", id=" << rtc::hex_encode(request->id())
>-                      << ", code=0"  // Makes logging easier to parse.
>-                      << ", rtt=" << rtt
>-                      << ", pings_since_last_response=" << pings;
>+    RTC_LOG_V(sev) << ToString() << ": Received STUN ping response, id="
>+                   << rtc::hex_encode(request->id())
>+                   << ", code=0"  // Makes logging easier to parse.
>+                      ", rtt="
>+                   << rtt << ", pings_since_last_response=" << pings;
>   }
>   ReceivedPingResponse(rtt, request->id());
> 
>@@ -1490,6 +1652,8 @@ void Connection::OnConnectionRequestResponse(ConnectionRequest* request,
>   packet_loss_estimator_.ReceivedResponse(request->id(), time_received);
> 
>   stats_.recv_ping_responses++;
>+  LogCandidatePairEvent(
>+      webrtc::IceCandidatePairEventType::kCheckResponseReceived);
> 
>   MaybeUpdateLocalCandidate(request, response);
> }
>@@ -1497,10 +1661,10 @@ void Connection::OnConnectionRequestResponse(ConnectionRequest* request,
> void Connection::OnConnectionRequestErrorResponse(ConnectionRequest* request,
>                                                   StunMessage* response) {
>   int error_code = response->GetErrorCodeValue();
>-  LOG_J(LS_INFO, this) << "Received STUN error response"
>-                       << " id=" << rtc::hex_encode(request->id())
>-                       << " code=" << error_code
>-                       << " rtt=" << request->Elapsed();
>+  RTC_LOG(LS_WARNING) << ToString() << ": Received STUN error response id="
>+                      << rtc::hex_encode(request->id())
>+                      << " code=" << error_code
>+                      << " rtt=" << request->Elapsed();
> 
>   if (error_code == STUN_ERROR_UNKNOWN_ATTRIBUTE ||
>       error_code == STUN_ERROR_SERVER_ERROR ||
>@@ -1512,8 +1676,9 @@ void Connection::OnConnectionRequestErrorResponse(ConnectionRequest* request,
>     HandleRoleConflictFromPeer();
>   } else {
>     // This is not a valid connection.
>-    LOG_J(LS_ERROR, this) << "Received STUN error response, code="
>-                          << error_code << "; killing connection";
>+    RTC_LOG(LS_ERROR) << ToString()
>+                      << ": Received STUN error response, code=" << error_code
>+                      << "; killing connection";
>     FailAndDestroy();
>   }
> }
>@@ -1521,19 +1686,20 @@ void Connection::OnConnectionRequestErrorResponse(ConnectionRequest* request,
> void Connection::OnConnectionRequestTimeout(ConnectionRequest* request) {
>   // Log at LS_INFO if we miss a ping on a writable connection.
>   rtc::LoggingSeverity sev = writable() ? rtc::LS_INFO : rtc::LS_VERBOSE;
>-  LOG_JV(sev, this) << "Timing-out STUN ping "
>-                    << rtc::hex_encode(request->id())
>-                    << " after " << request->Elapsed() << " ms";
>+  RTC_LOG_V(sev) << ToString() << ": Timing-out STUN ping "
>+                 << rtc::hex_encode(request->id()) << " after "
>+                 << request->Elapsed() << " ms";
> }
> 
> void Connection::OnConnectionRequestSent(ConnectionRequest* request) {
>   // Log at LS_INFO if we send a ping on an unwritable connection.
>   rtc::LoggingSeverity sev = !writable() ? rtc::LS_INFO : rtc::LS_VERBOSE;
>-  LOG_JV(sev, this) << "Sent STUN ping"
>-                    << ", id=" << rtc::hex_encode(request->id())
>-                    << ", use_candidate=" << use_candidate_attr()
>-                    << ", nomination=" << nomination();
>+  RTC_LOG_V(sev) << ToString()
>+                 << ": Sent STUN ping, id=" << rtc::hex_encode(request->id())
>+                 << ", use_candidate=" << use_candidate_attr()
>+                 << ", nomination=" << nomination();
>   stats_.sent_ping_requests_total++;
>+  LogCandidatePairEvent(webrtc::IceCandidatePairEventType::kCheckSent);
>   if (stats_.recv_ping_responses == 0) {
>     stats_.sent_ping_requests_before_first_response++;
>   }
>@@ -1552,7 +1718,7 @@ void Connection::MaybeSetRemoteIceParametersAndGeneration(
>   }
>   // TODO(deadbeef): A value of '0' for the generation is used for both
>   // generation 0 and "generation unknown". It should be changed to an
>-  // rtc::Optional to fix this.
>+  // absl::optional to fix this.
>   if (remote_candidate_.username() == ice_params.ufrag &&
>       remote_candidate_.password() == ice_params.pwd &&
>       remote_candidate_.generation() == 0) {
>@@ -1573,7 +1739,7 @@ void Connection::MaybeUpdatePeerReflexiveCandidate(
>   }
> }
> 
>-void Connection::OnMessage(rtc::Message *pmsg) {
>+void Connection::OnMessage(rtc::Message* pmsg) {
>   RTC_DCHECK(pmsg->message_id == MSG_DELETE);
>   RTC_LOG(LS_INFO) << "Connection deleted with number of pings sent: "
>                    << num_pings_sent_;
>@@ -1583,7 +1749,7 @@ void Connection::OnMessage(rtc::Message *pmsg) {
> 
> int64_t Connection::last_received() const {
>   return std::max(last_data_received_,
>-             std::max(last_ping_received_, last_ping_response_received_));
>+                  std::max(last_ping_received_, last_ping_response_received_));
> }
> 
> ConnectionInfo Connection::stats() {
>@@ -1619,15 +1785,16 @@ void Connection::MaybeUpdateLocalCandidate(ConnectionRequest* request,
>   if (!addr) {
>     RTC_LOG(LS_WARNING)
>         << "Connection::OnConnectionRequestResponse - "
>-        << "No MAPPED-ADDRESS or XOR-MAPPED-ADDRESS found in the "
>-        << "stun response message";
>+           "No MAPPED-ADDRESS or XOR-MAPPED-ADDRESS found in the "
>+           "stun response message";
>     return;
>   }
> 
>   for (size_t i = 0; i < port_->Candidates().size(); ++i) {
>     if (port_->Candidates()[i].address() == addr->GetAddress()) {
>       if (local_candidate_index_ != i) {
>-        LOG_J(LS_INFO, this) << "Updating local candidate type to srflx.";
>+        RTC_LOG(LS_INFO) << ToString()
>+                         << ": Updating local candidate type to srflx.";
>         local_candidate_index_ = i;
>         // SignalStateChange to force a re-sort in P2PTransportChannel as this
>         // Connection's local candidate has changed.
>@@ -1644,8 +1811,8 @@ void Connection::MaybeUpdateLocalCandidate(ConnectionRequest* request,
>       request->msg()->GetUInt32(STUN_ATTR_PRIORITY);
>   if (!priority_attr) {
>     RTC_LOG(LS_WARNING) << "Connection::OnConnectionRequestResponse - "
>-                        << "No STUN_ATTR_PRIORITY found in the "
>-                        << "stun response message";
>+                           "No STUN_ATTR_PRIORITY found in the "
>+                           "stun response message";
>     return;
>   }
>   const uint32_t priority = priority_attr->value();
>@@ -1671,7 +1838,7 @@ void Connection::MaybeUpdateLocalCandidate(ConnectionRequest* request,
>   new_local_candidate.set_network_cost(local_candidate().network_cost());
> 
>   // Change the local candidate of this Connection to the new prflx candidate.
>-  LOG_J(LS_INFO, this) << "Updating local candidate type to prflx.";
>+  RTC_LOG(LS_INFO) << ToString() << ": Updating local candidate type to prflx.";
>   local_candidate_index_ = port_->AddPrflxCandidate(new_local_candidate);
> 
>   // SignalStateChange to force a re-sort in P2PTransportChannel as this
>@@ -1697,11 +1864,12 @@ ProxyConnection::ProxyConnection(Port* port,
>                                  const Candidate& remote_candidate)
>     : Connection(port, index, remote_candidate) {}
> 
>-int ProxyConnection::Send(const void* data, size_t size,
>+int ProxyConnection::Send(const void* data,
>+                          size_t size,
>                           const rtc::PacketOptions& options) {
>   stats_.sent_total_packets++;
>-  int sent = port_->SendTo(data, size, remote_candidate_.address(),
>-                           options, true);
>+  int sent =
>+      port_->SendTo(data, size, remote_candidate_.address(), options, true);
>   if (sent <= 0) {
>     RTC_DCHECK(sent < 0);
>     error_ = port_->GetError();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/port.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/port.h
>index a68b236b602..d881fb8eafa 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/port.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/port.h
>@@ -17,9 +17,14 @@
> #include <string>
> #include <vector>
> 
>+#include "absl/types/optional.h"
> #include "api/candidate.h"
>-#include "api/optional.h"
>+#include "api/rtcerror.h"
>+#include "logging/rtc_event_log/events/rtc_event_ice_candidate_pair.h"
>+#include "logging/rtc_event_log/events/rtc_event_ice_candidate_pair_config.h"
>+#include "logging/rtc_event_log/icelogger.h"
> #include "p2p/base/candidatepairinterface.h"
>+#include "p2p/base/p2pconstants.h"
> #include "p2p/base/packetlossestimator.h"
> #include "p2p/base/packetsocketfactory.h"
> #include "p2p/base/portinterface.h"
>@@ -31,8 +36,8 @@
> #include "rtc_base/network.h"
> #include "rtc_base/proxyinfo.h"
> #include "rtc_base/ratetracker.h"
>-#include "rtc_base/sigslot.h"
> #include "rtc_base/socketaddress.h"
>+#include "rtc_base/third_party/sigslot/sigslot.h"
> #include "rtc_base/thread.h"
> 
> namespace cricket {
>@@ -51,35 +56,9 @@ extern const char TCPTYPE_ACTIVE_STR[];
> extern const char TCPTYPE_PASSIVE_STR[];
> extern const char TCPTYPE_SIMOPEN_STR[];
> 
>-// The minimum time we will wait before destroying a connection after creating
>-// it.
>-static const int MIN_CONNECTION_LIFETIME = 10 * 1000;  // 10 seconds.
>-
>-// A connection will be declared dead if it has not received anything for this
>-// long.
>-static const int DEAD_CONNECTION_RECEIVE_TIMEOUT = 30 * 1000;  // 30 seconds.
>-
>-// The timeout duration when a connection does not receive anything.
>-static const int WEAK_CONNECTION_RECEIVE_TIMEOUT = 2500;  // 2.5 seconds
>-
>-// The length of time we wait before timing out writability on a connection.
>-static const int CONNECTION_WRITE_TIMEOUT = 15 * 1000;  // 15 seconds
>-
>-// The length of time we wait before we become unwritable.
>-static const int CONNECTION_WRITE_CONNECT_TIMEOUT = 5 * 1000;  // 5 seconds
>-
>-// This is the length of time that we wait for a ping response to come back.
>-// There is no harm to keep this value high other than a small amount
>-// of increased memory.  But in some networks (2G),
>-// we observe up to 60s RTTs.
>-static const int CONNECTION_RESPONSE_TIMEOUT = 60 * 1000;  // 60 seconds
>-
>-// The number of pings that must fail to respond before we become unwritable.
>-static const uint32_t CONNECTION_WRITE_CONNECT_FAILURES = 5;
>-
> enum RelayType {
>-  RELAY_GTURN,   // Legacy google relay service.
>-  RELAY_TURN     // Standard (TURN) relay service.
>+  RELAY_GTURN,  // Legacy google relay service.
>+  RELAY_TURN    // Standard (TURN) relay service.
> };
> 
> enum IcePriorityValue {
>@@ -103,6 +82,36 @@ enum class IceCandidatePairState {
>   // frozen because we have not implemented ICE freezing logic.
> };
> 
>+// Stats that we can return about the port of a STUN candidate.
>+class StunStats {
>+ public:
>+  StunStats() = default;
>+  StunStats(const StunStats&) = default;
>+  ~StunStats() = default;
>+
>+  StunStats& operator=(const StunStats& other) = default;
>+
>+  int stun_binding_requests_sent = 0;
>+  int stun_binding_responses_received = 0;
>+  double stun_binding_rtt_ms_total = 0;
>+  double stun_binding_rtt_ms_squared_total = 0;
>+};
>+
>+// Stats that we can return about a candidate.
>+class CandidateStats {
>+ public:
>+  CandidateStats();
>+  explicit CandidateStats(Candidate candidate);
>+  CandidateStats(const CandidateStats&);
>+  ~CandidateStats();
>+
>+  Candidate candidate;
>+  // STUN port stats if this candidate is a STUN candidate.
>+  absl::optional<StunStats> stun_stats;
>+};
>+
>+typedef std::vector<CandidateStats> CandidateStatsList;
>+
> // Stats that we can return about the connections for a transport channel.
> // TODO(hta): Rename to ConnectionStats
> struct ConnectionInfo {
>@@ -143,10 +152,10 @@ struct ConnectionInfo {
>   // https://w3c.github.io/webrtc-stats/#dom-rtcicecandidatepairstats-totalroundtriptime
>   uint64_t total_round_trip_time_ms;
>   // https://w3c.github.io/webrtc-stats/#dom-rtcicecandidatepairstats-currentroundtriptime
>-  rtc::Optional<uint32_t> current_round_trip_time_ms;
>+  absl::optional<uint32_t> current_round_trip_time_ms;
> };
> 
>-// Information about all the connections of a channel.
>+// Information about all the candidate pairs of a channel.
> typedef std::vector<ConnectionInfo> ConnectionInfos;
> 
> const char* ProtoToString(ProtocolType proto);
>@@ -170,7 +179,8 @@ typedef std::set<rtc::SocketAddress> ServerAddresses;
> // Represents a local communication mechanism that can be used to create
> // connections to similar mechanisms of the other client.  Subclasses of this
> // one add support for specific mechanisms like local UDP ports.
>-class Port : public PortInterface, public rtc::MessageHandler,
>+class Port : public PortInterface,
>+             public rtc::MessageHandler,
>              public sigslot::has_slots<> {
>  public:
>   // INIT: The state when a port is just created.
>@@ -185,14 +195,6 @@ class Port : public PortInterface, public rtc::MessageHandler,
>        rtc::Network* network,
>        const std::string& username_fragment,
>        const std::string& password);
>-  // TODO(deadbeef): Delete this constructor once clients are moved off of it.
>-  Port(rtc::Thread* thread,
>-       const std::string& type,
>-       rtc::PacketSocketFactory* factory,
>-       rtc::Network* network,
>-       const rtc::IPAddress& ip,
>-       const std::string& username_fragment,
>-       const std::string& password);
>   Port(rtc::Thread* thread,
>        const std::string& type,
>        rtc::PacketSocketFactory* factory,
>@@ -203,6 +205,11 @@ class Port : public PortInterface, public rtc::MessageHandler,
>        const std::string& password);
>   ~Port() override;
> 
>+  // Note that the port type does NOT uniquely identify different subclasses of
>+  // Port. Use the 2-tuple of the port type AND the protocol (GetProtocol()) to
>+  // uniquely identify subclasses. Whenever a new subclass of Port introduces a
>+  // conflit in the value of the 2-tuple, make sure that the implementation that
>+  // relies on this 2-tuple for RTTI is properly changed.
>   const std::string& Type() const override;
>   rtc::Network* Network() const override;
> 
>@@ -314,8 +321,7 @@ class Port : public PortInterface, public rtc::MessageHandler,
>                                 int error_code,
>                                 const std::string& reason) override;
> 
>-  void set_proxy(const std::string& user_agent,
>-                 const rtc::ProxyInfo& proxy) {
>+  void set_proxy(const std::string& user_agent, const rtc::ProxyInfo& proxy) {
>     user_agent_ = user_agent;
>     proxy_ = proxy;
>   }
>@@ -365,6 +371,8 @@ class Port : public PortInterface, public rtc::MessageHandler,
> 
>   int16_t network_cost() const { return network_cost_; }
> 
>+  void GetStunStats(absl::optional<StunStats>* stats) override{};
>+
>  protected:
>   enum { MSG_DESTROY_IF_DEAD = 0, MSG_FIRST_AVAILABLE };
> 
>@@ -405,7 +413,8 @@ class Port : public PortInterface, public rtc::MessageHandler,
>   // Called when a packet is received from an unknown address that is not
>   // currently a connection.  If this is an authenticated STUN binding request,
>   // then we will signal the client.
>-  void OnReadPacket(const char* data, size_t size,
>+  void OnReadPacket(const char* data,
>+                    size_t size,
>                     const rtc::SocketAddress& addr,
>                     ProtocolType proto);
> 
>@@ -430,7 +439,9 @@ class Port : public PortInterface, public rtc::MessageHandler,
>   }
> 
>   // Extra work to be done in subclasses when a connection is destroyed.
>-  virtual void HandleConnectionDestroyed(Connection*) {}
>+  virtual void HandleConnectionDestroyed(Connection* conn) {}
>+
>+  void CopyPortInformationToPacketInfo(rtc::PacketInfo* info) const;
> 
>  private:
>   void Construct();
>@@ -497,6 +508,9 @@ class Connection : public CandidatePairInterface,
> 
>   ~Connection() override;
> 
>+  // A unique ID assigned when the connection is created.
>+  uint32_t id() { return id_; }
>+
>   // The local port where this connection sends and receives packets.
>   Port* port() { return port_; }
>   const Port* port() const { return port_; }
>@@ -511,10 +525,10 @@ class Connection : public CandidatePairInterface,
>   uint64_t priority() const;
> 
>   enum WriteState {
>-    STATE_WRITABLE          = 0,  // we have received ping responses recently
>-    STATE_WRITE_UNRELIABLE  = 1,  // we have had a few ping failures
>-    STATE_WRITE_INIT        = 2,  // we have yet to receive a ping response
>-    STATE_WRITE_TIMEOUT     = 3,  // we have had a large number of ping failures
>+    STATE_WRITABLE = 0,          // we have received ping responses recently
>+    STATE_WRITE_UNRELIABLE = 1,  // we have had a few ping failures
>+    STATE_WRITE_INIT = 2,        // we have yet to receive a ping response
>+    STATE_WRITE_TIMEOUT = 3,     // we have had a large number of ping failures
>   };
> 
>   WriteState write_state() const { return write_state_; }
>@@ -525,9 +539,7 @@ class Connection : public CandidatePairInterface,
>   // be false for TCP connections.
>   bool connected() const { return connected_; }
>   bool weak() const { return !(writable() && receiving() && connected()); }
>-  bool active() const {
>-    return write_state_ != STATE_WRITE_TIMEOUT;
>-  }
>+  bool active() const { return write_state_ != STATE_WRITE_TIMEOUT; }
> 
>   // A connection is dead if it can be safely deleted.
>   bool dead(int64_t now) const;
>@@ -535,6 +547,15 @@ class Connection : public CandidatePairInterface,
>   // Estimate of the round-trip time over this connection.
>   int rtt() const { return rtt_; }
> 
>+  int unwritable_timeout() const;
>+  void set_unwritable_timeout(const absl::optional<int>& value_ms) {
>+    unwritable_timeout_ = value_ms;
>+  }
>+  int unwritable_min_checks() const;
>+  void set_unwritable_min_checks(const absl::optional<int>& value) {
>+    unwritable_min_checks_ = value;
>+  }
>+
>   // Gets the |ConnectionInfo| stats, where |best_connection| has not been
>   // populated (default value false).
>   ConnectionInfo stats();
>@@ -548,7 +569,8 @@ class Connection : public CandidatePairInterface,
>   // The connection can send and receive packets asynchronously.  This matches
>   // the interface of AsyncPacketSocket, which may use UDP or TCP under the
>   // covers.
>-  virtual int Send(const void* data, size_t size,
>+  virtual int Send(const void* data,
>+                   size_t size,
>                    const rtc::PacketOptions& options) = 0;
> 
>   // Error if Send() returns < 0
>@@ -560,7 +582,8 @@ class Connection : public CandidatePairInterface,
>   sigslot::signal1<Connection*> SignalReadyToSend;
> 
>   // Called when a packet is received on this connection.
>-  void OnReadPacket(const char* data, size_t size,
>+  void OnReadPacket(const char* data,
>+                    size_t size,
>                     const rtc::PacketTime& packet_time);
> 
>   // Called when the socket is currently able to send.
>@@ -594,11 +617,10 @@ class Connection : public CandidatePairInterface,
>   // Public for unit tests.
>   uint32_t acked_nomination() const { return acked_nomination_; }
> 
>-  void set_remote_ice_mode(IceMode mode) {
>-    remote_ice_mode_ = mode;
>-  }
>+  void set_remote_ice_mode(IceMode mode) { remote_ice_mode_ = mode; }
> 
>-  void set_receiving_timeout(int receiving_timeout_ms) {
>+  int receiving_timeout() const;
>+  void set_receiving_timeout(absl::optional<int> receiving_timeout_ms) {
>     receiving_timeout_ = receiving_timeout_ms;
>   }
> 
>@@ -639,11 +661,21 @@ class Connection : public CandidatePairInterface,
>   std::string ToDebugId() const;
>   std::string ToString() const;
>   std::string ToSensitiveString() const;
>+  // Structured description of this candidate pair.
>+  const webrtc::IceCandidatePairDescription& ToLogDescription();
>+  void set_ice_event_log(webrtc::IceEventLog* ice_event_log) {
>+    ice_event_log_ = ice_event_log;
>+  }
>   // Prints pings_since_last_response_ into a string.
>   void PrintPingsSinceLastResponse(std::string* pings, size_t max);
> 
>   bool reported() const { return reported_; }
>-  void set_reported(bool reported) { reported_ = reported;}
>+  void set_reported(bool reported) { reported_ = reported; }
>+  // The following two methods are only used for logging in ToString above, and
>+  // this flag is set true by P2PTransportChannel for its selected candidate
>+  // pair.
>+  bool selected() const { return selected_; }
>+  void set_selected(bool selected) { selected_ = selected; }
> 
>   // This signal will be fired if this connection is nominated by the
>   // controlling side.
>@@ -714,6 +746,7 @@ class Connection : public CandidatePairInterface,
> 
>   void OnMessage(rtc::Message* pmsg) override;
> 
>+  uint32_t id_;
>   Port* port_;
>   size_t local_candidate_index_;
>   Candidate remote_candidate_;
>@@ -728,10 +761,14 @@ class Connection : public CandidatePairInterface,
>   void MaybeUpdateLocalCandidate(ConnectionRequest* request,
>                                  StunMessage* response);
> 
>+  void LogCandidatePairConfig(webrtc::IceCandidatePairConfigType type);
>+  void LogCandidatePairEvent(webrtc::IceCandidatePairEventType type);
>+
>   WriteState write_state_;
>   bool receiving_;
>   bool connected_;
>   bool pruned_;
>+  bool selected_ = false;
>   // By default |use_candidate_attr_| flag will be true,
>   // as we will be using aggressive nomination.
>   // But when peer is ice-lite, this flag "must" be initialized to false and
>@@ -757,7 +794,7 @@ class Connection : public CandidatePairInterface,
>   // https://w3c.github.io/webrtc-stats/#dom-rtcicecandidatepairstats-totalroundtriptime
>   uint64_t total_round_trip_time_ms_ = 0;
>   // https://w3c.github.io/webrtc-stats/#dom-rtcicecandidatepairstats-currentroundtriptime
>-  rtc::Optional<uint32_t> current_round_trip_time_ms_;
>+  absl::optional<uint32_t> current_round_trip_time_ms_;
>   int64_t last_ping_sent_;      // last time we sent a ping to the other side
>   int64_t last_ping_received_;  // last time we received a ping from the other
>                                 // side
>@@ -768,13 +805,19 @@ class Connection : public CandidatePairInterface,
> 
>   PacketLossEstimator packet_loss_estimator_;
> 
>+  absl::optional<int> unwritable_timeout_;
>+  absl::optional<int> unwritable_min_checks_;
>+
>   bool reported_;
>   IceCandidatePairState state_;
>   // Time duration to switch from receiving to not receiving.
>-  int receiving_timeout_;
>+  absl::optional<int> receiving_timeout_;
>   int64_t time_created_ms_;
>   int num_pings_sent_ = 0;
> 
>+  absl::optional<webrtc::IceCandidatePairDescription> log_description_;
>+  webrtc::IceEventLog* ice_event_log_ = nullptr;
>+
>   friend class Port;
>   friend class ConnectionRequest;
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/port_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/port_unittest.cc
>index 438e4e60082..5aaf315363b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/port_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/port_unittest.cc
>@@ -11,7 +11,9 @@
> #include <list>
> #include <memory>
> 
>+#include "absl/memory/memory.h"
> #include "p2p/base/basicpacketsocketfactory.h"
>+#include "p2p/base/p2pconstants.h"
> #include "p2p/base/relayport.h"
> #include "p2p/base/stunport.h"
> #include "p2p/base/tcpport.h"
>@@ -27,7 +29,6 @@
> #include "rtc_base/logging.h"
> #include "rtc_base/natserver.h"
> #include "rtc_base/natsocketfactory.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/socketaddress.h"
> #include "rtc_base/ssladapter.h"
> #include "rtc_base/stringutils.h"
>@@ -52,6 +53,7 @@ namespace {
> 
> constexpr int kDefaultTimeout = 3000;
> constexpr int kShortTimeout = 1000;
>+constexpr int kMaxExpectedSimulatedRtt = 200;
> const SocketAddress kLocalAddr1("192.168.1.2", 0);
> const SocketAddress kLocalAddr2("192.168.1.3", 0);
> const SocketAddress kNatAddr1("77.77.77.77", rtc::NAT_SERVER_UDP_PORT);
>@@ -182,9 +184,11 @@ class TestPort : public Port {
>     conn->set_use_candidate_attr(true);
>     return conn;
>   }
>-  virtual int SendTo(
>-      const void* data, size_t size, const rtc::SocketAddress& addr,
>-      const rtc::PacketOptions& options, bool payload) {
>+  virtual int SendTo(const void* data,
>+                     size_t size,
>+                     const rtc::SocketAddress& addr,
>+                     const rtc::PacketOptions& options,
>+                     bool payload) {
>     if (!payload) {
>       IceMessage* msg = new IceMessage;
>       Buffer* buf = new Buffer(static_cast<const char*>(data), size);
>@@ -199,15 +203,9 @@ class TestPort : public Port {
>     }
>     return static_cast<int>(size);
>   }
>-  virtual int SetOption(rtc::Socket::Option opt, int value) {
>-    return 0;
>-  }
>-  virtual int GetOption(rtc::Socket::Option opt, int* value) {
>-    return -1;
>-  }
>-  virtual int GetError() {
>-    return 0;
>-  }
>+  virtual int SetOption(rtc::Socket::Option opt, int value) { return 0; }
>+  virtual int GetOption(rtc::Socket::Option opt, int* value) { return -1; }
>+  virtual int GetError() { return 0; }
>   void Reset() {
>     last_stun_buf_.reset();
>     last_stun_msg_.reset();
>@@ -226,21 +224,22 @@ class TestPort : public Port {
>   int type_preference_ = 0;
> };
> 
>-static void SendPingAndReceiveResponse(
>-    Connection* lconn, TestPort* lport, Connection* rconn, TestPort* rport,
>-    rtc::ScopedFakeClock* clock, int64_t ms) {
>+static void SendPingAndReceiveResponse(Connection* lconn,
>+                                       TestPort* lport,
>+                                       Connection* rconn,
>+                                       TestPort* rport,
>+                                       rtc::ScopedFakeClock* clock,
>+                                       int64_t ms) {
>   lconn->Ping(rtc::TimeMillis());
>   ASSERT_TRUE_WAIT(lport->last_stun_msg(), kDefaultTimeout);
>   ASSERT_TRUE(lport->last_stun_buf());
>   rconn->OnReadPacket(lport->last_stun_buf()->data<char>(),
>-                      lport->last_stun_buf()->size(),
>-                      rtc::PacketTime());
>-  clock->AdvanceTime(rtc::TimeDelta::FromMilliseconds(ms));
>+                      lport->last_stun_buf()->size(), rtc::PacketTime());
>+  clock->AdvanceTime(webrtc::TimeDelta::ms(ms));
>   ASSERT_TRUE_WAIT(rport->last_stun_msg(), kDefaultTimeout);
>   ASSERT_TRUE(rport->last_stun_buf());
>   lconn->OnReadPacket(rport->last_stun_buf()->data<char>(),
>-                      rport->last_stun_buf()->size(),
>-                      rtc::PacketTime());
>+                      rport->last_stun_buf()->size(), rtc::PacketTime());
> }
> 
> class TestChannel : public sigslot::has_slots<> {
>@@ -270,8 +269,8 @@ class TestChannel : public sigslot::has_slots<> {
>         (ice_mode_ == ICEMODE_FULL) ? ICEMODE_LITE : ICEMODE_FULL;
>     conn_->set_remote_ice_mode(remote_ice_mode);
>     conn_->set_use_candidate_attr(remote_ice_mode == ICEMODE_FULL);
>-    conn_->SignalStateChange.connect(
>-        this, &TestChannel::OnConnectionStateChange);
>+    conn_->SignalStateChange.connect(this,
>+                                     &TestChannel::OnConnectionStateChange);
>     conn_->SignalDestroyed.connect(this, &TestChannel::OnDestroyed);
>     conn_->SignalReadyToSend.connect(this,
>                                      &TestChannel::OnConnectionReadyToSend);
>@@ -292,9 +291,7 @@ class TestChannel : public sigslot::has_slots<> {
>     port_->SendBindingResponse(remote_request_.get(), remote_address_);
>     remote_request_.reset();
>   }
>-  void Ping() {
>-    Ping(0);
>-  }
>+  void Ping() { Ping(0); }
>   void Ping(int64_t now) { conn_->Ping(now); }
>   void Stop() {
>     if (conn_) {
>@@ -302,21 +299,19 @@ class TestChannel : public sigslot::has_slots<> {
>     }
>   }
> 
>-  void OnPortComplete(Port* port) {
>-    complete_count_++;
>-  }
>-  void SetIceMode(IceMode ice_mode) {
>-    ice_mode_ = ice_mode;
>-  }
>+  void OnPortComplete(Port* port) { complete_count_++; }
>+  void SetIceMode(IceMode ice_mode) { ice_mode_ = ice_mode; }
> 
>   int SendData(const char* data, size_t len) {
>     rtc::PacketOptions options;
>     return conn_->Send(data, len, options);
>   }
> 
>-  void OnUnknownAddress(PortInterface* port, const SocketAddress& addr,
>+  void OnUnknownAddress(PortInterface* port,
>+                        const SocketAddress& addr,
>                         ProtocolType proto,
>-                        IceMessage* msg, const std::string& rf,
>+                        IceMessage* msg,
>+                        const std::string& rf,
>                         bool /*port_muxed*/) {
>     ASSERT_EQ(port_.get(), port);
>     if (!remote_address_.IsNil()) {
>@@ -358,9 +353,7 @@ class TestChannel : public sigslot::has_slots<> {
>   void set_connection_ready_to_send(bool ready) {
>     connection_ready_to_send_ = ready;
>   }
>-  bool connection_ready_to_send() const {
>-    return connection_ready_to_send_;
>-  }
>+  bool connection_ready_to_send() const { return connection_ready_to_send_; }
> 
>  private:
>   // ReadyToSend will only issue after a Connection recovers from ENOTCONN
>@@ -403,8 +396,7 @@ class PortTest : public testing::Test, public sigslot::has_slots<> {
>         username_(rtc::CreateRandomString(ICE_UFRAG_LENGTH)),
>         password_(rtc::CreateRandomString(ICE_PWD_LENGTH)),
>         role_conflict_(false),
>-        ports_destroyed_(0) {
>-  }
>+        ports_destroyed_(0) {}
> 
>  protected:
>   void TestLocalToLocal() {
>@@ -421,8 +413,8 @@ class PortTest : public testing::Test, public sigslot::has_slots<> {
>     Port* port2 = CreateStunPort(kLocalAddr2, &nat_socket_factory2_);
>     port2->SetIceRole(cricket::ICEROLE_CONTROLLED);
>     TestConnectivity("udp", port1, StunName(ntype), port2,
>-                     ntype == NAT_OPEN_CONE, true,
>-                     ntype != NAT_SYMMETRIC, true);
>+                     ntype == NAT_OPEN_CONE, true, ntype != NAT_SYMMETRIC,
>+                     true);
>   }
>   void TestLocalToRelay(RelayType rtype, ProtocolType proto) {
>     Port* port1 = CreateUdpPort(kLocalAddr1);
>@@ -438,8 +430,8 @@ class PortTest : public testing::Test, public sigslot::has_slots<> {
>     port1->SetIceRole(cricket::ICEROLE_CONTROLLING);
>     Port* port2 = CreateUdpPort(kLocalAddr2);
>     port2->SetIceRole(cricket::ICEROLE_CONTROLLED);
>-    TestConnectivity(StunName(ntype), port1, "udp", port2,
>-                     true, ntype != NAT_SYMMETRIC, true, true);
>+    TestConnectivity(StunName(ntype), port1, "udp", port2, true,
>+                     ntype != NAT_SYMMETRIC, true, true);
>   }
>   void TestStunToStun(NATType ntype1, NATType ntype2) {
>     nat_server1_.reset(CreateNatServer(kNatAddr1, ntype1));
>@@ -449,8 +441,8 @@ class PortTest : public testing::Test, public sigslot::has_slots<> {
>     Port* port2 = CreateStunPort(kLocalAddr2, &nat_socket_factory2_);
>     port2->SetIceRole(cricket::ICEROLE_CONTROLLED);
>     TestConnectivity(StunName(ntype1), port1, StunName(ntype2), port2,
>-                     ntype2 == NAT_OPEN_CONE,
>-                     ntype1 != NAT_SYMMETRIC, ntype2 != NAT_SYMMETRIC,
>+                     ntype2 == NAT_OPEN_CONE, ntype1 != NAT_SYMMETRIC,
>+                     ntype2 != NAT_SYMMETRIC,
>                      ntype1 + ntype2 < (NAT_PORT_RESTRICTED + NAT_SYMMETRIC));
>   }
>   void TestStunToRelay(NATType ntype, RelayType rtype, ProtocolType proto) {
>@@ -499,13 +491,14 @@ class PortTest : public testing::Test, public sigslot::has_slots<> {
>   UDPPort* CreateUdpPort(const SocketAddress& addr,
>                          PacketSocketFactory* socket_factory) {
>     return UDPPort::Create(&main_, socket_factory, MakeNetwork(addr), 0, 0,
>-                           username_, password_, std::string(), true);
>+                           username_, password_, std::string(), true,
>+                           absl::nullopt);
>   }
>   TCPPort* CreateTcpPort(const SocketAddress& addr) {
>     return CreateTcpPort(addr, &socket_factory_);
>   }
>   TCPPort* CreateTcpPort(const SocketAddress& addr,
>-                        PacketSocketFactory* socket_factory) {
>+                         PacketSocketFactory* socket_factory) {
>     return TCPPort::Create(&main_, socket_factory, MakeNetwork(addr), 0, 0,
>                            username_, password_, true);
>   }
>@@ -514,10 +507,13 @@ class PortTest : public testing::Test, public sigslot::has_slots<> {
>     ServerAddresses stun_servers;
>     stun_servers.insert(kStunAddr);
>     return StunPort::Create(&main_, factory, MakeNetwork(addr), 0, 0, username_,
>-                            password_, stun_servers, std::string());
>+                            password_, stun_servers, std::string(),
>+                            absl::nullopt);
>   }
>-  Port* CreateRelayPort(const SocketAddress& addr, RelayType rtype,
>-                        ProtocolType int_proto, ProtocolType ext_proto) {
>+  Port* CreateRelayPort(const SocketAddress& addr,
>+                        RelayType rtype,
>+                        ProtocolType int_proto,
>+                        ProtocolType ext_proto) {
>     if (rtype == RELAY_TURN) {
>       return CreateTurnPort(addr, &socket_factory_, int_proto, ext_proto);
>     } else {
>@@ -526,7 +522,8 @@ class PortTest : public testing::Test, public sigslot::has_slots<> {
>   }
>   TurnPort* CreateTurnPort(const SocketAddress& addr,
>                            PacketSocketFactory* socket_factory,
>-                           ProtocolType int_proto, ProtocolType ext_proto) {
>+                           ProtocolType int_proto,
>+                           ProtocolType ext_proto) {
>     SocketAddress server_addr =
>         int_proto == PROTO_TCP ? kTurnTcpIntAddr : kTurnUdpIntAddr;
>     return CreateTurnPort(addr, socket_factory, int_proto, ext_proto,
>@@ -534,19 +531,21 @@ class PortTest : public testing::Test, public sigslot::has_slots<> {
>   }
>   TurnPort* CreateTurnPort(const SocketAddress& addr,
>                            PacketSocketFactory* socket_factory,
>-                           ProtocolType int_proto, ProtocolType ext_proto,
>+                           ProtocolType int_proto,
>+                           ProtocolType ext_proto,
>                            const rtc::SocketAddress& server_addr) {
>     return TurnPort::Create(
>         &main_, socket_factory, MakeNetwork(addr), 0, 0, username_, password_,
>         ProtocolAddress(server_addr, int_proto), kRelayCredentials, 0,
>         std::string(), std::vector<std::string>(), std::vector<std::string>(),
>-        nullptr);
>+        nullptr, nullptr);
>   }
>   RelayPort* CreateGturnPort(const SocketAddress& addr,
>-                             ProtocolType int_proto, ProtocolType ext_proto) {
>+                             ProtocolType int_proto,
>+                             ProtocolType ext_proto) {
>     RelayPort* port = CreateGturnPort(addr);
>-    SocketAddress addrs[] =
>-        { kRelayUdpIntAddr, kRelayTcpIntAddr, kRelaySslTcpIntAddr };
>+    SocketAddress addrs[] = {kRelayUdpIntAddr, kRelayTcpIntAddr,
>+                             kRelaySslTcpIntAddr};
>     port->AddServerAddress(ProtocolAddress(addrs[int_proto], int_proto));
>     return port;
>   }
>@@ -560,7 +559,7 @@ class PortTest : public testing::Test, public sigslot::has_slots<> {
>     // other side can connect to this port using a non-UDP protocol.
>   }
>   rtc::NATServer* CreateNatServer(const SocketAddress& addr,
>-                                        rtc::NATType type) {
>+                                  rtc::NATType type) {
>     return new rtc::NATServer(type, ss_.get(), addr, addr, ss_.get(), addr);
>   }
>   static const char* StunName(NATType type) {
>@@ -612,10 +611,14 @@ class PortTest : public testing::Test, public sigslot::has_slots<> {
>   void ExpectPortsCanConnect(bool can_connect, Port* p1, Port* p2);
> 
>   // This does all the work and then deletes |port1| and |port2|.
>-  void TestConnectivity(const char* name1, Port* port1,
>-                        const char* name2, Port* port2,
>-                        bool accept, bool same_addr1,
>-                        bool same_addr2, bool possible);
>+  void TestConnectivity(const char* name1,
>+                        Port* port1,
>+                        const char* name2,
>+                        Port* port2,
>+                        bool accept,
>+                        bool same_addr1,
>+                        bool same_addr2,
>+                        bool possible);
> 
>   // This connects the provided channels which have already started.  |ch1|
>   // should have its Connection created (either through CreateConnection() or
>@@ -730,8 +733,7 @@ class PortTest : public testing::Test, public sigslot::has_slots<> {
> 
>       // Verify that we could still connect channels.
>       ConnectStartedChannels(&ch1, &ch2);
>-      EXPECT_TRUE_WAIT(ch1.connection_ready_to_send(),
>-                       kTcpReconnectTimeout);
>+      EXPECT_TRUE_WAIT(ch1.connection_ready_to_send(), kTcpReconnectTimeout);
>       // Channel2 is the passive one so a new connection is created during
>       // reconnect. This new connection should never have issued ENOTCONN
>       // hence the connection_ready_to_send() should be false.
>@@ -760,8 +762,8 @@ class PortTest : public testing::Test, public sigslot::has_slots<> {
>   IceMessage* CreateStunMessageWithUsername(int type,
>                                             const std::string& username) {
>     IceMessage* msg = CreateStunMessage(type);
>-    msg->AddAttribute(
>-        rtc::MakeUnique<StunByteStringAttribute>(STUN_ATTR_USERNAME, username));
>+    msg->AddAttribute(absl::make_unique<StunByteStringAttribute>(
>+        STUN_ATTR_USERNAME, username));
>     return msg;
>   }
>   TestPort* CreateTestPort(const rtc::SocketAddress& addr,
>@@ -792,9 +794,7 @@ class PortTest : public testing::Test, public sigslot::has_slots<> {
>     return port;
>   }
> 
>-  void OnRoleConflict(PortInterface* port) {
>-    role_conflict_ = true;
>-  }
>+  void OnRoleConflict(PortInterface* port) { role_conflict_ = true; }
>   bool role_conflict() const { return role_conflict_; }
> 
>   void ConnectToSignalDestroyed(PortInterface* port) {
>@@ -833,10 +833,14 @@ class PortTest : public testing::Test, public sigslot::has_slots<> {
>   int ports_destroyed_;
> };
> 
>-void PortTest::TestConnectivity(const char* name1, Port* port1,
>-                                const char* name2, Port* port2,
>-                                bool accept, bool same_addr1,
>-                                bool same_addr2, bool possible) {
>+void PortTest::TestConnectivity(const char* name1,
>+                                Port* port1,
>+                                const char* name2,
>+                                Port* port2,
>+                                bool accept,
>+                                bool same_addr1,
>+                                bool same_addr2,
>+                                bool possible) {
>   rtc::ScopedFakeClock clock;
>   RTC_LOG(LS_INFO) << "Test: " << name1 << " to " << name2 << ": ";
>   port1->set_component(cricket::ICE_CANDIDATE_COMPONENT_DEFAULT);
>@@ -870,7 +874,8 @@ void PortTest::TestConnectivity(const char* name1, Port* port1,
> 
>     // Ensure the ping came from the same address used for src.
>     // This is the case unless the source NAT was symmetric.
>-    if (same_addr1) EXPECT_EQ(ch2.remote_address(), GetAddress(port1));
>+    if (same_addr1)
>+      EXPECT_EQ(ch2.remote_address(), GetAddress(port1));
>     EXPECT_TRUE(same_addr2);
> 
>     // Send a ping from dst to src.
>@@ -976,9 +981,8 @@ class FakePacketSocketFactory : public rtc::PacketSocketFactory {
>   FakePacketSocketFactory()
>       : next_udp_socket_(NULL),
>         next_server_tcp_socket_(NULL),
>-        next_client_tcp_socket_(NULL) {
>-  }
>-  ~FakePacketSocketFactory() override { }
>+        next_client_tcp_socket_(NULL) {}
>+  ~FakePacketSocketFactory() override {}
> 
>   AsyncPacketSocket* CreateUdpSocket(const SocketAddress& address,
>                                      uint16_t min_port,
>@@ -1021,9 +1025,7 @@ class FakePacketSocketFactory : public rtc::PacketSocketFactory {
>   void set_next_client_tcp_socket(AsyncPacketSocket* next_client_tcp_socket) {
>     next_client_tcp_socket_ = next_client_tcp_socket;
>   }
>-  rtc::AsyncResolverInterface* CreateAsyncResolver() override {
>-    return NULL;
>-  }
>+  rtc::AsyncResolverInterface* CreateAsyncResolver() override { return NULL; }
> 
>  private:
>   AsyncPacketSocket* next_udp_socket_;
>@@ -1035,33 +1037,30 @@ class FakeAsyncPacketSocket : public AsyncPacketSocket {
>  public:
>   // Returns current local address. Address may be set to NULL if the
>   // socket is not bound yet (GetState() returns STATE_BINDING).
>-  virtual SocketAddress GetLocalAddress() const {
>-    return SocketAddress();
>-  }
>+  virtual SocketAddress GetLocalAddress() const { return SocketAddress(); }
> 
>   // Returns remote address. Returns zeroes if this is not a client TCP socket.
>-  virtual SocketAddress GetRemoteAddress() const {
>-    return SocketAddress();
>-  }
>+  virtual SocketAddress GetRemoteAddress() const { return SocketAddress(); }
> 
>   // Send a packet.
>-  virtual int Send(const void *pv, size_t cb,
>+  virtual int Send(const void* pv,
>+                   size_t cb,
>                    const rtc::PacketOptions& options) {
>     return static_cast<int>(cb);
>   }
>-  virtual int SendTo(const void *pv, size_t cb, const SocketAddress& addr,
>+  virtual int SendTo(const void* pv,
>+                     size_t cb,
>+                     const SocketAddress& addr,
>                      const rtc::PacketOptions& options) {
>     return static_cast<int>(cb);
>   }
>-  virtual int Close() {
>-    return 0;
>-  }
>+  virtual int Close() { return 0; }
> 
>   virtual State GetState() const { return state_; }
>   virtual int GetOption(Socket::Option opt, int* value) { return 0; }
>   virtual int SetOption(Socket::Option opt, int value) { return 0; }
>   virtual int GetError() const { return 0; }
>-  virtual void SetError(int error) { }
>+  virtual void SetError(int error) {}
> 
>   void set_state(State state) { state_ = state; }
> 
>@@ -1351,8 +1350,8 @@ TEST_F(PortTest, TestConnectionDead) {
>   conn->ReceivedPing();
>   int64_t after_last_receiving = rtc::TimeMillis();
>   // The connection will be dead after DEAD_CONNECTION_RECEIVE_TIMEOUT
>-  conn->UpdateState(
>-      before_last_receiving + DEAD_CONNECTION_RECEIVE_TIMEOUT - 1);
>+  conn->UpdateState(before_last_receiving + DEAD_CONNECTION_RECEIVE_TIMEOUT -
>+                    1);
>   rtc::Thread::Current()->ProcessMessages(100);
>   EXPECT_TRUE(ch1.conn() != nullptr);
>   conn->UpdateState(after_last_receiving + DEAD_CONNECTION_RECEIVE_TIMEOUT + 1);
>@@ -1384,16 +1383,15 @@ TEST_F(PortTest, TestLoopbackCall) {
>   lport->SetIceTiebreaker(kTiebreaker1);
>   lport->PrepareAddress();
>   ASSERT_FALSE(lport->Candidates().empty());
>-  Connection* conn = lport->CreateConnection(lport->Candidates()[0],
>-                                             Port::ORIGIN_MESSAGE);
>+  Connection* conn =
>+      lport->CreateConnection(lport->Candidates()[0], Port::ORIGIN_MESSAGE);
>   conn->Ping(0);
> 
>   ASSERT_TRUE_WAIT(lport->last_stun_msg() != NULL, kDefaultTimeout);
>   IceMessage* msg = lport->last_stun_msg();
>   EXPECT_EQ(STUN_BINDING_REQUEST, msg->type());
>   conn->OnReadPacket(lport->last_stun_buf()->data<char>(),
>-                     lport->last_stun_buf()->size(),
>-                     rtc::PacketTime());
>+                     lport->last_stun_buf()->size(), rtc::PacketTime());
>   ASSERT_TRUE_WAIT(lport->last_stun_msg() != NULL, kDefaultTimeout);
>   msg = lport->last_stun_msg();
>   EXPECT_EQ(STUN_BINDING_RESPONSE, msg->type());
>@@ -1403,8 +1401,8 @@ TEST_F(PortTest, TestLoopbackCall) {
>   lport->Reset();
>   lport->AddCandidateAddress(kLocalAddr2);
>   // Creating a different connection as |conn| is receiving.
>-  Connection* conn1 = lport->CreateConnection(lport->Candidates()[1],
>-                                              Port::ORIGIN_MESSAGE);
>+  Connection* conn1 =
>+      lport->CreateConnection(lport->Candidates()[1], Port::ORIGIN_MESSAGE);
>   conn1->Ping(0);
> 
>   ASSERT_TRUE_WAIT(lport->last_stun_msg() != NULL, kDefaultTimeout);
>@@ -1412,13 +1410,13 @@ TEST_F(PortTest, TestLoopbackCall) {
>   EXPECT_EQ(STUN_BINDING_REQUEST, msg->type());
>   std::unique_ptr<IceMessage> modified_req(
>       CreateStunMessage(STUN_BINDING_REQUEST));
>-  const StunByteStringAttribute* username_attr = msg->GetByteString(
>-      STUN_ATTR_USERNAME);
>-  modified_req->AddAttribute(rtc::MakeUnique<StunByteStringAttribute>(
>+  const StunByteStringAttribute* username_attr =
>+      msg->GetByteString(STUN_ATTR_USERNAME);
>+  modified_req->AddAttribute(absl::make_unique<StunByteStringAttribute>(
>       STUN_ATTR_USERNAME, username_attr->GetString()));
>   // To make sure we receive error response, adding tiebreaker less than
>   // what's present in request.
>-  modified_req->AddAttribute(rtc::MakeUnique<StunUInt64Attribute>(
>+  modified_req->AddAttribute(absl::make_unique<StunUInt64Attribute>(
>       STUN_ATTR_ICE_CONTROLLING, kTiebreaker1 - 1));
>   modified_req->AddMessageIntegrity("lpass");
>   modified_req->AddFingerprint();
>@@ -1451,10 +1449,10 @@ TEST_F(PortTest, TestIceRoleConflict) {
>   rport->PrepareAddress();
>   ASSERT_FALSE(lport->Candidates().empty());
>   ASSERT_FALSE(rport->Candidates().empty());
>-  Connection* lconn = lport->CreateConnection(rport->Candidates()[0],
>-                                              Port::ORIGIN_MESSAGE);
>-  Connection* rconn = rport->CreateConnection(lport->Candidates()[0],
>-                                              Port::ORIGIN_MESSAGE);
>+  Connection* lconn =
>+      lport->CreateConnection(rport->Candidates()[0], Port::ORIGIN_MESSAGE);
>+  Connection* rconn =
>+      rport->CreateConnection(lport->Candidates()[0], Port::ORIGIN_MESSAGE);
>   rconn->Ping(0);
> 
>   ASSERT_TRUE_WAIT(rport->last_stun_msg() != NULL, kDefaultTimeout);
>@@ -1462,8 +1460,7 @@ TEST_F(PortTest, TestIceRoleConflict) {
>   EXPECT_EQ(STUN_BINDING_REQUEST, msg->type());
>   // Send rport binding request to lport.
>   lconn->OnReadPacket(rport->last_stun_buf()->data<char>(),
>-                      rport->last_stun_buf()->size(),
>-                      rtc::PacketTime());
>+                      rport->last_stun_buf()->size(), rtc::PacketTime());
> 
>   ASSERT_TRUE_WAIT(lport->last_stun_msg() != NULL, kDefaultTimeout);
>   EXPECT_EQ(STUN_BINDING_RESPONSE, lport->last_stun_msg()->type());
>@@ -1474,15 +1471,14 @@ TEST_F(PortTest, TestTcpNoDelay) {
>   TCPPort* port1 = CreateTcpPort(kLocalAddr1);
>   port1->SetIceRole(cricket::ICEROLE_CONTROLLING);
>   int option_value = -1;
>-  int success = port1->GetOption(rtc::Socket::OPT_NODELAY,
>-                                 &option_value);
>+  int success = port1->GetOption(rtc::Socket::OPT_NODELAY, &option_value);
>   ASSERT_EQ(0, success);  // GetOption() should complete successfully w/ 0
>   ASSERT_EQ(1, option_value);
>   delete port1;
> }
> 
> TEST_F(PortTest, TestDelayedBindingUdp) {
>-  FakeAsyncPacketSocket *socket = new FakeAsyncPacketSocket();
>+  FakeAsyncPacketSocket* socket = new FakeAsyncPacketSocket();
>   FakePacketSocketFactory socket_factory;
> 
>   socket_factory.set_next_udp_socket(socket);
>@@ -1498,7 +1494,7 @@ TEST_F(PortTest, TestDelayedBindingUdp) {
> }
> 
> TEST_F(PortTest, TestDelayedBindingTcp) {
>-  FakeAsyncPacketSocket *socket = new FakeAsyncPacketSocket();
>+  FakeAsyncPacketSocket* socket = new FakeAsyncPacketSocket();
>   FakePacketSocketFactory socket_factory;
> 
>   socket_factory.set_next_server_tcp_socket(socket);
>@@ -1516,12 +1512,11 @@ TEST_F(PortTest, TestDelayedBindingTcp) {
> void PortTest::TestCrossFamilyPorts(int type) {
>   FakePacketSocketFactory factory;
>   std::unique_ptr<Port> ports[4];
>-  SocketAddress addresses[4] = {SocketAddress("192.168.1.3", 0),
>-                                SocketAddress("192.168.1.4", 0),
>-                                SocketAddress("2001:db8::1", 0),
>-                                SocketAddress("2001:db8::2", 0)};
>+  SocketAddress addresses[4] = {
>+      SocketAddress("192.168.1.3", 0), SocketAddress("192.168.1.4", 0),
>+      SocketAddress("2001:db8::1", 0), SocketAddress("2001:db8::2", 0)};
>   for (int i = 0; i < 4; i++) {
>-    FakeAsyncPacketSocket *socket = new FakeAsyncPacketSocket();
>+    FakeAsyncPacketSocket* socket = new FakeAsyncPacketSocket();
>     if (type == SOCK_DGRAM) {
>       factory.set_next_udp_socket(socket);
>       ports[i].reset(CreateUdpPort(addresses[i], &factory));
>@@ -1572,8 +1567,7 @@ TEST_F(PortTest, TestSkipCrossFamilyUdp) {
> }
> 
> void PortTest::ExpectPortsCanConnect(bool can_connect, Port* p1, Port* p2) {
>-  Connection* c = p1->CreateConnection(GetCandidate(p2),
>-                                       Port::ORIGIN_MESSAGE);
>+  Connection* c = p1->CreateConnection(GetCandidate(p2), Port::ORIGIN_MESSAGE);
>   if (can_connect) {
>     EXPECT_FALSE(NULL == c);
>     EXPECT_EQ(1U, p1->connections().size());
>@@ -1586,12 +1580,11 @@ void PortTest::ExpectPortsCanConnect(bool can_connect, Port* p1, Port* p2) {
> TEST_F(PortTest, TestUdpV6CrossTypePorts) {
>   FakePacketSocketFactory factory;
>   std::unique_ptr<Port> ports[4];
>-  SocketAddress addresses[4] = {SocketAddress("2001:db8::1", 0),
>-                                SocketAddress("fe80::1", 0),
>-                                SocketAddress("fe80::2", 0),
>-                                SocketAddress("::1", 0)};
>+  SocketAddress addresses[4] = {
>+      SocketAddress("2001:db8::1", 0), SocketAddress("fe80::1", 0),
>+      SocketAddress("fe80::2", 0), SocketAddress("::1", 0)};
>   for (int i = 0; i < 4; i++) {
>-    FakeAsyncPacketSocket *socket = new FakeAsyncPacketSocket();
>+    FakeAsyncPacketSocket* socket = new FakeAsyncPacketSocket();
>     factory.set_next_udp_socket(socket);
>     ports[i].reset(CreateUdpPort(addresses[i], &factory));
>     socket->set_state(AsyncPacketSocket::STATE_BINDING);
>@@ -1619,33 +1612,28 @@ TEST_F(PortTest, TestUdpV6CrossTypePorts) {
> TEST_F(PortTest, TestDefaultDscpValue) {
>   int dscp;
>   std::unique_ptr<UDPPort> udpport(CreateUdpPort(kLocalAddr1));
>-  EXPECT_EQ(0, udpport->SetOption(rtc::Socket::OPT_DSCP,
>-                                  rtc::DSCP_CS6));
>+  EXPECT_EQ(0, udpport->SetOption(rtc::Socket::OPT_DSCP, rtc::DSCP_CS6));
>   EXPECT_EQ(0, udpport->GetOption(rtc::Socket::OPT_DSCP, &dscp));
>   std::unique_ptr<TCPPort> tcpport(CreateTcpPort(kLocalAddr1));
>-  EXPECT_EQ(0, tcpport->SetOption(rtc::Socket::OPT_DSCP,
>-                                 rtc::DSCP_AF31));
>+  EXPECT_EQ(0, tcpport->SetOption(rtc::Socket::OPT_DSCP, rtc::DSCP_AF31));
>   EXPECT_EQ(0, tcpport->GetOption(rtc::Socket::OPT_DSCP, &dscp));
>   EXPECT_EQ(rtc::DSCP_AF31, dscp);
>   std::unique_ptr<StunPort> stunport(
>       CreateStunPort(kLocalAddr1, nat_socket_factory1()));
>-  EXPECT_EQ(0, stunport->SetOption(rtc::Socket::OPT_DSCP,
>-                                  rtc::DSCP_AF41));
>+  EXPECT_EQ(0, stunport->SetOption(rtc::Socket::OPT_DSCP, rtc::DSCP_AF41));
>   EXPECT_EQ(0, stunport->GetOption(rtc::Socket::OPT_DSCP, &dscp));
>   EXPECT_EQ(rtc::DSCP_AF41, dscp);
>   std::unique_ptr<TurnPort> turnport1(
>       CreateTurnPort(kLocalAddr1, nat_socket_factory1(), PROTO_UDP, PROTO_UDP));
>   // Socket is created in PrepareAddress.
>   turnport1->PrepareAddress();
>-  EXPECT_EQ(0, turnport1->SetOption(rtc::Socket::OPT_DSCP,
>-                                  rtc::DSCP_CS7));
>+  EXPECT_EQ(0, turnport1->SetOption(rtc::Socket::OPT_DSCP, rtc::DSCP_CS7));
>   EXPECT_EQ(0, turnport1->GetOption(rtc::Socket::OPT_DSCP, &dscp));
>   EXPECT_EQ(rtc::DSCP_CS7, dscp);
>   // This will verify correct value returned without the socket.
>   std::unique_ptr<TurnPort> turnport2(
>       CreateTurnPort(kLocalAddr1, nat_socket_factory1(), PROTO_UDP, PROTO_UDP));
>-  EXPECT_EQ(0, turnport2->SetOption(rtc::Socket::OPT_DSCP,
>-                                  rtc::DSCP_CS6));
>+  EXPECT_EQ(0, turnport2->SetOption(rtc::Socket::OPT_DSCP, rtc::DSCP_CS6));
>   EXPECT_EQ(0, turnport2->GetOption(rtc::Socket::OPT_DSCP, &dscp));
>   EXPECT_EQ(rtc::DSCP_CS6, dscp);
> }
>@@ -1665,10 +1653,10 @@ TEST_F(PortTest, TestSendStunMessage) {
>   lport->PrepareAddress();
>   rport->PrepareAddress();
>   ASSERT_FALSE(rport->Candidates().empty());
>-  Connection* lconn = lport->CreateConnection(
>-      rport->Candidates()[0], Port::ORIGIN_MESSAGE);
>-  Connection* rconn = rport->CreateConnection(
>-      lport->Candidates()[0], Port::ORIGIN_MESSAGE);
>+  Connection* lconn =
>+      lport->CreateConnection(rport->Candidates()[0], Port::ORIGIN_MESSAGE);
>+  Connection* rconn =
>+      rport->CreateConnection(lport->Candidates()[0], Port::ORIGIN_MESSAGE);
>   lconn->Ping(0);
> 
>   // Check that it's a proper BINDING-REQUEST.
>@@ -1720,8 +1708,8 @@ TEST_F(PortTest, TestSendStunMessage) {
>   EXPECT_EQ(1U, rconn->stats().sent_ping_responses);
> 
>   EXPECT_FALSE(msg->IsLegacy());
>-  const StunAddressAttribute* addr_attr = msg->GetAddress(
>-      STUN_ATTR_XOR_MAPPED_ADDRESS);
>+  const StunAddressAttribute* addr_attr =
>+      msg->GetAddress(STUN_ATTR_XOR_MAPPED_ADDRESS);
>   ASSERT_TRUE(addr_attr != NULL);
>   EXPECT_EQ(lport->Candidates()[0].address(), addr_attr->GetAddress());
>   EXPECT_TRUE(msg->GetByteString(STUN_ATTR_MESSAGE_INTEGRITY) != NULL);
>@@ -1744,10 +1732,9 @@ TEST_F(PortTest, TestSendStunMessage) {
> 
>   // Respond with a BINDING-ERROR-RESPONSE. This wouldn't happen in real life,
>   // but we can do it here.
>-  rport->SendBindingErrorResponse(request.get(),
>-                                  lport->Candidates()[0].address(),
>-                                  STUN_ERROR_SERVER_ERROR,
>-                                  STUN_ERROR_REASON_SERVER_ERROR);
>+  rport->SendBindingErrorResponse(
>+      request.get(), lport->Candidates()[0].address(), STUN_ERROR_SERVER_ERROR,
>+      STUN_ERROR_REASON_SERVER_ERROR);
>   msg = rport->last_stun_msg();
>   ASSERT_TRUE(msg != NULL);
>   EXPECT_EQ(STUN_BINDING_ERROR_RESPONSE, msg->type());
>@@ -1830,10 +1817,10 @@ TEST_F(PortTest, TestNomination) {
>   rport->PrepareAddress();
>   ASSERT_FALSE(lport->Candidates().empty());
>   ASSERT_FALSE(rport->Candidates().empty());
>-  Connection* lconn = lport->CreateConnection(rport->Candidates()[0],
>-                                              Port::ORIGIN_MESSAGE);
>-  Connection* rconn = rport->CreateConnection(lport->Candidates()[0],
>-                                              Port::ORIGIN_MESSAGE);
>+  Connection* lconn =
>+      lport->CreateConnection(rport->Candidates()[0], Port::ORIGIN_MESSAGE);
>+  Connection* rconn =
>+      rport->CreateConnection(lport->Candidates()[0], Port::ORIGIN_MESSAGE);
> 
>   // |lconn| is controlling, |rconn| is controlled.
>   uint32_t nomination = 1234;
>@@ -1850,8 +1837,7 @@ TEST_F(PortTest, TestNomination) {
>   ASSERT_TRUE_WAIT(lport->last_stun_msg(), kDefaultTimeout);
>   ASSERT_TRUE(lport->last_stun_buf());
>   rconn->OnReadPacket(lport->last_stun_buf()->data<char>(),
>-                      lport->last_stun_buf()->size(),
>-                      rtc::PacketTime());
>+                      lport->last_stun_buf()->size(), rtc::PacketTime());
>   EXPECT_EQ(nomination, rconn->remote_nomination());
>   EXPECT_FALSE(lconn->nominated());
>   EXPECT_TRUE(rconn->nominated());
>@@ -1863,8 +1849,7 @@ TEST_F(PortTest, TestNomination) {
>   ASSERT_TRUE_WAIT(rport->last_stun_msg(), kDefaultTimeout);
>   ASSERT_TRUE(rport->last_stun_buf());
>   lconn->OnReadPacket(rport->last_stun_buf()->data<char>(),
>-                      rport->last_stun_buf()->size(),
>-                      rtc::PacketTime());
>+                      rport->last_stun_buf()->size(), rtc::PacketTime());
>   EXPECT_EQ(nomination, lconn->acked_nomination());
>   EXPECT_TRUE(lconn->nominated());
>   EXPECT_TRUE(rconn->nominated());
>@@ -1888,28 +1873,28 @@ TEST_F(PortTest, TestRoundTripTime) {
>   rport->PrepareAddress();
>   ASSERT_FALSE(lport->Candidates().empty());
>   ASSERT_FALSE(rport->Candidates().empty());
>-  Connection* lconn = lport->CreateConnection(rport->Candidates()[0],
>-                                              Port::ORIGIN_MESSAGE);
>-  Connection* rconn = rport->CreateConnection(lport->Candidates()[0],
>-                                              Port::ORIGIN_MESSAGE);
>+  Connection* lconn =
>+      lport->CreateConnection(rport->Candidates()[0], Port::ORIGIN_MESSAGE);
>+  Connection* rconn =
>+      rport->CreateConnection(lport->Candidates()[0], Port::ORIGIN_MESSAGE);
> 
>   EXPECT_EQ(0u, lconn->stats().total_round_trip_time_ms);
>   EXPECT_FALSE(lconn->stats().current_round_trip_time_ms);
> 
>-  SendPingAndReceiveResponse(
>-      lconn, lport.get(), rconn, rport.get(), &clock, 10);
>+  SendPingAndReceiveResponse(lconn, lport.get(), rconn, rport.get(), &clock,
>+                             10);
>   EXPECT_EQ(10u, lconn->stats().total_round_trip_time_ms);
>   ASSERT_TRUE(lconn->stats().current_round_trip_time_ms);
>   EXPECT_EQ(10u, *lconn->stats().current_round_trip_time_ms);
> 
>-  SendPingAndReceiveResponse(
>-      lconn, lport.get(), rconn, rport.get(), &clock, 20);
>+  SendPingAndReceiveResponse(lconn, lport.get(), rconn, rport.get(), &clock,
>+                             20);
>   EXPECT_EQ(30u, lconn->stats().total_round_trip_time_ms);
>   ASSERT_TRUE(lconn->stats().current_round_trip_time_ms);
>   EXPECT_EQ(20u, *lconn->stats().current_round_trip_time_ms);
> 
>-  SendPingAndReceiveResponse(
>-      lconn, lport.get(), rconn, rport.get(), &clock, 30);
>+  SendPingAndReceiveResponse(lconn, lport.get(), rconn, rport.get(), &clock,
>+                             30);
>   EXPECT_EQ(60u, lconn->stats().total_round_trip_time_ms);
>   ASSERT_TRUE(lconn->stats().current_round_trip_time_ms);
>   EXPECT_EQ(30u, *lconn->stats().current_round_trip_time_ms);
>@@ -1929,16 +1914,16 @@ TEST_F(PortTest, TestUseCandidateAttribute) {
>   lport->PrepareAddress();
>   rport->PrepareAddress();
>   ASSERT_FALSE(rport->Candidates().empty());
>-  Connection* lconn = lport->CreateConnection(
>-      rport->Candidates()[0], Port::ORIGIN_MESSAGE);
>+  Connection* lconn =
>+      lport->CreateConnection(rport->Candidates()[0], Port::ORIGIN_MESSAGE);
>   lconn->Ping(0);
>   ASSERT_TRUE_WAIT(lport->last_stun_msg() != NULL, kDefaultTimeout);
>   IceMessage* msg = lport->last_stun_msg();
>   const StunUInt64Attribute* ice_controlling_attr =
>       msg->GetUInt64(STUN_ATTR_ICE_CONTROLLING);
>   ASSERT_TRUE(ice_controlling_attr != NULL);
>-  const StunByteStringAttribute* use_candidate_attr = msg->GetByteString(
>-      STUN_ATTR_USE_CANDIDATE);
>+  const StunByteStringAttribute* use_candidate_attr =
>+      msg->GetByteString(STUN_ATTR_USE_CANDIDATE);
>   ASSERT_TRUE(use_candidate_attr != NULL);
> }
> 
>@@ -2061,8 +2046,8 @@ TEST_F(PortTest, TestHandleStunMessage) {
> 
>   // BINDING-REQUEST from local to remote with valid ICE username,
>   // MESSAGE-INTEGRITY, and FINGERPRINT.
>-  in_msg.reset(CreateStunMessageWithUsername(STUN_BINDING_REQUEST,
>-                                             "rfrag:lfrag"));
>+  in_msg.reset(
>+      CreateStunMessageWithUsername(STUN_BINDING_REQUEST, "rfrag:lfrag"));
>   in_msg->AddMessageIntegrity("rpass");
>   in_msg->AddFingerprint();
>   WriteStunMessage(in_msg.get(), buf.get());
>@@ -2073,7 +2058,7 @@ TEST_F(PortTest, TestHandleStunMessage) {
> 
>   // BINDING-RESPONSE without username, with MESSAGE-INTEGRITY and FINGERPRINT.
>   in_msg.reset(CreateStunMessage(STUN_BINDING_RESPONSE));
>-  in_msg->AddAttribute(rtc::MakeUnique<StunXorAddressAttribute>(
>+  in_msg->AddAttribute(absl::make_unique<StunXorAddressAttribute>(
>       STUN_ATTR_XOR_MAPPED_ADDRESS, kLocalAddr2));
>   in_msg->AddMessageIntegrity("rpass");
>   in_msg->AddFingerprint();
>@@ -2085,7 +2070,7 @@ TEST_F(PortTest, TestHandleStunMessage) {
> 
>   // BINDING-ERROR-RESPONSE without username, with error, M-I, and FINGERPRINT.
>   in_msg.reset(CreateStunMessage(STUN_BINDING_ERROR_RESPONSE));
>-  in_msg->AddAttribute(rtc::MakeUnique<StunErrorCodeAttribute>(
>+  in_msg->AddAttribute(absl::make_unique<StunErrorCodeAttribute>(
>       STUN_ATTR_ERROR_CODE, STUN_ERROR_SERVER_ERROR,
>       STUN_ERROR_REASON_SERVER_ERROR));
>   in_msg->AddFingerprint();
>@@ -2097,7 +2082,7 @@ TEST_F(PortTest, TestHandleStunMessage) {
>   ASSERT_TRUE(out_msg->GetErrorCode() != NULL);
>   EXPECT_EQ(STUN_ERROR_SERVER_ERROR, out_msg->GetErrorCode()->code());
>   EXPECT_EQ(std::string(STUN_ERROR_REASON_SERVER_ERROR),
>-      out_msg->GetErrorCode()->reason());
>+            out_msg->GetErrorCode()->reason());
> }
> 
> // Tests handling of ICE binding requests with missing or incorrect usernames.
>@@ -2143,8 +2128,8 @@ TEST_F(PortTest, TestHandleStunMessageBadUsername) {
>   EXPECT_EQ(STUN_ERROR_UNAUTHORIZED, port->last_stun_error_code());
> 
>   // BINDING-REQUEST with reversed username.
>-  in_msg.reset(CreateStunMessageWithUsername(STUN_BINDING_REQUEST,
>-                                            "lfrag:rfrag"));
>+  in_msg.reset(
>+      CreateStunMessageWithUsername(STUN_BINDING_REQUEST, "lfrag:rfrag"));
>   in_msg->AddMessageIntegrity("rpass");
>   in_msg->AddFingerprint();
>   WriteStunMessage(in_msg.get(), buf.get());
>@@ -2155,8 +2140,8 @@ TEST_F(PortTest, TestHandleStunMessageBadUsername) {
>   EXPECT_EQ(STUN_ERROR_UNAUTHORIZED, port->last_stun_error_code());
> 
>   // BINDING-REQUEST with garbage username.
>-  in_msg.reset(CreateStunMessageWithUsername(STUN_BINDING_REQUEST,
>-                                             "abcd:efgh"));
>+  in_msg.reset(
>+      CreateStunMessageWithUsername(STUN_BINDING_REQUEST, "abcd:efgh"));
>   in_msg->AddMessageIntegrity("rpass");
>   in_msg->AddFingerprint();
>   WriteStunMessage(in_msg.get(), buf.get());
>@@ -2179,8 +2164,8 @@ TEST_F(PortTest, TestHandleStunMessageBadMessageIntegrity) {
> 
>   // BINDING-REQUEST from local to remote with valid ICE username and
>   // FINGERPRINT, but no MESSAGE-INTEGRITY.
>-  in_msg.reset(CreateStunMessageWithUsername(STUN_BINDING_REQUEST,
>-                                             "rfrag:lfrag"));
>+  in_msg.reset(
>+      CreateStunMessageWithUsername(STUN_BINDING_REQUEST, "rfrag:lfrag"));
>   in_msg->AddFingerprint();
>   WriteStunMessage(in_msg.get(), buf.get());
>   EXPECT_TRUE(port->GetStunMessage(buf->Data(), buf->Length(), addr, &out_msg,
>@@ -2191,8 +2176,8 @@ TEST_F(PortTest, TestHandleStunMessageBadMessageIntegrity) {
> 
>   // BINDING-REQUEST from local to remote with valid ICE username and
>   // FINGERPRINT, but invalid MESSAGE-INTEGRITY.
>-  in_msg.reset(CreateStunMessageWithUsername(STUN_BINDING_REQUEST,
>-                                             "rfrag:lfrag"));
>+  in_msg.reset(
>+      CreateStunMessageWithUsername(STUN_BINDING_REQUEST, "rfrag:lfrag"));
>   in_msg->AddMessageIntegrity("invalid");
>   in_msg->AddFingerprint();
>   WriteStunMessage(in_msg.get(), buf.get());
>@@ -2219,8 +2204,8 @@ TEST_F(PortTest, TestHandleStunMessageBadFingerprint) {
> 
>   // BINDING-REQUEST from local to remote with valid ICE username and
>   // MESSAGE-INTEGRITY, but no FINGERPRINT; GetStunMessage should fail.
>-  in_msg.reset(CreateStunMessageWithUsername(STUN_BINDING_REQUEST,
>-                                             "rfrag:lfrag"));
>+  in_msg.reset(
>+      CreateStunMessageWithUsername(STUN_BINDING_REQUEST, "rfrag:lfrag"));
>   in_msg->AddMessageIntegrity("rpass");
>   WriteStunMessage(in_msg.get(), buf.get());
>   EXPECT_FALSE(port->GetStunMessage(buf->Data(), buf->Length(), addr, &out_msg,
>@@ -2237,7 +2222,7 @@ TEST_F(PortTest, TestHandleStunMessageBadFingerprint) {
> 
>   // Valid BINDING-RESPONSE, except no FINGERPRINT.
>   in_msg.reset(CreateStunMessage(STUN_BINDING_RESPONSE));
>-  in_msg->AddAttribute(rtc::MakeUnique<StunXorAddressAttribute>(
>+  in_msg->AddAttribute(absl::make_unique<StunXorAddressAttribute>(
>       STUN_ATTR_XOR_MAPPED_ADDRESS, kLocalAddr2));
>   in_msg->AddMessageIntegrity("rpass");
>   WriteStunMessage(in_msg.get(), buf.get());
>@@ -2255,7 +2240,7 @@ TEST_F(PortTest, TestHandleStunMessageBadFingerprint) {
> 
>   // Valid BINDING-ERROR-RESPONSE, except no FINGERPRINT.
>   in_msg.reset(CreateStunMessage(STUN_BINDING_ERROR_RESPONSE));
>-  in_msg->AddAttribute(rtc::MakeUnique<StunErrorCodeAttribute>(
>+  in_msg->AddAttribute(absl::make_unique<StunErrorCodeAttribute>(
>       STUN_ATTR_ERROR_CODE, STUN_ERROR_SERVER_ERROR,
>       STUN_ERROR_REASON_SERVER_ERROR));
>   in_msg->AddMessageIntegrity("rpass");
>@@ -2308,10 +2293,10 @@ TEST_F(PortTest, TestHandleStunBindingIndication) {
>   ASSERT_FALSE(lport->Candidates().empty());
>   ASSERT_FALSE(rport->Candidates().empty());
> 
>-  Connection* lconn = lport->CreateConnection(rport->Candidates()[0],
>-                                              Port::ORIGIN_MESSAGE);
>-  Connection* rconn = rport->CreateConnection(lport->Candidates()[0],
>-                                              Port::ORIGIN_MESSAGE);
>+  Connection* lconn =
>+      lport->CreateConnection(rport->Candidates()[0], Port::ORIGIN_MESSAGE);
>+  Connection* rconn =
>+      rport->CreateConnection(lport->Candidates()[0], Port::ORIGIN_MESSAGE);
>   rconn->Ping(0);
> 
>   ASSERT_TRUE_WAIT(rport->last_stun_msg() != NULL, kDefaultTimeout);
>@@ -2319,8 +2304,7 @@ TEST_F(PortTest, TestHandleStunBindingIndication) {
>   EXPECT_EQ(STUN_BINDING_REQUEST, msg->type());
>   // Send rport binding request to lport.
>   lconn->OnReadPacket(rport->last_stun_buf()->data<char>(),
>-                      rport->last_stun_buf()->size(),
>-                      rtc::PacketTime());
>+                      rport->last_stun_buf()->size(), rtc::PacketTime());
>   ASSERT_TRUE_WAIT(lport->last_stun_msg() != NULL, kDefaultTimeout);
>   EXPECT_EQ(STUN_BINDING_RESPONSE, lport->last_stun_msg()->type());
>   int64_t last_ping_received1 = lconn->last_ping_received();
>@@ -2373,11 +2357,9 @@ TEST_F(PortTest, TestComputeCandidatePriority) {
> TEST_F(PortTest, TestFoundation) {
>   std::unique_ptr<TestPort> testport(
>       CreateTestPort(kLocalAddr1, "name", "pass"));
>-  testport->AddCandidateAddress(kLocalAddr1, kLocalAddr1,
>-                                LOCAL_PORT_TYPE,
>+  testport->AddCandidateAddress(kLocalAddr1, kLocalAddr1, LOCAL_PORT_TYPE,
>                                 cricket::ICE_TYPE_PREFERENCE_HOST, false);
>-  testport->AddCandidateAddress(kLocalAddr2, kLocalAddr1,
>-                                STUN_PORT_TYPE,
>+  testport->AddCandidateAddress(kLocalAddr2, kLocalAddr1, STUN_PORT_TYPE,
>                                 cricket::ICE_TYPE_PREFERENCE_SRFLX, true);
>   EXPECT_NE(testport->Candidates()[0].foundation(),
>             testport->Candidates()[1].foundation());
>@@ -2442,8 +2424,8 @@ TEST_F(PortTest, TestCandidateFoundation) {
>   // Running a second turn server, to get different base IP address.
>   SocketAddress kTurnUdpIntAddr2("99.99.98.4", STUN_SERVER_PORT);
>   SocketAddress kTurnUdpExtAddr2("99.99.98.5", 0);
>-  TestTurnServer turn_server2(
>-      rtc::Thread::Current(), kTurnUdpIntAddr2, kTurnUdpExtAddr2);
>+  TestTurnServer turn_server2(rtc::Thread::Current(), kTurnUdpIntAddr2,
>+                              kTurnUdpExtAddr2);
>   std::unique_ptr<Port> turnport3(
>       CreateTurnPort(kLocalAddr1, nat_socket_factory1(), PROTO_UDP, PROTO_UDP,
>                      kTurnUdpIntAddr2));
>@@ -2481,8 +2463,7 @@ TEST_F(PortTest, TestCandidateRelatedAddress) {
>   stunport->PrepareAddress();
>   ASSERT_EQ_WAIT(1U, stunport->Candidates().size(), kDefaultTimeout);
>   // Check STUN candidate address.
>-  EXPECT_EQ(stunport->Candidates()[0].address().ipaddr(),
>-            kNatAddr1.ipaddr());
>+  EXPECT_EQ(stunport->Candidates()[0].address().ipaddr(), kNatAddr1.ipaddr());
>   // Check STUN candidate related address.
>   EXPECT_EQ(stunport->Candidates()[0].related_address(),
>             stunport->GetLocalAddress());
>@@ -2495,8 +2476,7 @@ TEST_F(PortTest, TestCandidateRelatedAddress) {
>   relayport->PrepareAddress();
>   ASSERT_EQ_WAIT(1U, relayport->Candidates().size(), kDefaultTimeout);
>   // For Gturn related address is set to "0.0.0.0:0"
>-  EXPECT_EQ(rtc::SocketAddress(),
>-            relayport->Candidates()[0].related_address());
>+  EXPECT_EQ(rtc::SocketAddress(), relayport->Candidates()[0].related_address());
>   // Verifying the related address for TURN candidate.
>   // For TURN related address must be equal to the mapped address.
>   std::unique_ptr<Port> turnport(
>@@ -2538,8 +2518,8 @@ TEST_F(PortTest, TestConnectionPriority) {
>   // pair priority = 2^32*MIN(G,D) + 2*MAX(G,D) + (G>D?1:0)
>   lport->SetIceRole(cricket::ICEROLE_CONTROLLING);
>   rport->SetIceRole(cricket::ICEROLE_CONTROLLED);
>-  Connection* lconn = lport->CreateConnection(
>-      rport->Candidates()[0], Port::ORIGIN_MESSAGE);
>+  Connection* lconn =
>+      lport->CreateConnection(rport->Candidates()[0], Port::ORIGIN_MESSAGE);
> #if defined(WEBRTC_WIN)
>   EXPECT_EQ(0x2001EE9FC003D0BU, lconn->priority());
> #else
>@@ -2548,8 +2528,8 @@ TEST_F(PortTest, TestConnectionPriority) {
> 
>   lport->SetIceRole(cricket::ICEROLE_CONTROLLED);
>   rport->SetIceRole(cricket::ICEROLE_CONTROLLING);
>-  Connection* rconn = rport->CreateConnection(
>-      lport->Candidates()[0], Port::ORIGIN_MESSAGE);
>+  Connection* rconn =
>+      rport->CreateConnection(lport->Candidates()[0], Port::ORIGIN_MESSAGE);
> #if defined(WEBRTC_WIN)
>   EXPECT_EQ(0x2001EE9FC003D0AU, rconn->priority());
> #else
>@@ -2557,6 +2537,12 @@ TEST_F(PortTest, TestConnectionPriority) {
> #endif
> }
> 
>+// Note that UpdateState takes into account the estimated RTT, and the
>+// correctness of using |kMaxExpectedSimulatedRtt| as an upper bound of RTT in
>+// the following tests depends on the link rate and the delay distriubtion
>+// configured in VirtualSocketServer::AddPacketToNetwork. The tests below use
>+// the default setup where the RTT is deterministically one, which generates an
>+// estimate given by |MINIMUM_RTT| = 100.
> TEST_F(PortTest, TestWritableState) {
>   rtc::ScopedFakeClock clock;
>   UDPPort* port1 = CreateUdpPort(kLocalAddr1);
>@@ -2602,7 +2588,8 @@ TEST_F(PortTest, TestWritableState) {
>   for (uint32_t i = 1; i <= CONNECTION_WRITE_CONNECT_FAILURES; ++i) {
>     ch1.Ping(i);
>   }
>-  int unreliable_timeout_delay = CONNECTION_WRITE_CONNECT_TIMEOUT + 500;
>+  int unreliable_timeout_delay =
>+      CONNECTION_WRITE_CONNECT_TIMEOUT + kMaxExpectedSimulatedRtt;
>   ch1.conn()->UpdateState(unreliable_timeout_delay);
>   EXPECT_EQ(Connection::STATE_WRITE_UNRELIABLE, ch1.conn()->write_state());
> 
>@@ -2613,14 +2600,13 @@ TEST_F(PortTest, TestWritableState) {
>   // responses.
>   EXPECT_EQ_SIMULATED_WAIT(Connection::STATE_WRITABLE,
>                            ch1.conn()->write_state(), kDefaultTimeout, clock);
>-
>   // Wait long enough for a full timeout (past however long we've already
>   // waited).
>   for (uint32_t i = 1; i <= CONNECTION_WRITE_CONNECT_FAILURES; ++i) {
>     ch1.Ping(unreliable_timeout_delay + i);
>   }
>   ch1.conn()->UpdateState(unreliable_timeout_delay + CONNECTION_WRITE_TIMEOUT +
>-                          500u);
>+                          kMaxExpectedSimulatedRtt);
>   EXPECT_EQ(Connection::STATE_WRITE_TIMEOUT, ch1.conn()->write_state());
> 
>   // Even if the connection has timed out, the Connection shouldn't block
>@@ -2631,6 +2617,64 @@ TEST_F(PortTest, TestWritableState) {
>   ch2.Stop();
> }
> 
>+// Test writability states using the configured threshold value to replace
>+// the default value given by |CONNECTION_WRITE_CONNECT_TIMEOUT| and
>+// |CONNECTION_WRITE_CONNECT_FAILURES|.
>+TEST_F(PortTest, TestWritableStateWithConfiguredThreshold) {
>+  rtc::ScopedFakeClock clock;
>+  UDPPort* port1 = CreateUdpPort(kLocalAddr1);
>+  port1->SetIceRole(cricket::ICEROLE_CONTROLLING);
>+  UDPPort* port2 = CreateUdpPort(kLocalAddr2);
>+  port2->SetIceRole(cricket::ICEROLE_CONTROLLED);
>+
>+  // Set up channels.
>+  TestChannel ch1(port1);
>+  TestChannel ch2(port2);
>+
>+  // Acquire addresses.
>+  ch1.Start();
>+  ch2.Start();
>+  ASSERT_EQ_SIMULATED_WAIT(1, ch1.complete_count(), kDefaultTimeout, clock);
>+  ASSERT_EQ_SIMULATED_WAIT(1, ch2.complete_count(), kDefaultTimeout, clock);
>+
>+  // Send a ping from src to dst.
>+  ch1.CreateConnection(GetCandidate(port2));
>+  ASSERT_TRUE(ch1.conn() != NULL);
>+  ch1.Ping();
>+  SIMULATED_WAIT(!ch2.remote_address().IsNil(), kShortTimeout, clock);
>+
>+  // Accept the connection to return the binding response, transition to
>+  // writable, and allow data to be sent.
>+  ch2.AcceptConnection(GetCandidate(port1));
>+  EXPECT_EQ_SIMULATED_WAIT(Connection::STATE_WRITABLE,
>+                           ch1.conn()->write_state(), kDefaultTimeout, clock);
>+
>+  ch1.conn()->set_unwritable_timeout(1000);
>+  ch1.conn()->set_unwritable_min_checks(3);
>+  // Send two checks.
>+  ch1.Ping(1);
>+  ch1.Ping(2);
>+  // We have not reached the timeout nor have we sent the minimum number of
>+  // checks to change the state to Unreliable.
>+  ch1.conn()->UpdateState(999);
>+  EXPECT_EQ(Connection::STATE_WRITABLE, ch1.conn()->write_state());
>+  // We have not sent the minimum number of checks without responses.
>+  ch1.conn()->UpdateState(1000 + kMaxExpectedSimulatedRtt);
>+  EXPECT_EQ(Connection::STATE_WRITABLE, ch1.conn()->write_state());
>+  // Last ping after which the candidate pair should become Unreliable after
>+  // timeout.
>+  ch1.Ping(3);
>+  // We have not reached the timeout.
>+  ch1.conn()->UpdateState(999);
>+  EXPECT_EQ(Connection::STATE_WRITABLE, ch1.conn()->write_state());
>+  // We should be in the state Unreliable now.
>+  ch1.conn()->UpdateState(1000 + kMaxExpectedSimulatedRtt);
>+  EXPECT_EQ(Connection::STATE_WRITE_UNRELIABLE, ch1.conn()->write_state());
>+
>+  ch1.Stop();
>+  ch2.Stop();
>+}
>+
> TEST_F(PortTest, TestTimeoutForNeverWritable) {
>   UDPPort* port1 = CreateUdpPort(kLocalAddr1);
>   port1->SetIceRole(cricket::ICEROLE_CONTROLLING);
>@@ -2653,7 +2697,7 @@ TEST_F(PortTest, TestTimeoutForNeverWritable) {
>   for (uint32_t i = 1; i <= CONNECTION_WRITE_CONNECT_FAILURES; ++i) {
>     ch1.Ping(i);
>   }
>-  ch1.conn()->UpdateState(CONNECTION_WRITE_TIMEOUT + 500u);
>+  ch1.conn()->UpdateState(CONNECTION_WRITE_TIMEOUT + kMaxExpectedSimulatedRtt);
>   EXPECT_EQ(Connection::STATE_WRITE_TIMEOUT, ch1.conn()->write_state());
> }
> 
>@@ -2662,9 +2706,9 @@ TEST_F(PortTest, TestTimeoutForNeverWritable) {
> // In this test |ch1| behaves like FULL mode client and we have created
> // port which responds to the ping message just like LITE client.
> TEST_F(PortTest, TestIceLiteConnectivity) {
>-  TestPort* ice_full_port = CreateTestPort(
>-      kLocalAddr1, "lfrag", "lpass",
>-      cricket::ICEROLE_CONTROLLING, kTiebreaker1);
>+  TestPort* ice_full_port =
>+      CreateTestPort(kLocalAddr1, "lfrag", "lpass",
>+                     cricket::ICEROLE_CONTROLLING, kTiebreaker1);
> 
>   std::unique_ptr<TestPort> ice_lite_port(
>       CreateTestPort(kLocalAddr2, "rfrag", "rpass", cricket::ICEROLE_CONTROLLED,
>@@ -2698,11 +2742,11 @@ TEST_F(PortTest, TestIceLiteConnectivity) {
>   // NOTE: Ideally we should't create connection at this stage from lite
>   // port, as it should be done only after receiving ping with USE_CANDIDATE.
>   // But we need a connection to send a response message.
>-  ice_lite_port->CreateConnection(
>-      ice_full_port->Candidates()[0], cricket::Port::ORIGIN_MESSAGE);
>+  ice_lite_port->CreateConnection(ice_full_port->Candidates()[0],
>+                                  cricket::Port::ORIGIN_MESSAGE);
>   std::unique_ptr<IceMessage> request(CopyStunMessage(msg));
>-  ice_lite_port->SendBindingResponse(
>-      request.get(), ice_full_port->Candidates()[0].address());
>+  ice_lite_port->SendBindingResponse(request.get(),
>+                                     ice_full_port->Candidates()[0].address());
> 
>   // Feeding the respone message from litemode to the full mode connection.
>   ch1.conn()->OnReadPacket(ice_lite_port->last_stun_buf()->data<char>(),
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/portallocator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/portallocator.cc
>index 84c1df24da8..ee0466f7963 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/portallocator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/portallocator.cc
>@@ -79,6 +79,19 @@ bool PortAllocatorSession::IsStopped() const {
>   return false;
> }
> 
>+void PortAllocatorSession::GetCandidateStatsFromReadyPorts(
>+    CandidateStatsList* candidate_stats_list) const {
>+  auto ports = ReadyPorts();
>+  for (auto* port : ports) {
>+    auto candidates = port->Candidates();
>+    for (const auto& candidate : candidates) {
>+      CandidateStats candidate_stats(candidate);
>+      port->GetStunStats(&candidate_stats.stun_stats);
>+      candidate_stats_list->push_back(std::move(candidate_stats));
>+    }
>+  }
>+}
>+
> uint32_t PortAllocatorSession::generation() {
>   return generation_;
> }
>@@ -94,16 +107,28 @@ PortAllocator::PortAllocator()
>       max_ipv6_networks_(kDefaultMaxIPv6Networks),
>       step_delay_(kDefaultStepDelay),
>       allow_tcp_listen_(true),
>-      candidate_filter_(CF_ALL) {}
>+      candidate_filter_(CF_ALL) {
>+  // The allocator will be attached to a thread in Initialize.
>+  thread_checker_.DetachFromThread();
>+}
> 
>-PortAllocator::~PortAllocator() = default;
>+void PortAllocator::Initialize() {
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  initialized_ = true;
>+}
>+
>+PortAllocator::~PortAllocator() {
>+  CheckRunOnValidThreadIfInitialized();
>+}
> 
> bool PortAllocator::SetConfiguration(
>     const ServerAddresses& stun_servers,
>     const std::vector<RelayServerConfig>& turn_servers,
>     int candidate_pool_size,
>     bool prune_turn_ports,
>-    webrtc::TurnCustomizer* turn_customizer) {
>+    webrtc::TurnCustomizer* turn_customizer,
>+    const absl::optional<int>& stun_candidate_keepalive_interval) {
>+  CheckRunOnValidThreadIfInitialized();
>   bool ice_servers_changed =
>       (stun_servers != stun_servers_ || turn_servers != turn_servers_);
>   stun_servers_ = stun_servers;
>@@ -113,8 +138,7 @@ bool PortAllocator::SetConfiguration(
>   if (candidate_pool_frozen_) {
>     if (candidate_pool_size != candidate_pool_size_) {
>       RTC_LOG(LS_ERROR)
>-          << "Trying to change candidate pool size after pool was "
>-          << "frozen.";
>+          << "Trying to change candidate pool size after pool was frozen.";
>       return false;
>     }
>     return true;
>@@ -142,6 +166,16 @@ bool PortAllocator::SetConfiguration(
>     pooled_sessions_.pop_front();
>   }
> 
>+  // |stun_candidate_keepalive_interval_| will be used in STUN port allocation
>+  // in future sessions. We also update the ready ports in the pooled sessions.
>+  // Ports in sessions that are taken and owned by P2PTransportChannel will be
>+  // updated there via IceConfig.
>+  stun_candidate_keepalive_interval_ = stun_candidate_keepalive_interval;
>+  for (const auto& session : pooled_sessions_) {
>+    session->SetStunKeepaliveIntervalForReadyPorts(
>+        stun_candidate_keepalive_interval_);
>+  }
>+
>   // If |candidate_pool_size_| is greater than the number of pooled sessions,
>   // create new sessions.
>   while (static_cast<int>(pooled_sessions_.size()) < candidate_pool_size_) {
>@@ -158,6 +192,7 @@ std::unique_ptr<PortAllocatorSession> PortAllocator::CreateSession(
>     int component,
>     const std::string& ice_ufrag,
>     const std::string& ice_pwd) {
>+  CheckRunOnValidThreadAndInitialized();
>   auto session = std::unique_ptr<PortAllocatorSession>(
>       CreateSessionInternal(content_name, component, ice_ufrag, ice_pwd));
>   session->SetCandidateFilter(candidate_filter());
>@@ -169,6 +204,7 @@ std::unique_ptr<PortAllocatorSession> PortAllocator::TakePooledSession(
>     int component,
>     const std::string& ice_ufrag,
>     const std::string& ice_pwd) {
>+  CheckRunOnValidThreadAndInitialized();
>   RTC_DCHECK(!ice_ufrag.empty());
>   RTC_DCHECK(!ice_pwd.empty());
>   if (pooled_sessions_.empty()) {
>@@ -185,6 +221,7 @@ std::unique_ptr<PortAllocatorSession> PortAllocator::TakePooledSession(
> }
> 
> const PortAllocatorSession* PortAllocator::GetPooledSession() const {
>+  CheckRunOnValidThreadAndInitialized();
>   if (pooled_sessions_.empty()) {
>     return nullptr;
>   }
>@@ -192,11 +229,21 @@ const PortAllocatorSession* PortAllocator::GetPooledSession() const {
> }
> 
> void PortAllocator::FreezeCandidatePool() {
>+  CheckRunOnValidThreadAndInitialized();
>   candidate_pool_frozen_ = true;
> }
> 
> void PortAllocator::DiscardCandidatePool() {
>+  CheckRunOnValidThreadIfInitialized();
>   pooled_sessions_.clear();
> }
> 
>+void PortAllocator::GetCandidateStatsFromPooledSessions(
>+    CandidateStatsList* candidate_stats_list) {
>+  CheckRunOnValidThreadAndInitialized();
>+  for (const auto& session : pooled_sessions()) {
>+    session->GetCandidateStatsFromReadyPorts(candidate_stats_list);
>+  }
>+}
>+
> }  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/portallocator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/portallocator.h
>index f8e813f502a..355b3212f0a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/portallocator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/portallocator.h
>@@ -20,13 +20,14 @@
> #include "p2p/base/portinterface.h"
> #include "rtc_base/helpers.h"
> #include "rtc_base/proxyinfo.h"
>-#include "rtc_base/sigslot.h"
>+#include "rtc_base/sslcertificate.h"
>+#include "rtc_base/third_party/sigslot/sigslot.h"
> #include "rtc_base/thread.h"
>+#include "rtc_base/thread_checker.h"
> 
> namespace webrtc {
>-class MetricsObserverInterface;
> class TurnCustomizer;
>-}
>+}  // namespace webrtc
> 
> namespace cricket {
> 
>@@ -46,12 +47,6 @@ enum {
>   // servers.
>   PORTALLOCATOR_DISABLE_TCP = 0x08,
>   PORTALLOCATOR_ENABLE_IPV6 = 0x40,
>-  // TODO(pthatcher): Remove this once it's no longer used in:
>-  // remoting/client/plugin/pepper_port_allocator.cc
>-  // remoting/protocol/chromium_port_allocator.cc
>-  // remoting/test/fake_port_allocator.cc
>-  // It's a no-op and is no longer needed.
>-  PORTALLOCATOR_ENABLE_SHARED_UFRAG = 0x80,
>   PORTALLOCATOR_ENABLE_SHARED_SOCKET = 0x100,
>   PORTALLOCATOR_ENABLE_STUN_RETRANSMIT_ATTRIBUTE = 0x200,
>   // When specified, we'll only allocate the STUN candidate for the public
>@@ -79,14 +74,9 @@ enum {
>   // When specified, do not collect IPv6 ICE candidates on Wi-Fi.
>   PORTALLOCATOR_ENABLE_IPV6_ON_WIFI = 0x4000,
> 
>-  // When this flag is set, ports not bound to any specific network interface
>-  // will be used, in addition to normal ports bound to the enumerated
>-  // interfaces. Without this flag, these "any address" ports would only be
>-  // used when network enumeration fails or is disabled. But under certain
>-  // conditions, these ports may succeed where others fail, so they may allow
>-  // the application to work in a wider variety of environments, at the expense
>-  // of having to allocate additional candidates.
>-  PORTALLOCATOR_ENABLE_ANY_ADDRESS_PORTS = 0x8000,
>+  // Exclude link-local network interfaces
>+  // from considertaion after adapter enumeration.
>+  PORTALLOCATOR_DISABLE_LINK_LOCAL_NETWORKS = 0x8000,
> };
> 
> // Defines various reasons that have caused ICE regathering.
>@@ -181,6 +171,7 @@ struct RelayServerConfig {
>   TlsCertPolicy tls_cert_policy = TlsCertPolicy::TLS_CERT_POLICY_SECURE;
>   std::vector<std::string> tls_alpn_protocols;
>   std::vector<std::string> tls_elliptic_curves;
>+  rtc::SSLCertificateVerifier* tls_cert_verifier = nullptr;
> };
> 
> class PortAllocatorSession : public sigslot::has_slots<> {
>@@ -240,7 +231,16 @@ class PortAllocatorSession : public sigslot::has_slots<> {
>   virtual void RegatherOnFailedNetworks() {}
>   // Re-gathers candidates on all networks.
>   virtual void RegatherOnAllNetworks() {}
>-
>+  // Get candidate-level stats from all candidates on the ready ports and return
>+  // the stats to the given list.
>+  virtual void GetCandidateStatsFromReadyPorts(
>+      CandidateStatsList* candidate_stats_list) const;
>+  // Set the interval at which STUN candidates will resend STUN binding requests
>+  // on the underlying ports to keep NAT bindings open.
>+  // The default value of the interval in implementation is restored if a null
>+  // optional value is passed.
>+  virtual void SetStunKeepaliveIntervalForReadyPorts(
>+      const absl::optional<int>& stun_keepalive_interval) {}
>   // Another way of getting the information provided by the signals below.
>   //
>   // Ports and candidates are not guaranteed to be in the same order as the
>@@ -259,8 +259,8 @@ class PortAllocatorSession : public sigslot::has_slots<> {
>   // ready(pairable).
>   sigslot::signal2<PortAllocatorSession*, const std::vector<PortInterface*>&>
>       SignalPortsPruned;
>-  sigslot::signal2<PortAllocatorSession*,
>-                   const std::vector<Candidate>&> SignalCandidatesReady;
>+  sigslot::signal2<PortAllocatorSession*, const std::vector<Candidate>&>
>+      SignalCandidatesReady;
>   // Candidates should be signaled to be removed when the port that generated
>   // the candidates is removed.
>   sigslot::signal2<PortAllocatorSession*, const std::vector<Candidate>&>
>@@ -313,19 +313,18 @@ class PortAllocatorSession : public sigslot::has_slots<> {
> };
> 
> // Every method of PortAllocator (including the destructor) must be called on
>-// the same thread, except for the constructor which may be called on any
>-// thread.
>+// the same thread after Initialize is called.
> //
>-// This allows constructing a PortAllocator subclass on one thread and
>-// passing it into an object that uses it on a different thread.
>+// This allows a PortAllocator subclass to be constructed and configured on one
>+// thread, and passed into an object that uses it on a different thread.
> class PortAllocator : public sigslot::has_slots<> {
>  public:
>   PortAllocator();
>   ~PortAllocator() override;
> 
>-  // This should be called on the PortAllocator's thread before the
>-  // PortAllocator is used. Subclasses may override this if necessary.
>-  virtual void Initialize() {}
>+  // This MUST be called on the PortAllocator's thread after finishing
>+  // constructing and configuring the PortAllocator subclasses.
>+  virtual void Initialize();
> 
>   // Set STUN and TURN servers to be used in future sessions, and set
>   // candidate pool size, as described in JSEP.
>@@ -343,15 +342,29 @@ class PortAllocator : public sigslot::has_slots<> {
>                         const std::vector<RelayServerConfig>& turn_servers,
>                         int candidate_pool_size,
>                         bool prune_turn_ports,
>-                        webrtc::TurnCustomizer* turn_customizer = nullptr);
>+                        webrtc::TurnCustomizer* turn_customizer = nullptr,
>+                        const absl::optional<int>&
>+                            stun_candidate_keepalive_interval = absl::nullopt);
> 
>-  const ServerAddresses& stun_servers() const { return stun_servers_; }
>+  const ServerAddresses& stun_servers() const {
>+    CheckRunOnValidThreadIfInitialized();
>+    return stun_servers_;
>+  }
> 
>   const std::vector<RelayServerConfig>& turn_servers() const {
>+    CheckRunOnValidThreadIfInitialized();
>     return turn_servers_;
>   }
> 
>-  int candidate_pool_size() const { return candidate_pool_size_; }
>+  int candidate_pool_size() const {
>+    CheckRunOnValidThreadIfInitialized();
>+    return candidate_pool_size_;
>+  }
>+
>+  const absl::optional<int>& stun_candidate_keepalive_interval() const {
>+    CheckRunOnValidThreadIfInitialized();
>+    return stun_candidate_keepalive_interval_;
>+  }
> 
>   // Sets the network types to ignore.
>   // Values are defined by the AdapterType enum.
>@@ -392,23 +405,48 @@ class PortAllocator : public sigslot::has_slots<> {
>   // Discard any remaining pooled sessions.
>   void DiscardCandidatePool();
> 
>-  uint32_t flags() const { return flags_; }
>-  void set_flags(uint32_t flags) { flags_ = flags; }
>+  uint32_t flags() const {
>+    CheckRunOnValidThreadIfInitialized();
>+    return flags_;
>+  }
>+
>+  void set_flags(uint32_t flags) {
>+    CheckRunOnValidThreadIfInitialized();
>+    flags_ = flags;
>+  }
> 
>   // These three methods are deprecated. If connections need to go through a
>   // proxy, the application should create a BasicPortAllocator given a custom
>   // PacketSocketFactory that creates proxy sockets.
>-  const std::string& user_agent() const { return agent_; }
>-  const rtc::ProxyInfo& proxy() const { return proxy_; }
>+  const std::string& user_agent() const {
>+    CheckRunOnValidThreadIfInitialized();
>+    return agent_;
>+  }
>+
>+  const rtc::ProxyInfo& proxy() const {
>+    CheckRunOnValidThreadIfInitialized();
>+    return proxy_;
>+  }
>+
>   void set_proxy(const std::string& agent, const rtc::ProxyInfo& proxy) {
>+    CheckRunOnValidThreadIfInitialized();
>     agent_ = agent;
>     proxy_ = proxy;
>   }
> 
>   // Gets/Sets the port range to use when choosing client ports.
>-  int min_port() const { return min_port_; }
>-  int max_port() const { return max_port_; }
>+  int min_port() const {
>+    CheckRunOnValidThreadIfInitialized();
>+    return min_port_;
>+  }
>+
>+  int max_port() const {
>+    CheckRunOnValidThreadIfInitialized();
>+    return max_port_;
>+  }
>+
>   bool SetPortRange(int min_port, int max_port) {
>+    CheckRunOnValidThreadIfInitialized();
>     if (min_port > max_port) {
>       return false;
>     }
>@@ -426,8 +464,15 @@ class PortAllocator : public sigslot::has_slots<> {
>   // ICE down. We should work on making our ICE logic smarter (for example,
>   // prioritizing pinging connections that are most likely to work) so that
>   // every network interface can be used without impacting ICE's speed.
>-  void set_max_ipv6_networks(int networks) { max_ipv6_networks_ = networks; }
>-  int max_ipv6_networks() { return max_ipv6_networks_; }
>+  void set_max_ipv6_networks(int networks) {
>+    CheckRunOnValidThreadIfInitialized();
>+    max_ipv6_networks_ = networks;
>+  }
>+
>+  int max_ipv6_networks() {
>+    CheckRunOnValidThreadIfInitialized();
>+    return max_ipv6_networks_;
>+  }
> 
>   // Delay between different candidate gathering phases (UDP, TURN, TCP).
>   // Defaults to 1 second, but PeerConnection sets it to 50ms.
>@@ -435,33 +480,65 @@ class PortAllocator : public sigslot::has_slots<> {
>   // STUN transactions at once, but that's already happening if you configure
>   // multiple STUN servers or have multiple network interfaces. We should
>   // implement some global pacing logic instead if that's our goal.
>-  uint32_t step_delay() const { return step_delay_; }
>-  void set_step_delay(uint32_t delay) { step_delay_ = delay; }
>+  uint32_t step_delay() const {
>+    CheckRunOnValidThreadIfInitialized();
>+    return step_delay_;
>+  }
>+
>+  void set_step_delay(uint32_t delay) {
>+    CheckRunOnValidThreadIfInitialized();
>+    step_delay_ = delay;
>+  }
>+
>+  bool allow_tcp_listen() const {
>+    CheckRunOnValidThreadIfInitialized();
>+    return allow_tcp_listen_;
>+  }
> 
>-  bool allow_tcp_listen() const { return allow_tcp_listen_; }
>   void set_allow_tcp_listen(bool allow_tcp_listen) {
>+    CheckRunOnValidThreadIfInitialized();
>     allow_tcp_listen_ = allow_tcp_listen;
>   }
> 
>-  uint32_t candidate_filter() { return candidate_filter_; }
>+  uint32_t candidate_filter() {
>+    CheckRunOnValidThreadIfInitialized();
>+    return candidate_filter_;
>+  }
>+
>   void set_candidate_filter(uint32_t filter) {
>+    CheckRunOnValidThreadIfInitialized();
>     candidate_filter_ = filter;
>   }
> 
>-  bool prune_turn_ports() const { return prune_turn_ports_; }
>+  bool prune_turn_ports() const {
>+    CheckRunOnValidThreadIfInitialized();
>+    return prune_turn_ports_;
>+  }
> 
>   // Gets/Sets the Origin value used for WebRTC STUN requests.
>-  const std::string& origin() const { return origin_; }
>-  void set_origin(const std::string& origin) { origin_ = origin; }
>+  const std::string& origin() const {
>+    CheckRunOnValidThreadIfInitialized();
>+    return origin_;
>+  }
> 
>-  void SetMetricsObserver(webrtc::MetricsObserverInterface* observer) {
>-    metrics_observer_ = observer;
>+  void set_origin(const std::string& origin) {
>+    CheckRunOnValidThreadIfInitialized();
>+    origin_ = origin;
>   }
> 
>   webrtc::TurnCustomizer* turn_customizer() {
>+    CheckRunOnValidThreadIfInitialized();
>     return turn_customizer_;
>   }
> 
>+  // Collect candidate stats from pooled allocator sessions. This can be used to
>+  // collect candidate stats without creating an offer/answer or setting local
>+  // description. After the local description is set, the ownership of the
>+  // pooled session is taken by P2PTransportChannel, and the
>+  // candidate stats can be collected from P2PTransportChannel::GetStats.
>+  virtual void GetCandidateStatsFromPooledSessions(
>+      CandidateStatsList* candidate_stats_list);
>+
>  protected:
>   virtual PortAllocatorSession* CreateSessionInternal(
>       const std::string& content_name,
>@@ -469,14 +546,21 @@ class PortAllocator : public sigslot::has_slots<> {
>       const std::string& ice_ufrag,
>       const std::string& ice_pwd) = 0;
> 
>-  webrtc::MetricsObserverInterface* metrics_observer() {
>-    return metrics_observer_;
>-  }
>-
>   const std::deque<std::unique_ptr<PortAllocatorSession>>& pooled_sessions() {
>     return pooled_sessions_;
>   }
> 
>+  // The following thread checks are only done in DCHECK for the consistency
>+  // with the exsiting thread checks.
>+  void CheckRunOnValidThreadIfInitialized() const {
>+    RTC_DCHECK(!initialized_ || thread_checker_.CalledOnValidThread());
>+  }
>+
>+  void CheckRunOnValidThreadAndInitialized() const {
>+    RTC_DCHECK(initialized_ && thread_checker_.CalledOnValidThread());
>+  }
>+
>+  bool initialized_ = false;
>   uint32_t flags_;
>   std::string agent_;
>   rtc::ProxyInfo proxy_;
>@@ -487,6 +571,7 @@ class PortAllocator : public sigslot::has_slots<> {
>   bool allow_tcp_listen_;
>   uint32_t candidate_filter_;
>   std::string origin_;
>+  rtc::ThreadChecker thread_checker_;
> 
>  private:
>   ServerAddresses stun_servers_;
>@@ -496,12 +581,12 @@ class PortAllocator : public sigslot::has_slots<> {
>   bool candidate_pool_frozen_ = false;
>   bool prune_turn_ports_ = false;
> 
>-  webrtc::MetricsObserverInterface* metrics_observer_ = nullptr;
>-
>   // Customizer for TURN messages.
>   // The instance is owned by application and will be shared among
>   // all TurnPort(s) created.
>   webrtc::TurnCustomizer* turn_customizer_ = nullptr;
>+
>+  absl::optional<int> stun_candidate_keepalive_interval_;
> };
> 
> }  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/portinterface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/portinterface.h
>index e34e39abc34..60fa4444b3d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/portinterface.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/portinterface.h
>@@ -14,6 +14,7 @@
> #include <string>
> #include <vector>
> 
>+#include "absl/types/optional.h"
> #include "api/candidate.h"
> #include "p2p/base/transportdescription.h"
> #include "rtc_base/asyncpacketsocket.h"
>@@ -22,12 +23,13 @@
> namespace rtc {
> class Network;
> struct PacketOptions;
>-}
>+}  // namespace rtc
> 
> namespace cricket {
> class Connection;
> class IceMessage;
> class StunMessage;
>+class StunStats;
> 
> enum ProtocolType {
>   PROTO_UDP,
>@@ -66,13 +68,12 @@ class PortInterface {
>   virtual void PrepareAddress() = 0;
> 
>   // Returns the connection to the given address or NULL if none exists.
>-  virtual Connection* GetConnection(
>-      const rtc::SocketAddress& remote_addr) = 0;
>+  virtual Connection* GetConnection(const rtc::SocketAddress& remote_addr) = 0;
> 
>   // Creates a new connection to the given address.
>   enum CandidateOrigin { ORIGIN_THIS_PORT, ORIGIN_OTHER_PORT, ORIGIN_MESSAGE };
>-  virtual Connection* CreateConnection(
>-      const Candidate& remote_candidate, CandidateOrigin origin) = 0;
>+  virtual Connection* CreateConnection(const Candidate& remote_candidate,
>+                                       CandidateOrigin origin) = 0;
> 
>   // Functions on the underlying socket(s).
>   virtual int SetOption(rtc::Socket::Option opt, int value) = 0;
>@@ -85,25 +86,32 @@ class PortInterface {
> 
>   // Sends the given packet to the given address, provided that the address is
>   // that of a connection or an address that has sent to us already.
>-  virtual int SendTo(const void* data, size_t size,
>+  virtual int SendTo(const void* data,
>+                     size_t size,
>                      const rtc::SocketAddress& addr,
>-                     const rtc::PacketOptions& options, bool payload) = 0;
>+                     const rtc::PacketOptions& options,
>+                     bool payload) = 0;
> 
>   // Indicates that we received a successful STUN binding request from an
>   // address that doesn't correspond to any current connection.  To turn this
>   // into a real connection, call CreateConnection.
>-  sigslot::signal6<PortInterface*, const rtc::SocketAddress&,
>-                   ProtocolType, IceMessage*, const std::string&,
>-                   bool> SignalUnknownAddress;
>+  sigslot::signal6<PortInterface*,
>+                   const rtc::SocketAddress&,
>+                   ProtocolType,
>+                   IceMessage*,
>+                   const std::string&,
>+                   bool>
>+      SignalUnknownAddress;
> 
>   // Sends a response message (normal or error) to the given request.  One of
>   // these methods should be called as a response to SignalUnknownAddress.
>   // NOTE: You MUST call CreateConnection BEFORE SendBindingResponse.
>   virtual void SendBindingResponse(StunMessage* request,
>                                    const rtc::SocketAddress& addr) = 0;
>-  virtual void SendBindingErrorResponse(
>-      StunMessage* request, const rtc::SocketAddress& addr,
>-      int error_code, const std::string& reason) = 0;
>+  virtual void SendBindingErrorResponse(StunMessage* request,
>+                                        const rtc::SocketAddress& addr,
>+                                        int error_code,
>+                                        const std::string& reason) = 0;
> 
>   // Signaled when this port decides to delete itself because it no longer has
>   // any usefulness.
>@@ -117,14 +125,17 @@ class PortInterface {
>   // through their respective connection and instead delivers every packet
>   // through this port.
>   virtual void EnablePortPackets() = 0;
>-  sigslot::signal4<PortInterface*, const char*, size_t,
>-                   const rtc::SocketAddress&> SignalReadPacket;
>+  sigslot::
>+      signal4<PortInterface*, const char*, size_t, const rtc::SocketAddress&>
>+          SignalReadPacket;
> 
>   // Emitted each time a packet is sent on this port.
>   sigslot::signal1<const rtc::SentPacket&> SignalSentPacket;
> 
>   virtual std::string ToString() const = 0;
> 
>+  virtual void GetStunStats(absl::optional<StunStats>* stats) = 0;
>+
>  protected:
>   PortInterface();
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/pseudotcp.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/pseudotcp.cc
>index e15ec4056a6..42a4f0cee46 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/pseudotcp.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/pseudotcp.cc
>@@ -18,7 +18,6 @@
> #include <set>
> 
> #include "rtc_base/arraysize.h"
>-#include "rtc_base/basictypes.h"
> #include "rtc_base/bytebuffer.h"
> #include "rtc_base/byteorder.h"
> #include "rtc_base/checks.h"
>@@ -900,9 +899,28 @@ bool PseudoTcp::process(Segment& seg) {
>   bool bNewData = false;
> 
>   if (seg.len > 0) {
>+    bool bRecover = false;
>     if (bIgnoreData) {
>       if (seg.seq == m_rcv_nxt) {
>         m_rcv_nxt += seg.len;
>+        // If we received a data segment out of order relative to a control
>+        // segment, then we wrote it into the receive buffer at an offset (see
>+        // "WriteOffset") below. So we need to advance the position in the
>+        // buffer to avoid corrupting data. See bugs.webrtc.org/9208
>+        //
>+        // We advance the position in the buffer by N bytes by acting like we
>+        // wrote N bytes and then immediately read them. We can only do this if
>+        // there's not already data ready to read, but this should always be
>+        // true in the problematic scenario, since control frames are always
>+        // sent first in the stream.
>+        size_t rcv_buffered;
>+        if (m_rbuf.GetBuffered(&rcv_buffered) && rcv_buffered == 0) {
>+          m_rbuf.ConsumeWriteBuffer(seg.len);
>+          m_rbuf.ConsumeReadData(seg.len);
>+          // After shifting the position in the buffer, we may have
>+          // out-of-order packets ready to be recovered.
>+          bRecover = true;
>+        }
>       }
>     } else {
>       uint32_t nOffset = seg.seq - m_rcv_nxt;
>@@ -921,23 +939,9 @@ bool PseudoTcp::process(Segment& seg) {
>         m_rcv_nxt += seg.len;
>         m_rcv_wnd -= seg.len;
>         bNewData = true;
>-
>-        RList::iterator it = m_rlist.begin();
>-        while ((it != m_rlist.end()) && (it->seq <= m_rcv_nxt)) {
>-          if (it->seq + it->len > m_rcv_nxt) {
>-            sflags = sfImmediateAck;  // (Fast Recovery)
>-            uint32_t nAdjust = (it->seq + it->len) - m_rcv_nxt;
>-#if _DEBUGMSG >= _DBG_NORMAL
>-            RTC_LOG(LS_INFO)
>-                << "Recovered " << nAdjust << " bytes (" << m_rcv_nxt << " -> "
>-                << m_rcv_nxt + nAdjust << ")";
>-#endif  // _DEBUGMSG
>-            m_rbuf.ConsumeWriteBuffer(nAdjust);
>-            m_rcv_nxt += nAdjust;
>-            m_rcv_wnd -= nAdjust;
>-          }
>-          it = m_rlist.erase(it);
>-        }
>+        // May be able to recover packets previously received out-of-order
>+        // now.
>+        bRecover = true;
>       } else {
> #if _DEBUGMSG >= _DBG_NORMAL
>         RTC_LOG(LS_INFO) << "Saving " << seg.len << " bytes (" << seg.seq
>@@ -953,6 +957,24 @@ bool PseudoTcp::process(Segment& seg) {
>         m_rlist.insert(it, rseg);
>       }
>     }
>+    if (bRecover) {
>+      RList::iterator it = m_rlist.begin();
>+      while ((it != m_rlist.end()) && (it->seq <= m_rcv_nxt)) {
>+        if (it->seq + it->len > m_rcv_nxt) {
>+          sflags = sfImmediateAck;  // (Fast Recovery)
>+          uint32_t nAdjust = (it->seq + it->len) - m_rcv_nxt;
>+#if _DEBUGMSG >= _DBG_NORMAL
>+          RTC_LOG(LS_INFO) << "Recovered " << nAdjust << " bytes (" << m_rcv_nxt
>+                           << " -> " << m_rcv_nxt + nAdjust << ")";
>+#endif  // _DEBUGMSG
>+          m_rbuf.ConsumeWriteBuffer(nAdjust);
>+          m_rcv_nxt += nAdjust;
>+          m_rcv_wnd -= nAdjust;
>+          bNewData = true;
>+        }
>+        it = m_rlist.erase(it);
>+      }
>+    }
>   }
> 
>   attemptSend(sflags);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/pseudotcp.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/pseudotcp.h
>index b36a7d9ddb7..c363ef64b74 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/pseudotcp.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/pseudotcp.h
>@@ -13,7 +13,6 @@
> 
> #include <list>
> 
>-#include "rtc_base/basictypes.h"
> #include "rtc_base/stream.h"
> 
> namespace cricket {
>@@ -35,7 +34,8 @@ class IPseudoTcpNotify {
>   // Write the packet onto the network
>   enum WriteResult { WR_SUCCESS, WR_TOO_LARGE, WR_FAIL };
>   virtual WriteResult TcpWritePacket(PseudoTcp* tcp,
>-                                     const char* buffer, size_t len) = 0;
>+                                     const char* buffer,
>+                                     size_t len) = 0;
> 
>  protected:
>   virtual ~IPseudoTcpNotify() {}
>@@ -59,7 +59,11 @@ class PseudoTcp {
>   int GetError();
> 
>   enum TcpState {
>-    TCP_LISTEN, TCP_SYN_SENT, TCP_SYN_RECEIVED, TCP_ESTABLISHED, TCP_CLOSED
>+    TCP_LISTEN,
>+    TCP_SYN_SENT,
>+    TCP_SYN_RECEIVED,
>+    TCP_ESTABLISHED,
>+    TCP_CLOSED
>   };
>   TcpState State() const { return m_state; }
> 
>@@ -72,7 +76,7 @@ class PseudoTcp {
> 
>   // Call this whenever a packet arrives.
>   // Returns true if the packet was processed successfully.
>-  bool NotifyPacket(const char * buffer, size_t len);
>+  bool NotifyPacket(const char* buffer, size_t len);
> 
>   // Call this to determine the next time NotifyClock should be called.
>   // Returns false if the socket is ready to be destroyed.
>@@ -85,10 +89,10 @@ class PseudoTcp {
>   // Setting options for OPT_RCVBUF or OPT_SNDBUF after Connect() is called
>   // will result in an assertion.
>   enum Option {
>-    OPT_NODELAY,      // Whether to enable Nagle's algorithm (0 == off)
>-    OPT_ACKDELAY,     // The Delayed ACK timeout (0 == off).
>-    OPT_RCVBUF,       // Set the receive buffer size, in bytes.
>-    OPT_SNDBUF,       // Set the send buffer size, in bytes.
>+    OPT_NODELAY,   // Whether to enable Nagle's algorithm (0 == off)
>+    OPT_ACKDELAY,  // The Delayed ACK timeout (0 == off).
>+    OPT_RCVBUF,    // Set the receive buffer size, in bytes.
>+    OPT_SNDBUF,    // Set the send buffer size, in bytes.
>   };
>   void GetOption(Option opt, int* value);
>   void SetOption(Option opt, int value);
>@@ -114,7 +118,7 @@ class PseudoTcp {
>     uint32_t conv, seq, ack;
>     uint8_t flags;
>     uint16_t wnd;
>-    const char * data;
>+    const char* data;
>     uint32_t len;
>     uint32_t tsval, tsecr;
>   };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/pseudotcp_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/pseudotcp_unittest.cc
>index fb9a5034550..dd86f921d52 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/pseudotcp_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/pseudotcp_unittest.cc
>@@ -49,13 +49,18 @@ class PseudoTcpTestBase : public testing::Test,
>         remote_mtu_(65535),
>         delay_(0),
>         loss_(0) {
>-    // Set use of the test RNG to get predictable loss patterns.
>+    // Set use of the test RNG to get predictable loss patterns. Otherwise,
>+    // this test would occasionally get really unlucky loss and time out.
>     rtc::SetRandomTestMode(true);
>   }
>   ~PseudoTcpTestBase() {
>     // Put it back for the next test.
>     rtc::SetRandomTestMode(false);
>   }
>+  // If true, both endpoints will send the "connect" segment simultaneously,
>+  // rather than |local_| sending it followed by a response from |remote_|.
>+  // Note that this is what chromoting ends up doing.
>+  void SetSimultaneousOpen(bool enabled) { simultaneous_open_ = enabled; }
>   void SetLocalMtu(int mtu) {
>     local_.NotifyMTU(mtu);
>     local_mtu_ = mtu;
>@@ -66,6 +71,9 @@ class PseudoTcpTestBase : public testing::Test,
>   }
>   void SetDelay(int delay) { delay_ = delay; }
>   void SetLoss(int percent) { loss_ = percent; }
>+  // Used to cause the initial "connect" segment to be lost, needed for a
>+  // regression test.
>+  void DropNextPacket() { drop_next_packet_ = true; }
>   void SetOptNagling(bool enable_nagles) {
>     local_.SetOption(PseudoTcp::OPT_NODELAY, !enable_nagles);
>     remote_.SetOption(PseudoTcp::OPT_NODELAY, !enable_nagles);
>@@ -93,6 +101,12 @@ class PseudoTcpTestBase : public testing::Test,
>     if (ret == 0) {
>       UpdateLocalClock();
>     }
>+    if (simultaneous_open_) {
>+      ret = remote_.Connect();
>+      if (ret == 0) {
>+        UpdateRemoteClock();
>+      }
>+    }
>     return ret;
>   }
>   void Close() {
>@@ -134,19 +148,28 @@ class PseudoTcpTestBase : public testing::Test,
>   virtual WriteResult TcpWritePacket(PseudoTcp* tcp,
>                                      const char* buffer,
>                                      size_t len) {
>+    // Drop a packet if the test called DropNextPacket.
>+    if (drop_next_packet_) {
>+      drop_next_packet_ = false;
>+      RTC_LOG(LS_VERBOSE) << "Dropping packet due to DropNextPacket, size="
>+                          << len;
>+      return WR_SUCCESS;
>+    }
>     // Randomly drop the desired percentage of packets.
>-    // Also drop packets that are larger than the configured MTU.
>     if (rtc::CreateRandomId() % 100 < static_cast<uint32_t>(loss_)) {
>       RTC_LOG(LS_VERBOSE) << "Randomly dropping packet, size=" << len;
>-    } else if (len > static_cast<size_t>(std::min(local_mtu_, remote_mtu_))) {
>+      return WR_SUCCESS;
>+    }
>+    // Also drop packets that are larger than the configured MTU.
>+    if (len > static_cast<size_t>(std::min(local_mtu_, remote_mtu_))) {
>       RTC_LOG(LS_VERBOSE) << "Dropping packet that exceeds path MTU, size="
>                           << len;
>-    } else {
>-      int id = (tcp == &local_) ? MSG_RPACKET : MSG_LPACKET;
>-      std::string packet(buffer, len);
>-      rtc::Thread::Current()->PostDelayed(RTC_FROM_HERE, delay_, this, id,
>-                                          rtc::WrapMessageData(packet));
>+      return WR_SUCCESS;
>     }
>+    int id = (tcp == &local_) ? MSG_RPACKET : MSG_LPACKET;
>+    std::string packet(buffer, len);
>+    rtc::Thread::Current()->PostDelayed(RTC_FROM_HERE, delay_, this, id,
>+                                        rtc::WrapMessageData(packet));
>     return WR_SUCCESS;
>   }
> 
>@@ -198,6 +221,8 @@ class PseudoTcpTestBase : public testing::Test,
>   int remote_mtu_;
>   int delay_;
>   int loss_;
>+  bool drop_next_packet_ = false;
>+  bool simultaneous_open_ = false;
> };
> 
> class PseudoTcpTest : public PseudoTcpTestBase {
>@@ -554,8 +579,8 @@ class PseudoTcpTestReceiveWindow : public PseudoTcpTestBase {
>     } else {
>       if (!remote_.isReceiveBufferFull()) {
>         RTC_LOG(LS_ERROR) << "This shouldn't happen - the send buffer is full, "
>-                          << "the receive buffer is not, and there are no "
>-                          << "remaining messages to process.";
>+                             "the receive buffer is not, and there are no "
>+                             "remaining messages to process.";
>       }
>       send_stream_.GetPosition(&position);
>       send_position_.push_back(position);
>@@ -620,6 +645,27 @@ TEST_F(PseudoTcpTest, TestSendWithLossAndOptNaglingOff) {
>   TestTransfer(100000);  // less data so test runs faster
> }
> 
>+// Regression test for bugs.webrtc.org/9208.
>+//
>+// This bug resulted in corrupted data if a "connect" segment was received after
>+// a data segment. This is only possible if:
>+//
>+// * The initial "connect" segment is lost, and retransmitted later.
>+// * Both sides send "connect"s simultaneously, such that the local side thinks
>+//   a connection is established even before its "connect" has been
>+//   acknowledged.
>+// * Nagle algorithm disabled, allowing a data segment to be sent before the
>+//   "connect" has been acknowledged.
>+TEST_F(PseudoTcpTest,
>+       TestSendWhenFirstPacketLostWithOptNaglingOffAndSimultaneousOpen) {
>+  SetLocalMtu(1500);
>+  SetRemoteMtu(1500);
>+  DropNextPacket();
>+  SetOptNagling(false);
>+  SetSimultaneousOpen(true);
>+  TestTransfer(10000);
>+}
>+
> // Test sending data with 10% packet loss and Delayed ACK disabled.
> // Transmission should be slightly faster than with it enabled.
> TEST_F(PseudoTcpTest, TestSendWithLossAndOptAckDelayOff) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/regatheringcontroller.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/regatheringcontroller.cc
>new file mode 100644
>index 00000000000..6d4c4fd978c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/regatheringcontroller.cc
>@@ -0,0 +1,157 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "p2p/base/regatheringcontroller.h"
>+
>+namespace webrtc {
>+
>+using Config = BasicRegatheringController::Config;
>+
>+Config::Config(const absl::optional<rtc::IntervalRange>&
>+                   regather_on_all_networks_interval_range,
>+               int regather_on_failed_networks_interval)
>+    : regather_on_all_networks_interval_range(
>+          regather_on_all_networks_interval_range),
>+      regather_on_failed_networks_interval(
>+          regather_on_failed_networks_interval) {}
>+
>+Config::Config(const Config& other) = default;
>+
>+Config::~Config() = default;
>+Config& Config::operator=(const Config& other) = default;
>+
>+BasicRegatheringController::BasicRegatheringController(
>+    const Config& config,
>+    cricket::IceTransportInternal* ice_transport,
>+    rtc::Thread* thread)
>+    : config_(config),
>+      ice_transport_(ice_transport),
>+      thread_(thread),
>+      rand_(rtc::SystemTimeNanos()) {
>+  RTC_DCHECK(ice_transport_);
>+  RTC_DCHECK(thread_);
>+  ice_transport_->SignalStateChanged.connect(
>+      this, &BasicRegatheringController::OnIceTransportStateChanged);
>+  ice_transport->SignalWritableState.connect(
>+      this, &BasicRegatheringController::OnIceTransportWritableState);
>+  ice_transport->SignalReceivingState.connect(
>+      this, &BasicRegatheringController::OnIceTransportReceivingState);
>+  ice_transport->SignalNetworkRouteChanged.connect(
>+      this, &BasicRegatheringController::OnIceTransportNetworkRouteChanged);
>+}
>+
>+BasicRegatheringController::~BasicRegatheringController() = default;
>+
>+void BasicRegatheringController::Start() {
>+  ScheduleRecurringRegatheringOnFailedNetworks();
>+  if (config_.regather_on_all_networks_interval_range) {
>+    ScheduleRecurringRegatheringOnAllNetworks();
>+  }
>+}
>+
>+void BasicRegatheringController::SetConfig(const Config& config) {
>+  bool need_cancel_on_all_networks =
>+      has_recurring_schedule_on_all_networks_ &&
>+      (config_.regather_on_all_networks_interval_range !=
>+       config.regather_on_all_networks_interval_range);
>+  bool need_reschedule_on_all_networks =
>+      config.regather_on_all_networks_interval_range &&
>+      (config_.regather_on_all_networks_interval_range !=
>+       config.regather_on_all_networks_interval_range);
>+  bool need_cancel_and_reschedule_on_failed_networks =
>+      has_recurring_schedule_on_failed_networks_ &&
>+      (config_.regather_on_failed_networks_interval !=
>+       config.regather_on_failed_networks_interval);
>+  config_ = config;
>+  if (need_cancel_on_all_networks) {
>+    CancelScheduledRecurringRegatheringOnAllNetworks();
>+  }
>+  if (need_reschedule_on_all_networks) {
>+    ScheduleRecurringRegatheringOnAllNetworks();
>+  }
>+  if (need_cancel_and_reschedule_on_failed_networks) {
>+    CancelScheduledRecurringRegatheringOnFailedNetworks();
>+    ScheduleRecurringRegatheringOnFailedNetworks();
>+  }
>+}
>+
>+void BasicRegatheringController::ScheduleRecurringRegatheringOnAllNetworks() {
>+  RTC_DCHECK(config_.regather_on_all_networks_interval_range &&
>+             config_.regather_on_all_networks_interval_range.value().min() >=
>+                 0);
>+  int delay_ms = SampleRegatherAllNetworksInterval(
>+      config_.regather_on_all_networks_interval_range.value());
>+  CancelScheduledRecurringRegatheringOnAllNetworks();
>+  has_recurring_schedule_on_all_networks_ = true;
>+  invoker_for_all_networks_.AsyncInvokeDelayed<void>(
>+      RTC_FROM_HERE, thread(),
>+      rtc::Bind(
>+          &BasicRegatheringController::RegatherOnAllNetworksIfDoneGathering,
>+          this, true),
>+      delay_ms);
>+}
>+
>+void BasicRegatheringController::RegatherOnAllNetworksIfDoneGathering(
>+    bool repeated) {
>+  // Only regather when the current session is in the CLEARED state (i.e., not
>+  // running or stopped). It is only possible to enter this state when we gather
>+  // continually, so there is an implicit check on continual gathering here.
>+  if (allocator_session_ && allocator_session_->IsCleared()) {
>+    allocator_session_->RegatherOnAllNetworks();
>+  }
>+  if (repeated) {
>+    ScheduleRecurringRegatheringOnAllNetworks();
>+  }
>+}
>+
>+void BasicRegatheringController::
>+    ScheduleRecurringRegatheringOnFailedNetworks() {
>+  RTC_DCHECK(config_.regather_on_failed_networks_interval >= 0);
>+  CancelScheduledRecurringRegatheringOnFailedNetworks();
>+  has_recurring_schedule_on_failed_networks_ = true;
>+  invoker_for_failed_networks_.AsyncInvokeDelayed<void>(
>+      RTC_FROM_HERE, thread(),
>+      rtc::Bind(
>+          &BasicRegatheringController::RegatherOnFailedNetworksIfDoneGathering,
>+          this, true),
>+      config_.regather_on_failed_networks_interval);
>+}
>+
>+void BasicRegatheringController::RegatherOnFailedNetworksIfDoneGathering(
>+    bool repeated) {
>+  // Only regather when the current session is in the CLEARED state (i.e., not
>+  // running or stopped). It is only possible to enter this state when we gather
>+  // continually, so there is an implicit check on continual gathering here.
>+  if (allocator_session_ && allocator_session_->IsCleared()) {
>+    allocator_session_->RegatherOnFailedNetworks();
>+  }
>+  if (repeated) {
>+    ScheduleRecurringRegatheringOnFailedNetworks();
>+  }
>+}
>+
>+void BasicRegatheringController::
>+    CancelScheduledRecurringRegatheringOnAllNetworks() {
>+  invoker_for_all_networks_.Clear();
>+  has_recurring_schedule_on_all_networks_ = false;
>+}
>+
>+void BasicRegatheringController::
>+    CancelScheduledRecurringRegatheringOnFailedNetworks() {
>+  invoker_for_failed_networks_.Clear();
>+  has_recurring_schedule_on_failed_networks_ = false;
>+}
>+
>+int BasicRegatheringController::SampleRegatherAllNetworksInterval(
>+    const rtc::IntervalRange& range) {
>+  return rand_.Rand(range.min(), range.max());
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/regatheringcontroller.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/regatheringcontroller.h
>new file mode 100644
>index 00000000000..891ec85b88f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/regatheringcontroller.h
>@@ -0,0 +1,124 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef P2P_BASE_REGATHERINGCONTROLLER_H_
>+#define P2P_BASE_REGATHERINGCONTROLLER_H_
>+
>+#include "p2p/base/icetransportinternal.h"
>+#include "p2p/base/portallocator.h"
>+#include "rtc_base/asyncinvoker.h"
>+#include "rtc_base/random.h"
>+#include "rtc_base/thread.h"
>+
>+namespace webrtc {
>+
>+// Controls regathering of candidates for the ICE transport passed into it,
>+// reacting to signals like SignalWritableState, SignalNetworkRouteChange, etc.,
>+// using methods like GetStats to get additional information, and calling
>+// methods like RegatherOnAllNetworks on the PortAllocatorSession when
>+// regathering is desired.
>+//
>+// TODO(qingsi): Add the description of behavior when autonomous regathering is
>+// implemented.
>+//
>+// "Regathering" is defined as gathering additional candidates within a single
>+// ICE generation (or in other words, PortAllocatorSession), and is possible
>+// when "continual gathering" is enabled. This may allow connectivity to be
>+// maintained and/or restored without a full ICE restart.
>+//
>+// Regathering will only begin after PortAllocationSession is set via
>+// set_allocator_session. This should be called any time the "active"
>+// PortAllocatorSession is changed (in other words, when an ICE restart occurs),
>+// so that candidates are gathered for the "current" ICE generation.
>+//
>+// All methods of BasicRegatheringController should be called on the same
>+// thread as the one passed to the constructor, and this thread should be the
>+// same one where PortAllocatorSession runs, which is also identical to the
>+// network thread of the ICE transport, as given by
>+// P2PTransportChannel::thread().
>+class BasicRegatheringController : public sigslot::has_slots<> {
>+ public:
>+  struct Config {
>+    Config(const absl::optional<rtc::IntervalRange>&
>+               regather_on_all_networks_interval_range,
>+           int regather_on_failed_networks_interval);
>+    Config(const Config& other);
>+    ~Config();
>+    Config& operator=(const Config& other);
>+    absl::optional<rtc::IntervalRange> regather_on_all_networks_interval_range;
>+    int regather_on_failed_networks_interval;
>+  };
>+
>+  BasicRegatheringController() = delete;
>+  BasicRegatheringController(const Config& config,
>+                             cricket::IceTransportInternal* ice_transport,
>+                             rtc::Thread* thread);
>+  ~BasicRegatheringController() override;
>+  // TODO(qingsi): Remove this method after implementing a new signal in
>+  // P2PTransportChannel and reacting to that signal for the initial schedules
>+  // of regathering.
>+  void Start();
>+  void set_allocator_session(cricket::PortAllocatorSession* allocator_session) {
>+    allocator_session_ = allocator_session;
>+  }
>+  // Setting a different config of the regathering interval range on all
>+  // networks cancels and reschedules the recurring schedules, if any, of
>+  // regathering on all networks. The same applies to the change of the
>+  // regathering interval on the failed networks. This rescheduling behavior is
>+  // seperately defined for the two config parameters.
>+  void SetConfig(const Config& config);
>+
>+ private:
>+  // TODO(qingsi): Implement the following methods and use methods from the ICE
>+  // transport like GetStats to get additional information for the decision
>+  // making in regathering.
>+  void OnIceTransportStateChanged(cricket::IceTransportInternal*) {}
>+  void OnIceTransportWritableState(rtc::PacketTransportInternal*) {}
>+  void OnIceTransportReceivingState(rtc::PacketTransportInternal*) {}
>+  void OnIceTransportNetworkRouteChanged(absl::optional<rtc::NetworkRoute>) {}
>+  // Schedules delayed and repeated regathering of local candidates on all
>+  // networks, where the delay in milliseconds is randomly sampled from the
>+  // range in the config. The delay of each repetition is independently sampled
>+  // from the same range. When scheduled, all previous schedules are canceled.
>+  void ScheduleRecurringRegatheringOnAllNetworks();
>+  // Schedules delayed and repeated regathering of local candidates on failed
>+  // networks, where the delay in milliseconds is given by the config. Each
>+  // repetition is separated by the same delay. When scheduled, all previous
>+  // schedules are canceled.
>+  void ScheduleRecurringRegatheringOnFailedNetworks();
>+  // Cancels regathering scheduled by ScheduleRecurringRegatheringOnAllNetworks.
>+  void CancelScheduledRecurringRegatheringOnAllNetworks();
>+  // Cancels regathering scheduled by
>+  // ScheduleRecurringRegatheringOnFailedNetworks.
>+  void CancelScheduledRecurringRegatheringOnFailedNetworks();
>+
>+  rtc::Thread* thread() const { return thread_; }
>+  // The following two methods perform the actual regathering, if the recent
>+  // port allocator session has done the initial gathering.
>+  void RegatherOnAllNetworksIfDoneGathering(bool repeated);
>+  void RegatherOnFailedNetworksIfDoneGathering(bool repeated);
>+  // Samples a delay from the uniform distribution in the given range.
>+  int SampleRegatherAllNetworksInterval(const rtc::IntervalRange& range);
>+
>+  Config config_;
>+  cricket::IceTransportInternal* ice_transport_;
>+  cricket::PortAllocatorSession* allocator_session_ = nullptr;
>+  bool has_recurring_schedule_on_all_networks_ = false;
>+  bool has_recurring_schedule_on_failed_networks_ = false;
>+  rtc::Thread* thread_;
>+  rtc::AsyncInvoker invoker_for_all_networks_;
>+  rtc::AsyncInvoker invoker_for_failed_networks_;
>+  // Used to generate random intervals for regather_all_networks_interval_range.
>+  Random rand_;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // P2P_BASE_REGATHERINGCONTROLLER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/regatheringcontroller_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/regatheringcontroller_unittest.cc
>new file mode 100644
>index 00000000000..6ae64e8658d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/regatheringcontroller_unittest.cc
>@@ -0,0 +1,307 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <map>
>+#include <memory>
>+#include <string>
>+#include <vector>
>+
>+#include "p2p/base/fakeportallocator.h"
>+#include "p2p/base/mockicetransport.h"
>+#include "p2p/base/p2pconstants.h"
>+#include "p2p/base/port.h"
>+#include "p2p/base/regatheringcontroller.h"
>+#include "p2p/base/stunserver.h"
>+#include "rtc_base/gunit.h"
>+#include "rtc_base/refcountedobject.h"
>+#include "rtc_base/scoped_ref_ptr.h"
>+#include "rtc_base/socketaddress.h"
>+#include "rtc_base/thread.h"
>+#include "rtc_base/virtualsocketserver.h"
>+
>+namespace {
>+
>+const int kOnlyLocalPorts = cricket::PORTALLOCATOR_DISABLE_STUN |
>+                            cricket::PORTALLOCATOR_DISABLE_RELAY |
>+                            cricket::PORTALLOCATOR_DISABLE_TCP;
>+// The address of the public STUN server.
>+const rtc::SocketAddress kStunAddr("99.99.99.1", cricket::STUN_SERVER_PORT);
>+// The addresses for the public TURN server.
>+const rtc::SocketAddress kTurnUdpIntAddr("99.99.99.3",
>+                                         cricket::STUN_SERVER_PORT);
>+const cricket::RelayCredentials kRelayCredentials("test", "test");
>+const char kIceUfrag[] = "UF00";
>+const char kIcePwd[] = "TESTICEPWD00000000000000";
>+
>+}  // namespace
>+
>+namespace webrtc {
>+
>+class RegatheringControllerTest : public testing::Test,
>+                                  public sigslot::has_slots<> {
>+ public:
>+  RegatheringControllerTest()
>+      : vss_(new rtc::VirtualSocketServer()),
>+        thread_(vss_.get()),
>+        ice_transport_(new cricket::MockIceTransport()),
>+        allocator_(
>+            new cricket::FakePortAllocator(rtc::Thread::Current(), nullptr)) {
>+    BasicRegatheringController::Config regathering_config(absl::nullopt, 0);
>+    regathering_controller_.reset(new BasicRegatheringController(
>+        regathering_config, ice_transport_.get(), rtc::Thread::Current()));
>+  }
>+
>+  // Initializes the allocator and gathers candidates once by StartGettingPorts.
>+  void InitializeAndGatherOnce() {
>+    cricket::ServerAddresses stun_servers;
>+    stun_servers.insert(kStunAddr);
>+    cricket::RelayServerConfig turn_server(cricket::RELAY_TURN);
>+    turn_server.credentials = kRelayCredentials;
>+    turn_server.ports.push_back(
>+        cricket::ProtocolAddress(kTurnUdpIntAddr, cricket::PROTO_UDP));
>+    std::vector<cricket::RelayServerConfig> turn_servers(1, turn_server);
>+    allocator_->set_flags(kOnlyLocalPorts);
>+    allocator_->SetConfiguration(stun_servers, turn_servers, 0 /* pool size */,
>+                                 false /* prune turn ports */);
>+    allocator_session_ = allocator_->CreateSession(
>+        "test", cricket::ICE_CANDIDATE_COMPONENT_RTP, kIceUfrag, kIcePwd);
>+    // The gathering will take place on the current thread and the following
>+    // call of StartGettingPorts is blocking. We will not ClearGettingPorts
>+    // prematurely.
>+    allocator_session_->StartGettingPorts();
>+    allocator_session_->SignalIceRegathering.connect(
>+        this, &RegatheringControllerTest::OnIceRegathering);
>+    regathering_controller_->set_allocator_session(allocator_session_.get());
>+  }
>+
>+  // The regathering controller is initialized with the allocator session
>+  // cleared. Only after clearing the session, we would be able to regather. See
>+  // the comments for BasicRegatheringController in regatheringcontroller.h.
>+  void InitializeAndGatherOnceWithSessionCleared() {
>+    InitializeAndGatherOnce();
>+    allocator_session_->ClearGettingPorts();
>+  }
>+
>+  void OnIceRegathering(cricket::PortAllocatorSession* allocator_session,
>+                        cricket::IceRegatheringReason reason) {
>+    ++count_[reason];
>+  }
>+
>+  int GetRegatheringReasonCount(cricket::IceRegatheringReason reason) {
>+    return count_[reason];
>+  }
>+
>+  BasicRegatheringController* regathering_controller() {
>+    return regathering_controller_.get();
>+  }
>+
>+ private:
>+  std::unique_ptr<rtc::VirtualSocketServer> vss_;
>+  rtc::AutoSocketServerThread thread_;
>+  std::unique_ptr<cricket::IceTransportInternal> ice_transport_;
>+  std::unique_ptr<BasicRegatheringController> regathering_controller_;
>+  std::unique_ptr<cricket::PortAllocator> allocator_;
>+  std::unique_ptr<cricket::PortAllocatorSession> allocator_session_;
>+  std::map<cricket::IceRegatheringReason, int> count_;
>+};
>+
>+// Tests that ICE regathering occurs only if the port allocator session is
>+// cleared. A port allocation session is not cleared if the initial gathering is
>+// still in progress or the continual gathering is not enabled.
>+TEST_F(RegatheringControllerTest,
>+       IceRegatheringDoesNotOccurIfSessionNotCleared) {
>+  rtc::ScopedFakeClock clock;
>+  InitializeAndGatherOnce();  // Session not cleared.
>+
>+  rtc::IntervalRange regather_all_networks_interval_range(2000, 2000);
>+  BasicRegatheringController::Config config(
>+      regather_all_networks_interval_range, 2000);
>+  regathering_controller()->SetConfig(config);
>+  regathering_controller()->Start();
>+  SIMULATED_WAIT(false, 10000, clock);
>+  // Expect no regathering in the last 10s.
>+  EXPECT_EQ(0, GetRegatheringReasonCount(
>+                   cricket::IceRegatheringReason::OCCASIONAL_REFRESH));
>+  EXPECT_EQ(0, GetRegatheringReasonCount(
>+                   cricket::IceRegatheringReason::NETWORK_FAILURE));
>+}
>+
>+TEST_F(RegatheringControllerTest, IceRegatheringRepeatsAsScheduled) {
>+  rtc::ScopedFakeClock clock;
>+  InitializeAndGatherOnceWithSessionCleared();
>+
>+  rtc::IntervalRange regather_all_networks_interval_range(2000, 2000);
>+  BasicRegatheringController::Config config(
>+      regather_all_networks_interval_range, 2000);
>+  regathering_controller()->SetConfig(config);
>+  regathering_controller()->Start();
>+  SIMULATED_WAIT(false, 2000 - 1, clock);
>+  // Expect no regathering.
>+  EXPECT_EQ(0, GetRegatheringReasonCount(
>+                   cricket::IceRegatheringReason::OCCASIONAL_REFRESH));
>+  EXPECT_EQ(0, GetRegatheringReasonCount(
>+                   cricket::IceRegatheringReason::NETWORK_FAILURE));
>+  SIMULATED_WAIT(false, 2, clock);
>+  // Expect regathering on all networks and on failed networks to happen once
>+  // respectively in that last 2s with 2s interval.
>+  EXPECT_EQ(1, GetRegatheringReasonCount(
>+                   cricket::IceRegatheringReason::OCCASIONAL_REFRESH));
>+  EXPECT_EQ(1, GetRegatheringReasonCount(
>+                   cricket::IceRegatheringReason::NETWORK_FAILURE));
>+  SIMULATED_WAIT(false, 11000, clock);
>+  // Expect regathering to happen for another 5 times in 11s with 2s interval.
>+  EXPECT_EQ(6, GetRegatheringReasonCount(
>+                   cricket::IceRegatheringReason::OCCASIONAL_REFRESH));
>+  EXPECT_EQ(6, GetRegatheringReasonCount(
>+                   cricket::IceRegatheringReason::NETWORK_FAILURE));
>+}
>+
>+// Tests that the schedule of ICE regathering on all networks can be started
>+// when not scheduled initially.
>+TEST_F(RegatheringControllerTest,
>+       IceRegatheringOnAllNetworksCanBeScheduledAfterStart) {
>+  rtc::ScopedFakeClock clock;
>+  InitializeAndGatherOnceWithSessionCleared();
>+
>+  BasicRegatheringController::Config config(absl::nullopt, 2000);
>+  regathering_controller()->SetConfig(config);
>+  regathering_controller()->Start();
>+  SIMULATED_WAIT(false, 3000, clock);
>+  // Expect no regathering on all networks.
>+  EXPECT_EQ(0, GetRegatheringReasonCount(
>+                   cricket::IceRegatheringReason::OCCASIONAL_REFRESH));
>+  config.regather_on_all_networks_interval_range =
>+      rtc::IntervalRange(2000, 2000);
>+  regathering_controller()->SetConfig(config);
>+  SIMULATED_WAIT(false, 11000, clock);
>+  // Expect regathering to happen for 5 times on all networks in the last 11s
>+  // with 2s interval.
>+  EXPECT_EQ(5, GetRegatheringReasonCount(
>+                   cricket::IceRegatheringReason::OCCASIONAL_REFRESH));
>+}
>+
>+// Tests that ICE regathering on all networks can be canceled by changing the
>+// config.
>+TEST_F(RegatheringControllerTest, IceRegatheringOnAllNetworksCanBeCanceled) {
>+  rtc::ScopedFakeClock clock;
>+  InitializeAndGatherOnceWithSessionCleared();
>+
>+  rtc::IntervalRange regather_all_networks_interval_range(2000, 2000);
>+  BasicRegatheringController::Config config(
>+      regather_all_networks_interval_range, 2000);
>+  regathering_controller()->SetConfig(config);
>+  regathering_controller()->Start();
>+  config.regather_on_all_networks_interval_range.reset();
>+  // Set the regathering interval range on all networks to nullopt should cancel
>+  // the schedule on all networks.
>+  regathering_controller()->SetConfig(config);
>+  SIMULATED_WAIT(false, 10000, clock);
>+  // Expect no regathering on all networks happened in the last 10s.
>+  EXPECT_EQ(0, GetRegatheringReasonCount(
>+                   cricket::IceRegatheringReason::OCCASIONAL_REFRESH));
>+}
>+
>+// Tests that canceling the regathering on all networks does not cancel the
>+// schedule on failed networks.
>+TEST_F(RegatheringControllerTest,
>+       CancelingRegatheringOnAllNetworksDoesNotCancelOnFailedNetworks) {
>+  rtc::ScopedFakeClock clock;
>+  InitializeAndGatherOnceWithSessionCleared();
>+
>+  rtc::IntervalRange regather_all_networks_interval_range(2000, 2000);
>+  BasicRegatheringController::Config config(
>+      regather_all_networks_interval_range, 2000);
>+  regathering_controller()->SetConfig(config);
>+  regathering_controller()->Start();
>+  config.regather_on_all_networks_interval_range =
>+      rtc::IntervalRange(20000, 20000);
>+  // Canceling and rescheduling the regathering on all networks should not
>+  // impact the schedule for failed networks.
>+  regathering_controller()->SetConfig(config);
>+  SIMULATED_WAIT(false, 11000, clock);
>+  // Expect regathering to happen for 5 times for failed networks in the last
>+  // 11s with 2s interval.
>+  EXPECT_EQ(5, GetRegatheringReasonCount(
>+                   cricket::IceRegatheringReason::NETWORK_FAILURE));
>+}
>+
>+// Tests that canceling the regathering on failed networks does not cancel the
>+// schedule on all networks.
>+TEST_F(RegatheringControllerTest,
>+       CancelingRegatheringOnFailedNetworksDoesNotCancelOnAllNetworks) {
>+  rtc::ScopedFakeClock clock;
>+  InitializeAndGatherOnceWithSessionCleared();
>+
>+  rtc::IntervalRange regather_all_networks_interval_range(2000, 2000);
>+  BasicRegatheringController::Config config(
>+      regather_all_networks_interval_range, 2000);
>+  regathering_controller()->SetConfig(config);
>+  regathering_controller()->Start();
>+  config.regather_on_failed_networks_interval = 20000;
>+  // Canceling and rescheduling the regathering on failed networks should not
>+  // impact the schedule for all networks.
>+  regathering_controller()->SetConfig(config);
>+  SIMULATED_WAIT(false, 11000, clock);
>+  // Expect regathering to happen for 5 times for all networks in the last 11s
>+  // with 2s interval.
>+  EXPECT_EQ(5, GetRegatheringReasonCount(
>+                   cricket::IceRegatheringReason::OCCASIONAL_REFRESH));
>+}
>+
>+// Tests that the schedule of ICE regathering on all networks can be canceled
>+// and replaced by a new recurring schedule.
>+TEST_F(RegatheringControllerTest,
>+       ScheduleOfIceRegatheringOnAllNetworksCanBeReplaced) {
>+  rtc::ScopedFakeClock clock;
>+  InitializeAndGatherOnceWithSessionCleared();
>+
>+  rtc::IntervalRange regather_all_networks_interval_range(2000, 2000);
>+  BasicRegatheringController::Config config(
>+      regather_all_networks_interval_range, 2000);
>+  regathering_controller()->SetConfig(config);
>+  regathering_controller()->Start();
>+  config.regather_on_all_networks_interval_range =
>+      rtc::IntervalRange(5000, 5000);
>+  regathering_controller()->SetConfig(config);
>+  SIMULATED_WAIT(false, 3000, clock);
>+  // Expect no regathering from the previous schedule.
>+  EXPECT_EQ(0, GetRegatheringReasonCount(
>+                   cricket::IceRegatheringReason::OCCASIONAL_REFRESH));
>+  SIMULATED_WAIT(false, 11000 - 3000, clock);
>+  // Expect regathering to happen twice in the last 11s with 5s interval.
>+  EXPECT_EQ(2, GetRegatheringReasonCount(
>+                   cricket::IceRegatheringReason::OCCASIONAL_REFRESH));
>+}
>+
>+// Tests that the schedule of ICE regathering on failed networks can be canceled
>+// and replaced by a new recurring schedule.
>+TEST_F(RegatheringControllerTest,
>+       ScheduleOfIceRegatheringOnFailedNetworksCanBeReplaced) {
>+  rtc::ScopedFakeClock clock;
>+  InitializeAndGatherOnceWithSessionCleared();
>+
>+  rtc::IntervalRange regather_all_networks_interval_range(2000, 2000);
>+  BasicRegatheringController::Config config(
>+      regather_all_networks_interval_range, 2000);
>+  regathering_controller()->SetConfig(config);
>+  regathering_controller()->Start();
>+  config.regather_on_failed_networks_interval = 5000;
>+  regathering_controller()->SetConfig(config);
>+  SIMULATED_WAIT(false, 3000, clock);
>+  // Expect no regathering from the previous schedule.
>+  EXPECT_EQ(0, GetRegatheringReasonCount(
>+                   cricket::IceRegatheringReason::NETWORK_FAILURE));
>+  SIMULATED_WAIT(false, 11000 - 3000, clock);
>+  // Expect regathering to happen twice in the last 11s with 5s interval.
>+  EXPECT_EQ(2, GetRegatheringReasonCount(
>+                   cricket::IceRegatheringReason::NETWORK_FAILURE));
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/relayport.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/relayport.cc
>index 045f0c36448..7413f3b6e2d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/relayport.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/relayport.cc
>@@ -18,11 +18,11 @@
> namespace cricket {
> 
> static const int kMessageConnectTimeout = 1;
>-static const int kKeepAliveDelay           = 10 * 60 * 1000;
>-static const int kRetryTimeout             = 50 * 1000;  // ICE says 50 secs
>+static const int kKeepAliveDelay = 10 * 60 * 1000;
>+static const int kRetryTimeout = 50 * 1000;  // ICE says 50 secs
> // How long to wait for a socket to connect to remote host in milliseconds
> // before trying another connection.
>-static const int kSoftConnectTimeoutMs     = 3 * 1000;
>+static const int kSoftConnectTimeoutMs = 3 * 1000;
> 
> // Handles a connection to one address/port/protocol combination for a
> // particular RelayEntry.
>@@ -34,17 +34,11 @@ class RelayConnection : public sigslot::has_slots<> {
>   ~RelayConnection() override;
>   rtc::AsyncPacketSocket* socket() const { return socket_; }
> 
>-  const ProtocolAddress* protocol_address() {
>-    return protocol_address_;
>-  }
>+  const ProtocolAddress* protocol_address() { return protocol_address_; }
> 
>-  rtc::SocketAddress GetAddress() const {
>-    return protocol_address_->address;
>-  }
>+  rtc::SocketAddress GetAddress() const { return protocol_address_->address; }
> 
>-  ProtocolType GetProtocol() const {
>-    return protocol_address_->proto;
>-  }
>+  ProtocolType GetProtocol() const { return protocol_address_->proto; }
> 
>   int SetSocketOption(rtc::Socket::Option opt, int value);
> 
>@@ -67,15 +61,14 @@ class RelayConnection : public sigslot::has_slots<> {
>  private:
>   rtc::AsyncPacketSocket* socket_;
>   const ProtocolAddress* protocol_address_;
>-  StunRequestManager *request_manager_;
>+  StunRequestManager* request_manager_;
> };
> 
> // Manages a number of connections to the relayserver, one for each
> // available protocol. We aim to use each connection for only a
> // specific destination address so that we can avoid wrapping every
> // packet in a STUN send / data indication.
>-class RelayEntry : public rtc::MessageHandler,
>-                   public sigslot::has_slots<> {
>+class RelayEntry : public rtc::MessageHandler, public sigslot::has_slots<> {
>  public:
>   RelayEntry(RelayPort* port, const rtc::SocketAddress& ext_addr);
>   ~RelayEntry() override;
>@@ -107,7 +100,8 @@ class RelayEntry : public rtc::MessageHandler,
> 
>   // Sends a packet to the given destination address using the socket of this
>   // entry.  This will wrap the packet in STUN if necessary.
>-  int SendTo(const void* data, size_t size,
>+  int SendTo(const void* data,
>+             size_t size,
>              const rtc::SocketAddress& addr,
>              const rtc::PacketOptions& options);
> 
>@@ -140,11 +134,11 @@ class RelayEntry : public rtc::MessageHandler,
>   void OnSocketClose(rtc::AsyncPacketSocket* socket, int error);
> 
>   // Called when a packet is received on this socket.
>-  void OnReadPacket(
>-    rtc::AsyncPacketSocket* socket,
>-    const char* data, size_t size,
>-    const rtc::SocketAddress& remote_addr,
>-    const rtc::PacketTime& packet_time);
>+  void OnReadPacket(rtc::AsyncPacketSocket* socket,
>+                    const char* data,
>+                    size_t size,
>+                    const rtc::SocketAddress& remote_addr,
>+                    const rtc::PacketTime& packet_time);
> 
>   void OnSentPacket(rtc::AsyncPacketSocket* socket,
>                     const rtc::SentPacket& sent_packet);
>@@ -154,7 +148,8 @@ class RelayEntry : public rtc::MessageHandler,
> 
>   // Sends the given data on the socket to the server with no wrapping.  This
>   // returns the number of bytes written or -1 if an error occurred.
>-  int SendPacket(const void* data, size_t size,
>+  int SendPacket(const void* data,
>+                 size_t size,
>                  const rtc::PacketOptions& options);
> };
> 
>@@ -196,8 +191,7 @@ RelayPort::RelayPort(rtc::Thread* thread,
>            password),
>       ready_(false),
>       error_(0) {
>-  entries_.push_back(
>-      new RelayEntry(this, rtc::SocketAddress()));
>+  entries_.push_back(new RelayEntry(this, rtc::SocketAddress()));
>   // TODO(?): set local preference value for TCP based candidates.
> }
> 
>@@ -210,9 +204,8 @@ RelayPort::~RelayPort() {
> void RelayPort::AddServerAddress(const ProtocolAddress& addr) {
>   // Since HTTP proxies usually only allow 443,
>   // let's up the priority on PROTO_SSLTCP
>-  if (addr.proto == PROTO_SSLTCP &&
>-      (proxy().type == rtc::PROXY_HTTPS ||
>-       proxy().type == rtc::PROXY_UNKNOWN)) {
>+  if (addr.proto == PROTO_SSLTCP && (proxy().type == rtc::PROXY_HTTPS ||
>+                                     proxy().type == rtc::PROXY_UNKNOWN)) {
>     server_addr_.push_front(addr);
>   } else {
>     server_addr_.push_back(addr);
>@@ -235,8 +228,7 @@ void RelayPort::AddExternalAddress(const ProtocolAddress& addr) {
> void RelayPort::SetReady() {
>   if (!ready_) {
>     std::vector<ProtocolAddress>::iterator iter;
>-    for (iter = external_addr_.begin();
>-         iter != external_addr_.end(); ++iter) {
>+    for (iter = external_addr_.begin(); iter != external_addr_.end(); ++iter) {
>       std::string proto_name = ProtoToString(iter->proto);
>       // In case of Gturn, related address is set to null socket address.
>       // This is due to as mapped address stun attribute is used for allocated
>@@ -250,7 +242,7 @@ void RelayPort::SetReady() {
>   }
> }
> 
>-const ProtocolAddress * RelayPort::ServerAddress(size_t index) const {
>+const ProtocolAddress* RelayPort::ServerAddress(size_t index) const {
>   if (index < server_addr_.size())
>     return &server_addr_[index];
>   return NULL;
>@@ -260,8 +252,7 @@ bool RelayPort::HasMagicCookie(const char* data, size_t size) {
>   if (size < 24 + sizeof(TURN_MAGIC_COOKIE_VALUE)) {
>     return false;
>   } else {
>-    return memcmp(data + 24,
>-                  TURN_MAGIC_COOKIE_VALUE,
>+    return memcmp(data + 24, TURN_MAGIC_COOKIE_VALUE,
>                   sizeof(TURN_MAGIC_COOKIE_VALUE)) == 0;
>   }
> }
>@@ -300,12 +291,13 @@ Connection* RelayPort::CreateConnection(const Candidate& address,
>     }
>   }
> 
>-  Connection * conn = new ProxyConnection(this, index, address);
>+  Connection* conn = new ProxyConnection(this, index, address);
>   AddOrReplaceConnection(conn);
>   return conn;
> }
> 
>-int RelayPort::SendTo(const void* data, size_t size,
>+int RelayPort::SendTo(const void* data,
>+                      size_t size,
>                       const rtc::SocketAddress& addr,
>                       const rtc::PacketOptions& options,
>                       bool payload) {
>@@ -349,7 +341,9 @@ int RelayPort::SendTo(const void* data, size_t size,
>   }
> 
>   // Send the actual contents to the server using the usual mechanism.
>-  int sent = entry->SendTo(data, size, addr, options);
>+  rtc::PacketOptions modified_options(options);
>+  CopyPortInformationToPacketInfo(&modified_options.info_signaled_after_sent);
>+  int sent = entry->SendTo(data, size, addr, modified_options);
>   if (sent <= 0) {
>     RTC_DCHECK(sent < 0);
>     error_ = entry->GetError();
>@@ -398,11 +392,11 @@ ProtocolType RelayPort::GetProtocol() const {
>   return PROTO_UDP;
> }
> 
>-void RelayPort::OnReadPacket(
>-    const char* data, size_t size,
>-    const rtc::SocketAddress& remote_addr,
>-    ProtocolType proto,
>-    const rtc::PacketTime& packet_time) {
>+void RelayPort::OnReadPacket(const char* data,
>+                             size_t size,
>+                             const rtc::SocketAddress& remote_addr,
>+                             ProtocolType proto,
>+                             const rtc::PacketTime& packet_time) {
>   if (Connection* conn = GetConnection(remote_addr)) {
>     conn->OnReadPacket(data, size, packet_time);
>   } else {
>@@ -413,8 +407,7 @@ void RelayPort::OnReadPacket(
> RelayConnection::RelayConnection(const ProtocolAddress* protocol_address,
>                                  rtc::AsyncPacketSocket* socket,
>                                  rtc::Thread* thread)
>-    : socket_(socket),
>-      protocol_address_(protocol_address) {
>+    : socket_(socket), protocol_address_(protocol_address) {
>   request_manager_ = new StunRequestManager(thread);
>   request_manager_->SignalSendPacket.connect(this,
>                                              &RelayConnection::OnSendPacket);
>@@ -425,8 +418,7 @@ RelayConnection::~RelayConnection() {
>   delete socket_;
> }
> 
>-int RelayConnection::SetSocketOption(rtc::Socket::Option opt,
>-                                     int value) {
>+int RelayConnection::SetSocketOption(rtc::Socket::Option opt, int value) {
>   if (socket_) {
>     return socket_->SetOption(opt, value);
>   }
>@@ -437,19 +429,22 @@ bool RelayConnection::CheckResponse(StunMessage* msg) {
>   return request_manager_->CheckResponse(msg);
> }
> 
>-void RelayConnection::OnSendPacket(const void* data, size_t size,
>+void RelayConnection::OnSendPacket(const void* data,
>+                                   size_t size,
>                                    StunRequest* req) {
>   // TODO(mallinath) Find a way to get DSCP value from Port.
>   rtc::PacketOptions options;  // Default dscp set to NO_CHANGE.
>   int sent = socket_->SendTo(data, size, GetAddress(), options);
>   if (sent <= 0) {
>-    RTC_LOG(LS_VERBOSE) << "OnSendPacket: failed sending to " << GetAddress()
>+    RTC_LOG(LS_VERBOSE) << "OnSendPacket: failed sending to "
>+                        << GetAddress().ToString()
>                         << strerror(socket_->GetError());
>     RTC_DCHECK(sent < 0);
>   }
> }
> 
>-int RelayConnection::Send(const void* pv, size_t cb,
>+int RelayConnection::Send(const void* pv,
>+                          size_t cb,
>                           const rtc::PacketOptions& options) {
>   return socket_->SendTo(pv, cb, GetAddress(), options);
> }
>@@ -458,12 +453,13 @@ void RelayConnection::SendAllocateRequest(RelayEntry* entry, int delay) {
>   request_manager_->SendDelayed(new AllocateRequest(entry, this), delay);
> }
> 
>-RelayEntry::RelayEntry(RelayPort* port,
>-                       const rtc::SocketAddress& ext_addr)
>-    : port_(port), ext_addr_(ext_addr),
>-      server_index_(0), connected_(false), locked_(false),
>-      current_connection_(NULL) {
>-}
>+RelayEntry::RelayEntry(RelayPort* port, const rtc::SocketAddress& ext_addr)
>+    : port_(port),
>+      ext_addr_(ext_addr),
>+      server_index_(0),
>+      connected_(false),
>+      locked_(false),
>+      current_connection_(NULL) {}
> 
> RelayEntry::~RelayEntry() {
>   // Remove all RelayConnections and dispose sockets.
>@@ -508,7 +504,7 @@ void RelayEntry::Connect() {
>         rtc::SocketAddress(port_->Network()->GetBestIP(), 0), ra->address,
>         port_->proxy(), port_->user_agent(), opts);
>   } else {
>-    RTC_LOG(LS_WARNING) << "Unknown protocol (" << ra->proto << ")";
>+    RTC_LOG(LS_WARNING) << "Unknown protocol: " << ra->proto;
>   }
> 
>   // If we failed to get a socket, move on to the next protocol.
>@@ -564,7 +560,8 @@ void RelayEntry::OnConnect(const rtc::SocketAddress& mapped_addr,
>   port_->SetReady();
> }
> 
>-int RelayEntry::SendTo(const void* data, size_t size,
>+int RelayEntry::SendTo(const void* data,
>+                       size_t size,
>                        const rtc::SocketAddress& addr,
>                        const rtc::PacketOptions& options) {
>   // If this connection is locked to the address given, then we can send the
>@@ -632,8 +629,7 @@ int RelayEntry::SetSocketOption(rtc::Socket::Option opt, int value) {
>   return socket_error;
> }
> 
>-void RelayEntry::HandleConnectFailure(
>-    rtc::AsyncPacketSocket* socket) {
>+void RelayEntry::HandleConnectFailure(rtc::AsyncPacketSocket* socket) {
>   // Make sure it's the current connection that has failed, it might
>   // be an old socked that has not yet been disposed.
>   if (!socket ||
>@@ -647,12 +643,12 @@ void RelayEntry::HandleConnectFailure(
>   }
> }
> 
>-void RelayEntry::OnMessage(rtc::Message *pmsg) {
>+void RelayEntry::OnMessage(rtc::Message* pmsg) {
>   RTC_DCHECK(pmsg->message_id == kMessageConnectTimeout);
>   if (current_connection_) {
>     const ProtocolAddress* ra = current_connection_->protocol_address();
>     RTC_LOG(LS_WARNING) << "Relay " << ra->proto << " connection to "
>-                        << ra->address << " timed out";
>+                        << ra->address.ToString() << " timed out";
> 
>     // Currently we connect to each server address in sequence. If we
>     // have more addresses to try, treat this is an error and move on to
>@@ -676,17 +672,16 @@ void RelayEntry::OnSocketConnect(rtc::AsyncPacketSocket* socket) {
>   }
> }
> 
>-void RelayEntry::OnSocketClose(rtc::AsyncPacketSocket* socket,
>-                               int error) {
>-  RTC_PLOG(LERROR, error) << "Relay connection failed: socket closed";
>+void RelayEntry::OnSocketClose(rtc::AsyncPacketSocket* socket, int error) {
>+  RTC_LOG_ERR_EX(LERROR, error) << "Relay connection failed: socket closed";
>   HandleConnectFailure(socket);
> }
> 
>-void RelayEntry::OnReadPacket(
>-    rtc::AsyncPacketSocket* socket,
>-    const char* data, size_t size,
>-    const rtc::SocketAddress& remote_addr,
>-    const rtc::PacketTime& packet_time) {
>+void RelayEntry::OnReadPacket(rtc::AsyncPacketSocket* socket,
>+                              const char* data,
>+                              size_t size,
>+                              const rtc::SocketAddress& remote_addr,
>+                              const rtc::PacketTime& packet_time) {
>   // RTC_DCHECK(remote_addr == port_->server_addr());
>   // TODO(?): are we worried about this?
> 
>@@ -720,7 +715,7 @@ void RelayEntry::OnReadPacket(
>     return;
>   } else if (msg.type() == STUN_SEND_RESPONSE) {
>     if (const StunUInt32Attribute* options_attr =
>-        msg.GetUInt32(STUN_ATTR_OPTIONS)) {
>+            msg.GetUInt32(STUN_ATTR_OPTIONS)) {
>       if (options_attr->value() & 0x1) {
>         locked_ = true;
>       }
>@@ -767,7 +762,8 @@ void RelayEntry::OnReadyToSend(rtc::AsyncPacketSocket* socket) {
>   }
> }
> 
>-int RelayEntry::SendPacket(const void* data, size_t size,
>+int RelayEntry::SendPacket(const void* data,
>+                           size_t size,
>                            const rtc::PacketOptions& options) {
>   int sent = 0;
>   if (current_connection_) {
>@@ -778,11 +774,8 @@ int RelayEntry::SendPacket(const void* data, size_t size,
>   return sent;
> }
> 
>-AllocateRequest::AllocateRequest(RelayEntry* entry,
>-                                 RelayConnection* connection)
>-    : StunRequest(new RelayMessage()),
>-      entry_(entry),
>-      connection_(connection) {
>+AllocateRequest::AllocateRequest(RelayEntry* entry, RelayConnection* connection)
>+    : StunRequest(new RelayMessage()), entry_(entry), connection_(connection) {
>   start_time_ = rtc::TimeMillis();
> }
> 
>@@ -790,9 +783,8 @@ void AllocateRequest::Prepare(StunMessage* request) {
>   request->SetType(STUN_ALLOCATE_REQUEST);
> 
>   auto username_attr = StunAttribute::CreateByteString(STUN_ATTR_USERNAME);
>-  username_attr->CopyBytes(
>-      entry_->port()->username_fragment().c_str(),
>-      entry_->port()->username_fragment().size());
>+  username_attr->CopyBytes(entry_->port()->username_fragment().c_str(),
>+                           entry_->port()->username_fragment().size());
>   request->AddAttribute(std::move(username_attr));
> }
> 
>@@ -806,10 +798,9 @@ int AllocateRequest::resend_delay() {
>   if (count_ == 0) {
>     return 0;
>   }
>-  return 100 * std::max(1 << (count_-1), 2);
>+  return 100 * std::max(1 << (count_ - 1), 2);
> }
> 
>-
> void AllocateRequest::OnResponse(StunMessage* response) {
>   const StunAddressAttribute* addr_attr =
>       response->GetAddress(STUN_ATTR_MAPPED_ADDRESS);
>@@ -832,9 +823,8 @@ void AllocateRequest::OnErrorResponse(StunMessage* response) {
>   if (!attr) {
>     RTC_LOG(LS_ERROR) << "Missing allocate response error code.";
>   } else {
>-    RTC_LOG(INFO) << "Allocate error response:"
>-                  << " code=" << attr->code() << " reason='" << attr->reason()
>-                  << "'";
>+    RTC_LOG(INFO) << "Allocate error response: code=" << attr->code()
>+                  << " reason=" << attr->reason();
>   }
> 
>   if (rtc::TimeMillis() - start_time_ <= kRetryTimeout)
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/relayport.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/relayport.h
>index 3e1416df776..48a939e0cfd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/relayport.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/relayport.h
>@@ -62,7 +62,7 @@ class RelayPort : public Port {
>   bool SupportsProtocol(const std::string& protocol) const override;
>   ProtocolType GetProtocol() const override;
> 
>-  const ProtocolAddress * ServerAddress(size_t index) const;
>+  const ProtocolAddress* ServerAddress(size_t index) const;
>   bool IsReady() { return ready_; }
> 
>   // Used for testing.
>@@ -88,7 +88,8 @@ class RelayPort : public Port {
>              bool payload) override;
> 
>   // Dispatches the given packet to the port or connection as appropriate.
>-  void OnReadPacket(const char* data, size_t size,
>+  void OnReadPacket(const char* data,
>+                    size_t size,
>                     const rtc::SocketAddress& remote_addr,
>                     ProtocolType proto,
>                     const rtc::PacketTime& packet_time);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/relayport_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/relayport_unittest.cc
>index 274bab369f9..e6e922fdaf7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/relayport_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/relayport_unittest.cc
>@@ -41,8 +41,7 @@ static const int kMaxTimeoutMs = 5000;
> // RelayPort and created sockets by listening for signals such as,
> // SignalConnectFailure, SignalConnectTimeout, SignalSocketClosed and
> // SignalReadPacket.
>-class RelayPortTest : public testing::Test,
>-                      public sigslot::has_slots<> {
>+class RelayPortTest : public testing::Test, public sigslot::has_slots<> {
>  public:
>   RelayPortTest()
>       : virtual_socket_server_(new rtc::VirtualSocketServer()),
>@@ -63,7 +62,8 @@ class RelayPortTest : public testing::Test,
>   }
> 
>   void OnReadPacket(rtc::AsyncPacketSocket* socket,
>-                    const char* data, size_t size,
>+                    const char* data,
>+                    size_t size,
>                     const rtc::SocketAddress& remote_addr,
>                     const rtc::PacketTime& packet_time) {
>     received_packet_count_[socket]++;
>@@ -80,17 +80,15 @@ class RelayPortTest : public testing::Test,
>  protected:
>   virtual void SetUp() {
>     // The relay server needs an external socket to work properly.
>-    rtc::AsyncUDPSocket* ext_socket =
>-        CreateAsyncUdpSocket(kRelayExtAddr);
>+    rtc::AsyncUDPSocket* ext_socket = CreateAsyncUdpSocket(kRelayExtAddr);
>     relay_server_->AddExternalSocket(ext_socket);
> 
>     // Listen for failures.
>-    relay_port_->SignalConnectFailure.
>-        connect(this, &RelayPortTest::OnConnectFailure);
>+    relay_port_->SignalConnectFailure.connect(this,
>+                                              &RelayPortTest::OnConnectFailure);
> 
>     // Listen for soft timeouts.
>-    relay_port_->SignalSoftTimeout.
>-        connect(this, &RelayPortTest::OnSoftTimeout);
>+    relay_port_->SignalSoftTimeout.connect(this, &RelayPortTest::OnSoftTimeout);
>   }
> 
>   // Udp has the highest 'goodness' value of the three different
>@@ -177,8 +175,7 @@ class RelayPortTest : public testing::Test,
>         cricket::ProtocolAddress(kRelayTcpAddr, cricket::PROTO_TCP);
> 
>     // Create a ssl server socket for the RelayServer.
>-    rtc::AsyncSocket* ssl_server_socket =
>-        CreateServerSocket(kRelaySslAddr);
>+    rtc::AsyncSocket* ssl_server_socket = CreateServerSocket(kRelaySslAddr);
>     relay_server_->AddInternalServerSocket(ssl_server_socket,
>                                            cricket::PROTO_SSLTCP);
> 
>@@ -210,7 +207,7 @@ class RelayPortTest : public testing::Test,
>  private:
>   rtc::AsyncUDPSocket* CreateAsyncUdpSocket(const SocketAddress addr) {
>     rtc::AsyncSocket* socket =
>-        virtual_socket_server_->CreateAsyncSocket(SOCK_DGRAM);
>+        virtual_socket_server_->CreateAsyncSocket(AF_INET, SOCK_DGRAM);
>     rtc::AsyncUDPSocket* packet_socket =
>         rtc::AsyncUDPSocket::Create(socket, addr);
>     EXPECT_TRUE(packet_socket != NULL);
>@@ -220,7 +217,7 @@ class RelayPortTest : public testing::Test,
> 
>   rtc::AsyncSocket* CreateServerSocket(const SocketAddress addr) {
>     rtc::AsyncSocket* socket =
>-        virtual_socket_server_->CreateAsyncSocket(SOCK_STREAM);
>+        virtual_socket_server_->CreateAsyncSocket(AF_INET, SOCK_STREAM);
>     EXPECT_GE(socket->Bind(addr), 0);
>     EXPECT_GE(socket->Listen(5), 0);
>     return socket;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/relayserver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/relayserver.cc
>index 9bb53ad45aa..8e17a79db8d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/relayserver.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/relayserver.cc
>@@ -33,7 +33,9 @@ const int MAX_LIFETIME = 15 * 60 * 1000;
> const uint32_t USERNAME_LENGTH = 16;
> 
> // Calls SendTo on the given socket and logs any bad results.
>-void Send(rtc::AsyncPacketSocket* socket, const char* bytes, size_t size,
>+void Send(rtc::AsyncPacketSocket* socket,
>+          const char* bytes,
>+          size_t size,
>           const rtc::SocketAddress& addr) {
>   rtc::PacketOptions options;
>   int result = socket->SendTo(bytes, size, addr, options);
>@@ -55,9 +57,12 @@ void SendStun(const StunMessage& msg,
> }
> 
> // Constructs a STUN error response and sends it on the given socket.
>-void SendStunError(const StunMessage& msg, rtc::AsyncPacketSocket* socket,
>-                   const rtc::SocketAddress& remote_addr, int error_code,
>-                   const char* error_desc, const std::string& magic_cookie) {
>+void SendStunError(const StunMessage& msg,
>+                   rtc::AsyncPacketSocket* socket,
>+                   const rtc::SocketAddress& remote_addr,
>+                   int error_code,
>+                   const char* error_desc,
>+                   const std::string& magic_cookie) {
>   RelayMessage err_msg;
>   err_msg.SetType(GetStunErrorResponseType(msg.type()));
>   err_msg.SetTransactionID(msg.transaction_id());
>@@ -82,8 +87,7 @@ void SendStunError(const StunMessage& msg, rtc::AsyncPacketSocket* socket,
> }
> 
> RelayServer::RelayServer(rtc::Thread* thread)
>-  : thread_(thread), random_(rtc::SystemTimeNanos()), log_bindings_(true) {
>-}
>+    : thread_(thread), random_(rtc::SystemTimeNanos()), log_bindings_(true) {}
> 
> RelayServer::~RelayServer() {
>   // Deleting the binding will cause it to be removed from the map.
>@@ -143,8 +147,7 @@ void RelayServer::AddInternalServerSocket(rtc::AsyncSocket* socket,
>   socket->SignalReadEvent.connect(this, &RelayServer::OnReadEvent);
> }
> 
>-void RelayServer::RemoveInternalServerSocket(
>-    rtc::AsyncSocket* socket) {
>+void RelayServer::RemoveInternalServerSocket(rtc::AsyncSocket* socket) {
>   ServerSocketMap::iterator iter = server_sockets_.find(socket);
>   RTC_DCHECK(iter != server_sockets_.end());
>   server_sockets_.erase(iter);
>@@ -182,11 +185,11 @@ void RelayServer::OnReadEvent(rtc::AsyncSocket* socket) {
>   AcceptConnection(socket);
> }
> 
>-void RelayServer::OnInternalPacket(
>-    rtc::AsyncPacketSocket* socket, const char* bytes, size_t size,
>-    const rtc::SocketAddress& remote_addr,
>-    const rtc::PacketTime& packet_time) {
>-
>+void RelayServer::OnInternalPacket(rtc::AsyncPacketSocket* socket,
>+                                   const char* bytes,
>+                                   size_t size,
>+                                   const rtc::SocketAddress& remote_addr,
>+                                   const rtc::PacketTime& packet_time) {
>   // Get the address of the connection we just received on.
>   rtc::SocketAddressPair ap(remote_addr, socket->GetLocalAddress());
>   RTC_DCHECK(!ap.destination().IsNil());
>@@ -227,11 +230,11 @@ void RelayServer::OnInternalPacket(
>   }
> }
> 
>-void RelayServer::OnExternalPacket(
>-    rtc::AsyncPacketSocket* socket, const char* bytes, size_t size,
>-    const rtc::SocketAddress& remote_addr,
>-    const rtc::PacketTime& packet_time) {
>-
>+void RelayServer::OnExternalPacket(rtc::AsyncPacketSocket* socket,
>+                                   const char* bytes,
>+                                   size_t size,
>+                                   const rtc::SocketAddress& remote_addr,
>+                                   const rtc::PacketTime& packet_time) {
>   // Get the address of the connection we just received on.
>   rtc::SocketAddressPair ap(remote_addr, socket->GetLocalAddress());
>   RTC_DCHECK(!ap.destination().IsNil());
>@@ -296,11 +299,12 @@ void RelayServer::OnExternalPacket(
>   int_conn->Send(bytes, size, ext_conn->addr_pair().source());
> }
> 
>-bool RelayServer::HandleStun(
>-    const char* bytes, size_t size, const rtc::SocketAddress& remote_addr,
>-    rtc::AsyncPacketSocket* socket, std::string* username,
>-    StunMessage* msg) {
>-
>+bool RelayServer::HandleStun(const char* bytes,
>+                             size_t size,
>+                             const rtc::SocketAddress& remote_addr,
>+                             rtc::AsyncPacketSocket* socket,
>+                             std::string* username,
>+                             StunMessage* msg) {
>   // Parse this into a stun message. Eat the message if this fails.
>   rtc::ByteBufferReader buf(bytes, size);
>   if (!msg->Read(&buf)) {
>@@ -324,10 +328,10 @@ bool RelayServer::HandleStun(
>   return true;
> }
> 
>-void RelayServer::HandleStunAllocate(
>-    const char* bytes, size_t size, const rtc::SocketAddressPair& ap,
>-    rtc::AsyncPacketSocket* socket) {
>-
>+void RelayServer::HandleStunAllocate(const char* bytes,
>+                                     size_t size,
>+                                     const rtc::SocketAddressPair& ap,
>+                                     rtc::AsyncPacketSocket* socket) {
>   // Make sure this is a valid STUN request.
>   RelayMessage request;
>   std::string username;
>@@ -336,11 +340,7 @@ void RelayServer::HandleStunAllocate(
> 
>   // Make sure this is a an allocate request.
>   if (request.type() != STUN_ALLOCATE_REQUEST) {
>-    SendStunError(request,
>-                  socket,
>-                  ap.source(),
>-                  600,
>-                  "Operation Not Supported",
>+    SendStunError(request, socket, ap.source(), 600, "Operation Not Supported",
>                   "");
>     return;
>   }
>@@ -386,9 +386,9 @@ void RelayServer::HandleStunAllocate(
>   HandleStunAllocate(int_conn, request);
> }
> 
>-void RelayServer::HandleStun(
>-    RelayServerConnection* int_conn, const char* bytes, size_t size) {
>-
>+void RelayServer::HandleStun(RelayServerConnection* int_conn,
>+                             const char* bytes,
>+                             size_t size) {
>   // Make sure this is a valid STUN request.
>   RelayMessage request;
>   std::string username;
>@@ -413,9 +413,8 @@ void RelayServer::HandleStun(
>     int_conn->SendStunError(request, 600, "Operation Not Supported");
> }
> 
>-void RelayServer::HandleStunAllocate(
>-    RelayServerConnection* int_conn, const StunMessage& request) {
>-
>+void RelayServer::HandleStunAllocate(RelayServerConnection* int_conn,
>+                                     const StunMessage& request) {
>   // Create a response message that includes an address with which external
>   // clients can communicate.
> 
>@@ -430,10 +429,9 @@ void RelayServer::HandleStunAllocate(
>   response.AddAttribute(std::move(magic_cookie_attr));
> 
>   RTC_DCHECK_GT(external_sockets_.size(), 0);
>-  size_t index = random_.Rand(rtc::dchecked_cast<uint32_t>(
>-      external_sockets_.size() - 1));
>-  rtc::SocketAddress ext_addr =
>-      external_sockets_[index]->GetLocalAddress();
>+  size_t index =
>+      random_.Rand(rtc::dchecked_cast<uint32_t>(external_sockets_.size() - 1));
>+  rtc::SocketAddress ext_addr = external_sockets_[index]->GetLocalAddress();
> 
>   auto addr_attr = StunAttribute::CreateAddress(STUN_ATTR_MAPPED_ADDRESS);
>   addr_attr->SetIP(ext_addr.ipaddr());
>@@ -452,9 +450,8 @@ void RelayServer::HandleStunAllocate(
>   int_conn->SendStun(response);
> }
> 
>-void RelayServer::HandleStunSend(
>-    RelayServerConnection* int_conn, const StunMessage& request) {
>-
>+void RelayServer::HandleStunSend(RelayServerConnection* int_conn,
>+                                 const StunMessage& request) {
>   const StunAddressAttribute* addr_attr =
>       request.GetAddress(STUN_ATTR_DESTINATION_ADDRESS);
>   if (!addr_attr) {
>@@ -533,14 +530,13 @@ void RelayServer::RemoveBinding(RelayServerBinding* binding) {
>   }
> }
> 
>-void RelayServer::OnMessage(rtc::Message *pmsg) {
>+void RelayServer::OnMessage(rtc::Message* pmsg) {
>   static const uint32_t kMessageAcceptConnection = 1;
>   RTC_DCHECK(pmsg->message_id == kMessageAcceptConnection);
> 
>   rtc::MessageData* data = pmsg->pdata;
>   rtc::AsyncSocket* socket =
>-      static_cast <rtc::TypedMessageData<rtc::AsyncSocket*>*>
>-      (data)->data();
>+      static_cast<rtc::TypedMessageData<rtc::AsyncSocket*>*>(data)->data();
>   AcceptConnection(socket);
>   delete data;
> }
>@@ -554,8 +550,7 @@ void RelayServer::OnTimeout(RelayServerBinding* binding) {
> void RelayServer::AcceptConnection(rtc::AsyncSocket* server_socket) {
>   // Check if someone is trying to connect to us.
>   rtc::SocketAddress accept_addr;
>-  rtc::AsyncSocket* accepted_socket =
>-      server_socket->Accept(&accept_addr);
>+  rtc::AsyncSocket* accepted_socket = server_socket->Accept(&accept_addr);
>   if (accepted_socket != NULL) {
>     // We had someone trying to connect, now check which protocol to
>     // use and create a packet socket.
>@@ -573,9 +568,10 @@ void RelayServer::AcceptConnection(rtc::AsyncSocket* server_socket) {
> }
> 
> RelayServerConnection::RelayServerConnection(
>-    RelayServerBinding* binding, const rtc::SocketAddressPair& addrs,
>+    RelayServerBinding* binding,
>+    const rtc::SocketAddressPair& addrs,
>     rtc::AsyncPacketSocket* socket)
>-  : binding_(binding), addr_pair_(addrs), socket_(socket), locked_(false) {
>+    : binding_(binding), addr_pair_(addrs), socket_(socket), locked_(false) {
>   // The creation of a new connection constitutes a use of the binding.
>   binding_->NoteUsed();
> }
>@@ -592,8 +588,9 @@ void RelayServerConnection::Send(const char* data, size_t size) {
>   cricket::Send(socket_, data, size, addr_pair_.source());
> }
> 
>-void RelayServerConnection::Send(
>-    const char* data, size_t size, const rtc::SocketAddress& from_addr) {
>+void RelayServerConnection::Send(const char* data,
>+                                 size_t size,
>+                                 const rtc::SocketAddress& from_addr) {
>   // If the from address is known to the client, we don't need to send it.
>   if (locked() && (from_addr == default_dest_)) {
>     Send(data, size);
>@@ -631,14 +628,14 @@ void RelayServerConnection::SendStun(const StunMessage& msg) {
>   cricket::SendStun(msg, socket_, addr_pair_.source());
> }
> 
>-void RelayServerConnection::SendStunError(
>-      const StunMessage& request, int error_code, const char* error_desc) {
>+void RelayServerConnection::SendStunError(const StunMessage& request,
>+                                          int error_code,
>+                                          const char* error_desc) {
>   // An error does not indicate use.  If no legitimate use off the binding
>   // occurs, we want it to be cleaned up even if errors are still occuring.
> 
>-  cricket::SendStunError(
>-      request, socket_, addr_pair_.source(), error_code, error_desc,
>-      binding_->magic_cookie());
>+  cricket::SendStunError(request, socket_, addr_pair_.source(), error_code,
>+                         error_desc, binding_->magic_cookie());
> }
> 
> void RelayServerConnection::Lock() {
>@@ -661,9 +658,8 @@ RelayServerBinding::RelayServerBinding(RelayServer* server,
>       password_(password),
>       lifetime_(lifetime) {
>   // For now, every connection uses the standard magic cookie value.
>-  magic_cookie_.append(
>-      reinterpret_cast<const char*>(TURN_MAGIC_COOKIE_VALUE),
>-      sizeof(TURN_MAGIC_COOKIE_VALUE));
>+  magic_cookie_.append(reinterpret_cast<const char*>(TURN_MAGIC_COOKIE_VALUE),
>+                       sizeof(TURN_MAGIC_COOKIE_VALUE));
> 
>   // Initialize the last-used time to now.
>   NoteUsed();
>@@ -709,7 +705,6 @@ bool RelayServerBinding::HasMagicCookie(const char* bytes, size_t size) const {
> 
> RelayServerConnection* RelayServerBinding::GetInternalConnection(
>     const rtc::SocketAddress& ext_addr) {
>-
>   // Look for an internal connection that is locked to this address.
>   for (size_t i = 0; i < internal_connections_.size(); ++i) {
>     if (internal_connections_[i]->locked() &&
>@@ -731,7 +726,7 @@ RelayServerConnection* RelayServerBinding::GetExternalConnection(
>   return 0;
> }
> 
>-void RelayServerBinding::OnMessage(rtc::Message *pmsg) {
>+void RelayServerBinding::OnMessage(rtc::Message* pmsg) {
>   if (pmsg->message_id == MSG_LIFETIME_TIMER) {
>     RTC_DCHECK(!pmsg->pdata);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/relayserver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/relayserver.h
>index 54ef1cb07e6..c7661204f81 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/relayserver.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/relayserver.h
>@@ -30,8 +30,7 @@ class RelayServerConnection;
> 
> // Relays traffic between connections to the server that are "bound" together.
> // All connections created with the same username/password are bound together.
>-class RelayServer : public rtc::MessageHandler,
>-                    public sigslot::has_slots<> {
>+class RelayServer : public rtc::MessageHandler, public sigslot::has_slots<> {
>  public:
>   // Creates a server, which will use this thread to post messages to itself.
>   explicit RelayServer(rtc::Thread* thread);
>@@ -70,11 +69,10 @@ class RelayServer : public rtc::MessageHandler,
> 
>  private:
>   typedef std::vector<rtc::AsyncPacketSocket*> SocketList;
>-  typedef std::map<rtc::AsyncSocket*,
>-                   cricket::ProtocolType> ServerSocketMap;
>+  typedef std::map<rtc::AsyncSocket*, cricket::ProtocolType> ServerSocketMap;
>   typedef std::map<std::string, RelayServerBinding*> BindingMap;
>-  typedef std::map<rtc::SocketAddressPair,
>-                   RelayServerConnection*> ConnectionMap;
>+  typedef std::map<rtc::SocketAddressPair, RelayServerConnection*>
>+      ConnectionMap;
> 
>   rtc::Thread* thread_;
>   webrtc::Random random_;
>@@ -88,25 +86,31 @@ class RelayServer : public rtc::MessageHandler,
> 
>   // Called when a packet is received by the server on one of its sockets.
>   void OnInternalPacket(rtc::AsyncPacketSocket* socket,
>-                        const char* bytes, size_t size,
>+                        const char* bytes,
>+                        size_t size,
>                         const rtc::SocketAddress& remote_addr,
>                         const rtc::PacketTime& packet_time);
>   void OnExternalPacket(rtc::AsyncPacketSocket* socket,
>-                        const char* bytes, size_t size,
>+                        const char* bytes,
>+                        size_t size,
>                         const rtc::SocketAddress& remote_addr,
>                         const rtc::PacketTime& packet_time);
> 
>   void OnReadEvent(rtc::AsyncSocket* socket);
> 
>   // Processes the relevant STUN request types from the client.
>-  bool HandleStun(const char* bytes, size_t size,
>+  bool HandleStun(const char* bytes,
>+                  size_t size,
>                   const rtc::SocketAddress& remote_addr,
>                   rtc::AsyncPacketSocket* socket,
>-                  std::string* username, StunMessage* msg);
>-  void HandleStunAllocate(const char* bytes, size_t size,
>+                  std::string* username,
>+                  StunMessage* msg);
>+  void HandleStunAllocate(const char* bytes,
>+                          size_t size,
>                           const rtc::SocketAddressPair& ap,
>                           rtc::AsyncPacketSocket* socket);
>-  void HandleStun(RelayServerConnection* int_conn, const char* bytes,
>+  void HandleStun(RelayServerConnection* int_conn,
>+                  const char* bytes,
>                   size_t size);
>   void HandleStunAllocate(RelayServerConnection* int_conn,
>                           const StunMessage& msg);
>@@ -149,8 +153,7 @@ class RelayServerConnection {
>   // Sends a packet to the connected client.  If an address is provided, then
>   // we make sure the internal client receives it, wrapping if necessary.
>   void Send(const char* data, size_t size);
>-  void Send(const char* data, size_t size,
>-            const rtc::SocketAddress& ext_addr);
>+  void Send(const char* data, size_t size, const rtc::SocketAddress& ext_addr);
> 
>   // Sends a STUN message to the connected client with no wrapping.
>   void SendStun(const StunMessage& msg);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/relayserver_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/relayserver_unittest.cc
>index 80240166692..c4c026a241b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/relayserver_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/relayserver_unittest.cc
>@@ -12,11 +12,11 @@
> #include <string>
> #include <utility>
> 
>+#include "absl/memory/memory.h"
> #include "p2p/base/relayserver.h"
> #include "rtc_base/gunit.h"
> #include "rtc_base/helpers.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/socketaddress.h"
> #include "rtc_base/ssladapter.h"
> #include "rtc_base/testclient.h"
>@@ -59,10 +59,10 @@ class RelayServerTest : public testing::Test {
>     server_->AddExternalSocket(
>         rtc::AsyncUDPSocket::Create(ss_.get(), server_ext_addr));
> 
>-    client1_.reset(new rtc::TestClient(
>-        WrapUnique(rtc::AsyncUDPSocket::Create(ss_.get(), client1_addr))));
>-    client2_.reset(new rtc::TestClient(
>-        WrapUnique(rtc::AsyncUDPSocket::Create(ss_.get(), client2_addr))));
>+    client1_.reset(new rtc::TestClient(absl::WrapUnique(
>+        rtc::AsyncUDPSocket::Create(ss_.get(), client1_addr))));
>+    client2_.reset(new rtc::TestClient(absl::WrapUnique(
>+        rtc::AsyncUDPSocket::Create(ss_.get(), client2_addr))));
>   }
> 
>   void Allocate() {
>@@ -95,30 +95,20 @@ class RelayServerTest : public testing::Test {
>   void SendRaw2(const char* data, int len) {
>     return Send(client2_.get(), data, len, server_ext_addr);
>   }
>-  void Send(rtc::TestClient* client, const char* data,
>-            int len, const SocketAddress& addr) {
>+  void Send(rtc::TestClient* client,
>+            const char* data,
>+            int len,
>+            const SocketAddress& addr) {
>     client->SendTo(data, len, addr);
>   }
> 
>-  bool Receive1Fails() {
>-    return client1_.get()->CheckNoPacket();
>-  }
>-  bool Receive2Fails() {
>-    return client2_.get()->CheckNoPacket();
>-  }
>+  bool Receive1Fails() { return client1_.get()->CheckNoPacket(); }
>+  bool Receive2Fails() { return client2_.get()->CheckNoPacket(); }
> 
>-  StunMessage* Receive1() {
>-    return Receive(client1_.get());
>-  }
>-  StunMessage* Receive2() {
>-    return Receive(client2_.get());
>-  }
>-  std::string ReceiveRaw1() {
>-    return ReceiveRaw(client1_.get());
>-  }
>-  std::string ReceiveRaw2() {
>-    return ReceiveRaw(client2_.get());
>-  }
>+  StunMessage* Receive1() { return Receive(client1_.get()); }
>+  StunMessage* Receive2() { return Receive(client2_.get()); }
>+  std::string ReceiveRaw1() { return ReceiveRaw(client1_.get()); }
>+  std::string ReceiveRaw2() { return ReceiveRaw(client2_.get()); }
>   StunMessage* Receive(rtc::TestClient* client) {
>     StunMessage* msg = NULL;
>     std::unique_ptr<rtc::TestClient::Packet> packet =
>@@ -144,8 +134,7 @@ class RelayServerTest : public testing::Test {
>   static StunMessage* CreateStunMessage(int type) {
>     StunMessage* msg = new RelayMessage();
>     msg->SetType(type);
>-    msg->SetTransactionID(
>-        rtc::CreateRandomString(kStunTransactionIdLength));
>+    msg->SetTransactionID(rtc::CreateRandomString(kStunTransactionIdLength));
>     return msg;
>   }
>   static void AddMagicCookieAttr(StunMessage* msg) {
>@@ -275,8 +264,7 @@ TEST_F(RelayServerTest, TestReallocate) {
>   EXPECT_EQ(server_ext_addr.port(), mapped_addr->port());
>   EXPECT_EQ(server_ext_addr.ipaddr(), mapped_addr->ipaddr());
> 
>-  const StunUInt32Attribute* lifetime_attr =
>-      res->GetUInt32(STUN_ATTR_LIFETIME);
>+  const StunUInt32Attribute* lifetime_attr = res->GetUInt32(STUN_ATTR_LIFETIME);
>   ASSERT_TRUE(lifetime_attr != NULL);
>   EXPECT_EQ(LIFETIME, lifetime_attr->value());
> }
>@@ -296,8 +284,7 @@ TEST_F(RelayServerTest, TestRemoteBind) {
>   ASSERT_TRUE(res);
>   EXPECT_EQ(STUN_DATA_INDICATION, res->type());
> 
>-  const StunByteStringAttribute* recv_data =
>-      res->GetByteString(STUN_ATTR_DATA);
>+  const StunByteStringAttribute* recv_data = res->GetByteString(STUN_ATTR_DATA);
>   ASSERT_TRUE(recv_data != NULL);
> 
>   rtc::ByteBufferReader buf(recv_data->bytes(), recv_data->length());
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stun.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stun.cc
>index 6e8f0982687..cc9fb4e4c09 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stun.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stun.cc
>@@ -12,15 +12,16 @@
> 
> #include <string.h>
> 
>+#include <algorithm>
> #include <memory>
> #include <utility>
> 
>+#include "absl/memory/memory.h"
> #include "rtc_base/byteorder.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/crc32.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/messagedigest.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/stringencode.h"
> 
> using rtc::ByteBufferReader;
>@@ -40,7 +41,7 @@ const char STUN_ERROR_REASON_UNSUPPORTED_PROTOCOL[] = "Unsupported Protocol";
> const char STUN_ERROR_REASON_ROLE_CONFLICT[] = "Role Conflict";
> const char STUN_ERROR_REASON_SERVER_ERROR[] = "Server Error";
> 
>-const char TURN_MAGIC_COOKIE_VALUE[] = { '\x72', '\xC6', '\x4B', '\xC6' };
>+const char TURN_MAGIC_COOKIE_VALUE[] = {'\x72', '\xC6', '\x4B', '\xC6'};
> const char EMPTY_TRANSACTION_ID[] = "0000000000000000";
> const uint32_t STUN_FINGERPRINT_XOR_VALUE = 0x5354554E;
> 
>@@ -49,7 +50,8 @@ const uint32_t STUN_FINGERPRINT_XOR_VALUE = 0x5354554E;
> StunMessage::StunMessage()
>     : type_(0),
>       length_(0),
>-      transaction_id_(EMPTY_TRANSACTION_ID) {
>+      transaction_id_(EMPTY_TRANSACTION_ID),
>+      stun_magic_cookie_(kStunMagicCookie) {
>   RTC_DCHECK(IsValidTransactionId(transaction_id_));
> }
> 
>@@ -70,15 +72,16 @@ bool StunMessage::SetTransactionID(const std::string& str) {
>   return true;
> }
> 
>-static bool ImplementationDefinedRange(int attr_type) {
>-  return attr_type >= 0xC000 && attr_type <= 0xFFFF;
>+static bool DesignatedExpertRange(int attr_type) {
>+  return (attr_type >= 0x4000 && attr_type <= 0x7FFF) ||
>+         (attr_type >= 0xC000 && attr_type <= 0xFFFF);
> }
> 
> void StunMessage::AddAttribute(std::unique_ptr<StunAttribute> attr) {
>   // Fail any attributes that aren't valid for this type of message,
>-  // but allow any type for the range that is "implementation defined"
>-  // in the RFC.
>-  if (!ImplementationDefinedRange(attr->type())) {
>+  // but allow any type for the range that in the RFC is reserved for
>+  // the "designated experts".
>+  if (!DesignatedExpertRange(attr->type())) {
>     RTC_DCHECK_EQ(attr->value_type(), GetAttributeValueType(attr->type()));
>   }
> 
>@@ -95,8 +98,8 @@ void StunMessage::AddAttribute(std::unique_ptr<StunAttribute> attr) {
> std::unique_ptr<StunAttribute> StunMessage::RemoveAttribute(int type) {
>   std::unique_ptr<StunAttribute> attribute;
>   for (auto it = attrs_.rbegin(); it != attrs_.rend(); ++it) {
>-    if ((* it)->type() == type) {
>-      attribute = std::move(* it);
>+    if ((*it)->type() == type) {
>+      attribute = std::move(*it);
>       attrs_.erase(std::next(it).base());
>       break;
>     }
>@@ -158,7 +161,8 @@ const StunUInt16ListAttribute* StunMessage::GetUnknownAttributes() const {
> 
> // Verifies a STUN message has a valid MESSAGE-INTEGRITY attribute, using the
> // procedure outlined in RFC 5389, section 15.4.
>-bool StunMessage::ValidateMessageIntegrity(const char* data, size_t size,
>+bool StunMessage::ValidateMessageIntegrity(const char* data,
>+                                           size_t size,
>                                            const std::string& password) {
>   // Verifying the size of the message.
>   if ((size % 4) != 0 || size < kStunHeaderSize) {
>@@ -209,8 +213,8 @@ bool StunMessage::ValidateMessageIntegrity(const char* data, size_t size,
>   if (size > mi_pos + kStunAttributeHeaderSize + kStunMessageIntegritySize) {
>     // Stun message has other attributes after message integrity.
>     // Adjust the length parameter in stun message to calculate HMAC.
>-    size_t extra_offset = size -
>-        (mi_pos + kStunAttributeHeaderSize + kStunMessageIntegritySize);
>+    size_t extra_offset =
>+        size - (mi_pos + kStunAttributeHeaderSize + kStunMessageIntegritySize);
>     size_t new_adjusted_len = size - extra_offset - kStunHeaderSize;
> 
>     // Writing new length of the STUN message @ Message Length in temp buffer.
>@@ -223,17 +227,15 @@ bool StunMessage::ValidateMessageIntegrity(const char* data, size_t size,
>   }
> 
>   char hmac[kStunMessageIntegritySize];
>-  size_t ret = rtc::ComputeHmac(rtc::DIGEST_SHA_1,
>-                                      password.c_str(), password.size(),
>-                                      temp_data.get(), mi_pos,
>-                                      hmac, sizeof(hmac));
>+  size_t ret =
>+      rtc::ComputeHmac(rtc::DIGEST_SHA_1, password.c_str(), password.size(),
>+                       temp_data.get(), mi_pos, hmac, sizeof(hmac));
>   RTC_DCHECK(ret == sizeof(hmac));
>   if (ret != sizeof(hmac))
>     return false;
> 
>   // Comparing the calculated HMAC with the one present in the message.
>-  return memcmp(data + current_pos + kStunAttributeHeaderSize,
>-                hmac,
>+  return memcmp(data + current_pos + kStunAttributeHeaderSize, hmac,
>                 sizeof(hmac)) == 0;
> }
> 
>@@ -241,11 +243,10 @@ bool StunMessage::AddMessageIntegrity(const std::string& password) {
>   return AddMessageIntegrity(password.c_str(), password.size());
> }
> 
>-bool StunMessage::AddMessageIntegrity(const char* key,
>-                                      size_t keylen) {
>+bool StunMessage::AddMessageIntegrity(const char* key, size_t keylen) {
>   // Add the attribute with a dummy value. Since this is a known attribute, it
>   // can't fail.
>-  auto msg_integrity_attr_ptr = rtc::MakeUnique<StunByteStringAttribute>(
>+  auto msg_integrity_attr_ptr = absl::make_unique<StunByteStringAttribute>(
>       STUN_ATTR_MESSAGE_INTEGRITY, std::string(kStunMessageIntegritySize, '0'));
>   auto* msg_integrity_attr = msg_integrity_attr_ptr.get();
>   AddAttribute(std::move(msg_integrity_attr_ptr));
>@@ -258,14 +259,12 @@ bool StunMessage::AddMessageIntegrity(const char* key,
>   int msg_len_for_hmac = static_cast<int>(
>       buf.Length() - kStunAttributeHeaderSize - msg_integrity_attr->length());
>   char hmac[kStunMessageIntegritySize];
>-  size_t ret = rtc::ComputeHmac(rtc::DIGEST_SHA_1,
>-                                      key, keylen,
>-                                      buf.Data(), msg_len_for_hmac,
>-                                      hmac, sizeof(hmac));
>+  size_t ret = rtc::ComputeHmac(rtc::DIGEST_SHA_1, key, keylen, buf.Data(),
>+                                msg_len_for_hmac, hmac, sizeof(hmac));
>   RTC_DCHECK(ret == sizeof(hmac));
>   if (ret != sizeof(hmac)) {
>     RTC_LOG(LS_ERROR) << "HMAC computation failed. Message-Integrity "
>-                      << "has dummy value.";
>+                         "has dummy value.";
>     return false;
>   }
> 
>@@ -301,14 +300,14 @@ bool StunMessage::ValidateFingerprint(const char* data, size_t size) {
>   uint32_t fingerprint =
>       rtc::GetBE32(fingerprint_attr_data + kStunAttributeHeaderSize);
>   return ((fingerprint ^ STUN_FINGERPRINT_XOR_VALUE) ==
>-      rtc::ComputeCrc32(data, size - fingerprint_attr_size));
>+          rtc::ComputeCrc32(data, size - fingerprint_attr_size));
> }
> 
> bool StunMessage::AddFingerprint() {
>   // Add the attribute with a dummy value. Since this is a known attribute,
>   // it can't fail.
>   auto fingerprint_attr_ptr =
>-      rtc::MakeUnique<StunUInt32Attribute>(STUN_ATTR_FINGERPRINT, 0);
>+      absl::make_unique<StunUInt32Attribute>(STUN_ATTR_FINGERPRINT, 0);
>   auto* fingerprint_attr = fingerprint_attr_ptr.get();
>   AddAttribute(std::move(fingerprint_attr_ptr));
> 
>@@ -394,7 +393,7 @@ bool StunMessage::Write(ByteBufferWriter* buf) const {
>   buf->WriteUInt16(type_);
>   buf->WriteUInt16(length_);
>   if (!IsLegacy())
>-    buf->WriteUInt32(kStunMagicCookie);
>+    buf->WriteUInt32(stun_magic_cookie_);
>   buf->WriteString(transaction_id_);
> 
>   for (const auto& attr : attrs_) {
>@@ -412,22 +411,40 @@ StunMessage* StunMessage::CreateNew() const {
>   return new StunMessage();
> }
> 
>+void StunMessage::SetStunMagicCookie(uint32_t val) {
>+  stun_magic_cookie_ = val;
>+}
>+
> StunAttributeValueType StunMessage::GetAttributeValueType(int type) const {
>   switch (type) {
>-    case STUN_ATTR_MAPPED_ADDRESS:      return STUN_VALUE_ADDRESS;
>-    case STUN_ATTR_USERNAME:            return STUN_VALUE_BYTE_STRING;
>-    case STUN_ATTR_MESSAGE_INTEGRITY:   return STUN_VALUE_BYTE_STRING;
>-    case STUN_ATTR_ERROR_CODE:          return STUN_VALUE_ERROR_CODE;
>-    case STUN_ATTR_UNKNOWN_ATTRIBUTES:  return STUN_VALUE_UINT16_LIST;
>-    case STUN_ATTR_REALM:               return STUN_VALUE_BYTE_STRING;
>-    case STUN_ATTR_NONCE:               return STUN_VALUE_BYTE_STRING;
>-    case STUN_ATTR_XOR_MAPPED_ADDRESS:  return STUN_VALUE_XOR_ADDRESS;
>-    case STUN_ATTR_SOFTWARE:            return STUN_VALUE_BYTE_STRING;
>-    case STUN_ATTR_ALTERNATE_SERVER:    return STUN_VALUE_ADDRESS;
>-    case STUN_ATTR_FINGERPRINT:         return STUN_VALUE_UINT32;
>-    case STUN_ATTR_ORIGIN:              return STUN_VALUE_BYTE_STRING;
>-    case STUN_ATTR_RETRANSMIT_COUNT:    return STUN_VALUE_UINT32;
>-    default:                            return STUN_VALUE_UNKNOWN;
>+    case STUN_ATTR_MAPPED_ADDRESS:
>+      return STUN_VALUE_ADDRESS;
>+    case STUN_ATTR_USERNAME:
>+      return STUN_VALUE_BYTE_STRING;
>+    case STUN_ATTR_MESSAGE_INTEGRITY:
>+      return STUN_VALUE_BYTE_STRING;
>+    case STUN_ATTR_ERROR_CODE:
>+      return STUN_VALUE_ERROR_CODE;
>+    case STUN_ATTR_UNKNOWN_ATTRIBUTES:
>+      return STUN_VALUE_UINT16_LIST;
>+    case STUN_ATTR_REALM:
>+      return STUN_VALUE_BYTE_STRING;
>+    case STUN_ATTR_NONCE:
>+      return STUN_VALUE_BYTE_STRING;
>+    case STUN_ATTR_XOR_MAPPED_ADDRESS:
>+      return STUN_VALUE_XOR_ADDRESS;
>+    case STUN_ATTR_SOFTWARE:
>+      return STUN_VALUE_BYTE_STRING;
>+    case STUN_ATTR_ALTERNATE_SERVER:
>+      return STUN_VALUE_ADDRESS;
>+    case STUN_ATTR_FINGERPRINT:
>+      return STUN_VALUE_UINT32;
>+    case STUN_ATTR_ORIGIN:
>+      return STUN_VALUE_BYTE_STRING;
>+    case STUN_ATTR_RETRANSMIT_COUNT:
>+      return STUN_VALUE_UINT32;
>+    default:
>+      return STUN_VALUE_UNKNOWN;
>   }
> }
> 
>@@ -436,7 +453,7 @@ StunAttribute* StunMessage::CreateAttribute(int type, size_t length) /*const*/ {
>   if (value_type != STUN_VALUE_UNKNOWN) {
>     return StunAttribute::Create(value_type, type,
>                                  static_cast<uint16_t>(length), this);
>-  } else if (ImplementationDefinedRange(type)) {
>+  } else if (DesignatedExpertRange(type)) {
>     // Read unknown attributes as STUN_VALUE_BYTE_STRING
>     return StunAttribute::Create(STUN_VALUE_BYTE_STRING, type,
>                                  static_cast<uint16_t>(length), this);
>@@ -456,14 +473,13 @@ const StunAttribute* StunMessage::GetAttribute(int type) const {
> 
> bool StunMessage::IsValidTransactionId(const std::string& transaction_id) {
>   return transaction_id.size() == kStunTransactionIdLength ||
>-      transaction_id.size() == kStunLegacyTransactionIdLength;
>+         transaction_id.size() == kStunLegacyTransactionIdLength;
> }
> 
> // StunAttribute
> 
> StunAttribute::StunAttribute(uint16_t type, uint16_t length)
>-    : type_(type), length_(length) {
>-}
>+    : type_(type), length_(length) {}
> 
> void StunAttribute::ConsumePadding(ByteBufferReader* buf) const {
>   int remainder = length_ % 4;
>@@ -506,38 +522,38 @@ StunAttribute* StunAttribute::Create(StunAttributeValueType value_type,
> 
> std::unique_ptr<StunAddressAttribute> StunAttribute::CreateAddress(
>     uint16_t type) {
>-  return rtc::MakeUnique<StunAddressAttribute>(type, 0);
>+  return absl::make_unique<StunAddressAttribute>(type, 0);
> }
> 
> std::unique_ptr<StunXorAddressAttribute> StunAttribute::CreateXorAddress(
>     uint16_t type) {
>-  return rtc::MakeUnique<StunXorAddressAttribute>(type, 0, nullptr);
>+  return absl::make_unique<StunXorAddressAttribute>(type, 0, nullptr);
> }
> 
> std::unique_ptr<StunUInt64Attribute> StunAttribute::CreateUInt64(
>     uint16_t type) {
>-  return rtc::MakeUnique<StunUInt64Attribute>(type);
>+  return absl::make_unique<StunUInt64Attribute>(type);
> }
> 
> std::unique_ptr<StunUInt32Attribute> StunAttribute::CreateUInt32(
>     uint16_t type) {
>-  return rtc::MakeUnique<StunUInt32Attribute>(type);
>+  return absl::make_unique<StunUInt32Attribute>(type);
> }
> 
> std::unique_ptr<StunByteStringAttribute> StunAttribute::CreateByteString(
>     uint16_t type) {
>-  return rtc::MakeUnique<StunByteStringAttribute>(type, 0);
>+  return absl::make_unique<StunByteStringAttribute>(type, 0);
> }
> 
> std::unique_ptr<StunErrorCodeAttribute> StunAttribute::CreateErrorCode() {
>-  return rtc::MakeUnique<StunErrorCodeAttribute>(
>+  return absl::make_unique<StunErrorCodeAttribute>(
>       STUN_ATTR_ERROR_CODE, StunErrorCodeAttribute::MIN_SIZE);
> }
> 
> std::unique_ptr<StunUInt16ListAttribute>
> StunAttribute::CreateUnknownAttributes() {
>-  return rtc::MakeUnique<StunUInt16ListAttribute>(STUN_ATTR_UNKNOWN_ATTRIBUTES,
>-                                                  0);
>+  return absl::make_unique<StunUInt16ListAttribute>(
>+      STUN_ATTR_UNKNOWN_ATTRIBUTES, 0);
> }
> 
> StunAddressAttribute::StunAddressAttribute(uint16_t type,
>@@ -547,8 +563,7 @@ StunAddressAttribute::StunAddressAttribute(uint16_t type,
> }
> 
> StunAddressAttribute::StunAddressAttribute(uint16_t type, uint16_t length)
>-    : StunAttribute(type, length) {
>-}
>+    : StunAttribute(type, length) {}
> 
> StunAttributeValueType StunAddressAttribute::value_type() const {
>   return STUN_VALUE_ADDRESS;
>@@ -618,14 +633,12 @@ bool StunAddressAttribute::Write(ByteBufferWriter* buf) const {
> 
> StunXorAddressAttribute::StunXorAddressAttribute(uint16_t type,
>                                                  const rtc::SocketAddress& addr)
>-    : StunAddressAttribute(type, addr), owner_(NULL) {
>-}
>+    : StunAddressAttribute(type, addr), owner_(NULL) {}
> 
> StunXorAddressAttribute::StunXorAddressAttribute(uint16_t type,
>                                                  uint16_t length,
>                                                  StunMessage* owner)
>-    : StunAddressAttribute(type, length), owner_(owner) {
>-}
>+    : StunAddressAttribute(type, length), owner_(owner) {}
> 
> StunAttributeValueType StunXorAddressAttribute::value_type() const {
>   return STUN_VALUE_XOR_ADDRESS;
>@@ -709,12 +722,10 @@ bool StunXorAddressAttribute::Write(ByteBufferWriter* buf) const {
> }
> 
> StunUInt32Attribute::StunUInt32Attribute(uint16_t type, uint32_t value)
>-    : StunAttribute(type, SIZE), bits_(value) {
>-}
>+    : StunAttribute(type, SIZE), bits_(value) {}
> 
> StunUInt32Attribute::StunUInt32Attribute(uint16_t type)
>-    : StunAttribute(type, SIZE), bits_(0) {
>-}
>+    : StunAttribute(type, SIZE), bits_(0) {}
> 
> StunAttributeValueType StunUInt32Attribute::value_type() const {
>   return STUN_VALUE_UINT32;
>@@ -743,12 +754,10 @@ bool StunUInt32Attribute::Write(ByteBufferWriter* buf) const {
> }
> 
> StunUInt64Attribute::StunUInt64Attribute(uint16_t type, uint64_t value)
>-    : StunAttribute(type, SIZE), bits_(value) {
>-}
>+    : StunAttribute(type, SIZE), bits_(value) {}
> 
> StunUInt64Attribute::StunUInt64Attribute(uint16_t type)
>-    : StunAttribute(type, SIZE), bits_(0) {
>-}
>+    : StunAttribute(type, SIZE), bits_(0) {}
> 
> StunAttributeValueType StunUInt64Attribute::value_type() const {
>   return STUN_VALUE_UINT64;
>@@ -766,8 +775,7 @@ bool StunUInt64Attribute::Write(ByteBufferWriter* buf) const {
> }
> 
> StunByteStringAttribute::StunByteStringAttribute(uint16_t type)
>-    : StunAttribute(type, 0), bytes_(NULL) {
>-}
>+    : StunAttribute(type, 0), bytes_(NULL) {}
> 
> StunByteStringAttribute::StunByteStringAttribute(uint16_t type,
>                                                  const std::string& str)
>@@ -783,11 +791,10 @@ StunByteStringAttribute::StunByteStringAttribute(uint16_t type,
> }
> 
> StunByteStringAttribute::StunByteStringAttribute(uint16_t type, uint16_t length)
>-    : StunAttribute(type, length), bytes_(NULL) {
>-}
>+    : StunAttribute(type, length), bytes_(NULL) {}
> 
> StunByteStringAttribute::~StunByteStringAttribute() {
>-  delete [] bytes_;
>+  delete[] bytes_;
> }
> 
> StunAttributeValueType StunByteStringAttribute::value_type() const {
>@@ -833,7 +840,7 @@ bool StunByteStringAttribute::Write(ByteBufferWriter* buf) const {
> }
> 
> void StunByteStringAttribute::SetBytes(char* bytes, size_t length) {
>-  delete [] bytes_;
>+  delete[] bytes_;
>   bytes_ = bytes;
>   SetLength(static_cast<uint16_t>(length));
> }
>@@ -849,11 +856,9 @@ StunErrorCodeAttribute::StunErrorCodeAttribute(uint16_t type,
> }
> 
> StunErrorCodeAttribute::StunErrorCodeAttribute(uint16_t type, uint16_t length)
>-    : StunAttribute(type, length), class_(0), number_(0) {
>-}
>+    : StunAttribute(type, length), class_(0), number_(0) {}
> 
>-StunErrorCodeAttribute::~StunErrorCodeAttribute() {
>-}
>+StunErrorCodeAttribute::~StunErrorCodeAttribute() {}
> 
> StunAttributeValueType StunErrorCodeAttribute::value_type() const {
>   return STUN_VALUE_ERROR_CODE;
>@@ -993,9 +998,8 @@ bool ComputeStunCredentialHash(const std::string& username,
>   input += password;
> 
>   char digest[rtc::MessageDigest::kMaxSize];
>-  size_t size = rtc::ComputeDigest(
>-      rtc::DIGEST_MD5, input.c_str(), input.size(),
>-      digest, sizeof(digest));
>+  size_t size = rtc::ComputeDigest(rtc::DIGEST_MD5, input.c_str(), input.size(),
>+                                   digest, sizeof(digest));
>   if (size == 0) {
>     return false;
>   }
>@@ -1012,11 +1016,9 @@ std::unique_ptr<StunAttribute> CopyStunAttribute(
>     tmp_buffer_ptr = &tmpBuffer;
>   }
> 
>-  std::unique_ptr<StunAttribute> copy(
>-      StunAttribute::Create(attribute.value_type(),
>-                            attribute.type(),
>-                            static_cast<uint16_t>(attribute.length()),
>-                            nullptr));
>+  std::unique_ptr<StunAttribute> copy(StunAttribute::Create(
>+      attribute.value_type(), attribute.type(),
>+      static_cast<uint16_t>(attribute.length()), nullptr));
> 
>   if (!copy) {
>     return nullptr;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stun.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stun.h
>index b455bb0f99a..dfb56e0c0ec 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stun.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stun.h
>@@ -18,7 +18,6 @@
> #include <string>
> #include <vector>
> 
>-#include "rtc_base/basictypes.h"
> #include "rtc_base/bytebuffer.h"
> #include "rtc_base/socketaddress.h"
> 
>@@ -26,10 +25,10 @@ namespace cricket {
> 
> // These are the types of STUN messages defined in RFC 5389.
> enum StunMessageType {
>-  STUN_BINDING_REQUEST                  = 0x0001,
>-  STUN_BINDING_INDICATION               = 0x0011,
>-  STUN_BINDING_RESPONSE                 = 0x0101,
>-  STUN_BINDING_ERROR_RESPONSE           = 0x0111,
>+  STUN_BINDING_REQUEST = 0x0001,
>+  STUN_BINDING_INDICATION = 0x0011,
>+  STUN_BINDING_RESPONSE = 0x0101,
>+  STUN_BINDING_ERROR_RESPONSE = 0x0111,
> };
> 
> // These are all known STUN attributes, defined in RFC 5389 and elsewhere.
>@@ -38,19 +37,19 @@ enum StunMessageType {
> // RETRANSMIT_COUNT is the number of outstanding pings without a response at
> // the time the packet is generated.
> enum StunAttributeType {
>-  STUN_ATTR_MAPPED_ADDRESS              = 0x0001,  // Address
>-  STUN_ATTR_USERNAME                    = 0x0006,  // ByteString
>-  STUN_ATTR_MESSAGE_INTEGRITY           = 0x0008,  // ByteString, 20 bytes
>-  STUN_ATTR_ERROR_CODE                  = 0x0009,  // ErrorCode
>-  STUN_ATTR_UNKNOWN_ATTRIBUTES          = 0x000a,  // UInt16List
>-  STUN_ATTR_REALM                       = 0x0014,  // ByteString
>-  STUN_ATTR_NONCE                       = 0x0015,  // ByteString
>-  STUN_ATTR_XOR_MAPPED_ADDRESS          = 0x0020,  // XorAddress
>-  STUN_ATTR_SOFTWARE                    = 0x8022,  // ByteString
>-  STUN_ATTR_ALTERNATE_SERVER            = 0x8023,  // Address
>-  STUN_ATTR_FINGERPRINT                 = 0x8028,  // UInt32
>-  STUN_ATTR_ORIGIN                      = 0x802F,  // ByteString
>-  STUN_ATTR_RETRANSMIT_COUNT            = 0xFF00   // UInt32
>+  STUN_ATTR_MAPPED_ADDRESS = 0x0001,      // Address
>+  STUN_ATTR_USERNAME = 0x0006,            // ByteString
>+  STUN_ATTR_MESSAGE_INTEGRITY = 0x0008,   // ByteString, 20 bytes
>+  STUN_ATTR_ERROR_CODE = 0x0009,          // ErrorCode
>+  STUN_ATTR_UNKNOWN_ATTRIBUTES = 0x000a,  // UInt16List
>+  STUN_ATTR_REALM = 0x0014,               // ByteString
>+  STUN_ATTR_NONCE = 0x0015,               // ByteString
>+  STUN_ATTR_XOR_MAPPED_ADDRESS = 0x0020,  // XorAddress
>+  STUN_ATTR_SOFTWARE = 0x8022,            // ByteString
>+  STUN_ATTR_ALTERNATE_SERVER = 0x8023,    // Address
>+  STUN_ATTR_FINGERPRINT = 0x8028,         // UInt32
>+  STUN_ATTR_ORIGIN = 0x802F,              // ByteString
>+  STUN_ATTR_RETRANSMIT_COUNT = 0xFF00     // UInt32
> };
> 
> // These are the types of the values associated with the attributes above.
>@@ -58,34 +57,34 @@ enum StunAttributeType {
> // attributes. Note that these values are for our own use, and not defined in
> // RFC 5389.
> enum StunAttributeValueType {
>-  STUN_VALUE_UNKNOWN                    = 0,
>-  STUN_VALUE_ADDRESS                    = 1,
>-  STUN_VALUE_XOR_ADDRESS                = 2,
>-  STUN_VALUE_UINT32                     = 3,
>-  STUN_VALUE_UINT64                     = 4,
>-  STUN_VALUE_BYTE_STRING                = 5,
>-  STUN_VALUE_ERROR_CODE                 = 6,
>-  STUN_VALUE_UINT16_LIST                = 7
>+  STUN_VALUE_UNKNOWN = 0,
>+  STUN_VALUE_ADDRESS = 1,
>+  STUN_VALUE_XOR_ADDRESS = 2,
>+  STUN_VALUE_UINT32 = 3,
>+  STUN_VALUE_UINT64 = 4,
>+  STUN_VALUE_BYTE_STRING = 5,
>+  STUN_VALUE_ERROR_CODE = 6,
>+  STUN_VALUE_UINT16_LIST = 7
> };
> 
> // These are the types of STUN addresses defined in RFC 5389.
> enum StunAddressFamily {
>   // NB: UNDEF is not part of the STUN spec.
>-  STUN_ADDRESS_UNDEF                    = 0,
>-  STUN_ADDRESS_IPV4                     = 1,
>-  STUN_ADDRESS_IPV6                     = 2
>+  STUN_ADDRESS_UNDEF = 0,
>+  STUN_ADDRESS_IPV4 = 1,
>+  STUN_ADDRESS_IPV6 = 2
> };
> 
> // These are the types of STUN error codes defined in RFC 5389.
> enum StunErrorCode {
>-  STUN_ERROR_TRY_ALTERNATE              = 300,
>-  STUN_ERROR_BAD_REQUEST                = 400,
>-  STUN_ERROR_UNAUTHORIZED               = 401,
>-  STUN_ERROR_UNKNOWN_ATTRIBUTE          = 420,
>-  STUN_ERROR_STALE_CREDENTIALS          = 430,  // GICE only
>-  STUN_ERROR_STALE_NONCE                = 438,
>-  STUN_ERROR_SERVER_ERROR               = 500,
>-  STUN_ERROR_GLOBAL_FAILURE             = 600
>+  STUN_ERROR_TRY_ALTERNATE = 300,
>+  STUN_ERROR_BAD_REQUEST = 400,
>+  STUN_ERROR_UNAUTHORIZED = 401,
>+  STUN_ERROR_UNKNOWN_ATTRIBUTE = 420,
>+  STUN_ERROR_STALE_CREDENTIALS = 430,  // GICE only
>+  STUN_ERROR_STALE_NONCE = 438,
>+  STUN_ERROR_SERVER_ERROR = 500,
>+  STUN_ERROR_GLOBAL_FAILURE = 600
> };
> 
> // Strings for the error codes above.
>@@ -171,7 +170,8 @@ class StunMessage {
>   // Validates that a raw STUN message has a correct MESSAGE-INTEGRITY value.
>   // This can't currently be done on a StunMessage, since it is affected by
>   // padding data (which we discard when reading a StunMessage).
>-  static bool ValidateMessageIntegrity(const char* data, size_t size,
>+  static bool ValidateMessageIntegrity(const char* data,
>+                                       size_t size,
>                                        const std::string& password);
>   // Adds a MESSAGE-INTEGRITY attribute that is valid for the current message.
>   bool AddMessageIntegrity(const std::string& password);
>@@ -194,6 +194,10 @@ class StunMessage {
>   // Creates an empty message. Overridable by derived classes.
>   virtual StunMessage* CreateNew() const;
> 
>+  // Modify the stun magic cookie used for this STUN message.
>+  // This is used for testing.
>+  void SetStunMagicCookie(uint32_t val);
>+
>  protected:
>   // Verifies that the given attribute is allowed for this message.
>   virtual StunAttributeValueType GetAttributeValueType(int type) const;
>@@ -207,13 +211,13 @@ class StunMessage {
>   uint16_t length_;
>   std::string transaction_id_;
>   std::vector<std::unique_ptr<StunAttribute>> attrs_;
>+  uint32_t stun_magic_cookie_;
> };
> 
> // Base class for all STUN/TURN attributes.
> class StunAttribute {
>  public:
>-  virtual ~StunAttribute() {
>-  }
>+  virtual ~StunAttribute() {}
> 
>   int type() const { return type_; }
>   size_t length() const { return length_; }
>@@ -222,7 +226,7 @@ class StunAttribute {
>   virtual StunAttributeValueType value_type() const = 0;
> 
>   // Only XorAddressAttribute needs this so far.
>-  virtual void SetOwner(StunMessage*) {}
>+  virtual void SetOwner(StunMessage* owner) {}
> 
>   // Reads the body (not the type or length) for this type of attribute from
>   // the given buffer.  Return value is true if successful.
>@@ -478,7 +482,9 @@ bool IsStunErrorResponseType(int msg_type);
> 
> // Computes the STUN long-term credential hash.
> bool ComputeStunCredentialHash(const std::string& username,
>-    const std::string& realm, const std::string& password, std::string* hash);
>+                               const std::string& realm,
>+                               const std::string& password,
>+                               std::string* hash);
> 
> // Make a copy af |attribute| and return a new StunAttribute.
> //   This is useful if you don't care about what kind of attribute you
>@@ -503,22 +509,22 @@ enum RelayMessageType {
>   // STUN_ALLOCATE_REQUEST              = 0x0003,
>   // STUN_ALLOCATE_RESPONSE             = 0x0103,
>   // STUN_ALLOCATE_ERROR_RESPONSE       = 0x0113,
>-  STUN_SEND_REQUEST                     = 0x0004,
>-  STUN_SEND_RESPONSE                    = 0x0104,
>-  STUN_SEND_ERROR_RESPONSE              = 0x0114,
>-  STUN_DATA_INDICATION                  = 0x0115,
>+  STUN_SEND_REQUEST = 0x0004,
>+  STUN_SEND_RESPONSE = 0x0104,
>+  STUN_SEND_ERROR_RESPONSE = 0x0114,
>+  STUN_DATA_INDICATION = 0x0115,
> };
> 
> // "GTURN"-specific STUN attributes.
> // TODO(?): Rename these attributes to GTURN_ to avoid conflicts.
> enum RelayAttributeType {
>-  STUN_ATTR_LIFETIME                    = 0x000d,  // UInt32
>-  STUN_ATTR_MAGIC_COOKIE                = 0x000f,  // ByteString, 4 bytes
>-  STUN_ATTR_BANDWIDTH                   = 0x0010,  // UInt32
>-  STUN_ATTR_DESTINATION_ADDRESS         = 0x0011,  // Address
>-  STUN_ATTR_SOURCE_ADDRESS2             = 0x0012,  // Address
>-  STUN_ATTR_DATA                        = 0x0013,  // ByteString
>-  STUN_ATTR_OPTIONS                     = 0x8001,  // UInt32
>+  STUN_ATTR_LIFETIME = 0x000d,             // UInt32
>+  STUN_ATTR_MAGIC_COOKIE = 0x000f,         // ByteString, 4 bytes
>+  STUN_ATTR_BANDWIDTH = 0x0010,            // UInt32
>+  STUN_ATTR_DESTINATION_ADDRESS = 0x0011,  // Address
>+  STUN_ATTR_SOURCE_ADDRESS2 = 0x0012,      // Address
>+  STUN_ATTR_DATA = 0x0013,                 // ByteString
>+  STUN_ATTR_OPTIONS = 0x8001,              // UInt32
> };
> 
> // A "GTURN" STUN message.
>@@ -530,33 +536,33 @@ class RelayMessage : public StunMessage {
> 
> // Defined in TURN RFC 5766.
> enum TurnMessageType {
>-  STUN_ALLOCATE_REQUEST                 = 0x0003,
>-  STUN_ALLOCATE_RESPONSE                = 0x0103,
>-  STUN_ALLOCATE_ERROR_RESPONSE          = 0x0113,
>-  TURN_REFRESH_REQUEST                  = 0x0004,
>-  TURN_REFRESH_RESPONSE                 = 0x0104,
>-  TURN_REFRESH_ERROR_RESPONSE           = 0x0114,
>-  TURN_SEND_INDICATION                  = 0x0016,
>-  TURN_DATA_INDICATION                  = 0x0017,
>-  TURN_CREATE_PERMISSION_REQUEST        = 0x0008,
>-  TURN_CREATE_PERMISSION_RESPONSE       = 0x0108,
>+  STUN_ALLOCATE_REQUEST = 0x0003,
>+  STUN_ALLOCATE_RESPONSE = 0x0103,
>+  STUN_ALLOCATE_ERROR_RESPONSE = 0x0113,
>+  TURN_REFRESH_REQUEST = 0x0004,
>+  TURN_REFRESH_RESPONSE = 0x0104,
>+  TURN_REFRESH_ERROR_RESPONSE = 0x0114,
>+  TURN_SEND_INDICATION = 0x0016,
>+  TURN_DATA_INDICATION = 0x0017,
>+  TURN_CREATE_PERMISSION_REQUEST = 0x0008,
>+  TURN_CREATE_PERMISSION_RESPONSE = 0x0108,
>   TURN_CREATE_PERMISSION_ERROR_RESPONSE = 0x0118,
>-  TURN_CHANNEL_BIND_REQUEST             = 0x0009,
>-  TURN_CHANNEL_BIND_RESPONSE            = 0x0109,
>-  TURN_CHANNEL_BIND_ERROR_RESPONSE      = 0x0119,
>+  TURN_CHANNEL_BIND_REQUEST = 0x0009,
>+  TURN_CHANNEL_BIND_RESPONSE = 0x0109,
>+  TURN_CHANNEL_BIND_ERROR_RESPONSE = 0x0119,
> };
> 
> enum TurnAttributeType {
>-  STUN_ATTR_CHANNEL_NUMBER              = 0x000C,  // UInt32
>-  STUN_ATTR_TURN_LIFETIME               = 0x000d,  // UInt32
>-  STUN_ATTR_XOR_PEER_ADDRESS            = 0x0012,  // XorAddress
>+  STUN_ATTR_CHANNEL_NUMBER = 0x000C,    // UInt32
>+  STUN_ATTR_TURN_LIFETIME = 0x000d,     // UInt32
>+  STUN_ATTR_XOR_PEER_ADDRESS = 0x0012,  // XorAddress
>   // TODO(mallinath) - Uncomment after RelayAttributes are renamed.
>   // STUN_ATTR_DATA                     = 0x0013,  // ByteString
>-  STUN_ATTR_XOR_RELAYED_ADDRESS         = 0x0016,  // XorAddress
>-  STUN_ATTR_EVEN_PORT                   = 0x0018,  // ByteString, 1 byte.
>-  STUN_ATTR_REQUESTED_TRANSPORT         = 0x0019,  // UInt32
>-  STUN_ATTR_DONT_FRAGMENT               = 0x001A,  // No content, Length = 0
>-  STUN_ATTR_RESERVATION_TOKEN           = 0x0022,  // ByteString, 8 bytes.
>+  STUN_ATTR_XOR_RELAYED_ADDRESS = 0x0016,  // XorAddress
>+  STUN_ATTR_EVEN_PORT = 0x0018,            // ByteString, 1 byte.
>+  STUN_ATTR_REQUESTED_TRANSPORT = 0x0019,  // UInt32
>+  STUN_ATTR_DONT_FRAGMENT = 0x001A,        // No content, Length = 0
>+  STUN_ATTR_RESERVATION_TOKEN = 0x0022,    // ByteString, 8 bytes.
>   // TODO(mallinath) - Rename STUN_ATTR_TURN_LIFETIME to STUN_ATTR_LIFETIME and
>   // STUN_ATTR_TURN_DATA to STUN_ATTR_DATA. Also rename RelayMessage attributes
>   // by appending G to attribute name.
>@@ -564,10 +570,10 @@ enum TurnAttributeType {
> 
> // RFC 5766-defined errors.
> enum TurnErrorType {
>-  STUN_ERROR_FORBIDDEN                  = 403,
>-  STUN_ERROR_ALLOCATION_MISMATCH        = 437,
>-  STUN_ERROR_WRONG_CREDENTIALS          = 441,
>-  STUN_ERROR_UNSUPPORTED_PROTOCOL       = 442
>+  STUN_ERROR_FORBIDDEN = 403,
>+  STUN_ERROR_ALLOCATION_MISMATCH = 437,
>+  STUN_ERROR_WRONG_CREDENTIALS = 441,
>+  STUN_ERROR_UNSUPPORTED_PROTOCOL = 442
> };
> extern const char STUN_ERROR_REASON_FORBIDDEN[];
> extern const char STUN_ERROR_REASON_ALLOCATION_MISMATCH[];
>@@ -593,7 +599,7 @@ enum IceAttributeType {
> 
> // RFC 5245-defined errors.
> enum IceErrorCode {
>-  STUN_ERROR_ROLE_CONFLICT              = 487,
>+  STUN_ERROR_ROLE_CONFLICT = 487,
> };
> extern const char STUN_ERROR_REASON_ROLE_CONFLICT[];
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stun_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stun_unittest.cc
>index 5c805dc7702..6d9510882d1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stun_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stun_unittest.cc
>@@ -11,27 +11,29 @@
> #include <string>
> #include <utility>
> 
>+#include "absl/memory/memory.h"
> #include "p2p/base/stun.h"
> #include "rtc_base/arraysize.h"
> #include "rtc_base/bytebuffer.h"
> #include "rtc_base/gunit.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/messagedigest.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/socketaddress.h"
> 
> namespace cricket {
> 
> class StunTest : public ::testing::Test {
>  protected:
>-  void CheckStunHeader(const StunMessage& msg, StunMessageType expected_type,
>+  void CheckStunHeader(const StunMessage& msg,
>+                       StunMessageType expected_type,
>                        size_t expected_length) {
>     ASSERT_EQ(expected_type, msg.type());
>     ASSERT_EQ(expected_length, msg.length());
>   }
> 
>   void CheckStunTransactionID(const StunMessage& msg,
>-                              const unsigned char* expectedID, size_t length) {
>+                              const unsigned char* expectedID,
>+                              size_t length) {
>     ASSERT_EQ(length, msg.transaction_id().size());
>     ASSERT_EQ(length == kStunTransactionIdLength + 4, msg.IsLegacy());
>     ASSERT_EQ(length == kStunTransactionIdLength, !msg.IsLegacy());
>@@ -73,11 +75,13 @@ class StunTest : public ::testing::Test {
>   }
> };
> 
>-
> // Sample STUN packets with various attributes
> // Gathered by wiresharking pjproject's pjnath test programs
> // pjproject available at www.pjsip.org
> 
>+// clang-format off
>+// clang formatting doesn't respect inline comments.
>+
> static const unsigned char kStunMessageWithIPv6MappedAddress[] = {
>   0x00, 0x01, 0x00, 0x18,  // message header
>   0x21, 0x12, 0xa4, 0x42,  // transaction id
>@@ -267,6 +271,7 @@ static const unsigned char kRtcpPacket[] = {
>   0x00, 0x03, 0x73, 0x50,
> };
> 
>+
> // RFC5769 Test Vectors
> // Software name (request):  "STUN test client" (without quotes)
> // Software name (response): "test vector" (without quotes)
>@@ -472,31 +477,31 @@ static const unsigned char kCalculatedHmac2[] = {
>   0x43, 0x14, 0x10, 0x28   // }
> };
> 
>+// clang-format on
>+
> // A transaction ID without the 'magic cookie' portion
> // pjnat's test programs use this transaction ID a lot.
>-const unsigned char kTestTransactionId1[] = { 0x029, 0x01f, 0x0cd, 0x07c,
>-                                              0x0ba, 0x058, 0x0ab, 0x0d7,
>-                                              0x0f2, 0x041, 0x001, 0x000 };
>+const unsigned char kTestTransactionId1[] = {0x029, 0x01f, 0x0cd, 0x07c,
>+                                             0x0ba, 0x058, 0x0ab, 0x0d7,
>+                                             0x0f2, 0x041, 0x001, 0x000};
> 
> // They use this one sometimes too.
>-const unsigned char kTestTransactionId2[] = { 0x0e3, 0x0a9, 0x046, 0x0e1,
>-                                              0x07c, 0x000, 0x0c2, 0x062,
>-                                              0x054, 0x008, 0x001, 0x000 };
>-
>-const in6_addr kIPv6TestAddress1 = { { { 0x24, 0x01, 0xfa, 0x00,
>-                                         0x00, 0x04, 0x10, 0x00,
>-                                         0xbe, 0x30, 0x5b, 0xff,
>-                                         0xfe, 0xe5, 0x00, 0xc3 } } };
>-const in6_addr kIPv6TestAddress2 = { { { 0x24, 0x01, 0xfa, 0x00,
>-                                         0x00, 0x04, 0x10, 0x12,
>-                                         0x06, 0x0c, 0xce, 0xff,
>-                                         0xfe, 0x1f, 0x61, 0xa4 } } };
>+const unsigned char kTestTransactionId2[] = {0x0e3, 0x0a9, 0x046, 0x0e1,
>+                                             0x07c, 0x000, 0x0c2, 0x062,
>+                                             0x054, 0x008, 0x001, 0x000};
>+
>+const in6_addr kIPv6TestAddress1 = {
>+    {{0x24, 0x01, 0xfa, 0x00, 0x00, 0x04, 0x10, 0x00, 0xbe, 0x30, 0x5b, 0xff,
>+      0xfe, 0xe5, 0x00, 0xc3}}};
>+const in6_addr kIPv6TestAddress2 = {
>+    {{0x24, 0x01, 0xfa, 0x00, 0x00, 0x04, 0x10, 0x12, 0x06, 0x0c, 0xce, 0xff,
>+      0xfe, 0x1f, 0x61, 0xa4}}};
> 
> #ifdef WEBRTC_POSIX
>-const in_addr kIPv4TestAddress1 =  { 0xe64417ac };
>+const in_addr kIPv4TestAddress1 = {0xe64417ac};
> #elif defined WEBRTC_WIN
> // Windows in_addr has a union with a uchar[] array first.
>-const in_addr kIPv4TestAddress1 =  { { { 0x0ac, 0x017, 0x044, 0x0e6 } } };
>+const in_addr kIPv4TestAddress1 = {{{0x0ac, 0x017, 0x044, 0x0e6}}};
> #endif
> const char kTestUserName1[] = "abcdefgh";
> const char kTestUserName2[] = "abc";
>@@ -516,9 +521,9 @@ const int kTestMessagePort4 = 40444;
> // Test that the GetStun*Type and IsStun*Type methods work as expected.
> TEST_F(StunTest, MessageTypes) {
>   EXPECT_EQ(STUN_BINDING_RESPONSE,
>-      GetStunSuccessResponseType(STUN_BINDING_REQUEST));
>+            GetStunSuccessResponseType(STUN_BINDING_REQUEST));
>   EXPECT_EQ(STUN_BINDING_ERROR_RESPONSE,
>-      GetStunErrorResponseType(STUN_BINDING_REQUEST));
>+            GetStunErrorResponseType(STUN_BINDING_REQUEST));
>   EXPECT_EQ(-1, GetStunSuccessResponseType(STUN_BINDING_INDICATION));
>   EXPECT_EQ(-1, GetStunSuccessResponseType(STUN_BINDING_RESPONSE));
>   EXPECT_EQ(-1, GetStunSuccessResponseType(STUN_BINDING_ERROR_RESPONSE));
>@@ -526,10 +531,8 @@ TEST_F(StunTest, MessageTypes) {
>   EXPECT_EQ(-1, GetStunErrorResponseType(STUN_BINDING_RESPONSE));
>   EXPECT_EQ(-1, GetStunErrorResponseType(STUN_BINDING_ERROR_RESPONSE));
> 
>-  int types[] = {
>-    STUN_BINDING_REQUEST, STUN_BINDING_INDICATION,
>-    STUN_BINDING_RESPONSE, STUN_BINDING_ERROR_RESPONSE
>-  };
>+  int types[] = {STUN_BINDING_REQUEST, STUN_BINDING_INDICATION,
>+                 STUN_BINDING_RESPONSE, STUN_BINDING_ERROR_RESPONSE};
>   for (size_t i = 0; i < arraysize(types); ++i) {
>     EXPECT_EQ(i == 0U, IsStunRequestType(types[i]));
>     EXPECT_EQ(i == 1U, IsStunIndicationType(types[i]));
>@@ -547,8 +550,8 @@ TEST_F(StunTest, ReadMessageWithIPv4AddressAttribute) {
> 
>   const StunAddressAttribute* addr = msg.GetAddress(STUN_ATTR_MAPPED_ADDRESS);
>   rtc::IPAddress test_address(kIPv4TestAddress1);
>-  CheckStunAddressAttribute(addr, STUN_ADDRESS_IPV4,
>-                            kTestMessagePort4, test_address);
>+  CheckStunAddressAttribute(addr, STUN_ADDRESS_IPV4, kTestMessagePort4,
>+                            test_address);
> }
> 
> TEST_F(StunTest, ReadMessageWithIPv4XorAddressAttribute) {
>@@ -561,8 +564,8 @@ TEST_F(StunTest, ReadMessageWithIPv4XorAddressAttribute) {
>   const StunAddressAttribute* addr =
>       msg.GetAddress(STUN_ATTR_XOR_MAPPED_ADDRESS);
>   rtc::IPAddress test_address(kIPv4TestAddress1);
>-  CheckStunAddressAttribute(addr, STUN_ADDRESS_IPV4,
>-                            kTestMessagePort3, test_address);
>+  CheckStunAddressAttribute(addr, STUN_ADDRESS_IPV4, kTestMessagePort3,
>+                            test_address);
> }
> 
> TEST_F(StunTest, ReadMessageWithIPv6AddressAttribute) {
>@@ -574,8 +577,8 @@ TEST_F(StunTest, ReadMessageWithIPv6AddressAttribute) {
>   rtc::IPAddress test_address(kIPv6TestAddress1);
> 
>   const StunAddressAttribute* addr = msg.GetAddress(STUN_ATTR_MAPPED_ADDRESS);
>-  CheckStunAddressAttribute(addr, STUN_ADDRESS_IPV6,
>-                            kTestMessagePort2, test_address);
>+  CheckStunAddressAttribute(addr, STUN_ADDRESS_IPV6, kTestMessagePort2,
>+                            test_address);
> }
> 
> TEST_F(StunTest, ReadMessageWithInvalidAddressAttribute) {
>@@ -587,8 +590,8 @@ TEST_F(StunTest, ReadMessageWithInvalidAddressAttribute) {
>   rtc::IPAddress test_address(kIPv6TestAddress1);
> 
>   const StunAddressAttribute* addr = msg.GetAddress(STUN_ATTR_MAPPED_ADDRESS);
>-  CheckStunAddressAttribute(addr, STUN_ADDRESS_IPV6,
>-                            kTestMessagePort2, test_address);
>+  CheckStunAddressAttribute(addr, STUN_ADDRESS_IPV6, kTestMessagePort2,
>+                            test_address);
> }
> 
> TEST_F(StunTest, ReadMessageWithIPv6XorAddressAttribute) {
>@@ -602,8 +605,8 @@ TEST_F(StunTest, ReadMessageWithIPv6XorAddressAttribute) {
> 
>   const StunAddressAttribute* addr =
>       msg.GetAddress(STUN_ATTR_XOR_MAPPED_ADDRESS);
>-  CheckStunAddressAttribute(addr, STUN_ADDRESS_IPV6,
>-                            kTestMessagePort1, test_address);
>+  CheckStunAddressAttribute(addr, STUN_ADDRESS_IPV6, kTestMessagePort1,
>+                            test_address);
> }
> 
> // Read the RFC5389 fields from the RFC5769 sample STUN request.
>@@ -628,8 +631,7 @@ TEST_F(StunTest, ReadRfc5769RequestMessage) {
>   ASSERT_TRUE(msg.GetByteString(STUN_ATTR_MESSAGE_INTEGRITY) != NULL);
> 
>   // Fingerprint checked in a later test, but double-check the value here.
>-  const StunUInt32Attribute* fingerprint =
>-      msg.GetUInt32(STUN_ATTR_FINGERPRINT);
>+  const StunUInt32Attribute* fingerprint = msg.GetUInt32(STUN_ATTR_FINGERPRINT);
>   ASSERT_TRUE(fingerprint != NULL);
>   EXPECT_EQ(0xe57a3bcf, fingerprint->value());
> }
>@@ -693,13 +695,11 @@ TEST_F(StunTest, ReadRfc5769RequestMessageLongTermAuth) {
>   ASSERT_TRUE(username != NULL);
>   EXPECT_EQ(kRfc5769SampleMsgWithAuthUsername, username->GetString());
> 
>-  const StunByteStringAttribute* nonce =
>-      msg.GetByteString(STUN_ATTR_NONCE);
>+  const StunByteStringAttribute* nonce = msg.GetByteString(STUN_ATTR_NONCE);
>   ASSERT_TRUE(nonce != NULL);
>   EXPECT_EQ(kRfc5769SampleMsgWithAuthNonce, nonce->GetString());
> 
>-  const StunByteStringAttribute* realm =
>-      msg.GetByteString(STUN_ATTR_REALM);
>+  const StunByteStringAttribute* realm = msg.GetByteString(STUN_ATTR_REALM);
>   ASSERT_TRUE(realm != NULL);
>   EXPECT_EQ(kRfc5769SampleMsgWithAuthRealm, realm->GetString());
> 
>@@ -714,7 +714,7 @@ TEST_F(StunTest, ReadRfc5769RequestMessageLongTermAuth) {
> TEST_F(StunTest, ReadLegacyMessage) {
>   unsigned char rfc3489_packet[sizeof(kStunMessageWithIPv4MappedAddress)];
>   memcpy(rfc3489_packet, kStunMessageWithIPv4MappedAddress,
>-      sizeof(kStunMessageWithIPv4MappedAddress));
>+         sizeof(kStunMessageWithIPv4MappedAddress));
>   // Overwrite the magic cookie here.
>   memcpy(&rfc3489_packet[4], "ABCD", 4);
> 
>@@ -725,8 +725,8 @@ TEST_F(StunTest, ReadLegacyMessage) {
> 
>   const StunAddressAttribute* addr = msg.GetAddress(STUN_ATTR_MAPPED_ADDRESS);
>   rtc::IPAddress test_address(kIPv4TestAddress1);
>-  CheckStunAddressAttribute(addr, STUN_ADDRESS_IPV4,
>-                            kTestMessagePort4, test_address);
>+  CheckStunAddressAttribute(addr, STUN_ADDRESS_IPV4, kTestMessagePort4,
>+                            test_address);
> }
> 
> TEST_F(StunTest, SetIPv6XorAddressAttributeOwner) {
>@@ -741,8 +741,8 @@ TEST_F(StunTest, SetIPv6XorAddressAttributeOwner) {
> 
>   const StunAddressAttribute* addr =
>       msg.GetAddress(STUN_ATTR_XOR_MAPPED_ADDRESS);
>-  CheckStunAddressAttribute(addr, STUN_ADDRESS_IPV6,
>-                            kTestMessagePort1, test_address);
>+  CheckStunAddressAttribute(addr, STUN_ADDRESS_IPV6, kTestMessagePort1,
>+                            test_address);
> 
>   // Owner with a different transaction ID.
>   msg2.SetTransactionID("ABCDABCDABCD");
>@@ -789,8 +789,8 @@ TEST_F(StunTest, SetIPv4XorAddressAttributeOwner) {
> 
>   const StunAddressAttribute* addr =
>       msg.GetAddress(STUN_ATTR_XOR_MAPPED_ADDRESS);
>-  CheckStunAddressAttribute(addr, STUN_ADDRESS_IPV4,
>-                            kTestMessagePort3, test_address);
>+  CheckStunAddressAttribute(addr, STUN_ADDRESS_IPV4, kTestMessagePort3,
>+                            test_address);
> 
>   // Owner with a different transaction ID.
>   msg2.SetTransactionID("ABCDABCDABCD");
>@@ -994,8 +994,8 @@ TEST_F(StunTest, ReadByteStringAttribute) {
> 
> TEST_F(StunTest, ReadPaddedByteStringAttribute) {
>   StunMessage msg;
>-  size_t size = ReadStunMessage(&msg,
>-                                kStunMessageWithPaddedByteStringAttribute);
>+  size_t size =
>+      ReadStunMessage(&msg, kStunMessageWithPaddedByteStringAttribute);
>   ASSERT_NE(0U, size);
>   CheckStunHeader(msg, STUN_BINDING_REQUEST, size);
>   CheckStunTransactionID(msg, kTestTransactionId2, kStunTransactionIdLength);
>@@ -1056,8 +1056,7 @@ TEST_F(StunTest, ReadMessageWithOriginAttribute) {
>   StunMessage msg;
>   size_t size = ReadStunMessage(&msg, kStunMessageWithOriginAttribute);
>   CheckStunHeader(msg, STUN_BINDING_REQUEST, size);
>-  const StunByteStringAttribute* origin =
>-      msg.GetByteString(STUN_ATTR_ORIGIN);
>+  const StunByteStringAttribute* origin = msg.GetByteString(STUN_ATTR_ORIGIN);
>   ASSERT_TRUE(origin != NULL);
>   EXPECT_EQ(kTestOrigin, origin->GetString());
> }
>@@ -1117,7 +1116,7 @@ TEST_F(StunTest, WriteMessageWithOriginAttribute) {
>       std::string(reinterpret_cast<const char*>(kTestTransactionId1),
>                   kStunTransactionIdLength));
>   auto origin =
>-      rtc::MakeUnique<StunByteStringAttribute>(STUN_ATTR_ORIGIN, kTestOrigin);
>+      absl::make_unique<StunByteStringAttribute>(STUN_ATTR_ORIGIN, kTestOrigin);
>   msg.AddAttribute(std::move(origin));
> 
>   rtc::ByteBufferWriter out;
>@@ -1154,56 +1153,47 @@ TEST_F(StunTest, ValidateMessageIntegrity) {
>   // Try the messages from RFC 5769.
>   EXPECT_TRUE(StunMessage::ValidateMessageIntegrity(
>       reinterpret_cast<const char*>(kRfc5769SampleRequest),
>-      sizeof(kRfc5769SampleRequest),
>-      kRfc5769SampleMsgPassword));
>+      sizeof(kRfc5769SampleRequest), kRfc5769SampleMsgPassword));
>   EXPECT_FALSE(StunMessage::ValidateMessageIntegrity(
>       reinterpret_cast<const char*>(kRfc5769SampleRequest),
>-      sizeof(kRfc5769SampleRequest),
>-      "InvalidPassword"));
>+      sizeof(kRfc5769SampleRequest), "InvalidPassword"));
> 
>   EXPECT_TRUE(StunMessage::ValidateMessageIntegrity(
>       reinterpret_cast<const char*>(kRfc5769SampleResponse),
>-      sizeof(kRfc5769SampleResponse),
>-      kRfc5769SampleMsgPassword));
>+      sizeof(kRfc5769SampleResponse), kRfc5769SampleMsgPassword));
>   EXPECT_FALSE(StunMessage::ValidateMessageIntegrity(
>       reinterpret_cast<const char*>(kRfc5769SampleResponse),
>-      sizeof(kRfc5769SampleResponse),
>-      "InvalidPassword"));
>+      sizeof(kRfc5769SampleResponse), "InvalidPassword"));
> 
>   EXPECT_TRUE(StunMessage::ValidateMessageIntegrity(
>       reinterpret_cast<const char*>(kRfc5769SampleResponseIPv6),
>-      sizeof(kRfc5769SampleResponseIPv6),
>-      kRfc5769SampleMsgPassword));
>+      sizeof(kRfc5769SampleResponseIPv6), kRfc5769SampleMsgPassword));
>   EXPECT_FALSE(StunMessage::ValidateMessageIntegrity(
>       reinterpret_cast<const char*>(kRfc5769SampleResponseIPv6),
>-      sizeof(kRfc5769SampleResponseIPv6),
>-      "InvalidPassword"));
>+      sizeof(kRfc5769SampleResponseIPv6), "InvalidPassword"));
> 
>   // We first need to compute the key for the long-term authentication HMAC.
>   std::string key;
>   ComputeStunCredentialHash(kRfc5769SampleMsgWithAuthUsername,
>-      kRfc5769SampleMsgWithAuthRealm, kRfc5769SampleMsgWithAuthPassword, &key);
>+                            kRfc5769SampleMsgWithAuthRealm,
>+                            kRfc5769SampleMsgWithAuthPassword, &key);
>   EXPECT_TRUE(StunMessage::ValidateMessageIntegrity(
>       reinterpret_cast<const char*>(kRfc5769SampleRequestLongTermAuth),
>       sizeof(kRfc5769SampleRequestLongTermAuth), key));
>   EXPECT_FALSE(StunMessage::ValidateMessageIntegrity(
>       reinterpret_cast<const char*>(kRfc5769SampleRequestLongTermAuth),
>-      sizeof(kRfc5769SampleRequestLongTermAuth),
>-      "InvalidPassword"));
>+      sizeof(kRfc5769SampleRequestLongTermAuth), "InvalidPassword"));
> 
>   // Try some edge cases.
>   EXPECT_FALSE(StunMessage::ValidateMessageIntegrity(
>       reinterpret_cast<const char*>(kStunMessageWithZeroLength),
>-      sizeof(kStunMessageWithZeroLength),
>-      kRfc5769SampleMsgPassword));
>+      sizeof(kStunMessageWithZeroLength), kRfc5769SampleMsgPassword));
>   EXPECT_FALSE(StunMessage::ValidateMessageIntegrity(
>       reinterpret_cast<const char*>(kStunMessageWithExcessLength),
>-      sizeof(kStunMessageWithExcessLength),
>-      kRfc5769SampleMsgPassword));
>+      sizeof(kStunMessageWithExcessLength), kRfc5769SampleMsgPassword));
>   EXPECT_FALSE(StunMessage::ValidateMessageIntegrity(
>       reinterpret_cast<const char*>(kStunMessageWithSmallLength),
>-      sizeof(kStunMessageWithSmallLength),
>-      kRfc5769SampleMsgPassword));
>+      sizeof(kStunMessageWithSmallLength), kRfc5769SampleMsgPassword));
> 
>   // Again, but with the lengths matching what is claimed in the headers.
>   EXPECT_FALSE(StunMessage::ValidateMessageIntegrity(
>@@ -1222,8 +1212,7 @@ TEST_F(StunTest, ValidateMessageIntegrity) {
>   // Check that a too-short HMAC doesn't cause buffer overflow.
>   EXPECT_FALSE(StunMessage::ValidateMessageIntegrity(
>       reinterpret_cast<const char*>(kStunMessageWithBadHmacAtEnd),
>-      sizeof(kStunMessageWithBadHmacAtEnd),
>-      kRfc5769SampleMsgPassword));
>+      sizeof(kStunMessageWithBadHmacAtEnd), kRfc5769SampleMsgPassword));
> 
>   // Test that munging a single bit anywhere in the message causes the
>   // message-integrity check to fail, unless it is after the M-I attribute.
>@@ -1233,8 +1222,9 @@ TEST_F(StunTest, ValidateMessageIntegrity) {
>     buf[i] ^= 0x01;
>     if (i > 0)
>       buf[i - 1] ^= 0x01;
>-    EXPECT_EQ(i >= sizeof(buf) - 8, StunMessage::ValidateMessageIntegrity(
>-        buf, sizeof(buf), kRfc5769SampleMsgPassword));
>+    EXPECT_EQ(i >= sizeof(buf) - 8,
>+              StunMessage::ValidateMessageIntegrity(buf, sizeof(buf),
>+                                                    kRfc5769SampleMsgPassword));
>   }
> }
> 
>@@ -1251,14 +1241,14 @@ TEST_F(StunTest, AddMessageIntegrity) {
>   const StunByteStringAttribute* mi_attr =
>       msg.GetByteString(STUN_ATTR_MESSAGE_INTEGRITY);
>   EXPECT_EQ(20U, mi_attr->length());
>-  EXPECT_EQ(0, memcmp(
>-      mi_attr->bytes(), kCalculatedHmac1, sizeof(kCalculatedHmac1)));
>+  EXPECT_EQ(
>+      0, memcmp(mi_attr->bytes(), kCalculatedHmac1, sizeof(kCalculatedHmac1)));
> 
>   rtc::ByteBufferWriter buf1;
>   EXPECT_TRUE(msg.Write(&buf1));
>   EXPECT_TRUE(StunMessage::ValidateMessageIntegrity(
>-        reinterpret_cast<const char*>(buf1.Data()), buf1.Length(),
>-        kRfc5769SampleMsgPassword));
>+      reinterpret_cast<const char*>(buf1.Data()), buf1.Length(),
>+      kRfc5769SampleMsgPassword));
> 
>   IceMessage msg2;
>   rtc::ByteBufferReader buf2(
>@@ -1275,8 +1265,8 @@ TEST_F(StunTest, AddMessageIntegrity) {
>   rtc::ByteBufferWriter buf3;
>   EXPECT_TRUE(msg2.Write(&buf3));
>   EXPECT_TRUE(StunMessage::ValidateMessageIntegrity(
>-        reinterpret_cast<const char*>(buf3.Data()), buf3.Length(),
>-        kRfc5769SampleMsgPassword));
>+      reinterpret_cast<const char*>(buf3.Data()), buf3.Length(),
>+      kRfc5769SampleMsgPassword));
> }
> 
> // Check our STUN message validation code against the RFC5769 test messages.
>@@ -1331,6 +1321,8 @@ TEST_F(StunTest, AddFingerprint) {
> }
> 
> // Sample "GTURN" relay message.
>+// clang-format off
>+// clang formatting doesn't respect inline comments.
> static const unsigned char kRelayMessage[] = {
>   0x00, 0x01, 0x00, 88,    // message header
>   0x21, 0x12, 0xA4, 0x42,  // magic cookie
>@@ -1360,6 +1352,7 @@ static const unsigned char kRelayMessage[] = {
>   'a', 'b', 'c', 'd',
>   'e', 'f', 'g', 0         // DATA must be padded per rfc5766.
> };
>+// clang-format on
> 
> // Test that we can read the GTURN-specific fields.
> TEST_F(StunTest, ReadRelayMessage) {
>@@ -1412,10 +1405,8 @@ TEST_F(StunTest, ReadRelayMessage) {
>   bytes = msg.GetByteString(STUN_ATTR_MAGIC_COOKIE);
>   ASSERT_TRUE(bytes != NULL);
>   EXPECT_EQ(4U, bytes->length());
>-  EXPECT_EQ(0,
>-            memcmp(bytes->bytes(),
>-                   TURN_MAGIC_COOKIE_VALUE,
>-                   sizeof(TURN_MAGIC_COOKIE_VALUE)));
>+  EXPECT_EQ(0, memcmp(bytes->bytes(), TURN_MAGIC_COOKIE_VALUE,
>+                      sizeof(TURN_MAGIC_COOKIE_VALUE)));
> 
>   bytes2 = StunAttribute::CreateByteString(STUN_ATTR_MAGIC_COOKIE);
>   bytes2->CopyBytes(reinterpret_cast<const char*>(TURN_MAGIC_COOKIE_VALUE),
>@@ -1541,10 +1532,10 @@ TEST_F(StunTest, RemoveAttribute) {
> // Test CopyStunAttribute
> TEST_F(StunTest, CopyAttribute) {
>   rtc::ByteBufferWriter buf;
>-  rtc::ByteBufferWriter* buffer_ptrs[] = { &buf, nullptr };
>+  rtc::ByteBufferWriter* buffer_ptrs[] = {&buf, nullptr};
>   // Test both with and without supplied ByteBufferWriter.
>   for (auto buffer_ptr : buffer_ptrs) {
>-    { // Test StunByteStringAttribute.
>+    {  // Test StunByteStringAttribute.
>       auto attr = StunAttribute::CreateByteString(STUN_ATTR_USERNAME);
>       attr->CopyBytes("kes", sizeof("kes"));
> 
>@@ -1554,13 +1545,13 @@ TEST_F(StunTest, CopyAttribute) {
>                    static_cast<StunByteStringAttribute*>(copy.get())->bytes());
>     }
> 
>-    { // Test StunAddressAttribute.
>+    {  // Test StunAddressAttribute.
>       rtc::IPAddress test_ip(kIPv6TestAddress2);
>       auto addr = StunAttribute::CreateAddress(STUN_ATTR_MAPPED_ADDRESS);
>       rtc::SocketAddress test_addr(test_ip, kTestMessagePort2);
>       addr->SetAddress(test_addr);
>-      CheckStunAddressAttribute(addr.get(),
>-                                STUN_ADDRESS_IPV6, kTestMessagePort2, test_ip);
>+      CheckStunAddressAttribute(addr.get(), STUN_ADDRESS_IPV6,
>+                                kTestMessagePort2, test_ip);
> 
>       auto copy = CopyStunAttribute(*addr.get(), buffer_ptr);
>       ASSERT_EQ(copy->value_type(), STUN_VALUE_ADDRESS);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunport.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunport.cc
>index d450d8e94d6..9574314c769 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunport.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunport.cc
>@@ -13,7 +13,7 @@
> #include <utility>
> #include <vector>
> 
>-#include "p2p/base/common.h"
>+#include "p2p/base/p2pconstants.h"
> #include "p2p/base/portallocator.h"
> #include "p2p/base/stun.h"
> #include "rtc_base/checks.h"
>@@ -25,8 +25,7 @@
> namespace cricket {
> 
> // TODO(?): Move these to a common place (used in relayport too)
>-const int KEEPALIVE_DELAY = 10 * 1000;  // 10 seconds - sort timeouts
>-const int RETRY_TIMEOUT = 50 * 1000;    // 50 seconds
>+const int RETRY_TIMEOUT = 50 * 1000;  // 50 seconds
> 
> // Handles a binding request sent to the STUN server.
> class StunBindingRequest : public StunRequest {
>@@ -52,7 +51,7 @@ class StunBindingRequest : public StunRequest {
>       RTC_LOG(LS_ERROR) << "Binding address has bad family";
>     } else {
>       rtc::SocketAddress addr(addr_attr->ipaddr(), addr_attr->port());
>-      port_->OnStunBindingRequestSucceeded(server_addr_, addr);
>+      port_->OnStunBindingRequestSucceeded(this->Elapsed(), server_addr_, addr);
>     }
> 
>     // The keep-alive requests will be stopped after its lifetime has passed.
>@@ -69,9 +68,9 @@ class StunBindingRequest : public StunRequest {
>       RTC_LOG(LS_ERROR) << "Missing binding response error code.";
>     } else {
>       RTC_LOG(LS_ERROR) << "Binding error response:"
>-                        << " class=" << attr->eclass()
>-                        << " number=" << attr->number() << " reason='"
>-                        << attr->reason() << "'";
>+                           " class="
>+                        << attr->eclass() << " number=" << attr->number()
>+                        << " reason=" << attr->reason();
>     }
> 
>     port_->OnStunBindingOrResolveRequestFailed(server_addr_);
>@@ -105,13 +104,12 @@ class StunBindingRequest : public StunRequest {
>   int64_t start_time_;
> };
> 
>-UDPPort::AddressResolver::AddressResolver(
>-    rtc::PacketSocketFactory* factory)
>+UDPPort::AddressResolver::AddressResolver(rtc::PacketSocketFactory* factory)
>     : socket_factory_(factory) {}
> 
> UDPPort::AddressResolver::~AddressResolver() {
>-  for (ResolverMap::iterator it = resolvers_.begin();
>-       it != resolvers_.end(); ++it) {
>+  for (ResolverMap::iterator it = resolvers_.begin(); it != resolvers_.end();
>+       ++it) {
>     // TODO(guoweis): Change to asynchronous DNS resolution to prevent the hang
>     // when passing true to the Destroy() which is a safer way to avoid the code
>     // unloaded before the thread exits. Please see webrtc bug 5139.
>@@ -119,16 +117,14 @@ UDPPort::AddressResolver::~AddressResolver() {
>   }
> }
> 
>-void UDPPort::AddressResolver::Resolve(
>-    const rtc::SocketAddress& address) {
>+void UDPPort::AddressResolver::Resolve(const rtc::SocketAddress& address) {
>   if (resolvers_.find(address) != resolvers_.end())
>     return;
> 
>   rtc::AsyncResolverInterface* resolver =
>       socket_factory_->CreateAsyncResolver();
>-  resolvers_.insert(
>-      std::pair<rtc::SocketAddress, rtc::AsyncResolverInterface*>(
>-          address, resolver));
>+  resolvers_.insert(std::pair<rtc::SocketAddress, rtc::AsyncResolverInterface*>(
>+      address, resolver));
> 
>   resolver->SignalDone.connect(this,
>                                &UDPPort::AddressResolver::OnResolveResult);
>@@ -149,8 +145,8 @@ bool UDPPort::AddressResolver::GetResolvedAddress(
> 
> void UDPPort::AddressResolver::OnResolveResult(
>     rtc::AsyncResolverInterface* resolver) {
>-  for (ResolverMap::iterator it = resolvers_.begin();
>-       it != resolvers_.end(); ++it) {
>+  for (ResolverMap::iterator it = resolvers_.begin(); it != resolvers_.end();
>+       ++it) {
>     if (it->second == resolver) {
>       SignalDone(it->first, resolver->GetError());
>       return;
>@@ -166,17 +162,12 @@ UDPPort::UDPPort(rtc::Thread* thread,
>                  const std::string& password,
>                  const std::string& origin,
>                  bool emit_local_for_anyaddress)
>-    : Port(thread,
>-           LOCAL_PORT_TYPE,
>-           factory,
>-           network,
>-           username,
>-           password),
>+    : Port(thread, LOCAL_PORT_TYPE, factory, network, username, password),
>       requests_(thread),
>       socket_(socket),
>       error_(0),
>       ready_(false),
>-      stun_keepalive_delay_(KEEPALIVE_DELAY),
>+      stun_keepalive_delay_(STUN_KEEPALIVE_INTERVAL),
>       emit_local_for_anyaddress_(emit_local_for_anyaddress) {
>   requests_.set_origin(origin);
> }
>@@ -202,7 +193,7 @@ UDPPort::UDPPort(rtc::Thread* thread,
>       socket_(NULL),
>       error_(0),
>       ready_(false),
>-      stun_keepalive_delay_(KEEPALIVE_DELAY),
>+      stun_keepalive_delay_(STUN_KEEPALIVE_INTERVAL),
>       emit_local_for_anyaddress_(emit_local_for_anyaddress) {
>   requests_.set_origin(origin);
> }
>@@ -214,7 +205,7 @@ bool UDPPort::Init() {
>     socket_ = socket_factory()->CreateUdpSocket(
>         rtc::SocketAddress(Network()->GetBestIP(), 0), min_port(), max_port());
>     if (!socket_) {
>-      LOG_J(LS_WARNING, this) << "UDP socket creation failed";
>+      RTC_LOG(LS_WARNING) << ToString() << ": UDP socket creation failed";
>       return false;
>     }
>     socket_->SignalReadPacket.connect(this, &UDPPort::OnReadPacket);
>@@ -269,15 +260,18 @@ Connection* UDPPort::CreateConnection(const Candidate& address,
>   return conn;
> }
> 
>-int UDPPort::SendTo(const void* data, size_t size,
>+int UDPPort::SendTo(const void* data,
>+                    size_t size,
>                     const rtc::SocketAddress& addr,
>                     const rtc::PacketOptions& options,
>                     bool payload) {
>-  int sent = socket_->SendTo(data, size, addr, options);
>+  rtc::PacketOptions modified_options(options);
>+  CopyPortInformationToPacketInfo(&modified_options.info_signaled_after_sent);
>+  int sent = socket_->SendTo(data, size, addr, modified_options);
>   if (sent < 0) {
>     error_ = socket_->GetError();
>-    LOG_J(LS_ERROR, this) << "UDP send of " << size
>-                          << " bytes failed with error " << error_;
>+    RTC_LOG(LS_ERROR) << ToString() << ": UDP send of " << size
>+                      << " bytes failed with error " << error_;
>   }
>   return sent;
> }
>@@ -317,6 +311,14 @@ ProtocolType UDPPort::GetProtocol() const {
>   return PROTO_UDP;
> }
> 
>+void UDPPort::GetStunStats(absl::optional<StunStats>* stats) {
>+  *stats = stats_;
>+}
>+
>+void UDPPort::set_stun_keepalive_delay(const absl::optional<int>& delay) {
>+  stun_keepalive_delay_ = delay.value_or(STUN_KEEPALIVE_INTERVAL);
>+}
>+
> void UDPPort::OnLocalAddressReady(rtc::AsyncPacketSocket* socket,
>                                   const rtc::SocketAddress& address) {
>   // When adapter enumeration is disabled and binding to the any address, the
>@@ -384,20 +386,20 @@ void UDPPort::ResolveStunAddress(const rtc::SocketAddress& stun_addr) {
>     resolver_->SignalDone.connect(this, &UDPPort::OnResolveResult);
>   }
> 
>-  LOG_J(LS_INFO, this) << "Starting STUN host lookup for "
>-                       << stun_addr.ToSensitiveString();
>+  RTC_LOG(LS_INFO) << ToString() << ": Starting STUN host lookup for "
>+                   << stun_addr.ToSensitiveString();
>   resolver_->Resolve(stun_addr);
> }
> 
>-void UDPPort::OnResolveResult(const rtc::SocketAddress& input,
>-                              int error) {
>+void UDPPort::OnResolveResult(const rtc::SocketAddress& input, int error) {
>   RTC_DCHECK(resolver_.get() != NULL);
> 
>   rtc::SocketAddress resolved;
>   if (error != 0 || !resolver_->GetResolvedAddress(
>                         input, Network()->GetBestIP().family(), &resolved)) {
>-    LOG_J(LS_WARNING, this) << "StunPort: stun host lookup received error "
>-                            << error;
>+    RTC_LOG(LS_WARNING) << ToString()
>+                        << ": StunPort: stun host lookup received error "
>+                        << error;
>     OnStunBindingOrResolveRequestFailed(input);
>     return;
>   }
>@@ -446,14 +448,19 @@ bool UDPPort::MaybeSetDefaultLocalAddress(rtc::SocketAddress* addr) const {
> }
> 
> void UDPPort::OnStunBindingRequestSucceeded(
>+    int rtt_ms,
>     const rtc::SocketAddress& stun_server_addr,
>     const rtc::SocketAddress& stun_reflected_addr) {
>+  RTC_DCHECK(stats_.stun_binding_responses_received <
>+             stats_.stun_binding_requests_sent);
>+  stats_.stun_binding_responses_received++;
>+  stats_.stun_binding_rtt_ms_total += rtt_ms;
>+  stats_.stun_binding_rtt_ms_squared_total += rtt_ms * rtt_ms;
>   if (bind_request_succeeded_servers_.find(stun_server_addr) !=
>-          bind_request_succeeded_servers_.end()) {
>+      bind_request_succeeded_servers_.end()) {
>     return;
>   }
>   bind_request_succeeded_servers_.insert(stun_server_addr);
>-
>   // If socket is shared and |stun_reflected_addr| is equal to local socket
>   // address, or if the same address has been added by another STUN server,
>   // then discarding the stun address.
>@@ -463,8 +470,8 @@ void UDPPort::OnStunBindingRequestSucceeded(
>     rtc::SocketAddress related_address = socket_->GetLocalAddress();
>     // If we can't stamp the related address correctly, empty it to avoid leak.
>     if (!MaybeSetDefaultLocalAddress(&related_address)) {
>-      related_address = rtc::EmptySocketAddressWithFamily(
>-          related_address.family());
>+      related_address =
>+          rtc::EmptySocketAddressWithFamily(related_address.family());
>     }
> 
>     std::ostringstream url;
>@@ -480,7 +487,7 @@ void UDPPort::OnStunBindingRequestSucceeded(
> void UDPPort::OnStunBindingOrResolveRequestFailed(
>     const rtc::SocketAddress& stun_server_addr) {
>   if (bind_request_failed_servers_.find(stun_server_addr) !=
>-          bind_request_failed_servers_.end()) {
>+      bind_request_failed_servers_.end()) {
>     return;
>   }
>   bind_request_failed_servers_.insert(stun_server_addr);
>@@ -492,7 +499,8 @@ void UDPPort::MaybeSetPortCompleteOrError() {
>     return;
> 
>   // Do not set port ready if we are still waiting for bind responses.
>-  const size_t servers_done_bind_request = bind_request_failed_servers_.size() +
>+  const size_t servers_done_bind_request =
>+      bind_request_failed_servers_.size() +
>       bind_request_succeeded_servers_.size();
>   if (server_addresses_.size() != servers_done_bind_request) {
>     return;
>@@ -503,8 +511,7 @@ void UDPPort::MaybeSetPortCompleteOrError() {
> 
>   // The port is "completed" if there is no stun server provided, or the bind
>   // request succeeded for any stun server, or the socket is shared.
>-  if (server_addresses_.empty() ||
>-      bind_request_succeeded_servers_.size() > 0 ||
>+  if (server_addresses_.empty() || bind_request_succeeded_servers_.size() > 0 ||
>       SharedSocket()) {
>     SignalPortComplete(this);
>   } else {
>@@ -516,8 +523,12 @@ void UDPPort::MaybeSetPortCompleteOrError() {
> void UDPPort::OnSendPacket(const void* data, size_t size, StunRequest* req) {
>   StunBindingRequest* sreq = static_cast<StunBindingRequest*>(req);
>   rtc::PacketOptions options(DefaultDscpValue());
>-  if (socket_->SendTo(data, size, sreq->server_addr(), options) < 0)
>-    RTC_PLOG(LERROR, socket_->GetError()) << "sendto";
>+  options.info_signaled_after_sent.packet_type = rtc::PacketType::kStunMessage;
>+  CopyPortInformationToPacketInfo(&options.info_signaled_after_sent);
>+  if (socket_->SendTo(data, size, sreq->server_addr(), options) < 0) {
>+    RTC_LOG_ERR_EX(LERROR, socket_->GetError()) << "sendto";
>+  }
>+  stats_.stun_binding_requests_sent++;
> }
> 
> bool UDPPort::HasCandidateWithAddress(const rtc::SocketAddress& addr) const {
>@@ -538,9 +549,11 @@ StunPort* StunPort::Create(rtc::Thread* thread,
>                            const std::string& username,
>                            const std::string& password,
>                            const ServerAddresses& servers,
>-                           const std::string& origin) {
>+                           const std::string& origin,
>+                           absl::optional<int> stun_keepalive_interval) {
>   StunPort* port = new StunPort(thread, factory, network, min_port, max_port,
>                                 username, password, servers, origin);
>+  port->set_stun_keepalive_delay(stun_keepalive_interval);
>   if (!port->Init()) {
>     delete port;
>     port = NULL;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunport.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunport.h
>index cf45d1df596..9cb479df0e2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunport.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunport.h
>@@ -42,9 +42,11 @@ class UDPPort : public Port {
>                          const std::string& username,
>                          const std::string& password,
>                          const std::string& origin,
>-                         bool emit_local_for_anyaddress) {
>+                         bool emit_local_for_anyaddress,
>+                         absl::optional<int> stun_keepalive_interval) {
>     UDPPort* port = new UDPPort(thread, factory, network, socket, username,
>                                 password, origin, emit_local_for_anyaddress);
>+    port->set_stun_keepalive_delay(stun_keepalive_interval);
>     if (!port->Init()) {
>       delete port;
>       port = NULL;
>@@ -60,10 +62,12 @@ class UDPPort : public Port {
>                          const std::string& username,
>                          const std::string& password,
>                          const std::string& origin,
>-                         bool emit_local_for_anyaddress) {
>+                         bool emit_local_for_anyaddress,
>+                         absl::optional<int> stun_keepalive_interval) {
>     UDPPort* port =
>         new UDPPort(thread, factory, network, min_port, max_port, username,
>                     password, origin, emit_local_for_anyaddress);
>+    port->set_stun_keepalive_delay(stun_keepalive_interval);
>     if (!port->Init()) {
>       delete port;
>       port = NULL;
>@@ -101,9 +105,9 @@ class UDPPort : public Port {
>   bool SupportsProtocol(const std::string& protocol) const override;
>   ProtocolType GetProtocol() const override;
> 
>-  void set_stun_keepalive_delay(int delay) {
>-    stun_keepalive_delay_ = delay;
>-  }
>+  void GetStunStats(absl::optional<StunStats>* stats) override;
>+
>+  void set_stun_keepalive_delay(const absl::optional<int>& delay);
>   int stun_keepalive_delay() const {
>     return stun_keepalive_delay_;
>   }
>@@ -208,6 +212,7 @@ class UDPPort : public Port {
> 
>   // Below methods handles binding request responses.
>   void OnStunBindingRequestSucceeded(
>+      int rtt_ms,
>       const rtc::SocketAddress& stun_server_addr,
>       const rtc::SocketAddress& stun_reflected_addr);
>   void OnStunBindingOrResolveRequestFailed(
>@@ -242,6 +247,8 @@ class UDPPort : public Port {
>   int stun_keepalive_delay_;
>   int stun_keepalive_lifetime_ = INFINITE_LIFETIME;
> 
>+  StunStats stats_;
>+
>   // This is true by default and false when
>   // PORTALLOCATOR_DISABLE_DEFAULT_LOCAL_CANDIDATE is specified.
>   bool emit_local_for_anyaddress_;
>@@ -259,7 +266,8 @@ class StunPort : public UDPPort {
>                           const std::string& username,
>                           const std::string& password,
>                           const ServerAddresses& servers,
>-                          const std::string& origin);
>+                          const std::string& origin,
>+                          absl::optional<int> stun_keepalive_interval);
> 
>   void PrepareAddress() override;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunport_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunport_unittest.cc
>index e9acdb11a27..2c18bd8d8f3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunport_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunport_unittest.cc
>@@ -74,7 +74,7 @@ class StunPortTestBase : public testing::Test, public sigslot::has_slots<> {
>     stun_port_.reset(cricket::StunPort::Create(
>         rtc::Thread::Current(), &socket_factory_, &network_, 0, 0,
>         rtc::CreateRandomString(16), rtc::CreateRandomString(22), stun_servers,
>-        std::string()));
>+        std::string(), absl::nullopt));
>     stun_port_->set_stun_keepalive_delay(stun_keepalive_delay_);
>     // If |stun_keepalive_lifetime_| is negative, let the stun port
>     // choose its lifetime from the network type.
>@@ -92,10 +92,9 @@ class StunPortTestBase : public testing::Test, public sigslot::has_slots<> {
>     ASSERT_TRUE(socket_ != NULL);
>     socket_->SignalReadPacket.connect(this, &StunPortTestBase::OnReadPacket);
>     stun_port_.reset(cricket::UDPPort::Create(
>-        rtc::Thread::Current(), &socket_factory_,
>-        &network_, socket_.get(),
>-        rtc::CreateRandomString(16), rtc::CreateRandomString(22),
>-        std::string(), false));
>+        rtc::Thread::Current(), &socket_factory_, &network_, socket_.get(),
>+        rtc::CreateRandomString(16), rtc::CreateRandomString(22), std::string(),
>+        false, absl::nullopt));
>     ASSERT_TRUE(stun_port_ != NULL);
>     ServerAddresses stun_servers;
>     stun_servers.insert(server_addr);
>@@ -105,21 +104,21 @@ class StunPortTestBase : public testing::Test, public sigslot::has_slots<> {
>     stun_port_->SignalPortError.connect(this, &StunPortTestBase::OnPortError);
>   }
> 
>-  void PrepareAddress() {
>-    stun_port_->PrepareAddress();
>-  }
>+  void PrepareAddress() { stun_port_->PrepareAddress(); }
> 
>-  void OnReadPacket(rtc::AsyncPacketSocket* socket, const char* data,
>-                    size_t size, const rtc::SocketAddress& remote_addr,
>+  void OnReadPacket(rtc::AsyncPacketSocket* socket,
>+                    const char* data,
>+                    size_t size,
>+                    const rtc::SocketAddress& remote_addr,
>                     const rtc::PacketTime& packet_time) {
>-    stun_port_->HandleIncomingPacket(
>-        socket, data, size, remote_addr, rtc::PacketTime());
>+    stun_port_->HandleIncomingPacket(socket, data, size, remote_addr,
>+                                     rtc::PacketTime());
>   }
> 
>   void SendData(const char* data, size_t len) {
>-    stun_port_->HandleIncomingPacket(
>-        socket_.get(), data, len, rtc::SocketAddress("22.22.22.22", 0),
>-        rtc::PacketTime());
>+    stun_port_->HandleIncomingPacket(socket_.get(), data, len,
>+                                     rtc::SocketAddress("22.22.22.22", 0),
>+                                     rtc::PacketTime());
>   }
> 
>  protected:
>@@ -137,20 +136,14 @@ class StunPortTestBase : public testing::Test, public sigslot::has_slots<> {
>     done_ = true;
>     error_ = true;
>   }
>-  void SetKeepaliveDelay(int delay) {
>-    stun_keepalive_delay_ = delay;
>-  }
>+  void SetKeepaliveDelay(int delay) { stun_keepalive_delay_ = delay; }
> 
>   void SetKeepaliveLifetime(int lifetime) {
>     stun_keepalive_lifetime_ = lifetime;
>   }
> 
>-  cricket::TestStunServer* stun_server_1() {
>-    return stun_server_1_.get();
>-  }
>-  cricket::TestStunServer* stun_server_2() {
>-    return stun_server_2_.get();
>-  }
>+  cricket::TestStunServer* stun_server_1() { return stun_server_1_.get(); }
>+  cricket::TestStunServer* stun_server_2() { return stun_server_2_.get(); }
> 
>  private:
>   std::unique_ptr<rtc::VirtualSocketServer> ss_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunrequest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunrequest.cc
>index 851bd70b890..56410bea089 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunrequest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunrequest.cc
>@@ -14,10 +14,10 @@
> #include <memory>
> #include <vector>
> 
>+#include "absl/memory/memory.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/helpers.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/stringencode.h"
> 
> namespace cricket {
>@@ -41,13 +41,11 @@ const int STUN_MAX_RETRANSMISSIONS = 8;  // Total sends: 9
> // work well.
> const int STUN_MAX_RTO = 8000;  // milliseconds, or 5 doublings
> 
>-StunRequestManager::StunRequestManager(rtc::Thread* thread)
>-    : thread_(thread) {
>-}
>+StunRequestManager::StunRequestManager(rtc::Thread* thread) : thread_(thread) {}
> 
> StunRequestManager::~StunRequestManager() {
>   while (requests_.begin() != requests_.end()) {
>-    StunRequest *request = requests_.begin()->second;
>+    StunRequest* request = requests_.begin()->second;
>     requests_.erase(requests_.begin());
>     delete request;
>   }
>@@ -167,17 +165,17 @@ bool StunRequestManager::CheckResponse(const char* data, size_t size) {
> }
> 
> StunRequest::StunRequest()
>-    : count_(0), timeout_(false), manager_(0),
>-      msg_(new StunMessage()), tstamp_(0) {
>-  msg_->SetTransactionID(
>-      rtc::CreateRandomString(kStunTransactionIdLength));
>+    : count_(0),
>+      timeout_(false),
>+      manager_(0),
>+      msg_(new StunMessage()),
>+      tstamp_(0) {
>+  msg_->SetTransactionID(rtc::CreateRandomString(kStunTransactionIdLength));
> }
> 
> StunRequest::StunRequest(StunMessage* request)
>-    : count_(0), timeout_(false), manager_(0),
>-      msg_(request), tstamp_(0) {
>-  msg_->SetTransactionID(
>-      rtc::CreateRandomString(kStunTransactionIdLength));
>+    : count_(0), timeout_(false), manager_(0), msg_(request), tstamp_(0) {
>+  msg_->SetTransactionID(rtc::CreateRandomString(kStunTransactionIdLength));
> }
> 
> StunRequest::~StunRequest() {
>@@ -192,8 +190,8 @@ StunRequest::~StunRequest() {
> void StunRequest::Construct() {
>   if (msg_->type() == 0) {
>     if (!origin_.empty()) {
>-      msg_->AddAttribute(
>-          rtc::MakeUnique<StunByteStringAttribute>(STUN_ATTR_ORIGIN, origin_));
>+      msg_->AddAttribute(absl::make_unique<StunByteStringAttribute>(
>+          STUN_ATTR_ORIGIN, origin_));
>     }
>     Prepare(msg_);
>     RTC_DCHECK(msg_->type() != 0);
>@@ -217,7 +215,6 @@ int StunRequest::Elapsed() const {
>   return static_cast<int>(rtc::TimeMillis() - tstamp_);
> }
> 
>-
> void StunRequest::set_manager(StunRequestManager* manager) {
>   RTC_DCHECK(!manager_);
>   manager_ = manager;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunrequest.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunrequest.h
>index 79b2f306d13..a88b36fecc0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunrequest.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunrequest.h
>@@ -14,7 +14,7 @@
> #include <map>
> #include <string>
> #include "p2p/base/stun.h"
>-#include "rtc_base/sigslot.h"
>+#include "rtc_base/third_party/sigslot/sigslot.h"
> #include "rtc_base/thread.h"
> 
> namespace cricket {
>@@ -118,11 +118,11 @@ class StunRequest : public rtc::MessageHandler {
> 
>   // Fills in a request object to be sent.  Note that request's transaction ID
>   // will already be set and cannot be changed.
>-  virtual void Prepare(StunMessage*) {}
>+  virtual void Prepare(StunMessage* request) {}
> 
>   // Called when the message receives a response or times out.
>-  virtual void OnResponse(StunMessage*) {}
>-  virtual void OnErrorResponse(StunMessage*) {}
>+  virtual void OnResponse(StunMessage* response) {}
>+  virtual void OnErrorResponse(StunMessage* response) {}
>   virtual void OnTimeout() {}
>   // Called when the message is sent.
>   virtual void OnSent();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunrequest_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunrequest_unittest.cc
>index f777b85f854..07b40d68502 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunrequest_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunrequest_unittest.cc
>@@ -20,13 +20,15 @@
> 
> namespace cricket {
> 
>-class StunRequestTest : public testing::Test,
>-                        public sigslot::has_slots<> {
>+class StunRequestTest : public testing::Test, public sigslot::has_slots<> {
>  public:
>   StunRequestTest()
>       : manager_(rtc::Thread::Current()),
>-        request_count_(0), response_(NULL),
>-        success_(false), failure_(false), timeout_(false) {
>+        request_count_(0),
>+        response_(NULL),
>+        success_(false),
>+        failure_(false),
>+        timeout_(false) {
>     manager_.SignalSendPacket.connect(this, &StunRequestTest::OnSendPacket);
>   }
> 
>@@ -42,9 +44,7 @@ class StunRequestTest : public testing::Test,
>     response_ = res;
>     failure_ = true;
>   }
>-  void OnTimeout() {
>-    timeout_ = true;
>-  }
>+  void OnTimeout() { timeout_ = true; }
> 
>  protected:
>   static StunMessage* CreateStunMessage(StunMessageType type,
>@@ -76,16 +76,13 @@ class StunRequestThunker : public StunRequest {
>   StunRequestThunker(StunMessage* msg, StunRequestTest* test)
>       : StunRequest(msg), test_(test) {}
>   explicit StunRequestThunker(StunRequestTest* test) : test_(test) {}
>+
>  private:
>-  virtual void OnResponse(StunMessage* res) {
>-    test_->OnResponse(res);
>-  }
>+  virtual void OnResponse(StunMessage* res) { test_->OnResponse(res); }
>   virtual void OnErrorResponse(StunMessage* res) {
>     test_->OnErrorResponse(res);
>   }
>-  virtual void OnTimeout() {
>-    test_->OnTimeout();
>-  }
>+  virtual void OnTimeout() { test_->OnTimeout(); }
> 
>   virtual void Prepare(StunMessage* request) {
>     request->SetType(STUN_BINDING_REQUEST);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunserver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunserver.cc
>index a9f0e99cd13..50cb6546744 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunserver.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunserver.cc
>@@ -25,10 +25,11 @@ StunServer::~StunServer() {
>   socket_->SignalReadPacket.disconnect(this);
> }
> 
>-void StunServer::OnPacket(
>-    rtc::AsyncPacketSocket* socket, const char* buf, size_t size,
>-    const rtc::SocketAddress& remote_addr,
>-    const rtc::PacketTime& packet_time) {
>+void StunServer::OnPacket(rtc::AsyncPacketSocket* socket,
>+                          const char* buf,
>+                          size_t size,
>+                          const rtc::SocketAddress& remote_addr,
>+                          const rtc::PacketTime& packet_time) {
>   // Parse the STUN message; eat any messages that fail to parse.
>   rtc::ByteBufferReader bbuf(buf, size);
>   StunMessage msg;
>@@ -50,16 +51,17 @@ void StunServer::OnPacket(
>   }
> }
> 
>-void StunServer::OnBindingRequest(
>-    StunMessage* msg, const rtc::SocketAddress& remote_addr) {
>+void StunServer::OnBindingRequest(StunMessage* msg,
>+                                  const rtc::SocketAddress& remote_addr) {
>   StunMessage response;
>-  GetStunBindReqponse(msg, remote_addr, &response);
>+  GetStunBindResponse(msg, remote_addr, &response);
>   SendResponse(response, remote_addr);
> }
> 
>-void StunServer::SendErrorResponse(
>-    const StunMessage& msg, const rtc::SocketAddress& addr,
>-    int error_code, const char* error_desc) {
>+void StunServer::SendErrorResponse(const StunMessage& msg,
>+                                   const rtc::SocketAddress& addr,
>+                                   int error_code,
>+                                   const char* error_desc) {
>   StunMessage err_msg;
>   err_msg.SetType(GetStunErrorResponseType(msg.type()));
>   err_msg.SetTransactionID(msg.transaction_id());
>@@ -72,8 +74,8 @@ void StunServer::SendErrorResponse(
>   SendResponse(err_msg, addr);
> }
> 
>-void StunServer::SendResponse(
>-    const StunMessage& msg, const rtc::SocketAddress& addr) {
>+void StunServer::SendResponse(const StunMessage& msg,
>+                              const rtc::SocketAddress& addr) {
>   rtc::ByteBufferWriter buf;
>   msg.Write(&buf);
>   rtc::PacketOptions options;
>@@ -81,7 +83,7 @@ void StunServer::SendResponse(
>     RTC_LOG_ERR(LS_ERROR) << "sendto";
> }
> 
>-void StunServer::GetStunBindReqponse(StunMessage* request,
>+void StunServer::GetStunBindResponse(StunMessage* request,
>                                      const rtc::SocketAddress& remote_addr,
>                                      StunMessage* response) const {
>   response->SetType(STUN_BINDING_RESPONSE);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunserver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunserver.h
>index 7489ce94169..5d0f1305abb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunserver.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunserver.h
>@@ -29,32 +29,30 @@ class StunServer : public sigslot::has_slots<> {
> 
>  protected:
>   // Slot for AsyncSocket.PacketRead:
>-  void OnPacket(
>-      rtc::AsyncPacketSocket* socket, const char* buf, size_t size,
>-      const rtc::SocketAddress& remote_addr,
>-      const rtc::PacketTime& packet_time);
>+  void OnPacket(rtc::AsyncPacketSocket* socket,
>+                const char* buf,
>+                size_t size,
>+                const rtc::SocketAddress& remote_addr,
>+                const rtc::PacketTime& packet_time);
> 
>   // Handlers for the different types of STUN/TURN requests:
>   virtual void OnBindingRequest(StunMessage* msg,
>-      const rtc::SocketAddress& addr);
>-  void OnAllocateRequest(StunMessage* msg,
>-      const rtc::SocketAddress& addr);
>-  void OnSharedSecretRequest(StunMessage* msg,
>-      const rtc::SocketAddress& addr);
>-  void OnSendRequest(StunMessage* msg,
>-      const rtc::SocketAddress& addr);
>+                                const rtc::SocketAddress& addr);
>+  void OnAllocateRequest(StunMessage* msg, const rtc::SocketAddress& addr);
>+  void OnSharedSecretRequest(StunMessage* msg, const rtc::SocketAddress& addr);
>+  void OnSendRequest(StunMessage* msg, const rtc::SocketAddress& addr);
> 
>   // Sends an error response to the given message back to the user.
>-  void SendErrorResponse(
>-      const StunMessage& msg, const rtc::SocketAddress& addr,
>-      int error_code, const char* error_desc);
>+  void SendErrorResponse(const StunMessage& msg,
>+                         const rtc::SocketAddress& addr,
>+                         int error_code,
>+                         const char* error_desc);
> 
>   // Sends the given message to the appropriate destination.
>-  void SendResponse(const StunMessage& msg,
>-       const rtc::SocketAddress& addr);
>+  void SendResponse(const StunMessage& msg, const rtc::SocketAddress& addr);
> 
>   // A helper method to compose a STUN binding response.
>-  void GetStunBindReqponse(StunMessage* request,
>+  void GetStunBindResponse(StunMessage* request,
>                            const rtc::SocketAddress& remote_addr,
>                            StunMessage* response) const;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunserver_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunserver_unittest.cc
>index d1b65b2da31..eb94e720c12 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunserver_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunserver_unittest.cc
>@@ -11,10 +11,10 @@
> #include <memory>
> #include <string>
> 
>+#include "absl/memory/memory.h"
> #include "p2p/base/stunserver.h"
> #include "rtc_base/gunit.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/testclient.h"
> #include "rtc_base/thread.h"
> #include "rtc_base/virtualsocketserver.h"
>@@ -30,10 +30,10 @@ class StunServerTest : public testing::Test {
>  public:
>   StunServerTest() : ss_(new rtc::VirtualSocketServer()), network_(ss_.get()) {}
>   virtual void SetUp() {
>-    server_.reset(new StunServer(
>-        rtc::AsyncUDPSocket::Create(ss_.get(), server_addr)));
>+    server_.reset(
>+        new StunServer(rtc::AsyncUDPSocket::Create(ss_.get(), server_addr)));
>     client_.reset(new rtc::TestClient(
>-        WrapUnique(rtc::AsyncUDPSocket::Create(ss_.get(), client_addr))));
>+        absl::WrapUnique(rtc::AsyncUDPSocket::Create(ss_.get(), client_addr))));
> 
>     network_.Start();
>   }
>@@ -45,9 +45,7 @@ class StunServerTest : public testing::Test {
>   void Send(const char* buf, int len) {
>     client_->SendTo(buf, len, server_addr);
>   }
>-  bool ReceiveFails() {
>-    return(client_->CheckNoPacket());
>-  }
>+  bool ReceiveFails() { return (client_->CheckNoPacket()); }
>   StunMessage* Receive() {
>     StunMessage* msg = NULL;
>     std::unique_ptr<rtc::TestClient::Packet> packet =
>@@ -89,8 +87,9 @@ TEST_F(StunServerTest, TestGood) {
>   EXPECT_EQ(1, mapped_addr->family());
>   EXPECT_EQ(client_addr.port(), mapped_addr->port());
>   if (mapped_addr->ipaddr() != client_addr.ipaddr()) {
>-    RTC_LOG(LS_WARNING) << "Warning: mapped IP (" << mapped_addr->ipaddr()
>-                        << ") != local IP (" << client_addr.ipaddr() << ")";
>+    RTC_LOG(LS_WARNING) << "Warning: mapped IP ("
>+                        << mapped_addr->ipaddr().ToString() << ") != local IP ("
>+                        << client_addr.ipaddr().ToString() << ")";
>   }
> 
>   delete msg;
>@@ -99,9 +98,10 @@ TEST_F(StunServerTest, TestGood) {
> #endif  // if !defined(THREAD_SANITIZER)
> 
> TEST_F(StunServerTest, TestBad) {
>-  const char* bad = "this is a completely nonsensical message whose only "
>-                    "purpose is to make the parser go 'ack'.  it doesn't "
>-                    "look anything like a normal stun message";
>+  const char* bad =
>+      "this is a completely nonsensical message whose only "
>+      "purpose is to make the parser go 'ack'.  it doesn't "
>+      "look anything like a normal stun message";
>   Send(bad, static_cast<int>(strlen(bad)));
> 
>   ASSERT_TRUE(ReceiveFails());
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/tcpport.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/tcpport.cc
>index 89d60d8d1b1..d070f6ecee8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/tcpport.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/tcpport.cc
>@@ -68,7 +68,6 @@
> 
> #include <vector>
> 
>-#include "p2p/base/common.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> 
>@@ -141,8 +140,7 @@ Connection* TCPPort::CreateConnection(const Candidate& address,
>   }
> 
>   TCPConnection* conn = NULL;
>-  if (rtc::AsyncPacketSocket* socket =
>-      GetIncoming(address.address(), true)) {
>+  if (rtc::AsyncPacketSocket* socket = GetIncoming(address.address(), true)) {
>     // Incoming connection; we already created a socket and connected signals,
>     // so we need to hand off the "read packet" responsibility to
>     // TCPConnection.
>@@ -175,7 +173,8 @@ void TCPPort::PrepareAddress() {
>                  TCPTYPE_PASSIVE_STR, LOCAL_PORT_TYPE,
>                  ICE_TYPE_PREFERENCE_HOST_TCP, 0, "", true);
>   } else {
>-    LOG_J(LS_INFO, this) << "Not listening due to firewall restrictions.";
>+    RTC_LOG(LS_INFO) << ToString()
>+                     << ": Not listening due to firewall restrictions.";
>     // Note: We still add the address, since otherwise the remote side won't
>     // recognize our incoming TCP connections. According to
>     // https://tools.ietf.org/html/rfc6544#section-4.5, for active candidate,
>@@ -191,11 +190,12 @@ void TCPPort::PrepareAddress() {
>   }
> }
> 
>-int TCPPort::SendTo(const void* data, size_t size,
>+int TCPPort::SendTo(const void* data,
>+                    size_t size,
>                     const rtc::SocketAddress& addr,
>                     const rtc::PacketOptions& options,
>                     bool payload) {
>-  rtc::AsyncPacketSocket * socket = NULL;
>+  rtc::AsyncPacketSocket* socket = NULL;
>   TCPConnection* conn = static_cast<TCPConnection*>(GetConnection(addr));
> 
>   // For Connection, this is the code path used by Ping() to establish
>@@ -211,19 +211,21 @@ int TCPPort::SendTo(const void* data, size_t size,
>     socket = GetIncoming(addr);
>   }
>   if (!socket) {
>-    LOG_J(LS_ERROR, this) << "Attempted to send to an unknown destination, "
>-                          << addr.ToSensitiveString();
>+    RTC_LOG(LS_ERROR) << ToString()
>+                      << ": Attempted to send to an unknown destination: "
>+                      << addr.ToSensitiveString();
>     return SOCKET_ERROR;  // TODO(tbd): Set error_
>   }
>-
>-  int sent = socket->Send(data, size, options);
>+  rtc::PacketOptions modified_options(options);
>+  CopyPortInformationToPacketInfo(&modified_options.info_signaled_after_sent);
>+  int sent = socket->Send(data, size, modified_options);
>   if (sent < 0) {
>     error_ = socket->GetError();
>     // Error from this code path for a Connection (instead of from a bare
>     // socket) will not trigger reconnecting. In theory, this shouldn't matter
>     // as OnClose should always be called and set connected to false.
>-    LOG_J(LS_ERROR, this) << "TCP send of " << size
>-                          << " bytes failed with error " << error_;
>+    RTC_LOG(LS_ERROR) << ToString() << ": TCP send of " << size
>+                      << " bytes failed with error " << error_;
>   }
>   return sent;
> }
>@@ -267,8 +269,8 @@ void TCPPort::OnNewConnection(rtc::AsyncPacketSocket* socket,
>   incoming.socket->SignalReadyToSend.connect(this, &TCPPort::OnReadyToSend);
>   incoming.socket->SignalSentPacket.connect(this, &TCPPort::OnSentPacket);
> 
>-  LOG_J(LS_VERBOSE, this) << "Accepted connection from "
>-                          << incoming.addr.ToSensitiveString();
>+  RTC_LOG(LS_VERBOSE) << ToString() << ": Accepted connection from "
>+                      << incoming.addr.ToSensitiveString();
>   incoming_.push_back(incoming);
> }
> 
>@@ -277,16 +279,17 @@ void TCPPort::TryCreateServerSocket() {
>       rtc::SocketAddress(Network()->GetBestIP(), 0), min_port(), max_port(),
>       false /* ssl */);
>   if (!socket_) {
>-    LOG_J(LS_WARNING, this)
>-        << "TCP server socket creation failed; continuing anyway.";
>+    RTC_LOG(LS_WARNING)
>+        << ToString()
>+        << ": TCP server socket creation failed; continuing anyway.";
>     return;
>   }
>   socket_->SignalNewConnection.connect(this, &TCPPort::OnNewConnection);
>   socket_->SignalAddressReady.connect(this, &TCPPort::OnAddressReady);
> }
> 
>-rtc::AsyncPacketSocket* TCPPort::GetIncoming(
>-    const rtc::SocketAddress& addr, bool remove) {
>+rtc::AsyncPacketSocket* TCPPort::GetIncoming(const rtc::SocketAddress& addr,
>+                                             bool remove) {
>   rtc::AsyncPacketSocket* socket = NULL;
>   for (std::list<Incoming>::iterator it = incoming_.begin();
>        it != incoming_.end(); ++it) {
>@@ -301,7 +304,8 @@ rtc::AsyncPacketSocket* TCPPort::GetIncoming(
> }
> 
> void TCPPort::OnReadPacket(rtc::AsyncPacketSocket* socket,
>-                           const char* data, size_t size,
>+                           const char* data,
>+                           size_t size,
>                            const rtc::SocketAddress& remote_addr,
>                            const rtc::PacketTime& packet_time) {
>   Port::OnReadPacket(data, size, remote_addr, PROTO_TCP);
>@@ -323,6 +327,10 @@ void TCPPort::OnAddressReady(rtc::AsyncPacketSocket* socket,
>              0, "", true);
> }
> 
>+// TODO(qingsi): |CONNECTION_WRITE_CONNECT_TIMEOUT| is overriden by
>+// |ice_unwritable_timeout| in IceConfig when determining the writability state.
>+// Replace this constant with the config parameter assuming the default value if
>+// we decide it is also applicable here.
> TCPConnection::TCPConnection(TCPPort* port,
>                              const Candidate& candidate,
>                              rtc::AsyncPacketSocket* socket)
>@@ -338,22 +346,24 @@ TCPConnection::TCPConnection(TCPPort* port,
>   } else {
>     // Incoming connections should match one of the network addresses. Same as
>     // what's being checked in OnConnect, but just DCHECKing here.
>-    LOG_J(LS_VERBOSE, this)
>-        << "socket ipaddr: " << socket_->GetLocalAddress().ToString()
>-        << ", port() Network:" << port->Network()->ToString();
>+    RTC_LOG(LS_VERBOSE) << ToString() << ": socket ipaddr: "
>+                        << socket_->GetLocalAddress().ToString()
>+                        << ", port() Network:" << port->Network()->ToString();
>     const std::vector<rtc::InterfaceAddress>& desired_addresses =
>         port_->Network()->GetIPs();
>-    RTC_DCHECK(socket->GetLocalAddress().IsLoopbackIP() ||
>-        std::find(desired_addresses.begin(), desired_addresses.end(),
>-        socket_->GetLocalAddress().ipaddr()) != desired_addresses.end());
>+    RTC_DCHECK(std::find_if(desired_addresses.begin(), desired_addresses.end(),
>+                            [this](const rtc::InterfaceAddress& addr) {
>+                              return socket_->GetLocalAddress().ipaddr() ==
>+                                     addr;
>+                            }) != desired_addresses.end());
>     ConnectSocketSignals(socket);
>   }
> }
> 
>-TCPConnection::~TCPConnection() {
>-}
>+TCPConnection::~TCPConnection() {}
> 
>-int TCPConnection::Send(const void* data, size_t size,
>+int TCPConnection::Send(const void* data,
>+                        size_t size,
>                         const rtc::PacketOptions& options) {
>   if (!socket_) {
>     error_ = ENOTCONN;
>@@ -377,7 +387,10 @@ int TCPConnection::Send(const void* data, size_t size,
>     return SOCKET_ERROR;
>   }
>   stats_.sent_total_packets++;
>-  int sent = socket_->Send(data, size, options);
>+  rtc::PacketOptions modified_options(options);
>+  static_cast<TCPPort*>(port_)->CopyPortInformationToPacketInfo(
>+      &modified_options.info_signaled_after_sent);
>+  int sent = socket_->Send(data, size, modified_options);
>   if (sent < 0) {
>     stats_.sent_discarded_packets++;
>     error_ = socket_->GetError();
>@@ -425,29 +438,31 @@ void TCPConnection::OnConnect(rtc::AsyncPacketSocket* socket) {
>   const rtc::SocketAddress& socket_address = socket->GetLocalAddress();
>   const std::vector<rtc::InterfaceAddress>& desired_addresses =
>       port_->Network()->GetIPs();
>-  if (std::find(desired_addresses.begin(), desired_addresses.end(),
>-                socket_address.ipaddr()) != desired_addresses.end()) {
>-    LOG_J(LS_VERBOSE, this) << "Connection established to "
>-                            << socket->GetRemoteAddress().ToSensitiveString();
>+  if (std::find_if(desired_addresses.begin(), desired_addresses.end(),
>+                   [socket_address](const rtc::InterfaceAddress& addr) {
>+                     return socket_address.ipaddr() == addr;
>+                   }) != desired_addresses.end()) {
>+    RTC_LOG(LS_VERBOSE) << ToString() << ": Connection established to "
>+                        << socket->GetRemoteAddress().ToSensitiveString();
>   } else {
>     if (socket->GetLocalAddress().IsLoopbackIP()) {
>       RTC_LOG(LS_WARNING) << "Socket is bound to the address:"
>                           << socket_address.ipaddr().ToString()
>-                          << ", rather then an address associated with network:"
>+                          << ", rather than an address associated with network:"
>                           << port_->Network()->ToString()
>                           << ". Still allowing it since it's localhost.";
>     } else if (IPIsAny(port_->Network()->GetBestIP())) {
>       RTC_LOG(LS_WARNING)
>           << "Socket is bound to the address:"
>           << socket_address.ipaddr().ToString()
>-          << ", rather then an address associated with network:"
>+          << ", rather than an address associated with network:"
>           << port_->Network()->ToString()
>           << ". Still allowing it since it's the 'any' address"
>-          << ", possibly caused by multiple_routes being disabled.";
>+             ", possibly caused by multiple_routes being disabled.";
>     } else {
>       RTC_LOG(LS_WARNING) << "Dropping connection as TCP socket bound to IP "
>                           << socket_address.ipaddr().ToString()
>-                          << ", rather then an address associated with network:"
>+                          << ", rather than an address associated with network:"
>                           << port_->Network()->ToString();
>       OnClose(socket, 0);
>       return;
>@@ -461,7 +476,7 @@ void TCPConnection::OnConnect(rtc::AsyncPacketSocket* socket) {
> 
> void TCPConnection::OnClose(rtc::AsyncPacketSocket* socket, int error) {
>   RTC_DCHECK(socket == socket_.get());
>-  LOG_J(LS_INFO, this) << "Connection closed with error " << error;
>+  RTC_LOG(LS_INFO) << ToString() << ": Connection closed with error " << error;
> 
>   // Guard against the condition where IPC socket will call OnClose for every
>   // packet it can't send.
>@@ -508,17 +523,19 @@ void TCPConnection::MaybeReconnect() {
>     return;
>   }
> 
>-  LOG_J(LS_INFO, this) << "TCP Connection with remote is closed, "
>-                       << "trying to reconnect";
>+  RTC_LOG(LS_INFO) << ToString()
>+                   << ": TCP Connection with remote is closed, "
>+                      "trying to reconnect";
> 
>   CreateOutgoingTcpSocket();
>   error_ = EPIPE;
> }
> 
>-void TCPConnection::OnReadPacket(
>-  rtc::AsyncPacketSocket* socket, const char* data, size_t size,
>-  const rtc::SocketAddress& remote_addr,
>-  const rtc::PacketTime& packet_time) {
>+void TCPConnection::OnReadPacket(rtc::AsyncPacketSocket* socket,
>+                                 const char* data,
>+                                 size_t size,
>+                                 const rtc::SocketAddress& remote_addr,
>+                                 const rtc::PacketTime& packet_time) {
>   RTC_DCHECK(socket == socket_.get());
>   Connection::OnReadPacket(data, size, packet_time);
> }
>@@ -539,15 +556,16 @@ void TCPConnection::CreateOutgoingTcpSocket() {
>       remote_candidate().address(), port()->proxy(), port()->user_agent(),
>       opts));
>   if (socket_) {
>-    LOG_J(LS_VERBOSE, this)
>-        << "Connecting from " << socket_->GetLocalAddress().ToSensitiveString()
>-        << " to " << remote_candidate().address().ToSensitiveString();
>+    RTC_LOG(LS_VERBOSE) << ToString() << ": Connecting from "
>+                        << socket_->GetLocalAddress().ToSensitiveString()
>+                        << " to "
>+                        << remote_candidate().address().ToSensitiveString();
>     set_connected(false);
>     connection_pending_ = true;
>     ConnectSocketSignals(socket_.get());
>   } else {
>-    LOG_J(LS_WARNING, this) << "Failed to create connection to "
>-                            << remote_candidate().address().ToSensitiveString();
>+    RTC_LOG(LS_WARNING) << ToString() << ": Failed to create connection to "
>+                        << remote_candidate().address().ToSensitiveString();
>   }
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/tcpport.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/tcpport.h
>index d20a8de1df8..fa5c958c21c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/tcpport.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/tcpport.h
>@@ -83,12 +83,13 @@ class TCPPort : public Port {
> 
>   void TryCreateServerSocket();
> 
>-  rtc::AsyncPacketSocket* GetIncoming(
>-      const rtc::SocketAddress& addr, bool remove = false);
>+  rtc::AsyncPacketSocket* GetIncoming(const rtc::SocketAddress& addr,
>+                                      bool remove = false);
> 
>   // Receives packet signal from the local TCP Socket.
>   void OnReadPacket(rtc::AsyncPacketSocket* socket,
>-                    const char* data, size_t size,
>+                    const char* data,
>+                    size_t size,
>                     const rtc::SocketAddress& remote_addr,
>                     const rtc::PacketTime& packet_time);
> 
>@@ -113,7 +114,8 @@ class TCPPort : public Port {
> class TCPConnection : public Connection {
>  public:
>   // Connection is outgoing unless socket is specified
>-  TCPConnection(TCPPort* port, const Candidate& candidate,
>+  TCPConnection(TCPPort* port,
>+                const Candidate& candidate,
>                 rtc::AsyncPacketSocket* socket = 0);
>   ~TCPConnection() override;
> 
>@@ -154,7 +156,8 @@ class TCPConnection : public Connection {
>   void OnConnect(rtc::AsyncPacketSocket* socket);
>   void OnClose(rtc::AsyncPacketSocket* socket, int error);
>   void OnReadPacket(rtc::AsyncPacketSocket* socket,
>-                    const char* data, size_t size,
>+                    const char* data,
>+                    size_t size,
>                     const rtc::SocketAddress& remote_addr,
>                     const rtc::PacketTime& packet_time);
>   void OnReadyToSend(rtc::AsyncPacketSocket* socket);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/tcpport_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/tcpport_unittest.cc
>index 12bb437517a..ef302f90260 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/tcpport_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/tcpport_unittest.cc
>@@ -26,8 +26,12 @@ using cricket::ICE_PWD_LENGTH;
> 
> static int kTimeout = 1000;
> static const SocketAddress kLocalAddr("11.11.11.11", 0);
>+static const SocketAddress kLocalIPv6Addr("2401:fa00:4:1000:be30:5bff:fee5:c3",
>+                                          0);
> static const SocketAddress kAlternateLocalAddr("1.2.3.4", 0);
> static const SocketAddress kRemoteAddr("22.22.22.22", 0);
>+static const SocketAddress kRemoteIPv6Addr("2401:fa00:4:1000:be30:5bff:fee5:c4",
>+                                           0);
> 
> class ConnectionObserver : public sigslot::has_slots<> {
>  public:
>@@ -50,8 +54,7 @@ class TCPPortTest : public testing::Test, public sigslot::has_slots<> {
>         main_(ss_.get()),
>         socket_factory_(rtc::Thread::Current()),
>         username_(rtc::CreateRandomString(ICE_UFRAG_LENGTH)),
>-        password_(rtc::CreateRandomString(ICE_PWD_LENGTH)) {
>-  }
>+        password_(rtc::CreateRandomString(ICE_PWD_LENGTH)) {}
> 
>   rtc::Network* MakeNetwork(const SocketAddress& addr) {
>     networks_.emplace_back("unittest", "unittest", addr.ipaddr(), 32);
>@@ -155,6 +158,25 @@ TEST_F(TCPPortTest, TCPPortNotDiscardedIfNotBoundToBestIP) {
>             local_port->Candidates()[0].address().ipaddr());
> }
> 
>+// Regression test for crbug.com/webrtc/8972, caused by buggy comparison
>+// between rtc::IPAddress and rtc::InterfaceAddress.
>+TEST_F(TCPPortTest, TCPPortNotDiscardedIfBoundToTemporaryIP) {
>+  networks_.emplace_back("unittest", "unittest", kLocalIPv6Addr.ipaddr(), 32);
>+  networks_.back().AddIP(rtc::InterfaceAddress(
>+      kLocalIPv6Addr.ipaddr(), rtc::IPV6_ADDRESS_FLAG_TEMPORARY));
>+
>+  auto local_port = CreateTCPPort(&networks_.back());
>+  auto remote_port = CreateTCPPort(kRemoteIPv6Addr);
>+  local_port->PrepareAddress();
>+  remote_port->PrepareAddress();
>+
>+  // Connection should succeed if the port isn't discarded.
>+  Connection* conn = local_port->CreateConnection(remote_port->Candidates()[0],
>+                                                  Port::ORIGIN_MESSAGE);
>+  ASSERT_NE(nullptr, conn);
>+  EXPECT_TRUE_WAIT(conn->connected(), kTimeout);
>+}
>+
> class SentPacketCounter : public sigslot::has_slots<> {
>  public:
>   explicit SentPacketCounter(TCPPort* p) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/testrelayserver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/testrelayserver.h
>index 0d360cf2487..5efffc8bd04 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/testrelayserver.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/testrelayserver.h
>@@ -15,8 +15,8 @@
> 
> #include "p2p/base/relayserver.h"
> #include "rtc_base/asynctcpsocket.h"
>-#include "rtc_base/sigslot.h"
> #include "rtc_base/socketadapters.h"
>+#include "rtc_base/third_party/sigslot/sigslot.h"
> #include "rtc_base/thread.h"
> 
> namespace cricket {
>@@ -32,19 +32,17 @@ class TestRelayServer : public sigslot::has_slots<> {
>                   const rtc::SocketAddress& ssl_int_addr,
>                   const rtc::SocketAddress& ssl_ext_addr)
>       : server_(thread) {
>-    server_.AddInternalSocket(rtc::AsyncUDPSocket::Create(
>-        thread->socketserver(), udp_int_addr));
>-    server_.AddExternalSocket(rtc::AsyncUDPSocket::Create(
>-        thread->socketserver(), udp_ext_addr));
>+    server_.AddInternalSocket(
>+        rtc::AsyncUDPSocket::Create(thread->socketserver(), udp_int_addr));
>+    server_.AddExternalSocket(
>+        rtc::AsyncUDPSocket::Create(thread->socketserver(), udp_ext_addr));
> 
>     tcp_int_socket_.reset(CreateListenSocket(thread, tcp_int_addr));
>     tcp_ext_socket_.reset(CreateListenSocket(thread, tcp_ext_addr));
>     ssl_int_socket_.reset(CreateListenSocket(thread, ssl_int_addr));
>     ssl_ext_socket_.reset(CreateListenSocket(thread, ssl_ext_addr));
>   }
>-  int GetConnectionCount() const {
>-    return server_.GetConnectionCount();
>-  }
>+  int GetConnectionCount() const { return server_.GetConnectionCount(); }
>   rtc::SocketAddressPair GetConnection(int connection) const {
>     return server_.GetConnection(connection);
>   }
>@@ -54,7 +52,7 @@ class TestRelayServer : public sigslot::has_slots<> {
> 
>  private:
>   rtc::AsyncSocket* CreateListenSocket(rtc::Thread* thread,
>-      const rtc::SocketAddress& addr) {
>+                                       const rtc::SocketAddress& addr) {
>     rtc::AsyncSocket* socket =
>         thread->socketserver()->CreateAsyncSocket(addr.family(), SOCK_STREAM);
>     socket->Bind(addr);
>@@ -63,22 +61,22 @@ class TestRelayServer : public sigslot::has_slots<> {
>     return socket;
>   }
>   void OnAccept(rtc::AsyncSocket* socket) {
>-    bool external = (socket == tcp_ext_socket_.get() ||
>-                     socket == ssl_ext_socket_.get());
>-    bool ssl = (socket == ssl_int_socket_.get() ||
>-                socket == ssl_ext_socket_.get());
>+    bool external =
>+        (socket == tcp_ext_socket_.get() || socket == ssl_ext_socket_.get());
>+    bool ssl =
>+        (socket == ssl_int_socket_.get() || socket == ssl_ext_socket_.get());
>     rtc::AsyncSocket* raw_socket = socket->Accept(NULL);
>     if (raw_socket) {
>       rtc::AsyncTCPSocket* packet_socket = new rtc::AsyncTCPSocket(
>-          (!ssl) ? raw_socket :
>-          new rtc::AsyncSSLServerSocket(raw_socket), false);
>+          (!ssl) ? raw_socket : new rtc::AsyncSSLServerSocket(raw_socket),
>+          false);
>       if (!external) {
>         packet_socket->SignalClose.connect(this,
>-            &TestRelayServer::OnInternalClose);
>+                                           &TestRelayServer::OnInternalClose);
>         server_.AddInternalSocket(packet_socket);
>       } else {
>         packet_socket->SignalClose.connect(this,
>-            &TestRelayServer::OnExternalClose);
>+                                           &TestRelayServer::OnExternalClose);
>         server_.AddExternalSocket(packet_socket);
>       }
>     }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/teststunserver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/teststunserver.cc
>index 13bc5773d94..7a18d7ad80b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/teststunserver.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/teststunserver.cc
>@@ -27,7 +27,7 @@ void TestStunServer::OnBindingRequest(StunMessage* msg,
>     StunServer::OnBindingRequest(msg, remote_addr);
>   } else {
>     StunMessage response;
>-    GetStunBindReqponse(msg, fake_stun_addr_, &response);
>+    GetStunBindResponse(msg, fake_stun_addr_, &response);
>     SendResponse(response, remote_addr);
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/testturncustomizer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/testturncustomizer.h
>index d05569657c7..7ca4579e98a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/testturncustomizer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/testturncustomizer.h
>@@ -11,9 +11,9 @@
> #ifndef P2P_BASE_TESTTURNCUSTOMIZER_H_
> #define P2P_BASE_TESTTURNCUSTOMIZER_H_
> 
>+#include "absl/memory/memory.h"
> #include "api/turncustomizer.h"
> #include "rtc_base/gunit.h"
>-#include "rtc_base/ptr_util.h"
> 
> namespace cricket {
> 
>@@ -24,17 +24,16 @@ class TestTurnCustomizer : public webrtc::TurnCustomizer {
> 
>   enum TestTurnAttributeExtensions {
>     // Test only attribute
>-    STUN_ATTR_COUNTER                     = 0xFF02   // Number
>+    STUN_ATTR_COUNTER = 0xFF02  // Number
>   };
> 
>-  void MaybeModifyOutgoingStunMessage(
>-      cricket::PortInterface* port,
>-      cricket::StunMessage* message) override {
>+  void MaybeModifyOutgoingStunMessage(cricket::PortInterface* port,
>+                                      cricket::StunMessage* message) override {
>     modify_cnt_++;
> 
>     ASSERT_NE(0, message->type());
>     if (add_counter_) {
>-      message->AddAttribute(rtc::MakeUnique<cricket::StunUInt32Attribute>(
>+      message->AddAttribute(absl::make_unique<cricket::StunUInt32Attribute>(
>           STUN_ATTR_COUNTER, modify_cnt_));
>     }
>     return;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/testturnserver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/testturnserver.h
>index 0edc4765113..734ca4d9721 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/testturnserver.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/testturnserver.h
>@@ -21,6 +21,7 @@
> #include "rtc_base/ssladapter.h"
> #include "rtc_base/sslidentity.h"
> #include "rtc_base/thread.h"
>+#include "rtc_base/thread_checker.h"
> 
> namespace cricket {
> 
>@@ -31,8 +32,7 @@ class TestTurnRedirector : public TurnRedirectInterface {
>  public:
>   explicit TestTurnRedirector(const std::vector<rtc::SocketAddress>& addresses)
>       : alternate_server_addresses_(addresses),
>-        iter_(alternate_server_addresses_.begin()) {
>-  }
>+        iter_(alternate_server_addresses_.begin()) {}
> 
>   virtual bool ShouldRedirect(const rtc::SocketAddress&,
>                               rtc::SocketAddress* out) {
>@@ -53,9 +53,11 @@ class TestTurnServer : public TurnAuthInterface {
>   TestTurnServer(rtc::Thread* thread,
>                  const rtc::SocketAddress& int_addr,
>                  const rtc::SocketAddress& udp_ext_addr,
>-                 ProtocolType int_protocol = PROTO_UDP)
>+                 ProtocolType int_protocol = PROTO_UDP,
>+                 bool ignore_bad_cert = true,
>+                 const std::string& common_name = "test turn server")
>       : server_(thread), thread_(thread) {
>-    AddInternalSocket(int_addr, int_protocol);
>+    AddInternalSocket(int_addr, int_protocol, ignore_bad_cert, common_name);
>     server_.SetExternalSocketFactory(new rtc::BasicPacketSocketFactory(thread),
>                                      udp_ext_addr);
>     server_.set_realm(kTestRealm);
>@@ -63,22 +65,33 @@ class TestTurnServer : public TurnAuthInterface {
>     server_.set_auth_hook(this);
>   }
> 
>+  ~TestTurnServer() { RTC_DCHECK(thread_checker_.CalledOnValidThread()); }
>+
>   void set_enable_otu_nonce(bool enable) {
>+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
>     server_.set_enable_otu_nonce(enable);
>   }
> 
>-  TurnServer* server() { return &server_; }
>+  TurnServer* server() {
>+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+    return &server_;
>+  }
> 
>   void set_redirect_hook(TurnRedirectInterface* redirect_hook) {
>+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
>     server_.set_redirect_hook(redirect_hook);
>   }
> 
>   void set_enable_permission_checks(bool enable) {
>+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
>     server_.set_enable_permission_checks(enable);
>   }
> 
>   void AddInternalSocket(const rtc::SocketAddress& int_addr,
>-                         ProtocolType proto) {
>+                         ProtocolType proto,
>+                         bool ignore_bad_cert = true,
>+                         const std::string& common_name = "test turn server") {
>+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
>     if (proto == cricket::PROTO_UDP) {
>       server_.AddInternalSocket(
>           rtc::AsyncUDPSocket::Create(thread_->socketserver(), int_addr),
>@@ -87,7 +100,7 @@ class TestTurnServer : public TurnAuthInterface {
>       // For TCP we need to create a server socket which can listen for incoming
>       // new connections.
>       rtc::AsyncSocket* socket =
>-          thread_->socketserver()->CreateAsyncSocket(SOCK_STREAM);
>+          thread_->socketserver()->CreateAsyncSocket(AF_INET, SOCK_STREAM);
>       if (proto == cricket::PROTO_TLS) {
>         // For TLS, wrap the TCP socket with an SSL adapter. The adapter must
>         // be configured with a self-signed certificate for testing.
>@@ -96,8 +109,8 @@ class TestTurnServer : public TurnAuthInterface {
>         rtc::SSLAdapter* adapter = rtc::SSLAdapter::Create(socket);
>         adapter->SetRole(rtc::SSL_SERVER);
>         adapter->SetIdentity(
>-            rtc::SSLIdentity::Generate("test turn server", rtc::KeyParams()));
>-        adapter->SetIgnoreBadCert(true);
>+            rtc::SSLIdentity::Generate(common_name, rtc::KeyParams()));
>+        adapter->SetIgnoreBadCert(ignore_bad_cert);
>         socket = adapter;
>       }
>       socket->Bind(int_addr);
>@@ -111,9 +124,10 @@ class TestTurnServer : public TurnAuthInterface {
>   // Finds the first allocation in the server allocation map with a source
>   // ip and port matching the socket address provided.
>   TurnServerAllocation* FindAllocation(const rtc::SocketAddress& src) {
>+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
>     const TurnServer::AllocationMap& map = server_.allocations();
>     for (TurnServer::AllocationMap::const_iterator it = map.begin();
>-        it != map.end(); ++it) {
>+         it != map.end(); ++it) {
>       if (src == it->first.src()) {
>         return it->second.get();
>       }
>@@ -124,13 +138,16 @@ class TestTurnServer : public TurnAuthInterface {
>  private:
>   // For this test server, succeed if the password is the same as the username.
>   // Obviously, do not use this in a production environment.
>-  virtual bool GetKey(const std::string& username, const std::string& realm,
>+  virtual bool GetKey(const std::string& username,
>+                      const std::string& realm,
>                       std::string* key) {
>+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
>     return ComputeStunCredentialHash(username, realm, username, key);
>   }
> 
>   TurnServer server_;
>   rtc::Thread* thread_;
>+  rtc::ThreadChecker thread_checker_;
> };
> 
> }  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transportdescription.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transportdescription.cc
>index 9cbf383d5d1..ad29a1199e7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transportdescription.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transportdescription.cc
>@@ -18,11 +18,8 @@ namespace cricket {
> 
> bool StringToConnectionRole(const std::string& role_str, ConnectionRole* role) {
>   const char* const roles[] = {
>-      CONNECTIONROLE_ACTIVE_STR,
>-      CONNECTIONROLE_PASSIVE_STR,
>-      CONNECTIONROLE_ACTPASS_STR,
>-      CONNECTIONROLE_HOLDCONN_STR
>-  };
>+      CONNECTIONROLE_ACTIVE_STR, CONNECTIONROLE_PASSIVE_STR,
>+      CONNECTIONROLE_ACTPASS_STR, CONNECTIONROLE_HOLDCONN_STR};
> 
>   for (size_t i = 0; i < arraysize(roles); ++i) {
>     if (_stricmp(roles[i], role_str.c_str()) == 0) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transportdescription.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transportdescription.h
>index 664b7d3a23b..3bffdf971a0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transportdescription.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transportdescription.h
>@@ -30,18 +30,10 @@ namespace cricket {
> //               or unsupported crypto.
> // TODO(deadbeef): Remove this or rename it to something more appropriate, like
> // SdesPolicy.
>-enum SecurePolicy {
>-  SEC_DISABLED,
>-  SEC_ENABLED,
>-  SEC_REQUIRED
>-};
>+enum SecurePolicy { SEC_DISABLED, SEC_ENABLED, SEC_REQUIRED };
> 
> // Whether our side of the call is driving the negotiation, or the other side.
>-enum IceRole {
>-  ICEROLE_CONTROLLING = 0,
>-  ICEROLE_CONTROLLED,
>-  ICEROLE_UNKNOWN
>-};
>+enum IceRole { ICEROLE_CONTROLLING = 0, ICEROLE_CONTROLLED, ICEROLE_UNKNOWN };
> 
> // ICE RFC 5245 implementation type.
> enum IceMode {
>@@ -123,8 +115,7 @@ struct TransportDescription {
>                          HasOption(ICE_OPTION_RENOMINATION));
>   }
> 
>-  static rtc::SSLFingerprint* CopyFingerprint(
>-      const rtc::SSLFingerprint* from) {
>+  static rtc::SSLFingerprint* CopyFingerprint(const rtc::SSLFingerprint* from) {
>     if (!from)
>       return NULL;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transportdescriptionfactory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transportdescriptionfactory.cc
>index d5f8561b6b9..618726e8411 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transportdescriptionfactory.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transportdescriptionfactory.cc
>@@ -21,8 +21,7 @@
> namespace cricket {
> 
> TransportDescriptionFactory::TransportDescriptionFactory()
>-    : secure_(SEC_DISABLED) {
>-}
>+    : secure_(SEC_DISABLED) {}
> 
> TransportDescriptionFactory::~TransportDescriptionFactory() = default;
> 
>@@ -64,7 +63,7 @@ TransportDescription* TransportDescriptionFactory::CreateAnswer(
>   // TODO(juberti): Figure out why we get NULL offers, and fix this upstream.
>   if (!offer) {
>     RTC_LOG(LS_WARNING) << "Failed to create TransportDescription answer "
>-                        << "because offer is NULL";
>+                           "because offer is NULL";
>     return NULL;
>   }
> 
>@@ -89,8 +88,9 @@ TransportDescription* TransportDescriptionFactory::CreateAnswer(
>     if (secure_ == SEC_ENABLED || secure_ == SEC_REQUIRED) {
>       // Fail if we can't create the fingerprint.
>       // Setting DTLS role to active.
>-      ConnectionRole role = (options.prefer_passive_role) ?
>-          CONNECTIONROLE_PASSIVE : CONNECTIONROLE_ACTIVE;
>+      ConnectionRole role = (options.prefer_passive_role)
>+                                ? CONNECTIONROLE_PASSIVE
>+                                : CONNECTIONROLE_ACTIVE;
> 
>       if (!SetSecurityInfo(desc.get(), role)) {
>         return NULL;
>@@ -106,8 +106,8 @@ TransportDescription* TransportDescriptionFactory::CreateAnswer(
>   return desc.release();
> }
> 
>-bool TransportDescriptionFactory::SetSecurityInfo(
>-    TransportDescription* desc, ConnectionRole role) const {
>+bool TransportDescriptionFactory::SetSecurityInfo(TransportDescription* desc,
>+                                                  ConnectionRole role) const {
>   if (!certificate_) {
>     RTC_LOG(LS_ERROR) << "Cannot create identity digest with no certificate";
>     return false;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transportdescriptionfactory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transportdescriptionfactory.h
>index f8b031a683e..937c5fa1fea 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transportdescriptionfactory.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transportdescriptionfactory.h
>@@ -52,7 +52,8 @@ class TransportDescriptionFactory {
>   }
> 
>   // Creates a transport description suitable for use in an offer.
>-  TransportDescription* CreateOffer(const TransportOptions& options,
>+  TransportDescription* CreateOffer(
>+      const TransportOptions& options,
>       const TransportDescription* current_description) const;
>   // Create a transport description that is a response to an offer.
>   //
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transportdescriptionfactory_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transportdescriptionfactory_unittest.cc
>index df01871067e..a7c34b56a04 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transportdescriptionfactory_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transportdescriptionfactory_unittest.cc
>@@ -32,8 +32,10 @@ class TransportDescriptionFactoryTest : public testing::Test {
>             new rtc::FakeSSLIdentity("User2")))) {}
> 
>   void CheckDesc(const TransportDescription* desc,
>-                 const std::string& opt, const std::string& ice_ufrag,
>-                 const std::string& ice_pwd, const std::string& dtls_alg) {
>+                 const std::string& opt,
>+                 const std::string& ice_ufrag,
>+                 const std::string& ice_pwd,
>+                 const std::string& dtls_alg) {
>     ASSERT_TRUE(desc != NULL);
>     EXPECT_EQ(!opt.empty(), desc->HasOption(opt));
>     if (ice_ufrag.empty() && ice_pwd.empty()) {
>@@ -160,8 +162,8 @@ TEST_F(TransportDescriptionFactoryTest, TestOfferDtls) {
>   f1_.set_secure(cricket::SEC_ENABLED);
>   f1_.set_certificate(cert1_);
>   std::string digest_alg;
>-  ASSERT_TRUE(cert1_->ssl_certificate().GetSignatureDigestAlgorithm(
>-      &digest_alg));
>+  ASSERT_TRUE(
>+      cert1_->ssl_certificate().GetSignatureDigestAlgorithm(&digest_alg));
>   std::unique_ptr<TransportDescription> desc(
>       f1_.CreateOffer(TransportOptions(), NULL));
>   CheckDesc(desc.get(), "", "", "", digest_alg);
>@@ -185,15 +187,14 @@ TEST_F(TransportDescriptionFactoryTest, TestOfferDtlsReofferDtls) {
>   f1_.set_secure(cricket::SEC_ENABLED);
>   f1_.set_certificate(cert1_);
>   std::string digest_alg;
>-  ASSERT_TRUE(cert1_->ssl_certificate().GetSignatureDigestAlgorithm(
>-      &digest_alg));
>+  ASSERT_TRUE(
>+      cert1_->ssl_certificate().GetSignatureDigestAlgorithm(&digest_alg));
>   std::unique_ptr<TransportDescription> old_desc(
>       f1_.CreateOffer(TransportOptions(), NULL));
>   ASSERT_TRUE(old_desc.get() != NULL);
>   std::unique_ptr<TransportDescription> desc(
>       f1_.CreateOffer(TransportOptions(), old_desc.get()));
>-  CheckDesc(desc.get(), "",
>-            old_desc->ice_ufrag, old_desc->ice_pwd, digest_alg);
>+  CheckDesc(desc.get(), "", old_desc->ice_ufrag, old_desc->ice_pwd, digest_alg);
> }
> 
> TEST_F(TransportDescriptionFactoryTest, TestAnswerDefault) {
>@@ -218,8 +219,7 @@ TEST_F(TransportDescriptionFactoryTest, TestReanswer) {
>   std::unique_ptr<TransportDescription> desc(
>       f2_.CreateAnswer(offer.get(), TransportOptions(), true, old_desc.get()));
>   ASSERT_TRUE(desc.get() != NULL);
>-  CheckDesc(desc.get(), "",
>-            old_desc->ice_ufrag, old_desc->ice_pwd, "");
>+  CheckDesc(desc.get(), "", old_desc->ice_ufrag, old_desc->ice_pwd, "");
> }
> 
> // Test that we handle answering an offer with DTLS with no DTLS.
>@@ -261,8 +261,8 @@ TEST_F(TransportDescriptionFactoryTest, TestAnswerDtlsToDtls) {
>   // f2_ produces the answer that is being checked in this test, so the
>   // answer must contain fingerprint lines with cert2_'s digest algorithm.
>   std::string digest_alg2;
>-  ASSERT_TRUE(cert2_->ssl_certificate().GetSignatureDigestAlgorithm(
>-      &digest_alg2));
>+  ASSERT_TRUE(
>+      cert2_->ssl_certificate().GetSignatureDigestAlgorithm(&digest_alg2));
> 
>   std::unique_ptr<TransportDescription> offer(
>       f1_.CreateOffer(TransportOptions(), NULL));
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transportfactoryinterface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transportfactoryinterface.h
>new file mode 100644
>index 00000000000..ce32ee8483e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transportfactoryinterface.h
>@@ -0,0 +1,42 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef P2P_BASE_TRANSPORTFACTORYINTERFACE_H_
>+#define P2P_BASE_TRANSPORTFACTORYINTERFACE_H_
>+
>+#include <memory>
>+#include <string>
>+
>+#include "p2p/base/dtlstransportinternal.h"
>+#include "p2p/base/icetransportinternal.h"
>+
>+namespace cricket {
>+
>+// This interface is used to create DTLS/ICE transports. The external transports
>+// can be injected into the JsepTransportController through it. For example, the
>+// FakeIceTransport/FakeDtlsTransport can be injected by setting a
>+// FakeTransportFactory which implements this interface to the
>+// JsepTransportController.
>+class TransportFactoryInterface {
>+ public:
>+  virtual ~TransportFactoryInterface() {}
>+
>+  virtual std::unique_ptr<IceTransportInternal> CreateIceTransport(
>+      const std::string& transport_name,
>+      int component) = 0;
>+
>+  virtual std::unique_ptr<DtlsTransportInternal> CreateDtlsTransport(
>+      std::unique_ptr<IceTransportInternal> ice,
>+      const rtc::CryptoOptions& crypto_options) = 0;
>+};
>+
>+}  // namespace cricket
>+
>+#endif  // P2P_BASE_TRANSPORTFACTORYINTERFACE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transportinfo.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transportinfo.h
>index 3502423c125..627c0f781f6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transportinfo.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/transportinfo.h
>@@ -29,8 +29,7 @@ struct TransportInfo {
> 
>   TransportInfo(const std::string& content_name,
>                 const TransportDescription& description)
>-      : content_name(content_name),
>-        description(description) {}
>+      : content_name(content_name), description(description) {}
> 
>   std::string content_name;
>   TransportDescription description;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/turnport.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/turnport.cc
>index 285d41bf394..13c9a4691a7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/turnport.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/turnport.cc
>@@ -15,15 +15,14 @@
> #include <utility>
> #include <vector>
> 
>-#include "api/optional.h"
>-#include "p2p/base/common.h"
>+#include "absl/memory/memory.h"
>+#include "absl/types/optional.h"
> #include "p2p/base/stun.h"
> #include "rtc_base/asyncpacketsocket.h"
> #include "rtc_base/byteorder.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/nethelpers.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/socketaddress.h"
> #include "rtc_base/stringencode.h"
> 
>@@ -97,7 +96,8 @@ class TurnRefreshRequest : public StunRequest {
> class TurnCreatePermissionRequest : public StunRequest,
>                                     public sigslot::has_slots<> {
>  public:
>-  TurnCreatePermissionRequest(TurnPort* port, TurnEntry* entry,
>+  TurnCreatePermissionRequest(TurnPort* port,
>+                              TurnEntry* entry,
>                               const rtc::SocketAddress& ext_addr);
>   void Prepare(StunMessage* request) override;
>   void OnSent() override;
>@@ -113,10 +113,11 @@ class TurnCreatePermissionRequest : public StunRequest,
>   rtc::SocketAddress ext_addr_;
> };
> 
>-class TurnChannelBindRequest : public StunRequest,
>-                               public sigslot::has_slots<> {
>+class TurnChannelBindRequest : public StunRequest, public sigslot::has_slots<> {
>  public:
>-  TurnChannelBindRequest(TurnPort* port, TurnEntry* entry, int channel_id,
>+  TurnChannelBindRequest(TurnPort* port,
>+                         TurnEntry* entry,
>+                         int channel_id,
>                          const rtc::SocketAddress& ext_addr);
>   void Prepare(StunMessage* request) override;
>   void OnSent() override;
>@@ -138,8 +139,7 @@ class TurnChannelBindRequest : public StunRequest,
> class TurnEntry : public sigslot::has_slots<> {
>  public:
>   enum BindState { STATE_UNBOUND, STATE_BINDING, STATE_BOUND };
>-  TurnEntry(TurnPort* port, int channel_id,
>-            const rtc::SocketAddress& ext_addr);
>+  TurnEntry(TurnPort* port, int channel_id, const rtc::SocketAddress& ext_addr);
> 
>   TurnPort* port() { return port_; }
> 
>@@ -152,7 +152,7 @@ class TurnEntry : public sigslot::has_slots<> {
> 
>   // If the destruction timestamp is set, that means destruction has been
>   // scheduled (will occur TURN_PERMISSION_TIMEOUT after it's scheduled).
>-  rtc::Optional<int64_t> destruction_timestamp() {
>+  absl::optional<int64_t> destruction_timestamp() {
>     return destruction_timestamp_;
>   }
>   void set_destruction_timestamp(int64_t destruction_timestamp) {
>@@ -165,7 +165,9 @@ class TurnEntry : public sigslot::has_slots<> {
>   void SendChannelBindRequest(int delay);
>   // Sends a packet to the given destination address.
>   // This will wrap the packet in STUN if necessary.
>-  int Send(const void* data, size_t size, bool payload,
>+  int Send(const void* data,
>+           size_t size,
>+           bool payload,
>            const rtc::PacketOptions& options);
> 
>   void OnCreatePermissionSuccess();
>@@ -186,7 +188,7 @@ class TurnEntry : public sigslot::has_slots<> {
>   // is also used as an ID of the event scheduling. When the destruction event
>   // actually fires, the TurnEntry will be destroyed only if the timestamp here
>   // matches the one in the firing event.
>-  rtc::Optional<int64_t> destruction_timestamp_;
>+  absl::optional<int64_t> destruction_timestamp_;
> };
> 
> TurnPort::TurnPort(rtc::Thread* thread,
>@@ -200,13 +202,9 @@ TurnPort::TurnPort(rtc::Thread* thread,
>                    int server_priority,
>                    const std::string& origin,
>                    webrtc::TurnCustomizer* customizer)
>-    : Port(thread,
>-           RELAY_PORT_TYPE,
>-           factory,
>-           network,
>-           username,
>-           password),
>+    : Port(thread, RELAY_PORT_TYPE, factory, network, username, password),
>       server_address_(server_address),
>+      tls_cert_verifier_(nullptr),
>       credentials_(credentials),
>       socket_(socket),
>       resolver_(NULL),
>@@ -234,7 +232,8 @@ TurnPort::TurnPort(rtc::Thread* thread,
>                    const std::string& origin,
>                    const std::vector<std::string>& tls_alpn_protocols,
>                    const std::vector<std::string>& tls_elliptic_curves,
>-                   webrtc::TurnCustomizer* customizer)
>+                   webrtc::TurnCustomizer* customizer,
>+                   rtc::SSLCertificateVerifier* tls_cert_verifier)
>     : Port(thread,
>            RELAY_PORT_TYPE,
>            factory,
>@@ -246,6 +245,7 @@ TurnPort::TurnPort(rtc::Thread* thread,
>       server_address_(server_address),
>       tls_alpn_protocols_(tls_alpn_protocols),
>       tls_elliptic_curves_(tls_elliptic_curves),
>+      tls_cert_verifier_(tls_cert_verifier),
>       credentials_(credentials),
>       socket_(NULL),
>       resolver_(NULL),
>@@ -266,9 +266,7 @@ TurnPort::~TurnPort() {
>   // release the allocation by sending a refresh with
>   // lifetime 0.
>   if (ready()) {
>-    TurnRefreshRequest bye(this);
>-    bye.set_lifetime(0);
>-    SendRequest(&bye, 0);
>+    Release();
>   }
> 
>   while (!entries_.empty()) {
>@@ -307,10 +305,9 @@ std::vector<std::string> TurnPort::GetTlsEllipticCurves() const {
> }
> 
> void TurnPort::PrepareAddress() {
>-  if (credentials_.username.empty() ||
>-      credentials_.password.empty()) {
>+  if (credentials_.username.empty() || credentials_.password.empty()) {
>     RTC_LOG(LS_ERROR) << "Allocation can't be started without setting the"
>-                      << " TURN server credentials for the user.";
>+                         " TURN server credentials for the user.";
>     OnAllocateError();
>     return;
>   }
>@@ -325,8 +322,8 @@ void TurnPort::PrepareAddress() {
>   } else {
>     // If protocol family of server address doesn't match with local, return.
>     if (!IsCompatibleAddress(server_address_.address)) {
>-      RTC_LOG(LS_ERROR) << "IP address family does not match: "
>-                        << "server: " << server_address_.address.family()
>+      RTC_LOG(LS_ERROR) << "IP address family does not match. server: "
>+                        << server_address_.address.family()
>                         << " local: " << Network()->GetBestIP().family();
>       OnAllocateError();
>       return;
>@@ -335,9 +332,9 @@ void TurnPort::PrepareAddress() {
>     // Insert the current address to prevent redirection pingpong.
>     attempted_server_addresses_.insert(server_address_.address);
> 
>-    LOG_J(LS_INFO, this) << "Trying to connect to TURN server via "
>-                         << ProtoToString(server_address_.proto) << " @ "
>-                         << server_address_.address.ToSensitiveString();
>+    RTC_LOG(LS_INFO) << ToString() << ": Trying to connect to TURN server via "
>+                     << ProtoToString(server_address_.proto) << " @ "
>+                     << server_address_.address.ToSensitiveString();
>     if (!CreateTurnClientSocket()) {
>       RTC_LOG(LS_ERROR) << "Failed to create TURN client socket";
>       OnAllocateError();
>@@ -376,6 +373,7 @@ bool TurnPort::CreateTurnClientSocket() {
>     tcp_options.opts = opts;
>     tcp_options.tls_alpn_protocols = tls_alpn_protocols_;
>     tcp_options.tls_elliptic_curves = tls_elliptic_curves_;
>+    tcp_options.tls_cert_verifier = tls_cert_verifier_;
>     socket_ = socket_factory()->CreateClientTcpSocket(
>         rtc::SocketAddress(Network()->GetBestIP(), 0), server_address_.address,
>         proxy(), user_agent(), tcp_options);
>@@ -436,26 +434,28 @@ void TurnPort::OnSocketConnect(rtc::AsyncPacketSocket* socket) {
>   const rtc::SocketAddress& socket_address = socket->GetLocalAddress();
>   const std::vector<rtc::InterfaceAddress>& desired_addresses =
>       Network()->GetIPs();
>-  if (std::find(desired_addresses.begin(), desired_addresses.end(),
>-                socket_address.ipaddr()) == desired_addresses.end()) {
>+  if (std::find_if(desired_addresses.begin(), desired_addresses.end(),
>+                   [socket_address](const rtc::InterfaceAddress& addr) {
>+                     return socket_address.ipaddr() == addr;
>+                   }) == desired_addresses.end()) {
>     if (socket->GetLocalAddress().IsLoopbackIP()) {
>       RTC_LOG(LS_WARNING) << "Socket is bound to the address:"
>                           << socket_address.ipaddr().ToString()
>-                          << ", rather then an address associated with network:"
>+                          << ", rather than an address associated with network:"
>                           << Network()->ToString()
>                           << ". Still allowing it since it's localhost.";
>     } else if (IPIsAny(Network()->GetBestIP())) {
>       RTC_LOG(LS_WARNING)
>           << "Socket is bound to the address:"
>           << socket_address.ipaddr().ToString()
>-          << ", rather then an address associated with network:"
>+          << ", rather than an address associated with network:"
>           << Network()->ToString()
>           << ". Still allowing it since it's the 'any' address"
>-          << ", possibly caused by multiple_routes being disabled.";
>+             ", possibly caused by multiple_routes being disabled.";
>     } else {
>       RTC_LOG(LS_WARNING) << "Socket is bound to the address:"
>                           << socket_address.ipaddr().ToString()
>-                          << ", rather then an address associated with network:"
>+                          << ", rather than an address associated with network:"
>                           << Network()->ToString() << ". Discarding TURN port.";
>       OnAllocateError();
>       return;
>@@ -467,29 +467,32 @@ void TurnPort::OnSocketConnect(rtc::AsyncPacketSocket* socket) {
>     server_address_.address = socket_->GetRemoteAddress();
>   }
> 
>-  RTC_LOG(LS_INFO) << "TurnPort connected to " << socket->GetRemoteAddress()
>-                   << " using tcp.";
>+  RTC_LOG(LS_INFO) << "TurnPort connected to "
>+                   << socket->GetRemoteAddress().ToString() << " using tcp.";
>   SendRequest(new TurnAllocateRequest(this), 0);
> }
> 
> void TurnPort::OnSocketClose(rtc::AsyncPacketSocket* socket, int error) {
>-  LOG_J(LS_WARNING, this) << "Connection with server failed, error=" << error;
>+  RTC_LOG(LS_WARNING) << ToString()
>+                      << ": Connection with server failed with error: "
>+                      << error;
>   RTC_DCHECK(socket == socket_);
>   Close();
> }
> 
> void TurnPort::OnAllocateMismatch() {
>   if (allocate_mismatch_retries_ >= MAX_ALLOCATE_MISMATCH_RETRIES) {
>-    LOG_J(LS_WARNING, this) << "Giving up on the port after "
>-                            << allocate_mismatch_retries_
>-                            << " retries for STUN_ERROR_ALLOCATION_MISMATCH";
>+    RTC_LOG(LS_WARNING) << ToString() << ": Giving up on the port after "
>+                        << allocate_mismatch_retries_
>+                        << " retries for STUN_ERROR_ALLOCATION_MISMATCH";
>     OnAllocateError();
>     return;
>   }
> 
>-  LOG_J(LS_INFO, this) << "Allocating a new socket after "
>-                       << "STUN_ERROR_ALLOCATION_MISMATCH, retry = "
>-                       << allocate_mismatch_retries_ + 1;
>+  RTC_LOG(LS_INFO) << ToString()
>+                   << ": Allocating a new socket after "
>+                      "STUN_ERROR_ALLOCATION_MISMATCH, retry: "
>+                   << allocate_mismatch_retries_ + 1;
>   if (SharedSocket()) {
>     ResetSharedSocket();
>   } else {
>@@ -573,14 +576,16 @@ int TurnPort::GetError() {
>   return error_;
> }
> 
>-int TurnPort::SendTo(const void* data, size_t size,
>+int TurnPort::SendTo(const void* data,
>+                     size_t size,
>                      const rtc::SocketAddress& addr,
>                      const rtc::PacketOptions& options,
>                      bool payload) {
>   // Try to find an entry for this specific address; we should have one.
>   TurnEntry* entry = FindEntry(addr);
>   if (!entry) {
>-    RTC_LOG(LS_ERROR) << "Did not find the TurnEntry for address " << addr;
>+    RTC_LOG(LS_ERROR) << "Did not find the TurnEntry for address "
>+                      << addr.ToString();
>     return 0;
>   }
> 
>@@ -590,7 +595,9 @@ int TurnPort::SendTo(const void* data, size_t size,
>   }
> 
>   // Send the actual contents to the server using the usual mechanism.
>-  int sent = entry->Send(data, size, payload, options);
>+  rtc::PacketOptions modified_options(options);
>+  CopyPortInformationToPacketInfo(&modified_options.info_signaled_after_sent);
>+  int sent = entry->Send(data, size, payload, modified_options);
>   if (sent <= 0) {
>     return SOCKET_ERROR;
>   }
>@@ -606,7 +613,8 @@ bool TurnPort::CanHandleIncomingPacketsFrom(
> }
> 
> bool TurnPort::HandleIncomingPacket(rtc::AsyncPacketSocket* socket,
>-                                    const char* data, size_t size,
>+                                    const char* data,
>+                                    size_t size,
>                                     const rtc::SocketAddress& remote_addr,
>                                     const rtc::PacketTime& packet_time) {
>   if (socket != socket_) {
>@@ -619,22 +627,24 @@ bool TurnPort::HandleIncomingPacket(rtc::AsyncPacketSocket* socket,
>   // alternative server redirection. TODO(guoweis): add a unit test for this
>   // race condition.
>   if (remote_addr != server_address_.address) {
>-    LOG_J(LS_WARNING, this) << "Discarding TURN message from unknown address:"
>-                            << remote_addr.ToString()
>-                            << ", server_address_:"
>-                            << server_address_.address.ToString();
>+    RTC_LOG(LS_WARNING) << ToString()
>+                        << ": Discarding TURN message from unknown address: "
>+                        << remote_addr.ToString() << " server_address_: "
>+                        << server_address_.address.ToString();
>     return false;
>   }
> 
>   // The message must be at least the size of a channel header.
>   if (size < TURN_CHANNEL_HEADER_SIZE) {
>-    LOG_J(LS_WARNING, this) << "Received TURN message that was too short";
>+    RTC_LOG(LS_WARNING) << ToString()
>+                        << ": Received TURN message that was too short";
>     return false;
>   }
> 
>   if (state_ == STATE_DISCONNECTED) {
>-    LOG_J(LS_WARNING, this)
>-        << "Received TURN message while the TURN port is disconnected";
>+    RTC_LOG(LS_WARNING)
>+        << ToString()
>+        << ": Received TURN message while the TURN port is disconnected";
>     return false;
>   }
> 
>@@ -654,8 +664,9 @@ bool TurnPort::HandleIncomingPacket(rtc::AsyncPacketSocket* socket,
> 
>   if (SharedSocket() && (msg_type == STUN_BINDING_RESPONSE ||
>                          msg_type == STUN_BINDING_ERROR_RESPONSE)) {
>-    LOG_J(LS_VERBOSE, this) <<
>-        "Ignoring STUN binding response message on shared socket.";
>+    RTC_LOG(LS_VERBOSE)
>+        << ToString()
>+        << ": Ignoring STUN binding response message on shared socket.";
>     return false;
>   }
> 
>@@ -663,8 +674,10 @@ bool TurnPort::HandleIncomingPacket(rtc::AsyncPacketSocket* socket,
>   // Check success responses, but not errors, for MESSAGE-INTEGRITY.
>   if (IsStunSuccessResponseType(msg_type) &&
>       !StunMessage::ValidateMessageIntegrity(data, size, hash())) {
>-    LOG_J(LS_WARNING, this) << "Received TURN message with invalid "
>-                            << "message integrity, msg_type=" << msg_type;
>+    RTC_LOG(LS_WARNING) << ToString()
>+                        << ": Received TURN message with invalid "
>+                           "message integrity, msg_type: "
>+                        << msg_type;
>     return true;
>   }
>   request_manager_.CheckResponse(data, size);
>@@ -701,32 +714,31 @@ bool TurnPort::SetAlternateServer(const rtc::SocketAddress& address) {
>   // Check if we have seen this address before and reject if we did.
>   AttemptedServerSet::iterator iter = attempted_server_addresses_.find(address);
>   if (iter != attempted_server_addresses_.end()) {
>-    LOG_J(LS_WARNING, this) << "Redirection to ["
>-                            << address.ToSensitiveString()
>-                            << "] ignored, allocation failed.";
>+    RTC_LOG(LS_WARNING) << ToString() << ": Redirection to ["
>+                        << address.ToSensitiveString()
>+                        << "] ignored, allocation failed.";
>     return false;
>   }
> 
>   // If protocol family of server address doesn't match with local, return.
>   if (!IsCompatibleAddress(address)) {
>     RTC_LOG(LS_WARNING) << "Server IP address family does not match with "
>-                        << "local host address family type";
>+                           "local host address family type";
>     return false;
>   }
> 
>   // Block redirects to a loopback address.
>   // See: https://bugs.chromium.org/p/chromium/issues/detail?id=649118
>   if (address.IsLoopbackIP()) {
>-    LOG_J(LS_WARNING, this)
>-        << "Blocking attempted redirect to loopback address.";
>+    RTC_LOG(LS_WARNING) << ToString()
>+                        << ": Blocking attempted redirect to loopback address.";
>     return false;
>   }
> 
>-  LOG_J(LS_INFO, this) << "Redirecting from TURN server ["
>-                       << server_address_.address.ToSensitiveString()
>-                       << "] to TURN server ["
>-                       << address.ToSensitiveString()
>-                       << "]";
>+  RTC_LOG(LS_INFO) << ToString() << ": Redirecting from TURN server ["
>+                   << server_address_.address.ToSensitiveString()
>+                   << "] to TURN server [" << address.ToSensitiveString()
>+                   << "]";
>   server_address_ = ProtocolAddress(address, server_address_.proto);
> 
>   // Insert the current address to prevent redirection pingpong.
>@@ -738,8 +750,8 @@ void TurnPort::ResolveTurnAddress(const rtc::SocketAddress& address) {
>   if (resolver_)
>     return;
> 
>-  LOG_J(LS_INFO, this) << "Starting TURN host lookup for "
>-                       << address.ToSensitiveString();
>+  RTC_LOG(LS_INFO) << ToString() << ": Starting TURN host lookup for "
>+                   << address.ToSensitiveString();
>   resolver_ = socket_factory()->CreateAsyncResolver();
>   resolver_->SignalDone.connect(this, &TurnPort::OnResolveResult);
>   resolver_->Start(address);
>@@ -765,27 +777,29 @@ void TurnPort::OnResolveResult(rtc::AsyncResolverInterface* resolver) {
>   if (resolver_->GetError() != 0 ||
>       !resolver_->GetResolvedAddress(Network()->GetBestIP().family(),
>                                      &resolved_address)) {
>-    LOG_J(LS_WARNING, this) << "TURN host lookup received error "
>-                            << resolver_->GetError();
>+    RTC_LOG(LS_WARNING) << ToString() << ": TURN host lookup received error "
>+                        << resolver_->GetError();
>     error_ = resolver_->GetError();
>     OnAllocateError();
>     return;
>   }
>   // Signal needs both resolved and unresolved address. After signal is sent
>   // we can copy resolved address back into |server_address_|.
>-  SignalResolvedServerAddress(this, server_address_.address,
>-                              resolved_address);
>+  SignalResolvedServerAddress(this, server_address_.address, resolved_address);
>   server_address_.address = resolved_address;
>   PrepareAddress();
> }
> 
>-void TurnPort::OnSendStunPacket(const void* data, size_t size,
>+void TurnPort::OnSendStunPacket(const void* data,
>+                                size_t size,
>                                 StunRequest* request) {
>   RTC_DCHECK(connected());
>   rtc::PacketOptions options(DefaultDscpValue());
>+  options.info_signaled_after_sent.packet_type = rtc::PacketType::kTurnMessage;
>+  CopyPortInformationToPacketInfo(&options.info_signaled_after_sent);
>   if (Send(data, size, options) < 0) {
>-    LOG_J(LS_ERROR, this) << "Failed to send TURN message, err="
>-                          << socket_->GetError();
>+    RTC_LOG(LS_ERROR) << ToString() << ": Failed to send TURN message, error: "
>+                      << socket_->GetError();
>   }
> }
> 
>@@ -839,6 +853,18 @@ void TurnPort::HandleRefreshError() {
>   }
> }
> 
>+void TurnPort::Release() {
>+  // Remove any pending refresh requests.
>+  request_manager_.Clear();
>+
>+  // Send refresh with lifetime 0.
>+  TurnRefreshRequest* req = new TurnRefreshRequest(this);
>+  req->set_lifetime(0);
>+  SendRequest(req, 0);
>+
>+  state_ = STATE_RECEIVEONLY;
>+}
>+
> void TurnPort::Close() {
>   if (!ready()) {
>     OnAllocateError();
>@@ -850,6 +876,8 @@ void TurnPort::Close() {
>   for (auto kv : connections()) {
>     kv.second->Destroy();
>   }
>+
>+  SignalTurnPortClosed(this);
> }
> 
> void TurnPort::OnMessage(rtc::Message* message) {
>@@ -880,6 +908,9 @@ void TurnPort::OnMessage(rtc::Message* message) {
>         PrepareAddress();
>       }
>       break;
>+    case MSG_ALLOCATION_RELEASED:
>+      Close();
>+      break;
>     default:
>       Port::OnMessage(message);
>   }
>@@ -889,13 +920,15 @@ void TurnPort::OnAllocateRequestTimeout() {
>   OnAllocateError();
> }
> 
>-void TurnPort::HandleDataIndication(const char* data, size_t size,
>+void TurnPort::HandleDataIndication(const char* data,
>+                                    size_t size,
>                                     const rtc::PacketTime& packet_time) {
>   // Read in the message, and process according to RFC5766, Section 10.4.
>   rtc::ByteBufferReader buf(data, size);
>   TurnMessage msg;
>   if (!msg.Read(&buf)) {
>-    LOG_J(LS_WARNING, this) << "Received invalid TURN data indication";
>+    RTC_LOG(LS_WARNING) << ToString()
>+                        << ": Received invalid TURN data indication";
>     return;
>   }
> 
>@@ -903,16 +936,17 @@ void TurnPort::HandleDataIndication(const char* data, size_t size,
>   const StunAddressAttribute* addr_attr =
>       msg.GetAddress(STUN_ATTR_XOR_PEER_ADDRESS);
>   if (!addr_attr) {
>-    LOG_J(LS_WARNING, this) << "Missing STUN_ATTR_XOR_PEER_ADDRESS attribute "
>-                            << "in data indication.";
>+    RTC_LOG(LS_WARNING) << ToString()
>+                        << ": Missing STUN_ATTR_XOR_PEER_ADDRESS attribute "
>+                           "in data indication.";
>     return;
>   }
> 
>-  const StunByteStringAttribute* data_attr =
>-      msg.GetByteString(STUN_ATTR_DATA);
>+  const StunByteStringAttribute* data_attr = msg.GetByteString(STUN_ATTR_DATA);
>   if (!data_attr) {
>-    LOG_J(LS_WARNING, this) << "Missing STUN_ATTR_DATA attribute in "
>-                            << "data indication.";
>+    RTC_LOG(LS_WARNING) << ToString()
>+                        << ": Missing STUN_ATTR_DATA attribute in "
>+                           "data indication.";
>     return;
>   }
> 
>@@ -920,16 +954,18 @@ void TurnPort::HandleDataIndication(const char* data, size_t size,
>   // a permission for.
>   rtc::SocketAddress ext_addr(addr_attr->GetAddress());
>   if (!HasPermission(ext_addr.ipaddr())) {
>-    LOG_J(LS_WARNING, this)
>-        << "Received TURN data indication with unknown "
>-        << "peer address, addr=" << ext_addr.ToSensitiveString();
>+    RTC_LOG(LS_WARNING) << ToString()
>+                        << ": Received TURN data indication with unknown "
>+                           "peer address, addr: "
>+                        << ext_addr.ToSensitiveString();
>   }
> 
>-  DispatchPacket(data_attr->bytes(), data_attr->length(), ext_addr,
>-                 PROTO_UDP, packet_time);
>+  DispatchPacket(data_attr->bytes(), data_attr->length(), ext_addr, PROTO_UDP,
>+                 packet_time);
> }
> 
>-void TurnPort::HandleChannelData(int channel_id, const char* data,
>+void TurnPort::HandleChannelData(int channel_id,
>+                                 const char* data,
>                                  size_t size,
>                                  const rtc::PacketTime& packet_time) {
>   // Read the message, and process according to RFC5766, Section 11.6.
>@@ -949,16 +985,20 @@ void TurnPort::HandleChannelData(int channel_id, const char* data,
>   // Extract header fields from the message.
>   uint16_t len = rtc::GetBE16(data + 2);
>   if (len > size - TURN_CHANNEL_HEADER_SIZE) {
>-    LOG_J(LS_WARNING, this) << "Received TURN channel data message with "
>-                            << "incorrect length, len=" << len;
>+    RTC_LOG(LS_WARNING) << ToString()
>+                        << ": Received TURN channel data message with "
>+                           "incorrect length, len: "
>+                        << len;
>     return;
>   }
>   // Allowing messages larger than |len|, as ChannelData can be padded.
> 
>   TurnEntry* entry = FindEntry(channel_id);
>   if (!entry) {
>-    LOG_J(LS_WARNING, this) << "Received TURN channel data message for invalid "
>-                            << "channel, channel_id=" << channel_id;
>+    RTC_LOG(LS_WARNING) << ToString()
>+                        << ": Received TURN channel data message for invalid "
>+                           "channel, channel_id: "
>+                        << channel_id;
>     return;
>   }
> 
>@@ -966,9 +1006,11 @@ void TurnPort::HandleChannelData(int channel_id, const char* data,
>                  PROTO_UDP, packet_time);
> }
> 
>-void TurnPort::DispatchPacket(const char* data, size_t size,
>-    const rtc::SocketAddress& remote_addr,
>-    ProtocolType proto, const rtc::PacketTime& packet_time) {
>+void TurnPort::DispatchPacket(const char* data,
>+                              size_t size,
>+                              const rtc::SocketAddress& remote_addr,
>+                              ProtocolType proto,
>+                              const rtc::PacketTime& packet_time) {
>   if (Connection* conn = GetConnection(remote_addr)) {
>     conn->OnReadPacket(data, size, packet_time);
>   } else {
>@@ -976,17 +1018,36 @@ void TurnPort::DispatchPacket(const char* data, size_t size,
>   }
> }
> 
>-bool TurnPort::ScheduleRefresh(int lifetime) {
>-  // Lifetime is in seconds; we schedule a refresh for one minute less.
>+bool TurnPort::ScheduleRefresh(uint32_t lifetime) {
>+  // Lifetime is in seconds, delay is in milliseconds.
>+  int delay = 1 * 60 * 1000;
>+
>+  // Cutoff lifetime bigger than 1h.
>+  constexpr uint32_t max_lifetime = 60 * 60;
>+
>   if (lifetime < 2 * 60) {
>-    LOG_J(LS_WARNING, this) << "Received response with lifetime that was "
>-                            << "too short, lifetime=" << lifetime;
>-    return false;
>+    // The RFC does not mention a lower limit on lifetime.
>+    // So if server sends a value less than 2 minutes, we schedule a refresh
>+    // for half lifetime.
>+    RTC_LOG(LS_WARNING) << ToString()
>+                        << ": Received response with short lifetime: "
>+                        << lifetime << " seconds.";
>+    delay = (lifetime * 1000) / 2;
>+  } else if (lifetime > max_lifetime) {
>+    // Make 1 hour largest delay, and then sce
>+    // we schedule a refresh for one minute less than max lifetime.
>+    RTC_LOG(LS_WARNING) << ToString()
>+                        << ": Received response with long lifetime: "
>+                        << lifetime << " seconds.";
>+    delay = (max_lifetime - 60) * 1000;
>+  } else {
>+    // Normal case,
>+    // we schedule a refresh for one minute less than requested lifetime.
>+    delay = (lifetime - 60) * 1000;
>   }
> 
>-  int delay = (lifetime - 60) * 1000;
>   SendRequest(new TurnRefreshRequest(this), delay);
>-  LOG_J(LS_INFO, this) << "Scheduled refresh in " << delay << "ms.";
>+  RTC_LOG(LS_INFO) << ToString() << ": Scheduled refresh in " << delay << "ms.";
>   return true;
> }
> 
>@@ -997,17 +1058,18 @@ void TurnPort::SendRequest(StunRequest* req, int delay) {
> void TurnPort::AddRequestAuthInfo(StunMessage* msg) {
>   // If we've gotten the necessary data from the server, add it to our request.
>   RTC_DCHECK(!hash_.empty());
>-  msg->AddAttribute(rtc::MakeUnique<StunByteStringAttribute>(
>+  msg->AddAttribute(absl::make_unique<StunByteStringAttribute>(
>       STUN_ATTR_USERNAME, credentials_.username));
>   msg->AddAttribute(
>-      rtc::MakeUnique<StunByteStringAttribute>(STUN_ATTR_REALM, realm_));
>+      absl::make_unique<StunByteStringAttribute>(STUN_ATTR_REALM, realm_));
>   msg->AddAttribute(
>-      rtc::MakeUnique<StunByteStringAttribute>(STUN_ATTR_NONCE, nonce_));
>+      absl::make_unique<StunByteStringAttribute>(STUN_ATTR_NONCE, nonce_));
>   const bool success = msg->AddMessageIntegrity(hash());
>   RTC_DCHECK(success);
> }
> 
>-int TurnPort::Send(const void* data, size_t len,
>+int TurnPort::Send(const void* data,
>+                   size_t len,
>                    const rtc::PacketOptions& options) {
>   return socket_->SendTo(data, len, server_address_.address, options);
> }
>@@ -1026,7 +1088,7 @@ bool TurnPort::UpdateNonce(StunMessage* response) {
>       response->GetByteString(STUN_ATTR_REALM);
>   if (!realm_attr) {
>     RTC_LOG(LS_ERROR) << "Missing STUN_ATTR_REALM attribute in "
>-                      << "stale nonce error response.";
>+                         "stale nonce error response.";
>     return false;
>   }
>   set_realm(realm_attr->GetString());
>@@ -1035,7 +1097,7 @@ bool TurnPort::UpdateNonce(StunMessage* response) {
>       response->GetByteString(STUN_ATTR_NONCE);
>   if (!nonce_attr) {
>     RTC_LOG(LS_ERROR) << "Missing STUN_ATTR_NONCE attribute in "
>-                      << "stale nonce error response.";
>+                         "stale nonce error response.";
>     return false;
>   }
>   set_nonce(nonce_attr->GetString());
>@@ -1048,29 +1110,25 @@ void TurnPort::ResetNonce() {
>   realm_.clear();
> }
> 
>-static bool MatchesIP(TurnEntry* e, rtc::IPAddress ipaddr) {
>-  return e->address().ipaddr() == ipaddr;
>-}
> bool TurnPort::HasPermission(const rtc::IPAddress& ipaddr) const {
>   return (std::find_if(entries_.begin(), entries_.end(),
>-      std::bind2nd(std::ptr_fun(MatchesIP), ipaddr)) != entries_.end());
>+                       [&ipaddr](const TurnEntry* e) {
>+                         return e->address().ipaddr() == ipaddr;
>+                       }) != entries_.end());
> }
> 
>-static bool MatchesAddress(TurnEntry* e, rtc::SocketAddress addr) {
>-  return e->address() == addr;
>-}
> TurnEntry* TurnPort::FindEntry(const rtc::SocketAddress& addr) const {
>-  EntryList::const_iterator it = std::find_if(entries_.begin(), entries_.end(),
>-      std::bind2nd(std::ptr_fun(MatchesAddress), addr));
>+  auto it = std::find_if(
>+      entries_.begin(), entries_.end(),
>+      [&addr](const TurnEntry* e) { return e->address() == addr; });
>   return (it != entries_.end()) ? *it : NULL;
> }
> 
>-static bool MatchesChannelId(TurnEntry* e, int id) {
>-  return e->channel_id() == id;
>-}
> TurnEntry* TurnPort::FindEntry(int channel_id) const {
>-  EntryList::const_iterator it = std::find_if(entries_.begin(), entries_.end(),
>-      std::bind2nd(std::ptr_fun(MatchesChannelId), channel_id));
>+  auto it = std::find_if(entries_.begin(), entries_.end(),
>+                         [&channel_id](const TurnEntry* e) {
>+                           return e->channel_id() == channel_id;
>+                         });
>   return (it != entries_.end()) ? *it : NULL;
> }
> 
>@@ -1193,8 +1251,9 @@ void TurnPort::TurnCustomizerMaybeModifyOutgoingStunMessage(
>   turn_customizer_->MaybeModifyOutgoingStunMessage(this, message);
> }
> 
>-bool TurnPort::TurnCustomizerAllowChannelData(
>-    const void* data, size_t size, bool payload) {
>+bool TurnPort::TurnCustomizerAllowChannelData(const void* data,
>+                                              size_t size,
>+                                              bool payload) {
>   if (turn_customizer_ == nullptr) {
>     return true;
>   }
>@@ -1203,9 +1262,7 @@ bool TurnPort::TurnCustomizerAllowChannelData(
> }
> 
> TurnAllocateRequest::TurnAllocateRequest(TurnPort* port)
>-    : StunRequest(new TurnMessage()),
>-      port_(port) {
>-}
>+    : StunRequest(new TurnMessage()), port_(port) {}
> 
> void TurnAllocateRequest::Prepare(StunMessage* request) {
>   // Create the request as indicated in RFC 5766, Section 6.1.
>@@ -1221,23 +1278,26 @@ void TurnAllocateRequest::Prepare(StunMessage* request) {
> }
> 
> void TurnAllocateRequest::OnSent() {
>-  LOG_J(LS_INFO, port_) << "TURN allocate request sent"
>-                        << ", id=" << rtc::hex_encode(id());
>+  RTC_LOG(LS_INFO) << port_->ToString() << ": TURN allocate request sent, id="
>+                   << rtc::hex_encode(id());
>   StunRequest::OnSent();
> }
> 
> void TurnAllocateRequest::OnResponse(StunMessage* response) {
>-  LOG_J(LS_INFO, port_) << "TURN allocate requested successfully"
>-                        << ", id=" << rtc::hex_encode(id())
>-                        << ", code=0"  // Makes logging easier to parse.
>-                        << ", rtt=" << Elapsed();
>+  RTC_LOG(LS_INFO) << port_->ToString()
>+                   << ": TURN allocate requested successfully, id="
>+                   << rtc::hex_encode(id())
>+                   << ", code=0"  // Makes logging easier to parse.
>+                      ", rtt="
>+                   << Elapsed();
> 
>   // Check mandatory attributes as indicated in RFC5766, Section 6.3.
>   const StunAddressAttribute* mapped_attr =
>       response->GetAddress(STUN_ATTR_XOR_MAPPED_ADDRESS);
>   if (!mapped_attr) {
>-    LOG_J(LS_WARNING, port_) << "Missing STUN_ATTR_XOR_MAPPED_ADDRESS "
>-                             << "attribute in allocate success response";
>+    RTC_LOG(LS_WARNING) << port_->ToString()
>+                        << ": Missing STUN_ATTR_XOR_MAPPED_ADDRESS "
>+                           "attribute in allocate success response";
>     return;
>   }
>   // Using XOR-Mapped-Address for stun.
>@@ -1246,16 +1306,18 @@ void TurnAllocateRequest::OnResponse(StunMessage* response) {
>   const StunAddressAttribute* relayed_attr =
>       response->GetAddress(STUN_ATTR_XOR_RELAYED_ADDRESS);
>   if (!relayed_attr) {
>-    LOG_J(LS_WARNING, port_) << "Missing STUN_ATTR_XOR_RELAYED_ADDRESS "
>-                             << "attribute in allocate success response";
>+    RTC_LOG(LS_WARNING) << port_->ToString()
>+                        << ": Missing STUN_ATTR_XOR_RELAYED_ADDRESS "
>+                           "attribute in allocate success response";
>     return;
>   }
> 
>   const StunUInt32Attribute* lifetime_attr =
>       response->GetUInt32(STUN_ATTR_TURN_LIFETIME);
>   if (!lifetime_attr) {
>-    LOG_J(LS_WARNING, port_) << "Missing STUN_ATTR_TURN_LIFETIME attribute in "
>-                             << "allocate success response";
>+    RTC_LOG(LS_WARNING) << port_->ToString()
>+                        << ": Missing STUN_ATTR_TURN_LIFETIME attribute in "
>+                           "allocate success response";
>     return;
>   }
>   // Notify the port the allocate succeeded, and schedule a refresh request.
>@@ -1268,12 +1330,13 @@ void TurnAllocateRequest::OnErrorResponse(StunMessage* response) {
>   // Process error response according to RFC5766, Section 6.4.
>   int error_code = response->GetErrorCodeValue();
> 
>-  LOG_J(LS_INFO, port_) << "Received TURN allocate error response"
>-                        << ", id=" << rtc::hex_encode(id())
>-                        << ", code=" << error_code << ", rtt=" << Elapsed();
>+  RTC_LOG(LS_INFO) << port_->ToString()
>+                   << ": Received TURN allocate error response, id="
>+                   << rtc::hex_encode(id()) << ", code=" << error_code
>+                   << ", rtt=" << Elapsed();
> 
>   switch (error_code) {
>-    case STUN_ERROR_UNAUTHORIZED:       // Unauthrorized.
>+    case STUN_ERROR_UNAUTHORIZED:  // Unauthrorized.
>       OnAuthChallenge(response, error_code);
>       break;
>     case STUN_ERROR_TRY_ALTERNATE:
>@@ -1286,25 +1349,26 @@ void TurnAllocateRequest::OnErrorResponse(StunMessage* response) {
>                             TurnPort::MSG_ALLOCATE_MISMATCH);
>       break;
>     default:
>-      LOG_J(LS_WARNING, port_)
>-          << "Received TURN allocate error response"
>-          << ", id=" << rtc::hex_encode(id()) << ", code=" << error_code
>-          << ", rtt=" << Elapsed();
>+      RTC_LOG(LS_WARNING) << port_->ToString()
>+                          << ": Received TURN allocate error response, id="
>+                          << rtc::hex_encode(id()) << ", code=" << error_code
>+                          << ", rtt=" << Elapsed();
>       port_->OnAllocateError();
>   }
> }
> 
> void TurnAllocateRequest::OnTimeout() {
>-  LOG_J(LS_WARNING, port_) << "TURN allocate request "
>-                           << rtc::hex_encode(id()) << " timeout";
>+  RTC_LOG(LS_WARNING) << port_->ToString() << ": TURN allocate request "
>+                      << rtc::hex_encode(id()) << " timeout";
>   port_->OnAllocateRequestTimeout();
> }
> 
> void TurnAllocateRequest::OnAuthChallenge(StunMessage* response, int code) {
>   // If we failed to authenticate even after we sent our credentials, fail hard.
>   if (code == STUN_ERROR_UNAUTHORIZED && !port_->hash().empty()) {
>-    LOG_J(LS_WARNING, port_) << "Failed to authenticate with the server "
>-                             << "after challenge.";
>+    RTC_LOG(LS_WARNING) << port_->ToString()
>+                        << ": Failed to authenticate with the server "
>+                           "after challenge.";
>     port_->OnAllocateError();
>     return;
>   }
>@@ -1313,8 +1377,9 @@ void TurnAllocateRequest::OnAuthChallenge(StunMessage* response, int code) {
>   const StunByteStringAttribute* realm_attr =
>       response->GetByteString(STUN_ATTR_REALM);
>   if (!realm_attr) {
>-    LOG_J(LS_WARNING, port_) << "Missing STUN_ATTR_REALM attribute in "
>-                             << "allocate unauthorized response.";
>+    RTC_LOG(LS_WARNING) << port_->ToString()
>+                        << ": Missing STUN_ATTR_REALM attribute in "
>+                           "allocate unauthorized response.";
>     return;
>   }
>   port_->set_realm(realm_attr->GetString());
>@@ -1322,8 +1387,9 @@ void TurnAllocateRequest::OnAuthChallenge(StunMessage* response, int code) {
>   const StunByteStringAttribute* nonce_attr =
>       response->GetByteString(STUN_ATTR_NONCE);
>   if (!nonce_attr) {
>-    LOG_J(LS_WARNING, port_) << "Missing STUN_ATTR_NONCE attribute in "
>-                             << "allocate unauthorized response.";
>+    RTC_LOG(LS_WARNING) << port_->ToString()
>+                        << ": Missing STUN_ATTR_NONCE attribute in "
>+                           "allocate unauthorized response.";
>     return;
>   }
>   port_->set_nonce(nonce_attr->GetString());
>@@ -1341,8 +1407,9 @@ void TurnAllocateRequest::OnTryAlternate(StunMessage* response, int code) {
>   const StunAddressAttribute* alternate_server_attr =
>       response->GetAddress(STUN_ATTR_ALTERNATE_SERVER);
>   if (!alternate_server_attr) {
>-    LOG_J(LS_WARNING, port_) << "Missing STUN_ATTR_ALTERNATE_SERVER "
>-                             << "attribute in try alternate error response";
>+    RTC_LOG(LS_WARNING) << port_->ToString()
>+                        << ": Missing STUN_ATTR_ALTERNATE_SERVER "
>+                           "attribute in try alternate error response";
>     port_->OnAllocateError();
>     return;
>   }
>@@ -1355,16 +1422,18 @@ void TurnAllocateRequest::OnTryAlternate(StunMessage* response, int code) {
>   const StunByteStringAttribute* realm_attr =
>       response->GetByteString(STUN_ATTR_REALM);
>   if (realm_attr) {
>-    LOG_J(LS_INFO, port_) << "Applying STUN_ATTR_REALM attribute in "
>-                          << "try alternate error response.";
>+    RTC_LOG(LS_INFO) << port_->ToString()
>+                     << ": Applying STUN_ATTR_REALM attribute in "
>+                        "try alternate error response.";
>     port_->set_realm(realm_attr->GetString());
>   }
> 
>   const StunByteStringAttribute* nonce_attr =
>       response->GetByteString(STUN_ATTR_NONCE);
>   if (nonce_attr) {
>-    LOG_J(LS_INFO, port_) << "Applying STUN_ATTR_NONCE attribute in "
>-                          << "try alternate error response.";
>+    RTC_LOG(LS_INFO) << port_->ToString()
>+                     << ": Applying STUN_ATTR_NONCE attribute in "
>+                        "try alternate error response.";
>     port_->set_nonce(nonce_attr->GetString());
>   }
> 
>@@ -1376,10 +1445,7 @@ void TurnAllocateRequest::OnTryAlternate(StunMessage* response, int code) {
> }
> 
> TurnRefreshRequest::TurnRefreshRequest(TurnPort* port)
>-    : StunRequest(new TurnMessage()),
>-      port_(port),
>-      lifetime_(-1) {
>-}
>+    : StunRequest(new TurnMessage()), port_(port), lifetime_(-1) {}
> 
> void TurnRefreshRequest::Prepare(StunMessage* request) {
>   // Create the request as indicated in RFC 5766, Section 7.1.
>@@ -1387,7 +1453,7 @@ void TurnRefreshRequest::Prepare(StunMessage* request) {
>   request->SetType(TURN_REFRESH_REQUEST);
>   if (lifetime_ > -1) {
>     request->AddAttribute(
>-        rtc::MakeUnique<StunUInt32Attribute>(STUN_ATTR_LIFETIME, lifetime_));
>+        absl::make_unique<StunUInt32Attribute>(STUN_ATTR_LIFETIME, lifetime_));
>   }
> 
>   port_->AddRequestAuthInfo(request);
>@@ -1395,28 +1461,39 @@ void TurnRefreshRequest::Prepare(StunMessage* request) {
> }
> 
> void TurnRefreshRequest::OnSent() {
>-  LOG_J(LS_INFO, port_) << "TURN refresh request sent"
>-                        << ", id=" << rtc::hex_encode(id());
>+  RTC_LOG(LS_INFO) << port_->ToString() << ": TURN refresh request sent, id="
>+                   << rtc::hex_encode(id());
>   StunRequest::OnSent();
> }
> 
> void TurnRefreshRequest::OnResponse(StunMessage* response) {
>-  LOG_J(LS_INFO, port_) << "TURN refresh requested successfully"
>-                        << ", id=" << rtc::hex_encode(id())
>-                        << ", code=0"  // Makes logging easier to parse.
>-                        << ", rtt=" << Elapsed();
>+  RTC_LOG(LS_INFO) << port_->ToString()
>+                   << ": TURN refresh requested successfully, id="
>+                   << rtc::hex_encode(id())
>+                   << ", code=0"  // Makes logging easier to parse.
>+                      ", rtt="
>+                   << Elapsed();
> 
>   // Check mandatory attributes as indicated in RFC5766, Section 7.3.
>   const StunUInt32Attribute* lifetime_attr =
>       response->GetUInt32(STUN_ATTR_TURN_LIFETIME);
>   if (!lifetime_attr) {
>-    LOG_J(LS_WARNING, port_) << "Missing STUN_ATTR_TURN_LIFETIME attribute in "
>-                             << "refresh success response.";
>+    RTC_LOG(LS_WARNING) << port_->ToString()
>+                        << ": Missing STUN_ATTR_TURN_LIFETIME attribute in "
>+                           "refresh success response.";
>     return;
>   }
> 
>-  // Schedule a refresh based on the returned lifetime value.
>-  port_->ScheduleRefresh(lifetime_attr->value());
>+  if (lifetime_attr->value() > 0) {
>+    // Schedule a refresh based on the returned lifetime value.
>+    port_->ScheduleRefresh(lifetime_attr->value());
>+  } else {
>+    // If we scheduled a refresh with lifetime 0, we're releasing this
>+    // allocation; see TurnPort::Release.
>+    port_->thread()->Post(RTC_FROM_HERE, port_,
>+                          TurnPort::MSG_ALLOCATION_RELEASED);
>+  }
>+
>   port_->SignalTurnRefreshResult(port_, TURN_SUCCESS_RESULT_CODE);
> }
> 
>@@ -1429,22 +1506,24 @@ void TurnRefreshRequest::OnErrorResponse(StunMessage* response) {
>       port_->SendRequest(new TurnRefreshRequest(port_), 0);
>     }
>   } else {
>-    LOG_J(LS_WARNING, port_)
>-        << "Received TURN refresh error response"
>-        << ", id=" << rtc::hex_encode(id()) << ", code=" << error_code
>-        << ", rtt=" << Elapsed();
>+    RTC_LOG(LS_WARNING) << port_->ToString()
>+                        << ": Received TURN refresh error response, id="
>+                        << rtc::hex_encode(id()) << ", code=" << error_code
>+                        << ", rtt=" << Elapsed();
>     port_->OnRefreshError();
>     port_->SignalTurnRefreshResult(port_, error_code);
>   }
> }
> 
> void TurnRefreshRequest::OnTimeout() {
>-  LOG_J(LS_WARNING, port_) << "TURN refresh timeout " << rtc::hex_encode(id());
>+  RTC_LOG(LS_WARNING) << port_->ToString() << ": TURN refresh timeout "
>+                      << rtc::hex_encode(id());
>   port_->OnRefreshError();
> }
> 
> TurnCreatePermissionRequest::TurnCreatePermissionRequest(
>-    TurnPort* port, TurnEntry* entry,
>+    TurnPort* port,
>+    TurnEntry* entry,
>     const rtc::SocketAddress& ext_addr)
>     : StunRequest(new TurnMessage()),
>       port_(port),
>@@ -1457,23 +1536,26 @@ TurnCreatePermissionRequest::TurnCreatePermissionRequest(
> void TurnCreatePermissionRequest::Prepare(StunMessage* request) {
>   // Create the request as indicated in RFC5766, Section 9.1.
>   request->SetType(TURN_CREATE_PERMISSION_REQUEST);
>-  request->AddAttribute(rtc::MakeUnique<StunXorAddressAttribute>(
>+  request->AddAttribute(absl::make_unique<StunXorAddressAttribute>(
>       STUN_ATTR_XOR_PEER_ADDRESS, ext_addr_));
>   port_->AddRequestAuthInfo(request);
>   port_->TurnCustomizerMaybeModifyOutgoingStunMessage(request);
> }
> 
> void TurnCreatePermissionRequest::OnSent() {
>-  LOG_J(LS_INFO, port_) << "TURN create permission request sent"
>-                        << ", id=" << rtc::hex_encode(id());
>+  RTC_LOG(LS_INFO) << port_->ToString()
>+                   << ": TURN create permission request sent, id="
>+                   << rtc::hex_encode(id());
>   StunRequest::OnSent();
> }
> 
> void TurnCreatePermissionRequest::OnResponse(StunMessage* response) {
>-  LOG_J(LS_INFO, port_) << "TURN permission requested successfully"
>-                        << ", id=" << rtc::hex_encode(id())
>-                        << ", code=0"  // Makes logging easier to parse.
>-                        << ", rtt=" << Elapsed();
>+  RTC_LOG(LS_INFO) << port_->ToString()
>+                   << ": TURN permission requested successfully, id="
>+                   << rtc::hex_encode(id())
>+                   << ", code=0"  // Makes logging easier to parse.
>+                      ", rtt="
>+                   << Elapsed();
> 
>   if (entry_) {
>     entry_->OnCreatePermissionSuccess();
>@@ -1482,17 +1564,19 @@ void TurnCreatePermissionRequest::OnResponse(StunMessage* response) {
> 
> void TurnCreatePermissionRequest::OnErrorResponse(StunMessage* response) {
>   int error_code = response->GetErrorCodeValue();
>-  LOG_J(LS_WARNING, port_) << "Received TURN create permission error response"
>-                           << ", id=" << rtc::hex_encode(id())
>-                           << ", code=" << error_code << ", rtt=" << Elapsed();
>+  RTC_LOG(LS_WARNING) << port_->ToString()
>+                      << ": Received TURN create permission error response, id="
>+                      << rtc::hex_encode(id()) << ", code=" << error_code
>+                      << ", rtt=" << Elapsed();
>   if (entry_) {
>     entry_->OnCreatePermissionError(response, error_code);
>   }
> }
> 
> void TurnCreatePermissionRequest::OnTimeout() {
>-  LOG_J(LS_WARNING, port_) << "TURN create permission timeout "
>-                           << rtc::hex_encode(id());
>+  RTC_LOG(LS_WARNING) << port_->ToString()
>+                      << ": TURN create permission timeout "
>+                      << rtc::hex_encode(id());
>   if (entry_) {
>     entry_->OnCreatePermissionTimeout();
>   }
>@@ -1504,39 +1588,44 @@ void TurnCreatePermissionRequest::OnEntryDestroyed(TurnEntry* entry) {
> }
> 
> TurnChannelBindRequest::TurnChannelBindRequest(
>-    TurnPort* port, TurnEntry* entry,
>-    int channel_id, const rtc::SocketAddress& ext_addr)
>+    TurnPort* port,
>+    TurnEntry* entry,
>+    int channel_id,
>+    const rtc::SocketAddress& ext_addr)
>     : StunRequest(new TurnMessage()),
>       port_(port),
>       entry_(entry),
>       channel_id_(channel_id),
>       ext_addr_(ext_addr) {
>-  entry_->SignalDestroyed.connect(
>-      this, &TurnChannelBindRequest::OnEntryDestroyed);
>+  entry_->SignalDestroyed.connect(this,
>+                                  &TurnChannelBindRequest::OnEntryDestroyed);
> }
> 
> void TurnChannelBindRequest::Prepare(StunMessage* request) {
>   // Create the request as indicated in RFC5766, Section 11.1.
>   request->SetType(TURN_CHANNEL_BIND_REQUEST);
>-  request->AddAttribute(rtc::MakeUnique<StunUInt32Attribute>(
>+  request->AddAttribute(absl::make_unique<StunUInt32Attribute>(
>       STUN_ATTR_CHANNEL_NUMBER, channel_id_ << 16));
>-  request->AddAttribute(rtc::MakeUnique<StunXorAddressAttribute>(
>+  request->AddAttribute(absl::make_unique<StunXorAddressAttribute>(
>       STUN_ATTR_XOR_PEER_ADDRESS, ext_addr_));
>   port_->AddRequestAuthInfo(request);
>   port_->TurnCustomizerMaybeModifyOutgoingStunMessage(request);
> }
> 
> void TurnChannelBindRequest::OnSent() {
>-  LOG_J(LS_INFO, port_) << "TURN channel bind request sent"
>-                        << ", id=" << rtc::hex_encode(id());
>+  RTC_LOG(LS_INFO) << port_->ToString()
>+                   << ": TURN channel bind request sent, id="
>+                   << rtc::hex_encode(id());
>   StunRequest::OnSent();
> }
> 
> void TurnChannelBindRequest::OnResponse(StunMessage* response) {
>-  LOG_J(LS_INFO, port_) << "TURN channel bind requested successfully"
>-                        << ", id=" << rtc::hex_encode(id())
>-                        << ", code=0"  // Makes logging easier to parse.
>-                        << ", rtt=" << Elapsed();
>+  RTC_LOG(LS_INFO) << port_->ToString()
>+                   << ": TURN channel bind requested successfully, id="
>+                   << rtc::hex_encode(id())
>+                   << ", code=0"  // Makes logging easier to parse.
>+                      ", rtt="
>+                   << Elapsed();
> 
>   if (entry_) {
>     entry_->OnChannelBindSuccess();
>@@ -1546,23 +1635,25 @@ void TurnChannelBindRequest::OnResponse(StunMessage* response) {
>     // permission from expiring.
>     int delay = TURN_PERMISSION_TIMEOUT - 60000;
>     entry_->SendChannelBindRequest(delay);
>-    LOG_J(LS_INFO, port_) << "Scheduled channel bind in " << delay << "ms.";
>+    RTC_LOG(LS_INFO) << port_->ToString() << ": Scheduled channel bind in "
>+                     << delay << "ms.";
>   }
> }
> 
> void TurnChannelBindRequest::OnErrorResponse(StunMessage* response) {
>   int error_code = response->GetErrorCodeValue();
>-  LOG_J(LS_WARNING, port_) << "Received TURN channel bind error response"
>-                           << ", id=" << rtc::hex_encode(id())
>-                           << ", code=" << error_code << ", rtt=" << Elapsed();
>+  RTC_LOG(LS_WARNING) << port_->ToString()
>+                      << ": Received TURN channel bind error response, id="
>+                      << rtc::hex_encode(id()) << ", code=" << error_code
>+                      << ", rtt=" << Elapsed();
>   if (entry_) {
>     entry_->OnChannelBindError(response, error_code);
>   }
> }
> 
> void TurnChannelBindRequest::OnTimeout() {
>-  LOG_J(LS_WARNING, port_) << "TURN channel bind timeout "
>-                           << rtc::hex_encode(id());
>+  RTC_LOG(LS_WARNING) << port_->ToString() << ": TURN channel bind timeout "
>+                      << rtc::hex_encode(id());
>   if (entry_) {
>     entry_->OnChannelBindTimeout();
>   }
>@@ -1573,7 +1664,8 @@ void TurnChannelBindRequest::OnEntryDestroyed(TurnEntry* entry) {
>   entry_ = NULL;
> }
> 
>-TurnEntry::TurnEntry(TurnPort* port, int channel_id,
>+TurnEntry::TurnEntry(TurnPort* port,
>+                     int channel_id,
>                      const rtc::SocketAddress& ext_addr)
>     : port_(port),
>       channel_id_(channel_id),
>@@ -1589,11 +1681,13 @@ void TurnEntry::SendCreatePermissionRequest(int delay) {
> }
> 
> void TurnEntry::SendChannelBindRequest(int delay) {
>-  port_->SendRequest(new TurnChannelBindRequest(
>-      port_, this, channel_id_, ext_addr_), delay);
>+  port_->SendRequest(
>+      new TurnChannelBindRequest(port_, this, channel_id_, ext_addr_), delay);
> }
> 
>-int TurnEntry::Send(const void* data, size_t size, bool payload,
>+int TurnEntry::Send(const void* data,
>+                    size_t size,
>+                    bool payload,
>                     const rtc::PacketOptions& options) {
>   rtc::ByteBufferWriter buf;
>   if (state_ != STATE_BOUND ||
>@@ -1602,12 +1696,11 @@ int TurnEntry::Send(const void* data, size_t size, bool payload,
>     // The turn_customizer_ can also make us use Send Indication.
>     TurnMessage msg;
>     msg.SetType(TURN_SEND_INDICATION);
>-    msg.SetTransactionID(
>-        rtc::CreateRandomString(kStunTransactionIdLength));
>-    msg.AddAttribute(rtc::MakeUnique<StunXorAddressAttribute>(
>+    msg.SetTransactionID(rtc::CreateRandomString(kStunTransactionIdLength));
>+    msg.AddAttribute(absl::make_unique<StunXorAddressAttribute>(
>         STUN_ATTR_XOR_PEER_ADDRESS, ext_addr_));
>     msg.AddAttribute(
>-        rtc::MakeUnique<StunByteStringAttribute>(STUN_ATTR_DATA, data, size));
>+        absl::make_unique<StunByteStringAttribute>(STUN_ATTR_DATA, data, size));
> 
>     port_->TurnCustomizerMaybeModifyOutgoingStunMessage(&msg);
> 
>@@ -1625,13 +1718,15 @@ int TurnEntry::Send(const void* data, size_t size, bool payload,
>     buf.WriteUInt16(static_cast<uint16_t>(size));
>     buf.WriteBytes(reinterpret_cast<const char*>(data), size);
>   }
>-  return port_->Send(buf.Data(), buf.Length(), options);
>+  rtc::PacketOptions modified_options(options);
>+  modified_options.info_signaled_after_sent.turn_overhead_bytes =
>+      buf.Length() - size;
>+  return port_->Send(buf.Data(), buf.Length(), modified_options);
> }
> 
> void TurnEntry::OnCreatePermissionSuccess() {
>-  LOG_J(LS_INFO, port_) << "Create permission for "
>-                        << ext_addr_.ToSensitiveString()
>-                        << " succeeded";
>+  RTC_LOG(LS_INFO) << port_->ToString() << ": Create permission for "
>+                   << ext_addr_.ToSensitiveString() << " succeeded";
>   port_->SignalCreatePermissionResult(port_, ext_addr_,
>                                       TURN_SUCCESS_RESULT_CODE);
> 
>@@ -1642,8 +1737,9 @@ void TurnEntry::OnCreatePermissionSuccess() {
>     // times out.
>     int delay = TURN_PERMISSION_TIMEOUT - 60000;
>     SendCreatePermissionRequest(delay);
>-    LOG_J(LS_INFO, port_) << "Scheduled create-permission-request in "
>-                          << delay << "ms.";
>+    RTC_LOG(LS_INFO) << port_->ToString()
>+                     << ": Scheduled create-permission-request in " << delay
>+                     << "ms.";
>   }
> }
> 
>@@ -1656,7 +1752,8 @@ void TurnEntry::OnCreatePermissionError(StunMessage* response, int code) {
>     bool found = port_->FailAndPruneConnection(ext_addr_);
>     if (found) {
>       RTC_LOG(LS_ERROR) << "Received TURN CreatePermission error response, "
>-                        << "code=" << code << "; pruned connection.";
>+                           "code="
>+                        << code << "; pruned connection.";
>     }
>     // Send signal with error code.
>     port_->SignalCreatePermissionResult(port_, ext_addr_, code);
>@@ -1668,8 +1765,8 @@ void TurnEntry::OnCreatePermissionTimeout() {
> }
> 
> void TurnEntry::OnChannelBindSuccess() {
>-  LOG_J(LS_INFO, port_) << "Channel bind for " << ext_addr_.ToSensitiveString()
>-                        << " succeeded";
>+  RTC_LOG(LS_INFO) << port_->ToString() << ": Successful channel bind for "
>+                   << ext_addr_.ToSensitiveString();
>   RTC_DCHECK(state_ == STATE_BINDING || state_ == STATE_BOUND);
>   state_ = STATE_BOUND;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/turnport.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/turnport.h
>index 4ee5f891fad..5955aafda13 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/turnport.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/turnport.h
>@@ -22,6 +22,7 @@
> #include "p2p/client/basicportallocator.h"
> #include "rtc_base/asyncinvoker.h"
> #include "rtc_base/asyncpacketsocket.h"
>+#include "rtc_base/sslcertificate.h"
> 
> namespace rtc {
> class AsyncResolver;
>@@ -67,24 +68,26 @@ class TurnPort : public Port {
> 
>   // Create a TURN port that will use a new socket, bound to |network| and
>   // using a port in the range between |min_port| and |max_port|.
>-  static TurnPort* Create(rtc::Thread* thread,
>-                          rtc::PacketSocketFactory* factory,
>-                          rtc::Network* network,
>-                          uint16_t min_port,
>-                          uint16_t max_port,
>-                          const std::string& username,  // ice username.
>-                          const std::string& password,  // ice password.
>-                          const ProtocolAddress& server_address,
>-                          const RelayCredentials& credentials,
>-                          int server_priority,
>-                          const std::string& origin,
>-                          const std::vector<std::string>& tls_alpn_protocols,
>-                          const std::vector<std::string>& tls_elliptic_curves,
>-                          webrtc::TurnCustomizer* customizer) {
>+  static TurnPort* Create(
>+      rtc::Thread* thread,
>+      rtc::PacketSocketFactory* factory,
>+      rtc::Network* network,
>+      uint16_t min_port,
>+      uint16_t max_port,
>+      const std::string& username,  // ice username.
>+      const std::string& password,  // ice password.
>+      const ProtocolAddress& server_address,
>+      const RelayCredentials& credentials,
>+      int server_priority,
>+      const std::string& origin,
>+      const std::vector<std::string>& tls_alpn_protocols,
>+      const std::vector<std::string>& tls_elliptic_curves,
>+      webrtc::TurnCustomizer* customizer,
>+      rtc::SSLCertificateVerifier* tls_cert_verifier = nullptr) {
>     return new TurnPort(thread, factory, network, min_port, max_port, username,
>                         password, server_address, credentials, server_priority,
>                         origin, tls_alpn_protocols, tls_elliptic_curves,
>-                        customizer);
>+                        customizer, tls_cert_verifier);
>   }
> 
>   ~TurnPort() override;
>@@ -107,6 +110,10 @@ class TurnPort : public Port {
>   virtual std::vector<std::string> GetTlsAlpnProtocols() const;
>   virtual std::vector<std::string> GetTlsEllipticCurves() const;
> 
>+  // Release a TURN allocation by sending a refresh with lifetime 0.
>+  // Sets state to STATE_RECEIVEONLY.
>+  void Release();
>+
>   void PrepareAddress() override;
>   Connection* CreateConnection(const Candidate& c,
>                                PortInterface::CandidateOrigin origin) override;
>@@ -161,6 +168,11 @@ class TurnPort : public Port {
>                    const rtc::SocketAddress&,
>                    const rtc::SocketAddress&> SignalResolvedServerAddress;
> 
>+  // Signal when TurnPort is closed,
>+  // e.g remote socket closed (TCP)
>+  //  or receiveing a REFRESH response with lifetime 0.
>+  sigslot::signal1<TurnPort*> SignalTurnPortClosed;
>+
>   // All public methods/signals below are for testing only.
>   sigslot::signal2<TurnPort*, int> SignalTurnRefreshResult;
>   sigslot::signal3<TurnPort*, const rtc::SocketAddress&, int>
>@@ -205,7 +217,8 @@ class TurnPort : public Port {
>            const std::string& origin,
>            const std::vector<std::string>& tls_alpn_protocols,
>            const std::vector<std::string>& tls_elliptic_curves,
>-           webrtc::TurnCustomizer* customizer);
>+           webrtc::TurnCustomizer* customizer,
>+           rtc::SSLCertificateVerifier* tls_cert_verifier = nullptr);
> 
>   // NOTE: This method needs to be accessible for StacPort
>   // return true if entry was created (i.e channel_number consumed).
>@@ -217,7 +230,8 @@ class TurnPort : public Port {
>     MSG_ALLOCATE_ERROR = MSG_FIRST_AVAILABLE,
>     MSG_ALLOCATE_MISMATCH,
>     MSG_TRY_ALTERNATE_SERVER,
>-    MSG_REFRESH_ERROR
>+    MSG_REFRESH_ERROR,
>+    MSG_ALLOCATION_RELEASED
>   };
> 
>   typedef std::list<TurnEntry*> EntryList;
>@@ -260,7 +274,7 @@ class TurnPort : public Port {
>       const rtc::SocketAddress& remote_addr,
>       ProtocolType proto, const rtc::PacketTime& packet_time);
> 
>-  bool ScheduleRefresh(int lifetime);
>+  bool ScheduleRefresh(uint32_t lifetime);
>   void SendRequest(StunRequest* request, int delay);
>   int Send(const void* data, size_t size,
>            const rtc::PacketOptions& options);
>@@ -293,6 +307,7 @@ class TurnPort : public Port {
>   TlsCertPolicy tls_cert_policy_ = TlsCertPolicy::TLS_CERT_POLICY_SECURE;
>   std::vector<std::string> tls_alpn_protocols_;
>   std::vector<std::string> tls_elliptic_curves_;
>+  rtc::SSLCertificateVerifier* tls_cert_verifier_;
>   RelayCredentials credentials_;
>   AttemptedServerSet attempted_server_addresses_;
> 
>@@ -319,7 +334,8 @@ class TurnPort : public Port {
> 
>   rtc::AsyncInvoker invoker_;
> 
>-  // Optional TurnCustomizer that can modify outgoing messages.
>+  // Optional TurnCustomizer that can modify outgoing messages. Once set, this
>+  // must outlive the TurnPort's lifetime.
>   webrtc::TurnCustomizer *turn_customizer_ = nullptr;
> 
>   friend class TurnEntry;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/turnport_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/turnport_unittest.cc
>index 0e9d1e2112e..bac35e891cd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/turnport_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/turnport_unittest.cc
>@@ -16,6 +16,7 @@
> #include <utility>
> #include <vector>
> 
>+#include "absl/memory/memory.h"
> #include "p2p/base/basicpacketsocketfactory.h"
> #include "p2p/base/p2pconstants.h"
> #include "p2p/base/portallocator.h"
>@@ -33,7 +34,6 @@
> #include "rtc_base/gunit.h"
> #include "rtc_base/helpers.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/socketadapters.h"
> #include "rtc_base/socketaddress.h"
> #include "rtc_base/ssladapter.h"
>@@ -44,10 +44,10 @@ using rtc::SocketAddress;
> 
> static const SocketAddress kLocalAddr1("11.11.11.11", 0);
> static const SocketAddress kLocalAddr2("22.22.22.22", 0);
>-static const SocketAddress kLocalIPv6Addr(
>-    "2401:fa00:4:1000:be30:5bff:fee5:c3", 0);
>-static const SocketAddress kLocalIPv6Addr2(
>-    "2401:fa00:4:2000:be30:5bff:fee5:d4", 0);
>+static const SocketAddress kLocalIPv6Addr("2401:fa00:4:1000:be30:5bff:fee5:c3",
>+                                          0);
>+static const SocketAddress kLocalIPv6Addr2("2401:fa00:4:2000:be30:5bff:fee5:d4",
>+                                           0);
> static const SocketAddress kTurnUdpIntAddr("99.99.99.3",
>                                            cricket::TURN_SERVER_PORT);
> static const SocketAddress kTurnTcpIntAddr("99.99.99.4",
>@@ -61,7 +61,8 @@ static const SocketAddress kTurnIPv6IntAddr(
>     "2400:4030:2:2c00:be30:abcd:efab:cdef",
>     cricket::TURN_SERVER_PORT);
> static const SocketAddress kTurnUdpIPv6IntAddr(
>-    "2400:4030:1:2c00:be30:abcd:efab:cdef", cricket::TURN_SERVER_PORT);
>+    "2400:4030:1:2c00:be30:abcd:efab:cdef",
>+    cricket::TURN_SERVER_PORT);
> 
> static const char kCandidateFoundation[] = "foundation";
> static const char kIceUfrag1[] = "TESTICEUFRAG0001";
>@@ -81,22 +82,22 @@ static constexpr unsigned int kConnectionDestructionDelay = 1;
> // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=5191
> static constexpr unsigned int kResolverTimeout = 10000;
> 
>-static const cricket::ProtocolAddress kTurnUdpProtoAddr(
>-    kTurnUdpIntAddr, cricket::PROTO_UDP);
>-static const cricket::ProtocolAddress kTurnTcpProtoAddr(
>-    kTurnTcpIntAddr, cricket::PROTO_TCP);
>+static const cricket::ProtocolAddress kTurnUdpProtoAddr(kTurnUdpIntAddr,
>+                                                        cricket::PROTO_UDP);
>+static const cricket::ProtocolAddress kTurnTcpProtoAddr(kTurnTcpIntAddr,
>+                                                        cricket::PROTO_TCP);
> static const cricket::ProtocolAddress kTurnTlsProtoAddr(kTurnTcpIntAddr,
>                                                         cricket::PROTO_TLS);
>-static const cricket::ProtocolAddress kTurnUdpIPv6ProtoAddr(
>-    kTurnUdpIPv6IntAddr, cricket::PROTO_UDP);
>+static const cricket::ProtocolAddress kTurnUdpIPv6ProtoAddr(kTurnUdpIPv6IntAddr,
>+                                                            cricket::PROTO_UDP);
> 
> static const unsigned int MSG_TESTFINISH = 0;
> 
> #if defined(WEBRTC_LINUX) && !defined(WEBRTC_ANDROID)
> static int GetFDCount() {
>-  struct dirent *dp;
>+  struct dirent* dp;
>   int fd_count = 0;
>-  DIR *dir = opendir("/proc/self/fd/");
>+  DIR* dir = opendir("/proc/self/fd/");
>   while ((dp = readdir(dir)) != NULL) {
>     if (dp->d_name[0] == '.')
>       continue;
>@@ -154,12 +155,14 @@ class TurnPortTest : public testing::Test,
>         turn_error_(false),
>         turn_unknown_address_(false),
>         turn_create_permission_success_(false),
>+        turn_port_closed_(false),
>+        turn_port_destroyed_(false),
>         udp_ready_(false),
>         test_finish_(false) {
>     // Some code uses "last received time == 0" to represent "nothing received
>     // so far", so we need to start the fake clock at a nonzero time...
>     // TODO(deadbeef): Fix this.
>-    fake_clock_.AdvanceTime(rtc::TimeDelta::FromSeconds(1));
>+    fake_clock_.AdvanceTime(webrtc::TimeDelta::seconds(1));
>   }
> 
>   virtual void OnMessage(rtc::Message* msg) {
>@@ -168,12 +171,8 @@ class TurnPortTest : public testing::Test,
>       test_finish_ = true;
>   }
> 
>-  void OnTurnPortComplete(Port* port) {
>-    turn_ready_ = true;
>-  }
>-  void OnTurnPortError(Port* port) {
>-    turn_error_ = true;
>-  }
>+  void OnTurnPortComplete(Port* port) { turn_ready_ = true; }
>+  void OnTurnPortError(Port* port) { turn_error_ = true; }
>   void OnTurnUnknownAddress(PortInterface* port,
>                             const SocketAddress& addr,
>                             ProtocolType proto,
>@@ -192,26 +191,32 @@ class TurnPortTest : public testing::Test,
>   void OnTurnRefreshResult(TurnPort* port, int code) {
>     turn_refresh_success_ = (code == 0);
>   }
>-  void OnTurnReadPacket(Connection* conn, const char* data, size_t size,
>+  void OnTurnReadPacket(Connection* conn,
>+                        const char* data,
>+                        size_t size,
>                         const rtc::PacketTime& packet_time) {
>     turn_packets_.push_back(rtc::Buffer(data, size));
>   }
>-  void OnUdpPortComplete(Port* port) {
>-    udp_ready_ = true;
>-  }
>-  void OnUdpReadPacket(Connection* conn, const char* data, size_t size,
>+  void OnUdpPortComplete(Port* port) { udp_ready_ = true; }
>+  void OnUdpReadPacket(Connection* conn,
>+                       const char* data,
>+                       size_t size,
>                        const rtc::PacketTime& packet_time) {
>     udp_packets_.push_back(rtc::Buffer(data, size));
>   }
>   void OnSocketReadPacket(rtc::AsyncPacketSocket* socket,
>-                          const char* data, size_t size,
>+                          const char* data,
>+                          size_t size,
>                           const rtc::SocketAddress& remote_addr,
>                           const rtc::PacketTime& packet_time) {
>     turn_port_->HandleIncomingPacket(socket, data, size, remote_addr,
>                                      packet_time);
>   }
>+  void OnTurnPortClosed(TurnPort* port) { turn_port_closed_ = true; }
>+  void OnTurnPortDestroyed(PortInterface* port) { turn_port_destroyed_ = true; }
>+
>   rtc::AsyncSocket* CreateServerSocket(const SocketAddress addr) {
>-    rtc::AsyncSocket* socket = ss_->CreateAsyncSocket(SOCK_STREAM);
>+    rtc::AsyncSocket* socket = ss_->CreateAsyncSocket(AF_INET, SOCK_STREAM);
>     EXPECT_GE(socket->Bind(addr), 0);
>     EXPECT_GE(socket->Listen(5), 0);
>     return socket;
>@@ -291,15 +296,15 @@ class TurnPortTest : public testing::Test,
>       socket_.reset(socket_factory_.CreateUdpSocket(
>           rtc::SocketAddress(kLocalAddr1.ipaddr(), 0), 0, 0));
>       ASSERT_TRUE(socket_ != NULL);
>-      socket_->SignalReadPacket.connect(
>-          this, &TurnPortTest::OnSocketReadPacket);
>+      socket_->SignalReadPacket.connect(this,
>+                                        &TurnPortTest::OnSocketReadPacket);
>     }
> 
>     RelayCredentials credentials(username, password);
>-    turn_port_.reset(TurnPort::Create(
>-        &main_, &socket_factory_, MakeNetwork(kLocalAddr1), socket_.get(),
>-        kIceUfrag1, kIcePwd1, server_address, credentials, 0, std::string(),
>-        nullptr));
>+    turn_port_.reset(TurnPort::Create(&main_, &socket_factory_,
>+                                      MakeNetwork(kLocalAddr1), socket_.get(),
>+                                      kIceUfrag1, kIcePwd1, server_address,
>+                                      credentials, 0, std::string(), nullptr));
>     // This TURN port will be the controlling.
>     turn_port_->SetIceRole(ICEROLE_CONTROLLING);
>     ConnectSignals();
>@@ -307,27 +312,30 @@ class TurnPortTest : public testing::Test,
> 
>   void ConnectSignals() {
>     turn_port_->SignalPortComplete.connect(this,
>-        &TurnPortTest::OnTurnPortComplete);
>-    turn_port_->SignalPortError.connect(this,
>-        &TurnPortTest::OnTurnPortError);
>-    turn_port_->SignalUnknownAddress.connect(this,
>-        &TurnPortTest::OnTurnUnknownAddress);
>-    turn_port_->SignalCreatePermissionResult.connect(this,
>-        &TurnPortTest::OnTurnCreatePermissionResult);
>+                                           &TurnPortTest::OnTurnPortComplete);
>+    turn_port_->SignalPortError.connect(this, &TurnPortTest::OnTurnPortError);
>+    turn_port_->SignalUnknownAddress.connect(
>+        this, &TurnPortTest::OnTurnUnknownAddress);
>+    turn_port_->SignalCreatePermissionResult.connect(
>+        this, &TurnPortTest::OnTurnCreatePermissionResult);
>     turn_port_->SignalTurnRefreshResult.connect(
>         this, &TurnPortTest::OnTurnRefreshResult);
>+    turn_port_->SignalTurnPortClosed.connect(this,
>+                                             &TurnPortTest::OnTurnPortClosed);
>+    turn_port_->SignalDestroyed.connect(this,
>+                                        &TurnPortTest::OnTurnPortDestroyed);
>   }
> 
>   void CreateUdpPort() { CreateUdpPort(kLocalAddr2); }
> 
>   void CreateUdpPort(const SocketAddress& address) {
>-    udp_port_.reset(UDPPort::Create(&main_, &socket_factory_,
>-                                    MakeNetwork(address), 0, 0, kIceUfrag2,
>-                                    kIcePwd2, std::string(), false));
>+    udp_port_.reset(UDPPort::Create(
>+        &main_, &socket_factory_, MakeNetwork(address), 0, 0, kIceUfrag2,
>+        kIcePwd2, std::string(), false, absl::nullopt));
>     // UDP port will be controlled.
>     udp_port_->SetIceRole(ICEROLE_CONTROLLED);
>-    udp_port_->SignalPortComplete.connect(
>-        this, &TurnPortTest::OnUdpPortComplete);
>+    udp_port_->SignalPortComplete.connect(this,
>+                                          &TurnPortTest::OnUdpPortComplete);
>   }
> 
>   void PrepareTurnAndUdpPorts(ProtocolType protocol_type) {
>@@ -517,7 +525,7 @@ class TurnPortTest : public testing::Test,
>     // Make a socket and bind it to the local port, to make extra sure no
>     // packet is sent to this address.
>     std::unique_ptr<rtc::Socket> loopback_socket(ss_->CreateSocket(
>-        protocol_type == PROTO_UDP ? SOCK_DGRAM : SOCK_STREAM));
>+        AF_INET, protocol_type == PROTO_UDP ? SOCK_DGRAM : SOCK_STREAM));
>     ASSERT_NE(nullptr, loopback_socket.get());
>     ASSERT_EQ(0, loopback_socket->Bind(loopback_address));
>     if (protocol_type == PROTO_TCP) {
>@@ -555,8 +563,8 @@ class TurnPortTest : public testing::Test,
> 
>     // Send ping from UDP to TURN.
>     ASSERT_GE(turn_port_->Candidates().size(), 1U);
>-    Connection* conn1 = udp_port_->CreateConnection(
>-                    turn_port_->Candidates()[0], Port::ORIGIN_MESSAGE);
>+    Connection* conn1 = udp_port_->CreateConnection(turn_port_->Candidates()[0],
>+                                                    Port::ORIGIN_MESSAGE);
>     ASSERT_TRUE(conn1 != NULL);
>     conn1->Ping(0);
>     SIMULATED_WAIT(!turn_unknown_address_, kSimulatedRtt * 2, fake_clock_);
>@@ -565,8 +573,8 @@ class TurnPortTest : public testing::Test,
>     EXPECT_EQ(Connection::STATE_WRITE_INIT, conn1->write_state());
> 
>     // Send ping from TURN to UDP.
>-    Connection* conn2 = turn_port_->CreateConnection(
>-                    udp_port_->Candidates()[0], Port::ORIGIN_MESSAGE);
>+    Connection* conn2 = turn_port_->CreateConnection(udp_port_->Candidates()[0],
>+                                                     Port::ORIGIN_MESSAGE);
>     ASSERT_TRUE(conn2 != NULL);
>     ASSERT_TRUE_SIMULATED_WAIT(turn_create_permission_success_, kSimulatedRtt,
>                                fake_clock_);
>@@ -630,10 +638,10 @@ class TurnPortTest : public testing::Test,
>     PrepareTurnAndUdpPorts(protocol_type);
> 
>     // Create connections and send pings.
>-    Connection* conn1 = turn_port_->CreateConnection(
>-        udp_port_->Candidates()[0], Port::ORIGIN_MESSAGE);
>-    Connection* conn2 = udp_port_->CreateConnection(
>-        turn_port_->Candidates()[0], Port::ORIGIN_MESSAGE);
>+    Connection* conn1 = turn_port_->CreateConnection(udp_port_->Candidates()[0],
>+                                                     Port::ORIGIN_MESSAGE);
>+    Connection* conn2 = udp_port_->CreateConnection(turn_port_->Candidates()[0],
>+                                                    Port::ORIGIN_MESSAGE);
>     ASSERT_TRUE(conn1 != NULL);
>     ASSERT_TRUE(conn2 != NULL);
>     conn1->SignalReadPacket.connect(static_cast<TurnPortTest*>(this),
>@@ -650,7 +658,7 @@ class TurnPortTest : public testing::Test,
>     // Send some data.
>     size_t num_packets = 256;
>     for (size_t i = 0; i < num_packets; ++i) {
>-      unsigned char buf[256] = { 0 };
>+      unsigned char buf[256] = {0};
>       for (size_t j = 0; j < i + 1; ++j) {
>         buf[j] = 0xFF - static_cast<unsigned char>(j);
>       }
>@@ -677,6 +685,48 @@ class TurnPortTest : public testing::Test,
>                              kSimulatedRtt, fake_clock_);
>   }
> 
>+  // Test that the TURN allocation is released by sending a refresh request
>+  // with lifetime 0 when Release is called.
>+  void TestTurnGracefulReleaseAllocation(ProtocolType protocol_type) {
>+    PrepareTurnAndUdpPorts(protocol_type);
>+
>+    // Create connections and send pings.
>+    Connection* conn1 = turn_port_->CreateConnection(udp_port_->Candidates()[0],
>+                                                     Port::ORIGIN_MESSAGE);
>+    Connection* conn2 = udp_port_->CreateConnection(turn_port_->Candidates()[0],
>+                                                    Port::ORIGIN_MESSAGE);
>+    ASSERT_TRUE(conn1 != NULL);
>+    ASSERT_TRUE(conn2 != NULL);
>+    conn1->SignalReadPacket.connect(static_cast<TurnPortTest*>(this),
>+                                    &TurnPortTest::OnTurnReadPacket);
>+    conn2->SignalReadPacket.connect(static_cast<TurnPortTest*>(this),
>+                                    &TurnPortTest::OnUdpReadPacket);
>+    conn1->Ping(0);
>+    EXPECT_EQ_SIMULATED_WAIT(Connection::STATE_WRITABLE, conn1->write_state(),
>+                             kSimulatedRtt * 2, fake_clock_);
>+    conn2->Ping(0);
>+    EXPECT_EQ_SIMULATED_WAIT(Connection::STATE_WRITABLE, conn2->write_state(),
>+                             kSimulatedRtt * 2, fake_clock_);
>+
>+    // Send some data from Udp to TurnPort.
>+    unsigned char buf[256] = {0};
>+    conn2->Send(buf, sizeof(buf), options);
>+
>+    // Now release the TurnPort allocation.
>+    // This will send a REFRESH with lifetime 0 to server.
>+    turn_port_->Release();
>+
>+    // Wait for the TurnPort to signal closed.
>+    ASSERT_TRUE_SIMULATED_WAIT(turn_port_closed_, kSimulatedRtt, fake_clock_);
>+
>+    // But the data should have arrived first.
>+    ASSERT_EQ(1ul, turn_packets_.size());
>+    EXPECT_EQ(sizeof(buf), turn_packets_[0].size());
>+
>+    // The allocation is released at server.
>+    EXPECT_EQ(0U, turn_server_.server()->allocations().size());
>+  }
>+
>  protected:
>   rtc::ScopedFakeClock fake_clock_;
>   // When a "create port" helper method is called with an IP, we create a
>@@ -694,6 +744,8 @@ class TurnPortTest : public testing::Test,
>   bool turn_error_;
>   bool turn_unknown_address_;
>   bool turn_create_permission_success_;
>+  bool turn_port_closed_;
>+  bool turn_port_destroyed_;
>   bool udp_ready_;
>   bool test_finish_;
>   bool turn_refresh_success_ = false;
>@@ -739,7 +791,7 @@ TEST_F(TurnPortTest, TestReconstructedServerUrlForTls) {
> // Do a normal TURN allocation.
> TEST_F(TurnPortTest, TestTurnAllocate) {
>   CreateTurnPort(kTurnUsername, kTurnPassword, kTurnUdpProtoAddr);
>-  EXPECT_EQ(0, turn_port_->SetOption(rtc::Socket::OPT_SNDBUF, 10*1024));
>+  EXPECT_EQ(0, turn_port_->SetOption(rtc::Socket::OPT_SNDBUF, 10 * 1024));
>   turn_port_->PrepareAddress();
>   EXPECT_TRUE_SIMULATED_WAIT(turn_ready_, kSimulatedRtt * 2, fake_clock_);
>   ASSERT_EQ(1U, turn_port_->Candidates().size());
>@@ -752,7 +804,7 @@ TEST_F(TurnPortTest, TestTurnAllocate) {
> TEST_F(TurnPortTest, TestTurnTcpAllocate) {
>   turn_server_.AddInternalSocket(kTurnTcpIntAddr, PROTO_TCP);
>   CreateTurnPort(kTurnUsername, kTurnPassword, kTurnTcpProtoAddr);
>-  EXPECT_EQ(0, turn_port_->SetOption(rtc::Socket::OPT_SNDBUF, 10*1024));
>+  EXPECT_EQ(0, turn_port_->SetOption(rtc::Socket::OPT_SNDBUF, 10 * 1024));
>   turn_port_->PrepareAddress();
>   EXPECT_TRUE_SIMULATED_WAIT(turn_ready_, kSimulatedRtt * 3, fake_clock_);
>   ASSERT_EQ(1U, turn_port_->Candidates().size());
>@@ -840,6 +892,28 @@ TEST_F(TurnPortTest, TurnTcpAllocationNotDiscardedIfNotBoundToBestIP) {
>             turn_port_->Candidates()[0].related_address().ipaddr());
> }
> 
>+// Regression test for crbug.com/webrtc/8972, caused by buggy comparison
>+// between rtc::IPAddress and rtc::InterfaceAddress.
>+TEST_F(TurnPortTest, TCPPortNotDiscardedIfBoundToTemporaryIP) {
>+  networks_.emplace_back("unittest", "unittest", kLocalIPv6Addr.ipaddr(), 32);
>+  networks_.back().AddIP(rtc::InterfaceAddress(
>+      kLocalIPv6Addr.ipaddr(), rtc::IPV6_ADDRESS_FLAG_TEMPORARY));
>+
>+  // Set up TURN server to use TCP (this logic only exists for TCP).
>+  turn_server_.AddInternalSocket(kTurnIPv6IntAddr, PROTO_TCP);
>+
>+  // Create TURN port using our special Network, and tell it to start
>+  // allocation.
>+  CreateTurnPortWithNetwork(
>+      &networks_.back(), kTurnUsername, kTurnPassword,
>+      cricket::ProtocolAddress(kTurnIPv6IntAddr, PROTO_TCP));
>+  turn_port_->PrepareAddress();
>+
>+  // Candidate should be gathered as normally.
>+  EXPECT_TRUE_SIMULATED_WAIT(turn_ready_, kSimulatedRtt * 3, fake_clock_);
>+  ASSERT_EQ(1U, turn_port_->Candidates().size());
>+}
>+
> // Testing turn port will attempt to create TCP socket on address resolution
> // failure.
> TEST_F(TurnPortTest, TestTurnTcpOnAddressResolveFailure) {
>@@ -1429,6 +1503,24 @@ TEST_F(TurnPortTest, TestTurnTLSReleaseAllocation) {
>   TestTurnReleaseAllocation(PROTO_TLS);
> }
> 
>+TEST_F(TurnPortTest, TestTurnUDPGracefulReleaseAllocation) {
>+  turn_server_.AddInternalSocket(kTurnTcpIntAddr, PROTO_UDP);
>+  CreateTurnPort(kTurnUsername, kTurnPassword, kTurnUdpProtoAddr);
>+  TestTurnGracefulReleaseAllocation(PROTO_UDP);
>+}
>+
>+TEST_F(TurnPortTest, TestTurnTCPGracefulReleaseAllocation) {
>+  turn_server_.AddInternalSocket(kTurnTcpIntAddr, PROTO_TCP);
>+  CreateTurnPort(kTurnUsername, kTurnPassword, kTurnTcpProtoAddr);
>+  TestTurnGracefulReleaseAllocation(PROTO_TCP);
>+}
>+
>+TEST_F(TurnPortTest, TestTurnTLSGracefulReleaseAllocation) {
>+  turn_server_.AddInternalSocket(kTurnTcpIntAddr, PROTO_TLS);
>+  CreateTurnPort(kTurnUsername, kTurnPassword, kTurnTlsProtoAddr);
>+  TestTurnGracefulReleaseAllocation(PROTO_TLS);
>+}
>+
> // Test that nothing bad happens if we try to create a connection to the same
> // remote address twice. Previously there was a bug that caused this to hit a
> // DCHECK.
>@@ -1464,11 +1556,11 @@ TEST_F(TurnPortTest, TestResolverShutdown) {
> }
> #endif
> 
>-class MessageObserver : public StunMessageObserver{
>+class MessageObserver : public StunMessageObserver {
>  public:
>-  MessageObserver(unsigned int *message_counter,
>+  MessageObserver(unsigned int* message_counter,
>                   unsigned int* channel_data_counter,
>-                  unsigned int *attr_counter)
>+                  unsigned int* attr_counter)
>       : message_counter_(message_counter),
>         channel_data_counter_(channel_data_counter),
>         attr_counter_(attr_counter) {}
>@@ -1478,8 +1570,8 @@ class MessageObserver : public StunMessageObserver{
>       (*message_counter_)++;
>     }
>     // Implementation defined attributes are returned as ByteString
>-    const StunByteStringAttribute* attr = msg->GetByteString(
>-        TestTurnCustomizer::STUN_ATTR_COUNTER);
>+    const StunByteStringAttribute* attr =
>+        msg->GetByteString(TestTurnCustomizer::STUN_ATTR_COUNTER);
>     if (attr != nullptr && attr_counter_ != nullptr) {
>       rtc::ByteBufferReader buf(attr->bytes(), attr->length());
>       unsigned int val = ~0u;
>@@ -1512,8 +1604,7 @@ TEST_F(TurnPortTest, TestTurnCustomizerCount) {
>   unsigned int observer_attr_counter = 0;
>   TestTurnCustomizer* customizer = new TestTurnCustomizer();
>   std::unique_ptr<MessageObserver> validator(new MessageObserver(
>-      &observer_message_counter,
>-      &observer_channel_data_counter,
>+      &observer_message_counter, &observer_channel_data_counter,
>       &observer_attr_counter));
> 
>   turn_server_.AddInternalSocket(kTurnTcpIntAddr, PROTO_TLS);
>@@ -1543,8 +1634,7 @@ TEST_F(TurnPortTest, TestTurnCustomizerDisallowChannelData) {
>   unsigned int observer_attr_counter = 0;
>   TestTurnCustomizer* customizer = new TestTurnCustomizer();
>   std::unique_ptr<MessageObserver> validator(new MessageObserver(
>-      &observer_message_counter,
>-      &observer_channel_data_counter,
>+      &observer_message_counter, &observer_channel_data_counter,
>       &observer_attr_counter));
>   customizer->allow_channel_data_ = false;
>   turn_server_.AddInternalSocket(kTurnTcpIntAddr, PROTO_TLS);
>@@ -1573,8 +1663,7 @@ TEST_F(TurnPortTest, TestTurnCustomizerAddAttribute) {
>   unsigned int observer_attr_counter = 0;
>   TestTurnCustomizer* customizer = new TestTurnCustomizer();
>   std::unique_ptr<MessageObserver> validator(new MessageObserver(
>-      &observer_message_counter,
>-      &observer_channel_data_counter,
>+      &observer_message_counter, &observer_channel_data_counter,
>       &observer_attr_counter));
>   customizer->allow_channel_data_ = false;
>   customizer->add_counter_ = true;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/turnserver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/turnserver.cc
>index 902e889057b..a7a8c572325 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/turnserver.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/turnserver.cc
>@@ -13,8 +13,8 @@
> #include <tuple>  // for std::tie
> #include <utility>
> 
>+#include "absl/memory/memory.h"
> #include "p2p/base/asyncstuntcpsocket.h"
>-#include "p2p/base/common.h"
> #include "p2p/base/packetsocketfactory.h"
> #include "p2p/base/stun.h"
> #include "rtc_base/bind.h"
>@@ -23,7 +23,6 @@
> #include "rtc_base/helpers.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/messagedigest.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/socketadapters.h"
> #include "rtc_base/stringencode.h"
> #include "rtc_base/thread.h"
>@@ -115,7 +114,7 @@ static bool InitErrorResponse(const StunMessage* req, int code,
>     return false;
>   resp->SetType(resp_type);
>   resp->SetTransactionID(req->transaction_id());
>-  resp->AddAttribute(rtc::MakeUnique<cricket::StunErrorCodeAttribute>(
>+  resp->AddAttribute(absl::make_unique<cricket::StunErrorCodeAttribute>(
>       STUN_ATTR_ERROR_CODE, code, reason));
>   return true;
> }
>@@ -130,6 +129,7 @@ TurnServer::TurnServer(rtc::Thread* thread)
> }
> 
> TurnServer::~TurnServer() {
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>   for (InternalSocketMap::iterator it = server_sockets_.begin();
>        it != server_sockets_.end(); ++it) {
>     rtc::AsyncPacketSocket* socket = it->first;
>@@ -145,6 +145,7 @@ TurnServer::~TurnServer() {
> 
> void TurnServer::AddInternalSocket(rtc::AsyncPacketSocket* socket,
>                                    ProtocolType proto) {
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>   RTC_DCHECK(server_sockets_.end() == server_sockets_.find(socket));
>   server_sockets_[socket] = proto;
>   socket->SignalReadPacket.connect(this, &TurnServer::OnInternalPacket);
>@@ -152,6 +153,7 @@ void TurnServer::AddInternalSocket(rtc::AsyncPacketSocket* socket,
> 
> void TurnServer::AddInternalServerSocket(rtc::AsyncSocket* socket,
>                                          ProtocolType proto) {
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>   RTC_DCHECK(server_listen_sockets_.end() ==
>              server_listen_sockets_.find(socket));
>   server_listen_sockets_[socket] = proto;
>@@ -161,17 +163,20 @@ void TurnServer::AddInternalServerSocket(rtc::AsyncSocket* socket,
> void TurnServer::SetExternalSocketFactory(
>     rtc::PacketSocketFactory* factory,
>     const rtc::SocketAddress& external_addr) {
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>   external_socket_factory_.reset(factory);
>   external_addr_ = external_addr;
> }
> 
> void TurnServer::OnNewInternalConnection(rtc::AsyncSocket* socket) {
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>   RTC_DCHECK(server_listen_sockets_.find(socket) !=
>              server_listen_sockets_.end());
>   AcceptConnection(socket);
> }
> 
> void TurnServer::AcceptConnection(rtc::AsyncSocket* server_socket) {
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>   // Check if someone is trying to connect to us.
>   rtc::SocketAddress accept_addr;
>   rtc::AsyncSocket* accepted_socket = server_socket->Accept(&accept_addr);
>@@ -188,6 +193,7 @@ void TurnServer::AcceptConnection(rtc::AsyncSocket* server_socket) {
> 
> void TurnServer::OnInternalSocketClose(rtc::AsyncPacketSocket* socket,
>                                        int err) {
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>   DestroyInternalSocket(socket);
> }
> 
>@@ -195,6 +201,7 @@ void TurnServer::OnInternalPacket(rtc::AsyncPacketSocket* socket,
>                                   const char* data, size_t size,
>                                   const rtc::SocketAddress& addr,
>                                   const rtc::PacketTime& packet_time) {
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>   // Fail if the packet is too small to even contain a channel header.
>   if (size < TURN_CHANNEL_HEADER_SIZE) {
>     return;
>@@ -220,6 +227,7 @@ void TurnServer::OnInternalPacket(rtc::AsyncPacketSocket* socket,
> 
> void TurnServer::HandleStunMessage(TurnServerConnection* conn, const char* data,
>                                    size_t size) {
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>   TurnMessage msg;
>   rtc::ByteBufferReader buf(data, size);
>   if (!msg.Read(&buf) || (buf.Length() > 0)) {
>@@ -286,6 +294,7 @@ void TurnServer::HandleStunMessage(TurnServerConnection* conn, const char* data,
> }
> 
> bool TurnServer::GetKey(const StunMessage* msg, std::string* key) {
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>   const StunByteStringAttribute* username_attr =
>       msg->GetByteString(STUN_ATTR_USERNAME);
>   if (!username_attr) {
>@@ -300,6 +309,7 @@ bool TurnServer::CheckAuthorization(TurnServerConnection* conn,
>                                     const StunMessage* msg,
>                                     const char* data, size_t size,
>                                     const std::string& key) {
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>   // RFC 5389, 10.2.2.
>   RTC_DCHECK(IsStunRequestType(msg->type()));
>   const StunByteStringAttribute* mi_attr =
>@@ -358,11 +368,12 @@ bool TurnServer::CheckAuthorization(TurnServerConnection* conn,
> 
> void TurnServer::HandleBindingRequest(TurnServerConnection* conn,
>                                       const StunMessage* req) {
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>   StunMessage response;
>   InitResponse(req, &response);
> 
>   // Tell the user the address that we received their request from.
>-  auto mapped_addr_attr = rtc::MakeUnique<StunXorAddressAttribute>(
>+  auto mapped_addr_attr = absl::make_unique<StunXorAddressAttribute>(
>       STUN_ATTR_XOR_MAPPED_ADDRESS, conn->src());
>   response.AddAttribute(std::move(mapped_addr_attr));
> 
>@@ -372,6 +383,7 @@ void TurnServer::HandleBindingRequest(TurnServerConnection* conn,
> void TurnServer::HandleAllocateRequest(TurnServerConnection* conn,
>                                        const TurnMessage* msg,
>                                        const std::string& key) {
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>   // Check the parameters in the request.
>   const StunUInt32Attribute* transport_attr =
>       msg->GetUInt32(STUN_ATTR_REQUESTED_TRANSPORT);
>@@ -401,6 +413,7 @@ void TurnServer::HandleAllocateRequest(TurnServerConnection* conn,
> }
> 
> std::string TurnServer::GenerateNonce(int64_t now) const {
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>   // Generate a nonce of the form hex(now + HMAC-MD5(nonce_key_, now))
>   std::string input(reinterpret_cast<const char*>(&now), sizeof(now));
>   std::string nonce = rtc::hex_encode(input.c_str(), input.size());
>@@ -411,6 +424,7 @@ std::string TurnServer::GenerateNonce(int64_t now) const {
> }
> 
> bool TurnServer::ValidateNonce(const std::string& nonce) const {
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>   // Check the size.
>   if (nonce.size() != kNonceSize) {
>     return false;
>@@ -436,6 +450,7 @@ bool TurnServer::ValidateNonce(const std::string& nonce) const {
> }
> 
> TurnServerAllocation* TurnServer::FindAllocation(TurnServerConnection* conn) {
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>   AllocationMap::const_iterator it = allocations_.find(*conn);
>   return (it != allocations_.end()) ? it->second.get() : nullptr;
> }
>@@ -443,6 +458,7 @@ TurnServerAllocation* TurnServer::FindAllocation(TurnServerConnection* conn) {
> TurnServerAllocation* TurnServer::CreateAllocation(TurnServerConnection* conn,
>                                                    int proto,
>                                                    const std::string& key) {
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>   rtc::AsyncPacketSocket* external_socket = (external_socket_factory_) ?
>       external_socket_factory_->CreateUdpSocket(external_addr_, 0, 0) : NULL;
>   if (!external_socket) {
>@@ -460,6 +476,7 @@ TurnServerAllocation* TurnServer::CreateAllocation(TurnServerConnection* conn,
> void TurnServer::SendErrorResponse(TurnServerConnection* conn,
>                                    const StunMessage* req,
>                                    int code, const std::string& reason) {
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>   TurnMessage resp;
>   InitErrorResponse(req, code, reason, &resp);
>   RTC_LOG(LS_INFO) << "Sending error response, type=" << resp.type()
>@@ -470,6 +487,7 @@ void TurnServer::SendErrorResponse(TurnServerConnection* conn,
> void TurnServer::SendErrorResponseWithRealmAndNonce(
>     TurnServerConnection* conn, const StunMessage* msg,
>     int code, const std::string& reason) {
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>   TurnMessage resp;
>   InitErrorResponse(msg, code, reason, &resp);
> 
>@@ -478,29 +496,31 @@ void TurnServer::SendErrorResponseWithRealmAndNonce(
>     timestamp = ts_for_next_nonce_;
>     ts_for_next_nonce_ = 0;
>   }
>-  resp.AddAttribute(rtc::MakeUnique<StunByteStringAttribute>(
>+  resp.AddAttribute(absl::make_unique<StunByteStringAttribute>(
>       STUN_ATTR_NONCE, GenerateNonce(timestamp)));
>   resp.AddAttribute(
>-      rtc::MakeUnique<StunByteStringAttribute>(STUN_ATTR_REALM, realm_));
>+      absl::make_unique<StunByteStringAttribute>(STUN_ATTR_REALM, realm_));
>   SendStun(conn, &resp);
> }
> 
> void TurnServer::SendErrorResponseWithAlternateServer(
>     TurnServerConnection* conn, const StunMessage* msg,
>     const rtc::SocketAddress& addr) {
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>   TurnMessage resp;
>   InitErrorResponse(msg, STUN_ERROR_TRY_ALTERNATE,
>                     STUN_ERROR_REASON_TRY_ALTERNATE_SERVER, &resp);
>-  resp.AddAttribute(
>-      rtc::MakeUnique<StunAddressAttribute>(STUN_ATTR_ALTERNATE_SERVER, addr));
>+  resp.AddAttribute(absl::make_unique<StunAddressAttribute>(
>+      STUN_ATTR_ALTERNATE_SERVER, addr));
>   SendStun(conn, &resp);
> }
> 
> void TurnServer::SendStun(TurnServerConnection* conn, StunMessage* msg) {
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>   rtc::ByteBufferWriter buf;
>   // Add a SOFTWARE attribute if one is set.
>   if (!software_.empty()) {
>-    msg->AddAttribute(rtc::MakeUnique<StunByteStringAttribute>(
>+    msg->AddAttribute(absl::make_unique<StunByteStringAttribute>(
>         STUN_ATTR_SOFTWARE, software_));
>   }
>   msg->Write(&buf);
>@@ -509,11 +529,13 @@ void TurnServer::SendStun(TurnServerConnection* conn, StunMessage* msg) {
> 
> void TurnServer::Send(TurnServerConnection* conn,
>                       const rtc::ByteBufferWriter& buf) {
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>   rtc::PacketOptions options;
>   conn->socket()->SendTo(buf.Data(), buf.Length(), conn->src(), options);
> }
> 
> void TurnServer::OnAllocationDestroyed(TurnServerAllocation* allocation) {
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>   // Removing the internal socket if the connection is not udp.
>   rtc::AsyncPacketSocket* socket = allocation->conn()->socket();
>   InternalSocketMap::iterator iter = server_sockets_.find(socket);
>@@ -533,9 +555,11 @@ void TurnServer::OnAllocationDestroyed(TurnServerAllocation* allocation) {
> }
> 
> void TurnServer::DestroyInternalSocket(rtc::AsyncPacketSocket* socket) {
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>   InternalSocketMap::iterator iter = server_sockets_.find(socket);
>   if (iter != server_sockets_.end()) {
>     rtc::AsyncPacketSocket* socket = iter->first;
>+    socket->SignalReadPacket.disconnect(this);
>     server_sockets_.erase(iter);
>     // We must destroy the socket async to avoid invalidating the sigslot
>     // callback list iterator inside a sigslot callback. (In other words,
>@@ -548,6 +572,7 @@ void TurnServer::DestroyInternalSocket(rtc::AsyncPacketSocket* socket) {
> }
> 
> void TurnServer::FreeSockets() {
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>   sockets_to_delete_.clear();
> }
> 
>@@ -601,7 +626,7 @@ TurnServerAllocation::~TurnServerAllocation() {
>     delete *it;
>   }
>   thread_->Clear(this, MSG_ALLOCATION_TIMEOUT);
>-  LOG_J(LS_INFO, this) << "Allocation destroyed";
>+  RTC_LOG(LS_INFO) << ToString() << ": Allocation destroyed";
> }
> 
> std::string TurnServerAllocation::ToString() const {
>@@ -630,8 +655,9 @@ void TurnServerAllocation::HandleTurnMessage(const TurnMessage* msg) {
>       break;
>     default:
>       // Not sure what to do with this, just eat it.
>-      LOG_J(LS_WARNING, this) << "Invalid TURN message type received: "
>-                              << msg->type();
>+      RTC_LOG(LS_WARNING) << ToString()
>+                          << ": Invalid TURN message type received: "
>+                          << msg->type();
>   }
> }
> 
>@@ -653,18 +679,19 @@ void TurnServerAllocation::HandleAllocateRequest(const TurnMessage* msg) {
>   thread_->PostDelayed(RTC_FROM_HERE, lifetime_secs * 1000, this,
>                        MSG_ALLOCATION_TIMEOUT);
> 
>-  LOG_J(LS_INFO, this) << "Created allocation, lifetime=" << lifetime_secs;
>+  RTC_LOG(LS_INFO) << ToString()
>+                   << ": Created allocation with lifetime=" << lifetime_secs;
> 
>   // We've already validated all the important bits; just send a response here.
>   TurnMessage response;
>   InitResponse(msg, &response);
> 
>-  auto mapped_addr_attr = rtc::MakeUnique<StunXorAddressAttribute>(
>+  auto mapped_addr_attr = absl::make_unique<StunXorAddressAttribute>(
>       STUN_ATTR_XOR_MAPPED_ADDRESS, conn_.src());
>-  auto relayed_addr_attr = rtc::MakeUnique<StunXorAddressAttribute>(
>+  auto relayed_addr_attr = absl::make_unique<StunXorAddressAttribute>(
>       STUN_ATTR_XOR_RELAYED_ADDRESS, external_socket_->GetLocalAddress());
>   auto lifetime_attr =
>-      rtc::MakeUnique<StunUInt32Attribute>(STUN_ATTR_LIFETIME, lifetime_secs);
>+      absl::make_unique<StunUInt32Attribute>(STUN_ATTR_LIFETIME, lifetime_secs);
>   response.AddAttribute(std::move(mapped_addr_attr));
>   response.AddAttribute(std::move(relayed_addr_attr));
>   response.AddAttribute(std::move(lifetime_attr));
>@@ -681,14 +708,15 @@ void TurnServerAllocation::HandleRefreshRequest(const TurnMessage* msg) {
>   thread_->PostDelayed(RTC_FROM_HERE, lifetime_secs * 1000, this,
>                        MSG_ALLOCATION_TIMEOUT);
> 
>-  LOG_J(LS_INFO, this) << "Refreshed allocation, lifetime=" << lifetime_secs;
>+  RTC_LOG(LS_INFO) << ToString()
>+                   << ": Refreshed allocation, lifetime=" << lifetime_secs;
> 
>   // Send a success response with a LIFETIME attribute.
>   TurnMessage response;
>   InitResponse(msg, &response);
> 
>   auto lifetime_attr =
>-      rtc::MakeUnique<StunUInt32Attribute>(STUN_ATTR_LIFETIME, lifetime_secs);
>+      absl::make_unique<StunUInt32Attribute>(STUN_ATTR_LIFETIME, lifetime_secs);
>   response.AddAttribute(std::move(lifetime_attr));
> 
>   SendResponse(&response);
>@@ -700,7 +728,8 @@ void TurnServerAllocation::HandleSendIndication(const TurnMessage* msg) {
>   const StunAddressAttribute* peer_attr =
>       msg->GetAddress(STUN_ATTR_XOR_PEER_ADDRESS);
>   if (!data_attr || !peer_attr) {
>-    LOG_J(LS_WARNING, this) << "Received invalid send indication";
>+    RTC_LOG(LS_WARNING) << ToString()
>+                        << ": Received invalid send indication";
>     return;
>   }
> 
>@@ -709,8 +738,10 @@ void TurnServerAllocation::HandleSendIndication(const TurnMessage* msg) {
>     SendExternal(data_attr->bytes(), data_attr->length(),
>                  peer_attr->GetAddress());
>   } else {
>-    LOG_J(LS_WARNING, this) << "Received send indication without permission"
>-                            << "peer=" << peer_attr->GetAddress();
>+    RTC_LOG(LS_WARNING) << ToString()
>+                        << ": Received send indication without permission"
>+                           " peer="
>+                        << peer_attr->GetAddress().ToString();
>   }
> }
> 
>@@ -733,8 +764,9 @@ void TurnServerAllocation::HandleCreatePermissionRequest(
>   // Add this permission.
>   AddPermission(peer_attr->GetAddress().ipaddr());
> 
>-  LOG_J(LS_INFO, this) << "Created permission, peer="
>-                       << peer_attr->GetAddress();
>+  RTC_LOG(LS_INFO) << ToString()
>+                   << ": Created permission, peer="
>+                   << peer_attr->GetAddress().ToString();
> 
>   // Send a success response.
>   TurnMessage response;
>@@ -782,8 +814,9 @@ void TurnServerAllocation::HandleChannelBindRequest(const TurnMessage* msg) {
>   // Channel binds also refresh permissions.
>   AddPermission(peer_attr->GetAddress().ipaddr());
> 
>-  LOG_J(LS_INFO, this) << "Bound channel, id=" << channel_id
>-                       << ", peer=" << peer_attr->GetAddress();
>+  RTC_LOG(LS_INFO) << ToString()
>+                   << ": Bound channel, id=" << channel_id
>+                   << ", peer=" << peer_attr->GetAddress().ToString();
> 
>   // Send a success response.
>   TurnMessage response;
>@@ -800,8 +833,9 @@ void TurnServerAllocation::HandleChannelData(const char* data, size_t size) {
>     SendExternal(data + TURN_CHANNEL_HEADER_SIZE,
>                  size - TURN_CHANNEL_HEADER_SIZE, channel->peer());
>   } else {
>-    LOG_J(LS_WARNING, this) << "Received channel data for invalid channel, id="
>-                            << channel_id;
>+    RTC_LOG(LS_WARNING) << ToString()
>+                        << ": Received channel data for invalid channel, id="
>+                        << channel_id;
>   }
> }
> 
>@@ -826,14 +860,16 @@ void TurnServerAllocation::OnExternalPacket(
>     msg.SetType(TURN_DATA_INDICATION);
>     msg.SetTransactionID(
>         rtc::CreateRandomString(kStunTransactionIdLength));
>-    msg.AddAttribute(rtc::MakeUnique<StunXorAddressAttribute>(
>+    msg.AddAttribute(absl::make_unique<StunXorAddressAttribute>(
>         STUN_ATTR_XOR_PEER_ADDRESS, addr));
>     msg.AddAttribute(
>-        rtc::MakeUnique<StunByteStringAttribute>(STUN_ATTR_DATA, data, size));
>+        absl::make_unique<StunByteStringAttribute>(STUN_ATTR_DATA, data, size));
>     server_->SendStun(&conn_, &msg);
>   } else {
>-    LOG_J(LS_WARNING, this) << "Received external packet without permission, "
>-                            << "peer=" << addr;
>+    RTC_LOG(LS_WARNING)
>+        << ToString()
>+        << ": Received external packet without permission, peer="
>+        << addr.ToString();
>   }
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/turnserver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/turnserver.h
>index 3ead996c253..e7f886cf8fb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/turnserver.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/turnserver.h
>@@ -23,8 +23,9 @@
> #include "rtc_base/asyncinvoker.h"
> #include "rtc_base/asyncpacketsocket.h"
> #include "rtc_base/messagequeue.h"
>-#include "rtc_base/sigslot.h"
> #include "rtc_base/socketaddress.h"
>+#include "rtc_base/third_party/sigslot/sigslot.h"
>+#include "rtc_base/thread_checker.h"
> 
> namespace rtc {
> class ByteBufferWriter;
>@@ -178,30 +179,54 @@ class TurnServer : public sigslot::has_slots<> {
>   ~TurnServer() override;
> 
>   // Gets/sets the realm value to use for the server.
>-  const std::string& realm() const { return realm_; }
>-  void set_realm(const std::string& realm) { realm_ = realm; }
>+  const std::string& realm() const {
>+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+    return realm_;
>+  }
>+  void set_realm(const std::string& realm) {
>+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+    realm_ = realm;
>+  }
> 
>   // Gets/sets the value for the SOFTWARE attribute for TURN messages.
>-  const std::string& software() const { return software_; }
>-  void set_software(const std::string& software) { software_ = software; }
>+  const std::string& software() const {
>+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+    return software_;
>+  }
>+  void set_software(const std::string& software) {
>+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+    software_ = software;
>+  }
> 
>-  const AllocationMap& allocations() const { return allocations_; }
>+  const AllocationMap& allocations() const {
>+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+    return allocations_;
>+  }
> 
>   // Sets the authentication callback; does not take ownership.
>-  void set_auth_hook(TurnAuthInterface* auth_hook) { auth_hook_ = auth_hook; }
>+  void set_auth_hook(TurnAuthInterface* auth_hook) {
>+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+    auth_hook_ = auth_hook;
>+  }
> 
>   void set_redirect_hook(TurnRedirectInterface* redirect_hook) {
>+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
>     redirect_hook_ = redirect_hook;
>   }
> 
>-  void set_enable_otu_nonce(bool enable) { enable_otu_nonce_ = enable; }
>+  void set_enable_otu_nonce(bool enable) {
>+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+    enable_otu_nonce_ = enable;
>+  }
> 
>   // If set to true, reject CreatePermission requests to RFC1918 addresses.
>   void set_reject_private_addresses(bool filter) {
>+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
>     reject_private_addresses_ = filter;
>   }
> 
>   void set_enable_permission_checks(bool enable) {
>+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
>     enable_permission_checks_ = enable;
>   }
> 
>@@ -218,12 +243,14 @@ class TurnServer : public sigslot::has_slots<> {
>                                 const rtc::SocketAddress& address);
>   // For testing only.
>   std::string SetTimestampForNextNonce(int64_t timestamp) {
>+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
>     ts_for_next_nonce_ = timestamp;
>     return GenerateNonce(timestamp);
>   }
> 
>   void SetStunMessageObserver(
>       std::unique_ptr<StunMessageObserver> observer) {
>+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
>     stun_message_observer_ = std::move(observer);
>   }
> 
>@@ -282,6 +309,7 @@ class TurnServer : public sigslot::has_slots<> {
>                    ProtocolType> ServerSocketMap;
> 
>   rtc::Thread* thread_;
>+  rtc::ThreadChecker thread_checker_;
>   std::string nonce_key_;
>   std::string realm_;
>   std::string software_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/udptransport.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/udptransport.cc
>index 8b4ce006894..c21e5b8769e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/udptransport.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/udptransport.cc
>@@ -96,11 +96,11 @@ int UdpTransport::SendPacket(const char* data,
>   return result;
> }
> 
>-rtc::Optional<rtc::NetworkRoute> UdpTransport::network_route() const {
>+absl::optional<rtc::NetworkRoute> UdpTransport::network_route() const {
>   rtc::NetworkRoute network_route;
>   network_route.packet_overhead =
>       /*kUdpOverhead=*/8 + GetIpOverhead(GetLocalAddress().family());
>-  return rtc::Optional<rtc::NetworkRoute>(network_route);
>+  return absl::optional<rtc::NetworkRoute>(network_route);
> }
> 
> int UdpTransport::SetOption(rtc::Socket::Option opt, int value) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/udptransport.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/udptransport.h
>index 118b596e91f..06a795f2e1f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/udptransport.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/udptransport.h
>@@ -14,7 +14,7 @@
> #include <memory>
> #include <string>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> #include "api/ortc/udptransportinterface.h"
> #include "p2p/base/packettransportinternal.h"
> #include "rtc_base/asyncpacketsocket.h"  // For PacketOptions.
>@@ -25,7 +25,7 @@ class AsyncPacketSocket;
> struct PacketTime;
> struct SentPacket;
> class SocketAddress;
>-}
>+}  // namespace rtc
> 
> namespace cricket {
> 
>@@ -61,7 +61,7 @@ class UdpTransport : public rtc::PacketTransportInternal,
> 
>   int GetError() override;
> 
>-  rtc::Optional<rtc::NetworkRoute> network_route() const override;
>+  absl::optional<rtc::NetworkRoute> network_route() const override;
> 
>  protected:
>   PacketTransportInternal* GetInternal() override;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/client/basicportallocator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/client/basicportallocator.cc
>index 5f27a7db996..ec9abb73ddd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/client/basicportallocator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/client/basicportallocator.cc
>@@ -11,13 +11,12 @@
> #include "p2p/client/basicportallocator.h"
> 
> #include <algorithm>
>+#include <functional>
> #include <set>
> #include <string>
> #include <vector>
> 
>-#include "api/umametrics.h"
> #include "p2p/base/basicpacketsocketfactory.h"
>-#include "p2p/base/common.h"
> #include "p2p/base/port.h"
> #include "p2p/base/relayport.h"
> #include "p2p/base/stunport.h"
>@@ -26,7 +25,9 @@
> #include "p2p/base/udpport.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/helpers.h"
>+#include "rtc_base/ipaddress.h"
> #include "rtc_base/logging.h"
>+#include "system_wrappers/include/metrics.h"
> 
> using rtc::CreateRandomId;
> 
>@@ -39,6 +40,7 @@ enum {
>   MSG_ALLOCATION_PHASE,
>   MSG_SEQUENCEOBJECTS_CREATED,
>   MSG_CONFIG_STOP,
>+  MSG_SIGNAL_ANY_ADDRESS_PORTS,
> };
> 
> const int PHASE_UDP = 0;
>@@ -89,6 +91,32 @@ int ComparePort(const cricket::Port* a, const cricket::Port* b) {
>   return a_family - b_family;
> }
> 
>+struct NetworkFilter {
>+  using Predicate = std::function<bool(rtc::Network*)>;
>+  NetworkFilter(Predicate pred, const std::string& description)
>+      : pred(pred), description(description) {}
>+  Predicate pred;
>+  const std::string description;
>+};
>+
>+using NetworkList = rtc::NetworkManager::NetworkList;
>+void FilterNetworks(NetworkList* networks, NetworkFilter filter) {
>+  auto start_to_remove =
>+      std::remove_if(networks->begin(), networks->end(), filter.pred);
>+  if (start_to_remove == networks->end()) {
>+    return;
>+  }
>+  RTC_LOG(INFO) << "Filtered out " << filter.description << " networks:";
>+  for (auto it = start_to_remove; it != networks->end(); ++it) {
>+    RTC_LOG(INFO) << (*it)->ToString();
>+  }
>+  networks->erase(start_to_remove, networks->end());
>+}
>+
>+bool IsAnyAddressPort(const cricket::Port* port) {
>+  return rtc::IPIsAny(port->Network()->GetBestIP());
>+}
>+
> }  // namespace
> 
> namespace cricket {
>@@ -107,13 +135,12 @@ BasicPortAllocator::BasicPortAllocator(
>   RTC_DCHECK(relay_port_factory_ != nullptr);
>   RTC_DCHECK(network_manager_ != nullptr);
>   RTC_DCHECK(socket_factory_ != nullptr);
>-  SetConfiguration(ServerAddresses(), std::vector<RelayServerConfig>(),
>-                   0, false, customizer);
>+  SetConfiguration(ServerAddresses(), std::vector<RelayServerConfig>(), 0,
>+                   false, customizer);
>   Construct();
> }
> 
>-BasicPortAllocator::BasicPortAllocator(
>-    rtc::NetworkManager* network_manager)
>+BasicPortAllocator::BasicPortAllocator(rtc::NetworkManager* network_manager)
>     : network_manager_(network_manager), socket_factory_(nullptr) {
>   InitRelayPortFactory(nullptr);
>   RTC_DCHECK(relay_port_factory_ != nullptr);
>@@ -121,14 +148,13 @@ BasicPortAllocator::BasicPortAllocator(
>   Construct();
> }
> 
>-BasicPortAllocator::BasicPortAllocator(
>-    rtc::NetworkManager* network_manager,
>-    rtc::PacketSocketFactory* socket_factory,
>-    const ServerAddresses& stun_servers)
>+BasicPortAllocator::BasicPortAllocator(rtc::NetworkManager* network_manager,
>+                                       rtc::PacketSocketFactory* socket_factory,
>+                                       const ServerAddresses& stun_servers)
>     : network_manager_(network_manager), socket_factory_(socket_factory) {
>   InitRelayPortFactory(nullptr);
>   RTC_DCHECK(relay_port_factory_ != nullptr);
>-  RTC_DCHECK(socket_factory_ != NULL);
>+  RTC_DCHECK(socket_factory_ != nullptr);
>   SetConfiguration(stun_servers, std::vector<RelayServerConfig>(), 0, false,
>                    nullptr);
>   Construct();
>@@ -140,7 +166,7 @@ BasicPortAllocator::BasicPortAllocator(
>     const rtc::SocketAddress& relay_address_udp,
>     const rtc::SocketAddress& relay_address_tcp,
>     const rtc::SocketAddress& relay_address_ssl)
>-    : network_manager_(network_manager), socket_factory_(NULL) {
>+    : network_manager_(network_manager), socket_factory_(nullptr) {
>   InitRelayPortFactory(nullptr);
>   RTC_DCHECK(relay_port_factory_ != nullptr);
>   RTC_DCHECK(network_manager_ != nullptr);
>@@ -170,9 +196,6 @@ void BasicPortAllocator::Construct() {
> 
> void BasicPortAllocator::OnIceRegathering(PortAllocatorSession* session,
>                                           IceRegatheringReason reason) {
>-  if (!metrics_observer()) {
>-    return;
>-  }
>   // If the session has not been taken by an active channel, do not report the
>   // metric.
>   for (auto& allocator_session : pooled_sessions()) {
>@@ -181,12 +204,13 @@ void BasicPortAllocator::OnIceRegathering(PortAllocatorSession* session,
>     }
>   }
> 
>-  metrics_observer()->IncrementEnumCounter(
>-      webrtc::kEnumCounterIceRegathering, static_cast<int>(reason),
>-      static_cast<int>(IceRegatheringReason::MAX_VALUE));
>+  RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.IceRegatheringReason",
>+                            static_cast<int>(reason),
>+                            static_cast<int>(IceRegatheringReason::MAX_VALUE));
> }
> 
> BasicPortAllocator::~BasicPortAllocator() {
>+  CheckRunOnValidThreadIfInitialized();
>   // Our created port allocator sessions depend on us, so destroy our remaining
>   // pooled sessions before anything else.
>   DiscardCandidatePool();
>@@ -196,12 +220,16 @@ void BasicPortAllocator::SetNetworkIgnoreMask(int network_ignore_mask) {
>   // TODO(phoglund): implement support for other types than loopback.
>   // See https://code.google.com/p/webrtc/issues/detail?id=4288.
>   // Then remove set_network_ignore_list from NetworkManager.
>+  CheckRunOnValidThreadIfInitialized();
>   network_ignore_mask_ = network_ignore_mask;
> }
> 
> PortAllocatorSession* BasicPortAllocator::CreateSessionInternal(
>-    const std::string& content_name, int component,
>-    const std::string& ice_ufrag, const std::string& ice_pwd) {
>+    const std::string& content_name,
>+    int component,
>+    const std::string& ice_ufrag,
>+    const std::string& ice_pwd) {
>+  CheckRunOnValidThreadAndInitialized();
>   PortAllocatorSession* session = new BasicPortAllocatorSession(
>       this, content_name, component, ice_ufrag, ice_pwd);
>   session->SignalIceRegathering.connect(this,
>@@ -210,6 +238,7 @@ PortAllocatorSession* BasicPortAllocator::CreateSessionInternal(
> }
> 
> void BasicPortAllocator::AddTurnServer(const RelayServerConfig& turn_server) {
>+  CheckRunOnValidThreadAndInitialized();
>   std::vector<RelayServerConfig> new_turn_servers = turn_servers();
>   new_turn_servers.push_back(turn_server);
>   SetConfiguration(stun_servers(), new_turn_servers, candidate_pool_size(),
>@@ -239,7 +268,7 @@ BasicPortAllocatorSession::BasicPortAllocatorSession(
>                            ice_pwd,
>                            allocator->flags()),
>       allocator_(allocator),
>-      network_thread_(NULL),
>+      network_thread_(nullptr),
>       socket_factory_(allocator->socket_factory()),
>       allocation_started_(false),
>       network_manager_started_(false),
>@@ -252,7 +281,7 @@ BasicPortAllocatorSession::BasicPortAllocatorSession(
> 
> BasicPortAllocatorSession::~BasicPortAllocatorSession() {
>   allocator_->network_manager()->StopUpdating();
>-  if (network_thread_ != NULL)
>+  if (network_thread_ != nullptr)
>     network_thread_->Clear(this);
> 
>   for (uint32_t i = 0; i < sequences_.size(); ++i) {
>@@ -419,7 +448,7 @@ void BasicPortAllocatorSession::Regather(
>   std::vector<PortData*> ports_to_prune = GetUnprunedPorts(networks);
>   if (!ports_to_prune.empty()) {
>     RTC_LOG(LS_INFO) << "Prune " << ports_to_prune.size() << " ports";
>-    PrunePortsAndRemoveCandidates(ports_to_prune);
>+    PrunePortsAndSignalCandidatesRemoval(ports_to_prune);
>   }
> 
>   if (allocation_started_ && network_manager_started_ && !IsStopped()) {
>@@ -429,6 +458,21 @@ void BasicPortAllocatorSession::Regather(
>   }
> }
> 
>+void BasicPortAllocatorSession::SetStunKeepaliveIntervalForReadyPorts(
>+    const absl::optional<int>& stun_keepalive_interval) {
>+  auto ports = ReadyPorts();
>+  for (PortInterface* port : ports) {
>+    // The port type and protocol can be used to identify different subclasses
>+    // of Port in the current implementation. Note that a TCPPort has the type
>+    // LOCAL_PORT_TYPE but uses the protocol PROTO_TCP.
>+    if (port->Type() == STUN_PORT_TYPE ||
>+        (port->Type() == LOCAL_PORT_TYPE && port->GetProtocol() == PROTO_UDP)) {
>+      static_cast<UDPPort*>(port)->set_stun_keepalive_delay(
>+          stun_keepalive_interval);
>+    }
>+  }
>+}
>+
> std::vector<PortInterface*> BasicPortAllocatorSession::ReadyPorts() const {
>   std::vector<PortInterface*> ret;
>   for (const PortData& data : ports_) {
>@@ -504,30 +548,34 @@ bool BasicPortAllocatorSession::CandidatesAllocationDone() const {
>                       [](const PortData& port) { return port.inprogress(); });
> }
> 
>-void BasicPortAllocatorSession::OnMessage(rtc::Message *message) {
>+void BasicPortAllocatorSession::OnMessage(rtc::Message* message) {
>   switch (message->message_id) {
>-  case MSG_CONFIG_START:
>-    RTC_DCHECK(rtc::Thread::Current() == network_thread_);
>-    GetPortConfigurations();
>-    break;
>-  case MSG_CONFIG_READY:
>-    RTC_DCHECK(rtc::Thread::Current() == network_thread_);
>-    OnConfigReady(static_cast<PortConfiguration*>(message->pdata));
>-    break;
>-  case MSG_ALLOCATE:
>-    RTC_DCHECK(rtc::Thread::Current() == network_thread_);
>-    OnAllocate();
>-    break;
>-  case MSG_SEQUENCEOBJECTS_CREATED:
>-    RTC_DCHECK(rtc::Thread::Current() == network_thread_);
>-    OnAllocationSequenceObjectsCreated();
>-    break;
>-  case MSG_CONFIG_STOP:
>-    RTC_DCHECK(rtc::Thread::Current() == network_thread_);
>-    OnConfigStop();
>-    break;
>-  default:
>-    RTC_NOTREACHED();
>+    case MSG_CONFIG_START:
>+      RTC_DCHECK(rtc::Thread::Current() == network_thread_);
>+      GetPortConfigurations();
>+      break;
>+    case MSG_CONFIG_READY:
>+      RTC_DCHECK(rtc::Thread::Current() == network_thread_);
>+      OnConfigReady(static_cast<PortConfiguration*>(message->pdata));
>+      break;
>+    case MSG_ALLOCATE:
>+      RTC_DCHECK(rtc::Thread::Current() == network_thread_);
>+      OnAllocate();
>+      break;
>+    case MSG_SEQUENCEOBJECTS_CREATED:
>+      RTC_DCHECK(rtc::Thread::Current() == network_thread_);
>+      OnAllocationSequenceObjectsCreated();
>+      break;
>+    case MSG_CONFIG_STOP:
>+      RTC_DCHECK(rtc::Thread::Current() == network_thread_);
>+      OnConfigStop();
>+      break;
>+    case MSG_SIGNAL_ANY_ADDRESS_PORTS:
>+      RTC_DCHECK(rtc::Thread::Current() == network_thread_);
>+      SignalAnyAddressPortsAndCandidatesReadyIfNotRedundant();
>+      break;
>+    default:
>+      RTC_NOTREACHED();
>   }
> }
> 
>@@ -539,9 +587,8 @@ void BasicPortAllocatorSession::UpdateIceParametersInternal() {
> }
> 
> void BasicPortAllocatorSession::GetPortConfigurations() {
>-  PortConfiguration* config = new PortConfiguration(allocator_->stun_servers(),
>-                                                    username(),
>-                                                    password());
>+  PortConfiguration* config =
>+      new PortConfiguration(allocator_->stun_servers(), username(), password());
> 
>   for (const RelayServerConfig& turn_server : allocator_->turn_servers()) {
>     config->AddRelay(turn_server);
>@@ -569,8 +616,8 @@ void BasicPortAllocatorSession::OnConfigStop() {
>   // mark those as error. Since session doesn't need any new candidates
>   // at this stage of the allocation, it's safe to discard any new candidates.
>   bool send_signal = false;
>-  for (std::vector<PortData>::iterator it = ports_.begin();
>-       it != ports_.end(); ++it) {
>+  for (std::vector<PortData>::iterator it = ports_.begin(); it != ports_.end();
>+       ++it) {
>     if (it->inprogress()) {
>       // Updating port state to error, which didn't finish allocating candidates
>       // yet.
>@@ -589,7 +636,7 @@ void BasicPortAllocatorSession::OnConfigStop() {
> 
>   // If we stopped anything that was running, send a done signal now.
>   if (send_signal) {
>-    MaybeSignalCandidatesAllocationDone();
>+    FireAllocationStatusSignalsIfNeeded();
>   }
> }
> 
>@@ -617,41 +664,58 @@ std::vector<rtc::Network*> BasicPortAllocatorSession::GetNetworks() {
>       rtc::NetworkManager::ENUMERATION_BLOCKED) {
>     set_flags(flags() | PORTALLOCATOR_DISABLE_ADAPTER_ENUMERATION);
>   }
>-  // If the adapter enumeration is disabled, we'll just bind to any address
>-  // instead of specific NIC. This is to ensure the same routing for http
>-  // traffic by OS is also used here to avoid any local or public IP leakage
>-  // during stun process.
>-  if (flags() & PORTALLOCATOR_DISABLE_ADAPTER_ENUMERATION) {
>-    network_manager->GetAnyAddressNetworks(&networks);
>-  } else {
>+
>+  // If adapter enumeration is disabled, we'll just bind to any address
>+  // instead of a specific NIC. This is to ensure that WebRTC traffic is routed
>+  // by the OS in the same way that HTTP traffic would be, and no additional
>+  // local or public IPs are leaked during ICE processing.
>+  //
>+  // Even when adapter enumeration is enabled, we still bind to the "any"
>+  // address as a fallback, since this may potentially reveal network
>+  // interfaces that weren't otherwise accessible. Note that the candidates
>+  // gathered by binding to the "any" address won't be surfaced to the
>+  // application if they're determined to be redundant (if they have the same
>+  // address as a candidate gathered by binding to an interface explicitly).
>+  if (!(flags() & PORTALLOCATOR_DISABLE_ADAPTER_ENUMERATION)) {
>     network_manager->GetNetworks(&networks);
>-    // If network enumeration fails, use the ANY address as a fallback, so we
>-    // can at least try gathering candidates using the default route chosen by
>-    // the OS. Or, if the PORTALLOCATOR_ENABLE_ANY_ADDRESS_PORTS flag is
>-    // set, we'll use ANY address candidates either way.
>-    if (networks.empty() || flags() & PORTALLOCATOR_ENABLE_ANY_ADDRESS_PORTS) {
>-      network_manager->GetAnyAddressNetworks(&networks);
>-    }
>+  }
>+
>+  network_manager->GetAnyAddressNetworks(&networks);
>+
>+  // Filter out link-local networks if needed.
>+  if (flags() & PORTALLOCATOR_DISABLE_LINK_LOCAL_NETWORKS) {
>+    NetworkFilter link_local_filter(
>+        [](rtc::Network* network) { return IPIsLinkLocal(network->prefix()); },
>+        "link-local");
>+    FilterNetworks(&networks, link_local_filter);
>   }
>   // Do some more filtering, depending on the network ignore mask and "disable
>   // costly networks" flag.
>-  networks.erase(std::remove_if(networks.begin(), networks.end(),
>-                                [this](rtc::Network* network) {
>-                                  return allocator_->network_ignore_mask() &
>-                                         network->type();
>-                                }),
>-                 networks.end());
>+  NetworkFilter ignored_filter(
>+      [this](rtc::Network* network) {
>+        return allocator_->network_ignore_mask() & network->type();
>+      },
>+      "ignored");
>+  FilterNetworks(&networks, ignored_filter);
>   if (flags() & PORTALLOCATOR_DISABLE_COSTLY_NETWORKS) {
>     uint16_t lowest_cost = rtc::kNetworkCostMax;
>     for (rtc::Network* network : networks) {
>+      // Don't determine the lowest cost from a link-local or any address
>+      // network. On iOS, a device connected to the computer will get a
>+      // link-local network for communicating with the computer, however this
>+      // network can't be used to connect to a peer outside the network.
>+      if (rtc::IPIsLinkLocal(network->GetBestIP()) ||
>+          rtc::IPIsAny(network->GetBestIP())) {
>+        continue;
>+      }
>       lowest_cost = std::min<uint16_t>(lowest_cost, network->GetCost());
>     }
>-    networks.erase(std::remove_if(networks.begin(), networks.end(),
>-                                  [lowest_cost](rtc::Network* network) {
>-                                    return network->GetCost() >
>-                                           lowest_cost + rtc::kNetworkCostLow;
>-                                  }),
>-                   networks.end());
>+    NetworkFilter costly_filter(
>+        [lowest_cost](rtc::Network* network) {
>+          return network->GetCost() > lowest_cost + rtc::kNetworkCostLow;
>+        },
>+        "costly");
>+    FilterNetworks(&networks, costly_filter);
>   }
>   // Lastly, if we have a limit for the number of IPv6 network interfaces (by
>   // default, it's 5), remove networks to ensure that limit is satisfied.
>@@ -739,6 +803,19 @@ void BasicPortAllocatorSession::DoAllocate(bool disable_equivalent) {
>   if (done_signal_needed) {
>     network_thread_->Post(RTC_FROM_HERE, this, MSG_SEQUENCEOBJECTS_CREATED);
>   }
>+
>+  // If adapter enumeration is enabled, then we prefer binding to individual
>+  // network adapters, only using ports bound to the "any" address (0.0.0.0) if
>+  // they reveal an interface not otherwise accessible. Normally these will be
>+  // surfaced when candidate allocation completes, but sometimes candidate
>+  // allocation can take a long time, if a STUN transaction times out for
>+  // instance. So as a backup, we'll surface these ports/candidates after
>+  // |kMaxWaitMsBeforeSignalingAnyAddressPortsAndCandidates| passes.
>+  if (!(flags() & PORTALLOCATOR_DISABLE_ADAPTER_ENUMERATION)) {
>+    network_thread_->PostDelayed(
>+        RTC_FROM_HERE, kMaxWaitMsBeforeSignalingAnyAddressPortsAndCandidates,
>+        this, MSG_SIGNAL_ANY_ADDRESS_PORTS);
>+  }
> }
> 
> void BasicPortAllocatorSession::OnNetworksChanged() {
>@@ -758,7 +835,7 @@ void BasicPortAllocatorSession::OnNetworksChanged() {
>   if (!ports_to_prune.empty()) {
>     RTC_LOG(LS_INFO) << "Prune " << ports_to_prune.size()
>                      << " ports because their networks were gone";
>-    PrunePortsAndRemoveCandidates(ports_to_prune);
>+    PrunePortsAndSignalCandidatesRemoval(ports_to_prune);
>   }
> 
>   if (allocation_started_ && !IsStopped()) {
>@@ -781,14 +858,14 @@ void BasicPortAllocatorSession::DisableEquivalentPhases(
>     PortConfiguration* config,
>     uint32_t* flags) {
>   for (uint32_t i = 0; i < sequences_.size() &&
>-                           (*flags & DISABLE_ALL_PHASES) != DISABLE_ALL_PHASES;
>+                       (*flags & DISABLE_ALL_PHASES) != DISABLE_ALL_PHASES;
>        ++i) {
>     sequences_[i]->DisableEquivalentPhases(network, config, flags);
>   }
> }
> 
> void BasicPortAllocatorSession::AddAllocatedPort(Port* port,
>-                                                 AllocationSequence * seq,
>+                                                 AllocationSequence* seq,
>                                                  bool prepare_address) {
>   if (!port)
>     return;
>@@ -808,12 +885,11 @@ void BasicPortAllocatorSession::AddAllocatedPort(Port* port,
>   port->SignalCandidateReady.connect(
>       this, &BasicPortAllocatorSession::OnCandidateReady);
>   port->SignalPortComplete.connect(this,
>-      &BasicPortAllocatorSession::OnPortComplete);
>+                                   &BasicPortAllocatorSession::OnPortComplete);
>   port->SignalDestroyed.connect(this,
>-      &BasicPortAllocatorSession::OnPortDestroyed);
>-  port->SignalPortError.connect(
>-      this, &BasicPortAllocatorSession::OnPortError);
>-  LOG_J(LS_INFO, port) << "Added port to allocator";
>+                                &BasicPortAllocatorSession::OnPortDestroyed);
>+  port->SignalPortError.connect(this, &BasicPortAllocatorSession::OnPortError);
>+  RTC_LOG(LS_INFO) << port->ToString() << ": Added port to allocator";
> 
>   if (prepare_address)
>     port->PrepareAddress();
>@@ -822,15 +898,16 @@ void BasicPortAllocatorSession::AddAllocatedPort(Port* port,
> void BasicPortAllocatorSession::OnAllocationSequenceObjectsCreated() {
>   allocation_sequences_created_ = true;
>   // Send candidate allocation complete signal if we have no sequences.
>-  MaybeSignalCandidatesAllocationDone();
>+  FireAllocationStatusSignalsIfNeeded();
> }
> 
>-void BasicPortAllocatorSession::OnCandidateReady(
>-    Port* port, const Candidate& c) {
>+void BasicPortAllocatorSession::OnCandidateReady(Port* port,
>+                                                 const Candidate& c) {
>   RTC_DCHECK(rtc::Thread::Current() == network_thread_);
>   PortData* data = FindPort(port);
>-  RTC_DCHECK(data != NULL);
>-  LOG_J(LS_INFO, port) << "Gathered candidate: " << c.ToSensitiveString();
>+  RTC_DCHECK(data != nullptr);
>+  RTC_LOG(LS_INFO) << port->ToString()
>+                   << ": Gathered candidate: " << c.ToSensitiveString();
>   // Discarding any candidate signal if port allocation status is
>   // already done with gathering.
>   if (!data->inprogress()) {
>@@ -857,23 +934,56 @@ void BasicPortAllocatorSession::OnCandidateReady(
>     }
>     // If the current port is not pruned yet, SignalPortReady.
>     if (!data->pruned()) {
>-      LOG_J(LS_INFO, port) << "Port ready.";
>-      SignalPortReady(this, port);
>       port->KeepAliveUntilPruned();
>+      // We postpone the signaling of any address ports to when the candidates
>+      // allocation is done or the candidate allocation process has start for
>+      // more than kMaxWaitMsBeforeSignalingAnyAddressPortsAndCandidates, and
>+      // we check whether they are redundant or not (in
>+      // SignalAnyAddressPortsAndCandidatesReadyIfNotRedundant). Otherwise,
>+      // connectivity checks will be sent from these possibly redundant ports,
>+      // likely also resulting in "prflx" candidate pairs being created on the
>+      // other side if not pruned in time. The signaling of any address ports
>+      // that are not redundant happens in
>+      // SignalAnyAddressPortsAndCandidatesReadyIfNotRedundant.
>+      //
>+      // If adapter enumeration is disabled, these "any" address ports
>+      // are all we'll get, so we can signal them immediately.
>+      //
>+      // Same logic applies to candidates below.
>+
>+      if (!IsAnyAddressPort(port) ||
>+          (flags() & PORTALLOCATOR_DISABLE_ADAPTER_ENUMERATION)) {
>+        RTC_LOG(INFO) << port->ToString() << ": Port ready.";
>+        SignalPortReady(this, port);
>+        data->set_signaled();
>+      }
>     }
>   }
> 
>   if (data->ready() && CheckCandidateFilter(c)) {
>-    std::vector<Candidate> candidates;
>-    candidates.push_back(SanitizeRelatedAddress(c));
>-    SignalCandidatesReady(this, candidates);
>+    // See comment above about why we delay signaling candidates from "any
>+    // address" ports.
>+    //
>+    // For candidates gathered after the any address port is signaled, we will
>+    // not perform the redundancy check anymore. Note that late candiates
>+    // gathered from the any address port should be a srflx candidate from a
>+    // late STUN binding response.
>+    if (data->signaled()) {
>+      std::vector<Candidate> candidates;
>+      candidates.push_back(SanitizeRelatedAddress(c));
>+      SignalCandidatesReady(this, candidates);
>+    } else {
>+      RTC_LOG(INFO) << "Candidate not signaled yet because it is from the "
>+                       "any address port: "
>+                    << c.ToSensitiveString();
>+    }
>   } else {
>     RTC_LOG(LS_INFO) << "Discarding candidate because it doesn't match filter.";
>   }
> 
>   // If we have pruned any port, maybe need to signal port allocation done.
>   if (pruned) {
>-    MaybeSignalCandidatesAllocationDone();
>+    FireAllocationStatusSignalsIfNeeded();
>   }
> }
> 
>@@ -907,7 +1017,7 @@ bool BasicPortAllocatorSession::PruneTurnPorts(Port* newly_pairable_turn_port) {
>         ComparePort(data.port(), best_turn_port) < 0) {
>       pruned = true;
>       if (data.port() != newly_pairable_turn_port) {
>-        // These ports will be pruned in PrunePortsAndRemoveCandidates.
>+        // These ports will be pruned in PrunePortsAndSignalCandidatesRemoval.
>         ports_to_prune.push_back(&data);
>       } else {
>         data.Prune();
>@@ -918,7 +1028,7 @@ bool BasicPortAllocatorSession::PruneTurnPorts(Port* newly_pairable_turn_port) {
>   if (!ports_to_prune.empty()) {
>     RTC_LOG(LS_INFO) << "Prune " << ports_to_prune.size()
>                      << " low-priority TURN ports";
>-    PrunePortsAndRemoveCandidates(ports_to_prune);
>+    PrunePortsAndSignalCandidatesRemoval(ports_to_prune);
>   }
>   return pruned;
> }
>@@ -931,9 +1041,10 @@ void BasicPortAllocatorSession::PruneAllPorts() {
> 
> void BasicPortAllocatorSession::OnPortComplete(Port* port) {
>   RTC_DCHECK(rtc::Thread::Current() == network_thread_);
>-  LOG_J(LS_INFO, port) << "Port completed gathering candidates.";
>+  RTC_LOG(LS_INFO) << port->ToString()
>+                   << ": Port completed gathering candidates.";
>   PortData* data = FindPort(port);
>-  RTC_DCHECK(data != NULL);
>+  RTC_DCHECK(data != nullptr);
> 
>   // Ignore any late signals.
>   if (!data->inprogress()) {
>@@ -943,14 +1054,15 @@ void BasicPortAllocatorSession::OnPortComplete(Port* port) {
>   // Moving to COMPLETE state.
>   data->set_complete();
>   // Send candidate allocation complete signal if this was the last port.
>-  MaybeSignalCandidatesAllocationDone();
>+  FireAllocationStatusSignalsIfNeeded();
> }
> 
> void BasicPortAllocatorSession::OnPortError(Port* port) {
>   RTC_DCHECK(rtc::Thread::Current() == network_thread_);
>-  LOG_J(LS_INFO, port) << "Port encountered error while gathering candidates.";
>+  RTC_LOG(LS_INFO) << port->ToString()
>+                   << ": Port encountered error while gathering candidates.";
>   PortData* data = FindPort(port);
>-  RTC_DCHECK(data != NULL);
>+  RTC_DCHECK(data != nullptr);
>   // We might have already given up on this port and stopped it.
>   if (!data->inprogress()) {
>     return;
>@@ -960,7 +1072,7 @@ void BasicPortAllocatorSession::OnPortError(Port* port) {
>   // But this signal itself is generic.
>   data->set_error();
>   // Send candidate allocation complete signal if this was the last port.
>-  MaybeSignalCandidatesAllocationDone();
>+  FireAllocationStatusSignalsIfNeeded();
> }
> 
> bool BasicPortAllocatorSession::CheckCandidateFilter(const Candidate& c) const {
>@@ -1004,24 +1116,34 @@ bool BasicPortAllocatorSession::CandidatePairable(const Candidate& c,
>   // prevent even default IP addresses from leaking), we still don't want to
>   // ping from them, even if device enumeration is disabled.  Thus, we check for
>   // both device enumeration and host candidates being disabled.
>-  bool network_enumeration_disabled = c.address().IsAnyIP();
>+  bool candidate_has_any_address = c.address().IsAnyIP();
>   bool can_ping_from_candidate =
>       (port->SharedSocket() || c.protocol() == TCP_PROTOCOL_NAME);
>   bool host_candidates_disabled = !(candidate_filter_ & CF_HOST);
> 
>   return candidate_signalable ||
>-         (network_enumeration_disabled && can_ping_from_candidate &&
>+         (candidate_has_any_address && can_ping_from_candidate &&
>           !host_candidates_disabled);
> }
> 
> void BasicPortAllocatorSession::OnPortAllocationComplete(
>     AllocationSequence* seq) {
>   // Send candidate allocation complete signal if all ports are done.
>-  MaybeSignalCandidatesAllocationDone();
>+  FireAllocationStatusSignalsIfNeeded();
> }
> 
>-void BasicPortAllocatorSession::MaybeSignalCandidatesAllocationDone() {
>+void BasicPortAllocatorSession::FireAllocationStatusSignalsIfNeeded() {
>   if (CandidatesAllocationDone()) {
>+    // Now that allocation is done, we can surface any ports bound to the "any"
>+    // address if they're not redundant (if they don't have the same address as
>+    // a port bound to a specific interface). We don't surface them as soon as
>+    // they're gathered because we may not know yet whether they're redundant.
>+    //
>+    // This also happens after a timeout of 2 seconds (see comment in
>+    // DoAllocate); if allocation completes first we clear that timer since
>+    // it's not needed.
>+    network_thread_->Clear(this, MSG_SIGNAL_ANY_ADDRESS_PORTS);
>+    SignalAnyAddressPortsAndCandidatesReadyIfNotRedundant();
>     if (pooled()) {
>       RTC_LOG(LS_INFO) << "All candidates gathered for pooled session.";
>     } else {
>@@ -1032,6 +1154,100 @@ void BasicPortAllocatorSession::MaybeSignalCandidatesAllocationDone() {
>   }
> }
> 
>+// We detect the redundancy in any address ports as follows:
>+//
>+// 1. Delay the signaling of all any address ports and candidates gathered from
>+// these ports, which happens in OnCandidateReady.
>+//
>+// 2. For all non-any address ports, collect the IPs of their candidates
>+// (ignoring "active" TCP candidates, since no sockets are created for them
>+// until a connection is made and there's no guarantee they'll work).
>+//
>+// 3. For each any address port, compare their candidates to the existing IPs
>+// collected from step 2, and this port can be signaled if it has candidates
>+// with unseen IPs.
>+void BasicPortAllocatorSession::
>+    SignalAnyAddressPortsAndCandidatesReadyIfNotRedundant() {
>+  // Note that this is called either when allocation completes, or after a
>+  // timeout, so some ports may still be waiting for STUN transactions to
>+  // finish.
>+  //
>+  // First, get a list of all "any address" ports that have not yet been
>+  // signaled, and a list of candidate IP addresses from all other ports.
>+  std::vector<PortData*> maybe_signalable_any_address_ports;
>+  std::set<rtc::IPAddress> ips_from_non_any_address_ports;
>+  for (PortData& port_data : ports_) {
>+    if (!port_data.ready()) {
>+      continue;
>+    }
>+    if (IsAnyAddressPort(port_data.port())) {
>+      if (!port_data.signaled()) {
>+        maybe_signalable_any_address_ports.push_back(&port_data);
>+      }
>+    } else {
>+      for (const Candidate& c : port_data.port()->Candidates()) {
>+        // If the port of the candidate is |DISCARD_PORT| (9), this is an
>+        // "active" TCP candidate and it doesn't mean we actually bound a
>+        // socket to this address, so ignore it.
>+        if (c.address().port() != DISCARD_PORT) {
>+          ips_from_non_any_address_ports.insert(c.address().ipaddr());
>+          if (port_data.port()->Type() == RELAY_PORT_TYPE) {
>+            // The related address of a relay candidate is the server reflexive
>+            // address obtained from the TURN allocation response.
>+            ips_from_non_any_address_ports.insert(c.related_address().ipaddr());
>+          }
>+        }
>+      }
>+    }
>+  }
>+  // Now signal "any" address ports that have a unique address, and prune any
>+  // that don't.
>+  std::vector<PortData*> signalable_any_address_ports;
>+  std::vector<PortData*> prunable_any_address_ports;
>+  std::vector<Candidate> signalable_candidates_from_any_address_ports;
>+  for (PortData* port_data : maybe_signalable_any_address_ports) {
>+    bool port_signalable = false;
>+    for (const Candidate& c : port_data->port()->Candidates()) {
>+      if (!CandidatePairable(c, port_data->port()) ||
>+          ips_from_non_any_address_ports.count(c.address().ipaddr()) ||
>+          ips_from_non_any_address_ports.count(c.related_address().ipaddr())) {
>+        continue;
>+      }
>+      // Even when a port is bound to the "any" address, it should normally
>+      // still have an associated IP (determined by calling "connect" and then
>+      // "getsockaddr"). Though sometimes even this fails (meaning |is_any_ip|
>+      // will be true), and thus we have no way of knowing whether the port is
>+      // redundant or not. In that case, we'll use the port if we have
>+      // *no* ports bound to specific addresses. This is needed for corner
>+      // cases such as bugs.webrtc.org/7798.
>+      bool is_any_ip = rtc::IPIsAny(c.address().ipaddr());
>+      if (is_any_ip && !ips_from_non_any_address_ports.empty()) {
>+        continue;
>+      }
>+      port_signalable = true;
>+      // Still need to check the candidiate filter and sanitize the related
>+      // address before signaling the candidate itself.
>+      if (CheckCandidateFilter(c)) {
>+        signalable_candidates_from_any_address_ports.push_back(
>+            SanitizeRelatedAddress(c));
>+      }
>+    }
>+    if (port_signalable) {
>+      signalable_any_address_ports.push_back(port_data);
>+    } else {
>+      prunable_any_address_ports.push_back(port_data);
>+    }
>+  }
>+  PrunePorts(prunable_any_address_ports);
>+  for (PortData* port_data : signalable_any_address_ports) {
>+    RTC_LOG(INFO) << port_data->port()->ToString() << ": Port ready.";
>+    SignalPortReady(this, port_data->port());
>+    port_data->set_signaled();
>+  }
>+  RTC_LOG(INFO) << "Signaling candidates from the any address ports.";
>+  SignalCandidatesReady(this, signalable_candidates_from_any_address_ports);
>+}
>+
> void BasicPortAllocatorSession::OnPortDestroyed(
>     PortInterface* port) {
>   RTC_DCHECK(rtc::Thread::Current() == network_thread_);
>@@ -1039,8 +1255,8 @@ void BasicPortAllocatorSession::OnPortDestroyed(
>        iter != ports_.end(); ++iter) {
>     if (port == iter->port()) {
>       ports_.erase(iter);
>-      LOG_J(LS_INFO, port) << "Removed port from allocator ("
>-                           << static_cast<int>(ports_.size()) << " remaining)";
>+      RTC_LOG(LS_INFO) << port->ToString() << ": Removed port from allocator ("
>+                       << static_cast<int>(ports_.size()) << " remaining)";
>       return;
>     }
>   }
>@@ -1049,13 +1265,13 @@ void BasicPortAllocatorSession::OnPortDestroyed(
> 
> BasicPortAllocatorSession::PortData* BasicPortAllocatorSession::FindPort(
>     Port* port) {
>-  for (std::vector<PortData>::iterator it = ports_.begin();
>-       it != ports_.end(); ++it) {
>+  for (std::vector<PortData>::iterator it = ports_.begin(); it != ports_.end();
>+       ++it) {
>     if (it->port() == port) {
>       return &*it;
>     }
>   }
>-  return NULL;
>+  return nullptr;
> }
> 
> std::vector<BasicPortAllocatorSession::PortData*>
>@@ -1072,7 +1288,7 @@ BasicPortAllocatorSession::GetUnprunedPorts(
>   return unpruned_ports;
> }
> 
>-void BasicPortAllocatorSession::PrunePortsAndRemoveCandidates(
>+std::vector<Candidate> BasicPortAllocatorSession::PrunePorts(
>     const std::vector<PortData*>& port_data_list) {
>   std::vector<PortInterface*> pruned_ports;
>   std::vector<Candidate> removed_candidates;
>@@ -1090,6 +1306,12 @@ void BasicPortAllocatorSession::PrunePortsAndRemoveCandidates(
>   if (!pruned_ports.empty()) {
>     SignalPortsPruned(this, pruned_ports);
>   }
>+  return removed_candidates;
>+}
>+
>+void BasicPortAllocatorSession::PrunePortsAndSignalCandidatesRemoval(
>+    const std::vector<PortData*>& port_data_list) {
>+  std::vector<Candidate> removed_candidates = PrunePorts(port_data_list);
>   if (!removed_candidates.empty()) {
>     RTC_LOG(LS_INFO) << "Removed " << removed_candidates.size()
>                      << " candidates";
>@@ -1109,9 +1331,8 @@ AllocationSequence::AllocationSequence(BasicPortAllocatorSession* session,
>       state_(kInit),
>       flags_(flags),
>       udp_socket_(),
>-      udp_port_(NULL),
>-      phase_(0) {
>-}
>+      udp_port_(nullptr),
>+      phase_(0) {}
> 
> void AllocationSequence::Init() {
>   if (IsFlagSet(PORTALLOCATOR_ENABLE_SHARED_SOCKET)) {
>@@ -1119,16 +1340,16 @@ void AllocationSequence::Init() {
>         rtc::SocketAddress(network_->GetBestIP(), 0),
>         session_->allocator()->min_port(), session_->allocator()->max_port()));
>     if (udp_socket_) {
>-      udp_socket_->SignalReadPacket.connect(
>-          this, &AllocationSequence::OnReadPacket);
>+      udp_socket_->SignalReadPacket.connect(this,
>+                                            &AllocationSequence::OnReadPacket);
>     }
>-    // Continuing if |udp_socket_| is NULL, as local TCP and RelayPort using TCP
>-    // are next available options to setup a communication channel.
>+    // Continuing if |udp_socket_| is null, as local TCP and RelayPort using
>+    // TCP are next available options to setup a communication channel.
>   }
> }
> 
> void AllocationSequence::Clear() {
>-  udp_port_ = NULL;
>+  udp_port_ = nullptr;
>   relay_ports_.clear();
> }
> 
>@@ -1144,7 +1365,8 @@ AllocationSequence::~AllocationSequence() {
> }
> 
> void AllocationSequence::DisableEquivalentPhases(rtc::Network* network,
>-    PortConfiguration* config, uint32_t* flags) {
>+                                                 PortConfiguration* config,
>+                                                 uint32_t* flags) {
>   if (network_failed_) {
>     // If the network of this allocation sequence has ever become failed,
>     // it won't be equivalent to the new network.
>@@ -1221,8 +1443,8 @@ void AllocationSequence::OnMessage(rtc::Message* msg) {
>   const char* const PHASE_NAMES[kNumPhases] = {"Udp", "Relay", "Tcp"};
> 
>   // Perform all of the phases in the current step.
>-  LOG_J(LS_INFO, network_) << "Allocation Phase="
>-                           << PHASE_NAMES[phase_];
>+  RTC_LOG(LS_INFO) << network_->ToString()
>+                   << ": Allocation Phase=" << PHASE_NAMES[phase_];
> 
>   switch (phase_) {
>     case PHASE_UDP:
>@@ -1264,20 +1486,22 @@ void AllocationSequence::CreateUDPPorts() {
> 
>   // TODO(mallinath) - Remove UDPPort creating socket after shared socket
>   // is enabled completely.
>-  UDPPort* port = NULL;
>+  UDPPort* port = nullptr;
>   bool emit_local_candidate_for_anyaddress =
>       !IsFlagSet(PORTALLOCATOR_DISABLE_DEFAULT_LOCAL_CANDIDATE);
>   if (IsFlagSet(PORTALLOCATOR_ENABLE_SHARED_SOCKET) && udp_socket_) {
>     port = UDPPort::Create(
>         session_->network_thread(), session_->socket_factory(), network_,
>         udp_socket_.get(), session_->username(), session_->password(),
>-        session_->allocator()->origin(), emit_local_candidate_for_anyaddress);
>+        session_->allocator()->origin(), emit_local_candidate_for_anyaddress,
>+        session_->allocator()->stun_candidate_keepalive_interval());
>   } else {
>     port = UDPPort::Create(
>         session_->network_thread(), session_->socket_factory(), network_,
>         session_->allocator()->min_port(), session_->allocator()->max_port(),
>         session_->username(), session_->password(),
>-        session_->allocator()->origin(), emit_local_candidate_for_anyaddress);
>+        session_->allocator()->origin(), emit_local_candidate_for_anyaddress,
>+        session_->allocator()->stun_candidate_keepalive_interval());
>   }
> 
>   if (port) {
>@@ -1292,7 +1516,7 @@ void AllocationSequence::CreateUDPPorts() {
>         if (config_ && !config_->StunServers().empty()) {
>           RTC_LOG(LS_INFO)
>               << "AllocationSequence: UDPPort will be handling the "
>-              << "STUN candidate generation.";
>+                 "STUN candidate generation.";
>           port->set_server_addresses(config_->StunServers());
>         }
>       }
>@@ -1340,7 +1564,8 @@ void AllocationSequence::CreateStunPorts() {
>       session_->network_thread(), session_->socket_factory(), network_,
>       session_->allocator()->min_port(), session_->allocator()->max_port(),
>       session_->username(), session_->password(), config_->StunServers(),
>-      session_->allocator()->origin());
>+      session_->allocator()->origin(),
>+      session_->allocator()->stun_candidate_keepalive_interval());
>   if (port) {
>     session_->AddAllocatedPort(port, this, true);
>     // Since StunPort is not created using shared socket, |port| will not be
>@@ -1394,8 +1619,7 @@ void AllocationSequence::CreateGturnPort(const RelayServerConfig& config) {
> 
>     // Add the addresses of this protocol.
>     PortList::const_iterator relay_port;
>-    for (relay_port = config.ports.begin();
>-         relay_port != config.ports.end();
>+    for (relay_port = config.ports.begin(); relay_port != config.ports.end();
>          ++relay_port) {
>       port->AddServerAddress(*relay_port);
>       port->AddExternalAddress(*relay_port);
>@@ -1407,8 +1631,8 @@ void AllocationSequence::CreateGturnPort(const RelayServerConfig& config) {
> 
> void AllocationSequence::CreateTurnPort(const RelayServerConfig& config) {
>   PortList::const_iterator relay_port;
>-  for (relay_port = config.ports.begin();
>-       relay_port != config.ports.end(); ++relay_port) {
>+  for (relay_port = config.ports.begin(); relay_port != config.ports.end();
>+       ++relay_port) {
>     // Skip UDP connections to relay servers if it's disallowed.
>     if (IsFlagSet(PORTALLOCATOR_DISABLE_UDP_RELAY) &&
>         relay_port->proto == PROTO_UDP) {
>@@ -1422,7 +1646,8 @@ void AllocationSequence::CreateTurnPort(const RelayServerConfig& config) {
>     if (server_ip_family != AF_UNSPEC && server_ip_family != local_ip_family) {
>       RTC_LOG(LS_INFO)
>           << "Server and local address families are not compatible. "
>-          << "Server address: " << relay_port->address.ipaddr().ToString()
>+             "Server address: "
>+          << relay_port->address.ipaddr().ToString()
>           << " Local address: " << network_->GetBestIP().ToString();
>       continue;
>     }
>@@ -1449,9 +1674,8 @@ void AllocationSequence::CreateTurnPort(const RelayServerConfig& config) {
>           args, udp_socket_.get());
> 
>       if (!port) {
>-        RTC_LOG(LS_WARNING)
>-            << "Failed to create relay port with "
>-            << args.server_address->address.ToString();
>+        RTC_LOG(LS_WARNING) << "Failed to create relay port with "
>+                            << args.server_address->address.ToString();
>         continue;
>       }
> 
>@@ -1461,26 +1685,25 @@ void AllocationSequence::CreateTurnPort(const RelayServerConfig& config) {
>       port->SignalDestroyed.connect(this, &AllocationSequence::OnPortDestroyed);
>     } else {
>       port = session_->allocator()->relay_port_factory()->Create(
>-          args,
>-          session_->allocator()->min_port(),
>+          args, session_->allocator()->min_port(),
>           session_->allocator()->max_port());
> 
>       if (!port) {
>-        RTC_LOG(LS_WARNING)
>-            << "Failed to create relay port with "
>-            << args.server_address->address.ToString();
>+        RTC_LOG(LS_WARNING) << "Failed to create relay port with "
>+                            << args.server_address->address.ToString();
>         continue;
>       }
>     }
>-    RTC_DCHECK(port != NULL);
>+    RTC_DCHECK(port != nullptr);
>     session_->AddAllocatedPort(port.release(), this, true);
>   }
> }
> 
>-void AllocationSequence::OnReadPacket(
>-    rtc::AsyncPacketSocket* socket, const char* data, size_t size,
>-    const rtc::SocketAddress& remote_addr,
>-    const rtc::PacketTime& packet_time) {
>+void AllocationSequence::OnReadPacket(rtc::AsyncPacketSocket* socket,
>+                                      const char* data,
>+                                      size_t size,
>+                                      const rtc::SocketAddress& remote_addr,
>+                                      const rtc::PacketTime& packet_time) {
>   RTC_DCHECK(socket == udp_socket_.get());
> 
>   bool turn_port_found = false;
>@@ -1517,7 +1740,7 @@ void AllocationSequence::OnReadPacket(
> 
> void AllocationSequence::OnPortDestroyed(PortInterface* port) {
>   if (udp_port_ == port) {
>-    udp_port_ = NULL;
>+    udp_port_ = nullptr;
>     return;
>   }
> 
>@@ -1531,10 +1754,9 @@ void AllocationSequence::OnPortDestroyed(PortInterface* port) {
> }
> 
> // PortConfiguration
>-PortConfiguration::PortConfiguration(
>-    const rtc::SocketAddress& stun_address,
>-    const std::string& username,
>-    const std::string& password)
>+PortConfiguration::PortConfiguration(const rtc::SocketAddress& stun_address,
>+                                     const std::string& username,
>+                                     const std::string& password)
>     : stun_address(stun_address), username(username), password(password) {
>   if (!stun_address.IsNil())
>     stun_servers.insert(stun_address);
>@@ -1543,9 +1765,7 @@ PortConfiguration::PortConfiguration(
> PortConfiguration::PortConfiguration(const ServerAddresses& stun_servers,
>                                      const std::string& username,
>                                      const std::string& password)
>-    : stun_servers(stun_servers),
>-      username(username),
>-      password(password) {
>+    : stun_servers(stun_servers), username(username), password(password) {
>   if (!stun_servers.empty())
>     stun_address = *(stun_servers.begin());
> }
>@@ -1571,12 +1791,11 @@ void PortConfiguration::AddRelay(const RelayServerConfig& config) {
>   relays.push_back(config);
> }
> 
>-bool PortConfiguration::SupportsProtocol(
>-    const RelayServerConfig& relay, ProtocolType type) const {
>+bool PortConfiguration::SupportsProtocol(const RelayServerConfig& relay,
>+                                         ProtocolType type) const {
>   PortList::const_iterator relay_port;
>-  for (relay_port = relay.ports.begin();
>-        relay_port != relay.ports.end();
>-        ++relay_port) {
>+  for (relay_port = relay.ports.begin(); relay_port != relay.ports.end();
>+       ++relay_port) {
>     if (relay_port->proto == type)
>       return true;
>   }
>@@ -1586,15 +1805,15 @@ bool PortConfiguration::SupportsProtocol(
> bool PortConfiguration::SupportsProtocol(RelayType turn_type,
>                                          ProtocolType type) const {
>   for (size_t i = 0; i < relays.size(); ++i) {
>-    if (relays[i].type == turn_type &&
>-        SupportsProtocol(relays[i], type))
>+    if (relays[i].type == turn_type && SupportsProtocol(relays[i], type))
>       return true;
>   }
>   return false;
> }
> 
> ServerAddresses PortConfiguration::GetRelayServerAddresses(
>-    RelayType turn_type, ProtocolType type) const {
>+    RelayType turn_type,
>+    ProtocolType type) const {
>   ServerAddresses servers;
>   for (size_t i = 0; i < relays.size(); ++i) {
>     if (relays[i].type == turn_type && SupportsProtocol(relays[i], type)) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/client/basicportallocator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/client/basicportallocator.h
>index c9b93be227f..2105172e4e9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/client/basicportallocator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/client/basicportallocator.h
>@@ -17,8 +17,8 @@
> 
> #include "api/turncustomizer.h"
> #include "p2p/base/portallocator.h"
>-#include "p2p/client/turnportfactory.h"
> #include "p2p/client/relayportfactoryinterface.h"
>+#include "p2p/client/turnportfactory.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/messagequeue.h"
> #include "rtc_base/network.h"
>@@ -47,13 +47,22 @@ class BasicPortAllocator : public PortAllocator {
> 
>   // Set to kDefaultNetworkIgnoreMask by default.
>   void SetNetworkIgnoreMask(int network_ignore_mask) override;
>-  int network_ignore_mask() const { return network_ignore_mask_; }
>+  int network_ignore_mask() const {
>+    CheckRunOnValidThreadIfInitialized();
>+    return network_ignore_mask_;
>+  }
> 
>-  rtc::NetworkManager* network_manager() const { return network_manager_; }
>+  rtc::NetworkManager* network_manager() const {
>+    CheckRunOnValidThreadIfInitialized();
>+    return network_manager_;
>+  }
> 
>   // If socket_factory() is set to NULL each PortAllocatorSession
>   // creates its own socket factory.
>-  rtc::PacketSocketFactory* socket_factory() { return socket_factory_; }
>+  rtc::PacketSocketFactory* socket_factory() {
>+    CheckRunOnValidThreadIfInitialized();
>+    return socket_factory_;
>+  }
> 
>   PortAllocatorSession* CreateSessionInternal(
>       const std::string& content_name,
>@@ -65,6 +74,7 @@ class BasicPortAllocator : public PortAllocator {
>   void AddTurnServer(const RelayServerConfig& turn_server);
> 
>   RelayPortFactoryInterface* relay_port_factory() {
>+    CheckRunOnValidThreadIfInitialized();
>     return relay_port_factory_;
>   }
> 
>@@ -127,6 +137,8 @@ class BasicPortAllocatorSession : public PortAllocatorSession,
>   bool CandidatesAllocationDone() const override;
>   void RegatherOnFailedNetworks() override;
>   void RegatherOnAllNetworks() override;
>+  void SetStunKeepaliveIntervalForReadyPorts(
>+      const absl::optional<int>& stun_keepalive_interval) override;
>   void PruneAllPorts() override;
> 
>  protected:
>@@ -139,7 +151,7 @@ class BasicPortAllocatorSession : public PortAllocatorSession,
>   // network (or a timeout occurs), we will start allocating ports.
>   virtual void ConfigReady(PortConfiguration* config);
> 
>-  // MessageHandler.  Can be overridden if message IDs do not conflict.
>+  // MessageHandler.  Can be overriden if message IDs do not conflict.
>   void OnMessage(rtc::Message* message) override;
> 
>  private:
>@@ -156,6 +168,10 @@ class BasicPortAllocatorSession : public PortAllocatorSession,
>     bool error() const { return state_ == STATE_ERROR; }
>     bool pruned() const { return state_ == STATE_PRUNED; }
>     bool inprogress() const { return state_ == STATE_INPROGRESS; }
>+    // True if this port has been fired in SignalPortReady. This may be false
>+    // even if ready() is true if the port was bound to the "any" address; see
>+    // comment above SignalAnyAddressPortsAndCandidatesReadyIfNotRedundant.
>+    bool signaled() const { return signaled_; }
>     // Returns true if this port is ready to be used.
>     bool ready() const {
>       return has_pairable_candidate_ && state_ != STATE_ERROR &&
>@@ -174,13 +190,12 @@ class BasicPortAllocatorSession : public PortAllocatorSession,
>       }
>       has_pairable_candidate_ = has_pairable_candidate;
>     }
>-    void set_complete() {
>-      state_ = STATE_COMPLETE;
>-    }
>+    void set_complete() { state_ = STATE_COMPLETE; }
>     void set_error() {
>       RTC_DCHECK(state_ == STATE_INPROGRESS);
>       state_ = STATE_ERROR;
>     }
>+    void set_signaled() { signaled_ = true; }
> 
>    private:
>     enum State {
>@@ -193,6 +208,7 @@ class BasicPortAllocatorSession : public PortAllocatorSession,
>     Port* port_ = nullptr;
>     AllocationSequence* sequence_ = nullptr;
>     bool has_pairable_candidate_ = false;
>+    bool signaled_ = false;
>     State state_ = STATE_INPROGRESS;
>   };
> 
>@@ -206,14 +222,14 @@ class BasicPortAllocatorSession : public PortAllocatorSession,
>   void DisableEquivalentPhases(rtc::Network* network,
>                                PortConfiguration* config,
>                                uint32_t* flags);
>-  void AddAllocatedPort(Port* port, AllocationSequence* seq,
>+  void AddAllocatedPort(Port* port,
>+                        AllocationSequence* seq,
>                         bool prepare_address);
>   void OnCandidateReady(Port* port, const Candidate& c);
>   void OnPortComplete(Port* port);
>   void OnPortError(Port* port);
>   void OnProtocolEnabled(AllocationSequence* seq, ProtocolType proto);
>   void OnPortDestroyed(PortInterface* port);
>-  void MaybeSignalCandidatesAllocationDone();
>   void OnPortAllocationComplete(AllocationSequence* seq);
>   PortData* FindPort(Port* port);
>   std::vector<rtc::Network*> GetNetworks();
>@@ -230,9 +246,13 @@ class BasicPortAllocatorSession : public PortAllocatorSession,
> 
>   std::vector<PortData*> GetUnprunedPorts(
>       const std::vector<rtc::Network*>& networks);
>+  // Prunes ports and removes candidates gathered from these ports locally. The
>+  // list of the removed candidates are returned.
>+  std::vector<Candidate> PrunePorts(
>+      const std::vector<PortData*>& port_data_list);
>   // Prunes ports and signal the remote side to remove the candidates that
>   // were previously signaled from these ports.
>-  void PrunePortsAndRemoveCandidates(
>+  void PrunePortsAndSignalCandidatesRemoval(
>       const std::vector<PortData*>& port_data_list);
>   // Gets filtered and sanitized candidates generated from a port and
>   // append to |candidates|.
>@@ -241,6 +261,12 @@ class BasicPortAllocatorSession : public PortAllocatorSession,
>   Port* GetBestTurnPortForNetwork(const std::string& network_name) const;
>   // Returns true if at least one TURN port is pruned.
>   bool PruneTurnPorts(Port* newly_pairable_turn_port);
>+  // Fires signals related to aggregate status update in the allocation,
>+  // including candidates allocation done, and any address ports and their
>+  // candidates ready.
>+  void FireAllocationStatusSignalsIfNeeded();
>+  // TODO(qingsi): Rename "any address" to "wildcard address" in p2p/.
>+  void SignalAnyAddressPortsAndCandidatesReadyIfNotRedundant();
> 
>   BasicPortAllocator* allocator_;
>   rtc::Thread* network_thread_;
>@@ -296,8 +322,8 @@ struct PortConfiguration : public rtc::MessageData {
>   bool SupportsProtocol(RelayType turn_type, ProtocolType type) const;
>   // Helper method returns the server addresses for the matching RelayType and
>   // Protocol type.
>-  ServerAddresses GetRelayServerAddresses(
>-      RelayType turn_type, ProtocolType type) const;
>+  ServerAddresses GetRelayServerAddresses(RelayType turn_type,
>+                                          ProtocolType type) const;
> };
> 
> class UDPPort;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/client/basicportallocator_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/client/basicportallocator_unittest.cc
>index a1413285868..b3e52ffec51 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/client/basicportallocator_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/client/basicportallocator_unittest.cc
>@@ -14,6 +14,7 @@
> #include "p2p/base/basicpacketsocketfactory.h"
> #include "p2p/base/p2pconstants.h"
> #include "p2p/base/p2ptransportchannel.h"
>+#include "p2p/base/stunport.h"
> #include "p2p/base/testrelayserver.h"
> #include "p2p/base/teststunserver.h"
> #include "p2p/base/testturnserver.h"
>@@ -33,6 +34,7 @@
> #include "rtc_base/ssladapter.h"
> #include "rtc_base/thread.h"
> #include "rtc_base/virtualsocketserver.h"
>+#include "system_wrappers/include/metrics_default.h"
> 
> using rtc::IPAddress;
> using rtc::SocketAddress;
>@@ -97,6 +99,25 @@ static const char kTurnPassword[] = "test";
> // Add some margin of error for slow bots.
> static const int kStunTimeoutMs = cricket::STUN_TOTAL_TIMEOUT;
> 
>+namespace {
>+
>+void CheckStunKeepaliveIntervalOfAllReadyPorts(
>+    const cricket::PortAllocatorSession* allocator_session,
>+    int expected) {
>+  auto ready_ports = allocator_session->ReadyPorts();
>+  for (const auto* port : ready_ports) {
>+    if (port->Type() == cricket::STUN_PORT_TYPE ||
>+        (port->Type() == cricket::LOCAL_PORT_TYPE &&
>+         port->GetProtocol() == cricket::PROTO_UDP)) {
>+      EXPECT_EQ(
>+          static_cast<const cricket::UDPPort*>(port)->stun_keepalive_delay(),
>+          expected);
>+    }
>+  }
>+}
>+
>+}  // namespace
>+
> namespace cricket {
> 
> // Helper for dumping candidates
>@@ -146,7 +167,9 @@ class BasicPortAllocatorTestBase : public testing::Test,
>     allocator_.reset(new BasicPortAllocator(&network_manager_, stun_servers,
>                                             kRelayUdpIntAddr, kRelayTcpIntAddr,
>                                             kRelaySslTcpIntAddr));
>+    allocator_->Initialize();
>     allocator_->set_step_delay(kMinimumStepDelay);
>+    webrtc::metrics::Reset();
>   }
> 
>   void AddInterface(const SocketAddress& addr) {
>@@ -181,6 +204,7 @@ class BasicPortAllocatorTestBase : public testing::Test,
>   // Endpoint is on the public network. No STUN or TURN.
>   void ResetWithNoServersOrNat() {
>     allocator_.reset(new BasicPortAllocator(&network_manager_));
>+    allocator_->Initialize();
>     allocator_->set_step_delay(kMinimumStepDelay);
>   }
>   // Endpoint is behind a NAT, with STUN specified.
>@@ -463,7 +487,8 @@ class BasicPortAllocatorTestBase : public testing::Test,
>     }
>     allocator_.reset(new BasicPortAllocator(
>         &network_manager_, nat_socket_factory_.get(), stun_servers));
>-    allocator().set_step_delay(kMinimumStepDelay);
>+    allocator_->Initialize();
>+    allocator_->set_step_delay(kMinimumStepDelay);
>   }
> 
>   std::unique_ptr<rtc::VirtualSocketServer> vss_;
>@@ -508,7 +533,7 @@ class BasicPortAllocatorTest : public FakeClockBase,
>     network_manager_.set_default_local_addresses(kPrivateAddr.ipaddr(),
>                                                  rtc::IPAddress());
>     if (!session_) {
>-      EXPECT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+      ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>     }
>     session_->set_flags(session_->flags() |
>                         PORTALLOCATOR_DISABLE_ADAPTER_ENUMERATION |
>@@ -520,8 +545,8 @@ class BasicPortAllocatorTest : public FakeClockBase,
> 
>     uint32_t total_candidates = 0;
>     if (!host_candidate_addr.IsNil()) {
>-      EXPECT_PRED4(HasCandidate, candidates_, "local", "udp",
>-                   rtc::SocketAddress(kPrivateAddr.ipaddr(), 0));
>+      EXPECT_TRUE(HasCandidate(candidates_, "local", "udp",
>+                               rtc::SocketAddress(kPrivateAddr.ipaddr(), 0)));
>       ++total_candidates;
>     }
>     if (!stun_candidate_addr.IsNil()) {
>@@ -529,20 +554,23 @@ class BasicPortAllocatorTest : public FakeClockBase,
>       if (host_candidate_addr.IsNil()) {
>         related_address.SetIP(rtc::GetAnyIP(stun_candidate_addr.family()));
>       }
>-      EXPECT_PRED5(HasCandidateWithRelatedAddr, candidates_, "stun", "udp",
>-                   rtc::SocketAddress(stun_candidate_addr, 0), related_address);
>+      EXPECT_TRUE(HasCandidateWithRelatedAddr(
>+          candidates_, "stun", "udp",
>+          rtc::SocketAddress(stun_candidate_addr, 0), related_address));
>       ++total_candidates;
>     }
>     if (!relay_candidate_udp_transport_addr.IsNil()) {
>-      EXPECT_PRED5(HasCandidateWithRelatedAddr, candidates_, "relay", "udp",
>-                   rtc::SocketAddress(relay_candidate_udp_transport_addr, 0),
>-                   rtc::SocketAddress(stun_candidate_addr, 0));
>+      EXPECT_TRUE(HasCandidateWithRelatedAddr(
>+          candidates_, "relay", "udp",
>+          rtc::SocketAddress(relay_candidate_udp_transport_addr, 0),
>+          rtc::SocketAddress(stun_candidate_addr, 0)));
>       ++total_candidates;
>     }
>     if (!relay_candidate_tcp_transport_addr.IsNil()) {
>-      EXPECT_PRED5(HasCandidateWithRelatedAddr, candidates_, "relay", "udp",
>-                   rtc::SocketAddress(relay_candidate_tcp_transport_addr, 0),
>-                   rtc::SocketAddress(stun_candidate_addr, 0));
>+      EXPECT_TRUE(HasCandidateWithRelatedAddr(
>+          candidates_, "relay", "udp",
>+          rtc::SocketAddress(relay_candidate_tcp_transport_addr, 0),
>+          rtc::SocketAddress(stun_candidate_addr, 0)));
>       ++total_candidates;
>     }
> 
>@@ -556,6 +584,7 @@ class BasicPortAllocatorTest : public FakeClockBase,
>     AddInterface(kClientAddr, "net1");
>     AddInterface(kClientIPv6Addr, "net1");
>     allocator_.reset(new BasicPortAllocator(&network_manager_));
>+    allocator_->Initialize();
>     allocator_->SetConfiguration(allocator_->stun_servers(),
>                                  allocator_->turn_servers(), 0, true);
>     AddTurnServers(kTurnUdpIntIPv6Addr, rtc::SocketAddress());
>@@ -566,7 +595,7 @@ class BasicPortAllocatorTest : public FakeClockBase,
>         allocator().flags() | PORTALLOCATOR_ENABLE_SHARED_SOCKET |
>         PORTALLOCATOR_ENABLE_IPV6 | PORTALLOCATOR_DISABLE_TCP);
> 
>-    EXPECT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+    ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>     session_->StartGettingPorts();
>     EXPECT_TRUE_SIMULATED_WAIT(candidate_allocation_done_,
>                                kDefaultAllocationTimeout, fake_clock);
>@@ -584,15 +613,16 @@ class BasicPortAllocatorTest : public FakeClockBase,
>     const std::vector<Candidate>& ready_candidates =
>         session_->ReadyCandidates();
>     EXPECT_EQ(3U, ready_candidates.size());
>-    EXPECT_PRED4(HasCandidate, ready_candidates, "local", "udp", kClientAddr);
>-    EXPECT_PRED4(HasCandidate, ready_candidates, "relay", "udp",
>-                 rtc::SocketAddress(kTurnUdpExtAddr.ipaddr(), 0));
>+    EXPECT_TRUE(HasCandidate(ready_candidates, "local", "udp", kClientAddr));
>+    EXPECT_TRUE(HasCandidate(ready_candidates, "relay", "udp",
>+                             rtc::SocketAddress(kTurnUdpExtAddr.ipaddr(), 0)));
>   }
> 
>   void TestUdpTurnPortPrunesTcpTurnPort() {
>     turn_server_.AddInternalSocket(kTurnTcpIntAddr, PROTO_TCP);
>     AddInterface(kClientAddr);
>     allocator_.reset(new BasicPortAllocator(&network_manager_));
>+    allocator_->Initialize();
>     allocator_->SetConfiguration(allocator_->stun_servers(),
>                                  allocator_->turn_servers(), 0, true);
>     AddTurnServers(kTurnUdpIntAddr, kTurnTcpIntAddr);
>@@ -601,7 +631,7 @@ class BasicPortAllocatorTest : public FakeClockBase,
>                           PORTALLOCATOR_ENABLE_SHARED_SOCKET |
>                           PORTALLOCATOR_DISABLE_TCP);
> 
>-    EXPECT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+    ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>     session_->StartGettingPorts();
>     EXPECT_TRUE_SIMULATED_WAIT(candidate_allocation_done_,
>                                kDefaultAllocationTimeout, fake_clock);
>@@ -624,9 +654,9 @@ class BasicPortAllocatorTest : public FakeClockBase,
>     const std::vector<Candidate>& ready_candidates =
>         session_->ReadyCandidates();
>     EXPECT_EQ(2U, ready_candidates.size());
>-    EXPECT_PRED4(HasCandidate, ready_candidates, "local", "udp", kClientAddr);
>-    EXPECT_PRED4(HasCandidate, ready_candidates, "relay", "udp",
>-                 rtc::SocketAddress(kTurnUdpExtAddr.ipaddr(), 0));
>+    EXPECT_TRUE(HasCandidate(ready_candidates, "local", "udp", kClientAddr));
>+    EXPECT_TRUE(HasCandidate(ready_candidates, "relay", "udp",
>+                             rtc::SocketAddress(kTurnUdpExtAddr.ipaddr(), 0)));
>   }
> 
>   void TestEachInterfaceHasItsOwnTurnPorts() {
>@@ -639,6 +669,7 @@ class BasicPortAllocatorTest : public FakeClockBase,
>     AddInterface(kClientAddr2, "net2", rtc::ADAPTER_TYPE_CELLULAR);
>     AddInterface(kClientIPv6Addr2, "net2", rtc::ADAPTER_TYPE_CELLULAR);
>     allocator_.reset(new BasicPortAllocator(&network_manager_));
>+    allocator_->Initialize();
>     allocator_->SetConfiguration(allocator_->stun_servers(),
>                                  allocator_->turn_servers(), 0, true);
>     // Have both UDP/TCP and IPv4/IPv6 TURN ports.
>@@ -649,7 +680,7 @@ class BasicPortAllocatorTest : public FakeClockBase,
>     allocator_->set_flags(
>         allocator().flags() | PORTALLOCATOR_ENABLE_SHARED_SOCKET |
>         PORTALLOCATOR_ENABLE_IPV6 | PORTALLOCATOR_ENABLE_IPV6_ON_WIFI);
>-    EXPECT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+    ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>     session_->StartGettingPorts();
>     EXPECT_TRUE_SIMULATED_WAIT(candidate_allocation_done_,
>                                kDefaultAllocationTimeout, fake_clock);
>@@ -674,20 +705,20 @@ class BasicPortAllocatorTest : public FakeClockBase,
>     const std::vector<Candidate>& ready_candidates =
>         session_->ReadyCandidates();
>     EXPECT_EQ(10U, ready_candidates.size());
>-    EXPECT_PRED4(HasCandidate, ready_candidates, "local", "udp", kClientAddr);
>-    EXPECT_PRED4(HasCandidate, ready_candidates, "local", "udp", kClientAddr2);
>-    EXPECT_PRED4(HasCandidate, ready_candidates, "local", "udp",
>-                 kClientIPv6Addr);
>-    EXPECT_PRED4(HasCandidate, ready_candidates, "local", "udp",
>-                 kClientIPv6Addr2);
>-    EXPECT_PRED4(HasCandidate, ready_candidates, "local", "tcp", kClientAddr);
>-    EXPECT_PRED4(HasCandidate, ready_candidates, "local", "tcp", kClientAddr2);
>-    EXPECT_PRED4(HasCandidate, ready_candidates, "local", "tcp",
>-                 kClientIPv6Addr);
>-    EXPECT_PRED4(HasCandidate, ready_candidates, "local", "tcp",
>-                 kClientIPv6Addr2);
>-    EXPECT_PRED4(HasCandidate, ready_candidates, "relay", "udp",
>-                 rtc::SocketAddress(kTurnUdpExtAddr.ipaddr(), 0));
>+    EXPECT_TRUE(HasCandidate(ready_candidates, "local", "udp", kClientAddr));
>+    EXPECT_TRUE(HasCandidate(ready_candidates, "local", "udp", kClientAddr2));
>+    EXPECT_TRUE(
>+        HasCandidate(ready_candidates, "local", "udp", kClientIPv6Addr));
>+    EXPECT_TRUE(
>+        HasCandidate(ready_candidates, "local", "udp", kClientIPv6Addr2));
>+    EXPECT_TRUE(HasCandidate(ready_candidates, "local", "tcp", kClientAddr));
>+    EXPECT_TRUE(HasCandidate(ready_candidates, "local", "tcp", kClientAddr2));
>+    EXPECT_TRUE(
>+        HasCandidate(ready_candidates, "local", "tcp", kClientIPv6Addr));
>+    EXPECT_TRUE(
>+        HasCandidate(ready_candidates, "local", "tcp", kClientIPv6Addr2));
>+    EXPECT_TRUE(HasCandidate(ready_candidates, "relay", "udp",
>+                             rtc::SocketAddress(kTurnUdpExtAddr.ipaddr(), 0)));
>   }
> };
> 
>@@ -704,7 +735,7 @@ TEST_F(BasicPortAllocatorTest, TestBasic) {
>   EXPECT_TRUE(HasRelayAddress(ProtocolAddress(kRelayTcpIntAddr, PROTO_TCP)));
>   EXPECT_TRUE(
>       HasRelayAddress(ProtocolAddress(kRelaySslTcpIntAddr, PROTO_SSLTCP)));
>-  EXPECT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+  ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>   EXPECT_FALSE(session_->CandidatesAllocationDone());
> }
> 
>@@ -720,7 +751,7 @@ TEST_F(BasicPortAllocatorTest, TestIgnoreOnlyLoopbackNetworkByDefault) {
>                rtc::ADAPTER_TYPE_VPN);
>   AddInterface(SocketAddress(IPAddress(0x12345604U), 0), "test_lo",
>                rtc::ADAPTER_TYPE_LOOPBACK);
>-  EXPECT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+  ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>   session_->set_flags(PORTALLOCATOR_DISABLE_STUN | PORTALLOCATOR_DISABLE_RELAY |
>                       PORTALLOCATOR_DISABLE_TCP);
>   session_->StartGettingPorts();
>@@ -742,7 +773,7 @@ TEST_F(BasicPortAllocatorTest, TestIgnoreNetworksAccordingToIgnoreMask) {
>   allocator_->SetNetworkIgnoreMask(rtc::ADAPTER_TYPE_ETHERNET |
>                                    rtc::ADAPTER_TYPE_LOOPBACK |
>                                    rtc::ADAPTER_TYPE_WIFI);
>-  EXPECT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+  ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>   session_->set_flags(PORTALLOCATOR_DISABLE_STUN | PORTALLOCATOR_DISABLE_RELAY |
>                       PORTALLOCATOR_DISABLE_TCP);
>   session_->StartGettingPorts();
>@@ -765,13 +796,13 @@ TEST_F(BasicPortAllocatorTest,
>                         cricket::PORTALLOCATOR_DISABLE_RELAY |
>                         cricket::PORTALLOCATOR_DISABLE_TCP |
>                         cricket::PORTALLOCATOR_DISABLE_COSTLY_NETWORKS);
>-  EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
>+  ASSERT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
>   session_->StartGettingPorts();
>   EXPECT_TRUE_SIMULATED_WAIT(candidate_allocation_done_,
>                              kDefaultAllocationTimeout, fake_clock);
>   // Should only get one Wi-Fi candidate.
>   EXPECT_EQ(1U, candidates_.size());
>-  EXPECT_PRED4(HasCandidate, candidates_, "local", "udp", wifi);
>+  EXPECT_TRUE(HasCandidate(candidates_, "local", "udp", wifi));
> }
> 
> // Test that when the PORTALLOCATOR_DISABLE_COSTLY_NETWORKS flag is set and
>@@ -791,14 +822,14 @@ TEST_F(BasicPortAllocatorTest,
>                         cricket::PORTALLOCATOR_DISABLE_RELAY |
>                         cricket::PORTALLOCATOR_DISABLE_TCP |
>                         cricket::PORTALLOCATOR_DISABLE_COSTLY_NETWORKS);
>-  EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
>+  ASSERT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
>   session_->StartGettingPorts();
>   EXPECT_TRUE_SIMULATED_WAIT(candidate_allocation_done_,
>                              kDefaultAllocationTimeout, fake_clock);
>   // Should only get two candidates, none of which is cell.
>   EXPECT_EQ(2U, candidates_.size());
>-  EXPECT_PRED4(HasCandidate, candidates_, "local", "udp", unknown1);
>-  EXPECT_PRED4(HasCandidate, candidates_, "local", "udp", unknown2);
>+  EXPECT_TRUE(HasCandidate(candidates_, "local", "udp", unknown1));
>+  EXPECT_TRUE(HasCandidate(candidates_, "local", "udp", unknown2));
> }
> 
> // Test that when the PORTALLOCATOR_DISABLE_COSTLY_NETWORKS flag is set and
>@@ -819,13 +850,13 @@ TEST_F(BasicPortAllocatorTest,
>                         cricket::PORTALLOCATOR_DISABLE_RELAY |
>                         cricket::PORTALLOCATOR_DISABLE_TCP |
>                         cricket::PORTALLOCATOR_DISABLE_COSTLY_NETWORKS);
>-  EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
>+  ASSERT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
>   session_->StartGettingPorts();
>   EXPECT_TRUE_SIMULATED_WAIT(candidate_allocation_done_,
>                              kDefaultAllocationTimeout, fake_clock);
>   // Should only get one Wi-Fi candidate.
>   EXPECT_EQ(1U, candidates_.size());
>-  EXPECT_PRED4(HasCandidate, candidates_, "local", "udp", wifi);
>+  EXPECT_TRUE(HasCandidate(candidates_, "local", "udp", wifi));
> }
> 
> // Test that if the PORTALLOCATOR_DISABLE_COSTLY_NETWORKS flag is set, but the
>@@ -840,13 +871,86 @@ TEST_F(BasicPortAllocatorTest,
>                         cricket::PORTALLOCATOR_DISABLE_RELAY |
>                         cricket::PORTALLOCATOR_DISABLE_TCP |
>                         cricket::PORTALLOCATOR_DISABLE_COSTLY_NETWORKS);
>-  EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
>+  ASSERT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
>   session_->StartGettingPorts();
>   EXPECT_TRUE_SIMULATED_WAIT(candidate_allocation_done_,
>                              kDefaultAllocationTimeout, fake_clock);
>   // Make sure we got the cell candidate.
>   EXPECT_EQ(1U, candidates_.size());
>-  EXPECT_PRED4(HasCandidate, candidates_, "local", "udp", cellular);
>+  EXPECT_TRUE(HasCandidate(candidates_, "local", "udp", cellular));
>+}
>+
>+// Test that if both PORTALLOCATOR_DISABLE_COSTLY_NETWORKS is set, and there is
>+// a WiFi network with link-local IP address and a cellular network, then the
>+// cellular candidate will still be gathered.
>+TEST_F(BasicPortAllocatorTest,
>+       CellNotRemovedWhenCostlyNetworksDisabledAndWifiIsLinkLocal) {
>+  SocketAddress wifi_link_local("169.254.0.1", 0);
>+  SocketAddress cellular(IPAddress(0x12345601U), 0);
>+  AddInterface(wifi_link_local, "test_wlan0", rtc::ADAPTER_TYPE_WIFI);
>+  AddInterface(cellular, "test_cell0", rtc::ADAPTER_TYPE_CELLULAR);
>+
>+  allocator().set_flags(cricket::PORTALLOCATOR_DISABLE_STUN |
>+                        cricket::PORTALLOCATOR_DISABLE_RELAY |
>+                        cricket::PORTALLOCATOR_DISABLE_TCP |
>+                        cricket::PORTALLOCATOR_DISABLE_COSTLY_NETWORKS);
>+  ASSERT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
>+  session_->StartGettingPorts();
>+  EXPECT_TRUE_SIMULATED_WAIT(candidate_allocation_done_,
>+                             kDefaultAllocationTimeout, fake_clock);
>+  // Make sure we got both wifi and cell candidates.
>+  EXPECT_EQ(2U, candidates_.size());
>+  EXPECT_TRUE(HasCandidate(candidates_, "local", "udp", wifi_link_local));
>+  EXPECT_TRUE(HasCandidate(candidates_, "local", "udp", cellular));
>+}
>+
>+// Test that if both PORTALLOCATOR_DISABLE_COSTLY_NETWORKS is set, and there is
>+// a WiFi network with link-local IP address, a WiFi network with a normal IP
>+// address and a cellular network, then the cellular candidate will not be
>+// gathered.
>+TEST_F(BasicPortAllocatorTest,
>+       CellRemovedWhenCostlyNetworksDisabledAndBothWifisPresent) {
>+  SocketAddress wifi(IPAddress(0x12345600U), 0);
>+  SocketAddress wifi_link_local("169.254.0.1", 0);
>+  SocketAddress cellular(IPAddress(0x12345601U), 0);
>+  AddInterface(wifi, "test_wlan0", rtc::ADAPTER_TYPE_WIFI);
>+  AddInterface(wifi_link_local, "test_wlan1", rtc::ADAPTER_TYPE_WIFI);
>+  AddInterface(cellular, "test_cell0", rtc::ADAPTER_TYPE_CELLULAR);
>+
>+  allocator().set_flags(cricket::PORTALLOCATOR_DISABLE_STUN |
>+                        cricket::PORTALLOCATOR_DISABLE_RELAY |
>+                        cricket::PORTALLOCATOR_DISABLE_TCP |
>+                        cricket::PORTALLOCATOR_DISABLE_COSTLY_NETWORKS);
>+  ASSERT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
>+  session_->StartGettingPorts();
>+  EXPECT_TRUE_SIMULATED_WAIT(candidate_allocation_done_,
>+                             kDefaultAllocationTimeout, fake_clock);
>+  // Make sure we got only wifi candidates.
>+  EXPECT_EQ(2U, candidates_.size());
>+  EXPECT_TRUE(HasCandidate(candidates_, "local", "udp", wifi));
>+  EXPECT_TRUE(HasCandidate(candidates_, "local", "udp", wifi_link_local));
>+}
>+
>+// Test that the adapter types of the Ethernet and the VPN can be correctly
>+// identified so that the Ethernet has a lower network cost than the VPN, and
>+// the Ethernet is not filtered out if PORTALLOCATOR_DISABLE_COSTLY_NETWORKS is
>+// set.
>+TEST_F(BasicPortAllocatorTest,
>+       EthernetIsNotFilteredOutWhenCostlyNetworksDisabledAndVpnPresent) {
>+  AddInterface(kClientAddr, "eth0", rtc::ADAPTER_TYPE_ETHERNET);
>+  AddInterface(kClientAddr2, "tap0", rtc::ADAPTER_TYPE_VPN);
>+  allocator().set_flags(PORTALLOCATOR_DISABLE_COSTLY_NETWORKS |
>+                        PORTALLOCATOR_DISABLE_RELAY |
>+                        PORTALLOCATOR_DISABLE_TCP);
>+  ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+  session_->StartGettingPorts();
>+  ASSERT_TRUE_SIMULATED_WAIT(candidate_allocation_done_,
>+                             kDefaultAllocationTimeout, fake_clock);
>+  // The VPN tap0 network should be filtered out as a costly network, and we
>+  // should have a UDP port and a STUN port from the Ethernet eth0.
>+  ASSERT_EQ(2U, ports_.size());
>+  EXPECT_EQ(ports_[0]->Network()->name(), "eth0");
>+  EXPECT_EQ(ports_[1]->Network()->name(), "eth0");
> }
> 
> // Test that no more than allocator.max_ipv6_networks() IPv6 networks are used
>@@ -863,14 +967,14 @@ TEST_F(BasicPortAllocatorTest, MaxIpv6NetworksLimitEnforced) {
>                         PORTALLOCATOR_DISABLE_STUN |
>                         PORTALLOCATOR_DISABLE_RELAY);
> 
>-  EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
>+  ASSERT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
>   session_->StartGettingPorts();
>   EXPECT_TRUE_SIMULATED_WAIT(candidate_allocation_done_,
>                              kDefaultAllocationTimeout, fake_clock);
>   EXPECT_EQ(2U, candidates_.size());
>   // Ensure the expected two interfaces (eth0 and eth1) were used.
>-  EXPECT_PRED4(HasCandidate, candidates_, "local", "udp", kClientIPv6Addr);
>-  EXPECT_PRED4(HasCandidate, candidates_, "local", "udp", kClientIPv6Addr2);
>+  EXPECT_TRUE(HasCandidate(candidates_, "local", "udp", kClientIPv6Addr));
>+  EXPECT_TRUE(HasCandidate(candidates_, "local", "udp", kClientIPv6Addr2));
> }
> 
> // Ensure that allocator.max_ipv6_networks() doesn't prevent IPv4 networks from
>@@ -888,23 +992,23 @@ TEST_F(BasicPortAllocatorTest, MaxIpv6NetworksLimitDoesNotImpactIpv4Networks) {
>                         PORTALLOCATOR_DISABLE_STUN |
>                         PORTALLOCATOR_DISABLE_RELAY);
> 
>-  EXPECT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
>+  ASSERT_TRUE(CreateSession(cricket::ICE_CANDIDATE_COMPONENT_RTP));
>   session_->StartGettingPorts();
>   EXPECT_TRUE_SIMULATED_WAIT(candidate_allocation_done_,
>                              kDefaultAllocationTimeout, fake_clock);
>   EXPECT_EQ(3U, candidates_.size());
>   // Ensure that only one IPv6 interface was used, but both IPv4 interfaces
>   // were used.
>-  EXPECT_PRED4(HasCandidate, candidates_, "local", "udp", kClientIPv6Addr);
>-  EXPECT_PRED4(HasCandidate, candidates_, "local", "udp", kClientAddr);
>-  EXPECT_PRED4(HasCandidate, candidates_, "local", "udp", kClientAddr2);
>+  EXPECT_TRUE(HasCandidate(candidates_, "local", "udp", kClientIPv6Addr));
>+  EXPECT_TRUE(HasCandidate(candidates_, "local", "udp", kClientAddr));
>+  EXPECT_TRUE(HasCandidate(candidates_, "local", "udp", kClientAddr2));
> }
> 
> // Test that we could use loopback interface as host candidate.
> TEST_F(BasicPortAllocatorTest, TestLoopbackNetworkInterface) {
>   AddInterface(kLoopbackAddr, "test_loopback", rtc::ADAPTER_TYPE_LOOPBACK);
>   allocator_->SetNetworkIgnoreMask(0);
>-  EXPECT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+  ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>   session_->set_flags(PORTALLOCATOR_DISABLE_STUN | PORTALLOCATOR_DISABLE_RELAY |
>                       PORTALLOCATOR_DISABLE_TCP);
>   session_->StartGettingPorts();
>@@ -916,20 +1020,20 @@ TEST_F(BasicPortAllocatorTest, TestLoopbackNetworkInterface) {
> // Tests that we can get all the desired addresses successfully.
> TEST_F(BasicPortAllocatorTest, TestGetAllPortsWithMinimumStepDelay) {
>   AddInterface(kClientAddr);
>-  EXPECT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+  ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>   session_->StartGettingPorts();
>   ASSERT_TRUE_SIMULATED_WAIT(candidate_allocation_done_,
>                              kDefaultAllocationTimeout, fake_clock);
>   EXPECT_EQ(7U, candidates_.size());
>   EXPECT_EQ(4U, ports_.size());
>-  EXPECT_PRED4(HasCandidate, candidates_, "local", "udp", kClientAddr);
>-  EXPECT_PRED4(HasCandidate, candidates_, "stun", "udp", kClientAddr);
>-  EXPECT_PRED4(HasCandidate, candidates_, "relay", "udp", kRelayUdpIntAddr);
>-  EXPECT_PRED4(HasCandidate, candidates_, "relay", "udp", kRelayUdpExtAddr);
>-  EXPECT_PRED4(HasCandidate, candidates_, "relay", "tcp", kRelayTcpIntAddr);
>-  EXPECT_PRED4(HasCandidate, candidates_, "local", "tcp", kClientAddr);
>-  EXPECT_PRED4(HasCandidate, candidates_, "relay", "ssltcp",
>-               kRelaySslTcpIntAddr);
>+  EXPECT_TRUE(HasCandidate(candidates_, "local", "udp", kClientAddr));
>+  EXPECT_TRUE(HasCandidate(candidates_, "stun", "udp", kClientAddr));
>+  EXPECT_TRUE(HasCandidate(candidates_, "relay", "udp", kRelayUdpIntAddr));
>+  EXPECT_TRUE(HasCandidate(candidates_, "relay", "udp", kRelayUdpExtAddr));
>+  EXPECT_TRUE(HasCandidate(candidates_, "relay", "tcp", kRelayTcpIntAddr));
>+  EXPECT_TRUE(HasCandidate(candidates_, "local", "tcp", kClientAddr));
>+  EXPECT_TRUE(
>+      HasCandidate(candidates_, "relay", "ssltcp", kRelaySslTcpIntAddr));
> }
> 
> // Test that when the same network interface is brought down and up, the
>@@ -938,7 +1042,7 @@ TEST_F(BasicPortAllocatorTest, TestGetAllPortsWithMinimumStepDelay) {
> TEST_F(BasicPortAllocatorTest, TestSameNetworkDownAndUpWhenSessionNotStopped) {
>   std::string if_name("test_net0");
>   AddInterface(kClientAddr, if_name);
>-  EXPECT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+  ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>   session_->StartGettingPorts();
>   ASSERT_TRUE_SIMULATED_WAIT(candidate_allocation_done_,
>                              kDefaultAllocationTimeout, fake_clock);
>@@ -975,7 +1079,7 @@ TEST_F(BasicPortAllocatorTest, TestSameNetworkDownAndUpWhenSessionNotStopped) {
> TEST_F(BasicPortAllocatorTest, TestSameNetworkDownAndUpWhenSessionStopped) {
>   std::string if_name("test_net0");
>   AddInterface(kClientAddr, if_name);
>-  EXPECT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+  ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>   session_->StartGettingPorts();
>   ASSERT_TRUE_SIMULATED_WAIT(candidate_allocation_done_,
>                              kDefaultAllocationTimeout, fake_clock);
>@@ -1013,7 +1117,7 @@ TEST_F(BasicPortAllocatorTest, CandidatesRegatheredAfterBindingFails) {
>   AddInterface(kClientAddr, if_name);
>   fss_->set_tcp_sockets_enabled(false);
>   fss_->set_udp_sockets_enabled(false);
>-  EXPECT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+  ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>   session_->StartGettingPorts();
>   ASSERT_TRUE_SIMULATED_WAIT(candidate_allocation_done_,
>                              kDefaultAllocationTimeout, fake_clock);
>@@ -1021,7 +1125,7 @@ TEST_F(BasicPortAllocatorTest, CandidatesRegatheredAfterBindingFails) {
>   // a single TCP active candidate, since that doesn't require creating a
>   // socket).
>   ASSERT_EQ(1U, candidates_.size());
>-  EXPECT_PRED4(HasCandidate, candidates_, "local", "tcp", kClientAddr);
>+  EXPECT_TRUE(HasCandidate(candidates_, "local", "tcp", kClientAddr));
>   candidate_allocation_done_ = false;
> 
>   // Now simulate the interface coming up, with the newfound ability to bind
>@@ -1033,30 +1137,35 @@ TEST_F(BasicPortAllocatorTest, CandidatesRegatheredAfterBindingFails) {
>                              kDefaultAllocationTimeout, fake_clock);
>   // Should get UDP and TCP candidate.
>   ASSERT_EQ(2U, candidates_.size());
>-  EXPECT_PRED4(HasCandidate, candidates_, "local", "udp", kClientAddr);
>+  EXPECT_TRUE(HasCandidate(candidates_, "local", "udp", kClientAddr));
>   // TODO(deadbeef): This is actually the same active TCP candidate as before.
>   // We should extend this test to also verify that a server candidate is
>   // gathered.
>-  EXPECT_PRED4(HasCandidate, candidates_, "local", "tcp", kClientAddr);
>+  EXPECT_TRUE(HasCandidate(candidates_, "local", "tcp", kClientAddr));
> }
> 
> // Verify candidates with default step delay of 1sec.
> TEST_F(BasicPortAllocatorTest, TestGetAllPortsWithOneSecondStepDelay) {
>   AddInterface(kClientAddr);
>   allocator_->set_step_delay(kDefaultStepDelay);
>-  EXPECT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+  ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>   session_->StartGettingPorts();
>+  // Host and STUN candidates from kClientAddr.
>   ASSERT_EQ_SIMULATED_WAIT(2U, candidates_.size(), 1000, fake_clock);
>+  // UDP and STUN ports on kClientAddr.
>   EXPECT_EQ(2U, ports_.size());
>+  // Host, STUN and relay candidates from kClientAddr.
>   ASSERT_EQ_SIMULATED_WAIT(6U, candidates_.size(), 2000, fake_clock);
>+  // UDP, STUN and relay ports on kClientAddr.
>   EXPECT_EQ(3U, ports_.size());
>-  EXPECT_PRED4(HasCandidate, candidates_, "relay", "udp", kRelayUdpIntAddr);
>-  EXPECT_PRED4(HasCandidate, candidates_, "relay", "udp", kRelayUdpExtAddr);
>-  EXPECT_PRED4(HasCandidate, candidates_, "relay", "tcp", kRelayTcpIntAddr);
>-  EXPECT_PRED4(HasCandidate, candidates_, "relay", "ssltcp",
>-               kRelaySslTcpIntAddr);
>+  EXPECT_TRUE(HasCandidate(candidates_, "relay", "udp", kRelayUdpIntAddr));
>+  EXPECT_TRUE(HasCandidate(candidates_, "relay", "udp", kRelayUdpExtAddr));
>+  EXPECT_TRUE(HasCandidate(candidates_, "relay", "tcp", kRelayTcpIntAddr));
>+  EXPECT_TRUE(
>+      HasCandidate(candidates_, "relay", "ssltcp", kRelaySslTcpIntAddr));
>+  // One more TCP candidate from kClientAddr.
>   ASSERT_EQ_SIMULATED_WAIT(7U, candidates_.size(), 1500, fake_clock);
>-  EXPECT_PRED4(HasCandidate, candidates_, "local", "tcp", kClientAddr);
>+  EXPECT_TRUE(HasCandidate(candidates_, "local", "tcp", kClientAddr));
>   EXPECT_EQ(4U, ports_.size());
>   EXPECT_TRUE(candidate_allocation_done_);
>   // If we Stop gathering now, we shouldn't get a second "done" callback.
>@@ -1065,7 +1174,7 @@ TEST_F(BasicPortAllocatorTest, TestGetAllPortsWithOneSecondStepDelay) {
> 
> TEST_F(BasicPortAllocatorTest, TestSetupVideoRtpPortsWithNormalSendBuffers) {
>   AddInterface(kClientAddr);
>-  EXPECT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP, CN_VIDEO));
>+  ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP, CN_VIDEO));
>   session_->StartGettingPorts();
>   ASSERT_TRUE_SIMULATED_WAIT(candidate_allocation_done_,
>                              kDefaultAllocationTimeout, fake_clock);
>@@ -1081,7 +1190,7 @@ TEST_F(BasicPortAllocatorTest, TestSetupVideoRtpPortsWithNormalSendBuffers) {
> // middle of gathering.
> TEST_F(BasicPortAllocatorTest, TestStopGetAllPorts) {
>   AddInterface(kClientAddr);
>-  EXPECT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+  ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>   session_->StartGettingPorts();
>   ASSERT_EQ_SIMULATED_WAIT(2U, candidates_.size(), kDefaultAllocationTimeout,
>                            fake_clock);
>@@ -1102,7 +1211,7 @@ TEST_F(BasicPortAllocatorTest, TestGetAllPortsPortRange) {
>   EXPECT_TRUE(SetPortRange(0, 0));
>   // Check that a valid port range succeeds.
>   EXPECT_TRUE(SetPortRange(kMinPort, kMaxPort));
>-  EXPECT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+  ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>   session_->StartGettingPorts();
>   ASSERT_TRUE_SIMULATED_WAIT(candidate_allocation_done_,
>                              kDefaultAllocationTimeout, fake_clock);
>@@ -1113,14 +1222,14 @@ TEST_F(BasicPortAllocatorTest, TestGetAllPortsPortRange) {
>   for (const Candidate& candidate : candidates_) {
>     // Check the port number for the UDP/STUN/TCP port objects.
>     if (candidate.type() != RELAY_PORT_TYPE) {
>-      EXPECT_PRED3(CheckPort, candidate.address(), kMinPort, kMaxPort);
>+      EXPECT_TRUE(CheckPort(candidate.address(), kMinPort, kMaxPort));
>       ++num_nonrelay_candidates;
>     }
>   }
>   EXPECT_EQ(3, num_nonrelay_candidates);
>   // Check the port number used to connect to the relay server.
>-  EXPECT_PRED3(CheckPort, relay_server_.GetConnection(0).source(), kMinPort,
>-               kMaxPort);
>+  EXPECT_TRUE(
>+      CheckPort(relay_server_.GetConnection(0).source(), kMinPort, kMaxPort));
> }
> 
> // Test that if we have no network adapters, we bind to the ANY address and
>@@ -1136,7 +1245,7 @@ TEST_F(BasicPortAllocatorTest, TestGetAllPortsNoAdapters) {
>   // behind a NAT but IPv6 not, or having an IPv6 NAT.
>   // TODO(deadbeef): Fix this.
>   network_manager_.set_ipv6_enabled(false);
>-  EXPECT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+  ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>   session_->StartGettingPorts();
>   EXPECT_TRUE_SIMULATED_WAIT(candidate_allocation_done_,
>                              kDefaultAllocationTimeout, fake_clock);
>@@ -1150,8 +1259,8 @@ TEST_F(BasicPortAllocatorTest, TestGetAllPortsNoAdapters) {
>   // candidate for it is useless and shouldn't be signaled. So we only have
>   // STUN/TURN candidates.
>   EXPECT_EQ(3U, candidates_.size());
>-  EXPECT_PRED4(HasCandidate, candidates_, "stun", "udp",
>-               rtc::SocketAddress(kNatUdpAddr.ipaddr(), 0));
>+  EXPECT_TRUE(HasCandidate(candidates_, "stun", "udp",
>+                           rtc::SocketAddress(kNatUdpAddr.ipaddr(), 0)));
>   // Again, two TURN candidates, using UDP/TCP for the first hop to the TURN
>   // server.
>   EXPECT_EQ(2,
>@@ -1183,7 +1292,7 @@ TEST_F(BasicPortAllocatorTest,
>   // Enable IPv6 here. Since the network_manager doesn't have IPv6 default
>   // address set and we have no IPv6 STUN server, there should be no IPv6
>   // candidates.
>-  EXPECT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+  ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>   session_->set_flags(PORTALLOCATOR_ENABLE_IPV6);
> 
>   // Expect to see 3 ports for IPv4: HOST/STUN, TURN/UDP and TCP ports, 2 ports
>@@ -1224,7 +1333,7 @@ TEST_F(BasicPortAllocatorTest,
> TEST_F(BasicPortAllocatorTest,
>        TestDisableAdapterEnumerationWithoutNatLocalhostCandidateDisabled) {
>   ResetWithStunServerNoNat(kStunAddr);
>-  EXPECT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+  ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>   session_->set_flags(PORTALLOCATOR_DISABLE_DEFAULT_LOCAL_CANDIDATE);
>   // Expect to see 2 ports: STUN and TCP ports, localhost candidate and STUN
>   // candidate.
>@@ -1237,12 +1346,11 @@ TEST_F(BasicPortAllocatorTest,
> // a NAT, there is no local candidate. However, this specified default route
> // (kClientAddr) which was discovered when sending STUN requests, will become
> // the srflx addresses.
>-TEST_F(
>-    BasicPortAllocatorTest,
>-    TestDisableAdapterEnumerationWithoutNatLocalhostCandDisabledDiffRoute) {
>+TEST_F(BasicPortAllocatorTest,
>+       TestDisableAdapterEnumerationWithoutNatLocalhostCandDisabledDiffRoute) {
>   ResetWithStunServerNoNat(kStunAddr);
>   AddInterfaceAsDefaultRoute(kClientAddr);
>-  EXPECT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+  ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>   session_->set_flags(PORTALLOCATOR_DISABLE_DEFAULT_LOCAL_CANDIDATE);
>   // Expect to see 2 ports: STUN and TCP ports, localhost candidate and STUN
>   // candidate.
>@@ -1256,7 +1364,7 @@ TEST_F(
> TEST_F(BasicPortAllocatorTest,
>        TestDisableAdapterEnumerationWithNatLocalhostCandidateDisabled) {
>   ResetWithStunServerAndNat(kStunAddr);
>-  EXPECT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+  ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>   session_->set_flags(PORTALLOCATOR_DISABLE_DEFAULT_LOCAL_CANDIDATE);
>   // Expect to see 2 ports: STUN and TCP ports, and single STUN candidate.
>   CheckDisableAdapterEnumeration(2U, rtc::IPAddress(), kNatUdpAddr.ipaddr(),
>@@ -1270,7 +1378,7 @@ TEST_F(BasicPortAllocatorTest, TestDisableUdpTurn) {
>   AddInterface(kClientAddr);
>   ResetWithStunServerAndNat(kStunAddr);
>   AddTurnServers(kTurnUdpIntAddr, kTurnTcpIntAddr);
>-  EXPECT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+  ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>   session_->set_flags(PORTALLOCATOR_DISABLE_UDP_RELAY |
>                       PORTALLOCATOR_DISABLE_UDP | PORTALLOCATOR_DISABLE_STUN |
>                       PORTALLOCATOR_ENABLE_SHARED_SOCKET);
>@@ -1284,11 +1392,11 @@ TEST_F(BasicPortAllocatorTest, TestDisableUdpTurn) {
>   EXPECT_EQ(2U, ports_.size());
>   EXPECT_EQ(2U, candidates_.size());
>   Candidate turn_candidate;
>-  EXPECT_PRED5(FindCandidate, candidates_, "relay", "udp", kTurnUdpExtAddr,
>-               &turn_candidate);
>+  EXPECT_TRUE(FindCandidate(candidates_, "relay", "udp", kTurnUdpExtAddr,
>+                            &turn_candidate));
>   // The TURN candidate should use TCP to contact the TURN server.
>   EXPECT_EQ(TCP_PROTOCOL_NAME, turn_candidate.relay_protocol());
>-  EXPECT_PRED4(HasCandidate, candidates_, "local", "tcp", kClientAddr);
>+  EXPECT_TRUE(HasCandidate(candidates_, "local", "tcp", kClientAddr));
> }
> 
> // Disable for asan, see
>@@ -1299,7 +1407,7 @@ TEST_F(BasicPortAllocatorTest, TestDisableUdpTurn) {
> // are disabled.
> TEST_F(BasicPortAllocatorTest, TestDisableAllPorts) {
>   AddInterface(kClientAddr);
>-  EXPECT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+  ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>   session_->set_flags(PORTALLOCATOR_DISABLE_UDP | PORTALLOCATOR_DISABLE_STUN |
>                       PORTALLOCATOR_DISABLE_RELAY | PORTALLOCATOR_DISABLE_TCP);
>   session_->StartGettingPorts();
>@@ -1311,18 +1419,18 @@ TEST_F(BasicPortAllocatorTest, TestDisableAllPorts) {
> TEST_F(BasicPortAllocatorTest, TestGetAllPortsNoUdpSockets) {
>   AddInterface(kClientAddr);
>   fss_->set_udp_sockets_enabled(false);
>-  EXPECT_TRUE(CreateSession(1));
>+  ASSERT_TRUE(CreateSession(1));
>   session_->StartGettingPorts();
>   ASSERT_TRUE_SIMULATED_WAIT(candidate_allocation_done_,
>                              kDefaultAllocationTimeout, fake_clock);
>   EXPECT_EQ(5U, candidates_.size());
>   EXPECT_EQ(2U, ports_.size());
>-  EXPECT_PRED4(HasCandidate, candidates_, "relay", "udp", kRelayUdpIntAddr);
>-  EXPECT_PRED4(HasCandidate, candidates_, "relay", "udp", kRelayUdpExtAddr);
>-  EXPECT_PRED4(HasCandidate, candidates_, "relay", "tcp", kRelayTcpIntAddr);
>-  EXPECT_PRED4(HasCandidate, candidates_, "local", "tcp", kClientAddr);
>-  EXPECT_PRED4(HasCandidate, candidates_, "relay", "ssltcp",
>-               kRelaySslTcpIntAddr);
>+  EXPECT_TRUE(HasCandidate(candidates_, "relay", "udp", kRelayUdpIntAddr));
>+  EXPECT_TRUE(HasCandidate(candidates_, "relay", "udp", kRelayUdpExtAddr));
>+  EXPECT_TRUE(HasCandidate(candidates_, "relay", "tcp", kRelayTcpIntAddr));
>+  EXPECT_TRUE(HasCandidate(candidates_, "local", "tcp", kClientAddr));
>+  EXPECT_TRUE(
>+      HasCandidate(candidates_, "relay", "ssltcp", kRelaySslTcpIntAddr));
> }
> 
> #endif  // if !defined(ADDRESS_SANITIZER)
>@@ -1334,18 +1442,18 @@ TEST_F(BasicPortAllocatorTest, TestGetAllPortsNoUdpSocketsNoTcpListen) {
>   AddInterface(kClientAddr);
>   fss_->set_udp_sockets_enabled(false);
>   fss_->set_tcp_listen_enabled(false);
>-  EXPECT_TRUE(CreateSession(1));
>+  ASSERT_TRUE(CreateSession(1));
>   session_->StartGettingPorts();
>   ASSERT_TRUE_SIMULATED_WAIT(candidate_allocation_done_,
>                              kDefaultAllocationTimeout, fake_clock);
>   EXPECT_EQ(5U, candidates_.size());
>   EXPECT_EQ(2U, ports_.size());
>-  EXPECT_PRED4(HasCandidate, candidates_, "relay", "udp", kRelayUdpIntAddr);
>-  EXPECT_PRED4(HasCandidate, candidates_, "relay", "udp", kRelayUdpExtAddr);
>-  EXPECT_PRED4(HasCandidate, candidates_, "relay", "tcp", kRelayTcpIntAddr);
>-  EXPECT_PRED4(HasCandidate, candidates_, "local", "tcp", kClientAddr);
>-  EXPECT_PRED4(HasCandidate, candidates_, "relay", "ssltcp",
>-               kRelaySslTcpIntAddr);
>+  EXPECT_TRUE(HasCandidate(candidates_, "relay", "udp", kRelayUdpIntAddr));
>+  EXPECT_TRUE(HasCandidate(candidates_, "relay", "udp", kRelayUdpExtAddr));
>+  EXPECT_TRUE(HasCandidate(candidates_, "relay", "tcp", kRelayTcpIntAddr));
>+  EXPECT_TRUE(HasCandidate(candidates_, "local", "tcp", kClientAddr));
>+  EXPECT_TRUE(
>+      HasCandidate(candidates_, "relay", "ssltcp", kRelaySslTcpIntAddr));
> }
> 
> // Test that we don't crash or malfunction if we can't create any sockets.
>@@ -1354,7 +1462,7 @@ TEST_F(BasicPortAllocatorTest, TestGetAllPortsNoSockets) {
>   AddInterface(kClientAddr);
>   fss_->set_tcp_sockets_enabled(false);
>   fss_->set_udp_sockets_enabled(false);
>-  EXPECT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+  ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>   session_->StartGettingPorts();
>   SIMULATED_WAIT(candidates_.size() > 0, 2000, fake_clock);
>   // TODO(deadbeef): Check candidate_allocation_done signal.
>@@ -1365,23 +1473,26 @@ TEST_F(BasicPortAllocatorTest, TestGetAllPortsNoSockets) {
> // Testing STUN timeout.
> TEST_F(BasicPortAllocatorTest, TestGetAllPortsNoUdpAllowed) {
>   fss_->AddRule(false, rtc::FP_UDP, rtc::FD_ANY, kClientAddr);
>+  fss_->AddRule(false, rtc::FP_UDP, rtc::FD_ANY, kAnyAddr);
>   AddInterface(kClientAddr);
>-  EXPECT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+  ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>   session_->StartGettingPorts();
>   EXPECT_EQ_SIMULATED_WAIT(2U, candidates_.size(), kDefaultAllocationTimeout,
>                            fake_clock);
>+  // UDP and TCP ports on kClientAddr.
>   EXPECT_EQ(2U, ports_.size());
>-  EXPECT_PRED4(HasCandidate, candidates_, "local", "udp", kClientAddr);
>-  EXPECT_PRED4(HasCandidate, candidates_, "local", "tcp", kClientAddr);
>+  EXPECT_TRUE(HasCandidate(candidates_, "local", "udp", kClientAddr));
>+  EXPECT_TRUE(HasCandidate(candidates_, "local", "tcp", kClientAddr));
>   // RelayPort connection timeout is 3sec. TCP connection with RelayServer
>   // will be tried after about 3 seconds.
>   EXPECT_EQ_SIMULATED_WAIT(6U, candidates_.size(), 3500, fake_clock);
>+  // UDP, TCP and relay ports on kClientAddr.
>   EXPECT_EQ(3U, ports_.size());
>-  EXPECT_PRED4(HasCandidate, candidates_, "relay", "udp", kRelayUdpIntAddr);
>-  EXPECT_PRED4(HasCandidate, candidates_, "relay", "tcp", kRelayTcpIntAddr);
>-  EXPECT_PRED4(HasCandidate, candidates_, "relay", "ssltcp",
>-               kRelaySslTcpIntAddr);
>-  EXPECT_PRED4(HasCandidate, candidates_, "relay", "udp", kRelayUdpExtAddr);
>+  EXPECT_TRUE(HasCandidate(candidates_, "relay", "udp", kRelayUdpIntAddr));
>+  EXPECT_TRUE(HasCandidate(candidates_, "relay", "tcp", kRelayTcpIntAddr));
>+  EXPECT_TRUE(
>+      HasCandidate(candidates_, "relay", "ssltcp", kRelaySslTcpIntAddr));
>+  EXPECT_TRUE(HasCandidate(candidates_, "relay", "udp", kRelayUdpExtAddr));
>   // We wait at least for a full STUN timeout, which
>   // cricket::STUN_TOTAL_TIMEOUT seconds.  But since 3-3.5 seconds
>   // already passed (see above), we wait 3 seconds less than that.
>@@ -1396,7 +1507,7 @@ TEST_F(BasicPortAllocatorTest, TestCandidatePriorityOfMultipleInterfaces) {
>   // convenience.
>   allocator().set_flags(PORTALLOCATOR_DISABLE_TCP | PORTALLOCATOR_DISABLE_STUN |
>                         PORTALLOCATOR_DISABLE_RELAY);
>-  EXPECT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+  ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>   session_->StartGettingPorts();
>   EXPECT_TRUE_SIMULATED_WAIT(candidate_allocation_done_,
>                              kDefaultAllocationTimeout, fake_clock);
>@@ -1409,7 +1520,7 @@ TEST_F(BasicPortAllocatorTest, TestCandidatePriorityOfMultipleInterfaces) {
> // Test to verify ICE restart process.
> TEST_F(BasicPortAllocatorTest, TestGetAllPortsRestarts) {
>   AddInterface(kClientAddr);
>-  EXPECT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+  ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>   session_->StartGettingPorts();
>   EXPECT_TRUE_SIMULATED_WAIT(candidate_allocation_done_,
>                              kDefaultAllocationTimeout, fake_clock);
>@@ -1424,7 +1535,7 @@ TEST_F(BasicPortAllocatorTest, TestGetAllPortsRestarts) {
> // according to JSEP, which means the *next* allocator session returned.
> TEST_F(BasicPortAllocatorTest, TestSessionUsesOwnCandidateFilter) {
>   AddInterface(kClientAddr);
>-  EXPECT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+  ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>   // Set candidate filter *after* creating the session. Should have no effect.
>   allocator().set_candidate_filter(CF_RELAY);
>   session_->StartGettingPorts();
>@@ -1446,12 +1557,12 @@ TEST_F(BasicPortAllocatorTest, TestCandidateFilterWithRelayOnly) {
>   // GTURN is not configured here.
>   ResetWithTurnServersNoNat(kTurnUdpIntAddr, rtc::SocketAddress());
>   allocator().set_candidate_filter(CF_RELAY);
>-  EXPECT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+  ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>   session_->StartGettingPorts();
>   EXPECT_TRUE_SIMULATED_WAIT(candidate_allocation_done_,
>                              kDefaultAllocationTimeout, fake_clock);
>-  EXPECT_PRED4(HasCandidate, candidates_, "relay", "udp",
>-               rtc::SocketAddress(kTurnUdpExtAddr.ipaddr(), 0));
>+  EXPECT_TRUE(HasCandidate(candidates_, "relay", "udp",
>+                           rtc::SocketAddress(kTurnUdpExtAddr.ipaddr(), 0)));
> 
>   EXPECT_EQ(1U, candidates_.size());
>   EXPECT_EQ(1U, ports_.size());  // Only Relay port will be in ready state.
>@@ -1465,7 +1576,7 @@ TEST_F(BasicPortAllocatorTest, TestCandidateFilterWithHostOnly) {
>   AddInterface(kClientAddr);
>   allocator().set_flags(PORTALLOCATOR_ENABLE_SHARED_SOCKET);
>   allocator().set_candidate_filter(CF_HOST);
>-  EXPECT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+  ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>   session_->StartGettingPorts();
>   EXPECT_TRUE_SIMULATED_WAIT(candidate_allocation_done_,
>                              kDefaultAllocationTimeout, fake_clock);
>@@ -1483,7 +1594,7 @@ TEST_F(BasicPortAllocatorTest, TestCandidateFilterWithReflexiveOnly) {
> 
>   allocator().set_flags(PORTALLOCATOR_ENABLE_SHARED_SOCKET);
>   allocator().set_candidate_filter(CF_REFLEXIVE);
>-  EXPECT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+  ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>   session_->StartGettingPorts();
>   EXPECT_TRUE_SIMULATED_WAIT(candidate_allocation_done_,
>                              kDefaultAllocationTimeout, fake_clock);
>@@ -1502,7 +1613,7 @@ TEST_F(BasicPortAllocatorTest, TestCandidateFilterWithReflexiveOnlyAndNoNAT) {
>   AddInterface(kClientAddr);
>   allocator().set_flags(PORTALLOCATOR_ENABLE_SHARED_SOCKET);
>   allocator().set_candidate_filter(CF_REFLEXIVE);
>-  EXPECT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+  ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>   session_->StartGettingPorts();
>   EXPECT_TRUE_SIMULATED_WAIT(candidate_allocation_done_,
>                              kDefaultAllocationTimeout, fake_clock);
>@@ -1517,14 +1628,14 @@ TEST_F(BasicPortAllocatorTest, TestCandidateFilterWithReflexiveOnlyAndNoNAT) {
> // Test that we get the same ufrag and pwd for all candidates.
> TEST_F(BasicPortAllocatorTest, TestEnableSharedUfrag) {
>   AddInterface(kClientAddr);
>-  EXPECT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+  ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>   session_->StartGettingPorts();
>   ASSERT_TRUE_SIMULATED_WAIT(candidate_allocation_done_,
>                              kDefaultAllocationTimeout, fake_clock);
>   EXPECT_EQ(7U, candidates_.size());
>-  EXPECT_PRED4(HasCandidate, candidates_, "local", "udp", kClientAddr);
>-  EXPECT_PRED4(HasCandidate, candidates_, "stun", "udp", kClientAddr);
>-  EXPECT_PRED4(HasCandidate, candidates_, "local", "tcp", kClientAddr);
>+  EXPECT_TRUE(HasCandidate(candidates_, "local", "udp", kClientAddr));
>+  EXPECT_TRUE(HasCandidate(candidates_, "stun", "udp", kClientAddr));
>+  EXPECT_TRUE(HasCandidate(candidates_, "local", "tcp", kClientAddr));
>   EXPECT_EQ(4U, ports_.size());
>   for (const Candidate& candidate : candidates_) {
>     EXPECT_EQ(kIceUfrag0, candidate.username());
>@@ -1540,12 +1651,12 @@ TEST_F(BasicPortAllocatorTest, TestSharedSocketWithoutNat) {
>   AddInterface(kClientAddr);
>   allocator_->set_flags(allocator().flags() |
>                         PORTALLOCATOR_ENABLE_SHARED_SOCKET);
>-  EXPECT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+  ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>   session_->StartGettingPorts();
>   ASSERT_EQ_SIMULATED_WAIT(6U, candidates_.size(), kDefaultAllocationTimeout,
>                            fake_clock);
>   EXPECT_EQ(3U, ports_.size());
>-  EXPECT_PRED4(HasCandidate, candidates_, "local", "udp", kClientAddr);
>+  EXPECT_TRUE(HasCandidate(candidates_, "local", "udp", kClientAddr));
>   EXPECT_TRUE_SIMULATED_WAIT(candidate_allocation_done_,
>                              kDefaultAllocationTimeout, fake_clock);
> }
>@@ -1559,14 +1670,14 @@ TEST_F(BasicPortAllocatorTest, TestSharedSocketWithNat) {
> 
>   allocator_->set_flags(allocator().flags() |
>                         PORTALLOCATOR_ENABLE_SHARED_SOCKET);
>-  EXPECT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+  ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>   session_->StartGettingPorts();
>   ASSERT_EQ_SIMULATED_WAIT(3U, candidates_.size(), kDefaultAllocationTimeout,
>                            fake_clock);
>   ASSERT_EQ(2U, ports_.size());
>-  EXPECT_PRED4(HasCandidate, candidates_, "local", "udp", kClientAddr);
>-  EXPECT_PRED4(HasCandidate, candidates_, "stun", "udp",
>-               rtc::SocketAddress(kNatUdpAddr.ipaddr(), 0));
>+  EXPECT_TRUE(HasCandidate(candidates_, "local", "udp", kClientAddr));
>+  EXPECT_TRUE(HasCandidate(candidates_, "stun", "udp",
>+                           rtc::SocketAddress(kNatUdpAddr.ipaddr(), 0)));
>   EXPECT_TRUE_SIMULATED_WAIT(candidate_allocation_done_,
>                              kDefaultAllocationTimeout, fake_clock);
>   EXPECT_EQ(3U, candidates_.size());
>@@ -1577,6 +1688,7 @@ TEST_F(BasicPortAllocatorTest, TestSharedSocketWithoutNatUsingTurn) {
>   turn_server_.AddInternalSocket(kTurnTcpIntAddr, PROTO_TCP);
>   AddInterface(kClientAddr);
>   allocator_.reset(new BasicPortAllocator(&network_manager_));
>+  allocator_->Initialize();
> 
>   AddTurnServers(kTurnUdpIntAddr, kTurnTcpIntAddr);
> 
>@@ -1585,18 +1697,18 @@ TEST_F(BasicPortAllocatorTest, TestSharedSocketWithoutNatUsingTurn) {
>                         PORTALLOCATOR_ENABLE_SHARED_SOCKET |
>                         PORTALLOCATOR_DISABLE_TCP);
> 
>-  EXPECT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+  ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>   session_->StartGettingPorts();
> 
>   ASSERT_TRUE_SIMULATED_WAIT(candidate_allocation_done_,
>                              kDefaultAllocationTimeout, fake_clock);
>   ASSERT_EQ(3U, candidates_.size());
>   ASSERT_EQ(3U, ports_.size());
>-  EXPECT_PRED4(HasCandidate, candidates_, "local", "udp", kClientAddr);
>-  EXPECT_PRED4(HasCandidate, candidates_, "relay", "udp",
>-               rtc::SocketAddress(kTurnUdpExtAddr.ipaddr(), 0));
>-  EXPECT_PRED4(HasCandidate, candidates_, "relay", "udp",
>-               rtc::SocketAddress(kTurnUdpExtAddr.ipaddr(), 0));
>+  EXPECT_TRUE(HasCandidate(candidates_, "local", "udp", kClientAddr));
>+  EXPECT_TRUE(HasCandidate(candidates_, "relay", "udp",
>+                           rtc::SocketAddress(kTurnUdpExtAddr.ipaddr(), 0)));
>+  EXPECT_TRUE(HasCandidate(candidates_, "relay", "udp",
>+                           rtc::SocketAddress(kTurnUdpExtAddr.ipaddr(), 0)));
> }
> 
> // Test that if prune_turn_ports is set, TCP TURN port will not be used
>@@ -1681,6 +1793,7 @@ TEST_F(BasicPortAllocatorTestWithRealClock,
>                                  PROTO_UDP);
>   AddInterface(kClientAddr);
>   allocator_.reset(new BasicPortAllocator(&network_manager_));
>+  allocator_->Initialize();
>   RelayServerConfig turn_server(RELAY_TURN);
>   RelayCredentials credentials(kTurnUsername, kTurnPassword);
>   turn_server.credentials = credentials;
>@@ -1693,7 +1806,7 @@ TEST_F(BasicPortAllocatorTestWithRealClock,
>                         PORTALLOCATOR_ENABLE_SHARED_SOCKET |
>                         PORTALLOCATOR_DISABLE_TCP);
> 
>-  EXPECT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+  ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>   session_->StartGettingPorts();
> 
>   EXPECT_EQ_WAIT(2U, ports_.size(), kDefaultAllocationTimeout);
>@@ -1712,18 +1825,18 @@ TEST_F(BasicPortAllocatorTest, TestSharedSocketWithNatUsingTurn) {
>                         PORTALLOCATOR_ENABLE_SHARED_SOCKET |
>                         PORTALLOCATOR_DISABLE_TCP);
> 
>-  EXPECT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+  ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>   session_->StartGettingPorts();
> 
>   ASSERT_TRUE_SIMULATED_WAIT(candidate_allocation_done_,
>                              kDefaultAllocationTimeout, fake_clock);
>   EXPECT_EQ(3U, candidates_.size());
>   ASSERT_EQ(2U, ports_.size());
>-  EXPECT_PRED4(HasCandidate, candidates_, "local", "udp", kClientAddr);
>-  EXPECT_PRED4(HasCandidate, candidates_, "stun", "udp",
>-               rtc::SocketAddress(kNatUdpAddr.ipaddr(), 0));
>-  EXPECT_PRED4(HasCandidate, candidates_, "relay", "udp",
>-               rtc::SocketAddress(kTurnUdpExtAddr.ipaddr(), 0));
>+  EXPECT_TRUE(HasCandidate(candidates_, "local", "udp", kClientAddr));
>+  EXPECT_TRUE(HasCandidate(candidates_, "stun", "udp",
>+                           rtc::SocketAddress(kNatUdpAddr.ipaddr(), 0)));
>+  EXPECT_TRUE(HasCandidate(candidates_, "relay", "udp",
>+                           rtc::SocketAddress(kTurnUdpExtAddr.ipaddr(), 0)));
>   EXPECT_TRUE_SIMULATED_WAIT(candidate_allocation_done_,
>                              kDefaultAllocationTimeout, fake_clock);
>   // Local port will be created first and then TURN port.
>@@ -1751,19 +1864,21 @@ TEST_F(BasicPortAllocatorTest, TestSharedSocketWithNatUsingTurnAsStun) {
>                         PORTALLOCATOR_ENABLE_SHARED_SOCKET |
>                         PORTALLOCATOR_DISABLE_TCP);
> 
>-  EXPECT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+  ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>   session_->StartGettingPorts();
> 
>   ASSERT_TRUE_SIMULATED_WAIT(candidate_allocation_done_,
>                              kDefaultAllocationTimeout, fake_clock);
>   EXPECT_EQ(3U, candidates_.size());
>-  EXPECT_PRED4(HasCandidate, candidates_, "local", "udp", kClientAddr);
>+  EXPECT_TRUE(HasCandidate(candidates_, "local", "udp", kClientAddr));
>   Candidate stun_candidate;
>-  EXPECT_PRED5(FindCandidate, candidates_, "stun", "udp",
>-               rtc::SocketAddress(kNatUdpAddr.ipaddr(), 0), &stun_candidate);
>-  EXPECT_PRED5(HasCandidateWithRelatedAddr, candidates_, "relay", "udp",
>-               rtc::SocketAddress(kTurnUdpExtAddr.ipaddr(), 0),
>-               stun_candidate.address());
>+  EXPECT_TRUE(FindCandidate(candidates_, "stun", "udp",
>+                            rtc::SocketAddress(kNatUdpAddr.ipaddr(), 0),
>+                            &stun_candidate));
>+  EXPECT_TRUE(HasCandidateWithRelatedAddr(
>+      candidates_, "relay", "udp",
>+      rtc::SocketAddress(kTurnUdpExtAddr.ipaddr(), 0),
>+      stun_candidate.address()));
> 
>   // Local port will be created first and then TURN port.
>   // TODO(deadbeef): This isn't something the BasicPortAllocator API contract
>@@ -1785,16 +1900,16 @@ TEST_F(BasicPortAllocatorTest, TestSharedSocketWithNatUsingTurnTcpOnly) {
>                         PORTALLOCATOR_ENABLE_SHARED_SOCKET |
>                         PORTALLOCATOR_DISABLE_TCP);
> 
>-  EXPECT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+  ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>   session_->StartGettingPorts();
> 
>   ASSERT_TRUE_SIMULATED_WAIT(candidate_allocation_done_,
>                              kDefaultAllocationTimeout, fake_clock);
>   EXPECT_EQ(2U, candidates_.size());
>   ASSERT_EQ(2U, ports_.size());
>-  EXPECT_PRED4(HasCandidate, candidates_, "local", "udp", kClientAddr);
>-  EXPECT_PRED4(HasCandidate, candidates_, "relay", "udp",
>-               rtc::SocketAddress(kTurnUdpExtAddr.ipaddr(), 0));
>+  EXPECT_TRUE(HasCandidate(candidates_, "local", "udp", kClientAddr));
>+  EXPECT_TRUE(HasCandidate(candidates_, "relay", "udp",
>+                           rtc::SocketAddress(kTurnUdpExtAddr.ipaddr(), 0)));
>   EXPECT_EQ(1U, ports_[0]->Candidates().size());
>   EXPECT_EQ(1U, ports_[1]->Candidates().size());
> }
>@@ -1812,21 +1927,22 @@ TEST_F(BasicPortAllocatorTest, TestNonSharedSocketWithNatUsingTurnAsStun) {
> 
>   allocator_->set_flags(allocator().flags() | PORTALLOCATOR_DISABLE_TCP);
> 
>-  EXPECT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+  ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>   session_->StartGettingPorts();
> 
>   ASSERT_TRUE_SIMULATED_WAIT(candidate_allocation_done_,
>                              kDefaultAllocationTimeout, fake_clock);
>   EXPECT_EQ(3U, candidates_.size());
>   ASSERT_EQ(3U, ports_.size());
>-  EXPECT_PRED4(HasCandidate, candidates_, "local", "udp", kClientAddr);
>+  EXPECT_TRUE(HasCandidate(candidates_, "local", "udp", kClientAddr));
>   Candidate stun_candidate;
>-  EXPECT_PRED5(FindCandidate, candidates_, "stun", "udp",
>-               rtc::SocketAddress(kNatUdpAddr.ipaddr(), 0), &stun_candidate);
>+  EXPECT_TRUE(FindCandidate(candidates_, "stun", "udp",
>+                            rtc::SocketAddress(kNatUdpAddr.ipaddr(), 0),
>+                            &stun_candidate));
>   Candidate turn_candidate;
>-  EXPECT_PRED5(FindCandidate, candidates_, "relay", "udp",
>-               rtc::SocketAddress(kTurnUdpExtAddr.ipaddr(), 0),
>-               &turn_candidate);
>+  EXPECT_TRUE(FindCandidate(candidates_, "relay", "udp",
>+                            rtc::SocketAddress(kTurnUdpExtAddr.ipaddr(), 0),
>+                            &turn_candidate));
>   // Not using shared socket, so the STUN request's server reflexive address
>   // should be different than the TURN request's server reflexive address.
>   EXPECT_NE(turn_candidate.related_address(), stun_candidate.address());
>@@ -1850,18 +1966,20 @@ TEST_F(BasicPortAllocatorTest, TestSharedSocketWithNatUsingTurnAndStun) {
>                         PORTALLOCATOR_ENABLE_SHARED_SOCKET |
>                         PORTALLOCATOR_DISABLE_TCP);
> 
>-  EXPECT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+  ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>   session_->StartGettingPorts();
> 
>   ASSERT_EQ_SIMULATED_WAIT(3U, candidates_.size(), kDefaultAllocationTimeout,
>                            fake_clock);
>-  EXPECT_PRED4(HasCandidate, candidates_, "local", "udp", kClientAddr);
>+  EXPECT_TRUE(HasCandidate(candidates_, "local", "udp", kClientAddr));
>   Candidate stun_candidate;
>-  EXPECT_PRED5(FindCandidate, candidates_, "stun", "udp",
>-               rtc::SocketAddress(kNatUdpAddr.ipaddr(), 0), &stun_candidate);
>-  EXPECT_PRED5(HasCandidateWithRelatedAddr, candidates_, "relay", "udp",
>-               rtc::SocketAddress(kTurnUdpExtAddr.ipaddr(), 0),
>-               stun_candidate.address());
>+  EXPECT_TRUE(FindCandidate(candidates_, "stun", "udp",
>+                            rtc::SocketAddress(kNatUdpAddr.ipaddr(), 0),
>+                            &stun_candidate));
>+  EXPECT_TRUE(HasCandidateWithRelatedAddr(
>+      candidates_, "relay", "udp",
>+      rtc::SocketAddress(kTurnUdpExtAddr.ipaddr(), 0),
>+      stun_candidate.address()));
> 
>   // Don't bother waiting for STUN timeout, since we already verified
>   // that we got a STUN candidate from the TURN server.
>@@ -1875,19 +1993,152 @@ TEST_F(BasicPortAllocatorTest, TestSharedSocketNoUdpAllowed) {
>                         PORTALLOCATOR_DISABLE_TCP |
>                         PORTALLOCATOR_ENABLE_SHARED_SOCKET);
>   fss_->AddRule(false, rtc::FP_UDP, rtc::FD_ANY, kClientAddr);
>+  fss_->AddRule(false, rtc::FP_UDP, rtc::FD_ANY, kAnyAddr);
>   AddInterface(kClientAddr);
>-  EXPECT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+  ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>   session_->StartGettingPorts();
>-  ASSERT_EQ_SIMULATED_WAIT(1U, ports_.size(), kDefaultAllocationTimeout,
>+  ASSERT_EQ_SIMULATED_WAIT(1U, candidates_.size(), kDefaultAllocationTimeout,
>                            fake_clock);
>-  EXPECT_EQ(1U, candidates_.size());
>-  EXPECT_PRED4(HasCandidate, candidates_, "local", "udp", kClientAddr);
>+  // UDP ports on kClientAddr.
>+  EXPECT_EQ(1U, ports_.size());
>+  EXPECT_TRUE(HasCandidate(candidates_, "local", "udp", kClientAddr));
>   // STUN timeout is 9.5sec. We need to wait to get candidate done signal.
>   EXPECT_TRUE_SIMULATED_WAIT(candidate_allocation_done_, kStunTimeoutMs,
>                              fake_clock);
>   EXPECT_EQ(1U, candidates_.size());
> }
> 
>+// Test that any address ports that are redundant do not surface.
>+TEST_F(BasicPortAllocatorTest, RedundantAnyAddressPortsDoNotSurface) {
>+  allocator().set_flags(allocator().flags() | PORTALLOCATOR_DISABLE_RELAY |
>+                        PORTALLOCATOR_DISABLE_TCP |
>+                        PORTALLOCATOR_ENABLE_SHARED_SOCKET);
>+  AddInterface(kClientAddr);
>+  // The any address ports will be duplicates of kClientAddr.
>+  vss_->SetAlternativeLocalAddress(kAnyAddr.ipaddr(), kClientAddr.ipaddr());
>+  ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+  session_->StartGettingPorts();
>+  EXPECT_TRUE_SIMULATED_WAIT(candidate_allocation_done_, kStunTimeoutMs,
>+                             fake_clock);
>+  EXPECT_EQ(1U, ports_.size());
>+  EXPECT_EQ(1, CountPorts(ports_, "local", PROTO_UDP, kClientAddr));
>+}
>+
>+// Test that candidates from the any address ports are not pruned if the
>+// explicit binding to enumerated networks fails.
>+TEST_F(BasicPortAllocatorTest,
>+       CandidatesFromAnyAddressPortsCanSurfaceWhenExplicitBindingFails) {
>+  allocator().set_flags(allocator().flags() | PORTALLOCATOR_DISABLE_RELAY |
>+                        PORTALLOCATOR_DISABLE_TCP |
>+                        PORTALLOCATOR_ENABLE_SHARED_SOCKET);
>+  AddInterface(kClientAddr);
>+  fss_->SetUnbindableIps({kClientAddr.ipaddr()});
>+  // The any address ports will be duplicates of kClientAddr, but the explict
>+  // binding will fail.
>+  vss_->SetAlternativeLocalAddress(kAnyAddr.ipaddr(), kClientAddr.ipaddr());
>+  ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+  session_->StartGettingPorts();
>+  EXPECT_TRUE_SIMULATED_WAIT(candidate_allocation_done_, kStunTimeoutMs,
>+                             fake_clock);
>+  EXPECT_EQ(1, CountPorts(ports_, "local", PROTO_UDP, kAnyAddr));
>+  EXPECT_EQ(1U, candidates_.size());
>+  EXPECT_TRUE(HasCandidate(candidates_, "local", "udp", kClientAddr));
>+}
>+
>+// Test that for an endpoint whose network enumeration only reveals one address
>+// (kClientAddr), it can observe a different address when binding to the "any"
>+// address. BasicPortAllocator should detect this and surface candidates for
>+// each address.
>+TEST_F(BasicPortAllocatorTest,
>+       CandidatesFromAnyAddressPortsCanSurfaceIfNotRedundant) {
>+  allocator().set_flags(allocator().flags() | PORTALLOCATOR_DISABLE_RELAY |
>+                        PORTALLOCATOR_DISABLE_TCP |
>+                        PORTALLOCATOR_ENABLE_SHARED_SOCKET);
>+  AddInterface(kClientAddr);
>+  // When bound to the any address, the port allocator should discover the
>+  // alternative local address.
>+  vss_->SetAlternativeLocalAddress(kAnyAddr.ipaddr(), kClientAddr2.ipaddr());
>+  ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+  session_->StartGettingPorts();
>+  EXPECT_TRUE_SIMULATED_WAIT(candidate_allocation_done_, kStunTimeoutMs,
>+                             fake_clock);
>+  EXPECT_EQ(1, CountPorts(ports_, "local", PROTO_UDP, kAnyAddr));
>+  EXPECT_EQ(2U, candidates_.size());
>+  EXPECT_TRUE(HasCandidate(candidates_, "local", "udp", kClientAddr));
>+  EXPECT_TRUE(HasCandidate(candidates_, "local", "udp", kClientAddr2));
>+}
>+
>+// Test that any address ports and their candidates are eventually signaled
>+// after the maximum wait interval for the completion of candidate allocation,
>+// when the any address ports and candidates are not redundant.
>+TEST_F(BasicPortAllocatorTest,
>+       GetAnyAddressPortsAfterMaximumWaitForCandidateAllocationDone) {
>+  ResetWithTurnServersNoNat(kTurnUdpIntAddr, rtc::SocketAddress());
>+  AddInterface(kClientAddr);
>+  vss_->SetAlternativeLocalAddress(kAnyAddr.ipaddr(), kClientAddr2.ipaddr());
>+  // STUN binding request and TURN allocation request will time out.
>+  fss_->AddRule(false, rtc::FP_UDP, rtc::FD_ANY, kClientAddr);
>+  ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+  session_->StartGettingPorts();
>+  SIMULATED_WAIT(false, kMaxWaitMsBeforeSignalingAnyAddressPortsAndCandidates,
>+                 fake_clock);
>+  EXPECT_EQ(2U, candidates_.size());
>+  EXPECT_TRUE(HasCandidate(candidates_, "local", "udp", kClientAddr));
>+  EXPECT_TRUE(HasCandidate(candidates_, "local", "tcp", kClientAddr));
>+  SIMULATED_WAIT(false, 1, fake_clock);
>+  EXPECT_FALSE(candidate_allocation_done_);
>+  // Candidates from the any address ports.
>+  EXPECT_EQ(6U, candidates_.size());
>+  EXPECT_TRUE(HasCandidate(candidates_, "local", "udp", kClientAddr2));
>+  EXPECT_TRUE(HasCandidate(candidates_, "stun", "udp", kClientAddr2));
>+  EXPECT_TRUE(HasCandidate(candidates_, "relay", "udp", kTurnUdpExtAddr));
>+  EXPECT_TRUE(HasCandidate(candidates_, "local", "tcp", kClientAddr2));
>+}
>+
>+// Test that the TCP port with the wildcard address is signaled if no ports are
>+// bound to enuemrated networks.
>+TEST_F(BasicPortAllocatorTest,
>+       GetAnyAddressTcpPortWhenNoPortsBoundToEnumeratedNetworks) {
>+  ResetWithTurnServersNoNat(kTurnUdpIntAddr, rtc::SocketAddress());
>+  AddInterface(kClientAddr);
>+  fss_->SetUnbindableIps({kClientAddr.ipaddr()});
>+  ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+  session_->StartGettingPorts();
>+  SIMULATED_WAIT(false,
>+                 kMaxWaitMsBeforeSignalingAnyAddressPortsAndCandidates + 1,
>+                 fake_clock);
>+  EXPECT_EQ(1, CountPorts(ports_, "local", PROTO_TCP, kAnyAddr));
>+}
>+
>+// Test that the STUN candidate from the any address port can still surface, if
>+// it is gathered after this port is signaled, .
>+TEST_F(BasicPortAllocatorTest,
>+       StunCandidateFromAnyAddressPortsGatheredLateCanBeSignaled) {
>+  ResetWithTurnServersNoNat(kTurnUdpIntAddr, rtc::SocketAddress());
>+  AddInterface(kClientAddr);
>+  vss_->SetAlternativeLocalAddress(kAnyAddr.ipaddr(), kClientAddr2.ipaddr());
>+  fss_->AddRule(false, rtc::FP_UDP, rtc::FD_ANY, kClientAddr);
>+  fss_->AddRule(false, rtc::FP_UDP, rtc::FD_ANY, kClientAddr2);
>+  ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+  session_->StartGettingPorts();
>+  SIMULATED_WAIT(false,
>+                 kMaxWaitMsBeforeSignalingAnyAddressPortsAndCandidates + 1000,
>+                 fake_clock);
>+  EXPECT_FALSE(candidate_allocation_done_);
>+  EXPECT_EQ(4U, candidates_.size());
>+  EXPECT_TRUE(HasCandidate(candidates_, "local", "udp", kClientAddr));
>+  EXPECT_TRUE(HasCandidate(candidates_, "local", "tcp", kClientAddr));
>+  EXPECT_TRUE(HasCandidate(candidates_, "local", "udp", kClientAddr2));
>+  EXPECT_TRUE(HasCandidate(candidates_, "local", "tcp", kClientAddr2));
>+  fss_->ClearRules();
>+  EXPECT_TRUE_SIMULATED_WAIT(candidate_allocation_done_,
>+                             kDefaultAllocationTimeout, fake_clock);
>+  EXPECT_EQ(7U, candidates_.size());
>+  EXPECT_TRUE(HasCandidate(candidates_, "stun", "udp", kClientAddr));
>+  EXPECT_TRUE(HasCandidate(candidates_, "stun", "udp", kClientAddr2));
>+  EXPECT_TRUE(HasCandidate(candidates_, "relay", "udp", kTurnUdpExtAddr));
>+}
>+
> // Test that when the NetworkManager doesn't have permission to enumerate
> // adapters, the PORTALLOCATOR_DISABLE_ADAPTER_ENUMERATION is specified
> // automatically.
>@@ -1901,13 +2152,13 @@ TEST_F(BasicPortAllocatorTest, TestNetworkPermissionBlocked) {
>                         PORTALLOCATOR_ENABLE_SHARED_SOCKET);
>   EXPECT_EQ(0U,
>             allocator_->flags() & PORTALLOCATOR_DISABLE_ADAPTER_ENUMERATION);
>-  EXPECT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+  ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>   EXPECT_EQ(0U, session_->flags() & PORTALLOCATOR_DISABLE_ADAPTER_ENUMERATION);
>   session_->StartGettingPorts();
>   EXPECT_EQ_SIMULATED_WAIT(1U, ports_.size(), kDefaultAllocationTimeout,
>                            fake_clock);
>   EXPECT_EQ(1U, candidates_.size());
>-  EXPECT_PRED4(HasCandidate, candidates_, "local", "udp", kPrivateAddr);
>+  EXPECT_TRUE(HasCandidate(candidates_, "local", "udp", kPrivateAddr));
>   EXPECT_NE(0U, session_->flags() & PORTALLOCATOR_DISABLE_ADAPTER_ENUMERATION);
> }
> 
>@@ -1919,22 +2170,22 @@ TEST_F(BasicPortAllocatorTest, TestEnableIPv6Addresses) {
>   AddInterface(kClientIPv6Addr);
>   AddInterface(kClientAddr);
>   allocator_->set_step_delay(kMinimumStepDelay);
>-  EXPECT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+  ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>   session_->StartGettingPorts();
>   ASSERT_TRUE_SIMULATED_WAIT(candidate_allocation_done_,
>                              kDefaultAllocationTimeout, fake_clock);
>   EXPECT_EQ(4U, ports_.size());
>   EXPECT_EQ(4U, candidates_.size());
>-  EXPECT_PRED4(HasCandidate, candidates_, "local", "udp", kClientIPv6Addr);
>-  EXPECT_PRED4(HasCandidate, candidates_, "local", "udp", kClientAddr);
>-  EXPECT_PRED4(HasCandidate, candidates_, "local", "tcp", kClientIPv6Addr);
>-  EXPECT_PRED4(HasCandidate, candidates_, "local", "tcp", kClientAddr);
>+  EXPECT_TRUE(HasCandidate(candidates_, "local", "udp", kClientIPv6Addr));
>+  EXPECT_TRUE(HasCandidate(candidates_, "local", "udp", kClientAddr));
>+  EXPECT_TRUE(HasCandidate(candidates_, "local", "tcp", kClientIPv6Addr));
>+  EXPECT_TRUE(HasCandidate(candidates_, "local", "tcp", kClientAddr));
> }
> 
> TEST_F(BasicPortAllocatorTest, TestStopGettingPorts) {
>   AddInterface(kClientAddr);
>   allocator_->set_step_delay(kDefaultStepDelay);
>-  EXPECT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+  ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>   session_->StartGettingPorts();
>   ASSERT_EQ_SIMULATED_WAIT(2U, candidates_.size(), 1000, fake_clock);
>   EXPECT_EQ(2U, ports_.size());
>@@ -1956,7 +2207,7 @@ TEST_F(BasicPortAllocatorTest, TestStopGettingPorts) {
> TEST_F(BasicPortAllocatorTest, TestClearGettingPorts) {
>   AddInterface(kClientAddr);
>   allocator_->set_step_delay(kDefaultStepDelay);
>-  EXPECT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+  ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>   session_->StartGettingPorts();
>   ASSERT_EQ_SIMULATED_WAIT(2U, candidates_.size(), 1000, fake_clock);
>   EXPECT_EQ(2U, ports_.size());
>@@ -2049,4 +2300,91 @@ TEST_F(BasicPortAllocatorTest, TestSetCandidateFilterAfterCandidatesGathered) {
>   }
> }
> 
>+TEST_F(BasicPortAllocatorTest, SetStunKeepaliveIntervalForPorts) {
>+  const int pool_size = 1;
>+  const int expected_stun_keepalive_interval = 123;
>+  AddInterface(kClientAddr);
>+  allocator_->SetConfiguration(allocator_->stun_servers(),
>+                               allocator_->turn_servers(), pool_size, false,
>+                               nullptr, expected_stun_keepalive_interval);
>+  auto* pooled_session = allocator_->GetPooledSession();
>+  ASSERT_NE(nullptr, pooled_session);
>+  EXPECT_EQ_SIMULATED_WAIT(true, pooled_session->CandidatesAllocationDone(),
>+                           kDefaultAllocationTimeout, fake_clock);
>+  CheckStunKeepaliveIntervalOfAllReadyPorts(pooled_session,
>+                                            expected_stun_keepalive_interval);
>+}
>+
>+TEST_F(BasicPortAllocatorTest,
>+       ChangeStunKeepaliveIntervalForPortsAfterInitialConfig) {
>+  const int pool_size = 1;
>+  AddInterface(kClientAddr);
>+  allocator_->SetConfiguration(allocator_->stun_servers(),
>+                               allocator_->turn_servers(), pool_size, false,
>+                               nullptr, 123 /* stun keepalive interval */);
>+  auto* pooled_session = allocator_->GetPooledSession();
>+  ASSERT_NE(nullptr, pooled_session);
>+  EXPECT_EQ_SIMULATED_WAIT(true, pooled_session->CandidatesAllocationDone(),
>+                           kDefaultAllocationTimeout, fake_clock);
>+  const int expected_stun_keepalive_interval = 321;
>+  allocator_->SetConfiguration(allocator_->stun_servers(),
>+                               allocator_->turn_servers(), pool_size, false,
>+                               nullptr, expected_stun_keepalive_interval);
>+  CheckStunKeepaliveIntervalOfAllReadyPorts(pooled_session,
>+                                            expected_stun_keepalive_interval);
>+}
>+
>+TEST_F(BasicPortAllocatorTest,
>+       SetStunKeepaliveIntervalForPortsWithSharedSocket) {
>+  const int pool_size = 1;
>+  const int expected_stun_keepalive_interval = 123;
>+  AddInterface(kClientAddr);
>+  allocator_->set_flags(allocator().flags() |
>+                        PORTALLOCATOR_ENABLE_SHARED_SOCKET);
>+  allocator_->SetConfiguration(allocator_->stun_servers(),
>+                               allocator_->turn_servers(), pool_size, false,
>+                               nullptr, expected_stun_keepalive_interval);
>+  ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+  session_->StartGettingPorts();
>+  EXPECT_TRUE_SIMULATED_WAIT(candidate_allocation_done_,
>+                             kDefaultAllocationTimeout, fake_clock);
>+  CheckStunKeepaliveIntervalOfAllReadyPorts(session_.get(),
>+                                            expected_stun_keepalive_interval);
>+}
>+
>+TEST_F(BasicPortAllocatorTest,
>+       SetStunKeepaliveIntervalForPortsWithoutSharedSocket) {
>+  const int pool_size = 1;
>+  const int expected_stun_keepalive_interval = 123;
>+  AddInterface(kClientAddr);
>+  allocator_->set_flags(allocator().flags() &
>+                        ~(PORTALLOCATOR_ENABLE_SHARED_SOCKET));
>+  allocator_->SetConfiguration(allocator_->stun_servers(),
>+                               allocator_->turn_servers(), pool_size, false,
>+                               nullptr, expected_stun_keepalive_interval);
>+  ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+  session_->StartGettingPorts();
>+  EXPECT_TRUE_SIMULATED_WAIT(candidate_allocation_done_,
>+                             kDefaultAllocationTimeout, fake_clock);
>+  CheckStunKeepaliveIntervalOfAllReadyPorts(session_.get(),
>+                                            expected_stun_keepalive_interval);
>+}
>+
>+TEST_F(BasicPortAllocatorTest, IceRegatheringMetricsLoggedWhenNetworkChanges) {
>+  // Only test local ports to simplify test.
>+  ResetWithNoServersOrNat();
>+  AddInterface(kClientAddr, "test_net0");
>+  ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
>+  session_->StartGettingPorts();
>+  EXPECT_TRUE_SIMULATED_WAIT(candidate_allocation_done_,
>+                             kDefaultAllocationTimeout, fake_clock);
>+  candidate_allocation_done_ = false;
>+  AddInterface(kClientAddr2, "test_net1");
>+  EXPECT_TRUE_SIMULATED_WAIT(candidate_allocation_done_,
>+                             kDefaultAllocationTimeout, fake_clock);
>+  EXPECT_EQ(1, webrtc::metrics::NumEvents(
>+                   "WebRTC.PeerConnection.IceRegatheringReason",
>+                   static_cast<int>(IceRegatheringReason::NETWORK_CHANGE)));
>+}
>+
> }  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/client/relayportfactoryinterface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/client/relayportfactoryinterface.h
>index 5890bcd9bd2..2afe6512fe8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/client/relayportfactoryinterface.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/client/relayportfactoryinterface.h
>@@ -56,15 +56,13 @@ class RelayPortFactoryInterface {
> 
>   // This variant is used for UDP connection to the relay server
>   // using a already existing shared socket.
>-  virtual std::unique_ptr<Port> Create(
>-      const CreateRelayPortArgs& args,
>-      rtc::AsyncPacketSocket* udp_socket) = 0;
>+  virtual std::unique_ptr<Port> Create(const CreateRelayPortArgs& args,
>+                                       rtc::AsyncPacketSocket* udp_socket) = 0;
> 
>   // This variant is used for the other cases.
>-  virtual std::unique_ptr<Port> Create(
>-      const CreateRelayPortArgs& args,
>-      int min_port,
>-      int max_port) = 0;
>+  virtual std::unique_ptr<Port> Create(const CreateRelayPortArgs& args,
>+                                       int min_port,
>+                                       int max_port) = 0;
> };
> 
> }  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/client/socketmonitor.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/client/socketmonitor.cc
>deleted file mode 100644
>index 6bfa186fae1..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/client/socketmonitor.cc
>+++ /dev/null
>@@ -1,96 +0,0 @@
>-/*
>- *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "p2p/client/socketmonitor.h"
>-
>-#include "rtc_base/checks.h"
>-
>-namespace cricket {
>-
>-enum {
>-  MSG_MONITOR_POLL,
>-  MSG_MONITOR_START,
>-  MSG_MONITOR_STOP,
>-  MSG_MONITOR_SIGNAL
>-};
>-
>-ConnectionMonitor::ConnectionMonitor(ConnectionStatsGetter* stats_getter,
>-                                     rtc::Thread* network_thread,
>-                                     rtc::Thread* monitoring_thread) {
>-  stats_getter_ = stats_getter;
>-  network_thread_ = network_thread;
>-  monitoring_thread_ = monitoring_thread;
>-  monitoring_ = false;
>-}
>-
>-ConnectionMonitor::~ConnectionMonitor() {
>-  network_thread_->Clear(this);
>-  monitoring_thread_->Clear(this);
>-}
>-
>-void ConnectionMonitor::Start(int milliseconds) {
>-  rate_ = milliseconds;
>-  if (rate_ < 250)
>-    rate_ = 250;
>-  network_thread_->Post(RTC_FROM_HERE, this, MSG_MONITOR_START);
>-}
>-
>-void ConnectionMonitor::Stop() {
>-  network_thread_->Post(RTC_FROM_HERE, this, MSG_MONITOR_STOP);
>-}
>-
>-void ConnectionMonitor::OnMessage(rtc::Message *message) {
>-  rtc::CritScope cs(&crit_);
>-  switch (message->message_id) {
>-    case MSG_MONITOR_START:
>-      RTC_DCHECK(rtc::Thread::Current() == network_thread_);
>-      if (!monitoring_) {
>-        monitoring_ = true;
>-        PollConnectionStats_w();
>-      }
>-      break;
>-
>-    case MSG_MONITOR_STOP:
>-      RTC_DCHECK(rtc::Thread::Current() == network_thread_);
>-      if (monitoring_) {
>-        monitoring_ = false;
>-        network_thread_->Clear(this);
>-      }
>-      break;
>-
>-    case MSG_MONITOR_POLL:
>-      RTC_DCHECK(rtc::Thread::Current() == network_thread_);
>-      PollConnectionStats_w();
>-      break;
>-
>-    case MSG_MONITOR_SIGNAL: {
>-      RTC_DCHECK(rtc::Thread::Current() == monitoring_thread_);
>-      std::vector<ConnectionInfo> infos = connection_infos_;
>-      crit_.Leave();
>-      SignalUpdate(this, infos);
>-      crit_.Enter();
>-      break;
>-    }
>-  }
>-}
>-
>-void ConnectionMonitor::PollConnectionStats_w() {
>-  RTC_DCHECK(rtc::Thread::Current() == network_thread_);
>-  rtc::CritScope cs(&crit_);
>-
>-  // Gather connection infos
>-  stats_getter_->GetConnectionStats(&connection_infos_);
>-
>-  // Signal the monitoring thread, start another poll timer
>-  monitoring_thread_->Post(RTC_FROM_HERE, this, MSG_MONITOR_SIGNAL);
>-  network_thread_->PostDelayed(RTC_FROM_HERE, rate_, this, MSG_MONITOR_POLL);
>-}
>-
>-}  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/client/socketmonitor.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/client/socketmonitor.h
>deleted file mode 100644
>index 3428b0335b6..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/client/socketmonitor.h
>+++ /dev/null
>@@ -1,63 +0,0 @@
>-/*
>- *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef P2P_CLIENT_SOCKETMONITOR_H_
>-#define P2P_CLIENT_SOCKETMONITOR_H_
>-
>-#include <vector>
>-
>-#include "p2p/base/port.h"  // for ConnectionInfos
>-#include "rtc_base/criticalsection.h"
>-#include "rtc_base/sigslot.h"
>-#include "rtc_base/thread.h"
>-
>-// TODO(pthatcher): Move these to connectionmonitor.h and
>-// connectionmonitor.cc, or just move them into channel.cc
>-
>-namespace cricket {
>-
>-class ConnectionStatsGetter {
>- public:
>-  virtual ~ConnectionStatsGetter() {}
>-  virtual bool GetConnectionStats(ConnectionInfos* infos) = 0;
>-};
>-
>-class ConnectionMonitor : public rtc::MessageHandler,
>-                          public sigslot::has_slots<> {
>- public:
>-  ConnectionMonitor(ConnectionStatsGetter* stats_getter,
>-                    rtc::Thread* network_thread,
>-                    rtc::Thread* monitoring_thread);
>-  ~ConnectionMonitor() override;
>-
>-  void Start(int cms);
>-  void Stop();
>-
>-  sigslot::signal2<ConnectionMonitor*,
>-                   const std::vector<ConnectionInfo>&> SignalUpdate;
>-
>- protected:
>-  void OnMessage(rtc::Message* message) override;
>-
>- private:
>-  void PollConnectionStats_w();
>-
>-  std::vector<ConnectionInfo> connection_infos_;
>-  ConnectionStatsGetter* stats_getter_;
>-  rtc::Thread* network_thread_;
>-  rtc::Thread* monitoring_thread_;
>-  rtc::CriticalSection crit_;
>-  uint32_t rate_;
>-  bool monitoring_;
>-};
>-
>-}  // namespace cricket
>-
>-#endif  // P2P_CLIENT_SOCKETMONITOR_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/client/turnportfactory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/client/turnportfactory.cc
>index 9f24f5f4b60..6404134c4a1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/client/turnportfactory.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/client/turnportfactory.cc
>@@ -16,49 +16,29 @@
> 
> namespace cricket {
> 
>-TurnPortFactory::~TurnPortFactory() {
>-}
>+TurnPortFactory::~TurnPortFactory() {}
> 
> std::unique_ptr<Port> TurnPortFactory::Create(
>     const CreateRelayPortArgs& args,
>     rtc::AsyncPacketSocket* udp_socket) {
>-
>   TurnPort* port = TurnPort::Create(
>-      args.network_thread,
>-      args.socket_factory,
>-      args.network,
>-      udp_socket,
>-      args.username,
>-      args.password,
>-      *args.server_address,
>-      args.config->credentials,
>-      args.config->priority,
>-      args.origin,
>+      args.network_thread, args.socket_factory, args.network, udp_socket,
>+      args.username, args.password, *args.server_address,
>+      args.config->credentials, args.config->priority, args.origin,
>       args.turn_customizer);
>   port->SetTlsCertPolicy(args.config->tls_cert_policy);
>   return std::unique_ptr<Port>(port);
> }
> 
>-std::unique_ptr<Port> TurnPortFactory::Create(
>-    const CreateRelayPortArgs& args,
>-    int min_port,
>-    int max_port) {
>-
>+std::unique_ptr<Port> TurnPortFactory::Create(const CreateRelayPortArgs& args,
>+                                              int min_port,
>+                                              int max_port) {
>   TurnPort* port = TurnPort::Create(
>-      args.network_thread,
>-      args.socket_factory,
>-      args.network,
>-      min_port,
>-      max_port,
>-      args.username,
>-      args.password,
>-      *args.server_address,
>-      args.config->credentials,
>-      args.config->priority,
>-      args.origin,
>-      args.config->tls_alpn_protocols,
>-      args.config->tls_elliptic_curves,
>-      args.turn_customizer);
>+      args.network_thread, args.socket_factory, args.network, min_port,
>+      max_port, args.username, args.password, *args.server_address,
>+      args.config->credentials, args.config->priority, args.origin,
>+      args.config->tls_alpn_protocols, args.config->tls_elliptic_curves,
>+      args.turn_customizer, args.config->tls_cert_verifier);
>   port->SetTlsCertPolicy(args.config->tls_cert_policy);
>   return std::unique_ptr<Port>(port);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/client/turnportfactory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/client/turnportfactory.h
>index d5c510b83fe..3e45a6acb00 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/client/turnportfactory.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/client/turnportfactory.h
>@@ -22,14 +22,12 @@ class TurnPortFactory : public RelayPortFactoryInterface {
>  public:
>   ~TurnPortFactory() override;
> 
>-  std::unique_ptr<Port> Create(
>-      const CreateRelayPortArgs& args,
>-      rtc::AsyncPacketSocket* udp_socket) override;
>-
>-  std::unique_ptr<Port> Create(
>-      const CreateRelayPortArgs& args,
>-      int min_port,
>-      int max_port) override;
>+  std::unique_ptr<Port> Create(const CreateRelayPortArgs& args,
>+                               rtc::AsyncPacketSocket* udp_socket) override;
>+
>+  std::unique_ptr<Port> Create(const CreateRelayPortArgs& args,
>+                               int min_port,
>+                               int max_port) override;
> };
> 
> }  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/stunprober/stunprober.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/stunprober/stunprober.cc
>index 2a02cd02f1a..51506b670d7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/stunprober/stunprober.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/stunprober/stunprober.cc
>@@ -258,8 +258,7 @@ StunProber::StunProber(rtc::PacketSocketFactory* socket_factory,
>     : interval_ms_(0),
>       socket_factory_(socket_factory),
>       thread_(thread),
>-      networks_(networks) {
>-}
>+      networks_(networks) {}
> 
> StunProber::~StunProber() {
>   for (auto* req : requesters_) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/stunprober/stunprober.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/stunprober/stunprober.h
>index 718e202f017..9f1d42f32ce 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/stunprober/stunprober.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/stunprober/stunprober.h
>@@ -16,7 +16,6 @@
> #include <vector>
> 
> #include "rtc_base/asyncinvoker.h"
>-#include "rtc_base/basictypes.h"
> #include "rtc_base/bytebuffer.h"
> #include "rtc_base/callback.h"
> #include "rtc_base/constructormagic.h"
>@@ -25,7 +24,6 @@
> #include "rtc_base/socketaddress.h"
> #include "rtc_base/thread.h"
> #include "rtc_base/thread_checker.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace rtc {
> class AsyncPacketSocket;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/BUILD.gn
>index fb9e65a2691..03b6d80040b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/BUILD.gn
>@@ -29,24 +29,18 @@ rtc_static_library("rtc_pc_base") {
>   visibility = [ "*" ]
>   defines = []
>   sources = [
>-    "audiomonitor.cc",
>-    "audiomonitor.h",
>-    "bundlefilter.cc",
>-    "bundlefilter.h",
>     "channel.cc",
>     "channel.h",
>     "channelmanager.cc",
>     "channelmanager.h",
>-    "currentspeakermonitor.cc",
>-    "currentspeakermonitor.h",
>     "dtlssrtptransport.cc",
>     "dtlssrtptransport.h",
>     "externalhmac.cc",
>     "externalhmac.h",
>     "jseptransport.cc",
>     "jseptransport.h",
>-    "mediamonitor.cc",
>-    "mediamonitor.h",
>+    "jseptransportcontroller.cc",
>+    "jseptransportcontroller.h",
>     "mediasession.cc",
>     "mediasession.h",
>     "rtcpmuxfilter.cc",
>@@ -65,26 +59,35 @@ rtc_static_library("rtc_pc_base") {
>     "srtpsession.h",
>     "srtptransport.cc",
>     "srtptransport.h",
>-    "transportcontroller.cc",
>-    "transportcontroller.h",
>-    "voicechannel.h",
>+    "transportstats.cc",
>+    "transportstats.h",
>   ]
> 
>   deps = [
>     "..:webrtc_common",
>+    "../api:array_view",
>     "../api:call_api",
>     "../api:libjingle_peerconnection_api",
>-    "../api:optional",
>     "../api:ortc_api",
>+    "../api/video:video_frame",
>+    "../call:rtp_interfaces",
>+    "../call:rtp_receiver",
>     "../common_video:common_video",
>+    "../logging:rtc_event_log_api",
>     "../media:rtc_data",
>     "../media:rtc_h264_profile_id",
>     "../media:rtc_media_base",
>+    "../modules/rtp_rtcp:rtp_rtcp_format",
>     "../p2p:rtc_p2p",
>     "../rtc_base:checks",
>     "../rtc_base:rtc_base",
>     "../rtc_base:rtc_task_queue",
>     "../rtc_base:stringutils",
>+    "../rtc_base/third_party/base64",
>+    "../rtc_base/third_party/sigslot",
>+    "../system_wrappers:metrics_api",
>+    "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> 
>   if (rtc_build_libsrtp) {
>@@ -101,21 +104,16 @@ rtc_static_library("rtc_pc_base") {
> 
> rtc_source_set("rtc_pc") {
>   visibility = [ "*" ]
>+  allow_poison = [
>+    "audio_codecs",  # TODO(bugs.webrtc.org/8396): Remove.
>+    "software_video_codecs",  # TODO(bugs.webrtc.org/7925): Remove.
>+  ]
>   deps = [
>     ":rtc_pc_base",
>     "../media:rtc_audio_video",
>   ]
> }
> 
>-config("libjingle_peerconnection_warnings_config") {
>-  # GN orders flags on a target before flags from configs. The default config
>-  # adds these flags so to cancel them out they need to come from a config and
>-  # cannot be on the target directly.
>-  if (!is_win && !is_clang) {
>-    cflags = [ "-Wno-maybe-uninitialized" ]  # Only exists for GCC.
>-  }
>-}
>-
> rtc_static_library("peerconnection") {
>   visibility = [ "*" ]
>   cflags = []
>@@ -141,10 +139,15 @@ rtc_static_library("peerconnection") {
>     "peerconnection.h",
>     "peerconnectionfactory.cc",
>     "peerconnectionfactory.h",
>+    "peerconnectioninternal.h",
>     "remoteaudiosource.cc",
>     "remoteaudiosource.h",
>     "rtcstatscollector.cc",
>     "rtcstatscollector.h",
>+    "rtcstatstraversal.cc",
>+    "rtcstatstraversal.h",
>+    "rtpparametersconversion.cc",
>+    "rtpparametersconversion.h",
>     "rtpreceiver.cc",
>     "rtpreceiver.h",
>     "rtpsender.cc",
>@@ -172,8 +175,6 @@ rtc_static_library("peerconnection") {
>     "webrtcsessiondescriptionfactory.h",
>   ]
> 
>-  configs += [ ":libjingle_peerconnection_warnings_config" ]
>-
>   if (!build_with_chromium && is_clang) {
>     # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>     suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>@@ -183,24 +184,33 @@ rtc_static_library("peerconnection") {
>     ":rtc_pc_base",
>     "..:webrtc_common",
>     "../api:call_api",
>+    "../api:fec_controller_api",
>     "../api:libjingle_peerconnection_api",
>-    "../api:optional",
>-    "../api:peerconnection_and_implicit_call_api",
>     "../api:rtc_stats_api",
>+    "../api/video:video_frame",
>     "../api/video_codecs:video_codecs_api",
>     "../call:call_interfaces",
>     "../common_video:common_video",
>+    "../logging:ice_log",
>     "../logging:rtc_event_log_api",
>+    "../logging:rtc_event_log_impl_output",
>     "../media:rtc_data",
>     "../media:rtc_media_base",
>+    "../modules/congestion_controller/bbr",
>     "../p2p:rtc_p2p",
>     "../rtc_base:checks",
>     "../rtc_base:rtc_base",
>     "../rtc_base:rtc_base_approved",
>     "../rtc_base:stringutils",
>+    "../rtc_base/experiments:congestion_controller_experiment",
>+    "../rtc_base/third_party/base64",
>+    "../rtc_base/third_party/sigslot",
>     "../stats",
>     "../system_wrappers",
>     "../system_wrappers:field_trial_api",
>+    "../system_wrappers:metrics_api",
>+    "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
> 
>@@ -216,14 +226,15 @@ rtc_static_library("create_pc_factory") {
>   ]
> 
>   deps = [
>-    "../api:audio_mixer_api",
>+    "../api:callfactory_api",
>     "../api:libjingle_peerconnection_api",
>-    "../api:peerconnection_and_implicit_call_api",
>+    "../api/audio:audio_mixer_api",
>     "../api/audio_codecs:audio_codecs_api",
>     "../api/video_codecs:video_codecs_api",
>     "../call",
>     "../call:call_interfaces",
>     "../logging:rtc_event_log_api",
>+    "../logging:rtc_event_log_impl_base",
>     "../media:rtc_audio_video",
>     "../media:rtc_media_base",
>     "../modules/audio_device:audio_device",
>@@ -232,8 +243,6 @@ rtc_static_library("create_pc_factory") {
>     "../rtc_base:rtc_base_approved",
>   ]
> 
>-  configs += [ ":libjingle_peerconnection_warnings_config" ]
>-
>   if (!build_with_chromium && is_clang) {
>     # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>     suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>@@ -242,6 +251,10 @@ rtc_static_library("create_pc_factory") {
> 
> rtc_source_set("libjingle_peerconnection") {
>   visibility = [ "*" ]
>+  allow_poison = [
>+    "audio_codecs",  # TODO(bugs.webrtc.org/8396): Remove.
>+    "software_video_codecs",  # TODO(bugs.webrtc.org/7925): Remove.
>+  ]
>   deps = [
>     ":create_pc_factory",
>     ":peerconnection",
>@@ -250,25 +263,15 @@ rtc_source_set("libjingle_peerconnection") {
> }
> 
> if (rtc_include_tests) {
>-  config("rtc_pc_unittests_config") {
>-    # GN orders flags on a target before flags from configs. The default config
>-    # adds -Wall, and this flag have to be after -Wall -- so they need to
>-    # come from a config and can't be on the target directly.
>-    if (!is_win && !is_clang) {
>-      cflags = [ "-Wno-maybe-uninitialized" ]  # Only exists for GCC.
>-    }
>-  }
>-
>   rtc_test("rtc_pc_unittests") {
>     testonly = true
> 
>     sources = [
>-      "bundlefilter_unittest.cc",
>       "channel_unittest.cc",
>       "channelmanager_unittest.cc",
>-      "currentspeakermonitor_unittest.cc",
>       "dtlssrtptransport_unittest.cc",
>       "jseptransport_unittest.cc",
>+      "jseptransportcontroller_unittest.cc",
>       "mediasession_unittest.cc",
>       "rtcpmuxfilter_unittest.cc",
>       "rtptransport_unittest.cc",
>@@ -277,13 +280,10 @@ if (rtc_include_tests) {
>       "srtpsession_unittest.cc",
>       "srtptestutil.h",
>       "srtptransport_unittest.cc",
>-      "transportcontroller_unittest.cc",
>     ]
> 
>     include_dirs = [ "//third_party/libsrtp/srtp" ]
> 
>-    configs += [ ":rtc_pc_unittests_config" ]
>-
>     if (!build_with_chromium && is_clang) {
>       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>@@ -300,9 +300,11 @@ if (rtc_include_tests) {
>       ":rtc_pc_base",
>       "../api:array_view",
>       "../api:libjingle_peerconnection_api",
>+      "../call:rtp_interfaces",
>       "../logging:rtc_event_log_api",
>       "../media:rtc_media_base",
>       "../media:rtc_media_tests_utils",
>+      "../modules/rtp_rtcp:rtp_rtcp_format",
>       "../p2p:p2p_test_utils",
>       "../p2p:rtc_p2p",
>       "../rtc_base:checks",
>@@ -310,9 +312,11 @@ if (rtc_include_tests) {
>       "../rtc_base:rtc_base_approved",
>       "../rtc_base:rtc_base_tests_main",
>       "../rtc_base:rtc_base_tests_utils",
>+      "../rtc_base/third_party/sigslot",
>       "../system_wrappers:metrics_default",
>       "../system_wrappers:runtime_enabled_features_default",
>       "../test:test_support",
>+      "//third_party/abseil-cpp/absl/memory",
>     ]
> 
>     if (rtc_build_libsrtp) {
>@@ -324,20 +328,58 @@ if (rtc_include_tests) {
>     }
>   }
> 
>+  rtc_source_set("peerconnection_perf_tests") {
>+    testonly = true
>+    sources = [
>+      "peerconnection_rampup_tests.cc",
>+      "peerconnectionwrapper.cc",
>+      "peerconnectionwrapper.h",
>+    ]
>+    deps = [
>+      ":pc_test_utils",
>+      "../api:libjingle_peerconnection_api",
>+      "../api:libjingle_peerconnection_test_api",
>+      "../api:rtc_stats_api",
>+      "../api/audio_codecs:builtin_audio_decoder_factory",
>+      "../api/audio_codecs:builtin_audio_encoder_factory",
>+      "../api/video_codecs:builtin_video_decoder_factory",
>+      "../api/video_codecs:builtin_video_encoder_factory",
>+      "../media:rtc_media_tests_utils",
>+      "../p2p:p2p_test_utils",
>+      "../p2p:rtc_p2p",
>+      "../pc:peerconnection",
>+      "../rtc_base:rtc_base",
>+      "../rtc_base:rtc_base_approved",
>+      "../rtc_base:rtc_base_tests_utils",
>+      "../test:perf_test",
>+      "../test:test_support",
>+      "//third_party/abseil-cpp/absl/memory",
>+    ]
>+    if (!build_with_chromium && is_clang) {
>+      # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>+      suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>+    }
>+  }
>+
>   rtc_source_set("pc_test_utils") {
>     testonly = true
>     sources = [
>       "test/fakeaudiocapturemodule.cc",
>       "test/fakeaudiocapturemodule.h",
>       "test/fakedatachannelprovider.h",
>-      "test/fakeperiodicvideocapturer.h",
>+      "test/fakepeerconnectionbase.h",
>+      "test/fakepeerconnectionforstats.h",
>+      "test/fakeperiodicvideosource.h",
>+      "test/fakeperiodicvideotracksource.h",
>       "test/fakertccertificategenerator.h",
>       "test/fakesctptransport.h",
>-      "test/faketransportcontroller.h",
>       "test/fakevideotrackrenderer.h",
>       "test/fakevideotracksource.h",
>+      "test/framegeneratorcapturervideotracksource.h",
>       "test/mock_datachannel.h",
>       "test/mock_peerconnection.h",
>+      "test/mock_rtpreceiverinternal.h",
>+      "test/mock_rtpsenderinternal.h",
>       "test/mockpeerconnectionobservers.h",
>       "test/peerconnectiontestwrapper.cc",
>       "test/peerconnectiontestwrapper.h",
>@@ -352,8 +394,10 @@ if (rtc_include_tests) {
>       "..:webrtc_common",
>       "../api:libjingle_peerconnection_api",
>       "../api:libjingle_peerconnection_test_api",
>-      "../api:peerconnection_and_implicit_call_api",
>       "../api:rtc_stats_api",
>+      "../api/video:video_frame",
>+      "../api/video_codecs:builtin_video_decoder_factory",
>+      "../api/video_codecs:builtin_video_encoder_factory",
>       "../call:call_interfaces",
>       "../logging:rtc_event_log_api",
>       "../media:rtc_data",
>@@ -361,13 +405,17 @@ if (rtc_include_tests) {
>       "../media:rtc_media_base",
>       "../media:rtc_media_tests_utils",
>       "../modules/audio_device:audio_device",
>+      "../modules/audio_processing:audio_processing",
>       "../p2p:p2p_test_utils",
>       "../rtc_base:checks",
>       "../rtc_base:rtc_base",
>       "../rtc_base:rtc_base_approved",
>       "../rtc_base:rtc_base_tests_utils",
>+      "../rtc_base:rtc_task_queue",
>+      "../rtc_base/third_party/sigslot",
>       "../test:test_support",
>-      "//testing/gmock",
>+      "../test:video_test_common",
>+      "//third_party/abseil-cpp/absl/memory",
>     ]
> 
>     if (!build_with_chromium && is_clang) {
>@@ -376,25 +424,6 @@ if (rtc_include_tests) {
>     }
>   }
> 
>-  config("peerconnection_unittests_config") {
>-    # The warnings below are enabled by default. Since GN orders compiler flags
>-    # for a target before flags from configs, the only way to disable such
>-    # warnings is by having them in a separate config, loaded from the target.
>-    # TODO(kjellander): Make the code compile without disabling these flags.
>-    # See https://bugs.webrtc.org/3307.
>-    if (is_clang && is_win) {
>-      cflags = [
>-        # See https://bugs.chromium.org/p/webrtc/issues/detail?id=6267
>-        # for -Wno-sign-compare
>-        "-Wno-sign-compare",
>-      ]
>-    }
>-
>-    if (!is_win) {
>-      cflags = [ "-Wno-sign-compare" ]
>-    }
>-  }
>-
>   rtc_test("peerconnection_unittests") {
>     testonly = true
>     sources = [
>@@ -408,6 +437,7 @@ if (rtc_include_tests) {
>       "peerconnection_bundle_unittest.cc",
>       "peerconnection_crypto_unittest.cc",
>       "peerconnection_datachannel_unittest.cc",
>+      "peerconnection_histogram_unittest.cc",
>       "peerconnection_ice_unittest.cc",
>       "peerconnection_integrationtest.cc",
>       "peerconnection_jsep_unittest.cc",
>@@ -422,7 +452,9 @@ if (rtc_include_tests) {
>       "proxy_unittest.cc",
>       "rtcstats_integrationtest.cc",
>       "rtcstatscollector_unittest.cc",
>+      "rtcstatstraversal_unittest.cc",
>       "rtpmediautils_unittest.cc",
>+      "rtpparametersconversion_unittest.cc",
>       "rtpsenderreceiver_unittest.cc",
>       "sctputils_unittest.cc",
>       "statscollector_unittest.cc",
>@@ -438,31 +470,23 @@ if (rtc_include_tests) {
>       defines = [ "HAVE_SCTP" ]
>     }
> 
>-    configs += [ ":peerconnection_unittests_config" ]
>-
>     if (!build_with_chromium && is_clang) {
>       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>     }
> 
>-    # TODO(jschuh): Bug 1348: fix this warning.
>-    configs += [ "//build/config/compiler:no_size_t_to_int_warning" ]
>-
>-    if (is_win) {
>-      cflags = [
>-        "/wd4245",  # conversion from int to size_t, signed/unsigned mismatch.
>-        "/wd4389",  # signed/unsigned mismatch.
>-      ]
>-    }
>-
>     deps = [
>       ":peerconnection",
>       ":rtc_pc_base",
>       "../api:libjingle_peerconnection_api",
>       "../api:mock_rtp",
>-      "../api:peerconnection_and_implicit_call_api",
>+      "../api/units:time_delta",
>+      "../logging:fake_rtc_event_log",
>       "../rtc_base:checks",
>       "../rtc_base:stringutils",
>+      "../rtc_base/third_party/base64",
>+      "../test:fileutils",
>+      "//third_party/abseil-cpp/absl/memory",
>     ]
>     if (is_android) {
>       deps += [ ":android_black_magic" ]
>@@ -472,18 +496,21 @@ if (rtc_include_tests) {
>       ":libjingle_peerconnection",
>       ":pc_test_utils",
>       "..:webrtc_common",
>-      "../api:fakemetricsobserver",
>+      "../api:callfactory_api",
>       "../api:libjingle_peerconnection_test_api",
>-      "../api:optional",
>       "../api:rtc_stats_api",
>       "../api/audio_codecs:audio_codecs_api",
>       "../api/audio_codecs:builtin_audio_decoder_factory",
>       "../api/audio_codecs:builtin_audio_encoder_factory",
>       "../api/audio_codecs/L16:audio_decoder_L16",
>       "../api/audio_codecs/L16:audio_encoder_L16",
>+      "../api/video_codecs:builtin_video_decoder_factory",
>+      "../api/video_codecs:builtin_video_encoder_factory",
>+      "../api/video_codecs:video_codecs_api",
>       "../call:call_interfaces",
>       "../logging:rtc_event_log_api",
>-      "../logging:rtc_event_log_impl",
>+      "../logging:rtc_event_log_impl_base",
>+      "../logging:rtc_event_log_impl_output",
>       "../media:rtc_audio_video",
>       "../media:rtc_data",  # TODO(phoglund): AFAIK only used for one sctp constant.
>       "../media:rtc_media_base",
>@@ -497,10 +524,13 @@ if (rtc_include_tests) {
>       "../rtc_base:rtc_base_approved",
>       "../rtc_base:rtc_base_tests_main",
>       "../rtc_base:rtc_base_tests_utils",
>+      "../rtc_base:rtc_task_queue",
>+      "../rtc_base:safe_conversions",
>       "../system_wrappers:metrics_default",
>       "../system_wrappers:runtime_enabled_features_default",
>       "../test:audio_codec_mocks",
>       "../test:test_support",
>+      "//third_party/abseil-cpp/absl/types:optional",
>     ]
> 
>     if (is_android) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/DEPS b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/DEPS
>index 1992af013e1..1f9d15272dc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/DEPS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/DEPS
>@@ -8,6 +8,7 @@ include_rules = [
>   "+media",
>   "+modules/audio_device",
>   "+modules/audio_processing",
>+  "+modules/congestion_controller",
>   "+modules/rtp_rtcp",
>   "+modules/video_coding",
>   "+modules/video_render",
>@@ -18,18 +19,8 @@ include_rules = [
> specific_include_rules = {
>   "androidtestinitializer\.cc": [
>     "+base/android",  # Allowed only for Android tests.
>-    "+voice_engine",
>   ],
>   "srtpfilter_unittest\.cc": [
>     "+crypto",
>   ],
>-
>-  # TODO(ossu): Remove these exceptions when audio_encoder_factory.h
>-  # has moved to api/.
>-  "peerconnectionfactory\.cc": [
>-    "+modules/audio_coding/codecs/builtin_audio_encoder_factory.h",
>-  ],
>-  "peerconnectioninterface_unittest\.cc": [
>-    "+modules/audio_coding/codecs/builtin_audio_encoder_factory.h",
>-  ],
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/OWNERS
>index f698e832a17..d8555c8847e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/OWNERS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/OWNERS
>@@ -1,12 +1,12 @@
>-deadbeef@webrtc.org
> honghaiz@webrtc.org
> hta@webrtc.org
>-jiayl@webrtc.org
> juberti@webrtc.org
> perkj@webrtc.org
> pthatcher@webrtc.org
>-sergeyu@chromium.org
>+shampson@webrtc.org
>+steveanton@webrtc.org
> tommi@webrtc.org
>+zhihuang@webrtc.org
> 
> per-file peerconnection*=hbos@webrtc.org
> per-file rtcstats*=hbos@webrtc.org
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/audiomonitor.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/audiomonitor.cc
>deleted file mode 100644
>index a5bffd7b581..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/audiomonitor.cc
>+++ /dev/null
>@@ -1,105 +0,0 @@
>-/*
>- *  Copyright 2004 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include <assert.h>
>-#include "pc/audiomonitor.h"
>-#include "pc/voicechannel.h"
>-
>-namespace cricket {
>-
>-const uint32_t MSG_MONITOR_POLL = 1;
>-const uint32_t MSG_MONITOR_START = 2;
>-const uint32_t MSG_MONITOR_STOP = 3;
>-const uint32_t MSG_MONITOR_SIGNAL = 4;
>-
>-AudioMonitor::AudioMonitor(VoiceChannel *voice_channel,
>-                           rtc::Thread *monitor_thread) {
>-  voice_channel_ = voice_channel;
>-  monitoring_thread_ = monitor_thread;
>-  monitoring_ = false;
>-}
>-
>-AudioMonitor::~AudioMonitor() {
>-  voice_channel_->worker_thread()->Clear(this);
>-  monitoring_thread_->Clear(this);
>-}
>-
>-void AudioMonitor::Start(int milliseconds) {
>-  rate_ = milliseconds;
>-  if (rate_ < 100)
>-    rate_ = 100;
>-  voice_channel_->worker_thread()->Post(RTC_FROM_HERE, this, MSG_MONITOR_START);
>-}
>-
>-void AudioMonitor::Stop() {
>-  voice_channel_->worker_thread()->Post(RTC_FROM_HERE, this, MSG_MONITOR_STOP);
>-}
>-
>-void AudioMonitor::OnMessage(rtc::Message *message) {
>-  rtc::CritScope cs(&crit_);
>-
>-  switch (message->message_id) {
>-  case MSG_MONITOR_START:
>-    assert(rtc::Thread::Current() == voice_channel_->worker_thread());
>-    if (!monitoring_) {
>-      monitoring_ = true;
>-      PollVoiceChannel();
>-    }
>-    break;
>-
>-  case MSG_MONITOR_STOP:
>-    assert(rtc::Thread::Current() == voice_channel_->worker_thread());
>-    if (monitoring_) {
>-      monitoring_ = false;
>-      voice_channel_->worker_thread()->Clear(this);
>-    }
>-    break;
>-
>-  case MSG_MONITOR_POLL:
>-    assert(rtc::Thread::Current() == voice_channel_->worker_thread());
>-    PollVoiceChannel();
>-    break;
>-
>-  case MSG_MONITOR_SIGNAL:
>-    {
>-      assert(rtc::Thread::Current() == monitoring_thread_);
>-      AudioInfo info = audio_info_;
>-      crit_.Leave();
>-      SignalUpdate(this, info);
>-      crit_.Enter();
>-    }
>-    break;
>-  }
>-}
>-
>-void AudioMonitor::PollVoiceChannel() {
>-  rtc::CritScope cs(&crit_);
>-  assert(rtc::Thread::Current() == voice_channel_->worker_thread());
>-
>-  // Gather connection infos
>-  audio_info_.input_level = voice_channel_->GetInputLevel_w();
>-  audio_info_.output_level = voice_channel_->GetOutputLevel_w();
>-  voice_channel_->GetActiveStreams_w(&audio_info_.active_streams);
>-
>-  // Signal the monitoring thread, start another poll timer
>-  monitoring_thread_->Post(RTC_FROM_HERE, this, MSG_MONITOR_SIGNAL);
>-  voice_channel_->worker_thread()->PostDelayed(RTC_FROM_HERE, rate_, this,
>-                                               MSG_MONITOR_POLL);
>-}
>-
>-VoiceChannel *AudioMonitor::voice_channel() {
>-  return voice_channel_;
>-}
>-
>-rtc::Thread *AudioMonitor::monitor_thread() {
>-  return monitoring_thread_;
>-}
>-
>-}  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/audiomonitor.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/audiomonitor.h
>deleted file mode 100644
>index 26212afc728..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/audiomonitor.h
>+++ /dev/null
>@@ -1,60 +0,0 @@
>-/*
>- *  Copyright 2004 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef PC_AUDIOMONITOR_H_
>-#define PC_AUDIOMONITOR_H_
>-
>-#include <vector>
>-#include <utility>
>-
>-#include "p2p/base/port.h"
>-#include "rtc_base/sigslot.h"
>-#include "rtc_base/thread.h"
>-
>-namespace cricket {
>-
>-class VoiceChannel;
>-
>-struct AudioInfo {
>-  int input_level;
>-  int output_level;
>-  typedef std::vector<std::pair<uint32_t, int> > StreamList;
>-  StreamList active_streams;  // ssrcs contributing to output_level
>-};
>-
>-class AudioMonitor : public rtc::MessageHandler,
>-    public sigslot::has_slots<> {
>- public:
>-  AudioMonitor(VoiceChannel* voice_channel, rtc::Thread *monitor_thread);
>-  ~AudioMonitor();
>-
>-  void Start(int cms);
>-  void Stop();
>-
>-  VoiceChannel* voice_channel();
>-  rtc::Thread *monitor_thread();
>-
>-  sigslot::signal2<AudioMonitor*, const AudioInfo&> SignalUpdate;
>-
>- protected:
>-  void OnMessage(rtc::Message *message);
>-  void PollVoiceChannel();
>-
>-  AudioInfo audio_info_;
>-  VoiceChannel* voice_channel_;
>-  rtc::Thread* monitoring_thread_;
>-  rtc::CriticalSection crit_;
>-  uint32_t rate_;
>-  bool monitoring_;
>-};
>-
>-}  // namespace cricket
>-
>-#endif  // PC_AUDIOMONITOR_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/bundlefilter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/bundlefilter.cc
>index 7791da62749..201d2bf95ca 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/bundlefilter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/bundlefilter.cc
>@@ -15,11 +15,9 @@
> 
> namespace cricket {
> 
>-BundleFilter::BundleFilter() {
>-}
>+BundleFilter::BundleFilter() {}
> 
>-BundleFilter::~BundleFilter() {
>-}
>+BundleFilter::~BundleFilter() {}
> 
> bool BundleFilter::DemuxPacket(const uint8_t* data, size_t len) {
>   // For RTP packets, we check whether the payload type can be found.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/bundlefilter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/bundlefilter.h
>index 7decbba8a4f..de477f457f2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/bundlefilter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/bundlefilter.h
>@@ -17,7 +17,7 @@
> #include <vector>
> 
> #include "media/base/streamparams.h"
>-#include "rtc_base/basictypes.h"
>+//#include "rtc_base/basictypes.h"
> 
> namespace cricket {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/bundlefilter_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/bundlefilter_unittest.cc
>index 2b1af5c3a39..e7cadbfbac6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/bundlefilter_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/bundlefilter_unittest.cc
>@@ -19,29 +19,30 @@ static const int kPayloadType3 = 0x33;
> 
> // SSRC = 0x1111, Payload type = 0x11
> static const unsigned char kRtpPacketPt1Ssrc1[] = {
>-    0x80, kPayloadType1, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x11,
>-    0x11,
>+    0x80, kPayloadType1, 0x00, 0x01, 0x00, 0x00,
>+    0x00, 0x00,          0x00, 0x00, 0x11, 0x11,
> };
> 
> // SSRC = 0x2222, Payload type = 0x22
> static const unsigned char kRtpPacketPt2Ssrc2[] = {
>-    0x80, 0x80 + kPayloadType2, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00,
>+    0x80, 0x80 + kPayloadType2,
>+    0x00, 0x01,
>+    0x00, 0x00,
>+    0x00, 0x00,
>+    0x00, 0x00,
>     0x22, 0x22,
> };
> 
> // SSRC = 0x2222, Payload type = 0x33
> static const unsigned char kRtpPacketPt3Ssrc2[] = {
>-    0x80, kPayloadType3, 0x00, 0x01, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x22,
>-    0x22,
>+    0x80, kPayloadType3, 0x00, 0x01, 0x00, 0x00,
>+    0x00, 0x00,          0x00, 0x00, 0x22, 0x22,
> };
> 
> // An SCTP packet.
> static const unsigned char kSctpPacket[] = {
>-    0x00, 0x01, 0x00, 0x01,
>-    0xff, 0xff, 0xff, 0xff,
>-    0x00, 0x00, 0x00, 0x00,
>-    0x03, 0x00, 0x00, 0x04,
>-    0x00, 0x00, 0x00, 0x00,
>+    0x00, 0x01, 0x00, 0x01, 0xff, 0xff, 0xff, 0xff, 0x00, 0x00,
>+    0x00, 0x00, 0x03, 0x00, 0x00, 0x04, 0x00, 0x00, 0x00, 0x00,
> };
> 
> TEST(BundleFilterTest, RtpPacketTest) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/channel.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/channel.cc
>index e1c103b39ba..281a4992f46 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/channel.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/channel.cc
>@@ -14,9 +14,11 @@
> 
> #include "pc/channel.h"
> 
>+#include "absl/memory/memory.h"
> #include "api/call/audio_sink.h"
> #include "media/base/mediaconstants.h"
> #include "media/base/rtputils.h"
>+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
> #include "rtc_base/bind.h"
> #include "rtc_base/byteorder.h"
> #include "rtc_base/checks.h"
>@@ -24,7 +26,6 @@
> #include "rtc_base/dscp.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/networkroute.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/trace_event.h"
> // Adding 'nogncheck' to disable the gn include headers check to support modular
> // WebRTC build targets.
>@@ -47,8 +48,7 @@ struct SendPacketMessageData : public rtc::MessageData {
> }  // namespace
> 
> enum {
>-  MSG_EARLYMEDIATIMEOUT = 1,
>-  MSG_SEND_RTP_PACKET,
>+  MSG_SEND_RTP_PACKET = 1,
>   MSG_SEND_RTCP_PACKET,
>   MSG_READYTOSENDDATA,
>   MSG_DATARECEIVED,
>@@ -99,28 +99,23 @@ BaseChannel::BaseChannel(rtc::Thread* worker_thread,
>                          rtc::Thread* signaling_thread,
>                          std::unique_ptr<MediaChannel> media_channel,
>                          const std::string& content_name,
>-                         bool rtcp_mux_required,
>-                         bool srtp_required)
>+                         bool srtp_required,
>+                         rtc::CryptoOptions crypto_options)
>     : worker_thread_(worker_thread),
>       network_thread_(network_thread),
>       signaling_thread_(signaling_thread),
>       content_name_(content_name),
>-      rtcp_mux_required_(rtcp_mux_required),
>-      unencrypted_rtp_transport_(
>-          rtc::MakeUnique<webrtc::RtpTransport>(rtcp_mux_required)),
>       srtp_required_(srtp_required),
>+      crypto_options_(crypto_options),
>       media_channel_(std::move(media_channel)) {
>   RTC_DCHECK_RUN_ON(worker_thread_);
>-  rtp_transport_ = unencrypted_rtp_transport_.get();
>-  ConnectToRtpTransport();
>+  demuxer_criteria_.mid = content_name;
>   RTC_LOG(LS_INFO) << "Created channel for " << content_name;
> }
> 
> BaseChannel::~BaseChannel() {
>   TRACE_EVENT0("webrtc", "BaseChannel::~BaseChannel");
>   RTC_DCHECK_RUN_ON(worker_thread_);
>-  Deinit();
>-  StopConnectionMonitor();
>   // Eats any outstanding messages or packets.
>   worker_thread_->Clear(&invoker_);
>   worker_thread_->Clear(this);
>@@ -131,60 +126,38 @@ BaseChannel::~BaseChannel() {
>   RTC_LOG(LS_INFO) << "Destroyed channel: " << content_name_;
> }
> 
>-void BaseChannel::ConnectToRtpTransport() {
>+bool BaseChannel::ConnectToRtpTransport() {
>   RTC_DCHECK(rtp_transport_);
>+  if (!RegisterRtpDemuxerSink()) {
>+    return false;
>+  }
>   rtp_transport_->SignalReadyToSend.connect(
>       this, &BaseChannel::OnTransportReadyToSend);
>-  // TODO(zstein):  RtpTransport::SignalPacketReceived will probably be replaced
>-  // with a callback interface later so that the demuxer can select which
>-  // channel to signal.
>-  rtp_transport_->SignalPacketReceived.connect(this,
>-                                               &BaseChannel::OnPacketReceived);
>+  rtp_transport_->SignalRtcpPacketReceived.connect(
>+      this, &BaseChannel::OnRtcpPacketReceived);
>   rtp_transport_->SignalNetworkRouteChanged.connect(
>       this, &BaseChannel::OnNetworkRouteChanged);
>   rtp_transport_->SignalWritableState.connect(this,
>                                               &BaseChannel::OnWritableState);
>   rtp_transport_->SignalSentPacket.connect(this,
>                                            &BaseChannel::SignalSentPacket_n);
>+  return true;
> }
> 
> void BaseChannel::DisconnectFromRtpTransport() {
>   RTC_DCHECK(rtp_transport_);
>+  rtp_transport_->UnregisterRtpDemuxerSink(this);
>   rtp_transport_->SignalReadyToSend.disconnect(this);
>-  rtp_transport_->SignalPacketReceived.disconnect(this);
>+  rtp_transport_->SignalRtcpPacketReceived.disconnect(this);
>   rtp_transport_->SignalNetworkRouteChanged.disconnect(this);
>   rtp_transport_->SignalWritableState.disconnect(this);
>   rtp_transport_->SignalSentPacket.disconnect(this);
> }
> 
>-void BaseChannel::Init_w(DtlsTransportInternal* rtp_dtls_transport,
>-                         DtlsTransportInternal* rtcp_dtls_transport,
>-                         rtc::PacketTransportInternal* rtp_packet_transport,
>-                         rtc::PacketTransportInternal* rtcp_packet_transport) {
>-  RTC_DCHECK_RUN_ON(worker_thread_);
>-  network_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
>-    SetTransports_n(rtp_dtls_transport, rtcp_dtls_transport,
>-                    rtp_packet_transport, rtcp_packet_transport);
>-
>-    if (rtcp_mux_required_) {
>-      rtcp_mux_filter_.SetActive();
>-    }
>-  });
>-
>-  // Both RTP and RTCP channels should be set, we can call SetInterface on
>-  // the media channel and it can set network options.
>-  media_channel_->SetInterface(this);
>-}
>-
> void BaseChannel::Init_w(webrtc::RtpTransportInternal* rtp_transport) {
>   RTC_DCHECK_RUN_ON(worker_thread_);
>-  network_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
>-    SetRtpTransport(rtp_transport);
>-
>-    if (rtcp_mux_required_) {
>-      rtcp_mux_filter_.SetActive();
>-    }
>-  });
>+  network_thread_->Invoke<void>(
>+      RTC_FROM_HERE, [this, rtp_transport] { SetRtpTransport(rtp_transport); });
> 
>   // Both RTP and RTCP channels should be set, we can call SetInterface on
>   // the media channel and it can set network options.
>@@ -200,11 +173,8 @@ void BaseChannel::Deinit() {
>   network_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
>     FlushRtcpMessages_n();
> 
>-    if (dtls_srtp_transport_) {
>-      dtls_srtp_transport_->SetDtlsTransports(nullptr, nullptr);
>-    } else {
>-      rtp_transport_->SetRtpPacketTransport(nullptr);
>-      rtp_transport_->SetRtcpPacketTransport(nullptr);
>+    if (rtp_transport_) {
>+      DisconnectFromRtpTransport();
>     }
>     // Clear pending read packets/messages.
>     network_thread_->Clear(&invoker_);
>@@ -212,148 +182,46 @@ void BaseChannel::Deinit() {
>   });
> }
> 
>-void BaseChannel::SetRtpTransport(webrtc::RtpTransportInternal* rtp_transport) {
>+bool BaseChannel::SetRtpTransport(webrtc::RtpTransportInternal* rtp_transport) {
>+  if (rtp_transport == rtp_transport_) {
>+    return true;
>+  }
>+
>   if (!network_thread_->IsCurrent()) {
>-    network_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
>-      SetRtpTransport(rtp_transport);
>-      return;
>+    return network_thread_->Invoke<bool>(RTC_FROM_HERE, [this, rtp_transport] {
>+      return SetRtpTransport(rtp_transport);
>     });
>   }
> 
>-  RTC_DCHECK(rtp_transport);
>-
>   if (rtp_transport_) {
>     DisconnectFromRtpTransport();
>   }
>-  rtp_transport_ = rtp_transport;
>-  RTC_LOG(LS_INFO) << "Setting the RtpTransport for " << content_name();
>-  ConnectToRtpTransport();
>-
>-  UpdateWritableState_n();
>-}
> 
>-void BaseChannel::SetTransports(DtlsTransportInternal* rtp_dtls_transport,
>-                                DtlsTransportInternal* rtcp_dtls_transport) {
>-  network_thread_->Invoke<void>(
>-      RTC_FROM_HERE,
>-      Bind(&BaseChannel::SetTransports_n, this, rtp_dtls_transport,
>-           rtcp_dtls_transport, rtp_dtls_transport, rtcp_dtls_transport));
>-}
>-
>-void BaseChannel::SetTransports(
>-    rtc::PacketTransportInternal* rtp_packet_transport,
>-    rtc::PacketTransportInternal* rtcp_packet_transport) {
>-  network_thread_->Invoke<void>(
>-      RTC_FROM_HERE, Bind(&BaseChannel::SetTransports_n, this, nullptr, nullptr,
>-                          rtp_packet_transport, rtcp_packet_transport));
>-}
>-
>-void BaseChannel::SetTransports_n(
>-    DtlsTransportInternal* rtp_dtls_transport,
>-    DtlsTransportInternal* rtcp_dtls_transport,
>-    rtc::PacketTransportInternal* rtp_packet_transport,
>-    rtc::PacketTransportInternal* rtcp_packet_transport) {
>-  RTC_DCHECK(network_thread_->IsCurrent());
>-  // Validate some assertions about the input.
>-  RTC_DCHECK(rtp_packet_transport);
>-  RTC_DCHECK_EQ(NeedsRtcpTransport(), rtcp_packet_transport != nullptr);
>-  if (rtp_dtls_transport || rtcp_dtls_transport) {
>-    // DTLS/non-DTLS pointers should be to the same object.
>-    RTC_DCHECK(rtp_dtls_transport == rtp_packet_transport);
>-    RTC_DCHECK(rtcp_dtls_transport == rtcp_packet_transport);
>-    // Can't go from non-DTLS to DTLS.
>-    RTC_DCHECK(!rtp_transport_->rtp_packet_transport() || rtp_dtls_transport_);
>-  } else {
>-    // Can't go from DTLS to non-DTLS.
>-    RTC_DCHECK(!rtp_dtls_transport_);
>-  }
>-  // Transport names should be the same.
>-  if (rtp_dtls_transport && rtcp_dtls_transport) {
>-    RTC_DCHECK(rtp_dtls_transport->transport_name() ==
>-               rtcp_dtls_transport->transport_name());
>-  }
>-
>-  if (rtp_packet_transport == rtp_transport_->rtp_packet_transport()) {
>-    // Nothing to do if transport isn't changing.
>-    return;
>-  }
>-
>-  std::string debug_name;
>-  if (rtp_dtls_transport) {
>-    transport_name_ = rtp_dtls_transport->transport_name();
>-    debug_name = transport_name_;
>-  } else {
>-    debug_name = rtp_packet_transport->transport_name();
>-  }
>-  // If this BaseChannel doesn't require RTCP mux and we haven't fully
>-  // negotiated RTCP mux, we need an RTCP transport.
>-  if (rtcp_packet_transport) {
>-    RTC_LOG(LS_INFO) << "Setting RTCP Transport for " << content_name()
>-                     << " on " << debug_name << " transport "
>-                     << rtcp_packet_transport;
>-    SetTransport_n(/*rtcp=*/true, rtcp_dtls_transport, rtcp_packet_transport);
>-  }
>-
>-  RTC_LOG(LS_INFO) << "Setting RTP Transport for " << content_name() << " on "
>-                   << debug_name << " transport " << rtp_packet_transport;
>-  SetTransport_n(/*rtcp=*/false, rtp_dtls_transport, rtp_packet_transport);
>-
>-  // Set DtlsTransport/PacketTransport for RTP-level transport.
>-  if ((rtp_dtls_transport_ || rtcp_dtls_transport_) && dtls_srtp_transport_) {
>-    // When setting the transport with non-null |dtls_srtp_transport_|, we are
>-    // using DTLS-SRTP. This could happen for bundling. If the
>-    // |dtls_srtp_transport| is null, we cannot tell if it doing DTLS-SRTP or
>-    // SDES until the description is set. So don't call |EnableDtlsSrtp_n| here.
>-    dtls_srtp_transport_->SetDtlsTransports(rtp_dtls_transport,
>-                                            rtcp_dtls_transport);
>-  } else {
>-    rtp_transport_->SetRtpPacketTransport(rtp_packet_transport);
>-    rtp_transport_->SetRtcpPacketTransport(rtcp_packet_transport);
>-  }
>-
>-  // Update aggregate writable/ready-to-send state between RTP and RTCP upon
>-  // setting new transport channels.
>-  UpdateWritableState_n();
>-}
>-
>-void BaseChannel::SetTransport_n(
>-    bool rtcp,
>-    DtlsTransportInternal* new_dtls_transport,
>-    rtc::PacketTransportInternal* new_packet_transport) {
>-  RTC_DCHECK(network_thread_->IsCurrent());
>-  if (new_dtls_transport) {
>-    RTC_DCHECK(new_dtls_transport == new_packet_transport);
>-  }
>-  DtlsTransportInternal*& old_dtls_transport =
>-      rtcp ? rtcp_dtls_transport_ : rtp_dtls_transport_;
>-  rtc::PacketTransportInternal* old_packet_transport =
>-      rtcp ? rtp_transport_->rtcp_packet_transport()
>-           : rtp_transport_->rtp_packet_transport();
>-
>-  if (!old_packet_transport && !new_packet_transport) {
>-    // Nothing to do.
>-    return;
>-  }
>-
>-  RTC_DCHECK(old_packet_transport != new_packet_transport);
>-
>-  old_dtls_transport = new_dtls_transport;
>-
>-  // If there's no new transport, we're done.
>-  if (!new_packet_transport) {
>-    return;
>-  }
>+  rtp_transport_ = rtp_transport;
>+  if (rtp_transport_) {
>+    RTC_DCHECK(rtp_transport_->rtp_packet_transport());
>+    transport_name_ = rtp_transport_->rtp_packet_transport()->transport_name();
> 
>-  if (rtcp && new_dtls_transport) {
>-    RTC_CHECK(!(ShouldSetupDtlsSrtp_n() && srtp_active()))
>-        << "Setting RTCP for DTLS/SRTP after the DTLS is active "
>-        << "should never happen.";
>-  }
>+    if (!ConnectToRtpTransport()) {
>+      RTC_LOG(LS_ERROR) << "Failed to connect to the new RtpTransport.";
>+      return false;
>+    }
>+    OnTransportReadyToSend(rtp_transport_->IsReadyToSend());
>+    UpdateWritableState_n();
> 
>-  auto& socket_options = rtcp ? rtcp_socket_options_ : socket_options_;
>-  for (const auto& pair : socket_options) {
>-    new_packet_transport->SetOption(pair.first, pair.second);
>+    // Set the cached socket options.
>+    for (const auto& pair : socket_options_) {
>+      rtp_transport_->rtp_packet_transport()->SetOption(pair.first,
>+                                                        pair.second);
>+    }
>+    if (rtp_transport_->rtcp_packet_transport()) {
>+      for (const auto& pair : rtcp_socket_options_) {
>+        rtp_transport_->rtp_packet_transport()->SetOption(pair.first,
>+                                                          pair.second);
>+      }
>+    }
>   }
>+  return true;
> }
> 
> bool BaseChannel::Enable(bool enable) {
>@@ -365,11 +233,19 @@ bool BaseChannel::Enable(bool enable) {
> }
> 
> bool BaseChannel::AddRecvStream(const StreamParams& sp) {
>+  demuxer_criteria_.ssrcs.insert(sp.first_ssrc());
>+  if (!RegisterRtpDemuxerSink()) {
>+    return false;
>+  }
>   return InvokeOnWorker<bool>(RTC_FROM_HERE,
>                               Bind(&BaseChannel::AddRecvStream_w, this, sp));
> }
> 
> bool BaseChannel::RemoveRecvStream(uint32_t ssrc) {
>+  demuxer_criteria_.ssrcs.erase(ssrc);
>+  if (!RegisterRtpDemuxerSink()) {
>+    return false;
>+  }
>   return InvokeOnWorker<bool>(
>       RTC_FROM_HERE, Bind(&BaseChannel::RemoveRecvStream_w, this, ssrc));
> }
>@@ -403,41 +279,6 @@ bool BaseChannel::SetRemoteContent(const MediaContentDescription* content,
>       Bind(&BaseChannel::SetRemoteContent_w, this, content, type, error_desc));
> }
> 
>-void BaseChannel::StartConnectionMonitor(int cms) {
>-  // We pass in the BaseChannel instead of the rtp_dtls_transport_
>-  // because if the rtp_dtls_transport_ changes, the ConnectionMonitor
>-  // would be pointing to the wrong TransportChannel.
>-  // We pass in the network thread because on that thread connection monitor
>-  // will call BaseChannel::GetConnectionStats which must be called on the
>-  // network thread.
>-  connection_monitor_.reset(
>-      new ConnectionMonitor(this, network_thread(), rtc::Thread::Current()));
>-  connection_monitor_->SignalUpdate.connect(
>-      this, &BaseChannel::OnConnectionMonitorUpdate);
>-  connection_monitor_->Start(cms);
>-}
>-
>-void BaseChannel::StopConnectionMonitor() {
>-  if (connection_monitor_) {
>-    connection_monitor_->Stop();
>-    connection_monitor_.reset();
>-  }
>-}
>-
>-bool BaseChannel::GetConnectionStats(ConnectionInfos* infos) {
>-  RTC_DCHECK(network_thread_->IsCurrent());
>-  if (!rtp_dtls_transport_) {
>-    return false;
>-  }
>-  return rtp_dtls_transport_->ice_transport()->GetStats(infos);
>-}
>-
>-bool BaseChannel::NeedsRtcpTransport() {
>-  // If this BaseChannel doesn't require RTCP mux and we haven't fully
>-  // negotiated RTCP mux, we need an RTCP transport.
>-  return !rtcp_mux_required_ && !rtcp_mux_filter_.IsFullyActive();
>-}
>-
> bool BaseChannel::IsReadyToReceiveMedia_w() const {
>   // Receive data if we are enabled and have local content,
>   return enabled() &&
>@@ -456,7 +297,7 @@ bool BaseChannel::IsReadyToSendMedia_n() const {
>   return enabled() &&
>          webrtc::RtpTransceiverDirectionHasRecv(remote_content_direction_) &&
>          webrtc::RtpTransceiverDirectionHasSend(local_content_direction_) &&
>-         was_ever_writable() && (srtp_active() || !ShouldSetupDtlsSrtp_n());
>+         was_ever_writable();
> }
> 
> bool BaseChannel::SendPacket(rtc::CopyOnWriteBuffer* packet,
>@@ -469,7 +310,8 @@ bool BaseChannel::SendRtcp(rtc::CopyOnWriteBuffer* packet,
>   return SendPacket(true, packet, options);
> }
> 
>-int BaseChannel::SetOption(SocketType type, rtc::Socket::Option opt,
>+int BaseChannel::SetOption(SocketType type,
>+                           rtc::Socket::Option opt,
>                            int value) {
>   return network_thread_->Invoke<int>(
>       RTC_FROM_HERE, Bind(&BaseChannel::SetOption_n, this, type, opt, value));
>@@ -479,6 +321,7 @@ int BaseChannel::SetOption_n(SocketType type,
>                              rtc::Socket::Option opt,
>                              int value) {
>   RTC_DCHECK(network_thread_->IsCurrent());
>+  RTC_DCHECK(rtp_transport_);
>   rtc::PacketTransportInternal* transport = nullptr;
>   switch (type) {
>     case ST_RTP:
>@@ -498,11 +341,6 @@ int BaseChannel::SetOption_n(SocketType type,
> void BaseChannel::OnWritableState(bool writable) {
>   RTC_DCHECK(network_thread_->IsCurrent());
>   if (writable) {
>-    // This is used to cover the scenario when the DTLS handshake is completed
>-    // and DtlsTransport becomes writable before the remote description is set.
>-    if (ShouldSetupDtlsSrtp_n()) {
>-      EnableDtlsSrtp_n();
>-    }
>     ChannelWritable_n();
>   } else {
>     ChannelNotWritable_n();
>@@ -510,7 +348,7 @@ void BaseChannel::OnWritableState(bool writable) {
> }
> 
> void BaseChannel::OnNetworkRouteChanged(
>-    rtc::Optional<rtc::NetworkRoute> network_route) {
>+    absl::optional<rtc::NetworkRoute> network_route) {
>   RTC_DCHECK(network_thread_->IsCurrent());
>   rtc::NetworkRoute new_route;
>   if (network_route) {
>@@ -549,13 +387,14 @@ bool BaseChannel::SendPacket(bool rtcp,
>     network_thread_->Post(RTC_FROM_HERE, this, message_id, data);
>     return true;
>   }
>+
>   TRACE_EVENT0("webrtc", "BaseChannel::SendPacket");
> 
>   // Now that we are on the correct thread, ensure we have a place to send this
>   // packet before doing anything. (We might get RTCP packets that we don't
>   // intend to send.) If we've negotiated RTCP mux, send RTCP over the RTP
>   // transport.
>-  if (!rtp_transport_->IsWritable(rtcp)) {
>+  if (!rtp_transport_ || !rtp_transport_->IsWritable(rtcp)) {
>     return false;
>   }
> 
>@@ -587,22 +426,57 @@ bool BaseChannel::SendPacket(bool rtcp,
>     std::string packet_type = rtcp ? "RTCP" : "RTP";
>     RTC_LOG(LS_WARNING) << "Sending an " << packet_type
>                         << " packet without encryption.";
>-  } else {
>-    // Make sure we didn't accidentally send any packets without encryption.
>-    RTC_DCHECK(rtp_transport_ == sdes_transport_.get() ||
>-               rtp_transport_ == dtls_srtp_transport_.get());
>   }
>+
>   // Bon voyage.
>   return rtcp ? rtp_transport_->SendRtcpPacket(packet, options, PF_SRTP_BYPASS)
>               : rtp_transport_->SendRtpPacket(packet, options, PF_SRTP_BYPASS);
> }
> 
>-bool BaseChannel::HandlesPayloadType(int packet_type) const {
>-  return rtp_transport_->HandlesPayloadType(packet_type);
>+void BaseChannel::OnRtpPacket(const webrtc::RtpPacketReceived& parsed_packet) {
>+  // Reconstruct the PacketTime from the |parsed_packet|.
>+  // RtpPacketReceived.arrival_time_ms = (PacketTime + 500) / 1000;
>+  // Note: The |not_before| field is always 0 here. This field is not currently
>+  // used, so it should be fine.
>+  int64_t timestamp = -1;
>+  if (parsed_packet.arrival_time_ms() > 0) {
>+    timestamp = parsed_packet.arrival_time_ms() * 1000;
>+  }
>+  rtc::PacketTime packet_time(timestamp, /*not_before=*/0);
>+
>+  OnPacketReceived(/*rtcp=*/false, parsed_packet.Buffer(), packet_time);
>+}
>+
>+void BaseChannel::UpdateRtpHeaderExtensionMap(
>+    const RtpHeaderExtensions& header_extensions) {
>+  RTC_DCHECK(rtp_transport_);
>+  // Update the header extension map on network thread in case there is data
>+  // race.
>+  // TODO(zhihuang): Add an rtc::ThreadChecker make sure to RtpTransport won't
>+  // be accessed from different threads.
>+  //
>+  // NOTE: This doesn't take the BUNDLE case in account meaning the RTP header
>+  // extension maps are not merged when BUNDLE is enabled. This is fine because
>+  // the ID for MID should be consistent among all the RTP transports.
>+  network_thread_->Invoke<void>(RTC_FROM_HERE, [this, &header_extensions] {
>+    rtp_transport_->UpdateRtpHeaderExtensionMap(header_extensions);
>+  });
>+}
>+
>+bool BaseChannel::RegisterRtpDemuxerSink() {
>+  RTC_DCHECK(rtp_transport_);
>+  return network_thread_->Invoke<bool>(RTC_FROM_HERE, [this] {
>+    return rtp_transport_->RegisterRtpDemuxerSink(demuxer_criteria_, this);
>+  });
>+}
>+
>+void BaseChannel::OnRtcpPacketReceived(rtc::CopyOnWriteBuffer* packet,
>+                                       const rtc::PacketTime& packet_time) {
>+  OnPacketReceived(/*rtcp=*/true, *packet, packet_time);
> }
> 
> void BaseChannel::OnPacketReceived(bool rtcp,
>-                                   rtc::CopyOnWriteBuffer* packet,
>+                                   const rtc::CopyOnWriteBuffer& packet,
>                                    const rtc::PacketTime& packet_time) {
>   if (!has_received_packet_ && !rtcp) {
>     has_received_packet_ = true;
>@@ -629,7 +503,7 @@ void BaseChannel::OnPacketReceived(bool rtcp,
> 
>   invoker_.AsyncInvoke<void>(
>       RTC_FROM_HERE, worker_thread_,
>-      Bind(&BaseChannel::ProcessPacket, this, rtcp, *packet, packet_time));
>+      Bind(&BaseChannel::ProcessPacket, this, rtcp, packet, packet_time));
> }
> 
> void BaseChannel::ProcessPacket(bool rtcp,
>@@ -668,12 +542,8 @@ void BaseChannel::DisableMedia_w() {
> }
> 
> void BaseChannel::UpdateWritableState_n() {
>-  rtc::PacketTransportInternal* rtp_packet_transport =
>-      rtp_transport_->rtp_packet_transport();
>-  rtc::PacketTransportInternal* rtcp_packet_transport =
>-      rtp_transport_->rtcp_packet_transport();
>-  if (rtp_packet_transport && rtp_packet_transport->writable() &&
>-      (!rtcp_packet_transport || rtcp_packet_transport->writable())) {
>+  if (rtp_transport_->IsWritable(/*rtcp=*/true) &&
>+      rtp_transport_->IsWritable(/*rtcp=*/false)) {
>     ChannelWritable_n();
>   } else {
>     ChannelNotWritable_n();
>@@ -694,11 +564,6 @@ void BaseChannel::ChannelWritable_n() {
>   UpdateMediaSendRecvState();
> }
> 
>-bool BaseChannel::ShouldSetupDtlsSrtp_n() const {
>-  // Since DTLS is applied to all transports, checking RTP should be enough.
>-  return rtp_dtls_transport_ && rtp_dtls_transport_->IsDtlsActive();
>-}
>-
> void BaseChannel::ChannelNotWritable_n() {
>   RTC_DCHECK(network_thread_->IsCurrent());
>   if (!writable_)
>@@ -709,238 +574,6 @@ void BaseChannel::ChannelNotWritable_n() {
>   UpdateMediaSendRecvState();
> }
> 
>-bool BaseChannel::SetRtpTransportParameters(
>-    const MediaContentDescription* content,
>-    SdpType type,
>-    ContentSource src,
>-    const RtpHeaderExtensions& extensions,
>-    std::string* error_desc) {
>-  std::vector<int> encrypted_extension_ids;
>-  for (const webrtc::RtpExtension& extension : extensions) {
>-    if (extension.encrypt) {
>-      RTC_LOG(LS_INFO) << "Using " << (src == CS_LOCAL ? "local" : "remote")
>-                       << " encrypted extension: " << extension.ToString();
>-      encrypted_extension_ids.push_back(extension.id);
>-    }
>-  }
>-
>-  // Cache srtp_required_ for belt and suspenders check on SendPacket
>-  return network_thread_->Invoke<bool>(
>-      RTC_FROM_HERE,
>-      Bind(&BaseChannel::SetRtpTransportParameters_n, this, content, type, src,
>-           encrypted_extension_ids, error_desc));
>-}
>-
>-bool BaseChannel::SetRtpTransportParameters_n(
>-    const MediaContentDescription* content,
>-    SdpType type,
>-    ContentSource src,
>-    const std::vector<int>& encrypted_extension_ids,
>-    std::string* error_desc) {
>-  RTC_DCHECK(network_thread_->IsCurrent());
>-
>-  if (!SetSrtp_n(content->cryptos(), type, src, encrypted_extension_ids,
>-                 error_desc)) {
>-    return false;
>-  }
>-
>-  if (!SetRtcpMux_n(content->rtcp_mux(), type, src, error_desc)) {
>-    return false;
>-  }
>-
>-  return true;
>-}
>-
>-// |dtls| will be set to true if DTLS is active for transport and crypto is
>-// empty.
>-bool BaseChannel::CheckSrtpConfig_n(const std::vector<CryptoParams>& cryptos,
>-                                    bool* dtls,
>-                                    std::string* error_desc) {
>-  *dtls = rtp_dtls_transport_ && rtp_dtls_transport_->IsDtlsActive();
>-  if (*dtls && !cryptos.empty()) {
>-    SafeSetError("Cryptos must be empty when DTLS is active.", error_desc);
>-    return false;
>-  }
>-  return true;
>-}
>-
>-void BaseChannel::EnableSdes_n() {
>-  if (sdes_transport_) {
>-    return;
>-  }
>-  // DtlsSrtpTransport and SrtpTransport shouldn't be enabled at the same
>-  // time.
>-  RTC_DCHECK(!dtls_srtp_transport_);
>-  RTC_DCHECK(unencrypted_rtp_transport_);
>-  sdes_transport_ = rtc::MakeUnique<webrtc::SrtpTransport>(
>-      std::move(unencrypted_rtp_transport_));
>-#if defined(ENABLE_EXTERNAL_AUTH)
>-  sdes_transport_->EnableExternalAuth();
>-#endif
>-  SetRtpTransport(sdes_transport_.get());
>-  RTC_LOG(LS_INFO) << "Wrapping RtpTransport in SrtpTransport.";
>-}
>-
>-void BaseChannel::EnableDtlsSrtp_n() {
>-  if (dtls_srtp_transport_) {
>-    return;
>-  }
>-  // DtlsSrtpTransport and SrtpTransport shouldn't be enabled at the same
>-  // time.
>-  RTC_DCHECK(!sdes_transport_);
>-  RTC_DCHECK(unencrypted_rtp_transport_);
>-
>-  auto srtp_transport = rtc::MakeUnique<webrtc::SrtpTransport>(
>-      std::move(unencrypted_rtp_transport_));
>-#if defined(ENABLE_EXTERNAL_AUTH)
>-  srtp_transport->EnableExternalAuth();
>-#endif
>-  dtls_srtp_transport_ =
>-      rtc::MakeUnique<webrtc::DtlsSrtpTransport>(std::move(srtp_transport));
>-
>-  SetRtpTransport(dtls_srtp_transport_.get());
>-  if (cached_send_extension_ids_) {
>-    dtls_srtp_transport_->UpdateSendEncryptedHeaderExtensionIds(
>-        *cached_send_extension_ids_);
>-  }
>-  if (cached_recv_extension_ids_) {
>-    dtls_srtp_transport_->UpdateRecvEncryptedHeaderExtensionIds(
>-        *cached_recv_extension_ids_);
>-  }
>-  // Set the DtlsTransport and the |dtls_srtp_transport_| will handle the DTLS
>-  // relate signal internally.
>-  RTC_DCHECK(rtp_dtls_transport_);
>-  dtls_srtp_transport_->SetDtlsTransports(rtp_dtls_transport_,
>-                                          rtcp_dtls_transport_);
>-
>-  RTC_LOG(LS_INFO) << "Wrapping SrtpTransport in DtlsSrtpTransport.";
>-}
>-
>-bool BaseChannel::SetSrtp_n(const std::vector<CryptoParams>& cryptos,
>-                            SdpType type,
>-                            ContentSource src,
>-                            const std::vector<int>& encrypted_extension_ids,
>-                            std::string* error_desc) {
>-  TRACE_EVENT0("webrtc", "BaseChannel::SetSrtp_w");
>-  bool ret = false;
>-  bool dtls = false;
>-  ret = CheckSrtpConfig_n(cryptos, &dtls, error_desc);
>-  if (!ret) {
>-    return false;
>-  }
>-
>-  // If SRTP was not required, but we're setting a description that uses SDES,
>-  // we need to upgrade to an SrtpTransport.
>-  if (!sdes_transport_ && !dtls && !cryptos.empty()) {
>-    EnableSdes_n();
>-  }
>-
>-  if ((type == SdpType::kAnswer || type == SdpType::kPrAnswer) && dtls) {
>-    EnableDtlsSrtp_n();
>-  }
>-
>-  UpdateEncryptedHeaderExtensionIds(src, encrypted_extension_ids);
>-
>-  if (!dtls) {
>-    switch (type) {
>-      case SdpType::kOffer:
>-        ret = sdes_negotiator_.SetOffer(cryptos, src);
>-        break;
>-      case SdpType::kPrAnswer:
>-        ret = sdes_negotiator_.SetProvisionalAnswer(cryptos, src);
>-        break;
>-      case SdpType::kAnswer:
>-        ret = sdes_negotiator_.SetAnswer(cryptos, src);
>-        break;
>-      default:
>-        break;
>-    }
>-
>-    // If setting an SDES answer succeeded, apply the negotiated parameters
>-    // to the SRTP transport.
>-    if ((type == SdpType::kPrAnswer || type == SdpType::kAnswer) && ret) {
>-      if (sdes_negotiator_.send_cipher_suite() &&
>-          sdes_negotiator_.recv_cipher_suite()) {
>-        RTC_DCHECK(cached_send_extension_ids_);
>-        RTC_DCHECK(cached_recv_extension_ids_);
>-        ret = sdes_transport_->SetRtpParams(
>-            *(sdes_negotiator_.send_cipher_suite()),
>-            sdes_negotiator_.send_key().data(),
>-            static_cast<int>(sdes_negotiator_.send_key().size()),
>-            *(cached_send_extension_ids_),
>-            *(sdes_negotiator_.recv_cipher_suite()),
>-            sdes_negotiator_.recv_key().data(),
>-            static_cast<int>(sdes_negotiator_.recv_key().size()),
>-            *(cached_recv_extension_ids_));
>-      } else {
>-        RTC_LOG(LS_INFO) << "No crypto keys are provided for SDES.";
>-        if (type == SdpType::kAnswer && sdes_transport_) {
>-          // Explicitly reset the |sdes_transport_| if no crypto param is
>-          // provided in the answer. No need to call |ResetParams()| for
>-          // |sdes_negotiator_| because it resets the params inside |SetAnswer|.
>-          sdes_transport_->ResetParams();
>-        }
>-      }
>-    }
>-  }
>-
>-  if (!ret) {
>-    SafeSetError("Failed to setup SRTP.", error_desc);
>-    return false;
>-  }
>-  return true;
>-}
>-
>-bool BaseChannel::SetRtcpMux_n(bool enable,
>-                               SdpType type,
>-                               ContentSource src,
>-                               std::string* error_desc) {
>-  // Provide a more specific error message for the RTCP mux "require" policy
>-  // case.
>-  if (rtcp_mux_required_ && !enable) {
>-    SafeSetError(
>-        "rtcpMuxPolicy is 'require', but media description does not "
>-        "contain 'a=rtcp-mux'.",
>-        error_desc);
>-    return false;
>-  }
>-  bool ret = false;
>-  switch (type) {
>-    case SdpType::kOffer:
>-      ret = rtcp_mux_filter_.SetOffer(enable, src);
>-      break;
>-    case SdpType::kPrAnswer:
>-      // This may activate RTCP muxing, but we don't yet destroy the transport
>-      // because the final answer may deactivate it.
>-      ret = rtcp_mux_filter_.SetProvisionalAnswer(enable, src);
>-      break;
>-    case SdpType::kAnswer:
>-      ret = rtcp_mux_filter_.SetAnswer(enable, src);
>-      if (ret && rtcp_mux_filter_.IsActive()) {
>-        ActivateRtcpMux();
>-      }
>-      break;
>-    default:
>-      break;
>-  }
>-  if (!ret) {
>-    SafeSetError("Failed to setup RTCP mux filter.", error_desc);
>-    return false;
>-  }
>-  rtp_transport_->SetRtcpMuxEnabled(rtcp_mux_filter_.IsActive());
>-  // |rtcp_mux_filter_| can be active if |action| is SdpType::kPrAnswer or
>-  // SdpType::kAnswer, but we only want to tear down the RTCP transport if we
>-  // received a final answer.
>-  if (rtcp_mux_filter_.IsActive()) {
>-    // If the RTP transport is already writable, then so are we.
>-    if (rtp_transport_->rtp_packet_transport()->writable()) {
>-      ChannelWritable_n();
>-    }
>-  }
>-
>-  return true;
>-}
>-
> bool BaseChannel::AddRecvStream_w(const StreamParams& sp) {
>   RTC_DCHECK(worker_thread() == rtc::Thread::Current());
>   return media_channel()->AddRecvStream(sp);
>@@ -958,11 +591,11 @@ bool BaseChannel::UpdateLocalStreams_w(const std::vector<StreamParams>& streams,
>   bool ret = true;
>   for (StreamParamsVec::const_iterator it = local_streams_.begin();
>        it != local_streams_.end(); ++it) {
>-    if (!GetStreamBySsrc(streams, it->first_ssrc())) {
>+    if (it->has_ssrcs() && !GetStreamBySsrc(streams, it->first_ssrc())) {
>       if (!media_channel()->RemoveSendStream(it->first_ssrc())) {
>         std::ostringstream desc;
>-        desc << "Failed to remove send stream with ssrc "
>-             << it->first_ssrc() << ".";
>+        desc << "Failed to remove send stream with ssrc " << it->first_ssrc()
>+             << ".";
>         SafeSetError(desc.str(), error_desc);
>         ret = false;
>       }
>@@ -971,7 +604,7 @@ bool BaseChannel::UpdateLocalStreams_w(const std::vector<StreamParams>& streams,
>   // Check for new streams.
>   for (StreamParamsVec::const_iterator it = streams.begin();
>        it != streams.end(); ++it) {
>-    if (!GetStreamBySsrc(local_streams_, it->first_ssrc())) {
>+    if (it->has_ssrcs() && !GetStreamBySsrc(local_streams_, it->first_ssrc())) {
>       if (media_channel()->AddSendStream(*it)) {
>         RTC_LOG(LS_INFO) << "Add send stream ssrc: " << it->ssrcs[0];
>       } else {
>@@ -994,22 +627,32 @@ bool BaseChannel::UpdateRemoteStreams_w(
>   bool ret = true;
>   for (StreamParamsVec::const_iterator it = remote_streams_.begin();
>        it != remote_streams_.end(); ++it) {
>-    if (!GetStreamBySsrc(streams, it->first_ssrc())) {
>-      if (!RemoveRecvStream_w(it->first_ssrc())) {
>+    // If we no longer have an unsignaled stream, we would like to remove
>+    // the unsignaled stream params that are cached.
>+    if ((!it->has_ssrcs() && !HasStreamWithNoSsrcs(streams)) ||
>+        !GetStreamBySsrc(streams, it->first_ssrc())) {
>+      if (RemoveRecvStream_w(it->first_ssrc())) {
>+        RTC_LOG(LS_INFO) << "Remove remote ssrc: " << it->first_ssrc();
>+      } else {
>         std::ostringstream desc;
>-        desc << "Failed to remove remote stream with ssrc "
>-             << it->first_ssrc() << ".";
>+        desc << "Failed to remove remote stream with ssrc " << it->first_ssrc()
>+             << ".";
>         SafeSetError(desc.str(), error_desc);
>         ret = false;
>       }
>     }
>   }
>+  demuxer_criteria_.ssrcs.clear();
>   // Check for new streams.
>   for (StreamParamsVec::const_iterator it = streams.begin();
>-      it != streams.end(); ++it) {
>-    if (!GetStreamBySsrc(remote_streams_, it->first_ssrc())) {
>+       it != streams.end(); ++it) {
>+    // We allow a StreamParams with an empty list of SSRCs, in which case the
>+    // MediaChannel will cache the parameters and use them for any unsignaled
>+    // stream received later.
>+    if ((!it->has_ssrcs() && !HasStreamWithNoSsrcs(remote_streams_)) ||
>+        !GetStreamBySsrc(remote_streams_, it->first_ssrc())) {
>       if (AddRecvStream_w(*it)) {
>-        RTC_LOG(LS_INFO) << "Add remote ssrc: " << it->ssrcs[0];
>+        RTC_LOG(LS_INFO) << "Add remote ssrc: " << it->first_ssrc();
>       } else {
>         std::ostringstream desc;
>         desc << "Failed to add remote stream ssrc: " << it->first_ssrc();
>@@ -1017,62 +660,32 @@ bool BaseChannel::UpdateRemoteStreams_w(
>         ret = false;
>       }
>     }
>+    // Update the receiving SSRCs.
>+    demuxer_criteria_.ssrcs.insert(it->ssrcs.begin(), it->ssrcs.end());
>   }
>+  // Re-register the sink to update the receiving ssrcs.
>+  RegisterRtpDemuxerSink();
>   remote_streams_ = streams;
>   return ret;
> }
> 
> RtpHeaderExtensions BaseChannel::GetFilteredRtpHeaderExtensions(
>     const RtpHeaderExtensions& extensions) {
>-  if (!rtp_dtls_transport_ ||
>-      !rtp_dtls_transport_->crypto_options()
>-          .enable_encrypted_rtp_header_extensions) {
>+  RTC_DCHECK(rtp_transport_);
>+  if (crypto_options_.enable_encrypted_rtp_header_extensions) {
>     RtpHeaderExtensions filtered;
>     auto pred = [](const webrtc::RtpExtension& extension) {
>-        return !extension.encrypt;
>+      return !extension.encrypt;
>     };
>     std::copy_if(extensions.begin(), extensions.end(),
>-        std::back_inserter(filtered), pred);
>+                 std::back_inserter(filtered), pred);
>     return filtered;
>   }
> 
>   return webrtc::RtpExtension::FilterDuplicateNonEncrypted(extensions);
> }
> 
>-void BaseChannel::MaybeCacheRtpAbsSendTimeHeaderExtension_w(
>-    const std::vector<webrtc::RtpExtension>& extensions) {
>-// Absolute Send Time extension id is used only with external auth,
>-// so do not bother searching for it and making asyncronious call to set
>-// something that is not used.
>-#if defined(ENABLE_EXTERNAL_AUTH)
>-  const webrtc::RtpExtension* send_time_extension =
>-      webrtc::RtpExtension::FindHeaderExtensionByUri(
>-          extensions, webrtc::RtpExtension::kAbsSendTimeUri);
>-  int rtp_abs_sendtime_extn_id =
>-      send_time_extension ? send_time_extension->id : -1;
>-  invoker_.AsyncInvoke<void>(
>-      RTC_FROM_HERE, network_thread_,
>-      Bind(&BaseChannel::CacheRtpAbsSendTimeHeaderExtension_n, this,
>-           rtp_abs_sendtime_extn_id));
>-#endif
>-}
>-
>-void BaseChannel::CacheRtpAbsSendTimeHeaderExtension_n(
>-    int rtp_abs_sendtime_extn_id) {
>-  if (sdes_transport_) {
>-    sdes_transport_->CacheRtpAbsSendTimeHeaderExtension(
>-        rtp_abs_sendtime_extn_id);
>-  } else if (dtls_srtp_transport_) {
>-    dtls_srtp_transport_->CacheRtpAbsSendTimeHeaderExtension(
>-        rtp_abs_sendtime_extn_id);
>-  } else {
>-    RTC_LOG(LS_WARNING)
>-        << "Trying to cache the Absolute Send Time extension id "
>-           "but the SRTP is not active.";
>-  }
>-}
>-
>-void BaseChannel::OnMessage(rtc::Message *pmsg) {
>+void BaseChannel::OnMessage(rtc::Message* pmsg) {
>   TRACE_EVENT0("webrtc", "BaseChannel::OnMessage");
>   switch (pmsg->message_id) {
>     case MSG_SEND_RTP_PACKET:
>@@ -1093,7 +706,7 @@ void BaseChannel::OnMessage(rtc::Message *pmsg) {
> }
> 
> void BaseChannel::AddHandledPayloadType(int payload_type) {
>-  rtp_transport_->AddHandledPayloadType(payload_type);
>+  demuxer_criteria_.payload_types.insert(static_cast<uint8_t>(payload_type));
> }
> 
> void BaseChannel::FlushRtcpMessages_n() {
>@@ -1120,160 +733,30 @@ void BaseChannel::SignalSentPacket_w(const rtc::SentPacket& sent_packet) {
>   SignalSentPacket(sent_packet);
> }
> 
>-void BaseChannel::UpdateEncryptedHeaderExtensionIds(
>-    cricket::ContentSource source,
>-    const std::vector<int>& extension_ids) {
>-  if (source == ContentSource::CS_LOCAL) {
>-    cached_recv_extension_ids_ = std::move(extension_ids);
>-    if (dtls_srtp_transport_) {
>-      dtls_srtp_transport_->UpdateRecvEncryptedHeaderExtensionIds(
>-          extension_ids);
>-    }
>-  } else {
>-    cached_send_extension_ids_ = std::move(extension_ids);
>-    if (dtls_srtp_transport_) {
>-      dtls_srtp_transport_->UpdateSendEncryptedHeaderExtensionIds(
>-          extension_ids);
>-    }
>-  }
>-}
>-
>-void BaseChannel::ActivateRtcpMux() {
>-  // We permanently activated RTCP muxing; signal that we no longer need
>-  // the RTCP transport.
>-  std::string debug_name =
>-      transport_name_.empty()
>-          ? rtp_transport_->rtp_packet_transport()->transport_name()
>-          : transport_name_;
>-  RTC_LOG(LS_INFO) << "Enabling rtcp-mux for " << content_name()
>-                   << "; no longer need RTCP transport for " << debug_name;
>-  if (rtp_transport_->rtcp_packet_transport()) {
>-    SetTransport_n(/*rtcp=*/true, nullptr, nullptr);
>-    if (dtls_srtp_transport_) {
>-      RTC_DCHECK(rtp_dtls_transport_);
>-      dtls_srtp_transport_->SetDtlsTransports(rtp_dtls_transport_,
>-                                              /*rtcp_dtls_transport_=*/nullptr);
>-    } else {
>-      rtp_transport_->SetRtcpPacketTransport(nullptr);
>-    }
>-    SignalRtcpMuxFullyActive(transport_name_);
>-  }
>-  UpdateWritableState_n();
>-}
>-
> VoiceChannel::VoiceChannel(rtc::Thread* worker_thread,
>                            rtc::Thread* network_thread,
>                            rtc::Thread* signaling_thread,
>-                           MediaEngineInterface* media_engine,
>+                           // TODO(nisse): Delete unused argument.
>+                           MediaEngineInterface* /* media_engine */,
>                            std::unique_ptr<VoiceMediaChannel> media_channel,
>                            const std::string& content_name,
>-                           bool rtcp_mux_required,
>-                           bool srtp_required)
>+                           bool srtp_required,
>+                           rtc::CryptoOptions crypto_options)
>     : BaseChannel(worker_thread,
>                   network_thread,
>                   signaling_thread,
>                   std::move(media_channel),
>                   content_name,
>-                  rtcp_mux_required,
>-                  srtp_required),
>-      media_engine_(media_engine) {}
>+                  srtp_required,
>+                  crypto_options) {}
> 
> VoiceChannel::~VoiceChannel() {
>   TRACE_EVENT0("webrtc", "VoiceChannel::~VoiceChannel");
>-  StopAudioMonitor();
>-  StopMediaMonitor();
>   // this can't be done in the base class, since it calls a virtual
>   DisableMedia_w();
>   Deinit();
> }
> 
>-bool VoiceChannel::SetAudioSend(uint32_t ssrc,
>-                                bool enable,
>-                                const AudioOptions* options,
>-                                AudioSource* source) {
>-  return InvokeOnWorker<bool>(
>-      RTC_FROM_HERE, Bind(&VoiceMediaChannel::SetAudioSend, media_channel(),
>-                          ssrc, enable, options, source));
>-}
>-
>-// TODO(juberti): Handle early media the right way. We should get an explicit
>-// ringing message telling us to start playing local ringback, which we cancel
>-// if any early media actually arrives. For now, we do the opposite, which is
>-// to wait 1 second for early media, and start playing local ringback if none
>-// arrives.
>-void VoiceChannel::SetEarlyMedia(bool enable) {
>-  if (enable) {
>-    // Start the early media timeout
>-    worker_thread()->PostDelayed(RTC_FROM_HERE, kEarlyMediaTimeout, this,
>-                                 MSG_EARLYMEDIATIMEOUT);
>-  } else {
>-    // Stop the timeout if currently going.
>-    worker_thread()->Clear(this, MSG_EARLYMEDIATIMEOUT);
>-  }
>-}
>-
>-bool VoiceChannel::GetStats(VoiceMediaInfo* stats) {
>-  return InvokeOnWorker<bool>(RTC_FROM_HERE, Bind(&VoiceMediaChannel::GetStats,
>-                                                  media_channel(), stats));
>-}
>-
>-void VoiceChannel::StartMediaMonitor(int cms) {
>-  media_monitor_.reset(new VoiceMediaMonitor(media_channel(), worker_thread(),
>-      rtc::Thread::Current()));
>-  media_monitor_->SignalUpdate.connect(
>-      this, &VoiceChannel::OnMediaMonitorUpdate);
>-  media_monitor_->Start(cms);
>-}
>-
>-void VoiceChannel::StopMediaMonitor() {
>-  if (media_monitor_) {
>-    media_monitor_->Stop();
>-    media_monitor_->SignalUpdate.disconnect(this);
>-    media_monitor_.reset();
>-  }
>-}
>-
>-void VoiceChannel::StartAudioMonitor(int cms) {
>-  audio_monitor_.reset(new AudioMonitor(this, rtc::Thread::Current()));
>-  audio_monitor_
>-    ->SignalUpdate.connect(this, &VoiceChannel::OnAudioMonitorUpdate);
>-  audio_monitor_->Start(cms);
>-}
>-
>-void VoiceChannel::StopAudioMonitor() {
>-  if (audio_monitor_) {
>-    audio_monitor_->Stop();
>-    audio_monitor_.reset();
>-  }
>-}
>-
>-bool VoiceChannel::IsAudioMonitorRunning() const {
>-  return (audio_monitor_.get() != NULL);
>-}
>-
>-int VoiceChannel::GetInputLevel_w() {
>-  return media_engine_->GetInputLevel();
>-}
>-
>-int VoiceChannel::GetOutputLevel_w() {
>-  return media_channel()->GetOutputLevel();
>-}
>-
>-void VoiceChannel::GetActiveStreams_w(AudioInfo::StreamList* actives) {
>-  media_channel()->GetActiveStreams(actives);
>-}
>-
>-void VoiceChannel::OnPacketReceived(bool rtcp,
>-                                    rtc::CopyOnWriteBuffer* packet,
>-                                    const rtc::PacketTime& packet_time) {
>-  BaseChannel::OnPacketReceived(rtcp, packet, packet_time);
>-  // Set a flag when we've received an RTP packet. If we're waiting for early
>-  // media, this will disable the timeout.
>-  if (!received_media_ && !rtcp) {
>-    received_media_ = true;
>-  }
>-}
>-
> void BaseChannel::UpdateMediaSendRecvState() {
>   RTC_DCHECK(network_thread_->IsCurrent());
>   invoker_.AsyncInvoke<void>(
>@@ -1312,11 +795,7 @@ bool VoiceChannel::SetLocalContent_w(const MediaContentDescription* content,
> 
>   RtpHeaderExtensions rtp_header_extensions =
>       GetFilteredRtpHeaderExtensions(audio->rtp_header_extensions());
>-
>-  if (!SetRtpTransportParameters(content, type, CS_LOCAL, rtp_header_extensions,
>-                                 error_desc)) {
>-    return false;
>-  }
>+  UpdateRtpHeaderExtensionMap(rtp_header_extensions);
> 
>   AudioRecvParameters recv_params = last_recv_params_;
>   RtpParametersFromMediaDescription(audio, rtp_header_extensions, &recv_params);
>@@ -1328,6 +807,12 @@ bool VoiceChannel::SetLocalContent_w(const MediaContentDescription* content,
>   for (const AudioCodec& codec : audio->codecs()) {
>     AddHandledPayloadType(codec.id);
>   }
>+  // Need to re-register the sink to update the handled payload.
>+  if (!RegisterRtpDemuxerSink()) {
>+    RTC_LOG(LS_ERROR) << "Failed to set up audio demuxing.";
>+    return false;
>+  }
>+
>   last_recv_params_ = recv_params;
> 
>   // TODO(pthatcher): Move local streams into AudioSendParameters, and
>@@ -1362,14 +847,10 @@ bool VoiceChannel::SetRemoteContent_w(const MediaContentDescription* content,
>   RtpHeaderExtensions rtp_header_extensions =
>       GetFilteredRtpHeaderExtensions(audio->rtp_header_extensions());
> 
>-  if (!SetRtpTransportParameters(content, type, CS_REMOTE,
>-                                 rtp_header_extensions, error_desc)) {
>-    return false;
>-  }
>-
>   AudioSendParameters send_params = last_send_params_;
>   RtpSendParametersFromMediaDescription(audio, rtp_header_extensions,
>-      &send_params);
>+                                        &send_params);
>+  send_params.mid = content_name();
> 
>   bool parameters_applied = media_channel()->SetSendParameters(send_params);
>   if (!parameters_applied) {
>@@ -1388,71 +869,30 @@ bool VoiceChannel::SetRemoteContent_w(const MediaContentDescription* content,
>     return false;
>   }
> 
>-  if (audio->rtp_header_extensions_set()) {
>-    MaybeCacheRtpAbsSendTimeHeaderExtension_w(rtp_header_extensions);
>-  }
>-
>   set_remote_content_direction(content->direction());
>   UpdateMediaSendRecvState_w();
>   return true;
> }
> 
>-void VoiceChannel::HandleEarlyMediaTimeout() {
>-  // This occurs on the main thread, not the worker thread.
>-  if (!received_media_) {
>-    RTC_LOG(LS_INFO) << "No early media received before timeout";
>-    SignalEarlyMediaTimeout(this);
>-  }
>-}
>-
>-void VoiceChannel::OnMessage(rtc::Message *pmsg) {
>-  switch (pmsg->message_id) {
>-    case MSG_EARLYMEDIATIMEOUT:
>-      HandleEarlyMediaTimeout();
>-      break;
>-    default:
>-      BaseChannel::OnMessage(pmsg);
>-      break;
>-  }
>-}
>-
>-void VoiceChannel::OnConnectionMonitorUpdate(
>-    ConnectionMonitor* monitor, const std::vector<ConnectionInfo>& infos) {
>-  SignalConnectionMonitor(this, infos);
>-}
>-
>-void VoiceChannel::OnMediaMonitorUpdate(
>-    VoiceMediaChannel* media_channel, const VoiceMediaInfo& info) {
>-  RTC_DCHECK(media_channel == this->media_channel());
>-  SignalMediaMonitor(this, info);
>-}
>-
>-void VoiceChannel::OnAudioMonitorUpdate(AudioMonitor* monitor,
>-                                        const AudioInfo& info) {
>-  SignalAudioMonitor(this, info);
>-}
>-
> VideoChannel::VideoChannel(rtc::Thread* worker_thread,
>                            rtc::Thread* network_thread,
>                            rtc::Thread* signaling_thread,
>                            std::unique_ptr<VideoMediaChannel> media_channel,
>                            const std::string& content_name,
>-                           bool rtcp_mux_required,
>-                           bool srtp_required)
>+                           bool srtp_required,
>+                           rtc::CryptoOptions crypto_options)
>     : BaseChannel(worker_thread,
>                   network_thread,
>                   signaling_thread,
>                   std::move(media_channel),
>                   content_name,
>-                  rtcp_mux_required,
>-                  srtp_required) {}
>+                  srtp_required,
>+                  crypto_options) {}
> 
> VideoChannel::~VideoChannel() {
>   TRACE_EVENT0("webrtc", "VideoChannel::~VideoChannel");
>-  StopMediaMonitor();
>   // this can't be done in the base class, since it calls a virtual
>   DisableMedia_w();
>-
>   Deinit();
> }
> 
>@@ -1473,26 +913,6 @@ void VideoChannel::FillBitrateInfo(BandwidthEstimationInfo* bwe_info) {
>                                            media_channel(), bwe_info));
> }
> 
>-bool VideoChannel::GetStats(VideoMediaInfo* stats) {
>-  return InvokeOnWorker<bool>(RTC_FROM_HERE, Bind(&VideoMediaChannel::GetStats,
>-                                                  media_channel(), stats));
>-}
>-
>-void VideoChannel::StartMediaMonitor(int cms) {
>-  media_monitor_.reset(new VideoMediaMonitor(media_channel(), worker_thread(),
>-      rtc::Thread::Current()));
>-  media_monitor_->SignalUpdate.connect(
>-      this, &VideoChannel::OnMediaMonitorUpdate);
>-  media_monitor_->Start(cms);
>-}
>-
>-void VideoChannel::StopMediaMonitor() {
>-  if (media_monitor_) {
>-    media_monitor_->Stop();
>-    media_monitor_.reset();
>-  }
>-}
>-
> bool VideoChannel::SetLocalContent_w(const MediaContentDescription* content,
>                                      SdpType type,
>                                      std::string* error_desc) {
>@@ -1510,11 +930,7 @@ bool VideoChannel::SetLocalContent_w(const MediaContentDescription* content,
> 
>   RtpHeaderExtensions rtp_header_extensions =
>       GetFilteredRtpHeaderExtensions(video->rtp_header_extensions());
>-
>-  if (!SetRtpTransportParameters(content, type, CS_LOCAL, rtp_header_extensions,
>-                                 error_desc)) {
>-    return false;
>-  }
>+  UpdateRtpHeaderExtensionMap(rtp_header_extensions);
> 
>   VideoRecvParameters recv_params = last_recv_params_;
>   RtpParametersFromMediaDescription(video, rtp_header_extensions, &recv_params);
>@@ -1526,6 +942,12 @@ bool VideoChannel::SetLocalContent_w(const MediaContentDescription* content,
>   for (const VideoCodec& codec : video->codecs()) {
>     AddHandledPayloadType(codec.id);
>   }
>+  // Need to re-register the sink to update the handled payload.
>+  if (!RegisterRtpDemuxerSink()) {
>+    RTC_LOG(LS_ERROR) << "Failed to set up video demuxing.";
>+    return false;
>+  }
>+
>   last_recv_params_ = recv_params;
> 
>   // TODO(pthatcher): Move local streams into VideoSendParameters, and
>@@ -1560,17 +982,13 @@ bool VideoChannel::SetRemoteContent_w(const MediaContentDescription* content,
>   RtpHeaderExtensions rtp_header_extensions =
>       GetFilteredRtpHeaderExtensions(video->rtp_header_extensions());
> 
>-  if (!SetRtpTransportParameters(content, type, CS_REMOTE,
>-                                 rtp_header_extensions, error_desc)) {
>-    return false;
>-  }
>-
>   VideoSendParameters send_params = last_send_params_;
>   RtpSendParametersFromMediaDescription(video, rtp_header_extensions,
>-      &send_params);
>+                                        &send_params);
>   if (video->conference_mode()) {
>     send_params.conference_mode = true;
>   }
>+  send_params.mid = content_name();
> 
>   bool parameters_applied = media_channel()->SetSendParameters(send_params);
> 
>@@ -1589,67 +1007,33 @@ bool VideoChannel::SetRemoteContent_w(const MediaContentDescription* content,
>     SafeSetError("Failed to set remote video description streams.", error_desc);
>     return false;
>   }
>-
>-  if (video->rtp_header_extensions_set()) {
>-    MaybeCacheRtpAbsSendTimeHeaderExtension_w(rtp_header_extensions);
>-  }
>-
>   set_remote_content_direction(content->direction());
>   UpdateMediaSendRecvState_w();
>   return true;
> }
> 
>-void VideoChannel::OnConnectionMonitorUpdate(
>-    ConnectionMonitor* monitor, const std::vector<ConnectionInfo> &infos) {
>-  SignalConnectionMonitor(this, infos);
>-}
>-
>-// TODO(pthatcher): Look into removing duplicate code between
>-// audio, video, and data, perhaps by using templates.
>-void VideoChannel::OnMediaMonitorUpdate(
>-    VideoMediaChannel* media_channel, const VideoMediaInfo &info) {
>-  RTC_DCHECK(media_channel == this->media_channel());
>-  SignalMediaMonitor(this, info);
>-}
>-
> RtpDataChannel::RtpDataChannel(rtc::Thread* worker_thread,
>                                rtc::Thread* network_thread,
>                                rtc::Thread* signaling_thread,
>                                std::unique_ptr<DataMediaChannel> media_channel,
>                                const std::string& content_name,
>-                               bool rtcp_mux_required,
>-                               bool srtp_required)
>+                               bool srtp_required,
>+                               rtc::CryptoOptions crypto_options)
>     : BaseChannel(worker_thread,
>                   network_thread,
>                   signaling_thread,
>                   std::move(media_channel),
>                   content_name,
>-                  rtcp_mux_required,
>-                  srtp_required) {}
>+                  srtp_required,
>+                  crypto_options) {}
> 
> RtpDataChannel::~RtpDataChannel() {
>   TRACE_EVENT0("webrtc", "RtpDataChannel::~RtpDataChannel");
>-  StopMediaMonitor();
>   // this can't be done in the base class, since it calls a virtual
>   DisableMedia_w();
>-
>   Deinit();
> }
> 
>-void RtpDataChannel::Init_w(
>-    DtlsTransportInternal* rtp_dtls_transport,
>-    DtlsTransportInternal* rtcp_dtls_transport,
>-    rtc::PacketTransportInternal* rtp_packet_transport,
>-    rtc::PacketTransportInternal* rtcp_packet_transport) {
>-  BaseChannel::Init_w(rtp_dtls_transport, rtcp_dtls_transport,
>-                      rtp_packet_transport, rtcp_packet_transport);
>-
>-  media_channel()->SignalDataReceived.connect(this,
>-                                              &RtpDataChannel::OnDataReceived);
>-  media_channel()->SignalReadyToSend.connect(
>-      this, &RtpDataChannel::OnDataChannelReadyToSend);
>-}
>-
> void RtpDataChannel::Init_w(webrtc::RtpTransportInternal* rtp_transport) {
>   BaseChannel::Init_w(rtp_transport);
>   media_channel()->SignalDataReceived.connect(this,
>@@ -1702,11 +1086,6 @@ bool RtpDataChannel::SetLocalContent_w(const MediaContentDescription* content,
>   RtpHeaderExtensions rtp_header_extensions =
>       GetFilteredRtpHeaderExtensions(data->rtp_header_extensions());
> 
>-  if (!SetRtpTransportParameters(content, type, CS_LOCAL, rtp_header_extensions,
>-                                 error_desc)) {
>-    return false;
>-  }
>-
>   DataRecvParameters recv_params = last_recv_params_;
>   RtpParametersFromMediaDescription(data, rtp_header_extensions, &recv_params);
>   if (!media_channel()->SetRecvParameters(recv_params)) {
>@@ -1717,6 +1096,12 @@ bool RtpDataChannel::SetLocalContent_w(const MediaContentDescription* content,
>   for (const DataCodec& codec : data->codecs()) {
>     AddHandledPayloadType(codec.id);
>   }
>+  // Need to re-register the sink to update the handled payload.
>+  if (!RegisterRtpDemuxerSink()) {
>+    RTC_LOG(LS_ERROR) << "Failed to set up data demuxing.";
>+    return false;
>+  }
>+
>   last_recv_params_ = recv_params;
> 
>   // TODO(pthatcher): Move local streams into DataSendParameters, and
>@@ -1761,14 +1146,9 @@ bool RtpDataChannel::SetRemoteContent_w(const MediaContentDescription* content,
>       GetFilteredRtpHeaderExtensions(data->rtp_header_extensions());
> 
>   RTC_LOG(LS_INFO) << "Setting remote data description";
>-  if (!SetRtpTransportParameters(content, type, CS_REMOTE,
>-                                 rtp_header_extensions, error_desc)) {
>-    return false;
>-  }
>-
>   DataSendParameters send_params = last_send_params_;
>   RtpSendParametersFromMediaDescription<DataCodec>(data, rtp_header_extensions,
>-      &send_params);
>+                                                   &send_params);
>   if (!media_channel()->SetSendParameters(send_params)) {
>     SafeSetError("Failed to set remote data description send parameters.",
>                  error_desc);
>@@ -1781,8 +1161,7 @@ bool RtpDataChannel::SetRemoteContent_w(const MediaContentDescription* content,
>   // description too (without a local description, we won't be able to
>   // recv them anyway).
>   if (!UpdateRemoteStreams_w(data->streams(), type, error_desc)) {
>-    SafeSetError("Failed to set remote data description streams.",
>-                 error_desc);
>+    SafeSetError("Failed to set remote data description streams.", error_desc);
>     return false;
>   }
> 
>@@ -1835,39 +1214,10 @@ void RtpDataChannel::OnMessage(rtc::Message* pmsg) {
>   }
> }
> 
>-void RtpDataChannel::OnConnectionMonitorUpdate(
>-    ConnectionMonitor* monitor,
>-    const std::vector<ConnectionInfo>& infos) {
>-  SignalConnectionMonitor(this, infos);
>-}
>-
>-void RtpDataChannel::StartMediaMonitor(int cms) {
>-  media_monitor_.reset(new DataMediaMonitor(media_channel(), worker_thread(),
>-      rtc::Thread::Current()));
>-  media_monitor_->SignalUpdate.connect(this,
>-                                       &RtpDataChannel::OnMediaMonitorUpdate);
>-  media_monitor_->Start(cms);
>-}
>-
>-void RtpDataChannel::StopMediaMonitor() {
>-  if (media_monitor_) {
>-    media_monitor_->Stop();
>-    media_monitor_->SignalUpdate.disconnect(this);
>-    media_monitor_.reset();
>-  }
>-}
>-
>-void RtpDataChannel::OnMediaMonitorUpdate(DataMediaChannel* media_channel,
>-                                          const DataMediaInfo& info) {
>-  RTC_DCHECK(media_channel == this->media_channel());
>-  SignalMediaMonitor(this, info);
>-}
>-
> void RtpDataChannel::OnDataReceived(const ReceiveDataParams& params,
>                                     const char* data,
>                                     size_t len) {
>-  DataReceivedMessageData* msg = new DataReceivedMessageData(
>-      params, data, len);
>+  DataReceivedMessageData* msg = new DataReceivedMessageData(params, data, len);
>   signaling_thread()->Post(RTC_FROM_HERE, this, MSG_DATARECEIVED, msg);
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/channel.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/channel.h
>index ca36dcdfe1e..ba584690ba6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/channel.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/channel.h
>@@ -21,28 +21,24 @@
> #include "api/call/audio_sink.h"
> #include "api/jsep.h"
> #include "api/rtpreceiverinterface.h"
>-#include "api/videosinkinterface.h"
>-#include "api/videosourceinterface.h"
>+#include "api/video/video_sink_interface.h"
>+#include "api/video/video_source_interface.h"
>+#include "call/rtp_packet_sink_interface.h"
> #include "media/base/mediachannel.h"
> #include "media/base/mediaengine.h"
> #include "media/base/streamparams.h"
> #include "p2p/base/dtlstransportinternal.h"
> #include "p2p/base/packettransportinternal.h"
>-#include "p2p/client/socketmonitor.h"
>-#include "pc/audiomonitor.h"
> #include "pc/dtlssrtptransport.h"
>-#include "pc/mediamonitor.h"
> #include "pc/mediasession.h"
>-#include "pc/rtcpmuxfilter.h"
> #include "pc/rtptransport.h"
> #include "pc/srtpfilter.h"
> #include "pc/srtptransport.h"
>-#include "pc/transportcontroller.h"
> #include "rtc_base/asyncinvoker.h"
> #include "rtc_base/asyncudpsocket.h"
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/network.h"
>-#include "rtc_base/sigslot.h"
>+#include "rtc_base/third_party/sigslot/sigslot.h"
> 
> namespace webrtc {
> class AudioSinkInterface;
>@@ -71,27 +67,23 @@ class MediaContentDescription;
> // vtable, and the media channel's thread using BaseChannel as the
> // NetworkInterface.
> 
>-class BaseChannel
>-    : public rtc::MessageHandler, public sigslot::has_slots<>,
>-      public MediaChannel::NetworkInterface,
>-      public ConnectionStatsGetter {
>+class BaseChannel : public rtc::MessageHandler,
>+                    public sigslot::has_slots<>,
>+                    public MediaChannel::NetworkInterface,
>+                    public webrtc::RtpPacketSinkInterface {
>  public:
>   // If |srtp_required| is true, the channel will not send or receive any
>   // RTP/RTCP packets without using SRTP (either using SDES or DTLS-SRTP).
>+  // TODO(zhihuang:) Create a BaseChannel::Config struct for the parameter lists
>+  // which will make it easier to change the constructor.
>   BaseChannel(rtc::Thread* worker_thread,
>               rtc::Thread* network_thread,
>               rtc::Thread* signaling_thread,
>               std::unique_ptr<MediaChannel> media_channel,
>               const std::string& content_name,
>-              bool rtcp_mux_required,
>-              bool srtp_required);
>+              bool srtp_required,
>+              rtc::CryptoOptions crypto_options);
>   virtual ~BaseChannel();
>-  // TODO(zhihuang): Remove this once the RtpTransport can be shared between
>-  // BaseChannels.
>-  void Init_w(DtlsTransportInternal* rtp_dtls_transport,
>-              DtlsTransportInternal* rtcp_dtls_transport,
>-              rtc::PacketTransportInternal* rtp_packet_transport,
>-              rtc::PacketTransportInternal* rtcp_packet_transport);
>   void Init_w(webrtc::RtpTransportInternal* rtp_transport);
> 
>   // Deinit may be called multiple times and is simply ignored if it's already
>@@ -105,16 +97,10 @@ class BaseChannel
>   const std::string& transport_name() const { return transport_name_; }
>   bool enabled() const { return enabled_; }
> 
>-  // This function returns true if we are using SDES.
>-  bool sdes_active() const {
>-    return sdes_transport_ && sdes_negotiator_.IsActive();
>-  }
>-  // The following function returns true if we are using DTLS-based keying.
>-  bool dtls_active() const {
>-    return dtls_srtp_transport_ && dtls_srtp_transport_->IsActive();
>-  }
>   // This function returns true if using SRTP (DTLS-based keying or SDES).
>-  bool srtp_active() const { return sdes_active() || dtls_active(); }
>+  bool srtp_active() const {
>+    return rtp_transport_ && rtp_transport_->IsSrtpActive();
>+  }
> 
>   bool writable() const { return writable_; }
> 
>@@ -122,22 +108,8 @@ class BaseChannel
>   // encryption, an SrtpTransport for SDES or a DtlsSrtpTransport for DTLS-SRTP.
>   // This can be called from any thread and it hops to the network thread
>   // internally. It would replace the |SetTransports| and its variants.
>-  void SetRtpTransport(webrtc::RtpTransportInternal* rtp_transport);
>-
>-  // Set the transport(s), and update writability and "ready-to-send" state.
>-  // |rtp_transport| must be non-null.
>-  // |rtcp_transport| must be supplied if NeedsRtcpTransport() is true (meaning
>-  // RTCP muxing is not fully active yet).
>-  // |rtp_transport| and |rtcp_transport| must share the same transport name as
>-  // well.
>-  // Can not start with "rtc::PacketTransportInternal" and switch to
>-  // "DtlsTransportInternal", or vice-versa.
>-  // TODO(zhihuang): Remove these two once the RtpTransport can be shared
>-  // between BaseChannels.
>-  void SetTransports(DtlsTransportInternal* rtp_dtls_transport,
>-                     DtlsTransportInternal* rtcp_dtls_transport);
>-  void SetTransports(rtc::PacketTransportInternal* rtp_packet_transport,
>-                     rtc::PacketTransportInternal* rtcp_packet_transport);
>+  bool SetRtpTransport(webrtc::RtpTransportInternal* rtp_transport);
>+
>   // Channel control
>   bool SetLocalContent(const MediaContentDescription* content,
>                        webrtc::SdpType type,
>@@ -148,18 +120,12 @@ class BaseChannel
> 
>   bool Enable(bool enable);
> 
>-  // Multiplexing
>+  // TODO(zhihuang): These methods are used for testing and can be removed.
>   bool AddRecvStream(const StreamParams& sp);
>   bool RemoveRecvStream(uint32_t ssrc);
>   bool AddSendStream(const StreamParams& sp);
>   bool RemoveSendStream(uint32_t ssrc);
> 
>-  // Monitoring
>-  void StartConnectionMonitor(int cms);
>-  void StopConnectionMonitor();
>-  // For ConnectionStatsGetter, used by ConnectionMonitor
>-  bool GetConnectionStats(ConnectionInfos* infos) override;
>-
>   const std::vector<StreamParams>& local_streams() const {
>     return local_streams_;
>   }
>@@ -182,30 +148,31 @@ class BaseChannel
>   // Fired on the network thread.
>   sigslot::signal1<const std::string&> SignalRtcpMuxFullyActive;
> 
>-  // Only public for unit tests.  Otherwise, consider private.
>-  DtlsTransportInternal* rtp_dtls_transport() const {
>-    return rtp_dtls_transport_;
>-  }
>-  DtlsTransportInternal* rtcp_dtls_transport() const {
>-    return rtcp_dtls_transport_;
>+  rtc::PacketTransportInternal* rtp_packet_transport() {
>+    if (rtp_transport_) {
>+      return rtp_transport_->rtp_packet_transport();
>+    }
>+    return nullptr;
>   }
> 
>-  bool NeedsRtcpTransport();
>+  rtc::PacketTransportInternal* rtcp_packet_transport() {
>+    if (rtp_transport_) {
>+      return rtp_transport_->rtcp_packet_transport();
>+    }
>+    return nullptr;
>+  }
> 
>   // From RtpTransport - public for testing only
>   void OnTransportReadyToSend(bool ready);
> 
>   // Only public for unit tests.  Otherwise, consider protected.
>-  int SetOption(SocketType type, rtc::Socket::Option o, int val)
>-      override;
>+  int SetOption(SocketType type, rtc::Socket::Option o, int val) override;
>   int SetOption_n(SocketType type, rtc::Socket::Option o, int val);
> 
>   virtual cricket::MediaType media_type() = 0;
> 
>-  // Public for testing.
>-  // TODO(zstein): Remove this once channels register themselves with
>-  // an RtpTransport in a more explicit way.
>-  bool HandlesPayloadType(int payload_type) const;
>+  // RtpPacketSinkInterface overrides.
>+  void OnRtpPacket(const webrtc::RtpPacketReceived& packet) override;
> 
>   // Used by the RTCStatsCollector tests to set the transport name without
>   // creating RtpTransports.
>@@ -216,19 +183,6 @@ class BaseChannel
>  protected:
>   virtual MediaChannel* media_channel() const { return media_channel_.get(); }
> 
>-  void SetTransports_n(DtlsTransportInternal* rtp_dtls_transport,
>-                       DtlsTransportInternal* rtcp_dtls_transport,
>-                       rtc::PacketTransportInternal* rtp_packet_transport,
>-                       rtc::PacketTransportInternal* rtcp_packet_transport);
>-
>-  // This does not update writability or "ready-to-send" state; it just
>-  // disconnects from the old channel and connects to the new one.
>-  // TODO(zhihuang): Remove this once the RtpTransport can be shared between
>-  // BaseChannels.
>-  void SetTransport_n(bool rtcp,
>-                      DtlsTransportInternal* new_dtls_transport,
>-                      rtc::PacketTransportInternal* new_packet_transport);
>-
>   bool was_ever_writable() const { return was_ever_writable_; }
>   void set_local_content_direction(webrtc::RtpTransceiverDirection direction) {
>     local_content_direction_ = direction;
>@@ -261,7 +215,7 @@ class BaseChannel
>   // From RtpTransportInternal
>   void OnWritableState(bool writable);
> 
>-  void OnNetworkRouteChanged(rtc::Optional<rtc::NetworkRoute> network_route);
>+  void OnNetworkRouteChanged(absl::optional<rtc::NetworkRoute> network_route);
> 
>   bool PacketIsRtcp(const rtc::PacketTransportInternal* transport,
>                     const char* data,
>@@ -270,13 +224,12 @@ class BaseChannel
>                   rtc::CopyOnWriteBuffer* packet,
>                   const rtc::PacketOptions& options);
> 
>-  bool WantsPacket(bool rtcp, const rtc::CopyOnWriteBuffer* packet);
>-  void HandlePacket(bool rtcp, rtc::CopyOnWriteBuffer* packet,
>-                    const rtc::PacketTime& packet_time);
>-  // TODO(zstein): packet can be const once the RtpTransport handles protection.
>-  virtual void OnPacketReceived(bool rtcp,
>-                                rtc::CopyOnWriteBuffer* packet,
>-                                const rtc::PacketTime& packet_time);
>+  void OnRtcpPacketReceived(rtc::CopyOnWriteBuffer* packet,
>+                            const rtc::PacketTime& packet_time);
>+
>+  void OnPacketReceived(bool rtcp,
>+                        const rtc::CopyOnWriteBuffer& packet,
>+                        const rtc::PacketTime& packet_time);
>   void ProcessPacket(bool rtcp,
>                      const rtc::CopyOnWriteBuffer& packet,
>                      const rtc::PacketTime& packet_time);
>@@ -295,11 +248,6 @@ class BaseChannel
>   bool RemoveRecvStream_w(uint32_t ssrc);
>   bool AddSendStream_w(const StreamParams& sp);
>   bool RemoveSendStream_w(uint32_t ssrc);
>-  bool ShouldSetupDtlsSrtp_n() const;
>-  // Do the DTLS key expansion and impose it on the SRTP/SRTCP filters.
>-  // |rtcp_channel| indicates whether to set up the RTP or RTCP filter.
>-  bool SetupDtlsSrtp_n(bool rtcp);
>-  void MaybeSetupDtlsSrtp_n();
> 
>   // Should be called whenever the conditions for
>   // IsReadyToReceiveMedia/IsReadyToSendMedia are satisfied (or unsatisfied).
>@@ -319,49 +267,15 @@ class BaseChannel
>   virtual bool SetRemoteContent_w(const MediaContentDescription* content,
>                                   webrtc::SdpType type,
>                                   std::string* error_desc) = 0;
>-  bool SetRtpTransportParameters(const MediaContentDescription* content,
>-                                 webrtc::SdpType type,
>-                                 ContentSource src,
>-                                 const RtpHeaderExtensions& extensions,
>-                                 std::string* error_desc);
>-  bool SetRtpTransportParameters_n(
>-      const MediaContentDescription* content,
>-      webrtc::SdpType type,
>-      ContentSource src,
>-      const std::vector<int>& encrypted_extension_ids,
>-      std::string* error_desc);
>-
>   // Return a list of RTP header extensions with the non-encrypted extensions
>   // removed depending on the current crypto_options_ and only if both the
>   // non-encrypted and encrypted extension is present for the same URI.
>   RtpHeaderExtensions GetFilteredRtpHeaderExtensions(
>       const RtpHeaderExtensions& extensions);
> 
>-  // Helper method to get RTP Absoulute SendTime extension header id if
>-  // present in remote supported extensions list.
>-  void MaybeCacheRtpAbsSendTimeHeaderExtension_w(
>-      const std::vector<webrtc::RtpExtension>& extensions);
>-
>-  bool CheckSrtpConfig_n(const std::vector<CryptoParams>& cryptos,
>-                         bool* dtls,
>-                         std::string* error_desc);
>-  bool SetSrtp_n(const std::vector<CryptoParams>& params,
>-                 webrtc::SdpType type,
>-                 ContentSource src,
>-                 const std::vector<int>& encrypted_extension_ids,
>-                 std::string* error_desc);
>-  bool SetRtcpMux_n(bool enable,
>-                    webrtc::SdpType type,
>-                    ContentSource src,
>-                    std::string* error_desc);
>-
>   // From MessageHandler
>   void OnMessage(rtc::Message* pmsg) override;
> 
>-  // Handled in derived classes
>-  virtual void OnConnectionMonitorUpdate(ConnectionMonitor* monitor,
>-      const std::vector<ConnectionInfo>& infos) = 0;
>-
>   // Helper function template for invoking methods on the worker thread.
>   template <class T, class FunctorT>
>   T InvokeOnWorker(const rtc::Location& posted_from, const FunctorT& functor) {
>@@ -370,66 +284,36 @@ class BaseChannel
> 
>   void AddHandledPayloadType(int payload_type);
> 
>+  void UpdateRtpHeaderExtensionMap(
>+      const RtpHeaderExtensions& header_extensions);
>+
>+  bool RegisterRtpDemuxerSink();
>+
>  private:
>-  void ConnectToRtpTransport();
>+  bool ConnectToRtpTransport();
>   void DisconnectFromRtpTransport();
>   void SignalSentPacket_n(const rtc::SentPacket& sent_packet);
>   void SignalSentPacket_w(const rtc::SentPacket& sent_packet);
>   bool IsReadyToSendMedia_n() const;
>-  void CacheRtpAbsSendTimeHeaderExtension_n(int rtp_abs_sendtime_extn_id);
>-  // Wraps the existing RtpTransport in an SrtpTransport.
>-  void EnableSdes_n();
>-
>-  // Wraps the existing RtpTransport in a new SrtpTransport and wraps that in a
>-  // new DtlsSrtpTransport.
>-  void EnableDtlsSrtp_n();
>-
>-  // Update the encrypted header extension IDs when setting the local/remote
>-  // description and use them later together with other crypto parameters from
>-  // DtlsTransport. If DTLS-SRTP is enabled, it also update the encrypted header
>-  // extension IDs for DtlsSrtpTransport.
>-  void UpdateEncryptedHeaderExtensionIds(cricket::ContentSource source,
>-                                         const std::vector<int>& extension_ids);
>-
>-  // Permanently enable RTCP muxing. Set null RTCP PacketTransport for
>-  // BaseChannel and RtpTransport. If using DTLS-SRTP, set null DtlsTransport
>-  // for DtlsSrtpTransport.
>-  void ActivateRtcpMux();
>-
>   rtc::Thread* const worker_thread_;
>   rtc::Thread* const network_thread_;
>   rtc::Thread* const signaling_thread_;
>   rtc::AsyncInvoker invoker_;
> 
>   const std::string content_name_;
>-  std::unique_ptr<ConnectionMonitor> connection_monitor_;
> 
>   // Won't be set when using raw packet transports. SDP-specific thing.
>   std::string transport_name_;
> 
>-  const bool rtcp_mux_required_;
>-
>-  // Separate DTLS/non-DTLS pointers to support using BaseChannel without DTLS.
>-  // Temporary measure until more refactoring is done.
>-  // If non-null, "X_dtls_transport_" will always equal "X_packet_transport_".
>-  DtlsTransportInternal* rtp_dtls_transport_ = nullptr;
>-  DtlsTransportInternal* rtcp_dtls_transport_ = nullptr;
>-
>   webrtc::RtpTransportInternal* rtp_transport_ = nullptr;
>-  // Only one of these transports is non-null at a time. One for DTLS-SRTP, one
>-  // for SDES and one for unencrypted RTP.
>-  std::unique_ptr<webrtc::SrtpTransport> sdes_transport_;
>-  std::unique_ptr<webrtc::DtlsSrtpTransport> dtls_srtp_transport_;
>-  std::unique_ptr<webrtc::RtpTransport> unencrypted_rtp_transport_;
> 
>   std::vector<std::pair<rtc::Socket::Option, int> > socket_options_;
>   std::vector<std::pair<rtc::Socket::Option, int> > rtcp_socket_options_;
>-  SrtpFilter sdes_negotiator_;
>-  RtcpMuxFilter rtcp_mux_filter_;
>   bool writable_ = false;
>   bool was_ever_writable_ = false;
>   bool has_received_packet_ = false;
>   const bool srtp_required_ = true;
>+  rtc::CryptoOptions crypto_options_;
> 
>   // MediaChannel related members that should be accessed from the worker
>   // thread.
>@@ -445,9 +329,7 @@ class BaseChannel
>   webrtc::RtpTransceiverDirection remote_content_direction_ =
>       webrtc::RtpTransceiverDirection::kInactive;
> 
>-  // The cached encrypted header extension IDs.
>-  rtc::Optional<std::vector<int>> cached_send_extension_ids_;
>-  rtc::Optional<std::vector<int>> cached_recv_extension_ids_;
>+  webrtc::RtpDemuxerCriteria demuxer_criteria_;
> };
> 
> // VoiceChannel is a specialization that adds support for early media, DTMF,
>@@ -460,56 +342,22 @@ class VoiceChannel : public BaseChannel {
>                MediaEngineInterface* media_engine,
>                std::unique_ptr<VoiceMediaChannel> channel,
>                const std::string& content_name,
>-               bool rtcp_mux_required,
>-               bool srtp_required);
>+               bool srtp_required,
>+               rtc::CryptoOptions crypto_options);
>   ~VoiceChannel();
> 
>-  // Configure sending media on the stream with SSRC |ssrc|
>-  // If there is only one sending stream SSRC 0 can be used.
>-  bool SetAudioSend(uint32_t ssrc,
>-                    bool enable,
>-                    const AudioOptions* options,
>-                    AudioSource* source);
>-
>   // downcasts a MediaChannel
>   VoiceMediaChannel* media_channel() const override {
>     return static_cast<VoiceMediaChannel*>(BaseChannel::media_channel());
>   }
> 
>-  void SetEarlyMedia(bool enable);
>-  // This signal is emitted when we have gone a period of time without
>-  // receiving early media. When received, a UI should start playing its
>-  // own ringing sound
>-  sigslot::signal1<VoiceChannel*> SignalEarlyMediaTimeout;
>-
>-  // Get statistics about the current media session.
>-  bool GetStats(VoiceMediaInfo* stats);
>-
>-  // Monitoring functions
>-  sigslot::signal2<VoiceChannel*, const std::vector<ConnectionInfo>&>
>-      SignalConnectionMonitor;
>-
>-  void StartMediaMonitor(int cms);
>-  void StopMediaMonitor();
>-  sigslot::signal2<VoiceChannel*, const VoiceMediaInfo&> SignalMediaMonitor;
>-
>-  void StartAudioMonitor(int cms);
>-  void StopAudioMonitor();
>-  bool IsAudioMonitorRunning() const;
>-  sigslot::signal2<VoiceChannel*, const AudioInfo&> SignalAudioMonitor;
>-
>-  int GetInputLevel_w();
>-  int GetOutputLevel_w();
>-  void GetActiveStreams_w(AudioInfo::StreamList* actives);
>   webrtc::RtpParameters GetRtpSendParameters_w(uint32_t ssrc) const;
>-  bool SetRtpSendParameters_w(uint32_t ssrc, webrtc::RtpParameters parameters);
>+  webrtc::RTCError SetRtpSendParameters_w(uint32_t ssrc,
>+                                          webrtc::RtpParameters parameters);
>   cricket::MediaType media_type() override { return cricket::MEDIA_TYPE_AUDIO; }
> 
>  private:
>   // overrides from BaseChannel
>-  void OnPacketReceived(bool rtcp,
>-                        rtc::CopyOnWriteBuffer* packet,
>-                        const rtc::PacketTime& packet_time) override;
>   void UpdateMediaSendRecvState_w() override;
>   bool SetLocalContent_w(const MediaContentDescription* content,
>                          webrtc::SdpType type,
>@@ -517,21 +365,6 @@ class VoiceChannel : public BaseChannel {
>   bool SetRemoteContent_w(const MediaContentDescription* content,
>                           webrtc::SdpType type,
>                           std::string* error_desc) override;
>-  void HandleEarlyMediaTimeout();
>-
>-  void OnMessage(rtc::Message* pmsg) override;
>-  void OnConnectionMonitorUpdate(
>-      ConnectionMonitor* monitor,
>-      const std::vector<ConnectionInfo>& infos) override;
>-  void OnMediaMonitorUpdate(VoiceMediaChannel* media_channel,
>-                            const VoiceMediaInfo& info);
>-  void OnAudioMonitorUpdate(AudioMonitor* monitor, const AudioInfo& info);
>-
>-  static const int kEarlyMediaTimeout = 1000;
>-  MediaEngineInterface* media_engine_;
>-  bool received_media_ = false;
>-  std::unique_ptr<VoiceMediaMonitor> media_monitor_;
>-  std::unique_ptr<AudioMonitor> audio_monitor_;
> 
>   // Last AudioSendParameters sent down to the media_channel() via
>   // SetSendParameters.
>@@ -549,8 +382,8 @@ class VideoChannel : public BaseChannel {
>                rtc::Thread* signaling_thread,
>                std::unique_ptr<VideoMediaChannel> media_channel,
>                const std::string& content_name,
>-               bool rtcp_mux_required,
>-               bool srtp_required);
>+               bool srtp_required,
>+               rtc::CryptoOptions crypto_options);
>   ~VideoChannel();
> 
>   // downcasts a MediaChannel
>@@ -559,15 +392,6 @@ class VideoChannel : public BaseChannel {
>   }
> 
>   void FillBitrateInfo(BandwidthEstimationInfo* bwe_info);
>-  // Get statistics about the current media session.
>-  bool GetStats(VideoMediaInfo* stats);
>-
>-  sigslot::signal2<VideoChannel*, const std::vector<ConnectionInfo>&>
>-      SignalConnectionMonitor;
>-
>-  void StartMediaMonitor(int cms);
>-  void StopMediaMonitor();
>-  sigslot::signal2<VideoChannel*, const VideoMediaInfo&> SignalMediaMonitor;
> 
>   cricket::MediaType media_type() override { return cricket::MEDIA_TYPE_VIDEO; }
> 
>@@ -580,15 +404,6 @@ class VideoChannel : public BaseChannel {
>   bool SetRemoteContent_w(const MediaContentDescription* content,
>                           webrtc::SdpType type,
>                           std::string* error_desc) override;
>-  bool GetStats_w(VideoMediaInfo* stats);
>-
>-  void OnConnectionMonitorUpdate(
>-      ConnectionMonitor* monitor,
>-      const std::vector<ConnectionInfo>& infos) override;
>-  void OnMediaMonitorUpdate(VideoMediaChannel* media_channel,
>-                            const VideoMediaInfo& info);
>-
>-  std::unique_ptr<VideoMediaMonitor> media_monitor_;
> 
>   // Last VideoSendParameters sent down to the media_channel() via
>   // SetSendParameters.
>@@ -606,8 +421,8 @@ class RtpDataChannel : public BaseChannel {
>                  rtc::Thread* signaling_thread,
>                  std::unique_ptr<DataMediaChannel> channel,
>                  const std::string& content_name,
>-                 bool rtcp_mux_required,
>-                 bool srtp_required);
>+                 bool srtp_required,
>+                 rtc::CryptoOptions crypto_options);
>   ~RtpDataChannel();
>   // TODO(zhihuang): Remove this once the RtpTransport can be shared between
>   // BaseChannels.
>@@ -621,17 +436,8 @@ class RtpDataChannel : public BaseChannel {
>                         const rtc::CopyOnWriteBuffer& payload,
>                         SendDataResult* result);
> 
>-  void StartMediaMonitor(int cms);
>-  void StopMediaMonitor();
>-
>   // Should be called on the signaling thread only.
>-  bool ready_to_send_data() const {
>-    return ready_to_send_data_;
>-  }
>-
>-  sigslot::signal2<RtpDataChannel*, const DataMediaInfo&> SignalMediaMonitor;
>-  sigslot::signal2<RtpDataChannel*, const std::vector<ConnectionInfo>&>
>-      SignalConnectionMonitor;
>+  bool ready_to_send_data() const { return ready_to_send_data_; }
> 
>   sigslot::signal2<const ReceiveDataParams&, const rtc::CopyOnWriteBuffer&>
>       SignalDataReceived;
>@@ -652,11 +458,7 @@ class RtpDataChannel : public BaseChannel {
>     SendDataMessageData(const SendDataParams& params,
>                         const rtc::CopyOnWriteBuffer* payload,
>                         SendDataResult* result)
>-        : params(params),
>-          payload(payload),
>-          result(result),
>-          succeeded(false) {
>-    }
>+        : params(params), payload(payload), result(result), succeeded(false) {}
> 
>     const SendDataParams& params;
>     const rtc::CopyOnWriteBuffer* payload;
>@@ -668,11 +470,10 @@ class RtpDataChannel : public BaseChannel {
>     // We copy the data because the data will become invalid after we
>     // handle DataMediaChannel::SignalDataReceived but before we fire
>     // SignalDataReceived.
>-    DataReceivedMessageData(
>-        const ReceiveDataParams& params, const char* data, size_t len)
>-        : params(params),
>-          payload(data, len) {
>-    }
>+    DataReceivedMessageData(const ReceiveDataParams& params,
>+                            const char* data,
>+                            size_t len)
>+        : params(params), payload(data, len) {}
>     const ReceiveDataParams params;
>     const rtc::CopyOnWriteBuffer payload;
>   };
>@@ -692,16 +493,11 @@ class RtpDataChannel : public BaseChannel {
>   void UpdateMediaSendRecvState_w() override;
> 
>   void OnMessage(rtc::Message* pmsg) override;
>-  void OnConnectionMonitorUpdate(
>-      ConnectionMonitor* monitor,
>-      const std::vector<ConnectionInfo>& infos) override;
>-  void OnMediaMonitorUpdate(DataMediaChannel* media_channel,
>-                            const DataMediaInfo& info);
>-  void OnDataReceived(
>-      const ReceiveDataParams& params, const char* data, size_t len);
>+  void OnDataReceived(const ReceiveDataParams& params,
>+                      const char* data,
>+                      size_t len);
>   void OnDataChannelReadyToSend(bool writable);
> 
>-  std::unique_ptr<DataMediaMonitor> media_monitor_;
>   bool ready_to_send_data_ = false;
> 
>   // Last DataSendParameters sent down to the media_channel() via
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/channel_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/channel_unittest.cc
>index 7e3b45993ac..58cb17b8bbf 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/channel_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/channel_unittest.cc
>@@ -42,10 +42,12 @@ const cricket::VideoCodec kH264SvcCodec(99, "H264-SVC");
> const cricket::DataCodec kGoogleDataCodec(101, "google-data");
> const uint32_t kSsrc1 = 0x1111;
> const uint32_t kSsrc2 = 0x2222;
>+const uint32_t kSsrc3 = 0x3333;
>+const uint32_t kSsrc4 = 0x4444;
> const int kAudioPts[] = {0, 8};
> const int kVideoPts[] = {97, 99};
> enum class NetworkIsWorker { Yes, No };
>-const int kDefaultTimeout = 10000;  // 10 seconds.
>+
> }  // namespace
> 
> template <class ChannelT,
>@@ -86,20 +88,16 @@ class DataTraits : public Traits<cricket::RtpDataChannel,
>                                  cricket::DataOptions> {};
> 
> // Base class for Voice/Video/RtpDataChannel tests
>-template<class T>
>+template <class T>
> class ChannelTest : public testing::Test, public sigslot::has_slots<> {
>  public:
>   enum Flags {
>     RTCP_MUX = 0x1,
>-    RTCP_MUX_REQUIRED = 0x2,
>-    SECURE = 0x4,
>     SSRC_MUX = 0x8,
>     DTLS = 0x10,
>     // Use BaseChannel with PacketTransportInternal rather than
>     // DtlsTransportInternal.
>     RAW_PACKET_TRANSPORT = 0x20,
>-    GCM_CIPHER = 0x40,
>-    ENCRYPTED_HEADERS = 0x80,
>   };
> 
>   ChannelTest(bool verify_playout,
>@@ -119,9 +117,9 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
>   }
> 
>   void CreateChannels(int flags1, int flags2) {
>-    CreateChannels(rtc::MakeUnique<typename T::MediaChannel>(
>+    CreateChannels(absl::make_unique<typename T::MediaChannel>(
>                        nullptr, typename T::Options()),
>-                   rtc::MakeUnique<typename T::MediaChannel>(
>+                   absl::make_unique<typename T::MediaChannel>(
>                        nullptr, typename T::Options()),
>                    flags1, flags2);
>   }
>@@ -135,9 +133,7 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
>     if (network_thread_keeper_) {
>       network_thread_keeper_->Start();
>     }
>-    // Make sure RTCP_MUX_REQUIRED isn't set without RTCP_MUX.
>-    RTC_DCHECK_NE(RTCP_MUX_REQUIRED, flags1 & (RTCP_MUX | RTCP_MUX_REQUIRED));
>-    RTC_DCHECK_NE(RTCP_MUX_REQUIRED, flags2 & (RTCP_MUX | RTCP_MUX_REQUIRED));
>+
>     // Make sure if using raw packet transports, they're used for both
>     // channels.
>     RTC_DCHECK_EQ(flags1 & RAW_PACKET_TRANSPORT, flags2 & RAW_PACKET_TRANSPORT);
>@@ -153,7 +149,7 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
>       fake_rtp_packet_transport1_.reset(
>           new rtc::FakePacketTransport("channel1_rtp"));
>       rtp1 = fake_rtp_packet_transport1_.get();
>-      if (!(flags1 & RTCP_MUX_REQUIRED)) {
>+      if (!(flags1 & RTCP_MUX)) {
>         fake_rtcp_packet_transport1_.reset(
>             new rtc::FakePacketTransport("channel1_rtcp"));
>         rtcp1 = fake_rtcp_packet_transport1_.get();
>@@ -163,7 +159,7 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
>       fake_rtp_dtls_transport1_.reset(new cricket::FakeDtlsTransport(
>           "channel1", cricket::ICE_CANDIDATE_COMPONENT_RTP));
>       rtp1 = fake_rtp_dtls_transport1_.get();
>-      if (!(flags1 & RTCP_MUX_REQUIRED)) {
>+      if (!(flags1 & RTCP_MUX)) {
>         fake_rtcp_dtls_transport1_.reset(new cricket::FakeDtlsTransport(
>             "channel1", cricket::ICE_CANDIDATE_COMPONENT_RTCP));
>         rtcp1 = fake_rtcp_dtls_transport1_.get();
>@@ -177,29 +173,13 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
>           fake_rtcp_dtls_transport1_->SetLocalCertificate(cert1);
>         }
>       }
>-      if (flags1 & ENCRYPTED_HEADERS) {
>-        rtc::CryptoOptions crypto_options;
>-        crypto_options.enable_encrypted_rtp_header_extensions = true;
>-        fake_rtp_dtls_transport1_->SetCryptoOptions(crypto_options);
>-        if (fake_rtcp_dtls_transport1_) {
>-          fake_rtcp_dtls_transport1_->SetCryptoOptions(crypto_options);
>-        }
>-      }
>-      if (flags1 & GCM_CIPHER) {
>-        fake_rtp_dtls_transport1_->SetSrtpCryptoSuite(
>-            rtc::SRTP_AEAD_AES_256_GCM);
>-        if (fake_rtcp_dtls_transport1_) {
>-          fake_rtcp_dtls_transport1_->SetSrtpCryptoSuite(
>-              rtc::SRTP_AEAD_AES_256_GCM);
>-        }
>-      }
>     }
>     // Based on flags, create fake DTLS or raw packet transports.
>     if (flags2 & RAW_PACKET_TRANSPORT) {
>       fake_rtp_packet_transport2_.reset(
>           new rtc::FakePacketTransport("channel2_rtp"));
>       rtp2 = fake_rtp_packet_transport2_.get();
>-      if (!(flags2 & RTCP_MUX_REQUIRED)) {
>+      if (!(flags2 & RTCP_MUX)) {
>         fake_rtcp_packet_transport2_.reset(
>             new rtc::FakePacketTransport("channel2_rtcp"));
>         rtcp2 = fake_rtcp_packet_transport2_.get();
>@@ -209,7 +189,7 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
>       fake_rtp_dtls_transport2_.reset(new cricket::FakeDtlsTransport(
>           "channel2", cricket::ICE_CANDIDATE_COMPONENT_RTP));
>       rtp2 = fake_rtp_dtls_transport2_.get();
>-      if (!(flags2 & RTCP_MUX_REQUIRED)) {
>+      if (!(flags2 & RTCP_MUX)) {
>         fake_rtcp_dtls_transport2_.reset(new cricket::FakeDtlsTransport(
>             "channel2", cricket::ICE_CANDIDATE_COMPONENT_RTCP));
>         rtcp2 = fake_rtcp_dtls_transport2_.get();
>@@ -223,47 +203,26 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
>           fake_rtcp_dtls_transport2_->SetLocalCertificate(cert2);
>         }
>       }
>-      if (flags2 & ENCRYPTED_HEADERS) {
>-        rtc::CryptoOptions crypto_options;
>-        crypto_options.enable_encrypted_rtp_header_extensions = true;
>-        fake_rtp_dtls_transport2_->SetCryptoOptions(crypto_options);
>-        if (fake_rtcp_dtls_transport2_) {
>-          fake_rtcp_dtls_transport2_->SetCryptoOptions(crypto_options);
>-        }
>-      }
>-      if (flags2 & GCM_CIPHER) {
>-        fake_rtp_dtls_transport2_->SetSrtpCryptoSuite(
>-            rtc::SRTP_AEAD_AES_256_GCM);
>-        if (fake_rtcp_dtls_transport2_) {
>-          fake_rtcp_dtls_transport2_->SetSrtpCryptoSuite(
>-              rtc::SRTP_AEAD_AES_256_GCM);
>-        }
>-      }
>     }
>-    channel1_ =
>-        CreateChannel(worker_thread, network_thread_, &media_engine_,
>-                      std::move(ch1), fake_rtp_dtls_transport1_.get(),
>-                      fake_rtcp_dtls_transport1_.get(), rtp1, rtcp1, flags1);
>-    channel2_ =
>-        CreateChannel(worker_thread, network_thread_, &media_engine_,
>-                      std::move(ch2), fake_rtp_dtls_transport2_.get(),
>-                      fake_rtcp_dtls_transport2_.get(), rtp2, rtcp2, flags2);
>-    channel1_->SignalMediaMonitor.connect(this,
>-                                          &ChannelTest<T>::OnMediaMonitor1);
>-    channel2_->SignalMediaMonitor.connect(this,
>-                                          &ChannelTest<T>::OnMediaMonitor2);
>+    rtp_transport1_ = CreateRtpTransportBasedOnFlags(
>+        fake_rtp_packet_transport1_.get(), fake_rtcp_packet_transport1_.get(),
>+        fake_rtp_dtls_transport1_.get(), fake_rtcp_dtls_transport1_.get(),
>+        flags1);
>+    rtp_transport2_ = CreateRtpTransportBasedOnFlags(
>+        fake_rtp_packet_transport2_.get(), fake_rtcp_packet_transport2_.get(),
>+        fake_rtp_dtls_transport2_.get(), fake_rtcp_dtls_transport2_.get(),
>+        flags2);
>+
>+    channel1_ = CreateChannel(worker_thread, network_thread_, &media_engine_,
>+                              std::move(ch1), rtp_transport1_.get(), flags1);
>+    channel2_ = CreateChannel(worker_thread, network_thread_, &media_engine_,
>+                              std::move(ch2), rtp_transport2_.get(), flags2);
>     channel1_->SignalRtcpMuxFullyActive.connect(
>         this, &ChannelTest<T>::OnRtcpMuxFullyActive1);
>     channel2_->SignalRtcpMuxFullyActive.connect(
>         this, &ChannelTest<T>::OnRtcpMuxFullyActive2);
>-    if ((flags1 & DTLS) && (flags2 & DTLS)) {
>-      flags1 = (flags1 & ~SECURE);
>-      flags2 = (flags2 & ~SECURE);
>-    }
>-    CreateContent(flags1, kPcmuCodec, kH264Codec,
>-                  &local_media_content1_);
>-    CreateContent(flags2, kPcmuCodec, kH264Codec,
>-                  &local_media_content2_);
>+    CreateContent(flags1, kPcmuCodec, kH264Codec, &local_media_content1_);
>+    CreateContent(flags2, kPcmuCodec, kH264Codec, &local_media_content2_);
>     CopyContent(local_media_content1_, &remote_media_content1_);
>     CopyContent(local_media_content2_, &remote_media_content2_);
> 
>@@ -286,24 +245,64 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
>       rtc::Thread* network_thread,
>       cricket::MediaEngineInterface* engine,
>       std::unique_ptr<typename T::MediaChannel> ch,
>-      cricket::DtlsTransportInternal* fake_rtp_dtls_transport,
>-      cricket::DtlsTransportInternal* fake_rtcp_dtls_transport,
>-      rtc::PacketTransportInternal* fake_rtp_packet_transport,
>-      rtc::PacketTransportInternal* fake_rtcp_packet_transport,
>+      webrtc::RtpTransportInternal* rtp_transport,
>       int flags) {
>     rtc::Thread* signaling_thread = rtc::Thread::Current();
>-    auto channel = rtc::MakeUnique<typename T::Channel>(
>+    auto channel = absl::make_unique<typename T::Channel>(
>         worker_thread, network_thread, signaling_thread, engine, std::move(ch),
>-        cricket::CN_AUDIO, (flags & RTCP_MUX_REQUIRED) != 0,
>-        (flags & SECURE) != 0);
>-    if (!channel->NeedsRtcpTransport()) {
>-      fake_rtcp_dtls_transport = nullptr;
>-    }
>-    channel->Init_w(fake_rtp_dtls_transport, fake_rtcp_dtls_transport,
>-                    fake_rtp_packet_transport, fake_rtcp_packet_transport);
>+        cricket::CN_AUDIO, (flags & DTLS) != 0, rtc::CryptoOptions());
>+    channel->Init_w(rtp_transport);
>     return channel;
>   }
> 
>+  std::unique_ptr<webrtc::RtpTransportInternal> CreateRtpTransportBasedOnFlags(
>+      rtc::PacketTransportInternal* rtp_packet_transport,
>+      rtc::PacketTransportInternal* rtcp_packet_transport,
>+      DtlsTransportInternal* rtp_dtls_transport,
>+      DtlsTransportInternal* rtcp_dtls_transport,
>+      int flags) {
>+    if (flags & RTCP_MUX) {
>+      rtcp_packet_transport = nullptr;
>+      rtcp_dtls_transport = nullptr;
>+    }
>+
>+    if (flags & DTLS) {
>+      return CreateDtlsSrtpTransport(rtp_dtls_transport, rtcp_dtls_transport);
>+    } else {
>+      if (flags & RAW_PACKET_TRANSPORT) {
>+        return CreateUnencryptedTransport(rtp_packet_transport,
>+                                          rtcp_packet_transport);
>+      } else {
>+        return CreateUnencryptedTransport(rtp_dtls_transport,
>+                                          rtcp_dtls_transport);
>+      }
>+    }
>+  }
>+
>+  std::unique_ptr<webrtc::RtpTransport> CreateUnencryptedTransport(
>+      rtc::PacketTransportInternal* rtp_packet_transport,
>+      rtc::PacketTransportInternal* rtcp_packet_transport) {
>+    auto rtp_transport = absl::make_unique<webrtc::RtpTransport>(
>+        rtcp_packet_transport == nullptr);
>+
>+    rtp_transport->SetRtpPacketTransport(rtp_packet_transport);
>+    if (rtcp_packet_transport) {
>+      rtp_transport->SetRtcpPacketTransport(rtcp_packet_transport);
>+    }
>+    return rtp_transport;
>+  }
>+
>+  std::unique_ptr<webrtc::DtlsSrtpTransport> CreateDtlsSrtpTransport(
>+      cricket::DtlsTransportInternal* rtp_dtls_transport,
>+      cricket::DtlsTransportInternal* rtcp_dtls_transport) {
>+    auto dtls_srtp_transport = absl::make_unique<webrtc::DtlsSrtpTransport>(
>+        rtcp_dtls_transport == nullptr);
>+
>+    dtls_srtp_transport->SetDtlsTransports(rtp_dtls_transport,
>+                                           rtcp_dtls_transport);
>+    return dtls_srtp_transport;
>+  }
>+
>   void ConnectFakeTransports() {
>     network_thread_->Invoke<void>(RTC_FROM_HERE, [this] {
>       bool asymmetric = false;
>@@ -401,9 +400,7 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
>   bool AddStream1(int id) {
>     return channel1_->AddRecvStream(cricket::StreamParams::CreateLegacy(id));
>   }
>-  bool RemoveStream1(int id) {
>-    return channel1_->RemoveRecvStream(id);
>-  }
>+  bool RemoveStream1(int id) { return channel1_->RemoveRecvStream(id); }
> 
>   void SendRtp1() {
>     media_channel1_->SendRtp(rtp_packet_.data(), rtp_packet_.size(),
>@@ -483,18 +480,10 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
>     return data;
>   }
> 
>-  bool CheckNoRtp1() {
>-    return media_channel1_->CheckNoRtp();
>-  }
>-  bool CheckNoRtp2() {
>-    return media_channel2_->CheckNoRtp();
>-  }
>-  bool CheckNoRtcp1() {
>-    return media_channel1_->CheckNoRtcp();
>-  }
>-  bool CheckNoRtcp2() {
>-    return media_channel2_->CheckNoRtcp();
>-  }
>+  bool CheckNoRtp1() { return media_channel1_->CheckNoRtp(); }
>+  bool CheckNoRtp2() { return media_channel2_->CheckNoRtp(); }
>+  bool CheckNoRtcp1() { return media_channel1_->CheckNoRtcp(); }
>+  bool CheckNoRtcp2() { return media_channel2_->CheckNoRtcp(); }
> 
>   void CreateContent(int flags,
>                      const cricket::AudioCodec& audio_codec,
>@@ -511,7 +500,7 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
>   // kPcmuCodec is used as audio codec and kH264Codec is used as video codec.
>   typename T::Content* CreateMediaContentWithStream(uint32_t ssrc) {
>     typename T::Content* content = new typename T::Content();
>-    CreateContent(SECURE, kPcmuCodec, kH264Codec, content);
>+    CreateContent(0, kPcmuCodec, kH264Codec, content);
>     AddLegacyStreamInContent(ssrc, 0, content);
>     return content;
>   }
>@@ -541,16 +530,6 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
>     return false;  // overridden in specialized classes
>   }
> 
>-  void OnMediaMonitor1(typename T::Channel* channel,
>-                       const typename T::MediaInfo& info) {
>-    RTC_DCHECK_EQ(channel, channel1_.get());
>-    media_info_callbacks1_++;
>-  }
>-  void OnMediaMonitor2(typename T::Channel* channel,
>-                       const typename T::MediaInfo& info) {
>-    RTC_DCHECK_EQ(channel, channel2_.get());
>-    media_info_callbacks2_++;
>-  }
>   void OnRtcpMuxFullyActive1(const std::string&) {
>     rtcp_mux_activated_callbacks1_++;
>   }
>@@ -594,8 +573,8 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
>     EXPECT_EQ(0U, media_channel1_->codecs().size());
>     EXPECT_TRUE(channel1_->SetRemoteContent(&content, SdpType::kAnswer, NULL));
>     ASSERT_EQ(1U, media_channel1_->codecs().size());
>-    EXPECT_TRUE(CodecMatches(content.codecs()[0],
>-                             media_channel1_->codecs()[0]));
>+    EXPECT_TRUE(
>+        CodecMatches(content.codecs()[0], media_channel1_->codecs()[0]));
>   }
> 
>   // Test that SetLocalContent and SetRemoteContent properly deals
>@@ -608,8 +587,8 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
>     EXPECT_EQ(0U, media_channel1_->codecs().size());
>     EXPECT_TRUE(channel1_->SetRemoteContent(&content, SdpType::kAnswer, NULL));
>     ASSERT_EQ(1U, media_channel1_->codecs().size());
>-    EXPECT_TRUE(CodecMatches(content.codecs()[0],
>-                             media_channel1_->codecs()[0]));
>+    EXPECT_TRUE(
>+        CodecMatches(content.codecs()[0], media_channel1_->codecs()[0]));
>   }
> 
>   // Test that SetLocalContent and SetRemoteContent properly set RTCP
>@@ -707,7 +686,7 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
> 
>     // Let channel 2 update the content by sending |stream2| and enable SRTP.
>     typename T::Content content3;
>-    CreateContent(SECURE, kPcmuCodec, kH264Codec, &content3);
>+    CreateContent(0, kPcmuCodec, kH264Codec, &content3);
>     content3.AddStream(stream2);
>     EXPECT_TRUE(channel2_->SetLocalContent(&content3, SdpType::kOffer, NULL));
>     ASSERT_EQ(1u, media_channel2_->send_streams().size());
>@@ -719,197 +698,18 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
> 
>     // Channel 1 replies but stop sending stream1.
>     typename T::Content content4;
>-    CreateContent(SECURE, kPcmuCodec, kH264Codec, &content4);
>+    CreateContent(0, kPcmuCodec, kH264Codec, &content4);
>     EXPECT_TRUE(channel1_->SetLocalContent(&content4, SdpType::kAnswer, NULL));
>     EXPECT_EQ(0u, media_channel1_->send_streams().size());
> 
>     EXPECT_TRUE(channel2_->SetRemoteContent(&content4, SdpType::kAnswer, NULL));
>     EXPECT_EQ(0u, media_channel2_->recv_streams().size());
> 
>-    EXPECT_TRUE(channel1_->srtp_active());
>-    EXPECT_TRUE(channel2_->srtp_active());
>     SendCustomRtp2(kSsrc2, 0);
>     WaitForThreads();
>     EXPECT_TRUE(CheckCustomRtp1(kSsrc2, 0));
>   }
> 
>-  enum EncryptedHeaderTestScenario {
>-    // Offer/Answer are processed before DTLS completes.
>-    DEFAULT,
>-    // DTLS completes before any Offer/Answer have been sent.
>-    DTLS_BEFORE_OFFER_ANSWER,
>-    // DTLS completes after channel 2 has processed (remote) Offer and (local)
>-    // Answer.
>-    DTLS_AFTER_CHANNEL2_READY,
>-  };
>-
>-  // Test that encrypted header extensions are working and can be changed when
>-  // sending a new OFFER/ANSWER.
>-  void TestChangeEncryptedHeaderExtensions(int flags,
>-      EncryptedHeaderTestScenario scenario = DEFAULT) {
>-    RTC_CHECK(scenario == 0 || (flags & DTLS));
>-    struct PacketListener : public sigslot::has_slots<> {
>-      PacketListener() {}
>-      void OnReadPacket(rtc::PacketTransportInternal* transport,
>-          const char* data, size_t size, const rtc::PacketTime& time,
>-          int flags) {
>-        CompareHeaderExtensions(
>-            reinterpret_cast<const char*>(kPcmuFrameWithExtensions),
>-            sizeof(kPcmuFrameWithExtensions), data, size, encrypted_headers,
>-            false);
>-      }
>-      std::vector<int> encrypted_headers;
>-    } packet_listener1, packet_listener2;
>-
>-    cricket::StreamParams stream1;
>-    stream1.groupid = "group1";
>-    stream1.id = "stream1";
>-    stream1.ssrcs.push_back(kSsrc1);
>-    stream1.cname = "stream1_cname";
>-
>-    cricket::StreamParams stream2;
>-    stream2.groupid = "group1";
>-    stream2.id = "stream2";
>-    stream2.ssrcs.push_back(kSsrc2);
>-    stream2.cname = "stream2_cname";
>-
>-    // Use SRTP when testing encrypted extensions.
>-    int channel_flags = flags | SECURE | ENCRYPTED_HEADERS;
>-    // Enable SDES if channel is not using DTLS.
>-    int content_flags = (channel_flags & DTLS) == 0 ? SECURE : 0;
>-
>-    // kPcmuFrameWithExtensions contains RTP extension headers with ids 1-4.
>-    // Make sure to use URIs that are supported for encryption.
>-    cricket::RtpHeaderExtensions extensions1;
>-    extensions1.push_back(
>-        RtpExtension(RtpExtension::kAudioLevelUri, 10));
>-    extensions1.push_back(
>-        RtpExtension(RtpExtension::kAudioLevelUri, 1, true));
>-
>-    cricket::RtpHeaderExtensions extensions2;
>-    extensions2.push_back(
>-        RtpExtension(RtpExtension::kAudioLevelUri, 10));
>-    extensions2.push_back(
>-        RtpExtension(RtpExtension::kAudioLevelUri, 2, true));
>-    extensions2.push_back(
>-        RtpExtension(RtpExtension::kVideoRotationUri, 3));
>-    extensions2.push_back(
>-        RtpExtension(RtpExtension::kTimestampOffsetUri, 4, true));
>-
>-    // Setup a call where channel 1 send |stream1| to channel 2.
>-    CreateChannels(channel_flags, channel_flags);
>-    fake_rtp_dtls_transport1_->fake_ice_transport()->SignalReadPacket.connect(
>-        &packet_listener1, &PacketListener::OnReadPacket);
>-    fake_rtp_dtls_transport2_->fake_ice_transport()->SignalReadPacket.connect(
>-        &packet_listener2, &PacketListener::OnReadPacket);
>-
>-    if (scenario == DTLS_BEFORE_OFFER_ANSWER) {
>-      ConnectFakeTransports();
>-      WaitForThreads();
>-    }
>-
>-    typename T::Content content1;
>-    CreateContent(content_flags, kPcmuCodec, kH264Codec, &content1);
>-    content1.AddStream(stream1);
>-    content1.set_rtp_header_extensions(extensions1);
>-    EXPECT_TRUE(channel1_->SetLocalContent(&content1, SdpType::kOffer, NULL));
>-    EXPECT_TRUE(channel1_->Enable(true));
>-    EXPECT_EQ(1u, media_channel1_->send_streams().size());
>-    packet_listener1.encrypted_headers.push_back(1);
>-
>-    EXPECT_TRUE(channel2_->SetRemoteContent(&content1, SdpType::kOffer, NULL));
>-    EXPECT_EQ(1u, media_channel2_->recv_streams().size());
>-
>-    // Channel 2 sends back |stream2|.
>-    typename T::Content content2;
>-    CreateContent(content_flags, kPcmuCodec, kH264Codec, &content2);
>-    content2.AddStream(stream2);
>-    content2.set_rtp_header_extensions(extensions1);
>-    EXPECT_TRUE(channel2_->SetLocalContent(&content2, SdpType::kAnswer, NULL));
>-    EXPECT_TRUE(channel2_->Enable(true));
>-    EXPECT_EQ(1u, media_channel2_->send_streams().size());
>-    packet_listener2.encrypted_headers.push_back(1);
>-
>-    if (scenario == DTLS_AFTER_CHANNEL2_READY) {
>-      ConnectFakeTransports();
>-      WaitForThreads();
>-    }
>-
>-    if (scenario == DTLS_BEFORE_OFFER_ANSWER ||
>-        scenario == DTLS_AFTER_CHANNEL2_READY) {
>-      // In both scenarios with partially completed Offer/Answer, sending
>-      // packets from Channel 2 to Channel 1 should work.
>-      SendCustomRtp2(kSsrc2, 0);
>-      WaitForThreads();
>-      EXPECT_TRUE(CheckCustomRtp1(kSsrc2, 0));
>-    }
>-
>-    EXPECT_TRUE(channel1_->SetRemoteContent(&content2, SdpType::kAnswer, NULL));
>-    EXPECT_EQ(1u, media_channel1_->recv_streams().size());
>-
>-    if (scenario == DEFAULT) {
>-      ConnectFakeTransports();
>-      WaitForThreads();
>-    }
>-
>-    SendCustomRtp1(kSsrc1, 0);
>-    SendCustomRtp2(kSsrc2, 0);
>-    WaitForThreads();
>-    EXPECT_TRUE(CheckCustomRtp2(kSsrc1, 0));
>-    EXPECT_TRUE(CheckCustomRtp1(kSsrc2, 0));
>-
>-    // Let channel 2 update the encrypted header extensions.
>-    typename T::Content content3;
>-    CreateContent(content_flags, kPcmuCodec, kH264Codec, &content3);
>-    content3.AddStream(stream2);
>-    content3.set_rtp_header_extensions(extensions2);
>-    EXPECT_TRUE(channel2_->SetLocalContent(&content3, SdpType::kOffer, NULL));
>-    ASSERT_EQ(1u, media_channel2_->send_streams().size());
>-    EXPECT_EQ(stream2, media_channel2_->send_streams()[0]);
>-    packet_listener2.encrypted_headers.clear();
>-    packet_listener2.encrypted_headers.push_back(2);
>-    packet_listener2.encrypted_headers.push_back(4);
>-
>-    EXPECT_TRUE(channel1_->SetRemoteContent(&content3, SdpType::kOffer, NULL));
>-    ASSERT_EQ(1u, media_channel1_->recv_streams().size());
>-    EXPECT_EQ(stream2, media_channel1_->recv_streams()[0]);
>-
>-    // Channel 1 is already sending the new encrypted extensions. These
>-    // can be decrypted by channel 2. Channel 2 is still sending the old
>-    // encrypted extensions (which can be decrypted by channel 1).
>-
>-    if (flags & DTLS) {
>-      // DTLS supports updating the encrypted extensions with only the OFFER
>-      // being processed. For SDES both the OFFER and ANSWER must have been
>-      // processed to update encrypted extensions, so we can't check this case.
>-      SendCustomRtp1(kSsrc1, 0);
>-      SendCustomRtp2(kSsrc2, 0);
>-      WaitForThreads();
>-      EXPECT_TRUE(CheckCustomRtp2(kSsrc1, 0));
>-      EXPECT_TRUE(CheckCustomRtp1(kSsrc2, 0));
>-    }
>-
>-    // Channel 1 replies with the same extensions.
>-    typename T::Content content4;
>-    CreateContent(content_flags, kPcmuCodec, kH264Codec, &content4);
>-    content4.AddStream(stream1);
>-    content4.set_rtp_header_extensions(extensions2);
>-    EXPECT_TRUE(channel1_->SetLocalContent(&content4, SdpType::kAnswer, NULL));
>-    EXPECT_EQ(1u, media_channel1_->send_streams().size());
>-    packet_listener1.encrypted_headers.clear();
>-    packet_listener1.encrypted_headers.push_back(2);
>-    packet_listener1.encrypted_headers.push_back(4);
>-
>-    EXPECT_TRUE(channel2_->SetRemoteContent(&content4, SdpType::kAnswer, NULL));
>-    EXPECT_EQ(1u, media_channel2_->recv_streams().size());
>-
>-    SendCustomRtp1(kSsrc1, 0);
>-    SendCustomRtp2(kSsrc2, 0);
>-    WaitForThreads();
>-    EXPECT_TRUE(CheckCustomRtp2(kSsrc1, 0));
>-    EXPECT_TRUE(CheckCustomRtp1(kSsrc2, 0));
>-  }
>-
>   // Test that we only start playout and sending at the right times.
>   void TestPlayoutAndSendingStates() {
>     CreateChannels(0, 0);
>@@ -1044,8 +844,10 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
>     static constexpr int kLastPacketId = 100;
>     // Ipv4(20) + UDP(8).
>     static constexpr int kTransportOverheadPerPacket = 28;
>+    static constexpr int kSrtpOverheadPerPacket = 10;
> 
>-    CreateChannels(0, 0);
>+    CreateChannels(DTLS, DTLS);
>+    SendInitiate();
> 
>     typename T::MediaChannel* media_channel1 =
>         static_cast<typename T::MediaChannel*>(channel1_->media_channel());
>@@ -1061,7 +863,7 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
>       // The transport channel becomes disconnected.
>       fake_rtp_dtls_transport1_->ice_transport()->SignalNetworkRouteChanged(
> 
>-          rtc::Optional<rtc::NetworkRoute>(network_route));
>+          absl::optional<rtc::NetworkRoute>(network_route));
>     });
>     WaitForThreads();
>     EXPECT_EQ(1, media_channel1->num_network_route_changes());
>@@ -1078,16 +880,19 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
>       // The transport channel becomes connected.
>       fake_rtp_dtls_transport1_->ice_transport()->SignalNetworkRouteChanged(
> 
>-          rtc::Optional<rtc::NetworkRoute>(network_route));
>+          absl::optional<rtc::NetworkRoute>(network_route));
>     });
>     WaitForThreads();
>     EXPECT_EQ(1, media_channel1->num_network_route_changes());
>-    rtc::NetworkRoute expected_network_route(true, kLocalNetId, kRemoteNetId,
>-                                             kLastPacketId);
>+    rtc::NetworkRoute expected_network_route;
>+    expected_network_route.connected = true;
>+    expected_network_route.local_network_id = kLocalNetId;
>+    expected_network_route.remote_network_id = kRemoteNetId;
>+    expected_network_route.last_sent_packet_id = kLastPacketId;
>     EXPECT_EQ(expected_network_route, media_channel1->last_network_route());
>     EXPECT_EQ(kLastPacketId,
>               media_channel1->last_network_route().last_sent_packet_id);
>-    EXPECT_EQ(kTransportOverheadPerPacket,
>+    EXPECT_EQ(kTransportOverheadPerPacket + kSrtpOverheadPerPacket,
>               media_channel1->transport_overhead_per_packet());
>   }
> 
>@@ -1125,8 +930,9 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
>         T::MediaChannel::SendRtcp(kRtcpReport, sizeof(kRtcpReport));
>       }
>     };
>-    CreateChannels(rtc::MakeUnique<LastWordMediaChannel>(),
>-                   rtc::MakeUnique<LastWordMediaChannel>(), RTCP_MUX, RTCP_MUX);
>+    CreateChannels(absl::make_unique<LastWordMediaChannel>(),
>+                   absl::make_unique<LastWordMediaChannel>(), RTCP_MUX,
>+                   RTCP_MUX);
>     EXPECT_TRUE(SendInitiate());
>     EXPECT_TRUE(SendAccept());
>     EXPECT_TRUE(Terminate());
>@@ -1134,11 +940,11 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
> 
>   // Send voice RTP data to the other side and ensure it gets there.
>   void SendRtpToRtp() {
>-    CreateChannels(RTCP_MUX | RTCP_MUX_REQUIRED, RTCP_MUX | RTCP_MUX_REQUIRED);
>+    CreateChannels(RTCP_MUX, RTCP_MUX);
>     EXPECT_TRUE(SendInitiate());
>     EXPECT_TRUE(SendAccept());
>-    EXPECT_FALSE(channel1_->NeedsRtcpTransport());
>-    EXPECT_FALSE(channel2_->NeedsRtcpTransport());
>+    EXPECT_EQ(nullptr, channel1_->rtcp_packet_transport());
>+    EXPECT_EQ(nullptr, channel2_->rtcp_packet_transport());
>     SendRtp1();
>     SendRtp2();
>     WaitForThreads();
>@@ -1166,216 +972,19 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
>     CreateChannels(0, 0);
>     EXPECT_TRUE(SendInitiate());
>     EXPECT_TRUE(SendAccept());
>-    EXPECT_TRUE(channel1_->NeedsRtcpTransport());
>-    EXPECT_TRUE(channel2_->NeedsRtcpTransport());
>-    SendRtcp1();
>-    SendRtcp2();
>-    WaitForThreads();
>-    EXPECT_TRUE(CheckRtcp1());
>-    EXPECT_TRUE(CheckRtcp2());
>-    EXPECT_TRUE(CheckNoRtcp1());
>-    EXPECT_TRUE(CheckNoRtcp2());
>-  }
>-
>-  // Check that RTCP is transmitted if only the initiator supports mux.
>-  void SendRtcpMuxToRtcp() {
>-    CreateChannels(RTCP_MUX, 0);
>-    EXPECT_TRUE(SendInitiate());
>-    EXPECT_TRUE(SendAccept());
>-    EXPECT_TRUE(channel1_->NeedsRtcpTransport());
>-    EXPECT_TRUE(channel2_->NeedsRtcpTransport());
>-    SendRtcp1();
>-    SendRtcp2();
>-    WaitForThreads();
>-    EXPECT_TRUE(CheckRtcp1());
>-    EXPECT_TRUE(CheckRtcp2());
>-    EXPECT_TRUE(CheckNoRtcp1());
>-    EXPECT_TRUE(CheckNoRtcp2());
>-  }
>-
>-  // Check that RTP and RTCP are transmitted ok when both sides support mux.
>-  void SendRtcpMuxToRtcpMux() {
>-    CreateChannels(RTCP_MUX, RTCP_MUX);
>-    EXPECT_TRUE(SendInitiate());
>-    EXPECT_TRUE(channel1_->NeedsRtcpTransport());
>-    EXPECT_FALSE(channel2_->NeedsRtcpTransport());
>-    EXPECT_EQ(0, rtcp_mux_activated_callbacks1_);
>-    EXPECT_TRUE(SendAccept());
>-    EXPECT_FALSE(channel1_->NeedsRtcpTransport());
>-    EXPECT_EQ(1, rtcp_mux_activated_callbacks1_);
>-    SendRtp1();
>-    SendRtp2();
>+    EXPECT_NE(nullptr, channel1_->rtcp_packet_transport());
>+    EXPECT_NE(nullptr, channel2_->rtcp_packet_transport());
>     SendRtcp1();
>     SendRtcp2();
>     WaitForThreads();
>-    EXPECT_TRUE(CheckRtp1());
>-    EXPECT_TRUE(CheckRtp2());
>-    EXPECT_TRUE(CheckNoRtp1());
>-    EXPECT_TRUE(CheckNoRtp2());
>     EXPECT_TRUE(CheckRtcp1());
>     EXPECT_TRUE(CheckRtcp2());
>     EXPECT_TRUE(CheckNoRtcp1());
>     EXPECT_TRUE(CheckNoRtcp2());
>   }
> 
>-  // Check that RTP and RTCP are transmitted ok when both sides
>-  // support mux and one the offerer requires mux.
>-  void SendRequireRtcpMuxToRtcpMux() {
>-    CreateChannels(RTCP_MUX | RTCP_MUX_REQUIRED, RTCP_MUX);
>-    EXPECT_TRUE(SendInitiate());
>-    EXPECT_FALSE(channel1_->NeedsRtcpTransport());
>-    EXPECT_FALSE(channel2_->NeedsRtcpTransport());
>-    EXPECT_TRUE(SendAccept());
>-    SendRtp1();
>-    SendRtp2();
>-    SendRtcp1();
>-    SendRtcp2();
>-    WaitForThreads();
>-    EXPECT_TRUE(CheckRtp1());
>-    EXPECT_TRUE(CheckRtp2());
>-    EXPECT_TRUE(CheckNoRtp1());
>-    EXPECT_TRUE(CheckNoRtp2());
>-    EXPECT_TRUE(CheckRtcp1());
>-    EXPECT_TRUE(CheckRtcp2());
>-    EXPECT_TRUE(CheckNoRtcp1());
>-    EXPECT_TRUE(CheckNoRtcp2());
>-  }
>-
>-  // Check that RTP and RTCP are transmitted ok when both sides
>-  // support mux and only the answerer requires rtcp mux.
>-  void SendRtcpMuxToRequireRtcpMux() {
>-    CreateChannels(RTCP_MUX, RTCP_MUX | RTCP_MUX_REQUIRED);
>-    EXPECT_TRUE(SendInitiate());
>-    EXPECT_TRUE(channel1_->NeedsRtcpTransport());
>-    EXPECT_FALSE(channel2_->NeedsRtcpTransport());
>-    EXPECT_EQ(0, rtcp_mux_activated_callbacks1_);
>-    EXPECT_TRUE(SendAccept());
>-    EXPECT_FALSE(channel1_->NeedsRtcpTransport());
>-    EXPECT_EQ(1, rtcp_mux_activated_callbacks1_);
>-    SendRtp1();
>-    SendRtp2();
>-    SendRtcp1();
>-    SendRtcp2();
>-    WaitForThreads();
>-    EXPECT_TRUE(CheckRtp1());
>-    EXPECT_TRUE(CheckRtp2());
>-    EXPECT_TRUE(CheckNoRtp1());
>-    EXPECT_TRUE(CheckNoRtp2());
>-    EXPECT_TRUE(CheckRtcp1());
>-    EXPECT_TRUE(CheckRtcp2());
>-    EXPECT_TRUE(CheckNoRtcp1());
>-    EXPECT_TRUE(CheckNoRtcp2());
>-  }
>-
>-  // Check that RTP and RTCP are transmitted ok when both sides
>-  // require mux.
>-  void SendRequireRtcpMuxToRequireRtcpMux() {
>-    CreateChannels(RTCP_MUX | RTCP_MUX_REQUIRED, RTCP_MUX | RTCP_MUX_REQUIRED);
>-    EXPECT_TRUE(SendInitiate());
>-    EXPECT_FALSE(channel1_->NeedsRtcpTransport());
>-    EXPECT_FALSE(channel2_->NeedsRtcpTransport());
>-    EXPECT_TRUE(SendAccept());
>-    EXPECT_FALSE(channel1_->NeedsRtcpTransport());
>-    SendRtp1();
>-    SendRtp2();
>-    SendRtcp1();
>-    SendRtcp2();
>-    WaitForThreads();
>-    EXPECT_TRUE(CheckRtp1());
>-    EXPECT_TRUE(CheckRtp2());
>-    EXPECT_TRUE(CheckNoRtp1());
>-    EXPECT_TRUE(CheckNoRtp2());
>-    EXPECT_TRUE(CheckRtcp1());
>-    EXPECT_TRUE(CheckRtcp2());
>-    EXPECT_TRUE(CheckNoRtcp1());
>-    EXPECT_TRUE(CheckNoRtcp2());
>-  }
>-
>-  // Check that SendAccept fails if the answerer doesn't support mux
>-  // and the offerer requires it.
>-  void SendRequireRtcpMuxToNoRtcpMux() {
>-    CreateChannels(RTCP_MUX | RTCP_MUX_REQUIRED, 0);
>-    EXPECT_TRUE(SendInitiate());
>-    EXPECT_FALSE(channel1_->NeedsRtcpTransport());
>-    EXPECT_TRUE(channel2_->NeedsRtcpTransport());
>-    EXPECT_FALSE(SendAccept());
>-  }
>-
>-  // Check that RTCP data sent by the initiator before the accept is not muxed.
>-  void SendEarlyRtcpMuxToRtcp() {
>-    CreateChannels(RTCP_MUX, 0);
>-    EXPECT_TRUE(SendInitiate());
>-    EXPECT_TRUE(channel1_->NeedsRtcpTransport());
>-    EXPECT_TRUE(channel2_->NeedsRtcpTransport());
>-
>-    // RTCP can be sent before the call is accepted, if the transport is ready.
>-    // It should not be muxed though, as the remote side doesn't support mux.
>-    SendRtcp1();
>-    WaitForThreads();
>-    EXPECT_TRUE(CheckNoRtp2());
>-    EXPECT_TRUE(CheckRtcp2());
>-
>-    // Send RTCP packet from callee and verify that it is received.
>-    SendRtcp2();
>-    WaitForThreads();
>-    EXPECT_TRUE(CheckNoRtp1());
>-    EXPECT_TRUE(CheckRtcp1());
>-
>-    // Complete call setup and ensure everything is still OK.
>-    EXPECT_TRUE(SendAccept());
>-    EXPECT_TRUE(channel1_->NeedsRtcpTransport());
>-    SendRtcp1();
>-    SendRtcp2();
>-    WaitForThreads();
>-    EXPECT_TRUE(CheckRtcp2());
>-    EXPECT_TRUE(CheckRtcp1());
>-  }
>-
>-
>-  // Check that RTCP data is not muxed until both sides have enabled muxing,
>-  // but that we properly demux before we get the accept message, since there
>-  // is a race between RTP data and the jingle accept.
>-  void SendEarlyRtcpMuxToRtcpMux() {
>-    CreateChannels(RTCP_MUX, RTCP_MUX);
>-    EXPECT_TRUE(SendInitiate());
>-    EXPECT_TRUE(channel1_->NeedsRtcpTransport());
>-    EXPECT_FALSE(channel2_->NeedsRtcpTransport());
>-
>-    // RTCP can't be sent yet, since the RTCP transport isn't writable, and
>-    // we haven't yet received the accept that says we should mux.
>-    SendRtcp1();
>-    WaitForThreads();
>-    EXPECT_TRUE(CheckNoRtcp2());
>-
>-    // Send muxed RTCP packet from callee and verify that it is received.
>-    SendRtcp2();
>-    WaitForThreads();
>-    EXPECT_TRUE(CheckNoRtp1());
>-    EXPECT_TRUE(CheckRtcp1());
>-
>-    // Complete call setup and ensure everything is still OK.
>-    EXPECT_EQ(0, rtcp_mux_activated_callbacks1_);
>-    EXPECT_TRUE(SendAccept());
>-    EXPECT_FALSE(channel1_->NeedsRtcpTransport());
>-    EXPECT_EQ(1, rtcp_mux_activated_callbacks1_);
>-    SendRtcp1();
>-    SendRtcp2();
>-    WaitForThreads();
>-    EXPECT_TRUE(CheckRtcp2());
>-    EXPECT_TRUE(CheckRtcp1());
>-  }
>-
>-  // Test that we properly send SRTP with RTCP in both directions.
>-  // You can pass in DTLS, RTCP_MUX, and RAW_PACKET_TRANSPORT as flags.
>-  void SendSrtpToSrtp(int flags1_in = 0, int flags2_in = 0) {
>-    RTC_CHECK((flags1_in & ~(RTCP_MUX | DTLS | RAW_PACKET_TRANSPORT)) == 0);
>-    RTC_CHECK((flags2_in & ~(RTCP_MUX | DTLS | RAW_PACKET_TRANSPORT)) == 0);
>-
>-    int flags1 = SECURE | flags1_in;
>-    int flags2 = SECURE | flags2_in;
>-    bool dtls1 = !!(flags1_in & DTLS);
>-    bool dtls2 = !!(flags2_in & DTLS);
>-    CreateChannels(flags1, flags2);
>+  void SendDtlsSrtpToDtlsSrtp(int flags1, int flags2) {
>+    CreateChannels(flags1 | DTLS, flags2 | DTLS);
>     EXPECT_FALSE(channel1_->srtp_active());
>     EXPECT_FALSE(channel2_->srtp_active());
>     EXPECT_TRUE(SendInitiate());
>@@ -1385,43 +994,6 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
>     EXPECT_TRUE(SendAccept());
>     EXPECT_TRUE(channel1_->srtp_active());
>     EXPECT_TRUE(channel2_->srtp_active());
>-    EXPECT_EQ(dtls1 && dtls2, channel1_->dtls_active());
>-    EXPECT_EQ(dtls1 && dtls2, channel2_->dtls_active());
>-    SendRtp1();
>-    SendRtp2();
>-    SendRtcp1();
>-    SendRtcp2();
>-    WaitForThreads();
>-    EXPECT_TRUE(CheckRtp1());
>-    EXPECT_TRUE(CheckRtp2());
>-    EXPECT_TRUE(CheckNoRtp1());
>-    EXPECT_TRUE(CheckNoRtp2());
>-    EXPECT_TRUE(CheckRtcp1());
>-    EXPECT_TRUE(CheckRtcp2());
>-    EXPECT_TRUE(CheckNoRtcp1());
>-    EXPECT_TRUE(CheckNoRtcp2());
>-  }
>-
>-  // Test that the DTLS to SDES fallback is not supported and the negotiation
>-  // between DTLS to SDES end points will fail.
>-  void SendDtlsToSdesNotSupported() {
>-    int flags1 = SECURE | DTLS;
>-    int flags2 = SECURE;
>-    CreateChannels(flags1, flags2);
>-    EXPECT_FALSE(channel1_->srtp_active());
>-    EXPECT_FALSE(channel2_->srtp_active());
>-    EXPECT_FALSE(SendInitiate());
>-  }
>-
>-  // Test that we properly handling SRTP negotiating down to RTP.
>-  void SendSrtpToRtp() {
>-    CreateChannels(SECURE, 0);
>-    EXPECT_FALSE(channel1_->srtp_active());
>-    EXPECT_FALSE(channel2_->srtp_active());
>-    EXPECT_TRUE(SendInitiate());
>-    EXPECT_TRUE(SendAccept());
>-    EXPECT_FALSE(channel1_->srtp_active());
>-    EXPECT_FALSE(channel2_->srtp_active());
>     SendRtp1();
>     SendRtp2();
>     SendRtcp1();
>@@ -1440,49 +1012,42 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
>   // Test that we can send and receive early media when a provisional answer is
>   // sent and received. The test uses SRTP, RTCP mux and SSRC mux.
>   void SendEarlyMediaUsingRtcpMuxSrtp() {
>-      int sequence_number1_1 = 0, sequence_number2_2 = 0;
>-
>-      CreateChannels(SSRC_MUX | RTCP_MUX | SECURE,
>-                     SSRC_MUX | RTCP_MUX | SECURE);
>-      EXPECT_TRUE(SendOffer());
>-      EXPECT_TRUE(SendProvisionalAnswer());
>-      EXPECT_TRUE(channel1_->srtp_active());
>-      EXPECT_TRUE(channel2_->srtp_active());
>-      EXPECT_TRUE(channel1_->NeedsRtcpTransport());
>-      EXPECT_TRUE(channel2_->NeedsRtcpTransport());
>-      WaitForThreads();  // Wait for 'sending' flag go through network thread.
>-      SendCustomRtcp1(kSsrc1);
>-      SendCustomRtp1(kSsrc1, ++sequence_number1_1);
>-      WaitForThreads();
>-      EXPECT_TRUE(CheckCustomRtcp2(kSsrc1));
>-      EXPECT_TRUE(CheckCustomRtp2(kSsrc1, sequence_number1_1));
>-
>-      // Send packets from callee and verify that it is received.
>-      SendCustomRtcp2(kSsrc2);
>-      SendCustomRtp2(kSsrc2, ++sequence_number2_2);
>-      WaitForThreads();
>-      EXPECT_TRUE(CheckCustomRtcp1(kSsrc2));
>-      EXPECT_TRUE(CheckCustomRtp1(kSsrc2, sequence_number2_2));
>-
>-      // Complete call setup and ensure everything is still OK.
>-      EXPECT_EQ(0, rtcp_mux_activated_callbacks1_);
>-      EXPECT_EQ(0, rtcp_mux_activated_callbacks2_);
>-      EXPECT_TRUE(SendFinalAnswer());
>-      EXPECT_FALSE(channel1_->NeedsRtcpTransport());
>-      EXPECT_FALSE(channel2_->NeedsRtcpTransport());
>-      EXPECT_EQ(1, rtcp_mux_activated_callbacks1_);
>-      EXPECT_EQ(1, rtcp_mux_activated_callbacks2_);
>-      EXPECT_TRUE(channel1_->srtp_active());
>-      EXPECT_TRUE(channel2_->srtp_active());
>-      SendCustomRtcp1(kSsrc1);
>-      SendCustomRtp1(kSsrc1, ++sequence_number1_1);
>-      SendCustomRtcp2(kSsrc2);
>-      SendCustomRtp2(kSsrc2, ++sequence_number2_2);
>-      WaitForThreads();
>-      EXPECT_TRUE(CheckCustomRtcp2(kSsrc1));
>-      EXPECT_TRUE(CheckCustomRtp2(kSsrc1, sequence_number1_1));
>-      EXPECT_TRUE(CheckCustomRtcp1(kSsrc2));
>-      EXPECT_TRUE(CheckCustomRtp1(kSsrc2, sequence_number2_2));
>+    int sequence_number1_1 = 0, sequence_number2_2 = 0;
>+
>+    CreateChannels(SSRC_MUX | RTCP_MUX | DTLS, SSRC_MUX | RTCP_MUX | DTLS);
>+    EXPECT_TRUE(SendOffer());
>+    EXPECT_TRUE(SendProvisionalAnswer());
>+    EXPECT_TRUE(channel1_->srtp_active());
>+    EXPECT_TRUE(channel2_->srtp_active());
>+    EXPECT_EQ(nullptr, channel1_->rtcp_packet_transport());
>+    EXPECT_EQ(nullptr, channel2_->rtcp_packet_transport());
>+    WaitForThreads();  // Wait for 'sending' flag go through network thread.
>+    SendCustomRtcp1(kSsrc1);
>+    SendCustomRtp1(kSsrc1, ++sequence_number1_1);
>+    WaitForThreads();
>+    EXPECT_TRUE(CheckCustomRtcp2(kSsrc1));
>+    EXPECT_TRUE(CheckCustomRtp2(kSsrc1, sequence_number1_1));
>+
>+    // Send packets from callee and verify that it is received.
>+    SendCustomRtcp2(kSsrc2);
>+    SendCustomRtp2(kSsrc2, ++sequence_number2_2);
>+    WaitForThreads();
>+    EXPECT_TRUE(CheckCustomRtcp1(kSsrc2));
>+    EXPECT_TRUE(CheckCustomRtp1(kSsrc2, sequence_number2_2));
>+
>+    // Complete call setup and ensure everything is still OK.
>+    EXPECT_TRUE(SendFinalAnswer());
>+    EXPECT_TRUE(channel1_->srtp_active());
>+    EXPECT_TRUE(channel2_->srtp_active());
>+    SendCustomRtcp1(kSsrc1);
>+    SendCustomRtp1(kSsrc1, ++sequence_number1_1);
>+    SendCustomRtcp2(kSsrc2);
>+    SendCustomRtp2(kSsrc2, ++sequence_number2_2);
>+    WaitForThreads();
>+    EXPECT_TRUE(CheckCustomRtcp2(kSsrc1));
>+    EXPECT_TRUE(CheckCustomRtp2(kSsrc1, sequence_number1_1));
>+    EXPECT_TRUE(CheckCustomRtcp1(kSsrc2));
>+    EXPECT_TRUE(CheckCustomRtp1(kSsrc2, sequence_number2_2));
>   }
> 
>   // Test that we properly send RTP without SRTP from a thread.
>@@ -1508,37 +1073,14 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
>     EXPECT_TRUE(CheckNoRtcp2());
>   }
> 
>-  // Test that we properly send SRTP with RTCP from a thread.
>-  void SendSrtpToSrtpOnThread() {
>-    CreateChannels(SECURE, SECURE);
>-    EXPECT_TRUE(SendInitiate());
>-    EXPECT_TRUE(SendAccept());
>-    ScopedCallThread send_rtp1([this] { SendRtp1(); });
>-    ScopedCallThread send_rtp2([this] { SendRtp2(); });
>-    ScopedCallThread send_rtcp1([this] { SendRtcp1(); });
>-    ScopedCallThread send_rtcp2([this] { SendRtcp2(); });
>-    rtc::Thread* involved_threads[] = {send_rtp1.thread(), send_rtp2.thread(),
>-                                       send_rtcp1.thread(),
>-                                       send_rtcp2.thread()};
>-    WaitForThreads(involved_threads);
>-    EXPECT_TRUE(CheckRtp1());
>-    EXPECT_TRUE(CheckRtp2());
>-    EXPECT_TRUE(CheckNoRtp1());
>-    EXPECT_TRUE(CheckNoRtp2());
>-    EXPECT_TRUE(CheckRtcp1());
>-    EXPECT_TRUE(CheckRtcp2());
>-    EXPECT_TRUE(CheckNoRtcp1());
>-    EXPECT_TRUE(CheckNoRtcp2());
>-  }
>-
>   // Test that the mediachannel retains its sending state after the transport
>   // becomes non-writable.
>   void SendWithWritabilityLoss() {
>-    CreateChannels(RTCP_MUX | RTCP_MUX_REQUIRED, RTCP_MUX | RTCP_MUX_REQUIRED);
>+    CreateChannels(RTCP_MUX, RTCP_MUX);
>     EXPECT_TRUE(SendInitiate());
>     EXPECT_TRUE(SendAccept());
>-    EXPECT_FALSE(channel1_->NeedsRtcpTransport());
>-    EXPECT_FALSE(channel2_->NeedsRtcpTransport());
>+    EXPECT_EQ(nullptr, channel1_->rtcp_packet_transport());
>+    EXPECT_EQ(nullptr, channel2_->rtcp_packet_transport());
>     SendRtp1();
>     SendRtp2();
>     WaitForThreads();
>@@ -1601,8 +1143,10 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
>     EXPECT_TRUE(CheckNoRtp2());
>   }
> 
>-  void SendBundleToBundle(
>-      const int* pl_types, int len, bool rtcp_mux, bool secure) {
>+  void SendBundleToBundle(const int* pl_types,
>+                          int len,
>+                          bool rtcp_mux,
>+                          bool secure) {
>     ASSERT_EQ(2, len);
>     int sequence_number1_1 = 0, sequence_number2_2 = 0;
>     // Only pl_type1 was added to the bundle filter for both |channel1_|
>@@ -1610,21 +1154,14 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
>     int pl_type1 = pl_types[0];
>     int pl_type2 = pl_types[1];
>     int flags = SSRC_MUX;
>-    if (secure) flags |= SECURE;
>+    if (secure)
>+      flags |= DTLS;
>     if (rtcp_mux) {
>       flags |= RTCP_MUX;
>     }
>     CreateChannels(flags, flags);
>     EXPECT_TRUE(SendInitiate());
>-    EXPECT_TRUE(channel1_->NeedsRtcpTransport());
>-    EXPECT_EQ(rtcp_mux, !channel2_->NeedsRtcpTransport());
>     EXPECT_TRUE(SendAccept());
>-    EXPECT_EQ(rtcp_mux, !channel1_->NeedsRtcpTransport());
>-    EXPECT_EQ(rtcp_mux, !channel2_->NeedsRtcpTransport());
>-    EXPECT_TRUE(channel1_->HandlesPayloadType(pl_type1));
>-    EXPECT_TRUE(channel2_->HandlesPayloadType(pl_type1));
>-    EXPECT_FALSE(channel1_->HandlesPayloadType(pl_type2));
>-    EXPECT_FALSE(channel2_->HandlesPayloadType(pl_type2));
> 
>     // Both channels can receive pl_type1 only.
>     SendCustomRtp1(kSsrc1, ++sequence_number1_1, pl_type1);
>@@ -1635,13 +1172,13 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
>     EXPECT_TRUE(CheckNoRtp1());
>     EXPECT_TRUE(CheckNoRtp2());
> 
>-    // RTCP test
>-    SendCustomRtp1(kSsrc1, ++sequence_number1_1, pl_type2);
>-    SendCustomRtp2(kSsrc2, ++sequence_number2_2, pl_type2);
>+    SendCustomRtp1(kSsrc3, ++sequence_number1_1, pl_type2);
>+    SendCustomRtp2(kSsrc4, ++sequence_number2_2, pl_type2);
>     WaitForThreads();
>-    EXPECT_FALSE(CheckCustomRtp2(kSsrc1, sequence_number1_1, pl_type2));
>-    EXPECT_FALSE(CheckCustomRtp1(kSsrc2, sequence_number2_2, pl_type2));
>+    EXPECT_FALSE(CheckCustomRtp2(kSsrc3, sequence_number1_1, pl_type2));
>+    EXPECT_FALSE(CheckCustomRtp1(kSsrc4, sequence_number2_2, pl_type2));
> 
>+    // RTCP test
>     SendCustomRtcp1(kSsrc1);
>     SendCustomRtcp2(kSsrc2);
>     WaitForThreads();
>@@ -1658,26 +1195,6 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
>     EXPECT_TRUE(CheckCustomRtcp2(kSsrc2));
>   }
> 
>-  // Test that the media monitor can be run and gives callbacks.
>-  void TestMediaMonitor() {
>-    CreateChannels(0, 0);
>-    EXPECT_TRUE(SendInitiate());
>-    EXPECT_TRUE(SendAccept());
>-    channel1_->StartMediaMonitor(100);
>-    channel2_->StartMediaMonitor(100);
>-    // Ensure we get callbacks and stop.
>-    EXPECT_TRUE_WAIT(media_info_callbacks1_ > 0, kDefaultTimeout);
>-    EXPECT_TRUE_WAIT(media_info_callbacks2_ > 0, kDefaultTimeout);
>-    channel1_->StopMediaMonitor();
>-    channel2_->StopMediaMonitor();
>-    // Ensure a restart of a stopped monitor works.
>-    channel1_->StartMediaMonitor(100);
>-    EXPECT_TRUE_WAIT(media_info_callbacks1_ > 0, kDefaultTimeout);
>-    channel1_->StopMediaMonitor();
>-    // Ensure stopping a stopped monitor is OK.
>-    channel1_->StopMediaMonitor();
>-  }
>-
>   void TestSetContentFailure() {
>     CreateChannels(0, 0);
> 
>@@ -1798,8 +1315,8 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
>     CreateChannels(0, 0);
>     EXPECT_TRUE(SendInitiate());
>     EXPECT_TRUE(SendAccept());
>-    EXPECT_TRUE(channel1_->NeedsRtcpTransport());
>-    EXPECT_TRUE(channel2_->NeedsRtcpTransport());
>+    EXPECT_NE(nullptr, channel1_->rtcp_packet_transport());
>+    EXPECT_NE(nullptr, channel2_->rtcp_packet_transport());
> 
>     // Send RTCP1 from a different thread.
>     ScopedCallThread send_rtcp([this] { SendRtcp1(); });
>@@ -1828,33 +1345,6 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
>     EXPECT_FALSE(media_channel1_->ready_to_send());
>   }
> 
>-  void TestOnTransportReadyToSendWithRtcpMux() {
>-    CreateChannels(0, 0);
>-    typename T::Content content;
>-    CreateContent(0, kPcmuCodec, kH264Codec, &content);
>-    // Both sides agree on mux. Should signal that RTCP mux is fully active.
>-    content.set_rtcp_mux(true);
>-    EXPECT_TRUE(channel1_->SetLocalContent(&content, SdpType::kOffer, NULL));
>-    EXPECT_EQ(0, rtcp_mux_activated_callbacks1_);
>-    EXPECT_TRUE(channel1_->SetRemoteContent(&content, SdpType::kAnswer, NULL));
>-    EXPECT_EQ(1, rtcp_mux_activated_callbacks1_);
>-    cricket::FakeDtlsTransport* rtp = fake_rtp_dtls_transport1_.get();
>-    EXPECT_FALSE(media_channel1_->ready_to_send());
>-    // In the case of rtcp mux, the SignalReadyToSend() from rtp channel
>-    // should trigger the MediaChannel's OnReadyToSend.
>-    network_thread_->Invoke<void>(RTC_FROM_HERE,
>-                                  [rtp] { rtp->SignalReadyToSend(rtp); });
>-    WaitForThreads();
>-    EXPECT_TRUE(media_channel1_->ready_to_send());
>-
>-    // TODO(zstein): Find a way to test this without making
>-    // OnTransportReadyToSend public.
>-    network_thread_->Invoke<void>(
>-        RTC_FROM_HERE, [this] { channel1_->OnTransportReadyToSend(false); });
>-    WaitForThreads();
>-    EXPECT_FALSE(media_channel1_->ready_to_send());
>-  }
>-
>   bool SetRemoteContentWithBitrateLimit(int remote_limit) {
>     typename T::Content content;
>     CreateContent(0, kPcmuCodec, kH264Codec, &content);
>@@ -1862,7 +1352,7 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
>     return channel1_->SetRemoteContent(&content, SdpType::kOffer, NULL);
>   }
> 
>-  webrtc::RtpParameters BitrateLimitedParameters(rtc::Optional<int> limit) {
>+  webrtc::RtpParameters BitrateLimitedParameters(absl::optional<int> limit) {
>     webrtc::RtpParameters parameters;
>     webrtc::RtpEncodingParameters encoding;
>     encoding.max_bitrate_bps = std::move(limit);
>@@ -1871,7 +1361,7 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
>   }
> 
>   void VerifyMaxBitrate(const webrtc::RtpParameters& parameters,
>-                        rtc::Optional<int> expected_bitrate) {
>+                        absl::optional<int> expected_bitrate) {
>     EXPECT_EQ(1UL, parameters.encodings.size());
>     EXPECT_EQ(expected_bitrate, parameters.encodings[0].max_bitrate_bps);
>   }
>@@ -1882,12 +1372,13 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
>                                            SdpType::kOffer, NULL));
>     EXPECT_EQ(media_channel1_->max_bps(), -1);
>     VerifyMaxBitrate(media_channel1_->GetRtpSendParameters(kSsrc1),
>-                     rtc::nullopt);
>+                     absl::nullopt);
>   }
> 
>-  // Test that when a channel gets new transports with a call to
>-  // |SetTransports|, the socket options from the old transports are merged with
>-  // the options on the new transport.
>+  // Test that when a channel gets new RtpTransport with a call to
>+  // |SetRtpTransport|, the socket options from the old RtpTransport is merged
>+  // with the options on the new one.
>+
>   // For example, audio and video may use separate socket options, but initially
>   // be unbundled, then later become bundled. When this happens, their preferred
>   // socket options should be merged to the underlying transport they share.
>@@ -1895,22 +1386,27 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
>     constexpr int kSndBufSize = 4000;
>     constexpr int kRcvBufSize = 8000;
> 
>-    CreateChannels(0, 0);
>+    CreateChannels(DTLS, DTLS);
> 
>     channel1_->SetOption(cricket::BaseChannel::ST_RTP,
>                          rtc::Socket::Option::OPT_SNDBUF, kSndBufSize);
>     channel2_->SetOption(cricket::BaseChannel::ST_RTP,
>                          rtc::Socket::Option::OPT_RCVBUF, kRcvBufSize);
> 
>-    channel1_->SetTransports(channel2_->rtp_dtls_transport(),
>-                             channel2_->rtcp_dtls_transport());
>+    new_rtp_transport_ = CreateDtlsSrtpTransport(
>+        static_cast<DtlsTransportInternal*>(channel2_->rtp_packet_transport()),
>+        static_cast<DtlsTransportInternal*>(
>+            channel2_->rtcp_packet_transport()));
>+    channel1_->SetRtpTransport(new_rtp_transport_.get());
> 
>     int option_val;
>-    ASSERT_TRUE(channel1_->rtp_dtls_transport()->GetOption(
>-        rtc::Socket::Option::OPT_SNDBUF, &option_val));
>+    ASSERT_TRUE(
>+        static_cast<DtlsTransportInternal*>(channel1_->rtp_packet_transport())
>+            ->GetOption(rtc::Socket::Option::OPT_SNDBUF, &option_val));
>     EXPECT_EQ(kSndBufSize, option_val);
>-    ASSERT_TRUE(channel1_->rtp_dtls_transport()->GetOption(
>-        rtc::Socket::Option::OPT_RCVBUF, &option_val));
>+    ASSERT_TRUE(
>+        static_cast<DtlsTransportInternal*>(channel1_->rtp_packet_transport())
>+            ->GetOption(rtc::Socket::Option::OPT_RCVBUF, &option_val));
>     EXPECT_EQ(kRcvBufSize, option_val);
>   }
> 
>@@ -1950,6 +1446,9 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
>   std::unique_ptr<rtc::FakePacketTransport> fake_rtcp_packet_transport1_;
>   std::unique_ptr<rtc::FakePacketTransport> fake_rtp_packet_transport2_;
>   std::unique_ptr<rtc::FakePacketTransport> fake_rtcp_packet_transport2_;
>+  std::unique_ptr<webrtc::RtpTransportInternal> rtp_transport1_;
>+  std::unique_ptr<webrtc::RtpTransportInternal> rtp_transport2_;
>+  std::unique_ptr<webrtc::RtpTransportInternal> new_rtp_transport_;
>   cricket::FakeMediaEngine media_engine_;
>   // The media channels are owned by the voice channel objects below.
>   typename T::MediaChannel* media_channel1_ = nullptr;
>@@ -1963,14 +1462,12 @@ class ChannelTest : public testing::Test, public sigslot::has_slots<> {
>   // The RTP and RTCP packets to send in the tests.
>   rtc::Buffer rtp_packet_;
>   rtc::Buffer rtcp_packet_;
>-  int media_info_callbacks1_ = 0;
>-  int media_info_callbacks2_ = 0;
>   int rtcp_mux_activated_callbacks1_ = 0;
>   int rtcp_mux_activated_callbacks2_ = 0;
>   cricket::CandidatePairInterface* last_selected_candidate_pair_;
> };
> 
>-template<>
>+template <>
> void ChannelTest<VoiceTraits>::CreateContent(
>     int flags,
>     const cricket::AudioCodec& audio_codec,
>@@ -1978,25 +1475,20 @@ void ChannelTest<VoiceTraits>::CreateContent(
>     cricket::AudioContentDescription* audio) {
>   audio->AddCodec(audio_codec);
>   audio->set_rtcp_mux((flags & RTCP_MUX) != 0);
>-  if ((flags & SECURE) && !(flags & DTLS)) {
>-    audio->AddCrypto(cricket::CryptoParams(
>-        1, rtc::CS_AES_CM_128_HMAC_SHA1_32,
>-        "inline:" + rtc::CreateRandomString(40), std::string()));
>-  }
> }
> 
>-template<>
>+template <>
> void ChannelTest<VoiceTraits>::CopyContent(
>     const cricket::AudioContentDescription& source,
>     cricket::AudioContentDescription* audio) {
>   *audio = source;
> }
> 
>-template<>
>+template <>
> bool ChannelTest<VoiceTraits>::CodecMatches(const cricket::AudioCodec& c1,
>                                             const cricket::AudioCodec& c2) {
>   return c1.name == c2.name && c1.clockrate == c2.clockrate &&
>-      c1.bitrate == c2.bitrate && c1.channels == c2.channels;
>+         c1.bitrate == c2.bitrate && c1.channels == c2.channels;
> }
> 
> template <>
>@@ -2022,21 +1514,23 @@ class VoiceChannelDoubleThreadTest : public ChannelTest<VoiceTraits> {
> };
> 
> class VoiceChannelWithEncryptedRtpHeaderExtensionsSingleThreadTest
>-  : public ChannelTest<VoiceTraits> {
>+    : public ChannelTest<VoiceTraits> {
>  public:
>   typedef ChannelTest<VoiceTraits> Base;
>   VoiceChannelWithEncryptedRtpHeaderExtensionsSingleThreadTest()
>-      : Base(true, kPcmuFrameWithExtensions, kRtcpReport,
>-            NetworkIsWorker::Yes) {}
>+      : Base(true,
>+             kPcmuFrameWithExtensions,
>+             kRtcpReport,
>+             NetworkIsWorker::Yes) {}
> };
> 
> class VoiceChannelWithEncryptedRtpHeaderExtensionsDoubleThreadTest
>-  : public ChannelTest<VoiceTraits> {
>+    : public ChannelTest<VoiceTraits> {
>  public:
>   typedef ChannelTest<VoiceTraits> Base;
>   VoiceChannelWithEncryptedRtpHeaderExtensionsDoubleThreadTest()
>-      : Base(true, kPcmuFrameWithExtensions, kRtcpReport,
>-            NetworkIsWorker::No) {}
>+      : Base(true, kPcmuFrameWithExtensions, kRtcpReport, NetworkIsWorker::No) {
>+  }
> };
> 
> // override to add NULL parameter
>@@ -2046,31 +1540,23 @@ std::unique_ptr<cricket::VideoChannel> ChannelTest<VideoTraits>::CreateChannel(
>     rtc::Thread* network_thread,
>     cricket::MediaEngineInterface* engine,
>     std::unique_ptr<cricket::FakeVideoMediaChannel> ch,
>-    cricket::DtlsTransportInternal* fake_rtp_dtls_transport,
>-    cricket::DtlsTransportInternal* fake_rtcp_dtls_transport,
>-    rtc::PacketTransportInternal* fake_rtp_packet_transport,
>-    rtc::PacketTransportInternal* fake_rtcp_packet_transport,
>+    webrtc::RtpTransportInternal* rtp_transport,
>     int flags) {
>   rtc::Thread* signaling_thread = rtc::Thread::Current();
>-  auto channel = rtc::MakeUnique<cricket::VideoChannel>(
>+  auto channel = absl::make_unique<cricket::VideoChannel>(
>       worker_thread, network_thread, signaling_thread, std::move(ch),
>-      cricket::CN_VIDEO, (flags & RTCP_MUX_REQUIRED) != 0,
>-      (flags & SECURE) != 0);
>-  if (!channel->NeedsRtcpTransport()) {
>-    fake_rtcp_dtls_transport = nullptr;
>-  }
>-  channel->Init_w(fake_rtp_dtls_transport, fake_rtcp_dtls_transport,
>-                  fake_rtp_packet_transport, fake_rtcp_packet_transport);
>+      cricket::CN_VIDEO, (flags & DTLS) != 0, rtc::CryptoOptions());
>+  channel->Init_w(rtp_transport);
>   return channel;
> }
> 
> // override to add 0 parameter
>-template<>
>+template <>
> bool ChannelTest<VideoTraits>::AddStream1(int id) {
>   return channel1_->AddRecvStream(cricket::StreamParams::CreateLegacy(id));
> }
> 
>-template<>
>+template <>
> void ChannelTest<VideoTraits>::CreateContent(
>     int flags,
>     const cricket::AudioCodec& audio_codec,
>@@ -2078,21 +1564,16 @@ void ChannelTest<VideoTraits>::CreateContent(
>     cricket::VideoContentDescription* video) {
>   video->AddCodec(video_codec);
>   video->set_rtcp_mux((flags & RTCP_MUX) != 0);
>-  if (flags & SECURE) {
>-    video->AddCrypto(cricket::CryptoParams(
>-        1, rtc::CS_AES_CM_128_HMAC_SHA1_80,
>-        "inline:" + rtc::CreateRandomString(40), std::string()));
>-  }
> }
> 
>-template<>
>+template <>
> void ChannelTest<VideoTraits>::CopyContent(
>     const cricket::VideoContentDescription& source,
>     cricket::VideoContentDescription* video) {
>   *video = source;
> }
> 
>-template<>
>+template <>
> bool ChannelTest<VideoTraits>::CodecMatches(const cricket::VideoCodec& c1,
>                                             const cricket::VideoCodec& c2) {
>   return c1.name == c2.name;
>@@ -2154,74 +1635,10 @@ TEST_F(VoiceChannelSingleThreadTest, TestChangeStreamParamsInContent) {
>   Base::TestChangeStreamParamsInContent();
> }
> 
>-TEST_F(VoiceChannelWithEncryptedRtpHeaderExtensionsSingleThreadTest,
>-    TestChangeEncryptedHeaderExtensionsDtls) {
>-  int flags = DTLS;
>-  Base::TestChangeEncryptedHeaderExtensions(flags);
>-}
>-
>-TEST_F(VoiceChannelWithEncryptedRtpHeaderExtensionsSingleThreadTest,
>-    TestChangeEncryptedHeaderExtensionsDtlsScenario1) {
>-  int flags = DTLS;
>-  Base::TestChangeEncryptedHeaderExtensions(flags, DTLS_BEFORE_OFFER_ANSWER);
>-}
>-
>-TEST_F(VoiceChannelWithEncryptedRtpHeaderExtensionsSingleThreadTest,
>-    TestChangeEncryptedHeaderExtensionsDtlsScenario2) {
>-  int flags = DTLS;
>-  Base::TestChangeEncryptedHeaderExtensions(flags, DTLS_AFTER_CHANNEL2_READY);
>-}
>-
>-TEST_F(VoiceChannelWithEncryptedRtpHeaderExtensionsSingleThreadTest,
>-    TestChangeEncryptedHeaderExtensionsDtlsGcm) {
>-  int flags = DTLS | GCM_CIPHER;
>-  Base::TestChangeEncryptedHeaderExtensions(flags);
>-}
>-
>-TEST_F(VoiceChannelWithEncryptedRtpHeaderExtensionsSingleThreadTest,
>-    TestChangeEncryptedHeaderExtensionsDtlsGcmScenario1) {
>-  int flags = DTLS | GCM_CIPHER;
>-  Base::TestChangeEncryptedHeaderExtensions(flags, DTLS_BEFORE_OFFER_ANSWER);
>-}
>-
>-TEST_F(VoiceChannelWithEncryptedRtpHeaderExtensionsSingleThreadTest,
>-    TestChangeEncryptedHeaderExtensionsDtlsGcmScenario2) {
>-  int flags = DTLS | GCM_CIPHER;
>-  Base::TestChangeEncryptedHeaderExtensions(flags, DTLS_AFTER_CHANNEL2_READY);
>-}
>-
>-TEST_F(VoiceChannelWithEncryptedRtpHeaderExtensionsSingleThreadTest,
>-    TestChangeEncryptedHeaderExtensionsSDES) {
>-  int flags = 0;
>-  Base::TestChangeEncryptedHeaderExtensions(flags);
>-}
>-
> TEST_F(VoiceChannelSingleThreadTest, TestPlayoutAndSendingStates) {
>   Base::TestPlayoutAndSendingStates();
> }
> 
>-TEST_F(VoiceChannelSingleThreadTest, TestMuteStream) {
>-  CreateChannels(0, 0);
>-  // Test that we can Mute the default channel even though the sending SSRC
>-  // is unknown.
>-  EXPECT_FALSE(media_channel1_->IsStreamMuted(0));
>-  EXPECT_TRUE(channel1_->SetAudioSend(0, false, nullptr, nullptr));
>-  EXPECT_TRUE(media_channel1_->IsStreamMuted(0));
>-  EXPECT_TRUE(channel1_->SetAudioSend(0, true, nullptr, nullptr));
>-  EXPECT_FALSE(media_channel1_->IsStreamMuted(0));
>-
>-  // Test that we can not mute an unknown SSRC.
>-  EXPECT_FALSE(channel1_->SetAudioSend(kSsrc1, false, nullptr, nullptr));
>-
>-  SendInitiate();
>-  // After the local session description has been set, we can mute a stream
>-  // with its SSRC.
>-  EXPECT_TRUE(channel1_->SetAudioSend(kSsrc1, false, nullptr, nullptr));
>-  EXPECT_TRUE(media_channel1_->IsStreamMuted(kSsrc1));
>-  EXPECT_TRUE(channel1_->SetAudioSend(kSsrc1, true, nullptr, nullptr));
>-  EXPECT_FALSE(media_channel1_->IsStreamMuted(kSsrc1));
>-}
>-
> TEST_F(VoiceChannelSingleThreadTest, TestMediaContentDirection) {
>   Base::TestMediaContentDirection();
> }
>@@ -2246,66 +1663,12 @@ TEST_F(VoiceChannelSingleThreadTest, SendRtcpToRtcp) {
>   Base::SendRtcpToRtcp();
> }
> 
>-TEST_F(VoiceChannelSingleThreadTest, SendRtcpMuxToRtcp) {
>-  Base::SendRtcpMuxToRtcp();
>-}
>-
>-TEST_F(VoiceChannelSingleThreadTest, SendRtcpMuxToRtcpMux) {
>-  Base::SendRtcpMuxToRtcpMux();
>-}
>-
>-TEST_F(VoiceChannelSingleThreadTest, SendRequireRtcpMuxToRtcpMux) {
>-  Base::SendRequireRtcpMuxToRtcpMux();
>-}
>-
>-TEST_F(VoiceChannelSingleThreadTest, SendRtcpMuxToRequireRtcpMux) {
>-  Base::SendRtcpMuxToRequireRtcpMux();
>-}
>-
>-TEST_F(VoiceChannelSingleThreadTest, SendRequireRtcpMuxToRequireRtcpMux) {
>-  Base::SendRequireRtcpMuxToRequireRtcpMux();
>-}
>-
>-TEST_F(VoiceChannelSingleThreadTest, SendRequireRtcpMuxToNoRtcpMux) {
>-  Base::SendRequireRtcpMuxToNoRtcpMux();
>-}
>-
>-TEST_F(VoiceChannelSingleThreadTest, SendEarlyRtcpMuxToRtcp) {
>-  Base::SendEarlyRtcpMuxToRtcp();
>-}
>-
>-TEST_F(VoiceChannelSingleThreadTest, SendEarlyRtcpMuxToRtcpMux) {
>-  Base::SendEarlyRtcpMuxToRtcpMux();
>-}
>-
>-TEST_F(VoiceChannelSingleThreadTest, SendSrtpToSrtpRtcpMux) {
>-  Base::SendSrtpToSrtp(RTCP_MUX, RTCP_MUX);
>-}
>-
>-TEST_F(VoiceChannelSingleThreadTest, SendSrtpToRtp) {
>-  Base::SendSrtpToSrtp();
>-}
>-
>-TEST_F(VoiceChannelSingleThreadTest, SendSrtcpMux) {
>-  Base::SendSrtpToSrtp(RTCP_MUX, RTCP_MUX);
>-}
>-
>-TEST_F(VoiceChannelSingleThreadTest, SendDtlsSrtpToSrtp) {
>-  Base::SendDtlsToSdesNotSupported();
>-}
>-
> TEST_F(VoiceChannelSingleThreadTest, SendDtlsSrtpToDtlsSrtp) {
>-  Base::SendSrtpToSrtp(DTLS, DTLS);
>+  Base::SendDtlsSrtpToDtlsSrtp(0, 0);
> }
> 
> TEST_F(VoiceChannelSingleThreadTest, SendDtlsSrtpToDtlsSrtpRtcpMux) {
>-  Base::SendSrtpToSrtp(DTLS | RTCP_MUX, DTLS | RTCP_MUX);
>-}
>-
>-// Test using the channel with a raw packet interface, as opposed to a DTLS
>-// transport interface.
>-TEST_F(VoiceChannelSingleThreadTest, SendSrtpToSrtpWithRawPacketTransport) {
>-  Base::SendSrtpToSrtp(RAW_PACKET_TRANSPORT, RAW_PACKET_TRANSPORT);
>+  Base::SendDtlsSrtpToDtlsSrtp(RTCP_MUX, RTCP_MUX);
> }
> 
> TEST_F(VoiceChannelSingleThreadTest, SendEarlyMediaUsingRtcpMuxSrtp) {
>@@ -2316,18 +1679,10 @@ TEST_F(VoiceChannelSingleThreadTest, SendRtpToRtpOnThread) {
>   Base::SendRtpToRtpOnThread();
> }
> 
>-TEST_F(VoiceChannelSingleThreadTest, SendSrtpToSrtpOnThread) {
>-  Base::SendSrtpToSrtpOnThread();
>-}
>-
> TEST_F(VoiceChannelSingleThreadTest, SendWithWritabilityLoss) {
>   Base::SendWithWritabilityLoss();
> }
> 
>-TEST_F(VoiceChannelSingleThreadTest, TestMediaMonitor) {
>-  Base::TestMediaMonitor();
>-}
>-
> TEST_F(VoiceChannelSingleThreadTest, TestSetContentFailure) {
>   Base::TestSetContentFailure();
> }
>@@ -2356,10 +1711,6 @@ TEST_F(VoiceChannelSingleThreadTest, TestOnTransportReadyToSend) {
>   Base::TestOnTransportReadyToSend();
> }
> 
>-TEST_F(VoiceChannelSingleThreadTest, TestOnTransportReadyToSendWithRtcpMux) {
>-  Base::TestOnTransportReadyToSendWithRtcpMux();
>-}
>-
> TEST_F(VoiceChannelSingleThreadTest, SendBundleToBundle) {
>   Base::SendBundleToBundle(kAudioPts, arraysize(kAudioPts), false, false);
> }
>@@ -2419,74 +1770,10 @@ TEST_F(VoiceChannelDoubleThreadTest, TestChangeStreamParamsInContent) {
>   Base::TestChangeStreamParamsInContent();
> }
> 
>-TEST_F(VoiceChannelWithEncryptedRtpHeaderExtensionsDoubleThreadTest,
>-    TestChangeEncryptedHeaderExtensionsDtls) {
>-  int flags = DTLS;
>-  Base::TestChangeEncryptedHeaderExtensions(flags);
>-}
>-
>-TEST_F(VoiceChannelWithEncryptedRtpHeaderExtensionsDoubleThreadTest,
>-    TestChangeEncryptedHeaderExtensionsDtlsScenario1) {
>-  int flags = DTLS;
>-  Base::TestChangeEncryptedHeaderExtensions(flags, DTLS_BEFORE_OFFER_ANSWER);
>-}
>-
>-TEST_F(VoiceChannelWithEncryptedRtpHeaderExtensionsDoubleThreadTest,
>-    TestChangeEncryptedHeaderExtensionsDtlsScenario2) {
>-  int flags = DTLS;
>-  Base::TestChangeEncryptedHeaderExtensions(flags, DTLS_AFTER_CHANNEL2_READY);
>-}
>-
>-TEST_F(VoiceChannelWithEncryptedRtpHeaderExtensionsDoubleThreadTest,
>-    TestChangeEncryptedHeaderExtensionsDtlsGcm) {
>-  int flags = DTLS | GCM_CIPHER;
>-  Base::TestChangeEncryptedHeaderExtensions(flags);
>-}
>-
>-TEST_F(VoiceChannelWithEncryptedRtpHeaderExtensionsDoubleThreadTest,
>-    TestChangeEncryptedHeaderExtensionsDtlsGcmScenario1) {
>-  int flags = DTLS | GCM_CIPHER;
>-  Base::TestChangeEncryptedHeaderExtensions(flags, DTLS_BEFORE_OFFER_ANSWER);
>-}
>-
>-TEST_F(VoiceChannelWithEncryptedRtpHeaderExtensionsDoubleThreadTest,
>-    TestChangeEncryptedHeaderExtensionsDtlsGcmScenario2) {
>-  int flags = DTLS | GCM_CIPHER;
>-  Base::TestChangeEncryptedHeaderExtensions(flags, DTLS_AFTER_CHANNEL2_READY);
>-}
>-
>-TEST_F(VoiceChannelWithEncryptedRtpHeaderExtensionsDoubleThreadTest,
>-    TestChangeEncryptedHeaderExtensionsSDES) {
>-  int flags = 0;
>-  Base::TestChangeEncryptedHeaderExtensions(flags);
>-}
>-
> TEST_F(VoiceChannelDoubleThreadTest, TestPlayoutAndSendingStates) {
>   Base::TestPlayoutAndSendingStates();
> }
> 
>-TEST_F(VoiceChannelDoubleThreadTest, TestMuteStream) {
>-  CreateChannels(0, 0);
>-  // Test that we can Mute the default channel even though the sending SSRC
>-  // is unknown.
>-  EXPECT_FALSE(media_channel1_->IsStreamMuted(0));
>-  EXPECT_TRUE(channel1_->SetAudioSend(0, false, nullptr, nullptr));
>-  EXPECT_TRUE(media_channel1_->IsStreamMuted(0));
>-  EXPECT_TRUE(channel1_->SetAudioSend(0, true, nullptr, nullptr));
>-  EXPECT_FALSE(media_channel1_->IsStreamMuted(0));
>-
>-  // Test that we can not mute an unknown SSRC.
>-  EXPECT_FALSE(channel1_->SetAudioSend(kSsrc1, false, nullptr, nullptr));
>-
>-  SendInitiate();
>-  // After the local session description has been set, we can mute a stream
>-  // with its SSRC.
>-  EXPECT_TRUE(channel1_->SetAudioSend(kSsrc1, false, nullptr, nullptr));
>-  EXPECT_TRUE(media_channel1_->IsStreamMuted(kSsrc1));
>-  EXPECT_TRUE(channel1_->SetAudioSend(kSsrc1, true, nullptr, nullptr));
>-  EXPECT_FALSE(media_channel1_->IsStreamMuted(kSsrc1));
>-}
>-
> TEST_F(VoiceChannelDoubleThreadTest, TestMediaContentDirection) {
>   Base::TestMediaContentDirection();
> }
>@@ -2511,66 +1798,12 @@ TEST_F(VoiceChannelDoubleThreadTest, SendRtcpToRtcp) {
>   Base::SendRtcpToRtcp();
> }
> 
>-TEST_F(VoiceChannelDoubleThreadTest, SendRtcpMuxToRtcp) {
>-  Base::SendRtcpMuxToRtcp();
>-}
>-
>-TEST_F(VoiceChannelDoubleThreadTest, SendRtcpMuxToRtcpMux) {
>-  Base::SendRtcpMuxToRtcpMux();
>-}
>-
>-TEST_F(VoiceChannelDoubleThreadTest, SendRequireRtcpMuxToRtcpMux) {
>-  Base::SendRequireRtcpMuxToRtcpMux();
>-}
>-
>-TEST_F(VoiceChannelDoubleThreadTest, SendRtcpMuxToRequireRtcpMux) {
>-  Base::SendRtcpMuxToRequireRtcpMux();
>-}
>-
>-TEST_F(VoiceChannelDoubleThreadTest, SendRequireRtcpMuxToRequireRtcpMux) {
>-  Base::SendRequireRtcpMuxToRequireRtcpMux();
>-}
>-
>-TEST_F(VoiceChannelDoubleThreadTest, SendRequireRtcpMuxToNoRtcpMux) {
>-  Base::SendRequireRtcpMuxToNoRtcpMux();
>-}
>-
>-TEST_F(VoiceChannelDoubleThreadTest, SendEarlyRtcpMuxToRtcp) {
>-  Base::SendEarlyRtcpMuxToRtcp();
>-}
>-
>-TEST_F(VoiceChannelDoubleThreadTest, SendEarlyRtcpMuxToRtcpMux) {
>-  Base::SendEarlyRtcpMuxToRtcpMux();
>-}
>-
>-TEST_F(VoiceChannelDoubleThreadTest, SendSrtpToSrtpRtcpMux) {
>-  Base::SendSrtpToSrtp(RTCP_MUX, RTCP_MUX);
>-}
>-
>-TEST_F(VoiceChannelDoubleThreadTest, SendSrtpToRtp) {
>-  Base::SendSrtpToSrtp();
>-}
>-
>-TEST_F(VoiceChannelDoubleThreadTest, SendSrtcpMux) {
>-  Base::SendSrtpToSrtp(RTCP_MUX, RTCP_MUX);
>-}
>-
>-TEST_F(VoiceChannelDoubleThreadTest, SendDtlsSrtpToSrtp) {
>-  Base::SendDtlsToSdesNotSupported();
>-}
>-
> TEST_F(VoiceChannelDoubleThreadTest, SendDtlsSrtpToDtlsSrtp) {
>-  Base::SendSrtpToSrtp(DTLS, DTLS);
>+  Base::SendDtlsSrtpToDtlsSrtp(0, 0);
> }
> 
> TEST_F(VoiceChannelDoubleThreadTest, SendDtlsSrtpToDtlsSrtpRtcpMux) {
>-  Base::SendSrtpToSrtp(DTLS | RTCP_MUX, DTLS | RTCP_MUX);
>-}
>-
>-// Test using the channel with a raw packet interface, as opposed to a DTLS
>-// transport interface.
>-TEST_F(VoiceChannelDoubleThreadTest, SendSrtpToSrtpWithRawPacketTransport) {
>-  Base::SendSrtpToSrtp(RAW_PACKET_TRANSPORT, RAW_PACKET_TRANSPORT);
>+  Base::SendDtlsSrtpToDtlsSrtp(RTCP_MUX, RTCP_MUX);
> }
> 
> TEST_F(VoiceChannelDoubleThreadTest, SendEarlyMediaUsingRtcpMuxSrtp) {
>@@ -2581,18 +1814,10 @@ TEST_F(VoiceChannelDoubleThreadTest, SendRtpToRtpOnThread) {
>   Base::SendRtpToRtpOnThread();
> }
> 
>-TEST_F(VoiceChannelDoubleThreadTest, SendSrtpToSrtpOnThread) {
>-  Base::SendSrtpToSrtpOnThread();
>-}
>-
> TEST_F(VoiceChannelDoubleThreadTest, SendWithWritabilityLoss) {
>   Base::SendWithWritabilityLoss();
> }
> 
>-TEST_F(VoiceChannelDoubleThreadTest, TestMediaMonitor) {
>-  Base::TestMediaMonitor();
>-}
>-
> TEST_F(VoiceChannelDoubleThreadTest, TestSetContentFailure) {
>   Base::TestSetContentFailure();
> }
>@@ -2621,10 +1846,6 @@ TEST_F(VoiceChannelDoubleThreadTest, TestOnTransportReadyToSend) {
>   Base::TestOnTransportReadyToSend();
> }
> 
>-TEST_F(VoiceChannelDoubleThreadTest, TestOnTransportReadyToSendWithRtcpMux) {
>-  Base::TestOnTransportReadyToSendWithRtcpMux();
>-}
>-
> TEST_F(VoiceChannelDoubleThreadTest, SendBundleToBundle) {
>   Base::SendBundleToBundle(kAudioPts, arraysize(kAudioPts), false, false);
> }
>@@ -2710,66 +1931,12 @@ TEST_F(VideoChannelSingleThreadTest, SendRtcpToRtcp) {
>   Base::SendRtcpToRtcp();
> }
> 
>-TEST_F(VideoChannelSingleThreadTest, SendRtcpMuxToRtcp) {
>-  Base::SendRtcpMuxToRtcp();
>-}
>-
>-TEST_F(VideoChannelSingleThreadTest, SendRtcpMuxToRtcpMux) {
>-  Base::SendRtcpMuxToRtcpMux();
>-}
>-
>-TEST_F(VideoChannelSingleThreadTest, SendRequireRtcpMuxToRtcpMux) {
>-  Base::SendRequireRtcpMuxToRtcpMux();
>-}
>-
>-TEST_F(VideoChannelSingleThreadTest, SendRtcpMuxToRequireRtcpMux) {
>-  Base::SendRtcpMuxToRequireRtcpMux();
>-}
>-
>-TEST_F(VideoChannelSingleThreadTest, SendRequireRtcpMuxToRequireRtcpMux) {
>-  Base::SendRequireRtcpMuxToRequireRtcpMux();
>-}
>-
>-TEST_F(VideoChannelSingleThreadTest, SendRequireRtcpMuxToNoRtcpMux) {
>-  Base::SendRequireRtcpMuxToNoRtcpMux();
>-}
>-
>-TEST_F(VideoChannelSingleThreadTest, SendEarlyRtcpMuxToRtcp) {
>-  Base::SendEarlyRtcpMuxToRtcp();
>-}
>-
>-TEST_F(VideoChannelSingleThreadTest, SendEarlyRtcpMuxToRtcpMux) {
>-  Base::SendEarlyRtcpMuxToRtcpMux();
>-}
>-
>-TEST_F(VideoChannelSingleThreadTest, SendSrtpToSrtp) {
>-  Base::SendSrtpToSrtp();
>-}
>-
>-TEST_F(VideoChannelSingleThreadTest, SendSrtpToRtp) {
>-  Base::SendSrtpToSrtp();
>-}
>-
>-TEST_F(VideoChannelSingleThreadTest, SendDtlsSrtpToSrtp) {
>-  Base::SendDtlsToSdesNotSupported();
>-}
>-
> TEST_F(VideoChannelSingleThreadTest, SendDtlsSrtpToDtlsSrtp) {
>-  Base::SendSrtpToSrtp(DTLS, DTLS);
>+  Base::SendDtlsSrtpToDtlsSrtp(0, 0);
> }
> 
> TEST_F(VideoChannelSingleThreadTest, SendDtlsSrtpToDtlsSrtpRtcpMux) {
>-  Base::SendSrtpToSrtp(DTLS | RTCP_MUX, DTLS | RTCP_MUX);
>-}
>-
>-// Test using the channel with a raw packet interface, as opposed to a DTLS
>-// transport interface.
>-TEST_F(VideoChannelSingleThreadTest, SendSrtpToSrtpWithRawPacketTransport) {
>-  Base::SendSrtpToSrtp(RAW_PACKET_TRANSPORT, RAW_PACKET_TRANSPORT);
>-}
>-
>-TEST_F(VideoChannelSingleThreadTest, SendSrtcpMux) {
>-  Base::SendSrtpToSrtp(RTCP_MUX, RTCP_MUX);
>+  Base::SendDtlsSrtpToDtlsSrtp(RTCP_MUX, RTCP_MUX);
> }
> 
> TEST_F(VideoChannelSingleThreadTest, SendEarlyMediaUsingRtcpMuxSrtp) {
>@@ -2780,18 +1947,10 @@ TEST_F(VideoChannelSingleThreadTest, SendRtpToRtpOnThread) {
>   Base::SendRtpToRtpOnThread();
> }
> 
>-TEST_F(VideoChannelSingleThreadTest, SendSrtpToSrtpOnThread) {
>-  Base::SendSrtpToSrtpOnThread();
>-}
>-
> TEST_F(VideoChannelSingleThreadTest, SendWithWritabilityLoss) {
>   Base::SendWithWritabilityLoss();
> }
> 
>-TEST_F(VideoChannelSingleThreadTest, TestMediaMonitor) {
>-  Base::TestMediaMonitor();
>-}
>-
> TEST_F(VideoChannelSingleThreadTest, TestSetContentFailure) {
>   Base::TestSetContentFailure();
> }
>@@ -2836,10 +1995,6 @@ TEST_F(VideoChannelSingleThreadTest, TestOnTransportReadyToSend) {
>   Base::TestOnTransportReadyToSend();
> }
> 
>-TEST_F(VideoChannelSingleThreadTest, TestOnTransportReadyToSendWithRtcpMux) {
>-  Base::TestOnTransportReadyToSendWithRtcpMux();
>-}
>-
> TEST_F(VideoChannelSingleThreadTest, DefaultMaxBitrateIsUnlimited) {
>   Base::DefaultMaxBitrateIsUnlimited();
> }
>@@ -2909,66 +2064,12 @@ TEST_F(VideoChannelDoubleThreadTest, SendRtcpToRtcp) {
>   Base::SendRtcpToRtcp();
> }
> 
>-TEST_F(VideoChannelDoubleThreadTest, SendRtcpMuxToRtcp) {
>-  Base::SendRtcpMuxToRtcp();
>-}
>-
>-TEST_F(VideoChannelDoubleThreadTest, SendRtcpMuxToRtcpMux) {
>-  Base::SendRtcpMuxToRtcpMux();
>-}
>-
>-TEST_F(VideoChannelDoubleThreadTest, SendRequireRtcpMuxToRtcpMux) {
>-  Base::SendRequireRtcpMuxToRtcpMux();
>-}
>-
>-TEST_F(VideoChannelDoubleThreadTest, SendRtcpMuxToRequireRtcpMux) {
>-  Base::SendRtcpMuxToRequireRtcpMux();
>-}
>-
>-TEST_F(VideoChannelDoubleThreadTest, SendRequireRtcpMuxToRequireRtcpMux) {
>-  Base::SendRequireRtcpMuxToRequireRtcpMux();
>-}
>-
>-TEST_F(VideoChannelDoubleThreadTest, SendRequireRtcpMuxToNoRtcpMux) {
>-  Base::SendRequireRtcpMuxToNoRtcpMux();
>-}
>-
>-TEST_F(VideoChannelDoubleThreadTest, SendEarlyRtcpMuxToRtcp) {
>-  Base::SendEarlyRtcpMuxToRtcp();
>-}
>-
>-TEST_F(VideoChannelDoubleThreadTest, SendEarlyRtcpMuxToRtcpMux) {
>-  Base::SendEarlyRtcpMuxToRtcpMux();
>-}
>-
>-TEST_F(VideoChannelDoubleThreadTest, SendSrtpToSrtp) {
>-  Base::SendSrtpToSrtp();
>-}
>-
>-TEST_F(VideoChannelDoubleThreadTest, SendSrtpToRtp) {
>-  Base::SendSrtpToSrtp();
>-}
>-
>-TEST_F(VideoChannelDoubleThreadTest, SendDtlsSrtpToSrtp) {
>-  Base::SendDtlsToSdesNotSupported();
>-}
>-
> TEST_F(VideoChannelDoubleThreadTest, SendDtlsSrtpToDtlsSrtp) {
>-  Base::SendSrtpToSrtp(DTLS, DTLS);
>+  Base::SendDtlsSrtpToDtlsSrtp(0, 0);
> }
> 
> TEST_F(VideoChannelDoubleThreadTest, SendDtlsSrtpToDtlsSrtpRtcpMux) {
>-  Base::SendSrtpToSrtp(DTLS | RTCP_MUX, DTLS | RTCP_MUX);
>-}
>-
>-// Test using the channel with a raw packet interface, as opposed to a DTLS
>-// transport interface.
>-TEST_F(VideoChannelDoubleThreadTest, SendSrtpToSrtpWithRawPacketTransport) {
>-  Base::SendSrtpToSrtp(RAW_PACKET_TRANSPORT, RAW_PACKET_TRANSPORT);
>-}
>-
>-TEST_F(VideoChannelDoubleThreadTest, SendSrtcpMux) {
>-  Base::SendSrtpToSrtp(RTCP_MUX, RTCP_MUX);
>+  Base::SendDtlsSrtpToDtlsSrtp(RTCP_MUX, RTCP_MUX);
> }
> 
> TEST_F(VideoChannelDoubleThreadTest, SendEarlyMediaUsingRtcpMuxSrtp) {
>@@ -2979,18 +2080,10 @@ TEST_F(VideoChannelDoubleThreadTest, SendRtpToRtpOnThread) {
>   Base::SendRtpToRtpOnThread();
> }
> 
>-TEST_F(VideoChannelDoubleThreadTest, SendSrtpToSrtpOnThread) {
>-  Base::SendSrtpToSrtpOnThread();
>-}
>-
> TEST_F(VideoChannelDoubleThreadTest, SendWithWritabilityLoss) {
>   Base::SendWithWritabilityLoss();
> }
> 
>-TEST_F(VideoChannelDoubleThreadTest, TestMediaMonitor) {
>-  Base::TestMediaMonitor();
>-}
>-
> TEST_F(VideoChannelDoubleThreadTest, TestSetContentFailure) {
>   Base::TestSetContentFailure();
> }
>@@ -3035,10 +2128,6 @@ TEST_F(VideoChannelDoubleThreadTest, TestOnTransportReadyToSend) {
>   Base::TestOnTransportReadyToSend();
> }
> 
>-TEST_F(VideoChannelDoubleThreadTest, TestOnTransportReadyToSendWithRtcpMux) {
>-  Base::TestOnTransportReadyToSendWithRtcpMux();
>-}
>-
> TEST_F(VideoChannelDoubleThreadTest, DefaultMaxBitrateIsUnlimited) {
>   Base::DefaultMaxBitrateIsUnlimited();
> }
>@@ -3070,21 +2159,13 @@ std::unique_ptr<cricket::RtpDataChannel> ChannelTest<DataTraits>::CreateChannel(
>     rtc::Thread* network_thread,
>     cricket::MediaEngineInterface* engine,
>     std::unique_ptr<cricket::FakeDataMediaChannel> ch,
>-    cricket::DtlsTransportInternal* fake_rtp_dtls_transport,
>-    cricket::DtlsTransportInternal* fake_rtcp_dtls_transport,
>-    rtc::PacketTransportInternal* fake_rtp_packet_transport,
>-    rtc::PacketTransportInternal* fake_rtcp_packet_transport,
>+    webrtc::RtpTransportInternal* rtp_transport,
>     int flags) {
>   rtc::Thread* signaling_thread = rtc::Thread::Current();
>-  auto channel = rtc::MakeUnique<cricket::RtpDataChannel>(
>+  auto channel = absl::make_unique<cricket::RtpDataChannel>(
>       worker_thread, network_thread, signaling_thread, std::move(ch),
>-      cricket::CN_DATA, (flags & RTCP_MUX_REQUIRED) != 0,
>-      (flags & SECURE) != 0);
>-  if (!channel->NeedsRtcpTransport()) {
>-    fake_rtcp_dtls_transport = nullptr;
>-  }
>-  channel->Init_w(fake_rtp_dtls_transport, fake_rtcp_dtls_transport,
>-                  fake_rtp_packet_transport, fake_rtcp_packet_transport);
>+      cricket::CN_DATA, (flags & DTLS) != 0, rtc::CryptoOptions());
>+  channel->Init_w(rtp_transport);
>   return channel;
> }
> 
>@@ -3096,11 +2177,6 @@ void ChannelTest<DataTraits>::CreateContent(
>     cricket::DataContentDescription* data) {
>   data->AddCodec(kGoogleDataCodec);
>   data->set_rtcp_mux((flags & RTCP_MUX) != 0);
>-  if (flags & SECURE) {
>-    data->AddCrypto(cricket::CryptoParams(
>-        1, rtc::CS_AES_CM_128_HMAC_SHA1_32,
>-        "inline:" + rtc::CreateRandomString(40), std::string()));
>-  }
> }
> 
> template <>
>@@ -3173,10 +2249,6 @@ TEST_F(RtpDataChannelSingleThreadTest, TestOnTransportReadyToSend) {
>   Base::TestOnTransportReadyToSend();
> }
> 
>-TEST_F(RtpDataChannelSingleThreadTest, TestOnTransportReadyToSendWithRtcpMux) {
>-  Base::TestOnTransportReadyToSendWithRtcpMux();
>-}
>-
> TEST_F(RtpDataChannelSingleThreadTest, SendRtpToRtp) {
>   Base::SendRtpToRtp();
> }
>@@ -3185,50 +2257,14 @@ TEST_F(RtpDataChannelSingleThreadTest, SendRtcpToRtcp) {
>   Base::SendRtcpToRtcp();
> }
> 
>-TEST_F(RtpDataChannelSingleThreadTest, SendRtcpMuxToRtcp) {
>-  Base::SendRtcpMuxToRtcp();
>-}
>-
>-TEST_F(RtpDataChannelSingleThreadTest, SendRtcpMuxToRtcpMux) {
>-  Base::SendRtcpMuxToRtcpMux();
>-}
>-
>-TEST_F(RtpDataChannelSingleThreadTest, SendEarlyRtcpMuxToRtcp) {
>-  Base::SendEarlyRtcpMuxToRtcp();
>-}
>-
>-TEST_F(RtpDataChannelSingleThreadTest, SendEarlyRtcpMuxToRtcpMux) {
>-  Base::SendEarlyRtcpMuxToRtcpMux();
>-}
>-
>-TEST_F(RtpDataChannelSingleThreadTest, SendSrtpToSrtp) {
>-  Base::SendSrtpToSrtp();
>-}
>-
>-TEST_F(RtpDataChannelSingleThreadTest, SendSrtpToRtp) {
>-  Base::SendSrtpToSrtp();
>-}
>-
>-TEST_F(RtpDataChannelSingleThreadTest, SendSrtcpMux) {
>-  Base::SendSrtpToSrtp(RTCP_MUX, RTCP_MUX);
>-}
>-
> TEST_F(RtpDataChannelSingleThreadTest, SendRtpToRtpOnThread) {
>   Base::SendRtpToRtpOnThread();
> }
> 
>-TEST_F(RtpDataChannelSingleThreadTest, SendSrtpToSrtpOnThread) {
>-  Base::SendSrtpToSrtpOnThread();
>-}
>-
> TEST_F(RtpDataChannelSingleThreadTest, SendWithWritabilityLoss) {
>   Base::SendWithWritabilityLoss();
> }
> 
>-TEST_F(RtpDataChannelSingleThreadTest, TestMediaMonitor) {
>-  Base::TestMediaMonitor();
>-}
>-
> TEST_F(RtpDataChannelSingleThreadTest, SocketOptionsMergedOnSetTransport) {
>   Base::SocketOptionsMergedOnSetTransport();
> }
>@@ -3297,10 +2333,6 @@ TEST_F(RtpDataChannelDoubleThreadTest, TestOnTransportReadyToSend) {
>   Base::TestOnTransportReadyToSend();
> }
> 
>-TEST_F(RtpDataChannelDoubleThreadTest, TestOnTransportReadyToSendWithRtcpMux) {
>-  Base::TestOnTransportReadyToSendWithRtcpMux();
>-}
>-
> TEST_F(RtpDataChannelDoubleThreadTest, SendRtpToRtp) {
>   Base::SendRtpToRtp();
> }
>@@ -3309,50 +2341,14 @@ TEST_F(RtpDataChannelDoubleThreadTest, SendRtcpToRtcp) {
>   Base::SendRtcpToRtcp();
> }
> 
>-TEST_F(RtpDataChannelDoubleThreadTest, SendRtcpMuxToRtcp) {
>-  Base::SendRtcpMuxToRtcp();
>-}
>-
>-TEST_F(RtpDataChannelDoubleThreadTest, SendRtcpMuxToRtcpMux) {
>-  Base::SendRtcpMuxToRtcpMux();
>-}
>-
>-TEST_F(RtpDataChannelDoubleThreadTest, SendEarlyRtcpMuxToRtcp) {
>-  Base::SendEarlyRtcpMuxToRtcp();
>-}
>-
>-TEST_F(RtpDataChannelDoubleThreadTest, SendEarlyRtcpMuxToRtcpMux) {
>-  Base::SendEarlyRtcpMuxToRtcpMux();
>-}
>-
>-TEST_F(RtpDataChannelDoubleThreadTest, SendSrtpToSrtp) {
>-  Base::SendSrtpToSrtp();
>-}
>-
>-TEST_F(RtpDataChannelDoubleThreadTest, SendSrtpToRtp) {
>-  Base::SendSrtpToSrtp();
>-}
>-
>-TEST_F(RtpDataChannelDoubleThreadTest, SendSrtcpMux) {
>-  Base::SendSrtpToSrtp(RTCP_MUX, RTCP_MUX);
>-}
>-
> TEST_F(RtpDataChannelDoubleThreadTest, SendRtpToRtpOnThread) {
>   Base::SendRtpToRtpOnThread();
> }
> 
>-TEST_F(RtpDataChannelDoubleThreadTest, SendSrtpToSrtpOnThread) {
>-  Base::SendSrtpToSrtpOnThread();
>-}
>-
> TEST_F(RtpDataChannelDoubleThreadTest, SendWithWritabilityLoss) {
>   Base::SendWithWritabilityLoss();
> }
> 
>-TEST_F(RtpDataChannelDoubleThreadTest, TestMediaMonitor) {
>-  Base::TestMediaMonitor();
>-}
>-
> TEST_F(RtpDataChannelDoubleThreadTest, SocketOptionsMergedOnSetTransport) {
>   Base::SocketOptionsMergedOnSetTransport();
> }
>@@ -3364,118 +2360,12 @@ TEST_F(RtpDataChannelDoubleThreadTest, TestSendData) {
> 
>   cricket::SendDataParams params;
>   params.ssrc = 42;
>-  unsigned char data[] = {
>-    'f', 'o', 'o'
>-  };
>+  unsigned char data[] = {'f', 'o', 'o'};
>   rtc::CopyOnWriteBuffer payload(data, 3);
>   cricket::SendDataResult result;
>   ASSERT_TRUE(media_channel1_->SendData(params, payload, &result));
>-  EXPECT_EQ(params.ssrc,
>-            media_channel1_->last_sent_data_params().ssrc);
>+  EXPECT_EQ(params.ssrc, media_channel1_->last_sent_data_params().ssrc);
>   EXPECT_EQ("foo", media_channel1_->last_sent_data());
> }
> 
>-#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
>-
>-// Verifies some DCHECKs are in place.
>-// Uses VoiceChannel, but any BaseChannel subclass would work.
>-class BaseChannelDeathTest : public testing::Test {
>- public:
>-  BaseChannelDeathTest()
>-      : fake_rtp_dtls_transport_("foo", cricket::ICE_CANDIDATE_COMPONENT_RTP),
>-        fake_rtcp_dtls_transport_("foo", cricket::ICE_CANDIDATE_COMPONENT_RTCP),
>-        // RTCP mux not required, SRTP required.
>-        voice_channel_(rtc::Thread::Current(),
>-                       rtc::Thread::Current(),
>-                       rtc::Thread::Current(),
>-                       &fake_media_engine_,
>-                       rtc::MakeUnique<cricket::FakeVoiceMediaChannel>(
>-                           nullptr,
>-                           cricket::AudioOptions()),
>-                       cricket::CN_AUDIO,
>-                       false,
>-                       true) {}
>-
>- protected:
>-  cricket::FakeMediaEngine fake_media_engine_;
>-  cricket::FakeDtlsTransport fake_rtp_dtls_transport_;
>-  cricket::FakeDtlsTransport fake_rtcp_dtls_transport_;
>-  cricket::VoiceChannel voice_channel_;
>-};
>-
>-TEST_F(BaseChannelDeathTest, SetTransportsWithNullRtpTransport) {
>-  voice_channel_.Init_w(&fake_rtp_dtls_transport_, &fake_rtcp_dtls_transport_,
>-                        &fake_rtp_dtls_transport_, &fake_rtcp_dtls_transport_);
>-  cricket::FakeDtlsTransport new_rtcp_transport(
>-      "bar", cricket::ICE_CANDIDATE_COMPONENT_RTCP);
>-  EXPECT_DEATH(voice_channel_.SetTransports(nullptr, &new_rtcp_transport), "");
>-}
>-
>-TEST_F(BaseChannelDeathTest, SetTransportsWithMissingRtcpTransport) {
>-  voice_channel_.Init_w(&fake_rtp_dtls_transport_, &fake_rtcp_dtls_transport_,
>-                        &fake_rtp_dtls_transport_, &fake_rtcp_dtls_transport_);
>-  cricket::FakeDtlsTransport new_rtp_transport(
>-      "bar", cricket::ICE_CANDIDATE_COMPONENT_RTP);
>-  EXPECT_DEATH(voice_channel_.SetTransports(&new_rtp_transport, nullptr), "");
>-}
>-
>-TEST_F(BaseChannelDeathTest, SetTransportsWithUnneededRtcpTransport) {
>-  voice_channel_.Init_w(&fake_rtp_dtls_transport_, &fake_rtcp_dtls_transport_,
>-                        &fake_rtp_dtls_transport_, &fake_rtcp_dtls_transport_);
>-  // Activate RTCP muxing, simulating offer/answer negotiation.
>-  cricket::AudioContentDescription content;
>-  content.set_rtcp_mux(true);
>-  ASSERT_TRUE(
>-      voice_channel_.SetLocalContent(&content, SdpType::kOffer, nullptr));
>-  ASSERT_TRUE(
>-      voice_channel_.SetRemoteContent(&content, SdpType::kAnswer, nullptr));
>-  cricket::FakeDtlsTransport new_rtp_transport(
>-      "bar", cricket::ICE_CANDIDATE_COMPONENT_RTP);
>-  cricket::FakeDtlsTransport new_rtcp_transport(
>-      "bar", cricket::ICE_CANDIDATE_COMPONENT_RTCP);
>-  // After muxing is enabled, no RTCP transport should be passed in here.
>-  EXPECT_DEATH(
>-      voice_channel_.SetTransports(&new_rtp_transport, &new_rtcp_transport),
>-      "");
>-}
>-
>-// This test will probably go away if/when we move the transport name out of
>-// the transport classes and into their parent classes.
>-TEST_F(BaseChannelDeathTest, SetTransportsWithMismatchingTransportNames) {
>-  voice_channel_.Init_w(&fake_rtp_dtls_transport_, &fake_rtcp_dtls_transport_,
>-                        &fake_rtp_dtls_transport_, &fake_rtcp_dtls_transport_);
>-  cricket::FakeDtlsTransport new_rtp_transport(
>-      "bar", cricket::ICE_CANDIDATE_COMPONENT_RTP);
>-  cricket::FakeDtlsTransport new_rtcp_transport(
>-      "baz", cricket::ICE_CANDIDATE_COMPONENT_RTCP);
>-  EXPECT_DEATH(
>-      voice_channel_.SetTransports(&new_rtp_transport, &new_rtcp_transport),
>-      "");
>-}
>-
>-// Not expected to support going from DtlsTransportInternal to
>-// PacketTransportInternal.
>-TEST_F(BaseChannelDeathTest, SetTransportsDtlsToNonDtls) {
>-  voice_channel_.Init_w(&fake_rtp_dtls_transport_, &fake_rtcp_dtls_transport_,
>-                        &fake_rtp_dtls_transport_, &fake_rtcp_dtls_transport_);
>-  EXPECT_DEATH(
>-      voice_channel_.SetTransports(
>-          static_cast<rtc::PacketTransportInternal*>(&fake_rtp_dtls_transport_),
>-          static_cast<rtc::PacketTransportInternal*>(
>-              &fake_rtp_dtls_transport_)),
>-      "");
>-}
>-
>-// Not expected to support going from PacketTransportInternal to
>-// DtlsTransportInternal.
>-TEST_F(BaseChannelDeathTest, SetTransportsNonDtlsToDtls) {
>-  voice_channel_.Init_w(nullptr, nullptr, &fake_rtp_dtls_transport_,
>-                        &fake_rtcp_dtls_transport_);
>-  EXPECT_DEATH(voice_channel_.SetTransports(&fake_rtp_dtls_transport_,
>-                                            &fake_rtp_dtls_transport_),
>-               "");
>-}
>-
>-#endif  // RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
>-
> // TODO(pthatcher): TestSetReceiver?
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/channelmanager.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/channelmanager.cc
>index ead1da9cdd3..dab622ad70e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/channelmanager.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/channelmanager.cc
>@@ -13,10 +13,10 @@
> #include <algorithm>
> #include <utility>
> 
>+#include "absl/memory/memory.h"
> #include "media/base/rtpdataengine.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/stringutils.h"
> #include "rtc_base/trace_event.h"
> 
>@@ -152,38 +152,6 @@ void ChannelManager::Terminate() {
>   initialized_ = false;
> }
> 
>-VoiceChannel* ChannelManager::CreateVoiceChannel(
>-    webrtc::Call* call,
>-    const cricket::MediaConfig& media_config,
>-    DtlsTransportInternal* rtp_transport,
>-    DtlsTransportInternal* rtcp_transport,
>-    rtc::Thread* signaling_thread,
>-    const std::string& content_name,
>-    bool srtp_required,
>-    const AudioOptions& options) {
>-  return worker_thread_->Invoke<VoiceChannel*>(RTC_FROM_HERE, [&] {
>-    return CreateVoiceChannel_w(
>-        call, media_config, rtp_transport, rtcp_transport, rtp_transport,
>-        rtcp_transport, signaling_thread, content_name, srtp_required, options);
>-  });
>-}
>-
>-VoiceChannel* ChannelManager::CreateVoiceChannel(
>-    webrtc::Call* call,
>-    const cricket::MediaConfig& media_config,
>-    rtc::PacketTransportInternal* rtp_transport,
>-    rtc::PacketTransportInternal* rtcp_transport,
>-    rtc::Thread* signaling_thread,
>-    const std::string& content_name,
>-    bool srtp_required,
>-    const AudioOptions& options) {
>-  return worker_thread_->Invoke<VoiceChannel*>(RTC_FROM_HERE, [&] {
>-    return CreateVoiceChannel_w(call, media_config, nullptr, nullptr,
>-                                rtp_transport, rtcp_transport, signaling_thread,
>-                                content_name, srtp_required, options);
>-  });
>-}
>-
> VoiceChannel* ChannelManager::CreateVoiceChannel(
>     webrtc::Call* call,
>     const cricket::MediaConfig& media_config,
>@@ -191,12 +159,13 @@ VoiceChannel* ChannelManager::CreateVoiceChannel(
>     rtc::Thread* signaling_thread,
>     const std::string& content_name,
>     bool srtp_required,
>+    const rtc::CryptoOptions& crypto_options,
>     const AudioOptions& options) {
>   if (!worker_thread_->IsCurrent()) {
>     return worker_thread_->Invoke<VoiceChannel*>(RTC_FROM_HERE, [&] {
>       return CreateVoiceChannel(call, media_config, rtp_transport,
>                                 signaling_thread, content_name, srtp_required,
>-                                options);
>+                                crypto_options, options);
>     });
>   }
> 
>@@ -213,10 +182,10 @@ VoiceChannel* ChannelManager::CreateVoiceChannel(
>     return nullptr;
>   }
> 
>-  auto voice_channel = rtc::MakeUnique<VoiceChannel>(
>+  auto voice_channel = absl::make_unique<VoiceChannel>(
>       worker_thread_, network_thread_, signaling_thread, media_engine_.get(),
>-      rtc::WrapUnique(media_channel), content_name,
>-      rtp_transport->rtcp_packet_transport() == nullptr, srtp_required);
>+      absl::WrapUnique(media_channel), content_name, srtp_required,
>+      crypto_options);
> 
>   voice_channel->Init_w(rtp_transport);
> 
>@@ -225,42 +194,6 @@ VoiceChannel* ChannelManager::CreateVoiceChannel(
>   return voice_channel_ptr;
> }
> 
>-VoiceChannel* ChannelManager::CreateVoiceChannel_w(
>-    webrtc::Call* call,
>-    const cricket::MediaConfig& media_config,
>-    DtlsTransportInternal* rtp_dtls_transport,
>-    DtlsTransportInternal* rtcp_dtls_transport,
>-    rtc::PacketTransportInternal* rtp_packet_transport,
>-    rtc::PacketTransportInternal* rtcp_packet_transport,
>-    rtc::Thread* signaling_thread,
>-    const std::string& content_name,
>-    bool srtp_required,
>-    const AudioOptions& options) {
>-  RTC_DCHECK_RUN_ON(worker_thread_);
>-  RTC_DCHECK(initialized_);
>-  RTC_DCHECK(call);
>-  if (!media_engine_) {
>-    return nullptr;
>-  }
>-
>-  VoiceMediaChannel* media_channel = media_engine_->CreateChannel(
>-      call, media_config, options);
>-  if (!media_channel) {
>-    return nullptr;
>-  }
>-
>-  auto voice_channel = rtc::MakeUnique<VoiceChannel>(
>-      worker_thread_, network_thread_, signaling_thread, media_engine_.get(),
>-      rtc::WrapUnique(media_channel), content_name,
>-      rtcp_packet_transport == nullptr, srtp_required);
>-  voice_channel->Init_w(rtp_dtls_transport, rtcp_dtls_transport,
>-                        rtp_packet_transport, rtcp_packet_transport);
>-
>-  VoiceChannel* voice_channel_ptr = voice_channel.get();
>-  voice_channels_.push_back(std::move(voice_channel));
>-  return voice_channel_ptr;
>-}
>-
> void ChannelManager::DestroyVoiceChannel(VoiceChannel* voice_channel) {
>   TRACE_EVENT0("webrtc", "ChannelManager::DestroyVoiceChannel");
>   if (!voice_channel) {
>@@ -286,38 +219,6 @@ void ChannelManager::DestroyVoiceChannel(VoiceChannel* voice_channel) {
>   voice_channels_.erase(it);
> }
> 
>-VideoChannel* ChannelManager::CreateVideoChannel(
>-    webrtc::Call* call,
>-    const cricket::MediaConfig& media_config,
>-    DtlsTransportInternal* rtp_transport,
>-    DtlsTransportInternal* rtcp_transport,
>-    rtc::Thread* signaling_thread,
>-    const std::string& content_name,
>-    bool srtp_required,
>-    const VideoOptions& options) {
>-  return worker_thread_->Invoke<VideoChannel*>(RTC_FROM_HERE, [&] {
>-    return CreateVideoChannel_w(
>-        call, media_config, rtp_transport, rtcp_transport, rtp_transport,
>-        rtcp_transport, signaling_thread, content_name, srtp_required, options);
>-  });
>-}
>-
>-VideoChannel* ChannelManager::CreateVideoChannel(
>-    webrtc::Call* call,
>-    const cricket::MediaConfig& media_config,
>-    rtc::PacketTransportInternal* rtp_transport,
>-    rtc::PacketTransportInternal* rtcp_transport,
>-    rtc::Thread* signaling_thread,
>-    const std::string& content_name,
>-    bool srtp_required,
>-    const VideoOptions& options) {
>-  return worker_thread_->Invoke<VideoChannel*>(RTC_FROM_HERE, [&] {
>-    return CreateVideoChannel_w(call, media_config, nullptr, nullptr,
>-                                rtp_transport, rtcp_transport, signaling_thread,
>-                                content_name, srtp_required, options);
>-  });
>-}
>-
> VideoChannel* ChannelManager::CreateVideoChannel(
>     webrtc::Call* call,
>     const cricket::MediaConfig& media_config,
>@@ -325,12 +226,13 @@ VideoChannel* ChannelManager::CreateVideoChannel(
>     rtc::Thread* signaling_thread,
>     const std::string& content_name,
>     bool srtp_required,
>+    const rtc::CryptoOptions& crypto_options,
>     const VideoOptions& options) {
>   if (!worker_thread_->IsCurrent()) {
>     return worker_thread_->Invoke<VideoChannel*>(RTC_FROM_HERE, [&] {
>       return CreateVideoChannel(call, media_config, rtp_transport,
>                                 signaling_thread, content_name, srtp_required,
>-                                options);
>+                                crypto_options, options);
>     });
>   }
> 
>@@ -347,10 +249,10 @@ VideoChannel* ChannelManager::CreateVideoChannel(
>     return nullptr;
>   }
> 
>-  auto video_channel = rtc::MakeUnique<VideoChannel>(
>+  auto video_channel = absl::make_unique<VideoChannel>(
>       worker_thread_, network_thread_, signaling_thread,
>-      rtc::WrapUnique(media_channel), content_name,
>-      rtp_transport->rtcp_packet_transport() == nullptr, srtp_required);
>+      absl::WrapUnique(media_channel), content_name, srtp_required,
>+      crypto_options);
>   video_channel->Init_w(rtp_transport);
> 
>   VideoChannel* video_channel_ptr = video_channel.get();
>@@ -358,42 +260,6 @@ VideoChannel* ChannelManager::CreateVideoChannel(
>   return video_channel_ptr;
> }
> 
>-VideoChannel* ChannelManager::CreateVideoChannel_w(
>-    webrtc::Call* call,
>-    const cricket::MediaConfig& media_config,
>-    DtlsTransportInternal* rtp_dtls_transport,
>-    DtlsTransportInternal* rtcp_dtls_transport,
>-    rtc::PacketTransportInternal* rtp_packet_transport,
>-    rtc::PacketTransportInternal* rtcp_packet_transport,
>-    rtc::Thread* signaling_thread,
>-    const std::string& content_name,
>-    bool srtp_required,
>-    const VideoOptions& options) {
>-  RTC_DCHECK_RUN_ON(worker_thread_);
>-  RTC_DCHECK(initialized_);
>-  RTC_DCHECK(call);
>-  if (!media_engine_) {
>-    return nullptr;
>-  }
>-
>-  VideoMediaChannel* media_channel = media_engine_->CreateVideoChannel(
>-      call, media_config, options);
>-  if (!media_channel) {
>-    return nullptr;
>-  }
>-
>-  auto video_channel = rtc::MakeUnique<VideoChannel>(
>-      worker_thread_, network_thread_, signaling_thread,
>-      rtc::WrapUnique(media_channel), content_name,
>-      rtcp_packet_transport == nullptr, srtp_required);
>-  video_channel->Init_w(rtp_dtls_transport, rtcp_dtls_transport,
>-                        rtp_packet_transport, rtcp_packet_transport);
>-
>-  VideoChannel* video_channel_ptr = video_channel.get();
>-  video_channels_.push_back(std::move(video_channel));
>-  return video_channel_ptr;
>-}
>-
> void ChannelManager::DestroyVideoChannel(VideoChannel* video_channel) {
>   TRACE_EVENT0("webrtc", "ChannelManager::DestroyVideoChannel");
>   if (!video_channel) {
>@@ -419,51 +285,17 @@ void ChannelManager::DestroyVideoChannel(VideoChannel* video_channel) {
>   video_channels_.erase(it);
> }
> 
>-RtpDataChannel* ChannelManager::CreateRtpDataChannel(
>-    const cricket::MediaConfig& media_config,
>-    DtlsTransportInternal* rtp_transport,
>-    DtlsTransportInternal* rtcp_transport,
>-    rtc::Thread* signaling_thread,
>-    const std::string& content_name,
>-    bool srtp_required) {
>-  if (!worker_thread_->IsCurrent()) {
>-    return worker_thread_->Invoke<RtpDataChannel*>(RTC_FROM_HERE, [&] {
>-      return CreateRtpDataChannel(media_config, rtp_transport, rtcp_transport,
>-                                  signaling_thread, content_name,
>-                                  srtp_required);
>-    });
>-  }
>-
>-  // This is ok to alloc from a thread other than the worker thread.
>-  RTC_DCHECK(initialized_);
>-  DataMediaChannel* media_channel = data_engine_->CreateChannel(media_config);
>-  if (!media_channel) {
>-    RTC_LOG(LS_WARNING) << "Failed to create RTP data channel.";
>-    return nullptr;
>-  }
>-
>-  auto data_channel = rtc::MakeUnique<RtpDataChannel>(
>-      worker_thread_, network_thread_, signaling_thread,
>-      rtc::WrapUnique(media_channel), content_name, rtcp_transport == nullptr,
>-      srtp_required);
>-  data_channel->Init_w(rtp_transport, rtcp_transport, rtp_transport,
>-                       rtcp_transport);
>-
>-  RtpDataChannel* data_channel_ptr = data_channel.get();
>-  data_channels_.push_back(std::move(data_channel));
>-  return data_channel_ptr;
>-}
>-
> RtpDataChannel* ChannelManager::CreateRtpDataChannel(
>     const cricket::MediaConfig& media_config,
>     webrtc::RtpTransportInternal* rtp_transport,
>     rtc::Thread* signaling_thread,
>     const std::string& content_name,
>-    bool srtp_required) {
>+    bool srtp_required,
>+    const rtc::CryptoOptions& crypto_options) {
>   if (!worker_thread_->IsCurrent()) {
>     return worker_thread_->Invoke<RtpDataChannel*>(RTC_FROM_HERE, [&] {
>       return CreateRtpDataChannel(media_config, rtp_transport, signaling_thread,
>-                                  content_name, srtp_required);
>+                                  content_name, srtp_required, crypto_options);
>     });
>   }
> 
>@@ -475,10 +307,10 @@ RtpDataChannel* ChannelManager::CreateRtpDataChannel(
>     return nullptr;
>   }
> 
>-  auto data_channel = rtc::MakeUnique<RtpDataChannel>(
>+  auto data_channel = absl::make_unique<RtpDataChannel>(
>       worker_thread_, network_thread_, signaling_thread,
>-      rtc::WrapUnique(media_channel), content_name,
>-      rtp_transport->rtcp_packet_transport() == nullptr, srtp_required);
>+      absl::WrapUnique(media_channel), content_name, srtp_required,
>+      crypto_options);
>   data_channel->Init_w(rtp_transport);
> 
>   RtpDataChannel* data_channel_ptr = data_channel.get();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/channelmanager.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/channelmanager.h
>index 72d218a228d..c6b601ea8f6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/channelmanager.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/channelmanager.h
>@@ -80,90 +80,38 @@ class ChannelManager final {
>   // call the appropriate Destroy*Channel method when done.
> 
>   // Creates a voice channel, to be associated with the specified session.
>-  // TODO(zhihuang): Replace this with the method taking an
>-  // RtpTransportInternal*;
>-  VoiceChannel* CreateVoiceChannel(
>-      webrtc::Call* call,
>-      const cricket::MediaConfig& media_config,
>-      DtlsTransportInternal* rtp_transport,
>-      DtlsTransportInternal* rtcp_transport,
>-      rtc::Thread* signaling_thread,
>-      const std::string& content_name,
>-      bool srtp_required,
>-      const AudioOptions& options);
>-  // Version of the above that takes PacketTransportInternal.
>-  // TODO(zhihuang): Replace this with the method taking an
>-  // RtpTransportInternal*;
>-  VoiceChannel* CreateVoiceChannel(
>-      webrtc::Call* call,
>-      const cricket::MediaConfig& media_config,
>-      rtc::PacketTransportInternal* rtp_transport,
>-      rtc::PacketTransportInternal* rtcp_transport,
>-      rtc::Thread* signaling_thread,
>-      const std::string& content_name,
>-      bool srtp_required,
>-      const AudioOptions& options);
>-
>   VoiceChannel* CreateVoiceChannel(webrtc::Call* call,
>                                    const cricket::MediaConfig& media_config,
>                                    webrtc::RtpTransportInternal* rtp_transport,
>                                    rtc::Thread* signaling_thread,
>                                    const std::string& content_name,
>                                    bool srtp_required,
>+                                   const rtc::CryptoOptions& crypto_options,
>                                    const AudioOptions& options);
>   // Destroys a voice channel created by CreateVoiceChannel.
>   void DestroyVoiceChannel(VoiceChannel* voice_channel);
> 
>   // Creates a video channel, synced with the specified voice channel, and
>   // associated with the specified session.
>-  // TODO(zhihuang): Replace this with the method taking an
>-  // RtpTransportInternal*;
>-  VideoChannel* CreateVideoChannel(
>-      webrtc::Call* call,
>-      const cricket::MediaConfig& media_config,
>-      DtlsTransportInternal* rtp_transport,
>-      DtlsTransportInternal* rtcp_transport,
>-      rtc::Thread* signaling_thread,
>-      const std::string& content_name,
>-      bool srtp_required,
>-      const VideoOptions& options);
>   // Version of the above that takes PacketTransportInternal.
>-  // TODO(zhihuang): Replace this with the method taking an
>-  // RtpTransportInternal*;
>-  VideoChannel* CreateVideoChannel(
>-      webrtc::Call* call,
>-      const cricket::MediaConfig& media_config,
>-      rtc::PacketTransportInternal* rtp_transport,
>-      rtc::PacketTransportInternal* rtcp_transport,
>-      rtc::Thread* signaling_thread,
>-      const std::string& content_name,
>-      bool srtp_required,
>-      const VideoOptions& options);
>   VideoChannel* CreateVideoChannel(webrtc::Call* call,
>                                    const cricket::MediaConfig& media_config,
>                                    webrtc::RtpTransportInternal* rtp_transport,
>                                    rtc::Thread* signaling_thread,
>                                    const std::string& content_name,
>                                    bool srtp_required,
>+                                   const rtc::CryptoOptions& crypto_options,
>                                    const VideoOptions& options);
>   // Destroys a video channel created by CreateVideoChannel.
>   void DestroyVideoChannel(VideoChannel* video_channel);
> 
>-  // TODO(zhihuang): Replace this with the method taking an
>-  // RtpTransportInternal*;
>-  RtpDataChannel* CreateRtpDataChannel(
>-      const cricket::MediaConfig& media_config,
>-      DtlsTransportInternal* rtp_transport,
>-      DtlsTransportInternal* rtcp_transport,
>-      rtc::Thread* signaling_thread,
>-      const std::string& content_name,
>-      bool srtp_required);
>   RtpDataChannel* CreateRtpDataChannel(
>       const cricket::MediaConfig& media_config,
>       webrtc::RtpTransportInternal* rtp_transport,
>       rtc::Thread* signaling_thread,
>       const std::string& content_name,
>-      bool srtp_required);
>+      bool srtp_required,
>+      const rtc::CryptoOptions& crypto_options);
>   // Destroys a data channel created by CreateRtpDataChannel.
>   void DestroyRtpDataChannel(RtpDataChannel* data_channel);
> 
>@@ -191,29 +139,6 @@ class ChannelManager final {
>   void StopAecDump();
> 
>  private:
>-  VoiceChannel* CreateVoiceChannel_w(
>-      webrtc::Call* call,
>-      const cricket::MediaConfig& media_config,
>-      DtlsTransportInternal* rtp_dtls_transport,
>-      DtlsTransportInternal* rtcp_dtls_transport,
>-      rtc::PacketTransportInternal* rtp_packet_transport,
>-      rtc::PacketTransportInternal* rtcp_packet_transport,
>-      rtc::Thread* signaling_thread,
>-      const std::string& content_name,
>-      bool srtp_required,
>-      const AudioOptions& options);
>-  VideoChannel* CreateVideoChannel_w(
>-      webrtc::Call* call,
>-      const cricket::MediaConfig& media_config,
>-      DtlsTransportInternal* rtp_dtls_transport,
>-      DtlsTransportInternal* rtcp_dtls_transport,
>-      rtc::PacketTransportInternal* rtp_packet_transport,
>-      rtc::PacketTransportInternal* rtcp_packet_transport,
>-      rtc::Thread* signaling_thread,
>-      const std::string& content_name,
>-      bool srtp_required,
>-      const VideoOptions& options);
>-
>   std::unique_ptr<MediaEngineInterface> media_engine_;  // Nullable.
>   std::unique_ptr<DataEngineInterface> data_engine_;    // Non-null.
>   bool initialized_ = false;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/channelmanager_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/channelmanager_unittest.cc
>index 18f84752f78..47c95306ee7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/channelmanager_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/channelmanager_unittest.cc
>@@ -11,13 +11,11 @@
> #include <memory>
> #include <utility>
> 
>-#include "logging/rtc_event_log/rtc_event_log.h"
> #include "media/base/fakemediaengine.h"
>-#include "media/base/fakevideocapturer.h"
> #include "media/base/testutils.h"
> #include "media/engine/fakewebrtccall.h"
>+#include "p2p/base/fakedtlstransport.h"
> #include "pc/channelmanager.h"
>-#include "pc/test/faketransportcontroller.h"
> #include "rtc_base/gunit.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/thread.h"
>@@ -48,14 +46,43 @@ class ChannelManagerTest : public testing::Test {
>             std::unique_ptr<DataEngineInterface>(fdme_),
>             rtc::Thread::Current(),
>             rtc::Thread::Current())),
>-        fake_call_(webrtc::Call::Config(&event_log_)),
>-        transport_controller_(
>-            new cricket::FakeTransportController(ICEROLE_CONTROLLING)) {
>+        fake_call_() {
>     fme_->SetAudioCodecs(MAKE_VECTOR(kAudioCodecs));
>     fme_->SetVideoCodecs(MAKE_VECTOR(kVideoCodecs));
>   }
> 
>-  webrtc::RtcEventLogNullImpl event_log_;
>+  std::unique_ptr<webrtc::RtpTransportInternal> CreateDtlsSrtpTransport() {
>+    rtp_dtls_transport_ = absl::make_unique<FakeDtlsTransport>(
>+        "fake_dtls_transport", cricket::ICE_CANDIDATE_COMPONENT_RTP);
>+    auto dtls_srtp_transport = absl::make_unique<webrtc::DtlsSrtpTransport>(
>+        /*rtcp_mux_required=*/true);
>+    dtls_srtp_transport->SetDtlsTransports(rtp_dtls_transport_.get(),
>+                                           /*rtcp_dtls_transport=*/nullptr);
>+    return dtls_srtp_transport;
>+  }
>+
>+  void TestCreateDestroyChannels(webrtc::RtpTransportInternal* rtp_transport) {
>+    cricket::VoiceChannel* voice_channel = cm_->CreateVoiceChannel(
>+        &fake_call_, cricket::MediaConfig(), rtp_transport,
>+        rtc::Thread::Current(), cricket::CN_AUDIO, kDefaultSrtpRequired,
>+        rtc::CryptoOptions(), AudioOptions());
>+    EXPECT_TRUE(voice_channel != nullptr);
>+    cricket::VideoChannel* video_channel = cm_->CreateVideoChannel(
>+        &fake_call_, cricket::MediaConfig(), rtp_transport,
>+        rtc::Thread::Current(), cricket::CN_VIDEO, kDefaultSrtpRequired,
>+        rtc::CryptoOptions(), VideoOptions());
>+    EXPECT_TRUE(video_channel != nullptr);
>+    cricket::RtpDataChannel* rtp_data_channel = cm_->CreateRtpDataChannel(
>+        cricket::MediaConfig(), rtp_transport, rtc::Thread::Current(),
>+        cricket::CN_DATA, kDefaultSrtpRequired, rtc::CryptoOptions());
>+    EXPECT_TRUE(rtp_data_channel != nullptr);
>+    cm_->DestroyVideoChannel(video_channel);
>+    cm_->DestroyVoiceChannel(voice_channel);
>+    cm_->DestroyRtpDataChannel(rtp_data_channel);
>+    cm_->Terminate();
>+  }
>+
>+  std::unique_ptr<DtlsTransportInternal> rtp_dtls_transport_;
>   std::unique_ptr<rtc::Thread> network_;
>   std::unique_ptr<rtc::Thread> worker_;
>   // |fme_| and |fdme_| are actually owned by |cm_|.
>@@ -63,7 +90,6 @@ class ChannelManagerTest : public testing::Test {
>   cricket::FakeDataEngine* fdme_;
>   std::unique_ptr<cricket::ChannelManager> cm_;
>   cricket::FakeCall fake_call_;
>-  std::unique_ptr<cricket::FakeTransportController> transport_controller_;
> };
> 
> // Test that we startup/shutdown properly.
>@@ -95,68 +121,6 @@ TEST_F(ChannelManagerTest, StartupShutdownOnThread) {
>   EXPECT_FALSE(cm_->initialized());
> }
> 
>-// Test that we can create and destroy a voice and video channel.
>-TEST_F(ChannelManagerTest, CreateDestroyChannels) {
>-  EXPECT_TRUE(cm_->Init());
>-  cricket::DtlsTransportInternal* rtp_transport =
>-      transport_controller_->CreateDtlsTransport(
>-          cricket::CN_AUDIO, cricket::ICE_CANDIDATE_COMPONENT_RTP);
>-  cricket::VoiceChannel* voice_channel = cm_->CreateVoiceChannel(
>-      &fake_call_, cricket::MediaConfig(),
>-      rtp_transport, nullptr /*rtcp_transport*/,
>-      rtc::Thread::Current(), cricket::CN_AUDIO, kDefaultSrtpRequired,
>-      AudioOptions());
>-  EXPECT_TRUE(voice_channel != nullptr);
>-  cricket::VideoChannel* video_channel = cm_->CreateVideoChannel(
>-      &fake_call_, cricket::MediaConfig(),
>-      rtp_transport, nullptr /*rtcp_transport*/,
>-      rtc::Thread::Current(), cricket::CN_VIDEO, kDefaultSrtpRequired,
>-      VideoOptions());
>-  EXPECT_TRUE(video_channel != nullptr);
>-  cricket::RtpDataChannel* rtp_data_channel = cm_->CreateRtpDataChannel(
>-      cricket::MediaConfig(), rtp_transport, nullptr /*rtcp_transport*/,
>-      rtc::Thread::Current(), cricket::CN_DATA, kDefaultSrtpRequired);
>-  EXPECT_TRUE(rtp_data_channel != nullptr);
>-  cm_->DestroyVideoChannel(video_channel);
>-  cm_->DestroyVoiceChannel(voice_channel);
>-  cm_->DestroyRtpDataChannel(rtp_data_channel);
>-  cm_->Terminate();
>-}
>-
>-// Test that we can create and destroy a voice and video channel with a worker.
>-TEST_F(ChannelManagerTest, CreateDestroyChannelsOnThread) {
>-  network_->Start();
>-  worker_->Start();
>-  EXPECT_TRUE(cm_->set_worker_thread(worker_.get()));
>-  EXPECT_TRUE(cm_->set_network_thread(network_.get()));
>-  EXPECT_TRUE(cm_->Init());
>-  transport_controller_.reset(new cricket::FakeTransportController(
>-      network_.get(), ICEROLE_CONTROLLING));
>-  cricket::DtlsTransportInternal* rtp_transport =
>-      transport_controller_->CreateDtlsTransport(
>-          cricket::CN_AUDIO, cricket::ICE_CANDIDATE_COMPONENT_RTP);
>-  cricket::VoiceChannel* voice_channel = cm_->CreateVoiceChannel(
>-      &fake_call_, cricket::MediaConfig(),
>-      rtp_transport, nullptr /*rtcp_transport*/,
>-      rtc::Thread::Current(), cricket::CN_AUDIO, kDefaultSrtpRequired,
>-      AudioOptions());
>-  EXPECT_TRUE(voice_channel != nullptr);
>-  cricket::VideoChannel* video_channel = cm_->CreateVideoChannel(
>-      &fake_call_, cricket::MediaConfig(),
>-      rtp_transport, nullptr /*rtcp_transport*/,
>-      rtc::Thread::Current(), cricket::CN_VIDEO, kDefaultSrtpRequired,
>-      VideoOptions());
>-  EXPECT_TRUE(video_channel != nullptr);
>-  cricket::RtpDataChannel* rtp_data_channel = cm_->CreateRtpDataChannel(
>-      cricket::MediaConfig(), rtp_transport, nullptr /*rtcp_transport*/,
>-      rtc::Thread::Current(), cricket::CN_DATA, kDefaultSrtpRequired);
>-  EXPECT_TRUE(rtp_data_channel != nullptr);
>-  cm_->DestroyVideoChannel(video_channel);
>-  cm_->DestroyVoiceChannel(voice_channel);
>-  cm_->DestroyRtpDataChannel(rtp_data_channel);
>-  cm_->Terminate();
>-}
>-
> TEST_F(ChannelManagerTest, SetVideoRtxEnabled) {
>   std::vector<VideoCodec> codecs;
>   const VideoCodec rtx_codec(96, "rtx");
>@@ -187,89 +151,20 @@ TEST_F(ChannelManagerTest, SetVideoRtxEnabled) {
>   EXPECT_TRUE(ContainsMatchingCodec(codecs, rtx_codec));
> }
> 
>-enum class RTPTransportType { kRtp, kSrtp, kDtlsSrtp };
>-
>-class ChannelManagerTestWithRtpTransport
>-    : public ChannelManagerTest,
>-      public ::testing::WithParamInterface<RTPTransportType> {
>- public:
>-  std::unique_ptr<webrtc::RtpTransportInternal> CreateRtpTransport() {
>-    RTPTransportType type = GetParam();
>-    switch (type) {
>-      case RTPTransportType::kRtp:
>-        return CreatePlainRtpTransport();
>-      case RTPTransportType::kSrtp:
>-        return CreateSrtpTransport();
>-      case RTPTransportType::kDtlsSrtp:
>-        return CreateDtlsSrtpTransport();
>-    }
>-    return nullptr;
>-  }
>-
>-  void TestCreateDestroyChannels(webrtc::RtpTransportInternal* rtp_transport) {
>-    cricket::VoiceChannel* voice_channel = cm_->CreateVoiceChannel(
>-        &fake_call_, cricket::MediaConfig(), rtp_transport,
>-        rtc::Thread::Current(), cricket::CN_AUDIO, kDefaultSrtpRequired,
>-        AudioOptions());
>-    EXPECT_TRUE(voice_channel != nullptr);
>-    cricket::VideoChannel* video_channel = cm_->CreateVideoChannel(
>-        &fake_call_, cricket::MediaConfig(), rtp_transport,
>-        rtc::Thread::Current(), cricket::CN_VIDEO, kDefaultSrtpRequired,
>-        VideoOptions());
>-    EXPECT_TRUE(video_channel != nullptr);
>-    cricket::RtpDataChannel* rtp_data_channel = cm_->CreateRtpDataChannel(
>-        cricket::MediaConfig(), rtp_transport, rtc::Thread::Current(),
>-        cricket::CN_DATA, kDefaultSrtpRequired);
>-    EXPECT_TRUE(rtp_data_channel != nullptr);
>-    cm_->DestroyVideoChannel(video_channel);
>-    cm_->DestroyVoiceChannel(voice_channel);
>-    cm_->DestroyRtpDataChannel(rtp_data_channel);
>-    cm_->Terminate();
>-  }
>-
>- private:
>-  std::unique_ptr<webrtc::RtpTransportInternal> CreatePlainRtpTransport() {
>-    return rtc::MakeUnique<webrtc::RtpTransport>(/*rtcp_mux_required=*/true);
>-  }
>-
>-  std::unique_ptr<webrtc::RtpTransportInternal> CreateSrtpTransport() {
>-    auto rtp_transport =
>-        rtc::MakeUnique<webrtc::RtpTransport>(/*rtcp_mux_required=*/true);
>-    auto srtp_transport =
>-        rtc::MakeUnique<webrtc::SrtpTransport>(std::move(rtp_transport));
>-    return srtp_transport;
>-  }
>-
>-  std::unique_ptr<webrtc::RtpTransportInternal> CreateDtlsSrtpTransport() {
>-    auto rtp_transport =
>-        rtc::MakeUnique<webrtc::RtpTransport>(/*rtcp_mux_required=*/true);
>-    auto srtp_transport =
>-        rtc::MakeUnique<webrtc::SrtpTransport>(std::move(rtp_transport));
>-    auto dtls_srtp_transport_ =
>-        rtc::MakeUnique<webrtc::DtlsSrtpTransport>(std::move(srtp_transport));
>-    return dtls_srtp_transport_;
>-  }
>-};
>-
>-TEST_P(ChannelManagerTestWithRtpTransport, CreateDestroyChannels) {
>+TEST_F(ChannelManagerTest, CreateDestroyChannels) {
>   EXPECT_TRUE(cm_->Init());
>-  auto rtp_transport = CreateRtpTransport();
>+  auto rtp_transport = CreateDtlsSrtpTransport();
>   TestCreateDestroyChannels(rtp_transport.get());
> }
> 
>-TEST_P(ChannelManagerTestWithRtpTransport, CreateDestroyChannelsOnThread) {
>+TEST_F(ChannelManagerTest, CreateDestroyChannelsOnThread) {
>   network_->Start();
>   worker_->Start();
>   EXPECT_TRUE(cm_->set_worker_thread(worker_.get()));
>   EXPECT_TRUE(cm_->set_network_thread(network_.get()));
>   EXPECT_TRUE(cm_->Init());
>-  auto rtp_transport = CreateRtpTransport();
>+  auto rtp_transport = CreateDtlsSrtpTransport();
>   TestCreateDestroyChannels(rtp_transport.get());
> }
> 
>-INSTANTIATE_TEST_CASE_P(ChannelManagerTest,
>-                        ChannelManagerTestWithRtpTransport,
>-                        ::testing::Values(RTPTransportType::kRtp,
>-                                          RTPTransportType::kSrtp,
>-                                          RTPTransportType::kDtlsSrtp));
> }  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/createpeerconnectionfactory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/createpeerconnectionfactory.cc
>index 3f389af6412..bbc4ebbe730 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/createpeerconnectionfactory.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/createpeerconnectionfactory.cc
>@@ -8,10 +8,10 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>+#include "api/call/callfactoryinterface.h"
> #include "api/peerconnectioninterface.h"
> #include "api/video_codecs/video_decoder_factory.h"
> #include "api/video_codecs/video_encoder_factory.h"
>-#include "call/callfactoryinterface.h"
> #include "logging/rtc_event_log/rtc_event_log_factory_interface.h"
> #include "media/engine/webrtcmediaengine.h"
> #include "modules/audio_device/include/audio_device.h"
>@@ -22,6 +22,7 @@
> 
> namespace webrtc {
> 
>+#if defined(USE_BUILTIN_SW_CODECS)
> rtc::scoped_refptr<PeerConnectionFactoryInterface> CreatePeerConnectionFactory(
>     rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
>     rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory) {
>@@ -67,6 +68,43 @@ rtc::scoped_refptr<PeerConnectionFactoryInterface> CreatePeerConnectionFactory(
>       std::move(call_factory), std::move(event_log_factory));
> }
> 
>+rtc::scoped_refptr<PeerConnectionFactoryInterface> CreatePeerConnectionFactory(
>+    rtc::Thread* network_thread,
>+    rtc::Thread* worker_thread,
>+    rtc::Thread* signaling_thread,
>+    AudioDeviceModule* default_adm,
>+    rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
>+    rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory,
>+    cricket::WebRtcVideoEncoderFactory* video_encoder_factory,
>+    cricket::WebRtcVideoDecoderFactory* video_decoder_factory,
>+    rtc::scoped_refptr<AudioMixer> audio_mixer,
>+    rtc::scoped_refptr<AudioProcessing> audio_processing,
>+    std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory,
>+    std::unique_ptr<NetworkControllerFactoryInterface>
>+        network_controller_factory) {
>+  rtc::scoped_refptr<AudioProcessing> audio_processing_use = audio_processing;
>+  if (!audio_processing_use) {
>+    audio_processing_use = AudioProcessingBuilder().Create();
>+  }
>+
>+  std::unique_ptr<cricket::MediaEngineInterface> media_engine(
>+      cricket::WebRtcMediaEngineFactory::Create(
>+          default_adm, audio_encoder_factory, audio_decoder_factory,
>+          video_encoder_factory, video_decoder_factory, audio_mixer,
>+          audio_processing_use));
>+
>+  std::unique_ptr<CallFactoryInterface> call_factory = CreateCallFactory();
>+
>+  std::unique_ptr<RtcEventLogFactoryInterface> event_log_factory =
>+      CreateRtcEventLogFactory();
>+
>+  return CreateModularPeerConnectionFactory(
>+      network_thread, worker_thread, signaling_thread, std::move(media_engine),
>+      std::move(call_factory), std::move(event_log_factory),
>+      std::move(fec_controller_factory), std::move(network_controller_factory));
>+}
>+#endif
>+
> rtc::scoped_refptr<PeerConnectionFactoryInterface> CreatePeerConnectionFactory(
>     rtc::Thread* network_thread,
>     rtc::Thread* worker_thread,
>@@ -97,6 +135,7 @@ rtc::scoped_refptr<PeerConnectionFactoryInterface> CreatePeerConnectionFactory(
>       std::move(call_factory), std::move(event_log_factory));
> }
> 
>+#if defined(USE_BUILTIN_SW_CODECS)
> rtc::scoped_refptr<PeerConnectionFactoryInterface>
> CreatePeerConnectionFactoryWithAudioMixer(
>     rtc::Thread* network_thread,
>@@ -128,5 +167,6 @@ rtc::scoped_refptr<PeerConnectionFactoryInterface> CreatePeerConnectionFactory(
>       audio_encoder_factory, audio_decoder_factory, video_encoder_factory,
>       video_decoder_factory, nullptr);
> }
>+#endif
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/currentspeakermonitor.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/currentspeakermonitor.cc
>deleted file mode 100644
>index e210a45d61b..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/currentspeakermonitor.cc
>+++ /dev/null
>@@ -1,203 +0,0 @@
>-/*
>- *  Copyright 2011 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "pc/currentspeakermonitor.h"
>-
>-#include <vector>
>-
>-#include "media/base/streamparams.h"
>-#include "pc/audiomonitor.h"
>-#include "rtc_base/logging.h"
>-
>-namespace cricket {
>-
>-namespace {
>-const int kMaxAudioLevel = 9;
>-// To avoid overswitching, we disable switching for a period of time after a
>-// switch is done.
>-const int kDefaultMinTimeBetweenSwitches = 1000;
>-}
>-
>-CurrentSpeakerMonitor::CurrentSpeakerMonitor(
>-    AudioSourceContext* audio_source_context)
>-    : started_(false),
>-      audio_source_context_(audio_source_context),
>-      current_speaker_ssrc_(0),
>-      earliest_permitted_switch_time_(0),
>-      min_time_between_switches_(kDefaultMinTimeBetweenSwitches) {}
>-
>-CurrentSpeakerMonitor::~CurrentSpeakerMonitor() {
>-  Stop();
>-}
>-
>-void CurrentSpeakerMonitor::Start() {
>-  if (!started_) {
>-    audio_source_context_->SignalAudioMonitor.connect(
>-        this, &CurrentSpeakerMonitor::OnAudioMonitor);
>-    audio_source_context_->SignalMediaStreamsUpdate.connect(
>-        this, &CurrentSpeakerMonitor::OnMediaStreamsUpdate);
>-    audio_source_context_->SignalMediaStreamsReset.connect(
>-        this, &CurrentSpeakerMonitor::OnMediaStreamsReset);
>-
>-    started_ = true;
>-  }
>-}
>-
>-void CurrentSpeakerMonitor::Stop() {
>-  if (started_) {
>-    audio_source_context_->SignalAudioMonitor.disconnect(this);
>-    audio_source_context_->SignalMediaStreamsUpdate.disconnect(this);
>-
>-    started_ = false;
>-    ssrc_to_speaking_state_map_.clear();
>-    current_speaker_ssrc_ = 0;
>-    earliest_permitted_switch_time_ = 0;
>-  }
>-}
>-
>-void CurrentSpeakerMonitor::set_min_time_between_switches(
>-    int min_time_between_switches) {
>-  min_time_between_switches_ = min_time_between_switches;
>-}
>-
>-void CurrentSpeakerMonitor::OnAudioMonitor(
>-    AudioSourceContext* audio_source_context, const AudioInfo& info) {
>-  std::map<uint32_t, int> active_ssrc_to_level_map;
>-  cricket::AudioInfo::StreamList::const_iterator stream_list_it;
>-  for (stream_list_it = info.active_streams.begin();
>-       stream_list_it != info.active_streams.end(); ++stream_list_it) {
>-    uint32_t ssrc = stream_list_it->first;
>-    active_ssrc_to_level_map[ssrc] = stream_list_it->second;
>-
>-    // It's possible we haven't yet added this source to our map.  If so,
>-    // add it now with a "not speaking" state.
>-    if (ssrc_to_speaking_state_map_.find(ssrc) ==
>-        ssrc_to_speaking_state_map_.end()) {
>-      ssrc_to_speaking_state_map_[ssrc] = SS_NOT_SPEAKING;
>-    }
>-  }
>-
>-  int max_level = 0;
>-  uint32_t loudest_speaker_ssrc = 0;
>-
>-  // Update the speaking states of all participants based on the new audio
>-  // level information.  Also retain loudest speaker.
>-  std::map<uint32_t, SpeakingState>::iterator state_it;
>-  for (state_it = ssrc_to_speaking_state_map_.begin();
>-       state_it != ssrc_to_speaking_state_map_.end(); ++state_it) {
>-    bool is_previous_speaker = current_speaker_ssrc_ == state_it->first;
>-
>-    // This uses a state machine in order to gradually identify
>-    // members as having started or stopped speaking. Matches the
>-    // algorithm used by the hangouts js code.
>-
>-    std::map<uint32_t, int>::const_iterator level_it =
>-        active_ssrc_to_level_map.find(state_it->first);
>-    // Note that the stream map only contains streams with non-zero audio
>-    // levels.
>-    int level = (level_it != active_ssrc_to_level_map.end()) ?
>-        level_it->second : 0;
>-    switch (state_it->second) {
>-      case SS_NOT_SPEAKING:
>-        if (level > 0) {
>-          // Reset level because we don't think they're really speaking.
>-          level = 0;
>-          state_it->second = SS_MIGHT_BE_SPEAKING;
>-        } else {
>-          // State unchanged.
>-        }
>-        break;
>-      case SS_MIGHT_BE_SPEAKING:
>-        if (level > 0) {
>-          state_it->second = SS_SPEAKING;
>-        } else {
>-          state_it->second = SS_NOT_SPEAKING;
>-        }
>-        break;
>-      case SS_SPEAKING:
>-        if (level > 0) {
>-          // State unchanged.
>-        } else {
>-          state_it->second = SS_WAS_SPEAKING_RECENTLY1;
>-          if (is_previous_speaker) {
>-            // Assume this is an inter-word silence and assign him the highest
>-            // volume.
>-            level = kMaxAudioLevel;
>-          }
>-        }
>-        break;
>-      case SS_WAS_SPEAKING_RECENTLY1:
>-        if (level > 0) {
>-          state_it->second = SS_SPEAKING;
>-        } else {
>-          state_it->second = SS_WAS_SPEAKING_RECENTLY2;
>-          if (is_previous_speaker) {
>-            // Assume this is an inter-word silence and assign him the highest
>-            // volume.
>-            level = kMaxAudioLevel;
>-          }
>-        }
>-        break;
>-      case SS_WAS_SPEAKING_RECENTLY2:
>-        if (level > 0) {
>-          state_it->second = SS_SPEAKING;
>-        } else {
>-          state_it->second = SS_NOT_SPEAKING;
>-        }
>-        break;
>-    }
>-
>-    if (level > max_level) {
>-      loudest_speaker_ssrc = state_it->first;
>-      max_level = level;
>-    } else if (level > 0 && level == max_level && is_previous_speaker) {
>-      // Favor continuity of loudest speakers if audio levels are equal.
>-      loudest_speaker_ssrc = state_it->first;
>-    }
>-  }
>-
>-  // We avoid over-switching by disabling switching for a period of time after
>-  // a switch is done.
>-  int64_t now = rtc::TimeMillis();
>-  if (earliest_permitted_switch_time_ <= now &&
>-      current_speaker_ssrc_ != loudest_speaker_ssrc) {
>-    current_speaker_ssrc_ = loudest_speaker_ssrc;
>-    RTC_LOG(LS_INFO) << "Current speaker changed to " << current_speaker_ssrc_;
>-    earliest_permitted_switch_time_ = now + min_time_between_switches_;
>-    SignalUpdate(this, current_speaker_ssrc_);
>-  }
>-}
>-
>-void CurrentSpeakerMonitor::OnMediaStreamsUpdate(
>-    AudioSourceContext* audio_source_context,
>-    const MediaStreams& added,
>-    const MediaStreams& removed) {
>-  if (audio_source_context == audio_source_context_) {
>-    // Update the speaking state map based on added and removed streams.
>-    for (std::vector<cricket::StreamParams>::const_iterator
>-           it = removed.audio().begin(); it != removed.audio().end(); ++it) {
>-      ssrc_to_speaking_state_map_.erase(it->first_ssrc());
>-    }
>-
>-    for (std::vector<cricket::StreamParams>::const_iterator
>-           it = added.audio().begin(); it != added.audio().end(); ++it) {
>-      ssrc_to_speaking_state_map_[it->first_ssrc()] = SS_NOT_SPEAKING;
>-    }
>-  }
>-}
>-
>-void CurrentSpeakerMonitor::OnMediaStreamsReset(
>-    AudioSourceContext* audio_source_context) {
>-  if (audio_source_context == audio_source_context_) {
>-    ssrc_to_speaking_state_map_.clear();
>-  }
>-}
>-
>-}  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/currentspeakermonitor.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/currentspeakermonitor.h
>deleted file mode 100644
>index 589fb1b1d55..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/currentspeakermonitor.h
>+++ /dev/null
>@@ -1,96 +0,0 @@
>-/*
>- *  Copyright 2011 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-// CurrentSpeakerMonitor monitors the audio levels for a session and determines
>-// which participant is currently speaking.
>-
>-#ifndef PC_CURRENTSPEAKERMONITOR_H_
>-#define PC_CURRENTSPEAKERMONITOR_H_
>-
>-#include <stdint.h>
>-
>-#include <map>
>-
>-#include "rtc_base/sigslot.h"
>-
>-namespace cricket {
>-
>-struct AudioInfo;
>-struct MediaStreams;
>-
>-class AudioSourceContext {
>- public:
>-  sigslot::signal2<AudioSourceContext*, const cricket::AudioInfo&>
>-      SignalAudioMonitor;
>-  sigslot::signal1<AudioSourceContext*> SignalMediaStreamsReset;
>-  sigslot::signal3<AudioSourceContext*,
>-                   const cricket::MediaStreams&,
>-                   const cricket::MediaStreams&> SignalMediaStreamsUpdate;
>-};
>-
>-// CurrentSpeakerMonitor can be used to monitor the audio-levels from
>-// many audio-sources and report on changes in the loudest audio-source.
>-// Its a generic type and relies on an AudioSourceContext which is aware of
>-// the audio-sources. AudioSourceContext needs to provide two signals namely
>-// SignalAudioInfoMonitor - provides audio info of the all current speakers.
>-// SignalMediaSourcesUpdated - provides updates when a speaker leaves or joins.
>-// Note that the AudioSourceContext's audio monitor must be started
>-// before this is started.
>-// It's recommended that the audio monitor be started with a 100 ms period.
>-class CurrentSpeakerMonitor : public sigslot::has_slots<> {
>- public:
>-  explicit CurrentSpeakerMonitor(AudioSourceContext* audio_source_context);
>-  ~CurrentSpeakerMonitor();
>-
>-  void Start();
>-  void Stop();
>-
>-  // Used by tests.  Note that the actual minimum time between switches
>-  // enforced by the monitor will be the given value plus or minus the
>-  // resolution of the system clock.
>-  void set_min_time_between_switches(int min_time_between_switches);
>-
>-  // This is fired when the current speaker changes, and provides his audio
>-  // SSRC.  This only fires after the audio monitor on the underlying
>-  // AudioSourceContext has been started.
>-  sigslot::signal2<CurrentSpeakerMonitor*, uint32_t> SignalUpdate;
>-
>- private:
>-  void OnAudioMonitor(AudioSourceContext* audio_source_context,
>-                      const AudioInfo& info);
>-  void OnMediaStreamsUpdate(AudioSourceContext* audio_source_context,
>-                            const MediaStreams& added,
>-                            const MediaStreams& removed);
>-  void OnMediaStreamsReset(AudioSourceContext* audio_source_context);
>-
>-  // These are states that a participant will pass through so that we gradually
>-  // recognize that they have started and stopped speaking.  This avoids
>-  // "twitchiness".
>-  enum SpeakingState {
>-    SS_NOT_SPEAKING,
>-    SS_MIGHT_BE_SPEAKING,
>-    SS_SPEAKING,
>-    SS_WAS_SPEAKING_RECENTLY1,
>-    SS_WAS_SPEAKING_RECENTLY2
>-  };
>-
>-  bool started_;
>-  AudioSourceContext* audio_source_context_;
>-  std::map<uint32_t, SpeakingState> ssrc_to_speaking_state_map_;
>-  uint32_t current_speaker_ssrc_;
>-  // To prevent overswitching, switching is disabled for some time after a
>-  // switch is made.  This gives us the earliest time a switch is permitted.
>-  int64_t earliest_permitted_switch_time_;
>-  int min_time_between_switches_;
>-};
>-
>-}  // namespace cricket
>-
>-#endif  // PC_CURRENTSPEAKERMONITOR_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/currentspeakermonitor_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/currentspeakermonitor_unittest.cc
>deleted file mode 100644
>index fa3f0cb7d92..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/currentspeakermonitor_unittest.cc
>+++ /dev/null
>@@ -1,217 +0,0 @@
>-/*
>- *  Copyright 2004 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include <utility>
>-
>-#include "pc/currentspeakermonitor.h"
>-#include "pc/audiomonitor.h"
>-#include "rtc_base/gunit.h"
>-#include "rtc_base/thread.h"
>-
>-namespace cricket {
>-
>-static const uint32_t kSsrc1 = 1001;
>-static const uint32_t kSsrc2 = 1002;
>-static const uint32_t kMinTimeBetweenSwitches = 10;
>-// Due to limited system clock resolution, the CurrentSpeakerMonitor may
>-// actually require more or less time between switches than that specified
>-// in the call to set_min_time_between_switches.  To be safe, we sleep for
>-// 90 ms more than the min time between switches before checking for a switch.
>-// I am assuming system clocks do not have a coarser resolution than 90 ms.
>-static const uint32_t kSleepTimeBetweenSwitches = 100;
>-
>-class CurrentSpeakerMonitorTest : public testing::Test,
>-    public sigslot::has_slots<> {
>- public:
>-  CurrentSpeakerMonitorTest() {
>-    monitor_ = new CurrentSpeakerMonitor(&source_);
>-    // Shrink the minimum time betweeen switches to 10 ms so we don't have to
>-    // slow down our tests.
>-    monitor_->set_min_time_between_switches(kMinTimeBetweenSwitches);
>-    monitor_->SignalUpdate.connect(this, &CurrentSpeakerMonitorTest::OnUpdate);
>-    current_speaker_ = 0;
>-    num_changes_ = 0;
>-    monitor_->Start();
>-  }
>-
>-  ~CurrentSpeakerMonitorTest() {
>-    delete monitor_;
>-  }
>-
>-  void SignalAudioMonitor(const AudioInfo& info) {
>-    source_.SignalAudioMonitor(&source_, info);
>-  }
>-
>- protected:
>-  AudioSourceContext source_;
>-  CurrentSpeakerMonitor* monitor_;
>-  int num_changes_;
>-  uint32_t current_speaker_;
>-
>-  void OnUpdate(CurrentSpeakerMonitor* monitor, uint32_t current_speaker) {
>-    current_speaker_ = current_speaker;
>-    num_changes_++;
>-  }
>-};
>-
>-static void InitAudioInfo(AudioInfo* info, int input_level, int output_level) {
>-  info->input_level = input_level;
>-  info->output_level = output_level;
>-}
>-
>-TEST_F(CurrentSpeakerMonitorTest, NoActiveStreams) {
>-  AudioInfo info;
>-  InitAudioInfo(&info, 0, 0);
>-  SignalAudioMonitor(info);
>-
>-  EXPECT_EQ(current_speaker_, 0U);
>-  EXPECT_EQ(num_changes_, 0);
>-}
>-
>-TEST_F(CurrentSpeakerMonitorTest, MultipleActiveStreams) {
>-  AudioInfo info;
>-  InitAudioInfo(&info, 0, 0);
>-
>-  info.active_streams.push_back(std::make_pair(kSsrc1, 3));
>-  info.active_streams.push_back(std::make_pair(kSsrc2, 7));
>-  SignalAudioMonitor(info);
>-
>-  // No speaker recognized because the initial sample is treated as possibly
>-  // just noise and disregarded.
>-  EXPECT_EQ(current_speaker_, 0U);
>-  EXPECT_EQ(num_changes_, 0);
>-
>-  info.active_streams.push_back(std::make_pair(kSsrc1, 3));
>-  info.active_streams.push_back(std::make_pair(kSsrc2, 7));
>-  SignalAudioMonitor(info);
>-
>-  EXPECT_EQ(current_speaker_, kSsrc2);
>-  EXPECT_EQ(num_changes_, 1);
>-}
>-
>-// See: https://code.google.com/p/webrtc/issues/detail?id=2409
>-TEST_F(CurrentSpeakerMonitorTest, DISABLED_RapidSpeakerChange) {
>-  AudioInfo info;
>-  InitAudioInfo(&info, 0, 0);
>-
>-  info.active_streams.push_back(std::make_pair(kSsrc1, 3));
>-  info.active_streams.push_back(std::make_pair(kSsrc2, 7));
>-  SignalAudioMonitor(info);
>-
>-  EXPECT_EQ(current_speaker_, 0U);
>-  EXPECT_EQ(num_changes_, 0);
>-
>-  info.active_streams.push_back(std::make_pair(kSsrc1, 3));
>-  info.active_streams.push_back(std::make_pair(kSsrc2, 7));
>-  SignalAudioMonitor(info);
>-
>-  EXPECT_EQ(current_speaker_, kSsrc2);
>-  EXPECT_EQ(num_changes_, 1);
>-
>-  info.active_streams.push_back(std::make_pair(kSsrc1, 9));
>-  info.active_streams.push_back(std::make_pair(kSsrc2, 1));
>-  SignalAudioMonitor(info);
>-
>-  // We expect no speaker change because of the rapid change.
>-  EXPECT_EQ(current_speaker_, kSsrc2);
>-  EXPECT_EQ(num_changes_, 1);
>-}
>-
>-// Flaky on iOS: webrtc:7057.
>-#if defined(WEBRTC_IOS)
>-#define MAYBE_SpeakerChange DISABLED_SpeakerChange
>-#else
>-#define MAYBE_SpeakerChange SpeakerChange
>-#endif
>-TEST_F(CurrentSpeakerMonitorTest, MAYBE_SpeakerChange) {
>-  AudioInfo info;
>-  InitAudioInfo(&info, 0, 0);
>-
>-  info.active_streams.push_back(std::make_pair(kSsrc1, 3));
>-  info.active_streams.push_back(std::make_pair(kSsrc2, 7));
>-  SignalAudioMonitor(info);
>-
>-  EXPECT_EQ(current_speaker_, 0U);
>-  EXPECT_EQ(num_changes_, 0);
>-
>-  info.active_streams.push_back(std::make_pair(kSsrc1, 3));
>-  info.active_streams.push_back(std::make_pair(kSsrc2, 7));
>-  SignalAudioMonitor(info);
>-
>-  EXPECT_EQ(current_speaker_, kSsrc2);
>-  EXPECT_EQ(num_changes_, 1);
>-
>-  // Wait so the changes don't come so rapidly.
>-  rtc::Thread::SleepMs(kSleepTimeBetweenSwitches);
>-
>-  info.active_streams.push_back(std::make_pair(kSsrc1, 9));
>-  info.active_streams.push_back(std::make_pair(kSsrc2, 1));
>-  SignalAudioMonitor(info);
>-
>-  EXPECT_EQ(current_speaker_, kSsrc1);
>-  EXPECT_EQ(num_changes_, 2);
>-}
>-
>-TEST_F(CurrentSpeakerMonitorTest, InterwordSilence) {
>-  AudioInfo info;
>-  InitAudioInfo(&info, 0, 0);
>-
>-  info.active_streams.push_back(std::make_pair(kSsrc1, 3));
>-  info.active_streams.push_back(std::make_pair(kSsrc2, 7));
>-  SignalAudioMonitor(info);
>-
>-  EXPECT_EQ(current_speaker_, 0U);
>-  EXPECT_EQ(num_changes_, 0);
>-
>-  info.active_streams.push_back(std::make_pair(kSsrc1, 3));
>-  info.active_streams.push_back(std::make_pair(kSsrc2, 7));
>-  SignalAudioMonitor(info);
>-
>-  EXPECT_EQ(current_speaker_, kSsrc2);
>-  EXPECT_EQ(num_changes_, 1);
>-
>-  info.active_streams.push_back(std::make_pair(kSsrc1, 3));
>-  info.active_streams.push_back(std::make_pair(kSsrc2, 7));
>-  SignalAudioMonitor(info);
>-
>-  EXPECT_EQ(current_speaker_, kSsrc2);
>-  EXPECT_EQ(num_changes_, 1);
>-
>-  // Wait so the changes don't come so rapidly.
>-  rtc::Thread::SleepMs(kSleepTimeBetweenSwitches);
>-
>-  info.active_streams.push_back(std::make_pair(kSsrc1, 3));
>-  info.active_streams.push_back(std::make_pair(kSsrc2, 0));
>-  SignalAudioMonitor(info);
>-
>-  // Current speaker shouldn't have changed because we treat this as an inter-
>-  // word silence.
>-  EXPECT_EQ(current_speaker_, kSsrc2);
>-  EXPECT_EQ(num_changes_, 1);
>-
>-  info.active_streams.push_back(std::make_pair(kSsrc1, 3));
>-  info.active_streams.push_back(std::make_pair(kSsrc2, 0));
>-  SignalAudioMonitor(info);
>-
>-  // Current speaker shouldn't have changed because we treat this as an inter-
>-  // word silence.
>-  EXPECT_EQ(current_speaker_, kSsrc2);
>-  EXPECT_EQ(num_changes_, 1);
>-
>-  info.active_streams.push_back(std::make_pair(kSsrc1, 3));
>-  info.active_streams.push_back(std::make_pair(kSsrc2, 0));
>-  SignalAudioMonitor(info);
>-
>-  // At this point, we should have concluded that SSRC2 stopped speaking.
>-  EXPECT_EQ(current_speaker_, kSsrc1);
>-  EXPECT_EQ(num_changes_, 2);
>-}
>-
>-}  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/datachannel.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/datachannel.cc
>index 950fbeaea8b..6971ca8f66c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/datachannel.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/datachannel.cc
>@@ -122,10 +122,9 @@ rtc::scoped_refptr<DataChannel> DataChannel::Create(
>   return channel;
> }
> 
>-DataChannel::DataChannel(
>-    DataChannelProviderInterface* provider,
>-    cricket::DataChannelType dct,
>-    const std::string& label)
>+DataChannel::DataChannel(DataChannelProviderInterface* provider,
>+                         cricket::DataChannelType dct,
>+                         const std::string& label)
>     : label_(label),
>       observer_(nullptr),
>       state_(kConnecting),
>@@ -141,26 +140,22 @@ DataChannel::DataChannel(
>       receive_ssrc_set_(false),
>       writable_(false),
>       send_ssrc_(0),
>-      receive_ssrc_(0) {
>-}
>+      receive_ssrc_(0) {}
> 
> bool DataChannel::Init(const InternalDataChannelInit& config) {
>   if (data_channel_type_ == cricket::DCT_RTP) {
>-    if (config.reliable ||
>-        config.id != -1 ||
>-        config.maxRetransmits != -1 ||
>+    if (config.reliable || config.id != -1 || config.maxRetransmits != -1 ||
>         config.maxRetransmitTime != -1) {
>       RTC_LOG(LS_ERROR) << "Failed to initialize the RTP data channel due to "
>-                        << "invalid DataChannelInit.";
>+                           "invalid DataChannelInit.";
>       return false;
>     }
>     handshake_state_ = kHandshakeReady;
>   } else if (data_channel_type_ == cricket::DCT_SCTP) {
>-    if (config.id < -1 ||
>-        config.maxRetransmits < -1 ||
>+    if (config.id < -1 || config.maxRetransmits < -1 ||
>         config.maxRetransmitTime < -1) {
>       RTC_LOG(LS_ERROR) << "Failed to initialize the SCTP data channel due to "
>-                        << "invalid DataChannelInit.";
>+                           "invalid DataChannelInit.";
>       return false;
>     }
>     if (config.maxRetransmits != -1 && config.maxRetransmitTime != -1) {
>@@ -171,15 +166,15 @@ bool DataChannel::Init(const InternalDataChannelInit& config) {
>     config_ = config;
> 
>     switch (config_.open_handshake_role) {
>-    case webrtc::InternalDataChannelInit::kNone:  // pre-negotiated
>-      handshake_state_ = kHandshakeReady;
>-      break;
>-    case webrtc::InternalDataChannelInit::kOpener:
>-      handshake_state_ = kHandshakeShouldSendOpen;
>-      break;
>-    case webrtc::InternalDataChannelInit::kAcker:
>-      handshake_state_ = kHandshakeShouldSendAck;
>-      break;
>+      case webrtc::InternalDataChannelInit::kNone:  // pre-negotiated
>+        handshake_state_ = kHandshakeReady;
>+        break;
>+      case webrtc::InternalDataChannelInit::kOpener:
>+        handshake_state_ = kHandshakeShouldSendOpen;
>+        break;
>+      case webrtc::InternalDataChannelInit::kAcker:
>+        handshake_state_ = kHandshakeShouldSendAck;
>+        break;
>     }
> 
>     // Try to connect to the transport in case the transport channel already
>@@ -214,8 +209,7 @@ bool DataChannel::reliable() const {
>   if (data_channel_type_ == cricket::DCT_RTP) {
>     return false;
>   } else {
>-    return config_.maxRetransmits == -1 &&
>-           config_.maxRetransmitTime == -1;
>+    return config_.maxRetransmits == -1 && config_.maxRetransmitTime == -1;
>   }
> }
> 
>@@ -229,6 +223,7 @@ void DataChannel::Close() {
>   send_ssrc_ = 0;
>   send_ssrc_set_ = false;
>   SetState(kClosing);
>+  // Will send queued data before beginning the underlying closing procedure.
>   UpdateState();
> }
> 
>@@ -251,7 +246,9 @@ bool DataChannel::Send(const DataBuffer& buffer) {
>     // blocked.
>     RTC_DCHECK(data_channel_type_ == cricket::DCT_SCTP);
>     if (!QueueSendDataMessage(buffer)) {
>-      Close();
>+      RTC_LOG(LS_ERROR) << "Closing the DataChannel due to a failure to queue "
>+                           "additional data.";
>+      CloseAbruptly();
>     }
>     return true;
>   }
>@@ -276,11 +273,6 @@ void DataChannel::SetReceiveSsrc(uint32_t receive_ssrc) {
>   UpdateState();
> }
> 
>-// The remote peer request that this channel shall be closed.
>-void DataChannel::RemotePeerRequestClose() {
>-  DoClose();
>-}
>-
> void DataChannel::SetSctpSid(int sid) {
>   RTC_DCHECK_LT(config_.id, 0);
>   RTC_DCHECK_GE(sid, 0);
>@@ -293,6 +285,33 @@ void DataChannel::SetSctpSid(int sid) {
>   provider_->AddSctpDataStream(sid);
> }
> 
>+void DataChannel::OnClosingProcedureStartedRemotely(int sid) {
>+  if (data_channel_type_ == cricket::DCT_SCTP && sid == config_.id &&
>+      state_ != kClosing && state_ != kClosed) {
>+    // Don't bother sending queued data since the side that initiated the
>+    // closure wouldn't receive it anyway. See crbug.com/559394 for a lengthy
>+    // discussion about this.
>+    queued_send_data_.Clear();
>+    queued_control_data_.Clear();
>+    // Just need to change state to kClosing, SctpTransport will handle the
>+    // rest of the closing procedure and OnClosingProcedureComplete will be
>+    // called later.
>+    started_closing_procedure_ = true;
>+    SetState(kClosing);
>+  }
>+}
>+
>+void DataChannel::OnClosingProcedureComplete(int sid) {
>+  if (data_channel_type_ == cricket::DCT_SCTP && sid == config_.id) {
>+    // If the closing procedure is complete, we should have finished sending
>+    // all pending data and transitioned to kClosing already.
>+    RTC_DCHECK_EQ(state_, kClosing);
>+    RTC_DCHECK(queued_send_data_.Empty());
>+    DisconnectFromProvider();
>+    SetState(kClosed);
>+  }
>+}
>+
> void DataChannel::OnTransportChannelCreated() {
>   RTC_DCHECK(data_channel_type_ == cricket::DCT_SCTP);
>   if (!connected_to_provider_) {
>@@ -306,11 +325,15 @@ void DataChannel::OnTransportChannelCreated() {
> }
> 
> void DataChannel::OnTransportChannelDestroyed() {
>-  // This method needs to synchronously close the data channel, which means any
>-  // queued data needs to be discarded.
>-  queued_send_data_.Clear();
>-  queued_control_data_.Clear();
>-  DoClose();
>+  // The SctpTransport is going away (for example, because the SCTP m= section
>+  // was rejected), so we need to close abruptly.
>+  CloseAbruptly();
>+}
>+
>+// The remote peer request that this channel shall be closed.
>+void DataChannel::RemotePeerRequestClose() {
>+  RTC_DCHECK(data_channel_type_ == cricket::DCT_RTP);
>+  CloseAbruptly();
> }
> 
> void DataChannel::SetSendSsrc(uint32_t send_ssrc) {
>@@ -344,8 +367,9 @@ void DataChannel::OnDataReceived(const cricket::ReceiveDataParams& params,
>     RTC_DCHECK(data_channel_type_ == cricket::DCT_SCTP);
>     if (handshake_state_ != kHandshakeWaitingForAck) {
>       // Ignore it if we are not expecting an ACK message.
>-      RTC_LOG(LS_WARNING) << "DataChannel received unexpected CONTROL message, "
>-                          << "sid = " << params.sid;
>+      RTC_LOG(LS_WARNING)
>+          << "DataChannel received unexpected CONTROL message, sid = "
>+          << params.sid;
>       return;
>     }
>     if (ParseDataChannelOpenAckMessage(payload)) {
>@@ -386,7 +410,7 @@ void DataChannel::OnDataReceived(const cricket::ReceiveDataParams& params,
> 
>       queued_received_data_.Clear();
>       if (data_channel_type_ != cricket::DCT_RTP) {
>-        Close();
>+        CloseAbruptly();
>       }
> 
>       return;
>@@ -395,12 +419,6 @@ void DataChannel::OnDataReceived(const cricket::ReceiveDataParams& params,
>   }
> }
> 
>-void DataChannel::OnStreamClosedRemotely(int sid) {
>-  if (data_channel_type_ == cricket::DCT_SCTP && sid == config_.id) {
>-    Close();
>-  }
>-}
>-
> void DataChannel::OnChannelReady(bool writable) {
>   writable_ = writable;
>   if (!writable) {
>@@ -412,14 +430,23 @@ void DataChannel::OnChannelReady(bool writable) {
>   UpdateState();
> }
> 
>-void DataChannel::DoClose() {
>-  if (state_ == kClosed)
>+void DataChannel::CloseAbruptly() {
>+  if (state_ == kClosed) {
>     return;
>+  }
> 
>-  receive_ssrc_set_ = false;
>-  send_ssrc_set_ = false;
>+  if (connected_to_provider_) {
>+    DisconnectFromProvider();
>+  }
>+
>+  // Closing abruptly means any queued data gets thrown away.
>+  queued_send_data_.Clear();
>+  queued_control_data_.Clear();
>+
>+  // Still go to "kClosing" before "kClosed", since observers may be expecting
>+  // that.
>   SetState(kClosing);
>-  UpdateState();
>+  SetState(kClosed);
> }
> 
> void DataChannel::UpdateState() {
>@@ -443,9 +470,8 @@ void DataChannel::UpdateState() {
>             WriteDataChannelOpenAckMessage(&payload);
>             SendControlMessage(payload);
>           }
>-          if (writable_ &&
>-              (handshake_state_ == kHandshakeReady ||
>-               handshake_state_ == kHandshakeWaitingForAck)) {
>+          if (writable_ && (handshake_state_ == kHandshakeReady ||
>+                            handshake_state_ == kHandshakeWaitingForAck)) {
>             SetState(kOpen);
>             // If we have received buffers before the channel got writable.
>             // Deliver them now.
>@@ -459,13 +485,28 @@ void DataChannel::UpdateState() {
>       break;
>     }
>     case kClosing: {
>+      // Wait for all queued data to be sent before beginning the closing
>+      // procedure.
>       if (queued_send_data_.Empty() && queued_control_data_.Empty()) {
>-        if (connected_to_provider_) {
>-          DisconnectFromProvider();
>-        }
>-
>-        if (!connected_to_provider_ && !send_ssrc_set_ && !receive_ssrc_set_) {
>-          SetState(kClosed);
>+        if (data_channel_type_ == cricket::DCT_RTP) {
>+          // For RTP data channels, we can go to "closed" after we finish
>+          // sending data and the send/recv SSRCs are unset.
>+          if (connected_to_provider_) {
>+            DisconnectFromProvider();
>+          }
>+          if (!send_ssrc_set_ && !receive_ssrc_set_) {
>+            SetState(kClosed);
>+          }
>+        } else {
>+          // For SCTP data channels, we need to wait for the closing procedure
>+          // to complete; after calling RemoveSctpDataStream,
>+          // OnClosingProcedureComplete will end up called asynchronously
>+          // afterwards.
>+          if (connected_to_provider_ && !started_closing_procedure_ &&
>+              config_.id >= 0) {
>+            started_closing_procedure_ = true;
>+            provider_->RemoveSctpDataStream(config_.id);
>+          }
>         }
>       }
>       break;
>@@ -497,10 +538,6 @@ void DataChannel::DisconnectFromProvider() {
> 
>   provider_->DisconnectDataChannel(this);
>   connected_to_provider_ = false;
>-
>-  if (data_channel_type_ == cricket::DCT_SCTP && config_.id >= 0) {
>-    provider_->RemoveSctpDataStream(config_.id);
>-  }
> }
> 
> void DataChannel::DeliverQueuedReceivedData() {
>@@ -551,7 +588,7 @@ bool DataChannel::SendDataMessage(const DataBuffer& buffer,
>       send_params.ordered = true;
>       RTC_LOG(LS_VERBOSE)
>           << "Sending data as ordered for unordered DataChannel "
>-          << "because the OPEN_ACK message has not been received.";
>+             "because the OPEN_ACK message has not been received.";
>     }
> 
>     send_params.max_rtx_count = config_.maxRetransmits;
>@@ -583,8 +620,9 @@ bool DataChannel::SendDataMessage(const DataBuffer& buffer,
>   // Close the channel if the error is not SDR_BLOCK, or if queuing the
>   // message failed.
>   RTC_LOG(LS_ERROR) << "Closing the DataChannel due to a failure to send data, "
>-                    << "send_result = " << send_result;
>-  Close();
>+                       "send_result = "
>+                    << send_result;
>+  CloseAbruptly();
> 
>   return false;
> }
>@@ -649,8 +687,9 @@ bool DataChannel::SendControlMessage(const rtc::CopyOnWriteBuffer& buffer) {
>     QueueControlMessage(buffer);
>   } else {
>     RTC_LOG(LS_ERROR) << "Closing the DataChannel due to a failure to send"
>-                      << " the CONTROL message, send_result = " << send_result;
>-    Close();
>+                         " the CONTROL message, send_result = "
>+                      << send_result;
>+    CloseAbruptly();
>   }
>   return retval;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/datachannel.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/datachannel.h
>index 1af90dab6e3..d29ee59ce39 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/datachannel.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/datachannel.h
>@@ -21,12 +21,15 @@
> #include "pc/channel.h"
> #include "rtc_base/messagehandler.h"
> #include "rtc_base/scoped_ref_ptr.h"
>-#include "rtc_base/sigslot.h"
>+#include "rtc_base/third_party/sigslot/sigslot.h"
> 
> namespace webrtc {
> 
> class DataChannel;
> 
>+// TODO(deadbeef): Once RTP data channels go away, get rid of this and have
>+// DataChannel depend on SctpTransportInternal (pure virtual SctpTransport
>+// interface) instead.
> class DataChannelProviderInterface {
>  public:
>   // Sends the data to the transport.
>@@ -39,7 +42,8 @@ class DataChannelProviderInterface {
>   virtual void DisconnectDataChannel(DataChannel* data_channel) = 0;
>   // Adds the data channel SID to the transport for SCTP.
>   virtual void AddSctpDataStream(int sid) = 0;
>-  // Removes the data channel SID from the transport for SCTP.
>+  // Begins the closing procedure by sending an outgoing stream reset. Still
>+  // need to wait for callbacks to tell when this completes.
>   virtual void RemoveSctpDataStream(int sid) = 0;
>   // Returns true if the transport channel is ready to send data.
>   virtual bool ReadyToSendData() const = 0;
>@@ -49,11 +53,7 @@ class DataChannelProviderInterface {
> };
> 
> struct InternalDataChannelInit : public DataChannelInit {
>-  enum OpenHandshakeRole {
>-    kOpener,
>-    kAcker,
>-    kNone
>-  };
>+  enum OpenHandshakeRole { kOpener, kAcker, kNone };
>   // The default role is kOpener because the default |negotiated| is false.
>   InternalDataChannelInit() : open_handshake_role(kOpener) {}
>   explicit InternalDataChannelInit(const DataChannelInit& base)
>@@ -73,7 +73,7 @@ class SctpSidAllocator {
>   // Gets the first unused odd/even id based on the DTLS role. If |role| is
>   // SSL_CLIENT, the allocated id starts from 0 and takes even numbers;
>   // otherwise, the id starts from 1 and takes odd numbers.
>-  // Returns false if no id can be allocated.
>+  // Returns false if no ID can be allocated.
>   bool AllocateSid(rtc::SSLRole role, int* sid);
> 
>   // Attempts to reserve a specific sid. Returns false if it's unavailable.
>@@ -92,7 +92,7 @@ class SctpSidAllocator {
> // DataChannel is a an implementation of the DataChannelInterface based on
> // libjingle's data engine. It provides an implementation of unreliable or
> // reliabledata channels. Currently this class is specifically designed to use
>-// both RtpDataEngine and SctpDataEngine.
>+// both RtpDataChannel and SctpTransport.
> 
> // DataChannel states:
> // kConnecting: The channel has been created the transport might not yet be
>@@ -104,6 +104,16 @@ class SctpSidAllocator {
> //           has been called with SSRC==0
> // kClosed: Both UpdateReceiveSsrc and UpdateSendSsrc has been called with
> //          SSRC==0.
>+//
>+// How the closing procedure works for SCTP:
>+// 1. Alice calls Close(), state changes to kClosing.
>+// 2. Alice finishes sending any queued data.
>+// 3. Alice calls RemoveSctpDataStream, sends outgoing stream reset.
>+// 4. Bob receives incoming stream reset; OnClosingProcedureStartedRemotely
>+//    called.
>+// 5. Bob sends outgoing stream reset. 6. Alice receives incoming reset,
>+//    Bob receives acknowledgement. Both receive OnClosingProcedureComplete
>+//    callback and transition to kClosed.
> class DataChannel : public DataChannelInterface,
>                     public sigslot::has_slots<>,
>                     public rtc::MessageHandler {
>@@ -147,16 +157,21 @@ class DataChannel : public DataChannelInterface,
>   // Slots for provider to connect signals to.
>   void OnDataReceived(const cricket::ReceiveDataParams& params,
>                       const rtc::CopyOnWriteBuffer& payload);
>-  void OnStreamClosedRemotely(int sid);
>-
>-  // The remote peer request that this channel should be closed.
>-  void RemotePeerRequestClose();
> 
>-  // The following methods are for SCTP only.
>+  /********************************************
>+   * The following methods are for SCTP only. *
>+   ********************************************/
> 
>   // Sets the SCTP sid and adds to transport layer if not set yet. Should only
>   // be called once.
>   void SetSctpSid(int sid);
>+  // The remote side started the closing procedure by resetting its outgoing
>+  // stream (our incoming stream). Sets state to kClosing.
>+  void OnClosingProcedureStartedRemotely(int sid);
>+  // The closing procedure is complete; both incoming and outgoing stream
>+  // resets are done and the channel can transition to kClosed. Called
>+  // asynchronously after RemoveSctpDataStream.
>+  void OnClosingProcedureComplete(int sid);
>   // Called when the transport channel is created.
>   // Only needs to be called for SCTP data channels.
>   void OnTransportChannelCreated();
>@@ -165,8 +180,12 @@ class DataChannel : public DataChannelInterface,
>   // to kClosed.
>   void OnTransportChannelDestroyed();
> 
>-  // The following methods are for RTP only.
>+  /*******************************************
>+   * The following methods are for RTP only. *
>+   *******************************************/
> 
>+  // The remote peer requested that this channel should be closed.
>+  void RemotePeerRequestClose();
>   // Set the SSRC this channel should use to send data on the
>   // underlying data engine. |send_ssrc| == 0 means that the channel is no
>   // longer part of the session negotiation.
>@@ -200,9 +219,7 @@ class DataChannel : public DataChannelInterface,
>     PacketQueue();
>     ~PacketQueue();
> 
>-    size_t byte_count() const {
>-      return byte_count_;
>-    }
>+    size_t byte_count() const { return byte_count_; }
> 
>     bool Empty() const;
> 
>@@ -231,7 +248,11 @@ class DataChannel : public DataChannelInterface,
>   };
> 
>   bool Init(const InternalDataChannelInit& config);
>-  void DoClose();
>+  // Close immediately, ignoring any queued data or closing procedure.
>+  // This is called for RTP data channels when SDP indicates a channel should
>+  // be removed, or SCTP data channels when the underlying SctpTransport is
>+  // being destroyed.
>+  void CloseAbruptly();
>   void UpdateState();
>   void SetState(DataState state);
>   void DisconnectFromProvider();
>@@ -261,6 +282,8 @@ class DataChannel : public DataChannelInterface,
>   bool send_ssrc_set_;
>   bool receive_ssrc_set_;
>   bool writable_;
>+  // Did we already start the graceful SCTP closing procedure?
>+  bool started_closing_procedure_ = false;
>   uint32_t send_ssrc_;
>   uint32_t receive_ssrc_;
>   // Control messages that always have to get sent out before any queued
>@@ -272,25 +295,25 @@ class DataChannel : public DataChannelInterface,
> 
> // Define proxy for DataChannelInterface.
> BEGIN_SIGNALING_PROXY_MAP(DataChannel)
>-  PROXY_SIGNALING_THREAD_DESTRUCTOR()
>-  PROXY_METHOD1(void, RegisterObserver, DataChannelObserver*)
>-  PROXY_METHOD0(void, UnregisterObserver)
>-  PROXY_CONSTMETHOD0(std::string, label)
>-  PROXY_CONSTMETHOD0(bool, reliable)
>-  PROXY_CONSTMETHOD0(bool, ordered)
>-  PROXY_CONSTMETHOD0(uint16_t, maxRetransmitTime)
>-  PROXY_CONSTMETHOD0(uint16_t, maxRetransmits)
>-  PROXY_CONSTMETHOD0(std::string, protocol)
>-  PROXY_CONSTMETHOD0(bool, negotiated)
>-  PROXY_CONSTMETHOD0(int, id)
>-  PROXY_CONSTMETHOD0(DataState, state)
>-  PROXY_CONSTMETHOD0(uint32_t, messages_sent)
>-  PROXY_CONSTMETHOD0(uint64_t, bytes_sent)
>-  PROXY_CONSTMETHOD0(uint32_t, messages_received)
>-  PROXY_CONSTMETHOD0(uint64_t, bytes_received)
>-  PROXY_CONSTMETHOD0(uint64_t, buffered_amount)
>-  PROXY_METHOD0(void, Close)
>-  PROXY_METHOD1(bool, Send, const DataBuffer&)
>+PROXY_SIGNALING_THREAD_DESTRUCTOR()
>+PROXY_METHOD1(void, RegisterObserver, DataChannelObserver*)
>+PROXY_METHOD0(void, UnregisterObserver)
>+PROXY_CONSTMETHOD0(std::string, label)
>+PROXY_CONSTMETHOD0(bool, reliable)
>+PROXY_CONSTMETHOD0(bool, ordered)
>+PROXY_CONSTMETHOD0(uint16_t, maxRetransmitTime)
>+PROXY_CONSTMETHOD0(uint16_t, maxRetransmits)
>+PROXY_CONSTMETHOD0(std::string, protocol)
>+PROXY_CONSTMETHOD0(bool, negotiated)
>+PROXY_CONSTMETHOD0(int, id)
>+PROXY_CONSTMETHOD0(DataState, state)
>+PROXY_CONSTMETHOD0(uint32_t, messages_sent)
>+PROXY_CONSTMETHOD0(uint64_t, bytes_sent)
>+PROXY_CONSTMETHOD0(uint32_t, messages_received)
>+PROXY_CONSTMETHOD0(uint64_t, bytes_received)
>+PROXY_CONSTMETHOD0(uint64_t, buffered_amount)
>+PROXY_METHOD0(void, Close)
>+PROXY_METHOD1(bool, Send, const DataBuffer&)
> END_PROXY_MAP()
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/datachannel_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/datachannel_unittest.cc
>index f8e98808d55..eb5deb93cec 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/datachannel_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/datachannel_unittest.cc
>@@ -15,6 +15,7 @@
> #include "pc/sctputils.h"
> #include "pc/test/fakedatachannelprovider.h"
> #include "rtc_base/gunit.h"
>+#include "rtc_base/numerics/safe_conversions.h"
> 
> using webrtc::DataChannel;
> using webrtc::SctpSidAllocator;
>@@ -28,33 +29,23 @@ class FakeDataChannelObserver : public webrtc::DataChannelObserver {
>         on_state_change_count_(0),
>         on_buffered_amount_change_count_(0) {}
> 
>-  void OnStateChange() {
>-    ++on_state_change_count_;
>-  }
>+  void OnStateChange() { ++on_state_change_count_; }
> 
>   void OnBufferedAmountChange(uint64_t previous_amount) {
>     ++on_buffered_amount_change_count_;
>   }
> 
>-  void OnMessage(const webrtc::DataBuffer& buffer) {
>-    ++messages_received_;
>-  }
>+  void OnMessage(const webrtc::DataBuffer& buffer) { ++messages_received_; }
> 
>-  size_t messages_received() const {
>-    return messages_received_;
>-  }
>+  size_t messages_received() const { return messages_received_; }
> 
>-  void ResetOnStateChangeCount() {
>-    on_state_change_count_ = 0;
>-  }
>+  void ResetOnStateChangeCount() { on_state_change_count_ = 0; }
> 
>   void ResetOnBufferedAmountChangeCount() {
>     on_buffered_amount_change_count_ = 0;
>   }
> 
>-  size_t on_state_change_count() const {
>-    return on_state_change_count_;
>-  }
>+  size_t on_state_change_count() const { return on_state_change_count_; }
> 
>   size_t on_buffered_amount_change_count() const {
>     return on_buffered_amount_change_count_;
>@@ -66,6 +57,9 @@ class FakeDataChannelObserver : public webrtc::DataChannelObserver {
>   size_t on_buffered_amount_change_count_;
> };
> 
>+// TODO(deadbeef): The fact that these tests use a fake provider makes them not
>+// too valuable. Should rewrite using the
>+// peerconnection_datachannel_unittest.cc infrastructure.
> class SctpDataChannelTest : public testing::Test {
>  protected:
>   SctpDataChannelTest()
>@@ -100,13 +94,9 @@ class StateSignalsListener : public sigslot::has_slots<> {
>   int opened_count() const { return opened_count_; }
>   int closed_count() const { return closed_count_; }
> 
>-  void OnSignalOpened(DataChannel* data_channel) {
>-    ++opened_count_;
>-  }
>+  void OnSignalOpened(DataChannel* data_channel) { ++opened_count_; }
> 
>-  void OnSignalClosed(DataChannel* data_channel) {
>-    ++closed_count_;
>-  }
>+  void OnSignalClosed(DataChannel* data_channel) { ++closed_count_; }
> 
>  private:
>   int opened_count_ = 0;
>@@ -183,7 +173,8 @@ TEST_F(SctpDataChannelTest, BufferedAmountWhenBlocked) {
>   }
>   EXPECT_EQ(buffer.data.size() * number_of_packets,
>             webrtc_data_channel_->buffered_amount());
>-  EXPECT_EQ(number_of_packets, observer_->on_buffered_amount_change_count());
>+  EXPECT_EQ(rtc::checked_cast<size_t>(number_of_packets),
>+            observer_->on_buffered_amount_change_count());
> }
> 
> // Tests that the queued data are sent when the channel transitions from blocked
>@@ -231,12 +222,10 @@ TEST_F(SctpDataChannelTest, VerifyMessagesAndBytesSent) {
>   AddObserver();
>   SetChannelReady();
>   std::vector<webrtc::DataBuffer> buffers({
>-    webrtc::DataBuffer("message 1"),
>-    webrtc::DataBuffer("msg 2"),
>-    webrtc::DataBuffer("message three"),
>-    webrtc::DataBuffer("quadra message"),
>-    webrtc::DataBuffer("fifthmsg"),
>-    webrtc::DataBuffer("message of the beast"),
>+      webrtc::DataBuffer("message 1"), webrtc::DataBuffer("msg 2"),
>+      webrtc::DataBuffer("message three"), webrtc::DataBuffer("quadra message"),
>+      webrtc::DataBuffer("fifthmsg"),
>+      webrtc::DataBuffer("message of the beast"),
>   });
> 
>   // Default values.
>@@ -303,8 +292,7 @@ TEST_F(SctpDataChannelTest, LateCreatedChannelTransitionToOpen) {
>   rtc::scoped_refptr<DataChannel> dc =
>       DataChannel::Create(provider_.get(), cricket::DCT_SCTP, "test1", init);
>   EXPECT_EQ(webrtc::DataChannelInterface::kConnecting, dc->state());
>-  EXPECT_TRUE_WAIT(webrtc::DataChannelInterface::kOpen == dc->state(),
>-                   1000);
>+  EXPECT_TRUE_WAIT(webrtc::DataChannelInterface::kOpen == dc->state(), 1000);
> }
> 
> // Tests that an unordered DataChannel sends data as ordered until the OPEN_ACK
>@@ -456,12 +444,10 @@ TEST_F(SctpDataChannelTest, NoMsgSentIfNegotiatedAndNotFromOpenMsg) {
> TEST_F(SctpDataChannelTest, VerifyMessagesAndBytesReceived) {
>   AddObserver();
>   std::vector<webrtc::DataBuffer> buffers({
>-    webrtc::DataBuffer("message 1"),
>-    webrtc::DataBuffer("msg 2"),
>-    webrtc::DataBuffer("message three"),
>-    webrtc::DataBuffer("quadra message"),
>-    webrtc::DataBuffer("fifthmsg"),
>-    webrtc::DataBuffer("message of the beast"),
>+      webrtc::DataBuffer("message 1"), webrtc::DataBuffer("msg 2"),
>+      webrtc::DataBuffer("message three"), webrtc::DataBuffer("quadra message"),
>+      webrtc::DataBuffer("fifthmsg"),
>+      webrtc::DataBuffer("message of the beast"),
>   });
> 
>   webrtc_data_channel_->SetSctpSid(1);
>@@ -560,29 +546,6 @@ TEST_F(SctpDataChannelTest, ClosedOnTransportError) {
>             webrtc_data_channel_->state());
> }
> 
>-// Tests that a already closed DataChannel does not fire onStateChange again.
>-TEST_F(SctpDataChannelTest, ClosedDataChannelDoesNotFireOnStateChange) {
>-  AddObserver();
>-  webrtc_data_channel_->Close();
>-  // OnStateChange called for kClosing and kClosed.
>-  EXPECT_EQ(2U, observer_->on_state_change_count());
>-
>-  observer_->ResetOnStateChangeCount();
>-  webrtc_data_channel_->RemotePeerRequestClose();
>-  EXPECT_EQ(0U, observer_->on_state_change_count());
>-}
>-
>-// Tests that RemotePeerRequestClose closes the local DataChannel.
>-TEST_F(SctpDataChannelTest, RemotePeerRequestClose) {
>-  AddObserver();
>-  webrtc_data_channel_->RemotePeerRequestClose();
>-
>-  // OnStateChange called for kClosing and kClosed.
>-  EXPECT_EQ(2U, observer_->on_state_change_count());
>-  EXPECT_EQ(webrtc::DataChannelInterface::kClosed,
>-            webrtc_data_channel_->state());
>-}
>-
> // Tests that the DataChannel is closed if the received buffer is full.
> TEST_F(SctpDataChannelTest, ClosedWhenReceivedBufferFull) {
>   SetChannelReady();
>@@ -604,13 +567,11 @@ TEST_F(SctpDataChannelTest, ClosedWhenReceivedBufferFull) {
> TEST_F(SctpDataChannelTest, SendEmptyData) {
>   webrtc_data_channel_->SetSctpSid(1);
>   SetChannelReady();
>-  EXPECT_EQ(webrtc::DataChannelInterface::kOpen,
>-            webrtc_data_channel_->state());
>+  EXPECT_EQ(webrtc::DataChannelInterface::kOpen, webrtc_data_channel_->state());
> 
>   webrtc::DataBuffer buffer("");
>   EXPECT_TRUE(webrtc_data_channel_->Send(buffer));
>-  EXPECT_EQ(webrtc::DataChannelInterface::kOpen,
>-            webrtc_data_channel_->state());
>+  EXPECT_EQ(webrtc::DataChannelInterface::kOpen, webrtc_data_channel_->state());
> }
> 
> // Tests that a channel can be closed without being opened or assigned an sid.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/dtlssrtptransport.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/dtlssrtptransport.cc
>index bc5d3aa9002..e31b2f5872c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/dtlssrtptransport.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/dtlssrtptransport.cc
>@@ -24,20 +24,8 @@ static const char kDtlsSrtpExporterLabel[] = "EXTRACTOR-dtls_srtp";
> 
> namespace webrtc {
> 
>-DtlsSrtpTransport::DtlsSrtpTransport(
>-    std::unique_ptr<webrtc::SrtpTransport> srtp_transport)
>-    : RtpTransportInternalAdapter(srtp_transport.get()) {
>-  srtp_transport_ = std::move(srtp_transport);
>-  RTC_DCHECK(srtp_transport_);
>-  srtp_transport_->SignalPacketReceived.connect(
>-      this, &DtlsSrtpTransport::OnPacketReceived);
>-  srtp_transport_->SignalReadyToSend.connect(this,
>-                                             &DtlsSrtpTransport::OnReadyToSend);
>-  srtp_transport_->SignalWritableState.connect(
>-      this, &DtlsSrtpTransport::OnWritableState);
>-  srtp_transport_->SignalSentPacket.connect(this,
>-                                            &DtlsSrtpTransport::OnSentPacket);
>-}
>+DtlsSrtpTransport::DtlsSrtpTransport(bool rtcp_mux_enabled)
>+    : SrtpTransport(rtcp_mux_enabled) {}
> 
> void DtlsSrtpTransport::SetDtlsTransports(
>     cricket::DtlsTransportInternal* rtp_dtls_transport,
>@@ -51,18 +39,23 @@ void DtlsSrtpTransport::SetDtlsTransports(
>   // When using DTLS-SRTP, we must reset the SrtpTransport every time the
>   // DtlsTransport changes and wait until the DTLS handshake is complete to set
>   // the newly negotiated parameters.
>-  if (IsActive()) {
>-    srtp_transport_->ResetParams();
>+  // If |active_reset_srtp_params_| is true, intentionally reset the SRTP
>+  // parameter even though the DtlsTransport may not change.
>+  if (IsSrtpActive() && (rtp_dtls_transport != rtp_dtls_transport_ ||
>+                         active_reset_srtp_params_)) {
>+    ResetParams();
>   }
> 
>   const std::string transport_name =
>       rtp_dtls_transport ? rtp_dtls_transport->transport_name() : "null";
> 
>-  // This would only be possible if using BUNDLE but not rtcp-mux, which isn't
>-  // allowed according to the BUNDLE spec.
>-  RTC_CHECK(!(IsActive()))
>-      << "Setting RTCP for DTLS/SRTP after the DTLS is active "
>-      << "should never happen.";
>+  if (rtcp_dtls_transport && rtcp_dtls_transport != rtcp_dtls_transport_) {
>+    // This would only be possible if using BUNDLE but not rtcp-mux, which isn't
>+    // allowed according to the BUNDLE spec.
>+    RTC_CHECK(!(IsSrtpActive()))
>+        << "Setting RTCP for DTLS/SRTP after the DTLS is active "
>+           "should never happen.";
>+  }
> 
>   RTC_LOG(LS_INFO) << "Setting RTCP Transport on " << transport_name
>                    << " transport " << rtcp_dtls_transport;
>@@ -74,13 +67,13 @@ void DtlsSrtpTransport::SetDtlsTransports(
>   SetRtpDtlsTransport(rtp_dtls_transport);
>   SetRtpPacketTransport(rtp_dtls_transport);
> 
>-  UpdateWritableStateAndMaybeSetupDtlsSrtp();
>+  MaybeSetupDtlsSrtp();
> }
> 
> void DtlsSrtpTransport::SetRtcpMuxEnabled(bool enable) {
>-  srtp_transport_->SetRtcpMuxEnabled(enable);
>+  SrtpTransport::SetRtcpMuxEnabled(enable);
>   if (enable) {
>-    UpdateWritableStateAndMaybeSetupDtlsSrtp();
>+    MaybeSetupDtlsSrtp();
>   }
> }
> 
>@@ -126,10 +119,9 @@ bool DtlsSrtpTransport::IsDtlsConnected() {
> }
> 
> bool DtlsSrtpTransport::IsDtlsWritable() {
>-  auto rtp_packet_transport = srtp_transport_->rtp_packet_transport();
>   auto rtcp_packet_transport =
>-      rtcp_mux_enabled() ? nullptr : srtp_transport_->rtcp_packet_transport();
>-  return rtp_packet_transport && rtp_packet_transport->writable() &&
>+      rtcp_mux_enabled() ? nullptr : rtcp_dtls_transport_;
>+  return rtp_dtls_transport_ && rtp_dtls_transport_->writable() &&
>          (!rtcp_packet_transport || rtcp_packet_transport->writable());
> }
> 
>@@ -138,7 +130,7 @@ bool DtlsSrtpTransport::DtlsHandshakeCompleted() {
> }
> 
> void DtlsSrtpTransport::MaybeSetupDtlsSrtp() {
>-  if (IsActive() || !DtlsHandshakeCompleted()) {
>+  if (IsSrtpActive() || !IsDtlsWritable()) {
>     return;
>   }
> 
>@@ -163,16 +155,15 @@ void DtlsSrtpTransport::SetupRtpDtlsSrtp() {
>   }
> 
>   int selected_crypto_suite;
>-  std::vector<unsigned char> send_key;
>-  std::vector<unsigned char> recv_key;
>+  rtc::ZeroOnFreeBuffer<unsigned char> send_key;
>+  rtc::ZeroOnFreeBuffer<unsigned char> recv_key;
> 
>   if (!ExtractParams(rtp_dtls_transport_, &selected_crypto_suite, &send_key,
>                      &recv_key) ||
>-      !srtp_transport_->SetRtpParams(
>-          selected_crypto_suite, &send_key[0],
>-          static_cast<int>(send_key.size()), send_extension_ids,
>-          selected_crypto_suite, &recv_key[0],
>-          static_cast<int>(recv_key.size()), recv_extension_ids)) {
>+      !SetRtpParams(selected_crypto_suite, &send_key[0],
>+                    static_cast<int>(send_key.size()), send_extension_ids,
>+                    selected_crypto_suite, &recv_key[0],
>+                    static_cast<int>(recv_key.size()), recv_extension_ids)) {
>     SignalDtlsSrtpSetupFailure(this, /*rtcp=*/false);
>     RTC_LOG(LS_WARNING) << "DTLS-SRTP key installation for RTP failed";
>   }
>@@ -182,7 +173,7 @@ void DtlsSrtpTransport::SetupRtcpDtlsSrtp() {
>   // Return if the DTLS-SRTP is active because the encrypted header extension
>   // IDs don't need to be updated for RTCP and the crypto params don't need to
>   // be reset.
>-  if (IsActive()) {
>+  if (IsSrtpActive()) {
>     return;
>   }
> 
>@@ -196,15 +187,15 @@ void DtlsSrtpTransport::SetupRtcpDtlsSrtp() {
>   }
> 
>   int selected_crypto_suite;
>-  std::vector<unsigned char> rtcp_send_key;
>-  std::vector<unsigned char> rtcp_recv_key;
>+  rtc::ZeroOnFreeBuffer<unsigned char> rtcp_send_key;
>+  rtc::ZeroOnFreeBuffer<unsigned char> rtcp_recv_key;
>   if (!ExtractParams(rtcp_dtls_transport_, &selected_crypto_suite,
>                      &rtcp_send_key, &rtcp_recv_key) ||
>-      !srtp_transport_->SetRtcpParams(
>-          selected_crypto_suite, &rtcp_send_key[0],
>-          static_cast<int>(rtcp_send_key.size()), send_extension_ids,
>-          selected_crypto_suite, &rtcp_recv_key[0],
>-          static_cast<int>(rtcp_recv_key.size()), recv_extension_ids)) {
>+      !SetRtcpParams(selected_crypto_suite, &rtcp_send_key[0],
>+                     static_cast<int>(rtcp_send_key.size()), send_extension_ids,
>+                     selected_crypto_suite, &rtcp_recv_key[0],
>+                     static_cast<int>(rtcp_recv_key.size()),
>+                     recv_extension_ids)) {
>     SignalDtlsSrtpSetupFailure(this, /*rtcp=*/true);
>     RTC_LOG(LS_WARNING) << "DTLS-SRTP key installation for RTCP failed";
>   }
>@@ -213,8 +204,8 @@ void DtlsSrtpTransport::SetupRtcpDtlsSrtp() {
> bool DtlsSrtpTransport::ExtractParams(
>     cricket::DtlsTransportInternal* dtls_transport,
>     int* selected_crypto_suite,
>-    std::vector<unsigned char>* send_key,
>-    std::vector<unsigned char>* recv_key) {
>+    rtc::ZeroOnFreeBuffer<unsigned char>* send_key,
>+    rtc::ZeroOnFreeBuffer<unsigned char>* recv_key) {
>   if (!dtls_transport || !dtls_transport->IsDtlsActive()) {
>     return false;
>   }
>@@ -237,7 +228,7 @@ bool DtlsSrtpTransport::ExtractParams(
>   }
> 
>   // OK, we're now doing DTLS (RFC 5764)
>-  std::vector<unsigned char> dtls_buffer(key_len * 2 + salt_len * 2);
>+  rtc::ZeroOnFreeBuffer<unsigned char> dtls_buffer(key_len * 2 + salt_len * 2);
> 
>   // RFC 5705 exporter using the RFC 5764 parameters
>   if (!dtls_transport->ExportKeyingMaterial(kDtlsSrtpExporterLabel, NULL, 0,
>@@ -249,8 +240,8 @@ bool DtlsSrtpTransport::ExtractParams(
>   }
> 
>   // Sync up the keys with the DTLS-SRTP interface
>-  std::vector<unsigned char> client_write_key(key_len + salt_len);
>-  std::vector<unsigned char> server_write_key(key_len + salt_len);
>+  rtc::ZeroOnFreeBuffer<unsigned char> client_write_key(key_len + salt_len);
>+  rtc::ZeroOnFreeBuffer<unsigned char> server_write_key(key_len + salt_len);
>   size_t offset = 0;
>   memcpy(&client_write_key[0], &dtls_buffer[offset], key_len);
>   offset += key_len;
>@@ -261,19 +252,18 @@ bool DtlsSrtpTransport::ExtractParams(
>   memcpy(&server_write_key[key_len], &dtls_buffer[offset], salt_len);
> 
>   rtc::SSLRole role;
>-  if (!dtls_transport->GetSslRole(&role)) {
>-    RTC_LOG(LS_WARNING) << "GetSslRole failed";
>+  if (!dtls_transport->GetDtlsRole(&role)) {
>+    RTC_LOG(LS_WARNING) << "Failed to get the DTLS role.";
>     return false;
>   }
> 
>   if (role == rtc::SSL_SERVER) {
>-    *send_key = server_write_key;
>-    *recv_key = client_write_key;
>+    *send_key = std::move(server_write_key);
>+    *recv_key = std::move(client_write_key);
>   } else {
>-    *send_key = client_write_key;
>-    *recv_key = server_write_key;
>+    *send_key = std::move(client_write_key);
>+    *recv_key = std::move(server_write_key);
>   }
>-
>   return true;
> }
> 
>@@ -306,54 +296,22 @@ void DtlsSrtpTransport::SetRtcpDtlsTransport(
>   SetDtlsTransport(rtcp_dtls_transport, &rtcp_dtls_transport_);
> }
> 
>-void DtlsSrtpTransport::UpdateWritableStateAndMaybeSetupDtlsSrtp() {
>-  bool writable = IsDtlsWritable();
>-  SetWritable(writable);
>-  if (writable) {
>-    MaybeSetupDtlsSrtp();
>-  }
>-}
>-
>-void DtlsSrtpTransport::SetWritable(bool writable) {
>-  // Only fire the signal if the writable state changes.
>-  if (writable_ != writable) {
>-    writable_ = writable;
>-    SignalWritableState(writable_);
>-  }
>-}
>-
> void DtlsSrtpTransport::OnDtlsState(cricket::DtlsTransportInternal* transport,
>                                     cricket::DtlsTransportState state) {
>   RTC_DCHECK(transport == rtp_dtls_transport_ ||
>              transport == rtcp_dtls_transport_);
> 
>   if (state != cricket::DTLS_TRANSPORT_CONNECTED) {
>-    srtp_transport_->ResetParams();
>+    ResetParams();
>     return;
>   }
> 
>   MaybeSetupDtlsSrtp();
> }
> 
>-void DtlsSrtpTransport::OnWritableState(bool writable) {
>-  SetWritable(writable);
>-  if (writable) {
>-    MaybeSetupDtlsSrtp();
>-  }
>-}
>-
>-void DtlsSrtpTransport::OnSentPacket(const rtc::SentPacket& sent_packet) {
>-  SignalSentPacket(sent_packet);
>-}
>-
>-void DtlsSrtpTransport::OnPacketReceived(bool rtcp,
>-                                         rtc::CopyOnWriteBuffer* packet,
>-                                         const rtc::PacketTime& packet_time) {
>-  SignalPacketReceived(rtcp, packet, packet_time);
>-}
>-
>-void DtlsSrtpTransport::OnReadyToSend(bool ready) {
>-  SignalReadyToSend(ready);
>+void DtlsSrtpTransport::OnWritableState(
>+    rtc::PacketTransportInternal* packet_transport) {
>+  MaybeSetupDtlsSrtp();
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/dtlssrtptransport.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/dtlssrtptransport.h
>index 55b96c16450..498f02e9176 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/dtlssrtptransport.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/dtlssrtptransport.h
>@@ -16,19 +16,17 @@
> #include <vector>
> 
> #include "p2p/base/dtlstransportinternal.h"
>-#include "pc/rtptransportinternaladapter.h"
> #include "pc/srtptransport.h"
>+#include "rtc_base/buffer.h"
> 
> namespace webrtc {
> 
>-// This class is intended to be used as an RtpTransport and it wraps both an
>-// SrtpTransport and DtlsTransports(RTP/RTCP). When the DTLS handshake is
>-// finished, it extracts the keying materials from DtlsTransport and sets them
>-// to SrtpTransport.
>-class DtlsSrtpTransport : public RtpTransportInternalAdapter {
>+// The subclass of SrtpTransport is used for DTLS-SRTP. When the DTLS handshake
>+// is finished, it extracts the keying materials from DtlsTransport and
>+// configures the SrtpSessions in the base class.
>+class DtlsSrtpTransport : public SrtpTransport {
>  public:
>-  explicit DtlsSrtpTransport(
>-      std::unique_ptr<webrtc::SrtpTransport> srtp_transport);
>+  explicit DtlsSrtpTransport(bool rtcp_mux_enabled);
> 
>   // Set P2P layer RTP/RTCP DtlsTransports. When using RTCP-muxing,
>   // |rtcp_dtls_transport| is null.
>@@ -44,19 +42,22 @@ class DtlsSrtpTransport : public RtpTransportInternalAdapter {
>   void UpdateRecvEncryptedHeaderExtensionIds(
>       const std::vector<int>& recv_extension_ids);
> 
>-  bool IsActive() { return srtp_transport_->IsActive(); }
>+  sigslot::signal2<DtlsSrtpTransport*, bool> SignalDtlsSrtpSetupFailure;
> 
>-  // Cache RTP Absoulute SendTime extension header ID. This is only used when
>-  // external authentication is enabled.
>-  void CacheRtpAbsSendTimeHeaderExtension(int rtp_abs_sendtime_extn_id) {
>-    srtp_transport_->CacheRtpAbsSendTimeHeaderExtension(
>-        rtp_abs_sendtime_extn_id);
>+  RTCError SetSrtpSendKey(const cricket::CryptoParams& params) override {
>+    return RTCError(RTCErrorType::UNSUPPORTED_OPERATION,
>+                    "Set SRTP keys for DTLS-SRTP is not supported.");
>+  }
>+  RTCError SetSrtpReceiveKey(const cricket::CryptoParams& params) override {
>+    return RTCError(RTCErrorType::UNSUPPORTED_OPERATION,
>+                    "Set SRTP keys for DTLS-SRTP is not supported.");
>   }
> 
>-  // TODO(zhihuang): Remove this when we remove RtpTransportAdapter.
>-  RtpTransportAdapter* GetInternal() override { return nullptr; }
>-
>-  sigslot::signal2<DtlsSrtpTransport*, bool> SignalDtlsSrtpSetupFailure;
>+  // If |active_reset_srtp_params_| is set to be true, the SRTP parameters will
>+  // be reset whenever the DtlsTransports are reset.
>+  void SetActiveResetSrtpParams(bool active_reset_srtp_params) {
>+    active_reset_srtp_params_ = active_reset_srtp_params;
>+  }
> 
>  private:
>   bool IsDtlsActive();
>@@ -68,36 +69,29 @@ class DtlsSrtpTransport : public RtpTransportInternalAdapter {
>   void SetupRtcpDtlsSrtp();
>   bool ExtractParams(cricket::DtlsTransportInternal* dtls_transport,
>                      int* selected_crypto_suite,
>-                     std::vector<unsigned char>* send_key,
>-                     std::vector<unsigned char>* recv_key);
>+                     rtc::ZeroOnFreeBuffer<unsigned char>* send_key,
>+                     rtc::ZeroOnFreeBuffer<unsigned char>* recv_key);
>   void SetDtlsTransport(cricket::DtlsTransportInternal* new_dtls_transport,
>                         cricket::DtlsTransportInternal** old_dtls_transport);
>   void SetRtpDtlsTransport(cricket::DtlsTransportInternal* rtp_dtls_transport);
>   void SetRtcpDtlsTransport(
>       cricket::DtlsTransportInternal* rtcp_dtls_transport);
>-  void UpdateWritableStateAndMaybeSetupDtlsSrtp();
>-  // Set the writability and fire the SignalWritableState if the writability
>-  // changes.
>-  void SetWritable(bool writable);
> 
>   void OnDtlsState(cricket::DtlsTransportInternal* dtls_transport,
>                    cricket::DtlsTransportState state);
>-  void OnWritableState(bool writable);
>-  void OnSentPacket(const rtc::SentPacket& sent_packet);
>-  void OnPacketReceived(bool rtcp,
>-                        rtc::CopyOnWriteBuffer* packet,
>-                        const rtc::PacketTime& packet_time);
>-  void OnReadyToSend(bool ready);
>-
>-  bool writable_ = false;
>-  std::unique_ptr<SrtpTransport> srtp_transport_;
>+
>+  // Override the SrtpTransport::OnWritableState.
>+  void OnWritableState(rtc::PacketTransportInternal* packet_transport) override;
>+
>   // Owned by the TransportController.
>   cricket::DtlsTransportInternal* rtp_dtls_transport_ = nullptr;
>   cricket::DtlsTransportInternal* rtcp_dtls_transport_ = nullptr;
> 
>   // The encrypted header extension IDs.
>-  rtc::Optional<std::vector<int>> send_extension_ids_;
>-  rtc::Optional<std::vector<int>> recv_extension_ids_;
>+  absl::optional<std::vector<int>> send_extension_ids_;
>+  absl::optional<std::vector<int>> recv_extension_ids_;
>+
>+  bool active_reset_srtp_params_ = false;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/dtlssrtptransport_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/dtlssrtptransport_unittest.cc
>index 08a8151ee76..2012c49d741 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/dtlssrtptransport_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/dtlssrtptransport_unittest.cc
>@@ -13,6 +13,7 @@
> #include <memory>
> #include <utility>
> 
>+#include "absl/memory/memory.h"
> #include "media/base/fakertp.h"
> #include "p2p/base/dtlstransportinternal.h"
> #include "p2p/base/fakedtlstransport.h"
>@@ -22,7 +23,6 @@
> #include "pc/rtptransporttestutil.h"
> #include "rtc_base/asyncpacketsocket.h"
> #include "rtc_base/gunit.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/sslstreamadapter.h"
> 
> using cricket::FakeDtlsTransport;
>@@ -33,50 +33,26 @@ using webrtc::RtpTransport;
> 
> const int kRtpAuthTagLen = 10;
> 
>-class TransportObserver : public sigslot::has_slots<> {
>- public:
>-  void OnPacketReceived(bool rtcp,
>-                        rtc::CopyOnWriteBuffer* packet,
>-                        const rtc::PacketTime& packet_time) {
>-    rtcp ? last_recv_rtcp_packet_ = *packet : last_recv_rtp_packet_ = *packet;
>-  }
>-
>-  void OnReadyToSend(bool ready) { ready_to_send_ = ready; }
>-
>-  rtc::CopyOnWriteBuffer last_recv_rtp_packet() {
>-    return last_recv_rtp_packet_;
>-  }
>-
>-  rtc::CopyOnWriteBuffer last_recv_rtcp_packet() {
>-    return last_recv_rtcp_packet_;
>-  }
>-
>-  bool ready_to_send() { return ready_to_send_; }
>-
>- private:
>-  rtc::CopyOnWriteBuffer last_recv_rtp_packet_;
>-  rtc::CopyOnWriteBuffer last_recv_rtcp_packet_;
>-  bool ready_to_send_ = false;
>-};
>-
> class DtlsSrtpTransportTest : public testing::Test,
>                               public sigslot::has_slots<> {
>  protected:
>   DtlsSrtpTransportTest() {}
> 
>+  ~DtlsSrtpTransportTest() {
>+    if (dtls_srtp_transport1_) {
>+      dtls_srtp_transport1_->UnregisterRtpDemuxerSink(&transport_observer1_);
>+    }
>+    if (dtls_srtp_transport2_) {
>+      dtls_srtp_transport2_->UnregisterRtpDemuxerSink(&transport_observer2_);
>+    }
>+  }
>+
>   std::unique_ptr<DtlsSrtpTransport> MakeDtlsSrtpTransport(
>       FakeDtlsTransport* rtp_dtls,
>       FakeDtlsTransport* rtcp_dtls,
>       bool rtcp_mux_enabled) {
>-    auto rtp_transport = rtc::MakeUnique<RtpTransport>(rtcp_mux_enabled);
>-
>-    rtp_transport->AddHandledPayloadType(0x00);
>-    rtp_transport->AddHandledPayloadType(0xc9);
>-
>-    auto srtp_transport =
>-        rtc::MakeUnique<SrtpTransport>(std::move(rtp_transport));
>     auto dtls_srtp_transport =
>-        rtc::MakeUnique<DtlsSrtpTransport>(std::move(srtp_transport));
>+        absl::make_unique<DtlsSrtpTransport>(rtcp_mux_enabled);
> 
>     dtls_srtp_transport->SetDtlsTransports(rtp_dtls, rtcp_dtls);
> 
>@@ -93,15 +69,24 @@ class DtlsSrtpTransportTest : public testing::Test,
>     dtls_srtp_transport2_ =
>         MakeDtlsSrtpTransport(rtp_dtls2, rtcp_dtls2, rtcp_mux_enabled);
> 
>-    dtls_srtp_transport1_->SignalPacketReceived.connect(
>-        &transport_observer1_, &TransportObserver::OnPacketReceived);
>+    dtls_srtp_transport1_->SignalRtcpPacketReceived.connect(
>+        &transport_observer1_,
>+        &webrtc::TransportObserver::OnRtcpPacketReceived);
>     dtls_srtp_transport1_->SignalReadyToSend.connect(
>-        &transport_observer1_, &TransportObserver::OnReadyToSend);
>+        &transport_observer1_, &webrtc::TransportObserver::OnReadyToSend);
> 
>-    dtls_srtp_transport2_->SignalPacketReceived.connect(
>-        &transport_observer2_, &TransportObserver::OnPacketReceived);
>+    dtls_srtp_transport2_->SignalRtcpPacketReceived.connect(
>+        &transport_observer2_,
>+        &webrtc::TransportObserver::OnRtcpPacketReceived);
>     dtls_srtp_transport2_->SignalReadyToSend.connect(
>-        &transport_observer2_, &TransportObserver::OnReadyToSend);
>+        &transport_observer2_, &webrtc::TransportObserver::OnReadyToSend);
>+    webrtc::RtpDemuxerCriteria demuxer_criteria;
>+    // 0x00 is the payload type used in kPcmuFrame.
>+    demuxer_criteria.payload_types = {0x00};
>+    dtls_srtp_transport1_->RegisterRtpDemuxerSink(demuxer_criteria,
>+                                                  &transport_observer1_);
>+    dtls_srtp_transport2_->RegisterRtpDemuxerSink(demuxer_criteria,
>+                                                  &transport_observer2_);
>   }
> 
>   void CompleteDtlsHandshake(FakeDtlsTransport* fake_dtls1,
>@@ -118,8 +103,8 @@ class DtlsSrtpTransportTest : public testing::Test,
>   void SendRecvRtpPackets() {
>     ASSERT_TRUE(dtls_srtp_transport1_);
>     ASSERT_TRUE(dtls_srtp_transport2_);
>-    ASSERT_TRUE(dtls_srtp_transport1_->IsActive());
>-    ASSERT_TRUE(dtls_srtp_transport2_->IsActive());
>+    ASSERT_TRUE(dtls_srtp_transport1_->IsSrtpActive());
>+    ASSERT_TRUE(dtls_srtp_transport2_->IsSrtpActive());
> 
>     size_t rtp_len = sizeof(kPcmuFrame);
>     size_t packet_size = rtp_len + kRtpAuthTagLen;
>@@ -138,16 +123,21 @@ class DtlsSrtpTransportTest : public testing::Test,
>     rtc::PacketOptions options;
>     // Send a packet from |srtp_transport1_| to |srtp_transport2_| and verify
>     // that the packet can be successfully received and decrypted.
>+    int prev_received_packets = transport_observer2_.rtp_count();
>     ASSERT_TRUE(dtls_srtp_transport1_->SendRtpPacket(&rtp_packet1to2, options,
>                                                      cricket::PF_SRTP_BYPASS));
>     ASSERT_TRUE(transport_observer2_.last_recv_rtp_packet().data());
>     EXPECT_EQ(0, memcmp(transport_observer2_.last_recv_rtp_packet().data(),
>                         kPcmuFrame, rtp_len));
>+    EXPECT_EQ(prev_received_packets + 1, transport_observer2_.rtp_count());
>+
>+    prev_received_packets = transport_observer1_.rtp_count();
>     ASSERT_TRUE(dtls_srtp_transport2_->SendRtpPacket(&rtp_packet2to1, options,
>                                                      cricket::PF_SRTP_BYPASS));
>     ASSERT_TRUE(transport_observer1_.last_recv_rtp_packet().data());
>     EXPECT_EQ(0, memcmp(transport_observer1_.last_recv_rtp_packet().data(),
>                         kPcmuFrame, rtp_len));
>+    EXPECT_EQ(prev_received_packets + 1, transport_observer1_.rtp_count());
>   }
> 
>   void SendRecvRtcpPackets() {
>@@ -163,26 +153,30 @@ class DtlsSrtpTransportTest : public testing::Test,
>     rtc::PacketOptions options;
>     // Send a packet from |srtp_transport1_| to |srtp_transport2_| and verify
>     // that the packet can be successfully received and decrypted.
>+    int prev_received_packets = transport_observer2_.rtcp_count();
>     ASSERT_TRUE(dtls_srtp_transport1_->SendRtcpPacket(&rtcp_packet1to2, options,
>                                                       cricket::PF_SRTP_BYPASS));
>     ASSERT_TRUE(transport_observer2_.last_recv_rtcp_packet().data());
>     EXPECT_EQ(0, memcmp(transport_observer2_.last_recv_rtcp_packet().data(),
>                         kRtcpReport, rtcp_len));
>+    EXPECT_EQ(prev_received_packets + 1, transport_observer2_.rtcp_count());
> 
>     // Do the same thing in the opposite direction;
>+    prev_received_packets = transport_observer1_.rtcp_count();
>     ASSERT_TRUE(dtls_srtp_transport2_->SendRtcpPacket(&rtcp_packet2to1, options,
>                                                       cricket::PF_SRTP_BYPASS));
>     ASSERT_TRUE(transport_observer1_.last_recv_rtcp_packet().data());
>     EXPECT_EQ(0, memcmp(transport_observer1_.last_recv_rtcp_packet().data(),
>                         kRtcpReport, rtcp_len));
>+    EXPECT_EQ(prev_received_packets + 1, transport_observer1_.rtcp_count());
>   }
> 
>   void SendRecvRtpPacketsWithHeaderExtension(
>       const std::vector<int>& encrypted_header_ids) {
>     ASSERT_TRUE(dtls_srtp_transport1_);
>     ASSERT_TRUE(dtls_srtp_transport2_);
>-    ASSERT_TRUE(dtls_srtp_transport1_->IsActive());
>-    ASSERT_TRUE(dtls_srtp_transport2_->IsActive());
>+    ASSERT_TRUE(dtls_srtp_transport1_->IsSrtpActive());
>+    ASSERT_TRUE(dtls_srtp_transport2_->IsSrtpActive());
> 
>     size_t rtp_len = sizeof(kPcmuFrameWithExtensions);
>     size_t packet_size = rtp_len + kRtpAuthTagLen;
>@@ -251,8 +245,8 @@ class DtlsSrtpTransportTest : public testing::Test,
> 
>   std::unique_ptr<DtlsSrtpTransport> dtls_srtp_transport1_;
>   std::unique_ptr<DtlsSrtpTransport> dtls_srtp_transport2_;
>-  TransportObserver transport_observer1_;
>-  TransportObserver transport_observer2_;
>+  webrtc::TransportObserver transport_observer1_;
>+  webrtc::TransportObserver transport_observer2_;
> 
>   int sequence_number_ = 0;
> };
>@@ -260,17 +254,17 @@ class DtlsSrtpTransportTest : public testing::Test,
> // Tests that if RTCP muxing is enabled and transports are set after RTP
> // transport finished the handshake, SRTP is set up.
> TEST_F(DtlsSrtpTransportTest, SetTransportsAfterHandshakeCompleteWithRtcpMux) {
>-  auto rtp_dtls1 = rtc::MakeUnique<FakeDtlsTransport>(
>+  auto rtp_dtls1 = absl::make_unique<FakeDtlsTransport>(
>       "video", cricket::ICE_CANDIDATE_COMPONENT_RTP);
>-  auto rtp_dtls2 = rtc::MakeUnique<FakeDtlsTransport>(
>+  auto rtp_dtls2 = absl::make_unique<FakeDtlsTransport>(
>       "video", cricket::ICE_CANDIDATE_COMPONENT_RTP);
> 
>   MakeDtlsSrtpTransports(rtp_dtls1.get(), nullptr, rtp_dtls2.get(), nullptr,
>                          /*rtcp_mux_enabled=*/true);
> 
>-  auto rtp_dtls3 = rtc::MakeUnique<FakeDtlsTransport>(
>+  auto rtp_dtls3 = absl::make_unique<FakeDtlsTransport>(
>       "audio", cricket::ICE_CANDIDATE_COMPONENT_RTP);
>-  auto rtp_dtls4 = rtc::MakeUnique<FakeDtlsTransport>(
>+  auto rtp_dtls4 = absl::make_unique<FakeDtlsTransport>(
>       "audio", cricket::ICE_CANDIDATE_COMPONENT_RTP);
> 
>   CompleteDtlsHandshake(rtp_dtls3.get(), rtp_dtls4.get());
>@@ -285,25 +279,25 @@ TEST_F(DtlsSrtpTransportTest, SetTransportsAfterHandshakeCompleteWithRtcpMux) {
> // RTP and RTCP transports finished the handshake, SRTP is set up.
> TEST_F(DtlsSrtpTransportTest,
>        SetTransportsAfterHandshakeCompleteWithoutRtcpMux) {
>-  auto rtp_dtls1 = rtc::MakeUnique<FakeDtlsTransport>(
>+  auto rtp_dtls1 = absl::make_unique<FakeDtlsTransport>(
>       "video", cricket::ICE_CANDIDATE_COMPONENT_RTP);
>-  auto rtcp_dtls1 = rtc::MakeUnique<FakeDtlsTransport>(
>+  auto rtcp_dtls1 = absl::make_unique<FakeDtlsTransport>(
>       "video", cricket::ICE_CANDIDATE_COMPONENT_RTCP);
>-  auto rtp_dtls2 = rtc::MakeUnique<FakeDtlsTransport>(
>+  auto rtp_dtls2 = absl::make_unique<FakeDtlsTransport>(
>       "video", cricket::ICE_CANDIDATE_COMPONENT_RTP);
>-  auto rtcp_dtls2 = rtc::MakeUnique<FakeDtlsTransport>(
>+  auto rtcp_dtls2 = absl::make_unique<FakeDtlsTransport>(
>       "video", cricket::ICE_CANDIDATE_COMPONENT_RTCP);
> 
>   MakeDtlsSrtpTransports(rtp_dtls1.get(), rtcp_dtls1.get(), rtp_dtls2.get(),
>                          rtcp_dtls2.get(), /*rtcp_mux_enabled=*/false);
> 
>-  auto rtp_dtls3 = rtc::MakeUnique<FakeDtlsTransport>(
>+  auto rtp_dtls3 = absl::make_unique<FakeDtlsTransport>(
>       "audio", cricket::ICE_CANDIDATE_COMPONENT_RTP);
>-  auto rtcp_dtls3 = rtc::MakeUnique<FakeDtlsTransport>(
>+  auto rtcp_dtls3 = absl::make_unique<FakeDtlsTransport>(
>       "audio", cricket::ICE_CANDIDATE_COMPONENT_RTCP);
>-  auto rtp_dtls4 = rtc::MakeUnique<FakeDtlsTransport>(
>+  auto rtp_dtls4 = absl::make_unique<FakeDtlsTransport>(
>       "audio", cricket::ICE_CANDIDATE_COMPONENT_RTP);
>-  auto rtcp_dtls4 = rtc::MakeUnique<FakeDtlsTransport>(
>+  auto rtcp_dtls4 = absl::make_unique<FakeDtlsTransport>(
>       "audio", cricket::ICE_CANDIDATE_COMPONENT_RTCP);
>   CompleteDtlsHandshake(rtp_dtls3.get(), rtp_dtls4.get());
>   CompleteDtlsHandshake(rtcp_dtls3.get(), rtcp_dtls4.get());
>@@ -317,13 +311,13 @@ TEST_F(DtlsSrtpTransportTest,
> // Tests if RTCP muxing is enabled, SRTP is set up as soon as the RTP DTLS
> // handshake is finished.
> TEST_F(DtlsSrtpTransportTest, SetTransportsBeforeHandshakeCompleteWithRtcpMux) {
>-  auto rtp_dtls1 = rtc::MakeUnique<FakeDtlsTransport>(
>+  auto rtp_dtls1 = absl::make_unique<FakeDtlsTransport>(
>       "audio", cricket::ICE_CANDIDATE_COMPONENT_RTP);
>-  auto rtcp_dtls1 = rtc::MakeUnique<FakeDtlsTransport>(
>+  auto rtcp_dtls1 = absl::make_unique<FakeDtlsTransport>(
>       "audio", cricket::ICE_CANDIDATE_COMPONENT_RTCP);
>-  auto rtp_dtls2 = rtc::MakeUnique<FakeDtlsTransport>(
>+  auto rtp_dtls2 = absl::make_unique<FakeDtlsTransport>(
>       "audio", cricket::ICE_CANDIDATE_COMPONENT_RTP);
>-  auto rtcp_dtls2 = rtc::MakeUnique<FakeDtlsTransport>(
>+  auto rtcp_dtls2 = absl::make_unique<FakeDtlsTransport>(
>       "audio", cricket::ICE_CANDIDATE_COMPONENT_RTCP);
> 
>   MakeDtlsSrtpTransports(rtp_dtls1.get(), rtcp_dtls1.get(), rtp_dtls2.get(),
>@@ -340,21 +334,21 @@ TEST_F(DtlsSrtpTransportTest, SetTransportsBeforeHandshakeCompleteWithRtcpMux) {
> // RTCP DTLS handshake are finished.
> TEST_F(DtlsSrtpTransportTest,
>        SetTransportsBeforeHandshakeCompleteWithoutRtcpMux) {
>-  auto rtp_dtls1 = rtc::MakeUnique<FakeDtlsTransport>(
>+  auto rtp_dtls1 = absl::make_unique<FakeDtlsTransport>(
>       "audio", cricket::ICE_CANDIDATE_COMPONENT_RTP);
>-  auto rtcp_dtls1 = rtc::MakeUnique<FakeDtlsTransport>(
>+  auto rtcp_dtls1 = absl::make_unique<FakeDtlsTransport>(
>       "audio", cricket::ICE_CANDIDATE_COMPONENT_RTCP);
>-  auto rtp_dtls2 = rtc::MakeUnique<FakeDtlsTransport>(
>+  auto rtp_dtls2 = absl::make_unique<FakeDtlsTransport>(
>       "audio", cricket::ICE_CANDIDATE_COMPONENT_RTP);
>-  auto rtcp_dtls2 = rtc::MakeUnique<FakeDtlsTransport>(
>+  auto rtcp_dtls2 = absl::make_unique<FakeDtlsTransport>(
>       "audio", cricket::ICE_CANDIDATE_COMPONENT_RTCP);
> 
>   MakeDtlsSrtpTransports(rtp_dtls1.get(), rtcp_dtls1.get(), rtp_dtls2.get(),
>                          rtcp_dtls2.get(), /*rtcp_mux_enabled=*/false);
> 
>   CompleteDtlsHandshake(rtp_dtls1.get(), rtp_dtls2.get());
>-  EXPECT_FALSE(dtls_srtp_transport1_->IsActive());
>-  EXPECT_FALSE(dtls_srtp_transport2_->IsActive());
>+  EXPECT_FALSE(dtls_srtp_transport1_->IsSrtpActive());
>+  EXPECT_FALSE(dtls_srtp_transport2_->IsSrtpActive());
>   CompleteDtlsHandshake(rtcp_dtls1.get(), rtcp_dtls2.get());
>   SendRecvPackets();
> }
>@@ -363,28 +357,28 @@ TEST_F(DtlsSrtpTransportTest,
> // context will be reset and will be re-setup once the new transports' handshake
> // complete.
> TEST_F(DtlsSrtpTransportTest, DtlsSrtpResetAfterDtlsTransportChange) {
>-  auto rtp_dtls1 = rtc::MakeUnique<FakeDtlsTransport>(
>+  auto rtp_dtls1 = absl::make_unique<FakeDtlsTransport>(
>       "audio", cricket::ICE_CANDIDATE_COMPONENT_RTP);
>-  auto rtp_dtls2 = rtc::MakeUnique<FakeDtlsTransport>(
>+  auto rtp_dtls2 = absl::make_unique<FakeDtlsTransport>(
>       "audio", cricket::ICE_CANDIDATE_COMPONENT_RTP);
> 
>   MakeDtlsSrtpTransports(rtp_dtls1.get(), nullptr, rtp_dtls2.get(), nullptr,
>                          /*rtcp_mux_enabled=*/true);
> 
>   CompleteDtlsHandshake(rtp_dtls1.get(), rtp_dtls2.get());
>-  EXPECT_TRUE(dtls_srtp_transport1_->IsActive());
>-  EXPECT_TRUE(dtls_srtp_transport2_->IsActive());
>+  EXPECT_TRUE(dtls_srtp_transport1_->IsSrtpActive());
>+  EXPECT_TRUE(dtls_srtp_transport2_->IsSrtpActive());
> 
>-  auto rtp_dtls3 = rtc::MakeUnique<FakeDtlsTransport>(
>+  auto rtp_dtls3 = absl::make_unique<FakeDtlsTransport>(
>       "audio", cricket::ICE_CANDIDATE_COMPONENT_RTP);
>-  auto rtp_dtls4 = rtc::MakeUnique<FakeDtlsTransport>(
>+  auto rtp_dtls4 = absl::make_unique<FakeDtlsTransport>(
>       "audio", cricket::ICE_CANDIDATE_COMPONENT_RTP);
> 
>   // The previous context is reset.
>   dtls_srtp_transport1_->SetDtlsTransports(rtp_dtls3.get(), nullptr);
>   dtls_srtp_transport2_->SetDtlsTransports(rtp_dtls4.get(), nullptr);
>-  EXPECT_FALSE(dtls_srtp_transport1_->IsActive());
>-  EXPECT_FALSE(dtls_srtp_transport2_->IsActive());
>+  EXPECT_FALSE(dtls_srtp_transport1_->IsSrtpActive());
>+  EXPECT_FALSE(dtls_srtp_transport2_->IsSrtpActive());
> 
>   // Re-setup.
>   CompleteDtlsHandshake(rtp_dtls3.get(), rtp_dtls4.get());
>@@ -395,13 +389,13 @@ TEST_F(DtlsSrtpTransportTest, DtlsSrtpResetAfterDtlsTransportChange) {
> // enabled, SRTP is set up.
> TEST_F(DtlsSrtpTransportTest,
>        RtcpMuxEnabledAfterRtpTransportHandshakeComplete) {
>-  auto rtp_dtls1 = rtc::MakeUnique<FakeDtlsTransport>(
>+  auto rtp_dtls1 = absl::make_unique<FakeDtlsTransport>(
>       "audio", cricket::ICE_CANDIDATE_COMPONENT_RTP);
>-  auto rtcp_dtls1 = rtc::MakeUnique<FakeDtlsTransport>(
>+  auto rtcp_dtls1 = absl::make_unique<FakeDtlsTransport>(
>       "audio", cricket::ICE_CANDIDATE_COMPONENT_RTCP);
>-  auto rtp_dtls2 = rtc::MakeUnique<FakeDtlsTransport>(
>+  auto rtp_dtls2 = absl::make_unique<FakeDtlsTransport>(
>       "audio", cricket::ICE_CANDIDATE_COMPONENT_RTP);
>-  auto rtcp_dtls2 = rtc::MakeUnique<FakeDtlsTransport>(
>+  auto rtcp_dtls2 = absl::make_unique<FakeDtlsTransport>(
>       "audio", cricket::ICE_CANDIDATE_COMPONENT_RTCP);
> 
>   MakeDtlsSrtpTransports(rtp_dtls1.get(), rtcp_dtls1.get(), rtp_dtls2.get(),
>@@ -409,8 +403,8 @@ TEST_F(DtlsSrtpTransportTest,
> 
>   CompleteDtlsHandshake(rtp_dtls1.get(), rtp_dtls2.get());
>   // Inactive because the RTCP transport handshake didn't complete.
>-  EXPECT_FALSE(dtls_srtp_transport1_->IsActive());
>-  EXPECT_FALSE(dtls_srtp_transport2_->IsActive());
>+  EXPECT_FALSE(dtls_srtp_transport1_->IsSrtpActive());
>+  EXPECT_FALSE(dtls_srtp_transport2_->IsSrtpActive());
> 
>   dtls_srtp_transport1_->SetRtcpMuxEnabled(true);
>   dtls_srtp_transport2_->SetRtcpMuxEnabled(true);
>@@ -422,9 +416,9 @@ TEST_F(DtlsSrtpTransportTest,
> // Tests that when SetSend/RecvEncryptedHeaderExtensionIds is called, the SRTP
> // sessions are updated with new encryped header extension IDs immediately.
> TEST_F(DtlsSrtpTransportTest, EncryptedHeaderExtensionIdUpdated) {
>-  auto rtp_dtls1 = rtc::MakeUnique<FakeDtlsTransport>(
>+  auto rtp_dtls1 = absl::make_unique<FakeDtlsTransport>(
>       "audio", cricket::ICE_CANDIDATE_COMPONENT_RTP);
>-  auto rtp_dtls2 = rtc::MakeUnique<FakeDtlsTransport>(
>+  auto rtp_dtls2 = absl::make_unique<FakeDtlsTransport>(
>       "audio", cricket::ICE_CANDIDATE_COMPONENT_RTP);
> 
>   MakeDtlsSrtpTransports(rtp_dtls1.get(), nullptr, rtp_dtls2.get(), nullptr,
>@@ -448,9 +442,9 @@ TEST_F(DtlsSrtpTransportTest, EncryptedHeaderExtensionIdUpdated) {
> // Tests if RTCP muxing is enabled. DtlsSrtpTransport is ready to send once the
> // RTP DtlsTransport is ready.
> TEST_F(DtlsSrtpTransportTest, SignalReadyToSendFiredWithRtcpMux) {
>-  auto rtp_dtls1 = rtc::MakeUnique<FakeDtlsTransport>(
>+  auto rtp_dtls1 = absl::make_unique<FakeDtlsTransport>(
>       "audio", cricket::ICE_CANDIDATE_COMPONENT_RTP);
>-  auto rtp_dtls2 = rtc::MakeUnique<FakeDtlsTransport>(
>+  auto rtp_dtls2 = absl::make_unique<FakeDtlsTransport>(
>       "audio", cricket::ICE_CANDIDATE_COMPONENT_RTP);
> 
>   MakeDtlsSrtpTransports(rtp_dtls1.get(), nullptr, rtp_dtls2.get(), nullptr,
>@@ -464,13 +458,13 @@ TEST_F(DtlsSrtpTransportTest, SignalReadyToSendFiredWithRtcpMux) {
> // Tests if RTCP muxing is not enabled. DtlsSrtpTransport is ready to send once
> // both the RTP and RTCP DtlsTransport are ready.
> TEST_F(DtlsSrtpTransportTest, SignalReadyToSendFiredWithoutRtcpMux) {
>-  auto rtp_dtls1 = rtc::MakeUnique<FakeDtlsTransport>(
>+  auto rtp_dtls1 = absl::make_unique<FakeDtlsTransport>(
>       "audio", cricket::ICE_CANDIDATE_COMPONENT_RTP);
>-  auto rtcp_dtls1 = rtc::MakeUnique<FakeDtlsTransport>(
>+  auto rtcp_dtls1 = absl::make_unique<FakeDtlsTransport>(
>       "audio", cricket::ICE_CANDIDATE_COMPONENT_RTCP);
>-  auto rtp_dtls2 = rtc::MakeUnique<FakeDtlsTransport>(
>+  auto rtp_dtls2 = absl::make_unique<FakeDtlsTransport>(
>       "audio", cricket::ICE_CANDIDATE_COMPONENT_RTP);
>-  auto rtcp_dtls2 = rtc::MakeUnique<FakeDtlsTransport>(
>+  auto rtcp_dtls2 = absl::make_unique<FakeDtlsTransport>(
>       "audio", cricket::ICE_CANDIDATE_COMPONENT_RTCP);
> 
>   MakeDtlsSrtpTransports(rtp_dtls1.get(), rtcp_dtls1.get(), rtp_dtls2.get(),
>@@ -484,3 +478,83 @@ TEST_F(DtlsSrtpTransportTest, SignalReadyToSendFiredWithoutRtcpMux) {
>   EXPECT_TRUE(transport_observer1_.ready_to_send());
>   EXPECT_TRUE(transport_observer2_.ready_to_send());
> }
>+
>+// Test that if an endpoint "fully" enables RTCP mux, setting the RTCP
>+// transport to null, it *doesn't* reset its SRTP context. That would cause the
>+// ROC and SRTCP index to be reset, causing replay detection and other errors
>+// when attempting to unprotect packets.
>+// Regression test for bugs.webrtc.org/8996
>+TEST_F(DtlsSrtpTransportTest, SrtpSessionNotResetWhenRtcpTransportRemoved) {
>+  auto rtp_dtls1 = absl::make_unique<FakeDtlsTransport>(
>+      "audio", cricket::ICE_CANDIDATE_COMPONENT_RTP);
>+  auto rtcp_dtls1 = absl::make_unique<FakeDtlsTransport>(
>+      "audio", cricket::ICE_CANDIDATE_COMPONENT_RTCP);
>+  auto rtp_dtls2 = absl::make_unique<FakeDtlsTransport>(
>+      "audio", cricket::ICE_CANDIDATE_COMPONENT_RTP);
>+  auto rtcp_dtls2 = absl::make_unique<FakeDtlsTransport>(
>+      "audio", cricket::ICE_CANDIDATE_COMPONENT_RTCP);
>+
>+  MakeDtlsSrtpTransports(rtp_dtls1.get(), rtcp_dtls1.get(), rtp_dtls2.get(),
>+                         rtcp_dtls2.get(), /*rtcp_mux_enabled=*/true);
>+  CompleteDtlsHandshake(rtp_dtls1.get(), rtp_dtls2.get());
>+  CompleteDtlsHandshake(rtcp_dtls1.get(), rtcp_dtls2.get());
>+
>+  // Send some RTCP packets, causing the SRTCP index to be incremented.
>+  SendRecvRtcpPackets();
>+
>+  // Set RTCP transport to null, which previously would trigger this problem.
>+  dtls_srtp_transport1_->SetDtlsTransports(rtp_dtls1.get(), nullptr);
>+
>+  // Attempt to send more RTCP packets. If the issue occurred, one side would
>+  // reset its context while the other would not, causing replay detection
>+  // errors when a packet with a duplicate SRTCP index is received.
>+  SendRecvRtcpPackets();
>+}
>+
>+// Tests that RTCP packets can be sent and received if both sides actively reset
>+// the SRTP parameters with the |active_reset_srtp_params_| flag.
>+TEST_F(DtlsSrtpTransportTest, ActivelyResetSrtpParams) {
>+  auto rtp_dtls1 = absl::make_unique<FakeDtlsTransport>(
>+      "audio", cricket::ICE_CANDIDATE_COMPONENT_RTP);
>+  auto rtcp_dtls1 = absl::make_unique<FakeDtlsTransport>(
>+      "audio", cricket::ICE_CANDIDATE_COMPONENT_RTCP);
>+  auto rtp_dtls2 = absl::make_unique<FakeDtlsTransport>(
>+      "audio", cricket::ICE_CANDIDATE_COMPONENT_RTP);
>+  auto rtcp_dtls2 = absl::make_unique<FakeDtlsTransport>(
>+      "audio", cricket::ICE_CANDIDATE_COMPONENT_RTCP);
>+
>+  MakeDtlsSrtpTransports(rtp_dtls1.get(), rtcp_dtls1.get(), rtp_dtls2.get(),
>+                         rtcp_dtls2.get(), /*rtcp_mux_enabled=*/true);
>+  CompleteDtlsHandshake(rtp_dtls1.get(), rtp_dtls2.get());
>+  CompleteDtlsHandshake(rtcp_dtls1.get(), rtcp_dtls2.get());
>+
>+  // Send some RTCP packets, causing the SRTCP index to be incremented.
>+  SendRecvRtcpPackets();
>+
>+  // Only set the |active_reset_srtp_params_| flag to be true one side.
>+  dtls_srtp_transport1_->SetActiveResetSrtpParams(true);
>+  // Set RTCP transport to null to trigger the SRTP parameters update.
>+  dtls_srtp_transport1_->SetDtlsTransports(rtp_dtls1.get(), nullptr);
>+  dtls_srtp_transport2_->SetDtlsTransports(rtp_dtls2.get(), nullptr);
>+
>+  // Sending some RTCP packets.
>+  size_t rtcp_len = sizeof(kRtcpReport);
>+  size_t packet_size = rtcp_len + 4 + kRtpAuthTagLen;
>+  rtc::Buffer rtcp_packet_buffer(packet_size);
>+  rtc::CopyOnWriteBuffer rtcp_packet(kRtcpReport, rtcp_len, packet_size);
>+  int prev_received_packets = transport_observer2_.rtcp_count();
>+  ASSERT_TRUE(dtls_srtp_transport1_->SendRtcpPacket(
>+      &rtcp_packet, rtc::PacketOptions(), cricket::PF_SRTP_BYPASS));
>+  // The RTCP packet is not exepected to be received because the SRTP parameters
>+  // are only reset on one side and the SRTCP index is out of sync.
>+  EXPECT_EQ(prev_received_packets, transport_observer2_.rtcp_count());
>+
>+  // Set the flag to be true on the other side.
>+  dtls_srtp_transport2_->SetActiveResetSrtpParams(true);
>+  // Set RTCP transport to null to trigger the SRTP parameters update.
>+  dtls_srtp_transport1_->SetDtlsTransports(rtp_dtls1.get(), nullptr);
>+  dtls_srtp_transport2_->SetDtlsTransports(rtp_dtls2.get(), nullptr);
>+
>+  // RTCP packets flow is expected to work just fine.
>+  SendRecvRtcpPackets();
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/dtmfsender.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/dtmfsender.cc
>index 82644aacf0a..709612095e3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/dtmfsender.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/dtmfsender.cc
>@@ -41,11 +41,13 @@ static const char kDtmfValidTones[] = ",0123456789*#ABCDabcd";
> static const char kDtmfTonesTable[] = ",0123456789*#ABCD";
> // The duration cannot be more than 6000ms or less than 40ms. The gap between
> // tones must be at least 50 ms.
>+// Source for values: W3C WEBRTC specification.
>+// https://w3c.github.io/webrtc-pc/#dom-rtcdtmfsender-insertdtmf
> static const int kDtmfDefaultDurationMs = 100;
> static const int kDtmfMinDurationMs = 40;
> static const int kDtmfMaxDurationMs = 6000;
> static const int kDtmfDefaultGapMs = 50;
>-static const int kDtmfMinGapMs = 50;
>+static const int kDtmfMinGapMs = 30;
> 
> // Get DTMF code from the DTMF event character.
> bool GetDtmfCode(char tone, int* code) {
>@@ -60,32 +62,28 @@ bool GetDtmfCode(char tone, int* code) {
> }
> 
> rtc::scoped_refptr<DtmfSender> DtmfSender::Create(
>-    AudioTrackInterface* track,
>     rtc::Thread* signaling_thread,
>     DtmfProviderInterface* provider) {
>   if (!signaling_thread) {
>     return nullptr;
>   }
>   rtc::scoped_refptr<DtmfSender> dtmf_sender(
>-      new rtc::RefCountedObject<DtmfSender>(track, signaling_thread,
>-                                                  provider));
>+      new rtc::RefCountedObject<DtmfSender>(signaling_thread, provider));
>   return dtmf_sender;
> }
> 
>-DtmfSender::DtmfSender(AudioTrackInterface* track,
>-                       rtc::Thread* signaling_thread,
>+DtmfSender::DtmfSender(rtc::Thread* signaling_thread,
>                        DtmfProviderInterface* provider)
>-    : track_(track),
>-      observer_(NULL),
>+    : observer_(nullptr),
>       signaling_thread_(signaling_thread),
>       provider_(provider),
>       duration_(kDtmfDefaultDurationMs),
>       inter_tone_gap_(kDtmfDefaultGapMs) {
>-  RTC_DCHECK(signaling_thread_ != NULL);
>+  RTC_DCHECK(signaling_thread_);
>   // TODO(deadbeef): Once we can use shared_ptr and weak_ptr,
>   // do that instead of relying on a "destroyed" signal.
>   if (provider_) {
>-    RTC_DCHECK(provider_->GetOnDestroyedSignal() != NULL);
>+    RTC_DCHECK(provider_->GetOnDestroyedSignal());
>     provider_->GetOnDestroyedSignal()->connect(
>         this, &DtmfSender::OnProviderDestroyed);
>   }
>@@ -100,7 +98,7 @@ void DtmfSender::RegisterObserver(DtmfSenderObserverInterface* observer) {
> }
> 
> void DtmfSender::UnregisterObserver() {
>-  observer_ = NULL;
>+  observer_ = nullptr;
> }
> 
> bool DtmfSender::CanInsertDtmf() {
>@@ -111,18 +109,19 @@ bool DtmfSender::CanInsertDtmf() {
>   return provider_->CanInsertDtmf();
> }
> 
>-bool DtmfSender::InsertDtmf(const std::string& tones, int duration,
>+bool DtmfSender::InsertDtmf(const std::string& tones,
>+                            int duration,
>                             int inter_tone_gap) {
>   RTC_DCHECK(signaling_thread_->IsCurrent());
> 
>-  if (duration > kDtmfMaxDurationMs ||
>-      duration < kDtmfMinDurationMs ||
>+  if (duration > kDtmfMaxDurationMs || duration < kDtmfMinDurationMs ||
>       inter_tone_gap < kDtmfMinGapMs) {
>     RTC_LOG(LS_ERROR)
>         << "InsertDtmf is called with invalid duration or tones gap. "
>-        << "The duration cannot be more than " << kDtmfMaxDurationMs
>-        << "ms or less than " << kDtmfMinDurationMs << "ms. "
>-        << "The gap between tones must be at least " << kDtmfMinGapMs << "ms.";
>+           "The duration cannot be more than "
>+        << kDtmfMaxDurationMs << "ms or less than " << kDtmfMinDurationMs
>+        << "ms. The gap between tones must be at least " << kDtmfMinGapMs
>+        << "ms.";
>     return false;
>   }
> 
>@@ -138,14 +137,10 @@ bool DtmfSender::InsertDtmf(const std::string& tones, int duration,
>   // Clear the previous queue.
>   signaling_thread_->Clear(this, MSG_DO_INSERT_DTMF);
>   // Kick off a new DTMF task queue.
>-  signaling_thread_->Post(RTC_FROM_HERE, this, MSG_DO_INSERT_DTMF);
>+  signaling_thread_->PostDelayed(RTC_FROM_HERE, 1, this, MSG_DO_INSERT_DTMF);
>   return true;
> }
> 
>-const AudioTrackInterface* DtmfSender::track() const {
>-  return track_;
>-}
>-
> std::string DtmfSender::tones() const {
>   return tones_;
> }
>@@ -230,7 +225,7 @@ void DtmfSender::DoInsertDtmf() {
> void DtmfSender::OnProviderDestroyed() {
>   RTC_LOG(LS_INFO) << "The Dtmf provider is deleted. Clear the sending queue.";
>   StopSending();
>-  provider_ = NULL;
>+  provider_ = nullptr;
> }
> 
> void DtmfSender::StopSending() {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/dtmfsender.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/dtmfsender.h
>index ff7f0ac0490..7b45a1f7223 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/dtmfsender.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/dtmfsender.h
>@@ -14,20 +14,16 @@
> #include <string>
> 
> #include "api/dtmfsenderinterface.h"
>-#include "api/mediastreaminterface.h"
> #include "api/proxy.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/messagehandler.h"
> #include "rtc_base/refcount.h"
>+#include "rtc_base/thread.h"
> 
> // DtmfSender is the native implementation of the RTCDTMFSender defined by
> // the WebRTC W3C Editor's Draft.
> // http://dev.w3.org/2011/webrtc/editor/webrtc.html
> 
>-namespace rtc {
>-class Thread;
>-}
>-
> namespace webrtc {
> 
> // This interface is called by DtmfSender to talk to the actual audio channel
>@@ -49,17 +45,12 @@ class DtmfProviderInterface {
>   virtual ~DtmfProviderInterface() {}
> };
> 
>-class DtmfSender
>-    : public DtmfSenderInterface,
>-      public sigslot::has_slots<>,
>-      public rtc::MessageHandler {
>+class DtmfSender : public DtmfSenderInterface,
>+                   public sigslot::has_slots<>,
>+                   public rtc::MessageHandler {
>  public:
>-  // |track| is only there for backwards compatibility, since there's a track
>-  // accessor method.
>-  static rtc::scoped_refptr<DtmfSender> Create(
>-      AudioTrackInterface* track,
>-      rtc::Thread* signaling_thread,
>-      DtmfProviderInterface* provider);
>+  static rtc::scoped_refptr<DtmfSender> Create(rtc::Thread* signaling_thread,
>+                                               DtmfProviderInterface* provider);
> 
>   // Implements DtmfSenderInterface.
>   void RegisterObserver(DtmfSenderObserverInterface* observer) override;
>@@ -68,15 +59,12 @@ class DtmfSender
>   bool InsertDtmf(const std::string& tones,
>                   int duration,
>                   int inter_tone_gap) override;
>-  const AudioTrackInterface* track() const override;
>   std::string tones() const override;
>   int duration() const override;
>   int inter_tone_gap() const override;
> 
>  protected:
>-  DtmfSender(AudioTrackInterface* track,
>-             rtc::Thread* signaling_thread,
>-             DtmfProviderInterface* provider);
>+  DtmfSender(rtc::Thread* signaling_thread, DtmfProviderInterface* provider);
>   virtual ~DtmfSender();
> 
>  private:
>@@ -92,7 +80,6 @@ class DtmfSender
> 
>   void StopSending();
> 
>-  rtc::scoped_refptr<AudioTrackInterface> track_;
>   DtmfSenderObserverInterface* observer_;
>   rtc::Thread* signaling_thread_;
>   DtmfProviderInterface* provider_;
>@@ -105,15 +92,14 @@ class DtmfSender
> 
> // Define proxy for DtmfSenderInterface.
> BEGIN_SIGNALING_PROXY_MAP(DtmfSender)
>-  PROXY_SIGNALING_THREAD_DESTRUCTOR()
>-  PROXY_METHOD1(void, RegisterObserver, DtmfSenderObserverInterface*)
>-  PROXY_METHOD0(void, UnregisterObserver)
>-  PROXY_METHOD0(bool, CanInsertDtmf)
>-  PROXY_METHOD3(bool, InsertDtmf, const std::string&, int, int)
>-  PROXY_CONSTMETHOD0(const AudioTrackInterface*, track)
>-  PROXY_CONSTMETHOD0(std::string, tones)
>-  PROXY_CONSTMETHOD0(int, duration)
>-  PROXY_CONSTMETHOD0(int, inter_tone_gap)
>+PROXY_SIGNALING_THREAD_DESTRUCTOR()
>+PROXY_METHOD1(void, RegisterObserver, DtmfSenderObserverInterface*)
>+PROXY_METHOD0(void, UnregisterObserver)
>+PROXY_METHOD0(bool, CanInsertDtmf)
>+PROXY_METHOD3(bool, InsertDtmf, const std::string&, int, int)
>+PROXY_CONSTMETHOD0(std::string, tones)
>+PROXY_CONSTMETHOD0(int, duration)
>+PROXY_CONSTMETHOD0(int, inter_tone_gap)
> END_PROXY_MAP()
> 
> // Get DTMF code from the DTMF event character.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/dtmfsender_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/dtmfsender_unittest.cc
>index a5d102b5f2e..dc8cc115b8b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/dtmfsender_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/dtmfsender_unittest.cc
>@@ -15,19 +15,14 @@
> #include <string>
> #include <vector>
> 
>-
>-#include "pc/audiotrack.h"
> #include "rtc_base/fakeclock.h"
> #include "rtc_base/gunit.h"
> #include "rtc_base/timeutils.h"
> 
>-using webrtc::AudioTrackInterface;
>-using webrtc::AudioTrack;
> using webrtc::DtmfProviderInterface;
> using webrtc::DtmfSender;
> using webrtc::DtmfSenderObserverInterface;
> 
>-static const char kTestAudioLabel[] = "test_audio_track";
> // TODO(deadbeef): Even though this test now uses a fake clock, it has a
> // generous 3-second timeout for every test case. The timeout could be tuned
> // to each test based on the tones sent, instead.
>@@ -46,12 +41,8 @@ class FakeDtmfObserver : public DtmfSenderObserverInterface {
>   }
> 
>   // getters
>-  const std::vector<std::string>& tones() const {
>-    return tones_;
>-  }
>-  bool completed() const {
>-    return completed_;
>-  }
>+  const std::vector<std::string>& tones() const { return tones_; }
>+  bool completed() const { return completed_; }
> 
>  private:
>   std::vector<std::string> tones_;
>@@ -62,9 +53,7 @@ class FakeDtmfProvider : public DtmfProviderInterface {
>  public:
>   struct DtmfInfo {
>     DtmfInfo(int code, int duration, int gap)
>-      : code(code),
>-        duration(duration),
>-        gap(gap) {}
>+        : code(code), duration(duration), gap(gap) {}
>     int code;
>     int duration;
>     int gap;
>@@ -72,9 +61,7 @@ class FakeDtmfProvider : public DtmfProviderInterface {
> 
>   FakeDtmfProvider() : last_insert_dtmf_call_(0) {}
> 
>-  ~FakeDtmfProvider() {
>-    SignalDestroyed();
>-  }
>+  ~FakeDtmfProvider() { SignalDestroyed(); }
> 
>   // Implements DtmfProviderInterface.
>   bool CanInsertDtmf() override { return can_insert_; }
>@@ -114,12 +101,10 @@ class FakeDtmfProvider : public DtmfProviderInterface {
> class DtmfSenderTest : public testing::Test {
>  protected:
>   DtmfSenderTest()
>-      : track_(AudioTrack::Create(kTestAudioLabel, NULL)),
>-        observer_(new rtc::RefCountedObject<FakeDtmfObserver>()),
>+      : observer_(new rtc::RefCountedObject<FakeDtmfObserver>()),
>         provider_(new FakeDtmfProvider()) {
>     provider_->SetCanInsertDtmf(true);
>-    dtmf_ = DtmfSender::Create(track_, rtc::Thread::Current(),
>-                               provider_.get());
>+    dtmf_ = DtmfSender::Create(rtc::Thread::Current(), provider_.get());
>     dtmf_->RegisterObserver(observer_.get());
>   }
> 
>@@ -131,7 +116,8 @@ class DtmfSenderTest : public testing::Test {
> 
>   // Constructs a list of DtmfInfo from |tones|, |duration| and
>   // |inter_tone_gap|.
>-  void GetDtmfInfoFromString(const std::string& tones, int duration,
>+  void GetDtmfInfoFromString(const std::string& tones,
>+                             int duration,
>                              int inter_tone_gap,
>                              std::vector<FakeDtmfProvider::DtmfInfo>* dtmfs) {
>     // Init extra_delay as -inter_tone_gap - duration to ensure the first
>@@ -146,24 +132,24 @@ class DtmfSenderTest : public testing::Test {
>       if (tone == ',') {
>         extra_delay = 2000;  // 2 seconds
>       } else {
>-        dtmfs->push_back(FakeDtmfProvider::DtmfInfo(code, duration,
>-                         duration + inter_tone_gap + extra_delay));
>+        dtmfs->push_back(FakeDtmfProvider::DtmfInfo(
>+            code, duration, duration + inter_tone_gap + extra_delay));
>         extra_delay = 0;
>       }
>     }
>   }
> 
>-  void VerifyExpectedState(AudioTrackInterface* track,
>-                          const std::string& tones,
>-                          int duration, int inter_tone_gap) {
>-    EXPECT_EQ(track, dtmf_->track());
>+  void VerifyExpectedState(const std::string& tones,
>+                           int duration,
>+                           int inter_tone_gap) {
>     EXPECT_EQ(tones, dtmf_->tones());
>     EXPECT_EQ(duration, dtmf_->duration());
>     EXPECT_EQ(inter_tone_gap, dtmf_->inter_tone_gap());
>   }
> 
>   // Verify the provider got all the expected calls.
>-  void VerifyOnProvider(const std::string& tones, int duration,
>+  void VerifyOnProvider(const std::string& tones,
>+                        int duration,
>                         int inter_tone_gap) {
>     std::vector<FakeDtmfProvider::DtmfInfo> dtmf_queue_ref;
>     GetDtmfInfoFromString(tones, duration, inter_tone_gap, &dtmf_queue_ref);
>@@ -205,7 +191,6 @@ class DtmfSenderTest : public testing::Test {
>     }
>   }
> 
>-  rtc::scoped_refptr<AudioTrackInterface> track_;
>   std::unique_ptr<FakeDtmfObserver> observer_;
>   std::unique_ptr<FakeDtmfProvider> provider_;
>   rtc::scoped_refptr<DtmfSender> dtmf_;
>@@ -237,14 +222,14 @@ TEST_F(DtmfSenderTest, InsertDtmfTwice) {
>   int duration = 100;
>   int inter_tone_gap = 50;
>   EXPECT_TRUE(dtmf_->InsertDtmf(tones1, duration, inter_tone_gap));
>-  VerifyExpectedState(track_, tones1, duration, inter_tone_gap);
>+  VerifyExpectedState(tones1, duration, inter_tone_gap);
>   // Wait until the first tone got sent.
>   EXPECT_TRUE_SIMULATED_WAIT(observer_->tones().size() == 1, kMaxWaitMs,
>                              fake_clock_);
>-  VerifyExpectedState(track_, "2", duration, inter_tone_gap);
>+  VerifyExpectedState("2", duration, inter_tone_gap);
>   // Insert with another tone buffer.
>   EXPECT_TRUE(dtmf_->InsertDtmf(tones2, duration, inter_tone_gap));
>-  VerifyExpectedState(track_, tones2, duration, inter_tone_gap);
>+  VerifyExpectedState(tones2, duration, inter_tone_gap);
>   // Wait until it's completed.
>   EXPECT_TRUE_SIMULATED_WAIT(observer_->completed(), kMaxWaitMs, fake_clock_);
> 
>@@ -331,7 +316,19 @@ TEST_F(DtmfSenderTest, InsertDtmfWithInvalidDurationOrGap) {
> 
>   EXPECT_FALSE(dtmf_->InsertDtmf(tones, 6001, inter_tone_gap));
>   EXPECT_FALSE(dtmf_->InsertDtmf(tones, 39, inter_tone_gap));
>-  EXPECT_FALSE(dtmf_->InsertDtmf(tones, duration, 49));
>+  EXPECT_FALSE(dtmf_->InsertDtmf(tones, duration, 29));
>+
>+  EXPECT_TRUE(dtmf_->InsertDtmf(tones, duration, inter_tone_gap));
>+}
> 
>+TEST_F(DtmfSenderTest, InsertDtmfSendsAfterWait) {
>+  std::string tones = "ABC";
>+  int duration = 100;
>+  int inter_tone_gap = 50;
>   EXPECT_TRUE(dtmf_->InsertDtmf(tones, duration, inter_tone_gap));
>+  VerifyExpectedState("ABC", duration, inter_tone_gap);
>+  // Wait until the first tone got sent.
>+  EXPECT_TRUE_SIMULATED_WAIT(observer_->tones().size() == 1, kMaxWaitMs,
>+                             fake_clock_);
>+  VerifyExpectedState("BC", duration, inter_tone_gap);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/externalhmac.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/externalhmac.cc
>index 899c33822ca..d4383f1f4c4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/externalhmac.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/externalhmac.cc
>@@ -13,34 +13,32 @@
> #include <stdlib.h>  // For malloc/free.
> 
> #include "rtc_base/logging.h"
>+#include "rtc_base/zero_memory.h"
> 
> #include "third_party/libsrtp/crypto/include/crypto_kernel.h"
> #include "third_party/libsrtp/include/srtp.h"
> 
> // Begin test case 0 */
> static const uint8_t kExternalHmacTestCase0Key[20] = {
>-  0x0b, 0x0b, 0x0b, 0x0b, 0x0b, 0x0b, 0x0b, 0x0b,
>-  0x0b, 0x0b, 0x0b, 0x0b, 0x0b, 0x0b, 0x0b, 0x0b,
>-  0x0b, 0x0b, 0x0b, 0x0b
>-};
>+    0x0b, 0x0b, 0x0b, 0x0b, 0x0b, 0x0b, 0x0b, 0x0b, 0x0b, 0x0b,
>+    0x0b, 0x0b, 0x0b, 0x0b, 0x0b, 0x0b, 0x0b, 0x0b, 0x0b, 0x0b};
> 
> static const uint8_t kExternalHmacTestCase0Data[8] = {
>-  0x48, 0x69, 0x20, 0x54, 0x68, 0x65, 0x72, 0x65   // "Hi There"
>+    0x48, 0x69, 0x20, 0x54, 0x68, 0x65, 0x72, 0x65  // "Hi There"
> };
> 
>-static const uint8_t kExternalHmacFakeTag[10] = {
>-  0xba, 0xdd, 0xba, 0xdd, 0xba, 0xdd, 0xba, 0xdd, 0xba, 0xdd
>-};
>+static const uint8_t kExternalHmacFakeTag[10] = {0xba, 0xdd, 0xba, 0xdd, 0xba,
>+                                                 0xdd, 0xba, 0xdd, 0xba, 0xdd};
> 
> static const srtp_auth_test_case_t kExternalHmacTestCase0 = {
>-  20,                                                    // Octets in key
>-  const_cast<uint8_t*>(kExternalHmacTestCase0Key),   // Key
>-  8,                                                     // Octets in data
>-  const_cast<uint8_t*>(kExternalHmacTestCase0Data),  // Data
>-  10,                                                    // Octets in tag
>-  const_cast<uint8_t*>(kExternalHmacFakeTag),          // Tag
>-  NULL                                                   // Pointer to next
>-                                                         // testcase
>+    20,                                                // Octets in key
>+    const_cast<uint8_t*>(kExternalHmacTestCase0Key),   // Key
>+    8,                                                 // Octets in data
>+    const_cast<uint8_t*>(kExternalHmacTestCase0Data),  // Data
>+    10,                                                // Octets in tag
>+    const_cast<uint8_t*>(kExternalHmacFakeTag),        // Tag
>+    NULL                                               // Pointer to next
>+                                                       // testcase
> };
> 
> static const char kExternalHmacDescription[] =
>@@ -48,17 +46,16 @@ static const char kExternalHmacDescription[] =
> 
> // srtp_auth_type_t external_hmac is the hmac metaobject
> 
>-static const srtp_auth_type_t external_hmac  = {
>-  external_hmac_alloc,
>-  external_hmac_dealloc,
>-  external_hmac_init,
>-  external_hmac_compute,
>-  external_hmac_update,
>-  external_hmac_start,
>-  const_cast<char*>(kExternalHmacDescription),
>-  const_cast<srtp_auth_test_case_t*>(&kExternalHmacTestCase0),
>-  EXTERNAL_HMAC_SHA1
>-};
>+static const srtp_auth_type_t external_hmac = {
>+    external_hmac_alloc,
>+    external_hmac_dealloc,
>+    external_hmac_init,
>+    external_hmac_compute,
>+    external_hmac_update,
>+    external_hmac_start,
>+    const_cast<char*>(kExternalHmacDescription),
>+    const_cast<srtp_auth_test_case_t*>(&kExternalHmacTestCase0),
>+    EXTERNAL_HMAC_SHA1};
> 
> srtp_err_status_t external_hmac_alloc(srtp_auth_t** a,
>                                       int key_len,
>@@ -94,9 +91,7 @@ srtp_err_status_t external_hmac_alloc(srtp_auth_t** a,
> }
> 
> srtp_err_status_t external_hmac_dealloc(srtp_auth_t* a) {
>-  // Zeroize entire state
>-  memset(reinterpret_cast<uint8_t*>(a), 0,
>-         sizeof(ExternalHmacContext) + sizeof(srtp_auth_t));
>+  rtc::ExplicitZeroMemory(a, sizeof(ExternalHmacContext) + sizeof(srtp_auth_t));
> 
>   // Free memory
>   delete[] a;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/iceserverparsing.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/iceserverparsing.cc
>index b260954f04a..806fb3bc215 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/iceserverparsing.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/iceserverparsing.cc
>@@ -231,6 +231,7 @@ static RTCErrorType ParseIceServerUrl(
>       if (username.empty() || server.password.empty()) {
>         // The WebRTC spec requires throwing an InvalidAccessError when username
>         // or credential are ommitted; this is the native equivalent.
>+        RTC_LOG(LS_ERROR) << "TURN URL without username, or password empty";
>         return RTCErrorType::INVALID_PARAMETER;
>       }
>       // If the hostname field is not empty, then the server address must be
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jsepicecandidate.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jsepicecandidate.cc
>index 48feb89a69d..277f1bab4ff 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jsepicecandidate.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jsepicecandidate.cc
>@@ -12,6 +12,7 @@
> 
> #include <vector>
> 
>+#include "absl/memory/memory.h"
> #include "pc/webrtcsdp.h"
> #include "rtc_base/stringencode.h"
> 
>@@ -29,22 +30,26 @@ IceCandidateInterface* CreateIceCandidate(const std::string& sdp_mid,
>   return jsep_ice;
> }
> 
>+std::unique_ptr<IceCandidateInterface> CreateIceCandidate(
>+    const std::string& sdp_mid,
>+    int sdp_mline_index,
>+    const cricket::Candidate& candidate) {
>+  return absl::make_unique<JsepIceCandidate>(sdp_mid, sdp_mline_index,
>+                                             candidate);
>+}
>+
> JsepIceCandidate::JsepIceCandidate(const std::string& sdp_mid,
>                                    int sdp_mline_index)
>-    : sdp_mid_(sdp_mid),
>-      sdp_mline_index_(sdp_mline_index) {
>-}
>+    : sdp_mid_(sdp_mid), sdp_mline_index_(sdp_mline_index) {}
> 
> JsepIceCandidate::JsepIceCandidate(const std::string& sdp_mid,
>                                    int sdp_mline_index,
>                                    const cricket::Candidate& candidate)
>     : sdp_mid_(sdp_mid),
>       sdp_mline_index_(sdp_mline_index),
>-      candidate_(candidate) {
>-}
>+      candidate_(candidate) {}
> 
>-JsepIceCandidate::~JsepIceCandidate() {
>-}
>+JsepIceCandidate::~JsepIceCandidate() {}
> 
> bool JsepIceCandidate::Initialize(const std::string& sdp, SdpParseError* err) {
>   return SdpDeserializeCandidate(sdp, this, err);
>@@ -57,39 +62,4 @@ bool JsepIceCandidate::ToString(std::string* out) const {
>   return !out->empty();
> }
> 
>-JsepCandidateCollection::~JsepCandidateCollection() {
>-  for (std::vector<JsepIceCandidate*>::iterator it = candidates_.begin();
>-       it != candidates_.end(); ++it) {
>-    delete *it;
>-  }
>-}
>-
>-bool JsepCandidateCollection::HasCandidate(
>-    const IceCandidateInterface* candidate) const {
>-  bool ret = false;
>-  for (std::vector<JsepIceCandidate*>::const_iterator it = candidates_.begin();
>-      it != candidates_.end(); ++it) {
>-    if ((*it)->sdp_mid() == candidate->sdp_mid() &&
>-        (*it)->sdp_mline_index() == candidate->sdp_mline_index() &&
>-        (*it)->candidate().IsEquivalent(candidate->candidate())) {
>-      ret = true;
>-      break;
>-    }
>-  }
>-  return ret;
>-}
>-
>-size_t JsepCandidateCollection::remove(const cricket::Candidate& candidate) {
>-  auto iter = std::find_if(candidates_.begin(), candidates_.end(),
>-                           [candidate](JsepIceCandidate* c) {
>-                             return candidate.MatchesForRemoval(c->candidate());
>-                           });
>-  if (iter != candidates_.end()) {
>-    delete *iter;
>-    candidates_.erase(iter);
>-    return 1;
>-  }
>-  return 0;
>-}
>-
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jsepsessiondescription.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jsepsessiondescription.cc
>index 91840e36547..64a7a2bd591 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jsepsessiondescription.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jsepsessiondescription.cc
>@@ -12,11 +12,11 @@
> 
> #include <memory>
> 
>+#include "absl/memory/memory.h"
> #include "p2p/base/port.h"
> #include "pc/mediasession.h"
> #include "pc/webrtcsdp.h"
> #include "rtc_base/arraysize.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/stringencode.h"
> 
> using cricket::SessionDescription;
>@@ -112,7 +112,7 @@ const char* SdpTypeToString(SdpType type) {
>   return "";
> }
> 
>-rtc::Optional<SdpType> SdpTypeFromString(const std::string& type_str) {
>+absl::optional<SdpType> SdpTypeFromString(const std::string& type_str) {
>   if (type_str == SessionDescriptionInterface::kOffer) {
>     return SdpType::kOffer;
>   } else if (type_str == SessionDescriptionInterface::kPrAnswer) {
>@@ -120,14 +120,14 @@ rtc::Optional<SdpType> SdpTypeFromString(const std::string& type_str) {
>   } else if (type_str == SessionDescriptionInterface::kAnswer) {
>     return SdpType::kAnswer;
>   } else {
>-    return rtc::nullopt;
>+    return absl::nullopt;
>   }
> }
> 
> // TODO(steveanton): Remove this default implementation once Chromium has been
> // updated.
> SdpType SessionDescriptionInterface::GetType() const {
>-  rtc::Optional<SdpType> maybe_type = SdpTypeFromString(type());
>+  absl::optional<SdpType> maybe_type = SdpTypeFromString(type());
>   if (maybe_type) {
>     return *maybe_type;
>   } else {
>@@ -142,7 +142,7 @@ SdpType SessionDescriptionInterface::GetType() const {
> SessionDescriptionInterface* CreateSessionDescription(const std::string& type,
>                                                       const std::string& sdp,
>                                                       SdpParseError* error) {
>-  rtc::Optional<SdpType> maybe_type = SdpTypeFromString(type);
>+  absl::optional<SdpType> maybe_type = SdpTypeFromString(type);
>   if (!maybe_type) {
>     return nullptr;
>   }
>@@ -160,17 +160,29 @@ std::unique_ptr<SessionDescriptionInterface> CreateSessionDescription(
>     SdpType type,
>     const std::string& sdp,
>     SdpParseError* error_out) {
>-  auto jsep_desc = rtc::MakeUnique<JsepSessionDescription>(type);
>+  auto jsep_desc = absl::make_unique<JsepSessionDescription>(type);
>   if (!SdpDeserialize(sdp, jsep_desc.get(), error_out)) {
>     return nullptr;
>   }
>   return std::move(jsep_desc);
> }
> 
>+std::unique_ptr<SessionDescriptionInterface> CreateSessionDescription(
>+    SdpType type,
>+    const std::string& session_id,
>+    const std::string& session_version,
>+    std::unique_ptr<cricket::SessionDescription> description) {
>+  auto jsep_description = absl::make_unique<JsepSessionDescription>(type);
>+  bool initialize_success = jsep_description->Initialize(
>+      description.release(), session_id, session_version);
>+  RTC_DCHECK(initialize_success);
>+  return std::move(jsep_description);
>+}
>+
> JsepSessionDescription::JsepSessionDescription(SdpType type) : type_(type) {}
> 
> JsepSessionDescription::JsepSessionDescription(const std::string& type) {
>-  rtc::Optional<SdpType> maybe_type = SdpTypeFromString(type);
>+  absl::optional<SdpType> maybe_type = SdpTypeFromString(type);
>   if (maybe_type) {
>     type_ = *maybe_type;
>   } else {
>@@ -273,7 +285,7 @@ bool JsepSessionDescription::ToString(std::string* out) const {
>   if (!description_ || !out) {
>     return false;
>   }
>-  *out = SdpSerialize(*this, false);
>+  *out = SdpSerialize(*this);
>   return !out->empty();
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jsepsessiondescription_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jsepsessiondescription_unittest.cc
>index fbcfcb7161e..d9e2588a551 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jsepsessiondescription_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jsepsessiondescription_unittest.cc
>@@ -11,6 +11,7 @@
> #include <memory>
> #include <string>
> 
>+#include "absl/memory/memory.h"
> #include "api/candidate.h"
> #include "api/jsepicecandidate.h"
> #include "api/jsepsessiondescription.h"
>@@ -20,7 +21,6 @@
> #include "pc/sessiondescription.h"
> #include "pc/webrtcsdp.h"
> #include "rtc_base/gunit.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/stringencode.h"
> 
> using cricket::MediaProtocolType;
>@@ -81,13 +81,11 @@ class JsepSessionDescriptionTest : public testing::Test {
>     cricket::Candidate candidate(cricket::ICE_CANDIDATE_COMPONENT_RTP, "udp",
>                                  address, 1, "", "", "local", 0, "1");
>     candidate_ = candidate;
>-    const std::string session_id =
>-        rtc::ToString(rtc::CreateRandomId64());
>-    const std::string session_version =
>-        rtc::ToString(rtc::CreateRandomId());
>-    jsep_desc_ = rtc::MakeUnique<JsepSessionDescription>(SdpType::kOffer);
>+    const std::string session_id = rtc::ToString(rtc::CreateRandomId64());
>+    const std::string session_version = rtc::ToString(rtc::CreateRandomId());
>+    jsep_desc_ = absl::make_unique<JsepSessionDescription>(SdpType::kOffer);
>     ASSERT_TRUE(jsep_desc_->Initialize(CreateCricketSessionDescription(),
>-        session_id, session_version));
>+                                       session_id, session_version));
>   }
> 
>   std::string Serialize(const SessionDescriptionInterface* desc) {
>@@ -99,7 +97,7 @@ class JsepSessionDescriptionTest : public testing::Test {
> 
>   std::unique_ptr<SessionDescriptionInterface> DeSerialize(
>       const std::string& sdp) {
>-    auto jsep_desc = rtc::MakeUnique<JsepSessionDescription>(SdpType::kOffer);
>+    auto jsep_desc = absl::make_unique<JsepSessionDescription>(SdpType::kOffer);
>     EXPECT_TRUE(webrtc::SdpDeserialize(sdp, jsep_desc.get(), nullptr));
>     return std::move(jsep_desc);
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jseptransport.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jseptransport.cc
>index 12aad7a3f07..c3dfd32464d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jseptransport.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jseptransport.cc
>@@ -1,5 +1,5 @@
> /*
>- *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>  *
>  *  Use of this source code is governed by a BSD-style license
>  *  that can be found in the LICENSE file in the root of the source
>@@ -13,6 +13,7 @@
> #include <memory>
> #include <utility>  // for std::pair
> 
>+#include "absl/memory/memory.h"
> #include "api/candidate.h"
> #include "p2p/base/p2pconstants.h"
> #include "p2p/base/p2ptransportchannel.h"
>@@ -20,155 +21,170 @@
> #include "rtc_base/bind.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>+#include "rtc_base/strings/string_builder.h"
> 
> using webrtc::SdpType;
> 
> namespace cricket {
> 
>-TransportChannelStats::TransportChannelStats() = default;
>-
>-TransportChannelStats::TransportChannelStats(const TransportChannelStats&) =
>-    default;
>-
>-TransportChannelStats::~TransportChannelStats() = default;
>-
>-TransportStats::TransportStats() = default;
>-
>-TransportStats::~TransportStats() = default;
>-
>-bool BadTransportDescription(const std::string& desc, std::string* err_desc) {
>-  if (err_desc) {
>-    *err_desc = desc;
>-  }
>-  RTC_LOG(LS_ERROR) << desc;
>-  return false;
>-}
>-
>-static bool VerifyIceParams(const TransportDescription& desc) {
>+static bool VerifyIceParams(const JsepTransportDescription& jsep_description) {
>   // For legacy protocols.
>-  if (desc.ice_ufrag.empty() && desc.ice_pwd.empty())
>+  // TODO(zhihuang): Remove this once the legacy protocol is no longer
>+  // supported.
>+  if (jsep_description.transport_desc.ice_ufrag.empty() &&
>+      jsep_description.transport_desc.ice_pwd.empty()) {
>     return true;
>+  }
> 
>-  if (desc.ice_ufrag.length() < ICE_UFRAG_MIN_LENGTH ||
>-      desc.ice_ufrag.length() > ICE_UFRAG_MAX_LENGTH) {
>+  if (jsep_description.transport_desc.ice_ufrag.length() <
>+          ICE_UFRAG_MIN_LENGTH ||
>+      jsep_description.transport_desc.ice_ufrag.length() >
>+          ICE_UFRAG_MAX_LENGTH) {
>     return false;
>   }
>-  if (desc.ice_pwd.length() < ICE_PWD_MIN_LENGTH ||
>-      desc.ice_pwd.length() > ICE_PWD_MAX_LENGTH) {
>+  if (jsep_description.transport_desc.ice_pwd.length() < ICE_PWD_MIN_LENGTH ||
>+      jsep_description.transport_desc.ice_pwd.length() > ICE_PWD_MAX_LENGTH) {
>     return false;
>   }
>   return true;
> }
> 
>-JsepTransport::JsepTransport(
>-    const std::string& mid,
>-    const rtc::scoped_refptr<rtc::RTCCertificate>& certificate)
>-    : mid_(mid), certificate_(certificate) {}
>-
>-JsepTransport::~JsepTransport() = default;
>-
>-bool JsepTransport::AddChannel(DtlsTransportInternal* dtls, int component) {
>-  if (channels_.find(component) != channels_.end()) {
>-    RTC_LOG(LS_ERROR) << "Adding channel for component " << component
>-                      << " twice.";
>-    return false;
>-  }
>-  channels_[component] = dtls;
>-  // Something's wrong if a channel is being added after a description is set.
>-  // This may currently occur if rtcp-mux is negotiated, then a new m= section
>-  // is added in a later offer/answer. But this is suboptimal and should be
>-  // changed; we shouldn't support going from muxed to non-muxed.
>-  // TODO(deadbeef): Once this is fixed, make the warning an error, and remove
>-  // the calls to "ApplyXTransportDescription" below.
>-  if (local_description_set_ || remote_description_set_) {
>-    RTC_LOG(LS_WARNING) << "Adding new transport channel after "
>-                           "transport description already applied.";
>-  }
>-  bool ret = true;
>-  std::string err;
>-  if (local_description_set_) {
>-    ret &= ApplyLocalTransportDescription(channels_[component], &err);
>-  }
>-  if (remote_description_set_) {
>-    ret &= ApplyRemoteTransportDescription(channels_[component], &err);
>-  }
>-  if (local_description_set_ && remote_description_set_) {
>-    ret &= ApplyNegotiatedTransportDescription(channels_[component], &err);
>-  }
>-  return ret;
>+JsepTransportDescription::JsepTransportDescription() {}
>+
>+JsepTransportDescription::JsepTransportDescription(
>+    bool rtcp_mux_enabled,
>+    const std::vector<CryptoParams>& cryptos,
>+    const std::vector<int>& encrypted_header_extension_ids,
>+    int rtp_abs_sendtime_extn_id,
>+    const TransportDescription& transport_desc)
>+    : rtcp_mux_enabled(rtcp_mux_enabled),
>+      cryptos(cryptos),
>+      encrypted_header_extension_ids(encrypted_header_extension_ids),
>+      rtp_abs_sendtime_extn_id(rtp_abs_sendtime_extn_id),
>+      transport_desc(transport_desc) {}
>+
>+JsepTransportDescription::JsepTransportDescription(
>+    const JsepTransportDescription& from)
>+    : rtcp_mux_enabled(from.rtcp_mux_enabled),
>+      cryptos(from.cryptos),
>+      encrypted_header_extension_ids(from.encrypted_header_extension_ids),
>+      rtp_abs_sendtime_extn_id(from.rtp_abs_sendtime_extn_id),
>+      transport_desc(from.transport_desc) {}
>+
>+JsepTransportDescription::~JsepTransportDescription() = default;
>+
>+JsepTransportDescription& JsepTransportDescription::operator=(
>+    const JsepTransportDescription& from) {
>+  if (this == &from) {
>+    return *this;
>+  }
>+  rtcp_mux_enabled = from.rtcp_mux_enabled;
>+  cryptos = from.cryptos;
>+  encrypted_header_extension_ids = from.encrypted_header_extension_ids;
>+  rtp_abs_sendtime_extn_id = from.rtp_abs_sendtime_extn_id;
>+  transport_desc = from.transport_desc;
>+
>+  return *this;
> }
> 
>-bool JsepTransport::RemoveChannel(int component) {
>-  auto it = channels_.find(component);
>-  if (it == channels_.end()) {
>-    RTC_LOG(LS_ERROR) << "Trying to remove channel for component " << component
>-                      << ", which doesn't exist.";
>-    return false;
>+JsepTransport::JsepTransport(
>+    const std::string& mid,
>+    const rtc::scoped_refptr<rtc::RTCCertificate>& local_certificate,
>+    std::unique_ptr<webrtc::RtpTransport> unencrypted_rtp_transport,
>+    std::unique_ptr<webrtc::SrtpTransport> sdes_transport,
>+    std::unique_ptr<webrtc::DtlsSrtpTransport> dtls_srtp_transport,
>+    std::unique_ptr<DtlsTransportInternal> rtp_dtls_transport,
>+    std::unique_ptr<DtlsTransportInternal> rtcp_dtls_transport)
>+    : mid_(mid),
>+      local_certificate_(local_certificate),
>+      rtp_dtls_transport_(std::move(rtp_dtls_transport)),
>+      rtcp_dtls_transport_(std::move(rtcp_dtls_transport)) {
>+  RTC_DCHECK(rtp_dtls_transport_);
>+  if (unencrypted_rtp_transport) {
>+    RTC_DCHECK(!sdes_transport);
>+    RTC_DCHECK(!dtls_srtp_transport);
>+    unencrypted_rtp_transport_ = std::move(unencrypted_rtp_transport);
>+  } else if (sdes_transport) {
>+    RTC_DCHECK(!unencrypted_rtp_transport);
>+    RTC_DCHECK(!dtls_srtp_transport);
>+    sdes_transport_ = std::move(sdes_transport);
>+  } else {
>+    RTC_DCHECK(dtls_srtp_transport);
>+    RTC_DCHECK(!unencrypted_rtp_transport);
>+    RTC_DCHECK(!sdes_transport);
>+    dtls_srtp_transport_ = std::move(dtls_srtp_transport);
>   }
>-  channels_.erase(component);
>-  return true;
> }
> 
>-bool JsepTransport::HasChannels() const {
>-  return !channels_.empty();
>-}
>+JsepTransport::~JsepTransport() {}
> 
>-void JsepTransport::SetLocalCertificate(
>-    const rtc::scoped_refptr<rtc::RTCCertificate>& certificate) {
>-  certificate_ = certificate;
>-}
>+webrtc::RTCError JsepTransport::SetLocalJsepTransportDescription(
>+    const JsepTransportDescription& jsep_description,
>+    SdpType type) {
>+  webrtc::RTCError error;
> 
>-bool JsepTransport::GetLocalCertificate(
>-    rtc::scoped_refptr<rtc::RTCCertificate>* certificate) const {
>-  if (!certificate_) {
>-    return false;
>+  if (!VerifyIceParams(jsep_description)) {
>+    return webrtc::RTCError(webrtc::RTCErrorType::INVALID_PARAMETER,
>+                            "Invalid ice-ufrag or ice-pwd length.");
>   }
> 
>-  *certificate = certificate_;
>-  return true;
>-}
>-
>-bool JsepTransport::SetLocalTransportDescription(
>-    const TransportDescription& description,
>-    SdpType type,
>-    std::string* error_desc) {
>-  bool ret = true;
>+  if (!SetRtcpMux(jsep_description.rtcp_mux_enabled, type,
>+                  ContentSource::CS_LOCAL)) {
>+    return webrtc::RTCError(webrtc::RTCErrorType::INVALID_PARAMETER,
>+                            "Failed to setup RTCP mux.");
>+  }
> 
>-  if (!VerifyIceParams(description)) {
>-    return BadTransportDescription("Invalid ice-ufrag or ice-pwd length",
>-                                   error_desc);
>+  // If doing SDES, setup the SDES crypto parameters.
>+  if (sdes_transport_) {
>+    RTC_DCHECK(!unencrypted_rtp_transport_);
>+    RTC_DCHECK(!dtls_srtp_transport_);
>+    if (!SetSdes(jsep_description.cryptos,
>+                 jsep_description.encrypted_header_extension_ids, type,
>+                 ContentSource::CS_LOCAL)) {
>+      return webrtc::RTCError(webrtc::RTCErrorType::INVALID_PARAMETER,
>+                              "Failed to setup SDES crypto parameters.");
>+    }
>+  } else if (dtls_srtp_transport_) {
>+    RTC_DCHECK(!unencrypted_rtp_transport_);
>+    RTC_DCHECK(!sdes_transport_);
>+    dtls_srtp_transport_->UpdateRecvEncryptedHeaderExtensionIds(
>+        jsep_description.encrypted_header_extension_ids);
>   }
> 
>   bool ice_restarting =
>-      local_description_set_ &&
>-      IceCredentialsChanged(local_description_->ice_ufrag,
>-                            local_description_->ice_pwd, description.ice_ufrag,
>-                            description.ice_pwd);
>-  local_description_.reset(new TransportDescription(description));
>+      local_description_ != nullptr &&
>+      IceCredentialsChanged(local_description_->transport_desc.ice_ufrag,
>+                            local_description_->transport_desc.ice_pwd,
>+                            jsep_description.transport_desc.ice_ufrag,
>+                            jsep_description.transport_desc.ice_pwd);
>+  local_description_.reset(new JsepTransportDescription(jsep_description));
> 
>   rtc::SSLFingerprint* local_fp =
>-      local_description_->identity_fingerprint.get();
>+      local_description_->transport_desc.identity_fingerprint.get();
> 
>   if (!local_fp) {
>-    certificate_ = nullptr;
>-  } else if (!VerifyCertificateFingerprint(certificate_.get(), local_fp,
>-                                           error_desc)) {
>-    return false;
>+    local_certificate_ = nullptr;
>+  } else {
>+    error = VerifyCertificateFingerprint(local_certificate_.get(), local_fp);
>+    if (!error.ok()) {
>+      local_description_.reset();
>+      return error;
>+    }
>   }
> 
>-  for (const auto& kv : channels_) {
>-    ret &= ApplyLocalTransportDescription(kv.second, error_desc);
>-  }
>-  if (!ret) {
>-    return false;
>+  SetLocalIceParameters(rtp_dtls_transport_->ice_transport());
>+
>+  if (rtcp_dtls_transport_) {
>+    SetLocalIceParameters(rtcp_dtls_transport_->ice_transport());
>   }
> 
>   // If PRANSWER/ANSWER is set, we should decide transport protocol type.
>   if (type == SdpType::kPrAnswer || type == SdpType::kAnswer) {
>-    ret &= NegotiateTransportDescription(type, error_desc);
>+    error = NegotiateAndSetDtlsParameters(type);
>   }
>-  if (!ret) {
>-    return false;
>+  if (!error.ok()) {
>+    local_description_.reset();
>+    return error;
>   }
> 
>   if (needs_ice_restart_ && ice_restarting) {
>@@ -177,35 +193,88 @@ bool JsepTransport::SetLocalTransportDescription(
>                         << mid();
>   }
> 
>-  local_description_set_ = true;
>-  return true;
>+  return webrtc::RTCError::OK();
> }
> 
>-bool JsepTransport::SetRemoteTransportDescription(
>-    const TransportDescription& description,
>-    SdpType type,
>-    std::string* error_desc) {
>-  bool ret = true;
>+webrtc::RTCError JsepTransport::SetRemoteJsepTransportDescription(
>+    const JsepTransportDescription& jsep_description,
>+    webrtc::SdpType type) {
>+  webrtc::RTCError error;
> 
>-  if (!VerifyIceParams(description)) {
>-    return BadTransportDescription("Invalid ice-ufrag or ice-pwd length",
>-                                   error_desc);
>+  if (!VerifyIceParams(jsep_description)) {
>+    remote_description_.reset();
>+    return webrtc::RTCError(webrtc::RTCErrorType::INVALID_PARAMETER,
>+                            "Invalid ice-ufrag or ice-pwd length.");
>   }
> 
>-  remote_description_.reset(new TransportDescription(description));
>-  for (const auto& kv : channels_) {
>-    ret &= ApplyRemoteTransportDescription(kv.second, error_desc);
>+  if (!SetRtcpMux(jsep_description.rtcp_mux_enabled, type,
>+                  ContentSource::CS_REMOTE)) {
>+    return webrtc::RTCError(webrtc::RTCErrorType::INVALID_PARAMETER,
>+                            "Failed to setup RTCP mux.");
>+  }
>+
>+  // If doing SDES, setup the SDES crypto parameters.
>+  if (sdes_transport_) {
>+    RTC_DCHECK(!unencrypted_rtp_transport_);
>+    RTC_DCHECK(!dtls_srtp_transport_);
>+    if (!SetSdes(jsep_description.cryptos,
>+                 jsep_description.encrypted_header_extension_ids, type,
>+                 ContentSource::CS_REMOTE)) {
>+      return webrtc::RTCError(webrtc::RTCErrorType::INVALID_PARAMETER,
>+                              "Failed to setup SDES crypto parameters.");
>+    }
>+    sdes_transport_->CacheRtpAbsSendTimeHeaderExtension(
>+        jsep_description.rtp_abs_sendtime_extn_id);
>+  } else if (dtls_srtp_transport_) {
>+    RTC_DCHECK(!unencrypted_rtp_transport_);
>+    RTC_DCHECK(!sdes_transport_);
>+    dtls_srtp_transport_->UpdateSendEncryptedHeaderExtensionIds(
>+        jsep_description.encrypted_header_extension_ids);
>+    dtls_srtp_transport_->CacheRtpAbsSendTimeHeaderExtension(
>+        jsep_description.rtp_abs_sendtime_extn_id);
>+  }
>+
>+  remote_description_.reset(new JsepTransportDescription(jsep_description));
>+  SetRemoteIceParameters(rtp_dtls_transport_->ice_transport());
>+
>+  if (rtcp_dtls_transport_) {
>+    SetRemoteIceParameters(rtcp_dtls_transport_->ice_transport());
>   }
> 
>   // If PRANSWER/ANSWER is set, we should decide transport protocol type.
>   if (type == SdpType::kPrAnswer || type == SdpType::kAnswer) {
>-    ret = NegotiateTransportDescription(SdpType::kOffer, error_desc);
>+    error = NegotiateAndSetDtlsParameters(SdpType::kOffer);
>   }
>-  if (ret) {
>-    remote_description_set_ = true;
>+  if (!error.ok()) {
>+    remote_description_.reset();
>+    return error;
>   }
>+  return webrtc::RTCError::OK();
>+}
> 
>-  return ret;
>+webrtc::RTCError JsepTransport::AddRemoteCandidates(
>+    const Candidates& candidates) {
>+  if (!local_description_ || !remote_description_) {
>+    return webrtc::RTCError(webrtc::RTCErrorType::INVALID_STATE,
>+                            mid() +
>+                                " is not ready to use the remote candidate "
>+                                "because the local or remote description is "
>+                                "not set.");
>+  }
>+
>+  for (const cricket::Candidate& candidate : candidates) {
>+    auto transport =
>+        candidate.component() == cricket::ICE_CANDIDATE_COMPONENT_RTP
>+            ? rtp_dtls_transport_.get()
>+            : rtcp_dtls_transport_.get();
>+    if (!transport) {
>+      return webrtc::RTCError(webrtc::RTCErrorType::INVALID_PARAMETER,
>+                              "Candidate has an unknown component: " +
>+                                  candidate.ToString() + " for mid " + mid());
>+    }
>+    transport->ice_transport()->AddRemoteCandidate(candidate);
>+  }
>+  return webrtc::RTCError::OK();
> }
> 
> void JsepTransport::SetNeedsIceRestartFlag() {
>@@ -215,143 +284,254 @@ void JsepTransport::SetNeedsIceRestartFlag() {
>   }
> }
> 
>-bool JsepTransport::NeedsIceRestart() const {
>-  return needs_ice_restart_;
>-}
>+absl::optional<rtc::SSLRole> JsepTransport::GetDtlsRole() const {
>+  RTC_DCHECK(rtp_dtls_transport_);
>+  rtc::SSLRole dtls_role;
>+  if (!rtp_dtls_transport_->GetDtlsRole(&dtls_role)) {
>+    return absl::optional<rtc::SSLRole>();
>+  }
> 
>-rtc::Optional<rtc::SSLRole> JsepTransport::GetSslRole() const {
>-  return ssl_role_;
>+  return absl::optional<rtc::SSLRole>(dtls_role);
> }
> 
> bool JsepTransport::GetStats(TransportStats* stats) {
>   stats->transport_name = mid();
>   stats->channel_stats.clear();
>-  for (auto& kv : channels_) {
>-    DtlsTransportInternal* dtls_transport = kv.second;
>-    TransportChannelStats substats;
>-    substats.component = kv.first;
>-    dtls_transport->GetSrtpCryptoSuite(&substats.srtp_crypto_suite);
>-    dtls_transport->GetSslCipherSuite(&substats.ssl_cipher_suite);
>-    substats.dtls_state = dtls_transport->dtls_state();
>-    if (!dtls_transport->ice_transport()->GetStats(
>-            &substats.connection_infos)) {
>-      return false;
>-    }
>-    stats->channel_stats.push_back(substats);
>+  bool ret = GetTransportStats(rtp_dtls_transport_.get(), stats);
>+  if (rtcp_dtls_transport_) {
>+    ret &= GetTransportStats(rtcp_dtls_transport_.get(), stats);
>   }
>-  return true;
>+  return ret;
> }
> 
>-bool JsepTransport::VerifyCertificateFingerprint(
>+webrtc::RTCError JsepTransport::VerifyCertificateFingerprint(
>     const rtc::RTCCertificate* certificate,
>-    const rtc::SSLFingerprint* fingerprint,
>-    std::string* error_desc) const {
>+    const rtc::SSLFingerprint* fingerprint) const {
>   if (!fingerprint) {
>-    return BadTransportDescription("No fingerprint.", error_desc);
>+    return webrtc::RTCError(webrtc::RTCErrorType::INVALID_PARAMETER,
>+                            "No fingerprint");
>   }
>   if (!certificate) {
>-    return BadTransportDescription(
>-        "Fingerprint provided but no identity available.", error_desc);
>+    return webrtc::RTCError(webrtc::RTCErrorType::INVALID_PARAMETER,
>+                            "Fingerprint provided but no identity available.");
>   }
>   std::unique_ptr<rtc::SSLFingerprint> fp_tmp(rtc::SSLFingerprint::Create(
>       fingerprint->algorithm, certificate->identity()));
>   RTC_DCHECK(fp_tmp.get() != NULL);
>   if (*fp_tmp == *fingerprint) {
>-    return true;
>+    return webrtc::RTCError::OK();
>   }
>-  std::ostringstream desc;
>+  char ss_buf[1024];
>+  rtc::SimpleStringBuilder desc(ss_buf);
>   desc << "Local fingerprint does not match identity. Expected: ";
>   desc << fp_tmp->ToString();
>   desc << " Got: " << fingerprint->ToString();
>-  return BadTransportDescription(desc.str(), error_desc);
>+  return webrtc::RTCError(webrtc::RTCErrorType::INVALID_PARAMETER,
>+                          std::string(desc.str()));
> }
> 
>-bool JsepTransport::ApplyLocalTransportDescription(
>-    DtlsTransportInternal* dtls_transport,
>-    std::string* error_desc) {
>-  dtls_transport->ice_transport()->SetIceParameters(
>-      local_description_->GetIceParameters());
>-  return true;
>+void JsepTransport::SetActiveResetSrtpParams(bool active_reset_srtp_params) {
>+  if (dtls_srtp_transport_) {
>+    RTC_LOG(INFO)
>+        << "Setting active_reset_srtp_params of DtlsSrtpTransport to: "
>+        << active_reset_srtp_params;
>+    dtls_srtp_transport_->SetActiveResetSrtpParams(active_reset_srtp_params);
>+  }
> }
> 
>-bool JsepTransport::ApplyRemoteTransportDescription(
>-    DtlsTransportInternal* dtls_transport,
>-    std::string* error_desc) {
>-  dtls_transport->ice_transport()->SetRemoteIceParameters(
>-      remote_description_->GetIceParameters());
>-  dtls_transport->ice_transport()->SetRemoteIceMode(
>-      remote_description_->ice_mode);
>-  return true;
>+void JsepTransport::SetLocalIceParameters(IceTransportInternal* ice_transport) {
>+  RTC_DCHECK(ice_transport);
>+  RTC_DCHECK(local_description_);
>+  ice_transport->SetIceParameters(
>+      local_description_->transport_desc.GetIceParameters());
> }
> 
>-bool JsepTransport::ApplyNegotiatedTransportDescription(
>+void JsepTransport::SetRemoteIceParameters(
>+    IceTransportInternal* ice_transport) {
>+  RTC_DCHECK(ice_transport);
>+  RTC_DCHECK(remote_description_);
>+  ice_transport->SetRemoteIceParameters(
>+      remote_description_->transport_desc.GetIceParameters());
>+  ice_transport->SetRemoteIceMode(remote_description_->transport_desc.ice_mode);
>+}
>+
>+webrtc::RTCError JsepTransport::SetNegotiatedDtlsParameters(
>     DtlsTransportInternal* dtls_transport,
>-    std::string* error_desc) {
>+    absl::optional<rtc::SSLRole> dtls_role,
>+    rtc::SSLFingerprint* remote_fingerprint) {
>+  RTC_DCHECK(dtls_transport);
>   // Set SSL role. Role must be set before fingerprint is applied, which
>   // initiates DTLS setup.
>-  if (ssl_role_ && !dtls_transport->SetSslRole(*ssl_role_)) {
>-    return BadTransportDescription("Failed to set SSL role for the channel.",
>-                                   error_desc);
>+  if (dtls_role && !dtls_transport->SetDtlsRole(*dtls_role)) {
>+    return webrtc::RTCError(webrtc::RTCErrorType::INVALID_PARAMETER,
>+                            "Failed to set SSL role for the transport.");
>   }
>   // Apply remote fingerprint.
>-  if (!dtls_transport->SetRemoteFingerprint(
>-          remote_fingerprint_->algorithm,
>-          reinterpret_cast<const uint8_t*>(remote_fingerprint_->digest.data()),
>-          remote_fingerprint_->digest.size())) {
>-    return BadTransportDescription("Failed to apply remote fingerprint.",
>-                                   error_desc);
>+  if (!remote_fingerprint ||
>+      !dtls_transport->SetRemoteFingerprint(
>+          remote_fingerprint->algorithm,
>+          reinterpret_cast<const uint8_t*>(remote_fingerprint->digest.data()),
>+          remote_fingerprint->digest.size())) {
>+    return webrtc::RTCError(webrtc::RTCErrorType::INVALID_PARAMETER,
>+                            "Failed to apply remote fingerprint.");
>+  }
>+  return webrtc::RTCError::OK();
>+}
>+
>+bool JsepTransport::SetRtcpMux(bool enable,
>+                               webrtc::SdpType type,
>+                               ContentSource source) {
>+  bool ret = false;
>+  switch (type) {
>+    case SdpType::kOffer:
>+      ret = rtcp_mux_negotiator_.SetOffer(enable, source);
>+      break;
>+    case SdpType::kPrAnswer:
>+      // This may activate RTCP muxing, but we don't yet destroy the transport
>+      // because the final answer may deactivate it.
>+      ret = rtcp_mux_negotiator_.SetProvisionalAnswer(enable, source);
>+      break;
>+    case SdpType::kAnswer:
>+      ret = rtcp_mux_negotiator_.SetAnswer(enable, source);
>+      if (ret && rtcp_mux_negotiator_.IsActive()) {
>+        ActivateRtcpMux();
>+      }
>+      break;
>+    default:
>+      RTC_NOTREACHED();
>   }
>-  return true;
>+
>+  if (!ret) {
>+    return false;
>+  }
>+
>+  auto transport = rtp_transport();
>+  transport->SetRtcpMuxEnabled(rtcp_mux_negotiator_.IsActive());
>+  return ret;
> }
> 
>-bool JsepTransport::NegotiateTransportDescription(
>-    SdpType local_description_type,
>-    std::string* error_desc) {
>+void JsepTransport::ActivateRtcpMux() {
>+  if (unencrypted_rtp_transport_) {
>+    RTC_DCHECK(!sdes_transport_);
>+    RTC_DCHECK(!dtls_srtp_transport_);
>+    unencrypted_rtp_transport_->SetRtcpPacketTransport(nullptr);
>+  } else if (sdes_transport_) {
>+    RTC_DCHECK(!unencrypted_rtp_transport_);
>+    RTC_DCHECK(!dtls_srtp_transport_);
>+    sdes_transport_->SetRtcpPacketTransport(nullptr);
>+  } else {
>+    RTC_DCHECK(dtls_srtp_transport_);
>+    RTC_DCHECK(!unencrypted_rtp_transport_);
>+    RTC_DCHECK(!sdes_transport_);
>+    dtls_srtp_transport_->SetDtlsTransports(rtp_dtls_transport(),
>+                                            /*rtcp_dtls_transport=*/nullptr);
>+  }
>+  rtcp_dtls_transport_.reset();
>+  // Notify the JsepTransportController to update the aggregate states.
>+  SignalRtcpMuxActive();
>+}
>+
>+bool JsepTransport::SetSdes(const std::vector<CryptoParams>& cryptos,
>+                            const std::vector<int>& encrypted_extension_ids,
>+                            webrtc::SdpType type,
>+                            ContentSource source) {
>+  bool ret = false;
>+  ret = sdes_negotiator_.Process(cryptos, type, source);
>+  if (!ret) {
>+    return ret;
>+  }
>+
>+  if (source == ContentSource::CS_LOCAL) {
>+    recv_extension_ids_ = std::move(encrypted_extension_ids);
>+  } else {
>+    send_extension_ids_ = std::move(encrypted_extension_ids);
>+  }
>+
>+  // If setting an SDES answer succeeded, apply the negotiated parameters
>+  // to the SRTP transport.
>+  if ((type == SdpType::kPrAnswer || type == SdpType::kAnswer) && ret) {
>+    if (sdes_negotiator_.send_cipher_suite() &&
>+        sdes_negotiator_.recv_cipher_suite()) {
>+      RTC_DCHECK(send_extension_ids_);
>+      RTC_DCHECK(recv_extension_ids_);
>+      ret = sdes_transport_->SetRtpParams(
>+          *(sdes_negotiator_.send_cipher_suite()),
>+          sdes_negotiator_.send_key().data(),
>+          static_cast<int>(sdes_negotiator_.send_key().size()),
>+          *(send_extension_ids_), *(sdes_negotiator_.recv_cipher_suite()),
>+          sdes_negotiator_.recv_key().data(),
>+          static_cast<int>(sdes_negotiator_.recv_key().size()),
>+          *(recv_extension_ids_));
>+    } else {
>+      RTC_LOG(LS_INFO) << "No crypto keys are provided for SDES.";
>+      if (type == SdpType::kAnswer) {
>+        // Explicitly reset the |sdes_transport_| if no crypto param is
>+        // provided in the answer. No need to call |ResetParams()| for
>+        // |sdes_negotiator_| because it resets the params inside |SetAnswer|.
>+        sdes_transport_->ResetParams();
>+      }
>+    }
>+  }
>+  return ret;
>+}
>+
>+webrtc::RTCError JsepTransport::NegotiateAndSetDtlsParameters(
>+    SdpType local_description_type) {
>   if (!local_description_ || !remote_description_) {
>-    const std::string msg =
>-        "Applying an answer transport description "
>-        "without applying any offer.";
>-    return BadTransportDescription(msg, error_desc);
>+    return webrtc::RTCError(webrtc::RTCErrorType::INVALID_STATE,
>+                            "Applying an answer transport description "
>+                            "without applying any offer.");
>   }
>+  std::unique_ptr<rtc::SSLFingerprint> remote_fingerprint;
>+  absl::optional<rtc::SSLRole> negotiated_dtls_role;
>+
>   rtc::SSLFingerprint* local_fp =
>-      local_description_->identity_fingerprint.get();
>+      local_description_->transport_desc.identity_fingerprint.get();
>   rtc::SSLFingerprint* remote_fp =
>-      remote_description_->identity_fingerprint.get();
>+      remote_description_->transport_desc.identity_fingerprint.get();
>   if (remote_fp && local_fp) {
>-    remote_fingerprint_.reset(new rtc::SSLFingerprint(*remote_fp));
>-    if (!NegotiateRole(local_description_type, error_desc)) {
>-      return false;
>+    remote_fingerprint = absl::make_unique<rtc::SSLFingerprint>(*remote_fp);
>+    webrtc::RTCError error =
>+        NegotiateDtlsRole(local_description_type,
>+                          local_description_->transport_desc.connection_role,
>+                          remote_description_->transport_desc.connection_role,
>+                          &negotiated_dtls_role);
>+    if (!error.ok()) {
>+      return error;
>     }
>   } else if (local_fp && (local_description_type == SdpType::kAnswer)) {
>-    return BadTransportDescription(
>-        "Local fingerprint supplied when caller didn't offer DTLS.",
>-        error_desc);
>+    return webrtc::RTCError(
>+        webrtc::RTCErrorType::INVALID_PARAMETER,
>+        "Local fingerprint supplied when caller didn't offer DTLS.");
>   } else {
>     // We are not doing DTLS
>-    remote_fingerprint_.reset(new rtc::SSLFingerprint("", nullptr, 0));
>+    remote_fingerprint = absl::make_unique<rtc::SSLFingerprint>("", nullptr, 0);
>   }
>   // Now that we have negotiated everything, push it downward.
>   // Note that we cache the result so that if we have race conditions
>-  // between future SetRemote/SetLocal invocations and new channel
>+  // between future SetRemote/SetLocal invocations and new transport
>   // creation, we have the negotiation state saved until a new
>   // negotiation happens.
>-  for (const auto& kv : channels_) {
>-    if (!ApplyNegotiatedTransportDescription(kv.second, error_desc)) {
>-      return false;
>-    }
>+  webrtc::RTCError error = SetNegotiatedDtlsParameters(
>+      rtp_dtls_transport_.get(), negotiated_dtls_role,
>+      remote_fingerprint.get());
>+  if (!error.ok()) {
>+    return error;
>   }
>-  return true;
>-}
> 
>-bool JsepTransport::NegotiateRole(SdpType local_description_type,
>-                                  std::string* error_desc) {
>-  if (!local_description_ || !remote_description_) {
>-    const std::string msg =
>-        "Local and Remote description must be set before "
>-        "transport descriptions are negotiated";
>-    return BadTransportDescription(msg, error_desc);
>+  if (rtcp_dtls_transport_) {
>+    error = SetNegotiatedDtlsParameters(rtcp_dtls_transport_.get(),
>+                                        negotiated_dtls_role,
>+                                        remote_fingerprint.get());
>   }
>+  return error;
>+}
> 
>+webrtc::RTCError JsepTransport::NegotiateDtlsRole(
>+    SdpType local_description_type,
>+    ConnectionRole local_connection_role,
>+    ConnectionRole remote_connection_role,
>+    absl::optional<rtc::SSLRole>* negotiated_dtls_role) {
>   // From RFC 4145, section-4.1, The following are the values that the
>   // 'setup' attribute can take in an offer/answer exchange:
>   //       Offer      Answer
>@@ -375,14 +555,12 @@ bool JsepTransport::NegotiateRole(SdpType local_description_type,
>   // ClientHello over each flow (host/port quartet).
>   // IOW - actpass and passive modes should be treated as server and
>   // active as client.
>-  ConnectionRole local_connection_role = local_description_->connection_role;
>-  ConnectionRole remote_connection_role = remote_description_->connection_role;
>-
>   bool is_remote_server = false;
>   if (local_description_type == SdpType::kOffer) {
>     if (local_connection_role != CONNECTIONROLE_ACTPASS) {
>-      return BadTransportDescription(
>-          "Offerer must use actpass value for setup attribute.", error_desc);
>+      return webrtc::RTCError(
>+          webrtc::RTCErrorType::INVALID_PARAMETER,
>+          "Offerer must use actpass value for setup attribute.");
>     }
> 
>     if (remote_connection_role == CONNECTIONROLE_ACTIVE ||
>@@ -390,10 +568,10 @@ bool JsepTransport::NegotiateRole(SdpType local_description_type,
>         remote_connection_role == CONNECTIONROLE_NONE) {
>       is_remote_server = (remote_connection_role == CONNECTIONROLE_PASSIVE);
>     } else {
>-      const std::string msg =
>+      return webrtc::RTCError(
>+          webrtc::RTCErrorType::INVALID_PARAMETER,
>           "Answerer must use either active or passive value "
>-          "for setup attribute.";
>-      return BadTransportDescription(msg, error_desc);
>+          "for setup attribute.");
>     }
>     // If remote is NONE or ACTIVE it will act as client.
>   } else {
>@@ -406,15 +584,16 @@ bool JsepTransport::NegotiateRole(SdpType local_description_type,
>       //
>       // See https://datatracker.ietf.org/doc/html/draft-ietf-mmusic-dtls-sdp,
>       // section 5.5.
>-      if (!ssl_role_ ||
>-          (*ssl_role_ == rtc::SSL_CLIENT &&
>+      auto current_dtls_role = GetDtlsRole();
>+      if (!current_dtls_role ||
>+          (*current_dtls_role == rtc::SSL_CLIENT &&
>            remote_connection_role == CONNECTIONROLE_ACTIVE) ||
>-          (*ssl_role_ == rtc::SSL_SERVER &&
>+          (*current_dtls_role == rtc::SSL_SERVER &&
>            remote_connection_role == CONNECTIONROLE_PASSIVE)) {
>-        return BadTransportDescription(
>+        return webrtc::RTCError(
>+            webrtc::RTCErrorType::INVALID_PARAMETER,
>             "Offerer must use actpass value or current negotiated role for "
>-            "setup attribute.",
>-            error_desc);
>+            "setup attribute.");
>       }
>     }
> 
>@@ -422,16 +601,35 @@ bool JsepTransport::NegotiateRole(SdpType local_description_type,
>         local_connection_role == CONNECTIONROLE_PASSIVE) {
>       is_remote_server = (local_connection_role == CONNECTIONROLE_ACTIVE);
>     } else {
>-      const std::string msg =
>+      return webrtc::RTCError(
>+          webrtc::RTCErrorType::INVALID_PARAMETER,
>           "Answerer must use either active or passive value "
>-          "for setup attribute.";
>-      return BadTransportDescription(msg, error_desc);
>+          "for setup attribute.");
>     }
> 
>     // If local is passive, local will act as server.
>   }
> 
>-  ssl_role_.emplace(is_remote_server ? rtc::SSL_CLIENT : rtc::SSL_SERVER);
>+  *negotiated_dtls_role = (is_remote_server ? std::move(rtc::SSL_CLIENT)
>+                                            : std::move(rtc::SSL_SERVER));
>+  return webrtc::RTCError::OK();
>+}
>+
>+bool JsepTransport::GetTransportStats(DtlsTransportInternal* dtls_transport,
>+                                      TransportStats* stats) {
>+  RTC_DCHECK(dtls_transport);
>+  TransportChannelStats substats;
>+  substats.component = dtls_transport == rtcp_dtls_transport_.get()
>+                           ? ICE_CANDIDATE_COMPONENT_RTCP
>+                           : ICE_CANDIDATE_COMPONENT_RTP;
>+  dtls_transport->GetSrtpCryptoSuite(&substats.srtp_crypto_suite);
>+  dtls_transport->GetSslCipherSuite(&substats.ssl_cipher_suite);
>+  substats.dtls_state = dtls_transport->dtls_state();
>+  if (!dtls_transport->ice_transport()->GetStats(
>+          &substats.connection_infos, &substats.candidate_stats_list)) {
>+    return false;
>+  }
>+  stats->channel_stats.push_back(substats);
>   return true;
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jseptransport.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jseptransport.h
>index 972f91bf5be..8e898533ef9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jseptransport.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jseptransport.h
>@@ -1,5 +1,5 @@
> /*
>- *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>  *
>  *  Use of this source code is governed by a BSD-style license
>  *  that can be found in the LICENSE file in the root of the source
>@@ -16,101 +16,102 @@
> #include <string>
> #include <vector>
> 
>+#include "absl/types/optional.h"
> #include "api/candidate.h"
> #include "api/jsep.h"
>-#include "api/optional.h"
> #include "p2p/base/dtlstransport.h"
> #include "p2p/base/p2pconstants.h"
> #include "p2p/base/transportinfo.h"
>+#include "pc/dtlssrtptransport.h"
>+#include "pc/rtcpmuxfilter.h"
>+#include "pc/rtptransport.h"
> #include "pc/sessiondescription.h"
>+#include "pc/srtpfilter.h"
>+#include "pc/srtptransport.h"
>+#include "pc/transportstats.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/messagequeue.h"
> #include "rtc_base/rtccertificate.h"
>-#include "rtc_base/sigslot.h"
> #include "rtc_base/sslstreamadapter.h"
>+#include "rtc_base/third_party/sigslot/sigslot.h"
> 
> namespace cricket {
> 
> class DtlsTransportInternal;
> 
>-struct TransportChannelStats {
>-  TransportChannelStats();
>-  TransportChannelStats(const TransportChannelStats&);
>-  ~TransportChannelStats();
>-
>-  int component = 0;
>-  ConnectionInfos connection_infos;
>-  int srtp_crypto_suite = rtc::SRTP_INVALID_CRYPTO_SUITE;
>-  int ssl_cipher_suite = rtc::TLS_NULL_WITH_NULL_NULL;
>-  DtlsTransportState dtls_state = DTLS_TRANSPORT_NEW;
>-};
>-
>-// Information about all the channels of a transport.
>-// TODO(hta): Consider if a simple vector is as good as a map.
>-typedef std::vector<TransportChannelStats> TransportChannelStatsList;
>-
>-// Information about the stats of a transport.
>-struct TransportStats {
>-  TransportStats();
>-  ~TransportStats();
>-
>-  std::string transport_name;
>-  TransportChannelStatsList channel_stats;
>+struct JsepTransportDescription {
>+ public:
>+  JsepTransportDescription();
>+  JsepTransportDescription(
>+      bool rtcp_mux_enabled,
>+      const std::vector<CryptoParams>& cryptos,
>+      const std::vector<int>& encrypted_header_extension_ids,
>+      int rtp_abs_sendtime_extn_id,
>+      const TransportDescription& transport_description);
>+  JsepTransportDescription(const JsepTransportDescription& from);
>+  ~JsepTransportDescription();
>+
>+  JsepTransportDescription& operator=(const JsepTransportDescription& from);
>+
>+  bool rtcp_mux_enabled = true;
>+  std::vector<CryptoParams> cryptos;
>+  std::vector<int> encrypted_header_extension_ids;
>+  int rtp_abs_sendtime_extn_id = -1;
>+  // TODO(zhihuang): Add the ICE and DTLS related variables and methods from
>+  // TransportDescription and remove this extra layer of abstraction.
>+  TransportDescription transport_desc;
> };
> 
>-bool BadTransportDescription(const std::string& desc, std::string* err_desc);
>-
>-// Helper class used by TransportController that processes
>+// Helper class used by JsepTransportController that processes
> // TransportDescriptions. A TransportDescription represents the
> // transport-specific properties of an SDP m= section, processed according to
> // JSEP. Each transport consists of DTLS and ICE transport channels for RTP
> // (and possibly RTCP, if rtcp-mux isn't used).
> //
>-// On Threading:  Transport performs work solely on the network thread, and so
>-// its methods should only be called on the network thread.
>+// On Threading: JsepTransport performs work solely on the network thread, and
>+// so its methods should only be called on the network thread.
> class JsepTransport : public sigslot::has_slots<> {
>  public:
>   // |mid| is just used for log statements in order to identify the Transport.
>-  // Note that |certificate| is allowed to be null since a remote description
>-  // may be set before a local certificate is generated.
>-  JsepTransport(const std::string& mid,
>-                const rtc::scoped_refptr<rtc::RTCCertificate>& certificate);
>+  // Note that |local_certificate| is allowed to be null since a remote
>+  // description may be set before a local certificate is generated.
>+  JsepTransport(
>+      const std::string& mid,
>+      const rtc::scoped_refptr<rtc::RTCCertificate>& local_certificate,
>+      std::unique_ptr<webrtc::RtpTransport> unencrypted_rtp_transport,
>+      std::unique_ptr<webrtc::SrtpTransport> sdes_transport,
>+      std::unique_ptr<webrtc::DtlsSrtpTransport> dtls_srtp_transport,
>+      std::unique_ptr<DtlsTransportInternal> rtp_dtls_transport,
>+      std::unique_ptr<DtlsTransportInternal> rtcp_dtls_transport);
>+
>   ~JsepTransport() override;
> 
>-  // Returns the MID of this transport.
>+  // Returns the MID of this transport. This is only used for logging.
>   const std::string& mid() const { return mid_; }
> 
>-  // Add or remove channel that is affected when a local/remote transport
>-  // description is set on this transport. Need to add all channels before
>-  // setting a transport description.
>-  bool AddChannel(DtlsTransportInternal* dtls, int component);
>-  bool RemoveChannel(int component);
>-  bool HasChannels() const;
>-
>-  bool ready_for_remote_candidates() const {
>-    return local_description_set_ && remote_description_set_;
>-  }
>-
>   // Must be called before applying local session description.
>   // Needed in order to verify the local fingerprint.
>   void SetLocalCertificate(
>-      const rtc::scoped_refptr<rtc::RTCCertificate>& certificate);
>+      const rtc::scoped_refptr<rtc::RTCCertificate>& local_certificate) {
>+    local_certificate_ = local_certificate;
>+  }
> 
>-  // Get a copy of the local certificate provided by SetLocalCertificate.
>-  bool GetLocalCertificate(
>-      rtc::scoped_refptr<rtc::RTCCertificate>* certificate) const;
>+  // Return the local certificate provided by SetLocalCertificate.
>+  rtc::scoped_refptr<rtc::RTCCertificate> GetLocalCertificate() const {
>+    return local_certificate_;
>+  }
> 
>-  // Set the local TransportDescription to be used by DTLS and ICE channels
>-  // that are part of this Transport.
>-  bool SetLocalTransportDescription(const TransportDescription& description,
>-                                    webrtc::SdpType type,
>-                                    std::string* error_desc);
>+  webrtc::RTCError SetLocalJsepTransportDescription(
>+      const JsepTransportDescription& jsep_description,
>+      webrtc::SdpType type);
> 
>   // Set the remote TransportDescription to be used by DTLS and ICE channels
>   // that are part of this Transport.
>-  bool SetRemoteTransportDescription(const TransportDescription& description,
>-                                     webrtc::SdpType type,
>-                                     std::string* error_desc);
>+  webrtc::RTCError SetRemoteJsepTransportDescription(
>+      const JsepTransportDescription& jsep_description,
>+      webrtc::SdpType type);
>+
>+  webrtc::RTCError AddRemoteCandidates(const Candidates& candidates);
> 
>   // Set the "needs-ice-restart" flag as described in JSEP. After the flag is
>   // set, offers should generate new ufrags/passwords until an ICE restart
>@@ -122,80 +123,123 @@ class JsepTransport : public sigslot::has_slots<> {
>   // Returns true if the ICE restart flag above was set, and no ICE restart has
>   // occurred yet for this transport (by applying a local description with
>   // changed ufrag/password).
>-  bool NeedsIceRestart() const;
>+  bool needs_ice_restart() const { return needs_ice_restart_; }
> 
>-  // Returns role if negotiated, or empty Optional if it hasn't been negotiated
>-  // yet.
>-  rtc::Optional<rtc::SSLRole> GetSslRole() const;
>+  // Returns role if negotiated, or empty absl::optional if it hasn't been
>+  // negotiated yet.
>+  absl::optional<rtc::SSLRole> GetDtlsRole() const;
> 
>   // TODO(deadbeef): Make this const. See comment in transportcontroller.h.
>   bool GetStats(TransportStats* stats);
> 
>-  // The current local transport description, possibly used
>-  // by the transport controller.
>-  const TransportDescription* local_description() const {
>+  const JsepTransportDescription* local_description() const {
>     return local_description_.get();
>   }
> 
>-  // The current remote transport description, possibly used
>-  // by the transport controller.
>-  const TransportDescription* remote_description() const {
>+  const JsepTransportDescription* remote_description() const {
>     return remote_description_.get();
>   }
> 
>+  webrtc::RtpTransportInternal* rtp_transport() const {
>+    if (dtls_srtp_transport_) {
>+      return dtls_srtp_transport_.get();
>+    } else if (sdes_transport_) {
>+      return sdes_transport_.get();
>+    } else {
>+      return unencrypted_rtp_transport_.get();
>+    }
>+  }
>+
>+  DtlsTransportInternal* rtp_dtls_transport() const {
>+    return rtp_dtls_transport_.get();
>+  }
>+
>+  DtlsTransportInternal* rtcp_dtls_transport() const {
>+    return rtcp_dtls_transport_.get();
>+  }
>+
>+  // This is signaled when RTCP-mux becomes active and
>+  // |rtcp_dtls_transport_| is destroyed. The JsepTransportController will
>+  // handle the signal and update the aggregate transport states.
>+  sigslot::signal<> SignalRtcpMuxActive;
>+
>   // TODO(deadbeef): The methods below are only public for testing. Should make
>   // them utility functions or objects so they can be tested independently from
>   // this class.
> 
>-  // Returns false if the certificate's identity does not match the fingerprint,
>-  // or either is NULL.
>-  bool VerifyCertificateFingerprint(const rtc::RTCCertificate* certificate,
>-                                    const rtc::SSLFingerprint* fingerprint,
>-                                    std::string* error_desc) const;
>+  // Returns an error if the certificate's identity does not match the
>+  // fingerprint, or either is NULL.
>+  webrtc::RTCError VerifyCertificateFingerprint(
>+      const rtc::RTCCertificate* certificate,
>+      const rtc::SSLFingerprint* fingerprint) const;
>+
>+  void SetActiveResetSrtpParams(bool active_reset_srtp_params);
> 
>  private:
>-  // Negotiates the transport parameters based on the current local and remote
>-  // transport description, such as the ICE role to use, and whether DTLS
>-  // should be activated.
>+  bool SetRtcpMux(bool enable, webrtc::SdpType type, ContentSource source);
>+
>+  void ActivateRtcpMux();
>+
>+  bool SetSdes(const std::vector<CryptoParams>& cryptos,
>+               const std::vector<int>& encrypted_extension_ids,
>+               webrtc::SdpType type,
>+               ContentSource source);
>+
>+  // Negotiates and sets the DTLS parameters based on the current local and
>+  // remote transport description, such as the DTLS role to use, and whether
>+  // DTLS should be activated.
>   //
>   // Called when an answer TransportDescription is applied.
>-  bool NegotiateTransportDescription(webrtc::SdpType local_description_type,
>-                                     std::string* error_desc);
>+  webrtc::RTCError NegotiateAndSetDtlsParameters(
>+      webrtc::SdpType local_description_type);
> 
>-  // Negotiates the SSL role based off the offer and answer as specified by
>-  // RFC 4145, section-4.1. Returns false if the SSL role cannot be determined
>+  // Negotiates the DTLS role based off the offer and answer as specified by
>+  // RFC 4145, section-4.1. Returns an RTCError if role cannot be determined
>   // from the local description and remote description.
>-  bool NegotiateRole(webrtc::SdpType local_description_type,
>-                     std::string* error_desc);
>+  webrtc::RTCError NegotiateDtlsRole(
>+      webrtc::SdpType local_description_type,
>+      ConnectionRole local_connection_role,
>+      ConnectionRole remote_connection_role,
>+      absl::optional<rtc::SSLRole>* negotiated_dtls_role);
> 
>-  // Pushes down the transport parameters from the local description, such
>+  // Pushes down the ICE parameters from the local description, such
>   // as the ICE ufrag and pwd.
>-  bool ApplyLocalTransportDescription(DtlsTransportInternal* dtls_transport,
>-                                      std::string* error_desc);
>+  void SetLocalIceParameters(IceTransportInternal* ice);
> 
>-  // Pushes down the transport parameters from the remote description to the
>-  // transport channel.
>-  bool ApplyRemoteTransportDescription(DtlsTransportInternal* dtls_transport,
>-                                       std::string* error_desc);
>+  // Pushes down the ICE parameters from the remote description.
>+  void SetRemoteIceParameters(IceTransportInternal* ice);
> 
>-  // Pushes down the transport parameters obtained via negotiation.
>-  bool ApplyNegotiatedTransportDescription(
>+  // Pushes down the DTLS parameters obtained via negotiation.
>+  webrtc::RTCError SetNegotiatedDtlsParameters(
>       DtlsTransportInternal* dtls_transport,
>-      std::string* error_desc);
>+      absl::optional<rtc::SSLRole> dtls_role,
>+      rtc::SSLFingerprint* remote_fingerprint);
>+
>+  bool GetTransportStats(DtlsTransportInternal* dtls_transport,
>+                         TransportStats* stats);
> 
>   const std::string mid_;
>   // needs-ice-restart bit as described in JSEP.
>   bool needs_ice_restart_ = false;
>-  rtc::scoped_refptr<rtc::RTCCertificate> certificate_;
>-  rtc::Optional<rtc::SSLRole> ssl_role_;
>-  std::unique_ptr<rtc::SSLFingerprint> remote_fingerprint_;
>-  std::unique_ptr<TransportDescription> local_description_;
>-  std::unique_ptr<TransportDescription> remote_description_;
>-  bool local_description_set_ = false;
>-  bool remote_description_set_ = false;
>-
>-  // Candidate component => DTLS channel
>-  std::map<int, DtlsTransportInternal*> channels_;
>+  rtc::scoped_refptr<rtc::RTCCertificate> local_certificate_;
>+  std::unique_ptr<JsepTransportDescription> local_description_;
>+  std::unique_ptr<JsepTransportDescription> remote_description_;
>+
>+  // To avoid downcasting and make it type safe, keep three unique pointers for
>+  // different SRTP mode and only one of these is non-nullptr.
>+  std::unique_ptr<webrtc::RtpTransport> unencrypted_rtp_transport_;
>+  std::unique_ptr<webrtc::SrtpTransport> sdes_transport_;
>+  std::unique_ptr<webrtc::DtlsSrtpTransport> dtls_srtp_transport_;
>+
>+  std::unique_ptr<DtlsTransportInternal> rtp_dtls_transport_;
>+  std::unique_ptr<DtlsTransportInternal> rtcp_dtls_transport_;
>+
>+  SrtpFilter sdes_negotiator_;
>+  RtcpMuxFilter rtcp_mux_negotiator_;
>+
>+  // Cache the encrypted header extension IDs for SDES negoitation.
>+  absl::optional<std::vector<int>> send_extension_ids_;
>+  absl::optional<std::vector<int>> recv_extension_ids_;
> 
>   RTC_DISALLOW_COPY_AND_ASSIGN(JsepTransport);
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jseptransport_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jseptransport_unittest.cc
>index 1111f36b0b3..56a742c338b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jseptransport_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jseptransport_unittest.cc
>@@ -1,5 +1,5 @@
> /*
>- *  Copyright 2011 The WebRTC Project Authors. All rights reserved.
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>  *
>  *  Use of this source code is governed by a BSD-style license
>  *  that can be found in the LICENSE file in the root of the source
>@@ -7,95 +7,201 @@
>  *  in the file PATENTS.  All contributing project authors may
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
>-#include "pc/jseptransport.h"
> 
> #include <memory>
>+#include <tuple>
>+#include <utility>
> 
>+#include "media/base/fakertp.h"
> #include "p2p/base/fakedtlstransport.h"
> #include "p2p/base/fakeicetransport.h"
>-#include "rtc_base/fakesslidentity.h"
>+#include "pc/jseptransport.h"
> #include "rtc_base/gunit.h"
>-#include "rtc_base/network.h"
> 
> namespace cricket {
>-
>-using rtc::SocketAddress;
> using webrtc::SdpType;
> 
>-static const char kIceUfrag1[] = "TESTICEUFRAG0001";
>+static const char kIceUfrag1[] = "U001";
> static const char kIcePwd1[] = "TESTICEPWD00000000000001";
>+static const char kIceUfrag2[] = "U002";
>+static const char kIcePwd2[] = "TESTIEPWD00000000000002";
>+static const char kTransportName[] = "Test Transport";
>+
>+enum class SrtpMode {
>+  kSdes,
>+  kDtlsSrtp,
>+};
>+
>+struct NegotiateRoleParams {
>+  ConnectionRole local_role;
>+  ConnectionRole remote_role;
>+  SdpType local_type;
>+  SdpType remote_type;
>+};
> 
>-static const char kIceUfrag2[] = "TESTICEUFRAG0002";
>-static const char kIcePwd2[] = "TESTICEPWD00000000000002";
>-
>-TransportDescription MakeTransportDescription(
>-    const char* ufrag,
>-    const char* pwd,
>-    const rtc::scoped_refptr<rtc::RTCCertificate>& cert,
>-    ConnectionRole role = CONNECTIONROLE_NONE) {
>-  std::unique_ptr<rtc::SSLFingerprint> fingerprint;
>-  if (cert) {
>-    fingerprint.reset(rtc::SSLFingerprint::CreateFromCertificate(cert));
>+class JsepTransport2Test : public testing::Test, public sigslot::has_slots<> {
>+ protected:
>+  std::unique_ptr<webrtc::SrtpTransport> CreateSdesTransport(
>+      rtc::PacketTransportInternal* rtp_packet_transport,
>+      rtc::PacketTransportInternal* rtcp_packet_transport) {
>+    auto srtp_transport = absl::make_unique<webrtc::SrtpTransport>(
>+        rtcp_packet_transport == nullptr);
>+
>+    srtp_transport->SetRtpPacketTransport(rtp_packet_transport);
>+    if (rtcp_packet_transport) {
>+      srtp_transport->SetRtcpPacketTransport(rtp_packet_transport);
>+    }
>+    return srtp_transport;
>   }
>-  return TransportDescription(std::vector<std::string>(), ufrag, pwd,
>-                              ICEMODE_FULL, role, fingerprint.get());
>-}
> 
>-class JsepTransportTest : public testing::Test, public sigslot::has_slots<> {
>- public:
>-  JsepTransportTest() { RecreateTransport(); }
>-
>-  bool SetupFakeTransports(int component) {
>-    fake_ice_transports_.emplace_back(
>-        new FakeIceTransport(transport_->mid(), component));
>-    fake_dtls_transports_.emplace_back(
>-        new FakeDtlsTransport(fake_ice_transports_.back().get()));
>-    return transport_->AddChannel(fake_dtls_transports_.back().get(),
>-                                  component);
>+  std::unique_ptr<webrtc::DtlsSrtpTransport> CreateDtlsSrtpTransport(
>+      cricket::DtlsTransportInternal* rtp_dtls_transport,
>+      cricket::DtlsTransportInternal* rtcp_dtls_transport) {
>+    auto dtls_srtp_transport = absl::make_unique<webrtc::DtlsSrtpTransport>(
>+        rtcp_dtls_transport == nullptr);
>+    dtls_srtp_transport->SetDtlsTransports(rtp_dtls_transport,
>+                                           rtcp_dtls_transport);
>+    return dtls_srtp_transport;
>   }
> 
>-  void DestroyChannel(int component) { transport_->RemoveChannel(component); }
>+  // Create a new JsepTransport with a FakeDtlsTransport and a
>+  // FakeIceTransport.
>+  std::unique_ptr<JsepTransport> CreateJsepTransport2(bool rtcp_mux_enabled,
>+                                                      SrtpMode srtp_mode) {
>+    auto ice = absl::make_unique<FakeIceTransport>(kTransportName,
>+                                                   ICE_CANDIDATE_COMPONENT_RTP);
>+    auto rtp_dtls_transport =
>+        absl::make_unique<FakeDtlsTransport>(std::move(ice));
>+
>+    std::unique_ptr<FakeDtlsTransport> rtcp_dtls_transport;
>+    if (!rtcp_mux_enabled) {
>+      ice = absl::make_unique<FakeIceTransport>(kTransportName,
>+                                                ICE_CANDIDATE_COMPONENT_RTCP);
>+      rtcp_dtls_transport =
>+          absl::make_unique<FakeDtlsTransport>(std::move(ice));
>+    }
>+
>+    std::unique_ptr<webrtc::RtpTransport> unencrypted_rtp_transport;
>+    std::unique_ptr<webrtc::SrtpTransport> sdes_transport;
>+    std::unique_ptr<webrtc::DtlsSrtpTransport> dtls_srtp_transport;
>+    switch (srtp_mode) {
>+      case SrtpMode::kSdes:
>+        sdes_transport = CreateSdesTransport(rtp_dtls_transport.get(),
>+                                             rtcp_dtls_transport.get());
>+        sdes_transport_ = sdes_transport.get();
>+        break;
>+      case SrtpMode::kDtlsSrtp:
>+        dtls_srtp_transport = CreateDtlsSrtpTransport(
>+            rtp_dtls_transport.get(), rtcp_dtls_transport.get());
>+        break;
>+      default:
>+        RTC_NOTREACHED();
>+    }
>+
>+    auto jsep_transport = absl::make_unique<JsepTransport>(
>+        kTransportName, /*local_certificate=*/nullptr,
>+        std::move(unencrypted_rtp_transport), std::move(sdes_transport),
>+        std::move(dtls_srtp_transport), std::move(rtp_dtls_transport),
>+        std::move(rtcp_dtls_transport));
> 
>-  void RecreateTransport() {
>-    transport_.reset(new JsepTransport("test content name", nullptr));
>+    signal_rtcp_mux_active_received_ = false;
>+    jsep_transport->SignalRtcpMuxActive.connect(
>+        this, &JsepTransport2Test::OnRtcpMuxActive);
>+    return jsep_transport;
>   }
> 
>-  bool IceCredentialsChanged(const std::string& old_ufrag,
>-                             const std::string& old_pwd,
>-                             const std::string& new_ufrag,
>-                             const std::string& new_pwd) {
>-    return (old_ufrag != new_ufrag) || (old_pwd != new_pwd);
>+  JsepTransportDescription MakeJsepTransportDescription(
>+      bool rtcp_mux_enabled,
>+      const char* ufrag,
>+      const char* pwd,
>+      const rtc::scoped_refptr<rtc::RTCCertificate>& cert,
>+      ConnectionRole role = CONNECTIONROLE_NONE) {
>+    JsepTransportDescription jsep_description;
>+    jsep_description.rtcp_mux_enabled = rtcp_mux_enabled;
>+
>+    std::unique_ptr<rtc::SSLFingerprint> fingerprint;
>+    if (cert) {
>+      fingerprint.reset(rtc::SSLFingerprint::CreateFromCertificate(cert));
>+    }
>+    jsep_description.transport_desc =
>+        TransportDescription(std::vector<std::string>(), ufrag, pwd,
>+                             ICEMODE_FULL, role, fingerprint.get());
>+    return jsep_description;
>   }
> 
>- protected:
>-  std::vector<std::unique_ptr<FakeDtlsTransport>> fake_dtls_transports_;
>-  std::vector<std::unique_ptr<FakeIceTransport>> fake_ice_transports_;
>-  std::unique_ptr<JsepTransport> transport_;
>+  Candidate CreateCandidate(int component) {
>+    Candidate c;
>+    c.set_address(rtc::SocketAddress("192.168.1.1", 8000));
>+    c.set_component(component);
>+    c.set_protocol(UDP_PROTOCOL_NAME);
>+    c.set_priority(1);
>+    return c;
>+  }
>+
>+  void OnRtcpMuxActive() { signal_rtcp_mux_active_received_ = true; }
>+
>+  std::unique_ptr<JsepTransport> jsep_transport_;
>+  bool signal_rtcp_mux_active_received_ = false;
>+  // The SrtpTransport is owned by |jsep_transport_|. Keep a raw pointer here
>+  // for testing.
>+  webrtc::SrtpTransport* sdes_transport_ = nullptr;
> };
> 
>-// This test verifies transports are created with proper ICE ufrag/password
>-// after a transport description is applied.
>-TEST_F(JsepTransportTest, TestIceTransportParameters) {
>-  EXPECT_TRUE(SetupFakeTransports(1));
>-  TransportDescription local_desc(kIceUfrag1, kIcePwd1);
>-  ASSERT_TRUE(transport_->SetLocalTransportDescription(local_desc,
>-                                                       SdpType::kOffer, NULL));
>-  EXPECT_EQ(ICEMODE_FULL, fake_ice_transports_[0]->remote_ice_mode());
>-  EXPECT_EQ(kIceUfrag1, fake_ice_transports_[0]->ice_ufrag());
>-  EXPECT_EQ(kIcePwd1, fake_ice_transports_[0]->ice_pwd());
>-
>-  TransportDescription remote_desc(kIceUfrag2, kIcePwd2);
>-  ASSERT_TRUE(transport_->SetRemoteTransportDescription(
>-      remote_desc, SdpType::kAnswer, NULL));
>-  EXPECT_EQ(ICEMODE_FULL, fake_ice_transports_[0]->remote_ice_mode());
>-  EXPECT_EQ(kIceUfrag2, fake_ice_transports_[0]->remote_ice_ufrag());
>-  EXPECT_EQ(kIcePwd2, fake_ice_transports_[0]->remote_ice_pwd());
>+// The parameterized tests cover both cases when RTCP mux is enable and
>+// disabled.
>+class JsepTransport2WithRtcpMux : public JsepTransport2Test,
>+                                  public testing::WithParamInterface<bool> {};
>+
>+// This test verifies the ICE parameters are properly applied to the transports.
>+TEST_P(JsepTransport2WithRtcpMux, SetIceParameters) {
>+  bool rtcp_mux_enabled = GetParam();
>+  jsep_transport_ = CreateJsepTransport2(rtcp_mux_enabled, SrtpMode::kDtlsSrtp);
>+
>+  JsepTransportDescription jsep_description;
>+  jsep_description.transport_desc = TransportDescription(kIceUfrag1, kIcePwd1);
>+  jsep_description.rtcp_mux_enabled = rtcp_mux_enabled;
>+  ASSERT_TRUE(
>+      jsep_transport_
>+          ->SetLocalJsepTransportDescription(jsep_description, SdpType::kOffer)
>+          .ok());
>+  auto fake_ice_transport = static_cast<FakeIceTransport*>(
>+      jsep_transport_->rtp_dtls_transport()->ice_transport());
>+  EXPECT_EQ(ICEMODE_FULL, fake_ice_transport->remote_ice_mode());
>+  EXPECT_EQ(kIceUfrag1, fake_ice_transport->ice_ufrag());
>+  EXPECT_EQ(kIcePwd1, fake_ice_transport->ice_pwd());
>+  if (!rtcp_mux_enabled) {
>+    fake_ice_transport = static_cast<FakeIceTransport*>(
>+        jsep_transport_->rtcp_dtls_transport()->ice_transport());
>+    ASSERT_TRUE(fake_ice_transport);
>+    EXPECT_EQ(ICEMODE_FULL, fake_ice_transport->remote_ice_mode());
>+    EXPECT_EQ(kIceUfrag1, fake_ice_transport->ice_ufrag());
>+    EXPECT_EQ(kIcePwd1, fake_ice_transport->ice_pwd());
>+  }
>+
>+  jsep_description.transport_desc = TransportDescription(kIceUfrag2, kIcePwd2);
>+  ASSERT_TRUE(jsep_transport_
>+                  ->SetRemoteJsepTransportDescription(jsep_description,
>+                                                      SdpType::kAnswer)
>+                  .ok());
>+  fake_ice_transport = static_cast<FakeIceTransport*>(
>+      jsep_transport_->rtp_dtls_transport()->ice_transport());
>+  EXPECT_EQ(ICEMODE_FULL, fake_ice_transport->remote_ice_mode());
>+  EXPECT_EQ(kIceUfrag2, fake_ice_transport->remote_ice_ufrag());
>+  EXPECT_EQ(kIcePwd2, fake_ice_transport->remote_ice_pwd());
>+  if (!rtcp_mux_enabled) {
>+    fake_ice_transport = static_cast<FakeIceTransport*>(
>+        jsep_transport_->rtcp_dtls_transport()->ice_transport());
>+    ASSERT_TRUE(fake_ice_transport);
>+    EXPECT_EQ(ICEMODE_FULL, fake_ice_transport->remote_ice_mode());
>+    EXPECT_EQ(kIceUfrag2, fake_ice_transport->remote_ice_ufrag());
>+    EXPECT_EQ(kIcePwd2, fake_ice_transport->remote_ice_pwd());
>+  }
> }
> 
>-// Similarly, test that DTLS parameters are applied after a transport
>-// description is applied.
>-TEST_F(JsepTransportTest, TestDtlsTransportParameters) {
>-  EXPECT_TRUE(SetupFakeTransports(1));
>+// Similarly, test DTLS parameters are properly applied to the transports.
>+TEST_P(JsepTransport2WithRtcpMux, SetDtlsParameters) {
>+  bool rtcp_mux_enabled = GetParam();
>+  jsep_transport_ = CreateJsepTransport2(rtcp_mux_enabled, SrtpMode::kDtlsSrtp);
> 
>   // Create certificates.
>   rtc::scoped_refptr<rtc::RTCCertificate> local_cert =
>@@ -104,31 +210,49 @@ TEST_F(JsepTransportTest, TestDtlsTransportParameters) {
>   rtc::scoped_refptr<rtc::RTCCertificate> remote_cert =
>       rtc::RTCCertificate::Create(std::unique_ptr<rtc::SSLIdentity>(
>           rtc::SSLIdentity::Generate("remote", rtc::KT_DEFAULT)));
>-  transport_->SetLocalCertificate(local_cert);
>-
>-  // Apply offer/answer.
>-  TransportDescription local_desc = MakeTransportDescription(
>-      kIceUfrag1, kIcePwd1, local_cert, CONNECTIONROLE_ACTPASS);
>-  ASSERT_TRUE(transport_->SetLocalTransportDescription(
>-      local_desc, SdpType::kOffer, nullptr));
>-  TransportDescription remote_desc = MakeTransportDescription(
>-      kIceUfrag2, kIcePwd2, remote_cert, CONNECTIONROLE_ACTIVE);
>-  ASSERT_TRUE(transport_->SetRemoteTransportDescription(
>-      remote_desc, SdpType::kAnswer, nullptr));
>+  jsep_transport_->SetLocalCertificate(local_cert);
>+
>+  // Apply offer.
>+  JsepTransportDescription local_description =
>+      MakeJsepTransportDescription(rtcp_mux_enabled, kIceUfrag1, kIcePwd1,
>+                                   local_cert, CONNECTIONROLE_ACTPASS);
>+  ASSERT_TRUE(
>+      jsep_transport_
>+          ->SetLocalJsepTransportDescription(local_description, SdpType::kOffer)
>+          .ok());
>+  // Apply Answer.
>+  JsepTransportDescription remote_description =
>+      MakeJsepTransportDescription(rtcp_mux_enabled, kIceUfrag2, kIcePwd2,
>+                                   remote_cert, CONNECTIONROLE_ACTIVE);
>+  ASSERT_TRUE(jsep_transport_
>+                  ->SetRemoteJsepTransportDescription(remote_description,
>+                                                      SdpType::kAnswer)
>+                  .ok());
> 
>   // Verify that SSL role and remote fingerprint were set correctly based on
>   // transport descriptions.
>-  rtc::SSLRole role;
>-  EXPECT_TRUE(fake_dtls_transports_[0]->GetSslRole(&role));
>+  auto role = jsep_transport_->GetDtlsRole();
>+  ASSERT_TRUE(role);
>   EXPECT_EQ(rtc::SSL_SERVER, role);  // Because remote description was "active".
>-  EXPECT_EQ(remote_desc.identity_fingerprint->ToString(),
>-            fake_dtls_transports_[0]->dtls_fingerprint().ToString());
>+  auto fake_dtls =
>+      static_cast<FakeDtlsTransport*>(jsep_transport_->rtp_dtls_transport());
>+  EXPECT_EQ(remote_description.transport_desc.identity_fingerprint->ToString(),
>+            fake_dtls->dtls_fingerprint().ToString());
>+
>+  if (!rtcp_mux_enabled) {
>+    auto fake_rtcp_dtls =
>+        static_cast<FakeDtlsTransport*>(jsep_transport_->rtcp_dtls_transport());
>+    EXPECT_EQ(
>+        remote_description.transport_desc.identity_fingerprint->ToString(),
>+        fake_rtcp_dtls->dtls_fingerprint().ToString());
>+  }
> }
> 
> // Same as above test, but with remote transport description using
> // CONNECTIONROLE_PASSIVE, expecting SSL_CLIENT role.
>-TEST_F(JsepTransportTest, TestDtlsTransportParametersWithPassiveAnswer) {
>-  EXPECT_TRUE(SetupFakeTransports(1));
>+TEST_P(JsepTransport2WithRtcpMux, SetDtlsParametersWithPassiveAnswer) {
>+  bool rtcp_mux_enabled = GetParam();
>+  jsep_transport_ = CreateJsepTransport2(rtcp_mux_enabled, SrtpMode::kDtlsSrtp);
> 
>   // Create certificates.
>   rtc::scoped_refptr<rtc::RTCCertificate> local_cert =
>@@ -137,140 +261,114 @@ TEST_F(JsepTransportTest, TestDtlsTransportParametersWithPassiveAnswer) {
>   rtc::scoped_refptr<rtc::RTCCertificate> remote_cert =
>       rtc::RTCCertificate::Create(std::unique_ptr<rtc::SSLIdentity>(
>           rtc::SSLIdentity::Generate("remote", rtc::KT_DEFAULT)));
>-  transport_->SetLocalCertificate(local_cert);
>-
>-  // Apply offer/answer.
>-  TransportDescription local_desc = MakeTransportDescription(
>-      kIceUfrag1, kIcePwd1, local_cert, CONNECTIONROLE_ACTPASS);
>-  ASSERT_TRUE(transport_->SetLocalTransportDescription(
>-      local_desc, SdpType::kOffer, nullptr));
>-  TransportDescription remote_desc = MakeTransportDescription(
>-      kIceUfrag2, kIcePwd2, remote_cert, CONNECTIONROLE_PASSIVE);
>-  ASSERT_TRUE(transport_->SetRemoteTransportDescription(
>-      remote_desc, SdpType::kAnswer, nullptr));
>+  jsep_transport_->SetLocalCertificate(local_cert);
>+
>+  // Apply offer.
>+  JsepTransportDescription local_description =
>+      MakeJsepTransportDescription(rtcp_mux_enabled, kIceUfrag1, kIcePwd1,
>+                                   local_cert, CONNECTIONROLE_ACTPASS);
>+  ASSERT_TRUE(
>+      jsep_transport_
>+          ->SetLocalJsepTransportDescription(local_description, SdpType::kOffer)
>+          .ok());
>+  // Apply Answer.
>+  JsepTransportDescription remote_description =
>+      MakeJsepTransportDescription(rtcp_mux_enabled, kIceUfrag2, kIcePwd2,
>+                                   remote_cert, CONNECTIONROLE_PASSIVE);
>+  ASSERT_TRUE(jsep_transport_
>+                  ->SetRemoteJsepTransportDescription(remote_description,
>+                                                      SdpType::kAnswer)
>+                  .ok());
> 
>   // Verify that SSL role and remote fingerprint were set correctly based on
>   // transport descriptions.
>-  rtc::SSLRole role;
>-  EXPECT_TRUE(fake_dtls_transports_[0]->GetSslRole(&role));
>+  auto role = jsep_transport_->GetDtlsRole();
>+  ASSERT_TRUE(role);
>   EXPECT_EQ(rtc::SSL_CLIENT,
>             role);  // Because remote description was "passive".
>-  EXPECT_EQ(remote_desc.identity_fingerprint->ToString(),
>-            fake_dtls_transports_[0]->dtls_fingerprint().ToString());
>-}
>-
>-// Add two DtlsTransports/IceTransports and make sure parameters are applied to
>-// both of them. Applicable when RTP/RTCP are not multiplexed, so they share
>-// the same parameters but different connections.
>-TEST_F(JsepTransportTest, TestTransportParametersAppliedToTwoComponents) {
>-  EXPECT_TRUE(SetupFakeTransports(1));
>-  EXPECT_TRUE(SetupFakeTransports(2));
>-
>-  // Create certificates.
>-  rtc::scoped_refptr<rtc::RTCCertificate> local_cert =
>-      rtc::RTCCertificate::Create(std::unique_ptr<rtc::SSLIdentity>(
>-          rtc::SSLIdentity::Generate("local", rtc::KT_DEFAULT)));
>-  rtc::scoped_refptr<rtc::RTCCertificate> remote_cert =
>-      rtc::RTCCertificate::Create(std::unique_ptr<rtc::SSLIdentity>(
>-          rtc::SSLIdentity::Generate("remote", rtc::KT_DEFAULT)));
>-  transport_->SetLocalCertificate(local_cert);
>-
>-  // Apply offer/answer.
>-  TransportDescription local_desc = MakeTransportDescription(
>-      kIceUfrag1, kIcePwd1, local_cert, CONNECTIONROLE_ACTPASS);
>-  ASSERT_TRUE(transport_->SetLocalTransportDescription(
>-      local_desc, SdpType::kOffer, nullptr));
>-  TransportDescription remote_desc = MakeTransportDescription(
>-      kIceUfrag2, kIcePwd2, remote_cert, CONNECTIONROLE_ACTIVE);
>-  ASSERT_TRUE(transport_->SetRemoteTransportDescription(
>-      remote_desc, SdpType::kAnswer, nullptr));
>-
>-  for (int i = 0; i < 1; ++i) {
>-    // Verify parameters of ICE transports.
>-    EXPECT_EQ(ICEMODE_FULL, fake_ice_transports_[i]->remote_ice_mode());
>-    EXPECT_EQ(kIceUfrag1, fake_ice_transports_[i]->ice_ufrag());
>-    EXPECT_EQ(kIcePwd1, fake_ice_transports_[i]->ice_pwd());
>-    EXPECT_EQ(kIceUfrag2, fake_ice_transports_[i]->remote_ice_ufrag());
>-    EXPECT_EQ(kIcePwd2, fake_ice_transports_[i]->remote_ice_pwd());
>-    // Verify parameters of DTLS transports.
>-    rtc::SSLRole role;
>-    EXPECT_TRUE(fake_dtls_transports_[i]->GetSslRole(&role));
>-    EXPECT_EQ(rtc::SSL_SERVER,
>-              role);  // Because remote description was "active".
>-    EXPECT_EQ(remote_desc.identity_fingerprint->ToString(),
>-              fake_dtls_transports_[i]->dtls_fingerprint().ToString());
>+  auto fake_dtls =
>+      static_cast<FakeDtlsTransport*>(jsep_transport_->rtp_dtls_transport());
>+  EXPECT_EQ(remote_description.transport_desc.identity_fingerprint->ToString(),
>+            fake_dtls->dtls_fingerprint().ToString());
>+
>+  if (!rtcp_mux_enabled) {
>+    auto fake_rtcp_dtls =
>+        static_cast<FakeDtlsTransport*>(jsep_transport_->rtcp_dtls_transport());
>+    EXPECT_EQ(
>+        remote_description.transport_desc.identity_fingerprint->ToString(),
>+        fake_rtcp_dtls->dtls_fingerprint().ToString());
>   }
> }
> 
>-// Verifies that IceCredentialsChanged returns true when either ufrag or pwd
>-// changed, and false in other cases.
>-TEST_F(JsepTransportTest, TestIceCredentialsChanged) {
>-  EXPECT_TRUE(IceCredentialsChanged("u1", "p1", "u2", "p2"));
>-  EXPECT_TRUE(IceCredentialsChanged("u1", "p1", "u2", "p1"));
>-  EXPECT_TRUE(IceCredentialsChanged("u1", "p1", "u1", "p2"));
>-  EXPECT_FALSE(IceCredentialsChanged("u1", "p1", "u1", "p1"));
>-}
>-
>-// Tests SetNeedsIceRestartFlag and NeedsIceRestart, ensuring NeedsIceRestart
>+// Tests SetNeedsIceRestartFlag and need_ice_restart, ensuring needs_ice_restart
> // only starts returning "false" once an ICE restart has been initiated.
>-TEST_F(JsepTransportTest, NeedsIceRestart) {
>-  // Do initial offer/answer so there's something to restart.
>-  TransportDescription local_desc(kIceUfrag1, kIcePwd1);
>-  TransportDescription remote_desc(kIceUfrag1, kIcePwd1);
>-  ASSERT_TRUE(transport_->SetLocalTransportDescription(
>-      local_desc, SdpType::kOffer, nullptr));
>-  ASSERT_TRUE(transport_->SetRemoteTransportDescription(
>-      remote_desc, SdpType::kAnswer, nullptr));
>-
>+TEST_P(JsepTransport2WithRtcpMux, NeedsIceRestart) {
>+  bool rtcp_mux_enabled = GetParam();
>+  jsep_transport_ = CreateJsepTransport2(rtcp_mux_enabled, SrtpMode::kDtlsSrtp);
>+
>+  // Use the same JsepTransportDescription for both offer and answer.
>+  JsepTransportDescription description;
>+  description.transport_desc = TransportDescription(kIceUfrag1, kIcePwd1);
>+  ASSERT_TRUE(
>+      jsep_transport_
>+          ->SetLocalJsepTransportDescription(description, SdpType::kOffer)
>+          .ok());
>+  ASSERT_TRUE(
>+      jsep_transport_
>+          ->SetRemoteJsepTransportDescription(description, SdpType::kAnswer)
>+          .ok());
>   // Flag initially should be false.
>-  EXPECT_FALSE(transport_->NeedsIceRestart());
>+  EXPECT_FALSE(jsep_transport_->needs_ice_restart());
> 
>   // After setting flag, it should be true.
>-  transport_->SetNeedsIceRestartFlag();
>-  EXPECT_TRUE(transport_->NeedsIceRestart());
>-
>-  // Doing an identical offer/answer shouldn't clear the flag.
>-  ASSERT_TRUE(transport_->SetLocalTransportDescription(
>-      local_desc, SdpType::kOffer, nullptr));
>-  ASSERT_TRUE(transport_->SetRemoteTransportDescription(
>-      remote_desc, SdpType::kAnswer, nullptr));
>-  EXPECT_TRUE(transport_->NeedsIceRestart());
>+  jsep_transport_->SetNeedsIceRestartFlag();
>+  EXPECT_TRUE(jsep_transport_->needs_ice_restart());
>+
>+  ASSERT_TRUE(
>+      jsep_transport_
>+          ->SetLocalJsepTransportDescription(description, SdpType::kOffer)
>+          .ok());
>+  ASSERT_TRUE(
>+      jsep_transport_
>+          ->SetRemoteJsepTransportDescription(description, SdpType::kAnswer)
>+          .ok());
>+  EXPECT_TRUE(jsep_transport_->needs_ice_restart());
> 
>   // Doing an offer/answer that restarts ICE should clear the flag.
>-  TransportDescription ice_restart_local_desc(kIceUfrag2, kIcePwd2);
>-  TransportDescription ice_restart_remote_desc(kIceUfrag2, kIcePwd2);
>-  ASSERT_TRUE(transport_->SetLocalTransportDescription(
>-      ice_restart_local_desc, SdpType::kOffer, nullptr));
>-  ASSERT_TRUE(transport_->SetRemoteTransportDescription(
>-      ice_restart_remote_desc, SdpType::kAnswer, nullptr));
>-  EXPECT_FALSE(transport_->NeedsIceRestart());
>+  description.transport_desc = TransportDescription(kIceUfrag2, kIcePwd2);
>+  ASSERT_TRUE(
>+      jsep_transport_
>+          ->SetLocalJsepTransportDescription(description, SdpType::kOffer)
>+          .ok());
>+  ASSERT_TRUE(
>+      jsep_transport_
>+          ->SetRemoteJsepTransportDescription(description, SdpType::kAnswer)
>+          .ok());
>+  EXPECT_FALSE(jsep_transport_->needs_ice_restart());
> }
> 
>-TEST_F(JsepTransportTest, TestGetStats) {
>-  EXPECT_TRUE(SetupFakeTransports(1));
>+TEST_P(JsepTransport2WithRtcpMux, GetStats) {
>+  bool rtcp_mux_enabled = GetParam();
>+  jsep_transport_ = CreateJsepTransport2(rtcp_mux_enabled, SrtpMode::kDtlsSrtp);
>+
>+  size_t expected_stats_size = rtcp_mux_enabled ? 1u : 2u;
>   TransportStats stats;
>-  EXPECT_TRUE(transport_->GetStats(&stats));
>-  // Note that this tests the behavior of a FakeIceTransport.
>-  ASSERT_EQ(1U, stats.channel_stats.size());
>-  EXPECT_EQ(1, stats.channel_stats[0].component);
>-  // Set local transport description for FakeTransport before connecting.
>-  TransportDescription faketransport_desc(
>-      std::vector<std::string>(), rtc::CreateRandomString(ICE_UFRAG_LENGTH),
>-      rtc::CreateRandomString(ICE_PWD_LENGTH), ICEMODE_FULL,
>-      CONNECTIONROLE_NONE, nullptr);
>-  transport_->SetLocalTransportDescription(faketransport_desc, SdpType::kOffer,
>-                                           nullptr);
>-  EXPECT_TRUE(transport_->GetStats(&stats));
>-  ASSERT_EQ(1U, stats.channel_stats.size());
>-  EXPECT_EQ(1, stats.channel_stats[0].component);
>+  EXPECT_TRUE(jsep_transport_->GetStats(&stats));
>+  EXPECT_EQ(expected_stats_size, stats.channel_stats.size());
>+  EXPECT_EQ(ICE_CANDIDATE_COMPONENT_RTP, stats.channel_stats[0].component);
>+  if (!rtcp_mux_enabled) {
>+    EXPECT_EQ(ICE_CANDIDATE_COMPONENT_RTCP, stats.channel_stats[1].component);
>+  }
> }
> 
> // Tests that VerifyCertificateFingerprint only returns true when the
> // certificate matches the fingerprint.
>-TEST_F(JsepTransportTest, TestVerifyCertificateFingerprint) {
>-  std::string error_desc;
>+TEST_P(JsepTransport2WithRtcpMux, VerifyCertificateFingerprint) {
>+  bool rtcp_mux_enabled = GetParam();
>+  jsep_transport_ = CreateJsepTransport2(rtcp_mux_enabled, SrtpMode::kDtlsSrtp);
>+
>   EXPECT_FALSE(
>-      transport_->VerifyCertificateFingerprint(nullptr, nullptr, &error_desc));
>+      jsep_transport_->VerifyCertificateFingerprint(nullptr, nullptr).ok());
>   rtc::KeyType key_types[] = {rtc::KT_RSA, rtc::KT_ECDSA};
> 
>   for (auto& key_type : key_types) {
>@@ -287,41 +385,40 @@ TEST_F(JsepTransportTest, TestVerifyCertificateFingerprint) {
>         rtc::SSLFingerprint::Create(digest_algorithm, certificate->identity()));
>     ASSERT_NE(nullptr, good_fingerprint);
> 
>-    EXPECT_TRUE(transport_->VerifyCertificateFingerprint(
>-        certificate.get(), good_fingerprint.get(), &error_desc));
>-    EXPECT_FALSE(transport_->VerifyCertificateFingerprint(
>-        certificate.get(), nullptr, &error_desc));
>-    EXPECT_FALSE(transport_->VerifyCertificateFingerprint(
>-        nullptr, good_fingerprint.get(), &error_desc));
>+    EXPECT_TRUE(jsep_transport_
>+                    ->VerifyCertificateFingerprint(certificate.get(),
>+                                                   good_fingerprint.get())
>+                    .ok());
>+    EXPECT_FALSE(jsep_transport_
>+                     ->VerifyCertificateFingerprint(certificate.get(), nullptr)
>+                     .ok());
>+    EXPECT_FALSE(
>+        jsep_transport_
>+            ->VerifyCertificateFingerprint(nullptr, good_fingerprint.get())
>+            .ok());
> 
>     rtc::SSLFingerprint bad_fingerprint = *good_fingerprint;
>     bad_fingerprint.digest.AppendData("0", 1);
>-    EXPECT_FALSE(transport_->VerifyCertificateFingerprint(
>-        certificate.get(), &bad_fingerprint, &error_desc));
>+    EXPECT_FALSE(
>+        jsep_transport_
>+            ->VerifyCertificateFingerprint(certificate.get(), &bad_fingerprint)
>+            .ok());
>   }
> }
> 
> // Tests the logic of DTLS role negotiation for an initial offer/answer.
>-TEST_F(JsepTransportTest, DtlsRoleNegotiation) {
>+TEST_P(JsepTransport2WithRtcpMux, ValidDtlsRoleNegotiation) {
>+  bool rtcp_mux_enabled = GetParam();
>   // Just use the same certificate for both sides; doesn't really matter in a
>   // non end-to-end test.
>   rtc::scoped_refptr<rtc::RTCCertificate> certificate =
>       rtc::RTCCertificate::Create(std::unique_ptr<rtc::SSLIdentity>(
>           rtc::SSLIdentity::Generate("testing", rtc::KT_ECDSA)));
> 
>-  TransportDescription local_desc =
>-      MakeTransportDescription(kIceUfrag1, kIcePwd1, certificate);
>-  TransportDescription remote_desc =
>-      MakeTransportDescription(kIceUfrag2, kIcePwd2, certificate);
>-
>-  struct NegotiateRoleParams {
>-    ConnectionRole local_role;
>-    ConnectionRole remote_role;
>-    SdpType local_type;
>-    SdpType remote_type;
>-  };
>-
>-  std::string error_desc;
>+  JsepTransportDescription local_description = MakeJsepTransportDescription(
>+      rtcp_mux_enabled, kIceUfrag1, kIcePwd1, certificate);
>+  JsepTransportDescription remote_description = MakeJsepTransportDescription(
>+      rtcp_mux_enabled, kIceUfrag2, kIcePwd2, certificate);
> 
>   // Parameters which set the SSL role to SSL_CLIENT.
>   NegotiateRoleParams valid_client_params[] = {
>@@ -335,25 +432,34 @@ TEST_F(JsepTransportTest, DtlsRoleNegotiation) {
>        SdpType::kPrAnswer}};
> 
>   for (auto& param : valid_client_params) {
>-    RecreateTransport();
>-    transport_->SetLocalCertificate(certificate);
>+    jsep_transport_ =
>+        CreateJsepTransport2(rtcp_mux_enabled, SrtpMode::kDtlsSrtp);
>+    jsep_transport_->SetLocalCertificate(certificate);
> 
>-    local_desc.connection_role = param.local_role;
>-    remote_desc.connection_role = param.remote_role;
>+    local_description.transport_desc.connection_role = param.local_role;
>+    remote_description.transport_desc.connection_role = param.remote_role;
> 
>     // Set the offer first.
>     if (param.local_type == SdpType::kOffer) {
>-      EXPECT_TRUE(transport_->SetLocalTransportDescription(
>-          local_desc, param.local_type, nullptr));
>-      EXPECT_TRUE(transport_->SetRemoteTransportDescription(
>-          remote_desc, param.remote_type, nullptr));
>+      EXPECT_TRUE(jsep_transport_
>+                      ->SetLocalJsepTransportDescription(local_description,
>+                                                         param.local_type)
>+                      .ok());
>+      EXPECT_TRUE(jsep_transport_
>+                      ->SetRemoteJsepTransportDescription(remote_description,
>+                                                          param.remote_type)
>+                      .ok());
>     } else {
>-      EXPECT_TRUE(transport_->SetRemoteTransportDescription(
>-          remote_desc, param.remote_type, nullptr));
>-      EXPECT_TRUE(transport_->SetLocalTransportDescription(
>-          local_desc, param.local_type, nullptr));
>+      EXPECT_TRUE(jsep_transport_
>+                      ->SetRemoteJsepTransportDescription(remote_description,
>+                                                          param.remote_type)
>+                      .ok());
>+      EXPECT_TRUE(jsep_transport_
>+                      ->SetLocalJsepTransportDescription(local_description,
>+                                                         param.local_type)
>+                      .ok());
>     }
>-    EXPECT_EQ(rtc::SSL_CLIENT, *transport_->GetSslRole());
>+    EXPECT_EQ(rtc::SSL_CLIENT, *jsep_transport_->GetDtlsRole());
>   }
> 
>   // Parameters which set the SSL role to SSL_SERVER.
>@@ -368,28 +474,51 @@ TEST_F(JsepTransportTest, DtlsRoleNegotiation) {
>        SdpType::kPrAnswer}};
> 
>   for (auto& param : valid_server_params) {
>-    RecreateTransport();
>-    transport_->SetLocalCertificate(certificate);
>+    jsep_transport_ =
>+        CreateJsepTransport2(rtcp_mux_enabled, SrtpMode::kDtlsSrtp);
>+    jsep_transport_->SetLocalCertificate(certificate);
> 
>-    local_desc.connection_role = param.local_role;
>-    remote_desc.connection_role = param.remote_role;
>+    local_description.transport_desc.connection_role = param.local_role;
>+    remote_description.transport_desc.connection_role = param.remote_role;
> 
>     // Set the offer first.
>     if (param.local_type == SdpType::kOffer) {
>-      EXPECT_TRUE(transport_->SetLocalTransportDescription(
>-          local_desc, param.local_type, nullptr));
>-      EXPECT_TRUE(transport_->SetRemoteTransportDescription(
>-          remote_desc, param.remote_type, nullptr));
>+      EXPECT_TRUE(jsep_transport_
>+                      ->SetLocalJsepTransportDescription(local_description,
>+                                                         param.local_type)
>+                      .ok());
>+      EXPECT_TRUE(jsep_transport_
>+                      ->SetRemoteJsepTransportDescription(remote_description,
>+                                                          param.remote_type)
>+                      .ok());
>     } else {
>-      EXPECT_TRUE(transport_->SetRemoteTransportDescription(
>-          remote_desc, param.remote_type, nullptr));
>-      EXPECT_TRUE(transport_->SetLocalTransportDescription(
>-          local_desc, param.local_type, nullptr));
>+      EXPECT_TRUE(jsep_transport_
>+                      ->SetRemoteJsepTransportDescription(remote_description,
>+                                                          param.remote_type)
>+                      .ok());
>+      EXPECT_TRUE(jsep_transport_
>+                      ->SetLocalJsepTransportDescription(local_description,
>+                                                         param.local_type)
>+                      .ok());
>     }
>-    EXPECT_EQ(rtc::SSL_SERVER, *transport_->GetSslRole());
>+    EXPECT_EQ(rtc::SSL_SERVER, *jsep_transport_->GetDtlsRole());
>   }
>+}
>+
>+// Tests the logic of DTLS role negotiation for an initial offer/answer.
>+TEST_P(JsepTransport2WithRtcpMux, InvalidDtlsRoleNegotiation) {
>+  bool rtcp_mux_enabled = GetParam();
>+  // Just use the same certificate for both sides; doesn't really matter in a
>+  // non end-to-end test.
>+  rtc::scoped_refptr<rtc::RTCCertificate> certificate =
>+      rtc::RTCCertificate::Create(std::unique_ptr<rtc::SSLIdentity>(
>+          rtc::SSLIdentity::Generate("testing", rtc::KT_ECDSA)));
>+
>+  JsepTransportDescription local_description = MakeJsepTransportDescription(
>+      rtcp_mux_enabled, kIceUfrag1, kIcePwd1, certificate);
>+  JsepTransportDescription remote_description = MakeJsepTransportDescription(
>+      rtcp_mux_enabled, kIceUfrag2, kIcePwd2, certificate);
> 
>-  // Invalid parameters due to both peers having a duplicate role.
>   NegotiateRoleParams duplicate_params[] = {
>       {CONNECTIONROLE_ACTIVE, CONNECTIONROLE_ACTIVE, SdpType::kAnswer,
>        SdpType::kOffer},
>@@ -417,23 +546,31 @@ TEST_F(JsepTransportTest, DtlsRoleNegotiation) {
>        SdpType::kPrAnswer}};
> 
>   for (auto& param : duplicate_params) {
>-    RecreateTransport();
>-    transport_->SetLocalCertificate(certificate);
>+    jsep_transport_ =
>+        CreateJsepTransport2(rtcp_mux_enabled, SrtpMode::kDtlsSrtp);
>+    jsep_transport_->SetLocalCertificate(certificate);
> 
>-    local_desc.connection_role = param.local_role;
>-    remote_desc.connection_role = param.remote_role;
>+    local_description.transport_desc.connection_role = param.local_role;
>+    remote_description.transport_desc.connection_role = param.remote_role;
> 
>-    // Set the offer first.
>     if (param.local_type == SdpType::kOffer) {
>-      EXPECT_TRUE(transport_->SetLocalTransportDescription(
>-          local_desc, param.local_type, nullptr));
>-      EXPECT_FALSE(transport_->SetRemoteTransportDescription(
>-          remote_desc, param.remote_type, nullptr));
>+      EXPECT_TRUE(jsep_transport_
>+                      ->SetLocalJsepTransportDescription(local_description,
>+                                                         param.local_type)
>+                      .ok());
>+      EXPECT_FALSE(jsep_transport_
>+                       ->SetRemoteJsepTransportDescription(remote_description,
>+                                                           param.remote_type)
>+                       .ok());
>     } else {
>-      EXPECT_TRUE(transport_->SetRemoteTransportDescription(
>-          remote_desc, param.remote_type, nullptr));
>-      EXPECT_FALSE(transport_->SetLocalTransportDescription(
>-          local_desc, param.local_type, nullptr));
>+      EXPECT_TRUE(jsep_transport_
>+                      ->SetRemoteJsepTransportDescription(remote_description,
>+                                                          param.remote_type)
>+                      .ok());
>+      EXPECT_FALSE(jsep_transport_
>+                       ->SetLocalJsepTransportDescription(local_description,
>+                                                          param.local_type)
>+                       .ok());
>     }
>   }
> 
>@@ -465,209 +602,638 @@ TEST_F(JsepTransportTest, DtlsRoleNegotiation) {
>        SdpType::kPrAnswer}};
> 
>   for (auto& param : offerer_without_actpass_params) {
>-    RecreateTransport();
>-    transport_->SetLocalCertificate(certificate);
>+    jsep_transport_ =
>+        CreateJsepTransport2(rtcp_mux_enabled, SrtpMode::kDtlsSrtp);
>+    jsep_transport_->SetLocalCertificate(certificate);
> 
>-    local_desc.connection_role = param.local_role;
>-    remote_desc.connection_role = param.remote_role;
>+    local_description.transport_desc.connection_role = param.local_role;
>+    remote_description.transport_desc.connection_role = param.remote_role;
> 
>-    // Set the offer first.
>-    // TODO(deadbeef): Really this should fail as soon as the offer is
>-    // attempted to be applied, and not when the answer is applied.
>     if (param.local_type == SdpType::kOffer) {
>-      EXPECT_TRUE(transport_->SetLocalTransportDescription(
>-          local_desc, param.local_type, nullptr));
>-      EXPECT_FALSE(transport_->SetRemoteTransportDescription(
>-          remote_desc, param.remote_type, nullptr));
>+      EXPECT_TRUE(jsep_transport_
>+                      ->SetLocalJsepTransportDescription(local_description,
>+                                                         param.local_type)
>+                      .ok());
>+      EXPECT_FALSE(jsep_transport_
>+                       ->SetRemoteJsepTransportDescription(remote_description,
>+                                                           param.remote_type)
>+                       .ok());
>     } else {
>-      EXPECT_TRUE(transport_->SetRemoteTransportDescription(
>-          remote_desc, param.remote_type, nullptr));
>-      EXPECT_FALSE(transport_->SetLocalTransportDescription(
>-          local_desc, param.local_type, nullptr));
>+      EXPECT_TRUE(jsep_transport_
>+                      ->SetRemoteJsepTransportDescription(remote_description,
>+                                                          param.remote_type)
>+                      .ok());
>+      EXPECT_FALSE(jsep_transport_
>+                       ->SetLocalJsepTransportDescription(local_description,
>+                                                          param.local_type)
>+                       .ok());
>     }
>   }
> }
> 
>+INSTANTIATE_TEST_CASE_P(JsepTransport2Test,
>+                        JsepTransport2WithRtcpMux,
>+                        testing::Bool());
>+
> // Test that a reoffer in the opposite direction is successful as long as the
> // role isn't changing. Doesn't test every possible combination like the test
> // above.
>-TEST_F(JsepTransportTest, ValidDtlsReofferFromAnswerer) {
>+TEST_F(JsepTransport2Test, ValidDtlsReofferFromAnswerer) {
>   // Just use the same certificate for both sides; doesn't really matter in a
>   // non end-to-end test.
>   rtc::scoped_refptr<rtc::RTCCertificate> certificate =
>       rtc::RTCCertificate::Create(std::unique_ptr<rtc::SSLIdentity>(
>           rtc::SSLIdentity::Generate("testing", rtc::KT_ECDSA)));
>-  transport_->SetLocalCertificate(certificate);
>-
>-  TransportDescription local_offer = MakeTransportDescription(
>-      kIceUfrag1, kIcePwd1, certificate, CONNECTIONROLE_ACTPASS);
>-  TransportDescription remote_answer = MakeTransportDescription(
>-      kIceUfrag2, kIcePwd2, certificate, CONNECTIONROLE_ACTIVE);
>-
>-  EXPECT_TRUE(transport_->SetLocalTransportDescription(
>-      local_offer, SdpType::kOffer, nullptr));
>-  EXPECT_TRUE(transport_->SetRemoteTransportDescription(
>-      remote_answer, SdpType::kAnswer, nullptr));
>+  bool rtcp_mux_enabled = true;
>+  jsep_transport_ = CreateJsepTransport2(rtcp_mux_enabled, SrtpMode::kDtlsSrtp);
>+  jsep_transport_->SetLocalCertificate(certificate);
>+
>+  JsepTransportDescription local_offer =
>+      MakeJsepTransportDescription(rtcp_mux_enabled, kIceUfrag1, kIcePwd1,
>+                                   certificate, CONNECTIONROLE_ACTPASS);
>+  JsepTransportDescription remote_answer =
>+      MakeJsepTransportDescription(rtcp_mux_enabled, kIceUfrag2, kIcePwd2,
>+                                   certificate, CONNECTIONROLE_ACTIVE);
>+
>+  EXPECT_TRUE(
>+      jsep_transport_
>+          ->SetLocalJsepTransportDescription(local_offer, SdpType::kOffer)
>+          .ok());
>+  EXPECT_TRUE(
>+      jsep_transport_
>+          ->SetRemoteJsepTransportDescription(remote_answer, SdpType::kAnswer)
>+          .ok());
> 
>   // We were actpass->active previously, now in the other direction it's
>   // actpass->passive.
>-  TransportDescription remote_offer = MakeTransportDescription(
>-      kIceUfrag2, kIcePwd2, certificate, CONNECTIONROLE_ACTPASS);
>-  TransportDescription local_answer = MakeTransportDescription(
>-      kIceUfrag1, kIcePwd1, certificate, CONNECTIONROLE_PASSIVE);
>-
>-  EXPECT_TRUE(transport_->SetRemoteTransportDescription(
>-      remote_offer, SdpType::kOffer, nullptr));
>-  EXPECT_TRUE(transport_->SetLocalTransportDescription(
>-      local_answer, SdpType::kAnswer, nullptr));
>+  JsepTransportDescription remote_offer =
>+      MakeJsepTransportDescription(rtcp_mux_enabled, kIceUfrag2, kIcePwd2,
>+                                   certificate, CONNECTIONROLE_ACTPASS);
>+  JsepTransportDescription local_answer =
>+      MakeJsepTransportDescription(rtcp_mux_enabled, kIceUfrag1, kIcePwd1,
>+                                   certificate, CONNECTIONROLE_PASSIVE);
>+
>+  EXPECT_TRUE(
>+      jsep_transport_
>+          ->SetRemoteJsepTransportDescription(remote_offer, SdpType::kOffer)
>+          .ok());
>+  EXPECT_TRUE(
>+      jsep_transport_
>+          ->SetLocalJsepTransportDescription(local_answer, SdpType::kAnswer)
>+          .ok());
> }
> 
> // Test that a reoffer in the opposite direction fails if the role changes.
> // Inverse of test above.
>-TEST_F(JsepTransportTest, InvalidDtlsReofferFromAnswerer) {
>+TEST_F(JsepTransport2Test, InvalidDtlsReofferFromAnswerer) {
>   // Just use the same certificate for both sides; doesn't really matter in a
>   // non end-to-end test.
>   rtc::scoped_refptr<rtc::RTCCertificate> certificate =
>       rtc::RTCCertificate::Create(std::unique_ptr<rtc::SSLIdentity>(
>           rtc::SSLIdentity::Generate("testing", rtc::KT_ECDSA)));
>-  transport_->SetLocalCertificate(certificate);
>-
>-  TransportDescription local_offer = MakeTransportDescription(
>-      kIceUfrag1, kIcePwd1, certificate, CONNECTIONROLE_ACTPASS);
>-  TransportDescription remote_answer = MakeTransportDescription(
>-      kIceUfrag2, kIcePwd2, certificate, CONNECTIONROLE_ACTIVE);
>-
>-  EXPECT_TRUE(transport_->SetLocalTransportDescription(
>-      local_offer, SdpType::kOffer, nullptr));
>-  EXPECT_TRUE(transport_->SetRemoteTransportDescription(
>-      remote_answer, SdpType::kAnswer, nullptr));
>+  bool rtcp_mux_enabled = true;
>+  jsep_transport_ = CreateJsepTransport2(rtcp_mux_enabled, SrtpMode::kDtlsSrtp);
>+  jsep_transport_->SetLocalCertificate(certificate);
>+
>+  JsepTransportDescription local_offer =
>+      MakeJsepTransportDescription(rtcp_mux_enabled, kIceUfrag1, kIcePwd1,
>+                                   certificate, CONNECTIONROLE_ACTPASS);
>+  JsepTransportDescription remote_answer =
>+      MakeJsepTransportDescription(rtcp_mux_enabled, kIceUfrag2, kIcePwd2,
>+                                   certificate, CONNECTIONROLE_ACTIVE);
>+
>+  EXPECT_TRUE(
>+      jsep_transport_
>+          ->SetLocalJsepTransportDescription(local_offer, SdpType::kOffer)
>+          .ok());
>+  EXPECT_TRUE(
>+      jsep_transport_
>+          ->SetRemoteJsepTransportDescription(remote_answer, SdpType::kAnswer)
>+          .ok());
> 
>   // Changing role to passive here isn't allowed. Though for some reason this
>   // only fails in SetLocalTransportDescription.
>-  TransportDescription remote_offer = MakeTransportDescription(
>-      kIceUfrag2, kIcePwd2, certificate, CONNECTIONROLE_PASSIVE);
>-  TransportDescription local_answer = MakeTransportDescription(
>-      kIceUfrag1, kIcePwd1, certificate, CONNECTIONROLE_ACTIVE);
>-
>-  EXPECT_TRUE(transport_->SetRemoteTransportDescription(
>-      remote_offer, SdpType::kOffer, nullptr));
>-  EXPECT_FALSE(transport_->SetLocalTransportDescription(
>-      local_answer, SdpType::kAnswer, nullptr));
>+  JsepTransportDescription remote_offer =
>+      MakeJsepTransportDescription(rtcp_mux_enabled, kIceUfrag2, kIcePwd2,
>+                                   certificate, CONNECTIONROLE_PASSIVE);
>+  JsepTransportDescription local_answer =
>+      MakeJsepTransportDescription(rtcp_mux_enabled, kIceUfrag1, kIcePwd1,
>+                                   certificate, CONNECTIONROLE_ACTIVE);
>+
>+  EXPECT_TRUE(
>+      jsep_transport_
>+          ->SetRemoteJsepTransportDescription(remote_offer, SdpType::kOffer)
>+          .ok());
>+  EXPECT_FALSE(
>+      jsep_transport_
>+          ->SetLocalJsepTransportDescription(local_answer, SdpType::kAnswer)
>+          .ok());
> }
> 
> // Test that a remote offer with the current negotiated role can be accepted.
> // This is allowed by dtls-sdp, though we'll never generate such an offer,
> // since JSEP requires generating "actpass".
>-TEST_F(JsepTransportTest, RemoteOfferWithCurrentNegotiatedDtlsRole) {
>-  // Just use the same certificate in both descriptions; the remote fingerprint
>-  // doesn't matter in a non end-to-end test.
>+TEST_F(JsepTransport2Test, RemoteOfferWithCurrentNegotiatedDtlsRole) {
>   rtc::scoped_refptr<rtc::RTCCertificate> certificate =
>       rtc::RTCCertificate::Create(std::unique_ptr<rtc::SSLIdentity>(
>           rtc::SSLIdentity::Generate("testing", rtc::KT_ECDSA)));
>-  transport_->SetLocalCertificate(certificate);
>+  bool rtcp_mux_enabled = true;
>+  jsep_transport_ = CreateJsepTransport2(rtcp_mux_enabled, SrtpMode::kDtlsSrtp);
>+  jsep_transport_->SetLocalCertificate(certificate);
> 
>-  TransportDescription remote_desc = MakeTransportDescription(
>-      kIceUfrag1, kIcePwd1, certificate, CONNECTIONROLE_ACTPASS);
>-  TransportDescription local_desc = MakeTransportDescription(
>-      kIceUfrag2, kIcePwd2, certificate, CONNECTIONROLE_ACTIVE);
>+  JsepTransportDescription remote_desc =
>+      MakeJsepTransportDescription(rtcp_mux_enabled, kIceUfrag1, kIcePwd1,
>+                                   certificate, CONNECTIONROLE_ACTPASS);
>+  JsepTransportDescription local_desc =
>+      MakeJsepTransportDescription(rtcp_mux_enabled, kIceUfrag2, kIcePwd2,
>+                                   certificate, CONNECTIONROLE_ACTIVE);
> 
>   // Normal initial offer/answer with "actpass" in the offer and "active" in
>   // the answer.
>-  ASSERT_TRUE(transport_->SetRemoteTransportDescription(
>-      remote_desc, SdpType::kOffer, nullptr));
>-  ASSERT_TRUE(transport_->SetLocalTransportDescription(
>-      local_desc, SdpType::kAnswer, nullptr));
>+  ASSERT_TRUE(
>+      jsep_transport_
>+          ->SetRemoteJsepTransportDescription(remote_desc, SdpType::kOffer)
>+          .ok());
>+  ASSERT_TRUE(
>+      jsep_transport_
>+          ->SetLocalJsepTransportDescription(local_desc, SdpType::kAnswer)
>+          .ok());
> 
>   // Sanity check that role was actually negotiated.
>-  rtc::Optional<rtc::SSLRole> role = transport_->GetSslRole();
>+  absl::optional<rtc::SSLRole> role = jsep_transport_->GetDtlsRole();
>   ASSERT_TRUE(role);
>   EXPECT_EQ(rtc::SSL_CLIENT, *role);
> 
>   // Subsequent offer with current negotiated role of "passive".
>-  remote_desc.connection_role = CONNECTIONROLE_PASSIVE;
>-  EXPECT_TRUE(transport_->SetRemoteTransportDescription(
>-      remote_desc, SdpType::kOffer, nullptr));
>-  EXPECT_TRUE(transport_->SetLocalTransportDescription(
>-      local_desc, SdpType::kAnswer, nullptr));
>+  remote_desc.transport_desc.connection_role = CONNECTIONROLE_PASSIVE;
>+  EXPECT_TRUE(
>+      jsep_transport_
>+          ->SetRemoteJsepTransportDescription(remote_desc, SdpType::kOffer)
>+          .ok());
>+  EXPECT_TRUE(
>+      jsep_transport_
>+          ->SetLocalJsepTransportDescription(local_desc, SdpType::kAnswer)
>+          .ok());
> }
> 
> // Test that a remote offer with the inverse of the current negotiated DTLS
> // role is rejected.
>-TEST_F(JsepTransportTest, RemoteOfferThatChangesNegotiatedDtlsRole) {
>+TEST_F(JsepTransport2Test, RemoteOfferThatChangesNegotiatedDtlsRole) {
>   rtc::scoped_refptr<rtc::RTCCertificate> certificate =
>       rtc::RTCCertificate::Create(std::unique_ptr<rtc::SSLIdentity>(
>           rtc::SSLIdentity::Generate("testing", rtc::KT_ECDSA)));
>-  std::unique_ptr<rtc::SSLFingerprint> fingerprint(
>-      rtc::SSLFingerprint::CreateFromCertificate(certificate));
>-  transport_->SetLocalCertificate(certificate);
>-
>-  TransportDescription local_desc(kIceUfrag1, kIcePwd1);
>-  TransportDescription remote_desc(kIceUfrag2, kIcePwd2);
>-  // Just use the same fingerprint in both descriptions; the remote fingerprint
>-  // doesn't matter in a non end-to-end test.
>-  local_desc.identity_fingerprint.reset(
>-      TransportDescription::CopyFingerprint(fingerprint.get()));
>-  remote_desc.identity_fingerprint.reset(
>-      TransportDescription::CopyFingerprint(fingerprint.get()));
>-
>-  remote_desc.connection_role = CONNECTIONROLE_ACTPASS;
>-  local_desc.connection_role = CONNECTIONROLE_ACTIVE;
>+  bool rtcp_mux_enabled = true;
>+  jsep_transport_ = CreateJsepTransport2(rtcp_mux_enabled, SrtpMode::kDtlsSrtp);
>+  jsep_transport_->SetLocalCertificate(certificate);
>+
>+  JsepTransportDescription remote_desc =
>+      MakeJsepTransportDescription(rtcp_mux_enabled, kIceUfrag1, kIcePwd1,
>+                                   certificate, CONNECTIONROLE_ACTPASS);
>+  JsepTransportDescription local_desc =
>+      MakeJsepTransportDescription(rtcp_mux_enabled, kIceUfrag2, kIcePwd2,
>+                                   certificate, CONNECTIONROLE_ACTIVE);
> 
>   // Normal initial offer/answer with "actpass" in the offer and "active" in
>   // the answer.
>-  ASSERT_TRUE(transport_->SetRemoteTransportDescription(
>-      remote_desc, SdpType::kOffer, nullptr));
>-  ASSERT_TRUE(transport_->SetLocalTransportDescription(
>-      local_desc, SdpType::kAnswer, nullptr));
>+  ASSERT_TRUE(
>+      jsep_transport_
>+          ->SetRemoteJsepTransportDescription(remote_desc, SdpType::kOffer)
>+          .ok());
>+  ASSERT_TRUE(
>+      jsep_transport_
>+          ->SetLocalJsepTransportDescription(local_desc, SdpType::kAnswer)
>+          .ok());
> 
>   // Sanity check that role was actually negotiated.
>-  rtc::Optional<rtc::SSLRole> role = transport_->GetSslRole();
>+  absl::optional<rtc::SSLRole> role = jsep_transport_->GetDtlsRole();
>   ASSERT_TRUE(role);
>   EXPECT_EQ(rtc::SSL_CLIENT, *role);
> 
>-  // Subsequent offer with "active", which is the opposite of the remote
>-  // endpoint's negotiated role.
>-  // TODO(deadbeef): Really this should fail as soon as the offer is
>-  // attempted to be applied, and not when the answer is applied.
>-  remote_desc.connection_role = CONNECTIONROLE_ACTIVE;
>-  EXPECT_TRUE(transport_->SetRemoteTransportDescription(
>-      remote_desc, SdpType::kOffer, nullptr));
>-  EXPECT_FALSE(transport_->SetLocalTransportDescription(
>-      local_desc, SdpType::kAnswer, nullptr));
>+  // Subsequent offer with current negotiated role of "passive".
>+  remote_desc.transport_desc.connection_role = CONNECTIONROLE_ACTIVE;
>+  EXPECT_TRUE(
>+      jsep_transport_
>+          ->SetRemoteJsepTransportDescription(remote_desc, SdpType::kOffer)
>+          .ok());
>+  EXPECT_FALSE(
>+      jsep_transport_
>+          ->SetLocalJsepTransportDescription(local_desc, SdpType::kAnswer)
>+          .ok());
> }
> 
> // Testing that a legacy client that doesn't use the setup attribute will be
> // interpreted as having an active role.
>-TEST_F(JsepTransportTest, TestDtlsSetupWithLegacyAsAnswerer) {
>+TEST_F(JsepTransport2Test, DtlsSetupWithLegacyAsAnswerer) {
>   rtc::scoped_refptr<rtc::RTCCertificate> certificate =
>       rtc::RTCCertificate::Create(std::unique_ptr<rtc::SSLIdentity>(
>           rtc::SSLIdentity::Generate("testing", rtc::KT_ECDSA)));
>-  std::unique_ptr<rtc::SSLFingerprint> fingerprint(
>-      rtc::SSLFingerprint::CreateFromCertificate(certificate));
>-  transport_->SetLocalCertificate(certificate);
>-
>-  TransportDescription local_desc(kIceUfrag1, kIcePwd1);
>-  TransportDescription remote_desc(kIceUfrag2, kIcePwd2);
>-  // Just use the same fingerprint in both descriptions; the remote fingerprint
>-  // doesn't matter in a non end-to-end test.
>-  local_desc.identity_fingerprint.reset(
>-      TransportDescription::CopyFingerprint(fingerprint.get()));
>-  remote_desc.identity_fingerprint.reset(
>-      TransportDescription::CopyFingerprint(fingerprint.get()));
>-
>-  local_desc.connection_role = CONNECTIONROLE_ACTPASS;
>-  ASSERT_TRUE(transport_->SetLocalTransportDescription(
>-      local_desc, SdpType::kOffer, nullptr));
>+  bool rtcp_mux_enabled = true;
>+  jsep_transport_ = CreateJsepTransport2(rtcp_mux_enabled, SrtpMode::kDtlsSrtp);
>+  jsep_transport_->SetLocalCertificate(certificate);
>+
>+  JsepTransportDescription remote_desc =
>+      MakeJsepTransportDescription(rtcp_mux_enabled, kIceUfrag1, kIcePwd1,
>+                                   certificate, CONNECTIONROLE_ACTPASS);
>+  JsepTransportDescription local_desc =
>+      MakeJsepTransportDescription(rtcp_mux_enabled, kIceUfrag2, kIcePwd2,
>+                                   certificate, CONNECTIONROLE_ACTIVE);
>+
>+  local_desc.transport_desc.connection_role = CONNECTIONROLE_ACTPASS;
>+  ASSERT_TRUE(
>+      jsep_transport_
>+          ->SetLocalJsepTransportDescription(local_desc, SdpType::kOffer)
>+          .ok());
>   // Use CONNECTIONROLE_NONE to simulate legacy endpoint.
>-  remote_desc.connection_role = CONNECTIONROLE_NONE;
>-  ASSERT_TRUE(transport_->SetRemoteTransportDescription(
>-      remote_desc, SdpType::kAnswer, nullptr));
>+  remote_desc.transport_desc.connection_role = CONNECTIONROLE_NONE;
>+  ASSERT_TRUE(
>+      jsep_transport_
>+          ->SetRemoteJsepTransportDescription(remote_desc, SdpType::kAnswer)
>+          .ok());
> 
>-  rtc::Optional<rtc::SSLRole> role = transport_->GetSslRole();
>+  absl::optional<rtc::SSLRole> role = jsep_transport_->GetDtlsRole();
>   ASSERT_TRUE(role);
>   // Since legacy answer ommitted setup atribute, and we offered actpass, we
>   // should act as passive (server).
>   EXPECT_EQ(rtc::SSL_SERVER, *role);
> }
> 
>+// Tests that when the RTCP mux is successfully negotiated, the RTCP transport
>+// will be destroyed and the SignalRtpMuxActive will be fired.
>+TEST_F(JsepTransport2Test, RtcpMuxNegotiation) {
>+  jsep_transport_ =
>+      CreateJsepTransport2(/*rtcp_mux_enabled=*/false, SrtpMode::kDtlsSrtp);
>+  JsepTransportDescription local_desc;
>+  local_desc.rtcp_mux_enabled = true;
>+  EXPECT_NE(nullptr, jsep_transport_->rtcp_dtls_transport());
>+  EXPECT_FALSE(signal_rtcp_mux_active_received_);
>+
>+  // The remote side supports RTCP-mux.
>+  JsepTransportDescription remote_desc;
>+  remote_desc.rtcp_mux_enabled = true;
>+  ASSERT_TRUE(
>+      jsep_transport_
>+          ->SetLocalJsepTransportDescription(local_desc, SdpType::kOffer)
>+          .ok());
>+  ASSERT_TRUE(
>+      jsep_transport_
>+          ->SetRemoteJsepTransportDescription(remote_desc, SdpType::kAnswer)
>+          .ok());
>+
>+  EXPECT_EQ(nullptr, jsep_transport_->rtcp_dtls_transport());
>+  EXPECT_TRUE(signal_rtcp_mux_active_received_);
>+
>+  // The remote side doesn't support RTCP-mux.
>+  jsep_transport_ =
>+      CreateJsepTransport2(/*rtcp_mux_enabled=*/false, SrtpMode::kDtlsSrtp);
>+  signal_rtcp_mux_active_received_ = false;
>+  remote_desc.rtcp_mux_enabled = false;
>+  ASSERT_TRUE(
>+      jsep_transport_
>+          ->SetLocalJsepTransportDescription(local_desc, SdpType::kOffer)
>+          .ok());
>+  ASSERT_TRUE(
>+      jsep_transport_
>+          ->SetRemoteJsepTransportDescription(remote_desc, SdpType::kAnswer)
>+          .ok());
>+
>+  EXPECT_NE(nullptr, jsep_transport_->rtcp_dtls_transport());
>+  EXPECT_FALSE(signal_rtcp_mux_active_received_);
>+}
>+
>+TEST_F(JsepTransport2Test, SdesNegotiation) {
>+  jsep_transport_ =
>+      CreateJsepTransport2(/*rtcp_mux_enabled=*/true, SrtpMode::kSdes);
>+  ASSERT_TRUE(sdes_transport_);
>+  EXPECT_FALSE(sdes_transport_->IsSrtpActive());
>+
>+  JsepTransportDescription offer_desc;
>+  offer_desc.cryptos.push_back(cricket::CryptoParams(
>+      1, rtc::CS_AES_CM_128_HMAC_SHA1_32,
>+      "inline:" + rtc::CreateRandomString(40), std::string()));
>+  ASSERT_TRUE(
>+      jsep_transport_
>+          ->SetLocalJsepTransportDescription(offer_desc, SdpType::kOffer)
>+          .ok());
>+
>+  JsepTransportDescription answer_desc;
>+  answer_desc.cryptos.push_back(cricket::CryptoParams(
>+      1, rtc::CS_AES_CM_128_HMAC_SHA1_32,
>+      "inline:" + rtc::CreateRandomString(40), std::string()));
>+  ASSERT_TRUE(
>+      jsep_transport_
>+          ->SetRemoteJsepTransportDescription(answer_desc, SdpType::kAnswer)
>+          .ok());
>+  EXPECT_TRUE(sdes_transport_->IsSrtpActive());
>+}
>+
>+TEST_F(JsepTransport2Test, SdesNegotiationWithEmptyCryptosInAnswer) {
>+  jsep_transport_ =
>+      CreateJsepTransport2(/*rtcp_mux_enabled=*/true, SrtpMode::kSdes);
>+  ASSERT_TRUE(sdes_transport_);
>+  EXPECT_FALSE(sdes_transport_->IsSrtpActive());
>+
>+  JsepTransportDescription offer_desc;
>+  offer_desc.cryptos.push_back(cricket::CryptoParams(
>+      1, rtc::CS_AES_CM_128_HMAC_SHA1_32,
>+      "inline:" + rtc::CreateRandomString(40), std::string()));
>+  ASSERT_TRUE(
>+      jsep_transport_
>+          ->SetLocalJsepTransportDescription(offer_desc, SdpType::kOffer)
>+          .ok());
>+
>+  JsepTransportDescription answer_desc;
>+  ASSERT_TRUE(
>+      jsep_transport_
>+          ->SetRemoteJsepTransportDescription(answer_desc, SdpType::kAnswer)
>+          .ok());
>+  // SRTP is not active because the crypto parameter is answer is empty.
>+  EXPECT_FALSE(sdes_transport_->IsSrtpActive());
>+}
>+
>+TEST_F(JsepTransport2Test, SdesNegotiationWithMismatchedCryptos) {
>+  jsep_transport_ =
>+      CreateJsepTransport2(/*rtcp_mux_enabled=*/true, SrtpMode::kSdes);
>+  ASSERT_TRUE(sdes_transport_);
>+  EXPECT_FALSE(sdes_transport_->IsSrtpActive());
>+
>+  JsepTransportDescription offer_desc;
>+  offer_desc.cryptos.push_back(cricket::CryptoParams(
>+      1, rtc::CS_AES_CM_128_HMAC_SHA1_32,
>+      "inline:" + rtc::CreateRandomString(40), std::string()));
>+  ASSERT_TRUE(
>+      jsep_transport_
>+          ->SetLocalJsepTransportDescription(offer_desc, SdpType::kOffer)
>+          .ok());
>+
>+  JsepTransportDescription answer_desc;
>+  answer_desc.cryptos.push_back(cricket::CryptoParams(
>+      1, rtc::CS_AES_CM_128_HMAC_SHA1_80,
>+      "inline:" + rtc::CreateRandomString(40), std::string()));
>+  // Expected to fail because the crypto parameters don't match.
>+  ASSERT_FALSE(
>+      jsep_transport_
>+          ->SetRemoteJsepTransportDescription(answer_desc, SdpType::kAnswer)
>+          .ok());
>+}
>+
>+// Tests that the remote candidates can be added to the transports after both
>+// local and remote descriptions are set.
>+TEST_F(JsepTransport2Test, AddRemoteCandidates) {
>+  jsep_transport_ =
>+      CreateJsepTransport2(/*rtcp_mux_enabled=*/true, SrtpMode::kDtlsSrtp);
>+  auto fake_ice_transport = static_cast<FakeIceTransport*>(
>+      jsep_transport_->rtp_dtls_transport()->ice_transport());
>+
>+  Candidates candidates;
>+  candidates.push_back(CreateCandidate(/*COMPONENT_RTP*/ 1));
>+  candidates.push_back(CreateCandidate(/*COMPONENT_RTP*/ 1));
>+
>+  JsepTransportDescription desc;
>+  ASSERT_TRUE(
>+      jsep_transport_->SetLocalJsepTransportDescription(desc, SdpType::kOffer)
>+          .ok());
>+  // Expected to fail because the remote description is unset.
>+  EXPECT_FALSE(jsep_transport_->AddRemoteCandidates(candidates).ok());
>+
>+  ASSERT_TRUE(
>+      jsep_transport_->SetRemoteJsepTransportDescription(desc, SdpType::kAnswer)
>+          .ok());
>+  EXPECT_EQ(0u, fake_ice_transport->remote_candidates().size());
>+  EXPECT_TRUE(jsep_transport_->AddRemoteCandidates(candidates).ok());
>+  EXPECT_EQ(candidates.size(), fake_ice_transport->remote_candidates().size());
>+}
>+
>+enum class Scenario {
>+  kSdes,
>+  kDtlsBeforeCallerSendOffer,
>+  kDtlsBeforeCallerSetAnswer,
>+  kDtlsAfterCallerSetAnswer,
>+};
>+
>+class JsepTransport2HeaderExtensionTest
>+    : public JsepTransport2Test,
>+      public ::testing::WithParamInterface<std::tuple<Scenario, bool>> {
>+ protected:
>+  JsepTransport2HeaderExtensionTest() {}
>+
>+  void CreateJsepTransportPair(SrtpMode mode) {
>+    jsep_transport1_ = CreateJsepTransport2(/*rtcp_mux_enabled=*/true, mode);
>+    jsep_transport2_ = CreateJsepTransport2(/*rtcp_mux_enabled=*/true, mode);
>+
>+    auto fake_dtls1 =
>+        static_cast<FakeDtlsTransport*>(jsep_transport1_->rtp_dtls_transport());
>+    auto fake_dtls2 =
>+        static_cast<FakeDtlsTransport*>(jsep_transport2_->rtp_dtls_transport());
>+
>+    fake_dtls1->fake_ice_transport()->SignalReadPacket.connect(
>+        this, &JsepTransport2HeaderExtensionTest::OnReadPacket1);
>+    fake_dtls2->fake_ice_transport()->SignalReadPacket.connect(
>+        this, &JsepTransport2HeaderExtensionTest::OnReadPacket2);
>+
>+    if (mode == SrtpMode::kDtlsSrtp) {
>+      auto cert1 =
>+          rtc::RTCCertificate::Create(std::unique_ptr<rtc::SSLIdentity>(
>+              rtc::SSLIdentity::Generate("session1", rtc::KT_DEFAULT)));
>+      jsep_transport1_->rtp_dtls_transport()->SetLocalCertificate(cert1);
>+      auto cert2 =
>+          rtc::RTCCertificate::Create(std::unique_ptr<rtc::SSLIdentity>(
>+              rtc::SSLIdentity::Generate("session1", rtc::KT_DEFAULT)));
>+      jsep_transport2_->rtp_dtls_transport()->SetLocalCertificate(cert2);
>+    }
>+  }
>+
>+  void OnReadPacket1(rtc::PacketTransportInternal* transport,
>+                     const char* data,
>+                     size_t size,
>+                     const rtc::PacketTime& time,
>+                     int flags) {
>+    RTC_LOG(LS_INFO) << "JsepTransport 1 Received a packet.";
>+    CompareHeaderExtensions(
>+        reinterpret_cast<const char*>(kPcmuFrameWithExtensions),
>+        sizeof(kPcmuFrameWithExtensions), data, size, recv_encrypted_headers1_,
>+        false);
>+    received_packet_count_++;
>+  }
>+
>+  void OnReadPacket2(rtc::PacketTransportInternal* transport,
>+                     const char* data,
>+                     size_t size,
>+                     const rtc::PacketTime& time,
>+                     int flags) {
>+    RTC_LOG(LS_INFO) << "JsepTransport 2 Received a packet.";
>+    CompareHeaderExtensions(
>+        reinterpret_cast<const char*>(kPcmuFrameWithExtensions),
>+        sizeof(kPcmuFrameWithExtensions), data, size, recv_encrypted_headers2_,
>+        false);
>+    received_packet_count_++;
>+  }
>+
>+  void ConnectTransport() {
>+    auto rtp_dtls_transport1 =
>+        static_cast<FakeDtlsTransport*>(jsep_transport1_->rtp_dtls_transport());
>+    auto rtp_dtls_transport2 =
>+        static_cast<FakeDtlsTransport*>(jsep_transport2_->rtp_dtls_transport());
>+    rtp_dtls_transport1->SetDestination(rtp_dtls_transport2);
>+  }
>+
>+  int GetRtpAuthLen() {
>+    bool use_gcm = std::get<1>(GetParam());
>+    if (use_gcm) {
>+      return 16;
>+    }
>+    return 10;
>+  }
>+
>+  void TestSendRecvPacketWithEncryptedHeaderExtension() {
>+    TestOneWaySendRecvPacketWithEncryptedHeaderExtension(
>+        jsep_transport1_.get());
>+    TestOneWaySendRecvPacketWithEncryptedHeaderExtension(
>+        jsep_transport2_.get());
>+  }
>+
>+  void TestOneWaySendRecvPacketWithEncryptedHeaderExtension(
>+      JsepTransport* sender_transport) {
>+    size_t rtp_len = sizeof(kPcmuFrameWithExtensions);
>+    size_t packet_size = rtp_len + GetRtpAuthLen();
>+    rtc::Buffer rtp_packet_buffer(packet_size);
>+    char* rtp_packet_data = rtp_packet_buffer.data<char>();
>+    memcpy(rtp_packet_data, kPcmuFrameWithExtensions, rtp_len);
>+    // In order to be able to run this test function multiple times we can not
>+    // use the same sequence number twice. Increase the sequence number by one.
>+    rtc::SetBE16(reinterpret_cast<uint8_t*>(rtp_packet_data) + 2,
>+                 ++sequence_number_);
>+    rtc::CopyOnWriteBuffer rtp_packet(rtp_packet_data, rtp_len, packet_size);
>+
>+    int packet_count_before = received_packet_count_;
>+    rtc::PacketOptions options;
>+    // Send a packet and verify that the packet can be successfully received and
>+    // decrypted.
>+    ASSERT_TRUE(sender_transport->rtp_transport()->SendRtpPacket(
>+        &rtp_packet, options, cricket::PF_SRTP_BYPASS));
>+    EXPECT_EQ(packet_count_before + 1, received_packet_count_);
>+  }
>+
>+  int sequence_number_ = 0;
>+  int received_packet_count_ = 0;
>+  std::unique_ptr<JsepTransport> jsep_transport1_;
>+  std::unique_ptr<JsepTransport> jsep_transport2_;
>+  std::vector<int> recv_encrypted_headers1_;
>+  std::vector<int> recv_encrypted_headers2_;
>+};
>+
>+// Test that the encrypted header extension works and can be changed in
>+// different scenarios.
>+TEST_P(JsepTransport2HeaderExtensionTest, EncryptedHeaderExtensionNegotiation) {
>+  Scenario scenario = std::get<0>(GetParam());
>+  bool use_gcm = std::get<1>(GetParam());
>+  SrtpMode mode = SrtpMode ::kDtlsSrtp;
>+  if (scenario == Scenario::kSdes) {
>+    mode = SrtpMode::kSdes;
>+  }
>+  CreateJsepTransportPair(mode);
>+  recv_encrypted_headers1_.push_back(kHeaderExtensionIDs[0]);
>+  recv_encrypted_headers2_.push_back(kHeaderExtensionIDs[1]);
>+
>+  cricket::CryptoParams sdes_param(1, rtc::CS_AES_CM_128_HMAC_SHA1_80,
>+                                   "inline:" + rtc::CreateRandomString(40),
>+                                   std::string());
>+  if (use_gcm) {
>+    auto fake_dtls1 =
>+        static_cast<FakeDtlsTransport*>(jsep_transport1_->rtp_dtls_transport());
>+    auto fake_dtls2 =
>+        static_cast<FakeDtlsTransport*>(jsep_transport2_->rtp_dtls_transport());
>+
>+    fake_dtls1->SetSrtpCryptoSuite(rtc::SRTP_AEAD_AES_256_GCM);
>+    fake_dtls2->SetSrtpCryptoSuite(rtc::SRTP_AEAD_AES_256_GCM);
>+  }
>+
>+  if (scenario == Scenario::kDtlsBeforeCallerSendOffer) {
>+    ConnectTransport();
>+  }
>+
>+  JsepTransportDescription offer_desc;
>+  offer_desc.encrypted_header_extension_ids = recv_encrypted_headers1_;
>+  if (scenario == Scenario::kSdes) {
>+    offer_desc.cryptos.push_back(sdes_param);
>+  }
>+  ASSERT_TRUE(
>+      jsep_transport1_
>+          ->SetLocalJsepTransportDescription(offer_desc, SdpType::kOffer)
>+          .ok());
>+  ASSERT_TRUE(
>+      jsep_transport2_
>+          ->SetRemoteJsepTransportDescription(offer_desc, SdpType::kOffer)
>+          .ok());
>+
>+  JsepTransportDescription answer_desc;
>+  answer_desc.encrypted_header_extension_ids = recv_encrypted_headers2_;
>+  if (scenario == Scenario::kSdes) {
>+    answer_desc.cryptos.push_back(sdes_param);
>+  }
>+  ASSERT_TRUE(
>+      jsep_transport2_
>+          ->SetLocalJsepTransportDescription(answer_desc, SdpType::kAnswer)
>+          .ok());
>+
>+  if (scenario == Scenario::kDtlsBeforeCallerSetAnswer) {
>+    ConnectTransport();
>+    // Sending packet from transport2 to transport1 should work when they are
>+    // partially configured.
>+    TestOneWaySendRecvPacketWithEncryptedHeaderExtension(
>+        /*sender_transport=*/jsep_transport2_.get());
>+  }
>+
>+  ASSERT_TRUE(
>+      jsep_transport1_
>+          ->SetRemoteJsepTransportDescription(answer_desc, SdpType::kAnswer)
>+          .ok());
>+
>+  if (scenario == Scenario::kDtlsAfterCallerSetAnswer ||
>+      scenario == Scenario::kSdes) {
>+    ConnectTransport();
>+  }
>+  EXPECT_TRUE(jsep_transport1_->rtp_transport()->IsSrtpActive());
>+  EXPECT_TRUE(jsep_transport2_->rtp_transport()->IsSrtpActive());
>+  TestSendRecvPacketWithEncryptedHeaderExtension();
>+
>+  // Change the encrypted header extension in a new offer/answer exchange.
>+  recv_encrypted_headers1_.clear();
>+  recv_encrypted_headers2_.clear();
>+  recv_encrypted_headers1_.push_back(kHeaderExtensionIDs[1]);
>+  recv_encrypted_headers2_.push_back(kHeaderExtensionIDs[0]);
>+  offer_desc.encrypted_header_extension_ids = recv_encrypted_headers1_;
>+  answer_desc.encrypted_header_extension_ids = recv_encrypted_headers2_;
>+  ASSERT_TRUE(
>+      jsep_transport1_
>+          ->SetLocalJsepTransportDescription(offer_desc, SdpType::kOffer)
>+          .ok());
>+  ASSERT_TRUE(
>+      jsep_transport2_
>+          ->SetRemoteJsepTransportDescription(offer_desc, SdpType::kOffer)
>+          .ok());
>+  ASSERT_TRUE(
>+      jsep_transport2_
>+          ->SetLocalJsepTransportDescription(answer_desc, SdpType::kAnswer)
>+          .ok());
>+  ASSERT_TRUE(
>+      jsep_transport1_
>+          ->SetRemoteJsepTransportDescription(answer_desc, SdpType::kAnswer)
>+          .ok());
>+  EXPECT_TRUE(jsep_transport1_->rtp_transport()->IsSrtpActive());
>+  EXPECT_TRUE(jsep_transport2_->rtp_transport()->IsSrtpActive());
>+  TestSendRecvPacketWithEncryptedHeaderExtension();
>+}
>+
>+INSTANTIATE_TEST_CASE_P(
>+    JsepTransport2Test,
>+    JsepTransport2HeaderExtensionTest,
>+    ::testing::Values(
>+        std::make_tuple(Scenario::kSdes, false),
>+        std::make_tuple(Scenario::kDtlsBeforeCallerSendOffer, true),
>+        std::make_tuple(Scenario::kDtlsBeforeCallerSetAnswer, true),
>+        std::make_tuple(Scenario::kDtlsAfterCallerSetAnswer, true),
>+        std::make_tuple(Scenario::kDtlsBeforeCallerSendOffer, false),
>+        std::make_tuple(Scenario::kDtlsBeforeCallerSetAnswer, false),
>+        std::make_tuple(Scenario::kDtlsAfterCallerSetAnswer, false)));
>+
> }  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jseptransportcontroller.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jseptransportcontroller.cc
>new file mode 100644
>index 00000000000..1375ab0e59a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jseptransportcontroller.cc
>@@ -0,0 +1,1235 @@
>+/*
>+ *  Copyright 2017 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "pc/jseptransportcontroller.h"
>+
>+#include <algorithm>
>+#include <memory>
>+#include <utility>
>+
>+#include "absl/memory/memory.h"
>+#include "p2p/base/port.h"
>+#include "rtc_base/bind.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/thread.h"
>+
>+using webrtc::SdpType;
>+
>+namespace {
>+
>+enum {
>+  MSG_ICECONNECTIONSTATE,
>+  MSG_ICEGATHERINGSTATE,
>+  MSG_ICECANDIDATESGATHERED,
>+};
>+
>+struct CandidatesData : public rtc::MessageData {
>+  CandidatesData(const std::string& transport_name,
>+                 const cricket::Candidates& candidates)
>+      : transport_name(transport_name), candidates(candidates) {}
>+
>+  std::string transport_name;
>+  cricket::Candidates candidates;
>+};
>+
>+webrtc::RTCError VerifyCandidate(const cricket::Candidate& cand) {
>+  // No address zero.
>+  if (cand.address().IsNil() || cand.address().IsAnyIP()) {
>+    return webrtc::RTCError(webrtc::RTCErrorType::INVALID_PARAMETER,
>+                            "candidate has address of zero");
>+  }
>+
>+  // Disallow all ports below 1024, except for 80 and 443 on public addresses.
>+  int port = cand.address().port();
>+  if (cand.protocol() == cricket::TCP_PROTOCOL_NAME &&
>+      (cand.tcptype() == cricket::TCPTYPE_ACTIVE_STR || port == 0)) {
>+    // Expected for active-only candidates per
>+    // http://tools.ietf.org/html/rfc6544#section-4.5 so no error.
>+    // Libjingle clients emit port 0, in "active" mode.
>+    return webrtc::RTCError::OK();
>+  }
>+  if (port < 1024) {
>+    if ((port != 80) && (port != 443)) {
>+      return webrtc::RTCError(
>+          webrtc::RTCErrorType::INVALID_PARAMETER,
>+          "candidate has port below 1024, but not 80 or 443");
>+    }
>+
>+    if (cand.address().IsPrivateIP()) {
>+      return webrtc::RTCError(
>+          webrtc::RTCErrorType::INVALID_PARAMETER,
>+          "candidate has port of 80 or 443 with private IP address");
>+    }
>+  }
>+
>+  return webrtc::RTCError::OK();
>+}
>+
>+webrtc::RTCError VerifyCandidates(const cricket::Candidates& candidates) {
>+  for (const cricket::Candidate& candidate : candidates) {
>+    webrtc::RTCError error = VerifyCandidate(candidate);
>+    if (!error.ok()) {
>+      return error;
>+    }
>+  }
>+  return webrtc::RTCError::OK();
>+}
>+
>+}  // namespace
>+
>+namespace webrtc {
>+
>+JsepTransportController::JsepTransportController(
>+    rtc::Thread* signaling_thread,
>+    rtc::Thread* network_thread,
>+    cricket::PortAllocator* port_allocator,
>+    AsyncResolverFactory* async_resolver_factory,
>+    Config config)
>+    : signaling_thread_(signaling_thread),
>+      network_thread_(network_thread),
>+      port_allocator_(port_allocator),
>+      async_resolver_factory_(async_resolver_factory),
>+      config_(config) {
>+  // The |transport_observer| is assumed to be non-null.
>+  RTC_DCHECK(config_.transport_observer);
>+}
>+
>+JsepTransportController::~JsepTransportController() {
>+  // Channel destructors may try to send packets, so this needs to happen on
>+  // the network thread.
>+  network_thread_->Invoke<void>(
>+      RTC_FROM_HERE,
>+      rtc::Bind(&JsepTransportController::DestroyAllJsepTransports_n, this));
>+}
>+
>+RTCError JsepTransportController::SetLocalDescription(
>+    SdpType type,
>+    const cricket::SessionDescription* description) {
>+  if (!network_thread_->IsCurrent()) {
>+    return network_thread_->Invoke<RTCError>(
>+        RTC_FROM_HERE, [=] { return SetLocalDescription(type, description); });
>+  }
>+
>+  if (!initial_offerer_.has_value()) {
>+    initial_offerer_.emplace(type == SdpType::kOffer);
>+    if (*initial_offerer_) {
>+      SetIceRole_n(cricket::ICEROLE_CONTROLLING);
>+    } else {
>+      SetIceRole_n(cricket::ICEROLE_CONTROLLED);
>+    }
>+  }
>+  return ApplyDescription_n(/*local=*/true, type, description);
>+}
>+
>+RTCError JsepTransportController::SetRemoteDescription(
>+    SdpType type,
>+    const cricket::SessionDescription* description) {
>+  if (!network_thread_->IsCurrent()) {
>+    return network_thread_->Invoke<RTCError>(
>+        RTC_FROM_HERE, [=] { return SetRemoteDescription(type, description); });
>+  }
>+
>+  return ApplyDescription_n(/*local=*/false, type, description);
>+}
>+
>+RtpTransportInternal* JsepTransportController::GetRtpTransport(
>+    const std::string& mid) const {
>+  auto jsep_transport = GetJsepTransportForMid(mid);
>+  if (!jsep_transport) {
>+    return nullptr;
>+  }
>+  return jsep_transport->rtp_transport();
>+}
>+
>+cricket::DtlsTransportInternal* JsepTransportController::GetDtlsTransport(
>+    const std::string& mid) const {
>+  auto jsep_transport = GetJsepTransportForMid(mid);
>+  if (!jsep_transport) {
>+    return nullptr;
>+  }
>+  return jsep_transport->rtp_dtls_transport();
>+}
>+
>+cricket::DtlsTransportInternal* JsepTransportController::GetRtcpDtlsTransport(
>+    const std::string& mid) const {
>+  auto jsep_transport = GetJsepTransportForMid(mid);
>+  if (!jsep_transport) {
>+    return nullptr;
>+  }
>+  return jsep_transport->rtcp_dtls_transport();
>+}
>+
>+void JsepTransportController::SetIceConfig(const cricket::IceConfig& config) {
>+  if (!network_thread_->IsCurrent()) {
>+    network_thread_->Invoke<void>(RTC_FROM_HERE, [&] { SetIceConfig(config); });
>+    return;
>+  }
>+
>+  ice_config_ = config;
>+  for (auto& dtls : GetDtlsTransports()) {
>+    dtls->ice_transport()->SetIceConfig(ice_config_);
>+  }
>+}
>+
>+void JsepTransportController::SetNeedsIceRestartFlag() {
>+  for (auto& kv : jsep_transports_by_name_) {
>+    kv.second->SetNeedsIceRestartFlag();
>+  }
>+}
>+
>+bool JsepTransportController::NeedsIceRestart(
>+    const std::string& transport_name) const {
>+  const cricket::JsepTransport* transport =
>+      GetJsepTransportByName(transport_name);
>+  if (!transport) {
>+    return false;
>+  }
>+  return transport->needs_ice_restart();
>+}
>+
>+absl::optional<rtc::SSLRole> JsepTransportController::GetDtlsRole(
>+    const std::string& mid) const {
>+  if (!network_thread_->IsCurrent()) {
>+    return network_thread_->Invoke<absl::optional<rtc::SSLRole>>(
>+        RTC_FROM_HERE, [&] { return GetDtlsRole(mid); });
>+  }
>+
>+  const cricket::JsepTransport* t = GetJsepTransportForMid(mid);
>+  if (!t) {
>+    return absl::optional<rtc::SSLRole>();
>+  }
>+  return t->GetDtlsRole();
>+}
>+
>+bool JsepTransportController::SetLocalCertificate(
>+    const rtc::scoped_refptr<rtc::RTCCertificate>& certificate) {
>+  if (!network_thread_->IsCurrent()) {
>+    return network_thread_->Invoke<bool>(
>+        RTC_FROM_HERE, [&] { return SetLocalCertificate(certificate); });
>+  }
>+
>+  // Can't change a certificate, or set a null certificate.
>+  if (certificate_ || !certificate) {
>+    return false;
>+  }
>+  certificate_ = certificate;
>+
>+  // Set certificate for JsepTransport, which verifies it matches the
>+  // fingerprint in SDP, and DTLS transport.
>+  // Fallback from DTLS to SDES is not supported.
>+  for (auto& kv : jsep_transports_by_name_) {
>+    kv.second->SetLocalCertificate(certificate_);
>+  }
>+  for (auto& dtls : GetDtlsTransports()) {
>+    bool set_cert_success = dtls->SetLocalCertificate(certificate_);
>+    RTC_DCHECK(set_cert_success);
>+  }
>+  return true;
>+}
>+
>+rtc::scoped_refptr<rtc::RTCCertificate>
>+JsepTransportController::GetLocalCertificate(
>+    const std::string& transport_name) const {
>+  if (!network_thread_->IsCurrent()) {
>+    return network_thread_->Invoke<rtc::scoped_refptr<rtc::RTCCertificate>>(
>+        RTC_FROM_HERE, [&] { return GetLocalCertificate(transport_name); });
>+  }
>+
>+  const cricket::JsepTransport* t = GetJsepTransportByName(transport_name);
>+  if (!t) {
>+    return nullptr;
>+  }
>+  return t->GetLocalCertificate();
>+}
>+
>+std::unique_ptr<rtc::SSLCertChain>
>+JsepTransportController::GetRemoteSSLCertChain(
>+    const std::string& transport_name) const {
>+  if (!network_thread_->IsCurrent()) {
>+    return network_thread_->Invoke<std::unique_ptr<rtc::SSLCertChain>>(
>+        RTC_FROM_HERE, [&] { return GetRemoteSSLCertChain(transport_name); });
>+  }
>+
>+  // Get the certificate from the RTP transport's DTLS handshake. Should be
>+  // identical to the RTCP transport's, since they were given the same remote
>+  // fingerprint.
>+  auto jsep_transport = GetJsepTransportByName(transport_name);
>+  if (!jsep_transport) {
>+    return nullptr;
>+  }
>+  auto dtls = jsep_transport->rtp_dtls_transport();
>+  if (!dtls) {
>+    return nullptr;
>+  }
>+
>+  return dtls->GetRemoteSSLCertChain();
>+}
>+
>+void JsepTransportController::MaybeStartGathering() {
>+  if (!network_thread_->IsCurrent()) {
>+    network_thread_->Invoke<void>(RTC_FROM_HERE,
>+                                  [&] { MaybeStartGathering(); });
>+    return;
>+  }
>+
>+  for (auto& dtls : GetDtlsTransports()) {
>+    dtls->ice_transport()->MaybeStartGathering();
>+  }
>+}
>+
>+RTCError JsepTransportController::AddRemoteCandidates(
>+    const std::string& transport_name,
>+    const cricket::Candidates& candidates) {
>+  if (!network_thread_->IsCurrent()) {
>+    return network_thread_->Invoke<RTCError>(RTC_FROM_HERE, [&] {
>+      return AddRemoteCandidates(transport_name, candidates);
>+    });
>+  }
>+
>+  // Verify each candidate before passing down to the transport layer.
>+  RTCError error = VerifyCandidates(candidates);
>+  if (!error.ok()) {
>+    return error;
>+  }
>+  auto jsep_transport = GetJsepTransportByName(transport_name);
>+  if (!jsep_transport) {
>+    RTC_LOG(LS_WARNING) << "Not adding candidate because the JsepTransport "
>+                           "doesn't exist. Ignore it.";
>+    return RTCError::OK();
>+  }
>+  return jsep_transport->AddRemoteCandidates(candidates);
>+}
>+
>+RTCError JsepTransportController::RemoveRemoteCandidates(
>+    const cricket::Candidates& candidates) {
>+  if (!network_thread_->IsCurrent()) {
>+    return network_thread_->Invoke<RTCError>(
>+        RTC_FROM_HERE, [&] { return RemoveRemoteCandidates(candidates); });
>+  }
>+
>+  // Verify each candidate before passing down to the transport layer.
>+  RTCError error = VerifyCandidates(candidates);
>+  if (!error.ok()) {
>+    return error;
>+  }
>+
>+  std::map<std::string, cricket::Candidates> candidates_by_transport_name;
>+  for (const cricket::Candidate& cand : candidates) {
>+    if (!cand.transport_name().empty()) {
>+      candidates_by_transport_name[cand.transport_name()].push_back(cand);
>+    } else {
>+      RTC_LOG(LS_ERROR) << "Not removing candidate because it does not have a "
>+                           "transport name set: "
>+                        << cand.ToString();
>+    }
>+  }
>+
>+  for (const auto& kv : candidates_by_transport_name) {
>+    const std::string& transport_name = kv.first;
>+    const cricket::Candidates& candidates = kv.second;
>+    cricket::JsepTransport* jsep_transport =
>+        GetJsepTransportByName(transport_name);
>+    if (!jsep_transport) {
>+      RTC_LOG(LS_WARNING)
>+          << "Not removing candidate because the JsepTransport doesn't exist.";
>+      continue;
>+    }
>+    for (const cricket::Candidate& candidate : candidates) {
>+      auto dtls = candidate.component() == cricket::ICE_CANDIDATE_COMPONENT_RTP
>+                      ? jsep_transport->rtp_dtls_transport()
>+                      : jsep_transport->rtcp_dtls_transport();
>+      if (dtls) {
>+        dtls->ice_transport()->RemoveRemoteCandidate(candidate);
>+      }
>+    }
>+  }
>+  return RTCError::OK();
>+}
>+
>+bool JsepTransportController::GetStats(const std::string& transport_name,
>+                                       cricket::TransportStats* stats) {
>+  if (!network_thread_->IsCurrent()) {
>+    return network_thread_->Invoke<bool>(
>+        RTC_FROM_HERE, [=] { return GetStats(transport_name, stats); });
>+  }
>+
>+  cricket::JsepTransport* transport = GetJsepTransportByName(transport_name);
>+  if (!transport) {
>+    return false;
>+  }
>+  return transport->GetStats(stats);
>+}
>+
>+void JsepTransportController::SetActiveResetSrtpParams(
>+    bool active_reset_srtp_params) {
>+  if (!network_thread_->IsCurrent()) {
>+    network_thread_->Invoke<void>(RTC_FROM_HERE, [=] {
>+      SetActiveResetSrtpParams(active_reset_srtp_params);
>+    });
>+    return;
>+  }
>+
>+  RTC_LOG(INFO)
>+      << "Updating the active_reset_srtp_params for JsepTransportController: "
>+      << active_reset_srtp_params;
>+  config_.active_reset_srtp_params = active_reset_srtp_params;
>+  for (auto& kv : jsep_transports_by_name_) {
>+    kv.second->SetActiveResetSrtpParams(active_reset_srtp_params);
>+  }
>+}
>+
>+std::unique_ptr<cricket::DtlsTransportInternal>
>+JsepTransportController::CreateDtlsTransport(const std::string& transport_name,
>+                                             bool rtcp) {
>+  RTC_DCHECK(network_thread_->IsCurrent());
>+  int component = rtcp ? cricket::ICE_CANDIDATE_COMPONENT_RTCP
>+                       : cricket::ICE_CANDIDATE_COMPONENT_RTP;
>+
>+  std::unique_ptr<cricket::DtlsTransportInternal> dtls;
>+  if (config_.external_transport_factory) {
>+    auto ice = config_.external_transport_factory->CreateIceTransport(
>+        transport_name, component);
>+    dtls = config_.external_transport_factory->CreateDtlsTransport(
>+        std::move(ice), config_.crypto_options);
>+  } else {
>+    auto ice = absl::make_unique<cricket::P2PTransportChannel>(
>+        transport_name, component, port_allocator_, async_resolver_factory_,
>+        config_.event_log);
>+    dtls = absl::make_unique<cricket::DtlsTransport>(std::move(ice),
>+                                                     config_.crypto_options);
>+  }
>+
>+  RTC_DCHECK(dtls);
>+  dtls->SetSslMaxProtocolVersion(config_.ssl_max_version);
>+  dtls->ice_transport()->SetIceRole(ice_role_);
>+  dtls->ice_transport()->SetIceTiebreaker(ice_tiebreaker_);
>+  dtls->ice_transport()->SetIceConfig(ice_config_);
>+  if (certificate_) {
>+    bool set_cert_success = dtls->SetLocalCertificate(certificate_);
>+    RTC_DCHECK(set_cert_success);
>+  }
>+
>+  // Connect to signals offered by the DTLS and ICE transport.
>+  dtls->SignalWritableState.connect(
>+      this, &JsepTransportController::OnTransportWritableState_n);
>+  dtls->SignalReceivingState.connect(
>+      this, &JsepTransportController::OnTransportReceivingState_n);
>+  dtls->SignalDtlsHandshakeError.connect(
>+      this, &JsepTransportController::OnDtlsHandshakeError);
>+  dtls->ice_transport()->SignalGatheringState.connect(
>+      this, &JsepTransportController::OnTransportGatheringState_n);
>+  dtls->ice_transport()->SignalCandidateGathered.connect(
>+      this, &JsepTransportController::OnTransportCandidateGathered_n);
>+  dtls->ice_transport()->SignalCandidatesRemoved.connect(
>+      this, &JsepTransportController::OnTransportCandidatesRemoved_n);
>+  dtls->ice_transport()->SignalRoleConflict.connect(
>+      this, &JsepTransportController::OnTransportRoleConflict_n);
>+  dtls->ice_transport()->SignalStateChanged.connect(
>+      this, &JsepTransportController::OnTransportStateChanged_n);
>+  return dtls;
>+}
>+
>+std::unique_ptr<webrtc::RtpTransport>
>+JsepTransportController::CreateUnencryptedRtpTransport(
>+    const std::string& transport_name,
>+    rtc::PacketTransportInternal* rtp_packet_transport,
>+    rtc::PacketTransportInternal* rtcp_packet_transport) {
>+  RTC_DCHECK(network_thread_->IsCurrent());
>+  auto unencrypted_rtp_transport =
>+      absl::make_unique<RtpTransport>(rtcp_packet_transport == nullptr);
>+  unencrypted_rtp_transport->SetRtpPacketTransport(rtp_packet_transport);
>+  if (rtcp_packet_transport) {
>+    unencrypted_rtp_transport->SetRtcpPacketTransport(rtcp_packet_transport);
>+  }
>+  return unencrypted_rtp_transport;
>+}
>+
>+std::unique_ptr<webrtc::SrtpTransport>
>+JsepTransportController::CreateSdesTransport(
>+    const std::string& transport_name,
>+    cricket::DtlsTransportInternal* rtp_dtls_transport,
>+    cricket::DtlsTransportInternal* rtcp_dtls_transport) {
>+  RTC_DCHECK(network_thread_->IsCurrent());
>+  auto srtp_transport =
>+      absl::make_unique<webrtc::SrtpTransport>(rtcp_dtls_transport == nullptr);
>+  RTC_DCHECK(rtp_dtls_transport);
>+  srtp_transport->SetRtpPacketTransport(rtp_dtls_transport);
>+  if (rtcp_dtls_transport) {
>+    srtp_transport->SetRtcpPacketTransport(rtcp_dtls_transport);
>+  }
>+  if (config_.enable_external_auth) {
>+    srtp_transport->EnableExternalAuth();
>+  }
>+  return srtp_transport;
>+}
>+
>+std::unique_ptr<webrtc::DtlsSrtpTransport>
>+JsepTransportController::CreateDtlsSrtpTransport(
>+    const std::string& transport_name,
>+    cricket::DtlsTransportInternal* rtp_dtls_transport,
>+    cricket::DtlsTransportInternal* rtcp_dtls_transport) {
>+  RTC_DCHECK(network_thread_->IsCurrent());
>+  auto dtls_srtp_transport = absl::make_unique<webrtc::DtlsSrtpTransport>(
>+      rtcp_dtls_transport == nullptr);
>+  if (config_.enable_external_auth) {
>+    dtls_srtp_transport->EnableExternalAuth();
>+  }
>+
>+  dtls_srtp_transport->SetDtlsTransports(rtp_dtls_transport,
>+                                         rtcp_dtls_transport);
>+  dtls_srtp_transport->SetActiveResetSrtpParams(
>+      config_.active_reset_srtp_params);
>+  return dtls_srtp_transport;
>+}
>+
>+std::vector<cricket::DtlsTransportInternal*>
>+JsepTransportController::GetDtlsTransports() {
>+  std::vector<cricket::DtlsTransportInternal*> dtls_transports;
>+  for (auto it = jsep_transports_by_name_.begin();
>+       it != jsep_transports_by_name_.end(); ++it) {
>+    auto jsep_transport = it->second.get();
>+    RTC_DCHECK(jsep_transport);
>+    if (jsep_transport->rtp_dtls_transport()) {
>+      dtls_transports.push_back(jsep_transport->rtp_dtls_transport());
>+    }
>+
>+    if (jsep_transport->rtcp_dtls_transport()) {
>+      dtls_transports.push_back(jsep_transport->rtcp_dtls_transport());
>+    }
>+  }
>+  return dtls_transports;
>+}
>+
>+void JsepTransportController::OnMessage(rtc::Message* pmsg) {
>+  RTC_DCHECK(signaling_thread_->IsCurrent());
>+
>+  switch (pmsg->message_id) {
>+    case MSG_ICECONNECTIONSTATE: {
>+      rtc::TypedMessageData<cricket::IceConnectionState>* data =
>+          static_cast<rtc::TypedMessageData<cricket::IceConnectionState>*>(
>+              pmsg->pdata);
>+      SignalIceConnectionState(data->data());
>+      delete data;
>+      break;
>+    }
>+    case MSG_ICEGATHERINGSTATE: {
>+      rtc::TypedMessageData<cricket::IceGatheringState>* data =
>+          static_cast<rtc::TypedMessageData<cricket::IceGatheringState>*>(
>+              pmsg->pdata);
>+      SignalIceGatheringState(data->data());
>+      delete data;
>+      break;
>+    }
>+    case MSG_ICECANDIDATESGATHERED: {
>+      CandidatesData* data = static_cast<CandidatesData*>(pmsg->pdata);
>+      SignalIceCandidatesGathered(data->transport_name, data->candidates);
>+      delete data;
>+      break;
>+    }
>+    default:
>+      RTC_NOTREACHED();
>+  }
>+}
>+
>+RTCError JsepTransportController::ApplyDescription_n(
>+    bool local,
>+    SdpType type,
>+    const cricket::SessionDescription* description) {
>+  RTC_DCHECK(network_thread_->IsCurrent());
>+  RTC_DCHECK(description);
>+
>+  if (local) {
>+    local_desc_ = description;
>+  } else {
>+    remote_desc_ = description;
>+  }
>+
>+  RTCError error;
>+  error = ValidateAndMaybeUpdateBundleGroup(local, type, description);
>+  if (!error.ok()) {
>+    return error;
>+  }
>+
>+  std::vector<int> merged_encrypted_extension_ids;
>+  if (bundle_group_) {
>+    merged_encrypted_extension_ids =
>+        MergeEncryptedHeaderExtensionIdsForBundle(description);
>+  }
>+
>+  for (const cricket::ContentInfo& content_info : description->contents()) {
>+    // Don't create transports for rejected m-lines and bundled m-lines."
>+    if (content_info.rejected ||
>+        (IsBundled(content_info.name) && content_info.name != *bundled_mid())) {
>+      continue;
>+    }
>+    error = MaybeCreateJsepTransport(content_info);
>+    if (!error.ok()) {
>+      return error;
>+    }
>+  }
>+
>+  RTC_DCHECK(description->contents().size() ==
>+             description->transport_infos().size());
>+  for (size_t i = 0; i < description->contents().size(); ++i) {
>+    const cricket::ContentInfo& content_info = description->contents()[i];
>+    const cricket::TransportInfo& transport_info =
>+        description->transport_infos()[i];
>+    if (content_info.rejected) {
>+      HandleRejectedContent(content_info, description);
>+      continue;
>+    }
>+
>+    if (IsBundled(content_info.name) && content_info.name != *bundled_mid()) {
>+      if (!HandleBundledContent(content_info)) {
>+        return RTCError(RTCErrorType::INVALID_PARAMETER,
>+                        "Failed to process the bundled m= section.");
>+      }
>+      continue;
>+    }
>+
>+    error = ValidateContent(content_info);
>+    if (!error.ok()) {
>+      return error;
>+    }
>+
>+    std::vector<int> extension_ids;
>+    if (bundled_mid() && content_info.name == *bundled_mid()) {
>+      extension_ids = merged_encrypted_extension_ids;
>+    } else {
>+      extension_ids = GetEncryptedHeaderExtensionIds(content_info);
>+    }
>+
>+    int rtp_abs_sendtime_extn_id =
>+        GetRtpAbsSendTimeHeaderExtensionId(content_info);
>+
>+    cricket::JsepTransport* transport =
>+        GetJsepTransportForMid(content_info.name);
>+    RTC_DCHECK(transport);
>+
>+    SetIceRole_n(DetermineIceRole(transport, transport_info, type, local));
>+
>+    cricket::JsepTransportDescription jsep_description =
>+        CreateJsepTransportDescription(content_info, transport_info,
>+                                       extension_ids, rtp_abs_sendtime_extn_id);
>+    if (local) {
>+      error =
>+          transport->SetLocalJsepTransportDescription(jsep_description, type);
>+    } else {
>+      error =
>+          transport->SetRemoteJsepTransportDescription(jsep_description, type);
>+    }
>+
>+    if (!error.ok()) {
>+      LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
>+                           "Failed to apply the description for " +
>+                               content_info.name + ": " + error.message());
>+    }
>+  }
>+  return RTCError::OK();
>+}
>+
>+RTCError JsepTransportController::ValidateAndMaybeUpdateBundleGroup(
>+    bool local,
>+    SdpType type,
>+    const cricket::SessionDescription* description) {
>+  RTC_DCHECK(description);
>+  const cricket::ContentGroup* new_bundle_group =
>+      description->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
>+
>+  // The BUNDLE group containing a MID that no m= section has is invalid.
>+  if (new_bundle_group) {
>+    for (auto content_name : new_bundle_group->content_names()) {
>+      if (!description->GetContentByName(content_name)) {
>+        return RTCError(RTCErrorType::INVALID_PARAMETER,
>+                        "The BUNDLE group contains MID:" + content_name +
>+                            " matching no m= section.");
>+      }
>+    }
>+  }
>+
>+  if (type == SdpType::kAnswer) {
>+    const cricket::ContentGroup* offered_bundle_group =
>+        local ? remote_desc_->GetGroupByName(cricket::GROUP_TYPE_BUNDLE)
>+              : local_desc_->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
>+
>+    if (new_bundle_group) {
>+      // The BUNDLE group in answer should be a subset of offered group.
>+      for (auto content_name : new_bundle_group->content_names()) {
>+        if (!offered_bundle_group ||
>+            !offered_bundle_group->HasContentName(content_name)) {
>+          return RTCError(RTCErrorType::INVALID_PARAMETER,
>+                          "The BUNDLE group in answer contains a MID that was "
>+                          "not in the offered group.");
>+        }
>+      }
>+    }
>+
>+    if (bundle_group_) {
>+      for (auto content_name : bundle_group_->content_names()) {
>+        // An answer that removes m= sections from pre-negotiated BUNDLE group
>+        // without rejecting it, is invalid.
>+        if (!new_bundle_group ||
>+            !new_bundle_group->HasContentName(content_name)) {
>+          auto* content_info = description->GetContentByName(content_name);
>+          if (!content_info || !content_info->rejected) {
>+            return RTCError(RTCErrorType::INVALID_PARAMETER,
>+                            "Answer cannot remove m= section  " + content_name +
>+                                " from already-established BUNDLE group.");
>+          }
>+        }
>+      }
>+    }
>+  }
>+
>+  if (config_.bundle_policy ==
>+          PeerConnectionInterface::kBundlePolicyMaxBundle &&
>+      !description->HasGroup(cricket::GROUP_TYPE_BUNDLE)) {
>+    return RTCError(RTCErrorType::INVALID_PARAMETER,
>+                    "max-bundle is used but no bundle group found.");
>+  }
>+
>+  if (ShouldUpdateBundleGroup(type, description)) {
>+    const std::string* new_bundled_mid = new_bundle_group->FirstContentName();
>+    if (bundled_mid() && new_bundled_mid &&
>+        *bundled_mid() != *new_bundled_mid) {
>+      return RTCError(RTCErrorType::UNSUPPORTED_OPERATION,
>+                      "Changing the negotiated BUNDLE-tag is not supported.");
>+    }
>+
>+    bundle_group_ = *new_bundle_group;
>+  }
>+
>+  if (!bundled_mid()) {
>+    return RTCError::OK();
>+  }
>+
>+  auto bundled_content = description->GetContentByName(*bundled_mid());
>+  if (!bundled_content) {
>+    return RTCError(
>+        RTCErrorType::INVALID_PARAMETER,
>+        "An m= section associated with the BUNDLE-tag doesn't exist.");
>+  }
>+
>+  // If the |bundled_content| is rejected, other contents in the bundle group
>+  // should be rejected.
>+  if (bundled_content->rejected) {
>+    for (auto content_name : bundle_group_->content_names()) {
>+      auto other_content = description->GetContentByName(content_name);
>+      if (!other_content->rejected) {
>+        return RTCError(
>+            RTCErrorType::INVALID_PARAMETER,
>+            "The m= section:" + content_name + " should be rejected.");
>+      }
>+    }
>+  }
>+
>+  return RTCError::OK();
>+}
>+
>+RTCError JsepTransportController::ValidateContent(
>+    const cricket::ContentInfo& content_info) {
>+  if (config_.rtcp_mux_policy ==
>+          PeerConnectionInterface::kRtcpMuxPolicyRequire &&
>+      content_info.type == cricket::MediaProtocolType::kRtp &&
>+      !content_info.media_description()->rtcp_mux()) {
>+    return RTCError(RTCErrorType::INVALID_PARAMETER,
>+                    "The m= section:" + content_info.name +
>+                        " is invalid. RTCP-MUX is not "
>+                        "enabled when it is required.");
>+  }
>+  return RTCError::OK();
>+}
>+
>+void JsepTransportController::HandleRejectedContent(
>+    const cricket::ContentInfo& content_info,
>+    const cricket::SessionDescription* description) {
>+  // If the content is rejected, let the
>+  // BaseChannel/SctpTransport change the RtpTransport/DtlsTransport first,
>+  // then destroy the cricket::JsepTransport.
>+  RemoveTransportForMid(content_info.name);
>+  if (content_info.name == bundled_mid()) {
>+    for (auto content_name : bundle_group_->content_names()) {
>+      RemoveTransportForMid(content_name);
>+    }
>+    bundle_group_.reset();
>+  } else if (IsBundled(content_info.name)) {
>+    // Remove the rejected content from the |bundle_group_|.
>+    bundle_group_->RemoveContentName(content_info.name);
>+    // Reset the bundle group if nothing left.
>+    if (!bundle_group_->FirstContentName()) {
>+      bundle_group_.reset();
>+    }
>+  }
>+  MaybeDestroyJsepTransport(content_info.name);
>+}
>+
>+bool JsepTransportController::HandleBundledContent(
>+    const cricket::ContentInfo& content_info) {
>+  auto jsep_transport = GetJsepTransportByName(*bundled_mid());
>+  RTC_DCHECK(jsep_transport);
>+  // If the content is bundled, let the
>+  // BaseChannel/SctpTransport change the RtpTransport/DtlsTransport first,
>+  // then destroy the cricket::JsepTransport.
>+  if (SetTransportForMid(content_info.name, jsep_transport)) {
>+    MaybeDestroyJsepTransport(content_info.name);
>+    return true;
>+  }
>+  return false;
>+}
>+
>+bool JsepTransportController::SetTransportForMid(
>+    const std::string& mid,
>+    cricket::JsepTransport* jsep_transport) {
>+  RTC_DCHECK(jsep_transport);
>+  if (mid_to_transport_[mid] == jsep_transport) {
>+    return true;
>+  }
>+
>+  mid_to_transport_[mid] = jsep_transport;
>+  return config_.transport_observer->OnTransportChanged(
>+      mid, jsep_transport->rtp_transport(),
>+      jsep_transport->rtp_dtls_transport());
>+}
>+
>+void JsepTransportController::RemoveTransportForMid(const std::string& mid) {
>+  bool ret =
>+      config_.transport_observer->OnTransportChanged(mid, nullptr, nullptr);
>+  // Calling OnTransportChanged with nullptr should always succeed, since it is
>+  // only expected to fail when adding media to a transport (not removing).
>+  RTC_DCHECK(ret);
>+  mid_to_transport_.erase(mid);
>+}
>+
>+cricket::JsepTransportDescription
>+JsepTransportController::CreateJsepTransportDescription(
>+    cricket::ContentInfo content_info,
>+    cricket::TransportInfo transport_info,
>+    const std::vector<int>& encrypted_extension_ids,
>+    int rtp_abs_sendtime_extn_id) {
>+  const cricket::MediaContentDescription* content_desc =
>+      static_cast<const cricket::MediaContentDescription*>(
>+          content_info.description);
>+  RTC_DCHECK(content_desc);
>+  bool rtcp_mux_enabled = content_info.type == cricket::MediaProtocolType::kSctp
>+                              ? true
>+                              : content_desc->rtcp_mux();
>+
>+  return cricket::JsepTransportDescription(
>+      rtcp_mux_enabled, content_desc->cryptos(), encrypted_extension_ids,
>+      rtp_abs_sendtime_extn_id, transport_info.description);
>+}
>+
>+bool JsepTransportController::ShouldUpdateBundleGroup(
>+    SdpType type,
>+    const cricket::SessionDescription* description) {
>+  if (config_.bundle_policy ==
>+      PeerConnectionInterface::kBundlePolicyMaxBundle) {
>+    return true;
>+  }
>+
>+  if (type != SdpType::kAnswer) {
>+    return false;
>+  }
>+
>+  RTC_DCHECK(local_desc_ && remote_desc_);
>+  const cricket::ContentGroup* local_bundle =
>+      local_desc_->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
>+  const cricket::ContentGroup* remote_bundle =
>+      remote_desc_->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
>+  return local_bundle && remote_bundle;
>+}
>+
>+std::vector<int> JsepTransportController::GetEncryptedHeaderExtensionIds(
>+    const cricket::ContentInfo& content_info) {
>+  const cricket::MediaContentDescription* content_desc =
>+      static_cast<const cricket::MediaContentDescription*>(
>+          content_info.description);
>+
>+  if (!config_.crypto_options.enable_encrypted_rtp_header_extensions) {
>+    return std::vector<int>();
>+  }
>+
>+  std::vector<int> encrypted_header_extension_ids;
>+  for (auto extension : content_desc->rtp_header_extensions()) {
>+    if (!extension.encrypt) {
>+      continue;
>+    }
>+    auto it = std::find(encrypted_header_extension_ids.begin(),
>+                        encrypted_header_extension_ids.end(), extension.id);
>+    if (it == encrypted_header_extension_ids.end()) {
>+      encrypted_header_extension_ids.push_back(extension.id);
>+    }
>+  }
>+  return encrypted_header_extension_ids;
>+}
>+
>+std::vector<int>
>+JsepTransportController::MergeEncryptedHeaderExtensionIdsForBundle(
>+    const cricket::SessionDescription* description) {
>+  RTC_DCHECK(description);
>+  RTC_DCHECK(bundle_group_);
>+
>+  std::vector<int> merged_ids;
>+  // Union the encrypted header IDs in the group when bundle is enabled.
>+  for (const cricket::ContentInfo& content_info : description->contents()) {
>+    if (bundle_group_->HasContentName(content_info.name)) {
>+      std::vector<int> extension_ids =
>+          GetEncryptedHeaderExtensionIds(content_info);
>+      for (int id : extension_ids) {
>+        auto it = std::find(merged_ids.begin(), merged_ids.end(), id);
>+        if (it == merged_ids.end()) {
>+          merged_ids.push_back(id);
>+        }
>+      }
>+    }
>+  }
>+  return merged_ids;
>+}
>+
>+int JsepTransportController::GetRtpAbsSendTimeHeaderExtensionId(
>+    const cricket::ContentInfo& content_info) {
>+  if (!config_.enable_external_auth) {
>+    return -1;
>+  }
>+
>+  const cricket::MediaContentDescription* content_desc =
>+      static_cast<const cricket::MediaContentDescription*>(
>+          content_info.description);
>+
>+  const webrtc::RtpExtension* send_time_extension =
>+      webrtc::RtpExtension::FindHeaderExtensionByUri(
>+          content_desc->rtp_header_extensions(),
>+          webrtc::RtpExtension::kAbsSendTimeUri);
>+  return send_time_extension ? send_time_extension->id : -1;
>+}
>+
>+const cricket::JsepTransport* JsepTransportController::GetJsepTransportForMid(
>+    const std::string& mid) const {
>+  auto it = mid_to_transport_.find(mid);
>+  return it == mid_to_transport_.end() ? nullptr : it->second;
>+}
>+
>+cricket::JsepTransport* JsepTransportController::GetJsepTransportForMid(
>+    const std::string& mid) {
>+  auto it = mid_to_transport_.find(mid);
>+  return it == mid_to_transport_.end() ? nullptr : it->second;
>+}
>+
>+const cricket::JsepTransport* JsepTransportController::GetJsepTransportByName(
>+    const std::string& transport_name) const {
>+  auto it = jsep_transports_by_name_.find(transport_name);
>+  return (it == jsep_transports_by_name_.end()) ? nullptr : it->second.get();
>+}
>+
>+cricket::JsepTransport* JsepTransportController::GetJsepTransportByName(
>+    const std::string& transport_name) {
>+  auto it = jsep_transports_by_name_.find(transport_name);
>+  return (it == jsep_transports_by_name_.end()) ? nullptr : it->second.get();
>+}
>+
>+RTCError JsepTransportController::MaybeCreateJsepTransport(
>+    const cricket::ContentInfo& content_info) {
>+  RTC_DCHECK(network_thread_->IsCurrent());
>+  cricket::JsepTransport* transport = GetJsepTransportByName(content_info.name);
>+  if (transport) {
>+    return RTCError::OK();
>+  }
>+
>+  const cricket::MediaContentDescription* content_desc =
>+      static_cast<const cricket::MediaContentDescription*>(
>+          content_info.description);
>+  if (certificate_ && !content_desc->cryptos().empty()) {
>+    return RTCError(RTCErrorType::INVALID_PARAMETER,
>+                    "SDES and DTLS-SRTP cannot be enabled at the same time.");
>+  }
>+
>+  std::unique_ptr<cricket::DtlsTransportInternal> rtp_dtls_transport =
>+      CreateDtlsTransport(content_info.name, /*rtcp =*/false);
>+  std::unique_ptr<cricket::DtlsTransportInternal> rtcp_dtls_transport;
>+  if (config_.rtcp_mux_policy !=
>+          PeerConnectionInterface::kRtcpMuxPolicyRequire &&
>+      content_info.type == cricket::MediaProtocolType::kRtp) {
>+    rtcp_dtls_transport =
>+        CreateDtlsTransport(content_info.name, /*rtcp =*/true);
>+  }
>+
>+  std::unique_ptr<RtpTransport> unencrypted_rtp_transport;
>+  std::unique_ptr<SrtpTransport> sdes_transport;
>+  std::unique_ptr<DtlsSrtpTransport> dtls_srtp_transport;
>+  if (config_.disable_encryption) {
>+    unencrypted_rtp_transport = CreateUnencryptedRtpTransport(
>+        content_info.name, rtp_dtls_transport.get(), rtcp_dtls_transport.get());
>+  } else if (!content_desc->cryptos().empty()) {
>+    sdes_transport = CreateSdesTransport(
>+        content_info.name, rtp_dtls_transport.get(), rtcp_dtls_transport.get());
>+  } else {
>+    dtls_srtp_transport = CreateDtlsSrtpTransport(
>+        content_info.name, rtp_dtls_transport.get(), rtcp_dtls_transport.get());
>+  }
>+
>+  std::unique_ptr<cricket::JsepTransport> jsep_transport =
>+      absl::make_unique<cricket::JsepTransport>(
>+          content_info.name, certificate_, std::move(unencrypted_rtp_transport),
>+          std::move(sdes_transport), std::move(dtls_srtp_transport),
>+          std::move(rtp_dtls_transport), std::move(rtcp_dtls_transport));
>+  jsep_transport->SignalRtcpMuxActive.connect(
>+      this, &JsepTransportController::UpdateAggregateStates_n);
>+  SetTransportForMid(content_info.name, jsep_transport.get());
>+
>+  jsep_transports_by_name_[content_info.name] = std::move(jsep_transport);
>+  UpdateAggregateStates_n();
>+  return RTCError::OK();
>+}
>+
>+void JsepTransportController::MaybeDestroyJsepTransport(
>+    const std::string& mid) {
>+  auto jsep_transport = GetJsepTransportByName(mid);
>+  if (!jsep_transport) {
>+    return;
>+  }
>+
>+  // Don't destroy the JsepTransport if there are still media sections referring
>+  // to it.
>+  for (const auto& kv : mid_to_transport_) {
>+    if (kv.second == jsep_transport) {
>+      return;
>+    }
>+  }
>+  jsep_transports_by_name_.erase(mid);
>+  UpdateAggregateStates_n();
>+}
>+
>+void JsepTransportController::DestroyAllJsepTransports_n() {
>+  RTC_DCHECK(network_thread_->IsCurrent());
>+  jsep_transports_by_name_.clear();
>+}
>+
>+void JsepTransportController::SetIceRole_n(cricket::IceRole ice_role) {
>+  RTC_DCHECK(network_thread_->IsCurrent());
>+
>+  ice_role_ = ice_role;
>+  for (auto& dtls : GetDtlsTransports()) {
>+    dtls->ice_transport()->SetIceRole(ice_role_);
>+  }
>+}
>+
>+cricket::IceRole JsepTransportController::DetermineIceRole(
>+    cricket::JsepTransport* jsep_transport,
>+    const cricket::TransportInfo& transport_info,
>+    SdpType type,
>+    bool local) {
>+  cricket::IceRole ice_role = ice_role_;
>+  auto tdesc = transport_info.description;
>+  if (local) {
>+    // The initial offer side may use ICE Lite, in which case, per RFC5245
>+    // Section 5.1.1, the answer side should take the controlling role if it is
>+    // in the full ICE mode.
>+    //
>+    // When both sides use ICE Lite, the initial offer side must take the
>+    // controlling role, and this is the default logic implemented in
>+    // SetLocalDescription in JsepTransportController.
>+    if (jsep_transport->remote_description() &&
>+        jsep_transport->remote_description()->transport_desc.ice_mode ==
>+            cricket::ICEMODE_LITE &&
>+        ice_role_ == cricket::ICEROLE_CONTROLLED &&
>+        tdesc.ice_mode == cricket::ICEMODE_FULL) {
>+      ice_role = cricket::ICEROLE_CONTROLLING;
>+    }
>+
>+    // Older versions of Chrome expect the ICE role to be re-determined when an
>+    // ICE restart occurs, and also don't perform conflict resolution correctly,
>+    // so for now we can't safely stop doing this, unless the application opts
>+    // in by setting |config_.redetermine_role_on_ice_restart_| to false. See:
>+    // https://bugs.chromium.org/p/chromium/issues/detail?id=628676
>+    // TODO(deadbeef): Remove this when these old versions of Chrome reach a low
>+    // enough population.
>+    if (config_.redetermine_role_on_ice_restart &&
>+        jsep_transport->local_description() &&
>+        cricket::IceCredentialsChanged(
>+            jsep_transport->local_description()->transport_desc.ice_ufrag,
>+            jsep_transport->local_description()->transport_desc.ice_pwd,
>+            tdesc.ice_ufrag, tdesc.ice_pwd) &&
>+        // Don't change the ICE role if the remote endpoint is ICE lite; we
>+        // should always be controlling in that case.
>+        (!jsep_transport->remote_description() ||
>+         jsep_transport->remote_description()->transport_desc.ice_mode !=
>+             cricket::ICEMODE_LITE)) {
>+      ice_role = (type == SdpType::kOffer) ? cricket::ICEROLE_CONTROLLING
>+                                           : cricket::ICEROLE_CONTROLLED;
>+    }
>+  } else {
>+    // If our role is cricket::ICEROLE_CONTROLLED and the remote endpoint
>+    // supports only ice_lite, this local endpoint should take the CONTROLLING
>+    // role.
>+    // TODO(deadbeef): This is a session-level attribute, so it really shouldn't
>+    // be in a TransportDescription in the first place...
>+    if (ice_role_ == cricket::ICEROLE_CONTROLLED &&
>+        tdesc.ice_mode == cricket::ICEMODE_LITE) {
>+      ice_role = cricket::ICEROLE_CONTROLLING;
>+    }
>+
>+    // If we use ICE Lite and the remote endpoint uses the full implementation
>+    // of ICE, the local endpoint must take the controlled role, and the other
>+    // side must be the controlling role.
>+    if (jsep_transport->local_description() &&
>+        jsep_transport->local_description()->transport_desc.ice_mode ==
>+            cricket::ICEMODE_LITE &&
>+        ice_role_ == cricket::ICEROLE_CONTROLLING &&
>+        tdesc.ice_mode == cricket::ICEMODE_FULL) {
>+      ice_role = cricket::ICEROLE_CONTROLLED;
>+    }
>+  }
>+
>+  return ice_role;
>+}
>+
>+void JsepTransportController::OnTransportWritableState_n(
>+    rtc::PacketTransportInternal* transport) {
>+  RTC_DCHECK(network_thread_->IsCurrent());
>+  RTC_LOG(LS_INFO) << " Transport " << transport->transport_name()
>+                   << " writability changed to " << transport->writable()
>+                   << ".";
>+  UpdateAggregateStates_n();
>+}
>+
>+void JsepTransportController::OnTransportReceivingState_n(
>+    rtc::PacketTransportInternal* transport) {
>+  RTC_DCHECK(network_thread_->IsCurrent());
>+  UpdateAggregateStates_n();
>+}
>+
>+void JsepTransportController::OnTransportGatheringState_n(
>+    cricket::IceTransportInternal* transport) {
>+  RTC_DCHECK(network_thread_->IsCurrent());
>+  UpdateAggregateStates_n();
>+}
>+
>+void JsepTransportController::OnTransportCandidateGathered_n(
>+    cricket::IceTransportInternal* transport,
>+    const cricket::Candidate& candidate) {
>+  RTC_DCHECK(network_thread_->IsCurrent());
>+
>+  // We should never signal peer-reflexive candidates.
>+  if (candidate.type() == cricket::PRFLX_PORT_TYPE) {
>+    RTC_NOTREACHED();
>+    return;
>+  }
>+  std::vector<cricket::Candidate> candidates;
>+  candidates.push_back(candidate);
>+  CandidatesData* data =
>+      new CandidatesData(transport->transport_name(), candidates);
>+  signaling_thread_->Post(RTC_FROM_HERE, this, MSG_ICECANDIDATESGATHERED, data);
>+}
>+
>+void JsepTransportController::OnTransportCandidatesRemoved_n(
>+    cricket::IceTransportInternal* transport,
>+    const cricket::Candidates& candidates) {
>+  invoker_.AsyncInvoke<void>(
>+      RTC_FROM_HERE, signaling_thread_,
>+      rtc::Bind(&JsepTransportController::OnTransportCandidatesRemoved, this,
>+                candidates));
>+}
>+
>+void JsepTransportController::OnTransportCandidatesRemoved(
>+    const cricket::Candidates& candidates) {
>+  RTC_DCHECK(signaling_thread_->IsCurrent());
>+  SignalIceCandidatesRemoved(candidates);
>+}
>+
>+void JsepTransportController::OnTransportRoleConflict_n(
>+    cricket::IceTransportInternal* transport) {
>+  RTC_DCHECK(network_thread_->IsCurrent());
>+  // Note: since the role conflict is handled entirely on the network thread,
>+  // we don't need to worry about role conflicts occurring on two ports at
>+  // once. The first one encountered should immediately reverse the role.
>+  cricket::IceRole reversed_role = (ice_role_ == cricket::ICEROLE_CONTROLLING)
>+                                       ? cricket::ICEROLE_CONTROLLED
>+                                       : cricket::ICEROLE_CONTROLLING;
>+  RTC_LOG(LS_INFO) << "Got role conflict; switching to "
>+                   << (reversed_role == cricket::ICEROLE_CONTROLLING
>+                           ? "controlling"
>+                           : "controlled")
>+                   << " role.";
>+  SetIceRole_n(reversed_role);
>+}
>+
>+void JsepTransportController::OnTransportStateChanged_n(
>+    cricket::IceTransportInternal* transport) {
>+  RTC_DCHECK(network_thread_->IsCurrent());
>+  RTC_LOG(LS_INFO) << transport->transport_name() << " Transport "
>+                   << transport->component()
>+                   << " state changed. Check if state is complete.";
>+  UpdateAggregateStates_n();
>+}
>+
>+void JsepTransportController::UpdateAggregateStates_n() {
>+  RTC_DCHECK(network_thread_->IsCurrent());
>+
>+  auto dtls_transports = GetDtlsTransports();
>+  cricket::IceConnectionState new_connection_state =
>+      cricket::kIceConnectionConnecting;
>+  cricket::IceGatheringState new_gathering_state = cricket::kIceGatheringNew;
>+  bool any_failed = false;
>+  bool all_connected = !dtls_transports.empty();
>+  bool all_completed = !dtls_transports.empty();
>+  bool any_gathering = false;
>+  bool all_done_gathering = !dtls_transports.empty();
>+  for (const auto& dtls : dtls_transports) {
>+    any_failed = any_failed || dtls->ice_transport()->GetState() ==
>+                                   cricket::IceTransportState::STATE_FAILED;
>+    all_connected = all_connected && dtls->writable();
>+    all_completed =
>+        all_completed && dtls->writable() &&
>+        dtls->ice_transport()->GetState() ==
>+            cricket::IceTransportState::STATE_COMPLETED &&
>+        dtls->ice_transport()->GetIceRole() == cricket::ICEROLE_CONTROLLING &&
>+        dtls->ice_transport()->gathering_state() ==
>+            cricket::kIceGatheringComplete;
>+    any_gathering = any_gathering || dtls->ice_transport()->gathering_state() !=
>+                                         cricket::kIceGatheringNew;
>+    all_done_gathering =
>+        all_done_gathering && dtls->ice_transport()->gathering_state() ==
>+                                  cricket::kIceGatheringComplete;
>+  }
>+  if (any_failed) {
>+    new_connection_state = cricket::kIceConnectionFailed;
>+  } else if (all_completed) {
>+    new_connection_state = cricket::kIceConnectionCompleted;
>+  } else if (all_connected) {
>+    new_connection_state = cricket::kIceConnectionConnected;
>+  }
>+  if (ice_connection_state_ != new_connection_state) {
>+    ice_connection_state_ = new_connection_state;
>+    signaling_thread_->Post(
>+        RTC_FROM_HERE, this, MSG_ICECONNECTIONSTATE,
>+        new rtc::TypedMessageData<cricket::IceConnectionState>(
>+            new_connection_state));
>+  }
>+
>+  if (all_done_gathering) {
>+    new_gathering_state = cricket::kIceGatheringComplete;
>+  } else if (any_gathering) {
>+    new_gathering_state = cricket::kIceGatheringGathering;
>+  }
>+  if (ice_gathering_state_ != new_gathering_state) {
>+    ice_gathering_state_ = new_gathering_state;
>+    signaling_thread_->Post(
>+        RTC_FROM_HERE, this, MSG_ICEGATHERINGSTATE,
>+        new rtc::TypedMessageData<cricket::IceGatheringState>(
>+            new_gathering_state));
>+  }
>+}
>+
>+void JsepTransportController::OnDtlsHandshakeError(
>+    rtc::SSLHandshakeError error) {
>+  SignalDtlsHandshakeError(error);
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jseptransportcontroller.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jseptransportcontroller.h
>new file mode 100644
>index 00000000000..c8effd746b7
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jseptransportcontroller.h
>@@ -0,0 +1,331 @@
>+/*
>+ *  Copyright 2017 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef PC_JSEPTRANSPORTCONTROLLER_H_
>+#define PC_JSEPTRANSPORTCONTROLLER_H_
>+
>+#include <map>
>+#include <memory>
>+#include <string>
>+#include <utility>
>+#include <vector>
>+
>+#include "api/candidate.h"
>+#include "api/peerconnectioninterface.h"
>+#include "logging/rtc_event_log/rtc_event_log.h"
>+#include "media/sctp/sctptransportinternal.h"
>+#include "p2p/base/dtlstransport.h"
>+#include "p2p/base/p2ptransportchannel.h"
>+#include "p2p/base/transportfactoryinterface.h"
>+#include "pc/channel.h"
>+#include "pc/dtlssrtptransport.h"
>+#include "pc/jseptransport.h"
>+#include "pc/rtptransport.h"
>+#include "pc/srtptransport.h"
>+#include "rtc_base/asyncinvoker.h"
>+#include "rtc_base/constructormagic.h"
>+#include "rtc_base/refcountedobject.h"
>+#include "rtc_base/sslstreamadapter.h"
>+#include "rtc_base/third_party/sigslot/sigslot.h"
>+
>+namespace rtc {
>+class Thread;
>+class PacketTransportInternal;
>+}  // namespace rtc
>+
>+namespace webrtc {
>+
>+class JsepTransportController : public sigslot::has_slots<>,
>+                                public rtc::MessageHandler {
>+ public:
>+  // Used when the RtpTransport/DtlsTransport of the m= section is changed
>+  // because the section is rejected or BUNDLE is enabled.
>+  class Observer {
>+   public:
>+    virtual ~Observer() {}
>+
>+    // Returns true if media associated with |mid| was successfully set up to be
>+    // demultiplexed on |rtp_transport|. Could return false if two bundled m=
>+    // sections use the same SSRC, for example.
>+    virtual bool OnTransportChanged(
>+        const std::string& mid,
>+        RtpTransportInternal* rtp_transport,
>+        cricket::DtlsTransportInternal* dtls_transport) = 0;
>+  };
>+
>+  struct Config {
>+    // If |redetermine_role_on_ice_restart| is true, ICE role is redetermined
>+    // upon setting a local transport description that indicates an ICE
>+    // restart.
>+    bool redetermine_role_on_ice_restart = true;
>+    rtc::SSLProtocolVersion ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
>+    // |crypto_options| is used to determine if created DTLS transports
>+    // negotiate GCM crypto suites or not.
>+    rtc::CryptoOptions crypto_options;
>+    PeerConnectionInterface::BundlePolicy bundle_policy =
>+        PeerConnectionInterface::kBundlePolicyBalanced;
>+    PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy =
>+        PeerConnectionInterface::kRtcpMuxPolicyRequire;
>+    bool disable_encryption = false;
>+    bool enable_external_auth = false;
>+    // Used to inject the ICE/DTLS transports created externally.
>+    cricket::TransportFactoryInterface* external_transport_factory = nullptr;
>+    Observer* transport_observer = nullptr;
>+    bool active_reset_srtp_params = false;
>+    RtcEventLog* event_log = nullptr;
>+  };
>+
>+  // The ICE related events are signaled on the |signaling_thread|.
>+  // All the transport related methods are called on the |network_thread|.
>+  JsepTransportController(rtc::Thread* signaling_thread,
>+                          rtc::Thread* network_thread,
>+                          cricket::PortAllocator* port_allocator,
>+                          AsyncResolverFactory* async_resolver_factory,
>+                          Config config);
>+  virtual ~JsepTransportController();
>+
>+  // The main method to be called; applies a description at the transport
>+  // level, creating/destroying transport objects as needed and updating their
>+  // properties. This includes RTP, DTLS, and ICE (but not SCTP). At least not
>+  // yet? May make sense to in the future.
>+  RTCError SetLocalDescription(SdpType type,
>+                               const cricket::SessionDescription* description);
>+
>+  RTCError SetRemoteDescription(SdpType type,
>+                                const cricket::SessionDescription* description);
>+
>+  // Get transports to be used for the provided |mid|. If bundling is enabled,
>+  // calling GetRtpTransport for multiple MIDs may yield the same object.
>+  RtpTransportInternal* GetRtpTransport(const std::string& mid) const;
>+  cricket::DtlsTransportInternal* GetDtlsTransport(
>+      const std::string& mid) const;
>+  cricket::DtlsTransportInternal* GetRtcpDtlsTransport(
>+      const std::string& mid) const;
>+
>+  /*********************
>+   * ICE-related methods
>+   ********************/
>+  // This method is public to allow PeerConnection to update it from
>+  // SetConfiguration.
>+  void SetIceConfig(const cricket::IceConfig& config);
>+  // Set the "needs-ice-restart" flag as described in JSEP. After the flag is
>+  // set, offers should generate new ufrags/passwords until an ICE restart
>+  // occurs.
>+  void SetNeedsIceRestartFlag();
>+  // Returns true if the ICE restart flag above was set, and no ICE restart has
>+  // occurred yet for this transport (by applying a local description with
>+  // changed ufrag/password). If the transport has been deleted as a result of
>+  // bundling, returns false.
>+  bool NeedsIceRestart(const std::string& mid) const;
>+  // Start gathering candidates for any new transports, or transports doing an
>+  // ICE restart.
>+  void MaybeStartGathering();
>+  RTCError AddRemoteCandidates(
>+      const std::string& mid,
>+      const std::vector<cricket::Candidate>& candidates);
>+  RTCError RemoveRemoteCandidates(
>+      const std::vector<cricket::Candidate>& candidates);
>+
>+  /**********************
>+   * DTLS-related methods
>+   *********************/
>+  // Specifies the identity to use in this session.
>+  // Can only be called once.
>+  bool SetLocalCertificate(
>+      const rtc::scoped_refptr<rtc::RTCCertificate>& certificate);
>+  rtc::scoped_refptr<rtc::RTCCertificate> GetLocalCertificate(
>+      const std::string& mid) const;
>+  // Caller owns returned certificate chain. This method mainly exists for
>+  // stats reporting.
>+  std::unique_ptr<rtc::SSLCertChain> GetRemoteSSLCertChain(
>+      const std::string& mid) const;
>+  // Get negotiated role, if one has been negotiated.
>+  absl::optional<rtc::SSLRole> GetDtlsRole(const std::string& mid) const;
>+
>+  // TODO(deadbeef): GetStats isn't const because all the way down to
>+  // OpenSSLStreamAdapter, GetSslCipherSuite and GetDtlsSrtpCryptoSuite are not
>+  // const. Fix this.
>+  bool GetStats(const std::string& mid, cricket::TransportStats* stats);
>+
>+  bool initial_offerer() const { return initial_offerer_ && *initial_offerer_; }
>+
>+  void SetActiveResetSrtpParams(bool active_reset_srtp_params);
>+
>+  // All of these signals are fired on the signaling thread.
>+
>+  // If any transport failed => failed,
>+  // Else if all completed => completed,
>+  // Else if all connected => connected,
>+  // Else => connecting
>+  sigslot::signal1<cricket::IceConnectionState> SignalIceConnectionState;
>+
>+  // If all transports done gathering => complete,
>+  // Else if any are gathering => gathering,
>+  // Else => new
>+  sigslot::signal1<cricket::IceGatheringState> SignalIceGatheringState;
>+
>+  // (mid, candidates)
>+  sigslot::signal2<const std::string&, const std::vector<cricket::Candidate>&>
>+      SignalIceCandidatesGathered;
>+
>+  sigslot::signal1<const std::vector<cricket::Candidate>&>
>+      SignalIceCandidatesRemoved;
>+
>+  sigslot::signal1<rtc::SSLHandshakeError> SignalDtlsHandshakeError;
>+
>+ private:
>+  void OnMessage(rtc::Message* pmsg) override;
>+
>+  RTCError ApplyDescription_n(bool local,
>+                              SdpType type,
>+                              const cricket::SessionDescription* description);
>+  RTCError ValidateAndMaybeUpdateBundleGroup(
>+      bool local,
>+      SdpType type,
>+      const cricket::SessionDescription* description);
>+  RTCError ValidateContent(const cricket::ContentInfo& content_info);
>+
>+  void HandleRejectedContent(const cricket::ContentInfo& content_info,
>+                             const cricket::SessionDescription* description);
>+  bool HandleBundledContent(const cricket::ContentInfo& content_info);
>+
>+  bool SetTransportForMid(const std::string& mid,
>+                          cricket::JsepTransport* jsep_transport);
>+  void RemoveTransportForMid(const std::string& mid);
>+
>+  cricket::JsepTransportDescription CreateJsepTransportDescription(
>+      cricket::ContentInfo content_info,
>+      cricket::TransportInfo transport_info,
>+      const std::vector<int>& encrypted_extension_ids,
>+      int rtp_abs_sendtime_extn_id);
>+
>+  absl::optional<std::string> bundled_mid() const {
>+    absl::optional<std::string> bundled_mid;
>+    if (bundle_group_ && bundle_group_->FirstContentName()) {
>+      bundled_mid = *(bundle_group_->FirstContentName());
>+    }
>+    return bundled_mid;
>+  }
>+
>+  bool IsBundled(const std::string& mid) const {
>+    return bundle_group_ && bundle_group_->HasContentName(mid);
>+  }
>+
>+  bool ShouldUpdateBundleGroup(SdpType type,
>+                               const cricket::SessionDescription* description);
>+
>+  std::vector<int> MergeEncryptedHeaderExtensionIdsForBundle(
>+      const cricket::SessionDescription* description);
>+  std::vector<int> GetEncryptedHeaderExtensionIds(
>+      const cricket::ContentInfo& content_info);
>+
>+  int GetRtpAbsSendTimeHeaderExtensionId(
>+      const cricket::ContentInfo& content_info);
>+
>+  // This method takes the BUNDLE group into account. If the JsepTransport is
>+  // destroyed because of BUNDLE, it would return the transport which other
>+  // transports are bundled on (In current implementation, it is the first
>+  // content in the BUNDLE group).
>+  const cricket::JsepTransport* GetJsepTransportForMid(
>+      const std::string& mid) const;
>+  cricket::JsepTransport* GetJsepTransportForMid(const std::string& mid);
>+
>+  // Get the JsepTransport without considering the BUNDLE group. Return nullptr
>+  // if the JsepTransport is destroyed.
>+  const cricket::JsepTransport* GetJsepTransportByName(
>+      const std::string& transport_name) const;
>+  cricket::JsepTransport* GetJsepTransportByName(
>+      const std::string& transport_name);
>+
>+  RTCError MaybeCreateJsepTransport(const cricket::ContentInfo& content_info);
>+  void MaybeDestroyJsepTransport(const std::string& mid);
>+  void DestroyAllJsepTransports_n();
>+
>+  void SetIceRole_n(cricket::IceRole ice_role);
>+
>+  cricket::IceRole DetermineIceRole(
>+      cricket::JsepTransport* jsep_transport,
>+      const cricket::TransportInfo& transport_info,
>+      SdpType type,
>+      bool local);
>+
>+  std::unique_ptr<cricket::DtlsTransportInternal> CreateDtlsTransport(
>+      const std::string& transport_name,
>+      bool rtcp);
>+
>+  std::unique_ptr<webrtc::RtpTransport> CreateUnencryptedRtpTransport(
>+      const std::string& transport_name,
>+      rtc::PacketTransportInternal* rtp_packet_transport,
>+      rtc::PacketTransportInternal* rtcp_packet_transport);
>+  std::unique_ptr<webrtc::SrtpTransport> CreateSdesTransport(
>+      const std::string& transport_name,
>+      cricket::DtlsTransportInternal* rtp_dtls_transport,
>+      cricket::DtlsTransportInternal* rtcp_dtls_transport);
>+  std::unique_ptr<webrtc::DtlsSrtpTransport> CreateDtlsSrtpTransport(
>+      const std::string& transport_name,
>+      cricket::DtlsTransportInternal* rtp_dtls_transport,
>+      cricket::DtlsTransportInternal* rtcp_dtls_transport);
>+
>+  // Collect all the DtlsTransports, including RTP and RTCP, from the
>+  // JsepTransports. JsepTransportController can iterate all the DtlsTransports
>+  // and update the aggregate states.
>+  std::vector<cricket::DtlsTransportInternal*> GetDtlsTransports();
>+
>+  // Handlers for signals from Transport.
>+  void OnTransportWritableState_n(rtc::PacketTransportInternal* transport);
>+  void OnTransportReceivingState_n(rtc::PacketTransportInternal* transport);
>+  void OnTransportGatheringState_n(cricket::IceTransportInternal* transport);
>+  void OnTransportCandidateGathered_n(cricket::IceTransportInternal* transport,
>+                                      const cricket::Candidate& candidate);
>+  void OnTransportCandidatesRemoved(const cricket::Candidates& candidates);
>+  void OnTransportCandidatesRemoved_n(cricket::IceTransportInternal* transport,
>+                                      const cricket::Candidates& candidates);
>+  void OnTransportRoleConflict_n(cricket::IceTransportInternal* transport);
>+  void OnTransportStateChanged_n(cricket::IceTransportInternal* transport);
>+
>+  void UpdateAggregateStates_n();
>+
>+  void OnDtlsHandshakeError(rtc::SSLHandshakeError error);
>+
>+  rtc::Thread* const signaling_thread_ = nullptr;
>+  rtc::Thread* const network_thread_ = nullptr;
>+  cricket::PortAllocator* const port_allocator_ = nullptr;
>+  AsyncResolverFactory* const async_resolver_factory_ = nullptr;
>+
>+  std::map<std::string, std::unique_ptr<cricket::JsepTransport>>
>+      jsep_transports_by_name_;
>+  // This keeps track of the mapping between media section
>+  // (BaseChannel/SctpTransport) and the JsepTransport underneath.
>+  std::map<std::string, cricket::JsepTransport*> mid_to_transport_;
>+
>+  // Aggregate state for Transports.
>+  cricket::IceConnectionState ice_connection_state_ =
>+      cricket::kIceConnectionConnecting;
>+  cricket::IceGatheringState ice_gathering_state_ = cricket::kIceGatheringNew;
>+
>+  Config config_;
>+  const cricket::SessionDescription* local_desc_ = nullptr;
>+  const cricket::SessionDescription* remote_desc_ = nullptr;
>+  absl::optional<bool> initial_offerer_;
>+
>+  absl::optional<cricket::ContentGroup> bundle_group_;
>+
>+  cricket::IceConfig ice_config_;
>+  cricket::IceRole ice_role_ = cricket::ICEROLE_CONTROLLING;
>+  uint64_t ice_tiebreaker_ = rtc::CreateRandomId64();
>+  rtc::scoped_refptr<rtc::RTCCertificate> certificate_;
>+  rtc::AsyncInvoker invoker_;
>+
>+  RTC_DISALLOW_COPY_AND_ASSIGN(JsepTransportController);
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // PC_JSEPTRANSPORTCONTROLLER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jseptransportcontroller_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jseptransportcontroller_unittest.cc
>new file mode 100644
>index 00000000000..6f8693bea7a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/jseptransportcontroller_unittest.cc
>@@ -0,0 +1,1411 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <map>
>+#include <memory>
>+
>+#include "absl/memory/memory.h"
>+#include "p2p/base/fakedtlstransport.h"
>+#include "p2p/base/fakeicetransport.h"
>+#include "p2p/base/transportfactoryinterface.h"
>+#include "p2p/base/transportinfo.h"
>+#include "pc/jseptransportcontroller.h"
>+#include "rtc_base/gunit.h"
>+#include "rtc_base/thread.h"
>+#include "test/gtest.h"
>+
>+using cricket::FakeDtlsTransport;
>+using cricket::Candidate;
>+using cricket::Candidates;
>+using webrtc::SdpType;
>+
>+static const int kTimeout = 100;
>+static const char kIceUfrag1[] = "u0001";
>+static const char kIcePwd1[] = "TESTICEPWD00000000000001";
>+static const char kIceUfrag2[] = "u0002";
>+static const char kIcePwd2[] = "TESTICEPWD00000000000002";
>+static const char kIceUfrag3[] = "u0003";
>+static const char kIcePwd3[] = "TESTICEPWD00000000000003";
>+static const char kAudioMid1[] = "audio1";
>+static const char kAudioMid2[] = "audio2";
>+static const char kVideoMid1[] = "video1";
>+static const char kVideoMid2[] = "video2";
>+static const char kDataMid1[] = "data1";
>+
>+namespace webrtc {
>+
>+class FakeTransportFactory : public cricket::TransportFactoryInterface {
>+ public:
>+  std::unique_ptr<cricket::IceTransportInternal> CreateIceTransport(
>+      const std::string& transport_name,
>+      int component) override {
>+    return absl::make_unique<cricket::FakeIceTransport>(transport_name,
>+                                                        component);
>+  }
>+
>+  std::unique_ptr<cricket::DtlsTransportInternal> CreateDtlsTransport(
>+      std::unique_ptr<cricket::IceTransportInternal> ice,
>+      const rtc::CryptoOptions& crypto_options) override {
>+    std::unique_ptr<cricket::FakeIceTransport> fake_ice(
>+        static_cast<cricket::FakeIceTransport*>(ice.release()));
>+    return absl::make_unique<FakeDtlsTransport>(std::move(fake_ice));
>+  }
>+};
>+
>+class JsepTransportControllerTest : public JsepTransportController::Observer,
>+                                    public testing::Test,
>+                                    public sigslot::has_slots<> {
>+ public:
>+  JsepTransportControllerTest() : signaling_thread_(rtc::Thread::Current()) {
>+    fake_transport_factory_ = absl::make_unique<FakeTransportFactory>();
>+  }
>+
>+  void CreateJsepTransportController(
>+      JsepTransportController::Config config,
>+      rtc::Thread* signaling_thread = rtc::Thread::Current(),
>+      rtc::Thread* network_thread = rtc::Thread::Current(),
>+      cricket::PortAllocator* port_allocator = nullptr) {
>+    config.transport_observer = this;
>+    // The tests only works with |fake_transport_factory|;
>+    config.external_transport_factory = fake_transport_factory_.get();
>+    // TODO(zstein): Provide an AsyncResolverFactory once it is required.
>+    transport_controller_ = absl::make_unique<JsepTransportController>(
>+        signaling_thread, network_thread, port_allocator, nullptr, config);
>+    ConnectTransportControllerSignals();
>+  }
>+
>+  void ConnectTransportControllerSignals() {
>+    transport_controller_->SignalIceConnectionState.connect(
>+        this, &JsepTransportControllerTest::OnConnectionState);
>+    transport_controller_->SignalIceGatheringState.connect(
>+        this, &JsepTransportControllerTest::OnGatheringState);
>+    transport_controller_->SignalIceCandidatesGathered.connect(
>+        this, &JsepTransportControllerTest::OnCandidatesGathered);
>+  }
>+
>+  std::unique_ptr<cricket::SessionDescription>
>+  CreateSessionDescriptionWithoutBundle() {
>+    auto description = absl::make_unique<cricket::SessionDescription>();
>+    AddAudioSection(description.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
>+                    cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
>+                    nullptr);
>+    AddVideoSection(description.get(), kVideoMid1, kIceUfrag1, kIcePwd1,
>+                    cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
>+                    nullptr);
>+    return description;
>+  }
>+
>+  std::unique_ptr<cricket::SessionDescription>
>+  CreateSessionDescriptionWithBundleGroup() {
>+    auto description = CreateSessionDescriptionWithoutBundle();
>+    cricket::ContentGroup bundle_group(cricket::GROUP_TYPE_BUNDLE);
>+    bundle_group.AddContentName(kAudioMid1);
>+    bundle_group.AddContentName(kVideoMid1);
>+    description->AddGroup(bundle_group);
>+
>+    return description;
>+  }
>+
>+  void AddAudioSection(cricket::SessionDescription* description,
>+                       const std::string& mid,
>+                       const std::string& ufrag,
>+                       const std::string& pwd,
>+                       cricket::IceMode ice_mode,
>+                       cricket::ConnectionRole conn_role,
>+                       rtc::scoped_refptr<rtc::RTCCertificate> cert) {
>+    std::unique_ptr<cricket::AudioContentDescription> audio(
>+        new cricket::AudioContentDescription());
>+    // Set RTCP-mux to be true because the default policy is "mux required".
>+    audio->set_rtcp_mux(true);
>+    description->AddContent(mid, cricket::MediaProtocolType::kRtp,
>+                            /*rejected=*/false, audio.release());
>+    AddTransportInfo(description, mid, ufrag, pwd, ice_mode, conn_role, cert);
>+  }
>+
>+  void AddVideoSection(cricket::SessionDescription* description,
>+                       const std::string& mid,
>+                       const std::string& ufrag,
>+                       const std::string& pwd,
>+                       cricket::IceMode ice_mode,
>+                       cricket::ConnectionRole conn_role,
>+                       rtc::scoped_refptr<rtc::RTCCertificate> cert) {
>+    std::unique_ptr<cricket::VideoContentDescription> video(
>+        new cricket::VideoContentDescription());
>+    // Set RTCP-mux to be true because the default policy is "mux required".
>+    video->set_rtcp_mux(true);
>+    description->AddContent(mid, cricket::MediaProtocolType::kRtp,
>+                            /*rejected=*/false, video.release());
>+    AddTransportInfo(description, mid, ufrag, pwd, ice_mode, conn_role, cert);
>+  }
>+
>+  void AddDataSection(cricket::SessionDescription* description,
>+                      const std::string& mid,
>+                      cricket::MediaProtocolType protocol_type,
>+                      const std::string& ufrag,
>+                      const std::string& pwd,
>+                      cricket::IceMode ice_mode,
>+                      cricket::ConnectionRole conn_role,
>+                      rtc::scoped_refptr<rtc::RTCCertificate> cert) {
>+    std::unique_ptr<cricket::DataContentDescription> data(
>+        new cricket::DataContentDescription());
>+    data->set_rtcp_mux(true);
>+    description->AddContent(mid, protocol_type,
>+                            /*rejected=*/false, data.release());
>+    AddTransportInfo(description, mid, ufrag, pwd, ice_mode, conn_role, cert);
>+  }
>+
>+  void AddTransportInfo(cricket::SessionDescription* description,
>+                        const std::string& mid,
>+                        const std::string& ufrag,
>+                        const std::string& pwd,
>+                        cricket::IceMode ice_mode,
>+                        cricket::ConnectionRole conn_role,
>+                        rtc::scoped_refptr<rtc::RTCCertificate> cert) {
>+    std::unique_ptr<rtc::SSLFingerprint> fingerprint;
>+    if (cert) {
>+      fingerprint.reset(rtc::SSLFingerprint::CreateFromCertificate(cert));
>+    }
>+
>+    cricket::TransportDescription transport_desc(std::vector<std::string>(),
>+                                                 ufrag, pwd, ice_mode,
>+                                                 conn_role, fingerprint.get());
>+    description->AddTransportInfo(cricket::TransportInfo(mid, transport_desc));
>+  }
>+
>+  cricket::IceConfig CreateIceConfig(
>+      int receiving_timeout,
>+      cricket::ContinualGatheringPolicy continual_gathering_policy) {
>+    cricket::IceConfig config;
>+    config.receiving_timeout = receiving_timeout;
>+    config.continual_gathering_policy = continual_gathering_policy;
>+    return config;
>+  }
>+
>+  Candidate CreateCandidate(const std::string& transport_name, int component) {
>+    Candidate c;
>+    c.set_transport_name(transport_name);
>+    c.set_address(rtc::SocketAddress("192.168.1.1", 8000));
>+    c.set_component(component);
>+    c.set_protocol(cricket::UDP_PROTOCOL_NAME);
>+    c.set_priority(1);
>+    return c;
>+  }
>+
>+  void CreateLocalDescriptionAndCompleteConnectionOnNetworkThread() {
>+    if (!network_thread_->IsCurrent()) {
>+      network_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
>+        CreateLocalDescriptionAndCompleteConnectionOnNetworkThread();
>+      });
>+      return;
>+    }
>+
>+    auto description = CreateSessionDescriptionWithBundleGroup();
>+    EXPECT_TRUE(transport_controller_
>+                    ->SetLocalDescription(SdpType::kOffer, description.get())
>+                    .ok());
>+
>+    transport_controller_->MaybeStartGathering();
>+    auto fake_audio_dtls = static_cast<FakeDtlsTransport*>(
>+        transport_controller_->GetDtlsTransport(kAudioMid1));
>+    auto fake_video_dtls = static_cast<FakeDtlsTransport*>(
>+        transport_controller_->GetDtlsTransport(kVideoMid1));
>+    fake_audio_dtls->fake_ice_transport()->SignalCandidateGathered(
>+        fake_audio_dtls->fake_ice_transport(),
>+        CreateCandidate(kAudioMid1, /*component=*/1));
>+    fake_video_dtls->fake_ice_transport()->SignalCandidateGathered(
>+        fake_video_dtls->fake_ice_transport(),
>+        CreateCandidate(kVideoMid1, /*component=*/1));
>+    fake_audio_dtls->fake_ice_transport()->SetCandidatesGatheringComplete();
>+    fake_video_dtls->fake_ice_transport()->SetCandidatesGatheringComplete();
>+    fake_audio_dtls->fake_ice_transport()->SetConnectionCount(2);
>+    fake_video_dtls->fake_ice_transport()->SetConnectionCount(2);
>+    fake_audio_dtls->SetReceiving(true);
>+    fake_video_dtls->SetReceiving(true);
>+    fake_audio_dtls->SetWritable(true);
>+    fake_video_dtls->SetWritable(true);
>+    fake_audio_dtls->fake_ice_transport()->SetConnectionCount(1);
>+    fake_video_dtls->fake_ice_transport()->SetConnectionCount(1);
>+  }
>+
>+ protected:
>+  void OnConnectionState(cricket::IceConnectionState state) {
>+    if (!signaling_thread_->IsCurrent()) {
>+      signaled_on_non_signaling_thread_ = true;
>+    }
>+    connection_state_ = state;
>+    ++connection_state_signal_count_;
>+  }
>+
>+  void OnGatheringState(cricket::IceGatheringState state) {
>+    if (!signaling_thread_->IsCurrent()) {
>+      signaled_on_non_signaling_thread_ = true;
>+    }
>+    gathering_state_ = state;
>+    ++gathering_state_signal_count_;
>+  }
>+
>+  void OnCandidatesGathered(const std::string& transport_name,
>+                            const Candidates& candidates) {
>+    if (!signaling_thread_->IsCurrent()) {
>+      signaled_on_non_signaling_thread_ = true;
>+    }
>+    candidates_[transport_name].insert(candidates_[transport_name].end(),
>+                                       candidates.begin(), candidates.end());
>+    ++candidates_signal_count_;
>+  }
>+
>+  // JsepTransportController::Observer overrides.
>+  bool OnTransportChanged(
>+      const std::string& mid,
>+      RtpTransportInternal* rtp_transport,
>+      cricket::DtlsTransportInternal* dtls_transport) override {
>+    changed_rtp_transport_by_mid_[mid] = rtp_transport;
>+    changed_dtls_transport_by_mid_[mid] = dtls_transport;
>+    return true;
>+  }
>+
>+  // Information received from signals from transport controller.
>+  cricket::IceConnectionState connection_state_ =
>+      cricket::kIceConnectionConnecting;
>+  bool receiving_ = false;
>+  cricket::IceGatheringState gathering_state_ = cricket::kIceGatheringNew;
>+  // transport_name => candidates
>+  std::map<std::string, Candidates> candidates_;
>+  // Counts of each signal emitted.
>+  int connection_state_signal_count_ = 0;
>+  int receiving_signal_count_ = 0;
>+  int gathering_state_signal_count_ = 0;
>+  int candidates_signal_count_ = 0;
>+
>+  // |network_thread_| should be destroyed after |transport_controller_|
>+  std::unique_ptr<rtc::Thread> network_thread_;
>+  std::unique_ptr<JsepTransportController> transport_controller_;
>+  std::unique_ptr<FakeTransportFactory> fake_transport_factory_;
>+  rtc::Thread* const signaling_thread_ = nullptr;
>+  bool signaled_on_non_signaling_thread_ = false;
>+  // Used to verify the SignalRtpTransportChanged/SignalDtlsTransportChanged are
>+  // signaled correctly.
>+  std::map<std::string, RtpTransportInternal*> changed_rtp_transport_by_mid_;
>+  std::map<std::string, cricket::DtlsTransportInternal*>
>+      changed_dtls_transport_by_mid_;
>+};
>+
>+TEST_F(JsepTransportControllerTest, GetRtpTransport) {
>+  CreateJsepTransportController(JsepTransportController::Config());
>+  auto description = CreateSessionDescriptionWithoutBundle();
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetLocalDescription(SdpType::kOffer, description.get())
>+                  .ok());
>+  auto audio_rtp_transport = transport_controller_->GetRtpTransport(kAudioMid1);
>+  auto video_rtp_transport = transport_controller_->GetRtpTransport(kVideoMid1);
>+  EXPECT_NE(nullptr, audio_rtp_transport);
>+  EXPECT_NE(nullptr, video_rtp_transport);
>+  EXPECT_NE(audio_rtp_transport, video_rtp_transport);
>+  // Return nullptr for non-existing ones.
>+  EXPECT_EQ(nullptr, transport_controller_->GetRtpTransport(kAudioMid2));
>+}
>+
>+TEST_F(JsepTransportControllerTest, GetDtlsTransport) {
>+  JsepTransportController::Config config;
>+  config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyNegotiate;
>+  CreateJsepTransportController(config);
>+  auto description = CreateSessionDescriptionWithoutBundle();
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetLocalDescription(SdpType::kOffer, description.get())
>+                  .ok());
>+  EXPECT_NE(nullptr, transport_controller_->GetDtlsTransport(kAudioMid1));
>+  EXPECT_NE(nullptr, transport_controller_->GetRtcpDtlsTransport(kAudioMid1));
>+  EXPECT_NE(nullptr, transport_controller_->GetDtlsTransport(kVideoMid1));
>+  EXPECT_NE(nullptr, transport_controller_->GetRtcpDtlsTransport(kVideoMid1));
>+  // Return nullptr for non-existing ones.
>+  EXPECT_EQ(nullptr, transport_controller_->GetDtlsTransport(kVideoMid2));
>+  EXPECT_EQ(nullptr, transport_controller_->GetRtcpDtlsTransport(kVideoMid2));
>+}
>+
>+TEST_F(JsepTransportControllerTest, GetDtlsTransportWithRtcpMux) {
>+  JsepTransportController::Config config;
>+  config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
>+  CreateJsepTransportController(config);
>+  auto description = CreateSessionDescriptionWithoutBundle();
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetLocalDescription(SdpType::kOffer, description.get())
>+                  .ok());
>+  EXPECT_NE(nullptr, transport_controller_->GetDtlsTransport(kAudioMid1));
>+  EXPECT_EQ(nullptr, transport_controller_->GetRtcpDtlsTransport(kAudioMid1));
>+  EXPECT_NE(nullptr, transport_controller_->GetDtlsTransport(kVideoMid1));
>+  EXPECT_EQ(nullptr, transport_controller_->GetRtcpDtlsTransport(kVideoMid1));
>+}
>+
>+TEST_F(JsepTransportControllerTest, SetIceConfig) {
>+  CreateJsepTransportController(JsepTransportController::Config());
>+  auto description = CreateSessionDescriptionWithoutBundle();
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetLocalDescription(SdpType::kOffer, description.get())
>+                  .ok());
>+
>+  transport_controller_->SetIceConfig(
>+      CreateIceConfig(kTimeout, cricket::GATHER_CONTINUALLY));
>+  FakeDtlsTransport* fake_audio_dtls = static_cast<FakeDtlsTransport*>(
>+      transport_controller_->GetDtlsTransport(kAudioMid1));
>+  ASSERT_NE(nullptr, fake_audio_dtls);
>+  EXPECT_EQ(kTimeout,
>+            fake_audio_dtls->fake_ice_transport()->receiving_timeout());
>+  EXPECT_TRUE(fake_audio_dtls->fake_ice_transport()->gather_continually());
>+
>+  // Test that value stored in controller is applied to new transports.
>+  AddAudioSection(description.get(), kAudioMid2, kIceUfrag1, kIcePwd1,
>+                  cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
>+                  nullptr);
>+
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetLocalDescription(SdpType::kOffer, description.get())
>+                  .ok());
>+  fake_audio_dtls = static_cast<FakeDtlsTransport*>(
>+      transport_controller_->GetDtlsTransport(kAudioMid2));
>+  ASSERT_NE(nullptr, fake_audio_dtls);
>+  EXPECT_EQ(kTimeout,
>+            fake_audio_dtls->fake_ice_transport()->receiving_timeout());
>+  EXPECT_TRUE(fake_audio_dtls->fake_ice_transport()->gather_continually());
>+}
>+
>+// Tests the getter and setter of the ICE restart flag.
>+TEST_F(JsepTransportControllerTest, NeedIceRestart) {
>+  CreateJsepTransportController(JsepTransportController::Config());
>+  auto description = CreateSessionDescriptionWithoutBundle();
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetLocalDescription(SdpType::kOffer, description.get())
>+                  .ok());
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetRemoteDescription(SdpType::kAnswer, description.get())
>+                  .ok());
>+
>+  // Initially NeedsIceRestart should return false.
>+  EXPECT_FALSE(transport_controller_->NeedsIceRestart(kAudioMid1));
>+  EXPECT_FALSE(transport_controller_->NeedsIceRestart(kVideoMid1));
>+  // Set the needs-ice-restart flag and verify NeedsIceRestart starts returning
>+  // true.
>+  transport_controller_->SetNeedsIceRestartFlag();
>+  EXPECT_TRUE(transport_controller_->NeedsIceRestart(kAudioMid1));
>+  EXPECT_TRUE(transport_controller_->NeedsIceRestart(kVideoMid1));
>+  // For a nonexistent transport, false should be returned.
>+  EXPECT_FALSE(transport_controller_->NeedsIceRestart(kVideoMid2));
>+
>+  // Reset the ice_ufrag/ice_pwd for audio.
>+  auto audio_transport_info = description->GetTransportInfoByName(kAudioMid1);
>+  audio_transport_info->description.ice_ufrag = kIceUfrag2;
>+  audio_transport_info->description.ice_pwd = kIcePwd2;
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetLocalDescription(SdpType::kOffer, description.get())
>+                  .ok());
>+  // Because the ICE is only restarted for audio, NeedsIceRestart is expected to
>+  // return false for audio and true for video.
>+  EXPECT_FALSE(transport_controller_->NeedsIceRestart(kAudioMid1));
>+  EXPECT_TRUE(transport_controller_->NeedsIceRestart(kVideoMid1));
>+}
>+
>+TEST_F(JsepTransportControllerTest, MaybeStartGathering) {
>+  CreateJsepTransportController(JsepTransportController::Config());
>+  auto description = CreateSessionDescriptionWithBundleGroup();
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetLocalDescription(SdpType::kOffer, description.get())
>+                  .ok());
>+  // After setting the local description, we should be able to start gathering
>+  // candidates.
>+  transport_controller_->MaybeStartGathering();
>+  EXPECT_EQ_WAIT(cricket::kIceGatheringGathering, gathering_state_, kTimeout);
>+  EXPECT_EQ(1, gathering_state_signal_count_);
>+}
>+
>+TEST_F(JsepTransportControllerTest, AddRemoveRemoteCandidates) {
>+  CreateJsepTransportController(JsepTransportController::Config());
>+  auto description = CreateSessionDescriptionWithoutBundle();
>+  transport_controller_->SetLocalDescription(SdpType::kOffer,
>+                                             description.get());
>+  transport_controller_->SetRemoteDescription(SdpType::kAnswer,
>+                                              description.get());
>+  auto fake_audio_dtls = static_cast<FakeDtlsTransport*>(
>+      transport_controller_->GetDtlsTransport(kAudioMid1));
>+  ASSERT_NE(nullptr, fake_audio_dtls);
>+  Candidates candidates;
>+  candidates.push_back(
>+      CreateCandidate(kAudioMid1, cricket::ICE_CANDIDATE_COMPONENT_RTP));
>+  EXPECT_TRUE(
>+      transport_controller_->AddRemoteCandidates(kAudioMid1, candidates).ok());
>+  EXPECT_EQ(1U,
>+            fake_audio_dtls->fake_ice_transport()->remote_candidates().size());
>+
>+  EXPECT_TRUE(transport_controller_->RemoveRemoteCandidates(candidates).ok());
>+  EXPECT_EQ(0U,
>+            fake_audio_dtls->fake_ice_transport()->remote_candidates().size());
>+}
>+
>+TEST_F(JsepTransportControllerTest, SetAndGetLocalCertificate) {
>+  CreateJsepTransportController(JsepTransportController::Config());
>+
>+  rtc::scoped_refptr<rtc::RTCCertificate> certificate1 =
>+      rtc::RTCCertificate::Create(std::unique_ptr<rtc::SSLIdentity>(
>+          rtc::SSLIdentity::Generate("session1", rtc::KT_DEFAULT)));
>+  rtc::scoped_refptr<rtc::RTCCertificate> returned_certificate;
>+
>+  auto description = absl::make_unique<cricket::SessionDescription>();
>+  AddAudioSection(description.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
>+                  cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
>+                  certificate1);
>+
>+  // Apply the local certificate.
>+  EXPECT_TRUE(transport_controller_->SetLocalCertificate(certificate1));
>+  // Apply the local description.
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetLocalDescription(SdpType::kOffer, description.get())
>+                  .ok());
>+  returned_certificate = transport_controller_->GetLocalCertificate(kAudioMid1);
>+  EXPECT_TRUE(returned_certificate);
>+  EXPECT_EQ(certificate1->identity()->certificate().ToPEMString(),
>+            returned_certificate->identity()->certificate().ToPEMString());
>+
>+  // Should fail if called for a nonexistant transport.
>+  EXPECT_EQ(nullptr, transport_controller_->GetLocalCertificate(kVideoMid1));
>+
>+  // Shouldn't be able to change the identity once set.
>+  rtc::scoped_refptr<rtc::RTCCertificate> certificate2 =
>+      rtc::RTCCertificate::Create(std::unique_ptr<rtc::SSLIdentity>(
>+          rtc::SSLIdentity::Generate("session2", rtc::KT_DEFAULT)));
>+  EXPECT_FALSE(transport_controller_->SetLocalCertificate(certificate2));
>+}
>+
>+TEST_F(JsepTransportControllerTest, GetRemoteSSLCertChain) {
>+  CreateJsepTransportController(JsepTransportController::Config());
>+  auto description = CreateSessionDescriptionWithBundleGroup();
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetLocalDescription(SdpType::kOffer, description.get())
>+                  .ok());
>+  rtc::FakeSSLCertificate fake_certificate("fake_data");
>+
>+  auto fake_audio_dtls = static_cast<FakeDtlsTransport*>(
>+      transport_controller_->GetDtlsTransport(kAudioMid1));
>+  fake_audio_dtls->SetRemoteSSLCertificate(&fake_certificate);
>+  std::unique_ptr<rtc::SSLCertChain> returned_cert_chain =
>+      transport_controller_->GetRemoteSSLCertChain(kAudioMid1);
>+  ASSERT_TRUE(returned_cert_chain);
>+  ASSERT_EQ(1u, returned_cert_chain->GetSize());
>+  EXPECT_EQ(fake_certificate.ToPEMString(),
>+            returned_cert_chain->Get(0).ToPEMString());
>+
>+  // Should fail if called for a nonexistant transport.
>+  EXPECT_FALSE(transport_controller_->GetRemoteSSLCertChain(kAudioMid2));
>+}
>+
>+TEST_F(JsepTransportControllerTest, GetDtlsRole) {
>+  CreateJsepTransportController(JsepTransportController::Config());
>+  auto offer_certificate =
>+      rtc::RTCCertificate::Create(std::unique_ptr<rtc::SSLIdentity>(
>+          rtc::SSLIdentity::Generate("offer", rtc::KT_DEFAULT)));
>+  auto answer_certificate =
>+      rtc::RTCCertificate::Create(std::unique_ptr<rtc::SSLIdentity>(
>+          rtc::SSLIdentity::Generate("answer", rtc::KT_DEFAULT)));
>+  transport_controller_->SetLocalCertificate(offer_certificate);
>+
>+  auto offer_desc = absl::make_unique<cricket::SessionDescription>();
>+  AddAudioSection(offer_desc.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
>+                  cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
>+                  offer_certificate);
>+  auto answer_desc = absl::make_unique<cricket::SessionDescription>();
>+  AddAudioSection(answer_desc.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
>+                  cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
>+                  answer_certificate);
>+
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetLocalDescription(SdpType::kOffer, offer_desc.get())
>+                  .ok());
>+
>+  absl::optional<rtc::SSLRole> role =
>+      transport_controller_->GetDtlsRole(kAudioMid1);
>+  // The DTLS role is not decided yet.
>+  EXPECT_FALSE(role);
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetRemoteDescription(SdpType::kAnswer, answer_desc.get())
>+                  .ok());
>+  role = transport_controller_->GetDtlsRole(kAudioMid1);
>+
>+  ASSERT_TRUE(role);
>+  EXPECT_EQ(rtc::SSL_CLIENT, *role);
>+}
>+
>+TEST_F(JsepTransportControllerTest, GetStats) {
>+  CreateJsepTransportController(JsepTransportController::Config());
>+  auto description = CreateSessionDescriptionWithBundleGroup();
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetLocalDescription(SdpType::kOffer, description.get())
>+                  .ok());
>+
>+  cricket::TransportStats stats;
>+  EXPECT_TRUE(transport_controller_->GetStats(kAudioMid1, &stats));
>+  EXPECT_EQ(kAudioMid1, stats.transport_name);
>+  EXPECT_EQ(1u, stats.channel_stats.size());
>+  // Return false for non-existing transport.
>+  EXPECT_FALSE(transport_controller_->GetStats(kAudioMid2, &stats));
>+}
>+
>+TEST_F(JsepTransportControllerTest, SignalConnectionStateFailed) {
>+  CreateJsepTransportController(JsepTransportController::Config());
>+  auto description = CreateSessionDescriptionWithoutBundle();
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetLocalDescription(SdpType::kOffer, description.get())
>+                  .ok());
>+
>+  auto fake_ice = static_cast<cricket::FakeIceTransport*>(
>+      transport_controller_->GetDtlsTransport(kAudioMid1)->ice_transport());
>+  fake_ice->SetCandidatesGatheringComplete();
>+  fake_ice->SetConnectionCount(1);
>+  // The connection stats will be failed if there is no active connection.
>+  fake_ice->SetConnectionCount(0);
>+  EXPECT_EQ_WAIT(cricket::kIceConnectionFailed, connection_state_, kTimeout);
>+  EXPECT_EQ(1, connection_state_signal_count_);
>+}
>+
>+TEST_F(JsepTransportControllerTest, SignalConnectionStateConnected) {
>+  CreateJsepTransportController(JsepTransportController::Config());
>+  auto description = CreateSessionDescriptionWithoutBundle();
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetLocalDescription(SdpType::kOffer, description.get())
>+                  .ok());
>+
>+  auto fake_audio_dtls = static_cast<FakeDtlsTransport*>(
>+      transport_controller_->GetDtlsTransport(kAudioMid1));
>+  auto fake_video_dtls = static_cast<FakeDtlsTransport*>(
>+      transport_controller_->GetDtlsTransport(kVideoMid1));
>+
>+  // First, have one transport connect, and another fail, to ensure that
>+  // the first transport connecting didn't trigger a "connected" state signal.
>+  // We should only get a signal when all are connected.
>+  fake_audio_dtls->fake_ice_transport()->SetConnectionCount(1);
>+  fake_audio_dtls->SetWritable(true);
>+  fake_audio_dtls->fake_ice_transport()->SetCandidatesGatheringComplete();
>+  // Decrease the number of the connection to trigger the signal.
>+  fake_video_dtls->fake_ice_transport()->SetConnectionCount(1);
>+  fake_video_dtls->fake_ice_transport()->SetConnectionCount(0);
>+  fake_video_dtls->fake_ice_transport()->SetCandidatesGatheringComplete();
>+
>+  EXPECT_EQ_WAIT(cricket::kIceConnectionFailed, connection_state_, kTimeout);
>+  EXPECT_EQ(1, connection_state_signal_count_);
>+
>+  // Set the connection count to be 2 and the cricket::FakeIceTransport will set
>+  // the transport state to be STATE_CONNECTING.
>+  fake_video_dtls->fake_ice_transport()->SetConnectionCount(2);
>+  fake_video_dtls->SetWritable(true);
>+  EXPECT_EQ_WAIT(cricket::kIceConnectionConnected, connection_state_, kTimeout);
>+  EXPECT_EQ(2, connection_state_signal_count_);
>+}
>+
>+TEST_F(JsepTransportControllerTest, SignalConnectionStateComplete) {
>+  CreateJsepTransportController(JsepTransportController::Config());
>+  auto description = CreateSessionDescriptionWithoutBundle();
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetLocalDescription(SdpType::kOffer, description.get())
>+                  .ok());
>+
>+  auto fake_audio_dtls = static_cast<FakeDtlsTransport*>(
>+      transport_controller_->GetDtlsTransport(kAudioMid1));
>+  auto fake_video_dtls = static_cast<FakeDtlsTransport*>(
>+      transport_controller_->GetDtlsTransport(kVideoMid1));
>+
>+  // First, have one transport connect, and another fail, to ensure that
>+  // the first transport connecting didn't trigger a "connected" state signal.
>+  // We should only get a signal when all are connected.
>+  fake_audio_dtls->fake_ice_transport()->SetConnectionCount(1);
>+  fake_audio_dtls->SetWritable(true);
>+  fake_audio_dtls->fake_ice_transport()->SetCandidatesGatheringComplete();
>+  // Decrease the number of the connection to trigger the signal.
>+  fake_video_dtls->fake_ice_transport()->SetConnectionCount(1);
>+  fake_video_dtls->fake_ice_transport()->SetConnectionCount(0);
>+  fake_video_dtls->fake_ice_transport()->SetCandidatesGatheringComplete();
>+
>+  EXPECT_EQ_WAIT(cricket::kIceConnectionFailed, connection_state_, kTimeout);
>+  EXPECT_EQ(1, connection_state_signal_count_);
>+
>+  // Set the connection count to be 1 and the cricket::FakeIceTransport will set
>+  // the transport state to be STATE_COMPLETED.
>+  fake_video_dtls->fake_ice_transport()->SetConnectionCount(1);
>+  fake_video_dtls->SetWritable(true);
>+  EXPECT_EQ_WAIT(cricket::kIceConnectionCompleted, connection_state_, kTimeout);
>+  EXPECT_EQ(2, connection_state_signal_count_);
>+}
>+
>+TEST_F(JsepTransportControllerTest, SignalIceGatheringStateGathering) {
>+  CreateJsepTransportController(JsepTransportController::Config());
>+  auto description = CreateSessionDescriptionWithoutBundle();
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetLocalDescription(SdpType::kOffer, description.get())
>+                  .ok());
>+
>+  auto fake_audio_dtls = static_cast<FakeDtlsTransport*>(
>+      transport_controller_->GetDtlsTransport(kAudioMid1));
>+  fake_audio_dtls->fake_ice_transport()->MaybeStartGathering();
>+  // Should be in the gathering state as soon as any transport starts gathering.
>+  EXPECT_EQ_WAIT(cricket::kIceGatheringGathering, gathering_state_, kTimeout);
>+  EXPECT_EQ(1, gathering_state_signal_count_);
>+}
>+
>+TEST_F(JsepTransportControllerTest, SignalIceGatheringStateComplete) {
>+  CreateJsepTransportController(JsepTransportController::Config());
>+  auto description = CreateSessionDescriptionWithoutBundle();
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetLocalDescription(SdpType::kOffer, description.get())
>+                  .ok());
>+
>+  auto fake_audio_dtls = static_cast<FakeDtlsTransport*>(
>+      transport_controller_->GetDtlsTransport(kAudioMid1));
>+  auto fake_video_dtls = static_cast<FakeDtlsTransport*>(
>+      transport_controller_->GetDtlsTransport(kVideoMid1));
>+
>+  fake_audio_dtls->fake_ice_transport()->MaybeStartGathering();
>+  EXPECT_EQ_WAIT(cricket::kIceGatheringGathering, gathering_state_, kTimeout);
>+  EXPECT_EQ(1, gathering_state_signal_count_);
>+
>+  // Have one transport finish gathering, to make sure gathering
>+  // completion wasn't signalled if only one transport finished gathering.
>+  fake_audio_dtls->fake_ice_transport()->SetCandidatesGatheringComplete();
>+  EXPECT_EQ(1, gathering_state_signal_count_);
>+
>+  fake_video_dtls->fake_ice_transport()->MaybeStartGathering();
>+  EXPECT_EQ_WAIT(cricket::kIceGatheringGathering, gathering_state_, kTimeout);
>+  EXPECT_EQ(1, gathering_state_signal_count_);
>+
>+  fake_video_dtls->fake_ice_transport()->SetCandidatesGatheringComplete();
>+  EXPECT_EQ_WAIT(cricket::kIceGatheringComplete, gathering_state_, kTimeout);
>+  EXPECT_EQ(2, gathering_state_signal_count_);
>+}
>+
>+// Test that when the last transport that hasn't finished connecting and/or
>+// gathering is destroyed, the aggregate state jumps to "completed". This can
>+// happen if, for example, we have an audio and video transport, the audio
>+// transport completes, then we start bundling video on the audio transport.
>+TEST_F(JsepTransportControllerTest,
>+       SignalingWhenLastIncompleteTransportDestroyed) {
>+  CreateJsepTransportController(JsepTransportController::Config());
>+  auto description = CreateSessionDescriptionWithBundleGroup();
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetLocalDescription(SdpType::kOffer, description.get())
>+                  .ok());
>+
>+  auto fake_audio_dtls = static_cast<FakeDtlsTransport*>(
>+      transport_controller_->GetDtlsTransport(kAudioMid1));
>+  auto fake_video_dtls = static_cast<FakeDtlsTransport*>(
>+      transport_controller_->GetDtlsTransport(kVideoMid1));
>+  EXPECT_NE(fake_audio_dtls, fake_video_dtls);
>+
>+  fake_audio_dtls->fake_ice_transport()->MaybeStartGathering();
>+  EXPECT_EQ_WAIT(cricket::kIceGatheringGathering, gathering_state_, kTimeout);
>+  EXPECT_EQ(1, gathering_state_signal_count_);
>+
>+  // Let the audio transport complete.
>+  fake_audio_dtls->SetWritable(true);
>+  fake_audio_dtls->fake_ice_transport()->SetCandidatesGatheringComplete();
>+  fake_audio_dtls->fake_ice_transport()->SetConnectionCount(1);
>+  EXPECT_EQ(1, gathering_state_signal_count_);
>+
>+  // Set the remote description and enable the bundle.
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetRemoteDescription(SdpType::kAnswer, description.get())
>+                  .ok());
>+  // The BUNDLE should be enabled, the incomplete video transport should be
>+  // deleted and the states shoud be updated.
>+  fake_video_dtls = static_cast<FakeDtlsTransport*>(
>+      transport_controller_->GetDtlsTransport(kVideoMid1));
>+  EXPECT_EQ(fake_audio_dtls, fake_video_dtls);
>+  EXPECT_EQ_WAIT(cricket::kIceConnectionCompleted, connection_state_, kTimeout);
>+  EXPECT_EQ_WAIT(cricket::kIceGatheringComplete, gathering_state_, kTimeout);
>+  EXPECT_EQ(2, gathering_state_signal_count_);
>+}
>+
>+TEST_F(JsepTransportControllerTest, SignalCandidatesGathered) {
>+  CreateJsepTransportController(JsepTransportController::Config());
>+  auto description = CreateSessionDescriptionWithBundleGroup();
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetLocalDescription(SdpType::kOffer, description.get())
>+                  .ok());
>+  transport_controller_->MaybeStartGathering();
>+
>+  auto fake_audio_dtls = static_cast<FakeDtlsTransport*>(
>+      transport_controller_->GetDtlsTransport(kAudioMid1));
>+  fake_audio_dtls->fake_ice_transport()->SignalCandidateGathered(
>+      fake_audio_dtls->fake_ice_transport(), CreateCandidate(kAudioMid1, 1));
>+  EXPECT_EQ_WAIT(1, candidates_signal_count_, kTimeout);
>+  EXPECT_EQ(1u, candidates_[kAudioMid1].size());
>+}
>+
>+TEST_F(JsepTransportControllerTest, IceSignalingOccursOnSignalingThread) {
>+  network_thread_ = rtc::Thread::CreateWithSocketServer();
>+  network_thread_->Start();
>+  CreateJsepTransportController(JsepTransportController::Config(),
>+                                signaling_thread_, network_thread_.get(),
>+                                /*PortAllocator=*/nullptr);
>+  CreateLocalDescriptionAndCompleteConnectionOnNetworkThread();
>+
>+  // connecting --> connected --> completed
>+  EXPECT_EQ_WAIT(cricket::kIceConnectionCompleted, connection_state_, kTimeout);
>+  EXPECT_EQ(2, connection_state_signal_count_);
>+
>+  // new --> gathering --> complete
>+  EXPECT_EQ_WAIT(cricket::kIceGatheringComplete, gathering_state_, kTimeout);
>+  EXPECT_EQ(2, gathering_state_signal_count_);
>+
>+  EXPECT_EQ_WAIT(1u, candidates_[kAudioMid1].size(), kTimeout);
>+  EXPECT_EQ_WAIT(1u, candidates_[kVideoMid1].size(), kTimeout);
>+  EXPECT_EQ(2, candidates_signal_count_);
>+
>+  EXPECT_TRUE(!signaled_on_non_signaling_thread_);
>+}
>+
>+// Older versions of Chrome expect the ICE role to be re-determined when an
>+// ICE restart occurs, and also don't perform conflict resolution correctly,
>+// so for now we can't safely stop doing this.
>+// See: https://bugs.chromium.org/p/chromium/issues/detail?id=628676
>+// TODO(deadbeef): Remove this when these old versions of Chrome reach a low
>+// enough population.
>+TEST_F(JsepTransportControllerTest, IceRoleRedeterminedOnIceRestartByDefault) {
>+  CreateJsepTransportController(JsepTransportController::Config());
>+  // Let the |transport_controller_| be the controlled side initially.
>+  auto remote_offer = absl::make_unique<cricket::SessionDescription>();
>+  AddAudioSection(remote_offer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
>+                  cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
>+                  nullptr);
>+  auto local_answer = absl::make_unique<cricket::SessionDescription>();
>+  AddAudioSection(local_answer.get(), kAudioMid1, kIceUfrag2, kIcePwd2,
>+                  cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
>+                  nullptr);
>+
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetRemoteDescription(SdpType::kOffer, remote_offer.get())
>+                  .ok());
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetLocalDescription(SdpType::kAnswer, local_answer.get())
>+                  .ok());
>+
>+  auto fake_dtls = static_cast<FakeDtlsTransport*>(
>+      transport_controller_->GetDtlsTransport(kAudioMid1));
>+  EXPECT_EQ(cricket::ICEROLE_CONTROLLED,
>+            fake_dtls->fake_ice_transport()->GetIceRole());
>+
>+  // New offer will trigger the ICE restart.
>+  auto restart_local_offer = absl::make_unique<cricket::SessionDescription>();
>+  AddAudioSection(restart_local_offer.get(), kAudioMid1, kIceUfrag3, kIcePwd3,
>+                  cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
>+                  nullptr);
>+  EXPECT_TRUE(
>+      transport_controller_
>+          ->SetLocalDescription(SdpType::kOffer, restart_local_offer.get())
>+          .ok());
>+  EXPECT_EQ(cricket::ICEROLE_CONTROLLING,
>+            fake_dtls->fake_ice_transport()->GetIceRole());
>+}
>+
>+// Test that if the TransportController was created with the
>+// |redetermine_role_on_ice_restart| parameter set to false, the role is *not*
>+// redetermined on an ICE restart.
>+TEST_F(JsepTransportControllerTest, IceRoleNotRedetermined) {
>+  JsepTransportController::Config config;
>+  config.redetermine_role_on_ice_restart = false;
>+
>+  CreateJsepTransportController(config);
>+  // Let the |transport_controller_| be the controlled side initially.
>+  auto remote_offer = absl::make_unique<cricket::SessionDescription>();
>+  AddAudioSection(remote_offer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
>+                  cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
>+                  nullptr);
>+  auto local_answer = absl::make_unique<cricket::SessionDescription>();
>+  AddAudioSection(local_answer.get(), kAudioMid1, kIceUfrag2, kIcePwd2,
>+                  cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
>+                  nullptr);
>+
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetRemoteDescription(SdpType::kOffer, remote_offer.get())
>+                  .ok());
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetLocalDescription(SdpType::kAnswer, local_answer.get())
>+                  .ok());
>+
>+  auto fake_dtls = static_cast<FakeDtlsTransport*>(
>+      transport_controller_->GetDtlsTransport(kAudioMid1));
>+  EXPECT_EQ(cricket::ICEROLE_CONTROLLED,
>+            fake_dtls->fake_ice_transport()->GetIceRole());
>+
>+  // New offer will trigger the ICE restart.
>+  auto restart_local_offer = absl::make_unique<cricket::SessionDescription>();
>+  AddAudioSection(restart_local_offer.get(), kAudioMid1, kIceUfrag3, kIcePwd3,
>+                  cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
>+                  nullptr);
>+  EXPECT_TRUE(
>+      transport_controller_
>+          ->SetLocalDescription(SdpType::kOffer, restart_local_offer.get())
>+          .ok());
>+  EXPECT_EQ(cricket::ICEROLE_CONTROLLED,
>+            fake_dtls->fake_ice_transport()->GetIceRole());
>+}
>+
>+// Tests ICE-Lite mode in remote answer.
>+TEST_F(JsepTransportControllerTest, SetIceRoleWhenIceLiteInRemoteAnswer) {
>+  CreateJsepTransportController(JsepTransportController::Config());
>+  auto local_offer = absl::make_unique<cricket::SessionDescription>();
>+  AddAudioSection(local_offer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
>+                  cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
>+                  nullptr);
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetLocalDescription(SdpType::kOffer, local_offer.get())
>+                  .ok());
>+  auto fake_dtls = static_cast<FakeDtlsTransport*>(
>+      transport_controller_->GetDtlsTransport(kAudioMid1));
>+  EXPECT_EQ(cricket::ICEROLE_CONTROLLING,
>+            fake_dtls->fake_ice_transport()->GetIceRole());
>+  EXPECT_EQ(cricket::ICEMODE_FULL,
>+            fake_dtls->fake_ice_transport()->remote_ice_mode());
>+
>+  auto remote_answer = absl::make_unique<cricket::SessionDescription>();
>+  AddAudioSection(remote_answer.get(), kAudioMid1, kIceUfrag2, kIcePwd2,
>+                  cricket::ICEMODE_LITE, cricket::CONNECTIONROLE_PASSIVE,
>+                  nullptr);
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
>+                  .ok());
>+  EXPECT_EQ(cricket::ICEROLE_CONTROLLING,
>+            fake_dtls->fake_ice_transport()->GetIceRole());
>+  EXPECT_EQ(cricket::ICEMODE_LITE,
>+            fake_dtls->fake_ice_transport()->remote_ice_mode());
>+}
>+
>+// Tests that the ICE role remains "controlling" if a subsequent offer that
>+// does an ICE restart is received from an ICE lite endpoint. Regression test
>+// for: https://crbug.com/710760
>+TEST_F(JsepTransportControllerTest,
>+       IceRoleIsControllingAfterIceRestartFromIceLiteEndpoint) {
>+  CreateJsepTransportController(JsepTransportController::Config());
>+  auto remote_offer = absl::make_unique<cricket::SessionDescription>();
>+  AddAudioSection(remote_offer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
>+                  cricket::ICEMODE_LITE, cricket::CONNECTIONROLE_ACTPASS,
>+                  nullptr);
>+  auto local_answer = absl::make_unique<cricket::SessionDescription>();
>+  AddAudioSection(local_answer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
>+                  cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
>+                  nullptr);
>+  // Initial Offer/Answer exchange. If the remote offerer is ICE-Lite, then the
>+  // local side is the controlling.
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetRemoteDescription(SdpType::kOffer, remote_offer.get())
>+                  .ok());
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetLocalDescription(SdpType::kAnswer, local_answer.get())
>+                  .ok());
>+  auto fake_dtls = static_cast<FakeDtlsTransport*>(
>+      transport_controller_->GetDtlsTransport(kAudioMid1));
>+  EXPECT_EQ(cricket::ICEROLE_CONTROLLING,
>+            fake_dtls->fake_ice_transport()->GetIceRole());
>+
>+  // In the subsequence remote offer triggers an ICE restart.
>+  auto remote_offer2 = absl::make_unique<cricket::SessionDescription>();
>+  AddAudioSection(remote_offer2.get(), kAudioMid1, kIceUfrag2, kIcePwd2,
>+                  cricket::ICEMODE_LITE, cricket::CONNECTIONROLE_ACTPASS,
>+                  nullptr);
>+  auto local_answer2 = absl::make_unique<cricket::SessionDescription>();
>+  AddAudioSection(local_answer2.get(), kAudioMid1, kIceUfrag2, kIcePwd2,
>+                  cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
>+                  nullptr);
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetRemoteDescription(SdpType::kOffer, remote_offer2.get())
>+                  .ok());
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetLocalDescription(SdpType::kAnswer, local_answer2.get())
>+                  .ok());
>+  fake_dtls = static_cast<FakeDtlsTransport*>(
>+      transport_controller_->GetDtlsTransport(kAudioMid1));
>+  // The local side is still the controlling role since the remote side is using
>+  // ICE-Lite.
>+  EXPECT_EQ(cricket::ICEROLE_CONTROLLING,
>+            fake_dtls->fake_ice_transport()->GetIceRole());
>+}
>+
>+// Tests that the SDP has more than one audio/video m= sections.
>+TEST_F(JsepTransportControllerTest, MultipleMediaSectionsOfSameTypeWithBundle) {
>+  CreateJsepTransportController(JsepTransportController::Config());
>+  cricket::ContentGroup bundle_group(cricket::GROUP_TYPE_BUNDLE);
>+  bundle_group.AddContentName(kAudioMid1);
>+  bundle_group.AddContentName(kAudioMid2);
>+  bundle_group.AddContentName(kVideoMid1);
>+  bundle_group.AddContentName(kDataMid1);
>+
>+  auto local_offer = absl::make_unique<cricket::SessionDescription>();
>+  AddAudioSection(local_offer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
>+                  cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
>+                  nullptr);
>+  AddAudioSection(local_offer.get(), kAudioMid2, kIceUfrag2, kIcePwd2,
>+                  cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
>+                  nullptr);
>+  AddVideoSection(local_offer.get(), kVideoMid1, kIceUfrag3, kIcePwd3,
>+                  cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
>+                  nullptr);
>+  AddDataSection(local_offer.get(), kDataMid1,
>+                 cricket::MediaProtocolType::kSctp, kIceUfrag1, kIcePwd1,
>+                 cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
>+                 nullptr);
>+
>+  auto remote_answer = absl::make_unique<cricket::SessionDescription>();
>+  AddAudioSection(remote_answer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
>+                  cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
>+                  nullptr);
>+  AddAudioSection(remote_answer.get(), kAudioMid2, kIceUfrag2, kIcePwd2,
>+                  cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
>+                  nullptr);
>+  AddVideoSection(remote_answer.get(), kVideoMid1, kIceUfrag3, kIcePwd3,
>+                  cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
>+                  nullptr);
>+  AddDataSection(remote_answer.get(), kDataMid1,
>+                 cricket::MediaProtocolType::kSctp, kIceUfrag1, kIcePwd1,
>+                 cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
>+                 nullptr);
>+
>+  local_offer->AddGroup(bundle_group);
>+  remote_answer->AddGroup(bundle_group);
>+
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetLocalDescription(SdpType::kOffer, local_offer.get())
>+                  .ok());
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
>+                  .ok());
>+  // Verify that all the sections are bundled on kAudio1.
>+  auto transport1 = transport_controller_->GetRtpTransport(kAudioMid1);
>+  auto transport2 = transport_controller_->GetRtpTransport(kAudioMid2);
>+  auto transport3 = transport_controller_->GetRtpTransport(kVideoMid1);
>+  auto transport4 = transport_controller_->GetRtpTransport(kDataMid1);
>+  EXPECT_EQ(transport1, transport2);
>+  EXPECT_EQ(transport1, transport3);
>+  EXPECT_EQ(transport1, transport4);
>+
>+  // Verify the OnRtpTransport/DtlsTransportChanged signals are fired correctly.
>+  auto it = changed_rtp_transport_by_mid_.find(kAudioMid2);
>+  ASSERT_TRUE(it != changed_rtp_transport_by_mid_.end());
>+  EXPECT_EQ(transport1, it->second);
>+  it = changed_rtp_transport_by_mid_.find(kAudioMid2);
>+  ASSERT_TRUE(it != changed_rtp_transport_by_mid_.end());
>+  EXPECT_EQ(transport1, it->second);
>+  it = changed_rtp_transport_by_mid_.find(kVideoMid1);
>+  ASSERT_TRUE(it != changed_rtp_transport_by_mid_.end());
>+  EXPECT_EQ(transport1, it->second);
>+  // Verify the DtlsTransport for the SCTP data channel is reset correctly.
>+  auto it2 = changed_dtls_transport_by_mid_.find(kDataMid1);
>+  ASSERT_TRUE(it2 != changed_dtls_transport_by_mid_.end());
>+  EXPECT_EQ(transport1->rtp_packet_transport(), it2->second);
>+}
>+
>+// Tests that only a subset of all the m= sections are bundled.
>+TEST_F(JsepTransportControllerTest, BundleSubsetOfMediaSections) {
>+  CreateJsepTransportController(JsepTransportController::Config());
>+  cricket::ContentGroup bundle_group(cricket::GROUP_TYPE_BUNDLE);
>+  bundle_group.AddContentName(kAudioMid1);
>+  bundle_group.AddContentName(kVideoMid1);
>+
>+  auto local_offer = absl::make_unique<cricket::SessionDescription>();
>+  AddAudioSection(local_offer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
>+                  cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
>+                  nullptr);
>+  AddAudioSection(local_offer.get(), kAudioMid2, kIceUfrag2, kIcePwd2,
>+                  cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
>+                  nullptr);
>+  AddVideoSection(local_offer.get(), kVideoMid1, kIceUfrag3, kIcePwd3,
>+                  cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
>+                  nullptr);
>+
>+  auto remote_answer = absl::make_unique<cricket::SessionDescription>();
>+  AddAudioSection(remote_answer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
>+                  cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
>+                  nullptr);
>+  AddAudioSection(remote_answer.get(), kAudioMid2, kIceUfrag2, kIcePwd2,
>+                  cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
>+                  nullptr);
>+  AddVideoSection(remote_answer.get(), kVideoMid1, kIceUfrag3, kIcePwd3,
>+                  cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
>+                  nullptr);
>+
>+  local_offer->AddGroup(bundle_group);
>+  remote_answer->AddGroup(bundle_group);
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetLocalDescription(SdpType::kOffer, local_offer.get())
>+                  .ok());
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
>+                  .ok());
>+
>+  // Verifiy that only |kAudio1| and |kVideo1| are bundled.
>+  auto transport1 = transport_controller_->GetRtpTransport(kAudioMid1);
>+  auto transport2 = transport_controller_->GetRtpTransport(kAudioMid2);
>+  auto transport3 = transport_controller_->GetRtpTransport(kVideoMid1);
>+  EXPECT_NE(transport1, transport2);
>+  EXPECT_EQ(transport1, transport3);
>+
>+  auto it = changed_rtp_transport_by_mid_.find(kVideoMid1);
>+  ASSERT_TRUE(it != changed_rtp_transport_by_mid_.end());
>+  EXPECT_EQ(transport1, it->second);
>+  it = changed_rtp_transport_by_mid_.find(kAudioMid2);
>+  EXPECT_TRUE(transport2 == it->second);
>+}
>+
>+// Tests that the initial offer/answer only have data section and audio/video
>+// sections are added in the subsequent offer.
>+TEST_F(JsepTransportControllerTest, BundleOnDataSectionInSubsequentOffer) {
>+  CreateJsepTransportController(JsepTransportController::Config());
>+  cricket::ContentGroup bundle_group(cricket::GROUP_TYPE_BUNDLE);
>+  bundle_group.AddContentName(kDataMid1);
>+
>+  auto local_offer = absl::make_unique<cricket::SessionDescription>();
>+  AddDataSection(local_offer.get(), kDataMid1,
>+                 cricket::MediaProtocolType::kSctp, kIceUfrag1, kIcePwd1,
>+                 cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
>+                 nullptr);
>+  auto remote_answer = absl::make_unique<cricket::SessionDescription>();
>+  AddDataSection(remote_answer.get(), kDataMid1,
>+                 cricket::MediaProtocolType::kSctp, kIceUfrag1, kIcePwd1,
>+                 cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
>+                 nullptr);
>+  local_offer->AddGroup(bundle_group);
>+  remote_answer->AddGroup(bundle_group);
>+
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetLocalDescription(SdpType::kOffer, local_offer.get())
>+                  .ok());
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
>+                  .ok());
>+  auto data_transport = transport_controller_->GetRtpTransport(kDataMid1);
>+
>+  // Add audio/video sections in subsequent offer.
>+  AddAudioSection(local_offer.get(), kAudioMid1, kIceUfrag2, kIcePwd2,
>+                  cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
>+                  nullptr);
>+  AddVideoSection(local_offer.get(), kVideoMid1, kIceUfrag3, kIcePwd3,
>+                  cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
>+                  nullptr);
>+  AddAudioSection(remote_answer.get(), kAudioMid1, kIceUfrag2, kIcePwd2,
>+                  cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
>+                  nullptr);
>+  AddVideoSection(remote_answer.get(), kVideoMid1, kIceUfrag3, kIcePwd3,
>+                  cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
>+                  nullptr);
>+
>+  // Reset the bundle group and do another offer/answer exchange.
>+  bundle_group.AddContentName(kAudioMid1);
>+  bundle_group.AddContentName(kVideoMid1);
>+  local_offer->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE);
>+  remote_answer->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE);
>+  local_offer->AddGroup(bundle_group);
>+  remote_answer->AddGroup(bundle_group);
>+
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetLocalDescription(SdpType::kOffer, local_offer.get())
>+                  .ok());
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
>+                  .ok());
>+
>+  auto audio_transport = transport_controller_->GetRtpTransport(kAudioMid1);
>+  auto video_transport = transport_controller_->GetRtpTransport(kVideoMid1);
>+  EXPECT_EQ(data_transport, audio_transport);
>+  EXPECT_EQ(data_transport, video_transport);
>+}
>+
>+TEST_F(JsepTransportControllerTest, VideoDataRejectedInAnswer) {
>+  CreateJsepTransportController(JsepTransportController::Config());
>+  cricket::ContentGroup bundle_group(cricket::GROUP_TYPE_BUNDLE);
>+  bundle_group.AddContentName(kAudioMid1);
>+  bundle_group.AddContentName(kVideoMid1);
>+  bundle_group.AddContentName(kDataMid1);
>+
>+  auto local_offer = absl::make_unique<cricket::SessionDescription>();
>+  AddAudioSection(local_offer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
>+                  cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
>+                  nullptr);
>+  AddVideoSection(local_offer.get(), kVideoMid1, kIceUfrag2, kIcePwd2,
>+                  cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
>+                  nullptr);
>+  AddDataSection(local_offer.get(), kDataMid1,
>+                 cricket::MediaProtocolType::kSctp, kIceUfrag3, kIcePwd3,
>+                 cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
>+                 nullptr);
>+
>+  auto remote_answer = absl::make_unique<cricket::SessionDescription>();
>+  AddAudioSection(remote_answer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
>+                  cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
>+                  nullptr);
>+  AddVideoSection(remote_answer.get(), kVideoMid1, kIceUfrag2, kIcePwd2,
>+                  cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
>+                  nullptr);
>+  AddDataSection(remote_answer.get(), kDataMid1,
>+                 cricket::MediaProtocolType::kSctp, kIceUfrag3, kIcePwd3,
>+                 cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
>+                 nullptr);
>+  // Reject video and data section.
>+  remote_answer->contents()[1].rejected = true;
>+  remote_answer->contents()[2].rejected = true;
>+
>+  local_offer->AddGroup(bundle_group);
>+  remote_answer->AddGroup(bundle_group);
>+
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetLocalDescription(SdpType::kOffer, local_offer.get())
>+                  .ok());
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
>+                  .ok());
>+
>+  // Verify the RtpTransport/DtlsTransport is destroyed correctly.
>+  EXPECT_EQ(nullptr, transport_controller_->GetRtpTransport(kVideoMid1));
>+  EXPECT_EQ(nullptr, transport_controller_->GetDtlsTransport(kDataMid1));
>+  // Verify the signals are fired correctly.
>+  auto it = changed_rtp_transport_by_mid_.find(kVideoMid1);
>+  ASSERT_TRUE(it != changed_rtp_transport_by_mid_.end());
>+  EXPECT_EQ(nullptr, it->second);
>+  auto it2 = changed_dtls_transport_by_mid_.find(kDataMid1);
>+  ASSERT_TRUE(it2 != changed_dtls_transport_by_mid_.end());
>+  EXPECT_EQ(nullptr, it2->second);
>+}
>+
>+// Tests that changing the bundled MID in subsequent offer/answer exchange is
>+// not supported.
>+// TODO(bugs.webrtc.org/6704): Change this test to expect success once issue is
>+// fixed
>+TEST_F(JsepTransportControllerTest, ChangeBundledMidNotSupported) {
>+  CreateJsepTransportController(JsepTransportController::Config());
>+  cricket::ContentGroup bundle_group(cricket::GROUP_TYPE_BUNDLE);
>+  bundle_group.AddContentName(kAudioMid1);
>+  bundle_group.AddContentName(kVideoMid1);
>+
>+  auto local_offer = absl::make_unique<cricket::SessionDescription>();
>+  AddAudioSection(local_offer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
>+                  cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
>+                  nullptr);
>+  AddVideoSection(local_offer.get(), kVideoMid1, kIceUfrag2, kIcePwd2,
>+                  cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
>+                  nullptr);
>+
>+  auto remote_answer = absl::make_unique<cricket::SessionDescription>();
>+  AddAudioSection(remote_answer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
>+                  cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
>+                  nullptr);
>+  AddVideoSection(remote_answer.get(), kVideoMid1, kIceUfrag2, kIcePwd2,
>+                  cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
>+                  nullptr);
>+
>+  local_offer->AddGroup(bundle_group);
>+  remote_answer->AddGroup(bundle_group);
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetLocalDescription(SdpType::kOffer, local_offer.get())
>+                  .ok());
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
>+                  .ok());
>+  EXPECT_EQ(transport_controller_->GetRtpTransport(kAudioMid1),
>+            transport_controller_->GetRtpTransport(kVideoMid1));
>+
>+  // Reorder the bundle group.
>+  EXPECT_TRUE(bundle_group.RemoveContentName(kAudioMid1));
>+  bundle_group.AddContentName(kAudioMid1);
>+  // The answerer uses the new bundle group and now the bundle mid is changed to
>+  // |kVideo1|.
>+  remote_answer->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE);
>+  remote_answer->AddGroup(bundle_group);
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetLocalDescription(SdpType::kOffer, local_offer.get())
>+                  .ok());
>+  EXPECT_FALSE(transport_controller_
>+                   ->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
>+                   .ok());
>+}
>+// Test that rejecting only the first m= section of a BUNDLE group is treated as
>+// an error, but rejecting all of them works as expected.
>+TEST_F(JsepTransportControllerTest, RejectFirstContentInBundleGroup) {
>+  CreateJsepTransportController(JsepTransportController::Config());
>+  cricket::ContentGroup bundle_group(cricket::GROUP_TYPE_BUNDLE);
>+  bundle_group.AddContentName(kAudioMid1);
>+  bundle_group.AddContentName(kVideoMid1);
>+  bundle_group.AddContentName(kDataMid1);
>+
>+  auto local_offer = absl::make_unique<cricket::SessionDescription>();
>+  AddAudioSection(local_offer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
>+                  cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
>+                  nullptr);
>+  AddVideoSection(local_offer.get(), kVideoMid1, kIceUfrag2, kIcePwd2,
>+                  cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
>+                  nullptr);
>+  AddDataSection(local_offer.get(), kDataMid1,
>+                 cricket::MediaProtocolType::kSctp, kIceUfrag3, kIcePwd3,
>+                 cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
>+                 nullptr);
>+
>+  auto remote_answer = absl::make_unique<cricket::SessionDescription>();
>+  AddAudioSection(remote_answer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
>+                  cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
>+                  nullptr);
>+  AddVideoSection(remote_answer.get(), kVideoMid1, kIceUfrag2, kIcePwd2,
>+                  cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
>+                  nullptr);
>+  AddDataSection(remote_answer.get(), kDataMid1,
>+                 cricket::MediaProtocolType::kSctp, kIceUfrag3, kIcePwd3,
>+                 cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
>+                 nullptr);
>+  // Reject audio content in answer.
>+  remote_answer->contents()[0].rejected = true;
>+
>+  local_offer->AddGroup(bundle_group);
>+  remote_answer->AddGroup(bundle_group);
>+
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetLocalDescription(SdpType::kOffer, local_offer.get())
>+                  .ok());
>+  EXPECT_FALSE(transport_controller_
>+                   ->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
>+                   .ok());
>+
>+  // Reject all the contents.
>+  remote_answer->contents()[1].rejected = true;
>+  remote_answer->contents()[2].rejected = true;
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
>+                  .ok());
>+  EXPECT_EQ(nullptr, transport_controller_->GetRtpTransport(kAudioMid1));
>+  EXPECT_EQ(nullptr, transport_controller_->GetRtpTransport(kVideoMid1));
>+  EXPECT_EQ(nullptr, transport_controller_->GetDtlsTransport(kDataMid1));
>+}
>+
>+// Tests that applying non-RTCP-mux offer would fail when kRtcpMuxPolicyRequire
>+// is used.
>+TEST_F(JsepTransportControllerTest, ApplyNonRtcpMuxOfferWhenMuxingRequired) {
>+  JsepTransportController::Config config;
>+  config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
>+  CreateJsepTransportController(config);
>+  auto local_offer = absl::make_unique<cricket::SessionDescription>();
>+  AddAudioSection(local_offer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
>+                  cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
>+                  nullptr);
>+
>+  local_offer->contents()[0].media_description()->set_rtcp_mux(false);
>+  // Applying a non-RTCP-mux offer is expected to fail.
>+  EXPECT_FALSE(transport_controller_
>+                   ->SetLocalDescription(SdpType::kOffer, local_offer.get())
>+                   .ok());
>+}
>+
>+// Tests that applying non-RTCP-mux answer would fail when kRtcpMuxPolicyRequire
>+// is used.
>+TEST_F(JsepTransportControllerTest, ApplyNonRtcpMuxAnswerWhenMuxingRequired) {
>+  JsepTransportController::Config config;
>+  config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyRequire;
>+  CreateJsepTransportController(config);
>+  auto local_offer = absl::make_unique<cricket::SessionDescription>();
>+  AddAudioSection(local_offer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
>+                  cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_ACTPASS,
>+                  nullptr);
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetLocalDescription(SdpType::kOffer, local_offer.get())
>+                  .ok());
>+
>+  auto remote_answer = absl::make_unique<cricket::SessionDescription>();
>+  AddAudioSection(remote_answer.get(), kAudioMid1, kIceUfrag1, kIcePwd1,
>+                  cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_PASSIVE,
>+                  nullptr);
>+  // Applying a non-RTCP-mux answer is expected to fail.
>+  remote_answer->contents()[0].media_description()->set_rtcp_mux(false);
>+  EXPECT_FALSE(transport_controller_
>+                   ->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
>+                   .ok());
>+}
>+
>+// This tests that the BUNDLE group in answer should be a subset of the offered
>+// group.
>+TEST_F(JsepTransportControllerTest,
>+       AddContentToBundleGroupInAnswerNotSupported) {
>+  CreateJsepTransportController(JsepTransportController::Config());
>+  auto local_offer = CreateSessionDescriptionWithoutBundle();
>+  auto remote_answer = CreateSessionDescriptionWithoutBundle();
>+
>+  cricket::ContentGroup offer_bundle_group(cricket::GROUP_TYPE_BUNDLE);
>+  offer_bundle_group.AddContentName(kAudioMid1);
>+  local_offer->AddGroup(offer_bundle_group);
>+
>+  cricket::ContentGroup answer_bundle_group(cricket::GROUP_TYPE_BUNDLE);
>+  answer_bundle_group.AddContentName(kAudioMid1);
>+  answer_bundle_group.AddContentName(kVideoMid1);
>+  remote_answer->AddGroup(answer_bundle_group);
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetLocalDescription(SdpType::kOffer, local_offer.get())
>+                  .ok());
>+  EXPECT_FALSE(transport_controller_
>+                   ->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
>+                   .ok());
>+}
>+
>+// This tests that the BUNDLE group with non-existing MID should be rejectd.
>+TEST_F(JsepTransportControllerTest, RejectBundleGroupWithNonExistingMid) {
>+  CreateJsepTransportController(JsepTransportController::Config());
>+  auto local_offer = CreateSessionDescriptionWithoutBundle();
>+  auto remote_answer = CreateSessionDescriptionWithoutBundle();
>+
>+  cricket::ContentGroup invalid_bundle_group(cricket::GROUP_TYPE_BUNDLE);
>+  // The BUNDLE group is invalid because there is no data section in the
>+  // description.
>+  invalid_bundle_group.AddContentName(kDataMid1);
>+  local_offer->AddGroup(invalid_bundle_group);
>+  remote_answer->AddGroup(invalid_bundle_group);
>+
>+  EXPECT_FALSE(transport_controller_
>+                   ->SetLocalDescription(SdpType::kOffer, local_offer.get())
>+                   .ok());
>+  EXPECT_FALSE(transport_controller_
>+                   ->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
>+                   .ok());
>+}
>+
>+// This tests that an answer shouldn't be able to remove an m= section from an
>+// established group without rejecting it.
>+TEST_F(JsepTransportControllerTest, RemoveContentFromBundleGroup) {
>+  CreateJsepTransportController(JsepTransportController::Config());
>+
>+  auto local_offer = CreateSessionDescriptionWithBundleGroup();
>+  auto remote_answer = CreateSessionDescriptionWithBundleGroup();
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetLocalDescription(SdpType::kOffer, local_offer.get())
>+                  .ok());
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetRemoteDescription(SdpType::kAnswer, remote_answer.get())
>+                  .ok());
>+
>+  // Do an re-offer/answer.
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetLocalDescription(SdpType::kOffer, local_offer.get())
>+                  .ok());
>+  auto new_answer = CreateSessionDescriptionWithoutBundle();
>+  cricket::ContentGroup new_bundle_group(cricket::GROUP_TYPE_BUNDLE);
>+  //  The answer removes video from the BUNDLE group without rejecting it is
>+  //  invalid.
>+  new_bundle_group.AddContentName(kAudioMid1);
>+  new_answer->AddGroup(new_bundle_group);
>+
>+  // Applying invalid answer is expected to fail.
>+  EXPECT_FALSE(transport_controller_
>+                   ->SetRemoteDescription(SdpType::kAnswer, new_answer.get())
>+                   .ok());
>+
>+  // Rejected the video content.
>+  auto video_content = new_answer->GetContentByName(kVideoMid1);
>+  ASSERT_TRUE(video_content);
>+  video_content->rejected = true;
>+  EXPECT_TRUE(transport_controller_
>+                  ->SetRemoteDescription(SdpType::kAnswer, new_answer.get())
>+                  .ok());
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/localaudiosource.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/localaudiosource.cc
>index 5c86dd30f96..b8dbfaab1cd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/localaudiosource.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/localaudiosource.cc
>@@ -12,22 +12,12 @@
> 
> #include <vector>
> 
>-#include "api/mediaconstraintsinterface.h"
> #include "media/base/mediaengine.h"
> 
>-using webrtc::MediaConstraintsInterface;
> using webrtc::MediaSourceInterface;
> 
> namespace webrtc {
> 
>-rtc::scoped_refptr<LocalAudioSource> LocalAudioSource::Create(
>-    const MediaConstraintsInterface* constraints) {
>-  rtc::scoped_refptr<LocalAudioSource> source(
>-      new rtc::RefCountedObject<LocalAudioSource>());
>-  source->Initialize(constraints);
>-  return source;
>-}
>-
> rtc::scoped_refptr<LocalAudioSource> LocalAudioSource::Create(
>     const cricket::AudioOptions* audio_options) {
>   rtc::scoped_refptr<LocalAudioSource> source(
>@@ -36,13 +26,7 @@ rtc::scoped_refptr<LocalAudioSource> LocalAudioSource::Create(
>   return source;
> }
> 
>-void LocalAudioSource::Initialize(
>-    const MediaConstraintsInterface* constraints) {
>-  CopyConstraintsIntoAudioOptions(constraints, &options_);
>-}
>-
>-void LocalAudioSource::Initialize(
>-    const cricket::AudioOptions* audio_options) {
>+void LocalAudioSource::Initialize(const cricket::AudioOptions* audio_options) {
>   if (!audio_options)
>     return;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/localaudiosource.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/localaudiosource.h
>index cd17330e1fd..c5f65304d0b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/localaudiosource.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/localaudiosource.h
>@@ -20,14 +20,9 @@
> 
> namespace webrtc {
> 
>-class MediaConstraintsInterface;
>-
> class LocalAudioSource : public Notifier<AudioSourceInterface> {
>  public:
>   // Creates an instance of LocalAudioSource.
>-  static rtc::scoped_refptr<LocalAudioSource> Create(
>-      const MediaConstraintsInterface* constraints);
>-
>   static rtc::scoped_refptr<LocalAudioSource> Create(
>       const cricket::AudioOptions* audio_options);
> 
>@@ -44,7 +39,6 @@ class LocalAudioSource : public Notifier<AudioSourceInterface> {
>   ~LocalAudioSource() override {}
> 
>  private:
>-  void Initialize(const MediaConstraintsInterface* constraints);
>   void Initialize(const cricket::AudioOptions* audio_options);
> 
>   cricket::AudioOptions options_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/localaudiosource_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/localaudiosource_unittest.cc
>index bf566c627e2..54766caf396 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/localaudiosource_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/localaudiosource_unittest.cc
>@@ -22,75 +22,6 @@ using webrtc::LocalAudioSource;
> using webrtc::MediaConstraintsInterface;
> using webrtc::MediaSourceInterface;
> 
>-TEST(LocalAudioSourceTest, SetValidOptions) {
>-  webrtc::FakeConstraints constraints;
>-  constraints.AddMandatory(
>-      MediaConstraintsInterface::kGoogEchoCancellation, false);
>-  constraints.AddOptional(
>-      MediaConstraintsInterface::kExtendedFilterEchoCancellation, true);
>-  constraints.AddOptional(MediaConstraintsInterface::kDAEchoCancellation, true);
>-  constraints.AddOptional(MediaConstraintsInterface::kAutoGainControl, true);
>-  constraints.AddOptional(
>-      MediaConstraintsInterface::kExperimentalAutoGainControl, true);
>-  constraints.AddMandatory(MediaConstraintsInterface::kNoiseSuppression, false);
>-  constraints.AddOptional(MediaConstraintsInterface::kHighpassFilter, true);
>-
>-  rtc::scoped_refptr<LocalAudioSource> source =
>-      LocalAudioSource::Create(&constraints);
>-
>-  EXPECT_EQ(false, source->options().echo_cancellation);
>-  EXPECT_EQ(true, source->options().extended_filter_aec);
>-  EXPECT_EQ(true, source->options().delay_agnostic_aec);
>-  EXPECT_EQ(true, source->options().auto_gain_control);
>-  EXPECT_EQ(true, source->options().experimental_agc);
>-  EXPECT_EQ(false, source->options().noise_suppression);
>-  EXPECT_EQ(true, source->options().highpass_filter);
>-}
>-
>-TEST(LocalAudioSourceTest, OptionNotSet) {
>-  webrtc::FakeConstraints constraints;
>-  rtc::scoped_refptr<LocalAudioSource> source =
>-      LocalAudioSource::Create(&constraints);
>-  EXPECT_EQ(rtc::nullopt, source->options().highpass_filter);
>-}
>-
>-TEST(LocalAudioSourceTest, MandatoryOverridesOptional) {
>-  webrtc::FakeConstraints constraints;
>-  constraints.AddMandatory(
>-      MediaConstraintsInterface::kGoogEchoCancellation, false);
>-  constraints.AddOptional(
>-      MediaConstraintsInterface::kGoogEchoCancellation, true);
>-
>-  rtc::scoped_refptr<LocalAudioSource> source =
>-      LocalAudioSource::Create(&constraints);
>-
>-  EXPECT_EQ(false, source->options().echo_cancellation);
>-}
>-
>-TEST(LocalAudioSourceTest, InvalidOptional) {
>-  webrtc::FakeConstraints constraints;
>-  constraints.AddOptional(MediaConstraintsInterface::kHighpassFilter, false);
>-  constraints.AddOptional("invalidKey", false);
>-
>-  rtc::scoped_refptr<LocalAudioSource> source =
>-      LocalAudioSource::Create(&constraints);
>-
>-  EXPECT_EQ(MediaSourceInterface::kLive, source->state());
>-  EXPECT_EQ(false, source->options().highpass_filter);
>-}
>-
>-TEST(LocalAudioSourceTest, InvalidMandatory) {
>-  webrtc::FakeConstraints constraints;
>-  constraints.AddMandatory(MediaConstraintsInterface::kHighpassFilter, false);
>-  constraints.AddMandatory("invalidKey", false);
>-
>-  rtc::scoped_refptr<LocalAudioSource> source =
>-      LocalAudioSource::Create(&constraints);
>-
>-  EXPECT_EQ(MediaSourceInterface::kLive, source->state());
>-  EXPECT_EQ(false, source->options().highpass_filter);
>-}
>-
> TEST(LocalAudioSourceTest, InitWithAudioOptions) {
>   cricket::AudioOptions audio_options;
>   audio_options.highpass_filter = true;
>@@ -101,6 +32,6 @@ TEST(LocalAudioSourceTest, InitWithAudioOptions) {
> 
> TEST(LocalAudioSourceTest, InitWithNoOptions) {
>   rtc::scoped_refptr<LocalAudioSource> source =
>-      LocalAudioSource::Create((cricket::AudioOptions*)nullptr);
>-  EXPECT_EQ(rtc::nullopt, source->options().highpass_filter);
>+      LocalAudioSource::Create(nullptr);
>+  EXPECT_EQ(absl::nullopt, source->options().highpass_filter);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediaconstraintsinterface_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediaconstraintsinterface_unittest.cc
>index 882d09748ed..cf6716b7376 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediaconstraintsinterface_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediaconstraintsinterface_unittest.cc
>@@ -28,13 +28,7 @@ bool Matches(const PeerConnectionInterface::RTCConfiguration& a,
>          a.screencast_min_bitrate == b.screencast_min_bitrate &&
>          a.combined_audio_video_bwe == b.combined_audio_video_bwe &&
>          a.enable_dtls_srtp == b.enable_dtls_srtp &&
>-         a.media_config.enable_dscp == b.media_config.enable_dscp &&
>-         a.media_config.video.enable_cpu_overuse_detection ==
>-             b.media_config.video.enable_cpu_overuse_detection &&
>-         a.media_config.video.disable_prerenderer_smoothing ==
>-             b.media_config.video.disable_prerenderer_smoothing &&
>-         a.media_config.video.suspend_below_min_bitrate ==
>-             b.media_config.video.suspend_below_min_bitrate;
>+         a.media_config == b.media_config;
> }
> 
> TEST(MediaConstraintsInterface, CopyConstraintsIntoRtcConfiguration) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediamonitor.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediamonitor.cc
>deleted file mode 100644
>index e2e3da8b209..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediamonitor.cc
>+++ /dev/null
>@@ -1,91 +0,0 @@
>-/*
>- *  Copyright 2005 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "pc/mediamonitor.h"
>-#include "pc/channelmanager.h"
>-#include "rtc_base/checks.h"
>-
>-namespace cricket {
>-
>-enum {
>-  MSG_MONITOR_POLL = 1,
>-  MSG_MONITOR_START = 2,
>-  MSG_MONITOR_STOP = 3,
>-  MSG_MONITOR_SIGNAL = 4
>-};
>-
>-MediaMonitor::MediaMonitor(rtc::Thread* worker_thread,
>-                           rtc::Thread* monitor_thread)
>-    : worker_thread_(worker_thread),
>-      monitor_thread_(monitor_thread), monitoring_(false), rate_(0) {
>-}
>-
>-MediaMonitor::~MediaMonitor() {
>-  monitoring_ = false;
>-  monitor_thread_->Clear(this);
>-  worker_thread_->Clear(this);
>-}
>-
>-void MediaMonitor::Start(uint32_t milliseconds) {
>-  rate_ = milliseconds;
>-  if (rate_ < 100)
>-    rate_ = 100;
>-  worker_thread_->Post(RTC_FROM_HERE, this, MSG_MONITOR_START);
>-}
>-
>-void MediaMonitor::Stop() {
>-  worker_thread_->Post(RTC_FROM_HERE, this, MSG_MONITOR_STOP);
>-  rate_ = 0;
>-}
>-
>-void MediaMonitor::OnMessage(rtc::Message* message) {
>-  rtc::CritScope cs(&crit_);
>-
>-  switch (message->message_id) {
>-  case MSG_MONITOR_START:
>-    RTC_DCHECK(rtc::Thread::Current() == worker_thread_);
>-    if (!monitoring_) {
>-      monitoring_ = true;
>-      PollMediaChannel();
>-    }
>-    break;
>-
>-  case MSG_MONITOR_STOP:
>-    RTC_DCHECK(rtc::Thread::Current() == worker_thread_);
>-    if (monitoring_) {
>-      monitoring_ = false;
>-      worker_thread_->Clear(this);
>-    }
>-    break;
>-
>-  case MSG_MONITOR_POLL:
>-    RTC_DCHECK(rtc::Thread::Current() == worker_thread_);
>-    PollMediaChannel();
>-    break;
>-
>-  case MSG_MONITOR_SIGNAL:
>-    RTC_DCHECK(rtc::Thread::Current() == monitor_thread_);
>-    Update();
>-    break;
>-  }
>-}
>-
>-void MediaMonitor::PollMediaChannel() {
>-  rtc::CritScope cs(&crit_);
>-  RTC_DCHECK(rtc::Thread::Current() == worker_thread_);
>-
>-  GetStats();
>-
>-  // Signal the monitoring thread, start another poll timer
>-  monitor_thread_->Post(RTC_FROM_HERE, this, MSG_MONITOR_SIGNAL);
>-  worker_thread_->PostDelayed(RTC_FROM_HERE, rate_, this, MSG_MONITOR_POLL);
>-}
>-
>-}  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediamonitor.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediamonitor.h
>deleted file mode 100644
>index fcd24106aac..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediamonitor.h
>+++ /dev/null
>@@ -1,82 +0,0 @@
>-/*
>- *  Copyright 2005 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-// Class to collect statistics from a media channel
>-
>-#ifndef PC_MEDIAMONITOR_H_
>-#define PC_MEDIAMONITOR_H_
>-
>-#include "media/base/mediachannel.h"
>-#include "rtc_base/criticalsection.h"
>-#include "rtc_base/sigslot.h"
>-#include "rtc_base/thread.h"
>-#include "rtc_base/thread_annotations.h"
>-
>-namespace cricket {
>-
>-// The base MediaMonitor class, independent of voice and video.
>-class MediaMonitor : public rtc::MessageHandler,
>-    public sigslot::has_slots<> {
>- public:
>-  MediaMonitor(rtc::Thread* worker_thread,
>-               rtc::Thread* monitor_thread);
>-  ~MediaMonitor();
>-
>-  void Start(uint32_t milliseconds);
>-  void Stop();
>-
>- protected:
>-  void OnMessage(rtc::Message *message);
>-  void PollMediaChannel();
>-  virtual void GetStats() = 0;
>-  virtual void Update() = 0;
>-
>-  rtc::CriticalSection crit_;
>-  rtc::Thread* worker_thread_;
>-  rtc::Thread* monitor_thread_;
>-  bool monitoring_;
>-  uint32_t rate_;
>-};
>-
>-// Templatized MediaMonitor that can deal with different kinds of media.
>-template<class MC, class MI>
>-class MediaMonitorT : public MediaMonitor {
>- public:
>-  MediaMonitorT(MC* media_channel, rtc::Thread* worker_thread,
>-                rtc::Thread* monitor_thread)
>-      : MediaMonitor(worker_thread, monitor_thread),
>-        media_channel_(media_channel) {}
>-  sigslot::signal2<MC*, const MI&> SignalUpdate;
>-
>- protected:
>-  // These routines assume the crit_ lock is held by the calling thread.
>-  virtual void GetStats() {
>-    media_info_.Clear();
>-    media_channel_->GetStats(&media_info_);
>-  }
>-  virtual void Update() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_) {
>-    MI stats(media_info_);
>-    crit_.Leave();
>-    SignalUpdate(media_channel_, stats);
>-    crit_.Enter();
>-  }
>-
>- private:
>-  MC* media_channel_;
>-  MI media_info_;
>-};
>-
>-typedef MediaMonitorT<VoiceMediaChannel, VoiceMediaInfo> VoiceMediaMonitor;
>-typedef MediaMonitorT<VideoMediaChannel, VideoMediaInfo> VideoMediaMonitor;
>-typedef MediaMonitorT<DataMediaChannel, DataMediaInfo> DataMediaMonitor;
>-
>-}  // namespace cricket
>-
>-#endif  // PC_MEDIAMONITOR_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediasession.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediasession.cc
>index f53785df932..23706be6df3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediasession.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediasession.cc
>@@ -18,8 +18,8 @@
> #include <unordered_map>
> #include <utility>
> 
>+#include "absl/types/optional.h"
> #include "api/cryptoparams.h"
>-#include "api/optional.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "media/base/h264_profile_level_id.h"
> #include "media/base/mediaconstants.h"
>@@ -27,11 +27,11 @@
> #include "pc/channelmanager.h"
> #include "pc/rtpmediautils.h"
> #include "pc/srtpfilter.h"
>-#include "rtc_base/base64.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/helpers.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/stringutils.h"
>+#include "rtc_base/third_party/base64/base64.h"
> 
> namespace {
> 
>@@ -120,12 +120,13 @@ static bool IsMediaContentOfType(const ContentInfo* content,
>   return content->media_description()->type() == media_type;
> }
> 
>-static bool CreateCryptoParams(int tag, const std::string& cipher,
>-                               CryptoParams *out) {
>+static bool CreateCryptoParams(int tag,
>+                               const std::string& cipher,
>+                               CryptoParams* out) {
>   int key_len;
>   int salt_len;
>-  if (!rtc::GetSrtpKeyAndSaltLengths(
>-      rtc::SrtpCryptoSuiteFromName(cipher), &key_len, &salt_len)) {
>+  if (!rtc::GetSrtpKeyAndSaltLengths(rtc::SrtpCryptoSuiteFromName(cipher),
>+                                     &key_len, &salt_len)) {
>     return false;
>   }
> 
>@@ -146,7 +147,7 @@ static bool CreateCryptoParams(int tag, const std::string& cipher,
> }
> 
> static bool AddCryptoParams(const std::string& cipher_suite,
>-                            CryptoParamsVec *out) {
>+                            CryptoParamsVec* out) {
>   int size = static_cast<int>(out->size());
> 
>   out->resize(size + 1);
>@@ -194,14 +195,17 @@ bool FindMatchingCrypto(const CryptoParamsVec& cryptos,
>   return false;
> }
> 
>-// For audio, HMAC 32 is prefered over HMAC 80 because of the low overhead.
>+// For audio, HMAC 32 (if enabled) is prefered over HMAC 80 because of the
>+// low overhead.
> void GetSupportedAudioSdesCryptoSuites(const rtc::CryptoOptions& crypto_options,
>                                        std::vector<int>* crypto_suites) {
>   if (crypto_options.enable_gcm_crypto_suites) {
>     crypto_suites->push_back(rtc::SRTP_AEAD_AES_256_GCM);
>     crypto_suites->push_back(rtc::SRTP_AEAD_AES_128_GCM);
>   }
>-  crypto_suites->push_back(rtc::SRTP_AES128_CM_SHA1_32);
>+  if (crypto_options.enable_aes128_sha1_32_crypto_cipher) {
>+    crypto_suites->push_back(rtc::SRTP_AES128_CM_SHA1_32);
>+  }
>   crypto_suites->push_back(rtc::SRTP_AES128_CM_SHA1_80);
> }
> 
>@@ -245,23 +249,23 @@ void GetSupportedDataSdesCryptoSuiteNames(
> }
> 
> // Support any GCM cipher (if enabled through options). For video support only
>-// 80-bit SHA1 HMAC. For audio 32-bit HMAC is tolerated unless bundle is enabled
>-// because it is low overhead.
>+// 80-bit SHA1 HMAC. For audio 32-bit HMAC is tolerated (if enabled) unless
>+// bundle is enabled because it is low overhead.
> // Pick the crypto in the list that is supported.
> static bool SelectCrypto(const MediaContentDescription* offer,
>                          bool bundle,
>                          const rtc::CryptoOptions& crypto_options,
>-                         CryptoParams *crypto) {
>+                         CryptoParams* crypto) {
>   bool audio = offer->type() == MEDIA_TYPE_AUDIO;
>   const CryptoParamsVec& cryptos = offer->cryptos();
> 
>-  for (CryptoParamsVec::const_iterator i = cryptos.begin();
>-       i != cryptos.end(); ++i) {
>+  for (CryptoParamsVec::const_iterator i = cryptos.begin(); i != cryptos.end();
>+       ++i) {
>     if ((crypto_options.enable_gcm_crypto_suites &&
>          rtc::IsGcmCryptoSuiteName(i->cipher_suite)) ||
>         rtc::CS_AES_CM_128_HMAC_SHA1_80 == i->cipher_suite ||
>         (rtc::CS_AES_CM_128_HMAC_SHA1_32 == i->cipher_suite && audio &&
>-         !bundle)) {
>+         !bundle && crypto_options.enable_aes128_sha1_32_crypto_cipher)) {
>       return CreateCryptoParams(i->tag, i->cipher_suite, crypto);
>     }
>   }
>@@ -322,8 +326,7 @@ class UsedIds {
>   UsedIds(int min_allowed_id, int max_allowed_id)
>       : min_allowed_id_(min_allowed_id),
>         max_allowed_id_(max_allowed_id),
>-        next_id_(max_allowed_id) {
>-  }
>+        next_id_(max_allowed_id) {}
> 
>   // Loops through all Id in |ids| and changes its id if it is
>   // already in use by another IdStruct. Call this methods with all Id
>@@ -331,8 +334,8 @@ class UsedIds {
>   // Note that typename Id must be a type of IdStruct.
>   template <typename Id>
>   void FindAndSetIdUsed(std::vector<Id>* ids) {
>-    for (typename std::vector<Id>::iterator it = ids->begin();
>-         it != ids->end(); ++it) {
>+    for (typename std::vector<Id>::iterator it = ids->begin(); it != ids->end();
>+         ++it) {
>       FindAndSetIdUsed(&*it);
>     }
>   }
>@@ -369,13 +372,9 @@ class UsedIds {
>     return next_id_;
>   }
> 
>-  bool IsIdUsed(int new_id) {
>-    return id_set_.find(new_id) != id_set_.end();
>-  }
>+  bool IsIdUsed(int new_id) { return id_set_.find(new_id) != id_set_.end(); }
> 
>-  void SetIdUsed(int new_id) {
>-    id_set_.insert(new_id);
>-  }
>+  void SetIdUsed(int new_id) { id_set_.insert(new_id); }
> 
>   const int min_allowed_id_;
>   const int max_allowed_id_;
>@@ -388,9 +387,7 @@ class UsedIds {
> class UsedPayloadTypes : public UsedIds<Codec> {
>  public:
>   UsedPayloadTypes()
>-      : UsedIds<Codec>(kDynamicPayloadTypeMin, kDynamicPayloadTypeMax) {
>-  }
>-
>+      : UsedIds<Codec>(kDynamicPayloadTypeMin, kDynamicPayloadTypeMax) {}
> 
>  private:
>   static const int kDynamicPayloadTypeMin = 96;
>@@ -425,7 +422,6 @@ static bool AddStreamParams(
>   const bool include_rtx_streams =
>       ContainsRtxCodec(content_description->codecs());
> 
>-
>   const bool include_flexfec_stream =
>       ContainsFlexfecCodec(content_description->codecs());
> 
>@@ -468,14 +464,12 @@ static bool AddStreamParams(
>         } else if (!ssrcs.empty()) {
>           RTC_LOG(LS_WARNING)
>               << "Our FlexFEC implementation only supports protecting "
>-              << "a single media streams. This session has multiple "
>-              << "media streams however, so no FlexFEC SSRC will be generated.";
>+                 "a single media streams. This session has multiple "
>+                 "media streams however, so no FlexFEC SSRC will be generated.";
>         }
>       }
>       stream_param.cname = rtcp_cname;
>-      // TODO(steveanton): Support any number of stream ids.
>-      RTC_CHECK(sender.stream_ids.size() == 1U);
>-      stream_param.sync_label = sender.stream_ids[0];
>+      stream_param.set_stream_ids(sender.stream_ids);
>       content_description->AddStream(stream_param);
> 
>       // Store the new StreamParams in current_streams.
>@@ -485,9 +479,7 @@ static bool AddStreamParams(
>       // Use existing generated SSRCs/groups, but update the sync_label if
>       // necessary. This may be needed if a MediaStreamTrack was moved from one
>       // MediaStream to another.
>-      // TODO(steveanton): Support any number of stream ids.
>-      RTC_CHECK(sender.stream_ids.size() == 1U);
>-      param->sync_label = sender.stream_ids[0];
>+      param->set_stream_ids(sender.stream_ids);
>       content_description->AddStream(*param);
>     }
>   }
>@@ -520,8 +512,7 @@ static bool UpdateTransportInfoForBundle(const ContentGroup& bundle_group,
>       selected_transport_info->description.ice_pwd;
>   ConnectionRole selected_connection_role =
>       selected_transport_info->description.connection_role;
>-  for (TransportInfos::iterator it =
>-           sdesc->transport_infos().begin();
>+  for (TransportInfos::iterator it = sdesc->transport_infos().begin();
>        it != sdesc->transport_infos().end(); ++it) {
>     if (bundle_group.HasContentName(it->content_name) &&
>         it->content_name != selected_content_name) {
>@@ -549,22 +540,6 @@ static bool GetCryptosByName(const SessionDescription* sdesc,
>   return true;
> }
> 
>-// Predicate function used by the remove_if.
>-// Returns true if the |crypto|'s cipher_suite is not found in |filter|.
>-static bool CryptoNotFound(const CryptoParams crypto,
>-                           const CryptoParamsVec* filter) {
>-  if (filter == NULL) {
>-    return true;
>-  }
>-  for (CryptoParamsVec::const_iterator it = filter->begin();
>-       it != filter->end(); ++it) {
>-    if (it->cipher_suite == crypto.cipher_suite) {
>-      return false;
>-    }
>-  }
>-  return true;
>-}
>-
> // Prunes the |target_cryptos| by removing the crypto params (cipher_suite)
> // which are not available in |filter|.
> static void PruneCryptos(const CryptoParamsVec& filter,
>@@ -572,11 +547,19 @@ static void PruneCryptos(const CryptoParamsVec& filter,
>   if (!target_cryptos) {
>     return;
>   }
>-  target_cryptos->erase(std::remove_if(target_cryptos->begin(),
>-                                       target_cryptos->end(),
>-                                       bind2nd(ptr_fun(CryptoNotFound),
>-                                               &filter)),
>-                        target_cryptos->end());
>+
>+  target_cryptos->erase(
>+      std::remove_if(target_cryptos->begin(), target_cryptos->end(),
>+                     // Returns true if the |crypto|'s cipher_suite is not
>+                     // found in |filter|.
>+                     [&filter](const CryptoParams& crypto) {
>+                       for (const CryptoParams& entry : filter) {
>+                         if (entry.cipher_suite == crypto.cipher_suite)
>+                           return false;
>+                       }
>+                       return true;
>+                     }),
>+      target_cryptos->end());
> }
> 
> bool IsRtpProtocol(const std::string& protocol) {
>@@ -940,11 +923,12 @@ static bool FindByUri(const RtpHeaderExtensions& extensions,
> 
> static bool FindByUriWithEncryptionPreference(
>     const RtpHeaderExtensions& extensions,
>-    const webrtc::RtpExtension& ext_to_match, bool encryption_preference,
>+    const webrtc::RtpExtension& ext_to_match,
>+    bool encryption_preference,
>     webrtc::RtpExtension* found_extension) {
>   const webrtc::RtpExtension* unencrypted_extension = nullptr;
>   for (RtpHeaderExtensions::const_iterator it = extensions.begin();
>-       it  != extensions.end(); ++it) {
>+       it != extensions.end(); ++it) {
>     // We assume that all URIs are given in a canonical format.
>     if (it->uri == ext_to_match.uri) {
>       if (!encryption_preference || it->encrypt) {
>@@ -1017,7 +1001,8 @@ static void AddEncryptedVersionsOfHdrExts(RtpHeaderExtensions* extensions,
>     if (extension.encrypt ||
>         !webrtc::RtpExtension::IsEncryptionSupported(extension.uri) ||
>         (FindByUriWithEncryptionPreference(*extensions, extension, true,
>-            &existing) && existing.encrypt)) {
>+                                           &existing) &&
>+         existing.encrypt)) {
>       continue;
>     }
> 
>@@ -1032,7 +1017,7 @@ static void AddEncryptedVersionsOfHdrExts(RtpHeaderExtensions* extensions,
>     }
>   }
>   extensions->insert(extensions->end(), encrypted_extensions.begin(),
>-      encrypted_extensions.end());
>+                     encrypted_extensions.end());
> }
> 
> static void NegotiateRtpHeaderExtensions(
>@@ -1041,11 +1026,12 @@ static void NegotiateRtpHeaderExtensions(
>     bool enable_encrypted_rtp_header_extensions,
>     RtpHeaderExtensions* negotiated_extenstions) {
>   RtpHeaderExtensions::const_iterator ours;
>-  for (ours = local_extensions.begin();
>-       ours != local_extensions.end(); ++ours) {
>+  for (ours = local_extensions.begin(); ours != local_extensions.end();
>+       ++ours) {
>     webrtc::RtpExtension theirs;
>-    if (FindByUriWithEncryptionPreference(offered_extensions, *ours,
>-        enable_encrypted_rtp_header_extensions, &theirs)) {
>+    if (FindByUriWithEncryptionPreference(
>+            offered_extensions, *ours, enable_encrypted_rtp_header_extensions,
>+            &theirs)) {
>       // We respond with their RTP header extension id.
>       negotiated_extenstions->push_back(theirs);
>     }
>@@ -1088,10 +1074,9 @@ static bool CreateMediaContentAnswer(
>   answer->AddCodecs(negotiated_codecs);
>   answer->set_protocol(offer->protocol());
>   RtpHeaderExtensions negotiated_rtp_extensions;
>-  NegotiateRtpHeaderExtensions(local_rtp_extenstions,
>-                               offer->rtp_header_extensions(),
>-                               enable_encrypted_rtp_header_extensions,
>-                               &negotiated_rtp_extensions);
>+  NegotiateRtpHeaderExtensions(
>+      local_rtp_extenstions, offer->rtp_header_extensions(),
>+      enable_encrypted_rtp_header_extensions, &negotiated_rtp_extensions);
>   answer->set_rtp_header_extensions(negotiated_rtp_extensions);
> 
>   answer->set_rtcp_mux(session_options.rtcp_mux_enabled && offer->rtcp_mux());
>@@ -1269,7 +1254,8 @@ const AudioCodecs& MediaSessionDescriptionFactory::audio_recv_codecs() const {
> }
> 
> void MediaSessionDescriptionFactory::set_audio_codecs(
>-    const AudioCodecs& send_codecs, const AudioCodecs& recv_codecs) {
>+    const AudioCodecs& send_codecs,
>+    const AudioCodecs& recv_codecs) {
>   audio_send_codecs_ = send_codecs;
>   audio_recv_codecs_ = recv_codecs;
>   ComputeAudioCodecsIntersectionAndUnion();
>@@ -1298,8 +1284,8 @@ SessionDescription* MediaSessionDescriptionFactory::CreateOffer(
> 
>   RtpHeaderExtensions audio_rtp_extensions;
>   RtpHeaderExtensions video_rtp_extensions;
>-  GetRtpHdrExtsToOffer(current_description, &audio_rtp_extensions,
>-                       &video_rtp_extensions);
>+  GetRtpHdrExtsToOffer(session_options, current_description,
>+                       &audio_rtp_extensions, &video_rtp_extensions);
> 
>   // Must have options for each existing section.
>   if (current_description) {
>@@ -1375,6 +1361,20 @@ SessionDescription* MediaSessionDescriptionFactory::CreateOffer(
>       return nullptr;
>     }
>   }
>+
>+  // The following determines how to signal MSIDs to ensure compatibility with
>+  // older endpoints (in particular, older Plan B endpoints).
>+  if (session_options.is_unified_plan) {
>+    // Be conservative and signal using both a=msid and a=ssrc lines. Unified
>+    // Plan answerers will look at a=msid and Plan B answerers will look at the
>+    // a=ssrc MSID line.
>+    offer->set_msid_signaling(cricket::kMsidSignalingMediaSection |
>+                              cricket::kMsidSignalingSsrcAttribute);
>+  } else {
>+    // Plan B always signals MSID using a=ssrc lines.
>+    offer->set_msid_signaling(cricket::kMsidSignalingSsrcAttribute);
>+  }
>+
>   return offer.release();
> }
> 
>@@ -1481,10 +1481,17 @@ SessionDescription* MediaSessionDescriptionFactory::CreateAnswer(
>     }
>   }
> 
>-  // Only put BUNDLE group in answer if nonempty.
>-  if (answer_bundle.FirstContentName()) {
>+  // If a BUNDLE group was offered, put a BUNDLE group in the answer even if
>+  // it's empty. RFC5888 says:
>+  //
>+  //   A SIP entity that receives an offer that contains an "a=group" line
>+  //   with semantics that are understood MUST return an answer that
>+  //   contains an "a=group" line with the same semantics.
>+  if (offer_bundle) {
>     answer->AddGroup(answer_bundle);
>+  }
> 
>+  if (answer_bundle.FirstContentName()) {
>     // Share the same ICE credentials and crypto params across all contents,
>     // as BUNDLE requires.
>     if (!UpdateTransportInfoForBundle(answer_bundle, answer.get())) {
>@@ -1500,6 +1507,39 @@ SessionDescription* MediaSessionDescriptionFactory::CreateAnswer(
>     }
>   }
> 
>+  // The following determines how to signal MSIDs to ensure compatibility with
>+  // older endpoints (in particular, older Plan B endpoints).
>+  if (session_options.is_unified_plan) {
>+    // Unified Plan needs to look at what the offer included to find the most
>+    // compatible answer.
>+    if (offer->msid_signaling() == 0) {
>+      // We end up here in one of three cases:
>+      // 1. An empty offer. We'll reply with an empty answer so it doesn't
>+      //    matter what we pick here.
>+      // 2. A data channel only offer. We won't add any MSIDs to the answer so
>+      //    it also doesn't matter what we pick here.
>+      // 3. Media that's either sendonly or inactive from the remote endpoint.
>+      //    We don't have any information to say whether the endpoint is Plan B
>+      //    or Unified Plan, so be conservative and send both.
>+      answer->set_msid_signaling(cricket::kMsidSignalingMediaSection |
>+                                 cricket::kMsidSignalingSsrcAttribute);
>+    } else if (offer->msid_signaling() ==
>+               (cricket::kMsidSignalingMediaSection |
>+                cricket::kMsidSignalingSsrcAttribute)) {
>+      // If both a=msid and a=ssrc MSID signaling methods were used, we're
>+      // probably talking to a Unified Plan endpoint so respond with just
>+      // a=msid.
>+      answer->set_msid_signaling(cricket::kMsidSignalingMediaSection);
>+    } else {
>+      // Otherwise, it's clear which method the offerer is using so repeat that
>+      // back to them.
>+      answer->set_msid_signaling(offer->msid_signaling());
>+    }
>+  } else {
>+    // Plan B always signals MSID using a=ssrc lines.
>+    answer->set_msid_signaling(cricket::kMsidSignalingSsrcAttribute);
>+  }
>+
>   return answer.release();
> }
> 
>@@ -1672,6 +1712,7 @@ void MediaSessionDescriptionFactory::GetCodecsForAnswer(
> }
> 
> void MediaSessionDescriptionFactory::GetRtpHdrExtsToOffer(
>+    const MediaSessionOptions& session_options,
>     const SessionDescription* current_description,
>     RtpHeaderExtensions* offer_audio_extensions,
>     RtpHeaderExtensions* offer_video_extensions) const {
>@@ -1707,12 +1748,12 @@ void MediaSessionDescriptionFactory::GetRtpHdrExtsToOffer(
> 
>   // Add our default RTP header extensions that are not in
>   // |current_description|.
>-  MergeRtpHdrExts(audio_rtp_header_extensions(), offer_audio_extensions,
>-                  &all_regular_extensions, &all_encrypted_extensions,
>-                  &used_ids);
>-  MergeRtpHdrExts(video_rtp_header_extensions(), offer_video_extensions,
>-                  &all_regular_extensions, &all_encrypted_extensions,
>-                  &used_ids);
>+  MergeRtpHdrExts(audio_rtp_header_extensions(session_options.is_unified_plan),
>+                  offer_audio_extensions, &all_regular_extensions,
>+                  &all_encrypted_extensions, &used_ids);
>+  MergeRtpHdrExts(video_rtp_header_extensions(session_options.is_unified_plan),
>+                  offer_video_extensions, &all_regular_extensions,
>+                  &all_encrypted_extensions, &used_ids);
> 
>   // TODO(jbauch): Support adding encrypted header extensions to existing
>   // sessions.
>@@ -1725,18 +1766,19 @@ void MediaSessionDescriptionFactory::GetRtpHdrExtsToOffer(
> }
> 
> bool MediaSessionDescriptionFactory::AddTransportOffer(
>-  const std::string& content_name,
>-  const TransportOptions& transport_options,
>-  const SessionDescription* current_desc,
>-  SessionDescription* offer_desc) const {
>+    const std::string& content_name,
>+    const TransportOptions& transport_options,
>+    const SessionDescription* current_desc,
>+    SessionDescription* offer_desc) const {
>   if (!transport_desc_factory_)
>-     return false;
>+    return false;
>   const TransportDescription* current_tdesc =
>       GetTransportDescription(content_name, current_desc);
>   std::unique_ptr<TransportDescription> new_tdesc(
>       transport_desc_factory_->CreateOffer(transport_options, current_tdesc));
>-  bool ret = (new_tdesc.get() != NULL &&
>-      offer_desc->AddTransportInfo(TransportInfo(content_name, *new_tdesc)));
>+  bool ret =
>+      (new_tdesc.get() != NULL &&
>+       offer_desc->AddTransportInfo(TransportInfo(content_name, *new_tdesc)));
>   if (!ret) {
>     RTC_LOG(LS_ERROR) << "Failed to AddTransportOffer, content name="
>                       << content_name;
>@@ -1765,8 +1807,8 @@ bool MediaSessionDescriptionFactory::AddTransportAnswer(
>     const std::string& content_name,
>     const TransportDescription& transport_desc,
>     SessionDescription* answer_desc) const {
>-  if (!answer_desc->AddTransportInfo(TransportInfo(content_name,
>-                                                   transport_desc))) {
>+  if (!answer_desc->AddTransportInfo(
>+          TransportInfo(content_name, transport_desc))) {
>     RTC_LOG(LS_ERROR) << "Failed to AddTransportAnswer, content name="
>                       << content_name;
>     return false;
>@@ -1960,8 +2002,8 @@ bool MediaSessionDescriptionFactory::AddDataContentForOffer(
>     // TODO(deadbeef): Offer kMediaProtocolUdpDtlsSctp (or TcpDtlsSctp), once
>     // it's safe to do so. Older versions of webrtc would reject these
>     // protocols; see https://bugs.chromium.org/p/webrtc/issues/detail?id=7706.
>-    data->set_protocol(
>-        secure_transport ? kMediaProtocolDtlsSctp : kMediaProtocolSctp);
>+    data->set_protocol(secure_transport ? kMediaProtocolDtlsSctp
>+                                        : kMediaProtocolSctp);
>   } else {
>     GetSupportedDataSdesCryptoSuiteNames(session_options.crypto_options,
>                                          &crypto_suites);
>@@ -2071,8 +2113,9 @@ bool MediaSessionDescriptionFactory::AddAudioContentForAnswer(
>   if (!CreateMediaContentAnswer(
>           offer_audio_description, media_description_options, session_options,
>           filtered_codecs, sdes_policy, GetCryptos(current_content),
>-          audio_rtp_extensions_, enable_encrypted_rtp_header_extensions_,
>-          current_streams, bundle_enabled, audio_answer.get())) {
>+          audio_rtp_header_extensions(session_options.is_unified_plan),
>+          enable_encrypted_rtp_header_extensions_, current_streams,
>+          bundle_enabled, audio_answer.get())) {
>     return false;  // Fails the session setup.
>   }
> 
>@@ -2082,10 +2125,12 @@ bool MediaSessionDescriptionFactory::AddAudioContentForAnswer(
>                   offer_content->rejected ||
>                   !IsMediaProtocolSupported(MEDIA_TYPE_AUDIO,
>                                             audio_answer->protocol(), secure);
>-  if (!rejected) {
>-    AddTransportAnswer(media_description_options.mid, *(audio_transport.get()),
>-                       answer);
>-  } else {
>+  if (!AddTransportAnswer(media_description_options.mid,
>+                          *(audio_transport.get()), answer)) {
>+    return false;
>+  }
>+
>+  if (rejected) {
>     RTC_LOG(LS_INFO) << "Audio m= section '" << media_description_options.mid
>                      << "' being rejected in answer.";
>   }
>@@ -2154,8 +2199,9 @@ bool MediaSessionDescriptionFactory::AddVideoContentForAnswer(
>   if (!CreateMediaContentAnswer(
>           offer_video_description, media_description_options, session_options,
>           filtered_codecs, sdes_policy, GetCryptos(current_content),
>-          video_rtp_extensions_, enable_encrypted_rtp_header_extensions_,
>-          current_streams, bundle_enabled, video_answer.get())) {
>+          video_rtp_header_extensions(session_options.is_unified_plan),
>+          enable_encrypted_rtp_header_extensions_, current_streams,
>+          bundle_enabled, video_answer.get())) {
>     return false;  // Failed the sessin setup.
>   }
>   bool secure = bundle_transport ? bundle_transport->description.secure()
>@@ -2164,11 +2210,12 @@ bool MediaSessionDescriptionFactory::AddVideoContentForAnswer(
>                   offer_content->rejected ||
>                   !IsMediaProtocolSupported(MEDIA_TYPE_VIDEO,
>                                             video_answer->protocol(), secure);
>+  if (!AddTransportAnswer(media_description_options.mid,
>+                          *(video_transport.get()), answer)) {
>+    return false;
>+  }
>+
>   if (!rejected) {
>-    if (!AddTransportAnswer(media_description_options.mid,
>-                            *(video_transport.get()), answer)) {
>-      return false;
>-    }
>     video_answer->set_bandwidth(kAutoBandwidth);
>   } else {
>     RTC_LOG(LS_INFO) << "Video m= section '" << media_description_options.mid
>@@ -2228,12 +2275,13 @@ bool MediaSessionDescriptionFactory::AddDataContentForAnswer(
>                   offer_content->rejected ||
>                   !IsMediaProtocolSupported(MEDIA_TYPE_DATA,
>                                             data_answer->protocol(), secure);
>+  if (!AddTransportAnswer(media_description_options.mid,
>+                          *(data_transport.get()), answer)) {
>+    return false;
>+  }
>+
>   if (!rejected) {
>     data_answer->set_bandwidth(kDataMaxBandwidth);
>-    if (!AddTransportAnswer(media_description_options.mid,
>-                            *(data_transport.get()), answer)) {
>-      return false;
>-    }
>   } else {
>     // RFC 3264
>     // The answer MUST contain the same number of m-lines as the offer.
>@@ -2311,8 +2359,8 @@ const ContentInfo* GetFirstDataContent(const ContentInfos& contents) {
>   return GetFirstMediaContent(contents, MEDIA_TYPE_DATA);
> }
> 
>-static const ContentInfo* GetFirstMediaContent(const SessionDescription* sdesc,
>-                                               MediaType media_type) {
>+const ContentInfo* GetFirstMediaContent(const SessionDescription* sdesc,
>+                                        MediaType media_type) {
>   if (sdesc == nullptr) {
>     return nullptr;
>   }
>@@ -2333,7 +2381,8 @@ const ContentInfo* GetFirstDataContent(const SessionDescription* sdesc) {
> }
> 
> const MediaContentDescription* GetFirstMediaContentDescription(
>-    const SessionDescription* sdesc, MediaType media_type) {
>+    const SessionDescription* sdesc,
>+    MediaType media_type) {
>   const ContentInfo* content = GetFirstMediaContent(sdesc, media_type);
>   return (content ? content->media_description() : nullptr);
> }
>@@ -2382,8 +2431,8 @@ ContentInfo* GetFirstDataContent(ContentInfos* contents) {
>   return GetFirstMediaContent(contents, MEDIA_TYPE_DATA);
> }
> 
>-static ContentInfo* GetFirstMediaContent(SessionDescription* sdesc,
>-                                         MediaType media_type) {
>+ContentInfo* GetFirstMediaContent(SessionDescription* sdesc,
>+                                  MediaType media_type) {
>   if (sdesc == nullptr) {
>     return nullptr;
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediasession.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediasession.h
>index 7c531aaee6a..b40974039b8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediasession.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediasession.h
>@@ -35,8 +35,6 @@ const char kDefaultRtcpCname[] = "DefaultRtcpCname";
> // Options for an RtpSender contained with an media description/"m=" section.
> struct SenderOptions {
>   std::string track_id;
>-  // TODO(steveanton): As part of work towards Unified Plan, this has been
>-  // changed to be a vector. But for now this can only have exactly one.
>   std::vector<std::string> stream_ids;
>   int num_sim_layers;
> };
>@@ -94,6 +92,7 @@ struct MediaSessionOptions {
>   bool vad_enabled = true;  // When disabled, removes all CN codecs from SDP.
>   bool rtcp_mux_enabled = true;
>   bool bundle_enabled = false;
>+  bool is_unified_plan = false;
>   std::string rtcp_cname = kDefaultRtcpCname;
>   rtc::CryptoOptions crypto_options;
>   // List of media description options in the same order that the media
>@@ -125,16 +124,28 @@ class MediaSessionDescriptionFactory {
>   void set_audio_rtp_header_extensions(const RtpHeaderExtensions& extensions) {
>     audio_rtp_extensions_ = extensions;
>   }
>-  const RtpHeaderExtensions& audio_rtp_header_extensions() const {
>-    return audio_rtp_extensions_;
>+  RtpHeaderExtensions audio_rtp_header_extensions(bool unified_plan) const {
>+    RtpHeaderExtensions extensions = audio_rtp_extensions_;
>+    // If we are Unified Plan, also offer the MID header extension.
>+    if (unified_plan) {
>+      extensions.push_back(webrtc::RtpExtension(
>+          webrtc::RtpExtension::kMidUri, webrtc::RtpExtension::kMidDefaultId));
>+    }
>+    return extensions;
>   }
>   const VideoCodecs& video_codecs() const { return video_codecs_; }
>   void set_video_codecs(const VideoCodecs& codecs) { video_codecs_ = codecs; }
>   void set_video_rtp_header_extensions(const RtpHeaderExtensions& extensions) {
>     video_rtp_extensions_ = extensions;
>   }
>-  const RtpHeaderExtensions& video_rtp_header_extensions() const {
>-    return video_rtp_extensions_;
>+  RtpHeaderExtensions video_rtp_header_extensions(bool unified_plan) const {
>+    RtpHeaderExtensions extensions = video_rtp_extensions_;
>+    // If we are Unified Plan, also offer the MID header extension.
>+    if (unified_plan) {
>+      extensions.push_back(webrtc::RtpExtension(
>+          webrtc::RtpExtension::kMidUri, webrtc::RtpExtension::kMidDefaultId));
>+    }
>+    return extensions;
>   }
>   const DataCodecs& data_codecs() const { return data_codecs_; }
>   void set_data_codecs(const DataCodecs& codecs) { data_codecs_ = codecs; }
>@@ -168,14 +179,14 @@ class MediaSessionDescriptionFactory {
>                           AudioCodecs* audio_codecs,
>                           VideoCodecs* video_codecs,
>                           DataCodecs* data_codecs) const;
>-  void GetRtpHdrExtsToOffer(const SessionDescription* current_description,
>+  void GetRtpHdrExtsToOffer(const MediaSessionOptions& session_options,
>+                            const SessionDescription* current_description,
>                             RtpHeaderExtensions* audio_extensions,
>                             RtpHeaderExtensions* video_extensions) const;
>-  bool AddTransportOffer(
>-      const std::string& content_name,
>-      const TransportOptions& transport_options,
>-      const SessionDescription* current_desc,
>-      SessionDescription* offer) const;
>+  bool AddTransportOffer(const std::string& content_name,
>+                         const TransportOptions& transport_options,
>+                         const SessionDescription* current_desc,
>+                         SessionDescription* offer) const;
> 
>   TransportDescription* CreateTransportAnswer(
>       const std::string& content_name,
>@@ -184,10 +195,9 @@ class MediaSessionDescriptionFactory {
>       const SessionDescription* current_desc,
>       bool require_transport_attributes) const;
> 
>-  bool AddTransportAnswer(
>-      const std::string& content_name,
>-      const TransportDescription& transport_desc,
>-      SessionDescription* answer_desc) const;
>+  bool AddTransportAnswer(const std::string& content_name,
>+                          const TransportDescription& transport_desc,
>+                          SessionDescription* answer_desc) const;
> 
>   // Helpers for adding media contents to the SessionDescription. Returns true
>   // it succeeds or the media content is not needed, or false if there is any
>@@ -287,6 +297,8 @@ const ContentInfo* GetFirstMediaContent(const ContentInfos& contents,
> const ContentInfo* GetFirstAudioContent(const ContentInfos& contents);
> const ContentInfo* GetFirstVideoContent(const ContentInfos& contents);
> const ContentInfo* GetFirstDataContent(const ContentInfos& contents);
>+const ContentInfo* GetFirstMediaContent(const SessionDescription* sdesc,
>+                                        MediaType media_type);
> const ContentInfo* GetFirstAudioContent(const SessionDescription* sdesc);
> const ContentInfo* GetFirstVideoContent(const SessionDescription* sdesc);
> const ContentInfo* GetFirstDataContent(const SessionDescription* sdesc);
>@@ -302,6 +314,8 @@ ContentInfo* GetFirstMediaContent(ContentInfos* contents, MediaType media_type);
> ContentInfo* GetFirstAudioContent(ContentInfos* contents);
> ContentInfo* GetFirstVideoContent(ContentInfos* contents);
> ContentInfo* GetFirstDataContent(ContentInfos* contents);
>+ContentInfo* GetFirstMediaContent(SessionDescription* sdesc,
>+                                  MediaType media_type);
> ContentInfo* GetFirstAudioContent(SessionDescription* sdesc);
> ContentInfo* GetFirstVideoContent(SessionDescription* sdesc);
> ContentInfo* GetFirstDataContent(SessionDescription* sdesc);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediasession_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediasession_unittest.cc
>index 4176e0e76e9..269ad78a708 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediasession_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediasession_unittest.cc
>@@ -26,9 +26,9 @@
> #include "rtc_base/messagedigest.h"
> #include "rtc_base/ssladapter.h"
> 
>-#define ASSERT_CRYPTO(cd, s, cs) \
>-    ASSERT_EQ(s, cd->cryptos().size()); \
>-    ASSERT_EQ(std::string(cs), cd->cryptos()[0].cipher_suite)
>+#define ASSERT_CRYPTO(cd, s, cs)      \
>+  ASSERT_EQ(s, cd->cryptos().size()); \
>+  ASSERT_EQ(std::string(cs), cd->cryptos()[0].cipher_suite)
> 
> typedef std::vector<cricket::Candidate> Candidates;
> 
>@@ -82,7 +82,7 @@ static const AudioCodec kAudioCodecs1[] = {
>     AudioCodec(107, "CN", 48000, 0, 1)};
> 
> static const AudioCodec kAudioCodecs2[] = {
>-    AudioCodec(126, "speex", 16000, 22000, 1),
>+    AudioCodec(126, "foo", 16000, 22000, 1),
>     AudioCodec(0, "PCMU", 8000, 64000, 1),
>     AudioCodec(127, "iLBC", 8000, 13300, 1),
> };
>@@ -213,6 +213,11 @@ static const char* kMediaProtocolsDtls[] = {
>     "TCP/TLS/RTP/SAVPF", "TCP/TLS/RTP/SAVP", "UDP/TLS/RTP/SAVPF",
>     "UDP/TLS/RTP/SAVP"};
> 
>+// SRTP cipher name negotiated by the tests. This must be updated if the
>+// default changes.
>+static const char* kDefaultSrtpCryptoSuite = CS_AES_CM_128_HMAC_SHA1_80;
>+static const char* kDefaultSrtpCryptoSuiteGcm = CS_AEAD_AES_256_GCM;
>+
> // These constants are used to make the code using "AddMediaSection" more
> // readable.
> static constexpr bool kStopped = true;
>@@ -375,7 +380,7 @@ class MediaSessionDescriptionFactoryTest : public testing::Test {
>     simulcast_params.ssrcs = MAKE_VECTOR(kSimulcastParamsSsrc);
>     simulcast_params.ssrc_groups = ssrc_groups;
>     simulcast_params.cname = "Video_SIM_FEC";
>-    simulcast_params.sync_label = kMediaStream1;
>+    simulcast_params.set_stream_ids({kMediaStream1});
> 
>     StreamParamsVec video_streams;
>     video_streams.push_back(simulcast_params);
>@@ -418,18 +423,14 @@ class MediaSessionDescriptionFactoryTest : public testing::Test {
>     std::unique_ptr<SessionDescription> desc;
>     if (has_current_desc) {
>       current_desc.reset(new SessionDescription());
>-      EXPECT_TRUE(current_desc->AddTransportInfo(
>-          TransportInfo("audio",
>-                        TransportDescription(current_audio_ufrag,
>-                                             current_audio_pwd))));
>-      EXPECT_TRUE(current_desc->AddTransportInfo(
>-          TransportInfo("video",
>-                        TransportDescription(current_video_ufrag,
>-                                             current_video_pwd))));
>-      EXPECT_TRUE(current_desc->AddTransportInfo(
>-          TransportInfo("data",
>-                        TransportDescription(current_data_ufrag,
>-                                             current_data_pwd))));
>+      EXPECT_TRUE(current_desc->AddTransportInfo(TransportInfo(
>+          "audio",
>+          TransportDescription(current_audio_ufrag, current_audio_pwd))));
>+      EXPECT_TRUE(current_desc->AddTransportInfo(TransportInfo(
>+          "video",
>+          TransportDescription(current_video_ufrag, current_video_pwd))));
>+      EXPECT_TRUE(current_desc->AddTransportInfo(TransportInfo(
>+          "data", TransportDescription(current_data_ufrag, current_data_pwd))));
>     }
>     if (offer) {
>       desc.reset(f1_.CreateOffer(options, current_desc.get()));
>@@ -466,8 +467,7 @@ class MediaSessionDescriptionFactoryTest : public testing::Test {
>       if (options.bundle_enabled) {
>         EXPECT_EQ(ti_audio->description.ice_ufrag,
>                   ti_video->description.ice_ufrag);
>-        EXPECT_EQ(ti_audio->description.ice_pwd,
>-                  ti_video->description.ice_pwd);
>+        EXPECT_EQ(ti_audio->description.ice_pwd, ti_video->description.ice_pwd);
>       } else {
>         if (has_current_desc) {
>           EXPECT_EQ(current_video_ufrag, ti_video->description.ice_ufrag);
>@@ -493,8 +493,7 @@ class MediaSessionDescriptionFactoryTest : public testing::Test {
>       if (options.bundle_enabled) {
>         EXPECT_EQ(ti_audio->description.ice_ufrag,
>                   ti_data->description.ice_ufrag);
>-        EXPECT_EQ(ti_audio->description.ice_pwd,
>-                  ti_data->description.ice_pwd);
>+        EXPECT_EQ(ti_audio->description.ice_pwd, ti_data->description.ice_pwd);
>       } else {
>         if (has_current_desc) {
>           EXPECT_EQ(current_data_ufrag, ti_data->description.ice_ufrag);
>@@ -545,7 +544,7 @@ class MediaSessionDescriptionFactoryTest : public testing::Test {
>     EXPECT_TRUE(CompareCryptoParams(audio_media_desc->cryptos(),
>                                     video_media_desc->cryptos()));
>     EXPECT_EQ(1u, audio_media_desc->cryptos().size());
>-    EXPECT_EQ(std::string(CS_AES_CM_128_HMAC_SHA1_80),
>+    EXPECT_EQ(std::string(kDefaultSrtpCryptoSuite),
>               audio_media_desc->cryptos()[0].cipher_suite);
> 
>     // Verify the selected crypto is one from the reference audio
>@@ -555,7 +554,7 @@ class MediaSessionDescriptionFactoryTest : public testing::Test {
>     bool found = false;
>     for (size_t i = 0; i < ref_audio_media_desc->cryptos().size(); ++i) {
>       if (ref_audio_media_desc->cryptos()[i].Matches(
>-          audio_media_desc->cryptos()[0])) {
>+              audio_media_desc->cryptos()[0])) {
>         found = true;
>         break;
>       }
>@@ -636,18 +635,18 @@ class MediaSessionDescriptionFactoryTest : public testing::Test {
>     EXPECT_EQ(0U, acd->first_ssrc());             // no sender is attached
>     EXPECT_TRUE(acd->rtcp_mux());                 // negotiated rtcp-mux
>     if (gcm_offer && gcm_answer) {
>-      ASSERT_CRYPTO(acd, 1U, CS_AEAD_AES_256_GCM);
>+      ASSERT_CRYPTO(acd, 1U, kDefaultSrtpCryptoSuiteGcm);
>     } else {
>-      ASSERT_CRYPTO(acd, 1U, CS_AES_CM_128_HMAC_SHA1_32);
>+      ASSERT_CRYPTO(acd, 1U, kDefaultSrtpCryptoSuite);
>     }
>     EXPECT_EQ(MEDIA_TYPE_VIDEO, vcd->type());
>     EXPECT_EQ(MAKE_VECTOR(kVideoCodecsAnswer), vcd->codecs());
>-    EXPECT_EQ(0U, vcd->first_ssrc());             // no sender is attached
>-    EXPECT_TRUE(vcd->rtcp_mux());                 // negotiated rtcp-mux
>+    EXPECT_EQ(0U, vcd->first_ssrc());  // no sender is attached
>+    EXPECT_TRUE(vcd->rtcp_mux());      // negotiated rtcp-mux
>     if (gcm_offer && gcm_answer) {
>-      ASSERT_CRYPTO(vcd, 1U, CS_AEAD_AES_256_GCM);
>+      ASSERT_CRYPTO(vcd, 1U, kDefaultSrtpCryptoSuiteGcm);
>     } else {
>-      ASSERT_CRYPTO(vcd, 1U, CS_AES_CM_128_HMAC_SHA1_80);
>+      ASSERT_CRYPTO(vcd, 1U, kDefaultSrtpCryptoSuite);
>     }
>     EXPECT_EQ(std::string(cricket::kMediaProtocolSavpf), vcd->protocol());
>   }
>@@ -676,7 +675,7 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateAudioOffer) {
>   EXPECT_EQ(0U, acd->first_ssrc());             // no sender is attached.
>   EXPECT_EQ(kAutoBandwidth, acd->bandwidth());  // default bandwidth (auto)
>   EXPECT_TRUE(acd->rtcp_mux());                 // rtcp-mux defaults on
>-  ASSERT_CRYPTO(acd, 2U, CS_AES_CM_128_HMAC_SHA1_32);
>+  ASSERT_CRYPTO(acd, 1U, kDefaultSrtpCryptoSuite);
>   EXPECT_EQ(std::string(cricket::kMediaProtocolSavpf), acd->protocol());
> }
> 
>@@ -700,14 +699,14 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateVideoOffer) {
>   EXPECT_EQ(0U, acd->first_ssrc());             // no sender is attached
>   EXPECT_EQ(kAutoBandwidth, acd->bandwidth());  // default bandwidth (auto)
>   EXPECT_TRUE(acd->rtcp_mux());                 // rtcp-mux defaults on
>-  ASSERT_CRYPTO(acd, 2U, CS_AES_CM_128_HMAC_SHA1_32);
>+  ASSERT_CRYPTO(acd, 1U, kDefaultSrtpCryptoSuite);
>   EXPECT_EQ(std::string(cricket::kMediaProtocolSavpf), acd->protocol());
>   EXPECT_EQ(MEDIA_TYPE_VIDEO, vcd->type());
>   EXPECT_EQ(f1_.video_codecs(), vcd->codecs());
>   EXPECT_EQ(0U, vcd->first_ssrc());             // no sender is attached
>   EXPECT_EQ(kAutoBandwidth, vcd->bandwidth());  // default bandwidth (auto)
>   EXPECT_TRUE(vcd->rtcp_mux());                 // rtcp-mux defaults on
>-  ASSERT_CRYPTO(vcd, 1U, CS_AES_CM_128_HMAC_SHA1_80);
>+  ASSERT_CRYPTO(vcd, 1U, kDefaultSrtpCryptoSuite);
>   EXPECT_EQ(std::string(cricket::kMediaProtocolSavpf), vcd->protocol());
> }
> 
>@@ -778,11 +777,11 @@ TEST_F(MediaSessionDescriptionFactoryTest,
>   EXPECT_TRUE(NULL != acd);
>   EXPECT_TRUE(NULL != dcd);
> 
>-  ASSERT_CRYPTO(acd, 1U, CS_AES_CM_128_HMAC_SHA1_80);
>+  ASSERT_CRYPTO(acd, 1U, kDefaultSrtpCryptoSuite);
>   EXPECT_EQ(std::string(cricket::kMediaProtocolSavpf), acd->protocol());
>-  ASSERT_CRYPTO(vcd, 1U, CS_AES_CM_128_HMAC_SHA1_80);
>+  ASSERT_CRYPTO(vcd, 1U, kDefaultSrtpCryptoSuite);
>   EXPECT_EQ(std::string(cricket::kMediaProtocolSavpf), vcd->protocol());
>-  ASSERT_CRYPTO(dcd, 1U, CS_AES_CM_128_HMAC_SHA1_80);
>+  ASSERT_CRYPTO(dcd, 1U, kDefaultSrtpCryptoSuite);
>   EXPECT_EQ(std::string(cricket::kMediaProtocolSavpf), dcd->protocol());
> }
> 
>@@ -807,15 +806,15 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateRtpDataOffer) {
>   EXPECT_EQ(0U, acd->first_ssrc());             // no sender is attched.
>   EXPECT_EQ(kAutoBandwidth, acd->bandwidth());  // default bandwidth (auto)
>   EXPECT_TRUE(acd->rtcp_mux());                 // rtcp-mux defaults on
>-  ASSERT_CRYPTO(acd, 2U, CS_AES_CM_128_HMAC_SHA1_32);
>+  ASSERT_CRYPTO(acd, 1U, kDefaultSrtpCryptoSuite);
>   EXPECT_EQ(std::string(cricket::kMediaProtocolSavpf), acd->protocol());
>   EXPECT_EQ(MEDIA_TYPE_DATA, dcd->type());
>   EXPECT_EQ(f1_.data_codecs(), dcd->codecs());
>   EXPECT_EQ(0U, dcd->first_ssrc());  // no sender is attached.
>   EXPECT_EQ(cricket::kDataMaxBandwidth,
>-            dcd->bandwidth());                  // default bandwidth (auto)
>-  EXPECT_TRUE(dcd->rtcp_mux());                 // rtcp-mux defaults on
>-  ASSERT_CRYPTO(dcd, 1U, CS_AES_CM_128_HMAC_SHA1_80);
>+            dcd->bandwidth());   // default bandwidth (auto)
>+  EXPECT_TRUE(dcd->rtcp_mux());  // rtcp-mux defaults on
>+  ASSERT_CRYPTO(dcd, 1U, kDefaultSrtpCryptoSuite);
>   EXPECT_EQ(std::string(cricket::kMediaProtocolSavpf), dcd->protocol());
> }
> 
>@@ -867,8 +866,8 @@ TEST_F(MediaSessionDescriptionFactoryTest,
>   const AudioContentDescription* acd = ac->media_description()->as_audio();
>   const VideoContentDescription* vcd = vc->media_description()->as_video();
> 
>-  EXPECT_FALSE(vcd->has_ssrcs());             // No StreamParams.
>-  EXPECT_FALSE(acd->has_ssrcs());             // No StreamParams.
>+  EXPECT_FALSE(vcd->has_ssrcs());  // No StreamParams.
>+  EXPECT_FALSE(acd->has_ssrcs());  // No StreamParams.
> }
> 
> // Creates an audio+video sendonly offer.
>@@ -943,7 +942,7 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateAudioAnswer) {
>   EXPECT_EQ(0U, acd->first_ssrc());             // no sender is attached
>   EXPECT_EQ(kAutoBandwidth, acd->bandwidth());  // negotiated auto bw
>   EXPECT_TRUE(acd->rtcp_mux());                 // negotiated rtcp-mux
>-  ASSERT_CRYPTO(acd, 1U, CS_AES_CM_128_HMAC_SHA1_32);
>+  ASSERT_CRYPTO(acd, 1U, kDefaultSrtpCryptoSuite);
>   EXPECT_EQ(std::string(cricket::kMediaProtocolSavpf), acd->protocol());
> }
> 
>@@ -969,7 +968,7 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateAudioAnswerGcm) {
>   EXPECT_EQ(0U, acd->first_ssrc());             // no sender is attached
>   EXPECT_EQ(kAutoBandwidth, acd->bandwidth());  // negotiated auto bw
>   EXPECT_TRUE(acd->rtcp_mux());                 // negotiated rtcp-mux
>-  ASSERT_CRYPTO(acd, 1U, CS_AEAD_AES_256_GCM);
>+  ASSERT_CRYPTO(acd, 1U, kDefaultSrtpCryptoSuiteGcm);
>   EXPECT_EQ(std::string(cricket::kMediaProtocolSavpf), acd->protocol());
> }
> 
>@@ -996,12 +995,12 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateVideoAnswer) {
>   EXPECT_EQ(kAutoBandwidth, acd->bandwidth());  // negotiated auto bw
>   EXPECT_EQ(0U, acd->first_ssrc());             // no sender is attached
>   EXPECT_TRUE(acd->rtcp_mux());                 // negotiated rtcp-mux
>-  ASSERT_CRYPTO(acd, 1U, CS_AES_CM_128_HMAC_SHA1_32);
>+  ASSERT_CRYPTO(acd, 1U, kDefaultSrtpCryptoSuite);
>   EXPECT_EQ(MEDIA_TYPE_VIDEO, vcd->type());
>   EXPECT_EQ(MAKE_VECTOR(kVideoCodecsAnswer), vcd->codecs());
>-  EXPECT_EQ(0U, vcd->first_ssrc());             // no sender is attached
>-  EXPECT_TRUE(vcd->rtcp_mux());                 // negotiated rtcp-mux
>-  ASSERT_CRYPTO(vcd, 1U, CS_AES_CM_128_HMAC_SHA1_80);
>+  EXPECT_EQ(0U, vcd->first_ssrc());  // no sender is attached
>+  EXPECT_TRUE(vcd->rtcp_mux());      // negotiated rtcp-mux
>+  ASSERT_CRYPTO(vcd, 1U, kDefaultSrtpCryptoSuite);
>   EXPECT_EQ(std::string(cricket::kMediaProtocolSavpf), vcd->protocol());
> }
> 
>@@ -1045,12 +1044,12 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateDataAnswer) {
>   EXPECT_EQ(kAutoBandwidth, acd->bandwidth());  // negotiated auto bw
>   EXPECT_EQ(0U, acd->first_ssrc());             // no sender is attached
>   EXPECT_TRUE(acd->rtcp_mux());                 // negotiated rtcp-mux
>-  ASSERT_CRYPTO(acd, 1U, CS_AES_CM_128_HMAC_SHA1_32);
>+  ASSERT_CRYPTO(acd, 1U, kDefaultSrtpCryptoSuite);
>   EXPECT_EQ(MEDIA_TYPE_DATA, dcd->type());
>   EXPECT_EQ(MAKE_VECTOR(kDataCodecsAnswer), dcd->codecs());
>   EXPECT_EQ(0U, dcd->first_ssrc());  // no sender is attached
>   EXPECT_TRUE(dcd->rtcp_mux());      // negotiated rtcp-mux
>-  ASSERT_CRYPTO(dcd, 1U, CS_AES_CM_128_HMAC_SHA1_80);
>+  ASSERT_CRYPTO(dcd, 1U, kDefaultSrtpCryptoSuite);
>   EXPECT_EQ(std::string(cricket::kMediaProtocolSavpf), dcd->protocol());
> }
> 
>@@ -1077,12 +1076,12 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateDataAnswerGcm) {
>   EXPECT_EQ(kAutoBandwidth, acd->bandwidth());  // negotiated auto bw
>   EXPECT_EQ(0U, acd->first_ssrc());             // no sender is attached
>   EXPECT_TRUE(acd->rtcp_mux());                 // negotiated rtcp-mux
>-  ASSERT_CRYPTO(acd, 1U, CS_AEAD_AES_256_GCM);
>+  ASSERT_CRYPTO(acd, 1U, kDefaultSrtpCryptoSuiteGcm);
>   EXPECT_EQ(MEDIA_TYPE_DATA, dcd->type());
>   EXPECT_EQ(MAKE_VECTOR(kDataCodecsAnswer), dcd->codecs());
>   EXPECT_EQ(0U, dcd->first_ssrc());  // no sender is attached
>   EXPECT_TRUE(dcd->rtcp_mux());      // negotiated rtcp-mux
>-  ASSERT_CRYPTO(dcd, 1U, CS_AEAD_AES_256_GCM);
>+  ASSERT_CRYPTO(dcd, 1U, kDefaultSrtpCryptoSuiteGcm);
>   EXPECT_EQ(std::string(cricket::kMediaProtocolSavpf), dcd->protocol());
> }
> 
>@@ -1294,124 +1293,112 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestOfferAnswerWithRtpExtensions) {
>   std::unique_ptr<SessionDescription> answer(
>       f2_.CreateAnswer(offer.get(), opts, NULL));
> 
>-  EXPECT_EQ(MAKE_VECTOR(kAudioRtpExtension1),
>-            GetFirstAudioContentDescription(
>-                offer.get())->rtp_header_extensions());
>-  EXPECT_EQ(MAKE_VECTOR(kVideoRtpExtension1),
>-            GetFirstVideoContentDescription(
>-                offer.get())->rtp_header_extensions());
>-  EXPECT_EQ(MAKE_VECTOR(kAudioRtpExtensionAnswer),
>-            GetFirstAudioContentDescription(
>-                answer.get())->rtp_header_extensions());
>-  EXPECT_EQ(MAKE_VECTOR(kVideoRtpExtensionAnswer),
>-            GetFirstVideoContentDescription(
>-                answer.get())->rtp_header_extensions());
>+  EXPECT_EQ(
>+      MAKE_VECTOR(kAudioRtpExtension1),
>+      GetFirstAudioContentDescription(offer.get())->rtp_header_extensions());
>+  EXPECT_EQ(
>+      MAKE_VECTOR(kVideoRtpExtension1),
>+      GetFirstVideoContentDescription(offer.get())->rtp_header_extensions());
>+  EXPECT_EQ(
>+      MAKE_VECTOR(kAudioRtpExtensionAnswer),
>+      GetFirstAudioContentDescription(answer.get())->rtp_header_extensions());
>+  EXPECT_EQ(
>+      MAKE_VECTOR(kVideoRtpExtensionAnswer),
>+      GetFirstVideoContentDescription(answer.get())->rtp_header_extensions());
> }
> 
> TEST_F(MediaSessionDescriptionFactoryTest,
>-    TestOfferAnswerWithEncryptedRtpExtensionsBoth) {
>+       TestOfferAnswerWithEncryptedRtpExtensionsBoth) {
>   MediaSessionOptions opts;
>   AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &opts);
> 
>   f1_.set_enable_encrypted_rtp_header_extensions(true);
>   f2_.set_enable_encrypted_rtp_header_extensions(true);
> 
>-  f1_.set_audio_rtp_header_extensions(
>-      MAKE_VECTOR(kAudioRtpExtension1));
>-  f1_.set_video_rtp_header_extensions(
>-      MAKE_VECTOR(kVideoRtpExtension1));
>-  f2_.set_audio_rtp_header_extensions(
>-      MAKE_VECTOR(kAudioRtpExtension2));
>-  f2_.set_video_rtp_header_extensions(
>-      MAKE_VECTOR(kVideoRtpExtension2));
>+  f1_.set_audio_rtp_header_extensions(MAKE_VECTOR(kAudioRtpExtension1));
>+  f1_.set_video_rtp_header_extensions(MAKE_VECTOR(kVideoRtpExtension1));
>+  f2_.set_audio_rtp_header_extensions(MAKE_VECTOR(kAudioRtpExtension2));
>+  f2_.set_video_rtp_header_extensions(MAKE_VECTOR(kVideoRtpExtension2));
> 
>   std::unique_ptr<SessionDescription> offer(f1_.CreateOffer(opts, NULL));
>   ASSERT_TRUE(offer.get() != NULL);
>   std::unique_ptr<SessionDescription> answer(
>       f2_.CreateAnswer(offer.get(), opts, NULL));
> 
>-  EXPECT_EQ(MAKE_VECTOR(kAudioRtpExtensionEncrypted1),
>-            GetFirstAudioContentDescription(
>-                offer.get())->rtp_header_extensions());
>-  EXPECT_EQ(MAKE_VECTOR(kVideoRtpExtensionEncrypted1),
>-            GetFirstVideoContentDescription(
>-                offer.get())->rtp_header_extensions());
>-  EXPECT_EQ(MAKE_VECTOR(kAudioRtpExtensionEncryptedAnswer),
>-            GetFirstAudioContentDescription(
>-                answer.get())->rtp_header_extensions());
>-  EXPECT_EQ(MAKE_VECTOR(kVideoRtpExtensionEncryptedAnswer),
>-            GetFirstVideoContentDescription(
>-                answer.get())->rtp_header_extensions());
>+  EXPECT_EQ(
>+      MAKE_VECTOR(kAudioRtpExtensionEncrypted1),
>+      GetFirstAudioContentDescription(offer.get())->rtp_header_extensions());
>+  EXPECT_EQ(
>+      MAKE_VECTOR(kVideoRtpExtensionEncrypted1),
>+      GetFirstVideoContentDescription(offer.get())->rtp_header_extensions());
>+  EXPECT_EQ(
>+      MAKE_VECTOR(kAudioRtpExtensionEncryptedAnswer),
>+      GetFirstAudioContentDescription(answer.get())->rtp_header_extensions());
>+  EXPECT_EQ(
>+      MAKE_VECTOR(kVideoRtpExtensionEncryptedAnswer),
>+      GetFirstVideoContentDescription(answer.get())->rtp_header_extensions());
> }
> 
> TEST_F(MediaSessionDescriptionFactoryTest,
>-    TestOfferAnswerWithEncryptedRtpExtensionsOffer) {
>+       TestOfferAnswerWithEncryptedRtpExtensionsOffer) {
>   MediaSessionOptions opts;
>   AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &opts);
> 
>   f1_.set_enable_encrypted_rtp_header_extensions(true);
> 
>-  f1_.set_audio_rtp_header_extensions(
>-      MAKE_VECTOR(kAudioRtpExtension1));
>-  f1_.set_video_rtp_header_extensions(
>-      MAKE_VECTOR(kVideoRtpExtension1));
>-  f2_.set_audio_rtp_header_extensions(
>-      MAKE_VECTOR(kAudioRtpExtension2));
>-  f2_.set_video_rtp_header_extensions(
>-      MAKE_VECTOR(kVideoRtpExtension2));
>+  f1_.set_audio_rtp_header_extensions(MAKE_VECTOR(kAudioRtpExtension1));
>+  f1_.set_video_rtp_header_extensions(MAKE_VECTOR(kVideoRtpExtension1));
>+  f2_.set_audio_rtp_header_extensions(MAKE_VECTOR(kAudioRtpExtension2));
>+  f2_.set_video_rtp_header_extensions(MAKE_VECTOR(kVideoRtpExtension2));
> 
>   std::unique_ptr<SessionDescription> offer(f1_.CreateOffer(opts, NULL));
>   ASSERT_TRUE(offer.get() != NULL);
>   std::unique_ptr<SessionDescription> answer(
>       f2_.CreateAnswer(offer.get(), opts, NULL));
> 
>-  EXPECT_EQ(MAKE_VECTOR(kAudioRtpExtensionEncrypted1),
>-            GetFirstAudioContentDescription(
>-                offer.get())->rtp_header_extensions());
>-  EXPECT_EQ(MAKE_VECTOR(kVideoRtpExtensionEncrypted1),
>-            GetFirstVideoContentDescription(
>-                offer.get())->rtp_header_extensions());
>-  EXPECT_EQ(MAKE_VECTOR(kAudioRtpExtensionAnswer),
>-            GetFirstAudioContentDescription(
>-                answer.get())->rtp_header_extensions());
>-  EXPECT_EQ(MAKE_VECTOR(kVideoRtpExtensionAnswer),
>-            GetFirstVideoContentDescription(
>-                answer.get())->rtp_header_extensions());
>+  EXPECT_EQ(
>+      MAKE_VECTOR(kAudioRtpExtensionEncrypted1),
>+      GetFirstAudioContentDescription(offer.get())->rtp_header_extensions());
>+  EXPECT_EQ(
>+      MAKE_VECTOR(kVideoRtpExtensionEncrypted1),
>+      GetFirstVideoContentDescription(offer.get())->rtp_header_extensions());
>+  EXPECT_EQ(
>+      MAKE_VECTOR(kAudioRtpExtensionAnswer),
>+      GetFirstAudioContentDescription(answer.get())->rtp_header_extensions());
>+  EXPECT_EQ(
>+      MAKE_VECTOR(kVideoRtpExtensionAnswer),
>+      GetFirstVideoContentDescription(answer.get())->rtp_header_extensions());
> }
> 
> TEST_F(MediaSessionDescriptionFactoryTest,
>-    TestOfferAnswerWithEncryptedRtpExtensionsAnswer) {
>+       TestOfferAnswerWithEncryptedRtpExtensionsAnswer) {
>   MediaSessionOptions opts;
>   AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &opts);
> 
>   f2_.set_enable_encrypted_rtp_header_extensions(true);
> 
>-  f1_.set_audio_rtp_header_extensions(
>-      MAKE_VECTOR(kAudioRtpExtension1));
>-  f1_.set_video_rtp_header_extensions(
>-      MAKE_VECTOR(kVideoRtpExtension1));
>-  f2_.set_audio_rtp_header_extensions(
>-      MAKE_VECTOR(kAudioRtpExtension2));
>-  f2_.set_video_rtp_header_extensions(
>-      MAKE_VECTOR(kVideoRtpExtension2));
>+  f1_.set_audio_rtp_header_extensions(MAKE_VECTOR(kAudioRtpExtension1));
>+  f1_.set_video_rtp_header_extensions(MAKE_VECTOR(kVideoRtpExtension1));
>+  f2_.set_audio_rtp_header_extensions(MAKE_VECTOR(kAudioRtpExtension2));
>+  f2_.set_video_rtp_header_extensions(MAKE_VECTOR(kVideoRtpExtension2));
> 
>   std::unique_ptr<SessionDescription> offer(f1_.CreateOffer(opts, NULL));
>   ASSERT_TRUE(offer.get() != NULL);
>   std::unique_ptr<SessionDescription> answer(
>       f2_.CreateAnswer(offer.get(), opts, NULL));
> 
>-  EXPECT_EQ(MAKE_VECTOR(kAudioRtpExtension1),
>-            GetFirstAudioContentDescription(
>-                offer.get())->rtp_header_extensions());
>-  EXPECT_EQ(MAKE_VECTOR(kVideoRtpExtension1),
>-            GetFirstVideoContentDescription(
>-                offer.get())->rtp_header_extensions());
>-  EXPECT_EQ(MAKE_VECTOR(kAudioRtpExtensionAnswer),
>-            GetFirstAudioContentDescription(
>-                answer.get())->rtp_header_extensions());
>-  EXPECT_EQ(MAKE_VECTOR(kVideoRtpExtensionAnswer),
>-            GetFirstVideoContentDescription(
>-                answer.get())->rtp_header_extensions());
>+  EXPECT_EQ(
>+      MAKE_VECTOR(kAudioRtpExtension1),
>+      GetFirstAudioContentDescription(offer.get())->rtp_header_extensions());
>+  EXPECT_EQ(
>+      MAKE_VECTOR(kVideoRtpExtension1),
>+      GetFirstVideoContentDescription(offer.get())->rtp_header_extensions());
>+  EXPECT_EQ(
>+      MAKE_VECTOR(kAudioRtpExtensionAnswer),
>+      GetFirstAudioContentDescription(answer.get())->rtp_header_extensions());
>+  EXPECT_EQ(
>+      MAKE_VECTOR(kVideoRtpExtensionAnswer),
>+      GetFirstVideoContentDescription(answer.get())->rtp_header_extensions());
> }
> 
> // Create an audio, video, data answer without legacy StreamParams.
>@@ -1633,7 +1620,7 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateMultiStreamVideoOffer) {
> 
>   const StreamParamsVec& audio_streams = acd->streams();
>   ASSERT_EQ(2U, audio_streams.size());
>-  EXPECT_EQ(audio_streams[0].cname , audio_streams[1].cname);
>+  EXPECT_EQ(audio_streams[0].cname, audio_streams[1].cname);
>   EXPECT_EQ(kAudioTrack1, audio_streams[0].id);
>   ASSERT_EQ(1U, audio_streams[0].ssrcs.size());
>   EXPECT_NE(0U, audio_streams[0].ssrcs[0]);
>@@ -1643,11 +1630,11 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateMultiStreamVideoOffer) {
> 
>   EXPECT_EQ(kAutoBandwidth, acd->bandwidth());  // default bandwidth (auto)
>   EXPECT_TRUE(acd->rtcp_mux());                 // rtcp-mux defaults on
>-  ASSERT_CRYPTO(acd, 2U, CS_AES_CM_128_HMAC_SHA1_32);
>+  ASSERT_CRYPTO(acd, 1U, kDefaultSrtpCryptoSuite);
> 
>   EXPECT_EQ(MEDIA_TYPE_VIDEO, vcd->type());
>   EXPECT_EQ(f1_.video_codecs(), vcd->codecs());
>-  ASSERT_CRYPTO(vcd, 1U, CS_AES_CM_128_HMAC_SHA1_80);
>+  ASSERT_CRYPTO(vcd, 1U, kDefaultSrtpCryptoSuite);
> 
>   const StreamParamsVec& video_streams = vcd->streams();
>   ASSERT_EQ(1U, video_streams.size());
>@@ -1658,11 +1645,11 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateMultiStreamVideoOffer) {
> 
>   EXPECT_EQ(MEDIA_TYPE_DATA, dcd->type());
>   EXPECT_EQ(f1_.data_codecs(), dcd->codecs());
>-  ASSERT_CRYPTO(dcd, 1U, CS_AES_CM_128_HMAC_SHA1_80);
>+  ASSERT_CRYPTO(dcd, 1U, kDefaultSrtpCryptoSuite);
> 
>   const StreamParamsVec& data_streams = dcd->streams();
>   ASSERT_EQ(2U, data_streams.size());
>-  EXPECT_EQ(data_streams[0].cname , data_streams[1].cname);
>+  EXPECT_EQ(data_streams[0].cname, data_streams[1].cname);
>   EXPECT_EQ(kDataTrack1, data_streams[0].id);
>   ASSERT_EQ(1U, data_streams[0].ssrcs.size());
>   EXPECT_NE(0U, data_streams[0].ssrcs[0]);
>@@ -1671,9 +1658,9 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateMultiStreamVideoOffer) {
>   EXPECT_NE(0U, data_streams[1].ssrcs[0]);
> 
>   EXPECT_EQ(cricket::kDataMaxBandwidth,
>-            dcd->bandwidth());                  // default bandwidth (auto)
>-  EXPECT_TRUE(dcd->rtcp_mux());                 // rtcp-mux defaults on
>-  ASSERT_CRYPTO(dcd, 1U, CS_AES_CM_128_HMAC_SHA1_80);
>+            dcd->bandwidth());   // default bandwidth (auto)
>+  EXPECT_TRUE(dcd->rtcp_mux());  // rtcp-mux defaults on
>+  ASSERT_CRYPTO(dcd, 1U, kDefaultSrtpCryptoSuite);
> 
>   // Update the offer. Add a new video track that is not synched to the
>   // other tracks and replace audio track 2 with audio track 3.
>@@ -1708,11 +1695,11 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateMultiStreamVideoOffer) {
>   EXPECT_EQ(vcd->codecs(), updated_vcd->codecs());
>   EXPECT_EQ(dcd->type(), updated_dcd->type());
>   EXPECT_EQ(dcd->codecs(), updated_dcd->codecs());
>-  ASSERT_CRYPTO(updated_acd, 2U, CS_AES_CM_128_HMAC_SHA1_32);
>+  ASSERT_CRYPTO(updated_acd, 1U, kDefaultSrtpCryptoSuite);
>   EXPECT_TRUE(CompareCryptoParams(acd->cryptos(), updated_acd->cryptos()));
>-  ASSERT_CRYPTO(updated_vcd, 1U, CS_AES_CM_128_HMAC_SHA1_80);
>+  ASSERT_CRYPTO(updated_vcd, 1U, kDefaultSrtpCryptoSuite);
>   EXPECT_TRUE(CompareCryptoParams(vcd->cryptos(), updated_vcd->cryptos()));
>-  ASSERT_CRYPTO(updated_dcd, 1U, CS_AES_CM_128_HMAC_SHA1_80);
>+  ASSERT_CRYPTO(updated_dcd, 1U, kDefaultSrtpCryptoSuite);
>   EXPECT_TRUE(CompareCryptoParams(dcd->cryptos(), updated_dcd->cryptos()));
> 
>   const StreamParamsVec& updated_audio_streams = updated_acd->streams();
>@@ -1821,16 +1808,16 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateMultiStreamVideoAnswer) {
>   const AudioContentDescription* acd = ac->media_description()->as_audio();
>   const VideoContentDescription* vcd = vc->media_description()->as_video();
>   const DataContentDescription* dcd = dc->media_description()->as_data();
>-  ASSERT_CRYPTO(acd, 1U, CS_AES_CM_128_HMAC_SHA1_32);
>-  ASSERT_CRYPTO(vcd, 1U, CS_AES_CM_128_HMAC_SHA1_80);
>-  ASSERT_CRYPTO(dcd, 1U, CS_AES_CM_128_HMAC_SHA1_80);
>+  ASSERT_CRYPTO(acd, 1U, kDefaultSrtpCryptoSuite);
>+  ASSERT_CRYPTO(vcd, 1U, kDefaultSrtpCryptoSuite);
>+  ASSERT_CRYPTO(dcd, 1U, kDefaultSrtpCryptoSuite);
> 
>   EXPECT_EQ(MEDIA_TYPE_AUDIO, acd->type());
>   EXPECT_EQ(MAKE_VECTOR(kAudioCodecsAnswer), acd->codecs());
> 
>   const StreamParamsVec& audio_streams = acd->streams();
>   ASSERT_EQ(2U, audio_streams.size());
>-  EXPECT_TRUE(audio_streams[0].cname ==  audio_streams[1].cname);
>+  EXPECT_TRUE(audio_streams[0].cname == audio_streams[1].cname);
>   EXPECT_EQ(kAudioTrack1, audio_streams[0].id);
>   ASSERT_EQ(1U, audio_streams[0].ssrcs.size());
>   EXPECT_NE(0U, audio_streams[0].ssrcs[0]);
>@@ -1856,7 +1843,7 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateMultiStreamVideoAnswer) {
> 
>   const StreamParamsVec& data_streams = dcd->streams();
>   ASSERT_EQ(2U, data_streams.size());
>-  EXPECT_TRUE(data_streams[0].cname ==  data_streams[1].cname);
>+  EXPECT_TRUE(data_streams[0].cname == data_streams[1].cname);
>   EXPECT_EQ(kDataTrack1, data_streams[0].id);
>   ASSERT_EQ(1U, data_streams[0].ssrcs.size());
>   EXPECT_NE(0U, data_streams[0].ssrcs[0]);
>@@ -1865,8 +1852,8 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateMultiStreamVideoAnswer) {
>   EXPECT_NE(0U, data_streams[1].ssrcs[0]);
> 
>   EXPECT_EQ(cricket::kDataMaxBandwidth,
>-            dcd->bandwidth());                  // default bandwidth (auto)
>-  EXPECT_TRUE(dcd->rtcp_mux());                 // rtcp-mux defaults on
>+            dcd->bandwidth());   // default bandwidth (auto)
>+  EXPECT_TRUE(dcd->rtcp_mux());  // rtcp-mux defaults on
> 
>   // Update the answer. Add a new video track that is not synched to the
>   // other tracks and remove 1 audio track.
>@@ -1891,11 +1878,11 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateMultiStreamVideoAnswer) {
>   const DataContentDescription* updated_dcd =
>       dc->media_description()->as_data();
> 
>-  ASSERT_CRYPTO(updated_acd, 1U, CS_AES_CM_128_HMAC_SHA1_32);
>+  ASSERT_CRYPTO(updated_acd, 1U, kDefaultSrtpCryptoSuite);
>   EXPECT_TRUE(CompareCryptoParams(acd->cryptos(), updated_acd->cryptos()));
>-  ASSERT_CRYPTO(updated_vcd, 1U, CS_AES_CM_128_HMAC_SHA1_80);
>+  ASSERT_CRYPTO(updated_vcd, 1U, kDefaultSrtpCryptoSuite);
>   EXPECT_TRUE(CompareCryptoParams(vcd->cryptos(), updated_vcd->cryptos()));
>-  ASSERT_CRYPTO(updated_dcd, 1U, CS_AES_CM_128_HMAC_SHA1_80);
>+  ASSERT_CRYPTO(updated_dcd, 1U, kDefaultSrtpCryptoSuite);
>   EXPECT_TRUE(CompareCryptoParams(dcd->cryptos(), updated_dcd->cryptos()));
> 
>   EXPECT_EQ(acd->type(), updated_acd->type());
>@@ -1907,7 +1894,7 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCreateMultiStreamVideoAnswer) {
> 
>   const StreamParamsVec& updated_audio_streams = updated_acd->streams();
>   ASSERT_EQ(1U, updated_audio_streams.size());
>-  EXPECT_TRUE(audio_streams[0] ==  updated_audio_streams[0]);
>+  EXPECT_TRUE(audio_streams[0] == updated_audio_streams[0]);
> 
>   const StreamParamsVec& updated_video_streams = updated_vcd->streams();
>   ASSERT_EQ(2U, updated_video_streams.size());
>@@ -1950,17 +1937,14 @@ TEST_F(MediaSessionDescriptionFactoryTest,
>   // TODO(wu): |updated_offer| should not include the codec
>   // (i.e. |kAudioCodecs2[0]|) the other side doesn't support.
>   const AudioCodec kUpdatedAudioCodecOffer[] = {
>-    kAudioCodecsAnswer[0],
>-    kAudioCodecsAnswer[1],
>-    kAudioCodecs2[0],
>+      kAudioCodecsAnswer[0], kAudioCodecsAnswer[1], kAudioCodecs2[0],
>   };
> 
>   // The expected video codecs are the common video codecs from the first
>   // offer/answer exchange plus the video codecs only |f2_| offer, sorted in
>   // preference order.
>   const VideoCodec kUpdatedVideoCodecOffer[] = {
>-    kVideoCodecsAnswer[0],
>-    kVideoCodecs2[1],
>+      kVideoCodecsAnswer[0], kVideoCodecs2[1],
>   };
> 
>   const AudioContentDescription* updated_acd =
>@@ -2123,7 +2107,7 @@ TEST_F(MediaSessionDescriptionFactoryTest,
> 
>   ASSERT_EQ("H264", updated_vcd->codecs()[0].name);
>   ASSERT_EQ(std::string(cricket::kRtxCodecName), updated_vcd->codecs()[1].name);
>-  int new_h264_pl_type =  updated_vcd->codecs()[0].id;
>+  int new_h264_pl_type = updated_vcd->codecs()[0].id;
>   EXPECT_NE(used_pl_type, new_h264_pl_type);
>   VideoCodec rtx = updated_vcd->codecs()[1];
>   int pt_referenced_by_rtx = rtc::FromString<int>(
>@@ -2455,12 +2439,12 @@ TEST_F(MediaSessionDescriptionFactoryTest,
>   std::unique_ptr<SessionDescription> answer(
>       f2_.CreateAnswer(offer.get(), opts, NULL));
> 
>-  EXPECT_EQ(MAKE_VECTOR(kAudioRtpExtensionAnswer),
>-            GetFirstAudioContentDescription(
>-                answer.get())->rtp_header_extensions());
>-  EXPECT_EQ(MAKE_VECTOR(kVideoRtpExtensionAnswer),
>-            GetFirstVideoContentDescription(
>-                answer.get())->rtp_header_extensions());
>+  EXPECT_EQ(
>+      MAKE_VECTOR(kAudioRtpExtensionAnswer),
>+      GetFirstAudioContentDescription(answer.get())->rtp_header_extensions());
>+  EXPECT_EQ(
>+      MAKE_VECTOR(kVideoRtpExtensionAnswer),
>+      GetFirstVideoContentDescription(answer.get())->rtp_header_extensions());
> 
>   std::unique_ptr<SessionDescription> updated_offer(
>       f2_.CreateOffer(opts, answer.get()));
>@@ -2511,23 +2495,23 @@ TEST_F(MediaSessionDescriptionFactoryTest, RtpExtensionIdReused) {
>       kVideoRtpExtension3[0], kAudioRtpExtension3[1],
>   };
> 
>-  EXPECT_EQ(MAKE_VECTOR(kAudioRtpExtension3),
>-            GetFirstAudioContentDescription(
>-                offer.get())->rtp_header_extensions());
>-  EXPECT_EQ(MAKE_VECTOR(kExpectedVideoRtpExtension),
>-            GetFirstVideoContentDescription(
>-                offer.get())->rtp_header_extensions());
>+  EXPECT_EQ(
>+      MAKE_VECTOR(kAudioRtpExtension3),
>+      GetFirstAudioContentDescription(offer.get())->rtp_header_extensions());
>+  EXPECT_EQ(
>+      MAKE_VECTOR(kExpectedVideoRtpExtension),
>+      GetFirstVideoContentDescription(offer.get())->rtp_header_extensions());
> 
>   // Nothing should change when creating a new offer
>   std::unique_ptr<SessionDescription> updated_offer(
>       f1_.CreateOffer(opts, offer.get()));
> 
>   EXPECT_EQ(MAKE_VECTOR(kAudioRtpExtension3),
>-            GetFirstAudioContentDescription(
>-                updated_offer.get())->rtp_header_extensions());
>+            GetFirstAudioContentDescription(updated_offer.get())
>+                ->rtp_header_extensions());
>   EXPECT_EQ(MAKE_VECTOR(kExpectedVideoRtpExtension),
>-            GetFirstVideoContentDescription(
>-                updated_offer.get())->rtp_header_extensions());
>+            GetFirstVideoContentDescription(updated_offer.get())
>+                ->rtp_header_extensions());
> }
> 
> // Same as "RtpExtensionIdReused" above for encrypted RTP extensions.
>@@ -2553,23 +2537,23 @@ TEST_F(MediaSessionDescriptionFactoryTest, RtpExtensionIdReusedEncrypted) {
>       kAudioRtpExtension3ForEncryptionOffer[2],
>   };
> 
>-  EXPECT_EQ(MAKE_VECTOR(kAudioRtpExtension3ForEncryptionOffer),
>-            GetFirstAudioContentDescription(
>-                offer.get())->rtp_header_extensions());
>-  EXPECT_EQ(MAKE_VECTOR(kExpectedVideoRtpExtension),
>-            GetFirstVideoContentDescription(
>-                offer.get())->rtp_header_extensions());
>+  EXPECT_EQ(
>+      MAKE_VECTOR(kAudioRtpExtension3ForEncryptionOffer),
>+      GetFirstAudioContentDescription(offer.get())->rtp_header_extensions());
>+  EXPECT_EQ(
>+      MAKE_VECTOR(kExpectedVideoRtpExtension),
>+      GetFirstVideoContentDescription(offer.get())->rtp_header_extensions());
> 
>   // Nothing should change when creating a new offer
>   std::unique_ptr<SessionDescription> updated_offer(
>       f1_.CreateOffer(opts, offer.get()));
> 
>   EXPECT_EQ(MAKE_VECTOR(kAudioRtpExtension3ForEncryptionOffer),
>-            GetFirstAudioContentDescription(
>-                updated_offer.get())->rtp_header_extensions());
>+            GetFirstAudioContentDescription(updated_offer.get())
>+                ->rtp_header_extensions());
>   EXPECT_EQ(MAKE_VECTOR(kExpectedVideoRtpExtension),
>-            GetFirstVideoContentDescription(
>-                updated_offer.get())->rtp_header_extensions());
>+            GetFirstVideoContentDescription(updated_offer.get())
>+                ->rtp_header_extensions());
> }
> 
> TEST(MediaSessionDescription, CopySessionDescription) {
>@@ -2638,7 +2622,7 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestTransportInfoOfferMultimedia) {
> }
> 
> TEST_F(MediaSessionDescriptionFactoryTest,
>-    TestTransportInfoOfferMultimediaCurrent) {
>+       TestTransportInfoOfferMultimediaCurrent) {
>   MediaSessionOptions options;
>   AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &options);
>   AddDataSection(cricket::DCT_RTP, RtpTransceiverDirection::kRecvOnly,
>@@ -2699,7 +2683,7 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestTransportInfoAnswerMultimedia) {
> }
> 
> TEST_F(MediaSessionDescriptionFactoryTest,
>-    TestTransportInfoAnswerMultimediaCurrent) {
>+       TestTransportInfoAnswerMultimediaCurrent) {
>   MediaSessionOptions options;
>   AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &options);
>   AddDataSection(cricket::DCT_RTP, RtpTransceiverDirection::kRecvOnly,
>@@ -2717,7 +2701,7 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestTransportInfoAnswerBundle) {
> }
> 
> TEST_F(MediaSessionDescriptionFactoryTest,
>-    TestTransportInfoAnswerBundleCurrent) {
>+       TestTransportInfoAnswerBundleCurrent) {
>   MediaSessionOptions options;
>   AddAudioVideoSections(RtpTransceiverDirection::kRecvOnly, &options);
>   AddDataSection(cricket::DCT_RTP, RtpTransceiverDirection::kRecvOnly,
>@@ -2793,7 +2777,7 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestOfferDtlsSavpfCreateAnswer) {
>   const AudioContentDescription* answer_audio_desc =
>       answer_content->media_description()->as_audio();
>   EXPECT_EQ(std::string(cricket::kMediaProtocolDtlsSavpf),
>-                        answer_audio_desc->protocol());
>+            answer_audio_desc->protocol());
> }
> 
> // Test that we include both SDES and DTLS in the offer, but only include SDES
>@@ -2819,7 +2803,7 @@ TEST_F(MediaSessionDescriptionFactoryTest, TestCryptoDtls) {
>   ASSERT_TRUE(audio_media_desc != NULL);
>   video_media_desc = offer->GetContentDescriptionByName("video");
>   ASSERT_TRUE(video_media_desc != NULL);
>-  EXPECT_EQ(2u, audio_media_desc->cryptos().size());
>+  EXPECT_EQ(1u, audio_media_desc->cryptos().size());
>   EXPECT_EQ(1u, video_media_desc->cryptos().size());
> 
>   audio_trans_desc = offer->GetTransportDescriptionByName("audio");
>@@ -3196,6 +3180,10 @@ TEST_F(MediaSessionDescriptionFactoryTest,
>   ASSERT_EQ(2u, answer->contents().size());
>   EXPECT_FALSE(answer->contents()[0].rejected);
>   EXPECT_TRUE(answer->contents()[1].rejected);
>+
>+  // The TransportInfo of the rejected m= section is expected to be added in the
>+  // answer.
>+  EXPECT_EQ(offer->transport_infos().size(), answer->transport_infos().size());
> }
> 
> // Test the generated media sections has the same order of the
>@@ -3376,6 +3364,47 @@ TEST_F(MediaSessionDescriptionFactoryTest, CreateAnswerWithLocalCodecParams) {
>   EXPECT_EQ(video_value2, value);
> }
> 
>+// Test that matching packetization-mode is part of the criteria for matching
>+// H264 codecs (in addition to profile-level-id). Previously, this was not the
>+// case, so the first H264 codec with the same profile-level-id would match and
>+// the payload type in the answer would be incorrect.
>+// This is a regression test for bugs.webrtc.org/8808
>+TEST_F(MediaSessionDescriptionFactoryTest,
>+       H264MatchCriteriaIncludesPacketizationMode) {
>+  // Create two H264 codecs with the same profile level ID and different
>+  // packetization modes.
>+  VideoCodec h264_pm0(96, "H264");
>+  h264_pm0.params[cricket::kH264FmtpProfileLevelId] = "42c01f";
>+  h264_pm0.params[cricket::kH264FmtpPacketizationMode] = "0";
>+  VideoCodec h264_pm1(97, "H264");
>+  h264_pm1.params[cricket::kH264FmtpProfileLevelId] = "42c01f";
>+  h264_pm1.params[cricket::kH264FmtpPacketizationMode] = "1";
>+
>+  // Offerer will send both codecs, answerer should choose the one with matching
>+  // packetization mode (and not the first one it sees).
>+  f1_.set_video_codecs({h264_pm0, h264_pm1});
>+  f2_.set_video_codecs({h264_pm1});
>+
>+  MediaSessionOptions opts;
>+  AddMediaSection(MEDIA_TYPE_VIDEO, "video", RtpTransceiverDirection::kSendRecv,
>+                  kActive, &opts);
>+
>+  std::unique_ptr<SessionDescription> offer(f1_.CreateOffer(opts, nullptr));
>+  ASSERT_TRUE(offer);
>+
>+  std::unique_ptr<SessionDescription> answer(
>+      f2_.CreateAnswer(offer.get(), opts, nullptr));
>+  ASSERT_TRUE(answer);
>+
>+  // Answer should have one negotiated codec with packetization-mode=1 using the
>+  // offered payload type.
>+  ASSERT_EQ(1u, answer->contents().size());
>+  auto answer_vcd = answer->contents()[0].media_description()->as_video();
>+  ASSERT_EQ(1u, answer_vcd->codecs().size());
>+  auto answer_codec = answer_vcd->codecs()[0];
>+  EXPECT_EQ(h264_pm1.id, answer_codec.id);
>+}
>+
> class MediaProtocolTest : public ::testing::TestWithParam<const char*> {
>  public:
>   MediaProtocolTest() : f1_(&tdf1_), f2_(&tdf2_) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediastream.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediastream.cc
>index eb78a8bf853..d5924095cdb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediastream.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediastream.cc
>@@ -16,8 +16,7 @@
> namespace webrtc {
> 
> template <class V>
>-static typename V::iterator FindTrack(V* vector,
>-                                      const std::string& track_id) {
>+static typename V::iterator FindTrack(V* vector, const std::string& track_id) {
>   typename V::iterator it = vector->begin();
>   for (; it != vector->end(); ++it) {
>     if ((*it)->id() == track_id) {
>@@ -27,16 +26,13 @@ static typename V::iterator FindTrack(V* vector,
>   return it;
> }
> 
>-rtc::scoped_refptr<MediaStream> MediaStream::Create(
>-    const std::string& label) {
>+rtc::scoped_refptr<MediaStream> MediaStream::Create(const std::string& id) {
>   rtc::RefCountedObject<MediaStream>* stream =
>-      new rtc::RefCountedObject<MediaStream>(label);
>+      new rtc::RefCountedObject<MediaStream>(id);
>   return stream;
> }
> 
>-MediaStream::MediaStream(const std::string& label)
>-    : label_(label) {
>-}
>+MediaStream::MediaStream(const std::string& id) : id_(id) {}
> 
> bool MediaStream::AddTrack(AudioTrackInterface* track) {
>   return AddTrack<AudioTrackVector, AudioTrackInterface>(&audio_tracks_, track);
>@@ -54,16 +50,16 @@ bool MediaStream::RemoveTrack(VideoTrackInterface* track) {
>   return RemoveTrack<VideoTrackVector>(&video_tracks_, track);
> }
> 
>-rtc::scoped_refptr<AudioTrackInterface>
>-MediaStream::FindAudioTrack(const std::string& track_id) {
>+rtc::scoped_refptr<AudioTrackInterface> MediaStream::FindAudioTrack(
>+    const std::string& track_id) {
>   AudioTrackVector::iterator it = FindTrack(&audio_tracks_, track_id);
>   if (it == audio_tracks_.end())
>     return NULL;
>   return *it;
> }
> 
>-rtc::scoped_refptr<VideoTrackInterface>
>-MediaStream::FindVideoTrack(const std::string& track_id) {
>+rtc::scoped_refptr<VideoTrackInterface> MediaStream::FindVideoTrack(
>+    const std::string& track_id) {
>   VideoTrackVector::iterator it = FindTrack(&video_tracks_, track_id);
>   if (it == video_tracks_.end())
>     return NULL;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediastream.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediastream.h
>index 081b8c91168..63b35108b6c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediastream.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediastream.h
>@@ -23,24 +23,24 @@ namespace webrtc {
> 
> class MediaStream : public Notifier<MediaStreamInterface> {
>  public:
>-  static rtc::scoped_refptr<MediaStream> Create(const std::string& label);
>+  static rtc::scoped_refptr<MediaStream> Create(const std::string& id);
> 
>-  std::string label() const override { return label_; }
>+  std::string id() const override { return id_; }
> 
>   bool AddTrack(AudioTrackInterface* track) override;
>   bool AddTrack(VideoTrackInterface* track) override;
>   bool RemoveTrack(AudioTrackInterface* track) override;
>   bool RemoveTrack(VideoTrackInterface* track) override;
>-  rtc::scoped_refptr<AudioTrackInterface>
>-      FindAudioTrack(const std::string& track_id) override;
>-  rtc::scoped_refptr<VideoTrackInterface>
>-      FindVideoTrack(const std::string& track_id) override;
>+  rtc::scoped_refptr<AudioTrackInterface> FindAudioTrack(
>+      const std::string& track_id) override;
>+  rtc::scoped_refptr<VideoTrackInterface> FindVideoTrack(
>+      const std::string& track_id) override;
> 
>   AudioTrackVector GetAudioTracks() override { return audio_tracks_; }
>   VideoTrackVector GetVideoTracks() override { return video_tracks_; }
> 
>  protected:
>-  explicit MediaStream(const std::string& label);
>+  explicit MediaStream(const std::string& id);
> 
>  private:
>   template <typename TrackVector, typename Track>
>@@ -48,7 +48,7 @@ class MediaStream : public Notifier<MediaStreamInterface> {
>   template <typename TrackVector>
>   bool RemoveTrack(TrackVector* Tracks, MediaStreamTrackInterface* track);
> 
>-  std::string label_;
>+  std::string id_;
>   AudioTrackVector audio_tracks_;
>   VideoTrackVector video_tracks_;
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediastream_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediastream_unittest.cc
>index cfd1b50e95b..b81e55c824a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediastream_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediastream_unittest.cc
>@@ -19,7 +19,7 @@
> #include "test/gmock.h"
> #include "test/gtest.h"
> 
>-static const char kStreamLabel1[] = "local_stream_1";
>+static const char kStreamId1[] = "local_stream_1";
> static const char kVideoTrackId[] = "dummy_video_cam_1";
> static const char kAudioTrackId[] = "dummy_microphone_1";
> 
>@@ -50,10 +50,10 @@ class MockObserver : public ObserverInterface {
>   NotifierInterface* notifier_;
> };
> 
>-class MediaStreamTest: public testing::Test {
>+class MediaStreamTest : public testing::Test {
>  protected:
>   virtual void SetUp() {
>-    stream_ = MediaStream::Create(kStreamLabel1);
>+    stream_ = MediaStream::Create(kStreamId1);
>     ASSERT_TRUE(stream_.get() != NULL);
> 
>     video_track_ = VideoTrack::Create(
>@@ -75,8 +75,7 @@ class MediaStreamTest: public testing::Test {
>   void ChangeTrack(MediaStreamTrackInterface* track) {
>     MockObserver observer(track);
> 
>-    EXPECT_CALL(observer, OnChanged())
>-        .Times(Exactly(1));
>+    EXPECT_CALL(observer, OnChanged()).Times(Exactly(1));
>     track->set_enabled(false);
>     EXPECT_FALSE(track->enabled());
>   }
>@@ -98,8 +97,8 @@ TEST_F(MediaStreamTest, GetTrackInfo) {
> 
>   ASSERT_EQ(1u, stream_->GetVideoTracks().size());
>   EXPECT_TRUE(stream_->GetVideoTracks()[0].get() == video_track.get());
>-  EXPECT_TRUE(stream_->FindVideoTrack(video_track->id()).get()
>-              == video_track.get());
>+  EXPECT_TRUE(stream_->FindVideoTrack(video_track->id()).get() ==
>+              video_track.get());
>   video_track = stream_->GetVideoTracks()[0];
>   EXPECT_EQ(0, video_track->id().compare(kVideoTrackId));
>   EXPECT_TRUE(video_track->enabled());
>@@ -111,8 +110,8 @@ TEST_F(MediaStreamTest, GetTrackInfo) {
>   EXPECT_TRUE(audio_track->enabled());
>   ASSERT_EQ(1u, stream_->GetAudioTracks().size());
>   EXPECT_TRUE(stream_->GetAudioTracks()[0].get() == audio_track.get());
>-  EXPECT_TRUE(stream_->FindAudioTrack(audio_track->id()).get()
>-              == audio_track.get());
>+  EXPECT_TRUE(stream_->FindAudioTrack(audio_track->id()).get() ==
>+              audio_track.get());
>   audio_track = stream_->GetAudioTracks()[0];
>   EXPECT_EQ(0, audio_track->id().compare(kAudioTrackId));
>   EXPECT_TRUE(audio_track->enabled());
>@@ -121,8 +120,7 @@ TEST_F(MediaStreamTest, GetTrackInfo) {
> TEST_F(MediaStreamTest, RemoveTrack) {
>   MockObserver observer(stream_);
> 
>-  EXPECT_CALL(observer, OnChanged())
>-      .Times(Exactly(2));
>+  EXPECT_CALL(observer, OnChanged()).Times(Exactly(2));
> 
>   EXPECT_TRUE(stream_->RemoveTrack(audio_track_));
>   EXPECT_FALSE(stream_->RemoveTrack(audio_track_));
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediastreamobserver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediastreamobserver.h
>index 1aed50b50eb..217c9b6468d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediastreamobserver.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/mediastreamobserver.h
>@@ -13,7 +13,7 @@
> 
> #include "api/mediastreaminterface.h"
> #include "rtc_base/scoped_ref_ptr.h"
>-#include "rtc_base/sigslot.h"
>+#include "rtc_base/third_party/sigslot/sigslot.h"
> 
> namespace webrtc {
> 
>@@ -22,7 +22,7 @@ namespace webrtc {
> class MediaStreamObserver : public ObserverInterface {
>  public:
>   explicit MediaStreamObserver(MediaStreamInterface* stream);
>-  ~MediaStreamObserver();
>+  ~MediaStreamObserver() override;
> 
>   const MediaStreamInterface* stream() const { return stream_; }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection.cc
>index 1bebb0509a1..e068f60de15 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection.cc
>@@ -11,17 +11,20 @@
> #include "pc/peerconnection.h"
> 
> #include <algorithm>
>+#include <limits>
> #include <queue>
> #include <set>
> #include <utility>
> #include <vector>
> 
>+#include "absl/memory/memory.h"
> #include "api/jsepicecandidate.h"
> #include "api/jsepsessiondescription.h"
> #include "api/mediaconstraintsinterface.h"
> #include "api/mediastreamproxy.h"
> #include "api/mediastreamtrackproxy.h"
> #include "call/call.h"
>+#include "logging/rtc_event_log/icelogger.h"
> #include "logging/rtc_event_log/output/rtc_event_log_output_file.h"
> #include "logging/rtc_event_log/rtc_event_log.h"
> #include "media/sctp/sctptransport.h"
>@@ -44,12 +47,12 @@
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/numerics/safe_conversions.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/stringencode.h"
> #include "rtc_base/stringutils.h"
> #include "rtc_base/trace_event.h"
> #include "system_wrappers/include/clock.h"
> #include "system_wrappers/include/field_trial.h"
>+#include "system_wrappers/include/metrics.h"
> 
> using cricket::ContentInfo;
> using cricket::ContentInfos;
>@@ -88,11 +91,10 @@ const char kDtlsSrtpSetupFailureRtp[] =
>     "Couldn't set up DTLS-SRTP on RTP channel.";
> const char kDtlsSrtpSetupFailureRtcp[] =
>     "Couldn't set up DTLS-SRTP on RTCP channel.";
>-const char kEnableBundleFailed[] = "Failed to enable BUNDLE.";
> 
> namespace {
> 
>-static const char kDefaultStreamLabel[] = "default";
>+static const char kDefaultStreamId[] = "default";
> static const char kDefaultAudioSenderId[] = "defaulta0";
> static const char kDefaultVideoSenderId[] = "defaultv0";
> 
>@@ -105,16 +107,18 @@ enum {
>   MSG_CREATE_SESSIONDESCRIPTION_FAILED,
>   MSG_GETSTATS,
>   MSG_FREE_DATACHANNELS,
>+  MSG_REPORT_USAGE_PATTERN,
> };
> 
>+static const int REPORT_USAGE_PATTERN_DELAY_MS = 60000;
>+
> struct SetSessionDescriptionMsg : public rtc::MessageData {
>   explicit SetSessionDescriptionMsg(
>       webrtc::SetSessionDescriptionObserver* observer)
>-      : observer(observer) {
>-  }
>+      : observer(observer) {}
> 
>   rtc::scoped_refptr<webrtc::SetSessionDescriptionObserver> observer;
>-  std::string error;
>+  RTCError error;
> };
> 
> struct CreateSessionDescriptionMsg : public rtc::MessageData {
>@@ -123,14 +127,13 @@ struct CreateSessionDescriptionMsg : public rtc::MessageData {
>       : observer(observer) {}
> 
>   rtc::scoped_refptr<webrtc::CreateSessionDescriptionObserver> observer;
>-  std::string error;
>+  RTCError error;
> };
> 
> struct GetStatsMsg : public rtc::MessageData {
>   GetStatsMsg(webrtc::StatsObserver* observer,
>               webrtc::MediaStreamTrackInterface* track)
>-      : observer(observer), track(track) {
>-  }
>+      : observer(observer), track(track) {}
>   rtc::scoped_refptr<webrtc::StatsObserver> observer;
>   rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> track;
> };
>@@ -141,8 +144,8 @@ bool CanAddLocalMediaStream(webrtc::StreamCollectionInterface* current_streams,
>   if (!new_stream || !current_streams) {
>     return false;
>   }
>-  if (current_streams->find(new_stream->label()) != nullptr) {
>-    RTC_LOG(LS_ERROR) << "MediaStream with label " << new_stream->label()
>+  if (current_streams->find(new_stream->id()) != nullptr) {
>+    RTC_LOG(LS_ERROR) << "MediaStream with ID " << new_stream->id()
>                       << " is already added.";
>     return false;
>   }
>@@ -226,19 +229,19 @@ uint32_t ConvertIceTransportTypeToCandidateFilter(
> }
> 
> // Helper to set an error and return from a method.
>-static bool SafeSetError(webrtc::RTCErrorType type, webrtc::RTCError* error) {
>+bool SafeSetError(webrtc::RTCErrorType type, webrtc::RTCError* error) {
>   if (error) {
>     error->set_type(type);
>   }
>   return type == webrtc::RTCErrorType::NONE;
> }
> 
>-static bool SafeSetError(webrtc::RTCError&& error, webrtc::RTCError* error_out) {
>-  bool result = error.ok();
>+bool SafeSetError(webrtc::RTCError error, webrtc::RTCError* error_out) {
>+  bool ok = error.ok();
>   if (error_out) {
>     *error_out = std::move(error);
>   }
>-  return result;
>+  return ok;
> }
> 
> std::string GetSignalingStateString(
>@@ -318,45 +321,100 @@ IceCandidatePairType GetIceCandidatePairCounter(
>   return kIceCandidatePairMax;
> }
> 
>-// Verify that the order of media sections in |new_desc| matches
>-// |existing_desc|. The number of m= sections in |new_desc| should be no less
>-// than |existing_desc|.
>-bool MediaSectionsInSameOrder(const SessionDescription* existing_desc,
>-                              const SessionDescription* new_desc) {
>-  if (!existing_desc || !new_desc) {
>-    return false;
>-  }
>+// Logic to decide if an m= section can be recycled. This means that the new
>+// m= section is not rejected, but the old local or remote m= section is
>+// rejected. |old_content_one| and |old_content_two| refer to the m= section
>+// of the old remote and old local descriptions in no particular order.
>+// We need to check both the old local and remote because either
>+// could be the most current from the latest negotation.
>+bool IsMediaSectionBeingRecycled(SdpType type,
>+                                 const ContentInfo& content,
>+                                 const ContentInfo* old_content_one,
>+                                 const ContentInfo* old_content_two) {
>+  return type == SdpType::kOffer && !content.rejected &&
>+         ((old_content_one && old_content_one->rejected) ||
>+          (old_content_two && old_content_two->rejected));
>+}
> 
>-  if (existing_desc->contents().size() > new_desc->contents().size()) {
>+// Verify that the order of media sections in |new_desc| matches
>+// |current_desc|. The number of m= sections in |new_desc| should be no
>+// less than |current_desc|. In the case of checking an answer's
>+// |new_desc|, the |current_desc| is the last offer that was set as the
>+// local or remote. In the case of checking an offer's |new_desc| we
>+// check against the local and remote descriptions stored from the last
>+// negotiation, because either of these could be the most up to date for
>+// possible rejected m sections. These are the |current_desc| and
>+// |secondary_current_desc|.
>+bool MediaSectionsInSameOrder(const SessionDescription& current_desc,
>+                              const SessionDescription* secondary_current_desc,
>+                              const SessionDescription& new_desc,
>+                              const SdpType type) {
>+  if (current_desc.contents().size() > new_desc.contents().size()) {
>     return false;
>   }
> 
>-  for (size_t i = 0; i < existing_desc->contents().size(); ++i) {
>-    if (existing_desc->contents()[i].rejected) {
>-      // If the media section can be recycled, it's valid for the MID and media
>-      // type to change.
>+  for (size_t i = 0; i < current_desc.contents().size(); ++i) {
>+    const cricket::ContentInfo* secondary_content_info = nullptr;
>+    if (secondary_current_desc &&
>+        i < secondary_current_desc->contents().size()) {
>+      secondary_content_info = &secondary_current_desc->contents()[i];
>+    }
>+    if (IsMediaSectionBeingRecycled(type, new_desc.contents()[i],
>+                                    &current_desc.contents()[i],
>+                                    secondary_content_info)) {
>+      // For new offer descriptions, if the media section can be recycled, it's
>+      // valid for the MID and media type to change.
>       continue;
>     }
>-    if (new_desc->contents()[i].name != existing_desc->contents()[i].name) {
>+    if (new_desc.contents()[i].name != current_desc.contents()[i].name) {
>       return false;
>     }
>     const MediaContentDescription* new_desc_mdesc =
>-        new_desc->contents()[i].media_description();
>-    const MediaContentDescription* existing_desc_mdesc =
>-        existing_desc->contents()[i].media_description();
>-    if (new_desc_mdesc->type() != existing_desc_mdesc->type()) {
>+        new_desc.contents()[i].media_description();
>+    const MediaContentDescription* current_desc_mdesc =
>+        current_desc.contents()[i].media_description();
>+    if (new_desc_mdesc->type() != current_desc_mdesc->type()) {
>       return false;
>     }
>   }
>   return true;
> }
> 
>-bool MediaSectionsHaveSameCount(const SessionDescription* desc1,
>-                                const SessionDescription* desc2) {
>-  if (!desc1 || !desc2) {
>-    return false;
>+bool MediaSectionsHaveSameCount(const SessionDescription& desc1,
>+                                const SessionDescription& desc2) {
>+  return desc1.contents().size() == desc2.contents().size();
>+}
>+
>+void NoteKeyProtocolAndMedia(KeyExchangeProtocolType protocol_type,
>+                             cricket::MediaType media_type) {
>+  RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.KeyProtocol", protocol_type,
>+                            kEnumCounterKeyProtocolMax);
>+  static const std::map<std::pair<KeyExchangeProtocolType, cricket::MediaType>,
>+                        KeyExchangeProtocolMedia>
>+      proto_media_counter_map = {
>+          {{kEnumCounterKeyProtocolDtls, cricket::MEDIA_TYPE_AUDIO},
>+           kEnumCounterKeyProtocolMediaTypeDtlsAudio},
>+          {{kEnumCounterKeyProtocolDtls, cricket::MEDIA_TYPE_VIDEO},
>+           kEnumCounterKeyProtocolMediaTypeDtlsVideo},
>+          {{kEnumCounterKeyProtocolDtls, cricket::MEDIA_TYPE_DATA},
>+           kEnumCounterKeyProtocolMediaTypeDtlsData},
>+          {{kEnumCounterKeyProtocolSdes, cricket::MEDIA_TYPE_AUDIO},
>+           kEnumCounterKeyProtocolMediaTypeSdesAudio},
>+          {{kEnumCounterKeyProtocolSdes, cricket::MEDIA_TYPE_VIDEO},
>+           kEnumCounterKeyProtocolMediaTypeSdesVideo},
>+          {{kEnumCounterKeyProtocolSdes, cricket::MEDIA_TYPE_DATA},
>+           kEnumCounterKeyProtocolMediaTypeSdesData}};
>+
>+  auto it = proto_media_counter_map.find({protocol_type, media_type});
>+  if (it != proto_media_counter_map.end()) {
>+    RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.KeyProtocolByMedia",
>+                              it->second, kEnumCounterKeyProtocolMediaTypeMax);
>   }
>-  return desc1->contents().size() == desc2->contents().size();
>+}
>+
>+void NoteAddIceCandidateResult(int result) {
>+  RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.AddIceCandidate", result,
>+                            kAddIceCandidateMax);
> }
> 
> // Checks that each non-rejected content has SDES crypto keys or a DTLS
>@@ -372,6 +430,10 @@ RTCError VerifyCrypto(const SessionDescription* desc, bool dtls_enabled) {
>     if (content_info.rejected) {
>       continue;
>     }
>+    // Note what media is used with each crypto protocol, for all sections.
>+    NoteKeyProtocolAndMedia(dtls_enabled ? webrtc::kEnumCounterKeyProtocolDtls
>+                                         : webrtc::kEnumCounterKeyProtocolSdes,
>+                            content_info.media_description()->type());
>     const std::string& mid = content_info.name;
>     if (bundle && bundle->HasContentName(mid) &&
>         mid != *(bundle->FirstContentName())) {
>@@ -526,6 +588,37 @@ bool CheckForRemoteIceRestart(const SessionDescriptionInterface* old_desc,
>   return false;
> }
> 
>+// Generates a string error message for SetLocalDescription/SetRemoteDescription
>+// from an RTCError.
>+std::string GetSetDescriptionErrorMessage(cricket::ContentSource source,
>+                                          SdpType type,
>+                                          const RTCError& error) {
>+  std::ostringstream oss;
>+  oss << "Failed to set " << (source == cricket::CS_LOCAL ? "local" : "remote")
>+      << " " << SdpTypeToString(type) << " sdp: " << error.message();
>+  return oss.str();
>+}
>+
>+std::string GetStreamIdsString(rtc::ArrayView<const std::string> stream_ids) {
>+  std::string output = "streams=[";
>+  const char* separator = "";
>+  for (const auto& stream_id : stream_ids) {
>+    output.append(separator).append(stream_id);
>+    separator = ", ";
>+  }
>+  output.append("]");
>+  return output;
>+}
>+
>+absl::optional<int> RTCConfigurationToIceConfigOptionalInt(
>+    int rtc_configuration_parameter) {
>+  if (rtc_configuration_parameter ==
>+      webrtc::PeerConnectionInterface::RTCConfiguration::kUndefined) {
>+    return absl::nullopt;
>+  }
>+  return rtc_configuration_parameter;
>+}
>+
> }  // namespace
> 
> // Upon completion, posts a task to execute the callback of the
>@@ -549,7 +642,7 @@ class PeerConnection::SetRemoteDescriptionObserverAdapter
>     if (error.ok())
>       pc_->PostSetSessionDescriptionSuccess(wrapper_);
>     else
>-      pc_->PostSetSessionDescriptionFailure(wrapper_, error.message());
>+      pc_->PostSetSessionDescriptionFailure(wrapper_, std::move(error));
>   }
> 
>  private:
>@@ -571,10 +664,11 @@ bool PeerConnectionInterface::RTCConfiguration::operator==(
>     bool disable_ipv6;
>     bool disable_ipv6_on_wifi;
>     int max_ipv6_networks;
>+    bool disable_link_local_networks;
>     bool enable_rtp_data_channel;
>-    rtc::Optional<int> screencast_min_bitrate;
>-    rtc::Optional<bool> combined_audio_video_bwe;
>-    rtc::Optional<bool> enable_dtls_srtp;
>+    absl::optional<int> screencast_min_bitrate;
>+    absl::optional<bool> combined_audio_video_bwe;
>+    absl::optional<bool> enable_dtls_srtp;
>     TcpCandidatePolicy tcp_candidate_policy;
>     CandidateNetworkPolicy candidate_network_policy;
>     int audio_jitter_buffer_max_packets;
>@@ -588,10 +682,17 @@ bool PeerConnectionInterface::RTCConfiguration::operator==(
>     bool presume_writable_when_fully_relayed;
>     bool enable_ice_renomination;
>     bool redetermine_role_on_ice_restart;
>-    rtc::Optional<int> ice_check_min_interval;
>-    rtc::Optional<rtc::IntervalRange> ice_regather_interval_range;
>+    absl::optional<int> ice_check_interval_strong_connectivity;
>+    absl::optional<int> ice_check_interval_weak_connectivity;
>+    absl::optional<int> ice_check_min_interval;
>+    absl::optional<int> ice_unwritable_timeout;
>+    absl::optional<int> ice_unwritable_min_checks;
>+    absl::optional<int> stun_candidate_keepalive_interval;
>+    absl::optional<rtc::IntervalRange> ice_regather_interval_range;
>     webrtc::TurnCustomizer* turn_customizer;
>     SdpSemantics sdp_semantics;
>+    absl::optional<rtc::AdapterType> network_preference;
>+    bool active_reset_srtp_params;
>   };
>   static_assert(sizeof(stuff_being_tested_for_equality) == sizeof(*this),
>                 "Did you add something to RTCConfiguration and forget to "
>@@ -615,6 +716,7 @@ bool PeerConnectionInterface::RTCConfiguration::operator==(
>          media_config == o.media_config && disable_ipv6 == o.disable_ipv6 &&
>          disable_ipv6_on_wifi == o.disable_ipv6_on_wifi &&
>          max_ipv6_networks == o.max_ipv6_networks &&
>+         disable_link_local_networks == o.disable_link_local_networks &&
>          enable_rtp_data_channel == o.enable_rtp_data_channel &&
>          screencast_min_bitrate == o.screencast_min_bitrate &&
>          combined_audio_video_bwe == o.combined_audio_video_bwe &&
>@@ -625,10 +727,20 @@ bool PeerConnectionInterface::RTCConfiguration::operator==(
>              o.presume_writable_when_fully_relayed &&
>          enable_ice_renomination == o.enable_ice_renomination &&
>          redetermine_role_on_ice_restart == o.redetermine_role_on_ice_restart &&
>+         ice_check_interval_strong_connectivity ==
>+             o.ice_check_interval_strong_connectivity &&
>+         ice_check_interval_weak_connectivity ==
>+             o.ice_check_interval_weak_connectivity &&
>          ice_check_min_interval == o.ice_check_min_interval &&
>+         ice_unwritable_timeout == o.ice_unwritable_timeout &&
>+         ice_unwritable_min_checks == o.ice_unwritable_min_checks &&
>+         stun_candidate_keepalive_interval ==
>+             o.stun_candidate_keepalive_interval &&
>          ice_regather_interval_range == o.ice_regather_interval_range &&
>          turn_customizer == o.turn_customizer &&
>-         sdp_semantics == o.sdp_semantics;
>+         sdp_semantics == o.sdp_semantics &&
>+         network_preference == o.network_preference &&
>+         active_reset_srtp_params == o.active_reset_srtp_params;
> }
> 
> bool PeerConnectionInterface::RTCConfiguration::operator!=(
>@@ -749,6 +861,7 @@ PeerConnection::~PeerConnection() {
>   // call_ and event_log_ must be destroyed on the worker thread.
>   worker_thread()->Invoke<void>(RTC_FROM_HERE, [this] {
>     call_.reset();
>+    // The event log must outlive call (and any other object that uses it).
>     event_log_.reset();
>   });
> }
>@@ -757,12 +870,12 @@ void PeerConnection::DestroyAllChannels() {
>   // Destroy video channels first since they may have a pointer to a voice
>   // channel.
>   for (auto transceiver : transceivers_) {
>-    if (transceiver->internal()->media_type() == cricket::MEDIA_TYPE_VIDEO) {
>+    if (transceiver->media_type() == cricket::MEDIA_TYPE_VIDEO) {
>       DestroyTransceiverChannel(transceiver);
>     }
>   }
>   for (auto transceiver : transceivers_) {
>-    if (transceiver->internal()->media_type() == cricket::MEDIA_TYPE_AUDIO) {
>+    if (transceiver->media_type() == cricket::MEDIA_TYPE_AUDIO) {
>       DestroyTransceiverChannel(transceiver);
>     }
>   }
>@@ -771,9 +884,7 @@ void PeerConnection::DestroyAllChannels() {
> 
> bool PeerConnection::Initialize(
>     const PeerConnectionInterface::RTCConfiguration& configuration,
>-    std::unique_ptr<cricket::PortAllocator> allocator,
>-    std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
>-    PeerConnectionObserver* observer) {
>+    PeerConnectionDependencies dependencies) {
>   TRACE_EVENT0("webrtc", "PeerConnection::Initialize");
> 
>   RTCError config_error = ValidateConfiguration(configuration);
>@@ -782,45 +893,92 @@ bool PeerConnection::Initialize(
>     return false;
>   }
> 
>-  if (!allocator) {
>+  if (!dependencies.allocator) {
>     RTC_LOG(LS_ERROR)
>         << "PeerConnection initialized without a PortAllocator? "
>-        << "This shouldn't happen if using PeerConnectionFactory.";
>+           "This shouldn't happen if using PeerConnectionFactory.";
>     return false;
>   }
> 
>-  if (!observer) {
>+  if (!dependencies.observer) {
>     // TODO(deadbeef): Why do we do this?
>     RTC_LOG(LS_ERROR) << "PeerConnection initialized without a "
>-                      << "PeerConnectionObserver";
>+                         "PeerConnectionObserver";
>+    return false;
>+  }
>+
>+  observer_ = dependencies.observer;
>+  async_resolver_factory_ = std::move(dependencies.async_resolver_factory);
>+  port_allocator_ = std::move(dependencies.allocator);
>+  tls_cert_verifier_ = std::move(dependencies.tls_cert_verifier);
>+
>+  cricket::ServerAddresses stun_servers;
>+  std::vector<cricket::RelayServerConfig> turn_servers;
>+
>+  RTCErrorType parse_error =
>+      ParseIceServers(configuration.servers, &stun_servers, &turn_servers);
>+  if (parse_error != RTCErrorType::NONE) {
>     return false;
>   }
>-  observer_ = observer;
>-  port_allocator_ = std::move(allocator);
> 
>   // The port allocator lives on the network thread and should be initialized
>   // there.
>   if (!network_thread()->Invoke<bool>(
>-          RTC_FROM_HERE, rtc::Bind(&PeerConnection::InitializePortAllocator_n,
>-                                   this, configuration))) {
>+          RTC_FROM_HERE,
>+          rtc::Bind(&PeerConnection::InitializePortAllocator_n, this,
>+                    stun_servers, turn_servers, configuration))) {
>     return false;
>   }
>+  // If initialization was successful, note if STUN or TURN servers
>+  // were supplied.
>+  if (!stun_servers.empty()) {
>+    NoteUsageEvent(UsageEvent::STUN_SERVER_ADDED);
>+  }
>+  if (!turn_servers.empty()) {
>+    NoteUsageEvent(UsageEvent::TURN_SERVER_ADDED);
>+  }
>+
>+  // Send information about IPv4/IPv6 status.
>+  PeerConnectionAddressFamilyCounter address_family;
>+  if (port_allocator_flags_ & cricket::PORTALLOCATOR_ENABLE_IPV6) {
>+    address_family = kPeerConnection_IPv6;
>+  } else {
>+    address_family = kPeerConnection_IPv4;
>+  }
>+  RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.IPMetrics", address_family,
>+                            kPeerConnectionAddressFamilyCounter_Max);
>+
>+  const PeerConnectionFactoryInterface::Options& options = factory_->options();
> 
>   // RFC 3264: The numeric value of the session id and version in the
>   // o line MUST be representable with a "64 bit signed integer".
>   // Due to this constraint session id |session_id_| is max limited to
>   // LLONG_MAX.
>   session_id_ = rtc::ToString(rtc::CreateRandomId64() & LLONG_MAX);
>-  transport_controller_.reset(factory_->CreateTransportController(
>-      port_allocator_.get(), configuration.redetermine_role_on_ice_restart));
>-  transport_controller_->SetIceRole(cricket::ICEROLE_CONTROLLED);
>-  transport_controller_->SignalConnectionState.connect(
>+  JsepTransportController::Config config;
>+  config.redetermine_role_on_ice_restart =
>+      configuration.redetermine_role_on_ice_restart;
>+  config.ssl_max_version = factory_->options().ssl_max_version;
>+  config.disable_encryption = options.disable_encryption;
>+  config.bundle_policy = configuration.bundle_policy;
>+  config.rtcp_mux_policy = configuration.rtcp_mux_policy;
>+  config.crypto_options = options.crypto_options;
>+  config.transport_observer = this;
>+  config.event_log = event_log_.get();
>+#if defined(ENABLE_EXTERNAL_AUTH)
>+  config.enable_external_auth = true;
>+#endif
>+  config.active_reset_srtp_params = configuration.active_reset_srtp_params;
>+  transport_controller_.reset(new JsepTransportController(
>+      signaling_thread(), network_thread(), port_allocator_.get(),
>+      async_resolver_factory_.get(), config));
>+  transport_controller_->SignalIceConnectionState.connect(
>       this, &PeerConnection::OnTransportControllerConnectionState);
>-  transport_controller_->SignalGatheringState.connect(
>+  transport_controller_->SignalIceGatheringState.connect(
>       this, &PeerConnection::OnTransportControllerGatheringState);
>-  transport_controller_->SignalCandidatesGathered.connect(
>+  transport_controller_->SignalIceCandidatesGathered.connect(
>       this, &PeerConnection::OnTransportControllerCandidatesGathered);
>-  transport_controller_->SignalCandidatesRemoved.connect(
>+  transport_controller_->SignalIceCandidatesRemoved.connect(
>       this, &PeerConnection::OnTransportControllerCandidatesRemoved);
>   transport_controller_->SignalDtlsHandshakeError.connect(
>       this, &PeerConnection::OnTransportControllerDtlsHandshakeError);
>@@ -832,10 +990,6 @@ bool PeerConnection::Initialize(
> 
>   configuration_ = configuration;
> 
>-  const PeerConnectionFactoryInterface::Options& options = factory_->options();
>-
>-  transport_controller_->SetSslMaxProtocolVersion(options.ssl_max_version);
>-
>   // Obtain a certificate from RTCConfiguration if any were provided (optional).
>   rtc::scoped_refptr<rtc::RTCCertificate> certificate;
>   if (!configuration.certificates.empty()) {
>@@ -851,7 +1005,7 @@ bool PeerConnection::Initialize(
>     dtls_enabled_ = false;
>   } else {
>     // Enable DTLS by default if we have an identity store or a certificate.
>-    dtls_enabled_ = (cert_generator || certificate);
>+    dtls_enabled_ = (dependencies.cert_generator || certificate);
>     // |configuration| can override the default |dtls_enabled_| value.
>     if (configuration.enable_dtls_srtp) {
>       dtls_enabled_ = *(configuration.enable_dtls_srtp);
>@@ -885,16 +1039,16 @@ bool PeerConnection::Initialize(
>   // what PeerConnectionDescriptionFactory will do, so make sure that we give it
>   // the right instructions by clearing the variables if needed.
>   if (!dtls_enabled_) {
>-    cert_generator.reset();
>+    dependencies.cert_generator.reset();
>     certificate = nullptr;
>   } else if (certificate) {
>     // Favor generated certificate over the certificate generator.
>-    cert_generator.reset();
>+    dependencies.cert_generator.reset();
>   }
> 
>   webrtc_session_desc_factory_.reset(new WebRtcSessionDescriptionFactory(
>       signaling_thread(), channel_manager(), this, session_id(),
>-      std::move(cert_generator), certificate));
>+      std::move(dependencies.cert_generator), certificate));
>   webrtc_session_desc_factory_->SignalCertificateReady.connect(
>       this, &PeerConnection::OnCertificateReady);
> 
>@@ -914,7 +1068,10 @@ bool PeerConnection::Initialize(
>         RtpTransceiverProxyWithInternal<RtpTransceiver>::Create(
>             signaling_thread(), new RtpTransceiver(cricket::MEDIA_TYPE_VIDEO)));
>   }
>-
>+  int delay_ms =
>+      return_histogram_very_quickly_ ? 0 : REPORT_USAGE_PATTERN_DELAY_MS;
>+  signaling_thread()->PostDelayed(RTC_FROM_HERE, delay_ms, this,
>+                                  MSG_REPORT_USAGE_PATTERN, nullptr);
>   return true;
> }
> 
>@@ -926,20 +1083,28 @@ RTCError PeerConnection::ValidateConfiguration(
>                     "ice_regather_interval_range specified but continual "
>                     "gathering policy is GATHER_ONCE");
>   }
>-  return RTCError::OK();
>+  auto result =
>+      cricket::P2PTransportChannel::ValidateIceConfig(ParseIceConfig(config));
>+  return result;
> }
> 
>-rtc::scoped_refptr<StreamCollectionInterface>
>-PeerConnection::local_streams() {
>+rtc::scoped_refptr<StreamCollectionInterface> PeerConnection::local_streams() {
>+  RTC_CHECK(!IsUnifiedPlan()) << "local_streams is not available with Unified "
>+                                 "Plan SdpSemantics. Please use GetSenders "
>+                                 "instead.";
>   return local_streams_;
> }
> 
>-rtc::scoped_refptr<StreamCollectionInterface>
>-PeerConnection::remote_streams() {
>+rtc::scoped_refptr<StreamCollectionInterface> PeerConnection::remote_streams() {
>+  RTC_CHECK(!IsUnifiedPlan()) << "remote_streams is not available with Unified "
>+                                 "Plan SdpSemantics. Please use GetReceivers "
>+                                 "instead.";
>   return remote_streams_;
> }
> 
> bool PeerConnection::AddStream(MediaStreamInterface* local_stream) {
>+  RTC_CHECK(!IsUnifiedPlan()) << "AddStream is not available with Unified Plan "
>+                                 "SdpSemantics. Please use AddTrack instead.";
>   TRACE_EVENT0("webrtc", "PeerConnection::AddStream");
>   if (IsClosed()) {
>     return false;
>@@ -968,11 +1133,14 @@ bool PeerConnection::AddStream(MediaStreamInterface* local_stream) {
>   }
> 
>   stats_->AddStream(local_stream);
>-  observer_->OnRenegotiationNeeded();
>+  Observer()->OnRenegotiationNeeded();
>   return true;
> }
> 
> void PeerConnection::RemoveStream(MediaStreamInterface* local_stream) {
>+  RTC_CHECK(!IsUnifiedPlan()) << "RemoveStream is not available with Unified "
>+                                 "Plan SdpSemantics. Please use RemoveTrack "
>+                                 "instead.";
>   TRACE_EVENT0("webrtc", "PeerConnection::RemoveStream");
>   if (!IsClosed()) {
>     for (const auto& track : local_stream->GetAudioTracks()) {
>@@ -987,39 +1155,19 @@ void PeerConnection::RemoveStream(MediaStreamInterface* local_stream) {
>       std::remove_if(
>           stream_observers_.begin(), stream_observers_.end(),
>           [local_stream](const std::unique_ptr<MediaStreamObserver>& observer) {
>-            return observer->stream()->label().compare(local_stream->label()) ==
>-                   0;
>+            return observer->stream()->id().compare(local_stream->id()) == 0;
>           }),
>       stream_observers_.end());
> 
>   if (IsClosed()) {
>     return;
>   }
>-  observer_->OnRenegotiationNeeded();
>-}
>-
>-rtc::scoped_refptr<RtpSenderInterface> PeerConnection::AddTrack(
>-    MediaStreamTrackInterface* track,
>-    std::vector<MediaStreamInterface*> streams) {
>-  TRACE_EVENT0("webrtc", "PeerConnection::AddTrack");
>-  std::vector<std::string> stream_labels;
>-  for (auto* stream : streams) {
>-    if (!stream) {
>-      RTC_LOG(LS_ERROR) << "Stream list has null element.";
>-      return nullptr;
>-    }
>-    stream_labels.push_back(stream->label());
>-  }
>-  auto sender_or_error = AddTrack(track, stream_labels);
>-  if (!sender_or_error.ok()) {
>-    return nullptr;
>-  }
>-  return sender_or_error.MoveValue();
>+  Observer()->OnRenegotiationNeeded();
> }
> 
> RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>> PeerConnection::AddTrack(
>     rtc::scoped_refptr<MediaStreamTrackInterface> track,
>-    const std::vector<std::string>& stream_labels) {
>+    const std::vector<std::string>& stream_ids) {
>   TRACE_EVENT0("webrtc", "PeerConnection::AddTrack");
>   if (!track) {
>     LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, "Track is null.");
>@@ -1029,12 +1177,6 @@ RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>> PeerConnection::AddTrack(
>     LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
>                          "Track has invalid kind: " + track->kind());
>   }
>-  // TODO(bugs.webrtc.org/7932): Support adding a track to multiple streams.
>-  if (stream_labels.size() > 1u) {
>-    LOG_AND_RETURN_ERROR(
>-        RTCErrorType::UNSUPPORTED_OPERATION,
>-        "AddTrack with more than one stream is not currently supported.");
>-  }
>   if (IsClosed()) {
>     LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_STATE,
>                          "PeerConnection is closed.");
>@@ -1044,20 +1186,11 @@ RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>> PeerConnection::AddTrack(
>         RTCErrorType::INVALID_PARAMETER,
>         "Sender already exists for track " + track->id() + ".");
>   }
>-  // TODO(bugs.webrtc.org/7933): MediaSession expects the sender to have exactly
>-  // one stream. AddTrackInternal will return an error if there is more than one
>-  // stream, but if the caller specifies none then we need to generate a random
>-  // stream label.
>-  std::vector<std::string> adjusted_stream_labels = stream_labels;
>-  if (stream_labels.empty()) {
>-    adjusted_stream_labels.push_back(rtc::CreateRandomUuid());
>-  }
>-  RTC_DCHECK_EQ(1, adjusted_stream_labels.size());
>   auto sender_or_error =
>-      (IsUnifiedPlan() ? AddTrackUnifiedPlan(track, adjusted_stream_labels)
>-                       : AddTrackPlanB(track, adjusted_stream_labels));
>+      (IsUnifiedPlan() ? AddTrackUnifiedPlan(track, stream_ids)
>+                       : AddTrackPlanB(track, stream_ids));
>   if (sender_or_error.ok()) {
>-    observer_->OnRenegotiationNeeded();
>+    Observer()->OnRenegotiationNeeded();
>     stats_->AddTrack(track);
>   }
>   return sender_or_error;
>@@ -1066,30 +1199,38 @@ RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>> PeerConnection::AddTrack(
> RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>>
> PeerConnection::AddTrackPlanB(
>     rtc::scoped_refptr<MediaStreamTrackInterface> track,
>-    const std::vector<std::string>& stream_labels) {
>+    const std::vector<std::string>& stream_ids) {
>+  if (stream_ids.size() > 1u) {
>+    LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_OPERATION,
>+                         "AddTrack with more than one stream is not "
>+                         "supported with Plan B semantics.");
>+  }
>+  std::vector<std::string> adjusted_stream_ids = stream_ids;
>+  if (adjusted_stream_ids.empty()) {
>+    adjusted_stream_ids.push_back(rtc::CreateRandomUuid());
>+  }
>   cricket::MediaType media_type =
>       (track->kind() == MediaStreamTrackInterface::kAudioKind
>            ? cricket::MEDIA_TYPE_AUDIO
>            : cricket::MEDIA_TYPE_VIDEO);
>-  auto new_sender = CreateSender(media_type, track, stream_labels);
>+  auto new_sender =
>+      CreateSender(media_type, track->id(), track, adjusted_stream_ids);
>   if (track->kind() == MediaStreamTrackInterface::kAudioKind) {
>-    static_cast<AudioRtpSender*>(new_sender->internal())
>-        ->SetMediaChannel(voice_media_channel());
>+    new_sender->internal()->SetVoiceMediaChannel(voice_media_channel());
>     GetAudioTransceiver()->internal()->AddSender(new_sender);
>     const RtpSenderInfo* sender_info =
>         FindSenderInfo(local_audio_sender_infos_,
>-                       new_sender->internal()->stream_id(), track->id());
>+                       new_sender->internal()->stream_ids()[0], track->id());
>     if (sender_info) {
>       new_sender->internal()->SetSsrc(sender_info->first_ssrc);
>     }
>   } else {
>     RTC_DCHECK_EQ(MediaStreamTrackInterface::kVideoKind, track->kind());
>-    static_cast<VideoRtpSender*>(new_sender->internal())
>-        ->SetMediaChannel(video_media_channel());
>+    new_sender->internal()->SetVideoMediaChannel(video_media_channel());
>     GetVideoTransceiver()->internal()->AddSender(new_sender);
>     const RtpSenderInfo* sender_info =
>         FindSenderInfo(local_video_sender_infos_,
>-                       new_sender->internal()->stream_id(), track->id());
>+                       new_sender->internal()->stream_ids()[0], track->id());
>     if (sender_info) {
>       new_sender->internal()->SetSsrc(sender_info->first_ssrc);
>     }
>@@ -1100,26 +1241,40 @@ PeerConnection::AddTrackPlanB(
> RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>>
> PeerConnection::AddTrackUnifiedPlan(
>     rtc::scoped_refptr<MediaStreamTrackInterface> track,
>-    const std::vector<std::string>& stream_labels) {
>+    const std::vector<std::string>& stream_ids) {
>   auto transceiver = FindFirstTransceiverForAddedTrack(track);
>   if (transceiver) {
>+    RTC_LOG(LS_INFO) << "Reusing an existing "
>+                     << cricket::MediaTypeToString(transceiver->media_type())
>+                     << " transceiver for AddTrack.";
>     if (transceiver->direction() == RtpTransceiverDirection::kRecvOnly) {
>-      transceiver->SetDirection(RtpTransceiverDirection::kSendRecv);
>+      transceiver->internal()->set_direction(
>+          RtpTransceiverDirection::kSendRecv);
>     } else if (transceiver->direction() == RtpTransceiverDirection::kInactive) {
>-      transceiver->SetDirection(RtpTransceiverDirection::kSendOnly);
>+      transceiver->internal()->set_direction(
>+          RtpTransceiverDirection::kSendOnly);
>     }
>     transceiver->sender()->SetTrack(track);
>-    transceiver->internal()->sender_internal()->set_stream_ids(stream_labels);
>+    transceiver->internal()->sender_internal()->set_stream_ids(stream_ids);
>   } else {
>     cricket::MediaType media_type =
>         (track->kind() == MediaStreamTrackInterface::kAudioKind
>              ? cricket::MEDIA_TYPE_AUDIO
>              : cricket::MEDIA_TYPE_VIDEO);
>-    auto sender = CreateSender(media_type, track, stream_labels);
>+    RTC_LOG(LS_INFO) << "Adding " << cricket::MediaTypeToString(media_type)
>+                     << " transceiver in response to a call to AddTrack.";
>+    std::string sender_id = track->id();
>+    // Avoid creating a sender with an existing ID by generating a random ID.
>+    // This can happen if this is the second time AddTrack has created a sender
>+    // for this track.
>+    if (FindSenderById(sender_id)) {
>+      sender_id = rtc::CreateRandomUuid();
>+    }
>+    auto sender = CreateSender(media_type, sender_id, track, stream_ids);
>     auto receiver = CreateReceiver(media_type, rtc::CreateRandomUuid());
>     transceiver = CreateAndAddTransceiver(sender, receiver);
>     transceiver->internal()->set_created_by_addtrack(true);
>-    transceiver->SetDirection(RtpTransceiverDirection::kSendRecv);
>+    transceiver->internal()->set_direction(RtpTransceiverDirection::kSendRecv);
>   }
>   return transceiver->sender();
> }
>@@ -1130,9 +1285,10 @@ PeerConnection::FindFirstTransceiverForAddedTrack(
>   RTC_DCHECK(track);
>   for (auto transceiver : transceivers_) {
>     if (!transceiver->sender()->track() &&
>-        cricket::MediaTypeToString(transceiver->internal()->media_type()) ==
>+        cricket::MediaTypeToString(transceiver->media_type()) ==
>             track->kind() &&
>-        !transceiver->internal()->has_ever_been_used_to_send()) {
>+        !transceiver->internal()->has_ever_been_used_to_send() &&
>+        !transceiver->stopped()) {
>       return transceiver;
>     }
>   }
>@@ -1141,10 +1297,10 @@ PeerConnection::FindFirstTransceiverForAddedTrack(
> 
> bool PeerConnection::RemoveTrack(RtpSenderInterface* sender) {
>   TRACE_EVENT0("webrtc", "PeerConnection::RemoveTrack");
>-  return RemoveTrackInternal(sender).ok();
>+  return RemoveTrackNew(sender).ok();
> }
> 
>-RTCError PeerConnection::RemoveTrackInternal(
>+RTCError PeerConnection::RemoveTrackNew(
>     rtc::scoped_refptr<RtpSenderInterface> sender) {
>   if (!sender) {
>     LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, "Sender is null.");
>@@ -1160,9 +1316,11 @@ RTCError PeerConnection::RemoveTrackInternal(
>     }
>     sender->SetTrack(nullptr);
>     if (transceiver->direction() == RtpTransceiverDirection::kSendRecv) {
>-      transceiver->internal()->SetDirection(RtpTransceiverDirection::kRecvOnly);
>+      transceiver->internal()->set_direction(
>+          RtpTransceiverDirection::kRecvOnly);
>     } else if (transceiver->direction() == RtpTransceiverDirection::kSendOnly) {
>-      transceiver->internal()->SetDirection(RtpTransceiverDirection::kInactive);
>+      transceiver->internal()->set_direction(
>+          RtpTransceiverDirection::kInactive);
>     }
>   } else {
>     bool removed;
>@@ -1178,7 +1336,7 @@ RTCError PeerConnection::RemoveTrackInternal(
>           "Couldn't find sender " + sender->id() + " to remove.");
>     }
>   }
>-  observer_->OnRenegotiationNeeded();
>+  Observer()->OnRenegotiationNeeded();
>   return RTCError::OK();
> }
> 
>@@ -1203,11 +1361,8 @@ RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
> PeerConnection::AddTransceiver(
>     rtc::scoped_refptr<MediaStreamTrackInterface> track,
>     const RtpTransceiverInit& init) {
>-  if (!IsUnifiedPlan()) {
>-    LOG_AND_RETURN_ERROR(
>-        RTCErrorType::INTERNAL_ERROR,
>-        "AddTransceiver only supported when Unified Plan is enabled.");
>-  }
>+  RTC_CHECK(IsUnifiedPlan())
>+      << "AddTransceiver is only available with Unified Plan SdpSemantics";
>   if (!track) {
>     LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, "track is null");
>   }
>@@ -1231,11 +1386,8 @@ PeerConnection::AddTransceiver(cricket::MediaType media_type) {
> RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
> PeerConnection::AddTransceiver(cricket::MediaType media_type,
>                                const RtpTransceiverInit& init) {
>-  if (!IsUnifiedPlan()) {
>-    LOG_AND_RETURN_ERROR(
>-        RTCErrorType::INTERNAL_ERROR,
>-        "AddTransceiver only supported when Unified Plan is enabled.");
>-  }
>+  RTC_CHECK(IsUnifiedPlan())
>+      << "AddTransceiver is only available with Unified Plan SdpSemantics";
>   if (!(media_type == cricket::MEDIA_TYPE_AUDIO ||
>         media_type == cricket::MEDIA_TYPE_VIDEO)) {
>     LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
>@@ -1248,7 +1400,8 @@ RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
> PeerConnection::AddTransceiver(
>     cricket::MediaType media_type,
>     rtc::scoped_refptr<MediaStreamTrackInterface> track,
>-    const RtpTransceiverInit& init) {
>+    const RtpTransceiverInit& init,
>+    bool fire_callback) {
>   RTC_DCHECK((media_type == cricket::MEDIA_TYPE_AUDIO ||
>               media_type == cricket::MEDIA_TYPE_VIDEO));
>   if (track) {
>@@ -1260,21 +1413,21 @@ PeerConnection::AddTransceiver(
> 
>   // TODO(bugs.webrtc.org/7600): Verify init.
> 
>-  if (init.stream_labels.size() > 1u) {
>-    LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_PARAMETER,
>-                         "More than one stream is not yet supported.");
>-  }
>-
>-  std::vector<std::string> stream_labels = {!init.stream_labels.empty()
>-                                                ? init.stream_labels[0]
>-                                                : rtc::CreateRandomUuid()};
>-
>-  auto sender = CreateSender(media_type, track, stream_labels);
>+  RTC_LOG(LS_INFO) << "Adding " << cricket::MediaTypeToString(media_type)
>+                   << " transceiver in response to a call to AddTransceiver.";
>+  // Set the sender ID equal to the track ID if the track is specified unless
>+  // that sender ID is already in use.
>+  std::string sender_id =
>+      (track && !FindSenderById(track->id()) ? track->id()
>+                                             : rtc::CreateRandomUuid());
>+  auto sender = CreateSender(media_type, sender_id, track, init.stream_ids);
>   auto receiver = CreateReceiver(media_type, rtc::CreateRandomUuid());
>   auto transceiver = CreateAndAddTransceiver(sender, receiver);
>   transceiver->internal()->set_direction(init.direction);
> 
>-  observer_->OnRenegotiationNeeded();
>+  if (fire_callback) {
>+    Observer()->OnRenegotiationNeeded();
>+  }
> 
>   return rtc::scoped_refptr<RtpTransceiverInterface>(transceiver);
> }
>@@ -1282,28 +1435,28 @@ PeerConnection::AddTransceiver(
> rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>
> PeerConnection::CreateSender(
>     cricket::MediaType media_type,
>+    const std::string& id,
>     rtc::scoped_refptr<MediaStreamTrackInterface> track,
>-    const std::vector<std::string>& stream_labels) {
>+    const std::vector<std::string>& stream_ids) {
>   rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> sender;
>   if (media_type == cricket::MEDIA_TYPE_AUDIO) {
>     RTC_DCHECK(!track ||
>                (track->kind() == MediaStreamTrackInterface::kAudioKind));
>     sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
>         signaling_thread(),
>-        new AudioRtpSender(worker_thread(),
>-                           static_cast<AudioTrackInterface*>(track.get()),
>-                           stream_labels, stats_.get()));
>+        new AudioRtpSender(worker_thread(), id, stats_.get()));
>+    NoteUsageEvent(UsageEvent::AUDIO_ADDED);
>   } else {
>     RTC_DCHECK_EQ(media_type, cricket::MEDIA_TYPE_VIDEO);
>     RTC_DCHECK(!track ||
>                (track->kind() == MediaStreamTrackInterface::kVideoKind));
>     sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
>-        signaling_thread(),
>-        new VideoRtpSender(worker_thread(),
>-                           static_cast<VideoTrackInterface*>(track.get()),
>-                           stream_labels));
>+        signaling_thread(), new VideoRtpSender(worker_thread(), id));
>+    NoteUsageEvent(UsageEvent::VIDEO_ADDED);
>   }
>-  sender->internal()->set_stream_ids(stream_labels);
>+  bool set_track_succeeded = sender->SetTrack(track);
>+  RTC_DCHECK(set_track_succeeded);
>+  sender->internal()->set_stream_ids(stream_ids);
>   return sender;
> }
> 
>@@ -1314,13 +1467,15 @@ PeerConnection::CreateReceiver(cricket::MediaType media_type,
>       receiver;
>   if (media_type == cricket::MEDIA_TYPE_AUDIO) {
>     receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
>-        signaling_thread(),
>-        new AudioRtpReceiver(worker_thread(), receiver_id, {}, 0, nullptr));
>+        signaling_thread(), new AudioRtpReceiver(worker_thread(), receiver_id,
>+                                                 std::vector<std::string>({})));
>+    NoteUsageEvent(UsageEvent::AUDIO_ADDED);
>   } else {
>     RTC_DCHECK_EQ(media_type, cricket::MEDIA_TYPE_VIDEO);
>     receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
>-        signaling_thread(),
>-        new VideoRtpReceiver(worker_thread(), receiver_id, {}, 0, nullptr));
>+        signaling_thread(), new VideoRtpReceiver(worker_thread(), receiver_id,
>+                                                 std::vector<std::string>({})));
>+    NoteUsageEvent(UsageEvent::VIDEO_ADDED);
>   }
>   return receiver;
> }
>@@ -1330,57 +1485,61 @@ PeerConnection::CreateAndAddTransceiver(
>     rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> sender,
>     rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
>         receiver) {
>+  // Ensure that the new sender does not have an ID that is already in use by
>+  // another sender.
>+  // Allow receiver IDs to conflict since those come from remote SDP (which
>+  // could be invalid, but should not cause a crash).
>+  RTC_DCHECK(!FindSenderById(sender->id()));
>   auto transceiver = RtpTransceiverProxyWithInternal<RtpTransceiver>::Create(
>       signaling_thread(), new RtpTransceiver(sender, receiver));
>   transceivers_.push_back(transceiver);
>+  transceiver->internal()->SignalNegotiationNeeded.connect(
>+      this, &PeerConnection::OnNegotiationNeeded);
>   return transceiver;
> }
> 
>-rtc::scoped_refptr<DtmfSenderInterface> PeerConnection::CreateDtmfSender(
>-    AudioTrackInterface* track) {
>-  TRACE_EVENT0("webrtc", "PeerConnection::CreateDtmfSender");
>-  if (IsClosed()) {
>-    return nullptr;
>-  }
>-  if (!track) {
>-    RTC_LOG(LS_ERROR) << "CreateDtmfSender - track is NULL.";
>-    return nullptr;
>-  }
>-  auto track_sender = FindSenderForTrack(track);
>-  if (!track_sender) {
>-    RTC_LOG(LS_ERROR) << "CreateDtmfSender called with a non-added track.";
>-    return nullptr;
>-  }
>-
>-  return track_sender->GetDtmfSender();
>+void PeerConnection::OnNegotiationNeeded() {
>+  RTC_DCHECK_RUN_ON(signaling_thread());
>+  RTC_DCHECK(!IsClosed());
>+  Observer()->OnRenegotiationNeeded();
> }
> 
> rtc::scoped_refptr<RtpSenderInterface> PeerConnection::CreateSender(
>     const std::string& kind,
>     const std::string& stream_id) {
>+  RTC_CHECK(!IsUnifiedPlan()) << "CreateSender is not available with Unified "
>+                                 "Plan SdpSemantics. Please use AddTransceiver "
>+                                 "instead.";
>   TRACE_EVENT0("webrtc", "PeerConnection::CreateSender");
>   if (IsClosed()) {
>     return nullptr;
>   }
> 
>-  std::vector<std::string> stream_labels;
>-  if (!stream_id.empty()) {
>-    stream_labels.push_back(stream_id);
>+  // Internally we need to have one stream with Plan B semantics, so we
>+  // generate a random stream ID if not specified.
>+  std::vector<std::string> stream_ids;
>+  if (stream_id.empty()) {
>+    stream_ids.push_back(rtc::CreateRandomUuid());
>+    RTC_LOG(LS_INFO)
>+        << "No stream_id specified for sender. Generated stream ID: "
>+        << stream_ids[0];
>+  } else {
>+    stream_ids.push_back(stream_id);
>   }
> 
>   // TODO(steveanton): Move construction of the RtpSenders to RtpTransceiver.
>   rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> new_sender;
>   if (kind == MediaStreamTrackInterface::kAudioKind) {
>-    auto* audio_sender = new AudioRtpSender(worker_thread(), nullptr,
>-                                            stream_labels, stats_.get());
>-    audio_sender->SetMediaChannel(voice_media_channel());
>+    auto* audio_sender = new AudioRtpSender(
>+        worker_thread(), rtc::CreateRandomUuid(), stats_.get());
>+    audio_sender->SetVoiceMediaChannel(voice_media_channel());
>     new_sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
>         signaling_thread(), audio_sender);
>     GetAudioTransceiver()->internal()->AddSender(new_sender);
>   } else if (kind == MediaStreamTrackInterface::kVideoKind) {
>     auto* video_sender =
>-        new VideoRtpSender(worker_thread(), nullptr, stream_labels);
>-    video_sender->SetMediaChannel(video_media_channel());
>+        new VideoRtpSender(worker_thread(), rtc::CreateRandomUuid());
>+    video_sender->SetVideoMediaChannel(video_media_channel());
>     new_sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
>         signaling_thread(), video_sender);
>     GetVideoTransceiver()->internal()->AddSender(new_sender);
>@@ -1388,6 +1547,7 @@ rtc::scoped_refptr<RtpSenderInterface> PeerConnection::CreateSender(
>     RTC_LOG(LS_ERROR) << "CreateSender called with invalid kind: " << kind;
>     return nullptr;
>   }
>+  new_sender->internal()->set_stream_ids(stream_ids);
> 
>   return new_sender;
> }
>@@ -1437,7 +1597,8 @@ PeerConnection::GetReceiversInternal() const {
> 
> std::vector<rtc::scoped_refptr<RtpTransceiverInterface>>
> PeerConnection::GetTransceivers() const {
>-  RTC_DCHECK(IsUnifiedPlan());
>+  RTC_CHECK(IsUnifiedPlan())
>+      << "GetTransceivers is only supported with Unified Plan SdpSemantics.";
>   std::vector<rtc::scoped_refptr<RtpTransceiverInterface>> all_transceivers;
>   for (auto transceiver : transceivers_) {
>     all_transceivers.push_back(transceiver);
>@@ -1469,10 +1630,68 @@ bool PeerConnection::GetStats(StatsObserver* observer,
> }
> 
> void PeerConnection::GetStats(RTCStatsCollectorCallback* callback) {
>+  TRACE_EVENT0("webrtc", "PeerConnection::GetStats");
>   RTC_DCHECK(stats_collector_);
>+  RTC_DCHECK(callback);
>   stats_collector_->GetStatsReport(callback);
> }
> 
>+void PeerConnection::GetStats(
>+    rtc::scoped_refptr<RtpSenderInterface> selector,
>+    rtc::scoped_refptr<RTCStatsCollectorCallback> callback) {
>+  TRACE_EVENT0("webrtc", "PeerConnection::GetStats");
>+  RTC_DCHECK(callback);
>+  RTC_DCHECK(stats_collector_);
>+  rtc::scoped_refptr<RtpSenderInternal> internal_sender;
>+  if (selector) {
>+    for (const auto& proxy_transceiver : transceivers_) {
>+      for (const auto& proxy_sender :
>+           proxy_transceiver->internal()->senders()) {
>+        if (proxy_sender == selector) {
>+          internal_sender = proxy_sender->internal();
>+          break;
>+        }
>+      }
>+      if (internal_sender)
>+        break;
>+    }
>+  }
>+  // If there is no |internal_sender|Â then |selector| is either null or does not
>+  // belong to the PeerConnection (in Plan B, senders can be removed from the
>+  // PeerConnection). This means that "all the stats objects representing the
>+  // selector" is an empty set. Invoking GetStatsReport() with a null selector
>+  // produces an empty stats report.
>+  stats_collector_->GetStatsReport(internal_sender, callback);
>+}
>+
>+void PeerConnection::GetStats(
>+    rtc::scoped_refptr<RtpReceiverInterface> selector,
>+    rtc::scoped_refptr<RTCStatsCollectorCallback> callback) {
>+  TRACE_EVENT0("webrtc", "PeerConnection::GetStats");
>+  RTC_DCHECK(callback);
>+  RTC_DCHECK(stats_collector_);
>+  rtc::scoped_refptr<RtpReceiverInternal> internal_receiver;
>+  if (selector) {
>+    for (const auto& proxy_transceiver : transceivers_) {
>+      for (const auto& proxy_receiver :
>+           proxy_transceiver->internal()->receivers()) {
>+        if (proxy_receiver == selector) {
>+          internal_receiver = proxy_receiver->internal();
>+          break;
>+        }
>+      }
>+      if (internal_receiver)
>+        break;
>+    }
>+  }
>+  // If there is no |internal_receiver|Â then |selector| is either null or does
>+  // not belong to the PeerConnection (in Plan B, receivers can be removed from
>+  // the PeerConnection). This means that "all the stats objects representing
>+  // the selector" is an empty set. Invoking GetStatsReport() with a null
>+  // selector produces an empty stats report.
>+  stats_collector_->GetStatsReport(internal_receiver, callback);
>+}
>+
> PeerConnectionInterface::SignalingState PeerConnection::signaling_state() {
>   return signaling_state_;
> }
>@@ -1487,8 +1706,7 @@ PeerConnection::ice_gathering_state() {
>   return ice_gathering_state_;
> }
> 
>-rtc::scoped_refptr<DataChannelInterface>
>-PeerConnection::CreateDataChannel(
>+rtc::scoped_refptr<DataChannelInterface> PeerConnection::CreateDataChannel(
>     const std::string& label,
>     const DataChannelInit* config) {
>   TRACE_EVENT0("webrtc", "PeerConnection::CreateDataChannel");
>@@ -1508,9 +1726,9 @@ PeerConnection::CreateDataChannel(
>   // Trigger the onRenegotiationNeeded event for every new RTP DataChannel, or
>   // the first SCTP DataChannel.
>   if (data_channel_type() == cricket::DCT_RTP || first_datachannel) {
>-    observer_->OnRenegotiationNeeded();
>+    Observer()->OnRenegotiationNeeded();
>   }
>-
>+  NoteUsageEvent(UsageEvent::DATA_ADDED);
>   return DataChannelProxy::Create(signaling_thread(), channel.get());
> }
> 
>@@ -1543,22 +1761,106 @@ void PeerConnection::CreateOffer(CreateSessionDescriptionObserver* observer,
>   if (IsClosed()) {
>     std::string error = "CreateOffer called when PeerConnection is closed.";
>     RTC_LOG(LS_ERROR) << error;
>-    PostCreateSessionDescriptionFailure(observer, error);
>+    PostCreateSessionDescriptionFailure(
>+        observer, RTCError(RTCErrorType::INVALID_STATE, std::move(error)));
>     return;
>   }
> 
>   if (!ValidateOfferAnswerOptions(options)) {
>     std::string error = "CreateOffer called with invalid options.";
>     RTC_LOG(LS_ERROR) << error;
>-    PostCreateSessionDescriptionFailure(observer, error);
>+    PostCreateSessionDescriptionFailure(
>+        observer, RTCError(RTCErrorType::INVALID_PARAMETER, std::move(error)));
>     return;
>   }
> 
>+  // Legacy handling for offer_to_receive_audio and offer_to_receive_video.
>+  // Specified in WebRTC section 4.4.3.2 "Legacy configuration extensions".
>+  if (IsUnifiedPlan()) {
>+    RTCError error = HandleLegacyOfferOptions(options);
>+    if (!error.ok()) {
>+      PostCreateSessionDescriptionFailure(observer, std::move(error));
>+      return;
>+    }
>+  }
>+
>   cricket::MediaSessionOptions session_options;
>   GetOptionsForOffer(options, &session_options);
>   webrtc_session_desc_factory_->CreateOffer(observer, options, session_options);
> }
> 
>+RTCError PeerConnection::HandleLegacyOfferOptions(
>+    const RTCOfferAnswerOptions& options) {
>+  RTC_DCHECK(IsUnifiedPlan());
>+
>+  if (options.offer_to_receive_audio == 0) {
>+    RemoveRecvDirectionFromReceivingTransceiversOfType(
>+        cricket::MEDIA_TYPE_AUDIO);
>+  } else if (options.offer_to_receive_audio == 1) {
>+    AddUpToOneReceivingTransceiverOfType(cricket::MEDIA_TYPE_AUDIO);
>+  } else if (options.offer_to_receive_audio > 1) {
>+    LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_PARAMETER,
>+                         "offer_to_receive_audio > 1 is not supported.");
>+  }
>+
>+  if (options.offer_to_receive_video == 0) {
>+    RemoveRecvDirectionFromReceivingTransceiversOfType(
>+        cricket::MEDIA_TYPE_VIDEO);
>+  } else if (options.offer_to_receive_video == 1) {
>+    AddUpToOneReceivingTransceiverOfType(cricket::MEDIA_TYPE_VIDEO);
>+  } else if (options.offer_to_receive_video > 1) {
>+    LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_PARAMETER,
>+                         "offer_to_receive_video > 1 is not supported.");
>+  }
>+
>+  return RTCError::OK();
>+}
>+
>+void PeerConnection::RemoveRecvDirectionFromReceivingTransceiversOfType(
>+    cricket::MediaType media_type) {
>+  for (auto transceiver : GetReceivingTransceiversOfType(media_type)) {
>+    RtpTransceiverDirection new_direction =
>+        RtpTransceiverDirectionWithRecvSet(transceiver->direction(), false);
>+    if (new_direction != transceiver->direction()) {
>+      RTC_LOG(LS_INFO) << "Changing " << cricket::MediaTypeToString(media_type)
>+                       << " transceiver (MID="
>+                       << transceiver->mid().value_or("<not set>") << ") from "
>+                       << RtpTransceiverDirectionToString(
>+                              transceiver->direction())
>+                       << " to "
>+                       << RtpTransceiverDirectionToString(new_direction)
>+                       << " since CreateOffer specified offer_to_receive=0";
>+      transceiver->internal()->set_direction(new_direction);
>+    }
>+  }
>+}
>+
>+void PeerConnection::AddUpToOneReceivingTransceiverOfType(
>+    cricket::MediaType media_type) {
>+  if (GetReceivingTransceiversOfType(media_type).empty()) {
>+    RTC_LOG(LS_INFO)
>+        << "Adding one recvonly " << cricket::MediaTypeToString(media_type)
>+        << " transceiver since CreateOffer specified offer_to_receive=1";
>+    RtpTransceiverInit init;
>+    init.direction = RtpTransceiverDirection::kRecvOnly;
>+    AddTransceiver(media_type, nullptr, init, /*fire_callback=*/false);
>+  }
>+}
>+
>+std::vector<rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>>
>+PeerConnection::GetReceivingTransceiversOfType(cricket::MediaType media_type) {
>+  std::vector<
>+      rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>>
>+      receiving_transceivers;
>+  for (auto transceiver : transceivers_) {
>+    if (!transceiver->stopped() && transceiver->media_type() == media_type &&
>+        RtpTransceiverDirectionHasRecv(transceiver->direction())) {
>+      receiving_transceivers.push_back(transceiver);
>+    }
>+  }
>+  return receiving_transceivers;
>+}
>+
> void PeerConnection::CreateAnswer(
>     CreateSessionDescriptionObserver* observer,
>     const MediaConstraintsInterface* constraints) {
>@@ -1574,7 +1876,8 @@ void PeerConnection::CreateAnswer(
>                                               &offer_answer_options)) {
>     std::string error = "CreateAnswer called with invalid constraints.";
>     RTC_LOG(LS_ERROR) << error;
>-    PostCreateSessionDescriptionFailure(observer, error);
>+    PostCreateSessionDescriptionFailure(
>+        observer, RTCError(RTCErrorType::INVALID_PARAMETER, std::move(error)));
>     return;
>   }
> 
>@@ -1589,19 +1892,31 @@ void PeerConnection::CreateAnswer(CreateSessionDescriptionObserver* observer,
>     return;
>   }
> 
>-  if (IsClosed()) {
>-    std::string error = "CreateAnswer called when PeerConnection is closed.";
>+  if (!(signaling_state_ == kHaveRemoteOffer ||
>+        signaling_state_ == kHaveLocalPrAnswer)) {
>+    std::string error =
>+        "PeerConnection cannot create an answer in a state other than "
>+        "have-remote-offer or have-local-pranswer.";
>     RTC_LOG(LS_ERROR) << error;
>-    PostCreateSessionDescriptionFailure(observer, error);
>+    PostCreateSessionDescriptionFailure(
>+        observer, RTCError(RTCErrorType::INVALID_STATE, std::move(error)));
>     return;
>   }
> 
>-  if (remote_description() &&
>-      remote_description()->GetType() != SdpType::kOffer) {
>-    std::string error = "CreateAnswer called without remote offer.";
>-    RTC_LOG(LS_ERROR) << error;
>-    PostCreateSessionDescriptionFailure(observer, error);
>-    return;
>+  // The remote description should be set if we're in the right state.
>+  RTC_DCHECK(remote_description());
>+
>+  if (IsUnifiedPlan()) {
>+    if (options.offer_to_receive_audio != RTCOfferAnswerOptions::kUndefined) {
>+      RTC_LOG(LS_WARNING) << "CreateAnswer: offer_to_receive_audio is not "
>+                             "supported with Unified Plan semantics. Use the "
>+                             "RtpTransceiver API instead.";
>+    }
>+    if (options.offer_to_receive_video != RTCOfferAnswerOptions::kUndefined) {
>+      RTC_LOG(LS_WARNING) << "CreateAnswer: offer_to_receive_video is not "
>+                             "supported with Unified Plan semantics. Use the "
>+                             "RtpTransceiver API instead.";
>+    }
>   }
> 
>   cricket::MediaSessionOptions session_options;
>@@ -1612,42 +1927,73 @@ void PeerConnection::CreateAnswer(CreateSessionDescriptionObserver* observer,
> 
> void PeerConnection::SetLocalDescription(
>     SetSessionDescriptionObserver* observer,
>-    SessionDescriptionInterface* desc) {
>+    SessionDescriptionInterface* desc_ptr) {
>   TRACE_EVENT0("webrtc", "PeerConnection::SetLocalDescription");
> 
>+  // The SetLocalDescription contract is that we take ownership of the session
>+  // description regardless of the outcome, so wrap it in a unique_ptr right
>+  // away. Ideally, SetLocalDescription's signature will be changed to take the
>+  // description as a unique_ptr argument to formalize this agreement.
>+  std::unique_ptr<SessionDescriptionInterface> desc(desc_ptr);
>+
>   if (!observer) {
>     RTC_LOG(LS_ERROR) << "SetLocalDescription - observer is NULL.";
>     return;
>   }
> 
>   if (!desc) {
>-    PostSetSessionDescriptionFailure(observer, "SessionDescription is NULL.");
>+    PostSetSessionDescriptionFailure(
>+        observer,
>+        RTCError(RTCErrorType::INTERNAL_ERROR, "SessionDescription is NULL."));
>     return;
>   }
> 
>-  SdpType type = desc->GetType();
>+  // If a session error has occurred the PeerConnection is in a possibly
>+  // inconsistent state so fail right away.
>+  if (session_error() != SessionError::kNone) {
>+    std::string error_message = GetSessionErrorMsg();
>+    RTC_LOG(LS_ERROR) << "SetLocalDescription: " << error_message;
>+    PostSetSessionDescriptionFailure(
>+        observer,
>+        RTCError(RTCErrorType::INTERNAL_ERROR, std::move(error_message)));
>+    return;
>+  }
>+
>+  RTCError error = ValidateSessionDescription(desc.get(), cricket::CS_LOCAL);
>+  if (!error.ok()) {
>+    std::string error_message = GetSetDescriptionErrorMessage(
>+        cricket::CS_LOCAL, desc->GetType(), error);
>+    RTC_LOG(LS_ERROR) << error_message;
>+    PostSetSessionDescriptionFailure(
>+        observer,
>+        RTCError(RTCErrorType::INTERNAL_ERROR, std::move(error_message)));
>+    return;
>+  }
> 
>-  RTCError error = ApplyLocalDescription(rtc::WrapUnique(desc));
>+  // Grab the description type before moving ownership to ApplyLocalDescription,
>+  // which may destroy it before returning.
>+  const SdpType type = desc->GetType();
>+
>+  error = ApplyLocalDescription(std::move(desc));
>   // |desc| may be destroyed at this point.
> 
>   if (!error.ok()) {
>-    std::ostringstream oss;
>-    oss << "Failed to set local " << SdpTypeToString(type)
>-        << " sdp: " << error.message();
>-    std::string error_message = oss.str();
>-    RTC_LOG(LS_ERROR) << error_message << " (" << error.type() << ")";
>-    PostSetSessionDescriptionFailure(observer, std::move(error_message));
>+    // If ApplyLocalDescription fails, the PeerConnection could be in an
>+    // inconsistent state, so act conservatively here and set the session error
>+    // so that future calls to SetLocalDescription/SetRemoteDescription fail.
>+    SetSessionError(SessionError::kContent, error.message());
>+    std::string error_message =
>+        GetSetDescriptionErrorMessage(cricket::CS_LOCAL, type, error);
>+    RTC_LOG(LS_ERROR) << error_message;
>+    PostSetSessionDescriptionFailure(
>+        observer,
>+        RTCError(RTCErrorType::INTERNAL_ERROR, std::move(error_message)));
>     return;
>   }
>   RTC_DCHECK(local_description());
> 
>   PostSetSessionDescriptionSuccess(observer);
> 
>-  // According to JSEP, after setLocalDescription, changing the candidate pool
>-  // size is not allowed, and changing the set of ICE servers will not result
>-  // in new candidates being gathered.
>-  port_allocator_->FreezeCandidatePool();
>-
>   // MaybeStartGathering needs to be called after posting
>   // MSG_SET_SESSIONDESCRIPTION_SUCCESS, so that we don't signal any candidates
>   // before signaling that SetLocalDescription completed.
>@@ -1659,7 +2005,10 @@ void PeerConnection::SetLocalDescription(
>     network_thread()->Invoke<void>(
>         RTC_FROM_HERE, rtc::Bind(&cricket::PortAllocator::DiscardCandidatePool,
>                                  port_allocator_.get()));
>+    // Make UMA notes about what was agreed to.
>+    ReportNegotiatedSdpSemantics(*local_description());
>   }
>+  NoteUsageEvent(UsageEvent::SET_LOCAL_DESCRIPTION_CALLED);
> }
> 
> RTCError PeerConnection::ApplyLocalDescription(
>@@ -1667,26 +2016,10 @@ RTCError PeerConnection::ApplyLocalDescription(
>   RTC_DCHECK_RUN_ON(signaling_thread());
>   RTC_DCHECK(desc);
> 
>-  RTCError error = ValidateSessionDescription(desc.get(), cricket::CS_LOCAL);
>-  if (!error.ok()) {
>-    return error;
>-  }
>-
>   // Update stats here so that we have the most recent stats for tracks and
>   // streams that might be removed by updating the session description.
>   stats_->UpdateStats(kStatsOutputLevelStandard);
> 
>-  // Update the initial_offerer flag if this session is the initial_offerer.
>-  SdpType type = desc->GetType();
>-  if (!initial_offerer_.has_value()) {
>-    initial_offerer_.emplace(type == SdpType::kOffer);
>-    if (*initial_offerer_) {
>-      transport_controller_->SetIceRole(cricket::ICEROLE_CONTROLLING);
>-    } else {
>-      transport_controller_->SetIceRole(cricket::ICEROLE_CONTROLLED);
>-    }
>-  }
>-
>   // Take a reference to the old local description since it's used below to
>   // compare against the new local description. When setting the new local
>   // description, grab ownership of the replaced session description in case it
>@@ -1695,6 +2028,7 @@ RTCError PeerConnection::ApplyLocalDescription(
>   const SessionDescriptionInterface* old_local_description =
>       local_description();
>   std::unique_ptr<SessionDescriptionInterface> replaced_local_description;
>+  SdpType type = desc->GetType();
>   if (type == SdpType::kAnswer) {
>     replaced_local_description = pending_local_description_
>                                      ? std::move(pending_local_description_)
>@@ -1710,12 +2044,20 @@ RTCError PeerConnection::ApplyLocalDescription(
>   // |local_description()|.
>   RTC_DCHECK(local_description());
> 
>+  RTCError error = PushdownTransportDescription(cricket::CS_LOCAL, type);
>+  if (!error.ok()) {
>+    return error;
>+  }
>+
>   if (IsUnifiedPlan()) {
>     RTCError error = UpdateTransceiversAndDataChannels(
>-        cricket::CS_LOCAL, old_local_description, *local_description());
>+        cricket::CS_LOCAL, *local_description(), old_local_description,
>+        remote_description());
>     if (!error.ok()) {
>       return error;
>     }
>+    std::vector<rtc::scoped_refptr<RtpTransceiverInterface>> remove_list;
>+    std::vector<rtc::scoped_refptr<MediaStreamInterface>> removed_streams;
>     for (auto transceiver : transceivers_) {
>       const ContentInfo* content =
>           FindMediaSectionForTransceiver(transceiver, local_description());
>@@ -1723,15 +2065,36 @@ RTCError PeerConnection::ApplyLocalDescription(
>         continue;
>       }
>       const MediaContentDescription* media_desc = content->media_description();
>+      // 2.2.7.1.6: If description is of type "answer" or "pranswer", then run
>+      // the following steps:
>       if (type == SdpType::kPrAnswer || type == SdpType::kAnswer) {
>+        // 2.2.7.1.6.1: If direction is "sendonly" or "inactive", and
>+        // transceiver's [[FiredDirection]] slot is either "sendrecv" or
>+        // "recvonly", process the removal of a remote track for the media
>+        // description, given transceiver, removeList, and muteTracks.
>+        if (!RtpTransceiverDirectionHasRecv(media_desc->direction()) &&
>+            (transceiver->internal()->fired_direction() &&
>+             RtpTransceiverDirectionHasRecv(
>+                 *transceiver->internal()->fired_direction()))) {
>+          ProcessRemovalOfRemoteTrack(transceiver, &remove_list,
>+                                      &removed_streams);
>+        }
>+        // 2.2.7.1.6.2: Set transceiver's [[CurrentDirection]] and
>+        // [[FiredDirection]] slots to direction.
>         transceiver->internal()->set_current_direction(media_desc->direction());
>+        transceiver->internal()->set_fired_direction(media_desc->direction());
>       }
>-      if (content->rejected && !transceiver->stopped()) {
>-        transceiver->Stop();
>-      }
>+    }
>+    auto observer = Observer();
>+    for (auto transceiver : remove_list) {
>+      observer->OnRemoveTrack(transceiver->receiver());
>+    }
>+    for (auto stream : removed_streams) {
>+      observer->OnRemoveStream(stream);
>     }
>   } else {
>-    // Transport and Media channels will be created only when offer is set.
>+    // Media channels will be created only when offer is set. These may use new
>+    // transports just created by PushdownTransportDescription.
>     if (type == SdpType::kOffer) {
>       // TODO(bugs.webrtc.org/4676) - Handle CreateChannel failure, as new local
>       // description is applied. Restore back to old description.
>@@ -1740,12 +2103,12 @@ RTCError PeerConnection::ApplyLocalDescription(
>         return error;
>       }
>     }
>-
>     // Remove unused channels if MediaContentDescription is rejected.
>     RemoveUnusedChannels(local_description()->description());
>   }
> 
>-  error = UpdateSessionState(type, cricket::CS_LOCAL);
>+  error = UpdateSessionState(type, cricket::CS_LOCAL,
>+                             local_description()->description());
>   if (!error.ok()) {
>     return error;
>   }
>@@ -1767,7 +2130,30 @@ RTCError PeerConnection::ApplyLocalDescription(
>     AllocateSctpSids(role);
>   }
> 
>-  if (!IsUnifiedPlan()) {
>+  if (IsUnifiedPlan()) {
>+    for (auto transceiver : transceivers_) {
>+      const ContentInfo* content =
>+          FindMediaSectionForTransceiver(transceiver, local_description());
>+      if (!content) {
>+        continue;
>+      }
>+      const auto& streams = content->media_description()->streams();
>+      if (!content->rejected && !streams.empty()) {
>+        transceiver->internal()->sender_internal()->set_stream_ids(
>+            streams[0].stream_ids());
>+        transceiver->internal()->sender_internal()->SetSsrc(
>+            streams[0].first_ssrc());
>+      } else {
>+        // 0 is a special value meaning "this sender has no associated send
>+        // stream". Need to call this so the sender won't attempt to configure
>+        // a no longer existing stream and run into DCHECKs in the lower
>+        // layers.
>+        transceiver->internal()->sender_internal()->SetSsrc(0);
>+      }
>+    }
>+  } else {
>+    // Plan B semantics.
>+
>     // Update state and SSRC of local MediaStreams and DataChannels based on the
>     // local session description.
>     const cricket::ContentInfo* audio_content =
>@@ -1834,31 +2220,64 @@ void PeerConnection::SetRemoteDescription(
>     return;
>   }
> 
>+  // If a session error has occurred the PeerConnection is in a possibly
>+  // inconsistent state so fail right away.
>+  if (session_error() != SessionError::kNone) {
>+    std::string error_message = GetSessionErrorMsg();
>+    RTC_LOG(LS_ERROR) << "SetRemoteDescription: " << error_message;
>+    observer->OnSetRemoteDescriptionComplete(
>+        RTCError(RTCErrorType::INTERNAL_ERROR, std::move(error_message)));
>+    return;
>+  }
>+
>+  if (desc->GetType() == SdpType::kOffer) {
>+    // Report to UMA the format of the received offer.
>+    ReportSdpFormatReceived(*desc);
>+  }
>+
>+  RTCError error = ValidateSessionDescription(desc.get(), cricket::CS_REMOTE);
>+  if (!error.ok()) {
>+    std::string error_message = GetSetDescriptionErrorMessage(
>+        cricket::CS_REMOTE, desc->GetType(), error);
>+    RTC_LOG(LS_ERROR) << error_message;
>+    observer->OnSetRemoteDescriptionComplete(
>+        RTCError(error.type(), std::move(error_message)));
>+    return;
>+  }
>+
>+  // Grab the description type before moving ownership to
>+  // ApplyRemoteDescription, which may destroy it before returning.
>   const SdpType type = desc->GetType();
> 
>-  RTCError error = ApplyRemoteDescription(std::move(desc));
>+  error = ApplyRemoteDescription(std::move(desc));
>   // |desc| may be destroyed at this point.
> 
>   if (!error.ok()) {
>-    std::ostringstream oss;
>-    oss << "Failed to set remote " << SdpTypeToString(type)
>-        << " sdp: " << error.message();
>-    std::string error_message = oss.str();
>-    RTC_LOG(LS_ERROR) << error_message << " (" << error.type() << ")";
>+    // If ApplyRemoteDescription fails, the PeerConnection could be in an
>+    // inconsistent state, so act conservatively here and set the session error
>+    // so that future calls to SetLocalDescription/SetRemoteDescription fail.
>+    SetSessionError(SessionError::kContent, error.message());
>+    std::string error_message =
>+        GetSetDescriptionErrorMessage(cricket::CS_REMOTE, type, error);
>+    RTC_LOG(LS_ERROR) << error_message;
>     observer->OnSetRemoteDescriptionComplete(
>         RTCError(error.type(), std::move(error_message)));
>     return;
>   }
>+  RTC_DCHECK(remote_description());
> 
>-  if (remote_description()->GetType() == SdpType::kAnswer) {
>+  if (type == SdpType::kAnswer) {
>     // TODO(deadbeef): We already had to hop to the network thread for
>     // MaybeStartGathering...
>     network_thread()->Invoke<void>(
>         RTC_FROM_HERE, rtc::Bind(&cricket::PortAllocator::DiscardCandidatePool,
>                                  port_allocator_.get()));
>+    // Make UMA notes about what was agreed to.
>+    ReportNegotiatedSdpSemantics(*remote_description());
>   }
> 
>   observer->OnSetRemoteDescriptionComplete(RTCError::OK());
>+  NoteUsageEvent(UsageEvent::SET_REMOTE_DESCRIPTION_CALLED);
> }
> 
> RTCError PeerConnection::ApplyRemoteDescription(
>@@ -1866,11 +2285,6 @@ RTCError PeerConnection::ApplyRemoteDescription(
>   RTC_DCHECK_RUN_ON(signaling_thread());
>   RTC_DCHECK(desc);
> 
>-  RTCError error = ValidateSessionDescription(desc.get(), cricket::CS_REMOTE);
>-  if (!error.ok()) {
>-    return error;
>-  }
>-
>   // Update stats here so that we have the most recent stats for tracks and
>   // streams that might be removed by updating the session description.
>   stats_->UpdateStats(kStatsOutputLevelStandard);
>@@ -1899,30 +2313,37 @@ RTCError PeerConnection::ApplyRemoteDescription(
>   // |remote_description()|.
>   RTC_DCHECK(remote_description());
> 
>+  RTCError error = PushdownTransportDescription(cricket::CS_REMOTE, type);
>+  if (!error.ok()) {
>+    return error;
>+  }
>   // Transport and Media channels will be created only when offer is set.
>   if (IsUnifiedPlan()) {
>     RTCError error = UpdateTransceiversAndDataChannels(
>-        cricket::CS_REMOTE, old_remote_description, *remote_description());
>+        cricket::CS_REMOTE, *remote_description(), local_description(),
>+        old_remote_description);
>     if (!error.ok()) {
>       return error;
>     }
>   } else {
>+    // Media channels will be created only when offer is set. These may use new
>+    // transports just created by PushdownTransportDescription.
>     if (type == SdpType::kOffer) {
>-      // TODO(bugs.webrtc.org/4676) - Handle CreateChannel failure, as new local
>+      // TODO(mallinath) - Handle CreateChannel failure, as new local
>       // description is applied. Restore back to old description.
>       RTCError error = CreateChannels(*remote_description()->description());
>       if (!error.ok()) {
>         return error;
>       }
>     }
>-
>     // Remove unused channels if MediaContentDescription is rejected.
>     RemoveUnusedChannels(remote_description()->description());
>   }
> 
>-  // NOTE: Candidates allocation will be initiated only when SetLocalDescription
>-  // is called.
>-  error = UpdateSessionState(type, cricket::CS_REMOTE);
>+  // NOTE: Candidates allocation will be initiated only when
>+  // SetLocalDescription is called.
>+  error = UpdateSessionState(type, cricket::CS_REMOTE,
>+                             remote_description()->description());
>   if (!error.ok()) {
>     return error;
>   }
>@@ -1972,6 +2393,7 @@ RTCError PeerConnection::ApplyRemoteDescription(
>   // read to determine the current checking state. The existing SignalConnecting
>   // actually means "gathering candidates", so cannot be be used here.
>   if (remote_description()->GetType() != SdpType::kOffer &&
>+      remote_description()->number_of_mediasections() > 0u &&
>       ice_connection_state() == PeerConnectionInterface::kIceConnectionNew) {
>     SetIceConnectionState(PeerConnectionInterface::kIceConnectionChecking);
>   }
>@@ -1984,7 +2406,11 @@ RTCError PeerConnection::ApplyRemoteDescription(
>   }
> 
>   if (IsUnifiedPlan()) {
>-    std::vector<TrackEvent> track_events;
>+    std::vector<rtc::scoped_refptr<RtpTransceiverInterface>>
>+        now_receiving_transceivers;
>+    std::vector<rtc::scoped_refptr<RtpTransceiverInterface>> remove_list;
>+    std::vector<rtc::scoped_refptr<MediaStreamInterface>> added_streams;
>+    std::vector<rtc::scoped_refptr<MediaStreamInterface>> removed_streams;
>     for (auto transceiver : transceivers_) {
>       const ContentInfo* content =
>           FindMediaSectionForTransceiver(transceiver, remote_description());
>@@ -1998,41 +2424,89 @@ RTCError PeerConnection::ApplyRemoteDescription(
>       // the RTCSessionDescription: If direction is sendrecv or recvonly, and
>       // transceiver's current direction is neither sendrecv nor recvonly,
>       // process the addition of a remote track for the media description.
>+      std::vector<std::string> stream_ids;
>+      if (!media_desc->streams().empty()) {
>+        // The remote description has signaled the stream IDs.
>+        stream_ids = media_desc->streams()[0].stream_ids();
>+      }
>       if (RtpTransceiverDirectionHasRecv(local_direction) &&
>-          (!transceiver->current_direction() ||
>-           !RtpTransceiverDirectionHasRecv(
>-               *transceiver->current_direction()))) {
>-        const std::string& sync_label = media_desc->streams()[0].sync_label;
>-        rtc::scoped_refptr<MediaStreamInterface> stream =
>-            remote_streams_->find(sync_label);
>-        if (!stream) {
>-          stream = MediaStreamProxy::Create(rtc::Thread::Current(),
>-                                            MediaStream::Create(sync_label));
>-          remote_streams_->AddStream(stream);
>+          (!transceiver->fired_direction() ||
>+           !RtpTransceiverDirectionHasRecv(*transceiver->fired_direction()))) {
>+        RTC_LOG(LS_INFO) << "Processing the addition of a new track for MID="
>+                         << content->name << " (added to "
>+                         << GetStreamIdsString(stream_ids) << ").";
>+
>+        std::vector<rtc::scoped_refptr<MediaStreamInterface>> media_streams;
>+        for (const std::string& stream_id : stream_ids) {
>+          rtc::scoped_refptr<MediaStreamInterface> stream =
>+              remote_streams_->find(stream_id);
>+          if (!stream) {
>+            stream = MediaStreamProxy::Create(rtc::Thread::Current(),
>+                                              MediaStream::Create(stream_id));
>+            remote_streams_->AddStream(stream);
>+            added_streams.push_back(stream);
>+          }
>+          media_streams.push_back(stream);
>         }
>-        transceiver->internal()->receiver_internal()->SetStreams({stream});
>-        TrackEvent track_event;
>-        track_event.receiver = transceiver->receiver();
>-        track_event.streams = transceiver->receiver()->streams();
>-        track_events.push_back(std::move(track_event));
>+        // This will add the remote track to the streams.
>+        // TODO(hbos): When we remove remote_streams(), use set_stream_ids()
>+        // instead. https://crbug.com/webrtc/9480
>+        transceiver->internal()->receiver_internal()->SetStreams(media_streams);
>+        now_receiving_transceivers.push_back(transceiver);
>       }
>-      // If direction is sendonly or inactive, and transceiver's current
>-      // direction is neither sendonly nor inactive, process the removal of a
>-      // remote track for the media description.
>+      // 2.2.8.1.7: If direction is "sendonly" or "inactive", and transceiver's
>+      // [[FiredDirection]] slot is either "sendrecv" or "recvonly", process the
>+      // removal of a remote track for the media description, given transceiver,
>+      // removeList, and muteTracks.
>       if (!RtpTransceiverDirectionHasRecv(local_direction) &&
>-          (transceiver->current_direction() &&
>-           RtpTransceiverDirectionHasRecv(*transceiver->current_direction()))) {
>-        transceiver->internal()->receiver_internal()->SetStreams({});
>+          (transceiver->fired_direction() &&
>+           RtpTransceiverDirectionHasRecv(*transceiver->fired_direction()))) {
>+        ProcessRemovalOfRemoteTrack(transceiver, &remove_list,
>+                                    &removed_streams);
>       }
>+      // 2.2.8.1.8: Set transceiver's [[FiredDirection]] slot to direction.
>+      transceiver->internal()->set_fired_direction(local_direction);
>+      // 2.2.8.1.9: If description is of type "answer" or "pranswer", then run
>+      // the following steps:
>       if (type == SdpType::kPrAnswer || type == SdpType::kAnswer) {
>+        // 2.2.8.1.9.1: Set transceiver's [[CurrentDirection]] slot to
>+        // direction.
>         transceiver->internal()->set_current_direction(local_direction);
>       }
>+      // 2.2.8.1.10: If the media description is rejected, and transceiver is
>+      // not already stopped, stop the RTCRtpTransceiver transceiver.
>       if (content->rejected && !transceiver->stopped()) {
>+        RTC_LOG(LS_INFO) << "Stopping transceiver for MID=" << content->name
>+                         << " since the media section was rejected.";
>         transceiver->Stop();
>       }
>+      if (!content->rejected &&
>+          RtpTransceiverDirectionHasRecv(local_direction)) {
>+        // Set ssrc to 0 in the case of an unsignalled ssrc.
>+        uint32_t ssrc = 0;
>+        if (!media_desc->streams().empty() &&
>+            media_desc->streams()[0].has_ssrcs()) {
>+          ssrc = media_desc->streams()[0].first_ssrc();
>+        }
>+        transceiver->internal()->receiver_internal()->SetupMediaChannel(ssrc);
>+      }
>+    }
>+    // Once all processing has finished, fire off callbacks.
>+    auto observer = Observer();
>+    for (auto transceiver : now_receiving_transceivers) {
>+      stats_->AddTrack(transceiver->receiver()->track());
>+      observer->OnTrack(transceiver);
>+      observer->OnAddTrack(transceiver->receiver(),
>+                           transceiver->receiver()->streams());
>     }
>-    for (auto event : track_events) {
>-      observer_->OnAddTrack(event.receiver, event.streams);
>+    for (auto stream : added_streams) {
>+      observer->OnAddStream(stream);
>+    }
>+    for (auto transceiver : remove_list) {
>+      observer->OnRemoveTrack(transceiver->receiver());
>+    }
>+    for (auto stream : removed_streams) {
>+      observer->OnRemoveStream(stream);
>     }
>   }
> 
>@@ -2107,10 +2581,11 @@ RTCError PeerConnection::ApplyRemoteDescription(
>     }
> 
>     // Iterate new_streams and notify the observer about new MediaStreams.
>+    auto observer = Observer();
>     for (size_t i = 0; i < new_streams->count(); ++i) {
>       MediaStreamInterface* new_stream = new_streams->at(i);
>       stats_->AddStream(new_stream);
>-      observer_->OnAddStream(
>+      observer->OnAddStream(
>           rtc::scoped_refptr<MediaStreamInterface>(new_stream));
>     }
> 
>@@ -2120,39 +2595,74 @@ RTCError PeerConnection::ApplyRemoteDescription(
>   return RTCError::OK();
> }
> 
>+void PeerConnection::ProcessRemovalOfRemoteTrack(
>+    rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
>+        transceiver,
>+    std::vector<rtc::scoped_refptr<RtpTransceiverInterface>>* remove_list,
>+    std::vector<rtc::scoped_refptr<MediaStreamInterface>>* removed_streams) {
>+  RTC_DCHECK(transceiver->mid());
>+  RTC_LOG(LS_INFO) << "Processing the removal of a track for MID="
>+                   << *transceiver->mid();
>+  std::vector<rtc::scoped_refptr<MediaStreamInterface>> media_streams =
>+      transceiver->internal()->receiver_internal()->streams();
>+  // This will remove the remote track from the streams.
>+  transceiver->internal()->receiver_internal()->set_stream_ids({});
>+  remove_list->push_back(transceiver);
>+  // Remove any streams that no longer have tracks.
>+  // TODO(https://crbug.com/webrtc/9480): When we use stream IDs instead
>+  // of streams, see if the stream was removed by checking if this was the
>+  // last receiver with that stream ID.
>+  for (auto stream : media_streams) {
>+    if (stream->GetAudioTracks().empty() && stream->GetVideoTracks().empty()) {
>+      remote_streams_->RemoveStream(stream);
>+      removed_streams->push_back(stream);
>+    }
>+  }
>+}
>+
> RTCError PeerConnection::UpdateTransceiversAndDataChannels(
>     cricket::ContentSource source,
>-    const SessionDescriptionInterface* old_session,
>-    const SessionDescriptionInterface& new_session) {
>+    const SessionDescriptionInterface& new_session,
>+    const SessionDescriptionInterface* old_local_description,
>+    const SessionDescriptionInterface* old_remote_description) {
>   RTC_DCHECK(IsUnifiedPlan());
> 
>-  auto bundle_group_or_error = GetEarlyBundleGroup(*new_session.description());
>-  if (!bundle_group_or_error.ok()) {
>-    return bundle_group_or_error.MoveError();
>+  const cricket::ContentGroup* bundle_group = nullptr;
>+  if (new_session.GetType() == SdpType::kOffer) {
>+    auto bundle_group_or_error =
>+        GetEarlyBundleGroup(*new_session.description());
>+    if (!bundle_group_or_error.ok()) {
>+      return bundle_group_or_error.MoveError();
>+    }
>+    bundle_group = bundle_group_or_error.MoveValue();
>   }
>-  const cricket::ContentGroup* bundle_group = bundle_group_or_error.MoveValue();
> 
>-  const ContentInfos& old_contents =
>-      (old_session ? old_session->description()->contents() : ContentInfos());
>   const ContentInfos& new_contents = new_session.description()->contents();
>-
>   for (size_t i = 0; i < new_contents.size(); ++i) {
>     const cricket::ContentInfo& new_content = new_contents[i];
>-    const cricket::ContentInfo* old_content =
>-        (i < old_contents.size() ? &old_contents[i] : nullptr);
>     cricket::MediaType media_type = new_content.media_description()->type();
>     seen_mids_.insert(new_content.name);
>     if (media_type == cricket::MEDIA_TYPE_AUDIO ||
>         media_type == cricket::MEDIA_TYPE_VIDEO) {
>+      const cricket::ContentInfo* old_local_content = nullptr;
>+      if (old_local_description &&
>+          i < old_local_description->description()->contents().size()) {
>+        old_local_content =
>+            &old_local_description->description()->contents()[i];
>+      }
>+      const cricket::ContentInfo* old_remote_content = nullptr;
>+      if (old_remote_description &&
>+          i < old_remote_description->description()->contents().size()) {
>+        old_remote_content =
>+            &old_remote_description->description()->contents()[i];
>+      }
>       auto transceiver_or_error =
>-          AssociateTransceiver(source, i, new_content, old_content);
>+          AssociateTransceiver(source, new_session.GetType(), i, new_content,
>+                               old_local_content, old_remote_content);
>       if (!transceiver_or_error.ok()) {
>         return transceiver_or_error.MoveError();
>       }
>       auto transceiver = transceiver_or_error.MoveValue();
>-      if (source == cricket::CS_LOCAL && transceiver->stopped()) {
>-        continue;
>-      }
>       RTCError error =
>           UpdateTransceiverChannel(transceiver, new_content, bundle_group);
>       if (!error.ok()) {
>@@ -2161,6 +2671,8 @@ RTCError PeerConnection::UpdateTransceiversAndDataChannels(
>     } else if (media_type == cricket::MEDIA_TYPE_DATA) {
>       if (GetDataMid() && new_content.name != *GetDataMid()) {
>         // Ignore all but the first data section.
>+        RTC_LOG(LS_INFO) << "Ignoring data media section with MID="
>+                         << new_content.name;
>         continue;
>       }
>       RTCError error = UpdateDataChannel(source, new_content, bundle_group);
>@@ -2191,16 +2703,11 @@ RTCError PeerConnection::UpdateTransceiverChannel(
>     }
>   } else {
>     if (!channel) {
>-      if (transceiver->internal()->media_type() == cricket::MEDIA_TYPE_AUDIO) {
>-        channel = CreateVoiceChannel(
>-            content.name,
>-            GetTransportNameForMediaSection(content.name, bundle_group));
>+      if (transceiver->media_type() == cricket::MEDIA_TYPE_AUDIO) {
>+        channel = CreateVoiceChannel(content.name);
>       } else {
>-        RTC_DCHECK_EQ(cricket::MEDIA_TYPE_VIDEO,
>-                      transceiver->internal()->media_type());
>-        channel = CreateVideoChannel(
>-            content.name,
>-            GetTransportNameForMediaSection(content.name, bundle_group));
>+        RTC_DCHECK_EQ(cricket::MEDIA_TYPE_VIDEO, transceiver->media_type());
>+        channel = CreateVideoChannel(content.name);
>       }
>       if (!channel) {
>         LOG_AND_RETURN_ERROR(
>@@ -2218,15 +2725,15 @@ RTCError PeerConnection::UpdateDataChannel(
>     const cricket::ContentInfo& content,
>     const cricket::ContentGroup* bundle_group) {
>   if (data_channel_type_ == cricket::DCT_NONE) {
>-    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
>-                         "Data channels are not supported.");
>+    // If data channels are disabled, ignore this media section. CreateAnswer
>+    // will take care of rejecting it.
>+    return RTCError::OK();
>   }
>   if (content.rejected) {
>     DestroyDataChannel();
>   } else {
>     if (!rtp_data_channel_ && !sctp_transport_) {
>-      if (!CreateDataChannel(content.name, GetTransportNameForMediaSection(
>-                                               content.name, bundle_group))) {
>+      if (!CreateDataChannel(content.name)) {
>         LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
>                              "Failed to create data channel.");
>       }
>@@ -2243,19 +2750,30 @@ RTCError PeerConnection::UpdateDataChannel(
> 
> RTCErrorOr<rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>>
> PeerConnection::AssociateTransceiver(cricket::ContentSource source,
>+                                     SdpType type,
>                                      size_t mline_index,
>                                      const ContentInfo& content,
>-                                     const ContentInfo* old_content) {
>+                                     const ContentInfo* old_local_content,
>+                                     const ContentInfo* old_remote_content) {
>   RTC_DCHECK(IsUnifiedPlan());
>-  // If the m= section is being recycled (rejected in previous remote
>-  // description, not rejected in current description), dissociate the currently
>-  // associated RtpTransceiver by setting its mid property to null, and discard
>-  // the mapping between the transceiver and its m= section index.
>-  if (old_content && old_content->rejected && !content.rejected) {
>-    auto old_transceiver = GetAssociatedTransceiver(old_content->name);
>+  // If this is an offer then the m= section might be recycled. If the m=
>+  // section is being recycled (defined as: rejected in the current local or
>+  // remote description and not rejected in new description), dissociate the
>+  // currently associated RtpTransceiver by setting its mid property to null,
>+  // and discard the mapping between the transceiver and its m= section index.
>+  if (IsMediaSectionBeingRecycled(type, content, old_local_content,
>+                                  old_remote_content)) {
>+    // We want to dissociate the transceiver that has the rejected mid.
>+    const std::string& old_mid =
>+        (old_local_content && old_local_content->rejected)
>+            ? old_local_content->name
>+            : old_remote_content->name;
>+    auto old_transceiver = GetAssociatedTransceiver(old_mid);
>     if (old_transceiver) {
>-      old_transceiver->internal()->set_mid(rtc::nullopt);
>-      old_transceiver->internal()->set_mline_index(rtc::nullopt);
>+      RTC_LOG(LS_INFO) << "Dissociating transceiver for MID=" << old_mid
>+                       << " since the media section is being recycled.";
>+      old_transceiver->internal()->set_mid(absl::nullopt);
>+      old_transceiver->internal()->set_mline_index(absl::nullopt);
>     }
>   }
>   const MediaContentDescription* media_desc = content.media_description();
>@@ -2282,17 +2800,27 @@ PeerConnection::AssociateTransceiver(cricket::ContentSource source,
>     // If no RtpTransceiver was found in the previous step, create one with a
>     // recvonly direction.
>     if (!transceiver) {
>-      auto sender =
>-          CreateSender(media_desc->type(), nullptr, {rtc::CreateRandomUuid()});
>-      auto receiver =
>-          CreateReceiver(media_desc->type(), media_desc->streams()[0].id);
>+      RTC_LOG(LS_INFO) << "Adding "
>+                       << cricket::MediaTypeToString(media_desc->type())
>+                       << " transceiver for MID=" << content.name
>+                       << " at i=" << mline_index
>+                       << " in response to the remote description.";
>+      std::string sender_id = rtc::CreateRandomUuid();
>+      auto sender = CreateSender(media_desc->type(), sender_id, nullptr, {});
>+      std::string receiver_id;
>+      if (!media_desc->streams().empty()) {
>+        receiver_id = media_desc->streams()[0].id;
>+      } else {
>+        receiver_id = rtc::CreateRandomUuid();
>+      }
>+      auto receiver = CreateReceiver(media_desc->type(), receiver_id);
>       transceiver = CreateAndAddTransceiver(sender, receiver);
>       transceiver->internal()->set_direction(
>           RtpTransceiverDirection::kRecvOnly);
>     }
>   }
>   RTC_DCHECK(transceiver);
>-  if (transceiver->internal()->media_type() != media_desc->type()) {
>+  if (transceiver->media_type() != media_desc->type()) {
>     LOG_AND_RETURN_ERROR(
>         RTCErrorType::INVALID_PARAMETER,
>         "Transceiver type does not match media description type.");
>@@ -2338,7 +2866,7 @@ PeerConnection::FindAvailableTransceiverToReceive(
>   // associated with any m= section and are not stopped, find the first such
>   // RtpTransceiver.
>   for (auto transceiver : transceivers_) {
>-    if (transceiver->internal()->media_type() == media_type &&
>+    if (transceiver->media_type() == media_type &&
>         transceiver->internal()->created_by_addtrack() && !transceiver->mid() &&
>         !transceiver->stopped()) {
>       return transceiver;
>@@ -2364,7 +2892,7 @@ const cricket::ContentInfo* PeerConnection::FindMediaSectionForTransceiver(
>     // Plan B only allows at most one audio and one video section, so use the
>     // first media section of that type.
>     return cricket::GetFirstMediaContent(sdesc->description()->contents(),
>-                                         transceiver->internal()->media_type());
>+                                         transceiver->media_type());
>   }
> }
> 
>@@ -2375,7 +2903,14 @@ PeerConnectionInterface::RTCConfiguration PeerConnection::GetConfiguration() {
> bool PeerConnection::SetConfiguration(const RTCConfiguration& configuration,
>                                       RTCError* error) {
>   TRACE_EVENT0("webrtc", "PeerConnection::SetConfiguration");
>+  if (IsClosed()) {
>+    RTC_LOG(LS_ERROR) << "SetConfiguration: PeerConnection is closed.";
>+    return SafeSetError(RTCErrorType::INVALID_STATE, error);
>+  }
> 
>+  // According to JSEP, after setLocalDescription, changing the candidate pool
>+  // size is not allowed, and changing the set of ICE servers will not result
>+  // in new candidates being gathered.
>   if (local_description() && configuration.ice_candidate_pool_size !=
>                                  configuration_.ice_candidate_pool_size) {
>     RTC_LOG(LS_ERROR) << "Can't change candidate pool size after calling "
>@@ -2394,7 +2929,19 @@ bool PeerConnection::SetConfiguration(const RTCConfiguration& configuration,
>       configuration.ice_candidate_pool_size;
>   modified_config.prune_turn_ports = configuration.prune_turn_ports;
>   modified_config.ice_check_min_interval = configuration.ice_check_min_interval;
>+  modified_config.ice_check_interval_strong_connectivity =
>+      configuration.ice_check_interval_strong_connectivity;
>+  modified_config.ice_check_interval_weak_connectivity =
>+      configuration.ice_check_interval_weak_connectivity;
>+  modified_config.ice_unwritable_timeout = configuration.ice_unwritable_timeout;
>+  modified_config.ice_unwritable_min_checks =
>+      configuration.ice_unwritable_min_checks;
>+  modified_config.stun_candidate_keepalive_interval =
>+      configuration.stun_candidate_keepalive_interval;
>   modified_config.turn_customizer = configuration.turn_customizer;
>+  modified_config.network_preference = configuration.network_preference;
>+  modified_config.active_reset_srtp_params =
>+      configuration.active_reset_srtp_params;
>   if (configuration != modified_config) {
>     RTC_LOG(LS_ERROR) << "Modifying the configuration in an unsupported way.";
>     return SafeSetError(RTCErrorType::INVALID_MODIFICATION, error);
>@@ -2409,7 +2956,7 @@ bool PeerConnection::SetConfiguration(const RTCConfiguration& configuration,
>   // Note that this isn't possible through chromium, since it's an unsigned
>   // short in WebIDL.
>   if (configuration.ice_candidate_pool_size < 0 ||
>-      configuration.ice_candidate_pool_size > UINT16_MAX) {
>+      configuration.ice_candidate_pool_size > static_cast<int>(UINT16_MAX)) {
>     return SafeSetError(RTCErrorType::INVALID_RANGE, error);
>   }
> 
>@@ -2421,6 +2968,13 @@ bool PeerConnection::SetConfiguration(const RTCConfiguration& configuration,
>   if (parse_error != RTCErrorType::NONE) {
>     return SafeSetError(parse_error, error);
>   }
>+  // Note if STUN or TURN servers were supplied.
>+  if (!stun_servers.empty()) {
>+    NoteUsageEvent(UsageEvent::STUN_SERVER_ADDED);
>+  }
>+  if (!turn_servers.empty()) {
>+    NoteUsageEvent(UsageEvent::TURN_SERVER_ADDED);
>+  }
> 
>   // In theory this shouldn't fail.
>   if (!network_thread()->Invoke<bool>(
>@@ -2429,7 +2983,8 @@ bool PeerConnection::SetConfiguration(const RTCConfiguration& configuration,
>                     stun_servers, turn_servers, modified_config.type,
>                     modified_config.ice_candidate_pool_size,
>                     modified_config.prune_turn_ports,
>-                    modified_config.turn_customizer))) {
>+                    modified_config.turn_customizer,
>+                    modified_config.stun_candidate_keepalive_interval))) {
>     RTC_LOG(LS_ERROR) << "Failed to apply configuration to PortAllocator.";
>     return SafeSetError(RTCErrorType::INTERNAL_ERROR, error);
>   }
>@@ -2443,9 +2998,12 @@ bool PeerConnection::SetConfiguration(const RTCConfiguration& configuration,
>     transport_controller_->SetNeedsIceRestartFlag();
>   }
> 
>-  if (modified_config.ice_check_min_interval !=
>-      configuration_.ice_check_min_interval) {
>-    transport_controller_->SetIceConfig(ParseIceConfig(modified_config));
>+  transport_controller_->SetIceConfig(ParseIceConfig(modified_config));
>+
>+  if (configuration_.active_reset_srtp_params !=
>+      modified_config.active_reset_srtp_params) {
>+    transport_controller_->SetActiveResetSrtpParams(
>+        modified_config.active_reset_srtp_params);
>   }
> 
>   configuration_ = modified_config;
>@@ -2456,36 +3014,50 @@ bool PeerConnection::AddIceCandidate(
>     const IceCandidateInterface* ice_candidate) {
>   TRACE_EVENT0("webrtc", "PeerConnection::AddIceCandidate");
>   if (IsClosed()) {
>+    RTC_LOG(LS_ERROR) << "AddIceCandidate: PeerConnection is closed.";
>+    NoteAddIceCandidateResult(kAddIceCandidateFailClosed);
>     return false;
>   }
> 
>   if (!remote_description()) {
>-    RTC_LOG(LS_ERROR) << "ProcessIceMessage: ICE candidates can't be added "
>-                      << "without any remote session description.";
>+    RTC_LOG(LS_ERROR) << "AddIceCandidate: ICE candidates can't be added "
>+                         "without any remote session description.";
>+    NoteAddIceCandidateResult(kAddIceCandidateFailNoRemoteDescription);
>     return false;
>   }
> 
>   if (!ice_candidate) {
>-    RTC_LOG(LS_ERROR) << "ProcessIceMessage: Candidate is NULL.";
>+    RTC_LOG(LS_ERROR) << "AddIceCandidate: Candidate is null.";
>+    NoteAddIceCandidateResult(kAddIceCandidateFailNullCandidate);
>     return false;
>   }
> 
>   bool valid = false;
>   bool ready = ReadyToUseRemoteCandidate(ice_candidate, nullptr, &valid);
>   if (!valid) {
>+    NoteAddIceCandidateResult(kAddIceCandidateFailNotValid);
>     return false;
>   }
> 
>   // Add this candidate to the remote session description.
>   if (!mutable_remote_description()->AddCandidate(ice_candidate)) {
>-    RTC_LOG(LS_ERROR) << "ProcessIceMessage: Candidate cannot be used.";
>+    RTC_LOG(LS_ERROR) << "AddIceCandidate: Candidate cannot be used.";
>+    NoteAddIceCandidateResult(kAddIceCandidateFailInAddition);
>     return false;
>   }
> 
>   if (ready) {
>-    return UseCandidate(ice_candidate);
>+    bool result = UseCandidate(ice_candidate);
>+    if (result) {
>+      NoteUsageEvent(UsageEvent::REMOTE_CANDIDATE_ADDED);
>+      NoteAddIceCandidateResult(kAddIceCandidateSuccess);
>+    } else {
>+      NoteAddIceCandidateResult(kAddIceCandidateFailNotUsable);
>+    }
>+    return result;
>   } else {
>-    RTC_LOG(LS_INFO) << "ProcessIceMessage: Not ready to use candidate.";
>+    RTC_LOG(LS_INFO) << "AddIceCandidate: Not ready to use candidate.";
>+    NoteAddIceCandidateResult(kAddIceCandidateFailNotReady);
>     return true;
>   }
> }
>@@ -2493,14 +3065,19 @@ bool PeerConnection::AddIceCandidate(
> bool PeerConnection::RemoveIceCandidates(
>     const std::vector<cricket::Candidate>& candidates) {
>   TRACE_EVENT0("webrtc", "PeerConnection::RemoveIceCandidates");
>+  if (IsClosed()) {
>+    RTC_LOG(LS_ERROR) << "RemoveIceCandidates: PeerConnection is closed.";
>+    return false;
>+  }
>+
>   if (!remote_description()) {
>-    RTC_LOG(LS_ERROR) << "RemoveRemoteIceCandidates: ICE candidates can't be "
>-                      << "removed without any remote session description.";
>+    RTC_LOG(LS_ERROR) << "RemoveIceCandidates: ICE candidates can't be removed "
>+                         "without any remote session description.";
>     return false;
>   }
> 
>   if (candidates.empty()) {
>-    RTC_LOG(LS_ERROR) << "RemoveRemoteIceCandidates: candidates are empty.";
>+    RTC_LOG(LS_ERROR) << "RemoveIceCandidates: candidates are empty.";
>     return false;
>   }
> 
>@@ -2508,70 +3085,47 @@ bool PeerConnection::RemoveIceCandidates(
>       mutable_remote_description()->RemoveCandidates(candidates);
>   if (number_removed != candidates.size()) {
>     RTC_LOG(LS_ERROR)
>-        << "RemoveRemoteIceCandidates: Failed to remove candidates. "
>-        << "Requested " << candidates.size() << " but only " << number_removed
>+        << "RemoveIceCandidates: Failed to remove candidates. Requested "
>+        << candidates.size() << " but only " << number_removed
>         << " are removed.";
>   }
> 
>   // Remove the candidates from the transport controller.
>-  std::string error;
>-  bool res = transport_controller_->RemoveRemoteCandidates(candidates, &error);
>-  if (!res && !error.empty()) {
>-    RTC_LOG(LS_ERROR) << "Error when removing remote candidates: " << error;
>+  RTCError error = transport_controller_->RemoveRemoteCandidates(candidates);
>+  if (!error.ok()) {
>+    RTC_LOG(LS_ERROR)
>+        << "RemoveIceCandidates: Error when removing remote candidates: "
>+        << error.message();
>   }
>   return true;
> }
> 
>-void PeerConnection::RegisterUMAObserver(UMAObserver* observer) {
>-  TRACE_EVENT0("webrtc", "PeerConnection::RegisterUmaObserver");
>-  uma_observer_ = observer;
>-
>-  if (transport_controller()) {
>-    transport_controller()->SetMetricsObserver(uma_observer_);
>-  }
>-
>-  // Send information about IPv4/IPv6 status.
>-  if (uma_observer_) {
>-    port_allocator_->SetMetricsObserver(uma_observer_);
>-    if (port_allocator_->flags() & cricket::PORTALLOCATOR_ENABLE_IPV6) {
>-      uma_observer_->IncrementEnumCounter(
>-          kEnumCounterAddressFamily, kPeerConnection_IPv6,
>-          kPeerConnectionAddressFamilyCounter_Max);
>-    } else {
>-      uma_observer_->IncrementEnumCounter(
>-          kEnumCounterAddressFamily, kPeerConnection_IPv4,
>-          kPeerConnectionAddressFamilyCounter_Max);
>-    }
>-  }
>-}
>-
>-RTCError PeerConnection::SetBitrate(const BitrateParameters& bitrate) {
>+RTCError PeerConnection::SetBitrate(const BitrateSettings& bitrate) {
>   if (!worker_thread()->IsCurrent()) {
>     return worker_thread()->Invoke<RTCError>(
>-        RTC_FROM_HERE, rtc::Bind(&PeerConnection::SetBitrate, this, bitrate));
>+        RTC_FROM_HERE, [&]() { return SetBitrate(bitrate); });
>   }
> 
>-  const bool has_min = static_cast<bool>(bitrate.min_bitrate_bps);
>-  const bool has_current = static_cast<bool>(bitrate.current_bitrate_bps);
>-  const bool has_max = static_cast<bool>(bitrate.max_bitrate_bps);
>+  const bool has_min = bitrate.min_bitrate_bps.has_value();
>+  const bool has_start = bitrate.start_bitrate_bps.has_value();
>+  const bool has_max = bitrate.max_bitrate_bps.has_value();
>   if (has_min && *bitrate.min_bitrate_bps < 0) {
>     LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
>                          "min_bitrate_bps <= 0");
>   }
>-  if (has_current) {
>-    if (has_min && *bitrate.current_bitrate_bps < *bitrate.min_bitrate_bps) {
>+  if (has_start) {
>+    if (has_min && *bitrate.start_bitrate_bps < *bitrate.min_bitrate_bps) {
>       LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
>-                           "current_bitrate_bps < min_bitrate_bps");
>-    } else if (*bitrate.current_bitrate_bps < 0) {
>+                           "start_bitrate_bps < min_bitrate_bps");
>+    } else if (*bitrate.start_bitrate_bps < 0) {
>       LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
>                            "curent_bitrate_bps < 0");
>     }
>   }
>   if (has_max) {
>-    if (has_current &&
>-        *bitrate.max_bitrate_bps < *bitrate.current_bitrate_bps) {
>+    if (has_start && *bitrate.max_bitrate_bps < *bitrate.start_bitrate_bps) {
>       LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
>-                           "max_bitrate_bps < current_bitrate_bps");
>+                           "max_bitrate_bps < start_bitrate_bps");
>     } else if (has_min && *bitrate.max_bitrate_bps < *bitrate.min_bitrate_bps) {
>       LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
>                            "max_bitrate_bps < min_bitrate_bps");
>@@ -2581,13 +3135,8 @@ RTCError PeerConnection::SetBitrate(const BitrateParameters& bitrate) {
>     }
>   }
> 
>-  Call::Config::BitrateConfigMask mask;
>-  mask.min_bitrate_bps = bitrate.min_bitrate_bps;
>-  mask.start_bitrate_bps = bitrate.current_bitrate_bps;
>-  mask.max_bitrate_bps = bitrate.max_bitrate_bps;
>-
>   RTC_DCHECK(call_.get());
>-  call_->SetBitrateConfigMask(mask);
>+  call_->GetTransportControllerSend()->SetClientBitratePreferences(bitrate);
> 
>   return RTCError::OK();
> }
>@@ -2601,7 +3150,7 @@ void PeerConnection::SetBitrateAllocationStrategy(
>         bitrate_allocation_strategy.release();
>     auto functor = [this, strategy_raw]() {
>       call_->SetBitrateAllocationStrategy(
>-          rtc::WrapUnique<rtc::BitrateAllocationStrategy>(strategy_raw));
>+          absl::WrapUnique<rtc::BitrateAllocationStrategy>(strategy_raw));
>     };
>     worker_thread->Invoke<void>(RTC_FROM_HERE, functor);
>     return;
>@@ -2636,10 +3185,31 @@ void PeerConnection::SetAudioRecording(bool recording) {
> 
> std::unique_ptr<rtc::SSLCertificate>
> PeerConnection::GetRemoteAudioSSLCertificate() {
>-  if (!voice_channel()) {
>+  std::unique_ptr<rtc::SSLCertChain> chain = GetRemoteAudioSSLCertChain();
>+  if (!chain || !chain->GetSize()) {
>+    return nullptr;
>+  }
>+  return chain->Get(0).GetUniqueReference();
>+}
>+
>+std::unique_ptr<rtc::SSLCertChain>
>+PeerConnection::GetRemoteAudioSSLCertChain() {
>+  auto audio_transceiver = GetFirstAudioTransceiver();
>+  if (!audio_transceiver || !audio_transceiver->internal()->channel()) {
>     return nullptr;
>   }
>-  return GetRemoteSSLCertificate(voice_channel()->transport_name());
>+  return transport_controller_->GetRemoteSSLCertChain(
>+      audio_transceiver->internal()->channel()->transport_name());
>+}
>+
>+rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
>+PeerConnection::GetFirstAudioTransceiver() const {
>+  for (auto transceiver : transceivers_) {
>+    if (transceiver->media_type() == cricket::MEDIA_TYPE_AUDIO) {
>+      return transceiver;
>+    }
>+  }
>+  return nullptr;
> }
> 
> bool PeerConnection::StartRtcEventLog(rtc::PlatformFile file,
>@@ -2649,7 +3219,7 @@ bool PeerConnection::StartRtcEventLog(rtc::PlatformFile file,
>                               ? RtcEventLog::kUnlimitedOutput
>                               : rtc::saturated_cast<size_t>(max_size_bytes);
>   return StartRtcEventLog(
>-      rtc::MakeUnique<RtcEventLogOutputFile>(file, max_size),
>+      absl::make_unique<RtcEventLogOutputFile>(file, max_size),
>       webrtc::RtcEventLog::kImmediateOutput);
> }
> 
>@@ -2710,22 +3280,43 @@ void PeerConnection::Close() {
>   stats_->UpdateStats(kStatsOutputLevelStandard);
> 
>   ChangeSignalingState(PeerConnectionInterface::kClosed);
>+  NoteUsageEvent(UsageEvent::CLOSE_CALLED);
> 
>   for (auto transceiver : transceivers_) {
>     transceiver->Stop();
>   }
>+
>+  // Ensure that all asynchronous stats requests are completed before destroying
>+  // the transport controller below.
>+  if (stats_collector_) {
>+    stats_collector_->WaitForPendingRequest();
>+  }
>+
>+  // Don't destroy BaseChannels until after stats has been cleaned up so that
>+  // the last stats request can still read from the channels.
>   DestroyAllChannels();
> 
>+  // The event log is used in the transport controller, which must be outlived
>+  // by the former. CreateOffer by the peer connection is implemented
>+  // asynchronously and if the peer connection is closed without resetting the
>+  // WebRTC session description factory, the session description factory would
>+  // call the transport controller.
>+  webrtc_session_desc_factory_.reset();
>+  transport_controller_.reset();
>+
>   network_thread()->Invoke<void>(
>-      RTC_FROM_HERE,
>-      rtc::Bind(&cricket::PortAllocator::DiscardCandidatePool,
>-                port_allocator_.get()));
>+      RTC_FROM_HERE, rtc::Bind(&cricket::PortAllocator::DiscardCandidatePool,
>+                               port_allocator_.get()));
> 
>   worker_thread()->Invoke<void>(RTC_FROM_HERE, [this] {
>     call_.reset();
>     // The event log must outlive call (and any other object that uses it).
>     event_log_.reset();
>   });
>+  ReportUsagePattern();
>+  // The .h file says that observer can be discarded after close() returns.
>+  // Make sure this is true.
>+  observer_ = nullptr;
> }
> 
> void PeerConnection::OnMessage(rtc::Message* msg) {
>@@ -2740,14 +3331,14 @@ void PeerConnection::OnMessage(rtc::Message* msg) {
>     case MSG_SET_SESSIONDESCRIPTION_FAILED: {
>       SetSessionDescriptionMsg* param =
>           static_cast<SetSessionDescriptionMsg*>(msg->pdata);
>-      param->observer->OnFailure(param->error);
>+      param->observer->OnFailure(std::move(param->error));
>       delete param;
>       break;
>     }
>     case MSG_CREATE_SESSIONDESCRIPTION_FAILED: {
>       CreateSessionDescriptionMsg* param =
>           static_cast<CreateSessionDescriptionMsg*>(msg->pdata);
>-      param->observer->OnFailure(param->error);
>+      param->observer->OnFailure(std::move(param->error));
>       delete param;
>       break;
>     }
>@@ -2763,25 +3354,54 @@ void PeerConnection::OnMessage(rtc::Message* msg) {
>       sctp_data_channels_to_free_.clear();
>       break;
>     }
>+    case MSG_REPORT_USAGE_PATTERN: {
>+      ReportUsagePattern();
>+      break;
>+    }
>     default:
>       RTC_NOTREACHED() << "Not implemented";
>       break;
>   }
> }
> 
>+cricket::VoiceMediaChannel* PeerConnection::voice_media_channel() const {
>+  RTC_DCHECK(!IsUnifiedPlan());
>+  auto* voice_channel = static_cast<cricket::VoiceChannel*>(
>+      GetAudioTransceiver()->internal()->channel());
>+  if (voice_channel) {
>+    return voice_channel->media_channel();
>+  } else {
>+    return nullptr;
>+  }
>+}
>+
>+cricket::VideoMediaChannel* PeerConnection::video_media_channel() const {
>+  RTC_DCHECK(!IsUnifiedPlan());
>+  auto* video_channel = static_cast<cricket::VideoChannel*>(
>+      GetVideoTransceiver()->internal()->channel());
>+  if (video_channel) {
>+    return video_channel->media_channel();
>+  } else {
>+    return nullptr;
>+  }
>+}
>+
> void PeerConnection::CreateAudioReceiver(
>     MediaStreamInterface* stream,
>     const RtpSenderInfo& remote_sender_info) {
>   std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams;
>   streams.push_back(rtc::scoped_refptr<MediaStreamInterface>(stream));
>-  rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
>-      receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
>-          signaling_thread(),
>-          new AudioRtpReceiver(worker_thread(), remote_sender_info.sender_id,
>-                               streams, remote_sender_info.first_ssrc,
>-                               voice_media_channel()));
>+  // TODO(https://crbug.com/webrtc/9480): When we remove remote_streams(), use
>+  // the constructor taking stream IDs instead.
>+  auto* audio_receiver = new AudioRtpReceiver(
>+      worker_thread(), remote_sender_info.sender_id, streams);
>+  audio_receiver->SetVoiceMediaChannel(voice_media_channel());
>+  audio_receiver->SetupMediaChannel(remote_sender_info.first_ssrc);
>+  auto receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
>+      signaling_thread(), audio_receiver);
>   GetAudioTransceiver()->internal()->AddReceiver(receiver);
>-  observer_->OnAddTrack(receiver, std::move(streams));
>+  Observer()->OnAddTrack(receiver, std::move(streams));
>+  NoteUsageEvent(UsageEvent::AUDIO_ADDED);
> }
> 
> void PeerConnection::CreateVideoReceiver(
>@@ -2789,14 +3409,17 @@ void PeerConnection::CreateVideoReceiver(
>     const RtpSenderInfo& remote_sender_info) {
>   std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams;
>   streams.push_back(rtc::scoped_refptr<MediaStreamInterface>(stream));
>-  rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
>-      receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
>-          signaling_thread(),
>-          new VideoRtpReceiver(worker_thread(), remote_sender_info.sender_id,
>-                               streams, remote_sender_info.first_ssrc,
>-                               video_media_channel()));
>+  // TODO(https://crbug.com/webrtc/9480): When we remove remote_streams(), use
>+  // the constructor taking stream IDs instead.
>+  auto* video_receiver = new VideoRtpReceiver(
>+      worker_thread(), remote_sender_info.sender_id, streams);
>+  video_receiver->SetVideoMediaChannel(video_media_channel());
>+  video_receiver->SetupMediaChannel(remote_sender_info.first_ssrc);
>+  auto receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
>+      signaling_thread(), video_receiver);
>   GetVideoTransceiver()->internal()->AddReceiver(receiver);
>-  observer_->OnAddTrack(receiver, std::move(streams));
>+  Observer()->OnAddTrack(receiver, std::move(streams));
>+  NoteUsageEvent(UsageEvent::VIDEO_ADDED);
> }
> 
> // TODO(deadbeef): Keep RtpReceivers around even if track goes away in remote
>@@ -2820,19 +3443,20 @@ rtc::scoped_refptr<RtpReceiverInterface> PeerConnection::RemoveAndStopReceiver(
> void PeerConnection::AddAudioTrack(AudioTrackInterface* track,
>                                    MediaStreamInterface* stream) {
>   RTC_DCHECK(!IsClosed());
>+  RTC_DCHECK(track);
>+  RTC_DCHECK(stream);
>   auto sender = FindSenderForTrack(track);
>   if (sender) {
>     // We already have a sender for this track, so just change the stream_id
>     // so that it's correct in the next call to CreateOffer.
>-    sender->internal()->set_stream_id(stream->label());
>+    sender->internal()->set_stream_ids({stream->id()});
>     return;
>   }
> 
>   // Normal case; we've never seen this track before.
>-  auto new_sender =
>-      CreateSender(cricket::MEDIA_TYPE_AUDIO, track, {stream->label()});
>-  static_cast<AudioRtpSender*>(new_sender->internal())
>-      ->SetMediaChannel(voice_media_channel());
>+  auto new_sender = CreateSender(cricket::MEDIA_TYPE_AUDIO, track->id(), track,
>+                                 {stream->id()});
>+  new_sender->internal()->SetVoiceMediaChannel(voice_media_channel());
>   GetAudioTransceiver()->internal()->AddSender(new_sender);
>   // If the sender has already been configured in SDP, we call SetSsrc,
>   // which will connect the sender to the underlying transport. This can
>@@ -2841,7 +3465,7 @@ void PeerConnection::AddAudioTrack(AudioTrackInterface* track,
>   // session description is not changed and RemoveStream is called, and
>   // later AddStream is called again with the same stream.
>   const RtpSenderInfo* sender_info =
>-      FindSenderInfo(local_audio_sender_infos_, stream->label(), track->id());
>+      FindSenderInfo(local_audio_sender_infos_, stream->id(), track->id());
>   if (sender_info) {
>     new_sender->internal()->SetSsrc(sender_info->first_ssrc);
>   }
>@@ -2864,22 +3488,23 @@ void PeerConnection::RemoveAudioTrack(AudioTrackInterface* track,
> void PeerConnection::AddVideoTrack(VideoTrackInterface* track,
>                                    MediaStreamInterface* stream) {
>   RTC_DCHECK(!IsClosed());
>+  RTC_DCHECK(track);
>+  RTC_DCHECK(stream);
>   auto sender = FindSenderForTrack(track);
>   if (sender) {
>     // We already have a sender for this track, so just change the stream_id
>     // so that it's correct in the next call to CreateOffer.
>-    sender->internal()->set_stream_id(stream->label());
>+    sender->internal()->set_stream_ids({stream->id()});
>     return;
>   }
> 
>   // Normal case; we've never seen this track before.
>-  auto new_sender =
>-      CreateSender(cricket::MEDIA_TYPE_VIDEO, track, {stream->label()});
>-  static_cast<VideoRtpSender*>(new_sender->internal())
>-      ->SetMediaChannel(video_media_channel());
>+  auto new_sender = CreateSender(cricket::MEDIA_TYPE_VIDEO, track->id(), track,
>+                                 {stream->id()});
>+  new_sender->internal()->SetVideoMediaChannel(video_media_channel());
>   GetVideoTransceiver()->internal()->AddSender(new_sender);
>   const RtpSenderInfo* sender_info =
>-      FindSenderInfo(local_video_sender_infos_, stream->label(), track->id());
>+      FindSenderInfo(local_video_sender_infos_, stream->id(), track->id());
>   if (sender_info) {
>     new_sender->internal()->SetSsrc(sender_info->first_ssrc);
>   }
>@@ -2915,7 +3540,7 @@ void PeerConnection::SetIceConnectionState(IceConnectionState new_state) {
>              PeerConnectionInterface::kIceConnectionClosed);
> 
>   ice_connection_state_ = new_state;
>-  observer_->OnIceConnectionChange(ice_connection_state_);
>+  Observer()->OnIceConnectionChange(ice_connection_state_);
> }
> 
> void PeerConnection::OnIceGatheringChange(
>@@ -2925,7 +3550,7 @@ void PeerConnection::OnIceGatheringChange(
>     return;
>   }
>   ice_gathering_state_ = new_state;
>-  observer_->OnIceGatheringChange(ice_gathering_state_);
>+  Observer()->OnIceGatheringChange(ice_gathering_state_);
> }
> 
> void PeerConnection::OnIceCandidate(
>@@ -2934,7 +3559,12 @@ void PeerConnection::OnIceCandidate(
>   if (IsClosed()) {
>     return;
>   }
>-  observer_->OnIceCandidate(candidate.get());
>+  NoteUsageEvent(UsageEvent::CANDIDATE_COLLECTED);
>+  if (candidate->candidate().type() == LOCAL_PORT_TYPE &&
>+      candidate->candidate().address().IsPrivateIP()) {
>+    NoteUsageEvent(UsageEvent::PRIVATE_CANDIDATE_COLLECTED);
>+  }
>+  Observer()->OnIceCandidate(candidate.get());
> }
> 
> void PeerConnection::OnIceCandidatesRemoved(
>@@ -2943,7 +3573,7 @@ void PeerConnection::OnIceCandidatesRemoved(
>   if (IsClosed()) {
>     return;
>   }
>-  observer_->OnIceCandidatesRemoved(candidates);
>+  Observer()->OnIceCandidatesRemoved(candidates);
> }
> 
> void PeerConnection::ChangeSignalingState(
>@@ -2959,13 +3589,13 @@ void PeerConnection::ChangeSignalingState(
>   signaling_state_ = signaling_state;
>   if (signaling_state == kClosed) {
>     ice_connection_state_ = kIceConnectionClosed;
>-    observer_->OnIceConnectionChange(ice_connection_state_);
>+    Observer()->OnIceConnectionChange(ice_connection_state_);
>     if (ice_gathering_state_ != kIceGatheringComplete) {
>       ice_gathering_state_ = kIceGatheringComplete;
>-      observer_->OnIceGatheringChange(ice_gathering_state_);
>+      Observer()->OnIceGatheringChange(ice_gathering_state_);
>     }
>   }
>-  observer_->OnSignalingChange(signaling_state_);
>+  Observer()->OnSignalingChange(signaling_state_);
> }
> 
> void PeerConnection::OnAudioTrackAdded(AudioTrackInterface* track,
>@@ -2974,7 +3604,7 @@ void PeerConnection::OnAudioTrackAdded(AudioTrackInterface* track,
>     return;
>   }
>   AddAudioTrack(track, stream);
>-  observer_->OnRenegotiationNeeded();
>+  Observer()->OnRenegotiationNeeded();
> }
> 
> void PeerConnection::OnAudioTrackRemoved(AudioTrackInterface* track,
>@@ -2983,7 +3613,7 @@ void PeerConnection::OnAudioTrackRemoved(AudioTrackInterface* track,
>     return;
>   }
>   RemoveAudioTrack(track, stream);
>-  observer_->OnRenegotiationNeeded();
>+  Observer()->OnRenegotiationNeeded();
> }
> 
> void PeerConnection::OnVideoTrackAdded(VideoTrackInterface* track,
>@@ -2992,7 +3622,7 @@ void PeerConnection::OnVideoTrackAdded(VideoTrackInterface* track,
>     return;
>   }
>   AddVideoTrack(track, stream);
>-  observer_->OnRenegotiationNeeded();
>+  Observer()->OnRenegotiationNeeded();
> }
> 
> void PeerConnection::OnVideoTrackRemoved(VideoTrackInterface* track,
>@@ -3001,7 +3631,7 @@ void PeerConnection::OnVideoTrackRemoved(VideoTrackInterface* track,
>     return;
>   }
>   RemoveVideoTrack(track, stream);
>-  observer_->OnRenegotiationNeeded();
>+  Observer()->OnRenegotiationNeeded();
> }
> 
> void PeerConnection::PostSetSessionDescriptionSuccess(
>@@ -3013,18 +3643,20 @@ void PeerConnection::PostSetSessionDescriptionSuccess(
> 
> void PeerConnection::PostSetSessionDescriptionFailure(
>     SetSessionDescriptionObserver* observer,
>-    const std::string& error) {
>+    RTCError&& error) {
>+  RTC_DCHECK(!error.ok());
>   SetSessionDescriptionMsg* msg = new SetSessionDescriptionMsg(observer);
>-  msg->error = error;
>+  msg->error = std::move(error);
>   signaling_thread()->Post(RTC_FROM_HERE, this,
>                            MSG_SET_SESSIONDESCRIPTION_FAILED, msg);
> }
> 
> void PeerConnection::PostCreateSessionDescriptionFailure(
>     CreateSessionDescriptionObserver* observer,
>-    const std::string& error) {
>+    RTCError error) {
>+  RTC_DCHECK(!error.ok());
>   CreateSessionDescriptionMsg* msg = new CreateSessionDescriptionMsg(observer);
>-  msg->error = error;
>+  msg->error = std::move(error);
>   signaling_thread()->Post(RTC_FROM_HERE, this,
>                            MSG_CREATE_SESSIONDESCRIPTION_FAILED, msg);
> }
>@@ -3058,6 +3690,7 @@ void PeerConnection::GetOptionsForOffer(
> 
>   session_options->rtcp_cname = rtcp_cname_;
>   session_options->crypto_options = factory_->options().crypto_options;
>+  session_options->is_unified_plan = IsUnifiedPlan();
> }
> 
> void PeerConnection::GetOptionsForPlanBOffer(
>@@ -3092,9 +3725,9 @@ void PeerConnection::GetOptionsForPlanBOffer(
>         (offer_answer_options.offer_to_receive_video > 0);
>   }
> 
>-  rtc::Optional<size_t> audio_index;
>-  rtc::Optional<size_t> video_index;
>-  rtc::Optional<size_t> data_index;
>+  absl::optional<size_t> audio_index;
>+  absl::optional<size_t> video_index;
>+  absl::optional<size_t> data_index;
>   // If a current description exists, generate m= sections in the same order,
>   // using the first audio/video/data section that appears and rejecting
>   // extraneous ones.
>@@ -3165,15 +3798,24 @@ GetMediaDescriptionOptionsForTransceiver(
>         transceiver,
>     const std::string& mid) {
>   cricket::MediaDescriptionOptions media_description_options(
>-      transceiver->internal()->media_type(), mid, transceiver->direction(),
>+      transceiver->media_type(), mid, transceiver->direction(),
>       transceiver->stopped());
>-  cricket::SenderOptions sender_options;
>-  sender_options.track_id = transceiver->sender()->id();
>-  sender_options.stream_ids = transceiver->sender()->stream_ids();
>-  // TODO(bugs.webrtc.org/7600): Set num_sim_layers to the number of encodings
>-  // set in the RTP parameters when the transceiver was added.
>-  sender_options.num_sim_layers = 1;
>-  media_description_options.sender_options.push_back(sender_options);
>+  // This behavior is specified in JSEP. The gist is that:
>+  // 1. The MSID is included if the RtpTransceiver's direction is sendonly or
>+  //    sendrecv.
>+  // 2. If the MSID is included, then it must be included in any subsequent
>+  //    offer/answer exactly the same until the RtpTransceiver is stopped.
>+  if (!transceiver->stopped() &&
>+      (RtpTransceiverDirectionHasSend(transceiver->direction()) ||
>+       transceiver->internal()->has_ever_been_used_to_send())) {
>+    cricket::SenderOptions sender_options;
>+    sender_options.track_id = transceiver->sender()->id();
>+    sender_options.stream_ids = transceiver->sender()->stream_ids();
>+    // TODO(bugs.webrtc.org/7600): Set num_sim_layers to the number of encodings
>+    // set in the RTP parameters when the transceiver was added.
>+    sender_options.num_sim_layers = 1;
>+    media_description_options.sender_options.push_back(sender_options);
>+  }
>   return media_description_options;
> }
> 
>@@ -3219,12 +3861,11 @@ void PeerConnection::GetOptionsForUnifiedPlanOffer(
>       RTC_CHECK(transceiver);
>       // A media section is considered eligible for recycling if it is marked as
>       // rejected in either the local or remote description.
>-      if (had_been_rejected) {
>+      if (had_been_rejected && transceiver->stopped()) {
>         session_options->media_description_options.push_back(
>-            cricket::MediaDescriptionOptions(
>-                transceiver->internal()->media_type(), mid,
>-                RtpTransceiverDirection::kInactive,
>-                /*stopped=*/true));
>+            cricket::MediaDescriptionOptions(transceiver->media_type(), mid,
>+                                             RtpTransceiverDirection::kInactive,
>+                                             /*stopped=*/true));
>         recycleable_mline_indices.push(i);
>       } else {
>         session_options->media_description_options.push_back(
>@@ -3307,6 +3948,7 @@ void PeerConnection::GetOptionsForAnswer(
> 
>   session_options->rtcp_cname = rtcp_cname_;
>   session_options->crypto_options = factory_->options().crypto_options;
>+  session_options->is_unified_plan = IsUnifiedPlan();
> }
> 
> void PeerConnection::GetOptionsForPlanBAnswer(
>@@ -3331,21 +3973,18 @@ void PeerConnection::GetOptionsForPlanBAnswer(
>     recv_video = (offer_answer_options.offer_to_receive_video > 0);
>   }
> 
>-  rtc::Optional<size_t> audio_index;
>-  rtc::Optional<size_t> video_index;
>-  rtc::Optional<size_t> data_index;
>-  if (remote_description()) {
>-    // The pending remote description should be an offer.
>-    RTC_DCHECK(remote_description()->GetType() == SdpType::kOffer);
>-    // Generate m= sections that match those in the offer.
>-    // Note that mediasession.cc will handle intersection our preferred
>-    // direction with the offered direction.
>-    GenerateMediaDescriptionOptions(
>-        remote_description(),
>-        RtpTransceiverDirectionFromSendRecv(send_audio, recv_audio),
>-        RtpTransceiverDirectionFromSendRecv(send_video, recv_video),
>-        &audio_index, &video_index, &data_index, session_options);
>-  }
>+  absl::optional<size_t> audio_index;
>+  absl::optional<size_t> video_index;
>+  absl::optional<size_t> data_index;
>+
>+  // Generate m= sections that match those in the offer.
>+  // Note that mediasession.cc will handle intersection our preferred
>+  // direction with the offered direction.
>+  GenerateMediaDescriptionOptions(
>+      remote_description(),
>+      RtpTransceiverDirectionFromSendRecv(send_audio, recv_audio),
>+      RtpTransceiverDirectionFromSendRecv(send_video, recv_video), &audio_index,
>+      &video_index, &data_index, session_options);
> 
>   cricket::MediaDescriptionOptions* audio_media_description_options =
>       !audio_index ? nullptr
>@@ -3376,10 +4015,11 @@ void PeerConnection::GetOptionsForUnifiedPlanAnswer(
>           GetMediaDescriptionOptionsForTransceiver(transceiver, content.name));
>     } else {
>       RTC_CHECK_EQ(cricket::MEDIA_TYPE_DATA, media_type);
>-      RTC_CHECK(GetDataMid());
>-      // Always reject a data section if it has already been rejected.
>-      // Additionally, reject all data sections except for the first one.
>-      if (content.rejected || content.name != *GetDataMid()) {
>+      // Reject all data sections if data channels are disabled.
>+      // Reject a data section if it has already been rejected.
>+      // Reject all data sections except for the first one.
>+      if (data_channel_type_ == cricket::DCT_NONE || content.rejected ||
>+          content.name != *GetDataMid()) {
>         session_options->media_description_options.push_back(
>             GetMediaDescriptionOptionsForRejectedData(content.name));
>       } else {
>@@ -3394,9 +4034,9 @@ void PeerConnection::GenerateMediaDescriptionOptions(
>     const SessionDescriptionInterface* session_desc,
>     RtpTransceiverDirection audio_direction,
>     RtpTransceiverDirection video_direction,
>-    rtc::Optional<size_t>* audio_index,
>-    rtc::Optional<size_t>* video_index,
>-    rtc::Optional<size_t>* data_index,
>+    absl::optional<size_t>* audio_index,
>+    absl::optional<size_t>* video_index,
>+    absl::optional<size_t>* data_index,
>     cricket::MediaSessionOptions* session_options) {
>   for (const cricket::ContentInfo& content :
>        session_desc->description()->contents()) {
>@@ -3465,17 +4105,17 @@ PeerConnection::GetMediaDescriptionOptionsForRejectedData(
>   return options;
> }
> 
>-rtc::Optional<std::string> PeerConnection::GetDataMid() const {
>+absl::optional<std::string> PeerConnection::GetDataMid() const {
>   switch (data_channel_type_) {
>     case cricket::DCT_RTP:
>       if (!rtp_data_channel_) {
>-        return rtc::nullopt;
>+        return absl::nullopt;
>       }
>       return rtp_data_channel_->content_name();
>     case cricket::DCT_SCTP:
>-      return sctp_content_name_;
>+      return sctp_mid_;
>     default:
>-      return rtc::nullopt;
>+      return absl::nullopt;
>   }
> }
> 
>@@ -3490,6 +4130,8 @@ void PeerConnection::UpdateRemoteSendersList(
>     bool default_sender_needed,
>     cricket::MediaType media_type,
>     StreamCollection* new_streams) {
>+  RTC_DCHECK(!IsUnifiedPlan());
>+
>   std::vector<RtpSenderInfo>* current_senders =
>       GetRemoteSenderInfos(media_type);
> 
>@@ -3501,9 +4143,16 @@ void PeerConnection::UpdateRemoteSendersList(
>     const RtpSenderInfo& info = *sender_it;
>     const cricket::StreamParams* params =
>         cricket::GetStreamBySsrc(streams, info.first_ssrc);
>-    bool sender_exists = params && params->id == info.sender_id;
>+    std::string params_stream_id;
>+    if (params) {
>+      params_stream_id =
>+          (!params->first_stream_id().empty() ? params->first_stream_id()
>+                                              : kDefaultStreamId);
>+    }
>+    bool sender_exists = params && params->id == info.sender_id &&
>+                         params_stream_id == info.stream_id;
>     // If this is a default track, and we still need it, don't remove it.
>-    if ((info.stream_label == kDefaultStreamLabel && default_sender_needed) ||
>+    if ((info.stream_id == kDefaultStreamId && default_sender_needed) ||
>         sender_exists) {
>       ++sender_it;
>     } else {
>@@ -3514,26 +4163,39 @@ void PeerConnection::UpdateRemoteSendersList(
> 
>   // Find new and active senders.
>   for (const cricket::StreamParams& params : streams) {
>-    // The sync_label is the MediaStream label and the |stream.id| is the
>-    // sender id.
>-    const std::string& stream_label = params.sync_label;
>+    if (!params.has_ssrcs()) {
>+      // The remote endpoint has streams, but didn't signal ssrcs. For an active
>+      // sender, this means it is coming from a Unified Plan endpoint,so we just
>+      // create a default.
>+      default_sender_needed = true;
>+      break;
>+    }
>+
>+    // |params.id| is the sender id and the stream id uses the first of
>+    // |params.stream_ids|. The remote description could come from a Unified
>+    // Plan endpoint, with multiple or no stream_ids() signaled. Since this is
>+    // not supported in Plan B, we just take the first here and create the
>+    // default stream ID if none is specified.
>+    const std::string& stream_id =
>+        (!params.first_stream_id().empty() ? params.first_stream_id()
>+                                           : kDefaultStreamId);
>     const std::string& sender_id = params.id;
>     uint32_t ssrc = params.first_ssrc();
> 
>     rtc::scoped_refptr<MediaStreamInterface> stream =
>-        remote_streams_->find(stream_label);
>+        remote_streams_->find(stream_id);
>     if (!stream) {
>       // This is a new MediaStream. Create a new remote MediaStream.
>       stream = MediaStreamProxy::Create(rtc::Thread::Current(),
>-                                        MediaStream::Create(stream_label));
>+                                        MediaStream::Create(stream_id));
>       remote_streams_->AddStream(stream);
>       new_streams->AddStream(stream);
>     }
> 
>     const RtpSenderInfo* sender_info =
>-        FindSenderInfo(*current_senders, stream_label, sender_id);
>+        FindSenderInfo(*current_senders, stream_id, sender_id);
>     if (!sender_info) {
>-      current_senders->push_back(RtpSenderInfo(stream_label, sender_id, ssrc));
>+      current_senders->push_back(RtpSenderInfo(stream_id, sender_id, ssrc));
>       OnRemoteSenderAdded(current_senders->back(), media_type);
>     }
>   }
>@@ -3541,22 +4203,22 @@ void PeerConnection::UpdateRemoteSendersList(
>   // Add default sender if necessary.
>   if (default_sender_needed) {
>     rtc::scoped_refptr<MediaStreamInterface> default_stream =
>-        remote_streams_->find(kDefaultStreamLabel);
>+        remote_streams_->find(kDefaultStreamId);
>     if (!default_stream) {
>       // Create the new default MediaStream.
>       default_stream = MediaStreamProxy::Create(
>-          rtc::Thread::Current(), MediaStream::Create(kDefaultStreamLabel));
>+          rtc::Thread::Current(), MediaStream::Create(kDefaultStreamId));
>       remote_streams_->AddStream(default_stream);
>       new_streams->AddStream(default_stream);
>     }
>     std::string default_sender_id = (media_type == cricket::MEDIA_TYPE_AUDIO)
>                                         ? kDefaultAudioSenderId
>                                         : kDefaultVideoSenderId;
>-    const RtpSenderInfo* default_sender_info = FindSenderInfo(
>-        *current_senders, kDefaultStreamLabel, default_sender_id);
>+    const RtpSenderInfo* default_sender_info =
>+        FindSenderInfo(*current_senders, kDefaultStreamId, default_sender_id);
>     if (!default_sender_info) {
>       current_senders->push_back(
>-          RtpSenderInfo(kDefaultStreamLabel, default_sender_id, 0));
>+          RtpSenderInfo(kDefaultStreamId, default_sender_id, 0));
>       OnRemoteSenderAdded(current_senders->back(), media_type);
>     }
>   }
>@@ -3564,9 +4226,11 @@ void PeerConnection::UpdateRemoteSendersList(
> 
> void PeerConnection::OnRemoteSenderAdded(const RtpSenderInfo& sender_info,
>                                          cricket::MediaType media_type) {
>-  MediaStreamInterface* stream =
>-      remote_streams_->find(sender_info.stream_label);
>+  RTC_LOG(LS_INFO) << "Creating " << cricket::MediaTypeToString(media_type)
>+                   << " receiver for track_id=" << sender_info.sender_id
>+                   << " and stream_id=" << sender_info.stream_id;
> 
>+  MediaStreamInterface* stream = remote_streams_->find(sender_info.stream_id);
>   if (media_type == cricket::MEDIA_TYPE_AUDIO) {
>     CreateAudioReceiver(stream, sender_info);
>   } else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
>@@ -3578,8 +4242,11 @@ void PeerConnection::OnRemoteSenderAdded(const RtpSenderInfo& sender_info,
> 
> void PeerConnection::OnRemoteSenderRemoved(const RtpSenderInfo& sender_info,
>                                            cricket::MediaType media_type) {
>-  MediaStreamInterface* stream =
>-      remote_streams_->find(sender_info.stream_label);
>+  RTC_LOG(LS_INFO) << "Removing " << cricket::MediaTypeToString(media_type)
>+                   << " receiver for track_id=" << sender_info.sender_id
>+                   << " and stream_id=" << sender_info.stream_id;
>+
>+  MediaStreamInterface* stream = remote_streams_->find(sender_info.stream_id);
> 
>   rtc::scoped_refptr<RtpReceiverInterface> receiver;
>   if (media_type == cricket::MEDIA_TYPE_AUDIO) {
>@@ -3606,7 +4273,7 @@ void PeerConnection::OnRemoteSenderRemoved(const RtpSenderInfo& sender_info,
>     RTC_NOTREACHED() << "Invalid media type";
>   }
>   if (receiver) {
>-    observer_->OnRemoveTrack(receiver);
>+    Observer()->OnRemoveTrack(receiver);
>   }
> }
> 
>@@ -3621,7 +4288,7 @@ void PeerConnection::UpdateEndedRemoteMediaStreams() {
> 
>   for (auto& stream : streams_to_remove) {
>     remote_streams_->RemoveStream(stream);
>-    observer_->OnRemoveStream(std::move(stream));
>+    Observer()->OnRemoveStream(std::move(stream));
>   }
> }
> 
>@@ -3630,7 +4297,7 @@ void PeerConnection::UpdateLocalSenders(
>     cricket::MediaType media_type) {
>   std::vector<RtpSenderInfo>* current_senders = GetLocalSenderInfos(media_type);
> 
>-  // Find removed tracks. I.e., tracks where the track id, stream label or ssrc
>+  // Find removed tracks. I.e., tracks where the track id, stream id or ssrc
>   // don't match the new StreamParam.
>   for (auto sender_it = current_senders->begin();
>        sender_it != current_senders->end();
>@@ -3639,7 +4306,7 @@ void PeerConnection::UpdateLocalSenders(
>     const cricket::StreamParams* params =
>         cricket::GetStreamBySsrc(streams, info.first_ssrc);
>     if (!params || params->id != info.sender_id ||
>-        params->sync_label != info.stream_label) {
>+        params->first_stream_id() != info.stream_id) {
>       OnLocalSenderRemoved(info, media_type);
>       sender_it = current_senders->erase(sender_it);
>     } else {
>@@ -3651,13 +4318,13 @@ void PeerConnection::UpdateLocalSenders(
>   for (const cricket::StreamParams& params : streams) {
>     // The sync_label is the MediaStream label and the |stream.id| is the
>     // sender id.
>-    const std::string& stream_label = params.sync_label;
>+    const std::string& stream_id = params.first_stream_id();
>     const std::string& sender_id = params.id;
>     uint32_t ssrc = params.first_ssrc();
>     const RtpSenderInfo* sender_info =
>-        FindSenderInfo(*current_senders, stream_label, sender_id);
>+        FindSenderInfo(*current_senders, stream_id, sender_id);
>     if (!sender_info) {
>-      current_senders->push_back(RtpSenderInfo(stream_label, sender_id, ssrc));
>+      current_senders->push_back(RtpSenderInfo(stream_id, sender_id, ssrc));
>       OnLocalSenderAdded(current_senders->back(), media_type);
>     }
>   }
>@@ -3665,6 +4332,7 @@ void PeerConnection::UpdateLocalSenders(
> 
> void PeerConnection::OnLocalSenderAdded(const RtpSenderInfo& sender_info,
>                                         cricket::MediaType media_type) {
>+  RTC_DCHECK(!IsUnifiedPlan());
>   auto sender = FindSenderById(sender_info.sender_id);
>   if (!sender) {
>     RTC_LOG(LS_WARNING) << "An unknown RtpSender with id "
>@@ -3675,11 +4343,11 @@ void PeerConnection::OnLocalSenderAdded(const RtpSenderInfo& sender_info,
> 
>   if (sender->media_type() != media_type) {
>     RTC_LOG(LS_WARNING) << "An RtpSender has been configured in the local"
>-                        << " description with an unexpected media type.";
>+                           " description with an unexpected media type.";
>     return;
>   }
> 
>-  sender->internal()->set_stream_id(sender_info.stream_label);
>+  sender->internal()->set_stream_ids({sender_info.stream_id});
>   sender->internal()->SetSsrc(sender_info.first_ssrc);
> }
> 
>@@ -3697,7 +4365,7 @@ void PeerConnection::OnLocalSenderRemoved(const RtpSenderInfo& sender_info,
>   // match with the calls to CreateSender, AddStream and RemoveStream.
>   if (sender->media_type() != media_type) {
>     RTC_LOG(LS_WARNING) << "An RtpSender has been configured in the local"
>-                        << " description with an unexpected media type.";
>+                           " description with an unexpected media type.";
>     return;
>   }
> 
>@@ -3715,7 +4383,7 @@ void PeerConnection::UpdateLocalRtpDataChannels(
>     // MediaStreams and Tracks.
>     // For MediaStreams, the sync_label is the MediaStream label and the
>     // track label is the same as |streamid|.
>-    const std::string& channel_label = params.sync_label;
>+    const std::string& channel_label = params.first_stream_id();
>     auto data_channel_it = rtp_data_channels_.find(channel_label);
>     if (data_channel_it == rtp_data_channels_.end()) {
>       RTC_LOG(LS_ERROR) << "channel label not found";
>@@ -3737,9 +4405,9 @@ void PeerConnection::UpdateRemoteRtpDataChannels(
>   for (const cricket::StreamParams& params : streams) {
>     // The data channel label is either the mslabel or the SSRC if the mslabel
>     // does not exist. Ex a=ssrc:444330170 mslabel:test1.
>-    std::string label = params.sync_label.empty()
>+    std::string label = params.first_stream_id().empty()
>                             ? rtc::ToString(params.first_ssrc())
>-                            : params.sync_label;
>+                            : params.first_stream_id();
>     auto data_channel_it = rtp_data_channels_.find(label);
>     if (data_channel_it == rtp_data_channels_.end()) {
>       // This is a new data channel.
>@@ -3786,13 +4454,13 @@ void PeerConnection::CreateRemoteRtpDataChannel(const std::string& label,
>       InternalCreateDataChannel(label, nullptr));
>   if (!channel.get()) {
>     RTC_LOG(LS_WARNING) << "Remote peer requested a DataChannel but"
>-                        << "CreateDataChannel failed.";
>+                           "CreateDataChannel failed.";
>     return;
>   }
>   channel->SetReceiveSsrc(remote_ssrc);
>   rtc::scoped_refptr<DataChannelInterface> proxy_channel =
>       DataChannelProxy::Create(signaling_thread(), channel);
>-  observer_->OnDataChannel(std::move(proxy_channel));
>+  Observer()->OnDataChannel(std::move(proxy_channel));
> }
> 
> rtc::scoped_refptr<DataChannel> PeerConnection::InternalCreateDataChannel(
>@@ -3819,7 +4487,7 @@ rtc::scoped_refptr<DataChannel> PeerConnection::InternalCreateDataChannel(
>       }
>     } else if (!sid_allocator_.ReserveSid(new_config.id)) {
>       RTC_LOG(LS_ERROR) << "Failed to create a SCTP data channel "
>-                        << "because the id is already in use or out of range.";
>+                           "because the id is already in use or out of range.";
>       return nullptr;
>     }
>   }
>@@ -3845,7 +4513,7 @@ rtc::scoped_refptr<DataChannel> PeerConnection::InternalCreateDataChannel(
>                                   &PeerConnection::OnSctpDataChannelClosed);
>   }
> 
>-  SignalDataChannelCreated(channel.get());
>+  SignalDataChannelCreated_(channel.get());
>   return channel;
> }
> 
>@@ -3872,6 +4540,8 @@ void PeerConnection::OnSctpDataChannelClosed(DataChannel* channel) {
>        ++it) {
>     if (it->get() == channel) {
>       if (channel->id() >= 0) {
>+        // After the closing procedure is done, it's safe to use this ID for
>+        // another data channel.
>         sid_allocator_.ReleaseSid(channel->id());
>       }
>       // Since this method is triggered by a signal from the DataChannel,
>@@ -3913,7 +4583,8 @@ void PeerConnection::OnDataChannelOpenMessage(
> 
>   rtc::scoped_refptr<DataChannelInterface> proxy_channel =
>       DataChannelProxy::Create(signaling_thread(), channel);
>-  observer_->OnDataChannel(std::move(proxy_channel));
>+  Observer()->OnDataChannel(std::move(proxy_channel));
>+  NoteUsageEvent(UsageEvent::DATA_ADDED);
> }
> 
> rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
>@@ -3922,7 +4593,7 @@ PeerConnection::GetAudioTransceiver() const {
>   // audio/video transceiver.
>   RTC_DCHECK(!IsUnifiedPlan());
>   for (auto transceiver : transceivers_) {
>-    if (transceiver->internal()->media_type() == cricket::MEDIA_TYPE_AUDIO) {
>+    if (transceiver->media_type() == cricket::MEDIA_TYPE_AUDIO) {
>       return transceiver;
>     }
>   }
>@@ -3936,7 +4607,7 @@ PeerConnection::GetVideoTransceiver() const {
>   // audio/video transceiver.
>   RTC_DCHECK(!IsUnifiedPlan());
>   for (auto transceiver : transceivers_) {
>-    if (transceiver->internal()->media_type() == cricket::MEDIA_TYPE_VIDEO) {
>+    if (transceiver->media_type() == cricket::MEDIA_TYPE_VIDEO) {
>       return transceiver;
>     }
>   }
>@@ -4014,10 +4685,10 @@ std::vector<PeerConnection::RtpSenderInfo>* PeerConnection::GetLocalSenderInfos(
> 
> const PeerConnection::RtpSenderInfo* PeerConnection::FindSenderInfo(
>     const std::vector<PeerConnection::RtpSenderInfo>& infos,
>-    const std::string& stream_label,
>+    const std::string& stream_id,
>     const std::string sender_id) const {
>   for (const RtpSenderInfo& sender_info : infos) {
>-    if (sender_info.stream_label == stream_label &&
>+    if (sender_info.stream_id == stream_id &&
>         sender_info.sender_id == sender_id) {
>       return &sender_info;
>     }
>@@ -4035,60 +4706,64 @@ DataChannel* PeerConnection::FindDataChannelBySid(int sid) const {
> }
> 
> bool PeerConnection::InitializePortAllocator_n(
>+    const cricket::ServerAddresses& stun_servers,
>+    const std::vector<cricket::RelayServerConfig>& turn_servers,
>     const RTCConfiguration& configuration) {
>-  cricket::ServerAddresses stun_servers;
>-  std::vector<cricket::RelayServerConfig> turn_servers;
>-  if (ParseIceServers(configuration.servers, &stun_servers, &turn_servers) !=
>-      RTCErrorType::NONE) {
>-    return false;
>-  }
>-
>   port_allocator_->Initialize();
>-
>   // To handle both internal and externally created port allocator, we will
>   // enable BUNDLE here.
>-  int portallocator_flags = port_allocator_->flags();
>-  portallocator_flags |= cricket::PORTALLOCATOR_ENABLE_SHARED_SOCKET |
>-                         cricket::PORTALLOCATOR_ENABLE_IPV6 |
>-                         cricket::PORTALLOCATOR_ENABLE_IPV6_ON_WIFI;
>+  port_allocator_flags_ = port_allocator_->flags();
>+  port_allocator_flags_ |= cricket::PORTALLOCATOR_ENABLE_SHARED_SOCKET |
>+                           cricket::PORTALLOCATOR_ENABLE_IPV6 |
>+                           cricket::PORTALLOCATOR_ENABLE_IPV6_ON_WIFI;
>   // If the disable-IPv6 flag was specified, we'll not override it
>   // by experiment.
>   if (configuration.disable_ipv6) {
>-    portallocator_flags &= ~(cricket::PORTALLOCATOR_ENABLE_IPV6);
>+    port_allocator_flags_ &= ~(cricket::PORTALLOCATOR_ENABLE_IPV6);
>   } else if (webrtc::field_trial::FindFullName("WebRTC-IPv6Default")
>                  .find("Disabled") == 0) {
>-    portallocator_flags &= ~(cricket::PORTALLOCATOR_ENABLE_IPV6);
>+    port_allocator_flags_ &= ~(cricket::PORTALLOCATOR_ENABLE_IPV6);
>   }
> 
>   if (configuration.disable_ipv6_on_wifi) {
>-    portallocator_flags &= ~(cricket::PORTALLOCATOR_ENABLE_IPV6_ON_WIFI);
>+    port_allocator_flags_ &= ~(cricket::PORTALLOCATOR_ENABLE_IPV6_ON_WIFI);
>     RTC_LOG(LS_INFO) << "IPv6 candidates on Wi-Fi are disabled.";
>   }
> 
>   if (configuration.tcp_candidate_policy == kTcpCandidatePolicyDisabled) {
>-    portallocator_flags |= cricket::PORTALLOCATOR_DISABLE_TCP;
>+    port_allocator_flags_ |= cricket::PORTALLOCATOR_DISABLE_TCP;
>     RTC_LOG(LS_INFO) << "TCP candidates are disabled.";
>   }
> 
>   if (configuration.candidate_network_policy ==
>       kCandidateNetworkPolicyLowCost) {
>-    portallocator_flags |= cricket::PORTALLOCATOR_DISABLE_COSTLY_NETWORKS;
>+    port_allocator_flags_ |= cricket::PORTALLOCATOR_DISABLE_COSTLY_NETWORKS;
>     RTC_LOG(LS_INFO) << "Do not gather candidates on high-cost networks";
>   }
> 
>-  port_allocator_->set_flags(portallocator_flags);
>+  if (configuration.disable_link_local_networks) {
>+    port_allocator_flags_ |= cricket::PORTALLOCATOR_DISABLE_LINK_LOCAL_NETWORKS;
>+    RTC_LOG(LS_INFO) << "Disable candidates on link-local network interfaces.";
>+  }
>+
>+  port_allocator_->set_flags(port_allocator_flags_);
>   // No step delay is used while allocating ports.
>   port_allocator_->set_step_delay(cricket::kMinimumStepDelay);
>   port_allocator_->set_candidate_filter(
>       ConvertIceTransportTypeToCandidateFilter(configuration.type));
>   port_allocator_->set_max_ipv6_networks(configuration.max_ipv6_networks);
> 
>+  auto turn_servers_copy = turn_servers;
>+  for (auto& turn_server : turn_servers_copy) {
>+    turn_server.tls_cert_verifier = tls_cert_verifier_.get();
>+  }
>   // Call this last since it may create pooled allocator sessions using the
>   // properties set above.
>-  port_allocator_->SetConfiguration(stun_servers, turn_servers,
>-                                    configuration.ice_candidate_pool_size,
>-                                    configuration.prune_turn_ports,
>-                                    configuration.turn_customizer);
>+  port_allocator_->SetConfiguration(
>+      stun_servers, std::move(turn_servers_copy),
>+      configuration.ice_candidate_pool_size, configuration.prune_turn_ports,
>+      configuration.turn_customizer,
>+      configuration.stun_candidate_keepalive_interval);
>   return true;
> }
> 
>@@ -4098,24 +4773,32 @@ bool PeerConnection::ReconfigurePortAllocator_n(
>     IceTransportsType type,
>     int candidate_pool_size,
>     bool prune_turn_ports,
>-    webrtc::TurnCustomizer* turn_customizer) {
>+    webrtc::TurnCustomizer* turn_customizer,
>+    absl::optional<int> stun_candidate_keepalive_interval) {
>   port_allocator_->set_candidate_filter(
>       ConvertIceTransportTypeToCandidateFilter(type));
>+  // According to JSEP, after setLocalDescription, changing the candidate pool
>+  // size is not allowed, and changing the set of ICE servers will not result
>+  // in new candidates being gathered.
>+  if (local_description()) {
>+    port_allocator_->FreezeCandidatePool();
>+  }
>+  // Add the custom tls turn servers if they exist.
>+  auto turn_servers_copy = turn_servers;
>+  for (auto& turn_server : turn_servers_copy) {
>+    turn_server.tls_cert_verifier = tls_cert_verifier_.get();
>+  }
>   // Call this last since it may create pooled allocator sessions using the
>   // candidate filter set above.
>   return port_allocator_->SetConfiguration(
>-      stun_servers, turn_servers, candidate_pool_size, prune_turn_ports,
>-      turn_customizer);
>+      stun_servers, std::move(turn_servers_copy), candidate_pool_size,
>+      prune_turn_ports, turn_customizer, stun_candidate_keepalive_interval);
> }
> 
> cricket::ChannelManager* PeerConnection::channel_manager() const {
>   return factory_->channel_manager();
> }
> 
>-MetricsObserverInterface* PeerConnection::metrics_observer() const {
>-  return uma_observer_;
>-}
>-
> bool PeerConnection::StartRtcEventLog_w(
>     std::unique_ptr<RtcEventLogOutput> output,
>     int64_t output_period_ms) {
>@@ -4150,16 +4833,21 @@ bool PeerConnection::GetSctpSslRole(rtc::SSLRole* role) {
>   if (!local_description() || !remote_description()) {
>     RTC_LOG(LS_INFO)
>         << "Local and Remote descriptions must be applied to get the "
>-        << "SSL Role of the SCTP transport.";
>+           "SSL Role of the SCTP transport.";
>     return false;
>   }
>   if (!sctp_transport_) {
>     RTC_LOG(LS_INFO) << "Non-rejected SCTP m= section is needed to get the "
>-                     << "SSL Role of the SCTP transport.";
>+                        "SSL Role of the SCTP transport.";
>     return false;
>   }
> 
>-  return transport_controller_->GetSslRole(*sctp_transport_name_, role);
>+  auto dtls_role = transport_controller_->GetDtlsRole(*sctp_mid_);
>+  if (dtls_role) {
>+    *role = *dtls_role;
>+    return true;
>+  }
>+  return false;
> }
> 
> bool PeerConnection::GetSslRole(const std::string& content_name,
>@@ -4167,29 +4855,16 @@ bool PeerConnection::GetSslRole(const std::string& content_name,
>   if (!local_description() || !remote_description()) {
>     RTC_LOG(LS_INFO)
>         << "Local and Remote descriptions must be applied to get the "
>-        << "SSL Role of the session.";
>+           "SSL Role of the session.";
>     return false;
>   }
> 
>-  return transport_controller_->GetSslRole(GetTransportName(content_name),
>-                                           role);
>-}
>-
>-// TODO(steveanton): Eventually it'd be nice to store the channels as a single
>-// vector of BaseChannel pointers instead of separate voice and video channel
>-// vectors. At that point, this will become a simple getter.
>-std::vector<cricket::BaseChannel*> PeerConnection::Channels() const {
>-  std::vector<cricket::BaseChannel*> channels;
>-  if (voice_channel()) {
>-    channels.push_back(voice_channel());
>-  }
>-  if (video_channel()) {
>-    channels.push_back(video_channel());
>-  }
>-  if (rtp_data_channel_) {
>-    channels.push_back(rtp_data_channel_);
>+  auto dtls_role = transport_controller_->GetDtlsRole(content_name);
>+  if (dtls_role) {
>+    *role = *dtls_role;
>+    return true;
>   }
>-  return channels;
>+  return false;
> }
> 
> void PeerConnection::SetSessionError(SessionError error,
>@@ -4201,42 +4876,16 @@ void PeerConnection::SetSessionError(SessionError error,
>   }
> }
> 
>-RTCError PeerConnection::UpdateSessionState(SdpType type,
>-                                            cricket::ContentSource source) {
>+RTCError PeerConnection::UpdateSessionState(
>+    SdpType type,
>+    cricket::ContentSource source,
>+    const cricket::SessionDescription* description) {
>   RTC_DCHECK_RUN_ON(signaling_thread());
> 
>   // If there's already a pending error then no state transition should happen.
>   // But all call-sites should be verifying this before calling us!
>   RTC_DCHECK(session_error() == SessionError::kNone);
> 
>-  // If this is an answer then we know whether to BUNDLE or not. If both the
>-  // local and remote side have agreed to BUNDLE, go ahead and enable it.
>-  if (type == SdpType::kAnswer) {
>-    const cricket::ContentGroup* local_bundle =
>-        local_description()->description()->GetGroupByName(
>-            cricket::GROUP_TYPE_BUNDLE);
>-    const cricket::ContentGroup* remote_bundle =
>-        remote_description()->description()->GetGroupByName(
>-            cricket::GROUP_TYPE_BUNDLE);
>-    if (local_bundle && remote_bundle) {
>-      // The answerer decides the transport to bundle on.
>-      const cricket::ContentGroup* answer_bundle =
>-          (source == cricket::CS_LOCAL ? local_bundle : remote_bundle);
>-      if (!EnableBundle(*answer_bundle)) {
>-        LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
>-                             kEnableBundleFailed);
>-      }
>-    }
>-  }
>-
>-  // Only push down the transport description after potentially enabling BUNDLE;
>-  // we don't want to push down a description on a transport about to be
>-  // destroyed.
>-  RTCError error = PushdownTransportDescription(source, type);
>-  if (!error.ok()) {
>-    return error;
>-  }
>-
>   // If this is answer-ish we're ready to let media flow.
>   if (type == SdpType::kPrAnswer || type == SdpType::kAnswer) {
>     EnableSending();
>@@ -4259,12 +4908,9 @@ RTCError PeerConnection::UpdateSessionState(SdpType type,
> 
>   // Update internal objects according to the session description's media
>   // descriptions.
>-  error = PushdownMediaDescription(type, source);
>+  RTCError error = PushdownMediaDescription(type, source);
>   if (!error.ok()) {
>-    SetSessionError(SessionError::kContent, error.message());
>-  }
>-  if (session_error() != SessionError::kNone) {
>-    LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR, GetSessionErrorMsg());
>+    return error;
>   }
> 
>   return RTCError::OK();
>@@ -4292,10 +4938,9 @@ RTCError PeerConnection::PushdownMediaDescription(
>       continue;
>     }
>     std::string error;
>-    bool success =
>-        (source == cricket::CS_LOCAL)
>-            ? channel->SetLocalContent(content_desc, type, &error)
>-            : channel->SetRemoteContent(content_desc, type, &error);
>+    bool success = (source == cricket::CS_LOCAL)
>+                       ? channel->SetLocalContent(content_desc, type, &error)
>+                       : channel->SetRemoteContent(content_desc, type, &error);
>     if (!success) {
>       LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, std::move(error));
>     }
>@@ -4313,8 +4958,7 @@ RTCError PeerConnection::PushdownMediaDescription(
>         bool success =
>             (source == cricket::CS_LOCAL)
>                 ? rtp_data_channel_->SetLocalContent(data_desc, type, &error)
>-                : rtp_data_channel_->SetRemoteContent(data_desc, type,
>-                                                      &error);
>+                : rtp_data_channel_->SetRemoteContent(data_desc, type, &error);
>         if (!success) {
>           LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
>                                std::move(error));
>@@ -4356,29 +5000,17 @@ RTCError PeerConnection::PushdownTransportDescription(
>     SdpType type) {
>   RTC_DCHECK_RUN_ON(signaling_thread());
> 
>-  const SessionDescriptionInterface* sdesc =
>-      (source == cricket::CS_LOCAL ? local_description()
>-                                   : remote_description());
>-  RTC_DCHECK(sdesc);
>-  for (const cricket::TransportInfo& tinfo :
>-       sdesc->description()->transport_infos()) {
>-    std::string error;
>-    bool success;
>-    if (source == cricket::CS_LOCAL) {
>-      success = transport_controller_->SetLocalTransportDescription(
>-          tinfo.content_name, tinfo.description, type, &error);
>-    } else {
>-      success = transport_controller_->SetRemoteTransportDescription(
>-          tinfo.content_name, tinfo.description, type, &error);
>-    }
>-    if (!success) {
>-      LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
>-                           "Failed to push down transport description for " +
>-                               tinfo.content_name + ": " + error);
>-    }
>+  if (source == cricket::CS_LOCAL) {
>+    const SessionDescriptionInterface* sdesc = local_description();
>+    RTC_DCHECK(sdesc);
>+    return transport_controller_->SetLocalDescription(type,
>+                                                      sdesc->description());
>+  } else {
>+    const SessionDescriptionInterface* sdesc = remote_description();
>+    RTC_DCHECK(sdesc);
>+    return transport_controller_->SetRemoteDescription(type,
>+                                                       sdesc->description());
>   }
>-
>-  return RTCError::OK();
> }
> 
> bool PeerConnection::GetTransportDescription(
>@@ -4397,76 +5029,9 @@ bool PeerConnection::GetTransportDescription(
>   return true;
> }
> 
>-bool PeerConnection::EnableBundle(const cricket::ContentGroup& bundle) {
>-  const std::string* first_content_name = bundle.FirstContentName();
>-  if (!first_content_name) {
>-    RTC_LOG(LS_WARNING) << "Tried to BUNDLE with no contents.";
>-    return false;
>-  }
>-  const std::string& transport_name = *first_content_name;
>-
>-  auto maybe_set_transport = [this, bundle,
>-                              transport_name](cricket::BaseChannel* ch) {
>-    if (!ch || !bundle.HasContentName(ch->content_name())) {
>-      return;
>-    }
>-
>-    std::string old_transport_name = ch->transport_name();
>-    if (old_transport_name == transport_name) {
>-      RTC_LOG(LS_INFO) << "BUNDLE already enabled for " << ch->content_name()
>-                       << " on " << transport_name << ".";
>-      return;
>-    }
>-
>-    cricket::DtlsTransportInternal* rtp_dtls_transport =
>-        transport_controller_->CreateDtlsTransport(
>-            transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
>-    bool need_rtcp = (ch->rtcp_dtls_transport() != nullptr);
>-    cricket::DtlsTransportInternal* rtcp_dtls_transport = nullptr;
>-    if (need_rtcp) {
>-      rtcp_dtls_transport = transport_controller_->CreateDtlsTransport(
>-          transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
>-    }
>-
>-    ch->SetTransports(rtp_dtls_transport, rtcp_dtls_transport);
>-    RTC_LOG(LS_INFO) << "Enabled BUNDLE for " << ch->content_name() << " on "
>-                     << transport_name << ".";
>-    transport_controller_->DestroyDtlsTransport(
>-        old_transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
>-    // If the channel needs rtcp, it means that the channel used to have a
>-    // rtcp transport which needs to be deleted now.
>-    if (need_rtcp) {
>-      transport_controller_->DestroyDtlsTransport(
>-          old_transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
>-    }
>-  };
>-
>-  for (auto transceiver : transceivers_) {
>-    maybe_set_transport(transceiver->internal()->channel());
>-  }
>-  maybe_set_transport(rtp_data_channel_);
>-
>-  // For SCTP, transport creation/deletion happens here instead of in the
>-  // object itself.
>-  if (sctp_transport_) {
>-    RTC_DCHECK(sctp_transport_name_);
>-    RTC_DCHECK(sctp_content_name_);
>-    if (transport_name != *sctp_transport_name_ &&
>-        bundle.HasContentName(*sctp_content_name_)) {
>-      network_thread()->Invoke<void>(
>-          RTC_FROM_HERE, rtc::Bind(&PeerConnection::ChangeSctpTransport_n, this,
>-                                   transport_name));
>-    }
>-  }
>-
>-  return true;
>-}
>-
> cricket::IceConfig PeerConnection::ParseIceConfig(
>     const PeerConnectionInterface::RTCConfiguration& config) const {
>   cricket::ContinualGatheringPolicy gathering_policy;
>-  // TODO(honghaiz): Add the third continual gathering policy in
>-  // PeerConnectionInterface and map it to GATHER_CONTINUALLY_AND_RECOVER.
>   switch (config.continual_gathering_policy) {
>     case PeerConnectionInterface::GATHER_ONCE:
>       gathering_policy = cricket::GATHER_ONCE;
>@@ -4478,18 +5043,27 @@ cricket::IceConfig PeerConnection::ParseIceConfig(
>       RTC_NOTREACHED();
>       gathering_policy = cricket::GATHER_ONCE;
>   }
>+
>   cricket::IceConfig ice_config;
>-  ice_config.receiving_timeout = config.ice_connection_receiving_timeout;
>+  ice_config.receiving_timeout = RTCConfigurationToIceConfigOptionalInt(
>+      config.ice_connection_receiving_timeout);
>   ice_config.prioritize_most_likely_candidate_pairs =
>       config.prioritize_most_likely_ice_candidate_pairs;
>   ice_config.backup_connection_ping_interval =
>-      config.ice_backup_candidate_pair_ping_interval;
>+      RTCConfigurationToIceConfigOptionalInt(
>+          config.ice_backup_candidate_pair_ping_interval);
>   ice_config.continual_gathering_policy = gathering_policy;
>   ice_config.presume_writable_when_fully_relayed =
>       config.presume_writable_when_fully_relayed;
>+  ice_config.ice_check_interval_strong_connectivity =
>+      config.ice_check_interval_strong_connectivity;
>+  ice_config.ice_check_interval_weak_connectivity =
>+      config.ice_check_interval_weak_connectivity;
>   ice_config.ice_check_min_interval = config.ice_check_min_interval;
>+  ice_config.stun_keepalive_interval = config.stun_candidate_keepalive_interval;
>   ice_config.regather_all_networks_interval_range =
>       config.ice_regather_interval_range;
>+  ice_config.network_preference = config.network_preference;
>   return ice_config;
> }
> 
>@@ -4516,7 +5090,7 @@ bool PeerConnection::SendData(const cricket::SendDataParams& params,
>                               cricket::SendDataResult* result) {
>   if (!rtp_data_channel_ && !sctp_transport_) {
>     RTC_LOG(LS_ERROR) << "SendData called when rtp_data_channel_ "
>-                      << "and sctp_transport_ are NULL.";
>+                         "and sctp_transport_ are NULL.";
>     return false;
>   }
>   return rtp_data_channel_
>@@ -4544,8 +5118,10 @@ bool PeerConnection::ConnectDataChannel(DataChannel* webrtc_data_channel) {
>                                       &DataChannel::OnChannelReady);
>     SignalSctpDataReceived.connect(webrtc_data_channel,
>                                    &DataChannel::OnDataReceived);
>-    SignalSctpStreamClosedRemotely.connect(
>-        webrtc_data_channel, &DataChannel::OnStreamClosedRemotely);
>+    SignalSctpClosingProcedureStartedRemotely.connect(
>+        webrtc_data_channel, &DataChannel::OnClosingProcedureStartedRemotely);
>+    SignalSctpClosingProcedureComplete.connect(
>+        webrtc_data_channel, &DataChannel::OnClosingProcedureComplete);
>   }
>   return true;
> }
>@@ -4563,7 +5139,8 @@ void PeerConnection::DisconnectDataChannel(DataChannel* webrtc_data_channel) {
>   } else {
>     SignalSctpReadyToSendData.disconnect(webrtc_data_channel);
>     SignalSctpDataReceived.disconnect(webrtc_data_channel);
>-    SignalSctpStreamClosedRemotely.disconnect(webrtc_data_channel);
>+    SignalSctpClosingProcedureStartedRemotely.disconnect(webrtc_data_channel);
>+    SignalSctpClosingProcedureComplete.disconnect(webrtc_data_channel);
>   }
> }
> 
>@@ -4581,7 +5158,7 @@ void PeerConnection::AddSctpDataStream(int sid) {
> void PeerConnection::RemoveSctpDataStream(int sid) {
>   if (!sctp_transport_) {
>     RTC_LOG(LS_ERROR) << "RemoveSctpDataStream called when sctp_transport_ is "
>-                      << "NULL.";
>+                         "NULL.";
>     return;
>   }
>   network_thread()->Invoke<void>(
>@@ -4594,51 +5171,85 @@ bool PeerConnection::ReadyToSendData() const {
>          sctp_ready_to_send_data_;
> }
> 
>-std::unique_ptr<SessionStats> PeerConnection::GetSessionStats_s() {
>-  RTC_DCHECK(signaling_thread()->IsCurrent());
>-  ChannelNamePairs channel_name_pairs;
>-  if (voice_channel()) {
>-    channel_name_pairs.voice = ChannelNamePair(
>-        voice_channel()->content_name(), voice_channel()->transport_name());
>+absl::optional<std::string> PeerConnection::sctp_transport_name() const {
>+  if (sctp_mid_ && transport_controller_) {
>+    auto dtls_transport = transport_controller_->GetDtlsTransport(*sctp_mid_);
>+    if (dtls_transport) {
>+      return dtls_transport->transport_name();
>+    }
>+    return absl::optional<std::string>();
>   }
>-  if (video_channel()) {
>-    channel_name_pairs.video = ChannelNamePair(
>-        video_channel()->content_name(), video_channel()->transport_name());
>+  return absl::optional<std::string>();
>+}
>+
>+cricket::CandidateStatsList PeerConnection::GetPooledCandidateStats() const {
>+  cricket::CandidateStatsList candidate_states_list;
>+  network_thread()->Invoke<void>(
>+      RTC_FROM_HERE,
>+      rtc::Bind(&cricket::PortAllocator::GetCandidateStatsFromPooledSessions,
>+                port_allocator_.get(), &candidate_states_list));
>+  return candidate_states_list;
>+}
>+
>+std::map<std::string, std::string> PeerConnection::GetTransportNamesByMid()
>+    const {
>+  std::map<std::string, std::string> transport_names_by_mid;
>+  for (auto transceiver : transceivers_) {
>+    cricket::BaseChannel* channel = transceiver->internal()->channel();
>+    if (channel) {
>+      transport_names_by_mid[channel->content_name()] =
>+          channel->transport_name();
>+    }
>   }
>-  if (rtp_data_channel()) {
>-    channel_name_pairs.data =
>-        ChannelNamePair(rtp_data_channel()->content_name(),
>-                        rtp_data_channel()->transport_name());
>+  if (rtp_data_channel_) {
>+    transport_names_by_mid[rtp_data_channel_->content_name()] =
>+        rtp_data_channel_->transport_name();
>   }
>   if (sctp_transport_) {
>-    RTC_DCHECK(sctp_content_name_);
>-    RTC_DCHECK(sctp_transport_name_);
>-    channel_name_pairs.data =
>-        ChannelNamePair(*sctp_content_name_, *sctp_transport_name_);
>+    absl::optional<std::string> transport_name = sctp_transport_name();
>+    RTC_DCHECK(transport_name);
>+    transport_names_by_mid[*sctp_mid_] = *transport_name;
>   }
>-  return GetSessionStats(channel_name_pairs);
>+  return transport_names_by_mid;
> }
> 
>-std::unique_ptr<SessionStats> PeerConnection::GetSessionStats(
>-    const ChannelNamePairs& channel_name_pairs) {
>-  if (network_thread()->IsCurrent()) {
>-    return GetSessionStats_n(channel_name_pairs);
>+std::map<std::string, cricket::TransportStats>
>+PeerConnection::GetTransportStatsByNames(
>+    const std::set<std::string>& transport_names) {
>+  if (!network_thread()->IsCurrent()) {
>+    return network_thread()
>+        ->Invoke<std::map<std::string, cricket::TransportStats>>(
>+            RTC_FROM_HERE,
>+            [&] { return GetTransportStatsByNames(transport_names); });
>   }
>-  return network_thread()->Invoke<std::unique_ptr<SessionStats>>(
>-      RTC_FROM_HERE,
>-      rtc::Bind(&PeerConnection::GetSessionStats_n, this, channel_name_pairs));
>+  std::map<std::string, cricket::TransportStats> transport_stats_by_name;
>+  for (const std::string& transport_name : transport_names) {
>+    cricket::TransportStats transport_stats;
>+    bool success =
>+        transport_controller_->GetStats(transport_name, &transport_stats);
>+    if (success) {
>+      transport_stats_by_name[transport_name] = std::move(transport_stats);
>+    } else {
>+      RTC_LOG(LS_ERROR) << "Failed to get transport stats for transport_name="
>+                        << transport_name;
>+    }
>+  }
>+  return transport_stats_by_name;
> }
> 
> bool PeerConnection::GetLocalCertificate(
>     const std::string& transport_name,
>     rtc::scoped_refptr<rtc::RTCCertificate>* certificate) {
>-  return transport_controller_->GetLocalCertificate(transport_name,
>-                                                    certificate);
>+  if (!certificate) {
>+    return false;
>+  }
>+  *certificate = transport_controller_->GetLocalCertificate(transport_name);
>+  return *certificate != nullptr;
> }
> 
>-std::unique_ptr<rtc::SSLCertificate> PeerConnection::GetRemoteSSLCertificate(
>+std::unique_ptr<rtc::SSLCertChain> PeerConnection::GetRemoteSSLCertChain(
>     const std::string& transport_name) {
>-  return transport_controller_->GetRemoteSSLCertificate(transport_name);
>+  return transport_controller_->GetRemoteSSLCertChain(transport_name);
> }
> 
> cricket::DataChannelType PeerConnection::data_channel_type() const {
>@@ -4687,12 +5298,13 @@ void PeerConnection::OnTransportControllerConnectionState(
>       break;
>     case cricket::kIceConnectionConnected:
>       RTC_LOG(LS_INFO) << "Changing to ICE connected state because "
>-                       << "all transports are writable.";
>+                          "all transports are writable.";
>       SetIceConnectionState(PeerConnectionInterface::kIceConnectionConnected);
>+      NoteUsageEvent(UsageEvent::ICE_STATE_CONNECTED);
>       break;
>     case cricket::kIceConnectionCompleted:
>       RTC_LOG(LS_INFO) << "Changing to ICE completed state because "
>-                       << "all transports are complete.";
>+                          "all transports are complete.";
>       if (ice_connection_state_ !=
>           PeerConnectionInterface::kIceConnectionConnected) {
>         // If jumping directly from "checking" to "connected",
>@@ -4700,9 +5312,8 @@ void PeerConnection::OnTransportControllerConnectionState(
>         SetIceConnectionState(PeerConnectionInterface::kIceConnectionConnected);
>       }
>       SetIceConnectionState(PeerConnectionInterface::kIceConnectionCompleted);
>-      if (metrics_observer()) {
>-        ReportTransportStats();
>-      }
>+      NoteUsageEvent(UsageEvent::ICE_STATE_CONNECTED);
>+      ReportTransportStats();
>       break;
>     default:
>       RTC_NOTREACHED();
>@@ -4740,7 +5351,7 @@ void PeerConnection::OnTransportControllerCandidatesRemoved(
>   for (const cricket::Candidate& candidate : candidates) {
>     if (candidate.transport_name().empty()) {
>       RTC_LOG(LS_ERROR) << "OnTransportControllerCandidatesRemoved: "
>-                        << "empty content name in candidate "
>+                           "empty content name in candidate "
>                         << candidate.ToString();
>       return;
>     }
>@@ -4754,11 +5365,9 @@ void PeerConnection::OnTransportControllerCandidatesRemoved(
> 
> void PeerConnection::OnTransportControllerDtlsHandshakeError(
>     rtc::SSLHandshakeError error) {
>-  if (metrics_observer()) {
>-    metrics_observer()->IncrementEnumCounter(
>-        webrtc::kEnumCounterDtlsHandshakeError, static_cast<int>(error),
>-        static_cast<int>(rtc::SSLHandshakeError::MAX_VALUE));
>-  }
>+  RTC_HISTOGRAM_ENUMERATION(
>+      "WebRTC.PeerConnection.DtlsHandshakeError", static_cast<int>(error),
>+      static_cast<int>(rtc::SSLHandshakeError::MAX_VALUE));
> }
> 
> void PeerConnection::EnableSending() {
>@@ -4810,7 +5419,7 @@ bool PeerConnection::UseCandidatesInSessionDescription(
>         if (valid) {
>           RTC_LOG(LS_INFO)
>               << "UseCandidatesInSessionDescription: Not ready to use "
>-              << "candidate.";
>+                 "candidate.";
>         }
>         continue;
>       }
>@@ -4837,9 +5446,9 @@ bool PeerConnection::UseCandidate(const IceCandidateInterface* candidate) {
>   std::vector<cricket::Candidate> candidates;
>   candidates.push_back(candidate->candidate());
>   // Invoking BaseSession method to handle remote candidates.
>-  std::string error;
>-  if (transport_controller_->AddRemoteCandidates(content.name, candidates,
>-                                                 &error)) {
>+  RTCError error =
>+      transport_controller_->AddRemoteCandidates(content.name, candidates);
>+  if (error.ok()) {
>     // Candidates successfully submitted for checking.
>     if (ice_connection_state_ == PeerConnectionInterface::kIceConnectionNew ||
>         ice_connection_state_ ==
>@@ -4855,10 +5464,8 @@ bool PeerConnection::UseCandidate(const IceCandidateInterface* candidate) {
>       SetIceConnectionState(PeerConnectionInterface::kIceConnectionChecking);
>     }
>     // TODO(bemasc): If state is Completed, go back to Connected.
>-  } else {
>-    if (!error.empty()) {
>-      RTC_LOG(LS_WARNING) << error;
>-    }
>+  } else if (error.message()) {
>+    RTC_LOG(LS_WARNING) << error.message();
>   }
>   return true;
> }
>@@ -4882,25 +5489,6 @@ void PeerConnection::RemoveUnusedChannels(const SessionDescription* desc) {
>   }
> }
> 
>-std::string PeerConnection::GetTransportNameForMediaSection(
>-    const std::string& mid,
>-    const cricket::ContentGroup* bundle_group) const {
>-  if (!bundle_group) {
>-    return mid;
>-  }
>-  const std::string* first_content_name = bundle_group->FirstContentName();
>-  if (!first_content_name) {
>-    RTC_LOG(LS_WARNING) << "Tried to BUNDLE with no contents.";
>-    return mid;
>-  }
>-  if (!bundle_group->HasContentName(mid)) {
>-    RTC_LOG(LS_WARNING) << mid << " is not part of any bundle group";
>-    return mid;
>-  }
>-  RTC_LOG(LS_INFO) << "Bundling " << mid << " on " << *first_content_name;
>-  return *first_content_name;
>-}
>-
> RTCErrorOr<const cricket::ContentGroup*> PeerConnection::GetEarlyBundleGroup(
>     const SessionDescription& desc) const {
>   const cricket::ContentGroup* bundle_group = nullptr;
>@@ -4917,19 +5505,12 @@ RTCErrorOr<const cricket::ContentGroup*> PeerConnection::GetEarlyBundleGroup(
> }
> 
> RTCError PeerConnection::CreateChannels(const SessionDescription& desc) {
>-  auto bundle_group_or_error = GetEarlyBundleGroup(desc);
>-  if (!bundle_group_or_error.ok()) {
>-    return bundle_group_or_error.MoveError();
>-  }
>-  const cricket::ContentGroup* bundle_group = bundle_group_or_error.MoveValue();
>-
>-  // Creating the media channels and transport proxies.
>+  // Creating the media channels. Transports should already have been created
>+  // at this point.
>   const cricket::ContentInfo* voice = cricket::GetFirstAudioContent(&desc);
>   if (voice && !voice->rejected &&
>       !GetAudioTransceiver()->internal()->channel()) {
>-    cricket::VoiceChannel* voice_channel = CreateVoiceChannel(
>-        voice->name,
>-        GetTransportNameForMediaSection(voice->name, bundle_group));
>+    cricket::VoiceChannel* voice_channel = CreateVoiceChannel(voice->name);
>     if (!voice_channel) {
>       LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
>                            "Failed to create voice channel.");
>@@ -4940,9 +5521,7 @@ RTCError PeerConnection::CreateChannels(const SessionDescription& desc) {
>   const cricket::ContentInfo* video = cricket::GetFirstVideoContent(&desc);
>   if (video && !video->rejected &&
>       !GetVideoTransceiver()->internal()->channel()) {
>-    cricket::VideoChannel* video_channel = CreateVideoChannel(
>-        video->name,
>-        GetTransportNameForMediaSection(video->name, bundle_group));
>+    cricket::VideoChannel* video_channel = CreateVideoChannel(video->name);
>     if (!video_channel) {
>       LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
>                            "Failed to create video channel.");
>@@ -4953,8 +5532,7 @@ RTCError PeerConnection::CreateChannels(const SessionDescription& desc) {
>   const cricket::ContentInfo* data = cricket::GetFirstDataContent(&desc);
>   if (data_channel_type_ != cricket::DCT_NONE && data && !data->rejected &&
>       !rtp_data_channel_ && !sctp_transport_) {
>-    if (!CreateDataChannel(data->name, GetTransportNameForMediaSection(
>-                                           data->name, bundle_group))) {
>+    if (!CreateDataChannel(data->name)) {
>       LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
>                            "Failed to create data channel.");
>     }
>@@ -4965,81 +5543,49 @@ RTCError PeerConnection::CreateChannels(const SessionDescription& desc) {
> 
> // TODO(steveanton): Perhaps this should be managed by the RtpTransceiver.
> cricket::VoiceChannel* PeerConnection::CreateVoiceChannel(
>-    const std::string& mid,
>-    const std::string& transport_name) {
>-  cricket::DtlsTransportInternal* rtp_dtls_transport =
>-      transport_controller_->CreateDtlsTransport(
>-          transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
>-  cricket::DtlsTransportInternal* rtcp_dtls_transport = nullptr;
>-  if (configuration_.rtcp_mux_policy !=
>-      PeerConnectionInterface::kRtcpMuxPolicyRequire) {
>-    rtcp_dtls_transport = transport_controller_->CreateDtlsTransport(
>-        transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
>-  }
>-
>+    const std::string& mid) {
>+  RtpTransportInternal* rtp_transport =
>+      transport_controller_->GetRtpTransport(mid);
>+  RTC_DCHECK(rtp_transport);
>   cricket::VoiceChannel* voice_channel = channel_manager()->CreateVoiceChannel(
>-      call_.get(), configuration_.media_config, rtp_dtls_transport,
>-      rtcp_dtls_transport, signaling_thread(), mid, SrtpRequired(),
>-      audio_options_);
>+      call_.get(), configuration_.media_config, rtp_transport,
>+      signaling_thread(), mid, SrtpRequired(),
>+      factory_->options().crypto_options, audio_options_);
>   if (!voice_channel) {
>-    transport_controller_->DestroyDtlsTransport(
>-        transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
>-    if (rtcp_dtls_transport) {
>-      transport_controller_->DestroyDtlsTransport(
>-          transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
>-    }
>     return nullptr;
>   }
>-  voice_channel->SignalRtcpMuxFullyActive.connect(
>-      this, &PeerConnection::DestroyRtcpTransport_n);
>   voice_channel->SignalDtlsSrtpSetupFailure.connect(
>       this, &PeerConnection::OnDtlsSrtpSetupFailure);
>   voice_channel->SignalSentPacket.connect(this,
>                                           &PeerConnection::OnSentPacket_w);
>+  voice_channel->SetRtpTransport(rtp_transport);
> 
>   return voice_channel;
> }
> 
> // TODO(steveanton): Perhaps this should be managed by the RtpTransceiver.
> cricket::VideoChannel* PeerConnection::CreateVideoChannel(
>-    const std::string& mid,
>-    const std::string& transport_name) {
>-  cricket::DtlsTransportInternal* rtp_dtls_transport =
>-      transport_controller_->CreateDtlsTransport(
>-          transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
>-  cricket::DtlsTransportInternal* rtcp_dtls_transport = nullptr;
>-  if (configuration_.rtcp_mux_policy !=
>-      PeerConnectionInterface::kRtcpMuxPolicyRequire) {
>-    rtcp_dtls_transport = transport_controller_->CreateDtlsTransport(
>-        transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
>-  }
>-
>+    const std::string& mid) {
>+  RtpTransportInternal* rtp_transport =
>+      transport_controller_->GetRtpTransport(mid);
>+  RTC_DCHECK(rtp_transport);
>   cricket::VideoChannel* video_channel = channel_manager()->CreateVideoChannel(
>-      call_.get(), configuration_.media_config, rtp_dtls_transport,
>-      rtcp_dtls_transport, signaling_thread(), mid, SrtpRequired(),
>-      video_options_);
>-
>+      call_.get(), configuration_.media_config, rtp_transport,
>+      signaling_thread(), mid, SrtpRequired(),
>+      factory_->options().crypto_options, video_options_);
>   if (!video_channel) {
>-    transport_controller_->DestroyDtlsTransport(
>-        transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
>-    if (rtcp_dtls_transport) {
>-      transport_controller_->DestroyDtlsTransport(
>-          transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
>-    }
>     return nullptr;
>   }
>-  video_channel->SignalRtcpMuxFullyActive.connect(
>-      this, &PeerConnection::DestroyRtcpTransport_n);
>   video_channel->SignalDtlsSrtpSetupFailure.connect(
>       this, &PeerConnection::OnDtlsSrtpSetupFailure);
>   video_channel->SignalSentPacket.connect(this,
>                                           &PeerConnection::OnSentPacket_w);
>+  video_channel->SetRtpTransport(rtp_transport);
> 
>   return video_channel;
> }
> 
>-bool PeerConnection::CreateDataChannel(const std::string& mid,
>-                                       const std::string& transport_name) {
>+bool PeerConnection::CreateDataChannel(const std::string& mid) {
>   bool sctp = (data_channel_type_ == cricket::DCT_SCTP);
>   if (sctp) {
>     if (!sctp_factory_) {
>@@ -5049,44 +5595,28 @@ bool PeerConnection::CreateDataChannel(const std::string& mid,
>       return false;
>     }
>     if (!network_thread()->Invoke<bool>(
>-            RTC_FROM_HERE, rtc::Bind(&PeerConnection::CreateSctpTransport_n,
>-                                     this, mid, transport_name))) {
>+            RTC_FROM_HERE,
>+            rtc::Bind(&PeerConnection::CreateSctpTransport_n, this, mid))) {
>       return false;
>     }
>     for (const auto& channel : sctp_data_channels_) {
>       channel->OnTransportChannelCreated();
>     }
>   } else {
>-    cricket::DtlsTransportInternal* rtp_dtls_transport =
>-        transport_controller_->CreateDtlsTransport(
>-            transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
>-    cricket::DtlsTransportInternal* rtcp_dtls_transport = nullptr;
>-    if (configuration_.rtcp_mux_policy !=
>-        PeerConnectionInterface::kRtcpMuxPolicyRequire) {
>-      rtcp_dtls_transport = transport_controller_->CreateDtlsTransport(
>-          transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
>-    }
>-
>+    RtpTransportInternal* rtp_transport =
>+        transport_controller_->GetRtpTransport(mid);
>+    RTC_DCHECK(rtp_transport);
>     rtp_data_channel_ = channel_manager()->CreateRtpDataChannel(
>-        configuration_.media_config, rtp_dtls_transport, rtcp_dtls_transport,
>-        signaling_thread(), mid, SrtpRequired());
>-
>+        configuration_.media_config, rtp_transport, signaling_thread(), mid,
>+        SrtpRequired(), factory_->options().crypto_options);
>     if (!rtp_data_channel_) {
>-      transport_controller_->DestroyDtlsTransport(
>-          transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
>-      if (rtcp_dtls_transport) {
>-        transport_controller_->DestroyDtlsTransport(
>-            transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
>-      }
>       return false;
>     }
>-
>-    rtp_data_channel_->SignalRtcpMuxFullyActive.connect(
>-        this, &PeerConnection::DestroyRtcpTransport_n);
>     rtp_data_channel_->SignalDtlsSrtpSetupFailure.connect(
>         this, &PeerConnection::OnDtlsSrtpSetupFailure);
>     rtp_data_channel_->SignalSentPacket.connect(
>         this, &PeerConnection::OnSentPacket_w);
>+    rtp_data_channel_->SetRtpTransport(rtp_transport);
>   }
> 
>   return true;
>@@ -5104,68 +5634,36 @@ Call::Stats PeerConnection::GetCallStats() {
>   }
> }
> 
>-std::unique_ptr<SessionStats> PeerConnection::GetSessionStats_n(
>-    const ChannelNamePairs& channel_name_pairs) {
>-  RTC_DCHECK(network_thread()->IsCurrent());
>-  std::unique_ptr<SessionStats> session_stats(new SessionStats());
>-  for (const auto channel_name_pair :
>-       {&channel_name_pairs.voice, &channel_name_pairs.video,
>-        &channel_name_pairs.data}) {
>-    if (*channel_name_pair) {
>-      cricket::TransportStats transport_stats;
>-      if (!transport_controller_->GetStats((*channel_name_pair)->transport_name,
>-                                           &transport_stats)) {
>-        return nullptr;
>-      }
>-      session_stats->transport_stats[(*channel_name_pair)->transport_name] =
>-          std::move(transport_stats);
>-    }
>-  }
>-  return session_stats;
>-}
>-
>-bool PeerConnection::CreateSctpTransport_n(const std::string& content_name,
>-                                           const std::string& transport_name) {
>+bool PeerConnection::CreateSctpTransport_n(const std::string& mid) {
>   RTC_DCHECK(network_thread()->IsCurrent());
>   RTC_DCHECK(sctp_factory_);
>-  cricket::DtlsTransportInternal* tc =
>-      transport_controller_->CreateDtlsTransport_n(
>-          transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
>-  sctp_transport_ = sctp_factory_->CreateSctpTransport(tc);
>+  cricket::DtlsTransportInternal* dtls_transport =
>+      transport_controller_->GetDtlsTransport(mid);
>+  RTC_DCHECK(dtls_transport);
>+  sctp_transport_ = sctp_factory_->CreateSctpTransport(dtls_transport);
>   RTC_DCHECK(sctp_transport_);
>   sctp_invoker_.reset(new rtc::AsyncInvoker());
>   sctp_transport_->SignalReadyToSendData.connect(
>       this, &PeerConnection::OnSctpTransportReadyToSendData_n);
>   sctp_transport_->SignalDataReceived.connect(
>       this, &PeerConnection::OnSctpTransportDataReceived_n);
>-  sctp_transport_->SignalStreamClosedRemotely.connect(
>-      this, &PeerConnection::OnSctpStreamClosedRemotely_n);
>-  sctp_transport_name_ = transport_name;
>-  sctp_content_name_ = content_name;
>+  // TODO(deadbeef): All we do here is AsyncInvoke to fire the signal on
>+  // another thread. Would be nice if there was a helper class similar to
>+  // sigslot::repeater that did this for us, eliminating a bunch of boilerplate
>+  // code.
>+  sctp_transport_->SignalClosingProcedureStartedRemotely.connect(
>+      this, &PeerConnection::OnSctpClosingProcedureStartedRemotely_n);
>+  sctp_transport_->SignalClosingProcedureComplete.connect(
>+      this, &PeerConnection::OnSctpClosingProcedureComplete_n);
>+  sctp_mid_ = mid;
>+  sctp_transport_->SetDtlsTransport(dtls_transport);
>   return true;
> }
> 
>-void PeerConnection::ChangeSctpTransport_n(const std::string& transport_name) {
>-  RTC_DCHECK(network_thread()->IsCurrent());
>-  RTC_DCHECK(sctp_transport_);
>-  RTC_DCHECK(sctp_transport_name_);
>-  std::string old_sctp_transport_name = *sctp_transport_name_;
>-  sctp_transport_name_ = transport_name;
>-  cricket::DtlsTransportInternal* tc =
>-      transport_controller_->CreateDtlsTransport_n(
>-          transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
>-  sctp_transport_->SetTransportChannel(tc);
>-  transport_controller_->DestroyDtlsTransport_n(
>-      old_sctp_transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
>-}
>-
> void PeerConnection::DestroySctpTransport_n() {
>   RTC_DCHECK(network_thread()->IsCurrent());
>   sctp_transport_.reset(nullptr);
>-  transport_controller_->DestroyDtlsTransport_n(
>-      *sctp_transport_name_, cricket::ICE_CANDIDATE_COMPONENT_RTP);
>-  sctp_content_name_.reset();
>-  sctp_transport_name_.reset();
>+  sctp_mid_.reset();
>   sctp_invoker_.reset(nullptr);
>   sctp_ready_to_send_data_ = false;
> }
>@@ -5228,13 +5726,22 @@ void PeerConnection::OnSctpTransportDataReceived_s(
>   }
> }
> 
>-void PeerConnection::OnSctpStreamClosedRemotely_n(int sid) {
>+void PeerConnection::OnSctpClosingProcedureStartedRemotely_n(int sid) {
>+  RTC_DCHECK(data_channel_type_ == cricket::DCT_SCTP);
>+  RTC_DCHECK(network_thread()->IsCurrent());
>+  sctp_invoker_->AsyncInvoke<void>(
>+      RTC_FROM_HERE, signaling_thread(),
>+      rtc::Bind(&sigslot::signal1<int>::operator(),
>+                &SignalSctpClosingProcedureStartedRemotely, sid));
>+}
>+
>+void PeerConnection::OnSctpClosingProcedureComplete_n(int sid) {
>   RTC_DCHECK(data_channel_type_ == cricket::DCT_SCTP);
>   RTC_DCHECK(network_thread()->IsCurrent());
>   sctp_invoker_->AsyncInvoke<void>(
>       RTC_FROM_HERE, signaling_thread(),
>       rtc::Bind(&sigslot::signal1<int>::operator(),
>-                &SignalSctpStreamClosedRemotely, sid));
>+                &SignalSctpClosingProcedureComplete, sid));
> }
> 
> // Returns false if bundle is enabled and rtcp_mux is disabled.
>@@ -5281,7 +5788,7 @@ RTCError PeerConnection::ValidateSessionDescription(
>   if ((source == cricket::CS_LOCAL && !ExpectSetLocalDescription(type)) ||
>       (source == cricket::CS_REMOTE && !ExpectSetRemoteDescription(type))) {
>     LOG_AND_RETURN_ERROR(
>-        RTCErrorType::INVALID_PARAMETER,
>+        RTCErrorType::INVALID_STATE,
>         "Called in wrong state: " + GetSignalingStateString(signaling_state()));
>   }
> 
>@@ -5311,50 +5818,58 @@ RTCError PeerConnection::ValidateSessionDescription(
> 
>   // Verify m-lines in Answer when compared against Offer.
>   if (type == SdpType::kPrAnswer || type == SdpType::kAnswer) {
>+    // With an answer we want to compare the new answer session description with
>+    // the offer's session description from the current negotiation.
>     const cricket::SessionDescription* offer_desc =
>         (source == cricket::CS_LOCAL) ? remote_description()->description()
>                                       : local_description()->description();
>-    if (!MediaSectionsHaveSameCount(offer_desc, sdesc->description()) ||
>-        !MediaSectionsInSameOrder(offer_desc, sdesc->description())) {
>+    if (!MediaSectionsHaveSameCount(*offer_desc, *sdesc->description()) ||
>+        !MediaSectionsInSameOrder(*offer_desc, nullptr, *sdesc->description(),
>+                                  type)) {
>       LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
>                            kMlineMismatchInAnswer);
>     }
>   } else {
>+    // The re-offers should respect the order of m= sections in current
>+    // description. See RFC3264 Section 8 paragraph 4 for more details.
>+    // With a re-offer, either the current local or current remote descriptions
>+    // could be the most up to date, so we would like to check against both of
>+    // them if they exist. It could be the case that one of them has a 0 port
>+    // for a media section, but the other does not. This is important to check
>+    // against in the case that we are recycling an m= section.
>     const cricket::SessionDescription* current_desc = nullptr;
>-    if (source == cricket::CS_LOCAL && local_description()) {
>+    const cricket::SessionDescription* secondary_current_desc = nullptr;
>+    if (local_description()) {
>       current_desc = local_description()->description();
>-    } else if (source == cricket::CS_REMOTE && remote_description()) {
>+      if (remote_description()) {
>+        secondary_current_desc = remote_description()->description();
>+      }
>+    } else if (remote_description()) {
>       current_desc = remote_description()->description();
>     }
>-    // The re-offers should respect the order of m= sections in current
>-    // description. See RFC3264 Section 8 paragraph 4 for more details.
>     if (current_desc &&
>-        !MediaSectionsInSameOrder(current_desc, sdesc->description())) {
>+        !MediaSectionsInSameOrder(*current_desc, secondary_current_desc,
>+                                  *sdesc->description(), type)) {
>       LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
>                            kMlineMismatchInSubsequentOffer);
>     }
>   }
> 
>-  // Unified Plan SDP should have exactly one stream per m= section for audio
>-  // and video.
>   if (IsUnifiedPlan()) {
>+    // Ensure that each audio and video media section has at most one
>+    // "StreamParams". This will return an error if receiving a session
>+    // description from a "Plan B" endpoint which adds multiple tracks of the
>+    // same type. With Unified Plan, there can only be at most one track per
>+    // media section.
>     for (const ContentInfo& content : sdesc->description()->contents()) {
>-      if (content.rejected) {
>-        continue;
>-      }
>-      if (content.media_description()) {
>-        cricket::MediaType media_type = content.media_description()->type();
>-        if (!(media_type == cricket::MEDIA_TYPE_AUDIO ||
>-              media_type == cricket::MEDIA_TYPE_VIDEO)) {
>-          continue;
>-        }
>-        const cricket::StreamParamsVec& streams =
>-            content.media_description()->streams();
>-        if (streams.size() != 1u) {
>-          LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
>-                               "Unified Plan SDP should have exactly one "
>-                               "track per media section for audio and video.");
>-        }
>+      const MediaContentDescription& desc = *content.description;
>+      if ((desc.type() == cricket::MEDIA_TYPE_AUDIO ||
>+           desc.type() == cricket::MEDIA_TYPE_VIDEO) &&
>+          desc.streams().size() > 1u) {
>+        LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
>+                             "Media section has more than one track specified "
>+                             "with a=ssrc lines which is not supported with "
>+                             "Unified Plan.");
>       }
>     }
>   }
>@@ -5406,6 +5921,91 @@ std::string PeerConnection::GetSessionErrorMsg() {
>   return desc.str();
> }
> 
>+void PeerConnection::ReportSdpFormatReceived(
>+    const SessionDescriptionInterface& remote_offer) {
>+  int num_audio_mlines = 0;
>+  int num_video_mlines = 0;
>+  int num_audio_tracks = 0;
>+  int num_video_tracks = 0;
>+  for (const ContentInfo& content : remote_offer.description()->contents()) {
>+    cricket::MediaType media_type = content.media_description()->type();
>+    int num_tracks = std::max(
>+        1, static_cast<int>(content.media_description()->streams().size()));
>+    if (media_type == cricket::MEDIA_TYPE_AUDIO) {
>+      num_audio_mlines += 1;
>+      num_audio_tracks += num_tracks;
>+    } else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
>+      num_video_mlines += 1;
>+      num_video_tracks += num_tracks;
>+    }
>+  }
>+  SdpFormatReceived format = kSdpFormatReceivedNoTracks;
>+  if (num_audio_mlines > 1 || num_video_mlines > 1) {
>+    format = kSdpFormatReceivedComplexUnifiedPlan;
>+  } else if (num_audio_tracks > 1 || num_video_tracks > 1) {
>+    format = kSdpFormatReceivedComplexPlanB;
>+  } else if (num_audio_tracks > 0 || num_video_tracks > 0) {
>+    format = kSdpFormatReceivedSimple;
>+  }
>+  RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.SdpFormatReceived", format,
>+                            kSdpFormatReceivedMax);
>+}
>+
>+void PeerConnection::NoteUsageEvent(UsageEvent event) {
>+  RTC_DCHECK_RUN_ON(signaling_thread());
>+  usage_event_accumulator_ |= static_cast<int>(event);
>+}
>+
>+void PeerConnection::ReportUsagePattern() const {
>+  RTC_DLOG(LS_INFO) << "Usage signature is " << usage_event_accumulator_;
>+  RTC_HISTOGRAM_ENUMERATION_SPARSE("WebRTC.PeerConnection.UsagePattern",
>+                                   usage_event_accumulator_,
>+                                   static_cast<int>(UsageEvent::MAX_VALUE));
>+  const int bad_bits =
>+      static_cast<int>(UsageEvent::SET_LOCAL_DESCRIPTION_CALLED) |
>+      static_cast<int>(UsageEvent::CANDIDATE_COLLECTED);
>+  const int good_bits =
>+      static_cast<int>(UsageEvent::SET_REMOTE_DESCRIPTION_CALLED) |
>+      static_cast<int>(UsageEvent::REMOTE_CANDIDATE_ADDED) |
>+      static_cast<int>(UsageEvent::ICE_STATE_CONNECTED);
>+  if ((usage_event_accumulator_ & bad_bits) == bad_bits &&
>+      (usage_event_accumulator_ & good_bits) == 0) {
>+    // If called after close(), we can't report, because observer may have
>+    // been deallocated, and therefore pointer is null. Write to log instead.
>+    if (observer_) {
>+      Observer()->OnInterestingUsage(usage_event_accumulator_);
>+    } else {
>+      RTC_LOG(LS_INFO) << "Interesting usage signature "
>+                       << usage_event_accumulator_
>+                       << " observed after observer shutdown";
>+    }
>+  }
>+}
>+
>+void PeerConnection::ReportNegotiatedSdpSemantics(
>+    const SessionDescriptionInterface& answer) {
>+  SdpSemanticNegotiated semantics_negotiated;
>+  switch (answer.description()->msid_signaling()) {
>+    case 0:
>+      semantics_negotiated = kSdpSemanticNegotiatedNone;
>+      break;
>+    case cricket::kMsidSignalingMediaSection:
>+      semantics_negotiated = kSdpSemanticNegotiatedUnifiedPlan;
>+      break;
>+    case cricket::kMsidSignalingSsrcAttribute:
>+      semantics_negotiated = kSdpSemanticNegotiatedPlanB;
>+      break;
>+    case cricket::kMsidSignalingMediaSection |
>+        cricket::kMsidSignalingSsrcAttribute:
>+      semantics_negotiated = kSdpSemanticNegotiatedMixed;
>+      break;
>+    default:
>+      RTC_NOTREACHED();
>+  }
>+  RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.SdpSemanticNegotiated",
>+                            semantics_negotiated, kSdpSemanticNegotiatedMax);
>+}
>+
> // We need to check the local/remote description for the Transport instead of
> // the session, because a new Transport added during renegotiation may have
> // them unset while the session has them set from the previous negotiation.
>@@ -5443,7 +6043,7 @@ bool PeerConnection::ReadyToUseRemoteCandidate(
>   if (transport_name.empty()) {
>     return false;
>   }
>-  return transport_controller_->ReadyForRemoteCandidates(transport_name);
>+  return true;
> }
> 
> bool PeerConnection::SrtpRequired() const {
>@@ -5462,26 +6062,34 @@ void PeerConnection::OnTransportControllerGatheringState(
> }
> 
> void PeerConnection::ReportTransportStats() {
>-  // Use a set so we don't report the same stats twice if two channels share
>-  // a transport.
>-  std::set<std::string> transport_names;
>-  if (voice_channel()) {
>-    transport_names.insert(voice_channel()->transport_name());
>-  }
>-  if (video_channel()) {
>-    transport_names.insert(video_channel()->transport_name());
>+  std::map<std::string, std::set<cricket::MediaType>>
>+      media_types_by_transport_name;
>+  for (auto transceiver : transceivers_) {
>+    if (transceiver->internal()->channel()) {
>+      const std::string& transport_name =
>+          transceiver->internal()->channel()->transport_name();
>+      media_types_by_transport_name[transport_name].insert(
>+          transceiver->media_type());
>+    }
>   }
>   if (rtp_data_channel()) {
>-    transport_names.insert(rtp_data_channel()->transport_name());
>+    media_types_by_transport_name[rtp_data_channel()->transport_name()].insert(
>+        cricket::MEDIA_TYPE_DATA);
>   }
>-  if (sctp_transport_name_) {
>-    transport_names.insert(*sctp_transport_name_);
>+
>+  absl::optional<std::string> transport_name = sctp_transport_name();
>+  if (transport_name) {
>+    media_types_by_transport_name[*transport_name].insert(
>+        cricket::MEDIA_TYPE_DATA);
>   }
>-  for (const auto& name : transport_names) {
>+
>+  for (const auto& entry : media_types_by_transport_name) {
>+    const std::string& transport_name = entry.first;
>+    const std::set<cricket::MediaType> media_types = entry.second;
>     cricket::TransportStats stats;
>-    if (transport_controller_->GetStats(name, &stats)) {
>+    if (transport_controller_->GetStats(transport_name, &stats)) {
>       ReportBestConnectionState(stats);
>-      ReportNegotiatedCiphers(stats);
>+      ReportNegotiatedCiphers(stats, media_types);
>     }
>   }
> }
>@@ -5489,45 +6097,41 @@ void PeerConnection::ReportTransportStats() {
> // for IPv4 and IPv6.
> void PeerConnection::ReportBestConnectionState(
>     const cricket::TransportStats& stats) {
>-  RTC_DCHECK(metrics_observer());
>-  for (cricket::TransportChannelStatsList::const_iterator it =
>-           stats.channel_stats.begin();
>-       it != stats.channel_stats.end(); ++it) {
>-    for (cricket::ConnectionInfos::const_iterator it_info =
>-             it->connection_infos.begin();
>-         it_info != it->connection_infos.end(); ++it_info) {
>-      if (!it_info->best_connection) {
>+  for (const cricket::TransportChannelStats& channel_stats :
>+       stats.channel_stats) {
>+    for (const cricket::ConnectionInfo& connection_info :
>+         channel_stats.connection_infos) {
>+      if (!connection_info.best_connection) {
>         continue;
>       }
> 
>-      PeerConnectionEnumCounterType type = kPeerConnectionEnumCounterMax;
>-      const cricket::Candidate& local = it_info->local_candidate;
>-      const cricket::Candidate& remote = it_info->remote_candidate;
>+      const cricket::Candidate& local = connection_info.local_candidate;
>+      const cricket::Candidate& remote = connection_info.remote_candidate;
> 
>       // Increment the counter for IceCandidatePairType.
>       if (local.protocol() == cricket::TCP_PROTOCOL_NAME ||
>           (local.type() == RELAY_PORT_TYPE &&
>            local.relay_protocol() == cricket::TCP_PROTOCOL_NAME)) {
>-        type = kEnumCounterIceCandidatePairTypeTcp;
>+        RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.CandidatePairType_TCP",
>+                                  GetIceCandidatePairCounter(local, remote),
>+                                  kIceCandidatePairMax);
>       } else if (local.protocol() == cricket::UDP_PROTOCOL_NAME) {
>-        type = kEnumCounterIceCandidatePairTypeUdp;
>+        RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.CandidatePairType_UDP",
>+                                  GetIceCandidatePairCounter(local, remote),
>+                                  kIceCandidatePairMax);
>       } else {
>         RTC_CHECK(0);
>       }
>-      metrics_observer()->IncrementEnumCounter(
>-          type, GetIceCandidatePairCounter(local, remote),
>-          kIceCandidatePairMax);
> 
>       // Increment the counter for IP type.
>       if (local.address().family() == AF_INET) {
>-        metrics_observer()->IncrementEnumCounter(
>-            kEnumCounterAddressFamily, kBestConnections_IPv4,
>-            kPeerConnectionAddressFamilyCounter_Max);
>-
>+        RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.IPMetrics",
>+                                  kBestConnections_IPv4,
>+                                  kPeerConnectionAddressFamilyCounter_Max);
>       } else if (local.address().family() == AF_INET6) {
>-        metrics_observer()->IncrementEnumCounter(
>-            kEnumCounterAddressFamily, kBestConnections_IPv6,
>-            kPeerConnectionAddressFamilyCounter_Max);
>+        RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.IPMetrics",
>+                                  kBestConnections_IPv6,
>+                                  kPeerConnectionAddressFamilyCounter_Max);
>       } else {
>         RTC_CHECK(0);
>       }
>@@ -5538,8 +6142,8 @@ void PeerConnection::ReportBestConnectionState(
> }
> 
> void PeerConnection::ReportNegotiatedCiphers(
>-    const cricket::TransportStats& stats) {
>-  RTC_DCHECK(metrics_observer());
>+    const cricket::TransportStats& stats,
>+    const std::set<cricket::MediaType>& media_types) {
>   if (!dtls_enabled_ || stats.channel_stats.empty()) {
>     return;
>   }
>@@ -5551,29 +6155,54 @@ void PeerConnection::ReportNegotiatedCiphers(
>     return;
>   }
> 
>-  PeerConnectionEnumCounterType srtp_counter_type;
>-  PeerConnectionEnumCounterType ssl_counter_type;
>-  if (stats.transport_name == cricket::CN_AUDIO) {
>-    srtp_counter_type = kEnumCounterAudioSrtpCipher;
>-    ssl_counter_type = kEnumCounterAudioSslCipher;
>-  } else if (stats.transport_name == cricket::CN_VIDEO) {
>-    srtp_counter_type = kEnumCounterVideoSrtpCipher;
>-    ssl_counter_type = kEnumCounterVideoSslCipher;
>-  } else if (stats.transport_name == cricket::CN_DATA) {
>-    srtp_counter_type = kEnumCounterDataSrtpCipher;
>-    ssl_counter_type = kEnumCounterDataSslCipher;
>-  } else {
>-    RTC_NOTREACHED();
>-    return;
>-  }
>-
>   if (srtp_crypto_suite != rtc::SRTP_INVALID_CRYPTO_SUITE) {
>-    metrics_observer()->IncrementSparseEnumCounter(srtp_counter_type,
>-                                                   srtp_crypto_suite);
>+    for (cricket::MediaType media_type : media_types) {
>+      switch (media_type) {
>+        case cricket::MEDIA_TYPE_AUDIO:
>+          RTC_HISTOGRAM_ENUMERATION_SPARSE(
>+              "WebRTC.PeerConnection.SrtpCryptoSuite.Audio", srtp_crypto_suite,
>+              rtc::SRTP_CRYPTO_SUITE_MAX_VALUE);
>+          break;
>+        case cricket::MEDIA_TYPE_VIDEO:
>+          RTC_HISTOGRAM_ENUMERATION_SPARSE(
>+              "WebRTC.PeerConnection.SrtpCryptoSuite.Video", srtp_crypto_suite,
>+              rtc::SRTP_CRYPTO_SUITE_MAX_VALUE);
>+          break;
>+        case cricket::MEDIA_TYPE_DATA:
>+          RTC_HISTOGRAM_ENUMERATION_SPARSE(
>+              "WebRTC.PeerConnection.SrtpCryptoSuite.Data", srtp_crypto_suite,
>+              rtc::SRTP_CRYPTO_SUITE_MAX_VALUE);
>+          break;
>+        default:
>+          RTC_NOTREACHED();
>+          continue;
>+      }
>+    }
>   }
>+
>   if (ssl_cipher_suite != rtc::TLS_NULL_WITH_NULL_NULL) {
>-    metrics_observer()->IncrementSparseEnumCounter(ssl_counter_type,
>-                                                   ssl_cipher_suite);
>+    for (cricket::MediaType media_type : media_types) {
>+      switch (media_type) {
>+        case cricket::MEDIA_TYPE_AUDIO:
>+          RTC_HISTOGRAM_ENUMERATION_SPARSE(
>+              "WebRTC.PeerConnection.SslCipherSuite.Audio", ssl_cipher_suite,
>+              rtc::SSL_CIPHER_SUITE_MAX_VALUE);
>+          break;
>+        case cricket::MEDIA_TYPE_VIDEO:
>+          RTC_HISTOGRAM_ENUMERATION_SPARSE(
>+              "WebRTC.PeerConnection.SslCipherSuite.Video", ssl_cipher_suite,
>+              rtc::SSL_CIPHER_SUITE_MAX_VALUE);
>+          break;
>+        case cricket::MEDIA_TYPE_DATA:
>+          RTC_HISTOGRAM_ENUMERATION_SPARSE(
>+              "WebRTC.PeerConnection.SslCipherSuite.Data", ssl_cipher_suite,
>+              rtc::SSL_CIPHER_SUITE_MAX_VALUE);
>+          break;
>+        default:
>+          RTC_NOTREACHED();
>+          continue;
>+      }
>+    }
>   }
> }
> 
>@@ -5590,22 +6219,15 @@ const std::string PeerConnection::GetTransportName(
>     return channel->transport_name();
>   }
>   if (sctp_transport_) {
>-    RTC_DCHECK(sctp_content_name_);
>-    RTC_DCHECK(sctp_transport_name_);
>-    if (content_name == *sctp_content_name_) {
>-      return *sctp_transport_name_;
>+    RTC_DCHECK(sctp_mid_);
>+    if (content_name == *sctp_mid_) {
>+      return *sctp_transport_name();
>     }
>   }
>   // Return an empty string if failed to retrieve the transport name.
>   return "";
> }
> 
>-void PeerConnection::DestroyRtcpTransport_n(const std::string& transport_name) {
>-  RTC_DCHECK(network_thread()->IsCurrent());
>-  transport_controller_->DestroyDtlsTransport_n(
>-      transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
>-}
>-
> void PeerConnection::DestroyTransceiverChannel(
>     rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
>         transceiver) {
>@@ -5640,14 +6262,6 @@ void PeerConnection::DestroyDataChannel() {
> 
> void PeerConnection::DestroyBaseChannel(cricket::BaseChannel* channel) {
>   RTC_DCHECK(channel);
>-  RTC_DCHECK(channel->rtp_dtls_transport());
>-
>-  // Need to cache these before destroying the base channel so that we do not
>-  // access uninitialized memory.
>-  const std::string transport_name =
>-      channel->rtp_dtls_transport()->transport_name();
>-  const bool need_to_delete_rtcp = (channel->rtcp_dtls_transport() != nullptr);
>-
>   switch (channel->media_type()) {
>     case cricket::MEDIA_TYPE_AUDIO:
>       channel_manager()->DestroyVoiceChannel(
>@@ -5665,15 +6279,31 @@ void PeerConnection::DestroyBaseChannel(cricket::BaseChannel* channel) {
>       RTC_NOTREACHED() << "Unknown media type: " << channel->media_type();
>       break;
>   }
>+}
> 
>-  // |channel| can no longer be used.
>-
>-  transport_controller_->DestroyDtlsTransport(
>-      transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
>-  if (need_to_delete_rtcp) {
>-    transport_controller_->DestroyDtlsTransport(
>-        transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP);
>+bool PeerConnection::OnTransportChanged(
>+    const std::string& mid,
>+    RtpTransportInternal* rtp_transport,
>+    cricket::DtlsTransportInternal* dtls_transport) {
>+  bool ret = true;
>+  auto base_channel = GetChannel(mid);
>+  if (base_channel) {
>+    ret = base_channel->SetRtpTransport(rtp_transport);
>   }
>+  if (sctp_transport_ && mid == sctp_mid_) {
>+    sctp_transport_->SetDtlsTransport(dtls_transport);
>+  }
>+  return ret;
>+}
>+
>+PeerConnectionObserver* PeerConnection::Observer() const {
>+  // In earlier production code, the pointer was not cleared on close,
>+  // which might have led to undefined behavior if the observer was not
>+  // deallocated, or strange crashes if it was.
>+  // We use CHECK in order to catch such behavior if it exists.
>+  // TODO(hta): Remove or replace with DCHECK if nothing is found.
>+  RTC_CHECK(observer_);
>+  return observer_;
> }
> 
> void PeerConnection::ClearStatsCache() {
>@@ -5682,4 +6312,9 @@ void PeerConnection::ClearStatsCache() {
>   }
> }
> 
>+void PeerConnection::RequestUsagePatternReportForTesting() {
>+  signaling_thread()->Post(RTC_FROM_HERE, this, MSG_REPORT_USAGE_PATTERN,
>+                           nullptr);
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection.h
>index 705fff50014..25fbe1fc207 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection.h
>@@ -20,7 +20,9 @@
> #include "api/peerconnectioninterface.h"
> #include "api/turncustomizer.h"
> #include "pc/iceserverparsing.h"
>+#include "pc/jseptransportcontroller.h"
> #include "pc/peerconnectionfactory.h"
>+#include "pc/peerconnectioninternal.h"
> #include "pc/rtcstatscollector.h"
> #include "pc/rtptransceiver.h"
> #include "pc/statscollector.h"
>@@ -33,27 +35,6 @@ class MediaStreamObserver;
> class VideoRtpReceiver;
> class RtcEventLog;
> 
>-// Statistics for all the transports of the session.
>-// TODO(pthatcher): Think of a better name for this.  We already have
>-// a TransportStats in transport.h.  Perhaps TransportsStats?
>-struct SessionStats {
>-  std::map<std::string, cricket::TransportStats> transport_stats;
>-};
>-
>-struct ChannelNamePair {
>-  ChannelNamePair(const std::string& content_name,
>-                  const std::string& transport_name)
>-      : content_name(content_name), transport_name(transport_name) {}
>-  std::string content_name;
>-  std::string transport_name;
>-};
>-
>-struct ChannelNamePairs {
>-  rtc::Optional<ChannelNamePair> voice;
>-  rtc::Optional<ChannelNamePair> video;
>-  rtc::Optional<ChannelNamePair> data;
>-};
>-
> // PeerConnection is the implementation of the PeerConnection object as defined
> // by the PeerConnectionInterface API surface.
> // The class currently is solely responsible for the following:
>@@ -68,20 +49,35 @@ struct ChannelNamePairs {
> // - Generating offers and answers based on the current state.
> // - The ICE state machine.
> // - Generating stats.
>-class PeerConnection : public PeerConnectionInterface,
>+class PeerConnection : public PeerConnectionInternal,
>                        public DataChannelProviderInterface,
>+                       public JsepTransportController::Observer,
>                        public rtc::MessageHandler,
>                        public sigslot::has_slots<> {
>  public:
>+  enum class UsageEvent : int {
>+    TURN_SERVER_ADDED = 0x01,
>+    STUN_SERVER_ADDED = 0x02,
>+    DATA_ADDED = 0x04,
>+    AUDIO_ADDED = 0x08,
>+    VIDEO_ADDED = 0x10,
>+    SET_LOCAL_DESCRIPTION_CALLED = 0x20,
>+    SET_REMOTE_DESCRIPTION_CALLED = 0x40,
>+    CANDIDATE_COLLECTED = 0x80,
>+    REMOTE_CANDIDATE_ADDED = 0x100,
>+    ICE_STATE_CONNECTED = 0x200,
>+    CLOSE_CALLED = 0x400,
>+    PRIVATE_CANDIDATE_COLLECTED = 0x800,
>+    MAX_VALUE = 0x1000,
>+  };
>+
>   explicit PeerConnection(PeerConnectionFactory* factory,
>                           std::unique_ptr<RtcEventLog> event_log,
>                           std::unique_ptr<Call> call);
> 
>   bool Initialize(
>       const PeerConnectionInterface::RTCConfiguration& configuration,
>-      std::unique_ptr<cricket::PortAllocator> allocator,
>-      std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
>-      PeerConnectionObserver* observer);
>+      PeerConnectionDependencies dependencies);
> 
>   rtc::scoped_refptr<StreamCollectionInterface> local_streams() override;
>   rtc::scoped_refptr<StreamCollectionInterface> remote_streams() override;
>@@ -90,11 +86,10 @@ class PeerConnection : public PeerConnectionInterface,
> 
>   RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>> AddTrack(
>       rtc::scoped_refptr<MediaStreamTrackInterface> track,
>-      const std::vector<std::string>& stream_labels) override;
>-  rtc::scoped_refptr<RtpSenderInterface> AddTrack(
>-      MediaStreamTrackInterface* track,
>-      std::vector<MediaStreamInterface*> streams) override;
>+      const std::vector<std::string>& stream_ids) override;
>   bool RemoveTrack(RtpSenderInterface* sender) override;
>+  RTCError RemoveTrackNew(
>+      rtc::scoped_refptr<RtpSenderInterface> sender) override;
> 
>   RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> AddTransceiver(
>       rtc::scoped_refptr<MediaStreamTrackInterface> track) override;
>@@ -116,8 +111,8 @@ class PeerConnection : public PeerConnectionInterface,
>   // See https://www.w3.org/TR/webrtc/#rtcdtlstransport-interface
>   std::unique_ptr<rtc::SSLCertificate> GetRemoteAudioSSLCertificate();
> 
>-  rtc::scoped_refptr<DtmfSenderInterface> CreateDtmfSender(
>-      AudioTrackInterface* track) override;
>+  // Version of the above method that returns the full certificate chain.
>+  std::unique_ptr<rtc::SSLCertChain> GetRemoteAudioSSLCertChain();
> 
>   rtc::scoped_refptr<RtpSenderInterface> CreateSender(
>       const std::string& kind,
>@@ -133,10 +128,18 @@ class PeerConnection : public PeerConnectionInterface,
>   rtc::scoped_refptr<DataChannelInterface> CreateDataChannel(
>       const std::string& label,
>       const DataChannelInit* config) override;
>+  // WARNING: LEGACY. See peerconnectioninterface.h
>   bool GetStats(StatsObserver* observer,
>                 webrtc::MediaStreamTrackInterface* track,
>                 StatsOutputLevel level) override;
>+  // Spec-complaint GetStats(). See peerconnectioninterface.h
>   void GetStats(RTCStatsCollectorCallback* callback) override;
>+  void GetStats(
>+      rtc::scoped_refptr<RtpSenderInterface> selector,
>+      rtc::scoped_refptr<RTCStatsCollectorCallback> callback) override;
>+  void GetStats(
>+      rtc::scoped_refptr<RtpReceiverInterface> selector,
>+      rtc::scoped_refptr<RTCStatsCollectorCallback> callback) override;
>   void ClearStatsCache() override;
> 
>   SignalingState signaling_state() override;
>@@ -184,9 +187,7 @@ class PeerConnection : public PeerConnectionInterface,
>   bool RemoveIceCandidates(
>       const std::vector<cricket::Candidate>& candidates) override;
> 
>-  void RegisterUMAObserver(UMAObserver* observer) override;
>-
>-  RTCError SetBitrate(const BitrateParameters& bitrate) override;
>+  RTCError SetBitrate(const BitrateSettings& bitrate) override;
> 
>   void SetBitrateAllocationStrategy(
>       std::unique_ptr<rtc::BitrateAllocationStrategy>
>@@ -203,91 +204,70 @@ class PeerConnection : public PeerConnectionInterface,
> 
>   void Close() override;
> 
>-  sigslot::signal1<DataChannel*> SignalDataChannelCreated;
>+  // PeerConnectionInternal implementation.
>+  rtc::Thread* network_thread() const override {
>+    return factory_->network_thread();
>+  }
>+  rtc::Thread* worker_thread() const override {
>+    return factory_->worker_thread();
>+  }
>+  rtc::Thread* signaling_thread() const override {
>+    return factory_->signaling_thread();
>+  }
>+
>+  std::string session_id() const override { return session_id_; }
> 
>-  // Virtual for unit tests.
>-  virtual const std::vector<rtc::scoped_refptr<DataChannel>>&
>-  sctp_data_channels() const {
>-    return sctp_data_channels_;
>+  bool initial_offerer() const override {
>+    return transport_controller_ && transport_controller_->initial_offerer();
>   }
> 
>-  rtc::Thread* network_thread() const { return factory_->network_thread(); }
>-  rtc::Thread* worker_thread() const { return factory_->worker_thread(); }
>-  rtc::Thread* signaling_thread() const { return factory_->signaling_thread(); }
>-
>-  // The SDP session ID as defined by RFC 3264.
>-  virtual const std::string& session_id() const { return session_id_; }
>-
>-  // Returns true if we were the initial offerer.
>-  bool initial_offerer() const { return initial_offerer_ && *initial_offerer_; }
>-
>-  // Returns stats for all channels of all transports.
>-  // This avoids exposing the internal structures used to track them.
>-  // The parameterless version creates |ChannelNamePairs| from |voice_channel|,
>-  // |video_channel| and |voice_channel| if available - this requires it to be
>-  // called on the signaling thread - and invokes the other |GetStats|. The
>-  // other |GetStats| can be invoked on any thread; if not invoked on the
>-  // network thread a thread hop will happen.
>-  std::unique_ptr<SessionStats> GetSessionStats_s();
>-  virtual std::unique_ptr<SessionStats> GetSessionStats(
>-      const ChannelNamePairs& channel_name_pairs);
>-
>-  // virtual so it can be mocked in unit tests
>-  virtual bool GetLocalCertificate(
>-      const std::string& transport_name,
>-      rtc::scoped_refptr<rtc::RTCCertificate>* certificate);
>-  virtual std::unique_ptr<rtc::SSLCertificate> GetRemoteSSLCertificate(
>-      const std::string& transport_name);
>-
>-  virtual Call::Stats GetCallStats();
>-
>-  // Exposed for stats collecting.
>-  // TODO(steveanton): Switch callers to use the plural form and remove these.
>-  virtual cricket::VoiceChannel* voice_channel() const {
>-    if (IsUnifiedPlan()) {
>-      // TODO(steveanton): Change stats collection to work with transceivers.
>-      return nullptr;
>-    }
>-    return static_cast<cricket::VoiceChannel*>(
>-        GetAudioTransceiver()->internal()->channel());
>+  std::vector<
>+      rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>>
>+  GetTransceiversInternal() const override {
>+    return transceivers_;
>   }
>-  virtual cricket::VideoChannel* video_channel() const {
>-    if (IsUnifiedPlan()) {
>-      // TODO(steveanton): Change stats collection to work with transceivers.
>-      return nullptr;
>-    }
>-    return static_cast<cricket::VideoChannel*>(
>-        GetVideoTransceiver()->internal()->channel());
>+
>+  bool GetLocalTrackIdBySsrc(uint32_t ssrc, std::string* track_id) override;
>+  bool GetRemoteTrackIdBySsrc(uint32_t ssrc, std::string* track_id) override;
>+
>+  sigslot::signal1<DataChannel*>& SignalDataChannelCreated() override {
>+    return SignalDataChannelCreated_;
>   }
> 
>-  // Only valid when using deprecated RTP data channels.
>-  virtual cricket::RtpDataChannel* rtp_data_channel() {
>+  cricket::RtpDataChannel* rtp_data_channel() const override {
>     return rtp_data_channel_;
>   }
>-  virtual rtc::Optional<std::string> sctp_content_name() const {
>-    return sctp_content_name_;
>-  }
>-  virtual rtc::Optional<std::string> sctp_transport_name() const {
>-    return sctp_transport_name_;
>+
>+  std::vector<rtc::scoped_refptr<DataChannel>> sctp_data_channels()
>+      const override {
>+    return sctp_data_channels_;
>   }
> 
>-  // Get the id used as a media stream track's "id" field from ssrc.
>-  virtual bool GetLocalTrackIdBySsrc(uint32_t ssrc, std::string* track_id);
>-  virtual bool GetRemoteTrackIdBySsrc(uint32_t ssrc, std::string* track_id);
>+  absl::optional<std::string> sctp_content_name() const override {
>+    return sctp_mid_;
>+  }
> 
>-  // Returns true if there was an ICE restart initiated by the remote offer.
>-  bool IceRestartPending(const std::string& content_name) const;
>+  absl::optional<std::string> sctp_transport_name() const override;
> 
>-  // Returns true if the ICE restart flag above was set, and no ICE restart has
>-  // occurred yet for this transport (by applying a local description with
>-  // changed ufrag/password). If the transport has been deleted as a result of
>-  // bundling, returns false.
>-  bool NeedsIceRestart(const std::string& content_name) const;
>+  cricket::CandidateStatsList GetPooledCandidateStats() const override;
>+  std::map<std::string, std::string> GetTransportNamesByMid() const override;
>+  std::map<std::string, cricket::TransportStats> GetTransportStatsByNames(
>+      const std::set<std::string>& transport_names) override;
>+  Call::Stats GetCallStats() override;
> 
>-  // Get SSL role for an arbitrary m= section (handles bundling correctly).
>-  // TODO(deadbeef): This is only used internally by the session description
>-  // factory, it shouldn't really be public).
>-  bool GetSslRole(const std::string& content_name, rtc::SSLRole* role);
>+  bool GetLocalCertificate(
>+      const std::string& transport_name,
>+      rtc::scoped_refptr<rtc::RTCCertificate>* certificate) override;
>+  std::unique_ptr<rtc::SSLCertChain> GetRemoteSSLCertChain(
>+      const std::string& transport_name) override;
>+  bool IceRestartPending(const std::string& content_name) const override;
>+  bool NeedsIceRestart(const std::string& content_name) const override;
>+  bool GetSslRole(const std::string& content_name, rtc::SSLRole* role) override;
>+
>+  void ReturnHistogramVeryQuicklyForTesting() {
>+    return_histogram_very_quickly_ = true;
>+  }
>+  void RequestUsagePatternReportForTesting();
> 
>  protected:
>   ~PeerConnection() override;
>@@ -298,37 +278,29 @@ class PeerConnection : public PeerConnectionInterface,
> 
>   struct RtpSenderInfo {
>     RtpSenderInfo() : first_ssrc(0) {}
>-    RtpSenderInfo(const std::string& stream_label,
>+    RtpSenderInfo(const std::string& stream_id,
>                   const std::string sender_id,
>                   uint32_t ssrc)
>-        : stream_label(stream_label), sender_id(sender_id), first_ssrc(ssrc) {}
>+        : stream_id(stream_id), sender_id(sender_id), first_ssrc(ssrc) {}
>     bool operator==(const RtpSenderInfo& other) {
>-      return this->stream_label == other.stream_label &&
>+      return this->stream_id == other.stream_id &&
>              this->sender_id == other.sender_id &&
>              this->first_ssrc == other.first_ssrc;
>     }
>-    std::string stream_label;
>+    std::string stream_id;
>     std::string sender_id;
>     // An RtpSender can have many SSRCs. The first one is used as a sort of ID
>     // for communicating with the lower layers.
>     uint32_t first_ssrc;
>   };
> 
>-  struct TrackEvent {
>-    rtc::scoped_refptr<RtpReceiverInterface> receiver;
>-    std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams;
>-  };
>-
>   // Implements MessageHandler.
>   void OnMessage(rtc::Message* msg) override;
> 
>-  cricket::VoiceMediaChannel* voice_media_channel() const {
>-    return voice_channel() ? voice_channel()->media_channel() : nullptr;
>-  }
>-
>-  cricket::VideoMediaChannel* video_media_channel() const {
>-    return video_channel() ? video_channel()->media_channel() : nullptr;
>-  }
>+  // Plan B helpers for getting the voice/video media channels for the single
>+  // audio/video transceiver, if it exists.
>+  cricket::VoiceMediaChannel* voice_media_channel() const;
>+  cricket::VideoMediaChannel* video_media_channel() const;
> 
>   std::vector<rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>>
>   GetSendersInternal() const;
>@@ -341,6 +313,9 @@ class PeerConnection : public PeerConnectionInterface,
>   rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
>   GetVideoTransceiver() const;
> 
>+  rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
>+  GetFirstAudioTransceiver() const;
>+
>   void CreateAudioReceiver(MediaStreamInterface* stream,
>                            const RtpSenderInfo& remote_sender_info);
> 
>@@ -361,11 +336,11 @@ class PeerConnection : public PeerConnectionInterface,
>   // AddTrack implementation when Unified Plan is specified.
>   RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>> AddTrackUnifiedPlan(
>       rtc::scoped_refptr<MediaStreamTrackInterface> track,
>-      const std::vector<std::string>& stream_labels);
>+      const std::vector<std::string>& stream_ids);
>   // AddTrack implementation when Plan B is specified.
>   RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>> AddTrackPlanB(
>       rtc::scoped_refptr<MediaStreamTrackInterface> track,
>-      const std::vector<std::string>& stream_labels);
>+      const std::vector<std::string>& stream_ids);
> 
>   // Returns the first RtpTransceiver suitable for a newly added track, if such
>   // transceiver is available.
>@@ -373,21 +348,22 @@ class PeerConnection : public PeerConnectionInterface,
>   FindFirstTransceiverForAddedTrack(
>       rtc::scoped_refptr<MediaStreamTrackInterface> track);
> 
>-  // RemoveTrack that returns an RTCError.
>-  RTCError RemoveTrackInternal(rtc::scoped_refptr<RtpSenderInterface> sender);
>-
>   rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
>   FindTransceiverBySender(rtc::scoped_refptr<RtpSenderInterface> sender);
> 
>+  // Internal implementation for AddTransceiver family of methods. If
>+  // |fire_callback| is set, fires OnRenegotiationNeeded callback if successful.
>   RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> AddTransceiver(
>       cricket::MediaType media_type,
>       rtc::scoped_refptr<MediaStreamTrackInterface> track,
>-      const RtpTransceiverInit& init);
>+      const RtpTransceiverInit& init,
>+      bool fire_callback = true);
> 
>   rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>
>   CreateSender(cricket::MediaType media_type,
>+               const std::string& id,
>                rtc::scoped_refptr<MediaStreamTrackInterface> track,
>-               const std::vector<std::string>& stream_labels);
>+               const std::vector<std::string>& stream_ids);
> 
>   rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
>   CreateReceiver(cricket::MediaType media_type, const std::string& receiver_id);
>@@ -425,10 +401,10 @@ class PeerConnection : public PeerConnectionInterface,
>   void PostSetSessionDescriptionSuccess(
>       SetSessionDescriptionObserver* observer);
>   void PostSetSessionDescriptionFailure(SetSessionDescriptionObserver* observer,
>-                                        const std::string& error);
>+                                        RTCError&& error);
>   void PostCreateSessionDescriptionFailure(
>       CreateSessionDescriptionObserver* observer,
>-      const std::string& error);
>+      RTCError error);
> 
>   // Synchronous implementations of SetLocalDescription/SetRemoteDescription
>   // that return an RTCError instead of invoking a callback.
>@@ -441,8 +417,9 @@ class PeerConnection : public PeerConnectionInterface,
>   // part of setting the local/remote description.
>   RTCError UpdateTransceiversAndDataChannels(
>       cricket::ContentSource source,
>-      const SessionDescriptionInterface* old_session,
>-      const SessionDescriptionInterface& new_session);
>+      const SessionDescriptionInterface& new_session,
>+      const SessionDescriptionInterface* old_local_description,
>+      const SessionDescriptionInterface* old_remote_description);
> 
>   // Either creates or destroys the transceiver's BaseChannel according to the
>   // given media section.
>@@ -462,9 +439,11 @@ class PeerConnection : public PeerConnectionInterface,
>   RTCErrorOr<
>       rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>>
>   AssociateTransceiver(cricket::ContentSource source,
>+                       SdpType type,
>                        size_t mline_index,
>                        const cricket::ContentInfo& content,
>-                       const cricket::ContentInfo* old_content);
>+                       const cricket::ContentInfo* old_local_content,
>+                       const cricket::ContentInfo* old_remote_content);
> 
>   // Returns the RtpTransceiver, if found, that is associated to the given MID.
>   rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
>@@ -489,6 +468,22 @@ class PeerConnection : public PeerConnectionInterface,
>           transceiver,
>       const SessionDescriptionInterface* sdesc) const;
> 
>+  // Runs the algorithm **process the removal of a remote track** specified in
>+  // the WebRTC specification.
>+  // This method will update the following lists:
>+  // |remove_list| is the list of transceivers for which the receiving track is
>+  //     being removed.
>+  // |removed_streams| is the list of streams which no longer have a receiving
>+  //     track so should be removed.
>+  // https://w3c.github.io/webrtc-pc/#process-remote-track-removal
>+  void ProcessRemovalOfRemoteTrack(
>+      rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
>+          transceiver,
>+      std::vector<rtc::scoped_refptr<RtpTransceiverInterface>>* remove_list,
>+      std::vector<rtc::scoped_refptr<MediaStreamInterface>>* removed_streams);
>+
>+  void OnNegotiationNeeded();
>+
>   bool IsClosed() const {
>     return signaling_state_ == PeerConnectionInterface::kClosed;
>   }
>@@ -507,6 +502,14 @@ class PeerConnection : public PeerConnectionInterface,
>           offer_answer_options,
>       cricket::MediaSessionOptions* session_options);
> 
>+  RTCError HandleLegacyOfferOptions(const RTCOfferAnswerOptions& options);
>+  void RemoveRecvDirectionFromReceivingTransceiversOfType(
>+      cricket::MediaType media_type);
>+  void AddUpToOneReceivingTransceiverOfType(cricket::MediaType media_type);
>+  std::vector<
>+      rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>>
>+  GetReceivingTransceiversOfType(cricket::MediaType media_type);
>+
>   // Returns a MediaSessionOptions struct with options decided by
>   // |constraints|, the local MediaStreams and DataChannels.
>   void GetOptionsForAnswer(const RTCOfferAnswerOptions& offer_answer_options,
>@@ -526,9 +529,9 @@ class PeerConnection : public PeerConnectionInterface,
>       const SessionDescriptionInterface* session_desc,
>       RtpTransceiverDirection audio_direction,
>       RtpTransceiverDirection video_direction,
>-      rtc::Optional<size_t>* audio_index,
>-      rtc::Optional<size_t>* video_index,
>-      rtc::Optional<size_t>* data_index,
>+      absl::optional<size_t>* audio_index,
>+      absl::optional<size_t>* video_index,
>+      absl::optional<size_t>* data_index,
>       cricket::MediaSessionOptions* session_options);
> 
>   // Generates the active MediaDescriptionOptions for the local data channel
>@@ -544,7 +547,7 @@ class PeerConnection : public PeerConnectionInterface,
>   // Returns the MID for the data section associated with either the
>   // RtpDataChannel or SCTP data channel, if it has been set. If no data
>   // channels are configured this will return nullopt.
>-  rtc::Optional<std::string> GetDataMid() const;
>+  absl::optional<std::string> GetDataMid() const;
> 
>   // Remove all local and remote senders of type |media_type|.
>   // Called when a media type is rejected (m-line set to port 0).
>@@ -657,7 +660,7 @@ class PeerConnection : public PeerConnectionInterface,
>   std::vector<RtpSenderInfo>* GetLocalSenderInfos(
>       cricket::MediaType media_type);
>   const RtpSenderInfo* FindSenderInfo(const std::vector<RtpSenderInfo>& infos,
>-                                      const std::string& stream_label,
>+                                      const std::string& stream_id,
>                                       const std::string sender_id) const;
> 
>   // Returns the specified SCTP DataChannel in sctp_data_channels_,
>@@ -665,7 +668,10 @@ class PeerConnection : public PeerConnectionInterface,
>   DataChannel* FindDataChannelBySid(int sid) const;
> 
>   // Called when first configuring the port allocator.
>-  bool InitializePortAllocator_n(const RTCConfiguration& configuration);
>+  bool InitializePortAllocator_n(
>+      const cricket::ServerAddresses& stun_servers,
>+      const std::vector<cricket::RelayServerConfig>& turn_servers,
>+      const RTCConfiguration& configuration);
>   // Called when SetConfiguration is called to apply the supported subset
>   // of the configuration on the network thread.
>   bool ReconfigurePortAllocator_n(
>@@ -674,7 +680,8 @@ class PeerConnection : public PeerConnectionInterface,
>       IceTransportsType type,
>       int candidate_pool_size,
>       bool prune_turn_ports,
>-      webrtc::TurnCustomizer* turn_customizer);
>+      webrtc::TurnCustomizer* turn_customizer,
>+      absl::optional<int> stun_candidate_keepalive_interval);
> 
>   // Starts output of an RTC event log to the given output object.
>   // This function should only be called from the worker thread.
>@@ -692,7 +699,6 @@ class PeerConnection : public PeerConnectionInterface,
>   RTCError ValidateConfiguration(const RTCConfiguration& config) const;
> 
>   cricket::ChannelManager* channel_manager() const;
>-  MetricsObserverInterface* metrics_observer() const;
> 
>   enum class SessionError {
>     kNone,       // No error.
>@@ -730,13 +736,6 @@ class PeerConnection : public PeerConnectionInterface,
>       const rtc::scoped_refptr<rtc::RTCCertificate>& certificate);
>   void OnDtlsSrtpSetupFailure(cricket::BaseChannel*, bool rtcp);
> 
>-  cricket::TransportController* transport_controller() const {
>-    return transport_controller_.get();
>-  }
>-
>-  // Return all managed, non-null channels.
>-  std::vector<cricket::BaseChannel*> Channels() const;
>-
>   // Non-const versions of local_description()/remote_description(), for use
>   // internally.
>   SessionDescriptionInterface* mutable_local_description() {
>@@ -751,7 +750,9 @@ class PeerConnection : public PeerConnectionInterface,
>   // Updates the error state, signaling if necessary.
>   void SetSessionError(SessionError error, const std::string& error_desc);
> 
>-  RTCError UpdateSessionState(SdpType type, cricket::ContentSource source);
>+  RTCError UpdateSessionState(SdpType type,
>+                              cricket::ContentSource source,
>+                              const cricket::SessionDescription* description);
>   // Push the media parts of the local or remote session description
>   // down to all of the channels.
>   RTCError PushdownMediaDescription(SdpType type,
>@@ -768,18 +769,6 @@ class PeerConnection : public PeerConnectionInterface,
>       const std::string& content_name,
>       cricket::TransportDescription* info);
> 
>-  // Returns the transport name for the given media section identified by |mid|.
>-  // If BUNDLE is enabled and the media section is part of the bundle group,
>-  // the transport name will be the first mid in the bundle group. Otherwise,
>-  // the transport name will be the mid of the media section.
>-  std::string GetTransportNameForMediaSection(
>-      const std::string& mid,
>-      const cricket::ContentGroup* bundle_group) const;
>-
>-  // Cause all the BaseChannels in the bundle group to have the same
>-  // transport channel.
>-  bool EnableBundle(const cricket::ContentGroup& bundle);
>-
>   // Enables media channels to allow sending of media.
>   // This enables media to flow on all configured audio/video channels and the
>   // RtpDataChannel.
>@@ -816,20 +805,12 @@ class PeerConnection : public PeerConnectionInterface,
>       const cricket::SessionDescription& desc) const;
> 
>   // Helper methods to create media channels.
>-  cricket::VoiceChannel* CreateVoiceChannel(const std::string& mid,
>-                                            const std::string& transport_name);
>-  cricket::VideoChannel* CreateVideoChannel(const std::string& mid,
>-                                            const std::string& transport_name);
>-  bool CreateDataChannel(const std::string& mid,
>-                         const std::string& transport_name);
>-
>-  std::unique_ptr<SessionStats> GetSessionStats_n(
>-      const ChannelNamePairs& channel_name_pairs);
>-
>-  bool CreateSctpTransport_n(const std::string& content_name,
>-                             const std::string& transport_name);
>+  cricket::VoiceChannel* CreateVoiceChannel(const std::string& mid);
>+  cricket::VideoChannel* CreateVideoChannel(const std::string& mid);
>+  bool CreateDataChannel(const std::string& mid);
>+
>+  bool CreateSctpTransport_n(const std::string& mid);
>   // For bundling.
>-  void ChangeSctpTransport_n(const std::string& transport_name);
>   void DestroySctpTransport_n();
>   // SctpTransport signal handlers. Needed to marshal signals from the network
>   // to signaling thread.
>@@ -843,7 +824,8 @@ class PeerConnection : public PeerConnectionInterface,
>   // CONTROL messages on unused SIDs and processes them as OPEN messages.
>   void OnSctpTransportDataReceived_s(const cricket::ReceiveDataParams& params,
>                                      const rtc::CopyOnWriteBuffer& payload);
>-  void OnSctpStreamClosedRemotely_n(int sid);
>+  void OnSctpClosingProcedureStartedRemotely_n(int sid);
>+  void OnSctpClosingProcedureComplete_n(int sid);
> 
>   bool ValidateBundleSettings(const cricket::SessionDescription* desc);
>   bool HasRtcpMuxEnabled(const cricket::ContentInfo* content);
>@@ -873,7 +855,7 @@ class PeerConnection : public PeerConnectionInterface,
>   // this session.
>   bool SrtpRequired() const;
> 
>-  // TransportController signal handlers.
>+  // JsepTransportController signal handlers.
>   void OnTransportControllerConnectionState(cricket::IceConnectionState state);
>   void OnTransportControllerGatheringState(cricket::IceGatheringState state);
>   void OnTransportControllerCandidatesGathered(
>@@ -886,6 +868,13 @@ class PeerConnection : public PeerConnectionInterface,
>   const char* SessionErrorToString(SessionError error) const;
>   std::string GetSessionErrorMsg();
> 
>+  // Report the UMA metric SdpFormatReceived for the given remote offer.
>+  void ReportSdpFormatReceived(const SessionDescriptionInterface& remote_offer);
>+
>+  // Report inferred negotiated SDP semantics from a local/remote answer to the
>+  // UMA observer.
>+  void ReportNegotiatedSdpSemantics(const SessionDescriptionInterface& answer);
>+
>   // Invoked when TransportController connection completion is signaled.
>   // Reports stats for all transports in use.
>   void ReportTransportStats();
>@@ -893,14 +882,16 @@ class PeerConnection : public PeerConnectionInterface,
>   // Gather the usage of IPv4/IPv6 as best connection.
>   void ReportBestConnectionState(const cricket::TransportStats& stats);
> 
>-  void ReportNegotiatedCiphers(const cricket::TransportStats& stats);
>+  void ReportNegotiatedCiphers(const cricket::TransportStats& stats,
>+                               const std::set<cricket::MediaType>& media_types);
>+
>+  void NoteUsageEvent(UsageEvent event);
>+  void ReportUsagePattern() const;
> 
>   void OnSentPacket_w(const rtc::SentPacket& sent_packet);
> 
>   const std::string GetTransportName(const std::string& content_name);
> 
>-  void DestroyRtcpTransport_n(const std::string& transport_name);
>-
>   // Destroys and clears the BaseChannel associated with the given transceiver,
>   // if such channel is set.
>   void DestroyTransceiverChannel(
>@@ -914,6 +905,22 @@ class PeerConnection : public PeerConnectionInterface,
>   // method is called.
>   void DestroyBaseChannel(cricket::BaseChannel* channel);
> 
>+  // JsepTransportController::Observer override.
>+  //
>+  // Called by |transport_controller_| when processing transport information
>+  // from a session description, and the mapping from m= sections to transports
>+  // changed (as a result of BUNDLE negotiation, or m= sections being
>+  // rejected).
>+  bool OnTransportChanged(
>+      const std::string& mid,
>+      RtpTransportInternal* rtp_transport,
>+      cricket::DtlsTransportInternal* dtls_transport) override;
>+
>+  // Returns the observer. Will crash on CHECK if the observer is removed.
>+  PeerConnectionObserver* Observer() const;
>+
>+  sigslot::signal1<DataChannel*> SignalDataChannelCreated_;
>+
>   // Storing the factory as a scoped reference pointer ensures that the memory
>   // in the PeerConnectionFactoryImpl remains available as long as the
>   // PeerConnection is running. It is passed to PeerConnection as a raw pointer.
>@@ -922,7 +929,6 @@ class PeerConnection : public PeerConnectionInterface,
>   // will refer to the same reference count.
>   rtc::scoped_refptr<PeerConnectionFactory> factory_;
>   PeerConnectionObserver* observer_ = nullptr;
>-  rtc::scoped_refptr<UMAObserver> uma_observer_ = nullptr;
> 
>   // The EventLog needs to outlive |call_| (and any other object that uses it).
>   std::unique_ptr<RtcEventLog> event_log_;
>@@ -932,7 +938,12 @@ class PeerConnection : public PeerConnectionInterface,
>   IceGatheringState ice_gathering_state_ = kIceGatheringNew;
>   PeerConnectionInterface::RTCConfiguration configuration_;
> 
>+  // TODO(zstein): |async_resolver_factory_| can currently be nullptr if it
>+  // is not injected. It should be required once chromium supplies it.
>+  std::unique_ptr<AsyncResolverFactory> async_resolver_factory_;
>   std::unique_ptr<cricket::PortAllocator> port_allocator_;
>+  std::unique_ptr<rtc::SSLCertificateVerifier> tls_cert_verifier_;
>+  int port_allocator_flags_ = 0;
> 
>   // One PeerConnection has only one RTCP CNAME.
>   // https://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-26#section-4.9
>@@ -974,20 +985,16 @@ class PeerConnection : public PeerConnectionInterface,
>   std::string session_error_desc_;
> 
>   std::string session_id_;
>-  rtc::Optional<bool> initial_offerer_;
> 
>-  std::unique_ptr<cricket::TransportController> transport_controller_;
>+  std::unique_ptr<JsepTransportController> transport_controller_;
>   std::unique_ptr<cricket::SctpTransportInternalFactory> sctp_factory_;
>   // |rtp_data_channel_| is used if in RTP data channel mode, |sctp_transport_|
>   // when using SCTP.
>   cricket::RtpDataChannel* rtp_data_channel_ = nullptr;
> 
>   std::unique_ptr<cricket::SctpTransportInternal> sctp_transport_;
>-  // |sctp_transport_name_| keeps track of what DTLS transport the SCTP
>-  // transport is using (which can change due to bundling).
>-  rtc::Optional<std::string> sctp_transport_name_;
>-  // |sctp_content_name_| is the content name (MID) in SDP.
>-  rtc::Optional<std::string> sctp_content_name_;
>+  // |sctp_mid_| is the content name (MID) in SDP.
>+  absl::optional<std::string> sctp_mid_;
>   // Value cached on signaling thread. Only updated when SctpReadyToSendData
>   // fires on the signaling thread.
>   bool sctp_ready_to_send_data_ = false;
>@@ -1005,7 +1012,8 @@ class PeerConnection : public PeerConnectionInterface,
>   sigslot::signal2<const cricket::ReceiveDataParams&,
>                    const rtc::CopyOnWriteBuffer&>
>       SignalSctpDataReceived;
>-  sigslot::signal1<int> SignalSctpStreamClosedRemotely;
>+  sigslot::signal1<int> SignalSctpClosingProcedureStartedRemotely;
>+  sigslot::signal1<int> SignalSctpClosingProcedureComplete;
> 
>   std::unique_ptr<SessionDescriptionInterface> current_local_description_;
>   std::unique_ptr<SessionDescriptionInterface> pending_local_description_;
>@@ -1028,6 +1036,9 @@ class PeerConnection : public PeerConnectionInterface,
>   // Member variables for caching global options.
>   cricket::AudioOptions audio_options_;
>   cricket::VideoOptions video_options_;
>+
>+  int usage_event_accumulator_ = 0;
>+  bool return_histogram_very_quickly_ = false;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_bundle_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_bundle_unittest.cc
>index 925156f97a7..907669b5ad3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_bundle_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_bundle_unittest.cc
>@@ -11,6 +11,8 @@
> #include "api/audio_codecs/builtin_audio_decoder_factory.h"
> #include "api/audio_codecs/builtin_audio_encoder_factory.h"
> #include "api/peerconnectionproxy.h"
>+#include "api/video_codecs/builtin_video_decoder_factory.h"
>+#include "api/video_codecs/builtin_video_encoder_factory.h"
> #include "p2p/base/fakeportallocator.h"
> #include "p2p/base/teststunserver.h"
> #include "p2p/client/basicportallocator.h"
>@@ -21,10 +23,10 @@
> #ifdef WEBRTC_ANDROID
> #include "pc/test/androidtestinitializer.h"
> #endif
>+#include "absl/memory/memory.h"
> #include "pc/test/fakeaudiocapturemodule.h"
> #include "rtc_base/fakenetwork.h"
> #include "rtc_base/gunit.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/virtualsocketserver.h"
> #include "test/gmock.h"
> 
>@@ -35,6 +37,7 @@ using RTCConfiguration = PeerConnectionInterface::RTCConfiguration;
> using RTCOfferAnswerOptions = PeerConnectionInterface::RTCOfferAnswerOptions;
> using RtcpMuxPolicy = PeerConnectionInterface::RtcpMuxPolicy;
> using rtc::SocketAddress;
>+using ::testing::Combine;
> using ::testing::ElementsAre;
> using ::testing::UnorderedElementsAre;
> using ::testing::Values;
>@@ -45,6 +48,7 @@ constexpr int kDefaultTimeout = 10000;
> // RtpSenderInterface/DtlsTransportInterface objects once they're available in
> // the API. The RtpSender can be used to determine which transport a given media
> // will use: https://www.w3.org/TR/webrtc/#dom-rtcrtpsender-transport
>+// Should also be able to remove GetTransceiversForTesting at that point.
> 
> class PeerConnectionWrapperForBundleTest : public PeerConnectionWrapper {
>  public:
>@@ -57,36 +61,55 @@ class PeerConnectionWrapperForBundleTest : public PeerConnectionWrapper {
>       const auto& content = desc->contents()[i];
>       if (content.media_description()->type() == media_type) {
>         candidate->set_transport_name(content.name);
>-        JsepIceCandidate jsep_candidate(content.name, i, *candidate);
>-        return pc()->AddIceCandidate(&jsep_candidate);
>+        std::unique_ptr<IceCandidateInterface> jsep_candidate =
>+            CreateIceCandidate(content.name, i, *candidate);
>+        return pc()->AddIceCandidate(jsep_candidate.get());
>       }
>     }
>     RTC_NOTREACHED();
>     return false;
>   }
> 
>-  rtc::PacketTransportInternal* voice_rtp_transport_channel() {
>-    return (voice_channel() ? voice_channel()->rtp_dtls_transport() : nullptr);
>+  rtc::PacketTransportInternal* voice_rtp_transport() {
>+    return (voice_channel() ? voice_channel()->rtp_packet_transport()
>+                            : nullptr);
>   }
> 
>-  rtc::PacketTransportInternal* voice_rtcp_transport_channel() {
>-    return (voice_channel() ? voice_channel()->rtcp_dtls_transport() : nullptr);
>+  rtc::PacketTransportInternal* voice_rtcp_transport() {
>+    return (voice_channel() ? voice_channel()->rtcp_packet_transport()
>+                            : nullptr);
>   }
> 
>   cricket::VoiceChannel* voice_channel() {
>-    return GetInternalPeerConnection()->voice_channel();
>+    auto transceivers = GetInternalPeerConnection()->GetTransceiversInternal();
>+    for (auto transceiver : transceivers) {
>+      if (transceiver->media_type() == cricket::MEDIA_TYPE_AUDIO) {
>+        return static_cast<cricket::VoiceChannel*>(
>+            transceiver->internal()->channel());
>+      }
>+    }
>+    return nullptr;
>   }
> 
>-  rtc::PacketTransportInternal* video_rtp_transport_channel() {
>-    return (video_channel() ? video_channel()->rtp_dtls_transport() : nullptr);
>+  rtc::PacketTransportInternal* video_rtp_transport() {
>+    return (video_channel() ? video_channel()->rtp_packet_transport()
>+                            : nullptr);
>   }
> 
>-  rtc::PacketTransportInternal* video_rtcp_transport_channel() {
>-    return (video_channel() ? video_channel()->rtcp_dtls_transport() : nullptr);
>+  rtc::PacketTransportInternal* video_rtcp_transport() {
>+    return (video_channel() ? video_channel()->rtcp_packet_transport()
>+                            : nullptr);
>   }
> 
>   cricket::VideoChannel* video_channel() {
>-    return GetInternalPeerConnection()->video_channel();
>+    auto transceivers = GetInternalPeerConnection()->GetTransceiversInternal();
>+    for (auto transceiver : transceivers) {
>+      if (transceiver->media_type() == cricket::MEDIA_TYPE_VIDEO) {
>+        return static_cast<cricket::VideoChannel*>(
>+            transceiver->internal()->channel());
>+      }
>+    }
>+    return nullptr;
>   }
> 
>   PeerConnection* GetInternalPeerConnection() {
>@@ -135,19 +158,23 @@ class PeerConnectionWrapperForBundleTest : public PeerConnectionWrapper {
>   rtc::FakeNetworkManager* network_;
> };
> 
>-class PeerConnectionBundleTest : public ::testing::Test {
>+class PeerConnectionBundleBaseTest : public ::testing::Test {
>  protected:
>   typedef std::unique_ptr<PeerConnectionWrapperForBundleTest> WrapperPtr;
> 
>-  PeerConnectionBundleTest()
>-      : vss_(new rtc::VirtualSocketServer()), main_(vss_.get()) {
>+  explicit PeerConnectionBundleBaseTest(SdpSemantics sdp_semantics)
>+      : vss_(new rtc::VirtualSocketServer()),
>+        main_(vss_.get()),
>+        sdp_semantics_(sdp_semantics) {
> #ifdef WEBRTC_ANDROID
>     InitializeAndroidObjects();
> #endif
>     pc_factory_ = CreatePeerConnectionFactory(
>         rtc::Thread::Current(), rtc::Thread::Current(), rtc::Thread::Current(),
>-        FakeAudioCaptureModule::Create(), CreateBuiltinAudioEncoderFactory(),
>-        CreateBuiltinAudioDecoderFactory(), nullptr, nullptr);
>+        rtc::scoped_refptr<AudioDeviceModule>(FakeAudioCaptureModule::Create()),
>+        CreateBuiltinAudioEncoderFactory(), CreateBuiltinAudioDecoderFactory(),
>+        CreateBuiltinVideoEncoderFactory(), CreateBuiltinVideoDecoderFactory(),
>+        nullptr /* audio_mixer */, nullptr /* audio_processing */);
>   }
> 
>   WrapperPtr CreatePeerConnection() {
>@@ -157,18 +184,20 @@ class PeerConnectionBundleTest : public ::testing::Test {
>   WrapperPtr CreatePeerConnection(const RTCConfiguration& config) {
>     auto* fake_network = NewFakeNetwork();
>     auto port_allocator =
>-        rtc::MakeUnique<cricket::BasicPortAllocator>(fake_network);
>+        absl::make_unique<cricket::BasicPortAllocator>(fake_network);
>     port_allocator->set_flags(cricket::PORTALLOCATOR_DISABLE_TCP |
>                               cricket::PORTALLOCATOR_DISABLE_RELAY);
>     port_allocator->set_step_delay(cricket::kMinimumStepDelay);
>-    auto observer = rtc::MakeUnique<MockPeerConnectionObserver>();
>+    auto observer = absl::make_unique<MockPeerConnectionObserver>();
>+    RTCConfiguration modified_config = config;
>+    modified_config.sdp_semantics = sdp_semantics_;
>     auto pc = pc_factory_->CreatePeerConnection(
>-        config, std::move(port_allocator), nullptr, observer.get());
>+        modified_config, std::move(port_allocator), nullptr, observer.get());
>     if (!pc) {
>       return nullptr;
>     }
> 
>-    auto wrapper = rtc::MakeUnique<PeerConnectionWrapperForBundleTest>(
>+    auto wrapper = absl::make_unique<PeerConnectionWrapperForBundleTest>(
>         pc_factory_, pc, std::move(observer));
>     wrapper->set_network(fake_network);
>     return wrapper;
>@@ -214,6 +243,21 @@ class PeerConnectionBundleTest : public ::testing::Test {
>   rtc::AutoSocketServerThread main_;
>   rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory_;
>   std::vector<std::unique_ptr<rtc::FakeNetworkManager>> fake_networks_;
>+  const SdpSemantics sdp_semantics_;
>+};
>+
>+class PeerConnectionBundleTest
>+    : public PeerConnectionBundleBaseTest,
>+      public ::testing::WithParamInterface<SdpSemantics> {
>+ protected:
>+  PeerConnectionBundleTest() : PeerConnectionBundleBaseTest(GetParam()) {}
>+};
>+
>+class PeerConnectionBundleTestUnifiedPlan
>+    : public PeerConnectionBundleBaseTest {
>+ protected:
>+  PeerConnectionBundleTestUnifiedPlan()
>+      : PeerConnectionBundleBaseTest(SdpSemantics::kUnifiedPlan) {}
> };
> 
> SdpContentMutator RemoveRtcpMux() {
>@@ -233,7 +277,7 @@ std::vector<int> GetCandidateComponents(
> 
> // Test that there are 2 local UDP candidates (1 RTP and 1 RTCP candidate) for
> // each media section when disabling bundling and disabling RTCP multiplexing.
>-TEST_F(PeerConnectionBundleTest,
>+TEST_P(PeerConnectionBundleTest,
>        TwoCandidatesForEachTransportWhenNoBundleNoRtcpMux) {
>   const SocketAddress kCallerAddress("1.1.1.1", 0);
>   const SocketAddress kCalleeAddress("2.2.2.2", 0);
>@@ -279,7 +323,7 @@ TEST_F(PeerConnectionBundleTest,
> 
> // Test that there is 1 local UDP candidate for both RTP and RTCP for each media
> // section when disabling bundle but enabling RTCP multiplexing.
>-TEST_F(PeerConnectionBundleTest,
>+TEST_P(PeerConnectionBundleTest,
>        OneCandidateForEachTransportWhenNoBundleButRtcpMux) {
>   const SocketAddress kCallerAddress("1.1.1.1", 0);
> 
>@@ -301,7 +345,7 @@ TEST_F(PeerConnectionBundleTest,
> 
> // Test that there is 1 local UDP candidate in only the first media section when
> // bundling and enabling RTCP multiplexing.
>-TEST_F(PeerConnectionBundleTest,
>+TEST_P(PeerConnectionBundleTest,
>        OneCandidateOnlyOnFirstTransportWhenBundleAndRtcpMux) {
>   const SocketAddress kCallerAddress("1.1.1.1", 0);
> 
>@@ -320,6 +364,24 @@ TEST_F(PeerConnectionBundleTest,
>   EXPECT_EQ(0u, caller->observer()->GetCandidatesByMline(1).size());
> }
> 
>+// It will fail if the offerer uses the mux-BUNDLE policy but the answerer
>+// doesn't support BUNDLE.
>+TEST_P(PeerConnectionBundleTest, MaxBundleNotSupportedInAnswer) {
>+  RTCConfiguration config;
>+  config.bundle_policy = BundlePolicy::kBundlePolicyMaxBundle;
>+  auto caller = CreatePeerConnectionWithAudioVideo(config);
>+  auto callee = CreatePeerConnectionWithAudioVideo();
>+
>+  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
>+  bool equal_before =
>+      (caller->voice_rtp_transport() == caller->video_rtp_transport());
>+  EXPECT_EQ(true, equal_before);
>+  RTCOfferAnswerOptions options;
>+  options.use_rtp_mux = false;
>+  EXPECT_FALSE(
>+      caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal(options)));
>+}
>+
> // The following parameterized test verifies that an offer/answer with varying
> // bundle policies and either bundle in the answer or not will produce the
> // expected RTP transports for audio and video. In particular, for bundling we
>@@ -337,15 +399,18 @@ std::ostream& operator<<(std::ostream& out, BundleIncluded value) {
> }
> 
> class PeerConnectionBundleMatrixTest
>-    : public PeerConnectionBundleTest,
>+    : public PeerConnectionBundleBaseTest,
>       public ::testing::WithParamInterface<
>-          std::tuple<BundlePolicy, BundleIncluded, bool, bool>> {
>+          std::tuple<SdpSemantics,
>+                     std::tuple<BundlePolicy, BundleIncluded, bool, bool>>> {
>  protected:
>-  PeerConnectionBundleMatrixTest() {
>-    bundle_policy_ = std::get<0>(GetParam());
>-    bundle_included_ = std::get<1>(GetParam());
>-    expected_same_before_ = std::get<2>(GetParam());
>-    expected_same_after_ = std::get<3>(GetParam());
>+  PeerConnectionBundleMatrixTest()
>+      : PeerConnectionBundleBaseTest(std::get<0>(GetParam())) {
>+    auto param = std::get<1>(GetParam());
>+    bundle_policy_ = std::get<0>(param);
>+    bundle_included_ = std::get<1>(param);
>+    expected_same_before_ = std::get<2>(param);
>+    expected_same_after_ = std::get<3>(param);
>   }
> 
>   PeerConnectionInterface::BundlePolicy bundle_policy_;
>@@ -362,16 +427,16 @@ TEST_P(PeerConnectionBundleMatrixTest,
>   auto callee = CreatePeerConnectionWithAudioVideo();
> 
>   ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
>-  bool equal_before = (caller->voice_rtp_transport_channel() ==
>-                       caller->video_rtp_transport_channel());
>+  bool equal_before =
>+      (caller->voice_rtp_transport() == caller->video_rtp_transport());
>   EXPECT_EQ(expected_same_before_, equal_before);
> 
>   RTCOfferAnswerOptions options;
>   options.use_rtp_mux = (bundle_included_ == BundleIncluded::kBundleInAnswer);
>   ASSERT_TRUE(
>       caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal(options)));
>-  bool equal_after = (caller->voice_rtp_transport_channel() ==
>-                      caller->video_rtp_transport_channel());
>+  bool equal_after =
>+      (caller->voice_rtp_transport() == caller->video_rtp_transport());
>   EXPECT_EQ(expected_same_after_, equal_after);
> }
> 
>@@ -382,35 +447,32 @@ TEST_P(PeerConnectionBundleMatrixTest,
> INSTANTIATE_TEST_CASE_P(
>     PeerConnectionBundleTest,
>     PeerConnectionBundleMatrixTest,
>-    Values(std::make_tuple(BundlePolicy::kBundlePolicyBalanced,
>-                           BundleIncluded::kBundleInAnswer,
>-                           false,
>-                           true),
>-           std::make_tuple(BundlePolicy::kBundlePolicyBalanced,
>-                           BundleIncluded::kBundleNotInAnswer,
>-                           false,
>-                           false),
>-           std::make_tuple(BundlePolicy::kBundlePolicyMaxBundle,
>-                           BundleIncluded::kBundleInAnswer,
>-                           true,
>-                           true),
>-           std::make_tuple(BundlePolicy::kBundlePolicyMaxBundle,
>-                           BundleIncluded::kBundleNotInAnswer,
>-                           true,
>-                           true),
>-           std::make_tuple(BundlePolicy::kBundlePolicyMaxCompat,
>-                           BundleIncluded::kBundleInAnswer,
>-                           false,
>-                           true),
>-           std::make_tuple(BundlePolicy::kBundlePolicyMaxCompat,
>-                           BundleIncluded::kBundleNotInAnswer,
>-                           false,
>-                           false)));
>+    Combine(Values(SdpSemantics::kPlanB, SdpSemantics::kUnifiedPlan),
>+            Values(std::make_tuple(BundlePolicy::kBundlePolicyBalanced,
>+                                   BundleIncluded::kBundleInAnswer,
>+                                   false,
>+                                   true),
>+                   std::make_tuple(BundlePolicy::kBundlePolicyBalanced,
>+                                   BundleIncluded::kBundleNotInAnswer,
>+                                   false,
>+                                   false),
>+                   std::make_tuple(BundlePolicy::kBundlePolicyMaxBundle,
>+                                   BundleIncluded::kBundleInAnswer,
>+                                   true,
>+                                   true),
>+                   std::make_tuple(BundlePolicy::kBundlePolicyMaxCompat,
>+                                   BundleIncluded::kBundleInAnswer,
>+                                   false,
>+                                   true),
>+                   std::make_tuple(BundlePolicy::kBundlePolicyMaxCompat,
>+                                   BundleIncluded::kBundleNotInAnswer,
>+                                   false,
>+                                   false))));
> 
> // Test that the audio/video transports on the callee side are the same before
> // and after setting a local answer when max BUNDLE is enabled and an offer with
> // BUNDLE is received.
>-TEST_F(PeerConnectionBundleTest,
>+TEST_P(PeerConnectionBundleTest,
>        TransportsSameForMaxBundleWithBundleInRemoteOffer) {
>   auto caller = CreatePeerConnectionWithAudioVideo();
>   RTCConfiguration config;
>@@ -422,16 +484,14 @@ TEST_F(PeerConnectionBundleTest,
>   ASSERT_TRUE(callee->SetRemoteDescription(
>       caller->CreateOfferAndSetAsLocal(options_with_bundle)));
> 
>-  EXPECT_EQ(callee->voice_rtp_transport_channel(),
>-            callee->video_rtp_transport_channel());
>+  EXPECT_EQ(callee->voice_rtp_transport(), callee->video_rtp_transport());
> 
>   ASSERT_TRUE(callee->SetLocalDescription(callee->CreateAnswer()));
> 
>-  EXPECT_EQ(callee->voice_rtp_transport_channel(),
>-            callee->video_rtp_transport_channel());
>+  EXPECT_EQ(callee->voice_rtp_transport(), callee->video_rtp_transport());
> }
> 
>-TEST_F(PeerConnectionBundleTest,
>+TEST_P(PeerConnectionBundleTest,
>        FailToSetRemoteOfferWithNoBundleWhenBundlePolicyMaxBundle) {
>   auto caller = CreatePeerConnectionWithAudioVideo();
>   RTCConfiguration config;
>@@ -449,7 +509,7 @@ TEST_F(PeerConnectionBundleTest,
> // media section.
> // Note: This is currently failing because of the following bug:
> // https://bugs.chromium.org/p/webrtc/issues/detail?id=6280
>-TEST_F(PeerConnectionBundleTest,
>+TEST_P(PeerConnectionBundleTest,
>        DISABLED_SuccessfullyNegotiateMaxBundleIfBundleTransportMediaRejected) {
>   RTCConfiguration config;
>   config.bundle_policy = BundlePolicy::kBundlePolicyMaxBundle;
>@@ -464,13 +524,13 @@ TEST_F(PeerConnectionBundleTest,
>   ASSERT_TRUE(
>       caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal(options)));
> 
>-  EXPECT_FALSE(caller->voice_rtp_transport_channel());
>-  EXPECT_TRUE(caller->video_rtp_transport_channel());
>+  EXPECT_FALSE(caller->voice_rtp_transport());
>+  EXPECT_TRUE(caller->video_rtp_transport());
> }
> 
> // When requiring RTCP multiplexing, the PeerConnection never makes RTCP
> // transport channels.
>-TEST_F(PeerConnectionBundleTest, NeverCreateRtcpTransportWithRtcpMuxRequired) {
>+TEST_P(PeerConnectionBundleTest, NeverCreateRtcpTransportWithRtcpMuxRequired) {
>   RTCConfiguration config;
>   config.rtcp_mux_policy = RtcpMuxPolicy::kRtcpMuxPolicyRequire;
>   auto caller = CreatePeerConnectionWithAudioVideo(config);
>@@ -478,20 +538,19 @@ TEST_F(PeerConnectionBundleTest, NeverCreateRtcpTransportWithRtcpMuxRequired) {
> 
>   ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
> 
>-  EXPECT_FALSE(caller->voice_rtcp_transport_channel());
>-  EXPECT_FALSE(caller->video_rtcp_transport_channel());
>+  EXPECT_FALSE(caller->voice_rtcp_transport());
>+  EXPECT_FALSE(caller->video_rtcp_transport());
> 
>   ASSERT_TRUE(
>       caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));
> 
>-  EXPECT_FALSE(caller->voice_rtcp_transport_channel());
>-  EXPECT_FALSE(caller->video_rtcp_transport_channel());
>+  EXPECT_FALSE(caller->voice_rtcp_transport());
>+  EXPECT_FALSE(caller->video_rtcp_transport());
> }
> 
>-// When negotiating RTCP multiplexing, the PeerConnection makes RTCP transport
>-// channels when the offer is sent, but will destroy them once the remote answer
>-// is set.
>-TEST_F(PeerConnectionBundleTest,
>+// When negotiating RTCP multiplexing, the PeerConnection makes RTCP transports
>+// when the offer is sent, but will destroy them once the remote answer is set.
>+TEST_P(PeerConnectionBundleTest,
>        CreateRtcpTransportOnlyBeforeAnswerWithRtcpMuxNegotiate) {
>   RTCConfiguration config;
>   config.rtcp_mux_policy = RtcpMuxPolicy::kRtcpMuxPolicyNegotiate;
>@@ -500,17 +559,17 @@ TEST_F(PeerConnectionBundleTest,
> 
>   ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
> 
>-  EXPECT_TRUE(caller->voice_rtcp_transport_channel());
>-  EXPECT_TRUE(caller->video_rtcp_transport_channel());
>+  EXPECT_TRUE(caller->voice_rtcp_transport());
>+  EXPECT_TRUE(caller->video_rtcp_transport());
> 
>   ASSERT_TRUE(
>       caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));
> 
>-  EXPECT_FALSE(caller->voice_rtcp_transport_channel());
>-  EXPECT_FALSE(caller->video_rtcp_transport_channel());
>+  EXPECT_FALSE(caller->voice_rtcp_transport());
>+  EXPECT_FALSE(caller->video_rtcp_transport());
> }
> 
>-TEST_F(PeerConnectionBundleTest, FailToSetDescriptionWithBundleAndNoRtcpMux) {
>+TEST_P(PeerConnectionBundleTest, FailToSetDescriptionWithBundleAndNoRtcpMux) {
>   auto caller = CreatePeerConnectionWithAudioVideo();
>   auto callee = CreatePeerConnectionWithAudioVideo();
> 
>@@ -537,7 +596,7 @@ TEST_F(PeerConnectionBundleTest, FailToSetDescriptionWithBundleAndNoRtcpMux) {
> 
> // Test that candidates sent to the "video" transport do not get pushed down to
> // the "audio" transport channel when bundling.
>-TEST_F(PeerConnectionBundleTest,
>+TEST_P(PeerConnectionBundleTest,
>        IgnoreCandidatesForUnusedTransportWhenBundling) {
>   const SocketAddress kAudioAddress1("1.1.1.1", 1111);
>   const SocketAddress kAudioAddress2("2.2.2.2", 2222);
>@@ -556,7 +615,7 @@ TEST_F(PeerConnectionBundleTest,
> 
>   ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
>   ASSERT_TRUE(
>-      caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));
>+      caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal(options)));
> 
>   // The way the *_WAIT checks work is they only wait if the condition fails,
>   // which does not help in the case where state is not changing. This is
>@@ -586,31 +645,241 @@ TEST_F(PeerConnectionBundleTest,
> // Test that the transport used by both audio and video is the transport
> // associated with the first MID in the answer BUNDLE group, even if it's in a
> // different order from the offer.
>-TEST_F(PeerConnectionBundleTest, BundleOnFirstMidInAnswer) {
>+TEST_P(PeerConnectionBundleTest, BundleOnFirstMidInAnswer) {
>   auto caller = CreatePeerConnectionWithAudioVideo();
>   auto callee = CreatePeerConnectionWithAudioVideo();
> 
>   ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
> 
>-  auto* old_video_transport = caller->video_rtp_transport_channel();
>+  auto* old_video_transport = caller->video_rtp_transport();
> 
>   auto answer = callee->CreateAnswer();
>   auto* old_bundle_group =
>       answer->description()->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
>-  ASSERT_THAT(old_bundle_group->content_names(),
>-              ElementsAre(cricket::CN_AUDIO, cricket::CN_VIDEO));
>+  std::string first_mid = old_bundle_group->content_names()[0];
>+  std::string second_mid = old_bundle_group->content_names()[1];
>   answer->description()->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE);
> 
>   cricket::ContentGroup new_bundle_group(cricket::GROUP_TYPE_BUNDLE);
>-  new_bundle_group.AddContentName(cricket::CN_VIDEO);
>-  new_bundle_group.AddContentName(cricket::CN_AUDIO);
>+  new_bundle_group.AddContentName(second_mid);
>+  new_bundle_group.AddContentName(first_mid);
>   answer->description()->AddGroup(new_bundle_group);
> 
>   ASSERT_TRUE(caller->SetRemoteDescription(std::move(answer)));
> 
>-  EXPECT_EQ(old_video_transport, caller->video_rtp_transport_channel());
>-  EXPECT_EQ(caller->voice_rtp_transport_channel(),
>-            caller->video_rtp_transport_channel());
>+  EXPECT_EQ(old_video_transport, caller->video_rtp_transport());
>+  EXPECT_EQ(caller->voice_rtp_transport(), caller->video_rtp_transport());
>+}
>+
>+// This tests that applying description with conflicted RTP demuxing criteria
>+// will fail.
>+TEST_P(PeerConnectionBundleTest,
>+       ApplyDescriptionWithConflictedDemuxCriteriaFail) {
>+  auto caller = CreatePeerConnectionWithAudioVideo();
>+  auto callee = CreatePeerConnectionWithAudioVideo();
>+
>+  RTCOfferAnswerOptions options;
>+  options.use_rtp_mux = false;
>+  auto offer = caller->CreateOffer(options);
>+  // Modified the SDP to make two m= sections have the same SSRC.
>+  ASSERT_GE(offer->description()->contents().size(), 2U);
>+  offer->description()
>+      ->contents()[0]
>+      .description->mutable_streams()[0]
>+      .ssrcs[0] = 1111222;
>+  offer->description()
>+      ->contents()[1]
>+      .description->mutable_streams()[0]
>+      .ssrcs[0] = 1111222;
>+  EXPECT_TRUE(
>+      caller->SetLocalDescription(CloneSessionDescription(offer.get())));
>+  EXPECT_TRUE(callee->SetRemoteDescription(std::move(offer)));
>+  EXPECT_TRUE(callee->CreateAnswerAndSetAsLocal(options));
>+
>+  // Enable BUNDLE in subsequent offer/answer exchange and two m= sections are
>+  // expectd to use one RtpTransport underneath.
>+  options.use_rtp_mux = true;
>+  EXPECT_TRUE(
>+      callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(options)));
>+  auto answer = callee->CreateAnswer(options);
>+  // When BUNDLE is enabled, applying the description is expected to fail
>+  // because the demuxing criteria is conflicted.
>+  EXPECT_FALSE(callee->SetLocalDescription(std::move(answer)));
>+}
>+
>+// This tests that changing the pre-negotiated BUNDLE tag is not supported.
>+TEST_P(PeerConnectionBundleTest, RejectDescriptionChangingBundleTag) {
>+  RTCConfiguration config;
>+  config.bundle_policy = BundlePolicy::kBundlePolicyMaxBundle;
>+  auto caller = CreatePeerConnectionWithAudioVideo(config);
>+  auto callee = CreatePeerConnectionWithAudioVideo(config);
>+
>+  RTCOfferAnswerOptions options;
>+  options.use_rtp_mux = true;
>+  auto offer = caller->CreateOfferAndSetAsLocal(options);
>+
>+  // Create a new bundle-group with different bundled_mid.
>+  auto* old_bundle_group =
>+      offer->description()->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
>+  std::string first_mid = old_bundle_group->content_names()[0];
>+  std::string second_mid = old_bundle_group->content_names()[1];
>+  cricket::ContentGroup new_bundle_group(cricket::GROUP_TYPE_BUNDLE);
>+  new_bundle_group.AddContentName(second_mid);
>+
>+  auto re_offer = CloneSessionDescription(offer.get());
>+  callee->SetRemoteDescription(std::move(offer));
>+  auto answer = callee->CreateAnswer(options);
>+  // Reject the first MID.
>+  answer->description()->contents()[0].rejected = true;
>+  // Remove the first MID from the bundle group.
>+  answer->description()->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE);
>+  answer->description()->AddGroup(new_bundle_group);
>+  // The answer is expected to be rejected.
>+  EXPECT_FALSE(caller->SetRemoteDescription(std::move(answer)));
>+
>+  // Do the same thing for re-offer.
>+  re_offer->description()->contents()[0].rejected = true;
>+  re_offer->description()->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE);
>+  re_offer->description()->AddGroup(new_bundle_group);
>+  // The re-offer is expected to be rejected.
>+  EXPECT_FALSE(caller->SetLocalDescription(std::move(re_offer)));
>+}
>+
>+// This tests that removing contents from BUNDLE group and reject the whole
>+// BUNDLE group could work. This is a regression test for
>+// (https://bugs.chromium.org/p/chromium/issues/detail?id=827917)
>+TEST_P(PeerConnectionBundleTest, RemovingContentAndRejectBundleGroup) {
>+  RTCConfiguration config;
>+#ifndef HAVE_SCTP
>+  config.enable_rtp_data_channel = true;
>+#endif
>+  config.bundle_policy = BundlePolicy::kBundlePolicyMaxBundle;
>+  auto caller = CreatePeerConnectionWithAudioVideo(config);
>+  caller->CreateDataChannel("dc");
>+
>+  auto offer = caller->CreateOfferAndSetAsLocal();
>+  auto re_offer = CloneSessionDescription(offer.get());
>+
>+  // Removing the second MID from the BUNDLE group.
>+  auto* old_bundle_group =
>+      offer->description()->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
>+  std::string first_mid = old_bundle_group->content_names()[0];
>+  std::string third_mid = old_bundle_group->content_names()[2];
>+  cricket::ContentGroup new_bundle_group(cricket::GROUP_TYPE_BUNDLE);
>+  new_bundle_group.AddContentName(first_mid);
>+  new_bundle_group.AddContentName(third_mid);
>+
>+  // Reject the entire new bundle group.
>+  re_offer->description()->contents()[0].rejected = true;
>+  re_offer->description()->contents()[2].rejected = true;
>+  re_offer->description()->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE);
>+  re_offer->description()->AddGroup(new_bundle_group);
>+
>+  EXPECT_TRUE(caller->SetLocalDescription(std::move(re_offer)));
>+}
>+
>+// This tests that the BUNDLE group in answer should be a subset of the offered
>+// group.
>+TEST_P(PeerConnectionBundleTest, AddContentToBundleGroupInAnswerNotSupported) {
>+  auto caller = CreatePeerConnectionWithAudioVideo();
>+  auto callee = CreatePeerConnectionWithAudioVideo();
>+
>+  auto offer = caller->CreateOffer();
>+  std::string first_mid = offer->description()->contents()[0].name;
>+  std::string second_mid = offer->description()->contents()[1].name;
>+
>+  cricket::ContentGroup bundle_group(cricket::GROUP_TYPE_BUNDLE);
>+  bundle_group.AddContentName(first_mid);
>+  offer->description()->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE);
>+  offer->description()->AddGroup(bundle_group);
>+  EXPECT_TRUE(
>+      caller->SetLocalDescription(CloneSessionDescription(offer.get())));
>+  EXPECT_TRUE(callee->SetRemoteDescription(std::move(offer)));
>+
>+  auto answer = callee->CreateAnswer();
>+  bundle_group.AddContentName(second_mid);
>+  answer->description()->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE);
>+  answer->description()->AddGroup(bundle_group);
>+
>+  // The answer is expected to be rejected because second mid is not in the
>+  // offered BUNDLE group.
>+  EXPECT_FALSE(callee->SetLocalDescription(std::move(answer)));
>+}
>+
>+// This tests that the BUNDLE group with non-existing MID should be rejectd.
>+TEST_P(PeerConnectionBundleTest, RejectBundleGroupWithNonExistingMid) {
>+  auto caller = CreatePeerConnectionWithAudioVideo();
>+  auto callee = CreatePeerConnectionWithAudioVideo();
>+
>+  auto offer = caller->CreateOffer();
>+  auto invalid_bundle_group =
>+      *offer->description()->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
>+  invalid_bundle_group.AddContentName("non-existing-MID");
>+  offer->description()->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE);
>+  offer->description()->AddGroup(invalid_bundle_group);
>+
>+  EXPECT_FALSE(
>+      caller->SetLocalDescription(CloneSessionDescription(offer.get())));
>+  EXPECT_FALSE(callee->SetRemoteDescription(std::move(offer)));
>+}
>+
>+// This tests that an answer shouldn't be able to remove an m= section from an
>+// established group without rejecting it.
>+TEST_P(PeerConnectionBundleTest, RemoveContentFromBundleGroup) {
>+  auto caller = CreatePeerConnectionWithAudioVideo();
>+  auto callee = CreatePeerConnectionWithAudioVideo();
>+
>+  EXPECT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
>+  EXPECT_TRUE(
>+      caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));
>+
>+  EXPECT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
>+  auto answer = callee->CreateAnswer();
>+  std::string second_mid = answer->description()->contents()[1].name;
>+
>+  auto invalid_bundle_group =
>+      *answer->description()->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
>+  invalid_bundle_group.RemoveContentName(second_mid);
>+  answer->description()->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE);
>+  answer->description()->AddGroup(invalid_bundle_group);
>+
>+  EXPECT_FALSE(
>+      callee->SetLocalDescription(CloneSessionDescription(answer.get())));
>+}
>+
>+INSTANTIATE_TEST_CASE_P(PeerConnectionBundleTest,
>+                        PeerConnectionBundleTest,
>+                        Values(SdpSemantics::kPlanB,
>+                               SdpSemantics::kUnifiedPlan));
>+
>+// According to RFC5888, if an endpoint understands the semantics of an
>+// "a=group", it MUST return an answer with that group. So, an empty BUNDLE
>+// group is valid when the answerer rejects all m= sections (by stopping all
>+// transceivers), meaning there's nothing to bundle.
>+//
>+// Only writing this test for Unified Plan mode, since there's no way to reject
>+// m= sections in answers for Plan B without SDP munging.
>+TEST_F(PeerConnectionBundleTestUnifiedPlan,
>+       EmptyBundleGroupCreatedInAnswerWhenAppropriate) {
>+  auto caller = CreatePeerConnectionWithAudioVideo();
>+  auto callee = CreatePeerConnection();
>+
>+  EXPECT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
>+
>+  // Stop all transceivers, causing all m= sections to be rejected.
>+  for (const auto& transceiver : callee->pc()->GetTransceivers()) {
>+    transceiver->Stop();
>+  }
>+  EXPECT_TRUE(
>+      caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));
>+
>+  // Verify that the answer actually contained an empty bundle group.
>+  const SessionDescriptionInterface* desc = callee->pc()->local_description();
>+  ASSERT_NE(nullptr, desc);
>+  const cricket::ContentGroup* bundle_group =
>+      desc->description()->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
>+  ASSERT_NE(nullptr, bundle_group);
>+  EXPECT_TRUE(bundle_group->content_names().empty());
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_crypto_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_crypto_unittest.cc
>index eb447d68b16..5a8b5b8238c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_crypto_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_crypto_unittest.cc
>@@ -10,6 +10,8 @@
> 
> #include "api/audio_codecs/builtin_audio_decoder_factory.h"
> #include "api/audio_codecs/builtin_audio_encoder_factory.h"
>+#include "api/video_codecs/builtin_video_decoder_factory.h"
>+#include "api/video_codecs/builtin_video_encoder_factory.h"
> #include "p2p/base/fakeportallocator.h"
> #include "pc/mediasession.h"
> #include "pc/peerconnectionwrapper.h"
>@@ -17,10 +19,10 @@
> #ifdef WEBRTC_ANDROID
> #include "pc/test/androidtestinitializer.h"
> #endif
>+#include "absl/memory/memory.h"
> #include "pc/test/fakeaudiocapturemodule.h"
> #include "pc/test/fakertccertificategenerator.h"
> #include "rtc_base/gunit.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/virtualsocketserver.h"
> 
> namespace webrtc {
>@@ -32,19 +34,23 @@ using ::testing::Combine;
> 
> constexpr int kGenerateCertTimeout = 1000;
> 
>-class PeerConnectionCryptoUnitTest : public ::testing::Test {
>+class PeerConnectionCryptoBaseTest : public ::testing::Test {
>  protected:
>   typedef std::unique_ptr<PeerConnectionWrapper> WrapperPtr;
> 
>-  PeerConnectionCryptoUnitTest()
>-      : vss_(new rtc::VirtualSocketServer()), main_(vss_.get()) {
>+  explicit PeerConnectionCryptoBaseTest(SdpSemantics sdp_semantics)
>+      : vss_(new rtc::VirtualSocketServer()),
>+        main_(vss_.get()),
>+        sdp_semantics_(sdp_semantics) {
> #ifdef WEBRTC_ANDROID
>     InitializeAndroidObjects();
> #endif
>     pc_factory_ = CreatePeerConnectionFactory(
>         rtc::Thread::Current(), rtc::Thread::Current(), rtc::Thread::Current(),
>         FakeAudioCaptureModule::Create(), CreateBuiltinAudioEncoderFactory(),
>-        CreateBuiltinAudioDecoderFactory(), nullptr, nullptr);
>+        CreateBuiltinAudioDecoderFactory(), CreateBuiltinVideoEncoderFactory(),
>+        CreateBuiltinVideoDecoderFactory(), nullptr /* audio_mixer */,
>+        nullptr /* audio_processing */);
>   }
> 
>   WrapperPtr CreatePeerConnection() {
>@@ -58,18 +64,20 @@ class PeerConnectionCryptoUnitTest : public ::testing::Test {
>   WrapperPtr CreatePeerConnection(
>       const RTCConfiguration& config,
>       std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_gen) {
>-    auto fake_port_allocator = rtc::MakeUnique<cricket::FakePortAllocator>(
>+    auto fake_port_allocator = absl::make_unique<cricket::FakePortAllocator>(
>         rtc::Thread::Current(), nullptr);
>-    auto observer = rtc::MakeUnique<MockPeerConnectionObserver>();
>+    auto observer = absl::make_unique<MockPeerConnectionObserver>();
>+    RTCConfiguration modified_config = config;
>+    modified_config.sdp_semantics = sdp_semantics_;
>     auto pc = pc_factory_->CreatePeerConnection(
>-        config, std::move(fake_port_allocator), std::move(cert_gen),
>+        modified_config, std::move(fake_port_allocator), std::move(cert_gen),
>         observer.get());
>     if (!pc) {
>       return nullptr;
>     }
> 
>-    return rtc::MakeUnique<PeerConnectionWrapper>(pc_factory_, pc,
>-                                                  std::move(observer));
>+    return absl::make_unique<PeerConnectionWrapper>(pc_factory_, pc,
>+                                                    std::move(observer));
>   }
> 
>   // Accepts the same arguments as CreatePeerConnection and adds default audio
>@@ -107,6 +115,7 @@ class PeerConnectionCryptoUnitTest : public ::testing::Test {
>   std::unique_ptr<rtc::VirtualSocketServer> vss_;
>   rtc::AutoSocketServerThread main_;
>   rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory_;
>+  const SdpSemantics sdp_semantics_;
> };
> 
> SdpContentPredicate HaveDtlsFingerprint() {
>@@ -143,6 +152,13 @@ SdpContentPredicate HaveSdesGcmCryptos(size_t num_crypto_suites) {
>   };
> }
> 
>+class PeerConnectionCryptoTest
>+    : public PeerConnectionCryptoBaseTest,
>+      public ::testing::WithParamInterface<SdpSemantics> {
>+ protected:
>+  PeerConnectionCryptoTest() : PeerConnectionCryptoBaseTest(GetParam()) {}
>+};
>+
> SdpContentMutator RemoveSdesCryptos() {
>   return [](cricket::ContentInfo* content, cricket::TransportInfo* transport) {
>     content->media_description()->set_cryptos({});
>@@ -157,7 +173,7 @@ SdpContentMutator RemoveDtlsFingerprint() {
> 
> // When DTLS is enabled, the SDP offer/answer should have a DTLS fingerprint and
> // no SDES cryptos.
>-TEST_F(PeerConnectionCryptoUnitTest, CorrectCryptoInOfferWhenDtlsEnabled) {
>+TEST_P(PeerConnectionCryptoTest, CorrectCryptoInOfferWhenDtlsEnabled) {
>   RTCConfiguration config;
>   config.enable_dtls_srtp.emplace(true);
>   auto caller = CreatePeerConnectionWithAudioVideo(config);
>@@ -171,7 +187,7 @@ TEST_F(PeerConnectionCryptoUnitTest, CorrectCryptoInOfferWhenDtlsEnabled) {
>   EXPECT_TRUE(SdpContentsAll(HaveProtocol(cricket::kMediaProtocolDtlsSavpf),
>                              offer->description()));
> }
>-TEST_F(PeerConnectionCryptoUnitTest, CorrectCryptoInAnswerWhenDtlsEnabled) {
>+TEST_P(PeerConnectionCryptoTest, CorrectCryptoInAnswerWhenDtlsEnabled) {
>   RTCConfiguration config;
>   config.enable_dtls_srtp.emplace(true);
>   auto caller = CreatePeerConnectionWithAudioVideo(config);
>@@ -190,7 +206,7 @@ TEST_F(PeerConnectionCryptoUnitTest, CorrectCryptoInAnswerWhenDtlsEnabled) {
> 
> // When DTLS is disabled, the SDP offer/answer should include SDES cryptos and
> // should not have a DTLS fingerprint.
>-TEST_F(PeerConnectionCryptoUnitTest, CorrectCryptoInOfferWhenDtlsDisabled) {
>+TEST_P(PeerConnectionCryptoTest, CorrectCryptoInOfferWhenDtlsDisabled) {
>   RTCConfiguration config;
>   config.enable_dtls_srtp.emplace(false);
>   auto caller = CreatePeerConnectionWithAudioVideo(config);
>@@ -204,7 +220,7 @@ TEST_F(PeerConnectionCryptoUnitTest, CorrectCryptoInOfferWhenDtlsDisabled) {
>   EXPECT_TRUE(SdpContentsAll(HaveProtocol(cricket::kMediaProtocolSavpf),
>                              offer->description()));
> }
>-TEST_F(PeerConnectionCryptoUnitTest, CorrectCryptoInAnswerWhenDtlsDisabled) {
>+TEST_P(PeerConnectionCryptoTest, CorrectCryptoInAnswerWhenDtlsDisabled) {
>   RTCConfiguration config;
>   config.enable_dtls_srtp.emplace(false);
>   auto caller = CreatePeerConnectionWithAudioVideo(config);
>@@ -223,8 +239,7 @@ TEST_F(PeerConnectionCryptoUnitTest, CorrectCryptoInAnswerWhenDtlsDisabled) {
> 
> // When encryption is disabled, the SDP offer/answer should have neither a DTLS
> // fingerprint nor any SDES crypto options.
>-TEST_F(PeerConnectionCryptoUnitTest,
>-       CorrectCryptoInOfferWhenEncryptionDisabled) {
>+TEST_P(PeerConnectionCryptoTest, CorrectCryptoInOfferWhenEncryptionDisabled) {
>   PeerConnectionFactoryInterface::Options options;
>   options.disable_encryption = true;
>   pc_factory_->SetOptions(options);
>@@ -242,8 +257,7 @@ TEST_F(PeerConnectionCryptoUnitTest,
>   EXPECT_TRUE(SdpContentsAll(HaveProtocol(cricket::kMediaProtocolAvpf),
>                              offer->description()));
> }
>-TEST_F(PeerConnectionCryptoUnitTest,
>-       CorrectCryptoInAnswerWhenEncryptionDisabled) {
>+TEST_P(PeerConnectionCryptoTest, CorrectCryptoInAnswerWhenEncryptionDisabled) {
>   PeerConnectionFactoryInterface::Options options;
>   options.disable_encryption = true;
>   pc_factory_->SetOptions(options);
>@@ -268,7 +282,7 @@ TEST_F(PeerConnectionCryptoUnitTest,
> // should have the correct ciphers in the SDES crypto options.
> // With GCM cipher suites enabled, there will be 3 cryptos in the offer and 1
> // in the answer.
>-TEST_F(PeerConnectionCryptoUnitTest, CorrectCryptoInOfferWithSdesAndGcm) {
>+TEST_P(PeerConnectionCryptoTest, CorrectCryptoInOfferWithSdesAndGcm) {
>   PeerConnectionFactoryInterface::Options options;
>   options.crypto_options.enable_gcm_crypto_suites = true;
>   pc_factory_->SetOptions(options);
>@@ -283,7 +297,7 @@ TEST_F(PeerConnectionCryptoUnitTest, CorrectCryptoInOfferWithSdesAndGcm) {
>   ASSERT_FALSE(offer->description()->contents().empty());
>   EXPECT_TRUE(SdpContentsAll(HaveSdesGcmCryptos(3), offer->description()));
> }
>-TEST_F(PeerConnectionCryptoUnitTest, CorrectCryptoInAnswerWithSdesAndGcm) {
>+TEST_P(PeerConnectionCryptoTest, CorrectCryptoInAnswerWithSdesAndGcm) {
>   PeerConnectionFactoryInterface::Options options;
>   options.crypto_options.enable_gcm_crypto_suites = true;
>   pc_factory_->SetOptions(options);
>@@ -301,7 +315,7 @@ TEST_F(PeerConnectionCryptoUnitTest, CorrectCryptoInAnswerWithSdesAndGcm) {
>   EXPECT_TRUE(SdpContentsAll(HaveSdesGcmCryptos(1), answer->description()));
> }
> 
>-TEST_F(PeerConnectionCryptoUnitTest, CanSetSdesGcmRemoteOfferAndLocalAnswer) {
>+TEST_P(PeerConnectionCryptoTest, CanSetSdesGcmRemoteOfferAndLocalAnswer) {
>   PeerConnectionFactoryInterface::Options options;
>   options.crypto_options.enable_gcm_crypto_suites = true;
>   pc_factory_->SetOptions(options);
>@@ -323,7 +337,7 @@ TEST_F(PeerConnectionCryptoUnitTest, CanSetSdesGcmRemoteOfferAndLocalAnswer) {
> // The following group tests that two PeerConnections can successfully exchange
> // an offer/answer when DTLS is off and that they will refuse any offer/answer
> // applied locally/remotely if it does not include SDES cryptos.
>-TEST_F(PeerConnectionCryptoUnitTest, ExchangeOfferAnswerWhenSdesOn) {
>+TEST_P(PeerConnectionCryptoTest, ExchangeOfferAnswerWhenSdesOn) {
>   RTCConfiguration config;
>   config.enable_dtls_srtp.emplace(false);
>   auto caller = CreatePeerConnectionWithAudioVideo(config);
>@@ -337,8 +351,7 @@ TEST_F(PeerConnectionCryptoUnitTest, ExchangeOfferAnswerWhenSdesOn) {
>   ASSERT_TRUE(answer);
>   ASSERT_TRUE(caller->SetRemoteDescription(std::move(answer)));
> }
>-TEST_F(PeerConnectionCryptoUnitTest,
>-       FailToSetLocalOfferWithNoCryptosWhenSdesOn) {
>+TEST_P(PeerConnectionCryptoTest, FailToSetLocalOfferWithNoCryptosWhenSdesOn) {
>   RTCConfiguration config;
>   config.enable_dtls_srtp.emplace(false);
>   auto caller = CreatePeerConnectionWithAudioVideo(config);
>@@ -348,8 +361,7 @@ TEST_F(PeerConnectionCryptoUnitTest,
> 
>   EXPECT_FALSE(caller->SetLocalDescription(std::move(offer)));
> }
>-TEST_F(PeerConnectionCryptoUnitTest,
>-       FailToSetRemoteOfferWithNoCryptosWhenSdesOn) {
>+TEST_P(PeerConnectionCryptoTest, FailToSetRemoteOfferWithNoCryptosWhenSdesOn) {
>   RTCConfiguration config;
>   config.enable_dtls_srtp.emplace(false);
>   auto caller = CreatePeerConnectionWithAudioVideo(config);
>@@ -360,8 +372,7 @@ TEST_F(PeerConnectionCryptoUnitTest,
> 
>   EXPECT_FALSE(callee->SetRemoteDescription(std::move(offer)));
> }
>-TEST_F(PeerConnectionCryptoUnitTest,
>-       FailToSetLocalAnswerWithNoCryptosWhenSdesOn) {
>+TEST_P(PeerConnectionCryptoTest, FailToSetLocalAnswerWithNoCryptosWhenSdesOn) {
>   RTCConfiguration config;
>   config.enable_dtls_srtp.emplace(false);
>   auto caller = CreatePeerConnectionWithAudioVideo(config);
>@@ -373,8 +384,7 @@ TEST_F(PeerConnectionCryptoUnitTest,
> 
>   EXPECT_FALSE(callee->SetLocalDescription(std::move(answer)));
> }
>-TEST_F(PeerConnectionCryptoUnitTest,
>-       FailToSetRemoteAnswerWithNoCryptosWhenSdesOn) {
>+TEST_P(PeerConnectionCryptoTest, FailToSetRemoteAnswerWithNoCryptosWhenSdesOn) {
>   RTCConfiguration config;
>   config.enable_dtls_srtp.emplace(false);
>   auto caller = CreatePeerConnectionWithAudioVideo(config);
>@@ -390,7 +400,7 @@ TEST_F(PeerConnectionCryptoUnitTest,
> // The following group tests that two PeerConnections can successfully exchange
> // an offer/answer when DTLS is on and that they will refuse any offer/answer
> // applied locally/remotely if it does not include a DTLS fingerprint.
>-TEST_F(PeerConnectionCryptoUnitTest, ExchangeOfferAnswerWhenDtlsOn) {
>+TEST_P(PeerConnectionCryptoTest, ExchangeOfferAnswerWhenDtlsOn) {
>   RTCConfiguration config;
>   config.enable_dtls_srtp.emplace(true);
>   auto caller = CreatePeerConnectionWithAudioVideo(config);
>@@ -404,7 +414,7 @@ TEST_F(PeerConnectionCryptoUnitTest, ExchangeOfferAnswerWhenDtlsOn) {
>   ASSERT_TRUE(answer);
>   ASSERT_TRUE(caller->SetRemoteDescription(std::move(answer)));
> }
>-TEST_F(PeerConnectionCryptoUnitTest,
>+TEST_P(PeerConnectionCryptoTest,
>        FailToSetLocalOfferWithNoFingerprintWhenDtlsOn) {
>   RTCConfiguration config;
>   config.enable_dtls_srtp.emplace(true);
>@@ -415,7 +425,7 @@ TEST_F(PeerConnectionCryptoUnitTest,
> 
>   EXPECT_FALSE(caller->SetLocalDescription(std::move(offer)));
> }
>-TEST_F(PeerConnectionCryptoUnitTest,
>+TEST_P(PeerConnectionCryptoTest,
>        FailToSetRemoteOfferWithNoFingerprintWhenDtlsOn) {
>   RTCConfiguration config;
>   config.enable_dtls_srtp.emplace(true);
>@@ -427,7 +437,7 @@ TEST_F(PeerConnectionCryptoUnitTest,
> 
>   EXPECT_FALSE(callee->SetRemoteDescription(std::move(offer)));
> }
>-TEST_F(PeerConnectionCryptoUnitTest,
>+TEST_P(PeerConnectionCryptoTest,
>        FailToSetLocalAnswerWithNoFingerprintWhenDtlsOn) {
>   RTCConfiguration config;
>   config.enable_dtls_srtp.emplace(true);
>@@ -438,7 +448,7 @@ TEST_F(PeerConnectionCryptoUnitTest,
>   auto answer = callee->CreateAnswer();
>   SdpContentsForEach(RemoveDtlsFingerprint(), answer->description());
> }
>-TEST_F(PeerConnectionCryptoUnitTest,
>+TEST_P(PeerConnectionCryptoTest,
>        FailToSetRemoteAnswerWithNoFingerprintWhenDtlsOn) {
>   RTCConfiguration config;
>   config.enable_dtls_srtp.emplace(true);
>@@ -453,7 +463,7 @@ TEST_F(PeerConnectionCryptoUnitTest,
> }
> 
> // Test that an offer/answer can be exchanged when encryption is disabled.
>-TEST_F(PeerConnectionCryptoUnitTest, ExchangeOfferAnswerWhenNoEncryption) {
>+TEST_P(PeerConnectionCryptoTest, ExchangeOfferAnswerWhenNoEncryption) {
>   PeerConnectionFactoryInterface::Options options;
>   options.disable_encryption = true;
>   pc_factory_->SetOptions(options);
>@@ -474,7 +484,7 @@ TEST_F(PeerConnectionCryptoUnitTest, ExchangeOfferAnswerWhenNoEncryption) {
> 
> // Tests that a DTLS call can be established when the certificate is specified
> // in the PeerConnection config and no certificate generator is specified.
>-TEST_F(PeerConnectionCryptoUnitTest,
>+TEST_P(PeerConnectionCryptoTest,
>        ExchangeOfferAnswerWhenDtlsCertificateInConfig) {
>   RTCConfiguration caller_config;
>   caller_config.enable_dtls_srtp.emplace(true);
>@@ -530,16 +540,20 @@ std::ostream& operator<<(std::ostream& out, CertGenResult value) {
>   }
> }
> 
>-class PeerConnectionCryptoDtlsCertGenUnitTest
>-    : public PeerConnectionCryptoUnitTest,
>-      public ::testing::WithParamInterface<
>-          ::testing::tuple<SdpType, CertGenTime, CertGenResult, size_t>> {
>+class PeerConnectionCryptoDtlsCertGenTest
>+    : public PeerConnectionCryptoBaseTest,
>+      public ::testing::WithParamInterface<std::tuple<SdpSemantics,
>+                                                      SdpType,
>+                                                      CertGenTime,
>+                                                      CertGenResult,
>+                                                      size_t>> {
>  protected:
>-  PeerConnectionCryptoDtlsCertGenUnitTest() {
>-    sdp_type_ = ::testing::get<0>(GetParam());
>-    cert_gen_time_ = ::testing::get<1>(GetParam());
>-    cert_gen_result_ = ::testing::get<2>(GetParam());
>-    concurrent_calls_ = ::testing::get<3>(GetParam());
>+  PeerConnectionCryptoDtlsCertGenTest()
>+      : PeerConnectionCryptoBaseTest(std::get<0>(GetParam())) {
>+    sdp_type_ = std::get<1>(GetParam());
>+    cert_gen_time_ = std::get<2>(GetParam());
>+    cert_gen_result_ = std::get<3>(GetParam());
>+    concurrent_calls_ = std::get<4>(GetParam());
>   }
> 
>   SdpType sdp_type_;
>@@ -548,11 +562,11 @@ class PeerConnectionCryptoDtlsCertGenUnitTest
>   size_t concurrent_calls_;
> };
> 
>-TEST_P(PeerConnectionCryptoDtlsCertGenUnitTest, TestCertificateGeneration) {
>+TEST_P(PeerConnectionCryptoDtlsCertGenTest, TestCertificateGeneration) {
>   RTCConfiguration config;
>   config.enable_dtls_srtp.emplace(true);
>   auto owned_fake_certificate_generator =
>-      rtc::MakeUnique<FakeRTCCertificateGenerator>();
>+      absl::make_unique<FakeRTCCertificateGenerator>();
>   auto* fake_certificate_generator = owned_fake_certificate_generator.get();
>   fake_certificate_generator->set_should_fail(cert_gen_result_ ==
>                                               CertGenResult::kFail);
>@@ -600,9 +614,10 @@ TEST_P(PeerConnectionCryptoDtlsCertGenUnitTest, TestCertificateGeneration) {
> }
> 
> INSTANTIATE_TEST_CASE_P(
>-    PeerConnectionCryptoUnitTest,
>-    PeerConnectionCryptoDtlsCertGenUnitTest,
>-    Combine(Values(SdpType::kOffer, SdpType::kAnswer),
>+    PeerConnectionCryptoTest,
>+    PeerConnectionCryptoDtlsCertGenTest,
>+    Combine(Values(SdpSemantics::kPlanB, SdpSemantics::kUnifiedPlan),
>+            Values(SdpType::kOffer, SdpType::kAnswer),
>             Values(CertGenTime::kBefore, CertGenTime::kDuring),
>             Values(CertGenResult::kSucceed, CertGenResult::kFail),
>             Values(1, 3)));
>@@ -610,7 +625,7 @@ INSTANTIATE_TEST_CASE_P(
> // Test that we can create and set an answer correctly when different
> // SSL roles have been negotiated for different transports.
> // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4525
>-TEST_F(PeerConnectionCryptoUnitTest, CreateAnswerWithDifferentSslRoles) {
>+TEST_P(PeerConnectionCryptoTest, CreateAnswerWithDifferentSslRoles) {
>   auto caller = CreatePeerConnectionWithAudioVideo();
>   auto callee = CreatePeerConnectionWithAudioVideo();
> 
>@@ -661,4 +676,52 @@ TEST_F(PeerConnectionCryptoUnitTest, CreateAnswerWithDifferentSslRoles) {
>   ASSERT_TRUE(callee->SetRemoteDescription(std::move(answer)));
> }
> 
>+// Tests that if the DTLS fingerprint is invalid then all future calls to
>+// SetLocalDescription and SetRemoteDescription will fail due to a session
>+// error.
>+// This is a regression test for crbug.com/800775
>+TEST_P(PeerConnectionCryptoTest, SessionErrorIfFingerprintInvalid) {
>+  auto callee_certificate = rtc::RTCCertificate::FromPEM(kRsaPems[0]);
>+  auto other_certificate = rtc::RTCCertificate::FromPEM(kRsaPems[1]);
>+
>+  auto caller = CreatePeerConnectionWithAudioVideo();
>+  RTCConfiguration callee_config;
>+  callee_config.enable_dtls_srtp.emplace(true);
>+  callee_config.certificates.push_back(callee_certificate);
>+  auto callee = CreatePeerConnectionWithAudioVideo(callee_config);
>+
>+  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
>+
>+  // Create an invalid answer with the other certificate's fingerprint.
>+  auto invalid_answer = callee->CreateAnswer();
>+  auto* audio_content =
>+      cricket::GetFirstAudioContent(invalid_answer->description());
>+  ASSERT_TRUE(audio_content);
>+  auto* audio_transport_info =
>+      invalid_answer->description()->GetTransportInfoByName(
>+          audio_content->name);
>+  ASSERT_TRUE(audio_transport_info);
>+  audio_transport_info->description.identity_fingerprint.reset(
>+      rtc::SSLFingerprint::CreateFromCertificate(other_certificate));
>+
>+  // Set the invalid answer and expect a fingerprint error.
>+  std::string error;
>+  ASSERT_FALSE(callee->SetLocalDescription(std::move(invalid_answer), &error));
>+  EXPECT_PRED_FORMAT2(AssertStringContains, error,
>+                      "Local fingerprint does not match identity.");
>+
>+  // Make sure that setting a valid remote offer or local answer also fails now.
>+  ASSERT_FALSE(callee->SetRemoteDescription(caller->CreateOffer(), &error));
>+  EXPECT_PRED_FORMAT2(AssertStringContains, error,
>+                      "Session error code: ERROR_CONTENT.");
>+  ASSERT_FALSE(callee->SetLocalDescription(callee->CreateAnswer(), &error));
>+  EXPECT_PRED_FORMAT2(AssertStringContains, error,
>+                      "Session error code: ERROR_CONTENT.");
>+}
>+
>+INSTANTIATE_TEST_CASE_P(PeerConnectionCryptoTest,
>+                        PeerConnectionCryptoTest,
>+                        Values(SdpSemantics::kPlanB,
>+                               SdpSemantics::kUnifiedPlan));
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_datachannel_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_datachannel_unittest.cc
>index ca3a3f11ca4..033a609f98a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_datachannel_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_datachannel_unittest.cc
>@@ -16,12 +16,13 @@
> #include "pc/peerconnection.h"
> #include "pc/peerconnectionfactory.h"
> #include "pc/peerconnectionwrapper.h"
>+#include "pc/sdputils.h"
> #ifdef WEBRTC_ANDROID
> #include "pc/test/androidtestinitializer.h"
> #endif
>+#include "absl/memory/memory.h"
> #include "pc/test/fakesctptransport.h"
> #include "rtc_base/gunit.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/virtualsocketserver.h"
> 
> namespace webrtc {
>@@ -38,13 +39,13 @@ class PeerConnectionFactoryForDataChannelTest
>             rtc::Thread::Current(),
>             rtc::Thread::Current(),
>             rtc::Thread::Current(),
>-            rtc::MakeUnique<cricket::FakeMediaEngine>(),
>+            absl::make_unique<cricket::FakeMediaEngine>(),
>             CreateCallFactory(),
>             nullptr) {}
> 
>   std::unique_ptr<cricket::SctpTransportInternalFactory>
>   CreateSctpTransportInternalFactory() {
>-    auto factory = rtc::MakeUnique<FakeSctpTransportFactory>();
>+    auto factory = absl::make_unique<FakeSctpTransportFactory>();
>     last_fake_sctp_transport_factory_ = factory.get();
>     return factory;
>   }
>@@ -65,11 +66,11 @@ class PeerConnectionWrapperForDataChannelTest : public PeerConnectionWrapper {
>     sctp_transport_factory_ = sctp_transport_factory;
>   }
> 
>-  rtc::Optional<std::string> sctp_content_name() {
>+  absl::optional<std::string> sctp_content_name() {
>     return GetInternalPeerConnection()->sctp_content_name();
>   }
> 
>-  rtc::Optional<std::string> sctp_transport_name() {
>+  absl::optional<std::string> sctp_transport_name() {
>     return GetInternalPeerConnection()->sctp_transport_name();
>   }
> 
>@@ -84,12 +85,14 @@ class PeerConnectionWrapperForDataChannelTest : public PeerConnectionWrapper {
>   FakeSctpTransportFactory* sctp_transport_factory_ = nullptr;
> };
> 
>-class PeerConnectionDataChannelTest : public ::testing::Test {
>+class PeerConnectionDataChannelBaseTest : public ::testing::Test {
>  protected:
>   typedef std::unique_ptr<PeerConnectionWrapperForDataChannelTest> WrapperPtr;
> 
>-  PeerConnectionDataChannelTest()
>-      : vss_(new rtc::VirtualSocketServer()), main_(vss_.get()) {
>+  explicit PeerConnectionDataChannelBaseTest(SdpSemantics sdp_semantics)
>+      : vss_(new rtc::VirtualSocketServer()),
>+        main_(vss_.get()),
>+        sdp_semantics_(sdp_semantics) {
> #ifdef WEBRTC_ANDROID
>     InitializeAndroidObjects();
> #endif
>@@ -111,14 +114,16 @@ class PeerConnectionDataChannelTest : public ::testing::Test {
>         new PeerConnectionFactoryForDataChannelTest());
>     pc_factory->SetOptions(factory_options);
>     RTC_CHECK(pc_factory->Initialize());
>-    auto observer = rtc::MakeUnique<MockPeerConnectionObserver>();
>-    auto pc = pc_factory->CreatePeerConnection(config, nullptr, nullptr,
>-                                               observer.get());
>+    auto observer = absl::make_unique<MockPeerConnectionObserver>();
>+    RTCConfiguration modified_config = config;
>+    modified_config.sdp_semantics = sdp_semantics_;
>+    auto pc = pc_factory->CreatePeerConnection(modified_config, nullptr,
>+                                               nullptr, observer.get());
>     if (!pc) {
>       return nullptr;
>     }
> 
>-    auto wrapper = rtc::MakeUnique<PeerConnectionWrapperForDataChannelTest>(
>+    auto wrapper = absl::make_unique<PeerConnectionWrapperForDataChannelTest>(
>         pc_factory, pc, std::move(observer));
>     RTC_DCHECK(pc_factory->last_fake_sctp_transport_factory_);
>     wrapper->set_sctp_transport_factory(
>@@ -153,9 +158,18 @@ class PeerConnectionDataChannelTest : public ::testing::Test {
> 
>   std::unique_ptr<rtc::VirtualSocketServer> vss_;
>   rtc::AutoSocketServerThread main_;
>+  const SdpSemantics sdp_semantics_;
>+};
>+
>+class PeerConnectionDataChannelTest
>+    : public PeerConnectionDataChannelBaseTest,
>+      public ::testing::WithParamInterface<SdpSemantics> {
>+ protected:
>+  PeerConnectionDataChannelTest()
>+      : PeerConnectionDataChannelBaseTest(GetParam()) {}
> };
> 
>-TEST_F(PeerConnectionDataChannelTest,
>+TEST_P(PeerConnectionDataChannelTest,
>        NoSctpTransportCreatedIfRtpDataChannelEnabled) {
>   RTCConfiguration config;
>   config.enable_rtp_data_channel = true;
>@@ -165,7 +179,7 @@ TEST_F(PeerConnectionDataChannelTest,
>   EXPECT_FALSE(caller->sctp_transport_factory()->last_fake_sctp_transport());
> }
> 
>-TEST_F(PeerConnectionDataChannelTest,
>+TEST_P(PeerConnectionDataChannelTest,
>        RtpDataChannelCreatedEvenIfSctpAvailable) {
>   RTCConfiguration config;
>   config.enable_rtp_data_channel = true;
>@@ -179,7 +193,7 @@ TEST_F(PeerConnectionDataChannelTest,
> 
> // Test that sctp_content_name/sctp_transport_name (used for stats) are correct
> // before and after BUNDLE is negotiated.
>-TEST_F(PeerConnectionDataChannelTest, SctpContentAndTransportNameSetCorrectly) {
>+TEST_P(PeerConnectionDataChannelTest, SctpContentAndTransportNameSetCorrectly) {
>   auto caller = CreatePeerConnection();
>   auto callee = CreatePeerConnection();
> 
>@@ -193,12 +207,24 @@ TEST_F(PeerConnectionDataChannelTest, SctpContentAndTransportNameSetCorrectly) {
>   caller->AddAudioTrack("a");
>   caller->AddVideoTrack("v");
>   caller->pc()->CreateDataChannel("dc", nullptr);
>-  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
>+
>+  auto offer = caller->CreateOffer();
>+  const auto& offer_contents = offer->description()->contents();
>+  ASSERT_EQ(cricket::MEDIA_TYPE_AUDIO,
>+            offer_contents[0].media_description()->type());
>+  std::string audio_mid = offer_contents[0].name;
>+  ASSERT_EQ(cricket::MEDIA_TYPE_DATA,
>+            offer_contents[2].media_description()->type());
>+  std::string data_mid = offer_contents[2].name;
>+
>+  ASSERT_TRUE(
>+      caller->SetLocalDescription(CloneSessionDescription(offer.get())));
>+  ASSERT_TRUE(callee->SetRemoteDescription(std::move(offer)));
> 
>   ASSERT_TRUE(caller->sctp_content_name());
>-  EXPECT_EQ(cricket::CN_DATA, *caller->sctp_content_name());
>+  EXPECT_EQ(data_mid, *caller->sctp_content_name());
>   ASSERT_TRUE(caller->sctp_transport_name());
>-  EXPECT_EQ(cricket::CN_DATA, *caller->sctp_transport_name());
>+  EXPECT_EQ(data_mid, *caller->sctp_transport_name());
> 
>   // Create answer that finishes BUNDLE negotiation, which means everything
>   // should be bundled on the first transport (audio).
>@@ -208,12 +234,12 @@ TEST_F(PeerConnectionDataChannelTest, SctpContentAndTransportNameSetCorrectly) {
>       caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));
> 
>   ASSERT_TRUE(caller->sctp_content_name());
>-  EXPECT_EQ(cricket::CN_DATA, *caller->sctp_content_name());
>+  EXPECT_EQ(data_mid, *caller->sctp_content_name());
>   ASSERT_TRUE(caller->sctp_transport_name());
>-  EXPECT_EQ(cricket::CN_AUDIO, *caller->sctp_transport_name());
>+  EXPECT_EQ(audio_mid, *caller->sctp_transport_name());
> }
> 
>-TEST_F(PeerConnectionDataChannelTest,
>+TEST_P(PeerConnectionDataChannelTest,
>        CreateOfferWithNoDataChannelsGivesNoDataSection) {
>   auto caller = CreatePeerConnection();
>   auto offer = caller->CreateOffer();
>@@ -222,7 +248,7 @@ TEST_F(PeerConnectionDataChannelTest,
>   EXPECT_FALSE(offer->description()->GetTransportInfoByName(cricket::CN_DATA));
> }
> 
>-TEST_F(PeerConnectionDataChannelTest,
>+TEST_P(PeerConnectionDataChannelTest,
>        CreateAnswerWithRemoteSctpDataChannelIncludesDataSection) {
>   auto caller = CreatePeerConnectionWithDataChannel();
>   auto callee = CreatePeerConnection();
>@@ -231,14 +257,14 @@ TEST_F(PeerConnectionDataChannelTest,
> 
>   auto answer = callee->CreateAnswer();
>   ASSERT_TRUE(answer);
>-  auto* data_content =
>-      answer->description()->GetContentByName(cricket::CN_DATA);
>+  auto* data_content = cricket::GetFirstDataContent(answer->description());
>   ASSERT_TRUE(data_content);
>   EXPECT_FALSE(data_content->rejected);
>-  EXPECT_TRUE(answer->description()->GetTransportInfoByName(cricket::CN_DATA));
>+  EXPECT_TRUE(
>+      answer->description()->GetTransportInfoByName(data_content->name));
> }
> 
>-TEST_F(PeerConnectionDataChannelTest,
>+TEST_P(PeerConnectionDataChannelTest,
>        CreateDataChannelWithDtlsDisabledSucceeds) {
>   RTCConfiguration config;
>   config.enable_dtls_srtp.emplace(false);
>@@ -247,7 +273,7 @@ TEST_F(PeerConnectionDataChannelTest,
>   EXPECT_TRUE(caller->pc()->CreateDataChannel("dc", nullptr));
> }
> 
>-TEST_F(PeerConnectionDataChannelTest, CreateDataChannelWithSctpDisabledFails) {
>+TEST_P(PeerConnectionDataChannelTest, CreateDataChannelWithSctpDisabledFails) {
>   PeerConnectionFactoryInterface::Options options;
>   options.disable_sctp_data_channels = true;
>   auto caller = CreatePeerConnection(RTCConfiguration(), options);
>@@ -258,7 +284,7 @@ TEST_F(PeerConnectionDataChannelTest, CreateDataChannelWithSctpDisabledFails) {
> // Test that if a callee has SCTP disabled and receives an offer with an SCTP
> // data channel, the data section is rejected and no SCTP transport is created
> // on the callee.
>-TEST_F(PeerConnectionDataChannelTest,
>+TEST_P(PeerConnectionDataChannelTest,
>        DataSectionRejectedIfCalleeHasSctpDisabled) {
>   auto caller = CreatePeerConnectionWithDataChannel();
>   PeerConnectionFactoryInterface::Options options;
>@@ -270,13 +296,12 @@ TEST_F(PeerConnectionDataChannelTest,
>   EXPECT_FALSE(callee->sctp_transport_factory()->last_fake_sctp_transport());
> 
>   auto answer = callee->CreateAnswer();
>-  auto* data_content =
>-      answer->description()->GetContentByName(cricket::CN_DATA);
>+  auto* data_content = cricket::GetFirstDataContent(answer->description());
>   ASSERT_TRUE(data_content);
>   EXPECT_TRUE(data_content->rejected);
> }
> 
>-TEST_F(PeerConnectionDataChannelTest, SctpPortPropagatedFromSdpToTransport) {
>+TEST_P(PeerConnectionDataChannelTest, SctpPortPropagatedFromSdpToTransport) {
>   constexpr int kNewSendPort = 9998;
>   constexpr int kNewRecvPort = 7775;
> 
>@@ -298,4 +323,9 @@ TEST_F(PeerConnectionDataChannelTest, SctpPortPropagatedFromSdpToTransport) {
>   EXPECT_EQ(kNewRecvPort, callee_transport->local_port());
> }
> 
>+INSTANTIATE_TEST_CASE_P(PeerConnectionDataChannelTest,
>+                        PeerConnectionDataChannelTest,
>+                        Values(SdpSemantics::kPlanB,
>+                               SdpSemantics::kUnifiedPlan));
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_histogram_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_histogram_unittest.cc
>new file mode 100644
>index 00000000000..f1203e4cb49
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_histogram_unittest.cc
>@@ -0,0 +1,511 @@
>+/*
>+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <tuple>
>+
>+#include "absl/memory/memory.h"
>+#include "api/jsep.h"
>+#include "api/peerconnectionproxy.h"
>+#include "media/base/fakemediaengine.h"
>+#include "p2p/client/basicportallocator.h"
>+#include "pc/mediasession.h"
>+#include "pc/peerconnection.h"
>+#include "pc/peerconnectionfactory.h"
>+#include "pc/peerconnectionwrapper.h"
>+#include "pc/sdputils.h"
>+#include "pc/test/fakesctptransport.h"
>+#include "rtc_base/fakenetwork.h"
>+#include "rtc_base/gunit.h"
>+#include "rtc_base/virtualsocketserver.h"
>+#include "system_wrappers/include/metrics_default.h"
>+
>+namespace webrtc {
>+
>+using RTCConfiguration = PeerConnectionInterface::RTCConfiguration;
>+using RTCOfferAnswerOptions = PeerConnectionInterface::RTCOfferAnswerOptions;
>+using ::testing::Values;
>+
>+static const char kUsagePatternMetric[] = "WebRTC.PeerConnection.UsagePattern";
>+static constexpr int kDefaultTimeout = 10000;
>+static const rtc::SocketAddress kDefaultLocalAddress("1.1.1.1", 0);
>+static const rtc::SocketAddress kPrivateLocalAddress("10.1.1.1", 0);
>+
>+int MakeUsageFingerprint(std::set<PeerConnection::UsageEvent> events) {
>+  int signature = 0;
>+  for (const auto it : events) {
>+    signature |= static_cast<int>(it);
>+  }
>+  return signature;
>+}
>+
>+class PeerConnectionFactoryForUsageHistogramTest
>+    : public rtc::RefCountedObject<PeerConnectionFactory> {
>+ public:
>+  PeerConnectionFactoryForUsageHistogramTest()
>+      : rtc::RefCountedObject<PeerConnectionFactory>(
>+            rtc::Thread::Current(),
>+            rtc::Thread::Current(),
>+            rtc::Thread::Current(),
>+            absl::make_unique<cricket::FakeMediaEngine>(),
>+            CreateCallFactory(),
>+            nullptr) {}
>+
>+  void ActionsBeforeInitializeForTesting(PeerConnectionInterface* pc) override {
>+    PeerConnection* internal_pc = static_cast<PeerConnection*>(pc);
>+    if (return_histogram_very_quickly_) {
>+      internal_pc->ReturnHistogramVeryQuicklyForTesting();
>+    }
>+  }
>+
>+  void ReturnHistogramVeryQuickly() { return_histogram_very_quickly_ = true; }
>+
>+ private:
>+  bool return_histogram_very_quickly_ = false;
>+};
>+
>+class PeerConnectionWrapperForUsageHistogramTest;
>+typedef PeerConnectionWrapperForUsageHistogramTest* RawWrapperPtr;
>+
>+class ObserverForUsageHistogramTest : public MockPeerConnectionObserver {
>+ public:
>+  void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override;
>+
>+  void OnInterestingUsage(int usage_pattern) override {
>+    interesting_usage_detected_ = usage_pattern;
>+  }
>+
>+  void PrepareToExchangeCandidates(RawWrapperPtr other) {
>+    candidate_target_ = other;
>+  }
>+
>+  bool HaveDataChannel() { return last_datachannel_; }
>+
>+  absl::optional<int> interesting_usage_detected() {
>+    return interesting_usage_detected_;
>+  }
>+
>+  void ClearInterestingUsageDetector() {
>+    interesting_usage_detected_ = absl::optional<int>();
>+  }
>+
>+ private:
>+  absl::optional<int> interesting_usage_detected_;
>+  RawWrapperPtr candidate_target_;  // Note: Not thread-safe against deletions.
>+};
>+
>+class PeerConnectionWrapperForUsageHistogramTest
>+    : public PeerConnectionWrapper {
>+ public:
>+  using PeerConnectionWrapper::PeerConnectionWrapper;
>+
>+  PeerConnection* GetInternalPeerConnection() {
>+    auto* pci =
>+        static_cast<PeerConnectionProxyWithInternal<PeerConnectionInterface>*>(
>+            pc());
>+    return static_cast<PeerConnection*>(pci->internal());
>+  }
>+
>+  // Override with different return type
>+  ObserverForUsageHistogramTest* observer() {
>+    return static_cast<ObserverForUsageHistogramTest*>(
>+        PeerConnectionWrapper::observer());
>+  }
>+
>+  void PrepareToExchangeCandidates(
>+      PeerConnectionWrapperForUsageHistogramTest* other) {
>+    observer()->PrepareToExchangeCandidates(other);
>+    other->observer()->PrepareToExchangeCandidates(this);
>+  }
>+
>+  bool IsConnected() {
>+    return pc()->ice_connection_state() ==
>+               PeerConnectionInterface::kIceConnectionConnected ||
>+           pc()->ice_connection_state() ==
>+               PeerConnectionInterface::kIceConnectionCompleted;
>+  }
>+
>+  bool HaveDataChannel() {
>+    return static_cast<ObserverForUsageHistogramTest*>(observer())
>+        ->HaveDataChannel();
>+  }
>+  void AddOrBufferIceCandidate(const webrtc::IceCandidateInterface* candidate) {
>+    if (!pc()->AddIceCandidate(candidate)) {
>+      std::string sdp;
>+      EXPECT_TRUE(candidate->ToString(&sdp));
>+      std::unique_ptr<webrtc::IceCandidateInterface> candidate_copy(
>+          CreateIceCandidate(candidate->sdp_mid(), candidate->sdp_mline_index(),
>+                             sdp, nullptr));
>+      buffered_candidates_.push_back(std::move(candidate_copy));
>+    }
>+  }
>+
>+  void AddBufferedIceCandidates() {
>+    for (const auto& candidate : buffered_candidates_) {
>+      EXPECT_TRUE(pc()->AddIceCandidate(candidate.get()));
>+    }
>+    buffered_candidates_.clear();
>+  }
>+
>+  bool ConnectTo(PeerConnectionWrapperForUsageHistogramTest* callee) {
>+    PrepareToExchangeCandidates(callee);
>+    if (!ExchangeOfferAnswerWith(callee)) {
>+      return false;
>+    }
>+    AddBufferedIceCandidates();
>+    callee->AddBufferedIceCandidates();
>+    WAIT(IsConnected(), kDefaultTimeout);
>+    WAIT(callee->IsConnected(), kDefaultTimeout);
>+    return IsConnected() && callee->IsConnected();
>+  }
>+
>+  bool GenerateOfferAndCollectCandidates() {
>+    auto offer = CreateOffer(RTCOfferAnswerOptions());
>+    if (!offer) {
>+      return false;
>+    }
>+    bool set_local_offer =
>+        SetLocalDescription(CloneSessionDescription(offer.get()));
>+    EXPECT_TRUE(set_local_offer);
>+    if (!set_local_offer) {
>+      return false;
>+    }
>+    EXPECT_TRUE_WAIT(observer()->ice_gathering_complete_, kDefaultTimeout);
>+    return true;
>+  }
>+
>+ private:
>+  // Candidates that have been sent but not yet configured
>+  std::vector<std::unique_ptr<webrtc::IceCandidateInterface>>
>+      buffered_candidates_;
>+};
>+
>+void ObserverForUsageHistogramTest::OnIceCandidate(
>+    const webrtc::IceCandidateInterface* candidate) {
>+  if (candidate_target_) {
>+    this->candidate_target_->AddOrBufferIceCandidate(candidate);
>+  }
>+  // If target is not set, ignore. This happens in one-ended unit tests.
>+}
>+
>+class PeerConnectionUsageHistogramTest : public ::testing::Test {
>+ protected:
>+  typedef std::unique_ptr<PeerConnectionWrapperForUsageHistogramTest>
>+      WrapperPtr;
>+
>+  PeerConnectionUsageHistogramTest()
>+      : vss_(new rtc::VirtualSocketServer()), main_(vss_.get()) {
>+    webrtc::metrics::Reset();
>+  }
>+
>+  WrapperPtr CreatePeerConnection() {
>+    return CreatePeerConnection(RTCConfiguration(),
>+                                PeerConnectionFactoryInterface::Options(),
>+                                nullptr, false);
>+  }
>+
>+  WrapperPtr CreatePeerConnection(const RTCConfiguration& config) {
>+    return CreatePeerConnection(
>+        config, PeerConnectionFactoryInterface::Options(), nullptr, false);
>+  }
>+
>+  WrapperPtr CreatePeerConnectionWithImmediateReport() {
>+    return CreatePeerConnection(RTCConfiguration(),
>+                                PeerConnectionFactoryInterface::Options(),
>+                                nullptr, true);
>+  }
>+
>+  WrapperPtr CreatePeerConnectionWithPrivateLocalAddresses() {
>+    fake_network_manager_.reset(new rtc::FakeNetworkManager());
>+    fake_network_manager_->AddInterface(kDefaultLocalAddress);
>+    fake_network_manager_->AddInterface(kPrivateLocalAddress);
>+    std::unique_ptr<cricket::BasicPortAllocator> port_allocator(
>+        new cricket::BasicPortAllocator(fake_network_manager_.get()));
>+    return CreatePeerConnection(RTCConfiguration(),
>+                                PeerConnectionFactoryInterface::Options(),
>+                                std::move(port_allocator), false);
>+  }
>+
>+  WrapperPtr CreatePeerConnection(
>+      const RTCConfiguration& config,
>+      const PeerConnectionFactoryInterface::Options factory_options,
>+      std::unique_ptr<cricket::PortAllocator> allocator,
>+      bool immediate_report) {
>+    rtc::scoped_refptr<PeerConnectionFactoryForUsageHistogramTest> pc_factory(
>+        new PeerConnectionFactoryForUsageHistogramTest());
>+    pc_factory->SetOptions(factory_options);
>+    RTC_CHECK(pc_factory->Initialize());
>+    if (immediate_report) {
>+      pc_factory->ReturnHistogramVeryQuickly();
>+    }
>+    auto observer = absl::make_unique<ObserverForUsageHistogramTest>();
>+    auto pc = pc_factory->CreatePeerConnection(config, std::move(allocator),
>+                                               nullptr, observer.get());
>+    if (!pc) {
>+      return nullptr;
>+    }
>+
>+    auto wrapper =
>+        absl::make_unique<PeerConnectionWrapperForUsageHistogramTest>(
>+            pc_factory, pc, std::move(observer));
>+    return wrapper;
>+  }
>+
>+  int ObservedFingerprint() {
>+    // This works correctly only if there is only one sample value
>+    // that has been counted.
>+    // Returns -1 for "not found".
>+    return webrtc::metrics::MinSample(kUsagePatternMetric);
>+  }
>+
>+  std::unique_ptr<rtc::FakeNetworkManager> fake_network_manager_;
>+  std::unique_ptr<rtc::VirtualSocketServer> vss_;
>+  rtc::AutoSocketServerThread main_;
>+};
>+
>+TEST_F(PeerConnectionUsageHistogramTest, UsageFingerprintHistogramFromTimeout) {
>+  auto pc = CreatePeerConnectionWithImmediateReport();
>+
>+  int expected_fingerprint = MakeUsageFingerprint({});
>+  ASSERT_EQ_WAIT(1, webrtc::metrics::NumSamples(kUsagePatternMetric),
>+                 kDefaultTimeout);
>+  EXPECT_EQ(
>+      1, webrtc::metrics::NumEvents(kUsagePatternMetric, expected_fingerprint));
>+}
>+
>+#ifndef WEBRTC_ANDROID
>+// These tests do not work on Android. Why is unclear.
>+// https://bugs.webrtc.org/9461
>+
>+// Test getting the usage fingerprint for an audio/video connection.
>+TEST_F(PeerConnectionUsageHistogramTest, FingerprintAudioVideo) {
>+  auto caller = CreatePeerConnection();
>+  auto callee = CreatePeerConnection();
>+  caller->AddAudioTrack("audio");
>+  caller->AddVideoTrack("video");
>+  ASSERT_TRUE(caller->ConnectTo(callee.get()));
>+  caller->pc()->Close();
>+  callee->pc()->Close();
>+  int expected_fingerprint = MakeUsageFingerprint(
>+      {PeerConnection::UsageEvent::AUDIO_ADDED,
>+       PeerConnection::UsageEvent::VIDEO_ADDED,
>+       PeerConnection::UsageEvent::SET_LOCAL_DESCRIPTION_CALLED,
>+       PeerConnection::UsageEvent::SET_REMOTE_DESCRIPTION_CALLED,
>+       PeerConnection::UsageEvent::CANDIDATE_COLLECTED,
>+       PeerConnection::UsageEvent::REMOTE_CANDIDATE_ADDED,
>+       PeerConnection::UsageEvent::ICE_STATE_CONNECTED,
>+       PeerConnection::UsageEvent::CLOSE_CALLED});
>+  // In this case, we may or may not have PRIVATE_CANDIDATE_COLLECTED,
>+  // depending on the machine configuration.
>+  EXPECT_EQ(2, webrtc::metrics::NumSamples(kUsagePatternMetric));
>+  EXPECT_TRUE(
>+      webrtc::metrics::NumEvents(kUsagePatternMetric, expected_fingerprint) ==
>+          2 ||
>+      webrtc::metrics::NumEvents(
>+          kUsagePatternMetric,
>+          expected_fingerprint |
>+              static_cast<int>(
>+                  PeerConnection::UsageEvent::PRIVATE_CANDIDATE_COLLECTED)) ==
>+          2);
>+}
>+
>+// Test getting the usage fingerprint when there are no host candidates.
>+TEST_F(PeerConnectionUsageHistogramTest, FingerprintWithNoHostCandidates) {
>+  RTCConfiguration config;
>+  config.type = PeerConnectionInterface::kNoHost;
>+  auto caller = CreatePeerConnection(config);
>+  auto callee = CreatePeerConnection(config);
>+  caller->AddAudioTrack("audio");
>+  caller->AddVideoTrack("video");
>+  // Under some bot configurations, this will fail - presumably bots where
>+  // no working non-host addresses exist.
>+  if (!caller->ConnectTo(callee.get())) {
>+    return;
>+  }
>+  // If we manage to connect, we should get this precise fingerprint.
>+  caller->pc()->Close();
>+  callee->pc()->Close();
>+  int expected_fingerprint = MakeUsageFingerprint(
>+      {PeerConnection::UsageEvent::AUDIO_ADDED,
>+       PeerConnection::UsageEvent::VIDEO_ADDED,
>+       PeerConnection::UsageEvent::SET_LOCAL_DESCRIPTION_CALLED,
>+       PeerConnection::UsageEvent::SET_REMOTE_DESCRIPTION_CALLED,
>+       PeerConnection::UsageEvent::CANDIDATE_COLLECTED,
>+       PeerConnection::UsageEvent::REMOTE_CANDIDATE_ADDED,
>+       PeerConnection::UsageEvent::ICE_STATE_CONNECTED,
>+       PeerConnection::UsageEvent::CLOSE_CALLED});
>+  EXPECT_EQ(2, webrtc::metrics::NumSamples(kUsagePatternMetric));
>+  EXPECT_EQ(
>+      2, webrtc::metrics::NumEvents(kUsagePatternMetric, expected_fingerprint));
>+}
>+
>+#ifdef HAVE_SCTP
>+TEST_F(PeerConnectionUsageHistogramTest, FingerprintDataOnly) {
>+  auto caller = CreatePeerConnection();
>+  auto callee = CreatePeerConnection();
>+  caller->CreateDataChannel("foodata");
>+  ASSERT_TRUE(caller->ConnectTo(callee.get()));
>+  ASSERT_TRUE_WAIT(callee->HaveDataChannel(), kDefaultTimeout);
>+  caller->pc()->Close();
>+  callee->pc()->Close();
>+  int expected_fingerprint = MakeUsageFingerprint(
>+      {PeerConnection::UsageEvent::DATA_ADDED,
>+       PeerConnection::UsageEvent::SET_LOCAL_DESCRIPTION_CALLED,
>+       PeerConnection::UsageEvent::SET_REMOTE_DESCRIPTION_CALLED,
>+       PeerConnection::UsageEvent::CANDIDATE_COLLECTED,
>+       PeerConnection::UsageEvent::REMOTE_CANDIDATE_ADDED,
>+       PeerConnection::UsageEvent::ICE_STATE_CONNECTED,
>+       PeerConnection::UsageEvent::CLOSE_CALLED});
>+  EXPECT_EQ(2, webrtc::metrics::NumSamples(kUsagePatternMetric));
>+  EXPECT_TRUE(
>+      webrtc::metrics::NumEvents(kUsagePatternMetric, expected_fingerprint) ==
>+          2 ||
>+      webrtc::metrics::NumEvents(
>+          kUsagePatternMetric,
>+          expected_fingerprint |
>+              static_cast<int>(
>+                  PeerConnection::UsageEvent::PRIVATE_CANDIDATE_COLLECTED)) ==
>+          2);
>+}
>+#endif  // HAVE_SCTP
>+#endif  // WEBRTC_ANDROID
>+
>+TEST_F(PeerConnectionUsageHistogramTest, FingerprintStunTurn) {
>+  RTCConfiguration configuration;
>+  PeerConnection::IceServer server;
>+  server.urls = {"stun:dummy.stun.server/"};
>+  configuration.servers.push_back(server);
>+  server.urls = {"turn:dummy.turn.server/"};
>+  server.username = "username";
>+  server.password = "password";
>+  configuration.servers.push_back(server);
>+  auto caller = CreatePeerConnection(configuration);
>+  ASSERT_TRUE(caller);
>+  caller->pc()->Close();
>+  int expected_fingerprint =
>+      MakeUsageFingerprint({PeerConnection::UsageEvent::STUN_SERVER_ADDED,
>+                            PeerConnection::UsageEvent::TURN_SERVER_ADDED,
>+                            PeerConnection::UsageEvent::CLOSE_CALLED});
>+  EXPECT_EQ(1, webrtc::metrics::NumSamples(kUsagePatternMetric));
>+  EXPECT_EQ(
>+      1, webrtc::metrics::NumEvents(kUsagePatternMetric, expected_fingerprint));
>+}
>+
>+TEST_F(PeerConnectionUsageHistogramTest, FingerprintStunTurnInReconfiguration) {
>+  RTCConfiguration configuration;
>+  PeerConnection::IceServer server;
>+  server.urls = {"stun:dummy.stun.server/"};
>+  configuration.servers.push_back(server);
>+  server.urls = {"turn:dummy.turn.server/"};
>+  server.username = "username";
>+  server.password = "password";
>+  configuration.servers.push_back(server);
>+  auto caller = CreatePeerConnection();
>+  ASSERT_TRUE(caller);
>+  RTCError error;
>+  caller->pc()->SetConfiguration(configuration, &error);
>+  ASSERT_TRUE(error.ok());
>+  caller->pc()->Close();
>+  int expected_fingerprint =
>+      MakeUsageFingerprint({PeerConnection::UsageEvent::STUN_SERVER_ADDED,
>+                            PeerConnection::UsageEvent::TURN_SERVER_ADDED,
>+                            PeerConnection::UsageEvent::CLOSE_CALLED});
>+  EXPECT_EQ(1, webrtc::metrics::NumSamples(kUsagePatternMetric));
>+  EXPECT_EQ(
>+      1, webrtc::metrics::NumEvents(kUsagePatternMetric, expected_fingerprint));
>+}
>+
>+TEST_F(PeerConnectionUsageHistogramTest, FingerprintWithPrivateIP) {
>+  auto caller = CreatePeerConnectionWithPrivateLocalAddresses();
>+  caller->AddAudioTrack("audio");
>+  ASSERT_TRUE(caller->GenerateOfferAndCollectCandidates());
>+  caller->pc()->Close();
>+  int expected_fingerprint = MakeUsageFingerprint(
>+      {PeerConnection::UsageEvent::AUDIO_ADDED,
>+       PeerConnection::UsageEvent::SET_LOCAL_DESCRIPTION_CALLED,
>+       PeerConnection::UsageEvent::CANDIDATE_COLLECTED,
>+       PeerConnection::UsageEvent::CLOSE_CALLED,
>+       PeerConnection::UsageEvent::PRIVATE_CANDIDATE_COLLECTED});
>+  EXPECT_EQ(1, webrtc::metrics::NumSamples(kUsagePatternMetric));
>+  EXPECT_EQ(
>+      1, webrtc::metrics::NumEvents(kUsagePatternMetric, expected_fingerprint));
>+}
>+
>+#ifndef WEBRTC_ANDROID
>+#ifdef HAVE_SCTP
>+TEST_F(PeerConnectionUsageHistogramTest, NotableUsageNoted) {
>+  auto caller = CreatePeerConnection();
>+  caller->CreateDataChannel("foo");
>+  caller->GenerateOfferAndCollectCandidates();
>+  caller->pc()->Close();
>+  int expected_fingerprint = MakeUsageFingerprint(
>+      {PeerConnection::UsageEvent::DATA_ADDED,
>+       PeerConnection::UsageEvent::SET_LOCAL_DESCRIPTION_CALLED,
>+       PeerConnection::UsageEvent::CANDIDATE_COLLECTED,
>+       PeerConnection::UsageEvent::CLOSE_CALLED});
>+  EXPECT_EQ(1, webrtc::metrics::NumSamples(kUsagePatternMetric));
>+  EXPECT_TRUE(expected_fingerprint == ObservedFingerprint() ||
>+              (expected_fingerprint |
>+               static_cast<int>(
>+                   PeerConnection::UsageEvent::PRIVATE_CANDIDATE_COLLECTED)) ==
>+                  ObservedFingerprint());
>+  EXPECT_EQ(absl::make_optional(ObservedFingerprint()),
>+            caller->observer()->interesting_usage_detected());
>+}
>+
>+TEST_F(PeerConnectionUsageHistogramTest, NotableUsageOnEventFiring) {
>+  auto caller = CreatePeerConnection();
>+  caller->CreateDataChannel("foo");
>+  caller->GenerateOfferAndCollectCandidates();
>+  int expected_fingerprint = MakeUsageFingerprint(
>+      {PeerConnection::UsageEvent::DATA_ADDED,
>+       PeerConnection::UsageEvent::SET_LOCAL_DESCRIPTION_CALLED,
>+       PeerConnection::UsageEvent::CANDIDATE_COLLECTED});
>+  EXPECT_EQ(0, webrtc::metrics::NumSamples(kUsagePatternMetric));
>+  caller->GetInternalPeerConnection()->RequestUsagePatternReportForTesting();
>+  EXPECT_EQ_WAIT(1, webrtc::metrics::NumSamples(kUsagePatternMetric),
>+                 kDefaultTimeout);
>+  EXPECT_TRUE(expected_fingerprint == ObservedFingerprint() ||
>+              (expected_fingerprint |
>+               static_cast<int>(
>+                   PeerConnection::UsageEvent::PRIVATE_CANDIDATE_COLLECTED)) ==
>+                  ObservedFingerprint());
>+  EXPECT_EQ(absl::make_optional(ObservedFingerprint()),
>+            caller->observer()->interesting_usage_detected());
>+}
>+
>+TEST_F(PeerConnectionUsageHistogramTest,
>+       NoNotableUsageOnEventFiringAfterClose) {
>+  auto caller = CreatePeerConnection();
>+  caller->CreateDataChannel("foo");
>+  caller->GenerateOfferAndCollectCandidates();
>+  int expected_fingerprint = MakeUsageFingerprint(
>+      {PeerConnection::UsageEvent::DATA_ADDED,
>+       PeerConnection::UsageEvent::SET_LOCAL_DESCRIPTION_CALLED,
>+       PeerConnection::UsageEvent::CANDIDATE_COLLECTED,
>+       PeerConnection::UsageEvent::CLOSE_CALLED});
>+  EXPECT_EQ(0, webrtc::metrics::NumSamples(kUsagePatternMetric));
>+  caller->pc()->Close();
>+  EXPECT_EQ(1, webrtc::metrics::NumSamples(kUsagePatternMetric));
>+  caller->GetInternalPeerConnection()->RequestUsagePatternReportForTesting();
>+  caller->observer()->ClearInterestingUsageDetector();
>+  EXPECT_EQ_WAIT(2, webrtc::metrics::NumSamples(kUsagePatternMetric),
>+                 kDefaultTimeout);
>+  EXPECT_TRUE(expected_fingerprint == ObservedFingerprint() ||
>+              (expected_fingerprint |
>+               static_cast<int>(
>+                   PeerConnection::UsageEvent::PRIVATE_CANDIDATE_COLLECTED)) ==
>+                  ObservedFingerprint());
>+  // After close, the usage-detection callback should NOT have been called.
>+  EXPECT_FALSE(caller->observer()->interesting_usage_detected());
>+}
>+#endif
>+#endif
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_ice_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_ice_unittest.cc
>index e6ba598e442..36e0a098dd4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_ice_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_ice_unittest.cc
>@@ -18,25 +18,29 @@
> #ifdef WEBRTC_ANDROID
> #include "pc/test/androidtestinitializer.h"
> #endif
>+#include "absl/memory/memory.h"
> #include "api/audio_codecs/builtin_audio_decoder_factory.h"
> #include "api/audio_codecs/builtin_audio_encoder_factory.h"
> #include "api/peerconnectionproxy.h"
>+#include "api/video_codecs/builtin_video_decoder_factory.h"
>+#include "api/video_codecs/builtin_video_encoder_factory.h"
> #include "pc/test/fakeaudiocapturemodule.h"
> #include "rtc_base/fakenetwork.h"
> #include "rtc_base/gunit.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/virtualsocketserver.h"
>+#include "system_wrappers/include/metrics_default.h"
> 
> namespace webrtc {
> 
> using RTCConfiguration = PeerConnectionInterface::RTCConfiguration;
> using RTCOfferAnswerOptions = PeerConnectionInterface::RTCOfferAnswerOptions;
> using rtc::SocketAddress;
>+using ::testing::Combine;
> using ::testing::Values;
> 
> constexpr int kIceCandidatesTimeout = 10000;
> 
>-class PeerConnectionWrapperForIceUnitTest : public PeerConnectionWrapper {
>+class PeerConnectionWrapperForIceTest : public PeerConnectionWrapper {
>  public:
>   using PeerConnectionWrapper::PeerConnectionWrapper;
> 
>@@ -47,8 +51,9 @@ class PeerConnectionWrapperForIceUnitTest : public PeerConnectionWrapper {
>     RTC_DCHECK(desc->contents().size() > 0);
>     const auto& first_content = desc->contents()[0];
>     candidate->set_transport_name(first_content.name);
>-    JsepIceCandidate jsep_candidate(first_content.name, 0, *candidate);
>-    return pc()->AddIceCandidate(&jsep_candidate);
>+    std::unique_ptr<IceCandidateInterface> jsep_candidate =
>+        CreateIceCandidate(first_content.name, 0, *candidate);
>+    return pc()->AddIceCandidate(jsep_candidate.get());
>   }
> 
>   // Returns ICE candidates from the remote session description.
>@@ -75,19 +80,23 @@ class PeerConnectionWrapperForIceUnitTest : public PeerConnectionWrapper {
>   rtc::FakeNetworkManager* network_;
> };
> 
>-class PeerConnectionIceUnitTest : public ::testing::Test {
>+class PeerConnectionIceBaseTest : public ::testing::Test {
>  protected:
>-  typedef std::unique_ptr<PeerConnectionWrapperForIceUnitTest> WrapperPtr;
>+  typedef std::unique_ptr<PeerConnectionWrapperForIceTest> WrapperPtr;
> 
>-  PeerConnectionIceUnitTest()
>-      : vss_(new rtc::VirtualSocketServer()), main_(vss_.get()) {
>+  explicit PeerConnectionIceBaseTest(SdpSemantics sdp_semantics)
>+      : vss_(new rtc::VirtualSocketServer()),
>+        main_(vss_.get()),
>+        sdp_semantics_(sdp_semantics) {
> #ifdef WEBRTC_ANDROID
>     InitializeAndroidObjects();
> #endif
>     pc_factory_ = CreatePeerConnectionFactory(
>         rtc::Thread::Current(), rtc::Thread::Current(), rtc::Thread::Current(),
>-        FakeAudioCaptureModule::Create(), CreateBuiltinAudioEncoderFactory(),
>-        CreateBuiltinAudioDecoderFactory(), nullptr, nullptr);
>+        rtc::scoped_refptr<AudioDeviceModule>(FakeAudioCaptureModule::Create()),
>+        CreateBuiltinAudioEncoderFactory(), CreateBuiltinAudioDecoderFactory(),
>+        CreateBuiltinVideoEncoderFactory(), CreateBuiltinVideoDecoderFactory(),
>+        nullptr /* audio_mixer */, nullptr /* audio_processing */);
>   }
> 
>   WrapperPtr CreatePeerConnection() {
>@@ -97,18 +106,20 @@ class PeerConnectionIceUnitTest : public ::testing::Test {
>   WrapperPtr CreatePeerConnection(const RTCConfiguration& config) {
>     auto* fake_network = NewFakeNetwork();
>     auto port_allocator =
>-        rtc::MakeUnique<cricket::BasicPortAllocator>(fake_network);
>+        absl::make_unique<cricket::BasicPortAllocator>(fake_network);
>     port_allocator->set_flags(cricket::PORTALLOCATOR_DISABLE_TCP |
>                               cricket::PORTALLOCATOR_DISABLE_RELAY);
>     port_allocator->set_step_delay(cricket::kMinimumStepDelay);
>-    auto observer = rtc::MakeUnique<MockPeerConnectionObserver>();
>+    RTCConfiguration modified_config = config;
>+    modified_config.sdp_semantics = sdp_semantics_;
>+    auto observer = absl::make_unique<MockPeerConnectionObserver>();
>     auto pc = pc_factory_->CreatePeerConnection(
>-        config, std::move(port_allocator), nullptr, observer.get());
>+        modified_config, std::move(port_allocator), nullptr, observer.get());
>     if (!pc) {
>       return nullptr;
>     }
> 
>-    auto wrapper = rtc::MakeUnique<PeerConnectionWrapperForIceUnitTest>(
>+    auto wrapper = absl::make_unique<PeerConnectionWrapperForIceTest>(
>         pc_factory_, pc, std::move(observer));
>     wrapper->set_network(fake_network);
>     return wrapper;
>@@ -191,10 +202,18 @@ class PeerConnectionIceUnitTest : public ::testing::Test {
>         static_cast<PeerConnectionProxyWithInternal<PeerConnectionInterface>*>(
>             pc_wrapper_ptr->pc());
>     PeerConnection* pc = static_cast<PeerConnection*>(pc_proxy->internal());
>-    return pc->voice_channel()
>-        ->rtp_dtls_transport()
>-        ->ice_transport()
>-        ->GetIceRole();
>+    for (auto transceiver : pc->GetTransceiversInternal()) {
>+      if (transceiver->media_type() == cricket::MEDIA_TYPE_AUDIO) {
>+        cricket::BaseChannel* channel = transceiver->internal()->channel();
>+        if (channel) {
>+          auto dtls_transport = static_cast<cricket::DtlsTransportInternal*>(
>+              channel->rtp_packet_transport());
>+          return dtls_transport->ice_transport()->GetIceRole();
>+        }
>+      }
>+    }
>+    RTC_NOTREACHED();
>+    return cricket::ICEROLE_UNKNOWN;
>   }
> 
>   bool AddCandidateToFirstTransport(cricket::Candidate* candidate,
>@@ -203,8 +222,9 @@ class PeerConnectionIceUnitTest : public ::testing::Test {
>     RTC_DCHECK(desc->contents().size() > 0);
>     const auto& first_content = desc->contents()[0];
>     candidate->set_transport_name(first_content.name);
>-    JsepIceCandidate jsep_candidate(first_content.name, 0, *candidate);
>-    return sdesc->AddCandidate(&jsep_candidate);
>+    std::unique_ptr<IceCandidateInterface> jsep_candidate =
>+        CreateIceCandidate(first_content.name, 0, *candidate);
>+    return sdesc->AddCandidate(jsep_candidate.get());
>   }
> 
>   rtc::FakeNetworkManager* NewFakeNetwork() {
>@@ -222,6 +242,16 @@ class PeerConnectionIceUnitTest : public ::testing::Test {
>   rtc::AutoSocketServerThread main_;
>   rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory_;
>   std::vector<std::unique_ptr<rtc::FakeNetworkManager>> fake_networks_;
>+  const SdpSemantics sdp_semantics_;
>+};
>+
>+class PeerConnectionIceTest
>+    : public PeerConnectionIceBaseTest,
>+      public ::testing::WithParamInterface<SdpSemantics> {
>+ protected:
>+  PeerConnectionIceTest() : PeerConnectionIceBaseTest(GetParam()) {
>+    webrtc::metrics::Reset();
>+  }
> };
> 
> ::testing::AssertionResult AssertCandidatesEqual(const char* a_expr,
>@@ -252,7 +282,7 @@ class PeerConnectionIceUnitTest : public ::testing::Test {
>   }
> }
> 
>-TEST_F(PeerConnectionIceUnitTest, OfferContainsGatheredCandidates) {
>+TEST_P(PeerConnectionIceTest, OfferContainsGatheredCandidates) {
>   const SocketAddress kLocalAddress("1.1.1.1", 0);
> 
>   auto caller = CreatePeerConnectionWithAudioVideo();
>@@ -272,7 +302,7 @@ TEST_F(PeerConnectionIceUnitTest, OfferContainsGatheredCandidates) {
>             offer->candidates(1)->count());
> }
> 
>-TEST_F(PeerConnectionIceUnitTest, AnswerContainsGatheredCandidates) {
>+TEST_P(PeerConnectionIceTest, AnswerContainsGatheredCandidates) {
>   const SocketAddress kCallerAddress("1.1.1.1", 0);
> 
>   auto caller = CreatePeerConnectionWithAudioVideo();
>@@ -284,7 +314,7 @@ TEST_F(PeerConnectionIceUnitTest, AnswerContainsGatheredCandidates) {
> 
>   EXPECT_TRUE_WAIT(callee->IsIceGatheringDone(), kIceCandidatesTimeout);
> 
>-  auto answer = callee->CreateAnswer();
>+  auto* answer = callee->pc()->local_description();
>   EXPECT_LT(0u, caller->observer()->GetCandidatesByMline(0).size());
>   EXPECT_EQ(callee->observer()->GetCandidatesByMline(0).size(),
>             answer->candidates(0)->count());
>@@ -293,7 +323,7 @@ TEST_F(PeerConnectionIceUnitTest, AnswerContainsGatheredCandidates) {
>             answer->candidates(1)->count());
> }
> 
>-TEST_F(PeerConnectionIceUnitTest,
>+TEST_P(PeerConnectionIceTest,
>        CanSetRemoteSessionDescriptionWithRemoteCandidates) {
>   const SocketAddress kCallerAddress("1.1.1.1", 1111);
> 
>@@ -311,7 +341,7 @@ TEST_F(PeerConnectionIceUnitTest,
>                       remote_candidates[0]->candidate());
> }
> 
>-TEST_F(PeerConnectionIceUnitTest, SetLocalDescriptionFailsIfNoIceCredentials) {
>+TEST_P(PeerConnectionIceTest, SetLocalDescriptionFailsIfNoIceCredentials) {
>   auto caller = CreatePeerConnectionWithAudioVideo();
> 
>   auto offer = caller->CreateOffer();
>@@ -320,7 +350,7 @@ TEST_F(PeerConnectionIceUnitTest, SetLocalDescriptionFailsIfNoIceCredentials) {
>   EXPECT_FALSE(caller->SetLocalDescription(std::move(offer)));
> }
> 
>-TEST_F(PeerConnectionIceUnitTest, SetRemoteDescriptionFailsIfNoIceCredentials) {
>+TEST_P(PeerConnectionIceTest, SetRemoteDescriptionFailsIfNoIceCredentials) {
>   auto caller = CreatePeerConnectionWithAudioVideo();
>   auto callee = CreatePeerConnectionWithAudioVideo();
> 
>@@ -330,10 +360,25 @@ TEST_F(PeerConnectionIceUnitTest, SetRemoteDescriptionFailsIfNoIceCredentials) {
>   EXPECT_FALSE(callee->SetRemoteDescription(std::move(offer)));
> }
> 
>+// Test that doing an offer/answer exchange with no transport (i.e., no data
>+// channel or media) results in the ICE connection state staying at New.
>+TEST_P(PeerConnectionIceTest,
>+       OfferAnswerWithNoTransportsDoesNotChangeIceConnectionState) {
>+  auto caller = CreatePeerConnection();
>+  auto callee = CreatePeerConnection();
>+
>+  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
>+
>+  EXPECT_EQ(PeerConnectionInterface::kIceConnectionNew,
>+            caller->pc()->ice_connection_state());
>+  EXPECT_EQ(PeerConnectionInterface::kIceConnectionNew,
>+            callee->pc()->ice_connection_state());
>+}
>+
> // The following group tests that ICE candidates are not generated before
> // SetLocalDescription is called on a PeerConnection.
> 
>-TEST_F(PeerConnectionIceUnitTest, NoIceCandidatesBeforeSetLocalDescription) {
>+TEST_P(PeerConnectionIceTest, NoIceCandidatesBeforeSetLocalDescription) {
>   const SocketAddress kLocalAddress("1.1.1.1", 0);
> 
>   auto caller = CreatePeerConnectionWithAudioVideo();
>@@ -344,7 +389,7 @@ TEST_F(PeerConnectionIceUnitTest, NoIceCandidatesBeforeSetLocalDescription) {
> 
>   EXPECT_EQ(0u, caller->observer()->candidates_.size());
> }
>-TEST_F(PeerConnectionIceUnitTest,
>+TEST_P(PeerConnectionIceTest,
>        NoIceCandidatesBeforeAnswerSetAsLocalDescription) {
>   const SocketAddress kCallerAddress("1.1.1.1", 1111);
> 
>@@ -363,22 +408,46 @@ TEST_F(PeerConnectionIceUnitTest,
>   EXPECT_EQ(0u, callee->observer()->candidates_.size());
> }
> 
>-TEST_F(PeerConnectionIceUnitTest,
>-       CannotAddCandidateWhenRemoteDescriptionNotSet) {
>+TEST_P(PeerConnectionIceTest, CannotAddCandidateWhenRemoteDescriptionNotSet) {
>   const SocketAddress kCalleeAddress("1.1.1.1", 1111);
> 
>   auto caller = CreatePeerConnectionWithAudioVideo();
>   cricket::Candidate candidate = CreateLocalUdpCandidate(kCalleeAddress);
>-  JsepIceCandidate jsep_candidate(cricket::CN_AUDIO, 0, candidate);
>+  std::unique_ptr<IceCandidateInterface> jsep_candidate =
>+      CreateIceCandidate(cricket::CN_AUDIO, 0, candidate);
> 
>-  EXPECT_FALSE(caller->pc()->AddIceCandidate(&jsep_candidate));
>+  EXPECT_FALSE(caller->pc()->AddIceCandidate(jsep_candidate.get()));
> 
>   caller->CreateOfferAndSetAsLocal();
> 
>-  EXPECT_FALSE(caller->pc()->AddIceCandidate(&jsep_candidate));
>+  EXPECT_FALSE(caller->pc()->AddIceCandidate(jsep_candidate.get()));
>+  EXPECT_EQ(
>+      2, webrtc::metrics::NumSamples("WebRTC.PeerConnection.AddIceCandidate"));
>+  EXPECT_EQ(
>+      2, webrtc::metrics::NumEvents("WebRTC.PeerConnection.AddIceCandidate",
>+                                    kAddIceCandidateFailNoRemoteDescription));
>+}
>+
>+TEST_P(PeerConnectionIceTest, CannotAddCandidateWhenPeerConnectionClosed) {
>+  const SocketAddress kCalleeAddress("1.1.1.1", 1111);
>+
>+  auto caller = CreatePeerConnectionWithAudioVideo();
>+  auto callee = CreatePeerConnectionWithAudioVideo();
>+
>+  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
>+
>+  cricket::Candidate candidate = CreateLocalUdpCandidate(kCalleeAddress);
>+  auto* audio_content = cricket::GetFirstAudioContent(
>+      caller->pc()->local_description()->description());
>+  std::unique_ptr<IceCandidateInterface> jsep_candidate =
>+      CreateIceCandidate(audio_content->name, 0, candidate);
>+
>+  caller->pc()->Close();
>+
>+  EXPECT_FALSE(caller->pc()->AddIceCandidate(jsep_candidate.get()));
> }
> 
>-TEST_F(PeerConnectionIceUnitTest, DuplicateIceCandidateIgnoredWhenAdded) {
>+TEST_P(PeerConnectionIceTest, DuplicateIceCandidateIgnoredWhenAdded) {
>   const SocketAddress kCalleeAddress("1.1.1.1", 1111);
> 
>   auto caller = CreatePeerConnectionWithAudioVideo();
>@@ -394,7 +463,29 @@ TEST_F(PeerConnectionIceUnitTest, DuplicateIceCandidateIgnoredWhenAdded) {
>   EXPECT_EQ(1u, caller->GetIceCandidatesFromRemoteDescription().size());
> }
> 
>-TEST_F(PeerConnectionIceUnitTest,
>+TEST_P(PeerConnectionIceTest,
>+       CannotRemoveIceCandidatesWhenPeerConnectionClosed) {
>+  const SocketAddress kCalleeAddress("1.1.1.1", 1111);
>+
>+  auto caller = CreatePeerConnectionWithAudioVideo();
>+  auto callee = CreatePeerConnectionWithAudioVideo();
>+
>+  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
>+
>+  cricket::Candidate candidate = CreateLocalUdpCandidate(kCalleeAddress);
>+  auto* audio_content = cricket::GetFirstAudioContent(
>+      caller->pc()->local_description()->description());
>+  std::unique_ptr<IceCandidateInterface> ice_candidate =
>+      CreateIceCandidate(audio_content->name, 0, candidate);
>+
>+  ASSERT_TRUE(caller->pc()->AddIceCandidate(ice_candidate.get()));
>+
>+  caller->pc()->Close();
>+
>+  EXPECT_FALSE(caller->pc()->RemoveIceCandidates({candidate}));
>+}
>+
>+TEST_P(PeerConnectionIceTest,
>        AddRemoveCandidateWithEmptyTransportDoesNotCrash) {
>   const SocketAddress kCalleeAddress("1.1.1.1", 1111);
> 
>@@ -407,12 +498,15 @@ TEST_F(PeerConnectionIceUnitTest,
> 
>   // |candidate.transport_name()| is empty.
>   cricket::Candidate candidate = CreateLocalUdpCandidate(kCalleeAddress);
>-  JsepIceCandidate ice_candidate(cricket::CN_AUDIO, 0, candidate);
>-  EXPECT_TRUE(caller->pc()->AddIceCandidate(&ice_candidate));
>+  auto* audio_content = cricket::GetFirstAudioContent(
>+      caller->pc()->local_description()->description());
>+  std::unique_ptr<IceCandidateInterface> ice_candidate =
>+      CreateIceCandidate(audio_content->name, 0, candidate);
>+  EXPECT_TRUE(caller->pc()->AddIceCandidate(ice_candidate.get()));
>   EXPECT_TRUE(caller->pc()->RemoveIceCandidates({candidate}));
> }
> 
>-TEST_F(PeerConnectionIceUnitTest, RemoveCandidateRemovesFromRemoteDescription) {
>+TEST_P(PeerConnectionIceTest, RemoveCandidateRemovesFromRemoteDescription) {
>   const SocketAddress kCalleeAddress("1.1.1.1", 1111);
> 
>   auto caller = CreatePeerConnectionWithAudioVideo();
>@@ -431,7 +525,7 @@ TEST_F(PeerConnectionIceUnitTest, RemoveCandidateRemovesFromRemoteDescription) {
> // Test that if a candidate is added via AddIceCandidate and via an updated
> // remote description, then both candidates appear in the stored remote
> // description.
>-TEST_F(PeerConnectionIceUnitTest,
>+TEST_P(PeerConnectionIceTest,
>        CandidateInSubsequentOfferIsAddedToRemoteDescription) {
>   const SocketAddress kCallerAddress1("1.1.1.1", 1111);
>   const SocketAddress kCallerAddress2("2.2.2.2", 2222);
>@@ -461,25 +555,24 @@ TEST_F(PeerConnectionIceUnitTest,
> // has either ICE ufrag/pwd too short or too long and succeeds otherwise.
> // The standard (https://tools.ietf.org/html/rfc5245#section-15.4) says that
> // pwd must be 22-256 characters and ufrag must be 4-256 characters.
>-TEST_F(PeerConnectionIceUnitTest, VerifyUfragPwdLength) {
>-  auto caller = CreatePeerConnectionWithAudioVideo();
>-  auto callee = CreatePeerConnectionWithAudioVideo();
>-
>-  auto set_local_description_with_ufrag_pwd_length =
>-      [this, &caller](int ufrag_len, int pwd_len) {
>-        auto offer = caller->CreateOffer();
>-        SetIceUfragPwd(offer.get(), std::string(ufrag_len, 'x'),
>-                       std::string(pwd_len, 'x'));
>-        return caller->SetLocalDescription(std::move(offer));
>-      };
>-
>-  auto set_remote_description_with_ufrag_pwd_length =
>-      [this, &caller, &callee](int ufrag_len, int pwd_len) {
>-        auto offer = caller->CreateOffer();
>-        SetIceUfragPwd(offer.get(), std::string(ufrag_len, 'x'),
>-                       std::string(pwd_len, 'x'));
>-        return callee->SetRemoteDescription(std::move(offer));
>-      };
>+TEST_P(PeerConnectionIceTest, VerifyUfragPwdLength) {
>+  auto set_local_description_with_ufrag_pwd_length = [this](int ufrag_len,
>+                                                            int pwd_len) {
>+    auto pc = CreatePeerConnectionWithAudioVideo();
>+    auto offer = pc->CreateOffer();
>+    SetIceUfragPwd(offer.get(), std::string(ufrag_len, 'x'),
>+                   std::string(pwd_len, 'x'));
>+    return pc->SetLocalDescription(std::move(offer));
>+  };
>+
>+  auto set_remote_description_with_ufrag_pwd_length = [this](int ufrag_len,
>+                                                             int pwd_len) {
>+    auto pc = CreatePeerConnectionWithAudioVideo();
>+    auto offer = pc->CreateOffer();
>+    SetIceUfragPwd(offer.get(), std::string(ufrag_len, 'x'),
>+                   std::string(pwd_len, 'x'));
>+    return pc->SetRemoteDescription(std::move(offer));
>+  };
> 
>   EXPECT_FALSE(set_local_description_with_ufrag_pwd_length(3, 22));
>   EXPECT_FALSE(set_remote_description_with_ufrag_pwd_length(3, 22));
>@@ -506,7 +599,7 @@ TEST_F(PeerConnectionIceUnitTest, VerifyUfragPwdLength) {
>     if (candidate_ip == address.ipaddr()) {
>       return ::testing::AssertionSuccess();
>     }
>-    candidate_hosts << "\n" << candidate_ip;
>+    candidate_hosts << "\n" << candidate_ip.ToString();
>   }
>   return ::testing::AssertionFailure()
>          << address_expr << " (host " << address.HostAsURIString()
>@@ -514,7 +607,7 @@ TEST_F(PeerConnectionIceUnitTest, VerifyUfragPwdLength) {
>          << " which have the following address hosts:" << candidate_hosts.str();
> }
> 
>-TEST_F(PeerConnectionIceUnitTest, CandidatesGeneratedForEachLocalInterface) {
>+TEST_P(PeerConnectionIceTest, CandidatesGeneratedForEachLocalInterface) {
>   const SocketAddress kLocalAddress1("1.1.1.1", 0);
>   const SocketAddress kLocalAddress2("2.2.2.2", 0);
> 
>@@ -530,8 +623,7 @@ TEST_F(PeerConnectionIceUnitTest, CandidatesGeneratedForEachLocalInterface) {
>   EXPECT_PRED_FORMAT2(AssertIpInCandidates, kLocalAddress2, candidates);
> }
> 
>-TEST_F(PeerConnectionIceUnitTest,
>-       TrickledSingleCandidateAddedToRemoteDescription) {
>+TEST_P(PeerConnectionIceTest, TrickledSingleCandidateAddedToRemoteDescription) {
>   const SocketAddress kCallerAddress("1.1.1.1", 1111);
> 
>   auto caller = CreatePeerConnectionWithAudioVideo();
>@@ -547,8 +639,7 @@ TEST_F(PeerConnectionIceUnitTest,
>                       candidates[0]->candidate());
> }
> 
>-TEST_F(PeerConnectionIceUnitTest,
>-       TwoTrickledCandidatesAddedToRemoteDescription) {
>+TEST_P(PeerConnectionIceTest, TwoTrickledCandidatesAddedToRemoteDescription) {
>   const SocketAddress kCalleeAddress1("1.1.1.1", 1111);
>   const SocketAddress kCalleeAddress2("2.2.2.2", 2222);
> 
>@@ -573,8 +664,7 @@ TEST_F(PeerConnectionIceUnitTest,
>                       candidates[1]->candidate());
> }
> 
>-TEST_F(PeerConnectionIceUnitTest,
>-       LocalDescriptionUpdatedWhenContinualGathering) {
>+TEST_P(PeerConnectionIceTest, LocalDescriptionUpdatedWhenContinualGathering) {
>   const SocketAddress kLocalAddress("1.1.1.1", 0);
> 
>   RTCConfiguration config;
>@@ -595,7 +685,7 @@ TEST_F(PeerConnectionIceUnitTest,
> // Test that when continual gathering is enabled, and a network interface goes
> // down, the candidate is signaled as removed and removed from the local
> // description.
>-TEST_F(PeerConnectionIceUnitTest,
>+TEST_P(PeerConnectionIceTest,
>        LocalCandidatesRemovedWhenNetworkDownIfGatheringContinually) {
>   const SocketAddress kLocalAddress("1.1.1.1", 0);
> 
>@@ -621,7 +711,7 @@ TEST_F(PeerConnectionIceUnitTest,
>   EXPECT_LT(0, caller->observer()->num_candidates_removed_);
> }
> 
>-TEST_F(PeerConnectionIceUnitTest,
>+TEST_P(PeerConnectionIceTest,
>        LocalCandidatesNotRemovedWhenNetworkDownIfGatheringOnce) {
>   const SocketAddress kLocalAddress("1.1.1.1", 0);
> 
>@@ -646,7 +736,7 @@ TEST_F(PeerConnectionIceUnitTest,
> // (indicating an ICE restart) the old candidates are removed and new candidates
> // added to the remote description.
> 
>-TEST_F(PeerConnectionIceUnitTest, IceRestartOfferClearsExistingCandidate) {
>+TEST_P(PeerConnectionIceTest, IceRestartOfferClearsExistingCandidate) {
>   const SocketAddress kCallerAddress("1.1.1.1", 1111);
> 
>   auto caller = CreatePeerConnectionWithAudioVideo();
>@@ -664,7 +754,7 @@ TEST_F(PeerConnectionIceUnitTest, IceRestartOfferClearsExistingCandidate) {
> 
>   EXPECT_EQ(0u, callee->GetIceCandidatesFromRemoteDescription().size());
> }
>-TEST_F(PeerConnectionIceUnitTest,
>+TEST_P(PeerConnectionIceTest,
>        IceRestartOfferCandidateReplacesExistingCandidate) {
>   const SocketAddress kFirstCallerAddress("1.1.1.1", 1111);
>   const SocketAddress kRestartedCallerAddress("2.2.2.2", 2222);
>@@ -696,8 +786,7 @@ TEST_F(PeerConnectionIceUnitTest,
> 
> // Test that if there is not an ICE restart (i.e., nothing changes), then the
> // answer to a later offer should have the same ufrag/pwd as the first answer.
>-TEST_F(PeerConnectionIceUnitTest,
>-       LaterAnswerHasSameIceCredentialsIfNoIceRestart) {
>+TEST_P(PeerConnectionIceTest, LaterAnswerHasSameIceCredentialsIfNoIceRestart) {
>   auto caller = CreatePeerConnectionWithAudioVideo();
>   auto callee = CreatePeerConnectionWithAudioVideo();
> 
>@@ -724,20 +813,23 @@ TEST_F(PeerConnectionIceUnitTest,
> // a=ice-pwd attributes compared to the previous SDP from the peer, it
> // indicates that ICE is restarting for this media stream."
> 
>-class PeerConnectionIceUfragPwdAnswerUnitTest
>-    : public PeerConnectionIceUnitTest,
>-      public ::testing::WithParamInterface<std::pair<bool, bool>> {
>+class PeerConnectionIceUfragPwdAnswerTest
>+    : public PeerConnectionIceBaseTest,
>+      public ::testing::WithParamInterface<
>+          std::tuple<SdpSemantics, std::tuple<bool, bool>>> {
>  protected:
>-  PeerConnectionIceUfragPwdAnswerUnitTest() {
>-    offer_new_ufrag_ = GetParam().first;
>-    offer_new_pwd_ = GetParam().second;
>+  PeerConnectionIceUfragPwdAnswerTest()
>+      : PeerConnectionIceBaseTest(std::get<0>(GetParam())) {
>+    auto param = std::get<1>(GetParam());
>+    offer_new_ufrag_ = std::get<0>(param);
>+    offer_new_pwd_ = std::get<1>(param);
>   }
> 
>   bool offer_new_ufrag_;
>   bool offer_new_pwd_;
> };
> 
>-TEST_P(PeerConnectionIceUfragPwdAnswerUnitTest, TestIncludedInAnswer) {
>+TEST_P(PeerConnectionIceUfragPwdAnswerTest, TestIncludedInAnswer) {
>   auto caller = CreatePeerConnectionWithAudioVideo();
>   auto callee = CreatePeerConnectionWithAudioVideo();
> 
>@@ -766,18 +858,19 @@ TEST_P(PeerConnectionIceUfragPwdAnswerUnitTest, TestIncludedInAnswer) {
> }
> 
> INSTANTIATE_TEST_CASE_P(
>-    PeerConnectionIceUnitTest,
>-    PeerConnectionIceUfragPwdAnswerUnitTest,
>-    Values(std::make_pair(true, true),     // Both changed.
>-           std::make_pair(true, false),    // Only ufrag changed.
>-           std::make_pair(false, true)));  // Only pwd changed.
>+    PeerConnectionIceTest,
>+    PeerConnectionIceUfragPwdAnswerTest,
>+    Combine(Values(SdpSemantics::kPlanB, SdpSemantics::kUnifiedPlan),
>+            Values(std::make_pair(true, true),      // Both changed.
>+                   std::make_pair(true, false),     // Only ufrag changed.
>+                   std::make_pair(false, true))));  // Only pwd changed.
> 
> // Test that if an ICE restart is offered on one media section, then the answer
> // will only change ICE ufrag/pwd for that section and keep the other sections
> // the same.
> // Note that this only works if we have disabled BUNDLE, otherwise all media
> // sections will share the same transport.
>-TEST_F(PeerConnectionIceUnitTest,
>+TEST_P(PeerConnectionIceTest,
>        CreateAnswerHasNewUfragPwdForOnlyMediaSectionWhichRestarted) {
>   auto caller = CreatePeerConnectionWithAudioVideo();
>   auto callee = CreatePeerConnectionWithAudioVideo();
>@@ -817,7 +910,7 @@ TEST_F(PeerConnectionIceUnitTest,
> // ICE and the callee uses the full implementation, the caller takes the
> // CONTROLLED role and the callee takes the CONTROLLING role. This is specified
> // in RFC5245 Section 5.1.1.
>-TEST_F(PeerConnectionIceUnitTest,
>+TEST_P(PeerConnectionIceTest,
>        OfferFromLiteIceControlledAndAnswerFromFullIceControlling) {
>   auto caller = CreatePeerConnectionWithAudioVideo();
>   auto callee = CreatePeerConnectionWithAudioVideo();
>@@ -841,7 +934,7 @@ TEST_F(PeerConnectionIceUnitTest,
> // Test that when the caller and the callee both use the lite implementation of
> // ICE, the initial offerer (caller) takes the CONTROLLING role and the callee
> // takes the CONTROLLED role. This is specified in RFC5245 Section 5.1.1.
>-TEST_F(PeerConnectionIceUnitTest,
>+TEST_P(PeerConnectionIceTest,
>        OfferFromLiteIceControllingAndAnswerFromLiteIceControlled) {
>   auto caller = CreatePeerConnectionWithAudioVideo();
>   auto callee = CreatePeerConnectionWithAudioVideo();
>@@ -862,4 +955,55 @@ TEST_F(PeerConnectionIceUnitTest,
>   EXPECT_EQ(cricket::ICEROLE_CONTROLLED, GetIceRole(callee));
> }
> 
>+INSTANTIATE_TEST_CASE_P(PeerConnectionIceTest,
>+                        PeerConnectionIceTest,
>+                        Values(SdpSemantics::kPlanB,
>+                               SdpSemantics::kUnifiedPlan));
>+
>+class PeerConnectionIceConfigTest : public testing::Test {
>+ protected:
>+  void SetUp() override {
>+    pc_factory_ = CreatePeerConnectionFactory(
>+        rtc::Thread::Current(), rtc::Thread::Current(), rtc::Thread::Current(),
>+        FakeAudioCaptureModule::Create(), CreateBuiltinAudioEncoderFactory(),
>+        CreateBuiltinAudioDecoderFactory(), CreateBuiltinVideoEncoderFactory(),
>+        CreateBuiltinVideoDecoderFactory(), nullptr /* audio_mixer */,
>+        nullptr /* audio_processing */);
>+  }
>+  void CreatePeerConnection(const RTCConfiguration& config) {
>+    std::unique_ptr<cricket::FakePortAllocator> port_allocator(
>+        new cricket::FakePortAllocator(rtc::Thread::Current(), nullptr));
>+    port_allocator_ = port_allocator.get();
>+    rtc::scoped_refptr<PeerConnectionInterface> pc(
>+        pc_factory_->CreatePeerConnection(
>+            config, nullptr /* constraint */, std::move(port_allocator),
>+            nullptr /* cert_generator */, &observer_));
>+    EXPECT_TRUE(pc.get());
>+    pc_ = std::move(pc.get());
>+  }
>+
>+  rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory_ = nullptr;
>+  rtc::scoped_refptr<PeerConnectionInterface> pc_ = nullptr;
>+  cricket::FakePortAllocator* port_allocator_ = nullptr;
>+
>+  MockPeerConnectionObserver observer_;
>+};
>+
>+TEST_F(PeerConnectionIceConfigTest, SetStunCandidateKeepaliveInterval) {
>+  RTCConfiguration config;
>+  config.stun_candidate_keepalive_interval = 123;
>+  config.ice_candidate_pool_size = 1;
>+  CreatePeerConnection(config);
>+  ASSERT_NE(port_allocator_, nullptr);
>+  absl::optional<int> actual_stun_keepalive_interval =
>+      port_allocator_->stun_candidate_keepalive_interval();
>+  EXPECT_EQ(actual_stun_keepalive_interval.value_or(-1), 123);
>+  config.stun_candidate_keepalive_interval = 321;
>+  RTCError error;
>+  pc_->SetConfiguration(config, &error);
>+  actual_stun_keepalive_interval =
>+      port_allocator_->stun_candidate_keepalive_interval();
>+  EXPECT_EQ(actual_stun_keepalive_interval.value_or(-1), 321);
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_integrationtest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_integrationtest.cc
>index 0bc68a24ee7..6def1983968 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_integrationtest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_integrationtest.cc
>@@ -24,13 +24,20 @@
> 
> #include "api/audio_codecs/builtin_audio_decoder_factory.h"
> #include "api/audio_codecs/builtin_audio_encoder_factory.h"
>-#include "api/fakemetricsobserver.h"
> #include "api/mediastreaminterface.h"
> #include "api/peerconnectioninterface.h"
> #include "api/peerconnectionproxy.h"
> #include "api/rtpreceiverinterface.h"
> #include "api/test/fakeconstraints.h"
>+#include "api/video_codecs/builtin_video_decoder_factory.h"
>+#include "api/video_codecs/builtin_video_encoder_factory.h"
>+#include "api/video_codecs/sdp_video_format.h"
>+#include "call/call.h"
>+#include "logging/rtc_event_log/fake_rtc_event_log_factory.h"
>+#include "logging/rtc_event_log/rtc_event_log_factory_interface.h"
> #include "media/engine/fakewebrtcvideoengine.h"
>+#include "media/engine/webrtcmediaengine.h"
>+#include "modules/audio_processing/include/audio_processing.h"
> #include "p2p/base/p2pconstants.h"
> #include "p2p/base/portinterface.h"
> #include "p2p/base/teststunserver.h"
>@@ -42,16 +49,20 @@
> #include "pc/mediasession.h"
> #include "pc/peerconnection.h"
> #include "pc/peerconnectionfactory.h"
>+#include "pc/rtpmediautils.h"
> #include "pc/sessiondescription.h"
> #include "pc/test/fakeaudiocapturemodule.h"
>-#include "pc/test/fakeperiodicvideocapturer.h"
>+#include "pc/test/fakeperiodicvideotracksource.h"
> #include "pc/test/fakertccertificategenerator.h"
> #include "pc/test/fakevideotrackrenderer.h"
> #include "pc/test/mockpeerconnectionobservers.h"
> #include "rtc_base/fakenetwork.h"
> #include "rtc_base/firewallsocketserver.h"
> #include "rtc_base/gunit.h"
>+#include "rtc_base/numerics/safe_conversions.h"
>+#include "rtc_base/testcertificateverifier.h"
> #include "rtc_base/virtualsocketserver.h"
>+#include "system_wrappers/include/metrics_default.h"
> #include "test/gmock.h"
> 
> using cricket::ContentInfo;
>@@ -60,7 +71,9 @@ using cricket::FakeWebRtcVideoDecoderFactory;
> using cricket::FakeWebRtcVideoEncoder;
> using cricket::FakeWebRtcVideoEncoderFactory;
> using cricket::MediaContentDescription;
>+using cricket::StreamParams;
> using rtc::SocketAddress;
>+using ::testing::Combine;
> using ::testing::ElementsAre;
> using ::testing::Values;
> using webrtc::DataBuffer;
>@@ -69,6 +82,7 @@ using webrtc::DtmfSender;
> using webrtc::DtmfSenderInterface;
> using webrtc::DtmfSenderObserverInterface;
> using webrtc::FakeConstraints;
>+using webrtc::FakeVideoTrackRenderer;
> using webrtc::MediaConstraintsInterface;
> using webrtc::MediaStreamInterface;
> using webrtc::MediaStreamTrackInterface;
>@@ -79,12 +93,22 @@ using webrtc::MockStatsObserver;
> using webrtc::ObserverInterface;
> using webrtc::PeerConnection;
> using webrtc::PeerConnectionInterface;
>+using RTCConfiguration = PeerConnectionInterface::RTCConfiguration;
> using webrtc::PeerConnectionFactory;
> using webrtc::PeerConnectionProxy;
>+using webrtc::RTCErrorType;
>+using webrtc::RTCTransportStats;
>+using webrtc::RtpSenderInterface;
> using webrtc::RtpReceiverInterface;
>+using webrtc::RtpSenderInterface;
>+using webrtc::RtpTransceiverDirection;
>+using webrtc::RtpTransceiverInit;
>+using webrtc::RtpTransceiverInterface;
>+using webrtc::SdpSemantics;
> using webrtc::SdpType;
> using webrtc::SessionDescriptionInterface;
> using webrtc::StreamCollectionInterface;
>+using webrtc::VideoTrackInterface;
> 
> namespace {
> 
>@@ -101,7 +125,7 @@ static const char kDataChannelLabel[] = "data_channel";
> 
> // SRTP cipher name negotiated by the tests. This must be updated if the
> // default changes.
>-static const int kDefaultSrtpCryptoSuite = rtc::SRTP_AES128_CM_SHA1_32;
>+static const int kDefaultSrtpCryptoSuite = rtc::SRTP_AES128_CM_SHA1_80;
> static const int kDefaultSrtpCryptoSuiteGcm = rtc::SRTP_AEAD_AES_256_GCM;
> 
> static const SocketAddress kDefaultLocalAddress("192.168.1.1", 0);
>@@ -123,6 +147,26 @@ void RemoveSsrcsAndMsids(cricket::SessionDescription* desc) {
>   desc->set_msid_supported(false);
> }
> 
>+// Removes all stream information besides the stream ids, simulating an
>+// endpoint that only signals a=msid lines to convey stream_ids.
>+void RemoveSsrcsAndKeepMsids(cricket::SessionDescription* desc) {
>+  for (ContentInfo& content : desc->contents()) {
>+    std::string track_id;
>+    std::vector<std::string> stream_ids;
>+    if (!content.media_description()->streams().empty()) {
>+      const StreamParams& first_stream =
>+          content.media_description()->streams()[0];
>+      track_id = first_stream.id;
>+      stream_ids = first_stream.stream_ids();
>+    }
>+    content.media_description()->mutable_streams().clear();
>+    StreamParams new_stream;
>+    new_stream.id = track_id;
>+    new_stream.set_stream_ids(stream_ids);
>+    content.media_description()->AddStream(new_stream);
>+  }
>+}
>+
> int FindFirstMediaStatsIndexByKind(
>     const std::string& kind,
>     const std::vector<const webrtc::RTCMediaStreamTrackStats*>&
>@@ -173,10 +217,9 @@ class MockRtpReceiverObserver : public webrtc::RtpReceiverObserverInterface {
> // encoders/decoders, though they aren't used by default since they don't
> // advertise support of any codecs.
> // TODO(steveanton): See how this could become a subclass of
>-// PeerConnectionWrapper defined in peerconnectionwrapper.h .
>+// PeerConnectionWrapper defined in peerconnectionwrapper.h.
> class PeerConnectionWrapper : public webrtc::PeerConnectionObserver,
>-                              public SignalingMessageReceiver,
>-                              public ObserverInterface {
>+                              public SignalingMessageReceiver {
>  public:
>   // Different factory methods for convenience.
>   // TODO(deadbeef): Could use the pattern of:
>@@ -191,56 +234,10 @@ class PeerConnectionWrapper : public webrtc::PeerConnectionObserver,
>       rtc::Thread* network_thread,
>       rtc::Thread* worker_thread) {
>     PeerConnectionWrapper* client(new PeerConnectionWrapper(debug_name));
>-    if (!client->Init(nullptr, nullptr, nullptr, std::move(cert_generator),
>-                      network_thread, worker_thread)) {
>-      delete client;
>-      return nullptr;
>-    }
>-    return client;
>-  }
>-
>-  static PeerConnectionWrapper* CreateWithConfig(
>-      const std::string& debug_name,
>-      const PeerConnectionInterface::RTCConfiguration& config,
>-      rtc::Thread* network_thread,
>-      rtc::Thread* worker_thread) {
>-    std::unique_ptr<FakeRTCCertificateGenerator> cert_generator(
>-        new FakeRTCCertificateGenerator());
>-    PeerConnectionWrapper* client(new PeerConnectionWrapper(debug_name));
>-    if (!client->Init(nullptr, nullptr, &config, std::move(cert_generator),
>-                      network_thread, worker_thread)) {
>-      delete client;
>-      return nullptr;
>-    }
>-    return client;
>-  }
>-
>-  static PeerConnectionWrapper* CreateWithOptions(
>-      const std::string& debug_name,
>-      const PeerConnectionFactory::Options& options,
>-      rtc::Thread* network_thread,
>-      rtc::Thread* worker_thread) {
>-    std::unique_ptr<FakeRTCCertificateGenerator> cert_generator(
>-        new FakeRTCCertificateGenerator());
>-    PeerConnectionWrapper* client(new PeerConnectionWrapper(debug_name));
>-    if (!client->Init(nullptr, &options, nullptr, std::move(cert_generator),
>-                      network_thread, worker_thread)) {
>-      delete client;
>-      return nullptr;
>-    }
>-    return client;
>-  }
>-
>-  static PeerConnectionWrapper* CreateWithConstraints(
>-      const std::string& debug_name,
>-      const MediaConstraintsInterface* constraints,
>-      rtc::Thread* network_thread,
>-      rtc::Thread* worker_thread) {
>-    std::unique_ptr<FakeRTCCertificateGenerator> cert_generator(
>-        new FakeRTCCertificateGenerator());
>-    PeerConnectionWrapper* client(new PeerConnectionWrapper(debug_name));
>-    if (!client->Init(constraints, nullptr, nullptr, std::move(cert_generator),
>-                      network_thread, worker_thread)) {
>+    webrtc::PeerConnectionDependencies dependencies(nullptr);
>+    dependencies.cert_generator = std::move(cert_generator);
>+    if (!client->Init(nullptr, nullptr, nullptr, std::move(dependencies),
>+                      network_thread, worker_thread, nullptr)) {
>       delete client;
>       return nullptr;
>     }
>@@ -286,6 +283,13 @@ class PeerConnectionWrapper : public webrtc::PeerConnectionObserver,
>     generated_sdp_munger_ = std::move(munger);
>   }
> 
>+  // Set a callback to be invoked when a remote offer is received via the fake
>+  // signaling channel. This provides an opportunity to change the
>+  // PeerConnection state before an answer is created and sent to the caller.
>+  void SetRemoteOfferHandler(std::function<void()> handler) {
>+    remote_offer_handler_ = std::move(handler);
>+  }
>+
>   // Every ICE connection state in order that has been seen by the observer.
>   std::vector<PeerConnectionInterface::IceConnectionState>
>   ice_connection_state_history() const {
>@@ -301,26 +305,25 @@ class PeerConnectionWrapper : public webrtc::PeerConnectionObserver,
>     return ice_gathering_state_history_;
>   }
> 
>-  // TODO(deadbeef): Switch the majority of these tests to use AddTrack instead
>-  // of AddStream since AddStream is deprecated.
>-  void AddAudioVideoMediaStream() {
>-    AddMediaStreamFromTracks(CreateLocalAudioTrack(), CreateLocalVideoTrack());
>+  void AddAudioVideoTracks() {
>+    AddAudioTrack();
>+    AddVideoTrack();
>   }
> 
>-  void AddAudioOnlyMediaStream() {
>-    AddMediaStreamFromTracks(CreateLocalAudioTrack(), nullptr);
>+  rtc::scoped_refptr<RtpSenderInterface> AddAudioTrack() {
>+    return AddTrack(CreateLocalAudioTrack());
>   }
> 
>-  void AddVideoOnlyMediaStream() {
>-    AddMediaStreamFromTracks(nullptr, CreateLocalVideoTrack());
>+  rtc::scoped_refptr<RtpSenderInterface> AddVideoTrack() {
>+    return AddTrack(CreateLocalVideoTrack());
>   }
> 
>   rtc::scoped_refptr<webrtc::AudioTrackInterface> CreateLocalAudioTrack() {
>-    FakeConstraints constraints;
>+    cricket::AudioOptions options;
>     // Disable highpass filter so that we can get all the test audio frames.
>-    constraints.AddMandatory(MediaConstraintsInterface::kHighpassFilter, false);
>+    options.highpass_filter = false;
>     rtc::scoped_refptr<webrtc::AudioSourceInterface> source =
>-        peer_connection_factory_->CreateAudioSource(&constraints);
>+        peer_connection_factory_->CreateAudioSource(options);
>     // TODO(perkj): Test audio source when it is implemented. Currently audio
>     // always use the default input.
>     return peer_connection_factory_->CreateAudioTrack(rtc::CreateRandomUuid(),
>@@ -328,33 +331,50 @@ class PeerConnectionWrapper : public webrtc::PeerConnectionObserver,
>   }
> 
>   rtc::scoped_refptr<webrtc::VideoTrackInterface> CreateLocalVideoTrack() {
>-    return CreateLocalVideoTrackInternal(FakeConstraints(),
>-                                         webrtc::kVideoRotation_0);
>+    return CreateLocalVideoTrackInternal(
>+        webrtc::FakePeriodicVideoSource::Config());
>   }
> 
>   rtc::scoped_refptr<webrtc::VideoTrackInterface>
>-  CreateLocalVideoTrackWithConstraints(const FakeConstraints& constraints) {
>-    return CreateLocalVideoTrackInternal(constraints, webrtc::kVideoRotation_0);
>+  CreateLocalVideoTrackWithConfig(
>+      webrtc::FakePeriodicVideoSource::Config config) {
>+    return CreateLocalVideoTrackInternal(config);
>   }
> 
>   rtc::scoped_refptr<webrtc::VideoTrackInterface>
>   CreateLocalVideoTrackWithRotation(webrtc::VideoRotation rotation) {
>-    return CreateLocalVideoTrackInternal(FakeConstraints(), rotation);
>+    webrtc::FakePeriodicVideoSource::Config config;
>+    config.rotation = rotation;
>+    return CreateLocalVideoTrackInternal(config);
>+  }
>+
>+  rtc::scoped_refptr<RtpSenderInterface> AddTrack(
>+      rtc::scoped_refptr<MediaStreamTrackInterface> track,
>+      const std::vector<std::string>& stream_ids = {}) {
>+    auto result = pc()->AddTrack(track, stream_ids);
>+    EXPECT_EQ(RTCErrorType::NONE, result.error().type());
>+    return result.MoveValue();
>   }
> 
>-  void AddMediaStreamFromTracks(
>-      const rtc::scoped_refptr<webrtc::AudioTrackInterface>& audio,
>-      const rtc::scoped_refptr<webrtc::VideoTrackInterface>& video) {
>-    rtc::scoped_refptr<MediaStreamInterface> stream =
>-        peer_connection_factory_->CreateLocalMediaStream(
>-            rtc::CreateRandomUuid());
>-    if (audio) {
>-      stream->AddTrack(audio);
>+  std::vector<rtc::scoped_refptr<RtpReceiverInterface>> GetReceiversOfType(
>+      cricket::MediaType media_type) {
>+    std::vector<rtc::scoped_refptr<RtpReceiverInterface>> receivers;
>+    for (auto receiver : pc()->GetReceivers()) {
>+      if (receiver->media_type() == media_type) {
>+        receivers.push_back(receiver);
>+      }
>     }
>-    if (video) {
>-      stream->AddTrack(video);
>+    return receivers;
>+  }
>+
>+  rtc::scoped_refptr<RtpTransceiverInterface> GetFirstTransceiverOfType(
>+      cricket::MediaType media_type) {
>+    for (auto transceiver : pc()->GetTransceivers()) {
>+      if (transceiver->receiver()->media_type() == media_type) {
>+        return transceiver;
>+      }
>     }
>-    EXPECT_TRUE(pc()->AddStream(stream));
>+    return nullptr;
>   }
> 
>   bool SignalingStateStable() {
>@@ -392,48 +412,14 @@ class PeerConnectionWrapper : public webrtc::PeerConnectionObserver,
>   // frames.
>   int min_video_frames_received_per_track() const {
>     int min_frames = INT_MAX;
>-    if (video_decoder_factory_enabled_) {
>-      const std::vector<FakeWebRtcVideoDecoder*>& decoders =
>-          fake_video_decoder_factory_->decoders();
>-      if (decoders.empty()) {
>-        return 0;
>-      }
>-      for (FakeWebRtcVideoDecoder* decoder : decoders) {
>-        min_frames = std::min(min_frames, decoder->GetNumFramesReceived());
>-      }
>-      return min_frames;
>-    } else {
>-      if (fake_video_renderers_.empty()) {
>-        return 0;
>-      }
>-
>-      for (const auto& pair : fake_video_renderers_) {
>-        min_frames = std::min(min_frames, pair.second->num_rendered_frames());
>-      }
>-      return min_frames;
>+    if (fake_video_renderers_.empty()) {
>+      return 0;
>     }
>-  }
> 
>-  // In contrast to the above, sums the video frames received for all tracks.
>-  // Can be used to verify that no video frames were received, or that the
>-  // counts didn't increase.
>-  int total_video_frames_received() const {
>-    int total = 0;
>-    if (video_decoder_factory_enabled_) {
>-      const std::vector<FakeWebRtcVideoDecoder*>& decoders =
>-          fake_video_decoder_factory_->decoders();
>-      for (const FakeWebRtcVideoDecoder* decoder : decoders) {
>-        total += decoder->GetNumFramesReceived();
>-      }
>-    } else {
>-      for (const auto& pair : fake_video_renderers_) {
>-        total += pair.second->num_rendered_frames();
>-      }
>-      for (const auto& renderer : removed_fake_video_renderers_) {
>-        total += renderer->num_rendered_frames();
>-      }
>+    for (const auto& pair : fake_video_renderers_) {
>+      min_frames = std::min(min_frames, pair.second->num_rendered_frames());
>     }
>-    return total;
>+    return min_frames;
>   }
> 
>   // Returns a MockStatsObserver in a state after stats gathering finished,
>@@ -567,17 +553,21 @@ class PeerConnectionWrapper : public webrtc::PeerConnectionObserver,
>   }
>   cricket::PortAllocator* port_allocator() const { return port_allocator_; }
> 
>+  webrtc::FakeRtcEventLogFactory* event_log_factory() const {
>+    return event_log_factory_;
>+  }
>+
>  private:
>   explicit PeerConnectionWrapper(const std::string& debug_name)
>       : debug_name_(debug_name) {}
> 
>-  bool Init(
>-      const MediaConstraintsInterface* constraints,
>-      const PeerConnectionFactory::Options* options,
>-      const PeerConnectionInterface::RTCConfiguration* config,
>-      std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
>-      rtc::Thread* network_thread,
>-      rtc::Thread* worker_thread) {
>+  bool Init(const MediaConstraintsInterface* constraints,
>+            const PeerConnectionFactory::Options* options,
>+            const PeerConnectionInterface::RTCConfiguration* config,
>+            webrtc::PeerConnectionDependencies dependencies,
>+            rtc::Thread* network_thread,
>+            rtc::Thread* worker_thread,
>+            std::unique_ptr<webrtc::FakeRtcEventLogFactory> event_log_factory) {
>     // There's an error in this test code if Init ends up being called twice.
>     RTC_DCHECK(!peer_connection_);
>     RTC_DCHECK(!peer_connection_factory_);
>@@ -592,33 +582,52 @@ class PeerConnectionWrapper : public webrtc::PeerConnectionObserver,
>     if (!fake_audio_capture_module_) {
>       return false;
>     }
>-    // Note that these factories don't end up getting used unless supported
>-    // codecs are added to them.
>-    fake_video_decoder_factory_ = new FakeWebRtcVideoDecoderFactory();
>-    fake_video_encoder_factory_ = new FakeWebRtcVideoEncoderFactory();
>     rtc::Thread* const signaling_thread = rtc::Thread::Current();
>-    peer_connection_factory_ = webrtc::CreatePeerConnectionFactory(
>-        network_thread, worker_thread, signaling_thread,
>-        fake_audio_capture_module_, webrtc::CreateBuiltinAudioEncoderFactory(),
>-        webrtc::CreateBuiltinAudioDecoderFactory(), fake_video_encoder_factory_,
>-        fake_video_decoder_factory_);
>+
>+    webrtc::PeerConnectionFactoryDependencies pc_factory_dependencies;
>+    pc_factory_dependencies.network_thread = network_thread;
>+    pc_factory_dependencies.worker_thread = worker_thread;
>+    pc_factory_dependencies.signaling_thread = signaling_thread;
>+    pc_factory_dependencies.media_engine =
>+        cricket::WebRtcMediaEngineFactory::Create(
>+            rtc::scoped_refptr<webrtc::AudioDeviceModule>(
>+                fake_audio_capture_module_),
>+            webrtc::CreateBuiltinAudioEncoderFactory(),
>+            webrtc::CreateBuiltinAudioDecoderFactory(),
>+            webrtc::CreateBuiltinVideoEncoderFactory(),
>+            webrtc::CreateBuiltinVideoDecoderFactory(), nullptr,
>+            webrtc::AudioProcessingBuilder().Create());
>+    pc_factory_dependencies.call_factory = webrtc::CreateCallFactory();
>+    if (event_log_factory) {
>+      event_log_factory_ = event_log_factory.get();
>+      pc_factory_dependencies.event_log_factory = std::move(event_log_factory);
>+    } else {
>+      pc_factory_dependencies.event_log_factory =
>+          webrtc::CreateRtcEventLogFactory();
>+    }
>+    peer_connection_factory_ = webrtc::CreateModularPeerConnectionFactory(
>+        std::move(pc_factory_dependencies));
>+
>     if (!peer_connection_factory_) {
>       return false;
>     }
>     if (options) {
>       peer_connection_factory_->SetOptions(*options);
>     }
>+    if (config) {
>+      sdp_semantics_ = config->sdp_semantics;
>+    }
>+
>+    dependencies.allocator = std::move(port_allocator);
>     peer_connection_ =
>-        CreatePeerConnection(std::move(port_allocator), constraints, config,
>-                             std::move(cert_generator));
>+        CreatePeerConnection(constraints, config, std::move(dependencies));
>     return peer_connection_.get() != nullptr;
>   }
> 
>   rtc::scoped_refptr<webrtc::PeerConnectionInterface> CreatePeerConnection(
>-      std::unique_ptr<cricket::PortAllocator> port_allocator,
>       const MediaConstraintsInterface* constraints,
>       const PeerConnectionInterface::RTCConfiguration* config,
>-      std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator) {
>+      webrtc::PeerConnectionDependencies dependencies) {
>     PeerConnectionInterface::RTCConfiguration modified_config;
>     // If |config| is null, this will result in a default configuration being
>     // used.
>@@ -631,9 +640,15 @@ class PeerConnectionWrapper : public webrtc::PeerConnectionObserver,
>     // ratios and not specific resolutions, is this even necessary?
>     modified_config.set_cpu_adaptation(false);
> 
>+    // Use the legacy interface.
>+    if (constraints != nullptr) {
>+      return peer_connection_factory_->CreatePeerConnection(
>+          modified_config, constraints, std::move(dependencies.allocator),
>+          std::move(dependencies.cert_generator), this);
>+    }
>+    dependencies.observer = this;
>     return peer_connection_factory_->CreatePeerConnection(
>-        modified_config, constraints, std::move(port_allocator),
>-        std::move(cert_generator), this);
>+        modified_config, std::move(dependencies));
>   }
> 
>   void set_signaling_message_receiver(
>@@ -647,30 +662,18 @@ class PeerConnectionWrapper : public webrtc::PeerConnectionObserver,
>     signal_ice_candidates_ = signal;
>   }
> 
>-  void EnableVideoDecoderFactory() {
>-    video_decoder_factory_enabled_ = true;
>-    fake_video_decoder_factory_->AddSupportedVideoCodecType(
>-        webrtc::kVideoCodecVP8);
>-  }
>-
>   rtc::scoped_refptr<webrtc::VideoTrackInterface> CreateLocalVideoTrackInternal(
>-      const FakeConstraints& constraints,
>-      webrtc::VideoRotation rotation) {
>+      webrtc::FakePeriodicVideoSource::Config config) {
>     // Set max frame rate to 10fps to reduce the risk of test flakiness.
>     // TODO(deadbeef): Do something more robust.
>-    FakeConstraints source_constraints = constraints;
>-    source_constraints.SetMandatoryMaxFrameRate(10);
>-
>-    cricket::FakeVideoCapturer* fake_capturer =
>-        new webrtc::FakePeriodicVideoCapturer();
>-    fake_capturer->SetRotation(rotation);
>-    video_capturers_.push_back(fake_capturer);
>-    rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> source =
>-        peer_connection_factory_->CreateVideoSource(fake_capturer,
>-                                                    &source_constraints);
>+    config.frame_interval_ms = 100;
>+
>+    video_track_sources_.emplace_back(
>+        new rtc::RefCountedObject<webrtc::FakePeriodicVideoTrackSource>(
>+            config, false /* remote */));
>     rtc::scoped_refptr<webrtc::VideoTrackInterface> track(
>-        peer_connection_factory_->CreateVideoTrack(rtc::CreateRandomUuid(),
>-                                                   source));
>+        peer_connection_factory_->CreateVideoTrack(
>+            rtc::CreateRandomUuid(), video_track_sources_.back()));
>     if (!local_video_renderer_) {
>       local_video_renderer_.reset(new webrtc::FakeVideoTrackRenderer(track));
>     }
>@@ -689,6 +692,9 @@ class PeerConnectionWrapper : public webrtc::PeerConnectionObserver,
>     // Setting a remote description may have changed the number of receivers,
>     // so reset the receiver observers.
>     ResetRtpReceiverObservers();
>+    if (remote_offer_handler_) {
>+      remote_offer_handler_();
>+    }
>     auto answer = CreateAnswer();
>     ASSERT_NE(nullptr, answer);
>     EXPECT_TRUE(SetLocalDescriptionAndSendSdpMessage(std::move(answer)));
>@@ -750,6 +756,9 @@ class PeerConnectionWrapper : public webrtc::PeerConnectionObserver,
>     std::string sdp;
>     EXPECT_TRUE(desc->ToString(&sdp));
>     pc()->SetLocalDescription(observer, desc.release());
>+    if (sdp_semantics_ == SdpSemantics::kUnifiedPlan) {
>+      RemoveUnusedVideoRenderers();
>+    }
>     // As mentioned above, we need to send the message immediately after
>     // SetLocalDescription.
>     SendSdpMessage(type, sdp);
>@@ -762,10 +771,43 @@ class PeerConnectionWrapper : public webrtc::PeerConnectionObserver,
>         new rtc::RefCountedObject<MockSetSessionDescriptionObserver>());
>     RTC_LOG(LS_INFO) << debug_name_ << ": SetRemoteDescription";
>     pc()->SetRemoteDescription(observer, desc.release());
>+    if (sdp_semantics_ == SdpSemantics::kUnifiedPlan) {
>+      RemoveUnusedVideoRenderers();
>+    }
>     EXPECT_TRUE_WAIT(observer->called(), kDefaultTimeout);
>     return observer->result();
>   }
> 
>+  // This is a work around to remove unused fake_video_renderers from
>+  // transceivers that have either stopped or are no longer receiving.
>+  void RemoveUnusedVideoRenderers() {
>+    auto transceivers = pc()->GetTransceivers();
>+    for (auto& transceiver : transceivers) {
>+      if (transceiver->receiver()->media_type() != cricket::MEDIA_TYPE_VIDEO) {
>+        continue;
>+      }
>+      // Remove fake video renderers from any stopped transceivers.
>+      if (transceiver->stopped()) {
>+        auto it =
>+            fake_video_renderers_.find(transceiver->receiver()->track()->id());
>+        if (it != fake_video_renderers_.end()) {
>+          fake_video_renderers_.erase(it);
>+        }
>+      }
>+      // Remove fake video renderers from any transceivers that are no longer
>+      // receiving.
>+      if ((transceiver->current_direction() &&
>+           !webrtc::RtpTransceiverDirectionHasRecv(
>+               *transceiver->current_direction()))) {
>+        auto it =
>+            fake_video_renderers_.find(transceiver->receiver()->track()->id());
>+        if (it != fake_video_renderers_.end()) {
>+          fake_video_renderers_.erase(it);
>+        }
>+      }
>+    }
>+  }
>+
>   // Simulate sending a blob of SDP with delay |signaling_delay_ms_| (0 by
>   // default).
>   void SendSdpMessage(SdpType type, const std::string& msg) {
>@@ -834,19 +876,26 @@ class PeerConnectionWrapper : public webrtc::PeerConnectionObserver,
>       webrtc::PeerConnectionInterface::SignalingState new_state) override {
>     EXPECT_EQ(pc()->signaling_state(), new_state);
>   }
>-  void OnAddStream(
>-      rtc::scoped_refptr<MediaStreamInterface> media_stream) override {
>-    media_stream->RegisterObserver(this);
>-    for (size_t i = 0; i < media_stream->GetVideoTracks().size(); ++i) {
>-      const std::string id = media_stream->GetVideoTracks()[i]->id();
>-      ASSERT_TRUE(fake_video_renderers_.find(id) ==
>+  void OnAddTrack(rtc::scoped_refptr<RtpReceiverInterface> receiver,
>+                  const std::vector<rtc::scoped_refptr<MediaStreamInterface>>&
>+                      streams) override {
>+    if (receiver->media_type() == cricket::MEDIA_TYPE_VIDEO) {
>+      rtc::scoped_refptr<VideoTrackInterface> video_track(
>+          static_cast<VideoTrackInterface*>(receiver->track().get()));
>+      ASSERT_TRUE(fake_video_renderers_.find(video_track->id()) ==
>                   fake_video_renderers_.end());
>-      fake_video_renderers_[id].reset(new webrtc::FakeVideoTrackRenderer(
>-          media_stream->GetVideoTracks()[i]));
>+      fake_video_renderers_[video_track->id()] =
>+          absl::make_unique<FakeVideoTrackRenderer>(video_track);
>+    }
>+  }
>+  void OnRemoveTrack(
>+      rtc::scoped_refptr<RtpReceiverInterface> receiver) override {
>+    if (receiver->media_type() == cricket::MEDIA_TYPE_VIDEO) {
>+      auto it = fake_video_renderers_.find(receiver->track()->id());
>+      RTC_DCHECK(it != fake_video_renderers_.end());
>+      fake_video_renderers_.erase(it);
>     }
>   }
>-  void OnRemoveStream(
>-      rtc::scoped_refptr<MediaStreamInterface> media_stream) override {}
>   void OnRenegotiationNeeded() override {}
>   void OnIceConnectionChange(
>       webrtc::PeerConnectionInterface::IceConnectionState new_state) override {
>@@ -876,40 +925,6 @@ class PeerConnectionWrapper : public webrtc::PeerConnectionObserver,
>     data_observer_.reset(new MockDataChannelObserver(data_channel));
>   }
> 
>-  // MediaStreamInterface callback
>-  void OnChanged() override {
>-    // Track added or removed from MediaStream, so update our renderers.
>-    rtc::scoped_refptr<StreamCollectionInterface> remote_streams =
>-        pc()->remote_streams();
>-    // Remove renderers for tracks that were removed.
>-    for (auto it = fake_video_renderers_.begin();
>-         it != fake_video_renderers_.end();) {
>-      if (remote_streams->FindVideoTrack(it->first) == nullptr) {
>-        auto to_remove = it++;
>-        removed_fake_video_renderers_.push_back(std::move(to_remove->second));
>-        fake_video_renderers_.erase(to_remove);
>-      } else {
>-        ++it;
>-      }
>-    }
>-    // Create renderers for new video tracks.
>-    for (size_t stream_index = 0; stream_index < remote_streams->count();
>-         ++stream_index) {
>-      MediaStreamInterface* remote_stream = remote_streams->at(stream_index);
>-      for (size_t track_index = 0;
>-           track_index < remote_stream->GetVideoTracks().size();
>-           ++track_index) {
>-        const std::string id =
>-            remote_stream->GetVideoTracks()[track_index]->id();
>-        if (fake_video_renderers_.find(id) != fake_video_renderers_.end()) {
>-          continue;
>-        }
>-        fake_video_renderers_[id].reset(new webrtc::FakeVideoTrackRenderer(
>-            remote_stream->GetVideoTracks()[track_index]));
>-      }
>-    }
>-  }
>-
>   std::string debug_name_;
> 
>   std::unique_ptr<rtc::FakeNetworkManager> fake_network_manager_;
>@@ -927,26 +942,24 @@ class PeerConnectionWrapper : public webrtc::PeerConnectionObserver,
>   // Needed to ensure frames aren't received for removed tracks.
>   std::vector<std::unique_ptr<webrtc::FakeVideoTrackRenderer>>
>       removed_fake_video_renderers_;
>-  // Needed to keep track of number of frames received when external decoder
>-  // used.
>-  FakeWebRtcVideoDecoderFactory* fake_video_decoder_factory_ = nullptr;
>-  FakeWebRtcVideoEncoderFactory* fake_video_encoder_factory_ = nullptr;
>-  bool video_decoder_factory_enabled_ = false;
> 
>   // For remote peer communication.
>   SignalingMessageReceiver* signaling_message_receiver_ = nullptr;
>   int signaling_delay_ms_ = 0;
>   bool signal_ice_candidates_ = true;
> 
>-  // Store references to the video capturers we've created, so that we can stop
>+  // Store references to the video sources we've created, so that we can stop
>   // them, if required.
>-  std::vector<cricket::FakeVideoCapturer*> video_capturers_;
>+  std::vector<rtc::scoped_refptr<webrtc::VideoTrackSource>>
>+      video_track_sources_;
>   // |local_video_renderer_| attached to the first created local video track.
>   std::unique_ptr<webrtc::FakeVideoTrackRenderer> local_video_renderer_;
> 
>+  SdpSemantics sdp_semantics_;
>   PeerConnectionInterface::RTCOfferAnswerOptions offer_answer_options_;
>   std::function<void(cricket::SessionDescription*)> received_sdp_munger_;
>   std::function<void(cricket::SessionDescription*)> generated_sdp_munger_;
>+  std::function<void()> remote_offer_handler_;
> 
>   rtc::scoped_refptr<DataChannelInterface> data_channel_;
>   std::unique_ptr<MockDataChannelObserver> data_observer_;
>@@ -958,9 +971,11 @@ class PeerConnectionWrapper : public webrtc::PeerConnectionObserver,
>   std::vector<PeerConnectionInterface::IceGatheringState>
>       ice_gathering_state_history_;
> 
>+  webrtc::FakeRtcEventLogFactory* event_log_factory_;
>+
>   rtc::AsyncInvoker invoker_;
> 
>-  friend class PeerConnectionIntegrationTest;
>+  friend class PeerConnectionIntegrationBaseTest;
> };
> 
> class MockRtcEventLogOutput : public webrtc::RtcEventLogOutput {
>@@ -970,28 +985,151 @@ class MockRtcEventLogOutput : public webrtc::RtcEventLogOutput {
>   MOCK_METHOD1(Write, bool(const std::string&));
> };
> 
>+// This helper object is used for both specifying how many audio/video frames
>+// are expected to be received for a caller/callee. It provides helper functions
>+// to specify these expectations. The object initially starts in a state of no
>+// expectations.
>+class MediaExpectations {
>+ public:
>+  enum ExpectFrames {
>+    kExpectSomeFrames,
>+    kExpectNoFrames,
>+    kNoExpectation,
>+  };
>+
>+  void ExpectBidirectionalAudioAndVideo() {
>+    ExpectBidirectionalAudio();
>+    ExpectBidirectionalVideo();
>+  }
>+
>+  void ExpectBidirectionalAudio() {
>+    CallerExpectsSomeAudio();
>+    CalleeExpectsSomeAudio();
>+  }
>+
>+  void ExpectNoAudio() {
>+    CallerExpectsNoAudio();
>+    CalleeExpectsNoAudio();
>+  }
>+
>+  void ExpectBidirectionalVideo() {
>+    CallerExpectsSomeVideo();
>+    CalleeExpectsSomeVideo();
>+  }
>+
>+  void ExpectNoVideo() {
>+    CallerExpectsNoVideo();
>+    CalleeExpectsNoVideo();
>+  }
>+
>+  void CallerExpectsSomeAudioAndVideo() {
>+    CallerExpectsSomeAudio();
>+    CallerExpectsSomeVideo();
>+  }
>+
>+  void CalleeExpectsSomeAudioAndVideo() {
>+    CalleeExpectsSomeAudio();
>+    CalleeExpectsSomeVideo();
>+  }
>+
>+  // Caller's audio functions.
>+  void CallerExpectsSomeAudio(
>+      int expected_audio_frames = kDefaultExpectedAudioFrameCount) {
>+    caller_audio_expectation_ = kExpectSomeFrames;
>+    caller_audio_frames_expected_ = expected_audio_frames;
>+  }
>+
>+  void CallerExpectsNoAudio() {
>+    caller_audio_expectation_ = kExpectNoFrames;
>+    caller_audio_frames_expected_ = 0;
>+  }
>+
>+  // Caller's video functions.
>+  void CallerExpectsSomeVideo(
>+      int expected_video_frames = kDefaultExpectedVideoFrameCount) {
>+    caller_video_expectation_ = kExpectSomeFrames;
>+    caller_video_frames_expected_ = expected_video_frames;
>+  }
>+
>+  void CallerExpectsNoVideo() {
>+    caller_video_expectation_ = kExpectNoFrames;
>+    caller_video_frames_expected_ = 0;
>+  }
>+
>+  // Callee's audio functions.
>+  void CalleeExpectsSomeAudio(
>+      int expected_audio_frames = kDefaultExpectedAudioFrameCount) {
>+    callee_audio_expectation_ = kExpectSomeFrames;
>+    callee_audio_frames_expected_ = expected_audio_frames;
>+  }
>+
>+  void CalleeExpectsNoAudio() {
>+    callee_audio_expectation_ = kExpectNoFrames;
>+    callee_audio_frames_expected_ = 0;
>+  }
>+
>+  // Callee's video functions.
>+  void CalleeExpectsSomeVideo(
>+      int expected_video_frames = kDefaultExpectedVideoFrameCount) {
>+    callee_video_expectation_ = kExpectSomeFrames;
>+    callee_video_frames_expected_ = expected_video_frames;
>+  }
>+
>+  void CalleeExpectsNoVideo() {
>+    callee_video_expectation_ = kExpectNoFrames;
>+    callee_video_frames_expected_ = 0;
>+  }
>+
>+  ExpectFrames caller_audio_expectation_ = kNoExpectation;
>+  ExpectFrames caller_video_expectation_ = kNoExpectation;
>+  ExpectFrames callee_audio_expectation_ = kNoExpectation;
>+  ExpectFrames callee_video_expectation_ = kNoExpectation;
>+  int caller_audio_frames_expected_ = 0;
>+  int caller_video_frames_expected_ = 0;
>+  int callee_audio_frames_expected_ = 0;
>+  int callee_video_frames_expected_ = 0;
>+};
>+
> // Tests two PeerConnections connecting to each other end-to-end, using a
> // virtual network, fake A/V capture and fake encoder/decoders. The
> // PeerConnections share the threads/socket servers, but use separate versions
> // of everything else (including "PeerConnectionFactory"s).
>-class PeerConnectionIntegrationTest : public testing::Test {
>+class PeerConnectionIntegrationBaseTest : public testing::Test {
>  public:
>-  PeerConnectionIntegrationTest()
>-      : ss_(new rtc::VirtualSocketServer()),
>+  explicit PeerConnectionIntegrationBaseTest(SdpSemantics sdp_semantics)
>+      : sdp_semantics_(sdp_semantics),
>+        ss_(new rtc::VirtualSocketServer()),
>         fss_(new rtc::FirewallSocketServer(ss_.get())),
>         network_thread_(new rtc::Thread(fss_.get())),
>         worker_thread_(rtc::Thread::Create()) {
>+    network_thread_->SetName("PCNetworkThread", this);
>+    worker_thread_->SetName("PCWorkerThread", this);
>     RTC_CHECK(network_thread_->Start());
>     RTC_CHECK(worker_thread_->Start());
>+    webrtc::metrics::Reset();
>   }
> 
>-  ~PeerConnectionIntegrationTest() {
>+  ~PeerConnectionIntegrationBaseTest() {
>+    // The PeerConnections should deleted before the TurnCustomizers.
>+    // A TurnPort is created with a raw pointer to a TurnCustomizer. The
>+    // TurnPort has the same lifetime as the PeerConnection, so it's expected
>+    // that the TurnCustomizer outlives the life of the PeerConnection or else
>+    // when Send() is called it will hit a seg fault.
>     if (caller_) {
>       caller_->set_signaling_message_receiver(nullptr);
>+      delete SetCallerPcWrapperAndReturnCurrent(nullptr);
>     }
>     if (callee_) {
>       callee_->set_signaling_message_receiver(nullptr);
>+      delete SetCalleePcWrapperAndReturnCurrent(nullptr);
>     }
>+
>+    // If turn servers were created for the test they need to be destroyed on
>+    // the network thread.
>+    network_thread()->Invoke<void>(RTC_FROM_HERE, [this] {
>+      turn_servers_.clear();
>+      turn_customizers_.clear();
>+    });
>   }
> 
>   bool SignalingStateStable() {
>@@ -1012,53 +1150,194 @@ class PeerConnectionIntegrationTest : public testing::Test {
>                 webrtc::PeerConnectionInterface::kIceConnectionCompleted);
>   }
> 
>+  // When |event_log_factory| is null, the default implementation of the event
>+  // log factory will be used.
>+  std::unique_ptr<PeerConnectionWrapper> CreatePeerConnectionWrapper(
>+      const std::string& debug_name,
>+      const MediaConstraintsInterface* constraints,
>+      const PeerConnectionFactory::Options* options,
>+      const RTCConfiguration* config,
>+      webrtc::PeerConnectionDependencies dependencies,
>+      std::unique_ptr<webrtc::FakeRtcEventLogFactory> event_log_factory) {
>+    RTCConfiguration modified_config;
>+    if (config) {
>+      modified_config = *config;
>+    }
>+    modified_config.sdp_semantics = sdp_semantics_;
>+    if (!dependencies.cert_generator) {
>+      dependencies.cert_generator =
>+          absl::make_unique<FakeRTCCertificateGenerator>();
>+    }
>+    std::unique_ptr<PeerConnectionWrapper> client(
>+        new PeerConnectionWrapper(debug_name));
>+
>+    if (!client->Init(constraints, options, &modified_config,
>+                      std::move(dependencies), network_thread_.get(),
>+                      worker_thread_.get(), std::move(event_log_factory))) {
>+      return nullptr;
>+    }
>+    return client;
>+  }
>+
>+  std::unique_ptr<PeerConnectionWrapper>
>+  CreatePeerConnectionWrapperWithFakeRtcEventLog(
>+      const std::string& debug_name,
>+      const MediaConstraintsInterface* constraints,
>+      const PeerConnectionFactory::Options* options,
>+      const RTCConfiguration* config,
>+      webrtc::PeerConnectionDependencies dependencies) {
>+    std::unique_ptr<webrtc::FakeRtcEventLogFactory> event_log_factory(
>+        new webrtc::FakeRtcEventLogFactory(rtc::Thread::Current()));
>+    return CreatePeerConnectionWrapper(debug_name, constraints, options, config,
>+                                       std::move(dependencies),
>+                                       std::move(event_log_factory));
>+  }
>+
>   bool CreatePeerConnectionWrappers() {
>     return CreatePeerConnectionWrappersWithConfig(
>         PeerConnectionInterface::RTCConfiguration(),
>         PeerConnectionInterface::RTCConfiguration());
>   }
> 
>+  bool CreatePeerConnectionWrappersWithSdpSemantics(
>+      SdpSemantics caller_semantics,
>+      SdpSemantics callee_semantics) {
>+    // Can't specify the sdp_semantics in the passed-in configuration since it
>+    // will be overwritten by CreatePeerConnectionWrapper with whatever is
>+    // stored in sdp_semantics_. So get around this by modifying the instance
>+    // variable before calling CreatePeerConnectionWrapper for the caller and
>+    // callee PeerConnections.
>+    SdpSemantics original_semantics = sdp_semantics_;
>+    sdp_semantics_ = caller_semantics;
>+    caller_ = CreatePeerConnectionWrapper(
>+        "Caller", nullptr, nullptr, nullptr,
>+        webrtc::PeerConnectionDependencies(nullptr), nullptr);
>+    sdp_semantics_ = callee_semantics;
>+    callee_ = CreatePeerConnectionWrapper(
>+        "Callee", nullptr, nullptr, nullptr,
>+        webrtc::PeerConnectionDependencies(nullptr), nullptr);
>+    sdp_semantics_ = original_semantics;
>+    return caller_ && callee_;
>+  }
>+
>   bool CreatePeerConnectionWrappersWithConstraints(
>       MediaConstraintsInterface* caller_constraints,
>       MediaConstraintsInterface* callee_constraints) {
>-    caller_.reset(PeerConnectionWrapper::CreateWithConstraints(
>-        "Caller", caller_constraints, network_thread_.get(),
>-        worker_thread_.get()));
>-    callee_.reset(PeerConnectionWrapper::CreateWithConstraints(
>-        "Callee", callee_constraints, network_thread_.get(),
>-        worker_thread_.get()));
>+    caller_ = CreatePeerConnectionWrapper(
>+        "Caller", caller_constraints, nullptr, nullptr,
>+        webrtc::PeerConnectionDependencies(nullptr), nullptr);
>+    callee_ = CreatePeerConnectionWrapper(
>+        "Callee", callee_constraints, nullptr, nullptr,
>+        webrtc::PeerConnectionDependencies(nullptr), nullptr);
>+
>     return caller_ && callee_;
>   }
> 
>   bool CreatePeerConnectionWrappersWithConfig(
>       const PeerConnectionInterface::RTCConfiguration& caller_config,
>       const PeerConnectionInterface::RTCConfiguration& callee_config) {
>-    caller_.reset(PeerConnectionWrapper::CreateWithConfig(
>-        "Caller", caller_config, network_thread_.get(), worker_thread_.get()));
>-    callee_.reset(PeerConnectionWrapper::CreateWithConfig(
>-        "Callee", callee_config, network_thread_.get(), worker_thread_.get()));
>+    caller_ = CreatePeerConnectionWrapper(
>+        "Caller", nullptr, nullptr, &caller_config,
>+        webrtc::PeerConnectionDependencies(nullptr), nullptr);
>+    callee_ = CreatePeerConnectionWrapper(
>+        "Callee", nullptr, nullptr, &callee_config,
>+        webrtc::PeerConnectionDependencies(nullptr), nullptr);
>+    return caller_ && callee_;
>+  }
>+
>+  bool CreatePeerConnectionWrappersWithConfigAndDeps(
>+      const PeerConnectionInterface::RTCConfiguration& caller_config,
>+      webrtc::PeerConnectionDependencies caller_dependencies,
>+      const PeerConnectionInterface::RTCConfiguration& callee_config,
>+      webrtc::PeerConnectionDependencies callee_dependencies) {
>+    caller_ =
>+        CreatePeerConnectionWrapper("Caller", nullptr, nullptr, &caller_config,
>+                                    std::move(caller_dependencies), nullptr);
>+    callee_ =
>+        CreatePeerConnectionWrapper("Callee", nullptr, nullptr, &callee_config,
>+                                    std::move(callee_dependencies), nullptr);
>     return caller_ && callee_;
>   }
> 
>   bool CreatePeerConnectionWrappersWithOptions(
>       const PeerConnectionFactory::Options& caller_options,
>       const PeerConnectionFactory::Options& callee_options) {
>-    caller_.reset(PeerConnectionWrapper::CreateWithOptions(
>-        "Caller", caller_options, network_thread_.get(), worker_thread_.get()));
>-    callee_.reset(PeerConnectionWrapper::CreateWithOptions(
>-        "Callee", callee_options, network_thread_.get(), worker_thread_.get()));
>+    caller_ = CreatePeerConnectionWrapper(
>+        "Caller", nullptr, &caller_options, nullptr,
>+        webrtc::PeerConnectionDependencies(nullptr), nullptr);
>+    callee_ = CreatePeerConnectionWrapper(
>+        "Callee", nullptr, &callee_options, nullptr,
>+        webrtc::PeerConnectionDependencies(nullptr), nullptr);
>     return caller_ && callee_;
>   }
> 
>-  PeerConnectionWrapper* CreatePeerConnectionWrapperWithAlternateKey() {
>+  bool CreatePeerConnectionWrappersWithFakeRtcEventLog() {
>+    PeerConnectionInterface::RTCConfiguration default_config;
>+    caller_ = CreatePeerConnectionWrapperWithFakeRtcEventLog(
>+        "Caller", nullptr, nullptr, &default_config,
>+        webrtc::PeerConnectionDependencies(nullptr));
>+    callee_ = CreatePeerConnectionWrapperWithFakeRtcEventLog(
>+        "Callee", nullptr, nullptr, &default_config,
>+        webrtc::PeerConnectionDependencies(nullptr));
>+    return caller_ && callee_;
>+  }
>+
>+  std::unique_ptr<PeerConnectionWrapper>
>+  CreatePeerConnectionWrapperWithAlternateKey() {
>     std::unique_ptr<FakeRTCCertificateGenerator> cert_generator(
>         new FakeRTCCertificateGenerator());
>     cert_generator->use_alternate_key();
> 
>-    // Make sure the new client is using a different certificate.
>-    return PeerConnectionWrapper::CreateWithDtlsIdentityStore(
>-        "New Peer", std::move(cert_generator), network_thread_.get(),
>-        worker_thread_.get());
>+    webrtc::PeerConnectionDependencies dependencies(nullptr);
>+    dependencies.cert_generator = std::move(cert_generator);
>+    return CreatePeerConnectionWrapper("New Peer", nullptr, nullptr, nullptr,
>+                                       std::move(dependencies), nullptr);
>+  }
>+
>+  cricket::TestTurnServer* CreateTurnServer(
>+      rtc::SocketAddress internal_address,
>+      rtc::SocketAddress external_address,
>+      cricket::ProtocolType type = cricket::ProtocolType::PROTO_UDP,
>+      const std::string& common_name = "test turn server") {
>+    rtc::Thread* thread = network_thread();
>+    std::unique_ptr<cricket::TestTurnServer> turn_server =
>+        network_thread()->Invoke<std::unique_ptr<cricket::TestTurnServer>>(
>+            RTC_FROM_HERE,
>+            [thread, internal_address, external_address, type, common_name] {
>+              return absl::make_unique<cricket::TestTurnServer>(
>+                  thread, internal_address, external_address, type,
>+                  /*ignore_bad_certs=*/true, common_name);
>+            });
>+    turn_servers_.push_back(std::move(turn_server));
>+    // Interactions with the turn server should be done on the network thread.
>+    return turn_servers_.back().get();
>+  }
>+
>+  cricket::TestTurnCustomizer* CreateTurnCustomizer() {
>+    std::unique_ptr<cricket::TestTurnCustomizer> turn_customizer =
>+        network_thread()->Invoke<std::unique_ptr<cricket::TestTurnCustomizer>>(
>+            RTC_FROM_HERE,
>+            [] { return absl::make_unique<cricket::TestTurnCustomizer>(); });
>+    turn_customizers_.push_back(std::move(turn_customizer));
>+    // Interactions with the turn customizer should be done on the network
>+    // thread.
>+    return turn_customizers_.back().get();
>+  }
>+
>+  // Checks that the function counters for a TestTurnCustomizer are greater than
>+  // 0.
>+  void ExpectTurnCustomizerCountersIncremented(
>+      cricket::TestTurnCustomizer* turn_customizer) {
>+    unsigned int allow_channel_data_counter =
>+        network_thread()->Invoke<unsigned int>(
>+            RTC_FROM_HERE, [turn_customizer] {
>+              return turn_customizer->allow_channel_data_cnt_;
>+            });
>+    EXPECT_GT(allow_channel_data_counter, 0u);
>+    unsigned int modify_counter = network_thread()->Invoke<unsigned int>(
>+        RTC_FROM_HERE,
>+        [turn_customizer] { return turn_customizer->modify_cnt_; });
>+    EXPECT_GT(modify_counter, 0u);
>   }
> 
>   // Once called, SDP blobs and ICE candidates will be automatically signaled
>@@ -1086,11 +1365,6 @@ class PeerConnectionIntegrationTest : public testing::Test {
>     callee_->set_signal_ice_candidates(signal);
>   }
> 
>-  void EnableVideoDecoderFactory() {
>-    caller_->EnableVideoDecoderFactory();
>-    callee_->EnableVideoDecoderFactory();
>-  }
>-
>   // Messages may get lost on the unreliable DataChannel, so we send multiple
>   // times to avoid test flakiness.
>   void SendRtpDataWithRetries(webrtc::DataChannelInterface* dc,
>@@ -1129,46 +1403,128 @@ class PeerConnectionIntegrationTest : public testing::Test {
> 
>   rtc::FirewallSocketServer* firewall() const { return fss_.get(); }
> 
>-  // Expects the provided number of new frames to be received within |wait_ms|.
>-  // "New frames" meaning that it waits for the current frame counts to
>-  // *increase* by the provided values. For video, uses
>-  // RecievedVideoFramesForEachTrack for the case of multiple video tracks
>-  // being received.
>-  void ExpectNewFramesReceivedWithWait(
>-      int expected_caller_received_audio_frames,
>-      int expected_caller_received_video_frames,
>-      int expected_callee_received_audio_frames,
>-      int expected_callee_received_video_frames,
>-      int wait_ms) {
>-    // Add current frame counts to the provided values, in order to wait for
>-    // the frame count to increase.
>-    expected_caller_received_audio_frames += caller()->audio_frames_received();
>-    expected_caller_received_video_frames +=
>+  // Expects the provided number of new frames to be received within
>+  // kMaxWaitForFramesMs. The new expected frames are specified in
>+  // |media_expectations|. Returns false if any of the expectations were
>+  // not met.
>+  bool ExpectNewFrames(const MediaExpectations& media_expectations) {
>+    // First initialize the expected frame counts based upon the current
>+    // frame count.
>+    int total_caller_audio_frames_expected = caller()->audio_frames_received();
>+    if (media_expectations.caller_audio_expectation_ ==
>+        MediaExpectations::kExpectSomeFrames) {
>+      total_caller_audio_frames_expected +=
>+          media_expectations.caller_audio_frames_expected_;
>+    }
>+    int total_caller_video_frames_expected =
>         caller()->min_video_frames_received_per_track();
>-    expected_callee_received_audio_frames += callee()->audio_frames_received();
>-    expected_callee_received_video_frames +=
>+    if (media_expectations.caller_video_expectation_ ==
>+        MediaExpectations::kExpectSomeFrames) {
>+      total_caller_video_frames_expected +=
>+          media_expectations.caller_video_frames_expected_;
>+    }
>+    int total_callee_audio_frames_expected = callee()->audio_frames_received();
>+    if (media_expectations.callee_audio_expectation_ ==
>+        MediaExpectations::kExpectSomeFrames) {
>+      total_callee_audio_frames_expected +=
>+          media_expectations.callee_audio_frames_expected_;
>+    }
>+    int total_callee_video_frames_expected =
>         callee()->min_video_frames_received_per_track();
>+    if (media_expectations.callee_video_expectation_ ==
>+        MediaExpectations::kExpectSomeFrames) {
>+      total_callee_video_frames_expected +=
>+          media_expectations.callee_video_frames_expected_;
>+    }
> 
>+    // Wait for the expected frames.
>     EXPECT_TRUE_WAIT(caller()->audio_frames_received() >=
>-                             expected_caller_received_audio_frames &&
>+                             total_caller_audio_frames_expected &&
>                          caller()->min_video_frames_received_per_track() >=
>-                             expected_caller_received_video_frames &&
>+                             total_caller_video_frames_expected &&
>                          callee()->audio_frames_received() >=
>-                             expected_callee_received_audio_frames &&
>+                             total_callee_audio_frames_expected &&
>                          callee()->min_video_frames_received_per_track() >=
>-                             expected_callee_received_video_frames,
>-                     wait_ms);
>-
>-    // After the combined wait, do an "expect" for each individual count, to
>-    // print out a more detailed message upon failure.
>+                             total_callee_video_frames_expected,
>+                     kMaxWaitForFramesMs);
>+    bool expectations_correct =
>+        caller()->audio_frames_received() >=
>+            total_caller_audio_frames_expected &&
>+        caller()->min_video_frames_received_per_track() >=
>+            total_caller_video_frames_expected &&
>+        callee()->audio_frames_received() >=
>+            total_callee_audio_frames_expected &&
>+        callee()->min_video_frames_received_per_track() >=
>+            total_callee_video_frames_expected;
>+
>+    // After the combined wait, print out a more detailed message upon
>+    // failure.
>     EXPECT_GE(caller()->audio_frames_received(),
>-              expected_caller_received_audio_frames);
>+              total_caller_audio_frames_expected);
>     EXPECT_GE(caller()->min_video_frames_received_per_track(),
>-              expected_caller_received_video_frames);
>+              total_caller_video_frames_expected);
>     EXPECT_GE(callee()->audio_frames_received(),
>-              expected_callee_received_audio_frames);
>+              total_callee_audio_frames_expected);
>     EXPECT_GE(callee()->min_video_frames_received_per_track(),
>-              expected_callee_received_video_frames);
>+              total_callee_video_frames_expected);
>+
>+    // We want to make sure nothing unexpected was received.
>+    if (media_expectations.caller_audio_expectation_ ==
>+        MediaExpectations::kExpectNoFrames) {
>+      EXPECT_EQ(caller()->audio_frames_received(),
>+                total_caller_audio_frames_expected);
>+      if (caller()->audio_frames_received() !=
>+          total_caller_audio_frames_expected) {
>+        expectations_correct = false;
>+      }
>+    }
>+    if (media_expectations.caller_video_expectation_ ==
>+        MediaExpectations::kExpectNoFrames) {
>+      EXPECT_EQ(caller()->min_video_frames_received_per_track(),
>+                total_caller_video_frames_expected);
>+      if (caller()->min_video_frames_received_per_track() !=
>+          total_caller_video_frames_expected) {
>+        expectations_correct = false;
>+      }
>+    }
>+    if (media_expectations.callee_audio_expectation_ ==
>+        MediaExpectations::kExpectNoFrames) {
>+      EXPECT_EQ(callee()->audio_frames_received(),
>+                total_callee_audio_frames_expected);
>+      if (callee()->audio_frames_received() !=
>+          total_callee_audio_frames_expected) {
>+        expectations_correct = false;
>+      }
>+    }
>+    if (media_expectations.callee_video_expectation_ ==
>+        MediaExpectations::kExpectNoFrames) {
>+      EXPECT_EQ(callee()->min_video_frames_received_per_track(),
>+                total_callee_video_frames_expected);
>+      if (callee()->min_video_frames_received_per_track() !=
>+          total_callee_video_frames_expected) {
>+        expectations_correct = false;
>+      }
>+    }
>+    return expectations_correct;
>+  }
>+
>+  void TestNegotiatedCipherSuite(
>+      const PeerConnectionFactory::Options& caller_options,
>+      const PeerConnectionFactory::Options& callee_options,
>+      int expected_cipher_suite) {
>+    ASSERT_TRUE(CreatePeerConnectionWrappersWithOptions(caller_options,
>+                                                        callee_options));
>+    ConnectFakeSignaling();
>+    caller()->AddAudioVideoTracks();
>+    callee()->AddAudioVideoTracks();
>+    caller()->CreateAndSetAndSignalOffer();
>+    ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
>+    EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(expected_cipher_suite),
>+                   caller()->OldGetStats()->SrtpCipher(), kDefaultTimeout);
>+    // TODO(bugs.webrtc.org/9456): Fix it.
>+    EXPECT_EQ(1, webrtc::metrics::NumEvents(
>+                     "WebRTC.PeerConnection.SrtpCryptoSuite.Audio",
>+                     expected_cipher_suite));
>   }
> 
>   void TestGcmNegotiationUsesCipherSuite(bool local_gcm_enabled,
>@@ -1178,24 +1534,13 @@ class PeerConnectionIntegrationTest : public testing::Test {
>     caller_options.crypto_options.enable_gcm_crypto_suites = local_gcm_enabled;
>     PeerConnectionFactory::Options callee_options;
>     callee_options.crypto_options.enable_gcm_crypto_suites = remote_gcm_enabled;
>-    ASSERT_TRUE(CreatePeerConnectionWrappersWithOptions(caller_options,
>-                                                        callee_options));
>-    rtc::scoped_refptr<webrtc::FakeMetricsObserver> caller_observer =
>-        new rtc::RefCountedObject<webrtc::FakeMetricsObserver>();
>-    caller()->pc()->RegisterUMAObserver(caller_observer);
>-    ConnectFakeSignaling();
>-    caller()->AddAudioVideoMediaStream();
>-    callee()->AddAudioVideoMediaStream();
>-    caller()->CreateAndSetAndSignalOffer();
>-    ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>-    EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(expected_cipher_suite),
>-                   caller()->OldGetStats()->SrtpCipher(), kDefaultTimeout);
>-    EXPECT_EQ(
>-        1, caller_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher,
>-                                           expected_cipher_suite));
>-    caller()->pc()->RegisterUMAObserver(nullptr);
>+    TestNegotiatedCipherSuite(caller_options, callee_options,
>+                              expected_cipher_suite);
>   }
> 
>+ protected:
>+  SdpSemantics sdp_semantics_;
>+
>  private:
>   // |ss_| is used by |network_thread_| so it must be destroyed later.
>   std::unique_ptr<rtc::VirtualSocketServer> ss_;
>@@ -1205,25 +1550,52 @@ class PeerConnectionIntegrationTest : public testing::Test {
>   // later.
>   std::unique_ptr<rtc::Thread> network_thread_;
>   std::unique_ptr<rtc::Thread> worker_thread_;
>+  // The turn servers and turn customizers should be accessed & deleted on the
>+  // network thread to avoid a race with the socket read/write that occurs
>+  // on the network thread.
>+  std::vector<std::unique_ptr<cricket::TestTurnServer>> turn_servers_;
>+  std::vector<std::unique_ptr<cricket::TestTurnCustomizer>> turn_customizers_;
>   std::unique_ptr<PeerConnectionWrapper> caller_;
>   std::unique_ptr<PeerConnectionWrapper> callee_;
> };
> 
>+class PeerConnectionIntegrationTest
>+    : public PeerConnectionIntegrationBaseTest,
>+      public ::testing::WithParamInterface<SdpSemantics> {
>+ protected:
>+  PeerConnectionIntegrationTest()
>+      : PeerConnectionIntegrationBaseTest(GetParam()) {}
>+};
>+
>+class PeerConnectionIntegrationTestPlanB
>+    : public PeerConnectionIntegrationBaseTest {
>+ protected:
>+  PeerConnectionIntegrationTestPlanB()
>+      : PeerConnectionIntegrationBaseTest(SdpSemantics::kPlanB) {}
>+};
>+
>+class PeerConnectionIntegrationTestUnifiedPlan
>+    : public PeerConnectionIntegrationBaseTest {
>+ protected:
>+  PeerConnectionIntegrationTestUnifiedPlan()
>+      : PeerConnectionIntegrationBaseTest(SdpSemantics::kUnifiedPlan) {}
>+};
>+
> // Test the OnFirstPacketReceived callback from audio/video RtpReceivers.  This
> // includes testing that the callback is invoked if an observer is connected
> // after the first packet has already been received.
>-TEST_F(PeerConnectionIntegrationTest,
>+TEST_P(PeerConnectionIntegrationTest,
>        RtpReceiverObserverOnFirstPacketReceived) {
>   ASSERT_TRUE(CreatePeerConnectionWrappers());
>   ConnectFakeSignaling();
>-  caller()->AddAudioVideoMediaStream();
>-  callee()->AddAudioVideoMediaStream();
>+  caller()->AddAudioVideoTracks();
>+  callee()->AddAudioVideoTracks();
>   // Start offer/answer exchange and wait for it to complete.
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>   // Should be one receiver each for audio/video.
>-  EXPECT_EQ(2, caller()->rtp_receiver_observers().size());
>-  EXPECT_EQ(2, callee()->rtp_receiver_observers().size());
>+  EXPECT_EQ(2U, caller()->rtp_receiver_observers().size());
>+  EXPECT_EQ(2U, callee()->rtp_receiver_observers().size());
>   // Wait for all "first packet received" callbacks to be fired.
>   EXPECT_TRUE_WAIT(
>       std::all_of(caller()->rtp_receiver_observers().begin(),
>@@ -1243,8 +1615,8 @@ TEST_F(PeerConnectionIntegrationTest,
>   // callback should still be invoked.
>   caller()->ResetRtpReceiverObservers();
>   callee()->ResetRtpReceiverObservers();
>-  EXPECT_EQ(2, caller()->rtp_receiver_observers().size());
>-  EXPECT_EQ(2, callee()->rtp_receiver_observers().size());
>+  EXPECT_EQ(2U, caller()->rtp_receiver_observers().size());
>+  EXPECT_EQ(2U, callee()->rtp_receiver_observers().size());
>   EXPECT_TRUE(
>       std::all_of(caller()->rtp_receiver_observers().begin(),
>                   caller()->rtp_receiver_observers().end(),
>@@ -1283,14 +1655,11 @@ class DummyDtmfObserver : public DtmfSenderObserverInterface {
> // exchange is done.
> void TestDtmfFromSenderToReceiver(PeerConnectionWrapper* sender,
>                                   PeerConnectionWrapper* receiver) {
>+  // We should be able to get a DTMF sender from the local sender.
>+  rtc::scoped_refptr<DtmfSenderInterface> dtmf_sender =
>+      sender->pc()->GetSenders().at(0)->GetDtmfSender();
>+  ASSERT_TRUE(dtmf_sender);
>   DummyDtmfObserver observer;
>-  rtc::scoped_refptr<DtmfSenderInterface> dtmf_sender;
>-
>-  // We should be able to create a DTMF sender from a local track.
>-  webrtc::AudioTrackInterface* localtrack =
>-      sender->local_streams()->at(0)->GetAudioTracks()[0];
>-  dtmf_sender = sender->pc()->CreateDtmfSender(localtrack);
>-  ASSERT_NE(nullptr, dtmf_sender.get());
>   dtmf_sender->RegisterObserver(&observer);
> 
>   // Test the DtmfSender object just created.
>@@ -1306,12 +1675,12 @@ void TestDtmfFromSenderToReceiver(PeerConnectionWrapper* sender,
> 
> // Verifies the DtmfSenderObserver callbacks for a DtmfSender (one in each
> // direction).
>-TEST_F(PeerConnectionIntegrationTest, DtmfSenderObserver) {
>+TEST_P(PeerConnectionIntegrationTest, DtmfSenderObserver) {
>   ASSERT_TRUE(CreatePeerConnectionWrappers());
>   ConnectFakeSignaling();
>   // Only need audio for DTMF.
>-  caller()->AddAudioOnlyMediaStream();
>-  callee()->AddAudioOnlyMediaStream();
>+  caller()->AddAudioTrack();
>+  callee()->AddAudioTrack();
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>   // DTLS must finish before the DTMF sender can be used reliably.
>@@ -1322,23 +1691,27 @@ TEST_F(PeerConnectionIntegrationTest, DtmfSenderObserver) {
> 
> // Basic end-to-end test, verifying media can be encoded/transmitted/decoded
> // between two connections, using DTLS-SRTP.
>-TEST_F(PeerConnectionIntegrationTest, EndToEndCallWithDtls) {
>+TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithDtls) {
>   ASSERT_TRUE(CreatePeerConnectionWrappers());
>   ConnectFakeSignaling();
>+
>   // Do normal offer/answer and wait for some frames to be received in each
>   // direction.
>-  caller()->AddAudioVideoMediaStream();
>-  callee()->AddAudioVideoMediaStream();
>+  caller()->AddAudioVideoTracks();
>+  callee()->AddAudioVideoTracks();
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>-  ExpectNewFramesReceivedWithWait(
>-      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
>-      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
>-      kMaxWaitForFramesMs);
>+  MediaExpectations media_expectations;
>+  media_expectations.ExpectBidirectionalAudioAndVideo();
>+  ASSERT_TRUE(ExpectNewFrames(media_expectations));
>+  EXPECT_LE(2, webrtc::metrics::NumEvents("WebRTC.PeerConnection.KeyProtocol",
>+                                          webrtc::kEnumCounterKeyProtocolDtls));
>+  EXPECT_EQ(0, webrtc::metrics::NumEvents("WebRTC.PeerConnection.KeyProtocol",
>+                                          webrtc::kEnumCounterKeyProtocolSdes));
> }
> 
> // Uses SDES instead of DTLS for key agreement.
>-TEST_F(PeerConnectionIntegrationTest, EndToEndCallWithSdes) {
>+TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithSdes) {
>   PeerConnectionInterface::RTCConfiguration sdes_config;
>   sdes_config.enable_dtls_srtp.emplace(false);
>   ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(sdes_config, sdes_config));
>@@ -1346,25 +1719,33 @@ TEST_F(PeerConnectionIntegrationTest, EndToEndCallWithSdes) {
> 
>   // Do normal offer/answer and wait for some frames to be received in each
>   // direction.
>-  caller()->AddAudioVideoMediaStream();
>-  callee()->AddAudioVideoMediaStream();
>+  caller()->AddAudioVideoTracks();
>+  callee()->AddAudioVideoTracks();
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>-  ExpectNewFramesReceivedWithWait(
>-      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
>-      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
>-      kMaxWaitForFramesMs);
>+  MediaExpectations media_expectations;
>+  media_expectations.ExpectBidirectionalAudioAndVideo();
>+  ASSERT_TRUE(ExpectNewFrames(media_expectations));
>+  EXPECT_LE(2, webrtc::metrics::NumEvents("WebRTC.PeerConnection.KeyProtocol",
>+                                          webrtc::kEnumCounterKeyProtocolSdes));
>+  EXPECT_EQ(0, webrtc::metrics::NumEvents("WebRTC.PeerConnection.KeyProtocol",
>+                                          webrtc::kEnumCounterKeyProtocolDtls));
> }
> 
> // Tests that the GetRemoteAudioSSLCertificate method returns the remote DTLS
> // certificate once the DTLS handshake has finished.
>-TEST_F(PeerConnectionIntegrationTest,
>+TEST_P(PeerConnectionIntegrationTest,
>        GetRemoteAudioSSLCertificateReturnsExchangedCertificate) {
>   auto GetRemoteAudioSSLCertificate = [](PeerConnectionWrapper* wrapper) {
>     auto pci = reinterpret_cast<PeerConnectionProxy*>(wrapper->pc());
>     auto pc = reinterpret_cast<PeerConnection*>(pci->internal());
>     return pc->GetRemoteAudioSSLCertificate();
>   };
>+  auto GetRemoteAudioSSLCertChain = [](PeerConnectionWrapper* wrapper) {
>+    auto pci = reinterpret_cast<PeerConnectionProxy*>(wrapper->pc());
>+    auto pc = reinterpret_cast<PeerConnection*>(pci->internal());
>+    return pc->GetRemoteAudioSSLCertChain();
>+  };
> 
>   auto caller_cert = rtc::RTCCertificate::FromPEM(kRsaPems[0]);
>   auto callee_cert = rtc::RTCCertificate::FromPEM(kRsaPems[1]);
>@@ -1384,9 +1765,11 @@ TEST_F(PeerConnectionIntegrationTest,
>   // calling this method should not crash).
>   EXPECT_EQ(nullptr, GetRemoteAudioSSLCertificate(caller()));
>   EXPECT_EQ(nullptr, GetRemoteAudioSSLCertificate(callee()));
>+  EXPECT_EQ(nullptr, GetRemoteAudioSSLCertChain(caller()));
>+  EXPECT_EQ(nullptr, GetRemoteAudioSSLCertChain(callee()));
> 
>-  caller()->AddAudioOnlyMediaStream();
>-  callee()->AddAudioOnlyMediaStream();
>+  caller()->AddAudioTrack();
>+  callee()->AddAudioTrack();
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>   ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
>@@ -1403,53 +1786,35 @@ TEST_F(PeerConnectionIntegrationTest,
>   ASSERT_TRUE(callee_remote_cert);
>   EXPECT_EQ(caller_cert->ssl_certificate().ToPEMString(),
>             callee_remote_cert->ToPEMString());
>-}
>-
>-// This test sets up a call between two parties (using DTLS) and tests that we
>-// can get a video aspect ratio of 16:9.
>-TEST_F(PeerConnectionIntegrationTest, SendAndReceive16To9AspectRatio) {
>-  ASSERT_TRUE(CreatePeerConnectionWrappers());
>-  ConnectFakeSignaling();
>-
>-  // Add video tracks with 16:9 constraint.
>-  FakeConstraints constraints;
>-  double requested_ratio = 16.0 / 9;
>-  constraints.SetMandatoryMinAspectRatio(requested_ratio);
>-  caller()->AddMediaStreamFromTracks(
>-      nullptr, caller()->CreateLocalVideoTrackWithConstraints(constraints));
>-  callee()->AddMediaStreamFromTracks(
>-      nullptr, callee()->CreateLocalVideoTrackWithConstraints(constraints));
> 
>-  // Do normal offer/answer and wait for at least one frame to be received in
>-  // each direction.
>-  caller()->CreateAndSetAndSignalOffer();
>-  ASSERT_TRUE_WAIT(caller()->min_video_frames_received_per_track() > 0 &&
>-                       callee()->min_video_frames_received_per_track() > 0,
>-                   kMaxWaitForFramesMs);
>+  auto caller_remote_cert_chain = GetRemoteAudioSSLCertChain(caller());
>+  ASSERT_TRUE(caller_remote_cert_chain);
>+  ASSERT_EQ(1U, caller_remote_cert_chain->GetSize());
>+  auto remote_cert = &caller_remote_cert_chain->Get(0);
>+  EXPECT_EQ(callee_cert->ssl_certificate().ToPEMString(),
>+            remote_cert->ToPEMString());
> 
>-  // Check rendered aspect ratio.
>-  EXPECT_EQ(requested_ratio, caller()->local_rendered_aspect_ratio());
>-  EXPECT_EQ(requested_ratio, caller()->rendered_aspect_ratio());
>-  EXPECT_EQ(requested_ratio, callee()->local_rendered_aspect_ratio());
>-  EXPECT_EQ(requested_ratio, callee()->rendered_aspect_ratio());
>+  auto callee_remote_cert_chain = GetRemoteAudioSSLCertChain(callee());
>+  ASSERT_TRUE(callee_remote_cert_chain);
>+  ASSERT_EQ(1U, callee_remote_cert_chain->GetSize());
>+  remote_cert = &callee_remote_cert_chain->Get(0);
>+  EXPECT_EQ(caller_cert->ssl_certificate().ToPEMString(),
>+            remote_cert->ToPEMString());
> }
> 
> // This test sets up a call between two parties with a source resolution of
> // 1280x720 and verifies that a 16:9 aspect ratio is received.
>-TEST_F(PeerConnectionIntegrationTest,
>+TEST_P(PeerConnectionIntegrationTest,
>        Send1280By720ResolutionAndReceive16To9AspectRatio) {
>   ASSERT_TRUE(CreatePeerConnectionWrappers());
>   ConnectFakeSignaling();
> 
>-  // Similar to above test, but uses MandatoryMin[Width/Height] constraint
>-  // instead of aspect ratio constraint.
>-  FakeConstraints constraints;
>-  constraints.SetMandatoryMinWidth(1280);
>-  constraints.SetMandatoryMinHeight(720);
>-  caller()->AddMediaStreamFromTracks(
>-      nullptr, caller()->CreateLocalVideoTrackWithConstraints(constraints));
>-  callee()->AddMediaStreamFromTracks(
>-      nullptr, callee()->CreateLocalVideoTrackWithConstraints(constraints));
>+  // Add video tracks with 16:9 aspect ratio, size 1280 x 720.
>+  webrtc::FakePeriodicVideoSource::Config config;
>+  config.width = 1280;
>+  config.height = 720;
>+  caller()->AddTrack(caller()->CreateLocalVideoTrackWithConfig(config));
>+  callee()->AddTrack(callee()->CreateLocalVideoTrackWithConfig(config));
> 
>   // Do normal offer/answer and wait for at least one frame to be received in
>   // each direction.
>@@ -1467,86 +1832,111 @@ TEST_F(PeerConnectionIntegrationTest,
> 
> // This test sets up an one-way call, with media only from caller to
> // callee.
>-TEST_F(PeerConnectionIntegrationTest, OneWayMediaCall) {
>+TEST_P(PeerConnectionIntegrationTest, OneWayMediaCall) {
>   ASSERT_TRUE(CreatePeerConnectionWrappers());
>   ConnectFakeSignaling();
>-  caller()->AddAudioVideoMediaStream();
>+  caller()->AddAudioVideoTracks();
>   caller()->CreateAndSetAndSignalOffer();
>-  int caller_received_frames = 0;
>-  ExpectNewFramesReceivedWithWait(
>-      caller_received_frames, caller_received_frames,
>-      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
>-      kMaxWaitForFramesMs);
>+  MediaExpectations media_expectations;
>+  media_expectations.CalleeExpectsSomeAudioAndVideo();
>+  media_expectations.CallerExpectsNoAudio();
>+  media_expectations.CallerExpectsNoVideo();
>+  ASSERT_TRUE(ExpectNewFrames(media_expectations));
> }
> 
> // This test sets up a audio call initially, with the callee rejecting video
> // initially. Then later the callee decides to upgrade to audio/video, and
> // initiates a new offer/answer exchange.
>-TEST_F(PeerConnectionIntegrationTest, AudioToVideoUpgrade) {
>+TEST_P(PeerConnectionIntegrationTest, AudioToVideoUpgrade) {
>   ASSERT_TRUE(CreatePeerConnectionWrappers());
>   ConnectFakeSignaling();
>   // Initially, offer an audio/video stream from the caller, but refuse to
>   // send/receive video on the callee side.
>-  caller()->AddAudioVideoMediaStream();
>-  callee()->AddAudioOnlyMediaStream();
>-  PeerConnectionInterface::RTCOfferAnswerOptions options;
>-  options.offer_to_receive_video = 0;
>-  callee()->SetOfferAnswerOptions(options);
>+  caller()->AddAudioVideoTracks();
>+  callee()->AddAudioTrack();
>+  if (sdp_semantics_ == SdpSemantics::kPlanB) {
>+    PeerConnectionInterface::RTCOfferAnswerOptions options;
>+    options.offer_to_receive_video = 0;
>+    callee()->SetOfferAnswerOptions(options);
>+  } else {
>+    callee()->SetRemoteOfferHandler([this] {
>+      callee()->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_VIDEO)->Stop();
>+    });
>+  }
>   // Do offer/answer and make sure audio is still received end-to-end.
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>-  ExpectNewFramesReceivedWithWait(kDefaultExpectedAudioFrameCount, 0,
>-                                  kDefaultExpectedAudioFrameCount, 0,
>-                                  kMaxWaitForFramesMs);
>+  {
>+    MediaExpectations media_expectations;
>+    media_expectations.ExpectBidirectionalAudio();
>+    media_expectations.ExpectNoVideo();
>+    ASSERT_TRUE(ExpectNewFrames(media_expectations));
>+  }
>   // Sanity check that the callee's description has a rejected video section.
>   ASSERT_NE(nullptr, callee()->pc()->local_description());
>   const ContentInfo* callee_video_content =
>       GetFirstVideoContent(callee()->pc()->local_description()->description());
>   ASSERT_NE(nullptr, callee_video_content);
>   EXPECT_TRUE(callee_video_content->rejected);
>+
>   // Now negotiate with video and ensure negotiation succeeds, with video
>   // frames and additional audio frames being received.
>-  callee()->AddVideoOnlyMediaStream();
>-  options.offer_to_receive_video = 1;
>-  callee()->SetOfferAnswerOptions(options);
>+  callee()->AddVideoTrack();
>+  if (sdp_semantics_ == SdpSemantics::kPlanB) {
>+    PeerConnectionInterface::RTCOfferAnswerOptions options;
>+    options.offer_to_receive_video = 1;
>+    callee()->SetOfferAnswerOptions(options);
>+  } else {
>+    callee()->SetRemoteOfferHandler(nullptr);
>+    caller()->SetRemoteOfferHandler([this] {
>+      // The caller creates a new transceiver to receive video on when receiving
>+      // the offer, but by default it is send only.
>+      auto transceivers = caller()->pc()->GetTransceivers();
>+      ASSERT_EQ(3U, transceivers.size());
>+      ASSERT_EQ(cricket::MEDIA_TYPE_VIDEO,
>+                transceivers[2]->receiver()->media_type());
>+      transceivers[2]->sender()->SetTrack(caller()->CreateLocalVideoTrack());
>+      transceivers[2]->SetDirection(RtpTransceiverDirection::kSendRecv);
>+    });
>+  }
>   callee()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>-  // Expect additional audio frames to be received after the upgrade.
>-  ExpectNewFramesReceivedWithWait(
>-      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
>-      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
>-      kMaxWaitForFramesMs);
>+  {
>+    // Expect additional audio frames to be received after the upgrade.
>+    MediaExpectations media_expectations;
>+    media_expectations.ExpectBidirectionalAudioAndVideo();
>+    ASSERT_TRUE(ExpectNewFrames(media_expectations));
>+  }
> }
> 
> // Simpler than the above test; just add an audio track to an established
> // video-only connection.
>-TEST_F(PeerConnectionIntegrationTest, AddAudioToVideoOnlyCall) {
>+TEST_P(PeerConnectionIntegrationTest, AddAudioToVideoOnlyCall) {
>   ASSERT_TRUE(CreatePeerConnectionWrappers());
>   ConnectFakeSignaling();
>   // Do initial offer/answer with just a video track.
>-  caller()->AddVideoOnlyMediaStream();
>-  callee()->AddVideoOnlyMediaStream();
>+  caller()->AddVideoTrack();
>+  callee()->AddVideoTrack();
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>   // Now add an audio track and do another offer/answer.
>-  caller()->AddAudioOnlyMediaStream();
>-  callee()->AddAudioOnlyMediaStream();
>+  caller()->AddAudioTrack();
>+  callee()->AddAudioTrack();
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>   // Ensure both audio and video frames are received end-to-end.
>-  ExpectNewFramesReceivedWithWait(
>-      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
>-      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
>-      kMaxWaitForFramesMs);
>+  MediaExpectations media_expectations;
>+  media_expectations.ExpectBidirectionalAudioAndVideo();
>+  ASSERT_TRUE(ExpectNewFrames(media_expectations));
> }
> 
> // This test sets up a call that's transferred to a new caller with a different
> // DTLS fingerprint.
>-TEST_F(PeerConnectionIntegrationTest, CallTransferredForCallee) {
>+TEST_P(PeerConnectionIntegrationTest, CallTransferredForCallee) {
>   ASSERT_TRUE(CreatePeerConnectionWrappers());
>   ConnectFakeSignaling();
>-  caller()->AddAudioVideoMediaStream();
>-  callee()->AddAudioVideoMediaStream();
>+  caller()->AddAudioVideoTracks();
>+  callee()->AddAudioVideoTracks();
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
> 
>@@ -1554,29 +1944,28 @@ TEST_F(PeerConnectionIntegrationTest, CallTransferredForCallee) {
>   // receiving client. These SRTP packets will be dropped.
>   std::unique_ptr<PeerConnectionWrapper> original_peer(
>       SetCallerPcWrapperAndReturnCurrent(
>-          CreatePeerConnectionWrapperWithAlternateKey()));
>+          CreatePeerConnectionWrapperWithAlternateKey().release()));
>   // TODO(deadbeef): Why do we call Close here? That goes against the comment
>   // directly above.
>   original_peer->pc()->Close();
> 
>   ConnectFakeSignaling();
>-  caller()->AddAudioVideoMediaStream();
>+  caller()->AddAudioVideoTracks();
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>   // Wait for some additional frames to be transmitted end-to-end.
>-  ExpectNewFramesReceivedWithWait(
>-      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
>-      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
>-      kMaxWaitForFramesMs);
>+  MediaExpectations media_expectations;
>+  media_expectations.ExpectBidirectionalAudioAndVideo();
>+  ASSERT_TRUE(ExpectNewFrames(media_expectations));
> }
> 
> // This test sets up a call that's transferred to a new callee with a different
> // DTLS fingerprint.
>-TEST_F(PeerConnectionIntegrationTest, CallTransferredForCaller) {
>+TEST_P(PeerConnectionIntegrationTest, CallTransferredForCaller) {
>   ASSERT_TRUE(CreatePeerConnectionWrappers());
>   ConnectFakeSignaling();
>-  caller()->AddAudioVideoMediaStream();
>-  callee()->AddAudioVideoMediaStream();
>+  caller()->AddAudioVideoTracks();
>+  callee()->AddAudioVideoTracks();
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
> 
>@@ -1584,43 +1973,42 @@ TEST_F(PeerConnectionIntegrationTest, CallTransferredForCaller) {
>   // receiving client. These SRTP packets will be dropped.
>   std::unique_ptr<PeerConnectionWrapper> original_peer(
>       SetCalleePcWrapperAndReturnCurrent(
>-          CreatePeerConnectionWrapperWithAlternateKey()));
>+          CreatePeerConnectionWrapperWithAlternateKey().release()));
>   // TODO(deadbeef): Why do we call Close here? That goes against the comment
>   // directly above.
>   original_peer->pc()->Close();
> 
>   ConnectFakeSignaling();
>-  callee()->AddAudioVideoMediaStream();
>+  callee()->AddAudioVideoTracks();
>   caller()->SetOfferAnswerOptions(IceRestartOfferAnswerOptions());
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>   // Wait for some additional frames to be transmitted end-to-end.
>-  ExpectNewFramesReceivedWithWait(
>-      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
>-      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
>-      kMaxWaitForFramesMs);
>+  MediaExpectations media_expectations;
>+  media_expectations.ExpectBidirectionalAudioAndVideo();
>+  ASSERT_TRUE(ExpectNewFrames(media_expectations));
> }
> 
> // This test sets up a non-bundled call and negotiates bundling at the same
> // time as starting an ICE restart. When bundling is in effect in the restart,
> // the DTLS-SRTP context should be successfully reset.
>-TEST_F(PeerConnectionIntegrationTest, BundlingEnabledWhileIceRestartOccurs) {
>+TEST_P(PeerConnectionIntegrationTest, BundlingEnabledWhileIceRestartOccurs) {
>   ASSERT_TRUE(CreatePeerConnectionWrappers());
>   ConnectFakeSignaling();
> 
>-  caller()->AddAudioVideoMediaStream();
>-  callee()->AddAudioVideoMediaStream();
>+  caller()->AddAudioVideoTracks();
>+  callee()->AddAudioVideoTracks();
>   // Remove the bundle group from the SDP received by the callee.
>   callee()->SetReceivedSdpMunger([](cricket::SessionDescription* desc) {
>     desc->RemoveGroupByName("BUNDLE");
>   });
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>-  ExpectNewFramesReceivedWithWait(
>-      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
>-      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
>-      kMaxWaitForFramesMs);
>-
>+  {
>+    MediaExpectations media_expectations;
>+    media_expectations.ExpectBidirectionalAudioAndVideo();
>+    ASSERT_TRUE(ExpectNewFrames(media_expectations));
>+  }
>   // Now stop removing the BUNDLE group, and trigger an ICE restart.
>   callee()->SetReceivedSdpMunger(nullptr);
>   caller()->SetOfferAnswerOptions(IceRestartOfferAnswerOptions());
>@@ -1628,25 +2016,24 @@ TEST_F(PeerConnectionIntegrationTest, BundlingEnabledWhileIceRestartOccurs) {
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
> 
>   // Expect additional frames to be received after the ICE restart.
>-  ExpectNewFramesReceivedWithWait(
>-      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
>-      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
>-      kMaxWaitForFramesMs);
>+  {
>+    MediaExpectations media_expectations;
>+    media_expectations.ExpectBidirectionalAudioAndVideo();
>+    ASSERT_TRUE(ExpectNewFrames(media_expectations));
>+  }
> }
> 
> // Test CVO (Coordination of Video Orientation). If a video source is rotated
> // and both peers support the CVO RTP header extension, the actual video frames
> // don't need to be encoded in different resolutions, since the rotation is
> // communicated through the RTP header extension.
>-TEST_F(PeerConnectionIntegrationTest, RotatedVideoWithCVOExtension) {
>+TEST_P(PeerConnectionIntegrationTest, RotatedVideoWithCVOExtension) {
>   ASSERT_TRUE(CreatePeerConnectionWrappers());
>   ConnectFakeSignaling();
>   // Add rotated video tracks.
>-  caller()->AddMediaStreamFromTracks(
>-      nullptr,
>+  caller()->AddTrack(
>       caller()->CreateLocalVideoTrackWithRotation(webrtc::kVideoRotation_90));
>-  callee()->AddMediaStreamFromTracks(
>-      nullptr,
>+  callee()->AddTrack(
>       callee()->CreateLocalVideoTrackWithRotation(webrtc::kVideoRotation_270));
> 
>   // Wait for video frames to be received by both sides.
>@@ -1670,15 +2057,13 @@ TEST_F(PeerConnectionIntegrationTest, RotatedVideoWithCVOExtension) {
> 
> // Test that when the CVO extension isn't supported, video is rotated the
> // old-fashioned way, by encoding rotated frames.
>-TEST_F(PeerConnectionIntegrationTest, RotatedVideoWithoutCVOExtension) {
>+TEST_P(PeerConnectionIntegrationTest, RotatedVideoWithoutCVOExtension) {
>   ASSERT_TRUE(CreatePeerConnectionWrappers());
>   ConnectFakeSignaling();
>   // Add rotated video tracks.
>-  caller()->AddMediaStreamFromTracks(
>-      nullptr,
>+  caller()->AddTrack(
>       caller()->CreateLocalVideoTrackWithRotation(webrtc::kVideoRotation_90));
>-  callee()->AddMediaStreamFromTracks(
>-      nullptr,
>+  callee()->AddTrack(
>       callee()->CreateLocalVideoTrackWithRotation(webrtc::kVideoRotation_270));
> 
>   // Remove the CVO extension from the offered SDP.
>@@ -1707,68 +2092,88 @@ TEST_F(PeerConnectionIntegrationTest, RotatedVideoWithoutCVOExtension) {
>   EXPECT_EQ(webrtc::kVideoRotation_0, callee()->rendered_rotation());
> }
> 
>-// TODO(deadbeef): The tests below rely on RTCOfferAnswerOptions to reject an
>-// m= section. When we implement Unified Plan SDP, the right way to do this
>-// would be by stopping an RtpTransceiver.
>-
> // Test that if the answerer rejects the audio m= section, no audio is sent or
> // received, but video still can be.
>-TEST_F(PeerConnectionIntegrationTest, AnswererRejectsAudioSection) {
>+TEST_P(PeerConnectionIntegrationTest, AnswererRejectsAudioSection) {
>   ASSERT_TRUE(CreatePeerConnectionWrappers());
>   ConnectFakeSignaling();
>-  caller()->AddAudioVideoMediaStream();
>-  // Only add video track for callee, and set offer_to_receive_audio to 0, so
>-  // it will reject the audio m= section completely.
>-  PeerConnectionInterface::RTCOfferAnswerOptions options;
>-  options.offer_to_receive_audio = 0;
>-  callee()->SetOfferAnswerOptions(options);
>-  callee()->AddMediaStreamFromTracks(nullptr,
>-                                     callee()->CreateLocalVideoTrack());
>+  caller()->AddAudioVideoTracks();
>+  if (sdp_semantics_ == SdpSemantics::kPlanB) {
>+    // Only add video track for callee, and set offer_to_receive_audio to 0, so
>+    // it will reject the audio m= section completely.
>+    PeerConnectionInterface::RTCOfferAnswerOptions options;
>+    options.offer_to_receive_audio = 0;
>+    callee()->SetOfferAnswerOptions(options);
>+  } else {
>+    // Stopping the audio RtpTransceiver will cause the media section to be
>+    // rejected in the answer.
>+    callee()->SetRemoteOfferHandler([this] {
>+      callee()->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_AUDIO)->Stop();
>+    });
>+  }
>+  callee()->AddTrack(callee()->CreateLocalVideoTrack());
>   // Do offer/answer and wait for successful end-to-end video frames.
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>-  ExpectNewFramesReceivedWithWait(0, kDefaultExpectedVideoFrameCount, 0,
>-                                  kDefaultExpectedVideoFrameCount,
>-                                  kMaxWaitForFramesMs);
>-  // Shouldn't have received audio frames at any point.
>-  EXPECT_EQ(0, caller()->audio_frames_received());
>-  EXPECT_EQ(0, callee()->audio_frames_received());
>+  MediaExpectations media_expectations;
>+  media_expectations.ExpectBidirectionalVideo();
>+  media_expectations.ExpectNoAudio();
>+  ASSERT_TRUE(ExpectNewFrames(media_expectations));
>+
>   // Sanity check that the callee's description has a rejected audio section.
>   ASSERT_NE(nullptr, callee()->pc()->local_description());
>   const ContentInfo* callee_audio_content =
>       GetFirstAudioContent(callee()->pc()->local_description()->description());
>   ASSERT_NE(nullptr, callee_audio_content);
>   EXPECT_TRUE(callee_audio_content->rejected);
>+  if (sdp_semantics_ == SdpSemantics::kUnifiedPlan) {
>+    // The caller's transceiver should have stopped after receiving the answer.
>+    EXPECT_TRUE(caller()
>+                    ->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_AUDIO)
>+                    ->stopped());
>+  }
> }
> 
> // Test that if the answerer rejects the video m= section, no video is sent or
> // received, but audio still can be.
>-TEST_F(PeerConnectionIntegrationTest, AnswererRejectsVideoSection) {
>+TEST_P(PeerConnectionIntegrationTest, AnswererRejectsVideoSection) {
>   ASSERT_TRUE(CreatePeerConnectionWrappers());
>   ConnectFakeSignaling();
>-  caller()->AddAudioVideoMediaStream();
>-  // Only add audio track for callee, and set offer_to_receive_video to 0, so
>-  // it will reject the video m= section completely.
>-  PeerConnectionInterface::RTCOfferAnswerOptions options;
>-  options.offer_to_receive_video = 0;
>-  callee()->SetOfferAnswerOptions(options);
>-  callee()->AddMediaStreamFromTracks(callee()->CreateLocalAudioTrack(),
>-                                     nullptr);
>+  caller()->AddAudioVideoTracks();
>+  if (sdp_semantics_ == SdpSemantics::kPlanB) {
>+    // Only add audio track for callee, and set offer_to_receive_video to 0, so
>+    // it will reject the video m= section completely.
>+    PeerConnectionInterface::RTCOfferAnswerOptions options;
>+    options.offer_to_receive_video = 0;
>+    callee()->SetOfferAnswerOptions(options);
>+  } else {
>+    // Stopping the video RtpTransceiver will cause the media section to be
>+    // rejected in the answer.
>+    callee()->SetRemoteOfferHandler([this] {
>+      callee()->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_VIDEO)->Stop();
>+    });
>+  }
>+  callee()->AddTrack(callee()->CreateLocalAudioTrack());
>   // Do offer/answer and wait for successful end-to-end audio frames.
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>-  ExpectNewFramesReceivedWithWait(kDefaultExpectedAudioFrameCount, 0,
>-                                  kDefaultExpectedAudioFrameCount, 0,
>-                                  kMaxWaitForFramesMs);
>-  // Shouldn't have received video frames at any point.
>-  EXPECT_EQ(0, caller()->total_video_frames_received());
>-  EXPECT_EQ(0, callee()->total_video_frames_received());
>+  MediaExpectations media_expectations;
>+  media_expectations.ExpectBidirectionalAudio();
>+  media_expectations.ExpectNoVideo();
>+  ASSERT_TRUE(ExpectNewFrames(media_expectations));
>+
>   // Sanity check that the callee's description has a rejected video section.
>   ASSERT_NE(nullptr, callee()->pc()->local_description());
>   const ContentInfo* callee_video_content =
>       GetFirstVideoContent(callee()->pc()->local_description()->description());
>   ASSERT_NE(nullptr, callee_video_content);
>   EXPECT_TRUE(callee_video_content->rejected);
>+  if (sdp_semantics_ == SdpSemantics::kUnifiedPlan) {
>+    // The caller's transceiver should have stopped after receiving the answer.
>+    EXPECT_TRUE(caller()
>+                    ->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_VIDEO)
>+                    ->stopped());
>+  }
> }
> 
> // Test that if the answerer rejects both audio and video m= sections, nothing
>@@ -1776,19 +2181,29 @@ TEST_F(PeerConnectionIntegrationTest, AnswererRejectsVideoSection) {
> // TODO(deadbeef): Test that a data channel still works. Currently this doesn't
> // test anything but the fact that negotiation succeeds, which doesn't mean
> // much.
>-TEST_F(PeerConnectionIntegrationTest, AnswererRejectsAudioAndVideoSections) {
>+TEST_P(PeerConnectionIntegrationTest, AnswererRejectsAudioAndVideoSections) {
>   ASSERT_TRUE(CreatePeerConnectionWrappers());
>   ConnectFakeSignaling();
>-  caller()->AddAudioVideoMediaStream();
>-  // Don't give the callee any tracks, and set offer_to_receive_X to 0, so it
>-  // will reject both audio and video m= sections.
>-  PeerConnectionInterface::RTCOfferAnswerOptions options;
>-  options.offer_to_receive_audio = 0;
>-  options.offer_to_receive_video = 0;
>-  callee()->SetOfferAnswerOptions(options);
>+  caller()->AddAudioVideoTracks();
>+  if (sdp_semantics_ == SdpSemantics::kPlanB) {
>+    // Don't give the callee any tracks, and set offer_to_receive_X to 0, so it
>+    // will reject both audio and video m= sections.
>+    PeerConnectionInterface::RTCOfferAnswerOptions options;
>+    options.offer_to_receive_audio = 0;
>+    options.offer_to_receive_video = 0;
>+    callee()->SetOfferAnswerOptions(options);
>+  } else {
>+    callee()->SetRemoteOfferHandler([this] {
>+      // Stopping all transceivers will cause all media sections to be rejected.
>+      for (auto transceiver : callee()->pc()->GetTransceivers()) {
>+        transceiver->Stop();
>+      }
>+    });
>+  }
>   // Do offer/answer and wait for stable signaling state.
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>+
>   // Sanity check that the callee's description has rejected m= sections.
>   ASSERT_NE(nullptr, callee()->pc()->local_description());
>   const ContentInfo* callee_audio_content =
>@@ -1805,28 +2220,31 @@ TEST_F(PeerConnectionIntegrationTest, AnswererRejectsAudioAndVideoSections) {
> // call runs for a while, the caller sends an updated offer with video being
> // rejected. Once the re-negotiation is done, the video flow should stop and
> // the audio flow should continue.
>-TEST_F(PeerConnectionIntegrationTest, VideoRejectedInSubsequentOffer) {
>+TEST_P(PeerConnectionIntegrationTest, VideoRejectedInSubsequentOffer) {
>   ASSERT_TRUE(CreatePeerConnectionWrappers());
>   ConnectFakeSignaling();
>-  caller()->AddAudioVideoMediaStream();
>-  callee()->AddAudioVideoMediaStream();
>+  caller()->AddAudioVideoTracks();
>+  callee()->AddAudioVideoTracks();
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>-  ExpectNewFramesReceivedWithWait(
>-      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
>-      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
>-      kMaxWaitForFramesMs);
>-
>+  {
>+    MediaExpectations media_expectations;
>+    media_expectations.ExpectBidirectionalAudioAndVideo();
>+    ASSERT_TRUE(ExpectNewFrames(media_expectations));
>+  }
>   // Renegotiate, rejecting the video m= section.
>-  // TODO(deadbeef): When an RtpTransceiver API is available, use that to
>-  // reject the video m= section.
>-  caller()->SetGeneratedSdpMunger([](cricket::SessionDescription* description) {
>-    for (cricket::ContentInfo& content : description->contents()) {
>-      if (cricket::IsVideoContent(&content)) {
>-        content.rejected = true;
>-      }
>-    }
>-  });
>+  if (sdp_semantics_ == SdpSemantics::kPlanB) {
>+    caller()->SetGeneratedSdpMunger(
>+        [](cricket::SessionDescription* description) {
>+          for (cricket::ContentInfo& content : description->contents()) {
>+            if (cricket::IsVideoContent(&content)) {
>+              content.rejected = true;
>+            }
>+          }
>+        });
>+  } else {
>+    caller()->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_VIDEO)->Stop();
>+  }
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kMaxWaitForActivationMs);
> 
>@@ -1836,19 +2254,50 @@ TEST_F(PeerConnectionIntegrationTest, VideoRejectedInSubsequentOffer) {
>       GetFirstVideoContent(caller()->pc()->local_description()->description());
>   ASSERT_NE(nullptr, caller_video_content);
>   EXPECT_TRUE(caller_video_content->rejected);
>+  // Wait for some additional audio frames to be received.
>+  {
>+    MediaExpectations media_expectations;
>+    media_expectations.ExpectBidirectionalAudio();
>+    media_expectations.ExpectNoVideo();
>+    ASSERT_TRUE(ExpectNewFrames(media_expectations));
>+  }
>+}
>+
>+// Do one offer/answer with audio, another that disables it (rejecting the m=
>+// section), and another that re-enables it. Regression test for:
>+// bugs.webrtc.org/6023
>+TEST_F(PeerConnectionIntegrationTestPlanB, EnableAudioAfterRejecting) {
>+  ASSERT_TRUE(CreatePeerConnectionWrappers());
>+  ConnectFakeSignaling();
>+
>+  // Add audio track, do normal offer/answer.
>+  rtc::scoped_refptr<webrtc::AudioTrackInterface> track =
>+      caller()->CreateLocalAudioTrack();
>+  rtc::scoped_refptr<webrtc::RtpSenderInterface> sender =
>+      caller()->pc()->AddTrack(track, {"stream"}).MoveValue();
>+  caller()->CreateAndSetAndSignalOffer();
>+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
> 
>-  int caller_video_received = caller()->total_video_frames_received();
>-  int callee_video_received = callee()->total_video_frames_received();
>+  // Remove audio track, and set offer_to_receive_audio to false to cause the
>+  // m= section to be completely disabled, not just "recvonly".
>+  caller()->pc()->RemoveTrack(sender);
>+  PeerConnectionInterface::RTCOfferAnswerOptions options;
>+  options.offer_to_receive_audio = 0;
>+  caller()->SetOfferAnswerOptions(options);
>+  caller()->CreateAndSetAndSignalOffer();
>+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
> 
>-  // Wait for some additional audio frames to be received.
>-  ExpectNewFramesReceivedWithWait(kDefaultExpectedAudioFrameCount, 0,
>-                                  kDefaultExpectedAudioFrameCount, 0,
>-                                  kMaxWaitForFramesMs);
>+  // Add the audio track again, expecting negotiation to succeed and frames to
>+  // flow.
>+  sender = caller()->pc()->AddTrack(track, {"stream"}).MoveValue();
>+  options.offer_to_receive_audio = 1;
>+  caller()->SetOfferAnswerOptions(options);
>+  caller()->CreateAndSetAndSignalOffer();
>+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
> 
>-  // During this time, we shouldn't have received any additional video frames
>-  // for the rejected video tracks.
>-  EXPECT_EQ(caller_video_received, caller()->total_video_frames_received());
>-  EXPECT_EQ(callee_video_received, callee()->total_video_frames_received());
>+  MediaExpectations media_expectations;
>+  media_expectations.CalleeExpectsSomeAudio();
>+  EXPECT_TRUE(ExpectNewFrames(media_expectations));
> }
> 
> // Basic end-to-end test, but without SSRC/MSID signaling. This functionality
>@@ -1856,36 +2305,82 @@ TEST_F(PeerConnectionIntegrationTest, VideoRejectedInSubsequentOffer) {
> // TODO(deadbeef): When we support the MID extension and demuxing on MID, also
> // add a test for an end-to-end test without MID signaling either (basically,
> // the minimum acceptable SDP).
>-TEST_F(PeerConnectionIntegrationTest, EndToEndCallWithoutSsrcOrMsidSignaling) {
>+TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithoutSsrcOrMsidSignaling) {
>   ASSERT_TRUE(CreatePeerConnectionWrappers());
>   ConnectFakeSignaling();
>   // Add audio and video, testing that packets can be demuxed on payload type.
>-  caller()->AddAudioVideoMediaStream();
>-  callee()->AddAudioVideoMediaStream();
>+  caller()->AddAudioVideoTracks();
>+  callee()->AddAudioVideoTracks();
>   // Remove SSRCs and MSIDs from the received offer SDP.
>   callee()->SetReceivedSdpMunger(RemoveSsrcsAndMsids);
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>-  ExpectNewFramesReceivedWithWait(
>-      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
>-      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
>-      kMaxWaitForFramesMs);
>+  MediaExpectations media_expectations;
>+  media_expectations.ExpectBidirectionalAudioAndVideo();
>+  ASSERT_TRUE(ExpectNewFrames(media_expectations));
>+}
>+
>+// Basic end-to-end test, without SSRC signaling. This means that the track
>+// was created properly and frames are delivered when the MSIDs are communicated
>+// with a=msid lines and no a=ssrc lines.
>+TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
>+       EndToEndCallWithoutSsrcSignaling) {
>+  const char kStreamId[] = "streamId";
>+  ASSERT_TRUE(CreatePeerConnectionWrappers());
>+  ConnectFakeSignaling();
>+  // Add just audio tracks.
>+  caller()->AddTrack(caller()->CreateLocalAudioTrack(), {kStreamId});
>+  callee()->AddAudioTrack();
>+
>+  // Remove SSRCs from the received offer SDP.
>+  callee()->SetReceivedSdpMunger(RemoveSsrcsAndKeepMsids);
>+  caller()->CreateAndSetAndSignalOffer();
>+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>+  MediaExpectations media_expectations;
>+  media_expectations.ExpectBidirectionalAudio();
>+  ASSERT_TRUE(ExpectNewFrames(media_expectations));
>+}
>+
>+// Tests that video flows between multiple video tracks when SSRCs are not
>+// signaled. This exercises the MID RTP header extension which is needed to
>+// demux the incoming video tracks.
>+TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
>+       EndToEndCallWithTwoVideoTracksAndNoSignaledSsrc) {
>+  ASSERT_TRUE(CreatePeerConnectionWrappers());
>+  ConnectFakeSignaling();
>+  caller()->AddVideoTrack();
>+  caller()->AddVideoTrack();
>+  callee()->AddVideoTrack();
>+  callee()->AddVideoTrack();
>+
>+  caller()->SetReceivedSdpMunger(&RemoveSsrcsAndKeepMsids);
>+  callee()->SetReceivedSdpMunger(&RemoveSsrcsAndKeepMsids);
>+  caller()->CreateAndSetAndSignalOffer();
>+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>+  ASSERT_EQ(2u, caller()->pc()->GetReceivers().size());
>+  ASSERT_EQ(2u, callee()->pc()->GetReceivers().size());
>+
>+  // Expect video to be received in both directions on both tracks.
>+  MediaExpectations media_expectations;
>+  media_expectations.ExpectBidirectionalVideo();
>+  EXPECT_TRUE(ExpectNewFrames(media_expectations));
> }
> 
> // Test that if two video tracks are sent (from caller to callee, in this test),
> // they're transmitted correctly end-to-end.
>-TEST_F(PeerConnectionIntegrationTest, EndToEndCallWithTwoVideoTracks) {
>+TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithTwoVideoTracks) {
>   ASSERT_TRUE(CreatePeerConnectionWrappers());
>   ConnectFakeSignaling();
>   // Add one audio/video stream, and one video-only stream.
>-  caller()->AddAudioVideoMediaStream();
>-  caller()->AddVideoOnlyMediaStream();
>+  caller()->AddAudioVideoTracks();
>+  caller()->AddVideoTrack();
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>-  ASSERT_EQ(2u, callee()->number_of_remote_streams());
>-  int expected_callee_received_frames = kDefaultExpectedVideoFrameCount;
>-  ExpectNewFramesReceivedWithWait(0, 0, 0, expected_callee_received_frames,
>-                                  kMaxWaitForFramesMs);
>+  ASSERT_EQ(3u, callee()->pc()->GetReceivers().size());
>+
>+  MediaExpectations media_expectations;
>+  media_expectations.CalleeExpectsSomeAudioAndVideo();
>+  ASSERT_TRUE(ExpectNewFrames(media_expectations));
> }
> 
> static void MakeSpecCompliantMaxBundleOffer(cricket::SessionDescription* desc) {
>@@ -1917,33 +2412,31 @@ static void MakeSpecCompliantMaxBundleOffer(cricket::SessionDescription* desc) {
> // TODO(deadbeef): Update this test to also omit "a=rtcp-mux", once that works.
> // TODO(deadbeef): Won't need this test once we start generating actual
> // standards-compliant SDP.
>-TEST_F(PeerConnectionIntegrationTest,
>+TEST_P(PeerConnectionIntegrationTest,
>        EndToEndCallWithSpecCompliantMaxBundleOffer) {
>   ASSERT_TRUE(CreatePeerConnectionWrappers());
>   ConnectFakeSignaling();
>-  caller()->AddAudioVideoMediaStream();
>-  callee()->AddAudioVideoMediaStream();
>+  caller()->AddAudioVideoTracks();
>+  callee()->AddAudioVideoTracks();
>   // Do the equivalent of setting the port to 0, adding a=bundle-only, and
>   // removing a=ice-ufrag, a=ice-pwd, a=fingerprint and a=setup from all
>   // but the first m= section.
>   callee()->SetReceivedSdpMunger(MakeSpecCompliantMaxBundleOffer);
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>-  ExpectNewFramesReceivedWithWait(
>-      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
>-      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
>-      kMaxWaitForFramesMs);
>+  MediaExpectations media_expectations;
>+  media_expectations.ExpectBidirectionalAudioAndVideo();
>+  ASSERT_TRUE(ExpectNewFrames(media_expectations));
> }
> 
> // Test that we can receive the audio output level from a remote audio track.
> // TODO(deadbeef): Use a fake audio source and verify that the output level is
> // exactly what the source on the other side was configured with.
>-TEST_F(PeerConnectionIntegrationTest, GetAudioOutputLevelStatsWithOldStatsApi) {
>+TEST_P(PeerConnectionIntegrationTest, GetAudioOutputLevelStatsWithOldStatsApi) {
>   ASSERT_TRUE(CreatePeerConnectionWrappers());
>   ConnectFakeSignaling();
>   // Just add an audio track.
>-  caller()->AddMediaStreamFromTracks(caller()->CreateLocalAudioTrack(),
>-                                     nullptr);
>+  caller()->AddAudioTrack();
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
> 
>@@ -1956,12 +2449,11 @@ TEST_F(PeerConnectionIntegrationTest, GetAudioOutputLevelStatsWithOldStatsApi) {
> // Test that an audio input level is reported.
> // TODO(deadbeef): Use a fake audio source and verify that the input level is
> // exactly what the source was configured with.
>-TEST_F(PeerConnectionIntegrationTest, GetAudioInputLevelStatsWithOldStatsApi) {
>+TEST_P(PeerConnectionIntegrationTest, GetAudioInputLevelStatsWithOldStatsApi) {
>   ASSERT_TRUE(CreatePeerConnectionWrappers());
>   ConnectFakeSignaling();
>   // Just add an audio track.
>-  caller()->AddMediaStreamFromTracks(caller()->CreateLocalAudioTrack(),
>-                                     nullptr);
>+  caller()->AddAudioTrack();
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
> 
>@@ -1972,53 +2464,42 @@ TEST_F(PeerConnectionIntegrationTest, GetAudioInputLevelStatsWithOldStatsApi) {
> }
> 
> // Test that we can get incoming byte counts from both audio and video tracks.
>-TEST_F(PeerConnectionIntegrationTest, GetBytesReceivedStatsWithOldStatsApi) {
>+TEST_P(PeerConnectionIntegrationTest, GetBytesReceivedStatsWithOldStatsApi) {
>   ASSERT_TRUE(CreatePeerConnectionWrappers());
>   ConnectFakeSignaling();
>-  caller()->AddAudioVideoMediaStream();
>+  caller()->AddAudioVideoTracks();
>   // Do offer/answer, wait for the callee to receive some frames.
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>-  int expected_caller_received_frames = 0;
>-  ExpectNewFramesReceivedWithWait(
>-      expected_caller_received_frames, expected_caller_received_frames,
>-      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
>-      kMaxWaitForFramesMs);
>+
>+  MediaExpectations media_expectations;
>+  media_expectations.CalleeExpectsSomeAudioAndVideo();
>+  ASSERT_TRUE(ExpectNewFrames(media_expectations));
> 
>   // Get a handle to the remote tracks created, so they can be used as GetStats
>   // filters.
>-  StreamCollectionInterface* remote_streams = callee()->remote_streams();
>-  ASSERT_EQ(1u, remote_streams->count());
>-  ASSERT_EQ(1u, remote_streams->at(0)->GetAudioTracks().size());
>-  ASSERT_EQ(1u, remote_streams->at(0)->GetVideoTracks().size());
>-  MediaStreamTrackInterface* remote_audio_track =
>-      remote_streams->at(0)->GetAudioTracks()[0];
>-  MediaStreamTrackInterface* remote_video_track =
>-      remote_streams->at(0)->GetVideoTracks()[0];
>-
>-  // We received frames, so we definitely should have nonzero "received bytes"
>-  // stats at this point.
>-  EXPECT_GT(callee()->OldGetStatsForTrack(remote_audio_track)->BytesReceived(),
>-            0);
>-  EXPECT_GT(callee()->OldGetStatsForTrack(remote_video_track)->BytesReceived(),
>-            0);
>+  for (auto receiver : callee()->pc()->GetReceivers()) {
>+    // We received frames, so we definitely should have nonzero "received bytes"
>+    // stats at this point.
>+    EXPECT_GT(callee()->OldGetStatsForTrack(receiver->track())->BytesReceived(),
>+              0);
>+  }
> }
> 
> // Test that we can get outgoing byte counts from both audio and video tracks.
>-TEST_F(PeerConnectionIntegrationTest, GetBytesSentStatsWithOldStatsApi) {
>+TEST_P(PeerConnectionIntegrationTest, GetBytesSentStatsWithOldStatsApi) {
>   ASSERT_TRUE(CreatePeerConnectionWrappers());
>   ConnectFakeSignaling();
>   auto audio_track = caller()->CreateLocalAudioTrack();
>   auto video_track = caller()->CreateLocalVideoTrack();
>-  caller()->AddMediaStreamFromTracks(audio_track, video_track);
>+  caller()->AddTrack(audio_track);
>+  caller()->AddTrack(video_track);
>   // Do offer/answer, wait for the callee to receive some frames.
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>-  int expected_caller_received_frames = 0;
>-  ExpectNewFramesReceivedWithWait(
>-      expected_caller_received_frames, expected_caller_received_frames,
>-      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
>-      kMaxWaitForFramesMs);
>+  MediaExpectations media_expectations;
>+  media_expectations.CalleeExpectsSomeAudioAndVideo();
>+  ASSERT_TRUE(ExpectNewFrames(media_expectations));
> 
>   // The callee received frames, so we definitely should have nonzero "sent
>   // bytes" stats at this point.
>@@ -2027,13 +2508,12 @@ TEST_F(PeerConnectionIntegrationTest, GetBytesSentStatsWithOldStatsApi) {
> }
> 
> // Test that we can get capture start ntp time.
>-TEST_F(PeerConnectionIntegrationTest, GetCaptureStartNtpTimeWithOldStatsApi) {
>+TEST_P(PeerConnectionIntegrationTest, GetCaptureStartNtpTimeWithOldStatsApi) {
>   ASSERT_TRUE(CreatePeerConnectionWrappers());
>   ConnectFakeSignaling();
>-  caller()->AddAudioOnlyMediaStream();
>+  caller()->AddAudioTrack();
> 
>-  auto audio_track = callee()->CreateLocalAudioTrack();
>-  callee()->AddMediaStreamFromTracks(audio_track, nullptr);
>+  callee()->AddAudioTrack();
> 
>   // Do offer/answer, wait for the callee to receive some frames.
>   caller()->CreateAndSetAndSignalOffer();
>@@ -2041,32 +2521,33 @@ TEST_F(PeerConnectionIntegrationTest, GetCaptureStartNtpTimeWithOldStatsApi) {
> 
>   // Get the remote audio track created on the receiver, so they can be used as
>   // GetStats filters.
>-  StreamCollectionInterface* remote_streams = callee()->remote_streams();
>-  ASSERT_EQ(1u, remote_streams->count());
>-  ASSERT_EQ(1u, remote_streams->at(0)->GetAudioTracks().size());
>-  MediaStreamTrackInterface* remote_audio_track =
>-      remote_streams->at(0)->GetAudioTracks()[0];
>+  auto receivers = callee()->pc()->GetReceivers();
>+  ASSERT_EQ(1u, receivers.size());
>+  auto remote_audio_track = receivers[0]->track();
> 
>   // Get the audio output level stats. Note that the level is not available
>   // until an RTCP packet has been received.
>-  EXPECT_TRUE_WAIT(callee()->OldGetStatsForTrack(remote_audio_track)->
>-                   CaptureStartNtpTime() > 0, 2 * kMaxWaitForFramesMs);
>+  EXPECT_TRUE_WAIT(
>+      callee()->OldGetStatsForTrack(remote_audio_track)->CaptureStartNtpTime() >
>+          0,
>+      2 * kMaxWaitForFramesMs);
> }
> 
> // Test that we can get stats (using the new stats implemnetation) for
> // unsignaled streams. Meaning when SSRCs/MSIDs aren't signaled explicitly in
> // SDP.
>-TEST_F(PeerConnectionIntegrationTest,
>+TEST_P(PeerConnectionIntegrationTest,
>        GetStatsForUnsignaledStreamWithNewStatsApi) {
>   ASSERT_TRUE(CreatePeerConnectionWrappers());
>   ConnectFakeSignaling();
>-  caller()->AddAudioOnlyMediaStream();
>+  caller()->AddAudioTrack();
>   // Remove SSRCs and MSIDs from the received offer SDP.
>   callee()->SetReceivedSdpMunger(RemoveSsrcsAndMsids);
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>-  // Wait for one audio frame to be received by the callee.
>-  ExpectNewFramesReceivedWithWait(0, 0, 1, 0, kMaxWaitForFramesMs);
>+  MediaExpectations media_expectations;
>+  media_expectations.CalleeExpectsSomeAudio(1);
>+  ASSERT_TRUE(ExpectNewFrames(media_expectations));
> 
>   // We received a frame, so we should have nonzero "bytes received" stats for
>   // the unsignaled stream, if stats are working for it.
>@@ -2081,19 +2562,42 @@ TEST_F(PeerConnectionIntegrationTest,
>   ASSERT_TRUE(inbound_stream_stats[0]->track_id.is_defined());
> }
> 
>+// Same as above but for the legacy stats implementation.
>+TEST_P(PeerConnectionIntegrationTest,
>+       GetStatsForUnsignaledStreamWithOldStatsApi) {
>+  ASSERT_TRUE(CreatePeerConnectionWrappers());
>+  ConnectFakeSignaling();
>+  caller()->AddAudioTrack();
>+  // Remove SSRCs and MSIDs from the received offer SDP.
>+  callee()->SetReceivedSdpMunger(RemoveSsrcsAndMsids);
>+  caller()->CreateAndSetAndSignalOffer();
>+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>+
>+  // Note that, since the old stats implementation associates SSRCs with tracks
>+  // using SDP, when SSRCs aren't signaled in SDP these stats won't have an
>+  // associated track ID. So we can't use the track "selector" argument.
>+  //
>+  // Also, we use "EXPECT_TRUE_WAIT" because the stats collector may decide to
>+  // return cached stats if not enough time has passed since the last update.
>+  EXPECT_TRUE_WAIT(callee()->OldGetStats()->BytesReceived() > 0,
>+                   kDefaultTimeout);
>+}
>+
> // Test that we can successfully get the media related stats (audio level
> // etc.) for the unsignaled stream.
>-TEST_F(PeerConnectionIntegrationTest,
>+TEST_P(PeerConnectionIntegrationTest,
>        GetMediaStatsForUnsignaledStreamWithNewStatsApi) {
>   ASSERT_TRUE(CreatePeerConnectionWrappers());
>   ConnectFakeSignaling();
>-  caller()->AddAudioVideoMediaStream();
>+  caller()->AddAudioVideoTracks();
>   // Remove SSRCs and MSIDs from the received offer SDP.
>   callee()->SetReceivedSdpMunger(RemoveSsrcsAndMsids);
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>-  // Wait for one audio frame to be received by the callee.
>-  ExpectNewFramesReceivedWithWait(0, 0, 1, 1, kMaxWaitForFramesMs);
>+  MediaExpectations media_expectations;
>+  media_expectations.CalleeExpectsSomeAudio(1);
>+  media_expectations.CalleeExpectsSomeVideo(1);
>+  ASSERT_TRUE(ExpectNewFrames(media_expectations));
> 
>   rtc::scoped_refptr<const webrtc::RTCStatsReport> report =
>       callee()->NewGetStats();
>@@ -2108,7 +2612,7 @@ TEST_F(PeerConnectionIntegrationTest,
> // Helper for test below.
> void ModifySsrcs(cricket::SessionDescription* desc) {
>   for (ContentInfo& content : desc->contents()) {
>-    for (cricket::StreamParams& stream :
>+    for (StreamParams& stream :
>          content.media_description()->mutable_streams()) {
>       for (uint32_t& ssrc : stream.ssrcs) {
>         ssrc = rtc::CreateRandomId();
>@@ -2132,19 +2636,22 @@ void ModifySsrcs(cricket::SessionDescription* desc) {
> // received, and a 50% chance that they'll stop updating (while
> // "concealed_samples" continues increasing, due to silence being generated for
> // the inactive stream).
>-TEST_F(PeerConnectionIntegrationTest,
>+TEST_P(PeerConnectionIntegrationTest,
>        TrackStatsUpdatedCorrectlyWhenUnsignaledSsrcChanges) {
>   ASSERT_TRUE(CreatePeerConnectionWrappers());
>   ConnectFakeSignaling();
>-  caller()->AddAudioOnlyMediaStream();
>+  caller()->AddAudioTrack();
>   // Remove SSRCs and MSIDs from the received offer SDP, simulating an endpoint
>   // that doesn't signal SSRCs (from the callee's perspective).
>   callee()->SetReceivedSdpMunger(RemoveSsrcsAndMsids);
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>   // Wait for 50 audio frames (500ms of audio) to be received by the callee.
>-  ExpectNewFramesReceivedWithWait(0, 0, 25, 0, kMaxWaitForFramesMs);
>-
>+  {
>+    MediaExpectations media_expectations;
>+    media_expectations.CalleeExpectsSomeAudio(50);
>+    ASSERT_TRUE(ExpectNewFrames(media_expectations));
>+  }
>   // Some audio frames were received, so we should have nonzero "samples
>   // received" for the track.
>   rtc::scoped_refptr<const webrtc::RTCStatsReport> report =
>@@ -2162,7 +2669,11 @@ TEST_F(PeerConnectionIntegrationTest,
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>   // Wait for 25 more audio frames (250ms of audio) to be received, from the new
>   // SSRC.
>-  ExpectNewFramesReceivedWithWait(0, 0, 25, 0, kMaxWaitForFramesMs);
>+  {
>+    MediaExpectations media_expectations;
>+    media_expectations.CalleeExpectsSomeAudio(25);
>+    ASSERT_TRUE(ExpectNewFrames(media_expectations));
>+  }
> 
>   report = callee()->NewGetStats();
>   ASSERT_NE(nullptr, report);
>@@ -2196,7 +2707,7 @@ TEST_F(PeerConnectionIntegrationTest,
> }
> 
> // Test that DTLS 1.0 is used if both sides only support DTLS 1.0.
>-TEST_F(PeerConnectionIntegrationTest, EndToEndCallWithDtls10) {
>+TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithDtls10) {
>   PeerConnectionFactory::Options dtls_10_options;
>   dtls_10_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
>   ASSERT_TRUE(CreatePeerConnectionWrappersWithOptions(dtls_10_options,
>@@ -2204,69 +2715,62 @@ TEST_F(PeerConnectionIntegrationTest, EndToEndCallWithDtls10) {
>   ConnectFakeSignaling();
>   // Do normal offer/answer and wait for some frames to be received in each
>   // direction.
>-  caller()->AddAudioVideoMediaStream();
>-  callee()->AddAudioVideoMediaStream();
>+  caller()->AddAudioVideoTracks();
>+  callee()->AddAudioVideoTracks();
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>-  ExpectNewFramesReceivedWithWait(
>-      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
>-      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
>-      kMaxWaitForFramesMs);
>+  MediaExpectations media_expectations;
>+  media_expectations.ExpectBidirectionalAudioAndVideo();
>+  ASSERT_TRUE(ExpectNewFrames(media_expectations));
> }
> 
> // Test getting cipher stats and UMA metrics when DTLS 1.0 is negotiated.
>-TEST_F(PeerConnectionIntegrationTest, Dtls10CipherStatsAndUmaMetrics) {
>+TEST_P(PeerConnectionIntegrationTest, Dtls10CipherStatsAndUmaMetrics) {
>   PeerConnectionFactory::Options dtls_10_options;
>   dtls_10_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
>   ASSERT_TRUE(CreatePeerConnectionWrappersWithOptions(dtls_10_options,
>                                                       dtls_10_options));
>   ConnectFakeSignaling();
>-  // Register UMA observer before signaling begins.
>-  rtc::scoped_refptr<webrtc::FakeMetricsObserver> caller_observer =
>-      new rtc::RefCountedObject<webrtc::FakeMetricsObserver>();
>-  caller()->pc()->RegisterUMAObserver(caller_observer);
>-  caller()->AddAudioVideoMediaStream();
>-  callee()->AddAudioVideoMediaStream();
>+  caller()->AddAudioVideoTracks();
>+  callee()->AddAudioVideoTracks();
>   caller()->CreateAndSetAndSignalOffer();
>-  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>+  ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
>   EXPECT_TRUE_WAIT(rtc::SSLStreamAdapter::IsAcceptableCipher(
>                        caller()->OldGetStats()->DtlsCipher(), rtc::KT_DEFAULT),
>                    kDefaultTimeout);
>   EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite),
>                  caller()->OldGetStats()->SrtpCipher(), kDefaultTimeout);
>-  EXPECT_EQ(1,
>-            caller_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher,
>-                                            kDefaultSrtpCryptoSuite));
>+  // TODO(bugs.webrtc.org/9456): Fix it.
>+  EXPECT_EQ(1, webrtc::metrics::NumEvents(
>+                   "WebRTC.PeerConnection.SrtpCryptoSuite.Audio",
>+                   kDefaultSrtpCryptoSuite));
> }
> 
> // Test getting cipher stats and UMA metrics when DTLS 1.2 is negotiated.
>-TEST_F(PeerConnectionIntegrationTest, Dtls12CipherStatsAndUmaMetrics) {
>+TEST_P(PeerConnectionIntegrationTest, Dtls12CipherStatsAndUmaMetrics) {
>   PeerConnectionFactory::Options dtls_12_options;
>   dtls_12_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
>   ASSERT_TRUE(CreatePeerConnectionWrappersWithOptions(dtls_12_options,
>                                                       dtls_12_options));
>   ConnectFakeSignaling();
>-  // Register UMA observer before signaling begins.
>-  rtc::scoped_refptr<webrtc::FakeMetricsObserver> caller_observer =
>-      new rtc::RefCountedObject<webrtc::FakeMetricsObserver>();
>-  caller()->pc()->RegisterUMAObserver(caller_observer);
>-  caller()->AddAudioVideoMediaStream();
>-  callee()->AddAudioVideoMediaStream();
>+  caller()->AddAudioVideoTracks();
>+  callee()->AddAudioVideoTracks();
>   caller()->CreateAndSetAndSignalOffer();
>-  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>+  ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
>   EXPECT_TRUE_WAIT(rtc::SSLStreamAdapter::IsAcceptableCipher(
>                        caller()->OldGetStats()->DtlsCipher(), rtc::KT_DEFAULT),
>                    kDefaultTimeout);
>   EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite),
>                  caller()->OldGetStats()->SrtpCipher(), kDefaultTimeout);
>-  EXPECT_EQ(1,
>-            caller_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher,
>-                                            kDefaultSrtpCryptoSuite));
>+  // TODO(bugs.webrtc.org/9456): Fix it.
>+  EXPECT_EQ(1, webrtc::metrics::NumEvents(
>+                   "WebRTC.PeerConnection.SrtpCryptoSuite.Audio",
>+                   kDefaultSrtpCryptoSuite));
> }
> 
> // Test that DTLS 1.0 can be used if the caller supports DTLS 1.2 and the
> // callee only supports 1.0.
>-TEST_F(PeerConnectionIntegrationTest, CallerDtls12ToCalleeDtls10) {
>+TEST_P(PeerConnectionIntegrationTest, CallerDtls12ToCalleeDtls10) {
>   PeerConnectionFactory::Options caller_options;
>   caller_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_12;
>   PeerConnectionFactory::Options callee_options;
>@@ -2276,19 +2780,18 @@ TEST_F(PeerConnectionIntegrationTest, CallerDtls12ToCalleeDtls10) {
>   ConnectFakeSignaling();
>   // Do normal offer/answer and wait for some frames to be received in each
>   // direction.
>-  caller()->AddAudioVideoMediaStream();
>-  callee()->AddAudioVideoMediaStream();
>+  caller()->AddAudioVideoTracks();
>+  callee()->AddAudioVideoTracks();
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>-  ExpectNewFramesReceivedWithWait(
>-      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
>-      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
>-      kMaxWaitForFramesMs);
>+  MediaExpectations media_expectations;
>+  media_expectations.ExpectBidirectionalAudioAndVideo();
>+  ASSERT_TRUE(ExpectNewFrames(media_expectations));
> }
> 
> // Test that DTLS 1.0 can be used if the caller only supports DTLS 1.0 and the
> // callee supports 1.2.
>-TEST_F(PeerConnectionIntegrationTest, CallerDtls10ToCalleeDtls12) {
>+TEST_P(PeerConnectionIntegrationTest, CallerDtls10ToCalleeDtls12) {
>   PeerConnectionFactory::Options caller_options;
>   caller_options.ssl_max_version = rtc::SSL_PROTOCOL_DTLS_10;
>   PeerConnectionFactory::Options callee_options;
>@@ -2298,18 +2801,51 @@ TEST_F(PeerConnectionIntegrationTest, CallerDtls10ToCalleeDtls12) {
>   ConnectFakeSignaling();
>   // Do normal offer/answer and wait for some frames to be received in each
>   // direction.
>-  caller()->AddAudioVideoMediaStream();
>-  callee()->AddAudioVideoMediaStream();
>+  caller()->AddAudioVideoTracks();
>+  callee()->AddAudioVideoTracks();
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>-  ExpectNewFramesReceivedWithWait(
>-      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
>-      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
>-      kMaxWaitForFramesMs);
>+  MediaExpectations media_expectations;
>+  media_expectations.ExpectBidirectionalAudioAndVideo();
>+  ASSERT_TRUE(ExpectNewFrames(media_expectations));
>+}
>+
>+// The three tests below verify that "enable_aes128_sha1_32_crypto_cipher"
>+// works as expected; the cipher should only be used if enabled by both sides.
>+TEST_P(PeerConnectionIntegrationTest,
>+       Aes128Sha1_32_CipherNotUsedWhenOnlyCallerSupported) {
>+  PeerConnectionFactory::Options caller_options;
>+  caller_options.crypto_options.enable_aes128_sha1_32_crypto_cipher = true;
>+  PeerConnectionFactory::Options callee_options;
>+  callee_options.crypto_options.enable_aes128_sha1_32_crypto_cipher = false;
>+  int expected_cipher_suite = rtc::SRTP_AES128_CM_SHA1_80;
>+  TestNegotiatedCipherSuite(caller_options, callee_options,
>+                            expected_cipher_suite);
>+}
>+
>+TEST_P(PeerConnectionIntegrationTest,
>+       Aes128Sha1_32_CipherNotUsedWhenOnlyCalleeSupported) {
>+  PeerConnectionFactory::Options caller_options;
>+  caller_options.crypto_options.enable_aes128_sha1_32_crypto_cipher = false;
>+  PeerConnectionFactory::Options callee_options;
>+  callee_options.crypto_options.enable_aes128_sha1_32_crypto_cipher = true;
>+  int expected_cipher_suite = rtc::SRTP_AES128_CM_SHA1_80;
>+  TestNegotiatedCipherSuite(caller_options, callee_options,
>+                            expected_cipher_suite);
>+}
>+
>+TEST_P(PeerConnectionIntegrationTest, Aes128Sha1_32_CipherUsedWhenSupported) {
>+  PeerConnectionFactory::Options caller_options;
>+  caller_options.crypto_options.enable_aes128_sha1_32_crypto_cipher = true;
>+  PeerConnectionFactory::Options callee_options;
>+  callee_options.crypto_options.enable_aes128_sha1_32_crypto_cipher = true;
>+  int expected_cipher_suite = rtc::SRTP_AES128_CM_SHA1_32;
>+  TestNegotiatedCipherSuite(caller_options, callee_options,
>+                            expected_cipher_suite);
> }
> 
> // Test that a non-GCM cipher is used if both sides only support non-GCM.
>-TEST_F(PeerConnectionIntegrationTest, NonGcmCipherUsedWhenGcmNotSupported) {
>+TEST_P(PeerConnectionIntegrationTest, NonGcmCipherUsedWhenGcmNotSupported) {
>   bool local_gcm_enabled = false;
>   bool remote_gcm_enabled = false;
>   int expected_cipher_suite = kDefaultSrtpCryptoSuite;
>@@ -2318,7 +2854,7 @@ TEST_F(PeerConnectionIntegrationTest, NonGcmCipherUsedWhenGcmNotSupported) {
> }
> 
> // Test that a GCM cipher is used if both ends support it.
>-TEST_F(PeerConnectionIntegrationTest, GcmCipherUsedWhenGcmSupported) {
>+TEST_P(PeerConnectionIntegrationTest, GcmCipherUsedWhenGcmSupported) {
>   bool local_gcm_enabled = true;
>   bool remote_gcm_enabled = true;
>   int expected_cipher_suite = kDefaultSrtpCryptoSuiteGcm;
>@@ -2327,7 +2863,7 @@ TEST_F(PeerConnectionIntegrationTest, GcmCipherUsedWhenGcmSupported) {
> }
> 
> // Test that GCM isn't used if only the offerer supports it.
>-TEST_F(PeerConnectionIntegrationTest,
>+TEST_P(PeerConnectionIntegrationTest,
>        NonGcmCipherUsedWhenOnlyCallerSupportsGcm) {
>   bool local_gcm_enabled = true;
>   bool remote_gcm_enabled = false;
>@@ -2337,7 +2873,7 @@ TEST_F(PeerConnectionIntegrationTest,
> }
> 
> // Test that GCM isn't used if only the answerer supports it.
>-TEST_F(PeerConnectionIntegrationTest,
>+TEST_P(PeerConnectionIntegrationTest,
>        NonGcmCipherUsedWhenOnlyCalleeSupportsGcm) {
>   bool local_gcm_enabled = false;
>   bool remote_gcm_enabled = true;
>@@ -2350,7 +2886,7 @@ TEST_F(PeerConnectionIntegrationTest,
> // enabled. Note that the above tests, such as GcmCipherUsedWhenGcmSupported,
> // only verify that a GCM cipher is negotiated, and not necessarily that SRTP
> // works with it.
>-TEST_F(PeerConnectionIntegrationTest, EndToEndCallWithGcmCipher) {
>+TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithGcmCipher) {
>   PeerConnectionFactory::Options gcm_options;
>   gcm_options.crypto_options.enable_gcm_crypto_suites = true;
>   ASSERT_TRUE(
>@@ -2358,19 +2894,18 @@ TEST_F(PeerConnectionIntegrationTest, EndToEndCallWithGcmCipher) {
>   ConnectFakeSignaling();
>   // Do normal offer/answer and wait for some frames to be received in each
>   // direction.
>-  caller()->AddAudioVideoMediaStream();
>-  callee()->AddAudioVideoMediaStream();
>+  caller()->AddAudioVideoTracks();
>+  callee()->AddAudioVideoTracks();
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>-  ExpectNewFramesReceivedWithWait(
>-      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
>-      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
>-      kMaxWaitForFramesMs);
>+  MediaExpectations media_expectations;
>+  media_expectations.ExpectBidirectionalAudioAndVideo();
>+  ASSERT_TRUE(ExpectNewFrames(media_expectations));
> }
> 
> // This test sets up a call between two parties with audio, video and an RTP
> // data channel.
>-TEST_F(PeerConnectionIntegrationTest, EndToEndCallWithRtpDataChannel) {
>+TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithRtpDataChannel) {
>   FakeConstraints setup_constraints;
>   setup_constraints.SetAllowRtpDataChannels();
>   ASSERT_TRUE(CreatePeerConnectionWrappersWithConstraints(&setup_constraints,
>@@ -2379,15 +2914,14 @@ TEST_F(PeerConnectionIntegrationTest, EndToEndCallWithRtpDataChannel) {
>   // Expect that data channel created on caller side will show up for callee as
>   // well.
>   caller()->CreateDataChannel();
>-  caller()->AddAudioVideoMediaStream();
>-  callee()->AddAudioVideoMediaStream();
>+  caller()->AddAudioVideoTracks();
>+  callee()->AddAudioVideoTracks();
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>   // Ensure the existence of the RTP data channel didn't impede audio/video.
>-  ExpectNewFramesReceivedWithWait(
>-      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
>-      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
>-      kMaxWaitForFramesMs);
>+  MediaExpectations media_expectations;
>+  media_expectations.ExpectBidirectionalAudioAndVideo();
>+  ASSERT_TRUE(ExpectNewFrames(media_expectations));
>   ASSERT_NE(nullptr, caller()->data_channel());
>   ASSERT_NE(nullptr, callee()->data_channel());
>   EXPECT_TRUE_WAIT(caller()->data_observer()->IsOpen(), kDefaultTimeout);
>@@ -2405,7 +2939,7 @@ TEST_F(PeerConnectionIntegrationTest, EndToEndCallWithRtpDataChannel) {
> 
> // Ensure that an RTP data channel is signaled as closed for the caller when
> // the callee rejects it in a subsequent offer.
>-TEST_F(PeerConnectionIntegrationTest,
>+TEST_P(PeerConnectionIntegrationTest,
>        RtpDataChannelSignaledClosedInCalleeOffer) {
>   // Same procedure as above test.
>   FakeConstraints setup_constraints;
>@@ -2414,8 +2948,8 @@ TEST_F(PeerConnectionIntegrationTest,
>                                                           &setup_constraints));
>   ConnectFakeSignaling();
>   caller()->CreateDataChannel();
>-  caller()->AddAudioVideoMediaStream();
>-  callee()->AddAudioVideoMediaStream();
>+  caller()->AddAudioVideoTracks();
>+  callee()->AddAudioVideoTracks();
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>   ASSERT_NE(nullptr, caller()->data_channel());
>@@ -2438,7 +2972,7 @@ TEST_F(PeerConnectionIntegrationTest,
> // transport has detected that a channel is writable and thus data can be
> // received before the data channel state changes to open. That is hard to test
> // but the same buffering is expected to be used in that case.
>-TEST_F(PeerConnectionIntegrationTest,
>+TEST_P(PeerConnectionIntegrationTest,
>        DataBufferedUntilRtpDataChannelObserverRegistered) {
>   // Use fake clock and simulated network delay so that we predictably can wait
>   // until an SCTP message has been delivered without "sleep()"ing.
>@@ -2446,7 +2980,7 @@ TEST_F(PeerConnectionIntegrationTest,
>   // Some things use a time of "0" as a special value, so we need to start out
>   // the fake clock at a nonzero time.
>   // TODO(deadbeef): Fix this.
>-  fake_clock.AdvanceTime(rtc::TimeDelta::FromSeconds(1));
>+  fake_clock.AdvanceTime(webrtc::TimeDelta::seconds(1));
>   virtual_socket_server()->set_delay_mean(5);  // 5 ms per hop.
>   virtual_socket_server()->UpdateDelayDistribution();
> 
>@@ -2479,11 +3013,16 @@ TEST_F(PeerConnectionIntegrationTest,
>   MockDataChannelObserver new_observer(callee()->data_channel());
>   EXPECT_EQ_SIMULATED_WAIT(data, new_observer.last_message(), kDefaultTimeout,
>                            fake_clock);
>+  // Closing the PeerConnections destroys the ports before the ScopedFakeClock.
>+  // If this is not done a DCHECK can be hit in ports.cc, because a large
>+  // negative number is calculated for the rtt due to the global clock changing.
>+  caller()->pc()->Close();
>+  callee()->pc()->Close();
> }
> 
> // This test sets up a call between two parties with audio, video and but only
> // the caller client supports RTP data channels.
>-TEST_F(PeerConnectionIntegrationTest, RtpDataChannelsRejectedByCallee) {
>+TEST_P(PeerConnectionIntegrationTest, RtpDataChannelsRejectedByCallee) {
>   FakeConstraints setup_constraints_1;
>   setup_constraints_1.SetAllowRtpDataChannels();
>   // Must disable DTLS to make negotiation succeed.
>@@ -2496,8 +3035,8 @@ TEST_F(PeerConnectionIntegrationTest, RtpDataChannelsRejectedByCallee) {
>       &setup_constraints_1, &setup_constraints_2));
>   ConnectFakeSignaling();
>   caller()->CreateDataChannel();
>-  caller()->AddAudioVideoMediaStream();
>-  callee()->AddAudioVideoMediaStream();
>+  caller()->AddAudioVideoTracks();
>+  callee()->AddAudioVideoTracks();
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>   // The caller should still have a data channel, but it should be closed, and
>@@ -2509,15 +3048,15 @@ TEST_F(PeerConnectionIntegrationTest, RtpDataChannelsRejectedByCallee) {
> 
> // This test sets up a call between two parties with audio, and video. When
> // audio and video is setup and flowing, an RTP data channel is negotiated.
>-TEST_F(PeerConnectionIntegrationTest, AddRtpDataChannelInSubsequentOffer) {
>+TEST_P(PeerConnectionIntegrationTest, AddRtpDataChannelInSubsequentOffer) {
>   FakeConstraints setup_constraints;
>   setup_constraints.SetAllowRtpDataChannels();
>   ASSERT_TRUE(CreatePeerConnectionWrappersWithConstraints(&setup_constraints,
>                                                           &setup_constraints));
>   ConnectFakeSignaling();
>   // Do initial offer/answer with audio/video.
>-  caller()->AddAudioVideoMediaStream();
>-  callee()->AddAudioVideoMediaStream();
>+  caller()->AddAudioVideoTracks();
>+  callee()->AddAudioVideoTracks();
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>   // Create data channel and do new offer and answer.
>@@ -2542,21 +3081,20 @@ TEST_F(PeerConnectionIntegrationTest, AddRtpDataChannelInSubsequentOffer) {
> 
> // This test sets up a call between two parties with audio, video and an SCTP
> // data channel.
>-TEST_F(PeerConnectionIntegrationTest, EndToEndCallWithSctpDataChannel) {
>+TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithSctpDataChannel) {
>   ASSERT_TRUE(CreatePeerConnectionWrappers());
>   ConnectFakeSignaling();
>   // Expect that data channel created on caller side will show up for callee as
>   // well.
>   caller()->CreateDataChannel();
>-  caller()->AddAudioVideoMediaStream();
>-  callee()->AddAudioVideoMediaStream();
>+  caller()->AddAudioVideoTracks();
>+  callee()->AddAudioVideoTracks();
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>   // Ensure the existence of the SCTP data channel didn't impede audio/video.
>-  ExpectNewFramesReceivedWithWait(
>-      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
>-      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
>-      kMaxWaitForFramesMs);
>+  MediaExpectations media_expectations;
>+  media_expectations.ExpectBidirectionalAudioAndVideo();
>+  ASSERT_TRUE(ExpectNewFrames(media_expectations));
>   // Caller data channel should already exist (it created one). Callee data
>   // channel may not exist yet, since negotiation happens in-band, not in SDP.
>   ASSERT_NE(nullptr, caller()->data_channel());
>@@ -2576,13 +3114,13 @@ TEST_F(PeerConnectionIntegrationTest, EndToEndCallWithSctpDataChannel) {
> 
> // Ensure that when the callee closes an SCTP data channel, the closing
> // procedure results in the data channel being closed for the caller as well.
>-TEST_F(PeerConnectionIntegrationTest, CalleeClosesSctpDataChannel) {
>+TEST_P(PeerConnectionIntegrationTest, CalleeClosesSctpDataChannel) {
>   // Same procedure as above test.
>   ASSERT_TRUE(CreatePeerConnectionWrappers());
>   ConnectFakeSignaling();
>   caller()->CreateDataChannel();
>-  caller()->AddAudioVideoMediaStream();
>-  callee()->AddAudioVideoMediaStream();
>+  caller()->AddAudioVideoTracks();
>+  callee()->AddAudioVideoTracks();
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>   ASSERT_NE(nullptr, caller()->data_channel());
>@@ -2597,15 +3135,15 @@ TEST_F(PeerConnectionIntegrationTest, CalleeClosesSctpDataChannel) {
>   EXPECT_TRUE_WAIT(!callee()->data_observer()->IsOpen(), kDefaultTimeout);
> }
> 
>-TEST_F(PeerConnectionIntegrationTest, SctpDataChannelConfigSentToOtherSide) {
>+TEST_P(PeerConnectionIntegrationTest, SctpDataChannelConfigSentToOtherSide) {
>   ASSERT_TRUE(CreatePeerConnectionWrappers());
>   ConnectFakeSignaling();
>   webrtc::DataChannelInit init;
>   init.id = 53;
>   init.maxRetransmits = 52;
>   caller()->CreateDataChannel("data-channel", &init);
>-  caller()->AddAudioVideoMediaStream();
>-  callee()->AddAudioVideoMediaStream();
>+  caller()->AddAudioVideoTracks();
>+  callee()->AddAudioVideoTracks();
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>   ASSERT_TRUE_WAIT(callee()->data_channel() != nullptr, kDefaultTimeout);
>@@ -2619,7 +3157,7 @@ TEST_F(PeerConnectionIntegrationTest, SctpDataChannelConfigSentToOtherSide) {
> // Test usrsctp's ability to process unordered data stream, where data actually
> // arrives out of order using simulated delays. Previously there have been some
> // bugs in this area.
>-TEST_F(PeerConnectionIntegrationTest, StressTestUnorderedSctpDataChannel) {
>+TEST_P(PeerConnectionIntegrationTest, StressTestUnorderedSctpDataChannel) {
>   // Introduce random network delays.
>   // Otherwise it's not a true "unordered" test.
>   virtual_socket_server()->set_delay_mean(20);
>@@ -2654,10 +3192,10 @@ TEST_F(PeerConnectionIntegrationTest, StressTestUnorderedSctpDataChannel) {
>   }
> 
>   // Wait for all messages to be received.
>-  EXPECT_EQ_WAIT(kNumMessages,
>+  EXPECT_EQ_WAIT(rtc::checked_cast<size_t>(kNumMessages),
>                  caller()->data_observer()->received_message_count(),
>                  kDefaultTimeout);
>-  EXPECT_EQ_WAIT(kNumMessages,
>+  EXPECT_EQ_WAIT(rtc::checked_cast<size_t>(kNumMessages),
>                  callee()->data_observer()->received_message_count(),
>                  kDefaultTimeout);
> 
>@@ -2675,12 +3213,12 @@ TEST_F(PeerConnectionIntegrationTest, StressTestUnorderedSctpDataChannel) {
> 
> // This test sets up a call between two parties with audio, and video. When
> // audio and video are setup and flowing, an SCTP data channel is negotiated.
>-TEST_F(PeerConnectionIntegrationTest, AddSctpDataChannelInSubsequentOffer) {
>+TEST_P(PeerConnectionIntegrationTest, AddSctpDataChannelInSubsequentOffer) {
>   ASSERT_TRUE(CreatePeerConnectionWrappers());
>   ConnectFakeSignaling();
>   // Do initial offer/answer with audio/video.
>-  caller()->AddAudioVideoMediaStream();
>-  callee()->AddAudioVideoMediaStream();
>+  caller()->AddAudioVideoTracks();
>+  callee()->AddAudioVideoTracks();
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>   // Create data channel and do new offer and answer.
>@@ -2707,7 +3245,7 @@ TEST_F(PeerConnectionIntegrationTest, AddSctpDataChannelInSubsequentOffer) {
> // to audio/video, ensuring frames are received end-to-end. Effectively the
> // inverse of the test above.
> // This was broken in M57; see https://crbug.com/711243
>-TEST_F(PeerConnectionIntegrationTest, SctpDataChannelToAudioVideoUpgrade) {
>+TEST_P(PeerConnectionIntegrationTest, SctpDataChannelToAudioVideoUpgrade) {
>   ASSERT_TRUE(CreatePeerConnectionWrappers());
>   ConnectFakeSignaling();
>   // Do initial offer/answer with just data channel.
>@@ -2721,14 +3259,13 @@ TEST_F(PeerConnectionIntegrationTest, SctpDataChannelToAudioVideoUpgrade) {
> 
>   // Do subsequent offer/answer with two-way audio and video. Audio and video
>   // should end up bundled on the DTLS/ICE transport already used for data.
>-  caller()->AddAudioVideoMediaStream();
>-  callee()->AddAudioVideoMediaStream();
>+  caller()->AddAudioVideoTracks();
>+  callee()->AddAudioVideoTracks();
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>-  ExpectNewFramesReceivedWithWait(
>-      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
>-      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
>-      kMaxWaitForFramesMs);
>+  MediaExpectations media_expectations;
>+  media_expectations.ExpectBidirectionalAudioAndVideo();
>+  ASSERT_TRUE(ExpectNewFrames(media_expectations));
> }
> 
> static void MakeSpecCompliantSctpOffer(cricket::SessionDescription* desc) {
>@@ -2742,7 +3279,7 @@ static void MakeSpecCompliantSctpOffer(cricket::SessionDescription* desc) {
> // Test that the data channel works when a spec-compliant SCTP m= section is
> // offered (using "a=sctp-port" instead of "a=sctpmap", and using
> // "UDP/DTLS/SCTP" as the protocol).
>-TEST_F(PeerConnectionIntegrationTest,
>+TEST_P(PeerConnectionIntegrationTest,
>        DataChannelWorksWhenSpecCompliantSctpOfferReceived) {
>   ASSERT_TRUE(CreatePeerConnectionWrappers());
>   ConnectFakeSignaling();
>@@ -2768,12 +3305,12 @@ TEST_F(PeerConnectionIntegrationTest,
> 
> // Test that the ICE connection and gathering states eventually reach
> // "complete".
>-TEST_F(PeerConnectionIntegrationTest, IceStatesReachCompletion) {
>+TEST_P(PeerConnectionIntegrationTest, IceStatesReachCompletion) {
>   ASSERT_TRUE(CreatePeerConnectionWrappers());
>   ConnectFakeSignaling();
>   // Do normal offer/answer.
>-  caller()->AddAudioVideoMediaStream();
>-  callee()->AddAudioVideoMediaStream();
>+  caller()->AddAudioVideoTracks();
>+  callee()->AddAudioVideoTracks();
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>   EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceGatheringComplete,
>@@ -2794,11 +3331,13 @@ TEST_F(PeerConnectionIntegrationTest, IceStatesReachCompletion) {
> // Test that firewalling the ICE connection causes the clients to identify the
> // disconnected state and then removing the firewall causes them to reconnect.
> class PeerConnectionIntegrationIceStatesTest
>-    : public PeerConnectionIntegrationTest,
>-      public ::testing::WithParamInterface<std::tuple<std::string, uint32_t>> {
>+    : public PeerConnectionIntegrationBaseTest,
>+      public ::testing::WithParamInterface<
>+          std::tuple<SdpSemantics, std::tuple<std::string, uint32_t>>> {
>  protected:
>-  PeerConnectionIntegrationIceStatesTest() {
>-    port_allocator_flags_ = std::get<1>(GetParam());
>+  PeerConnectionIntegrationIceStatesTest()
>+      : PeerConnectionIntegrationBaseTest(std::get<0>(GetParam())) {
>+    port_allocator_flags_ = std::get<1>(std::get<1>(GetParam()));
>   }
> 
>   void StartStunServer(const SocketAddress& server_address) {
>@@ -2811,8 +3350,14 @@ class PeerConnectionIntegrationIceStatesTest
>   }
> 
>   void SetPortAllocatorFlags() {
>-    caller()->port_allocator()->set_flags(port_allocator_flags_);
>-    callee()->port_allocator()->set_flags(port_allocator_flags_);
>+    network_thread()->Invoke<void>(
>+        RTC_FROM_HERE,
>+        rtc::Bind(&cricket::PortAllocator::set_flags,
>+                  caller()->port_allocator(), port_allocator_flags_));
>+    network_thread()->Invoke<void>(
>+        RTC_FROM_HERE,
>+        rtc::Bind(&cricket::PortAllocator::set_flags,
>+                  callee()->port_allocator(), port_allocator_flags_));
>   }
> 
>   std::vector<SocketAddress> CallerAddresses() {
>@@ -2876,8 +3421,8 @@ TEST_P(PeerConnectionIntegrationIceStatesTest, VerifyIceStates) {
>   ConnectFakeSignaling();
>   SetPortAllocatorFlags();
>   SetUpNetworkInterfaces();
>-  caller()->AddAudioVideoMediaStream();
>-  callee()->AddAudioVideoMediaStream();
>+  caller()->AddAudioVideoTracks();
>+  callee()->AddAudioVideoTracks();
> 
>   // Initial state before anything happens.
>   ASSERT_EQ(PeerConnectionInterface::kIceGatheringNew,
>@@ -2937,37 +3482,33 @@ TEST_P(PeerConnectionIntegrationIceStatesTest, VerifyBestConnection) {
>   ConnectFakeSignaling();
>   SetPortAllocatorFlags();
>   SetUpNetworkInterfaces();
>-  caller()->AddAudioVideoMediaStream();
>-  callee()->AddAudioVideoMediaStream();
>-
>-  rtc::scoped_refptr<webrtc::FakeMetricsObserver> metrics_observer(
>-      new rtc::RefCountedObject<webrtc::FakeMetricsObserver>());
>-  caller()->pc()->RegisterUMAObserver(metrics_observer.get());
>-
>+  caller()->AddAudioVideoTracks();
>+  callee()->AddAudioVideoTracks();
>   caller()->CreateAndSetAndSignalOffer();
> 
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
> 
>-  const int num_best_ipv4 = metrics_observer->GetEnumCounter(
>-      webrtc::kEnumCounterAddressFamily, webrtc::kBestConnections_IPv4);
>-  const int num_best_ipv6 = metrics_observer->GetEnumCounter(
>-      webrtc::kEnumCounterAddressFamily, webrtc::kBestConnections_IPv6);
>+  // TODO(bugs.webrtc.org/9456): Fix it.
>+  const int num_best_ipv4 = webrtc::metrics::NumEvents(
>+      "WebRTC.PeerConnection.IPMetrics", webrtc::kBestConnections_IPv4);
>+  const int num_best_ipv6 = webrtc::metrics::NumEvents(
>+      "WebRTC.PeerConnection.IPMetrics", webrtc::kBestConnections_IPv6);
>   if (TestIPv6()) {
>     // When IPv6 is enabled, we should prefer an IPv6 connection over an IPv4
>     // connection.
>-    EXPECT_EQ(0u, num_best_ipv4);
>-    EXPECT_EQ(1u, num_best_ipv6);
>+    EXPECT_EQ(0, num_best_ipv4);
>+    EXPECT_EQ(1, num_best_ipv6);
>   } else {
>-    EXPECT_EQ(1u, num_best_ipv4);
>-    EXPECT_EQ(0u, num_best_ipv6);
>+    EXPECT_EQ(1, num_best_ipv4);
>+    EXPECT_EQ(0, num_best_ipv6);
>   }
> 
>-  EXPECT_EQ(0u, metrics_observer->GetEnumCounter(
>-                    webrtc::kEnumCounterIceCandidatePairTypeUdp,
>-                    webrtc::kIceCandidatePairHostHost));
>-  EXPECT_EQ(1u, metrics_observer->GetEnumCounter(
>-                    webrtc::kEnumCounterIceCandidatePairTypeUdp,
>-                    webrtc::kIceCandidatePairHostPublicHostPublic));
>+  EXPECT_EQ(0, webrtc::metrics::NumEvents(
>+                   "WebRTC.PeerConnection.CandidatePairType_UDP",
>+                   webrtc::kIceCandidatePairHostHost));
>+  EXPECT_EQ(1, webrtc::metrics::NumEvents(
>+                   "WebRTC.PeerConnection.CandidatePairType_UDP",
>+                   webrtc::kIceCandidatePairHostPublicHostPublic));
> }
> 
> constexpr uint32_t kFlagsIPv4NoStun = cricket::PORTALLOCATOR_DISABLE_TCP |
>@@ -2979,23 +3520,24 @@ constexpr uint32_t kFlagsIPv6NoStun =
> constexpr uint32_t kFlagsIPv4Stun =
>     cricket::PORTALLOCATOR_DISABLE_TCP | cricket::PORTALLOCATOR_DISABLE_RELAY;
> 
>-INSTANTIATE_TEST_CASE_P(PeerConnectionIntegrationTest,
>-                        PeerConnectionIntegrationIceStatesTest,
>-                        Values(std::make_pair("IPv4 no STUN", kFlagsIPv4NoStun),
>-                               std::make_pair("IPv6 no STUN", kFlagsIPv6NoStun),
>-                               std::make_pair("IPv4 with STUN",
>-                                              kFlagsIPv4Stun)));
>+INSTANTIATE_TEST_CASE_P(
>+    PeerConnectionIntegrationTest,
>+    PeerConnectionIntegrationIceStatesTest,
>+    Combine(Values(SdpSemantics::kPlanB, SdpSemantics::kUnifiedPlan),
>+            Values(std::make_pair("IPv4 no STUN", kFlagsIPv4NoStun),
>+                   std::make_pair("IPv6 no STUN", kFlagsIPv6NoStun),
>+                   std::make_pair("IPv4 with STUN", kFlagsIPv4Stun))));
> 
> // This test sets up a call between two parties with audio and video.
> // During the call, the caller restarts ICE and the test verifies that
> // new ICE candidates are generated and audio and video still can flow, and the
> // ICE state reaches completed again.
>-TEST_F(PeerConnectionIntegrationTest, MediaContinuesFlowingAfterIceRestart) {
>+TEST_P(PeerConnectionIntegrationTest, MediaContinuesFlowingAfterIceRestart) {
>   ASSERT_TRUE(CreatePeerConnectionWrappers());
>   ConnectFakeSignaling();
>   // Do normal offer/answer and wait for ICE to complete.
>-  caller()->AddAudioVideoMediaStream();
>-  callee()->AddAudioVideoMediaStream();
>+  caller()->AddAudioVideoTracks();
>+  callee()->AddAudioVideoTracks();
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>   EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
>@@ -3059,23 +3601,22 @@ TEST_F(PeerConnectionIntegrationTest, MediaContinuesFlowingAfterIceRestart) {
>   ASSERT_NE(callee_ufrag_pre_restart, callee_ufrag_post_restart);
> 
>   // Ensure that additional frames are received after the ICE restart.
>-  ExpectNewFramesReceivedWithWait(
>-      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
>-      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
>-      kMaxWaitForFramesMs);
>+  MediaExpectations media_expectations;
>+  media_expectations.ExpectBidirectionalAudioAndVideo();
>+  ASSERT_TRUE(ExpectNewFrames(media_expectations));
> }
> 
> // Verify that audio/video can be received end-to-end when ICE renomination is
> // enabled.
>-TEST_F(PeerConnectionIntegrationTest, EndToEndCallWithIceRenomination) {
>+TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithIceRenomination) {
>   PeerConnectionInterface::RTCConfiguration config;
>   config.enable_ice_renomination = true;
>   ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(config, config));
>   ConnectFakeSignaling();
>   // Do normal offer/answer and wait for some frames to be received in each
>   // direction.
>-  caller()->AddAudioVideoMediaStream();
>-  callee()->AddAudioVideoMediaStream();
>+  caller()->AddAudioVideoTracks();
>+  callee()->AddAudioVideoTracks();
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>   // Sanity check that ICE renomination was actually negotiated.
>@@ -3094,16 +3635,15 @@ TEST_F(PeerConnectionIntegrationTest, EndToEndCallWithIceRenomination) {
>         std::find(info.description.transport_options.begin(),
>                   info.description.transport_options.end(), "renomination"));
>   }
>-  ExpectNewFramesReceivedWithWait(
>-      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
>-      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
>-      kMaxWaitForFramesMs);
>+  MediaExpectations media_expectations;
>+  media_expectations.ExpectBidirectionalAudioAndVideo();
>+  ASSERT_TRUE(ExpectNewFrames(media_expectations));
> }
> 
> // With a max bundle policy and RTCP muxing, adding a new media description to
> // the connection should not affect ICE at all because the new media will use
> // the existing connection.
>-TEST_F(PeerConnectionIntegrationTest,
>+TEST_P(PeerConnectionIntegrationTest,
>        AddMediaToConnectedBundleDoesNotRestartIce) {
>   PeerConnectionInterface::RTCConfiguration config;
>   config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
>@@ -3112,7 +3652,7 @@ TEST_F(PeerConnectionIntegrationTest,
>       config, PeerConnectionInterface::RTCConfiguration()));
>   ConnectFakeSignaling();
> 
>-  caller()->AddAudioOnlyMediaStream();
>+  caller()->AddAudioTrack();
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>   ASSERT_EQ_WAIT(PeerConnectionInterface::kIceConnectionCompleted,
>@@ -3120,7 +3660,7 @@ TEST_F(PeerConnectionIntegrationTest,
> 
>   caller()->clear_ice_connection_state_history();
> 
>-  caller()->AddVideoOnlyMediaStream();
>+  caller()->AddVideoTrack();
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
> 
>@@ -3130,22 +3670,28 @@ TEST_F(PeerConnectionIntegrationTest,
> // This test sets up a call between two parties with audio and video. It then
> // renegotiates setting the video m-line to "port 0", then later renegotiates
> // again, enabling video.
>-TEST_F(PeerConnectionIntegrationTest,
>+TEST_P(PeerConnectionIntegrationTest,
>        VideoFlowsAfterMediaSectionIsRejectedAndRecycled) {
>   ASSERT_TRUE(CreatePeerConnectionWrappers());
>   ConnectFakeSignaling();
> 
>   // Do initial negotiation, only sending media from the caller. Will result in
>   // video and audio recvonly "m=" sections.
>-  caller()->AddAudioVideoMediaStream();
>+  caller()->AddAudioVideoTracks();
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
> 
>   // Negotiate again, disabling the video "m=" section (the callee will set the
>   // port to 0 due to offer_to_receive_video = 0).
>-  PeerConnectionInterface::RTCOfferAnswerOptions options;
>-  options.offer_to_receive_video = 0;
>-  callee()->SetOfferAnswerOptions(options);
>+  if (sdp_semantics_ == SdpSemantics::kPlanB) {
>+    PeerConnectionInterface::RTCOfferAnswerOptions options;
>+    options.offer_to_receive_video = 0;
>+    callee()->SetOfferAnswerOptions(options);
>+  } else {
>+    callee()->SetRemoteOfferHandler([this] {
>+      callee()->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_VIDEO)->Stop();
>+    });
>+  }
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>   // Sanity check that video "m=" section was actually rejected.
>@@ -3156,44 +3702,34 @@ TEST_F(PeerConnectionIntegrationTest,
> 
>   // Enable video and do negotiation again, making sure video is received
>   // end-to-end, also adding media stream to callee.
>-  options.offer_to_receive_video = 1;
>-  callee()->SetOfferAnswerOptions(options);
>-  callee()->AddAudioVideoMediaStream();
>+  if (sdp_semantics_ == SdpSemantics::kPlanB) {
>+    PeerConnectionInterface::RTCOfferAnswerOptions options;
>+    options.offer_to_receive_video = 1;
>+    callee()->SetOfferAnswerOptions(options);
>+  } else {
>+    // The caller's transceiver is stopped, so we need to add another track.
>+    auto caller_transceiver =
>+        caller()->GetFirstTransceiverOfType(cricket::MEDIA_TYPE_VIDEO);
>+    EXPECT_TRUE(caller_transceiver->stopped());
>+    caller()->AddVideoTrack();
>+  }
>+  callee()->AddVideoTrack();
>+  callee()->SetRemoteOfferHandler(nullptr);
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>+
>   // Verify the caller receives frames from the newly added stream, and the
>   // callee receives additional frames from the re-enabled video m= section.
>-  ExpectNewFramesReceivedWithWait(
>-      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
>-      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
>-      kMaxWaitForFramesMs);
>-}
>-
>-// This test sets up a Jsep call between two parties with external
>-// VideoDecoderFactory.
>-// TODO(holmer): Disabled due to sometimes crashing on buildbots.
>-// See issue webrtc/2378.
>-TEST_F(PeerConnectionIntegrationTest,
>-       DISABLED_EndToEndCallWithVideoDecoderFactory) {
>-  ASSERT_TRUE(CreatePeerConnectionWrappers());
>-  EnableVideoDecoderFactory();
>-  ConnectFakeSignaling();
>-  caller()->AddAudioVideoMediaStream();
>-  callee()->AddAudioVideoMediaStream();
>-  caller()->CreateAndSetAndSignalOffer();
>-  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>-  ExpectNewFramesReceivedWithWait(
>-      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
>-      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
>-      kMaxWaitForFramesMs);
>+  MediaExpectations media_expectations;
>+  media_expectations.CalleeExpectsSomeAudio();
>+  media_expectations.ExpectBidirectionalVideo();
>+  ASSERT_TRUE(ExpectNewFrames(media_expectations));
> }
> 
> // This tests that if we negotiate after calling CreateSender but before we
> // have a track, then set a track later, frames from the newly-set track are
> // received end-to-end.
>-// TODO(deadbeef): Change this test to use AddTransceiver, once that's
>-// implemented.
>-TEST_F(PeerConnectionIntegrationTest,
>+TEST_F(PeerConnectionIntegrationTestPlanB,
>        MediaFlowsAfterEarlyWarmupWithCreateSender) {
>   ASSERT_TRUE(CreatePeerConnectionWrappers());
>   ConnectFakeSignaling();
>@@ -3217,25 +3753,62 @@ TEST_F(PeerConnectionIntegrationTest,
>   EXPECT_TRUE(caller_video_sender->SetTrack(caller()->CreateLocalVideoTrack()));
>   EXPECT_TRUE(callee_audio_sender->SetTrack(callee()->CreateLocalAudioTrack()));
>   EXPECT_TRUE(callee_video_sender->SetTrack(callee()->CreateLocalVideoTrack()));
>-  ExpectNewFramesReceivedWithWait(
>-      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
>-      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
>-      kMaxWaitForFramesMs);
>+  MediaExpectations media_expectations;
>+  media_expectations.ExpectBidirectionalAudioAndVideo();
>+  ASSERT_TRUE(ExpectNewFrames(media_expectations));
>+}
>+
>+// This tests that if we negotiate after calling AddTransceiver but before we
>+// have a track, then set a track later, frames from the newly-set tracks are
>+// received end-to-end.
>+TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
>+       MediaFlowsAfterEarlyWarmupWithAddTransceiver) {
>+  ASSERT_TRUE(CreatePeerConnectionWrappers());
>+  ConnectFakeSignaling();
>+  auto audio_result = caller()->pc()->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
>+  ASSERT_EQ(RTCErrorType::NONE, audio_result.error().type());
>+  auto caller_audio_sender = audio_result.MoveValue()->sender();
>+  auto video_result = caller()->pc()->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
>+  ASSERT_EQ(RTCErrorType::NONE, video_result.error().type());
>+  auto caller_video_sender = video_result.MoveValue()->sender();
>+  callee()->SetRemoteOfferHandler([this] {
>+    ASSERT_EQ(2u, callee()->pc()->GetTransceivers().size());
>+    callee()->pc()->GetTransceivers()[0]->SetDirection(
>+        RtpTransceiverDirection::kSendRecv);
>+    callee()->pc()->GetTransceivers()[1]->SetDirection(
>+        RtpTransceiverDirection::kSendRecv);
>+  });
>+  caller()->CreateAndSetAndSignalOffer();
>+  ASSERT_TRUE_WAIT(SignalingStateStable(), kMaxWaitForActivationMs);
>+  // Wait for ICE to complete, without any tracks being set.
>+  EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
>+                 caller()->ice_connection_state(), kMaxWaitForFramesMs);
>+  EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
>+                 callee()->ice_connection_state(), kMaxWaitForFramesMs);
>+  // Now set the tracks, and expect frames to immediately start flowing.
>+  auto callee_audio_sender = callee()->pc()->GetSenders()[0];
>+  auto callee_video_sender = callee()->pc()->GetSenders()[1];
>+  ASSERT_TRUE(caller_audio_sender->SetTrack(caller()->CreateLocalAudioTrack()));
>+  ASSERT_TRUE(caller_video_sender->SetTrack(caller()->CreateLocalVideoTrack()));
>+  ASSERT_TRUE(callee_audio_sender->SetTrack(callee()->CreateLocalAudioTrack()));
>+  ASSERT_TRUE(callee_video_sender->SetTrack(callee()->CreateLocalVideoTrack()));
>+  MediaExpectations media_expectations;
>+  media_expectations.ExpectBidirectionalAudioAndVideo();
>+  ASSERT_TRUE(ExpectNewFrames(media_expectations));
> }
> 
> // This test verifies that a remote video track can be added via AddStream,
> // and sent end-to-end. For this particular test, it's simply echoed back
> // from the caller to the callee, rather than being forwarded to a third
> // PeerConnection.
>-TEST_F(PeerConnectionIntegrationTest, CanSendRemoteVideoTrack) {
>+TEST_F(PeerConnectionIntegrationTestPlanB, CanSendRemoteVideoTrack) {
>   ASSERT_TRUE(CreatePeerConnectionWrappers());
>   ConnectFakeSignaling();
>   // Just send a video track from the caller.
>-  caller()->AddMediaStreamFromTracks(nullptr,
>-                                     caller()->CreateLocalVideoTrack());
>+  caller()->AddVideoTrack();
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kMaxWaitForActivationMs);
>-  ASSERT_EQ(1, callee()->remote_streams()->count());
>+  ASSERT_EQ(1U, callee()->remote_streams()->count());
> 
>   // Echo the stream back, and do a new offer/anwer (initiated by callee this
>   // time).
>@@ -3243,9 +3816,9 @@ TEST_F(PeerConnectionIntegrationTest, CanSendRemoteVideoTrack) {
>   callee()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kMaxWaitForActivationMs);
> 
>-  int expected_caller_received_video_frames = kDefaultExpectedVideoFrameCount;
>-  ExpectNewFramesReceivedWithWait(0, expected_caller_received_video_frames, 0,
>-                                  0, kMaxWaitForFramesMs);
>+  MediaExpectations media_expectations;
>+  media_expectations.ExpectBidirectionalVideo();
>+  ASSERT_TRUE(ExpectNewFrames(media_expectations));
> }
> 
> // Test that we achieve the expected end-to-end connection time, using a
>@@ -3261,12 +3834,12 @@ TEST_F(PeerConnectionIntegrationTest, CanSendRemoteVideoTrack) {
> // 2. 9 media hops: Rest of the DTLS handshake. 3 hops in each direction when
> //                  using TURN<->TURN pair, and DTLS exchange is 4 packets,
> //                  the first of which should have arrived before the answer.
>-TEST_F(PeerConnectionIntegrationTest, EndToEndConnectionTimeWithTurnTurnPair) {
>+TEST_P(PeerConnectionIntegrationTest, EndToEndConnectionTimeWithTurnTurnPair) {
>   rtc::ScopedFakeClock fake_clock;
>   // Some things use a time of "0" as a special value, so we need to start out
>   // the fake clock at a nonzero time.
>   // TODO(deadbeef): Fix this.
>-  fake_clock.AdvanceTime(rtc::TimeDelta::FromSeconds(1));
>+  fake_clock.AdvanceTime(webrtc::TimeDelta::seconds(1));
> 
>   static constexpr int media_hop_delay_ms = 50;
>   static constexpr int signaling_trip_delay_ms = 500;
>@@ -3288,17 +3861,19 @@ TEST_F(PeerConnectionIntegrationTest, EndToEndConnectionTimeWithTurnTurnPair) {
>                                                                  3478};
>   static const rtc::SocketAddress turn_server_2_external_address{"99.99.99.1",
>                                                                  0};
>-  cricket::TestTurnServer turn_server_1(network_thread(),
>-                                        turn_server_1_internal_address,
>-                                        turn_server_1_external_address);
>-  cricket::TestTurnServer turn_server_2(network_thread(),
>-                                        turn_server_2_internal_address,
>-                                        turn_server_2_external_address);
>+  cricket::TestTurnServer* turn_server_1 = CreateTurnServer(
>+      turn_server_1_internal_address, turn_server_1_external_address);
> 
>+  cricket::TestTurnServer* turn_server_2 = CreateTurnServer(
>+      turn_server_2_internal_address, turn_server_2_external_address);
>   // Bypass permission check on received packets so media can be sent before
>   // the candidate is signaled.
>-  turn_server_1.set_enable_permission_checks(false);
>-  turn_server_2.set_enable_permission_checks(false);
>+  network_thread()->Invoke<void>(RTC_FROM_HERE, [turn_server_1] {
>+    turn_server_1->set_enable_permission_checks(false);
>+  });
>+  network_thread()->Invoke<void>(RTC_FROM_HERE, [turn_server_2] {
>+    turn_server_2->set_enable_permission_checks(false);
>+  });
> 
>   PeerConnectionInterface::RTCConfiguration client_1_config;
>   webrtc::PeerConnectionInterface::IceServer ice_server_1;
>@@ -3335,17 +3910,17 @@ TEST_F(PeerConnectionIntegrationTest, EndToEndConnectionTimeWithTurnTurnPair) {
>   caller()->CreateAndSetAndSignalOffer();
>   EXPECT_TRUE_SIMULATED_WAIT(DtlsConnected(), total_connection_time_ms,
>                              fake_clock);
>-  // Need to free the clients here since they're using things we created on
>-  // the stack.
>-  delete SetCallerPcWrapperAndReturnCurrent(nullptr);
>-  delete SetCalleePcWrapperAndReturnCurrent(nullptr);
>+  // Closing the PeerConnections destroys the ports before the ScopedFakeClock.
>+  // If this is not done a DCHECK can be hit in ports.cc, because a large
>+  // negative number is calculated for the rtt due to the global clock changing.
>+  caller()->pc()->Close();
>+  callee()->pc()->Close();
> }
> 
> // Verify that a TurnCustomizer passed in through RTCConfiguration
> // is actually used by the underlying TURN candidate pair.
> // Note that turnport_unittest.cc contains more detailed, lower-level tests.
>-TEST_F(PeerConnectionIntegrationTest,           \
>-       TurnCustomizerUsedForTurnConnections) {
>+TEST_P(PeerConnectionIntegrationTest, TurnCustomizerUsedForTurnConnections) {
>   static const rtc::SocketAddress turn_server_1_internal_address{"88.88.88.0",
>                                                                  3478};
>   static const rtc::SocketAddress turn_server_1_external_address{"88.88.88.1",
>@@ -3354,12 +3929,10 @@ TEST_F(PeerConnectionIntegrationTest,           \
>                                                                  3478};
>   static const rtc::SocketAddress turn_server_2_external_address{"99.99.99.1",
>                                                                  0};
>-  cricket::TestTurnServer turn_server_1(network_thread(),
>-                                        turn_server_1_internal_address,
>-                                        turn_server_1_external_address);
>-  cricket::TestTurnServer turn_server_2(network_thread(),
>-                                        turn_server_2_internal_address,
>-                                        turn_server_2_external_address);
>+  CreateTurnServer(turn_server_1_internal_address,
>+                   turn_server_1_external_address);
>+  CreateTurnServer(turn_server_2_internal_address,
>+                   turn_server_2_external_address);
> 
>   PeerConnectionInterface::RTCConfiguration client_1_config;
>   webrtc::PeerConnectionInterface::IceServer ice_server_1;
>@@ -3368,8 +3941,8 @@ TEST_F(PeerConnectionIntegrationTest,           \
>   ice_server_1.password = "test";
>   client_1_config.servers.push_back(ice_server_1);
>   client_1_config.type = webrtc::PeerConnectionInterface::kRelay;
>-  auto customizer1 = rtc::MakeUnique<cricket::TestTurnCustomizer>();
>-  client_1_config.turn_customizer = customizer1.get();
>+  auto* customizer1 = CreateTurnCustomizer();
>+  client_1_config.turn_customizer = customizer1;
> 
>   PeerConnectionInterface::RTCConfiguration client_2_config;
>   webrtc::PeerConnectionInterface::IceServer ice_server_2;
>@@ -3378,11 +3951,117 @@ TEST_F(PeerConnectionIntegrationTest,           \
>   ice_server_2.password = "test";
>   client_2_config.servers.push_back(ice_server_2);
>   client_2_config.type = webrtc::PeerConnectionInterface::kRelay;
>-  auto customizer2 = rtc::MakeUnique<cricket::TestTurnCustomizer>();
>-  client_2_config.turn_customizer = customizer2.get();
>+  auto* customizer2 = CreateTurnCustomizer();
>+  client_2_config.turn_customizer = customizer2;
>+
>+  ASSERT_TRUE(
>+      CreatePeerConnectionWrappersWithConfig(client_1_config, client_2_config));
>+  ConnectFakeSignaling();
>+
>+  // Set "offer to receive audio/video" without adding any tracks, so we just
>+  // set up ICE/DTLS with no media.
>+  PeerConnectionInterface::RTCOfferAnswerOptions options;
>+  options.offer_to_receive_audio = 1;
>+  options.offer_to_receive_video = 1;
>+  caller()->SetOfferAnswerOptions(options);
>+  caller()->CreateAndSetAndSignalOffer();
>+  ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
>+
>+  ExpectTurnCustomizerCountersIncremented(customizer1);
>+  ExpectTurnCustomizerCountersIncremented(customizer2);
>+}
>+
>+// Verifies that you can use TCP instead of UDP to connect to a TURN server and
>+// send media between the caller and the callee.
>+TEST_P(PeerConnectionIntegrationTest, TCPUsedForTurnConnections) {
>+  static const rtc::SocketAddress turn_server_internal_address{"88.88.88.0",
>+                                                               3478};
>+  static const rtc::SocketAddress turn_server_external_address{"88.88.88.1", 0};
>+
>+  // Enable TCP for the fake turn server.
>+  CreateTurnServer(turn_server_internal_address, turn_server_external_address,
>+                   cricket::PROTO_TCP);
>+
>+  webrtc::PeerConnectionInterface::IceServer ice_server;
>+  ice_server.urls.push_back("turn:88.88.88.0:3478?transport=tcp");
>+  ice_server.username = "test";
>+  ice_server.password = "test";
>+
>+  PeerConnectionInterface::RTCConfiguration client_1_config;
>+  client_1_config.servers.push_back(ice_server);
>+  client_1_config.type = webrtc::PeerConnectionInterface::kRelay;
>+
>+  PeerConnectionInterface::RTCConfiguration client_2_config;
>+  client_2_config.servers.push_back(ice_server);
>+  client_2_config.type = webrtc::PeerConnectionInterface::kRelay;
> 
>   ASSERT_TRUE(
>       CreatePeerConnectionWrappersWithConfig(client_1_config, client_2_config));
>+
>+  // Do normal offer/answer and wait for ICE to complete.
>+  ConnectFakeSignaling();
>+  caller()->AddAudioVideoTracks();
>+  callee()->AddAudioVideoTracks();
>+  caller()->CreateAndSetAndSignalOffer();
>+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>+  EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
>+                 callee()->ice_connection_state(), kMaxWaitForFramesMs);
>+
>+  MediaExpectations media_expectations;
>+  media_expectations.ExpectBidirectionalAudioAndVideo();
>+  EXPECT_TRUE(ExpectNewFrames(media_expectations));
>+}
>+
>+// Verify that a SSLCertificateVerifier passed in through
>+// PeerConnectionDependencies is actually used by the underlying SSL
>+// implementation to determine whether a certificate presented by the TURN
>+// server is accepted by the client. Note that openssladapter_unittest.cc
>+// contains more detailed, lower-level tests.
>+TEST_P(PeerConnectionIntegrationTest,
>+       SSLCertificateVerifierUsedForTurnConnections) {
>+  static const rtc::SocketAddress turn_server_internal_address{"88.88.88.0",
>+                                                               3478};
>+  static const rtc::SocketAddress turn_server_external_address{"88.88.88.1", 0};
>+
>+  // Enable TCP-TLS for the fake turn server. We need to pass in 88.88.88.0 so
>+  // that host name verification passes on the fake certificate.
>+  CreateTurnServer(turn_server_internal_address, turn_server_external_address,
>+                   cricket::PROTO_TLS, "88.88.88.0");
>+
>+  webrtc::PeerConnectionInterface::IceServer ice_server;
>+  ice_server.urls.push_back("turns:88.88.88.0:3478?transport=tcp");
>+  ice_server.username = "test";
>+  ice_server.password = "test";
>+
>+  PeerConnectionInterface::RTCConfiguration client_1_config;
>+  client_1_config.servers.push_back(ice_server);
>+  client_1_config.type = webrtc::PeerConnectionInterface::kRelay;
>+
>+  PeerConnectionInterface::RTCConfiguration client_2_config;
>+  client_2_config.servers.push_back(ice_server);
>+  // Setting the type to kRelay forces the connection to go through a TURN
>+  // server.
>+  client_2_config.type = webrtc::PeerConnectionInterface::kRelay;
>+
>+  // Get a copy to the pointer so we can verify calls later.
>+  rtc::TestCertificateVerifier* client_1_cert_verifier =
>+      new rtc::TestCertificateVerifier();
>+  client_1_cert_verifier->verify_certificate_ = true;
>+  rtc::TestCertificateVerifier* client_2_cert_verifier =
>+      new rtc::TestCertificateVerifier();
>+  client_2_cert_verifier->verify_certificate_ = true;
>+
>+  // Create the dependencies with the test certificate verifier.
>+  webrtc::PeerConnectionDependencies client_1_deps(nullptr);
>+  client_1_deps.tls_cert_verifier =
>+      std::unique_ptr<rtc::TestCertificateVerifier>(client_1_cert_verifier);
>+  webrtc::PeerConnectionDependencies client_2_deps(nullptr);
>+  client_2_deps.tls_cert_verifier =
>+      std::unique_ptr<rtc::TestCertificateVerifier>(client_2_cert_verifier);
>+
>+  ASSERT_TRUE(CreatePeerConnectionWrappersWithConfigAndDeps(
>+      client_1_config, std::move(client_1_deps), client_2_config,
>+      std::move(client_2_deps)));
>   ConnectFakeSignaling();
> 
>   // Set "offer to receive audio/video" without adding any tracks, so we just
>@@ -3394,16 +4073,73 @@ TEST_F(PeerConnectionIntegrationTest,           \
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
> 
>-  EXPECT_GT(customizer1->allow_channel_data_cnt_, 0u);
>-  EXPECT_GT(customizer1->modify_cnt_, 0u);
>+  EXPECT_GT(client_1_cert_verifier->call_count_, 0u);
>+  EXPECT_GT(client_2_cert_verifier->call_count_, 0u);
>+}
>+
>+TEST_P(PeerConnectionIntegrationTest,
>+       SSLCertificateVerifierFailureUsedForTurnConnectionsFailsConnection) {
>+  static const rtc::SocketAddress turn_server_internal_address{"88.88.88.0",
>+                                                               3478};
>+  static const rtc::SocketAddress turn_server_external_address{"88.88.88.1", 0};
>+
>+  // Enable TCP-TLS for the fake turn server. We need to pass in 88.88.88.0 so
>+  // that host name verification passes on the fake certificate.
>+  CreateTurnServer(turn_server_internal_address, turn_server_external_address,
>+                   cricket::PROTO_TLS, "88.88.88.0");
>+
>+  webrtc::PeerConnectionInterface::IceServer ice_server;
>+  ice_server.urls.push_back("turns:88.88.88.0:3478?transport=tcp");
>+  ice_server.username = "test";
>+  ice_server.password = "test";
>+
>+  PeerConnectionInterface::RTCConfiguration client_1_config;
>+  client_1_config.servers.push_back(ice_server);
>+  client_1_config.type = webrtc::PeerConnectionInterface::kRelay;
>+
>+  PeerConnectionInterface::RTCConfiguration client_2_config;
>+  client_2_config.servers.push_back(ice_server);
>+  // Setting the type to kRelay forces the connection to go through a TURN
>+  // server.
>+  client_2_config.type = webrtc::PeerConnectionInterface::kRelay;
> 
>-  EXPECT_GT(customizer2->allow_channel_data_cnt_, 0u);
>-  EXPECT_GT(customizer2->modify_cnt_, 0u);
>+  // Get a copy to the pointer so we can verify calls later.
>+  rtc::TestCertificateVerifier* client_1_cert_verifier =
>+      new rtc::TestCertificateVerifier();
>+  client_1_cert_verifier->verify_certificate_ = false;
>+  rtc::TestCertificateVerifier* client_2_cert_verifier =
>+      new rtc::TestCertificateVerifier();
>+  client_2_cert_verifier->verify_certificate_ = false;
>+
>+  // Create the dependencies with the test certificate verifier.
>+  webrtc::PeerConnectionDependencies client_1_deps(nullptr);
>+  client_1_deps.tls_cert_verifier =
>+      std::unique_ptr<rtc::TestCertificateVerifier>(client_1_cert_verifier);
>+  webrtc::PeerConnectionDependencies client_2_deps(nullptr);
>+  client_2_deps.tls_cert_verifier =
>+      std::unique_ptr<rtc::TestCertificateVerifier>(client_2_cert_verifier);
>+
>+  ASSERT_TRUE(CreatePeerConnectionWrappersWithConfigAndDeps(
>+      client_1_config, std::move(client_1_deps), client_2_config,
>+      std::move(client_2_deps)));
>+  ConnectFakeSignaling();
> 
>-  // Need to free the clients here since they're using things we created on
>-  // the stack.
>-  delete SetCallerPcWrapperAndReturnCurrent(nullptr);
>-  delete SetCalleePcWrapperAndReturnCurrent(nullptr);
>+  // Set "offer to receive audio/video" without adding any tracks, so we just
>+  // set up ICE/DTLS with no media.
>+  PeerConnectionInterface::RTCOfferAnswerOptions options;
>+  options.offer_to_receive_audio = 1;
>+  options.offer_to_receive_video = 1;
>+  caller()->SetOfferAnswerOptions(options);
>+  caller()->CreateAndSetAndSignalOffer();
>+  bool wait_res = true;
>+  // TODO(bugs.webrtc.org/9219): When IceConnectionState is implemented
>+  // properly, should be able to just wait for a state of "failed" instead of
>+  // waiting a fixed 10 seconds.
>+  WAIT_(DtlsConnected(), kDefaultTimeout, wait_res);
>+  ASSERT_FALSE(wait_res);
>+
>+  EXPECT_GT(client_1_cert_verifier->call_count_, 0u);
>+  EXPECT_GT(client_2_cert_verifier->call_count_, 0u);
> }
> 
> // Test that audio and video flow end-to-end when codec names don't use the
>@@ -3414,11 +4150,11 @@ TEST_F(PeerConnectionIntegrationTest,           \
> // In the past, this has regressed and caused crashes/black video, due to the
> // fact that code at some layers was doing case-insensitive comparisons and
> // code at other layers was not.
>-TEST_F(PeerConnectionIntegrationTest, CodecNamesAreCaseInsensitive) {
>+TEST_P(PeerConnectionIntegrationTest, CodecNamesAreCaseInsensitive) {
>   ASSERT_TRUE(CreatePeerConnectionWrappers());
>   ConnectFakeSignaling();
>-  caller()->AddAudioVideoMediaStream();
>-  callee()->AddAudioVideoMediaStream();
>+  caller()->AddAudioVideoTracks();
>+  callee()->AddAudioVideoTracks();
> 
>   // Remove all but one audio/video codec (opus and VP8), and change the
>   // casing of the caller's generated offer.
>@@ -3454,20 +4190,21 @@ TEST_F(PeerConnectionIntegrationTest, CodecNamesAreCaseInsensitive) {
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
> 
>   // Verify frames are still received end-to-end.
>-  ExpectNewFramesReceivedWithWait(
>-      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
>-      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
>-      kMaxWaitForFramesMs);
>+  MediaExpectations media_expectations;
>+  media_expectations.ExpectBidirectionalAudioAndVideo();
>+  ASSERT_TRUE(ExpectNewFrames(media_expectations));
> }
> 
>-TEST_F(PeerConnectionIntegrationTest, GetSources) {
>+TEST_P(PeerConnectionIntegrationTest, GetSources) {
>   ASSERT_TRUE(CreatePeerConnectionWrappers());
>   ConnectFakeSignaling();
>-  caller()->AddAudioOnlyMediaStream();
>+  caller()->AddAudioTrack();
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>   // Wait for one audio frame to be received by the callee.
>-  ExpectNewFramesReceivedWithWait(0, 0, 1, 0, kMaxWaitForFramesMs);
>+  MediaExpectations media_expectations;
>+  media_expectations.CalleeExpectsSomeAudio(1);
>+  ASSERT_TRUE(ExpectNewFrames(media_expectations));
>   ASSERT_GT(callee()->pc()->GetReceivers().size(), 0u);
>   auto receiver = callee()->pc()->GetReceivers()[0];
>   ASSERT_EQ(receiver->media_type(), cricket::MEDIA_TYPE_AUDIO);
>@@ -3481,47 +4218,51 @@ TEST_F(PeerConnectionIntegrationTest, GetSources) {
> // Test that if a track is removed and added again with a different stream ID,
> // the new stream ID is successfully communicated in SDP and media continues to
> // flow end-to-end.
>-TEST_F(PeerConnectionIntegrationTest, RemoveAndAddTrackWithNewStreamId) {
>+// TODO(webrtc.bugs.org/8734): This test does not work for Unified Plan because
>+// it will not reuse a transceiver that has already been sending. After creating
>+// a new transceiver it tries to create an offer with two senders of the same
>+// track ids and it fails.
>+TEST_F(PeerConnectionIntegrationTestPlanB, RemoveAndAddTrackWithNewStreamId) {
>   ASSERT_TRUE(CreatePeerConnectionWrappers());
>   ConnectFakeSignaling();
> 
>-  rtc::scoped_refptr<MediaStreamInterface> stream_1 =
>-      caller()->pc_factory()->CreateLocalMediaStream("stream_1");
>-  rtc::scoped_refptr<MediaStreamInterface> stream_2 =
>-      caller()->pc_factory()->CreateLocalMediaStream("stream_2");
>-
>   // Add track using stream 1, do offer/answer.
>   rtc::scoped_refptr<webrtc::AudioTrackInterface> track =
>       caller()->CreateLocalAudioTrack();
>   rtc::scoped_refptr<webrtc::RtpSenderInterface> sender =
>-      caller()->pc()->AddTrack(track, {stream_1.get()});
>+      caller()->AddTrack(track, {"stream_1"});
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>-  // Wait for one audio frame to be received by the callee.
>-  ExpectNewFramesReceivedWithWait(0, 0, 1, 0, kMaxWaitForFramesMs);
>-
>+  {
>+    MediaExpectations media_expectations;
>+    media_expectations.CalleeExpectsSomeAudio(1);
>+    ASSERT_TRUE(ExpectNewFrames(media_expectations));
>+  }
>   // Remove the sender, and create a new one with the new stream.
>   caller()->pc()->RemoveTrack(sender);
>-  sender = caller()->pc()->AddTrack(track, {stream_2.get()});
>+  sender = caller()->AddTrack(track, {"stream_2"});
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>   // Wait for additional audio frames to be received by the callee.
>-  ExpectNewFramesReceivedWithWait(0, 0, kDefaultExpectedAudioFrameCount, 0,
>-                                  kMaxWaitForFramesMs);
>+  {
>+    MediaExpectations media_expectations;
>+    media_expectations.CalleeExpectsSomeAudio();
>+    ASSERT_TRUE(ExpectNewFrames(media_expectations));
>+  }
> }
> 
>-TEST_F(PeerConnectionIntegrationTest, RtcEventLogOutputWriteCalled) {
>+TEST_P(PeerConnectionIntegrationTest, RtcEventLogOutputWriteCalled) {
>   ASSERT_TRUE(CreatePeerConnectionWrappers());
>   ConnectFakeSignaling();
> 
>-  auto output = rtc::MakeUnique<testing::NiceMock<MockRtcEventLogOutput>>();
>+  auto output = absl::make_unique<testing::NiceMock<MockRtcEventLogOutput>>();
>   ON_CALL(*output, IsActive()).WillByDefault(testing::Return(true));
>   ON_CALL(*output, Write(::testing::_)).WillByDefault(testing::Return(true));
>   EXPECT_CALL(*output, Write(::testing::_)).Times(::testing::AtLeast(1));
>   EXPECT_TRUE(caller()->pc()->StartRtcEventLog(
>       std::move(output), webrtc::RtcEventLog::kImmediateOutput));
> 
>-  caller()->AddAudioVideoMediaStream();
>+  caller()->AddAudioVideoTracks();
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
> }
>@@ -3529,15 +4270,15 @@ TEST_F(PeerConnectionIntegrationTest, RtcEventLogOutputWriteCalled) {
> // Test that if candidates are only signaled by applying full session
> // descriptions (instead of using AddIceCandidate), the peers can connect to
> // each other and exchange media.
>-TEST_F(PeerConnectionIntegrationTest, MediaFlowsWhenCandidatesSetOnlyInSdp) {
>+TEST_P(PeerConnectionIntegrationTest, MediaFlowsWhenCandidatesSetOnlyInSdp) {
>   ASSERT_TRUE(CreatePeerConnectionWrappers());
>   // Each side will signal the session descriptions but not candidates.
>   ConnectFakeSignalingForSdpOnly();
> 
>   // Add audio video track and exchange the initial offer/answer with media
>   // information only. This will start ICE gathering on each side.
>-  caller()->AddAudioVideoMediaStream();
>-  callee()->AddAudioVideoMediaStream();
>+  caller()->AddAudioVideoTracks();
>+  callee()->AddAudioVideoTracks();
>   caller()->CreateAndSetAndSignalOffer();
> 
>   // Wait for all candidates to be gathered on both the caller and callee.
>@@ -3552,24 +4293,23 @@ TEST_F(PeerConnectionIntegrationTest, MediaFlowsWhenCandidatesSetOnlyInSdp) {
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
> 
>   // Ensure that media flows in both directions.
>-  ExpectNewFramesReceivedWithWait(
>-      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
>-      kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
>-      kMaxWaitForFramesMs);
>+  MediaExpectations media_expectations;
>+  media_expectations.ExpectBidirectionalAudioAndVideo();
>+  ASSERT_TRUE(ExpectNewFrames(media_expectations));
> }
> 
> // Test that SetAudioPlayout can be used to disable audio playout from the
> // start, then later enable it. This may be useful, for example, if the caller
> // needs to play a local ringtone until some event occurs, after which it
> // switches to playing the received audio.
>-TEST_F(PeerConnectionIntegrationTest, DisableAndEnableAudioPlayout) {
>+TEST_P(PeerConnectionIntegrationTest, DisableAndEnableAudioPlayout) {
>   ASSERT_TRUE(CreatePeerConnectionWrappers());
>   ConnectFakeSignaling();
> 
>   // Set up audio-only call where audio playout is disabled on caller's side.
>   caller()->pc()->SetAudioPlayout(false);
>-  caller()->AddAudioOnlyMediaStream();
>-  callee()->AddAudioOnlyMediaStream();
>+  caller()->AddAudioTrack();
>+  callee()->AddAudioTrack();
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
> 
>@@ -3584,9 +4324,9 @@ TEST_F(PeerConnectionIntegrationTest, DisableAndEnableAudioPlayout) {
> 
>   // Enable playout again, and ensure audio starts flowing.
>   caller()->pc()->SetAudioPlayout(true);
>-  ExpectNewFramesReceivedWithWait(kDefaultExpectedAudioFrameCount, 0,
>-                                  kDefaultExpectedAudioFrameCount, 0,
>-                                  kMaxWaitForFramesMs);
>+  MediaExpectations media_expectations;
>+  media_expectations.ExpectBidirectionalAudio();
>+  ASSERT_TRUE(ExpectNewFrames(media_expectations));
> }
> 
> double GetAudioEnergyStat(PeerConnectionWrapper* pc) {
>@@ -3610,15 +4350,15 @@ double GetAudioEnergyStat(PeerConnectionWrapper* pc) {
> 
> // Test that if audio playout is disabled via the SetAudioPlayout() method, then
> // incoming audio is still processed and statistics are generated.
>-TEST_F(PeerConnectionIntegrationTest,
>+TEST_P(PeerConnectionIntegrationTest,
>        DisableAudioPlayoutStillGeneratesAudioStats) {
>   ASSERT_TRUE(CreatePeerConnectionWrappers());
>   ConnectFakeSignaling();
> 
>   // Set up audio-only call where playout is disabled but audio-processing is
>   // still active.
>-  caller()->AddAudioOnlyMediaStream();
>-  callee()->AddAudioOnlyMediaStream();
>+  caller()->AddAudioTrack();
>+  callee()->AddAudioTrack();
>   caller()->pc()->SetAudioPlayout(false);
> 
>   caller()->CreateAndSetAndSignalOffer();
>@@ -3632,14 +4372,14 @@ TEST_F(PeerConnectionIntegrationTest,
> // start, then later enable it. This may be useful, for example, if the caller
> // wants to ensure that no audio resources are active before a certain state
> // is reached.
>-TEST_F(PeerConnectionIntegrationTest, DisableAndEnableAudioRecording) {
>+TEST_P(PeerConnectionIntegrationTest, DisableAndEnableAudioRecording) {
>   ASSERT_TRUE(CreatePeerConnectionWrappers());
>   ConnectFakeSignaling();
> 
>   // Set up audio-only call where audio recording is disabled on caller's side.
>   caller()->pc()->SetAudioRecording(false);
>-  caller()->AddAudioOnlyMediaStream();
>-  callee()->AddAudioOnlyMediaStream();
>+  caller()->AddAudioTrack();
>+  callee()->AddAudioTrack();
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
> 
>@@ -3654,26 +4394,26 @@ TEST_F(PeerConnectionIntegrationTest, DisableAndEnableAudioRecording) {
> 
>   // Enable audio recording again, and ensure audio starts flowing.
>   caller()->pc()->SetAudioRecording(true);
>-  ExpectNewFramesReceivedWithWait(kDefaultExpectedAudioFrameCount, 0,
>-                                  kDefaultExpectedAudioFrameCount, 0,
>-                                  kMaxWaitForFramesMs);
>+  MediaExpectations media_expectations;
>+  media_expectations.ExpectBidirectionalAudio();
>+  ASSERT_TRUE(ExpectNewFrames(media_expectations));
> }
> 
> // Test that after closing PeerConnections, they stop sending any packets (ICE,
> // DTLS, RTP...).
>-TEST_F(PeerConnectionIntegrationTest, ClosingConnectionStopsPacketFlow) {
>+TEST_P(PeerConnectionIntegrationTest, ClosingConnectionStopsPacketFlow) {
>   // Set up audio/video/data, wait for some frames to be received.
>   ASSERT_TRUE(CreatePeerConnectionWrappers());
>   ConnectFakeSignaling();
>-  caller()->AddAudioVideoMediaStream();
>+  caller()->AddAudioVideoTracks();
> #ifdef HAVE_SCTP
>   caller()->CreateDataChannel();
> #endif
>   caller()->CreateAndSetAndSignalOffer();
>   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>-  ExpectNewFramesReceivedWithWait(0, 0, kDefaultExpectedAudioFrameCount,
>-                                  kDefaultExpectedAudioFrameCount,
>-                                  kMaxWaitForFramesMs);
>+  MediaExpectations media_expectations;
>+  media_expectations.CalleeExpectsSomeAudioAndVideo();
>+  ASSERT_TRUE(ExpectNewFrames(media_expectations));
>   // Close PeerConnections.
>   caller()->pc()->Close();
>   callee()->pc()->Close();
>@@ -3684,6 +4424,211 @@ TEST_F(PeerConnectionIntegrationTest, ClosingConnectionStopsPacketFlow) {
>   EXPECT_EQ(sent_packets_a, sent_packets_b);
> }
> 
>+// Test that transport stats are generated by the RTCStatsCollector for a
>+// connection that only involves data channels. This is a regression test for
>+// crbug.com/826972.
>+#ifdef HAVE_SCTP
>+TEST_P(PeerConnectionIntegrationTest,
>+       TransportStatsReportedForDataChannelOnlyConnection) {
>+  ASSERT_TRUE(CreatePeerConnectionWrappers());
>+  ConnectFakeSignaling();
>+  caller()->CreateDataChannel();
>+
>+  caller()->CreateAndSetAndSignalOffer();
>+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>+  ASSERT_TRUE_WAIT(callee()->data_channel(), kDefaultTimeout);
>+
>+  auto caller_report = caller()->NewGetStats();
>+  EXPECT_EQ(1u, caller_report->GetStatsOfType<RTCTransportStats>().size());
>+  auto callee_report = callee()->NewGetStats();
>+  EXPECT_EQ(1u, callee_report->GetStatsOfType<RTCTransportStats>().size());
>+}
>+#endif  // HAVE_SCTP
>+
>+TEST_P(PeerConnectionIntegrationTest,
>+       IceEventsGeneratedAndLoggedInRtcEventLog) {
>+  ASSERT_TRUE(CreatePeerConnectionWrappersWithFakeRtcEventLog());
>+  ConnectFakeSignaling();
>+  PeerConnectionInterface::RTCOfferAnswerOptions options;
>+  options.offer_to_receive_audio = 1;
>+  caller()->SetOfferAnswerOptions(options);
>+  caller()->CreateAndSetAndSignalOffer();
>+  ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
>+  ASSERT_NE(nullptr, caller()->event_log_factory());
>+  ASSERT_NE(nullptr, callee()->event_log_factory());
>+  webrtc::FakeRtcEventLog* caller_event_log =
>+      static_cast<webrtc::FakeRtcEventLog*>(
>+          caller()->event_log_factory()->last_log_created());
>+  webrtc::FakeRtcEventLog* callee_event_log =
>+      static_cast<webrtc::FakeRtcEventLog*>(
>+          callee()->event_log_factory()->last_log_created());
>+  ASSERT_NE(nullptr, caller_event_log);
>+  ASSERT_NE(nullptr, callee_event_log);
>+  int caller_ice_config_count = caller_event_log->GetEventCount(
>+      webrtc::RtcEvent::Type::IceCandidatePairConfig);
>+  int caller_ice_event_count = caller_event_log->GetEventCount(
>+      webrtc::RtcEvent::Type::IceCandidatePairEvent);
>+  int callee_ice_config_count = callee_event_log->GetEventCount(
>+      webrtc::RtcEvent::Type::IceCandidatePairConfig);
>+  int callee_ice_event_count = callee_event_log->GetEventCount(
>+      webrtc::RtcEvent::Type::IceCandidatePairEvent);
>+  EXPECT_LT(0, caller_ice_config_count);
>+  EXPECT_LT(0, caller_ice_event_count);
>+  EXPECT_LT(0, callee_ice_config_count);
>+  EXPECT_LT(0, callee_ice_event_count);
>+}
>+
>+INSTANTIATE_TEST_CASE_P(PeerConnectionIntegrationTest,
>+                        PeerConnectionIntegrationTest,
>+                        Values(SdpSemantics::kPlanB,
>+                               SdpSemantics::kUnifiedPlan));
>+
>+// Tests that verify interoperability between Plan B and Unified Plan
>+// PeerConnections.
>+class PeerConnectionIntegrationInteropTest
>+    : public PeerConnectionIntegrationBaseTest,
>+      public ::testing::WithParamInterface<
>+          std::tuple<SdpSemantics, SdpSemantics>> {
>+ protected:
>+  // Setting the SdpSemantics for the base test to kDefault does not matter
>+  // because we specify not to use the test semantics when creating
>+  // PeerConnectionWrappers.
>+  PeerConnectionIntegrationInteropTest()
>+      : PeerConnectionIntegrationBaseTest(SdpSemantics::kPlanB),
>+        caller_semantics_(std::get<0>(GetParam())),
>+        callee_semantics_(std::get<1>(GetParam())) {}
>+
>+  bool CreatePeerConnectionWrappersWithSemantics() {
>+    return CreatePeerConnectionWrappersWithSdpSemantics(caller_semantics_,
>+                                                        callee_semantics_);
>+  }
>+
>+  const SdpSemantics caller_semantics_;
>+  const SdpSemantics callee_semantics_;
>+};
>+
>+TEST_P(PeerConnectionIntegrationInteropTest, NoMediaLocalToNoMediaRemote) {
>+  ASSERT_TRUE(CreatePeerConnectionWrappersWithSemantics());
>+  ConnectFakeSignaling();
>+
>+  caller()->CreateAndSetAndSignalOffer();
>+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>+}
>+
>+TEST_P(PeerConnectionIntegrationInteropTest, OneAudioLocalToNoMediaRemote) {
>+  ASSERT_TRUE(CreatePeerConnectionWrappersWithSemantics());
>+  ConnectFakeSignaling();
>+  auto audio_sender = caller()->AddAudioTrack();
>+
>+  caller()->CreateAndSetAndSignalOffer();
>+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>+
>+  // Verify that one audio receiver has been created on the remote and that it
>+  // has the same track ID as the sending track.
>+  auto receivers = callee()->pc()->GetReceivers();
>+  ASSERT_EQ(1u, receivers.size());
>+  EXPECT_EQ(cricket::MEDIA_TYPE_AUDIO, receivers[0]->media_type());
>+  EXPECT_EQ(receivers[0]->track()->id(), audio_sender->track()->id());
>+
>+  MediaExpectations media_expectations;
>+  media_expectations.CalleeExpectsSomeAudio();
>+  ASSERT_TRUE(ExpectNewFrames(media_expectations));
>+}
>+
>+TEST_P(PeerConnectionIntegrationInteropTest, OneAudioOneVideoToNoMediaRemote) {
>+  ASSERT_TRUE(CreatePeerConnectionWrappersWithSemantics());
>+  ConnectFakeSignaling();
>+  auto video_sender = caller()->AddVideoTrack();
>+  auto audio_sender = caller()->AddAudioTrack();
>+
>+  caller()->CreateAndSetAndSignalOffer();
>+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>+
>+  // Verify that one audio and one video receiver have been created on the
>+  // remote and that they have the same track IDs as the sending tracks.
>+  auto audio_receivers =
>+      callee()->GetReceiversOfType(cricket::MEDIA_TYPE_AUDIO);
>+  ASSERT_EQ(1u, audio_receivers.size());
>+  EXPECT_EQ(audio_receivers[0]->track()->id(), audio_sender->track()->id());
>+  auto video_receivers =
>+      callee()->GetReceiversOfType(cricket::MEDIA_TYPE_VIDEO);
>+  ASSERT_EQ(1u, video_receivers.size());
>+  EXPECT_EQ(video_receivers[0]->track()->id(), video_sender->track()->id());
>+
>+  MediaExpectations media_expectations;
>+  media_expectations.CalleeExpectsSomeAudioAndVideo();
>+  ASSERT_TRUE(ExpectNewFrames(media_expectations));
>+}
>+
>+TEST_P(PeerConnectionIntegrationInteropTest,
>+       OneAudioOneVideoLocalToOneAudioOneVideoRemote) {
>+  ASSERT_TRUE(CreatePeerConnectionWrappersWithSemantics());
>+  ConnectFakeSignaling();
>+  caller()->AddAudioVideoTracks();
>+  callee()->AddAudioVideoTracks();
>+
>+  caller()->CreateAndSetAndSignalOffer();
>+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>+
>+  MediaExpectations media_expectations;
>+  media_expectations.ExpectBidirectionalAudioAndVideo();
>+  ASSERT_TRUE(ExpectNewFrames(media_expectations));
>+}
>+
>+TEST_P(PeerConnectionIntegrationInteropTest,
>+       ReverseRolesOneAudioLocalToOneVideoRemote) {
>+  ASSERT_TRUE(CreatePeerConnectionWrappersWithSemantics());
>+  ConnectFakeSignaling();
>+  caller()->AddAudioTrack();
>+  callee()->AddVideoTrack();
>+
>+  caller()->CreateAndSetAndSignalOffer();
>+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>+
>+  // Verify that only the audio track has been negotiated.
>+  EXPECT_EQ(0u, caller()->GetReceiversOfType(cricket::MEDIA_TYPE_VIDEO).size());
>+  // Might also check that the callee's NegotiationNeeded flag is set.
>+
>+  // Reverse roles.
>+  callee()->CreateAndSetAndSignalOffer();
>+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>+
>+  MediaExpectations media_expectations;
>+  media_expectations.CallerExpectsSomeVideo();
>+  media_expectations.CalleeExpectsSomeAudio();
>+  ASSERT_TRUE(ExpectNewFrames(media_expectations));
>+}
>+
>+INSTANTIATE_TEST_CASE_P(
>+    PeerConnectionIntegrationTest,
>+    PeerConnectionIntegrationInteropTest,
>+    Values(std::make_tuple(SdpSemantics::kPlanB, SdpSemantics::kUnifiedPlan),
>+           std::make_tuple(SdpSemantics::kUnifiedPlan, SdpSemantics::kPlanB)));
>+
>+// Test that if the Unified Plan side offers two video tracks then the Plan B
>+// side will only see the first one and ignore the second.
>+TEST_F(PeerConnectionIntegrationTestPlanB, TwoVideoUnifiedPlanToNoMediaPlanB) {
>+  ASSERT_TRUE(CreatePeerConnectionWrappersWithSdpSemantics(
>+      SdpSemantics::kUnifiedPlan, SdpSemantics::kPlanB));
>+  ConnectFakeSignaling();
>+  auto first_sender = caller()->AddVideoTrack();
>+  caller()->AddVideoTrack();
>+
>+  caller()->CreateAndSetAndSignalOffer();
>+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
>+
>+  // Verify that there is only one receiver and it corresponds to the first
>+  // added track.
>+  auto receivers = callee()->pc()->GetReceivers();
>+  ASSERT_EQ(1u, receivers.size());
>+  EXPECT_TRUE(receivers[0]->track()->enabled());
>+  EXPECT_EQ(first_sender->track()->id(), receivers[0]->track()->id());
>+
>+  MediaExpectations media_expectations;
>+  media_expectations.CalleeExpectsSomeVideo();
>+  ASSERT_TRUE(ExpectNewFrames(media_expectations));
>+}
>+
> }  // namespace
> 
> #endif  // if !defined(THREAD_SANITIZER)
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_jsep_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_jsep_unittest.cc
>index 0ba96e1a3b9..7de0a3ff14a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_jsep_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_jsep_unittest.cc
>@@ -10,6 +10,8 @@
> 
> #include "api/audio_codecs/builtin_audio_decoder_factory.h"
> #include "api/audio_codecs/builtin_audio_encoder_factory.h"
>+#include "api/video_codecs/builtin_video_decoder_factory.h"
>+#include "api/video_codecs/builtin_video_encoder_factory.h"
> #include "media/engine/webrtcmediaengine.h"
> #include "modules/audio_processing/include/audio_processing.h"
> #include "pc/mediasession.h"
>@@ -19,10 +21,10 @@
> #ifdef WEBRTC_ANDROID
> #include "pc/test/androidtestinitializer.h"
> #endif
>+#include "absl/memory/memory.h"
> #include "pc/test/fakeaudiocapturemodule.h"
> #include "pc/test/fakesctptransport.h"
> #include "rtc_base/gunit.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/virtualsocketserver.h"
> #include "test/gmock.h"
> 
>@@ -44,24 +46,24 @@ using ::testing::UnorderedElementsAre;
> class PeerConnectionFactoryForJsepTest : public PeerConnectionFactory {
>  public:
>   PeerConnectionFactoryForJsepTest()
>-      : PeerConnectionFactory(
>-            rtc::Thread::Current(),
>-            rtc::Thread::Current(),
>-            rtc::Thread::Current(),
>-            rtc::WrapUnique(cricket::WebRtcMediaEngineFactory::Create(
>-                FakeAudioCaptureModule::Create(),
>-                CreateBuiltinAudioEncoderFactory(),
>-                CreateBuiltinAudioDecoderFactory(),
>-                nullptr,
>-                nullptr,
>-                nullptr,
>-                AudioProcessingBuilder().Create())),
>-            CreateCallFactory(),
>-            nullptr) {}
>+      : PeerConnectionFactory(rtc::Thread::Current(),
>+                              rtc::Thread::Current(),
>+                              rtc::Thread::Current(),
>+                              cricket::WebRtcMediaEngineFactory::Create(
>+                                  rtc::scoped_refptr<AudioDeviceModule>(
>+                                      FakeAudioCaptureModule::Create()),
>+                                  CreateBuiltinAudioEncoderFactory(),
>+                                  CreateBuiltinAudioDecoderFactory(),
>+                                  CreateBuiltinVideoEncoderFactory(),
>+                                  CreateBuiltinVideoDecoderFactory(),
>+                                  nullptr,
>+                                  AudioProcessingBuilder().Create()),
>+                              CreateCallFactory(),
>+                              nullptr) {}
> 
>   std::unique_ptr<cricket::SctpTransportInternalFactory>
>   CreateSctpTransportInternalFactory() {
>-    return rtc::MakeUnique<FakeSctpTransportFactory>();
>+    return absl::make_unique<FakeSctpTransportFactory>();
>   }
> };
> 
>@@ -86,15 +88,15 @@ class PeerConnectionJsepTest : public ::testing::Test {
>     rtc::scoped_refptr<PeerConnectionFactory> pc_factory(
>         new rtc::RefCountedObject<PeerConnectionFactoryForJsepTest>());
>     RTC_CHECK(pc_factory->Initialize());
>-    auto observer = rtc::MakeUnique<MockPeerConnectionObserver>();
>+    auto observer = absl::make_unique<MockPeerConnectionObserver>();
>     auto pc = pc_factory->CreatePeerConnection(config, nullptr, nullptr,
>                                                observer.get());
>     if (!pc) {
>       return nullptr;
>     }
> 
>-    return rtc::MakeUnique<PeerConnectionWrapper>(pc_factory, pc,
>-                                                  std::move(observer));
>+    return absl::make_unique<PeerConnectionWrapper>(pc_factory, pc,
>+                                                    std::move(observer));
>   }
> 
>   std::unique_ptr<rtc::VirtualSocketServer> vss_;
>@@ -243,18 +245,6 @@ TEST_F(PeerConnectionJsepTest, SetLocalOfferSetsTransceiverMid) {
>   EXPECT_EQ(video_mid, video_transceiver->mid());
> }
> 
>-TEST_F(PeerConnectionJsepTest, SetLocalOfferFailsWithNoTrackInMediaSection) {
>-  auto caller = CreatePeerConnection();
>-  caller->AddAudioTrack("audio");
>-
>-  auto offer = caller->CreateOffer();
>-  auto& contents = offer->description()->contents();
>-  ASSERT_EQ(1u, contents.size());
>-  contents[0].description->mutable_streams().clear();
>-
>-  ASSERT_FALSE(caller->SetLocalDescription(std::move(offer)));
>-}
>-
> // Tests for JSEP SetRemoteDescription with a remote offer.
> 
> // Test that setting a remote offer with sendrecv audio and video creates two
>@@ -269,14 +259,16 @@ TEST_F(PeerConnectionJsepTest, SetRemoteOfferCreatesTransceivers) {
> 
>   auto transceivers = callee->pc()->GetTransceivers();
>   ASSERT_EQ(2u, transceivers.size());
>-  EXPECT_EQ(cricket::MEDIA_TYPE_AUDIO,
>-            transceivers[0]->receiver()->media_type());
>+
>+  EXPECT_EQ(cricket::MEDIA_TYPE_AUDIO, transceivers[0]->media_type());
>   EXPECT_EQ(caller_audio->mid(), transceivers[0]->mid());
>   EXPECT_EQ(RtpTransceiverDirection::kRecvOnly, transceivers[0]->direction());
>-  EXPECT_EQ(cricket::MEDIA_TYPE_VIDEO,
>-            transceivers[1]->receiver()->media_type());
>+  EXPECT_EQ(0u, transceivers[0]->sender()->stream_ids().size());
>+
>+  EXPECT_EQ(cricket::MEDIA_TYPE_VIDEO, transceivers[1]->media_type());
>   EXPECT_EQ(caller_video->mid(), transceivers[1]->mid());
>   EXPECT_EQ(RtpTransceiverDirection::kRecvOnly, transceivers[1]->direction());
>+  EXPECT_EQ(0u, transceivers[1]->sender()->stream_ids().size());
> }
> 
> // Test that setting a remote offer with an audio track will reuse the
>@@ -316,7 +308,7 @@ TEST_F(PeerConnectionJsepTest,
> 
>   auto transceivers = callee->pc()->GetTransceivers();
>   ASSERT_EQ(2u, transceivers.size());
>-  EXPECT_EQ(rtc::nullopt, transceivers[0]->mid());
>+  EXPECT_EQ(absl::nullopt, transceivers[0]->mid());
>   EXPECT_EQ(caller_audio->mid(), transceivers[1]->mid());
> }
> 
>@@ -334,7 +326,7 @@ TEST_F(PeerConnectionJsepTest,
> 
>   auto transceivers = callee->pc()->GetTransceivers();
>   ASSERT_EQ(2u, transceivers.size());
>-  EXPECT_EQ(rtc::nullopt, transceivers[0]->mid());
>+  EXPECT_EQ(absl::nullopt, transceivers[0]->mid());
>   EXPECT_EQ(caller->pc()->GetTransceivers()[0]->mid(), transceivers[1]->mid());
>   EXPECT_EQ(MediaStreamTrackInterface::kAudioKind,
>             transceivers[1]->receiver()->track()->kind());
>@@ -353,7 +345,7 @@ TEST_F(PeerConnectionJsepTest,
> 
>   auto transceivers = callee->pc()->GetTransceivers();
>   ASSERT_EQ(2u, transceivers.size());
>-  EXPECT_EQ(rtc::nullopt, transceivers[0]->mid());
>+  EXPECT_EQ(absl::nullopt, transceivers[0]->mid());
>   EXPECT_EQ(caller->pc()->GetTransceivers()[0]->mid(), transceivers[1]->mid());
>   EXPECT_EQ(MediaStreamTrackInterface::kAudioKind,
>             transceivers[1]->receiver()->track()->kind());
>@@ -372,7 +364,7 @@ TEST_F(PeerConnectionJsepTest, SetRemoteOfferDoesNotReuseStoppedTransceiver) {
> 
>   auto transceivers = callee->pc()->GetTransceivers();
>   ASSERT_EQ(2u, transceivers.size());
>-  EXPECT_EQ(rtc::nullopt, transceivers[0]->mid());
>+  EXPECT_EQ(absl::nullopt, transceivers[0]->mid());
>   EXPECT_TRUE(transceivers[0]->stopped());
>   EXPECT_EQ(caller->pc()->GetTransceivers()[0]->mid(), transceivers[1]->mid());
>   EXPECT_FALSE(transceivers[1]->stopped());
>@@ -603,6 +595,107 @@ TEST_F(PeerConnectionJsepTest,
>   EXPECT_FALSE(contents[1].rejected);
> }
> 
>+// Test that the offer/answer and the transceivers are correctly generated and
>+// updated when the media section is recycled after the callee stops a
>+// transceiver and sends an answer with a 0 port.
>+TEST_F(PeerConnectionJsepTest,
>+       RecycleMediaSectionWhenStoppingTransceiverOnAnswerer) {
>+  auto caller = CreatePeerConnection();
>+  auto first_transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
>+  auto callee = CreatePeerConnection();
>+
>+  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
>+  callee->pc()->GetTransceivers()[0]->Stop();
>+  ASSERT_TRUE(
>+      caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));
>+  EXPECT_TRUE(first_transceiver->stopped());
>+  // First transceivers aren't dissociated yet.
>+  ASSERT_NE(absl::nullopt, first_transceiver->mid());
>+  std::string first_mid = *first_transceiver->mid();
>+  EXPECT_EQ(first_mid, callee->pc()->GetTransceivers()[0]->mid());
>+
>+  // New offer exchange with new transceivers that recycles the m section
>+  // correctly.
>+  caller->AddAudioTrack("audio2");
>+  callee->AddAudioTrack("audio2");
>+  auto offer = caller->CreateOffer();
>+  auto offer_contents = offer->description()->contents();
>+  std::string second_mid = offer_contents[0].name;
>+  ASSERT_EQ(1u, offer_contents.size());
>+  EXPECT_FALSE(offer_contents[0].rejected);
>+  EXPECT_NE(first_mid, second_mid);
>+
>+  // Setting the offer on each side will dissociate the first transceivers and
>+  // associate the new transceivers.
>+  ASSERT_TRUE(
>+      caller->SetLocalDescription(CloneSessionDescription(offer.get())));
>+  EXPECT_EQ(absl::nullopt, first_transceiver->mid());
>+  EXPECT_EQ(second_mid, caller->pc()->GetTransceivers()[1]->mid());
>+  ASSERT_TRUE(callee->SetRemoteDescription(std::move(offer)));
>+  EXPECT_EQ(absl::nullopt, callee->pc()->GetTransceivers()[0]->mid());
>+  EXPECT_EQ(second_mid, callee->pc()->GetTransceivers()[1]->mid());
>+
>+  // The new answer should also recycle the m section correctly.
>+  auto answer = callee->CreateAnswer();
>+  auto answer_contents = answer->description()->contents();
>+  ASSERT_EQ(1u, answer_contents.size());
>+  EXPECT_FALSE(answer_contents[0].rejected);
>+  EXPECT_EQ(second_mid, answer_contents[0].name);
>+
>+  // Finishing the negotiation shouldn't add or dissociate any transceivers.
>+  ASSERT_TRUE(
>+      callee->SetLocalDescription(CloneSessionDescription(answer.get())));
>+  ASSERT_TRUE(caller->SetRemoteDescription(std::move(answer)));
>+  auto caller_transceivers = caller->pc()->GetTransceivers();
>+  ASSERT_EQ(2u, caller_transceivers.size());
>+  EXPECT_EQ(absl::nullopt, caller_transceivers[0]->mid());
>+  EXPECT_EQ(second_mid, caller_transceivers[1]->mid());
>+  auto callee_transceivers = callee->pc()->GetTransceivers();
>+  ASSERT_EQ(2u, callee_transceivers.size());
>+  EXPECT_EQ(absl::nullopt, callee_transceivers[0]->mid());
>+  EXPECT_EQ(second_mid, callee_transceivers[1]->mid());
>+}
>+
>+// Test that creating/setting a local offer that recycles an m= section is
>+// idempotent.
>+TEST_F(PeerConnectionJsepTest, CreateOfferRecyclesWhenOfferingTwice) {
>+  // Do a negotiation with a port 0 for the media section.
>+  auto caller = CreatePeerConnection();
>+  auto first_transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
>+  auto callee = CreatePeerConnection();
>+  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
>+  first_transceiver->Stop();
>+  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
>+  caller->AddAudioTrack("audio2");
>+
>+  // Create a new offer that recycles the media section and set it as a local
>+  // description.
>+  auto offer = caller->CreateOffer();
>+  auto offer_contents = offer->description()->contents();
>+  ASSERT_EQ(1u, offer_contents.size());
>+  EXPECT_FALSE(offer_contents[0].rejected);
>+  ASSERT_TRUE(caller->SetLocalDescription(std::move(offer)));
>+  EXPECT_FALSE(caller->pc()->GetTransceivers()[1]->stopped());
>+  std::string second_mid = offer_contents[0].name;
>+
>+  // Create another new offer and set the local description again without the
>+  // rest of any negotation ocurring.
>+  auto second_offer = caller->CreateOffer();
>+  auto second_offer_contents = second_offer->description()->contents();
>+  ASSERT_EQ(1u, second_offer_contents.size());
>+  EXPECT_FALSE(second_offer_contents[0].rejected);
>+  // The mid shouldn't change.
>+  EXPECT_EQ(second_mid, second_offer_contents[0].name);
>+
>+  ASSERT_TRUE(caller->SetLocalDescription(std::move(second_offer)));
>+  // Make sure that the caller's transceivers are associated correctly.
>+  auto caller_transceivers = caller->pc()->GetTransceivers();
>+  ASSERT_EQ(2u, caller_transceivers.size());
>+  EXPECT_EQ(absl::nullopt, caller_transceivers[0]->mid());
>+  EXPECT_EQ(second_mid, caller_transceivers[1]->mid());
>+  EXPECT_FALSE(caller_transceivers[1]->stopped());
>+}
>+
> // Test that the offer/answer and transceivers for both the caller and callee
> // side are generated/updated correctly when recycling an audio/video media
> // section as a media section of either the same or opposite type.
>@@ -648,7 +741,7 @@ TEST_P(RecycleMediaSectionTest, VerifyOfferAnswerAndTransceivers) {
>   // the MID for the new transceiver.
>   ASSERT_TRUE(
>       caller->SetLocalDescription(CloneSessionDescription(offer.get())));
>-  EXPECT_EQ(rtc::nullopt, first_transceiver->mid());
>+  EXPECT_EQ(absl::nullopt, first_transceiver->mid());
>   EXPECT_EQ(second_mid, second_transceiver->mid());
> 
>   // Setting the remote offer will dissociate the previous transceiver and
>@@ -656,10 +749,10 @@ TEST_P(RecycleMediaSectionTest, VerifyOfferAnswerAndTransceivers) {
>   ASSERT_TRUE(callee->SetRemoteDescription(std::move(offer)));
>   auto callee_transceivers = callee->pc()->GetTransceivers();
>   ASSERT_EQ(2u, callee_transceivers.size());
>-  EXPECT_EQ(rtc::nullopt, callee_transceivers[0]->mid());
>-  EXPECT_EQ(first_type_, callee_transceivers[0]->receiver()->media_type());
>+  EXPECT_EQ(absl::nullopt, callee_transceivers[0]->mid());
>+  EXPECT_EQ(first_type_, callee_transceivers[0]->media_type());
>   EXPECT_EQ(second_mid, callee_transceivers[1]->mid());
>-  EXPECT_EQ(second_type_, callee_transceivers[1]->receiver()->media_type());
>+  EXPECT_EQ(second_type_, callee_transceivers[1]->media_type());
> 
>   // The answer should have only one media section for the new transceiver.
>   auto answer = callee->CreateAnswer();
>@@ -673,8 +766,11 @@ TEST_P(RecycleMediaSectionTest, VerifyOfferAnswerAndTransceivers) {
>   ASSERT_TRUE(
>       callee->SetLocalDescription(CloneSessionDescription(answer.get())));
> 
>-  // Setting the remote answer should succeed.
>+  // Setting the remote answer should succeed and not create any new
>+  // transceivers.
>   ASSERT_TRUE(caller->SetRemoteDescription(std::move(answer)));
>+  ASSERT_EQ(2u, caller->pc()->GetTransceivers().size());
>+  ASSERT_EQ(2u, callee->pc()->GetTransceivers().size());
> }
> 
> // Test all combinations of audio and video as the first and second media type
>@@ -925,6 +1021,104 @@ TEST_F(PeerConnectionJsepTest,
>   EXPECT_NE(kTrackId, streams[0].id);
> }
> 
>+// Test that if the transceiver is recvonly or inactive, then no MSID
>+// information is included in the offer.
>+TEST_F(PeerConnectionJsepTest, NoMsidInOfferIfTransceiverDirectionHasNoSend) {
>+  auto caller = CreatePeerConnection();
>+
>+  RtpTransceiverInit init_recvonly;
>+  init_recvonly.direction = RtpTransceiverDirection::kRecvOnly;
>+  ASSERT_TRUE(caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO, init_recvonly));
>+
>+  RtpTransceiverInit init_inactive;
>+  init_inactive.direction = RtpTransceiverDirection::kInactive;
>+  ASSERT_TRUE(caller->AddTransceiver(cricket::MEDIA_TYPE_VIDEO, init_inactive));
>+
>+  auto offer = caller->CreateOffer();
>+  auto contents = offer->description()->contents();
>+  ASSERT_EQ(2u, contents.size());
>+  // MSID is specified in the first stream, so no streams means no MSID.
>+  EXPECT_EQ(0u, contents[0].media_description()->streams().size());
>+  EXPECT_EQ(0u, contents[1].media_description()->streams().size());
>+}
>+
>+// Test that if an answer negotiates transceiver directions of recvonly or
>+// inactive, then no MSID information is included in the answer.
>+TEST_F(PeerConnectionJsepTest, NoMsidInAnswerIfNoRespondingTracks) {
>+  auto caller = CreatePeerConnection();
>+  auto callee = CreatePeerConnection();
>+
>+  // recvonly transceiver will get negotiated to inactive since the callee has
>+  // no tracks to send in response.
>+  RtpTransceiverInit init_recvonly;
>+  init_recvonly.direction = RtpTransceiverDirection::kRecvOnly;
>+  caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO, init_recvonly);
>+
>+  // sendrecv transceiver will get negotiated to recvonly since the callee has
>+  // no tracks to send in response.
>+  RtpTransceiverInit init_sendrecv;
>+  init_sendrecv.direction = RtpTransceiverDirection::kSendRecv;
>+  caller->AddTransceiver(cricket::MEDIA_TYPE_VIDEO, init_sendrecv);
>+
>+  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
>+
>+  auto answer = callee->CreateAnswer();
>+  auto contents = answer->description()->contents();
>+  ASSERT_EQ(2u, contents.size());
>+  // MSID is specified in the first stream, so no streams means no MSID.
>+  EXPECT_EQ(0u, contents[0].media_description()->streams().size());
>+  EXPECT_EQ(0u, contents[1].media_description()->streams().size());
>+}
>+
>+// Test that the MSID is included even if the transceiver direction has changed
>+// to inactive if the transceiver had previously sent media.
>+TEST_F(PeerConnectionJsepTest, IncludeMsidEvenIfDirectionHasChanged) {
>+  auto caller = CreatePeerConnection();
>+  caller->AddAudioTrack("audio");
>+  auto callee = CreatePeerConnection();
>+  callee->AddAudioTrack("audio");
>+
>+  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
>+
>+  caller->pc()->GetTransceivers()[0]->SetDirection(
>+      RtpTransceiverDirection::kInactive);
>+
>+  // The transceiver direction on both sides will turn to inactive.
>+  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
>+
>+  auto* offer = callee->pc()->remote_description();
>+  auto offer_contents = offer->description()->contents();
>+  ASSERT_EQ(1u, offer_contents.size());
>+  // MSID is specified in the first stream. If it is present, assume that MSID
>+  // is there.
>+  EXPECT_EQ(1u, offer_contents[0].media_description()->streams().size());
>+
>+  auto* answer = caller->pc()->remote_description();
>+  auto answer_contents = answer->description()->contents();
>+  ASSERT_EQ(1u, answer_contents.size());
>+  EXPECT_EQ(1u, answer_contents[0].media_description()->streams().size());
>+}
>+
>+// Test that stopping a RtpTransceiver will cause future offers to not include
>+// any MSID information for that section.
>+TEST_F(PeerConnectionJsepTest, RemoveMsidIfTransceiverStopped) {
>+  auto caller = CreatePeerConnection();
>+  auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
>+  auto callee = CreatePeerConnection();
>+
>+  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
>+
>+  transceiver->Stop();
>+
>+  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
>+
>+  auto* offer = callee->pc()->remote_description();
>+  auto offer_contents = offer->description()->contents();
>+  ASSERT_EQ(1u, offer_contents.size());
>+  // MSID is specified in the first stream, so no streams means no MSID.
>+  EXPECT_EQ(0u, offer_contents[0].media_description()->streams().size());
>+}
>+
> // Test that the callee RtpReceiver created by a call to SetRemoteDescription
> // has its ID set to the signaled track ID.
> TEST_F(PeerConnectionJsepTest,
>@@ -974,11 +1168,11 @@ TEST_F(PeerConnectionJsepTest,
> 
>   ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
> 
>-  ASSERT_EQ(callee->observer()->add_track_events_.size(), 1u);
>-  EXPECT_EQ(kTrackLabel,
>-            callee->observer()->add_track_events_[0].receiver->track()->id());
>-  // TODO(bugs.webrtc.org/7933): Also verify that no stream was added to the
>-  // receiver.
>+  const auto& track_events = callee->observer()->add_track_events_;
>+  ASSERT_EQ(1u, track_events.size());
>+  const auto& event = track_events[0];
>+  EXPECT_EQ(kTrackLabel, event.receiver->track()->id());
>+  EXPECT_EQ(0u, event.streams.size());
> }
> 
> // Test that setting a remote offer with one track that has one stream fires off
>@@ -986,11 +1180,11 @@ TEST_F(PeerConnectionJsepTest,
> TEST_F(PeerConnectionJsepTest,
>        SetRemoteOfferWithOneTrackOneStreamFiresOnAddTrack) {
>   const std::string kTrackLabel = "audio_track";
>-  const std::string kStreamLabel = "audio_stream";
>+  const std::string kStreamId = "audio_stream";
> 
>   auto caller = CreatePeerConnection();
>   auto callee = CreatePeerConnection();
>-  ASSERT_TRUE(caller->AddAudioTrack(kTrackLabel, {kStreamLabel}));
>+  ASSERT_TRUE(caller->AddAudioTrack(kTrackLabel, {kStreamId}));
> 
>   ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
> 
>@@ -999,7 +1193,7 @@ TEST_F(PeerConnectionJsepTest,
>   const auto& event = track_events[0];
>   ASSERT_EQ(1u, event.streams.size());
>   auto stream = event.streams[0];
>-  EXPECT_EQ(kStreamLabel, stream->label());
>+  EXPECT_EQ(kStreamId, stream->id());
>   EXPECT_THAT(track_events[0].snapshotted_stream_tracks.at(stream),
>               ElementsAre(event.receiver->track()));
>   EXPECT_EQ(event.receiver->streams(), track_events[0].streams);
>@@ -1014,12 +1208,12 @@ TEST_F(PeerConnectionJsepTest,
>        SetRemoteOfferWithTwoTracksSameStreamFiresOnAddTrack) {
>   const std::string kTrack1Label = "audio_track1";
>   const std::string kTrack2Label = "audio_track2";
>-  const std::string kSharedStreamLabel = "stream";
>+  const std::string kSharedStreamId = "stream";
> 
>   auto caller = CreatePeerConnection();
>   auto callee = CreatePeerConnection();
>-  ASSERT_TRUE(caller->AddAudioTrack(kTrack1Label, {kSharedStreamLabel}));
>-  ASSERT_TRUE(caller->AddAudioTrack(kTrack2Label, {kSharedStreamLabel}));
>+  ASSERT_TRUE(caller->AddAudioTrack(kTrack1Label, {kSharedStreamId}));
>+  ASSERT_TRUE(caller->AddAudioTrack(kTrack2Label, {kSharedStreamId}));
> 
>   ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
> 
>@@ -1038,6 +1232,88 @@ TEST_F(PeerConnectionJsepTest,
>               UnorderedElementsAre(track1, track2));
> }
> 
>-// TODO(bugs.webrtc.org/7932): Also test multi-stream case.
>+// Test that setting a remote offer with one track that has two streams fires
>+// off the correct OnAddTrack event.
>+TEST_F(PeerConnectionJsepTest,
>+       SetRemoteOfferWithOneTrackTwoStreamFiresOnAddTrack) {
>+  const std::string kTrackLabel = "audio_track";
>+  const std::string kStreamId1 = "audio_stream1";
>+  const std::string kStreamId2 = "audio_stream2";
>+
>+  auto caller = CreatePeerConnection();
>+  auto callee = CreatePeerConnection();
>+  ASSERT_TRUE(caller->AddAudioTrack(kTrackLabel, {kStreamId1, kStreamId2}));
>+
>+  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
>+
>+  const auto& track_events = callee->observer()->add_track_events_;
>+  ASSERT_EQ(1u, track_events.size());
>+  const auto& event = track_events[0];
>+  ASSERT_EQ(2u, event.streams.size());
>+  EXPECT_EQ(kStreamId1, event.streams[0]->id());
>+  EXPECT_EQ(kStreamId2, event.streams[1]->id());
>+}
>+
>+// Test that if an RtpTransceiver with a current_direction set is stopped, then
>+// current_direction is changed to null.
>+TEST_F(PeerConnectionJsepTest, CurrentDirectionResetWhenRtpTransceiverStopped) {
>+  auto caller = CreatePeerConnection();
>+  auto callee = CreatePeerConnection();
>+
>+  auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
>+
>+  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
>+
>+  ASSERT_TRUE(transceiver->current_direction());
>+  transceiver->Stop();
>+  EXPECT_FALSE(transceiver->current_direction());
>+}
>+
>+// Test that you can't set an answer on a PeerConnection before setting the
>+// offer.
>+TEST_F(PeerConnectionJsepTest, AnswerBeforeOfferFails) {
>+  auto caller = CreatePeerConnection();
>+  auto callee = CreatePeerConnection();
>+  caller->AddAudioTrack("audio");
>+
>+  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOffer()));
>+
>+  RTCError error;
>+  ASSERT_FALSE(caller->SetRemoteDescription(callee->CreateAnswer(), &error));
>+  EXPECT_EQ(RTCErrorType::INVALID_STATE, error.type());
>+}
>+
>+// Test that a Unified Plan PeerConnection fails to set a Plan B offer if it has
>+// two video tracks.
>+TEST_F(PeerConnectionJsepTest, TwoVideoPlanBToUnifiedPlanFails) {
>+  RTCConfiguration config_planb;
>+  config_planb.sdp_semantics = SdpSemantics::kPlanB;
>+  auto caller = CreatePeerConnection(config_planb);
>+  auto callee = CreatePeerConnection();
>+  caller->AddVideoTrack("video1");
>+  caller->AddVideoTrack("video2");
>+
>+  RTCError error;
>+  ASSERT_FALSE(callee->SetRemoteDescription(caller->CreateOffer(), &error));
>+  EXPECT_EQ(RTCErrorType::INVALID_PARAMETER, error.type());
>+}
>+
>+// Test that a Unified Plan PeerConnection fails to set a Plan B answer if it
>+// has two video tracks.
>+TEST_F(PeerConnectionJsepTest, OneVideoUnifiedPlanToTwoVideoPlanBFails) {
>+  auto caller = CreatePeerConnection();
>+  RTCConfiguration config_planb;
>+  config_planb.sdp_semantics = SdpSemantics::kPlanB;
>+  auto callee = CreatePeerConnection(config_planb);
>+  caller->AddVideoTrack("video");
>+  callee->AddVideoTrack("video1");
>+  callee->AddVideoTrack("video2");
>+
>+  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
>+
>+  RTCError error;
>+  ASSERT_FALSE(caller->SetRemoteDescription(caller->CreateAnswer(), &error));
>+  EXPECT_EQ(RTCErrorType::INVALID_PARAMETER, error.type());
>+}
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_media_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_media_unittest.cc
>index 79ffb8d9350..cc7ae9afaf7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_media_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_media_unittest.cc
>@@ -14,7 +14,7 @@
> 
> #include <tuple>
> 
>-#include "call/callfactoryinterface.h"
>+#include "api/call/callfactoryinterface.h"
> #include "logging/rtc_event_log/rtc_event_log_factory.h"
> #include "media/base/fakemediaengine.h"
> #include "p2p/base/fakeportallocator.h"
>@@ -25,9 +25,9 @@
> #ifdef WEBRTC_ANDROID
> #include "pc/test/androidtestinitializer.h"
> #endif
>+#include "absl/memory/memory.h"
> #include "pc/test/fakertccertificategenerator.h"
> #include "rtc_base/gunit.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/virtualsocketserver.h"
> #include "test/gmock.h"
> 
>@@ -54,12 +54,14 @@ class PeerConnectionWrapperForMediaTest : public PeerConnectionWrapper {
>   FakeMediaEngine* media_engine_;
> };
> 
>-class PeerConnectionMediaTest : public ::testing::Test {
>+class PeerConnectionMediaBaseTest : public ::testing::Test {
>  protected:
>   typedef std::unique_ptr<PeerConnectionWrapperForMediaTest> WrapperPtr;
> 
>-  PeerConnectionMediaTest()
>-      : vss_(new rtc::VirtualSocketServer()), main_(vss_.get()) {
>+  explicit PeerConnectionMediaBaseTest(SdpSemantics sdp_semantics)
>+      : vss_(new rtc::VirtualSocketServer()),
>+        main_(vss_.get()),
>+        sdp_semantics_(sdp_semantics) {
> #ifdef WEBRTC_ANDROID
>     InitializeAndroidObjects();
> #endif
>@@ -70,23 +72,26 @@ class PeerConnectionMediaTest : public ::testing::Test {
>   }
> 
>   WrapperPtr CreatePeerConnection(const RTCConfiguration& config) {
>-    auto media_engine = rtc::MakeUnique<FakeMediaEngine>();
>+    auto media_engine = absl::make_unique<FakeMediaEngine>();
>     auto* media_engine_ptr = media_engine.get();
>     auto pc_factory = CreateModularPeerConnectionFactory(
>         rtc::Thread::Current(), rtc::Thread::Current(), rtc::Thread::Current(),
>         std::move(media_engine), CreateCallFactory(),
>         CreateRtcEventLogFactory());
> 
>-    auto fake_port_allocator = rtc::MakeUnique<cricket::FakePortAllocator>(
>+    auto fake_port_allocator = absl::make_unique<cricket::FakePortAllocator>(
>         rtc::Thread::Current(), nullptr);
>-    auto observer = rtc::MakeUnique<MockPeerConnectionObserver>();
>-    auto pc = pc_factory->CreatePeerConnection(
>-        config, std::move(fake_port_allocator), nullptr, observer.get());
>+    auto observer = absl::make_unique<MockPeerConnectionObserver>();
>+    auto modified_config = config;
>+    modified_config.sdp_semantics = sdp_semantics_;
>+    auto pc = pc_factory->CreatePeerConnection(modified_config,
>+                                               std::move(fake_port_allocator),
>+                                               nullptr, observer.get());
>     if (!pc) {
>       return nullptr;
>     }
> 
>-    auto wrapper = rtc::MakeUnique<PeerConnectionWrapperForMediaTest>(
>+    auto wrapper = absl::make_unique<PeerConnectionWrapperForMediaTest>(
>         pc_factory, pc, std::move(observer));
>     wrapper->set_media_engine(media_engine_ptr);
>     return wrapper;
>@@ -105,27 +110,44 @@ class PeerConnectionMediaTest : public ::testing::Test {
>     return wrapper;
>   }
> 
>-  const cricket::MediaContentDescription* GetMediaContent(
>+  RtpTransceiverDirection GetMediaContentDirection(
>       const SessionDescriptionInterface* sdesc,
>-      const std::string& mid) {
>-    const auto* content_desc =
>-        sdesc->description()->GetContentDescriptionByName(mid);
>-    return static_cast<const cricket::MediaContentDescription*>(content_desc);
>+      cricket::MediaType media_type) {
>+    auto* content =
>+        cricket::GetFirstMediaContent(sdesc->description(), media_type);
>+    RTC_DCHECK(content);
>+    return content->media_description()->direction();
>   }
> 
>-  RtpTransceiverDirection GetMediaContentDirection(
>-      const SessionDescriptionInterface* sdesc,
>-      const std::string& mid) {
>-    auto* media_content = GetMediaContent(sdesc, mid);
>-    RTC_DCHECK(media_content);
>-    return media_content->direction();
>+  bool IsUnifiedPlan() const {
>+    return sdp_semantics_ == SdpSemantics::kUnifiedPlan;
>   }
> 
>   std::unique_ptr<rtc::VirtualSocketServer> vss_;
>   rtc::AutoSocketServerThread main_;
>+  const SdpSemantics sdp_semantics_;
>+};
>+
>+class PeerConnectionMediaTest
>+    : public PeerConnectionMediaBaseTest,
>+      public ::testing::WithParamInterface<SdpSemantics> {
>+ protected:
>+  PeerConnectionMediaTest() : PeerConnectionMediaBaseTest(GetParam()) {}
> };
> 
>-TEST_F(PeerConnectionMediaTest,
>+class PeerConnectionMediaTestUnifiedPlan : public PeerConnectionMediaBaseTest {
>+ protected:
>+  PeerConnectionMediaTestUnifiedPlan()
>+      : PeerConnectionMediaBaseTest(SdpSemantics::kUnifiedPlan) {}
>+};
>+
>+class PeerConnectionMediaTestPlanB : public PeerConnectionMediaBaseTest {
>+ protected:
>+  PeerConnectionMediaTestPlanB()
>+      : PeerConnectionMediaBaseTest(SdpSemantics::kPlanB) {}
>+};
>+
>+TEST_P(PeerConnectionMediaTest,
>        FailToSetRemoteDescriptionIfCreateMediaChannelFails) {
>   auto caller = CreatePeerConnectionWithAudioVideo();
>   auto callee = CreatePeerConnectionWithAudioVideo();
>@@ -133,19 +155,19 @@ TEST_F(PeerConnectionMediaTest,
> 
>   std::string error;
>   ASSERT_FALSE(callee->SetRemoteDescription(caller->CreateOffer(), &error));
>-  EXPECT_EQ("Failed to set remote offer sdp: Failed to create voice channel.",
>-            error);
>+  EXPECT_PRED_FORMAT2(AssertStartsWith, error,
>+                      "Failed to set remote offer sdp: Failed to create");
> }
> 
>-TEST_F(PeerConnectionMediaTest,
>+TEST_P(PeerConnectionMediaTest,
>        FailToSetLocalDescriptionIfCreateMediaChannelFails) {
>   auto caller = CreatePeerConnectionWithAudioVideo();
>   caller->media_engine()->set_fail_create_channel(true);
> 
>   std::string error;
>   ASSERT_FALSE(caller->SetLocalDescription(caller->CreateOffer(), &error));
>-  EXPECT_EQ("Failed to set local offer sdp: Failed to create voice channel.",
>-            error);
>+  EXPECT_PRED_FORMAT2(AssertStartsWith, error,
>+                      "Failed to set local offer sdp: Failed to create");
> }
> 
> std::vector<std::string> GetIds(
>@@ -160,7 +182,7 @@ std::vector<std::string> GetIds(
> // Test that exchanging an offer and answer with each side having an audio and
> // video stream creates the appropriate send/recv streams in the underlying
> // media engine on both sides.
>-TEST_F(PeerConnectionMediaTest, AudioVideoOfferAnswerCreateSendRecvStreams) {
>+TEST_P(PeerConnectionMediaTest, AudioVideoOfferAnswerCreateSendRecvStreams) {
>   const std::string kCallerAudioId = "caller_a";
>   const std::string kCallerVideoId = "caller_v";
>   const std::string kCalleeAudioId = "callee_a";
>@@ -203,123 +225,179 @@ TEST_F(PeerConnectionMediaTest, AudioVideoOfferAnswerCreateSendRecvStreams) {
>               ElementsAre(kCalleeVideoId));
> }
> 
>+// Test that stopping the caller transceivers causes the media channels on the
>+// callee to be destroyed after calling SetRemoteDescription on the generated
>+// offer.
>+// See next test for equivalent behavior with Plan B semantics.
>+TEST_F(PeerConnectionMediaTestUnifiedPlan,
>+       StoppedRemoteTransceiversRemovesMediaChannels) {
>+  auto caller = CreatePeerConnectionWithAudioVideo();
>+  auto callee = CreatePeerConnection();
>+
>+  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
>+
>+  // Stop both audio and video transceivers on the caller.
>+  auto transceivers = caller->pc()->GetTransceivers();
>+  ASSERT_EQ(2u, transceivers.size());
>+  transceivers[0]->Stop();
>+  transceivers[1]->Stop();
>+
>+  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
>+
>+  ASSERT_FALSE(callee->media_engine()->GetVoiceChannel(0));
>+  ASSERT_FALSE(callee->media_engine()->GetVideoChannel(0));
>+}
>+
> // Test that removing streams from a subsequent offer causes the receive streams
> // on the callee to be removed.
>-TEST_F(PeerConnectionMediaTest, EmptyRemoteOfferRemovesRecvStreams) {
>+// See previous test for equivalent behavior with Unified Plan semantics.
>+TEST_F(PeerConnectionMediaTestPlanB, EmptyRemoteOfferRemovesRecvStreams) {
>   auto caller = CreatePeerConnection();
>   auto caller_audio_track = caller->AddAudioTrack("a");
>   auto caller_video_track = caller->AddVideoTrack("v");
>   auto callee = CreatePeerConnectionWithAudioVideo();
> 
>-  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
>-  ASSERT_TRUE(
>-      caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));
>+  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
> 
>   // Remove both tracks from caller.
>   caller->pc()->RemoveTrack(caller_audio_track);
>   caller->pc()->RemoveTrack(caller_video_track);
> 
>-  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
>-  ASSERT_TRUE(
>-      caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));
>+  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
> 
>   auto callee_voice = callee->media_engine()->GetVoiceChannel(0);
>+  auto callee_video = callee->media_engine()->GetVideoChannel(0);
>   EXPECT_EQ(1u, callee_voice->send_streams().size());
>   EXPECT_EQ(0u, callee_voice->recv_streams().size());
>-
>-  auto callee_video = callee->media_engine()->GetVideoChannel(0);
>   EXPECT_EQ(1u, callee_video->send_streams().size());
>   EXPECT_EQ(0u, callee_video->recv_streams().size());
> }
> 
>+// Test that stopping the callee transceivers causes the media channels to be
>+// destroyed on the callee after calling SetLocalDescription on the local
>+// answer.
>+// See next test for equivalent behavior with Plan B semantics.
>+TEST_F(PeerConnectionMediaTestUnifiedPlan,
>+       StoppedLocalTransceiversRemovesMediaChannels) {
>+  auto caller = CreatePeerConnectionWithAudioVideo();
>+  auto callee = CreatePeerConnectionWithAudioVideo();
>+
>+  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
>+
>+  // Stop both audio and video transceivers on the callee.
>+  auto transceivers = callee->pc()->GetTransceivers();
>+  ASSERT_EQ(2u, transceivers.size());
>+  transceivers[0]->Stop();
>+  transceivers[1]->Stop();
>+
>+  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
>+
>+  EXPECT_FALSE(callee->media_engine()->GetVoiceChannel(0));
>+  EXPECT_FALSE(callee->media_engine()->GetVideoChannel(0));
>+}
>+
> // Test that removing streams from a subsequent answer causes the send streams
> // on the callee to be removed when applied locally.
>-TEST_F(PeerConnectionMediaTest, EmptyLocalAnswerRemovesSendStreams) {
>+// See previous test for equivalent behavior with Unified Plan semantics.
>+TEST_F(PeerConnectionMediaTestPlanB, EmptyLocalAnswerRemovesSendStreams) {
>   auto caller = CreatePeerConnectionWithAudioVideo();
>   auto callee = CreatePeerConnection();
>   auto callee_audio_track = callee->AddAudioTrack("a");
>   auto callee_video_track = callee->AddVideoTrack("v");
> 
>-  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
>-  ASSERT_TRUE(
>-      caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));
>+  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
> 
>   // Remove both tracks from callee.
>   callee->pc()->RemoveTrack(callee_audio_track);
>   callee->pc()->RemoveTrack(callee_video_track);
> 
>-  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
>-  ASSERT_TRUE(
>-      caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));
>+  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
> 
>   auto callee_voice = callee->media_engine()->GetVoiceChannel(0);
>+  auto callee_video = callee->media_engine()->GetVideoChannel(0);
>   EXPECT_EQ(0u, callee_voice->send_streams().size());
>   EXPECT_EQ(1u, callee_voice->recv_streams().size());
>-
>-  auto callee_video = callee->media_engine()->GetVideoChannel(0);
>   EXPECT_EQ(0u, callee_video->send_streams().size());
>   EXPECT_EQ(1u, callee_video->recv_streams().size());
> }
> 
> // Test that a new stream in a subsequent offer causes a new receive stream to
> // be created on the callee.
>-TEST_F(PeerConnectionMediaTest, NewStreamInRemoteOfferAddsRecvStreams) {
>+TEST_P(PeerConnectionMediaTest, NewStreamInRemoteOfferAddsRecvStreams) {
>   auto caller = CreatePeerConnectionWithAudioVideo();
>   auto callee = CreatePeerConnection();
> 
>-  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
>-  ASSERT_TRUE(
>-      caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));
>+  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
> 
>   // Add second set of tracks to the caller.
>   caller->AddAudioTrack("a2");
>   caller->AddVideoTrack("v2");
> 
>-  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
>-  ASSERT_TRUE(
>-      caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));
>-
>-  auto callee_voice = callee->media_engine()->GetVoiceChannel(0);
>-  EXPECT_EQ(2u, callee_voice->recv_streams().size());
>-  auto callee_video = callee->media_engine()->GetVideoChannel(0);
>-  EXPECT_EQ(2u, callee_video->recv_streams().size());
>+  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
>+
>+  auto a1 = callee->media_engine()->GetVoiceChannel(0);
>+  auto a2 = callee->media_engine()->GetVoiceChannel(1);
>+  auto v1 = callee->media_engine()->GetVideoChannel(0);
>+  auto v2 = callee->media_engine()->GetVideoChannel(1);
>+  if (IsUnifiedPlan()) {
>+    ASSERT_TRUE(a1);
>+    EXPECT_EQ(1u, a1->recv_streams().size());
>+    ASSERT_TRUE(a2);
>+    EXPECT_EQ(1u, a2->recv_streams().size());
>+    ASSERT_TRUE(v1);
>+    EXPECT_EQ(1u, v1->recv_streams().size());
>+    ASSERT_TRUE(v2);
>+    EXPECT_EQ(1u, v2->recv_streams().size());
>+  } else {
>+    ASSERT_TRUE(a1);
>+    EXPECT_EQ(2u, a1->recv_streams().size());
>+    ASSERT_FALSE(a2);
>+    ASSERT_TRUE(v1);
>+    EXPECT_EQ(2u, v1->recv_streams().size());
>+    ASSERT_FALSE(v2);
>+  }
> }
> 
> // Test that a new stream in a subsequent answer causes a new send stream to be
> // created on the callee when added locally.
>-TEST_F(PeerConnectionMediaTest, NewStreamInLocalAnswerAddsSendStreams) {
>+TEST_P(PeerConnectionMediaTest, NewStreamInLocalAnswerAddsSendStreams) {
>   auto caller = CreatePeerConnection();
>   auto callee = CreatePeerConnectionWithAudioVideo();
> 
>-  RTCOfferAnswerOptions options;
>-  options.offer_to_receive_audio =
>+  RTCOfferAnswerOptions offer_options;
>+  offer_options.offer_to_receive_audio =
>       RTCOfferAnswerOptions::kOfferToReceiveMediaTrue;
>-  options.offer_to_receive_video =
>+  offer_options.offer_to_receive_video =
>       RTCOfferAnswerOptions::kOfferToReceiveMediaTrue;
>+  RTCOfferAnswerOptions answer_options;
> 
>-  ASSERT_TRUE(
>-      callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(options)));
>-  ASSERT_TRUE(
>-      caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));
>+  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get(), offer_options,
>+                                              answer_options));
> 
>   // Add second set of tracks to the callee.
>   callee->AddAudioTrack("a2");
>   callee->AddVideoTrack("v2");
> 
>-  ASSERT_TRUE(
>-      callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(options)));
>-  ASSERT_TRUE(
>-      caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));
>+  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get(), offer_options,
>+                                              answer_options));
> 
>   auto callee_voice = callee->media_engine()->GetVoiceChannel(0);
>-  EXPECT_EQ(2u, callee_voice->send_streams().size());
>+  ASSERT_TRUE(callee_voice);
>   auto callee_video = callee->media_engine()->GetVideoChannel(0);
>-  EXPECT_EQ(2u, callee_video->send_streams().size());
>+  ASSERT_TRUE(callee_video);
>+
>+  if (IsUnifiedPlan()) {
>+    EXPECT_EQ(1u, callee_voice->send_streams().size());
>+    EXPECT_EQ(1u, callee_video->send_streams().size());
>+  } else {
>+    EXPECT_EQ(2u, callee_voice->send_streams().size());
>+    EXPECT_EQ(2u, callee_video->send_streams().size());
>+  }
> }
> 
> // A PeerConnection with no local streams and no explicit answer constraints
> // should not reject any offered media sections.
>-TEST_F(PeerConnectionMediaTest,
>+TEST_P(PeerConnectionMediaTest,
>        CreateAnswerWithNoStreamsAndDefaultOptionsDoesNotReject) {
>   auto caller = CreatePeerConnectionWithAudioVideo();
>   auto callee = CreatePeerConnection();
>@@ -338,14 +416,17 @@ TEST_F(PeerConnectionMediaTest,
> }
> 
> class PeerConnectionMediaOfferDirectionTest
>-    : public PeerConnectionMediaTest,
>+    : public PeerConnectionMediaBaseTest,
>       public ::testing::WithParamInterface<
>-          std::tuple<bool, int, RtpTransceiverDirection>> {
>+          std::tuple<SdpSemantics,
>+                     std::tuple<bool, int, RtpTransceiverDirection>>> {
>  protected:
>-  PeerConnectionMediaOfferDirectionTest() {
>-    send_media_ = std::get<0>(GetParam());
>-    offer_to_receive_ = std::get<1>(GetParam());
>-    expected_direction_ = std::get<2>(GetParam());
>+  PeerConnectionMediaOfferDirectionTest()
>+      : PeerConnectionMediaBaseTest(std::get<0>(GetParam())) {
>+    auto param = std::get<1>(GetParam());
>+    send_media_ = std::get<0>(param);
>+    offer_to_receive_ = std::get<1>(param);
>+    expected_direction_ = std::get<2>(param);
>   }
> 
>   bool send_media_;
>@@ -365,11 +446,12 @@ TEST_P(PeerConnectionMediaOfferDirectionTest, VerifyDirection) {
>   options.offer_to_receive_audio = offer_to_receive_;
>   auto offer = caller->CreateOffer(options);
> 
>-  auto* media_content = GetMediaContent(offer.get(), cricket::CN_AUDIO);
>+  auto* content = cricket::GetFirstMediaContent(offer->description(),
>+                                                cricket::MEDIA_TYPE_AUDIO);
>   if (expected_direction_ == RtpTransceiverDirection::kInactive) {
>-    EXPECT_FALSE(media_content);
>+    EXPECT_FALSE(content);
>   } else {
>-    EXPECT_EQ(expected_direction_, media_content->direction());
>+    EXPECT_EQ(expected_direction_, content->media_description()->direction());
>   }
> }
> 
>@@ -378,22 +460,25 @@ TEST_P(PeerConnectionMediaOfferDirectionTest, VerifyDirection) {
> INSTANTIATE_TEST_CASE_P(
>     PeerConnectionMediaTest,
>     PeerConnectionMediaOfferDirectionTest,
>-    Values(std::make_tuple(false, -1, RtpTransceiverDirection::kInactive),
>-           std::make_tuple(false, 0, RtpTransceiverDirection::kInactive),
>-           std::make_tuple(false, 1, RtpTransceiverDirection::kRecvOnly),
>-           std::make_tuple(true, -1, RtpTransceiverDirection::kSendRecv),
>-           std::make_tuple(true, 0, RtpTransceiverDirection::kSendOnly),
>-           std::make_tuple(true, 1, RtpTransceiverDirection::kSendRecv)));
>+    Combine(
>+        Values(SdpSemantics::kPlanB, SdpSemantics::kUnifiedPlan),
>+        Values(std::make_tuple(false, -1, RtpTransceiverDirection::kInactive),
>+               std::make_tuple(false, 0, RtpTransceiverDirection::kInactive),
>+               std::make_tuple(false, 1, RtpTransceiverDirection::kRecvOnly),
>+               std::make_tuple(true, -1, RtpTransceiverDirection::kSendRecv),
>+               std::make_tuple(true, 0, RtpTransceiverDirection::kSendOnly),
>+               std::make_tuple(true, 1, RtpTransceiverDirection::kSendRecv))));
> 
> class PeerConnectionMediaAnswerDirectionTest
>-    : public PeerConnectionMediaTest,
>+    : public PeerConnectionMediaBaseTest,
>       public ::testing::WithParamInterface<
>-          std::tuple<RtpTransceiverDirection, bool, int>> {
>+          std::tuple<SdpSemantics, RtpTransceiverDirection, bool, int>> {
>  protected:
>-  PeerConnectionMediaAnswerDirectionTest() {
>-    offer_direction_ = std::get<0>(GetParam());
>-    send_media_ = std::get<1>(GetParam());
>-    offer_to_receive_ = std::get<2>(GetParam());
>+  PeerConnectionMediaAnswerDirectionTest()
>+      : PeerConnectionMediaBaseTest(std::get<0>(GetParam())) {
>+    offer_direction_ = std::get<1>(GetParam());
>+    send_media_ = std::get<2>(GetParam());
>+    offer_to_receive_ = std::get<3>(GetParam());
>   }
> 
>   RtpTransceiverDirection offer_direction_;
>@@ -405,6 +490,13 @@ class PeerConnectionMediaAnswerDirectionTest
> // in the offer, the presence of a local media track on the receive side and the
> // offer_to_receive setting.
> TEST_P(PeerConnectionMediaAnswerDirectionTest, VerifyDirection) {
>+  if (IsUnifiedPlan() &&
>+      offer_to_receive_ != RTCOfferAnswerOptions::kUndefined) {
>+    // offer_to_receive_ is not implemented when creating answers with Unified
>+    // Plan semantics specified.
>+    return;
>+  }
>+
>   auto caller = CreatePeerConnection();
>   caller->AddAudioTrack("a");
> 
>@@ -441,13 +533,20 @@ TEST_P(PeerConnectionMediaAnswerDirectionTest, VerifyDirection) {
>   auto expected_direction =
>       RtpTransceiverDirectionFromSendRecv(negotiate_send, negotiate_recv);
>   EXPECT_EQ(expected_direction,
>-            GetMediaContentDirection(answer.get(), cricket::CN_AUDIO));
>+            GetMediaContentDirection(answer.get(), cricket::MEDIA_TYPE_AUDIO));
> }
> 
> // Tests that the media section is rejected if and only if the callee has no
> // local media track and has set offer_to_receive to 0, no matter which
> // direction the caller indicated in the offer.
> TEST_P(PeerConnectionMediaAnswerDirectionTest, VerifyRejected) {
>+  if (IsUnifiedPlan() &&
>+      offer_to_receive_ != RTCOfferAnswerOptions::kUndefined) {
>+    // offer_to_receive_ is not implemented when creating answers with Unified
>+    // Plan semantics specified.
>+    return;
>+  }
>+
>   auto caller = CreatePeerConnection();
>   caller->AddAudioTrack("a");
> 
>@@ -476,14 +575,16 @@ TEST_P(PeerConnectionMediaAnswerDirectionTest, VerifyRejected) {
> 
> INSTANTIATE_TEST_CASE_P(PeerConnectionMediaTest,
>                         PeerConnectionMediaAnswerDirectionTest,
>-                        Combine(Values(RtpTransceiverDirection::kInactive,
>+                        Combine(Values(SdpSemantics::kPlanB,
>+                                       SdpSemantics::kUnifiedPlan),
>+                                Values(RtpTransceiverDirection::kInactive,
>                                        RtpTransceiverDirection::kSendOnly,
>                                        RtpTransceiverDirection::kRecvOnly,
>                                        RtpTransceiverDirection::kSendRecv),
>                                 Bool(),
>                                 Values(-1, 0, 1)));
> 
>-TEST_F(PeerConnectionMediaTest, OfferHasDifferentDirectionForAudioVideo) {
>+TEST_P(PeerConnectionMediaTest, OfferHasDifferentDirectionForAudioVideo) {
>   auto caller = CreatePeerConnection();
>   caller->AddVideoTrack("v");
> 
>@@ -493,12 +594,18 @@ TEST_F(PeerConnectionMediaTest, OfferHasDifferentDirectionForAudioVideo) {
>   auto offer = caller->CreateOffer(options);
> 
>   EXPECT_EQ(RtpTransceiverDirection::kRecvOnly,
>-            GetMediaContentDirection(offer.get(), cricket::CN_AUDIO));
>+            GetMediaContentDirection(offer.get(), cricket::MEDIA_TYPE_AUDIO));
>   EXPECT_EQ(RtpTransceiverDirection::kSendOnly,
>-            GetMediaContentDirection(offer.get(), cricket::CN_VIDEO));
>+            GetMediaContentDirection(offer.get(), cricket::MEDIA_TYPE_VIDEO));
> }
> 
>-TEST_F(PeerConnectionMediaTest, AnswerHasDifferentDirectionsForAudioVideo) {
>+TEST_P(PeerConnectionMediaTest, AnswerHasDifferentDirectionsForAudioVideo) {
>+  if (IsUnifiedPlan()) {
>+    // offer_to_receive_ is not implemented when creating answers with Unified
>+    // Plan semantics specified.
>+    return;
>+  }
>+
>   auto caller = CreatePeerConnectionWithAudioVideo();
>   auto callee = CreatePeerConnection();
>   callee->AddVideoTrack("v");
>@@ -511,9 +618,9 @@ TEST_F(PeerConnectionMediaTest, AnswerHasDifferentDirectionsForAudioVideo) {
>   auto answer = callee->CreateAnswer(options);
> 
>   EXPECT_EQ(RtpTransceiverDirection::kRecvOnly,
>-            GetMediaContentDirection(answer.get(), cricket::CN_AUDIO));
>+            GetMediaContentDirection(answer.get(), cricket::MEDIA_TYPE_AUDIO));
>   EXPECT_EQ(RtpTransceiverDirection::kSendOnly,
>-            GetMediaContentDirection(answer.get(), cricket::CN_VIDEO));
>+            GetMediaContentDirection(answer.get(), cricket::MEDIA_TYPE_VIDEO));
> }
> 
> void AddComfortNoiseCodecsToSend(cricket::FakeMediaEngine* media_engine) {
>@@ -536,7 +643,7 @@ bool HasAnyComfortNoiseCodecs(const cricket::SessionDescription* desc) {
>   return false;
> }
> 
>-TEST_F(PeerConnectionMediaTest,
>+TEST_P(PeerConnectionMediaTest,
>        CreateOfferWithNoVoiceActivityDetectionIncludesNoComfortNoiseCodecs) {
>   auto caller = CreatePeerConnectionWithAudioVideo();
>   AddComfortNoiseCodecsToSend(caller->media_engine());
>@@ -548,7 +655,7 @@ TEST_F(PeerConnectionMediaTest,
>   EXPECT_FALSE(HasAnyComfortNoiseCodecs(offer->description()));
> }
> 
>-TEST_F(PeerConnectionMediaTest,
>+TEST_P(PeerConnectionMediaTest,
>        CreateAnswerWithNoVoiceActivityDetectionIncludesNoComfortNoiseCodecs) {
>   auto caller = CreatePeerConnectionWithAudioVideo();
>   AddComfortNoiseCodecsToSend(caller->media_engine());
>@@ -568,15 +675,18 @@ TEST_F(PeerConnectionMediaTest,
> // sections as per RFC 3264.
> 
> class PeerConnectionMediaInvalidMediaTest
>-    : public PeerConnectionMediaTest,
>-      public ::testing::WithParamInterface<
>+    : public PeerConnectionMediaBaseTest,
>+      public ::testing::WithParamInterface<std::tuple<
>+          SdpSemantics,
>           std::tuple<std::string,
>                      std::function<void(cricket::SessionDescription*)>,
>-                     std::string>> {
>+                     std::string>>> {
>  protected:
>-  PeerConnectionMediaInvalidMediaTest() {
>-    mutator_ = std::get<1>(GetParam());
>-    expected_error_ = std::get<2>(GetParam());
>+  PeerConnectionMediaInvalidMediaTest()
>+      : PeerConnectionMediaBaseTest(std::get<0>(GetParam())) {
>+    auto param = std::get<1>(GetParam());
>+    mutator_ = std::get<1>(param);
>+    expected_error_ = std::get<2>(param);
>   }
> 
>   std::function<void(cricket::SessionDescription*)> mutator_;
>@@ -630,9 +740,10 @@ void ReverseMediaContent(cricket::SessionDescription* desc) {
> }
> 
> void ChangeMediaTypeAudioToVideo(cricket::SessionDescription* desc) {
>-  desc->RemoveContentByName(cricket::CN_AUDIO);
>-  auto* video_content = desc->GetContentByName(cricket::CN_VIDEO);
>-  desc->AddContent(cricket::CN_AUDIO, video_content->type,
>+  std::string audio_mid = cricket::GetFirstAudioContent(desc)->name;
>+  desc->RemoveContentByName(audio_mid);
>+  auto* video_content = cricket::GetFirstVideoContent(desc);
>+  desc->AddContent(audio_mid, video_content->type,
>                    video_content->media_description()->Copy());
> }
> 
>@@ -643,20 +754,30 @@ constexpr char kMLinesOutOfOrder[] =
> INSTANTIATE_TEST_CASE_P(
>     PeerConnectionMediaTest,
>     PeerConnectionMediaInvalidMediaTest,
>-    Values(
>-        std::make_tuple("remove video", RemoveVideoContent, kMLinesOutOfOrder),
>-        std::make_tuple("rename video", RenameVideoContent, kMLinesOutOfOrder),
>-        std::make_tuple("reverse media sections",
>-                        ReverseMediaContent,
>-                        kMLinesOutOfOrder),
>-        std::make_tuple("change audio type to video type",
>-                        ChangeMediaTypeAudioToVideo,
>-                        kMLinesOutOfOrder)));
>+    Combine(Values(SdpSemantics::kPlanB, SdpSemantics::kUnifiedPlan),
>+            Values(std::make_tuple("remove video",
>+                                   RemoveVideoContent,
>+                                   kMLinesOutOfOrder),
>+                   std::make_tuple("rename video",
>+                                   RenameVideoContent,
>+                                   kMLinesOutOfOrder),
>+                   std::make_tuple("reverse media sections",
>+                                   ReverseMediaContent,
>+                                   kMLinesOutOfOrder),
>+                   std::make_tuple("change audio type to video type",
>+                                   ChangeMediaTypeAudioToVideo,
>+                                   kMLinesOutOfOrder))));
> 
> // Test that the correct media engine send/recv streams are created when doing
> // a series of offer/answers where audio/video are both sent, then audio is
> // rejected, then both audio/video sent again.
>-TEST_F(PeerConnectionMediaTest, TestAVOfferWithAudioOnlyAnswer) {
>+TEST_P(PeerConnectionMediaTest, TestAVOfferWithAudioOnlyAnswer) {
>+  if (IsUnifiedPlan()) {
>+    // offer_to_receive_ is not implemented when creating answers with Unified
>+    // Plan semantics specified.
>+    return;
>+  }
>+
>   RTCOfferAnswerOptions options_reject_video;
>   options_reject_video.offer_to_receive_audio =
>       RTCOfferAnswerOptions::kOfferToReceiveMediaTrue;
>@@ -714,7 +835,13 @@ TEST_F(PeerConnectionMediaTest, TestAVOfferWithAudioOnlyAnswer) {
> // Test that the correct media engine send/recv streams are created when doing
> // a series of offer/answers where audio/video are both sent, then video is
> // rejected, then both audio/video sent again.
>-TEST_F(PeerConnectionMediaTest, TestAVOfferWithVideoOnlyAnswer) {
>+TEST_P(PeerConnectionMediaTest, TestAVOfferWithVideoOnlyAnswer) {
>+  if (IsUnifiedPlan()) {
>+    // offer_to_receive_ is not implemented when creating answers with Unified
>+    // Plan semantics specified.
>+    return;
>+  }
>+
>   // Disable the bundling here. If the media is bundled on audio
>   // transport, then we can't reject the audio because switching the bundled
>   // transport is not currently supported.
>@@ -778,7 +905,7 @@ TEST_F(PeerConnectionMediaTest, TestAVOfferWithVideoOnlyAnswer) {
> // Tests that if the underlying video encoder fails to be initialized (signaled
> // by failing to set send codecs), the PeerConnection signals the error to the
> // client.
>-TEST_F(PeerConnectionMediaTest, MediaEngineErrorPropagatedToClients) {
>+TEST_P(PeerConnectionMediaTest, MediaEngineErrorPropagatedToClients) {
>   auto caller = CreatePeerConnectionWithAudioVideo();
>   auto callee = CreatePeerConnectionWithAudioVideo();
> 
>@@ -791,16 +918,15 @@ TEST_F(PeerConnectionMediaTest, MediaEngineErrorPropagatedToClients) {
>   ASSERT_FALSE(caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal(),
>                                             &error));
>   EXPECT_EQ(
>-      "Failed to set remote answer sdp: Session error code: ERROR_CONTENT. "
>-      "Session error description: Failed to set remote video description send "
>-      "parameters..",
>+      "Failed to set remote answer sdp: Failed to set remote video description "
>+      "send parameters.",
>       error);
> }
> 
> // Tests that if the underlying video encoder fails once then subsequent
> // attempts at setting the local/remote description will also fail, even if
> // SetSendCodecs no longer fails.
>-TEST_F(PeerConnectionMediaTest,
>+TEST_P(PeerConnectionMediaTest,
>        FailToApplyDescriptionIfVideoEncoderHasEverFailed) {
>   auto caller = CreatePeerConnectionWithAudioVideo();
>   auto callee = CreatePeerConnectionWithAudioVideo();
>@@ -820,27 +946,36 @@ TEST_F(PeerConnectionMediaTest,
> }
> 
> void RenameContent(cricket::SessionDescription* desc,
>-                   const std::string& old_name,
>+                   cricket::MediaType media_type,
>                    const std::string& new_name) {
>-  auto* content = desc->GetContentByName(old_name);
>+  auto* content = cricket::GetFirstMediaContent(desc, media_type);
>   RTC_DCHECK(content);
>+  std::string old_name = content->name;
>   content->name = new_name;
>   auto* transport = desc->GetTransportInfoByName(old_name);
>   RTC_DCHECK(transport);
>   transport->content_name = new_name;
>+
>+  // Rename the content name in the BUNDLE group.
>+  cricket::ContentGroup new_bundle_group =
>+      *desc->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
>+  new_bundle_group.RemoveContentName(old_name);
>+  new_bundle_group.AddContentName(new_name);
>+  desc->RemoveGroupByName(cricket::GROUP_TYPE_BUNDLE);
>+  desc->AddGroup(new_bundle_group);
> }
> 
> // Tests that an answer responds with the same MIDs as the offer.
>-TEST_F(PeerConnectionMediaTest, AnswerHasSameMidsAsOffer) {
>-  const std::string kAudioMid = "not default1";
>-  const std::string kVideoMid = "not default2";
>+TEST_P(PeerConnectionMediaTest, AnswerHasSameMidsAsOffer) {
>+  const std::string kAudioMid = "notdefault1";
>+  const std::string kVideoMid = "notdefault2";
> 
>   auto caller = CreatePeerConnectionWithAudioVideo();
>   auto callee = CreatePeerConnectionWithAudioVideo();
> 
>   auto offer = caller->CreateOffer();
>-  RenameContent(offer->description(), cricket::CN_AUDIO, kAudioMid);
>-  RenameContent(offer->description(), cricket::CN_VIDEO, kVideoMid);
>+  RenameContent(offer->description(), cricket::MEDIA_TYPE_AUDIO, kAudioMid);
>+  RenameContent(offer->description(), cricket::MEDIA_TYPE_VIDEO, kVideoMid);
>   ASSERT_TRUE(callee->SetRemoteDescription(std::move(offer)));
> 
>   auto answer = callee->CreateAnswer();
>@@ -852,16 +987,16 @@ TEST_F(PeerConnectionMediaTest, AnswerHasSameMidsAsOffer) {
> 
> // Test that if the callee creates a re-offer, the MIDs are the same as the
> // original offer.
>-TEST_F(PeerConnectionMediaTest, ReOfferHasSameMidsAsFirstOffer) {
>-  const std::string kAudioMid = "not default1";
>-  const std::string kVideoMid = "not default2";
>+TEST_P(PeerConnectionMediaTest, ReOfferHasSameMidsAsFirstOffer) {
>+  const std::string kAudioMid = "notdefault1";
>+  const std::string kVideoMid = "notdefault2";
> 
>   auto caller = CreatePeerConnectionWithAudioVideo();
>   auto callee = CreatePeerConnectionWithAudioVideo();
> 
>   auto offer = caller->CreateOffer();
>-  RenameContent(offer->description(), cricket::CN_AUDIO, kAudioMid);
>-  RenameContent(offer->description(), cricket::CN_VIDEO, kVideoMid);
>+  RenameContent(offer->description(), cricket::MEDIA_TYPE_AUDIO, kAudioMid);
>+  RenameContent(offer->description(), cricket::MEDIA_TYPE_VIDEO, kVideoMid);
>   ASSERT_TRUE(callee->SetRemoteDescription(std::move(offer)));
>   ASSERT_TRUE(callee->SetLocalDescription(callee->CreateAnswer()));
> 
>@@ -872,7 +1007,7 @@ TEST_F(PeerConnectionMediaTest, ReOfferHasSameMidsAsFirstOffer) {
>             cricket::GetFirstVideoContent(reoffer->description())->name);
> }
> 
>-TEST_F(PeerConnectionMediaTest,
>+TEST_P(PeerConnectionMediaTest,
>        CombinedAudioVideoBweConfigPropagatedToMediaEngine) {
>   RTCConfiguration config;
>   config.combined_audio_video_bwe.emplace(true);
>@@ -887,4 +1022,9 @@ TEST_F(PeerConnectionMediaTest,
>             audio_options.combined_audio_video_bwe);
> }
> 
>+INSTANTIATE_TEST_CASE_P(PeerConnectionMediaTest,
>+                        PeerConnectionMediaTest,
>+                        Values(SdpSemantics::kPlanB,
>+                               SdpSemantics::kUnifiedPlan));
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_rampup_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_rampup_tests.cc
>new file mode 100644
>index 00000000000..6d6a9fc776d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_rampup_tests.cc
>@@ -0,0 +1,397 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
>+#include "api/audio_codecs/builtin_audio_encoder_factory.h"
>+#include "api/stats/rtcstats_objects.h"
>+#include "api/test/fakeconstraints.h"
>+#include "api/video_codecs/builtin_video_decoder_factory.h"
>+#include "api/video_codecs/builtin_video_encoder_factory.h"
>+#include "p2p/base/testturnserver.h"
>+#include "p2p/client/basicportallocator.h"
>+#include "pc/peerconnection.h"
>+#include "pc/peerconnectionwrapper.h"
>+#include "pc/test/fakeaudiocapturemodule.h"
>+#include "pc/test/fakeperiodicvideotracksource.h"
>+#include "pc/test/fakertccertificategenerator.h"
>+#include "pc/test/fakevideotrackrenderer.h"
>+#include "pc/test/framegeneratorcapturervideotracksource.h"
>+#include "rtc_base/fakenetwork.h"
>+#include "rtc_base/firewallsocketserver.h"
>+#include "rtc_base/gunit.h"
>+#include "rtc_base/platform_thread.h"
>+#include "rtc_base/socketaddress.h"
>+#include "rtc_base/testcertificateverifier.h"
>+#include "rtc_base/virtualsocketserver.h"
>+#include "test/gtest.h"
>+#include "test/testsupport/perf_test.h"
>+
>+namespace webrtc {
>+
>+namespace {
>+static const int kDefaultTestTimeMs = 15000;
>+static const int kRampUpTimeMs = 5000;
>+static const int kPollIntervalTimeMs = 50;
>+static const int kDefaultTimeoutMs = 10000;
>+static const rtc::SocketAddress kDefaultLocalAddress("1.1.1.1", 0);
>+static const char kTurnInternalAddress[] = "88.88.88.0";
>+static const char kTurnExternalAddress[] = "88.88.88.1";
>+static const int kTurnInternalPort = 3478;
>+static const int kTurnExternalPort = 0;
>+// The video's configured max bitrate in webrtcvideoengine.cc is 1.7 Mbps.
>+// Setting the network bandwidth to 1 Mbps allows the video's bitrate to push
>+// the network's limitations.
>+static const int kNetworkBandwidth = 1000000;
>+}  // namespace
>+
>+using RTCConfiguration = PeerConnectionInterface::RTCConfiguration;
>+
>+// This is an end to end test to verify that BWE is functioning when setting
>+// up a one to one call at the PeerConnection level. The intention of the test
>+// is to catch potential regressions for different ICE path configurations. The
>+// test uses a VirtualSocketServer for it's underlying simulated network and
>+// fake audio and video sources. The test is based upon rampup_tests.cc, but
>+// instead is at the PeerConnection level and uses a different fake network
>+// (rampup_tests.cc uses SimulatedNetwork). In the future, this test could
>+// potentially test different network conditions and test video quality as well
>+// (video_quality_test.cc does this, but at the call level).
>+//
>+// The perf test results are printed using the perf test support. If the
>+// isolated_script_test_perf_output flag is specified in test_main.cc, then
>+// the results are written to a JSON formatted file for the Chrome perf
>+// dashboard. Since this test is a webrtc_perf_test, it will be run in the perf
>+// console every webrtc commit.
>+class PeerConnectionWrapperForRampUpTest : public PeerConnectionWrapper {
>+ public:
>+  using PeerConnectionWrapper::PeerConnectionWrapper;
>+
>+  PeerConnectionWrapperForRampUpTest(
>+      rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory,
>+      rtc::scoped_refptr<PeerConnectionInterface> pc,
>+      std::unique_ptr<MockPeerConnectionObserver> observer)
>+      : PeerConnectionWrapper::PeerConnectionWrapper(pc_factory,
>+                                                     pc,
>+                                                     std::move(observer)) {}
>+
>+  bool AddIceCandidates(std::vector<const IceCandidateInterface*> candidates) {
>+    bool success = true;
>+    for (const auto candidate : candidates) {
>+      if (!pc()->AddIceCandidate(candidate)) {
>+        success = false;
>+      }
>+    }
>+    return success;
>+  }
>+
>+  rtc::scoped_refptr<VideoTrackInterface> CreateLocalVideoTrack(
>+      FrameGeneratorCapturerVideoTrackSource::Config config,
>+      Clock* clock) {
>+    video_track_sources_.emplace_back(
>+        new rtc::RefCountedObject<FrameGeneratorCapturerVideoTrackSource>(
>+            config, clock));
>+    video_track_sources_.back()->Start();
>+    return rtc::scoped_refptr<VideoTrackInterface>(
>+        pc_factory()->CreateVideoTrack(rtc::CreateRandomUuid(),
>+                                       video_track_sources_.back()));
>+  }
>+
>+  rtc::scoped_refptr<AudioTrackInterface> CreateLocalAudioTrack(
>+      const cricket::AudioOptions options) {
>+    rtc::scoped_refptr<AudioSourceInterface> source =
>+        pc_factory()->CreateAudioSource(options);
>+    return pc_factory()->CreateAudioTrack(rtc::CreateRandomUuid(), source);
>+  }
>+
>+ private:
>+  std::vector<rtc::scoped_refptr<FrameGeneratorCapturerVideoTrackSource>>
>+      video_track_sources_;
>+};
>+
>+// TODO(shampson): Paramaterize the test to run for both Plan B & Unified Plan.
>+class PeerConnectionRampUpTest : public ::testing::Test {
>+ public:
>+  PeerConnectionRampUpTest()
>+      : clock_(Clock::GetRealTimeClock()),
>+        virtual_socket_server_(new rtc::VirtualSocketServer()),
>+        firewall_socket_server_(
>+            new rtc::FirewallSocketServer(virtual_socket_server_.get())),
>+        network_thread_(new rtc::Thread(firewall_socket_server_.get())),
>+        worker_thread_(rtc::Thread::Create()) {
>+    network_thread_->SetName("PCNetworkThread", this);
>+    worker_thread_->SetName("PCWorkerThread", this);
>+    RTC_CHECK(network_thread_->Start());
>+    RTC_CHECK(worker_thread_->Start());
>+
>+    virtual_socket_server_->set_bandwidth(kNetworkBandwidth / 8);
>+    pc_factory_ = CreatePeerConnectionFactory(
>+        network_thread_.get(), worker_thread_.get(), rtc::Thread::Current(),
>+        rtc::scoped_refptr<AudioDeviceModule>(FakeAudioCaptureModule::Create()),
>+        CreateBuiltinAudioEncoderFactory(), CreateBuiltinAudioDecoderFactory(),
>+        CreateBuiltinVideoEncoderFactory(), CreateBuiltinVideoDecoderFactory(),
>+        nullptr /* audio_mixer */, nullptr /* audio_processing */);
>+  }
>+
>+  virtual ~PeerConnectionRampUpTest() {
>+    network_thread()->Invoke<void>(RTC_FROM_HERE,
>+                                   [this] { turn_servers_.clear(); });
>+  }
>+
>+  bool CreatePeerConnectionWrappers(const RTCConfiguration& caller_config,
>+                                    const RTCConfiguration& callee_config) {
>+    caller_ = CreatePeerConnectionWrapper(caller_config);
>+    callee_ = CreatePeerConnectionWrapper(callee_config);
>+    return caller_ && callee_;
>+  }
>+
>+  std::unique_ptr<PeerConnectionWrapperForRampUpTest>
>+  CreatePeerConnectionWrapper(const RTCConfiguration& config) {
>+    auto* fake_network_manager = new rtc::FakeNetworkManager();
>+    fake_network_manager->AddInterface(kDefaultLocalAddress);
>+    fake_network_managers_.emplace_back(fake_network_manager);
>+
>+    auto observer = absl::make_unique<MockPeerConnectionObserver>();
>+    webrtc::PeerConnectionDependencies dependencies(observer.get());
>+    cricket::BasicPortAllocator* port_allocator =
>+        new cricket::BasicPortAllocator(fake_network_manager);
>+    port_allocator->set_step_delay(cricket::kDefaultStepDelay);
>+    dependencies.allocator =
>+        std::unique_ptr<cricket::BasicPortAllocator>(port_allocator);
>+    dependencies.tls_cert_verifier =
>+        absl::make_unique<rtc::TestCertificateVerifier>();
>+
>+    auto pc =
>+        pc_factory_->CreatePeerConnection(config, std::move(dependencies));
>+    if (!pc) {
>+      return nullptr;
>+    }
>+
>+    return absl::make_unique<PeerConnectionWrapperForRampUpTest>(
>+        pc_factory_, pc, std::move(observer));
>+  }
>+
>+  void SetupOneWayCall() {
>+    ASSERT_TRUE(caller_);
>+    ASSERT_TRUE(callee_);
>+    FrameGeneratorCapturerVideoTrackSource::Config config;
>+    caller_->AddTrack(caller_->CreateLocalVideoTrack(config, clock_));
>+    // Disable highpass filter so that we can get all the test audio frames.
>+    cricket::AudioOptions options;
>+    options.highpass_filter = false;
>+    caller_->AddTrack(caller_->CreateLocalAudioTrack(options));
>+
>+    // Do the SDP negotiation, and also exchange ice candidates.
>+    ASSERT_TRUE(caller_->ExchangeOfferAnswerWith(callee_.get()));
>+    ASSERT_TRUE_WAIT(
>+        caller_->signaling_state() == PeerConnectionInterface::kStable,
>+        kDefaultTimeoutMs);
>+    ASSERT_TRUE_WAIT(caller_->IsIceGatheringDone(), kDefaultTimeoutMs);
>+    ASSERT_TRUE_WAIT(callee_->IsIceGatheringDone(), kDefaultTimeoutMs);
>+
>+    // Connect an ICE candidate pairs.
>+    ASSERT_TRUE(
>+        callee_->AddIceCandidates(caller_->observer()->GetAllCandidates()));
>+    ASSERT_TRUE(
>+        caller_->AddIceCandidates(callee_->observer()->GetAllCandidates()));
>+    // This means that ICE and DTLS are connected.
>+    ASSERT_TRUE_WAIT(callee_->IsIceConnected(), kDefaultTimeoutMs);
>+    ASSERT_TRUE_WAIT(caller_->IsIceConnected(), kDefaultTimeoutMs);
>+  }
>+
>+  void CreateTurnServer(cricket::ProtocolType type,
>+                        const std::string& common_name = "test turn server") {
>+    rtc::Thread* thread = network_thread();
>+    std::unique_ptr<cricket::TestTurnServer> turn_server =
>+        network_thread_->Invoke<std::unique_ptr<cricket::TestTurnServer>>(
>+            RTC_FROM_HERE, [thread, type, common_name] {
>+              static const rtc::SocketAddress turn_server_internal_address{
>+                  kTurnInternalAddress, kTurnInternalPort};
>+              static const rtc::SocketAddress turn_server_external_address{
>+                  kTurnExternalAddress, kTurnExternalPort};
>+              return absl::make_unique<cricket::TestTurnServer>(
>+                  thread, turn_server_internal_address,
>+                  turn_server_external_address, type,
>+                  true /*ignore_bad_certs=*/, common_name);
>+            });
>+    turn_servers_.push_back(std::move(turn_server));
>+  }
>+
>+  // First runs the call for kRampUpTimeMs to ramp up the bandwidth estimate.
>+  // Then runs the test for the remaining test time, grabbing the bandwidth
>+  // estimation stat, every kPollIntervalTimeMs. When finished, averages the
>+  // bandwidth estimations and prints the bandwidth estimation result as a perf
>+  // metric.
>+  void RunTest(const std::string& test_string) {
>+    rtc::Thread::Current()->ProcessMessages(kRampUpTimeMs);
>+    int number_of_polls =
>+        (kDefaultTestTimeMs - kRampUpTimeMs) / kPollIntervalTimeMs;
>+    int total_bwe = 0;
>+    for (int i = 0; i < number_of_polls; ++i) {
>+      rtc::Thread::Current()->ProcessMessages(kPollIntervalTimeMs);
>+      total_bwe += static_cast<int>(GetCallerAvailableBitrateEstimate());
>+    }
>+    double average_bandwidth_estimate = total_bwe / number_of_polls;
>+    std::string value_description =
>+        "bwe_after_" + std::to_string(kDefaultTestTimeMs / 1000) + "_seconds";
>+    test::PrintResult("peerconnection_ramp_up_", test_string, value_description,
>+                      average_bandwidth_estimate, "bwe", false);
>+  }
>+
>+  rtc::Thread* network_thread() { return network_thread_.get(); }
>+
>+  rtc::FirewallSocketServer* firewall_socket_server() {
>+    return firewall_socket_server_.get();
>+  }
>+
>+  PeerConnectionWrapperForRampUpTest* caller() { return caller_.get(); }
>+
>+  PeerConnectionWrapperForRampUpTest* callee() { return callee_.get(); }
>+
>+ private:
>+  // Gets the caller's outgoing available bitrate from the stats. Returns 0 if
>+  // something went wrong. It takes the outgoing bitrate from the current
>+  // selected ICE candidate pair's stats.
>+  double GetCallerAvailableBitrateEstimate() {
>+    auto stats = caller_->GetStats();
>+    auto transport_stats = stats->GetStatsOfType<RTCTransportStats>();
>+    if (transport_stats.size() == 0u ||
>+        !transport_stats[0]->selected_candidate_pair_id.is_defined()) {
>+      return 0;
>+    }
>+    std::string selected_ice_id =
>+        transport_stats[0]->selected_candidate_pair_id.ValueToString();
>+    // Use the selected ICE candidate pair ID to get the appropriate ICE stats.
>+    const RTCIceCandidatePairStats ice_candidate_pair_stats =
>+        stats->Get(selected_ice_id)->cast_to<const RTCIceCandidatePairStats>();
>+    if (ice_candidate_pair_stats.available_outgoing_bitrate.is_defined()) {
>+      return *ice_candidate_pair_stats.available_outgoing_bitrate;
>+    }
>+    // We couldn't get the |available_outgoing_bitrate| for the active candidate
>+    // pair.
>+    return 0;
>+  }
>+
>+  Clock* const clock_;
>+  // The turn servers should be accessed & deleted on the network thread to
>+  // avoid a race with the socket read/write which occurs on the network thread.
>+  std::vector<std::unique_ptr<cricket::TestTurnServer>> turn_servers_;
>+  // |virtual_socket_server_| is used by |network_thread_| so it must be
>+  // destroyed later.
>+  // TODO(bugs.webrtc.org/7668): We would like to update the virtual network we
>+  // use for this test. VirtualSocketServer isn't ideal because:
>+  // 1) It uses the same queue & network capacity for both directions.
>+  // 2) VirtualSocketServer implements how the network bandwidth affects the
>+  //    send delay differently than the SimulatedNetwork, used by the
>+  //    FakeNetworkPipe. It would be ideal if all of levels of virtual
>+  //    networks used in testing were consistent.
>+  // We would also like to update this test to record the time to ramp up,
>+  // down, and back up (similar to in rampup_tests.cc). This is problematic with
>+  // the VirtualSocketServer. The first ramp down time is very noisy and the
>+  // second ramp up time can take up to 300 seconds, most likely due to a built
>+  // up queue.
>+  std::unique_ptr<rtc::VirtualSocketServer> virtual_socket_server_;
>+  std::unique_ptr<rtc::FirewallSocketServer> firewall_socket_server_;
>+  std::unique_ptr<rtc::Thread> network_thread_;
>+  std::unique_ptr<rtc::Thread> worker_thread_;
>+  // The |pc_factory| uses |network_thread_| & |worker_thread_|, so it must be
>+  // destroyed first.
>+  std::vector<std::unique_ptr<rtc::FakeNetworkManager>> fake_network_managers_;
>+  rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory_;
>+  std::unique_ptr<PeerConnectionWrapperForRampUpTest> caller_;
>+  std::unique_ptr<PeerConnectionWrapperForRampUpTest> callee_;
>+};
>+
>+TEST_F(PeerConnectionRampUpTest, TurnOverTCP) {
>+  CreateTurnServer(cricket::ProtocolType::PROTO_TCP);
>+  PeerConnectionInterface::IceServer ice_server;
>+  std::string ice_server_url = "turn:" + std::string(kTurnInternalAddress) +
>+                               ":" + std::to_string(kTurnInternalPort) +
>+                               "?transport=tcp";
>+  ice_server.urls.push_back(ice_server_url);
>+  ice_server.username = "test";
>+  ice_server.password = "test";
>+  PeerConnectionInterface::RTCConfiguration client_1_config;
>+  client_1_config.servers.push_back(ice_server);
>+  client_1_config.type = PeerConnectionInterface::kRelay;
>+  PeerConnectionInterface::RTCConfiguration client_2_config;
>+  client_2_config.servers.push_back(ice_server);
>+  client_2_config.type = PeerConnectionInterface::kRelay;
>+  ASSERT_TRUE(CreatePeerConnectionWrappers(client_1_config, client_2_config));
>+
>+  SetupOneWayCall();
>+  RunTest("turn_over_tcp");
>+}
>+
>+TEST_F(PeerConnectionRampUpTest, TurnOverUDP) {
>+  CreateTurnServer(cricket::ProtocolType::PROTO_UDP);
>+  PeerConnectionInterface::IceServer ice_server;
>+  std::string ice_server_url = "turn:" + std::string(kTurnInternalAddress) +
>+                               ":" + std::to_string(kTurnInternalPort);
>+
>+  ice_server.urls.push_back(ice_server_url);
>+  ice_server.username = "test";
>+  ice_server.password = "test";
>+  PeerConnectionInterface::RTCConfiguration client_1_config;
>+  client_1_config.servers.push_back(ice_server);
>+  client_1_config.type = PeerConnectionInterface::kRelay;
>+  PeerConnectionInterface::RTCConfiguration client_2_config;
>+  client_2_config.servers.push_back(ice_server);
>+  client_2_config.type = PeerConnectionInterface::kRelay;
>+  ASSERT_TRUE(CreatePeerConnectionWrappers(client_1_config, client_2_config));
>+
>+  SetupOneWayCall();
>+  RunTest("turn_over_udp");
>+}
>+
>+TEST_F(PeerConnectionRampUpTest, TurnOverTLS) {
>+  CreateTurnServer(cricket::ProtocolType::PROTO_TLS, kTurnInternalAddress);
>+  PeerConnectionInterface::IceServer ice_server;
>+  std::string ice_server_url = "turns:" + std::string(kTurnInternalAddress) +
>+                               ":" + std::to_string(kTurnInternalPort) +
>+                               "?transport=tcp";
>+  ice_server.urls.push_back(ice_server_url);
>+  ice_server.username = "test";
>+  ice_server.password = "test";
>+  PeerConnectionInterface::RTCConfiguration client_1_config;
>+  client_1_config.servers.push_back(ice_server);
>+  client_1_config.type = PeerConnectionInterface::kRelay;
>+  PeerConnectionInterface::RTCConfiguration client_2_config;
>+  client_2_config.servers.push_back(ice_server);
>+  client_2_config.type = PeerConnectionInterface::kRelay;
>+
>+  ASSERT_TRUE(CreatePeerConnectionWrappers(client_1_config, client_2_config));
>+
>+  SetupOneWayCall();
>+  RunTest("turn_over_tls");
>+}
>+
>+TEST_F(PeerConnectionRampUpTest, UDPPeerToPeer) {
>+  PeerConnectionInterface::RTCConfiguration client_1_config;
>+  client_1_config.tcp_candidate_policy =
>+      PeerConnection::kTcpCandidatePolicyDisabled;
>+  PeerConnectionInterface::RTCConfiguration client_2_config;
>+  client_2_config.tcp_candidate_policy =
>+      PeerConnection::kTcpCandidatePolicyDisabled;
>+  ASSERT_TRUE(CreatePeerConnectionWrappers(client_1_config, client_2_config));
>+
>+  SetupOneWayCall();
>+  RunTest("udp_peer_to_peer");
>+}
>+
>+TEST_F(PeerConnectionRampUpTest, TCPPeerToPeer) {
>+  firewall_socket_server()->set_udp_sockets_enabled(false);
>+  ASSERT_TRUE(CreatePeerConnectionWrappers(
>+      PeerConnectionInterface::RTCConfiguration(),
>+      PeerConnectionInterface::RTCConfiguration()));
>+
>+  SetupOneWayCall();
>+  RunTest("tcp_peer_to_peer");
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_rtp_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_rtp_unittest.cc
>index f418f7ee9f9..1e4bf620036 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_rtp_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_rtp_unittest.cc
>@@ -11,22 +11,27 @@
> #include <memory>
> #include <vector>
> 
>+#include "absl/memory/memory.h"
> #include "api/audio_codecs/builtin_audio_decoder_factory.h"
> #include "api/audio_codecs/builtin_audio_encoder_factory.h"
> #include "api/jsep.h"
> #include "api/mediastreaminterface.h"
> #include "api/peerconnectioninterface.h"
>+#include "api/video_codecs/builtin_video_decoder_factory.h"
>+#include "api/video_codecs/builtin_video_encoder_factory.h"
>+#include "pc/mediasession.h"
> #include "pc/mediastream.h"
> #include "pc/mediastreamtrack.h"
> #include "pc/peerconnectionwrapper.h"
>+#include "pc/sdputils.h"
> #include "pc/test/fakeaudiocapturemodule.h"
> #include "pc/test/mockpeerconnectionobservers.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/gunit.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/refcountedobject.h"
> #include "rtc_base/scoped_ref_ptr.h"
> #include "rtc_base/thread.h"
>+#include "system_wrappers/include/metrics_default.h"
> #include "test/gmock.h"
> 
> // This file contains tests for RTP Media API-related behavior of
>@@ -37,6 +42,7 @@ namespace webrtc {
> using RTCConfiguration = PeerConnectionInterface::RTCConfiguration;
> using ::testing::ElementsAre;
> using ::testing::UnorderedElementsAre;
>+using ::testing::Values;
> 
> const uint32_t kDefaultTimeout = 10000u;
> 
>@@ -57,305 +63,568 @@ class OnSuccessObserver : public rtc::RefCountedObject<
>   MethodFunctor on_success_;
> };
> 
>-class PeerConnectionRtpTest : public testing::Test {
>+class PeerConnectionRtpBaseTest : public testing::Test {
>  public:
>-  PeerConnectionRtpTest()
>-      : pc_factory_(
>+  explicit PeerConnectionRtpBaseTest(SdpSemantics sdp_semantics)
>+      : sdp_semantics_(sdp_semantics),
>+        pc_factory_(
>             CreatePeerConnectionFactory(rtc::Thread::Current(),
>                                         rtc::Thread::Current(),
>                                         rtc::Thread::Current(),
>                                         FakeAudioCaptureModule::Create(),
>                                         CreateBuiltinAudioEncoderFactory(),
>                                         CreateBuiltinAudioDecoderFactory(),
>-                                        nullptr,
>-                                        nullptr)) {}
>+                                        CreateBuiltinVideoEncoderFactory(),
>+                                        CreateBuiltinVideoDecoderFactory(),
>+                                        nullptr /* audio_mixer */,
>+                                        nullptr /* audio_processing */)) {
>+    webrtc::metrics::Reset();
>+  }
> 
>   std::unique_ptr<PeerConnectionWrapper> CreatePeerConnection() {
>     return CreatePeerConnection(RTCConfiguration());
>   }
> 
>+  std::unique_ptr<PeerConnectionWrapper> CreatePeerConnectionWithPlanB() {
>+    RTCConfiguration config;
>+    config.sdp_semantics = SdpSemantics::kPlanB;
>+    return CreatePeerConnectionInternal(config);
>+  }
>+
>   std::unique_ptr<PeerConnectionWrapper> CreatePeerConnectionWithUnifiedPlan() {
>     RTCConfiguration config;
>     config.sdp_semantics = SdpSemantics::kUnifiedPlan;
>-    return CreatePeerConnection(config);
>+    return CreatePeerConnectionInternal(config);
>   }
> 
>   std::unique_ptr<PeerConnectionWrapper> CreatePeerConnection(
>       const RTCConfiguration& config) {
>-    auto observer = rtc::MakeUnique<MockPeerConnectionObserver>();
>+    RTCConfiguration modified_config = config;
>+    modified_config.sdp_semantics = sdp_semantics_;
>+    return CreatePeerConnectionInternal(modified_config);
>+  }
>+
>+ protected:
>+  const SdpSemantics sdp_semantics_;
>+  rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory_;
>+
>+ private:
>+  // Private so that tests don't accidentally bypass the SdpSemantics
>+  // adjustment.
>+  std::unique_ptr<PeerConnectionWrapper> CreatePeerConnectionInternal(
>+      const RTCConfiguration& config) {
>+    auto observer = absl::make_unique<MockPeerConnectionObserver>();
>     auto pc = pc_factory_->CreatePeerConnection(config, nullptr, nullptr,
>                                                 observer.get());
>-    return rtc::MakeUnique<PeerConnectionWrapper>(pc_factory_, pc,
>-                                                  std::move(observer));
>+    return absl::make_unique<PeerConnectionWrapper>(pc_factory_, pc,
>+                                                    std::move(observer));
>   }
>+};
> 
>+class PeerConnectionRtpTest
>+    : public PeerConnectionRtpBaseTest,
>+      public ::testing::WithParamInterface<SdpSemantics> {
>  protected:
>-  rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory_;
>+  PeerConnectionRtpTest() : PeerConnectionRtpBaseTest(GetParam()) {}
>+};
>+
>+class PeerConnectionRtpTestPlanB : public PeerConnectionRtpBaseTest {
>+ protected:
>+  PeerConnectionRtpTestPlanB()
>+      : PeerConnectionRtpBaseTest(SdpSemantics::kPlanB) {}
>+};
>+
>+class PeerConnectionRtpTestUnifiedPlan : public PeerConnectionRtpBaseTest {
>+ protected:
>+  PeerConnectionRtpTestUnifiedPlan()
>+      : PeerConnectionRtpBaseTest(SdpSemantics::kUnifiedPlan) {}
> };
> 
> // These tests cover |webrtc::PeerConnectionObserver| callbacks firing upon
> // setting the remote description.
>-class PeerConnectionRtpCallbacksTest : public PeerConnectionRtpTest {};
> 
>-TEST_F(PeerConnectionRtpCallbacksTest, AddTrackWithoutStreamFiresOnAddTrack) {
>+TEST_P(PeerConnectionRtpTest, AddTrackWithoutStreamFiresOnAddTrack) {
>   auto caller = CreatePeerConnection();
>   auto callee = CreatePeerConnection();
> 
>-  ASSERT_TRUE(caller->AddTrack(caller->CreateAudioTrack("audio_track")));
>-  ASSERT_TRUE(
>-      callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
>-                                   static_cast<webrtc::RTCError*>(nullptr)));
>+  ASSERT_TRUE(caller->AddAudioTrack("audio_track"));
>+  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
> 
>   ASSERT_EQ(callee->observer()->add_track_events_.size(), 1u);
>-  // TODO(hbos): When "no stream" is handled correctly we would expect
>-  // |add_track_events_[0].streams| to be empty. https://crbug.com/webrtc/7933
>-  auto& add_track_event = callee->observer()->add_track_events_[0];
>-  ASSERT_EQ(add_track_event.streams.size(), 1u);
>-  EXPECT_TRUE(add_track_event.streams[0]->FindAudioTrack("audio_track"));
>+  const auto& add_track_event = callee->observer()->add_track_events_[0];
>   EXPECT_EQ(add_track_event.streams, add_track_event.receiver->streams());
>+
>+  if (sdp_semantics_ == SdpSemantics::kPlanB) {
>+    // Since we are not supporting the no stream case with Plan B, there should
>+    // be a generated stream, even though we didn't set one with AddTrack.
>+    ASSERT_EQ(1u, add_track_event.streams.size());
>+    EXPECT_TRUE(add_track_event.streams[0]->FindAudioTrack("audio_track"));
>+  } else {
>+    EXPECT_EQ(0u, add_track_event.streams.size());
>+  }
> }
> 
>-TEST_F(PeerConnectionRtpCallbacksTest, AddTrackWithStreamFiresOnAddTrack) {
>+TEST_P(PeerConnectionRtpTest, AddTrackWithStreamFiresOnAddTrack) {
>   auto caller = CreatePeerConnection();
>   auto callee = CreatePeerConnection();
> 
>-  ASSERT_TRUE(caller->AddTrack(caller->CreateAudioTrack("audio_track"),
>-                               {"audio_stream"}));
>-  ASSERT_TRUE(
>-      callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
>-                                   static_cast<webrtc::RTCError*>(nullptr)));
>+  ASSERT_TRUE(caller->AddAudioTrack("audio_track", {"audio_stream"}));
>+  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
> 
>   ASSERT_EQ(callee->observer()->add_track_events_.size(), 1u);
>   auto& add_track_event = callee->observer()->add_track_events_[0];
>   ASSERT_EQ(add_track_event.streams.size(), 1u);
>-  EXPECT_EQ("audio_stream", add_track_event.streams[0]->label());
>+  EXPECT_EQ("audio_stream", add_track_event.streams[0]->id());
>   EXPECT_TRUE(add_track_event.streams[0]->FindAudioTrack("audio_track"));
>   EXPECT_EQ(add_track_event.streams, add_track_event.receiver->streams());
> }
> 
>-TEST_F(PeerConnectionRtpCallbacksTest,
>-       RemoveTrackWithoutStreamFiresOnRemoveTrack) {
>+TEST_P(PeerConnectionRtpTest, RemoveTrackWithoutStreamFiresOnRemoveTrack) {
>   auto caller = CreatePeerConnection();
>   auto callee = CreatePeerConnection();
> 
>-  auto sender = caller->AddTrack(caller->CreateAudioTrack("audio_track"), {});
>-  ASSERT_TRUE(
>-      callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
>-                                   static_cast<webrtc::RTCError*>(nullptr)));
>+  auto sender = caller->AddAudioTrack("audio_track", {});
>+  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
>   ASSERT_EQ(callee->observer()->add_track_events_.size(), 1u);
>-  EXPECT_TRUE(caller->pc()->RemoveTrack(sender));
>   ASSERT_TRUE(
>-      callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
>-                                   static_cast<webrtc::RTCError*>(nullptr)));
>+      caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));
>+
>+  EXPECT_TRUE(caller->pc()->RemoveTrack(sender));
>+  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
> 
>   ASSERT_EQ(callee->observer()->add_track_events_.size(), 1u);
>   EXPECT_EQ(callee->observer()->GetAddTrackReceivers(),
>             callee->observer()->remove_track_events_);
> }
> 
>-TEST_F(PeerConnectionRtpCallbacksTest,
>-       RemoveTrackWithStreamFiresOnRemoveTrack) {
>+TEST_P(PeerConnectionRtpTest, RemoveTrackWithStreamFiresOnRemoveTrack) {
>   auto caller = CreatePeerConnection();
>   auto callee = CreatePeerConnection();
> 
>-  auto sender = caller->AddTrack(caller->CreateAudioTrack("audio_track"),
>-                                 {"audio_stream"});
>-  ASSERT_TRUE(
>-      callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
>-                                   static_cast<webrtc::RTCError*>(nullptr)));
>+  auto sender = caller->AddAudioTrack("audio_track", {"audio_stream"});
>+  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
>   ASSERT_EQ(callee->observer()->add_track_events_.size(), 1u);
>-  EXPECT_TRUE(caller->pc()->RemoveTrack(sender));
>   ASSERT_TRUE(
>-      callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
>-                                   static_cast<webrtc::RTCError*>(nullptr)));
>+      caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));
>+
>+  EXPECT_TRUE(caller->pc()->RemoveTrack(sender));
>+  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
> 
>   ASSERT_EQ(callee->observer()->add_track_events_.size(), 1u);
>   EXPECT_EQ(callee->observer()->GetAddTrackReceivers(),
>             callee->observer()->remove_track_events_);
>+  EXPECT_EQ(0u, callee->observer()->remote_streams()->count());
> }
> 
>-TEST_F(PeerConnectionRtpCallbacksTest,
>-       RemoveTrackWithSharedStreamFiresOnRemoveTrack) {
>+TEST_P(PeerConnectionRtpTest, RemoveTrackWithSharedStreamFiresOnRemoveTrack) {
>   auto caller = CreatePeerConnection();
>   auto callee = CreatePeerConnection();
> 
>-  const char kSharedStreamLabel[] = "shared_audio_stream";
>-  auto sender1 = caller->AddTrack(caller->CreateAudioTrack("audio_track1"),
>-                                  {kSharedStreamLabel});
>-  auto sender2 = caller->AddTrack(caller->CreateAudioTrack("audio_track2"),
>-                                  {kSharedStreamLabel});
>-  ASSERT_TRUE(
>-      callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
>-                                   static_cast<webrtc::RTCError*>(nullptr)));
>-
>+  const char kSharedStreamId[] = "shared_audio_stream";
>+  auto sender1 = caller->AddAudioTrack("audio_track1", {kSharedStreamId});
>+  auto sender2 = caller->AddAudioTrack("audio_track2", {kSharedStreamId});
>+  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
>   ASSERT_EQ(callee->observer()->add_track_events_.size(), 2u);
>+  ASSERT_TRUE(
>+      caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));
> 
>   // Remove "audio_track1".
>   EXPECT_TRUE(caller->pc()->RemoveTrack(sender1));
>-  ASSERT_TRUE(
>-      callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
>-                                   static_cast<webrtc::RTCError*>(nullptr)));
>+  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
>   ASSERT_EQ(callee->observer()->add_track_events_.size(), 2u);
>   EXPECT_EQ(
>       std::vector<rtc::scoped_refptr<RtpReceiverInterface>>{
>           callee->observer()->add_track_events_[0].receiver},
>       callee->observer()->remove_track_events_);
>+  ASSERT_EQ(1u, callee->observer()->remote_streams()->count());
>+  ASSERT_TRUE(
>+      caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));
> 
>   // Remove "audio_track2".
>   EXPECT_TRUE(caller->pc()->RemoveTrack(sender2));
>-  ASSERT_TRUE(
>-      callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
>-                                   static_cast<webrtc::RTCError*>(nullptr)));
>+  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
>   ASSERT_EQ(callee->observer()->add_track_events_.size(), 2u);
>   EXPECT_EQ(callee->observer()->GetAddTrackReceivers(),
>             callee->observer()->remove_track_events_);
>+  EXPECT_EQ(0u, callee->observer()->remote_streams()->count());
>+}
>+
>+// Tests the edge case that if a stream ID changes for a given track that both
>+// OnRemoveTrack and OnAddTrack is fired.
>+TEST_F(PeerConnectionRtpTestPlanB,
>+       RemoteStreamIdChangesFiresOnRemoveAndOnAddTrack) {
>+  auto caller = CreatePeerConnection();
>+  auto callee = CreatePeerConnection();
>+
>+  const char kStreamId1[] = "stream1";
>+  const char kStreamId2[] = "stream2";
>+  caller->AddAudioTrack("audio_track1", {kStreamId1});
>+  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
>+  EXPECT_EQ(callee->observer()->add_track_events_.size(), 1u);
>+
>+  // Change the stream ID of the sender in the session description.
>+  auto offer = caller->CreateOfferAndSetAsLocal();
>+  auto* audio_desc =
>+      cricket::GetFirstAudioContentDescription(offer->description());
>+  ASSERT_EQ(audio_desc->mutable_streams().size(), 1u);
>+  audio_desc->mutable_streams()[0].set_stream_ids({kStreamId2});
>+  ASSERT_TRUE(callee->SetRemoteDescription(std::move(offer)));
>+
>+  ASSERT_EQ(callee->observer()->add_track_events_.size(), 2u);
>+  EXPECT_EQ(callee->observer()->add_track_events_[1].streams[0]->id(),
>+            kStreamId2);
>+  ASSERT_EQ(callee->observer()->remove_track_events_.size(), 1u);
>+  EXPECT_EQ(callee->observer()->remove_track_events_[0]->streams()[0]->id(),
>+            kStreamId1);
>+}
>+
>+// Tests that setting a remote description with sending transceivers will fire
>+// the OnTrack callback for each transceiver and setting a remote description
>+// with receive only transceivers will not call OnTrack. One transceiver is
>+// created without any stream_ids, while the other is created with multiple
>+// stream_ids.
>+TEST_F(PeerConnectionRtpTestUnifiedPlan, AddTransceiverCallsOnTrack) {
>+  const std::string kStreamId1 = "video_stream1";
>+  const std::string kStreamId2 = "video_stream2";
>+  auto caller = CreatePeerConnection();
>+  auto callee = CreatePeerConnection();
>+
>+  auto audio_transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
>+  RtpTransceiverInit video_transceiver_init;
>+  video_transceiver_init.stream_ids = {kStreamId1, kStreamId2};
>+  auto video_transceiver =
>+      caller->AddTransceiver(cricket::MEDIA_TYPE_VIDEO, video_transceiver_init);
>+
>+  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
>+
>+  ASSERT_EQ(0u, caller->observer()->on_track_transceivers_.size());
>+  ASSERT_EQ(2u, callee->observer()->on_track_transceivers_.size());
>+  EXPECT_EQ(audio_transceiver->mid(),
>+            callee->pc()->GetTransceivers()[0]->mid());
>+  EXPECT_EQ(video_transceiver->mid(),
>+            callee->pc()->GetTransceivers()[1]->mid());
>+  std::vector<rtc::scoped_refptr<MediaStreamInterface>> audio_streams =
>+      callee->pc()->GetTransceivers()[0]->receiver()->streams();
>+  std::vector<rtc::scoped_refptr<MediaStreamInterface>> video_streams =
>+      callee->pc()->GetTransceivers()[1]->receiver()->streams();
>+  ASSERT_EQ(0u, audio_streams.size());
>+  ASSERT_EQ(2u, video_streams.size());
>+  EXPECT_EQ(kStreamId1, video_streams[0]->id());
>+  EXPECT_EQ(kStreamId2, video_streams[1]->id());
>+}
>+
>+// Test that doing additional offer/answer exchanges with no changes to tracks
>+// will cause no additional OnTrack calls after the tracks have been negotiated.
>+TEST_F(PeerConnectionRtpTestUnifiedPlan, ReofferDoesNotCallOnTrack) {
>+  auto caller = CreatePeerConnection();
>+  auto callee = CreatePeerConnection();
>+
>+  caller->AddAudioTrack("audio");
>+  callee->AddAudioTrack("audio");
>+
>+  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
>+  EXPECT_EQ(1u, caller->observer()->on_track_transceivers_.size());
>+  EXPECT_EQ(1u, callee->observer()->on_track_transceivers_.size());
>+
>+  // If caller reoffers with no changes expect no additional OnTrack calls.
>+  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
>+  EXPECT_EQ(1u, caller->observer()->on_track_transceivers_.size());
>+  EXPECT_EQ(1u, callee->observer()->on_track_transceivers_.size());
>+
>+  // Also if callee reoffers with no changes expect no additional OnTrack calls.
>+  ASSERT_TRUE(callee->ExchangeOfferAnswerWith(caller.get()));
>+  EXPECT_EQ(1u, caller->observer()->on_track_transceivers_.size());
>+  EXPECT_EQ(1u, callee->observer()->on_track_transceivers_.size());
>+}
>+
>+// Test that OnTrack is called when the transceiver direction changes to send
>+// the track.
>+TEST_F(PeerConnectionRtpTestUnifiedPlan, SetDirectionCallsOnTrack) {
>+  auto caller = CreatePeerConnection();
>+  auto callee = CreatePeerConnection();
>+
>+  auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
>+  transceiver->SetDirection(RtpTransceiverDirection::kInactive);
>+  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
>+  EXPECT_EQ(0u, caller->observer()->on_track_transceivers_.size());
>+  EXPECT_EQ(0u, callee->observer()->on_track_transceivers_.size());
>+
>+  transceiver->SetDirection(RtpTransceiverDirection::kSendOnly);
>+  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
>+  EXPECT_EQ(0u, caller->observer()->on_track_transceivers_.size());
>+  EXPECT_EQ(1u, callee->observer()->on_track_transceivers_.size());
>+
>+  // If the direction changes but it is still receiving on the remote side, then
>+  // OnTrack should not be fired again.
>+  transceiver->SetDirection(RtpTransceiverDirection::kSendRecv);
>+  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
>+  EXPECT_EQ(0u, caller->observer()->on_track_transceivers_.size());
>+  EXPECT_EQ(1u, callee->observer()->on_track_transceivers_.size());
>+}
>+
>+// Test that OnTrack is called twice when a sendrecv call is started, the callee
>+// changes the direction to inactive, then changes it back to sendrecv.
>+TEST_F(PeerConnectionRtpTestUnifiedPlan, SetDirectionHoldCallsOnTrackTwice) {
>+  auto caller = CreatePeerConnection();
>+  auto callee = CreatePeerConnection();
>+
>+  auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
>+
>+  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
>+  EXPECT_EQ(0u, caller->observer()->on_track_transceivers_.size());
>+  EXPECT_EQ(1u, callee->observer()->on_track_transceivers_.size());
>+
>+  // Put the call on hold by no longer receiving the track.
>+  callee->pc()->GetTransceivers()[0]->SetDirection(
>+      RtpTransceiverDirection::kInactive);
>+
>+  ASSERT_TRUE(callee->ExchangeOfferAnswerWith(caller.get()));
>+  EXPECT_EQ(0u, caller->observer()->on_track_transceivers_.size());
>+  EXPECT_EQ(1u, callee->observer()->on_track_transceivers_.size());
>+
>+  // Resume the call by changing the direction to recvonly. This should call
>+  // OnTrack again on the callee side.
>+  callee->pc()->GetTransceivers()[0]->SetDirection(
>+      RtpTransceiverDirection::kRecvOnly);
>+
>+  ASSERT_TRUE(callee->ExchangeOfferAnswerWith(caller.get()));
>+  EXPECT_EQ(0u, caller->observer()->on_track_transceivers_.size());
>+  EXPECT_EQ(2u, callee->observer()->on_track_transceivers_.size());
>+}
>+
>+// Test that setting a remote offer twice with no answer in the middle results
>+// in OnAddTrack being fired only once.
>+TEST_F(PeerConnectionRtpTestUnifiedPlan,
>+       ApplyTwoRemoteOffersWithNoAnswerResultsInOneAddTrackEvent) {
>+  auto caller = CreatePeerConnection();
>+  auto callee = CreatePeerConnection();
>+
>+  caller->AddAudioTrack("audio_track", {});
>+
>+  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOffer()));
>+  ASSERT_EQ(1u, callee->observer()->add_track_events_.size());
>+
>+  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOffer()));
>+  EXPECT_EQ(1u, callee->observer()->add_track_events_.size());
>+}
>+
>+// Test that setting a remote offer twice with no answer in the middle and the
>+// track being removed between the two offers results in OnAddTrack being called
>+// once the first time and OnRemoveTrack being called once the second time.
>+TEST_F(PeerConnectionRtpTestUnifiedPlan,
>+       ApplyRemoteOfferAddThenRemoteOfferRemoveResultsInOneRemoveTrackEvent) {
>+  auto caller = CreatePeerConnection();
>+  auto callee = CreatePeerConnection();
>+
>+  auto sender = caller->AddAudioTrack("audio_track", {});
>+
>+  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOffer()));
>+  ASSERT_EQ(1u, callee->observer()->add_track_events_.size());
>+  EXPECT_EQ(0u, callee->observer()->remove_track_events_.size());
>+
>+  caller->pc()->RemoveTrack(sender);
>+
>+  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOffer()));
>+  EXPECT_EQ(1u, callee->observer()->add_track_events_.size());
>+  EXPECT_EQ(1u, callee->observer()->remove_track_events_.size());
>+}
>+
>+// Test that changing the direction from receiving to not receiving between
>+// setting the remote offer and creating / setting the local answer results in
>+// a remove track event when SetLocalDescription is called.
>+TEST_F(PeerConnectionRtpTestUnifiedPlan,
>+       ChangeDirectionInAnswerResultsInRemoveTrackEvent) {
>+  auto caller = CreatePeerConnection();
>+  caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
>+  auto callee = CreatePeerConnection();
>+  callee->AddAudioTrack("audio_track", {});
>+
>+  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOffer()));
>+  EXPECT_EQ(1u, callee->observer()->add_track_events_.size());
>+  EXPECT_EQ(0u, callee->observer()->remove_track_events_.size());
>+
>+  auto callee_transceiver = callee->pc()->GetTransceivers()[0];
>+  callee_transceiver->SetDirection(RtpTransceiverDirection::kSendOnly);
>+
>+  ASSERT_TRUE(callee->SetLocalDescription(callee->CreateAnswer()));
>+  EXPECT_EQ(1u, callee->observer()->add_track_events_.size());
>+  EXPECT_EQ(1u, callee->observer()->remove_track_events_.size());
> }
> 
> // These tests examine the state of the peer connection as a result of
> // performing SetRemoteDescription().
>-class PeerConnectionRtpObserverTest : public PeerConnectionRtpTest {};
> 
>-TEST_F(PeerConnectionRtpObserverTest, AddSenderWithoutStreamAddsReceiver) {
>+TEST_P(PeerConnectionRtpTest, AddTrackWithoutStreamAddsReceiver) {
>   auto caller = CreatePeerConnection();
>   auto callee = CreatePeerConnection();
> 
>-  ASSERT_TRUE(caller->AddTrack(caller->CreateAudioTrack("audio_track"), {}));
>-  ASSERT_TRUE(
>-      callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
>-                                   static_cast<webrtc::RTCError*>(nullptr)));
>+  ASSERT_TRUE(caller->AddAudioTrack("audio_track", {}));
>+  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
> 
>   EXPECT_EQ(callee->pc()->GetReceivers().size(), 1u);
>   auto receiver_added = callee->pc()->GetReceivers()[0];
>   EXPECT_EQ("audio_track", receiver_added->track()->id());
>-  // TODO(hbos): When "no stream" is handled correctly we would expect
>-  // |receiver_added->streams()| to be empty. https://crbug.com/webrtc/7933
>-  EXPECT_EQ(receiver_added->streams().size(), 1u);
>-  EXPECT_TRUE(receiver_added->streams()[0]->FindAudioTrack("audio_track"));
>+
>+  if (sdp_semantics_ == SdpSemantics::kPlanB) {
>+    // Since we are not supporting the no stream case with Plan B, there should
>+    // be a generated stream, even though we didn't set one with AddTrack.
>+    ASSERT_EQ(1u, receiver_added->streams().size());
>+    EXPECT_TRUE(receiver_added->streams()[0]->FindAudioTrack("audio_track"));
>+  } else {
>+    EXPECT_EQ(0u, receiver_added->streams().size());
>+  }
> }
> 
>-TEST_F(PeerConnectionRtpObserverTest, AddSenderWithStreamAddsReceiver) {
>+TEST_P(PeerConnectionRtpTest, AddTrackWithStreamAddsReceiver) {
>   auto caller = CreatePeerConnection();
>   auto callee = CreatePeerConnection();
> 
>-  ASSERT_TRUE(caller->AddTrack(caller->CreateAudioTrack("audio_track"),
>-                               {"audio_stream"}));
>-  ASSERT_TRUE(
>-      callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
>-                                   static_cast<webrtc::RTCError*>(nullptr)));
>+  ASSERT_TRUE(caller->AddAudioTrack("audio_track", {"audio_stream"}));
>+  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
> 
>   EXPECT_EQ(callee->pc()->GetReceivers().size(), 1u);
>   auto receiver_added = callee->pc()->GetReceivers()[0];
>   EXPECT_EQ("audio_track", receiver_added->track()->id());
>   EXPECT_EQ(receiver_added->streams().size(), 1u);
>-  EXPECT_EQ("audio_stream", receiver_added->streams()[0]->label());
>+  EXPECT_EQ("audio_stream", receiver_added->streams()[0]->id());
>   EXPECT_TRUE(receiver_added->streams()[0]->FindAudioTrack("audio_track"));
> }
> 
>-TEST_F(PeerConnectionRtpObserverTest,
>-       RemoveSenderWithoutStreamRemovesReceiver) {
>+TEST_P(PeerConnectionRtpTest, RemoveTrackWithoutStreamRemovesReceiver) {
>   auto caller = CreatePeerConnection();
>   auto callee = CreatePeerConnection();
> 
>-  auto sender = caller->AddTrack(caller->CreateAudioTrack("audio_track"), {});
>+  auto sender = caller->AddAudioTrack("audio_track", {});
>   ASSERT_TRUE(sender);
>-  ASSERT_TRUE(
>-      callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
>-                                   static_cast<webrtc::RTCError*>(nullptr)));
>+  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
>+
>   ASSERT_EQ(callee->pc()->GetReceivers().size(), 1u);
>   auto receiver = callee->pc()->GetReceivers()[0];
>   ASSERT_TRUE(caller->pc()->RemoveTrack(sender));
>-  ASSERT_TRUE(
>-      callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
>-                                   static_cast<webrtc::RTCError*>(nullptr)));
>-
>-  // TODO(hbos): When we implement Unified Plan, receivers will not be removed.
>-  // Instead, the transceiver owning the receiver will become inactive.
>-  EXPECT_EQ(callee->pc()->GetReceivers().size(), 0u);
>+  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
>+
>+  if (sdp_semantics_ == SdpSemantics::kUnifiedPlan) {
>+    // With Unified Plan the receiver stays but the transceiver transitions to
>+    // inactive.
>+    ASSERT_EQ(1u, callee->pc()->GetReceivers().size());
>+    EXPECT_EQ(RtpTransceiverDirection::kInactive,
>+              callee->pc()->GetTransceivers()[0]->current_direction());
>+  } else {
>+    // With Plan B the receiver is removed.
>+    ASSERT_EQ(0u, callee->pc()->GetReceivers().size());
>+  }
> }
> 
>-TEST_F(PeerConnectionRtpObserverTest, RemoveSenderWithStreamRemovesReceiver) {
>+TEST_P(PeerConnectionRtpTest, RemoveTrackWithStreamRemovesReceiver) {
>   auto caller = CreatePeerConnection();
>   auto callee = CreatePeerConnection();
> 
>-  auto sender = caller->AddTrack(caller->CreateAudioTrack("audio_track"),
>-                                 {"audio_stream"});
>+  auto sender = caller->AddAudioTrack("audio_track", {"audio_stream"});
>   ASSERT_TRUE(sender);
>-  ASSERT_TRUE(
>-      callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
>-                                   static_cast<webrtc::RTCError*>(nullptr)));
>+  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
>   ASSERT_EQ(callee->pc()->GetReceivers().size(), 1u);
>   auto receiver = callee->pc()->GetReceivers()[0];
>   ASSERT_TRUE(caller->pc()->RemoveTrack(sender));
>-  ASSERT_TRUE(
>-      callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
>-                                   static_cast<webrtc::RTCError*>(nullptr)));
>-
>-  // TODO(hbos): When we implement Unified Plan, receivers will not be removed.
>-  // Instead, the transceiver owning the receiver will become inactive.
>-  EXPECT_EQ(callee->pc()->GetReceivers().size(), 0u);
>+  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
>+
>+  if (sdp_semantics_ == SdpSemantics::kUnifiedPlan) {
>+    // With Unified Plan the receiver stays but the transceiver transitions to
>+    // inactive.
>+    EXPECT_EQ(1u, callee->pc()->GetReceivers().size());
>+    EXPECT_EQ(RtpTransceiverDirection::kInactive,
>+              callee->pc()->GetTransceivers()[0]->current_direction());
>+  } else {
>+    // With Plan B the receiver is removed.
>+    EXPECT_EQ(0u, callee->pc()->GetReceivers().size());
>+  }
> }
> 
>-TEST_F(PeerConnectionRtpObserverTest,
>-       RemoveSenderWithSharedStreamRemovesReceiver) {
>+TEST_P(PeerConnectionRtpTest, RemoveTrackWithSharedStreamRemovesReceiver) {
>   auto caller = CreatePeerConnection();
>   auto callee = CreatePeerConnection();
> 
>-  const char kSharedStreamLabel[] = "shared_audio_stream";
>-  auto sender1 = caller->AddTrack(caller->CreateAudioTrack("audio_track1"),
>-                                  {kSharedStreamLabel});
>-  auto sender2 = caller->AddTrack(caller->CreateAudioTrack("audio_track2"),
>-                                  {kSharedStreamLabel});
>-  ASSERT_TRUE(
>-      callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
>-                                   static_cast<webrtc::RTCError*>(nullptr)));
>-
>-  ASSERT_EQ(callee->pc()->GetReceivers().size(), 2u);
>-  rtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver1;
>-  rtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver2;
>-  if (callee->pc()->GetReceivers()[0]->track()->id() == "audio_track1") {
>-    receiver1 = callee->pc()->GetReceivers()[0];
>-    receiver2 = callee->pc()->GetReceivers()[1];
>-  } else {
>-    receiver1 = callee->pc()->GetReceivers()[1];
>-    receiver2 = callee->pc()->GetReceivers()[0];
>-  }
>-  EXPECT_EQ("audio_track1", receiver1->track()->id());
>-  EXPECT_EQ("audio_track2", receiver2->track()->id());
>+  const char kSharedStreamId[] = "shared_audio_stream";
>+  auto sender1 = caller->AddAudioTrack("audio_track1", {kSharedStreamId});
>+  auto sender2 = caller->AddAudioTrack("audio_track2", {kSharedStreamId});
>+  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
>+  ASSERT_EQ(2u, callee->pc()->GetReceivers().size());
> 
>   // Remove "audio_track1".
>   EXPECT_TRUE(caller->pc()->RemoveTrack(sender1));
>-  ASSERT_TRUE(
>-      callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
>-                                   static_cast<webrtc::RTCError*>(nullptr)));
>-  // Only |receiver2| should remain.
>-  // TODO(hbos): When we implement Unified Plan, receivers will not be removed.
>-  // Instead, the transceiver owning the receiver will become inactive.
>-  EXPECT_EQ(
>-      std::vector<rtc::scoped_refptr<webrtc::RtpReceiverInterface>>{receiver2},
>-      callee->pc()->GetReceivers());
>+  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
>+
>+  if (sdp_semantics_ == SdpSemantics::kUnifiedPlan) {
>+    // With Unified Plan the receiver stays but the transceiver transitions to
>+    // inactive.
>+    ASSERT_EQ(2u, callee->pc()->GetReceivers().size());
>+    auto transceiver = callee->pc()->GetTransceivers()[0];
>+    EXPECT_EQ("audio_track1", transceiver->receiver()->track()->id());
>+    EXPECT_EQ(RtpTransceiverDirection::kInactive,
>+              transceiver->current_direction());
>+  } else {
>+    // With Plan B the receiver is removed.
>+    ASSERT_EQ(1u, callee->pc()->GetReceivers().size());
>+    EXPECT_EQ("audio_track2", callee->pc()->GetReceivers()[0]->track()->id());
>+  }
> 
>   // Remove "audio_track2".
>   EXPECT_TRUE(caller->pc()->RemoveTrack(sender2));
>-  ASSERT_TRUE(
>-      callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal(),
>-                                   static_cast<webrtc::RTCError*>(nullptr)));
>-  // TODO(hbos): When we implement Unified Plan, receivers will not be removed.
>-  // Instead, the transceiver owning the receiver will become inactive.
>-  EXPECT_EQ(callee->pc()->GetReceivers().size(), 0u);
>+  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
>+
>+  if (sdp_semantics_ == SdpSemantics::kUnifiedPlan) {
>+    // With Unified Plan the receiver stays but the transceiver transitions to
>+    // inactive.
>+    ASSERT_EQ(2u, callee->pc()->GetReceivers().size());
>+    auto transceiver = callee->pc()->GetTransceivers()[1];
>+    EXPECT_EQ("audio_track2", transceiver->receiver()->track()->id());
>+    EXPECT_EQ(RtpTransceiverDirection::kInactive,
>+              transceiver->current_direction());
>+  } else {
>+    // With Plan B the receiver is removed.
>+    ASSERT_EQ(0u, callee->pc()->GetReceivers().size());
>+  }
>+}
>+
>+TEST_P(PeerConnectionRtpTest, AudioGetParametersHasHeaderExtensions) {
>+  auto caller = CreatePeerConnection();
>+  auto callee = CreatePeerConnection();
>+  auto sender = caller->AddAudioTrack("audio_track");
>+  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
>+
>+  ASSERT_GT(caller->pc()->GetSenders().size(), 0u);
>+  EXPECT_GT(sender->GetParameters().header_extensions.size(), 0u);
>+
>+  ASSERT_GT(callee->pc()->GetReceivers().size(), 0u);
>+  auto receiver = callee->pc()->GetReceivers()[0];
>+  EXPECT_GT(receiver->GetParameters().header_extensions.size(), 0u);
>+}
>+
>+TEST_P(PeerConnectionRtpTest, VideoGetParametersHasHeaderExtensions) {
>+  auto caller = CreatePeerConnection();
>+  auto callee = CreatePeerConnection();
>+  auto sender = caller->AddVideoTrack("video_track");
>+  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
>+
>+  ASSERT_GT(caller->pc()->GetSenders().size(), 0u);
>+  EXPECT_GT(sender->GetParameters().header_extensions.size(), 0u);
>+
>+  ASSERT_GT(callee->pc()->GetReceivers().size(), 0u);
>+  auto receiver = callee->pc()->GetReceivers()[0];
>+  EXPECT_GT(receiver->GetParameters().header_extensions.size(), 0u);
> }
> 
> // Invokes SetRemoteDescription() twice in a row without synchronizing the two
> // calls and examine the state of the peer connection inside the callbacks to
> // ensure that the second call does not occur prematurely, contaminating the
> // state of the peer connection of the first callback.
>-TEST_F(PeerConnectionRtpObserverTest,
>+TEST_F(PeerConnectionRtpTestPlanB,
>        StatesCorrelateWithSetRemoteDescriptionCall) {
>   auto caller = CreatePeerConnection();
>   auto callee = CreatePeerConnection();
> 
>   // Create SDP for adding a track and for removing it. This will be used in the
>   // first and second SetRemoteDescription() calls.
>-  auto sender = caller->AddTrack(caller->CreateAudioTrack("audio_track"), {});
>+  auto sender = caller->AddAudioTrack("audio_track", {});
>   auto srd1_sdp = caller->CreateOfferAndSetAsLocal();
>   EXPECT_TRUE(caller->pc()->RemoveTrack(sender));
>   auto srd2_sdp = caller->CreateOfferAndSetAsLocal();
>@@ -397,11 +666,112 @@ TEST_F(PeerConnectionRtpObserverTest,
>   EXPECT_TRUE_WAIT(srd2_callback_called, kDefaultTimeout);
> }
> 
>+// Tests that a remote track is created with the signaled MSIDs when they are
>+// communicated with a=msid and no SSRCs are signaled at all (i.e., no a=ssrc
>+// lines).
>+TEST_F(PeerConnectionRtpTestUnifiedPlan, UnsignaledSsrcCreatesReceiverStreams) {
>+  auto caller = CreatePeerConnection();
>+  auto callee = CreatePeerConnection();
>+  const char kStreamId1[] = "stream1";
>+  const char kStreamId2[] = "stream2";
>+  caller->AddTrack(caller->CreateAudioTrack("audio_track1"),
>+                   {kStreamId1, kStreamId2});
>+
>+  auto offer = caller->CreateOfferAndSetAsLocal();
>+  // Munge the offer to take out everything but the stream_ids.
>+  auto contents = offer->description()->contents();
>+  ASSERT_TRUE(!contents.empty());
>+  ASSERT_TRUE(!contents[0].media_description()->streams().empty());
>+  std::vector<std::string> stream_ids =
>+      contents[0].media_description()->streams()[0].stream_ids();
>+  contents[0].media_description()->mutable_streams().clear();
>+  cricket::StreamParams new_stream;
>+  new_stream.set_stream_ids(stream_ids);
>+  contents[0].media_description()->AddStream(new_stream);
>+
>+  // Set the remote description and verify that the streams were added to the
>+  // receiver correctly.
>+  ASSERT_TRUE(
>+      callee->SetRemoteDescription(CloneSessionDescription(offer.get())));
>+  auto receivers = callee->pc()->GetReceivers();
>+  ASSERT_EQ(receivers.size(), 1u);
>+  ASSERT_EQ(receivers[0]->streams().size(), 2u);
>+  EXPECT_EQ(receivers[0]->streams()[0]->id(), kStreamId1);
>+  EXPECT_EQ(receivers[0]->streams()[1]->id(), kStreamId2);
>+}
>+
>+// Tests that with Unified Plan if the the stream id changes for a track when
>+// when setting a new remote description, that the media stream is updated
>+// appropriately for the receiver.
>+// TODO(https://github.com/w3c/webrtc-pc/issues/1937): Resolve spec issue or fix
>+// test.
>+TEST_F(PeerConnectionRtpTestUnifiedPlan,
>+       DISABLED_RemoteStreamIdChangesUpdatesReceiver) {
>+  auto caller = CreatePeerConnection();
>+  auto callee = CreatePeerConnection();
>+
>+  const char kStreamId1[] = "stream1";
>+  const char kStreamId2[] = "stream2";
>+  caller->AddAudioTrack("audio_track1", {kStreamId1});
>+  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
>+  EXPECT_EQ(callee->observer()->add_track_events_.size(), 1u);
>+
>+  // Change the stream id of the sender in the session description.
>+  auto offer = caller->CreateOfferAndSetAsLocal();
>+  auto contents = offer->description()->contents();
>+  ASSERT_EQ(contents.size(), 1u);
>+  ASSERT_EQ(contents[0].media_description()->mutable_streams().size(), 1u);
>+  contents[0].media_description()->mutable_streams()[0].set_stream_ids(
>+      {kStreamId2});
>+
>+  // Set the remote description and verify that the stream was updated
>+  // properly.
>+  ASSERT_TRUE(callee->SetRemoteDescription(std::move(offer)));
>+  auto receivers = callee->pc()->GetReceivers();
>+  ASSERT_EQ(receivers.size(), 1u);
>+  ASSERT_EQ(receivers[0]->streams().size(), 1u);
>+  EXPECT_EQ(receivers[0]->streams()[0]->id(), kStreamId2);
>+}
>+
>+// This tests a regression caught by a downstream client, that occured when
>+// applying a remote description with a SessionDescription object that
>+// contained StreamParams that didn't have ids. Although there were multiple
>+// remote audio senders, FindSenderInfo didn't find them as unique, because
>+// it looked up by StreamParam.id, which none had. This meant only one
>+// AudioRtpReceiver was created, as opposed to one for each remote sender.
>+TEST_F(PeerConnectionRtpTestPlanB,
>+       MultipleRemoteSendersWithoutStreamParamIdAddsMultipleReceivers) {
>+  auto caller = CreatePeerConnection();
>+  auto callee = CreatePeerConnection();
>+
>+  const char kStreamId1[] = "stream1";
>+  const char kStreamId2[] = "stream2";
>+  caller->AddAudioTrack("audio_track1", {kStreamId1});
>+  caller->AddAudioTrack("audio_track2", {kStreamId2});
>+
>+  auto offer = caller->CreateOfferAndSetAsLocal();
>+  auto mutable_streams =
>+      cricket::GetFirstAudioContentDescription(offer->description())
>+          ->mutable_streams();
>+  ASSERT_EQ(mutable_streams.size(), 2u);
>+  // Clear the IDs in the StreamParams.
>+  mutable_streams[0].id.clear();
>+  mutable_streams[1].id.clear();
>+  ASSERT_TRUE(
>+      callee->SetRemoteDescription(CloneSessionDescription(offer.get())));
>+
>+  auto receivers = callee->pc()->GetReceivers();
>+  ASSERT_EQ(receivers.size(), 2u);
>+  ASSERT_EQ(receivers[0]->streams().size(), 1u);
>+  EXPECT_EQ(kStreamId1, receivers[0]->streams()[0]->id());
>+  ASSERT_EQ(receivers[1]->streams().size(), 1u);
>+  EXPECT_EQ(kStreamId2, receivers[1]->streams()[0]->id());
>+}
>+
> // Tests for the legacy SetRemoteDescription() function signature.
>-class PeerConnectionRtpLegacyObserverTest : public PeerConnectionRtpTest {};
> 
> // Sanity test making sure the callback is invoked.
>-TEST_F(PeerConnectionRtpLegacyObserverTest, OnSuccess) {
>+TEST_P(PeerConnectionRtpTest, LegacyObserverOnSuccess) {
>   auto caller = CreatePeerConnection();
>   auto callee = CreatePeerConnection();
> 
>@@ -413,8 +783,8 @@ TEST_F(PeerConnectionRtpLegacyObserverTest, OnSuccess) {
> // Verifies legacy behavior: The observer is not called if if the peer
> // connection is destroyed because the asynchronous callback is executed in the
> // peer connection's message handler.
>-TEST_F(PeerConnectionRtpLegacyObserverTest,
>-       ObserverNotCalledIfPeerConnectionDereferenced) {
>+TEST_P(PeerConnectionRtpTest,
>+       LegacyObserverNotCalledIfPeerConnectionDereferenced) {
>   auto caller = CreatePeerConnection();
>   auto callee = CreatePeerConnection();
> 
>@@ -431,30 +801,32 @@ TEST_F(PeerConnectionRtpLegacyObserverTest,
> // RtpTransceiver Tests.
> 
> // Test that by default there are no transceivers with Unified Plan.
>-TEST_F(PeerConnectionRtpTest, PeerConnectionHasNoTransceivers) {
>-  auto caller = CreatePeerConnectionWithUnifiedPlan();
>+TEST_F(PeerConnectionRtpTestUnifiedPlan, PeerConnectionHasNoTransceivers) {
>+  auto caller = CreatePeerConnection();
>   EXPECT_THAT(caller->pc()->GetTransceivers(), ElementsAre());
> }
> 
> // Test that a transceiver created with the audio kind has the correct initial
> // properties.
>-TEST_F(PeerConnectionRtpTest, AddTransceiverHasCorrectInitProperties) {
>-  auto caller = CreatePeerConnectionWithUnifiedPlan();
>+TEST_F(PeerConnectionRtpTestUnifiedPlan,
>+       AddTransceiverHasCorrectInitProperties) {
>+  auto caller = CreatePeerConnection();
> 
>   auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
>-  EXPECT_EQ(rtc::nullopt, transceiver->mid());
>+  EXPECT_EQ(absl::nullopt, transceiver->mid());
>   EXPECT_FALSE(transceiver->stopped());
>   EXPECT_EQ(RtpTransceiverDirection::kSendRecv, transceiver->direction());
>-  EXPECT_EQ(rtc::nullopt, transceiver->current_direction());
>+  EXPECT_EQ(absl::nullopt, transceiver->current_direction());
> }
> 
> // Test that adding a transceiver with the audio kind creates an audio sender
> // and audio receiver with the receiver having a live audio track.
>-TEST_F(PeerConnectionRtpTest,
>+TEST_F(PeerConnectionRtpTestUnifiedPlan,
>        AddAudioTransceiverCreatesAudioSenderAndReceiver) {
>-  auto caller = CreatePeerConnectionWithUnifiedPlan();
>+  auto caller = CreatePeerConnection();
> 
>   auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
>+  EXPECT_EQ(cricket::MEDIA_TYPE_AUDIO, transceiver->media_type());
> 
>   ASSERT_TRUE(transceiver->sender());
>   EXPECT_EQ(cricket::MEDIA_TYPE_AUDIO, transceiver->sender()->media_type());
>@@ -470,11 +842,12 @@ TEST_F(PeerConnectionRtpTest,
> 
> // Test that adding a transceiver with the video kind creates an video sender
> // and video receiver with the receiver having a live video track.
>-TEST_F(PeerConnectionRtpTest,
>+TEST_F(PeerConnectionRtpTestUnifiedPlan,
>        AddAudioTransceiverCreatesVideoSenderAndReceiver) {
>-  auto caller = CreatePeerConnectionWithUnifiedPlan();
>+  auto caller = CreatePeerConnection();
> 
>   auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
>+  EXPECT_EQ(cricket::MEDIA_TYPE_VIDEO, transceiver->media_type());
> 
>   ASSERT_TRUE(transceiver->sender());
>   EXPECT_EQ(cricket::MEDIA_TYPE_VIDEO, transceiver->sender()->media_type());
>@@ -491,8 +864,8 @@ TEST_F(PeerConnectionRtpTest,
> // Test that after a call to AddTransceiver, the transceiver shows in
> // GetTransceivers(), the transceiver's sender shows in GetSenders(), and the
> // transceiver's receiver shows in GetReceivers().
>-TEST_F(PeerConnectionRtpTest, AddTransceiverShowsInLists) {
>-  auto caller = CreatePeerConnectionWithUnifiedPlan();
>+TEST_F(PeerConnectionRtpTestUnifiedPlan, AddTransceiverShowsInLists) {
>+  auto caller = CreatePeerConnection();
> 
>   auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
>   EXPECT_EQ(
>@@ -510,8 +883,9 @@ TEST_F(PeerConnectionRtpTest, AddTransceiverShowsInLists) {
> 
> // Test that the direction passed in through the AddTransceiver init parameter
> // is set in the returned transceiver.
>-TEST_F(PeerConnectionRtpTest, AddTransceiverWithDirectionIsReflected) {
>-  auto caller = CreatePeerConnectionWithUnifiedPlan();
>+TEST_F(PeerConnectionRtpTestUnifiedPlan,
>+       AddTransceiverWithDirectionIsReflected) {
>+  auto caller = CreatePeerConnection();
> 
>   RtpTransceiverInit init;
>   init.direction = RtpTransceiverDirection::kSendOnly;
>@@ -521,8 +895,9 @@ TEST_F(PeerConnectionRtpTest, AddTransceiverWithDirectionIsReflected) {
> 
> // Test that calling AddTransceiver with a track creates a transceiver which has
> // its sender's track set to the passed-in track.
>-TEST_F(PeerConnectionRtpTest, AddTransceiverWithTrackCreatesSenderWithTrack) {
>-  auto caller = CreatePeerConnectionWithUnifiedPlan();
>+TEST_F(PeerConnectionRtpTestUnifiedPlan,
>+       AddTransceiverWithTrackCreatesSenderWithTrack) {
>+  auto caller = CreatePeerConnection();
> 
>   auto audio_track = caller->CreateAudioTrack("audio track");
>   auto transceiver = caller->AddTransceiver(audio_track);
>@@ -540,9 +915,9 @@ TEST_F(PeerConnectionRtpTest, AddTransceiverWithTrackCreatesSenderWithTrack) {
> 
> // Test that calling AddTransceiver twice with the same track creates distinct
> // transceivers, senders with the same track.
>-TEST_F(PeerConnectionRtpTest,
>+TEST_F(PeerConnectionRtpTestUnifiedPlan,
>        AddTransceiverTwiceWithSameTrackCreatesMultipleTransceivers) {
>-  auto caller = CreatePeerConnectionWithUnifiedPlan();
>+  auto caller = CreatePeerConnection();
> 
>   auto audio_track = caller->CreateAudioTrack("audio track");
> 
>@@ -565,27 +940,27 @@ TEST_F(PeerConnectionRtpTest,
> 
> // RtpTransceiver error handling tests.
> 
>-TEST_F(PeerConnectionRtpTest, AddTransceiverWithInvalidKindReturnsError) {
>-  auto caller = CreatePeerConnectionWithUnifiedPlan();
>+TEST_F(PeerConnectionRtpTestUnifiedPlan,
>+       AddTransceiverWithInvalidKindReturnsError) {
>+  auto caller = CreatePeerConnection();
> 
>   auto result = caller->pc()->AddTransceiver(cricket::MEDIA_TYPE_DATA);
>   EXPECT_EQ(RTCErrorType::INVALID_PARAMETER, result.error().type());
> }
> 
>-TEST_F(PeerConnectionRtpTest, UnifiedPlanCanClosePeerConnection) {
>-  auto caller = CreatePeerConnectionWithUnifiedPlan();
>+TEST_F(PeerConnectionRtpTestUnifiedPlan,
>+       CanClosePeerConnectionWithoutCrashing) {
>+  auto caller = CreatePeerConnection();
> 
>   caller->pc()->Close();
> }
> 
> // Unified Plan AddTrack tests.
> 
>-class PeerConnectionRtpUnifiedPlanTest : public PeerConnectionRtpTest {};
>-
> // Test that adding an audio track creates a new audio RtpSender with the given
> // track.
>-TEST_F(PeerConnectionRtpUnifiedPlanTest, AddAudioTrackCreatesAudioSender) {
>-  auto caller = CreatePeerConnectionWithUnifiedPlan();
>+TEST_F(PeerConnectionRtpTestUnifiedPlan, AddAudioTrackCreatesAudioSender) {
>+  auto caller = CreatePeerConnection();
> 
>   auto audio_track = caller->CreateAudioTrack("a");
>   auto sender = caller->AddTrack(audio_track);
>@@ -597,8 +972,8 @@ TEST_F(PeerConnectionRtpUnifiedPlanTest, AddAudioTrackCreatesAudioSender) {
> 
> // Test that adding a video track creates a new video RtpSender with the given
> // track.
>-TEST_F(PeerConnectionRtpUnifiedPlanTest, AddVideoTrackCreatesVideoSender) {
>-  auto caller = CreatePeerConnectionWithUnifiedPlan();
>+TEST_F(PeerConnectionRtpTestUnifiedPlan, AddVideoTrackCreatesVideoSender) {
>+  auto caller = CreatePeerConnection();
> 
>   auto video_track = caller->CreateVideoTrack("a");
>   auto sender = caller->AddTrack(video_track);
>@@ -610,8 +985,8 @@ TEST_F(PeerConnectionRtpUnifiedPlanTest, AddVideoTrackCreatesVideoSender) {
> 
> // Test that adding a track to a new PeerConnection creates an RtpTransceiver
> // with the sender that AddTrack returns and in the sendrecv direction.
>-TEST_F(PeerConnectionRtpUnifiedPlanTest, AddFirstTrackCreatesTransceiver) {
>-  auto caller = CreatePeerConnectionWithUnifiedPlan();
>+TEST_F(PeerConnectionRtpTestUnifiedPlan, AddFirstTrackCreatesTransceiver) {
>+  auto caller = CreatePeerConnection();
> 
>   auto sender = caller->AddAudioTrack("a");
>   ASSERT_TRUE(sender);
>@@ -625,8 +1000,8 @@ TEST_F(PeerConnectionRtpUnifiedPlanTest, AddFirstTrackCreatesTransceiver) {
> // Test that if a transceiver of the same type but no track had been added to
> // the PeerConnection and later a call to AddTrack is made, the resulting sender
> // is the transceiver's sender and the sender's track is the newly-added track.
>-TEST_F(PeerConnectionRtpUnifiedPlanTest, AddTrackReusesTransceiver) {
>-  auto caller = CreatePeerConnectionWithUnifiedPlan();
>+TEST_F(PeerConnectionRtpTestUnifiedPlan, AddTrackReusesTransceiver) {
>+  auto caller = CreatePeerConnection();
> 
>   auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
>   auto audio_track = caller->CreateAudioTrack("a");
>@@ -642,8 +1017,8 @@ TEST_F(PeerConnectionRtpUnifiedPlanTest, AddTrackReusesTransceiver) {
> 
> // Test that adding two tracks to a new PeerConnection creates two
> // RtpTransceivers in the same order.
>-TEST_F(PeerConnectionRtpUnifiedPlanTest, TwoAddTrackCreatesTwoTransceivers) {
>-  auto caller = CreatePeerConnectionWithUnifiedPlan();
>+TEST_F(PeerConnectionRtpTestUnifiedPlan, TwoAddTrackCreatesTwoTransceivers) {
>+  auto caller = CreatePeerConnection();
> 
>   auto sender1 = caller->AddAudioTrack("a");
>   auto sender2 = caller->AddVideoTrack("v");
>@@ -658,8 +1033,8 @@ TEST_F(PeerConnectionRtpUnifiedPlanTest, TwoAddTrackCreatesTwoTransceivers) {
> // Test that if there are multiple transceivers with no sending track then a
> // later call to AddTrack will use the one of the same type as the newly-added
> // track.
>-TEST_F(PeerConnectionRtpUnifiedPlanTest, AddTrackReusesTransceiverOfType) {
>-  auto caller = CreatePeerConnectionWithUnifiedPlan();
>+TEST_F(PeerConnectionRtpTestUnifiedPlan, AddTrackReusesTransceiverOfType) {
>+  auto caller = CreatePeerConnection();
> 
>   auto audio_transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
>   auto video_transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_VIDEO);
>@@ -673,9 +1048,9 @@ TEST_F(PeerConnectionRtpUnifiedPlanTest, AddTrackReusesTransceiverOfType) {
> // Test that if the only transceivers that do not have a sending track have a
> // different type from the added track, then AddTrack will create a new
> // transceiver for the track.
>-TEST_F(PeerConnectionRtpUnifiedPlanTest,
>+TEST_F(PeerConnectionRtpTestUnifiedPlan,
>        AddTrackDoesNotReuseTransceiverOfWrongType) {
>-  auto caller = CreatePeerConnectionWithUnifiedPlan();
>+  auto caller = CreatePeerConnection();
> 
>   caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
>   auto sender = caller->AddVideoTrack("v");
>@@ -688,9 +1063,9 @@ TEST_F(PeerConnectionRtpUnifiedPlanTest,
> 
> // Test that the first available transceiver is reused by AddTrack when multiple
> // are available.
>-TEST_F(PeerConnectionRtpUnifiedPlanTest,
>+TEST_F(PeerConnectionRtpTestUnifiedPlan,
>        AddTrackReusesFirstMatchingTransceiver) {
>-  auto caller = CreatePeerConnectionWithUnifiedPlan();
>+  auto caller = CreatePeerConnection();
> 
>   caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
>   caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
>@@ -704,9 +1079,9 @@ TEST_F(PeerConnectionRtpUnifiedPlanTest,
> 
> // Test that a call to AddTrack that reuses a transceiver will change the
> // direction from inactive to sendonly.
>-TEST_F(PeerConnectionRtpUnifiedPlanTest,
>+TEST_F(PeerConnectionRtpTestUnifiedPlan,
>        AddTrackChangesDirectionFromInactiveToSendOnly) {
>-  auto caller = CreatePeerConnectionWithUnifiedPlan();
>+  auto caller = CreatePeerConnection();
> 
>   RtpTransceiverInit init;
>   init.direction = RtpTransceiverDirection::kInactive;
>@@ -721,9 +1096,9 @@ TEST_F(PeerConnectionRtpUnifiedPlanTest,
> 
> // Test that a call to AddTrack that reuses a transceiver will change the
> // direction from recvonly to sendrecv.
>-TEST_F(PeerConnectionRtpUnifiedPlanTest,
>+TEST_F(PeerConnectionRtpTestUnifiedPlan,
>        AddTrackChangesDirectionFromRecvOnlyToSendRecv) {
>-  auto caller = CreatePeerConnectionWithUnifiedPlan();
>+  auto caller = CreatePeerConnection();
> 
>   RtpTransceiverInit init;
>   init.direction = RtpTransceiverDirection::kRecvOnly;
>@@ -736,10 +1111,10 @@ TEST_F(PeerConnectionRtpUnifiedPlanTest,
>   EXPECT_EQ(RtpTransceiverDirection::kSendRecv, transceiver->direction());
> }
> 
>-TEST_F(PeerConnectionRtpUnifiedPlanTest, AddTrackCreatesSenderWithTrackId) {
>+TEST_F(PeerConnectionRtpTestUnifiedPlan, AddTrackCreatesSenderWithTrackId) {
>   const std::string kTrackId = "audio_track";
> 
>-  auto caller = CreatePeerConnectionWithUnifiedPlan();
>+  auto caller = CreatePeerConnection();
> 
>   auto audio_track = caller->CreateAudioTrack(kTrackId);
>   auto sender = caller->AddTrack(audio_track);
>@@ -749,28 +1124,26 @@ TEST_F(PeerConnectionRtpUnifiedPlanTest, AddTrackCreatesSenderWithTrackId) {
> 
> // Unified Plan AddTrack error handling.
> 
>-TEST_F(PeerConnectionRtpUnifiedPlanTest, AddTrackErrorIfClosed) {
>-  auto caller = CreatePeerConnectionWithUnifiedPlan();
>+TEST_F(PeerConnectionRtpTestUnifiedPlan, AddTrackErrorIfClosed) {
>+  auto caller = CreatePeerConnection();
> 
>   auto audio_track = caller->CreateAudioTrack("a");
>   caller->pc()->Close();
> 
>   caller->observer()->clear_negotiation_needed();
>-  auto result = caller->pc()
>-                ->AddTrack(audio_track, std::vector<std::string>());
>+  auto result = caller->pc()->AddTrack(audio_track, std::vector<std::string>());
>   EXPECT_EQ(RTCErrorType::INVALID_STATE, result.error().type());
>   EXPECT_FALSE(caller->observer()->negotiation_needed());
> }
> 
>-TEST_F(PeerConnectionRtpUnifiedPlanTest, AddTrackErrorIfTrackAlreadyHasSender) {
>-  auto caller = CreatePeerConnectionWithUnifiedPlan();
>+TEST_F(PeerConnectionRtpTestUnifiedPlan, AddTrackErrorIfTrackAlreadyHasSender) {
>+  auto caller = CreatePeerConnection();
> 
>   auto audio_track = caller->CreateAudioTrack("a");
>   ASSERT_TRUE(caller->AddTrack(audio_track));
> 
>   caller->observer()->clear_negotiation_needed();
>-  auto result = caller->pc()
>-                ->AddTrack(audio_track, std::vector<std::string>());
>+  auto result = caller->pc()->AddTrack(audio_track, std::vector<std::string>());
>   EXPECT_EQ(RTCErrorType::INVALID_PARAMETER, result.error().type());
>   EXPECT_FALSE(caller->observer()->negotiation_needed());
> }
>@@ -779,8 +1152,8 @@ TEST_F(PeerConnectionRtpUnifiedPlanTest, AddTrackErrorIfTrackAlreadyHasSender) {
> 
> // Test that calling RemoveTrack on a sender with a previously-added track
> // clears the sender's track.
>-TEST_F(PeerConnectionRtpUnifiedPlanTest, RemoveTrackClearsSenderTrack) {
>-  auto caller = CreatePeerConnectionWithUnifiedPlan();
>+TEST_F(PeerConnectionRtpTestUnifiedPlan, RemoveTrackClearsSenderTrack) {
>+  auto caller = CreatePeerConnection();
> 
>   auto sender = caller->AddAudioTrack("a");
>   ASSERT_TRUE(caller->pc()->RemoveTrack(sender));
>@@ -790,9 +1163,9 @@ TEST_F(PeerConnectionRtpUnifiedPlanTest, RemoveTrackClearsSenderTrack) {
> 
> // Test that calling RemoveTrack on a sender where the transceiver is configured
> // in the sendrecv direction changes the transceiver's direction to recvonly.
>-TEST_F(PeerConnectionRtpUnifiedPlanTest,
>+TEST_F(PeerConnectionRtpTestUnifiedPlan,
>        RemoveTrackChangesDirectionFromSendRecvToRecvOnly) {
>-  auto caller = CreatePeerConnectionWithUnifiedPlan();
>+  auto caller = CreatePeerConnection();
> 
>   RtpTransceiverInit init;
>   init.direction = RtpTransceiverDirection::kSendRecv;
>@@ -809,9 +1182,9 @@ TEST_F(PeerConnectionRtpUnifiedPlanTest,
> 
> // Test that calling RemoveTrack on a sender where the transceiver is configured
> // in the sendonly direction changes the transceiver's direction to inactive.
>-TEST_F(PeerConnectionRtpUnifiedPlanTest,
>+TEST_F(PeerConnectionRtpTestUnifiedPlan,
>        RemoveTrackChangesDirectionFromSendOnlyToInactive) {
>-  auto caller = CreatePeerConnectionWithUnifiedPlan();
>+  auto caller = CreatePeerConnection();
> 
>   RtpTransceiverInit init;
>   init.direction = RtpTransceiverDirection::kSendOnly;
>@@ -827,8 +1200,8 @@ TEST_F(PeerConnectionRtpUnifiedPlanTest,
> 
> // Test that calling RemoveTrack with a sender that has a null track results in
> // no change in state.
>-TEST_F(PeerConnectionRtpUnifiedPlanTest, RemoveTrackWithNullSenderTrackIsNoOp) {
>-  auto caller = CreatePeerConnectionWithUnifiedPlan();
>+TEST_F(PeerConnectionRtpTestUnifiedPlan, RemoveTrackWithNullSenderTrackIsNoOp) {
>+  auto caller = CreatePeerConnection();
> 
>   auto sender = caller->AddAudioTrack("a");
>   auto transceiver = caller->pc()->GetTransceivers()[0];
>@@ -843,8 +1216,8 @@ TEST_F(PeerConnectionRtpUnifiedPlanTest, RemoveTrackWithNullSenderTrackIsNoOp) {
> 
> // Unified Plan RemoveTrack error handling.
> 
>-TEST_F(PeerConnectionRtpUnifiedPlanTest, RemoveTrackErrorIfClosed) {
>-  auto caller = CreatePeerConnectionWithUnifiedPlan();
>+TEST_F(PeerConnectionRtpTestUnifiedPlan, RemoveTrackErrorIfClosed) {
>+  auto caller = CreatePeerConnection();
> 
>   auto sender = caller->AddAudioTrack("a");
>   caller->pc()->Close();
>@@ -854,9 +1227,9 @@ TEST_F(PeerConnectionRtpUnifiedPlanTest, RemoveTrackErrorIfClosed) {
>   EXPECT_FALSE(caller->observer()->negotiation_needed());
> }
> 
>-TEST_F(PeerConnectionRtpUnifiedPlanTest,
>+TEST_F(PeerConnectionRtpTestUnifiedPlan,
>        RemoveTrackNoErrorIfTrackAlreadyRemoved) {
>-  auto caller = CreatePeerConnectionWithUnifiedPlan();
>+  auto caller = CreatePeerConnection();
> 
>   auto sender = caller->AddAudioTrack("a");
>   ASSERT_TRUE(caller->pc()->RemoveTrack(sender));
>@@ -866,4 +1239,363 @@ TEST_F(PeerConnectionRtpUnifiedPlanTest,
>   EXPECT_FALSE(caller->observer()->negotiation_needed());
> }
> 
>+// Test that setting offers that add/remove/add a track repeatedly without
>+// setting the appropriate answer in between works.
>+// These are regression tests for bugs.webrtc.org/9401
>+TEST_F(PeerConnectionRtpTestUnifiedPlan, AddRemoveAddTrackOffersWorksAudio) {
>+  auto caller = CreatePeerConnection();
>+
>+  auto sender1 = caller->AddAudioTrack("audio1");
>+  ASSERT_TRUE(caller->SetLocalDescription(caller->CreateOffer()));
>+
>+  caller->pc()->RemoveTrack(sender1);
>+  ASSERT_TRUE(caller->SetLocalDescription(caller->CreateOffer()));
>+
>+  // This will re-use the transceiver created by the first AddTrack.
>+  auto sender2 = caller->AddAudioTrack("audio2");
>+  ASSERT_TRUE(caller->SetLocalDescription(caller->CreateOffer()));
>+
>+  EXPECT_EQ(1u, caller->pc()->GetTransceivers().size());
>+  EXPECT_EQ(sender1, sender2);
>+}
>+TEST_F(PeerConnectionRtpTestUnifiedPlan, AddRemoveAddTrackOffersWorksVideo) {
>+  auto caller = CreatePeerConnection();
>+
>+  auto sender1 = caller->AddVideoTrack("video1");
>+  ASSERT_TRUE(caller->SetLocalDescription(caller->CreateOffer()));
>+
>+  caller->pc()->RemoveTrack(sender1);
>+  ASSERT_TRUE(caller->SetLocalDescription(caller->CreateOffer()));
>+
>+  // This will re-use the transceiver created by the first AddTrack.
>+  auto sender2 = caller->AddVideoTrack("video2");
>+  ASSERT_TRUE(caller->SetLocalDescription(caller->CreateOffer()));
>+
>+  EXPECT_EQ(1u, caller->pc()->GetTransceivers().size());
>+  EXPECT_EQ(sender1, sender2);
>+}
>+
>+// Test that CreateOffer succeeds if two tracks with the same label are added.
>+TEST_F(PeerConnectionRtpTestUnifiedPlan, CreateOfferSameTrackLabel) {
>+  auto caller = CreatePeerConnection();
>+
>+  auto audio_sender = caller->AddAudioTrack("track", {});
>+  auto video_sender = caller->AddVideoTrack("track", {});
>+
>+  EXPECT_TRUE(caller->CreateOffer());
>+
>+  EXPECT_EQ(audio_sender->track()->id(), video_sender->track()->id());
>+  EXPECT_NE(audio_sender->id(), video_sender->id());
>+}
>+
>+// Test that CreateAnswer succeeds if two tracks with the same label are added.
>+TEST_F(PeerConnectionRtpTestUnifiedPlan, CreateAnswerSameTrackLabel) {
>+  auto caller = CreatePeerConnection();
>+  auto callee = CreatePeerConnection();
>+
>+  RtpTransceiverInit recvonly;
>+  recvonly.direction = RtpTransceiverDirection::kRecvOnly;
>+  caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO, recvonly);
>+  caller->AddTransceiver(cricket::MEDIA_TYPE_VIDEO, recvonly);
>+
>+  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOffer()));
>+
>+  auto audio_sender = callee->AddAudioTrack("track", {});
>+  auto video_sender = callee->AddVideoTrack("track", {});
>+
>+  EXPECT_TRUE(callee->CreateAnswer());
>+
>+  EXPECT_EQ(audio_sender->track()->id(), video_sender->track()->id());
>+  EXPECT_NE(audio_sender->id(), video_sender->id());
>+}
>+
>+// Test that calling AddTrack, RemoveTrack and AddTrack again creates a second
>+// m= section with a random sender id (different from the first, now rejected,
>+// m= section).
>+TEST_F(PeerConnectionRtpTestUnifiedPlan,
>+       AddRemoveAddTrackGeneratesNewSenderId) {
>+  auto caller = CreatePeerConnection();
>+  auto callee = CreatePeerConnection();
>+
>+  auto track = caller->CreateVideoTrack("video");
>+  auto sender1 = caller->AddTrack(track);
>+  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
>+
>+  caller->pc()->RemoveTrack(sender1);
>+  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
>+
>+  auto sender2 = caller->AddTrack(track);
>+
>+  EXPECT_NE(sender1, sender2);
>+  EXPECT_NE(sender1->id(), sender2->id());
>+  std::string sender2_id = sender2->id();
>+
>+  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
>+
>+  // The sender's ID should not change after negotiation.
>+  EXPECT_EQ(sender2_id, sender2->id());
>+}
>+
>+// Test that OnRenegotiationNeeded is fired if SetDirection is called on an
>+// active RtpTransceiver with a new direction.
>+TEST_F(PeerConnectionRtpTestUnifiedPlan,
>+       RenegotiationNeededAfterTransceiverSetDirection) {
>+  auto caller = CreatePeerConnection();
>+
>+  auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
>+
>+  caller->observer()->clear_negotiation_needed();
>+  transceiver->SetDirection(RtpTransceiverDirection::kInactive);
>+  EXPECT_TRUE(caller->observer()->negotiation_needed());
>+}
>+
>+// Test that OnRenegotiationNeeded is not fired if SetDirection is called on an
>+// active RtpTransceiver with current direction.
>+TEST_F(PeerConnectionRtpTestUnifiedPlan,
>+       NoRenegotiationNeededAfterTransceiverSetSameDirection) {
>+  auto caller = CreatePeerConnection();
>+
>+  auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
>+
>+  caller->observer()->clear_negotiation_needed();
>+  transceiver->SetDirection(transceiver->direction());
>+  EXPECT_FALSE(caller->observer()->negotiation_needed());
>+}
>+
>+// Test that OnRenegotiationNeeded is not fired if SetDirection is called on a
>+// stopped RtpTransceiver.
>+TEST_F(PeerConnectionRtpTestUnifiedPlan,
>+       NoRenegotiationNeededAfterSetDirectionOnStoppedTransceiver) {
>+  auto caller = CreatePeerConnection();
>+
>+  auto transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
>+  transceiver->Stop();
>+
>+  caller->observer()->clear_negotiation_needed();
>+  transceiver->SetDirection(RtpTransceiverDirection::kInactive);
>+  EXPECT_FALSE(caller->observer()->negotiation_needed());
>+}
>+
>+// Test MSID signaling between Unified Plan and Plan B endpoints. There are two
>+// options for this kind of signaling: media section based (a=msid) and ssrc
>+// based (a=ssrc MSID). While JSEP only specifies media section MSID signaling,
>+// we want to ensure compatibility with older Plan B endpoints that might expect
>+// ssrc based MSID signaling. Thus we test here that Unified Plan offers both
>+// types but answers with the same type as the offer.
>+
>+class PeerConnectionMsidSignalingTest
>+    : public PeerConnectionRtpTestUnifiedPlan {};
>+
>+TEST_F(PeerConnectionMsidSignalingTest, UnifiedPlanTalkingToOurself) {
>+  auto caller = CreatePeerConnectionWithUnifiedPlan();
>+  caller->AddAudioTrack("caller_audio");
>+  auto callee = CreatePeerConnectionWithUnifiedPlan();
>+  callee->AddAudioTrack("callee_audio");
>+
>+  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
>+
>+  // Offer should have had both a=msid and a=ssrc MSID lines.
>+  auto* offer = callee->pc()->remote_description();
>+  EXPECT_EQ((cricket::kMsidSignalingMediaSection |
>+             cricket::kMsidSignalingSsrcAttribute),
>+            offer->description()->msid_signaling());
>+
>+  // Answer should have had only a=msid lines.
>+  auto* answer = caller->pc()->remote_description();
>+  EXPECT_EQ(cricket::kMsidSignalingMediaSection,
>+            answer->description()->msid_signaling());
>+  // Check that this is counted correctly
>+  EXPECT_EQ(2, webrtc::metrics::NumSamples(
>+                   "WebRTC.PeerConnection.SdpSemanticNegotiated"));
>+  EXPECT_EQ(2, webrtc::metrics::NumEvents(
>+                   "WebRTC.PeerConnection.SdpSemanticNegotiated",
>+                   kSdpSemanticNegotiatedUnifiedPlan));
>+}
>+
>+TEST_F(PeerConnectionMsidSignalingTest, PlanBOfferToUnifiedPlanAnswer) {
>+  auto caller = CreatePeerConnectionWithPlanB();
>+  caller->AddAudioTrack("caller_audio");
>+  auto callee = CreatePeerConnectionWithUnifiedPlan();
>+  callee->AddAudioTrack("callee_audio");
>+
>+  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
>+
>+  // Offer should have only a=ssrc MSID lines.
>+  auto* offer = callee->pc()->remote_description();
>+  EXPECT_EQ(cricket::kMsidSignalingSsrcAttribute,
>+            offer->description()->msid_signaling());
>+
>+  // Answer should have only a=ssrc MSID lines to match the offer.
>+  auto* answer = caller->pc()->remote_description();
>+  EXPECT_EQ(cricket::kMsidSignalingSsrcAttribute,
>+            answer->description()->msid_signaling());
>+}
>+
>+// This tests that a Plan B endpoint appropriately sets the remote description
>+// from a Unified Plan offer. When the Unified Plan offer contains a=msid lines
>+// that signal no stream ids or multiple stream ids we expect that the Plan B
>+// endpoint always has exactly one media stream per track.
>+TEST_F(PeerConnectionMsidSignalingTest, UnifiedPlanToPlanBAnswer) {
>+  const std::string kStreamId1 = "audio_stream_1";
>+  const std::string kStreamId2 = "audio_stream_2";
>+
>+  auto caller = CreatePeerConnectionWithUnifiedPlan();
>+  caller->AddAudioTrack("caller_audio", {kStreamId1, kStreamId2});
>+  caller->AddVideoTrack("caller_video", {});
>+  auto callee = CreatePeerConnectionWithPlanB();
>+  callee->AddAudioTrack("callee_audio");
>+  caller->AddVideoTrack("callee_video");
>+
>+  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
>+
>+  // Offer should have had both a=msid and a=ssrc MSID lines.
>+  auto* offer = callee->pc()->remote_description();
>+  EXPECT_EQ((cricket::kMsidSignalingMediaSection |
>+             cricket::kMsidSignalingSsrcAttribute),
>+            offer->description()->msid_signaling());
>+
>+  // Callee should always have 1 stream for all of it's receivers.
>+  const auto& track_events = callee->observer()->add_track_events_;
>+  ASSERT_EQ(2u, track_events.size());
>+  ASSERT_EQ(1u, track_events[0].streams.size());
>+  EXPECT_EQ(kStreamId1, track_events[0].streams[0]->id());
>+  ASSERT_EQ(1u, track_events[1].streams.size());
>+  // This autogenerated a stream id for the empty one signalled.
>+  EXPECT_FALSE(track_events[1].streams[0]->id().empty());
>+}
>+
>+TEST_F(PeerConnectionMsidSignalingTest, PureUnifiedPlanToUs) {
>+  auto caller = CreatePeerConnectionWithUnifiedPlan();
>+  caller->AddAudioTrack("caller_audio");
>+  auto callee = CreatePeerConnectionWithUnifiedPlan();
>+  callee->AddAudioTrack("callee_audio");
>+
>+  auto offer = caller->CreateOffer();
>+  // Simulate a pure Unified Plan offerer by setting the MSID signaling to media
>+  // section only.
>+  offer->description()->set_msid_signaling(cricket::kMsidSignalingMediaSection);
>+
>+  ASSERT_TRUE(
>+      caller->SetLocalDescription(CloneSessionDescription(offer.get())));
>+  ASSERT_TRUE(callee->SetRemoteDescription(std::move(offer)));
>+
>+  // Answer should have only a=msid to match the offer.
>+  auto answer = callee->CreateAnswer();
>+  EXPECT_EQ(cricket::kMsidSignalingMediaSection,
>+            answer->description()->msid_signaling());
>+}
>+
>+// Test that the correct UMA metrics are reported for simple/complex SDP.
>+
>+class SdpFormatReceivedTest : public PeerConnectionRtpTestUnifiedPlan {};
>+
>+#ifdef HAVE_SCTP
>+TEST_F(SdpFormatReceivedTest, DataChannelOnlyIsReportedAsNoTracks) {
>+  auto caller = CreatePeerConnectionWithUnifiedPlan();
>+  caller->CreateDataChannel("dc");
>+  auto callee = CreatePeerConnectionWithUnifiedPlan();
>+
>+  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOffer()));
>+  // Note that only the callee does ReportSdpFormatReceived.
>+  EXPECT_EQ(1, webrtc::metrics::NumSamples(
>+                   "WebRTC.PeerConnection.SdpFormatReceived"));
>+  EXPECT_EQ(
>+      1, webrtc::metrics::NumEvents("WebRTC.PeerConnection.SdpFormatReceived",
>+                                    kSdpFormatReceivedNoTracks));
>+}
>+#endif  // HAVE_SCTP
>+
>+TEST_F(SdpFormatReceivedTest, SimpleUnifiedPlanIsReportedAsSimple) {
>+  auto caller = CreatePeerConnectionWithUnifiedPlan();
>+  caller->AddAudioTrack("audio");
>+  caller->AddVideoTrack("video");
>+  auto callee = CreatePeerConnectionWithPlanB();
>+
>+  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOffer()));
>+  // Note that only the callee does ReportSdpFormatReceived.
>+  EXPECT_EQ(1, webrtc::metrics::NumSamples(
>+                   "WebRTC.PeerConnection.SdpFormatReceived"));
>+  EXPECT_EQ(
>+      1, webrtc::metrics::NumEvents("WebRTC.PeerConnection.SdpFormatReceived",
>+                                    kSdpFormatReceivedSimple));
>+}
>+
>+TEST_F(SdpFormatReceivedTest, SimplePlanBIsReportedAsSimple) {
>+  auto caller = CreatePeerConnectionWithPlanB();
>+  caller->AddVideoTrack("video");  // Video only.
>+  auto callee = CreatePeerConnectionWithUnifiedPlan();
>+
>+  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOffer()));
>+
>+  EXPECT_EQ(1, webrtc::metrics::NumSamples(
>+                   "WebRTC.PeerConnection.SdpFormatReceived"));
>+  EXPECT_EQ(
>+      1, webrtc::metrics::NumEvents("WebRTC.PeerConnection.SdpFormatReceived",
>+                                    kSdpFormatReceivedSimple));
>+}
>+
>+TEST_F(SdpFormatReceivedTest, ComplexUnifiedIsReportedAsComplexUnifiedPlan) {
>+  auto caller = CreatePeerConnectionWithUnifiedPlan();
>+  caller->AddAudioTrack("audio1");
>+  caller->AddAudioTrack("audio2");
>+  caller->AddVideoTrack("video");
>+  auto callee = CreatePeerConnectionWithPlanB();
>+
>+  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOffer()));
>+  // Note that only the callee does ReportSdpFormatReceived.
>+  EXPECT_EQ(1, webrtc::metrics::NumSamples(
>+                   "WebRTC.PeerConnection.SdpFormatReceived"));
>+  EXPECT_EQ(
>+      1, webrtc::metrics::NumEvents("WebRTC.PeerConnection.SdpFormatReceived",
>+                                    kSdpFormatReceivedComplexUnifiedPlan));
>+}
>+
>+TEST_F(SdpFormatReceivedTest, ComplexPlanBIsReportedAsComplexPlanB) {
>+  auto caller = CreatePeerConnectionWithPlanB();
>+  caller->AddVideoTrack("video1");
>+  caller->AddVideoTrack("video2");
>+  auto callee = CreatePeerConnectionWithUnifiedPlan();
>+
>+  // This fails since Unified Plan cannot set a session description with
>+  // multiple "Plan B tracks" in the same media section. But we still expect the
>+  // SDP Format to be recorded.
>+  ASSERT_FALSE(callee->SetRemoteDescription(caller->CreateOffer()));
>+  // Note that only the callee does ReportSdpFormatReceived.
>+  EXPECT_EQ(1, webrtc::metrics::NumSamples(
>+                   "WebRTC.PeerConnection.SdpFormatReceived"));
>+  EXPECT_EQ(
>+      1, webrtc::metrics::NumEvents("WebRTC.PeerConnection.SdpFormatReceived",
>+                                    kSdpFormatReceivedComplexPlanB));
>+}
>+
>+// Sender setups in a call.
>+
>+TEST_P(PeerConnectionRtpTest, CreateTwoSendersWithSameTrack) {
>+  auto caller = CreatePeerConnection();
>+  auto callee = CreatePeerConnection();
>+
>+  auto track = caller->CreateAudioTrack("audio_track");
>+  auto sender1 = caller->AddTrack(track);
>+  ASSERT_TRUE(sender1);
>+  // We need to temporarily reset the track for the subsequent AddTrack() to
>+  // succeed.
>+  EXPECT_TRUE(sender1->SetTrack(nullptr));
>+  auto sender2 = caller->AddTrack(track);
>+  EXPECT_TRUE(sender2);
>+  EXPECT_TRUE(sender1->SetTrack(track));
>+
>+  if (sdp_semantics_ == SdpSemantics::kPlanB) {
>+    // TODO(hbos): When https://crbug.com/webrtc/8734 is resolved, this should
>+    // return true, and doing |callee->SetRemoteDescription()| should work.
>+    EXPECT_FALSE(caller->CreateOfferAndSetAsLocal());
>+  } else {
>+    EXPECT_TRUE(caller->CreateOfferAndSetAsLocal());
>+  }
>+}
>+
>+INSTANTIATE_TEST_CASE_P(PeerConnectionRtpTest,
>+                        PeerConnectionRtpTest,
>+                        Values(SdpSemantics::kPlanB,
>+                               SdpSemantics::kUnifiedPlan));
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_signaling_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_signaling_unittest.cc
>index 282ce1ce53d..95f464b2ee2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_signaling_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection_signaling_unittest.cc
>@@ -16,16 +16,18 @@
> #include "api/audio_codecs/builtin_audio_decoder_factory.h"
> #include "api/audio_codecs/builtin_audio_encoder_factory.h"
> #include "api/peerconnectionproxy.h"
>+#include "api/video_codecs/builtin_video_decoder_factory.h"
>+#include "api/video_codecs/builtin_video_encoder_factory.h"
> #include "pc/peerconnection.h"
> #include "pc/peerconnectionwrapper.h"
> #include "pc/sdputils.h"
> #ifdef WEBRTC_ANDROID
> #include "pc/test/androidtestinitializer.h"
> #endif
>+#include "absl/memory/memory.h"
> #include "pc/test/fakeaudiocapturemodule.h"
> #include "pc/test/fakertccertificategenerator.h"
> #include "rtc_base/gunit.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/stringutils.h"
> #include "rtc_base/virtualsocketserver.h"
> #include "test/gmock.h"
>@@ -55,19 +57,23 @@ class PeerConnectionWrapperForSignalingTest : public PeerConnectionWrapper {
>   }
> };
> 
>-class PeerConnectionSignalingTest : public ::testing::Test {
>+class PeerConnectionSignalingBaseTest : public ::testing::Test {
>  protected:
>   typedef std::unique_ptr<PeerConnectionWrapperForSignalingTest> WrapperPtr;
> 
>-  PeerConnectionSignalingTest()
>-      : vss_(new rtc::VirtualSocketServer()), main_(vss_.get()) {
>+  explicit PeerConnectionSignalingBaseTest(SdpSemantics sdp_semantics)
>+      : vss_(new rtc::VirtualSocketServer()),
>+        main_(vss_.get()),
>+        sdp_semantics_(sdp_semantics) {
> #ifdef WEBRTC_ANDROID
>     InitializeAndroidObjects();
> #endif
>     pc_factory_ = CreatePeerConnectionFactory(
>         rtc::Thread::Current(), rtc::Thread::Current(), rtc::Thread::Current(),
>-        FakeAudioCaptureModule::Create(), CreateBuiltinAudioEncoderFactory(),
>-        CreateBuiltinAudioDecoderFactory(), nullptr, nullptr);
>+        rtc::scoped_refptr<AudioDeviceModule>(FakeAudioCaptureModule::Create()),
>+        CreateBuiltinAudioEncoderFactory(), CreateBuiltinAudioDecoderFactory(),
>+        CreateBuiltinVideoEncoderFactory(), CreateBuiltinVideoDecoderFactory(),
>+        nullptr /* audio_mixer */, nullptr /* audio_processing */);
>   }
> 
>   WrapperPtr CreatePeerConnection() {
>@@ -75,14 +81,16 @@ class PeerConnectionSignalingTest : public ::testing::Test {
>   }
> 
>   WrapperPtr CreatePeerConnection(const RTCConfiguration& config) {
>-    auto observer = rtc::MakeUnique<MockPeerConnectionObserver>();
>-    auto pc = pc_factory_->CreatePeerConnection(config, nullptr, nullptr,
>-                                                observer.get());
>+    auto observer = absl::make_unique<MockPeerConnectionObserver>();
>+    RTCConfiguration modified_config = config;
>+    modified_config.sdp_semantics = sdp_semantics_;
>+    auto pc = pc_factory_->CreatePeerConnection(modified_config, nullptr,
>+                                                nullptr, observer.get());
>     if (!pc) {
>       return nullptr;
>     }
> 
>-    return rtc::MakeUnique<PeerConnectionWrapperForSignalingTest>(
>+    return absl::make_unique<PeerConnectionWrapperForSignalingTest>(
>         pc_factory_, pc, std::move(observer));
>   }
> 
>@@ -102,16 +110,24 @@ class PeerConnectionSignalingTest : public ::testing::Test {
>   std::unique_ptr<rtc::VirtualSocketServer> vss_;
>   rtc::AutoSocketServerThread main_;
>   rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory_;
>+  const SdpSemantics sdp_semantics_;
> };
> 
>-TEST_F(PeerConnectionSignalingTest, SetLocalOfferTwiceWorks) {
>+class PeerConnectionSignalingTest
>+    : public PeerConnectionSignalingBaseTest,
>+      public ::testing::WithParamInterface<SdpSemantics> {
>+ protected:
>+  PeerConnectionSignalingTest() : PeerConnectionSignalingBaseTest(GetParam()) {}
>+};
>+
>+TEST_P(PeerConnectionSignalingTest, SetLocalOfferTwiceWorks) {
>   auto caller = CreatePeerConnection();
> 
>   EXPECT_TRUE(caller->SetLocalDescription(caller->CreateOffer()));
>   EXPECT_TRUE(caller->SetLocalDescription(caller->CreateOffer()));
> }
> 
>-TEST_F(PeerConnectionSignalingTest, SetRemoteOfferTwiceWorks) {
>+TEST_P(PeerConnectionSignalingTest, SetRemoteOfferTwiceWorks) {
>   auto caller = CreatePeerConnection();
>   auto callee = CreatePeerConnection();
> 
>@@ -119,14 +135,14 @@ TEST_F(PeerConnectionSignalingTest, SetRemoteOfferTwiceWorks) {
>   EXPECT_TRUE(callee->SetRemoteDescription(caller->CreateOffer()));
> }
> 
>-TEST_F(PeerConnectionSignalingTest, FailToSetNullLocalDescription) {
>+TEST_P(PeerConnectionSignalingTest, FailToSetNullLocalDescription) {
>   auto caller = CreatePeerConnection();
>   std::string error;
>   ASSERT_FALSE(caller->SetLocalDescription(nullptr, &error));
>   EXPECT_EQ("SessionDescription is NULL.", error);
> }
> 
>-TEST_F(PeerConnectionSignalingTest, FailToSetNullRemoteDescription) {
>+TEST_P(PeerConnectionSignalingTest, FailToSetNullRemoteDescription) {
>   auto caller = CreatePeerConnection();
>   std::string error;
>   ASSERT_FALSE(caller->SetRemoteDescription(nullptr, &error));
>@@ -142,9 +158,15 @@ TEST_F(PeerConnectionSignalingTest, FailToSetNullRemoteDescription) {
> // state the PeerConnection was created in before it was closed.
> 
> class PeerConnectionSignalingStateTest
>-    : public PeerConnectionSignalingTest,
>-      public ::testing::WithParamInterface<std::tuple<SignalingState, bool>> {
>+    : public PeerConnectionSignalingBaseTest,
>+      public ::testing::WithParamInterface<
>+          std::tuple<SdpSemantics, SignalingState, bool>> {
>  protected:
>+  PeerConnectionSignalingStateTest()
>+      : PeerConnectionSignalingBaseTest(std::get<0>(GetParam())),
>+        state_under_test_(std::make_tuple(std::get<1>(GetParam()),
>+                                          std::get<2>(GetParam()))) {}
>+
>   RTCConfiguration GetConfig() {
>     RTCConfiguration config;
>     config.certificates.push_back(
>@@ -152,6 +174,10 @@ class PeerConnectionSignalingStateTest
>     return config;
>   }
> 
>+  WrapperPtr CreatePeerConnectionUnderTest() {
>+    return CreatePeerConnectionInState(state_under_test_);
>+  }
>+
>   WrapperPtr CreatePeerConnectionInState(SignalingState state) {
>     return CreatePeerConnectionInState(std::make_tuple(state, false));
>   }
>@@ -207,23 +233,12 @@ class PeerConnectionSignalingStateTest
> 
>     return wrapper;
>   }
>-};
> 
>-::testing::AssertionResult AssertStartsWith(const char* str_expr,
>-                                            const char* prefix_expr,
>-                                            const std::string& str,
>-                                            const std::string& prefix) {
>-  if (rtc::starts_with(str.c_str(), prefix.c_str())) {
>-    return ::testing::AssertionSuccess();
>-  } else {
>-    return ::testing::AssertionFailure()
>-           << str_expr << "\nwhich is\n\"" << str << "\"\ndoes not start with\n"
>-           << prefix_expr << "\nwhich is\n\"" << prefix << "\"";
>-  }
>-}
>+  std::tuple<SignalingState, bool> state_under_test_;
>+};
> 
> TEST_P(PeerConnectionSignalingStateTest, CreateOffer) {
>-  auto wrapper = CreatePeerConnectionInState(GetParam());
>+  auto wrapper = CreatePeerConnectionUnderTest();
>   if (wrapper->signaling_state() != SignalingState::kClosed) {
>     EXPECT_TRUE(wrapper->CreateOffer());
>   } else {
>@@ -235,30 +250,21 @@ TEST_P(PeerConnectionSignalingStateTest, CreateOffer) {
> }
> 
> TEST_P(PeerConnectionSignalingStateTest, CreateAnswer) {
>-  auto wrapper = CreatePeerConnectionInState(GetParam());
>+  auto wrapper = CreatePeerConnectionUnderTest();
>   if (wrapper->signaling_state() == SignalingState::kHaveLocalPrAnswer ||
>       wrapper->signaling_state() == SignalingState::kHaveRemoteOffer) {
>     EXPECT_TRUE(wrapper->CreateAnswer());
>   } else {
>     std::string error;
>     ASSERT_FALSE(wrapper->CreateAnswer(RTCOfferAnswerOptions(), &error));
>-    if (wrapper->signaling_state() == SignalingState::kClosed) {
>-      EXPECT_PRED_FORMAT2(AssertStartsWith, error,
>-                          "CreateAnswer called when PeerConnection is closed.");
>-    } else if (wrapper->signaling_state() ==
>-               SignalingState::kHaveRemotePrAnswer) {
>-      EXPECT_PRED_FORMAT2(AssertStartsWith, error,
>-                          "CreateAnswer called without remote offer.");
>-    } else {
>-      EXPECT_PRED_FORMAT2(
>-          AssertStartsWith, error,
>-          "CreateAnswer can't be called before SetRemoteDescription.");
>-    }
>+    EXPECT_EQ(error,
>+              "PeerConnection cannot create an answer in a state other than "
>+              "have-remote-offer or have-local-pranswer.");
>   }
> }
> 
> TEST_P(PeerConnectionSignalingStateTest, SetLocalOffer) {
>-  auto wrapper = CreatePeerConnectionInState(GetParam());
>+  auto wrapper = CreatePeerConnectionUnderTest();
>   if (wrapper->signaling_state() == SignalingState::kStable ||
>       wrapper->signaling_state() == SignalingState::kHaveLocalOffer) {
>     // Need to call CreateOffer on the PeerConnection under test, otherwise when
>@@ -286,7 +292,7 @@ TEST_P(PeerConnectionSignalingStateTest, SetLocalPrAnswer) {
>   auto pranswer =
>       CloneSessionDescription(wrapper_for_pranswer->pc()->local_description());
> 
>-  auto wrapper = CreatePeerConnectionInState(GetParam());
>+  auto wrapper = CreatePeerConnectionUnderTest();
>   if (wrapper->signaling_state() == SignalingState::kHaveLocalPrAnswer ||
>       wrapper->signaling_state() == SignalingState::kHaveRemoteOffer) {
>     EXPECT_TRUE(wrapper->SetLocalDescription(std::move(pranswer)));
>@@ -304,7 +310,7 @@ TEST_P(PeerConnectionSignalingStateTest, SetLocalAnswer) {
>       CreatePeerConnectionInState(SignalingState::kHaveRemoteOffer);
>   auto answer = wrapper_for_answer->CreateAnswer();
> 
>-  auto wrapper = CreatePeerConnectionInState(GetParam());
>+  auto wrapper = CreatePeerConnectionUnderTest();
>   if (wrapper->signaling_state() == SignalingState::kHaveLocalPrAnswer ||
>       wrapper->signaling_state() == SignalingState::kHaveRemoteOffer) {
>     EXPECT_TRUE(wrapper->SetLocalDescription(std::move(answer)));
>@@ -323,7 +329,7 @@ TEST_P(PeerConnectionSignalingStateTest, SetRemoteOffer) {
>   auto offer =
>       CloneSessionDescription(wrapper_for_offer->pc()->remote_description());
> 
>-  auto wrapper = CreatePeerConnectionInState(GetParam());
>+  auto wrapper = CreatePeerConnectionUnderTest();
>   if (wrapper->signaling_state() == SignalingState::kStable ||
>       wrapper->signaling_state() == SignalingState::kHaveRemoteOffer) {
>     EXPECT_TRUE(wrapper->SetRemoteDescription(std::move(offer)));
>@@ -342,7 +348,7 @@ TEST_P(PeerConnectionSignalingStateTest, SetRemotePrAnswer) {
>   auto pranswer =
>       CloneSessionDescription(wrapper_for_pranswer->pc()->remote_description());
> 
>-  auto wrapper = CreatePeerConnectionInState(GetParam());
>+  auto wrapper = CreatePeerConnectionUnderTest();
>   if (wrapper->signaling_state() == SignalingState::kHaveLocalOffer ||
>       wrapper->signaling_state() == SignalingState::kHaveRemotePrAnswer) {
>     EXPECT_TRUE(wrapper->SetRemoteDescription(std::move(pranswer)));
>@@ -360,7 +366,7 @@ TEST_P(PeerConnectionSignalingStateTest, SetRemoteAnswer) {
>       CreatePeerConnectionInState(SignalingState::kHaveRemoteOffer);
>   auto answer = wrapper_for_answer->CreateAnswer();
> 
>-  auto wrapper = CreatePeerConnectionInState(GetParam());
>+  auto wrapper = CreatePeerConnectionUnderTest();
>   if (wrapper->signaling_state() == SignalingState::kHaveLocalOffer ||
>       wrapper->signaling_state() == SignalingState::kHaveRemotePrAnswer) {
>     EXPECT_TRUE(wrapper->SetRemoteDescription(std::move(answer)));
>@@ -375,46 +381,35 @@ TEST_P(PeerConnectionSignalingStateTest, SetRemoteAnswer) {
> 
> INSTANTIATE_TEST_CASE_P(PeerConnectionSignalingTest,
>                         PeerConnectionSignalingStateTest,
>-                        Combine(Values(SignalingState::kStable,
>+                        Combine(Values(SdpSemantics::kPlanB,
>+                                       SdpSemantics::kUnifiedPlan),
>+                                Values(SignalingState::kStable,
>                                        SignalingState::kHaveLocalOffer,
>                                        SignalingState::kHaveLocalPrAnswer,
>                                        SignalingState::kHaveRemoteOffer,
>                                        SignalingState::kHaveRemotePrAnswer),
>                                 Bool()));
> 
>-TEST_F(PeerConnectionSignalingTest,
>-       CreateAnswerSucceedsIfStableAndRemoteDescriptionIsOffer) {
>-  auto caller = CreatePeerConnection();
>-  auto callee = CreatePeerConnection();
>-
>-  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
>-  ASSERT_TRUE(
>-      caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));
>-
>-  ASSERT_EQ(SignalingState::kStable, callee->signaling_state());
>-  EXPECT_TRUE(callee->CreateAnswer());
>-}
>-
>-TEST_F(PeerConnectionSignalingTest,
>-       CreateAnswerFailsIfStableButRemoteDescriptionIsAnswer) {
>+// Test that CreateAnswer fails if a round of offer/answer has been done and
>+// the PeerConnection is in the stable state.
>+TEST_P(PeerConnectionSignalingTest, CreateAnswerFailsIfStable) {
>   auto caller = CreatePeerConnection();
>   auto callee = CreatePeerConnection();
> 
>-  ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
>-  ASSERT_TRUE(
>-      caller->SetRemoteDescription(callee->CreateAnswerAndSetAsLocal()));
>+  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
> 
>   ASSERT_EQ(SignalingState::kStable, caller->signaling_state());
>-  std::string error;
>-  ASSERT_FALSE(caller->CreateAnswer(RTCOfferAnswerOptions(), &error));
>-  EXPECT_EQ("CreateAnswer called without remote offer.", error);
>+  EXPECT_FALSE(caller->CreateAnswer());
>+
>+  ASSERT_EQ(SignalingState::kStable, callee->signaling_state());
>+  EXPECT_FALSE(callee->CreateAnswer());
> }
> 
> // According to https://tools.ietf.org/html/rfc3264#section-8, the session id
> // stays the same but the version must be incremented if a later, different
> // session description is generated. These two tests verify that is the case for
> // both offers and answers.
>-TEST_F(PeerConnectionSignalingTest,
>+TEST_P(PeerConnectionSignalingTest,
>        SessionVersionIncrementedInSubsequentDifferentOffer) {
>   auto caller = CreatePeerConnection();
>   auto callee = CreatePeerConnection();
>@@ -435,14 +430,14 @@ TEST_F(PeerConnectionSignalingTest,
>   EXPECT_LT(rtc::FromString<uint64_t>(original_version),
>             rtc::FromString<uint64_t>(later_offer->session_version()));
> }
>-TEST_F(PeerConnectionSignalingTest,
>+TEST_P(PeerConnectionSignalingTest,
>        SessionVersionIncrementedInSubsequentDifferentAnswer) {
>   auto caller = CreatePeerConnection();
>   auto callee = CreatePeerConnection();
> 
>   ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOfferAndSetAsLocal()));
> 
>-  auto original_answer = callee->CreateAnswerAndSetAsLocal();
>+  auto original_answer = callee->CreateAnswer();
>   const std::string original_id = original_answer->session_id();
>   const std::string original_version = original_answer->session_version();
> 
>@@ -456,7 +451,7 @@ TEST_F(PeerConnectionSignalingTest,
>             rtc::FromString<uint64_t>(later_answer->session_version()));
> }
> 
>-TEST_F(PeerConnectionSignalingTest, InitiatorFlagSetOnCallerAndNotOnCallee) {
>+TEST_P(PeerConnectionSignalingTest, InitiatorFlagSetOnCallerAndNotOnCallee) {
>   auto caller = CreatePeerConnectionWithAudioVideo();
>   auto callee = CreatePeerConnectionWithAudioVideo();
> 
>@@ -479,7 +474,7 @@ TEST_F(PeerConnectionSignalingTest, InitiatorFlagSetOnCallerAndNotOnCallee) {
> // PeerConnection and make sure we get success/failure callbacks for all of the
> // requests.
> // Background: crbug.com/507307
>-TEST_F(PeerConnectionSignalingTest, CreateOffersAndShutdown) {
>+TEST_P(PeerConnectionSignalingTest, CreateOffersAndShutdown) {
>   auto caller = CreatePeerConnection();
> 
>   RTCOfferAnswerOptions options;
>@@ -504,4 +499,9 @@ TEST_F(PeerConnectionSignalingTest, CreateOffersAndShutdown) {
>   }
> }
> 
>+INSTANTIATE_TEST_CASE_P(PeerConnectionSignalingTest,
>+                        PeerConnectionSignalingTest,
>+                        Values(SdpSemantics::kPlanB,
>+                               SdpSemantics::kUnifiedPlan));
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectionendtoend_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectionendtoend_unittest.cc
>index 17f48bf326e..7a8c2dbc12d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectionendtoend_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectionendtoend_unittest.cc
>@@ -10,15 +10,16 @@
> 
> #include <memory>
> 
>+#include "absl/memory/memory.h"
> #include "api/audio_codecs/L16/audio_decoder_L16.h"
> #include "api/audio_codecs/L16/audio_encoder_L16.h"
>+#include "api/audio_codecs/audio_codec_pair_id.h"
> #include "api/audio_codecs/audio_decoder_factory_template.h"
> #include "api/audio_codecs/audio_encoder_factory_template.h"
> #include "api/audio_codecs/builtin_audio_decoder_factory.h"
> #include "api/audio_codecs/builtin_audio_encoder_factory.h"
> #include "rtc_base/gunit.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/stringencode.h"
> #include "rtc_base/stringutils.h"
> 
>@@ -34,6 +35,7 @@
> using testing::AtLeast;
> using testing::Invoke;
> using testing::StrictMock;
>+using testing::Values;
> using testing::_;
> 
> using webrtc::DataChannelInterface;
>@@ -41,21 +43,20 @@ using webrtc::FakeConstraints;
> using webrtc::MediaConstraintsInterface;
> using webrtc::MediaStreamInterface;
> using webrtc::PeerConnectionInterface;
>+using webrtc::SdpSemantics;
> 
> namespace {
> 
>-const int kMaxWait = 10000;
>+const int kMaxWait = 25000;
> 
> }  // namespace
> 
>-class PeerConnectionEndToEndTest
>-    : public sigslot::has_slots<>,
>-      public testing::Test {
>+class PeerConnectionEndToEndBaseTest : public sigslot::has_slots<>,
>+                                       public testing::Test {
>  public:
>-  typedef std::vector<rtc::scoped_refptr<DataChannelInterface> >
>-      DataChannelList;
>+  typedef std::vector<rtc::scoped_refptr<DataChannelInterface>> DataChannelList;
> 
>-  PeerConnectionEndToEndTest() {
>+  explicit PeerConnectionEndToEndBaseTest(SdpSemantics sdp_semantics) {
>     network_thread_ = rtc::Thread::CreateWithSocketServer();
>     worker_thread_ = rtc::Thread::Create();
>     RTC_CHECK(network_thread_->Start());
>@@ -67,48 +68,56 @@ class PeerConnectionEndToEndTest
>     webrtc::PeerConnectionInterface::IceServer ice_server;
>     ice_server.uri = "stun:stun.l.google.com:19302";
>     config_.servers.push_back(ice_server);
>+    config_.sdp_semantics = sdp_semantics;
> 
> #ifdef WEBRTC_ANDROID
>     webrtc::InitializeAndroidObjects();
> #endif
>   }
> 
>-  void CreatePcs(const MediaConstraintsInterface* pc_constraints,
>-                 const rtc::scoped_refptr<webrtc::AudioEncoderFactory>&
>-                     audio_encoder_factory,
>-                 const rtc::scoped_refptr<webrtc::AudioDecoderFactory>&
>-                     audio_decoder_factory) {
>-    EXPECT_TRUE(caller_->CreatePc(
>-        pc_constraints, config_, audio_encoder_factory, audio_decoder_factory));
>-    EXPECT_TRUE(callee_->CreatePc(
>-        pc_constraints, config_, audio_encoder_factory, audio_decoder_factory));
>+  void CreatePcs(
>+      const MediaConstraintsInterface* pc_constraints,
>+      rtc::scoped_refptr<webrtc::AudioEncoderFactory> audio_encoder_factory1,
>+      rtc::scoped_refptr<webrtc::AudioDecoderFactory> audio_decoder_factory1,
>+      rtc::scoped_refptr<webrtc::AudioEncoderFactory> audio_encoder_factory2,
>+      rtc::scoped_refptr<webrtc::AudioDecoderFactory> audio_decoder_factory2) {
>+    EXPECT_TRUE(caller_->CreatePc(pc_constraints, config_,
>+                                  audio_encoder_factory1,
>+                                  audio_decoder_factory1));
>+    EXPECT_TRUE(callee_->CreatePc(pc_constraints, config_,
>+                                  audio_encoder_factory2,
>+                                  audio_decoder_factory2));
>     PeerConnectionTestWrapper::Connect(caller_.get(), callee_.get());
> 
>     caller_->SignalOnDataChannel.connect(
>-        this, &PeerConnectionEndToEndTest::OnCallerAddedDataChanel);
>+        this, &PeerConnectionEndToEndBaseTest::OnCallerAddedDataChanel);
>     callee_->SignalOnDataChannel.connect(
>-        this, &PeerConnectionEndToEndTest::OnCalleeAddedDataChannel);
>+        this, &PeerConnectionEndToEndBaseTest::OnCalleeAddedDataChannel);
>+  }
>+
>+  void CreatePcs(
>+      const MediaConstraintsInterface* pc_constraints,
>+      rtc::scoped_refptr<webrtc::AudioEncoderFactory> audio_encoder_factory,
>+      rtc::scoped_refptr<webrtc::AudioDecoderFactory> audio_decoder_factory) {
>+    CreatePcs(pc_constraints, audio_encoder_factory, audio_decoder_factory,
>+              audio_encoder_factory, audio_decoder_factory);
>   }
> 
>   void GetAndAddUserMedia() {
>-    FakeConstraints audio_constraints;
>+    cricket::AudioOptions audio_options;
>     FakeConstraints video_constraints;
>-    GetAndAddUserMedia(true, audio_constraints, true, video_constraints);
>+    GetAndAddUserMedia(true, audio_options, true, video_constraints);
>   }
> 
>   void GetAndAddUserMedia(bool audio,
>-                          const FakeConstraints& audio_constraints,
>+                          const cricket::AudioOptions& audio_options,
>                           bool video,
>                           const FakeConstraints& video_constraints) {
>-    caller_->GetAndAddUserMedia(audio, audio_constraints,
>-                                video, video_constraints);
>-    callee_->GetAndAddUserMedia(audio, audio_constraints,
>-                                video, video_constraints);
>+    caller_->GetAndAddUserMedia(audio, audio_options, video, video_constraints);
>+    callee_->GetAndAddUserMedia(audio, audio_options, video, video_constraints);
>   }
> 
>-  void Negotiate() {
>-    caller_->CreateOffer(NULL);
>-  }
>+  void Negotiate() { caller_->CreateOffer(NULL); }
> 
>   void WaitForCallEstablished() {
>     caller_->WaitForCallEstablished();
>@@ -129,24 +138,41 @@ class PeerConnectionEndToEndTest
>   }
> 
>   // Tests that |dc1| and |dc2| can send to and receive from each other.
>-  void TestDataChannelSendAndReceive(
>-      DataChannelInterface* dc1, DataChannelInterface* dc2) {
>+  void TestDataChannelSendAndReceive(DataChannelInterface* dc1,
>+                                     DataChannelInterface* dc2,
>+                                     size_t size = 6) {
>     std::unique_ptr<webrtc::MockDataChannelObserver> dc1_observer(
>         new webrtc::MockDataChannelObserver(dc1));
> 
>     std::unique_ptr<webrtc::MockDataChannelObserver> dc2_observer(
>         new webrtc::MockDataChannelObserver(dc2));
> 
>-    static const std::string kDummyData = "abcdefg";
>-    webrtc::DataBuffer buffer(kDummyData);
>+    static const std::string kDummyData =
>+        "ABCDEFGHIJKLMNOPQRSTUVWXYZabcdefghijklmnopqrstuvwxyz0123456789+/";
>+    webrtc::DataBuffer buffer("");
>+
>+    size_t sizeLeft = size;
>+    while (sizeLeft > 0) {
>+      size_t chunkSize =
>+          sizeLeft > kDummyData.length() ? kDummyData.length() : sizeLeft;
>+      buffer.data.AppendData(kDummyData.data(), chunkSize);
>+      sizeLeft -= chunkSize;
>+    }
>+
>     EXPECT_TRUE(dc1->Send(buffer));
>-    EXPECT_EQ_WAIT(kDummyData, dc2_observer->last_message(), kMaxWait);
>+    EXPECT_EQ_WAIT(buffer.data,
>+                   rtc::CopyOnWriteBuffer(dc2_observer->last_message()),
>+                   kMaxWait);
> 
>     EXPECT_TRUE(dc2->Send(buffer));
>-    EXPECT_EQ_WAIT(kDummyData, dc1_observer->last_message(), kMaxWait);
>+    EXPECT_EQ_WAIT(buffer.data,
>+                   rtc::CopyOnWriteBuffer(dc1_observer->last_message()),
>+                   kMaxWait);
> 
>     EXPECT_EQ(1U, dc1_observer->received_message_count());
>+    EXPECT_EQ(size, dc1_observer->last_message().length());
>     EXPECT_EQ(1U, dc2_observer->received_message_count());
>+    EXPECT_EQ(size, dc2_observer->last_message().length());
>   }
> 
>   void WaitForDataChannelsToOpen(DataChannelInterface* local_dc,
>@@ -154,10 +180,9 @@ class PeerConnectionEndToEndTest
>                                  size_t remote_dc_index) {
>     EXPECT_EQ_WAIT(DataChannelInterface::kOpen, local_dc->state(), kMaxWait);
> 
>-    EXPECT_TRUE_WAIT(remote_dc_list.size() > remote_dc_index, kMaxWait);
>+    ASSERT_TRUE_WAIT(remote_dc_list.size() > remote_dc_index, kMaxWait);
>     EXPECT_EQ_WAIT(DataChannelInterface::kOpen,
>-                   remote_dc_list[remote_dc_index]->state(),
>-                   kMaxWait);
>+                   remote_dc_list[remote_dc_index]->state(), kMaxWait);
>     EXPECT_EQ(local_dc->id(), remote_dc_list[remote_dc_index]->id());
>   }
> 
>@@ -167,8 +192,7 @@ class PeerConnectionEndToEndTest
>     local_dc->Close();
>     EXPECT_EQ_WAIT(DataChannelInterface::kClosed, local_dc->state(), kMaxWait);
>     EXPECT_EQ_WAIT(DataChannelInterface::kClosed,
>-                   remote_dc_list[remote_dc_index]->state(),
>-                   kMaxWait);
>+                   remote_dc_list[remote_dc_index]->state(), kMaxWait);
>   }
> 
>  protected:
>@@ -181,6 +205,13 @@ class PeerConnectionEndToEndTest
>   webrtc::PeerConnectionInterface::RTCConfiguration config_;
> };
> 
>+class PeerConnectionEndToEndTest
>+    : public PeerConnectionEndToEndBaseTest,
>+      public ::testing::WithParamInterface<SdpSemantics> {
>+ protected:
>+  PeerConnectionEndToEndTest() : PeerConnectionEndToEndBaseTest(GetParam()) {}
>+};
>+
> namespace {
> 
> std::unique_ptr<webrtc::AudioDecoder> CreateForwardingMockDecoder(
>@@ -196,7 +227,7 @@ std::unique_ptr<webrtc::AudioDecoder> CreateForwardingMockDecoder(
> 
>   const auto dec = real_decoder.get();  // For lambda capturing.
>   auto mock_decoder =
>-      rtc::MakeUnique<ForwardingMockDecoder>(std::move(real_decoder));
>+      absl::make_unique<ForwardingMockDecoder>(std::move(real_decoder));
>   EXPECT_CALL(*mock_decoder, Channels())
>       .Times(AtLeast(1))
>       .WillRepeatedly(Invoke([dec] { return dec->Channels(); }));
>@@ -251,13 +282,15 @@ CreateForwardingMockDecoderFactory(
>           Invoke([real_decoder_factory](const webrtc::SdpAudioFormat& format) {
>             return real_decoder_factory->IsSupportedDecoder(format);
>           }));
>-  EXPECT_CALL(*mock_decoder_factory, MakeAudioDecoderMock(_, _))
>+  EXPECT_CALL(*mock_decoder_factory, MakeAudioDecoderMock(_, _, _))
>       .Times(AtLeast(2))
>       .WillRepeatedly(
>           Invoke([real_decoder_factory](
>                      const webrtc::SdpAudioFormat& format,
>+                     absl::optional<webrtc::AudioCodecPairId> codec_pair_id,
>                      std::unique_ptr<webrtc::AudioDecoder>* return_value) {
>-            auto real_decoder = real_decoder_factory->MakeAudioDecoder(format);
>+            auto real_decoder =
>+                real_decoder_factory->MakeAudioDecoder(format, codec_pair_id);
>             *return_value =
>                 real_decoder
>                     ? CreateForwardingMockDecoder(std::move(real_decoder))
>@@ -268,7 +301,7 @@ CreateForwardingMockDecoderFactory(
> 
> struct AudioEncoderUnicornSparklesRainbow {
>   using Config = webrtc::AudioEncoderL16::Config;
>-  static rtc::Optional<Config> SdpToConfig(webrtc::SdpAudioFormat format) {
>+  static absl::optional<Config> SdpToConfig(webrtc::SdpAudioFormat format) {
>     if (STR_CASE_CMP(format.name.c_str(), "UnicornSparklesRainbow") == 0) {
>       const webrtc::SdpAudioFormat::Parameters expected_params = {
>           {"num_horns", "1"}};
>@@ -277,7 +310,7 @@ struct AudioEncoderUnicornSparklesRainbow {
>       format.name = "L16";
>       return webrtc::AudioEncoderL16::SdpToConfig(format);
>     } else {
>-      return rtc::nullopt;
>+      return absl::nullopt;
>     }
>   }
>   static void AppendSupportedEncoders(
>@@ -296,14 +329,16 @@ struct AudioEncoderUnicornSparklesRainbow {
>   }
>   static std::unique_ptr<webrtc::AudioEncoder> MakeAudioEncoder(
>       const Config& config,
>-      int payload_type) {
>-    return webrtc::AudioEncoderL16::MakeAudioEncoder(config, payload_type);
>+      int payload_type,
>+      absl::optional<webrtc::AudioCodecPairId> codec_pair_id = absl::nullopt) {
>+    return webrtc::AudioEncoderL16::MakeAudioEncoder(config, payload_type,
>+                                                     codec_pair_id);
>   }
> };
> 
> struct AudioDecoderUnicornSparklesRainbow {
>   using Config = webrtc::AudioDecoderL16::Config;
>-  static rtc::Optional<Config> SdpToConfig(webrtc::SdpAudioFormat format) {
>+  static absl::optional<Config> SdpToConfig(webrtc::SdpAudioFormat format) {
>     if (STR_CASE_CMP(format.name.c_str(), "UnicornSparklesRainbow") == 0) {
>       const webrtc::SdpAudioFormat::Parameters expected_params = {
>           {"num_horns", "1"}};
>@@ -312,7 +347,7 @@ struct AudioDecoderUnicornSparklesRainbow {
>       format.name = "L16";
>       return webrtc::AudioDecoderL16::SdpToConfig(format);
>     } else {
>-      return rtc::nullopt;
>+      return absl::nullopt;
>     }
>   }
>   static void AppendSupportedDecoders(
>@@ -327,23 +362,15 @@ struct AudioDecoderUnicornSparklesRainbow {
>     }
>   }
>   static std::unique_ptr<webrtc::AudioDecoder> MakeAudioDecoder(
>-      const Config& config) {
>-    return webrtc::AudioDecoderL16::MakeAudioDecoder(config);
>+      const Config& config,
>+      absl::optional<webrtc::AudioCodecPairId> codec_pair_id = absl::nullopt) {
>+    return webrtc::AudioDecoderL16::MakeAudioDecoder(config, codec_pair_id);
>   }
> };
> 
> }  // namespace
> 
>-// Disabled for TSan v2, see
>-// https://bugs.chromium.org/p/webrtc/issues/detail?id=4719 for details.
>-// Disabled for Mac, see
>-// https://bugs.chromium.org/p/webrtc/issues/detail?id=5231 for details.
>-#if defined(THREAD_SANITIZER) || defined(WEBRTC_MAC)
>-#define MAYBE_Call DISABLED_Call
>-#else
>-#define MAYBE_Call Call
>-#endif
>-TEST_F(PeerConnectionEndToEndTest, MAYBE_Call) {
>+TEST_P(PeerConnectionEndToEndTest, Call) {
>   rtc::scoped_refptr<webrtc::AudioDecoderFactory> real_decoder_factory =
>       webrtc::CreateBuiltinAudioDecoderFactory();
>   CreatePcs(nullptr, webrtc::CreateBuiltinAudioEncoderFactory(),
>@@ -353,8 +380,7 @@ TEST_F(PeerConnectionEndToEndTest, MAYBE_Call) {
>   WaitForCallEstablished();
> }
> 
>-#if !defined(ADDRESS_SANITIZER)
>-TEST_F(PeerConnectionEndToEndTest, CallWithLegacySdp) {
>+TEST_P(PeerConnectionEndToEndTest, CallWithLegacySdp) {
>   FakeConstraints pc_constraints;
>   pc_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
>                               false);
>@@ -364,22 +390,102 @@ TEST_F(PeerConnectionEndToEndTest, CallWithLegacySdp) {
>   Negotiate();
>   WaitForCallEstablished();
> }
>-#endif  // !defined(ADDRESS_SANITIZER)
> 
>-TEST_F(PeerConnectionEndToEndTest, CallWithCustomCodec) {
>-  CreatePcs(
>-      nullptr,
>-      webrtc::CreateAudioEncoderFactory<AudioEncoderUnicornSparklesRainbow>(),
>-      webrtc::CreateAudioDecoderFactory<AudioDecoderUnicornSparklesRainbow>());
>+TEST_P(PeerConnectionEndToEndTest, CallWithCustomCodec) {
>+  class IdLoggingAudioEncoderFactory : public webrtc::AudioEncoderFactory {
>+   public:
>+    IdLoggingAudioEncoderFactory(
>+        rtc::scoped_refptr<AudioEncoderFactory> real_factory,
>+        std::vector<webrtc::AudioCodecPairId>* const codec_ids)
>+        : fact_(real_factory), codec_ids_(codec_ids) {}
>+    std::vector<webrtc::AudioCodecSpec> GetSupportedEncoders() override {
>+      return fact_->GetSupportedEncoders();
>+    }
>+    absl::optional<webrtc::AudioCodecInfo> QueryAudioEncoder(
>+        const webrtc::SdpAudioFormat& format) override {
>+      return fact_->QueryAudioEncoder(format);
>+    }
>+    std::unique_ptr<webrtc::AudioEncoder> MakeAudioEncoder(
>+        int payload_type,
>+        const webrtc::SdpAudioFormat& format,
>+        absl::optional<webrtc::AudioCodecPairId> codec_pair_id) override {
>+      EXPECT_TRUE(codec_pair_id.has_value());
>+      codec_ids_->push_back(*codec_pair_id);
>+      return fact_->MakeAudioEncoder(payload_type, format, codec_pair_id);
>+    }
>+
>+   private:
>+    const rtc::scoped_refptr<webrtc::AudioEncoderFactory> fact_;
>+    std::vector<webrtc::AudioCodecPairId>* const codec_ids_;
>+  };
>+
>+  class IdLoggingAudioDecoderFactory : public webrtc::AudioDecoderFactory {
>+   public:
>+    IdLoggingAudioDecoderFactory(
>+        rtc::scoped_refptr<AudioDecoderFactory> real_factory,
>+        std::vector<webrtc::AudioCodecPairId>* const codec_ids)
>+        : fact_(real_factory), codec_ids_(codec_ids) {}
>+    std::vector<webrtc::AudioCodecSpec> GetSupportedDecoders() override {
>+      return fact_->GetSupportedDecoders();
>+    }
>+    bool IsSupportedDecoder(const webrtc::SdpAudioFormat& format) override {
>+      return fact_->IsSupportedDecoder(format);
>+    }
>+    std::unique_ptr<webrtc::AudioDecoder> MakeAudioDecoder(
>+        const webrtc::SdpAudioFormat& format,
>+        absl::optional<webrtc::AudioCodecPairId> codec_pair_id) override {
>+      EXPECT_TRUE(codec_pair_id.has_value());
>+      codec_ids_->push_back(*codec_pair_id);
>+      return fact_->MakeAudioDecoder(format, codec_pair_id);
>+    }
>+
>+   private:
>+    const rtc::scoped_refptr<webrtc::AudioDecoderFactory> fact_;
>+    std::vector<webrtc::AudioCodecPairId>* const codec_ids_;
>+  };
>+
>+  std::vector<webrtc::AudioCodecPairId> encoder_id1, encoder_id2, decoder_id1,
>+      decoder_id2;
>+  CreatePcs(nullptr,
>+            rtc::scoped_refptr<webrtc::AudioEncoderFactory>(
>+                new rtc::RefCountedObject<IdLoggingAudioEncoderFactory>(
>+                    webrtc::CreateAudioEncoderFactory<
>+                        AudioEncoderUnicornSparklesRainbow>(),
>+                    &encoder_id1)),
>+            rtc::scoped_refptr<webrtc::AudioDecoderFactory>(
>+                new rtc::RefCountedObject<IdLoggingAudioDecoderFactory>(
>+                    webrtc::CreateAudioDecoderFactory<
>+                        AudioDecoderUnicornSparklesRainbow>(),
>+                    &decoder_id1)),
>+            rtc::scoped_refptr<webrtc::AudioEncoderFactory>(
>+                new rtc::RefCountedObject<IdLoggingAudioEncoderFactory>(
>+                    webrtc::CreateAudioEncoderFactory<
>+                        AudioEncoderUnicornSparklesRainbow>(),
>+                    &encoder_id2)),
>+            rtc::scoped_refptr<webrtc::AudioDecoderFactory>(
>+                new rtc::RefCountedObject<IdLoggingAudioDecoderFactory>(
>+                    webrtc::CreateAudioDecoderFactory<
>+                        AudioDecoderUnicornSparklesRainbow>(),
>+                    &decoder_id2)));
>   GetAndAddUserMedia();
>   Negotiate();
>   WaitForCallEstablished();
>+
>+  // Each codec factory has been used to create one codec. The first pair got
>+  // the same ID because they were passed to the same PeerConnectionFactory,
>+  // and the second pair got the same ID---but these two IDs are not equal,
>+  // because each PeerConnectionFactory has its own ID.
>+  EXPECT_EQ(1U, encoder_id1.size());
>+  EXPECT_EQ(1U, encoder_id2.size());
>+  EXPECT_EQ(encoder_id1, decoder_id1);
>+  EXPECT_EQ(encoder_id2, decoder_id2);
>+  EXPECT_NE(encoder_id1, encoder_id2);
> }
> 
> #ifdef HAVE_SCTP
> // Verifies that a DataChannel created before the negotiation can transition to
> // "OPEN" and transfer data.
>-TEST_F(PeerConnectionEndToEndTest, CreateDataChannelBeforeNegotiate) {
>+TEST_P(PeerConnectionEndToEndTest, CreateDataChannelBeforeNegotiate) {
>   CreatePcs(nullptr, webrtc::CreateBuiltinAudioEncoderFactory(),
>             webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
> 
>@@ -404,7 +510,7 @@ TEST_F(PeerConnectionEndToEndTest, CreateDataChannelBeforeNegotiate) {
> 
> // Verifies that a DataChannel created after the negotiation can transition to
> // "OPEN" and transfer data.
>-TEST_F(PeerConnectionEndToEndTest, CreateDataChannelAfterNegotiate) {
>+TEST_P(PeerConnectionEndToEndTest, CreateDataChannelAfterNegotiate) {
>   CreatePcs(nullptr, webrtc::CreateBuiltinAudioEncoderFactory(),
>             webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
> 
>@@ -435,8 +541,42 @@ TEST_F(PeerConnectionEndToEndTest, CreateDataChannelAfterNegotiate) {
>   CloseDataChannels(callee_dc, caller_signaled_data_channels_, 0);
> }
> 
>+// Verifies that a DataChannel created can transfer large messages.
>+TEST_P(PeerConnectionEndToEndTest, CreateDataChannelLargeTransfer) {
>+  CreatePcs(nullptr, webrtc::CreateBuiltinAudioEncoderFactory(),
>+            webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
>+
>+  webrtc::DataChannelInit init;
>+
>+  // This DataChannel is for creating the data content in the negotiation.
>+  rtc::scoped_refptr<DataChannelInterface> dummy(
>+      caller_->CreateDataChannel("data", init));
>+  Negotiate();
>+  WaitForConnection();
>+
>+  // Wait for the data channel created pre-negotiation to be opened.
>+  WaitForDataChannelsToOpen(dummy, callee_signaled_data_channels_, 0);
>+
>+  // Create new DataChannels after the negotiation and verify their states.
>+  rtc::scoped_refptr<DataChannelInterface> caller_dc(
>+      caller_->CreateDataChannel("hello", init));
>+  rtc::scoped_refptr<DataChannelInterface> callee_dc(
>+      callee_->CreateDataChannel("hello", init));
>+
>+  WaitForDataChannelsToOpen(caller_dc, callee_signaled_data_channels_, 1);
>+  WaitForDataChannelsToOpen(callee_dc, caller_signaled_data_channels_, 0);
>+
>+  TestDataChannelSendAndReceive(caller_dc, callee_signaled_data_channels_[1],
>+                                256 * 1024);
>+  TestDataChannelSendAndReceive(callee_dc, caller_signaled_data_channels_[0],
>+                                256 * 1024);
>+
>+  CloseDataChannels(caller_dc, callee_signaled_data_channels_, 1);
>+  CloseDataChannels(callee_dc, caller_signaled_data_channels_, 0);
>+}
>+
> // Verifies that DataChannel IDs are even/odd based on the DTLS roles.
>-TEST_F(PeerConnectionEndToEndTest, DataChannelIdAssignment) {
>+TEST_P(PeerConnectionEndToEndTest, DataChannelIdAssignment) {
>   CreatePcs(nullptr, webrtc::CreateBuiltinAudioEncoderFactory(),
>             webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
> 
>@@ -449,21 +589,21 @@ TEST_F(PeerConnectionEndToEndTest, DataChannelIdAssignment) {
>   Negotiate();
>   WaitForConnection();
> 
>-  EXPECT_EQ(1U, caller_dc_1->id() % 2);
>-  EXPECT_EQ(0U, callee_dc_1->id() % 2);
>+  EXPECT_EQ(1, caller_dc_1->id() % 2);
>+  EXPECT_EQ(0, callee_dc_1->id() % 2);
> 
>   rtc::scoped_refptr<DataChannelInterface> caller_dc_2(
>       caller_->CreateDataChannel("data", init));
>   rtc::scoped_refptr<DataChannelInterface> callee_dc_2(
>       callee_->CreateDataChannel("data", init));
> 
>-  EXPECT_EQ(1U, caller_dc_2->id() % 2);
>-  EXPECT_EQ(0U, callee_dc_2->id() % 2);
>+  EXPECT_EQ(1, caller_dc_2->id() % 2);
>+  EXPECT_EQ(0, callee_dc_2->id() % 2);
> }
> 
> // Verifies that the message is received by the right remote DataChannel when
> // there are multiple DataChannels.
>-TEST_F(PeerConnectionEndToEndTest,
>+TEST_P(PeerConnectionEndToEndTest,
>        MessageTransferBetweenTwoPairsOfDataChannels) {
>   CreatePcs(nullptr, webrtc::CreateBuiltinAudioEncoderFactory(),
>             webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
>@@ -501,14 +641,10 @@ TEST_F(PeerConnectionEndToEndTest,
> 
> // Verifies that a DataChannel added from an OPEN message functions after
> // a channel has been previously closed (webrtc issue 3778).
>-// This previously failed because the new channel re-uses the ID of the closed
>-// channel, and the closed channel was incorrectly still assigned to the id.
>-// TODO(deadbeef): This is disabled because there's currently a race condition
>-// caused by the fact that a data channel signals that it's closed before it
>-// really is. Re-enable this test once that's fixed.
>-// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4453
>-TEST_F(PeerConnectionEndToEndTest,
>-       DISABLED_DataChannelFromOpenWorksAfterClose) {
>+// This previously failed because the new channel re-used the ID of the closed
>+// channel, and the closed channel was incorrectly still assigned to the ID.
>+TEST_P(PeerConnectionEndToEndTest,
>+       DataChannelFromOpenWorksAfterPreviousChannelClosed) {
>   CreatePcs(nullptr, webrtc::CreateBuiltinAudioEncoderFactory(),
>             webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
> 
>@@ -520,12 +656,49 @@ TEST_F(PeerConnectionEndToEndTest,
>   WaitForConnection();
> 
>   WaitForDataChannelsToOpen(caller_dc, callee_signaled_data_channels_, 0);
>-  CloseDataChannels(caller_dc, callee_signaled_data_channels_, 0);
>+  int first_channel_id = caller_dc->id();
>+  // Wait for the local side to say it's closed, but not the remote side.
>+  // Previously, the channel on which Close is called reported being closed
>+  // prematurely, and this caused issues; see bugs.webrtc.org/4453.
>+  caller_dc->Close();
>+  EXPECT_EQ_WAIT(DataChannelInterface::kClosed, caller_dc->state(), kMaxWait);
> 
>   // Create a new channel and ensure it works after closing the previous one.
>   caller_dc = caller_->CreateDataChannel("data2", init);
>+  WaitForDataChannelsToOpen(caller_dc, callee_signaled_data_channels_, 1);
>+  // Since the second channel was created after the first finished closing, it
>+  // should be able to re-use the first one's ID.
>+  EXPECT_EQ(first_channel_id, caller_dc->id());
>+  TestDataChannelSendAndReceive(caller_dc, callee_signaled_data_channels_[1]);
>+
>+  CloseDataChannels(caller_dc, callee_signaled_data_channels_, 1);
>+}
> 
>+// Similar to the above test, but don't wait for the first channel to finish
>+// closing before creating the second one.
>+TEST_P(PeerConnectionEndToEndTest,
>+       DataChannelFromOpenWorksWhilePreviousChannelClosing) {
>+  CreatePcs(nullptr, webrtc::CreateBuiltinAudioEncoderFactory(),
>+            webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
>+
>+  webrtc::DataChannelInit init;
>+  rtc::scoped_refptr<DataChannelInterface> caller_dc(
>+      caller_->CreateDataChannel("data", init));
>+
>+  Negotiate();
>+  WaitForConnection();
>+
>+  WaitForDataChannelsToOpen(caller_dc, callee_signaled_data_channels_, 0);
>+  int first_channel_id = caller_dc->id();
>+  caller_dc->Close();
>+
>+  // Immediately create a new channel, before waiting for the previous one to
>+  // transition to "closed".
>+  caller_dc = caller_->CreateDataChannel("data2", init);
>   WaitForDataChannelsToOpen(caller_dc, callee_signaled_data_channels_, 1);
>+  // Since the second channel was created while the first was still closing,
>+  // it should have been assigned a different ID.
>+  EXPECT_NE(first_channel_id, caller_dc->id());
>   TestDataChannelSendAndReceive(caller_dc, callee_signaled_data_channels_[1]);
> 
>   CloseDataChannels(caller_dc, callee_signaled_data_channels_, 1);
>@@ -535,7 +708,7 @@ TEST_F(PeerConnectionEndToEndTest,
> // by the application (meaning only the PeerConnection contributes to its
> // reference count), no memory access violation will occur.
> // See: https://code.google.com/p/chromium/issues/detail?id=565048
>-TEST_F(PeerConnectionEndToEndTest, CloseDataChannelRemotelyWhileNotReferenced) {
>+TEST_P(PeerConnectionEndToEndTest, CloseDataChannelRemotelyWhileNotReferenced) {
>   CreatePcs(nullptr, webrtc::CreateBuiltinAudioEncoderFactory(),
>             webrtc::MockAudioDecoderFactory::CreateEmptyFactory());
> 
>@@ -557,3 +730,8 @@ TEST_F(PeerConnectionEndToEndTest, CloseDataChannelRemotelyWhileNotReferenced) {
>   rtc::Thread::Current()->ProcessMessages(100);
> }
> #endif  // HAVE_SCTP
>+
>+INSTANTIATE_TEST_CASE_P(PeerConnectionEndToEndTest,
>+                        PeerConnectionEndToEndTest,
>+                        Values(SdpSemantics::kPlanB,
>+                               SdpSemantics::kUnifiedPlan));
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectionfactory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectionfactory.cc
>index d9fcce2ae63..a6db5198692 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectionfactory.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectionfactory.cc
>@@ -11,7 +11,10 @@
> #include "pc/peerconnectionfactory.h"
> 
> #include <utility>
>+#include <vector>
> 
>+#include "absl/memory/memory.h"
>+#include "api/fec_controller.h"
> #include "api/mediaconstraintsinterface.h"
> #include "api/mediastreamproxy.h"
> #include "api/mediastreamtrackproxy.h"
>@@ -22,9 +25,9 @@
> #include "logging/rtc_event_log/rtc_event_log.h"
> #include "media/base/rtpdataengine.h"
> #include "media/sctp/sctptransport.h"
>+#include "pc/rtpparametersconversion.h"
> #include "rtc_base/bind.h"
> #include "rtc_base/checks.h"
>-#include "rtc_base/ptr_util.h"
> // Adding 'nogncheck' to disable the gn include headers check to support modular
> // WebRTC build targets.
> // TODO(zhihuang): This wouldn't be necessary if the interface and
>@@ -33,6 +36,7 @@
> #include "media/engine/webrtcvideodecoderfactory.h"     // nogncheck
> #include "media/engine/webrtcvideoencoderfactory.h"     // nogncheck
> #include "modules/audio_device/include/audio_device.h"  // nogncheck
>+#include "modules/congestion_controller/bbr/bbr_factory.h"
> #include "p2p/base/basicpacketsocketfactory.h"
> #include "p2p/client/basicportallocator.h"
> #include "pc/audiotrack.h"
>@@ -41,6 +45,7 @@
> #include "pc/peerconnection.h"
> #include "pc/videocapturertracksource.h"
> #include "pc/videotrack.h"
>+#include "rtc_base/experiments/congestion_controller_experiment.h"
> 
> namespace webrtc {
> 
>@@ -52,12 +57,46 @@ CreateModularPeerConnectionFactory(
>     std::unique_ptr<cricket::MediaEngineInterface> media_engine,
>     std::unique_ptr<CallFactoryInterface> call_factory,
>     std::unique_ptr<RtcEventLogFactoryInterface> event_log_factory) {
>+  PeerConnectionFactoryDependencies dependencies;
>+  dependencies.network_thread = network_thread;
>+  dependencies.worker_thread = worker_thread;
>+  dependencies.signaling_thread = signaling_thread;
>+  dependencies.media_engine = std::move(media_engine);
>+  dependencies.call_factory = std::move(call_factory);
>+  dependencies.event_log_factory = std::move(event_log_factory);
>+  return CreateModularPeerConnectionFactory(std::move(dependencies));
>+}
>+
>+rtc::scoped_refptr<PeerConnectionFactoryInterface>
>+CreateModularPeerConnectionFactory(
>+    rtc::Thread* network_thread,
>+    rtc::Thread* worker_thread,
>+    rtc::Thread* signaling_thread,
>+    std::unique_ptr<cricket::MediaEngineInterface> media_engine,
>+    std::unique_ptr<CallFactoryInterface> call_factory,
>+    std::unique_ptr<RtcEventLogFactoryInterface> event_log_factory,
>+    std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory,
>+    std::unique_ptr<NetworkControllerFactoryInterface>
>+        network_controller_factory) {
>+  PeerConnectionFactoryDependencies dependencies;
>+  dependencies.network_thread = network_thread;
>+  dependencies.worker_thread = worker_thread;
>+  dependencies.signaling_thread = signaling_thread;
>+  dependencies.media_engine = std::move(media_engine);
>+  dependencies.call_factory = std::move(call_factory);
>+  dependencies.event_log_factory = std::move(event_log_factory);
>+  dependencies.fec_controller_factory = std::move(fec_controller_factory);
>+  dependencies.network_controller_factory =
>+      std::move(network_controller_factory);
>+  return CreateModularPeerConnectionFactory(std::move(dependencies));
>+}
>+
>+rtc::scoped_refptr<PeerConnectionFactoryInterface>
>+CreateModularPeerConnectionFactory(
>+    PeerConnectionFactoryDependencies dependencies) {
>   rtc::scoped_refptr<PeerConnectionFactory> pc_factory(
>       new rtc::RefCountedObject<PeerConnectionFactory>(
>-          network_thread, worker_thread, signaling_thread,
>-          std::move(media_engine), std::move(call_factory),
>-          std::move(event_log_factory)));
>-
>+          std::move(dependencies)));
>   // Call Initialize synchronously but make sure it is executed on
>   // |signaling_thread|.
>   MethodCall0<PeerConnectionFactory, bool> call(
>@@ -78,13 +117,37 @@ PeerConnectionFactory::PeerConnectionFactory(
>     std::unique_ptr<cricket::MediaEngineInterface> media_engine,
>     std::unique_ptr<webrtc::CallFactoryInterface> call_factory,
>     std::unique_ptr<RtcEventLogFactoryInterface> event_log_factory)
>+    : PeerConnectionFactory(network_thread,
>+                            worker_thread,
>+                            signaling_thread,
>+                            std::move(media_engine),
>+                            std::move(call_factory),
>+                            std::move(event_log_factory),
>+                            nullptr,
>+                            nullptr) {}
>+
>+PeerConnectionFactory::PeerConnectionFactory(
>+    rtc::Thread* network_thread,
>+    rtc::Thread* worker_thread,
>+    rtc::Thread* signaling_thread,
>+    std::unique_ptr<cricket::MediaEngineInterface> media_engine,
>+    std::unique_ptr<webrtc::CallFactoryInterface> call_factory,
>+    std::unique_ptr<RtcEventLogFactoryInterface> event_log_factory,
>+    std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory,
>+    std::unique_ptr<NetworkControllerFactoryInterface>
>+        network_controller_factory)
>     : wraps_current_thread_(false),
>       network_thread_(network_thread),
>       worker_thread_(worker_thread),
>       signaling_thread_(signaling_thread),
>       media_engine_(std::move(media_engine)),
>       call_factory_(std::move(call_factory)),
>-      event_log_factory_(std::move(event_log_factory)) {
>+      event_log_factory_(std::move(event_log_factory)),
>+      fec_controller_factory_(std::move(fec_controller_factory)),
>+      injected_network_controller_factory_(
>+          std::move(network_controller_factory)),
>+      bbr_network_controller_factory_(
>+          absl::make_unique<BbrNetworkControllerFactory>()) {
>   if (!network_thread_) {
>     owned_network_thread_ = rtc::Thread::CreateWithSocketServer();
>     owned_network_thread_->SetName("pc_network_thread", nullptr);
>@@ -114,6 +177,18 @@ PeerConnectionFactory::PeerConnectionFactory(
>   // RTC_DCHECK(default_adm != NULL);
> }
> 
>+PeerConnectionFactory::PeerConnectionFactory(
>+    PeerConnectionFactoryDependencies dependencies)
>+    : PeerConnectionFactory(
>+          dependencies.network_thread,
>+          dependencies.worker_thread,
>+          dependencies.signaling_thread,
>+          std::move(dependencies.media_engine),
>+          std::move(dependencies.call_factory),
>+          std::move(dependencies.event_log_factory),
>+          std::move(dependencies.fec_controller_factory),
>+          std::move(dependencies.network_controller_factory)) {}
>+
> PeerConnectionFactory::~PeerConnectionFactory() {
>   RTC_DCHECK(signaling_thread_->IsCurrent());
>   channel_manager_.reset(nullptr);
>@@ -142,8 +217,8 @@ bool PeerConnectionFactory::Initialize() {
>     return false;
>   }
> 
>-  channel_manager_ = rtc::MakeUnique<cricket::ChannelManager>(
>-      std::move(media_engine_), rtc::MakeUnique<cricket::RtpDataEngine>(),
>+  channel_manager_ = absl::make_unique<cricket::ChannelManager>(
>+      std::move(media_engine_), absl::make_unique<cricket::RtpDataEngine>(),
>       worker_thread_, network_thread_);
> 
>   channel_manager_->SetVideoRtxEnabled(true);
>@@ -158,13 +233,58 @@ void PeerConnectionFactory::SetOptions(const Options& options) {
>   options_ = options;
> }
> 
>-rtc::scoped_refptr<AudioSourceInterface>
>-PeerConnectionFactory::CreateAudioSource(
>-    const MediaConstraintsInterface* constraints) {
>-  RTC_DCHECK(signaling_thread_->IsCurrent());
>-  rtc::scoped_refptr<LocalAudioSource> source(
>-      LocalAudioSource::Create(constraints));
>-  return source;
>+RtpCapabilities PeerConnectionFactory::GetRtpSenderCapabilities(
>+    cricket::MediaType kind) const {
>+  RTC_DCHECK_RUN_ON(signaling_thread_);
>+  switch (kind) {
>+    case cricket::MEDIA_TYPE_AUDIO: {
>+      cricket::AudioCodecs cricket_codecs;
>+      cricket::RtpHeaderExtensions cricket_extensions;
>+      channel_manager_->GetSupportedAudioSendCodecs(&cricket_codecs);
>+      channel_manager_->GetSupportedAudioRtpHeaderExtensions(
>+          &cricket_extensions);
>+      return ToRtpCapabilities(cricket_codecs, cricket_extensions);
>+    }
>+    case cricket::MEDIA_TYPE_VIDEO: {
>+      cricket::VideoCodecs cricket_codecs;
>+      cricket::RtpHeaderExtensions cricket_extensions;
>+      channel_manager_->GetSupportedVideoCodecs(&cricket_codecs);
>+      channel_manager_->GetSupportedVideoRtpHeaderExtensions(
>+          &cricket_extensions);
>+      return ToRtpCapabilities(cricket_codecs, cricket_extensions);
>+    }
>+    case cricket::MEDIA_TYPE_DATA:
>+      return RtpCapabilities();
>+  }
>+  // Not reached; avoids compile warning.
>+  RTC_FATAL();
>+}
>+
>+RtpCapabilities PeerConnectionFactory::GetRtpReceiverCapabilities(
>+    cricket::MediaType kind) const {
>+  RTC_DCHECK_RUN_ON(signaling_thread_);
>+  switch (kind) {
>+    case cricket::MEDIA_TYPE_AUDIO: {
>+      cricket::AudioCodecs cricket_codecs;
>+      cricket::RtpHeaderExtensions cricket_extensions;
>+      channel_manager_->GetSupportedAudioReceiveCodecs(&cricket_codecs);
>+      channel_manager_->GetSupportedAudioRtpHeaderExtensions(
>+          &cricket_extensions);
>+      return ToRtpCapabilities(cricket_codecs, cricket_extensions);
>+    }
>+    case cricket::MEDIA_TYPE_VIDEO: {
>+      cricket::VideoCodecs cricket_codecs;
>+      cricket::RtpHeaderExtensions cricket_extensions;
>+      channel_manager_->GetSupportedVideoCodecs(&cricket_codecs);
>+      channel_manager_->GetSupportedVideoRtpHeaderExtensions(
>+          &cricket_extensions);
>+      return ToRtpCapabilities(cricket_codecs, cricket_extensions);
>+    }
>+    case cricket::MEDIA_TYPE_DATA:
>+      return RtpCapabilities();
>+  }
>+  // Not reached; avoids compile warning.
>+  RTC_FATAL();
> }
> 
> rtc::scoped_refptr<AudioSourceInterface>
>@@ -232,22 +352,40 @@ PeerConnectionFactory::CreatePeerConnection(
>     std::unique_ptr<cricket::PortAllocator> allocator,
>     std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
>     PeerConnectionObserver* observer) {
>+  // Convert the legacy API into the new depnedency structure.
>+  PeerConnectionDependencies dependencies(observer);
>+  dependencies.allocator = std::move(allocator);
>+  dependencies.cert_generator = std::move(cert_generator);
>+  // Pass that into the new API.
>+  return CreatePeerConnection(configuration, std::move(dependencies));
>+}
>+
>+rtc::scoped_refptr<PeerConnectionInterface>
>+PeerConnectionFactory::CreatePeerConnection(
>+    const PeerConnectionInterface::RTCConfiguration& configuration,
>+    PeerConnectionDependencies dependencies) {
>   RTC_DCHECK(signaling_thread_->IsCurrent());
> 
>-  if (!cert_generator.get()) {
>-    // No certificate generator specified, use the default one.
>-    cert_generator.reset(
>-        new rtc::RTCCertificateGenerator(signaling_thread_, network_thread_));
>+  // Set internal defaults if optional dependencies are not set.
>+  if (!dependencies.cert_generator) {
>+    dependencies.cert_generator =
>+        absl::make_unique<rtc::RTCCertificateGenerator>(signaling_thread_,
>+                                                        network_thread_);
>   }
>-
>-  if (!allocator) {
>-    allocator.reset(new cricket::BasicPortAllocator(
>+  if (!dependencies.allocator) {
>+    dependencies.allocator.reset(new cricket::BasicPortAllocator(
>         default_network_manager_.get(), default_socket_factory_.get(),
>         configuration.turn_customizer));
>   }
>+
>+  // TODO(zstein): Once chromium injects its own AsyncResolverFactory, set
>+  // |dependencies.async_resolver_factory| to a new
>+  // |rtc::BasicAsyncResolverFactory| if no factory is provided.
>+
>   network_thread_->Invoke<void>(
>-      RTC_FROM_HERE, rtc::Bind(&cricket::PortAllocator::SetNetworkIgnoreMask,
>-                               allocator.get(), options_.network_ignore_mask));
>+      RTC_FROM_HERE,
>+      rtc::Bind(&cricket::PortAllocator::SetNetworkIgnoreMask,
>+                dependencies.allocator.get(), options_.network_ignore_mask));
> 
>   std::unique_ptr<RtcEventLog> event_log =
>       worker_thread_->Invoke<std::unique_ptr<RtcEventLog>>(
>@@ -261,19 +399,18 @@ PeerConnectionFactory::CreatePeerConnection(
>   rtc::scoped_refptr<PeerConnection> pc(
>       new rtc::RefCountedObject<PeerConnection>(this, std::move(event_log),
>                                                 std::move(call)));
>-
>-  if (!pc->Initialize(configuration, std::move(allocator),
>-                      std::move(cert_generator), observer)) {
>+  ActionsBeforeInitializeForTesting(pc);
>+  if (!pc->Initialize(configuration, std::move(dependencies))) {
>     return nullptr;
>   }
>   return PeerConnectionProxy::Create(signaling_thread(), pc);
> }
> 
> rtc::scoped_refptr<MediaStreamInterface>
>-PeerConnectionFactory::CreateLocalMediaStream(const std::string& label) {
>+PeerConnectionFactory::CreateLocalMediaStream(const std::string& stream_id) {
>   RTC_DCHECK(signaling_thread_->IsCurrent());
>   return MediaStreamProxy::Create(signaling_thread_,
>-                                  MediaStream::Create(label));
>+                                  MediaStream::Create(stream_id));
> }
> 
> rtc::scoped_refptr<VideoTrackInterface> PeerConnectionFactory::CreateVideoTrack(
>@@ -285,27 +422,18 @@ rtc::scoped_refptr<VideoTrackInterface> PeerConnectionFactory::CreateVideoTrack(
>   return VideoTrackProxy::Create(signaling_thread_, worker_thread_, track);
> }
> 
>-rtc::scoped_refptr<AudioTrackInterface>
>-PeerConnectionFactory::CreateAudioTrack(const std::string& id,
>-                                        AudioSourceInterface* source) {
>+rtc::scoped_refptr<AudioTrackInterface> PeerConnectionFactory::CreateAudioTrack(
>+    const std::string& id,
>+    AudioSourceInterface* source) {
>   RTC_DCHECK(signaling_thread_->IsCurrent());
>   rtc::scoped_refptr<AudioTrackInterface> track(AudioTrack::Create(id, source));
>   return AudioTrackProxy::Create(signaling_thread_, track);
> }
> 
>-cricket::TransportController* PeerConnectionFactory::CreateTransportController(
>-    cricket::PortAllocator* port_allocator,
>-    bool redetermine_role_on_ice_restart) {
>-  RTC_DCHECK(signaling_thread_->IsCurrent());
>-  return new cricket::TransportController(
>-      signaling_thread_, network_thread_, port_allocator,
>-      redetermine_role_on_ice_restart, options_.crypto_options);
>-}
>-
> std::unique_ptr<cricket::SctpTransportInternalFactory>
> PeerConnectionFactory::CreateSctpTransportInternalFactory() {
> #ifdef HAVE_SCTP
>-  return rtc::MakeUnique<cricket::SctpTransportFactory>(network_thread());
>+  return absl::make_unique<cricket::SctpTransportFactory>(network_thread());
> #else
>   return nullptr;
> #endif
>@@ -334,7 +462,7 @@ std::unique_ptr<RtcEventLog> PeerConnectionFactory::CreateRtcEventLog_w() {
>   const auto encoding_type = RtcEventLog::EncodingType::Legacy;
>   return event_log_factory_
>              ? event_log_factory_->CreateRtcEventLog(encoding_type)
>-             : rtc::MakeUnique<RtcEventLogNullImpl>();
>+             : absl::make_unique<RtcEventLogNullImpl>();
> }
> 
> std::unique_ptr<Call> PeerConnectionFactory::CreateCall_w(
>@@ -354,6 +482,20 @@ std::unique_ptr<Call> PeerConnectionFactory::CreateCall_w(
>   call_config.bitrate_config.start_bitrate_bps = kStartBandwidthBps;
>   call_config.bitrate_config.max_bitrate_bps = kMaxBandwidthBps;
> 
>+  call_config.fec_controller_factory = fec_controller_factory_.get();
>+
>+  if (CongestionControllerExperiment::BbrControllerEnabled()) {
>+    RTC_LOG(LS_INFO) << "Using BBR network controller factory";
>+    call_config.network_controller_factory =
>+        bbr_network_controller_factory_.get();
>+  } else if (CongestionControllerExperiment::InjectedControllerEnabled()) {
>+    RTC_LOG(LS_INFO) << "Using injected network controller factory";
>+    call_config.network_controller_factory =
>+        injected_network_controller_factory_.get();
>+  } else {
>+    RTC_LOG(LS_INFO) << "Using default network controller factory";
>+  }
>+
>   return std::unique_ptr<Call>(call_factory_->CreateCall(call_config));
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectionfactory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectionfactory.h
>index 398221ffdee..82fa50859f7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectionfactory.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectionfactory.h
>@@ -1,3 +1,4 @@
>+
> /*
>  *  Copyright 2011 The WebRTC project authors. All Rights Reserved.
>  *
>@@ -25,7 +26,7 @@
> namespace rtc {
> class BasicNetworkManager;
> class BasicPacketSocketFactory;
>-}
>+}  // namespace rtc
> 
> namespace webrtc {
> 
>@@ -55,16 +56,23 @@ class PeerConnectionFactory : public PeerConnectionFactoryInterface {
>       std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
>       PeerConnectionObserver* observer) override;
> 
>+  rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection(
>+      const PeerConnectionInterface::RTCConfiguration& configuration,
>+      PeerConnectionDependencies dependencies) override;
>+
>   bool Initialize();
> 
>-  rtc::scoped_refptr<MediaStreamInterface>
>-      CreateLocalMediaStream(const std::string& label) override;
>+  RtpCapabilities GetRtpSenderCapabilities(
>+      cricket::MediaType kind) const override;
>+
>+  RtpCapabilities GetRtpReceiverCapabilities(
>+      cricket::MediaType kind) const override;
>+
>+  rtc::scoped_refptr<MediaStreamInterface> CreateLocalMediaStream(
>+      const std::string& stream_id) override;
> 
>   rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource(
>       const cricket::AudioOptions& options) override;
>-  // Deprecated, use version without constraints.
>-  rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource(
>-      const MediaConstraintsInterface* constraints) override;
> 
>   rtc::scoped_refptr<VideoTrackSourceInterface> CreateVideoSource(
>       std::unique_ptr<cricket::VideoCapturer> capturer) override;
>@@ -80,17 +88,13 @@ class PeerConnectionFactory : public PeerConnectionFactoryInterface {
>       const std::string& id,
>       VideoTrackSourceInterface* video_source) override;
> 
>-  rtc::scoped_refptr<AudioTrackInterface>
>-      CreateAudioTrack(const std::string& id,
>-                       AudioSourceInterface* audio_source) override;
>+  rtc::scoped_refptr<AudioTrackInterface> CreateAudioTrack(
>+      const std::string& id,
>+      AudioSourceInterface* audio_source) override;
> 
>   bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) override;
>   void StopAecDump() override;
> 
>-  virtual cricket::TransportController* CreateTransportController(
>-      cricket::PortAllocator* port_allocator,
>-      bool redetermine_role_on_ice_restart);
>-
>   virtual std::unique_ptr<cricket::SctpTransportInternalFactory>
>   CreateSctpTransportInternalFactory();
> 
>@@ -108,6 +112,26 @@ class PeerConnectionFactory : public PeerConnectionFactoryInterface {
>       std::unique_ptr<cricket::MediaEngineInterface> media_engine,
>       std::unique_ptr<webrtc::CallFactoryInterface> call_factory,
>       std::unique_ptr<RtcEventLogFactoryInterface> event_log_factory);
>+  PeerConnectionFactory(
>+      rtc::Thread* network_thread,
>+      rtc::Thread* worker_thread,
>+      rtc::Thread* signaling_thread,
>+      std::unique_ptr<cricket::MediaEngineInterface> media_engine,
>+      std::unique_ptr<webrtc::CallFactoryInterface> call_factory,
>+      std::unique_ptr<RtcEventLogFactoryInterface> event_log_factory,
>+      std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory,
>+      std::unique_ptr<NetworkControllerFactoryInterface>
>+          network_controller_factory);
>+  // Use this implementation for all future use. This structure allows simple
>+  // management of all new dependencies being added to the
>+  // PeerConnectionFactory.
>+  explicit PeerConnectionFactory(
>+      PeerConnectionFactoryDependencies dependencies);
>+
>+  // Hook to let testing framework insert actions between
>+  // "new RTCPeerConnection" and "pc.Initialize"
>+  virtual void ActionsBeforeInitializeForTesting(PeerConnectionInterface*) {}
>+
>   virtual ~PeerConnectionFactory();
> 
>  private:
>@@ -127,6 +151,11 @@ class PeerConnectionFactory : public PeerConnectionFactoryInterface {
>   std::unique_ptr<cricket::MediaEngineInterface> media_engine_;
>   std::unique_ptr<webrtc::CallFactoryInterface> call_factory_;
>   std::unique_ptr<RtcEventLogFactoryInterface> event_log_factory_;
>+  std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory_;
>+  std::unique_ptr<NetworkControllerFactoryInterface>
>+      injected_network_controller_factory_;
>+  std::unique_ptr<NetworkControllerFactoryInterface>
>+      bbr_network_controller_factory_;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectionfactory_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectionfactory_unittest.cc
>index a1bdbf5af5b..773ba0198e5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectionfactory_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectionfactory_unittest.cc
>@@ -16,6 +16,8 @@
> #include "api/audio_codecs/builtin_audio_decoder_factory.h"
> #include "api/audio_codecs/builtin_audio_encoder_factory.h"
> #include "api/mediastreaminterface.h"
>+#include "api/video_codecs/builtin_video_decoder_factory.h"
>+#include "api/video_codecs/builtin_video_encoder_factory.h"
> #include "media/base/fakevideocapturer.h"
> #include "p2p/base/fakeportallocator.h"
> #include "pc/peerconnectionfactory.h"
>@@ -43,14 +45,12 @@ static const char kStunIceServer[] = "stun:stun.l.google.com:19302";
> static const char kTurnIceServer[] = "turn:test%40hello.com@test.com:1234";
> static const char kTurnIceServerWithTransport[] =
>     "turn:test@hello.com?transport=tcp";
>-static const char kSecureTurnIceServer[] =
>-    "turns:test@hello.com?transport=tcp";
>+static const char kSecureTurnIceServer[] = "turns:test@hello.com?transport=tcp";
> static const char kSecureTurnIceServerWithoutTransportParam[] =
>     "turns:test_no_transport@hello.com:443";
> static const char kSecureTurnIceServerWithoutTransportAndPortParam[] =
>     "turns:test_no_transport@hello.com";
>-static const char kTurnIceServerWithNoUsernameInUri[] =
>-    "turn:test.com:1234";
>+static const char kTurnIceServerWithNoUsernameInUri[] = "turn:test.com:1234";
> static const char kTurnPassword[] = "turnpassword";
> static const int kDefaultStunPort = 3478;
> static const int kDefaultStunTlsPort = 5349;
>@@ -78,8 +78,8 @@ class NullPeerConnectionObserver : public PeerConnectionObserver {
>       PeerConnectionInterface::IceConnectionState new_state) override {}
>   void OnIceGatheringChange(
>       PeerConnectionInterface::IceGatheringState new_state) override {}
>-  void OnIceCandidate(
>-      const webrtc::IceCandidateInterface* candidate) override {}
>+  void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) override {
>+  }
> };
> 
> }  // namespace
>@@ -93,10 +93,14 @@ class PeerConnectionFactoryTest : public testing::Test {
>     // level, and using a real one could make tests flaky e.g. when run in
>     // parallel.
>     factory_ = webrtc::CreatePeerConnectionFactory(
>-        rtc::Thread::Current(), rtc::Thread::Current(),
>-        FakeAudioCaptureModule::Create(),
>+        rtc::Thread::Current(), rtc::Thread::Current(), rtc::Thread::Current(),
>+        rtc::scoped_refptr<webrtc::AudioDeviceModule>(
>+            FakeAudioCaptureModule::Create()),
>         webrtc::CreateBuiltinAudioEncoderFactory(),
>-        webrtc::CreateBuiltinAudioDecoderFactory(), nullptr, nullptr);
>+        webrtc::CreateBuiltinAudioDecoderFactory(),
>+        webrtc::CreateBuiltinVideoEncoderFactory(),
>+        webrtc::CreateBuiltinVideoDecoderFactory(), nullptr /* audio_mixer */,
>+        nullptr /* audio_processing */);
> 
>     ASSERT_TRUE(factory_.get() != NULL);
>     port_allocator_.reset(
>@@ -126,6 +130,19 @@ class PeerConnectionFactoryTest : public testing::Test {
>     }
>   }
> 
>+  void VerifyAudioCodecCapability(const webrtc::RtpCodecCapability& codec) {
>+    EXPECT_EQ(codec.kind, cricket::MEDIA_TYPE_AUDIO);
>+    EXPECT_FALSE(codec.name.empty());
>+    EXPECT_GT(codec.clock_rate, 0);
>+    EXPECT_GT(codec.num_channels, 0);
>+  }
>+
>+  void VerifyVideoCodecCapability(const webrtc::RtpCodecCapability& codec) {
>+    EXPECT_EQ(codec.kind, cricket::MEDIA_TYPE_VIDEO);
>+    EXPECT_FALSE(codec.name.empty());
>+    EXPECT_GT(codec.clock_rate, 0);
>+  }
>+
>   rtc::scoped_refptr<PeerConnectionFactoryInterface> factory_;
>   NullPeerConnectionObserver observer_;
>   std::unique_ptr<cricket::FakePortAllocator> port_allocator_;
>@@ -147,8 +164,13 @@ TEST(PeerConnectionFactoryTestInternal, DISABLED_CreatePCUsingInternalModules) {
> 
>   rtc::scoped_refptr<PeerConnectionFactoryInterface> factory(
>       webrtc::CreatePeerConnectionFactory(
>+          nullptr /* network_thread */, nullptr /* worker_thread */,
>+          nullptr /* signaling_thread */, nullptr /* default_adm */,
>           webrtc::CreateBuiltinAudioEncoderFactory(),
>-          webrtc::CreateBuiltinAudioDecoderFactory()));
>+          webrtc::CreateBuiltinAudioDecoderFactory(),
>+          webrtc::CreateBuiltinVideoEncoderFactory(),
>+          webrtc::CreateBuiltinVideoDecoderFactory(), nullptr /* audio_mixer */,
>+          nullptr /* audio_processing */));
> 
>   NullPeerConnectionObserver observer;
>   webrtc::PeerConnectionInterface::RTCConfiguration config;
>@@ -161,6 +183,72 @@ TEST(PeerConnectionFactoryTestInternal, DISABLED_CreatePCUsingInternalModules) {
>   EXPECT_TRUE(pc.get() != nullptr);
> }
> 
>+TEST_F(PeerConnectionFactoryTest, CheckRtpSenderAudioCapabilities) {
>+  webrtc::RtpCapabilities audio_capabilities =
>+      factory_->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_AUDIO);
>+  EXPECT_FALSE(audio_capabilities.codecs.empty());
>+  for (const auto& codec : audio_capabilities.codecs) {
>+    VerifyAudioCodecCapability(codec);
>+  }
>+  EXPECT_FALSE(audio_capabilities.header_extensions.empty());
>+  for (const auto& header_extension : audio_capabilities.header_extensions) {
>+    EXPECT_FALSE(header_extension.uri.empty());
>+  }
>+}
>+
>+TEST_F(PeerConnectionFactoryTest, CheckRtpSenderVideoCapabilities) {
>+  webrtc::RtpCapabilities video_capabilities =
>+      factory_->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_VIDEO);
>+  EXPECT_FALSE(video_capabilities.codecs.empty());
>+  for (const auto& codec : video_capabilities.codecs) {
>+    VerifyVideoCodecCapability(codec);
>+  }
>+  EXPECT_FALSE(video_capabilities.header_extensions.empty());
>+  for (const auto& header_extension : video_capabilities.header_extensions) {
>+    EXPECT_FALSE(header_extension.uri.empty());
>+  }
>+}
>+
>+TEST_F(PeerConnectionFactoryTest, CheckRtpSenderDataCapabilities) {
>+  webrtc::RtpCapabilities data_capabilities =
>+      factory_->GetRtpSenderCapabilities(cricket::MEDIA_TYPE_DATA);
>+  EXPECT_TRUE(data_capabilities.codecs.empty());
>+  EXPECT_TRUE(data_capabilities.header_extensions.empty());
>+}
>+
>+TEST_F(PeerConnectionFactoryTest, CheckRtpReceiverAudioCapabilities) {
>+  webrtc::RtpCapabilities audio_capabilities =
>+      factory_->GetRtpReceiverCapabilities(cricket::MEDIA_TYPE_AUDIO);
>+  EXPECT_FALSE(audio_capabilities.codecs.empty());
>+  for (const auto& codec : audio_capabilities.codecs) {
>+    VerifyAudioCodecCapability(codec);
>+  }
>+  EXPECT_FALSE(audio_capabilities.header_extensions.empty());
>+  for (const auto& header_extension : audio_capabilities.header_extensions) {
>+    EXPECT_FALSE(header_extension.uri.empty());
>+  }
>+}
>+
>+TEST_F(PeerConnectionFactoryTest, CheckRtpReceiverVideoCapabilities) {
>+  webrtc::RtpCapabilities video_capabilities =
>+      factory_->GetRtpReceiverCapabilities(cricket::MEDIA_TYPE_VIDEO);
>+  EXPECT_FALSE(video_capabilities.codecs.empty());
>+  for (const auto& codec : video_capabilities.codecs) {
>+    VerifyVideoCodecCapability(codec);
>+  }
>+  EXPECT_FALSE(video_capabilities.header_extensions.empty());
>+  for (const auto& header_extension : video_capabilities.header_extensions) {
>+    EXPECT_FALSE(header_extension.uri.empty());
>+  }
>+}
>+
>+TEST_F(PeerConnectionFactoryTest, CheckRtpReceiverDataCapabilities) {
>+  webrtc::RtpCapabilities data_capabilities =
>+      factory_->GetRtpReceiverCapabilities(cricket::MEDIA_TYPE_DATA);
>+  EXPECT_TRUE(data_capabilities.codecs.empty());
>+  EXPECT_TRUE(data_capabilities.header_extensions.empty());
>+}
>+
> // This test verifies creation of PeerConnection with valid STUN and TURN
> // configuration. Also verifies the URL's parsed correctly as expected.
> TEST_F(PeerConnectionFactoryTest, CreatePCUsingIceServers) {
>@@ -341,7 +429,8 @@ TEST_F(PeerConnectionFactoryTest, CreatePCUsingIPLiteralAddress) {
> // This test verifies the captured stream is rendered locally using a
> // local video track.
> TEST_F(PeerConnectionFactoryTest, LocalRendering) {
>-  cricket::FakeVideoCapturer* capturer = new cricket::FakeVideoCapturer();
>+  cricket::FakeVideoCapturerWithTaskQueue* capturer =
>+      new cricket::FakeVideoCapturerWithTaskQueue();
>   // The source takes ownership of |capturer|, but we keep a raw pointer to
>   // inject fake frames.
>   rtc::scoped_refptr<VideoTrackSourceInterface> source(
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectioninterface_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectioninterface_unittest.cc
>index a06ed50086e..1beb34721cc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectioninterface_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectioninterface_unittest.cc
>@@ -14,6 +14,7 @@
> #include <string>
> #include <utility>
> 
>+#include "absl/memory/memory.h"
> #include "api/audio_codecs/builtin_audio_decoder_factory.h"
> #include "api/audio_codecs/builtin_audio_encoder_factory.h"
> #include "api/jsepsessiondescription.h"
>@@ -22,6 +23,8 @@
> #include "api/rtpreceiverinterface.h"
> #include "api/rtpsenderinterface.h"
> #include "api/test/fakeconstraints.h"
>+#include "api/video_codecs/builtin_video_decoder_factory.h"
>+#include "api/video_codecs/builtin_video_encoder_factory.h"
> #include "logging/rtc_event_log/output/rtc_event_log_output_file.h"
> #include "media/base/fakevideocapturer.h"
> #include "media/engine/webrtcmediaengine.h"
>@@ -32,6 +35,7 @@
> #include "pc/mediasession.h"
> #include "pc/mediastream.h"
> #include "pc/peerconnection.h"
>+#include "pc/rtpsender.h"
> #include "pc/streamcollection.h"
> #include "pc/test/fakeaudiocapturemodule.h"
> #include "pc/test/fakertccertificategenerator.h"
>@@ -41,7 +45,6 @@
> #include "pc/videocapturertracksource.h"
> #include "pc/videotrack.h"
> #include "rtc_base/gunit.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/stringutils.h"
> #include "rtc_base/virtualsocketserver.h"
> #include "test/gmock.h"
>@@ -51,9 +54,9 @@
> #include "pc/test/androidtestinitializer.h"
> #endif
> 
>-static const char kStreamLabel1[] = "local_stream_1";
>-static const char kStreamLabel2[] = "local_stream_2";
>-static const char kStreamLabel3[] = "local_stream_3";
>+static const char kStreamId1[] = "local_stream_1";
>+static const char kStreamId2[] = "local_stream_2";
>+static const char kStreamId3[] = "local_stream_3";
> static const int kDefaultStunPort = 3478;
> static const char kStunAddressOnly[] = "stun:address";
> static const char kStunInvalidPort[] = "stun:address:-1";
>@@ -74,7 +77,7 @@ static const char kSendrecv[] = "sendrecv";
> 
> // Reference SDP with a MediaStream with label "stream1" and audio track with
> // id "audio_1" and a video track with id "video_1;
>-static const char kSdpStringWithStream1[] =
>+static const char kSdpStringWithStream1PlanB[] =
>     "v=0\r\n"
>     "o=- 0 0 IN IP4 127.0.0.1\r\n"
>     "s=-\r\n"
>@@ -103,6 +106,38 @@ static const char kSdpStringWithStream1[] =
>     "a=ssrc:2 cname:stream1\r\n"
>     "a=ssrc:2 mslabel:stream1\r\n"
>     "a=ssrc:2 label:videotrack0\r\n";
>+// Same string as above but with the MID changed to the Unified Plan default.
>+// This is needed so that this SDP can be used as an answer for a Unified Plan
>+// offer.
>+static const char kSdpStringWithStream1UnifiedPlan[] =
>+    "v=0\r\n"
>+    "o=- 0 0 IN IP4 127.0.0.1\r\n"
>+    "s=-\r\n"
>+    "t=0 0\r\n"
>+    "m=audio 1 RTP/AVPF 103\r\n"
>+    "a=ice-ufrag:e5785931\r\n"
>+    "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
>+    "a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
>+    "BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
>+    "a=mid:0\r\n"
>+    "a=sendrecv\r\n"
>+    "a=rtcp-mux\r\n"
>+    "a=rtpmap:103 ISAC/16000\r\n"
>+    "a=ssrc:1 cname:stream1\r\n"
>+    "a=ssrc:1 mslabel:stream1\r\n"
>+    "a=ssrc:1 label:audiotrack0\r\n"
>+    "m=video 1 RTP/AVPF 120\r\n"
>+    "a=ice-ufrag:e5785931\r\n"
>+    "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
>+    "a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
>+    "BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
>+    "a=mid:1\r\n"
>+    "a=sendrecv\r\n"
>+    "a=rtcp-mux\r\n"
>+    "a=rtpmap:120 VP8/90000\r\n"
>+    "a=ssrc:2 cname:stream1\r\n"
>+    "a=ssrc:2 mslabel:stream1\r\n"
>+    "a=ssrc:2 label:videotrack0\r\n";
> 
> // Reference SDP with a MediaStream with label "stream1" and audio track with
> // id "audio_1";
>@@ -127,7 +162,7 @@ static const char kSdpStringWithStream1AudioTrackOnly[] =
> // Reference SDP with two MediaStreams with label "stream1" and "stream2. Each
> // MediaStreams have one audio track and one video track.
> // This uses MSID.
>-static const char kSdpStringWithStream1And2[] =
>+static const char kSdpStringWithStream1And2PlanB[] =
>     "v=0\r\n"
>     "o=- 0 0 IN IP4 127.0.0.1\r\n"
>     "s=-\r\n"
>@@ -159,6 +194,56 @@ static const char kSdpStringWithStream1And2[] =
>     "a=ssrc:2 msid:stream1 videotrack0\r\n"
>     "a=ssrc:4 cname:stream2\r\n"
>     "a=ssrc:4 msid:stream2 videotrack1\r\n";
>+static const char kSdpStringWithStream1And2UnifiedPlan[] =
>+    "v=0\r\n"
>+    "o=- 0 0 IN IP4 127.0.0.1\r\n"
>+    "s=-\r\n"
>+    "t=0 0\r\n"
>+    "a=msid-semantic: WMS stream1 stream2\r\n"
>+    "m=audio 1 RTP/AVPF 103\r\n"
>+    "a=ice-ufrag:e5785931\r\n"
>+    "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
>+    "a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
>+    "BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
>+    "a=mid:0\r\n"
>+    "a=sendrecv\r\n"
>+    "a=rtcp-mux\r\n"
>+    "a=rtpmap:103 ISAC/16000\r\n"
>+    "a=ssrc:1 cname:stream1\r\n"
>+    "a=ssrc:1 msid:stream1 audiotrack0\r\n"
>+    "m=video 1 RTP/AVPF 120\r\n"
>+    "a=ice-ufrag:e5785931\r\n"
>+    "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
>+    "a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
>+    "BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
>+    "a=mid:1\r\n"
>+    "a=sendrecv\r\n"
>+    "a=rtcp-mux\r\n"
>+    "a=rtpmap:120 VP8/0\r\n"
>+    "a=ssrc:2 cname:stream1\r\n"
>+    "a=ssrc:2 msid:stream1 videotrack0\r\n"
>+    "m=audio 1 RTP/AVPF 103\r\n"
>+    "a=ice-ufrag:e5785931\r\n"
>+    "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
>+    "a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
>+    "BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
>+    "a=mid:2\r\n"
>+    "a=sendrecv\r\n"
>+    "a=rtcp-mux\r\n"
>+    "a=rtpmap:103 ISAC/16000\r\n"
>+    "a=ssrc:3 cname:stream2\r\n"
>+    "a=ssrc:3 msid:stream2 audiotrack1\r\n"
>+    "m=video 1 RTP/AVPF 120\r\n"
>+    "a=ice-ufrag:e5785931\r\n"
>+    "a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
>+    "a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
>+    "BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
>+    "a=mid:3\r\n"
>+    "a=sendrecv\r\n"
>+    "a=rtcp-mux\r\n"
>+    "a=rtpmap:120 VP8/0\r\n"
>+    "a=ssrc:4 cname:stream2\r\n"
>+    "a=ssrc:4 msid:stream2 videotrack1\r\n";
> 
> // Reference SDP without MediaStreams. Msid is not supported.
> static const char kSdpStringWithoutStreams[] =
>@@ -320,11 +405,12 @@ static const char kDtlsSdesFallbackSdp[] =
>     "a=fingerprint:sha-1 "
>     "4A:AD:B9:B1:3F:82:18:3B:54:02:12:DF:3E:5D:49:6B:19:E5:7C:AB\r\n"
>     "a=setup:actpass\r\n"
>-    "a=crypto:1 AES_CM_128_HMAC_SHA1_32 "
>+    "a=crypto:0 AES_CM_128_HMAC_SHA1_80 "
>     "inline:NzB4d1BINUAvLEw6UzF3WSJ+PSdFcGdUJShpX1Zj|2^20|1:32 "
>     "dummy_session_params\r\n";
> 
> using ::testing::Exactly;
>+using ::testing::Values;
> using cricket::StreamParams;
> using webrtc::AudioSourceInterface;
> using webrtc::AudioTrack;
>@@ -350,8 +436,11 @@ using webrtc::RTCError;
> using webrtc::RTCErrorType;
> using webrtc::RtpReceiverInterface;
> using webrtc::RtpSenderInterface;
>+using webrtc::RtpSenderProxyWithInternal;
>+using webrtc::RtpSenderInternal;
> using webrtc::RtpTransceiverDirection;
> using webrtc::SdpParseError;
>+using webrtc::SdpSemantics;
> using webrtc::SdpType;
> using webrtc::SessionDescriptionInterface;
> using webrtc::StreamCollection;
>@@ -360,7 +449,8 @@ using webrtc::VideoTrackSourceInterface;
> using webrtc::VideoTrack;
> using webrtc::VideoTrackInterface;
> 
>-typedef PeerConnectionInterface::RTCOfferAnswerOptions RTCOfferAnswerOptions;
>+using RTCConfiguration = PeerConnectionInterface::RTCConfiguration;
>+using RTCOfferAnswerOptions = PeerConnectionInterface::RTCOfferAnswerOptions;
> 
> namespace {
> 
>@@ -395,7 +485,7 @@ void SetSsrcToZero(std::string* sdp) {
>   const char kSdpSsrcAtributeZero[] = "a=ssrc:0";
>   size_t ssrc_pos = 0;
>   while ((ssrc_pos = sdp->find(kSdpSsrcAtribute, ssrc_pos)) !=
>-      std::string::npos) {
>+         std::string::npos) {
>     size_t end_ssrc = sdp->find(" ", ssrc_pos);
>     sdp->replace(ssrc_pos, end_ssrc - ssrc_pos, kSdpSsrcAtributeZero);
>     ssrc_pos = end_ssrc;
>@@ -404,10 +494,10 @@ void SetSsrcToZero(std::string* sdp) {
> 
> // Check if |streams| contains the specified track.
> bool ContainsTrack(const std::vector<cricket::StreamParams>& streams,
>-                   const std::string& stream_label,
>+                   const std::string& stream_id,
>                    const std::string& track_id) {
>   for (const cricket::StreamParams& params : streams) {
>-    if (params.sync_label == stream_label && params.id == track_id) {
>+    if (params.first_stream_id() == stream_id && params.id == track_id) {
>       return true;
>     }
>   }
>@@ -481,7 +571,7 @@ bool CompareStreamCollections(StreamCollectionInterface* s1,
>   }
> 
>   for (size_t i = 0; i != s1->count(); ++i) {
>-    if (s1->at(i)->label() != s2->at(i)->label()) {
>+    if (s1->at(i)->id() != s2->at(i)->id()) {
>       return false;
>     }
>     webrtc::AudioTrackVector audio_tracks1 = s1->at(i)->GetAudioTracks();
>@@ -544,13 +634,16 @@ class PeerConnectionFactoryForTest : public webrtc::PeerConnectionFactory {
>   CreatePeerConnectionFactoryForTest() {
>     auto audio_encoder_factory = webrtc::CreateBuiltinAudioEncoderFactory();
>     auto audio_decoder_factory = webrtc::CreateBuiltinAudioDecoderFactory();
>+    auto video_encoder_factory = webrtc::CreateBuiltinVideoEncoderFactory();
>+    auto video_decoder_factory = webrtc::CreateBuiltinVideoDecoderFactory();
> 
>     // Use fake audio device module since we're only testing the interface
>     // level, and using a real one could make tests flaky when run in parallel.
>     auto media_engine = std::unique_ptr<cricket::MediaEngineInterface>(
>         cricket::WebRtcMediaEngineFactory::Create(
>             FakeAudioCaptureModule::Create(), audio_encoder_factory,
>-            audio_decoder_factory, nullptr, nullptr, nullptr,
>+            audio_decoder_factory, std::move(video_encoder_factory),
>+            std::move(video_decoder_factory), nullptr,
>             webrtc::AudioProcessingBuilder().Create()));
> 
>     std::unique_ptr<webrtc::CallFactoryInterface> call_factory =
>@@ -579,25 +672,16 @@ class PeerConnectionFactoryForTest : public webrtc::PeerConnectionFactory {
>                                       std::move(call_factory),
>                                       std::move(event_log_factory)) {}
> 
>-  cricket::TransportController* CreateTransportController(
>-      cricket::PortAllocator* port_allocator,
>-      bool redetermine_role_on_ice_restart) override {
>-    transport_controller = new cricket::TransportController(
>-        rtc::Thread::Current(), rtc::Thread::Current(), port_allocator,
>-        redetermine_role_on_ice_restart, rtc::CryptoOptions());
>-    return transport_controller;
>-  }
>-
>   rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_;
>-  cricket::TransportController* transport_controller;
> };
> 
>-// TODO(steveanton): Convert to use the new PeerConnectionUnitTestFixture and
>-// PeerConnectionWrapper.
>-class PeerConnectionInterfaceTest : public testing::Test {
>+// TODO(steveanton): Convert to use the new PeerConnectionWrapper.
>+class PeerConnectionInterfaceBaseTest : public testing::Test {
>  protected:
>-  PeerConnectionInterfaceTest()
>-      : vss_(new rtc::VirtualSocketServer()), main_(vss_.get()) {
>+  explicit PeerConnectionInterfaceBaseTest(SdpSemantics sdp_semantics)
>+      : vss_(new rtc::VirtualSocketServer()),
>+        main_(vss_.get()),
>+        sdp_semantics_(sdp_semantics) {
> #ifdef WEBRTC_ANDROID
>     webrtc::InitializeAndroidObjects();
> #endif
>@@ -609,8 +693,13 @@ class PeerConnectionInterfaceTest : public testing::Test {
>     fake_audio_capture_module_ = FakeAudioCaptureModule::Create();
>     pc_factory_ = webrtc::CreatePeerConnectionFactory(
>         rtc::Thread::Current(), rtc::Thread::Current(), rtc::Thread::Current(),
>-        fake_audio_capture_module_, webrtc::CreateBuiltinAudioEncoderFactory(),
>-        webrtc::CreateBuiltinAudioDecoderFactory(), nullptr, nullptr);
>+        rtc::scoped_refptr<webrtc::AudioDeviceModule>(
>+            fake_audio_capture_module_),
>+        webrtc::CreateBuiltinAudioEncoderFactory(),
>+        webrtc::CreateBuiltinAudioDecoderFactory(),
>+        webrtc::CreateBuiltinVideoEncoderFactory(),
>+        webrtc::CreateBuiltinVideoDecoderFactory(), nullptr /* audio_mixer */,
>+        nullptr /* audio_processing */);
>     ASSERT_TRUE(pc_factory_);
>     pc_factory_for_test_ =
>         PeerConnectionFactoryForTest::CreatePeerConnectionFactoryForTest();
>@@ -667,13 +756,15 @@ class PeerConnectionInterfaceTest : public testing::Test {
>     bool dtls;
>     if (FindConstraint(constraints,
>                        webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
>-                       &dtls,
>-                       nullptr) && dtls) {
>+                       &dtls, nullptr) &&
>+        dtls) {
>       fake_certificate_generator_ = new FakeRTCCertificateGenerator();
>       cert_generator.reset(fake_certificate_generator_);
>     }
>+    RTCConfiguration modified_config = config;
>+    modified_config.sdp_semantics = sdp_semantics_;
>     pc_ = pc_factory_->CreatePeerConnection(
>-        config, constraints, std::move(port_allocator),
>+        modified_config, constraints, std::move(port_allocator),
>         std::move(cert_generator), &observer_);
>     ASSERT_TRUE(pc_.get() != NULL);
>     observer_.SetPeerConnectionInterface(pc_.get());
>@@ -685,10 +776,10 @@ class PeerConnectionInterfaceTest : public testing::Test {
>     PeerConnectionInterface::IceServer server;
>     server.uri = uri;
>     config.servers.push_back(server);
>-
>-    rtc::scoped_refptr<PeerConnectionInterface> pc;
>-    pc = pc_factory_->CreatePeerConnection(config, nullptr, nullptr, nullptr,
>-                                           &observer_);
>+    config.sdp_semantics = sdp_semantics_;
>+    rtc::scoped_refptr<PeerConnectionInterface> pc =
>+        pc_factory_->CreatePeerConnection(config, nullptr, nullptr, nullptr,
>+                                          &observer_);
>     EXPECT_EQ(nullptr, pc);
>   }
> 
>@@ -698,6 +789,7 @@ class PeerConnectionInterfaceTest : public testing::Test {
>     server.uri = kTurnIceServerUri;
>     server.password = kTurnPassword;
>     config.servers.push_back(server);
>+    config.sdp_semantics = sdp_semantics_;
>     rtc::scoped_refptr<PeerConnectionInterface> pc =
>         pc_factory_->CreatePeerConnection(config, nullptr, nullptr, &observer_);
>     EXPECT_EQ(nullptr, pc);
>@@ -731,61 +823,82 @@ class PeerConnectionInterfaceTest : public testing::Test {
>     observer_.SetPeerConnectionInterface(NULL);
>   }
> 
>+  rtc::scoped_refptr<VideoTrackInterface> CreateVideoTrack(
>+      const std::string& label) {
>+    auto video_source = pc_factory_->CreateVideoSource(
>+        absl::make_unique<cricket::FakeVideoCapturer>(), nullptr);
>+    return pc_factory_->CreateVideoTrack(label, video_source);
>+  }
>+
>+  void AddVideoTrack(const std::string& track_label,
>+                     const std::vector<std::string>& stream_ids = {}) {
>+    auto sender_or_error =
>+        pc_->AddTrack(CreateVideoTrack(track_label), stream_ids);
>+    ASSERT_EQ(RTCErrorType::NONE, sender_or_error.error().type());
>+    EXPECT_TRUE_WAIT(observer_.renegotiation_needed_, kTimeout);
>+    observer_.renegotiation_needed_ = false;
>+  }
>+
>   void AddVideoStream(const std::string& label) {
>-    // Create a local stream.
>     rtc::scoped_refptr<MediaStreamInterface> stream(
>         pc_factory_->CreateLocalMediaStream(label));
>-    rtc::scoped_refptr<VideoTrackSourceInterface> video_source(
>-        pc_factory_->CreateVideoSource(std::unique_ptr<cricket::VideoCapturer>(
>-                                           new cricket::FakeVideoCapturer()),
>-                                       NULL));
>-    rtc::scoped_refptr<VideoTrackInterface> video_track(
>-        pc_factory_->CreateVideoTrack(label + "v0", video_source));
>-    stream->AddTrack(video_track.get());
>-    EXPECT_TRUE(pc_->AddStream(stream));
>+    stream->AddTrack(CreateVideoTrack(label + "v0"));
>+    ASSERT_TRUE(pc_->AddStream(stream));
>     EXPECT_TRUE_WAIT(observer_.renegotiation_needed_, kTimeout);
>     observer_.renegotiation_needed_ = false;
>   }
> 
>-  void AddVoiceStream(const std::string& label) {
>-    // Create a local stream.
>+  rtc::scoped_refptr<AudioTrackInterface> CreateAudioTrack(
>+      const std::string& label) {
>+    return pc_factory_->CreateAudioTrack(label, nullptr);
>+  }
>+
>+  void AddAudioTrack(const std::string& track_label,
>+                     const std::vector<std::string>& stream_ids = {}) {
>+    auto sender_or_error =
>+        pc_->AddTrack(CreateAudioTrack(track_label), stream_ids);
>+    ASSERT_EQ(RTCErrorType::NONE, sender_or_error.error().type());
>+    EXPECT_TRUE_WAIT(observer_.renegotiation_needed_, kTimeout);
>+    observer_.renegotiation_needed_ = false;
>+  }
>+
>+  void AddAudioStream(const std::string& label) {
>     rtc::scoped_refptr<MediaStreamInterface> stream(
>         pc_factory_->CreateLocalMediaStream(label));
>-    rtc::scoped_refptr<AudioTrackInterface> audio_track(
>-        pc_factory_->CreateAudioTrack(label + "a0", NULL));
>-    stream->AddTrack(audio_track.get());
>-    EXPECT_TRUE(pc_->AddStream(stream));
>+    stream->AddTrack(CreateAudioTrack(label + "a0"));
>+    ASSERT_TRUE(pc_->AddStream(stream));
>     EXPECT_TRUE_WAIT(observer_.renegotiation_needed_, kTimeout);
>     observer_.renegotiation_needed_ = false;
>   }
> 
>-  void AddAudioVideoStream(const std::string& stream_label,
>+  void AddAudioVideoStream(const std::string& stream_id,
>                            const std::string& audio_track_label,
>                            const std::string& video_track_label) {
>     // Create a local stream.
>     rtc::scoped_refptr<MediaStreamInterface> stream(
>-        pc_factory_->CreateLocalMediaStream(stream_label));
>-    rtc::scoped_refptr<AudioTrackInterface> audio_track(
>-        pc_factory_->CreateAudioTrack(
>-            audio_track_label, static_cast<AudioSourceInterface*>(NULL)));
>-    stream->AddTrack(audio_track.get());
>-    rtc::scoped_refptr<VideoTrackInterface> video_track(
>-        pc_factory_->CreateVideoTrack(
>-            video_track_label, pc_factory_->CreateVideoSource(
>-                                   std::unique_ptr<cricket::VideoCapturer>(
>-                                       new cricket::FakeVideoCapturer()))));
>-    stream->AddTrack(video_track.get());
>-    EXPECT_TRUE(pc_->AddStream(stream));
>+        pc_factory_->CreateLocalMediaStream(stream_id));
>+    stream->AddTrack(CreateAudioTrack(audio_track_label));
>+    stream->AddTrack(CreateVideoTrack(video_track_label));
>+    ASSERT_TRUE(pc_->AddStream(stream));
>     EXPECT_TRUE_WAIT(observer_.renegotiation_needed_, kTimeout);
>     observer_.renegotiation_needed_ = false;
>   }
> 
>+  rtc::scoped_refptr<RtpReceiverInterface> GetFirstReceiverOfType(
>+      cricket::MediaType media_type) {
>+    for (auto receiver : pc_->GetReceivers()) {
>+      if (receiver->media_type() == media_type) {
>+        return receiver;
>+      }
>+    }
>+    return nullptr;
>+  }
>+
>   bool DoCreateOfferAnswer(std::unique_ptr<SessionDescriptionInterface>* desc,
>                            bool offer,
>                            MediaConstraintsInterface* constraints) {
>-    rtc::scoped_refptr<MockCreateSessionDescriptionObserver>
>-        observer(new rtc::RefCountedObject<
>-            MockCreateSessionDescriptionObserver>());
>+    rtc::scoped_refptr<MockCreateSessionDescriptionObserver> observer(
>+        new rtc::RefCountedObject<MockCreateSessionDescriptionObserver>());
>     if (offer) {
>       pc_->CreateOffer(observer, constraints);
>     } else {
>@@ -809,9 +922,8 @@ class PeerConnectionInterfaceTest : public testing::Test {
>   bool DoSetSessionDescription(
>       std::unique_ptr<SessionDescriptionInterface> desc,
>       bool local) {
>-    rtc::scoped_refptr<MockSetSessionDescriptionObserver>
>-        observer(new rtc::RefCountedObject<
>-            MockSetSessionDescriptionObserver>());
>+    rtc::scoped_refptr<MockSetSessionDescriptionObserver> observer(
>+        new rtc::RefCountedObject<MockSetSessionDescriptionObserver>());
>     if (local) {
>       pc_->SetLocalDescription(observer, desc.release());
>     } else {
>@@ -839,8 +951,8 @@ class PeerConnectionInterfaceTest : public testing::Test {
>   bool DoGetStats(MediaStreamTrackInterface* track) {
>     rtc::scoped_refptr<MockStatsObserver> observer(
>         new rtc::RefCountedObject<MockStatsObserver>());
>-    if (!pc_->GetStats(
>-        observer, track, PeerConnectionInterface::kStatsOutputLevelStandard))
>+    if (!pc_->GetStats(observer, track,
>+                       PeerConnectionInterface::kStatsOutputLevelStandard))
>       return false;
>     EXPECT_TRUE_WAIT(observer->called(), kTimeout);
>     return observer->called();
>@@ -858,7 +970,14 @@ class PeerConnectionInterfaceTest : public testing::Test {
>   void InitiateCall() {
>     CreatePeerConnectionWithoutDtls();
>     // Create a local stream with audio&video tracks.
>-    AddAudioVideoStream(kStreamLabel1, "audio_label", "video_label");
>+    if (sdp_semantics_ == SdpSemantics::kPlanB) {
>+      AddAudioVideoStream(kStreamId1, "audio_track", "video_track");
>+    } else {
>+      // Unified Plan does not support AddStream, so just add an audio and video
>+      // track.
>+      AddAudioTrack(kAudioTracks[0], {kStreamId1});
>+      AddVideoTrack(kVideoTracks[0], {kStreamId1});
>+    }
>     CreateOfferReceiveAnswer();
>   }
> 
>@@ -974,13 +1093,15 @@ class PeerConnectionInterfaceTest : public testing::Test {
>     EXPECT_EQ(PeerConnectionInterface::kStable, observer_.state_);
>   }
> 
>-  // Help function used for waiting until a the last signaled remote stream has
>-  // the same label as |stream_label|. In a few of the tests in this file we
>-  // answer with the same session description as we offer and thus we can
>-  // check if OnAddStream have been called with the same stream as we offer to
>-  // send.
>-  void WaitAndVerifyOnAddStream(const std::string& stream_label) {
>-    EXPECT_EQ_WAIT(stream_label, observer_.GetLastAddedStreamLabel(), kTimeout);
>+  // Waits until a remote stream with the given id is signaled. This helper
>+  // function will verify both OnAddTrack and OnAddStream (Plan B only) are
>+  // called with the given stream id and expected number of tracks.
>+  void WaitAndVerifyOnAddStream(const std::string& stream_id,
>+                                int expected_num_tracks) {
>+    // Verify that both OnAddStream and OnAddTrack are called.
>+    EXPECT_EQ_WAIT(stream_id, observer_.GetLastAddedStreamId(), kTimeout);
>+    EXPECT_EQ_WAIT(expected_num_tracks,
>+                   observer_.CountAddTrackEventsForStream(stream_id), kTimeout);
>   }
> 
>   // Creates an offer and applies it as a local session description.
>@@ -1008,10 +1129,10 @@ class PeerConnectionInterfaceTest : public testing::Test {
>     reference_collection_ = StreamCollection::Create();
>     std::string sdp_ms1 = std::string(kSdpStringInit);
> 
>-    std::string mediastream_label = kStreams[0];
>+    std::string mediastream_id = kStreams[0];
> 
>     rtc::scoped_refptr<webrtc::MediaStreamInterface> stream(
>-        webrtc::MediaStream::Create(mediastream_label));
>+        webrtc::MediaStream::Create(mediastream_id));
>     reference_collection_->AddStream(stream);
> 
>     if (number_of_audio_tracks > 0) {
>@@ -1054,14 +1175,29 @@ class PeerConnectionInterfaceTest : public testing::Test {
>     ASSERT_TRUE(stream->AddTrack(video_track));
>   }
> 
>+  std::unique_ptr<SessionDescriptionInterface> CreateOfferWithOneAudioTrack() {
>+    CreatePeerConnectionWithoutDtls();
>+    AddAudioTrack(kAudioTracks[0]);
>+    std::unique_ptr<SessionDescriptionInterface> offer;
>+    EXPECT_TRUE(DoCreateOffer(&offer, nullptr));
>+    return offer;
>+  }
>+
>   std::unique_ptr<SessionDescriptionInterface> CreateOfferWithOneAudioStream() {
>     CreatePeerConnectionWithoutDtls();
>-    AddVoiceStream(kStreamLabel1);
>+    AddAudioStream(kStreamId1);
>     std::unique_ptr<SessionDescriptionInterface> offer;
>     EXPECT_TRUE(DoCreateOffer(&offer, nullptr));
>     return offer;
>   }
> 
>+  std::unique_ptr<SessionDescriptionInterface> CreateAnswerWithOneAudioTrack() {
>+    EXPECT_TRUE(DoSetRemoteDescription(CreateOfferWithOneAudioTrack()));
>+    std::unique_ptr<SessionDescriptionInterface> answer;
>+    EXPECT_TRUE(DoCreateAnswer(&answer, nullptr));
>+    return answer;
>+  }
>+
>   std::unique_ptr<SessionDescriptionInterface>
>   CreateAnswerWithOneAudioStream() {
>     EXPECT_TRUE(DoSetRemoteDescription(CreateOfferWithOneAudioStream()));
>@@ -1126,6 +1262,22 @@ class PeerConnectionInterfaceTest : public testing::Test {
>     return false;
>   }
> 
>+  const char* GetSdpStringWithStream1() const {
>+    if (sdp_semantics_ == SdpSemantics::kPlanB) {
>+      return kSdpStringWithStream1PlanB;
>+    } else {
>+      return kSdpStringWithStream1UnifiedPlan;
>+    }
>+  }
>+
>+  const char* GetSdpStringWithStream1And2() const {
>+    if (sdp_semantics_ == SdpSemantics::kPlanB) {
>+      return kSdpStringWithStream1And2PlanB;
>+    } else {
>+      return kSdpStringWithStream1And2UnifiedPlan;
>+    }
>+  }
>+
>   std::unique_ptr<rtc::VirtualSocketServer> vss_;
>   rtc::AutoSocketServerThread main_;
>   rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_;
>@@ -1136,60 +1288,50 @@ class PeerConnectionInterfaceTest : public testing::Test {
>   rtc::scoped_refptr<PeerConnectionInterface> pc_;
>   MockPeerConnectionObserver observer_;
>   rtc::scoped_refptr<StreamCollection> reference_collection_;
>+  const SdpSemantics sdp_semantics_;
> };
> 
>-// Test that no callbacks on the PeerConnectionObserver are called after the
>-// PeerConnection is closed.
>-TEST_F(PeerConnectionInterfaceTest, CloseAndTestCallbackFunctions) {
>-  rtc::scoped_refptr<PeerConnectionInterface> pc(
>-      pc_factory_for_test_->CreatePeerConnection(
>-          PeerConnectionInterface::RTCConfiguration(), nullptr, nullptr,
>-          nullptr, &observer_));
>-  observer_.SetPeerConnectionInterface(pc.get());
>-  pc->Close();
>-
>-  // No callbacks is expected to be called.
>-  observer_.callback_triggered_ = false;
>-  std::vector<cricket::Candidate> candidates;
>-  pc_factory_for_test_->transport_controller->SignalGatheringState(
>-      cricket::IceGatheringState{});
>-  pc_factory_for_test_->transport_controller->SignalCandidatesGathered(
>-      "", candidates);
>-  pc_factory_for_test_->transport_controller->SignalConnectionState(
>-      cricket::IceConnectionState{});
>-  pc_factory_for_test_->transport_controller->SignalCandidatesRemoved(
>-      candidates);
>-  pc_factory_for_test_->transport_controller->SignalReceiving(false);
>-  EXPECT_FALSE(observer_.callback_triggered_);
>-}
>+class PeerConnectionInterfaceTest
>+    : public PeerConnectionInterfaceBaseTest,
>+      public ::testing::WithParamInterface<SdpSemantics> {
>+ protected:
>+  PeerConnectionInterfaceTest() : PeerConnectionInterfaceBaseTest(GetParam()) {}
>+};
>+
>+class PeerConnectionInterfaceTestPlanB
>+    : public PeerConnectionInterfaceBaseTest {
>+ protected:
>+  PeerConnectionInterfaceTestPlanB()
>+      : PeerConnectionInterfaceBaseTest(SdpSemantics::kPlanB) {}
>+};
> 
> // Generate different CNAMEs when PeerConnections are created.
> // The CNAMEs are expected to be generated randomly. It is possible
> // that the test fails, though the possibility is very low.
>-TEST_F(PeerConnectionInterfaceTest, CnameGenerationInOffer) {
>+TEST_P(PeerConnectionInterfaceTest, CnameGenerationInOffer) {
>   std::unique_ptr<SessionDescriptionInterface> offer1 =
>-      CreateOfferWithOneAudioStream();
>+      CreateOfferWithOneAudioTrack();
>   std::unique_ptr<SessionDescriptionInterface> offer2 =
>-      CreateOfferWithOneAudioStream();
>+      CreateOfferWithOneAudioTrack();
>   EXPECT_NE(GetFirstAudioStreamCname(offer1.get()),
>             GetFirstAudioStreamCname(offer2.get()));
> }
> 
>-TEST_F(PeerConnectionInterfaceTest, CnameGenerationInAnswer) {
>+TEST_P(PeerConnectionInterfaceTest, CnameGenerationInAnswer) {
>   std::unique_ptr<SessionDescriptionInterface> answer1 =
>-      CreateAnswerWithOneAudioStream();
>+      CreateAnswerWithOneAudioTrack();
>   std::unique_ptr<SessionDescriptionInterface> answer2 =
>-      CreateAnswerWithOneAudioStream();
>+      CreateAnswerWithOneAudioTrack();
>   EXPECT_NE(GetFirstAudioStreamCname(answer1.get()),
>             GetFirstAudioStreamCname(answer2.get()));
> }
> 
>-TEST_F(PeerConnectionInterfaceTest,
>+TEST_P(PeerConnectionInterfaceTest,
>        CreatePeerConnectionWithDifferentConfigurations) {
>   CreatePeerConnectionWithDifferentConfigurations();
> }
> 
>-TEST_F(PeerConnectionInterfaceTest,
>+TEST_P(PeerConnectionInterfaceTest,
>        CreatePeerConnectionWithDifferentIceTransportsTypes) {
>   CreatePeerConnectionWithIceTransportsType(PeerConnectionInterface::kNone);
>   EXPECT_EQ(cricket::CF_NONE, port_allocator_->candidate_filter());
>@@ -1205,7 +1347,7 @@ TEST_F(PeerConnectionInterfaceTest,
> // Test that when a PeerConnection is created with a nonzero candidate pool
> // size, the pooled PortAllocatorSession is created with all the attributes
> // in the RTCConfiguration.
>-TEST_F(PeerConnectionInterfaceTest, CreatePeerConnectionWithPooledCandidates) {
>+TEST_P(PeerConnectionInterfaceTest, CreatePeerConnectionWithPooledCandidates) {
>   PeerConnectionInterface::RTCConfiguration config;
>   PeerConnectionInterface::IceServer server;
>   server.uri = kStunAddressOnly;
>@@ -1240,7 +1382,7 @@ TEST_F(PeerConnectionInterfaceTest, CreatePeerConnectionWithPooledCandidates) {
> //
> // Note that the candidate filter (RTCConfiguration::type) is already tested
> // above.
>-TEST_F(PeerConnectionInterfaceTest,
>+TEST_P(PeerConnectionInterfaceTest,
>        CreatePeerConnectionAppliesNetworkConfigToPortAllocator) {
>   // Create fake port allocator.
>   std::unique_ptr<cricket::FakePortAllocator> port_allocator(
>@@ -1264,7 +1406,10 @@ TEST_F(PeerConnectionInterfaceTest,
>           rtc::Thread::Current(), rtc::Thread::Current(),
>           rtc::Thread::Current(), fake_audio_capture_module_,
>           webrtc::CreateBuiltinAudioEncoderFactory(),
>-          webrtc::CreateBuiltinAudioDecoderFactory(), nullptr, nullptr));
>+          webrtc::CreateBuiltinAudioDecoderFactory(),
>+          webrtc::CreateBuiltinVideoEncoderFactory(),
>+          webrtc::CreateBuiltinVideoDecoderFactory(), nullptr /* audio_mixer */,
>+          nullptr /* audio_processing */));
>   rtc::scoped_refptr<PeerConnectionInterface> pc(
>       pc_factory->CreatePeerConnection(
>           config, nullptr, std::move(port_allocator), nullptr, &observer_));
>@@ -1280,34 +1425,9 @@ TEST_F(PeerConnectionInterfaceTest,
>   EXPECT_TRUE(raw_port_allocator->prune_turn_ports());
> }
> 
>-// Test that the PeerConnection initializes the port allocator passed into it,
>-// and on the correct thread.
>-TEST_F(PeerConnectionInterfaceTest,
>-       CreatePeerConnectionInitializesPortAllocatorOnNetworkThread) {
>-  std::unique_ptr<rtc::Thread> network_thread(
>-      rtc::Thread::CreateWithSocketServer());
>-  network_thread->Start();
>-  rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> pc_factory(
>-      webrtc::CreatePeerConnectionFactory(
>-          network_thread.get(), rtc::Thread::Current(), rtc::Thread::Current(),
>-          fake_audio_capture_module_,
>-          webrtc::CreateBuiltinAudioEncoderFactory(),
>-          webrtc::CreateBuiltinAudioDecoderFactory(), nullptr, nullptr));
>-  std::unique_ptr<cricket::FakePortAllocator> port_allocator(
>-      new cricket::FakePortAllocator(network_thread.get(), nullptr));
>-  cricket::FakePortAllocator* raw_port_allocator = port_allocator.get();
>-  PeerConnectionInterface::RTCConfiguration config;
>-  rtc::scoped_refptr<PeerConnectionInterface> pc(
>-      pc_factory->CreatePeerConnection(
>-          config, nullptr, std::move(port_allocator), nullptr, &observer_));
>-  // FakePortAllocator RTC_CHECKs that it's initialized on the right thread,
>-  // so all we have to do here is check that it's initialized.
>-  EXPECT_TRUE(raw_port_allocator->initialized());
>-}
>-
> // Check that GetConfiguration returns the configuration the PeerConnection was
> // constructed with, before SetConfiguration is called.
>-TEST_F(PeerConnectionInterfaceTest, GetConfigurationAfterCreatePeerConnection) {
>+TEST_P(PeerConnectionInterfaceTest, GetConfigurationAfterCreatePeerConnection) {
>   PeerConnectionInterface::RTCConfiguration config;
>   config.type = PeerConnectionInterface::kRelay;
>   CreatePeerConnection(config, nullptr);
>@@ -1319,10 +1439,10 @@ TEST_F(PeerConnectionInterfaceTest, GetConfigurationAfterCreatePeerConnection) {
> 
> // Check that GetConfiguration returns the last configuration passed into
> // SetConfiguration.
>-TEST_F(PeerConnectionInterfaceTest, GetConfigurationAfterSetConfiguration) {
>+TEST_P(PeerConnectionInterfaceTest, GetConfigurationAfterSetConfiguration) {
>   CreatePeerConnection();
> 
>-  PeerConnectionInterface::RTCConfiguration config;
>+  PeerConnectionInterface::RTCConfiguration config = pc_->GetConfiguration();
>   config.type = PeerConnectionInterface::kRelay;
>   EXPECT_TRUE(pc_->SetConfiguration(config));
> 
>@@ -1331,17 +1451,26 @@ TEST_F(PeerConnectionInterfaceTest, GetConfigurationAfterSetConfiguration) {
>   EXPECT_EQ(PeerConnectionInterface::kRelay, returned_config.type);
> }
> 
>-TEST_F(PeerConnectionInterfaceTest, AddStreams) {
>+TEST_P(PeerConnectionInterfaceTest, SetConfigurationFailsAfterClose) {
>+  CreatePeerConnection();
>+
>+  pc_->Close();
>+
>+  EXPECT_FALSE(
>+      pc_->SetConfiguration(PeerConnectionInterface::RTCConfiguration()));
>+}
>+
>+TEST_F(PeerConnectionInterfaceTestPlanB, AddStreams) {
>   CreatePeerConnectionWithoutDtls();
>-  AddVideoStream(kStreamLabel1);
>-  AddVoiceStream(kStreamLabel2);
>+  AddVideoStream(kStreamId1);
>+  AddAudioStream(kStreamId2);
>   ASSERT_EQ(2u, pc_->local_streams()->count());
> 
>   // Test we can add multiple local streams to one peerconnection.
>   rtc::scoped_refptr<MediaStreamInterface> stream(
>-      pc_factory_->CreateLocalMediaStream(kStreamLabel3));
>+      pc_factory_->CreateLocalMediaStream(kStreamId3));
>   rtc::scoped_refptr<AudioTrackInterface> audio_track(
>-      pc_factory_->CreateAudioTrack(kStreamLabel3,
>+      pc_factory_->CreateAudioTrack(kStreamId3,
>                                     static_cast<AudioSourceInterface*>(NULL)));
>   stream->AddTrack(audio_track.get());
>   EXPECT_TRUE(pc_->AddStream(stream));
>@@ -1361,43 +1490,39 @@ TEST_F(PeerConnectionInterfaceTest, AddStreams) {
> }
> 
> // Test that the created offer includes streams we added.
>-TEST_F(PeerConnectionInterfaceTest, AddedStreamsPresentInOffer) {
>+// Don't run under Unified Plan since the stream API is not available.
>+TEST_F(PeerConnectionInterfaceTestPlanB, AddedStreamsPresentInOffer) {
>   CreatePeerConnectionWithoutDtls();
>-  AddAudioVideoStream(kStreamLabel1, "audio_track", "video_track");
>+  AddAudioVideoStream(kStreamId1, "audio_track", "video_track");
>   std::unique_ptr<SessionDescriptionInterface> offer;
>   ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
> 
>   const cricket::AudioContentDescription* audio_desc =
>       cricket::GetFirstAudioContentDescription(offer->description());
>-  EXPECT_TRUE(
>-      ContainsTrack(audio_desc->streams(), kStreamLabel1, "audio_track"));
>+  EXPECT_TRUE(ContainsTrack(audio_desc->streams(), kStreamId1, "audio_track"));
> 
>   const cricket::VideoContentDescription* video_desc =
>       cricket::GetFirstVideoContentDescription(offer->description());
>-  EXPECT_TRUE(
>-      ContainsTrack(video_desc->streams(), kStreamLabel1, "video_track"));
>+  EXPECT_TRUE(ContainsTrack(video_desc->streams(), kStreamId1, "video_track"));
> 
>   // Add another stream and ensure the offer includes both the old and new
>   // streams.
>-  AddAudioVideoStream(kStreamLabel2, "audio_track2", "video_track2");
>+  AddAudioVideoStream(kStreamId2, "audio_track2", "video_track2");
>   ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
> 
>   audio_desc = cricket::GetFirstAudioContentDescription(offer->description());
>-  EXPECT_TRUE(
>-      ContainsTrack(audio_desc->streams(), kStreamLabel1, "audio_track"));
>-  EXPECT_TRUE(
>-      ContainsTrack(audio_desc->streams(), kStreamLabel2, "audio_track2"));
>+  EXPECT_TRUE(ContainsTrack(audio_desc->streams(), kStreamId1, "audio_track"));
>+  EXPECT_TRUE(ContainsTrack(audio_desc->streams(), kStreamId2, "audio_track2"));
> 
>   video_desc = cricket::GetFirstVideoContentDescription(offer->description());
>-  EXPECT_TRUE(
>-      ContainsTrack(video_desc->streams(), kStreamLabel1, "video_track"));
>-  EXPECT_TRUE(
>-      ContainsTrack(video_desc->streams(), kStreamLabel2, "video_track2"));
>+  EXPECT_TRUE(ContainsTrack(video_desc->streams(), kStreamId1, "video_track"));
>+  EXPECT_TRUE(ContainsTrack(video_desc->streams(), kStreamId2, "video_track2"));
> }
> 
>-TEST_F(PeerConnectionInterfaceTest, RemoveStream) {
>+// Don't run under Unified Plan since the stream API is not available.
>+TEST_F(PeerConnectionInterfaceTestPlanB, RemoveStream) {
>   CreatePeerConnectionWithoutDtls();
>-  AddVideoStream(kStreamLabel1);
>+  AddVideoStream(kStreamId1);
>   ASSERT_EQ(1u, pc_->local_streams()->count());
>   pc_->RemoveStream(pc_->local_streams()->at(0));
>   EXPECT_EQ(0u, pc_->local_streams()->count());
>@@ -1407,7 +1532,9 @@ TEST_F(PeerConnectionInterfaceTest, RemoveStream) {
> // Tests that the created offer includes tracks we added,
> // and that the RtpSenders are created correctly.
> // Also tests that RemoveTrack removes the tracks from subsequent offers.
>-TEST_F(PeerConnectionInterfaceTest, AddTrackRemoveTrack) {
>+// Only tested with Plan B since Unified Plan is covered in more detail by tests
>+// in peerconnection_jsep_unittests.cc
>+TEST_F(PeerConnectionInterfaceTestPlanB, AddTrackRemoveTrack) {
>   CreatePeerConnectionWithoutDtls();
>   rtc::scoped_refptr<AudioTrackInterface> audio_track(
>       pc_factory_->CreateAudioTrack("audio_track", nullptr));
>@@ -1416,14 +1543,14 @@ TEST_F(PeerConnectionInterfaceTest, AddTrackRemoveTrack) {
>           "video_track", pc_factory_->CreateVideoSource(
>                              std::unique_ptr<cricket::VideoCapturer>(
>                                  new cricket::FakeVideoCapturer()))));
>-  auto audio_sender = pc_->AddTrack(audio_track, {kStreamLabel1}).MoveValue();
>-  auto video_sender = pc_->AddTrack(video_track, {kStreamLabel1}).MoveValue();
>+  auto audio_sender = pc_->AddTrack(audio_track, {kStreamId1}).MoveValue();
>+  auto video_sender = pc_->AddTrack(video_track, {kStreamId1}).MoveValue();
>   EXPECT_EQ(1UL, audio_sender->stream_ids().size());
>-  EXPECT_EQ(kStreamLabel1, audio_sender->stream_ids()[0]);
>+  EXPECT_EQ(kStreamId1, audio_sender->stream_ids()[0]);
>   EXPECT_EQ("audio_track", audio_sender->id());
>   EXPECT_EQ(audio_track, audio_sender->track());
>   EXPECT_EQ(1UL, video_sender->stream_ids().size());
>-  EXPECT_EQ(kStreamLabel1, video_sender->stream_ids()[0]);
>+  EXPECT_EQ(kStreamId1, video_sender->stream_ids()[0]);
>   EXPECT_EQ("video_track", video_sender->id());
>   EXPECT_EQ(video_track, video_sender->track());
> 
>@@ -1434,12 +1561,12 @@ TEST_F(PeerConnectionInterfaceTest, AddTrackRemoveTrack) {
>   const cricket::ContentInfo* audio_content =
>       cricket::GetFirstAudioContent(offer->description());
>   EXPECT_TRUE(ContainsTrack(audio_content->media_description()->streams(),
>-                            kStreamLabel1, "audio_track"));
>+                            kStreamId1, "audio_track"));
> 
>   const cricket::ContentInfo* video_content =
>       cricket::GetFirstVideoContent(offer->description());
>   EXPECT_TRUE(ContainsTrack(video_content->media_description()->streams(),
>-                            kStreamLabel1, "video_track"));
>+                            kStreamId1, "video_track"));
> 
>   EXPECT_TRUE(DoSetLocalDescription(std::move(offer)));
> 
>@@ -1452,11 +1579,11 @@ TEST_F(PeerConnectionInterfaceTest, AddTrackRemoveTrack) {
> 
>   audio_content = cricket::GetFirstAudioContent(offer->description());
>   EXPECT_FALSE(ContainsTrack(audio_content->media_description()->streams(),
>-                             kStreamLabel1, "audio_track"));
>+                             kStreamId1, "audio_track"));
> 
>   video_content = cricket::GetFirstVideoContent(offer->description());
>   EXPECT_FALSE(ContainsTrack(video_content->media_description()->streams(),
>-                             kStreamLabel1, "video_track"));
>+                             kStreamId1, "video_track"));
> 
>   EXPECT_TRUE(DoSetLocalDescription(std::move(offer)));
> 
>@@ -1468,7 +1595,7 @@ TEST_F(PeerConnectionInterfaceTest, AddTrackRemoveTrack) {
> 
> // Test creating senders without a stream specified,
> // expecting a random stream ID to be generated.
>-TEST_F(PeerConnectionInterfaceTest, AddTrackWithoutStream) {
>+TEST_P(PeerConnectionInterfaceTest, AddTrackWithoutStream) {
>   CreatePeerConnectionWithoutDtls();
>   rtc::scoped_refptr<AudioTrackInterface> audio_track(
>       pc_factory_->CreateAudioTrack("audio_track", nullptr));
>@@ -1485,14 +1612,21 @@ TEST_F(PeerConnectionInterfaceTest, AddTrackWithoutStream) {
>   EXPECT_EQ(audio_track, audio_sender->track());
>   EXPECT_EQ("video_track", video_sender->id());
>   EXPECT_EQ(video_track, video_sender->track());
>-  // If the ID is truly a random GUID, it should be infinitely unlikely they
>-  // will be the same.
>-  EXPECT_NE(video_sender->stream_ids(), audio_sender->stream_ids());
>+  if (sdp_semantics_ == SdpSemantics::kPlanB) {
>+    // If the ID is truly a random GUID, it should be infinitely unlikely they
>+    // will be the same.
>+    EXPECT_NE(video_sender->stream_ids(), audio_sender->stream_ids());
>+  } else {
>+    // We allows creating tracks without stream ids under Unified Plan
>+    // semantics.
>+    EXPECT_EQ(0u, video_sender->stream_ids().size());
>+    EXPECT_EQ(0u, audio_sender->stream_ids().size());
>+  }
> }
> 
> // Test that we can call GetStats() after AddTrack but before connecting
> // the PeerConnection to a peer.
>-TEST_F(PeerConnectionInterfaceTest, AddTrackBeforeConnecting) {
>+TEST_P(PeerConnectionInterfaceTest, AddTrackBeforeConnecting) {
>   CreatePeerConnectionWithoutDtls();
>   rtc::scoped_refptr<AudioTrackInterface> audio_track(
>       pc_factory_->CreateAudioTrack("audio_track", nullptr));
>@@ -1506,7 +1640,7 @@ TEST_F(PeerConnectionInterfaceTest, AddTrackBeforeConnecting) {
>   EXPECT_TRUE(DoGetStats(nullptr));
> }
> 
>-TEST_F(PeerConnectionInterfaceTest, AttachmentIdIsSetOnAddTrack) {
>+TEST_P(PeerConnectionInterfaceTest, AttachmentIdIsSetOnAddTrack) {
>   CreatePeerConnectionWithoutDtls();
>   rtc::scoped_refptr<AudioTrackInterface> audio_track(
>       pc_factory_->CreateAudioTrack("audio_track", nullptr));
>@@ -1516,73 +1650,85 @@ TEST_F(PeerConnectionInterfaceTest, AttachmentIdIsSetOnAddTrack) {
>                              std::unique_ptr<cricket::VideoCapturer>(
>                                  new cricket::FakeVideoCapturer()))));
>   auto audio_sender = pc_->AddTrack(audio_track, std::vector<std::string>());
>+  ASSERT_TRUE(audio_sender.ok());
>+  auto* audio_sender_proxy =
>+      static_cast<RtpSenderProxyWithInternal<RtpSenderInternal>*>(
>+          audio_sender.value().get());
>+  EXPECT_NE(0, audio_sender_proxy->internal()->AttachmentId());
>+
>   auto video_sender = pc_->AddTrack(video_track, std::vector<std::string>());
>-  EXPECT_TRUE(audio_sender.ok());
>-  EXPECT_TRUE(video_sender.ok());
>-  EXPECT_NE(0, video_sender.value()->AttachmentId());
>-  EXPECT_NE(0, audio_sender.value()->AttachmentId());
>+  ASSERT_TRUE(video_sender.ok());
>+  auto* video_sender_proxy =
>+      static_cast<RtpSenderProxyWithInternal<RtpSenderInternal>*>(
>+          video_sender.value().get());
>+  EXPECT_NE(0, video_sender_proxy->internal()->AttachmentId());
> }
> 
>-TEST_F(PeerConnectionInterfaceTest, AttachmentIdIsSetOnAddStream) {
>+// Don't run under Unified Plan since the stream API is not available.
>+TEST_F(PeerConnectionInterfaceTestPlanB, AttachmentIdIsSetOnAddStream) {
>   CreatePeerConnectionWithoutDtls();
>-  AddVideoStream(kStreamLabel1);
>+  AddVideoStream(kStreamId1);
>   auto senders = pc_->GetSenders();
>-  EXPECT_EQ(1u, senders.size());
>-  EXPECT_NE(0, senders[0]->AttachmentId());
>+  ASSERT_EQ(1u, senders.size());
>+  auto* sender_proxy =
>+      static_cast<RtpSenderProxyWithInternal<RtpSenderInternal>*>(
>+          senders[0].get());
>+  EXPECT_NE(0, sender_proxy->internal()->AttachmentId());
> }
> 
>-TEST_F(PeerConnectionInterfaceTest, CreateOfferReceiveAnswer) {
>+TEST_P(PeerConnectionInterfaceTest, CreateOfferReceiveAnswer) {
>   InitiateCall();
>-  WaitAndVerifyOnAddStream(kStreamLabel1);
>+  WaitAndVerifyOnAddStream(kStreamId1, 2);
>   VerifyRemoteRtpHeaderExtensions();
> }
> 
>-TEST_F(PeerConnectionInterfaceTest, CreateOfferReceivePrAnswerAndAnswer) {
>+TEST_P(PeerConnectionInterfaceTest, CreateOfferReceivePrAnswerAndAnswer) {
>   CreatePeerConnectionWithoutDtls();
>-  AddVideoStream(kStreamLabel1);
>+  AddVideoTrack(kVideoTracks[0], {kStreamId1});
>   CreateOfferAsLocalDescription();
>   std::string offer;
>   EXPECT_TRUE(pc_->local_description()->ToString(&offer));
>   CreatePrAnswerAndAnswerAsRemoteDescription(offer);
>-  WaitAndVerifyOnAddStream(kStreamLabel1);
>+  WaitAndVerifyOnAddStream(kStreamId1, 1);
> }
> 
>-TEST_F(PeerConnectionInterfaceTest, ReceiveOfferCreateAnswer) {
>+TEST_P(PeerConnectionInterfaceTest, ReceiveOfferCreateAnswer) {
>   CreatePeerConnectionWithoutDtls();
>-  AddVideoStream(kStreamLabel1);
>+  AddVideoTrack(kVideoTracks[0], {kStreamId1});
> 
>   CreateOfferAsRemoteDescription();
>   CreateAnswerAsLocalDescription();
> 
>-  WaitAndVerifyOnAddStream(kStreamLabel1);
>+  WaitAndVerifyOnAddStream(kStreamId1, 1);
> }
> 
>-TEST_F(PeerConnectionInterfaceTest, ReceiveOfferCreatePrAnswerAndAnswer) {
>+TEST_P(PeerConnectionInterfaceTest, ReceiveOfferCreatePrAnswerAndAnswer) {
>   CreatePeerConnectionWithoutDtls();
>-  AddVideoStream(kStreamLabel1);
>+  AddVideoTrack(kVideoTracks[0], {kStreamId1});
> 
>   CreateOfferAsRemoteDescription();
>   CreatePrAnswerAsLocalDescription();
>   CreateAnswerAsLocalDescription();
> 
>-  WaitAndVerifyOnAddStream(kStreamLabel1);
>+  WaitAndVerifyOnAddStream(kStreamId1, 1);
> }
> 
>-TEST_F(PeerConnectionInterfaceTest, Renegotiate) {
>+// Don't run under Unified Plan since the stream API is not available.
>+TEST_F(PeerConnectionInterfaceTestPlanB, Renegotiate) {
>   InitiateCall();
>   ASSERT_EQ(1u, pc_->remote_streams()->count());
>   pc_->RemoveStream(pc_->local_streams()->at(0));
>   CreateOfferReceiveAnswer();
>   EXPECT_EQ(0u, pc_->remote_streams()->count());
>-  AddVideoStream(kStreamLabel1);
>+  AddVideoStream(kStreamId1);
>   CreateOfferReceiveAnswer();
> }
> 
> // Tests that after negotiating an audio only call, the respondent can perform a
> // renegotiation that removes the audio stream.
>-TEST_F(PeerConnectionInterfaceTest, RenegotiateAudioOnly) {
>+TEST_F(PeerConnectionInterfaceTestPlanB, RenegotiateAudioOnly) {
>   CreatePeerConnectionWithoutDtls();
>-  AddVoiceStream(kStreamLabel1);
>+  AddAudioStream(kStreamId1);
>   CreateOfferAsRemoteDescription();
>   CreateAnswerAsLocalDescription();
> 
>@@ -1593,13 +1739,13 @@ TEST_F(PeerConnectionInterfaceTest, RenegotiateAudioOnly) {
> }
> 
> // Test that candidates are generated and that we can parse our own candidates.
>-TEST_F(PeerConnectionInterfaceTest, IceCandidates) {
>+TEST_P(PeerConnectionInterfaceTest, IceCandidates) {
>   CreatePeerConnectionWithoutDtls();
> 
>   EXPECT_FALSE(pc_->AddIceCandidate(observer_.last_candidate()));
>   // SetRemoteDescription takes ownership of offer.
>   std::unique_ptr<SessionDescriptionInterface> offer;
>-  AddVideoStream(kStreamLabel1);
>+  AddVideoTrack(kVideoTracks[0]);
>   EXPECT_TRUE(DoCreateOffer(&offer, nullptr));
>   EXPECT_TRUE(DoSetRemoteDescription(std::move(offer)));
> 
>@@ -1616,7 +1762,7 @@ TEST_F(PeerConnectionInterfaceTest, IceCandidates) {
> 
> // Test that CreateOffer and CreateAnswer will fail if the track labels are
> // not unique.
>-TEST_F(PeerConnectionInterfaceTest, CreateOfferAnswerWithInvalidStream) {
>+TEST_F(PeerConnectionInterfaceTestPlanB, CreateOfferAnswerWithInvalidStream) {
>   CreatePeerConnectionWithoutDtls();
>   // Create a regular offer for the CreateAnswer test later.
>   std::unique_ptr<SessionDescriptionInterface> offer;
>@@ -1625,7 +1771,8 @@ TEST_F(PeerConnectionInterfaceTest, CreateOfferAnswerWithInvalidStream) {
>   offer.reset();
> 
>   // Create a local stream with audio&video tracks having same label.
>-  AddAudioVideoStream(kStreamLabel1, "track_label", "track_label");
>+  AddAudioTrack("track_label", {kStreamId1});
>+  AddVideoTrack("track_label", {kStreamId1});
> 
>   // Test CreateOffer
>   EXPECT_FALSE(DoCreateOffer(&offer, nullptr));
>@@ -1637,20 +1784,21 @@ TEST_F(PeerConnectionInterfaceTest, CreateOfferAnswerWithInvalidStream) {
> 
> // Test that we will get different SSRCs for each tracks in the offer and answer
> // we created.
>-TEST_F(PeerConnectionInterfaceTest, SsrcInOfferAnswer) {
>+TEST_P(PeerConnectionInterfaceTest, SsrcInOfferAnswer) {
>   CreatePeerConnectionWithoutDtls();
>   // Create a local stream with audio&video tracks having different labels.
>-  AddAudioVideoStream(kStreamLabel1, "audio_label", "video_label");
>+  AddAudioTrack(kAudioTracks[0], {kStreamId1});
>+  AddVideoTrack(kVideoTracks[0], {kStreamId1});
> 
>   // Test CreateOffer
>   std::unique_ptr<SessionDescriptionInterface> offer;
>   ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
>   int audio_ssrc = 0;
>   int video_ssrc = 0;
>-  EXPECT_TRUE(GetFirstSsrc(GetFirstAudioContent(offer->description()),
>-                           &audio_ssrc));
>-  EXPECT_TRUE(GetFirstSsrc(GetFirstVideoContent(offer->description()),
>-                           &video_ssrc));
>+  EXPECT_TRUE(
>+      GetFirstSsrc(GetFirstAudioContent(offer->description()), &audio_ssrc));
>+  EXPECT_TRUE(
>+      GetFirstSsrc(GetFirstVideoContent(offer->description()), &video_ssrc));
>   EXPECT_NE(audio_ssrc, video_ssrc);
> 
>   // Test CreateAnswer
>@@ -1659,20 +1807,21 @@ TEST_F(PeerConnectionInterfaceTest, SsrcInOfferAnswer) {
>   ASSERT_TRUE(DoCreateAnswer(&answer, nullptr));
>   audio_ssrc = 0;
>   video_ssrc = 0;
>-  EXPECT_TRUE(GetFirstSsrc(GetFirstAudioContent(answer->description()),
>-                           &audio_ssrc));
>-  EXPECT_TRUE(GetFirstSsrc(GetFirstVideoContent(answer->description()),
>-                           &video_ssrc));
>+  EXPECT_TRUE(
>+      GetFirstSsrc(GetFirstAudioContent(answer->description()), &audio_ssrc));
>+  EXPECT_TRUE(
>+      GetFirstSsrc(GetFirstVideoContent(answer->description()), &video_ssrc));
>   EXPECT_NE(audio_ssrc, video_ssrc);
> }
> 
> // Test that it's possible to call AddTrack on a MediaStream after adding
> // the stream to a PeerConnection.
> // TODO(deadbeef): Remove this test once this behavior is no longer supported.
>-TEST_F(PeerConnectionInterfaceTest, AddTrackAfterAddStream) {
>+// Don't run under Unified Plan since the stream API is not available.
>+TEST_F(PeerConnectionInterfaceTestPlanB, AddTrackAfterAddStream) {
>   CreatePeerConnectionWithoutDtls();
>   // Create audio stream and add to PeerConnection.
>-  AddVoiceStream(kStreamLabel1);
>+  AddAudioStream(kStreamId1);
>   MediaStreamInterface* stream = pc_->local_streams()->at(0);
> 
>   // Add video track to the audio-only stream.
>@@ -1694,10 +1843,11 @@ TEST_F(PeerConnectionInterfaceTest, AddTrackAfterAddStream) {
> // Test that it's possible to call RemoveTrack on a MediaStream after adding
> // the stream to a PeerConnection.
> // TODO(deadbeef): Remove this test once this behavior is no longer supported.
>-TEST_F(PeerConnectionInterfaceTest, RemoveTrackAfterAddStream) {
>+// Don't run under Unified Plan since the stream API is not available.
>+TEST_F(PeerConnectionInterfaceTestPlanB, RemoveTrackAfterAddStream) {
>   CreatePeerConnectionWithoutDtls();
>   // Create audio/video stream and add to PeerConnection.
>-  AddAudioVideoStream(kStreamLabel1, "audio_label", "video_label");
>+  AddAudioVideoStream(kStreamId1, "audio_label", "video_label");
>   MediaStreamInterface* stream = pc_->local_streams()->at(0);
> 
>   // Remove the video track.
>@@ -1711,23 +1861,12 @@ TEST_F(PeerConnectionInterfaceTest, RemoveTrackAfterAddStream) {
>   EXPECT_TRUE(video_desc == nullptr);
> }
> 
>-// Verify that CreateDtmfSender only succeeds if called with a valid local
>-// track. Other aspects of DtmfSenders are tested in
>-// peerconnection_integrationtest.cc.
>-TEST_F(PeerConnectionInterfaceTest, CreateDtmfSenderWithInvalidParams) {
>-  CreatePeerConnection();
>-  AddAudioVideoStream(kStreamLabel1, "audio_label", "video_label");
>-  EXPECT_EQ(nullptr, pc_->CreateDtmfSender(nullptr));
>-  rtc::scoped_refptr<webrtc::AudioTrackInterface> non_localtrack(
>-      pc_factory_->CreateAudioTrack("dummy_track", nullptr));
>-  EXPECT_EQ(nullptr, pc_->CreateDtmfSender(non_localtrack));
>-}
>-
> // Test creating a sender with a stream ID, and ensure the ID is populated
> // in the offer.
>-TEST_F(PeerConnectionInterfaceTest, CreateSenderWithStream) {
>+// Don't run under Unified Plan since the stream API is not available.
>+TEST_F(PeerConnectionInterfaceTestPlanB, CreateSenderWithStream) {
>   CreatePeerConnectionWithoutDtls();
>-  pc_->CreateSender("video", kStreamLabel1);
>+  pc_->CreateSender("video", kStreamId1);
> 
>   std::unique_ptr<SessionDescriptionInterface> offer;
>   ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
>@@ -1736,56 +1875,53 @@ TEST_F(PeerConnectionInterfaceTest, CreateSenderWithStream) {
>       cricket::GetFirstVideoContentDescription(offer->description());
>   ASSERT_TRUE(video_desc != nullptr);
>   ASSERT_EQ(1u, video_desc->streams().size());
>-  EXPECT_EQ(kStreamLabel1, video_desc->streams()[0].sync_label);
>+  EXPECT_EQ(kStreamId1, video_desc->streams()[0].first_stream_id());
> }
> 
> // Test that we can specify a certain track that we want statistics about.
>-TEST_F(PeerConnectionInterfaceTest, GetStatsForSpecificTrack) {
>+TEST_P(PeerConnectionInterfaceTest, GetStatsForSpecificTrack) {
>   InitiateCall();
>-  ASSERT_LT(0u, pc_->remote_streams()->count());
>-  ASSERT_LT(0u, pc_->remote_streams()->at(0)->GetAudioTracks().size());
>+  ASSERT_LT(0u, pc_->GetSenders().size());
>+  ASSERT_LT(0u, pc_->GetReceivers().size());
>   rtc::scoped_refptr<MediaStreamTrackInterface> remote_audio =
>-      pc_->remote_streams()->at(0)->GetAudioTracks()[0];
>+      pc_->GetReceivers()[0]->track();
>   EXPECT_TRUE(DoGetStats(remote_audio));
> 
>   // Remove the stream. Since we are sending to our selves the local
>   // and the remote stream is the same.
>-  pc_->RemoveStream(pc_->local_streams()->at(0));
>+  pc_->RemoveTrack(pc_->GetSenders()[0]);
>   // Do a re-negotiation.
>   CreateOfferReceiveAnswer();
> 
>-  ASSERT_EQ(0u, pc_->remote_streams()->count());
>-
>   // Test that we still can get statistics for the old track. Even if it is not
>   // sent any longer.
>   EXPECT_TRUE(DoGetStats(remote_audio));
> }
> 
> // Test that we can get stats on a video track.
>-TEST_F(PeerConnectionInterfaceTest, GetStatsForVideoTrack) {
>+TEST_P(PeerConnectionInterfaceTest, GetStatsForVideoTrack) {
>   InitiateCall();
>-  ASSERT_LT(0u, pc_->remote_streams()->count());
>-  ASSERT_LT(0u, pc_->remote_streams()->at(0)->GetVideoTracks().size());
>-  rtc::scoped_refptr<MediaStreamTrackInterface> remote_video =
>-      pc_->remote_streams()->at(0)->GetVideoTracks()[0];
>-  EXPECT_TRUE(DoGetStats(remote_video));
>+  auto video_receiver = GetFirstReceiverOfType(cricket::MEDIA_TYPE_VIDEO);
>+  ASSERT_TRUE(video_receiver);
>+  EXPECT_TRUE(DoGetStats(video_receiver->track()));
> }
> 
> // Test that we don't get statistics for an invalid track.
>-TEST_F(PeerConnectionInterfaceTest, GetStatsForInvalidTrack) {
>+TEST_P(PeerConnectionInterfaceTest, GetStatsForInvalidTrack) {
>   InitiateCall();
>   rtc::scoped_refptr<AudioTrackInterface> unknown_audio_track(
>       pc_factory_->CreateAudioTrack("unknown track", NULL));
>   EXPECT_FALSE(DoGetStats(unknown_audio_track));
> }
> 
>-TEST_F(PeerConnectionInterfaceTest, GetRTCStatsBeforeAndAfterCalling) {
>+TEST_P(PeerConnectionInterfaceTest, GetRTCStatsBeforeAndAfterCalling) {
>   CreatePeerConnectionWithoutDtls();
>   EXPECT_TRUE(DoGetRTCStats());
>   // Clearing stats cache is needed now, but should be temporary.
>   // https://bugs.chromium.org/p/webrtc/issues/detail?id=8693
>   pc_->ClearStatsCache();
>-  AddAudioVideoStream(kStreamLabel1, "audio_track", "video_track");
>+  AddAudioTrack(kAudioTracks[0], {kStreamId1});
>+  AddVideoTrack(kVideoTracks[0], {kStreamId1});
>   EXPECT_TRUE(DoGetRTCStats());
>   pc_->ClearStatsCache();
>   CreateOfferReceiveAnswer();
>@@ -1793,7 +1929,7 @@ TEST_F(PeerConnectionInterfaceTest, GetRTCStatsBeforeAndAfterCalling) {
> }
> 
> // This test setup two RTP data channels in loop back.
>-TEST_F(PeerConnectionInterfaceTest, TestDataChannel) {
>+TEST_P(PeerConnectionInterfaceTest, TestDataChannel) {
>   FakeConstraints constraints;
>   constraints.SetAllowRtpDataChannels();
>   CreatePeerConnection(&constraints);
>@@ -1840,7 +1976,7 @@ TEST_F(PeerConnectionInterfaceTest, TestDataChannel) {
> 
> // This test verifies that sendnig binary data over RTP data channels should
> // fail.
>-TEST_F(PeerConnectionInterfaceTest, TestSendBinaryOnRtpDataChannel) {
>+TEST_P(PeerConnectionInterfaceTest, TestSendBinaryOnRtpDataChannel) {
>   FakeConstraints constraints;
>   constraints.SetAllowRtpDataChannels();
>   CreatePeerConnection(&constraints);
>@@ -1870,7 +2006,7 @@ TEST_F(PeerConnectionInterfaceTest, TestSendBinaryOnRtpDataChannel) {
> 
> // This test setup a RTP data channels in loop back and test that a channel is
> // opened even if the remote end answer with a zero SSRC.
>-TEST_F(PeerConnectionInterfaceTest, TestSendOnlyDataChannel) {
>+TEST_P(PeerConnectionInterfaceTest, TestSendOnlyDataChannel) {
>   FakeConstraints constraints;
>   constraints.SetAllowRtpDataChannels();
>   CreatePeerConnection(&constraints);
>@@ -1892,7 +2028,7 @@ TEST_F(PeerConnectionInterfaceTest, TestSendOnlyDataChannel) {
> 
> // This test that if a data channel is added in an answer a receive only channel
> // channel is created.
>-TEST_F(PeerConnectionInterfaceTest, TestReceiveOnlyDataChannel) {
>+TEST_P(PeerConnectionInterfaceTest, TestReceiveOnlyDataChannel) {
>   FakeConstraints constraints;
>   constraints.SetAllowRtpDataChannels();
>   CreatePeerConnection(&constraints);
>@@ -1908,8 +2044,7 @@ TEST_F(PeerConnectionInterfaceTest, TestReceiveOnlyDataChannel) {
>   std::string sdp;
>   EXPECT_TRUE(pc_->local_description()->ToString(&sdp));
>   rtc::replace_substrs(offer_label.c_str(), offer_label.length(),
>-                             receive_label.c_str(), receive_label.length(),
>-                             &sdp);
>+                       receive_label.c_str(), receive_label.length(), &sdp);
>   CreateAnswerAsRemoteDescription(sdp);
> 
>   // Verify that a new incoming data channel has been created and that
>@@ -1933,7 +2068,7 @@ TEST_F(PeerConnectionInterfaceTest, TestReceiveOnlyDataChannel) {
> // This test that no data channel is returned if a reliable channel is
> // requested.
> // TODO(perkj): Remove this test once reliable channels are implemented.
>-TEST_F(PeerConnectionInterfaceTest, CreateReliableRtpDataChannelShouldFail) {
>+TEST_P(PeerConnectionInterfaceTest, CreateReliableRtpDataChannelShouldFail) {
>   FakeConstraints constraints;
>   constraints.SetAllowRtpDataChannels();
>   CreatePeerConnection(&constraints);
>@@ -1947,7 +2082,7 @@ TEST_F(PeerConnectionInterfaceTest, CreateReliableRtpDataChannelShouldFail) {
> }
> 
> // Verifies that duplicated label is not allowed for RTP data channel.
>-TEST_F(PeerConnectionInterfaceTest, RtpDuplicatedLabelNotAllowed) {
>+TEST_P(PeerConnectionInterfaceTest, RtpDuplicatedLabelNotAllowed) {
>   FakeConstraints constraints;
>   constraints.SetAllowRtpDataChannels();
>   CreatePeerConnection(&constraints);
>@@ -1964,7 +2099,7 @@ TEST_F(PeerConnectionInterfaceTest, RtpDuplicatedLabelNotAllowed) {
> 
> // This tests that a SCTP data channel is returned using different
> // DataChannelInit configurations.
>-TEST_F(PeerConnectionInterfaceTest, CreateSctpDataChannel) {
>+TEST_P(PeerConnectionInterfaceTest, CreateSctpDataChannel) {
>   FakeConstraints constraints;
>   constraints.SetAllowDtlsSctpDataChannels();
>   CreatePeerConnection(&constraints);
>@@ -2001,7 +2136,7 @@ TEST_F(PeerConnectionInterfaceTest, CreateSctpDataChannel) {
> 
> // This tests that no data channel is returned if both maxRetransmits and
> // maxRetransmitTime are set for SCTP data channels.
>-TEST_F(PeerConnectionInterfaceTest,
>+TEST_P(PeerConnectionInterfaceTest,
>        CreateSctpDataChannelShouldFailForInvalidConfig) {
>   FakeConstraints constraints;
>   constraints.SetAllowDtlsSctpDataChannels();
>@@ -2019,7 +2154,7 @@ TEST_F(PeerConnectionInterfaceTest,
> 
> // The test verifies that creating a SCTP data channel with an id already in use
> // or out of range should fail.
>-TEST_F(PeerConnectionInterfaceTest,
>+TEST_P(PeerConnectionInterfaceTest,
>        CreateSctpDataChannelWithInvalidIdShouldFail) {
>   FakeConstraints constraints;
>   constraints.SetAllowDtlsSctpDataChannels();
>@@ -2047,7 +2182,7 @@ TEST_F(PeerConnectionInterfaceTest,
> }
> 
> // Verifies that duplicated label is allowed for SCTP data channel.
>-TEST_F(PeerConnectionInterfaceTest, SctpDuplicatedLabelAllowed) {
>+TEST_P(PeerConnectionInterfaceTest, SctpDuplicatedLabelAllowed) {
>   FakeConstraints constraints;
>   constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
>                            true);
>@@ -2065,7 +2200,7 @@ TEST_F(PeerConnectionInterfaceTest, SctpDuplicatedLabelAllowed) {
> 
> // This test verifies that OnRenegotiationNeeded is fired for every new RTP
> // DataChannel.
>-TEST_F(PeerConnectionInterfaceTest, RenegotiationNeededForNewRtpDataChannel) {
>+TEST_P(PeerConnectionInterfaceTest, RenegotiationNeededForNewRtpDataChannel) {
>   FakeConstraints constraints;
>   constraints.SetAllowRtpDataChannels();
>   CreatePeerConnection(&constraints);
>@@ -2081,7 +2216,7 @@ TEST_F(PeerConnectionInterfaceTest, RenegotiationNeededForNewRtpDataChannel) {
> }
> 
> // This test that a data channel closes when a PeerConnection is deleted/closed.
>-TEST_F(PeerConnectionInterfaceTest, DataChannelCloseWhenPeerConnectionClose) {
>+TEST_P(PeerConnectionInterfaceTest, DataChannelCloseWhenPeerConnectionClose) {
>   FakeConstraints constraints;
>   constraints.SetAllowRtpDataChannels();
>   CreatePeerConnection(&constraints);
>@@ -2105,8 +2240,8 @@ TEST_F(PeerConnectionInterfaceTest, DataChannelCloseWhenPeerConnectionClose) {
>   EXPECT_EQ(DataChannelInterface::kClosed, data2->state());
> }
> 
>-// This test that data channels can be rejected in an answer.
>-TEST_F(PeerConnectionInterfaceTest, TestRejectDataChannelInAnswer) {
>+// This tests that RTP data channels can be rejected in an answer.
>+TEST_P(PeerConnectionInterfaceTest, TestRejectRtpDataChannelInAnswer) {
>   FakeConstraints constraints;
>   constraints.SetAllowRtpDataChannels();
>   CreatePeerConnection(&constraints);
>@@ -2123,7 +2258,36 @@ TEST_F(PeerConnectionInterfaceTest, TestRejectDataChannelInAnswer) {
>       webrtc::CreateSessionDescription(SdpType::kAnswer, sdp));
>   ASSERT_TRUE(answer);
>   cricket::ContentInfo* data_info =
>-      answer->description()->GetContentByName("data");
>+      cricket::GetFirstDataContent(answer->description());
>+  data_info->rejected = true;
>+
>+  DoSetRemoteDescription(std::move(answer));
>+  EXPECT_EQ(DataChannelInterface::kClosed, offer_channel->state());
>+}
>+
>+#ifdef HAVE_SCTP
>+// This tests that SCTP data channels can be rejected in an answer.
>+TEST_P(PeerConnectionInterfaceTest, TestRejectSctpDataChannelInAnswer)
>+#else
>+TEST_P(PeerConnectionInterfaceTest, DISABLED_TestRejectSctpDataChannelInAnswer)
>+#endif
>+{
>+  FakeConstraints constraints;
>+  CreatePeerConnection(&constraints);
>+
>+  rtc::scoped_refptr<DataChannelInterface> offer_channel(
>+      pc_->CreateDataChannel("offer_channel", NULL));
>+
>+  CreateOfferAsLocalDescription();
>+
>+  // Create an answer where the m-line for data channels are rejected.
>+  std::string sdp;
>+  EXPECT_TRUE(pc_->local_description()->ToString(&sdp));
>+  std::unique_ptr<SessionDescriptionInterface> answer(
>+      webrtc::CreateSessionDescription(SdpType::kAnswer, sdp));
>+  ASSERT_TRUE(answer);
>+  cricket::ContentInfo* data_info =
>+      cricket::GetFirstDataContent(answer->description());
>   data_info->rejected = true;
> 
>   DoSetRemoteDescription(std::move(answer));
>@@ -2133,12 +2297,13 @@ TEST_F(PeerConnectionInterfaceTest, TestRejectDataChannelInAnswer) {
> // Test that we can create a session description from an SDP string from
> // FireFox, use it as a remote session description, generate an answer and use
> // the answer as a local description.
>-TEST_F(PeerConnectionInterfaceTest, ReceiveFireFoxOffer) {
>+TEST_P(PeerConnectionInterfaceTest, ReceiveFireFoxOffer) {
>   FakeConstraints constraints;
>   constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
>                            true);
>   CreatePeerConnection(&constraints);
>-  AddAudioVideoStream(kStreamLabel1, "audio_label", "video_label");
>+  AddAudioTrack("audio_label");
>+  AddVideoTrack("video_label");
>   std::unique_ptr<SessionDescriptionInterface> desc(
>       webrtc::CreateSessionDescription(SdpType::kOffer,
>                                        webrtc::kFireFoxSdpOffer, nullptr));
>@@ -2160,59 +2325,64 @@ TEST_F(PeerConnectionInterfaceTest, ReceiveFireFoxOffer) {
>   content =
>       cricket::GetFirstDataContent(pc_->local_description()->description());
>   ASSERT_TRUE(content != NULL);
>-  // Expected to fail since it's using an incompatible format.
>-  EXPECT_TRUE(content->rejected);
>+  EXPECT_FALSE(content->rejected);
> #endif
> }
> 
> // Test that fallback from DTLS to SDES is not supported.
> // The fallback was previously supported but was removed to simplify the code
> // and because it's non-standard.
>-TEST_F(PeerConnectionInterfaceTest, DtlsSdesFallbackNotSupported) {
>+TEST_P(PeerConnectionInterfaceTest, DtlsSdesFallbackNotSupported) {
>   FakeConstraints constraints;
>   constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
>                            true);
>   CreatePeerConnection(&constraints);
>   // Wait for fake certificate to be generated. Previously, this is what caused
>   // the "a=crypto" lines to be rejected.
>-  AddAudioVideoStream(kStreamLabel1, "audio_label", "video_label");
>+  AddAudioTrack("audio_label");
>+  AddVideoTrack("video_label");
>   ASSERT_NE(nullptr, fake_certificate_generator_);
>   EXPECT_EQ_WAIT(1, fake_certificate_generator_->generated_certificates(),
>                  kTimeout);
>   std::unique_ptr<SessionDescriptionInterface> desc(
>       webrtc::CreateSessionDescription(SdpType::kOffer, kDtlsSdesFallbackSdp,
>                                        nullptr));
>-  EXPECT_FALSE(DoSetSessionDescription(std::move(desc), false));
>+  EXPECT_FALSE(DoSetSessionDescription(std::move(desc), /*local=*/false));
> }
> 
> // Test that we can create an audio only offer and receive an answer with a
> // limited set of audio codecs and receive an updated offer with more audio
> // codecs, where the added codecs are not supported.
>-TEST_F(PeerConnectionInterfaceTest, ReceiveUpdatedAudioOfferWithBadCodecs) {
>+TEST_P(PeerConnectionInterfaceTest, ReceiveUpdatedAudioOfferWithBadCodecs) {
>   CreatePeerConnectionWithoutDtls();
>-  AddVoiceStream("audio_label");
>+  AddAudioTrack("audio_label");
>   CreateOfferAsLocalDescription();
> 
>+  const char* answer_sdp =
>+      (sdp_semantics_ == SdpSemantics::kPlanB ? webrtc::kAudioSdpPlanB
>+                                              : webrtc::kAudioSdpUnifiedPlan);
>   std::unique_ptr<SessionDescriptionInterface> answer(
>-      webrtc::CreateSessionDescription(SdpType::kAnswer, webrtc::kAudioSdp,
>-                                       nullptr));
>+      webrtc::CreateSessionDescription(SdpType::kAnswer, answer_sdp, nullptr));
>   EXPECT_TRUE(DoSetSessionDescription(std::move(answer), false));
> 
>+  const char* reoffer_sdp =
>+      (sdp_semantics_ == SdpSemantics::kPlanB
>+           ? webrtc::kAudioSdpWithUnsupportedCodecsPlanB
>+           : webrtc::kAudioSdpWithUnsupportedCodecsUnifiedPlan);
>   std::unique_ptr<SessionDescriptionInterface> updated_offer(
>-      webrtc::CreateSessionDescription(
>-          SdpType::kOffer, webrtc::kAudioSdpWithUnsupportedCodecs, nullptr));
>+      webrtc::CreateSessionDescription(SdpType::kOffer, reoffer_sdp, nullptr));
>   EXPECT_TRUE(DoSetSessionDescription(std::move(updated_offer), false));
>   CreateAnswerAsLocalDescription();
> }
> 
> // Test that if we're receiving (but not sending) a track, subsequent offers
> // will have m-lines with a=recvonly.
>-TEST_F(PeerConnectionInterfaceTest, CreateSubsequentRecvOnlyOffer) {
>+TEST_P(PeerConnectionInterfaceTest, CreateSubsequentRecvOnlyOffer) {
>   FakeConstraints constraints;
>   constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
>                            true);
>   CreatePeerConnection(&constraints);
>-  CreateAndSetRemoteOffer(kSdpStringWithStream1);
>+  CreateAndSetRemoteOffer(GetSdpStringWithStream1());
>   CreateAnswerAsLocalDescription();
> 
>   // At this point we should be receiving stream 1, but not sending anything.
>@@ -2234,12 +2404,12 @@ TEST_F(PeerConnectionInterfaceTest, CreateSubsequentRecvOnlyOffer) {
> // Test that if we're receiving (but not sending) a track, and the
> // offerToReceiveVideo/offerToReceiveAudio constraints are explicitly set to
> // false, the generated m-lines will be a=inactive.
>-TEST_F(PeerConnectionInterfaceTest, CreateSubsequentInactiveOffer) {
>+TEST_P(PeerConnectionInterfaceTest, CreateSubsequentInactiveOffer) {
>   FakeConstraints constraints;
>   constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
>                            true);
>   CreatePeerConnection(&constraints);
>-  CreateAndSetRemoteOffer(kSdpStringWithStream1);
>+  CreateAndSetRemoteOffer(GetSdpStringWithStream1());
>   CreateAnswerAsLocalDescription();
> 
>   // At this point we should be receiving stream 1, but not sending anything.
>@@ -2266,10 +2436,10 @@ TEST_F(PeerConnectionInterfaceTest, CreateSubsequentInactiveOffer) {
> 
> // Test that we can use SetConfiguration to change the ICE servers of the
> // PortAllocator.
>-TEST_F(PeerConnectionInterfaceTest, SetConfigurationChangesIceServers) {
>+TEST_P(PeerConnectionInterfaceTest, SetConfigurationChangesIceServers) {
>   CreatePeerConnection();
> 
>-  PeerConnectionInterface::RTCConfiguration config;
>+  PeerConnectionInterface::RTCConfiguration config = pc_->GetConfiguration();
>   PeerConnectionInterface::IceServer server;
>   server.uri = "stun:test_hostname";
>   config.servers.push_back(server);
>@@ -2280,18 +2450,19 @@ TEST_F(PeerConnectionInterfaceTest, SetConfigurationChangesIceServers) {
>             port_allocator_->stun_servers().begin()->hostname());
> }
> 
>-TEST_F(PeerConnectionInterfaceTest, SetConfigurationChangesCandidateFilter) {
>+TEST_P(PeerConnectionInterfaceTest, SetConfigurationChangesCandidateFilter) {
>   CreatePeerConnection();
>-  PeerConnectionInterface::RTCConfiguration config;
>+  PeerConnectionInterface::RTCConfiguration config = pc_->GetConfiguration();
>   config.type = PeerConnectionInterface::kRelay;
>   EXPECT_TRUE(pc_->SetConfiguration(config));
>   EXPECT_EQ(cricket::CF_RELAY, port_allocator_->candidate_filter());
> }
> 
>-TEST_F(PeerConnectionInterfaceTest, SetConfigurationChangesPruneTurnPortsFlag) {
>+TEST_P(PeerConnectionInterfaceTest, SetConfigurationChangesPruneTurnPortsFlag) {
>   PeerConnectionInterface::RTCConfiguration config;
>   config.prune_turn_ports = false;
>   CreatePeerConnection(config, nullptr);
>+  config = pc_->GetConfiguration();
>   EXPECT_FALSE(port_allocator_->prune_turn_ports());
> 
>   config.prune_turn_ports = true;
>@@ -2302,10 +2473,11 @@ TEST_F(PeerConnectionInterfaceTest, SetConfigurationChangesPruneTurnPortsFlag) {
> // Test that the ice check interval can be changed. This does not verify that
> // the setting makes it all the way to P2PTransportChannel, as that would
> // require a very complex set of mocks.
>-TEST_F(PeerConnectionInterfaceTest, SetConfigurationChangesIceCheckInterval) {
>+TEST_P(PeerConnectionInterfaceTest, SetConfigurationChangesIceCheckInterval) {
>   PeerConnectionInterface::RTCConfiguration config;
>-  config.ice_check_min_interval = rtc::nullopt;
>+  config.ice_check_min_interval = absl::nullopt;
>   CreatePeerConnection(config, nullptr);
>+  config = pc_->GetConfiguration();
>   config.ice_check_min_interval = 100;
>   EXPECT_TRUE(pc_->SetConfiguration(config));
>   PeerConnectionInterface::RTCConfiguration new_config =
>@@ -2315,10 +2487,10 @@ TEST_F(PeerConnectionInterfaceTest, SetConfigurationChangesIceCheckInterval) {
> 
> // Test that when SetConfiguration changes both the pool size and other
> // attributes, the pooled session is created with the updated attributes.
>-TEST_F(PeerConnectionInterfaceTest,
>+TEST_P(PeerConnectionInterfaceTest,
>        SetConfigurationCreatesPooledSessionCorrectly) {
>   CreatePeerConnection();
>-  PeerConnectionInterface::RTCConfiguration config;
>+  PeerConnectionInterface::RTCConfiguration config = pc_->GetConfiguration();
>   config.ice_candidate_pool_size = 1;
>   PeerConnectionInterface::IceServer server;
>   server.uri = kStunAddressOnly;
>@@ -2335,11 +2507,11 @@ TEST_F(PeerConnectionInterfaceTest,
> 
> // Test that after SetLocalDescription, changing the pool size is not allowed,
> // and an invalid modification error is returned.
>-TEST_F(PeerConnectionInterfaceTest,
>+TEST_P(PeerConnectionInterfaceTest,
>        CantChangePoolSizeAfterSetLocalDescription) {
>   CreatePeerConnection();
>   // Start by setting a size of 1.
>-  PeerConnectionInterface::RTCConfiguration config;
>+  PeerConnectionInterface::RTCConfiguration config = pc_->GetConfiguration();
>   config.ice_candidate_pool_size = 1;
>   EXPECT_TRUE(pc_->SetConfiguration(config));
> 
>@@ -2358,12 +2530,12 @@ TEST_F(PeerConnectionInterfaceTest,
> 
> // Test that after setting an answer, extra pooled sessions are discarded. The
> // ICE candidate pool is only intended to be used for the first offer/answer.
>-TEST_F(PeerConnectionInterfaceTest,
>+TEST_P(PeerConnectionInterfaceTest,
>        ExtraPooledSessionsDiscardedAfterApplyingAnswer) {
>   CreatePeerConnection();
> 
>   // Set a larger-than-necessary size.
>-  PeerConnectionInterface::RTCConfiguration config;
>+  PeerConnectionInterface::RTCConfiguration config = pc_->GetConfiguration();
>   config.ice_candidate_pool_size = 4;
>   EXPECT_TRUE(pc_->SetConfiguration(config));
> 
>@@ -2379,10 +2551,10 @@ TEST_F(PeerConnectionInterfaceTest,
> 
> // After Close is called, pooled candidates should be discarded so as to not
> // waste network resources.
>-TEST_F(PeerConnectionInterfaceTest, PooledSessionsDiscardedAfterClose) {
>+TEST_P(PeerConnectionInterfaceTest, PooledSessionsDiscardedAfterClose) {
>   CreatePeerConnection();
> 
>-  PeerConnectionInterface::RTCConfiguration config;
>+  PeerConnectionInterface::RTCConfiguration config = pc_->GetConfiguration();
>   config.ice_candidate_pool_size = 3;
>   EXPECT_TRUE(pc_->SetConfiguration(config));
>   pc_->Close();
>@@ -2395,7 +2567,7 @@ TEST_F(PeerConnectionInterfaceTest, PooledSessionsDiscardedAfterClose) {
> 
> // Test that SetConfiguration returns an invalid modification error if
> // modifying a field in the configuration that isn't allowed to be modified.
>-TEST_F(PeerConnectionInterfaceTest,
>+TEST_P(PeerConnectionInterfaceTest,
>        SetConfigurationReturnsInvalidModificationError) {
>   PeerConnectionInterface::RTCConfiguration config;
>   config.bundle_policy = PeerConnectionInterface::kBundlePolicyBalanced;
>@@ -2403,21 +2575,22 @@ TEST_F(PeerConnectionInterfaceTest,
>   config.continual_gathering_policy = PeerConnectionInterface::GATHER_ONCE;
>   CreatePeerConnection(config, nullptr);
> 
>-  PeerConnectionInterface::RTCConfiguration modified_config = config;
>+  PeerConnectionInterface::RTCConfiguration modified_config =
>+      pc_->GetConfiguration();
>   modified_config.bundle_policy =
>       PeerConnectionInterface::kBundlePolicyMaxBundle;
>   RTCError error;
>   EXPECT_FALSE(pc_->SetConfiguration(modified_config, &error));
>   EXPECT_EQ(RTCErrorType::INVALID_MODIFICATION, error.type());
> 
>-  modified_config = config;
>+  modified_config = pc_->GetConfiguration();
>   modified_config.rtcp_mux_policy =
>       PeerConnectionInterface::kRtcpMuxPolicyRequire;
>   error.set_type(RTCErrorType::NONE);
>   EXPECT_FALSE(pc_->SetConfiguration(modified_config, &error));
>   EXPECT_EQ(RTCErrorType::INVALID_MODIFICATION, error.type());
> 
>-  modified_config = config;
>+  modified_config = pc_->GetConfiguration();
>   modified_config.continual_gathering_policy =
>       PeerConnectionInterface::GATHER_CONTINUALLY;
>   error.set_type(RTCErrorType::NONE);
>@@ -2427,10 +2600,11 @@ TEST_F(PeerConnectionInterfaceTest,
> 
> // Test that SetConfiguration returns a range error if the candidate pool size
> // is negative or larger than allowed by the spec.
>-TEST_F(PeerConnectionInterfaceTest,
>+TEST_P(PeerConnectionInterfaceTest,
>        SetConfigurationReturnsRangeErrorForBadCandidatePoolSize) {
>   PeerConnectionInterface::RTCConfiguration config;
>   CreatePeerConnection(config, nullptr);
>+  config = pc_->GetConfiguration();
> 
>   config.ice_candidate_pool_size = -1;
>   RTCError error;
>@@ -2445,10 +2619,11 @@ TEST_F(PeerConnectionInterfaceTest,
> 
> // Test that SetConfiguration returns a syntax error if parsing an ICE server
> // URL failed.
>-TEST_F(PeerConnectionInterfaceTest,
>+TEST_P(PeerConnectionInterfaceTest,
>        SetConfigurationReturnsSyntaxErrorFromBadIceUrls) {
>   PeerConnectionInterface::RTCConfiguration config;
>   CreatePeerConnection(config, nullptr);
>+  config = pc_->GetConfiguration();
> 
>   PeerConnectionInterface::IceServer bad_server;
>   bad_server.uri = "stunn:www.example.com";
>@@ -2460,10 +2635,11 @@ TEST_F(PeerConnectionInterfaceTest,
> 
> // Test that SetConfiguration returns an invalid parameter error if a TURN
> // IceServer is missing a username or password.
>-TEST_F(PeerConnectionInterfaceTest,
>+TEST_P(PeerConnectionInterfaceTest,
>        SetConfigurationReturnsInvalidParameterIfCredentialsMissing) {
>   PeerConnectionInterface::RTCConfiguration config;
>   CreatePeerConnection(config, nullptr);
>+  config = pc_->GetConfiguration();
> 
>   PeerConnectionInterface::IceServer bad_server;
>   bad_server.uri = "turn:www.example.com";
>@@ -2477,12 +2653,19 @@ TEST_F(PeerConnectionInterfaceTest,
> 
> // Test that PeerConnection::Close changes the states to closed and all remote
> // tracks change state to ended.
>-TEST_F(PeerConnectionInterfaceTest, CloseAndTestStreamsAndStates) {
>+TEST_P(PeerConnectionInterfaceTest, CloseAndTestStreamsAndStates) {
>   // Initialize a PeerConnection and negotiate local and remote session
>   // description.
>   InitiateCall();
>-  ASSERT_EQ(1u, pc_->local_streams()->count());
>-  ASSERT_EQ(1u, pc_->remote_streams()->count());
>+
>+  // With Plan B, verify the stream count. The analog with Unified Plan is the
>+  // RtpTransceiver count.
>+  if (sdp_semantics_ == SdpSemantics::kPlanB) {
>+    ASSERT_EQ(1u, pc_->local_streams()->count());
>+    ASSERT_EQ(1u, pc_->remote_streams()->count());
>+  } else {
>+    ASSERT_EQ(2u, pc_->GetTransceivers().size());
>+  }
> 
>   pc_->Close();
> 
>@@ -2492,23 +2675,35 @@ TEST_F(PeerConnectionInterfaceTest, CloseAndTestStreamsAndStates) {
>   EXPECT_EQ(PeerConnectionInterface::kIceGatheringComplete,
>             pc_->ice_gathering_state());
> 
>-  EXPECT_EQ(1u, pc_->local_streams()->count());
>-  EXPECT_EQ(1u, pc_->remote_streams()->count());
>+  if (sdp_semantics_ == SdpSemantics::kPlanB) {
>+    EXPECT_EQ(1u, pc_->local_streams()->count());
>+    EXPECT_EQ(1u, pc_->remote_streams()->count());
>+  } else {
>+    // Verify that the RtpTransceivers are still present but all stopped.
>+    EXPECT_EQ(2u, pc_->GetTransceivers().size());
>+    for (auto transceiver : pc_->GetTransceivers()) {
>+      EXPECT_TRUE(transceiver->stopped());
>+    }
>+  }
>+
>+  auto audio_receiver = GetFirstReceiverOfType(cricket::MEDIA_TYPE_AUDIO);
>+  ASSERT_TRUE(audio_receiver);
>+  auto video_receiver = GetFirstReceiverOfType(cricket::MEDIA_TYPE_VIDEO);
>+  ASSERT_TRUE(video_receiver);
> 
>-  rtc::scoped_refptr<MediaStreamInterface> remote_stream =
>-      pc_->remote_streams()->at(0);
>   // Track state may be updated asynchronously.
>   EXPECT_EQ_WAIT(MediaStreamTrackInterface::kEnded,
>-                 remote_stream->GetAudioTracks()[0]->state(), kTimeout);
>+                 audio_receiver->track()->state(), kTimeout);
>   EXPECT_EQ_WAIT(MediaStreamTrackInterface::kEnded,
>-                 remote_stream->GetVideoTracks()[0]->state(), kTimeout);
>+                 video_receiver->track()->state(), kTimeout);
> }
> 
> // Test that PeerConnection methods fails gracefully after
> // PeerConnection::Close has been called.
>-TEST_F(PeerConnectionInterfaceTest, CloseAndTestMethods) {
>+// Don't run under Unified Plan since the stream API is not available.
>+TEST_F(PeerConnectionInterfaceTestPlanB, CloseAndTestMethods) {
>   CreatePeerConnectionWithoutDtls();
>-  AddAudioVideoStream(kStreamLabel1, "audio_label", "video_label");
>+  AddAudioVideoStream(kStreamId1, "audio_label", "video_label");
>   CreateOfferAsRemoteDescription();
>   CreateAnswerAsLocalDescription();
> 
>@@ -2521,11 +2716,6 @@ TEST_F(PeerConnectionInterfaceTest, CloseAndTestMethods) {
>   pc_->RemoveStream(local_stream);
>   EXPECT_FALSE(pc_->AddStream(local_stream));
> 
>-  ASSERT_FALSE(local_stream->GetAudioTracks().empty());
>-  rtc::scoped_refptr<webrtc::DtmfSenderInterface> dtmf_sender(
>-      pc_->CreateDtmfSender(local_stream->GetAudioTracks()[0]));
>-  EXPECT_TRUE(NULL == dtmf_sender);  // local stream has been removed.
>-
>   EXPECT_TRUE(pc_->CreateDataChannel("test", NULL) == NULL);
> 
>   EXPECT_TRUE(pc_->local_description() != NULL);
>@@ -2549,7 +2739,7 @@ TEST_F(PeerConnectionInterfaceTest, CloseAndTestMethods) {
> }
> 
> // Test that GetStats can still be called after PeerConnection::Close.
>-TEST_F(PeerConnectionInterfaceTest, CloseAndGetStats) {
>+TEST_P(PeerConnectionInterfaceTest, CloseAndGetStats) {
>   InitiateCall();
>   pc_->Close();
>   DoGetStats(NULL);
>@@ -2562,12 +2752,12 @@ TEST_F(PeerConnectionInterfaceTest, CloseAndGetStats) {
> // This test verifies that the remote MediaStreams corresponding to a received
> // SDP string is created. In this test the two separate MediaStreams are
> // signaled.
>-TEST_F(PeerConnectionInterfaceTest, UpdateRemoteStreams) {
>+TEST_P(PeerConnectionInterfaceTest, UpdateRemoteStreams) {
>   FakeConstraints constraints;
>   constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
>                            true);
>   CreatePeerConnection(&constraints);
>-  CreateAndSetRemoteOffer(kSdpStringWithStream1);
>+  CreateAndSetRemoteOffer(GetSdpStringWithStream1());
> 
>   rtc::scoped_refptr<StreamCollection> reference(CreateStreamCollection(1, 1));
>   EXPECT_TRUE(
>@@ -2577,7 +2767,7 @@ TEST_F(PeerConnectionInterfaceTest, UpdateRemoteStreams) {
> 
>   // Create a session description based on another SDP with another
>   // MediaStream.
>-  CreateAndSetRemoteOffer(kSdpStringWithStream1And2);
>+  CreateAndSetRemoteOffer(GetSdpStringWithStream1And2());
> 
>   rtc::scoped_refptr<StreamCollection> reference2(CreateStreamCollection(2, 1));
>   EXPECT_TRUE(
>@@ -2586,7 +2776,9 @@ TEST_F(PeerConnectionInterfaceTest, UpdateRemoteStreams) {
> 
> // This test verifies that when remote tracks are added/removed from SDP, the
> // created remote streams are updated appropriately.
>-TEST_F(PeerConnectionInterfaceTest,
>+// Don't run under Unified Plan since this test uses Plan B SDP to test Plan B
>+// specific behavior.
>+TEST_F(PeerConnectionInterfaceTestPlanB,
>        AddRemoveTrackFromExistingRemoteMediaStream) {
>   FakeConstraints constraints;
>   constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
>@@ -2631,25 +2823,25 @@ TEST_F(PeerConnectionInterfaceTest,
> 
> // This tests that remote tracks are ended if a local session description is set
> // that rejects the media content type.
>-TEST_F(PeerConnectionInterfaceTest, RejectMediaContent) {
>+TEST_P(PeerConnectionInterfaceTest, RejectMediaContent) {
>   FakeConstraints constraints;
>   constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
>                            true);
>   CreatePeerConnection(&constraints);
>   // First create and set a remote offer, then reject its video content in our
>   // answer.
>-  CreateAndSetRemoteOffer(kSdpStringWithStream1);
>-  ASSERT_EQ(1u, observer_.remote_streams()->count());
>-  MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
>-  ASSERT_EQ(1u, remote_stream->GetVideoTracks().size());
>-  ASSERT_EQ(1u, remote_stream->GetAudioTracks().size());
>+  CreateAndSetRemoteOffer(kSdpStringWithStream1PlanB);
>+  auto audio_receiver = GetFirstReceiverOfType(cricket::MEDIA_TYPE_AUDIO);
>+  ASSERT_TRUE(audio_receiver);
>+  auto video_receiver = GetFirstReceiverOfType(cricket::MEDIA_TYPE_VIDEO);
>+  ASSERT_TRUE(video_receiver);
> 
>-  rtc::scoped_refptr<webrtc::VideoTrackInterface> remote_video =
>-      remote_stream->GetVideoTracks()[0];
>-  EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, remote_video->state());
>-  rtc::scoped_refptr<webrtc::AudioTrackInterface> remote_audio =
>-      remote_stream->GetAudioTracks()[0];
>+  rtc::scoped_refptr<MediaStreamTrackInterface> remote_audio =
>+      audio_receiver->track();
>   EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, remote_audio->state());
>+  rtc::scoped_refptr<MediaStreamTrackInterface> remote_video =
>+      video_receiver->track();
>+  EXPECT_EQ(MediaStreamTrackInterface::kLive, remote_video->state());
> 
>   std::unique_ptr<SessionDescriptionInterface> local_answer;
>   EXPECT_TRUE(DoCreateAnswer(&local_answer, nullptr));
>@@ -2657,8 +2849,8 @@ TEST_F(PeerConnectionInterfaceTest, RejectMediaContent) {
>       local_answer->description()->GetContentByName("video");
>   video_info->rejected = true;
>   EXPECT_TRUE(DoSetLocalDescription(std::move(local_answer)));
>-  EXPECT_EQ(webrtc::MediaStreamTrackInterface::kEnded, remote_video->state());
>-  EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, remote_audio->state());
>+  EXPECT_EQ(MediaStreamTrackInterface::kEnded, remote_video->state());
>+  EXPECT_EQ(MediaStreamTrackInterface::kLive, remote_audio->state());
> 
>   // Now create an offer where we reject both video and audio.
>   std::unique_ptr<SessionDescriptionInterface> local_offer;
>@@ -2672,27 +2864,28 @@ TEST_F(PeerConnectionInterfaceTest, RejectMediaContent) {
>   audio_info->rejected = true;
>   EXPECT_TRUE(DoSetLocalDescription(std::move(local_offer)));
>   // Track state may be updated asynchronously.
>-  EXPECT_EQ_WAIT(webrtc::MediaStreamTrackInterface::kEnded,
>-                 remote_audio->state(), kTimeout);
>-  EXPECT_EQ_WAIT(webrtc::MediaStreamTrackInterface::kEnded,
>-                 remote_video->state(), kTimeout);
>+  EXPECT_EQ_WAIT(MediaStreamTrackInterface::kEnded, remote_audio->state(),
>+                 kTimeout);
>+  EXPECT_EQ_WAIT(MediaStreamTrackInterface::kEnded, remote_video->state(),
>+                 kTimeout);
> }
> 
> // This tests that we won't crash if the remote track has been removed outside
> // of PeerConnection and then PeerConnection tries to reject the track.
>-TEST_F(PeerConnectionInterfaceTest, RemoveTrackThenRejectMediaContent) {
>+// Don't run under Unified Plan since the stream API is not available.
>+TEST_F(PeerConnectionInterfaceTestPlanB, RemoveTrackThenRejectMediaContent) {
>   FakeConstraints constraints;
>   constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
>                            true);
>   CreatePeerConnection(&constraints);
>-  CreateAndSetRemoteOffer(kSdpStringWithStream1);
>+  CreateAndSetRemoteOffer(GetSdpStringWithStream1());
>   MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
>   remote_stream->RemoveTrack(remote_stream->GetVideoTracks()[0]);
>   remote_stream->RemoveTrack(remote_stream->GetAudioTracks()[0]);
> 
>   std::unique_ptr<SessionDescriptionInterface> local_answer(
>-      webrtc::CreateSessionDescription(SdpType::kAnswer, kSdpStringWithStream1,
>-                                       nullptr));
>+      webrtc::CreateSessionDescription(SdpType::kAnswer,
>+                                       GetSdpStringWithStream1(), nullptr));
>   cricket::ContentInfo* video_info =
>       local_answer->description()->GetContentByName("video");
>   video_info->rejected = true;
>@@ -2707,13 +2900,13 @@ TEST_F(PeerConnectionInterfaceTest, RemoveTrackThenRejectMediaContent) {
> // This tests that if a recvonly remote description is set, no remote streams
> // will be created, even if the description contains SSRCs/MSIDs.
> // See: https://code.google.com/p/webrtc/issues/detail?id=5054
>-TEST_F(PeerConnectionInterfaceTest, RecvonlyDescriptionDoesntCreateStream) {
>+TEST_P(PeerConnectionInterfaceTest, RecvonlyDescriptionDoesntCreateStream) {
>   FakeConstraints constraints;
>   constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
>                            true);
>   CreatePeerConnection(&constraints);
> 
>-  std::string recvonly_offer = kSdpStringWithStream1;
>+  std::string recvonly_offer = GetSdpStringWithStream1();
>   rtc::replace_substrs(kSendrecv, strlen(kSendrecv), kRecvonly,
>                        strlen(kRecvonly), &recvonly_offer);
>   CreateAndSetRemoteOffer(recvonly_offer);
>@@ -2725,7 +2918,8 @@ TEST_F(PeerConnectionInterfaceTest, RecvonlyDescriptionDoesntCreateStream) {
> // description doesn't contain any streams and no MSID support.
> // It also tests that the default stream is updated if a video m-line is added
> // in a subsequent session description.
>-TEST_F(PeerConnectionInterfaceTest, SdpWithoutMsidCreatesDefaultStream) {
>+// Don't run under Unified Plan since this behavior is Plan B specific.
>+TEST_F(PeerConnectionInterfaceTestPlanB, SdpWithoutMsidCreatesDefaultStream) {
>   FakeConstraints constraints;
>   constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
>                            true);
>@@ -2737,7 +2931,7 @@ TEST_F(PeerConnectionInterfaceTest, SdpWithoutMsidCreatesDefaultStream) {
> 
>   EXPECT_EQ(1u, remote_stream->GetAudioTracks().size());
>   EXPECT_EQ(0u, remote_stream->GetVideoTracks().size());
>-  EXPECT_EQ("default", remote_stream->label());
>+  EXPECT_EQ("default", remote_stream->id());
> 
>   CreateAndSetRemoteOffer(kSdpStringWithoutStreams);
>   ASSERT_EQ(1u, observer_.remote_streams()->count());
>@@ -2753,7 +2947,8 @@ TEST_F(PeerConnectionInterfaceTest, SdpWithoutMsidCreatesDefaultStream) {
> 
> // This tests that a default MediaStream is created if a remote session
> // description doesn't contain any streams and media direction is send only.
>-TEST_F(PeerConnectionInterfaceTest,
>+// Don't run under Unified Plan since this behavior is Plan B specific.
>+TEST_F(PeerConnectionInterfaceTestPlanB,
>        SendOnlySdpWithoutMsidCreatesDefaultStream) {
>   FakeConstraints constraints;
>   constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
>@@ -2766,17 +2961,18 @@ TEST_F(PeerConnectionInterfaceTest,
> 
>   EXPECT_EQ(1u, remote_stream->GetAudioTracks().size());
>   EXPECT_EQ(1u, remote_stream->GetVideoTracks().size());
>-  EXPECT_EQ("default", remote_stream->label());
>+  EXPECT_EQ("default", remote_stream->id());
> }
> 
> // This tests that it won't crash when PeerConnection tries to remove
> // a remote track that as already been removed from the MediaStream.
>-TEST_F(PeerConnectionInterfaceTest, RemoveAlreadyGoneRemoteStream) {
>+// Don't run under Unified Plan since this behavior is Plan B specific.
>+TEST_F(PeerConnectionInterfaceTestPlanB, RemoveAlreadyGoneRemoteStream) {
>   FakeConstraints constraints;
>   constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
>                            true);
>   CreatePeerConnection(&constraints);
>-  CreateAndSetRemoteOffer(kSdpStringWithStream1);
>+  CreateAndSetRemoteOffer(GetSdpStringWithStream1());
>   MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
>   remote_stream->RemoveTrack(remote_stream->GetAudioTracks()[0]);
>   remote_stream->RemoveTrack(remote_stream->GetVideoTracks()[0]);
>@@ -2789,7 +2985,8 @@ TEST_F(PeerConnectionInterfaceTest, RemoveAlreadyGoneRemoteStream) {
> // This tests that a default MediaStream is created if the remote session
> // description doesn't contain any streams and don't contain an indication if
> // MSID is supported.
>-TEST_F(PeerConnectionInterfaceTest,
>+// Don't run under Unified Plan since this behavior is Plan B specific.
>+TEST_F(PeerConnectionInterfaceTestPlanB,
>        SdpWithoutMsidAndStreamsCreatesDefaultStream) {
>   FakeConstraints constraints;
>   constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
>@@ -2805,7 +3002,8 @@ TEST_F(PeerConnectionInterfaceTest,
> 
> // This tests that a default MediaStream is not created if the remote session
> // description doesn't contain any streams but does support MSID.
>-TEST_F(PeerConnectionInterfaceTest, SdpWithMsidDontCreatesDefaultStream) {
>+// Don't run under Unified Plan since this behavior is Plan B specific.
>+TEST_F(PeerConnectionInterfaceTestPlanB, SdpWithMsidDontCreatesDefaultStream) {
>   FakeConstraints constraints;
>   constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
>                            true);
>@@ -2817,7 +3015,8 @@ TEST_F(PeerConnectionInterfaceTest, SdpWithMsidDontCreatesDefaultStream) {
> // This tests that when setting a new description, the old default tracks are
> // not destroyed and recreated.
> // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=5250
>-TEST_F(PeerConnectionInterfaceTest,
>+// Don't run under Unified Plan since this behavior is Plan B specific.
>+TEST_F(PeerConnectionInterfaceTestPlanB,
>        DefaultTracksNotDestroyedAndRecreated) {
>   FakeConstraints constraints;
>   constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
>@@ -2839,12 +3038,13 @@ TEST_F(PeerConnectionInterfaceTest,
> 
> // This tests that a default MediaStream is not created if a remote session
> // description is updated to not have any MediaStreams.
>-TEST_F(PeerConnectionInterfaceTest, VerifyDefaultStreamIsNotCreated) {
>+// Don't run under Unified Plan since this behavior is Plan B specific.
>+TEST_F(PeerConnectionInterfaceTestPlanB, VerifyDefaultStreamIsNotCreated) {
>   FakeConstraints constraints;
>   constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
>                            true);
>   CreatePeerConnection(&constraints);
>-  CreateAndSetRemoteOffer(kSdpStringWithStream1);
>+  CreateAndSetRemoteOffer(GetSdpStringWithStream1());
>   rtc::scoped_refptr<StreamCollection> reference(CreateStreamCollection(1, 1));
>   EXPECT_TRUE(
>       CompareStreamCollections(observer_.remote_streams(), reference.get()));
>@@ -2853,11 +3053,73 @@ TEST_F(PeerConnectionInterfaceTest, VerifyDefaultStreamIsNotCreated) {
>   EXPECT_EQ(0u, observer_.remote_streams()->count());
> }
> 
>+// This tests that a default MediaStream is created if a remote SDP comes from
>+// an endpoint that doesn't signal SSRCs, but signals media stream IDs.
>+TEST_F(PeerConnectionInterfaceTestPlanB,
>+       SdpWithMsidWithoutSsrcCreatesDefaultStream) {
>+  FakeConstraints constraints;
>+  constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
>+                           true);
>+  CreatePeerConnection(&constraints);
>+  std::string sdp_string = kSdpStringWithoutStreamsAudioOnly;
>+  // Add a=msid lines to simulate a Unified Plan endpoint that only
>+  // signals stream IDs with a=msid lines.
>+  sdp_string.append("a=msid:audio_stream_id audio_track_id\n");
>+
>+  CreateAndSetRemoteOffer(sdp_string);
>+
>+  ASSERT_EQ(1u, observer_.remote_streams()->count());
>+  MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
>+  EXPECT_EQ("default", remote_stream->id());
>+  ASSERT_EQ(1u, remote_stream->GetAudioTracks().size());
>+}
>+
>+// This tests that when a Plan B endpoint receives an SDP that signals no media
>+// stream IDs indicated by the special character "-" in the a=msid line, that
>+// a default stream ID will be used for the MediaStream ID. This can occur
>+// when a Unified Plan endpoint signals no media stream IDs, but signals both
>+// a=ssrc msid and a=msid lines for interop signaling with Plan B.
>+TEST_F(PeerConnectionInterfaceTestPlanB,
>+       SdpWithEmptyMsidAndSsrcCreatesDefaultStreamId) {
>+  FakeConstraints constraints;
>+  constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
>+                           true);
>+  CreatePeerConnection(&constraints);
>+  // Add a a=msid line to the SDP. This is prioritized when parsing the SDP, so
>+  // the sender's stream ID will be interpreted as no stream IDs.
>+  std::string sdp_string = kSdpStringWithStream1AudioTrackOnly;
>+  sdp_string.append("a=msid:- audiotrack0\n");
>+
>+  CreateAndSetRemoteOffer(sdp_string);
>+
>+  ASSERT_EQ(1u, observer_.remote_streams()->count());
>+  // Because SSRCs are signaled the track ID will be what was signaled in the
>+  // a=msid line.
>+  EXPECT_EQ("audiotrack0", observer_.last_added_track_label_);
>+  MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
>+  EXPECT_EQ("default", remote_stream->id());
>+  ASSERT_EQ(1u, remote_stream->GetAudioTracks().size());
>+
>+  // Previously a bug ocurred when setting the remote description a second time.
>+  // This is because we checked equality of the remote StreamParams stream ID
>+  // (empty), and the previously set stream ID for the remote sender
>+  // ("default"). This cause a track to be removed, then added, when really
>+  // nothing should occur because it is the same track.
>+  CreateAndSetRemoteOffer(sdp_string);
>+  EXPECT_EQ(0u, observer_.remove_track_events_.size());
>+  EXPECT_EQ(1u, observer_.add_track_events_.size());
>+  EXPECT_EQ("audiotrack0", observer_.last_added_track_label_);
>+  remote_stream = observer_.remote_streams()->at(0);
>+  EXPECT_EQ("default", remote_stream->id());
>+  ASSERT_EQ(1u, remote_stream->GetAudioTracks().size());
>+}
>+
> // This tests that an RtpSender is created when the local description is set
> // after adding a local stream.
> // TODO(deadbeef): This test and the one below it need to be updated when
> // an RtpSender's lifetime isn't determined by when a local description is set.
>-TEST_F(PeerConnectionInterfaceTest, LocalDescriptionChanged) {
>+// Don't run under Unified Plan since this behavior is Plan B specific.
>+TEST_F(PeerConnectionInterfaceTestPlanB, LocalDescriptionChanged) {
>   FakeConstraints constraints;
>   constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
>                            true);
>@@ -2895,7 +3157,8 @@ TEST_F(PeerConnectionInterfaceTest, LocalDescriptionChanged) {
> 
> // This tests that an RtpSender is created when the local description is set
> // before adding a local stream.
>-TEST_F(PeerConnectionInterfaceTest,
>+// Don't run under Unified Plan since this behavior is Plan B specific.
>+TEST_F(PeerConnectionInterfaceTestPlanB,
>        AddLocalStreamAfterLocalDescriptionChanged) {
>   FakeConstraints constraints;
>   constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
>@@ -2925,23 +3188,23 @@ TEST_F(PeerConnectionInterfaceTest,
> 
> // This tests that the expected behavior occurs if the SSRC on a local track is
> // changed when SetLocalDescription is called.
>-TEST_F(PeerConnectionInterfaceTest,
>+TEST_P(PeerConnectionInterfaceTest,
>        ChangeSsrcOnTrackInLocalSessionDescription) {
>   FakeConstraints constraints;
>   constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
>                            true);
>   CreatePeerConnection(&constraints);
> 
>-  rtc::scoped_refptr<StreamCollection> stream_collection =
>-      CreateStreamCollection(2, 1);
>-  pc_->AddStream(stream_collection->at(0));
>+  AddAudioTrack(kAudioTracks[0]);
>+  AddVideoTrack(kVideoTracks[0]);
>   std::unique_ptr<SessionDescriptionInterface> offer;
>   ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
>   // Grab a copy of the offer before it gets passed into the PC.
>-  auto modified_offer =
>-      rtc::MakeUnique<webrtc::JsepSessionDescription>(webrtc::SdpType::kOffer);
>-  modified_offer->Initialize(offer->description()->Copy(), offer->session_id(),
>-                             offer->session_version());
>+  std::unique_ptr<SessionDescriptionInterface> modified_offer =
>+      webrtc::CreateSessionDescription(
>+          webrtc::SdpType::kOffer, offer->session_id(),
>+          offer->session_version(),
>+          absl::WrapUnique(offer->description()->Copy()));
>   EXPECT_TRUE(DoSetLocalDescription(std::move(offer)));
> 
>   auto senders = pc_->GetSenders();
>@@ -2979,7 +3242,8 @@ TEST_F(PeerConnectionInterfaceTest,
> 
> // This tests that the expected behavior occurs if a new session description is
> // set with the same tracks, but on a different MediaStream.
>-TEST_F(PeerConnectionInterfaceTest,
>+// Don't run under Unified Plan since the stream API is not available.
>+TEST_F(PeerConnectionInterfaceTestPlanB,
>        SignalSameTracksInSeparateMediaStream) {
>   FakeConstraints constraints;
>   constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
>@@ -3018,7 +3282,7 @@ TEST_F(PeerConnectionInterfaceTest,
> }
> 
> // This tests that PeerConnectionObserver::OnAddTrack is correctly called.
>-TEST_F(PeerConnectionInterfaceTest, OnAddTrackCallback) {
>+TEST_P(PeerConnectionInterfaceTest, OnAddTrackCallback) {
>   FakeConstraints constraints;
>   constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
>                            true);
>@@ -3028,24 +3292,26 @@ TEST_F(PeerConnectionInterfaceTest, OnAddTrackCallback) {
>   EXPECT_EQ(observer_.last_added_track_label_, kAudioTracks[0]);
> 
>   // Create and set the updated remote SDP.
>-  CreateAndSetRemoteOffer(kSdpStringWithStream1);
>+  CreateAndSetRemoteOffer(kSdpStringWithStream1PlanB);
>   EXPECT_EQ(observer_.num_added_tracks_, 2);
>   EXPECT_EQ(observer_.last_added_track_label_, kVideoTracks[0]);
> }
> 
> // Test that when SetConfiguration is called and the configuration is
> // changing, the next offer causes an ICE restart.
>-TEST_F(PeerConnectionInterfaceTest, SetConfigurationCausingIceRetart) {
>+TEST_P(PeerConnectionInterfaceTest, SetConfigurationCausingIceRestart) {
>   PeerConnectionInterface::RTCConfiguration config;
>   config.type = PeerConnectionInterface::kRelay;
>   // Need to pass default constraints to prevent disabling of DTLS...
>   FakeConstraints default_constraints;
>   CreatePeerConnection(config, &default_constraints);
>-  AddAudioVideoStream(kStreamLabel1, "audio_label", "video_label");
>+  config = pc_->GetConfiguration();
>+  AddAudioTrack(kAudioTracks[0], {kStreamId1});
>+  AddVideoTrack(kVideoTracks[0], {kStreamId1});
> 
>   // Do initial offer/answer so there's something to restart.
>   CreateOfferAsLocalDescription();
>-  CreateAnswerAsRemoteDescription(kSdpStringWithStream1);
>+  CreateAnswerAsRemoteDescription(GetSdpStringWithStream1());
> 
>   // Grab the ufrags.
>   std::vector<std::string> initial_ufrags = GetUfrags(pc_->local_description());
>@@ -3069,17 +3335,19 @@ TEST_F(PeerConnectionInterfaceTest, SetConfigurationCausingIceRetart) {
> 
> // Test that when SetConfiguration is called and the configuration *isn't*
> // changing, the next offer does *not* cause an ICE restart.
>-TEST_F(PeerConnectionInterfaceTest, SetConfigurationNotCausingIceRetart) {
>+TEST_P(PeerConnectionInterfaceTest, SetConfigurationNotCausingIceRestart) {
>   PeerConnectionInterface::RTCConfiguration config;
>   config.type = PeerConnectionInterface::kRelay;
>   // Need to pass default constraints to prevent disabling of DTLS...
>   FakeConstraints default_constraints;
>   CreatePeerConnection(config, &default_constraints);
>-  AddAudioVideoStream(kStreamLabel1, "audio_label", "video_label");
>+  config = pc_->GetConfiguration();
>+  AddAudioTrack(kAudioTracks[0]);
>+  AddVideoTrack(kVideoTracks[0]);
> 
>   // Do initial offer/answer so there's something to restart.
>   CreateOfferAsLocalDescription();
>-  CreateAnswerAsRemoteDescription(kSdpStringWithStream1);
>+  CreateAnswerAsRemoteDescription(GetSdpStringWithStream1());
> 
>   // Grab the ufrags.
>   std::vector<std::string> initial_ufrags = GetUfrags(pc_->local_description());
>@@ -3103,17 +3371,19 @@ TEST_F(PeerConnectionInterfaceTest, SetConfigurationNotCausingIceRetart) {
> // 4. Next createOffer should initiate an ICE restart, but only for the other
> //    m= section; it would be pointless to do an ICE restart for the m= section
> //    that was already restarted.
>-TEST_F(PeerConnectionInterfaceTest, SetConfigurationCausingPartialIceRestart) {
>+TEST_P(PeerConnectionInterfaceTest, SetConfigurationCausingPartialIceRestart) {
>   PeerConnectionInterface::RTCConfiguration config;
>   config.type = PeerConnectionInterface::kRelay;
>   // Need to pass default constraints to prevent disabling of DTLS...
>   FakeConstraints default_constraints;
>   CreatePeerConnection(config, &default_constraints);
>-  AddAudioVideoStream(kStreamLabel1, "audio_label", "video_label");
>+  config = pc_->GetConfiguration();
>+  AddAudioTrack(kAudioTracks[0], {kStreamId1});
>+  AddVideoTrack(kVideoTracks[0], {kStreamId1});
> 
>   // Do initial offer/answer so there's something to restart.
>   CreateOfferAsLocalDescription();
>-  CreateAnswerAsRemoteDescription(kSdpStringWithStream1);
>+  CreateAnswerAsRemoteDescription(GetSdpStringWithStream1());
> 
>   // Change ICE policy, which should set the "needs-ice-restart" flag.
>   config.type = PeerConnectionInterface::kAll;
>@@ -3121,8 +3391,8 @@ TEST_F(PeerConnectionInterfaceTest, SetConfigurationCausingPartialIceRestart) {
> 
>   // Do ICE restart for the first m= section, initiated by remote peer.
>   std::unique_ptr<webrtc::SessionDescriptionInterface> remote_offer(
>-      webrtc::CreateSessionDescription(SdpType::kOffer, kSdpStringWithStream1,
>-                                       nullptr));
>+      webrtc::CreateSessionDescription(SdpType::kOffer,
>+                                       GetSdpStringWithStream1(), nullptr));
>   ASSERT_TRUE(remote_offer);
>   remote_offer->description()->transport_infos()[0].description.ice_ufrag =
>       "modified";
>@@ -3131,13 +3401,13 @@ TEST_F(PeerConnectionInterfaceTest, SetConfigurationCausingPartialIceRestart) {
> 
>   // Grab the ufrags.
>   std::vector<std::string> initial_ufrags = GetUfrags(pc_->local_description());
>-  ASSERT_EQ(2, initial_ufrags.size());
>+  ASSERT_EQ(2U, initial_ufrags.size());
> 
>   // Create offer and grab the new ufrags.
>   CreateOfferAsLocalDescription();
>   std::vector<std::string> subsequent_ufrags =
>       GetUfrags(pc_->local_description());
>-  ASSERT_EQ(2, subsequent_ufrags.size());
>+  ASSERT_EQ(2U, subsequent_ufrags.size());
> 
>   // Ensure that only the ufrag for the second m= section changed.
>   EXPECT_EQ(initial_ufrags[0], subsequent_ufrags[0]);
>@@ -3147,7 +3417,7 @@ TEST_F(PeerConnectionInterfaceTest, SetConfigurationCausingPartialIceRestart) {
> // Tests that the methods to return current/pending descriptions work as
> // expected at different points in the offer/answer exchange. This test does
> // one offer/answer exchange as the offerer, then another as the answerer.
>-TEST_F(PeerConnectionInterfaceTest, CurrentAndPendingDescriptions) {
>+TEST_P(PeerConnectionInterfaceTest, CurrentAndPendingDescriptions) {
>   // This disables DTLS so we can apply an answer to ourselves.
>   CreatePeerConnection();
> 
>@@ -3220,7 +3490,7 @@ TEST_F(PeerConnectionInterfaceTest, CurrentAndPendingDescriptions) {
> // Tests that it won't crash when calling StartRtcEventLog or StopRtcEventLog
> // after the PeerConnection is closed.
> // This version tests the StartRtcEventLog version that receives a file.
>-TEST_F(PeerConnectionInterfaceTest,
>+TEST_P(PeerConnectionInterfaceTest,
>        StartAndStopLoggingToFileAfterPeerConnectionClosed) {
>   CreatePeerConnection();
>   // The RtcEventLog will be reset when the PeerConnection is closed.
>@@ -3245,7 +3515,7 @@ TEST_F(PeerConnectionInterfaceTest,
> // after the PeerConnection is closed.
> // This version tests the StartRtcEventLog version that receives an object
> // of type |RtcEventLogOutput|.
>-TEST_F(PeerConnectionInterfaceTest,
>+TEST_P(PeerConnectionInterfaceTest,
>        StartAndStopLoggingToOutputAfterPeerConnectionClosed) {
>   CreatePeerConnection();
>   // The RtcEventLog will be reset when the PeerConnection is closed.
>@@ -3254,13 +3524,13 @@ TEST_F(PeerConnectionInterfaceTest,
>   rtc::PlatformFile file = 0;
>   int64_t max_size_bytes = 1024;
>   EXPECT_FALSE(pc_->StartRtcEventLog(
>-      rtc::MakeUnique<webrtc::RtcEventLogOutputFile>(file, max_size_bytes),
>+      absl::make_unique<webrtc::RtcEventLogOutputFile>(file, max_size_bytes),
>       webrtc::RtcEventLog::kImmediateOutput));
>   pc_->StopRtcEventLog();
> }
> 
> // Test that generated offers/answers include "ice-option:trickle".
>-TEST_F(PeerConnectionInterfaceTest, OffersAndAnswersHaveTrickleIceOption) {
>+TEST_P(PeerConnectionInterfaceTest, OffersAndAnswersHaveTrickleIceOption) {
>   CreatePeerConnection();
> 
>   // First, create an offer with audio/video.
>@@ -3286,11 +3556,11 @@ TEST_F(PeerConnectionInterfaceTest, OffersAndAnswersHaveTrickleIceOption) {
> 
> // Test that ICE renomination isn't offered if it's not enabled in the PC's
> // RTCConfiguration.
>-TEST_F(PeerConnectionInterfaceTest, IceRenominationNotOffered) {
>+TEST_P(PeerConnectionInterfaceTest, IceRenominationNotOffered) {
>   PeerConnectionInterface::RTCConfiguration config;
>   config.enable_ice_renomination = false;
>   CreatePeerConnection(config, nullptr);
>-  AddVoiceStream("foo");
>+  AddAudioTrack("foo");
> 
>   std::unique_ptr<SessionDescriptionInterface> offer;
>   ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
>@@ -3302,11 +3572,11 @@ TEST_F(PeerConnectionInterfaceTest, IceRenominationNotOffered) {
> 
> // Test that the ICE renomination option is present in generated offers/answers
> // if it's enabled in the PC's RTCConfiguration.
>-TEST_F(PeerConnectionInterfaceTest, IceRenominationOptionInOfferAndAnswer) {
>+TEST_P(PeerConnectionInterfaceTest, IceRenominationOptionInOfferAndAnswer) {
>   PeerConnectionInterface::RTCConfiguration config;
>   config.enable_ice_renomination = true;
>   CreatePeerConnection(config, nullptr);
>-  AddVoiceStream("foo");
>+  AddAudioTrack("foo");
> 
>   std::unique_ptr<SessionDescriptionInterface> offer;
>   ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
>@@ -3329,7 +3599,7 @@ TEST_F(PeerConnectionInterfaceTest, IceRenominationOptionInOfferAndAnswer) {
> // Test that if CreateOffer is called with the deprecated "offer to receive
> // audio/video" constraints, they're processed and result in an offer with
> // audio/video sections just as if RTCOfferAnswerOptions had been used.
>-TEST_F(PeerConnectionInterfaceTest, CreateOfferWithOfferToReceiveConstraints) {
>+TEST_P(PeerConnectionInterfaceTest, CreateOfferWithOfferToReceiveConstraints) {
>   CreatePeerConnection();
> 
>   FakeConstraints constraints;
>@@ -3350,7 +3620,9 @@ TEST_F(PeerConnectionInterfaceTest, CreateOfferWithOfferToReceiveConstraints) {
> // Test that if CreateAnswer is called with the deprecated "offer to receive
> // audio/video" constraints, they're processed and can be used to reject an
> // offered m= section just as can be done with RTCOfferAnswerOptions;
>-TEST_F(PeerConnectionInterfaceTest, CreateAnswerWithOfferToReceiveConstraints) {
>+// Don't run under Unified Plan since this behavior is not supported.
>+TEST_F(PeerConnectionInterfaceTestPlanB,
>+       CreateAnswerWithOfferToReceiveConstraints) {
>   CreatePeerConnection();
> 
>   // First, create an offer with audio/video and apply it as a remote
>@@ -3377,18 +3649,14 @@ TEST_F(PeerConnectionInterfaceTest, CreateAnswerWithOfferToReceiveConstraints) {
>   EXPECT_TRUE(video->rejected);
> }
> 
>-#ifdef HAVE_SCTP
>-#define MAYBE_DataChannelOnlyOfferWithMaxBundlePolicy \
>-  DataChannelOnlyOfferWithMaxBundlePolicy
>-#else
>-#define MAYBE_DataChannelOnlyOfferWithMaxBundlePolicy \
>-  DISABLED_DataChannelOnlyOfferWithMaxBundlePolicy
>-#endif
>-
> // Test that negotiation can succeed with a data channel only, and with the max
> // bundle policy. Previously there was a bug that prevented this.
>-TEST_F(PeerConnectionInterfaceTest,
>-       MAYBE_DataChannelOnlyOfferWithMaxBundlePolicy) {
>+#ifdef HAVE_SCTP
>+TEST_P(PeerConnectionInterfaceTest, DataChannelOnlyOfferWithMaxBundlePolicy) {
>+#else
>+TEST_P(PeerConnectionInterfaceTest,
>+       DISABLED_DataChannelOnlyOfferWithMaxBundlePolicy) {
>+#endif  // HAVE_SCTP
>   PeerConnectionInterface::RTCConfiguration config;
>   config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
>   CreatePeerConnection(config, nullptr);
>@@ -3406,21 +3674,21 @@ TEST_F(PeerConnectionInterfaceTest,
>   EXPECT_TRUE(DoSetLocalDescription(std::move(answer)));
> }
> 
>-TEST_F(PeerConnectionInterfaceTest, SetBitrateWithoutMinSucceeds) {
>+TEST_P(PeerConnectionInterfaceTest, SetBitrateWithoutMinSucceeds) {
>   CreatePeerConnection();
>   PeerConnectionInterface::BitrateParameters bitrate;
>   bitrate.current_bitrate_bps = 100000;
>   EXPECT_TRUE(pc_->SetBitrate(bitrate).ok());
> }
> 
>-TEST_F(PeerConnectionInterfaceTest, SetBitrateNegativeMinFails) {
>+TEST_P(PeerConnectionInterfaceTest, SetBitrateNegativeMinFails) {
>   CreatePeerConnection();
>   PeerConnectionInterface::BitrateParameters bitrate;
>   bitrate.min_bitrate_bps = -1;
>   EXPECT_FALSE(pc_->SetBitrate(bitrate).ok());
> }
> 
>-TEST_F(PeerConnectionInterfaceTest, SetBitrateCurrentLessThanMinFails) {
>+TEST_P(PeerConnectionInterfaceTest, SetBitrateCurrentLessThanMinFails) {
>   CreatePeerConnection();
>   PeerConnectionInterface::BitrateParameters bitrate;
>   bitrate.min_bitrate_bps = 5;
>@@ -3428,14 +3696,14 @@ TEST_F(PeerConnectionInterfaceTest, SetBitrateCurrentLessThanMinFails) {
>   EXPECT_FALSE(pc_->SetBitrate(bitrate).ok());
> }
> 
>-TEST_F(PeerConnectionInterfaceTest, SetBitrateCurrentNegativeFails) {
>+TEST_P(PeerConnectionInterfaceTest, SetBitrateCurrentNegativeFails) {
>   CreatePeerConnection();
>   PeerConnectionInterface::BitrateParameters bitrate;
>   bitrate.current_bitrate_bps = -1;
>   EXPECT_FALSE(pc_->SetBitrate(bitrate).ok());
> }
> 
>-TEST_F(PeerConnectionInterfaceTest, SetBitrateMaxLessThanCurrentFails) {
>+TEST_P(PeerConnectionInterfaceTest, SetBitrateMaxLessThanCurrentFails) {
>   CreatePeerConnection();
>   PeerConnectionInterface::BitrateParameters bitrate;
>   bitrate.current_bitrate_bps = 10;
>@@ -3443,7 +3711,7 @@ TEST_F(PeerConnectionInterfaceTest, SetBitrateMaxLessThanCurrentFails) {
>   EXPECT_FALSE(pc_->SetBitrate(bitrate).ok());
> }
> 
>-TEST_F(PeerConnectionInterfaceTest, SetBitrateMaxLessThanMinFails) {
>+TEST_P(PeerConnectionInterfaceTest, SetBitrateMaxLessThanMinFails) {
>   CreatePeerConnection();
>   PeerConnectionInterface::BitrateParameters bitrate;
>   bitrate.min_bitrate_bps = 10;
>@@ -3451,7 +3719,7 @@ TEST_F(PeerConnectionInterfaceTest, SetBitrateMaxLessThanMinFails) {
>   EXPECT_FALSE(pc_->SetBitrate(bitrate).ok());
> }
> 
>-TEST_F(PeerConnectionInterfaceTest, SetBitrateMaxNegativeFails) {
>+TEST_P(PeerConnectionInterfaceTest, SetBitrateMaxNegativeFails) {
>   CreatePeerConnection();
>   PeerConnectionInterface::BitrateParameters bitrate;
>   bitrate.max_bitrate_bps = -1;
>@@ -3460,7 +3728,7 @@ TEST_F(PeerConnectionInterfaceTest, SetBitrateMaxNegativeFails) {
> 
> // ice_regather_interval_range requires WebRTC to be configured for continual
> // gathering already.
>-TEST_F(PeerConnectionInterfaceTest,
>+TEST_P(PeerConnectionInterfaceTest,
>        SetIceRegatherIntervalRangeWithoutContinualGatheringFails) {
>   PeerConnectionInterface::RTCConfiguration config;
>   config.ice_regather_interval_range.emplace(1000, 2000);
>@@ -3471,7 +3739,7 @@ TEST_F(PeerConnectionInterfaceTest,
> 
> // Ensures that there is no error when ice_regather_interval_range is set with
> // continual gathering enabled.
>-TEST_F(PeerConnectionInterfaceTest,
>+TEST_P(PeerConnectionInterfaceTest,
>        SetIceRegatherIntervalRangeWithContinualGathering) {
>   PeerConnectionInterface::RTCConfiguration config;
>   config.ice_regather_interval_range.emplace(1000, 2000);
>@@ -3480,11 +3748,11 @@ TEST_F(PeerConnectionInterfaceTest,
>   CreatePeerConnection(config, nullptr);
> }
> 
>-// The current bitrate from Call's BitrateConfigMask is currently clamped by
>-// Call's BitrateConfig, which comes from the SDP or a default value. This test
>-// checks that a call to SetBitrate with a current bitrate that will be clamped
>-// succeeds.
>-TEST_F(PeerConnectionInterfaceTest, SetBitrateCurrentLessThanImplicitMin) {
>+// The current bitrate from BitrateSettings is currently clamped
>+// by Call's BitrateConstraints, which comes from the SDP or a default value.
>+// This test checks that a call to SetBitrate with a current bitrate that will
>+// be clamped succeeds.
>+TEST_P(PeerConnectionInterfaceTest, SetBitrateCurrentLessThanImplicitMin) {
>   CreatePeerConnection();
>   PeerConnectionInterface::BitrateParameters bitrate;
>   bitrate.current_bitrate_bps = 1;
>@@ -3492,7 +3760,7 @@ TEST_F(PeerConnectionInterfaceTest, SetBitrateCurrentLessThanImplicitMin) {
> }
> 
> // The following tests verify that the offer can be created correctly.
>-TEST_F(PeerConnectionInterfaceTest,
>+TEST_P(PeerConnectionInterfaceTest,
>        CreateOfferFailsWithInvalidOfferToReceiveAudio) {
>   RTCOfferAnswerOptions rtc_options;
> 
>@@ -3507,7 +3775,7 @@ TEST_F(PeerConnectionInterfaceTest,
>   EXPECT_FALSE(CreateOfferWithOptions(rtc_options));
> }
> 
>-TEST_F(PeerConnectionInterfaceTest,
>+TEST_P(PeerConnectionInterfaceTest,
>        CreateOfferFailsWithInvalidOfferToReceiveVideo) {
>   RTCOfferAnswerOptions rtc_options;
> 
>@@ -3524,7 +3792,7 @@ TEST_F(PeerConnectionInterfaceTest,
> 
> // Test that the audio and video content will be added to an offer if both
> // |offer_to_receive_audio| and |offer_to_receive_video| options are 1.
>-TEST_F(PeerConnectionInterfaceTest, CreateOfferWithAudioVideoOptions) {
>+TEST_P(PeerConnectionInterfaceTest, CreateOfferWithAudioVideoOptions) {
>   RTCOfferAnswerOptions rtc_options;
>   rtc_options.offer_to_receive_audio = 1;
>   rtc_options.offer_to_receive_video = 1;
>@@ -3539,7 +3807,7 @@ TEST_F(PeerConnectionInterfaceTest, CreateOfferWithAudioVideoOptions) {
> 
> // Test that only audio content will be added to the offer if only
> // |offer_to_receive_audio| options is 1.
>-TEST_F(PeerConnectionInterfaceTest, CreateOfferWithAudioOnlyOptions) {
>+TEST_P(PeerConnectionInterfaceTest, CreateOfferWithAudioOnlyOptions) {
>   RTCOfferAnswerOptions rtc_options;
>   rtc_options.offer_to_receive_audio = 1;
>   rtc_options.offer_to_receive_video = 0;
>@@ -3554,7 +3822,7 @@ TEST_F(PeerConnectionInterfaceTest, CreateOfferWithAudioOnlyOptions) {
> 
> // Test that only video content will be added if only |offer_to_receive_video|
> // options is 1.
>-TEST_F(PeerConnectionInterfaceTest, CreateOfferWithVideoOnlyOptions) {
>+TEST_P(PeerConnectionInterfaceTest, CreateOfferWithVideoOnlyOptions) {
>   RTCOfferAnswerOptions rtc_options;
>   rtc_options.offer_to_receive_audio = 0;
>   rtc_options.offer_to_receive_video = 1;
>@@ -3569,7 +3837,7 @@ TEST_F(PeerConnectionInterfaceTest, CreateOfferWithVideoOnlyOptions) {
> 
> // Test that no media content will be added to the offer if using default
> // RTCOfferAnswerOptions.
>-TEST_F(PeerConnectionInterfaceTest, CreateOfferWithDefaultOfferAnswerOptions) {
>+TEST_P(PeerConnectionInterfaceTest, CreateOfferWithDefaultOfferAnswerOptions) {
>   RTCOfferAnswerOptions rtc_options;
> 
>   std::unique_ptr<SessionDescriptionInterface> offer;
>@@ -3582,39 +3850,35 @@ TEST_F(PeerConnectionInterfaceTest, CreateOfferWithDefaultOfferAnswerOptions) {
> 
> // Test that if |ice_restart| is true, the ufrag/pwd will change, otherwise
> // ufrag/pwd will be the same in the new offer.
>-TEST_F(PeerConnectionInterfaceTest, CreateOfferWithIceRestart) {
>+TEST_P(PeerConnectionInterfaceTest, CreateOfferWithIceRestart) {
>+  CreatePeerConnection();
>+
>   RTCOfferAnswerOptions rtc_options;
>   rtc_options.ice_restart = false;
>   rtc_options.offer_to_receive_audio = 1;
> 
>   std::unique_ptr<SessionDescriptionInterface> offer;
>-  CreatePeerConnection();
>   CreateOfferWithOptionsAsLocalDescription(&offer, rtc_options);
>-  auto ufrag1 = offer->description()
>-                    ->GetTransportInfoByName(cricket::CN_AUDIO)
>-                    ->description.ice_ufrag;
>-  auto pwd1 = offer->description()
>-                  ->GetTransportInfoByName(cricket::CN_AUDIO)
>-                  ->description.ice_pwd;
>+  std::string mid = cricket::GetFirstAudioContent(offer->description())->name;
>+  auto ufrag1 =
>+      offer->description()->GetTransportInfoByName(mid)->description.ice_ufrag;
>+  auto pwd1 =
>+      offer->description()->GetTransportInfoByName(mid)->description.ice_pwd;
> 
>   // |ice_restart| is false, the ufrag/pwd shouldn't change.
>   CreateOfferWithOptionsAsLocalDescription(&offer, rtc_options);
>-  auto ufrag2 = offer->description()
>-                    ->GetTransportInfoByName(cricket::CN_AUDIO)
>-                    ->description.ice_ufrag;
>-  auto pwd2 = offer->description()
>-                  ->GetTransportInfoByName(cricket::CN_AUDIO)
>-                  ->description.ice_pwd;
>+  auto ufrag2 =
>+      offer->description()->GetTransportInfoByName(mid)->description.ice_ufrag;
>+  auto pwd2 =
>+      offer->description()->GetTransportInfoByName(mid)->description.ice_pwd;
> 
>   // |ice_restart| is true, the ufrag/pwd should change.
>   rtc_options.ice_restart = true;
>   CreateOfferWithOptionsAsLocalDescription(&offer, rtc_options);
>-  auto ufrag3 = offer->description()
>-                    ->GetTransportInfoByName(cricket::CN_AUDIO)
>-                    ->description.ice_ufrag;
>-  auto pwd3 = offer->description()
>-                  ->GetTransportInfoByName(cricket::CN_AUDIO)
>-                  ->description.ice_pwd;
>+  auto ufrag3 =
>+      offer->description()->GetTransportInfoByName(mid)->description.ice_ufrag;
>+  auto pwd3 =
>+      offer->description()->GetTransportInfoByName(mid)->description.ice_pwd;
> 
>   EXPECT_EQ(ufrag1, ufrag2);
>   EXPECT_EQ(pwd1, pwd2);
>@@ -3624,7 +3888,7 @@ TEST_F(PeerConnectionInterfaceTest, CreateOfferWithIceRestart) {
> 
> // Test that if |use_rtp_mux| is true, the bundling will be enabled in the
> // offer; if it is false, there won't be any bundle group in the offer.
>-TEST_F(PeerConnectionInterfaceTest, CreateOfferWithRtpMux) {
>+TEST_P(PeerConnectionInterfaceTest, CreateOfferWithRtpMux) {
>   RTCOfferAnswerOptions rtc_options;
>   rtc_options.offer_to_receive_audio = 1;
>   rtc_options.offer_to_receive_video = 1;
>@@ -3652,10 +3916,12 @@ TEST_F(PeerConnectionInterfaceTest, CreateOfferWithRtpMux) {
> // PeerConnection -> AddTrack.
> // The test can be removed once addStream is rewritten in terms of addTrack
> // https://bugs.chromium.org/p/webrtc/issues/detail?id=7815
>-TEST_F(PeerConnectionInterfaceTest, MediaStreamAddTrackRemoveTrackRenegotiate) {
>+// Don't run under Unified Plan since the stream API is not available.
>+TEST_F(PeerConnectionInterfaceTestPlanB,
>+       MediaStreamAddTrackRemoveTrackRenegotiate) {
>   CreatePeerConnectionWithoutDtls();
>   rtc::scoped_refptr<MediaStreamInterface> stream(
>-      pc_factory_->CreateLocalMediaStream(kStreamLabel1));
>+      pc_factory_->CreateLocalMediaStream(kStreamId1));
>   pc_->AddStream(stream);
>   rtc::scoped_refptr<AudioTrackInterface> audio_track(
>       pc_factory_->CreateAudioTrack("audio_track", nullptr));
>@@ -3683,18 +3949,18 @@ TEST_F(PeerConnectionInterfaceTest, MediaStreamAddTrackRemoveTrackRenegotiate) {
> 
> // Tests that an error is returned if a description is applied that has fewer
> // media sections than the existing description.
>-TEST_F(PeerConnectionInterfaceTest,
>+TEST_P(PeerConnectionInterfaceTest,
>        MediaSectionCountEnforcedForSubsequentOffer) {
>   CreatePeerConnection();
>-  FakeConstraints constraints;
>-  constraints.SetMandatoryReceiveAudio(true);
>-  constraints.SetMandatoryReceiveVideo(true);
>+  AddAudioTrack("audio_label");
>+  AddVideoTrack("video_label");
>+
>   std::unique_ptr<SessionDescriptionInterface> offer;
>-  ASSERT_TRUE(DoCreateOffer(&offer, &constraints));
>+  ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
>   EXPECT_TRUE(DoSetRemoteDescription(std::move(offer)));
> 
>   // A remote offer with fewer media sections should be rejected.
>-  ASSERT_TRUE(DoCreateOffer(&offer, &constraints));
>+  ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
>   offer->description()->contents().pop_back();
>   offer->description()->contents().pop_back();
>   ASSERT_TRUE(offer->description()->contents().empty());
>@@ -3705,13 +3971,18 @@ TEST_F(PeerConnectionInterfaceTest,
>   EXPECT_TRUE(DoSetLocalDescription(std::move(answer)));
> 
>   // A subsequent local offer with fewer media sections should be rejected.
>-  ASSERT_TRUE(DoCreateOffer(&offer, &constraints));
>+  ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
>   offer->description()->contents().pop_back();
>   offer->description()->contents().pop_back();
>   ASSERT_TRUE(offer->description()->contents().empty());
>   EXPECT_FALSE(DoSetLocalDescription(std::move(offer)));
> }
> 
>+INSTANTIATE_TEST_CASE_P(PeerConnectionInterfaceTest,
>+                        PeerConnectionInterfaceTest,
>+                        Values(SdpSemantics::kPlanB,
>+                               SdpSemantics::kUnifiedPlan));
>+
> class PeerConnectionMediaConfigTest : public testing::Test {
>  protected:
>   void SetUp() override {
>@@ -3741,9 +4012,10 @@ TEST_F(PeerConnectionMediaConfigTest, TestDefaults) {
>       TestCreatePeerConnection(config, &constraints);
> 
>   EXPECT_FALSE(media_config.enable_dscp);
>-  EXPECT_TRUE(media_config.video.enable_cpu_overuse_detection);
>-  EXPECT_FALSE(media_config.video.disable_prerenderer_smoothing);
>+  EXPECT_TRUE(media_config.video.enable_cpu_adaptation);
>+  EXPECT_TRUE(media_config.video.enable_prerenderer_smoothing);
>   EXPECT_FALSE(media_config.video.suspend_below_min_bitrate);
>+  EXPECT_FALSE(media_config.video.experiment_cpu_load_estimator);
> }
> 
> // This test verifies the DSCP constraint is recognized and passed to
>@@ -3770,10 +4042,10 @@ TEST_F(PeerConnectionMediaConfigTest, TestCpuOveruseConstraintFalse) {
>   const cricket::MediaConfig media_config =
>       TestCreatePeerConnection(config, &constraints);
> 
>-  EXPECT_FALSE(media_config.video.enable_cpu_overuse_detection);
>+  EXPECT_FALSE(media_config.video.enable_cpu_adaptation);
> }
> 
>-// This test verifies that the disable_prerenderer_smoothing flag is
>+// This test verifies that the enable_prerenderer_smoothing flag is
> // propagated from RTCConfiguration to the PeerConnection.
> TEST_F(PeerConnectionMediaConfigTest, TestDisablePrerendererSmoothingTrue) {
>   PeerConnectionInterface::RTCConfiguration config;
>@@ -3783,7 +4055,20 @@ TEST_F(PeerConnectionMediaConfigTest, TestDisablePrerendererSmoothingTrue) {
>   const cricket::MediaConfig& media_config =
>       TestCreatePeerConnection(config, &constraints);
> 
>-  EXPECT_TRUE(media_config.video.disable_prerenderer_smoothing);
>+  EXPECT_FALSE(media_config.video.enable_prerenderer_smoothing);
>+}
>+
>+// This test verifies that the experiment_cpu_load_estimator flag is
>+// propagated from RTCConfiguration to the PeerConnection.
>+TEST_F(PeerConnectionMediaConfigTest, TestEnableExperimentCpuLoadEstimator) {
>+  PeerConnectionInterface::RTCConfiguration config;
>+  FakeConstraints constraints;
>+
>+  config.set_experiment_cpu_load_estimator(true);
>+  const cricket::MediaConfig& media_config =
>+      TestCreatePeerConnection(config, &constraints);
>+
>+  EXPECT_TRUE(media_config.video.experiment_cpu_load_estimator);
> }
> 
> // This test verifies the suspend below min bitrate constraint is
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectioninternal.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectioninternal.h
>new file mode 100644
>index 00000000000..8a169547cc4
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectioninternal.h
>@@ -0,0 +1,93 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef PC_PEERCONNECTIONINTERNAL_H_
>+#define PC_PEERCONNECTIONINTERNAL_H_
>+
>+#include <map>
>+#include <memory>
>+#include <set>
>+#include <string>
>+#include <vector>
>+
>+#include "api/peerconnectioninterface.h"
>+#include "call/call.h"
>+#include "pc/datachannel.h"
>+#include "pc/rtptransceiver.h"
>+
>+namespace webrtc {
>+
>+// Internal interface for extra PeerConnection methods.
>+class PeerConnectionInternal : public PeerConnectionInterface {
>+ public:
>+  virtual rtc::Thread* network_thread() const = 0;
>+  virtual rtc::Thread* worker_thread() const = 0;
>+  virtual rtc::Thread* signaling_thread() const = 0;
>+
>+  // The SDP session ID as defined by RFC 3264.
>+  virtual std::string session_id() const = 0;
>+
>+  // Returns true if we were the initial offerer.
>+  virtual bool initial_offerer() const = 0;
>+
>+  virtual std::vector<
>+      rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>>
>+  GetTransceiversInternal() const = 0;
>+
>+  // Get the id used as a media stream track's "id" field from ssrc.
>+  virtual bool GetLocalTrackIdBySsrc(uint32_t ssrc, std::string* track_id) = 0;
>+  virtual bool GetRemoteTrackIdBySsrc(uint32_t ssrc, std::string* track_id) = 0;
>+
>+  virtual sigslot::signal1<DataChannel*>& SignalDataChannelCreated() = 0;
>+
>+  // Only valid when using deprecated RTP data channels.
>+  virtual cricket::RtpDataChannel* rtp_data_channel() const = 0;
>+
>+  virtual std::vector<rtc::scoped_refptr<DataChannel>> sctp_data_channels()
>+      const = 0;
>+
>+  virtual absl::optional<std::string> sctp_content_name() const = 0;
>+  virtual absl::optional<std::string> sctp_transport_name() const = 0;
>+
>+  virtual cricket::CandidateStatsList GetPooledCandidateStats() const = 0;
>+
>+  // Returns a map from MID to transport name for all active media sections.
>+  virtual std::map<std::string, std::string> GetTransportNamesByMid() const = 0;
>+
>+  // Returns a map from transport name to transport stats for all given
>+  // transport names.
>+  virtual std::map<std::string, cricket::TransportStats>
>+  GetTransportStatsByNames(const std::set<std::string>& transport_names) = 0;
>+
>+  virtual Call::Stats GetCallStats() = 0;
>+
>+  virtual bool GetLocalCertificate(
>+      const std::string& transport_name,
>+      rtc::scoped_refptr<rtc::RTCCertificate>* certificate) = 0;
>+  virtual std::unique_ptr<rtc::SSLCertChain> GetRemoteSSLCertChain(
>+      const std::string& transport_name) = 0;
>+
>+  // Returns true if there was an ICE restart initiated by the remote offer.
>+  virtual bool IceRestartPending(const std::string& content_name) const = 0;
>+
>+  // Returns true if the ICE restart flag above was set, and no ICE restart has
>+  // occurred yet for this transport (by applying a local description with
>+  // changed ufrag/password). If the transport has been deleted as a result of
>+  // bundling, returns false.
>+  virtual bool NeedsIceRestart(const std::string& content_name) const = 0;
>+
>+  // Get SSL role for an arbitrary m= section (handles bundling correctly).
>+  virtual bool GetSslRole(const std::string& content_name,
>+                          rtc::SSLRole* role) = 0;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // PC_PEERCONNECTIONINTERNAL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectionwrapper.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectionwrapper.cc
>index 21d154289f1..5006337f875 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectionwrapper.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectionwrapper.cc
>@@ -15,15 +15,17 @@
> #include <utility>
> #include <vector>
> 
>+#include "absl/memory/memory.h"
> #include "api/jsepsessiondescription.h"
>-#include "media/base/fakevideocapturer.h"
> #include "pc/sdputils.h"
>+#include "pc/test/fakevideotracksource.h"
> #include "rtc_base/function_view.h"
> #include "rtc_base/gunit.h"
>-#include "rtc_base/ptr_util.h"
> 
> namespace webrtc {
> 
>+using RTCOfferAnswerOptions = PeerConnectionInterface::RTCOfferAnswerOptions;
>+
> namespace {
> const uint32_t kDefaultTimeout = 10000U;
> }
>@@ -57,7 +59,7 @@ MockPeerConnectionObserver* PeerConnectionWrapper::observer() {
> 
> std::unique_ptr<SessionDescriptionInterface>
> PeerConnectionWrapper::CreateOffer() {
>-  return CreateOffer(PeerConnectionInterface::RTCOfferAnswerOptions());
>+  return CreateOffer(RTCOfferAnswerOptions());
> }
> 
> std::unique_ptr<SessionDescriptionInterface> PeerConnectionWrapper::CreateOffer(
>@@ -72,8 +74,7 @@ std::unique_ptr<SessionDescriptionInterface> PeerConnectionWrapper::CreateOffer(
> 
> std::unique_ptr<SessionDescriptionInterface>
> PeerConnectionWrapper::CreateOfferAndSetAsLocal() {
>-  return CreateOfferAndSetAsLocal(
>-      PeerConnectionInterface::RTCOfferAnswerOptions());
>+  return CreateOfferAndSetAsLocal(RTCOfferAnswerOptions());
> }
> 
> std::unique_ptr<SessionDescriptionInterface>
>@@ -89,7 +90,7 @@ PeerConnectionWrapper::CreateOfferAndSetAsLocal(
> 
> std::unique_ptr<SessionDescriptionInterface>
> PeerConnectionWrapper::CreateAnswer() {
>-  return CreateAnswer(PeerConnectionInterface::RTCOfferAnswerOptions());
>+  return CreateAnswer(RTCOfferAnswerOptions());
> }
> 
> std::unique_ptr<SessionDescriptionInterface>
>@@ -105,8 +106,7 @@ PeerConnectionWrapper::CreateAnswer(
> 
> std::unique_ptr<SessionDescriptionInterface>
> PeerConnectionWrapper::CreateAnswerAndSetAsLocal() {
>-  return CreateAnswerAndSetAsLocal(
>-      PeerConnectionInterface::RTCOfferAnswerOptions());
>+  return CreateAnswerAndSetAsLocal(RTCOfferAnswerOptions());
> }
> 
> std::unique_ptr<SessionDescriptionInterface>
>@@ -181,12 +181,20 @@ bool PeerConnectionWrapper::SetSdp(
> 
> bool PeerConnectionWrapper::ExchangeOfferAnswerWith(
>     PeerConnectionWrapper* answerer) {
>+  return ExchangeOfferAnswerWith(answerer, RTCOfferAnswerOptions(),
>+                                 RTCOfferAnswerOptions());
>+}
>+
>+bool PeerConnectionWrapper::ExchangeOfferAnswerWith(
>+    PeerConnectionWrapper* answerer,
>+    const PeerConnectionInterface::RTCOfferAnswerOptions& offer_options,
>+    const PeerConnectionInterface::RTCOfferAnswerOptions& answer_options) {
>   RTC_DCHECK(answerer);
>   if (answerer == this) {
>     RTC_LOG(LS_ERROR) << "Cannot exchange offer/answer with ourself!";
>     return false;
>   }
>-  auto offer = CreateOffer();
>+  auto offer = CreateOffer(offer_options);
>   EXPECT_TRUE(offer);
>   if (!offer) {
>     return false;
>@@ -202,7 +210,7 @@ bool PeerConnectionWrapper::ExchangeOfferAnswerWith(
>   if (!set_remote_offer) {
>     return false;
>   }
>-  auto answer = answerer->CreateAnswer();
>+  auto answer = answerer->CreateAnswer(answer_options);
>   EXPECT_TRUE(answer);
>   if (!answer) {
>     return false;
>@@ -261,30 +269,28 @@ rtc::scoped_refptr<AudioTrackInterface> PeerConnectionWrapper::CreateAudioTrack(
> 
> rtc::scoped_refptr<VideoTrackInterface> PeerConnectionWrapper::CreateVideoTrack(
>     const std::string& label) {
>-  auto video_source = pc_factory()->CreateVideoSource(
>-      rtc::MakeUnique<cricket::FakeVideoCapturer>());
>-  return pc_factory()->CreateVideoTrack(label, video_source);
>+  return pc_factory()->CreateVideoTrack(label, FakeVideoTrackSource::Create());
> }
> 
> rtc::scoped_refptr<RtpSenderInterface> PeerConnectionWrapper::AddTrack(
>     rtc::scoped_refptr<MediaStreamTrackInterface> track,
>-    const std::vector<std::string>& stream_labels) {
>+    const std::vector<std::string>& stream_ids) {
>   RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>> result =
>-      pc()->AddTrack(track, stream_labels);
>+      pc()->AddTrack(track, stream_ids);
>   EXPECT_EQ(RTCErrorType::NONE, result.error().type());
>   return result.MoveValue();
> }
> 
> rtc::scoped_refptr<RtpSenderInterface> PeerConnectionWrapper::AddAudioTrack(
>     const std::string& track_label,
>-    const std::vector<std::string>& stream_labels) {
>-  return AddTrack(CreateAudioTrack(track_label), stream_labels);
>+    const std::vector<std::string>& stream_ids) {
>+  return AddTrack(CreateAudioTrack(track_label), stream_ids);
> }
> 
> rtc::scoped_refptr<RtpSenderInterface> PeerConnectionWrapper::AddVideoTrack(
>     const std::string& track_label,
>-    const std::vector<std::string>& stream_labels) {
>-  return AddTrack(CreateVideoTrack(track_label), stream_labels);
>+    const std::vector<std::string>& stream_ids) {
>+  return AddTrack(CreateVideoTrack(track_label), stream_ids);
> }
> 
> rtc::scoped_refptr<DataChannelInterface>
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectionwrapper.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectionwrapper.h
>index b5aa16342b3..436460e8729 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectionwrapper.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnectionwrapper.h
>@@ -97,16 +97,20 @@ class PeerConnectionWrapper {
>   // generating the offer and the given PeerConnectionWrapper generating the
>   // answer.
>   // Equivalent to:
>-  // 1. this->CreateOffer()
>+  // 1. this->CreateOffer(offer_options)
>   // 2. this->SetLocalDescription(offer)
>   // 3. answerer->SetRemoteDescription(offer)
>-  // 4. answerer->CreateAnswer()
>+  // 4. answerer->CreateAnswer(answer_options)
>   // 5. answerer->SetLocalDescription(answer)
>   // 6. this->SetRemoteDescription(answer)
>   // Returns true if all steps succeed, false otherwise.
>   // Suggested usage:
>   //   ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
>   bool ExchangeOfferAnswerWith(PeerConnectionWrapper* answerer);
>+  bool ExchangeOfferAnswerWith(
>+      PeerConnectionWrapper* answerer,
>+      const PeerConnectionInterface::RTCOfferAnswerOptions& offer_options,
>+      const PeerConnectionInterface::RTCOfferAnswerOptions& answer_options);
> 
>   // The following are wrappers for the underlying PeerConnection's
>   // AddTransceiver method. They return the result of calling AddTransceiver
>@@ -134,19 +138,19 @@ class PeerConnectionWrapper {
>   // AddTrack fails.
>   rtc::scoped_refptr<RtpSenderInterface> AddTrack(
>       rtc::scoped_refptr<MediaStreamTrackInterface> track,
>-      const std::vector<std::string>& stream_labels = {});
>+      const std::vector<std::string>& stream_ids = {});
> 
>   // Calls the underlying PeerConnection's AddTrack method with an audio media
>   // stream track not bound to any source.
>   rtc::scoped_refptr<RtpSenderInterface> AddAudioTrack(
>       const std::string& track_label,
>-      const std::vector<std::string>& stream_labels = {});
>+      const std::vector<std::string>& stream_ids = {});
> 
>   // Calls the underlying PeerConnection's AddTrack method with a video media
>-  // stream track fed by a fake video capturer.
>+  // stream track fed by a FakeVideoTrackSource.
>   rtc::scoped_refptr<RtpSenderInterface> AddVideoTrack(
>       const std::string& track_label,
>-      const std::vector<std::string>& stream_labels = {});
>+      const std::vector<std::string>& stream_ids = {});
> 
>   // Calls the underlying PeerConnection's CreateDataChannel method with default
>   // initialization parameters.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/proxy_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/proxy_unittest.cc
>index 81f040adf64..072e606e308 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/proxy_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/proxy_unittest.cc
>@@ -64,26 +64,26 @@ class Fake : public FakeInterface {
> 
> // Proxies for the test interface.
> BEGIN_PROXY_MAP(Fake)
>-  PROXY_WORKER_THREAD_DESTRUCTOR()
>-  PROXY_METHOD0(void, VoidMethod0)
>-  PROXY_METHOD0(std::string, Method0)
>-  PROXY_CONSTMETHOD0(std::string, ConstMethod0)
>-  PROXY_WORKER_METHOD1(std::string, Method1, std::string)
>-  PROXY_CONSTMETHOD1(std::string, ConstMethod1, std::string)
>-  PROXY_WORKER_METHOD2(std::string, Method2, std::string, std::string)
>+PROXY_WORKER_THREAD_DESTRUCTOR()
>+PROXY_METHOD0(void, VoidMethod0)
>+PROXY_METHOD0(std::string, Method0)
>+PROXY_CONSTMETHOD0(std::string, ConstMethod0)
>+PROXY_WORKER_METHOD1(std::string, Method1, std::string)
>+PROXY_CONSTMETHOD1(std::string, ConstMethod1, std::string)
>+PROXY_WORKER_METHOD2(std::string, Method2, std::string, std::string)
> END_PROXY_MAP()
> 
> // Preprocessor hack to get a proxy class a name different than FakeProxy.
> #define FakeProxy FakeSignalingProxy
> #define FakeProxyWithInternal FakeSignalingProxyWithInternal
> BEGIN_SIGNALING_PROXY_MAP(Fake)
>-  PROXY_SIGNALING_THREAD_DESTRUCTOR()
>-  PROXY_METHOD0(void, VoidMethod0)
>-  PROXY_METHOD0(std::string, Method0)
>-  PROXY_CONSTMETHOD0(std::string, ConstMethod0)
>-  PROXY_METHOD1(std::string, Method1, std::string)
>-  PROXY_CONSTMETHOD1(std::string, ConstMethod1, std::string)
>-  PROXY_METHOD2(std::string, Method2, std::string, std::string)
>+PROXY_SIGNALING_THREAD_DESTRUCTOR()
>+PROXY_METHOD0(void, VoidMethod0)
>+PROXY_METHOD0(std::string, Method0)
>+PROXY_CONSTMETHOD0(std::string, ConstMethod0)
>+PROXY_METHOD1(std::string, Method1, std::string)
>+PROXY_CONSTMETHOD1(std::string, ConstMethod1, std::string)
>+PROXY_METHOD2(std::string, Method2, std::string, std::string)
> END_PROXY_MAP()
> #undef FakeProxy
> 
>@@ -215,21 +215,17 @@ TEST_F(ProxyTest, VoidMethod0) {
> TEST_F(ProxyTest, Method0) {
>   EXPECT_CALL(*fake_, Method0())
>       .Times(Exactly(1))
>-      .WillOnce(
>-          DoAll(InvokeWithoutArgs(this, &ProxyTest::CheckSignalingThread),
>-                Return("Method0")));
>-  EXPECT_EQ("Method0",
>-            fake_proxy_->Method0());
>+      .WillOnce(DoAll(InvokeWithoutArgs(this, &ProxyTest::CheckSignalingThread),
>+                      Return("Method0")));
>+  EXPECT_EQ("Method0", fake_proxy_->Method0());
> }
> 
> TEST_F(ProxyTest, ConstMethod0) {
>   EXPECT_CALL(*fake_, ConstMethod0())
>       .Times(Exactly(1))
>-      .WillOnce(
>-          DoAll(InvokeWithoutArgs(this, &ProxyTest::CheckSignalingThread),
>-                Return("ConstMethod0")));
>-  EXPECT_EQ("ConstMethod0",
>-            fake_proxy_->ConstMethod0());
>+      .WillOnce(DoAll(InvokeWithoutArgs(this, &ProxyTest::CheckSignalingThread),
>+                      Return("ConstMethod0")));
>+  EXPECT_EQ("ConstMethod0", fake_proxy_->ConstMethod0());
> }
> 
> TEST_F(ProxyTest, WorkerMethod1) {
>@@ -245,9 +241,8 @@ TEST_F(ProxyTest, ConstMethod1) {
>   const std::string arg1 = "arg1";
>   EXPECT_CALL(*fake_, ConstMethod1(arg1))
>       .Times(Exactly(1))
>-      .WillOnce(
>-          DoAll(InvokeWithoutArgs(this, &ProxyTest::CheckSignalingThread),
>-                Return("ConstMethod1")));
>+      .WillOnce(DoAll(InvokeWithoutArgs(this, &ProxyTest::CheckSignalingThread),
>+                      Return("ConstMethod1")));
>   EXPECT_EQ("ConstMethod1", fake_proxy_->ConstMethod1(arg1));
> }
> 
>@@ -275,8 +270,8 @@ class Foo : public FooInterface {
> };
> 
> BEGIN_OWNED_PROXY_MAP(Foo)
>-  PROXY_SIGNALING_THREAD_DESTRUCTOR()
>-  PROXY_METHOD0(void, Bar)
>+PROXY_SIGNALING_THREAD_DESTRUCTOR()
>+PROXY_METHOD0(void, Bar)
> END_PROXY_MAP()
> 
> class OwnedProxyTest : public testing::Test {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/remoteaudiosource.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/remoteaudiosource.cc
>index a260be89fd7..94fdb870d1d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/remoteaudiosource.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/remoteaudiosource.cc
>@@ -15,43 +15,41 @@
> #include <memory>
> #include <utility>
> 
>+#include "absl/memory/memory.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/thread.h"
> 
> namespace webrtc {
> 
>-class RemoteAudioSource::Sink : public AudioSinkInterface {
>+// This proxy is passed to the underlying media engine to receive audio data as
>+// they come in. The data will then be passed back up to the RemoteAudioSource
>+// which will fan it out to all the sinks that have been added to it.
>+class RemoteAudioSource::AudioDataProxy : public AudioSinkInterface {
>  public:
>-  explicit Sink(RemoteAudioSource* source) : source_(source) {}
>-  ~Sink() override { source_->OnAudioChannelGone(); }
>+  explicit AudioDataProxy(RemoteAudioSource* source) : source_(source) {
>+    RTC_DCHECK(source);
>+  }
>+  ~AudioDataProxy() override { source_->OnAudioChannelGone(); }
> 
>- private:
>+  // AudioSinkInterface implementation.
>   void OnData(const AudioSinkInterface::Data& audio) override {
>-    if (source_)
>-      source_->OnData(audio);
>+    source_->OnData(audio);
>   }
> 
>+ private:
>   const rtc::scoped_refptr<RemoteAudioSource> source_;
>-  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(Sink);
>-};
> 
>-rtc::scoped_refptr<RemoteAudioSource> RemoteAudioSource::Create(
>-    rtc::Thread* worker_thread,
>-    cricket::VoiceMediaChannel* media_channel,
>-    uint32_t ssrc) {
>-  rtc::scoped_refptr<RemoteAudioSource> ret(
>-      new rtc::RefCountedObject<RemoteAudioSource>());
>-  ret->Initialize(worker_thread, media_channel, ssrc);
>-  return ret;
>-}
>+  RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioDataProxy);
>+};
> 
>-RemoteAudioSource::RemoteAudioSource()
>+RemoteAudioSource::RemoteAudioSource(rtc::Thread* worker_thread)
>     : main_thread_(rtc::Thread::Current()),
>+      worker_thread_(worker_thread),
>       state_(MediaSourceInterface::kLive) {
>   RTC_DCHECK(main_thread_);
>+  RTC_DCHECK(worker_thread_);
> }
> 
> RemoteAudioSource::~RemoteAudioSource() {
>@@ -60,17 +58,25 @@ RemoteAudioSource::~RemoteAudioSource() {
>   RTC_DCHECK(sinks_.empty());
> }
> 
>-void RemoteAudioSource::Initialize(rtc::Thread* worker_thread,
>-                                   cricket::VoiceMediaChannel* media_channel,
>-                                   uint32_t ssrc) {
>-  RTC_DCHECK(main_thread_->IsCurrent());
>-  // To make sure we always get notified when the channel goes out of scope,
>-  // we register for callbacks here and not on demand in AddSink.
>-  if (media_channel) {  // May be null in tests.
>-    worker_thread->Invoke<void>(RTC_FROM_HERE, [&] {
>-      media_channel->SetRawAudioSink(ssrc, rtc::MakeUnique<Sink>(this));
>-    });
>-  }
>+void RemoteAudioSource::Start(cricket::VoiceMediaChannel* media_channel,
>+                              uint32_t ssrc) {
>+  RTC_DCHECK_RUN_ON(main_thread_);
>+  RTC_DCHECK(media_channel);
>+  // Register for callbacks immediately before AddSink so that we always get
>+  // notified when a channel goes out of scope (signaled when "AudioDataProxy"
>+  // is destroyed).
>+  worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
>+    media_channel->SetRawAudioSink(ssrc,
>+                                   absl::make_unique<AudioDataProxy>(this));
>+  });
>+}
>+
>+void RemoteAudioSource::Stop(cricket::VoiceMediaChannel* media_channel,
>+                             uint32_t ssrc) {
>+  RTC_DCHECK_RUN_ON(main_thread_);
>+  RTC_DCHECK(media_channel);
>+  worker_thread_->Invoke<void>(
>+      RTC_FROM_HERE, [&] { media_channel->SetRawAudioSink(ssrc, nullptr); });
> }
> 
> MediaSourceInterface::SourceState RemoteAudioSource::state() const {
>@@ -86,8 +92,9 @@ bool RemoteAudioSource::remote() const {
> void RemoteAudioSource::SetVolume(double volume) {
>   RTC_DCHECK_GE(volume, 0);
>   RTC_DCHECK_LE(volume, 10);
>-  for (auto* observer : audio_observers_)
>+  for (auto* observer : audio_observers_) {
>     observer->OnSetVolume(volume);
>+  }
> }
> 
> void RemoteAudioSource::RegisterAudioObserver(AudioObserver* observer) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/remoteaudiosource.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/remoteaudiosource.h
>index eadbccacc74..4d87331f91d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/remoteaudiosource.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/remoteaudiosource.h
>@@ -28,50 +28,46 @@ class Thread;
> namespace webrtc {
> 
> // This class implements the audio source used by the remote audio track.
>+// This class works by configuring itself as a sink with the underlying media
>+// engine, then when receiving data will fan out to all added sinks.
> class RemoteAudioSource : public Notifier<AudioSourceInterface>,
>                           rtc::MessageHandler {
>  public:
>-  // Creates an instance of RemoteAudioSource.
>-  static rtc::scoped_refptr<RemoteAudioSource> Create(
>-      rtc::Thread* worker_thread,
>-      cricket::VoiceMediaChannel* media_channel,
>-      uint32_t ssrc);
>+  explicit RemoteAudioSource(rtc::Thread* worker_thread);
>+
>+  // Register and unregister remote audio source with the underlying media
>+  // engine.
>+  void Start(cricket::VoiceMediaChannel* media_channel, uint32_t ssrc);
>+  void Stop(cricket::VoiceMediaChannel* media_channel, uint32_t ssrc);
> 
>   // MediaSourceInterface implementation.
>   MediaSourceInterface::SourceState state() const override;
>   bool remote() const override;
> 
>+  // AudioSourceInterface implementation.
>+  void SetVolume(double volume) override;
>+  void RegisterAudioObserver(AudioObserver* observer) override;
>+  void UnregisterAudioObserver(AudioObserver* observer) override;
>+
>   void AddSink(AudioTrackSinkInterface* sink) override;
>   void RemoveSink(AudioTrackSinkInterface* sink) override;
> 
>  protected:
>-  RemoteAudioSource();
>   ~RemoteAudioSource() override;
> 
>-  // Post construction initialize where we can do things like save a reference
>-  // to ourselves (need to be fully constructed).
>-  void Initialize(rtc::Thread* worker_thread,
>-                  cricket::VoiceMediaChannel* media_channel,
>-                  uint32_t ssrc);
>-
>  private:
>-  typedef std::list<AudioObserver*> AudioObserverList;
>-
>-  // AudioSourceInterface implementation.
>-  void SetVolume(double volume) override;
>-  void RegisterAudioObserver(AudioObserver* observer) override;
>-  void UnregisterAudioObserver(AudioObserver* observer) override;
>-
>-  class Sink;
>+  // These are callbacks from the media engine.
>+  class AudioDataProxy;
>   void OnData(const AudioSinkInterface::Data& audio);
>   void OnAudioChannelGone();
> 
>   void OnMessage(rtc::Message* msg) override;
> 
>-  AudioObserverList audio_observers_;
>+  rtc::Thread* const main_thread_;
>+  rtc::Thread* const worker_thread_;
>+  std::list<AudioObserver*> audio_observers_;
>   rtc::CriticalSection sink_lock_;
>   std::list<AudioTrackSinkInterface*> sinks_;
>-  rtc::Thread* const main_thread_;
>   SourceState state_;
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcpmuxfilter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcpmuxfilter.cc
>index 331c8df45cb..dceaf465f42 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcpmuxfilter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcpmuxfilter.cc
>@@ -14,8 +14,7 @@
> 
> namespace cricket {
> 
>-RtcpMuxFilter::RtcpMuxFilter() : state_(ST_INIT), offer_enable_(false) {
>-}
>+RtcpMuxFilter::RtcpMuxFilter() : state_(ST_INIT), offer_enable_(false) {}
> 
> bool RtcpMuxFilter::IsFullyActive() const {
>   return state_ == ST_ACTIVE;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcstats_integrationtest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcstats_integrationtest.cc
>index 158d54b41b7..355d1f7a8b3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcstats_integrationtest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcstats_integrationtest.cc
>@@ -8,6 +8,7 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>+#include <algorithm>
> #include <set>
> #include <vector>
> 
>@@ -17,6 +18,7 @@
> #include "api/peerconnectioninterface.h"
> #include "api/stats/rtcstats_objects.h"
> #include "api/stats/rtcstatsreport.h"
>+#include "pc/rtcstatstraversal.h"
> #include "pc/test/peerconnectiontestwrapper.h"
> #include "pc/test/rtcstatsobtainer.h"
> #include "rtc_base/checks.h"
>@@ -24,6 +26,7 @@
> #include "rtc_base/gunit.h"
> #include "rtc_base/refcountedobject.h"
> #include "rtc_base/scoped_ref_ptr.h"
>+#include "rtc_base/stringutils.h"
> #include "rtc_base/trace_event.h"
> #include "rtc_base/virtualsocketserver.h"
> 
>@@ -115,10 +118,10 @@ class RTCStatsIntegrationTest : public testing::Test {
>     PeerConnectionTestWrapper::Connect(caller_.get(), callee_.get());
> 
>     // Get user media for audio and video
>-    caller_->GetAndAddUserMedia(true, FakeConstraints(),
>-                                true, FakeConstraints());
>-    callee_->GetAndAddUserMedia(true, FakeConstraints(),
>-                                true, FakeConstraints());
>+    caller_->GetAndAddUserMedia(true, cricket::AudioOptions(), true,
>+                                FakeConstraints());
>+    callee_->GetAndAddUserMedia(true, cricket::AudioOptions(), true,
>+                                FakeConstraints());
> 
>     // Create data channels
>     DataChannelInit init;
>@@ -134,10 +137,26 @@ class RTCStatsIntegrationTest : public testing::Test {
>   rtc::scoped_refptr<const RTCStatsReport> GetStatsFromCaller() {
>     return GetStats(caller_->pc());
>   }
>+  rtc::scoped_refptr<const RTCStatsReport> GetStatsFromCaller(
>+      rtc::scoped_refptr<RtpSenderInterface> selector) {
>+    return GetStats(caller_->pc(), selector);
>+  }
>+  rtc::scoped_refptr<const RTCStatsReport> GetStatsFromCaller(
>+      rtc::scoped_refptr<RtpReceiverInterface> selector) {
>+    return GetStats(caller_->pc(), selector);
>+  }
> 
>   rtc::scoped_refptr<const RTCStatsReport> GetStatsFromCallee() {
>     return GetStats(callee_->pc());
>   }
>+  rtc::scoped_refptr<const RTCStatsReport> GetStatsFromCallee(
>+      rtc::scoped_refptr<RtpSenderInterface> selector) {
>+    return GetStats(callee_->pc(), selector);
>+  }
>+  rtc::scoped_refptr<const RTCStatsReport> GetStatsFromCallee(
>+      rtc::scoped_refptr<RtpReceiverInterface> selector) {
>+    return GetStats(callee_->pc(), selector);
>+  }
> 
>  protected:
>   static rtc::scoped_refptr<const RTCStatsReport> GetStats(
>@@ -149,6 +168,17 @@ class RTCStatsIntegrationTest : public testing::Test {
>     return stats_obtainer->report();
>   }
> 
>+  template <typename T>
>+  static rtc::scoped_refptr<const RTCStatsReport> GetStats(
>+      PeerConnectionInterface* pc,
>+      rtc::scoped_refptr<T> selector) {
>+    rtc::scoped_refptr<RTCStatsObtainer> stats_obtainer =
>+        RTCStatsObtainer::Create();
>+    pc->GetStats(selector, stats_obtainer);
>+    EXPECT_TRUE_WAIT(stats_obtainer->report(), kGetStatsTimeoutMs);
>+    return stats_obtainer->report();
>+  }
>+
>   // |network_thread_| uses |virtual_socket_server_| so they must be
>   // constructed/destructed in the correct order.
>   rtc::VirtualSocketServer virtual_socket_server_;
>@@ -169,67 +199,65 @@ class RTCStatsVerifier {
>     }
>   }
> 
>-  void MarkMemberTested(
>-      const RTCStatsMemberInterface& member, bool test_successful) {
>+  void MarkMemberTested(const RTCStatsMemberInterface& member,
>+                        bool test_successful) {
>     untested_members_.erase(&member);
>     all_tests_successful_ &= test_successful;
>   }
> 
>   void TestMemberIsDefined(const RTCStatsMemberInterface& member) {
>-    EXPECT_TRUE(member.is_defined()) <<
>-        stats_->type() << "." << member.name() << "[" << stats_->id() <<
>-        "] was undefined.";
>+    EXPECT_TRUE(member.is_defined())
>+        << stats_->type() << "." << member.name() << "[" << stats_->id()
>+        << "] was undefined.";
>     MarkMemberTested(member, member.is_defined());
>   }
> 
>   void TestMemberIsUndefined(const RTCStatsMemberInterface& member) {
>-    EXPECT_FALSE(member.is_defined()) <<
>-        stats_->type() << "." << member.name() << "[" << stats_->id() <<
>-        "] was defined (" << member.ValueToString() << ").";
>+    EXPECT_FALSE(member.is_defined())
>+        << stats_->type() << "." << member.name() << "[" << stats_->id()
>+        << "] was defined (" << member.ValueToString() << ").";
>     MarkMemberTested(member, !member.is_defined());
>   }
> 
>-  template<typename T>
>+  template <typename T>
>   void TestMemberIsPositive(const RTCStatsMemberInterface& member) {
>-    EXPECT_TRUE(member.is_defined()) <<
>-        stats_->type() << "." << member.name() << "[" << stats_->id() <<
>-        "] was undefined.";
>+    EXPECT_TRUE(member.is_defined())
>+        << stats_->type() << "." << member.name() << "[" << stats_->id()
>+        << "] was undefined.";
>     if (!member.is_defined()) {
>       MarkMemberTested(member, false);
>       return;
>     }
>     bool is_positive = *member.cast_to<RTCStatsMember<T>>() > T(0);
>-    EXPECT_TRUE(is_positive) <<
>-        stats_->type() << "." << member.name() << "[" << stats_->id() <<
>-        "] was not positive (" << member.ValueToString() << ").";
>+    EXPECT_TRUE(is_positive)
>+        << stats_->type() << "." << member.name() << "[" << stats_->id()
>+        << "] was not positive (" << member.ValueToString() << ").";
>     MarkMemberTested(member, is_positive);
>   }
> 
>-  template<typename T>
>+  template <typename T>
>   void TestMemberIsNonNegative(const RTCStatsMemberInterface& member) {
>-    EXPECT_TRUE(member.is_defined()) <<
>-        stats_->type() << "." << member.name() << "[" << stats_->id() <<
>-        "] was undefined.";
>+    EXPECT_TRUE(member.is_defined())
>+        << stats_->type() << "." << member.name() << "[" << stats_->id()
>+        << "] was undefined.";
>     if (!member.is_defined()) {
>       MarkMemberTested(member, false);
>       return;
>     }
>     bool is_non_negative = *member.cast_to<RTCStatsMember<T>>() >= T(0);
>-    EXPECT_TRUE(is_non_negative) <<
>-        stats_->type() << "." << member.name() << "[" << stats_->id() <<
>-        "] was not non-negative (" << member.ValueToString() << ").";
>+    EXPECT_TRUE(is_non_negative)
>+        << stats_->type() << "." << member.name() << "[" << stats_->id()
>+        << "] was not non-negative (" << member.ValueToString() << ").";
>     MarkMemberTested(member, is_non_negative);
>   }
> 
>-  void TestMemberIsIDReference(
>-      const RTCStatsMemberInterface& member,
>-      const char* expected_type) {
>+  void TestMemberIsIDReference(const RTCStatsMemberInterface& member,
>+                               const char* expected_type) {
>     TestMemberIsIDReference(member, expected_type, false);
>   }
> 
>-  void TestMemberIsOptionalIDReference(
>-      const RTCStatsMemberInterface& member,
>-      const char* expected_type) {
>+  void TestMemberIsOptionalIDReference(const RTCStatsMemberInterface& member,
>+                                       const char* expected_type) {
>     TestMemberIsIDReference(member, expected_type, true);
>   }
> 
>@@ -237,18 +265,16 @@ class RTCStatsVerifier {
>     if (untested_members_.empty())
>       return all_tests_successful_;
>     for (const RTCStatsMemberInterface* member : untested_members_) {
>-      EXPECT_TRUE(false) <<
>-          stats_->type() << "." << member->name() << "[" << stats_->id() <<
>-          "] was not tested.";
>+      EXPECT_TRUE(false) << stats_->type() << "." << member->name() << "["
>+                         << stats_->id() << "] was not tested.";
>     }
>     return false;
>   }
> 
>  private:
>-  void TestMemberIsIDReference(
>-      const RTCStatsMemberInterface& member,
>-      const char* expected_type,
>-      bool optional) {
>+  void TestMemberIsIDReference(const RTCStatsMemberInterface& member,
>+                               const char* expected_type,
>+                               bool optional) {
>     if (optional && !member.is_defined()) {
>       MarkMemberTested(member, true);
>       return;
>@@ -276,10 +302,11 @@ class RTCStatsVerifier {
>         }
>       }
>     }
>-    EXPECT_TRUE(valid_reference) <<
>-      stats_->type() << "." << member.name() << " is not a reference to an " <<
>-      "existing dictionary of type " << expected_type << " (" <<
>-      member.ValueToString() << ").";
>+    EXPECT_TRUE(valid_reference)
>+        << stats_->type() << "." << member.name()
>+        << " is not a reference to an "
>+        << "existing dictionary of type " << expected_type << " ("
>+        << member.ValueToString() << ").";
>     MarkMemberTested(member, valid_reference);
>   }
> 
>@@ -309,28 +336,27 @@ class RTCStatsReportVerifier {
>   }
> 
>   explicit RTCStatsReportVerifier(const RTCStatsReport* report)
>-      : report_(report) {
>-  }
>+      : report_(report) {}
> 
>-  void VerifyReport() {
>+  void VerifyReport(std::vector<const char*> allowed_missing_stats) {
>     std::set<const char*> missing_stats = StatsTypes();
>     bool verify_successful = true;
>     std::vector<const RTCTransportStats*> transport_stats =
>         report_->GetStatsOfType<RTCTransportStats>();
>-    EXPECT_EQ(transport_stats.size(), 1);
>+    EXPECT_EQ(transport_stats.size(), 1U);
>     std::string selected_candidate_pair_id =
>         *transport_stats[0]->selected_candidate_pair_id;
>     for (const RTCStats& stats : *report_) {
>       missing_stats.erase(stats.type());
>       if (stats.type() == RTCCertificateStats::kType) {
>-        verify_successful &= VerifyRTCCertificateStats(
>-            stats.cast_to<RTCCertificateStats>());
>+        verify_successful &=
>+            VerifyRTCCertificateStats(stats.cast_to<RTCCertificateStats>());
>       } else if (stats.type() == RTCCodecStats::kType) {
>-        verify_successful &= VerifyRTCCodecStats(
>-            stats.cast_to<RTCCodecStats>());
>+        verify_successful &=
>+            VerifyRTCCodecStats(stats.cast_to<RTCCodecStats>());
>       } else if (stats.type() == RTCDataChannelStats::kType) {
>-        verify_successful &= VerifyRTCDataChannelStats(
>-            stats.cast_to<RTCDataChannelStats>());
>+        verify_successful &=
>+            VerifyRTCDataChannelStats(stats.cast_to<RTCDataChannelStats>());
>       } else if (stats.type() == RTCIceCandidatePairStats::kType) {
>         verify_successful &= VerifyRTCIceCandidatePairStats(
>             stats.cast_to<RTCIceCandidatePairStats>(),
>@@ -342,8 +368,8 @@ class RTCStatsReportVerifier {
>         verify_successful &= VerifyRTCRemoteIceCandidateStats(
>             stats.cast_to<RTCRemoteIceCandidateStats>());
>       } else if (stats.type() == RTCMediaStreamStats::kType) {
>-        verify_successful &= VerifyRTCMediaStreamStats(
>-            stats.cast_to<RTCMediaStreamStats>());
>+        verify_successful &=
>+            VerifyRTCMediaStreamStats(stats.cast_to<RTCMediaStreamStats>());
>       } else if (stats.type() == RTCMediaStreamTrackStats::kType) {
>         verify_successful &= VerifyRTCMediaStreamTrackStats(
>             stats.cast_to<RTCMediaStreamTrackStats>());
>@@ -357,37 +383,36 @@ class RTCStatsReportVerifier {
>         verify_successful &= VerifyRTCOutboundRTPStreamStats(
>             stats.cast_to<RTCOutboundRTPStreamStats>());
>       } else if (stats.type() == RTCTransportStats::kType) {
>-        verify_successful &= VerifyRTCTransportStats(
>-            stats.cast_to<RTCTransportStats>());
>+        verify_successful &=
>+            VerifyRTCTransportStats(stats.cast_to<RTCTransportStats>());
>       } else {
>         EXPECT_TRUE(false) << "Unrecognized stats type: " << stats.type();
>         verify_successful = false;
>       }
>     }
>-    if (!missing_stats.empty()) {
>-      verify_successful = false;
>-      for (const char* missing : missing_stats) {
>+    for (const char* missing : missing_stats) {
>+      if (std::find(allowed_missing_stats.begin(), allowed_missing_stats.end(),
>+                    missing) == allowed_missing_stats.end()) {
>+        verify_successful = false;
>         EXPECT_TRUE(false) << "Missing expected stats type: " << missing;
>       }
>     }
>-    EXPECT_TRUE(verify_successful) <<
>-        "One or more problems with the stats. This is the report:\n" <<
>-        report_->ToJson();
>+    EXPECT_TRUE(verify_successful)
>+        << "One or more problems with the stats. This is the report:\n"
>+        << report_->ToJson();
>   }
> 
>-  bool VerifyRTCCertificateStats(
>-      const RTCCertificateStats& certificate) {
>+  bool VerifyRTCCertificateStats(const RTCCertificateStats& certificate) {
>     RTCStatsVerifier verifier(report_, &certificate);
>     verifier.TestMemberIsDefined(certificate.fingerprint);
>     verifier.TestMemberIsDefined(certificate.fingerprint_algorithm);
>     verifier.TestMemberIsDefined(certificate.base64_certificate);
>-    verifier.TestMemberIsOptionalIDReference(
>-        certificate.issuer_certificate_id, RTCCertificateStats::kType);
>+    verifier.TestMemberIsOptionalIDReference(certificate.issuer_certificate_id,
>+                                             RTCCertificateStats::kType);
>     return verifier.ExpectAllMembersSuccessfullyTested();
>   }
> 
>-  bool VerifyRTCCodecStats(
>-      const RTCCodecStats& codec) {
>+  bool VerifyRTCCodecStats(const RTCCodecStats& codec) {
>     RTCStatsVerifier verifier(report_, &codec);
>     verifier.TestMemberIsDefined(codec.payload_type);
>     verifier.TestMemberIsDefined(codec.mime_type);
>@@ -398,8 +423,7 @@ class RTCStatsReportVerifier {
>     return verifier.ExpectAllMembersSuccessfullyTested();
>   }
> 
>-  bool VerifyRTCDataChannelStats(
>-      const RTCDataChannelStats& data_channel) {
>+  bool VerifyRTCDataChannelStats(const RTCDataChannelStats& data_channel) {
>     RTCStatsVerifier verifier(report_, &data_channel);
>     verifier.TestMemberIsDefined(data_channel.label);
>     verifier.TestMemberIsDefined(data_channel.protocol);
>@@ -413,14 +437,15 @@ class RTCStatsReportVerifier {
>   }
> 
>   bool VerifyRTCIceCandidatePairStats(
>-      const RTCIceCandidatePairStats& candidate_pair, bool is_selected_pair) {
>+      const RTCIceCandidatePairStats& candidate_pair,
>+      bool is_selected_pair) {
>     RTCStatsVerifier verifier(report_, &candidate_pair);
>-    verifier.TestMemberIsIDReference(
>-        candidate_pair.transport_id, RTCTransportStats::kType);
>-    verifier.TestMemberIsIDReference(
>-        candidate_pair.local_candidate_id, RTCLocalIceCandidateStats::kType);
>-    verifier.TestMemberIsIDReference(
>-        candidate_pair.remote_candidate_id, RTCRemoteIceCandidateStats::kType);
>+    verifier.TestMemberIsIDReference(candidate_pair.transport_id,
>+                                     RTCTransportStats::kType);
>+    verifier.TestMemberIsIDReference(candidate_pair.local_candidate_id,
>+                                     RTCLocalIceCandidateStats::kType);
>+    verifier.TestMemberIsIDReference(candidate_pair.remote_candidate_id,
>+                                     RTCRemoteIceCandidateStats::kType);
>     verifier.TestMemberIsDefined(candidate_pair.state);
>     verifier.TestMemberIsNonNegative<uint64_t>(candidate_pair.priority);
>     verifier.TestMemberIsDefined(candidate_pair.nominated);
>@@ -457,11 +482,10 @@ class RTCStatsReportVerifier {
>     return verifier.ExpectAllMembersSuccessfullyTested();
>   }
> 
>-  bool VerifyRTCIceCandidateStats(
>-      const RTCIceCandidateStats& candidate) {
>+  bool VerifyRTCIceCandidateStats(const RTCIceCandidateStats& candidate) {
>     RTCStatsVerifier verifier(report_, &candidate);
>-    verifier.TestMemberIsIDReference(
>-        candidate.transport_id, RTCTransportStats::kType);
>+    verifier.TestMemberIsIDReference(candidate.transport_id,
>+                                     RTCTransportStats::kType);
>     verifier.TestMemberIsDefined(candidate.is_remote);
>     if (*candidate.is_remote) {
>       verifier.TestMemberIsUndefined(candidate.network_type);
>@@ -488,12 +512,11 @@ class RTCStatsReportVerifier {
>     return VerifyRTCIceCandidateStats(remote_candidate);
>   }
> 
>-  bool VerifyRTCMediaStreamStats(
>-      const RTCMediaStreamStats& media_stream) {
>+  bool VerifyRTCMediaStreamStats(const RTCMediaStreamStats& media_stream) {
>     RTCStatsVerifier verifier(report_, &media_stream);
>     verifier.TestMemberIsDefined(media_stream.stream_identifier);
>-    verifier.TestMemberIsIDReference(
>-        media_stream.track_ids, RTCMediaStreamTrackStats::kType);
>+    verifier.TestMemberIsIDReference(media_stream.track_ids,
>+                                     RTCMediaStreamTrackStats::kType);
>     return verifier.ExpectAllMembersSuccessfullyTested();
>   }
> 
>@@ -515,6 +538,7 @@ class RTCStatsReportVerifier {
>       verifier.TestMemberIsUndefined(media_stream_track.frames_per_second);
>       if (*media_stream_track.remote_source) {
>         verifier.TestMemberIsUndefined(media_stream_track.frames_sent);
>+        verifier.TestMemberIsUndefined(media_stream_track.huge_frames_sent);
>         verifier.TestMemberIsNonNegative<uint32_t>(
>             media_stream_track.frames_received);
>         verifier.TestMemberIsNonNegative<uint32_t>(
>@@ -524,6 +548,8 @@ class RTCStatsReportVerifier {
>       } else {
>         verifier.TestMemberIsNonNegative<uint32_t>(
>             media_stream_track.frames_sent);
>+        verifier.TestMemberIsNonNegative<uint32_t>(
>+            media_stream_track.huge_frames_sent);
>         verifier.TestMemberIsUndefined(media_stream_track.frames_received);
>         verifier.TestMemberIsUndefined(media_stream_track.frames_decoded);
>         verifier.TestMemberIsUndefined(media_stream_track.frames_dropped);
>@@ -539,13 +565,13 @@ class RTCStatsReportVerifier {
>       verifier.TestMemberIsUndefined(media_stream_track.total_audio_energy);
>       verifier.TestMemberIsUndefined(media_stream_track.total_samples_duration);
>     } else {
>-      RTC_DCHECK_EQ(*media_stream_track.kind,
>-                    RTCMediaStreamTrackKind::kAudio);
>+      RTC_DCHECK_EQ(*media_stream_track.kind, RTCMediaStreamTrackKind::kAudio);
>       // Video-only members should be undefined
>       verifier.TestMemberIsUndefined(media_stream_track.frame_width);
>       verifier.TestMemberIsUndefined(media_stream_track.frame_height);
>       verifier.TestMemberIsUndefined(media_stream_track.frames_per_second);
>       verifier.TestMemberIsUndefined(media_stream_track.frames_sent);
>+      verifier.TestMemberIsUndefined(media_stream_track.huge_frames_sent);
>       verifier.TestMemberIsUndefined(media_stream_track.frames_received);
>       verifier.TestMemberIsUndefined(media_stream_track.frames_decoded);
>       verifier.TestMemberIsUndefined(media_stream_track.frames_dropped);
>@@ -563,8 +589,8 @@ class RTCStatsReportVerifier {
>       // test run until available or is there a way to have it always be
>       // defined? crbug.com/627816
>       verifier.MarkMemberTested(media_stream_track.echo_return_loss, true);
>-      verifier.MarkMemberTested(
>-          media_stream_track.echo_return_loss_enhancement, true);
>+      verifier.MarkMemberTested(media_stream_track.echo_return_loss_enhancement,
>+                                true);
>     }
>     // totalSamplesReceived, concealedSamples and concealmentEvents are only
>     // present on inbound audio tracks.
>@@ -598,16 +624,16 @@ class RTCStatsReportVerifier {
>     return verifier.ExpectAllMembersSuccessfullyTested();
>   }
> 
>-  void VerifyRTCRTPStreamStats(
>-      const RTCRTPStreamStats& stream, RTCStatsVerifier* verifier) {
>+  void VerifyRTCRTPStreamStats(const RTCRTPStreamStats& stream,
>+                               RTCStatsVerifier* verifier) {
>     verifier->TestMemberIsDefined(stream.ssrc);
>     verifier->TestMemberIsUndefined(stream.associate_stats_id);
>     verifier->TestMemberIsDefined(stream.is_remote);
>     verifier->TestMemberIsDefined(stream.media_type);
>-    verifier->TestMemberIsIDReference(
>-        stream.track_id, RTCMediaStreamTrackStats::kType);
>-    verifier->TestMemberIsIDReference(
>-        stream.transport_id, RTCTransportStats::kType);
>+    verifier->TestMemberIsIDReference(stream.track_id,
>+                                      RTCMediaStreamTrackStats::kType);
>+    verifier->TestMemberIsIDReference(stream.transport_id,
>+                                      RTCTransportStats::kType);
>     verifier->TestMemberIsIDReference(stream.codec_id, RTCCodecStats::kType);
>     if (stream.media_type.is_defined() && *stream.media_type == "video") {
>       verifier->TestMemberIsNonNegative<uint32_t>(stream.fir_count);
>@@ -685,20 +711,19 @@ class RTCStatsReportVerifier {
>     return verifier.ExpectAllMembersSuccessfullyTested();
>   }
> 
>-  bool VerifyRTCTransportStats(
>-      const RTCTransportStats& transport) {
>+  bool VerifyRTCTransportStats(const RTCTransportStats& transport) {
>     RTCStatsVerifier verifier(report_, &transport);
>     verifier.TestMemberIsNonNegative<uint64_t>(transport.bytes_sent);
>     verifier.TestMemberIsNonNegative<uint64_t>(transport.bytes_received);
>-    verifier.TestMemberIsOptionalIDReference(
>-        transport.rtcp_transport_stats_id, RTCTransportStats::kType);
>+    verifier.TestMemberIsOptionalIDReference(transport.rtcp_transport_stats_id,
>+                                             RTCTransportStats::kType);
>     verifier.TestMemberIsDefined(transport.dtls_state);
>-    verifier.TestMemberIsIDReference(
>-        transport.selected_candidate_pair_id, RTCIceCandidatePairStats::kType);
>-    verifier.TestMemberIsIDReference(
>-        transport.local_certificate_id, RTCCertificateStats::kType);
>-    verifier.TestMemberIsIDReference(
>-        transport.remote_certificate_id, RTCCertificateStats::kType);
>+    verifier.TestMemberIsIDReference(transport.selected_candidate_pair_id,
>+                                     RTCIceCandidatePairStats::kType);
>+    verifier.TestMemberIsIDReference(transport.local_certificate_id,
>+                                     RTCCertificateStats::kType);
>+    verifier.TestMemberIsIDReference(transport.remote_certificate_id,
>+                                     RTCCertificateStats::kType);
>     return verifier.ExpectAllMembersSuccessfullyTested();
>   }
> 
>@@ -711,7 +736,7 @@ TEST_F(RTCStatsIntegrationTest, GetStatsFromCaller) {
>   StartCall();
> 
>   rtc::scoped_refptr<const RTCStatsReport> report = GetStatsFromCaller();
>-  RTCStatsReportVerifier(report.get()).VerifyReport();
>+  RTCStatsReportVerifier(report.get()).VerifyReport({});
>   EXPECT_EQ(report->ToJson(), RTCStatsReportTraceListener::last_trace());
> }
> 
>@@ -719,10 +744,68 @@ TEST_F(RTCStatsIntegrationTest, GetStatsFromCallee) {
>   StartCall();
> 
>   rtc::scoped_refptr<const RTCStatsReport> report = GetStatsFromCallee();
>-  RTCStatsReportVerifier(report.get()).VerifyReport();
>+  RTCStatsReportVerifier(report.get()).VerifyReport({});
>   EXPECT_EQ(report->ToJson(), RTCStatsReportTraceListener::last_trace());
> }
> 
>+// These tests exercise the integration of the stats selection algorithm inside
>+// of PeerConnection. See rtcstatstraveral_unittest.cc for more detailed stats
>+// traversal tests on particular stats graphs.
>+TEST_F(RTCStatsIntegrationTest, GetStatsWithSenderSelector) {
>+  StartCall();
>+  ASSERT_FALSE(caller_->pc()->GetSenders().empty());
>+  rtc::scoped_refptr<const RTCStatsReport> report =
>+      GetStatsFromCaller(caller_->pc()->GetSenders()[0]);
>+  std::vector<const char*> allowed_missing_stats = {
>+      // TODO(hbos): Include RTC[Audio/Video]ReceiverStats when implemented.
>+      // TODO(hbos): Include RTCRemoteOutboundRtpStreamStats when implemented.
>+      // TODO(hbos): Include RTCRtpContributingSourceStats when implemented.
>+      RTCInboundRTPStreamStats::kType, RTCPeerConnectionStats::kType,
>+      RTCMediaStreamStats::kType, RTCDataChannelStats::kType,
>+  };
>+  RTCStatsReportVerifier(report.get()).VerifyReport(allowed_missing_stats);
>+  EXPECT_TRUE(report->size());
>+}
>+
>+TEST_F(RTCStatsIntegrationTest, GetStatsWithReceiverSelector) {
>+  StartCall();
>+
>+  ASSERT_FALSE(caller_->pc()->GetReceivers().empty());
>+  rtc::scoped_refptr<const RTCStatsReport> report =
>+      GetStatsFromCaller(caller_->pc()->GetReceivers()[0]);
>+  std::vector<const char*> allowed_missing_stats = {
>+      // TODO(hbos): Include RTC[Audio/Video]SenderStats when implemented.
>+      // TODO(hbos): Include RTCRemoteInboundRtpStreamStats when implemented.
>+      // TODO(hbos): Include RTCRtpContributingSourceStats when implemented.
>+      RTCOutboundRTPStreamStats::kType, RTCPeerConnectionStats::kType,
>+      RTCMediaStreamStats::kType, RTCDataChannelStats::kType,
>+  };
>+  RTCStatsReportVerifier(report.get()).VerifyReport(allowed_missing_stats);
>+  EXPECT_TRUE(report->size());
>+}
>+
>+TEST_F(RTCStatsIntegrationTest, GetStatsWithInvalidSenderSelector) {
>+  StartCall();
>+
>+  ASSERT_FALSE(callee_->pc()->GetSenders().empty());
>+  // The selector is invalid for the caller because it belongs to the callee.
>+  auto invalid_selector = callee_->pc()->GetSenders()[0];
>+  rtc::scoped_refptr<const RTCStatsReport> report =
>+      GetStatsFromCaller(invalid_selector);
>+  EXPECT_FALSE(report->size());
>+}
>+
>+TEST_F(RTCStatsIntegrationTest, GetStatsWithInvalidReceiverSelector) {
>+  StartCall();
>+
>+  ASSERT_FALSE(callee_->pc()->GetReceivers().empty());
>+  // The selector is invalid for the caller because it belongs to the callee.
>+  auto invalid_selector = callee_->pc()->GetReceivers()[0];
>+  rtc::scoped_refptr<const RTCStatsReport> report =
>+      GetStatsFromCaller(invalid_selector);
>+  EXPECT_FALSE(report->size());
>+}
>+
> TEST_F(RTCStatsIntegrationTest, GetsStatsWhileDestroyingPeerConnections) {
>   StartCall();
> 
>@@ -733,10 +816,69 @@ TEST_F(RTCStatsIntegrationTest, GetsStatsWhileDestroyingPeerConnections) {
>   caller_ = nullptr;
>   // Any pending stats requests should have completed in the act of destroying
>   // the peer connection.
>-  EXPECT_TRUE(stats_obtainer->report());
>+  ASSERT_TRUE(stats_obtainer->report());
>+  EXPECT_EQ(stats_obtainer->report()->ToJson(),
>+            RTCStatsReportTraceListener::last_trace());
>+}
>+
>+TEST_F(RTCStatsIntegrationTest, GetsStatsWhileClosingPeerConnection) {
>+  StartCall();
>+
>+  rtc::scoped_refptr<RTCStatsObtainer> stats_obtainer =
>+      RTCStatsObtainer::Create();
>+  caller_->pc()->GetStats(stats_obtainer);
>+  caller_->pc()->Close();
>+
>+  ASSERT_TRUE(stats_obtainer->report());
>   EXPECT_EQ(stats_obtainer->report()->ToJson(),
>             RTCStatsReportTraceListener::last_trace());
> }
>+
>+// GetStatsReferencedIds() is optimized to recognize what is or isn't a
>+// referenced ID based on dictionary type information and knowing what members
>+// are used as references, as opposed to iterating all members to find the ones
>+// with the "Id" or "Ids" suffix. As such, GetStatsReferencedIds() is tested as
>+// an integration test instead of a unit test in order to guard against adding
>+// new references and forgetting to update GetStatsReferencedIds().
>+TEST_F(RTCStatsIntegrationTest, GetStatsReferencedIds) {
>+  StartCall();
>+
>+  rtc::scoped_refptr<const RTCStatsReport> report = GetStatsFromCallee();
>+  for (const RTCStats& stats : *report) {
>+    // Find all references by looking at all string members with the "Id" or
>+    // "Ids" suffix.
>+    std::set<const std::string*> expected_ids;
>+    for (const auto* member : stats.Members()) {
>+      if (!member->is_defined())
>+        continue;
>+      if (member->type() == RTCStatsMemberInterface::kString) {
>+        if (rtc::ends_with(member->name(), "Id")) {
>+          const auto& id = member->cast_to<const RTCStatsMember<std::string>>();
>+          expected_ids.insert(&(*id));
>+        }
>+      } else if (member->type() == RTCStatsMemberInterface::kSequenceString) {
>+        if (rtc::ends_with(member->name(), "Ids")) {
>+          const auto& ids =
>+              member->cast_to<const RTCStatsMember<std::vector<std::string>>>();
>+          for (const std::string& id : *ids)
>+            expected_ids.insert(&id);
>+        }
>+      }
>+    }
>+
>+    std::vector<const std::string*> neighbor_ids = GetStatsReferencedIds(stats);
>+    EXPECT_EQ(neighbor_ids.size(), expected_ids.size());
>+    for (const std::string* neighbor_id : neighbor_ids) {
>+      EXPECT_TRUE(expected_ids.find(neighbor_id) != expected_ids.end())
>+          << "Unexpected neighbor ID: " << *neighbor_id;
>+    }
>+    for (const std::string* expected_id : expected_ids) {
>+      EXPECT_TRUE(std::find(neighbor_ids.begin(), neighbor_ids.end(),
>+                            expected_id) != neighbor_ids.end())
>+          << "Missing expected neighbor ID: " << *expected_id;
>+    }
>+  }
>+}
> #endif  // HAVE_SCTP
> 
> }  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcstatscollector.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcstatscollector.cc
>index 81476435ed1..c7555fd97ee 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcstatscollector.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcstatscollector.cc
>@@ -11,11 +11,11 @@
> #include "pc/rtcstatscollector.h"
> 
> #include <memory>
>-#include <sstream>
> #include <string>
> #include <utility>
> #include <vector>
> 
>+#include "absl/memory/memory.h"
> #include "api/candidate.h"
> #include "api/mediastreaminterface.h"
> #include "api/peerconnectioninterface.h"
>@@ -23,8 +23,9 @@
> #include "p2p/base/p2pconstants.h"
> #include "p2p/base/port.h"
> #include "pc/peerconnection.h"
>+#include "pc/rtcstatstraversal.h"
> #include "rtc_base/checks.h"
>-#include "rtc_base/stringutils.h"
>+#include "rtc_base/strings/string_builder.h"
> #include "rtc_base/timeutils.h"
> #include "rtc_base/trace_event.h"
> 
>@@ -36,25 +37,23 @@ std::string RTCCertificateIDFromFingerprint(const std::string& fingerprint) {
>   return "RTCCertificate_" + fingerprint;
> }
> 
>-std::string RTCCodecStatsIDFromDirectionMediaAndPayload(
>-    bool inbound, bool audio, uint32_t payload_type) {
>-  // TODO(hbos): The present codec ID assignment is not sufficient to support
>-  // Unified Plan or unbundled connections in all cases. When we are able to
>-  // handle multiple m= lines of the same media type (and multiple BaseChannels
>-  // for the same type is possible?) this needs to be updated to differentiate
>-  // the transport being used, and stats need to be collected for all of them.
>-  if (inbound) {
>-    return audio ? "RTCCodec_InboundAudio_" + rtc::ToString<>(payload_type)
>-                 : "RTCCodec_InboundVideo_" + rtc::ToString<>(payload_type);
>-  }
>-  return audio ? "RTCCodec_OutboundAudio_" + rtc::ToString<>(payload_type)
>-               : "RTCCodec_OutboundVideo_" + rtc::ToString<>(payload_type);
>+std::string RTCCodecStatsIDFromMidDirectionAndPayload(const std::string& mid,
>+                                                      bool inbound,
>+                                                      uint32_t payload_type) {
>+  char buf[1024];
>+  rtc::SimpleStringBuilder sb(buf);
>+  sb << "RTCCodec_" << mid << (inbound ? "_Inbound_" : "_Outbound_")
>+     << payload_type;
>+  return sb.str();
> }
> 
> std::string RTCIceCandidatePairStatsIDFromConnectionInfo(
>     const cricket::ConnectionInfo& info) {
>-  return "RTCIceCandidatePair_" + info.local_candidate.id() + "_" +
>-      info.remote_candidate.id();
>+  char buf[4096];
>+  rtc::SimpleStringBuilder sb(buf);
>+  sb << "RTCIceCandidatePair_" << info.local_candidate.id() << "_"
>+     << info.remote_candidate.id();
>+  return sb.str();
> }
> 
> const char kSender[] = "sender";
>@@ -63,25 +62,32 @@ const char kReceiver[] = "receiver";
> std::string RTCMediaStreamTrackStatsIDFromDirectionAndAttachment(
>     const char* direction,
>     int attachment_id) {
>-  std::ostringstream oss;
>-  oss << "RTCMediaStreamTrack_" << direction << "_" << attachment_id;
>-  return oss.str();
>+  char buf[1024];
>+  rtc::SimpleStringBuilder sb(buf);
>+  sb << "RTCMediaStreamTrack_" << direction << "_" << attachment_id;
>+  return sb.str();
> }
> 
> std::string RTCTransportStatsIDFromTransportChannel(
>     const std::string& transport_name, int channel_component) {
>-  return "RTCTransport_" + transport_name + "_" +
>-      rtc::ToString<>(channel_component);
>+  char buf[1024];
>+  rtc::SimpleStringBuilder sb(buf);
>+  sb << "RTCTransport_" << transport_name << "_" << channel_component;
>+  return sb.str();
> }
> 
> std::string RTCInboundRTPStreamStatsIDFromSSRC(bool audio, uint32_t ssrc) {
>-  return audio ? "RTCInboundRTPAudioStream_" + rtc::ToString<>(ssrc)
>-               : "RTCInboundRTPVideoStream_" + rtc::ToString<>(ssrc);
>+  char buf[1024];
>+  rtc::SimpleStringBuilder sb(buf);
>+  sb << "RTCInboundRTP" << (audio ? "Audio" : "Video") << "Stream_" << ssrc;
>+  return sb.str();
> }
> 
> std::string RTCOutboundRTPStreamStatsIDFromSSRC(bool audio, uint32_t ssrc) {
>-  return audio ? "RTCOutboundRTPAudioStream_" + rtc::ToString<>(ssrc)
>-               : "RTCOutboundRTPVideoStream_" + rtc::ToString<>(ssrc);
>+  char buf[1024];
>+  rtc::SimpleStringBuilder sb(buf);
>+  sb << "RTCOutboundRTP" << (audio ? "Audio" : "Video") << "Stream_" << ssrc;
>+  return sb.str();
> }
> 
> const char* CandidateTypeToRTCIceCandidateType(const std::string& type) {
>@@ -162,6 +168,7 @@ const char* NetworkAdapterTypeToStatsType(rtc::AdapterType type) {
>       return RTCNetworkType::kVpn;
>     case rtc::ADAPTER_TYPE_UNKNOWN:
>     case rtc::ADAPTER_TYPE_LOOPBACK:
>+    case rtc::ADAPTER_TYPE_ANY:
>       return RTCNetworkType::kUnknown;
>   }
>   RTC_NOTREACHED();
>@@ -175,14 +182,16 @@ double DoubleAudioLevelFromIntAudioLevel(int audio_level) {
> }
> 
> std::unique_ptr<RTCCodecStats> CodecStatsFromRtpCodecParameters(
>-    uint64_t timestamp_us, bool inbound, bool audio,
>+    uint64_t timestamp_us,
>+    const std::string& mid,
>+    bool inbound,
>     const RtpCodecParameters& codec_params) {
>   RTC_DCHECK_GE(codec_params.payload_type, 0);
>   RTC_DCHECK_LE(codec_params.payload_type, 127);
>   RTC_DCHECK(codec_params.clock_rate);
>   uint32_t payload_type = static_cast<uint32_t>(codec_params.payload_type);
>   std::unique_ptr<RTCCodecStats> codec_stats(new RTCCodecStats(
>-      RTCCodecStatsIDFromDirectionMediaAndPayload(inbound, audio, payload_type),
>+      RTCCodecStatsIDFromMidDirectionAndPayload(mid, inbound, payload_type),
>       timestamp_us));
>   codec_stats->payload_type = payload_type;
>   codec_stats->mime_type = codec_params.mime_type();
>@@ -218,15 +227,15 @@ void SetInboundRTPStreamStatsFromMediaReceiverInfo(
> }
> 
> void SetInboundRTPStreamStatsFromVoiceReceiverInfo(
>+    const std::string& mid,
>     const cricket::VoiceReceiverInfo& voice_receiver_info,
>     RTCInboundRTPStreamStats* inbound_audio) {
>   SetInboundRTPStreamStatsFromMediaReceiverInfo(
>       voice_receiver_info, inbound_audio);
>   inbound_audio->media_type = "audio";
>   if (voice_receiver_info.codec_payload_type) {
>-    inbound_audio->codec_id =
>-        RTCCodecStatsIDFromDirectionMediaAndPayload(
>-            true, true, *voice_receiver_info.codec_payload_type);
>+    inbound_audio->codec_id = RTCCodecStatsIDFromMidDirectionAndPayload(
>+        mid, true, *voice_receiver_info.codec_payload_type);
>   }
>   inbound_audio->jitter =
>       static_cast<double>(voice_receiver_info.jitter_ms) /
>@@ -236,15 +245,15 @@ void SetInboundRTPStreamStatsFromVoiceReceiverInfo(
> }
> 
> void SetInboundRTPStreamStatsFromVideoReceiverInfo(
>+    const std::string& mid,
>     const cricket::VideoReceiverInfo& video_receiver_info,
>     RTCInboundRTPStreamStats* inbound_video) {
>   SetInboundRTPStreamStatsFromMediaReceiverInfo(
>       video_receiver_info, inbound_video);
>   inbound_video->media_type = "video";
>   if (video_receiver_info.codec_payload_type) {
>-    inbound_video->codec_id =
>-        RTCCodecStatsIDFromDirectionMediaAndPayload(
>-            true, false, *video_receiver_info.codec_payload_type);
>+    inbound_video->codec_id = RTCCodecStatsIDFromMidDirectionAndPayload(
>+        mid, true, *video_receiver_info.codec_payload_type);
>   }
>   inbound_video->fir_count =
>       static_cast<uint32_t>(video_receiver_info.firs_sent);
>@@ -272,30 +281,30 @@ void SetOutboundRTPStreamStatsFromMediaSenderInfo(
> }
> 
> void SetOutboundRTPStreamStatsFromVoiceSenderInfo(
>+    const std::string& mid,
>     const cricket::VoiceSenderInfo& voice_sender_info,
>     RTCOutboundRTPStreamStats* outbound_audio) {
>   SetOutboundRTPStreamStatsFromMediaSenderInfo(
>       voice_sender_info, outbound_audio);
>   outbound_audio->media_type = "audio";
>   if (voice_sender_info.codec_payload_type) {
>-    outbound_audio->codec_id =
>-        RTCCodecStatsIDFromDirectionMediaAndPayload(
>-            false, true, *voice_sender_info.codec_payload_type);
>+    outbound_audio->codec_id = RTCCodecStatsIDFromMidDirectionAndPayload(
>+        mid, false, *voice_sender_info.codec_payload_type);
>   }
>   // |fir_count|, |pli_count| and |sli_count| are only valid for video and are
>   // purposefully left undefined for audio.
> }
> 
> void SetOutboundRTPStreamStatsFromVideoSenderInfo(
>+    const std::string& mid,
>     const cricket::VideoSenderInfo& video_sender_info,
>     RTCOutboundRTPStreamStats* outbound_video) {
>   SetOutboundRTPStreamStatsFromMediaSenderInfo(
>       video_sender_info, outbound_video);
>   outbound_video->media_type = "video";
>   if (video_sender_info.codec_payload_type) {
>-    outbound_video->codec_id =
>-        RTCCodecStatsIDFromDirectionMediaAndPayload(
>-            false, false, *video_sender_info.codec_payload_type);
>+    outbound_video->codec_id = RTCCodecStatsIDFromMidDirectionAndPayload(
>+        mid, false, *video_sender_info.codec_payload_type);
>   }
>   outbound_video->fir_count =
>       static_cast<uint32_t>(video_sender_info.firs_rcvd);
>@@ -460,6 +469,7 @@ ProduceMediaStreamTrackStatsFromVideoSenderInfo(
>   // TODO(hbos): Will reduce this by frames dropped due to congestion control
>   // when available. https://crbug.com/659137
>   video_track_stats->frames_sent = video_sender_info.frames_encoded;
>+  video_track_stats->huge_frames_sent = video_sender_info.huge_frames_sent;
>   return video_track_stats;
> }
> 
>@@ -502,18 +512,14 @@ ProduceMediaStreamTrackStatsFromVideoReceiverInfo(
> void ProduceSenderMediaTrackStats(
>     int64_t timestamp_us,
>     const TrackMediaInfoMap& track_media_info_map,
>-    std::vector<rtc::scoped_refptr<RtpSenderInterface>> senders,
>+    std::vector<rtc::scoped_refptr<RtpSenderInternal>> senders,
>     RTCStatsReport* report) {
>   // This function iterates over the senders to generate outgoing track stats.
> 
>   // TODO(hbos): Return stats of detached tracks. We have to perform stats
>   // gathering at the time of detachment to get accurate stats and timestamps.
>   // https://crbug.com/659137
>-  for (auto const sender : senders) {
>-    // Don't report on tracks before starting to send.
>-    // https://bugs.webrtc.org/8673
>-    if (sender->ssrc() == 0)
>-      continue;
>+  for (auto sender : senders) {
>     if (sender->media_type() == cricket::MEDIA_TYPE_AUDIO) {
>       AudioTrackInterface* track =
>           static_cast<AudioTrackInterface*>(sender->track().get());
>@@ -524,11 +530,11 @@ void ProduceSenderMediaTrackStats(
>       // TODO(hta): Checking on ssrc is not proper. There should be a way
>       // to see from a sender whether it's connected or not.
>       // Related to https://crbug.com/8694 (using ssrc 0 to indicate "none")
>-      if (sender->ssrc()) {
>+      if (sender->ssrc() != 0) {
>         // When pc.close is called, sender info is discarded, so
>         // we generate zeroes instead. Bug: It should be retained.
>         // https://crbug.com/807174
>-        auto sender_info =
>+        const cricket::VoiceSenderInfo* sender_info =
>             track_media_info_map.GetVoiceSenderInfoBySsrc(sender->ssrc());
>         if (sender_info) {
>           voice_sender_info = sender_info;
>@@ -552,11 +558,11 @@ void ProduceSenderMediaTrackStats(
>       // TODO(hta): Check on state not ssrc when state is available
>       // Related to https://bugs.webrtc.org/8694 (using ssrc 0 to indicate
>       // "none")
>-      if (sender->ssrc()) {
>+      if (sender->ssrc() != 0) {
>         // When pc.close is called, sender info is discarded, so
>         // we generate zeroes instead. Bug: It should be retained.
>         // https://crbug.com/807174
>-        auto sender_info =
>+        const cricket::VideoSenderInfo* sender_info =
>             track_media_info_map.GetVideoSenderInfoBySsrc(sender->ssrc());
>         if (sender_info) {
>           video_sender_info = sender_info;
>@@ -578,10 +584,10 @@ void ProduceSenderMediaTrackStats(
> void ProduceReceiverMediaTrackStats(
>     int64_t timestamp_us,
>     const TrackMediaInfoMap& track_media_info_map,
>-    std::vector<rtc::scoped_refptr<RtpReceiverInterface>> receivers,
>+    std::vector<rtc::scoped_refptr<RtpReceiverInternal>> receivers,
>     RTCStatsReport* report) {
>   // This function iterates over the receivers to find the remote tracks.
>-  for (auto const receiver : receivers) {
>+  for (auto receiver : receivers) {
>     if (receiver->media_type() == cricket::MEDIA_TYPE_AUDIO) {
>       AudioTrackInterface* track =
>           static_cast<AudioTrackInterface*>(receiver->track().get());
>@@ -614,47 +620,101 @@ void ProduceReceiverMediaTrackStats(
>   }
> }
> 
>-void ProduceMediaStreamStats(
>-    int64_t timestamp_us,
>-    const std::vector<rtc::scoped_refptr<RtpSenderInterface>>& rtp_senders,
>-    const std::vector<rtc::scoped_refptr<RtpReceiverInterface>>& rtp_receivers,
>-    RTCStatsReport* report) {
>-  std::map<std::string, std::vector<std::string>> track_ids;
>-
>-  for (auto& sender : rtp_senders) {
>-    std::string track_id = RTCMediaStreamTrackStatsIDFromDirectionAndAttachment(
>-        kSender, sender->AttachmentId());
>-    for (auto& stream_id : sender->stream_ids()) {
>-      track_ids[stream_id].push_back(track_id);
>+rtc::scoped_refptr<RTCStatsReport> CreateReportFilteredBySelector(
>+    bool filter_by_sender_selector,
>+    rtc::scoped_refptr<const RTCStatsReport> report,
>+    rtc::scoped_refptr<RtpSenderInternal> sender_selector,
>+    rtc::scoped_refptr<RtpReceiverInternal> receiver_selector) {
>+  std::vector<std::string> rtpstream_ids;
>+  if (filter_by_sender_selector) {
>+    // Filter mode: RTCStatsCollector::RequestInfo::kSenderSelector
>+    if (sender_selector) {
>+      // Find outbound-rtp(s) of the sender, i.e. the outbound-rtp(s) that
>+      // reference the sender stats.
>+      // Because we do not implement sender stats, we look at outbound-rtp(s)
>+      // that reference the track attachment stats for the sender instead.
>+      std::string track_id =
>+          RTCMediaStreamTrackStatsIDFromDirectionAndAttachment(
>+              kSender, sender_selector->AttachmentId());
>+      for (const auto& stats : *report) {
>+        if (stats.type() != RTCOutboundRTPStreamStats::kType)
>+          continue;
>+        const auto& outbound_rtp = stats.cast_to<RTCOutboundRTPStreamStats>();
>+        if (outbound_rtp.track_id.is_defined() &&
>+            *outbound_rtp.track_id == track_id) {
>+          rtpstream_ids.push_back(outbound_rtp.id());
>+        }
>+      }
>     }
>-  }
>-  for (auto& receiver : rtp_receivers) {
>-    std::string track_id = RTCMediaStreamTrackStatsIDFromDirectionAndAttachment(
>-        kReceiver, receiver->AttachmentId());
>-    for (auto& stream : receiver->streams()) {
>-      track_ids[stream->label()].push_back(track_id);
>+  } else {
>+    // Filter mode: RTCStatsCollector::RequestInfo::kReceiverSelector
>+    if (receiver_selector) {
>+      // Find inbound-rtp(s) of the receiver, i.e. the inbound-rtp(s) that
>+      // reference the receiver stats.
>+      // Because we do not implement receiver stats, we look at inbound-rtp(s)
>+      // that reference the track attachment stats for the receiver instead.
>+      std::string track_id =
>+          RTCMediaStreamTrackStatsIDFromDirectionAndAttachment(
>+              kReceiver, receiver_selector->AttachmentId());
>+      for (const auto& stats : *report) {
>+        if (stats.type() != RTCInboundRTPStreamStats::kType)
>+          continue;
>+        const auto& inbound_rtp = stats.cast_to<RTCInboundRTPStreamStats>();
>+        if (inbound_rtp.track_id.is_defined() &&
>+            *inbound_rtp.track_id == track_id) {
>+          rtpstream_ids.push_back(inbound_rtp.id());
>+        }
>+      }
>     }
>   }
>-
>-  // Build stats for each stream ID known.
>-  for (auto& it : track_ids) {
>-    std::unique_ptr<RTCMediaStreamStats> stream_stats(
>-        new RTCMediaStreamStats("RTCMediaStream_" + it.first, timestamp_us));
>-    stream_stats->stream_identifier = it.first;
>-    stream_stats->track_ids = it.second;
>-    report->AddStats(std::move(stream_stats));
>-  }
>+  if (rtpstream_ids.empty())
>+    return RTCStatsReport::Create(report->timestamp_us());
>+  return TakeReferencedStats(report->Copy(), rtpstream_ids);
> }
> 
> }  // namespace
> 
>+RTCStatsCollector::RequestInfo::RequestInfo(
>+    rtc::scoped_refptr<RTCStatsCollectorCallback> callback)
>+    : RequestInfo(FilterMode::kAll, std::move(callback), nullptr, nullptr) {}
>+
>+RTCStatsCollector::RequestInfo::RequestInfo(
>+    rtc::scoped_refptr<RtpSenderInternal> selector,
>+    rtc::scoped_refptr<RTCStatsCollectorCallback> callback)
>+    : RequestInfo(FilterMode::kSenderSelector,
>+                  std::move(callback),
>+                  std::move(selector),
>+                  nullptr) {}
>+
>+RTCStatsCollector::RequestInfo::RequestInfo(
>+    rtc::scoped_refptr<RtpReceiverInternal> selector,
>+    rtc::scoped_refptr<RTCStatsCollectorCallback> callback)
>+    : RequestInfo(FilterMode::kReceiverSelector,
>+                  std::move(callback),
>+                  nullptr,
>+                  std::move(selector)) {}
>+
>+RTCStatsCollector::RequestInfo::RequestInfo(
>+    RTCStatsCollector::RequestInfo::FilterMode filter_mode,
>+    rtc::scoped_refptr<RTCStatsCollectorCallback> callback,
>+    rtc::scoped_refptr<RtpSenderInternal> sender_selector,
>+    rtc::scoped_refptr<RtpReceiverInternal> receiver_selector)
>+    : filter_mode_(filter_mode),
>+      callback_(std::move(callback)),
>+      sender_selector_(std::move(sender_selector)),
>+      receiver_selector_(std::move(receiver_selector)) {
>+  RTC_DCHECK(callback_);
>+  RTC_DCHECK(!sender_selector_ || !receiver_selector_);
>+}
>+
> rtc::scoped_refptr<RTCStatsCollector> RTCStatsCollector::Create(
>-    PeerConnection* pc, int64_t cache_lifetime_us) {
>+    PeerConnectionInternal* pc,
>+    int64_t cache_lifetime_us) {
>   return rtc::scoped_refptr<RTCStatsCollector>(
>       new rtc::RefCountedObject<RTCStatsCollector>(pc, cache_lifetime_us));
> }
> 
>-RTCStatsCollector::RTCStatsCollector(PeerConnection* pc,
>+RTCStatsCollector::RTCStatsCollector(PeerConnectionInternal* pc,
>                                      int64_t cache_lifetime_us)
>     : pc_(pc),
>       signaling_thread_(pc->signaling_thread()),
>@@ -669,7 +729,7 @@ RTCStatsCollector::RTCStatsCollector(PeerConnection* pc,
>   RTC_DCHECK(worker_thread_);
>   RTC_DCHECK(network_thread_);
>   RTC_DCHECK_GE(cache_lifetime_us_, 0);
>-  pc_->SignalDataChannelCreated.connect(
>+  pc_->SignalDataChannelCreated().connect(
>       this, &RTCStatsCollector::OnDataChannelCreated);
> }
> 
>@@ -679,16 +739,39 @@ RTCStatsCollector::~RTCStatsCollector() {
> 
> void RTCStatsCollector::GetStatsReport(
>     rtc::scoped_refptr<RTCStatsCollectorCallback> callback) {
>+  GetStatsReportInternal(RequestInfo(std::move(callback)));
>+}
>+
>+void RTCStatsCollector::GetStatsReport(
>+    rtc::scoped_refptr<RtpSenderInternal> selector,
>+    rtc::scoped_refptr<RTCStatsCollectorCallback> callback) {
>+  GetStatsReportInternal(RequestInfo(std::move(selector), std::move(callback)));
>+}
>+
>+void RTCStatsCollector::GetStatsReport(
>+    rtc::scoped_refptr<RtpReceiverInternal> selector,
>+    rtc::scoped_refptr<RTCStatsCollectorCallback> callback) {
>+  GetStatsReportInternal(RequestInfo(std::move(selector), std::move(callback)));
>+}
>+
>+void RTCStatsCollector::GetStatsReportInternal(
>+    RTCStatsCollector::RequestInfo request) {
>   RTC_DCHECK(signaling_thread_->IsCurrent());
>-  RTC_DCHECK(callback);
>-  callbacks_.push_back(callback);
>+  requests_.push_back(std::move(request));
> 
>   // "Now" using a monotonically increasing timer.
>   int64_t cache_now_us = rtc::TimeMicros();
>   if (cached_report_ &&
>       cache_now_us - cache_timestamp_us_ <= cache_lifetime_us_) {
>-    // We have a fresh cached report to deliver.
>-    DeliverCachedReport();
>+    // We have a fresh cached report to deliver. Deliver asynchronously, since
>+    // the caller may not be expecting a synchronous callback, and it avoids
>+    // reentrancy problems.
>+    std::vector<RequestInfo> requests;
>+    requests.swap(requests_);
>+    invoker_.AsyncInvoke<void>(
>+        RTC_FROM_HERE, signaling_thread_,
>+        rtc::Bind(&RTCStatsCollector::DeliverCachedReport, this, cached_report_,
>+                  std::move(requests)));
>   } else if (!num_pending_partial_reports_) {
>     // Only start gathering stats if we're not already gathering stats. In the
>     // case of already gathering stats, |callback_| will be invoked when there
>@@ -702,36 +785,13 @@ void RTCStatsCollector::GetStatsReport(
>     num_pending_partial_reports_ = 2;
>     partial_report_timestamp_us_ = cache_now_us;
> 
>-    // Prepare |channel_name_pairs_| for use in
>-    // |ProducePartialResultsOnNetworkThread|.
>-    channel_name_pairs_.reset(new ChannelNamePairs());
>-    if (pc_->voice_channel()) {
>-      channel_name_pairs_->voice =
>-          ChannelNamePair(pc_->voice_channel()->content_name(),
>-                          pc_->voice_channel()->transport_name());
>-    }
>-    if (pc_->video_channel()) {
>-      channel_name_pairs_->video =
>-          ChannelNamePair(pc_->video_channel()->content_name(),
>-                          pc_->video_channel()->transport_name());
>-    }
>-    if (pc_->rtp_data_channel()) {
>-      channel_name_pairs_->data =
>-          ChannelNamePair(pc_->rtp_data_channel()->content_name(),
>-                          pc_->rtp_data_channel()->transport_name());
>-    }
>-    if (pc_->sctp_content_name()) {
>-      channel_name_pairs_->data = ChannelNamePair(*pc_->sctp_content_name(),
>-                                                  *pc_->sctp_transport_name());
>-    }
>-    // Prepare |track_media_info_map_| for use in
>+    // Prepare |transceiver_stats_infos_| for use in
>     // |ProducePartialResultsOnNetworkThread| and
>     // |ProducePartialResultsOnSignalingThread|.
>-    track_media_info_map_.reset(PrepareTrackMediaInfoMap_s().release());
>-    // Prepare |track_to_id_| for use in |ProducePartialResultsOnNetworkThread|.
>-    // This avoids a possible deadlock if |MediaStreamTrackInterface::id| is
>-    // implemented to invoke on the signaling thread.
>-    track_to_id_ = PrepareTrackToID_s();
>+    transceiver_stats_infos_ = PrepareTransceiverStatsInfos_s();
>+    // Prepare |transport_names_| for use in
>+    // |ProducePartialResultsOnNetworkThread|.
>+    transport_names_ = PrepareTransportNames_s();
> 
>     // Prepare |call_stats_| here since GetCallStats() will hop to the worker
>     // thread.
>@@ -770,7 +830,8 @@ void RTCStatsCollector::ProducePartialResultsOnSignalingThread(
>       timestamp_us);
> 
>   ProduceDataChannelStats_s(timestamp_us, report.get());
>-  ProduceMediaStreamAndTrackStats_s(timestamp_us, report.get());
>+  ProduceMediaStreamStats_s(timestamp_us, report.get());
>+  ProduceMediaStreamTrackStats_s(timestamp_us, report.get());
>   ProducePeerConnectionStats_s(timestamp_us, report.get());
> 
>   AddPartialResults(report);
>@@ -782,24 +843,19 @@ void RTCStatsCollector::ProducePartialResultsOnNetworkThread(
>   rtc::scoped_refptr<RTCStatsReport> report = RTCStatsReport::Create(
>       timestamp_us);
> 
>-  std::unique_ptr<SessionStats> session_stats =
>-      pc_->GetSessionStats(*channel_name_pairs_);
>-  if (session_stats) {
>-    std::map<std::string, CertificateStatsPair> transport_cert_stats =
>-        PrepareTransportCertificateStats_n(*session_stats);
>-
>-    ProduceCertificateStats_n(
>-        timestamp_us, transport_cert_stats, report.get());
>-    ProduceCodecStats_n(
>-        timestamp_us, *track_media_info_map_, report.get());
>-    ProduceIceCandidateAndPairStats_n(timestamp_us, *session_stats,
>-                                      track_media_info_map_->video_media_info(),
>-                                      call_stats_, report.get());
>-    ProduceRTPStreamStats_n(timestamp_us, *session_stats, *channel_name_pairs_,
>-                            *track_media_info_map_, report.get());
>-    ProduceTransportStats_n(
>-        timestamp_us, *session_stats, transport_cert_stats, report.get());
>-  }
>+  std::map<std::string, cricket::TransportStats> transport_stats_by_name =
>+      pc_->GetTransportStatsByNames(transport_names_);
>+
>+  std::map<std::string, CertificateStatsPair> transport_cert_stats =
>+      PrepareTransportCertificateStats_n(transport_stats_by_name);
>+
>+  ProduceCertificateStats_n(timestamp_us, transport_cert_stats, report.get());
>+  ProduceCodecStats_n(timestamp_us, transceiver_stats_infos_, report.get());
>+  ProduceIceCandidateAndPairStats_n(timestamp_us, transport_stats_by_name,
>+                                    call_stats_, report.get());
>+  ProduceRTPStreamStats_n(timestamp_us, transceiver_stats_infos_, report.get());
>+  ProduceTransportStats_n(timestamp_us, transport_stats_by_name,
>+                          transport_cert_stats, report.get());
> 
>   AddPartialResults(report);
> }
>@@ -829,27 +885,48 @@ void RTCStatsCollector::AddPartialResults_s(
>     cache_timestamp_us_ = partial_report_timestamp_us_;
>     cached_report_ = partial_report_;
>     partial_report_ = nullptr;
>-    channel_name_pairs_.reset();
>-    track_media_info_map_.reset();
>-    track_to_id_.clear();
>+    transceiver_stats_infos_.clear();
>     // Trace WebRTC Stats when getStats is called on Javascript.
>     // This allows access to WebRTC stats from trace logs. To enable them,
>     // select the "webrtc_stats" category when recording traces.
>     TRACE_EVENT_INSTANT1("webrtc_stats", "webrtc_stats", "report",
>                          cached_report_->ToJson());
>-    DeliverCachedReport();
>+
>+    // Deliver report and clear |requests_|.
>+    std::vector<RequestInfo> requests;
>+    requests.swap(requests_);
>+    DeliverCachedReport(cached_report_, std::move(requests));
>   }
> }
> 
>-void RTCStatsCollector::DeliverCachedReport() {
>+void RTCStatsCollector::DeliverCachedReport(
>+    rtc::scoped_refptr<const RTCStatsReport> cached_report,
>+    std::vector<RTCStatsCollector::RequestInfo> requests) {
>   RTC_DCHECK(signaling_thread_->IsCurrent());
>-  RTC_DCHECK(!callbacks_.empty());
>-  RTC_DCHECK(cached_report_);
>-  for (const rtc::scoped_refptr<RTCStatsCollectorCallback>& callback :
>-       callbacks_) {
>-    callback->OnStatsDelivered(cached_report_);
>+  RTC_DCHECK(!requests.empty());
>+  RTC_DCHECK(cached_report);
>+
>+  for (const RequestInfo& request : requests) {
>+    if (request.filter_mode() == RequestInfo::FilterMode::kAll) {
>+      request.callback()->OnStatsDelivered(cached_report);
>+    } else {
>+      bool filter_by_sender_selector;
>+      rtc::scoped_refptr<RtpSenderInternal> sender_selector;
>+      rtc::scoped_refptr<RtpReceiverInternal> receiver_selector;
>+      if (request.filter_mode() == RequestInfo::FilterMode::kSenderSelector) {
>+        filter_by_sender_selector = true;
>+        sender_selector = request.sender_selector();
>+      } else {
>+        RTC_DCHECK(request.filter_mode() ==
>+                   RequestInfo::FilterMode::kReceiverSelector);
>+        filter_by_sender_selector = false;
>+        receiver_selector = request.receiver_selector();
>+      }
>+      request.callback()->OnStatsDelivered(CreateReportFilteredBySelector(
>+          filter_by_sender_selector, cached_report, sender_selector,
>+          receiver_selector));
>+    }
>   }
>-  callbacks_.clear();
> }
> 
> void RTCStatsCollector::ProduceCertificateStats_n(
>@@ -870,37 +947,43 @@ void RTCStatsCollector::ProduceCertificateStats_n(
> }
> 
> void RTCStatsCollector::ProduceCodecStats_n(
>-    int64_t timestamp_us, const TrackMediaInfoMap& track_media_info_map,
>+    int64_t timestamp_us,
>+    const std::vector<RtpTransceiverStatsInfo>& transceiver_stats_infos,
>     RTCStatsReport* report) const {
>   RTC_DCHECK(network_thread_->IsCurrent());
>-  // Audio
>-  if (track_media_info_map.voice_media_info()) {
>-    // Inbound
>-    for (const auto& pair :
>-         track_media_info_map.voice_media_info()->receive_codecs) {
>-      report->AddStats(CodecStatsFromRtpCodecParameters(
>-          timestamp_us, true, true, pair.second));
>-    }
>-    // Outbound
>-    for (const auto& pair :
>-         track_media_info_map.voice_media_info()->send_codecs) {
>-      report->AddStats(CodecStatsFromRtpCodecParameters(
>-          timestamp_us, false, true, pair.second));
>+  for (const auto& stats : transceiver_stats_infos) {
>+    if (!stats.mid) {
>+      continue;
>     }
>-  }
>-  // Video
>-  if (track_media_info_map.video_media_info()) {
>-    // Inbound
>-    for (const auto& pair :
>-         track_media_info_map.video_media_info()->receive_codecs) {
>-      report->AddStats(CodecStatsFromRtpCodecParameters(
>-          timestamp_us, true, false, pair.second));
>+    const cricket::VoiceMediaInfo* voice_media_info =
>+        stats.track_media_info_map->voice_media_info();
>+    const cricket::VideoMediaInfo* video_media_info =
>+        stats.track_media_info_map->video_media_info();
>+    // Audio
>+    if (voice_media_info) {
>+      // Inbound
>+      for (const auto& pair : voice_media_info->receive_codecs) {
>+        report->AddStats(CodecStatsFromRtpCodecParameters(
>+            timestamp_us, *stats.mid, true, pair.second));
>+      }
>+      // Outbound
>+      for (const auto& pair : voice_media_info->send_codecs) {
>+        report->AddStats(CodecStatsFromRtpCodecParameters(
>+            timestamp_us, *stats.mid, false, pair.second));
>+      }
>     }
>-    // Outbound
>-    for (const auto& pair :
>-         track_media_info_map.video_media_info()->send_codecs) {
>-      report->AddStats(CodecStatsFromRtpCodecParameters(
>-          timestamp_us, false, false, pair.second));
>+    // Video
>+    if (video_media_info) {
>+      // Inbound
>+      for (const auto& pair : video_media_info->receive_codecs) {
>+        report->AddStats(CodecStatsFromRtpCodecParameters(
>+            timestamp_us, *stats.mid, true, pair.second));
>+      }
>+      // Outbound
>+      for (const auto& pair : video_media_info->send_codecs) {
>+        report->AddStats(CodecStatsFromRtpCodecParameters(
>+            timestamp_us, *stats.mid, false, pair.second));
>+      }
>     }
>   }
> }
>@@ -912,7 +995,7 @@ void RTCStatsCollector::ProduceDataChannelStats_s(
>        pc_->sctp_data_channels()) {
>     std::unique_ptr<RTCDataChannelStats> data_channel_stats(
>         new RTCDataChannelStats(
>-            "RTCDataChannel_" + rtc::ToString<>(data_channel->id()),
>+            "RTCDataChannel_" + rtc::ToString(data_channel->id()),
>             timestamp_us));
>     data_channel_stats->label = data_channel->label();
>     data_channel_stats->protocol = data_channel->protocol();
>@@ -929,15 +1012,17 @@ void RTCStatsCollector::ProduceDataChannelStats_s(
> 
> void RTCStatsCollector::ProduceIceCandidateAndPairStats_n(
>     int64_t timestamp_us,
>-    const SessionStats& session_stats,
>-    const cricket::VideoMediaInfo* video_media_info,
>+    const std::map<std::string, cricket::TransportStats>&
>+        transport_stats_by_name,
>     const Call::Stats& call_stats,
>     RTCStatsReport* report) const {
>   RTC_DCHECK(network_thread_->IsCurrent());
>-  for (const auto& transport_stats : session_stats.transport_stats) {
>-    for (const auto& channel_stats : transport_stats.second.channel_stats) {
>+  for (const auto& entry : transport_stats_by_name) {
>+    const std::string& transport_name = entry.first;
>+    const cricket::TransportStats& transport_stats = entry.second;
>+    for (const auto& channel_stats : transport_stats.channel_stats) {
>       std::string transport_id = RTCTransportStatsIDFromTransportChannel(
>-          transport_stats.second.transport_name, channel_stats.component);
>+          transport_name, channel_stats.component);
>       for (const cricket::ConnectionInfo& info :
>            channel_stats.connection_infos) {
>         std::unique_ptr<RTCIceCandidatePairStats> candidate_pair_stats(
>@@ -1008,16 +1093,61 @@ void RTCStatsCollector::ProduceIceCandidateAndPairStats_n(
>   }
> }
> 
>-void RTCStatsCollector::ProduceMediaStreamAndTrackStats_s(
>-    int64_t timestamp_us, RTCStatsReport* report) const {
>+void RTCStatsCollector::ProduceMediaStreamStats_s(
>+    int64_t timestamp_us,
>+    RTCStatsReport* report) const {
>+  RTC_DCHECK(signaling_thread_->IsCurrent());
>+
>+  std::map<std::string, std::vector<std::string>> track_ids;
>+
>+  for (const auto& stats : transceiver_stats_infos_) {
>+    for (auto sender : stats.transceiver->senders()) {
>+      std::string track_id =
>+          RTCMediaStreamTrackStatsIDFromDirectionAndAttachment(
>+              kSender, sender->internal()->AttachmentId());
>+      for (auto& stream_id : sender->stream_ids()) {
>+        track_ids[stream_id].push_back(track_id);
>+      }
>+    }
>+    for (auto receiver : stats.transceiver->receivers()) {
>+      std::string track_id =
>+          RTCMediaStreamTrackStatsIDFromDirectionAndAttachment(
>+              kReceiver, receiver->internal()->AttachmentId());
>+      for (auto& stream : receiver->streams()) {
>+        track_ids[stream->id()].push_back(track_id);
>+      }
>+    }
>+  }
>+
>+  // Build stats for each stream ID known.
>+  for (auto& it : track_ids) {
>+    std::unique_ptr<RTCMediaStreamStats> stream_stats(
>+        new RTCMediaStreamStats("RTCMediaStream_" + it.first, timestamp_us));
>+    stream_stats->stream_identifier = it.first;
>+    stream_stats->track_ids = it.second;
>+    report->AddStats(std::move(stream_stats));
>+  }
>+}
>+
>+void RTCStatsCollector::ProduceMediaStreamTrackStats_s(
>+    int64_t timestamp_us,
>+    RTCStatsReport* report) const {
>   RTC_DCHECK(signaling_thread_->IsCurrent());
>-  RTC_DCHECK(track_media_info_map_);
>-  ProduceMediaStreamStats(timestamp_us, pc_->GetSenders(), pc_->GetReceivers(),
>-                          report);
>-  ProduceSenderMediaTrackStats(timestamp_us, *track_media_info_map_,
>-                               pc_->GetSenders(), report);
>-  ProduceReceiverMediaTrackStats(timestamp_us, *track_media_info_map_,
>-                                 pc_->GetReceivers(), report);
>+  for (const RtpTransceiverStatsInfo& stats : transceiver_stats_infos_) {
>+    std::vector<rtc::scoped_refptr<RtpSenderInternal>> senders;
>+    for (auto sender : stats.transceiver->senders()) {
>+      senders.push_back(sender->internal());
>+    }
>+    ProduceSenderMediaTrackStats(timestamp_us, *stats.track_media_info_map,
>+                                 senders, report);
>+
>+    std::vector<rtc::scoped_refptr<RtpReceiverInternal>> receivers;
>+    for (auto receiver : stats.transceiver->receivers()) {
>+      receivers.push_back(receiver->internal());
>+    }
>+    ProduceReceiverMediaTrackStats(timestamp_us, *stats.track_media_info_map,
>+                                   receivers, report);
>+  }
> }
> 
> void RTCStatsCollector::ProducePeerConnectionStats_s(
>@@ -1032,154 +1162,163 @@ void RTCStatsCollector::ProducePeerConnectionStats_s(
> 
> void RTCStatsCollector::ProduceRTPStreamStats_n(
>     int64_t timestamp_us,
>-    const SessionStats& session_stats,
>-    const ChannelNamePairs& channel_name_pairs,
>-    const TrackMediaInfoMap& track_media_info_map,
>+    const std::vector<RtpTransceiverStatsInfo>& transceiver_stats_infos,
>     RTCStatsReport* report) const {
>   RTC_DCHECK(network_thread_->IsCurrent());
> 
>-  // Audio
>-  if (track_media_info_map.voice_media_info()) {
>-    RTC_DCHECK(channel_name_pairs.voice);
>-    const std::string& transport_name =
>-        (*channel_name_pairs.voice).transport_name;
>-    std::string transport_id = RTCTransportStatsIDFromTransportChannel(
>-        transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
>-    // Inbound
>-    for (const cricket::VoiceReceiverInfo& voice_receiver_info :
>-         track_media_info_map.voice_media_info()->receivers) {
>-      // TODO(nisse): SSRC == 0 currently means none. Delete check when that
>-      // is fixed.
>-      if (voice_receiver_info.ssrc() == 0)
>-        continue;
>-      std::unique_ptr<RTCInboundRTPStreamStats> inbound_audio(
>-          new RTCInboundRTPStreamStats(RTCInboundRTPStreamStatsIDFromSSRC(
>-                                           true, voice_receiver_info.ssrc()),
>-                                       timestamp_us));
>-      SetInboundRTPStreamStatsFromVoiceReceiverInfo(
>-          voice_receiver_info, inbound_audio.get());
>-      // TODO(hta): This lookup should look for the sender, not the track.
>-      rtc::scoped_refptr<AudioTrackInterface> audio_track =
>-          track_media_info_map_->GetAudioTrack(voice_receiver_info);
>-      if (audio_track) {
>-        RTC_DCHECK(track_to_id_.find(audio_track.get()) != track_to_id_.end());
>-        inbound_audio
>-            ->track_id = RTCMediaStreamTrackStatsIDFromDirectionAndAttachment(
>-            kReceiver,
>-            track_media_info_map_->GetAttachmentIdByTrack(audio_track).value());
>-      }
>-      inbound_audio->transport_id = transport_id;
>-      report->AddStats(std::move(inbound_audio));
>+  for (const RtpTransceiverStatsInfo& stats : transceiver_stats_infos) {
>+    if (stats.media_type == cricket::MEDIA_TYPE_AUDIO) {
>+      ProduceAudioRTPStreamStats_n(timestamp_us, stats, report);
>+    } else if (stats.media_type == cricket::MEDIA_TYPE_VIDEO) {
>+      ProduceVideoRTPStreamStats_n(timestamp_us, stats, report);
>+    } else {
>+      RTC_NOTREACHED();
>     }
>-    // Outbound
>-    for (const cricket::VoiceSenderInfo& voice_sender_info :
>-         track_media_info_map.voice_media_info()->senders) {
>-      // TODO(nisse): SSRC == 0 currently means none. Delete check when that
>-      // is fixed.
>-      if (voice_sender_info.ssrc() == 0)
>-        continue;
>-      std::unique_ptr<RTCOutboundRTPStreamStats> outbound_audio(
>-          new RTCOutboundRTPStreamStats(
>-              RTCOutboundRTPStreamStatsIDFromSSRC(
>-                  true, voice_sender_info.ssrc()),
>-              timestamp_us));
>-      SetOutboundRTPStreamStatsFromVoiceSenderInfo(
>-          voice_sender_info, outbound_audio.get());
>-      rtc::scoped_refptr<AudioTrackInterface> audio_track =
>-          track_media_info_map_->GetAudioTrack(voice_sender_info);
>-      if (audio_track) {
>-        RTC_DCHECK(track_to_id_.find(audio_track.get()) != track_to_id_.end());
>-        outbound_audio
>-            ->track_id = RTCMediaStreamTrackStatsIDFromDirectionAndAttachment(
>-            kSender,
>-            track_media_info_map.GetAttachmentIdByTrack(audio_track).value());
>-      }
>-      outbound_audio->transport_id = transport_id;
>-      report->AddStats(std::move(outbound_audio));
>+  }
>+}
>+
>+void RTCStatsCollector::ProduceAudioRTPStreamStats_n(
>+    int64_t timestamp_us,
>+    const RtpTransceiverStatsInfo& stats,
>+    RTCStatsReport* report) const {
>+  if (!stats.mid || !stats.transport_name) {
>+    return;
>+  }
>+  RTC_DCHECK(stats.track_media_info_map);
>+  const TrackMediaInfoMap& track_media_info_map = *stats.track_media_info_map;
>+  RTC_DCHECK(track_media_info_map.voice_media_info());
>+  std::string mid = *stats.mid;
>+  std::string transport_id = RTCTransportStatsIDFromTransportChannel(
>+      *stats.transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
>+  // Inbound
>+  for (const cricket::VoiceReceiverInfo& voice_receiver_info :
>+       track_media_info_map.voice_media_info()->receivers) {
>+    if (!voice_receiver_info.connected())
>+      continue;
>+    auto inbound_audio = absl::make_unique<RTCInboundRTPStreamStats>(
>+        RTCInboundRTPStreamStatsIDFromSSRC(true, voice_receiver_info.ssrc()),
>+        timestamp_us);
>+    SetInboundRTPStreamStatsFromVoiceReceiverInfo(mid, voice_receiver_info,
>+                                                  inbound_audio.get());
>+    // TODO(hta): This lookup should look for the sender, not the track.
>+    rtc::scoped_refptr<AudioTrackInterface> audio_track =
>+        track_media_info_map.GetAudioTrack(voice_receiver_info);
>+    if (audio_track) {
>+      inbound_audio->track_id =
>+          RTCMediaStreamTrackStatsIDFromDirectionAndAttachment(
>+              kReceiver,
>+              track_media_info_map.GetAttachmentIdByTrack(audio_track).value());
>     }
>+    inbound_audio->transport_id = transport_id;
>+    report->AddStats(std::move(inbound_audio));
>   }
>-  // Video
>-  if (track_media_info_map.video_media_info()) {
>-    RTC_DCHECK(channel_name_pairs.video);
>-    const std::string& transport_name =
>-        (*channel_name_pairs.video).transport_name;
>-    std::string transport_id = RTCTransportStatsIDFromTransportChannel(
>-        transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
>-    // Inbound
>-    for (const cricket::VideoReceiverInfo& video_receiver_info :
>-         track_media_info_map.video_media_info()->receivers) {
>-      // TODO(nisse): SSRC == 0 currently means none. Delete check when that
>-      // is fixed.
>-      if (video_receiver_info.ssrc() == 0)
>-        continue;
>-      std::unique_ptr<RTCInboundRTPStreamStats> inbound_video(
>-          new RTCInboundRTPStreamStats(
>-              RTCInboundRTPStreamStatsIDFromSSRC(
>-                  false, video_receiver_info.ssrc()),
>-              timestamp_us));
>-      SetInboundRTPStreamStatsFromVideoReceiverInfo(
>-          video_receiver_info, inbound_video.get());
>-      rtc::scoped_refptr<VideoTrackInterface> video_track =
>-          track_media_info_map_->GetVideoTrack(video_receiver_info);
>-      if (video_track) {
>-        RTC_DCHECK(track_to_id_.find(video_track.get()) != track_to_id_.end());
>-        inbound_video
>-            ->track_id = RTCMediaStreamTrackStatsIDFromDirectionAndAttachment(
>-            kReceiver,
>-            track_media_info_map_->GetAttachmentIdByTrack(video_track).value());
>-      }
>-      inbound_video->transport_id = transport_id;
>-      report->AddStats(std::move(inbound_video));
>+  // Outbound
>+  for (const cricket::VoiceSenderInfo& voice_sender_info :
>+       track_media_info_map.voice_media_info()->senders) {
>+    if (!voice_sender_info.connected())
>+      continue;
>+    auto outbound_audio = absl::make_unique<RTCOutboundRTPStreamStats>(
>+        RTCOutboundRTPStreamStatsIDFromSSRC(true, voice_sender_info.ssrc()),
>+        timestamp_us);
>+    SetOutboundRTPStreamStatsFromVoiceSenderInfo(mid, voice_sender_info,
>+                                                 outbound_audio.get());
>+    rtc::scoped_refptr<AudioTrackInterface> audio_track =
>+        track_media_info_map.GetAudioTrack(voice_sender_info);
>+    if (audio_track) {
>+      outbound_audio->track_id =
>+          RTCMediaStreamTrackStatsIDFromDirectionAndAttachment(
>+              kSender,
>+              track_media_info_map.GetAttachmentIdByTrack(audio_track).value());
>     }
>-    // Outbound
>-    for (const cricket::VideoSenderInfo& video_sender_info :
>-         track_media_info_map.video_media_info()->senders) {
>-      // TODO(nisse): SSRC == 0 currently means none. Delete check when that
>-      // is fixed.
>-      if (video_sender_info.ssrc() == 0)
>-        continue;
>-      std::unique_ptr<RTCOutboundRTPStreamStats> outbound_video(
>-          new RTCOutboundRTPStreamStats(RTCOutboundRTPStreamStatsIDFromSSRC(
>-                                            false, video_sender_info.ssrc()),
>-                                        timestamp_us));
>-      SetOutboundRTPStreamStatsFromVideoSenderInfo(
>-          video_sender_info, outbound_video.get());
>-      rtc::scoped_refptr<VideoTrackInterface> video_track =
>-          track_media_info_map_->GetVideoTrack(video_sender_info);
>-      if (video_track) {
>-        RTC_DCHECK(track_to_id_.find(video_track.get()) != track_to_id_.end());
>-        outbound_video
>-            ->track_id = RTCMediaStreamTrackStatsIDFromDirectionAndAttachment(
>-            kSender,
>-            track_media_info_map_->GetAttachmentIdByTrack(video_track).value());
>-      }
>-      outbound_video->transport_id = transport_id;
>-      report->AddStats(std::move(outbound_video));
>+    outbound_audio->transport_id = transport_id;
>+    report->AddStats(std::move(outbound_audio));
>+  }
>+}
>+
>+void RTCStatsCollector::ProduceVideoRTPStreamStats_n(
>+    int64_t timestamp_us,
>+    const RtpTransceiverStatsInfo& stats,
>+    RTCStatsReport* report) const {
>+  if (!stats.mid || !stats.transport_name) {
>+    return;
>+  }
>+  RTC_DCHECK(stats.track_media_info_map);
>+  const TrackMediaInfoMap& track_media_info_map = *stats.track_media_info_map;
>+  RTC_DCHECK(track_media_info_map.video_media_info());
>+  std::string mid = *stats.mid;
>+  std::string transport_id = RTCTransportStatsIDFromTransportChannel(
>+      *stats.transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
>+  // Inbound
>+  for (const cricket::VideoReceiverInfo& video_receiver_info :
>+       track_media_info_map.video_media_info()->receivers) {
>+    if (!video_receiver_info.connected())
>+      continue;
>+    auto inbound_video = absl::make_unique<RTCInboundRTPStreamStats>(
>+        RTCInboundRTPStreamStatsIDFromSSRC(false, video_receiver_info.ssrc()),
>+        timestamp_us);
>+    SetInboundRTPStreamStatsFromVideoReceiverInfo(mid, video_receiver_info,
>+                                                  inbound_video.get());
>+    rtc::scoped_refptr<VideoTrackInterface> video_track =
>+        track_media_info_map.GetVideoTrack(video_receiver_info);
>+    if (video_track) {
>+      inbound_video->track_id =
>+          RTCMediaStreamTrackStatsIDFromDirectionAndAttachment(
>+              kReceiver,
>+              track_media_info_map.GetAttachmentIdByTrack(video_track).value());
>+    }
>+    inbound_video->transport_id = transport_id;
>+    report->AddStats(std::move(inbound_video));
>+  }
>+  // Outbound
>+  for (const cricket::VideoSenderInfo& video_sender_info :
>+       track_media_info_map.video_media_info()->senders) {
>+    if (!video_sender_info.connected())
>+      continue;
>+    auto outbound_video = absl::make_unique<RTCOutboundRTPStreamStats>(
>+        RTCOutboundRTPStreamStatsIDFromSSRC(false, video_sender_info.ssrc()),
>+        timestamp_us);
>+    SetOutboundRTPStreamStatsFromVideoSenderInfo(mid, video_sender_info,
>+                                                 outbound_video.get());
>+    rtc::scoped_refptr<VideoTrackInterface> video_track =
>+        track_media_info_map.GetVideoTrack(video_sender_info);
>+    if (video_track) {
>+      outbound_video->track_id =
>+          RTCMediaStreamTrackStatsIDFromDirectionAndAttachment(
>+              kSender,
>+              track_media_info_map.GetAttachmentIdByTrack(video_track).value());
>     }
>+    outbound_video->transport_id = transport_id;
>+    report->AddStats(std::move(outbound_video));
>   }
> }
> 
> void RTCStatsCollector::ProduceTransportStats_n(
>-    int64_t timestamp_us, const SessionStats& session_stats,
>+    int64_t timestamp_us,
>+    const std::map<std::string, cricket::TransportStats>&
>+        transport_stats_by_name,
>     const std::map<std::string, CertificateStatsPair>& transport_cert_stats,
>     RTCStatsReport* report) const {
>   RTC_DCHECK(network_thread_->IsCurrent());
>-  for (const auto& transport : session_stats.transport_stats) {
>+  for (const auto& entry : transport_stats_by_name) {
>+    const std::string& transport_name = entry.first;
>+    const cricket::TransportStats& transport_stats = entry.second;
>+
>     // Get reference to RTCP channel, if it exists.
>     std::string rtcp_transport_stats_id;
>-    for (const auto& channel_stats : transport.second.channel_stats) {
>+    for (const cricket::TransportChannelStats& channel_stats :
>+         transport_stats.channel_stats) {
>       if (channel_stats.component ==
>           cricket::ICE_CANDIDATE_COMPONENT_RTCP) {
>         rtcp_transport_stats_id = RTCTransportStatsIDFromTransportChannel(
>-            transport.second.transport_name, channel_stats.component);
>+            transport_name, channel_stats.component);
>         break;
>       }
>     }
> 
>     // Get reference to local and remote certificates of this transport, if they
>     // exist.
>-    const auto& certificate_stats_it = transport_cert_stats.find(
>-        transport.second.transport_name);
>+    const auto& certificate_stats_it =
>+        transport_cert_stats.find(transport_name);
>     RTC_DCHECK(certificate_stats_it != transport_cert_stats.cend());
>     std::string local_certificate_id;
>     if (certificate_stats_it->second.local) {
>@@ -1193,12 +1332,12 @@ void RTCStatsCollector::ProduceTransportStats_n(
>     }
> 
>     // There is one transport stats for each channel.
>-    for (const auto& channel_stats : transport.second.channel_stats) {
>+    for (const cricket::TransportChannelStats& channel_stats :
>+         transport_stats.channel_stats) {
>       std::unique_ptr<RTCTransportStats> transport_stats(
>-          new RTCTransportStats(
>-              RTCTransportStatsIDFromTransportChannel(
>-                  transport.second.transport_name, channel_stats.component),
>-              timestamp_us));
>+          new RTCTransportStats(RTCTransportStatsIDFromTransportChannel(
>+                                    transport_name, channel_stats.component),
>+                                timestamp_us));
>       transport_stats->bytes_sent = 0;
>       transport_stats->bytes_received = 0;
>       transport_stats->dtls_state = DtlsTransportStateToRTCDtlsTransportState(
>@@ -1227,72 +1366,148 @@ void RTCStatsCollector::ProduceTransportStats_n(
> 
> std::map<std::string, RTCStatsCollector::CertificateStatsPair>
> RTCStatsCollector::PrepareTransportCertificateStats_n(
>-    const SessionStats& session_stats) const {
>+    const std::map<std::string, cricket::TransportStats>&
>+        transport_stats_by_name) const {
>   RTC_DCHECK(network_thread_->IsCurrent());
>   std::map<std::string, CertificateStatsPair> transport_cert_stats;
>-  for (const auto& transport_stats : session_stats.transport_stats) {
>+  for (const auto& entry : transport_stats_by_name) {
>+    const std::string& transport_name = entry.first;
>+
>     CertificateStatsPair certificate_stats_pair;
>     rtc::scoped_refptr<rtc::RTCCertificate> local_certificate;
>-    if (pc_->GetLocalCertificate(transport_stats.second.transport_name,
>-                                 &local_certificate)) {
>+    if (pc_->GetLocalCertificate(transport_name, &local_certificate)) {
>       certificate_stats_pair.local =
>-          local_certificate->ssl_certificate().GetStats();
>+          local_certificate->ssl_cert_chain().GetStats();
>     }
>-    std::unique_ptr<rtc::SSLCertificate> remote_certificate =
>-        pc_->GetRemoteSSLCertificate(transport_stats.second.transport_name);
>-    if (remote_certificate) {
>-      certificate_stats_pair.remote = remote_certificate->GetStats();
>+
>+    std::unique_ptr<rtc::SSLCertChain> remote_cert_chain =
>+        pc_->GetRemoteSSLCertChain(transport_name);
>+    if (remote_cert_chain) {
>+      certificate_stats_pair.remote = remote_cert_chain->GetStats();
>     }
>+
>     transport_cert_stats.insert(
>-        std::make_pair(transport_stats.second.transport_name,
>-                       std::move(certificate_stats_pair)));
>+        std::make_pair(transport_name, std::move(certificate_stats_pair)));
>   }
>   return transport_cert_stats;
> }
> 
>-std::unique_ptr<TrackMediaInfoMap>
>-RTCStatsCollector::PrepareTrackMediaInfoMap_s() const {
>-  RTC_DCHECK(signaling_thread_->IsCurrent());
>-  std::unique_ptr<cricket::VoiceMediaInfo> voice_media_info;
>-  if (pc_->voice_channel()) {
>-    voice_media_info.reset(new cricket::VoiceMediaInfo());
>-    if (!pc_->voice_channel()->GetStats(voice_media_info.get())) {
>-      voice_media_info.reset();
>+std::vector<RTCStatsCollector::RtpTransceiverStatsInfo>
>+RTCStatsCollector::PrepareTransceiverStatsInfos_s() const {
>+  std::vector<RtpTransceiverStatsInfo> transceiver_stats_infos;
>+
>+  // These are used to invoke GetStats for all the media channels together in
>+  // one worker thread hop.
>+  std::map<cricket::VoiceMediaChannel*,
>+           std::unique_ptr<cricket::VoiceMediaInfo>>
>+      voice_stats;
>+  std::map<cricket::VideoMediaChannel*,
>+           std::unique_ptr<cricket::VideoMediaInfo>>
>+      video_stats;
>+
>+  for (auto transceiver : pc_->GetTransceiversInternal()) {
>+    cricket::MediaType media_type = transceiver->media_type();
>+
>+    // Prepare stats entry. The TrackMediaInfoMap will be filled in after the
>+    // stats have been fetched on the worker thread.
>+    transceiver_stats_infos.emplace_back();
>+    RtpTransceiverStatsInfo& stats = transceiver_stats_infos.back();
>+    stats.transceiver = transceiver->internal();
>+    stats.media_type = media_type;
>+
>+    cricket::BaseChannel* channel = transceiver->internal()->channel();
>+    if (!channel) {
>+      // The remaining fields require a BaseChannel.
>+      continue;
>+    }
>+
>+    stats.mid = channel->content_name();
>+    stats.transport_name = channel->transport_name();
>+
>+    if (media_type == cricket::MEDIA_TYPE_AUDIO) {
>+      auto* voice_channel = static_cast<cricket::VoiceChannel*>(channel);
>+      RTC_DCHECK(voice_stats.find(voice_channel->media_channel()) ==
>+                 voice_stats.end());
>+      voice_stats[voice_channel->media_channel()] =
>+          absl::make_unique<cricket::VoiceMediaInfo>();
>+    } else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
>+      auto* video_channel = static_cast<cricket::VideoChannel*>(channel);
>+      RTC_DCHECK(video_stats.find(video_channel->media_channel()) ==
>+                 video_stats.end());
>+      video_stats[video_channel->media_channel()] =
>+          absl::make_unique<cricket::VideoMediaInfo>();
>+    } else {
>+      RTC_NOTREACHED();
>     }
>   }
>-  std::unique_ptr<cricket::VideoMediaInfo> video_media_info;
>-  if (pc_->video_channel()) {
>-    video_media_info.reset(new cricket::VideoMediaInfo());
>-    if (!pc_->video_channel()->GetStats(video_media_info.get())) {
>-      video_media_info.reset();
>+
>+  // Call GetStats for all media channels together on the worker thread in one
>+  // hop.
>+  worker_thread_->Invoke<void>(RTC_FROM_HERE, [&] {
>+    for (const auto& entry : voice_stats) {
>+      if (!entry.first->GetStats(entry.second.get())) {
>+        RTC_LOG(LS_WARNING) << "Failed to get voice stats.";
>+      }
>+    }
>+    for (const auto& entry : video_stats) {
>+      if (!entry.first->GetStats(entry.second.get())) {
>+        RTC_LOG(LS_WARNING) << "Failed to get video stats.";
>+      }
>+    }
>+  });
>+
>+  // Create the TrackMediaInfoMap for each transceiver stats object.
>+  for (auto& stats : transceiver_stats_infos) {
>+    auto transceiver = stats.transceiver;
>+    std::unique_ptr<cricket::VoiceMediaInfo> voice_media_info;
>+    std::unique_ptr<cricket::VideoMediaInfo> video_media_info;
>+    if (transceiver->channel()) {
>+      cricket::MediaType media_type = transceiver->media_type();
>+      if (media_type == cricket::MEDIA_TYPE_AUDIO) {
>+        auto* voice_channel =
>+            static_cast<cricket::VoiceChannel*>(transceiver->channel());
>+        RTC_DCHECK(voice_stats[voice_channel->media_channel()]);
>+        voice_media_info =
>+            std::move(voice_stats[voice_channel->media_channel()]);
>+      } else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
>+        auto* video_channel =
>+            static_cast<cricket::VideoChannel*>(transceiver->channel());
>+        RTC_DCHECK(video_stats[video_channel->media_channel()]);
>+        video_media_info =
>+            std::move(video_stats[video_channel->media_channel()]);
>+      }
>     }
>+    std::vector<rtc::scoped_refptr<RtpSenderInternal>> senders;
>+    for (auto sender : transceiver->senders()) {
>+      senders.push_back(sender->internal());
>+    }
>+    std::vector<rtc::scoped_refptr<RtpReceiverInternal>> receivers;
>+    for (auto receiver : transceiver->receivers()) {
>+      receivers.push_back(receiver->internal());
>+    }
>+    stats.track_media_info_map = absl::make_unique<TrackMediaInfoMap>(
>+        std::move(voice_media_info), std::move(video_media_info), senders,
>+        receivers);
>   }
>-  std::unique_ptr<TrackMediaInfoMap> track_media_info_map(
>-      new TrackMediaInfoMap(std::move(voice_media_info),
>-                            std::move(video_media_info),
>-                            pc_->GetSenders(),
>-                            pc_->GetReceivers()));
>-  return track_media_info_map;
>-}
>-
>-std::map<MediaStreamTrackInterface*, std::string>
>-RTCStatsCollector::PrepareTrackToID_s() const {
>-  // TODO(hta): Remove this method, and vector its callers via
>-  // senders / receivers instead.
>-  // It ignores tracks that are multiply connected to the PC.
>-  RTC_DCHECK(signaling_thread_->IsCurrent());
>-  std::map<MediaStreamTrackInterface*, std::string> track_to_id;
>-  for (auto sender : pc_->GetSenders()) {
>-    auto track = sender->track();
>-    if (track)
>-      track_to_id[track.get()] = track->id();
>+
>+  return transceiver_stats_infos;
>+}
>+
>+std::set<std::string> RTCStatsCollector::PrepareTransportNames_s() const {
>+  std::set<std::string> transport_names;
>+  for (const auto& transceiver : pc_->GetTransceiversInternal()) {
>+    if (transceiver->internal()->channel()) {
>+      transport_names.insert(
>+          transceiver->internal()->channel()->transport_name());
>+    }
>+  }
>+  if (pc_->rtp_data_channel()) {
>+    transport_names.insert(pc_->rtp_data_channel()->transport_name());
>   }
>-  for (auto receiver : pc_->GetReceivers()) {
>-    auto track = receiver->track();
>-    if (track)
>-      track_to_id[track.get()] = track->id();
>+  if (pc_->sctp_transport_name()) {
>+    transport_names.insert(*pc_->sctp_transport_name());
>   }
>-  return track_to_id;
>+  return transport_names;
> }
> 
> void RTCStatsCollector::OnDataChannelCreated(DataChannel* channel) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcstatscollector.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcstatscollector.h
>index d73a76443f1..8dd16e39a91 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcstatscollector.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcstatscollector.h
>@@ -17,34 +17,26 @@
> #include <string>
> #include <vector>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> #include "api/stats/rtcstats_objects.h"
> #include "api/stats/rtcstatscollectorcallback.h"
> #include "api/stats/rtcstatsreport.h"
> #include "call/call.h"
> #include "media/base/mediachannel.h"
> #include "pc/datachannel.h"
>+#include "pc/peerconnectioninternal.h"
> #include "pc/trackmediainfomap.h"
> #include "rtc_base/asyncinvoker.h"
> #include "rtc_base/refcount.h"
> #include "rtc_base/scoped_ref_ptr.h"
>-#include "rtc_base/sigslot.h"
> #include "rtc_base/sslidentity.h"
>+#include "rtc_base/third_party/sigslot/sigslot.h"
> #include "rtc_base/timeutils.h"
> 
>-namespace cricket {
>-class Candidate;
>-}  // namespace cricket
>-
>-namespace rtc {
>-class SSLCertificate;
>-}  // namespace rtc
>-
> namespace webrtc {
> 
>-class PeerConnection;
>-struct SessionStats;
>-struct ChannelNamePairs;
>+class RtpSenderInternal;
>+class RtpReceiverInternal;
> 
> // All public methods of the collector are to be called on the signaling thread.
> // Stats are gathered on the signaling, worker and network threads
>@@ -54,14 +46,25 @@ class RTCStatsCollector : public virtual rtc::RefCountInterface,
>                           public sigslot::has_slots<> {
>  public:
>   static rtc::scoped_refptr<RTCStatsCollector> Create(
>-      PeerConnection* pc,
>+      PeerConnectionInternal* pc,
>       int64_t cache_lifetime_us = 50 * rtc::kNumMicrosecsPerMillisec);
> 
>   // Gets a recent stats report. If there is a report cached that is still fresh
>   // it is returned, otherwise new stats are gathered and returned. A report is
>   // considered fresh for |cache_lifetime_| ms. const RTCStatsReports are safe
>   // to use across multiple threads and may be destructed on any thread.
>+  // If the optional selector argument is used, stats are filtered according to
>+  // stats selection algorithm before delivery.
>+  // https://w3c.github.io/webrtc-pc/#dfn-stats-selection-algorithm
>   void GetStatsReport(rtc::scoped_refptr<RTCStatsCollectorCallback> callback);
>+  // If |selector| is null the selection algorithm is still applied (interpreted
>+  // as: no RTP streams are sent by selector). The result is empty.
>+  void GetStatsReport(rtc::scoped_refptr<RtpSenderInternal> selector,
>+                      rtc::scoped_refptr<RTCStatsCollectorCallback> callback);
>+  // If |selector| is null the selection algorithm is still applied (interpreted
>+  // as: no RTP streams are received by selector). The result is empty.
>+  void GetStatsReport(rtc::scoped_refptr<RtpReceiverInternal> selector,
>+                      rtc::scoped_refptr<RTCStatsCollectorCallback> callback);
>   // Clears the cache's reference to the most recent stats report. Subsequently
>   // calling |GetStatsReport| guarantees fresh stats.
>   void ClearCachedStatsReport();
>@@ -71,7 +74,7 @@ class RTCStatsCollector : public virtual rtc::RefCountInterface,
>   void WaitForPendingRequest();
> 
>  protected:
>-  RTCStatsCollector(PeerConnection* pc, int64_t cache_lifetime_us);
>+  RTCStatsCollector(PeerConnectionInternal* pc, int64_t cache_lifetime_us);
>   ~RTCStatsCollector();
> 
>   // Stats gathering on a particular thread. Calls |AddPartialResults| before
>@@ -84,13 +87,73 @@ class RTCStatsCollector : public virtual rtc::RefCountInterface,
>       const rtc::scoped_refptr<RTCStatsReport>& partial_report);
> 
>  private:
>+  class RequestInfo {
>+   public:
>+    enum class FilterMode { kAll, kSenderSelector, kReceiverSelector };
>+
>+    // Constructs with FilterMode::kAll.
>+    explicit RequestInfo(
>+        rtc::scoped_refptr<RTCStatsCollectorCallback> callback);
>+    // Constructs with FilterMode::kSenderSelector. The selection algorithm is
>+    // applied even if |selector| is null, resulting in an empty report.
>+    RequestInfo(rtc::scoped_refptr<RtpSenderInternal> selector,
>+                rtc::scoped_refptr<RTCStatsCollectorCallback> callback);
>+    // Constructs with FilterMode::kReceiverSelector. The selection algorithm is
>+    // applied even if |selector| is null, resulting in an empty report.
>+    RequestInfo(rtc::scoped_refptr<RtpReceiverInternal> selector,
>+                rtc::scoped_refptr<RTCStatsCollectorCallback> callback);
>+
>+    FilterMode filter_mode() const { return filter_mode_; }
>+    rtc::scoped_refptr<RTCStatsCollectorCallback> callback() const {
>+      return callback_;
>+    }
>+    rtc::scoped_refptr<RtpSenderInternal> sender_selector() const {
>+      RTC_DCHECK(filter_mode_ == FilterMode::kSenderSelector);
>+      return sender_selector_;
>+    }
>+    rtc::scoped_refptr<RtpReceiverInternal> receiver_selector() const {
>+      RTC_DCHECK(filter_mode_ == FilterMode::kReceiverSelector);
>+      return receiver_selector_;
>+    }
>+
>+   private:
>+    RequestInfo(FilterMode filter_mode,
>+                rtc::scoped_refptr<RTCStatsCollectorCallback> callback,
>+                rtc::scoped_refptr<RtpSenderInternal> sender_selector,
>+                rtc::scoped_refptr<RtpReceiverInternal> receiver_selector);
>+
>+    FilterMode filter_mode_;
>+    rtc::scoped_refptr<RTCStatsCollectorCallback> callback_;
>+    rtc::scoped_refptr<RtpSenderInternal> sender_selector_;
>+    rtc::scoped_refptr<RtpReceiverInternal> receiver_selector_;
>+  };
>+
>+  void GetStatsReportInternal(RequestInfo request);
>+
>   struct CertificateStatsPair {
>     std::unique_ptr<rtc::SSLCertificateStats> local;
>     std::unique_ptr<rtc::SSLCertificateStats> remote;
>   };
> 
>+  // Structure for tracking stats about each RtpTransceiver managed by the
>+  // PeerConnection. This can either by a Plan B style or Unified Plan style
>+  // transceiver (i.e., can have 0 or many senders and receivers).
>+  // Some fields are copied from the RtpTransceiver/BaseChannel object so that
>+  // they can be accessed safely on threads other than the signaling thread.
>+  // If a BaseChannel is not available (e.g., if signaling has not started),
>+  // then |mid| and |transport_name| will be null.
>+  struct RtpTransceiverStatsInfo {
>+    rtc::scoped_refptr<RtpTransceiver> transceiver;
>+    cricket::MediaType media_type;
>+    absl::optional<std::string> mid;
>+    absl::optional<std::string> transport_name;
>+    std::unique_ptr<TrackMediaInfoMap> track_media_info_map;
>+  };
>+
>   void AddPartialResults_s(rtc::scoped_refptr<RTCStatsReport> partial_report);
>-  void DeliverCachedReport();
>+  void DeliverCachedReport(
>+      rtc::scoped_refptr<const RTCStatsReport> cached_report,
>+      std::vector<RequestInfo> requests);
> 
>   // Produces |RTCCertificateStats|.
>   void ProduceCertificateStats_n(
>@@ -99,41 +162,54 @@ class RTCStatsCollector : public virtual rtc::RefCountInterface,
>       RTCStatsReport* report) const;
>   // Produces |RTCCodecStats|.
>   void ProduceCodecStats_n(
>-      int64_t timestamp_us, const TrackMediaInfoMap& track_media_info_map,
>+      int64_t timestamp_us,
>+      const std::vector<RtpTransceiverStatsInfo>& transceiver_stats_infos,
>       RTCStatsReport* report) const;
>   // Produces |RTCDataChannelStats|.
>-  void ProduceDataChannelStats_s(
>-      int64_t timestamp_us, RTCStatsReport* report) const;
>+  void ProduceDataChannelStats_s(int64_t timestamp_us,
>+                                 RTCStatsReport* report) const;
>   // Produces |RTCIceCandidatePairStats| and |RTCIceCandidateStats|.
>   void ProduceIceCandidateAndPairStats_n(
>       int64_t timestamp_us,
>-      const SessionStats& session_stats,
>-      const cricket::VideoMediaInfo* video_media_info,
>+      const std::map<std::string, cricket::TransportStats>&
>+          transport_stats_by_name,
>       const Call::Stats& call_stats,
>       RTCStatsReport* report) const;
>-  // Produces |RTCMediaStreamStats| and |RTCMediaStreamTrackStats|.
>-  void ProduceMediaStreamAndTrackStats_s(
>-      int64_t timestamp_us, RTCStatsReport* report) const;
>+  // Produces |RTCMediaStreamStats|.
>+  void ProduceMediaStreamStats_s(int64_t timestamp_us,
>+                                 RTCStatsReport* report) const;
>+  // Produces |RTCMediaStreamTrackStats|.
>+  void ProduceMediaStreamTrackStats_s(int64_t timestamp_us,
>+                                      RTCStatsReport* report) const;
>   // Produces |RTCPeerConnectionStats|.
>-  void ProducePeerConnectionStats_s(
>-      int64_t timestamp_us, RTCStatsReport* report) const;
>+  void ProducePeerConnectionStats_s(int64_t timestamp_us,
>+                                    RTCStatsReport* report) const;
>   // Produces |RTCInboundRTPStreamStats| and |RTCOutboundRTPStreamStats|.
>-  void ProduceRTPStreamStats_n(int64_t timestamp_us,
>-                               const SessionStats& session_stats,
>-                               const ChannelNamePairs& channel_name_pairs,
>-                               const TrackMediaInfoMap& track_media_info_map,
>-                               RTCStatsReport* report) const;
>+  void ProduceRTPStreamStats_n(
>+      int64_t timestamp_us,
>+      const std::vector<RtpTransceiverStatsInfo>& transceiver_stats_infos,
>+      RTCStatsReport* report) const;
>+  void ProduceAudioRTPStreamStats_n(int64_t timestamp_us,
>+                                    const RtpTransceiverStatsInfo& stats,
>+                                    RTCStatsReport* report) const;
>+  void ProduceVideoRTPStreamStats_n(int64_t timestamp_us,
>+                                    const RtpTransceiverStatsInfo& stats,
>+                                    RTCStatsReport* report) const;
>   // Produces |RTCTransportStats|.
>   void ProduceTransportStats_n(
>-      int64_t timestamp_us, const SessionStats& session_stats,
>+      int64_t timestamp_us,
>+      const std::map<std::string, cricket::TransportStats>&
>+          transport_stats_by_name,
>       const std::map<std::string, CertificateStatsPair>& transport_cert_stats,
>       RTCStatsReport* report) const;
> 
>   // Helper function to stats-producing functions.
>   std::map<std::string, CertificateStatsPair>
>-  PrepareTransportCertificateStats_n(const SessionStats& session_stats) const;
>-  std::unique_ptr<TrackMediaInfoMap> PrepareTrackMediaInfoMap_s() const;
>-  std::map<MediaStreamTrackInterface*, std::string> PrepareTrackToID_s() const;
>+  PrepareTransportCertificateStats_n(
>+      const std::map<std::string, cricket::TransportStats>&
>+          transport_stats_by_name) const;
>+  std::vector<RtpTransceiverStatsInfo> PrepareTransceiverStatsInfos_s() const;
>+  std::set<std::string> PrepareTransportNames_s() const;
> 
>   // Slots for signals (sigslot) that are wired up to |pc_|.
>   void OnDataChannelCreated(DataChannel* channel);
>@@ -141,7 +217,7 @@ class RTCStatsCollector : public virtual rtc::RefCountInterface,
>   void OnDataChannelOpened(DataChannel* channel);
>   void OnDataChannelClosed(DataChannel* channel);
> 
>-  PeerConnection* const pc_;
>+  PeerConnectionInternal* const pc_;
>   rtc::Thread* const signaling_thread_;
>   rtc::Thread* const worker_thread_;
>   rtc::Thread* const network_thread_;
>@@ -150,15 +226,14 @@ class RTCStatsCollector : public virtual rtc::RefCountInterface,
>   int num_pending_partial_reports_;
>   int64_t partial_report_timestamp_us_;
>   rtc::scoped_refptr<RTCStatsReport> partial_report_;
>-  std::vector<rtc::scoped_refptr<RTCStatsCollectorCallback>> callbacks_;
>+  std::vector<RequestInfo> requests_;
> 
>   // Set in |GetStatsReport|, read in |ProducePartialResultsOnNetworkThread| and
>   // |ProducePartialResultsOnSignalingThread|, reset after work is complete. Not
>   // passed as arguments to avoid copies. This is thread safe - when we
>   // set/reset we know there are no pending stats requests in progress.
>-  std::unique_ptr<ChannelNamePairs> channel_name_pairs_;
>-  std::unique_ptr<TrackMediaInfoMap> track_media_info_map_;
>-  std::map<MediaStreamTrackInterface*, std::string> track_to_id_;
>+  std::vector<RtpTransceiverStatsInfo> transceiver_stats_infos_;
>+  std::set<std::string> transport_names_;
> 
>   Call::Stats call_stats_;
> 
>@@ -173,8 +248,7 @@ class RTCStatsCollector : public virtual rtc::RefCountInterface,
>   // Data recorded and maintained by the stats collector during its lifetime.
>   // Some stats are produced from this record instead of other components.
>   struct InternalRecord {
>-    InternalRecord() : data_channels_opened(0),
>-                       data_channels_closed(0) {}
>+    InternalRecord() : data_channels_opened(0), data_channels_closed(0) {}
> 
>     // The opened count goes up when a channel is fully opened and the closed
>     // count goes up if a previously opened channel has fully closed. The opened
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcstatscollector_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcstatscollector_unittest.cc
>index 8d4774a6145..3ece85f4c8c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcstatscollector_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcstatscollector_unittest.cc
>@@ -8,8 +8,6 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "pc/rtcstatscollector.h"
>-
> #include <initializer_list>
> #include <memory>
> #include <ostream>
>@@ -17,36 +15,32 @@
> #include <utility>
> #include <vector>
> 
>-#include "api/jsepsessiondescription.h"
>+#include "absl/memory/memory.h"
> #include "api/rtpparameters.h"
> #include "api/stats/rtcstats_objects.h"
> #include "api/stats/rtcstatsreport.h"
>-#include "api/test/mock_rtpreceiver.h"
>-#include "api/test/mock_rtpsender.h"
>-#include "media/base/fakemediaengine.h"
>-#include "media/base/test/mock_mediachannel.h"
>+#include "api/units/time_delta.h"
> #include "p2p/base/p2pconstants.h"
> #include "p2p/base/port.h"
> #include "pc/mediastream.h"
> #include "pc/mediastreamtrack.h"
>+#include "pc/rtcstatscollector.h"
>+#include "pc/test/fakepeerconnectionforstats.h"
> #include "pc/test/mock_datachannel.h"
>-#include "pc/test/mock_peerconnection.h"
>+#include "pc/test/mock_rtpreceiverinternal.h"
>+#include "pc/test/mock_rtpsenderinternal.h"
> #include "pc/test/rtcstatsobtainer.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/fakeclock.h"
> #include "rtc_base/fakesslidentity.h"
> #include "rtc_base/gunit.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/thread_checker.h"
>-#include "rtc_base/timedelta.h"
>+#include "rtc_base/stringutils.h"
> #include "rtc_base/timeutils.h"
> 
>-using testing::_;
>+using testing::AtLeast;
> using testing::Invoke;
> using testing::Return;
>-using testing::ReturnNull;
>-using testing::ReturnRef;
>-using testing::SetArgPointee;
> 
> namespace webrtc {
> 
>@@ -102,8 +96,6 @@ void PrintTo(const RTCTransportStats& stats, ::std::ostream* os) {
> namespace {
> 
> const int64_t kGetStatsReportTimeoutMs = 1000;
>-const bool kDefaultRtcpMuxRequired = true;
>-const bool kDefaultSrtpRequired = true;
> 
> struct CertificateInfo {
>   rtc::scoped_refptr<rtc::RTCCertificate> certificate;
>@@ -118,7 +110,7 @@ template <typename T>
> std::string IdForType(const RTCStatsReport* report) {
>   auto stats_of_my_type = report->RTCStatsReport::GetStatsOfType<T>();
>   // We cannot use ASSERT here, since we're within a function.
>-  EXPECT_EQ(1, stats_of_my_type.size())
>+  EXPECT_EQ(1U, stats_of_my_type.size())
>       << "Unexpected number of stats of this type";
>   if (stats_of_my_type.size() == 1) {
>     return stats_of_my_type[0]->id();
>@@ -135,37 +127,28 @@ std::unique_ptr<CertificateInfo> CreateFakeCertificateAndInfoFromDers(
>   info->ders = ders;
>   for (const std::string& der : ders) {
>     info->pems.push_back(rtc::SSLIdentity::DerToPem(
>-        "CERTIFICATE",
>-        reinterpret_cast<const unsigned char*>(der.c_str()),
>+        "CERTIFICATE", reinterpret_cast<const unsigned char*>(der.c_str()),
>         der.length()));
>   }
>   info->certificate =
>       rtc::RTCCertificate::Create(std::unique_ptr<rtc::FakeSSLIdentity>(
>-          new rtc::FakeSSLIdentity(rtc::FakeSSLCertificate(info->pems))));
>+          new rtc::FakeSSLIdentity(info->pems)));
>   // Strip header/footer and newline characters of PEM strings.
>   for (size_t i = 0; i < info->pems.size(); ++i) {
>-    rtc::replace_substrs("-----BEGIN CERTIFICATE-----", 27,
>-                         "", 0, &info->pems[i]);
>-    rtc::replace_substrs("-----END CERTIFICATE-----", 25,
>-                         "", 0, &info->pems[i]);
>-    rtc::replace_substrs("\n", 1,
>-                         "", 0, &info->pems[i]);
>+    rtc::replace_substrs("-----BEGIN CERTIFICATE-----", 27, "", 0,
>+                         &info->pems[i]);
>+    rtc::replace_substrs("-----END CERTIFICATE-----", 25, "", 0,
>+                         &info->pems[i]);
>+    rtc::replace_substrs("\n", 1, "", 0, &info->pems[i]);
>   }
>-  // Fingerprint of leaf certificate.
>-  std::unique_ptr<rtc::SSLFingerprint> fp(
>-      rtc::SSLFingerprint::Create("sha-1",
>-                                  &info->certificate->ssl_certificate()));
>-  EXPECT_TRUE(fp);
>-  info->fingerprints.push_back(fp->GetRfc4572Fingerprint());
>-  // Fingerprints of the rest of the chain.
>-  std::unique_ptr<rtc::SSLCertChain> chain =
>-      info->certificate->ssl_certificate().GetChain();
>-  if (chain) {
>-    for (size_t i = 0; i < chain->GetSize(); i++) {
>-      fp.reset(rtc::SSLFingerprint::Create("sha-1", &chain->Get(i)));
>-      EXPECT_TRUE(fp);
>-      info->fingerprints.push_back(fp->GetRfc4572Fingerprint());
>-    }
>+  // Fingerprints for the whole certificate chain, starting with leaf
>+  // certificate.
>+  const rtc::SSLCertChain& chain = info->certificate->ssl_cert_chain();
>+  std::unique_ptr<rtc::SSLFingerprint> fp;
>+  for (size_t i = 0; i < chain.GetSize(); i++) {
>+    fp.reset(rtc::SSLFingerprint::Create("sha-1", &chain.Get(i)));
>+    EXPECT_TRUE(fp);
>+    info->fingerprints.push_back(fp->GetRfc4572Fingerprint());
>   }
>   EXPECT_EQ(info->ders.size(), info->fingerprints.size());
>   return info;
>@@ -187,8 +170,7 @@ std::unique_ptr<cricket::Candidate> CreateFakeCandidate(
>   return candidate;
> }
> 
>-class FakeAudioTrackForStats
>-    : public MediaStreamTrack<AudioTrackInterface> {
>+class FakeAudioTrackForStats : public MediaStreamTrack<AudioTrackInterface> {
>  public:
>   static rtc::scoped_refptr<FakeAudioTrackForStats> Create(
>       const std::string& id,
>@@ -214,8 +196,7 @@ class FakeAudioTrackForStats
>   }
> };
> 
>-class FakeVideoTrackForStats
>-    : public MediaStreamTrack<VideoTrackInterface> {
>+class FakeVideoTrackForStats : public MediaStreamTrack<VideoTrackInterface> {
>  public:
>   static rtc::scoped_refptr<FakeVideoTrackForStats> Create(
>       const std::string& id,
>@@ -252,112 +233,104 @@ rtc::scoped_refptr<MediaStreamTrackInterface> CreateFakeTrack(
>   }
> }
> 
>-rtc::scoped_refptr<MockRtpSender> CreateMockSender(
>+rtc::scoped_refptr<MockRtpSenderInternal> CreateMockSender(
>     const rtc::scoped_refptr<MediaStreamTrackInterface>& track,
>     uint32_t ssrc,
>     int attachment_id,
>     std::vector<std::string> local_stream_ids) {
>-  rtc::scoped_refptr<MockRtpSender> sender(
>-      new rtc::RefCountedObject<MockRtpSender>());
>+  rtc::scoped_refptr<MockRtpSenderInternal> sender(
>+      new rtc::RefCountedObject<MockRtpSenderInternal>());
>   EXPECT_CALL(*sender, track()).WillRepeatedly(Return(track));
>   EXPECT_CALL(*sender, ssrc()).WillRepeatedly(Return(ssrc));
>-  EXPECT_CALL(*sender, media_type()).WillRepeatedly(Return(
>-      track->kind() == MediaStreamTrackInterface::kAudioKind
>-          ? cricket::MEDIA_TYPE_AUDIO : cricket::MEDIA_TYPE_VIDEO));
>-  EXPECT_CALL(*sender, GetParameters()).WillRepeatedly(Invoke(
>-    [ssrc]() {
>-      RtpParameters params;
>-      params.encodings.push_back(RtpEncodingParameters());
>-      params.encodings[0].ssrc = ssrc;
>-      return params;
>-    }));
>+  EXPECT_CALL(*sender, media_type())
>+      .WillRepeatedly(
>+          Return(track->kind() == MediaStreamTrackInterface::kAudioKind
>+                     ? cricket::MEDIA_TYPE_AUDIO
>+                     : cricket::MEDIA_TYPE_VIDEO));
>+  EXPECT_CALL(*sender, GetParameters()).WillRepeatedly(Invoke([ssrc]() {
>+    RtpParameters params;
>+    params.encodings.push_back(RtpEncodingParameters());
>+    params.encodings[0].ssrc = ssrc;
>+    return params;
>+  }));
>   EXPECT_CALL(*sender, AttachmentId()).WillRepeatedly(Return(attachment_id));
>   EXPECT_CALL(*sender, stream_ids()).WillRepeatedly(Return(local_stream_ids));
>   return sender;
> }
> 
>-rtc::scoped_refptr<MockRtpReceiver> CreateMockReceiver(
>+rtc::scoped_refptr<MockRtpReceiverInternal> CreateMockReceiver(
>     const rtc::scoped_refptr<MediaStreamTrackInterface>& track,
>     uint32_t ssrc,
>     int attachment_id) {
>-  rtc::scoped_refptr<MockRtpReceiver> receiver(
>-      new rtc::RefCountedObject<MockRtpReceiver>());
>+  rtc::scoped_refptr<MockRtpReceiverInternal> receiver(
>+      new rtc::RefCountedObject<MockRtpReceiverInternal>());
>   EXPECT_CALL(*receiver, track()).WillRepeatedly(Return(track));
>   EXPECT_CALL(*receiver, streams())
>       .WillRepeatedly(
>           Return(std::vector<rtc::scoped_refptr<MediaStreamInterface>>({})));
> 
>-  EXPECT_CALL(*receiver, media_type()).WillRepeatedly(Return(
>-      track->kind() == MediaStreamTrackInterface::kAudioKind
>-          ? cricket::MEDIA_TYPE_AUDIO : cricket::MEDIA_TYPE_VIDEO));
>-  EXPECT_CALL(*receiver, GetParameters()).WillRepeatedly(Invoke(
>-    [ssrc]() {
>-      RtpParameters params;
>-      params.encodings.push_back(RtpEncodingParameters());
>-      params.encodings[0].ssrc = ssrc;
>-      return params;
>-    }));
>+  EXPECT_CALL(*receiver, media_type())
>+      .WillRepeatedly(
>+          Return(track->kind() == MediaStreamTrackInterface::kAudioKind
>+                     ? cricket::MEDIA_TYPE_AUDIO
>+                     : cricket::MEDIA_TYPE_VIDEO));
>+  EXPECT_CALL(*receiver, GetParameters()).WillRepeatedly(Invoke([ssrc]() {
>+    RtpParameters params;
>+    params.encodings.push_back(RtpEncodingParameters());
>+    params.encodings[0].ssrc = ssrc;
>+    return params;
>+  }));
>   EXPECT_CALL(*receiver, AttachmentId()).WillRepeatedly(Return(attachment_id));
>   return receiver;
> }
> 
>-class RTCStatsCollectorTestHelper : public SetSessionDescriptionObserver {
>+class RTCStatsCollectorWrapper {
>  public:
>-  RTCStatsCollectorTestHelper()
>-      : worker_thread_(rtc::Thread::Current()),
>-        network_thread_(rtc::Thread::Current()),
>-        signaling_thread_(rtc::Thread::Current()),
>-        media_engine_(new cricket::FakeMediaEngine()),
>-        pc_factory_(new FakePeerConnectionFactory(
>-            std::unique_ptr<cricket::MediaEngineInterface>(media_engine_))),
>-        pc_(pc_factory_) {
>-    // Default return values for mocks.
>-    EXPECT_CALL(pc_, GetCallStats()).WillRepeatedly(Return(Call::Stats()));
>-    EXPECT_CALL(pc_, local_streams()).WillRepeatedly(Return(nullptr));
>-    EXPECT_CALL(pc_, remote_streams()).WillRepeatedly(Return(nullptr));
>-    EXPECT_CALL(pc_, GetSenders()).WillRepeatedly(Return(
>-        std::vector<rtc::scoped_refptr<RtpSenderInterface>>()));
>-    EXPECT_CALL(pc_, GetReceivers()).WillRepeatedly(Return(
>-        std::vector<rtc::scoped_refptr<RtpReceiverInterface>>()));
>-    EXPECT_CALL(pc_, sctp_data_channels()).WillRepeatedly(
>-        ReturnRef(data_channels_));
>-    EXPECT_CALL(pc_, video_channel()).WillRepeatedly(ReturnNull());
>-    EXPECT_CALL(pc_, voice_channel()).WillRepeatedly(ReturnNull());
>-    EXPECT_CALL(pc_, GetSessionStats(_)).WillRepeatedly(ReturnNull());
>-    EXPECT_CALL(pc_, GetLocalCertificate(_, _)).WillRepeatedly(Return(false));
>-    EXPECT_CALL(pc_, GetRemoteSSLCertificate_ReturnsRawPointer(_))
>-        .WillRepeatedly(Return(nullptr));
>+  explicit RTCStatsCollectorWrapper(
>+      rtc::scoped_refptr<FakePeerConnectionForStats> pc)
>+      : pc_(pc),
>+        stats_collector_(
>+            RTCStatsCollector::Create(pc, 50 * rtc::kNumMicrosecsPerMillisec)) {
>   }
> 
>-  rtc::ScopedFakeClock& fake_clock() { return fake_clock_; }
>-  rtc::Thread* worker_thread() { return worker_thread_; }
>-  rtc::Thread* network_thread() { return network_thread_; }
>-  rtc::Thread* signaling_thread() { return signaling_thread_; }
>-  cricket::FakeMediaEngine* media_engine() { return media_engine_; }
>-  MockPeerConnection& pc() { return pc_; }
>-  std::vector<rtc::scoped_refptr<DataChannel>>& data_channels() {
>-    return data_channels_;
>+  rtc::scoped_refptr<RTCStatsCollector> stats_collector() {
>+    return stats_collector_;
>   }
> 
>-  // SetSessionDescriptionObserver overrides.
>-  void OnSuccess() override {}
>-  void OnFailure(const std::string& error) override {
>-    RTC_NOTREACHED() << error;
>+  rtc::scoped_refptr<const RTCStatsReport> GetStatsReport() {
>+    rtc::scoped_refptr<RTCStatsObtainer> callback = RTCStatsObtainer::Create();
>+    stats_collector_->GetStatsReport(callback);
>+    return WaitForReport(callback);
>   }
> 
>-  void SetupLocalTrackAndSender(cricket::MediaType media_type,
>-                                const std::string& track_id,
>-                                uint32_t ssrc,
>-                                bool add_stream) {
>-    rtc::scoped_refptr<StreamCollection> local_streams =
>-        StreamCollection::Create();
>-    EXPECT_CALL(pc_, local_streams())
>-        .WillRepeatedly(Return(local_streams));
>+  rtc::scoped_refptr<const RTCStatsReport> GetStatsReportWithSenderSelector(
>+      rtc::scoped_refptr<RtpSenderInternal> selector) {
>+    rtc::scoped_refptr<RTCStatsObtainer> callback = RTCStatsObtainer::Create();
>+    stats_collector_->GetStatsReport(selector, callback);
>+    return WaitForReport(callback);
>+  }
> 
>+  rtc::scoped_refptr<const RTCStatsReport> GetStatsReportWithReceiverSelector(
>+      rtc::scoped_refptr<RtpReceiverInternal> selector) {
>+    rtc::scoped_refptr<RTCStatsObtainer> callback = RTCStatsObtainer::Create();
>+    stats_collector_->GetStatsReport(selector, callback);
>+    return WaitForReport(callback);
>+  }
>+
>+  rtc::scoped_refptr<const RTCStatsReport> GetFreshStatsReport() {
>+    stats_collector_->ClearCachedStatsReport();
>+    return GetStatsReport();
>+  }
>+
>+  rtc::scoped_refptr<MockRtpSenderInternal> SetupLocalTrackAndSender(
>+      cricket::MediaType media_type,
>+      const std::string& track_id,
>+      uint32_t ssrc,
>+      bool add_stream) {
>     rtc::scoped_refptr<MediaStream> local_stream;
>     if (add_stream) {
>-      local_stream = MediaStream::Create("LocalStreamLabel");
>-      local_streams->AddStream(local_stream);
>+      local_stream = MediaStream::Create("LocalStreamId");
>+      pc_->mutable_local_streams()->AddStream(local_stream);
>     }
> 
>     rtc::scoped_refptr<MediaStreamTrackInterface> track;
>@@ -375,24 +348,20 @@ class RTCStatsCollectorTestHelper : public SetSessionDescriptionObserver {
>       }
>     }
> 
>-    rtc::scoped_refptr<MockRtpSender> sender =
>+    rtc::scoped_refptr<MockRtpSenderInternal> sender =
>         CreateMockSender(track, ssrc, 50, {});
>-    EXPECT_CALL(pc_, GetSenders()).WillRepeatedly(Return(
>-        std::vector<rtc::scoped_refptr<RtpSenderInterface>>({
>-            rtc::scoped_refptr<RtpSenderInterface>(sender.get()) })));
>+    pc_->AddSender(sender);
>+    return sender;
>   }
> 
>-  void SetupRemoteTrackAndReceiver(cricket::MediaType media_type,
>-                                   const std::string& track_id,
>-                                   uint32_t ssrc) {
>-    rtc::scoped_refptr<StreamCollection> remote_streams =
>-        StreamCollection::Create();
>-    EXPECT_CALL(pc_, remote_streams())
>-        .WillRepeatedly(Return(remote_streams));
>-
>+  rtc::scoped_refptr<MockRtpReceiverInternal> SetupRemoteTrackAndReceiver(
>+      cricket::MediaType media_type,
>+      const std::string& track_id,
>+      const std::string& stream_id,
>+      uint32_t ssrc) {
>     rtc::scoped_refptr<MediaStream> remote_stream =
>-        MediaStream::Create("RemoteStreamLabel");
>-    remote_streams->AddStream(remote_stream);
>+        MediaStream::Create(stream_id);
>+    pc_->mutable_remote_streams()->AddStream(remote_stream);
> 
>     rtc::scoped_refptr<MediaStreamTrackInterface> track;
>     if (media_type == cricket::MEDIA_TYPE_AUDIO) {
>@@ -405,15 +374,14 @@ class RTCStatsCollectorTestHelper : public SetSessionDescriptionObserver {
>       remote_stream->AddTrack(static_cast<VideoTrackInterface*>(track.get()));
>     }
> 
>-    rtc::scoped_refptr<MockRtpReceiver> receiver =
>+    rtc::scoped_refptr<MockRtpReceiverInternal> receiver =
>         CreateMockReceiver(track, ssrc, 62);
>     EXPECT_CALL(*receiver, streams())
>         .WillRepeatedly(
>             Return(std::vector<rtc::scoped_refptr<MediaStreamInterface>>(
>                 {remote_stream})));
>-    EXPECT_CALL(pc_, GetReceivers()).WillRepeatedly(Return(
>-        std::vector<rtc::scoped_refptr<RtpReceiverInterface>>({
>-            rtc::scoped_refptr<RtpReceiverInterface>(receiver.get()) })));
>+    pc_->AddReceiver(receiver);
>+    return receiver;
>   }
> 
>   // Attaches tracks to peer connections by configuring RTP senders and RTP
>@@ -436,10 +404,9 @@ class RTCStatsCollectorTestHelper : public SetSessionDescriptionObserver {
>           remote_video_track_info_pairs,
>       std::vector<std::string> local_stream_ids,
>       std::vector<rtc::scoped_refptr<MediaStreamInterface>> remote_streams) {
>-    voice_media_info_.reset(new cricket::VoiceMediaInfo());
>-    video_media_info_.reset(new cricket::VideoMediaInfo());
>-    rtp_senders_.clear();
>-    rtp_receivers_.clear();
>+    cricket::VoiceMediaInfo voice_media_info;
>+    cricket::VideoMediaInfo video_media_info;
>+
>     // Local audio tracks and voice sender infos
>     int attachment_id = 147;
>     for (auto& pair : local_audio_track_info_pairs) {
>@@ -448,14 +415,14 @@ class RTCStatsCollectorTestHelper : public SetSessionDescriptionObserver {
>       RTC_DCHECK_EQ(local_audio_track->kind(),
>                     MediaStreamTrackInterface::kAudioKind);
> 
>-      voice_media_info_->senders.push_back(voice_sender_info);
>-      rtc::scoped_refptr<MockRtpSender> rtp_sender = CreateMockSender(
>+      voice_media_info.senders.push_back(voice_sender_info);
>+      rtc::scoped_refptr<MockRtpSenderInternal> rtp_sender = CreateMockSender(
>           rtc::scoped_refptr<MediaStreamTrackInterface>(local_audio_track),
>           voice_sender_info.local_stats[0].ssrc, attachment_id++,
>           local_stream_ids);
>-      rtp_senders_.push_back(rtc::scoped_refptr<RtpSenderInterface>(
>-          rtp_sender.get()));
>+      pc_->AddSender(rtp_sender);
>     }
>+
>     // Remote audio tracks and voice receiver infos
>     attachment_id = 181;
>     for (auto& pair : remote_audio_track_info_pairs) {
>@@ -464,15 +431,16 @@ class RTCStatsCollectorTestHelper : public SetSessionDescriptionObserver {
>       RTC_DCHECK_EQ(remote_audio_track->kind(),
>                     MediaStreamTrackInterface::kAudioKind);
> 
>-      voice_media_info_->receivers.push_back(voice_receiver_info);
>-      rtc::scoped_refptr<MockRtpReceiver> rtp_receiver = CreateMockReceiver(
>-          rtc::scoped_refptr<MediaStreamTrackInterface>(remote_audio_track),
>-          voice_receiver_info.local_stats[0].ssrc, attachment_id++);
>+      voice_media_info.receivers.push_back(voice_receiver_info);
>+      rtc::scoped_refptr<MockRtpReceiverInternal> rtp_receiver =
>+          CreateMockReceiver(
>+              rtc::scoped_refptr<MediaStreamTrackInterface>(remote_audio_track),
>+              voice_receiver_info.local_stats[0].ssrc, attachment_id++);
>       EXPECT_CALL(*rtp_receiver, streams())
>           .WillRepeatedly(Return(remote_streams));
>-      rtp_receivers_.push_back(rtc::scoped_refptr<RtpReceiverInterface>(
>-          rtp_receiver.get()));
>+      pc_->AddReceiver(rtp_receiver);
>     }
>+
>     // Local video tracks and video sender infos
>     attachment_id = 151;
>     for (auto& pair : local_video_track_info_pairs) {
>@@ -481,14 +449,14 @@ class RTCStatsCollectorTestHelper : public SetSessionDescriptionObserver {
>       RTC_DCHECK_EQ(local_video_track->kind(),
>                     MediaStreamTrackInterface::kVideoKind);
> 
>-      video_media_info_->senders.push_back(video_sender_info);
>-      rtc::scoped_refptr<MockRtpSender> rtp_sender = CreateMockSender(
>+      video_media_info.senders.push_back(video_sender_info);
>+      rtc::scoped_refptr<MockRtpSenderInternal> rtp_sender = CreateMockSender(
>           rtc::scoped_refptr<MediaStreamTrackInterface>(local_video_track),
>           video_sender_info.local_stats[0].ssrc, attachment_id++,
>           local_stream_ids);
>-      rtp_senders_.push_back(rtc::scoped_refptr<RtpSenderInterface>(
>-          rtp_sender.get()));
>+      pc_->AddSender(rtp_sender);
>     }
>+
>     // Remote video tracks and video receiver infos
>     attachment_id = 191;
>     for (auto& pair : remote_video_track_info_pairs) {
>@@ -497,174 +465,26 @@ class RTCStatsCollectorTestHelper : public SetSessionDescriptionObserver {
>       RTC_DCHECK_EQ(remote_video_track->kind(),
>                     MediaStreamTrackInterface::kVideoKind);
> 
>-      video_media_info_->receivers.push_back(video_receiver_info);
>-      rtc::scoped_refptr<MockRtpReceiver> rtp_receiver = CreateMockReceiver(
>-          rtc::scoped_refptr<MediaStreamTrackInterface>(remote_video_track),
>-          video_receiver_info.local_stats[0].ssrc, attachment_id++);
>+      video_media_info.receivers.push_back(video_receiver_info);
>+      rtc::scoped_refptr<MockRtpReceiverInternal> rtp_receiver =
>+          CreateMockReceiver(
>+              rtc::scoped_refptr<MediaStreamTrackInterface>(remote_video_track),
>+              video_receiver_info.local_stats[0].ssrc, attachment_id++);
>       EXPECT_CALL(*rtp_receiver, streams())
>           .WillRepeatedly(Return(remote_streams));
>-      rtp_receivers_.push_back(rtc::scoped_refptr<RtpReceiverInterface>(
>-          rtp_receiver.get()));
>-    }
>-    EXPECT_CALL(pc_, GetSenders()).WillRepeatedly(Return(rtp_senders_));
>-    EXPECT_CALL(pc_, GetReceivers()).WillRepeatedly(Return(rtp_receivers_));
>-
>-    auto* voice_media_channel = new MockVoiceMediaChannel();
>-    voice_channel_.reset(new cricket::VoiceChannel(
>-        worker_thread_, network_thread_, nullptr, media_engine_,
>-        rtc::WrapUnique(voice_media_channel), "VoiceContentName",
>-        kDefaultRtcpMuxRequired, kDefaultSrtpRequired));
>-    EXPECT_CALL(pc_, voice_channel())
>-        .WillRepeatedly(Return(voice_channel_.get()));
>-    EXPECT_CALL(*voice_media_channel, GetStats(_))
>-        .WillOnce(DoAll(SetArgPointee<0>(*voice_media_info_), Return(true)));
>-
>-    auto* video_media_channel = new MockVideoMediaChannel();
>-    video_channel_.reset(new cricket::VideoChannel(
>-        worker_thread_, network_thread_, nullptr,
>-        rtc::WrapUnique(video_media_channel), "VideoContentName",
>-        kDefaultRtcpMuxRequired, kDefaultSrtpRequired));
>-    EXPECT_CALL(pc_, video_channel())
>-        .WillRepeatedly(Return(video_channel_.get()));
>-    EXPECT_CALL(*video_media_channel, GetStats(_))
>-        .WillOnce(DoAll(SetArgPointee<0>(*video_media_info_), Return(true)));
>-  }
>-
>- private:
>-  rtc::ScopedFakeClock fake_clock_;
>-  RtcEventLogNullImpl event_log_;
>-  rtc::Thread* const worker_thread_;
>-  rtc::Thread* const network_thread_;
>-  rtc::Thread* const signaling_thread_;
>-  // |media_engine_| is actually owned by |pc_factory_|.
>-  cricket::FakeMediaEngine* media_engine_;
>-  rtc::scoped_refptr<FakePeerConnectionFactory> pc_factory_;
>-  MockPeerConnection pc_;
>-
>-  std::vector<rtc::scoped_refptr<DataChannel>> data_channels_;
>-  std::unique_ptr<cricket::VoiceChannel> voice_channel_;
>-  std::unique_ptr<cricket::VideoChannel> video_channel_;
>-  std::unique_ptr<cricket::VoiceMediaInfo> voice_media_info_;
>-  std::unique_ptr<cricket::VideoMediaInfo> video_media_info_;
>-  std::vector<rtc::scoped_refptr<RtpSenderInterface>> rtp_senders_;
>-  std::vector<rtc::scoped_refptr<RtpReceiverInterface>> rtp_receivers_;
>-};
>-
>-class RTCTestStats : public RTCStats {
>- public:
>-  WEBRTC_RTCSTATS_DECL();
>-
>-  RTCTestStats(const std::string& id, int64_t timestamp_us)
>-      : RTCStats(id, timestamp_us),
>-        dummy_stat("dummyStat") {}
>-
>-  RTCStatsMember<int32_t> dummy_stat;
>-};
>-
>-WEBRTC_RTCSTATS_IMPL(RTCTestStats, RTCStats, "test-stats",
>-    &dummy_stat);
>-
>-// Overrides the stats collection to verify thread usage and that the resulting
>-// partial reports are merged.
>-class FakeRTCStatsCollector : public RTCStatsCollector,
>-                              public RTCStatsCollectorCallback {
>- public:
>-  static rtc::scoped_refptr<FakeRTCStatsCollector> Create(
>-      PeerConnection* pc,
>-      int64_t cache_lifetime_us) {
>-    return rtc::scoped_refptr<FakeRTCStatsCollector>(
>-        new rtc::RefCountedObject<FakeRTCStatsCollector>(
>-            pc, cache_lifetime_us));
>-  }
>-
>-  // RTCStatsCollectorCallback implementation.
>-  void OnStatsDelivered(
>-      const rtc::scoped_refptr<const RTCStatsReport>& report) override {
>-    EXPECT_TRUE(signaling_thread_->IsCurrent());
>-    rtc::CritScope cs(&lock_);
>-    delivered_report_ = report;
>-  }
>-
>-  void VerifyThreadUsageAndResultsMerging() {
>-    GetStatsReport(rtc::scoped_refptr<RTCStatsCollectorCallback>(this));
>-    EXPECT_TRUE_WAIT(HasVerifiedResults(), kGetStatsReportTimeoutMs);
>-  }
>-
>-  bool HasVerifiedResults() {
>-    EXPECT_TRUE(signaling_thread_->IsCurrent());
>-    rtc::CritScope cs(&lock_);
>-    if (!delivered_report_)
>-      return false;
>-    EXPECT_EQ(produced_on_signaling_thread_, 1);
>-    EXPECT_EQ(produced_on_network_thread_, 1);
>-
>-    EXPECT_TRUE(delivered_report_->Get("SignalingThreadStats"));
>-    EXPECT_TRUE(delivered_report_->Get("NetworkThreadStats"));
>-
>-    produced_on_signaling_thread_ = 0;
>-    produced_on_network_thread_ = 0;
>-    delivered_report_ = nullptr;
>-    return true;
>-  }
>-
>- protected:
>-  FakeRTCStatsCollector(PeerConnection* pc, int64_t cache_lifetime)
>-      : RTCStatsCollector(pc, cache_lifetime),
>-        signaling_thread_(pc->signaling_thread()),
>-        worker_thread_(pc->worker_thread()),
>-        network_thread_(pc->network_thread()) {}
>-
>-  void ProducePartialResultsOnSignalingThread(int64_t timestamp_us) override {
>-    EXPECT_TRUE(signaling_thread_->IsCurrent());
>-    {
>-      rtc::CritScope cs(&lock_);
>-      EXPECT_FALSE(delivered_report_);
>-      ++produced_on_signaling_thread_;
>+      pc_->AddReceiver(rtp_receiver);
>     }
> 
>-    rtc::scoped_refptr<RTCStatsReport> signaling_report =
>-        RTCStatsReport::Create(0);
>-    signaling_report->AddStats(std::unique_ptr<const RTCStats>(
>-        new RTCTestStats("SignalingThreadStats", timestamp_us)));
>-    AddPartialResults(signaling_report);
>-  }
>-  void ProducePartialResultsOnNetworkThread(int64_t timestamp_us) override {
>-    EXPECT_TRUE(network_thread_->IsCurrent());
>-    {
>-      rtc::CritScope cs(&lock_);
>-      EXPECT_FALSE(delivered_report_);
>-      ++produced_on_network_thread_;
>-    }
>+    auto* voice_media_channel = pc_->AddVoiceChannel("audio", "transport");
>+    voice_media_channel->SetStats(voice_media_info);
> 
>-    rtc::scoped_refptr<RTCStatsReport> network_report =
>-        RTCStatsReport::Create(0);
>-    network_report->AddStats(std::unique_ptr<const RTCStats>(
>-        new RTCTestStats("NetworkThreadStats", timestamp_us)));
>-    AddPartialResults(network_report);
>+    auto* video_media_channel = pc_->AddVideoChannel("video", "transport");
>+    video_media_channel->SetStats(video_media_info);
>   }
> 
>  private:
>-  rtc::Thread* const signaling_thread_;
>-  rtc::Thread* const worker_thread_;
>-  rtc::Thread* const network_thread_;
>-
>-  rtc::CriticalSection lock_;
>-  rtc::scoped_refptr<const RTCStatsReport> delivered_report_;
>-  int produced_on_signaling_thread_ = 0;
>-  int produced_on_network_thread_ = 0;
>-};
>-
>-class RTCStatsCollectorTest : public testing::Test {
>- public:
>-  RTCStatsCollectorTest()
>-    : test_(new rtc::RefCountedObject<RTCStatsCollectorTestHelper>()),
>-      collector_(RTCStatsCollector::Create(
>-          &test_->pc(), 50 * rtc::kNumMicrosecsPerMillisec)) {
>-  }
>-
>-  rtc::scoped_refptr<const RTCStatsReport> GetStatsReport() {
>-    rtc::scoped_refptr<RTCStatsObtainer> callback = RTCStatsObtainer::Create();
>-    collector_->GetStatsReport(callback);
>+  rtc::scoped_refptr<const RTCStatsReport> WaitForReport(
>+      rtc::scoped_refptr<RTCStatsObtainer> callback) {
>     EXPECT_TRUE_WAIT(callback->report(), kGetStatsReportTimeoutMs);
>     int64_t after = rtc::TimeUTCMicros();
>     for (const RTCStats& stats : *callback->report()) {
>@@ -673,13 +493,22 @@ class RTCStatsCollectorTest : public testing::Test {
>     return callback->report();
>   }
> 
>+  rtc::scoped_refptr<FakePeerConnectionForStats> pc_;
>+  rtc::scoped_refptr<RTCStatsCollector> stats_collector_;
>+};
>+
>+class RTCStatsCollectorTest : public testing::Test {
>+ public:
>+  RTCStatsCollectorTest()
>+      : pc_(new rtc::RefCountedObject<FakePeerConnectionForStats>()),
>+        stats_(new RTCStatsCollectorWrapper(pc_)) {}
>+
>   void ExpectReportContainsCertificateInfo(
>       const rtc::scoped_refptr<const RTCStatsReport>& report,
>       const CertificateInfo& certinfo) {
>     for (size_t i = 0; i < certinfo.fingerprints.size(); ++i) {
>       RTCCertificateStats expected_certificate_stats(
>-          "RTCCertificate_" + certinfo.fingerprints[i],
>-          report->timestamp_us());
>+          "RTCCertificate_" + certinfo.fingerprints[i], report->timestamp_us());
>       expected_certificate_stats.fingerprint = certinfo.fingerprints[i];
>       expected_certificate_stats.fingerprint_algorithm = "sha-1";
>       expected_certificate_stats.base64_certificate = certinfo.pems[i];
>@@ -689,61 +518,168 @@ class RTCStatsCollectorTest : public testing::Test {
>       }
>       ASSERT_TRUE(report->Get(expected_certificate_stats.id()));
>       EXPECT_EQ(expected_certificate_stats,
>-                report->Get(expected_certificate_stats.id())->cast_to<
>-                      RTCCertificateStats>());
>+                report->Get(expected_certificate_stats.id())
>+                    ->cast_to<RTCCertificateStats>());
>     }
>   }
> 
>+  struct ExampleStatsGraph {
>+    rtc::scoped_refptr<RtpSenderInternal> sender;
>+    rtc::scoped_refptr<RtpReceiverInternal> receiver;
>+
>+    rtc::scoped_refptr<const RTCStatsReport> full_report;
>+    std::string send_codec_id;
>+    std::string recv_codec_id;
>+    std::string outbound_rtp_id;
>+    std::string inbound_rtp_id;
>+    std::string transport_id;
>+    std::string sender_track_id;
>+    std::string receiver_track_id;
>+    std::string remote_stream_id;
>+    std::string peer_connection_id;
>+  };
>+
>+  // Sets up the example stats graph (see ASCII art below) used for testing the
>+  // stats selection algorithm,
>+  // https://w3c.github.io/webrtc-pc/#dfn-stats-selection-algorithm.
>+  // These tests test the integration of the stats traversal algorithm inside of
>+  // RTCStatsCollector. See rtcstatstraveral_unittest.cc for more stats
>+  // traversal tests.
>+  ExampleStatsGraph SetupExampleStatsGraphForSelectorTests() {
>+    ExampleStatsGraph graph;
>+
>+    // codec (send)
>+    graph.send_codec_id = "RTCCodec_VideoMid_Outbound_1";
>+    cricket::VideoMediaInfo video_media_info;
>+    RtpCodecParameters send_codec;
>+    send_codec.payload_type = 1;
>+    send_codec.clock_rate = 0;
>+    video_media_info.send_codecs.insert(
>+        std::make_pair(send_codec.payload_type, send_codec));
>+    // codec (recv)
>+    graph.recv_codec_id = "RTCCodec_VideoMid_Inbound_2";
>+    RtpCodecParameters recv_codec;
>+    recv_codec.payload_type = 2;
>+    recv_codec.clock_rate = 0;
>+    video_media_info.receive_codecs.insert(
>+        std::make_pair(recv_codec.payload_type, recv_codec));
>+    // outbound-rtp
>+    graph.outbound_rtp_id = "RTCOutboundRTPVideoStream_3";
>+    video_media_info.senders.push_back(cricket::VideoSenderInfo());
>+    video_media_info.senders[0].local_stats.push_back(
>+        cricket::SsrcSenderInfo());
>+    video_media_info.senders[0].local_stats[0].ssrc = 3;
>+    video_media_info.senders[0].codec_payload_type = send_codec.payload_type;
>+    // inbound-rtp
>+    graph.inbound_rtp_id = "RTCInboundRTPVideoStream_4";
>+    video_media_info.receivers.push_back(cricket::VideoReceiverInfo());
>+    video_media_info.receivers[0].local_stats.push_back(
>+        cricket::SsrcReceiverInfo());
>+    video_media_info.receivers[0].local_stats[0].ssrc = 4;
>+    video_media_info.receivers[0].codec_payload_type = recv_codec.payload_type;
>+    // transport
>+    graph.transport_id = "RTCTransport_TransportName_1";
>+    auto* video_media_channel =
>+        pc_->AddVideoChannel("VideoMid", "TransportName");
>+    video_media_channel->SetStats(video_media_info);
>+    // track (sender)
>+    graph.sender = stats_->SetupLocalTrackAndSender(
>+        cricket::MEDIA_TYPE_VIDEO, "LocalVideoTrackID", 3, false);
>+    graph.sender_track_id = "RTCMediaStreamTrack_sender_" +
>+                            rtc::ToString(graph.sender->AttachmentId());
>+    // track (receiver) and stream (remote stream)
>+    graph.receiver = stats_->SetupRemoteTrackAndReceiver(
>+        cricket::MEDIA_TYPE_VIDEO, "RemoteVideoTrackID", "RemoteStreamId", 4);
>+    graph.receiver_track_id = "RTCMediaStreamTrack_receiver_" +
>+                              rtc::ToString(graph.receiver->AttachmentId());
>+    graph.remote_stream_id = "RTCMediaStream_RemoteStreamId";
>+    // peer-connection
>+    graph.peer_connection_id = "RTCPeerConnection";
>+
>+    // Expected stats graph:
>+    //
>+    // track (sender)      stream (remote stream) ---> track (receiver)
>+    //          ^                                        ^
>+    //          |                                        |
>+    //         outbound-rtp   inbound-rtp ---------------+
>+    //          |        |     |       |
>+    //          v        v     v       v
>+    // codec (send)     transport     codec (recv)     peer-connection
>+
>+    // Verify the stats graph is set up correctly.
>+    graph.full_report = stats_->GetStatsReport();
>+    EXPECT_EQ(graph.full_report->size(), 9u);
>+    EXPECT_TRUE(graph.full_report->Get(graph.send_codec_id));
>+    EXPECT_TRUE(graph.full_report->Get(graph.recv_codec_id));
>+    EXPECT_TRUE(graph.full_report->Get(graph.outbound_rtp_id));
>+    EXPECT_TRUE(graph.full_report->Get(graph.inbound_rtp_id));
>+    EXPECT_TRUE(graph.full_report->Get(graph.transport_id));
>+    EXPECT_TRUE(graph.full_report->Get(graph.sender_track_id));
>+    EXPECT_TRUE(graph.full_report->Get(graph.receiver_track_id));
>+    EXPECT_TRUE(graph.full_report->Get(graph.remote_stream_id));
>+    EXPECT_TRUE(graph.full_report->Get(graph.peer_connection_id));
>+    const auto& outbound_rtp = graph.full_report->Get(graph.outbound_rtp_id)
>+                                   ->cast_to<RTCOutboundRTPStreamStats>();
>+    EXPECT_EQ(*outbound_rtp.codec_id, graph.send_codec_id);
>+    EXPECT_EQ(*outbound_rtp.track_id, graph.sender_track_id);
>+    EXPECT_EQ(*outbound_rtp.transport_id, graph.transport_id);
>+    const auto& inbound_rtp = graph.full_report->Get(graph.inbound_rtp_id)
>+                                  ->cast_to<RTCInboundRTPStreamStats>();
>+    EXPECT_EQ(*inbound_rtp.codec_id, graph.recv_codec_id);
>+    EXPECT_EQ(*inbound_rtp.track_id, graph.receiver_track_id);
>+    EXPECT_EQ(*inbound_rtp.transport_id, graph.transport_id);
>+
>+    return graph;
>+  }
>+
>  protected:
>-  rtc::scoped_refptr<RTCStatsCollectorTestHelper> test_;
>-  rtc::scoped_refptr<RTCStatsCollector> collector_;
>+  rtc::ScopedFakeClock fake_clock_;
>+  rtc::scoped_refptr<FakePeerConnectionForStats> pc_;
>+  std::unique_ptr<RTCStatsCollectorWrapper> stats_;
> };
> 
> TEST_F(RTCStatsCollectorTest, SingleCallback) {
>   rtc::scoped_refptr<const RTCStatsReport> result;
>-  collector_->GetStatsReport(RTCStatsObtainer::Create(&result));
>+  stats_->stats_collector()->GetStatsReport(RTCStatsObtainer::Create(&result));
>   EXPECT_TRUE_WAIT(result, kGetStatsReportTimeoutMs);
> }
> 
> TEST_F(RTCStatsCollectorTest, MultipleCallbacks) {
>-  rtc::scoped_refptr<const RTCStatsReport> a;
>-  rtc::scoped_refptr<const RTCStatsReport> b;
>-  rtc::scoped_refptr<const RTCStatsReport> c;
>-  collector_->GetStatsReport(RTCStatsObtainer::Create(&a));
>-  collector_->GetStatsReport(RTCStatsObtainer::Create(&b));
>-  collector_->GetStatsReport(RTCStatsObtainer::Create(&c));
>+  rtc::scoped_refptr<const RTCStatsReport> a, b, c;
>+  stats_->stats_collector()->GetStatsReport(RTCStatsObtainer::Create(&a));
>+  stats_->stats_collector()->GetStatsReport(RTCStatsObtainer::Create(&b));
>+  stats_->stats_collector()->GetStatsReport(RTCStatsObtainer::Create(&c));
>   EXPECT_TRUE_WAIT(a, kGetStatsReportTimeoutMs);
>   EXPECT_TRUE_WAIT(b, kGetStatsReportTimeoutMs);
>   EXPECT_TRUE_WAIT(c, kGetStatsReportTimeoutMs);
>+
>   EXPECT_EQ(a.get(), b.get());
>   EXPECT_EQ(b.get(), c.get());
> }
> 
> TEST_F(RTCStatsCollectorTest, CachedStatsReports) {
>   // Caching should ensure |a| and |b| are the same report.
>-  rtc::scoped_refptr<const RTCStatsReport> a = GetStatsReport();
>-  rtc::scoped_refptr<const RTCStatsReport> b = GetStatsReport();
>+  rtc::scoped_refptr<const RTCStatsReport> a = stats_->GetStatsReport();
>+  rtc::scoped_refptr<const RTCStatsReport> b = stats_->GetStatsReport();
>   EXPECT_EQ(a.get(), b.get());
>   // Invalidate cache by clearing it.
>-  collector_->ClearCachedStatsReport();
>-  rtc::scoped_refptr<const RTCStatsReport> c = GetStatsReport();
>+  stats_->stats_collector()->ClearCachedStatsReport();
>+  rtc::scoped_refptr<const RTCStatsReport> c = stats_->GetStatsReport();
>   EXPECT_NE(b.get(), c.get());
>   // Invalidate cache by advancing time.
>-  test_->fake_clock().AdvanceTime(rtc::TimeDelta::FromMilliseconds(51));
>-  rtc::scoped_refptr<const RTCStatsReport> d = GetStatsReport();
>+  fake_clock_.AdvanceTime(TimeDelta::ms(51));
>+  rtc::scoped_refptr<const RTCStatsReport> d = stats_->GetStatsReport();
>   EXPECT_TRUE(d);
>   EXPECT_NE(c.get(), d.get());
> }
> 
> TEST_F(RTCStatsCollectorTest, MultipleCallbacksWithInvalidatedCacheInBetween) {
>-  rtc::scoped_refptr<const RTCStatsReport> a;
>-  rtc::scoped_refptr<const RTCStatsReport> b;
>-  rtc::scoped_refptr<const RTCStatsReport> c;
>-  collector_->GetStatsReport(RTCStatsObtainer::Create(&a));
>-  collector_->GetStatsReport(RTCStatsObtainer::Create(&b));
>+  rtc::scoped_refptr<const RTCStatsReport> a, b, c;
>+  stats_->stats_collector()->GetStatsReport(RTCStatsObtainer::Create(&a));
>+  stats_->stats_collector()->GetStatsReport(RTCStatsObtainer::Create(&b));
>   // Cache is invalidated after 50 ms.
>-  test_->fake_clock().AdvanceTime(rtc::TimeDelta::FromMilliseconds(51));
>-  collector_->GetStatsReport(RTCStatsObtainer::Create(&c));
>+  fake_clock_.AdvanceTime(TimeDelta::ms(51));
>+  stats_->stats_collector()->GetStatsReport(RTCStatsObtainer::Create(&c));
>   EXPECT_TRUE_WAIT(a, kGetStatsReportTimeoutMs);
>   EXPECT_TRUE_WAIT(b, kGetStatsReportTimeoutMs);
>   EXPECT_TRUE_WAIT(c, kGetStatsReportTimeoutMs);
>@@ -754,58 +690,29 @@ TEST_F(RTCStatsCollectorTest, MultipleCallbacksWithInvalidatedCacheInBetween) {
> }
> 
> TEST_F(RTCStatsCollectorTest, CollectRTCCertificateStatsSingle) {
>+  const char kTransportName[] = "transport";
>+
>+  pc_->AddVoiceChannel("audio", kTransportName);
>+
>   std::unique_ptr<CertificateInfo> local_certinfo =
>       CreateFakeCertificateAndInfoFromDers(
>-          std::vector<std::string>({ "(local) single certificate" }));
>+          std::vector<std::string>({"(local) single certificate"}));
>+  pc_->SetLocalCertificate(kTransportName, local_certinfo->certificate);
>+
>   std::unique_ptr<CertificateInfo> remote_certinfo =
>       CreateFakeCertificateAndInfoFromDers(
>-          std::vector<std::string>({ "(remote) single certificate" }));
>-
>-  // Mock the session to return the local and remote certificates.
>-  EXPECT_CALL(test_->pc(), GetSessionStats(_))
>-      .WillRepeatedly(Invoke([](const ChannelNamePairs&) {
>-        std::unique_ptr<SessionStats> stats(new SessionStats());
>-        stats->transport_stats["transport"].transport_name = "transport";
>-        return stats;
>-      }));
>-  EXPECT_CALL(test_->pc(), GetLocalCertificate(_, _))
>-      .WillRepeatedly(
>-          Invoke([&local_certinfo](
>-                     const std::string& transport_name,
>-                     rtc::scoped_refptr<rtc::RTCCertificate>* certificate) {
>-            if (transport_name == "transport") {
>-              *certificate = local_certinfo->certificate;
>-              return true;
>-            }
>-            return false;
>-          }));
>-  EXPECT_CALL(test_->pc(), GetRemoteSSLCertificate_ReturnsRawPointer(_))
>-      .WillRepeatedly(Invoke([&remote_certinfo](
>-                                 const std::string& transport_name) {
>-        if (transport_name == "transport")
>-          return remote_certinfo->certificate->ssl_certificate().GetReference();
>-        return static_cast<rtc::SSLCertificate*>(nullptr);
>-      }));
>-
>-  rtc::scoped_refptr<const RTCStatsReport> report = GetStatsReport();
>+          std::vector<std::string>({"(remote) single certificate"}));
>+  pc_->SetRemoteCertChain(
>+      kTransportName,
>+      remote_certinfo->certificate->ssl_cert_chain().UniqueCopy());
>+
>+  rtc::scoped_refptr<const RTCStatsReport> report = stats_->GetStatsReport();
>+
>   ExpectReportContainsCertificateInfo(report, *local_certinfo);
>   ExpectReportContainsCertificateInfo(report, *remote_certinfo);
> }
> 
> TEST_F(RTCStatsCollectorTest, CollectRTCCodecStats) {
>-  auto* voice_media_channel = new MockVoiceMediaChannel();
>-  cricket::VoiceChannel voice_channel(
>-      test_->worker_thread(), test_->network_thread(),
>-      test_->signaling_thread(), test_->media_engine(),
>-      rtc::WrapUnique(voice_media_channel), "VoiceContentName",
>-      kDefaultRtcpMuxRequired, kDefaultSrtpRequired);
>-
>-  auto* video_media_channel = new MockVideoMediaChannel();
>-  cricket::VideoChannel video_channel(
>-      test_->worker_thread(), test_->network_thread(),
>-      test_->signaling_thread(), rtc::WrapUnique(video_media_channel),
>-      "VideoContentName", kDefaultRtcpMuxRequired, kDefaultSrtpRequired);
>-
>   // Audio
>   cricket::VoiceMediaInfo voice_media_info;
> 
>@@ -825,8 +732,8 @@ TEST_F(RTCStatsCollectorTest, CollectRTCCodecStats) {
>   voice_media_info.send_codecs.insert(
>       std::make_pair(outbound_audio_codec.payload_type, outbound_audio_codec));
> 
>-  EXPECT_CALL(*voice_media_channel, GetStats(_))
>-      .WillOnce(DoAll(SetArgPointee<0>(voice_media_info), Return(true)));
>+  auto* voice_media_channel = pc_->AddVoiceChannel("AudioMid", "TransportName");
>+  voice_media_channel->SetStats(voice_media_info);
> 
>   // Video
>   cricket::VideoMediaInfo video_media_info;
>@@ -847,130 +754,85 @@ TEST_F(RTCStatsCollectorTest, CollectRTCCodecStats) {
>   video_media_info.send_codecs.insert(
>       std::make_pair(outbound_video_codec.payload_type, outbound_video_codec));
> 
>-  EXPECT_CALL(*video_media_channel, GetStats(_))
>-      .WillOnce(DoAll(SetArgPointee<0>(video_media_info), Return(true)));
>-
>-  SessionStats session_stats;
>-  session_stats.transport_stats["TransportName"].transport_name =
>-      "TransportName";
>-
>-  EXPECT_CALL(test_->pc(), GetSessionStats(_))
>-      .WillRepeatedly(Invoke([&session_stats](const ChannelNamePairs&) {
>-        return std::unique_ptr<SessionStats>(new SessionStats(session_stats));
>-      }));
>-  EXPECT_CALL(test_->pc(), voice_channel())
>-      .WillRepeatedly(Return(&voice_channel));
>-  EXPECT_CALL(test_->pc(), video_channel())
>-      .WillRepeatedly(Return(&video_channel));
>+  auto* video_media_channel = pc_->AddVideoChannel("VideoMid", "TransportName");
>+  video_media_channel->SetStats(video_media_info);
> 
>-  rtc::scoped_refptr<const RTCStatsReport> report = GetStatsReport();
>+  rtc::scoped_refptr<const RTCStatsReport> report = stats_->GetStatsReport();
> 
>-  RTCCodecStats expected_inbound_audio_codec(
>-      "RTCCodec_InboundAudio_1", report->timestamp_us());
>+  RTCCodecStats expected_inbound_audio_codec("RTCCodec_AudioMid_Inbound_1",
>+                                             report->timestamp_us());
>   expected_inbound_audio_codec.payload_type = 1;
>   expected_inbound_audio_codec.mime_type = "audio/opus";
>   expected_inbound_audio_codec.clock_rate = 1337;
> 
>-  RTCCodecStats expected_outbound_audio_codec(
>-      "RTCCodec_OutboundAudio_2", report->timestamp_us());
>+  RTCCodecStats expected_outbound_audio_codec("RTCCodec_AudioMid_Outbound_2",
>+                                              report->timestamp_us());
>   expected_outbound_audio_codec.payload_type = 2;
>   expected_outbound_audio_codec.mime_type = "audio/isac";
>   expected_outbound_audio_codec.clock_rate = 1338;
> 
>-  RTCCodecStats expected_inbound_video_codec(
>-      "RTCCodec_InboundVideo_3", report->timestamp_us());
>+  RTCCodecStats expected_inbound_video_codec("RTCCodec_VideoMid_Inbound_3",
>+                                             report->timestamp_us());
>   expected_inbound_video_codec.payload_type = 3;
>   expected_inbound_video_codec.mime_type = "video/H264";
>   expected_inbound_video_codec.clock_rate = 1339;
> 
>-  RTCCodecStats expected_outbound_video_codec(
>-      "RTCCodec_OutboundVideo_4", report->timestamp_us());
>+  RTCCodecStats expected_outbound_video_codec("RTCCodec_VideoMid_Outbound_4",
>+                                              report->timestamp_us());
>   expected_outbound_video_codec.payload_type = 4;
>   expected_outbound_video_codec.mime_type = "video/VP8";
>   expected_outbound_video_codec.clock_rate = 1340;
> 
>   ASSERT_TRUE(report->Get(expected_inbound_audio_codec.id()));
>-  EXPECT_EQ(expected_inbound_audio_codec,
>-            report->Get(expected_inbound_audio_codec.id())->cast_to<
>-                  RTCCodecStats>());
>+  EXPECT_EQ(
>+      expected_inbound_audio_codec,
>+      report->Get(expected_inbound_audio_codec.id())->cast_to<RTCCodecStats>());
> 
>   ASSERT_TRUE(report->Get(expected_outbound_audio_codec.id()));
>   EXPECT_EQ(expected_outbound_audio_codec,
>-            report->Get(expected_outbound_audio_codec.id())->cast_to<
>-                  RTCCodecStats>());
>+            report->Get(expected_outbound_audio_codec.id())
>+                ->cast_to<RTCCodecStats>());
> 
>   ASSERT_TRUE(report->Get(expected_inbound_video_codec.id()));
>-  EXPECT_EQ(expected_inbound_video_codec,
>-            report->Get(expected_inbound_video_codec.id())->cast_to<
>-                  RTCCodecStats>());
>+  EXPECT_EQ(
>+      expected_inbound_video_codec,
>+      report->Get(expected_inbound_video_codec.id())->cast_to<RTCCodecStats>());
> 
>   ASSERT_TRUE(report->Get(expected_outbound_video_codec.id()));
>   EXPECT_EQ(expected_outbound_video_codec,
>-            report->Get(expected_outbound_video_codec.id())->cast_to<
>-                  RTCCodecStats>());
>+            report->Get(expected_outbound_video_codec.id())
>+                ->cast_to<RTCCodecStats>());
> }
> 
> TEST_F(RTCStatsCollectorTest, CollectRTCCertificateStatsMultiple) {
>+  const char kAudioTransport[] = "audio";
>+  const char kVideoTransport[] = "video";
>+
>+  pc_->AddVoiceChannel("audio", kAudioTransport);
>   std::unique_ptr<CertificateInfo> audio_local_certinfo =
>       CreateFakeCertificateAndInfoFromDers(
>-          std::vector<std::string>({ "(local) audio" }));
>-  audio_local_certinfo = CreateFakeCertificateAndInfoFromDers(
>-      audio_local_certinfo->ders);
>+          std::vector<std::string>({"(local) audio"}));
>+  pc_->SetLocalCertificate(kAudioTransport, audio_local_certinfo->certificate);
>   std::unique_ptr<CertificateInfo> audio_remote_certinfo =
>       CreateFakeCertificateAndInfoFromDers(
>-          std::vector<std::string>({ "(remote) audio" }));
>-  audio_remote_certinfo = CreateFakeCertificateAndInfoFromDers(
>-      audio_remote_certinfo->ders);
>+          std::vector<std::string>({"(remote) audio"}));
>+  pc_->SetRemoteCertChain(
>+      kAudioTransport,
>+      audio_remote_certinfo->certificate->ssl_cert_chain().UniqueCopy());
> 
>+  pc_->AddVideoChannel("video", kVideoTransport);
>   std::unique_ptr<CertificateInfo> video_local_certinfo =
>       CreateFakeCertificateAndInfoFromDers(
>-          std::vector<std::string>({ "(local) video" }));
>-  video_local_certinfo = CreateFakeCertificateAndInfoFromDers(
>-      video_local_certinfo->ders);
>+          std::vector<std::string>({"(local) video"}));
>+  pc_->SetLocalCertificate(kVideoTransport, video_local_certinfo->certificate);
>   std::unique_ptr<CertificateInfo> video_remote_certinfo =
>       CreateFakeCertificateAndInfoFromDers(
>-          std::vector<std::string>({ "(remote) video" }));
>-  video_remote_certinfo = CreateFakeCertificateAndInfoFromDers(
>-      video_remote_certinfo->ders);
>-
>-  // Mock the session to return the local and remote certificates.
>-  EXPECT_CALL(test_->pc(), GetSessionStats(_))
>-      .WillRepeatedly(Invoke([](const ChannelNamePairs&) {
>-        std::unique_ptr<SessionStats> stats(new SessionStats());
>-        stats->transport_stats["audio"].transport_name = "audio";
>-        stats->transport_stats["video"].transport_name = "video";
>-        return stats;
>-      }));
>-  EXPECT_CALL(test_->pc(), GetLocalCertificate(_, _))
>-      .WillRepeatedly(
>-          Invoke([&audio_local_certinfo, &video_local_certinfo](
>-                     const std::string& transport_name,
>-                     rtc::scoped_refptr<rtc::RTCCertificate>* certificate) {
>-            if (transport_name == "audio") {
>-              *certificate = audio_local_certinfo->certificate;
>-              return true;
>-            }
>-            if (transport_name == "video") {
>-              *certificate = video_local_certinfo->certificate;
>-              return true;
>-            }
>-            return false;
>-          }));
>-  EXPECT_CALL(test_->pc(), GetRemoteSSLCertificate_ReturnsRawPointer(_))
>-      .WillRepeatedly(Invoke([&audio_remote_certinfo, &video_remote_certinfo](
>-                                 const std::string& transport_name) {
>-        if (transport_name == "audio") {
>-          return audio_remote_certinfo->certificate->ssl_certificate()
>-              .GetReference();
>-        }
>-        if (transport_name == "video") {
>-          return video_remote_certinfo->certificate->ssl_certificate()
>-              .GetReference();
>-        }
>-        return static_cast<rtc::SSLCertificate*>(nullptr);
>-      }));
>-
>-  rtc::scoped_refptr<const RTCStatsReport> report = GetStatsReport();
>+          std::vector<std::string>({"(remote) video"}));
>+  pc_->SetRemoteCertChain(
>+      kVideoTransport,
>+      video_remote_certinfo->certificate->ssl_cert_chain().UniqueCopy());
>+
>+  rtc::scoped_refptr<const RTCStatsReport> report = stats_->GetStatsReport();
>   ExpectReportContainsCertificateInfo(report, *audio_local_certinfo);
>   ExpectReportContainsCertificateInfo(report, *audio_remote_certinfo);
>   ExpectReportContainsCertificateInfo(report, *video_local_certinfo);
>@@ -978,57 +840,32 @@ TEST_F(RTCStatsCollectorTest, CollectRTCCertificateStatsMultiple) {
> }
> 
> TEST_F(RTCStatsCollectorTest, CollectRTCCertificateStatsChain) {
>-  std::vector<std::string> local_ders;
>-  local_ders.push_back("(local) this");
>-  local_ders.push_back("(local) is");
>-  local_ders.push_back("(local) a");
>-  local_ders.push_back("(local) chain");
>+  const char kTransportName[] = "transport";
>+
>+  pc_->AddVoiceChannel("audio", kTransportName);
>+
>   std::unique_ptr<CertificateInfo> local_certinfo =
>-      CreateFakeCertificateAndInfoFromDers(local_ders);
>-  std::vector<std::string> remote_ders;
>-  remote_ders.push_back("(remote) this");
>-  remote_ders.push_back("(remote) is");
>-  remote_ders.push_back("(remote) another");
>-  remote_ders.push_back("(remote) chain");
>+      CreateFakeCertificateAndInfoFromDers(
>+          {"(local) this", "(local) is", "(local) a", "(local) chain"});
>+  pc_->SetLocalCertificate(kTransportName, local_certinfo->certificate);
>+
>   std::unique_ptr<CertificateInfo> remote_certinfo =
>-      CreateFakeCertificateAndInfoFromDers(remote_ders);
>-
>-  // Mock the session to return the local and remote certificates.
>-  EXPECT_CALL(test_->pc(), GetSessionStats(_))
>-      .WillRepeatedly(Invoke([](const ChannelNamePairs&) {
>-        std::unique_ptr<SessionStats> stats(new SessionStats());
>-        stats->transport_stats["transport"].transport_name = "transport";
>-        return stats;
>-      }));
>-  EXPECT_CALL(test_->pc(), GetLocalCertificate(_, _))
>-      .WillRepeatedly(
>-          Invoke([&local_certinfo](
>-                     const std::string& transport_name,
>-                     rtc::scoped_refptr<rtc::RTCCertificate>* certificate) {
>-            if (transport_name == "transport") {
>-              *certificate = local_certinfo->certificate;
>-              return true;
>-            }
>-            return false;
>-          }));
>-  EXPECT_CALL(test_->pc(), GetRemoteSSLCertificate_ReturnsRawPointer(_))
>-      .WillRepeatedly(Invoke([&remote_certinfo](
>-                                 const std::string& transport_name) {
>-        if (transport_name == "transport")
>-          return remote_certinfo->certificate->ssl_certificate().GetReference();
>-        return static_cast<rtc::SSLCertificate*>(nullptr);
>-      }));
>-
>-  rtc::scoped_refptr<const RTCStatsReport> report = GetStatsReport();
>+      CreateFakeCertificateAndInfoFromDers({"(remote) this", "(remote) is",
>+                                            "(remote) another",
>+                                            "(remote) chain"});
>+  pc_->SetRemoteCertChain(
>+      kTransportName,
>+      remote_certinfo->certificate->ssl_cert_chain().UniqueCopy());
>+
>+  rtc::scoped_refptr<const RTCStatsReport> report = stats_->GetStatsReport();
>   ExpectReportContainsCertificateInfo(report, *local_certinfo);
>   ExpectReportContainsCertificateInfo(report, *remote_certinfo);
> }
> 
> TEST_F(RTCStatsCollectorTest, CollectRTCDataChannelStats) {
>-  test_->data_channels().push_back(
>-      new MockDataChannel(
>-          0, "MockDataChannel0", DataChannelInterface::kConnecting, "udp",
>-          1, 2, 3, 4));
>+  pc_->AddSctpDataChannel(new MockDataChannel(0, "MockDataChannel0",
>+                                              DataChannelInterface::kConnecting,
>+                                              "udp", 1, 2, 3, 4));
>   RTCDataChannelStats expected_data_channel0("RTCDataChannel_0", 0);
>   expected_data_channel0.label = "MockDataChannel0";
>   expected_data_channel0.protocol = "udp";
>@@ -1039,10 +876,8 @@ TEST_F(RTCStatsCollectorTest, CollectRTCDataChannelStats) {
>   expected_data_channel0.messages_received = 3;
>   expected_data_channel0.bytes_received = 4;
> 
>-  test_->data_channels().push_back(
>-      new MockDataChannel(
>-          1, "MockDataChannel1", DataChannelInterface::kOpen, "tcp",
>-          5, 6, 7, 8));
>+  pc_->AddSctpDataChannel(new MockDataChannel(
>+      1, "MockDataChannel1", DataChannelInterface::kOpen, "tcp", 5, 6, 7, 8));
>   RTCDataChannelStats expected_data_channel1("RTCDataChannel_1", 0);
>   expected_data_channel1.label = "MockDataChannel1";
>   expected_data_channel1.protocol = "tcp";
>@@ -1053,10 +888,9 @@ TEST_F(RTCStatsCollectorTest, CollectRTCDataChannelStats) {
>   expected_data_channel1.messages_received = 7;
>   expected_data_channel1.bytes_received = 8;
> 
>-  test_->data_channels().push_back(
>-      new MockDataChannel(
>-          2, "MockDataChannel2", DataChannelInterface::kClosing, "udp",
>-          9, 10, 11, 12));
>+  pc_->AddSctpDataChannel(new MockDataChannel(2, "MockDataChannel2",
>+                                              DataChannelInterface::kClosing,
>+                                              "udp", 9, 10, 11, 12));
>   RTCDataChannelStats expected_data_channel2("RTCDataChannel_2", 0);
>   expected_data_channel2.label = "MockDataChannel2";
>   expected_data_channel2.protocol = "udp";
>@@ -1067,10 +901,9 @@ TEST_F(RTCStatsCollectorTest, CollectRTCDataChannelStats) {
>   expected_data_channel2.messages_received = 11;
>   expected_data_channel2.bytes_received = 12;
> 
>-  test_->data_channels().push_back(
>-      new MockDataChannel(
>-          3, "MockDataChannel3", DataChannelInterface::kClosed, "tcp",
>-          13, 14, 15, 16));
>+  pc_->AddSctpDataChannel(new MockDataChannel(3, "MockDataChannel3",
>+                                              DataChannelInterface::kClosed,
>+                                              "tcp", 13, 14, 15, 16));
>   RTCDataChannelStats expected_data_channel3("RTCDataChannel_3", 0);
>   expected_data_channel3.label = "MockDataChannel3";
>   expected_data_channel3.protocol = "tcp";
>@@ -1081,23 +914,24 @@ TEST_F(RTCStatsCollectorTest, CollectRTCDataChannelStats) {
>   expected_data_channel3.messages_received = 15;
>   expected_data_channel3.bytes_received = 16;
> 
>-  rtc::scoped_refptr<const RTCStatsReport> report = GetStatsReport();
>+  rtc::scoped_refptr<const RTCStatsReport> report = stats_->GetStatsReport();
>+
>   ASSERT_TRUE(report->Get(expected_data_channel0.id()));
>-  EXPECT_EQ(expected_data_channel0,
>-            report->Get(expected_data_channel0.id())->cast_to<
>-                RTCDataChannelStats>());
>+  EXPECT_EQ(
>+      expected_data_channel0,
>+      report->Get(expected_data_channel0.id())->cast_to<RTCDataChannelStats>());
>   ASSERT_TRUE(report->Get(expected_data_channel1.id()));
>-  EXPECT_EQ(expected_data_channel1,
>-            report->Get(expected_data_channel1.id())->cast_to<
>-                RTCDataChannelStats>());
>+  EXPECT_EQ(
>+      expected_data_channel1,
>+      report->Get(expected_data_channel1.id())->cast_to<RTCDataChannelStats>());
>   ASSERT_TRUE(report->Get(expected_data_channel2.id()));
>-  EXPECT_EQ(expected_data_channel2,
>-            report->Get(expected_data_channel2.id())->cast_to<
>-                RTCDataChannelStats>());
>+  EXPECT_EQ(
>+      expected_data_channel2,
>+      report->Get(expected_data_channel2.id())->cast_to<RTCDataChannelStats>());
>   ASSERT_TRUE(report->Get(expected_data_channel3.id()));
>-  EXPECT_EQ(expected_data_channel3,
>-            report->Get(expected_data_channel3.id())->cast_to<
>-                RTCDataChannelStats>());
>+  EXPECT_EQ(
>+      expected_data_channel3,
>+      report->Get(expected_data_channel3.id())->cast_to<RTCDataChannelStats>());
> }
> 
> TEST_F(RTCStatsCollectorTest, CollectRTCIceCandidateStats) {
>@@ -1163,8 +997,8 @@ TEST_F(RTCStatsCollectorTest, CollectRTCIceCandidateStats) {
>   std::unique_ptr<cricket::Candidate> b_local =
>       CreateFakeCandidate("42.42.42.42", 42, "b_local's protocol",
>                           rtc::ADAPTER_TYPE_WIFI, cricket::LOCAL_PORT_TYPE, 42);
>-  RTCLocalIceCandidateStats expected_b_local(
>-      "RTCIceCandidate_" + b_local->id(), 0);
>+  RTCLocalIceCandidateStats expected_b_local("RTCIceCandidate_" + b_local->id(),
>+                                             0);
>   expected_b_local.transport_id = "RTCTransport_b_0";
>   expected_b_local.network_type = "wifi";
>   expected_b_local.ip = "42.42.42.42";
>@@ -1189,8 +1023,6 @@ TEST_F(RTCStatsCollectorTest, CollectRTCIceCandidateStats) {
>   expected_b_remote.deleted = false;
>   EXPECT_TRUE(*expected_b_remote.is_remote);
> 
>-  SessionStats session_stats;
>-
>   cricket::TransportChannelStats a_transport_channel_stats;
>   a_transport_channel_stats.connection_infos.push_back(
>       cricket::ConnectionInfo());
>@@ -1204,9 +1036,9 @@ TEST_F(RTCStatsCollectorTest, CollectRTCIceCandidateStats) {
>       *a_local_prflx.get();
>   a_transport_channel_stats.connection_infos[1].remote_candidate =
>       *a_remote_relay.get();
>-  session_stats.transport_stats["a"].transport_name = "a";
>-  session_stats.transport_stats["a"].channel_stats.push_back(
>-      a_transport_channel_stats);
>+
>+  pc_->AddVoiceChannel("audio", "a");
>+  pc_->SetTransportStats("a", a_transport_channel_stats);
> 
>   cricket::TransportChannelStats b_transport_channel_stats;
>   b_transport_channel_stats.connection_infos.push_back(
>@@ -1215,52 +1047,39 @@ TEST_F(RTCStatsCollectorTest, CollectRTCIceCandidateStats) {
>       *b_local.get();
>   b_transport_channel_stats.connection_infos[0].remote_candidate =
>       *b_remote.get();
>-  session_stats.transport_stats["b"].transport_name = "b";
>-  session_stats.transport_stats["b"].channel_stats.push_back(
>-      b_transport_channel_stats);
> 
>-  // Mock the session to return the desired candidates.
>-  EXPECT_CALL(test_->pc(), GetSessionStats(_))
>-      .WillRepeatedly(Invoke([&session_stats](const ChannelNamePairs&) {
>-        return std::unique_ptr<SessionStats>(new SessionStats(session_stats));
>-      }));
>+  pc_->AddVideoChannel("video", "b");
>+  pc_->SetTransportStats("b", b_transport_channel_stats);
> 
>-  rtc::scoped_refptr<const RTCStatsReport> report = GetStatsReport();
>+  rtc::scoped_refptr<const RTCStatsReport> report = stats_->GetStatsReport();
> 
>   ASSERT_TRUE(report->Get(expected_a_local_host.id()));
>-  EXPECT_EQ(expected_a_local_host,
>-            report->Get(expected_a_local_host.id())->cast_to<
>-                RTCLocalIceCandidateStats>());
>+  EXPECT_EQ(expected_a_local_host, report->Get(expected_a_local_host.id())
>+                                       ->cast_to<RTCLocalIceCandidateStats>());
>   ASSERT_TRUE(report->Get(expected_a_remote_srflx.id()));
>   EXPECT_EQ(expected_a_remote_srflx,
>-            report->Get(expected_a_remote_srflx.id())->cast_to<
>-                RTCRemoteIceCandidateStats>());
>+            report->Get(expected_a_remote_srflx.id())
>+                ->cast_to<RTCRemoteIceCandidateStats>());
>   ASSERT_TRUE(report->Get(expected_a_local_prflx.id()));
>-  EXPECT_EQ(expected_a_local_prflx,
>-            report->Get(expected_a_local_prflx.id())->cast_to<
>-                RTCLocalIceCandidateStats>());
>+  EXPECT_EQ(expected_a_local_prflx, report->Get(expected_a_local_prflx.id())
>+                                        ->cast_to<RTCLocalIceCandidateStats>());
>   ASSERT_TRUE(report->Get(expected_a_remote_relay.id()));
>   EXPECT_EQ(expected_a_remote_relay,
>-            report->Get(expected_a_remote_relay.id())->cast_to<
>-                RTCRemoteIceCandidateStats>());
>+            report->Get(expected_a_remote_relay.id())
>+                ->cast_to<RTCRemoteIceCandidateStats>());
>   ASSERT_TRUE(report->Get(expected_b_local.id()));
>-  EXPECT_EQ(expected_b_local,
>-            report->Get(expected_b_local.id())->cast_to<
>-                RTCLocalIceCandidateStats>());
>+  EXPECT_EQ(
>+      expected_b_local,
>+      report->Get(expected_b_local.id())->cast_to<RTCLocalIceCandidateStats>());
>   ASSERT_TRUE(report->Get(expected_b_remote.id()));
>-  EXPECT_EQ(expected_b_remote,
>-            report->Get(expected_b_remote.id())->cast_to<
>-                RTCRemoteIceCandidateStats>());
>+  EXPECT_EQ(expected_b_remote, report->Get(expected_b_remote.id())
>+                                   ->cast_to<RTCRemoteIceCandidateStats>());
>   EXPECT_TRUE(report->Get("RTCTransport_a_0"));
>   EXPECT_TRUE(report->Get("RTCTransport_b_0"));
> }
> 
> TEST_F(RTCStatsCollectorTest, CollectRTCIceCandidatePairStats) {
>-  auto* video_media_channel = new MockVideoMediaChannel();
>-  cricket::VideoChannel video_channel(
>-      test_->worker_thread(), test_->network_thread(),
>-      test_->signaling_thread(), rtc::WrapUnique(video_media_channel),
>-      "VideoContentName", kDefaultRtcpMuxRequired, kDefaultSrtpRequired);
>+  const char kTransportName[] = "transport";
> 
>   std::unique_ptr<cricket::Candidate> local_candidate =
>       CreateFakeCandidate("42.42.42.42", 42, "protocol", rtc::ADAPTER_TYPE_WIFI,
>@@ -1269,8 +1088,6 @@ TEST_F(RTCStatsCollectorTest, CollectRTCIceCandidatePairStats) {
>       "42.42.42.42", 42, "protocol", rtc::ADAPTER_TYPE_UNKNOWN,
>       cricket::LOCAL_PORT_TYPE, 42);
> 
>-  SessionStats session_stats;
>-
>   cricket::ConnectionInfo connection_info;
>   connection_info.best_connection = false;
>   connection_info.local_candidate = *local_candidate.get();
>@@ -1279,7 +1096,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCIceCandidatePairStats) {
>   connection_info.sent_total_bytes = 42;
>   connection_info.recv_total_bytes = 1234;
>   connection_info.total_round_trip_time_ms = 0;
>-  connection_info.current_round_trip_time_ms = rtc::nullopt;
>+  connection_info.current_round_trip_time_ms = absl::nullopt;
>   connection_info.recv_ping_requests = 2020;
>   connection_info.sent_ping_requests_total = 2020;
>   connection_info.sent_ping_requests_before_first_response = 2000;
>@@ -1292,25 +1109,11 @@ TEST_F(RTCStatsCollectorTest, CollectRTCIceCandidatePairStats) {
>   cricket::TransportChannelStats transport_channel_stats;
>   transport_channel_stats.component = cricket::ICE_CANDIDATE_COMPONENT_RTP;
>   transport_channel_stats.connection_infos.push_back(connection_info);
>-  session_stats.transport_stats["transport"].transport_name = "transport";
>-  session_stats.transport_stats["transport"].channel_stats.push_back(
>-      transport_channel_stats);
>-
>-  // Mock the session to return the desired candidates.
>-  EXPECT_CALL(test_->pc(), GetSessionStats(_))
>-      .WillRepeatedly(Invoke([&session_stats](const ChannelNamePairs&) {
>-        return std::unique_ptr<SessionStats>(new SessionStats(session_stats));
>-      }));
>-
>-  // Mock the session to return bandwidth estimation info. These should only
>-  // be used for a selected candidate pair.
>-  cricket::VideoMediaInfo video_media_info;
>-  EXPECT_CALL(*video_media_channel, GetStats(_))
>-      .WillOnce(DoAll(SetArgPointee<0>(video_media_info), Return(true)));
>-  EXPECT_CALL(test_->pc(), video_channel())
>-      .WillRepeatedly(Return(&video_channel));
> 
>-  rtc::scoped_refptr<const RTCStatsReport> report = GetStatsReport();
>+  pc_->AddVideoChannel("video", kTransportName);
>+  pc_->SetTransportStats(kTransportName, transport_channel_stats);
>+
>+  rtc::scoped_refptr<const RTCStatsReport> report = stats_->GetStatsReport();
> 
>   RTCIceCandidatePairStats expected_pair("RTCIceCandidatePair_" +
>                                              local_candidate->id() + "_" +
>@@ -1318,7 +1121,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCIceCandidatePairStats) {
>                                          report->timestamp_us());
>   expected_pair.transport_id =
>       "RTCTransport_transport_" +
>-      rtc::ToString<>(cricket::ICE_CANDIDATE_COMPONENT_RTP);
>+      rtc::ToString(cricket::ICE_CANDIDATE_COMPONENT_RTP);
>   expected_pair.local_candidate_id = "RTCIceCandidate_" + local_candidate->id();
>   expected_pair.remote_candidate_id =
>       "RTCIceCandidate_" + remote_candidate->id();
>@@ -1346,14 +1149,9 @@ TEST_F(RTCStatsCollectorTest, CollectRTCIceCandidatePairStats) {
>   EXPECT_TRUE(report->Get(*expected_pair.transport_id));
> 
>   // Set nominated and "GetStats" again.
>-  session_stats.transport_stats["transport"]
>-      .channel_stats[0]
>-      .connection_infos[0]
>-      .nominated = true;
>-  EXPECT_CALL(*video_media_channel, GetStats(_))
>-      .WillOnce(DoAll(SetArgPointee<0>(video_media_info), Return(true)));
>-  collector_->ClearCachedStatsReport();
>-  report = GetStatsReport();
>+  transport_channel_stats.connection_infos[0].nominated = true;
>+  pc_->SetTransportStats(kTransportName, transport_channel_stats);
>+  report = stats_->GetFreshStatsReport();
>   expected_pair.nominated = true;
>   ASSERT_TRUE(report->Get(expected_pair.id()));
>   EXPECT_EQ(
>@@ -1362,14 +1160,10 @@ TEST_F(RTCStatsCollectorTest, CollectRTCIceCandidatePairStats) {
>   EXPECT_TRUE(report->Get(*expected_pair.transport_id));
> 
>   // Set round trip times and "GetStats" again.
>-  session_stats.transport_stats["transport"].channel_stats[0]
>-      .connection_infos[0].total_round_trip_time_ms = 7331;
>-  session_stats.transport_stats["transport"].channel_stats[0]
>-      .connection_infos[0].current_round_trip_time_ms = 1337;
>-  EXPECT_CALL(*video_media_channel, GetStats(_))
>-      .WillOnce(DoAll(SetArgPointee<0>(video_media_info), Return(true)));
>-  collector_->ClearCachedStatsReport();
>-  report = GetStatsReport();
>+  transport_channel_stats.connection_infos[0].total_round_trip_time_ms = 7331;
>+  transport_channel_stats.connection_infos[0].current_round_trip_time_ms = 1337;
>+  pc_->SetTransportStats(kTransportName, transport_channel_stats);
>+  report = stats_->GetFreshStatsReport();
>   expected_pair.total_round_trip_time = 7.331;
>   expected_pair.current_round_trip_time = 1.337;
>   ASSERT_TRUE(report->Get(expected_pair.id()));
>@@ -1379,14 +1173,9 @@ TEST_F(RTCStatsCollectorTest, CollectRTCIceCandidatePairStats) {
>   EXPECT_TRUE(report->Get(*expected_pair.transport_id));
> 
>   // Make pair the current pair, clear bandwidth and "GetStats" again.
>-  session_stats.transport_stats["transport"]
>-      .channel_stats[0]
>-      .connection_infos[0]
>-      .best_connection = true;
>-  EXPECT_CALL(*video_media_channel, GetStats(_))
>-      .WillOnce(DoAll(SetArgPointee<0>(video_media_info), Return(true)));
>-  collector_->ClearCachedStatsReport();
>-  report = GetStatsReport();
>+  transport_channel_stats.connection_infos[0].best_connection = true;
>+  pc_->SetTransportStats(kTransportName, transport_channel_stats);
>+  report = stats_->GetFreshStatsReport();
>   // |expected_pair.available_[outgoing/incoming]_bitrate| should still be
>   // undefined because bandwidth is not set.
>   ASSERT_TRUE(report->Get(expected_pair.id()));
>@@ -1401,11 +1190,8 @@ TEST_F(RTCStatsCollectorTest, CollectRTCIceCandidatePairStats) {
>   call_stats.send_bandwidth_bps = kSendBandwidth;
>   const int kRecvBandwidth = 999;
>   call_stats.recv_bandwidth_bps = kRecvBandwidth;
>-  EXPECT_CALL(test_->pc(), GetCallStats()).WillRepeatedly(Return(call_stats));
>-  EXPECT_CALL(*video_media_channel, GetStats(_))
>-      .WillOnce(DoAll(SetArgPointee<0>(video_media_info), Return(true)));
>-  collector_->ClearCachedStatsReport();
>-  report = GetStatsReport();
>+  pc_->SetCallStats(call_stats);
>+  report = stats_->GetFreshStatsReport();
>   expected_pair.available_outgoing_bitrate = kSendBandwidth;
>   expected_pair.available_incoming_bitrate = kRecvBandwidth;
>   ASSERT_TRUE(report->Get(expected_pair.id()));
>@@ -1427,8 +1213,8 @@ TEST_F(RTCStatsCollectorTest, CollectRTCIceCandidatePairStats) {
>   EXPECT_FALSE(*expected_local_candidate.is_remote);
>   ASSERT_TRUE(report->Get(expected_local_candidate.id()));
>   EXPECT_EQ(expected_local_candidate,
>-            report->Get(expected_local_candidate.id())->cast_to<
>-                RTCLocalIceCandidateStats>());
>+            report->Get(expected_local_candidate.id())
>+                ->cast_to<RTCLocalIceCandidateStats>());
> 
>   RTCRemoteIceCandidateStats expected_remote_candidate(
>       *expected_pair.remote_candidate_id, report->timestamp_us());
>@@ -1442,61 +1228,61 @@ TEST_F(RTCStatsCollectorTest, CollectRTCIceCandidatePairStats) {
>   EXPECT_TRUE(*expected_remote_candidate.is_remote);
>   ASSERT_TRUE(report->Get(expected_remote_candidate.id()));
>   EXPECT_EQ(expected_remote_candidate,
>-            report->Get(expected_remote_candidate.id())->cast_to<
>-                RTCRemoteIceCandidateStats>());
>+            report->Get(expected_remote_candidate.id())
>+                ->cast_to<RTCRemoteIceCandidateStats>());
> }
> 
> TEST_F(RTCStatsCollectorTest, CollectRTCPeerConnectionStats) {
>   {
>-    rtc::scoped_refptr<const RTCStatsReport> report = GetStatsReport();
>+    rtc::scoped_refptr<const RTCStatsReport> report = stats_->GetStatsReport();
>     RTCPeerConnectionStats expected("RTCPeerConnection",
>                                     report->timestamp_us());
>     expected.data_channels_opened = 0;
>     expected.data_channels_closed = 0;
>     ASSERT_TRUE(report->Get("RTCPeerConnection"));
>-    EXPECT_EQ(expected,
>-              report->Get("RTCPeerConnection")->cast_to<
>-                  RTCPeerConnectionStats>());
>+    EXPECT_EQ(
>+        expected,
>+        report->Get("RTCPeerConnection")->cast_to<RTCPeerConnectionStats>());
>   }
> 
>   rtc::scoped_refptr<DataChannel> dummy_channel_a = DataChannel::Create(
>       nullptr, cricket::DCT_NONE, "DummyChannelA", InternalDataChannelInit());
>-  test_->pc().SignalDataChannelCreated(dummy_channel_a.get());
>+  pc_->SignalDataChannelCreated()(dummy_channel_a.get());
>   rtc::scoped_refptr<DataChannel> dummy_channel_b = DataChannel::Create(
>       nullptr, cricket::DCT_NONE, "DummyChannelB", InternalDataChannelInit());
>-  test_->pc().SignalDataChannelCreated(dummy_channel_b.get());
>+  pc_->SignalDataChannelCreated()(dummy_channel_b.get());
> 
>   dummy_channel_a->SignalOpened(dummy_channel_a.get());
>   // Closing a channel that is not opened should not affect the counts.
>   dummy_channel_b->SignalClosed(dummy_channel_b.get());
> 
>   {
>-    collector_->ClearCachedStatsReport();
>-    rtc::scoped_refptr<const RTCStatsReport> report = GetStatsReport();
>+    rtc::scoped_refptr<const RTCStatsReport> report =
>+        stats_->GetFreshStatsReport();
>     RTCPeerConnectionStats expected("RTCPeerConnection",
>                                     report->timestamp_us());
>     expected.data_channels_opened = 1;
>     expected.data_channels_closed = 0;
>     ASSERT_TRUE(report->Get("RTCPeerConnection"));
>-    EXPECT_EQ(expected,
>-              report->Get("RTCPeerConnection")->cast_to<
>-                  RTCPeerConnectionStats>());
>+    EXPECT_EQ(
>+        expected,
>+        report->Get("RTCPeerConnection")->cast_to<RTCPeerConnectionStats>());
>   }
> 
>   dummy_channel_b->SignalOpened(dummy_channel_b.get());
>   dummy_channel_b->SignalClosed(dummy_channel_b.get());
> 
>   {
>-    collector_->ClearCachedStatsReport();
>-    rtc::scoped_refptr<const RTCStatsReport> report = GetStatsReport();
>+    rtc::scoped_refptr<const RTCStatsReport> report =
>+        stats_->GetFreshStatsReport();
>     RTCPeerConnectionStats expected("RTCPeerConnection",
>                                     report->timestamp_us());
>     expected.data_channels_opened = 2;
>     expected.data_channels_closed = 1;
>     ASSERT_TRUE(report->Get("RTCPeerConnection"));
>-    EXPECT_EQ(expected,
>-              report->Get("RTCPeerConnection")->cast_to<
>-                  RTCPeerConnectionStats>());
>+    EXPECT_EQ(
>+        expected,
>+        report->Get("RTCPeerConnection")->cast_to<RTCPeerConnectionStats>());
>   }
> 
>   // Re-opening a data channel (or opening a new data channel that is re-using
>@@ -1504,52 +1290,47 @@ TEST_F(RTCStatsCollectorTest, CollectRTCPeerConnectionStats) {
>   dummy_channel_b->SignalOpened(dummy_channel_b.get());
> 
>   {
>-    collector_->ClearCachedStatsReport();
>-    rtc::scoped_refptr<const RTCStatsReport> report = GetStatsReport();
>+    rtc::scoped_refptr<const RTCStatsReport> report =
>+        stats_->GetFreshStatsReport();
>     RTCPeerConnectionStats expected("RTCPeerConnection",
>                                     report->timestamp_us());
>     expected.data_channels_opened = 3;
>     expected.data_channels_closed = 1;
>     ASSERT_TRUE(report->Get("RTCPeerConnection"));
>-    EXPECT_EQ(expected,
>-              report->Get("RTCPeerConnection")->cast_to<
>-                  RTCPeerConnectionStats>());
>+    EXPECT_EQ(
>+        expected,
>+        report->Get("RTCPeerConnection")->cast_to<RTCPeerConnectionStats>());
>   }
> 
>   dummy_channel_a->SignalClosed(dummy_channel_a.get());
>   dummy_channel_b->SignalClosed(dummy_channel_b.get());
> 
>   {
>-    collector_->ClearCachedStatsReport();
>-    rtc::scoped_refptr<const RTCStatsReport> report = GetStatsReport();
>+    rtc::scoped_refptr<const RTCStatsReport> report =
>+        stats_->GetFreshStatsReport();
>     RTCPeerConnectionStats expected("RTCPeerConnection",
>                                     report->timestamp_us());
>     expected.data_channels_opened = 3;
>     expected.data_channels_closed = 3;
>     ASSERT_TRUE(report->Get("RTCPeerConnection"));
>-    EXPECT_EQ(expected,
>-              report->Get("RTCPeerConnection")->cast_to<
>-                  RTCPeerConnectionStats>());
>+    EXPECT_EQ(
>+        expected,
>+        report->Get("RTCPeerConnection")->cast_to<RTCPeerConnectionStats>());
>   }
> }
> 
> TEST_F(RTCStatsCollectorTest,
>        CollectLocalRTCMediaStreamStatsAndRTCMediaStreamTrackStats_Audio) {
>-  rtc::scoped_refptr<StreamCollection> local_streams =
>-      StreamCollection::Create();
>-  EXPECT_CALL(test_->pc(), local_streams())
>-      .WillRepeatedly(Return(local_streams));
>-
>   rtc::scoped_refptr<MediaStream> local_stream =
>-      MediaStream::Create("LocalStreamLabel");
>-  local_streams->AddStream(local_stream);
>+      MediaStream::Create("LocalStreamId");
>+  pc_->mutable_local_streams()->AddStream(local_stream);
> 
>   // Local audio track
>   rtc::scoped_refptr<MediaStreamTrackInterface> local_audio_track =
>       CreateFakeTrack(cricket::MEDIA_TYPE_AUDIO, "LocalAudioTrackID",
>                       MediaStreamTrackInterface::kEnded);
>-  local_stream->AddTrack(static_cast<AudioTrackInterface*>(
>-      local_audio_track.get()));
>+  local_stream->AddTrack(
>+      static_cast<AudioTrackInterface*>(local_audio_track.get()));
> 
>   cricket::VoiceSenderInfo voice_sender_info_ssrc1;
>   voice_sender_info_ssrc1.local_stats.push_back(cricket::SsrcSenderInfo());
>@@ -1560,15 +1341,15 @@ TEST_F(RTCStatsCollectorTest,
>   voice_sender_info_ssrc1.apm_statistics.echo_return_loss = 42.0;
>   voice_sender_info_ssrc1.apm_statistics.echo_return_loss_enhancement = 52.0;
> 
>-  test_->CreateMockRtpSendersReceiversAndChannels(
>+  stats_->CreateMockRtpSendersReceiversAndChannels(
>       {std::make_pair(local_audio_track.get(), voice_sender_info_ssrc1)}, {},
>-      {}, {}, {local_stream->label()}, {});
>+      {}, {}, {local_stream->id()}, {});
> 
>-  rtc::scoped_refptr<const RTCStatsReport> report = GetStatsReport();
>+  rtc::scoped_refptr<const RTCStatsReport> report = stats_->GetStatsReport();
> 
>   RTCMediaStreamStats expected_local_stream(
>       IdForType<RTCMediaStreamStats>(report), report->timestamp_us());
>-  expected_local_stream.stream_identifier = local_stream->label();
>+  expected_local_stream.stream_identifier = local_stream->id();
>   expected_local_stream.track_ids = {
>       IdForType<RTCMediaStreamTrackStats>(report)};
>   ASSERT_TRUE(report->Get(expected_local_stream.id()))
>@@ -1600,21 +1381,16 @@ TEST_F(RTCStatsCollectorTest,
> 
> TEST_F(RTCStatsCollectorTest,
>        CollectRemoteRTCMediaStreamStatsAndRTCMediaStreamTrackStats_Audio) {
>-  rtc::scoped_refptr<StreamCollection> remote_streams =
>-      StreamCollection::Create();
>-  EXPECT_CALL(test_->pc(), remote_streams())
>-      .WillRepeatedly(Return(remote_streams));
>-
>   rtc::scoped_refptr<MediaStream> remote_stream =
>-      MediaStream::Create("RemoteStreamLabel");
>-  remote_streams->AddStream(remote_stream);
>+      MediaStream::Create("RemoteStreamId");
>+  pc_->mutable_remote_streams()->AddStream(remote_stream);
> 
>   // Remote audio track
>   rtc::scoped_refptr<MediaStreamTrackInterface> remote_audio_track =
>       CreateFakeTrack(cricket::MEDIA_TYPE_AUDIO, "RemoteAudioTrackID",
>                       MediaStreamTrackInterface::kLive);
>-  remote_stream->AddTrack(static_cast<AudioTrackInterface*>(
>-      remote_audio_track.get()));
>+  remote_stream->AddTrack(
>+      static_cast<AudioTrackInterface*>(remote_audio_track.get()));
> 
>   cricket::VoiceReceiverInfo voice_receiver_info;
>   voice_receiver_info.local_stats.push_back(cricket::SsrcReceiverInfo());
>@@ -1627,23 +1403,23 @@ TEST_F(RTCStatsCollectorTest,
>   voice_receiver_info.concealment_events = 12;
>   voice_receiver_info.jitter_buffer_delay_seconds = 3456;
> 
>-  test_->CreateMockRtpSendersReceiversAndChannels(
>+  stats_->CreateMockRtpSendersReceiversAndChannels(
>       {}, {std::make_pair(remote_audio_track.get(), voice_receiver_info)}, {},
>       {}, {}, {remote_stream});
> 
>-  rtc::scoped_refptr<const RTCStatsReport> report = GetStatsReport();
>+  rtc::scoped_refptr<const RTCStatsReport> report = stats_->GetStatsReport();
> 
>   RTCMediaStreamStats expected_remote_stream(
>       IdForType<RTCMediaStreamStats>(report), report->timestamp_us());
>-  expected_remote_stream.stream_identifier = remote_stream->label();
>+  expected_remote_stream.stream_identifier = remote_stream->id();
>   expected_remote_stream.track_ids =
>       std::vector<std::string>({IdForType<RTCMediaStreamTrackStats>(report)});
>   ASSERT_TRUE(report->Get(expected_remote_stream.id()))
>       << "Did not find " << expected_remote_stream.id() << " in "
>       << report->ToJson();
>-  EXPECT_EQ(expected_remote_stream,
>-            report->Get(expected_remote_stream.id())->cast_to<
>-                RTCMediaStreamStats>());
>+  EXPECT_EQ(
>+      expected_remote_stream,
>+      report->Get(expected_remote_stream.id())->cast_to<RTCMediaStreamStats>());
> 
>   RTCMediaStreamTrackStats expected_remote_audio_track(
>       IdForType<RTCMediaStreamTrackStats>(report), report->timestamp_us(),
>@@ -1661,27 +1437,22 @@ TEST_F(RTCStatsCollectorTest,
>   expected_remote_audio_track.jitter_buffer_delay = 3456;
>   ASSERT_TRUE(report->Get(expected_remote_audio_track.id()));
>   EXPECT_EQ(expected_remote_audio_track,
>-            report->Get(expected_remote_audio_track.id())->cast_to<
>-                RTCMediaStreamTrackStats>());
>+            report->Get(expected_remote_audio_track.id())
>+                ->cast_to<RTCMediaStreamTrackStats>());
> }
> 
> TEST_F(RTCStatsCollectorTest,
>        CollectLocalRTCMediaStreamStatsAndRTCMediaStreamTrackStats_Video) {
>-  rtc::scoped_refptr<StreamCollection> local_streams =
>-      StreamCollection::Create();
>-  EXPECT_CALL(test_->pc(), local_streams())
>-      .WillRepeatedly(Return(local_streams));
>-
>   rtc::scoped_refptr<MediaStream> local_stream =
>-      MediaStream::Create("LocalStreamLabel");
>-  local_streams->AddStream(local_stream);
>+      MediaStream::Create("LocalStreamId");
>+  pc_->mutable_local_streams()->AddStream(local_stream);
> 
>   // Local video track
>   rtc::scoped_refptr<MediaStreamTrackInterface> local_video_track =
>       CreateFakeTrack(cricket::MEDIA_TYPE_VIDEO, "LocalVideoTrackID",
>                       MediaStreamTrackInterface::kLive);
>-  local_stream->AddTrack(static_cast<VideoTrackInterface*>(
>-      local_video_track.get()));
>+  local_stream->AddTrack(
>+      static_cast<VideoTrackInterface*>(local_video_track.get()));
> 
>   cricket::VideoSenderInfo video_sender_info_ssrc1;
>   video_sender_info_ssrc1.local_stats.push_back(cricket::SsrcSenderInfo());
>@@ -1689,23 +1460,24 @@ TEST_F(RTCStatsCollectorTest,
>   video_sender_info_ssrc1.send_frame_width = 1234;
>   video_sender_info_ssrc1.send_frame_height = 4321;
>   video_sender_info_ssrc1.frames_encoded = 11;
>+  video_sender_info_ssrc1.huge_frames_sent = 1;
> 
>-  test_->CreateMockRtpSendersReceiversAndChannels(
>+  stats_->CreateMockRtpSendersReceiversAndChannels(
>       {}, {},
>       {std::make_pair(local_video_track.get(), video_sender_info_ssrc1)}, {},
>-      {local_stream->label()}, {});
>+      {local_stream->id()}, {});
> 
>-  rtc::scoped_refptr<const RTCStatsReport> report = GetStatsReport();
>+  rtc::scoped_refptr<const RTCStatsReport> report = stats_->GetStatsReport();
> 
>   auto stats_of_my_type = report->GetStatsOfType<RTCMediaStreamStats>();
>-  ASSERT_EQ(1, stats_of_my_type.size()) << "No stream in " << report->ToJson();
>+  ASSERT_EQ(1U, stats_of_my_type.size()) << "No stream in " << report->ToJson();
>   auto stats_of_track_type = report->GetStatsOfType<RTCMediaStreamTrackStats>();
>-  ASSERT_EQ(1, stats_of_track_type.size())
>+  ASSERT_EQ(1U, stats_of_track_type.size())
>       << "Wrong number of tracks in " << report->ToJson();
> 
>   RTCMediaStreamStats expected_local_stream(stats_of_my_type[0]->id(),
>                                             report->timestamp_us());
>-  expected_local_stream.stream_identifier = local_stream->label();
>+  expected_local_stream.stream_identifier = local_stream->id();
>   expected_local_stream.track_ids =
>       std::vector<std::string>({stats_of_track_type[0]->id()});
>   ASSERT_TRUE(report->Get(expected_local_stream.id()));
>@@ -1723,6 +1495,7 @@ TEST_F(RTCStatsCollectorTest,
>   expected_local_video_track_ssrc1.frame_width = 1234;
>   expected_local_video_track_ssrc1.frame_height = 4321;
>   expected_local_video_track_ssrc1.frames_sent = 11;
>+  expected_local_video_track_ssrc1.huge_frames_sent = 1;
>   ASSERT_TRUE(report->Get(expected_local_video_track_ssrc1.id()));
>   EXPECT_EQ(expected_local_video_track_ssrc1,
>             report->Get(expected_local_video_track_ssrc1.id())
>@@ -1731,21 +1504,16 @@ TEST_F(RTCStatsCollectorTest,
> 
> TEST_F(RTCStatsCollectorTest,
>        CollectRemoteRTCMediaStreamStatsAndRTCMediaStreamTrackStats_Video) {
>-  rtc::scoped_refptr<StreamCollection> remote_streams =
>-      StreamCollection::Create();
>-  EXPECT_CALL(test_->pc(), remote_streams())
>-      .WillRepeatedly(Return(remote_streams));
>-
>   rtc::scoped_refptr<MediaStream> remote_stream =
>-      MediaStream::Create("RemoteStreamLabel");
>-  remote_streams->AddStream(remote_stream);
>+      MediaStream::Create("RemoteStreamId");
>+  pc_->mutable_remote_streams()->AddStream(remote_stream);
> 
>   // Remote video track with values
>   rtc::scoped_refptr<MediaStreamTrackInterface> remote_video_track_ssrc3 =
>       CreateFakeTrack(cricket::MEDIA_TYPE_VIDEO, "RemoteVideoTrackID3",
>                       MediaStreamTrackInterface::kEnded);
>-  remote_stream->AddTrack(static_cast<VideoTrackInterface*>(
>-      remote_video_track_ssrc3.get()));
>+  remote_stream->AddTrack(
>+      static_cast<VideoTrackInterface*>(remote_video_track_ssrc3.get()));
> 
>   cricket::VideoReceiverInfo video_receiver_info_ssrc3;
>   video_receiver_info_ssrc3.local_stats.push_back(cricket::SsrcReceiverInfo());
>@@ -1756,30 +1524,30 @@ TEST_F(RTCStatsCollectorTest,
>   video_receiver_info_ssrc3.frames_decoded = 995;
>   video_receiver_info_ssrc3.frames_rendered = 990;
> 
>-  test_->CreateMockRtpSendersReceiversAndChannels(
>+  stats_->CreateMockRtpSendersReceiversAndChannels(
>       {}, {}, {},
>       {std::make_pair(remote_video_track_ssrc3.get(),
>                       video_receiver_info_ssrc3)},
>       {}, {remote_stream});
> 
>-  rtc::scoped_refptr<const RTCStatsReport> report = GetStatsReport();
>+  rtc::scoped_refptr<const RTCStatsReport> report = stats_->GetStatsReport();
> 
>   auto stats_of_my_type = report->GetStatsOfType<RTCMediaStreamStats>();
>-  ASSERT_EQ(1, stats_of_my_type.size()) << "No stream in " << report->ToJson();
>+  ASSERT_EQ(1U, stats_of_my_type.size()) << "No stream in " << report->ToJson();
>   auto stats_of_track_type = report->GetStatsOfType<RTCMediaStreamTrackStats>();
>-  ASSERT_EQ(1, stats_of_track_type.size())
>+  ASSERT_EQ(1U, stats_of_track_type.size())
>       << "Wrong number of tracks in " << report->ToJson();
>   ASSERT_TRUE(*(stats_of_track_type[0]->remote_source));
> 
>   RTCMediaStreamStats expected_remote_stream(stats_of_my_type[0]->id(),
>                                              report->timestamp_us());
>-  expected_remote_stream.stream_identifier = remote_stream->label();
>+  expected_remote_stream.stream_identifier = remote_stream->id();
>   expected_remote_stream.track_ids =
>       std::vector<std::string>({stats_of_track_type[0]->id()});
>   ASSERT_TRUE(report->Get(expected_remote_stream.id()));
>-  EXPECT_EQ(expected_remote_stream,
>-            report->Get(expected_remote_stream.id())->cast_to<
>-                RTCMediaStreamStats>());
>+  EXPECT_EQ(
>+      expected_remote_stream,
>+      report->Get(expected_remote_stream.id())->cast_to<RTCMediaStreamStats>());
> 
>   RTCMediaStreamTrackStats expected_remote_video_track_ssrc3(
>       stats_of_track_type[0]->id(), report->timestamp_us(),
>@@ -1796,22 +1564,11 @@ TEST_F(RTCStatsCollectorTest,
>   expected_remote_video_track_ssrc3.frames_dropped = 1000 - 990;
>   ASSERT_TRUE(report->Get(expected_remote_video_track_ssrc3.id()));
>   EXPECT_EQ(expected_remote_video_track_ssrc3,
>-            report->Get(expected_remote_video_track_ssrc3.id())->cast_to<
>-                RTCMediaStreamTrackStats>());
>+            report->Get(expected_remote_video_track_ssrc3.id())
>+                ->cast_to<RTCMediaStreamTrackStats>());
> }
> 
> TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Audio) {
>-  auto* voice_media_channel = new MockVoiceMediaChannel();
>-  cricket::VoiceChannel voice_channel(
>-      test_->worker_thread(), test_->network_thread(),
>-      test_->signaling_thread(), test_->media_engine(),
>-      rtc::WrapUnique(voice_media_channel), "VoiceContentName",
>-      kDefaultRtcpMuxRequired, kDefaultSrtpRequired);
>-  voice_channel.set_transport_name_for_testing("TransportName");
>-
>-  test_->SetupRemoteTrackAndReceiver(
>-      cricket::MEDIA_TYPE_AUDIO, "RemoteAudioTrackID", 1);
>-
>   cricket::VoiceMediaInfo voice_media_info;
> 
>   voice_media_info.receivers.push_back(cricket::VoiceReceiverInfo());
>@@ -1833,40 +1590,24 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Audio) {
>   voice_media_info.receive_codecs.insert(
>       std::make_pair(codec_parameters.payload_type, codec_parameters));
> 
>-  EXPECT_CALL(*voice_media_channel, GetStats(_))
>-      .WillOnce(DoAll(SetArgPointee<0>(voice_media_info), Return(true)));
>-
>-  SessionStats session_stats;
>-  session_stats.transport_stats["TransportName"].transport_name =
>-      "TransportName";
>-
>-  // Make sure the associated |RTCTransportStats| is created.
>-  cricket::TransportChannelStats channel_stats;
>-  channel_stats.component = cricket::ICE_CANDIDATE_COMPONENT_RTP;
>-  session_stats.transport_stats["TransportName"].channel_stats.push_back(
>-      channel_stats);
>-
>-  EXPECT_CALL(test_->pc(), GetSessionStats(_))
>-      .WillRepeatedly(Invoke([&session_stats](const ChannelNamePairs&) {
>-        return std::unique_ptr<SessionStats>(new SessionStats(session_stats));
>-      }));
>-  EXPECT_CALL(test_->pc(), voice_channel())
>-      .WillRepeatedly(Return(&voice_channel));
>+  auto* voice_media_channel = pc_->AddVoiceChannel("AudioMid", "TransportName");
>+  voice_media_channel->SetStats(voice_media_info);
>+  stats_->SetupRemoteTrackAndReceiver(
>+      cricket::MEDIA_TYPE_AUDIO, "RemoteAudioTrackID", "RemoteStreamId", 1);
> 
>-  rtc::scoped_refptr<const RTCStatsReport> report = GetStatsReport();
>+  rtc::scoped_refptr<const RTCStatsReport> report = stats_->GetStatsReport();
> 
>   auto stats_of_track_type = report->GetStatsOfType<RTCMediaStreamTrackStats>();
>-  ASSERT_EQ(1, stats_of_track_type.size());
>+  ASSERT_EQ(1U, stats_of_track_type.size());
> 
>-  RTCInboundRTPStreamStats expected_audio(
>-      "RTCInboundRTPAudioStream_1", report->timestamp_us());
>+  RTCInboundRTPStreamStats expected_audio("RTCInboundRTPAudioStream_1",
>+                                          report->timestamp_us());
>   expected_audio.ssrc = 1;
>   expected_audio.is_remote = false;
>   expected_audio.media_type = "audio";
>   expected_audio.track_id = stats_of_track_type[0]->id();
>-  expected_audio.transport_id = "RTCTransport_TransportName_" +
>-      rtc::ToString<>(cricket::ICE_CANDIDATE_COMPONENT_RTP);
>-  expected_audio.codec_id = "RTCCodec_InboundAudio_42";
>+  expected_audio.transport_id = "RTCTransport_TransportName_1";
>+  expected_audio.codec_id = "RTCCodec_AudioMid_Inbound_42";
>   expected_audio.packets_received = 2;
>   expected_audio.bytes_received = 3;
>   expected_audio.packets_lost = -1;
>@@ -1882,16 +1623,6 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Audio) {
> }
> 
> TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Video) {
>-  auto* video_media_channel = new MockVideoMediaChannel();
>-  cricket::VideoChannel video_channel(
>-      test_->worker_thread(), test_->network_thread(),
>-      test_->signaling_thread(), rtc::WrapUnique(video_media_channel),
>-      "VideoContentName", kDefaultRtcpMuxRequired, kDefaultSrtpRequired);
>-  video_channel.set_transport_name_for_testing("TransportName");
>-
>-  test_->SetupRemoteTrackAndReceiver(
>-      cricket::MEDIA_TYPE_VIDEO, "RemoteVideoTrackID", 1);
>-
>   cricket::VideoMediaInfo video_media_info;
> 
>   video_media_info.receivers.push_back(cricket::VideoReceiverInfo());
>@@ -1907,7 +1638,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Video) {
>   video_media_info.receivers[0].plis_sent = 6;
>   video_media_info.receivers[0].nacks_sent = 7;
>   video_media_info.receivers[0].frames_decoded = 8;
>-  video_media_info.receivers[0].qp_sum = rtc::nullopt;
>+  video_media_info.receivers[0].qp_sum = absl::nullopt;
> 
>   RtpCodecParameters codec_parameters;
>   codec_parameters.payload_type = 42;
>@@ -1917,37 +1648,21 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Video) {
>   video_media_info.receive_codecs.insert(
>       std::make_pair(codec_parameters.payload_type, codec_parameters));
> 
>-  EXPECT_CALL(*video_media_channel, GetStats(_))
>-      .WillOnce(DoAll(SetArgPointee<0>(video_media_info), Return(true)));
>-
>-  SessionStats session_stats;
>-  session_stats.transport_stats["TransportName"].transport_name =
>-      "TransportName";
>-
>-  // Make sure the associated |RTCTransportStats| is created.
>-  cricket::TransportChannelStats channel_stats;
>-  channel_stats.component = cricket::ICE_CANDIDATE_COMPONENT_RTP;
>-  session_stats.transport_stats["TransportName"].channel_stats.push_back(
>-      channel_stats);
>-
>-  EXPECT_CALL(test_->pc(), GetSessionStats(_))
>-      .WillRepeatedly(Invoke([&session_stats](const ChannelNamePairs&) {
>-        return std::unique_ptr<SessionStats>(new SessionStats(session_stats));
>-      }));
>-  EXPECT_CALL(test_->pc(), video_channel())
>-      .WillRepeatedly(Return(&video_channel));
>+  auto* video_media_channel = pc_->AddVideoChannel("VideoMid", "TransportName");
>+  video_media_channel->SetStats(video_media_info);
>+  stats_->SetupRemoteTrackAndReceiver(
>+      cricket::MEDIA_TYPE_VIDEO, "RemoteVideoTrackID", "RemoteStreamId", 1);
> 
>-  rtc::scoped_refptr<const RTCStatsReport> report = GetStatsReport();
>+  rtc::scoped_refptr<const RTCStatsReport> report = stats_->GetStatsReport();
> 
>-  RTCInboundRTPStreamStats expected_video(
>-      "RTCInboundRTPVideoStream_1", report->timestamp_us());
>+  RTCInboundRTPStreamStats expected_video("RTCInboundRTPVideoStream_1",
>+                                          report->timestamp_us());
>   expected_video.ssrc = 1;
>   expected_video.is_remote = false;
>   expected_video.media_type = "video";
>   expected_video.track_id = IdForType<RTCMediaStreamTrackStats>(report);
>-  expected_video.transport_id = "RTCTransport_TransportName_" +
>-      rtc::ToString<>(cricket::ICE_CANDIDATE_COMPONENT_RTP);
>-  expected_video.codec_id = "RTCCodec_InboundVideo_42";
>+  expected_video.transport_id = "RTCTransport_TransportName_1";
>+  expected_video.codec_id = "RTCCodec_VideoMid_Inbound_42";
>   expected_video.fir_count = 5;
>   expected_video.pli_count = 6;
>   expected_video.nack_count = 7;
>@@ -1966,11 +1681,9 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Video) {
>   // Set previously undefined values and "GetStats" again.
>   video_media_info.receivers[0].qp_sum = 9;
>   expected_video.qp_sum = 9;
>+  video_media_channel->SetStats(video_media_info);
> 
>-  EXPECT_CALL(*video_media_channel, GetStats(_))
>-      .WillOnce(DoAll(SetArgPointee<0>(video_media_info), Return(true)));
>-  collector_->ClearCachedStatsReport();
>-  report = GetStatsReport();
>+  report = stats_->GetFreshStatsReport();
> 
>   ASSERT_TRUE(report->Get(expected_video.id()));
>   EXPECT_EQ(
>@@ -1982,17 +1695,6 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Video) {
> }
> 
> TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Audio) {
>-  auto* voice_media_channel = new MockVoiceMediaChannel();
>-  cricket::VoiceChannel voice_channel(
>-      test_->worker_thread(), test_->network_thread(),
>-      test_->signaling_thread(), test_->media_engine(),
>-      rtc::WrapUnique(voice_media_channel), "VoiceContentName",
>-      kDefaultRtcpMuxRequired, kDefaultSrtpRequired);
>-  voice_channel.set_transport_name_for_testing("TransportName");
>-
>-  test_->SetupLocalTrackAndSender(cricket::MEDIA_TYPE_AUDIO,
>-                                  "LocalAudioTrackID", 1, true);
>-
>   cricket::VoiceMediaInfo voice_media_info;
> 
>   voice_media_info.senders.push_back(cricket::VoiceSenderInfo());
>@@ -2010,37 +1712,21 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Audio) {
>   voice_media_info.send_codecs.insert(
>       std::make_pair(codec_parameters.payload_type, codec_parameters));
> 
>-  EXPECT_CALL(*voice_media_channel, GetStats(_))
>-      .WillOnce(DoAll(SetArgPointee<0>(voice_media_info), Return(true)));
>-
>-  SessionStats session_stats;
>-  session_stats.transport_stats["TransportName"].transport_name =
>-      "TransportName";
>+  auto* voice_media_channel = pc_->AddVoiceChannel("AudioMid", "TransportName");
>+  voice_media_channel->SetStats(voice_media_info);
>+  stats_->SetupLocalTrackAndSender(cricket::MEDIA_TYPE_AUDIO,
>+                                   "LocalAudioTrackID", 1, true);
> 
>-  // Make sure the associated |RTCTransportStats| is created.
>-  cricket::TransportChannelStats channel_stats;
>-  channel_stats.component = cricket::ICE_CANDIDATE_COMPONENT_RTP;
>-  session_stats.transport_stats["TransportName"].channel_stats.push_back(
>-      channel_stats);
>+  rtc::scoped_refptr<const RTCStatsReport> report = stats_->GetStatsReport();
> 
>-  EXPECT_CALL(test_->pc(), GetSessionStats(_))
>-      .WillRepeatedly(Invoke([&session_stats](const ChannelNamePairs&) {
>-        return std::unique_ptr<SessionStats>(new SessionStats(session_stats));
>-      }));
>-  EXPECT_CALL(test_->pc(), voice_channel())
>-      .WillRepeatedly(Return(&voice_channel));
>-
>-  rtc::scoped_refptr<const RTCStatsReport> report = GetStatsReport();
>-
>-  RTCOutboundRTPStreamStats expected_audio(
>-      "RTCOutboundRTPAudioStream_1", report->timestamp_us());
>+  RTCOutboundRTPStreamStats expected_audio("RTCOutboundRTPAudioStream_1",
>+                                           report->timestamp_us());
>   expected_audio.ssrc = 1;
>   expected_audio.is_remote = false;
>   expected_audio.media_type = "audio";
>   expected_audio.track_id = IdForType<RTCMediaStreamTrackStats>(report);
>-  expected_audio.transport_id = "RTCTransport_TransportName_" +
>-      rtc::ToString<>(cricket::ICE_CANDIDATE_COMPONENT_RTP);
>-  expected_audio.codec_id = "RTCCodec_OutboundAudio_42";
>+  expected_audio.transport_id = "RTCTransport_TransportName_1";
>+  expected_audio.codec_id = "RTCCodec_AudioMid_Outbound_42";
>   expected_audio.packets_sent = 2;
>   expected_audio.bytes_sent = 3;
> 
>@@ -2059,16 +1745,6 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Audio) {
> }
> 
> TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Video) {
>-  auto* video_media_channel = new MockVideoMediaChannel();
>-  cricket::VideoChannel video_channel(
>-      test_->worker_thread(), test_->network_thread(),
>-      test_->signaling_thread(), rtc::WrapUnique(video_media_channel),
>-      "VideoContentName", kDefaultRtcpMuxRequired, kDefaultSrtpRequired);
>-  video_channel.set_transport_name_for_testing("TransportName");
>-
>-  test_->SetupLocalTrackAndSender(cricket::MEDIA_TYPE_VIDEO,
>-                                  "LocalVideoTrackID", 1, true);
>-
>   cricket::VideoMediaInfo video_media_info;
> 
>   video_media_info.senders.push_back(cricket::VideoSenderInfo());
>@@ -2081,7 +1757,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Video) {
>   video_media_info.senders[0].bytes_sent = 6;
>   video_media_info.senders[0].codec_payload_type = 42;
>   video_media_info.senders[0].frames_encoded = 8;
>-  video_media_info.senders[0].qp_sum = rtc::nullopt;
>+  video_media_info.senders[0].qp_sum = absl::nullopt;
> 
>   RtpCodecParameters codec_parameters;
>   codec_parameters.payload_type = 42;
>@@ -2091,32 +1767,17 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Video) {
>   video_media_info.send_codecs.insert(
>       std::make_pair(codec_parameters.payload_type, codec_parameters));
> 
>-  EXPECT_CALL(*video_media_channel, GetStats(_))
>-      .WillOnce(DoAll(SetArgPointee<0>(video_media_info), Return(true)));
>-
>-  SessionStats session_stats;
>-  session_stats.transport_stats["TransportName"].transport_name =
>-      "TransportName";
>-
>-  // Make sure the associated |RTCTransportStats| is created.
>-  cricket::TransportChannelStats channel_stats;
>-  channel_stats.component = cricket::ICE_CANDIDATE_COMPONENT_RTP;
>-  session_stats.transport_stats["TransportName"].channel_stats.push_back(
>-      channel_stats);
>+  auto* video_media_channel = pc_->AddVideoChannel("VideoMid", "TransportName");
>+  video_media_channel->SetStats(video_media_info);
>+  stats_->SetupLocalTrackAndSender(cricket::MEDIA_TYPE_VIDEO,
>+                                   "LocalVideoTrackID", 1, true);
> 
>-  EXPECT_CALL(test_->pc(), GetSessionStats(_))
>-      .WillRepeatedly(Invoke([&session_stats](const ChannelNamePairs&) {
>-        return std::unique_ptr<SessionStats>(new SessionStats(session_stats));
>-      }));
>-  EXPECT_CALL(test_->pc(), video_channel())
>-      .WillRepeatedly(Return(&video_channel));
>-
>-  rtc::scoped_refptr<const RTCStatsReport> report = GetStatsReport();
>+  rtc::scoped_refptr<const RTCStatsReport> report = stats_->GetStatsReport();
> 
>   auto stats_of_my_type = report->GetStatsOfType<RTCOutboundRTPStreamStats>();
>-  ASSERT_EQ(1, stats_of_my_type.size());
>+  ASSERT_EQ(1U, stats_of_my_type.size());
>   auto stats_of_track_type = report->GetStatsOfType<RTCMediaStreamTrackStats>();
>-  ASSERT_EQ(1, stats_of_track_type.size());
>+  ASSERT_EQ(1U, stats_of_track_type.size());
> 
>   RTCOutboundRTPStreamStats expected_video(stats_of_my_type[0]->id(),
>                                            report->timestamp_us());
>@@ -2124,9 +1785,8 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Video) {
>   expected_video.is_remote = false;
>   expected_video.media_type = "video";
>   expected_video.track_id = stats_of_track_type[0]->id();
>-  expected_video.transport_id = "RTCTransport_TransportName_" +
>-      rtc::ToString<>(cricket::ICE_CANDIDATE_COMPONENT_RTP);
>-  expected_video.codec_id = "RTCCodec_OutboundVideo_42";
>+  expected_video.transport_id = "RTCTransport_TransportName_1";
>+  expected_video.codec_id = "RTCCodec_VideoMid_Outbound_42";
>   expected_video.fir_count = 2;
>   expected_video.pli_count = 3;
>   expected_video.nack_count = 4;
>@@ -2143,11 +1803,9 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Video) {
>   // Set previously undefined values and "GetStats" again.
>   video_media_info.senders[0].qp_sum = 9;
>   expected_video.qp_sum = 9;
>+  video_media_channel->SetStats(video_media_info);
> 
>-  EXPECT_CALL(*video_media_channel, GetStats(_))
>-      .WillOnce(DoAll(SetArgPointee<0>(video_media_info), Return(true)));
>-  collector_->ClearCachedStatsReport();
>-  report = GetStatsReport();
>+  report = stats_->GetFreshStatsReport();
> 
>   ASSERT_TRUE(report->Get(expected_video.id()));
>   EXPECT_EQ(
>@@ -2159,6 +1817,10 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Video) {
> }
> 
> TEST_F(RTCStatsCollectorTest, CollectRTCTransportStats) {
>+  const char kTransportName[] = "transport";
>+
>+  pc_->AddVoiceChannel("audio", kTransportName);
>+
>   std::unique_ptr<cricket::Candidate> rtp_local_candidate =
>       CreateFakeCandidate("42.42.42.42", 42, "protocol", rtc::ADAPTER_TYPE_WIFI,
>                           cricket::LOCAL_PORT_TYPE, 42);
>@@ -2174,9 +1836,6 @@ TEST_F(RTCStatsCollectorTest, CollectRTCTransportStats) {
>                           rtc::ADAPTER_TYPE_UNKNOWN, cricket::LOCAL_PORT_TYPE,
>                           42);
> 
>-  SessionStats session_stats;
>-  session_stats.transport_stats["transport"].transport_name = "transport";
>-
>   cricket::ConnectionInfo rtp_connection_info;
>   rtp_connection_info.best_connection = false;
>   rtp_connection_info.local_candidate = *rtp_local_candidate.get();
>@@ -2187,22 +1846,14 @@ TEST_F(RTCStatsCollectorTest, CollectRTCTransportStats) {
>   rtp_transport_channel_stats.component = cricket::ICE_CANDIDATE_COMPONENT_RTP;
>   rtp_transport_channel_stats.connection_infos.push_back(rtp_connection_info);
>   rtp_transport_channel_stats.dtls_state = cricket::DTLS_TRANSPORT_NEW;
>-  session_stats.transport_stats["transport"].channel_stats.push_back(
>-      rtp_transport_channel_stats);
>-
>-
>-  // Mock the session to return the desired candidates.
>-  EXPECT_CALL(test_->pc(), GetSessionStats(_))
>-      .WillRepeatedly(Invoke([&session_stats](const ChannelNamePairs&) {
>-        return std::unique_ptr<SessionStats>(new SessionStats(session_stats));
>-      }));
>+  pc_->SetTransportStats(kTransportName, {rtp_transport_channel_stats});
> 
>   // Get stats without RTCP, an active connection or certificates.
>-  rtc::scoped_refptr<const RTCStatsReport> report = GetStatsReport();
>+  rtc::scoped_refptr<const RTCStatsReport> report = stats_->GetStatsReport();
> 
>   RTCTransportStats expected_rtp_transport(
>       "RTCTransport_transport_" +
>-          rtc::ToString<>(cricket::ICE_CANDIDATE_COMPONENT_RTP),
>+          rtc::ToString(cricket::ICE_CANDIDATE_COMPONENT_RTP),
>       report->timestamp_us());
>   expected_rtp_transport.bytes_sent = 42;
>   expected_rtp_transport.bytes_received = 1337;
>@@ -2224,16 +1875,15 @@ TEST_F(RTCStatsCollectorTest, CollectRTCTransportStats) {
>       cricket::ICE_CANDIDATE_COMPONENT_RTCP;
>   rtcp_transport_channel_stats.connection_infos.push_back(rtcp_connection_info);
>   rtcp_transport_channel_stats.dtls_state = cricket::DTLS_TRANSPORT_CONNECTING;
>-  session_stats.transport_stats["transport"].channel_stats.push_back(
>-      rtcp_transport_channel_stats);
>+  pc_->SetTransportStats(kTransportName, {rtp_transport_channel_stats,
>+                                          rtcp_transport_channel_stats});
> 
>-  collector_->ClearCachedStatsReport();
>   // Get stats with RTCP and without an active connection or certificates.
>-  report = GetStatsReport();
>+  report = stats_->GetFreshStatsReport();
> 
>   RTCTransportStats expected_rtcp_transport(
>       "RTCTransport_transport_" +
>-          rtc::ToString<>(cricket::ICE_CANDIDATE_COMPONENT_RTCP),
>+          rtc::ToString(cricket::ICE_CANDIDATE_COMPONENT_RTCP),
>       report->timestamp_us());
>   expected_rtcp_transport.bytes_sent = 1337;
>   expected_rtcp_transport.bytes_received = 42;
>@@ -2251,13 +1901,11 @@ TEST_F(RTCStatsCollectorTest, CollectRTCTransportStats) {
>       report->Get(expected_rtcp_transport.id())->cast_to<RTCTransportStats>());
> 
>   // Get stats with an active connection (selected candidate pair).
>-  session_stats.transport_stats["transport"]
>-      .channel_stats[1]
>-      .connection_infos[0]
>-      .best_connection = true;
>+  rtcp_transport_channel_stats.connection_infos[0].best_connection = true;
>+  pc_->SetTransportStats(kTransportName, {rtp_transport_channel_stats,
>+                                          rtcp_transport_channel_stats});
> 
>-  collector_->ClearCachedStatsReport();
>-  report = GetStatsReport();
>+  report = stats_->GetFreshStatsReport();
> 
>   expected_rtcp_transport.selected_candidate_pair_id =
>       "RTCIceCandidatePair_" + rtcp_local_candidate->id() + "_" +
>@@ -2274,32 +1922,16 @@ TEST_F(RTCStatsCollectorTest, CollectRTCTransportStats) {
> 
>   // Get stats with certificates.
>   std::unique_ptr<CertificateInfo> local_certinfo =
>-      CreateFakeCertificateAndInfoFromDers(
>-          std::vector<std::string>({ "(local) local", "(local) chain" }));
>+      CreateFakeCertificateAndInfoFromDers({"(local) local", "(local) chain"});
>+  pc_->SetLocalCertificate(kTransportName, local_certinfo->certificate);
>   std::unique_ptr<CertificateInfo> remote_certinfo =
>       CreateFakeCertificateAndInfoFromDers(
>-          std::vector<std::string>({ "(remote) local", "(remote) chain" }));
>-  EXPECT_CALL(test_->pc(), GetLocalCertificate(_, _))
>-      .WillRepeatedly(
>-          Invoke([&local_certinfo](
>-                     const std::string& transport_name,
>-                     rtc::scoped_refptr<rtc::RTCCertificate>* certificate) {
>-            if (transport_name == "transport") {
>-              *certificate = local_certinfo->certificate;
>-              return true;
>-            }
>-            return false;
>-          }));
>-  EXPECT_CALL(test_->pc(), GetRemoteSSLCertificate_ReturnsRawPointer(_))
>-      .WillRepeatedly(Invoke([&remote_certinfo](
>-                                 const std::string& transport_name) {
>-        if (transport_name == "transport")
>-          return remote_certinfo->certificate->ssl_certificate().GetReference();
>-        return static_cast<rtc::SSLCertificate*>(nullptr);
>-      }));
>-
>-  collector_->ClearCachedStatsReport();
>-  report = GetStatsReport();
>+          {"(remote) local", "(remote) chain"});
>+  pc_->SetRemoteCertChain(
>+      kTransportName,
>+      remote_certinfo->certificate->ssl_cert_chain().UniqueCopy());
>+
>+  report = stats_->GetFreshStatsReport();
> 
>   expected_rtp_transport.local_certificate_id =
>       "RTCCertificate_" + local_certinfo->fingerprints[0];
>@@ -2322,18 +1954,6 @@ TEST_F(RTCStatsCollectorTest, CollectRTCTransportStats) {
> }
> 
> TEST_F(RTCStatsCollectorTest, CollectNoStreamRTCOutboundRTPStreamStats_Audio) {
>-  // Emulates the case where AddTrack is used without an associated MediaStream
>-  auto* voice_media_channel = new MockVoiceMediaChannel();
>-  cricket::VoiceChannel voice_channel(
>-      test_->worker_thread(), test_->network_thread(),
>-      test_->signaling_thread(), test_->media_engine(),
>-      rtc::WrapUnique(voice_media_channel), "VoiceContentName",
>-      kDefaultRtcpMuxRequired, kDefaultSrtpRequired);
>-  voice_channel.set_transport_name_for_testing("TransportName");
>-
>-  test_->SetupLocalTrackAndSender(cricket::MEDIA_TYPE_AUDIO,
>-                                  "LocalAudioTrackID", 1, false);
>-
>   cricket::VoiceMediaInfo voice_media_info;
> 
>   voice_media_info.senders.push_back(cricket::VoiceSenderInfo());
>@@ -2351,27 +1971,13 @@ TEST_F(RTCStatsCollectorTest, CollectNoStreamRTCOutboundRTPStreamStats_Audio) {
>   voice_media_info.send_codecs.insert(
>       std::make_pair(codec_parameters.payload_type, codec_parameters));
> 
>-  EXPECT_CALL(*voice_media_channel, GetStats(_))
>-      .WillOnce(DoAll(SetArgPointee<0>(voice_media_info), Return(true)));
>-
>-  SessionStats session_stats;
>-  session_stats.transport_stats["TransportName"].transport_name =
>-      "TransportName";
>-
>-  // Make sure the associated |RTCTransportStats| is created.
>-  cricket::TransportChannelStats channel_stats;
>-  channel_stats.component = cricket::ICE_CANDIDATE_COMPONENT_RTP;
>-  session_stats.transport_stats["TransportName"].channel_stats.push_back(
>-      channel_stats);
>-
>-  EXPECT_CALL(test_->pc(), GetSessionStats(_))
>-      .WillRepeatedly(Invoke([&session_stats](const ChannelNamePairs&) {
>-        return std::unique_ptr<SessionStats>(new SessionStats(session_stats));
>-      }));
>-  EXPECT_CALL(test_->pc(), voice_channel())
>-      .WillRepeatedly(Return(&voice_channel));
>+  // Emulates the case where AddTrack is used without an associated MediaStream
>+  auto* voice_media_channel = pc_->AddVoiceChannel("AudioMid", "TransportName");
>+  voice_media_channel->SetStats(voice_media_info);
>+  stats_->SetupLocalTrackAndSender(cricket::MEDIA_TYPE_AUDIO,
>+                                   "LocalAudioTrackID", 1, false);
> 
>-  rtc::scoped_refptr<const RTCStatsReport> report = GetStatsReport();
>+  rtc::scoped_refptr<const RTCStatsReport> report = stats_->GetStatsReport();
> 
>   RTCOutboundRTPStreamStats expected_audio("RTCOutboundRTPAudioStream_1",
>                                            report->timestamp_us());
>@@ -2379,10 +1985,8 @@ TEST_F(RTCStatsCollectorTest, CollectNoStreamRTCOutboundRTPStreamStats_Audio) {
>   expected_audio.is_remote = false;
>   expected_audio.media_type = "audio";
>   expected_audio.track_id = IdForType<RTCMediaStreamTrackStats>(report);
>-  expected_audio.transport_id =
>-      "RTCTransport_TransportName_" +
>-      rtc::ToString<>(cricket::ICE_CANDIDATE_COMPONENT_RTP);
>-  expected_audio.codec_id = "RTCCodec_OutboundAudio_42";
>+  expected_audio.transport_id = "RTCTransport_TransportName_1";
>+  expected_audio.codec_id = "RTCCodec_AudioMid_Outbound_42";
>   expected_audio.packets_sent = 2;
>   expected_audio.bytes_sent = 3;
> 
>@@ -2395,56 +1999,255 @@ TEST_F(RTCStatsCollectorTest, CollectNoStreamRTCOutboundRTPStreamStats_Audio) {
>   EXPECT_TRUE(report->Get(*expected_audio.codec_id));
> }
> 
>+TEST_F(RTCStatsCollectorTest, GetStatsWithSenderSelector) {
>+  ExampleStatsGraph graph = SetupExampleStatsGraphForSelectorTests();
>+  // Expected stats graph when filtered by sender:
>+  //
>+  // track (sender)
>+  //          ^
>+  //          |
>+  //         outbound-rtp
>+  //          |        |
>+  //          v        v
>+  // codec (send)     transport
>+  rtc::scoped_refptr<const RTCStatsReport> sender_report =
>+      stats_->GetStatsReportWithSenderSelector(graph.sender);
>+  EXPECT_TRUE(sender_report);
>+  EXPECT_EQ(sender_report->timestamp_us(), graph.full_report->timestamp_us());
>+  EXPECT_EQ(sender_report->size(), 4u);
>+  EXPECT_TRUE(sender_report->Get(graph.send_codec_id));
>+  EXPECT_FALSE(sender_report->Get(graph.recv_codec_id));
>+  EXPECT_TRUE(sender_report->Get(graph.outbound_rtp_id));
>+  EXPECT_FALSE(sender_report->Get(graph.inbound_rtp_id));
>+  EXPECT_TRUE(sender_report->Get(graph.transport_id));
>+  EXPECT_TRUE(sender_report->Get(graph.sender_track_id));
>+  EXPECT_FALSE(sender_report->Get(graph.receiver_track_id));
>+  EXPECT_FALSE(sender_report->Get(graph.remote_stream_id));
>+  EXPECT_FALSE(sender_report->Get(graph.peer_connection_id));
>+}
>+
>+TEST_F(RTCStatsCollectorTest, GetStatsWithReceiverSelector) {
>+  ExampleStatsGraph graph = SetupExampleStatsGraphForSelectorTests();
>+  // Expected stats graph when filtered by receiver:
>+  //
>+  //                                                 track (receiver)
>+  //                                                   ^
>+  //                                                   |
>+  //                        inbound-rtp ---------------+
>+  //                         |       |
>+  //                         v       v
>+  //                  transport     codec (recv)
>+  rtc::scoped_refptr<const RTCStatsReport> receiver_report =
>+      stats_->GetStatsReportWithReceiverSelector(graph.receiver);
>+  EXPECT_TRUE(receiver_report);
>+  EXPECT_EQ(receiver_report->size(), 4u);
>+  EXPECT_EQ(receiver_report->timestamp_us(), graph.full_report->timestamp_us());
>+  EXPECT_FALSE(receiver_report->Get(graph.send_codec_id));
>+  EXPECT_TRUE(receiver_report->Get(graph.recv_codec_id));
>+  EXPECT_FALSE(receiver_report->Get(graph.outbound_rtp_id));
>+  EXPECT_TRUE(receiver_report->Get(graph.inbound_rtp_id));
>+  EXPECT_TRUE(receiver_report->Get(graph.transport_id));
>+  EXPECT_FALSE(receiver_report->Get(graph.sender_track_id));
>+  EXPECT_TRUE(receiver_report->Get(graph.receiver_track_id));
>+  EXPECT_FALSE(receiver_report->Get(graph.remote_stream_id));
>+  EXPECT_FALSE(receiver_report->Get(graph.peer_connection_id));
>+}
>+
>+TEST_F(RTCStatsCollectorTest, GetStatsWithNullSenderSelector) {
>+  ExampleStatsGraph graph = SetupExampleStatsGraphForSelectorTests();
>+  rtc::scoped_refptr<const RTCStatsReport> empty_report =
>+      stats_->GetStatsReportWithSenderSelector(nullptr);
>+  EXPECT_TRUE(empty_report);
>+  EXPECT_EQ(empty_report->timestamp_us(), graph.full_report->timestamp_us());
>+  EXPECT_EQ(empty_report->size(), 0u);
>+}
>+
>+TEST_F(RTCStatsCollectorTest, GetStatsWithNullReceiverSelector) {
>+  ExampleStatsGraph graph = SetupExampleStatsGraphForSelectorTests();
>+  rtc::scoped_refptr<const RTCStatsReport> empty_report =
>+      stats_->GetStatsReportWithReceiverSelector(nullptr);
>+  EXPECT_TRUE(empty_report);
>+  EXPECT_EQ(empty_report->timestamp_us(), graph.full_report->timestamp_us());
>+  EXPECT_EQ(empty_report->size(), 0u);
>+}
>+
> // When the PC has not had SetLocalDescription done, tracks all have
> // SSRC 0, meaning "unconnected".
>-// We do not report stats on those tracks. https://bugs.webrtc.org/8673
>-TEST_F(RTCStatsCollectorTest, StatsNotReportedOnZeroSsrc) {
>+// In this state, we report on track stats, but not RTP stats.
>+TEST_F(RTCStatsCollectorTest, StatsReportedOnZeroSsrc) {
>   rtc::scoped_refptr<MediaStreamTrackInterface> track =
>       CreateFakeTrack(cricket::MEDIA_TYPE_AUDIO, "audioTrack",
>                       MediaStreamTrackInterface::kLive);
>-  rtc::scoped_refptr<MockRtpSender> sender = CreateMockSender(track, 0, 49, {});
>-  EXPECT_CALL(test_->pc(), GetSenders())
>-      .WillRepeatedly(
>-          Return(std::vector<rtc::scoped_refptr<RtpSenderInterface>>(
>-              {rtc::scoped_refptr<RtpSenderInterface>(sender.get())})));
>-  rtc::scoped_refptr<const RTCStatsReport> report = GetStatsReport();
>+  rtc::scoped_refptr<MockRtpSenderInternal> sender =
>+      CreateMockSender(track, 0, 49, {});
>+  pc_->AddSender(sender);
>+
>+  rtc::scoped_refptr<const RTCStatsReport> report = stats_->GetStatsReport();
>+
>   std::vector<const RTCMediaStreamTrackStats*> track_stats =
>       report->GetStatsOfType<RTCMediaStreamTrackStats>();
>-  EXPECT_EQ(0, track_stats.size());
>+  EXPECT_EQ(1U, track_stats.size());
>+
>+  std::vector<const RTCRTPStreamStats*> rtp_stream_stats =
>+      report->GetStatsOfType<RTCRTPStreamStats>();
>+  EXPECT_EQ(0U, rtp_stream_stats.size());
> }
> 
> TEST_F(RTCStatsCollectorTest, DoNotCrashOnSsrcChange) {
>   rtc::scoped_refptr<MediaStreamTrackInterface> track =
>       CreateFakeTrack(cricket::MEDIA_TYPE_AUDIO, "audioTrack",
>                       MediaStreamTrackInterface::kLive);
>-  rtc::scoped_refptr<MockRtpSender> sender =
>+  rtc::scoped_refptr<MockRtpSenderInternal> sender =
>       CreateMockSender(track, 4711, 49, {});
>-  EXPECT_CALL(test_->pc(), GetSenders())
>-      .WillRepeatedly(
>-          Return(std::vector<rtc::scoped_refptr<RtpSenderInterface>>(
>-              {rtc::scoped_refptr<RtpSenderInterface>(sender.get())})));
>+  pc_->AddSender(sender);
>+
>   // We do not generate any matching voice_sender_info stats.
>-  rtc::scoped_refptr<const RTCStatsReport> report = GetStatsReport();
>+  rtc::scoped_refptr<const RTCStatsReport> report = stats_->GetStatsReport();
>+
>   std::vector<const RTCMediaStreamTrackStats*> track_stats =
>       report->GetStatsOfType<RTCMediaStreamTrackStats>();
>-  EXPECT_EQ(1, track_stats.size());
>+  EXPECT_EQ(1U, track_stats.size());
> }
> 
>-class RTCStatsCollectorTestWithFakeCollector : public testing::Test {
>+// Used for test below, to test calling GetStatsReport during a callback.
>+class RecursiveCallback : public RTCStatsCollectorCallback {
>  public:
>-  RTCStatsCollectorTestWithFakeCollector()
>-    : test_(new rtc::RefCountedObject<RTCStatsCollectorTestHelper>()),
>-      collector_(FakeRTCStatsCollector::Create(
>-          &test_->pc(), 50 * rtc::kNumMicrosecsPerMillisec)) {
>+  explicit RecursiveCallback(RTCStatsCollectorWrapper* stats) : stats_(stats) {}
>+
>+  void OnStatsDelivered(
>+      const rtc::scoped_refptr<const RTCStatsReport>& report) override {
>+    stats_->GetStatsReport();
>+    called_ = true;
>+  }
>+
>+  bool called() const { return called_; }
>+
>+ private:
>+  RTCStatsCollectorWrapper* stats_;
>+  bool called_ = false;
>+};
>+
>+// Test that nothing bad happens if a callback causes GetStatsReport to be
>+// called again recursively. Regression test for crbug.com/webrtc/8973.
>+TEST_F(RTCStatsCollectorTest, DoNotCrashWhenGetStatsCalledDuringCallback) {
>+  rtc::scoped_refptr<RecursiveCallback> callback1(
>+      new rtc::RefCountedObject<RecursiveCallback>(stats_.get()));
>+  rtc::scoped_refptr<RecursiveCallback> callback2(
>+      new rtc::RefCountedObject<RecursiveCallback>(stats_.get()));
>+  stats_->stats_collector()->GetStatsReport(callback1);
>+  stats_->stats_collector()->GetStatsReport(callback2);
>+  EXPECT_TRUE_WAIT(callback1->called(), kGetStatsReportTimeoutMs);
>+  EXPECT_TRUE_WAIT(callback2->called(), kGetStatsReportTimeoutMs);
>+}
>+
>+class RTCTestStats : public RTCStats {
>+ public:
>+  WEBRTC_RTCSTATS_DECL();
>+
>+  RTCTestStats(const std::string& id, int64_t timestamp_us)
>+      : RTCStats(id, timestamp_us), dummy_stat("dummyStat") {}
>+
>+  RTCStatsMember<int32_t> dummy_stat;
>+};
>+
>+WEBRTC_RTCSTATS_IMPL(RTCTestStats, RTCStats, "test-stats", &dummy_stat);
>+
>+// Overrides the stats collection to verify thread usage and that the resulting
>+// partial reports are merged.
>+class FakeRTCStatsCollector : public RTCStatsCollector,
>+                              public RTCStatsCollectorCallback {
>+ public:
>+  static rtc::scoped_refptr<FakeRTCStatsCollector> Create(
>+      PeerConnectionInternal* pc,
>+      int64_t cache_lifetime_us) {
>+    return rtc::scoped_refptr<FakeRTCStatsCollector>(
>+        new rtc::RefCountedObject<FakeRTCStatsCollector>(pc,
>+                                                         cache_lifetime_us));
>+  }
>+
>+  // RTCStatsCollectorCallback implementation.
>+  void OnStatsDelivered(
>+      const rtc::scoped_refptr<const RTCStatsReport>& report) override {
>+    EXPECT_TRUE(signaling_thread_->IsCurrent());
>+    rtc::CritScope cs(&lock_);
>+    delivered_report_ = report;
>+  }
>+
>+  void VerifyThreadUsageAndResultsMerging() {
>+    GetStatsReport(rtc::scoped_refptr<RTCStatsCollectorCallback>(this));
>+    EXPECT_TRUE_WAIT(HasVerifiedResults(), kGetStatsReportTimeoutMs);
>+  }
>+
>+  bool HasVerifiedResults() {
>+    EXPECT_TRUE(signaling_thread_->IsCurrent());
>+    rtc::CritScope cs(&lock_);
>+    if (!delivered_report_)
>+      return false;
>+    EXPECT_EQ(produced_on_signaling_thread_, 1);
>+    EXPECT_EQ(produced_on_network_thread_, 1);
>+
>+    EXPECT_TRUE(delivered_report_->Get("SignalingThreadStats"));
>+    EXPECT_TRUE(delivered_report_->Get("NetworkThreadStats"));
>+
>+    produced_on_signaling_thread_ = 0;
>+    produced_on_network_thread_ = 0;
>+    delivered_report_ = nullptr;
>+    return true;
>   }
> 
>  protected:
>-  rtc::scoped_refptr<RTCStatsCollectorTestHelper> test_;
>-  rtc::scoped_refptr<FakeRTCStatsCollector> collector_;
>+  FakeRTCStatsCollector(PeerConnectionInternal* pc, int64_t cache_lifetime)
>+      : RTCStatsCollector(pc, cache_lifetime),
>+        signaling_thread_(pc->signaling_thread()),
>+        worker_thread_(pc->worker_thread()),
>+        network_thread_(pc->network_thread()) {}
>+
>+  void ProducePartialResultsOnSignalingThread(int64_t timestamp_us) override {
>+    EXPECT_TRUE(signaling_thread_->IsCurrent());
>+    {
>+      rtc::CritScope cs(&lock_);
>+      EXPECT_FALSE(delivered_report_);
>+      ++produced_on_signaling_thread_;
>+    }
>+
>+    rtc::scoped_refptr<RTCStatsReport> signaling_report =
>+        RTCStatsReport::Create(0);
>+    signaling_report->AddStats(std::unique_ptr<const RTCStats>(
>+        new RTCTestStats("SignalingThreadStats", timestamp_us)));
>+    AddPartialResults(signaling_report);
>+  }
>+  void ProducePartialResultsOnNetworkThread(int64_t timestamp_us) override {
>+    EXPECT_TRUE(network_thread_->IsCurrent());
>+    {
>+      rtc::CritScope cs(&lock_);
>+      EXPECT_FALSE(delivered_report_);
>+      ++produced_on_network_thread_;
>+    }
>+
>+    rtc::scoped_refptr<RTCStatsReport> network_report =
>+        RTCStatsReport::Create(0);
>+    network_report->AddStats(std::unique_ptr<const RTCStats>(
>+        new RTCTestStats("NetworkThreadStats", timestamp_us)));
>+    AddPartialResults(network_report);
>+  }
>+
>+ private:
>+  rtc::Thread* const signaling_thread_;
>+  rtc::Thread* const worker_thread_;
>+  rtc::Thread* const network_thread_;
>+
>+  rtc::CriticalSection lock_;
>+  rtc::scoped_refptr<const RTCStatsReport> delivered_report_;
>+  int produced_on_signaling_thread_ = 0;
>+  int produced_on_network_thread_ = 0;
> };
> 
>-TEST_F(RTCStatsCollectorTestWithFakeCollector, ThreadUsageAndResultsMerging) {
>-  collector_->VerifyThreadUsageAndResultsMerging();
>+TEST(RTCStatsCollectorTestWithFakeCollector, ThreadUsageAndResultsMerging) {
>+  rtc::scoped_refptr<FakePeerConnectionForStats> pc(
>+      new rtc::RefCountedObject<FakePeerConnectionForStats>());
>+  rtc::scoped_refptr<FakeRTCStatsCollector> stats_collector(
>+      FakeRTCStatsCollector::Create(pc, 50 * rtc::kNumMicrosecsPerMillisec));
>+  stats_collector->VerifyThreadUsageAndResultsMerging();
> }
> 
> }  // namespace
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcstatstraversal.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcstatstraversal.cc
>new file mode 100644
>index 00000000000..62b8c52a107
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcstatstraversal.cc
>@@ -0,0 +1,120 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "pc/rtcstatstraversal.h"
>+
>+#include <memory>
>+#include <string>
>+#include <utility>
>+#include <vector>
>+
>+#include "api/stats/rtcstats_objects.h"
>+#include "rtc_base/checks.h"
>+
>+namespace webrtc {
>+
>+namespace {
>+
>+void TraverseAndTakeVisitedStats(RTCStatsReport* report,
>+                                 RTCStatsReport* visited_report,
>+                                 const std::string& current_id) {
>+  // Mark current stats object as visited by moving it |report| to
>+  // |visited_report|.
>+  std::unique_ptr<const RTCStats> current = report->Take(current_id);
>+  if (!current) {
>+    // This node has already been visited (or it is an invalid id).
>+    return;
>+  }
>+  std::vector<const std::string*> neighbor_ids =
>+      GetStatsReferencedIds(*current);
>+  visited_report->AddStats(std::move(current));
>+
>+  // Recursively traverse all neighbors.
>+  for (const auto* neighbor_id : neighbor_ids) {
>+    TraverseAndTakeVisitedStats(report, visited_report, *neighbor_id);
>+  }
>+}
>+
>+void AddIdIfDefined(const RTCStatsMember<std::string>& id,
>+                    std::vector<const std::string*>* neighbor_ids) {
>+  if (id.is_defined())
>+    neighbor_ids->push_back(&(*id));
>+}
>+
>+void AddIdsIfDefined(const RTCStatsMember<std::vector<std::string>>& ids,
>+                     std::vector<const std::string*>* neighbor_ids) {
>+  if (ids.is_defined()) {
>+    for (const std::string& id : *ids)
>+      neighbor_ids->push_back(&id);
>+  }
>+}
>+
>+}  // namespace
>+
>+rtc::scoped_refptr<RTCStatsReport> TakeReferencedStats(
>+    rtc::scoped_refptr<RTCStatsReport> report,
>+    const std::vector<std::string>& ids) {
>+  rtc::scoped_refptr<RTCStatsReport> result =
>+      RTCStatsReport::Create(report->timestamp_us());
>+  for (const auto& id : ids) {
>+    TraverseAndTakeVisitedStats(report.get(), result.get(), id);
>+  }
>+  return result;
>+}
>+
>+std::vector<const std::string*> GetStatsReferencedIds(const RTCStats& stats) {
>+  std::vector<const std::string*> neighbor_ids;
>+  const char* type = stats.type();
>+  if (type == RTCCertificateStats::kType) {
>+    const auto& certificate = static_cast<const RTCCertificateStats&>(stats);
>+    AddIdIfDefined(certificate.issuer_certificate_id, &neighbor_ids);
>+  } else if (type == RTCCodecStats::kType) {
>+    // RTCCodecStats does not have any neighbor references.
>+  } else if (type == RTCDataChannelStats::kType) {
>+    // RTCDataChannelStats does not have any neighbor references.
>+  } else if (type == RTCIceCandidatePairStats::kType) {
>+    const auto& candidate_pair =
>+        static_cast<const RTCIceCandidatePairStats&>(stats);
>+    AddIdIfDefined(candidate_pair.transport_id, &neighbor_ids);
>+    AddIdIfDefined(candidate_pair.local_candidate_id, &neighbor_ids);
>+    AddIdIfDefined(candidate_pair.remote_candidate_id, &neighbor_ids);
>+  } else if (type == RTCLocalIceCandidateStats::kType ||
>+             type == RTCRemoteIceCandidateStats::kType) {
>+    const auto& local_or_remote_candidate =
>+        static_cast<const RTCIceCandidateStats&>(stats);
>+    AddIdIfDefined(local_or_remote_candidate.transport_id, &neighbor_ids);
>+  } else if (type == RTCMediaStreamStats::kType) {
>+    const auto& stream = static_cast<const RTCMediaStreamStats&>(stats);
>+    AddIdsIfDefined(stream.track_ids, &neighbor_ids);
>+  } else if (type == RTCMediaStreamTrackStats::kType) {
>+    // RTCMediaStreamTrackStats does not have any neighbor references.
>+  } else if (type == RTCPeerConnectionStats::kType) {
>+    // RTCPeerConnectionStats does not have any neighbor references.
>+  } else if (type == RTCInboundRTPStreamStats::kType ||
>+             type == RTCOutboundRTPStreamStats::kType) {
>+    const auto& rtp = static_cast<const RTCRTPStreamStats&>(stats);
>+    AddIdIfDefined(rtp.associate_stats_id, &neighbor_ids);
>+    AddIdIfDefined(rtp.track_id, &neighbor_ids);
>+    AddIdIfDefined(rtp.transport_id, &neighbor_ids);
>+    AddIdIfDefined(rtp.codec_id, &neighbor_ids);
>+  } else if (type == RTCTransportStats::kType) {
>+    // RTCTransportStats does not have any neighbor references.
>+    const auto& transport = static_cast<const RTCTransportStats&>(stats);
>+    AddIdIfDefined(transport.rtcp_transport_stats_id, &neighbor_ids);
>+    AddIdIfDefined(transport.selected_candidate_pair_id, &neighbor_ids);
>+    AddIdIfDefined(transport.local_certificate_id, &neighbor_ids);
>+    AddIdIfDefined(transport.remote_certificate_id, &neighbor_ids);
>+  } else {
>+    RTC_NOTREACHED() << "Unrecognized type: " << type;
>+  }
>+  return neighbor_ids;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcstatstraversal.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcstatstraversal.h
>new file mode 100644
>index 00000000000..1c50ae46ac6
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcstatstraversal.h
>@@ -0,0 +1,43 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef PC_RTCSTATSTRAVERSAL_H_
>+#define PC_RTCSTATSTRAVERSAL_H_
>+
>+#include <string>
>+#include <vector>
>+
>+#include "api/stats/rtcstatsreport.h"
>+#include "rtc_base/scoped_ref_ptr.h"
>+
>+namespace webrtc {
>+
>+// Traverses the stats graph, taking all stats objects that are directly or
>+// indirectly accessible from and including the stats objects identified by
>+// |ids|, returning them as a new stats report.
>+// This is meant to be used to implement the stats selection algorithm.
>+// https://w3c.github.io/webrtc-pc/#dfn-stats-selection-algorithm
>+rtc::scoped_refptr<RTCStatsReport> TakeReferencedStats(
>+    rtc::scoped_refptr<RTCStatsReport> report,
>+    const std::vector<std::string>& ids);
>+
>+// Gets pointers to the string values of any members in |stats| that are used as
>+// references for looking up other stats objects in the same report by ID. The
>+// pointers are valid for the lifetime of |stats| assumings its members are not
>+// modified.
>+//
>+// For example, RTCCodecStats contains "transportId"
>+// (RTCCodecStats::transport_id) referencing an RTCTransportStats.
>+// https://w3c.github.io/webrtc-stats/#dom-rtccodecstats-transportid
>+std::vector<const std::string*> GetStatsReferencedIds(const RTCStats& stats);
>+
>+}  // namespace webrtc
>+
>+#endif  // PC_RTCSTATSTRAVERSAL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcstatstraversal_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcstatstraversal_unittest.cc
>new file mode 100644
>index 00000000000..05529640231
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtcstatstraversal_unittest.cc
>@@ -0,0 +1,208 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "pc/rtcstatstraversal.h"
>+
>+#include <memory>
>+#include <string>
>+#include <vector>
>+
>+#include "api/stats/rtcstats_objects.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/gunit.h"
>+
>+// This file contains tests for TakeReferencedStats().
>+// GetStatsNeighborIds() is tested in rtcstats_integrationtest.cc.
>+
>+namespace webrtc {
>+
>+class RTCStatsTraversalTest : public testing::Test {
>+ public:
>+  RTCStatsTraversalTest() {
>+    transport_ = new RTCTransportStats("transport", 0);
>+    candidate_pair_ = new RTCIceCandidatePairStats("candidate-pair", 0);
>+    local_candidate_ = new RTCLocalIceCandidateStats("local-candidate", 0);
>+    remote_candidate_ = new RTCRemoteIceCandidateStats("remote-candidate", 0);
>+    initial_report_ = RTCStatsReport::Create(0);
>+    initial_report_->AddStats(std::unique_ptr<const RTCStats>(transport_));
>+    initial_report_->AddStats(std::unique_ptr<const RTCStats>(candidate_pair_));
>+    initial_report_->AddStats(
>+        std::unique_ptr<const RTCStats>(local_candidate_));
>+    initial_report_->AddStats(
>+        std::unique_ptr<const RTCStats>(remote_candidate_));
>+    result_ = RTCStatsReport::Create(0);
>+  }
>+
>+  void TakeReferencedStats(std::vector<const RTCStats*> start_nodes) {
>+    std::vector<std::string> start_ids;
>+    for (const RTCStats* start_node : start_nodes) {
>+      start_ids.push_back(start_node->id());
>+    }
>+    result_ = webrtc::TakeReferencedStats(initial_report_, start_ids);
>+  }
>+
>+  void EXPECT_VISITED(const RTCStats* stats) {
>+    EXPECT_FALSE(initial_report_->Get(stats->id()))
>+        << '"' << stats->id()
>+        << "\" should be visited but it was not removed from initial report.";
>+    EXPECT_TRUE(result_->Get(stats->id()))
>+        << '"' << stats->id()
>+        << "\" should be visited but it was not added to the resulting report.";
>+  }
>+
>+  void EXPECT_UNVISITED(const RTCStats* stats) {
>+    EXPECT_TRUE(initial_report_->Get(stats->id()))
>+        << '"' << stats->id()
>+        << "\" should not be visited but it was removed from initial report.";
>+    EXPECT_FALSE(result_->Get(stats->id()))
>+        << '"' << stats->id()
>+        << "\" should not be visited but it was added to the resulting report.";
>+  }
>+
>+ protected:
>+  rtc::scoped_refptr<RTCStatsReport> initial_report_;
>+  rtc::scoped_refptr<RTCStatsReport> result_;
>+  // Raw pointers to stats owned by the reports.
>+  RTCTransportStats* transport_;
>+  RTCIceCandidatePairStats* candidate_pair_;
>+  RTCIceCandidateStats* local_candidate_;
>+  RTCIceCandidateStats* remote_candidate_;
>+};
>+
>+TEST_F(RTCStatsTraversalTest, NoReachableConnections) {
>+  // Everything references transport but transport doesn't reference anything.
>+  //
>+  //          candidate-pair
>+  //            |    |  |
>+  //            v    |  v
>+  // local-candidate | remote-candidate
>+  //              |  |  |
>+  //              v  v  v
>+  //          start:transport
>+  candidate_pair_->transport_id = "transport";
>+  candidate_pair_->local_candidate_id = "local-candidate";
>+  candidate_pair_->remote_candidate_id = "remote-candidate";
>+  local_candidate_->transport_id = "transport";
>+  remote_candidate_->transport_id = "transport";
>+  TakeReferencedStats({transport_});
>+  EXPECT_VISITED(transport_);
>+  EXPECT_UNVISITED(candidate_pair_);
>+  EXPECT_UNVISITED(local_candidate_);
>+  EXPECT_UNVISITED(remote_candidate_);
>+}
>+
>+TEST_F(RTCStatsTraversalTest, SelfReference) {
>+  transport_->rtcp_transport_stats_id = "transport";
>+  TakeReferencedStats({transport_});
>+  EXPECT_VISITED(transport_);
>+  EXPECT_UNVISITED(candidate_pair_);
>+  EXPECT_UNVISITED(local_candidate_);
>+  EXPECT_UNVISITED(remote_candidate_);
>+}
>+
>+TEST_F(RTCStatsTraversalTest, BogusReference) {
>+  transport_->rtcp_transport_stats_id = "bogus-reference";
>+  TakeReferencedStats({transport_});
>+  EXPECT_VISITED(transport_);
>+  EXPECT_UNVISITED(candidate_pair_);
>+  EXPECT_UNVISITED(local_candidate_);
>+  EXPECT_UNVISITED(remote_candidate_);
>+}
>+
>+TEST_F(RTCStatsTraversalTest, Tree) {
>+  //     start:candidate-pair
>+  //        |            |
>+  //        v            v
>+  // local-candidate   remote-candidate
>+  //       |
>+  //       v
>+  //   transport
>+  candidate_pair_->local_candidate_id = "local-candidate";
>+  candidate_pair_->remote_candidate_id = "remote-candidate";
>+  local_candidate_->transport_id = "transport";
>+  TakeReferencedStats({candidate_pair_});
>+  EXPECT_VISITED(transport_);
>+  EXPECT_VISITED(candidate_pair_);
>+  EXPECT_VISITED(local_candidate_);
>+  EXPECT_VISITED(remote_candidate_);
>+}
>+
>+TEST_F(RTCStatsTraversalTest, MultiplePathsToSameNode) {
>+  //     start:candidate-pair
>+  //        |            |
>+  //        v            v
>+  // local-candidate   remote-candidate
>+  //              |     |
>+  //              v     v
>+  //             transport
>+  candidate_pair_->local_candidate_id = "local-candidate";
>+  candidate_pair_->remote_candidate_id = "remote-candidate";
>+  local_candidate_->transport_id = "transport";
>+  remote_candidate_->transport_id = "transport";
>+  TakeReferencedStats({candidate_pair_});
>+  EXPECT_VISITED(transport_);
>+  EXPECT_VISITED(candidate_pair_);
>+  EXPECT_VISITED(local_candidate_);
>+  EXPECT_VISITED(remote_candidate_);
>+}
>+
>+TEST_F(RTCStatsTraversalTest, CyclicGraph) {
>+  //               candidate-pair
>+  //                  |     ^
>+  //                  v     |
>+  // start:local-candidate  |    remote-candidate
>+  //                    |   |
>+  //                    v   |
>+  //                  transport
>+  local_candidate_->transport_id = "transport";
>+  transport_->selected_candidate_pair_id = "candidate-pair";
>+  candidate_pair_->local_candidate_id = "local-candidate";
>+  TakeReferencedStats({local_candidate_});
>+  EXPECT_VISITED(transport_);
>+  EXPECT_VISITED(candidate_pair_);
>+  EXPECT_VISITED(local_candidate_);
>+  EXPECT_UNVISITED(remote_candidate_);
>+}
>+
>+TEST_F(RTCStatsTraversalTest, MultipleStarts) {
>+  //           start:candidate-pair
>+  //                        |
>+  //                        v
>+  // local-candidate    remote-candidate
>+  //             |
>+  //             v
>+  //           start:transport
>+  candidate_pair_->remote_candidate_id = "remote-candidate";
>+  local_candidate_->transport_id = "transport";
>+  TakeReferencedStats({candidate_pair_, transport_});
>+  EXPECT_VISITED(transport_);
>+  EXPECT_VISITED(candidate_pair_);
>+  EXPECT_UNVISITED(local_candidate_);
>+  EXPECT_VISITED(remote_candidate_);
>+}
>+
>+TEST_F(RTCStatsTraversalTest, MultipleStartsLeadingToSameNode) {
>+  //                candidate-pair
>+  //
>+  //
>+  // start:local-candidate   start:remote-candidate
>+  //                    |     |
>+  //                    v     v
>+  //                   transport
>+  local_candidate_->transport_id = "transport";
>+  remote_candidate_->transport_id = "transport";
>+  TakeReferencedStats({local_candidate_, remote_candidate_});
>+  EXPECT_VISITED(transport_);
>+  EXPECT_UNVISITED(candidate_pair_);
>+  EXPECT_VISITED(local_candidate_);
>+  EXPECT_VISITED(remote_candidate_);
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpmediautils.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpmediautils.cc
>index f88861c3204..337aa2d090e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpmediautils.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpmediautils.cc
>@@ -50,6 +50,20 @@ RtpTransceiverDirection RtpTransceiverDirectionReversed(
>   return direction;
> }
> 
>+RtpTransceiverDirection RtpTransceiverDirectionWithSendSet(
>+    RtpTransceiverDirection direction,
>+    bool send) {
>+  return RtpTransceiverDirectionFromSendRecv(
>+      send, RtpTransceiverDirectionHasRecv(direction));
>+}
>+
>+RtpTransceiverDirection RtpTransceiverDirectionWithRecvSet(
>+    RtpTransceiverDirection direction,
>+    bool recv) {
>+  return RtpTransceiverDirectionFromSendRecv(
>+      RtpTransceiverDirectionHasSend(direction), recv);
>+}
>+
> const char* RtpTransceiverDirectionToString(RtpTransceiverDirection direction) {
>   switch (direction) {
>     case RtpTransceiverDirection::kSendRecv:
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpmediautils.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpmediautils.h
>index 6f547bd04ac..5f02d899074 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpmediautils.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpmediautils.h
>@@ -31,9 +31,27 @@ bool RtpTransceiverDirectionHasRecv(RtpTransceiverDirection direction);
> RtpTransceiverDirection RtpTransceiverDirectionReversed(
>     RtpTransceiverDirection direction);
> 
>+// Returns the RtpTransceiverDirection with its send component set to |send|.
>+RtpTransceiverDirection RtpTransceiverDirectionWithSendSet(
>+    RtpTransceiverDirection direction,
>+    bool send = true);
>+
>+// Returns the RtpTransceiverDirection with its recv component set to |recv|.
>+RtpTransceiverDirection RtpTransceiverDirectionWithRecvSet(
>+    RtpTransceiverDirection direction,
>+    bool recv = true);
>+
> // Returns an unspecified string representation of the given direction.
> const char* RtpTransceiverDirectionToString(RtpTransceiverDirection direction);
> 
>+#ifdef UNIT_TEST
>+inline std::ostream& operator<<(  // no-presubmit-check TODO(webrtc:8982)
>+    std::ostream& os,             // no-presubmit-check TODO(webrtc:8982)
>+    RtpTransceiverDirection direction) {
>+  return os << RtpTransceiverDirectionToString(direction);
>+}
>+#endif  // UNIT_TEST
>+
> }  // namespace webrtc
> 
> #endif  // PC_RTPMEDIAUTILS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpmediautils_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpmediautils_unittest.cc
>index d72f35be078..af7e8f671b2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpmediautils_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpmediautils_unittest.cc
>@@ -8,16 +8,24 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>+#include <tuple>
>+
> #include "pc/rtpmediautils.h"
> #include "test/gtest.h"
> 
> namespace webrtc {
> 
>+using ::testing::Bool;
>+using ::testing::Combine;
> using ::testing::Values;
>+using ::testing::ValuesIn;
>+
>+RtpTransceiverDirection kAllDirections[] = {
>+    RtpTransceiverDirection::kSendRecv, RtpTransceiverDirection::kSendOnly,
>+    RtpTransceiverDirection::kRecvOnly, RtpTransceiverDirection::kInactive};
> 
> class EnumerateAllDirectionsTest
>-    : public ::testing::Test,
>-      public ::testing::WithParamInterface<RtpTransceiverDirection> {};
>+    : public ::testing::TestWithParam<RtpTransceiverDirection> {};
> 
> // Test that converting the direction to send/recv and back again results in the
> // same direction.
>@@ -51,9 +59,38 @@ TEST_P(EnumerateAllDirectionsTest, TestReversedIdentity) {
> 
> INSTANTIATE_TEST_CASE_P(RtpTransceiverDirectionTest,
>                         EnumerateAllDirectionsTest,
>-                        Values(RtpTransceiverDirection::kSendRecv,
>-                               RtpTransceiverDirection::kSendOnly,
>-                               RtpTransceiverDirection::kRecvOnly,
>-                               RtpTransceiverDirection::kInactive));
>+                        ValuesIn(kAllDirections));
>+
>+class EnumerateAllDirectionsAndBool
>+    : public ::testing::TestWithParam<
>+          std::tuple<RtpTransceiverDirection, bool>> {};
>+
>+TEST_P(EnumerateAllDirectionsAndBool, TestWithSendSet) {
>+  RtpTransceiverDirection direction = std::get<0>(GetParam());
>+  bool send = std::get<1>(GetParam());
>+
>+  RtpTransceiverDirection result =
>+      RtpTransceiverDirectionWithSendSet(direction, send);
>+
>+  EXPECT_EQ(send, RtpTransceiverDirectionHasSend(result));
>+  EXPECT_EQ(RtpTransceiverDirectionHasRecv(direction),
>+            RtpTransceiverDirectionHasRecv(result));
>+}
>+
>+TEST_P(EnumerateAllDirectionsAndBool, TestWithRecvSet) {
>+  RtpTransceiverDirection direction = std::get<0>(GetParam());
>+  bool recv = std::get<1>(GetParam());
>+
>+  RtpTransceiverDirection result =
>+      RtpTransceiverDirectionWithRecvSet(direction, recv);
>+
>+  EXPECT_EQ(RtpTransceiverDirectionHasSend(direction),
>+            RtpTransceiverDirectionHasSend(result));
>+  EXPECT_EQ(recv, RtpTransceiverDirectionHasRecv(result));
>+}
>+
>+INSTANTIATE_TEST_CASE_P(RtpTransceiverDirectionTest,
>+                        EnumerateAllDirectionsAndBool,
>+                        Combine(ValuesIn(kAllDirections), Bool()));
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtpparametersconversion.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpparametersconversion.cc
>similarity index 92%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtpparametersconversion.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpparametersconversion.cc
>index 0a657eca8f3..e519b9b06de 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtpparametersconversion.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpparametersconversion.cc
>@@ -8,10 +8,9 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "ortc/rtpparametersconversion.h"
>+#include "pc/rtpparametersconversion.h"
> 
> #include <set>
>-#include <sstream>
> #include <utility>
> 
> #include "media/base/rtputils.h"
>@@ -62,7 +61,7 @@ RTCErrorOr<cricket::FeedbackParam> ToCricketFeedbackParam(
>       return cricket::FeedbackParam(cricket::kRtcpFbParamTransportCc);
>   }
>   // Not reached; avoids compile warning.
>-  FATAL();
>+  RTC_FATAL();
> }
> 
> template <typename C>
>@@ -136,9 +135,10 @@ RTCErrorOr<C> ToCricketCodec(const RtpCodecParameters& codec) {
>   }
>   cricket_codec.name = codec.name;
>   if (!cricket::IsValidRtpPayloadType(codec.payload_type)) {
>-    std::ostringstream oss;
>-    oss << "Invalid payload type: " << codec.payload_type;
>-    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_RANGE, oss.str());
>+    char buf[40];
>+    rtc::SimpleStringBuilder sb(buf);
>+    sb << "Invalid payload type: " << codec.payload_type;
>+    LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_RANGE, sb.str());
>   }
>   cricket_codec.id = codec.payload_type;
>   for (const RtcpFeedback& feedback : codec.rtcp_feedback) {
>@@ -168,9 +168,10 @@ RTCErrorOr<std::vector<C>> ToCricketCodecs(
>       return result.MoveError();
>     }
>     if (!seen_payload_types.insert(codec.payload_type).second) {
>-      std::ostringstream oss;
>-      oss << "Duplicate payload type: " << codec.payload_type;
>-      LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, oss.str());
>+      char buf[40];
>+      rtc::SimpleStringBuilder sb(buf);
>+      sb << "Duplicate payload type: " << codec.payload_type;
>+      LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, sb.str());
>     }
>     cricket_codecs.push_back(result.MoveValue());
>   }
>@@ -186,17 +187,20 @@ template RTCErrorOr<std::vector<cricket::VideoCodec>> ToCricketCodecs<
> RTCErrorOr<cricket::RtpHeaderExtensions> ToCricketRtpHeaderExtensions(
>     const std::vector<RtpHeaderExtensionParameters>& extensions) {
>   cricket::RtpHeaderExtensions cricket_extensions;
>-  std::ostringstream err_writer;
>   std::set<int> seen_header_extension_ids;
>   for (const RtpHeaderExtensionParameters& extension : extensions) {
>     if (extension.id < RtpHeaderExtensionParameters::kMinId ||
>         extension.id > RtpHeaderExtensionParameters::kMaxId) {
>-      err_writer << "Invalid header extension id: " << extension.id;
>-      LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_RANGE, err_writer.str());
>+      char buf[50];
>+      rtc::SimpleStringBuilder sb(buf);
>+      sb << "Invalid header extension id: " << extension.id;
>+      LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_RANGE, sb.str());
>     }
>     if (!seen_header_extension_ids.insert(extension.id).second) {
>-      err_writer << "Duplicate header extension id: " << extension.id;
>-      LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, err_writer.str());
>+      char buf[50];
>+      rtc::SimpleStringBuilder sb(buf);
>+      sb << "Duplicate header extension id: " << extension.id;
>+      LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, sb.str());
>     }
>     cricket_extensions.push_back(extension);
>   }
>@@ -230,7 +234,7 @@ RTCErrorOr<cricket::StreamParamsVec> ToCricketStreamParamsVec(
>   return std::move(cricket_streams);
> }
> 
>-rtc::Optional<RtcpFeedback> ToRtcpFeedback(
>+absl::optional<RtcpFeedback> ToRtcpFeedback(
>     const cricket::FeedbackParam& cricket_feedback) {
>   if (cricket_feedback.id() == cricket::kRtcpFbParamCcm) {
>     if (cricket_feedback.param() == cricket::kRtcpFbCcmParamFir) {
>@@ -238,7 +242,7 @@ rtc::Optional<RtcpFeedback> ToRtcpFeedback(
>     } else {
>       RTC_LOG(LS_WARNING) << "Unsupported parameter for CCM RTCP feedback: "
>                           << cricket_feedback.param();
>-      return rtc::nullopt;
>+      return absl::nullopt;
>     }
>   } else if (cricket_feedback.id() == cricket::kRtcpFbParamNack) {
>     if (cricket_feedback.param().empty()) {
>@@ -249,13 +253,13 @@ rtc::Optional<RtcpFeedback> ToRtcpFeedback(
>     } else {
>       RTC_LOG(LS_WARNING) << "Unsupported parameter for NACK RTCP feedback: "
>                           << cricket_feedback.param();
>-      return rtc::nullopt;
>+      return absl::nullopt;
>     }
>   } else if (cricket_feedback.id() == cricket::kRtcpFbParamRemb) {
>     if (!cricket_feedback.param().empty()) {
>       RTC_LOG(LS_WARNING) << "Unsupported parameter for REMB RTCP feedback: "
>                           << cricket_feedback.param();
>-      return rtc::nullopt;
>+      return absl::nullopt;
>     } else {
>       return RtcpFeedback(RtcpFeedbackType::REMB);
>     }
>@@ -264,14 +268,14 @@ rtc::Optional<RtcpFeedback> ToRtcpFeedback(
>       RTC_LOG(LS_WARNING)
>           << "Unsupported parameter for transport-cc RTCP feedback: "
>           << cricket_feedback.param();
>-      return rtc::nullopt;
>+      return absl::nullopt;
>     } else {
>       return RtcpFeedback(RtcpFeedbackType::TRANSPORT_CC);
>     }
>   }
>   RTC_LOG(LS_WARNING) << "Unsupported RTCP feedback type: "
>                       << cricket_feedback.id();
>-  return rtc::nullopt;
>+  return absl::nullopt;
> }
> 
> std::vector<RtpEncodingParameters> ToRtpEncodings(
>@@ -328,9 +332,9 @@ RtpCodecCapability ToRtpCodecCapability(const C& cricket_codec) {
>   codec.preferred_payload_type.emplace(cricket_codec.id);
>   for (const cricket::FeedbackParam& cricket_feedback :
>        cricket_codec.feedback_params.params()) {
>-    rtc::Optional<RtcpFeedback> feedback = ToRtcpFeedback(cricket_feedback);
>+    absl::optional<RtcpFeedback> feedback = ToRtcpFeedback(cricket_feedback);
>     if (feedback) {
>-      codec.rtcp_feedback.push_back(feedback.MoveValue());
>+      codec.rtcp_feedback.push_back(feedback.value());
>     }
>   }
>   ToRtpCodecCapabilityTypeSpecific(cricket_codec, &codec);
>@@ -368,9 +372,9 @@ RtpCodecParameters ToRtpCodecParameters(const C& cricket_codec) {
>   codec_param.payload_type = cricket_codec.id;
>   for (const cricket::FeedbackParam& cricket_feedback :
>        cricket_codec.feedback_params.params()) {
>-    rtc::Optional<RtcpFeedback> feedback = ToRtcpFeedback(cricket_feedback);
>+    absl::optional<RtcpFeedback> feedback = ToRtcpFeedback(cricket_feedback);
>     if (feedback) {
>-      codec_param.rtcp_feedback.push_back(feedback.MoveValue());
>+      codec_param.rtcp_feedback.push_back(feedback.value());
>     }
>   }
>   ToRtpCodecParametersTypeSpecific(cricket_codec, &codec_param);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtpparametersconversion.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpparametersconversion.h
>similarity index 95%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtpparametersconversion.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpparametersconversion.h
>index c75e35440ff..d2d13aee916 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtpparametersconversion.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpparametersconversion.h
>@@ -8,13 +8,13 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#ifndef ORTC_RTPPARAMETERSCONVERSION_H_
>-#define ORTC_RTPPARAMETERSCONVERSION_H_
>+#ifndef PC_RTPPARAMETERSCONVERSION_H_
>+#define PC_RTPPARAMETERSCONVERSION_H_
> 
> #include <memory>
> #include <vector>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> #include "api/rtcerror.h"
> #include "api/rtpparameters.h"
> #include "media/base/codec.h"
>@@ -80,7 +80,7 @@ RTCErrorOr<cricket::StreamParamsVec> ToCricketStreamParamsVec(
> 
> // Returns empty value if |cricket_feedback| is a feedback type not
> // supported/recognized.
>-rtc::Optional<RtcpFeedback> ToRtcpFeedback(
>+absl::optional<RtcpFeedback> ToRtcpFeedback(
>     const cricket::FeedbackParam& cricket_feedback);
> 
> std::vector<RtpEncodingParameters> ToRtpEncodings(
>@@ -105,4 +105,4 @@ RtpParameters ToRtpParameters(
> 
> }  // namespace webrtc
> 
>-#endif  // ORTC_RTPPARAMETERSCONVERSION_H_
>+#endif  // PC_RTPPARAMETERSCONVERSION_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtpparametersconversion_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpparametersconversion_unittest.cc
>similarity index 99%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtpparametersconversion_unittest.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpparametersconversion_unittest.cc
>index d08ce9a19c2..649fe971916 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtpparametersconversion_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpparametersconversion_unittest.cc
>@@ -10,8 +10,7 @@
> 
> #include <algorithm>
> 
>-#include "ortc/rtpparametersconversion.h"
>-#include "ortc/testrtpparameters.h"
>+#include "pc/rtpparametersconversion.h"
> #include "rtc_base/gunit.h"
> 
> namespace webrtc {
>@@ -373,7 +372,7 @@ TEST(RtpParametersConversionTest, ToCricketStreamParamsVecMultipleEncodings) {
> }
> 
> TEST(RtpParametersConversionTest, ToRtcpFeedback) {
>-  rtc::Optional<RtcpFeedback> result = ToRtcpFeedback({"ccm", "fir"});
>+  absl::optional<RtcpFeedback> result = ToRtcpFeedback({"ccm", "fir"});
>   EXPECT_EQ(RtcpFeedback(RtcpFeedbackType::CCM, RtcpFeedbackMessageType::FIR),
>             *result);
>   result = ToRtcpFeedback(cricket::FeedbackParam("nack"));
>@@ -391,7 +390,7 @@ TEST(RtpParametersConversionTest, ToRtcpFeedback) {
> 
> TEST(RtpParametersConversionTest, ToRtcpFeedbackErrors) {
>   // CCM with missing or invalid message type.
>-  rtc::Optional<RtcpFeedback> result = ToRtcpFeedback({"ccm", "pli"});
>+  absl::optional<RtcpFeedback> result = ToRtcpFeedback({"ccm", "pli"});
>   EXPECT_FALSE(result);
>   result = ToRtcpFeedback(cricket::FeedbackParam("ccm"));
>   EXPECT_FALSE(result);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpreceiver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpreceiver.cc
>index e318c17e144..da2f81b162b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpreceiver.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpreceiver.cc
>@@ -13,9 +13,11 @@
> #include <utility>
> #include <vector>
> 
>+#include "api/mediastreamproxy.h"
> #include "api/mediastreamtrackproxy.h"
> #include "api/videosourceproxy.h"
> #include "pc/audiotrack.h"
>+#include "pc/mediastream.h"
> #include "pc/videotrack.h"
> #include "rtc_base/trace_event.h"
> 
>@@ -30,22 +32,35 @@ int GenerateUniqueId() {
>   return ++g_unique_id;
> }
> 
>+std::vector<rtc::scoped_refptr<MediaStreamInterface>> CreateStreamsFromIds(
>+    std::vector<std::string> stream_ids) {
>+  std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams(
>+      stream_ids.size());
>+  for (size_t i = 0; i < stream_ids.size(); ++i) {
>+    streams[i] = MediaStreamProxy::Create(
>+        rtc::Thread::Current(), MediaStream::Create(std::move(stream_ids[i])));
>+  }
>+  return streams;
>+}
>+
> }  // namespace
> 
>+AudioRtpReceiver::AudioRtpReceiver(rtc::Thread* worker_thread,
>+                                   std::string receiver_id,
>+                                   std::vector<std::string> stream_ids)
>+    : AudioRtpReceiver(worker_thread,
>+                       receiver_id,
>+                       CreateStreamsFromIds(std::move(stream_ids))) {}
>+
> AudioRtpReceiver::AudioRtpReceiver(
>     rtc::Thread* worker_thread,
>     const std::string& receiver_id,
>-    const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams,
>-    uint32_t ssrc,
>-    cricket::VoiceMediaChannel* media_channel)
>+    const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams)
>     : worker_thread_(worker_thread),
>       id_(receiver_id),
>-      ssrc_(ssrc),
>-      track_(AudioTrackProxy::Create(
>-          rtc::Thread::Current(),
>-          AudioTrack::Create(
>-              receiver_id,
>-              RemoteAudioSource::Create(worker_thread, media_channel, ssrc)))),
>+      source_(new rtc::RefCountedObject<RemoteAudioSource>(worker_thread)),
>+      track_(AudioTrackProxy::Create(rtc::Thread::Current(),
>+                                     AudioTrack::Create(receiver_id, source_))),
>       cached_track_enabled_(track_->enabled()),
>       attachment_id_(GenerateUniqueId()) {
>   RTC_DCHECK(worker_thread_);
>@@ -53,8 +68,6 @@ AudioRtpReceiver::AudioRtpReceiver(
>   track_->RegisterObserver(this);
>   track_->GetSource()->RegisterAudioObserver(this);
>   SetStreams(streams);
>-  SetMediaChannel(media_channel);
>-  Reconfigure();
> }
> 
> AudioRtpReceiver::~AudioRtpReceiver() {
>@@ -74,8 +87,9 @@ bool AudioRtpReceiver::SetOutputVolume(double volume) {
>   RTC_DCHECK_GE(volume, 0.0);
>   RTC_DCHECK_LE(volume, 10.0);
>   RTC_DCHECK(media_channel_);
>+  RTC_DCHECK(ssrc_);
>   return worker_thread_->Invoke<bool>(RTC_FROM_HERE, [&] {
>-    return media_channel_->SetOutputVolume(ssrc_, volume);
>+    return media_channel_->SetOutputVolume(*ssrc_, volume);
>   });
> }
> 
>@@ -83,7 +97,7 @@ void AudioRtpReceiver::OnSetVolume(double volume) {
>   RTC_DCHECK_GE(volume, 0);
>   RTC_DCHECK_LE(volume, 10);
>   cached_volume_ = volume;
>-  if (!media_channel_) {
>+  if (!media_channel_ || !ssrc_) {
>     RTC_LOG(LS_ERROR)
>         << "AudioRtpReceiver::OnSetVolume: No audio channel exists.";
>     return;
>@@ -98,22 +112,29 @@ void AudioRtpReceiver::OnSetVolume(double volume) {
>   }
> }
> 
>+std::vector<std::string> AudioRtpReceiver::stream_ids() const {
>+  std::vector<std::string> stream_ids(streams_.size());
>+  for (size_t i = 0; i < streams_.size(); ++i)
>+    stream_ids[i] = streams_[i]->id();
>+  return stream_ids;
>+}
>+
> RtpParameters AudioRtpReceiver::GetParameters() const {
>-  if (!media_channel_ || stopped_) {
>+  if (!media_channel_ || !ssrc_ || stopped_) {
>     return RtpParameters();
>   }
>   return worker_thread_->Invoke<RtpParameters>(RTC_FROM_HERE, [&] {
>-    return media_channel_->GetRtpReceiveParameters(ssrc_);
>+    return media_channel_->GetRtpReceiveParameters(*ssrc_);
>   });
> }
> 
> bool AudioRtpReceiver::SetParameters(const RtpParameters& parameters) {
>   TRACE_EVENT0("webrtc", "AudioRtpReceiver::SetParameters");
>-  if (!media_channel_ || stopped_) {
>+  if (!media_channel_ || !ssrc_ || stopped_) {
>     return false;
>   }
>   return worker_thread_->Invoke<bool>(RTC_FROM_HERE, [&] {
>-    return media_channel_->SetRtpReceiveParameters(ssrc_, parameters);
>+    return media_channel_->SetRtpReceiveParameters(*ssrc_, parameters);
>   });
> }
> 
>@@ -122,7 +143,7 @@ void AudioRtpReceiver::Stop() {
>   if (stopped_) {
>     return;
>   }
>-  if (media_channel_) {
>+  if (media_channel_ && ssrc_) {
>     // Allow that SetOutputVolume fail. This is the normal case when the
>     // underlying media channel has already been deleted.
>     SetOutputVolume(0.0);
>@@ -130,13 +151,34 @@ void AudioRtpReceiver::Stop() {
>   stopped_ = true;
> }
> 
>+void AudioRtpReceiver::SetupMediaChannel(uint32_t ssrc) {
>+  if (!media_channel_) {
>+    RTC_LOG(LS_ERROR)
>+        << "AudioRtpReceiver::SetupMediaChannel: No audio channel exists.";
>+    return;
>+  }
>+  if (ssrc_ == ssrc) {
>+    return;
>+  }
>+  if (ssrc_) {
>+    source_->Stop(media_channel_, *ssrc_);
>+  }
>+  ssrc_ = ssrc;
>+  source_->Start(media_channel_, *ssrc_);
>+  Reconfigure();
>+}
>+
>+void AudioRtpReceiver::set_stream_ids(std::vector<std::string> stream_ids) {
>+  SetStreams(CreateStreamsFromIds(std::move(stream_ids)));
>+}
>+
> void AudioRtpReceiver::SetStreams(
>     const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams) {
>   // Remove remote track from any streams that are going away.
>   for (auto existing_stream : streams_) {
>     bool removed = true;
>     for (auto stream : streams) {
>-      if (existing_stream->label() == stream->label()) {
>+      if (existing_stream->id() == stream->id()) {
>         RTC_DCHECK_EQ(existing_stream.get(), stream.get());
>         removed = false;
>         break;
>@@ -150,7 +192,7 @@ void AudioRtpReceiver::SetStreams(
>   for (auto stream : streams) {
>     bool added = true;
>     for (auto existing_stream : streams_) {
>-      if (stream->label() == existing_stream->label()) {
>+      if (stream->id() == existing_stream->id()) {
>         RTC_DCHECK_EQ(stream.get(), existing_stream.get());
>         added = false;
>         break;
>@@ -164,13 +206,16 @@ void AudioRtpReceiver::SetStreams(
> }
> 
> std::vector<RtpSource> AudioRtpReceiver::GetSources() const {
>+  if (!media_channel_ || !ssrc_ || stopped_) {
>+    return {};
>+  }
>   return worker_thread_->Invoke<std::vector<RtpSource>>(
>-      RTC_FROM_HERE, [&] { return media_channel_->GetSources(ssrc_); });
>+      RTC_FROM_HERE, [&] { return media_channel_->GetSources(*ssrc_); });
> }
> 
> void AudioRtpReceiver::Reconfigure() {
>   RTC_DCHECK(!stopped_);
>-  if (!media_channel_) {
>+  if (!media_channel_ || !ssrc_) {
>     RTC_LOG(LS_ERROR)
>         << "AudioRtpReceiver::Reconfigure: No audio channel exists.";
>     return;
>@@ -188,11 +233,6 @@ void AudioRtpReceiver::SetObserver(RtpReceiverObserverInterface* observer) {
>   }
> }
> 
>-void AudioRtpReceiver::SetMediaChannel(
>-    cricket::VoiceMediaChannel* media_channel) {
>-  media_channel_ = media_channel;
>-}
>-
> void AudioRtpReceiver::NotifyFirstPacketReceived() {
>   if (observer_) {
>     observer_->OnFirstPacketReceived(media_type());
>@@ -200,17 +240,20 @@ void AudioRtpReceiver::NotifyFirstPacketReceived() {
>   received_first_packet_ = true;
> }
> 
>+VideoRtpReceiver::VideoRtpReceiver(rtc::Thread* worker_thread,
>+                                   std::string receiver_id,
>+                                   std::vector<std::string> stream_ids)
>+    : VideoRtpReceiver(worker_thread,
>+                       receiver_id,
>+                       CreateStreamsFromIds(std::move(stream_ids))) {}
>+
> VideoRtpReceiver::VideoRtpReceiver(
>     rtc::Thread* worker_thread,
>     const std::string& receiver_id,
>-    const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams,
>-    uint32_t ssrc,
>-    cricket::VideoMediaChannel* media_channel)
>+    const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams)
>     : worker_thread_(worker_thread),
>       id_(receiver_id),
>-      ssrc_(ssrc),
>-      source_(new RefCountedObject<VideoTrackSource>(&broadcaster_,
>-                                                     true /* remote */)),
>+      source_(new RefCountedObject<VideoRtpTrackSource>()),
>       track_(VideoTrackProxy::Create(
>           rtc::Thread::Current(),
>           worker_thread,
>@@ -224,7 +267,6 @@ VideoRtpReceiver::VideoRtpReceiver(
>   RTC_DCHECK(worker_thread_);
>   SetStreams(streams);
>   source_->SetState(MediaSourceInterface::kLive);
>-  SetMediaChannel(media_channel);
> }
> 
> VideoRtpReceiver::~VideoRtpReceiver() {
>@@ -233,28 +275,36 @@ VideoRtpReceiver::~VideoRtpReceiver() {
>   Stop();
> }
> 
>+std::vector<std::string> VideoRtpReceiver::stream_ids() const {
>+  std::vector<std::string> stream_ids(streams_.size());
>+  for (size_t i = 0; i < streams_.size(); ++i)
>+    stream_ids[i] = streams_[i]->id();
>+  return stream_ids;
>+}
>+
> bool VideoRtpReceiver::SetSink(rtc::VideoSinkInterface<VideoFrame>* sink) {
>   RTC_DCHECK(media_channel_);
>+  RTC_DCHECK(ssrc_);
>   return worker_thread_->Invoke<bool>(
>-      RTC_FROM_HERE, [&] { return media_channel_->SetSink(ssrc_, sink); });
>+      RTC_FROM_HERE, [&] { return media_channel_->SetSink(*ssrc_, sink); });
> }
> 
> RtpParameters VideoRtpReceiver::GetParameters() const {
>-  if (!media_channel_ || stopped_) {
>+  if (!media_channel_ || !ssrc_ || stopped_) {
>     return RtpParameters();
>   }
>   return worker_thread_->Invoke<RtpParameters>(RTC_FROM_HERE, [&] {
>-    return media_channel_->GetRtpReceiveParameters(ssrc_);
>+    return media_channel_->GetRtpReceiveParameters(*ssrc_);
>   });
> }
> 
> bool VideoRtpReceiver::SetParameters(const RtpParameters& parameters) {
>   TRACE_EVENT0("webrtc", "VideoRtpReceiver::SetParameters");
>-  if (!media_channel_ || stopped_) {
>+  if (!media_channel_ || !ssrc_ || stopped_) {
>     return false;
>   }
>   return worker_thread_->Invoke<bool>(RTC_FROM_HERE, [&] {
>-    return media_channel_->SetRtpReceiveParameters(ssrc_, parameters);
>+    return media_channel_->SetRtpReceiveParameters(*ssrc_, parameters);
>   });
> }
> 
>@@ -264,8 +314,7 @@ void VideoRtpReceiver::Stop() {
>     return;
>   }
>   source_->SetState(MediaSourceInterface::kEnded);
>-  source_->OnSourceDestroyed();
>-  if (!media_channel_) {
>+  if (!media_channel_ || !ssrc_) {
>     RTC_LOG(LS_WARNING) << "VideoRtpReceiver::Stop: No video channel exists.";
>   } else {
>     // Allow that SetSink fail. This is the normal case when the underlying
>@@ -275,13 +324,32 @@ void VideoRtpReceiver::Stop() {
>   stopped_ = true;
> }
> 
>+void VideoRtpReceiver::SetupMediaChannel(uint32_t ssrc) {
>+  if (!media_channel_) {
>+    RTC_LOG(LS_ERROR)
>+        << "VideoRtpReceiver::SetupMediaChannel: No video channel exists.";
>+  }
>+  if (ssrc_ == ssrc) {
>+    return;
>+  }
>+  if (ssrc_) {
>+    SetSink(nullptr);
>+  }
>+  ssrc_ = ssrc;
>+  SetSink(source_->sink());
>+}
>+
>+void VideoRtpReceiver::set_stream_ids(std::vector<std::string> stream_ids) {
>+  SetStreams(CreateStreamsFromIds(std::move(stream_ids)));
>+}
>+
> void VideoRtpReceiver::SetStreams(
>     const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams) {
>   // Remove remote track from any streams that are going away.
>   for (auto existing_stream : streams_) {
>     bool removed = true;
>     for (auto stream : streams) {
>-      if (existing_stream->label() == stream->label()) {
>+      if (existing_stream->id() == stream->id()) {
>         RTC_DCHECK_EQ(existing_stream.get(), stream.get());
>         removed = false;
>         break;
>@@ -295,7 +363,7 @@ void VideoRtpReceiver::SetStreams(
>   for (auto stream : streams) {
>     bool added = true;
>     for (auto existing_stream : streams_) {
>-      if (stream->label() == existing_stream->label()) {
>+      if (stream->id() == existing_stream->id()) {
>         RTC_DCHECK_EQ(stream.get(), existing_stream.get());
>         added = false;
>         break;
>@@ -316,19 +384,6 @@ void VideoRtpReceiver::SetObserver(RtpReceiverObserverInterface* observer) {
>   }
> }
> 
>-void VideoRtpReceiver::SetMediaChannel(
>-    cricket::VideoMediaChannel* media_channel) {
>-  if (media_channel_) {
>-    SetSink(nullptr);
>-  }
>-  media_channel_ = media_channel;
>-  if (media_channel_) {
>-    if (!SetSink(&broadcaster_)) {
>-      RTC_NOTREACHED();
>-    }
>-  }
>-}
>-
> void VideoRtpReceiver::NotifyFirstPacketReceived() {
>   if (observer_) {
>     observer_->OnFirstPacketReceived(media_type());
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpreceiver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpreceiver.h
>index 82c3134a26c..263ad94013e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpreceiver.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpreceiver.h
>@@ -25,7 +25,6 @@
> #include "media/base/videobroadcaster.h"
> #include "pc/remoteaudiosource.h"
> #include "pc/videotracksource.h"
>-#include "rtc_base/basictypes.h"
> 
> namespace webrtc {
> 
>@@ -34,6 +33,21 @@ class RtpReceiverInternal : public RtpReceiverInterface {
>  public:
>   virtual void Stop() = 0;
> 
>+  // Sets the underlying MediaEngine channel associated with this RtpSender.
>+  // SetVoiceMediaChannel should be used for audio RtpSenders and
>+  // SetVideoMediaChannel should be used for video RtpSenders. Must call the
>+  // appropriate SetXxxMediaChannel(nullptr) before the media channel is
>+  // destroyed.
>+  virtual void SetVoiceMediaChannel(
>+      cricket::VoiceMediaChannel* voice_media_channel) = 0;
>+  virtual void SetVideoMediaChannel(
>+      cricket::VideoMediaChannel* video_media_channel) = 0;
>+
>+  // Configures the RtpReceiver with the underlying media channel, with the
>+  // given SSRC as the stream identifier. If |ssrc| is 0, the receiver will
>+  // receive packets on unsignaled SSRCs.
>+  virtual void SetupMediaChannel(uint32_t ssrc) = 0;
>+
>   // This SSRC is used as an identifier for the receiver between the API layer
>   // and the WebRtcVideoEngine, WebRtcVoiceEngine layer.
>   virtual uint32_t ssrc() const = 0;
>@@ -45,24 +59,31 @@ class RtpReceiverInternal : public RtpReceiverInterface {
>   // Set the associated remote media streams for this receiver. The remote track
>   // will be removed from any streams that are no longer present and added to
>   // any new streams.
>+  virtual void set_stream_ids(std::vector<std::string> stream_ids) = 0;
>+  // TODO(https://crbug.com/webrtc/9480): Remove SetStreams() in favor of
>+  // set_stream_ids() as soon as downstream projects are no longer dependent on
>+  // stream objects.
>   virtual void SetStreams(
>       const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams) = 0;
>+
>+  // Returns an ID that changes if the attached track changes, but
>+  // otherwise remains constant. Used to generate IDs for stats.
>+  // The special value zero means that no track is attached.
>+  virtual int AttachmentId() const = 0;
> };
> 
> class AudioRtpReceiver : public ObserverInterface,
>                          public AudioSourceInterface::AudioObserver,
>                          public rtc::RefCountedObject<RtpReceiverInternal> {
>  public:
>-  // An SSRC of 0 will create a receiver that will match the first SSRC it
>-  // sees.
>-  // TODO(deadbeef): Use rtc::Optional, or have another constructor that
>-  // doesn't take an SSRC, and make this one DCHECK(ssrc != 0).
>+  AudioRtpReceiver(rtc::Thread* worker_thread,
>+                   std::string receiver_id,
>+                   std::vector<std::string> stream_ids);
>+  // TODO(https://crbug.com/webrtc/9480): Remove this when streams() is removed.
>   AudioRtpReceiver(
>       rtc::Thread* worker_thread,
>       const std::string& receiver_id,
>-      const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams,
>-      uint32_t ssrc,
>-      cricket::VoiceMediaChannel* media_channel);
>+      const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams);
>   virtual ~AudioRtpReceiver();
> 
>   // ObserverInterface implementation
>@@ -79,6 +100,7 @@ class AudioRtpReceiver : public ObserverInterface,
>   rtc::scoped_refptr<MediaStreamTrackInterface> track() const override {
>     return track_.get();
>   }
>+  std::vector<std::string> stream_ids() const override;
>   std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams()
>       const override {
>     return streams_;
>@@ -95,16 +117,23 @@ class AudioRtpReceiver : public ObserverInterface,
> 
>   // RtpReceiverInternal implementation.
>   void Stop() override;
>-  uint32_t ssrc() const override { return ssrc_; }
>+  void SetupMediaChannel(uint32_t ssrc) override;
>+  uint32_t ssrc() const override { return ssrc_.value_or(0); }
>   void NotifyFirstPacketReceived() override;
>+  void set_stream_ids(std::vector<std::string> stream_ids) override;
>   void SetStreams(const std::vector<rtc::scoped_refptr<MediaStreamInterface>>&
>                       streams) override;
> 
>   void SetObserver(RtpReceiverObserverInterface* observer) override;
> 
>-  // Does not take ownership.
>-  // Should call SetMediaChannel(nullptr) before |media_channel| is destroyed.
>-  void SetMediaChannel(cricket::VoiceMediaChannel* media_channel);
>+  void SetVoiceMediaChannel(
>+      cricket::VoiceMediaChannel* voice_media_channel) override {
>+    media_channel_ = voice_media_channel;
>+  }
>+  void SetVideoMediaChannel(
>+      cricket::VideoMediaChannel* video_media_channel) override {
>+    RTC_NOTREACHED();
>+  }
> 
>   std::vector<RtpSource> GetSources() const override;
>   int AttachmentId() const override { return attachment_id_; }
>@@ -115,9 +144,10 @@ class AudioRtpReceiver : public ObserverInterface,
> 
>   rtc::Thread* const worker_thread_;
>   const std::string id_;
>-  const uint32_t ssrc_;
>-  cricket::VoiceMediaChannel* media_channel_ = nullptr;
>+  const rtc::scoped_refptr<RemoteAudioSource> source_;
>   const rtc::scoped_refptr<AudioTrackInterface> track_;
>+  cricket::VoiceMediaChannel* media_channel_ = nullptr;
>+  absl::optional<uint32_t> ssrc_;
>   std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams_;
>   bool cached_track_enabled_;
>   double cached_volume_ = 1;
>@@ -131,12 +161,15 @@ class VideoRtpReceiver : public rtc::RefCountedObject<RtpReceiverInternal> {
>  public:
>   // An SSRC of 0 will create a receiver that will match the first SSRC it
>   // sees.
>+  VideoRtpReceiver(rtc::Thread* worker_thread,
>+                   std::string receiver_id,
>+                   std::vector<std::string> streams_ids);
>+  // TODO(hbos): Remove this when streams() is removed.
>+  // https://crbug.com/webrtc/9480
>   VideoRtpReceiver(
>       rtc::Thread* worker_thread,
>       const std::string& receiver_id,
>-      const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams,
>-      uint32_t ssrc,
>-      cricket::VideoMediaChannel* media_channel);
>+      const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams);
> 
>   virtual ~VideoRtpReceiver();
> 
>@@ -148,6 +181,7 @@ class VideoRtpReceiver : public rtc::RefCountedObject<RtpReceiverInternal> {
>   rtc::scoped_refptr<MediaStreamTrackInterface> track() const override {
>     return track_.get();
>   }
>+  std::vector<std::string> stream_ids() const override;
>   std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams()
>       const override {
>     return streams_;
>@@ -164,31 +198,52 @@ class VideoRtpReceiver : public rtc::RefCountedObject<RtpReceiverInternal> {
> 
>   // RtpReceiverInternal implementation.
>   void Stop() override;
>-  uint32_t ssrc() const override { return ssrc_; }
>+  void SetupMediaChannel(uint32_t ssrc) override;
>+  uint32_t ssrc() const override { return ssrc_.value_or(0); }
>   void NotifyFirstPacketReceived() override;
>+  void set_stream_ids(std::vector<std::string> stream_ids) override;
>   void SetStreams(const std::vector<rtc::scoped_refptr<MediaStreamInterface>>&
>                       streams) override;
> 
>   void SetObserver(RtpReceiverObserverInterface* observer) override;
> 
>-  // Does not take ownership.
>-  // Should call SetMediaChannel(nullptr) before |media_channel| is destroyed.
>-  void SetMediaChannel(cricket::VideoMediaChannel* media_channel);
>+  void SetVoiceMediaChannel(
>+      cricket::VoiceMediaChannel* voice_media_channel) override {
>+    RTC_NOTREACHED();
>+  }
>+  void SetVideoMediaChannel(
>+      cricket::VideoMediaChannel* video_media_channel) override {
>+    media_channel_ = video_media_channel;
>+  }
>+
>+  int AttachmentId() const override { return attachment_id_; }
> 
>  private:
>+  class VideoRtpTrackSource : public VideoTrackSource {
>+   public:
>+    VideoRtpTrackSource() : VideoTrackSource(true /* remote */) {}
>+
>+    rtc::VideoSourceInterface<VideoFrame>* source() override {
>+      return &broadcaster_;
>+    }
>+    rtc::VideoSinkInterface<VideoFrame>* sink() { return &broadcaster_; }
>+
>+   private:
>+    // |broadcaster_| is needed since the decoder can only handle one sink.
>+    // It might be better if the decoder can handle multiple sinks and consider
>+    // the VideoSinkWants.
>+    rtc::VideoBroadcaster broadcaster_;
>+  };
>+
>   bool SetSink(rtc::VideoSinkInterface<VideoFrame>* sink);
> 
>   rtc::Thread* const worker_thread_;
>   const std::string id_;
>-  uint32_t ssrc_;
>   cricket::VideoMediaChannel* media_channel_ = nullptr;
>-  // |broadcaster_| is needed since the decoder can only handle one sink.
>-  // It might be better if the decoder can handle multiple sinks and consider
>-  // the VideoSinkWants.
>-  rtc::VideoBroadcaster broadcaster_;
>+  absl::optional<uint32_t> ssrc_;
>   // |source_| is held here to be able to change the state of the source when
>   // the VideoRtpReceiver is stopped.
>-  rtc::scoped_refptr<VideoTrackSource> source_;
>+  rtc::scoped_refptr<VideoRtpTrackSource> source_;
>   rtc::scoped_refptr<VideoTrackInterface> track_;
>   std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams_;
>   bool stopped_ = false;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpsender.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpsender.cc
>index 7c857642793..91708fa225c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpsender.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpsender.cc
>@@ -14,6 +14,7 @@
> 
> #include "api/mediastreaminterface.h"
> #include "pc/localaudiosource.h"
>+#include "pc/statscollector.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/helpers.h"
> #include "rtc_base/trace_event.h"
>@@ -29,6 +30,60 @@ int GenerateUniqueId() {
>   return ++g_unique_id;
> }
> 
>+// Returns an true if any RtpEncodingParameters member that isn't implemented
>+// contains a value.
>+bool UnimplementedRtpEncodingParameterHasValue(
>+    const RtpEncodingParameters& encoding_params) {
>+  if (encoding_params.codec_payload_type.has_value() ||
>+      encoding_params.fec.has_value() || encoding_params.rtx.has_value() ||
>+      encoding_params.dtx.has_value() || encoding_params.ptime.has_value() ||
>+      encoding_params.max_framerate.has_value() ||
>+      !encoding_params.rid.empty() ||
>+      encoding_params.scale_resolution_down_by.has_value() ||
>+      encoding_params.scale_framerate_down_by.has_value() ||
>+      !encoding_params.dependency_rids.empty()) {
>+    return true;
>+  }
>+  return false;
>+}
>+
>+// Returns true if a "per-sender" encoding parameter contains a value that isn't
>+// its default. Currently max_bitrate_bps and bitrate_priority both are
>+// implemented "per-sender," meaning that these encoding parameters
>+// are used for the RtpSender as a whole, not for a specific encoding layer.
>+// This is done by setting these encoding parameters at index 0 of
>+// RtpParameters.encodings. This function can be used to check if these
>+// parameters are set at any index other than 0 of RtpParameters.encodings,
>+// because they are currently unimplemented to be used for a specific encoding
>+// layer.
>+bool PerSenderRtpEncodingParameterHasValue(
>+    const RtpEncodingParameters& encoding_params) {
>+  if (encoding_params.bitrate_priority != kDefaultBitratePriority) {
>+    return true;
>+  }
>+  return false;
>+}
>+
>+// Returns true if any RtpParameters member that isn't implemented contains a
>+// value.
>+bool UnimplementedRtpParameterHasValue(const RtpParameters& parameters) {
>+  if (!parameters.mid.empty()) {
>+    return true;
>+  }
>+  for (size_t i = 0; i < parameters.encodings.size(); ++i) {
>+    if (UnimplementedRtpEncodingParameterHasValue(parameters.encodings[i])) {
>+      return true;
>+    }
>+    // Encoding parameters that are per-sender should only contain value at
>+    // index 0.
>+    if (i != 0 &&
>+        PerSenderRtpEncodingParameterHasValue(parameters.encodings[i])) {
>+      return true;
>+    }
>+  }
>+  return false;
>+}
>+
> }  // namespace
> 
> LocalAudioSinkAdapter::LocalAudioSinkAdapter() : sink_(nullptr) {}
>@@ -58,33 +113,16 @@ void LocalAudioSinkAdapter::SetSink(cricket::AudioSource::Sink* sink) {
> }
> 
> AudioRtpSender::AudioRtpSender(rtc::Thread* worker_thread,
>-                               StatsCollector* stats)
>-    : AudioRtpSender(worker_thread, nullptr, {rtc::CreateRandomUuid()}, stats) {
>-}
>-
>-AudioRtpSender::AudioRtpSender(rtc::Thread* worker_thread,
>-                               rtc::scoped_refptr<AudioTrackInterface> track,
>-                               const std::vector<std::string>& stream_labels,
>+                               const std::string& id,
>                                StatsCollector* stats)
>     : worker_thread_(worker_thread),
>-      id_(track ? track->id() : rtc::CreateRandomUuid()),
>-      stream_ids_(stream_labels),
>+      id_(id),
>       stats_(stats),
>-      track_(track),
>       dtmf_sender_proxy_(DtmfSenderProxy::Create(
>           rtc::Thread::Current(),
>-          DtmfSender::Create(track_, rtc::Thread::Current(), this))),
>-      cached_track_enabled_(track ? track->enabled() : false),
>-      sink_adapter_(new LocalAudioSinkAdapter()),
>-      attachment_id_(track ? GenerateUniqueId() : 0) {
>+          DtmfSender::Create(rtc::Thread::Current(), this))),
>+      sink_adapter_(new LocalAudioSinkAdapter()) {
>   RTC_DCHECK(worker_thread);
>-  // TODO(bugs.webrtc.org/7932): Remove once zero or multiple streams are
>-  // supported.
>-  RTC_DCHECK_EQ(stream_labels.size(), 1u);
>-  if (track_) {
>-    track_->RegisterObserver(this);
>-    track_->AddSink(sink_adapter_.get());
>-  }
> }
> 
> AudioRtpSender::~AudioRtpSender() {
>@@ -110,11 +148,11 @@ bool AudioRtpSender::CanInsertDtmf() {
> 
> bool AudioRtpSender::InsertDtmf(int code, int duration) {
>   if (!media_channel_) {
>-    RTC_LOG(LS_ERROR) << "CanInsertDtmf: No audio channel exists.";
>+    RTC_LOG(LS_ERROR) << "InsertDtmf: No audio channel exists.";
>     return false;
>   }
>   if (!ssrc_) {
>-    RTC_LOG(LS_ERROR) << "CanInsertDtmf: Sender does not have SSRC.";
>+    RTC_LOG(LS_ERROR) << "InsertDtmf: Sender does not have SSRC.";
>     return false;
>   }
>   bool success = worker_thread_->Invoke<bool>(RTC_FROM_HERE, [&] {
>@@ -185,26 +223,49 @@ bool AudioRtpSender::SetTrack(MediaStreamTrackInterface* track) {
>   } else if (prev_can_send_track) {
>     ClearAudioSend();
>   }
>-  attachment_id_ = GenerateUniqueId();
>+  attachment_id_ = (track_ ? GenerateUniqueId() : 0);
>   return true;
> }
> 
>-RtpParameters AudioRtpSender::GetParameters() const {
>+RtpParameters AudioRtpSender::GetParameters() {
>   if (!media_channel_ || stopped_) {
>     return RtpParameters();
>   }
>   return worker_thread_->Invoke<RtpParameters>(RTC_FROM_HERE, [&] {
>-    return media_channel_->GetRtpSendParameters(ssrc_);
>+    RtpParameters result = media_channel_->GetRtpSendParameters(ssrc_);
>+    last_transaction_id_ = rtc::CreateRandomUuid();
>+    result.transaction_id = last_transaction_id_.value();
>+    return result;
>   });
> }
> 
>-bool AudioRtpSender::SetParameters(const RtpParameters& parameters) {
>+RTCError AudioRtpSender::SetParameters(const RtpParameters& parameters) {
>   TRACE_EVENT0("webrtc", "AudioRtpSender::SetParameters");
>   if (!media_channel_ || stopped_) {
>-    return false;
>-  }
>-  return worker_thread_->Invoke<bool>(RTC_FROM_HERE, [&] {
>-    return media_channel_->SetRtpSendParameters(ssrc_, parameters);
>+    return RTCError(RTCErrorType::INVALID_STATE);
>+  }
>+  if (!last_transaction_id_) {
>+    LOG_AND_RETURN_ERROR(
>+        RTCErrorType::INVALID_STATE,
>+        "Failed to set parameters since getParameters() has never been called"
>+        " on this sender");
>+  }
>+  if (last_transaction_id_ != parameters.transaction_id) {
>+    LOG_AND_RETURN_ERROR(
>+        RTCErrorType::INVALID_MODIFICATION,
>+        "Failed to set parameters since the transaction_id doesn't match"
>+        " the last value returned from getParameters()");
>+  }
>+
>+  if (UnimplementedRtpParameterHasValue(parameters)) {
>+    LOG_AND_RETURN_ERROR(
>+        RTCErrorType::UNSUPPORTED_PARAMETER,
>+        "Attempted to set an unimplemented parameter of RtpParameters.");
>+  }
>+  return worker_thread_->Invoke<RTCError>(RTC_FROM_HERE, [&] {
>+    RTCError result = media_channel_->SetRtpSendParameters(ssrc_, parameters);
>+    last_transaction_id_.reset();
>+    return result;
>   });
> }
> 
>@@ -249,6 +310,7 @@ void AudioRtpSender::Stop() {
>       stats_->RemoveLocalAudioTrack(track_.get(), ssrc_);
>     }
>   }
>+  media_channel_ = nullptr;
>   stopped_ = true;
> }
> 
>@@ -300,28 +362,10 @@ void AudioRtpSender::ClearAudioSend() {
>   }
> }
> 
>-VideoRtpSender::VideoRtpSender(rtc::Thread* worker_thread)
>-    : VideoRtpSender(worker_thread, nullptr, {rtc::CreateRandomUuid()}) {}
>-
> VideoRtpSender::VideoRtpSender(rtc::Thread* worker_thread,
>-                               rtc::scoped_refptr<VideoTrackInterface> track,
>-                               const std::vector<std::string>& stream_labels)
>-    : worker_thread_(worker_thread),
>-      id_(track ? track->id() : rtc::CreateRandomUuid()),
>-      stream_ids_(stream_labels),
>-      track_(track),
>-      cached_track_enabled_(track ? track->enabled() : false),
>-      cached_track_content_hint_(
>-          track ? track->content_hint()
>-                : VideoTrackInterface::ContentHint::kNone),
>-      attachment_id_(track ? GenerateUniqueId() : 0) {
>+                               const std::string& id)
>+    : worker_thread_(worker_thread), id_(id) {
>   RTC_DCHECK(worker_thread);
>-  // TODO(bugs.webrtc.org/7932): Remove once zero or multiple streams are
>-  // supported.
>-  RTC_DCHECK_EQ(stream_labels.size(), 1u);
>-  if (track_) {
>-    track_->RegisterObserver(this);
>-  }
> }
> 
> VideoRtpSender::~VideoRtpSender() {
>@@ -331,9 +375,7 @@ VideoRtpSender::~VideoRtpSender() {
> void VideoRtpSender::OnChanged() {
>   TRACE_EVENT0("webrtc", "VideoRtpSender::OnChanged");
>   RTC_DCHECK(!stopped_);
>-  if (cached_track_enabled_ != track_->enabled() ||
>-      cached_track_content_hint_ != track_->content_hint()) {
>-    cached_track_enabled_ = track_->enabled();
>+  if (cached_track_content_hint_ != track_->content_hint()) {
>     cached_track_content_hint_ = track_->content_hint();
>     if (can_send_track()) {
>       SetVideoSend();
>@@ -366,7 +408,6 @@ bool VideoRtpSender::SetTrack(MediaStreamTrackInterface* track) {
>   rtc::scoped_refptr<VideoTrackInterface> old_track = track_;
>   track_ = video_track;
>   if (track_) {
>-    cached_track_enabled_ = track_->enabled();
>     cached_track_content_hint_ = track_->content_hint();
>     track_->RegisterObserver(this);
>   }
>@@ -377,26 +418,49 @@ bool VideoRtpSender::SetTrack(MediaStreamTrackInterface* track) {
>   } else if (prev_can_send_track) {
>     ClearVideoSend();
>   }
>-  attachment_id_ = GenerateUniqueId();
>+  attachment_id_ = (track_ ? GenerateUniqueId() : 0);
>   return true;
> }
> 
>-RtpParameters VideoRtpSender::GetParameters() const {
>+RtpParameters VideoRtpSender::GetParameters() {
>   if (!media_channel_ || stopped_) {
>     return RtpParameters();
>   }
>   return worker_thread_->Invoke<RtpParameters>(RTC_FROM_HERE, [&] {
>-    return media_channel_->GetRtpSendParameters(ssrc_);
>+    RtpParameters result = media_channel_->GetRtpSendParameters(ssrc_);
>+    last_transaction_id_ = rtc::CreateRandomUuid();
>+    result.transaction_id = last_transaction_id_.value();
>+    return result;
>   });
> }
> 
>-bool VideoRtpSender::SetParameters(const RtpParameters& parameters) {
>+RTCError VideoRtpSender::SetParameters(const RtpParameters& parameters) {
>   TRACE_EVENT0("webrtc", "VideoRtpSender::SetParameters");
>   if (!media_channel_ || stopped_) {
>-    return false;
>-  }
>-  return worker_thread_->Invoke<bool>(RTC_FROM_HERE, [&] {
>-    return media_channel_->SetRtpSendParameters(ssrc_, parameters);
>+    return RTCError(RTCErrorType::INVALID_STATE);
>+  }
>+  if (!last_transaction_id_) {
>+    LOG_AND_RETURN_ERROR(
>+        RTCErrorType::INVALID_STATE,
>+        "Failed to set parameters since getParameters() has never been called"
>+        " on this sender");
>+  }
>+  if (last_transaction_id_ != parameters.transaction_id) {
>+    LOG_AND_RETURN_ERROR(
>+        RTCErrorType::INVALID_MODIFICATION,
>+        "Failed to set parameters since the transaction_id doesn't match"
>+        " the last value returned from getParameters()");
>+  }
>+
>+  if (UnimplementedRtpParameterHasValue(parameters)) {
>+    LOG_AND_RETURN_ERROR(
>+        RTCErrorType::UNSUPPORTED_PARAMETER,
>+        "Attempted to set an unimplemented parameter of RtpParameters.");
>+  }
>+  return worker_thread_->Invoke<RTCError>(RTC_FROM_HERE, [&] {
>+    RTCError result = media_channel_->SetRtpSendParameters(ssrc_, parameters);
>+    last_transaction_id_.reset();
>+    return result;
>   });
> }
> 
>@@ -432,6 +496,7 @@ void VideoRtpSender::Stop() {
>   if (can_send_track()) {
>     ClearVideoSend();
>   }
>+  media_channel_ = nullptr;
>   stopped_ = true;
> }
> 
>@@ -455,14 +520,12 @@ void VideoRtpSender::SetVideoSend() {
>       options.is_screencast = false;
>       break;
>     case VideoTrackInterface::ContentHint::kDetailed:
>+    case VideoTrackInterface::ContentHint::kText:
>       options.is_screencast = true;
>       break;
>   }
>-  // |track_->enabled()| hops to the signaling thread, so call it before we hop
>-  // to the worker thread or else it will deadlock.
>-  bool track_enabled = track_->enabled();
>   bool success = worker_thread_->Invoke<bool>(RTC_FROM_HERE, [&] {
>-    return media_channel_->SetVideoSend(ssrc_, track_enabled, &options, track_);
>+    return media_channel_->SetVideoSend(ssrc_, &options, track_);
>   });
>   RTC_DCHECK(success);
> }
>@@ -478,7 +541,7 @@ void VideoRtpSender::ClearVideoSend() {
>   // This the normal case when the underlying media channel has already been
>   // deleted.
>   worker_thread_->Invoke<bool>(RTC_FROM_HERE, [&] {
>-    return media_channel_->SetVideoSend(ssrc_, false, nullptr, nullptr);
>+    return media_channel_->SetVideoSend(ssrc_, nullptr, nullptr);
>   });
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpsender.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpsender.h
>index 119d91f11a8..983bb20ae0a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpsender.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpsender.h
>@@ -21,34 +21,42 @@
> 
> #include "api/mediastreaminterface.h"
> #include "api/rtpsenderinterface.h"
>-#include "rtc_base/basictypes.h"
>-#include "rtc_base/criticalsection.h"
>-// Adding 'nogncheck' to disable the gn include headers check to support modular
>-// WebRTC build targets.
>-#include "media/base/audiosource.h"  // nogncheck
>+#include "media/base/audiosource.h"
>+#include "media/base/mediachannel.h"
> #include "pc/dtmfsender.h"
>-#include "pc/statscollector.h"
>+#include "rtc_base/criticalsection.h"
> 
> namespace webrtc {
> 
>+class StatsCollector;
>+
> // Internal interface used by PeerConnection.
> class RtpSenderInternal : public RtpSenderInterface {
>  public:
>+  // Sets the underlying MediaEngine channel associated with this RtpSender.
>+  // SetVoiceMediaChannel should be used for audio RtpSenders and
>+  // SetVideoMediaChannel should be used for video RtpSenders. Must call the
>+  // appropriate SetXxxMediaChannel(nullptr) before the media channel is
>+  // destroyed.
>+  virtual void SetVoiceMediaChannel(
>+      cricket::VoiceMediaChannel* voice_media_channel) = 0;
>+  virtual void SetVideoMediaChannel(
>+      cricket::VideoMediaChannel* video_media_channel) = 0;
>+
>   // Used to set the SSRC of the sender, once a local description has been set.
>   // If |ssrc| is 0, this indiates that the sender should disconnect from the
>   // underlying transport (this occurs if the sender isn't seen in a local
>   // description).
>   virtual void SetSsrc(uint32_t ssrc) = 0;
> 
>-  // TODO(steveanton): With Unified Plan, a track/RTCRTPSender can be part of
>-  // multiple streams (or no stream at all). Replace these singular methods with
>-  // their corresponding plural methods.
>-  // Until these are removed, RtpSenders must have exactly one stream.
>-  virtual void set_stream_id(const std::string& stream_id) = 0;
>-  virtual std::string stream_id() const = 0;
>   virtual void set_stream_ids(const std::vector<std::string>& stream_ids) = 0;
> 
>   virtual void Stop() = 0;
>+
>+  // Returns an ID that changes every time SetTrack() is called, but
>+  // otherwise remains constant. Used to generate IDs for stats.
>+  // The special value zero means that no track is attached.
>+  virtual int AttachmentId() const = 0;
> };
> 
> // LocalAudioSinkAdapter receives data callback as a sink to the local
>@@ -82,15 +90,10 @@ class AudioRtpSender : public DtmfProviderInterface,
>   // StatsCollector provided so that Add/RemoveLocalAudioTrack can be called
>   // at the appropriate times.
> 
>-  // Construct an AudioRtpSender with a null track, a single, randomly generated
>-  // stream label, and a randomly generated ID.
>-  AudioRtpSender(rtc::Thread* worker_thread, StatsCollector* stats);
>-
>-  // Construct an AudioRtpSender with the given track and stream labels. The
>-  // sender ID will be set to the track's ID.
>+  // Construct an RtpSender for audio with the given sender ID.
>+  // The sender is initialized with no track to send and no associated streams.
>   AudioRtpSender(rtc::Thread* worker_thread,
>-                 rtc::scoped_refptr<AudioTrackInterface> track,
>-                 const std::vector<std::string>& stream_labels,
>+                 const std::string& id,
>                  StatsCollector* stats);
> 
>   virtual ~AudioRtpSender();
>@@ -119,18 +122,14 @@ class AudioRtpSender : public DtmfProviderInterface,
> 
>   std::vector<std::string> stream_ids() const override { return stream_ids_; }
> 
>-  RtpParameters GetParameters() const override;
>-  bool SetParameters(const RtpParameters& parameters) override;
>+  RtpParameters GetParameters() override;
>+  RTCError SetParameters(const RtpParameters& parameters) override;
> 
>   rtc::scoped_refptr<DtmfSenderInterface> GetDtmfSender() const override;
> 
>   // RtpSenderInternal implementation.
>   void SetSsrc(uint32_t ssrc) override;
> 
>-  void set_stream_id(const std::string& stream_id) override {
>-    stream_ids_ = {stream_id};
>-  }
>-  std::string stream_id() const override { return stream_ids_[0]; }
>   void set_stream_ids(const std::vector<std::string>& stream_ids) override {
>     stream_ids_ = stream_ids;
>   }
>@@ -139,10 +138,13 @@ class AudioRtpSender : public DtmfProviderInterface,
> 
>   int AttachmentId() const override { return attachment_id_; }
> 
>-  // Does not take ownership.
>-  // Should call SetMediaChannel(nullptr) before |media_channel| is destroyed.
>-  void SetMediaChannel(cricket::VoiceMediaChannel* media_channel) {
>-    media_channel_ = media_channel;
>+  void SetVoiceMediaChannel(
>+      cricket::VoiceMediaChannel* voice_media_channel) override {
>+    media_channel_ = voice_media_channel;
>+  }
>+  void SetVideoMediaChannel(
>+      cricket::VideoMediaChannel* video_media_channel) override {
>+    RTC_NOTREACHED();
>   }
> 
>  private:
>@@ -159,13 +161,12 @@ class AudioRtpSender : public DtmfProviderInterface,
> 
>   rtc::Thread* const worker_thread_;
>   const std::string id_;
>-  // TODO(steveanton): Until more Unified Plan work is done, this can only have
>-  // exactly one element.
>   std::vector<std::string> stream_ids_;
>   cricket::VoiceMediaChannel* media_channel_ = nullptr;
>-  StatsCollector* stats_;
>+  StatsCollector* stats_ = nullptr;
>   rtc::scoped_refptr<AudioTrackInterface> track_;
>   rtc::scoped_refptr<DtmfSenderInterface> dtmf_sender_proxy_;
>+  absl::optional<std::string> last_transaction_id_;
>   uint32_t ssrc_ = 0;
>   bool cached_track_enabled_ = false;
>   bool stopped_ = false;
>@@ -179,15 +180,9 @@ class AudioRtpSender : public DtmfProviderInterface,
> class VideoRtpSender : public ObserverInterface,
>                        public rtc::RefCountedObject<RtpSenderInternal> {
>  public:
>-  // Construct a VideoRtpSender with a null track, a single, randomly generated
>-  // stream label, and a randomly generated ID.
>-  explicit VideoRtpSender(rtc::Thread* worker_thread);
>-
>-  // Construct a VideoRtpSender with the given track and stream labels. The
>-  // sender ID will be set to the track's ID.
>-  VideoRtpSender(rtc::Thread* worker_thread,
>-                 rtc::scoped_refptr<VideoTrackInterface> track,
>-                 const std::vector<std::string>& stream_labels);
>+  // Construct an RtpSender for video with the given sender ID.
>+  // The sender is initialized with no track to send and no associated streams.
>+  VideoRtpSender(rtc::Thread* worker_thread, const std::string& id);
> 
>   virtual ~VideoRtpSender();
> 
>@@ -210,18 +205,14 @@ class VideoRtpSender : public ObserverInterface,
> 
>   std::vector<std::string> stream_ids() const override { return stream_ids_; }
> 
>-  RtpParameters GetParameters() const override;
>-  bool SetParameters(const RtpParameters& parameters) override;
>+  RtpParameters GetParameters() override;
>+  RTCError SetParameters(const RtpParameters& parameters) override;
> 
>   rtc::scoped_refptr<DtmfSenderInterface> GetDtmfSender() const override;
> 
>   // RtpSenderInternal implementation.
>   void SetSsrc(uint32_t ssrc) override;
> 
>-  void set_stream_id(const std::string& stream_id) override {
>-    stream_ids_ = {stream_id};
>-  }
>-  std::string stream_id() const override { return stream_ids_[0]; }
>   void set_stream_ids(const std::vector<std::string>& stream_ids) override {
>     stream_ids_ = stream_ids;
>   }
>@@ -229,10 +220,13 @@ class VideoRtpSender : public ObserverInterface,
>   void Stop() override;
>   int AttachmentId() const override { return attachment_id_; }
> 
>-  // Does not take ownership.
>-  // Should call SetMediaChannel(nullptr) before |media_channel| is destroyed.
>-  void SetMediaChannel(cricket::VideoMediaChannel* media_channel) {
>-    media_channel_ = media_channel;
>+  void SetVoiceMediaChannel(
>+      cricket::VoiceMediaChannel* voice_media_channel) override {
>+    RTC_NOTREACHED();
>+  }
>+  void SetVideoMediaChannel(
>+      cricket::VideoMediaChannel* video_media_channel) override {
>+    media_channel_ = video_media_channel;
>   }
> 
>  private:
>@@ -245,13 +239,11 @@ class VideoRtpSender : public ObserverInterface,
> 
>   rtc::Thread* worker_thread_;
>   const std::string id_;
>-  // TODO(steveanton): Until more Unified Plan work is done, this can only have
>-  // exactly one element.
>   std::vector<std::string> stream_ids_;
>   cricket::VideoMediaChannel* media_channel_ = nullptr;
>   rtc::scoped_refptr<VideoTrackInterface> track_;
>+  absl::optional<std::string> last_transaction_id_;
>   uint32_t ssrc_ = 0;
>-  bool cached_track_enabled_ = false;
>   VideoTrackInterface::ContentHint cached_track_content_hint_ =
>       VideoTrackInterface::ContentHint::kNone;
>   bool stopped_ = false;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpsenderreceiver_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpsenderreceiver_unittest.cc
>index 9da9c6dc015..4014df54cb4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpsenderreceiver_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtpsenderreceiver_unittest.cc
>@@ -15,7 +15,9 @@
> #include "api/rtpparameters.h"
> #include "media/base/fakemediaengine.h"
> #include "media/base/rtpdataengine.h"
>+#include "media/base/testutils.h"
> #include "media/engine/fakewebrtccall.h"
>+#include "p2p/base/fakedtlstransport.h"
> #include "pc/audiotrack.h"
> #include "pc/channelmanager.h"
> #include "pc/localaudiosource.h"
>@@ -24,7 +26,6 @@
> #include "pc/rtpreceiver.h"
> #include "pc/rtpsender.h"
> #include "pc/streamcollection.h"
>-#include "pc/test/faketransportcontroller.h"
> #include "pc/test/fakevideotracksource.h"
> #include "pc/videotrack.h"
> #include "pc/videotracksource.h"
>@@ -39,7 +40,7 @@ using ::testing::Return;
> 
> namespace {
> 
>-static const char kStreamLabel1[] = "local_stream_1";
>+static const char kStreamId1[] = "local_stream_1";
> static const char kVideoTrackId[] = "video_1";
> static const char kAudioTrackId[] = "audio_1";
> static const uint32_t kVideoSsrc = 98;
>@@ -60,26 +61,27 @@ class RtpSenderReceiverTest : public testing::Test,
>         // Create fake media engine/etc. so we can create channels to use to
>         // test RtpSenders/RtpReceivers.
>         media_engine_(new cricket::FakeMediaEngine()),
>-        channel_manager_(rtc::WrapUnique(media_engine_),
>-                         rtc::MakeUnique<cricket::RtpDataEngine>(),
>+        channel_manager_(absl::WrapUnique(media_engine_),
>+                         absl::make_unique<cricket::RtpDataEngine>(),
>                          worker_thread_,
>                          network_thread_),
>-        fake_call_(Call::Config(&event_log_)),
>-        local_stream_(MediaStream::Create(kStreamLabel1)) {
>+        fake_call_(),
>+        local_stream_(MediaStream::Create(kStreamId1)) {
>     // Create channels to be used by the RtpSenders and RtpReceivers.
>     channel_manager_.Init();
>     bool srtp_required = true;
>-    cricket::DtlsTransportInternal* rtp_transport =
>-        fake_transport_controller_.CreateDtlsTransport(
>-            cricket::CN_AUDIO, cricket::ICE_CANDIDATE_COMPONENT_RTP);
>+    rtp_dtls_transport_ = absl::make_unique<cricket::FakeDtlsTransport>(
>+        "fake_dtls_transport", cricket::ICE_CANDIDATE_COMPONENT_RTP);
>+    rtp_transport_ = CreateDtlsSrtpTransport();
>+
>     voice_channel_ = channel_manager_.CreateVoiceChannel(
>-        &fake_call_, cricket::MediaConfig(),
>-        rtp_transport, nullptr, rtc::Thread::Current(),
>-        cricket::CN_AUDIO, srtp_required, cricket::AudioOptions());
>+        &fake_call_, cricket::MediaConfig(), rtp_transport_.get(),
>+        rtc::Thread::Current(), cricket::CN_AUDIO, srtp_required,
>+        rtc::CryptoOptions(), cricket::AudioOptions());
>     video_channel_ = channel_manager_.CreateVideoChannel(
>-        &fake_call_, cricket::MediaConfig(),
>-        rtp_transport, nullptr, rtc::Thread::Current(),
>-        cricket::CN_VIDEO, srtp_required, cricket::VideoOptions());
>+        &fake_call_, cricket::MediaConfig(), rtp_transport_.get(),
>+        rtc::Thread::Current(), cricket::CN_VIDEO, srtp_required,
>+        rtc::CryptoOptions(), cricket::VideoOptions());
>     voice_channel_->Enable(true);
>     video_channel_->Enable(true);
>     voice_media_channel_ = media_engine_->GetVoiceChannel(0);
>@@ -111,6 +113,14 @@ class RtpSenderReceiverTest : public testing::Test,
>         cricket::StreamParams::CreateLegacy(kVideoSsrc2));
>   }
> 
>+  std::unique_ptr<webrtc::RtpTransportInternal> CreateDtlsSrtpTransport() {
>+    auto dtls_srtp_transport = absl::make_unique<webrtc::DtlsSrtpTransport>(
>+        /*rtcp_mux_required=*/true);
>+    dtls_srtp_transport->SetDtlsTransports(rtp_dtls_transport_.get(),
>+                                           /*rtcp_dtls_transport=*/nullptr);
>+    return dtls_srtp_transport;
>+  }
>+
>   // Needed to use DTMF sender.
>   void AddDtmfCodec() {
>     cricket::AudioSendParameters params;
>@@ -137,9 +147,10 @@ class RtpSenderReceiverTest : public testing::Test,
>     audio_track_ = AudioTrack::Create(kAudioTrackId, source);
>     EXPECT_TRUE(local_stream_->AddTrack(audio_track_));
>     audio_rtp_sender_ =
>-        new AudioRtpSender(worker_thread_, local_stream_->GetAudioTracks()[0],
>-                           {local_stream_->label()}, nullptr);
>-    audio_rtp_sender_->SetMediaChannel(voice_media_channel_);
>+        new AudioRtpSender(worker_thread_, audio_track_->id(), nullptr);
>+    ASSERT_TRUE(audio_rtp_sender_->SetTrack(audio_track_));
>+    audio_rtp_sender_->set_stream_ids({local_stream_->id()});
>+    audio_rtp_sender_->SetVoiceMediaChannel(voice_media_channel_);
>     audio_rtp_sender_->SetSsrc(kAudioSsrc);
>     audio_rtp_sender_->GetOnDestroyedSignal()->connect(
>         this, &RtpSenderReceiverTest::OnAudioSenderDestroyed);
>@@ -147,27 +158,31 @@ class RtpSenderReceiverTest : public testing::Test,
>   }
> 
>   void CreateAudioRtpSenderWithNoTrack() {
>-    audio_rtp_sender_ = new AudioRtpSender(worker_thread_, nullptr);
>-    audio_rtp_sender_->SetMediaChannel(voice_media_channel_);
>+    audio_rtp_sender_ = new AudioRtpSender(worker_thread_, /*id=*/"", nullptr);
>+    audio_rtp_sender_->SetVoiceMediaChannel(voice_media_channel_);
>   }
> 
>   void OnAudioSenderDestroyed() { audio_sender_destroyed_signal_fired_ = true; }
> 
>+  void CreateVideoRtpSender(uint32_t ssrc) {
>+    CreateVideoRtpSender(false, ssrc);
>+  }
>+
>   void CreateVideoRtpSender() { CreateVideoRtpSender(false); }
> 
>-  void CreateVideoRtpSender(bool is_screencast) {
>+  void CreateVideoRtpSender(bool is_screencast, uint32_t ssrc = kVideoSsrc) {
>     AddVideoTrack(is_screencast);
>-    video_rtp_sender_ =
>-        new VideoRtpSender(worker_thread_, local_stream_->GetVideoTracks()[0],
>-                           {local_stream_->label()});
>-    video_rtp_sender_->SetMediaChannel(video_media_channel_);
>-    video_rtp_sender_->SetSsrc(kVideoSsrc);
>-    VerifyVideoChannelInput();
>+    video_rtp_sender_ = new VideoRtpSender(worker_thread_, video_track_->id());
>+    ASSERT_TRUE(video_rtp_sender_->SetTrack(video_track_));
>+    video_rtp_sender_->set_stream_ids({local_stream_->id()});
>+    video_rtp_sender_->SetVideoMediaChannel(video_media_channel_);
>+    video_rtp_sender_->SetSsrc(ssrc);
>+    VerifyVideoChannelInput(ssrc);
>   }
> 
>   void CreateVideoRtpSenderWithNoTrack() {
>-    video_rtp_sender_ = new VideoRtpSender(worker_thread_);
>-    video_rtp_sender_->SetMediaChannel(video_media_channel_);
>+    video_rtp_sender_ = new VideoRtpSender(worker_thread_, /*id=*/"");
>+    video_rtp_sender_->SetVideoMediaChannel(video_media_channel_);
>   }
> 
>   void DestroyAudioRtpSender() {
>@@ -183,8 +198,9 @@ class RtpSenderReceiverTest : public testing::Test,
>   void CreateAudioRtpReceiver(
>       std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams = {}) {
>     audio_rtp_receiver_ = new AudioRtpReceiver(
>-        rtc::Thread::Current(), kAudioTrackId, std::move(streams), kAudioSsrc,
>-        voice_media_channel_);
>+        rtc::Thread::Current(), kAudioTrackId, std::move(streams));
>+    audio_rtp_receiver_->SetVoiceMediaChannel(voice_media_channel_);
>+    audio_rtp_receiver_->SetupMediaChannel(kAudioSsrc);
>     audio_track_ = audio_rtp_receiver_->audio_track();
>     VerifyVoiceChannelOutput();
>   }
>@@ -192,8 +208,9 @@ class RtpSenderReceiverTest : public testing::Test,
>   void CreateVideoRtpReceiver(
>       std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams = {}) {
>     video_rtp_receiver_ = new VideoRtpReceiver(
>-        rtc::Thread::Current(), kVideoTrackId, std::move(streams), kVideoSsrc,
>-        video_media_channel_);
>+        rtc::Thread::Current(), kVideoTrackId, std::move(streams));
>+    video_rtp_receiver_->SetVideoMediaChannel(video_media_channel_);
>+    video_rtp_receiver_->SetupMediaChannel(kVideoSsrc);
>     video_track_ = video_rtp_receiver_->video_track();
>     VerifyVideoChannelOutput();
>   }
>@@ -266,9 +283,12 @@ class RtpSenderReceiverTest : public testing::Test,
>   rtc::Thread* const network_thread_;
>   rtc::Thread* const worker_thread_;
>   webrtc::RtcEventLogNullImpl event_log_;
>+  // The |rtp_dtls_transport_| and |rtp_transport_| should be destroyed after
>+  // the |channel_manager|.
>+  std::unique_ptr<cricket::DtlsTransportInternal> rtp_dtls_transport_;
>+  std::unique_ptr<webrtc::RtpTransportInternal> rtp_transport_;
>   // |media_engine_| is actually owned by |channel_manager_|.
>   cricket::FakeMediaEngine* media_engine_;
>-  cricket::FakeTransportController fake_transport_controller_;
>   cricket::ChannelManager channel_manager_;
>   cricket::FakeCall fake_call_;
>   cricket::VoiceChannel* voice_channel_;
>@@ -587,7 +607,135 @@ TEST_F(RtpSenderReceiverTest, AudioSenderCanSetParameters) {
> 
>   RtpParameters params = audio_rtp_sender_->GetParameters();
>   EXPECT_EQ(1u, params.encodings.size());
>-  EXPECT_TRUE(audio_rtp_sender_->SetParameters(params));
>+  EXPECT_TRUE(audio_rtp_sender_->SetParameters(params).ok());
>+
>+  DestroyAudioRtpSender();
>+}
>+
>+TEST_F(RtpSenderReceiverTest,
>+       AudioSenderMustCallGetParametersBeforeSetParameters) {
>+  CreateAudioRtpSender();
>+
>+  RtpParameters params;
>+  RTCError result = audio_rtp_sender_->SetParameters(params);
>+  EXPECT_EQ(RTCErrorType::INVALID_STATE, result.type());
>+
>+  DestroyAudioRtpSender();
>+}
>+
>+TEST_F(RtpSenderReceiverTest,
>+       AudioSenderSetParametersInvalidatesTransactionId) {
>+  CreateAudioRtpSender();
>+
>+  RtpParameters params = audio_rtp_sender_->GetParameters();
>+  EXPECT_EQ(1u, params.encodings.size());
>+  EXPECT_TRUE(audio_rtp_sender_->SetParameters(params).ok());
>+  RTCError result = audio_rtp_sender_->SetParameters(params);
>+  EXPECT_EQ(RTCErrorType::INVALID_STATE, result.type());
>+
>+  DestroyAudioRtpSender();
>+}
>+
>+TEST_F(RtpSenderReceiverTest, AudioSenderDetectTransactionIdModification) {
>+  CreateAudioRtpSender();
>+
>+  RtpParameters params = audio_rtp_sender_->GetParameters();
>+  params.transaction_id = "";
>+  RTCError result = audio_rtp_sender_->SetParameters(params);
>+  EXPECT_EQ(RTCErrorType::INVALID_MODIFICATION, result.type());
>+
>+  DestroyAudioRtpSender();
>+}
>+
>+TEST_F(RtpSenderReceiverTest, AudioSenderCheckTransactionIdRefresh) {
>+  CreateAudioRtpSender();
>+
>+  RtpParameters params = audio_rtp_sender_->GetParameters();
>+  EXPECT_NE(params.transaction_id.size(), 0U);
>+  auto saved_transaction_id = params.transaction_id;
>+  params = audio_rtp_sender_->GetParameters();
>+  EXPECT_NE(saved_transaction_id, params.transaction_id);
>+
>+  DestroyAudioRtpSender();
>+}
>+
>+TEST_F(RtpSenderReceiverTest, AudioSenderSetParametersOldValueFail) {
>+  CreateAudioRtpSender();
>+
>+  RtpParameters params = audio_rtp_sender_->GetParameters();
>+  RtpParameters second_params = audio_rtp_sender_->GetParameters();
>+
>+  RTCError result = audio_rtp_sender_->SetParameters(params);
>+  EXPECT_EQ(RTCErrorType::INVALID_MODIFICATION, result.type());
>+  DestroyAudioRtpSender();
>+}
>+
>+TEST_F(RtpSenderReceiverTest, AudioSenderCantSetUnimplementedRtpParameters) {
>+  CreateAudioRtpSender();
>+  RtpParameters params = audio_rtp_sender_->GetParameters();
>+  EXPECT_EQ(1u, params.encodings.size());
>+
>+  // Unimplemented RtpParameters: mid
>+  params.mid = "dummy_mid";
>+  EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
>+            audio_rtp_sender_->SetParameters(params).type());
>+  params = audio_rtp_sender_->GetParameters();
>+
>+  DestroyAudioRtpSender();
>+}
>+
>+TEST_F(RtpSenderReceiverTest,
>+       AudioSenderCantSetUnimplementedRtpEncodingParameters) {
>+  CreateAudioRtpSender();
>+  RtpParameters params = audio_rtp_sender_->GetParameters();
>+  EXPECT_EQ(1u, params.encodings.size());
>+
>+  // Unimplemented RtpParameters: codec_payload_type, fec, rtx, dtx, ptime,
>+  // max_framerate, scale_resolution_down_by, scale_framerate_down_by, rid,
>+  // dependency_rids.
>+  params.encodings[0].codec_payload_type = 1;
>+  EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
>+            audio_rtp_sender_->SetParameters(params).type());
>+  params = audio_rtp_sender_->GetParameters();
>+
>+  params.encodings[0].fec = RtpFecParameters();
>+  EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
>+            audio_rtp_sender_->SetParameters(params).type());
>+  params = audio_rtp_sender_->GetParameters();
>+
>+  params.encodings[0].rtx = RtpRtxParameters();
>+  EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
>+            audio_rtp_sender_->SetParameters(params).type());
>+  params = audio_rtp_sender_->GetParameters();
>+
>+  params.encodings[0].dtx = DtxStatus::ENABLED;
>+  EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
>+            audio_rtp_sender_->SetParameters(params).type());
>+  params = audio_rtp_sender_->GetParameters();
>+
>+  params.encodings[0].ptime = 1;
>+  EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
>+            audio_rtp_sender_->SetParameters(params).type());
>+  params = audio_rtp_sender_->GetParameters();
>+
>+  params.encodings[0].max_framerate = 1;
>+  EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
>+            audio_rtp_sender_->SetParameters(params).type());
>+  params = audio_rtp_sender_->GetParameters();
>+
>+  params.encodings[0].scale_resolution_down_by = 2.0;
>+  EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
>+            audio_rtp_sender_->SetParameters(params).type());
>+  params = audio_rtp_sender_->GetParameters();
>+
>+  params.encodings[0].rid = "dummy_rid";
>+  EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
>+            audio_rtp_sender_->SetParameters(params).type());
>+  params = audio_rtp_sender_->GetParameters();
>+
>+  params.encodings[0].dependency_rids.push_back("dummy_rid");
>+  EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
>+            audio_rtp_sender_->SetParameters(params).type());
> 
>   DestroyAudioRtpSender();
> }
>@@ -597,19 +745,19 @@ TEST_F(RtpSenderReceiverTest, SetAudioMaxSendBitrate) {
> 
>   EXPECT_EQ(-1, voice_media_channel_->max_bps());
>   webrtc::RtpParameters params = audio_rtp_sender_->GetParameters();
>-  EXPECT_EQ(1, params.encodings.size());
>+  EXPECT_EQ(1U, params.encodings.size());
>   EXPECT_FALSE(params.encodings[0].max_bitrate_bps);
>   params.encodings[0].max_bitrate_bps = 1000;
>-  EXPECT_TRUE(audio_rtp_sender_->SetParameters(params));
>+  EXPECT_TRUE(audio_rtp_sender_->SetParameters(params).ok());
> 
>   // Read back the parameters and verify they have been changed.
>   params = audio_rtp_sender_->GetParameters();
>-  EXPECT_EQ(1, params.encodings.size());
>+  EXPECT_EQ(1U, params.encodings.size());
>   EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps);
> 
>   // Verify that the audio channel received the new parameters.
>   params = voice_media_channel_->GetRtpSendParameters(kAudioSsrc);
>-  EXPECT_EQ(1, params.encodings.size());
>+  EXPECT_EQ(1U, params.encodings.size());
>   EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps);
> 
>   // Verify that the global bitrate limit has not been changed.
>@@ -622,19 +770,19 @@ TEST_F(RtpSenderReceiverTest, SetAudioBitratePriority) {
>   CreateAudioRtpSender();
> 
>   webrtc::RtpParameters params = audio_rtp_sender_->GetParameters();
>-  EXPECT_EQ(1, params.encodings.size());
>+  EXPECT_EQ(1U, params.encodings.size());
>   EXPECT_EQ(webrtc::kDefaultBitratePriority,
>             params.encodings[0].bitrate_priority);
>   double new_bitrate_priority = 2.0;
>   params.encodings[0].bitrate_priority = new_bitrate_priority;
>-  EXPECT_TRUE(audio_rtp_sender_->SetParameters(params));
>+  EXPECT_TRUE(audio_rtp_sender_->SetParameters(params).ok());
> 
>   params = audio_rtp_sender_->GetParameters();
>-  EXPECT_EQ(1, params.encodings.size());
>+  EXPECT_EQ(1U, params.encodings.size());
>   EXPECT_EQ(new_bitrate_priority, params.encodings[0].bitrate_priority);
> 
>   params = voice_media_channel_->GetRtpSendParameters(kAudioSsrc);
>-  EXPECT_EQ(1, params.encodings.size());
>+  EXPECT_EQ(1U, params.encodings.size());
>   EXPECT_EQ(new_bitrate_priority, params.encodings[0].bitrate_priority);
> 
>   DestroyAudioRtpSender();
>@@ -645,29 +793,186 @@ TEST_F(RtpSenderReceiverTest, VideoSenderCanSetParameters) {
> 
>   RtpParameters params = video_rtp_sender_->GetParameters();
>   EXPECT_EQ(1u, params.encodings.size());
>-  EXPECT_TRUE(video_rtp_sender_->SetParameters(params));
>+  EXPECT_TRUE(video_rtp_sender_->SetParameters(params).ok());
>+
>+  DestroyVideoRtpSender();
>+}
>+
>+TEST_F(RtpSenderReceiverTest,
>+       VideoSenderMustCallGetParametersBeforeSetParameters) {
>+  CreateVideoRtpSender();
>+
>+  RtpParameters params;
>+  RTCError result = video_rtp_sender_->SetParameters(params);
>+  EXPECT_EQ(RTCErrorType::INVALID_STATE, result.type());
> 
>   DestroyVideoRtpSender();
> }
> 
>-TEST_F(RtpSenderReceiverTest, SetVideoMaxSendBitrate) {
>+TEST_F(RtpSenderReceiverTest,
>+       VideoSenderSetParametersInvalidatesTransactionId) {
>+  CreateVideoRtpSender();
>+
>+  RtpParameters params = video_rtp_sender_->GetParameters();
>+  EXPECT_EQ(1u, params.encodings.size());
>+  EXPECT_TRUE(video_rtp_sender_->SetParameters(params).ok());
>+  RTCError result = video_rtp_sender_->SetParameters(params);
>+  EXPECT_EQ(RTCErrorType::INVALID_STATE, result.type());
>+
>+  DestroyVideoRtpSender();
>+}
>+
>+TEST_F(RtpSenderReceiverTest, VideoSenderDetectTransactionIdModification) {
>+  CreateVideoRtpSender();
>+
>+  RtpParameters params = video_rtp_sender_->GetParameters();
>+  params.transaction_id = "";
>+  RTCError result = video_rtp_sender_->SetParameters(params);
>+  EXPECT_EQ(RTCErrorType::INVALID_MODIFICATION, result.type());
>+
>+  DestroyVideoRtpSender();
>+}
>+
>+TEST_F(RtpSenderReceiverTest, VideoSenderCheckTransactionIdRefresh) {
>+  CreateVideoRtpSender();
>+
>+  RtpParameters params = video_rtp_sender_->GetParameters();
>+  EXPECT_NE(params.transaction_id.size(), 0U);
>+  auto saved_transaction_id = params.transaction_id;
>+  params = video_rtp_sender_->GetParameters();
>+  EXPECT_NE(saved_transaction_id, params.transaction_id);
>+
>+  DestroyVideoRtpSender();
>+}
>+
>+TEST_F(RtpSenderReceiverTest, VideoSenderSetParametersOldValueFail) {
>+  CreateVideoRtpSender();
>+
>+  RtpParameters params = video_rtp_sender_->GetParameters();
>+  RtpParameters second_params = video_rtp_sender_->GetParameters();
>+
>+  RTCError result = video_rtp_sender_->SetParameters(params);
>+  EXPECT_EQ(RTCErrorType::INVALID_MODIFICATION, result.type());
>+
>+  DestroyVideoRtpSender();
>+}
>+
>+TEST_F(RtpSenderReceiverTest, VideoSenderCantSetUnimplementedRtpParameters) {
>+  CreateVideoRtpSender();
>+  RtpParameters params = video_rtp_sender_->GetParameters();
>+  EXPECT_EQ(1u, params.encodings.size());
>+
>+  // Unimplemented RtpParameters: mid
>+  params.mid = "dummy_mid";
>+  EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
>+            video_rtp_sender_->SetParameters(params).type());
>+  params = video_rtp_sender_->GetParameters();
>+
>+  DestroyVideoRtpSender();
>+}
>+
>+TEST_F(RtpSenderReceiverTest,
>+       VideoSenderCantSetUnimplementedEncodingParameters) {
>+  CreateVideoRtpSender();
>+  RtpParameters params = video_rtp_sender_->GetParameters();
>+  EXPECT_EQ(1u, params.encodings.size());
>+
>+  // Unimplemented RtpParameters: codec_payload_type, fec, rtx, dtx, ptime,
>+  // max_framerate, scale_resolution_down_by, scale_framerate_down_by, rid,
>+  // dependency_rids.
>+  params.encodings[0].codec_payload_type = 1;
>+  EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
>+            video_rtp_sender_->SetParameters(params).type());
>+  params = video_rtp_sender_->GetParameters();
>+
>+  params.encodings[0].fec = RtpFecParameters();
>+  EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
>+            video_rtp_sender_->SetParameters(params).type());
>+  params = video_rtp_sender_->GetParameters();
>+
>+  params.encodings[0].rtx = RtpRtxParameters();
>+  EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
>+            video_rtp_sender_->SetParameters(params).type());
>+  params = video_rtp_sender_->GetParameters();
>+
>+  params.encodings[0].dtx = DtxStatus::ENABLED;
>+  EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
>+            video_rtp_sender_->SetParameters(params).type());
>+  params = video_rtp_sender_->GetParameters();
>+
>+  params.encodings[0].ptime = 1;
>+  EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
>+            video_rtp_sender_->SetParameters(params).type());
>+  params = video_rtp_sender_->GetParameters();
>+
>+  params.encodings[0].max_framerate = 1;
>+  EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
>+            video_rtp_sender_->SetParameters(params).type());
>+  params = video_rtp_sender_->GetParameters();
>+
>+  params.encodings[0].scale_resolution_down_by = 2.0;
>+  EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
>+            video_rtp_sender_->SetParameters(params).type());
>+  params = video_rtp_sender_->GetParameters();
>+
>+  params.encodings[0].rid = "dummy_rid";
>+  EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
>+            video_rtp_sender_->SetParameters(params).type());
>+  params = video_rtp_sender_->GetParameters();
>+
>+  params.encodings[0].dependency_rids.push_back("dummy_rid");
>+  EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
>+            video_rtp_sender_->SetParameters(params).type());
>+
>+  DestroyVideoRtpSender();
>+}
>+
>+// A video sender can have multiple simulcast layers, in which case it will
>+// contain multiple RtpEncodingParameters. This tests that if this is the case
>+// (simulcast), then we can't set the bitrate_priority, or max_bitrate_bps
>+// for any encodings besides at index 0, because these are both implemented
>+// "per-sender."
>+TEST_F(RtpSenderReceiverTest, VideoSenderCantSetPerSenderEncodingParameters) {
>+  // Add a simulcast specific send stream that contains 2 encoding parameters.
>+  std::vector<uint32_t> ssrcs({1, 2});
>+  cricket::StreamParams stream_params =
>+      cricket::CreateSimStreamParams("cname", ssrcs);
>+  video_media_channel_->AddSendStream(stream_params);
>+  uint32_t primary_ssrc = stream_params.first_ssrc();
>+  CreateVideoRtpSender(primary_ssrc);
>+  RtpParameters params = video_rtp_sender_->GetParameters();
>+  EXPECT_EQ(ssrcs.size(), params.encodings.size());
>+
>+  params.encodings[1].bitrate_priority = 2.0;
>+  EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
>+            video_rtp_sender_->SetParameters(params).type());
>+  params = video_rtp_sender_->GetParameters();
>+
>+  DestroyVideoRtpSender();
>+}
>+
>+TEST_F(RtpSenderReceiverTest, SetVideoMinMaxSendBitrate) {
>   CreateVideoRtpSender();
> 
>   EXPECT_EQ(-1, video_media_channel_->max_bps());
>   webrtc::RtpParameters params = video_rtp_sender_->GetParameters();
>-  EXPECT_EQ(1, params.encodings.size());
>+  EXPECT_EQ(1U, params.encodings.size());
>+  EXPECT_FALSE(params.encodings[0].min_bitrate_bps);
>   EXPECT_FALSE(params.encodings[0].max_bitrate_bps);
>+  params.encodings[0].min_bitrate_bps = 100;
>   params.encodings[0].max_bitrate_bps = 1000;
>-  EXPECT_TRUE(video_rtp_sender_->SetParameters(params));
>+  EXPECT_TRUE(video_rtp_sender_->SetParameters(params).ok());
> 
>   // Read back the parameters and verify they have been changed.
>   params = video_rtp_sender_->GetParameters();
>-  EXPECT_EQ(1, params.encodings.size());
>+  EXPECT_EQ(1U, params.encodings.size());
>+  EXPECT_EQ(100, params.encodings[0].min_bitrate_bps);
>   EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps);
> 
>   // Verify that the video channel received the new parameters.
>   params = video_media_channel_->GetRtpSendParameters(kVideoSsrc);
>-  EXPECT_EQ(1, params.encodings.size());
>+  EXPECT_EQ(1U, params.encodings.size());
>+  EXPECT_EQ(100, params.encodings[0].min_bitrate_bps);
>   EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps);
> 
>   // Verify that the global bitrate limit has not been changed.
>@@ -676,23 +981,51 @@ TEST_F(RtpSenderReceiverTest, SetVideoMaxSendBitrate) {
>   DestroyVideoRtpSender();
> }
> 
>+TEST_F(RtpSenderReceiverTest, SetVideoMinMaxSendBitrateSimulcast) {
>+  // Add a simulcast specific send stream that contains 2 encoding parameters.
>+  std::vector<uint32_t> ssrcs({1, 2});
>+  cricket::StreamParams stream_params =
>+      cricket::CreateSimStreamParams("cname", ssrcs);
>+  video_media_channel_->AddSendStream(stream_params);
>+  uint32_t primary_ssrc = stream_params.first_ssrc();
>+  CreateVideoRtpSender(primary_ssrc);
>+
>+  RtpParameters params = video_rtp_sender_->GetParameters();
>+  EXPECT_EQ(ssrcs.size(), params.encodings.size());
>+  params.encodings[0].min_bitrate_bps = 100;
>+  params.encodings[0].max_bitrate_bps = 1000;
>+  params.encodings[1].min_bitrate_bps = 200;
>+  params.encodings[1].max_bitrate_bps = 2000;
>+  EXPECT_TRUE(video_rtp_sender_->SetParameters(params).ok());
>+
>+  // Verify that the video channel received the new parameters.
>+  params = video_media_channel_->GetRtpSendParameters(primary_ssrc);
>+  EXPECT_EQ(ssrcs.size(), params.encodings.size());
>+  EXPECT_EQ(100, params.encodings[0].min_bitrate_bps);
>+  EXPECT_EQ(1000, params.encodings[0].max_bitrate_bps);
>+  EXPECT_EQ(200, params.encodings[1].min_bitrate_bps);
>+  EXPECT_EQ(2000, params.encodings[1].max_bitrate_bps);
>+
>+  DestroyVideoRtpSender();
>+}
>+
> TEST_F(RtpSenderReceiverTest, SetVideoBitratePriority) {
>   CreateVideoRtpSender();
> 
>   webrtc::RtpParameters params = video_rtp_sender_->GetParameters();
>-  EXPECT_EQ(1, params.encodings.size());
>+  EXPECT_EQ(1U, params.encodings.size());
>   EXPECT_EQ(webrtc::kDefaultBitratePriority,
>             params.encodings[0].bitrate_priority);
>   double new_bitrate_priority = 2.0;
>   params.encodings[0].bitrate_priority = new_bitrate_priority;
>-  EXPECT_TRUE(video_rtp_sender_->SetParameters(params));
>+  EXPECT_TRUE(video_rtp_sender_->SetParameters(params).ok());
> 
>   params = video_rtp_sender_->GetParameters();
>-  EXPECT_EQ(1, params.encodings.size());
>+  EXPECT_EQ(1U, params.encodings.size());
>   EXPECT_EQ(new_bitrate_priority, params.encodings[0].bitrate_priority);
> 
>   params = video_media_channel_->GetRtpSendParameters(kVideoSsrc);
>-  EXPECT_EQ(1, params.encodings.size());
>+  EXPECT_EQ(1U, params.encodings.size());
>   EXPECT_EQ(new_bitrate_priority, params.encodings[0].bitrate_priority);
> 
>   DestroyVideoRtpSender();
>@@ -740,6 +1073,9 @@ TEST_F(RtpSenderReceiverTest, PropagatesVideoTrackContentHint) {
>   // Setting fluid should remain in non-screencast mode (its default).
>   video_track_->set_content_hint(VideoTrackInterface::ContentHint::kFluid);
>   EXPECT_EQ(false, video_media_channel_->options().is_screencast);
>+  // Setting text should have the same effect as Detailed
>+  video_track_->set_content_hint(VideoTrackInterface::ContentHint::kText);
>+  EXPECT_EQ(true, video_media_channel_->options().is_screencast);
> 
>   DestroyVideoRtpSender();
> }
>@@ -766,6 +1102,9 @@ TEST_F(RtpSenderReceiverTest,
>   // Setting detailed should still remain in screencast mode (its default).
>   video_track_->set_content_hint(VideoTrackInterface::ContentHint::kDetailed);
>   EXPECT_EQ(true, video_media_channel_->options().is_screencast);
>+  // Setting text should have the same effect as Detailed
>+  video_track_->set_content_hint(VideoTrackInterface::ContentHint::kText);
>+  EXPECT_EQ(true, video_media_channel_->options().is_screencast);
> 
>   DestroyVideoRtpSender();
> }
>@@ -778,14 +1117,14 @@ TEST_F(RtpSenderReceiverTest,
>   // Setting detailed overrides the default non-screencast mode. This should be
>   // applied even if the track is set on construction.
>   video_track_->set_content_hint(VideoTrackInterface::ContentHint::kDetailed);
>-  video_rtp_sender_ = new VideoRtpSender(worker_thread_,
>-                                         local_stream_->GetVideoTracks()[0],
>-                                         {local_stream_->label()});
>-  video_rtp_sender_->SetMediaChannel(video_media_channel_);
>+  video_rtp_sender_ = new VideoRtpSender(worker_thread_, video_track_->id());
>+  ASSERT_TRUE(video_rtp_sender_->SetTrack(video_track_));
>+  video_rtp_sender_->set_stream_ids({local_stream_->id()});
>+  video_rtp_sender_->SetVideoMediaChannel(video_media_channel_);
>   video_track_->set_enabled(true);
> 
>   // Sender is not ready to send (no SSRC) so no option should have been set.
>-  EXPECT_EQ(rtc::nullopt, video_media_channel_->options().is_screencast);
>+  EXPECT_EQ(absl::nullopt, video_media_channel_->options().is_screencast);
> 
>   // Verify that the content hint is accounted for when video_rtp_sender_ does
>   // get enabled.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransceiver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransceiver.cc
>index ecb8ec879fb..0fe8ea6f50a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransceiver.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransceiver.cc
>@@ -16,10 +16,6 @@
> 
> namespace webrtc {
> 
>-std::ostream& operator<<(std::ostream& os, RtpTransceiverDirection direction) {
>-  return os << RtpTransceiverDirectionToString(direction);
>-}
>-
> RtpTransceiver::RtpTransceiver(cricket::MediaType media_type)
>     : unified_plan_(false), media_type_(media_type) {
>   RTC_DCHECK(media_type == cricket::MEDIA_TYPE_AUDIO ||
>@@ -61,14 +57,12 @@ void RtpTransceiver::SetChannel(cricket::BaseChannel* channel) {
>   for (auto sender : senders_) {
>     if (media_type() == cricket::MEDIA_TYPE_AUDIO) {
>       auto* voice_channel = static_cast<cricket::VoiceChannel*>(channel);
>-      static_cast<AudioRtpSender*>(sender->internal())
>-          ->SetMediaChannel(voice_channel ? voice_channel->media_channel()
>-                                          : nullptr);
>+      sender->internal()->SetVoiceMediaChannel(
>+          voice_channel ? voice_channel->media_channel() : nullptr);
>     } else {
>       auto* video_channel = static_cast<cricket::VideoChannel*>(channel);
>-      static_cast<VideoRtpSender*>(sender->internal())
>-          ->SetMediaChannel(video_channel ? video_channel->media_channel()
>-                                          : nullptr);
>+      sender->internal()->SetVideoMediaChannel(
>+          video_channel ? video_channel->media_channel() : nullptr);
>     }
>   }
> 
>@@ -78,14 +72,12 @@ void RtpTransceiver::SetChannel(cricket::BaseChannel* channel) {
>     }
>     if (media_type() == cricket::MEDIA_TYPE_AUDIO) {
>       auto* voice_channel = static_cast<cricket::VoiceChannel*>(channel);
>-      static_cast<AudioRtpReceiver*>(receiver->internal())
>-          ->SetMediaChannel(voice_channel ? voice_channel->media_channel()
>-                                          : nullptr);
>+      receiver->internal()->SetVoiceMediaChannel(
>+          voice_channel ? voice_channel->media_channel() : nullptr);
>     } else {
>       auto* video_channel = static_cast<cricket::VideoChannel*>(channel);
>-      static_cast<VideoRtpReceiver*>(receiver->internal())
>-          ->SetMediaChannel(video_channel ? video_channel->media_channel()
>-                                          : nullptr);
>+      receiver->internal()->SetVideoMediaChannel(
>+          video_channel ? video_channel->media_channel() : nullptr);
>     }
>   }
> }
>@@ -94,7 +86,7 @@ void RtpTransceiver::AddSender(
>     rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> sender) {
>   RTC_DCHECK(!unified_plan_);
>   RTC_DCHECK(sender);
>-  RTC_DCHECK_EQ(media_type(), sender->internal()->media_type());
>+  RTC_DCHECK_EQ(media_type(), sender->media_type());
>   RTC_DCHECK(std::find(senders_.begin(), senders_.end(), sender) ==
>              senders_.end());
>   senders_.push_back(sender);
>@@ -119,7 +111,7 @@ void RtpTransceiver::AddReceiver(
>         receiver) {
>   RTC_DCHECK(!unified_plan_);
>   RTC_DCHECK(receiver);
>-  RTC_DCHECK_EQ(media_type(), receiver->internal()->media_type());
>+  RTC_DCHECK_EQ(media_type(), receiver->media_type());
>   RTC_DCHECK(std::find(receivers_.begin(), receivers_.end(), receiver) ==
>              receivers_.end());
>   receivers_.push_back(receiver);
>@@ -152,7 +144,11 @@ rtc::scoped_refptr<RtpReceiverInternal> RtpTransceiver::receiver_internal()
>   return receivers_[0]->internal();
> }
> 
>-rtc::Optional<std::string> RtpTransceiver::mid() const {
>+cricket::MediaType RtpTransceiver::media_type() const {
>+  return media_type_;
>+}
>+
>+absl::optional<std::string> RtpTransceiver::mid() const {
>   return mid_;
> }
> 
>@@ -175,12 +171,23 @@ rtc::scoped_refptr<RtpReceiverInterface> RtpTransceiver::receiver() const {
> }
> 
> void RtpTransceiver::set_current_direction(RtpTransceiverDirection direction) {
>+  RTC_LOG(LS_INFO) << "Changing transceiver (MID=" << mid_.value_or("<not set>")
>+                   << ") current direction from "
>+                   << (current_direction_ ? RtpTransceiverDirectionToString(
>+                                                *current_direction_)
>+                                          : "<not set>")
>+                   << " to " << RtpTransceiverDirectionToString(direction)
>+                   << ".";
>   current_direction_ = direction;
>   if (RtpTransceiverDirectionHasSend(*current_direction_)) {
>     has_ever_been_used_to_send_ = true;
>   }
> }
> 
>+void RtpTransceiver::set_fired_direction(RtpTransceiverDirection direction) {
>+  fired_direction_ = direction;
>+}
>+
> bool RtpTransceiver::stopped() const {
>   return stopped_;
> }
>@@ -190,15 +197,26 @@ RtpTransceiverDirection RtpTransceiver::direction() const {
> }
> 
> void RtpTransceiver::SetDirection(RtpTransceiverDirection new_direction) {
>-  // TODO(steveanton): This should fire OnNegotiationNeeded.
>-  set_direction(new_direction);
>+  if (stopped()) {
>+    return;
>+  }
>+  if (new_direction == direction_) {
>+    return;
>+  }
>+  direction_ = new_direction;
>+  SignalNegotiationNeeded();
> }
> 
>-rtc::Optional<RtpTransceiverDirection> RtpTransceiver::current_direction()
>+absl::optional<RtpTransceiverDirection> RtpTransceiver::current_direction()
>     const {
>   return current_direction_;
> }
> 
>+absl::optional<RtpTransceiverDirection> RtpTransceiver::fired_direction()
>+    const {
>+  return fired_direction_;
>+}
>+
> void RtpTransceiver::Stop() {
>   for (auto sender : senders_) {
>     sender->internal()->Stop();
>@@ -207,6 +225,7 @@ void RtpTransceiver::Stop() {
>     receiver->internal()->Stop();
>   }
>   stopped_ = true;
>+  current_direction_ = absl::nullopt;
> }
> 
> void RtpTransceiver::SetCodecPreferences(
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransceiver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransceiver.h
>index 40ab2f021ef..3e0d4337826 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransceiver.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransceiver.h
>@@ -68,8 +68,6 @@ class RtpTransceiver final
>           receiver);
>   ~RtpTransceiver() override;
> 
>-  cricket::MediaType media_type() const { return media_type_; }
>-
>   // Returns the Voice/VideoChannel set for this transceiver. May be null if
>   // the transceiver is not in the currently set local/remote description.
>   cricket::BaseChannel* channel() const { return channel_; }
>@@ -121,15 +119,15 @@ class RtpTransceiver final
>   // when setting a local offer we need a way to remember which transceiver was
>   // used to create which media section in the offer. Storing the mline index
>   // in CreateOffer is specified in JSEP to allow us to do that.
>-  rtc::Optional<size_t> mline_index() const { return mline_index_; }
>-  void set_mline_index(rtc::Optional<size_t> mline_index) {
>+  absl::optional<size_t> mline_index() const { return mline_index_; }
>+  void set_mline_index(absl::optional<size_t> mline_index) {
>     mline_index_ = mline_index;
>   }
> 
>   // Sets the MID for this transceiver. If the MID is not null, then the
>   // transceiver is considered "associated" with the media section that has the
>   // same MID.
>-  void set_mid(const rtc::Optional<std::string>& mid) { mid_ = mid; }
>+  void set_mid(const absl::optional<std::string>& mid) { mid_ = mid; }
> 
>   // Sets the intended direction for this transceiver. Intended to be used
>   // internally over SetDirection since this does not trigger a negotiation
>@@ -143,6 +141,11 @@ class RtpTransceiver final
>   // transceiver has been set.
>   void set_current_direction(RtpTransceiverDirection direction);
> 
>+  // Sets the fired direction for this transceiver. The fired direction is null
>+  // until SetRemoteDescription is called or an answer is set (either local or
>+  // remote).
>+  void set_fired_direction(RtpTransceiverDirection direction);
>+
>   // According to JSEP rules for SetRemoteDescription, RtpTransceivers can be
>   // reused only if they were added by AddTrack.
>   void set_created_by_addtrack(bool created_by_addtrack) {
>@@ -156,14 +159,20 @@ class RtpTransceiver final
>     return has_ever_been_used_to_send_;
>   }
> 
>+  // Fired when the RtpTransceiver state changes such that negotiation is now
>+  // needed (e.g., in response to a direction change).
>+  sigslot::signal0<> SignalNegotiationNeeded;
>+
>   // RtpTransceiverInterface implementation.
>-  rtc::Optional<std::string> mid() const override;
>+  cricket::MediaType media_type() const override;
>+  absl::optional<std::string> mid() const override;
>   rtc::scoped_refptr<RtpSenderInterface> sender() const override;
>   rtc::scoped_refptr<RtpReceiverInterface> receiver() const override;
>   bool stopped() const override;
>   RtpTransceiverDirection direction() const override;
>   void SetDirection(RtpTransceiverDirection new_direction) override;
>-  rtc::Optional<RtpTransceiverDirection> current_direction() const override;
>+  absl::optional<RtpTransceiverDirection> current_direction() const override;
>+  absl::optional<RtpTransceiverDirection> fired_direction() const override;
>   void Stop() override;
>   void SetCodecPreferences(rtc::ArrayView<RtpCodecCapability> codecs) override;
> 
>@@ -180,9 +189,10 @@ class RtpTransceiver final
> 
>   bool stopped_ = false;
>   RtpTransceiverDirection direction_ = RtpTransceiverDirection::kInactive;
>-  rtc::Optional<RtpTransceiverDirection> current_direction_;
>-  rtc::Optional<std::string> mid_;
>-  rtc::Optional<size_t> mline_index_;
>+  absl::optional<RtpTransceiverDirection> current_direction_;
>+  absl::optional<RtpTransceiverDirection> fired_direction_;
>+  absl::optional<std::string> mid_;
>+  absl::optional<size_t> mline_index_;
>   bool created_by_addtrack_ = false;
>   bool has_ever_been_used_to_send_ = false;
> 
>@@ -191,13 +201,15 @@ class RtpTransceiver final
> 
> BEGIN_SIGNALING_PROXY_MAP(RtpTransceiver)
> PROXY_SIGNALING_THREAD_DESTRUCTOR()
>-PROXY_CONSTMETHOD0(rtc::Optional<std::string>, mid);
>+PROXY_CONSTMETHOD0(cricket::MediaType, media_type);
>+PROXY_CONSTMETHOD0(absl::optional<std::string>, mid);
> PROXY_CONSTMETHOD0(rtc::scoped_refptr<RtpSenderInterface>, sender);
> PROXY_CONSTMETHOD0(rtc::scoped_refptr<RtpReceiverInterface>, receiver);
> PROXY_CONSTMETHOD0(bool, stopped);
> PROXY_CONSTMETHOD0(RtpTransceiverDirection, direction);
> PROXY_METHOD1(void, SetDirection, RtpTransceiverDirection);
>-PROXY_CONSTMETHOD0(rtc::Optional<RtpTransceiverDirection>, current_direction);
>+PROXY_CONSTMETHOD0(absl::optional<RtpTransceiverDirection>, current_direction);
>+PROXY_CONSTMETHOD0(absl::optional<RtpTransceiverDirection>, fired_direction);
> PROXY_METHOD0(void, Stop);
> PROXY_METHOD1(void, SetCodecPreferences, rtc::ArrayView<RtpCodecCapability>);
> END_PROXY_MAP();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransport.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransport.cc
>index 26f7e3e4c9b..9e994e9b980 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransport.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransport.cc
>@@ -10,7 +10,10 @@
> 
> #include "pc/rtptransport.h"
> 
>+#include <utility>
>+
> #include "media/base/rtputils.h"
>+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
> #include "p2p/base/p2pconstants.h"
> #include "p2p/base/packettransportinterface.h"
> #include "rtc_base/checks.h"
>@@ -36,7 +39,7 @@ void RtpTransport::SetRtpPacketTransport(
>     rtp_packet_transport_->SignalWritableState.disconnect(this);
>     rtp_packet_transport_->SignalSentPacket.disconnect(this);
>     // Reset the network route of the old transport.
>-    SignalNetworkRouteChanged(rtc::Optional<rtc::NetworkRoute>());
>+    SignalNetworkRouteChanged(absl::optional<rtc::NetworkRoute>());
>   }
>   if (new_packet_transport) {
>     new_packet_transport->SignalReadyToSend.connect(
>@@ -44,7 +47,7 @@ void RtpTransport::SetRtpPacketTransport(
>     new_packet_transport->SignalReadPacket.connect(this,
>                                                    &RtpTransport::OnReadPacket);
>     new_packet_transport->SignalNetworkRouteChanged.connect(
>-        this, &RtpTransport::OnNetworkRouteChange);
>+        this, &RtpTransport::OnNetworkRouteChanged);
>     new_packet_transport->SignalWritableState.connect(
>         this, &RtpTransport::OnWritableState);
>     new_packet_transport->SignalSentPacket.connect(this,
>@@ -72,7 +75,7 @@ void RtpTransport::SetRtcpPacketTransport(
>     rtcp_packet_transport_->SignalWritableState.disconnect(this);
>     rtcp_packet_transport_->SignalSentPacket.disconnect(this);
>     // Reset the network route of the old transport.
>-    SignalNetworkRouteChanged(rtc::Optional<rtc::NetworkRoute>());
>+    SignalNetworkRouteChanged(absl::optional<rtc::NetworkRoute>());
>   }
>   if (new_packet_transport) {
>     new_packet_transport->SignalReadyToSend.connect(
>@@ -80,7 +83,7 @@ void RtpTransport::SetRtcpPacketTransport(
>     new_packet_transport->SignalReadPacket.connect(this,
>                                                    &RtpTransport::OnReadPacket);
>     new_packet_transport->SignalNetworkRouteChanged.connect(
>-        this, &RtpTransport::OnNetworkRouteChange);
>+        this, &RtpTransport::OnNetworkRouteChanged);
>     new_packet_transport->SignalWritableState.connect(
>         this, &RtpTransport::OnWritableState);
>     new_packet_transport->SignalSentPacket.connect(this,
>@@ -134,24 +137,27 @@ bool RtpTransport::SendPacket(bool rtcp,
>   return true;
> }
> 
>-bool RtpTransport::HandlesPacket(const uint8_t* data, size_t len) {
>-  return bundle_filter_.DemuxPacket(data, len);
>-}
>-
>-bool RtpTransport::HandlesPayloadType(int payload_type) const {
>-  return bundle_filter_.FindPayloadType(payload_type);
>-}
>-
>-void RtpTransport::AddHandledPayloadType(int payload_type) {
>-  bundle_filter_.AddPayloadType(payload_type);
>+void RtpTransport::UpdateRtpHeaderExtensionMap(
>+    const cricket::RtpHeaderExtensions& header_extensions) {
>+  header_extension_map_ = RtpHeaderExtensionMap(header_extensions);
> }
> 
>-PacketTransportInterface* RtpTransport::GetRtpPacketTransport() const {
>-  return rtp_packet_transport_;
>+bool RtpTransport::RegisterRtpDemuxerSink(const RtpDemuxerCriteria& criteria,
>+                                          RtpPacketSinkInterface* sink) {
>+  rtp_demuxer_.RemoveSink(sink);
>+  if (!rtp_demuxer_.AddSink(criteria, sink)) {
>+    RTC_LOG(LS_ERROR) << "Failed to register the sink for RTP demuxer.";
>+    return false;
>+  }
>+  return true;
> }
> 
>-PacketTransportInterface* RtpTransport::GetRtcpPacketTransport() const {
>-  return rtcp_packet_transport_;
>+bool RtpTransport::UnregisterRtpDemuxerSink(RtpPacketSinkInterface* sink) {
>+  if (!rtp_demuxer_.RemoveSink(sink)) {
>+    RTC_LOG(LS_ERROR) << "Failed to unregister the sink for RTP demuxer.";
>+    return false;
>+  }
>+  return true;
> }
> 
> RTCError RtpTransport::SetParameters(const RtpTransportParameters& parameters) {
>@@ -180,11 +186,26 @@ RtpTransportParameters RtpTransport::GetParameters() const {
>   return parameters_;
> }
> 
>+void RtpTransport::DemuxPacket(rtc::CopyOnWriteBuffer* packet,
>+                               const rtc::PacketTime& time) {
>+  webrtc::RtpPacketReceived parsed_packet(&header_extension_map_);
>+  if (!parsed_packet.Parse(std::move(*packet))) {
>+    RTC_LOG(LS_ERROR)
>+        << "Failed to parse the incoming RTP packet before demuxing. Drop it.";
>+    return;
>+  }
>+
>+  if (time.timestamp != -1) {
>+    parsed_packet.set_arrival_time_ms((time.timestamp + 500) / 1000);
>+  }
>+  rtp_demuxer_.OnRtpPacket(parsed_packet);
>+}
>+
> RtpTransportAdapter* RtpTransport::GetInternal() {
>   return nullptr;
> }
> 
>-bool RtpTransport::IsRtpTransportWritable() {
>+bool RtpTransport::IsTransportWritable() {
>   auto rtcp_packet_transport =
>       rtcp_mux_enabled_ ? nullptr : rtcp_packet_transport_;
>   return rtp_packet_transport_ && rtp_packet_transport_->writable() &&
>@@ -195,8 +216,8 @@ void RtpTransport::OnReadyToSend(rtc::PacketTransportInternal* transport) {
>   SetReadyToSend(transport == rtcp_packet_transport_, true);
> }
> 
>-void RtpTransport::OnNetworkRouteChange(
>-    rtc::Optional<rtc::NetworkRoute> network_route) {
>+void RtpTransport::OnNetworkRouteChanged(
>+    absl::optional<rtc::NetworkRoute> network_route) {
>   SignalNetworkRouteChanged(network_route);
> }
> 
>@@ -204,7 +225,7 @@ void RtpTransport::OnWritableState(
>     rtc::PacketTransportInternal* packet_transport) {
>   RTC_DCHECK(packet_transport == rtp_packet_transport_ ||
>              packet_transport == rtcp_packet_transport_);
>-  SignalWritableState(IsRtpTransportWritable());
>+  SignalWritableState(IsTransportWritable());
> }
> 
> void RtpTransport::OnSentPacket(rtc::PacketTransportInternal* packet_transport,
>@@ -214,33 +235,14 @@ void RtpTransport::OnSentPacket(rtc::PacketTransportInternal* packet_transport,
>   SignalSentPacket(sent_packet);
> }
> 
>-void RtpTransport::SetReadyToSend(bool rtcp, bool ready) {
>-  if (rtcp) {
>-    rtcp_ready_to_send_ = ready;
>-  } else {
>-    rtp_ready_to_send_ = ready;
>-  }
>-
>-  MaybeSignalReadyToSend();
>-}
>-
>-void RtpTransport::MaybeSignalReadyToSend() {
>-  bool ready_to_send =
>-      rtp_ready_to_send_ && (rtcp_ready_to_send_ || rtcp_mux_enabled_);
>-  if (ready_to_send != ready_to_send_) {
>-    ready_to_send_ = ready_to_send;
>-    SignalReadyToSend(ready_to_send);
>-  }
>+void RtpTransport::OnRtpPacketReceived(rtc::CopyOnWriteBuffer* packet,
>+                                       const rtc::PacketTime& packet_time) {
>+  DemuxPacket(packet, packet_time);
> }
> 
>-// Check the RTP payload type.  If 63 < payload type < 96, it's RTCP.
>-// For additional details, see http://tools.ietf.org/html/rfc5761.
>-bool IsRtcp(const char* data, int len) {
>-  if (len < 2) {
>-    return false;
>-  }
>-  char pt = data[1] & 0x7F;
>-  return (63 < pt) && (pt < 96);
>+void RtpTransport::OnRtcpPacketReceived(rtc::CopyOnWriteBuffer* packet,
>+                                        const rtc::PacketTime& packet_time) {
>+  SignalRtcpPacketReceived(packet, packet_time);
> }
> 
> void RtpTransport::OnReadPacket(rtc::PacketTransportInternal* transport,
>@@ -252,32 +254,47 @@ void RtpTransport::OnReadPacket(rtc::PacketTransportInternal* transport,
> 
>   // When using RTCP multiplexing we might get RTCP packets on the RTP
>   // transport. We check the RTP payload type to determine if it is RTCP.
>-  bool rtcp = transport == rtcp_packet_transport() ||
>-              IsRtcp(data, static_cast<int>(len));
>-  rtc::CopyOnWriteBuffer packet(data, len);
>+  bool rtcp =
>+      transport == rtcp_packet_transport() || cricket::IsRtcpPacket(data, len);
> 
>-  if (!WantsPacket(rtcp, &packet)) {
>+  // Filter out the packet that is neither RTP nor RTCP.
>+  if (!rtcp && !cricket::IsRtpPacket(data, len)) {
>     return;
>   }
>-  // This mutates |packet| if it is protected.
>-  SignalPacketReceived(rtcp, &packet, packet_time);
>-}
> 
>-bool RtpTransport::WantsPacket(bool rtcp,
>-                               const rtc::CopyOnWriteBuffer* packet) {
>+  rtc::CopyOnWriteBuffer packet(data, len);
>   // Protect ourselves against crazy data.
>-  if (!packet || !cricket::IsValidRtpRtcpPacketSize(rtcp, packet->size())) {
>+  if (!cricket::IsValidRtpRtcpPacketSize(rtcp, packet.size())) {
>     RTC_LOG(LS_ERROR) << "Dropping incoming "
>                       << cricket::RtpRtcpStringLiteral(rtcp)
>-                      << " packet: wrong size=" << packet->size();
>-    return false;
>+                      << " packet: wrong size=" << packet.size();
>+    return;
>   }
>+
>   if (rtcp) {
>-    // Permit all (seemingly valid) RTCP packets.
>-    return true;
>+    OnRtcpPacketReceived(&packet, packet_time);
>+  } else {
>+    OnRtpPacketReceived(&packet, packet_time);
>+  }
>+}
>+
>+void RtpTransport::SetReadyToSend(bool rtcp, bool ready) {
>+  if (rtcp) {
>+    rtcp_ready_to_send_ = ready;
>+  } else {
>+    rtp_ready_to_send_ = ready;
>+  }
>+
>+  MaybeSignalReadyToSend();
>+}
>+
>+void RtpTransport::MaybeSignalReadyToSend() {
>+  bool ready_to_send =
>+      rtp_ready_to_send_ && (rtcp_ready_to_send_ || rtcp_mux_enabled_);
>+  if (ready_to_send != ready_to_send_) {
>+    ready_to_send_ = ready_to_send;
>+    SignalReadyToSend(ready_to_send);
>   }
>-  // Check whether we handle this payload.
>-  return HandlesPacket(packet->data(), packet->size());
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransport.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransport.h
>index 497a7487b44..793ac901aa5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransport.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransport.h
>@@ -13,15 +13,15 @@
> 
> #include <string>
> 
>-#include "pc/bundlefilter.h"
>+#include "call/rtp_demuxer.h"
>+#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
> #include "pc/rtptransportinternal.h"
>-#include "rtc_base/sigslot.h"
>+#include "rtc_base/third_party/sigslot/sigslot.h"
> 
> namespace rtc {
> 
> class CopyOnWriteBuffer;
> struct PacketOptions;
>-struct PacketTime;
> class PacketTransportInternal;
> 
> }  // namespace rtc
>@@ -49,13 +49,19 @@ class RtpTransport : public RtpTransportInternal {
>   }
>   void SetRtcpPacketTransport(rtc::PacketTransportInternal* rtcp) override;
> 
>-  PacketTransportInterface* GetRtpPacketTransport() const override;
>-  PacketTransportInterface* GetRtcpPacketTransport() const override;
>+  PacketTransportInterface* GetRtpPacketTransport() const override {
>+    return rtp_packet_transport_;
>+  }
>+  PacketTransportInterface* GetRtcpPacketTransport() const override {
>+    return rtcp_packet_transport_;
>+  }
> 
>   // TODO(zstein): Use these RtcpParameters for configuration elsewhere.
>   RTCError SetParameters(const RtpTransportParameters& parameters) override;
>   RtpTransportParameters GetParameters() const override;
> 
>+  bool IsReadyToSend() const override { return ready_to_send_; }
>+
>   bool IsWritable(bool rtcp) const override;
> 
>   bool SendRtpPacket(rtc::CopyOnWriteBuffer* packet,
>@@ -66,42 +72,67 @@ class RtpTransport : public RtpTransportInternal {
>                       const rtc::PacketOptions& options,
>                       int flags) override;
> 
>-  bool HandlesPayloadType(int payload_type) const override;
>+  bool IsSrtpActive() const override { return false; }
>+
>+  void UpdateRtpHeaderExtensionMap(
>+      const cricket::RtpHeaderExtensions& header_extensions) override;
> 
>-  void AddHandledPayloadType(int payload_type) override;
>+  bool RegisterRtpDemuxerSink(const RtpDemuxerCriteria& criteria,
>+                              RtpPacketSinkInterface* sink) override;
>+
>+  bool UnregisterRtpDemuxerSink(RtpPacketSinkInterface* sink) override;
> 
>  protected:
>   // TODO(zstein): Remove this when we remove RtpTransportAdapter.
>   RtpTransportAdapter* GetInternal() override;
> 
>- private:
>-  bool IsRtpTransportWritable();
>-  bool HandlesPacket(const uint8_t* data, size_t len);
>-
>-  void OnReadyToSend(rtc::PacketTransportInternal* transport);
>-  void OnNetworkRouteChange(rtc::Optional<rtc::NetworkRoute> network_route);
>-  void OnWritableState(rtc::PacketTransportInternal* packet_transport);
>-  void OnSentPacket(rtc::PacketTransportInternal* packet_transport,
>-                    const rtc::SentPacket& sent_packet);
>-
>-  // Updates "ready to send" for an individual channel and fires
>-  // SignalReadyToSend.
>-  void SetReadyToSend(bool rtcp, bool ready);
>-
>-  void MaybeSignalReadyToSend();
>+  // These methods will be used in the subclasses.
>+  void DemuxPacket(rtc::CopyOnWriteBuffer* packet, const rtc::PacketTime& time);
> 
>   bool SendPacket(bool rtcp,
>                   rtc::CopyOnWriteBuffer* packet,
>                   const rtc::PacketOptions& options,
>                   int flags);
> 
>+  // Overridden by SrtpTransport.
>+  virtual void OnNetworkRouteChanged(
>+      absl::optional<rtc::NetworkRoute> network_route);
>+  virtual void OnRtpPacketReceived(rtc::CopyOnWriteBuffer* packet,
>+                                   const rtc::PacketTime& packet_time);
>+  virtual void OnRtcpPacketReceived(rtc::CopyOnWriteBuffer* packet,
>+                                    const rtc::PacketTime& packet_time);
>+  // Overridden by SrtpTransport and DtlsSrtpTransport.
>+  virtual void OnWritableState(rtc::PacketTransportInternal* packet_transport);
>+
>+ private:
>+  void OnReadyToSend(rtc::PacketTransportInternal* transport);
>+  void OnSentPacket(rtc::PacketTransportInternal* packet_transport,
>+                    const rtc::SentPacket& sent_packet);
>   void OnReadPacket(rtc::PacketTransportInternal* transport,
>                     const char* data,
>                     size_t len,
>                     const rtc::PacketTime& packet_time,
>                     int flags);
> 
>-  bool WantsPacket(bool rtcp, const rtc::CopyOnWriteBuffer* packet);
>+  // Updates "ready to send" for an individual channel and fires
>+  // SignalReadyToSend.
>+  void SetReadyToSend(bool rtcp, bool ready);
>+
>+  void MaybeSignalReadyToSend();
>+
>+  bool IsTransportWritable();
>+
>+  // SRTP specific methods.
>+  // TODO(zhihuang): Improve the inheritance model so that the RtpTransport
>+  // doesn't need to implement SRTP specfic methods.
>+  RTCError SetSrtpSendKey(const cricket::CryptoParams& params) override {
>+    RTC_NOTREACHED();
>+    return RTCError::OK();
>+  }
>+  RTCError SetSrtpReceiveKey(const cricket::CryptoParams& params) override {
>+    RTC_NOTREACHED();
>+    return RTCError::OK();
>+  }
> 
>   bool rtcp_mux_enabled_;
> 
>@@ -113,8 +144,10 @@ class RtpTransport : public RtpTransportInternal {
>   bool rtcp_ready_to_send_ = false;
> 
>   RtpTransportParameters parameters_;
>+  RtpDemuxer rtp_demuxer_;
> 
>-  cricket::BundleFilter bundle_filter_;
>+  // Used for identifying the MID for RtpDemuxer.
>+  RtpHeaderExtensionMap header_extension_map_;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransport_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransport_unittest.cc
>index 3876aa39983..7e4eb84cd71 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransport_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransport_unittest.cc
>@@ -60,22 +60,49 @@ TEST(RtpTransportTest, SetRtpTransportKeepAliveNotSupported) {
> class SignalObserver : public sigslot::has_slots<> {
>  public:
>   explicit SignalObserver(RtpTransport* transport) {
>+    transport_ = transport;
>     transport->SignalReadyToSend.connect(this, &SignalObserver::OnReadyToSend);
>     transport->SignalNetworkRouteChanged.connect(
>         this, &SignalObserver::OnNetworkRouteChanged);
>+    if (transport->rtp_packet_transport()) {
>+      transport->rtp_packet_transport()->SignalSentPacket.connect(
>+          this, &SignalObserver::OnSentPacket);
>+    }
>+
>+    if (transport->rtcp_packet_transport()) {
>+      transport->rtcp_packet_transport()->SignalSentPacket.connect(
>+          this, &SignalObserver::OnSentPacket);
>+    }
>   }
> 
>   bool ready() const { return ready_; }
>   void OnReadyToSend(bool ready) { ready_ = ready; }
> 
>-  rtc::Optional<rtc::NetworkRoute> network_route() { return network_route_; }
>-  void OnNetworkRouteChanged(rtc::Optional<rtc::NetworkRoute> network_route) {
>+  absl::optional<rtc::NetworkRoute> network_route() { return network_route_; }
>+  void OnNetworkRouteChanged(absl::optional<rtc::NetworkRoute> network_route) {
>     network_route_ = std::move(network_route);
>   }
> 
>+  void OnSentPacket(rtc::PacketTransportInternal* packet_transport,
>+                    const rtc::SentPacket& sent_packet) {
>+    if (packet_transport == transport_->rtp_packet_transport()) {
>+      rtp_transport_sent_count_++;
>+    } else {
>+      ASSERT_EQ(transport_->rtcp_packet_transport(), packet_transport);
>+      rtcp_transport_sent_count_++;
>+    }
>+  }
>+
>+  int rtp_transport_sent_count() { return rtp_transport_sent_count_; }
>+
>+  int rtcp_transport_sent_count() { return rtcp_transport_sent_count_; }
>+
>  private:
>+  int rtp_transport_sent_count_ = 0;
>+  int rtcp_transport_sent_count_ = 0;
>+  RtpTransport* transport_ = nullptr;
>   bool ready_ = false;
>-  rtc::Optional<rtc::NetworkRoute> network_route_;
>+  absl::optional<rtc::NetworkRoute> network_route_;
> };
> 
> TEST(RtpTransportTest, SettingRtcpAndRtpSignalsReady) {
>@@ -157,7 +184,7 @@ TEST(RtpTransportTest, SetRtpTransportWithNetworkRouteChanged) {
>   network_route.remote_network_id = kRemoteNetId;
>   network_route.last_sent_packet_id = kLastPacketId;
>   network_route.packet_overhead = kTransportOverheadPerPacket;
>-  fake_rtp.SetNetworkRoute(rtc::Optional<rtc::NetworkRoute>(network_route));
>+  fake_rtp.SetNetworkRoute(absl::optional<rtc::NetworkRoute>(network_route));
>   transport.SetRtpPacketTransport(&fake_rtp);
>   ASSERT_TRUE(observer.network_route());
>   EXPECT_EQ(network_route, *(observer.network_route()));
>@@ -184,7 +211,7 @@ TEST(RtpTransportTest, SetRtcpTransportWithNetworkRouteChanged) {
>   network_route.remote_network_id = kRemoteNetId;
>   network_route.last_sent_packet_id = kLastPacketId;
>   network_route.packet_overhead = kTransportOverheadPerPacket;
>-  fake_rtcp.SetNetworkRoute(rtc::Optional<rtc::NetworkRoute>(network_route));
>+  fake_rtcp.SetNetworkRoute(absl::optional<rtc::NetworkRoute>(network_route));
>   transport.SetRtcpPacketTransport(&fake_rtcp);
>   ASSERT_TRUE(observer.network_route());
>   EXPECT_EQ(network_route, *(observer.network_route()));
>@@ -197,49 +224,63 @@ TEST(RtpTransportTest, SetRtcpTransportWithNetworkRouteChanged) {
>   EXPECT_FALSE(observer.network_route());
> }
> 
>-class SignalCounter : public sigslot::has_slots<> {
>- public:
>-  explicit SignalCounter(RtpTransport* transport) {
>-    transport->SignalReadyToSend.connect(this, &SignalCounter::OnReadyToSend);
>-  }
>-  int count() const { return count_; }
>-  void OnReadyToSend(bool ready) { ++count_; }
>+// Test that RTCP packets are sent over correct transport based on the RTCP-mux
>+// status.
>+TEST(RtpTransportTest, RtcpPacketSentOverCorrectTransport) {
>+  // If the RTCP-mux is not enabled, RTCP packets are expected to be sent over
>+  // the RtcpPacketTransport.
>+  RtpTransport transport(kMuxDisabled);
>+  rtc::FakePacketTransport fake_rtcp("fake_rtcp");
>+  rtc::FakePacketTransport fake_rtp("fake_rtp");
>+  transport.SetRtcpPacketTransport(&fake_rtcp);  // rtcp ready
>+  transport.SetRtpPacketTransport(&fake_rtp);    // rtp ready
>+  SignalObserver observer(&transport);
> 
>- private:
>-  int count_ = 0;
>-};
>+  fake_rtp.SetDestination(&fake_rtp, true);
>+  fake_rtcp.SetDestination(&fake_rtcp, true);
>+
>+  rtc::CopyOnWriteBuffer packet;
>+  EXPECT_TRUE(transport.SendRtcpPacket(&packet, rtc::PacketOptions(), 0));
>+  EXPECT_EQ(1, observer.rtcp_transport_sent_count());
>+
>+  // The RTCP packets are expected to be sent over RtpPacketTransport if
>+  // RTCP-mux is enabled.
>+  transport.SetRtcpMuxEnabled(true);
>+  EXPECT_TRUE(transport.SendRtcpPacket(&packet, rtc::PacketOptions(), 0));
>+  EXPECT_EQ(1, observer.rtp_transport_sent_count());
>+}
> 
> TEST(RtpTransportTest, ChangingReadyToSendStateOnlySignalsWhenChanged) {
>   RtpTransport transport(kMuxEnabled);
>-  SignalCounter observer(&transport);
>+  TransportObserver observer(&transport);
>   rtc::FakePacketTransport fake_rtp("fake_rtp");
>   fake_rtp.SetWritable(true);
> 
>   // State changes, so we should signal.
>   transport.SetRtpPacketTransport(&fake_rtp);
>-  EXPECT_EQ(observer.count(), 1);
>+  EXPECT_EQ(observer.ready_to_send_signal_count(), 1);
> 
>   // State does not change, so we should not signal.
>   transport.SetRtpPacketTransport(&fake_rtp);
>-  EXPECT_EQ(observer.count(), 1);
>+  EXPECT_EQ(observer.ready_to_send_signal_count(), 1);
> 
>   // State does not change, so we should not signal.
>   transport.SetRtcpMuxEnabled(true);
>-  EXPECT_EQ(observer.count(), 1);
>+  EXPECT_EQ(observer.ready_to_send_signal_count(), 1);
> 
>   // State changes, so we should signal.
>   transport.SetRtcpMuxEnabled(false);
>-  EXPECT_EQ(observer.count(), 2);
>+  EXPECT_EQ(observer.ready_to_send_signal_count(), 2);
> }
> 
> // Test that SignalPacketReceived fires with rtcp=true when a RTCP packet is
> // received.
> TEST(RtpTransportTest, SignalDemuxedRtcp) {
>   RtpTransport transport(kMuxDisabled);
>-  SignalPacketReceivedCounter observer(&transport);
>   rtc::FakePacketTransport fake_rtp("fake_rtp");
>   fake_rtp.SetDestination(&fake_rtp, true);
>   transport.SetRtpPacketTransport(&fake_rtp);
>+  TransportObserver observer(&transport);
> 
>   // An rtcp packet.
>   const char data[] = {0, 73, 0, 0};
>@@ -259,11 +300,14 @@ static const int kRtpLen = 12;
> // handled payload type is received.
> TEST(RtpTransportTest, SignalHandledRtpPayloadType) {
>   RtpTransport transport(kMuxDisabled);
>-  SignalPacketReceivedCounter observer(&transport);
>   rtc::FakePacketTransport fake_rtp("fake_rtp");
>   fake_rtp.SetDestination(&fake_rtp, true);
>   transport.SetRtpPacketTransport(&fake_rtp);
>-  transport.AddHandledPayloadType(0x11);
>+  TransportObserver observer(&transport);
>+  RtpDemuxerCriteria demuxer_criteria;
>+  // Add a handled payload type.
>+  demuxer_criteria.payload_types = {0x11};
>+  transport.RegisterRtpDemuxerSink(demuxer_criteria, &observer);
> 
>   // An rtp packet.
>   const rtc::PacketOptions options;
>@@ -272,16 +316,22 @@ TEST(RtpTransportTest, SignalHandledRtpPayloadType) {
>   fake_rtp.SendPacket(rtp_data.data<char>(), kRtpLen, options, flags);
>   EXPECT_EQ(1, observer.rtp_count());
>   EXPECT_EQ(0, observer.rtcp_count());
>+  // Remove the sink before destroying the transport.
>+  transport.UnregisterRtpDemuxerSink(&observer);
> }
> 
> // Test that SignalPacketReceived does not fire when a RTP packet with an
> // unhandled payload type is received.
> TEST(RtpTransportTest, DontSignalUnhandledRtpPayloadType) {
>   RtpTransport transport(kMuxDisabled);
>-  SignalPacketReceivedCounter observer(&transport);
>   rtc::FakePacketTransport fake_rtp("fake_rtp");
>   fake_rtp.SetDestination(&fake_rtp, true);
>   transport.SetRtpPacketTransport(&fake_rtp);
>+  TransportObserver observer(&transport);
>+  RtpDemuxerCriteria demuxer_criteria;
>+  // Add an unhandled payload type.
>+  demuxer_criteria.payload_types = {0x12};
>+  transport.RegisterRtpDemuxerSink(demuxer_criteria, &observer);
> 
>   const rtc::PacketOptions options;
>   const int flags = 0;
>@@ -289,6 +339,8 @@ TEST(RtpTransportTest, DontSignalUnhandledRtpPayloadType) {
>   fake_rtp.SendPacket(rtp_data.data<char>(), kRtpLen, options, flags);
>   EXPECT_EQ(0, observer.rtp_count());
>   EXPECT_EQ(0, observer.rtcp_count());
>+  // Remove the sink before destroying the transport.
>+  transport.UnregisterRtpDemuxerSink(&observer);
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransportinternal.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransportinternal.h
>index 2ad66ab1c34..ca056807108 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransportinternal.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransportinternal.h
>@@ -13,10 +13,13 @@
> 
> #include <string>
> 
>-#include "api/ortc/rtptransportinterface.h"
>+#include "api/ortc/srtptransportinterface.h"
>+#include "call/rtp_demuxer.h"
> #include "p2p/base/icetransportinternal.h"
>+#include "pc/sessiondescription.h"
> #include "rtc_base/networkroute.h"
>-#include "rtc_base/sigslot.h"
>+#include "rtc_base/sslstreamadapter.h"
>+#include "rtc_base/third_party/sigslot/sigslot.h"
> 
> namespace rtc {
> class CopyOnWriteBuffer;
>@@ -26,11 +29,11 @@ struct PacketTime;
> 
> namespace webrtc {
> 
>-// This represents the internal interface beneath RtpTransportInterface;
>+// This represents the internal interface beneath SrtpTransportInterface;
> // it is not accessible to API consumers but is accessible to internal classes
> // in order to send and receive RTP and RTCP packets belonging to a single RTP
> // session. Additional convenience and configuration methods are also provided.
>-class RtpTransportInternal : public RtpTransportInterface,
>+class RtpTransportInternal : public SrtpTransportInterface,
>                              public sigslot::has_slots<> {
>  public:
>   virtual void SetRtcpMuxEnabled(bool enable) = 0;
>@@ -46,20 +49,22 @@ class RtpTransportInternal : public RtpTransportInterface,
>   virtual rtc::PacketTransportInternal* rtcp_packet_transport() const = 0;
>   virtual void SetRtcpPacketTransport(rtc::PacketTransportInternal* rtcp) = 0;
> 
>+  virtual bool IsReadyToSend() const = 0;
>+
>   // Called whenever a transport's ready-to-send state changes. The argument
>   // is true if all used transports are ready to send. This is more specific
>   // than just "writable"; it means the last send didn't return ENOTCONN.
>   sigslot::signal1<bool> SignalReadyToSend;
> 
>-  // TODO(zstein): Consider having two signals - RtpPacketReceived and
>-  // RtcpPacketReceived.
>-  // The first argument is true for RTCP packets and false for RTP packets.
>-  sigslot::signal3<bool, rtc::CopyOnWriteBuffer*, const rtc::PacketTime&>
>-      SignalPacketReceived;
>+  // Called whenever an RTCP packet is received. There is no equivalent signal
>+  // for RTP packets because they would be forwarded to the BaseChannel through
>+  // the RtpDemuxer callback.
>+  sigslot::signal2<rtc::CopyOnWriteBuffer*, const rtc::PacketTime&>
>+      SignalRtcpPacketReceived;
> 
>   // Called whenever the network route of the P2P layer transport changes.
>   // The argument is an optional network route.
>-  sigslot::signal1<rtc::Optional<rtc::NetworkRoute>> SignalNetworkRouteChanged;
>+  sigslot::signal1<absl::optional<rtc::NetworkRoute>> SignalNetworkRouteChanged;
> 
>   // Called whenever a transport's writable state might change. The argument is
>   // true if the transport is writable, otherwise it is false.
>@@ -79,9 +84,25 @@ class RtpTransportInternal : public RtpTransportInterface,
>                               const rtc::PacketOptions& options,
>                               int flags) = 0;
> 
>-  virtual bool HandlesPayloadType(int payload_type) const = 0;
>-
>-  virtual void AddHandledPayloadType(int payload_type) = 0;
>+  // This method updates the RTP header extension map so that the RTP transport
>+  // can parse the received packets and identify the MID. This is called by the
>+  // BaseChannel when setting the content description.
>+  //
>+  // TODO(zhihuang): Merging and replacing following methods handling header
>+  // extensions with SetParameters:
>+  //   UpdateRtpHeaderExtensionMap,
>+  //   UpdateSendEncryptedHeaderExtensionIds,
>+  //   UpdateRecvEncryptedHeaderExtensionIds,
>+  //   CacheRtpAbsSendTimeHeaderExtension,
>+  virtual void UpdateRtpHeaderExtensionMap(
>+      const cricket::RtpHeaderExtensions& header_extensions) = 0;
>+
>+  virtual bool IsSrtpActive() const = 0;
>+
>+  virtual bool RegisterRtpDemuxerSink(const RtpDemuxerCriteria& criteria,
>+                                      RtpPacketSinkInterface* sink) = 0;
>+
>+  virtual bool UnregisterRtpDemuxerSink(RtpPacketSinkInterface* sink) = 0;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransportinternaladapter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransportinternaladapter.h
>index 402f9cf49bc..fd92c5127d7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransportinternaladapter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransportinternaladapter.h
>@@ -24,6 +24,8 @@ namespace webrtc {
> // methods.
> class RtpTransportInternalAdapter : public RtpTransportInternal {
>  public:
>+  RtpTransportInternalAdapter() {}
>+
>   explicit RtpTransportInternalAdapter(RtpTransportInternal* transport)
>       : transport_(transport) {
>     RTC_DCHECK(transport_);
>@@ -51,6 +53,8 @@ class RtpTransportInternalAdapter : public RtpTransportInternal {
>     transport_->SetRtcpPacketTransport(rtcp);
>   }
> 
>+  bool IsReadyToSend() const override { return transport_->IsReadyToSend(); }
>+
>   bool IsWritable(bool rtcp) const override {
>     return transport_->IsWritable(rtcp);
>   }
>@@ -67,12 +71,18 @@ class RtpTransportInternalAdapter : public RtpTransportInternal {
>     return transport_->SendRtcpPacket(packet, options, flags);
>   }
> 
>-  bool HandlesPayloadType(int payload_type) const override {
>-    return transport_->HandlesPayloadType(payload_type);
>+  void UpdateRtpHeaderExtensionMap(
>+      const cricket::RtpHeaderExtensions& header_extensions) override {
>+    transport_->UpdateRtpHeaderExtensionMap(header_extensions);
>   }
> 
>-  void AddHandledPayloadType(int payload_type) override {
>-    return transport_->AddHandledPayloadType(payload_type);
>+  bool RegisterRtpDemuxerSink(const RtpDemuxerCriteria& criteria,
>+                              RtpPacketSinkInterface* sink) override {
>+    return transport_->RegisterRtpDemuxerSink(criteria, sink);
>+  }
>+
>+  bool UnregisterRtpDemuxerSink(RtpPacketSinkInterface* sink) override {
>+    return transport_->UnregisterRtpDemuxerSink(sink);
>   }
> 
>   // RtpTransportInterface overrides.
>@@ -92,9 +102,11 @@ class RtpTransportInternalAdapter : public RtpTransportInternal {
>     return transport_->GetParameters();
>   }
> 
>+  RtpTransportAdapter* GetInternal() override { return nullptr; }
>+
>  protected:
>   // Owned by the subclasses.
>-  RtpTransportInternal* transport_;
>+  RtpTransportInternal* transport_ = nullptr;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransporttestutil.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransporttestutil.h
>index c2bdaad23e7..eb337730d10 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransporttestutil.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/rtptransporttestutil.h
>@@ -11,32 +11,66 @@
> #ifndef PC_RTPTRANSPORTTESTUTIL_H_
> #define PC_RTPTRANSPORTTESTUTIL_H_
> 
>+#include "call/rtp_packet_sink_interface.h"
>+#include "modules/rtp_rtcp/source/rtp_packet_received.h"
> #include "pc/rtptransportinternal.h"
>-#include "rtc_base/sigslot.h"
>+#include "rtc_base/third_party/sigslot/sigslot.h"
> 
> namespace webrtc {
> 
>-class SignalPacketReceivedCounter : public sigslot::has_slots<> {
>+// Used to handle the signals when the RtpTransport receives an RTP/RTCP packet.
>+// Used in Rtp/Srtp/DtlsTransport unit tests.
>+class TransportObserver : public RtpPacketSinkInterface,
>+                          public sigslot::has_slots<> {
>  public:
>-  explicit SignalPacketReceivedCounter(RtpTransportInternal* transport) {
>-    transport->SignalPacketReceived.connect(
>-        this, &SignalPacketReceivedCounter::OnPacketReceived);
>+  TransportObserver() {}
>+
>+  explicit TransportObserver(RtpTransportInternal* rtp_transport) {
>+    rtp_transport->SignalRtcpPacketReceived.connect(
>+        this, &TransportObserver::OnRtcpPacketReceived);
>+    rtp_transport->SignalReadyToSend.connect(this,
>+                                             &TransportObserver::OnReadyToSend);
>   }
>-  int rtcp_count() const { return rtcp_count_; }
>+
>+  // RtpPacketInterface override.
>+  void OnRtpPacket(const RtpPacketReceived& packet) override {
>+    rtp_count_++;
>+    last_recv_rtp_packet_ = packet.Buffer();
>+  }
>+
>+  void OnRtcpPacketReceived(rtc::CopyOnWriteBuffer* packet,
>+                            const rtc::PacketTime& packet_time) {
>+    rtcp_count_++;
>+    last_recv_rtcp_packet_ = *packet;
>+  }
>+
>   int rtp_count() const { return rtp_count_; }
>+  int rtcp_count() const { return rtcp_count_; }
> 
>- private:
>-  void OnPacketReceived(bool rtcp,
>-                        rtc::CopyOnWriteBuffer*,
>-                        const rtc::PacketTime&) {
>-    if (rtcp) {
>-      ++rtcp_count_;
>-    } else {
>-      ++rtp_count_;
>-    }
>+  rtc::CopyOnWriteBuffer last_recv_rtp_packet() {
>+    return last_recv_rtp_packet_;
>   }
>-  int rtcp_count_ = 0;
>+
>+  rtc::CopyOnWriteBuffer last_recv_rtcp_packet() {
>+    return last_recv_rtcp_packet_;
>+  }
>+
>+  void OnReadyToSend(bool ready) {
>+    ready_to_send_signal_count_++;
>+    ready_to_send_ = ready;
>+  }
>+
>+  bool ready_to_send() { return ready_to_send_; }
>+
>+  int ready_to_send_signal_count() { return ready_to_send_signal_count_; }
>+
>+ private:
>+  bool ready_to_send_ = false;
>   int rtp_count_ = 0;
>+  int rtcp_count_ = 0;
>+  int ready_to_send_signal_count_ = 0;
>+  rtc::CopyOnWriteBuffer last_recv_rtp_packet_;
>+  rtc::CopyOnWriteBuffer last_recv_rtcp_packet_;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/sctputils.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/sctputils.cc
>index 6f0ef016d93..1dfa4db0262 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/sctputils.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/sctputils.cc
>@@ -88,7 +88,7 @@ bool ParseDataChannelOpenMessage(const rtc::CopyOnWriteBuffer& payload,
>     RTC_LOG(LS_WARNING) << "Could not read OPEN message protocol length.";
>     return false;
>   }
>-  if (!buffer.ReadString(label, (size_t) label_length)) {
>+  if (!buffer.ReadString(label, (size_t)label_length)) {
>     RTC_LOG(LS_WARNING) << "Could not read OPEN message label";
>     return false;
>   }
>@@ -165,9 +165,9 @@ bool WriteDataChannelOpenMessage(const std::string& label,
>     }
>   }
> 
>-  rtc::ByteBufferWriter buffer(
>-      NULL, 20 + label.length() + config.protocol.length(),
>-      rtc::ByteBuffer::ORDER_NETWORK);
>+  rtc::ByteBufferWriter buffer(NULL,
>+                               20 + label.length() + config.protocol.length(),
>+                               rtc::ByteBuffer::ORDER_NETWORK);
>   // TODO(tommi): Add error handling and check resulting length.
>   buffer.WriteUInt8(DATA_CHANNEL_OPEN_MESSAGE_TYPE);
>   buffer.WriteUInt8(channel_type);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/sctputils_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/sctputils_unittest.cc
>index 86cdc47a5b1..d2f2de6b6d8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/sctputils_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/sctputils_unittest.cc
>@@ -31,12 +31,14 @@ class SctpUtilsTest : public testing::Test {
> 
>     ASSERT_TRUE(buffer.ReadUInt8(&channel_type));
>     if (config.ordered) {
>-      EXPECT_EQ(config.maxRetransmits > -1 ?
>-                    0x01 : (config.maxRetransmitTime > -1 ? 0x02 : 0),
>+      EXPECT_EQ(config.maxRetransmits > -1
>+                    ? 0x01
>+                    : (config.maxRetransmitTime > -1 ? 0x02 : 0),
>                 channel_type);
>     } else {
>-      EXPECT_EQ(config.maxRetransmits > -1 ?
>-                    0x81 : (config.maxRetransmitTime > -1 ? 0x82 : 0x80),
>+      EXPECT_EQ(config.maxRetransmits > -1
>+                    ? 0x81
>+                    : (config.maxRetransmitTime > -1 ? 0x82 : 0x80),
>                 channel_type);
>     }
> 
>@@ -44,8 +46,8 @@ class SctpUtilsTest : public testing::Test {
> 
>     ASSERT_TRUE(buffer.ReadUInt32(&reliability));
>     if (config.maxRetransmits > -1 || config.maxRetransmitTime > -1) {
>-      EXPECT_EQ(config.maxRetransmits > -1 ?
>-                    config.maxRetransmits : config.maxRetransmitTime,
>+      EXPECT_EQ(config.maxRetransmits > -1 ? config.maxRetransmits
>+                                           : config.maxRetransmitTime,
>                 static_cast<int>(reliability));
>     }
> 
>@@ -75,8 +77,8 @@ TEST_F(SctpUtilsTest, WriteParseOpenMessageWithOrderedReliable) {
> 
>   std::string output_label;
>   webrtc::DataChannelInit output_config;
>-  ASSERT_TRUE(webrtc::ParseDataChannelOpenMessage(
>-      packet, &output_label, &output_config));
>+  ASSERT_TRUE(webrtc::ParseDataChannelOpenMessage(packet, &output_label,
>+                                                  &output_config));
> 
>   EXPECT_EQ(label, output_label);
>   EXPECT_EQ(config.protocol, output_config.protocol);
>@@ -99,8 +101,8 @@ TEST_F(SctpUtilsTest, WriteParseOpenMessageWithMaxRetransmitTime) {
> 
>   std::string output_label;
>   webrtc::DataChannelInit output_config;
>-  ASSERT_TRUE(webrtc::ParseDataChannelOpenMessage(
>-      packet, &output_label, &output_config));
>+  ASSERT_TRUE(webrtc::ParseDataChannelOpenMessage(packet, &output_label,
>+                                                  &output_config));
> 
>   EXPECT_EQ(label, output_label);
>   EXPECT_EQ(config.protocol, output_config.protocol);
>@@ -122,8 +124,8 @@ TEST_F(SctpUtilsTest, WriteParseOpenMessageWithMaxRetransmits) {
> 
>   std::string output_label;
>   webrtc::DataChannelInit output_config;
>-  ASSERT_TRUE(webrtc::ParseDataChannelOpenMessage(
>-      packet, &output_label, &output_config));
>+  ASSERT_TRUE(webrtc::ParseDataChannelOpenMessage(packet, &output_label,
>+                                                  &output_config));
> 
>   EXPECT_EQ(label, output_label);
>   EXPECT_EQ(config.protocol, output_config.protocol);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/sdputils.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/sdputils.cc
>index de77d63bf58..07bff7bca2f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/sdputils.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/sdputils.cc
>@@ -13,8 +13,8 @@
> #include <string>
> #include <utility>
> 
>+#include "absl/memory/memory.h"
> #include "api/jsepsessiondescription.h"
>-#include "rtc_base/ptr_util.h"
> 
> namespace webrtc {
> 
>@@ -28,7 +28,7 @@ std::unique_ptr<SessionDescriptionInterface> CloneSessionDescriptionAsType(
>     const SessionDescriptionInterface* sdesc,
>     SdpType type) {
>   RTC_DCHECK(sdesc);
>-  auto clone = rtc::MakeUnique<JsepSessionDescription>(type);
>+  auto clone = absl::make_unique<JsepSessionDescription>(type);
>   clone->Initialize(sdesc->description()->Copy(), sdesc->session_id(),
>                     sdesc->session_version());
>   // As of writing, our version of GCC does not allow returning a unique_ptr of
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/sessiondescription.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/sessiondescription.h
>index b2cdee9482a..645ebc54f86 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/sessiondescription.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/sessiondescription.h
>@@ -363,6 +363,15 @@ const ContentInfo* FindContentInfoByName(const ContentInfos& contents,
> const ContentInfo* FindContentInfoByType(const ContentInfos& contents,
>                                          const std::string& type);
> 
>+// Determines how the MSID will be signaled in the SDP. These can be used as
>+// flags to indicate both or none.
>+enum MsidSignaling {
>+  // Signal MSID with one a=msid line in the media section.
>+  kMsidSignalingMediaSection = 0x1,
>+  // Signal MSID with a=ssrc: msid lines in the media section.
>+  kMsidSignalingSsrcAttribute = 0x2
>+};
>+
> // Describes a collection of contents, each with its own name and
> // type.  Analogous to a <jingle> or <session> stanza.  Assumes that
> // contents are unique be name, but doesn't enforce that.
>@@ -439,6 +448,13 @@ class SessionDescription {
>   void set_msid_supported(bool supported) { msid_supported_ = supported; }
>   bool msid_supported() const { return msid_supported_; }
> 
>+  // Determines how the MSIDs were/will be signaled. Flag value composed of
>+  // MsidSignaling bits (see enum above).
>+  void set_msid_signaling(int msid_signaling) {
>+    msid_signaling_ = msid_signaling;
>+  }
>+  int msid_signaling() const { return msid_signaling_; }
>+
>  private:
>   SessionDescription(const SessionDescription&);
> 
>@@ -446,6 +462,9 @@ class SessionDescription {
>   TransportInfos transport_infos_;
>   ContentGroups content_groups_;
>   bool msid_supported_ = true;
>+  // Default to what Plan B would do.
>+  // TODO(bugs.webrtc.org/8530): Change default to kMsidSignalingMediaSection.
>+  int msid_signaling_ = kMsidSignalingSsrcAttribute;
> };
> 
> // Indicates whether a session description was sent by the local client or
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtpfilter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtpfilter.cc
>index 157fdebff3d..5ffe34212e5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtpfilter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtpfilter.cc
>@@ -11,30 +11,54 @@
> #include "pc/srtpfilter.h"
> 
> #include <string.h>
>-
> #include <algorithm>
>+#include <utility>
> 
> #include "media/base/rtputils.h"
> #include "pc/srtpsession.h"
>-#include "rtc_base/base64.h"
> #include "rtc_base/byteorder.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/stringencode.h"
>+#include "rtc_base/third_party/base64/base64.h"
> #include "rtc_base/timeutils.h"
>+#include "rtc_base/zero_memory.h"
> 
> namespace cricket {
> 
>-SrtpFilter::SrtpFilter() {
>-}
>+SrtpFilter::SrtpFilter() {}
> 
>-SrtpFilter::~SrtpFilter() {
>-}
>+SrtpFilter::~SrtpFilter() {}
> 
> bool SrtpFilter::IsActive() const {
>   return state_ >= ST_ACTIVE;
> }
> 
>+bool SrtpFilter::Process(const std::vector<CryptoParams>& cryptos,
>+                         webrtc::SdpType type,
>+                         ContentSource source) {
>+  bool ret = false;
>+  switch (type) {
>+    case webrtc::SdpType::kOffer:
>+      ret = SetOffer(cryptos, source);
>+      break;
>+    case webrtc::SdpType::kPrAnswer:
>+      ret = SetProvisionalAnswer(cryptos, source);
>+      break;
>+    case webrtc::SdpType::kAnswer:
>+      ret = SetAnswer(cryptos, source);
>+      break;
>+    default:
>+      break;
>+  }
>+
>+  if (!ret) {
>+    return false;
>+  }
>+
>+  return true;
>+}
>+
> bool SrtpFilter::SetOffer(const std::vector<CryptoParams>& offer_params,
>                           ContentSource source) {
>   if (!ExpectOffer(source)) {
>@@ -56,10 +80,9 @@ bool SrtpFilter::SetProvisionalAnswer(
> }
> 
> bool SrtpFilter::ExpectOffer(ContentSource source) {
>-  return ((state_ == ST_INIT) ||
>-          (state_ == ST_ACTIVE) ||
>-          (state_  == ST_SENTOFFER && source == CS_LOCAL) ||
>-          (state_  == ST_SENTUPDATEDOFFER && source == CS_LOCAL) ||
>+  return ((state_ == ST_INIT) || (state_ == ST_ACTIVE) ||
>+          (state_ == ST_SENTOFFER && source == CS_LOCAL) ||
>+          (state_ == ST_SENTUPDATEDOFFER && source == CS_LOCAL) ||
>           (state_ == ST_RECEIVEDOFFER && source == CS_REMOTE) ||
>           (state_ == ST_RECEIVEDUPDATEDOFFER && source == CS_REMOTE));
> }
>@@ -104,8 +127,8 @@ bool SrtpFilter::DoSetAnswer(const std::vector<CryptoParams>& answer_params,
>     } else {
>       // Need to wait for the final answer to decide if
>       // we should go to Active state.
>-      state_ = (source == CS_LOCAL) ? ST_SENTPRANSWER_NO_CRYPTO :
>-                                      ST_RECEIVEDPRANSWER_NO_CRYPTO;
>+      state_ = (source == CS_LOCAL) ? ST_SENTPRANSWER_NO_CRYPTO
>+                                    : ST_RECEIVEDPRANSWER_NO_CRYPTO;
>       return true;
>     }
>   }
>@@ -127,8 +150,7 @@ bool SrtpFilter::DoSetAnswer(const std::vector<CryptoParams>& answer_params,
>     offer_params_.clear();
>     state_ = ST_ACTIVE;
>   } else {
>-    state_ =
>-        (source == CS_LOCAL) ? ST_SENTPRANSWER : ST_RECEIVEDPRANSWER;
>+    state_ = (source == CS_LOCAL) ? ST_SENTPRANSWER : ST_RECEIVEDPRANSWER;
>   }
>   return true;
> }
>@@ -164,8 +186,8 @@ bool SrtpFilter::ResetParams() {
>   offer_params_.clear();
>   applied_send_params_ = CryptoParams();
>   applied_recv_params_ = CryptoParams();
>-  send_cipher_suite_ = rtc::nullopt;
>-  recv_cipher_suite_ = rtc::nullopt;
>+  send_cipher_suite_ = absl::nullopt;
>+  recv_cipher_suite_ = absl::nullopt;
>   send_key_.Clear();
>   recv_key_.Clear();
>   state_ = ST_INIT;
>@@ -184,7 +206,8 @@ bool SrtpFilter::ApplySendParams(const CryptoParams& send_params) {
>   send_cipher_suite_ = rtc::SrtpCryptoSuiteFromName(send_params.cipher_suite);
>   if (send_cipher_suite_ == rtc::SRTP_INVALID_CRYPTO_SUITE) {
>     RTC_LOG(LS_WARNING) << "Unknown crypto suite(s) received:"
>-                        << " send cipher_suite " << send_params.cipher_suite;
>+                           " send cipher_suite "
>+                        << send_params.cipher_suite;
>     return false;
>   }
> 
>@@ -192,11 +215,12 @@ bool SrtpFilter::ApplySendParams(const CryptoParams& send_params) {
>   if (!rtc::GetSrtpKeyAndSaltLengths(*send_cipher_suite_, &send_key_len,
>                                      &send_salt_len)) {
>     RTC_LOG(LS_WARNING) << "Could not get lengths for crypto suite(s):"
>-                        << " send cipher_suite " << send_params.cipher_suite;
>+                           " send cipher_suite "
>+                        << send_params.cipher_suite;
>     return false;
>   }
> 
>-  send_key_ = rtc::Buffer(send_key_len + send_salt_len);
>+  send_key_ = rtc::ZeroOnFreeBuffer<uint8_t>(send_key_len + send_salt_len);
>   return ParseKeyParams(send_params.key_params, send_key_.data(),
>                         send_key_.size());
> }
>@@ -213,7 +237,8 @@ bool SrtpFilter::ApplyRecvParams(const CryptoParams& recv_params) {
>   recv_cipher_suite_ = rtc::SrtpCryptoSuiteFromName(recv_params.cipher_suite);
>   if (recv_cipher_suite_ == rtc::SRTP_INVALID_CRYPTO_SUITE) {
>     RTC_LOG(LS_WARNING) << "Unknown crypto suite(s) received:"
>-                        << " recv cipher_suite " << recv_params.cipher_suite;
>+                           " recv cipher_suite "
>+                        << recv_params.cipher_suite;
>     return false;
>   }
> 
>@@ -221,11 +246,12 @@ bool SrtpFilter::ApplyRecvParams(const CryptoParams& recv_params) {
>   if (!rtc::GetSrtpKeyAndSaltLengths(*recv_cipher_suite_, &recv_key_len,
>                                      &recv_salt_len)) {
>     RTC_LOG(LS_WARNING) << "Could not get lengths for crypto suite(s):"
>-                        << " recv cipher_suite " << recv_params.cipher_suite;
>+                           " recv cipher_suite "
>+                        << recv_params.cipher_suite;
>     return false;
>   }
> 
>-  recv_key_ = rtc::Buffer(recv_key_len + recv_salt_len);
>+  recv_key_ = rtc::ZeroOnFreeBuffer<uint8_t>(recv_key_len + recv_salt_len);
>   return ParseKeyParams(recv_params.key_params, recv_key_.data(),
>                         recv_key_.size());
> }
>@@ -249,6 +275,9 @@ bool SrtpFilter::ParseKeyParams(const std::string& key_params,
>   }
> 
>   memcpy(key, key_str.c_str(), len);
>+  // TODO(bugs.webrtc.org/8905): Switch to ZeroOnFreeBuffer for storing
>+  // sensitive data.
>+  rtc::ExplicitZeroMemory(&key_str[0], key_str.size());
>   return true;
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtpfilter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtpfilter.h
>index e89a68af996..a4dd54fa8bd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtpfilter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtpfilter.h
>@@ -17,10 +17,11 @@
> #include <string>
> #include <vector>
> 
>+#include "absl/types/optional.h"
>+#include "api/array_view.h"
> #include "api/cryptoparams.h"
>-#include "api/optional.h"
>+#include "api/jsep.h"
> #include "pc/sessiondescription.h"
>-#include "rtc_base/basictypes.h"
> #include "rtc_base/buffer.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/criticalsection.h"
>@@ -37,10 +38,7 @@ namespace cricket {
> // TODO(zhihuang): Find a better name for this class, like "SdesNegotiator".
> class SrtpFilter {
>  public:
>-  enum Mode {
>-    PROTECT,
>-    UNPROTECT
>-  };
>+  enum Mode { PROTECT, UNPROTECT };
>   enum Error {
>     ERROR_NONE,
>     ERROR_FAIL,
>@@ -54,6 +52,13 @@ class SrtpFilter {
>   // Whether the filter is active (i.e. crypto has been properly negotiated).
>   bool IsActive() const;
> 
>+  // Handle the offer/answer negotiation of the crypto parameters internally.
>+  // TODO(zhihuang): Make SetOffer/ProvisionalAnswer/Answer private as helper
>+  // methods once start using Process.
>+  bool Process(const std::vector<CryptoParams>& cryptos,
>+               webrtc::SdpType type,
>+               ContentSource source);
>+
>   // Indicates which crypto algorithms and keys were contained in the offer.
>   // offer_params should contain a list of available parameters to use, or none,
>   // if crypto is not desired. This must be called before SetAnswer.
>@@ -73,11 +78,11 @@ class SrtpFilter {
> 
>   bool ResetParams();
> 
>-  rtc::Optional<int> send_cipher_suite() { return send_cipher_suite_; }
>-  rtc::Optional<int> recv_cipher_suite() { return recv_cipher_suite_; }
>+  absl::optional<int> send_cipher_suite() { return send_cipher_suite_; }
>+  absl::optional<int> recv_cipher_suite() { return recv_cipher_suite_; }
> 
>-  const rtc::Buffer& send_key() { return send_key_; }
>-  const rtc::Buffer& recv_key() { return recv_key_; }
>+  rtc::ArrayView<const uint8_t> send_key() { return send_key_; }
>+  rtc::ArrayView<const uint8_t> recv_key() { return recv_key_; }
> 
>  protected:
>   bool ExpectOffer(ContentSource source);
>@@ -104,13 +109,13 @@ class SrtpFilter {
>                              size_t len);
> 
>   enum State {
>-    ST_INIT,           // SRTP filter unused.
>-    ST_SENTOFFER,      // Offer with SRTP parameters sent.
>-    ST_RECEIVEDOFFER,  // Offer with SRTP parameters received.
>+    ST_INIT,                    // SRTP filter unused.
>+    ST_SENTOFFER,               // Offer with SRTP parameters sent.
>+    ST_RECEIVEDOFFER,           // Offer with SRTP parameters received.
>     ST_SENTPRANSWER_NO_CRYPTO,  // Sent provisional answer without crypto.
>     // Received provisional answer without crypto.
>     ST_RECEIVEDPRANSWER_NO_CRYPTO,
>-    ST_ACTIVE,         // Offer and answer set.
>+    ST_ACTIVE,  // Offer and answer set.
>     // SRTP filter is active but new parameters are offered.
>     // When the answer is set, the state transitions to ST_ACTIVE or ST_INIT.
>     ST_SENTUPDATEDOFFER,
>@@ -130,10 +135,10 @@ class SrtpFilter {
>   std::vector<CryptoParams> offer_params_;
>   CryptoParams applied_send_params_;
>   CryptoParams applied_recv_params_;
>-  rtc::Optional<int> send_cipher_suite_;
>-  rtc::Optional<int> recv_cipher_suite_;
>-  rtc::Buffer send_key_;
>-  rtc::Buffer recv_key_;
>+  absl::optional<int> send_cipher_suite_;
>+  absl::optional<int> recv_cipher_suite_;
>+  rtc::ZeroOnFreeBuffer<uint8_t> send_key_;
>+  rtc::ZeroOnFreeBuffer<uint8_t> recv_key_;
> };
> 
> }  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtpfilter_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtpfilter_unittest.cc
>index 017df5baf1f..e901fcdf7cb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtpfilter_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtpfilter_unittest.cc
>@@ -37,18 +37,30 @@ static const char kTestKeyParamsGcm3[] =
>     "inline:YKlABGZWMgX32xuMotrG0v0T7G83veegaVzubQ==";
> static const char kTestKeyParamsGcm4[] =
>     "inline:gJ6tWoUym2v+/F6xjr7xaxiS3QbJJozl3ZD/0A==";
>-static const cricket::CryptoParams kTestCryptoParams1(
>-    1, "AES_CM_128_HMAC_SHA1_80", kTestKeyParams1, "");
>-static const cricket::CryptoParams kTestCryptoParams2(
>-    1, "AES_CM_128_HMAC_SHA1_80", kTestKeyParams2, "");
>-static const cricket::CryptoParams kTestCryptoParamsGcm1(
>-    1, "AEAD_AES_256_GCM", kTestKeyParamsGcm1, "");
>-static const cricket::CryptoParams kTestCryptoParamsGcm2(
>-    1, "AEAD_AES_256_GCM", kTestKeyParamsGcm2, "");
>-static const cricket::CryptoParams kTestCryptoParamsGcm3(
>-    1, "AEAD_AES_128_GCM", kTestKeyParamsGcm3, "");
>-static const cricket::CryptoParams kTestCryptoParamsGcm4(
>-    1, "AEAD_AES_128_GCM", kTestKeyParamsGcm4, "");
>+static const cricket::CryptoParams kTestCryptoParams1(1,
>+                                                      "AES_CM_128_HMAC_SHA1_80",
>+                                                      kTestKeyParams1,
>+                                                      "");
>+static const cricket::CryptoParams kTestCryptoParams2(1,
>+                                                      "AES_CM_128_HMAC_SHA1_80",
>+                                                      kTestKeyParams2,
>+                                                      "");
>+static const cricket::CryptoParams kTestCryptoParamsGcm1(1,
>+                                                         "AEAD_AES_256_GCM",
>+                                                         kTestKeyParamsGcm1,
>+                                                         "");
>+static const cricket::CryptoParams kTestCryptoParamsGcm2(1,
>+                                                         "AEAD_AES_256_GCM",
>+                                                         kTestKeyParamsGcm2,
>+                                                         "");
>+static const cricket::CryptoParams kTestCryptoParamsGcm3(1,
>+                                                         "AEAD_AES_128_GCM",
>+                                                         kTestKeyParamsGcm3,
>+                                                         "");
>+static const cricket::CryptoParams kTestCryptoParamsGcm4(1,
>+                                                         "AEAD_AES_128_GCM",
>+                                                         kTestKeyParamsGcm4,
>+                                                         "");
> 
> class SrtpFilterTest : public testing::Test {
>  protected:
>@@ -71,11 +83,17 @@ class SrtpFilterTest : public testing::Test {
>     EXPECT_TRUE(f2_.IsActive());
>   }
> 
>+  void VerifyKeysAreEqual(ArrayView<const uint8_t> key1,
>+                          ArrayView<const uint8_t> key2) {
>+    EXPECT_EQ(key1.size(), key2.size());
>+    EXPECT_EQ(0, memcmp(key1.data(), key2.data(), key1.size()));
>+  }
>+
>   void VerifyCryptoParamsMatch(const std::string& cs1, const std::string& cs2) {
>     EXPECT_EQ(rtc::SrtpCryptoSuiteFromName(cs1), f1_.send_cipher_suite());
>     EXPECT_EQ(rtc::SrtpCryptoSuiteFromName(cs2), f2_.send_cipher_suite());
>-    EXPECT_TRUE(f1_.send_key() == f2_.recv_key());
>-    EXPECT_TRUE(f2_.send_key() == f1_.recv_key());
>+    VerifyKeysAreEqual(f1_.send_key(), f2_.recv_key());
>+    VerifyKeysAreEqual(f2_.send_key(), f1_.recv_key());
>   }
> 
>   cricket::SrtpFilter f1_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtpsession.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtpsession.cc
>index a07848d475c..a5fda8897fe 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtpsession.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtpsession.cc
>@@ -15,11 +15,17 @@
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/sslstreamadapter.h"
>+#include "system_wrappers/include/metrics.h"
> #include "third_party/libsrtp/include/srtp.h"
> #include "third_party/libsrtp/include/srtp_priv.h"
> 
> namespace cricket {
> 
>+// One more than the maximum libsrtp error code. Required by
>+// RTC_HISTOGRAM_ENUMERATION. Keep this in sync with srtp_error_status_t defined
>+// in srtp.h.
>+constexpr int kSrtpErrorCodeBoundary = 28;
>+
> SrtpSession::SrtpSession() {}
> 
> SrtpSession::~SrtpSession() {
>@@ -133,6 +139,8 @@ bool SrtpSession::UnprotectRtp(void* p, int in_len, int* out_len) {
>   int err = srtp_unprotect(session_, p, out_len);
>   if (err != srtp_err_status_ok) {
>     RTC_LOG(LS_WARNING) << "Failed to unprotect SRTP packet, err=" << err;
>+    RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.SrtpUnprotectError",
>+                              static_cast<int>(err), kSrtpErrorCodeBoundary);
>     return false;
>   }
>   return true;
>@@ -149,6 +157,8 @@ bool SrtpSession::UnprotectRtcp(void* p, int in_len, int* out_len) {
>   int err = srtp_unprotect_rtcp(session_, p, out_len);
>   if (err != srtp_err_status_ok) {
>     RTC_LOG(LS_WARNING) << "Failed to unprotect SRTCP packet, err=" << err;
>+    RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.SrtcpUnprotectError",
>+                              static_cast<int>(err), kSrtpErrorCodeBoundary);
>     return false;
>   }
>   return true;
>@@ -248,6 +258,7 @@ bool SrtpSession::DoSetKey(int type,
>   if (!rtc::GetSrtpKeyAndSaltLengths(cs, &expected_key_len,
>                                      &expected_salt_len)) {
>     // This should never happen.
>+    RTC_NOTREACHED();
>     RTC_LOG(LS_WARNING)
>         << "Failed to " << (session_ ? "update" : "create")
>         << " SRTP session: unsupported cipher_suite without length information"
>@@ -314,7 +325,7 @@ bool SrtpSession::SetKey(int type,
>   RTC_DCHECK(thread_checker_.CalledOnValidThread());
>   if (session_) {
>     RTC_LOG(LS_ERROR) << "Failed to create SRTP session: "
>-                      << "SRTP session already created";
>+                         "SRTP session already created";
>     return false;
>   }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtpsession.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtpsession.h
>index a6e78fab6be..c3a01bd139b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtpsession.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtpsession.h
>@@ -13,7 +13,7 @@
> 
> #include <vector>
> 
>-#include "rtc_base/basictypes.h"
>+#include "rtc_base/scoped_ref_ptr.h"
> #include "rtc_base/thread_checker.h"
> 
> // Forward declaration to avoid pulling in libsrtp headers here
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtpsession_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtpsession_unittest.cc
>index dc325739e82..b1bc9f06760 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtpsession_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtpsession_unittest.cc
>@@ -12,16 +12,22 @@
> 
> #include <string>
> 
>+#include "absl/memory/memory.h"
> #include "media/base/fakertp.h"
> #include "pc/srtptestutil.h"
> #include "rtc_base/gunit.h"
> #include "rtc_base/sslstreamadapter.h"  // For rtc::SRTP_*
>+#include "system_wrappers/include/metrics_default.h"
>+#include "third_party/libsrtp/include/srtp.h"
> 
> namespace rtc {
> 
> std::vector<int> kEncryptedHeaderExtensionIds;
> 
> class SrtpSessionTest : public testing::Test {
>+ public:
>+  SrtpSessionTest() { webrtc::metrics::Reset(); }
>+
>  protected:
>   virtual void SetUp() {
>     rtp_len_ = sizeof(kPcmuFrame);
>@@ -141,7 +147,17 @@ TEST_F(SrtpSessionTest, TestTamperReject) {
>   rtp_packet_[0] = 0x12;
>   rtcp_packet_[1] = 0x34;
>   EXPECT_FALSE(s2_.UnprotectRtp(rtp_packet_, rtp_len_, &out_len));
>+  EXPECT_EQ(1, webrtc::metrics::NumSamples(
>+                   "WebRTC.PeerConnection.SrtpUnprotectError"));
>+  EXPECT_EQ(
>+      1, webrtc::metrics::NumEvents("WebRTC.PeerConnection.SrtpUnprotectError",
>+                                    srtp_err_status_bad_param));
>   EXPECT_FALSE(s2_.UnprotectRtcp(rtcp_packet_, rtcp_len_, &out_len));
>+  EXPECT_EQ(1, webrtc::metrics::NumSamples(
>+                   "WebRTC.PeerConnection.SrtcpUnprotectError"));
>+  EXPECT_EQ(
>+      1, webrtc::metrics::NumEvents("WebRTC.PeerConnection.SrtcpUnprotectError",
>+                                    srtp_err_status_auth_fail));
> }
> 
> // Test that we fail to unprotect if the payloads are not authenticated.
>@@ -152,7 +168,17 @@ TEST_F(SrtpSessionTest, TestUnencryptReject) {
>   EXPECT_TRUE(s2_.SetRecv(SRTP_AES128_CM_SHA1_80, kTestKey1, kTestKeyLen,
>                           kEncryptedHeaderExtensionIds));
>   EXPECT_FALSE(s2_.UnprotectRtp(rtp_packet_, rtp_len_, &out_len));
>+  EXPECT_EQ(1, webrtc::metrics::NumSamples(
>+                   "WebRTC.PeerConnection.SrtpUnprotectError"));
>+  EXPECT_EQ(
>+      1, webrtc::metrics::NumEvents("WebRTC.PeerConnection.SrtpUnprotectError",
>+                                    srtp_err_status_auth_fail));
>   EXPECT_FALSE(s2_.UnprotectRtcp(rtcp_packet_, rtcp_len_, &out_len));
>+  EXPECT_EQ(1, webrtc::metrics::NumSamples(
>+                   "WebRTC.PeerConnection.SrtcpUnprotectError"));
>+  EXPECT_EQ(
>+      1, webrtc::metrics::NumEvents("WebRTC.PeerConnection.SrtcpUnprotectError",
>+                                    srtp_err_status_cant_check));
> }
> 
> // Test that we fail when using buffers that are too small.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtptransport.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtptransport.cc
>index 149651cddae..6fd8be37c24 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtptransport.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtptransport.cc
>@@ -13,176 +13,235 @@
> #include <string>
> #include <vector>
> 
>+#include "absl/memory/memory.h"
> #include "media/base/rtputils.h"
> #include "pc/rtptransport.h"
> #include "pc/srtpsession.h"
> #include "rtc_base/asyncpacketsocket.h"
>-#include "rtc_base/base64.h"
> #include "rtc_base/copyonwritebuffer.h"
>-#include "rtc_base/ptr_util.h"
>+#include "rtc_base/numerics/safe_conversions.h"
>+#include "rtc_base/third_party/base64/base64.h"
> #include "rtc_base/trace_event.h"
>+#include "rtc_base/zero_memory.h"
> 
> namespace webrtc {
> 
> SrtpTransport::SrtpTransport(bool rtcp_mux_enabled)
>-    : RtpTransportInternalAdapter(new RtpTransport(rtcp_mux_enabled)) {
>-  // Own the raw pointer |transport| from the base class.
>-  rtp_transport_.reset(transport_);
>-  RTC_DCHECK(rtp_transport_);
>-  ConnectToRtpTransport();
>-}
>+    : RtpTransport(rtcp_mux_enabled) {}
>+
>+RTCError SrtpTransport::SetSrtpSendKey(const cricket::CryptoParams& params) {
>+  if (send_params_) {
>+    LOG_AND_RETURN_ERROR(
>+        webrtc::RTCErrorType::UNSUPPORTED_OPERATION,
>+        "Setting the SRTP send key twice is currently unsupported.");
>+  }
>+  if (recv_params_ && recv_params_->cipher_suite != params.cipher_suite) {
>+    LOG_AND_RETURN_ERROR(
>+        webrtc::RTCErrorType::UNSUPPORTED_OPERATION,
>+        "The send key and receive key must have the same cipher suite.");
>+  }
>+
>+  send_cipher_suite_ = rtc::SrtpCryptoSuiteFromName(params.cipher_suite);
>+  if (*send_cipher_suite_ == rtc::SRTP_INVALID_CRYPTO_SUITE) {
>+    return RTCError(RTCErrorType::INVALID_PARAMETER,
>+                    "Invalid SRTP crypto suite");
>+  }
> 
>-SrtpTransport::SrtpTransport(
>-    std::unique_ptr<RtpTransportInternal> rtp_transport)
>-    : RtpTransportInternalAdapter(rtp_transport.get()),
>-      rtp_transport_(std::move(rtp_transport)) {
>-  RTC_DCHECK(rtp_transport_);
>-  ConnectToRtpTransport();
>+  int send_key_len, send_salt_len;
>+  if (!rtc::GetSrtpKeyAndSaltLengths(*send_cipher_suite_, &send_key_len,
>+                                     &send_salt_len)) {
>+    return RTCError(RTCErrorType::INVALID_PARAMETER,
>+                    "Could not get lengths for crypto suite(s):"
>+                    " send cipher_suite ");
>+  }
>+
>+  send_key_ = rtc::ZeroOnFreeBuffer<uint8_t>(send_key_len + send_salt_len);
>+  if (!ParseKeyParams(params.key_params, send_key_.data(), send_key_.size())) {
>+    return RTCError(RTCErrorType::INVALID_PARAMETER,
>+                    "Failed to parse the crypto key params");
>+  }
>+
>+  if (!MaybeSetKeyParams()) {
>+    return RTCError(RTCErrorType::INVALID_PARAMETER,
>+                    "Failed to set the crypto key params");
>+  }
>+  send_params_ = params;
>+  return RTCError::OK();
> }
> 
>-void SrtpTransport::ConnectToRtpTransport() {
>-  rtp_transport_->SignalPacketReceived.connect(
>-      this, &SrtpTransport::OnPacketReceived);
>-  rtp_transport_->SignalReadyToSend.connect(this,
>-                                            &SrtpTransport::OnReadyToSend);
>-  rtp_transport_->SignalNetworkRouteChanged.connect(
>-      this, &SrtpTransport::OnNetworkRouteChanged);
>-  rtp_transport_->SignalWritableState.connect(this,
>-                                              &SrtpTransport::OnWritableState);
>-  rtp_transport_->SignalSentPacket.connect(this, &SrtpTransport::OnSentPacket);
>+RTCError SrtpTransport::SetSrtpReceiveKey(const cricket::CryptoParams& params) {
>+  if (recv_params_) {
>+    LOG_AND_RETURN_ERROR(
>+        webrtc::RTCErrorType::UNSUPPORTED_OPERATION,
>+        "Setting the SRTP send key twice is currently unsupported.");
>+  }
>+  if (send_params_ && send_params_->cipher_suite != params.cipher_suite) {
>+    LOG_AND_RETURN_ERROR(
>+        webrtc::RTCErrorType::UNSUPPORTED_OPERATION,
>+        "The send key and receive key must have the same cipher suite.");
>+  }
>+
>+  recv_cipher_suite_ = rtc::SrtpCryptoSuiteFromName(params.cipher_suite);
>+  if (*recv_cipher_suite_ == rtc::SRTP_INVALID_CRYPTO_SUITE) {
>+    return RTCError(RTCErrorType::INVALID_PARAMETER,
>+                    "Invalid SRTP crypto suite");
>+  }
>+
>+  int recv_key_len, recv_salt_len;
>+  if (!rtc::GetSrtpKeyAndSaltLengths(*recv_cipher_suite_, &recv_key_len,
>+                                     &recv_salt_len)) {
>+    return RTCError(RTCErrorType::INVALID_PARAMETER,
>+                    "Could not get lengths for crypto suite(s):"
>+                    " recv cipher_suite ");
>+  }
>+
>+  recv_key_ = rtc::ZeroOnFreeBuffer<uint8_t>(recv_key_len + recv_salt_len);
>+  if (!ParseKeyParams(params.key_params, recv_key_.data(), recv_key_.size())) {
>+    return RTCError(RTCErrorType::INVALID_PARAMETER,
>+                    "Failed to parse the crypto key params");
>+  }
>+
>+  if (!MaybeSetKeyParams()) {
>+    return RTCError(RTCErrorType::INVALID_PARAMETER,
>+                    "Failed to set the crypto key params");
>+  }
>+  recv_params_ = params;
>+  return RTCError::OK();
> }
> 
> bool SrtpTransport::SendRtpPacket(rtc::CopyOnWriteBuffer* packet,
>                                   const rtc::PacketOptions& options,
>                                   int flags) {
>-  return SendPacket(false, packet, options, flags);
>-}
>-
>-bool SrtpTransport::SendRtcpPacket(rtc::CopyOnWriteBuffer* packet,
>-                                   const rtc::PacketOptions& options,
>-                                   int flags) {
>-  return SendPacket(true, packet, options, flags);
>-}
>-
>-bool SrtpTransport::SendPacket(bool rtcp,
>-                               rtc::CopyOnWriteBuffer* packet,
>-                               const rtc::PacketOptions& options,
>-                               int flags) {
>-  if (!IsActive()) {
>+  if (!IsSrtpActive()) {
>     RTC_LOG(LS_ERROR)
>         << "Failed to send the packet because SRTP transport is inactive.";
>     return false;
>   }
>-
>   rtc::PacketOptions updated_options = options;
>-  rtc::CopyOnWriteBuffer cp = *packet;
>   TRACE_EVENT0("webrtc", "SRTP Encode");
>   bool res;
>   uint8_t* data = packet->data();
>-  int len = static_cast<int>(packet->size());
>-  if (!rtcp) {
>+  int len = rtc::checked_cast<int>(packet->size());
> // If ENABLE_EXTERNAL_AUTH flag is on then packet authentication is not done
> // inside libsrtp for a RTP packet. A external HMAC module will be writing
> // a fake HMAC value. This is ONLY done for a RTP packet.
> // Socket layer will update rtp sendtime extension header if present in
> // packet with current time before updating the HMAC.
> #if !defined(ENABLE_EXTERNAL_AUTH)
>-    res = ProtectRtp(data, len, static_cast<int>(packet->capacity()), &len);
>+  res = ProtectRtp(data, len, static_cast<int>(packet->capacity()), &len);
> #else
>-    if (!IsExternalAuthActive()) {
>-      res = ProtectRtp(data, len, static_cast<int>(packet->capacity()), &len);
>-    } else {
>-      updated_options.packet_time_params.rtp_sendtime_extension_id =
>-          rtp_abs_sendtime_extn_id_;
>-      res = ProtectRtp(data, len, static_cast<int>(packet->capacity()), &len,
>-                       &updated_options.packet_time_params.srtp_packet_index);
>-      // If protection succeeds, let's get auth params from srtp.
>+  if (!IsExternalAuthActive()) {
>+    res = ProtectRtp(data, len, static_cast<int>(packet->capacity()), &len);
>+  } else {
>+    updated_options.packet_time_params.rtp_sendtime_extension_id =
>+        rtp_abs_sendtime_extn_id_;
>+    res = ProtectRtp(data, len, static_cast<int>(packet->capacity()), &len,
>+                     &updated_options.packet_time_params.srtp_packet_index);
>+    // If protection succeeds, let's get auth params from srtp.
>+    if (res) {
>+      uint8_t* auth_key = nullptr;
>+      int key_len = 0;
>+      res = GetRtpAuthParams(
>+          &auth_key, &key_len,
>+          &updated_options.packet_time_params.srtp_auth_tag_len);
>       if (res) {
>-        uint8_t* auth_key = NULL;
>-        int key_len;
>-        res = GetRtpAuthParams(
>-            &auth_key, &key_len,
>-            &updated_options.packet_time_params.srtp_auth_tag_len);
>-        if (res) {
>-          updated_options.packet_time_params.srtp_auth_key.resize(key_len);
>-          updated_options.packet_time_params.srtp_auth_key.assign(
>-              auth_key, auth_key + key_len);
>-        }
>+        updated_options.packet_time_params.srtp_auth_key.resize(key_len);
>+        updated_options.packet_time_params.srtp_auth_key.assign(
>+            auth_key, auth_key + key_len);
>       }
>     }
>+  }
> #endif
>-    if (!res) {
>-      int seq_num = -1;
>-      uint32_t ssrc = 0;
>-      cricket::GetRtpSeqNum(data, len, &seq_num);
>-      cricket::GetRtpSsrc(data, len, &ssrc);
>-      RTC_LOG(LS_ERROR) << "Failed to protect RTP packet: size=" << len
>-                        << ", seqnum=" << seq_num << ", SSRC=" << ssrc;
>-      return false;
>-    }
>-  } else {
>-    res = ProtectRtcp(data, len, static_cast<int>(packet->capacity()), &len);
>-    if (!res) {
>-      int type = -1;
>-      cricket::GetRtcpType(data, len, &type);
>-      RTC_LOG(LS_ERROR) << "Failed to protect RTCP packet: size=" << len
>-                        << ", type=" << type;
>-      return false;
>-    }
>+  if (!res) {
>+    int seq_num = -1;
>+    uint32_t ssrc = 0;
>+    cricket::GetRtpSeqNum(data, len, &seq_num);
>+    cricket::GetRtpSsrc(data, len, &ssrc);
>+    RTC_LOG(LS_ERROR) << "Failed to protect RTP packet: size=" << len
>+                      << ", seqnum=" << seq_num << ", SSRC=" << ssrc;
>+    return false;
>   }
> 
>   // Update the length of the packet now that we've added the auth tag.
>   packet->SetSize(len);
>-  return rtcp ? rtp_transport_->SendRtcpPacket(packet, updated_options, flags)
>-              : rtp_transport_->SendRtpPacket(packet, updated_options, flags);
>+  return SendPacket(/*rtcp=*/false, packet, updated_options, flags);
> }
> 
>-void SrtpTransport::OnPacketReceived(bool rtcp,
>-                                     rtc::CopyOnWriteBuffer* packet,
>-                                     const rtc::PacketTime& packet_time) {
>-  if (!IsActive()) {
>+bool SrtpTransport::SendRtcpPacket(rtc::CopyOnWriteBuffer* packet,
>+                                   const rtc::PacketOptions& options,
>+                                   int flags) {
>+  if (!IsSrtpActive()) {
>+    RTC_LOG(LS_ERROR)
>+        << "Failed to send the packet because SRTP transport is inactive.";
>+    return false;
>+  }
>+
>+  TRACE_EVENT0("webrtc", "SRTP Encode");
>+  uint8_t* data = packet->data();
>+  int len = rtc::checked_cast<int>(packet->size());
>+  if (!ProtectRtcp(data, len, static_cast<int>(packet->capacity()), &len)) {
>+    int type = -1;
>+    cricket::GetRtcpType(data, len, &type);
>+    RTC_LOG(LS_ERROR) << "Failed to protect RTCP packet: size=" << len
>+                      << ", type=" << type;
>+    return false;
>+  }
>+  // Update the length of the packet now that we've added the auth tag.
>+  packet->SetSize(len);
>+
>+  return SendPacket(/*rtcp=*/true, packet, options, flags);
>+}
>+
>+void SrtpTransport::OnRtpPacketReceived(rtc::CopyOnWriteBuffer* packet,
>+                                        const rtc::PacketTime& packet_time) {
>+  if (!IsSrtpActive()) {
>     RTC_LOG(LS_WARNING)
>-        << "Inactive SRTP transport received a packet. Drop it.";
>+        << "Inactive SRTP transport received an RTP packet. Drop it.";
>     return;
>   }
>-
>   TRACE_EVENT0("webrtc", "SRTP Decode");
>   char* data = packet->data<char>();
>-  int len = static_cast<int>(packet->size());
>-  bool res;
>-  if (!rtcp) {
>-    res = UnprotectRtp(data, len, &len);
>-    if (!res) {
>-      int seq_num = -1;
>-      uint32_t ssrc = 0;
>-      cricket::GetRtpSeqNum(data, len, &seq_num);
>-      cricket::GetRtpSsrc(data, len, &ssrc);
>-      RTC_LOG(LS_ERROR) << "Failed to unprotect RTP packet: size=" << len
>-                        << ", seqnum=" << seq_num << ", SSRC=" << ssrc;
>-      return;
>-    }
>-  } else {
>-    res = UnprotectRtcp(data, len, &len);
>-    if (!res) {
>-      int type = -1;
>-      cricket::GetRtcpType(data, len, &type);
>-      RTC_LOG(LS_ERROR) << "Failed to unprotect RTCP packet: size=" << len
>-                        << ", type=" << type;
>-      return;
>-    }
>+  int len = rtc::checked_cast<int>(packet->size());
>+  if (!UnprotectRtp(data, len, &len)) {
>+    int seq_num = -1;
>+    uint32_t ssrc = 0;
>+    cricket::GetRtpSeqNum(data, len, &seq_num);
>+    cricket::GetRtpSsrc(data, len, &ssrc);
>+    RTC_LOG(LS_ERROR) << "Failed to unprotect RTP packet: size=" << len
>+                      << ", seqnum=" << seq_num << ", SSRC=" << ssrc;
>+    return;
>   }
>+  packet->SetSize(len);
>+  DemuxPacket(packet, packet_time);
>+}
> 
>+void SrtpTransport::OnRtcpPacketReceived(rtc::CopyOnWriteBuffer* packet,
>+                                         const rtc::PacketTime& packet_time) {
>+  if (!IsSrtpActive()) {
>+    RTC_LOG(LS_WARNING)
>+        << "Inactive SRTP transport received an RTCP packet. Drop it.";
>+    return;
>+  }
>+  TRACE_EVENT0("webrtc", "SRTP Decode");
>+  char* data = packet->data<char>();
>+  int len = rtc::checked_cast<int>(packet->size());
>+  if (!UnprotectRtcp(data, len, &len)) {
>+    int type = -1;
>+    cricket::GetRtcpType(data, len, &type);
>+    RTC_LOG(LS_ERROR) << "Failed to unprotect RTCP packet: size=" << len
>+                      << ", type=" << type;
>+    return;
>+  }
>   packet->SetSize(len);
>-  SignalPacketReceived(rtcp, packet, packet_time);
>+  SignalRtcpPacketReceived(packet, packet_time);
> }
> 
> void SrtpTransport::OnNetworkRouteChanged(
>-
>-    rtc::Optional<rtc::NetworkRoute> network_route) {
>+    absl::optional<rtc::NetworkRoute> network_route) {
>   // Only append the SRTP overhead when there is a selected network route.
>   if (network_route) {
>     int srtp_overhead = 0;
>-    if (IsActive()) {
>+    if (IsSrtpActive()) {
>       GetSrtpOverhead(&srtp_overhead);
>     }
>     network_route->packet_overhead += srtp_overhead;
>@@ -190,6 +249,11 @@ void SrtpTransport::OnNetworkRouteChanged(
>   SignalNetworkRouteChanged(network_route);
> }
> 
>+void SrtpTransport::OnWritableState(
>+    rtc::PacketTransportInternal* packet_transport) {
>+  SignalWritableState(IsWritable(/*rtcp=*/true) && IsWritable(/*rtcp=*/true));
>+}
>+
> bool SrtpTransport::SetRtpParams(int send_cs,
>                                  const uint8_t* send_key,
>                                  int send_key_len,
>@@ -228,9 +292,9 @@ bool SrtpTransport::SetRtpParams(int send_cs,
>   }
> 
>   RTC_LOG(LS_INFO) << "SRTP " << (new_sessions ? "activated" : "updated")
>-                   << " with negotiated parameters:"
>-                   << " send cipher_suite " << send_cs << " recv cipher_suite "
>-                   << recv_cs;
>+                   << " with negotiated parameters: send cipher_suite "
>+                   << send_cs << " recv cipher_suite " << recv_cs;
>+  MaybeUpdateWritableState();
>   return true;
> }
> 
>@@ -262,35 +326,39 @@ bool SrtpTransport::SetRtcpParams(int send_cs,
>   }
> 
>   RTC_LOG(LS_INFO) << "SRTCP activated with negotiated parameters:"
>-                   << " send cipher_suite " << send_cs << " recv cipher_suite "
>-                   << recv_cs;
>-
>+                      " send cipher_suite "
>+                   << send_cs << " recv cipher_suite " << recv_cs;
>+  MaybeUpdateWritableState();
>   return true;
> }
> 
>-bool SrtpTransport::IsActive() const {
>+bool SrtpTransport::IsSrtpActive() const {
>   return send_session_ && recv_session_;
> }
> 
>+bool SrtpTransport::IsWritable(bool rtcp) const {
>+  return IsSrtpActive() && RtpTransport::IsWritable(rtcp);
>+}
>+
> void SrtpTransport::ResetParams() {
>   send_session_ = nullptr;
>   recv_session_ = nullptr;
>   send_rtcp_session_ = nullptr;
>   recv_rtcp_session_ = nullptr;
>+  MaybeUpdateWritableState();
>   RTC_LOG(LS_INFO) << "The params in SRTP transport are reset.";
> }
> 
> void SrtpTransport::CreateSrtpSessions() {
>   send_session_.reset(new cricket::SrtpSession());
>   recv_session_.reset(new cricket::SrtpSession());
>-
>   if (external_auth_enabled_) {
>     send_session_->EnableExternalAuth();
>   }
> }
> 
> bool SrtpTransport::ProtectRtp(void* p, int in_len, int max_len, int* out_len) {
>-  if (!IsActive()) {
>+  if (!IsSrtpActive()) {
>     RTC_LOG(LS_WARNING) << "Failed to ProtectRtp: SRTP not active";
>     return false;
>   }
>@@ -303,7 +371,7 @@ bool SrtpTransport::ProtectRtp(void* p,
>                                int max_len,
>                                int* out_len,
>                                int64_t* index) {
>-  if (!IsActive()) {
>+  if (!IsSrtpActive()) {
>     RTC_LOG(LS_WARNING) << "Failed to ProtectRtp: SRTP not active";
>     return false;
>   }
>@@ -315,7 +383,7 @@ bool SrtpTransport::ProtectRtcp(void* p,
>                                 int in_len,
>                                 int max_len,
>                                 int* out_len) {
>-  if (!IsActive()) {
>+  if (!IsSrtpActive()) {
>     RTC_LOG(LS_WARNING) << "Failed to ProtectRtcp: SRTP not active";
>     return false;
>   }
>@@ -328,7 +396,7 @@ bool SrtpTransport::ProtectRtcp(void* p,
> }
> 
> bool SrtpTransport::UnprotectRtp(void* p, int in_len, int* out_len) {
>-  if (!IsActive()) {
>+  if (!IsSrtpActive()) {
>     RTC_LOG(LS_WARNING) << "Failed to UnprotectRtp: SRTP not active";
>     return false;
>   }
>@@ -337,7 +405,7 @@ bool SrtpTransport::UnprotectRtp(void* p, int in_len, int* out_len) {
> }
> 
> bool SrtpTransport::UnprotectRtcp(void* p, int in_len, int* out_len) {
>-  if (!IsActive()) {
>+  if (!IsSrtpActive()) {
>     RTC_LOG(LS_WARNING) << "Failed to UnprotectRtcp: SRTP not active";
>     return false;
>   }
>@@ -352,7 +420,7 @@ bool SrtpTransport::UnprotectRtcp(void* p, int in_len, int* out_len) {
> bool SrtpTransport::GetRtpAuthParams(uint8_t** key,
>                                      int* key_len,
>                                      int* tag_len) {
>-  if (!IsActive()) {
>+  if (!IsSrtpActive()) {
>     RTC_LOG(LS_WARNING) << "Failed to GetRtpAuthParams: SRTP not active";
>     return false;
>   }
>@@ -362,7 +430,7 @@ bool SrtpTransport::GetRtpAuthParams(uint8_t** key,
> }
> 
> bool SrtpTransport::GetSrtpOverhead(int* srtp_overhead) const {
>-  if (!IsActive()) {
>+  if (!IsSrtpActive()) {
>     RTC_LOG(LS_WARNING) << "Failed to GetSrtpOverhead: SRTP not active";
>     return false;
>   }
>@@ -373,7 +441,7 @@ bool SrtpTransport::GetSrtpOverhead(int* srtp_overhead) const {
> }
> 
> void SrtpTransport::EnableExternalAuth() {
>-  RTC_DCHECK(!IsActive());
>+  RTC_DCHECK(!IsSrtpActive());
>   external_auth_enabled_ = true;
> }
> 
>@@ -382,7 +450,7 @@ bool SrtpTransport::IsExternalAuthEnabled() const {
> }
> 
> bool SrtpTransport::IsExternalAuthActive() const {
>-  if (!IsActive()) {
>+  if (!IsSrtpActive()) {
>     RTC_LOG(LS_WARNING)
>         << "Failed to check IsExternalAuthActive: SRTP not active";
>     return false;
>@@ -392,4 +460,49 @@ bool SrtpTransport::IsExternalAuthActive() const {
>   return send_session_->IsExternalAuthActive();
> }
> 
>+bool SrtpTransport::MaybeSetKeyParams() {
>+  if (!send_cipher_suite_ || !recv_cipher_suite_) {
>+    return true;
>+  }
>+
>+  return SetRtpParams(*send_cipher_suite_, send_key_.data(),
>+                      static_cast<int>(send_key_.size()), std::vector<int>(),
>+                      *recv_cipher_suite_, recv_key_.data(),
>+                      static_cast<int>(recv_key_.size()), std::vector<int>());
>+}
>+
>+bool SrtpTransport::ParseKeyParams(const std::string& key_params,
>+                                   uint8_t* key,
>+                                   size_t len) {
>+  // example key_params: "inline:YUJDZGVmZ2hpSktMbW9QUXJzVHVWd3l6MTIzNDU2"
>+
>+  // Fail if key-method is wrong.
>+  if (key_params.find("inline:") != 0) {
>+    return false;
>+  }
>+
>+  // Fail if base64 decode fails, or the key is the wrong size.
>+  std::string key_b64(key_params.substr(7)), key_str;
>+  if (!rtc::Base64::Decode(key_b64, rtc::Base64::DO_STRICT, &key_str,
>+                           nullptr) ||
>+      key_str.size() != len) {
>+    return false;
>+  }
>+
>+  memcpy(key, key_str.c_str(), len);
>+  // TODO(bugs.webrtc.org/8905): Switch to ZeroOnFreeBuffer for storing
>+  // sensitive data.
>+  rtc::ExplicitZeroMemory(&key_str[0], key_str.size());
>+  return true;
>+}
>+
>+void SrtpTransport::MaybeUpdateWritableState() {
>+  bool writable = IsWritable(/*rtcp=*/true) && IsWritable(/*rtcp=*/false);
>+  // Only fire the signal if the writable state changes.
>+  if (writable_ != writable) {
>+    writable_ = writable;
>+    SignalWritableState(writable_);
>+  }
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtptransport.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtptransport.h
>index d8b0dbd35a5..198024f5a54 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtptransport.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtptransport.h
>@@ -16,21 +16,28 @@
> #include <utility>
> #include <vector>
> 
>+#include "api/ortc/srtptransportinterface.h"
>+#include "p2p/base/dtlstransportinternal.h"
> #include "p2p/base/icetransportinternal.h"
>-#include "pc/rtptransportinternaladapter.h"
>-#include "pc/srtpfilter.h"
>+#include "pc/rtptransport.h"
> #include "pc/srtpsession.h"
>+#include "rtc_base/buffer.h"
> #include "rtc_base/checks.h"
> 
> namespace webrtc {
> 
>-// This class will eventually be a wrapper around RtpTransportInternal
>-// that protects and unprotects sent and received RTP packets.
>-class SrtpTransport : public RtpTransportInternalAdapter {
>+// This subclass of the RtpTransport is used for SRTP which is reponsible for
>+// protecting/unprotecting the packets. It provides interfaces to set the crypto
>+// parameters for the SrtpSession underneath.
>+class SrtpTransport : public RtpTransport {
>  public:
>   explicit SrtpTransport(bool rtcp_mux_enabled);
> 
>-  explicit SrtpTransport(std::unique_ptr<RtpTransportInternal> rtp_transport);
>+  virtual ~SrtpTransport() = default;
>+
>+  // SrtpTransportInterface specific implementation.
>+  RTCError SetSrtpSendKey(const cricket::CryptoParams& params) override;
>+  RTCError SetSrtpReceiveKey(const cricket::CryptoParams& params) override;
> 
>   bool SendRtpPacket(rtc::CopyOnWriteBuffer* packet,
>                      const rtc::PacketOptions& options,
>@@ -42,10 +49,9 @@ class SrtpTransport : public RtpTransportInternalAdapter {
> 
>   // The transport becomes active if the send_session_ and recv_session_ are
>   // created.
>-  bool IsActive() const;
>+  bool IsSrtpActive() const override;
> 
>-  // TODO(zstein): Remove this when we remove RtpTransportAdapter.
>-  RtpTransportAdapter* GetInternal() override { return nullptr; }
>+  bool IsWritable(bool rtcp) const override;
> 
>   // Create new send/recv sessions and set the negotiated crypto keys for RTP
>   // packet encryption. The keys can either come from SDES negotiation or DTLS
>@@ -98,26 +104,23 @@ class SrtpTransport : public RtpTransportInternalAdapter {
>     rtp_abs_sendtime_extn_id_ = rtp_abs_sendtime_extn_id;
>   }
> 
>+ protected:
>+  // If the writable state changed, fire the SignalWritableState.
>+  void MaybeUpdateWritableState();
>+
>  private:
>   void ConnectToRtpTransport();
>   void CreateSrtpSessions();
> 
>-  bool SendPacket(bool rtcp,
>-                  rtc::CopyOnWriteBuffer* packet,
>-                  const rtc::PacketOptions& options,
>-                  int flags);
>-
>-  void OnPacketReceived(bool rtcp,
>-                        rtc::CopyOnWriteBuffer* packet,
>-                        const rtc::PacketTime& packet_time);
>-  void OnReadyToSend(bool ready) { SignalReadyToSend(ready); }
>-  void OnNetworkRouteChanged(rtc::Optional<rtc::NetworkRoute> network_route);
>-
>-  void OnWritableState(bool writable) { SignalWritableState(writable); }
>+  void OnRtpPacketReceived(rtc::CopyOnWriteBuffer* packet,
>+                           const rtc::PacketTime& packet_time) override;
>+  void OnRtcpPacketReceived(rtc::CopyOnWriteBuffer* packet,
>+                            const rtc::PacketTime& packet_time) override;
>+  void OnNetworkRouteChanged(
>+      absl::optional<rtc::NetworkRoute> network_route) override;
> 
>-  void OnSentPacket(const rtc::SentPacket& sent_packet) {
>-    SignalSentPacket(sent_packet);
>-  }
>+  // Override the RtpTransport::OnWritableState.
>+  void OnWritableState(rtc::PacketTransportInternal* packet_transport) override;
> 
>   bool ProtectRtp(void* data, int in_len, int max_len, int* out_len);
> 
>@@ -135,14 +138,25 @@ class SrtpTransport : public RtpTransportInternalAdapter {
> 
>   bool UnprotectRtcp(void* data, int in_len, int* out_len);
> 
>+  bool MaybeSetKeyParams();
>+  bool ParseKeyParams(const std::string& key_params, uint8_t* key, size_t len);
>+
>   const std::string content_name_;
>-  std::unique_ptr<RtpTransportInternal> rtp_transport_;
> 
>   std::unique_ptr<cricket::SrtpSession> send_session_;
>   std::unique_ptr<cricket::SrtpSession> recv_session_;
>   std::unique_ptr<cricket::SrtpSession> send_rtcp_session_;
>   std::unique_ptr<cricket::SrtpSession> recv_rtcp_session_;
> 
>+  absl::optional<cricket::CryptoParams> send_params_;
>+  absl::optional<cricket::CryptoParams> recv_params_;
>+  absl::optional<int> send_cipher_suite_;
>+  absl::optional<int> recv_cipher_suite_;
>+  rtc::ZeroOnFreeBuffer<uint8_t> send_key_;
>+  rtc::ZeroOnFreeBuffer<uint8_t> recv_key_;
>+
>+  bool writable_ = false;
>+
>   bool external_auth_enabled_ = false;
> 
>   int rtp_abs_sendtime_extn_id_ = -1;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtptransport_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtptransport_unittest.cc
>index e0b2302e6ff..2ac64c84f6f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtptransport_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/srtptransport_unittest.cc
>@@ -12,6 +12,7 @@
> 
> #include "pc/srtptransport.h"
> 
>+#include "absl/memory/memory.h"
> #include "media/base/fakertp.h"
> #include "p2p/base/dtlstransportinternal.h"
> #include "p2p/base/fakepackettransport.h"
>@@ -20,7 +21,6 @@
> #include "pc/srtptestutil.h"
> #include "rtc_base/asyncpacketsocket.h"
> #include "rtc_base/gunit.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/sslstreamadapter.h"
> 
> using rtc::kTestKey1;
>@@ -42,50 +42,42 @@ class SrtpTransportTest : public testing::Test, public sigslot::has_slots<> {
>  protected:
>   SrtpTransportTest() {
>     bool rtcp_mux_enabled = true;
>-    auto rtp_transport1 = rtc::MakeUnique<RtpTransport>(rtcp_mux_enabled);
>-    auto rtp_transport2 = rtc::MakeUnique<RtpTransport>(rtcp_mux_enabled);
> 
>     rtp_packet_transport1_ =
>-        rtc::MakeUnique<rtc::FakePacketTransport>("fake_packet_transport1");
>+        absl::make_unique<rtc::FakePacketTransport>("fake_packet_transport1");
>     rtp_packet_transport2_ =
>-        rtc::MakeUnique<rtc::FakePacketTransport>("fake_packet_transport2");
>+        absl::make_unique<rtc::FakePacketTransport>("fake_packet_transport2");
> 
>     bool asymmetric = false;
>     rtp_packet_transport1_->SetDestination(rtp_packet_transport2_.get(),
>                                            asymmetric);
> 
>-    rtp_transport1->SetRtpPacketTransport(rtp_packet_transport1_.get());
>-    rtp_transport2->SetRtpPacketTransport(rtp_packet_transport2_.get());
>+    srtp_transport1_ = absl::make_unique<SrtpTransport>(rtcp_mux_enabled);
>+    srtp_transport2_ = absl::make_unique<SrtpTransport>(rtcp_mux_enabled);
> 
>-    // Add payload type for RTP packet and RTCP packet.
>-    rtp_transport1->AddHandledPayloadType(0x00);
>-    rtp_transport2->AddHandledPayloadType(0x00);
>-    rtp_transport1->AddHandledPayloadType(0xc9);
>-    rtp_transport2->AddHandledPayloadType(0xc9);
>+    srtp_transport1_->SetRtpPacketTransport(rtp_packet_transport1_.get());
>+    srtp_transport2_->SetRtpPacketTransport(rtp_packet_transport2_.get());
> 
>-    srtp_transport1_ =
>-        rtc::MakeUnique<SrtpTransport>(std::move(rtp_transport1));
>-    srtp_transport2_ =
>-        rtc::MakeUnique<SrtpTransport>(std::move(rtp_transport2));
>+    srtp_transport1_->SignalRtcpPacketReceived.connect(
>+        &rtp_sink1_, &TransportObserver::OnRtcpPacketReceived);
>+    srtp_transport2_->SignalRtcpPacketReceived.connect(
>+        &rtp_sink2_, &TransportObserver::OnRtcpPacketReceived);
> 
>-    srtp_transport1_->SignalPacketReceived.connect(
>-        this, &SrtpTransportTest::OnPacketReceived1);
>-    srtp_transport2_->SignalPacketReceived.connect(
>-        this, &SrtpTransportTest::OnPacketReceived2);
>-  }
>+    RtpDemuxerCriteria demuxer_criteria;
>+    // 0x00 is the payload type used in kPcmuFrame.
>+    demuxer_criteria.payload_types = {0x00};
> 
>-  void OnPacketReceived1(bool rtcp,
>-                         rtc::CopyOnWriteBuffer* packet,
>-                         const rtc::PacketTime& packet_time) {
>-    RTC_LOG(LS_INFO) << "SrtpTransport1 Received a packet.";
>-    last_recv_packet1_ = *packet;
>+    srtp_transport1_->RegisterRtpDemuxerSink(demuxer_criteria, &rtp_sink1_);
>+    srtp_transport2_->RegisterRtpDemuxerSink(demuxer_criteria, &rtp_sink2_);
>   }
> 
>-  void OnPacketReceived2(bool rtcp,
>-                         rtc::CopyOnWriteBuffer* packet,
>-                         const rtc::PacketTime& packet_time) {
>-    RTC_LOG(LS_INFO) << "SrtpTransport2 Received a packet.";
>-    last_recv_packet2_ = *packet;
>+  ~SrtpTransportTest() {
>+    if (srtp_transport1_) {
>+      srtp_transport1_->UnregisterRtpDemuxerSink(&rtp_sink1_);
>+    }
>+    if (srtp_transport2_) {
>+      srtp_transport2_->UnregisterRtpDemuxerSink(&rtp_sink2_);
>+    }
>   }
> 
>   // With external auth enabled, SRTP doesn't write the auth tag and
>@@ -142,9 +134,9 @@ class SrtpTransportTest : public testing::Test, public sigslot::has_slots<> {
>     if (srtp_transport1_->IsExternalAuthActive()) {
>       TestRtpAuthParams(srtp_transport1_.get(), cipher_suite_name);
>     } else {
>-      ASSERT_TRUE(last_recv_packet2_.data());
>-      EXPECT_EQ(0,
>-                memcmp(last_recv_packet2_.data(), original_rtp_data, rtp_len));
>+      ASSERT_TRUE(rtp_sink2_.last_recv_rtp_packet().data());
>+      EXPECT_EQ(0, memcmp(rtp_sink2_.last_recv_rtp_packet().data(),
>+                          original_rtp_data, rtp_len));
>       // Get the encrypted packet from underneath packet transport and verify
>       // the data is actually encrypted.
>       auto fake_rtp_packet_transport = static_cast<rtc::FakePacketTransport*>(
>@@ -159,9 +151,9 @@ class SrtpTransportTest : public testing::Test, public sigslot::has_slots<> {
>     if (srtp_transport2_->IsExternalAuthActive()) {
>       TestRtpAuthParams(srtp_transport2_.get(), cipher_suite_name);
>     } else {
>-      ASSERT_TRUE(last_recv_packet1_.data());
>-      EXPECT_EQ(0,
>-                memcmp(last_recv_packet1_.data(), original_rtp_data, rtp_len));
>+      ASSERT_TRUE(rtp_sink1_.last_recv_rtp_packet().data());
>+      EXPECT_EQ(0, memcmp(rtp_sink1_.last_recv_rtp_packet().data(),
>+                          original_rtp_data, rtp_len));
>       auto fake_rtp_packet_transport = static_cast<rtc::FakePacketTransport*>(
>           srtp_transport2_->rtp_packet_transport());
>       EXPECT_NE(0, memcmp(fake_rtp_packet_transport->last_sent_packet()->data(),
>@@ -170,12 +162,12 @@ class SrtpTransportTest : public testing::Test, public sigslot::has_slots<> {
>   }
> 
>   void TestSendRecvRtcpPacket(const std::string& cipher_suite_name) {
>-    size_t rtcp_len = sizeof(kRtcpReport);
>+    size_t rtcp_len = sizeof(::kRtcpReport);
>     size_t packet_size =
>         rtcp_len + 4 + rtc::rtcp_auth_tag_len(cipher_suite_name);
>     rtc::Buffer rtcp_packet_buffer(packet_size);
>     char* rtcp_packet_data = rtcp_packet_buffer.data<char>();
>-    memcpy(rtcp_packet_data, kRtcpReport, rtcp_len);
>+    memcpy(rtcp_packet_data, ::kRtcpReport, rtcp_len);
> 
>     rtc::CopyOnWriteBuffer rtcp_packet1to2(rtcp_packet_data, rtcp_len,
>                                            packet_size);
>@@ -187,8 +179,9 @@ class SrtpTransportTest : public testing::Test, public sigslot::has_slots<> {
>     // that the packet can be successfully received and decrypted.
>     ASSERT_TRUE(srtp_transport1_->SendRtcpPacket(&rtcp_packet1to2, options,
>                                                  cricket::PF_SRTP_BYPASS));
>-    ASSERT_TRUE(last_recv_packet2_.data());
>-    EXPECT_EQ(0, memcmp(last_recv_packet2_.data(), rtcp_packet_data, rtcp_len));
>+    ASSERT_TRUE(rtp_sink2_.last_recv_rtcp_packet().data());
>+    EXPECT_EQ(0, memcmp(rtp_sink2_.last_recv_rtcp_packet().data(),
>+                        rtcp_packet_data, rtcp_len));
>     // Get the encrypted packet from underneath packet transport and verify the
>     // data is actually encrypted.
>     auto fake_rtp_packet_transport = static_cast<rtc::FakePacketTransport*>(
>@@ -199,8 +192,9 @@ class SrtpTransportTest : public testing::Test, public sigslot::has_slots<> {
>     // Do the same thing in the opposite direction;
>     ASSERT_TRUE(srtp_transport2_->SendRtcpPacket(&rtcp_packet2to1, options,
>                                                  cricket::PF_SRTP_BYPASS));
>-    ASSERT_TRUE(last_recv_packet1_.data());
>-    EXPECT_EQ(0, memcmp(last_recv_packet1_.data(), rtcp_packet_data, rtcp_len));
>+    ASSERT_TRUE(rtp_sink1_.last_recv_rtcp_packet().data());
>+    EXPECT_EQ(0, memcmp(rtp_sink1_.last_recv_rtcp_packet().data(),
>+                        rtcp_packet_data, rtcp_len));
>     fake_rtp_packet_transport = static_cast<rtc::FakePacketTransport*>(
>         srtp_transport2_->rtp_packet_transport());
>     EXPECT_NE(0, memcmp(fake_rtp_packet_transport->last_sent_packet()->data(),
>@@ -229,8 +223,8 @@ class SrtpTransportTest : public testing::Test, public sigslot::has_slots<> {
>         cs, key1, key1_len, extension_ids, cs, key2, key2_len, extension_ids));
>     EXPECT_TRUE(srtp_transport2_->SetRtcpParams(
>         cs, key2, key2_len, extension_ids, cs, key1, key1_len, extension_ids));
>-    EXPECT_TRUE(srtp_transport1_->IsActive());
>-    EXPECT_TRUE(srtp_transport2_->IsActive());
>+    EXPECT_TRUE(srtp_transport1_->IsSrtpActive());
>+    EXPECT_TRUE(srtp_transport2_->IsSrtpActive());
>     if (rtc::IsGcmCryptoSuite(cs)) {
>       EXPECT_FALSE(srtp_transport1_->IsExternalAuthActive());
>       EXPECT_FALSE(srtp_transport2_->IsExternalAuthActive());
>@@ -267,8 +261,9 @@ class SrtpTransportTest : public testing::Test, public sigslot::has_slots<> {
>     // that the packet can be successfully received and decrypted.
>     ASSERT_TRUE(srtp_transport1_->SendRtpPacket(&rtp_packet1to2, options,
>                                                 cricket::PF_SRTP_BYPASS));
>-    ASSERT_TRUE(last_recv_packet2_.data());
>-    EXPECT_EQ(0, memcmp(last_recv_packet2_.data(), original_rtp_data, rtp_len));
>+    ASSERT_TRUE(rtp_sink2_.last_recv_rtp_packet().data());
>+    EXPECT_EQ(0, memcmp(rtp_sink2_.last_recv_rtp_packet().data(),
>+                        original_rtp_data, rtp_len));
>     // Get the encrypted packet from underneath packet transport and verify the
>     // data and header extension are actually encrypted.
>     auto fake_rtp_packet_transport = static_cast<rtc::FakePacketTransport*>(
>@@ -284,8 +279,9 @@ class SrtpTransportTest : public testing::Test, public sigslot::has_slots<> {
>     // Do the same thing in the opposite direction;
>     ASSERT_TRUE(srtp_transport2_->SendRtpPacket(&rtp_packet2to1, options,
>                                                 cricket::PF_SRTP_BYPASS));
>-    ASSERT_TRUE(last_recv_packet1_.data());
>-    EXPECT_EQ(0, memcmp(last_recv_packet1_.data(), original_rtp_data, rtp_len));
>+    ASSERT_TRUE(rtp_sink1_.last_recv_rtp_packet().data());
>+    EXPECT_EQ(0, memcmp(rtp_sink1_.last_recv_rtp_packet().data(),
>+                        original_rtp_data, rtp_len));
>     fake_rtp_packet_transport = static_cast<rtc::FakePacketTransport*>(
>         srtp_transport2_->rtp_packet_transport());
>     EXPECT_NE(0, memcmp(fake_rtp_packet_transport->last_sent_packet()->data(),
>@@ -315,8 +311,8 @@ class SrtpTransportTest : public testing::Test, public sigslot::has_slots<> {
>     EXPECT_TRUE(srtp_transport2_->SetRtpParams(cs, key2, key2_len,
>                                                encrypted_headers, cs, key1,
>                                                key1_len, encrypted_headers));
>-    EXPECT_TRUE(srtp_transport1_->IsActive());
>-    EXPECT_TRUE(srtp_transport2_->IsActive());
>+    EXPECT_TRUE(srtp_transport1_->IsSrtpActive());
>+    EXPECT_TRUE(srtp_transport2_->IsSrtpActive());
>     EXPECT_FALSE(srtp_transport1_->IsExternalAuthActive());
>     EXPECT_FALSE(srtp_transport2_->IsExternalAuthActive());
>     TestSendRecvPacketWithEncryptedHeaderExtension(cs_name, encrypted_headers);
>@@ -328,8 +324,9 @@ class SrtpTransportTest : public testing::Test, public sigslot::has_slots<> {
>   std::unique_ptr<rtc::FakePacketTransport> rtp_packet_transport1_;
>   std::unique_ptr<rtc::FakePacketTransport> rtp_packet_transport2_;
> 
>-  rtc::CopyOnWriteBuffer last_recv_packet1_;
>-  rtc::CopyOnWriteBuffer last_recv_packet2_;
>+  TransportObserver rtp_sink1_;
>+  TransportObserver rtp_sink2_;
>+
>   int sequence_number_ = 0;
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/statscollector.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/statscollector.cc
>index a8d187ee163..23a449308c3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/statscollector.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/statscollector.cc
>@@ -11,13 +11,14 @@
> #include "pc/statscollector.h"
> 
> #include <memory>
>+#include <set>
> #include <utility>
> #include <vector>
> 
> #include "pc/channel.h"
> #include "pc/peerconnection.h"
>-#include "rtc_base/base64.h"
> #include "rtc_base/checks.h"
>+#include "rtc_base/third_party/base64/base64.h"
> 
> namespace webrtc {
> namespace {
>@@ -38,8 +39,9 @@ const char* STATSREPORT_ADAPTER_TYPE_WIFI = "wlan";
> const char* STATSREPORT_ADAPTER_TYPE_WWAN = "wwan";
> const char* STATSREPORT_ADAPTER_TYPE_VPN = "vpn";
> const char* STATSREPORT_ADAPTER_TYPE_LOOPBACK = "loopback";
>+const char* STATSREPORT_ADAPTER_TYPE_WILDCARD = "wildcard";
> 
>-template<typename ValueType>
>+template <typename ValueType>
> struct TypeForAdd {
>   const StatsReport::StatsValueName name;
>   const ValueType& value;
>@@ -132,37 +134,36 @@ void SetAudioProcessingStats(StatsReport* report,
> void ExtractStats(const cricket::VoiceReceiverInfo& info, StatsReport* report) {
>   ExtractCommonReceiveProperties(info, report);
>   const FloatForAdd floats[] = {
>-    { StatsReport::kStatsValueNameExpandRate, info.expand_rate },
>-    { StatsReport::kStatsValueNameSecondaryDecodedRate,
>-      info.secondary_decoded_rate },
>-    { StatsReport::kStatsValueNameSecondaryDiscardedRate,
>-      info.secondary_discarded_rate },
>-    { StatsReport::kStatsValueNameSpeechExpandRate, info.speech_expand_rate },
>-    { StatsReport::kStatsValueNameAccelerateRate, info.accelerate_rate },
>-    { StatsReport::kStatsValueNamePreemptiveExpandRate,
>-      info.preemptive_expand_rate },
>-    { StatsReport::kStatsValueNameTotalAudioEnergy, info.total_output_energy },
>-    { StatsReport::kStatsValueNameTotalSamplesDuration,
>-      info.total_output_duration }
>-  };
>+      {StatsReport::kStatsValueNameExpandRate, info.expand_rate},
>+      {StatsReport::kStatsValueNameSecondaryDecodedRate,
>+       info.secondary_decoded_rate},
>+      {StatsReport::kStatsValueNameSecondaryDiscardedRate,
>+       info.secondary_discarded_rate},
>+      {StatsReport::kStatsValueNameSpeechExpandRate, info.speech_expand_rate},
>+      {StatsReport::kStatsValueNameAccelerateRate, info.accelerate_rate},
>+      {StatsReport::kStatsValueNamePreemptiveExpandRate,
>+       info.preemptive_expand_rate},
>+      {StatsReport::kStatsValueNameTotalAudioEnergy, info.total_output_energy},
>+      {StatsReport::kStatsValueNameTotalSamplesDuration,
>+       info.total_output_duration}};
> 
>   const IntForAdd ints[] = {
>-    { StatsReport::kStatsValueNameCurrentDelayMs, info.delay_estimate_ms },
>-    { StatsReport::kStatsValueNameDecodingCNG, info.decoding_cng },
>-    { StatsReport::kStatsValueNameDecodingCTN, info.decoding_calls_to_neteq },
>-    { StatsReport::kStatsValueNameDecodingCTSG,
>-      info.decoding_calls_to_silence_generator },
>-    { StatsReport::kStatsValueNameDecodingMutedOutput,
>-      info.decoding_muted_output },
>-    { StatsReport::kStatsValueNameDecodingNormal, info.decoding_normal },
>-    { StatsReport::kStatsValueNameDecodingPLC, info.decoding_plc },
>-    { StatsReport::kStatsValueNameDecodingPLCCNG, info.decoding_plc_cng },
>-    { StatsReport::kStatsValueNameJitterBufferMs, info.jitter_buffer_ms },
>-    { StatsReport::kStatsValueNameJitterReceived, info.jitter_ms },
>-    { StatsReport::kStatsValueNamePacketsLost, info.packets_lost },
>-    { StatsReport::kStatsValueNamePacketsReceived, info.packets_rcvd },
>-    { StatsReport::kStatsValueNamePreferredJitterBufferMs,
>-      info.jitter_buffer_preferred_ms },
>+      {StatsReport::kStatsValueNameCurrentDelayMs, info.delay_estimate_ms},
>+      {StatsReport::kStatsValueNameDecodingCNG, info.decoding_cng},
>+      {StatsReport::kStatsValueNameDecodingCTN, info.decoding_calls_to_neteq},
>+      {StatsReport::kStatsValueNameDecodingCTSG,
>+       info.decoding_calls_to_silence_generator},
>+      {StatsReport::kStatsValueNameDecodingMutedOutput,
>+       info.decoding_muted_output},
>+      {StatsReport::kStatsValueNameDecodingNormal, info.decoding_normal},
>+      {StatsReport::kStatsValueNameDecodingPLC, info.decoding_plc},
>+      {StatsReport::kStatsValueNameDecodingPLCCNG, info.decoding_plc_cng},
>+      {StatsReport::kStatsValueNameJitterBufferMs, info.jitter_buffer_ms},
>+      {StatsReport::kStatsValueNameJitterReceived, info.jitter_ms},
>+      {StatsReport::kStatsValueNamePacketsLost, info.packets_lost},
>+      {StatsReport::kStatsValueNamePacketsReceived, info.packets_rcvd},
>+      {StatsReport::kStatsValueNamePreferredJitterBufferMs,
>+       info.jitter_buffer_preferred_ms},
>   };
> 
>   for (const auto& f : floats)
>@@ -175,8 +176,7 @@ void ExtractStats(const cricket::VoiceReceiverInfo& info, StatsReport* report) {
>                    info.audio_level);
>   }
> 
>-  report->AddInt64(StatsReport::kStatsValueNameBytesReceived,
>-                   info.bytes_rcvd);
>+  report->AddInt64(StatsReport::kStatsValueNameBytesReceived, info.bytes_rcvd);
>   if (info.capture_start_ntp_time_ms >= 0) {
>     report->AddInt64(StatsReport::kStatsValueNameCaptureStartNtpTimeMs,
>                      info.capture_start_ntp_time_ms);
>@@ -191,17 +191,16 @@ void ExtractStats(const cricket::VoiceSenderInfo& info, StatsReport* report) {
>                           info.apm_statistics);
> 
>   const FloatForAdd floats[] = {
>-    { StatsReport::kStatsValueNameTotalAudioEnergy, info.total_input_energy },
>-    { StatsReport::kStatsValueNameTotalSamplesDuration,
>-      info.total_input_duration }
>-  };
>+      {StatsReport::kStatsValueNameTotalAudioEnergy, info.total_input_energy},
>+      {StatsReport::kStatsValueNameTotalSamplesDuration,
>+       info.total_input_duration}};
> 
>   RTC_DCHECK_GE(info.audio_level, 0);
>   const IntForAdd ints[] = {
>-    { StatsReport::kStatsValueNameAudioInputLevel, info.audio_level},
>-    { StatsReport::kStatsValueNameJitterReceived, info.jitter_ms },
>-    { StatsReport::kStatsValueNamePacketsLost, info.packets_lost },
>-    { StatsReport::kStatsValueNamePacketsSent, info.packets_sent },
>+      {StatsReport::kStatsValueNameAudioInputLevel, info.audio_level},
>+      {StatsReport::kStatsValueNameJitterReceived, info.jitter_ms},
>+      {StatsReport::kStatsValueNamePacketsLost, info.packets_lost},
>+      {StatsReport::kStatsValueNamePacketsSent, info.packets_sent},
>   };
> 
>   for (const auto& f : floats) {
>@@ -248,8 +247,7 @@ void ExtractStats(const cricket::VideoReceiverInfo& info, StatsReport* report) {
>   ExtractCommonReceiveProperties(info, report);
>   report->AddString(StatsReport::kStatsValueNameCodecImplementationName,
>                     info.decoder_implementation_name);
>-  report->AddInt64(StatsReport::kStatsValueNameBytesReceived,
>-                   info.bytes_rcvd);
>+  report->AddInt64(StatsReport::kStatsValueNameBytesReceived, info.bytes_rcvd);
>   if (info.capture_start_ntp_time_ms >= 0) {
>     report->AddInt64(StatsReport::kStatsValueNameCaptureStartNtpTimeMs,
>                      info.capture_start_ntp_time_ms);
>@@ -258,25 +256,25 @@ void ExtractStats(const cricket::VideoReceiverInfo& info, StatsReport* report) {
>     report->AddInt64(StatsReport::kStatsValueNameQpSum, *info.qp_sum);
> 
>   const IntForAdd ints[] = {
>-    { StatsReport::kStatsValueNameCurrentDelayMs, info.current_delay_ms },
>-    { StatsReport::kStatsValueNameDecodeMs, info.decode_ms },
>-    { StatsReport::kStatsValueNameFirsSent, info.firs_sent },
>-    { StatsReport::kStatsValueNameFrameHeightReceived, info.frame_height },
>-    { StatsReport::kStatsValueNameFrameRateDecoded, info.framerate_decoded },
>-    { StatsReport::kStatsValueNameFrameRateOutput, info.framerate_output },
>-    { StatsReport::kStatsValueNameFrameRateReceived, info.framerate_rcvd },
>-    { StatsReport::kStatsValueNameFrameWidthReceived, info.frame_width },
>-    { StatsReport::kStatsValueNameJitterBufferMs, info.jitter_buffer_ms },
>-    { StatsReport::kStatsValueNameMaxDecodeMs, info.max_decode_ms },
>-    { StatsReport::kStatsValueNameMinPlayoutDelayMs,
>-      info.min_playout_delay_ms },
>-    { StatsReport::kStatsValueNameNacksSent, info.nacks_sent },
>-    { StatsReport::kStatsValueNamePacketsLost, info.packets_lost },
>-    { StatsReport::kStatsValueNamePacketsReceived, info.packets_rcvd },
>-    { StatsReport::kStatsValueNamePlisSent, info.plis_sent },
>-    { StatsReport::kStatsValueNameRenderDelayMs, info.render_delay_ms },
>-    { StatsReport::kStatsValueNameTargetDelayMs, info.target_delay_ms },
>-    { StatsReport::kStatsValueNameFramesDecoded, info.frames_decoded },
>+      {StatsReport::kStatsValueNameCurrentDelayMs, info.current_delay_ms},
>+      {StatsReport::kStatsValueNameDecodeMs, info.decode_ms},
>+      {StatsReport::kStatsValueNameFirsSent, info.firs_sent},
>+      {StatsReport::kStatsValueNameFrameHeightReceived, info.frame_height},
>+      {StatsReport::kStatsValueNameFrameRateDecoded, info.framerate_decoded},
>+      {StatsReport::kStatsValueNameFrameRateOutput, info.framerate_output},
>+      {StatsReport::kStatsValueNameFrameRateReceived, info.framerate_rcvd},
>+      {StatsReport::kStatsValueNameFrameWidthReceived, info.frame_width},
>+      {StatsReport::kStatsValueNameJitterBufferMs, info.jitter_buffer_ms},
>+      {StatsReport::kStatsValueNameMaxDecodeMs, info.max_decode_ms},
>+      {StatsReport::kStatsValueNameMinPlayoutDelayMs,
>+       info.min_playout_delay_ms},
>+      {StatsReport::kStatsValueNameNacksSent, info.nacks_sent},
>+      {StatsReport::kStatsValueNamePacketsLost, info.packets_lost},
>+      {StatsReport::kStatsValueNamePacketsReceived, info.packets_rcvd},
>+      {StatsReport::kStatsValueNamePlisSent, info.plis_sent},
>+      {StatsReport::kStatsValueNameRenderDelayMs, info.render_delay_ms},
>+      {StatsReport::kStatsValueNameTargetDelayMs, info.target_delay_ms},
>+      {StatsReport::kStatsValueNameFramesDecoded, info.frames_decoded},
>   };
> 
>   for (const auto& i : ints)
>@@ -312,20 +310,21 @@ void ExtractStats(const cricket::VideoSenderInfo& info, StatsReport* report) {
>     report->AddInt(StatsReport::kStatsValueNameQpSum, *info.qp_sum);
> 
>   const IntForAdd ints[] = {
>-    { StatsReport::kStatsValueNameAdaptationChanges, info.adapt_changes },
>-    { StatsReport::kStatsValueNameAvgEncodeMs, info.avg_encode_ms },
>-    { StatsReport::kStatsValueNameEncodeUsagePercent,
>-      info.encode_usage_percent },
>-    { StatsReport::kStatsValueNameFirsReceived, info.firs_rcvd },
>-    { StatsReport::kStatsValueNameFrameHeightSent, info.send_frame_height },
>-    { StatsReport::kStatsValueNameFrameRateInput, info.framerate_input },
>-    { StatsReport::kStatsValueNameFrameRateSent, info.framerate_sent },
>-    { StatsReport::kStatsValueNameFrameWidthSent, info.send_frame_width },
>-    { StatsReport::kStatsValueNameNacksReceived, info.nacks_rcvd },
>-    { StatsReport::kStatsValueNamePacketsLost, info.packets_lost },
>-    { StatsReport::kStatsValueNamePacketsSent, info.packets_sent },
>-    { StatsReport::kStatsValueNamePlisReceived, info.plis_rcvd },
>-    { StatsReport::kStatsValueNameFramesEncoded, info.frames_encoded },
>+      {StatsReport::kStatsValueNameAdaptationChanges, info.adapt_changes},
>+      {StatsReport::kStatsValueNameAvgEncodeMs, info.avg_encode_ms},
>+      {StatsReport::kStatsValueNameEncodeUsagePercent,
>+       info.encode_usage_percent},
>+      {StatsReport::kStatsValueNameFirsReceived, info.firs_rcvd},
>+      {StatsReport::kStatsValueNameFrameHeightSent, info.send_frame_height},
>+      {StatsReport::kStatsValueNameFrameRateInput, info.framerate_input},
>+      {StatsReport::kStatsValueNameFrameRateSent, info.framerate_sent},
>+      {StatsReport::kStatsValueNameFrameWidthSent, info.send_frame_width},
>+      {StatsReport::kStatsValueNameNacksReceived, info.nacks_rcvd},
>+      {StatsReport::kStatsValueNamePacketsLost, info.packets_lost},
>+      {StatsReport::kStatsValueNamePacketsSent, info.packets_sent},
>+      {StatsReport::kStatsValueNamePlisReceived, info.plis_rcvd},
>+      {StatsReport::kStatsValueNameFramesEncoded, info.frames_encoded},
>+      {StatsReport::kStatsValueNameHugeFramesSent, info.huge_frames_sent},
>   };
> 
>   for (const auto& i : ints)
>@@ -343,14 +342,14 @@ void ExtractStats(const cricket::BandwidthEstimationInfo& info,
> 
>   report->set_timestamp(stats_gathering_started);
>   const IntForAdd ints[] = {
>-    { StatsReport::kStatsValueNameAvailableSendBandwidth,
>-      info.available_send_bandwidth },
>-    { StatsReport::kStatsValueNameAvailableReceiveBandwidth,
>-      info.available_recv_bandwidth },
>-    { StatsReport::kStatsValueNameTargetEncBitrate, info.target_enc_bitrate },
>-    { StatsReport::kStatsValueNameActualEncBitrate, info.actual_enc_bitrate },
>-    { StatsReport::kStatsValueNameRetransmitBitrate, info.retransmit_bitrate },
>-    { StatsReport::kStatsValueNameTransmitBitrate, info.transmit_bitrate },
>+      {StatsReport::kStatsValueNameAvailableSendBandwidth,
>+       info.available_send_bandwidth},
>+      {StatsReport::kStatsValueNameAvailableReceiveBandwidth,
>+       info.available_recv_bandwidth},
>+      {StatsReport::kStatsValueNameTargetEncBitrate, info.target_enc_bitrate},
>+      {StatsReport::kStatsValueNameActualEncBitrate, info.actual_enc_bitrate},
>+      {StatsReport::kStatsValueNameRetransmitBitrate, info.retransmit_bitrate},
>+      {StatsReport::kStatsValueNameTransmitBitrate, info.transmit_bitrate},
>   };
>   for (const auto& i : ints)
>     report->AddInt(i.name, i.value);
>@@ -373,7 +372,7 @@ void ExtractRemoteStats(const cricket::MediaReceiverInfo& info,
> // In order to use the template, the functions that are called from it,
> // ExtractStats and ExtractRemoteStats, must be defined and overloaded
> // for each type.
>-template<typename T>
>+template <typename T>
> void ExtractStatsFromList(const std::vector<T>& data,
>                           const StatsReport::Id& transport_id,
>                           StatsCollector* collector,
>@@ -382,8 +381,8 @@ void ExtractStatsFromList(const std::vector<T>& data,
>     uint32_t ssrc = d.ssrc();
>     // Each track can have stats for both local and remote objects.
>     // TODO(hta): Handle the case of multiple SSRCs per object.
>-    StatsReport* report = collector->PrepareReport(true, ssrc, transport_id,
>-                                                   direction);
>+    StatsReport* report =
>+        collector->PrepareReport(true, ssrc, transport_id, direction);
>     if (report)
>       ExtractStats(d, report);
> 
>@@ -428,13 +427,15 @@ const char* AdapterTypeToStatsType(rtc::AdapterType type) {
>       return STATSREPORT_ADAPTER_TYPE_VPN;
>     case rtc::ADAPTER_TYPE_LOOPBACK:
>       return STATSREPORT_ADAPTER_TYPE_LOOPBACK;
>+    case rtc::ADAPTER_TYPE_ANY:
>+      return STATSREPORT_ADAPTER_TYPE_WILDCARD;
>     default:
>       RTC_NOTREACHED();
>       return "";
>   }
> }
> 
>-StatsCollector::StatsCollector(PeerConnection* pc)
>+StatsCollector::StatsCollector(PeerConnectionInternal* pc)
>     : pc_(pc), stats_gathering_started_(0) {
>   RTC_DCHECK(pc_);
> }
>@@ -445,8 +446,7 @@ StatsCollector::~StatsCollector() {
> 
> // Wallclock time in ms.
> double StatsCollector::GetTimeNow() {
>-  return rtc::TimeUTCMicros() /
>-         static_cast<double>(rtc::kNumMicrosecsPerMillisec);
>+  return static_cast<double>(rtc::TimeUTCMillis());
> }
> 
> // Adds a MediaStream with tracks that can be used as a |selector| in a call
>@@ -527,8 +527,8 @@ void StatsCollector::GetStats(MediaStreamTrackInterface* track,
>   if (report)
>     reports->push_back(report);
> 
>-  report = reports_.Find(StatsReport::NewTypedId(
>-      StatsReport::kStatsReportTypeTrack, track->id()));
>+  report = reports_.Find(
>+      StatsReport::NewTypedId(StatsReport::kStatsReportTypeTrack, track->id()));
> 
>   if (!report)
>     return;
>@@ -547,8 +547,8 @@ void StatsCollector::GetStats(MediaStreamTrackInterface* track,
>   }
> }
> 
>-void
>-StatsCollector::UpdateStats(PeerConnectionInterface::StatsOutputLevel level) {
>+void StatsCollector::UpdateStats(
>+    PeerConnectionInterface::StatsOutputLevel level) {
>   RTC_DCHECK(pc_->signaling_thread()->IsCurrent());
>   double time_now = GetTimeNow();
>   // Calls to UpdateStats() that occur less than kMinGatherStatsPeriod number of
>@@ -568,40 +568,35 @@ StatsCollector::UpdateStats(PeerConnectionInterface::StatsOutputLevel level) {
>   // the same thread (this class has no locks right now).
>   ExtractSessionInfo();
>   ExtractBweInfo();
>-  ExtractVoiceInfo();
>-  ExtractVideoInfo(level);
>+  ExtractMediaInfo();
>   ExtractSenderInfo();
>   ExtractDataInfo();
>   UpdateTrackReports();
> }
> 
>-StatsReport* StatsCollector::PrepareReport(
>-    bool local,
>-    uint32_t ssrc,
>-    const StatsReport::Id& transport_id,
>-    StatsReport::Direction direction) {
>+StatsReport* StatsCollector::PrepareReport(bool local,
>+                                           uint32_t ssrc,
>+                                           const StatsReport::Id& transport_id,
>+                                           StatsReport::Direction direction) {
>   RTC_DCHECK(pc_->signaling_thread()->IsCurrent());
>   StatsReport::Id id(StatsReport::NewIdWithDirection(
>       local ? StatsReport::kStatsReportTypeSsrc
>             : StatsReport::kStatsReportTypeRemoteSsrc,
>-      rtc::ToString<uint32_t>(ssrc), direction));
>+      rtc::ToString(ssrc), direction));
>   StatsReport* report = reports_.Find(id);
> 
>   // Use the ID of the track that is currently mapped to the SSRC, if any.
>   std::string track_id;
>   if (!GetTrackIdBySsrc(ssrc, &track_id, direction)) {
>-    if (!report) {
>-      // The ssrc is not used by any track or existing report, return NULL
>-      // in such case to indicate no report is prepared for the ssrc.
>-      return NULL;
>+    // The SSRC is not used by any existing track (or lookup failed since the
>+    // SSRC wasn't signaled in SDP). Try copying the track ID from a previous
>+    // report: if one exists.
>+    if (report) {
>+      const StatsReport::Value* v =
>+          report->FindValue(StatsReport::kStatsValueNameTrackId);
>+      if (v)
>+        track_id = v->string_val();
>     }
>-
>-    // The ssrc is not used by any existing track. Keeps the old track id
>-    // since we want to report the stats for inactive ssrc.
>-    const StatsReport::Value* v =
>-        report->FindValue(StatsReport::kStatsValueNameTrackId);
>-    if (v)
>-      track_id = v->string_val();
>   }
> 
>   if (!report)
>@@ -611,7 +606,9 @@ StatsReport* StatsCollector::PrepareReport(
>   report->set_timestamp(stats_gathering_started_);
> 
>   report->AddInt64(StatsReport::kStatsValueNameSsrc, ssrc);
>-  report->AddString(StatsReport::kStatsValueNameTrackId, track_id);
>+  if (!track_id.empty()) {
>+    report->AddString(StatsReport::kStatsValueNameTrackId, track_id);
>+  }
>   // Add the mapping of SSRC to transport.
>   report->AddId(StatsReport::kStatsValueNameTransportId, transport_id);
>   return report;
>@@ -623,14 +620,12 @@ bool StatsCollector::IsValidTrack(const std::string& track_id) {
> }
> 
> StatsReport* StatsCollector::AddCertificateReports(
>-    const rtc::SSLCertificate* cert) {
>+    std::unique_ptr<rtc::SSLCertificateStats> cert_stats) {
>   RTC_DCHECK(pc_->signaling_thread()->IsCurrent());
>-  RTC_DCHECK(cert != NULL);
> 
>-  std::unique_ptr<rtc::SSLCertificateStats> first_stats = cert->GetStats();
>   StatsReport* first_report = nullptr;
>   StatsReport* prev_report = nullptr;
>-  for (rtc::SSLCertificateStats* stats = first_stats.get(); stats;
>+  for (rtc::SSLCertificateStats* stats = cert_stats.get(); stats;
>        stats = stats->issuer.get()) {
>     StatsReport::Id id(StatsReport::NewTypedId(
>         StatsReport::kStatsReportTypeCertificate, stats->fingerprint));
>@@ -653,27 +648,31 @@ StatsReport* StatsCollector::AddCertificateReports(
> }
> 
> StatsReport* StatsCollector::AddConnectionInfoReport(
>-    const std::string& content_name, int component, int connection_id,
>+    const std::string& content_name,
>+    int component,
>+    int connection_id,
>     const StatsReport::Id& channel_report_id,
>     const cricket::ConnectionInfo& info) {
>-  StatsReport::Id id(StatsReport::NewCandidatePairId(content_name, component,
>-                                                     connection_id));
>+  StatsReport::Id id(
>+      StatsReport::NewCandidatePairId(content_name, component, connection_id));
>   StatsReport* report = reports_.ReplaceOrAddNew(id);
>   report->set_timestamp(stats_gathering_started_);
> 
>   const BoolForAdd bools[] = {
>-    {StatsReport::kStatsValueNameActiveConnection, info.best_connection},
>-    {StatsReport::kStatsValueNameReceiving, info.receiving},
>-    {StatsReport::kStatsValueNameWritable, info.writable},
>+      {StatsReport::kStatsValueNameActiveConnection, info.best_connection},
>+      {StatsReport::kStatsValueNameReceiving, info.receiving},
>+      {StatsReport::kStatsValueNameWritable, info.writable},
>   };
>   for (const auto& b : bools)
>     report->AddBoolean(b.name, b.value);
> 
>   report->AddId(StatsReport::kStatsValueNameChannelId, channel_report_id);
>+  cricket::CandidateStats local_candidate_stats(info.local_candidate);
>+  cricket::CandidateStats remote_candidate_stats(info.remote_candidate);
>   report->AddId(StatsReport::kStatsValueNameLocalCandidateId,
>-                AddCandidateReport(info.local_candidate, true)->id());
>+                AddCandidateReport(local_candidate_stats, true)->id());
>   report->AddId(StatsReport::kStatsValueNameRemoteCandidateId,
>-                AddCandidateReport(info.remote_candidate, false)->id());
>+                AddCandidateReport(remote_candidate_stats, false)->id());
> 
>   const Int64ForAdd int64s[] = {
>       {StatsReport::kStatsValueNameBytesReceived, info.recv_total_bytes},
>@@ -708,8 +707,9 @@ StatsReport* StatsCollector::AddConnectionInfoReport(
> }
> 
> StatsReport* StatsCollector::AddCandidateReport(
>-    const cricket::Candidate& candidate,
>+    const cricket::CandidateStats& candidate_stats,
>     bool local) {
>+  const auto& candidate = candidate_stats.candidate;
>   StatsReport::Id id(StatsReport::NewCandidateId(local, candidate.id()));
>   StatsReport* report = reports_.Find(id);
>   if (!report) {
>@@ -731,6 +731,18 @@ StatsReport* StatsCollector::AddCandidateReport(
>                       candidate.protocol());
>   }
> 
>+  if (local && candidate_stats.stun_stats.has_value()) {
>+    const auto& stun_stats = candidate_stats.stun_stats.value();
>+    report->AddInt64(StatsReport::kStatsValueNameSentStunKeepaliveRequests,
>+                     stun_stats.stun_binding_requests_sent);
>+    report->AddInt64(StatsReport::kStatsValueNameRecvStunKeepaliveResponses,
>+                     stun_stats.stun_binding_responses_received);
>+    report->AddFloat(StatsReport::kStatsValueNameStunKeepaliveRttTotal,
>+                     stun_stats.stun_binding_rtt_ms_total);
>+    report->AddFloat(StatsReport::kStatsValueNameStunKeepaliveRttSquaredTotal,
>+                     stun_stats.stun_binding_rtt_ms_squared_total);
>+  }
>+
>   return report;
> }
> 
>@@ -745,36 +757,49 @@ void StatsCollector::ExtractSessionInfo() {
>   report->AddBoolean(StatsReport::kStatsValueNameInitiator,
>                      pc_->initial_offerer());
> 
>-  std::unique_ptr<SessionStats> stats = pc_->GetSessionStats_s();
>-  if (!stats) {
>-    return;
>+  cricket::CandidateStatsList pooled_candidate_stats_list =
>+      pc_->GetPooledCandidateStats();
>+
>+  for (const cricket::CandidateStats& stats : pooled_candidate_stats_list) {
>+    AddCandidateReport(stats, true);
>   }
> 
>-  for (const auto& transport_iter : stats->transport_stats) {
>+  std::set<std::string> transport_names;
>+  for (const auto& entry : pc_->GetTransportNamesByMid()) {
>+    transport_names.insert(entry.second);
>+  }
>+
>+  std::map<std::string, cricket::TransportStats> transport_stats_by_name =
>+      pc_->GetTransportStatsByNames(transport_names);
>+
>+  for (const auto& entry : transport_stats_by_name) {
>+    const std::string& transport_name = entry.first;
>+    const cricket::TransportStats& transport_stats = entry.second;
>+
>     // Attempt to get a copy of the certificates from the transport and
>     // expose them in stats reports.  All channels in a transport share the
>     // same local and remote certificates.
>     //
>     StatsReport::Id local_cert_report_id, remote_cert_report_id;
>     rtc::scoped_refptr<rtc::RTCCertificate> certificate;
>-    if (pc_->GetLocalCertificate(transport_iter.second.transport_name,
>-                                 &certificate)) {
>-      StatsReport* r = AddCertificateReports(&(certificate->ssl_certificate()));
>+    if (pc_->GetLocalCertificate(transport_name, &certificate)) {
>+      StatsReport* r =
>+          AddCertificateReports(certificate->ssl_cert_chain().GetStats());
>       if (r)
>         local_cert_report_id = r->id();
>     }
> 
>-    std::unique_ptr<rtc::SSLCertificate> cert =
>-        pc_->GetRemoteSSLCertificate(transport_iter.second.transport_name);
>-    if (cert) {
>-      StatsReport* r = AddCertificateReports(cert.get());
>+    std::unique_ptr<rtc::SSLCertChain> remote_cert_chain =
>+        pc_->GetRemoteSSLCertChain(transport_name);
>+    if (remote_cert_chain) {
>+      StatsReport* r = AddCertificateReports(remote_cert_chain->GetStats());
>       if (r)
>         remote_cert_report_id = r->id();
>     }
> 
>-    for (const auto& channel_iter : transport_iter.second.channel_stats) {
>-      StatsReport::Id id(StatsReport::NewComponentId(
>-          transport_iter.second.transport_name, channel_iter.component));
>+    for (const auto& channel_iter : transport_stats.channel_stats) {
>+      StatsReport::Id id(
>+          StatsReport::NewComponentId(transport_name, channel_iter.component));
>       StatsReport* channel_report = reports_.ReplaceOrAddNew(id);
>       channel_report->set_timestamp(stats_gathering_started_);
>       channel_report->AddInt(StatsReport::kStatsValueNameComponent,
>@@ -803,11 +828,21 @@ void StatsCollector::ExtractSessionInfo() {
>             rtc::SSLStreamAdapter::SslCipherSuiteToName(ssl_cipher_suite));
>       }
> 
>+      // Collect stats for non-pooled candidates. Note that the reports
>+      // generated here supersedes the candidate reports generated in
>+      // AddConnectionInfoReport below, and they may report candidates that are
>+      // not paired. Also, the candidate report generated in
>+      // AddConnectionInfoReport do not report port stats like StunStats.
>+      for (const cricket::CandidateStats& stats :
>+           channel_iter.candidate_stats_list) {
>+        AddCandidateReport(stats, true);
>+      }
>+
>       int connection_id = 0;
>       for (const cricket::ConnectionInfo& info :
>-               channel_iter.connection_infos) {
>+           channel_iter.connection_infos) {
>         StatsReport* connection_report = AddConnectionInfoReport(
>-            transport_iter.first, channel_iter.component, connection_id++,
>+            transport_name, channel_iter.component, connection_id++,
>             channel_report->id(), info);
>         if (info.best_connection) {
>           channel_report->AddId(
>@@ -830,69 +865,120 @@ void StatsCollector::ExtractBweInfo() {
>   bwe_info.available_send_bandwidth = call_stats.send_bandwidth_bps;
>   bwe_info.available_recv_bandwidth = call_stats.recv_bandwidth_bps;
>   bwe_info.bucket_delay = call_stats.pacer_delay_ms;
>+
>   // Fill in target encoder bitrate, actual encoder bitrate, rtx bitrate, etc.
>   // TODO(holmer): Also fill this in for audio.
>-  if (pc_->video_channel()) {
>-    pc_->video_channel()->FillBitrateInfo(&bwe_info);
>+  for (auto transceiver : pc_->GetTransceiversInternal()) {
>+    if (transceiver->media_type() != cricket::MEDIA_TYPE_VIDEO) {
>+      continue;
>+    }
>+    auto* video_channel =
>+        static_cast<cricket::VideoChannel*>(transceiver->internal()->channel());
>+    if (!video_channel) {
>+      continue;
>+    }
>+    video_channel->FillBitrateInfo(&bwe_info);
>   }
>+
>   StatsReport::Id report_id(StatsReport::NewBandwidthEstimationId());
>   StatsReport* report = reports_.FindOrAddNew(report_id);
>   ExtractStats(bwe_info, stats_gathering_started_, report);
> }
> 
>-void StatsCollector::ExtractVoiceInfo() {
>-  RTC_DCHECK(pc_->signaling_thread()->IsCurrent());
>-
>-  if (!pc_->voice_channel()) {
>-    return;
>-  }
>-  cricket::VoiceMediaInfo voice_info;
>-  if (!pc_->voice_channel()->GetStats(&voice_info)) {
>-    RTC_LOG(LS_ERROR) << "Failed to get voice channel stats.";
>-    return;
>-  }
>+namespace {
> 
>-  // TODO(tommi): The above code should run on the worker thread and post the
>-  // results back to the signaling thread, where we can add data to the reports.
>-  rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
>+struct VoiceChannelStatsInfo {
>+  std::string transport_name;
>+  cricket::VoiceMediaChannel* voice_media_channel;
>+  cricket::VoiceMediaInfo voice_media_info;
>+};
> 
>-  StatsReport::Id transport_id =
>-      StatsReport::NewComponentId(pc_->voice_channel()->transport_name(),
>-                                  cricket::ICE_CANDIDATE_COMPONENT_RTP);
>+struct VideoChannelStatsInfo {
>+  std::string transport_name;
>+  cricket::VideoMediaChannel* video_media_channel;
>+  cricket::VideoMediaInfo video_media_info;
>+};
> 
>-  ExtractStatsFromList(voice_info.receivers, transport_id, this,
>-      StatsReport::kReceive);
>-  ExtractStatsFromList(voice_info.senders, transport_id, this,
>-      StatsReport::kSend);
>+}  // namespace
> 
>-  UpdateStatsFromExistingLocalAudioTracks(voice_info.receivers.size() > 0);
>-}
>+void StatsCollector::ExtractMediaInfo() {
>+  RTC_DCHECK_RUN_ON(pc_->signaling_thread());
> 
>-void StatsCollector::ExtractVideoInfo(
>-    PeerConnectionInterface::StatsOutputLevel level) {
>-  RTC_DCHECK(pc_->signaling_thread()->IsCurrent());
>+  std::vector<VoiceChannelStatsInfo> voice_channel_infos;
>+  std::vector<VideoChannelStatsInfo> video_channel_infos;
> 
>-  if (!pc_->video_channel()) {
>-    return;
>-  }
>-  cricket::VideoMediaInfo video_info;
>-  if (!pc_->video_channel()->GetStats(&video_info)) {
>-    RTC_LOG(LS_ERROR) << "Failed to get video channel stats.";
>-    return;
>+  {
>+    rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
>+    for (auto transceiver : pc_->GetTransceiversInternal()) {
>+      if (!transceiver->internal()->channel()) {
>+        continue;
>+      }
>+      cricket::MediaType media_type = transceiver->internal()->media_type();
>+      if (media_type == cricket::MEDIA_TYPE_AUDIO) {
>+        auto* voice_channel = static_cast<cricket::VoiceChannel*>(
>+            transceiver->internal()->channel());
>+        voice_channel_infos.emplace_back();
>+        VoiceChannelStatsInfo& info = voice_channel_infos.back();
>+        info.transport_name = voice_channel->transport_name();
>+        info.voice_media_channel = voice_channel->media_channel();
>+      } else {
>+        RTC_DCHECK_EQ(media_type, cricket::MEDIA_TYPE_VIDEO);
>+        auto* video_channel = static_cast<cricket::VideoChannel*>(
>+            transceiver->internal()->channel());
>+        video_channel_infos.emplace_back();
>+        VideoChannelStatsInfo& info = video_channel_infos.back();
>+        info.transport_name = video_channel->transport_name();
>+        info.video_media_channel = video_channel->media_channel();
>+      }
>+    }
>   }
> 
>-  // TODO(tommi): The above code should run on the worker thread and post the
>-  // results back to the signaling thread, where we can add data to the reports.
>+  pc_->worker_thread()->Invoke<void>(RTC_FROM_HERE, [&] {
>+    rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
>+    for (auto it = voice_channel_infos.begin(); it != voice_channel_infos.end();
>+         /* incremented manually */) {
>+      if (!it->voice_media_channel->GetStats(&it->voice_media_info)) {
>+        RTC_LOG(LS_ERROR) << "Failed to get voice channel stats";
>+        it = voice_channel_infos.erase(it);
>+        continue;
>+      }
>+      ++it;
>+    }
>+    for (auto it = video_channel_infos.begin(); it != video_channel_infos.end();
>+         /* incremented manually */) {
>+      if (!it->video_media_channel->GetStats(&it->video_media_info)) {
>+        RTC_LOG(LS_ERROR) << "Failed to get video channel stats";
>+        it = video_channel_infos.erase(it);
>+        continue;
>+      }
>+      ++it;
>+    }
>+  });
>+
>   rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
> 
>-  StatsReport::Id transport_id =
>-      StatsReport::NewComponentId(pc_->video_channel()->transport_name(),
>-                                  cricket::ICE_CANDIDATE_COMPONENT_RTP);
>+  bool has_remote_audio = false;
>+  for (const auto& info : voice_channel_infos) {
>+    StatsReport::Id transport_id = StatsReport::NewComponentId(
>+        info.transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
>+    ExtractStatsFromList(info.voice_media_info.receivers, transport_id, this,
>+                         StatsReport::kReceive);
>+    ExtractStatsFromList(info.voice_media_info.senders, transport_id, this,
>+                         StatsReport::kSend);
>+    if (!info.voice_media_info.receivers.empty()) {
>+      has_remote_audio = true;
>+    }
>+  }
>+  for (const auto& info : video_channel_infos) {
>+    StatsReport::Id transport_id = StatsReport::NewComponentId(
>+        info.transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
>+    ExtractStatsFromList(info.video_media_info.receivers, transport_id, this,
>+                         StatsReport::kReceive);
>+    ExtractStatsFromList(info.video_media_info.senders, transport_id, this,
>+                         StatsReport::kSend);
>+  }
> 
>-  ExtractStatsFromList(video_info.receivers, transport_id, this,
>-      StatsReport::kReceive);
>-  ExtractStatsFromList(video_info.senders, transport_id, this,
>-      StatsReport::kSend);
>+  UpdateStatsFromExistingLocalAudioTracks(has_remote_audio);
> }
> 
> void StatsCollector::ExtractSenderInfo() {
>@@ -918,8 +1004,8 @@ void StatsCollector::ExtractSenderInfo() {
>       continue;
>     }
>     const StatsReport::Id stats_id = StatsReport::NewIdWithDirection(
>-        StatsReport::kStatsReportTypeSsrc,
>-        rtc::ToString<uint32_t>(sender->ssrc()), StatsReport::kSend);
>+        StatsReport::kStatsReportTypeSsrc, rtc::ToString(sender->ssrc()),
>+        StatsReport::kSend);
>     StatsReport* report = reports_.FindOrAddNew(stats_id);
>     report->AddInt(StatsReport::kStatsValueNameFrameWidthInput,
>                    stats.input_width);
>@@ -965,9 +1051,8 @@ void StatsCollector::UpdateStatsFromExistingLocalAudioTracks(
>   for (const auto& it : local_audio_tracks_) {
>     AudioTrackInterface* track = it.first;
>     uint32_t ssrc = it.second;
>-    StatsReport* report =
>-        GetReport(StatsReport::kStatsReportTypeSsrc,
>-                  rtc::ToString<uint32_t>(ssrc), StatsReport::kSend);
>+    StatsReport* report = GetReport(StatsReport::kStatsReportTypeSsrc,
>+                                    rtc::ToString(ssrc), StatsReport::kSend);
>     if (report == NULL) {
>       // This can happen if a local audio track is added to a stream on the
>       // fly and the report has not been set up yet. Do nothing in this case.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/statscollector.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/statscollector.h
>index 1a3abfc9aaf..a3c5992afc6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/statscollector.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/statscollector.h
>@@ -15,6 +15,7 @@
> #define PC_STATSCOLLECTOR_H_
> 
> #include <map>
>+#include <memory>
> #include <string>
> #include <utility>
> #include <vector>
>@@ -22,11 +23,10 @@
> #include "api/mediastreaminterface.h"
> #include "api/peerconnectioninterface.h"
> #include "api/statstypes.h"
>+#include "pc/peerconnectioninternal.h"
> 
> namespace webrtc {
> 
>-class PeerConnection;
>-
> // Conversion function to convert candidate type string to the corresponding one
> // from  enum RTCStatsIceCandidateType.
> const char* IceCandidateTypeToStatsType(const std::string& candidate_type);
>@@ -43,7 +43,7 @@ class StatsCollector {
>  public:
>   // The caller is responsible for ensuring that the pc outlives the
>   // StatsCollector instance.
>-  explicit StatsCollector(PeerConnection* pc);
>+  explicit StatsCollector(PeerConnectionInternal* pc);
>   virtual ~StatsCollector();
> 
>   // Adds a MediaStream with tracks that can be used as a |selector| in a call
>@@ -69,8 +69,7 @@ class StatsCollector {
>   // of filling in |reports|.  As is, there's a requirement that the caller
>   // uses |reports| immediately without allowing any async activity on
>   // the thread (message handling etc) and then discard the results.
>-  void GetStats(MediaStreamTrackInterface* track,
>-                StatsReports* reports);
>+  void GetStats(MediaStreamTrackInterface* track, StatsReports* reports);
> 
>   // Prepare a local or remote SSRC report for the given ssrc. Used internally
>   // in the ExtractStatsFromList template.
>@@ -97,25 +96,26 @@ class StatsCollector {
> 
>   // Helper method for creating IceCandidate report. |is_local| indicates
>   // whether this candidate is local or remote.
>-  StatsReport* AddCandidateReport(const cricket::Candidate& candidate,
>-                                  bool local);
>+  StatsReport* AddCandidateReport(
>+      const cricket::CandidateStats& candidate_stats,
>+      bool local);
> 
>   // Adds a report for this certificate and every certificate in its chain, and
>-  // returns the leaf certificate's report (|cert|'s report).
>-  StatsReport* AddCertificateReports(const rtc::SSLCertificate* cert);
>+  // returns the leaf certificate's report (|cert_stats|'s report).
>+  StatsReport* AddCertificateReports(
>+      std::unique_ptr<rtc::SSLCertificateStats> cert_stats);
> 
>   StatsReport* AddConnectionInfoReport(const std::string& content_name,
>-      int component, int connection_id,
>-      const StatsReport::Id& channel_report_id,
>-      const cricket::ConnectionInfo& info);
>+                                       int component,
>+                                       int connection_id,
>+                                       const StatsReport::Id& channel_report_id,
>+                                       const cricket::ConnectionInfo& info);
> 
>   void ExtractDataInfo();
>   void ExtractSessionInfo();
>   void ExtractBweInfo();
>-  void ExtractVoiceInfo();
>-  void ExtractVideoInfo(PeerConnectionInterface::StatsOutputLevel level);
>+  void ExtractMediaInfo();
>   void ExtractSenderInfo();
>-  void BuildSsrcToTransportId();
>   webrtc::StatsReport* GetReport(const StatsReport::StatsType& type,
>                                  const std::string& id,
>                                  StatsReport::Direction direction);
>@@ -139,7 +139,7 @@ class StatsCollector {
>   StatsCollection reports_;
>   TrackIdMap track_ids_;
>   // Raw pointer to the peer connection the statistics are gathered from.
>-  PeerConnection* const pc_;
>+  PeerConnectionInternal* const pc_;
>   double stats_gathering_started_;
> 
>   // TODO(tommi): We appear to be holding on to raw pointers to reference
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/statscollector_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/statscollector_unittest.cc
>index 4973fd3f73f..fa4d3db81bd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/statscollector_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/statscollector_unittest.cc
>@@ -16,47 +16,24 @@
> 
> #include "pc/statscollector.h"
> 
>-#include "api/mediastreaminterface.h"
>-#include "media/base/fakemediaengine.h"
>-#include "media/base/test/mock_mediachannel.h"
>-#include "pc/channelmanager.h"
> #include "pc/mediastream.h"
> #include "pc/mediastreamtrack.h"
>-#include "pc/peerconnection.h"
>-#include "pc/peerconnectionfactory.h"
>-#include "pc/test/fakedatachannelprovider.h"
>+#include "pc/test/fakepeerconnectionforstats.h"
> #include "pc/test/fakevideotracksource.h"
>-#include "pc/test/mock_peerconnection.h"
> #include "pc/videotrack.h"
>-#include "rtc_base/base64.h"
> #include "rtc_base/fakesslidentity.h"
>-#include "rtc_base/gunit.h"
>-#include "rtc_base/stringencode.h"
>-#include "test/gmock.h"
>+#include "rtc_base/third_party/base64/base64.h"
> #include "test/gtest.h"
> 
>-using testing::_;
>-using testing::DoAll;
>-using testing::Field;
>-using testing::Invoke;
>-using testing::Return;
>-using testing::ReturnNull;
>-using testing::ReturnRef;
>-using testing::SetArgPointee;
>-using webrtc::PeerConnectionInterface;
>-using webrtc::StatsReport;
>-using webrtc::StatsReports;
>-
>-namespace {
>-const bool kDefaultRtcpMuxRequired = true;
>-const bool kDefaultSrtpRequired = true;
>-}
>-
>-namespace cricket {
>-
>-class ChannelManager;
>-
>-}  // namespace cricket
>+using cricket::ConnectionInfo;
>+using cricket::SsrcReceiverInfo;
>+using cricket::TransportChannelStats;
>+using cricket::VideoMediaInfo;
>+using cricket::VideoReceiverInfo;
>+using cricket::VideoSenderInfo;
>+using cricket::VoiceMediaInfo;
>+using cricket::VoiceReceiverInfo;
>+using cricket::VoiceSenderInfo;
> 
> namespace webrtc {
> 
>@@ -73,7 +50,7 @@ const char kLocalTrackId[] = "local_track_id";
> const char kRemoteTrackId[] = "remote_track_id";
> const uint32_t kSsrcOfTrack = 1234;
> 
>-class FakeAudioProcessor : public webrtc::AudioProcessorInterface {
>+class FakeAudioProcessor : public AudioProcessorInterface {
>  public:
>   FakeAudioProcessor() {}
>   ~FakeAudioProcessor() {}
>@@ -101,22 +78,20 @@ class FakeAudioProcessor : public webrtc::AudioProcessorInterface {
>   }
> };
> 
>-class FakeAudioTrack
>-    : public webrtc::MediaStreamTrack<webrtc::AudioTrackInterface> {
>+class FakeAudioTrack : public MediaStreamTrack<AudioTrackInterface> {
>  public:
>   explicit FakeAudioTrack(const std::string& id)
>-      : webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>(id),
>+      : MediaStreamTrack<AudioTrackInterface>(id),
>         processor_(new rtc::RefCountedObject<FakeAudioProcessor>()) {}
>   std::string kind() const override { return "audio"; }
>-  webrtc::AudioSourceInterface* GetSource() const override { return NULL; }
>-  void AddSink(webrtc::AudioTrackSinkInterface* sink) override {}
>-  void RemoveSink(webrtc::AudioTrackSinkInterface* sink) override {}
>+  AudioSourceInterface* GetSource() const override { return NULL; }
>+  void AddSink(AudioTrackSinkInterface* sink) override {}
>+  void RemoveSink(AudioTrackSinkInterface* sink) override {}
>   bool GetSignalLevel(int* level) override {
>     *level = 1;
>     return true;
>   }
>-  rtc::scoped_refptr<webrtc::AudioProcessorInterface> GetAudioProcessor()
>-      override {
>+  rtc::scoped_refptr<AudioProcessorInterface> GetAudioProcessor() override {
>     return processor_;
>   }
> 
>@@ -126,7 +101,7 @@ class FakeAudioTrack
> 
> // This fake audio processor is used to verify that the undesired initial values
> // (-1) will be filtered out.
>-class FakeAudioProcessorWithInitValue : public webrtc::AudioProcessorInterface {
>+class FakeAudioProcessorWithInitValue : public AudioProcessorInterface {
>  public:
>   FakeAudioProcessorWithInitValue() {}
>   ~FakeAudioProcessorWithInitValue() {}
>@@ -149,22 +124,21 @@ class FakeAudioProcessorWithInitValue : public webrtc::AudioProcessorInterface {
> };
> 
> class FakeAudioTrackWithInitValue
>-    : public webrtc::MediaStreamTrack<webrtc::AudioTrackInterface> {
>+    : public MediaStreamTrack<AudioTrackInterface> {
>  public:
>   explicit FakeAudioTrackWithInitValue(const std::string& id)
>-      : webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>(id),
>+      : MediaStreamTrack<AudioTrackInterface>(id),
>         processor_(
>             new rtc::RefCountedObject<FakeAudioProcessorWithInitValue>()) {}
>   std::string kind() const override { return "audio"; }
>-  webrtc::AudioSourceInterface* GetSource() const override { return NULL; }
>-  void AddSink(webrtc::AudioTrackSinkInterface* sink) override {}
>-  void RemoveSink(webrtc::AudioTrackSinkInterface* sink) override {}
>+  AudioSourceInterface* GetSource() const override { return NULL; }
>+  void AddSink(AudioTrackSinkInterface* sink) override {}
>+  void RemoveSink(AudioTrackSinkInterface* sink) override {}
>   bool GetSignalLevel(int* level) override {
>     *level = 1;
>     return true;
>   }
>-  rtc::scoped_refptr<webrtc::AudioProcessorInterface> GetAudioProcessor()
>-      override {
>+  rtc::scoped_refptr<AudioProcessorInterface> GetAudioProcessor() override {
>     return processor_;
>   }
> 
>@@ -220,8 +194,9 @@ StatsReport::Id IdFromCertIdString(const std::string& cert_id) {
> 
> // Finds the |n|-th report of type |type| in |reports|.
> // |n| starts from 1 for finding the first report.
>-const StatsReport* FindNthReportByType(
>-    const StatsReports& reports, const StatsReport::StatsType& type, int n) {
>+const StatsReport* FindNthReportByType(const StatsReports& reports,
>+                                       const StatsReport::StatsType& type,
>+                                       int n) {
>   for (size_t i = 0; i < reports.size(); ++i) {
>     if (reports[i]->type() == type) {
>       n--;
>@@ -248,19 +223,16 @@ std::string ExtractSsrcStatsValue(const StatsReports& reports,
> 
> std::string ExtractBweStatsValue(const StatsReports& reports,
>                                  StatsReport::StatsValueName name) {
>-  return ExtractStatsValue(
>-      StatsReport::kStatsReportTypeBwe, reports, name);
>+  return ExtractStatsValue(StatsReport::kStatsReportTypeBwe, reports, name);
> }
> 
> std::string DerToPem(const std::string& der) {
>   return rtc::SSLIdentity::DerToPem(
>-        rtc::kPemTypeCertificate,
>-        reinterpret_cast<const unsigned char*>(der.c_str()),
>-        der.length());
>+      rtc::kPemTypeCertificate,
>+      reinterpret_cast<const unsigned char*>(der.c_str()), der.length());
> }
> 
>-std::vector<std::string> DersToPems(
>-    const std::vector<std::string>& ders) {
>+std::vector<std::string> DersToPems(const std::vector<std::string>& ders) {
>   std::vector<std::string> pems(ders.size());
>   std::transform(ders.begin(), ders.end(), pems.begin(), DerToPem);
>   return pems;
>@@ -277,16 +249,14 @@ void CheckCertChainReports(const StatsReports& reports,
>     ASSERT_TRUE(report != NULL);
> 
>     std::string der_base64;
>-    EXPECT_TRUE(GetValue(
>-        report, StatsReport::kStatsValueNameDer, &der_base64));
>+    EXPECT_TRUE(GetValue(report, StatsReport::kStatsValueNameDer, &der_base64));
>     std::string der = rtc::Base64::Decode(der_base64, rtc::Base64::DO_STRICT);
>     EXPECT_EQ(ders[i], der);
> 
>     std::string fingerprint_algorithm;
>-    EXPECT_TRUE(GetValue(
>-        report,
>-        StatsReport::kStatsValueNameFingerprintAlgorithm,
>-        &fingerprint_algorithm));
>+    EXPECT_TRUE(GetValue(report,
>+                         StatsReport::kStatsValueNameFingerprintAlgorithm,
>+                         &fingerprint_algorithm));
>     // The digest algorithm for a FakeSSLCertificate is always SHA-1.
>     std::string sha_1_str = rtc::DIGEST_SHA_1;
>     EXPECT_EQ(sha_1_str, fingerprint_algorithm);
>@@ -298,8 +268,7 @@ void CheckCertChainReports(const StatsReports& reports,
> 
>     ++i;
>     std::string issuer_id;
>-    if (!GetValue(report, StatsReport::kStatsValueNameIssuerId,
>-                  &issuer_id)) {
>+    if (!GetValue(report, StatsReport::kStatsValueNameIssuerId, &issuer_id)) {
>       break;
>     }
> 
>@@ -309,109 +278,104 @@ void CheckCertChainReports(const StatsReports& reports,
>   EXPECT_EQ(ders.size(), i);
> }
> 
>-void VerifyVoiceReceiverInfoReport(
>-    const StatsReport* report,
>-    const cricket::VoiceReceiverInfo& info) {
>+void VerifyVoiceReceiverInfoReport(const StatsReport* report,
>+                                   const cricket::VoiceReceiverInfo& info) {
>   std::string value_in_report;
>-  EXPECT_TRUE(GetValue(
>-      report, StatsReport::kStatsValueNameAudioOutputLevel, &value_in_report));
>-  EXPECT_EQ(rtc::ToString<int>(info.audio_level), value_in_report);
>-  EXPECT_TRUE(GetValue(
>-      report, StatsReport::kStatsValueNameBytesReceived, &value_in_report));
>-  EXPECT_EQ(rtc::ToString<int64_t>(info.bytes_rcvd), value_in_report);
>-  EXPECT_TRUE(GetValue(
>-      report, StatsReport::kStatsValueNameJitterReceived, &value_in_report));
>-  EXPECT_EQ(rtc::ToString<int>(info.jitter_ms), value_in_report);
>-  EXPECT_TRUE(GetValue(
>-      report, StatsReport::kStatsValueNameJitterBufferMs, &value_in_report));
>-  EXPECT_EQ(rtc::ToString<int>(info.jitter_buffer_ms), value_in_report);
>-  EXPECT_TRUE(GetValue(
>-      report, StatsReport::kStatsValueNamePreferredJitterBufferMs,
>-      &value_in_report));
>-  EXPECT_EQ(rtc::ToString<int>(info.jitter_buffer_preferred_ms),
>-      value_in_report);
>-  EXPECT_TRUE(GetValue(
>-      report, StatsReport::kStatsValueNameCurrentDelayMs, &value_in_report));
>-  EXPECT_EQ(rtc::ToString<int>(info.delay_estimate_ms), value_in_report);
>-  EXPECT_TRUE(GetValue(
>-      report, StatsReport::kStatsValueNameExpandRate, &value_in_report));
>-  EXPECT_EQ(rtc::ToString<float>(info.expand_rate), value_in_report);
>-  EXPECT_TRUE(GetValue(
>-      report, StatsReport::kStatsValueNameSpeechExpandRate, &value_in_report));
>-  EXPECT_EQ(rtc::ToString<float>(info.speech_expand_rate), value_in_report);
>+  EXPECT_TRUE(GetValue(report, StatsReport::kStatsValueNameAudioOutputLevel,
>+                       &value_in_report));
>+  EXPECT_EQ(rtc::ToString(info.audio_level), value_in_report);
>+  EXPECT_TRUE(GetValue(report, StatsReport::kStatsValueNameBytesReceived,
>+                       &value_in_report));
>+  EXPECT_EQ(rtc::ToString(info.bytes_rcvd), value_in_report);
>+  EXPECT_TRUE(GetValue(report, StatsReport::kStatsValueNameJitterReceived,
>+                       &value_in_report));
>+  EXPECT_EQ(rtc::ToString(info.jitter_ms), value_in_report);
>+  EXPECT_TRUE(GetValue(report, StatsReport::kStatsValueNameJitterBufferMs,
>+                       &value_in_report));
>+  EXPECT_EQ(rtc::ToString(info.jitter_buffer_ms), value_in_report);
>+  EXPECT_TRUE(GetValue(report,
>+                       StatsReport::kStatsValueNamePreferredJitterBufferMs,
>+                       &value_in_report));
>+  EXPECT_EQ(rtc::ToString(info.jitter_buffer_preferred_ms), value_in_report);
>+  EXPECT_TRUE(GetValue(report, StatsReport::kStatsValueNameCurrentDelayMs,
>+                       &value_in_report));
>+  EXPECT_EQ(rtc::ToString(info.delay_estimate_ms), value_in_report);
>+  EXPECT_TRUE(GetValue(report, StatsReport::kStatsValueNameExpandRate,
>+                       &value_in_report));
>+  EXPECT_EQ(rtc::ToString(info.expand_rate), value_in_report);
>+  EXPECT_TRUE(GetValue(report, StatsReport::kStatsValueNameSpeechExpandRate,
>+                       &value_in_report));
>+  EXPECT_EQ(rtc::ToString(info.speech_expand_rate), value_in_report);
>   EXPECT_TRUE(GetValue(report, StatsReport::kStatsValueNameAccelerateRate,
>                        &value_in_report));
>-  EXPECT_EQ(rtc::ToString<float>(info.accelerate_rate), value_in_report);
>+  EXPECT_EQ(rtc::ToString(info.accelerate_rate), value_in_report);
>   EXPECT_TRUE(GetValue(report, StatsReport::kStatsValueNamePreemptiveExpandRate,
>                        &value_in_report));
>-  EXPECT_EQ(rtc::ToString<float>(info.preemptive_expand_rate), value_in_report);
>+  EXPECT_EQ(rtc::ToString(info.preemptive_expand_rate), value_in_report);
>   EXPECT_TRUE(GetValue(report, StatsReport::kStatsValueNameSecondaryDecodedRate,
>                        &value_in_report));
>-  EXPECT_EQ(rtc::ToString<float>(info.secondary_decoded_rate), value_in_report);
>+  EXPECT_EQ(rtc::ToString(info.secondary_decoded_rate), value_in_report);
>   EXPECT_TRUE(GetValue(report,
>                        StatsReport::kStatsValueNameSecondaryDiscardedRate,
>                        &value_in_report));
>-  EXPECT_EQ(rtc::ToString<float>(info.secondary_discarded_rate),
>+  EXPECT_EQ(rtc::ToString(info.secondary_discarded_rate), value_in_report);
>+  EXPECT_TRUE(GetValue(report, StatsReport::kStatsValueNamePacketsReceived,
>+                       &value_in_report));
>+  EXPECT_EQ(rtc::ToString(info.packets_rcvd), value_in_report);
>+  EXPECT_TRUE(GetValue(report, StatsReport::kStatsValueNameDecodingCTSG,
>+                       &value_in_report));
>+  EXPECT_EQ(rtc::ToString(info.decoding_calls_to_silence_generator),
>             value_in_report);
>-  EXPECT_TRUE(GetValue(
>-      report, StatsReport::kStatsValueNamePacketsReceived, &value_in_report));
>-  EXPECT_EQ(rtc::ToString<int>(info.packets_rcvd), value_in_report);
>-  EXPECT_TRUE(GetValue(
>-      report, StatsReport::kStatsValueNameDecodingCTSG, &value_in_report));
>-  EXPECT_EQ(rtc::ToString<int>(info.decoding_calls_to_silence_generator),
>-      value_in_report);
>-  EXPECT_TRUE(GetValue(
>-      report, StatsReport::kStatsValueNameDecodingCTN, &value_in_report));
>-  EXPECT_EQ(rtc::ToString<int>(info.decoding_calls_to_neteq),
>-      value_in_report);
>-  EXPECT_TRUE(GetValue(
>-      report, StatsReport::kStatsValueNameDecodingNormal, &value_in_report));
>-  EXPECT_EQ(rtc::ToString<int>(info.decoding_normal), value_in_report);
>-  EXPECT_TRUE(GetValue(
>-      report, StatsReport::kStatsValueNameDecodingPLC, &value_in_report));
>-  EXPECT_EQ(rtc::ToString<int>(info.decoding_plc), value_in_report);
>-  EXPECT_TRUE(GetValue(
>-      report, StatsReport::kStatsValueNameDecodingCNG, &value_in_report));
>-  EXPECT_EQ(rtc::ToString<int>(info.decoding_cng), value_in_report);
>-  EXPECT_TRUE(GetValue(
>-      report, StatsReport::kStatsValueNameDecodingPLCCNG, &value_in_report));
>-  EXPECT_EQ(rtc::ToString<int>(info.decoding_plc_cng), value_in_report);
>+  EXPECT_TRUE(GetValue(report, StatsReport::kStatsValueNameDecodingCTN,
>+                       &value_in_report));
>+  EXPECT_EQ(rtc::ToString(info.decoding_calls_to_neteq), value_in_report);
>+  EXPECT_TRUE(GetValue(report, StatsReport::kStatsValueNameDecodingNormal,
>+                       &value_in_report));
>+  EXPECT_EQ(rtc::ToString(info.decoding_normal), value_in_report);
>+  EXPECT_TRUE(GetValue(report, StatsReport::kStatsValueNameDecodingPLC,
>+                       &value_in_report));
>+  EXPECT_EQ(rtc::ToString(info.decoding_plc), value_in_report);
>+  EXPECT_TRUE(GetValue(report, StatsReport::kStatsValueNameDecodingCNG,
>+                       &value_in_report));
>+  EXPECT_EQ(rtc::ToString(info.decoding_cng), value_in_report);
>+  EXPECT_TRUE(GetValue(report, StatsReport::kStatsValueNameDecodingPLCCNG,
>+                       &value_in_report));
>+  EXPECT_EQ(rtc::ToString(info.decoding_plc_cng), value_in_report);
>   EXPECT_TRUE(GetValue(report, StatsReport::kStatsValueNameDecodingMutedOutput,
>                        &value_in_report));
>-  EXPECT_EQ(rtc::ToString<int>(info.decoding_muted_output), value_in_report);
>+  EXPECT_EQ(rtc::ToString(info.decoding_muted_output), value_in_report);
>   EXPECT_TRUE(GetValue(report, StatsReport::kStatsValueNameCodecName,
>                        &value_in_report));
> }
> 
>-
> void VerifyVoiceSenderInfoReport(const StatsReport* report,
>                                  const cricket::VoiceSenderInfo& sinfo) {
>   std::string value_in_report;
>-  EXPECT_TRUE(GetValue(
>-      report, StatsReport::kStatsValueNameCodecName, &value_in_report));
>+  EXPECT_TRUE(GetValue(report, StatsReport::kStatsValueNameCodecName,
>+                       &value_in_report));
>   EXPECT_EQ(sinfo.codec_name, value_in_report);
>-  EXPECT_TRUE(GetValue(
>-      report, StatsReport::kStatsValueNameBytesSent, &value_in_report));
>-  EXPECT_EQ(rtc::ToString<int64_t>(sinfo.bytes_sent), value_in_report);
>-  EXPECT_TRUE(GetValue(
>-      report, StatsReport::kStatsValueNamePacketsSent, &value_in_report));
>-  EXPECT_EQ(rtc::ToString<int>(sinfo.packets_sent), value_in_report);
>-  EXPECT_TRUE(GetValue(
>-      report, StatsReport::kStatsValueNamePacketsLost, &value_in_report));
>-  EXPECT_EQ(rtc::ToString<int>(sinfo.packets_lost), value_in_report);
>-  EXPECT_TRUE(GetValue(
>-      report, StatsReport::kStatsValueNameRtt, &value_in_report));
>-  EXPECT_EQ(rtc::ToString<int>(sinfo.rtt_ms), value_in_report);
>-  EXPECT_TRUE(GetValue(
>-      report, StatsReport::kStatsValueNameRtt, &value_in_report));
>-  EXPECT_EQ(rtc::ToString<int>(sinfo.rtt_ms), value_in_report);
>-  EXPECT_TRUE(GetValue(
>-      report, StatsReport::kStatsValueNameJitterReceived, &value_in_report));
>-  EXPECT_EQ(rtc::ToString<int>(sinfo.jitter_ms), value_in_report);
>+  EXPECT_TRUE(GetValue(report, StatsReport::kStatsValueNameBytesSent,
>+                       &value_in_report));
>+  EXPECT_EQ(rtc::ToString(sinfo.bytes_sent), value_in_report);
>+  EXPECT_TRUE(GetValue(report, StatsReport::kStatsValueNamePacketsSent,
>+                       &value_in_report));
>+  EXPECT_EQ(rtc::ToString(sinfo.packets_sent), value_in_report);
>+  EXPECT_TRUE(GetValue(report, StatsReport::kStatsValueNamePacketsLost,
>+                       &value_in_report));
>+  EXPECT_EQ(rtc::ToString(sinfo.packets_lost), value_in_report);
>+  EXPECT_TRUE(
>+      GetValue(report, StatsReport::kStatsValueNameRtt, &value_in_report));
>+  EXPECT_EQ(rtc::ToString(sinfo.rtt_ms), value_in_report);
>+  EXPECT_TRUE(
>+      GetValue(report, StatsReport::kStatsValueNameRtt, &value_in_report));
>+  EXPECT_EQ(rtc::ToString(sinfo.rtt_ms), value_in_report);
>+  EXPECT_TRUE(GetValue(report, StatsReport::kStatsValueNameJitterReceived,
>+                       &value_in_report));
>+  EXPECT_EQ(rtc::ToString(sinfo.jitter_ms), value_in_report);
>   if (sinfo.apm_statistics.delay_median_ms) {
>     EXPECT_TRUE(GetValue(report, StatsReport::kStatsValueNameEchoDelayMedian,
>                          &value_in_report));
>-    EXPECT_EQ(rtc::ToString<int>(*sinfo.apm_statistics.delay_median_ms),
>+    EXPECT_EQ(rtc::ToString(*sinfo.apm_statistics.delay_median_ms),
>               value_in_report);
>   } else {
>     EXPECT_FALSE(GetValue(report, StatsReport::kStatsValueNameEchoDelayMedian,
>@@ -420,9 +384,8 @@ void VerifyVoiceSenderInfoReport(const StatsReport* report,
>   if (sinfo.apm_statistics.delay_standard_deviation_ms) {
>     EXPECT_TRUE(GetValue(report, StatsReport::kStatsValueNameEchoDelayStdDev,
>                          &value_in_report));
>-    EXPECT_EQ(
>-        rtc::ToString<int>(*sinfo.apm_statistics.delay_standard_deviation_ms),
>-        value_in_report);
>+    EXPECT_EQ(rtc::ToString(*sinfo.apm_statistics.delay_standard_deviation_ms),
>+              value_in_report);
>   } else {
>     EXPECT_FALSE(GetValue(report, StatsReport::kStatsValueNameEchoDelayStdDev,
>                           &value_in_report));
>@@ -430,7 +393,7 @@ void VerifyVoiceSenderInfoReport(const StatsReport* report,
>   if (sinfo.apm_statistics.echo_return_loss) {
>     EXPECT_TRUE(GetValue(report, StatsReport::kStatsValueNameEchoReturnLoss,
>                          &value_in_report));
>-    EXPECT_EQ(rtc::ToString<int>(*sinfo.apm_statistics.echo_return_loss),
>+    EXPECT_EQ(rtc::ToString(*sinfo.apm_statistics.echo_return_loss),
>               value_in_report);
>   } else {
>     EXPECT_FALSE(GetValue(report, StatsReport::kStatsValueNameEchoReturnLoss,
>@@ -440,9 +403,8 @@ void VerifyVoiceSenderInfoReport(const StatsReport* report,
>     EXPECT_TRUE(GetValue(report,
>                          StatsReport::kStatsValueNameEchoReturnLossEnhancement,
>                          &value_in_report));
>-    EXPECT_EQ(
>-        rtc::ToString<int>(*sinfo.apm_statistics.echo_return_loss_enhancement),
>-        value_in_report);
>+    EXPECT_EQ(rtc::ToString(*sinfo.apm_statistics.echo_return_loss_enhancement),
>+              value_in_report);
>   } else {
>     EXPECT_FALSE(GetValue(report,
>                           StatsReport::kStatsValueNameEchoReturnLossEnhancement,
>@@ -452,9 +414,8 @@ void VerifyVoiceSenderInfoReport(const StatsReport* report,
>     EXPECT_TRUE(GetValue(report,
>                          StatsReport::kStatsValueNameResidualEchoLikelihood,
>                          &value_in_report));
>-    EXPECT_EQ(
>-        rtc::ToString<float>(*sinfo.apm_statistics.residual_echo_likelihood),
>-        value_in_report);
>+    EXPECT_EQ(rtc::ToString(*sinfo.apm_statistics.residual_echo_likelihood),
>+              value_in_report);
>   } else {
>     EXPECT_FALSE(GetValue(report,
>                           StatsReport::kStatsValueNameResidualEchoLikelihood,
>@@ -464,7 +425,7 @@ void VerifyVoiceSenderInfoReport(const StatsReport* report,
>     EXPECT_TRUE(GetValue(
>         report, StatsReport::kStatsValueNameResidualEchoLikelihoodRecentMax,
>         &value_in_report));
>-    EXPECT_EQ(rtc::ToString<float>(
>+    EXPECT_EQ(rtc::ToString(
>                   *sinfo.apm_statistics.residual_echo_likelihood_recent_max),
>               value_in_report);
>   } else {
>@@ -474,56 +435,51 @@ void VerifyVoiceSenderInfoReport(const StatsReport* report,
>   }
>   EXPECT_TRUE(GetValue(report, StatsReport::kStatsValueNameAudioInputLevel,
>                        &value_in_report));
>-  EXPECT_EQ(rtc::ToString<int>(sinfo.audio_level), value_in_report);
>-  EXPECT_TRUE(GetValue(
>-      report, StatsReport::kStatsValueNameTypingNoiseState, &value_in_report));
>+  EXPECT_EQ(rtc::ToString(sinfo.audio_level), value_in_report);
>+  EXPECT_TRUE(GetValue(report, StatsReport::kStatsValueNameTypingNoiseState,
>+                       &value_in_report));
>   std::string typing_detected = sinfo.typing_noise_detected ? "true" : "false";
>   EXPECT_EQ(typing_detected, value_in_report);
>   EXPECT_TRUE(GetValue(report,
>                        StatsReport::kStatsValueNameAnaBitrateActionCounter,
>                        &value_in_report));
>   ASSERT_TRUE(sinfo.ana_statistics.bitrate_action_counter);
>-  EXPECT_EQ(
>-      rtc::ToString<uint32_t>(*sinfo.ana_statistics.bitrate_action_counter),
>-      value_in_report);
>+  EXPECT_EQ(rtc::ToString(*sinfo.ana_statistics.bitrate_action_counter),
>+            value_in_report);
>   EXPECT_TRUE(GetValue(report,
>                        StatsReport::kStatsValueNameAnaChannelActionCounter,
>                        &value_in_report));
>   ASSERT_TRUE(sinfo.ana_statistics.channel_action_counter);
>-  EXPECT_EQ(
>-      rtc::ToString<uint32_t>(*sinfo.ana_statistics.channel_action_counter),
>-      value_in_report);
>+  EXPECT_EQ(rtc::ToString(*sinfo.ana_statistics.channel_action_counter),
>+            value_in_report);
>   EXPECT_TRUE(GetValue(report, StatsReport::kStatsValueNameAnaDtxActionCounter,
>                        &value_in_report));
>   ASSERT_TRUE(sinfo.ana_statistics.dtx_action_counter);
>-  EXPECT_EQ(rtc::ToString<uint32_t>(*sinfo.ana_statistics.dtx_action_counter),
>+  EXPECT_EQ(rtc::ToString(*sinfo.ana_statistics.dtx_action_counter),
>             value_in_report);
>   EXPECT_TRUE(GetValue(report, StatsReport::kStatsValueNameAnaFecActionCounter,
>                        &value_in_report));
>   ASSERT_TRUE(sinfo.ana_statistics.fec_action_counter);
>-  EXPECT_EQ(rtc::ToString<uint32_t>(*sinfo.ana_statistics.fec_action_counter),
>+  EXPECT_EQ(rtc::ToString(*sinfo.ana_statistics.fec_action_counter),
>             value_in_report);
>   EXPECT_TRUE(GetValue(
>       report, StatsReport::kStatsValueNameAnaFrameLengthIncreaseCounter,
>       &value_in_report));
>   ASSERT_TRUE(sinfo.ana_statistics.frame_length_increase_counter);
>-  EXPECT_EQ(rtc::ToString<uint32_t>(
>-                *sinfo.ana_statistics.frame_length_increase_counter),
>+  EXPECT_EQ(rtc::ToString(*sinfo.ana_statistics.frame_length_increase_counter),
>             value_in_report);
>   EXPECT_TRUE(GetValue(
>       report, StatsReport::kStatsValueNameAnaFrameLengthDecreaseCounter,
>       &value_in_report));
>   ASSERT_TRUE(sinfo.ana_statistics.frame_length_decrease_counter);
>-  EXPECT_EQ(rtc::ToString<uint32_t>(
>-                *sinfo.ana_statistics.frame_length_decrease_counter),
>+  EXPECT_EQ(rtc::ToString(*sinfo.ana_statistics.frame_length_decrease_counter),
>             value_in_report);
>   EXPECT_TRUE(GetValue(report,
>                        StatsReport::kStatsValueNameAnaUplinkPacketLossFraction,
>                        &value_in_report));
>   ASSERT_TRUE(sinfo.ana_statistics.uplink_packet_loss_fraction);
>-  EXPECT_EQ(
>-      rtc::ToString<float>(*sinfo.ana_statistics.uplink_packet_loss_fraction),
>-      value_in_report);
>+  EXPECT_EQ(rtc::ToString(*sinfo.ana_statistics.uplink_packet_loss_fraction),
>+            value_in_report);
> }
> 
> // Helper methods to avoid duplication of code.
>@@ -557,9 +513,8 @@ void UpdateVoiceSenderInfoFromAudioTrack(
>     cricket::VoiceSenderInfo* voice_sender_info,
>     bool has_remote_tracks) {
>   audio_track->GetSignalLevel(&voice_sender_info->audio_level);
>-  webrtc::AudioProcessorInterface::AudioProcessorStatistics
>-      audio_processor_stats =
>-          audio_track->GetAudioProcessor()->GetStats(has_remote_tracks);
>+  AudioProcessorInterface::AudioProcessorStatistics audio_processor_stats =
>+      audio_track->GetAudioProcessor()->GetStats(has_remote_tracks);
>   voice_sender_info->typing_noise_detected =
>       audio_processor_stats.typing_noise_detected;
>   voice_sender_info->apm_statistics = audio_processor_stats.apm_statistics;
>@@ -586,14 +541,12 @@ void InitVoiceReceiverInfo(cricket::VoiceReceiverInfo* voice_receiver_info) {
>   voice_receiver_info->secondary_discarded_rate = 126;
> }
> 
>-class StatsCollectorForTest : public webrtc::StatsCollector {
>+class StatsCollectorForTest : public StatsCollector {
>  public:
>-  explicit StatsCollectorForTest(PeerConnection* pc)
>+  explicit StatsCollectorForTest(PeerConnectionInternal* pc)
>       : StatsCollector(pc), time_now_(19477) {}
> 
>-  double GetTimeNow() override {
>-    return time_now_;
>-  }
>+  double GetTimeNow() override { return time_now_; }
> 
>  private:
>   double time_now_;
>@@ -601,122 +554,45 @@ class StatsCollectorForTest : public webrtc::StatsCollector {
> 
> class StatsCollectorTest : public testing::Test {
>  protected:
>-  StatsCollectorTest()
>-      : worker_thread_(rtc::Thread::Current()),
>-        network_thread_(rtc::Thread::Current()),
>-        media_engine_(new cricket::FakeMediaEngine()),
>-        pc_factory_(new FakePeerConnectionFactory(
>-            std::unique_ptr<cricket::MediaEngineInterface>(media_engine_))),
>-        pc_(pc_factory_) {
>-    // By default, we ignore session GetStats calls.
>-    EXPECT_CALL(pc_, GetSessionStats(_)).WillRepeatedly(ReturnNull());
>-    // Add default returns for mock classes.
>-    EXPECT_CALL(pc_, video_channel()).WillRepeatedly(ReturnNull());
>-    EXPECT_CALL(pc_, voice_channel()).WillRepeatedly(ReturnNull());
>-    EXPECT_CALL(pc_, sctp_data_channels())
>-        .WillRepeatedly(ReturnRef(data_channels_));
>-    EXPECT_CALL(pc_, GetSenders()).WillRepeatedly(Return(
>-        std::vector<rtc::scoped_refptr<RtpSenderInterface>>()));
>-    EXPECT_CALL(pc_, GetReceivers()).WillRepeatedly(Return(
>-        std::vector<rtc::scoped_refptr<RtpReceiverInterface>>()));
>-    // Boring calls
>-    EXPECT_CALL(pc_, GetCallStats()).WillRepeatedly(Return(Call::Stats()));
>-  }
>-
>-  ~StatsCollectorTest() {}
>-
>-  // This creates a standard setup with a transport called "trspname"
>-  // having one transport channel
>-  // and the specified virtual connection name.
>-  void InitSessionStats(cricket::BaseChannel* channel) {
>-    const std::string kTransportName("trspname");
>-    cricket::TransportStats transport_stats;
>-    cricket::TransportChannelStats channel_stats;
>-    channel_stats.component = 1;
>-    transport_stats.transport_name = kTransportName;
>-    transport_stats.channel_stats.push_back(channel_stats);
>-
>-    session_stats_.transport_stats[kTransportName] = transport_stats;
>-    channel->set_transport_name_for_testing(kTransportName);
>-  }
>-
>-  void AddDataChannel(cricket::DataChannelType type,
>-                      const std::string& label,
>-                      int id) {
>-    InternalDataChannelInit config;
>-    config.id = id;
>-
>-    data_channels_.push_back(DataChannel::Create(
>-        &data_channel_provider_, cricket::DCT_SCTP, label, config));
>+  rtc::scoped_refptr<FakePeerConnectionForStats> CreatePeerConnection() {
>+    return new rtc::RefCountedObject<FakePeerConnectionForStats>();
>   }
> 
>-  StatsReport* AddCandidateReport(StatsCollector* collector,
>-                                  const cricket::Candidate& candidate,
>-                                  bool local) {
>-    return collector->AddCandidateReport(candidate, local);
>+  std::unique_ptr<StatsCollectorForTest> CreateStatsCollector(
>+      PeerConnectionInternal* pc) {
>+    return absl::make_unique<StatsCollectorForTest>(pc);
>   }
> 
>-  void SetupAndVerifyAudioTrackStats(
>-      FakeAudioTrack* audio_track,
>-      webrtc::MediaStream* stream,
>-      webrtc::StatsCollector* stats,
>-      cricket::VoiceChannel* voice_channel,
>-      MockVoiceMediaChannel* media_channel,
>-      cricket::VoiceSenderInfo* voice_sender_info,
>-      cricket::VoiceReceiverInfo* voice_receiver_info,
>-      cricket::VoiceMediaInfo* stats_read,
>-      StatsReports* reports) {
>-    // A track can't have both sender report and recv report at the same time
>-    // for now, this might change in the future though.
>-    EXPECT_TRUE((voice_sender_info == NULL) ^ (voice_receiver_info == NULL));
>-
>-    // Instruct the session to return stats containing the transport channel.
>-    InitSessionStats(voice_channel);
>-    EXPECT_CALL(pc_, GetSessionStats(_))
>-        .WillRepeatedly(Invoke([this](const ChannelNamePairs&) {
>-          return std::unique_ptr<SessionStats>(
>-              new SessionStats(session_stats_));
>-        }));
>-
>-    // Constructs an ssrc stats update.
>-    if (voice_sender_info)
>-      stats_read->senders.push_back(*voice_sender_info);
>-    if (voice_receiver_info)
>-      stats_read->receivers.push_back(*voice_receiver_info);
>-
>-    EXPECT_CALL(pc_, voice_channel()).WillRepeatedly(Return(voice_channel));
>-    EXPECT_CALL(pc_, video_channel()).WillRepeatedly(ReturnNull());
>-    EXPECT_CALL(*media_channel, GetStats(_))
>-        .WillOnce(DoAll(SetArgPointee<0>(*stats_read), Return(true)));
>-
>+  void VerifyAudioTrackStats(FakeAudioTrack* audio_track,
>+                             StatsCollectorForTest* stats,
>+                             const VoiceMediaInfo& voice_info,
>+                             StatsReports* reports) {
>     stats->UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
>     stats->ClearUpdateStatsCacheForTest();
>-    stats->GetStats(NULL, reports);
>+    stats->GetStats(nullptr, reports);
> 
>     // Verify the existence of the track report.
>-    const StatsReport* report = FindNthReportByType(
>-        *reports, StatsReport::kStatsReportTypeSsrc, 1);
>-    EXPECT_FALSE(report == NULL);
>+    const StatsReport* report =
>+        FindNthReportByType(*reports, StatsReport::kStatsReportTypeSsrc, 1);
>+    ASSERT_TRUE(report);
>     EXPECT_EQ(stats->GetTimeNow(), report->timestamp());
>-    std::string track_id = ExtractSsrcStatsValue(
>-        *reports, StatsReport::kStatsValueNameTrackId);
>+    std::string track_id =
>+        ExtractSsrcStatsValue(*reports, StatsReport::kStatsValueNameTrackId);
>     EXPECT_EQ(audio_track->id(), track_id);
>-    std::string ssrc_id = ExtractSsrcStatsValue(
>-        *reports, StatsReport::kStatsValueNameSsrc);
>-    EXPECT_EQ(rtc::ToString<uint32_t>(kSsrcOfTrack), ssrc_id);
>+    std::string ssrc_id =
>+        ExtractSsrcStatsValue(*reports, StatsReport::kStatsValueNameSsrc);
>+    EXPECT_EQ(rtc::ToString(kSsrcOfTrack), ssrc_id);
> 
>-    std::string media_type = ExtractSsrcStatsValue(*reports,
>-        StatsReport::kStatsValueNameMediaType);
>+    std::string media_type =
>+        ExtractSsrcStatsValue(*reports, StatsReport::kStatsValueNameMediaType);
>     EXPECT_EQ("audio", media_type);
> 
>     // Verifies the values in the track report.
>-    if (voice_sender_info) {
>-      UpdateVoiceSenderInfoFromAudioTrack(audio_track, voice_sender_info,
>-                                          stats_read->receivers.size() > 0);
>-      VerifyVoiceSenderInfoReport(report, *voice_sender_info);
>+    if (!voice_info.senders.empty()) {
>+      VerifyVoiceSenderInfoReport(report, voice_info.senders[0]);
>     }
>-    if (voice_receiver_info) {
>-      VerifyVoiceReceiverInfoReport(report, *voice_receiver_info);
>+    if (!voice_info.receivers.empty()) {
>+      VerifyVoiceReceiverInfoReport(report, voice_info.receivers[0]);
>     }
> 
>     // Verify we get the same result by passing a track to GetStats().
>@@ -724,76 +600,62 @@ class StatsCollectorTest : public testing::Test {
>     stats->GetStats(audio_track, &track_reports);
>     const StatsReport* track_report = FindNthReportByType(
>         track_reports, StatsReport::kStatsReportTypeSsrc, 1);
>-    EXPECT_TRUE(track_report);
>-    if (!track_report)
>-      return;
>+    ASSERT_TRUE(track_report);
>     EXPECT_EQ(stats->GetTimeNow(), track_report->timestamp());
>     track_id = ExtractSsrcStatsValue(track_reports,
>                                      StatsReport::kStatsValueNameTrackId);
>     EXPECT_EQ(audio_track->id(), track_id);
>-    ssrc_id = ExtractSsrcStatsValue(track_reports,
>-                                    StatsReport::kStatsValueNameSsrc);
>-    EXPECT_EQ(rtc::ToString<uint32_t>(kSsrcOfTrack), ssrc_id);
>-    if (voice_sender_info)
>-      VerifyVoiceSenderInfoReport(track_report, *voice_sender_info);
>-    if (voice_receiver_info)
>-    VerifyVoiceReceiverInfoReport(track_report, *voice_receiver_info);
>+    ssrc_id =
>+        ExtractSsrcStatsValue(track_reports, StatsReport::kStatsValueNameSsrc);
>+    EXPECT_EQ(rtc::ToString(kSsrcOfTrack), ssrc_id);
>+    if (!voice_info.senders.empty()) {
>+      VerifyVoiceSenderInfoReport(track_report, voice_info.senders[0]);
>+    }
>+    if (!voice_info.receivers.empty()) {
>+      VerifyVoiceReceiverInfoReport(track_report, voice_info.receivers[0]);
>+    }
>   }
> 
>-  void TestCertificateReports(
>-      const rtc::FakeSSLCertificate& local_cert,
>-      const std::vector<std::string>& local_ders,
>-      std::unique_ptr<rtc::FakeSSLCertificate> remote_cert,
>-      const std::vector<std::string>& remote_ders) {
>-    StatsCollectorForTest stats(&pc_);
>+  void TestCertificateReports(const rtc::FakeSSLIdentity& local_identity,
>+                              const std::vector<std::string>& local_ders,
>+                              const rtc::FakeSSLIdentity& remote_identity,
>+                              const std::vector<std::string>& remote_ders) {
>+    const std::string kTransportName = "transport";
>+
>+    auto pc = CreatePeerConnection();
>+    auto stats = CreateStatsCollector(pc);
> 
>-    StatsReports reports;  // returned values.
>+    pc->AddVoiceChannel("audio", kTransportName);
> 
>     // Fake stats to process.
>-    cricket::TransportChannelStats channel_stats;
>+    TransportChannelStats channel_stats;
>     channel_stats.component = 1;
>     channel_stats.srtp_crypto_suite = rtc::SRTP_AES128_CM_SHA1_80;
>     channel_stats.ssl_cipher_suite =
>         internal::TLS_ECDHE_RSA_WITH_AES_256_CBC_SHA;
>+    pc->SetTransportStats(kTransportName, channel_stats);
> 
>-    cricket::TransportStats transport_stats;
>-    transport_stats.transport_name = "audio";
>-    transport_stats.channel_stats.push_back(channel_stats);
>-
>-    SessionStats session_stats;
>-    session_stats.transport_stats[transport_stats.transport_name] =
>-        transport_stats;
>-
>-    // Fake certificate to report
>+    // Fake certificate to report.
>     rtc::scoped_refptr<rtc::RTCCertificate> local_certificate(
>-        rtc::RTCCertificate::Create(std::unique_ptr<rtc::FakeSSLIdentity>(
>-            new rtc::FakeSSLIdentity(local_cert))));
>+        rtc::RTCCertificate::Create(
>+            std::unique_ptr<rtc::SSLIdentity>(local_identity.GetReference())));
>+    pc->SetLocalCertificate(kTransportName, local_certificate);
>+    pc->SetRemoteCertChain(kTransportName,
>+                           remote_identity.cert_chain().UniqueCopy());
> 
>-    // Configure MockWebRtcSession
>-    EXPECT_CALL(pc_, GetLocalCertificate(transport_stats.transport_name, _))
>-        .WillOnce(DoAll(SetArgPointee<1>(local_certificate), Return(true)));
>-    EXPECT_CALL(pc_, GetRemoteSSLCertificate_ReturnsRawPointer(
>-                         transport_stats.transport_name))
>-        .WillOnce(Return(remote_cert.release()));
>-    EXPECT_CALL(pc_, GetSessionStats(_))
>-        .WillOnce(Invoke([&session_stats](const ChannelNamePairs&) {
>-          return std::unique_ptr<SessionStats>(
>-              new SessionStats(session_stats));
>-        }));
>-
>-    stats.UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
>+    stats->UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
> 
>-    stats.GetStats(NULL, &reports);
>+    StatsReports reports;
>+    stats->GetStats(nullptr, &reports);
> 
>-    const StatsReport* channel_report = FindNthReportByType(
>-        reports, StatsReport::kStatsReportTypeComponent, 1);
>-    EXPECT_TRUE(channel_report != NULL);
>+    const StatsReport* channel_report =
>+        FindNthReportByType(reports, StatsReport::kStatsReportTypeComponent, 1);
>+    EXPECT_TRUE(channel_report);
> 
>     // Check local certificate chain.
>-    std::string local_certificate_id = ExtractStatsValue(
>-        StatsReport::kStatsReportTypeComponent,
>-        reports,
>-        StatsReport::kStatsValueNameLocalCertificateId);
>+    std::string local_certificate_id =
>+        ExtractStatsValue(StatsReport::kStatsReportTypeComponent, reports,
>+                          StatsReport::kStatsValueNameLocalCertificateId);
>     if (local_ders.size() > 0) {
>       EXPECT_NE(kNotFound, local_certificate_id);
>       StatsReport::Id id(IdFromCertIdString(local_certificate_id));
>@@ -803,10 +665,9 @@ class StatsCollectorTest : public testing::Test {
>     }
> 
>     // Check remote certificate chain.
>-    std::string remote_certificate_id = ExtractStatsValue(
>-        StatsReport::kStatsReportTypeComponent,
>-        reports,
>-        StatsReport::kStatsValueNameRemoteCertificateId);
>+    std::string remote_certificate_id =
>+        ExtractStatsValue(StatsReport::kStatsReportTypeComponent, reports,
>+                          StatsReport::kStatsValueNameRemoteCertificateId);
>     if (remote_ders.size() > 0) {
>       EXPECT_NE(kNotFound, remote_certificate_id);
>       StatsReport::Id id(IdFromCertIdString(remote_certificate_id));
>@@ -828,17 +689,6 @@ class StatsCollectorTest : public testing::Test {
>     EXPECT_EQ(rtc::SrtpCryptoSuiteToName(rtc::SRTP_AES128_CM_SHA1_80),
>               srtp_crypto_suite);
>   }
>-
>-  webrtc::RtcEventLogNullImpl event_log_;
>-  rtc::Thread* const worker_thread_;
>-  rtc::Thread* const network_thread_;
>-  // |media_engine_| is actually owned by |pc_factory_|.
>-  cricket::FakeMediaEngine* media_engine_;
>-  rtc::scoped_refptr<FakePeerConnectionFactory> pc_factory_;
>-  MockPeerConnection pc_;
>-  FakeDataChannelProvider data_channel_provider_;
>-  SessionStats session_stats_;
>-  std::vector<rtc::scoped_refptr<DataChannel>> data_channels_;
> };
> 
> class StatsCollectorTrackTest : public StatsCollectorTest,
>@@ -847,90 +697,83 @@ class StatsCollectorTrackTest : public StatsCollectorTest,
>   // Adds a outgoing video track with a given SSRC into the stats.
>   // If GetParam() returns true, the track is also inserted into the local
>   // stream, which is created if necessary.
>-  void AddOutgoingVideoTrackStats(StatsCollectorForTest* stats) {
>-    track_ = webrtc::VideoTrack::Create(kLocalTrackId,
>-                                        webrtc::FakeVideoTrackSource::Create(),
>-                                        rtc::Thread::Current());
>+  void AddOutgoingVideoTrack(FakePeerConnectionForStats* pc,
>+                             StatsCollectorForTest* stats) {
>+    track_ = VideoTrack::Create(kLocalTrackId, FakeVideoTrackSource::Create(),
>+                                rtc::Thread::Current());
>     if (GetParam()) {
>       if (!stream_)
>-        stream_ = webrtc::MediaStream::Create("streamlabel");
>+        stream_ = MediaStream::Create("streamid");
>       stream_->AddTrack(track_);
>       stats->AddStream(stream_);
>     } else {
>       stats->AddTrack(track_);
>     }
>-    EXPECT_CALL(pc_, GetLocalTrackIdBySsrc(kSsrcOfTrack, _))
>-        .WillRepeatedly(DoAll(SetArgPointee<1>(kLocalTrackId), Return(true)));
>+    pc->AddLocalTrack(kSsrcOfTrack, kLocalTrackId);
>   }
> 
>   // Adds a incoming video track with a given SSRC into the stats.
>-  void AddIncomingVideoTrackStats(StatsCollectorForTest* stats) {
>-    track_ = webrtc::VideoTrack::Create(kRemoteTrackId,
>-                                        webrtc::FakeVideoTrackSource::Create(),
>-                                        rtc::Thread::Current());
>+  void AddIncomingVideoTrack(FakePeerConnectionForStats* pc,
>+                             StatsCollectorForTest* stats) {
>+    track_ = VideoTrack::Create(kRemoteTrackId, FakeVideoTrackSource::Create(),
>+                                rtc::Thread::Current());
>     if (GetParam()) {
>-      stream_ = webrtc::MediaStream::Create("streamlabel");
>+      stream_ = MediaStream::Create("streamid");
>       stream_->AddTrack(track_);
>       stats->AddStream(stream_);
>     } else {
>       stats->AddTrack(track_);
>     }
>-    EXPECT_CALL(pc_, GetRemoteTrackIdBySsrc(kSsrcOfTrack, _))
>-        .WillRepeatedly(DoAll(SetArgPointee<1>(kRemoteTrackId), Return(true)));
>+    pc->AddRemoteTrack(kSsrcOfTrack, kRemoteTrackId);
>   }
> 
>   // Adds a outgoing audio track with a given SSRC into the stats,
>   // and register it into the stats object.
>   // If GetParam() returns true, the track is also inserted into the local
>   // stream, which is created if necessary.
>-  void AddOutgoingAudioTrackStats(StatsCollectorForTest* stats) {
>+  void AddOutgoingAudioTrack(FakePeerConnectionForStats* pc,
>+                             StatsCollectorForTest* stats) {
>     audio_track_ = new rtc::RefCountedObject<FakeAudioTrack>(kLocalTrackId);
>     if (GetParam()) {
>       if (!stream_)
>-        stream_ = webrtc::MediaStream::Create("streamlabel");
>+        stream_ = MediaStream::Create("streamid");
>       stream_->AddTrack(audio_track_);
>       stats->AddStream(stream_);
>     } else {
>       stats->AddTrack(audio_track_);
>     }
>-    EXPECT_CALL(pc_, GetLocalTrackIdBySsrc(kSsrcOfTrack, _))
>-        .WillOnce(DoAll(SetArgPointee<1>(kLocalTrackId), Return(true)));
>+    pc->AddLocalTrack(kSsrcOfTrack, kLocalTrackId);
>   }
> 
>   // Adds a incoming audio track with a given SSRC into the stats.
>-  void AddIncomingAudioTrackStats(StatsCollectorForTest* stats) {
>+  void AddIncomingAudioTrack(FakePeerConnectionForStats* pc,
>+                             StatsCollectorForTest* stats) {
>     audio_track_ = new rtc::RefCountedObject<FakeAudioTrack>(kRemoteTrackId);
>     if (GetParam()) {
>       if (stream_ == NULL)
>-        stream_ = webrtc::MediaStream::Create("streamlabel");
>+        stream_ = MediaStream::Create("streamid");
>       stream_->AddTrack(audio_track_);
>       stats->AddStream(stream_);
>     } else {
>       stats->AddTrack(audio_track_);
>     }
>-    EXPECT_CALL(pc_, GetRemoteTrackIdBySsrc(kSsrcOfTrack, _))
>-        .WillOnce(DoAll(SetArgPointee<1>(kRemoteTrackId), Return(true)));
>+    pc->AddRemoteTrack(kSsrcOfTrack, kRemoteTrackId);
>   }
> 
>-  rtc::scoped_refptr<webrtc::MediaStream> stream_;
>-  rtc::scoped_refptr<webrtc::VideoTrack> track_;
>+  rtc::scoped_refptr<MediaStream> stream_;
>+  rtc::scoped_refptr<VideoTrack> track_;
>   rtc::scoped_refptr<FakeAudioTrack> audio_track_;
> };
> 
> TEST_F(StatsCollectorTest, FilterOutNegativeDataChannelId) {
>-  const std::string label = "hacks";
>-  // The data channel id is from the Config which is -1 initially.
>-  const int id = -1;
>-  const std::string state = DataChannelInterface::DataStateString(
>-      DataChannelInterface::DataState::kConnecting);
>+  auto pc = CreatePeerConnection();
>+  auto stats = CreateStatsCollector(pc);
> 
>-  AddDataChannel(cricket::DCT_SCTP, label, id);
>-  StatsCollectorForTest stats(&pc_);
>-
>-  stats.UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
>+  pc->AddSctpDataChannel("hacks");
> 
>+  stats->UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
>   StatsReports reports;
>-  stats.GetStats(NULL, &reports);
>+  stats->GetStats(nullptr, &reports);
> 
>   const StatsReport* report =
>       FindNthReportByType(reports, StatsReport::kStatsReportTypeDataChannel, 1);
>@@ -942,267 +785,218 @@ TEST_F(StatsCollectorTest, FilterOutNegativeDataChannelId) {
> 
> // Verify that ExtractDataInfo populates reports.
> TEST_F(StatsCollectorTest, ExtractDataInfo) {
>-  const std::string label = "hacks";
>-  const int id = 31337;
>-  const std::string state = DataChannelInterface::DataStateString(
>+  const std::string kDataChannelLabel = "hacks";
>+  constexpr int kDataChannelId = 31337;
>+  const std::string kConnectingString = DataChannelInterface::DataStateString(
>       DataChannelInterface::DataState::kConnecting);
> 
>-  AddDataChannel(cricket::DCT_SCTP, label, id);
>-  StatsCollectorForTest stats(&pc_);
>+  auto pc = CreatePeerConnection();
>+  auto stats = CreateStatsCollector(pc);
> 
>-  stats.UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
>+  InternalDataChannelInit init;
>+  init.id = kDataChannelId;
>+  pc->AddSctpDataChannel(kDataChannelLabel, init);
> 
>+  stats->UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
>   StatsReports reports;
>-  stats.GetStats(NULL, &reports);
>+  stats->GetStats(nullptr, &reports);
> 
>   const StatsReport* report =
>       FindNthReportByType(reports, StatsReport::kStatsReportTypeDataChannel, 1);
> 
>-  StatsReport::Id reportId = StatsReport::NewTypedIntId(
>-      StatsReport::kStatsReportTypeDataChannel, id);
>+  StatsReport::Id report_id = StatsReport::NewTypedIntId(
>+      StatsReport::kStatsReportTypeDataChannel, kDataChannelId);
> 
>-  EXPECT_TRUE(reportId->Equals(report->id()));
>+  EXPECT_TRUE(report_id->Equals(report->id()));
> 
>-  EXPECT_EQ(stats.GetTimeNow(), report->timestamp());
>-  EXPECT_EQ(label, ExtractStatsValue(StatsReport::kStatsReportTypeDataChannel,
>-                                     reports,
>-                                     StatsReport::kStatsValueNameLabel));
>-  EXPECT_EQ(rtc::ToString<int64_t>(id),
>+  EXPECT_EQ(stats->GetTimeNow(), report->timestamp());
>+  EXPECT_EQ(kDataChannelLabel,
>+            ExtractStatsValue(StatsReport::kStatsReportTypeDataChannel, reports,
>+                              StatsReport::kStatsValueNameLabel));
>+  EXPECT_EQ(rtc::ToString(kDataChannelId),
>             ExtractStatsValue(StatsReport::kStatsReportTypeDataChannel, reports,
>                               StatsReport::kStatsValueNameDataChannelId));
>-  EXPECT_EQ(state, ExtractStatsValue(StatsReport::kStatsReportTypeDataChannel,
>-                                     reports,
>-                                     StatsReport::kStatsValueNameState));
>-  EXPECT_EQ("", ExtractStatsValue(StatsReport::kStatsReportTypeDataChannel,
>-                                  reports,
>-                                  StatsReport::kStatsValueNameProtocol));
>+  EXPECT_EQ(kConnectingString,
>+            ExtractStatsValue(StatsReport::kStatsReportTypeDataChannel, reports,
>+                              StatsReport::kStatsValueNameState));
>+  EXPECT_EQ("",
>+            ExtractStatsValue(StatsReport::kStatsReportTypeDataChannel, reports,
>+                              StatsReport::kStatsValueNameProtocol));
> }
> 
> // This test verifies that 64-bit counters are passed successfully.
> TEST_P(StatsCollectorTrackTest, BytesCounterHandles64Bits) {
>-  StatsCollectorForTest stats(&pc_);
>-
>-  EXPECT_CALL(pc_, GetLocalCertificate(_, _)).WillRepeatedly(Return(false));
>-  EXPECT_CALL(pc_, GetRemoteSSLCertificate_ReturnsRawPointer(_))
>-      .WillRepeatedly(Return(nullptr));
>-
>-  auto* media_channel = new MockVideoMediaChannel();
>-  cricket::VideoChannel video_channel(
>-      worker_thread_, network_thread_, nullptr, rtc::WrapUnique(media_channel),
>-      "video", kDefaultRtcpMuxRequired, kDefaultSrtpRequired);
>-
>-  InitSessionStats(&video_channel);
>-  EXPECT_CALL(pc_, GetSessionStats(_))
>-      .WillRepeatedly(Invoke([this](const ChannelNamePairs&) {
>-        return std::unique_ptr<SessionStats>(
>-            new SessionStats(session_stats_));
>-      }));
>-
>-  StatsReports reports;  // returned values.
>-  cricket::VideoSenderInfo video_sender_info;
>-  cricket::VideoMediaInfo stats_read;
>   // The number of bytes must be larger than 0xFFFFFFFF for this test.
>-  const int64_t kBytesSent = 12345678901234LL;
>-  const std::string kBytesSentString("12345678901234");
>+  constexpr int64_t kBytesSent = 12345678901234LL;
> 
>-  AddOutgoingVideoTrackStats(&stats);
>+  auto pc = CreatePeerConnection();
>+  auto stats = CreateStatsCollector(pc);
> 
>-  // Construct a stats value to read.
>+  VideoSenderInfo video_sender_info;
>   video_sender_info.add_ssrc(1234);
>   video_sender_info.bytes_sent = kBytesSent;
>-  stats_read.senders.push_back(video_sender_info);
>-
>-  EXPECT_CALL(pc_, video_channel()).WillRepeatedly(Return(&video_channel));
>-  EXPECT_CALL(pc_, voice_channel()).WillRepeatedly(ReturnNull());
>-  EXPECT_CALL(*media_channel, GetStats(_))
>-      .WillOnce(DoAll(SetArgPointee<0>(stats_read),
>-                      Return(true)));
>-  stats.UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
>-  stats.GetStats(NULL, &reports);
>-  std::string result = ExtractSsrcStatsValue(reports,
>-      StatsReport::kStatsValueNameBytesSent);
>-  EXPECT_EQ(kBytesSentString, result);
>-}
>+  VideoMediaInfo video_info;
>+  video_info.senders.push_back(video_sender_info);
> 
>-// Test that audio BWE information is reported via stats.
>-TEST_P(StatsCollectorTrackTest, AudioBandwidthEstimationInfoIsReported) {
>-  StatsCollectorForTest stats(&pc_);
>+  auto* video_media_channel = pc->AddVideoChannel("video", "transport");
>+  video_media_channel->SetStats(video_info);
> 
>-  EXPECT_CALL(pc_, GetLocalCertificate(_, _)).WillRepeatedly(Return(false));
>-  EXPECT_CALL(pc_, GetRemoteSSLCertificate_ReturnsRawPointer(_))
>-      .WillRepeatedly(Return(nullptr));
>+  AddOutgoingVideoTrack(pc, stats.get());
> 
>-  auto* media_channel = new MockVoiceMediaChannel();
>-  cricket::VoiceChannel voice_channel(worker_thread_, network_thread_, nullptr,
>-                                      nullptr, rtc::WrapUnique(media_channel),
>-                                      "audio", kDefaultRtcpMuxRequired,
>-                                      kDefaultSrtpRequired);
>+  stats->UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
>+  StatsReports reports;
>+  stats->GetStats(nullptr, &reports);
> 
>-  InitSessionStats(&voice_channel);
>-  EXPECT_CALL(pc_, GetSessionStats(_))
>-      .WillRepeatedly(Invoke([this](const ChannelNamePairs&) {
>-        return std::unique_ptr<SessionStats>(new SessionStats(session_stats_));
>-      }));
>+  EXPECT_EQ(
>+      rtc::ToString(kBytesSent),
>+      ExtractSsrcStatsValue(reports, StatsReport::kStatsValueNameBytesSent));
>+}
> 
>-  StatsReports reports;  // returned values.
>-  cricket::VoiceSenderInfo voice_sender_info;
>-  cricket::VoiceMediaInfo stats_read;
>+// Test that audio BWE information is reported via stats.
>+TEST_P(StatsCollectorTrackTest, AudioBandwidthEstimationInfoIsReported) {
>   // Set up an SSRC just to test that we get both kinds of stats back: SSRC and
>   // BWE.
>-  const int64_t kBytesSent = 12345678901234LL;
>-  const std::string kBytesSentString("12345678901234");
>+  constexpr int64_t kBytesSent = 12345678901234LL;
>+  constexpr int kSendBandwidth = 1234567;
>+  constexpr int kRecvBandwidth = 12345678;
>+  constexpr int kPacerDelay = 123;
> 
>-  AddOutgoingAudioTrackStats(&stats);
>+  auto pc = CreatePeerConnection();
>+  auto stats = CreateStatsCollector(pc);
> 
>-  // Construct a stats value to read.
>+  VoiceSenderInfo voice_sender_info;
>   voice_sender_info.add_ssrc(1234);
>   voice_sender_info.bytes_sent = kBytesSent;
>-  stats_read.senders.push_back(voice_sender_info);
>+  VoiceMediaInfo voice_info;
>+  voice_info.senders.push_back(voice_sender_info);
>+
>+  auto* voice_media_channel = pc->AddVoiceChannel("audio", "transport");
>+  voice_media_channel->SetStats(voice_info);
>+
>+  AddOutgoingAudioTrack(pc, stats.get());
> 
>-  webrtc::Call::Stats call_stats;
>-  const int kSendBandwidth = 1234567;
>-  const int kRecvBandwidth = 12345678;
>-  const int kPacerDelay = 123;
>+  Call::Stats call_stats;
>   call_stats.send_bandwidth_bps = kSendBandwidth;
>   call_stats.recv_bandwidth_bps = kRecvBandwidth;
>   call_stats.pacer_delay_ms = kPacerDelay;
>-  EXPECT_CALL(pc_, GetCallStats()).WillRepeatedly(Return(call_stats));
>-  EXPECT_CALL(pc_, voice_channel()).WillRepeatedly(Return(&voice_channel));
>-  EXPECT_CALL(pc_, video_channel()).WillRepeatedly(ReturnNull());
>-  EXPECT_CALL(*media_channel, GetStats(_))
>-      .WillOnce(DoAll(SetArgPointee<0>(stats_read), Return(true)));
>-
>-  stats.UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
>-  stats.GetStats(NULL, &reports);
>-  std::string result =
>-      ExtractSsrcStatsValue(reports, StatsReport::kStatsValueNameBytesSent);
>-  EXPECT_EQ(kBytesSentString, result);
>-  result = ExtractBweStatsValue(
>-      reports, StatsReport::kStatsValueNameAvailableSendBandwidth);
>-  EXPECT_EQ(rtc::ToString(kSendBandwidth), result);
>-  result = ExtractBweStatsValue(
>-      reports, StatsReport::kStatsValueNameAvailableReceiveBandwidth);
>-  EXPECT_EQ(rtc::ToString(kRecvBandwidth), result);
>-  result =
>-      ExtractBweStatsValue(reports, StatsReport::kStatsValueNameBucketDelay);
>-  EXPECT_EQ(rtc::ToString(kPacerDelay), result);
>+  pc->SetCallStats(call_stats);
>+
>+  stats->UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
>+  StatsReports reports;
>+  stats->GetStats(nullptr, &reports);
>+
>+  EXPECT_EQ(
>+      rtc::ToString(kBytesSent),
>+      ExtractSsrcStatsValue(reports, StatsReport::kStatsValueNameBytesSent));
>+  EXPECT_EQ(rtc::ToString(kSendBandwidth),
>+            ExtractBweStatsValue(
>+                reports, StatsReport::kStatsValueNameAvailableSendBandwidth));
>+  EXPECT_EQ(
>+      rtc::ToString(kRecvBandwidth),
>+      ExtractBweStatsValue(
>+          reports, StatsReport::kStatsValueNameAvailableReceiveBandwidth));
>+  EXPECT_EQ(
>+      rtc::ToString(kPacerDelay),
>+      ExtractBweStatsValue(reports, StatsReport::kStatsValueNameBucketDelay));
> }
> 
> // Test that video BWE information is reported via stats.
> TEST_P(StatsCollectorTrackTest, VideoBandwidthEstimationInfoIsReported) {
>-  StatsCollectorForTest stats(&pc_);
>-
>-  EXPECT_CALL(pc_, GetLocalCertificate(_, _)).WillRepeatedly(Return(false));
>-  EXPECT_CALL(pc_, GetRemoteSSLCertificate_ReturnsRawPointer(_))
>-      .WillRepeatedly(Return(nullptr));
>-
>-  auto* media_channel = new MockVideoMediaChannel();
>-  cricket::VideoChannel video_channel(
>-      worker_thread_, network_thread_, nullptr, rtc::WrapUnique(media_channel),
>-      "video", kDefaultRtcpMuxRequired, kDefaultSrtpRequired);
>-
>-  InitSessionStats(&video_channel);
>-  EXPECT_CALL(pc_, GetSessionStats(_))
>-      .WillRepeatedly(Invoke([this](const ChannelNamePairs&) {
>-        return std::unique_ptr<SessionStats>(
>-            new SessionStats(session_stats_));
>-      }));
>-
>-  StatsReports reports;  // returned values.
>-  cricket::VideoSenderInfo video_sender_info;
>-  cricket::VideoMediaInfo stats_read;
>   // Set up an SSRC just to test that we get both kinds of stats back: SSRC and
>   // BWE.
>-  const int64_t kBytesSent = 12345678901234LL;
>-  const std::string kBytesSentString("12345678901234");
>+  constexpr int64_t kBytesSent = 12345678901234LL;
>+  constexpr int kSendBandwidth = 1234567;
>+  constexpr int kRecvBandwidth = 12345678;
>+  constexpr int kPacerDelay = 123;
> 
>-  AddOutgoingVideoTrackStats(&stats);
>+  auto pc = CreatePeerConnection();
>+  auto stats = CreateStatsCollector(pc);
> 
>-  // Construct a stats value to read.
>+  VideoSenderInfo video_sender_info;
>   video_sender_info.add_ssrc(1234);
>   video_sender_info.bytes_sent = kBytesSent;
>-  stats_read.senders.push_back(video_sender_info);
>+  VideoMediaInfo video_info;
>+  video_info.senders.push_back(video_sender_info);
>+
>+  auto* video_media_channel = pc->AddVideoChannel("video", "transport");
>+  video_media_channel->SetStats(video_info);
> 
>-  webrtc::Call::Stats call_stats;
>-  const int kSendBandwidth = 1234567;
>-  const int kRecvBandwidth = 12345678;
>-  const int kPacerDelay = 123;
>+  AddOutgoingVideoTrack(pc, stats.get());
>+
>+  Call::Stats call_stats;
>   call_stats.send_bandwidth_bps = kSendBandwidth;
>   call_stats.recv_bandwidth_bps = kRecvBandwidth;
>   call_stats.pacer_delay_ms = kPacerDelay;
>-  EXPECT_CALL(pc_, GetCallStats()).WillRepeatedly(Return(call_stats));
>-  EXPECT_CALL(pc_, video_channel()).WillRepeatedly(Return(&video_channel));
>-  EXPECT_CALL(pc_, voice_channel()).WillRepeatedly(ReturnNull());
>-  EXPECT_CALL(*media_channel, GetStats(_))
>-      .WillOnce(DoAll(SetArgPointee<0>(stats_read), Return(true)));
>-
>-  stats.UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
>-  stats.GetStats(NULL, &reports);
>-  std::string result = ExtractSsrcStatsValue(reports,
>-      StatsReport::kStatsValueNameBytesSent);
>-  EXPECT_EQ(kBytesSentString, result);
>-  result = ExtractBweStatsValue(
>-      reports, StatsReport::kStatsValueNameAvailableSendBandwidth);
>-  EXPECT_EQ(rtc::ToString(kSendBandwidth), result);
>-  result = ExtractBweStatsValue(
>-      reports, StatsReport::kStatsValueNameAvailableReceiveBandwidth);
>-  EXPECT_EQ(rtc::ToString(kRecvBandwidth), result);
>-  result =
>-      ExtractBweStatsValue(reports, StatsReport::kStatsValueNameBucketDelay);
>-  EXPECT_EQ(rtc::ToString(kPacerDelay), result);
>+  pc->SetCallStats(call_stats);
>+
>+  stats->UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
>+  StatsReports reports;
>+  stats->GetStats(nullptr, &reports);
>+
>+  EXPECT_EQ(
>+      rtc::ToString(kBytesSent),
>+      ExtractSsrcStatsValue(reports, StatsReport::kStatsValueNameBytesSent));
>+  EXPECT_EQ(rtc::ToString(kSendBandwidth),
>+            ExtractBweStatsValue(
>+                reports, StatsReport::kStatsValueNameAvailableSendBandwidth));
>+  EXPECT_EQ(
>+      rtc::ToString(kRecvBandwidth),
>+      ExtractBweStatsValue(
>+          reports, StatsReport::kStatsValueNameAvailableReceiveBandwidth));
>+  EXPECT_EQ(
>+      rtc::ToString(kPacerDelay),
>+      ExtractBweStatsValue(reports, StatsReport::kStatsValueNameBucketDelay));
> }
> 
> // This test verifies that an object of type "googSession" always
> // exists in the returned stats.
> TEST_F(StatsCollectorTest, SessionObjectExists) {
>-  StatsCollectorForTest stats(&pc_);
>+  auto pc = CreatePeerConnection();
>+  auto stats = CreateStatsCollector(pc);
> 
>-  StatsReports reports;  // returned values.
>-  stats.UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
>-  stats.GetStats(NULL, &reports);
>-  const StatsReport* session_report = FindNthReportByType(
>-      reports, StatsReport::kStatsReportTypeSession, 1);
>-  EXPECT_FALSE(session_report == NULL);
>+  stats->UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
>+  StatsReports reports;
>+  stats->GetStats(nullptr, &reports);
>+
>+  EXPECT_TRUE(
>+      FindNthReportByType(reports, StatsReport::kStatsReportTypeSession, 1));
> }
> 
> // This test verifies that only one object of type "googSession" exists
> // in the returned stats.
> TEST_F(StatsCollectorTest, OnlyOneSessionObjectExists) {
>-  StatsCollectorForTest stats(&pc_);
>+  auto pc = CreatePeerConnection();
>+  auto stats = CreateStatsCollector(pc);
> 
>-  StatsReports reports;  // returned values.
>-  stats.UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
>-  stats.UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
>-  stats.GetStats(NULL, &reports);
>-  const StatsReport* session_report = FindNthReportByType(
>-      reports, StatsReport::kStatsReportTypeSession, 1);
>-  EXPECT_FALSE(session_report == NULL);
>-  session_report = FindNthReportByType(
>-      reports, StatsReport::kStatsReportTypeSession, 2);
>-  EXPECT_EQ(NULL, session_report);
>+  stats->UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
>+  stats->UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
>+  StatsReports reports;
>+  stats->GetStats(nullptr, &reports);
>+
>+  EXPECT_TRUE(
>+      FindNthReportByType(reports, StatsReport::kStatsReportTypeSession, 1));
>+  EXPECT_FALSE(
>+      FindNthReportByType(reports, StatsReport::kStatsReportTypeSession, 2));
> }
> 
> // This test verifies that the empty track report exists in the returned stats
> // without calling StatsCollector::UpdateStats.
> TEST_P(StatsCollectorTrackTest, TrackObjectExistsWithoutUpdateStats) {
>-  StatsCollectorForTest stats(&pc_);
>+  auto pc = CreatePeerConnection();
>+  auto stats = CreateStatsCollector(pc);
> 
>-  auto* media_channel = new MockVideoMediaChannel();
>-  cricket::VideoChannel video_channel(
>-      worker_thread_, network_thread_, nullptr, rtc::WrapUnique(media_channel),
>-      "video", kDefaultRtcpMuxRequired, kDefaultSrtpRequired);
>-  AddOutgoingVideoTrackStats(&stats);
>+  pc->AddVideoChannel("video", "transport");
>+  AddOutgoingVideoTrack(pc, stats.get());
> 
>   // Verfies the existence of the track report.
>   StatsReports reports;
>-  stats.GetStats(NULL, &reports);
>-  ASSERT_EQ((size_t)1, reports.size());
>+  stats->GetStats(nullptr, &reports);
>+  ASSERT_EQ(1u, reports.size());
>   EXPECT_EQ(StatsReport::kStatsReportTypeTrack, reports[0]->type());
>   EXPECT_EQ(0, reports[0]->timestamp());
> 
>   std::string trackValue =
>-      ExtractStatsValue(StatsReport::kStatsReportTypeTrack,
>-                        reports,
>+      ExtractStatsValue(StatsReport::kStatsReportTypeTrack, reports,
>                         StatsReport::kStatsValueNameTrackId);
>   EXPECT_EQ(kLocalTrackId, trackValue);
> }
>@@ -1210,122 +1004,85 @@ TEST_P(StatsCollectorTrackTest, TrackObjectExistsWithoutUpdateStats) {
> // This test verifies that the empty track report exists in the returned stats
> // when StatsCollector::UpdateStats is called with ssrc stats.
> TEST_P(StatsCollectorTrackTest, TrackAndSsrcObjectExistAfterUpdateSsrcStats) {
>-  StatsCollectorForTest stats(&pc_);
>-
>-  EXPECT_CALL(pc_, GetLocalCertificate(_, _)).WillRepeatedly(Return(false));
>-  EXPECT_CALL(pc_, GetRemoteSSLCertificate_ReturnsRawPointer(_))
>-      .WillRepeatedly(Return(nullptr));
>+  constexpr int64_t kBytesSent = 12345678901234LL;
> 
>-  auto* media_channel = new MockVideoMediaChannel();
>-  cricket::VideoChannel video_channel(
>-      worker_thread_, network_thread_, nullptr, rtc::WrapUnique(media_channel),
>-      "video", kDefaultRtcpMuxRequired, kDefaultSrtpRequired);
>-
>-  InitSessionStats(&video_channel);
>-  EXPECT_CALL(pc_, GetSessionStats(_))
>-      .WillRepeatedly(Invoke([this](const ChannelNamePairs&) {
>-        return std::unique_ptr<SessionStats>(
>-            new SessionStats(session_stats_));
>-      }));
>-
>-  AddOutgoingVideoTrackStats(&stats);
>-
>-  // Constructs an ssrc stats update.
>-  cricket::VideoSenderInfo video_sender_info;
>-  cricket::VideoMediaInfo stats_read;
>-  const int64_t kBytesSent = 12345678901234LL;
>+  auto pc = CreatePeerConnection();
>+  auto stats = CreateStatsCollector(pc);
> 
>-  // Construct a stats value to read.
>+  VideoSenderInfo video_sender_info;
>   video_sender_info.add_ssrc(1234);
>   video_sender_info.bytes_sent = kBytesSent;
>-  stats_read.senders.push_back(video_sender_info);
>+  VideoMediaInfo video_info;
>+  video_info.senders.push_back(video_sender_info);
> 
>-  EXPECT_CALL(pc_, video_channel()).WillRepeatedly(Return(&video_channel));
>-  EXPECT_CALL(pc_, voice_channel()).WillRepeatedly(ReturnNull());
>-  EXPECT_CALL(*media_channel, GetStats(_))
>-    .WillOnce(DoAll(SetArgPointee<0>(stats_read),
>-                    Return(true)));
>+  auto* video_media_channel = pc->AddVideoChannel("video", "transport");
>+  video_media_channel->SetStats(video_info);
> 
>-  stats.UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
>+  AddOutgoingVideoTrack(pc, stats.get());
>+
>+  stats->UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
>   StatsReports reports;
>-  stats.GetStats(NULL, &reports);
>+  stats->GetStats(nullptr, &reports);
>+
>   // |reports| should contain at least one session report, one track report,
>   // and one ssrc report.
>-  EXPECT_LE((size_t)3, reports.size());
>-  const StatsReport* track_report = FindNthReportByType(
>-      reports, StatsReport::kStatsReportTypeTrack, 1);
>+  EXPECT_LE(3u, reports.size());
>+  const StatsReport* track_report =
>+      FindNthReportByType(reports, StatsReport::kStatsReportTypeTrack, 1);
>   EXPECT_TRUE(track_report);
> 
>   // Get report for the specific |track|.
>   reports.clear();
>-  stats.GetStats(track_, &reports);
>+  stats->GetStats(track_, &reports);
>   // |reports| should contain at least one session report, one track report,
>   // and one ssrc report.
>-  EXPECT_LE((size_t)3, reports.size());
>-  track_report = FindNthReportByType(
>-      reports, StatsReport::kStatsReportTypeTrack, 1);
>+  EXPECT_LE(3u, reports.size());
>+  track_report =
>+      FindNthReportByType(reports, StatsReport::kStatsReportTypeTrack, 1);
>   ASSERT_TRUE(track_report);
>-  EXPECT_EQ(stats.GetTimeNow(), track_report->timestamp());
>+  EXPECT_EQ(stats->GetTimeNow(), track_report->timestamp());
> 
>-  std::string ssrc_id = ExtractSsrcStatsValue(
>-      reports, StatsReport::kStatsValueNameSsrc);
>-  EXPECT_EQ(rtc::ToString<uint32_t>(kSsrcOfTrack), ssrc_id);
>+  std::string ssrc_id =
>+      ExtractSsrcStatsValue(reports, StatsReport::kStatsValueNameSsrc);
>+  EXPECT_EQ(rtc::ToString(kSsrcOfTrack), ssrc_id);
> 
>-  std::string track_id = ExtractSsrcStatsValue(
>-      reports, StatsReport::kStatsValueNameTrackId);
>+  std::string track_id =
>+      ExtractSsrcStatsValue(reports, StatsReport::kStatsValueNameTrackId);
>   EXPECT_EQ(kLocalTrackId, track_id);
> 
>-  std::string media_type = ExtractSsrcStatsValue(reports,
>-      StatsReport::kStatsValueNameMediaType);
>+  std::string media_type =
>+      ExtractSsrcStatsValue(reports, StatsReport::kStatsValueNameMediaType);
>   EXPECT_EQ("video", media_type);
> }
> 
> // This test verifies that an SSRC object has the identifier of a Transport
> // stats object, and that this transport stats object exists in stats.
> TEST_P(StatsCollectorTrackTest, TransportObjectLinkedFromSsrcObject) {
>-  StatsCollectorForTest stats(&pc_);
>-
>-  EXPECT_CALL(pc_, GetLocalCertificate(_, _)).WillRepeatedly(Return(false));
>-  EXPECT_CALL(pc_, GetRemoteSSLCertificate_ReturnsRawPointer(_))
>-      .WillRepeatedly(Return(nullptr));
>-
>-  auto* media_channel = new MockVideoMediaChannel();
>-  cricket::VideoChannel video_channel(
>-      worker_thread_, network_thread_, nullptr, rtc::WrapUnique(media_channel),
>-      "vcname", kDefaultRtcpMuxRequired, kDefaultSrtpRequired);
>-  AddOutgoingVideoTrackStats(&stats);
>+  constexpr int64_t kBytesSent = 12345678901234LL;
> 
>-  // Constructs an ssrc stats update.
>-  cricket::VideoSenderInfo video_sender_info;
>-  cricket::VideoMediaInfo stats_read;
>-  const int64_t kBytesSent = 12345678901234LL;
>+  auto pc = CreatePeerConnection();
>+  auto stats = CreateStatsCollector(pc);
> 
>-  // Construct a stats value to read.
>+  VideoSenderInfo video_sender_info;
>   video_sender_info.add_ssrc(1234);
>   video_sender_info.bytes_sent = kBytesSent;
>-  stats_read.senders.push_back(video_sender_info);
>-
>-  EXPECT_CALL(pc_, video_channel()).WillRepeatedly(Return(&video_channel));
>-  EXPECT_CALL(pc_, voice_channel()).WillRepeatedly(ReturnNull());
>-  EXPECT_CALL(*media_channel, GetStats(_))
>-    .WillRepeatedly(DoAll(SetArgPointee<0>(stats_read),
>-                          Return(true)));
>-
>-  InitSessionStats(&video_channel);
>-  EXPECT_CALL(pc_, GetSessionStats(_))
>-      .WillRepeatedly(Invoke([this](const ChannelNamePairs&) {
>-        return std::unique_ptr<SessionStats>(
>-            new SessionStats(session_stats_));
>-      }));
>-
>-  stats.UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
>+  VideoMediaInfo video_info;
>+  video_info.senders.push_back(video_sender_info);
>+
>+  auto* video_media_channel = pc->AddVideoChannel("video", "transport");
>+  video_media_channel->SetStats(video_info);
>+
>+  AddOutgoingVideoTrack(pc, stats.get());
>+
>+  stats->UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
>   StatsReports reports;
>-  stats.GetStats(NULL, &reports);
>-  std::string transport_id = ExtractStatsValue(
>-      StatsReport::kStatsReportTypeSsrc,
>-      reports,
>-      StatsReport::kStatsValueNameTransportId);
>+  stats->GetStats(nullptr, &reports);
>+
>+  std::string transport_id =
>+      ExtractStatsValue(StatsReport::kStatsReportTypeSsrc, reports,
>+                        StatsReport::kStatsValueNameTransportId);
>   ASSERT_NE(kNotFound, transport_id);
>+
>   // Transport id component ID will always be 1.
>   // This has assumptions about how the ID is constructed.  As is, this is
>   // OK since this is for testing purposes only, but if we ever need this
>@@ -1339,184 +1096,155 @@ TEST_P(StatsCollectorTrackTest, TransportObjectLinkedFromSsrcObject) {
>   StatsReport::Id id(StatsReport::NewComponentId(content, 1));
>   ASSERT_EQ(transport_id, id->ToString());
>   const StatsReport* transport_report = FindReportById(reports, id);
>-  ASSERT_FALSE(transport_report == NULL);
>+  ASSERT_TRUE(transport_report);
> }
> 
> // This test verifies that a remote stats object will not be created for
> // an outgoing SSRC where remote stats are not returned.
> TEST_P(StatsCollectorTrackTest, RemoteSsrcInfoIsAbsent) {
>-  StatsCollectorForTest stats(&pc_);
>+  auto pc = CreatePeerConnection();
>+  auto stats = CreateStatsCollector(pc);
> 
>-  auto* media_channel = new MockVideoMediaChannel();
>-  cricket::VideoChannel video_channel(
>-      worker_thread_, network_thread_, nullptr, rtc::WrapUnique(media_channel),
>-      "vcname", kDefaultRtcpMuxRequired, kDefaultSrtpRequired);
>-  AddOutgoingVideoTrackStats(&stats);
>+  pc->AddVideoChannel("video", "transport");
>+  AddOutgoingVideoTrack(pc, stats.get());
> 
>-  stats.UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
>+  stats->UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
>   StatsReports reports;
>-  stats.GetStats(NULL, &reports);
>-  const StatsReport* remote_report = FindNthReportByType(reports,
>-      StatsReport::kStatsReportTypeRemoteSsrc, 1);
>-  EXPECT_TRUE(remote_report == NULL);
>+  stats->GetStats(nullptr, &reports);
>+
>+  const StatsReport* remote_report =
>+      FindNthReportByType(reports, StatsReport::kStatsReportTypeRemoteSsrc, 1);
>+  EXPECT_FALSE(remote_report);
> }
> 
> // This test verifies that a remote stats object will be created for
> // an outgoing SSRC where stats are returned.
> TEST_P(StatsCollectorTrackTest, RemoteSsrcInfoIsPresent) {
>-  StatsCollectorForTest stats(&pc_);
>+  auto pc = CreatePeerConnection();
>+  auto stats = CreateStatsCollector(pc);
> 
>-  EXPECT_CALL(pc_, GetLocalCertificate(_, _)).WillRepeatedly(Return(false));
>-  EXPECT_CALL(pc_, GetRemoteSSLCertificate_ReturnsRawPointer(_))
>-      .WillRepeatedly(Return(nullptr));
>-
>-  auto* media_channel = new MockVideoMediaChannel();
>-  // The transport_name known by the video channel.
>-  cricket::VideoChannel video_channel(
>-      worker_thread_, network_thread_, nullptr, rtc::WrapUnique(media_channel),
>-      "vcname", kDefaultRtcpMuxRequired, kDefaultSrtpRequired);
>-  AddOutgoingVideoTrackStats(&stats);
>-
>-  // Instruct the session to return stats containing the transport channel.
>-  InitSessionStats(&video_channel);
>-  EXPECT_CALL(pc_, GetSessionStats(_))
>-      .WillRepeatedly(Invoke([this](const ChannelNamePairs&) {
>-        return std::unique_ptr<SessionStats>(
>-            new SessionStats(session_stats_));
>-      }));
>-
>-  // Constructs an ssrc stats update.
>-  cricket::VideoMediaInfo stats_read;
>-
>-  cricket::SsrcReceiverInfo remote_ssrc_stats;
>+  SsrcReceiverInfo remote_ssrc_stats;
>   remote_ssrc_stats.timestamp = 12345.678;
>   remote_ssrc_stats.ssrc = kSsrcOfTrack;
>-  cricket::VideoSenderInfo video_sender_info;
>+  VideoSenderInfo video_sender_info;
>   video_sender_info.add_ssrc(kSsrcOfTrack);
>   video_sender_info.remote_stats.push_back(remote_ssrc_stats);
>-  stats_read.senders.push_back(video_sender_info);
>+  VideoMediaInfo video_info;
>+  video_info.senders.push_back(video_sender_info);
>+
>+  auto* video_media_channel = pc->AddVideoChannel("video", "transport");
>+  video_media_channel->SetStats(video_info);
> 
>-  EXPECT_CALL(pc_, video_channel()).WillRepeatedly(Return(&video_channel));
>-  EXPECT_CALL(pc_, voice_channel()).WillRepeatedly(ReturnNull());
>-  EXPECT_CALL(*media_channel, GetStats(_))
>-    .WillRepeatedly(DoAll(SetArgPointee<0>(stats_read),
>-                          Return(true)));
>+  AddOutgoingVideoTrack(pc, stats.get());
> 
>-  stats.UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
>+  stats->UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
>   StatsReports reports;
>-  stats.GetStats(NULL, &reports);
>+  stats->GetStats(nullptr, &reports);
> 
>-  const StatsReport* remote_report = FindNthReportByType(reports,
>-      StatsReport::kStatsReportTypeRemoteSsrc, 1);
>-  EXPECT_FALSE(remote_report == NULL);
>+  const StatsReport* remote_report =
>+      FindNthReportByType(reports, StatsReport::kStatsReportTypeRemoteSsrc, 1);
>+  ASSERT_TRUE(remote_report);
>   EXPECT_EQ(12345.678, remote_report->timestamp());
> }
> 
> // This test verifies that the empty track report exists in the returned stats
> // when StatsCollector::UpdateStats is called with ssrc stats.
> TEST_P(StatsCollectorTrackTest, ReportsFromRemoteTrack) {
>-  StatsCollectorForTest stats(&pc_);
>+  constexpr int64_t kNumOfPacketsConcealed = 54321;
> 
>-  EXPECT_CALL(pc_, GetLocalCertificate(_, _)).WillRepeatedly(Return(false));
>-  EXPECT_CALL(pc_, GetRemoteSSLCertificate_ReturnsRawPointer(_))
>-      .WillRepeatedly(Return(nullptr));
>+  auto pc = CreatePeerConnection();
>+  auto stats = CreateStatsCollector(pc);
> 
>-  auto* media_channel = new MockVideoMediaChannel();
>-  cricket::VideoChannel video_channel(
>-      worker_thread_, network_thread_, nullptr, rtc::WrapUnique(media_channel),
>-      "video", kDefaultRtcpMuxRequired, kDefaultSrtpRequired);
>-
>-  InitSessionStats(&video_channel);
>-  EXPECT_CALL(pc_, GetSessionStats(_))
>-      .WillRepeatedly(Invoke([this](const ChannelNamePairs&) {
>-        return std::unique_ptr<SessionStats>(
>-            new SessionStats(session_stats_));
>-      }));
>-
>-  AddIncomingVideoTrackStats(&stats);
>-
>-  // Constructs an ssrc stats update.
>-  cricket::VideoReceiverInfo video_receiver_info;
>-  cricket::VideoMediaInfo stats_read;
>-  const int64_t kNumOfPacketsConcealed = 54321;
>-
>-  // Construct a stats value to read.
>+  VideoReceiverInfo video_receiver_info;
>   video_receiver_info.add_ssrc(1234);
>   video_receiver_info.packets_concealed = kNumOfPacketsConcealed;
>-  stats_read.receivers.push_back(video_receiver_info);
>+  VideoMediaInfo video_info;
>+  video_info.receivers.push_back(video_receiver_info);
>+
>+  auto* video_media_info = pc->AddVideoChannel("video", "transport");
>+  video_media_info->SetStats(video_info);
> 
>-  EXPECT_CALL(pc_, video_channel()).WillRepeatedly(Return(&video_channel));
>-  EXPECT_CALL(pc_, voice_channel()).WillRepeatedly(ReturnNull());
>-  EXPECT_CALL(*media_channel, GetStats(_))
>-      .WillOnce(DoAll(SetArgPointee<0>(stats_read),
>-                      Return(true)));
>+  AddIncomingVideoTrack(pc, stats.get());
> 
>-  stats.UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
>+  stats->UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
>   StatsReports reports;
>-  stats.GetStats(NULL, &reports);
>+  stats->GetStats(nullptr, &reports);
>+
>   // |reports| should contain at least one session report, one track report,
>   // and one ssrc report.
>-  EXPECT_LE(static_cast<size_t>(3), reports.size());
>-  const StatsReport* track_report = FindNthReportByType(
>-      reports, StatsReport::kStatsReportTypeTrack, 1);
>+  EXPECT_LE(3u, reports.size());
>+  const StatsReport* track_report =
>+      FindNthReportByType(reports, StatsReport::kStatsReportTypeTrack, 1);
>   ASSERT_TRUE(track_report);
>-  EXPECT_EQ(stats.GetTimeNow(), track_report->timestamp());
>+  EXPECT_EQ(stats->GetTimeNow(), track_report->timestamp());
> 
>-  std::string ssrc_id = ExtractSsrcStatsValue(
>-      reports, StatsReport::kStatsValueNameSsrc);
>-  EXPECT_EQ(rtc::ToString<uint32_t>(kSsrcOfTrack), ssrc_id);
>+  std::string ssrc_id =
>+      ExtractSsrcStatsValue(reports, StatsReport::kStatsValueNameSsrc);
>+  EXPECT_EQ(rtc::ToString(kSsrcOfTrack), ssrc_id);
> 
>-  std::string track_id = ExtractSsrcStatsValue(
>-      reports, StatsReport::kStatsValueNameTrackId);
>+  std::string track_id =
>+      ExtractSsrcStatsValue(reports, StatsReport::kStatsValueNameTrackId);
>   EXPECT_EQ(kRemoteTrackId, track_id);
> }
> 
> // This test verifies the Ice Candidate report should contain the correct
> // information from local/remote candidates.
> TEST_F(StatsCollectorTest, IceCandidateReport) {
>-  StatsCollectorForTest stats(&pc_);
>-
>-  StatsReports reports;                     // returned values.
>-
>-  const int local_port = 2000;
>-  const char local_ip[] = "192.168.0.1";
>-  const int remote_port = 2001;
>-  const char remote_ip[] = "192.168.0.2";
>-
>-  rtc::SocketAddress local_address(local_ip, local_port);
>-  rtc::SocketAddress remote_address(remote_ip, remote_port);
>-  rtc::AdapterType network_type = rtc::ADAPTER_TYPE_ETHERNET;
>-  uint32_t priority = 1000;
>-
>-  cricket::Candidate c;
>-  EXPECT_GT(c.id().length(), 0u);
>-  c.set_type(cricket::LOCAL_PORT_TYPE);
>-  c.set_protocol(cricket::UDP_PROTOCOL_NAME);
>-  c.set_address(local_address);
>-  c.set_priority(priority);
>-  c.set_network_type(network_type);
>-  std::string report_id = AddCandidateReport(&stats, c, true)->id()->ToString();
>-  EXPECT_EQ("Cand-" + c.id(), report_id);
>-
>-  c = cricket::Candidate();
>-  EXPECT_GT(c.id().length(), 0u);
>-  c.set_type(cricket::PRFLX_PORT_TYPE);
>-  c.set_protocol(cricket::UDP_PROTOCOL_NAME);
>-  c.set_address(remote_address);
>-  c.set_priority(priority);
>-  c.set_network_type(network_type);
>-  report_id = AddCandidateReport(&stats, c, false)->id()->ToString();
>-  EXPECT_EQ("Cand-" + c.id(), report_id);
>-
>-  stats.GetStats(NULL, &reports);
>+  const std::string kTransportName = "transport";
>+  const rtc::AdapterType kNetworkType = rtc::ADAPTER_TYPE_ETHERNET;
>+  constexpr uint32_t kPriority = 1000;
>+
>+  constexpr int kLocalPort = 2000;
>+  const std::string kLocalIp = "192.168.0.1";
>+  const rtc::SocketAddress kLocalAddress(kLocalIp, kLocalPort);
>+
>+  constexpr int kRemotePort = 2001;
>+  const std::string kRemoteIp = "192.168.0.2";
>+  const rtc::SocketAddress kRemoteAddress(kRemoteIp, kRemotePort);
>+
>+  auto pc = CreatePeerConnection();
>+  auto stats = CreateStatsCollector(pc);
>+
>+  cricket::Candidate local;
>+  EXPECT_GT(local.id().length(), 0u);
>+  local.set_type(cricket::LOCAL_PORT_TYPE);
>+  local.set_protocol(cricket::UDP_PROTOCOL_NAME);
>+  local.set_address(kLocalAddress);
>+  local.set_priority(kPriority);
>+  local.set_network_type(kNetworkType);
>+
>+  cricket::Candidate remote;
>+  EXPECT_GT(remote.id().length(), 0u);
>+  remote.set_type(cricket::PRFLX_PORT_TYPE);
>+  remote.set_protocol(cricket::UDP_PROTOCOL_NAME);
>+  remote.set_address(kRemoteAddress);
>+  remote.set_priority(kPriority);
>+  remote.set_network_type(kNetworkType);
>+
>+  ConnectionInfo connection_info;
>+  connection_info.local_candidate = local;
>+  connection_info.remote_candidate = remote;
>+  TransportChannelStats channel_stats;
>+  channel_stats.connection_infos.push_back(connection_info);
>+
>+  pc->AddVoiceChannel("audio", kTransportName);
>+  pc->SetTransportStats(kTransportName, channel_stats);
>+
>+  stats->UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
>+  StatsReports reports;
>+  stats->GetStats(nullptr, &reports);
> 
>   // Verify the local candidate report is populated correctly.
>   EXPECT_EQ(
>-      local_ip,
>+      "Cand-" + local.id(),
>+      ExtractStatsValue(StatsReport::kStatsReportTypeCandidatePair, reports,
>+                        StatsReport::kStatsValueNameLocalCandidateId));
>+  EXPECT_EQ(
>+      kLocalIp,
>       ExtractStatsValue(StatsReport::kStatsReportTypeIceLocalCandidate, reports,
>                         StatsReport::kStatsValueNameCandidateIPAddress));
>   EXPECT_EQ(
>-      rtc::ToString<int>(local_port),
>+      rtc::ToString(kLocalPort),
>       ExtractStatsValue(StatsReport::kStatsReportTypeIceLocalCandidate, reports,
>                         StatsReport::kStatsValueNameCandidatePortNumber));
>   EXPECT_EQ(
>@@ -1524,7 +1252,7 @@ TEST_F(StatsCollectorTest, IceCandidateReport) {
>       ExtractStatsValue(StatsReport::kStatsReportTypeIceLocalCandidate, reports,
>                         StatsReport::kStatsValueNameCandidateTransportType));
>   EXPECT_EQ(
>-      rtc::ToString<int>(priority),
>+      rtc::ToString(kPriority),
>       ExtractStatsValue(StatsReport::kStatsReportTypeIceLocalCandidate, reports,
>                         StatsReport::kStatsValueNameCandidatePriority));
>   EXPECT_EQ(
>@@ -1532,16 +1260,20 @@ TEST_F(StatsCollectorTest, IceCandidateReport) {
>       ExtractStatsValue(StatsReport::kStatsReportTypeIceLocalCandidate, reports,
>                         StatsReport::kStatsValueNameCandidateType));
>   EXPECT_EQ(
>-      AdapterTypeToStatsType(network_type),
>+      AdapterTypeToStatsType(kNetworkType),
>       ExtractStatsValue(StatsReport::kStatsReportTypeIceLocalCandidate, reports,
>                         StatsReport::kStatsValueNameCandidateNetworkType));
> 
>   // Verify the remote candidate report is populated correctly.
>-  EXPECT_EQ(remote_ip,
>+  EXPECT_EQ(
>+      "Cand-" + remote.id(),
>+      ExtractStatsValue(StatsReport::kStatsReportTypeCandidatePair, reports,
>+                        StatsReport::kStatsValueNameRemoteCandidateId));
>+  EXPECT_EQ(kRemoteIp,
>             ExtractStatsValue(StatsReport::kStatsReportTypeIceRemoteCandidate,
>                               reports,
>                               StatsReport::kStatsValueNameCandidateIPAddress));
>-  EXPECT_EQ(rtc::ToString<int>(remote_port),
>+  EXPECT_EQ(rtc::ToString(kRemotePort),
>             ExtractStatsValue(StatsReport::kStatsReportTypeIceRemoteCandidate,
>                               reports,
>                               StatsReport::kStatsValueNameCandidatePortNumber));
>@@ -1549,7 +1281,7 @@ TEST_F(StatsCollectorTest, IceCandidateReport) {
>             ExtractStatsValue(
>                 StatsReport::kStatsReportTypeIceRemoteCandidate, reports,
>                 StatsReport::kStatsValueNameCandidateTransportType));
>-  EXPECT_EQ(rtc::ToString<int>(priority),
>+  EXPECT_EQ(rtc::ToString(kPriority),
>             ExtractStatsValue(StatsReport::kStatsReportTypeIceRemoteCandidate,
>                               reports,
>                               StatsReport::kStatsValueNameCandidatePriority));
>@@ -1573,7 +1305,7 @@ TEST_F(StatsCollectorTest, ChainedCertificateReportsCreated) {
>   local_ders[2] = "some";
>   local_ders[3] = "der";
>   local_ders[4] = "values";
>-  rtc::FakeSSLCertificate local_cert(DersToPems(local_ders));
>+  rtc::FakeSSLIdentity local_identity(DersToPems(local_ders));
> 
>   // Build remote certificate chain
>   std::vector<std::string> remote_ders(4);
>@@ -1581,10 +1313,9 @@ TEST_F(StatsCollectorTest, ChainedCertificateReportsCreated) {
>   remote_ders[1] = "non-";
>   remote_ders[2] = "intersecting";
>   remote_ders[3] = "set";
>-  std::unique_ptr<rtc::FakeSSLCertificate> remote_cert(
>-      new rtc::FakeSSLCertificate(DersToPems(remote_ders)));
>+  rtc::FakeSSLIdentity remote_identity(DersToPems(remote_ders));
> 
>-  TestCertificateReports(local_cert, local_ders, std::move(remote_cert),
>+  TestCertificateReports(local_identity, local_ders, remote_identity,
>                          remote_ders);
> }
> 
>@@ -1593,63 +1324,41 @@ TEST_F(StatsCollectorTest, ChainedCertificateReportsCreated) {
> TEST_F(StatsCollectorTest, ChainlessCertificateReportsCreated) {
>   // Build local certificate.
>   std::string local_der = "This is the local der.";
>-  rtc::FakeSSLCertificate local_cert(DerToPem(local_der));
>+  rtc::FakeSSLIdentity local_identity(DerToPem(local_der));
> 
>   // Build remote certificate.
>   std::string remote_der = "This is somebody else's der.";
>-  std::unique_ptr<rtc::FakeSSLCertificate> remote_cert(
>-      new rtc::FakeSSLCertificate(DerToPem(remote_der)));
>+  rtc::FakeSSLIdentity remote_identity(DerToPem(remote_der));
> 
>-  TestCertificateReports(local_cert, std::vector<std::string>(1, local_der),
>-                         std::move(remote_cert),
>+  TestCertificateReports(local_identity, std::vector<std::string>(1, local_der),
>+                         remote_identity,
>                          std::vector<std::string>(1, remote_der));
> }
> 
> // This test verifies that the stats are generated correctly when no
> // transport is present.
> TEST_F(StatsCollectorTest, NoTransport) {
>-  StatsCollectorForTest stats(&pc_);
>-
>-  EXPECT_CALL(pc_, GetLocalCertificate(_, _)).WillRepeatedly(Return(false));
>-  EXPECT_CALL(pc_, GetRemoteSSLCertificate_ReturnsRawPointer(_))
>-      .WillRepeatedly(Return(nullptr));
>+  auto pc = CreatePeerConnection();
>+  auto stats = CreateStatsCollector(pc);
> 
>-  StatsReports reports;  // returned values.
>-
>-  // Fake stats to process.
>-  cricket::TransportChannelStats channel_stats;
>-  channel_stats.component = 1;
>-
>-  cricket::TransportStats transport_stats;
>-  transport_stats.transport_name = "audio";
>-  transport_stats.channel_stats.push_back(channel_stats);
>-
>-  SessionStats session_stats;
>-  session_stats.transport_stats[transport_stats.transport_name] =
>-      transport_stats;
>+  // This will cause the fake PeerConnection to generate a TransportStats entry
>+  // but with only a single dummy TransportChannelStats.
>+  pc->AddVoiceChannel("audio", "transport");
> 
>-  // Configure MockWebRtcSession
>-  EXPECT_CALL(pc_, GetSessionStats(_))
>-      .WillRepeatedly(Invoke([&session_stats](const ChannelNamePairs&) {
>-        return std::unique_ptr<SessionStats>(
>-            new SessionStats(session_stats));
>-      }));
>-
>-  stats.UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
>-  stats.GetStats(NULL, &reports);
>+  stats->UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
>+  StatsReports reports;
>+  stats->GetStats(nullptr, &reports);
> 
>   // Check that the local certificate is absent.
>-  std::string local_certificate_id = ExtractStatsValue(
>-      StatsReport::kStatsReportTypeComponent,
>-      reports,
>-      StatsReport::kStatsValueNameLocalCertificateId);
>+  std::string local_certificate_id =
>+      ExtractStatsValue(StatsReport::kStatsReportTypeComponent, reports,
>+                        StatsReport::kStatsValueNameLocalCertificateId);
>   ASSERT_EQ(kNotFound, local_certificate_id);
> 
>   // Check that the remote certificate is absent.
>-  std::string remote_certificate_id = ExtractStatsValue(
>-      StatsReport::kStatsReportTypeComponent,
>-      reports,
>-      StatsReport::kStatsValueNameRemoteCertificateId);
>+  std::string remote_certificate_id =
>+      ExtractStatsValue(StatsReport::kStatsReportTypeComponent, reports,
>+                        StatsReport::kStatsValueNameRemoteCertificateId);
>   ASSERT_EQ(kNotFound, remote_certificate_id);
> 
>   // Check that the negotiated ciphers are absent.
>@@ -1663,122 +1372,57 @@ TEST_F(StatsCollectorTest, NoTransport) {
>   ASSERT_EQ(kNotFound, srtp_crypto_suite);
> }
> 
>-// This test verifies that the stats are generated correctly when the transport
>-// does not have any certificates.
>-TEST_F(StatsCollectorTest, NoCertificates) {
>-  StatsCollectorForTest stats(&pc_);
>-
>-  EXPECT_CALL(pc_, GetLocalCertificate(_, _)).WillRepeatedly(Return(false));
>-  EXPECT_CALL(pc_, GetRemoteSSLCertificate_ReturnsRawPointer(_))
>-      .WillRepeatedly(Return(nullptr));
>-
>-  StatsReports reports;  // returned values.
>-
>-  // Fake stats to process.
>-  cricket::TransportChannelStats channel_stats;
>-  channel_stats.component = 1;
>-
>-  cricket::TransportStats transport_stats;
>-  transport_stats.transport_name = "audio";
>-  transport_stats.channel_stats.push_back(channel_stats);
>-
>-  SessionStats session_stats;
>-  session_stats.transport_stats[transport_stats.transport_name] =
>-      transport_stats;
>-
>-  // Configure MockWebRtcSession
>-  EXPECT_CALL(pc_, GetSessionStats(_))
>-      .WillRepeatedly(Invoke([&session_stats](const ChannelNamePairs&) {
>-        return std::unique_ptr<SessionStats>(
>-            new SessionStats(session_stats));
>-      }));
>-  stats.UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
>-  stats.GetStats(NULL, &reports);
>-
>-  // Check that the local certificate is absent.
>-  std::string local_certificate_id = ExtractStatsValue(
>-      StatsReport::kStatsReportTypeComponent,
>-      reports,
>-      StatsReport::kStatsValueNameLocalCertificateId);
>-  ASSERT_EQ(kNotFound, local_certificate_id);
>-
>-  // Check that the remote certificate is absent.
>-  std::string remote_certificate_id = ExtractStatsValue(
>-      StatsReport::kStatsReportTypeComponent,
>-      reports,
>-      StatsReport::kStatsValueNameRemoteCertificateId);
>-  ASSERT_EQ(kNotFound, remote_certificate_id);
>-}
>-
> // This test verifies that a remote certificate with an unsupported digest
> // algorithm is correctly ignored.
> TEST_F(StatsCollectorTest, UnsupportedDigestIgnored) {
>   // Build a local certificate.
>   std::string local_der = "This is the local der.";
>-  rtc::FakeSSLCertificate local_cert(DerToPem(local_der));
>+  rtc::FakeSSLIdentity local_identity(DerToPem(local_der));
> 
>   // Build a remote certificate with an unsupported digest algorithm.
>   std::string remote_der = "This is somebody else's der.";
>-  std::unique_ptr<rtc::FakeSSLCertificate> remote_cert(
>-      new rtc::FakeSSLCertificate(DerToPem(remote_der)));
>-  remote_cert->set_digest_algorithm("foobar");
>+  rtc::FakeSSLCertificate remote_cert(DerToPem(remote_der));
>+  remote_cert.set_digest_algorithm("foobar");
>+  rtc::FakeSSLIdentity remote_identity(remote_cert);
> 
>-  TestCertificateReports(local_cert, std::vector<std::string>(1, local_der),
>-                         std::move(remote_cert), std::vector<std::string>());
>+  TestCertificateReports(local_identity, std::vector<std::string>(1, local_der),
>+                         remote_identity, std::vector<std::string>());
> }
> 
> // This test verifies that the audio/video related stats which are -1 initially
> // will be filtered out.
> TEST_P(StatsCollectorTrackTest, FilterOutNegativeInitialValues) {
>-  StatsCollectorForTest stats(&pc_);
>-
>   // This test uses streams, but only works for the stream case.
>-  if (!GetParam())
>+  if (!GetParam()) {
>     return;
>-  EXPECT_CALL(pc_, GetLocalCertificate(_, _)).WillRepeatedly(Return(false));
>-  EXPECT_CALL(pc_, GetRemoteSSLCertificate_ReturnsRawPointer(_))
>-      .WillRepeatedly(Return(nullptr));
>+  }
> 
>-  auto* media_channel = new MockVoiceMediaChannel();
>-  cricket::VoiceChannel voice_channel(
>-      worker_thread_, network_thread_, nullptr, media_engine_,
>-      rtc::WrapUnique(media_channel), "vcname", kDefaultRtcpMuxRequired,
>-      kDefaultSrtpRequired);
>+  auto pc = CreatePeerConnection();
>+  auto stats = CreateStatsCollector(pc);
> 
>   // Create a local stream with a local audio track and adds it to the stats.
>-  if (stream_ == NULL)
>-    stream_ = webrtc::MediaStream::Create("streamlabel");
>-
>+  stream_ = MediaStream::Create("streamid");
>   rtc::scoped_refptr<FakeAudioTrackWithInitValue> local_track(
>       new rtc::RefCountedObject<FakeAudioTrackWithInitValue>(kLocalTrackId));
>   stream_->AddTrack(local_track);
>-  EXPECT_CALL(pc_, GetLocalTrackIdBySsrc(kSsrcOfTrack, _))
>-      .WillOnce(DoAll(SetArgPointee<1>(kLocalTrackId), Return(true)));
>+  pc->AddLocalTrack(kSsrcOfTrack, kLocalTrackId);
>   if (GetParam()) {
>-    stats.AddStream(stream_);
>+    stats->AddStream(stream_);
>   }
>-  stats.AddLocalAudioTrack(local_track.get(), kSsrcOfTrack);
>+  stats->AddLocalAudioTrack(local_track.get(), kSsrcOfTrack);
> 
>   // Create a remote stream with a remote audio track and adds it to the stats.
>-  rtc::scoped_refptr<webrtc::MediaStream> remote_stream(
>-      webrtc::MediaStream::Create("remotestreamlabel"));
>+  rtc::scoped_refptr<MediaStream> remote_stream(
>+      MediaStream::Create("remotestreamid"));
>   rtc::scoped_refptr<FakeAudioTrackWithInitValue> remote_track(
>       new rtc::RefCountedObject<FakeAudioTrackWithInitValue>(kRemoteTrackId));
>-  EXPECT_CALL(pc_, GetRemoteTrackIdBySsrc(kSsrcOfTrack, _))
>-      .WillOnce(DoAll(SetArgPointee<1>(kRemoteTrackId), Return(true)));
>   remote_stream->AddTrack(remote_track);
>+  pc->AddRemoteTrack(kSsrcOfTrack, kRemoteTrackId);
>   if (GetParam()) {
>-    stats.AddStream(remote_stream);
>+    stats->AddStream(remote_stream);
>   }
> 
>-  InitSessionStats(&voice_channel);
>-  EXPECT_CALL(pc_, GetSessionStats(_))
>-      .WillRepeatedly(Invoke([this](const ChannelNamePairs&) {
>-        return std::unique_ptr<SessionStats>(
>-            new SessionStats(session_stats_));
>-      }));
>-
>-  cricket::VoiceSenderInfo voice_sender_info;
>+  VoiceSenderInfo voice_sender_info;
>   voice_sender_info.add_ssrc(kSsrcOfTrack);
>   // These values are set to -1 initially in audio_send_stream.
>   // The voice_sender_info will read the values from audio_send_stream.
>@@ -1791,26 +1435,24 @@ TEST_P(StatsCollectorTrackTest, FilterOutNegativeInitialValues) {
>   UpdateVoiceSenderInfoFromAudioTrack(local_track.get(), &voice_sender_info,
>                                       true);
> 
>-  cricket::VoiceReceiverInfo voice_receiver_info;
>+  VoiceReceiverInfo voice_receiver_info;
>   voice_receiver_info.add_ssrc(kSsrcOfTrack);
>   voice_receiver_info.capture_start_ntp_time_ms = -1;
>   voice_receiver_info.audio_level = -1;
> 
>   // Constructs an ssrc stats update.
>-  cricket::VoiceMediaInfo stats_read;
>-  stats_read.senders.push_back(voice_sender_info);
>-  stats_read.receivers.push_back(voice_receiver_info);
>+  VoiceMediaInfo voice_info;
>+  voice_info.senders.push_back(voice_sender_info);
>+  voice_info.receivers.push_back(voice_receiver_info);
> 
>-  EXPECT_CALL(pc_, voice_channel()).WillRepeatedly(Return(&voice_channel));
>-  EXPECT_CALL(pc_, video_channel()).WillRepeatedly(ReturnNull());
>-  EXPECT_CALL(*media_channel, GetStats(_))
>-      .WillRepeatedly(DoAll(SetArgPointee<0>(stats_read), Return(true)));
>+  auto* voice_media_channel = pc->AddVoiceChannel("voice", "transport");
>+  voice_media_channel->SetStats(voice_info);
> 
>-  StatsReports reports;
>-  stats.UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
>+  stats->UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
> 
>   // Get stats for the local track.
>-  stats.GetStats(local_track.get(), &reports);
>+  StatsReports reports;
>+  stats->GetStats(local_track.get(), &reports);
>   const StatsReport* report =
>       FindNthReportByType(reports, StatsReport::kStatsReportTypeSsrc, 1);
>   ASSERT_TRUE(report);
>@@ -1829,7 +1471,7 @@ TEST_P(StatsCollectorTrackTest, FilterOutNegativeInitialValues) {
> 
>   // Get stats for the remote track.
>   reports.clear();
>-  stats.GetStats(remote_track.get(), &reports);
>+  stats->GetStats(remote_track.get(), &reports);
>   report = FindNthReportByType(reports, StatsReport::kStatsReportTypeSsrc, 1);
>   ASSERT_TRUE(report);
>   EXPECT_FALSE(GetValue(report,
>@@ -1842,116 +1484,86 @@ TEST_P(StatsCollectorTrackTest, FilterOutNegativeInitialValues) {
> // This test verifies that a local stats object can get statistics via
> // AudioTrackInterface::GetStats() method.
> TEST_P(StatsCollectorTrackTest, GetStatsFromLocalAudioTrack) {
>-  StatsCollectorForTest stats(&pc_);
>-
>-  EXPECT_CALL(pc_, GetLocalCertificate(_, _)).WillRepeatedly(Return(false));
>-  EXPECT_CALL(pc_, GetRemoteSSLCertificate_ReturnsRawPointer(_))
>-      .WillRepeatedly(Return(nullptr));
>+  auto pc = CreatePeerConnection();
>+  auto stats = CreateStatsCollector(pc);
> 
>-  auto* media_channel = new MockVoiceMediaChannel();
>-  cricket::VoiceChannel voice_channel(
>-      worker_thread_, network_thread_, nullptr, media_engine_,
>-      rtc::WrapUnique(media_channel), "vcname", kDefaultRtcpMuxRequired,
>-      kDefaultSrtpRequired);
>-  AddOutgoingAudioTrackStats(&stats);
>-  stats.AddLocalAudioTrack(audio_track_, kSsrcOfTrack);
>+  AddOutgoingAudioTrack(pc, stats.get());
>+  stats->AddLocalAudioTrack(audio_track_, kSsrcOfTrack);
> 
>-  cricket::VoiceSenderInfo voice_sender_info;
>+  VoiceSenderInfo voice_sender_info;
>   InitVoiceSenderInfo(&voice_sender_info);
>+  UpdateVoiceSenderInfoFromAudioTrack(audio_track_, &voice_sender_info, false);
>+  VoiceMediaInfo voice_info;
>+  voice_info.senders.push_back(voice_sender_info);
>+
>+  auto* voice_media_channel = pc->AddVoiceChannel("audio", "transport");
>+  voice_media_channel->SetStats(voice_info);
> 
>-  cricket::VoiceMediaInfo stats_read;
>   StatsReports reports;  // returned values.
>-  SetupAndVerifyAudioTrackStats(
>-      audio_track_.get(), stream_.get(), &stats, &voice_channel, media_channel,
>-      &voice_sender_info, NULL, &stats_read, &reports);
>+  VerifyAudioTrackStats(audio_track_, stats.get(), voice_info, &reports);
> 
>   // Verify that there is no remote report for the local audio track because
>   // we did not set it up.
>-  const StatsReport* remote_report = FindNthReportByType(reports,
>-      StatsReport::kStatsReportTypeRemoteSsrc, 1);
>+  const StatsReport* remote_report =
>+      FindNthReportByType(reports, StatsReport::kStatsReportTypeRemoteSsrc, 1);
>   EXPECT_TRUE(remote_report == NULL);
> }
> 
> // This test verifies that audio receive streams populate stats reports
> // correctly.
> TEST_P(StatsCollectorTrackTest, GetStatsFromRemoteStream) {
>-  StatsCollectorForTest stats(&pc_);
>-
>-  EXPECT_CALL(pc_, GetLocalCertificate(_, _)).WillRepeatedly(Return(false));
>-  EXPECT_CALL(pc_, GetRemoteSSLCertificate_ReturnsRawPointer(_))
>-      .WillRepeatedly(Return(nullptr));
>+  auto pc = CreatePeerConnection();
>+  auto stats = CreateStatsCollector(pc);
> 
>-  auto* media_channel = new MockVoiceMediaChannel();
>-  cricket::VoiceChannel voice_channel(
>-      worker_thread_, network_thread_, nullptr, media_engine_,
>-      rtc::WrapUnique(media_channel), "vcname", kDefaultRtcpMuxRequired,
>-      kDefaultSrtpRequired);
>-  AddIncomingAudioTrackStats(&stats);
>+  AddIncomingAudioTrack(pc, stats.get());
> 
>-  cricket::VoiceReceiverInfo voice_receiver_info;
>+  VoiceReceiverInfo voice_receiver_info;
>   InitVoiceReceiverInfo(&voice_receiver_info);
>   voice_receiver_info.codec_name = "fake_codec";
>+  VoiceMediaInfo voice_info;
>+  voice_info.receivers.push_back(voice_receiver_info);
>+
>+  auto* voice_media_channel = pc->AddVoiceChannel("audio", "transport");
>+  voice_media_channel->SetStats(voice_info);
> 
>-  cricket::VoiceMediaInfo stats_read;
>   StatsReports reports;  // returned values.
>-  SetupAndVerifyAudioTrackStats(audio_track_.get(), stream_.get(), &stats,
>-                                &voice_channel, media_channel, NULL,
>-                                &voice_receiver_info, &stats_read, &reports);
>+  VerifyAudioTrackStats(audio_track_, stats.get(), voice_info, &reports);
> }
> 
> // This test verifies that a local stats object won't update its statistics
> // after a RemoveLocalAudioTrack() call.
> TEST_P(StatsCollectorTrackTest, GetStatsAfterRemoveAudioStream) {
>-  StatsCollectorForTest stats(&pc_);
>-
>-  EXPECT_CALL(pc_, GetLocalCertificate(_, _)).WillRepeatedly(Return(false));
>-  EXPECT_CALL(pc_, GetRemoteSSLCertificate_ReturnsRawPointer(_))
>-      .WillRepeatedly(Return(nullptr));
>-
>-  auto* media_channel = new MockVoiceMediaChannel();
>-  cricket::VoiceChannel voice_channel(
>-      worker_thread_, network_thread_, nullptr, media_engine_,
>-      rtc::WrapUnique(media_channel), "vcname", kDefaultRtcpMuxRequired,
>-      kDefaultSrtpRequired);
>-  AddOutgoingAudioTrackStats(&stats);
>-  stats.AddLocalAudioTrack(audio_track_.get(), kSsrcOfTrack);
>-
>-  InitSessionStats(&voice_channel);
>-  EXPECT_CALL(pc_, GetSessionStats(_))
>-      .WillRepeatedly(Invoke([this](const ChannelNamePairs&) {
>-        return std::unique_ptr<SessionStats>(
>-            new SessionStats(session_stats_));
>-      }));
>-
>-  stats.RemoveLocalAudioTrack(audio_track_.get(), kSsrcOfTrack);
>-  cricket::VoiceSenderInfo voice_sender_info;
>+  auto pc = CreatePeerConnection();
>+  auto stats = CreateStatsCollector(pc);
>+
>+  AddOutgoingAudioTrack(pc, stats.get());
>+  stats->AddLocalAudioTrack(audio_track_.get(), kSsrcOfTrack);
>+
>+  VoiceSenderInfo voice_sender_info;
>   InitVoiceSenderInfo(&voice_sender_info);
>+  VoiceMediaInfo voice_info;
>+  voice_info.senders.push_back(voice_sender_info);
> 
>-  // Constructs an ssrc stats update.
>-  cricket::VoiceMediaInfo stats_read;
>-  stats_read.senders.push_back(voice_sender_info);
>+  auto* voice_media_channel = pc->AddVoiceChannel("audio", "transport");
>+  voice_media_channel->SetStats(voice_info);
> 
>-  EXPECT_CALL(pc_, voice_channel()).WillRepeatedly(Return(&voice_channel));
>-  EXPECT_CALL(pc_, video_channel()).WillRepeatedly(ReturnNull());
>-  EXPECT_CALL(*media_channel, GetStats(_))
>-      .WillRepeatedly(DoAll(SetArgPointee<0>(stats_read),
>-                            Return(true)));
>+  stats->RemoveLocalAudioTrack(audio_track_.get(), kSsrcOfTrack);
> 
>-  StatsReports reports;  // returned values.
>-  stats.UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
>-  stats.GetStats(NULL, &reports);
>+  stats->UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
>+  StatsReports reports;
>+  stats->GetStats(nullptr, &reports);
> 
>   // The report will exist since we don't remove them in RemoveStream().
>-  const StatsReport* report = FindNthReportByType(
>-      reports, StatsReport::kStatsReportTypeSsrc, 1);
>-  EXPECT_FALSE(report == NULL);
>-  EXPECT_EQ(stats.GetTimeNow(), report->timestamp());
>-  std::string track_id = ExtractSsrcStatsValue(
>-      reports, StatsReport::kStatsValueNameTrackId);
>+  const StatsReport* report =
>+      FindNthReportByType(reports, StatsReport::kStatsReportTypeSsrc, 1);
>+  ASSERT_TRUE(report);
>+  EXPECT_EQ(stats->GetTimeNow(), report->timestamp());
>+  std::string track_id =
>+      ExtractSsrcStatsValue(reports, StatsReport::kStatsValueNameTrackId);
>   EXPECT_EQ(kLocalTrackId, track_id);
>-  std::string ssrc_id = ExtractSsrcStatsValue(
>-      reports, StatsReport::kStatsValueNameSsrc);
>-  EXPECT_EQ(rtc::ToString<uint32_t>(kSsrcOfTrack), ssrc_id);
>+  std::string ssrc_id =
>+      ExtractSsrcStatsValue(reports, StatsReport::kStatsValueNameSsrc);
>+  EXPECT_EQ(rtc::ToString(kSsrcOfTrack), ssrc_id);
> 
>   // Verifies the values in the track report, no value will be changed by the
>   // AudioTrackInterface::GetSignalValue() and
>@@ -1962,85 +1574,65 @@ TEST_P(StatsCollectorTrackTest, GetStatsAfterRemoveAudioStream) {
> // This test verifies that when ongoing and incoming audio tracks are using
> // the same ssrc, they populate stats reports correctly.
> TEST_P(StatsCollectorTrackTest, LocalAndRemoteTracksWithSameSsrc) {
>-  StatsCollectorForTest stats(&pc_);
>-
>-  EXPECT_CALL(pc_, GetLocalCertificate(_, _)).WillRepeatedly(Return(false));
>-  EXPECT_CALL(pc_, GetRemoteSSLCertificate_ReturnsRawPointer(_))
>-      .WillRepeatedly(Return(nullptr));
>-
>-  auto* media_channel = new MockVoiceMediaChannel();
>-  cricket::VoiceChannel voice_channel(
>-      worker_thread_, network_thread_, nullptr, media_engine_,
>-      rtc::WrapUnique(media_channel), "vcname", kDefaultRtcpMuxRequired,
>-      kDefaultSrtpRequired);
>+  auto pc = CreatePeerConnection();
>+  auto stats = CreateStatsCollector(pc);
> 
>   // Create a local stream with a local audio track and adds it to the stats.
>-  AddOutgoingAudioTrackStats(&stats);
>-  stats.AddLocalAudioTrack(audio_track_.get(), kSsrcOfTrack);
>+  AddOutgoingAudioTrack(pc, stats.get());
>+  stats->AddLocalAudioTrack(audio_track_.get(), kSsrcOfTrack);
> 
>   // Create a remote stream with a remote audio track and adds it to the stats.
>-  rtc::scoped_refptr<webrtc::MediaStream> remote_stream(
>-      webrtc::MediaStream::Create("remotestreamlabel"));
>+  rtc::scoped_refptr<MediaStream> remote_stream(
>+      MediaStream::Create("remotestreamid"));
>   rtc::scoped_refptr<FakeAudioTrack> remote_track(
>       new rtc::RefCountedObject<FakeAudioTrack>(kRemoteTrackId));
>-  EXPECT_CALL(pc_, GetRemoteTrackIdBySsrc(kSsrcOfTrack, _))
>-      .WillOnce(DoAll(SetArgPointee<1>(kRemoteTrackId), Return(true)));
>+  pc->AddRemoteTrack(kSsrcOfTrack, kRemoteTrackId);
>   remote_stream->AddTrack(remote_track);
>-  stats.AddStream(remote_stream);
>+  stats->AddStream(remote_stream);
> 
>-  // Instruct the session to return stats containing the transport channel.
>-  InitSessionStats(&voice_channel);
>-  EXPECT_CALL(pc_, GetSessionStats(_))
>-      .WillRepeatedly(Invoke([this](const ChannelNamePairs&) {
>-        return std::unique_ptr<SessionStats>(
>-            new SessionStats(session_stats_));
>-      }));
>-
>-  cricket::VoiceSenderInfo voice_sender_info;
>+  VoiceSenderInfo voice_sender_info;
>   InitVoiceSenderInfo(&voice_sender_info);
>-
>   // Some of the contents in |voice_sender_info| needs to be updated from the
>   // |audio_track_|.
>   UpdateVoiceSenderInfoFromAudioTrack(audio_track_.get(), &voice_sender_info,
>                                       true);
> 
>-  cricket::VoiceReceiverInfo voice_receiver_info;
>+  VoiceReceiverInfo voice_receiver_info;
>   InitVoiceReceiverInfo(&voice_receiver_info);
> 
>   // Constructs an ssrc stats update.
>-  cricket::VoiceMediaInfo stats_read;
>-  stats_read.senders.push_back(voice_sender_info);
>-  stats_read.receivers.push_back(voice_receiver_info);
>+  VoiceMediaInfo voice_info;
>+  voice_info.senders.push_back(voice_sender_info);
>+  voice_info.receivers.push_back(voice_receiver_info);
> 
>-  EXPECT_CALL(pc_, voice_channel()).WillRepeatedly(Return(&voice_channel));
>-  EXPECT_CALL(pc_, video_channel()).WillRepeatedly(ReturnNull());
>-  EXPECT_CALL(*media_channel, GetStats(_))
>-      .WillRepeatedly(DoAll(SetArgPointee<0>(stats_read),
>-                            Return(true)));
>+  // Instruct the session to return stats containing the transport channel.
>+  auto* voice_media_channel = pc->AddVoiceChannel("audio", "transport");
>+  voice_media_channel->SetStats(voice_info);
> 
>-  StatsReports reports;  // returned values.
>-  stats.UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
>+  stats->UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
> 
>   // Get stats for the local track.
>-  stats.GetStats(audio_track_.get(), &reports);
>-  const StatsReport* track_report = FindNthReportByType(
>-      reports, StatsReport::kStatsReportTypeSsrc, 1);
>+  StatsReports reports;  // returned values.
>+  stats->GetStats(audio_track_.get(), &reports);
>+
>+  const StatsReport* track_report =
>+      FindNthReportByType(reports, StatsReport::kStatsReportTypeSsrc, 1);
>   ASSERT_TRUE(track_report);
>-  EXPECT_EQ(stats.GetTimeNow(), track_report->timestamp());
>-  std::string track_id = ExtractSsrcStatsValue(
>-      reports, StatsReport::kStatsValueNameTrackId);
>+  EXPECT_EQ(stats->GetTimeNow(), track_report->timestamp());
>+  std::string track_id =
>+      ExtractSsrcStatsValue(reports, StatsReport::kStatsValueNameTrackId);
>   EXPECT_EQ(kLocalTrackId, track_id);
>   VerifyVoiceSenderInfoReport(track_report, voice_sender_info);
> 
>   // Get stats for the remote track.
>   reports.clear();
>-  stats.GetStats(remote_track.get(), &reports);
>-  track_report = FindNthReportByType(reports,
>-                                     StatsReport::kStatsReportTypeSsrc, 1);
>+  stats->GetStats(remote_track.get(), &reports);
>+  track_report =
>+      FindNthReportByType(reports, StatsReport::kStatsReportTypeSsrc, 1);
>   ASSERT_TRUE(track_report);
>-  EXPECT_EQ(stats.GetTimeNow(), track_report->timestamp());
>-  track_id = ExtractSsrcStatsValue(reports,
>-                                   StatsReport::kStatsValueNameTrackId);
>+  EXPECT_EQ(stats->GetTimeNow(), track_report->timestamp());
>+  track_id =
>+      ExtractSsrcStatsValue(reports, StatsReport::kStatsValueNameTrackId);
>   EXPECT_EQ(kRemoteTrackId, track_id);
>   VerifyVoiceReceiverInfoReport(track_report, voice_receiver_info);
> }
>@@ -2050,97 +1642,78 @@ TEST_P(StatsCollectorTrackTest, LocalAndRemoteTracksWithSameSsrc) {
> // TODO(xians): Figure out if it is possible to encapsulate the setup and
> // avoid duplication of code in test cases.
> TEST_P(StatsCollectorTrackTest, TwoLocalTracksWithSameSsrc) {
>-  StatsCollectorForTest stats(&pc_);
>-
>   // This test only makes sense when we're using streams.
>-  if (!GetParam())
>+  if (!GetParam()) {
>     return;
>+  }
> 
>-  EXPECT_CALL(pc_, GetLocalCertificate(_, _)).WillRepeatedly(Return(false));
>-  EXPECT_CALL(pc_, GetRemoteSSLCertificate_ReturnsRawPointer(_))
>-      .WillRepeatedly(Return(nullptr));
>-
>-  auto* media_channel = new MockVoiceMediaChannel();
>-  cricket::VoiceChannel voice_channel(
>-      worker_thread_, network_thread_, nullptr, media_engine_,
>-      rtc::WrapUnique(media_channel), "vcname", kDefaultRtcpMuxRequired,
>-      kDefaultSrtpRequired);
>+  auto pc = CreatePeerConnection();
>+  auto stats = CreateStatsCollector(pc);
> 
>   // Create a local stream with a local audio track and adds it to the stats.
>-  AddOutgoingAudioTrackStats(&stats);
>-  stats.AddLocalAudioTrack(audio_track_, kSsrcOfTrack);
>+  AddOutgoingAudioTrack(pc, stats.get());
>+  stats->AddLocalAudioTrack(audio_track_, kSsrcOfTrack);
> 
>-  cricket::VoiceSenderInfo voice_sender_info;
>+  VoiceSenderInfo voice_sender_info;
>   InitVoiceSenderInfo(&voice_sender_info);
>+  UpdateVoiceSenderInfoFromAudioTrack(audio_track_, &voice_sender_info, false);
>   voice_sender_info.add_ssrc(kSsrcOfTrack);
>+  VoiceMediaInfo voice_info;
>+  voice_info.senders.push_back(voice_sender_info);
>+
>+  auto* voice_media_channel = pc->AddVoiceChannel("voice", "transport");
>+  voice_media_channel->SetStats(voice_info);
> 
>-  cricket::VoiceMediaInfo stats_read;
>   StatsReports reports;  // returned values.
>-  SetupAndVerifyAudioTrackStats(
>-      audio_track_.get(), stream_.get(), &stats, &voice_channel, media_channel,
>-      &voice_sender_info, NULL, &stats_read, &reports);
>+  VerifyAudioTrackStats(audio_track_, stats.get(), voice_info, &reports);
> 
>   // Remove the previous audio track from the stream.
>   stream_->RemoveTrack(audio_track_.get());
>-  stats.RemoveLocalAudioTrack(audio_track_.get(), kSsrcOfTrack);
>+  stats->RemoveLocalAudioTrack(audio_track_.get(), kSsrcOfTrack);
> 
>   // Create a new audio track and adds it to the stream and stats.
>   static const std::string kNewTrackId = "new_track_id";
>   rtc::scoped_refptr<FakeAudioTrack> new_audio_track(
>       new rtc::RefCountedObject<FakeAudioTrack>(kNewTrackId));
>-  EXPECT_CALL(pc_, GetLocalTrackIdBySsrc(kSsrcOfTrack, _))
>-      .WillOnce(DoAll(SetArgPointee<1>(kNewTrackId), Return(true)));
>+  pc->AddLocalTrack(kSsrcOfTrack, kNewTrackId);
>   stream_->AddTrack(new_audio_track);
> 
>-  stats.AddLocalAudioTrack(new_audio_track, kSsrcOfTrack);
>-  stats.ClearUpdateStatsCacheForTest();
>-  cricket::VoiceSenderInfo new_voice_sender_info;
>+  stats->AddLocalAudioTrack(new_audio_track, kSsrcOfTrack);
>+  stats->ClearUpdateStatsCacheForTest();
>+
>+  VoiceSenderInfo new_voice_sender_info;
>   InitVoiceSenderInfo(&new_voice_sender_info);
>-  cricket::VoiceMediaInfo new_stats_read;
>+  UpdateVoiceSenderInfoFromAudioTrack(new_audio_track, &new_voice_sender_info,
>+                                      false);
>+  VoiceMediaInfo new_voice_info;
>+  new_voice_info.senders.push_back(new_voice_sender_info);
>+  voice_media_channel->SetStats(new_voice_info);
>+
>   reports.clear();
>-  SetupAndVerifyAudioTrackStats(
>-      new_audio_track.get(), stream_.get(), &stats, &voice_channel,
>-      media_channel, &new_voice_sender_info, NULL, &new_stats_read, &reports);
>+  VerifyAudioTrackStats(new_audio_track, stats.get(), new_voice_info, &reports);
> }
> 
> // This test verifies that stats are correctly set in video send ssrc stats.
> TEST_P(StatsCollectorTrackTest, VerifyVideoSendSsrcStats) {
>-  StatsCollectorForTest stats(&pc_);
>-
>-  EXPECT_CALL(pc_, GetLocalCertificate(_, _)).WillRepeatedly(Return(false));
>-  EXPECT_CALL(pc_, GetRemoteSSLCertificate_ReturnsRawPointer(_))
>-      .WillRepeatedly(Return(nullptr));
>-
>-  auto* media_channel = new MockVideoMediaChannel();
>-  cricket::VideoChannel video_channel(
>-      worker_thread_, network_thread_, nullptr, rtc::WrapUnique(media_channel),
>-      "video", kDefaultRtcpMuxRequired, kDefaultSrtpRequired);
>-
>-  InitSessionStats(&video_channel);
>-  EXPECT_CALL(pc_, GetSessionStats(_))
>-      .WillRepeatedly(Invoke([this](const ChannelNamePairs&) {
>-        return std::unique_ptr<SessionStats>(
>-            new SessionStats(session_stats_));
>-      }));
>+  auto pc = CreatePeerConnection();
>+  auto stats = CreateStatsCollector(pc);
> 
>-  StatsReports reports;  // returned values.
>-  cricket::VideoSenderInfo video_sender_info;
>-  cricket::VideoMediaInfo stats_read;
>-
>-  AddOutgoingVideoTrackStats(&stats);
>+  AddOutgoingVideoTrack(pc, stats.get());
> 
>-  // Construct a stats value to read.
>+  VideoSenderInfo video_sender_info;
>   video_sender_info.add_ssrc(1234);
>   video_sender_info.frames_encoded = 10;
>   video_sender_info.qp_sum = 11;
>-  stats_read.senders.push_back(video_sender_info);
>-
>-  EXPECT_CALL(pc_, video_channel()).WillRepeatedly(Return(&video_channel));
>-  EXPECT_CALL(pc_, voice_channel()).WillRepeatedly(ReturnNull());
>-  EXPECT_CALL(*media_channel, GetStats(_))
>-      .WillOnce(DoAll(SetArgPointee<0>(stats_read), Return(true)));
>-  stats.UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
>-  stats.GetStats(NULL, &reports);
>+  VideoMediaInfo video_info;
>+  video_info.senders.push_back(video_sender_info);
>+
>+  auto* video_media_channel = pc->AddVideoChannel("video", "transport");
>+  video_media_channel->SetStats(video_info);
>+
>+  stats->UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
>+  StatsReports reports;
>+  stats->GetStats(nullptr, &reports);
>+
>   EXPECT_EQ(rtc::ToString(video_sender_info.frames_encoded),
>             ExtractSsrcStatsValue(reports,
>                                   StatsReport::kStatsValueNameFramesEncoded));
>@@ -2149,43 +1722,26 @@ TEST_P(StatsCollectorTrackTest, VerifyVideoSendSsrcStats) {
> }
> 
> // This test verifies that stats are correctly set in video receive ssrc stats.
>-TEST_P(StatsCollectorTrackTest, VerifyVideoReceiveSsrcStats) {
>-  StatsCollectorForTest stats(&pc_);
>-
>-  EXPECT_CALL(pc_, GetLocalCertificate(_, _)).WillRepeatedly(Return(false));
>-  EXPECT_CALL(pc_, GetRemoteSSLCertificate_ReturnsRawPointer(_))
>-      .WillRepeatedly(Return(nullptr));
>-
>-  auto* media_channel = new MockVideoMediaChannel();
>-  cricket::VideoChannel video_channel(
>-      worker_thread_, network_thread_, nullptr, rtc::WrapUnique(media_channel),
>-      "video", kDefaultRtcpMuxRequired, kDefaultSrtpRequired);
>+TEST_P(StatsCollectorTrackTest, VerifyVideoReceiveSsrcStatsNew) {
>+  auto pc = CreatePeerConnection();
>+  auto stats = CreateStatsCollector(pc);
> 
>-  InitSessionStats(&video_channel);
>-  EXPECT_CALL(pc_, GetSessionStats(_))
>-      .WillRepeatedly(Invoke([this](const ChannelNamePairs&) {
>-        return std::unique_ptr<SessionStats>(
>-            new SessionStats(session_stats_));
>-      }));
>+  AddIncomingVideoTrack(pc, stats.get());
> 
>-  StatsReports reports;  // returned values.
>-  cricket::VideoReceiverInfo video_receiver_info;
>-  cricket::VideoMediaInfo stats_read;
>-
>-  AddIncomingVideoTrackStats(&stats);
>-
>-  // Construct a stats value to read.
>+  VideoReceiverInfo video_receiver_info;
>   video_receiver_info.add_ssrc(1234);
>   video_receiver_info.frames_decoded = 10;
>   video_receiver_info.qp_sum = 11;
>-  stats_read.receivers.push_back(video_receiver_info);
>-
>-  EXPECT_CALL(pc_, video_channel()).WillRepeatedly(Return(&video_channel));
>-  EXPECT_CALL(pc_, voice_channel()).WillRepeatedly(ReturnNull());
>-  EXPECT_CALL(*media_channel, GetStats(_))
>-      .WillOnce(DoAll(SetArgPointee<0>(stats_read), Return(true)));
>-  stats.UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
>-  stats.GetStats(NULL, &reports);
>+  VideoMediaInfo video_info;
>+  video_info.receivers.push_back(video_receiver_info);
>+
>+  auto* video_media_channel = pc->AddVideoChannel("video", "transport");
>+  video_media_channel->SetStats(video_info);
>+
>+  stats->UpdateStats(PeerConnectionInterface::kStatsOutputLevelStandard);
>+  StatsReports reports;
>+  stats->GetStats(nullptr, &reports);
>+
>   EXPECT_EQ(rtc::ToString(video_receiver_info.frames_decoded),
>             ExtractSsrcStatsValue(reports,
>                                   StatsReport::kStatsValueNameFramesDecoded));
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/streamcollection.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/streamcollection.h
>index 19ddca991f9..76317f91e1a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/streamcollection.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/streamcollection.h
>@@ -23,37 +23,34 @@ class StreamCollection : public StreamCollectionInterface {
>  public:
>   static rtc::scoped_refptr<StreamCollection> Create() {
>     rtc::RefCountedObject<StreamCollection>* implementation =
>-         new rtc::RefCountedObject<StreamCollection>();
>+        new rtc::RefCountedObject<StreamCollection>();
>     return implementation;
>   }
> 
>   static rtc::scoped_refptr<StreamCollection> Create(
>       StreamCollection* streams) {
>     rtc::RefCountedObject<StreamCollection>* implementation =
>-         new rtc::RefCountedObject<StreamCollection>(streams);
>+        new rtc::RefCountedObject<StreamCollection>(streams);
>     return implementation;
>   }
> 
>-  virtual size_t count() {
>-    return media_streams_.size();
>-  }
>+  virtual size_t count() { return media_streams_.size(); }
> 
>   virtual MediaStreamInterface* at(size_t index) {
>     return media_streams_.at(index);
>   }
> 
>-  virtual MediaStreamInterface* find(const std::string& label) {
>+  virtual MediaStreamInterface* find(const std::string& id) {
>     for (StreamVector::iterator it = media_streams_.begin();
>          it != media_streams_.end(); ++it) {
>-      if ((*it)->label().compare(label) == 0) {
>+      if ((*it)->id().compare(id) == 0) {
>         return (*it);
>       }
>     }
>     return NULL;
>   }
> 
>-  virtual MediaStreamTrackInterface* FindAudioTrack(
>-      const std::string& id) {
>+  virtual MediaStreamTrackInterface* FindAudioTrack(const std::string& id) {
>     for (size_t i = 0; i < media_streams_.size(); ++i) {
>       MediaStreamTrackInterface* track = media_streams_[i]->FindAudioTrack(id);
>       if (track) {
>@@ -63,8 +60,7 @@ class StreamCollection : public StreamCollectionInterface {
>     return NULL;
>   }
> 
>-  virtual MediaStreamTrackInterface* FindVideoTrack(
>-      const std::string& id) {
>+  virtual MediaStreamTrackInterface* FindVideoTrack(const std::string& id) {
>     for (size_t i = 0; i < media_streams_.size(); ++i) {
>       MediaStreamTrackInterface* track = media_streams_[i]->FindVideoTrack(id);
>       if (track) {
>@@ -77,7 +73,7 @@ class StreamCollection : public StreamCollectionInterface {
>   void AddStream(MediaStreamInterface* stream) {
>     for (StreamVector::iterator it = media_streams_.begin();
>          it != media_streams_.end(); ++it) {
>-      if ((*it)->label().compare(stream->label()) == 0)
>+      if ((*it)->id().compare(stream->id()) == 0)
>         return;
>     }
>     media_streams_.push_back(stream);
>@@ -86,7 +82,7 @@ class StreamCollection : public StreamCollectionInterface {
>   void RemoveStream(MediaStreamInterface* remove_stream) {
>     for (StreamVector::iterator it = media_streams_.begin();
>          it != media_streams_.end(); ++it) {
>-      if ((*it)->label().compare(remove_stream->label()) == 0) {
>+      if ((*it)->id().compare(remove_stream->id()) == 0) {
>         media_streams_.erase(it);
>         break;
>       }
>@@ -96,10 +92,8 @@ class StreamCollection : public StreamCollectionInterface {
>  protected:
>   StreamCollection() {}
>   explicit StreamCollection(StreamCollection* original)
>-      : media_streams_(original->media_streams_) {
>-  }
>-  typedef std::vector<rtc::scoped_refptr<MediaStreamInterface> >
>-      StreamVector;
>+      : media_streams_(original->media_streams_) {}
>+  typedef std::vector<rtc::scoped_refptr<MediaStreamInterface> > StreamVector;
>   StreamVector media_streams_;
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/androidtestinitializer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/androidtestinitializer.cc
>index f39a3e7dd63..a67716dc23e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/androidtestinitializer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/androidtestinitializer.cc
>@@ -12,7 +12,6 @@
> 
> #include <pthread.h>
> 
>-
> #include "rtc_base/ignore_wundef.h"
> 
> // Note: this dependency is dangerous since it reaches into Chromium's base.
>@@ -23,7 +22,6 @@ RTC_PUSH_IGNORING_WUNDEF()
> #include "base/android/jni_android.h"
> RTC_POP_IGNORING_WUNDEF()
> 
>-
> #include "modules/utility/include/jvm_android.h"
> #include "rtc_base/checks.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/fakeaudiocapturemodule.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/fakeaudiocapturemodule.cc
>index a129ae5f739..c9c19f10271 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/fakeaudiocapturemodule.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/fakeaudiocapturemodule.cc
>@@ -45,8 +45,7 @@ FakeAudioCaptureModule::FakeAudioCaptureModule()
>       current_mic_level_(kMaxVolume),
>       started_(false),
>       next_frame_time_(0),
>-      frames_received_(0) {
>-}
>+      frames_received_(0) {}
> 
> FakeAudioCaptureModule::~FakeAudioCaptureModule() {
>   if (process_thread_) {
>@@ -426,7 +425,8 @@ bool FakeAudioCaptureModule::CheckRecBuffer(int value) {
>   const size_t buffer_size_in_samples =
>       sizeof(rec_buffer_) / kNumberBytesPerSample;
>   for (size_t i = 0; i < buffer_size_in_samples; ++i) {
>-    if (buffer_ptr[i] >= value) return true;
>+    if (buffer_ptr[i] >= value)
>+      return true;
>   }
>   return false;
> }
>@@ -496,10 +496,10 @@ void FakeAudioCaptureModule::ReceiveFrameP() {
>     size_t nSamplesOut = 0;
>     int64_t elapsed_time_ms = 0;
>     int64_t ntp_time_ms = 0;
>-    if (audio_callback_->NeedMorePlayData(kNumberSamples, kNumberBytesPerSample,
>-                                         kNumberOfChannels, kSamplesPerSecond,
>-                                         rec_buffer_, nSamplesOut,
>-                                         &elapsed_time_ms, &ntp_time_ms) != 0) {
>+    if (audio_callback_->NeedMorePlayData(
>+            kNumberSamples, kNumberBytesPerSample, kNumberOfChannels,
>+            kSamplesPerSecond, rec_buffer_, nSamplesOut, &elapsed_time_ms,
>+            &ntp_time_ms) != 0) {
>       RTC_NOTREACHED();
>     }
>     RTC_CHECK(nSamplesOut == kNumberSamples);
>@@ -525,13 +525,10 @@ void FakeAudioCaptureModule::SendFrameP() {
>   bool key_pressed = false;
>   uint32_t current_mic_level = 0;
>   MicrophoneVolume(&current_mic_level);
>-  if (audio_callback_->RecordedDataIsAvailable(send_buffer_, kNumberSamples,
>-                                              kNumberBytesPerSample,
>-                                              kNumberOfChannels,
>-                                              kSamplesPerSecond, kTotalDelayMs,
>-                                              kClockDriftMs, current_mic_level,
>-                                              key_pressed,
>-                                              current_mic_level) != 0) {
>+  if (audio_callback_->RecordedDataIsAvailable(
>+          send_buffer_, kNumberSamples, kNumberBytesPerSample,
>+          kNumberOfChannels, kSamplesPerSecond, kTotalDelayMs, kClockDriftMs,
>+          current_mic_level, key_pressed, current_mic_level) != 0) {
>     RTC_NOTREACHED();
>   }
>   SetMicrophoneVolume(current_mic_level);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/fakeaudiocapturemodule.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/fakeaudiocapturemodule.h
>index db346d06c8d..2f56ef4d5d3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/fakeaudiocapturemodule.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/fakeaudiocapturemodule.h
>@@ -24,7 +24,6 @@
> 
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/audio_device/include/audio_device.h"
>-#include "rtc_base/basictypes.h"
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/messagehandler.h"
> #include "rtc_base/scoped_ref_ptr.h"
>@@ -33,9 +32,8 @@ namespace rtc {
> class Thread;
> }  // namespace rtc
> 
>-class FakeAudioCaptureModule
>-    : public webrtc::AudioDeviceModule,
>-      public rtc::MessageHandler {
>+class FakeAudioCaptureModule : public webrtc::AudioDeviceModule,
>+                               public rtc::MessageHandler {
>  public:
>   typedef uint16_t Sample;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/fakeaudiocapturemodule_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/fakeaudiocapturemodule_unittest.cc
>index 0e7574f4c64..6ef76853474 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/fakeaudiocapturemodule_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/fakeaudiocapturemodule_unittest.cc
>@@ -18,15 +18,12 @@
> 
> using std::min;
> 
>-class FakeAdmTest : public testing::Test,
>-                    public webrtc::AudioTransport {
>+class FakeAdmTest : public testing::Test, public webrtc::AudioTransport {
>  protected:
>   static const int kMsInSecond = 1000;
> 
>   FakeAdmTest()
>-      : push_iterations_(0),
>-        pull_iterations_(0),
>-        rec_buffer_bytes_(0) {
>+      : push_iterations_(0), pull_iterations_(0), rec_buffer_bytes_(0) {
>     memset(rec_buffer_, 0, sizeof(rec_buffer_));
>   }
> 
>@@ -50,8 +47,9 @@ class FakeAdmTest : public testing::Test,
>     rtc::CritScope cs(&crit_);
>     rec_buffer_bytes_ = nSamples * nBytesPerSample;
>     if ((rec_buffer_bytes_ == 0) ||
>-        (rec_buffer_bytes_ > FakeAudioCaptureModule::kNumberSamples *
>-         FakeAudioCaptureModule::kNumberBytesPerSample)) {
>+        (rec_buffer_bytes_ >
>+         FakeAudioCaptureModule::kNumberSamples *
>+             FakeAudioCaptureModule::kNumberBytesPerSample)) {
>       ADD_FAILURE();
>       return -1;
>     }
>@@ -61,13 +59,6 @@ class FakeAdmTest : public testing::Test,
>     return 0;
>   }
> 
>-  void PushCaptureData(int voe_channel,
>-                       const void* audio_data,
>-                       int bits_per_sample,
>-                       int sample_rate,
>-                       size_t number_of_channels,
>-                       size_t number_of_frames) override {}
>-
>   void PullRenderData(int bits_per_sample,
>                       int sample_rate,
>                       size_t number_of_channels,
>@@ -88,9 +79,10 @@ class FakeAdmTest : public testing::Test,
>     rtc::CritScope cs(&crit_);
>     ++pull_iterations_;
>     const size_t audio_buffer_size = nSamples * nBytesPerSample;
>-    const size_t bytes_out = RecordedDataReceived() ?
>-        CopyFromRecBuffer(audioSamples, audio_buffer_size):
>-        GenerateZeroBuffer(audioSamples, audio_buffer_size);
>+    const size_t bytes_out =
>+        RecordedDataReceived()
>+            ? CopyFromRecBuffer(audioSamples, audio_buffer_size)
>+            : GenerateZeroBuffer(audioSamples, audio_buffer_size);
>     nSamplesOut = bytes_out / nBytesPerSample;
>     *elapsed_time_ms = 0;
>     *ntp_time_ms = 0;
>@@ -109,9 +101,7 @@ class FakeAdmTest : public testing::Test,
>   rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_;
> 
>  private:
>-  bool RecordedDataReceived() const {
>-    return rec_buffer_bytes_ != 0;
>-  }
>+  bool RecordedDataReceived() const { return rec_buffer_bytes_ != 0; }
>   size_t GenerateZeroBuffer(void* audio_buffer, size_t audio_buffer_size) {
>     memset(audio_buffer, 0, audio_buffer_size);
>     return audio_buffer_size;
>@@ -137,9 +127,8 @@ TEST_F(FakeAdmTest, PlayoutTest) {
>   EXPECT_EQ(0, fake_audio_capture_module_->RegisterAudioCallback(this));
> 
>   bool stereo_available = false;
>-  EXPECT_EQ(0,
>-            fake_audio_capture_module_->StereoPlayoutIsAvailable(
>-                &stereo_available));
>+  EXPECT_EQ(0, fake_audio_capture_module_->StereoPlayoutIsAvailable(
>+                   &stereo_available));
>   EXPECT_TRUE(stereo_available);
> 
>   EXPECT_NE(0, fake_audio_capture_module_->StartPlayout());
>@@ -170,7 +159,7 @@ TEST_F(FakeAdmTest, RecordTest) {
> 
>   bool stereo_available = false;
>   EXPECT_EQ(0, fake_audio_capture_module_->StereoRecordingIsAvailable(
>-      &stereo_available));
>+                   &stereo_available));
>   EXPECT_FALSE(stereo_available);
> 
>   EXPECT_NE(0, fake_audio_capture_module_->StartRecording());
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/fakedatachannelprovider.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/fakedatachannelprovider.h
>index 9aa02711760..83c5cb3924e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/fakedatachannelprovider.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/fakedatachannelprovider.h
>@@ -75,6 +75,14 @@ class FakeDataChannelProvider : public webrtc::DataChannelProviderInterface {
>     RTC_CHECK(sid >= 0);
>     send_ssrcs_.erase(sid);
>     recv_ssrcs_.erase(sid);
>+    // Unlike the real SCTP transport, act like the closing procedure finished
>+    // instantly, doing the same snapshot thing as below.
>+    for (webrtc::DataChannel* ch : std::set<webrtc::DataChannel*>(
>+             connected_channels_.begin(), connected_channels_.end())) {
>+      if (connected_channels_.count(ch)) {
>+        ch->OnClosingProcedureComplete(sid);
>+      }
>+    }
>   }
> 
>   bool ReadyToSendData() const override { return ready_to_send_; }
>@@ -87,7 +95,7 @@ class FakeDataChannelProvider : public webrtc::DataChannelProviderInterface {
>       // each value is still in connected_channels_ before calling
>       // OnChannelReady().  This avoids problems where the set gets modified
>       // in response to OnChannelReady().
>-      for (webrtc::DataChannel *ch : std::set<webrtc::DataChannel*>(
>+      for (webrtc::DataChannel* ch : std::set<webrtc::DataChannel*>(
>                connected_channels_.begin(), connected_channels_.end())) {
>         if (connected_channels_.count(ch)) {
>           ch->OnChannelReady(true);
>@@ -109,17 +117,14 @@ class FakeDataChannelProvider : public webrtc::DataChannelProviderInterface {
>     ready_to_send_ = ready;
>     if (ready) {
>       std::set<webrtc::DataChannel*>::iterator it;
>-      for (it = connected_channels_.begin();
>-           it != connected_channels_.end();
>+      for (it = connected_channels_.begin(); it != connected_channels_.end();
>            ++it) {
>         (*it)->OnChannelReady(true);
>       }
>     }
>   }
> 
>-  void set_transport_error() {
>-    transport_error_ = true;
>-  }
>+  void set_transport_error() { transport_error_ = true; }
> 
>   cricket::SendDataParams last_send_data_params() const {
>     return last_send_data_params_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/fakepeerconnectionbase.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/fakepeerconnectionbase.h
>new file mode 100644
>index 00000000000..db8596c84ef
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/fakepeerconnectionbase.h
>@@ -0,0 +1,305 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef PC_TEST_FAKEPEERCONNECTIONBASE_H_
>+#define PC_TEST_FAKEPEERCONNECTIONBASE_H_
>+
>+#include <map>
>+#include <memory>
>+#include <set>
>+#include <string>
>+#include <vector>
>+
>+#include "pc/peerconnectioninternal.h"
>+
>+namespace webrtc {
>+
>+// Customized PeerConnection fakes can be created by subclassing
>+// FakePeerConnectionBase then overriding the interesting methods. This class
>+// takes care of providing default implementations for all the pure virtual
>+// functions specified in the interfaces.
>+class FakePeerConnectionBase : public PeerConnectionInternal {
>+ public:
>+  // PeerConnectionInterface implementation.
>+
>+  rtc::scoped_refptr<StreamCollectionInterface> local_streams() override {
>+    return nullptr;
>+  }
>+
>+  rtc::scoped_refptr<StreamCollectionInterface> remote_streams() override {
>+    return nullptr;
>+  }
>+
>+  bool AddStream(MediaStreamInterface* stream) override { return false; }
>+
>+  void RemoveStream(MediaStreamInterface* stream) override {}
>+
>+  RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>> AddTrack(
>+      rtc::scoped_refptr<MediaStreamTrackInterface> track,
>+      const std::vector<std::string>& stream_ids) override {
>+    return RTCError(RTCErrorType::UNSUPPORTED_OPERATION, "Not implemented");
>+  }
>+
>+  bool RemoveTrack(RtpSenderInterface* sender) override { return false; }
>+
>+  RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> AddTransceiver(
>+      rtc::scoped_refptr<MediaStreamTrackInterface> track) override {
>+    return RTCError(RTCErrorType::UNSUPPORTED_OPERATION, "Not implemented");
>+  }
>+
>+  RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> AddTransceiver(
>+      rtc::scoped_refptr<MediaStreamTrackInterface> track,
>+      const RtpTransceiverInit& init) override {
>+    return RTCError(RTCErrorType::UNSUPPORTED_OPERATION, "Not implemented");
>+  }
>+
>+  RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> AddTransceiver(
>+      cricket::MediaType media_type) override {
>+    return RTCError(RTCErrorType::UNSUPPORTED_OPERATION, "Not implemented");
>+  }
>+
>+  RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> AddTransceiver(
>+      cricket::MediaType media_type,
>+      const RtpTransceiverInit& init) override {
>+    return RTCError(RTCErrorType::UNSUPPORTED_OPERATION, "Not implemented");
>+  }
>+
>+  rtc::scoped_refptr<RtpSenderInterface> CreateSender(
>+      const std::string& kind,
>+      const std::string& stream_id) override {
>+    return nullptr;
>+  }
>+
>+  std::vector<rtc::scoped_refptr<RtpSenderInterface>> GetSenders()
>+      const override {
>+    return {};
>+  }
>+
>+  std::vector<rtc::scoped_refptr<RtpReceiverInterface>> GetReceivers()
>+      const override {
>+    return {};
>+  }
>+
>+  std::vector<rtc::scoped_refptr<RtpTransceiverInterface>> GetTransceivers()
>+      const override {
>+    return {};
>+  }
>+
>+  bool GetStats(StatsObserver* observer,
>+                MediaStreamTrackInterface* track,
>+                StatsOutputLevel level) override {
>+    return false;
>+  }
>+
>+  void GetStats(RTCStatsCollectorCallback* callback) override {}
>+  void GetStats(
>+      rtc::scoped_refptr<RtpSenderInterface> selector,
>+      rtc::scoped_refptr<RTCStatsCollectorCallback> callback) override {}
>+  void GetStats(
>+      rtc::scoped_refptr<RtpReceiverInterface> selector,
>+      rtc::scoped_refptr<RTCStatsCollectorCallback> callback) override {}
>+
>+  void ClearStatsCache() override {}
>+
>+  rtc::scoped_refptr<DataChannelInterface> CreateDataChannel(
>+      const std::string& label,
>+      const DataChannelInit* config) override {
>+    return nullptr;
>+  }
>+
>+  const SessionDescriptionInterface* local_description() const override {
>+    return nullptr;
>+  }
>+  const SessionDescriptionInterface* remote_description() const override {
>+    return nullptr;
>+  }
>+
>+  const SessionDescriptionInterface* current_local_description()
>+      const override {
>+    return nullptr;
>+  }
>+  const SessionDescriptionInterface* current_remote_description()
>+      const override {
>+    return nullptr;
>+  }
>+
>+  const SessionDescriptionInterface* pending_local_description()
>+      const override {
>+    return nullptr;
>+  }
>+  const SessionDescriptionInterface* pending_remote_description()
>+      const override {
>+    return nullptr;
>+  }
>+
>+  void CreateOffer(CreateSessionDescriptionObserver* observer,
>+                   const MediaConstraintsInterface* constraints) override {}
>+
>+  void CreateOffer(CreateSessionDescriptionObserver* observer,
>+                   const RTCOfferAnswerOptions& options) override {}
>+
>+  void CreateAnswer(CreateSessionDescriptionObserver* observer,
>+                    const RTCOfferAnswerOptions& options) override {}
>+
>+  void CreateAnswer(CreateSessionDescriptionObserver* observer,
>+                    const MediaConstraintsInterface* constraints) override {}
>+
>+  void SetLocalDescription(SetSessionDescriptionObserver* observer,
>+                           SessionDescriptionInterface* desc) override {}
>+
>+  void SetRemoteDescription(SetSessionDescriptionObserver* observer,
>+                            SessionDescriptionInterface* desc) override {}
>+
>+  void SetRemoteDescription(
>+      std::unique_ptr<SessionDescriptionInterface> desc,
>+      rtc::scoped_refptr<SetRemoteDescriptionObserverInterface> observer)
>+      override {}
>+
>+  RTCConfiguration GetConfiguration() override { return RTCConfiguration(); }
>+
>+  bool SetConfiguration(const PeerConnectionInterface::RTCConfiguration& config,
>+                        RTCError* error) override {
>+    return false;
>+  }
>+
>+  bool SetConfiguration(
>+      const PeerConnectionInterface::RTCConfiguration& config) override {
>+    return false;
>+  }
>+
>+  bool AddIceCandidate(const IceCandidateInterface* candidate) override {
>+    return false;
>+  }
>+
>+  bool RemoveIceCandidates(
>+      const std::vector<cricket::Candidate>& candidates) override {
>+    return false;
>+  }
>+
>+  RTCError SetBitrate(const BitrateSettings& bitrate) override {
>+    return RTCError(RTCErrorType::UNSUPPORTED_OPERATION, "Not implemented");
>+  }
>+
>+  void SetBitrateAllocationStrategy(
>+      std::unique_ptr<rtc::BitrateAllocationStrategy>
>+          bitrate_allocation_strategy) override {}
>+
>+  void SetAudioPlayout(bool playout) override {}
>+
>+  void SetAudioRecording(bool recording) override {}
>+
>+  SignalingState signaling_state() override { return SignalingState::kStable; }
>+
>+  IceConnectionState ice_connection_state() override {
>+    return IceConnectionState::kIceConnectionNew;
>+  }
>+
>+  IceGatheringState ice_gathering_state() override {
>+    return IceGatheringState::kIceGatheringNew;
>+  }
>+
>+  bool StartRtcEventLog(rtc::PlatformFile file,
>+                        int64_t max_size_bytes) override {
>+    return false;
>+  }
>+
>+  bool StartRtcEventLog(std::unique_ptr<RtcEventLogOutput> output,
>+                        int64_t output_period_ms) override {
>+    return false;
>+  }
>+
>+  void StopRtcEventLog() override {}
>+
>+  void Close() override {}
>+
>+  // PeerConnectionInternal implementation.
>+
>+  rtc::Thread* network_thread() const override { return nullptr; }
>+  rtc::Thread* worker_thread() const override { return nullptr; }
>+  rtc::Thread* signaling_thread() const override { return nullptr; }
>+
>+  std::string session_id() const override { return ""; }
>+
>+  bool initial_offerer() const override { return false; }
>+
>+  std::vector<
>+      rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>>
>+  GetTransceiversInternal() const override {
>+    return {};
>+  }
>+
>+  bool GetLocalTrackIdBySsrc(uint32_t ssrc, std::string* track_id) override {
>+    return false;
>+  }
>+  bool GetRemoteTrackIdBySsrc(uint32_t ssrc, std::string* track_id) override {
>+    return false;
>+  }
>+
>+  sigslot::signal1<DataChannel*>& SignalDataChannelCreated() override {
>+    return SignalDataChannelCreated_;
>+  }
>+
>+  cricket::RtpDataChannel* rtp_data_channel() const override { return nullptr; }
>+
>+  std::vector<rtc::scoped_refptr<DataChannel>> sctp_data_channels()
>+      const override {
>+    return {};
>+  }
>+
>+  absl::optional<std::string> sctp_content_name() const override {
>+    return absl::nullopt;
>+  }
>+
>+  absl::optional<std::string> sctp_transport_name() const override {
>+    return absl::nullopt;
>+  }
>+
>+  std::map<std::string, std::string> GetTransportNamesByMid() const override {
>+    return {};
>+  }
>+
>+  std::map<std::string, cricket::TransportStats> GetTransportStatsByNames(
>+      const std::set<std::string>& transport_names) override {
>+    return {};
>+  }
>+
>+  Call::Stats GetCallStats() override { return Call::Stats(); }
>+
>+  bool GetLocalCertificate(
>+      const std::string& transport_name,
>+      rtc::scoped_refptr<rtc::RTCCertificate>* certificate) override {
>+    return false;
>+  }
>+
>+  std::unique_ptr<rtc::SSLCertChain> GetRemoteSSLCertChain(
>+      const std::string& transport_name) override {
>+    return nullptr;
>+  }
>+
>+  bool IceRestartPending(const std::string& content_name) const override {
>+    return false;
>+  }
>+
>+  bool NeedsIceRestart(const std::string& content_name) const override {
>+    return false;
>+  }
>+
>+  bool GetSslRole(const std::string& content_name,
>+                  rtc::SSLRole* role) override {
>+    return false;
>+  }
>+
>+ protected:
>+  sigslot::signal1<DataChannel*> SignalDataChannelCreated_;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // PC_TEST_FAKEPEERCONNECTIONBASE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/fakepeerconnectionforstats.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/fakepeerconnectionforstats.h
>new file mode 100644
>index 00000000000..642fa011eb4
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/fakepeerconnectionforstats.h
>@@ -0,0 +1,390 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef PC_TEST_FAKEPEERCONNECTIONFORSTATS_H_
>+#define PC_TEST_FAKEPEERCONNECTIONFORSTATS_H_
>+
>+#include <map>
>+#include <memory>
>+#include <set>
>+#include <string>
>+#include <utility>
>+#include <vector>
>+
>+#include "media/base/fakemediaengine.h"
>+#include "pc/streamcollection.h"
>+#include "pc/test/fakedatachannelprovider.h"
>+#include "pc/test/fakepeerconnectionbase.h"
>+
>+namespace webrtc {
>+
>+// Fake VoiceMediaChannel where the result of GetStats can be configured.
>+class FakeVoiceMediaChannelForStats : public cricket::FakeVoiceMediaChannel {
>+ public:
>+  FakeVoiceMediaChannelForStats()
>+      : cricket::FakeVoiceMediaChannel(nullptr, cricket::AudioOptions()) {}
>+
>+  void SetStats(const cricket::VoiceMediaInfo& voice_info) {
>+    stats_ = voice_info;
>+  }
>+
>+  // VoiceMediaChannel overrides.
>+  bool GetStats(cricket::VoiceMediaInfo* info) override {
>+    if (stats_) {
>+      *info = *stats_;
>+      return true;
>+    }
>+    return false;
>+  }
>+
>+ private:
>+  absl::optional<cricket::VoiceMediaInfo> stats_;
>+};
>+
>+// Fake VideoMediaChannel where the result of GetStats can be configured.
>+class FakeVideoMediaChannelForStats : public cricket::FakeVideoMediaChannel {
>+ public:
>+  FakeVideoMediaChannelForStats()
>+      : cricket::FakeVideoMediaChannel(nullptr, cricket::VideoOptions()) {}
>+
>+  void SetStats(const cricket::VideoMediaInfo& video_info) {
>+    stats_ = video_info;
>+  }
>+
>+  // VideoMediaChannel overrides.
>+  bool GetStats(cricket::VideoMediaInfo* info) override {
>+    if (stats_) {
>+      *info = *stats_;
>+      return true;
>+    }
>+    return false;
>+  }
>+
>+ private:
>+  absl::optional<cricket::VideoMediaInfo> stats_;
>+};
>+
>+constexpr bool kDefaultRtcpMuxRequired = true;
>+constexpr bool kDefaultSrtpRequired = true;
>+
>+// This class is intended to be fed into the StatsCollector and
>+// RTCStatsCollector so that the stats functionality can be unit tested.
>+// Individual tests can configure this fake as needed to simulate scenarios
>+// under which to test the stats collectors.
>+class FakePeerConnectionForStats : public FakePeerConnectionBase {
>+ public:
>+  // TODO(steveanton): Add support for specifying separate threads to test
>+  // multi-threading correctness.
>+  FakePeerConnectionForStats()
>+      : network_thread_(rtc::Thread::Current()),
>+        worker_thread_(rtc::Thread::Current()),
>+        signaling_thread_(rtc::Thread::Current()),
>+        local_streams_(StreamCollection::Create()),
>+        remote_streams_(StreamCollection::Create()) {}
>+
>+  rtc::scoped_refptr<StreamCollection> mutable_local_streams() {
>+    return local_streams_;
>+  }
>+
>+  rtc::scoped_refptr<StreamCollection> mutable_remote_streams() {
>+    return remote_streams_;
>+  }
>+
>+  void AddSender(rtc::scoped_refptr<RtpSenderInternal> sender) {
>+    // TODO(steveanton): Switch tests to use RtpTransceivers directly.
>+    auto sender_proxy = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
>+        signaling_thread_, sender);
>+    GetOrCreateFirstTransceiverOfType(sender->media_type())
>+        ->internal()
>+        ->AddSender(sender_proxy);
>+  }
>+
>+  void AddReceiver(rtc::scoped_refptr<RtpReceiverInternal> receiver) {
>+    // TODO(steveanton): Switch tests to use RtpTransceivers directly.
>+    auto receiver_proxy =
>+        RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
>+            signaling_thread_, receiver);
>+    GetOrCreateFirstTransceiverOfType(receiver->media_type())
>+        ->internal()
>+        ->AddReceiver(receiver_proxy);
>+  }
>+
>+  FakeVoiceMediaChannelForStats* AddVoiceChannel(
>+      const std::string& mid,
>+      const std::string& transport_name) {
>+    RTC_DCHECK(!voice_channel_);
>+    auto voice_media_channel =
>+        absl::make_unique<FakeVoiceMediaChannelForStats>();
>+    auto* voice_media_channel_ptr = voice_media_channel.get();
>+    voice_channel_ = absl::make_unique<cricket::VoiceChannel>(
>+        worker_thread_, network_thread_, signaling_thread_, nullptr,
>+        std::move(voice_media_channel), mid, kDefaultSrtpRequired,
>+        rtc::CryptoOptions());
>+    voice_channel_->set_transport_name_for_testing(transport_name);
>+    GetOrCreateFirstTransceiverOfType(cricket::MEDIA_TYPE_AUDIO)
>+        ->internal()
>+        ->SetChannel(voice_channel_.get());
>+    return voice_media_channel_ptr;
>+  }
>+
>+  FakeVideoMediaChannelForStats* AddVideoChannel(
>+      const std::string& mid,
>+      const std::string& transport_name) {
>+    RTC_DCHECK(!video_channel_);
>+    auto video_media_channel =
>+        absl::make_unique<FakeVideoMediaChannelForStats>();
>+    auto video_media_channel_ptr = video_media_channel.get();
>+    video_channel_ = absl::make_unique<cricket::VideoChannel>(
>+        worker_thread_, network_thread_, signaling_thread_,
>+        std::move(video_media_channel), mid, kDefaultSrtpRequired,
>+        rtc::CryptoOptions());
>+    video_channel_->set_transport_name_for_testing(transport_name);
>+    GetOrCreateFirstTransceiverOfType(cricket::MEDIA_TYPE_VIDEO)
>+        ->internal()
>+        ->SetChannel(video_channel_.get());
>+    return video_media_channel_ptr;
>+  }
>+
>+  void AddLocalTrack(uint32_t ssrc, const std::string& track_id) {
>+    local_track_id_by_ssrc_[ssrc] = track_id;
>+  }
>+
>+  void AddRemoteTrack(uint32_t ssrc, const std::string& track_id) {
>+    remote_track_id_by_ssrc_[ssrc] = track_id;
>+  }
>+
>+  void AddSctpDataChannel(const std::string& label) {
>+    AddSctpDataChannel(label, InternalDataChannelInit());
>+  }
>+
>+  void AddSctpDataChannel(const std::string& label,
>+                          const InternalDataChannelInit& init) {
>+    AddSctpDataChannel(DataChannel::Create(&data_channel_provider_,
>+                                           cricket::DCT_SCTP, label, init));
>+  }
>+
>+  void AddSctpDataChannel(rtc::scoped_refptr<DataChannel> data_channel) {
>+    sctp_data_channels_.push_back(data_channel);
>+  }
>+
>+  void SetTransportStats(const std::string& transport_name,
>+                         const cricket::TransportChannelStats& channel_stats) {
>+    SetTransportStats(
>+        transport_name,
>+        std::vector<cricket::TransportChannelStats>{channel_stats});
>+  }
>+
>+  void SetTransportStats(
>+      const std::string& transport_name,
>+      const std::vector<cricket::TransportChannelStats>& channel_stats_list) {
>+    cricket::TransportStats transport_stats;
>+    transport_stats.transport_name = transport_name;
>+    transport_stats.channel_stats = channel_stats_list;
>+    transport_stats_by_name_[transport_name] = transport_stats;
>+  }
>+
>+  void SetCallStats(const Call::Stats& call_stats) { call_stats_ = call_stats; }
>+
>+  void SetLocalCertificate(
>+      const std::string& transport_name,
>+      rtc::scoped_refptr<rtc::RTCCertificate> certificate) {
>+    local_certificates_by_transport_[transport_name] = certificate;
>+  }
>+
>+  void SetRemoteCertChain(const std::string& transport_name,
>+                          std::unique_ptr<rtc::SSLCertChain> chain) {
>+    remote_cert_chains_by_transport_[transport_name] = std::move(chain);
>+  }
>+
>+  // PeerConnectionInterface overrides.
>+
>+  rtc::scoped_refptr<StreamCollectionInterface> local_streams() override {
>+    return local_streams_;
>+  }
>+
>+  rtc::scoped_refptr<StreamCollectionInterface> remote_streams() override {
>+    return remote_streams_;
>+  }
>+
>+  std::vector<rtc::scoped_refptr<RtpSenderInterface>> GetSenders()
>+      const override {
>+    std::vector<rtc::scoped_refptr<RtpSenderInterface>> senders;
>+    for (auto transceiver : transceivers_) {
>+      for (auto sender : transceiver->internal()->senders()) {
>+        senders.push_back(sender);
>+      }
>+    }
>+    return senders;
>+  }
>+
>+  std::vector<rtc::scoped_refptr<RtpReceiverInterface>> GetReceivers()
>+      const override {
>+    std::vector<rtc::scoped_refptr<RtpReceiverInterface>> receivers;
>+    for (auto transceiver : transceivers_) {
>+      for (auto receiver : transceiver->internal()->receivers()) {
>+        receivers.push_back(receiver);
>+      }
>+    }
>+    return receivers;
>+  }
>+
>+  // PeerConnectionInternal overrides.
>+
>+  rtc::Thread* network_thread() const override { return network_thread_; }
>+
>+  rtc::Thread* worker_thread() const override { return worker_thread_; }
>+
>+  rtc::Thread* signaling_thread() const override { return signaling_thread_; }
>+
>+  std::vector<
>+      rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>>
>+  GetTransceiversInternal() const override {
>+    return transceivers_;
>+  }
>+
>+  bool GetLocalTrackIdBySsrc(uint32_t ssrc, std::string* track_id) override {
>+    auto it = local_track_id_by_ssrc_.find(ssrc);
>+    if (it != local_track_id_by_ssrc_.end()) {
>+      *track_id = it->second;
>+      return true;
>+    } else {
>+      return false;
>+    }
>+  }
>+
>+  bool GetRemoteTrackIdBySsrc(uint32_t ssrc, std::string* track_id) override {
>+    auto it = remote_track_id_by_ssrc_.find(ssrc);
>+    if (it != remote_track_id_by_ssrc_.end()) {
>+      *track_id = it->second;
>+      return true;
>+    } else {
>+      return false;
>+    }
>+  }
>+
>+  std::vector<rtc::scoped_refptr<DataChannel>> sctp_data_channels()
>+      const override {
>+    return sctp_data_channels_;
>+  }
>+
>+  cricket::CandidateStatsList GetPooledCandidateStats() const override {
>+    return {};
>+  }
>+
>+  std::map<std::string, std::string> GetTransportNamesByMid() const override {
>+    std::map<std::string, std::string> transport_names_by_mid;
>+    if (voice_channel_) {
>+      transport_names_by_mid[voice_channel_->content_name()] =
>+          voice_channel_->transport_name();
>+    }
>+    if (video_channel_) {
>+      transport_names_by_mid[video_channel_->content_name()] =
>+          video_channel_->transport_name();
>+    }
>+    return transport_names_by_mid;
>+  }
>+
>+  std::map<std::string, cricket::TransportStats> GetTransportStatsByNames(
>+      const std::set<std::string>& transport_names) override {
>+    std::map<std::string, cricket::TransportStats> transport_stats_by_name;
>+    for (const std::string& transport_name : transport_names) {
>+      transport_stats_by_name[transport_name] =
>+          GetTransportStatsByName(transport_name);
>+    }
>+    return transport_stats_by_name;
>+  }
>+
>+  Call::Stats GetCallStats() override { return call_stats_; }
>+
>+  bool GetLocalCertificate(
>+      const std::string& transport_name,
>+      rtc::scoped_refptr<rtc::RTCCertificate>* certificate) override {
>+    auto it = local_certificates_by_transport_.find(transport_name);
>+    if (it != local_certificates_by_transport_.end()) {
>+      *certificate = it->second;
>+      return true;
>+    } else {
>+      return false;
>+    }
>+  }
>+
>+  std::unique_ptr<rtc::SSLCertChain> GetRemoteSSLCertChain(
>+      const std::string& transport_name) override {
>+    auto it = remote_cert_chains_by_transport_.find(transport_name);
>+    if (it != remote_cert_chains_by_transport_.end()) {
>+      return it->second->UniqueCopy();
>+    } else {
>+      return nullptr;
>+    }
>+  }
>+
>+ private:
>+  cricket::TransportStats GetTransportStatsByName(
>+      const std::string& transport_name) {
>+    auto it = transport_stats_by_name_.find(transport_name);
>+    if (it != transport_stats_by_name_.end()) {
>+      // If specific transport stats have been specified, return those.
>+      return it->second;
>+    }
>+    // Otherwise, generate some dummy stats.
>+    cricket::TransportChannelStats channel_stats;
>+    channel_stats.component = cricket::ICE_CANDIDATE_COMPONENT_RTP;
>+    cricket::TransportStats transport_stats;
>+    transport_stats.transport_name = transport_name;
>+    transport_stats.channel_stats.push_back(channel_stats);
>+    return transport_stats;
>+  }
>+
>+  rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
>+  GetOrCreateFirstTransceiverOfType(cricket::MediaType media_type) {
>+    for (auto transceiver : transceivers_) {
>+      if (transceiver->internal()->media_type() == media_type) {
>+        return transceiver;
>+      }
>+    }
>+    auto transceiver = RtpTransceiverProxyWithInternal<RtpTransceiver>::Create(
>+        signaling_thread_, new RtpTransceiver(media_type));
>+    transceivers_.push_back(transceiver);
>+    return transceiver;
>+  }
>+
>+  rtc::Thread* const network_thread_;
>+  rtc::Thread* const worker_thread_;
>+  rtc::Thread* const signaling_thread_;
>+
>+  rtc::scoped_refptr<StreamCollection> local_streams_;
>+  rtc::scoped_refptr<StreamCollection> remote_streams_;
>+
>+  std::vector<
>+      rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>>
>+      transceivers_;
>+
>+  FakeDataChannelProvider data_channel_provider_;
>+
>+  std::unique_ptr<cricket::VoiceChannel> voice_channel_;
>+  std::unique_ptr<cricket::VideoChannel> video_channel_;
>+  std::map<uint32_t, std::string> local_track_id_by_ssrc_;
>+  std::map<uint32_t, std::string> remote_track_id_by_ssrc_;
>+
>+  std::vector<rtc::scoped_refptr<DataChannel>> sctp_data_channels_;
>+
>+  std::map<std::string, cricket::TransportStats> transport_stats_by_name_;
>+
>+  Call::Stats call_stats_;
>+
>+  std::map<std::string, rtc::scoped_refptr<rtc::RTCCertificate>>
>+      local_certificates_by_transport_;
>+  std::map<std::string, std::unique_ptr<rtc::SSLCertChain>>
>+      remote_cert_chains_by_transport_;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // PC_TEST_FAKEPEERCONNECTIONFORSTATS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/fakeperiodicvideocapturer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/fakeperiodicvideocapturer.h
>deleted file mode 100644
>index 38f2c75057c..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/fakeperiodicvideocapturer.h
>+++ /dev/null
>@@ -1,74 +0,0 @@
>-/*
>- *  Copyright 2012 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-// FakePeriodicVideoCapturer implements a fake cricket::VideoCapturer that
>-// creates video frames periodically after it has been started.
>-
>-#ifndef PC_TEST_FAKEPERIODICVIDEOCAPTURER_H_
>-#define PC_TEST_FAKEPERIODICVIDEOCAPTURER_H_
>-
>-#include <vector>
>-
>-#include "media/base/fakevideocapturer.h"
>-#include "rtc_base/thread.h"
>-
>-namespace webrtc {
>-
>-class FakePeriodicVideoCapturer : public cricket::FakeVideoCapturer,
>-                                  public rtc::MessageHandler {
>- public:
>-  FakePeriodicVideoCapturer() {
>-    std::vector<cricket::VideoFormat> formats;
>-    formats.push_back(cricket::VideoFormat(1280, 720,
>-            cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>-    formats.push_back(cricket::VideoFormat(640, 480,
>-        cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>-    formats.push_back(cricket::VideoFormat(640, 360,
>-            cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>-    formats.push_back(cricket::VideoFormat(320, 240,
>-        cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>-    formats.push_back(cricket::VideoFormat(160, 120,
>-        cricket::VideoFormat::FpsToInterval(30), cricket::FOURCC_I420));
>-    ResetSupportedFormats(formats);
>-  }
>-
>-  virtual cricket::CaptureState Start(const cricket::VideoFormat& format) {
>-    cricket::CaptureState state = FakeVideoCapturer::Start(format);
>-    if (state != cricket::CS_FAILED) {
>-      rtc::Thread::Current()->Post(RTC_FROM_HERE, this, MSG_CREATEFRAME);
>-    }
>-    return state;
>-  }
>-  virtual void Stop() {
>-    rtc::Thread::Current()->Clear(this);
>-  }
>-  // Inherited from MesageHandler.
>-  virtual void OnMessage(rtc::Message* msg) {
>-    if (msg->message_id == MSG_CREATEFRAME) {
>-      if (IsRunning()) {
>-        CaptureFrame();
>-        rtc::Thread::Current()->PostDelayed(
>-            RTC_FROM_HERE, static_cast<int>(GetCaptureFormat()->interval /
>-                                            rtc::kNumNanosecsPerMillisec),
>-            this, MSG_CREATEFRAME);
>-        }
>-    }
>-  }
>-
>- private:
>-  enum {
>-    // Offset  0xFF to make sure this don't collide with base class messages.
>-    MSG_CREATEFRAME = 0xFF
>-  };
>-};
>-
>-}  // namespace webrtc
>-
>-#endif  //  PC_TEST_FAKEPERIODICVIDEOCAPTURER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/fakeperiodicvideosource.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/fakeperiodicvideosource.h
>new file mode 100644
>index 00000000000..087deeb5439
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/fakeperiodicvideosource.h
>@@ -0,0 +1,100 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef PC_TEST_FAKEPERIODICVIDEOSOURCE_H_
>+#define PC_TEST_FAKEPERIODICVIDEOSOURCE_H_
>+
>+#include <memory>
>+
>+#include "absl/memory/memory.h"
>+#include "api/video/video_source_interface.h"
>+#include "media/base/fakeframesource.h"
>+#include "media/base/videobroadcaster.h"
>+#include "rtc_base/task_queue.h"
>+
>+namespace webrtc {
>+
>+class FakePeriodicVideoSource final
>+    : public rtc::VideoSourceInterface<VideoFrame> {
>+ public:
>+  static constexpr int kDefaultFrameIntervalMs = 33;
>+  static constexpr int kDefaultWidth = 640;
>+  static constexpr int kDefaultHeight = 480;
>+
>+  struct Config {
>+    int width = kDefaultWidth;
>+    int height = kDefaultHeight;
>+    int frame_interval_ms = kDefaultFrameIntervalMs;
>+    VideoRotation rotation = kVideoRotation_0;
>+  };
>+
>+  FakePeriodicVideoSource() : FakePeriodicVideoSource(Config()) {}
>+  explicit FakePeriodicVideoSource(Config config)
>+      : task_queue_(
>+            absl::make_unique<rtc::TaskQueue>("FakePeriodicVideoTrackSource")) {
>+    thread_checker_.DetachFromThread();
>+    task_queue_->PostTask(absl::make_unique<FrameTask>(config, &broadcaster_));
>+  }
>+
>+  void RemoveSink(rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) override {
>+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+    broadcaster_.RemoveSink(sink);
>+  }
>+
>+  void AddOrUpdateSink(rtc::VideoSinkInterface<webrtc::VideoFrame>* sink,
>+                       const rtc::VideoSinkWants& wants) override {
>+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+    broadcaster_.AddOrUpdateSink(sink, wants);
>+  }
>+
>+  void Stop() {
>+    RTC_DCHECK(task_queue_);
>+    task_queue_.reset();
>+  }
>+
>+ private:
>+  class FrameTask : public rtc::QueuedTask {
>+   public:
>+    FrameTask(Config config, rtc::VideoBroadcaster* broadcaster)
>+        : frame_interval_ms_(config.frame_interval_ms),
>+          frame_source_(
>+              config.width,
>+              config.height,
>+              config.frame_interval_ms * rtc::kNumMicrosecsPerMillisec),
>+          broadcaster_(broadcaster) {
>+      frame_source_.SetRotation(config.rotation);
>+    }
>+
>+    bool Run() override {
>+      if (broadcaster_->wants().rotation_applied) {
>+        broadcaster_->OnFrame(frame_source_.GetFrameRotationApplied());
>+      } else {
>+        broadcaster_->OnFrame(frame_source_.GetFrame());
>+      }
>+
>+      rtc::TaskQueue::Current()->PostDelayedTask(absl::WrapUnique(this),
>+                                                 frame_interval_ms_);
>+      return false;
>+    }
>+    int frame_interval_ms_;
>+    cricket::FakeFrameSource frame_source_;
>+    rtc::VideoBroadcaster* broadcaster_;
>+  };
>+
>+  rtc::ThreadChecker thread_checker_;
>+
>+  rtc::VideoBroadcaster broadcaster_;
>+
>+  std::unique_ptr<rtc::TaskQueue> task_queue_;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // PC_TEST_FAKEPERIODICVIDEOSOURCE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/fakeperiodicvideotracksource.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/fakeperiodicvideotracksource.h
>new file mode 100644
>index 00000000000..ab428d806a5
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/fakeperiodicvideotracksource.h
>@@ -0,0 +1,41 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef PC_TEST_FAKEPERIODICVIDEOTRACKSOURCE_H_
>+#define PC_TEST_FAKEPERIODICVIDEOTRACKSOURCE_H_
>+
>+#include "pc/test/fakeperiodicvideosource.h"
>+#include "pc/videotracksource.h"
>+
>+namespace webrtc {
>+
>+// A VideoTrackSource generating frames with configured size and frame interval.
>+class FakePeriodicVideoTrackSource : public VideoTrackSource {
>+ public:
>+  explicit FakePeriodicVideoTrackSource(bool remote)
>+      : FakePeriodicVideoTrackSource(FakePeriodicVideoSource::Config(),
>+                                     remote) {}
>+
>+  FakePeriodicVideoTrackSource(FakePeriodicVideoSource::Config config,
>+                               bool remote)
>+      : VideoTrackSource(remote), source_(config) {}
>+
>+  ~FakePeriodicVideoTrackSource() = default;
>+
>+ protected:
>+  rtc::VideoSourceInterface<VideoFrame>* source() override { return &source_; }
>+
>+ private:
>+  FakePeriodicVideoSource source_;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // PC_TEST_FAKEPERIODICVIDEOTRACKSOURCE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/fakertccertificategenerator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/fakertccertificategenerator.h
>index 9cc79284eaa..a3e7654a208 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/fakertccertificategenerator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/fakertccertificategenerator.h
>@@ -23,64 +23,63 @@
> static const rtc::RTCCertificatePEM kRsaPems[] = {
>     rtc::RTCCertificatePEM(
>         "-----BEGIN RSA PRIVATE KEY-----\n"
>-         "MIICdwIBADANBgkqhkiG9w0BAQEFAASCAmEwggJdAgEAAoGBAMYRkbhmI7kVA/rM\n"
>-         "czsZ+6JDhDvnkF+vn6yCAGuRPV03zuRqZtDy4N4to7PZu9PjqrRl7nDMXrG3YG9y\n"
>-         "rlIAZ72KjcKKFAJxQyAKLCIdawKRyp8RdK3LEySWEZb0AV58IadqPZDTNHHRX8dz\n"
>-         "5aTSMsbbkZ+C/OzTnbiMqLL/vg6jAgMBAAECgYAvgOs4FJcgvp+TuREx7YtiYVsH\n"
>-         "mwQPTum2z/8VzWGwR8BBHBvIpVe1MbD/Y4seyI2aco/7UaisatSgJhsU46/9Y4fq\n"
>-         "2TwXH9QANf4at4d9n/R6rzwpAJOpgwZgKvdQjkfrKTtgLV+/dawvpxUYkRH4JZM1\n"
>-         "CVGukMfKNrSVH4Ap4QJBAOJmGV1ASPnB4r4nc99at7JuIJmd7fmuVUwUgYi4XgaR\n"
>-         "WhScBsgYwZ/JoywdyZJgnbcrTDuVcWG56B3vXbhdpMsCQQDf9zeJrjnPZ3Cqm79y\n"
>-         "kdqANep0uwZciiNiWxsQrCHztywOvbFhdp8iYVFG9EK8DMY41Y5TxUwsHD+67zao\n"
>-         "ZNqJAkEA1suLUP/GvL8IwuRneQd2tWDqqRQ/Td3qq03hP7e77XtF/buya3Ghclo5\n"
>-         "54czUR89QyVfJEC6278nzA7n2h1uVQJAcG6mztNL6ja/dKZjYZye2CY44QjSlLo0\n"
>-         "MTgTSjdfg/28fFn2Jjtqf9Pi/X+50LWI/RcYMC2no606wRk9kyOuIQJBAK6VSAim\n"
>-         "1pOEjsYQn0X5KEIrz1G3bfCbB848Ime3U2/FWlCHMr6ch8kCZ5d1WUeJD3LbwMNG\n"
>-         "UCXiYxSsu20QNVw=\n"
>-         "-----END RSA PRIVATE KEY-----\n",
>-         "-----BEGIN CERTIFICATE-----\n"
>-         "MIIBmTCCAQKgAwIBAgIEbzBSAjANBgkqhkiG9w0BAQsFADARMQ8wDQYDVQQDEwZX\n"
>-         "ZWJSVEMwHhcNMTQwMTAyMTgyNDQ3WhcNMTQwMjAxMTgyNDQ3WjARMQ8wDQYDVQQD\n"
>-         "EwZXZWJSVEMwgZ8wDQYJKoZIhvcNAQEBBQADgY0AMIGJAoGBAMYRkbhmI7kVA/rM\n"
>-         "czsZ+6JDhDvnkF+vn6yCAGuRPV03zuRqZtDy4N4to7PZu9PjqrRl7nDMXrG3YG9y\n"
>-         "rlIAZ72KjcKKFAJxQyAKLCIdawKRyp8RdK3LEySWEZb0AV58IadqPZDTNHHRX8dz\n"
>-         "5aTSMsbbkZ+C/OzTnbiMqLL/vg6jAgMBAAEwDQYJKoZIhvcNAQELBQADgYEAUflI\n"
>-         "VUe5Krqf5RVa5C3u/UTAOAUJBiDS3VANTCLBxjuMsvqOG0WvaYWP3HYPgrz0jXK2\n"
>-         "LJE/mGw3MyFHEqi81jh95J+ypl6xKW6Rm8jKLR87gUvCaVYn/Z4/P3AqcQTB7wOv\n"
>-         "UD0A8qfhfDM+LK6rPAnCsVN0NRDY3jvd6rzix9M=\n"
>-         "-----END CERTIFICATE-----\n"),
>+        "MIICdwIBADANBgkqhkiG9w0BAQEFAASCAmEwggJdAgEAAoGBAMYRkbhmI7kVA/rM\n"
>+        "czsZ+6JDhDvnkF+vn6yCAGuRPV03zuRqZtDy4N4to7PZu9PjqrRl7nDMXrG3YG9y\n"
>+        "rlIAZ72KjcKKFAJxQyAKLCIdawKRyp8RdK3LEySWEZb0AV58IadqPZDTNHHRX8dz\n"
>+        "5aTSMsbbkZ+C/OzTnbiMqLL/vg6jAgMBAAECgYAvgOs4FJcgvp+TuREx7YtiYVsH\n"
>+        "mwQPTum2z/8VzWGwR8BBHBvIpVe1MbD/Y4seyI2aco/7UaisatSgJhsU46/9Y4fq\n"
>+        "2TwXH9QANf4at4d9n/R6rzwpAJOpgwZgKvdQjkfrKTtgLV+/dawvpxUYkRH4JZM1\n"
>+        "CVGukMfKNrSVH4Ap4QJBAOJmGV1ASPnB4r4nc99at7JuIJmd7fmuVUwUgYi4XgaR\n"
>+        "WhScBsgYwZ/JoywdyZJgnbcrTDuVcWG56B3vXbhdpMsCQQDf9zeJrjnPZ3Cqm79y\n"
>+        "kdqANep0uwZciiNiWxsQrCHztywOvbFhdp8iYVFG9EK8DMY41Y5TxUwsHD+67zao\n"
>+        "ZNqJAkEA1suLUP/GvL8IwuRneQd2tWDqqRQ/Td3qq03hP7e77XtF/buya3Ghclo5\n"
>+        "54czUR89QyVfJEC6278nzA7n2h1uVQJAcG6mztNL6ja/dKZjYZye2CY44QjSlLo0\n"
>+        "MTgTSjdfg/28fFn2Jjtqf9Pi/X+50LWI/RcYMC2no606wRk9kyOuIQJBAK6VSAim\n"
>+        "1pOEjsYQn0X5KEIrz1G3bfCbB848Ime3U2/FWlCHMr6ch8kCZ5d1WUeJD3LbwMNG\n"
>+        "UCXiYxSsu20QNVw=\n"
>+        "-----END RSA PRIVATE KEY-----\n",
>+        "-----BEGIN CERTIFICATE-----\n"
>+        "MIIBmTCCAQKgAwIBAgIEbzBSAjANBgkqhkiG9w0BAQsFADARMQ8wDQYDVQQDEwZX\n"
>+        "ZWJSVEMwHhcNMTQwMTAyMTgyNDQ3WhcNMTQwMjAxMTgyNDQ3WjARMQ8wDQYDVQQD\n"
>+        "EwZXZWJSVEMwgZ8wDQYJKoZIhvcNAQEBBQADgY0AMIGJAoGBAMYRkbhmI7kVA/rM\n"
>+        "czsZ+6JDhDvnkF+vn6yCAGuRPV03zuRqZtDy4N4to7PZu9PjqrRl7nDMXrG3YG9y\n"
>+        "rlIAZ72KjcKKFAJxQyAKLCIdawKRyp8RdK3LEySWEZb0AV58IadqPZDTNHHRX8dz\n"
>+        "5aTSMsbbkZ+C/OzTnbiMqLL/vg6jAgMBAAEwDQYJKoZIhvcNAQELBQADgYEAUflI\n"
>+        "VUe5Krqf5RVa5C3u/UTAOAUJBiDS3VANTCLBxjuMsvqOG0WvaYWP3HYPgrz0jXK2\n"
>+        "LJE/mGw3MyFHEqi81jh95J+ypl6xKW6Rm8jKLR87gUvCaVYn/Z4/P3AqcQTB7wOv\n"
>+        "UD0A8qfhfDM+LK6rPAnCsVN0NRDY3jvd6rzix9M=\n"
>+        "-----END CERTIFICATE-----\n"),
>     rtc::RTCCertificatePEM(
>-         "-----BEGIN RSA PRIVATE KEY-----\n"
>-         "MIICXQIBAAKBgQDeYqlyJ1wuiMsi905e3X81/WA/G3ym50PIDZBVtSwZi7JVQPgj\n"
>-         "Bl8CPZMvDh9EwB4Ji9ytA8dZZbQ4WbJWPr73zPpJSCvQqz6sOXSlenBRi72acNaQ\n"
>-         "sOR/qPvviJx5I6Hqo4qemfnjZhAW85a5BpgrAwKgMLIQTHCTLWwVSyrDrwIDAQAB\n"
>-         "AoGARni9eY8/hv+SX+I+05EdXt6MQXNUbQ+cSykBNCfVccLzIFEWUQMT2IHqwl6X\n"
>-         "ShIXcq7/n1QzOAEiuzixauM3YHg4xZ1Um2Ha9a7ig5Xg4v6b43bmMkNE6LkoAtYs\n"
>-         "qnQdfMh442b1liDud6IMb1Qk0amt3fSrgRMc547TZQVx4QECQQDxUeDm94r3p4ng\n"
>-         "5rCLLC1K5/6HSTZsh7jatKPlz7GfP/IZlYV7iE5784/n0wRiCjZOS7hQRy/8m2Gp\n"
>-         "pf4aZq+DAkEA6+np4d36FYikydvUrupLT3FkdRHGn/v83qOll/VmeNh+L1xMZlIP\n"
>-         "tM26hAXCcQb7O5+J9y3cx2CAQsBS11ZXZQJAfGgTo76WG9p5UEJdXUInD2jOZPwv\n"
>-         "XIATolxh6kXKcijLLLlSmT7KB0inNYIpzkkpee+7U1d/u6B3FriGaSHq9QJBAM/J\n"
>-         "ICnDdLCgwNvWVraVQC3BpwSB2pswvCFwq7py94V60XFvbw80Ogc6qIv98qvQxVlX\n"
>-         "hJIEgA/PjEi+0ng94Q0CQQDm8XSDby35gmjO+6eRmJtAjtB7nguLvrPXM6CPXRmD\n"
>-         "sRoBocpHw6j9UdzZ6qYG0FkdXZghezXFY58ro2BYYRR3\n"
>-         "-----END RSA PRIVATE KEY-----\n",
>-         "-----BEGIN CERTIFICATE-----\n"
>-         "MIICWDCCAcGgAwIBAgIJALgDjxMbBOhbMA0GCSqGSIb3DQEBCwUAMEUxCzAJBgNV\n"
>-         "BAYTAkFVMRMwEQYDVQQIDApTb21lLVN0YXRlMSEwHwYDVQQKDBhJbnRlcm5ldCBX\n"
>-         "aWRnaXRzIFB0eSBMdGQwHhcNMTUxMTEzMjIzMjEzWhcNMTYxMTEyMjIzMjEzWjBF\n"
>-         "MQswCQYDVQQGEwJBVTETMBEGA1UECAwKU29tZS1TdGF0ZTEhMB8GA1UECgwYSW50\n"
>-         "ZXJuZXQgV2lkZ2l0cyBQdHkgTHRkMIGfMA0GCSqGSIb3DQEBAQUAA4GNADCBiQKB\n"
>-         "gQDeYqlyJ1wuiMsi905e3X81/WA/G3ym50PIDZBVtSwZi7JVQPgjBl8CPZMvDh9E\n"
>-         "wB4Ji9ytA8dZZbQ4WbJWPr73zPpJSCvQqz6sOXSlenBRi72acNaQsOR/qPvviJx5\n"
>-         "I6Hqo4qemfnjZhAW85a5BpgrAwKgMLIQTHCTLWwVSyrDrwIDAQABo1AwTjAdBgNV\n"
>-         "HQ4EFgQUx2tbJdlcSTCepn09UdYORXKuSTAwHwYDVR0jBBgwFoAUx2tbJdlcSTCe\n"
>-         "pn09UdYORXKuSTAwDAYDVR0TBAUwAwEB/zANBgkqhkiG9w0BAQsFAAOBgQAmp9Id\n"
>-         "E716gHMqeBG4S2FCgVFCr0a0ugkaneQAN/c2L9CbMemEN9W6jvucUIVOtYd90dDW\n"
>-         "lXuowWmT/JctPe3D2qt4yvYW3puECHk2tVQmrJOZiZiTRtWm6HxkmoUYHYp/DtaS\n"
>-         "1Xe29gSTnZtI5sQCrGMzk3SGRSSs7ejLKiVDBQ==\n"
>-         "-----END CERTIFICATE-----\n")
>-};
>+        "-----BEGIN RSA PRIVATE KEY-----\n"
>+        "MIICXQIBAAKBgQDeYqlyJ1wuiMsi905e3X81/WA/G3ym50PIDZBVtSwZi7JVQPgj\n"
>+        "Bl8CPZMvDh9EwB4Ji9ytA8dZZbQ4WbJWPr73zPpJSCvQqz6sOXSlenBRi72acNaQ\n"
>+        "sOR/qPvviJx5I6Hqo4qemfnjZhAW85a5BpgrAwKgMLIQTHCTLWwVSyrDrwIDAQAB\n"
>+        "AoGARni9eY8/hv+SX+I+05EdXt6MQXNUbQ+cSykBNCfVccLzIFEWUQMT2IHqwl6X\n"
>+        "ShIXcq7/n1QzOAEiuzixauM3YHg4xZ1Um2Ha9a7ig5Xg4v6b43bmMkNE6LkoAtYs\n"
>+        "qnQdfMh442b1liDud6IMb1Qk0amt3fSrgRMc547TZQVx4QECQQDxUeDm94r3p4ng\n"
>+        "5rCLLC1K5/6HSTZsh7jatKPlz7GfP/IZlYV7iE5784/n0wRiCjZOS7hQRy/8m2Gp\n"
>+        "pf4aZq+DAkEA6+np4d36FYikydvUrupLT3FkdRHGn/v83qOll/VmeNh+L1xMZlIP\n"
>+        "tM26hAXCcQb7O5+J9y3cx2CAQsBS11ZXZQJAfGgTo76WG9p5UEJdXUInD2jOZPwv\n"
>+        "XIATolxh6kXKcijLLLlSmT7KB0inNYIpzkkpee+7U1d/u6B3FriGaSHq9QJBAM/J\n"
>+        "ICnDdLCgwNvWVraVQC3BpwSB2pswvCFwq7py94V60XFvbw80Ogc6qIv98qvQxVlX\n"
>+        "hJIEgA/PjEi+0ng94Q0CQQDm8XSDby35gmjO+6eRmJtAjtB7nguLvrPXM6CPXRmD\n"
>+        "sRoBocpHw6j9UdzZ6qYG0FkdXZghezXFY58ro2BYYRR3\n"
>+        "-----END RSA PRIVATE KEY-----\n",
>+        "-----BEGIN CERTIFICATE-----\n"
>+        "MIICWDCCAcGgAwIBAgIJALgDjxMbBOhbMA0GCSqGSIb3DQEBCwUAMEUxCzAJBgNV\n"
>+        "BAYTAkFVMRMwEQYDVQQIDApTb21lLVN0YXRlMSEwHwYDVQQKDBhJbnRlcm5ldCBX\n"
>+        "aWRnaXRzIFB0eSBMdGQwHhcNMTUxMTEzMjIzMjEzWhcNMTYxMTEyMjIzMjEzWjBF\n"
>+        "MQswCQYDVQQGEwJBVTETMBEGA1UECAwKU29tZS1TdGF0ZTEhMB8GA1UECgwYSW50\n"
>+        "ZXJuZXQgV2lkZ2l0cyBQdHkgTHRkMIGfMA0GCSqGSIb3DQEBAQUAA4GNADCBiQKB\n"
>+        "gQDeYqlyJ1wuiMsi905e3X81/WA/G3ym50PIDZBVtSwZi7JVQPgjBl8CPZMvDh9E\n"
>+        "wB4Ji9ytA8dZZbQ4WbJWPr73zPpJSCvQqz6sOXSlenBRi72acNaQsOR/qPvviJx5\n"
>+        "I6Hqo4qemfnjZhAW85a5BpgrAwKgMLIQTHCTLWwVSyrDrwIDAQABo1AwTjAdBgNV\n"
>+        "HQ4EFgQUx2tbJdlcSTCepn09UdYORXKuSTAwHwYDVR0jBBgwFoAUx2tbJdlcSTCe\n"
>+        "pn09UdYORXKuSTAwDAYDVR0TBAUwAwEB/zANBgkqhkiG9w0BAQsFAAOBgQAmp9Id\n"
>+        "E716gHMqeBG4S2FCgVFCr0a0ugkaneQAN/c2L9CbMemEN9W6jvucUIVOtYd90dDW\n"
>+        "lXuowWmT/JctPe3D2qt4yvYW3puECHk2tVQmrJOZiZiTRtWm6HxkmoUYHYp/DtaS\n"
>+        "1Xe29gSTnZtI5sQCrGMzk3SGRSSs7ejLKiVDBQ==\n"
>+        "-----END CERTIFICATE-----\n")};
> 
> // ECDSA with EC_NIST_P256.
> // These PEM strings were created by generating an identity with
>@@ -115,21 +114,19 @@ static const rtc::RTCCertificatePEM kEcdsaPems[] = {
>         "IkzTd8/pHHkZjV0RnTT5nKSMkjzdV8MDK+alG4ZqE20EfSFz7vGQMJt6jpMzS03f\n"
>         "nGswCgYIKoZIzj0EAwIDSAAwRQIgb/LBc8OtsC5lEDyjCP6M9xt5mwzUNrQBOFWZ\n"
>         "1fE/g68CIQD7uoFfbiq6dTp8ZwzbwQ8jJf08KjriamqA9OW/4268Dw==\n"
>-        "-----END CERTIFICATE-----\n")
>-};
>+        "-----END CERTIFICATE-----\n")};
> 
> class FakeRTCCertificateGenerator
>     : public rtc::RTCCertificateGeneratorInterface,
>       public rtc::MessageHandler {
>  public:
>-  typedef rtc::TypedMessageData<rtc::scoped_refptr<
>-      rtc::RTCCertificateGeneratorCallback> > MessageData;
>+  typedef rtc::TypedMessageData<
>+      rtc::scoped_refptr<rtc::RTCCertificateGeneratorCallback> >
>+      MessageData;
> 
>   FakeRTCCertificateGenerator() : should_fail_(false), should_wait_(false) {}
> 
>-  void set_should_fail(bool should_fail) {
>-    should_fail_ = should_fail;
>-  }
>+  void set_should_fail(bool should_fail) { should_fail_ = should_fail; }
> 
>   // If set to true, stalls the generation of the fake certificate until it is
>   // set to false.
>@@ -143,9 +140,9 @@ class FakeRTCCertificateGenerator
> 
>   void GenerateCertificateAsync(
>       const rtc::KeyParams& key_params,
>-      const rtc::Optional<uint64_t>& expires_ms,
>-      const rtc::scoped_refptr<rtc::RTCCertificateGeneratorCallback>&
>-          callback) override {
>+      const absl::optional<uint64_t>& expires_ms,
>+      const rtc::scoped_refptr<rtc::RTCCertificateGeneratorCallback>& callback)
>+      override {
>     // The certificates are created from constant PEM strings and use its coded
>     // expiration time, we do not support modifying it.
>     RTC_DCHECK(!expires_ms);
>@@ -169,13 +166,13 @@ class FakeRTCCertificateGenerator
> 
>   static rtc::scoped_refptr<rtc::RTCCertificate> GenerateCertificate() {
>     switch (rtc::KT_DEFAULT) {
>-    case rtc::KT_RSA:
>-      return rtc::RTCCertificate::FromPEM(kRsaPems[0]);
>-    case rtc::KT_ECDSA:
>-      return rtc::RTCCertificate::FromPEM(kEcdsaPems[0]);
>-    default:
>-      RTC_NOTREACHED();
>-      return nullptr;
>+      case rtc::KT_RSA:
>+        return rtc::RTCCertificate::FromPEM(kRsaPems[0]);
>+      case rtc::KT_ECDSA:
>+        return rtc::RTCCertificate::FromPEM(kEcdsaPems[0]);
>+      default:
>+        RTC_NOTREACHED();
>+        return nullptr;
>     }
>   }
> 
>@@ -188,13 +185,13 @@ class FakeRTCCertificateGenerator
> 
>   const rtc::RTCCertificatePEM& get_pem(const rtc::KeyType& key_type) const {
>     switch (key_type) {
>-    case rtc::KT_RSA:
>-      return kRsaPems[key_index_];
>-    case rtc::KT_ECDSA:
>-      return kEcdsaPems[key_index_];
>-    default:
>-      RTC_NOTREACHED();
>-      return kEcdsaPems[key_index_];
>+      case rtc::KT_RSA:
>+        return kRsaPems[key_index_];
>+      case rtc::KT_ECDSA:
>+        return kEcdsaPems[key_index_];
>+      default:
>+        RTC_NOTREACHED();
>+        return kEcdsaPems[key_index_];
>     }
>   }
>   const std::string& get_key(const rtc::KeyType& key_type) const {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/fakesctptransport.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/fakesctptransport.h
>index 17143ed66c4..294376d91a4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/fakesctptransport.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/fakesctptransport.h
>@@ -20,7 +20,7 @@
> // local/remote ports.
> class FakeSctpTransport : public cricket::SctpTransportInternal {
>  public:
>-  void SetTransportChannel(rtc::PacketTransportInternal* channel) override {}
>+  void SetDtlsTransport(rtc::PacketTransportInternal* transport) override {}
>   bool Start(int local_port, int remote_port) override {
>     local_port_.emplace(local_port);
>     remote_port_.emplace(remote_port);
>@@ -40,8 +40,8 @@ class FakeSctpTransport : public cricket::SctpTransportInternal {
>   int remote_port() const { return *remote_port_; }
> 
>  private:
>-  rtc::Optional<int> local_port_;
>-  rtc::Optional<int> remote_port_;
>+  absl::optional<int> local_port_;
>+  absl::optional<int> remote_port_;
> };
> 
> class FakeSctpTransportFactory : public cricket::SctpTransportInternalFactory {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/faketransportcontroller.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/faketransportcontroller.h
>deleted file mode 100644
>index 4a4cc96d8ea..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/faketransportcontroller.h
>+++ /dev/null
>@@ -1,159 +0,0 @@
>-/*
>- *  Copyright 2009 The WebRTC Project Authors. All rights reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef PC_TEST_FAKETRANSPORTCONTROLLER_H_
>-#define PC_TEST_FAKETRANSPORTCONTROLLER_H_
>-
>-#include <memory>
>-#include <string>
>-#include <vector>
>-
>-#include "p2p/base/fakedtlstransport.h"
>-#include "p2p/base/fakeicetransport.h"
>-#include "pc/transportcontroller.h"
>-#include "rtc_base/bind.h"
>-#include "rtc_base/sslfingerprint.h"
>-#include "rtc_base/thread.h"
>-
>-namespace cricket {
>-
>-// Fake TransportController class, which can be passed into a WebRtcSession
>-// object for test purposes. Can be connected to other FakeTransportControllers
>-// via Connect().
>-//
>-// This fake is unusual in that for the most part, it's implemented with the
>-// real TransportController code, but with fake TransportChannels underneath.
>-class FakeTransportController : public TransportController {
>- public:
>-  FakeTransportController()
>-      : TransportController(rtc::Thread::Current(),
>-                            rtc::Thread::Current(),
>-                            nullptr,
>-                            /*redetermine_role_on_ice_restart=*/true,
>-                            rtc::CryptoOptions()) {}
>-
>-  explicit FakeTransportController(bool redetermine_role_on_ice_restart)
>-      : TransportController(rtc::Thread::Current(),
>-                            rtc::Thread::Current(),
>-                            nullptr,
>-                            redetermine_role_on_ice_restart,
>-                            rtc::CryptoOptions()) {}
>-
>-  explicit FakeTransportController(IceRole role)
>-      : TransportController(rtc::Thread::Current(),
>-                            rtc::Thread::Current(),
>-                            nullptr,
>-                            /*redetermine_role_on_ice_restart=*/true,
>-                            rtc::CryptoOptions()) {
>-    SetIceRole(role);
>-  }
>-
>-  explicit FakeTransportController(rtc::Thread* network_thread)
>-      : TransportController(rtc::Thread::Current(),
>-                            network_thread,
>-                            nullptr,
>-                            /*redetermine_role_on_ice_restart=*/true,
>-                            rtc::CryptoOptions()) {}
>-
>-  FakeTransportController(rtc::Thread* network_thread, IceRole role)
>-      : TransportController(rtc::Thread::Current(),
>-                            network_thread,
>-                            nullptr,
>-                            /*redetermine_role_on_ice_restart=*/true,
>-                            rtc::CryptoOptions()) {
>-    SetIceRole(role);
>-  }
>-
>-  FakeDtlsTransport* GetFakeDtlsTransport_n(const std::string& transport_name,
>-                                            int component) {
>-    return static_cast<FakeDtlsTransport*>(
>-        get_channel_for_testing(transport_name, component));
>-  }
>-
>-  // Simulate the exchange of transport descriptions, and the gathering and
>-  // exchange of ICE candidates.
>-  void Connect(FakeTransportController* dest) {
>-    for (const std::string& transport_name : transport_names_for_testing()) {
>-      std::unique_ptr<rtc::SSLFingerprint> local_fingerprint;
>-      std::unique_ptr<rtc::SSLFingerprint> remote_fingerprint;
>-      if (certificate_for_testing()) {
>-        local_fingerprint.reset(rtc::SSLFingerprint::CreateFromCertificate(
>-            certificate_for_testing()));
>-      }
>-      if (dest->certificate_for_testing()) {
>-        remote_fingerprint.reset(rtc::SSLFingerprint::CreateFromCertificate(
>-            dest->certificate_for_testing()));
>-      }
>-      TransportDescription local_desc(
>-          std::vector<std::string>(),
>-          rtc::CreateRandomString(cricket::ICE_UFRAG_LENGTH),
>-          rtc::CreateRandomString(cricket::ICE_PWD_LENGTH),
>-          cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_NONE,
>-          local_fingerprint.get());
>-      TransportDescription remote_desc(
>-          std::vector<std::string>(),
>-          rtc::CreateRandomString(cricket::ICE_UFRAG_LENGTH),
>-          rtc::CreateRandomString(cricket::ICE_PWD_LENGTH),
>-          cricket::ICEMODE_FULL, cricket::CONNECTIONROLE_NONE,
>-          remote_fingerprint.get());
>-      std::string err;
>-      SetLocalTransportDescription(transport_name, local_desc,
>-                                   webrtc::SdpType::kOffer, &err);
>-      dest->SetRemoteTransportDescription(transport_name, local_desc,
>-                                          webrtc::SdpType::kOffer, &err);
>-      dest->SetLocalTransportDescription(transport_name, remote_desc,
>-                                         webrtc::SdpType::kAnswer, &err);
>-      SetRemoteTransportDescription(transport_name, remote_desc,
>-                                    webrtc::SdpType::kAnswer, &err);
>-    }
>-    MaybeStartGathering();
>-    dest->MaybeStartGathering();
>-    network_thread()->Invoke<void>(
>-        RTC_FROM_HERE,
>-        rtc::Bind(&FakeTransportController::SetChannelDestinations_n, this,
>-                  dest));
>-  }
>-
>-  void DestroyRtcpTransport(const std::string& transport_name) {
>-    DestroyDtlsTransport_n(transport_name,
>-                           cricket::ICE_CANDIDATE_COMPONENT_RTCP);
>-  }
>-
>- protected:
>-  IceTransportInternal* CreateIceTransportChannel_n(
>-      const std::string& transport_name,
>-      int component) override {
>-    return new FakeIceTransport(transport_name, component);
>-  }
>-
>-  DtlsTransportInternal* CreateDtlsTransportChannel_n(
>-      const std::string& transport_name,
>-      int component,
>-      IceTransportInternal* ice) override {
>-    return new FakeDtlsTransport(static_cast<FakeIceTransport*>(ice));
>-  }
>-
>- private:
>-  void SetChannelDestinations_n(FakeTransportController* dest) {
>-    for (DtlsTransportInternal* tc : channels_for_testing()) {
>-      FakeDtlsTransport* local = static_cast<FakeDtlsTransport*>(tc);
>-      FakeDtlsTransport* remote = dest->GetFakeDtlsTransport_n(
>-          local->transport_name(), local->component());
>-      if (remote) {
>-        bool asymmetric = false;
>-        local->SetDestination(remote, asymmetric);
>-      }
>-    }
>-  }
>-};
>-
>-}  // namespace cricket
>-
>-#endif  // PC_TEST_FAKETRANSPORTCONTROLLER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/fakevideotracksource.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/fakevideotracksource.h
>index 46216c73666..1b5bd87c531 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/fakevideotracksource.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/fakevideotracksource.h
>@@ -12,11 +12,12 @@
> #define PC_TEST_FAKEVIDEOTRACKSOURCE_H_
> 
> #include "api/mediastreaminterface.h"
>-#include "media/base/fakevideocapturer.h"
> #include "pc/videotracksource.h"
> 
> namespace webrtc {
> 
>+// A minimal implementation of VideoTrackSource, which doesn't produce
>+// any frames.
> class FakeVideoTrackSource : public VideoTrackSource {
>  public:
>   static rtc::scoped_refptr<FakeVideoTrackSource> Create(bool is_screencast) {
>@@ -27,20 +28,20 @@ class FakeVideoTrackSource : public VideoTrackSource {
>     return Create(false);
>   }
> 
>-  cricket::FakeVideoCapturer* fake_video_capturer() {
>-    return &fake_video_capturer_;
>-  }
>-
>   bool is_screencast() const override { return is_screencast_; }
>+  void AddOrUpdateSink(rtc::VideoSinkInterface<VideoFrame>* sink,
>+                       const rtc::VideoSinkWants& wants) override {}
>+  void RemoveSink(rtc::VideoSinkInterface<VideoFrame>* sink) {}
> 
>  protected:
>   explicit FakeVideoTrackSource(bool is_screencast)
>-      : VideoTrackSource(&fake_video_capturer_, false /* remote */),
>-        is_screencast_(is_screencast) {}
>-  virtual ~FakeVideoTrackSource() {}
>+      : VideoTrackSource(false /* remote */), is_screencast_(is_screencast) {}
>+  ~FakeVideoTrackSource() override = default;
>+
>+  // Unused, since we override AddOrUpdateSink and RemoveSink above.
>+  rtc::VideoSourceInterface<VideoFrame>* source() override { return nullptr; }
> 
>  private:
>-  cricket::FakeVideoCapturer fake_video_capturer_;
>   const bool is_screencast_;
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/framegeneratorcapturervideotracksource.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/framegeneratorcapturervideotracksource.h
>new file mode 100644
>index 00000000000..77e78e0c986
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/framegeneratorcapturervideotracksource.h
>@@ -0,0 +1,72 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef PC_TEST_FRAMEGENERATORCAPTURERVIDEOTRACKSOURCE_H_
>+#define PC_TEST_FRAMEGENERATORCAPTURERVIDEOTRACKSOURCE_H_
>+
>+#include <memory>
>+
>+#include "pc/videotracksource.h"
>+#include "test/frame_generator_capturer.h"
>+
>+namespace webrtc {
>+
>+// Implements a VideoTrackSourceInterface to be used for creating VideoTracks.
>+// The video source is generated using a FrameGeneratorCapturer, specifically
>+// a SquareGenerator that generates frames with randomly sized and colored
>+// squares.
>+class FrameGeneratorCapturerVideoTrackSource : public VideoTrackSource {
>+ public:
>+  static const int kDefaultFramesPerSecond = 30;
>+  static const int kDefaultWidth = 640;
>+  static const int kDefaultHeight = 480;
>+  static const int kNumSquaresGenerated = 50;
>+
>+  struct Config {
>+    int frames_per_second = kDefaultFramesPerSecond;
>+    int width = kDefaultWidth;
>+    int height = kDefaultHeight;
>+    int num_squares_generated = 50;
>+  };
>+
>+  explicit FrameGeneratorCapturerVideoTrackSource(Clock* clock)
>+      : FrameGeneratorCapturerVideoTrackSource(Config(), clock) {}
>+
>+  FrameGeneratorCapturerVideoTrackSource(Config config, Clock* clock)
>+      : VideoTrackSource(false /* remote */) {
>+    video_capturer_.reset(test::FrameGeneratorCapturer::Create(
>+        config.width, config.height, absl::nullopt,
>+        config.num_squares_generated, config.frames_per_second, clock));
>+  }
>+
>+  ~FrameGeneratorCapturerVideoTrackSource() = default;
>+
>+  void Start() {
>+    video_capturer_->Start();
>+    SetState(kLive);
>+  }
>+
>+  void Stop() {
>+    video_capturer_->Stop();
>+    SetState(kMuted);
>+  }
>+
>+ protected:
>+  rtc::VideoSourceInterface<VideoFrame>* source() override {
>+    return video_capturer_.get();
>+  }
>+
>+ private:
>+  std::unique_ptr<test::FrameGeneratorCapturer> video_capturer_;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // PC_TEST_FRAMEGENERATORCAPTURERVIDEOTRACKSOURCE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/mock_datachannel.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/mock_datachannel.h
>index 20a4da664a2..fe887146fea 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/mock_datachannel.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/mock_datachannel.h
>@@ -21,30 +21,27 @@ namespace webrtc {
> class MockDataChannel : public rtc::RefCountedObject<DataChannel> {
>  public:
>   MockDataChannel(int id, DataState state)
>-      : MockDataChannel(id, "MockDataChannel", state, "udp", 0, 0, 0, 0) {
>-  }
>-  MockDataChannel(
>-      int id,
>-      const std::string& label,
>-      DataState state,
>-      const std::string& protocol,
>-      uint32_t messages_sent,
>-      uint64_t bytes_sent,
>-      uint32_t messages_received,
>-      uint64_t bytes_received)
>-      : rtc::RefCountedObject<DataChannel>(
>-            nullptr, cricket::DCT_NONE, label) {
>+      : MockDataChannel(id, "MockDataChannel", state, "udp", 0, 0, 0, 0) {}
>+  MockDataChannel(int id,
>+                  const std::string& label,
>+                  DataState state,
>+                  const std::string& protocol,
>+                  uint32_t messages_sent,
>+                  uint64_t bytes_sent,
>+                  uint32_t messages_received,
>+                  uint64_t bytes_received)
>+      : rtc::RefCountedObject<DataChannel>(nullptr, cricket::DCT_NONE, label) {
>     EXPECT_CALL(*this, id()).WillRepeatedly(testing::Return(id));
>     EXPECT_CALL(*this, state()).WillRepeatedly(testing::Return(state));
>     EXPECT_CALL(*this, protocol()).WillRepeatedly(testing::Return(protocol));
>-    EXPECT_CALL(*this, messages_sent()).WillRepeatedly(
>-        testing::Return(messages_sent));
>-    EXPECT_CALL(*this, bytes_sent()).WillRepeatedly(
>-        testing::Return(bytes_sent));
>-    EXPECT_CALL(*this, messages_received()).WillRepeatedly(
>-        testing::Return(messages_received));
>-    EXPECT_CALL(*this, bytes_received()).WillRepeatedly(
>-        testing::Return(bytes_received));
>+    EXPECT_CALL(*this, messages_sent())
>+        .WillRepeatedly(testing::Return(messages_sent));
>+    EXPECT_CALL(*this, bytes_sent())
>+        .WillRepeatedly(testing::Return(bytes_sent));
>+    EXPECT_CALL(*this, messages_received())
>+        .WillRepeatedly(testing::Return(messages_received));
>+    EXPECT_CALL(*this, bytes_received())
>+        .WillRepeatedly(testing::Return(bytes_received));
>   }
>   MOCK_CONST_METHOD0(id, int());
>   MOCK_CONST_METHOD0(state, DataState());
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/mock_peerconnection.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/mock_peerconnection.h
>index cf115932cc6..d8cb5e381b8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/mock_peerconnection.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/mock_peerconnection.h
>@@ -11,7 +11,9 @@
> #ifndef PC_TEST_MOCK_PEERCONNECTION_H_
> #define PC_TEST_MOCK_PEERCONNECTION_H_
> 
>+#include <map>
> #include <memory>
>+#include <set>
> #include <string>
> #include <utility>
> #include <vector>
>@@ -51,37 +53,151 @@ class MockPeerConnection
>             factory,
>             std::unique_ptr<RtcEventLog>(),
>             std::unique_ptr<Call>()) {}
>-  MOCK_METHOD0(local_streams,
>-               rtc::scoped_refptr<StreamCollectionInterface>());
>-  MOCK_METHOD0(remote_streams,
>-               rtc::scoped_refptr<StreamCollectionInterface>());
>+
>+  // PeerConnectionInterface
>+  MOCK_METHOD0(local_streams, rtc::scoped_refptr<StreamCollectionInterface>());
>+  MOCK_METHOD0(remote_streams, rtc::scoped_refptr<StreamCollectionInterface>());
>+  MOCK_METHOD1(AddStream, bool(MediaStreamInterface*));
>+  MOCK_METHOD1(RemoveStream, void(MediaStreamInterface*));
>+  MOCK_METHOD2(AddTrack,
>+               RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>>(
>+                   rtc::scoped_refptr<MediaStreamTrackInterface>,
>+                   const std::vector<std::string>&));
>+  MOCK_METHOD2(AddTrack,
>+               rtc::scoped_refptr<RtpSenderInterface>(
>+                   MediaStreamTrackInterface*,
>+                   std::vector<MediaStreamInterface*>));
>+  MOCK_METHOD1(RemoveTrack, bool(RtpSenderInterface*));
>+  MOCK_METHOD1(AddTransceiver,
>+               RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>(
>+                   rtc::scoped_refptr<MediaStreamTrackInterface>));
>+  MOCK_METHOD2(AddTransceiver,
>+               RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>(
>+                   rtc::scoped_refptr<MediaStreamTrackInterface>,
>+                   const RtpTransceiverInit&));
>+  MOCK_METHOD1(AddTransceiver,
>+               RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>(
>+                   cricket::MediaType));
>+  MOCK_METHOD2(AddTransceiver,
>+               RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>(
>+                   cricket::MediaType,
>+                   const RtpTransceiverInit&));
>+  MOCK_METHOD2(CreateSender,
>+               rtc::scoped_refptr<RtpSenderInterface>(const std::string&,
>+                                                      const std::string&));
>   MOCK_CONST_METHOD0(GetSenders,
>                      std::vector<rtc::scoped_refptr<RtpSenderInterface>>());
>   MOCK_CONST_METHOD0(GetReceivers,
>                      std::vector<rtc::scoped_refptr<RtpReceiverInterface>>());
>-  MOCK_CONST_METHOD0(sctp_data_channels,
>-                     const std::vector<rtc::scoped_refptr<DataChannel>>&());
>-  MOCK_CONST_METHOD0(voice_channel, cricket::VoiceChannel*());
>-  MOCK_CONST_METHOD0(video_channel, cricket::VideoChannel*());
>-  // Libjingle uses "local" for a outgoing track, and "remote" for a incoming
>-  // track.
>+  MOCK_CONST_METHOD0(
>+      GetTransceivers,
>+      std::vector<rtc::scoped_refptr<RtpTransceiverInterface>>());
>+  MOCK_METHOD3(GetStats,
>+               bool(StatsObserver*,
>+                    MediaStreamTrackInterface*,
>+                    StatsOutputLevel));
>+  MOCK_METHOD1(GetStats, void(RTCStatsCollectorCallback*));
>+  MOCK_METHOD2(GetStats,
>+               void(rtc::scoped_refptr<RtpSenderInterface>,
>+                    rtc::scoped_refptr<RTCStatsCollectorCallback>));
>+  MOCK_METHOD2(GetStats,
>+               void(rtc::scoped_refptr<RtpReceiverInterface>,
>+                    rtc::scoped_refptr<RTCStatsCollectorCallback>));
>+  MOCK_METHOD0(ClearStatsCache, void());
>+  MOCK_METHOD2(
>+      CreateDataChannel,
>+      rtc::scoped_refptr<DataChannelInterface>(const std::string&,
>+                                               const DataChannelInit*));
>+  MOCK_CONST_METHOD0(local_description, const SessionDescriptionInterface*());
>+  MOCK_CONST_METHOD0(remote_description, const SessionDescriptionInterface*());
>+  MOCK_CONST_METHOD0(current_local_description,
>+                     const SessionDescriptionInterface*());
>+  MOCK_CONST_METHOD0(current_remote_description,
>+                     const SessionDescriptionInterface*());
>+  MOCK_CONST_METHOD0(pending_local_description,
>+                     const SessionDescriptionInterface*());
>+  MOCK_CONST_METHOD0(pending_remote_description,
>+                     const SessionDescriptionInterface*());
>+  MOCK_METHOD2(CreateOffer,
>+               void(CreateSessionDescriptionObserver*,
>+                    const MediaConstraintsInterface*));
>+  MOCK_METHOD2(CreateOffer,
>+               void(CreateSessionDescriptionObserver*,
>+                    const RTCOfferAnswerOptions&));
>+  MOCK_METHOD2(CreateAnswer,
>+               void(CreateSessionDescriptionObserver*,
>+                    const RTCOfferAnswerOptions&));
>+  MOCK_METHOD2(CreateAnswer,
>+               void(CreateSessionDescriptionObserver*,
>+                    const MediaConstraintsInterface*));
>+  MOCK_METHOD2(SetLocalDescription,
>+               void(SetSessionDescriptionObserver*,
>+                    SessionDescriptionInterface*));
>+  MOCK_METHOD2(SetRemoteDescription,
>+               void(SetSessionDescriptionObserver*,
>+                    SessionDescriptionInterface*));
>+  MOCK_METHOD2(SetRemoteDescription,
>+               void(std::unique_ptr<SessionDescriptionInterface>,
>+                    rtc::scoped_refptr<SetRemoteDescriptionObserverInterface>));
>+  MOCK_METHOD0(GetConfiguration, PeerConnectionInterface::RTCConfiguration());
>+  MOCK_METHOD2(SetConfiguration,
>+               bool(const PeerConnectionInterface::RTCConfiguration&,
>+                    RTCError*));
>+  MOCK_METHOD1(SetConfiguration,
>+               bool(const PeerConnectionInterface::RTCConfiguration&));
>+  MOCK_METHOD1(AddIceCandidate, bool(const IceCandidateInterface*));
>+  MOCK_METHOD1(RemoveIceCandidates,
>+               bool(const std::vector<cricket::Candidate>&));
>+  MOCK_METHOD1(SetBitrate, RTCError(const BitrateSettings&));
>+  MOCK_METHOD1(SetBitrate, RTCError(const BitrateParameters&));
>+  MOCK_METHOD1(SetBitrateAllocationStrategy,
>+               void(std::unique_ptr<rtc::BitrateAllocationStrategy>));
>+  MOCK_METHOD1(SetAudioPlayout, void(bool));
>+  MOCK_METHOD1(SetAudioRecording, void(bool));
>+  MOCK_METHOD0(signaling_state, SignalingState());
>+  MOCK_METHOD0(ice_connection_state, IceConnectionState());
>+  MOCK_METHOD0(ice_gathering_state, IceGatheringState());
>+  MOCK_METHOD2(StartRtcEventLog, bool(rtc::PlatformFile, int64_t));
>+  MOCK_METHOD2(StartRtcEventLog,
>+               bool(std::unique_ptr<RtcEventLogOutput>, int64_t));
>+  MOCK_METHOD0(StopRtcEventLog, void());
>+  MOCK_METHOD0(Close, void());
>+
>+  // PeerConnectionInternal
>+  MOCK_CONST_METHOD0(network_thread, rtc::Thread*());
>+  MOCK_CONST_METHOD0(worker_thread, rtc::Thread*());
>+  MOCK_CONST_METHOD0(signaling_thread, rtc::Thread*());
>+  MOCK_CONST_METHOD0(session_id, std::string());
>+  MOCK_CONST_METHOD0(initial_offerer, bool());
>+  MOCK_CONST_METHOD0(GetTransceiversInternal,
>+                     std::vector<rtc::scoped_refptr<
>+                         RtpTransceiverProxyWithInternal<RtpTransceiver>>>());
>   MOCK_METHOD2(GetLocalTrackIdBySsrc, bool(uint32_t, std::string*));
>   MOCK_METHOD2(GetRemoteTrackIdBySsrc, bool(uint32_t, std::string*));
>+  MOCK_METHOD0(SignalDataChannelCreated, sigslot::signal1<DataChannel*>&());
>+  MOCK_CONST_METHOD0(rtp_data_channel, cricket::RtpDataChannel*());
>+  MOCK_CONST_METHOD0(sctp_data_channels,
>+                     std::vector<rtc::scoped_refptr<DataChannel>>());
>+  MOCK_CONST_METHOD0(sctp_content_name, absl::optional<std::string>());
>+  MOCK_CONST_METHOD0(sctp_transport_name, absl::optional<std::string>());
>+  MOCK_CONST_METHOD0(GetPooledCandidateStats, cricket::CandidateStatsList());
>+
>+  typedef std::map<std::string, std::string> TransportNamesByMid;
>+  MOCK_CONST_METHOD0(GetTransportNamesByMid, TransportNamesByMid());
>+
>+  typedef std::map<std::string, cricket::TransportStats> TransportStatsByNames;
>+  MOCK_METHOD1(GetTransportStatsByNames,
>+               TransportStatsByNames(const std::set<std::string>&));
>   MOCK_METHOD0(GetCallStats, Call::Stats());
>-  MOCK_METHOD1(GetSessionStats,
>-               std::unique_ptr<SessionStats>(const ChannelNamePairs&));
>   MOCK_METHOD2(GetLocalCertificate,
>-               bool(const std::string& transport_name,
>-                    rtc::scoped_refptr<rtc::RTCCertificate>* certificate));
>-
>-  // Workaround for gmock's inability to cope with move-only return values.
>-  std::unique_ptr<rtc::SSLCertificate> GetRemoteSSLCertificate(
>-      const std::string& transport_name) /* override */ {
>-    return std::unique_ptr<rtc::SSLCertificate>(
>-        GetRemoteSSLCertificate_ReturnsRawPointer(transport_name));
>-  }
>-  MOCK_METHOD1(GetRemoteSSLCertificate_ReturnsRawPointer,
>-               rtc::SSLCertificate*(const std::string& transport_name));
>+               bool(const std::string&,
>+                    rtc::scoped_refptr<rtc::RTCCertificate>*));
>+  MOCK_METHOD1(GetRemoteSSLCertChain,
>+               std::unique_ptr<rtc::SSLCertChain>(const std::string&));
>+  MOCK_CONST_METHOD1(IceRestartPending, bool(const std::string&));
>+  MOCK_CONST_METHOD1(NeedsIceRestart, bool(const std::string&));
>+  MOCK_METHOD2(GetSslRole,
>+               bool(const std::string& content_name, rtc::SSLRole*));
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/mock_rtpreceiverinternal.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/mock_rtpreceiverinternal.h
>new file mode 100644
>index 00000000000..8b12e8fb483
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/mock_rtpreceiverinternal.h
>@@ -0,0 +1,54 @@
>+/*
>+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef PC_TEST_MOCK_RTPRECEIVERINTERNAL_H_
>+#define PC_TEST_MOCK_RTPRECEIVERINTERNAL_H_
>+
>+#include <string>
>+#include <vector>
>+
>+#include "pc/rtpreceiver.h"
>+#include "test/gmock.h"
>+
>+namespace webrtc {
>+
>+// The definition of MockRtpReceiver is copied in to avoid multiple inheritance.
>+class MockRtpReceiverInternal : public RtpReceiverInternal {
>+ public:
>+  // RtpReceiverInterface methods.
>+  MOCK_METHOD1(SetTrack, void(MediaStreamTrackInterface*));
>+  MOCK_CONST_METHOD0(track, rtc::scoped_refptr<MediaStreamTrackInterface>());
>+  MOCK_CONST_METHOD0(stream_ids, std::vector<std::string>());
>+  MOCK_CONST_METHOD0(streams,
>+                     std::vector<rtc::scoped_refptr<MediaStreamInterface>>());
>+  MOCK_CONST_METHOD0(media_type, cricket::MediaType());
>+  MOCK_CONST_METHOD0(id, std::string());
>+  MOCK_CONST_METHOD0(GetParameters, RtpParameters());
>+  MOCK_METHOD1(SetParameters, bool(const RtpParameters&));
>+  MOCK_METHOD1(SetObserver, void(RtpReceiverObserverInterface*));
>+  MOCK_CONST_METHOD0(GetSources, std::vector<RtpSource>());
>+
>+  // RtpReceiverInternal methods.
>+  MOCK_METHOD0(Stop, void());
>+  MOCK_METHOD1(SetVoiceMediaChannel, void(cricket::VoiceMediaChannel*));
>+  MOCK_METHOD1(SetVideoMediaChannel, void(cricket::VideoMediaChannel*));
>+  MOCK_METHOD1(SetupMediaChannel, void(uint32_t));
>+  MOCK_CONST_METHOD0(ssrc, uint32_t());
>+  MOCK_METHOD0(NotifyFirstPacketReceived, void());
>+  MOCK_METHOD1(set_stream_ids, void(std::vector<std::string>));
>+  MOCK_METHOD1(
>+      SetStreams,
>+      void(const std::vector<rtc::scoped_refptr<MediaStreamInterface>>&));
>+  MOCK_CONST_METHOD0(AttachmentId, int());
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // PC_TEST_MOCK_RTPRECEIVERINTERNAL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/mock_rtpsenderinternal.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/mock_rtpsenderinternal.h
>new file mode 100644
>index 00000000000..3ddffec7a1b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/mock_rtpsenderinternal.h
>@@ -0,0 +1,47 @@
>+/*
>+ *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef PC_TEST_MOCK_RTPSENDERINTERNAL_H_
>+#define PC_TEST_MOCK_RTPSENDERINTERNAL_H_
>+
>+#include <string>
>+#include <vector>
>+
>+#include "pc/rtpsender.h"
>+#include "test/gmock.h"
>+
>+namespace webrtc {
>+
>+// The definition of MockRtpSender is copied in to avoid multiple inheritance.
>+class MockRtpSenderInternal : public RtpSenderInternal {
>+ public:
>+  // RtpSenderInterface methods.
>+  MOCK_METHOD1(SetTrack, bool(MediaStreamTrackInterface*));
>+  MOCK_CONST_METHOD0(track, rtc::scoped_refptr<MediaStreamTrackInterface>());
>+  MOCK_CONST_METHOD0(ssrc, uint32_t());
>+  MOCK_CONST_METHOD0(media_type, cricket::MediaType());
>+  MOCK_CONST_METHOD0(id, std::string());
>+  MOCK_CONST_METHOD0(stream_ids, std::vector<std::string>());
>+  MOCK_METHOD0(GetParameters, RtpParameters());
>+  MOCK_METHOD1(SetParameters, RTCError(const RtpParameters&));
>+  MOCK_CONST_METHOD0(GetDtmfSender, rtc::scoped_refptr<DtmfSenderInterface>());
>+
>+  // RtpSenderInternal methods.
>+  MOCK_METHOD1(SetVoiceMediaChannel, void(cricket::VoiceMediaChannel*));
>+  MOCK_METHOD1(SetVideoMediaChannel, void(cricket::VideoMediaChannel*));
>+  MOCK_METHOD1(SetSsrc, void(uint32_t));
>+  MOCK_METHOD1(set_stream_ids, void(const std::vector<std::string>&));
>+  MOCK_METHOD0(Stop, void());
>+  MOCK_CONST_METHOD0(AttachmentId, int());
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // PC_TEST_MOCK_RTPSENDERINTERNAL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/mockpeerconnectionobservers.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/mockpeerconnectionobservers.h
>index 86f22368d21..6f5e9bc6652 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/mockpeerconnectionobservers.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/mockpeerconnectionobservers.h
>@@ -20,11 +20,11 @@
> #include <utility>
> #include <vector>
> 
>+#include "absl/memory/memory.h"
> #include "api/datachannelinterface.h"
> #include "api/jsepicecandidate.h"
> #include "pc/streamcollection.h"
> #include "rtc_base/checks.h"
>-#include "rtc_base/ptr_util.h"
> 
> namespace webrtc {
> 
>@@ -93,8 +93,13 @@ class MockPeerConnectionObserver : public PeerConnectionObserver {
>   void OnIceConnectionChange(
>       PeerConnectionInterface::IceConnectionState new_state) override {
>     RTC_DCHECK(pc_->ice_connection_state() == new_state);
>+    // When ICE is finished, the caller will get to a kIceConnectionCompleted
>+    // state, because it has the ICE controlling role, while the callee
>+    // will get to a kIceConnectionConnected state. This means that both ICE
>+    // and DTLS are connected.
>     ice_connected_ =
>-        (new_state == PeerConnectionInterface::kIceConnectionConnected);
>+        (new_state == PeerConnectionInterface::kIceConnectionConnected) ||
>+        (new_state == PeerConnectionInterface::kIceConnectionCompleted);
>     callback_triggered_ = true;
>   }
>   void OnIceGatheringChange(
>@@ -107,7 +112,7 @@ class MockPeerConnectionObserver : public PeerConnectionObserver {
>   void OnIceCandidate(const IceCandidateInterface* candidate) override {
>     RTC_DCHECK(PeerConnectionInterface::kIceGatheringNew !=
>                pc_->ice_gathering_state());
>-    candidates_.push_back(rtc::MakeUnique<JsepIceCandidate>(
>+    candidates_.push_back(absl::make_unique<JsepIceCandidate>(
>         candidate->sdp_mid(), candidate->sdp_mline_index(),
>         candidate->candidate()));
>     callback_triggered_ = true;
>@@ -132,6 +137,11 @@ class MockPeerConnectionObserver : public PeerConnectionObserver {
>     add_track_events_.push_back(AddTrackEvent(receiver, streams));
>   }
> 
>+  void OnTrack(
>+      rtc::scoped_refptr<RtpTransceiverInterface> transceiver) override {
>+    on_track_transceivers_.push_back(transceiver);
>+  }
>+
>   void OnRemoveTrack(
>       rtc::scoped_refptr<RtpReceiverInterface> receiver) override {
>     remove_track_events_.push_back(receiver);
>@@ -145,16 +155,33 @@ class MockPeerConnectionObserver : public PeerConnectionObserver {
>     return receivers;
>   }
> 
>-  // Returns the label of the last added stream.
>+  int CountAddTrackEventsForStream(const std::string& stream_id) {
>+    int found_tracks = 0;
>+    for (const AddTrackEvent& event : add_track_events_) {
>+      bool has_stream_id = false;
>+      for (auto stream : event.streams) {
>+        if (stream->id() == stream_id) {
>+          has_stream_id = true;
>+          break;
>+        }
>+      }
>+      if (has_stream_id) {
>+        ++found_tracks;
>+      }
>+    }
>+    return found_tracks;
>+  }
>+
>+  // Returns the id of the last added stream.
>   // Empty string if no stream have been added.
>-  std::string GetLastAddedStreamLabel() {
>+  std::string GetLastAddedStreamId() {
>     if (last_added_stream_.get())
>-      return last_added_stream_->label();
>+      return last_added_stream_->id();
>     return "";
>   }
>-  std::string GetLastRemovedStreamLabel() {
>+  std::string GetLastRemovedStreamId() {
>     if (last_removed_stream_.get())
>-      return last_removed_stream_->label();
>+      return last_removed_stream_->id();
>     return "";
>   }
> 
>@@ -166,6 +193,14 @@ class MockPeerConnectionObserver : public PeerConnectionObserver {
>     }
>   }
> 
>+  std::vector<const IceCandidateInterface*> GetAllCandidates() {
>+    std::vector<const IceCandidateInterface*> candidates;
>+    for (const auto& candidate : candidates_) {
>+      candidates.push_back(candidate.get());
>+    }
>+    return candidates;
>+  }
>+
>   std::vector<IceCandidateInterface*> GetCandidatesByMline(int mline_index) {
>     std::vector<IceCandidateInterface*> candidates;
>     for (const auto& candidate : candidates_) {
>@@ -192,6 +227,8 @@ class MockPeerConnectionObserver : public PeerConnectionObserver {
>   std::string last_added_track_label_;
>   std::vector<AddTrackEvent> add_track_events_;
>   std::vector<rtc::scoped_refptr<RtpReceiverInterface>> remove_track_events_;
>+  std::vector<rtc::scoped_refptr<RtpTransceiverInterface>>
>+      on_track_transceivers_;
>   int num_candidates_removed_ = 0;
> 
>  private:
>@@ -206,14 +243,14 @@ class MockCreateSessionDescriptionObserver
>       : called_(false),
>         error_("MockCreateSessionDescriptionObserver not called") {}
>   virtual ~MockCreateSessionDescriptionObserver() {}
>-  virtual void OnSuccess(SessionDescriptionInterface* desc) {
>+  void OnSuccess(SessionDescriptionInterface* desc) override {
>     called_ = true;
>     error_ = "";
>     desc_.reset(desc);
>   }
>-  virtual void OnFailure(const std::string& error) {
>+  void OnFailure(webrtc::RTCError error) override {
>     called_ = true;
>-    error_ = error;
>+    error_ = error.message();
>   }
>   bool called() const { return called_; }
>   bool result() const { return error_.empty(); }
>@@ -239,10 +276,11 @@ class MockSetSessionDescriptionObserver
>     called_ = true;
>     error_ = "";
>   }
>-  void OnFailure(const std::string& error) override {
>+  void OnFailure(webrtc::RTCError error) override {
>     called_ = true;
>-    error_ = error;
>+    error_ = error.message();
>   }
>+
>   bool called() const { return called_; }
>   bool result() const { return error_.empty(); }
>   const std::string& error() const { return error_; }
>@@ -268,7 +306,7 @@ class MockSetRemoteDescriptionObserver
> 
>  private:
>   // Set on complete, on success this is set to an RTCError::OK() error.
>-  rtc::Optional<RTCError> error_;
>+  absl::optional<RTCError> error_;
> };
> 
> class MockDataChannelObserver : public webrtc::DataChannelObserver {
>@@ -278,9 +316,7 @@ class MockDataChannelObserver : public webrtc::DataChannelObserver {
>     channel_->RegisterObserver(this);
>     state_ = channel_->state();
>   }
>-  virtual ~MockDataChannelObserver() {
>-    channel_->UnregisterObserver();
>-  }
>+  virtual ~MockDataChannelObserver() { channel_->UnregisterObserver(); }
> 
>   void OnBufferedAmountChange(uint64_t previous_amount) override {}
> 
>@@ -317,25 +353,25 @@ class MockStatsObserver : public webrtc::StatsObserver {
>       if (r->type() == StatsReport::kStatsReportTypeSsrc) {
>         stats_.timestamp = r->timestamp();
>         GetIntValue(r, StatsReport::kStatsValueNameAudioOutputLevel,
>-            &stats_.audio_output_level);
>+                    &stats_.audio_output_level);
>         GetIntValue(r, StatsReport::kStatsValueNameAudioInputLevel,
>-            &stats_.audio_input_level);
>+                    &stats_.audio_input_level);
>         GetIntValue(r, StatsReport::kStatsValueNameBytesReceived,
>-            &stats_.bytes_received);
>+                    &stats_.bytes_received);
>         GetIntValue(r, StatsReport::kStatsValueNameBytesSent,
>-            &stats_.bytes_sent);
>+                    &stats_.bytes_sent);
>         GetInt64Value(r, StatsReport::kStatsValueNameCaptureStartNtpTimeMs,
>-            &stats_.capture_start_ntp_time);
>+                      &stats_.capture_start_ntp_time);
>       } else if (r->type() == StatsReport::kStatsReportTypeBwe) {
>         stats_.timestamp = r->timestamp();
>         GetIntValue(r, StatsReport::kStatsValueNameAvailableReceiveBandwidth,
>-            &stats_.available_receive_bandwidth);
>+                    &stats_.available_receive_bandwidth);
>       } else if (r->type() == StatsReport::kStatsReportTypeComponent) {
>         stats_.timestamp = r->timestamp();
>         GetStringValue(r, StatsReport::kStatsValueNameDtlsCipher,
>-            &stats_.dtls_cipher);
>+                       &stats_.dtls_cipher);
>         GetStringValue(r, StatsReport::kStatsValueNameSrtpCipher,
>-            &stats_.srtp_cipher);
>+                       &stats_.srtp_cipher);
>       }
>     }
>   }
>@@ -397,8 +433,8 @@ class MockStatsObserver : public webrtc::StatsObserver {
>   }
> 
>   bool GetInt64Value(const StatsReport* report,
>-                   StatsReport::StatsValueName name,
>-                   int64_t* value) {
>+                     StatsReport::StatsValueName name,
>+                     int64_t* value) {
>     const StatsReport::Value* v = report->FindValue(name);
>     if (v) {
>       // TODO(tommi): We should really just be using an int here :-/
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/peerconnectiontestwrapper.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/peerconnectiontestwrapper.cc
>index aa2f79ea0af..145a09559d5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/peerconnectiontestwrapper.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/peerconnectiontestwrapper.cc
>@@ -10,10 +10,14 @@
> 
> #include <string>
> #include <utility>
>+#include <vector>
> 
>+#include "api/video_codecs/builtin_video_decoder_factory.h"
>+#include "api/video_codecs/builtin_video_encoder_factory.h"
>+#include "modules/audio_processing/include/audio_processing.h"
> #include "p2p/base/fakeportallocator.h"
> #include "pc/sdputils.h"
>-#include "pc/test/fakeperiodicvideocapturer.h"
>+#include "pc/test/fakeperiodicvideotracksource.h"
> #include "pc/test/fakertccertificategenerator.h"
> #include "pc/test/mockpeerconnectionobservers.h"
> #include "pc/test/peerconnectiontestwrapper.h"
>@@ -24,14 +28,16 @@ using webrtc::FakeVideoTrackRenderer;
> using webrtc::IceCandidateInterface;
> using webrtc::MediaConstraintsInterface;
> using webrtc::MediaStreamInterface;
>+using webrtc::MediaStreamTrackInterface;
> using webrtc::MockSetSessionDescriptionObserver;
> using webrtc::PeerConnectionInterface;
>+using webrtc::RtpReceiverInterface;
> using webrtc::SdpType;
> using webrtc::SessionDescriptionInterface;
> using webrtc::VideoTrackInterface;
> 
> namespace {
>-const char kStreamLabelBase[] = "stream_label";
>+const char kStreamIdBase[] = "stream_id";
> const char kVideoTrackLabelBase[] = "video_track";
> const char kAudioTrackLabelBase[] = "audio_track";
> constexpr int kMaxWait = 10000;
>@@ -46,8 +52,8 @@ void PeerConnectionTestWrapper::Connect(PeerConnectionTestWrapper* caller,
>   callee->SignalOnIceCandidateReady.connect(
>       caller, &PeerConnectionTestWrapper::AddIceCandidate);
> 
>-  caller->SignalOnSdpReady.connect(
>-      callee, &PeerConnectionTestWrapper::ReceiveOfferSdp);
>+  caller->SignalOnSdpReady.connect(callee,
>+                                   &PeerConnectionTestWrapper::ReceiveOfferSdp);
>   callee->SignalOnSdpReady.connect(
>       caller, &PeerConnectionTestWrapper::ReceiveAnswerSdp);
> }
>@@ -77,8 +83,11 @@ bool PeerConnectionTestWrapper::CreatePc(
> 
>   peer_connection_factory_ = webrtc::CreatePeerConnectionFactory(
>       network_thread_, worker_thread_, rtc::Thread::Current(),
>-      fake_audio_capture_module_, audio_encoder_factory, audio_decoder_factory,
>-      nullptr, nullptr);
>+      rtc::scoped_refptr<webrtc::AudioDeviceModule>(fake_audio_capture_module_),
>+      audio_encoder_factory, audio_decoder_factory,
>+      webrtc::CreateBuiltinVideoEncoderFactory(),
>+      webrtc::CreateBuiltinVideoDecoderFactory(), nullptr /* audio_mixer */,
>+      nullptr /* audio_processing */);
>   if (!peer_connection_factory_) {
>     return false;
>   }
>@@ -99,12 +108,14 @@ PeerConnectionTestWrapper::CreateDataChannel(
>   return peer_connection_->CreateDataChannel(label, &init);
> }
> 
>-void PeerConnectionTestWrapper::OnAddStream(
>-    rtc::scoped_refptr<MediaStreamInterface> stream) {
>-  RTC_LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_ << ": OnAddStream";
>-  // TODO(ronghuawu): support multiple streams.
>-  if (stream->GetVideoTracks().size() > 0) {
>-    renderer_.reset(new FakeVideoTrackRenderer(stream->GetVideoTracks()[0]));
>+void PeerConnectionTestWrapper::OnAddTrack(
>+    rtc::scoped_refptr<RtpReceiverInterface> receiver,
>+    const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams) {
>+  RTC_LOG(LS_INFO) << "PeerConnectionTestWrapper " << name_ << ": OnAddTrack";
>+  if (receiver->track()->kind() == MediaStreamTrackInterface::kVideoKind) {
>+    auto* video_track =
>+        static_cast<VideoTrackInterface*>(receiver->track().get());
>+    renderer_ = absl::make_unique<FakeVideoTrackRenderer>(video_track);
>   }
> }
> 
>@@ -169,9 +180,8 @@ void PeerConnectionTestWrapper::SetLocalDescription(SdpType type,
>                    << ": SetLocalDescription " << webrtc::SdpTypeToString(type)
>                    << " " << sdp;
> 
>-  rtc::scoped_refptr<MockSetSessionDescriptionObserver>
>-      observer(new rtc::RefCountedObject<
>-                   MockSetSessionDescriptionObserver>());
>+  rtc::scoped_refptr<MockSetSessionDescriptionObserver> observer(
>+      new rtc::RefCountedObject<MockSetSessionDescriptionObserver>());
>   peer_connection_->SetLocalDescription(
>       observer, webrtc::CreateSessionDescription(type, sdp).release());
> }
>@@ -182,9 +192,8 @@ void PeerConnectionTestWrapper::SetRemoteDescription(SdpType type,
>                    << ": SetRemoteDescription " << webrtc::SdpTypeToString(type)
>                    << " " << sdp;
> 
>-  rtc::scoped_refptr<MockSetSessionDescriptionObserver>
>-      observer(new rtc::RefCountedObject<
>-                   MockSetSessionDescriptionObserver>());
>+  rtc::scoped_refptr<MockSetSessionDescriptionObserver> observer(
>+      new rtc::RefCountedObject<MockSetSessionDescriptionObserver>());
>   peer_connection_->SetRemoteDescription(
>       observer, webrtc::CreateSessionDescription(type, sdp).release());
> }
>@@ -240,30 +249,37 @@ bool PeerConnectionTestWrapper::CheckForVideo() {
> }
> 
> void PeerConnectionTestWrapper::GetAndAddUserMedia(
>-    bool audio, const webrtc::FakeConstraints& audio_constraints,
>-    bool video, const webrtc::FakeConstraints& video_constraints) {
>+    bool audio,
>+    const cricket::AudioOptions& audio_options,
>+    bool video,
>+    const webrtc::FakeConstraints& video_constraints) {
>   rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
>-      GetUserMedia(audio, audio_constraints, video, video_constraints);
>-  EXPECT_TRUE(peer_connection_->AddStream(stream));
>+      GetUserMedia(audio, audio_options, video, video_constraints);
>+  for (auto audio_track : stream->GetAudioTracks()) {
>+    EXPECT_TRUE(peer_connection_->AddTrack(audio_track, {stream->id()}).ok());
>+  }
>+  for (auto video_track : stream->GetVideoTracks()) {
>+    EXPECT_TRUE(peer_connection_->AddTrack(video_track, {stream->id()}).ok());
>+  }
> }
> 
> rtc::scoped_refptr<webrtc::MediaStreamInterface>
>-    PeerConnectionTestWrapper::GetUserMedia(
>-        bool audio, const webrtc::FakeConstraints& audio_constraints,
>-        bool video, const webrtc::FakeConstraints& video_constraints) {
>-  std::string label = kStreamLabelBase +
>-      rtc::ToString<int>(
>-          static_cast<int>(peer_connection_->local_streams()->count()));
>+PeerConnectionTestWrapper::GetUserMedia(
>+    bool audio,
>+    const cricket::AudioOptions& audio_options,
>+    bool video,
>+    const webrtc::FakeConstraints& video_constraints) {
>+  std::string stream_id =
>+      kStreamIdBase + rtc::ToString(num_get_user_media_calls_++);
>   rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
>-      peer_connection_factory_->CreateLocalMediaStream(label);
>+      peer_connection_factory_->CreateLocalMediaStream(stream_id);
> 
>   if (audio) {
>-    FakeConstraints constraints = audio_constraints;
>+    cricket::AudioOptions options = audio_options;
>     // Disable highpass filter so that we can get all the test audio frames.
>-    constraints.AddMandatory(
>-        MediaConstraintsInterface::kHighpassFilter, false);
>+    options.highpass_filter = false;
>     rtc::scoped_refptr<webrtc::AudioSourceInterface> source =
>-        peer_connection_factory_->CreateAudioSource(&constraints);
>+        peer_connection_factory_->CreateAudioSource(options);
>     rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
>         peer_connection_factory_->CreateAudioTrack(kAudioTrackLabelBase,
>                                                    source));
>@@ -272,15 +288,14 @@ rtc::scoped_refptr<webrtc::MediaStreamInterface>
> 
>   if (video) {
>     // Set max frame rate to 10fps to reduce the risk of the tests to be flaky.
>-    FakeConstraints constraints = video_constraints;
>-    constraints.SetMandatoryMaxFrameRate(10);
>+    webrtc::FakePeriodicVideoSource::Config config;
>+    config.frame_interval_ms = 100;
> 
>     rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> source =
>-        peer_connection_factory_->CreateVideoSource(
>-            std::unique_ptr<cricket::VideoCapturer>(
>-                new webrtc::FakePeriodicVideoCapturer()),
>-            &constraints);
>-    std::string videotrack_label = label + kVideoTrackLabelBase;
>+        new rtc::RefCountedObject<webrtc::FakePeriodicVideoTrackSource>(
>+            config, /* remote */ false);
>+
>+    std::string videotrack_label = stream_id + kVideoTrackLabelBase;
>     rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track(
>         peer_connection_factory_->CreateVideoTrack(videotrack_label, source));
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/peerconnectiontestwrapper.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/peerconnectiontestwrapper.h
>index 8d43ea99211..f82bcf235ad 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/peerconnectiontestwrapper.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/peerconnectiontestwrapper.h
>@@ -13,12 +13,13 @@
> 
> #include <memory>
> #include <string>
>+#include <vector>
> 
> #include "api/peerconnectioninterface.h"
> #include "api/test/fakeconstraints.h"
> #include "pc/test/fakeaudiocapturemodule.h"
> #include "pc/test/fakevideotrackrenderer.h"
>-#include "rtc_base/sigslot.h"
>+#include "rtc_base/third_party/sigslot/sigslot.h"
> 
> class PeerConnectionTestWrapper
>     : public webrtc::PeerConnectionObserver,
>@@ -47,11 +48,11 @@ class PeerConnectionTestWrapper
> 
>   // Implements PeerConnectionObserver.
>   void OnSignalingChange(
>-     webrtc::PeerConnectionInterface::SignalingState new_state) override {}
>-  void OnAddStream(
>-      rtc::scoped_refptr<webrtc::MediaStreamInterface> stream) override;
>-  void OnRemoveStream(
>-      rtc::scoped_refptr<webrtc::MediaStreamInterface> stream) override {}
>+      webrtc::PeerConnectionInterface::SignalingState new_state) override {}
>+  void OnAddTrack(
>+      rtc::scoped_refptr<webrtc::RtpReceiverInterface> receiver,
>+      const std::vector<rtc::scoped_refptr<webrtc::MediaStreamInterface>>&
>+          streams) override;
>   void OnDataChannel(
>       rtc::scoped_refptr<webrtc::DataChannelInterface> data_channel) override;
>   void OnRenegotiationNeeded() override {}
>@@ -63,27 +64,28 @@ class PeerConnectionTestWrapper
> 
>   // Implements CreateSessionDescriptionObserver.
>   void OnSuccess(webrtc::SessionDescriptionInterface* desc) override;
>-  void OnFailure(const std::string& error) override {}
>+  void OnFailure(webrtc::RTCError) override {}
> 
>   void CreateOffer(const webrtc::MediaConstraintsInterface* constraints);
>   void CreateAnswer(const webrtc::MediaConstraintsInterface* constraints);
>   void ReceiveOfferSdp(const std::string& sdp);
>   void ReceiveAnswerSdp(const std::string& sdp);
>-  void AddIceCandidate(const std::string& sdp_mid, int sdp_mline_index,
>+  void AddIceCandidate(const std::string& sdp_mid,
>+                       int sdp_mline_index,
>                        const std::string& candidate);
>   void WaitForCallEstablished();
>   void WaitForConnection();
>   void WaitForAudio();
>   void WaitForVideo();
>-  void GetAndAddUserMedia(
>-    bool audio, const webrtc::FakeConstraints& audio_constraints,
>-    bool video, const webrtc::FakeConstraints& video_constraints);
>+  void GetAndAddUserMedia(bool audio,
>+                          const cricket::AudioOptions& audio_options,
>+                          bool video,
>+                          const webrtc::FakeConstraints& video_constraints);
> 
>   // sigslots
>   sigslot::signal1<std::string*> SignalOnIceCandidateCreated;
>-  sigslot::signal3<const std::string&,
>-                   int,
>-                   const std::string&> SignalOnIceCandidateReady;
>+  sigslot::signal3<const std::string&, int, const std::string&>
>+      SignalOnIceCandidateReady;
>   sigslot::signal1<std::string*> SignalOnSdpCreated;
>   sigslot::signal1<const std::string&> SignalOnSdpReady;
>   sigslot::signal1<webrtc::DataChannelInterface*> SignalOnDataChannel;
>@@ -95,8 +97,10 @@ class PeerConnectionTestWrapper
>   bool CheckForAudio();
>   bool CheckForVideo();
>   rtc::scoped_refptr<webrtc::MediaStreamInterface> GetUserMedia(
>-      bool audio, const webrtc::FakeConstraints& audio_constraints,
>-      bool video, const webrtc::FakeConstraints& video_constraints);
>+      bool audio,
>+      const cricket::AudioOptions& audio_options,
>+      bool video,
>+      const webrtc::FakeConstraints& video_constraints);
> 
>   std::string name_;
>   rtc::Thread* const network_thread_;
>@@ -106,6 +110,7 @@ class PeerConnectionTestWrapper
>       peer_connection_factory_;
>   rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_;
>   std::unique_ptr<webrtc::FakeVideoTrackRenderer> renderer_;
>+  int num_get_user_media_calls_ = 0;
> };
> 
> #endif  // PC_TEST_PEERCONNECTIONTESTWRAPPER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/testsdpstrings.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/testsdpstrings.h
>index fc884a1de34..ba139e21915 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/testsdpstrings.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/test/testsdpstrings.h
>@@ -61,7 +61,7 @@ static const char kFireFoxSdpOffer[] =
>     "a=candidate:5 2 UDP 1694302206 74.95.2.170 33611 typ srflx raddr"
>     " 10.0.254.2 rport 58890\r\n"
> #ifdef HAVE_SCTP
>-    "m=application 45536 SCTP/DTLS 5000\r\n"
>+    "m=application 45536 DTLS/SCTP 5000\r\n"
>     "c=IN IP4 74.95.2.170\r\n"
>     "a=fmtp:5000 protocol=webrtc-datachannel;streams=16\r\n"
>     "a=sendrecv\r\n"
>@@ -79,7 +79,7 @@ static const char kFireFoxSdpOffer[] =
>     ;  // NOLINT(whitespace/semicolon)
> 
> // Audio SDP with a limited set of audio codecs.
>-static const char kAudioSdp[] =
>+static const char kAudioSdpPlanB[] =
>     "v=0\r\n"
>     "o=- 7859371131 2 IN IP4 192.168.30.208\r\n"
>     "s=-\r\n"
>@@ -92,7 +92,7 @@ static const char kAudioSdp[] =
>     "a=sendrecv\r\n"
>     "a=rtcp:16000 IN IP4 192.168.30.208\r\n"
>     "a=rtcp-mux\r\n"
>-    "a=crypto:1 AES_CM_128_HMAC_SHA1_80 "
>+    "a=crypto:0 AES_CM_128_HMAC_SHA1_80 "
>     "inline:tvKIFjbMQ7W0/C2RzhwN0oQglj/7GJg+frdsNRxt\r\n"
>     "a=ice-ufrag:AI2sRT3r\r\n"
>     "a=ice-pwd:lByS9z2RSQlSE9XurlvjYmEm\r\n"
>@@ -101,8 +101,33 @@ static const char kAudioSdp[] =
>     "a=ssrc:4227871655 mslabel:1NFAV3iD08ioO2339rQS9pfOI9mDf6GeG9F4\r\n"
>     "a=ssrc:4227871655 label:1NFAV3iD08ioO2339rQS9pfOI9mDf6GeG9F4a0\r\n"
>     "a=mid:audio\r\n";
>+// Same string as above but with the MID changed to the Unified Plan default.
>+// This is needed so that this SDP can be used as an answer for a Unified Plan
>+// offer.
>+static const char kAudioSdpUnifiedPlan[] =
>+    "v=0\r\n"
>+    "o=- 7859371131 2 IN IP4 192.168.30.208\r\n"
>+    "s=-\r\n"
>+    "c=IN IP4 192.168.30.208\r\n"
>+    "t=0 0\r\n"
>+    "m=audio 16000 RTP/SAVPF 0 8 126\r\n"
>+    "a=rtpmap:0 PCMU/8000\r\n"
>+    "a=rtpmap:8 PCMA/8000\r\n"
>+    "a=rtpmap:126 telephone-event/8000\r\n"
>+    "a=sendrecv\r\n"
>+    "a=rtcp:16000 IN IP4 192.168.30.208\r\n"
>+    "a=rtcp-mux\r\n"
>+    "a=crypto:0 AES_CM_128_HMAC_SHA1_80 "
>+    "inline:tvKIFjbMQ7W0/C2RzhwN0oQglj/7GJg+frdsNRxt\r\n"
>+    "a=ice-ufrag:AI2sRT3r\r\n"
>+    "a=ice-pwd:lByS9z2RSQlSE9XurlvjYmEm\r\n"
>+    "a=ssrc:4227871655 cname:GeAAgb6XCPNLVMX5\r\n"
>+    "a=ssrc:4227871655 msid:1NFAV3iD08ioO2339rQS9pfOI9mDf6GeG9F4 a0\r\n"
>+    "a=ssrc:4227871655 mslabel:1NFAV3iD08ioO2339rQS9pfOI9mDf6GeG9F4\r\n"
>+    "a=ssrc:4227871655 label:1NFAV3iD08ioO2339rQS9pfOI9mDf6GeG9F4a0\r\n"
>+    "a=mid:0\r\n";
> 
>-static const char kAudioSdpWithUnsupportedCodecs[] =
>+static const char kAudioSdpWithUnsupportedCodecsPlanB[] =
>     "v=0\r\n"
>     "o=- 6858750541 2 IN IP4 192.168.30.208\r\n"
>     "s=-\r\n"
>@@ -117,7 +142,7 @@ static const char kAudioSdpWithUnsupportedCodecs[] =
>     "a=sendonly\r\n"
>     "a=rtcp:16000 IN IP4 192.168.30.208\r\n"
>     "a=rtcp-mux\r\n"
>-    "a=crypto:1 AES_CM_128_HMAC_SHA1_80 "
>+    "a=crypto:0 AES_CM_128_HMAC_SHA1_80 "
>     "inline:tvKIFjbMQ7W0/C2RzhwN0oQglj/7GJg+frdsNRxt\r\n"
>     "a=ice-ufrag:AI2sRT3r\r\n"
>     "a=ice-pwd:lByS9z2RSQlSE9XurlvjYmEm\r\n"
>@@ -126,6 +151,33 @@ static const char kAudioSdpWithUnsupportedCodecs[] =
>     "a=ssrc:4227871655 mslabel:7nU0TApbB-n4dfPlCplWT9QTEsbBDS1IlpW3\r\n"
>     "a=ssrc:4227871655 label:7nU0TApbB-n4dfPlCplWT9QTEsbBDS1IlpW3a0\r\n"
>     "a=mid:audio\r\n";
>+// Same string as above but with the MID changed to the Unified Plan default.
>+// This is needed so that this SDP can be used as an answer for a Unified Plan
>+// offer.
>+static const char kAudioSdpWithUnsupportedCodecsUnifiedPlan[] =
>+    "v=0\r\n"
>+    "o=- 6858750541 2 IN IP4 192.168.30.208\r\n"
>+    "s=-\r\n"
>+    "c=IN IP4 192.168.30.208\r\n"
>+    "t=0 0\r\n"
>+    "m=audio 16000 RTP/SAVPF 0 8 18 110 126\r\n"
>+    "a=rtpmap:0 PCMU/8000\r\n"
>+    "a=rtpmap:8 PCMA/8000\r\n"
>+    "a=rtpmap:18 WeirdCodec1/8000\r\n"
>+    "a=rtpmap:110 WeirdCodec2/8000\r\n"
>+    "a=rtpmap:126 telephone-event/8000\r\n"
>+    "a=sendonly\r\n"
>+    "a=rtcp:16000 IN IP4 192.168.30.208\r\n"
>+    "a=rtcp-mux\r\n"
>+    "a=crypto:0 AES_CM_128_HMAC_SHA1_80 "
>+    "inline:tvKIFjbMQ7W0/C2RzhwN0oQglj/7GJg+frdsNRxt\r\n"
>+    "a=ice-ufrag:AI2sRT3r\r\n"
>+    "a=ice-pwd:lByS9z2RSQlSE9XurlvjYmEm\r\n"
>+    "a=ssrc:4227871655 cname:TsmD02HRfhkJBm4m\r\n"
>+    "a=ssrc:4227871655 msid:7nU0TApbB-n4dfPlCplWT9QTEsbBDS1IlpW3 a0\r\n"
>+    "a=ssrc:4227871655 mslabel:7nU0TApbB-n4dfPlCplWT9QTEsbBDS1IlpW3\r\n"
>+    "a=ssrc:4227871655 label:7nU0TApbB-n4dfPlCplWT9QTEsbBDS1IlpW3a0\r\n"
>+    "a=mid:0\r\n";
> 
> }  // namespace webrtc
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/trackmediainfomap.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/trackmediainfomap.cc
>index 36e4eb3a953..b5abb7e9365 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/trackmediainfomap.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/trackmediainfomap.cc
>@@ -10,6 +10,7 @@
> 
> #include "pc/trackmediainfomap.h"
> 
>+#include <string>
> #include <utility>
> 
> namespace webrtc {
>@@ -29,8 +30,8 @@ const V* FindAddressOrNull(const std::map<K, V>& map, const K& key) {
> }
> 
> void GetAudioAndVideoTrackBySsrc(
>-    const std::vector<rtc::scoped_refptr<RtpSenderInterface>>& rtp_senders,
>-    const std::vector<rtc::scoped_refptr<RtpReceiverInterface>>& rtp_receivers,
>+    const std::vector<rtc::scoped_refptr<RtpSenderInternal>>& rtp_senders,
>+    const std::vector<rtc::scoped_refptr<RtpReceiverInternal>>& rtp_receivers,
>     std::map<uint32_t, AudioTrackInterface*>* local_audio_track_by_ssrc,
>     std::map<uint32_t, VideoTrackInterface*>* local_video_track_by_ssrc,
>     std::map<uint32_t, AudioTrackInterface*>* remote_audio_track_by_ssrc,
>@@ -46,7 +47,7 @@ void GetAudioAndVideoTrackBySsrc(
>   // means one thread jump if on signaling thread and two thread jumps if on any
>   // other threads). Is there a way to avoid thread jump(s) on a per
>   // sender/receiver, per method basis?
>-  for (const rtc::scoped_refptr<RtpSenderInterface>& rtp_sender : rtp_senders) {
>+  for (auto rtp_sender : rtp_senders) {
>     cricket::MediaType media_type = rtp_sender->media_type();
>     MediaStreamTrackInterface* track = rtp_sender->track();
>     if (!track) {
>@@ -71,8 +72,7 @@ void GetAudioAndVideoTrackBySsrc(
>       }
>     }
>   }
>-  for (const rtc::scoped_refptr<RtpReceiverInterface>& rtp_receiver :
>-       rtp_receivers) {
>+  for (auto rtp_receiver : rtp_receivers) {
>     cricket::MediaType media_type = rtp_receiver->media_type();
>     MediaStreamTrackInterface* track = rtp_receiver->track();
>     RTC_DCHECK(track);
>@@ -110,8 +110,8 @@ void GetAudioAndVideoTrackBySsrc(
> TrackMediaInfoMap::TrackMediaInfoMap(
>     std::unique_ptr<cricket::VoiceMediaInfo> voice_media_info,
>     std::unique_ptr<cricket::VideoMediaInfo> video_media_info,
>-    const std::vector<rtc::scoped_refptr<RtpSenderInterface>>& rtp_senders,
>-    const std::vector<rtc::scoped_refptr<RtpReceiverInterface>>& rtp_receivers)
>+    const std::vector<rtc::scoped_refptr<RtpSenderInternal>>& rtp_senders,
>+    const std::vector<rtc::scoped_refptr<RtpReceiverInternal>>& rtp_receivers)
>     : voice_media_info_(std::move(voice_media_info)),
>       video_media_info_(std::move(video_media_info)) {
>   std::map<uint32_t, AudioTrackInterface*> local_audio_track_by_ssrc;
>@@ -126,10 +126,10 @@ TrackMediaInfoMap::TrackMediaInfoMap(
>       &remote_video_track_by_ssrc, &unsignaled_audio_track,
>       &unsignaled_video_track);
> 
>-  for (auto& sender : rtp_senders) {
>+  for (auto sender : rtp_senders) {
>     attachment_id_by_track_[sender->track()] = sender->AttachmentId();
>   }
>-  for (auto& receiver : rtp_receivers) {
>+  for (auto receiver : rtp_receivers) {
>     attachment_id_by_track_[receiver->track()] = receiver->AttachmentId();
>   }
> 
>@@ -266,11 +266,11 @@ rtc::scoped_refptr<VideoTrackInterface> TrackMediaInfoMap::GetVideoTrack(
>   return FindValueOrNull(video_track_by_receiver_info_, &video_receiver_info);
> }
> 
>-rtc::Optional<int> TrackMediaInfoMap::GetAttachmentIdByTrack(
>+absl::optional<int> TrackMediaInfoMap::GetAttachmentIdByTrack(
>     const MediaStreamTrackInterface* track) const {
>   auto it = attachment_id_by_track_.find(track);
>-  return it != attachment_id_by_track_.end() ? rtc::Optional<int>(it->second)
>-                                             : rtc::nullopt;
>+  return it != attachment_id_by_track_.end() ? absl::optional<int>(it->second)
>+                                             : absl::nullopt;
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/trackmediainfomap.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/trackmediainfomap.h
>index c427c47998e..fa1f7e7128d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/trackmediainfomap.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/trackmediainfomap.h
>@@ -13,12 +13,13 @@
> 
> #include <map>
> #include <memory>
>+#include <string>
> #include <vector>
> 
> #include "api/mediastreaminterface.h"
>-#include "api/rtpreceiverinterface.h"
>-#include "api/rtpsenderinterface.h"
> #include "media/base/mediachannel.h"
>+#include "pc/rtpreceiver.h"
>+#include "pc/rtpsender.h"
> #include "rtc_base/refcount.h"
> 
> namespace webrtc {
>@@ -38,8 +39,8 @@ class TrackMediaInfoMap {
>   TrackMediaInfoMap(
>       std::unique_ptr<cricket::VoiceMediaInfo> voice_media_info,
>       std::unique_ptr<cricket::VideoMediaInfo> video_media_info,
>-      const std::vector<rtc::scoped_refptr<RtpSenderInterface>>& rtp_senders,
>-      const std::vector<rtc::scoped_refptr<RtpReceiverInterface>>&
>+      const std::vector<rtc::scoped_refptr<RtpSenderInternal>>& rtp_senders,
>+      const std::vector<rtc::scoped_refptr<RtpReceiverInternal>>&
>           rtp_receivers);
> 
>   const cricket::VoiceMediaInfo* voice_media_info() const {
>@@ -78,10 +79,12 @@ class TrackMediaInfoMap {
>   // It is not going to work if a track is attached multiple times, and
>   // it is not going to work if a received track is attached as a sending
>   // track (loopback).
>-  rtc::Optional<int> GetAttachmentIdByTrack(
>+  absl::optional<int> GetAttachmentIdByTrack(
>       const MediaStreamTrackInterface* track) const;
> 
>  private:
>+  absl::optional<std::string> voice_mid_;
>+  absl::optional<std::string> video_mid_;
>   std::unique_ptr<cricket::VoiceMediaInfo> voice_media_info_;
>   std::unique_ptr<cricket::VideoMediaInfo> video_media_info_;
>   // These maps map tracks (identified by a pointer) to their corresponding info
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/trackmediainfomap_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/trackmediainfomap_unittest.cc
>index 583aa89d739..1b320d021cf 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/trackmediainfomap_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/trackmediainfomap_unittest.cc
>@@ -12,16 +12,17 @@
> 
> #include <initializer_list>
> #include <memory>
>+#include <string>
> #include <utility>
> #include <vector>
> 
> #include "api/rtpreceiverinterface.h"
> #include "api/rtpsenderinterface.h"
>-#include "api/test/mock_rtpreceiver.h"
>-#include "api/test/mock_rtpsender.h"
> #include "media/base/mediachannel.h"
> #include "pc/audiotrack.h"
> #include "pc/test/fakevideotracksource.h"
>+#include "pc/test/mock_rtpreceiverinternal.h"
>+#include "pc/test/mock_rtpsenderinternal.h"
> #include "pc/videotrack.h"
> #include "rtc_base/refcount.h"
> #include "test/gtest.h"
>@@ -41,7 +42,7 @@ RtpParameters CreateRtpParametersWithSsrcs(
>   return params;
> }
> 
>-rtc::scoped_refptr<MockRtpSender> CreateMockRtpSender(
>+rtc::scoped_refptr<MockRtpSenderInternal> CreateMockRtpSender(
>     cricket::MediaType media_type,
>     std::initializer_list<uint32_t> ssrcs,
>     rtc::scoped_refptr<MediaStreamTrackInterface> track) {
>@@ -51,8 +52,8 @@ rtc::scoped_refptr<MockRtpSender> CreateMockRtpSender(
>   } else {
>     first_ssrc = 0;
>   }
>-  rtc::scoped_refptr<MockRtpSender> sender(
>-      new rtc::RefCountedObject<MockRtpSender>());
>+  rtc::scoped_refptr<MockRtpSenderInternal> sender(
>+      new rtc::RefCountedObject<MockRtpSenderInternal>());
>   EXPECT_CALL(*sender, track())
>       .WillRepeatedly(testing::Return(std::move(track)));
>   EXPECT_CALL(*sender, ssrc()).WillRepeatedly(testing::Return(first_ssrc));
>@@ -64,12 +65,12 @@ rtc::scoped_refptr<MockRtpSender> CreateMockRtpSender(
>   return sender;
> }
> 
>-rtc::scoped_refptr<MockRtpReceiver> CreateMockRtpReceiver(
>+rtc::scoped_refptr<MockRtpReceiverInternal> CreateMockRtpReceiver(
>     cricket::MediaType media_type,
>     std::initializer_list<uint32_t> ssrcs,
>     rtc::scoped_refptr<MediaStreamTrackInterface> track) {
>-  rtc::scoped_refptr<MockRtpReceiver> receiver(
>-      new rtc::RefCountedObject<MockRtpReceiver>());
>+  rtc::scoped_refptr<MockRtpReceiverInternal> receiver(
>+      new rtc::RefCountedObject<MockRtpReceiverInternal>());
>   EXPECT_CALL(*receiver, track())
>       .WillRepeatedly(testing::Return(std::move(track)));
>   EXPECT_CALL(*receiver, media_type())
>@@ -107,7 +108,7 @@ class TrackMediaInfoMapTest : public testing::Test {
> 
>   void AddRtpSenderWithSsrcs(std::initializer_list<uint32_t> ssrcs,
>                              MediaStreamTrackInterface* local_track) {
>-    rtc::scoped_refptr<MockRtpSender> rtp_sender = CreateMockRtpSender(
>+    rtc::scoped_refptr<MockRtpSenderInternal> rtp_sender = CreateMockRtpSender(
>         local_track->kind() == MediaStreamTrackInterface::kAudioKind
>             ? cricket::MEDIA_TYPE_AUDIO
>             : cricket::MEDIA_TYPE_VIDEO,
>@@ -135,7 +136,7 @@ class TrackMediaInfoMapTest : public testing::Test {
> 
>   void AddRtpReceiverWithSsrcs(std::initializer_list<uint32_t> ssrcs,
>                                MediaStreamTrackInterface* remote_track) {
>-    rtc::scoped_refptr<MockRtpReceiver> rtp_receiver = CreateMockRtpReceiver(
>+    auto rtp_receiver = CreateMockRtpReceiver(
>         remote_track->kind() == MediaStreamTrackInterface::kAudioKind
>             ? cricket::MEDIA_TYPE_AUDIO
>             : cricket::MEDIA_TYPE_VIDEO,
>@@ -172,8 +173,8 @@ class TrackMediaInfoMapTest : public testing::Test {
>  protected:
>   cricket::VoiceMediaInfo* voice_media_info_;
>   cricket::VideoMediaInfo* video_media_info_;
>-  std::vector<rtc::scoped_refptr<RtpSenderInterface>> rtp_senders_;
>-  std::vector<rtc::scoped_refptr<RtpReceiverInterface>> rtp_receivers_;
>+  std::vector<rtc::scoped_refptr<RtpSenderInternal>> rtp_senders_;
>+  std::vector<rtc::scoped_refptr<RtpReceiverInternal>> rtp_receivers_;
>   std::unique_ptr<TrackMediaInfoMap> map_;
>   rtc::scoped_refptr<AudioTrack> local_audio_track_;
>   rtc::scoped_refptr<AudioTrack> remote_audio_track_;
>@@ -407,7 +408,7 @@ TEST_F(TrackMediaInfoMapTest, GetAttachmentIdByTrack) {
>   CreateMap();
>   EXPECT_EQ(rtp_senders_[0]->AttachmentId(),
>             map_->GetAttachmentIdByTrack(local_audio_track_));
>-  EXPECT_EQ(rtc::nullopt, map_->GetAttachmentIdByTrack(local_video_track_));
>+  EXPECT_EQ(absl::nullopt, map_->GetAttachmentIdByTrack(local_video_track_));
> }
> 
> // Death tests.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/transportcontroller.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/transportcontroller.cc
>deleted file mode 100644
>index bd9ea7505a7..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/transportcontroller.cc
>+++ /dev/null
>@@ -1,1106 +0,0 @@
>-/*
>- *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "pc/transportcontroller.h"
>-
>-#include <algorithm>
>-#include <memory>
>-#include <utility>
>-
>-#include "p2p/base/port.h"
>-#include "rtc_base/bind.h"
>-#include "rtc_base/checks.h"
>-#include "rtc_base/ptr_util.h"
>-#include "rtc_base/thread.h"
>-
>-using webrtc::SdpType;
>-
>-namespace {
>-
>-enum {
>-  MSG_ICECONNECTIONSTATE,
>-  MSG_RECEIVING,
>-  MSG_ICEGATHERINGSTATE,
>-  MSG_CANDIDATESGATHERED,
>-};
>-
>-struct CandidatesData : public rtc::MessageData {
>-  CandidatesData(const std::string& transport_name,
>-                 const cricket::Candidates& candidates)
>-      : transport_name(transport_name), candidates(candidates) {}
>-
>-  std::string transport_name;
>-  cricket::Candidates candidates;
>-};
>-
>-bool VerifyCandidate(const cricket::Candidate& cand, std::string* error) {
>-  // No address zero.
>-  if (cand.address().IsNil() || cand.address().IsAnyIP()) {
>-    *error = "candidate has address of zero";
>-    return false;
>-  }
>-
>-  // Disallow all ports below 1024, except for 80 and 443 on public addresses.
>-  int port = cand.address().port();
>-  if (cand.protocol() == cricket::TCP_PROTOCOL_NAME &&
>-      (cand.tcptype() == cricket::TCPTYPE_ACTIVE_STR || port == 0)) {
>-    // Expected for active-only candidates per
>-    // http://tools.ietf.org/html/rfc6544#section-4.5 so no error.
>-    // Libjingle clients emit port 0, in "active" mode.
>-    return true;
>-  }
>-  if (port < 1024) {
>-    if ((port != 80) && (port != 443)) {
>-      *error = "candidate has port below 1024, but not 80 or 443";
>-      return false;
>-    }
>-
>-    if (cand.address().IsPrivateIP()) {
>-      *error = "candidate has port of 80 or 443 with private IP address";
>-      return false;
>-    }
>-  }
>-
>-  return true;
>-}
>-
>-bool VerifyCandidates(const cricket::Candidates& candidates,
>-                      std::string* error) {
>-  for (const cricket::Candidate& candidate : candidates) {
>-    if (!VerifyCandidate(candidate, error)) {
>-      return false;
>-    }
>-  }
>-  return true;
>-}
>-
>-}  // namespace
>-
>-namespace cricket {
>-
>-// This class groups the DTLS and ICE channels, and helps keep track of
>-// how many external objects (BaseChannels) reference each channel.
>-class TransportController::ChannelPair {
>- public:
>-  // TODO(deadbeef): Change the types of |dtls| and |ice| to
>-  // DtlsTransport and P2PTransportChannelWrapper, once TransportChannelImpl is
>-  // removed.
>-  ChannelPair(DtlsTransportInternal* dtls, IceTransportInternal* ice)
>-      : ice_(ice), dtls_(dtls) {}
>-
>-  // Currently, all ICE-related calls still go through this DTLS channel. But
>-  // that will change once we get rid of TransportChannelImpl, and the DTLS
>-  // channel interface no longer includes ICE-specific methods.
>-  const DtlsTransportInternal* dtls() const { return dtls_.get(); }
>-  DtlsTransportInternal* dtls() { return dtls_.get(); }
>-  const IceTransportInternal* ice() const { return ice_.get(); }
>-  IceTransportInternal* ice() { return ice_.get(); }
>-
>- private:
>-  std::unique_ptr<IceTransportInternal> ice_;
>-  std::unique_ptr<DtlsTransportInternal> dtls_;
>-
>-  RTC_DISALLOW_COPY_AND_ASSIGN(ChannelPair);
>-};
>-
>-TransportController::TransportController(
>-    rtc::Thread* signaling_thread,
>-    rtc::Thread* network_thread,
>-    PortAllocator* port_allocator,
>-    bool redetermine_role_on_ice_restart,
>-    const rtc::CryptoOptions& crypto_options)
>-    : signaling_thread_(signaling_thread),
>-      network_thread_(network_thread),
>-      port_allocator_(port_allocator),
>-      redetermine_role_on_ice_restart_(redetermine_role_on_ice_restart),
>-      crypto_options_(crypto_options) {}
>-
>-TransportController::~TransportController() {
>-  // Channel destructors may try to send packets, so this needs to happen on
>-  // the network thread.
>-  network_thread_->Invoke<void>(
>-      RTC_FROM_HERE,
>-      rtc::Bind(&TransportController::DestroyAllChannels_n, this));
>-}
>-
>-bool TransportController::SetSslMaxProtocolVersion(
>-    rtc::SSLProtocolVersion version) {
>-  return network_thread_->Invoke<bool>(
>-      RTC_FROM_HERE, rtc::Bind(&TransportController::SetSslMaxProtocolVersion_n,
>-                               this, version));
>-}
>-
>-void TransportController::SetIceConfig(const IceConfig& config) {
>-  network_thread_->Invoke<void>(
>-      RTC_FROM_HERE,
>-      rtc::Bind(&TransportController::SetIceConfig_n, this, config));
>-}
>-
>-void TransportController::SetIceRole(IceRole ice_role) {
>-  network_thread_->Invoke<void>(
>-      RTC_FROM_HERE,
>-      rtc::Bind(&TransportController::SetIceRole_n, this, ice_role));
>-}
>-
>-void TransportController::SetNeedsIceRestartFlag() {
>-  for (auto& kv : transports_) {
>-    kv.second->SetNeedsIceRestartFlag();
>-  }
>-}
>-
>-bool TransportController::NeedsIceRestart(
>-    const std::string& transport_name) const {
>-  const JsepTransport* transport = GetJsepTransport(transport_name);
>-  if (!transport) {
>-    return false;
>-  }
>-  return transport->NeedsIceRestart();
>-}
>-
>-bool TransportController::GetSslRole(const std::string& transport_name,
>-                                     rtc::SSLRole* role) const {
>-  return network_thread_->Invoke<bool>(
>-      RTC_FROM_HERE, rtc::Bind(&TransportController::GetSslRole_n, this,
>-                               transport_name, role));
>-}
>-
>-bool TransportController::SetLocalCertificate(
>-    const rtc::scoped_refptr<rtc::RTCCertificate>& certificate) {
>-  return network_thread_->Invoke<bool>(
>-      RTC_FROM_HERE, rtc::Bind(&TransportController::SetLocalCertificate_n,
>-                               this, certificate));
>-}
>-
>-bool TransportController::GetLocalCertificate(
>-    const std::string& transport_name,
>-    rtc::scoped_refptr<rtc::RTCCertificate>* certificate) const {
>-  if (network_thread_->IsCurrent()) {
>-    return GetLocalCertificate_n(transport_name, certificate);
>-  }
>-  return network_thread_->Invoke<bool>(
>-      RTC_FROM_HERE, rtc::Bind(&TransportController::GetLocalCertificate_n,
>-                               this, transport_name, certificate));
>-}
>-
>-std::unique_ptr<rtc::SSLCertificate>
>-TransportController::GetRemoteSSLCertificate(
>-    const std::string& transport_name) const {
>-  if (network_thread_->IsCurrent()) {
>-    return GetRemoteSSLCertificate_n(transport_name);
>-  }
>-  return network_thread_->Invoke<std::unique_ptr<rtc::SSLCertificate>>(
>-      RTC_FROM_HERE, rtc::Bind(&TransportController::GetRemoteSSLCertificate_n,
>-                               this, transport_name));
>-}
>-
>-bool TransportController::SetLocalTransportDescription(
>-    const std::string& transport_name,
>-    const TransportDescription& tdesc,
>-    SdpType type,
>-    std::string* err) {
>-  return network_thread_->Invoke<bool>(
>-      RTC_FROM_HERE,
>-      rtc::Bind(&TransportController::SetLocalTransportDescription_n, this,
>-                transport_name, tdesc, type, err));
>-}
>-
>-bool TransportController::SetRemoteTransportDescription(
>-    const std::string& transport_name,
>-    const TransportDescription& tdesc,
>-    SdpType type,
>-    std::string* err) {
>-  return network_thread_->Invoke<bool>(
>-      RTC_FROM_HERE,
>-      rtc::Bind(&TransportController::SetRemoteTransportDescription_n, this,
>-                transport_name, tdesc, type, err));
>-}
>-
>-void TransportController::MaybeStartGathering() {
>-  network_thread_->Invoke<void>(
>-      RTC_FROM_HERE,
>-      rtc::Bind(&TransportController::MaybeStartGathering_n, this));
>-}
>-
>-bool TransportController::AddRemoteCandidates(const std::string& transport_name,
>-                                              const Candidates& candidates,
>-                                              std::string* err) {
>-  return network_thread_->Invoke<bool>(
>-      RTC_FROM_HERE, rtc::Bind(&TransportController::AddRemoteCandidates_n,
>-                               this, transport_name, candidates, err));
>-}
>-
>-bool TransportController::RemoveRemoteCandidates(const Candidates& candidates,
>-                                                 std::string* err) {
>-  return network_thread_->Invoke<bool>(
>-      RTC_FROM_HERE, rtc::Bind(&TransportController::RemoveRemoteCandidates_n,
>-                               this, candidates, err));
>-}
>-
>-bool TransportController::ReadyForRemoteCandidates(
>-    const std::string& transport_name) const {
>-  return network_thread_->Invoke<bool>(
>-      RTC_FROM_HERE, rtc::Bind(&TransportController::ReadyForRemoteCandidates_n,
>-                               this, transport_name));
>-}
>-
>-bool TransportController::GetStats(const std::string& transport_name,
>-                                   TransportStats* stats) {
>-  if (network_thread_->IsCurrent()) {
>-    return GetStats_n(transport_name, stats);
>-  }
>-  return network_thread_->Invoke<bool>(
>-      RTC_FROM_HERE,
>-      rtc::Bind(&TransportController::GetStats_n, this, transport_name, stats));
>-}
>-
>-void TransportController::SetMetricsObserver(
>-    webrtc::MetricsObserverInterface* metrics_observer) {
>-  return network_thread_->Invoke<void>(
>-      RTC_FROM_HERE, rtc::Bind(&TransportController::SetMetricsObserver_n, this,
>-                               metrics_observer));
>-}
>-
>-DtlsTransportInternal* TransportController::CreateDtlsTransport(
>-    const std::string& transport_name,
>-    int component) {
>-  return network_thread_->Invoke<DtlsTransportInternal*>(
>-      RTC_FROM_HERE, rtc::Bind(&TransportController::CreateDtlsTransport_n,
>-                               this, transport_name, component));
>-}
>-
>-DtlsTransportInternal* TransportController::CreateDtlsTransport_n(
>-    const std::string& transport_name,
>-    int component) {
>-  RTC_DCHECK(network_thread_->IsCurrent());
>-
>-  RefCountedChannel* existing_channel = GetChannel_n(transport_name, component);
>-  if (existing_channel) {
>-    // Channel already exists; increment reference count and return.
>-    existing_channel->AddRef();
>-    return existing_channel->dtls();
>-  }
>-
>-  // Need to create a new channel.
>-  JsepTransport* transport = GetOrCreateJsepTransport(transport_name);
>-
>-  // Create DTLS channel wrapping ICE channel, and configure it.
>-  IceTransportInternal* ice =
>-      CreateIceTransportChannel_n(transport_name, component);
>-  DtlsTransportInternal* dtls =
>-      CreateDtlsTransportChannel_n(transport_name, component, ice);
>-  dtls->ice_transport()->SetMetricsObserver(metrics_observer_);
>-  dtls->ice_transport()->SetIceRole(ice_role_);
>-  dtls->ice_transport()->SetIceTiebreaker(ice_tiebreaker_);
>-  dtls->ice_transport()->SetIceConfig(ice_config_);
>-  if (certificate_) {
>-    bool set_cert_success = dtls->SetLocalCertificate(certificate_);
>-    RTC_DCHECK(set_cert_success);
>-  }
>-
>-  // Connect to signals offered by the channels. Currently, the DTLS channel
>-  // forwards signals from the ICE channel, so we only need to connect to the
>-  // DTLS channel. In the future this won't be the case.
>-  dtls->SignalWritableState.connect(
>-      this, &TransportController::OnChannelWritableState_n);
>-  dtls->SignalReceivingState.connect(
>-      this, &TransportController::OnChannelReceivingState_n);
>-  dtls->SignalDtlsHandshakeError.connect(
>-      this, &TransportController::OnDtlsHandshakeError);
>-  dtls->ice_transport()->SignalGatheringState.connect(
>-      this, &TransportController::OnChannelGatheringState_n);
>-  dtls->ice_transport()->SignalCandidateGathered.connect(
>-      this, &TransportController::OnChannelCandidateGathered_n);
>-  dtls->ice_transport()->SignalCandidatesRemoved.connect(
>-      this, &TransportController::OnChannelCandidatesRemoved_n);
>-  dtls->ice_transport()->SignalRoleConflict.connect(
>-      this, &TransportController::OnChannelRoleConflict_n);
>-  dtls->ice_transport()->SignalStateChanged.connect(
>-      this, &TransportController::OnChannelStateChanged_n);
>-  RefCountedChannel* new_pair = new RefCountedChannel(dtls, ice);
>-  new_pair->AddRef();
>-  channels_.insert(channels_.end(), new_pair);
>-  bool channel_added = transport->AddChannel(dtls, component);
>-  RTC_DCHECK(channel_added);
>-  // Adding a channel could cause aggregate state to change.
>-  UpdateAggregateStates_n();
>-  return dtls;
>-}
>-
>-void TransportController::DestroyDtlsTransport(
>-    const std::string& transport_name,
>-    int component) {
>-  network_thread_->Invoke<void>(
>-      RTC_FROM_HERE, rtc::Bind(&TransportController::DestroyDtlsTransport_n,
>-                               this, transport_name, component));
>-}
>-
>-void TransportController::DestroyDtlsTransport_n(
>-    const std::string& transport_name,
>-    int component) {
>-  RTC_DCHECK(network_thread_->IsCurrent());
>-  auto it = GetChannelIterator_n(transport_name, component);
>-  if (it == channels_.end()) {
>-    RTC_LOG(LS_WARNING) << "Attempting to delete " << transport_name
>-                        << " TransportChannel " << component
>-                        << ", which doesn't exist.";
>-    return;
>-  }
>-  // Release one reference to the RefCountedChannel, and do additional cleanup
>-  // only if it was the last one. Matches the AddRef logic in
>-  // CreateDtlsTransport_n.
>-  if ((*it)->Release() == rtc::RefCountReleaseStatus::kOtherRefsRemained) {
>-    return;
>-  }
>-  channels_.erase(it);
>-
>-  JsepTransport* t = GetJsepTransport(transport_name);
>-  bool channel_removed = t->RemoveChannel(component);
>-  RTC_DCHECK(channel_removed);
>-  // Just as we create a Transport when its first channel is created,
>-  // we delete it when its last channel is deleted.
>-  if (!t->HasChannels()) {
>-    transports_.erase(transport_name);
>-  }
>-  // Removing a channel could cause aggregate state to change.
>-  UpdateAggregateStates_n();
>-}
>-
>-webrtc::SrtpTransport* TransportController::CreateSdesTransport(
>-    const std::string& transport_name,
>-    bool rtcp_mux_enabled) {
>-  if (!network_thread_->IsCurrent()) {
>-    return network_thread_->Invoke<webrtc::SrtpTransport*>(RTC_FROM_HERE, [&] {
>-      return CreateSdesTransport(transport_name, rtcp_mux_enabled);
>-    });
>-  }
>-
>-  auto existing_rtp_transport = FindRtpTransport(transport_name);
>-
>-  if (existing_rtp_transport) {
>-    // For SRTP transport wrapper, the |srtp_transport| is expected to be
>-    // non-null and |dtls_srtp_transport| is expected to be a nullptr.
>-    if (!existing_rtp_transport->srtp_transport ||
>-        existing_rtp_transport->dtls_srtp_transport) {
>-      RTC_LOG(LS_ERROR)
>-          << "Failed to create an RTP transport for SDES using name: "
>-          << transport_name << " because the type doesn't match.";
>-      return nullptr;
>-    }
>-    existing_rtp_transport->AddRef();
>-    return existing_rtp_transport->srtp_transport;
>-  }
>-
>-  auto new_srtp_transport =
>-      rtc::MakeUnique<webrtc::SrtpTransport>(rtcp_mux_enabled);
>-
>-  // The SDES should use an IceTransport rather than a DtlsTransport. We call
>-  // |CreateDtlsTransport_n| here because the DtlsTransport will downgrade to an
>-  // wrapper over IceTransport if we don't set the certificates and it will just
>-  // forward the packets and signals without using DTLS. The support of SDES
>-  // will be removed once all the downstream application stop using it.
>-  new_srtp_transport->SetRtpPacketTransport(CreateDtlsTransport_n(
>-      transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP));
>-  if (!rtcp_mux_enabled) {
>-    new_srtp_transport->SetRtcpPacketTransport(CreateDtlsTransport_n(
>-        transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTCP));
>-  }
>-
>-#if defined(ENABLE_EXTERNAL_AUTH)
>-  new_srtp_transport->EnableExternalAuth();
>-#endif
>-
>-  auto new_rtp_transport_wrapper = new RefCountedRtpTransport();
>-  new_rtp_transport_wrapper->srtp_transport = new_srtp_transport.get();
>-  new_rtp_transport_wrapper->rtp_transport = std::move(new_srtp_transport);
>-  new_rtp_transport_wrapper->AddRef();
>-  rtp_transports_[transport_name] = new_rtp_transport_wrapper;
>-  return rtp_transports_[transport_name]->srtp_transport;
>-}
>-
>-webrtc::DtlsSrtpTransport* TransportController::CreateDtlsSrtpTransport(
>-    const std::string& transport_name,
>-    bool rtcp_mux_enabled) {
>-  if (!network_thread_->IsCurrent()) {
>-    return network_thread_->Invoke<webrtc::DtlsSrtpTransport*>(
>-        RTC_FROM_HERE, [&] {
>-          return CreateDtlsSrtpTransport(transport_name, rtcp_mux_enabled);
>-        });
>-  }
>-  auto existing_rtp_transport = FindRtpTransport(transport_name);
>-
>-  if (existing_rtp_transport) {
>-    // For DTLS-SRTP transport wrapper, the |dtls_srtp_transport| is expected to
>-    // be non-null and |srtp_transport| is expected to be a nullptr.
>-    if (existing_rtp_transport->srtp_transport ||
>-        !existing_rtp_transport->dtls_srtp_transport) {
>-      RTC_LOG(LS_ERROR)
>-          << "Failed to create an RTP transport for DTLS-SRTP using name: "
>-          << transport_name << " because the type doesn't match.";
>-      return nullptr;
>-    }
>-    existing_rtp_transport->AddRef();
>-    return existing_rtp_transport->dtls_srtp_transport;
>-  }
>-
>-  auto new_srtp_transport =
>-      rtc::MakeUnique<webrtc::SrtpTransport>(rtcp_mux_enabled);
>-
>-#if defined(ENABLE_EXTERNAL_AUTH)
>-  new_srtp_transport->EnableExternalAuth();
>-#endif
>-
>-  auto new_dtls_srtp_transport =
>-      rtc::MakeUnique<webrtc::DtlsSrtpTransport>(std::move(new_srtp_transport));
>-
>-  auto rtp_dtls_transport = CreateDtlsTransport_n(
>-      transport_name, cricket::ICE_CANDIDATE_COMPONENT_RTP);
>-  auto rtcp_dtls_transport =
>-      rtcp_mux_enabled
>-          ? nullptr
>-          : CreateDtlsTransport_n(transport_name,
>-                                  cricket::ICE_CANDIDATE_COMPONENT_RTCP);
>-
>-  new_dtls_srtp_transport->SetDtlsTransports(rtp_dtls_transport,
>-                                             rtcp_dtls_transport);
>-
>-  auto new_rtp_transport_wrapper = new RefCountedRtpTransport();
>-  new_rtp_transport_wrapper->dtls_srtp_transport =
>-      new_dtls_srtp_transport.get();
>-  new_rtp_transport_wrapper->rtp_transport = std::move(new_dtls_srtp_transport);
>-  new_rtp_transport_wrapper->AddRef();
>-  rtp_transports_[transport_name] = new_rtp_transport_wrapper;
>-  return rtp_transports_[transport_name]->dtls_srtp_transport;
>-}
>-
>-void TransportController::DestroyTransport(const std::string& transport_name) {
>-  if (!network_thread_->IsCurrent()) {
>-    network_thread_->Invoke<void>(RTC_FROM_HERE,
>-                                  [&] { DestroyTransport(transport_name); });
>-    return;
>-  }
>-
>-  auto existing_rtp_transport = FindRtpTransport(transport_name);
>-  if (!existing_rtp_transport) {
>-    RTC_LOG(LS_WARNING) << "Attempting to delete " << transport_name
>-                        << " transport , which doesn't exist.";
>-    return;
>-  }
>-  if (existing_rtp_transport->Release() ==
>-      rtc::RefCountReleaseStatus::kDroppedLastRef) {
>-    rtp_transports_.erase(transport_name);
>-  }
>-  return;
>-}
>-
>-std::vector<std::string> TransportController::transport_names_for_testing() {
>-  std::vector<std::string> ret;
>-  for (const auto& kv : transports_) {
>-    ret.push_back(kv.first);
>-  }
>-  return ret;
>-}
>-
>-std::vector<DtlsTransportInternal*>
>-TransportController::channels_for_testing() {
>-  std::vector<DtlsTransportInternal*> ret;
>-  for (RefCountedChannel* channel : channels_) {
>-    ret.push_back(channel->dtls());
>-  }
>-  return ret;
>-}
>-
>-DtlsTransportInternal* TransportController::get_channel_for_testing(
>-    const std::string& transport_name,
>-    int component) {
>-  RefCountedChannel* ch = GetChannel_n(transport_name, component);
>-  return ch ? ch->dtls() : nullptr;
>-}
>-
>-IceTransportInternal* TransportController::CreateIceTransportChannel_n(
>-    const std::string& transport_name,
>-    int component) {
>-  return new P2PTransportChannel(transport_name, component, port_allocator_);
>-}
>-
>-DtlsTransportInternal* TransportController::CreateDtlsTransportChannel_n(
>-    const std::string&,
>-    int,
>-    IceTransportInternal* ice) {
>-  DtlsTransport* dtls = new DtlsTransport(ice, crypto_options_);
>-  dtls->SetSslMaxProtocolVersion(ssl_max_version_);
>-  return dtls;
>-}
>-
>-void TransportController::OnMessage(rtc::Message* pmsg) {
>-  RTC_DCHECK(signaling_thread_->IsCurrent());
>-
>-  switch (pmsg->message_id) {
>-    case MSG_ICECONNECTIONSTATE: {
>-      rtc::TypedMessageData<IceConnectionState>* data =
>-          static_cast<rtc::TypedMessageData<IceConnectionState>*>(pmsg->pdata);
>-      SignalConnectionState(data->data());
>-      delete data;
>-      break;
>-    }
>-    case MSG_RECEIVING: {
>-      rtc::TypedMessageData<bool>* data =
>-          static_cast<rtc::TypedMessageData<bool>*>(pmsg->pdata);
>-      SignalReceiving(data->data());
>-      delete data;
>-      break;
>-    }
>-    case MSG_ICEGATHERINGSTATE: {
>-      rtc::TypedMessageData<IceGatheringState>* data =
>-          static_cast<rtc::TypedMessageData<IceGatheringState>*>(pmsg->pdata);
>-      SignalGatheringState(data->data());
>-      delete data;
>-      break;
>-    }
>-    case MSG_CANDIDATESGATHERED: {
>-      CandidatesData* data = static_cast<CandidatesData*>(pmsg->pdata);
>-      SignalCandidatesGathered(data->transport_name, data->candidates);
>-      delete data;
>-      break;
>-    }
>-    default:
>-      RTC_NOTREACHED();
>-  }
>-}
>-
>-const TransportController::RefCountedRtpTransport*
>-TransportController::FindRtpTransport(const std::string& transport_name) {
>-  auto it = rtp_transports_.find(transport_name);
>-  return it == rtp_transports_.end() ? nullptr : it->second;
>-}
>-
>-std::vector<TransportController::RefCountedChannel*>::iterator
>-TransportController::GetChannelIterator_n(const std::string& transport_name,
>-                                          int component) {
>-  RTC_DCHECK(network_thread_->IsCurrent());
>-  return std::find_if(channels_.begin(), channels_.end(),
>-                      [transport_name, component](RefCountedChannel* channel) {
>-                        return channel->dtls()->transport_name() ==
>-                                   transport_name &&
>-                               channel->dtls()->component() == component;
>-                      });
>-}
>-
>-std::vector<TransportController::RefCountedChannel*>::const_iterator
>-TransportController::GetChannelIterator_n(const std::string& transport_name,
>-                                          int component) const {
>-  RTC_DCHECK(network_thread_->IsCurrent());
>-  return std::find_if(
>-      channels_.begin(), channels_.end(),
>-      [transport_name, component](const RefCountedChannel* channel) {
>-        return channel->dtls()->transport_name() == transport_name &&
>-               channel->dtls()->component() == component;
>-      });
>-}
>-
>-const JsepTransport* TransportController::GetJsepTransport(
>-    const std::string& transport_name) const {
>-  auto it = transports_.find(transport_name);
>-  return (it == transports_.end()) ? nullptr : it->second.get();
>-}
>-
>-JsepTransport* TransportController::GetJsepTransport(
>-    const std::string& transport_name) {
>-  auto it = transports_.find(transport_name);
>-  return (it == transports_.end()) ? nullptr : it->second.get();
>-}
>-
>-const TransportController::RefCountedChannel* TransportController::GetChannel_n(
>-    const std::string& transport_name,
>-    int component) const {
>-  RTC_DCHECK(network_thread_->IsCurrent());
>-  auto it = GetChannelIterator_n(transport_name, component);
>-  return (it == channels_.end()) ? nullptr : *it;
>-}
>-
>-TransportController::RefCountedChannel* TransportController::GetChannel_n(
>-    const std::string& transport_name,
>-    int component) {
>-  RTC_DCHECK(network_thread_->IsCurrent());
>-  auto it = GetChannelIterator_n(transport_name, component);
>-  return (it == channels_.end()) ? nullptr : *it;
>-}
>-
>-JsepTransport* TransportController::GetOrCreateJsepTransport(
>-    const std::string& transport_name) {
>-  RTC_DCHECK(network_thread_->IsCurrent());
>-
>-  JsepTransport* transport = GetJsepTransport(transport_name);
>-  if (transport) {
>-    return transport;
>-  }
>-
>-  transport = new JsepTransport(transport_name, certificate_);
>-  transports_[transport_name] = std::unique_ptr<JsepTransport>(transport);
>-  return transport;
>-}
>-
>-void TransportController::DestroyAllChannels_n() {
>-  RTC_DCHECK(network_thread_->IsCurrent());
>-  transports_.clear();
>-  // TODO(nisse): If |channels_| were a vector of scoped_refptr, we
>-  // wouldn't need this strange hack.
>-  for (RefCountedChannel* channel : channels_) {
>-    // Even though these objects are normally ref-counted, if
>-    // TransportController is deleted while they still have references, just
>-    // remove all references.
>-    while (channel->Release() ==
>-           rtc::RefCountReleaseStatus::kOtherRefsRemained) {
>-    }
>-  }
>-  channels_.clear();
>-}
>-
>-bool TransportController::SetSslMaxProtocolVersion_n(
>-    rtc::SSLProtocolVersion version) {
>-  RTC_DCHECK(network_thread_->IsCurrent());
>-
>-  // Max SSL version can only be set before transports are created.
>-  if (!transports_.empty()) {
>-    return false;
>-  }
>-
>-  ssl_max_version_ = version;
>-  return true;
>-}
>-
>-void TransportController::SetIceConfig_n(const IceConfig& config) {
>-  RTC_DCHECK(network_thread_->IsCurrent());
>-
>-  ice_config_ = config;
>-  for (auto& channel : channels_) {
>-    channel->dtls()->ice_transport()->SetIceConfig(ice_config_);
>-  }
>-}
>-
>-void TransportController::SetIceRole_n(IceRole ice_role) {
>-  RTC_DCHECK(network_thread_->IsCurrent());
>-
>-  ice_role_ = ice_role;
>-  for (auto& channel : channels_) {
>-    channel->dtls()->ice_transport()->SetIceRole(ice_role_);
>-  }
>-}
>-
>-bool TransportController::GetSslRole_n(const std::string& transport_name,
>-                                       rtc::SSLRole* role) const {
>-  RTC_DCHECK(network_thread_->IsCurrent());
>-
>-  const JsepTransport* t = GetJsepTransport(transport_name);
>-  if (!t) {
>-    return false;
>-  }
>-  rtc::Optional<rtc::SSLRole> current_role = t->GetSslRole();
>-  if (!current_role) {
>-    return false;
>-  }
>-  *role = *current_role;
>-  return true;
>-}
>-
>-bool TransportController::SetLocalCertificate_n(
>-    const rtc::scoped_refptr<rtc::RTCCertificate>& certificate) {
>-  RTC_DCHECK(network_thread_->IsCurrent());
>-
>-  // Can't change a certificate, or set a null certificate.
>-  if (certificate_ || !certificate) {
>-    return false;
>-  }
>-  certificate_ = certificate;
>-
>-  // Set certificate for JsepTransport, which verifies it matches the
>-  // fingerprint in SDP, and DTLS transport.
>-  // Fallback from DTLS to SDES is not supported.
>-  for (auto& kv : transports_) {
>-    kv.second->SetLocalCertificate(certificate_);
>-  }
>-  for (auto& channel : channels_) {
>-    bool set_cert_success = channel->dtls()->SetLocalCertificate(certificate_);
>-    RTC_DCHECK(set_cert_success);
>-  }
>-  return true;
>-}
>-
>-bool TransportController::GetLocalCertificate_n(
>-    const std::string& transport_name,
>-    rtc::scoped_refptr<rtc::RTCCertificate>* certificate) const {
>-  RTC_DCHECK(network_thread_->IsCurrent());
>-
>-  const JsepTransport* t = GetJsepTransport(transport_name);
>-  if (!t) {
>-    return false;
>-  }
>-  return t->GetLocalCertificate(certificate);
>-}
>-
>-std::unique_ptr<rtc::SSLCertificate>
>-TransportController::GetRemoteSSLCertificate_n(
>-    const std::string& transport_name) const {
>-  RTC_DCHECK(network_thread_->IsCurrent());
>-
>-  // Get the certificate from the RTP channel's DTLS handshake. Should be
>-  // identical to the RTCP channel's, since they were given the same remote
>-  // fingerprint.
>-  const RefCountedChannel* ch = GetChannel_n(transport_name, 1);
>-  if (!ch) {
>-    return nullptr;
>-  }
>-  return ch->dtls()->GetRemoteSSLCertificate();
>-}
>-
>-bool TransportController::SetLocalTransportDescription_n(
>-    const std::string& transport_name,
>-    const TransportDescription& tdesc,
>-    SdpType type,
>-    std::string* err) {
>-  RTC_DCHECK(network_thread_->IsCurrent());
>-
>-  JsepTransport* transport = GetJsepTransport(transport_name);
>-  if (!transport) {
>-    // If we didn't find a transport, that's not an error;
>-    // it could have been deleted as a result of bundling.
>-    // TODO(deadbeef): Make callers smarter so they won't attempt to set a
>-    // description on a deleted transport.
>-    return true;
>-  }
>-
>-  // The initial offer side may use ICE Lite, in which case, per RFC5245 Section
>-  // 5.1.1, the answer side should take the controlling role if it is in the
>-  // full ICE mode.
>-  //
>-  // When both sides use ICE Lite, the initial offer side must take the
>-  // controlling role, and this is the default logic implemented in
>-  // SetLocalDescription in PeerConnection.
>-  if (transport->remote_description() &&
>-      transport->remote_description()->ice_mode == ICEMODE_LITE &&
>-      ice_role_ == ICEROLE_CONTROLLED && tdesc.ice_mode == ICEMODE_FULL) {
>-    SetIceRole_n(ICEROLE_CONTROLLING);
>-  }
>-
>-  // Older versions of Chrome expect the ICE role to be re-determined when an
>-  // ICE restart occurs, and also don't perform conflict resolution correctly,
>-  // so for now we can't safely stop doing this, unless the application opts in
>-  // by setting |redetermine_role_on_ice_restart_| to false.
>-  // See: https://bugs.chromium.org/p/chromium/issues/detail?id=628676
>-  // TODO(deadbeef): Remove this when these old versions of Chrome reach a low
>-  // enough population.
>-  if (redetermine_role_on_ice_restart_ && transport->local_description() &&
>-      IceCredentialsChanged(transport->local_description()->ice_ufrag,
>-                            transport->local_description()->ice_pwd,
>-                            tdesc.ice_ufrag, tdesc.ice_pwd) &&
>-      // Don't change the ICE role if the remote endpoint is ICE lite; we
>-      // should always be controlling in that case.
>-      (!transport->remote_description() ||
>-       transport->remote_description()->ice_mode != ICEMODE_LITE)) {
>-    IceRole new_ice_role =
>-        (type == SdpType::kOffer) ? ICEROLE_CONTROLLING : ICEROLE_CONTROLLED;
>-    SetIceRole(new_ice_role);
>-  }
>-
>-  RTC_LOG(LS_INFO) << "Set local transport description on " << transport_name;
>-  return transport->SetLocalTransportDescription(tdesc, type, err);
>-}
>-
>-bool TransportController::SetRemoteTransportDescription_n(
>-    const std::string& transport_name,
>-    const TransportDescription& tdesc,
>-    SdpType type,
>-    std::string* err) {
>-  RTC_DCHECK(network_thread_->IsCurrent());
>-
>-  // If our role is ICEROLE_CONTROLLED and the remote endpoint supports only
>-  // ice_lite, this local endpoint should take the CONTROLLING role.
>-  // TODO(deadbeef): This is a session-level attribute, so it really shouldn't
>-  // be in a TransportDescription in the first place...
>-  if (ice_role_ == ICEROLE_CONTROLLED && tdesc.ice_mode == ICEMODE_LITE) {
>-    SetIceRole_n(ICEROLE_CONTROLLING);
>-  }
>-
>-  JsepTransport* transport = GetJsepTransport(transport_name);
>-  if (!transport) {
>-    // If we didn't find a transport, that's not an error;
>-    // it could have been deleted as a result of bundling.
>-    // TODO(deadbeef): Make callers smarter so they won't attempt to set a
>-    // description on a deleted transport.
>-    return true;
>-  }
>-
>-  // If we use ICE Lite and the remote endpoint uses the full implementation of
>-  // ICE, the local endpoint must take the controlled role, and the other side
>-  // must be the controlling role.
>-  if (transport->local_description() &&
>-      transport->local_description()->ice_mode == ICEMODE_LITE &&
>-      ice_role_ == ICEROLE_CONTROLLING && tdesc.ice_mode == ICEMODE_FULL) {
>-    SetIceRole_n(ICEROLE_CONTROLLED);
>-  }
>-
>-  RTC_LOG(LS_INFO) << "Set remote transport description on " << transport_name;
>-  return transport->SetRemoteTransportDescription(tdesc, type, err);
>-}
>-
>-void TransportController::MaybeStartGathering_n() {
>-  for (auto& channel : channels_) {
>-    channel->dtls()->ice_transport()->MaybeStartGathering();
>-  }
>-}
>-
>-bool TransportController::AddRemoteCandidates_n(
>-    const std::string& transport_name,
>-    const Candidates& candidates,
>-    std::string* err) {
>-  RTC_DCHECK(network_thread_->IsCurrent());
>-
>-  // Verify each candidate before passing down to the transport layer.
>-  if (!VerifyCandidates(candidates, err)) {
>-    return false;
>-  }
>-
>-  JsepTransport* transport = GetJsepTransport(transport_name);
>-  if (!transport) {
>-    // If we didn't find a transport, that's not an error;
>-    // it could have been deleted as a result of bundling.
>-    return true;
>-  }
>-
>-  for (const Candidate& candidate : candidates) {
>-    RefCountedChannel* channel =
>-        GetChannel_n(transport_name, candidate.component());
>-    if (!channel) {
>-      *err = "Candidate has an unknown component: " + candidate.ToString() +
>-             " for content: " + transport_name;
>-      return false;
>-    }
>-    channel->dtls()->ice_transport()->AddRemoteCandidate(candidate);
>-  }
>-  return true;
>-}
>-
>-bool TransportController::RemoveRemoteCandidates_n(const Candidates& candidates,
>-                                                   std::string* err) {
>-  RTC_DCHECK(network_thread_->IsCurrent());
>-
>-  // Verify each candidate before passing down to the transport layer.
>-  if (!VerifyCandidates(candidates, err)) {
>-    return false;
>-  }
>-
>-  std::map<std::string, Candidates> candidates_by_transport_name;
>-  for (const Candidate& cand : candidates) {
>-    if (!cand.transport_name().empty()) {
>-      candidates_by_transport_name[cand.transport_name()].push_back(cand);
>-    } else {
>-      RTC_LOG(LS_ERROR) << "Not removing candidate because it does not have a "
>-                           "transport name set: "
>-                        << cand.ToString();
>-    }
>-  }
>-
>-  bool result = true;
>-  for (const auto& kv : candidates_by_transport_name) {
>-    const std::string& transport_name = kv.first;
>-    const Candidates& candidates = kv.second;
>-    JsepTransport* transport = GetJsepTransport(transport_name);
>-    if (!transport) {
>-      // If we didn't find a transport, that's not an error;
>-      // it could have been deleted as a result of bundling.
>-      continue;
>-    }
>-    for (const Candidate& candidate : candidates) {
>-      RefCountedChannel* channel =
>-          GetChannel_n(transport_name, candidate.component());
>-      if (channel) {
>-        channel->dtls()->ice_transport()->RemoveRemoteCandidate(candidate);
>-      }
>-    }
>-  }
>-  return result;
>-}
>-
>-bool TransportController::ReadyForRemoteCandidates_n(
>-    const std::string& transport_name) const {
>-  RTC_DCHECK(network_thread_->IsCurrent());
>-
>-  const JsepTransport* transport = GetJsepTransport(transport_name);
>-  if (!transport) {
>-    return false;
>-  }
>-  return transport->ready_for_remote_candidates();
>-}
>-
>-bool TransportController::GetStats_n(const std::string& transport_name,
>-                                     TransportStats* stats) {
>-  RTC_DCHECK(network_thread_->IsCurrent());
>-
>-  JsepTransport* transport = GetJsepTransport(transport_name);
>-  if (!transport) {
>-    return false;
>-  }
>-  return transport->GetStats(stats);
>-}
>-
>-void TransportController::SetMetricsObserver_n(
>-    webrtc::MetricsObserverInterface* metrics_observer) {
>-  RTC_DCHECK(network_thread_->IsCurrent());
>-  metrics_observer_ = metrics_observer;
>-  for (auto& channel : channels_) {
>-    channel->dtls()->ice_transport()->SetMetricsObserver(metrics_observer);
>-  }
>-}
>-
>-void TransportController::OnChannelWritableState_n(
>-    rtc::PacketTransportInternal* transport) {
>-  RTC_DCHECK(network_thread_->IsCurrent());
>-  RTC_LOG(LS_INFO) << " Transport " << transport->transport_name()
>-                   << " writability changed to " << transport->writable()
>-                   << ".";
>-  UpdateAggregateStates_n();
>-}
>-
>-void TransportController::OnChannelReceivingState_n(
>-    rtc::PacketTransportInternal* transport) {
>-  RTC_DCHECK(network_thread_->IsCurrent());
>-  UpdateAggregateStates_n();
>-}
>-
>-void TransportController::OnChannelGatheringState_n(
>-    IceTransportInternal* channel) {
>-  RTC_DCHECK(network_thread_->IsCurrent());
>-  UpdateAggregateStates_n();
>-}
>-
>-void TransportController::OnChannelCandidateGathered_n(
>-    IceTransportInternal* channel,
>-    const Candidate& candidate) {
>-  RTC_DCHECK(network_thread_->IsCurrent());
>-
>-  // We should never signal peer-reflexive candidates.
>-  if (candidate.type() == PRFLX_PORT_TYPE) {
>-    RTC_NOTREACHED();
>-    return;
>-  }
>-  std::vector<Candidate> candidates;
>-  candidates.push_back(candidate);
>-  CandidatesData* data =
>-      new CandidatesData(channel->transport_name(), candidates);
>-  signaling_thread_->Post(RTC_FROM_HERE, this, MSG_CANDIDATESGATHERED, data);
>-}
>-
>-void TransportController::OnChannelCandidatesRemoved_n(
>-    IceTransportInternal* channel,
>-    const Candidates& candidates) {
>-  invoker_.AsyncInvoke<void>(
>-      RTC_FROM_HERE, signaling_thread_,
>-      rtc::Bind(&TransportController::OnChannelCandidatesRemoved, this,
>-                candidates));
>-}
>-
>-void TransportController::OnChannelCandidatesRemoved(
>-    const Candidates& candidates) {
>-  RTC_DCHECK(signaling_thread_->IsCurrent());
>-  SignalCandidatesRemoved(candidates);
>-}
>-
>-void TransportController::OnChannelRoleConflict_n(
>-    IceTransportInternal* channel) {
>-  RTC_DCHECK(network_thread_->IsCurrent());
>-  // Note: since the role conflict is handled entirely on the network thread,
>-  // we don't need to worry about role conflicts occurring on two ports at once.
>-  // The first one encountered should immediately reverse the role.
>-  IceRole reversed_role = (ice_role_ == ICEROLE_CONTROLLING)
>-                              ? ICEROLE_CONTROLLED
>-                              : ICEROLE_CONTROLLING;
>-  RTC_LOG(LS_INFO) << "Got role conflict; switching to "
>-                   << (reversed_role == ICEROLE_CONTROLLING ? "controlling"
>-                                                            : "controlled")
>-                   << " role.";
>-  SetIceRole_n(reversed_role);
>-}
>-
>-void TransportController::OnChannelStateChanged_n(
>-    IceTransportInternal* channel) {
>-  RTC_DCHECK(network_thread_->IsCurrent());
>-  RTC_LOG(LS_INFO) << channel->transport_name() << " TransportChannel "
>-                   << channel->component()
>-                   << " state changed. Check if state is complete.";
>-  UpdateAggregateStates_n();
>-}
>-
>-void TransportController::UpdateAggregateStates_n() {
>-  RTC_DCHECK(network_thread_->IsCurrent());
>-
>-  IceConnectionState new_connection_state = kIceConnectionConnecting;
>-  IceGatheringState new_gathering_state = kIceGatheringNew;
>-  bool any_receiving = false;
>-  bool any_failed = false;
>-  bool all_connected = !channels_.empty();
>-  bool all_completed = !channels_.empty();
>-  bool any_gathering = false;
>-  bool all_done_gathering = !channels_.empty();
>-  for (const auto& channel : channels_) {
>-    any_receiving = any_receiving || channel->dtls()->receiving();
>-    any_failed = any_failed || channel->dtls()->ice_transport()->GetState() ==
>-                                   IceTransportState::STATE_FAILED;
>-    all_connected = all_connected && channel->dtls()->writable();
>-    all_completed =
>-        all_completed && channel->dtls()->writable() &&
>-        channel->dtls()->ice_transport()->GetState() ==
>-            IceTransportState::STATE_COMPLETED &&
>-        channel->dtls()->ice_transport()->GetIceRole() == ICEROLE_CONTROLLING &&
>-        channel->dtls()->ice_transport()->gathering_state() ==
>-            kIceGatheringComplete;
>-    any_gathering =
>-        any_gathering ||
>-        channel->dtls()->ice_transport()->gathering_state() != kIceGatheringNew;
>-    all_done_gathering = all_done_gathering &&
>-                         channel->dtls()->ice_transport()->gathering_state() ==
>-                             kIceGatheringComplete;
>-  }
>-  if (any_failed) {
>-    new_connection_state = kIceConnectionFailed;
>-  } else if (all_completed) {
>-    new_connection_state = kIceConnectionCompleted;
>-  } else if (all_connected) {
>-    new_connection_state = kIceConnectionConnected;
>-  }
>-  if (connection_state_ != new_connection_state) {
>-    connection_state_ = new_connection_state;
>-    signaling_thread_->Post(
>-        RTC_FROM_HERE, this, MSG_ICECONNECTIONSTATE,
>-        new rtc::TypedMessageData<IceConnectionState>(new_connection_state));
>-  }
>-
>-  if (receiving_ != any_receiving) {
>-    receiving_ = any_receiving;
>-    signaling_thread_->Post(RTC_FROM_HERE, this, MSG_RECEIVING,
>-                            new rtc::TypedMessageData<bool>(any_receiving));
>-  }
>-
>-  if (all_done_gathering) {
>-    new_gathering_state = kIceGatheringComplete;
>-  } else if (any_gathering) {
>-    new_gathering_state = kIceGatheringGathering;
>-  }
>-  if (gathering_state_ != new_gathering_state) {
>-    gathering_state_ = new_gathering_state;
>-    signaling_thread_->Post(
>-        RTC_FROM_HERE, this, MSG_ICEGATHERINGSTATE,
>-        new rtc::TypedMessageData<IceGatheringState>(new_gathering_state));
>-  }
>-}
>-
>-void TransportController::OnDtlsHandshakeError(rtc::SSLHandshakeError error) {
>-  SignalDtlsHandshakeError(error);
>-}
>-
>-}  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/transportcontroller.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/transportcontroller.h
>deleted file mode 100644
>index 67fab734b9b..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/transportcontroller.h
>+++ /dev/null
>@@ -1,312 +0,0 @@
>-/*
>- *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef PC_TRANSPORTCONTROLLER_H_
>-#define PC_TRANSPORTCONTROLLER_H_
>-
>-#include <map>
>-#include <memory>
>-#include <string>
>-#include <vector>
>-
>-#include "api/candidate.h"
>-#include "p2p/base/dtlstransport.h"
>-#include "p2p/base/p2ptransportchannel.h"
>-#include "pc/dtlssrtptransport.h"
>-#include "pc/jseptransport.h"
>-#include "pc/rtptransport.h"
>-#include "pc/srtptransport.h"
>-#include "rtc_base/asyncinvoker.h"
>-#include "rtc_base/constructormagic.h"
>-#include "rtc_base/refcountedobject.h"
>-#include "rtc_base/sigslot.h"
>-#include "rtc_base/sslstreamadapter.h"
>-
>-namespace rtc {
>-class Thread;
>-class PacketTransportInternal;
>-}  // namespace rtc
>-
>-namespace webrtc {
>-class MetricsObserverInterface;
>-}  // namespace webrtc
>-
>-namespace cricket {
>-
>-class TransportController : public sigslot::has_slots<>,
>-                            public rtc::MessageHandler {
>- public:
>-  // If |redetermine_role_on_ice_restart| is true, ICE role is redetermined
>-  // upon setting a local transport description that indicates an ICE restart.
>-  // For the constructor that doesn't take this parameter, it defaults to true.
>-  //
>-  // |crypto_options| is used to determine if created DTLS transports negotiate
>-  // GCM crypto suites or not.
>-  TransportController(rtc::Thread* signaling_thread,
>-                      rtc::Thread* network_thread,
>-                      PortAllocator* port_allocator,
>-                      bool redetermine_role_on_ice_restart,
>-                      const rtc::CryptoOptions& crypto_options);
>-
>-  virtual ~TransportController();
>-
>-  rtc::Thread* signaling_thread() const { return signaling_thread_; }
>-  rtc::Thread* network_thread() const { return network_thread_; }
>-
>-  PortAllocator* port_allocator() const { return port_allocator_; }
>-
>-  // Can only be set before transports are created.
>-  // TODO(deadbeef): Make this an argument to the constructor once BaseSession
>-  // and WebRtcSession are combined
>-  bool SetSslMaxProtocolVersion(rtc::SSLProtocolVersion version);
>-
>-  void SetIceConfig(const IceConfig& config);
>-  void SetIceRole(IceRole ice_role);
>-
>-  // Set the "needs-ice-restart" flag as described in JSEP. After the flag is
>-  // set, offers should generate new ufrags/passwords until an ICE restart
>-  // occurs.
>-  void SetNeedsIceRestartFlag();
>-  // Returns true if the ICE restart flag above was set, and no ICE restart has
>-  // occurred yet for this transport (by applying a local description with
>-  // changed ufrag/password). If the transport has been deleted as a result of
>-  // bundling, returns false.
>-  bool NeedsIceRestart(const std::string& transport_name) const;
>-
>-  bool GetSslRole(const std::string& transport_name, rtc::SSLRole* role) const;
>-
>-  // Specifies the identity to use in this session.
>-  // Can only be called once.
>-  bool SetLocalCertificate(
>-      const rtc::scoped_refptr<rtc::RTCCertificate>& certificate);
>-  bool GetLocalCertificate(
>-      const std::string& transport_name,
>-      rtc::scoped_refptr<rtc::RTCCertificate>* certificate) const;
>-  // Caller owns returned certificate. This method mainly exists for stats
>-  // reporting.
>-  std::unique_ptr<rtc::SSLCertificate> GetRemoteSSLCertificate(
>-      const std::string& transport_name) const;
>-  bool SetLocalTransportDescription(const std::string& transport_name,
>-                                    const TransportDescription& tdesc,
>-                                    webrtc::SdpType type,
>-                                    std::string* err);
>-  bool SetRemoteTransportDescription(const std::string& transport_name,
>-                                     const TransportDescription& tdesc,
>-                                     webrtc::SdpType type,
>-                                     std::string* err);
>-  // Start gathering candidates for any new transports, or transports doing an
>-  // ICE restart.
>-  void MaybeStartGathering();
>-  bool AddRemoteCandidates(const std::string& transport_name,
>-                           const Candidates& candidates,
>-                           std::string* err);
>-  bool RemoveRemoteCandidates(const Candidates& candidates, std::string* err);
>-  bool ReadyForRemoteCandidates(const std::string& transport_name) const;
>-  // TODO(deadbeef): GetStats isn't const because all the way down to
>-  // OpenSSLStreamAdapter,
>-  // GetSslCipherSuite and GetDtlsSrtpCryptoSuite are not const. Fix this.
>-  bool GetStats(const std::string& transport_name, TransportStats* stats);
>-  void SetMetricsObserver(webrtc::MetricsObserverInterface* metrics_observer);
>-
>-  // Creates a channel if it doesn't exist. Otherwise, increments a reference
>-  // count and returns an existing channel.
>-  DtlsTransportInternal* CreateDtlsTransport(const std::string& transport_name,
>-                                             int component);
>-  virtual DtlsTransportInternal* CreateDtlsTransport_n(
>-      const std::string& transport_name,
>-      int component);
>-
>-  // Decrements a channel's reference count, and destroys the channel if
>-  // nothing is referencing it.
>-  virtual void DestroyDtlsTransport(const std::string& transport_name,
>-                                    int component);
>-  virtual void DestroyDtlsTransport_n(const std::string& transport_name,
>-                                      int component);
>-
>-  // Create an SrtpTransport/DtlsSrtpTransport if it doesn't exist.
>-  // Otherwise, increments a reference count and returns the existing one.
>-  // These methods are not currently used but the plan is to transition
>-  // PeerConnection and BaseChannel to use them instead of CreateDtlsTransport.
>-  webrtc::SrtpTransport* CreateSdesTransport(const std::string& transport_name,
>-                                             bool rtcp_mux_enabled);
>-  webrtc::DtlsSrtpTransport* CreateDtlsSrtpTransport(
>-      const std::string& transport_name,
>-      bool rtcp_mux_enabled);
>-
>-  // Destroy an RTP level transport which can be an RtpTransport, an
>-  // SrtpTransport or a DtlsSrtpTransport.
>-  void DestroyTransport(const std::string& transport_name);
>-
>-  // TODO(deadbeef): Remove all for_testing methods!
>-  const rtc::scoped_refptr<rtc::RTCCertificate>& certificate_for_testing()
>-      const {
>-    return certificate_;
>-  }
>-  std::vector<std::string> transport_names_for_testing();
>-  std::vector<DtlsTransportInternal*> channels_for_testing();
>-  DtlsTransportInternal* get_channel_for_testing(
>-      const std::string& transport_name,
>-      int component);
>-
>-  // All of these signals are fired on the signalling thread.
>-
>-  // If any transport failed => failed,
>-  // Else if all completed => completed,
>-  // Else if all connected => connected,
>-  // Else => connecting
>-  sigslot::signal1<IceConnectionState> SignalConnectionState;
>-
>-  // Receiving if any transport is receiving
>-  sigslot::signal1<bool> SignalReceiving;
>-
>-  // If all transports done gathering => complete,
>-  // Else if any are gathering => gathering,
>-  // Else => new
>-  sigslot::signal1<IceGatheringState> SignalGatheringState;
>-
>-  // (transport_name, candidates)
>-  sigslot::signal2<const std::string&, const Candidates&>
>-      SignalCandidatesGathered;
>-
>-  sigslot::signal1<const Candidates&> SignalCandidatesRemoved;
>-
>-  sigslot::signal1<rtc::SSLHandshakeError> SignalDtlsHandshakeError;
>-
>- protected:
>-  // TODO(deadbeef): Get rid of these virtual methods. Used by
>-  // FakeTransportController currently, but FakeTransportController shouldn't
>-  // even be functioning by subclassing TransportController.
>-  virtual IceTransportInternal* CreateIceTransportChannel_n(
>-      const std::string& transport_name,
>-      int component);
>-  virtual DtlsTransportInternal* CreateDtlsTransportChannel_n(
>-      const std::string& transport_name,
>-      int component,
>-      IceTransportInternal* ice);
>-
>- private:
>-  void OnMessage(rtc::Message* pmsg) override;
>-
>-  class ChannelPair;
>-  typedef rtc::RefCountedObject<ChannelPair> RefCountedChannel;
>-
>-  // Wrapper for RtpTransport that keeps a reference count.
>-  // When using SDES, |srtp_transport| is non-null, |dtls_srtp_transport| is
>-  // null and |rtp_transport.get()| == |srtp_transport|,
>-  // When using DTLS-SRTP, |dtls_srtp_transport| is non-null, |srtp_transport|
>-  // is null and |rtp_transport.get()| == |dtls_srtp_transport|,
>-  // When using unencrypted RTP, only |rtp_transport| is non-null.
>-  struct RtpTransportWrapper {
>-    // |rtp_transport| is always non-null.
>-    std::unique_ptr<webrtc::RtpTransportInternal> rtp_transport;
>-    webrtc::SrtpTransport* srtp_transport = nullptr;
>-    webrtc::DtlsSrtpTransport* dtls_srtp_transport = nullptr;
>-  };
>-
>-  typedef rtc::RefCountedObject<RtpTransportWrapper> RefCountedRtpTransport;
>-
>-  const RefCountedRtpTransport* FindRtpTransport(
>-      const std::string& transport_name);
>-
>-  // Helper functions to get a channel or transport, or iterator to it (in case
>-  // it needs to be erased).
>-  std::vector<RefCountedChannel*>::iterator GetChannelIterator_n(
>-      const std::string& transport_name,
>-      int component);
>-  std::vector<RefCountedChannel*>::const_iterator GetChannelIterator_n(
>-      const std::string& transport_name,
>-      int component) const;
>-  const JsepTransport* GetJsepTransport(
>-      const std::string& transport_name) const;
>-  JsepTransport* GetJsepTransport(const std::string& transport_name);
>-  const RefCountedChannel* GetChannel_n(const std::string& transport_name,
>-                                        int component) const;
>-  RefCountedChannel* GetChannel_n(const std::string& transport_name,
>-                                  int component);
>-
>-  JsepTransport* GetOrCreateJsepTransport(const std::string& transport_name);
>-  void DestroyAllChannels_n();
>-
>-  bool SetSslMaxProtocolVersion_n(rtc::SSLProtocolVersion version);
>-  void SetIceConfig_n(const IceConfig& config);
>-  void SetIceRole_n(IceRole ice_role);
>-  bool GetSslRole_n(const std::string& transport_name,
>-                    rtc::SSLRole* role) const;
>-  bool SetLocalCertificate_n(
>-      const rtc::scoped_refptr<rtc::RTCCertificate>& certificate);
>-  bool GetLocalCertificate_n(
>-      const std::string& transport_name,
>-      rtc::scoped_refptr<rtc::RTCCertificate>* certificate) const;
>-  std::unique_ptr<rtc::SSLCertificate> GetRemoteSSLCertificate_n(
>-      const std::string& transport_name) const;
>-  bool SetLocalTransportDescription_n(const std::string& transport_name,
>-                                      const TransportDescription& tdesc,
>-                                      webrtc::SdpType type,
>-                                      std::string* err);
>-  bool SetRemoteTransportDescription_n(const std::string& transport_name,
>-                                       const TransportDescription& tdesc,
>-                                       webrtc::SdpType type,
>-                                       std::string* err);
>-  void MaybeStartGathering_n();
>-  bool AddRemoteCandidates_n(const std::string& transport_name,
>-                             const Candidates& candidates,
>-                             std::string* err);
>-  bool RemoveRemoteCandidates_n(const Candidates& candidates, std::string* err);
>-  bool ReadyForRemoteCandidates_n(const std::string& transport_name) const;
>-  bool GetStats_n(const std::string& transport_name, TransportStats* stats);
>-  void SetMetricsObserver_n(webrtc::MetricsObserverInterface* metrics_observer);
>-
>-  // Handlers for signals from Transport.
>-  void OnChannelWritableState_n(rtc::PacketTransportInternal* transport);
>-  void OnChannelReceivingState_n(rtc::PacketTransportInternal* transport);
>-  void OnChannelGatheringState_n(IceTransportInternal* channel);
>-  void OnChannelCandidateGathered_n(IceTransportInternal* channel,
>-                                    const Candidate& candidate);
>-  void OnChannelCandidatesRemoved(const Candidates& candidates);
>-  void OnChannelCandidatesRemoved_n(IceTransportInternal* channel,
>-                                    const Candidates& candidates);
>-  void OnChannelRoleConflict_n(IceTransportInternal* channel);
>-  void OnChannelStateChanged_n(IceTransportInternal* channel);
>-
>-  void UpdateAggregateStates_n();
>-
>-  void OnDtlsHandshakeError(rtc::SSLHandshakeError error);
>-
>-  rtc::Thread* const signaling_thread_ = nullptr;
>-  rtc::Thread* const network_thread_ = nullptr;
>-  PortAllocator* const port_allocator_ = nullptr;
>-
>-  std::map<std::string, std::unique_ptr<JsepTransport>> transports_;
>-  std::vector<RefCountedChannel*> channels_;
>-
>-  std::map<std::string, RefCountedRtpTransport*> rtp_transports_;
>-
>-  // Aggregate state for TransportChannelImpls.
>-  IceConnectionState connection_state_ = kIceConnectionConnecting;
>-  bool receiving_ = false;
>-  IceGatheringState gathering_state_ = kIceGatheringNew;
>-
>-  IceConfig ice_config_;
>-  IceRole ice_role_ = ICEROLE_CONTROLLING;
>-  bool redetermine_role_on_ice_restart_;
>-  uint64_t ice_tiebreaker_ = rtc::CreateRandomId64();
>-  rtc::CryptoOptions crypto_options_;
>-  rtc::SSLProtocolVersion ssl_max_version_ = rtc::SSL_PROTOCOL_DTLS_12;
>-  rtc::scoped_refptr<rtc::RTCCertificate> certificate_;
>-  rtc::AsyncInvoker invoker_;
>-
>-  webrtc::MetricsObserverInterface* metrics_observer_ = nullptr;
>-
>-  RTC_DISALLOW_COPY_AND_ASSIGN(TransportController);
>-};
>-
>-}  // namespace cricket
>-
>-#endif  // PC_TRANSPORTCONTROLLER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/transportcontroller_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/transportcontroller_unittest.cc
>deleted file mode 100644
>index 05c30c41b6f..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/transportcontroller_unittest.cc
>+++ /dev/null
>@@ -1,1019 +0,0 @@
>-/*
>- *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include <map>
>-#include <memory>
>-
>-#include "p2p/base/dtlstransport.h"
>-#include "p2p/base/fakeportallocator.h"
>-#include "p2p/base/p2ptransportchannel.h"
>-#include "p2p/base/portallocator.h"
>-#include "pc/test/faketransportcontroller.h"
>-#include "pc/transportcontroller.h"
>-#include "rtc_base/fakesslidentity.h"
>-#include "rtc_base/gunit.h"
>-#include "rtc_base/helpers.h"
>-#include "rtc_base/sslidentity.h"
>-#include "rtc_base/thread.h"
>-#include "test/gtest.h"
>-
>-using webrtc::SdpType;
>-
>-static const int kTimeout = 100;
>-static const char kIceUfrag1[] = "TESTICEUFRAG0001";
>-static const char kIcePwd1[] = "TESTICEPWD00000000000001";
>-static const char kIceUfrag2[] = "TESTICEUFRAG0002";
>-static const char kIcePwd2[] = "TESTICEPWD00000000000002";
>-static const char kIceUfrag3[] = "TESTICEUFRAG0003";
>-static const char kIcePwd3[] = "TESTICEPWD00000000000003";
>-static const bool kRtcpMuxEnabled = true;
>-
>-namespace cricket {
>-
>-// Only subclassing from FakeTransportController because currently that's the
>-// only way to have a TransportController with fake ICE/DTLS transports.
>-//
>-// TODO(deadbeef): Pass a "TransportFactory" or something similar into
>-// TransportController, instead of using inheritance in this way for testing.
>-typedef FakeTransportController TransportControllerForTest;
>-
>-class TransportControllerTest : public testing::Test,
>-                                public sigslot::has_slots<> {
>- public:
>-  TransportControllerTest()
>-      : transport_controller_(new TransportControllerForTest()),
>-        signaling_thread_(rtc::Thread::Current()) {
>-    ConnectTransportControllerSignals();
>-  }
>-
>-  void CreateTransportControllerWithNetworkThread() {
>-    if (!network_thread_) {
>-      network_thread_ = rtc::Thread::CreateWithSocketServer();
>-      network_thread_->Start();
>-    }
>-    transport_controller_.reset(
>-        new TransportControllerForTest(network_thread_.get()));
>-    ConnectTransportControllerSignals();
>-  }
>-
>-  void ConnectTransportControllerSignals() {
>-    transport_controller_->SignalConnectionState.connect(
>-        this, &TransportControllerTest::OnConnectionState);
>-    transport_controller_->SignalReceiving.connect(
>-        this, &TransportControllerTest::OnReceiving);
>-    transport_controller_->SignalGatheringState.connect(
>-        this, &TransportControllerTest::OnGatheringState);
>-    transport_controller_->SignalCandidatesGathered.connect(
>-        this, &TransportControllerTest::OnCandidatesGathered);
>-  }
>-
>-  FakeDtlsTransport* CreateFakeDtlsTransport(const std::string& content,
>-                                             int component) {
>-    DtlsTransportInternal* transport =
>-        transport_controller_->CreateDtlsTransport_n(content, component);
>-    return static_cast<FakeDtlsTransport*>(transport);
>-  }
>-
>-  void DestroyFakeDtlsTransport(const std::string& content, int component) {
>-    transport_controller_->DestroyDtlsTransport_n(content, component);
>-  }
>-
>-  Candidate CreateCandidate(int component) {
>-    Candidate c;
>-    c.set_address(rtc::SocketAddress("192.168.1.1", 8000));
>-    c.set_component(1);
>-    c.set_protocol(UDP_PROTOCOL_NAME);
>-    c.set_priority(1);
>-    return c;
>-  }
>-
>-  // Used for thread hopping test.
>-  void CreateFakeDtlsTransportsAndCompleteConnectionOnNetworkThread() {
>-    network_thread_->Invoke<void>(
>-        RTC_FROM_HERE,
>-        rtc::Bind(&TransportControllerTest::
>-                      CreateFakeDtlsTransportsAndCompleteConnection_w,
>-                  this));
>-  }
>-
>-  void CreateFakeDtlsTransportsAndCompleteConnection_w() {
>-    transport_controller_->SetIceRole(ICEROLE_CONTROLLING);
>-    FakeDtlsTransport* transport1 = CreateFakeDtlsTransport("audio", 1);
>-    ASSERT_NE(nullptr, transport1);
>-    FakeDtlsTransport* transport2 = CreateFakeDtlsTransport("video", 1);
>-    ASSERT_NE(nullptr, transport2);
>-
>-    TransportDescription local_desc(std::vector<std::string>(), kIceUfrag1,
>-                                    kIcePwd1, ICEMODE_FULL,
>-                                    CONNECTIONROLE_ACTPASS, nullptr);
>-    std::string err;
>-    transport_controller_->SetLocalTransportDescription("audio", local_desc,
>-                                                        SdpType::kOffer, &err);
>-    transport_controller_->SetLocalTransportDescription("video", local_desc,
>-                                                        SdpType::kOffer, &err);
>-    transport_controller_->MaybeStartGathering();
>-    transport1->fake_ice_transport()->SignalCandidateGathered(
>-        transport1->fake_ice_transport(), CreateCandidate(1));
>-    transport2->fake_ice_transport()->SignalCandidateGathered(
>-        transport2->fake_ice_transport(), CreateCandidate(1));
>-    transport1->fake_ice_transport()->SetCandidatesGatheringComplete();
>-    transport2->fake_ice_transport()->SetCandidatesGatheringComplete();
>-    transport1->fake_ice_transport()->SetConnectionCount(2);
>-    transport2->fake_ice_transport()->SetConnectionCount(2);
>-    transport1->SetReceiving(true);
>-    transport2->SetReceiving(true);
>-    transport1->SetWritable(true);
>-    transport2->SetWritable(true);
>-    transport1->fake_ice_transport()->SetConnectionCount(1);
>-    transport2->fake_ice_transport()->SetConnectionCount(1);
>-  }
>-
>-  IceConfig CreateIceConfig(
>-      int receiving_timeout,
>-      ContinualGatheringPolicy continual_gathering_policy) {
>-    IceConfig config;
>-    config.receiving_timeout = receiving_timeout;
>-    config.continual_gathering_policy = continual_gathering_policy;
>-    return config;
>-  }
>-
>- protected:
>-  void OnConnectionState(IceConnectionState state) {
>-    if (!signaling_thread_->IsCurrent()) {
>-      signaled_on_non_signaling_thread_ = true;
>-    }
>-    connection_state_ = state;
>-    ++connection_state_signal_count_;
>-  }
>-
>-  void OnReceiving(bool receiving) {
>-    if (!signaling_thread_->IsCurrent()) {
>-      signaled_on_non_signaling_thread_ = true;
>-    }
>-    receiving_ = receiving;
>-    ++receiving_signal_count_;
>-  }
>-
>-  void OnGatheringState(IceGatheringState state) {
>-    if (!signaling_thread_->IsCurrent()) {
>-      signaled_on_non_signaling_thread_ = true;
>-    }
>-    gathering_state_ = state;
>-    ++gathering_state_signal_count_;
>-  }
>-
>-  void OnCandidatesGathered(const std::string& transport_name,
>-                            const Candidates& candidates) {
>-    if (!signaling_thread_->IsCurrent()) {
>-      signaled_on_non_signaling_thread_ = true;
>-    }
>-    candidates_[transport_name].insert(candidates_[transport_name].end(),
>-                                       candidates.begin(), candidates.end());
>-    ++candidates_signal_count_;
>-  }
>-
>-  std::unique_ptr<rtc::Thread> network_thread_;  // Not used for most tests.
>-  std::unique_ptr<TransportControllerForTest> transport_controller_;
>-
>-  // Information received from signals from transport controller.
>-  IceConnectionState connection_state_ = kIceConnectionConnecting;
>-  bool receiving_ = false;
>-  IceGatheringState gathering_state_ = kIceGatheringNew;
>-  // transport_name => candidates
>-  std::map<std::string, Candidates> candidates_;
>-  // Counts of each signal emitted.
>-  int connection_state_signal_count_ = 0;
>-  int receiving_signal_count_ = 0;
>-  int gathering_state_signal_count_ = 0;
>-  int candidates_signal_count_ = 0;
>-
>-  // Used to make sure signals only come on signaling thread.
>-  rtc::Thread* const signaling_thread_ = nullptr;
>-  bool signaled_on_non_signaling_thread_ = false;
>-};
>-
>-TEST_F(TransportControllerTest, TestSetIceConfig) {
>-  FakeDtlsTransport* transport1 = CreateFakeDtlsTransport("audio", 1);
>-  ASSERT_NE(nullptr, transport1);
>-
>-  transport_controller_->SetIceConfig(
>-      CreateIceConfig(1000, GATHER_CONTINUALLY));
>-  EXPECT_EQ(1000, transport1->fake_ice_transport()->receiving_timeout());
>-  EXPECT_TRUE(transport1->fake_ice_transport()->gather_continually());
>-
>-  transport_controller_->SetIceConfig(
>-      CreateIceConfig(1000, GATHER_CONTINUALLY_AND_RECOVER));
>-  // Test that value stored in controller is applied to new transports.
>-  FakeDtlsTransport* transport2 = CreateFakeDtlsTransport("video", 1);
>-  ASSERT_NE(nullptr, transport2);
>-  EXPECT_EQ(1000, transport2->fake_ice_transport()->receiving_timeout());
>-  EXPECT_TRUE(transport2->fake_ice_transport()->gather_continually());
>-}
>-
>-TEST_F(TransportControllerTest, TestSetSslMaxProtocolVersion) {
>-  EXPECT_TRUE(transport_controller_->SetSslMaxProtocolVersion(
>-      rtc::SSL_PROTOCOL_DTLS_12));
>-  FakeDtlsTransport* transport = CreateFakeDtlsTransport("audio", 1);
>-
>-  ASSERT_NE(nullptr, transport);
>-  EXPECT_EQ(rtc::SSL_PROTOCOL_DTLS_12, transport->ssl_max_protocol_version());
>-
>-  // Setting max version after transport is created should fail.
>-  EXPECT_FALSE(transport_controller_->SetSslMaxProtocolVersion(
>-      rtc::SSL_PROTOCOL_DTLS_10));
>-}
>-
>-TEST_F(TransportControllerTest, TestSetIceRole) {
>-  FakeDtlsTransport* transport1 = CreateFakeDtlsTransport("audio", 1);
>-  ASSERT_NE(nullptr, transport1);
>-
>-  transport_controller_->SetIceRole(ICEROLE_CONTROLLING);
>-  EXPECT_EQ(ICEROLE_CONTROLLING,
>-            transport1->fake_ice_transport()->GetIceRole());
>-  transport_controller_->SetIceRole(ICEROLE_CONTROLLED);
>-  EXPECT_EQ(ICEROLE_CONTROLLED, transport1->fake_ice_transport()->GetIceRole());
>-
>-  // Test that value stored in controller is applied to new transports.
>-  FakeDtlsTransport* transport2 = CreateFakeDtlsTransport("video", 1);
>-  ASSERT_NE(nullptr, transport2);
>-  EXPECT_EQ(ICEROLE_CONTROLLED, transport2->fake_ice_transport()->GetIceRole());
>-}
>-
>-// Test that when one transport encounters a role conflict, the ICE role is
>-// swapped on every transport.
>-TEST_F(TransportControllerTest, TestIceRoleConflict) {
>-  FakeDtlsTransport* transport1 = CreateFakeDtlsTransport("audio", 1);
>-  ASSERT_NE(nullptr, transport1);
>-  FakeDtlsTransport* transport2 = CreateFakeDtlsTransport("video", 1);
>-  ASSERT_NE(nullptr, transport2);
>-
>-  transport_controller_->SetIceRole(ICEROLE_CONTROLLING);
>-  EXPECT_EQ(ICEROLE_CONTROLLING,
>-            transport1->fake_ice_transport()->GetIceRole());
>-  EXPECT_EQ(ICEROLE_CONTROLLING,
>-            transport2->fake_ice_transport()->GetIceRole());
>-
>-  transport1->fake_ice_transport()->SignalRoleConflict(
>-      transport1->fake_ice_transport());
>-  EXPECT_EQ(ICEROLE_CONTROLLED, transport1->fake_ice_transport()->GetIceRole());
>-  EXPECT_EQ(ICEROLE_CONTROLLED, transport2->fake_ice_transport()->GetIceRole());
>-
>-  // Should be able to handle a second role conflict. The remote endpoint can
>-  // change its role/tie-breaker when it does an ICE restart.
>-  transport2->fake_ice_transport()->SignalRoleConflict(
>-      transport2->fake_ice_transport());
>-  EXPECT_EQ(ICEROLE_CONTROLLING,
>-            transport1->fake_ice_transport()->GetIceRole());
>-  EXPECT_EQ(ICEROLE_CONTROLLING,
>-            transport2->fake_ice_transport()->GetIceRole());
>-}
>-
>-TEST_F(TransportControllerTest, TestGetSslRole) {
>-  rtc::SSLRole role;
>-  CreateFakeDtlsTransport("audio", 1);
>-
>-  // Should return false before role has been negotiated.
>-  EXPECT_FALSE(transport_controller_->GetSslRole("audio", &role));
>-
>-  // To negotiate an SSL role, need to set a local certificate, and
>-  // local/remote transport descriptions with DTLS info.
>-  rtc::scoped_refptr<rtc::RTCCertificate> certificate =
>-      rtc::RTCCertificate::Create(std::unique_ptr<rtc::SSLIdentity>(
>-          rtc::SSLIdentity::Generate("testing", rtc::KT_ECDSA)));
>-  std::unique_ptr<rtc::SSLFingerprint> fingerprint(
>-      rtc::SSLFingerprint::CreateFromCertificate(certificate));
>-  transport_controller_->SetLocalCertificate(certificate);
>-
>-  // Set the same fingerprint on both sides since the remote fingerprint
>-  // doesn't really matter for this test.
>-  TransportDescription local_desc(std::vector<std::string>(), kIceUfrag1,
>-                                  kIcePwd1, ICEMODE_FULL,
>-                                  CONNECTIONROLE_ACTPASS, fingerprint.get());
>-  TransportDescription remote_desc(std::vector<std::string>(), kIceUfrag2,
>-                                   kIcePwd2, ICEMODE_FULL,
>-                                   CONNECTIONROLE_ACTIVE, fingerprint.get());
>-  std::string err;
>-  EXPECT_TRUE(transport_controller_->SetLocalTransportDescription(
>-      "audio", local_desc, SdpType::kOffer, &err));
>-  EXPECT_TRUE(transport_controller_->SetRemoteTransportDescription(
>-      "audio", remote_desc, SdpType::kAnswer, &err));
>-
>-  // Finally we can get the role. Should be "server" since the remote
>-  // endpoint's role was "active".
>-  EXPECT_TRUE(transport_controller_->GetSslRole("audio", &role));
>-  EXPECT_EQ(rtc::SSL_SERVER, role);
>-
>-  // Lastly, test that GetSslRole returns false for a nonexistent transport.
>-  EXPECT_FALSE(transport_controller_->GetSslRole("video", &role));
>-}
>-
>-TEST_F(TransportControllerTest, TestSetAndGetLocalCertificate) {
>-  rtc::scoped_refptr<rtc::RTCCertificate> certificate1 =
>-      rtc::RTCCertificate::Create(std::unique_ptr<rtc::SSLIdentity>(
>-          rtc::SSLIdentity::Generate("session1", rtc::KT_DEFAULT)));
>-  rtc::scoped_refptr<rtc::RTCCertificate> certificate2 =
>-      rtc::RTCCertificate::Create(std::unique_ptr<rtc::SSLIdentity>(
>-          rtc::SSLIdentity::Generate("session2", rtc::KT_DEFAULT)));
>-  rtc::scoped_refptr<rtc::RTCCertificate> returned_certificate;
>-
>-  FakeDtlsTransport* transport1 = CreateFakeDtlsTransport("audio", 1);
>-  ASSERT_NE(nullptr, transport1);
>-
>-  EXPECT_TRUE(transport_controller_->SetLocalCertificate(certificate1));
>-  EXPECT_TRUE(transport_controller_->GetLocalCertificate(
>-      "audio", &returned_certificate));
>-  EXPECT_EQ(certificate1->identity()->certificate().ToPEMString(),
>-            returned_certificate->identity()->certificate().ToPEMString());
>-
>-  // Should fail if called for a nonexistant transport.
>-  EXPECT_FALSE(transport_controller_->GetLocalCertificate(
>-      "video", &returned_certificate));
>-
>-  // Test that identity stored in controller is applied to new transports.
>-  FakeDtlsTransport* transport2 = CreateFakeDtlsTransport("video", 1);
>-  ASSERT_NE(nullptr, transport2);
>-  EXPECT_TRUE(transport_controller_->GetLocalCertificate(
>-      "video", &returned_certificate));
>-  EXPECT_EQ(certificate1->identity()->certificate().ToPEMString(),
>-            returned_certificate->identity()->certificate().ToPEMString());
>-
>-  // Shouldn't be able to change the identity once set.
>-  EXPECT_FALSE(transport_controller_->SetLocalCertificate(certificate2));
>-}
>-
>-TEST_F(TransportControllerTest, TestGetRemoteSSLCertificate) {
>-  rtc::FakeSSLCertificate fake_certificate("fake_data");
>-
>-  FakeDtlsTransport* transport = CreateFakeDtlsTransport("audio", 1);
>-  ASSERT_NE(nullptr, transport);
>-
>-  transport->SetRemoteSSLCertificate(&fake_certificate);
>-  std::unique_ptr<rtc::SSLCertificate> returned_certificate =
>-      transport_controller_->GetRemoteSSLCertificate("audio");
>-  EXPECT_TRUE(returned_certificate);
>-  EXPECT_EQ(fake_certificate.ToPEMString(),
>-            returned_certificate->ToPEMString());
>-
>-  // Should fail if called for a nonexistant transport.
>-  EXPECT_FALSE(transport_controller_->GetRemoteSSLCertificate("video"));
>-}
>-
>-TEST_F(TransportControllerTest, TestSetLocalTransportDescription) {
>-  FakeDtlsTransport* transport = CreateFakeDtlsTransport("audio", 1);
>-  ASSERT_NE(nullptr, transport);
>-  TransportDescription local_desc(std::vector<std::string>(), kIceUfrag1,
>-                                  kIcePwd1, ICEMODE_FULL,
>-                                  CONNECTIONROLE_ACTPASS, nullptr);
>-  std::string err;
>-  EXPECT_TRUE(transport_controller_->SetLocalTransportDescription(
>-      "audio", local_desc, SdpType::kOffer, &err));
>-  // Check that ICE ufrag and pwd were propagated to transport.
>-  EXPECT_EQ(kIceUfrag1, transport->fake_ice_transport()->ice_ufrag());
>-  EXPECT_EQ(kIcePwd1, transport->fake_ice_transport()->ice_pwd());
>-  // After setting local description, we should be able to start gathering
>-  // candidates.
>-  transport_controller_->MaybeStartGathering();
>-  EXPECT_EQ_WAIT(kIceGatheringGathering, gathering_state_, kTimeout);
>-  EXPECT_EQ(1, gathering_state_signal_count_);
>-}
>-
>-TEST_F(TransportControllerTest, TestSetRemoteTransportDescription) {
>-  FakeDtlsTransport* transport = CreateFakeDtlsTransport("audio", 1);
>-  ASSERT_NE(nullptr, transport);
>-  TransportDescription remote_desc(std::vector<std::string>(), kIceUfrag1,
>-                                   kIcePwd1, ICEMODE_FULL,
>-                                   CONNECTIONROLE_ACTPASS, nullptr);
>-  std::string err;
>-  EXPECT_TRUE(transport_controller_->SetRemoteTransportDescription(
>-      "audio", remote_desc, SdpType::kOffer, &err));
>-  // Check that ICE ufrag and pwd were propagated to transport.
>-  EXPECT_EQ(kIceUfrag1, transport->fake_ice_transport()->remote_ice_ufrag());
>-  EXPECT_EQ(kIcePwd1, transport->fake_ice_transport()->remote_ice_pwd());
>-}
>-
>-TEST_F(TransportControllerTest, TestAddRemoteCandidates) {
>-  FakeDtlsTransport* transport = CreateFakeDtlsTransport("audio", 1);
>-  ASSERT_NE(nullptr, transport);
>-  Candidates candidates;
>-  candidates.push_back(CreateCandidate(1));
>-  std::string err;
>-  EXPECT_TRUE(
>-      transport_controller_->AddRemoteCandidates("audio", candidates, &err));
>-  EXPECT_EQ(1U, transport->fake_ice_transport()->remote_candidates().size());
>-}
>-
>-TEST_F(TransportControllerTest, TestReadyForRemoteCandidates) {
>-  FakeDtlsTransport* transport = CreateFakeDtlsTransport("audio", 1);
>-  ASSERT_NE(nullptr, transport);
>-  // We expect to be ready for remote candidates only after local and remote
>-  // descriptions are set.
>-  EXPECT_FALSE(transport_controller_->ReadyForRemoteCandidates("audio"));
>-
>-  std::string err;
>-  TransportDescription remote_desc(std::vector<std::string>(), kIceUfrag1,
>-                                   kIcePwd1, ICEMODE_FULL,
>-                                   CONNECTIONROLE_ACTPASS, nullptr);
>-  EXPECT_TRUE(transport_controller_->SetRemoteTransportDescription(
>-      "audio", remote_desc, SdpType::kOffer, &err));
>-  EXPECT_FALSE(transport_controller_->ReadyForRemoteCandidates("audio"));
>-
>-  TransportDescription local_desc(std::vector<std::string>(), kIceUfrag2,
>-                                  kIcePwd2, ICEMODE_FULL,
>-                                  CONNECTIONROLE_ACTPASS, nullptr);
>-  EXPECT_TRUE(transport_controller_->SetLocalTransportDescription(
>-      "audio", local_desc, SdpType::kAnswer, &err));
>-  EXPECT_TRUE(transport_controller_->ReadyForRemoteCandidates("audio"));
>-}
>-
>-TEST_F(TransportControllerTest, TestGetStats) {
>-  FakeDtlsTransport* transport1 = CreateFakeDtlsTransport("audio", 1);
>-  ASSERT_NE(nullptr, transport1);
>-  FakeDtlsTransport* transport2 = CreateFakeDtlsTransport("audio", 2);
>-  ASSERT_NE(nullptr, transport2);
>-  FakeDtlsTransport* transport3 = CreateFakeDtlsTransport("video", 1);
>-  ASSERT_NE(nullptr, transport3);
>-
>-  TransportStats stats;
>-  EXPECT_TRUE(transport_controller_->GetStats("audio", &stats));
>-  EXPECT_EQ("audio", stats.transport_name);
>-  EXPECT_EQ(2U, stats.channel_stats.size());
>-}
>-
>-// Test that a "transport" from a stats perspective (combination of RTP/RTCP
>-// transports) goes away when all references to its transports are gone.
>-TEST_F(TransportControllerTest, TestCreateAndDestroyFakeDtlsTransport) {
>-  FakeDtlsTransport* transport1 = CreateFakeDtlsTransport("audio", 1);
>-  ASSERT_NE(nullptr, transport1);
>-  FakeDtlsTransport* transport2 = CreateFakeDtlsTransport("audio", 1);
>-  ASSERT_NE(nullptr, transport2);
>-  ASSERT_EQ(transport1, transport2);
>-  FakeDtlsTransport* transport3 = CreateFakeDtlsTransport("audio", 2);
>-  ASSERT_NE(nullptr, transport3);
>-
>-  // Using GetStats to check if transport is destroyed from an outside class's
>-  // perspective.
>-  TransportStats stats;
>-  EXPECT_TRUE(transport_controller_->GetStats("audio", &stats));
>-  DestroyFakeDtlsTransport("audio", 2);
>-  DestroyFakeDtlsTransport("audio", 1);
>-  EXPECT_TRUE(transport_controller_->GetStats("audio", &stats));
>-  DestroyFakeDtlsTransport("audio", 1);
>-  EXPECT_FALSE(transport_controller_->GetStats("audio", &stats));
>-}
>-
>-TEST_F(TransportControllerTest, TestSignalConnectionStateFailed) {
>-  // Need controlling ICE role to get in failed state.
>-  transport_controller_->SetIceRole(ICEROLE_CONTROLLING);
>-  FakeDtlsTransport* transport1 = CreateFakeDtlsTransport("audio", 1);
>-  ASSERT_NE(nullptr, transport1);
>-  FakeDtlsTransport* transport2 = CreateFakeDtlsTransport("video", 1);
>-  ASSERT_NE(nullptr, transport2);
>-
>-  // Should signal "failed" if any transport failed; transport is considered
>-  // failed
>-  // if it previously had a connection but now has none, and gathering is
>-  // complete.
>-  transport1->fake_ice_transport()->SetCandidatesGatheringComplete();
>-  transport1->fake_ice_transport()->SetConnectionCount(1);
>-  transport1->fake_ice_transport()->SetConnectionCount(0);
>-  EXPECT_EQ_WAIT(kIceConnectionFailed, connection_state_, kTimeout);
>-  EXPECT_EQ(1, connection_state_signal_count_);
>-}
>-
>-TEST_F(TransportControllerTest, TestSignalConnectionStateConnected) {
>-  transport_controller_->SetIceRole(ICEROLE_CONTROLLING);
>-  FakeDtlsTransport* transport1 = CreateFakeDtlsTransport("audio", 1);
>-  ASSERT_NE(nullptr, transport1);
>-  FakeDtlsTransport* transport2 = CreateFakeDtlsTransport("video", 1);
>-  ASSERT_NE(nullptr, transport2);
>-  FakeDtlsTransport* transport3 = CreateFakeDtlsTransport("video", 2);
>-  ASSERT_NE(nullptr, transport3);
>-
>-  // First, have one transport connect, and another fail, to ensure that
>-  // the first transport connecting didn't trigger a "connected" state signal.
>-  // We should only get a signal when all are connected.
>-  transport1->fake_ice_transport()->SetConnectionCount(2);
>-  transport1->SetWritable(true);
>-  transport3->fake_ice_transport()->SetCandidatesGatheringComplete();
>-  transport3->fake_ice_transport()->SetConnectionCount(1);
>-  transport3->fake_ice_transport()->SetConnectionCount(0);
>-  EXPECT_EQ_WAIT(kIceConnectionFailed, connection_state_, kTimeout);
>-  // Signal count of 1 means that the only signal emitted was "failed".
>-  EXPECT_EQ(1, connection_state_signal_count_);
>-
>-  // Destroy the failed transport to return to "connecting" state.
>-  DestroyFakeDtlsTransport("video", 2);
>-  EXPECT_EQ_WAIT(kIceConnectionConnecting, connection_state_, kTimeout);
>-  EXPECT_EQ(2, connection_state_signal_count_);
>-
>-  // Make the remaining transport reach a connected state.
>-  transport2->fake_ice_transport()->SetConnectionCount(2);
>-  transport2->SetWritable(true);
>-  EXPECT_EQ_WAIT(kIceConnectionConnected, connection_state_, kTimeout);
>-  EXPECT_EQ(3, connection_state_signal_count_);
>-}
>-
>-TEST_F(TransportControllerTest, TestSignalConnectionStateComplete) {
>-  transport_controller_->SetIceRole(ICEROLE_CONTROLLING);
>-  FakeDtlsTransport* transport1 = CreateFakeDtlsTransport("audio", 1);
>-  ASSERT_NE(nullptr, transport1);
>-  FakeDtlsTransport* transport2 = CreateFakeDtlsTransport("video", 1);
>-  ASSERT_NE(nullptr, transport2);
>-  FakeDtlsTransport* transport3 = CreateFakeDtlsTransport("video", 2);
>-  ASSERT_NE(nullptr, transport3);
>-
>-  // Similar to above test, but we're now reaching the completed state, which
>-  // means only one connection per FakeDtlsTransport.
>-  transport1->fake_ice_transport()->SetCandidatesGatheringComplete();
>-  transport1->fake_ice_transport()->SetConnectionCount(1);
>-  transport1->SetWritable(true);
>-  transport3->fake_ice_transport()->SetCandidatesGatheringComplete();
>-  transport3->fake_ice_transport()->SetConnectionCount(1);
>-  transport3->fake_ice_transport()->SetConnectionCount(0);
>-  EXPECT_EQ_WAIT(kIceConnectionFailed, connection_state_, kTimeout);
>-  // Signal count of 1 means that the only signal emitted was "failed".
>-  EXPECT_EQ(1, connection_state_signal_count_);
>-
>-  // Destroy the failed transport to return to "connecting" state.
>-  DestroyFakeDtlsTransport("video", 2);
>-  EXPECT_EQ_WAIT(kIceConnectionConnecting, connection_state_, kTimeout);
>-  EXPECT_EQ(2, connection_state_signal_count_);
>-
>-  // Make the remaining transport reach a connected state.
>-  transport2->fake_ice_transport()->SetCandidatesGatheringComplete();
>-  transport2->fake_ice_transport()->SetConnectionCount(2);
>-  transport2->SetWritable(true);
>-  EXPECT_EQ_WAIT(kIceConnectionConnected, connection_state_, kTimeout);
>-  EXPECT_EQ(3, connection_state_signal_count_);
>-
>-  // Finally, transition to completed state.
>-  transport2->fake_ice_transport()->SetConnectionCount(1);
>-  EXPECT_EQ_WAIT(kIceConnectionCompleted, connection_state_, kTimeout);
>-  EXPECT_EQ(4, connection_state_signal_count_);
>-}
>-
>-// Make sure that if we're "connected" and remove a transport, we stay in the
>-// "connected" state.
>-TEST_F(TransportControllerTest, TestDestroyTransportAndStayConnected) {
>-  FakeDtlsTransport* transport1 = CreateFakeDtlsTransport("audio", 1);
>-  ASSERT_NE(nullptr, transport1);
>-  FakeDtlsTransport* transport2 = CreateFakeDtlsTransport("video", 1);
>-  ASSERT_NE(nullptr, transport2);
>-
>-  transport1->fake_ice_transport()->SetCandidatesGatheringComplete();
>-  transport1->fake_ice_transport()->SetConnectionCount(2);
>-  transport1->SetWritable(true);
>-  transport2->fake_ice_transport()->SetCandidatesGatheringComplete();
>-  transport2->fake_ice_transport()->SetConnectionCount(2);
>-  transport2->SetWritable(true);
>-  EXPECT_EQ_WAIT(kIceConnectionConnected, connection_state_, kTimeout);
>-  EXPECT_EQ(1, connection_state_signal_count_);
>-
>-  // Destroy one transport, then "complete" the other one, so we reach
>-  // a known state.
>-  DestroyFakeDtlsTransport("video", 1);
>-  transport1->fake_ice_transport()->SetConnectionCount(1);
>-  EXPECT_EQ_WAIT(kIceConnectionCompleted, connection_state_, kTimeout);
>-  // Signal count of 2 means the deletion didn't cause any unexpected signals
>-  EXPECT_EQ(2, connection_state_signal_count_);
>-}
>-
>-// If we destroy the last/only transport, we should simply transition to
>-// "connecting".
>-TEST_F(TransportControllerTest, TestDestroyLastTransportWhileConnected) {
>-  FakeDtlsTransport* transport = CreateFakeDtlsTransport("audio", 1);
>-  ASSERT_NE(nullptr, transport);
>-
>-  transport->fake_ice_transport()->SetCandidatesGatheringComplete();
>-  transport->fake_ice_transport()->SetConnectionCount(2);
>-  transport->SetWritable(true);
>-  EXPECT_EQ_WAIT(kIceConnectionConnected, connection_state_, kTimeout);
>-  EXPECT_EQ(1, connection_state_signal_count_);
>-
>-  DestroyFakeDtlsTransport("audio", 1);
>-  EXPECT_EQ_WAIT(kIceConnectionConnecting, connection_state_, kTimeout);
>-  // Signal count of 2 means the deletion didn't cause any unexpected signals
>-  EXPECT_EQ(2, connection_state_signal_count_);
>-}
>-
>-TEST_F(TransportControllerTest, TestSignalReceiving) {
>-  FakeDtlsTransport* transport1 = CreateFakeDtlsTransport("audio", 1);
>-  ASSERT_NE(nullptr, transport1);
>-  FakeDtlsTransport* transport2 = CreateFakeDtlsTransport("video", 1);
>-  ASSERT_NE(nullptr, transport2);
>-
>-  // Should signal receiving as soon as any transport is receiving.
>-  transport1->SetReceiving(true);
>-  EXPECT_TRUE_WAIT(receiving_, kTimeout);
>-  EXPECT_EQ(1, receiving_signal_count_);
>-
>-  transport2->SetReceiving(true);
>-  transport1->SetReceiving(false);
>-  transport2->SetReceiving(false);
>-  EXPECT_TRUE_WAIT(!receiving_, kTimeout);
>-  EXPECT_EQ(2, receiving_signal_count_);
>-}
>-
>-TEST_F(TransportControllerTest, TestSignalGatheringStateGathering) {
>-  FakeDtlsTransport* transport = CreateFakeDtlsTransport("audio", 1);
>-  ASSERT_NE(nullptr, transport);
>-  transport->fake_ice_transport()->MaybeStartGathering();
>-  // Should be in the gathering state as soon as any transport starts gathering.
>-  EXPECT_EQ_WAIT(kIceGatheringGathering, gathering_state_, kTimeout);
>-  EXPECT_EQ(1, gathering_state_signal_count_);
>-}
>-
>-TEST_F(TransportControllerTest, TestSignalGatheringStateComplete) {
>-  FakeDtlsTransport* transport1 = CreateFakeDtlsTransport("audio", 1);
>-  ASSERT_NE(nullptr, transport1);
>-  FakeDtlsTransport* transport2 = CreateFakeDtlsTransport("video", 1);
>-  ASSERT_NE(nullptr, transport2);
>-  FakeDtlsTransport* transport3 = CreateFakeDtlsTransport("data", 1);
>-  ASSERT_NE(nullptr, transport3);
>-
>-  transport3->fake_ice_transport()->MaybeStartGathering();
>-  EXPECT_EQ_WAIT(kIceGatheringGathering, gathering_state_, kTimeout);
>-  EXPECT_EQ(1, gathering_state_signal_count_);
>-
>-  // Have one transport finish gathering, then destroy it, to make sure
>-  // gathering
>-  // completion wasn't signalled if only one transport finished gathering.
>-  transport3->fake_ice_transport()->SetCandidatesGatheringComplete();
>-  DestroyFakeDtlsTransport("data", 1);
>-  EXPECT_EQ_WAIT(kIceGatheringNew, gathering_state_, kTimeout);
>-  EXPECT_EQ(2, gathering_state_signal_count_);
>-
>-  // Make remaining transports start and then finish gathering.
>-  transport1->fake_ice_transport()->MaybeStartGathering();
>-  transport2->fake_ice_transport()->MaybeStartGathering();
>-  EXPECT_EQ_WAIT(kIceGatheringGathering, gathering_state_, kTimeout);
>-  EXPECT_EQ(3, gathering_state_signal_count_);
>-
>-  transport1->fake_ice_transport()->SetCandidatesGatheringComplete();
>-  transport2->fake_ice_transport()->SetCandidatesGatheringComplete();
>-  EXPECT_EQ_WAIT(kIceGatheringComplete, gathering_state_, kTimeout);
>-  EXPECT_EQ(4, gathering_state_signal_count_);
>-}
>-
>-// Test that when the last transport that hasn't finished connecting and/or
>-// gathering is destroyed, the aggregate state jumps to "completed". This can
>-// happen if, for example, we have an audio and video transport, the audio
>-// transport completes, then we start bundling video on the audio transport.
>-TEST_F(TransportControllerTest,
>-       TestSignalingWhenLastIncompleteTransportDestroyed) {
>-  transport_controller_->SetIceRole(ICEROLE_CONTROLLING);
>-  FakeDtlsTransport* transport1 = CreateFakeDtlsTransport("audio", 1);
>-  ASSERT_NE(nullptr, transport1);
>-  FakeDtlsTransport* transport2 = CreateFakeDtlsTransport("video", 1);
>-  ASSERT_NE(nullptr, transport2);
>-
>-  transport1->fake_ice_transport()->SetCandidatesGatheringComplete();
>-  EXPECT_EQ_WAIT(kIceGatheringGathering, gathering_state_, kTimeout);
>-  EXPECT_EQ(1, gathering_state_signal_count_);
>-
>-  transport1->fake_ice_transport()->SetConnectionCount(1);
>-  transport1->SetWritable(true);
>-  DestroyFakeDtlsTransport("video", 1);
>-  EXPECT_EQ_WAIT(kIceConnectionCompleted, connection_state_, kTimeout);
>-  EXPECT_EQ(1, connection_state_signal_count_);
>-  EXPECT_EQ_WAIT(kIceGatheringComplete, gathering_state_, kTimeout);
>-  EXPECT_EQ(2, gathering_state_signal_count_);
>-}
>-
>-TEST_F(TransportControllerTest, TestSignalCandidatesGathered) {
>-  FakeDtlsTransport* transport = CreateFakeDtlsTransport("audio", 1);
>-  ASSERT_NE(nullptr, transport);
>-
>-  // Transport won't signal candidates until it has a local description.
>-  TransportDescription local_desc(std::vector<std::string>(), kIceUfrag1,
>-                                  kIcePwd1, ICEMODE_FULL,
>-                                  CONNECTIONROLE_ACTPASS, nullptr);
>-  std::string err;
>-  EXPECT_TRUE(transport_controller_->SetLocalTransportDescription(
>-      "audio", local_desc, SdpType::kOffer, &err));
>-  transport_controller_->MaybeStartGathering();
>-
>-  transport->fake_ice_transport()->SignalCandidateGathered(
>-      transport->fake_ice_transport(), CreateCandidate(1));
>-  EXPECT_EQ_WAIT(1, candidates_signal_count_, kTimeout);
>-  EXPECT_EQ(1U, candidates_["audio"].size());
>-}
>-
>-TEST_F(TransportControllerTest, TestSignalingOccursOnSignalingThread) {
>-  CreateTransportControllerWithNetworkThread();
>-  CreateFakeDtlsTransportsAndCompleteConnectionOnNetworkThread();
>-
>-  // connecting --> connected --> completed
>-  EXPECT_EQ_WAIT(kIceConnectionCompleted, connection_state_, kTimeout);
>-  EXPECT_EQ(2, connection_state_signal_count_);
>-
>-  EXPECT_TRUE_WAIT(receiving_, kTimeout);
>-  EXPECT_EQ(1, receiving_signal_count_);
>-
>-  // new --> gathering --> complete
>-  EXPECT_EQ_WAIT(kIceGatheringComplete, gathering_state_, kTimeout);
>-  EXPECT_EQ(2, gathering_state_signal_count_);
>-
>-  EXPECT_EQ_WAIT(1U, candidates_["audio"].size(), kTimeout);
>-  EXPECT_EQ_WAIT(1U, candidates_["video"].size(), kTimeout);
>-  EXPECT_EQ(2, candidates_signal_count_);
>-
>-  EXPECT_TRUE(!signaled_on_non_signaling_thread_);
>-}
>-
>-// Older versions of Chrome expect the ICE role to be re-determined when an
>-// ICE restart occurs, and also don't perform conflict resolution correctly,
>-// so for now we can't safely stop doing this.
>-// See: https://bugs.chromium.org/p/chromium/issues/detail?id=628676
>-// TODO(deadbeef): Remove this when these old versions of Chrome reach a low
>-// enough population.
>-TEST_F(TransportControllerTest, IceRoleRedeterminedOnIceRestartByDefault) {
>-  FakeDtlsTransport* transport = CreateFakeDtlsTransport("audio", 1);
>-  ASSERT_NE(nullptr, transport);
>-  std::string err;
>-  // Do an initial offer answer, so that the next offer is an ICE restart.
>-  transport_controller_->SetIceRole(ICEROLE_CONTROLLED);
>-  TransportDescription remote_desc(std::vector<std::string>(), kIceUfrag1,
>-                                   kIcePwd1, ICEMODE_FULL,
>-                                   CONNECTIONROLE_ACTPASS, nullptr);
>-  EXPECT_TRUE(transport_controller_->SetRemoteTransportDescription(
>-      "audio", remote_desc, SdpType::kOffer, &err));
>-  TransportDescription local_desc(std::vector<std::string>(), kIceUfrag2,
>-                                  kIcePwd2, ICEMODE_FULL,
>-                                  CONNECTIONROLE_ACTPASS, nullptr);
>-  EXPECT_TRUE(transport_controller_->SetLocalTransportDescription(
>-      "audio", local_desc, SdpType::kAnswer, &err));
>-  EXPECT_EQ(ICEROLE_CONTROLLED, transport->fake_ice_transport()->GetIceRole());
>-
>-  // The endpoint that initiated an ICE restart should take the controlling
>-  // role.
>-  TransportDescription ice_restart_desc(std::vector<std::string>(), kIceUfrag3,
>-                                        kIcePwd3, ICEMODE_FULL,
>-                                        CONNECTIONROLE_ACTPASS, nullptr);
>-  EXPECT_TRUE(transport_controller_->SetLocalTransportDescription(
>-      "audio", ice_restart_desc, SdpType::kOffer, &err));
>-  EXPECT_EQ(ICEROLE_CONTROLLING, transport->fake_ice_transport()->GetIceRole());
>-}
>-
>-// Test that if the TransportController was created with the
>-// |redetermine_role_on_ice_restart| parameter set to false, the role is *not*
>-// redetermined on an ICE restart.
>-TEST_F(TransportControllerTest, IceRoleNotRedetermined) {
>-  bool redetermine_role = false;
>-  transport_controller_.reset(new TransportControllerForTest(redetermine_role));
>-  FakeDtlsTransport* transport = CreateFakeDtlsTransport("audio", 1);
>-  ASSERT_NE(nullptr, transport);
>-  std::string err;
>-  // Do an initial offer answer, so that the next offer is an ICE restart.
>-  transport_controller_->SetIceRole(ICEROLE_CONTROLLED);
>-  TransportDescription remote_desc(std::vector<std::string>(), kIceUfrag1,
>-                                   kIcePwd1, ICEMODE_FULL,
>-                                   CONNECTIONROLE_ACTPASS, nullptr);
>-  EXPECT_TRUE(transport_controller_->SetRemoteTransportDescription(
>-      "audio", remote_desc, SdpType::kOffer, &err));
>-  TransportDescription local_desc(std::vector<std::string>(), kIceUfrag2,
>-                                  kIcePwd2, ICEMODE_FULL,
>-                                  CONNECTIONROLE_ACTPASS, nullptr);
>-  EXPECT_TRUE(transport_controller_->SetLocalTransportDescription(
>-      "audio", local_desc, SdpType::kAnswer, &err));
>-  EXPECT_EQ(ICEROLE_CONTROLLED, transport->fake_ice_transport()->GetIceRole());
>-
>-  // The endpoint that initiated an ICE restart should keep the existing role.
>-  TransportDescription ice_restart_desc(std::vector<std::string>(), kIceUfrag3,
>-                                        kIcePwd3, ICEMODE_FULL,
>-                                        CONNECTIONROLE_ACTPASS, nullptr);
>-  EXPECT_TRUE(transport_controller_->SetLocalTransportDescription(
>-      "audio", ice_restart_desc, SdpType::kOffer, &err));
>-  EXPECT_EQ(ICEROLE_CONTROLLED, transport->fake_ice_transport()->GetIceRole());
>-}
>-
>-// Tests ICE role is reversed after receiving ice-lite from remote.
>-TEST_F(TransportControllerTest, TestSetRemoteIceLiteInOffer) {
>-  FakeDtlsTransport* transport = CreateFakeDtlsTransport("audio", 1);
>-  ASSERT_NE(nullptr, transport);
>-  std::string err;
>-
>-  transport_controller_->SetIceRole(ICEROLE_CONTROLLED);
>-  TransportDescription remote_desc(std::vector<std::string>(), kIceUfrag1,
>-                                   kIcePwd1, ICEMODE_LITE,
>-                                   CONNECTIONROLE_ACTPASS, nullptr);
>-  EXPECT_TRUE(transport_controller_->SetRemoteTransportDescription(
>-      "audio", remote_desc, SdpType::kOffer, &err));
>-  TransportDescription local_desc(kIceUfrag1, kIcePwd1);
>-  ASSERT_TRUE(transport_controller_->SetLocalTransportDescription(
>-      "audio", local_desc, SdpType::kAnswer, nullptr));
>-
>-  EXPECT_EQ(ICEROLE_CONTROLLING, transport->fake_ice_transport()->GetIceRole());
>-  EXPECT_EQ(ICEMODE_LITE, transport->fake_ice_transport()->remote_ice_mode());
>-}
>-
>-// Tests ice-lite in remote answer.
>-TEST_F(TransportControllerTest, TestSetRemoteIceLiteInAnswer) {
>-  FakeDtlsTransport* transport = CreateFakeDtlsTransport("audio", 1);
>-  ASSERT_NE(nullptr, transport);
>-  std::string err;
>-
>-  transport_controller_->SetIceRole(ICEROLE_CONTROLLING);
>-  TransportDescription local_desc(kIceUfrag1, kIcePwd1);
>-  ASSERT_TRUE(transport_controller_->SetLocalTransportDescription(
>-      "audio", local_desc, SdpType::kOffer, nullptr));
>-  EXPECT_EQ(ICEROLE_CONTROLLING, transport->fake_ice_transport()->GetIceRole());
>-  // Transports will be created in ICEFULL_MODE.
>-  EXPECT_EQ(ICEMODE_FULL, transport->fake_ice_transport()->remote_ice_mode());
>-  TransportDescription remote_desc(std::vector<std::string>(), kIceUfrag1,
>-                                   kIcePwd1, ICEMODE_LITE, CONNECTIONROLE_NONE,
>-                                   nullptr);
>-  ASSERT_TRUE(transport_controller_->SetRemoteTransportDescription(
>-      "audio", remote_desc, SdpType::kAnswer, nullptr));
>-  EXPECT_EQ(ICEROLE_CONTROLLING, transport->fake_ice_transport()->GetIceRole());
>-  // After receiving remote description with ICEMODE_LITE, transport should
>-  // have mode set to ICEMODE_LITE.
>-  EXPECT_EQ(ICEMODE_LITE, transport->fake_ice_transport()->remote_ice_mode());
>-}
>-
>-// Tests that the ICE role remains "controlling" if a subsequent offer that
>-// does an ICE restart is received from an ICE lite endpoint. Regression test
>-// for: https://crbug.com/710760
>-TEST_F(TransportControllerTest,
>-       IceRoleIsControllingAfterIceRestartFromIceLiteEndpoint) {
>-  FakeDtlsTransport* transport = CreateFakeDtlsTransport("audio", 1);
>-  ASSERT_NE(nullptr, transport);
>-  std::string err;
>-
>-  // Initial offer/answer.
>-  TransportDescription remote_desc(std::vector<std::string>(), kIceUfrag1,
>-                                   kIcePwd1, ICEMODE_LITE,
>-                                   CONNECTIONROLE_ACTPASS, nullptr);
>-  TransportDescription local_desc(kIceUfrag1, kIcePwd1);
>-  ASSERT_TRUE(transport_controller_->SetRemoteTransportDescription(
>-      "audio", remote_desc, SdpType::kOffer, &err));
>-  ASSERT_TRUE(transport_controller_->SetLocalTransportDescription(
>-      "audio", local_desc, SdpType::kAnswer, nullptr));
>-  // Subsequent ICE restart offer/answer.
>-  remote_desc.ice_ufrag = kIceUfrag2;
>-  remote_desc.ice_pwd = kIcePwd2;
>-  local_desc.ice_ufrag = kIceUfrag2;
>-  local_desc.ice_pwd = kIcePwd2;
>-  ASSERT_TRUE(transport_controller_->SetRemoteTransportDescription(
>-      "audio", remote_desc, SdpType::kOffer, &err));
>-  ASSERT_TRUE(transport_controller_->SetLocalTransportDescription(
>-      "audio", local_desc, SdpType::kAnswer, nullptr));
>-
>-  EXPECT_EQ(ICEROLE_CONTROLLING, transport->fake_ice_transport()->GetIceRole());
>-}
>-
>-// Tests SetNeedsIceRestartFlag and NeedsIceRestart, setting the flag and then
>-// initiating an ICE restart for one of the transports.
>-TEST_F(TransportControllerTest, NeedsIceRestart) {
>-  CreateFakeDtlsTransport("audio", 1);
>-  CreateFakeDtlsTransport("video", 1);
>-
>-  // Do initial offer/answer so there's something to restart.
>-  TransportDescription local_desc(kIceUfrag1, kIcePwd1);
>-  TransportDescription remote_desc(kIceUfrag1, kIcePwd1);
>-  ASSERT_TRUE(transport_controller_->SetLocalTransportDescription(
>-      "audio", local_desc, SdpType::kOffer, nullptr));
>-  ASSERT_TRUE(transport_controller_->SetLocalTransportDescription(
>-      "video", local_desc, SdpType::kOffer, nullptr));
>-  ASSERT_TRUE(transport_controller_->SetRemoteTransportDescription(
>-      "audio", remote_desc, SdpType::kAnswer, nullptr));
>-  ASSERT_TRUE(transport_controller_->SetRemoteTransportDescription(
>-      "video", remote_desc, SdpType::kAnswer, nullptr));
>-
>-  // Initially NeedsIceRestart should return false.
>-  EXPECT_FALSE(transport_controller_->NeedsIceRestart("audio"));
>-  EXPECT_FALSE(transport_controller_->NeedsIceRestart("video"));
>-
>-  // Set the needs-ice-restart flag and verify NeedsIceRestart starts returning
>-  // true.
>-  transport_controller_->SetNeedsIceRestartFlag();
>-  EXPECT_TRUE(transport_controller_->NeedsIceRestart("audio"));
>-  EXPECT_TRUE(transport_controller_->NeedsIceRestart("video"));
>-  // For a nonexistent transport, false should be returned.
>-  EXPECT_FALSE(transport_controller_->NeedsIceRestart("deadbeef"));
>-
>-  // Do ICE restart but only for audio.
>-  TransportDescription ice_restart_local_desc(kIceUfrag2, kIcePwd2);
>-  ASSERT_TRUE(transport_controller_->SetLocalTransportDescription(
>-      "audio", ice_restart_local_desc, SdpType::kOffer, nullptr));
>-  ASSERT_TRUE(transport_controller_->SetLocalTransportDescription(
>-      "video", local_desc, SdpType::kOffer, nullptr));
>-  // NeedsIceRestart should still be true for video.
>-  EXPECT_FALSE(transport_controller_->NeedsIceRestart("audio"));
>-  EXPECT_TRUE(transport_controller_->NeedsIceRestart("video"));
>-}
>-
>-enum class RTPTransportType { kSdes, kDtlsSrtp };
>-std::ostream& operator<<(std::ostream& out, RTPTransportType value) {
>-  switch (value) {
>-    case RTPTransportType::kSdes:
>-      return out << "SDES";
>-    case RTPTransportType::kDtlsSrtp:
>-      return out << "DTLS-SRTP";
>-  }
>-  return out;
>-}
>-
>-// Tests the TransportController can correctly create and destroy the RTP
>-// transports.
>-class TransportControllerRTPTransportTest
>-    : public TransportControllerTest,
>-      public ::testing::WithParamInterface<RTPTransportType> {
>- protected:
>-  // Helper function used to create an RTP layer transport.
>-  webrtc::RtpTransportInternal* CreateRtpTransport(
>-      const std::string& transport_name) {
>-    RTPTransportType type = GetParam();
>-    switch (type) {
>-      case RTPTransportType::kSdes:
>-        return transport_controller_->CreateSdesTransport(transport_name,
>-                                                          kRtcpMuxEnabled);
>-      case RTPTransportType::kDtlsSrtp:
>-        return transport_controller_->CreateDtlsSrtpTransport(transport_name,
>-                                                              kRtcpMuxEnabled);
>-    }
>-    return nullptr;
>-  }
>-};
>-
>-// Tests that creating transports with the same name will cause the
>-// second call to re-use the transport created in the first call.
>-TEST_P(TransportControllerRTPTransportTest, CreateTransportsWithReference) {
>-  const std::string transport_name = "transport";
>-  webrtc::RtpTransportInternal* transport1 = CreateRtpTransport(transport_name);
>-  webrtc::RtpTransportInternal* transport2 = CreateRtpTransport(transport_name);
>-  EXPECT_NE(nullptr, transport1);
>-  EXPECT_NE(nullptr, transport2);
>-  // The TransportController is expected to return the existing one when using
>-  // the same transport name.
>-  EXPECT_EQ(transport1, transport2);
>-  transport_controller_->DestroyTransport(transport_name);
>-  transport_controller_->DestroyTransport(transport_name);
>-}
>-
>-// Tests that creating different type of RTP transports with same name is not
>-// allowed.
>-TEST_P(TransportControllerRTPTransportTest,
>-       CreateDifferentTypeOfTransportsWithSameName) {
>-  const std::string transport_name = "transport";
>-  webrtc::RtpTransportInternal* transport1 = CreateRtpTransport(transport_name);
>-  EXPECT_NE(nullptr, transport1);
>-  RTPTransportType type = GetParam();
>-  switch (type) {
>-    case RTPTransportType::kSdes:
>-      EXPECT_EQ(nullptr, transport_controller_->CreateDtlsSrtpTransport(
>-                             transport_name, kRtcpMuxEnabled));
>-      break;
>-    case RTPTransportType::kDtlsSrtp:
>-      EXPECT_EQ(nullptr, transport_controller_->CreateSdesTransport(
>-                             transport_name, kRtcpMuxEnabled));
>-      break;
>-    default:
>-      ASSERT_TRUE(false);
>-  }
>-  transport_controller_->DestroyTransport(transport_name);
>-}
>-
>-// Tests the RTP transport is not actually destroyed if references still exist.
>-TEST_P(TransportControllerRTPTransportTest, DestroyTransportWithReference) {
>-  const std::string transport_name = "transport";
>-  webrtc::RtpTransportInternal* transport1 = CreateRtpTransport(transport_name);
>-  webrtc::RtpTransportInternal* transport2 = CreateRtpTransport(transport_name);
>-  EXPECT_NE(nullptr, transport1);
>-  EXPECT_NE(nullptr, transport2);
>-  transport_controller_->DestroyTransport(transport_name);
>-  EXPECT_NE(nullptr, transport1->rtp_packet_transport());
>-  EXPECT_EQ(nullptr, transport1->rtcp_packet_transport());
>-  transport_controller_->DestroyTransport(transport_name);
>-}
>-
>-// Tests the RTP is actually destroyed if there is no reference to it.
>-// Disabled because of sometimes not working on Windows.
>-// https://bugs.webrtc.org/34942
>-TEST_P(TransportControllerRTPTransportTest,
>-       DISABLED_DestroyTransportWithNoReference) {
>-  const std::string transport_name = "transport";
>-  webrtc::RtpTransportInternal* transport1 = CreateRtpTransport(transport_name);
>-  webrtc::RtpTransportInternal* transport2 = CreateRtpTransport(transport_name);
>-  EXPECT_NE(nullptr, transport1);
>-  EXPECT_NE(nullptr, transport2);
>-  transport_controller_->DestroyTransport(transport_name);
>-  transport_controller_->DestroyTransport(transport_name);
>-#if RTC_DCHECK_IS_ON && GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
>-  EXPECT_DEATH(transport1->IsWritable(false), /*error_message=*/"");
>-#endif
>-}
>-
>-INSTANTIATE_TEST_CASE_P(TransportControllerTest,
>-                        TransportControllerRTPTransportTest,
>-                        ::testing::Values(RTPTransportType::kSdes,
>-                                          RTPTransportType::kDtlsSrtp));
>-
>-}  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/transportstats.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/transportstats.cc
>new file mode 100644
>index 00000000000..df2df5892fe
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/transportstats.cc
>@@ -0,0 +1,25 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#include "pc/transportstats.h"
>+
>+namespace cricket {
>+
>+TransportChannelStats::TransportChannelStats() = default;
>+
>+TransportChannelStats::TransportChannelStats(const TransportChannelStats&) =
>+    default;
>+
>+TransportChannelStats::~TransportChannelStats() = default;
>+
>+TransportStats::TransportStats() = default;
>+
>+TransportStats::~TransportStats() = default;
>+
>+}  // namespace cricket
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/transportstats.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/transportstats.h
>new file mode 100644
>index 00000000000..cff608ade19
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/transportstats.h
>@@ -0,0 +1,50 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef PC_TRANSPORTSTATS_H_
>+#define PC_TRANSPORTSTATS_H_
>+
>+#include <string>
>+#include <vector>
>+
>+#include "p2p/base/dtlstransport.h"
>+#include "p2p/base/port.h"
>+
>+namespace cricket {
>+
>+struct TransportChannelStats {
>+  TransportChannelStats();
>+  TransportChannelStats(const TransportChannelStats&);
>+  ~TransportChannelStats();
>+
>+  int component = 0;
>+  CandidateStatsList candidate_stats_list;
>+  ConnectionInfos connection_infos;
>+  int srtp_crypto_suite = rtc::SRTP_INVALID_CRYPTO_SUITE;
>+  int ssl_cipher_suite = rtc::TLS_NULL_WITH_NULL_NULL;
>+  DtlsTransportState dtls_state = DTLS_TRANSPORT_NEW;
>+};
>+
>+// Information about all the channels of a transport.
>+// TODO(hta): Consider if a simple vector is as good as a map.
>+typedef std::vector<TransportChannelStats> TransportChannelStatsList;
>+
>+// Information about the stats of a transport.
>+struct TransportStats {
>+  TransportStats();
>+  ~TransportStats();
>+
>+  std::string transport_name;
>+  TransportChannelStatsList channel_stats;
>+};
>+
>+}  // namespace cricket
>+
>+#endif  // PC_TRANSPORTSTATS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/videocapturertracksource.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/videocapturertracksource.cc
>index bbfb3348bec..0767dcffffb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/videocapturertracksource.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/videocapturertracksource.cc
>@@ -242,7 +242,7 @@ const cricket::VideoFormat& GetBestCaptureFormat(
> // Return false if the key is mandatory, and the value is invalid.
> bool ExtractOption(const MediaConstraintsInterface* all_constraints,
>                    const std::string& key,
>-                   rtc::Optional<bool>* option) {
>+                   absl::optional<bool>* option) {
>   size_t mandatory = 0;
>   bool value;
>   if (FindConstraint(all_constraints, key, &value, &mandatory)) {
>@@ -288,7 +288,7 @@ VideoCapturerTrackSource::VideoCapturerTrackSource(
>     rtc::Thread* worker_thread,
>     std::unique_ptr<cricket::VideoCapturer> capturer,
>     bool remote)
>-    : VideoTrackSource(capturer.get(), remote),
>+    : VideoTrackSource(remote),
>       signaling_thread_(rtc::Thread::Current()),
>       worker_thread_(worker_thread),
>       video_capturer_(std::move(capturer)),
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/videocapturertracksource.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/videocapturertracksource.h
>index b80323f6b0c..0a254dba570 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/videocapturertracksource.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/videocapturertracksource.h
>@@ -18,7 +18,7 @@
> #include "media/base/videocommon.h"
> #include "pc/videotracksource.h"
> #include "rtc_base/asyncinvoker.h"
>-#include "rtc_base/sigslot.h"
>+#include "rtc_base/third_party/sigslot/sigslot.h"
> 
> // VideoCapturerTrackSource implements VideoTrackSourceInterface. It owns a
> // cricket::VideoCapturer and make sure the camera is started at a resolution
>@@ -48,7 +48,9 @@ class VideoCapturerTrackSource : public VideoTrackSource,
>       bool remote);
> 
>   bool is_screencast() const final { return video_capturer_->IsScreencast(); }
>-  rtc::Optional<bool> needs_denoising() const final { return needs_denoising_; }
>+  absl::optional<bool> needs_denoising() const final {
>+    return needs_denoising_;
>+  }
> 
>   bool GetStats(Stats* stats) final;
> 
>@@ -57,6 +59,11 @@ class VideoCapturerTrackSource : public VideoTrackSource,
>                            std::unique_ptr<cricket::VideoCapturer> capturer,
>                            bool remote);
>   virtual ~VideoCapturerTrackSource();
>+
>+  rtc::VideoSourceInterface<VideoFrame>* source() override {
>+    return video_capturer_.get();
>+  }
>+
>   void Initialize(const webrtc::MediaConstraintsInterface* constraints);
> 
>  private:
>@@ -71,7 +78,7 @@ class VideoCapturerTrackSource : public VideoTrackSource,
>   std::unique_ptr<cricket::VideoCapturer> video_capturer_;
>   bool started_;
>   cricket::VideoFormat format_;
>-  rtc::Optional<bool> needs_denoising_;
>+  absl::optional<bool> needs_denoising_;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/videocapturertracksource_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/videocapturertracksource_unittest.cc
>index 819a7224ce6..d2b8e1574ce 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/videocapturertracksource_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/videocapturertracksource_unittest.cc
>@@ -18,13 +18,18 @@
> #include "media/base/fakevideocapturer.h"
> #include "media/base/fakevideorenderer.h"
> #include "pc/videocapturertracksource.h"
>+#include "rtc_base/arraysize.h"
>+#include "rtc_base/event.h"
> #include "rtc_base/gunit.h"
>+#include "rtc_base/task_queue.h"
> 
>+using cricket::FOURCC_I420;
>+using cricket::VideoFormat;
> using webrtc::FakeConstraints;
>-using webrtc::VideoCapturerTrackSource;
> using webrtc::MediaConstraintsInterface;
> using webrtc::MediaSourceInterface;
> using webrtc::ObserverInterface;
>+using webrtc::VideoCapturerTrackSource;
> using webrtc::VideoTrackSourceInterface;
> 
> namespace {
>@@ -42,23 +47,13 @@ class TestVideoCapturer : public cricket::FakeVideoCapturer {
>  public:
>   explicit TestVideoCapturer(bool is_screencast)
>       : FakeVideoCapturer(is_screencast), test_without_formats_(false) {
>-    std::vector<cricket::VideoFormat> formats;
>-    formats.push_back(
>-        cricket::VideoFormat(1280, 720, cricket::VideoFormat::FpsToInterval(30),
>-                             cricket::FOURCC_I420));
>-    formats.push_back(
>-        cricket::VideoFormat(640, 480, cricket::VideoFormat::FpsToInterval(30),
>-                             cricket::FOURCC_I420));
>-    formats.push_back(
>-        cricket::VideoFormat(640, 400, cricket::VideoFormat::FpsToInterval(30),
>-                             cricket::FOURCC_I420));
>-    formats.push_back(
>-        cricket::VideoFormat(320, 240, cricket::VideoFormat::FpsToInterval(30),
>-                             cricket::FOURCC_I420));
>-    formats.push_back(
>-        cricket::VideoFormat(352, 288, cricket::VideoFormat::FpsToInterval(30),
>-                             cricket::FOURCC_I420));
>-    ResetSupportedFormats(formats);
>+    static const auto fps = VideoFormat::FpsToInterval(30);
>+    static const VideoFormat formats[] = {
>+        {1280, 720, fps, FOURCC_I420}, {640, 480, fps, FOURCC_I420},
>+        {640, 400, fps, FOURCC_I420},  {320, 240, fps, FOURCC_I420},
>+        {352, 288, fps, FOURCC_I420},
>+    };
>+    ResetSupportedFormats({&formats[0], &formats[arraysize(formats)]});
>   }
> 
>   // This function is used for resetting the supported capture formats and
>@@ -67,22 +62,21 @@ class TestVideoCapturer : public cricket::FakeVideoCapturer {
>   // Chrome implementation.
>   void TestWithoutCameraFormats() {
>     test_without_formats_ = true;
>-    std::vector<cricket::VideoFormat> formats;
>+    std::vector<VideoFormat> formats;
>     ResetSupportedFormats(formats);
>   }
> 
>-  virtual cricket::CaptureState Start(
>-      const cricket::VideoFormat& capture_format) {
>+  virtual cricket::CaptureState Start(const VideoFormat& capture_format) {
>     if (test_without_formats_) {
>-      std::vector<cricket::VideoFormat> formats;
>+      std::vector<VideoFormat> formats;
>       formats.push_back(capture_format);
>       ResetSupportedFormats(formats);
>     }
>     return FakeVideoCapturer::Start(capture_format);
>   }
> 
>-  virtual bool GetBestCaptureFormat(const cricket::VideoFormat& desired,
>-                                    cricket::VideoFormat* best_format) {
>+  virtual bool GetBestCaptureFormat(const VideoFormat& desired,
>+                                    VideoFormat* best_format) {
>     if (test_without_formats_) {
>       *best_format = desired;
>       return true;
>@@ -131,6 +125,16 @@ class VideoCapturerTrackSourceTest : public testing::Test {
>     source_->AddOrUpdateSink(&renderer_, rtc::VideoSinkWants());
>   }
> 
>+  void CaptureSingleFrame() {
>+    rtc::Event event(false, false);
>+    task_queue_.PostTask([this, &event]() {
>+      ASSERT_TRUE(capturer_->CaptureFrame());
>+      event.Set();
>+    });
>+    event.Wait(rtc::Event::kForever);
>+  }
>+
>+  rtc::TaskQueue task_queue_{"VideoCapturerTrackSourceTest"};
>   std::unique_ptr<cricket::VideoCapturer> capturer_cleanup_;
>   TestVideoCapturer* capturer_;
>   cricket::FakeVideoRenderer renderer_;
>@@ -147,7 +151,7 @@ TEST_F(VideoCapturerTrackSourceTest, CapturerStartStop) {
>   EXPECT_EQ_WAIT(MediaSourceInterface::kLive, state_observer_->state(),
>                  kMaxWaitMs);
> 
>-  ASSERT_TRUE(capturer_->CaptureFrame());
>+  CaptureSingleFrame();
>   EXPECT_EQ(1, renderer_.num_rendered_frames());
> 
>   capturer_->Stop();
>@@ -178,7 +182,7 @@ TEST_F(VideoCapturerTrackSourceTest, MandatoryConstraintCif5Fps) {
>   CreateVideoCapturerSource(&constraints);
>   EXPECT_EQ_WAIT(MediaSourceInterface::kLive, state_observer_->state(),
>                  kMaxWaitMs);
>-  const cricket::VideoFormat* format = capturer_->GetCaptureFormat();
>+  const VideoFormat* format = capturer_->GetCaptureFormat();
>   ASSERT_TRUE(format != NULL);
>   EXPECT_EQ(352, format->width);
>   EXPECT_EQ(288, format->height);
>@@ -198,7 +202,7 @@ TEST_F(VideoCapturerTrackSourceTest, MandatoryMinVgaOptional720P) {
>   CreateVideoCapturerSource(&constraints);
>   EXPECT_EQ_WAIT(MediaSourceInterface::kLive, state_observer_->state(),
>                  kMaxWaitMs);
>-  const cricket::VideoFormat* format = capturer_->GetCaptureFormat();
>+  const VideoFormat* format = capturer_->GetCaptureFormat();
>   ASSERT_TRUE(format != NULL);
>   EXPECT_EQ(1280, format->width);
>   EXPECT_EQ(720, format->height);
>@@ -219,7 +223,7 @@ TEST_F(VideoCapturerTrackSourceTest, MandatoryAspectRatio4To3) {
>   CreateVideoCapturerSource(&constraints);
>   EXPECT_EQ_WAIT(MediaSourceInterface::kLive, state_observer_->state(),
>                  kMaxWaitMs);
>-  const cricket::VideoFormat* format = capturer_->GetCaptureFormat();
>+  const VideoFormat* format = capturer_->GetCaptureFormat();
>   ASSERT_TRUE(format != NULL);
>   EXPECT_EQ(640, format->width);
>   EXPECT_EQ(480, format->height);
>@@ -244,7 +248,7 @@ TEST_F(VideoCapturerTrackSourceTest, OptionalAspectRatioTooHigh) {
>   CreateVideoCapturerSource(&constraints);
>   EXPECT_EQ_WAIT(MediaSourceInterface::kLive, state_observer_->state(),
>                  kMaxWaitMs);
>-  const cricket::VideoFormat* format = capturer_->GetCaptureFormat();
>+  const VideoFormat* format = capturer_->GetCaptureFormat();
>   ASSERT_TRUE(format != NULL);
>   double aspect_ratio = static_cast<double>(format->width) / format->height;
>   EXPECT_LT(aspect_ratio, 2);
>@@ -258,7 +262,7 @@ TEST_F(VideoCapturerTrackSourceTest, NoCameraCapability) {
>   CreateVideoCapturerSource();
>   EXPECT_EQ_WAIT(MediaSourceInterface::kLive, state_observer_->state(),
>                  kMaxWaitMs);
>-  const cricket::VideoFormat* format = capturer_->GetCaptureFormat();
>+  const VideoFormat* format = capturer_->GetCaptureFormat();
>   ASSERT_TRUE(format != NULL);
>   EXPECT_EQ(640, format->width);
>   EXPECT_EQ(480, format->height);
>@@ -280,7 +284,7 @@ TEST_F(VideoCapturerTrackSourceTest, NoCameraCapability16To9Ratio) {
>   CreateVideoCapturerSource(&constraints);
>   EXPECT_EQ_WAIT(MediaSourceInterface::kLive, state_observer_->state(),
>                  kMaxWaitMs);
>-  const cricket::VideoFormat* format = capturer_->GetCaptureFormat();
>+  const VideoFormat* format = capturer_->GetCaptureFormat();
>   double aspect_ratio = static_cast<double>(format->width) / format->height;
>   EXPECT_LE(requested_aspect_ratio, aspect_ratio);
> }
>@@ -318,7 +322,7 @@ TEST_F(VideoCapturerTrackSourceTest, SetValidDenoisingConstraint) {
> TEST_F(VideoCapturerTrackSourceTest, NoiseReductionConstraintNotSet) {
>   FakeConstraints constraints;
>   CreateVideoCapturerSource(&constraints);
>-  EXPECT_EQ(rtc::nullopt, source_->needs_denoising());
>+  EXPECT_EQ(absl::nullopt, source_->needs_denoising());
> }
> 
> TEST_F(VideoCapturerTrackSourceTest,
>@@ -353,7 +357,7 @@ TEST_F(VideoCapturerTrackSourceTest, NoiseReductionAndInvalidKeyMandatory) {
> 
>   EXPECT_EQ_WAIT(MediaSourceInterface::kEnded, state_observer_->state(),
>                  kMaxWaitMs);
>-  EXPECT_EQ(rtc::nullopt, source_->needs_denoising());
>+  EXPECT_EQ(absl::nullopt, source_->needs_denoising());
> }
> 
> TEST_F(VideoCapturerTrackSourceTest, InvalidDenoisingValueOptional) {
>@@ -366,12 +370,13 @@ TEST_F(VideoCapturerTrackSourceTest, InvalidDenoisingValueOptional) {
>   EXPECT_EQ_WAIT(MediaSourceInterface::kLive, state_observer_->state(),
>                  kMaxWaitMs);
> 
>-  EXPECT_EQ(rtc::nullopt, source_->needs_denoising());
>+  EXPECT_EQ(absl::nullopt, source_->needs_denoising());
> }
> 
> TEST_F(VideoCapturerTrackSourceTest, InvalidDenoisingValueMandatory) {
>   FakeConstraints constraints;
>-  // Optional constraints should be ignored if the mandatory constraints fail.
>+  // absl::optional constraints should be ignored if the mandatory constraints
>+  // fail.
>   constraints.AddOptional(MediaConstraintsInterface::kNoiseReduction, "false");
>   // Values are case-sensitive and must be all lower-case.
>   constraints.AddMandatory(MediaConstraintsInterface::kNoiseReduction, "True");
>@@ -380,7 +385,7 @@ TEST_F(VideoCapturerTrackSourceTest, InvalidDenoisingValueMandatory) {
> 
>   EXPECT_EQ_WAIT(MediaSourceInterface::kEnded, state_observer_->state(),
>                  kMaxWaitMs);
>-  EXPECT_EQ(rtc::nullopt, source_->needs_denoising());
>+  EXPECT_EQ(absl::nullopt, source_->needs_denoising());
> }
> 
> TEST_F(VideoCapturerTrackSourceTest, MixedOptionsAndConstraints) {
>@@ -395,7 +400,7 @@ TEST_F(VideoCapturerTrackSourceTest, MixedOptionsAndConstraints) {
>   CreateVideoCapturerSource(&constraints);
>   EXPECT_EQ_WAIT(MediaSourceInterface::kLive, state_observer_->state(),
>                  kMaxWaitMs);
>-  const cricket::VideoFormat* format = capturer_->GetCaptureFormat();
>+  const VideoFormat* format = capturer_->GetCaptureFormat();
>   ASSERT_TRUE(format != NULL);
>   EXPECT_EQ(352, format->width);
>   EXPECT_EQ(288, format->height);
>@@ -414,7 +419,7 @@ TEST_F(VideoCapturerTrackSourceTest, ScreencastResolutionNoConstraint) {
>   ASSERT_TRUE(source_->is_screencast());
>   EXPECT_EQ_WAIT(MediaSourceInterface::kLive, state_observer_->state(),
>                  kMaxWaitMs);
>-  const cricket::VideoFormat* format = capturer_->GetCaptureFormat();
>+  const VideoFormat* format = capturer_->GetCaptureFormat();
>   ASSERT_TRUE(format != NULL);
>   EXPECT_EQ(640, format->width);
>   EXPECT_EQ(480, format->height);
>@@ -435,7 +440,7 @@ TEST_F(VideoCapturerTrackSourceTest, ScreencastResolutionWithConstraint) {
>   ASSERT_TRUE(source_->is_screencast());
>   EXPECT_EQ_WAIT(MediaSourceInterface::kLive, state_observer_->state(),
>                  kMaxWaitMs);
>-  const cricket::VideoFormat* format = capturer_->GetCaptureFormat();
>+  const VideoFormat* format = capturer_->GetCaptureFormat();
>   ASSERT_TRUE(format != NULL);
>   EXPECT_EQ(480, format->width);
>   EXPECT_EQ(270, format->height);
>@@ -444,7 +449,7 @@ TEST_F(VideoCapturerTrackSourceTest, ScreencastResolutionWithConstraint) {
> 
> TEST_F(VideoCapturerTrackSourceTest, MandatorySubOneFpsConstraints) {
>   FakeConstraints constraints;
>-  constraints.AddMandatory(MediaConstraintsInterface::kMaxFrameRate, 0.5);
>+  constraints.AddMandatory(MediaConstraintsInterface::kMaxFrameRate, 0);
> 
>   CreateVideoCapturerSource(&constraints);
>   EXPECT_EQ_WAIT(MediaSourceInterface::kEnded, state_observer_->state(),
>@@ -454,12 +459,12 @@ TEST_F(VideoCapturerTrackSourceTest, MandatorySubOneFpsConstraints) {
> 
> TEST_F(VideoCapturerTrackSourceTest, OptionalSubOneFpsConstraints) {
>   FakeConstraints constraints;
>-  constraints.AddOptional(MediaConstraintsInterface::kMaxFrameRate, 0.5);
>+  constraints.AddOptional(MediaConstraintsInterface::kMaxFrameRate, 0);
> 
>   CreateVideoCapturerSource(&constraints);
>   EXPECT_EQ_WAIT(MediaSourceInterface::kLive, state_observer_->state(),
>                  kMaxWaitMs);
>-  const cricket::VideoFormat* format = capturer_->GetCaptureFormat();
>+  const VideoFormat* format = capturer_->GetCaptureFormat();
>   ASSERT_TRUE(format != NULL);
>   EXPECT_EQ(1, format->framerate());
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/videotrack_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/videotrack_unittest.cc
>index ceaac8c44fa..e833d6ddf01 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/videotrack_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/videotrack_unittest.cc
>@@ -25,22 +25,31 @@ using webrtc::VideoTrackSource;
> using webrtc::VideoTrack;
> using webrtc::VideoTrackInterface;
> 
>+class TestVideoTrackSource : public VideoTrackSource {
>+ public:
>+  TestVideoTrackSource() : VideoTrackSource(true /* remote */) {}
>+  rtc::VideoSourceInterface<webrtc::VideoFrame>* source() override {
>+    return &capturer_;
>+  }
>+  cricket::FakeVideoCapturerWithTaskQueue* capturer() { return &capturer_; }
>+
>+ private:
>+  cricket::FakeVideoCapturerWithTaskQueue capturer_;
>+};
> class VideoTrackTest : public testing::Test {
>  public:
>   VideoTrackTest() {
>     static const char kVideoTrackId[] = "track_id";
>-    video_track_source_ = new rtc::RefCountedObject<VideoTrackSource>(
>-        &capturer_, true /* remote */);
>+    video_track_source_ = new rtc::RefCountedObject<TestVideoTrackSource>();
>     video_track_ = VideoTrack::Create(kVideoTrackId, video_track_source_,
>                                       rtc::Thread::Current());
>-    capturer_.Start(
>+    video_track_source_->capturer()->Start(
>         cricket::VideoFormat(640, 480, cricket::VideoFormat::FpsToInterval(30),
>                              cricket::FOURCC_I420));
>   }
> 
>  protected:
>-  cricket::FakeVideoCapturer capturer_;
>-  rtc::scoped_refptr<VideoTrackSource> video_track_source_;
>+  rtc::scoped_refptr<TestVideoTrackSource> video_track_source_;
>   rtc::scoped_refptr<VideoTrackInterface> video_track_;
> };
> 
>@@ -58,18 +67,18 @@ TEST_F(VideoTrackTest, RenderVideo) {
>   std::unique_ptr<FakeVideoTrackRenderer> renderer_1(
>       new FakeVideoTrackRenderer(video_track_.get()));
> 
>-  capturer_.CaptureFrame();
>+  video_track_source_->capturer()->CaptureFrame();
>   EXPECT_EQ(1, renderer_1->num_rendered_frames());
> 
>   // FakeVideoTrackRenderer register itself to |video_track_|
>   std::unique_ptr<FakeVideoTrackRenderer> renderer_2(
>       new FakeVideoTrackRenderer(video_track_.get()));
>-  capturer_.CaptureFrame();
>+  video_track_source_->capturer()->CaptureFrame();
>   EXPECT_EQ(2, renderer_1->num_rendered_frames());
>   EXPECT_EQ(1, renderer_2->num_rendered_frames());
> 
>   renderer_1.reset(nullptr);
>-  capturer_.CaptureFrame();
>+  video_track_source_->capturer()->CaptureFrame();
>   EXPECT_EQ(2, renderer_2->num_rendered_frames());
> }
> 
>@@ -78,17 +87,17 @@ TEST_F(VideoTrackTest, DisableTrackBlackout) {
>   std::unique_ptr<FakeVideoTrackRenderer> renderer(
>       new FakeVideoTrackRenderer(video_track_.get()));
> 
>-  capturer_.CaptureFrame();
>+  video_track_source_->capturer()->CaptureFrame();
>   EXPECT_EQ(1, renderer->num_rendered_frames());
>   EXPECT_FALSE(renderer->black_frame());
> 
>   video_track_->set_enabled(false);
>-  capturer_.CaptureFrame();
>+  video_track_source_->capturer()->CaptureFrame();
>   EXPECT_EQ(2, renderer->num_rendered_frames());
>   EXPECT_TRUE(renderer->black_frame());
> 
>   video_track_->set_enabled(true);
>-  capturer_.CaptureFrame();
>+  video_track_source_->capturer()->CaptureFrame();
>   EXPECT_EQ(3, renderer->num_rendered_frames());
>   EXPECT_FALSE(renderer->black_frame());
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/videotracksource.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/videotracksource.cc
>index 56f48e7375c..bf4fd643a5a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/videotracksource.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/videotracksource.cc
>@@ -15,9 +15,8 @@
> namespace webrtc {
> 
> VideoTrackSource::VideoTrackSource(
>-    rtc::VideoSourceInterface<VideoFrame>* source,
>     bool remote)
>-    : source_(source), state_(kInitializing), remote_(remote) {
>+    : state_(kInitializing), remote_(remote) {
>   worker_thread_checker_.DetachFromThread();
> }
> 
>@@ -28,26 +27,16 @@ void VideoTrackSource::SetState(SourceState new_state) {
>   }
> }
> 
>-void VideoTrackSource::OnSourceDestroyed() {
>-  source_ = nullptr;
>-}
>-
> void VideoTrackSource::AddOrUpdateSink(
>     rtc::VideoSinkInterface<VideoFrame>* sink,
>     const rtc::VideoSinkWants& wants) {
>   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  if (!source_) {
>-    return;
>-  }
>-  source_->AddOrUpdateSink(sink, wants);
>+  source()->AddOrUpdateSink(sink, wants);
> }
> 
> void VideoTrackSource::RemoveSink(rtc::VideoSinkInterface<VideoFrame>* sink) {
>   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
>-  if (!source_) {
>-    return;
>-  }
>-  source_->RemoveSink(sink);
>+  source()->RemoveSink(sink);
> }
> 
> }  //  namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/videotracksource.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/videotracksource.h
>index d4731d029e3..50488dd2c60 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/videotracksource.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/videotracksource.h
>@@ -13,27 +13,26 @@
> 
> #include "api/mediastreaminterface.h"
> #include "api/notifier.h"
>-#include "api/videosinkinterface.h"
>+#include "api/video/video_sink_interface.h"
> #include "media/base/mediachannel.h"
> #include "rtc_base/thread_checker.h"
> 
>-// VideoTrackSource implements VideoTrackSourceInterface.
> namespace webrtc {
> 
>+// VideoTrackSource is a convenience base class for implementations of
>+// VideoTrackSourceInterface.
> class VideoTrackSource : public Notifier<VideoTrackSourceInterface> {
>  public:
>-  VideoTrackSource(rtc::VideoSourceInterface<VideoFrame>* source, bool remote);
>+  explicit VideoTrackSource(bool remote);
>   void SetState(SourceState new_state);
>-  // OnSourceDestroyed clears this instance pointer to |source_|. It is useful
>-  // when the underlying rtc::VideoSourceInterface is destroyed before the
>-  // reference counted VideoTrackSource.
>-  void OnSourceDestroyed();
> 
>   SourceState state() const override { return state_; }
>   bool remote() const override { return remote_; }
> 
>   bool is_screencast() const override { return false; }
>-  rtc::Optional<bool> needs_denoising() const override { return rtc::nullopt; }
>+  absl::optional<bool> needs_denoising() const override {
>+    return absl::nullopt;
>+  }
> 
>   bool GetStats(Stats* stats) override { return false; }
> 
>@@ -41,10 +40,11 @@ class VideoTrackSource : public Notifier<VideoTrackSourceInterface> {
>                        const rtc::VideoSinkWants& wants) override;
>   void RemoveSink(rtc::VideoSinkInterface<VideoFrame>* sink) override;
> 
>+ protected:
>+  virtual rtc::VideoSourceInterface<VideoFrame>* source() = 0;
>+
>  private:
>   rtc::ThreadChecker worker_thread_checker_;
>-  rtc::VideoSourceInterface<VideoFrame>* source_;
>-  cricket::VideoOptions options_;
>   SourceState state_;
>   const bool remote_;
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/webrtcsdp.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/webrtcsdp.cc
>index 56f4c991caa..73df354dd5d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/webrtcsdp.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/webrtcsdp.cc
>@@ -22,6 +22,7 @@
> #include <unordered_map>
> #include <vector>
> 
>+#include "api/mediatypes.h"
> #include "api/candidate.h"
> #include "api/cryptoparams.h"
> #include "api/jsepicecandidate.h"
>@@ -120,10 +121,12 @@ static const char kSsrcAttributeCname[] = "cname";
> static const char kAttributeExtmap[] = "extmap";
> // draft-alvestrand-mmusic-msid-01
> // a=msid-semantic: WMS
>+// This is a legacy field supported only for Plan B semantics.
> static const char kAttributeMsidSemantics[] = "msid-semantic";
> static const char kMediaStreamSemantic[] = "WMS";
> static const char kSsrcAttributeMsid[] = "msid";
> static const char kDefaultMsid[] = "default";
>+static const char kNoStreamMsid[] = "-";
> static const char kSsrcAttributeMslabel[] = "mslabel";
> static const char kSSrcAttributeLabel[] = "label";
> static const char kAttributeSsrcGroup[] = "ssrc-group";
>@@ -230,19 +233,19 @@ template <class T>
> static void AddFmtpLine(const T& codec, std::string* message);
> static void BuildMediaDescription(const ContentInfo* content_info,
>                                   const TransportInfo* transport_info,
>-                                  const MediaType media_type,
>+                                  const cricket::MediaType media_type,
>                                   const std::vector<Candidate>& candidates,
>-                                  bool unified_plan_sdp,
>+                                  int msid_signaling,
>                                   std::string* message);
> static void BuildSctpContentAttributes(std::string* message,
>                                        int sctp_port,
>                                        bool use_sctpmap);
> static void BuildRtpContentAttributes(const MediaContentDescription* media_desc,
>-                                      const MediaType media_type,
>-                                      bool unified_plan_sdp,
>+                                      const cricket::MediaType media_type,
>+                                      int msid_signaling,
>                                       std::string* message);
> static void BuildRtpMap(const MediaContentDescription* media_desc,
>-                        const MediaType media_type,
>+                        const cricket::MediaType media_type,
>                         std::string* message);
> static void BuildCandidate(const std::vector<Candidate>& candidates,
>                            bool include_ufrag,
>@@ -273,19 +276,21 @@ static bool ParseMediaDescription(
>     std::vector<JsepIceCandidate*>* candidates,
>     SdpParseError* error);
> static bool ParseContent(const std::string& message,
>-                         const MediaType media_type,
>+                         const cricket::MediaType media_type,
>                          int mline_index,
>                          const std::string& protocol,
>                          const std::vector<int>& payload_types,
>                          size_t* pos,
>                          std::string* content_name,
>                          bool* bundle_only,
>+                         int* msid_signaling,
>                          MediaContentDescription* media_desc,
>                          TransportDescription* transport,
>                          std::vector<JsepIceCandidate*>* candidates,
>                          SdpParseError* error);
> static bool ParseSsrcAttribute(const std::string& line,
>                                SsrcInfoVec* ssrc_infos,
>+                               int* msid_signaling,
>                                SdpParseError* error);
> static bool ParseSsrcGroupAttribute(const std::string& line,
>                                     SsrcGroupVec* ssrc_groups,
>@@ -294,20 +299,24 @@ static bool ParseCryptoAttribute(const std::string& line,
>                                  MediaContentDescription* media_desc,
>                                  SdpParseError* error);
> static bool ParseRtpmapAttribute(const std::string& line,
>-                                 const MediaType media_type,
>+                                 const cricket::MediaType media_type,
>                                  const std::vector<int>& payload_types,
>                                  MediaContentDescription* media_desc,
>                                  SdpParseError* error);
> static bool ParseFmtpAttributes(const std::string& line,
>-                                const MediaType media_type,
>+                                const cricket::MediaType media_type,
>                                 MediaContentDescription* media_desc,
>                                 SdpParseError* error);
>-static bool ParseFmtpParam(const std::string& line, std::string* parameter,
>-                           std::string* value, SdpParseError* error);
>-static bool ParseCandidate(const std::string& message, Candidate* candidate,
>-                           SdpParseError* error, bool is_raw);
>+static bool ParseFmtpParam(const std::string& line,
>+                           std::string* parameter,
>+                           std::string* value,
>+                           SdpParseError* error);
>+static bool ParseCandidate(const std::string& message,
>+                           Candidate* candidate,
>+                           SdpParseError* error,
>+                           bool is_raw);
> static bool ParseRtcpFbAttribute(const std::string& line,
>-                                 const MediaType media_type,
>+                                 const cricket::MediaType media_type,
>                                  MediaContentDescription* media_desc,
>                                  SdpParseError* error);
> static bool ParseIceOptions(const std::string& line,
>@@ -323,7 +332,7 @@ static bool ParseDtlsSetup(const std::string& line,
>                            cricket::ConnectionRole* role,
>                            SdpParseError* error);
> static bool ParseMsidAttribute(const std::string& line,
>-                               std::string* stream_id,
>+                               std::vector<std::string>* stream_ids,
>                                std::string* track_id,
>                                SdpParseError* error);
> 
>@@ -373,8 +382,7 @@ static bool ParseFailed(const std::string& line,
> 
> // Parses failure where the failing SDP line isn't know or there are multiple
> // failing lines.
>-static bool ParseFailed(const std::string& description,
>-                        SdpParseError* error) {
>+static bool ParseFailed(const std::string& description, SdpParseError* error) {
>   return ParseFailed("", description, error);
> }
> 
>@@ -453,10 +461,15 @@ static bool GetLine(const std::string& message,
>   // where <type> MUST be exactly one case-significant character and
>   // <value> is structured text whose format depends on <type>.
>   // Whitespace MUST NOT be used on either side of the "=" sign.
>-  if (line->length() < 3 ||
>-      !islower(cline[0]) ||
>+  //
>+  // However, an exception to the whitespace rule is made for "s=", since
>+  // RFC4566 also says:
>+  //
>+  //   If a session has no meaningful name, the value "s= " SHOULD be used
>+  //   (i.e., a single space as the session name).
>+  if (line->length() < 3 || !islower(cline[0]) ||
>       cline[1] != kSdpDelimiterEqual ||
>-      cline[2] == kSdpDelimiterSpace) {
>+      (cline[0] != kLineTypeSessionName && cline[2] == kSdpDelimiterSpace)) {
>     *pos = line_begin;
>     return false;
>   }
>@@ -477,7 +490,8 @@ static void InitAttrLine(const std::string& attribute, std::ostringstream* os) {
> }
> 
> // Writes a SDP attribute line based on |attribute| and |value| to |message|.
>-static void AddAttributeLine(const std::string& attribute, int value,
>+static void AddAttributeLine(const std::string& attribute,
>+                             int value,
>                              std::string* message) {
>   std::ostringstream os;
>   InitAttrLine(attribute, &os);
>@@ -496,13 +510,14 @@ static bool IsLineType(const std::string& message,
>           cmessage[line_start + 1] == kSdpDelimiterEqual);
> }
> 
>-static bool IsLineType(const std::string& line,
>-                       const char type) {
>+static bool IsLineType(const std::string& line, const char type) {
>   return IsLineType(line, type, 0);
> }
> 
>-static bool GetLineWithType(const std::string& message, size_t* pos,
>-                            std::string* line, const char type) {
>+static bool GetLineWithType(const std::string& message,
>+                            size_t* pos,
>+                            std::string* line,
>+                            const char type) {
>   if (!IsLineType(message, type, *pos)) {
>     return false;
>   }
>@@ -526,14 +541,16 @@ static bool AddSsrcLine(uint32_t ssrc_id,
>   // a=ssrc:<ssrc-id> <attribute>:<value>
>   std::ostringstream os;
>   InitAttrLine(kAttributeSsrc, &os);
>-  os << kSdpDelimiterColon << ssrc_id << kSdpDelimiterSpace
>-     << attribute << kSdpDelimiterColon << value;
>+  os << kSdpDelimiterColon << ssrc_id << kSdpDelimiterSpace << attribute
>+     << kSdpDelimiterColon << value;
>   return AddLine(os.str(), message);
> }
> 
> // Get value only from <attribute>:<value>.
>-static bool GetValue(const std::string& message, const std::string& attribute,
>-                     std::string* value, SdpParseError* error) {
>+static bool GetValue(const std::string& message,
>+                     const std::string& attribute,
>+                     std::string* value,
>+                     SdpParseError* error) {
>   std::string leftpart;
>   if (!rtc::tokenize_first(message, kSdpDelimiterColon, &leftpart, value)) {
>     return ParseFailedGetValue(message, attribute, error);
>@@ -548,10 +565,8 @@ static bool GetValue(const std::string& message, const std::string& attribute,
> }
> 
> static bool CaseInsensitiveFind(std::string str1, std::string str2) {
>-  std::transform(str1.begin(), str1.end(), str1.begin(),
>-                 ::tolower);
>-  std::transform(str2.begin(), str2.end(), str2.begin(),
>-                 ::tolower);
>+  std::transform(str1.begin(), str1.end(), str1.begin(), ::tolower);
>+  std::transform(str2.begin(), str2.end(), str2.begin(), ::tolower);
>   return str1.find(str2) != std::string::npos;
> }
> 
>@@ -573,46 +588,64 @@ static bool GetPayloadTypeFromString(const std::string& line,
>                                      int* payload_type,
>                                      SdpParseError* error) {
>   return GetValueFromString(line, s, payload_type, error) &&
>-      cricket::IsValidRtpPayloadType(*payload_type);
>+         cricket::IsValidRtpPayloadType(*payload_type);
>+}
>+
>+// Creates a StreamParams track in the case when no SSRC lines are signaled.
>+// This is a track that does not contain SSRCs and only contains
>+// stream_ids/track_id if it's signaled with a=msid lines.
>+void CreateTrackWithNoSsrcs(const std::vector<std::string>& msid_stream_ids,
>+                            const std::string& msid_track_id,
>+                            StreamParamsVec* tracks) {
>+  StreamParams track;
>+  if (msid_track_id.empty() || msid_stream_ids.empty()) {
>+    // We only create an unsignaled track if a=msid lines were signaled.
>+    return;
>+  }
>+  track.set_stream_ids(msid_stream_ids);
>+  track.id = msid_track_id;
>+  tracks->push_back(track);
> }
> 
>-// |msid_stream_id| and |msid_track_id| represent the stream/track ID from the
>+// Creates the StreamParams tracks, for the case when SSRC lines are signaled.
>+// |msid_stream_ids| and |msid_track_id| represent the stream/track ID from the
> // "a=msid" attribute, if it exists. They are empty if the attribute does not
>-// exist.
>+// exist. We prioritize getting stream_ids/track_ids signaled in a=msid lines.
> void CreateTracksFromSsrcInfos(const SsrcInfoVec& ssrc_infos,
>-                               const std::string& msid_stream_id,
>+                               const std::vector<std::string>& msid_stream_ids,
>                                const std::string& msid_track_id,
>-                               StreamParamsVec* tracks) {
>+                               StreamParamsVec* tracks,
>+                               int msid_signaling) {
>   RTC_DCHECK(tracks != NULL);
>-  RTC_DCHECK(msid_stream_id.empty() == msid_track_id.empty());
>   for (SsrcInfoVec::const_iterator ssrc_info = ssrc_infos.begin();
>        ssrc_info != ssrc_infos.end(); ++ssrc_info) {
>     if (ssrc_info->cname.empty()) {
>       continue;
>     }
> 
>-    std::string stream_id;
>+    std::vector<std::string> stream_ids;
>     std::string track_id;
>-    if (ssrc_info->stream_id.empty() && !ssrc_info->mslabel.empty()) {
>-      // If there's no msid and there's mslabel, we consider this is a sdp from
>-      // a older version of client that doesn't support msid.
>+    if (msid_signaling & cricket::kMsidSignalingMediaSection) {
>+      // This is the case with Unified Plan SDP msid signaling.
>+      stream_ids = msid_stream_ids;
>+      track_id = msid_track_id;
>+    } else if (msid_signaling & cricket::kMsidSignalingSsrcAttribute) {
>+      // This is the case with Plan B SDP msid signaling.
>+      stream_ids.push_back(ssrc_info->stream_id);
>+      track_id = ssrc_info->track_id;
>+    } else if (!ssrc_info->mslabel.empty()) {
>+      // Since there's no a=msid or a=ssrc msid signaling, this is a sdp from
>+      // an older version of client that doesn't support msid.
>       // In that case, we use the mslabel and label to construct the track.
>-      stream_id = ssrc_info->mslabel;
>+      stream_ids.push_back(ssrc_info->mslabel);
>       track_id = ssrc_info->label;
>-    } else if (ssrc_info->stream_id.empty() && !msid_stream_id.empty()) {
>-      // If there's no msid in the SSRC attributes, but there's a global one
>-      // (from a=msid), use that. This is the case with unified plan SDP.
>-      stream_id = msid_stream_id;
>-      track_id = msid_track_id;
>     } else {
>-      stream_id = ssrc_info->stream_id;
>-      track_id = ssrc_info->track_id;
>+      // Since no media streams isn't supported with older SDP signaling, we
>+      // use a default a stream id.
>+      stream_ids.push_back(kDefaultMsid);
>     }
>-    // If a stream/track ID wasn't populated from the SSRC attributes OR the
>+    // If a track ID wasn't populated from the SSRC attributes OR the
>     // msid attribute, use default/random values.
>-    if (stream_id.empty()) {
>-      stream_id = kDefaultMsid;
>-    }
>     if (track_id.empty()) {
>       // TODO(ronghuawu): What should we do if the track id doesn't appear?
>       // Create random string (which will be used as track label later)?
>@@ -632,16 +665,18 @@ void CreateTracksFromSsrcInfos(const SsrcInfoVec& ssrc_infos,
>     }
>     track->add_ssrc(ssrc_info->ssrc_id);
>     track->cname = ssrc_info->cname;
>-    track->sync_label = stream_id;
>+    track->set_stream_ids(stream_ids);
>     track->id = track_id;
>   }
> }
> 
>-void GetMediaStreamLabels(const ContentInfo* content,
>-                          std::set<std::string>* labels) {
>+void GetMediaStreamIds(const ContentInfo* content,
>+                       std::set<std::string>* labels) {
>   for (const StreamParams& stream_params :
>        content->media_description()->streams()) {
>-    labels->insert(stream_params.sync_label);
>+    for (const std::string& stream_id : stream_params.stream_ids()) {
>+      labels->insert(stream_id);
>+    }
>   }
> }
> 
>@@ -675,10 +710,11 @@ static int GetCandidatePreferenceFromType(const std::string& type) {
> // The value of |component_id| currently supported are 1 (RTP) and 2 (RTCP).
> // TODO(deadbeef): Decide the default destination in webrtcsession and
> // pass it down via SessionDescription.
>-static void GetDefaultDestination(
>-    const std::vector<Candidate>& candidates,
>-    int component_id, std::string* port,
>-    std::string* ip, std::string* addr_type) {
>+static void GetDefaultDestination(const std::vector<Candidate>& candidates,
>+                                  int component_id,
>+                                  std::string* port,
>+                                  std::string* ip,
>+                                  std::string* addr_type) {
>   *addr_type = kConnectionIpv4Addrtype;
>   *port = kDummyPort;
>   *ip = kDummyAddress;
>@@ -718,8 +754,8 @@ static void GetDefaultDestination(
> // Gets "a=rtcp" line if found default RTCP candidate from |candidates|.
> static std::string GetRtcpLine(const std::vector<Candidate>& candidates) {
>   std::string rtcp_line, rtcp_port, rtcp_ip, addr_type;
>-  GetDefaultDestination(candidates, ICE_CANDIDATE_COMPONENT_RTCP,
>-                        &rtcp_port, &rtcp_ip, &addr_type);
>+  GetDefaultDestination(candidates, ICE_CANDIDATE_COMPONENT_RTCP, &rtcp_port,
>+                        &rtcp_ip, &addr_type);
>   // Found default RTCP candidate.
>   // RFC 5245
>   // If the agent is utilizing RTCP, it MUST encode the RTCP candidate
>@@ -730,11 +766,8 @@ static std::string GetRtcpLine(const std::vector<Candidate>& candidates) {
>   // connection-address] CRLF
>   std::ostringstream os;
>   InitAttrLine(kAttributeRtcp, &os);
>-  os << kSdpDelimiterColon
>-     << rtcp_port << " "
>-     << kConnectionNettype << " "
>-     << addr_type << " "
>-     << rtcp_ip;
>+  os << kSdpDelimiterColon << rtcp_port << " " << kConnectionNettype << " "
>+     << addr_type << " " << rtcp_ip;
>   rtcp_line = os.str();
>   return rtcp_line;
> }
>@@ -757,8 +790,7 @@ static bool IsValidPort(int port) {
>   return port >= 0 && port <= 65535;
> }
> 
>-std::string SdpSerialize(const JsepSessionDescription& jdesc,
>-                         bool unified_plan_sdp) {
>+std::string SdpSerialize(const JsepSessionDescription& jdesc) {
>   const cricket::SessionDescription* desc = jdesc.description();
>   if (!desc) {
>     return "";
>@@ -774,10 +806,11 @@ std::string SdpSerialize(const JsepSessionDescription& jdesc,
>   // <unicast-address>
>   std::ostringstream os;
>   InitLine(kLineTypeOrigin, kSessionOriginUsername, &os);
>-  const std::string& session_id = jdesc.session_id().empty() ?
>-      kSessionOriginSessionId : jdesc.session_id();
>-  const std::string& session_version = jdesc.session_version().empty() ?
>-      kSessionOriginSessionVersion : jdesc.session_version();
>+  const std::string& session_id =
>+      jdesc.session_id().empty() ? kSessionOriginSessionId : jdesc.session_id();
>+  const std::string& session_version = jdesc.session_version().empty()
>+                                           ? kSessionOriginSessionVersion
>+                                           : jdesc.session_version();
>   os << " " << session_id << " " << session_version << " "
>      << kSessionOriginNettype << " " << kSessionOriginAddrtype << " "
>      << kSessionOriginAddress;
>@@ -806,21 +839,34 @@ std::string SdpSerialize(const JsepSessionDescription& jdesc,
>   InitAttrLine(kAttributeMsidSemantics, &os);
>   os << kSdpDelimiterColon << " " << kMediaStreamSemantic;
> 
>-  std::set<std::string> media_stream_labels;
>+  std::set<std::string> media_stream_ids;
>   const ContentInfo* audio_content = GetFirstAudioContent(desc);
>   if (audio_content)
>-    GetMediaStreamLabels(audio_content, &media_stream_labels);
>+    GetMediaStreamIds(audio_content, &media_stream_ids);
> 
>   const ContentInfo* video_content = GetFirstVideoContent(desc);
>   if (video_content)
>-    GetMediaStreamLabels(video_content, &media_stream_labels);
>+    GetMediaStreamIds(video_content, &media_stream_ids);
> 
>-  for (std::set<std::string>::const_iterator it =
>-      media_stream_labels.begin(); it != media_stream_labels.end(); ++it) {
>+  for (std::set<std::string>::const_iterator it = media_stream_ids.begin();
>+       it != media_stream_ids.end(); ++it) {
>     os << " " << *it;
>   }
>   AddLine(os.str(), &message);
> 
>+  // a=ice-lite
>+  //
>+  // TODO(deadbeef): It's weird that we need to iterate TransportInfos for
>+  // this, when it's a session-level attribute. It really should be moved to a
>+  // session-level structure like SessionDescription.
>+  for (const cricket::TransportInfo& transport : desc->transport_infos()) {
>+    if (transport.description.ice_mode == cricket::ICEMODE_LITE) {
>+      InitAttrLine(kAttributeIceLite, &os);
>+      AddLine(os.str(), &message);
>+      break;
>+    }
>+  }
>+
>   // Preserve the order of the media contents.
>   int mline_index = -1;
>   for (cricket::ContentInfos::const_iterator it = desc->contents().begin();
>@@ -829,7 +875,7 @@ std::string SdpSerialize(const JsepSessionDescription& jdesc,
>     std::vector<Candidate> candidates;
>     GetCandidatesByMindex(jdesc, ++mline_index, &candidates);
>     BuildMediaDescription(&*it, desc->GetTransportInfoByName(it->name),
>-                          mdesc->type(), candidates, unified_plan_sdp,
>+                          mdesc->type(), candidates, desc->msid_signaling(),
>                           &message);
>   }
>   return message;
>@@ -880,8 +926,8 @@ bool SdpDeserialize(const std::string& message,
>                              session_connection_addr, desc, &candidates,
>                              error)) {
>     delete desc;
>-    for (std::vector<JsepIceCandidate*>::const_iterator
>-         it = candidates.begin(); it != candidates.end(); ++it) {
>+    for (std::vector<JsepIceCandidate*>::const_iterator it = candidates.begin();
>+         it != candidates.end(); ++it) {
>       delete *it;
>     }
>     return false;
>@@ -889,8 +935,8 @@ bool SdpDeserialize(const std::string& message,
> 
>   jdesc->Initialize(desc, session_id, session_version);
> 
>-  for (std::vector<JsepIceCandidate*>::const_iterator
>-       it = candidates.begin(); it != candidates.end(); ++it) {
>+  for (std::vector<JsepIceCandidate*>::const_iterator it = candidates.begin();
>+       it != candidates.end(); ++it) {
>     jdesc->AddCandidate(*it);
>     delete *it;
>   }
>@@ -921,8 +967,10 @@ bool SdpDeserializeCandidate(const std::string& transport_name,
>   return true;
> }
> 
>-bool ParseCandidate(const std::string& message, Candidate* candidate,
>-                    SdpParseError* error, bool is_raw) {
>+bool ParseCandidate(const std::string& message,
>+                    Candidate* candidate,
>+                    SdpParseError* error,
>+                    bool is_raw) {
>   RTC_DCHECK(candidate != NULL);
> 
>   // Get the first line from |message|.
>@@ -956,8 +1004,8 @@ bool ParseCandidate(const std::string& message, Candidate* candidate,
>       attribute_candidate != kAttributeCandidate) {
>     if (is_raw) {
>       std::ostringstream description;
>-      description << "Expect line: " << kAttributeCandidate
>-                  << ":" << "<candidate-str>";
>+      description << "Expect line: " << kAttributeCandidate << ":"
>+                  << "<candidate-str>";
>       return ParseFailed(first_line, 0, description.str(), error);
>     } else {
>       return ParseFailedExpectLine(first_line, 0, kLineTypeAttributes,
>@@ -1039,8 +1087,8 @@ bool ParseCandidate(const std::string& message, Candidate* candidate,
>   if (fields.size() >= (current_position + 2) &&
>       fields[current_position] == kAttributeCandidateRport) {
>     int port = 0;
>-    if (!GetValueFromString(
>-        first_line, fields[++current_position], &port, error)) {
>+    if (!GetValueFromString(first_line, fields[++current_position], &port,
>+                            error)) {
>       return false;
>     }
>     if (!IsValidPort(port)) {
>@@ -1154,8 +1202,7 @@ bool ParseExtmap(const std::string& line,
>   // RFC 5285
>   // a=extmap:<value>["/"<direction>] <URI> <extensionattributes>
>   std::vector<std::string> fields;
>-  rtc::split(line.substr(kLinePrefixLength),
>-                   kSdpDelimiterSpace, &fields);
>+  rtc::split(line.substr(kLinePrefixLength), kSdpDelimiterSpace, &fields);
>   const size_t expected_min_fields = 2;
>   if (fields.size() < expected_min_fields) {
>     return ParseFailedExpectMinFieldNum(line, expected_min_fields, error);
>@@ -1181,7 +1228,7 @@ bool ParseExtmap(const std::string& line,
>     const size_t expected_min_fields_encrypted = expected_min_fields + 1;
>     if (fields.size() < expected_min_fields_encrypted) {
>       return ParseFailedExpectMinFieldNum(line, expected_min_fields_encrypted,
>-          error);
>+                                          error);
>     }
> 
>     encrypted = true;
>@@ -1197,9 +1244,9 @@ bool ParseExtmap(const std::string& line,
> 
> void BuildMediaDescription(const ContentInfo* content_info,
>                            const TransportInfo* transport_info,
>-                           const MediaType media_type,
>+                           const cricket::MediaType media_type,
>                            const std::vector<Candidate>& candidates,
>-                           bool unified_plan_sdp,
>+                           int msid_signaling,
>                            std::string* message) {
>   RTC_DCHECK(message != NULL);
>   if (content_info == NULL || message == NULL) {
>@@ -1235,7 +1282,7 @@ void BuildMediaDescription(const ContentInfo* content_info,
>              video_desc->codecs().begin();
>          it != video_desc->codecs().end(); ++it) {
>       fmt.append(" ");
>-      fmt.append(rtc::ToString<int>(it->id));
>+      fmt.append(rtc::ToString(it->id));
>     }
>   } else if (media_type == cricket::MEDIA_TYPE_AUDIO) {
>     const AudioContentDescription* audio_desc = media_desc->as_audio();
>@@ -1243,7 +1290,7 @@ void BuildMediaDescription(const ContentInfo* content_info,
>              audio_desc->codecs().begin();
>          it != audio_desc->codecs().end(); ++it) {
>       fmt.append(" ");
>-      fmt.append(rtc::ToString<int>(it->id));
>+      fmt.append(rtc::ToString(it->id));
>     }
>   } else if (media_type == cricket::MEDIA_TYPE_DATA) {
>     const DataContentDescription* data_desc = media_desc->as_data();
>@@ -1261,16 +1308,16 @@ void BuildMediaDescription(const ContentInfo* content_info,
>           }
>         }
> 
>-        fmt.append(rtc::ToString<int>(sctp_port));
>+        fmt.append(rtc::ToString(sctp_port));
>       } else {
>         fmt.append(kDefaultSctpmapProtocol);
>       }
>     } else {
>       for (std::vector<cricket::DataCodec>::const_iterator it =
>-           data_desc->codecs().begin();
>+               data_desc->codecs().begin();
>            it != data_desc->codecs().end(); ++it) {
>         fmt.append(" ");
>-        fmt.append(rtc::ToString<int>(it->id));
>+        fmt.append(rtc::ToString(it->id));
>       }
>     }
>   }
>@@ -1297,8 +1344,9 @@ void BuildMediaDescription(const ContentInfo* content_info,
>     port = rtc::ToString(media_desc->connection_address().port());
>   }
> 
>-  rtc::SSLFingerprint* fp = (transport_info) ?
>-      transport_info->description.identity_fingerprint.get() : NULL;
>+  rtc::SSLFingerprint* fp =
>+      (transport_info) ? transport_info->description.identity_fingerprint.get()
>+                       : NULL;
> 
>   // Add the m and c lines.
>   InitLine(kLineTypeMedia, type, &os);
>@@ -1370,14 +1418,13 @@ void BuildMediaDescription(const ContentInfo* content_info,
>     if (fp) {
>       // Insert the fingerprint attribute.
>       InitAttrLine(kAttributeFingerprint, &os);
>-      os << kSdpDelimiterColon
>-         << fp->algorithm << kSdpDelimiterSpace
>+      os << kSdpDelimiterColon << fp->algorithm << kSdpDelimiterSpace
>          << fp->GetRfc4572Fingerprint();
>       AddLine(os.str(), message);
> 
>       // Inserting setup attribute.
>       if (transport_info->description.connection_role !=
>-              cricket::CONNECTIONROLE_NONE) {
>+          cricket::CONNECTIONROLE_NONE) {
>         // Making sure we are not using "passive" mode.
>         cricket::ConnectionRole role =
>             transport_info->description.connection_role;
>@@ -1405,8 +1452,7 @@ void BuildMediaDescription(const ContentInfo* content_info,
>     bool use_sctpmap = data_desc->use_sctpmap();
>     BuildSctpContentAttributes(message, sctp_port, use_sctpmap);
>   } else if (IsRtp(media_desc->protocol())) {
>-    BuildRtpContentAttributes(media_desc, media_type, unified_plan_sdp,
>-                              message);
>+    BuildRtpContentAttributes(media_desc, media_type, msid_signaling, message);
>   }
> }
> 
>@@ -1431,10 +1477,9 @@ void BuildSctpContentAttributes(std::string* message,
>   AddLine(os.str(), message);
> }
> 
>-// If unified_plan_sdp is true, will use "a=msid".
> void BuildRtpContentAttributes(const MediaContentDescription* media_desc,
>-                               const MediaType media_type,
>-                               bool unified_plan_sdp,
>+                               const cricket::MediaType media_type,
>+                               int msid_signaling,
>                                std::string* message) {
>   std::ostringstream os;
>   // RFC 5285
>@@ -1471,19 +1516,30 @@ void BuildRtpContentAttributes(const MediaContentDescription* media_desc,
>   }
>   AddLine(os.str(), message);
> 
>-  // draft-ietf-mmusic-msid-11
>-  // a=msid:<stream id> <track id>
>-  if (unified_plan_sdp && !media_desc->streams().empty()) {
>-    if (media_desc->streams().size() > 1u) {
>+  // Specified in https://datatracker.ietf.org/doc/draft-ietf-mmusic-msid/16/
>+  // a=msid:<msid-id> <msid-appdata>
>+  // The msid-id is a 1*64 token char representing the media stream id, and the
>+  // msid-appdata is a 1*64 token char representing the track id. There is a
>+  // line for every media stream, with a special msid-id value of "-"
>+  // representing no streams. The value of "msid-appdata" MUST be identical for
>+  // all lines.
>+  if (msid_signaling & cricket::kMsidSignalingMediaSection) {
>+    const StreamParamsVec& streams = media_desc->streams();
>+    if (streams.size() == 1u) {
>+      const StreamParams& track = streams[0];
>+      std::vector<std::string> stream_ids = track.stream_ids();
>+      if (stream_ids.empty()) {
>+        stream_ids.push_back(kNoStreamMsid);
>+      }
>+      for (const std::string& stream_id : stream_ids) {
>+        InitAttrLine(kAttributeMsid, &os);
>+        os << kSdpDelimiterColon << stream_id << kSdpDelimiterSpace << track.id;
>+        AddLine(os.str(), message);
>+      }
>+    } else if (streams.size() > 1u) {
>       RTC_LOG(LS_WARNING)
>-          << "Trying to serialize unified plan SDP with more than "
>-          << "one track in a media section. Omitting 'a=msid'.";
>-    } else {
>-      auto track = media_desc->streams().begin();
>-      const std::string& stream_id = track->sync_label;
>-      InitAttrLine(kAttributeMsid, &os);
>-      os << kSdpDelimiterColon << stream_id << kSdpDelimiterSpace << track->id;
>-      AddLine(os.str(), message);
>+          << "Trying to serialize Unified Plan SDP with more than "
>+             "one track in a media section. Omitting 'a=msid'.";
>     }
>   }
> 
>@@ -1528,12 +1584,6 @@ void BuildRtpContentAttributes(const MediaContentDescription* media_desc,
> 
>   for (StreamParamsVec::const_iterator track = media_desc->streams().begin();
>        track != media_desc->streams().end(); ++track) {
>-    // Require that the track belongs to a media stream,
>-    // ie the sync_label is set. This extra check is necessary since the
>-    // MediaContentDescription always contains a streamparam with an ssrc even
>-    // if no track or media stream have been created.
>-    if (track->sync_label.empty()) continue;
>-
>     // Build the ssrc-group lines.
>     for (size_t i = 0; i < track->ssrc_groups.size(); ++i) {
>       // RFC 5576
>@@ -1546,7 +1596,7 @@ void BuildRtpContentAttributes(const MediaContentDescription* media_desc,
>       std::vector<uint32_t>::const_iterator ssrc =
>           track->ssrc_groups[i].ssrcs.begin();
>       for (; ssrc != track->ssrc_groups[i].ssrcs.end(); ++ssrc) {
>-        os << kSdpDelimiterSpace << rtc::ToString<uint32_t>(*ssrc);
>+        os << kSdpDelimiterSpace << rtc::ToString(*ssrc);
>       }
>       AddLine(os.str(), message);
>     }
>@@ -1555,28 +1605,32 @@ void BuildRtpContentAttributes(const MediaContentDescription* media_desc,
>       uint32_t ssrc = track->ssrcs[i];
>       // RFC 5576
>       // a=ssrc:<ssrc-id> cname:<value>
>-      AddSsrcLine(ssrc, kSsrcAttributeCname,
>-                  track->cname, message);
>-
>-      // draft-alvestrand-mmusic-msid-00
>-      // a=ssrc:<ssrc-id> msid:identifier [appdata]
>-      // The appdata consists of the "id" attribute of a MediaStreamTrack,
>-      // which corresponds to the "id" attribute of StreamParams.
>-      const std::string& stream_id = track->sync_label;
>-      InitAttrLine(kAttributeSsrc, &os);
>-      os << kSdpDelimiterColon << ssrc << kSdpDelimiterSpace
>-         << kSsrcAttributeMsid << kSdpDelimiterColon << stream_id
>-         << kSdpDelimiterSpace << track->id;
>-      AddLine(os.str(), message);
>+      AddSsrcLine(ssrc, kSsrcAttributeCname, track->cname, message);
>+
>+      if (msid_signaling & cricket::kMsidSignalingSsrcAttribute) {
>+        // draft-alvestrand-mmusic-msid-00
>+        // a=ssrc:<ssrc-id> msid:identifier [appdata]
>+        // The appdata consists of the "id" attribute of a MediaStreamTrack,
>+        // which corresponds to the "id" attribute of StreamParams.
>+        // Since a=ssrc msid signaling is used in Plan B SDP semantics, and
>+        // multiple stream ids are not supported for Plan B, we are only adding
>+        // a line for the first media stream id here.
>+        const std::string& stream_id = track->first_stream_id();
>+        InitAttrLine(kAttributeSsrc, &os);
>+        os << kSdpDelimiterColon << ssrc << kSdpDelimiterSpace
>+           << kSsrcAttributeMsid << kSdpDelimiterColon << stream_id
>+           << kSdpDelimiterSpace << track->id;
>+        AddLine(os.str(), message);
> 
>-      // TODO(ronghuawu): Remove below code which is for backward
>-      // compatibility.
>-      // draft-alvestrand-rtcweb-mid-01
>-      // a=ssrc:<ssrc-id> mslabel:<value>
>-      // The label isn't yet defined.
>-      // a=ssrc:<ssrc-id> label:<value>
>-      AddSsrcLine(ssrc, kSsrcAttributeMslabel, track->sync_label, message);
>-      AddSsrcLine(ssrc, kSSrcAttributeLabel, track->id, message);
>+        // TODO(ronghuawu): Remove below code which is for backward
>+        // compatibility.
>+        // draft-alvestrand-rtcweb-mid-01
>+        // a=ssrc:<ssrc-id> mslabel:<value>
>+        // The label isn't yet defined.
>+        // a=ssrc:<ssrc-id> label:<value>
>+        AddSsrcLine(ssrc, kSsrcAttributeMslabel, stream_id, message);
>+        AddSsrcLine(ssrc, kSSrcAttributeLabel, track->id, message);
>+      }
>     }
>   }
> }
>@@ -1675,13 +1729,10 @@ void AddRtcpFbLines(const T& codec, std::string* message) {
>   }
> }
> 
>-bool AddSctpDataCodec(DataContentDescription* media_desc,
>-                      int sctp_port) {
>+bool AddSctpDataCodec(DataContentDescription* media_desc, int sctp_port) {
>   for (const auto& codec : media_desc->codecs()) {
>     if (cricket::CodecNamesEq(codec.name, cricket::kGoogleSctpDataCodecName)) {
>-      return ParseFailed("",
>-                         "Can't have multiple sctp port attributes.",
>-                         NULL);
>+      return ParseFailed("", "Can't have multiple sctp port attributes.", NULL);
>     }
>   }
>   // Add the SCTP Port number as a pseudo-codec "port" parameter
>@@ -1704,9 +1755,9 @@ bool GetMinValue(const std::vector<int>& values, int* value) {
> }
> 
> bool GetParameter(const std::string& name,
>-                  const cricket::CodecParameterMap& params, int* value) {
>-  std::map<std::string, std::string>::const_iterator found =
>-      params.find(name);
>+                  const cricket::CodecParameterMap& params,
>+                  int* value) {
>+  std::map<std::string, std::string>::const_iterator found = params.find(name);
>   if (found == params.end()) {
>     return false;
>   }
>@@ -1717,7 +1768,7 @@ bool GetParameter(const std::string& name,
> }
> 
> void BuildRtpMap(const MediaContentDescription* media_desc,
>-                 const MediaType media_type,
>+                 const cricket::MediaType media_type,
>                  std::string* message) {
>   RTC_DCHECK(message != NULL);
>   RTC_DCHECK(media_desc != NULL);
>@@ -1791,14 +1842,14 @@ void BuildRtpMap(const MediaContentDescription* media_desc,
>   } else if (media_type == cricket::MEDIA_TYPE_DATA) {
>     const DataContentDescription* data_desc = media_desc->as_data();
>     for (std::vector<cricket::DataCodec>::const_iterator it =
>-         data_desc->codecs().begin();
>+             data_desc->codecs().begin();
>          it != data_desc->codecs().end(); ++it) {
>       // RFC 4566
>       // a=rtpmap:<payload type> <encoding name>/<clock rate>
>       // [/<encodingparameters>]
>       InitAttrLine(kAttributeRtpmap, &os);
>-      os << kSdpDelimiterColon << it->id << " "
>-         << it->name << "/" << it->clockrate;
>+      os << kSdpDelimiterColon << it->id << " " << it->name << "/"
>+         << it->clockrate;
>       AddLine(os.str(), message);
>     }
>   }
>@@ -1834,15 +1885,12 @@ void BuildCandidate(const std::vector<Candidate>& candidates,
>     }
> 
>     InitAttrLine(kAttributeCandidate, &os);
>-    os << kSdpDelimiterColon
>-       << it->foundation() << " "
>-       << it->component() << " "
>-       << it->protocol() << " "
>-       << it->priority() << " "
>-       << it->address().ipaddr().ToString() << " "
>-       << it->address().PortAsString() << " "
>-       << kAttributeCandidateTyp << " "
>-       << type << " ";
>+    os << kSdpDelimiterColon << it->foundation() << " " << it->component()
>+       << " " << it->protocol() << " " << it->priority() << " "
>+       << (it->address().ipaddr().IsNil() ? it->address().hostname()
>+                                          : it->address().ipaddr().ToString())
>+       << " " << it->address().PortAsString() << " " << kAttributeCandidateTyp
>+       << " " << type << " ";
> 
>     // Related address
>     if (!it->related_address().IsNil()) {
>@@ -1887,7 +1935,7 @@ void BuildIceOptions(const std::vector<std::string>& transport_options,
> 
> bool IsRtp(const std::string& protocol) {
>   return protocol.empty() ||
>-      (protocol.find(cricket::kMediaProtocolRtpPrefix) != std::string::npos);
>+         (protocol.find(cricket::kMediaProtocolRtpPrefix) != std::string::npos);
> }
> 
> bool IsDtlsSctp(const std::string& protocol) {
>@@ -1960,19 +2008,18 @@ bool ParseSessionDescription(const std::string& message,
>   // RFC 4566
>   // v=  (protocol version)
>   if (!GetLineWithType(message, pos, &line, kLineTypeVersion)) {
>-    return ParseFailedExpectLine(message, *pos, kLineTypeVersion,
>-                                 std::string(), error);
>+    return ParseFailedExpectLine(message, *pos, kLineTypeVersion, std::string(),
>+                                 error);
>   }
>   // RFC 4566
>   // o=<username> <sess-id> <sess-version> <nettype> <addrtype>
>   // <unicast-address>
>   if (!GetLineWithType(message, pos, &line, kLineTypeOrigin)) {
>-    return ParseFailedExpectLine(message, *pos, kLineTypeOrigin,
>-                                 std::string(), error);
>+    return ParseFailedExpectLine(message, *pos, kLineTypeOrigin, std::string(),
>+                                 error);
>   }
>   std::vector<std::string> fields;
>-  rtc::split(line.substr(kLinePrefixLength),
>-                   kSdpDelimiterSpace, &fields);
>+  rtc::split(line.substr(kLinePrefixLength), kSdpDelimiterSpace, &fields);
>   const size_t expected_fields = 6;
>   if (fields.size() != expected_fields) {
>     return ParseFailedExpectFieldNum(line, expected_fields, error);
>@@ -1987,7 +2034,7 @@ bool ParseSessionDescription(const std::string& message,
>                                  std::string(), error);
>   }
> 
>-  // Optional lines
>+  // absl::optional lines
>   // Those are the optional lines, so shouldn't return false if not present.
>   // RFC 4566
>   // i=* (session information)
>@@ -2057,8 +2104,8 @@ bool ParseSessionDescription(const std::string& message,
>         return false;
>       }
>     } else if (HasAttribute(line, kAttributeIceUfrag)) {
>-      if (!GetValue(line, kAttributeIceUfrag,
>-                    &(session_td->ice_ufrag), error)) {
>+      if (!GetValue(line, kAttributeIceUfrag, &(session_td->ice_ufrag),
>+                    error)) {
>         return false;
>       }
>     } else if (HasAttribute(line, kAttributeIcePwd)) {
>@@ -2114,8 +2161,7 @@ bool ParseGroupAttribute(const std::string& line,
>   // RFC 5888 and draft-holmberg-mmusic-sdp-bundle-negotiation-00
>   // a=group:BUNDLE video voice
>   std::vector<std::string> fields;
>-  rtc::split(line.substr(kLinePrefixLength),
>-                   kSdpDelimiterSpace, &fields);
>+  rtc::split(line.substr(kLinePrefixLength), kSdpDelimiterSpace, &fields);
>   std::string semantics;
>   if (!GetValue(fields[0], kAttributeGroup, &semantics, error)) {
>     return false;
>@@ -2138,8 +2184,7 @@ static bool ParseFingerprintAttribute(const std::string& line,
>   }
> 
>   std::vector<std::string> fields;
>-  rtc::split(line.substr(kLinePrefixLength),
>-                   kSdpDelimiterSpace, &fields);
>+  rtc::split(line.substr(kLinePrefixLength), kSdpDelimiterSpace, &fields);
>   const size_t expected_fields = 2;
>   if (fields.size() != expected_fields) {
>     return ParseFailedExpectFieldNum(line, expected_fields, error);
>@@ -2157,11 +2202,9 @@ static bool ParseFingerprintAttribute(const std::string& line,
>                  ::tolower);
> 
>   // The second field is the digest value. De-hexify it.
>-  *fingerprint = rtc::SSLFingerprint::CreateFromRfc4572(
>-      algorithm, fields[1]);
>+  *fingerprint = rtc::SSLFingerprint::CreateFromRfc4572(algorithm, fields[1]);
>   if (!*fingerprint) {
>-    return ParseFailed(line,
>-                       "Failed to create fingerprint from the digest.",
>+    return ParseFailed(line, "Failed to create fingerprint from the digest.",
>                        error);
>   }
> 
>@@ -2187,32 +2230,45 @@ static bool ParseDtlsSetup(const std::string& line,
> }
> 
> static bool ParseMsidAttribute(const std::string& line,
>-                               std::string* stream_id,
>+                               std::vector<std::string>* stream_ids,
>                                std::string* track_id,
>                                SdpParseError* error) {
>-  // draft-ietf-mmusic-msid-11
>+  // https://datatracker.ietf.org/doc/draft-ietf-mmusic-msid/16/
>   // a=msid:<stream id> <track id>
>   // msid-value = msid-id [ SP msid-appdata ]
>   // msid-id = 1*64token-char ; see RFC 4566
>   // msid-appdata = 1*64token-char  ; see RFC 4566
>   std::string field1;
>+  std::string new_stream_id;
>+  std::string new_track_id;
>   if (!rtc::tokenize_first(line.substr(kLinePrefixLength), kSdpDelimiterSpace,
>-                           &field1, track_id)) {
>+                           &field1, &new_track_id)) {
>     const size_t expected_fields = 2;
>     return ParseFailedExpectFieldNum(line, expected_fields, error);
>   }
> 
>-  if (track_id->empty()) {
>+  if (new_track_id.empty()) {
>     return ParseFailed(line, "Missing track ID in msid attribute.", error);
>   }
>+  // All track ids should be the same within an m section in a Unified Plan SDP.
>+  if (!track_id->empty() && new_track_id.compare(*track_id) != 0) {
>+    return ParseFailed(
>+        line, "Two different track IDs in msid attribute in one m= section",
>+        error);
>+  }
>+  *track_id = new_track_id;
> 
>   // msid:<msid-id>
>-  if (!GetValue(field1, kAttributeMsid, stream_id, error)) {
>+  if (!GetValue(field1, kAttributeMsid, &new_stream_id, error)) {
>     return false;
>   }
>-  if (stream_id->empty()) {
>+  if (new_stream_id.empty()) {
>     return ParseFailed(line, "Missing stream ID in msid attribute.", error);
>   }
>+  // The special value "-" indicates "no MediaStream".
>+  if (new_stream_id.compare(kNoStreamMsid) != 0) {
>+    stream_ids->push_back(new_stream_id);
>+  }
>   return true;
> }
> 
>@@ -2244,36 +2300,23 @@ struct StaticPayloadAudioCodec {
>   size_t channels;
> };
> static const StaticPayloadAudioCodec kStaticPayloadAudioCodecs[] = {
>-  { "PCMU", 8000, 1 },
>-  { "reserved", 0, 0 },
>-  { "reserved", 0, 0 },
>-  { "GSM", 8000, 1 },
>-  { "G723", 8000, 1 },
>-  { "DVI4", 8000, 1 },
>-  { "DVI4", 16000, 1 },
>-  { "LPC", 8000, 1 },
>-  { "PCMA", 8000, 1 },
>-  { "G722", 8000, 1 },
>-  { "L16", 44100, 2 },
>-  { "L16", 44100, 1 },
>-  { "QCELP", 8000, 1 },
>-  { "CN", 8000, 1 },
>-  { "MPA", 90000, 1 },
>-  { "G728", 8000, 1 },
>-  { "DVI4", 11025, 1 },
>-  { "DVI4", 22050, 1 },
>-  { "G729", 8000, 1 },
>+    {"PCMU", 8000, 1},  {"reserved", 0, 0}, {"reserved", 0, 0},
>+    {"GSM", 8000, 1},   {"G723", 8000, 1},  {"DVI4", 8000, 1},
>+    {"DVI4", 16000, 1}, {"LPC", 8000, 1},   {"PCMA", 8000, 1},
>+    {"G722", 8000, 1},  {"L16", 44100, 2},  {"L16", 44100, 1},
>+    {"QCELP", 8000, 1}, {"CN", 8000, 1},    {"MPA", 90000, 1},
>+    {"G728", 8000, 1},  {"DVI4", 11025, 1}, {"DVI4", 22050, 1},
>+    {"G729", 8000, 1},
> };
> 
>-void MaybeCreateStaticPayloadAudioCodecs(
>-    const std::vector<int>& fmts, AudioContentDescription* media_desc) {
>+void MaybeCreateStaticPayloadAudioCodecs(const std::vector<int>& fmts,
>+                                         AudioContentDescription* media_desc) {
>   if (!media_desc) {
>     return;
>   }
>   RTC_DCHECK(media_desc->codecs().empty());
>   for (int payload_type : fmts) {
>-    if (!media_desc->HasCodec(payload_type) &&
>-        payload_type >= 0 &&
>+    if (!media_desc->HasCodec(payload_type) && payload_type >= 0 &&
>         static_cast<uint32_t>(payload_type) <
>             arraysize(kStaticPayloadAudioCodecs)) {
>       std::string encoding_name = kStaticPayloadAudioCodecs[payload_type].name;
>@@ -2287,13 +2330,14 @@ void MaybeCreateStaticPayloadAudioCodecs(
> 
> template <class C>
> static C* ParseContentDescription(const std::string& message,
>-                                  const MediaType media_type,
>+                                  const cricket::MediaType media_type,
>                                   int mline_index,
>                                   const std::string& protocol,
>                                   const std::vector<int>& payload_types,
>                                   size_t* pos,
>                                   std::string* content_name,
>                                   bool* bundle_only,
>+                                  int* msid_signaling,
>                                   TransportDescription* transport,
>                                   std::vector<JsepIceCandidate*>* candidates,
>                                   webrtc::SdpParseError* error) {
>@@ -2313,8 +2357,8 @@ static C* ParseContentDescription(const std::string& message,
>       break;
>   }
>   if (!ParseContent(message, media_type, mline_index, protocol, payload_types,
>-                    pos, content_name, bundle_only, media_desc, transport,
>-                    candidates, error)) {
>+                    pos, content_name, bundle_only, msid_signaling, media_desc,
>+                    transport, candidates, error)) {
>     delete media_desc;
>     return nullptr;
>   }
>@@ -2328,11 +2372,12 @@ static C* ParseContentDescription(const std::string& message,
>     payload_type_preferences[pt] = preference--;
>   }
>   std::vector<typename C::CodecType> codecs = media_desc->codecs();
>-  std::sort(codecs.begin(), codecs.end(), [&payload_type_preferences](
>-                                              const typename C::CodecType& a,
>-                                              const typename C::CodecType& b) {
>-    return payload_type_preferences[a.id] > payload_type_preferences[b.id];
>-  });
>+  std::sort(codecs.begin(), codecs.end(),
>+            [&payload_type_preferences](const typename C::CodecType& a,
>+                                        const typename C::CodecType& b) {
>+              return payload_type_preferences[a.id] >
>+                     payload_type_preferences[b.id];
>+            });
>   media_desc->set_codecs(codecs);
>   return media_desc;
> }
>@@ -2348,6 +2393,7 @@ bool ParseMediaDescription(const std::string& message,
>   RTC_DCHECK(desc != NULL);
>   std::string line;
>   int mline_index = -1;
>+  int msid_signaling = 0;
> 
>   // Zero or more media descriptions
>   // RFC 4566
>@@ -2356,8 +2402,7 @@ bool ParseMediaDescription(const std::string& message,
>     ++mline_index;
> 
>     std::vector<std::string> fields;
>-    rtc::split(line.substr(kLinePrefixLength),
>-                     kSdpDelimiterSpace, &fields);
>+    rtc::split(line.substr(kLinePrefixLength), kSdpDelimiterSpace, &fields);
> 
>     const size_t expected_min_fields = 4;
>     if (fields.size() < expected_min_fields) {
>@@ -2380,7 +2425,7 @@ bool ParseMediaDescription(const std::string& message,
>     // <fmt>
>     std::vector<int> payload_types;
>     if (IsRtp(protocol)) {
>-      for (size_t j = 3 ; j < fields.size(); ++j) {
>+      for (size_t j = 3; j < fields.size(); ++j) {
>         // TODO(wu): Remove when below bug is fixed.
>         // https://bugzilla.mozilla.org/show_bug.cgi?id=996329
>         if (fields[j].empty() && j == fields.size() - 1) {
>@@ -2405,22 +2450,23 @@ bool ParseMediaDescription(const std::string& message,
>     std::unique_ptr<MediaContentDescription> content;
>     std::string content_name;
>     bool bundle_only = false;
>+    int section_msid_signaling = 0;
>     if (HasAttribute(line, kMediaTypeVideo)) {
>       content.reset(ParseContentDescription<VideoContentDescription>(
>           message, cricket::MEDIA_TYPE_VIDEO, mline_index, protocol,
>-          payload_types, pos, &content_name, &bundle_only, &transport,
>-          candidates, error));
>+          payload_types, pos, &content_name, &bundle_only,
>+          &section_msid_signaling, &transport, candidates, error));
>     } else if (HasAttribute(line, kMediaTypeAudio)) {
>       content.reset(ParseContentDescription<AudioContentDescription>(
>           message, cricket::MEDIA_TYPE_AUDIO, mline_index, protocol,
>-          payload_types, pos, &content_name, &bundle_only, &transport,
>-          candidates, error));
>+          payload_types, pos, &content_name, &bundle_only,
>+          &section_msid_signaling, &transport, candidates, error));
>     } else if (HasAttribute(line, kMediaTypeData)) {
>       DataContentDescription* data_desc =
>           ParseContentDescription<DataContentDescription>(
>               message, cricket::MEDIA_TYPE_DATA, mline_index, protocol,
>-              payload_types, pos, &content_name, &bundle_only, &transport,
>-              candidates, error);
>+              payload_types, pos, &content_name, &bundle_only,
>+              &section_msid_signaling, &transport, candidates, error);
>       content.reset(data_desc);
> 
>       if (data_desc && IsDtlsSctp(protocol)) {
>@@ -2442,6 +2488,8 @@ bool ParseMediaDescription(const std::string& message,
>       return false;
>     }
> 
>+    msid_signaling |= section_msid_signaling;
>+
>     bool content_rejected = false;
>     // A port of 0 is not interpreted as a rejected m= section when it's
>     // used along with a=bundle-only.
>@@ -2452,8 +2500,8 @@ bool ParseMediaDescription(const std::string& message,
>         bundle_only = false;
>         RTC_LOG(LS_WARNING)
>             << "a=bundle-only attribute observed with a nonzero "
>-            << "port; this usage is unspecified so the attribute is being "
>-            << "ignored.";
>+               "port; this usage is unspecified so the attribute is being "
>+               "ignored.";
>       }
>     } else {
>       // If not using bundle-only, interpret port 0 in the normal way; the m=
>@@ -2500,6 +2548,8 @@ bool ParseMediaDescription(const std::string& message,
>     }
>   }
> 
>+  desc->set_msid_signaling(msid_signaling);
>+
>   size_t end_of_message = message.size();
>   if (mline_index == -1 && *pos != end_of_message) {
>     ParseFailed(message, *pos, "Expects m line.", error);
>@@ -2539,8 +2589,8 @@ bool VerifyVideoCodecs(const VideoContentDescription* video_desc) {
> 
> void AddParameters(const cricket::CodecParameterMap& parameters,
>                    cricket::Codec* codec) {
>-  for (cricket::CodecParameterMap::const_iterator iter =
>-           parameters.begin(); iter != parameters.end(); ++iter) {
>+  for (cricket::CodecParameterMap::const_iterator iter = parameters.begin();
>+       iter != parameters.end(); ++iter) {
>     codec->SetParam(iter->first, iter->second);
>   }
> }
>@@ -2598,7 +2648,8 @@ void AddOrReplaceCodec(MediaContentDescription* content_desc, const U& codec) {
> // Adds or updates existing codec corresponding to |payload_type| according
> // to |parameters|.
> template <class T, class U>
>-void UpdateCodec(MediaContentDescription* content_desc, int payload_type,
>+void UpdateCodec(MediaContentDescription* content_desc,
>+                 int payload_type,
>                  const cricket::CodecParameterMap& parameters) {
>   // Codec might already have been populated (from rtpmap).
>   U new_codec = GetCodecWithPayloadType(static_cast<T*>(content_desc)->codecs(),
>@@ -2610,7 +2661,8 @@ void UpdateCodec(MediaContentDescription* content_desc, int payload_type,
> // Adds or updates existing codec corresponding to |payload_type| according
> // to |feedback_param|.
> template <class T, class U>
>-void UpdateCodec(MediaContentDescription* content_desc, int payload_type,
>+void UpdateCodec(MediaContentDescription* content_desc,
>+                 int payload_type,
>                  const cricket::FeedbackParam& feedback_param) {
>   // Codec might already have been populated (from rtpmap).
>   U new_codec = GetCodecWithPayloadType(static_cast<T*>(content_desc)->codecs(),
>@@ -2631,7 +2683,7 @@ bool PopWildcardCodec(std::vector<T>* codecs, T* wildcard_codec) {
>   return false;
> }
> 
>-template<class T>
>+template <class T>
> void UpdateFromWildcardCodecs(cricket::MediaContentDescriptionImpl<T>* desc) {
>   auto codecs = desc->codecs();
>   T wildcard_codec;
>@@ -2644,7 +2696,8 @@ void UpdateFromWildcardCodecs(cricket::MediaContentDescriptionImpl<T>* desc) {
>   desc->set_codecs(codecs);
> }
> 
>-void AddAudioAttribute(const std::string& name, const std::string& value,
>+void AddAudioAttribute(const std::string& name,
>+                       const std::string& value,
>                        AudioContentDescription* audio_desc) {
>   if (value.empty()) {
>     return;
>@@ -2658,13 +2711,14 @@ void AddAudioAttribute(const std::string& name, const std::string& value,
> }
> 
> bool ParseContent(const std::string& message,
>-                  const MediaType media_type,
>+                  const cricket::MediaType media_type,
>                   int mline_index,
>                   const std::string& protocol,
>                   const std::vector<int>& payload_types,
>                   size_t* pos,
>                   std::string* content_name,
>                   bool* bundle_only,
>+                  int* msid_signaling,
>                   MediaContentDescription* media_desc,
>                   TransportDescription* transport,
>                   std::vector<JsepIceCandidate*>* candidates,
>@@ -2688,7 +2742,7 @@ bool ParseContent(const std::string& message,
>   SsrcGroupVec ssrc_groups;
>   std::string maxptime_as_string;
>   std::string ptime_as_string;
>-  std::string stream_id;
>+  std::vector<std::string> stream_ids;
>   std::string track_id;
> 
>   // Loop until the next m line
>@@ -2839,7 +2893,7 @@ bool ParseContent(const std::string& message,
>           return false;
>         }
>       } else if (HasAttribute(line, kAttributeSsrc)) {
>-        if (!ParseSsrcAttribute(line, &ssrc_infos, error)) {
>+        if (!ParseSsrcAttribute(line, &ssrc_infos, msid_signaling, error)) {
>           return false;
>         }
>       } else if (HasAttribute(line, kAttributeCrypto)) {
>@@ -2888,9 +2942,10 @@ bool ParseContent(const std::string& message,
>         if (flag_value.compare(kValueConference) == 0)
>           media_desc->set_conference_mode(true);
>       } else if (HasAttribute(line, kAttributeMsid)) {
>-        if (!ParseMsidAttribute(line, &stream_id, &track_id, error)) {
>+        if (!ParseMsidAttribute(line, &stream_ids, &track_id, error)) {
>           return false;
>         }
>+        *msid_signaling |= cricket::kMsidSignalingMediaSection;
>       }
>     } else {
>       // Only parse lines that we are interested of.
>@@ -2903,7 +2958,17 @@ bool ParseContent(const std::string& message,
>   // If the stream_id/track_id for all SSRCS are identical, one StreamParams
>   // will be created in CreateTracksFromSsrcInfos, containing all the SSRCs from
>   // the m= section.
>-  CreateTracksFromSsrcInfos(ssrc_infos, stream_id, track_id, &tracks);
>+  if (!ssrc_infos.empty()) {
>+    CreateTracksFromSsrcInfos(ssrc_infos, stream_ids, track_id, &tracks,
>+                              *msid_signaling);
>+  } else if (media_type != cricket::MEDIA_TYPE_DATA &&
>+             (*msid_signaling & cricket::kMsidSignalingMediaSection)) {
>+    // If the stream_ids/track_id was signaled but SSRCs were unsignaled we
>+    // still create a track. This isn't done for data media types because
>+    // StreamParams aren't used for SCTP streams, and RTP data channels don't
>+    // support unsignaled SSRCs.
>+    CreateTrackWithNoSsrcs(stream_ids, track_id, &tracks);
>+  }
> 
>   // Add the ssrc group to the track.
>   for (SsrcGroupVec::iterator ssrc_group = ssrc_groups.begin();
>@@ -2957,13 +3022,14 @@ bool ParseContent(const std::string& message,
>     (*it).set_username(transport->ice_ufrag);
>     RTC_DCHECK((*it).password().empty());
>     (*it).set_password(transport->ice_pwd);
>-    candidates->push_back(
>-        new JsepIceCandidate(mline_id, mline_index, *it));
>+    candidates->push_back(new JsepIceCandidate(mline_id, mline_index, *it));
>   }
>   return true;
> }
> 
>-bool ParseSsrcAttribute(const std::string& line, SsrcInfoVec* ssrc_infos,
>+bool ParseSsrcAttribute(const std::string& line,
>+                        SsrcInfoVec* ssrc_infos,
>+                        int* msid_signaling,
>                         SdpParseError* error) {
>   RTC_DCHECK(ssrc_infos != NULL);
>   // RFC 5576
>@@ -3017,18 +3083,18 @@ bool ParseSsrcAttribute(const std::string& line, SsrcInfoVec* ssrc_infos,
>     ssrc_info->cname = value;
>   } else if (attribute == kSsrcAttributeMsid) {
>     // draft-alvestrand-mmusic-msid-00
>-    // "msid:" identifier [ " " appdata ]
>+    // msid:identifier [appdata]
>     std::vector<std::string> fields;
>     rtc::split(value, kSdpDelimiterSpace, &fields);
>     if (fields.size() < 1 || fields.size() > 2) {
>-      return ParseFailed(line,
>-                         "Expected format \"msid:<identifier>[ <appdata>]\".",
>-                         error);
>+      return ParseFailed(
>+          line, "Expected format \"msid:<identifier>[ <appdata>]\".", error);
>     }
>     ssrc_info->stream_id = fields[0];
>     if (fields.size() == 2) {
>       ssrc_info->track_id = fields[1];
>     }
>+    *msid_signaling |= cricket::kMsidSignalingSsrcAttribute;
>   } else if (attribute == kSsrcAttributeMslabel) {
>     // draft-alvestrand-rtcweb-mid-01
>     // mslabel:<value>
>@@ -3048,8 +3114,7 @@ bool ParseSsrcGroupAttribute(const std::string& line,
>   // RFC 5576
>   // a=ssrc-group:<semantics> <ssrc-id> ...
>   std::vector<std::string> fields;
>-  rtc::split(line.substr(kLinePrefixLength),
>-                   kSdpDelimiterSpace, &fields);
>+  rtc::split(line.substr(kLinePrefixLength), kSdpDelimiterSpace, &fields);
>   const size_t expected_min_fields = 2;
>   if (fields.size() < expected_min_fields) {
>     return ParseFailedExpectMinFieldNum(line, expected_min_fields, error);
>@@ -3074,8 +3139,7 @@ bool ParseCryptoAttribute(const std::string& line,
>                           MediaContentDescription* media_desc,
>                           SdpParseError* error) {
>   std::vector<std::string> fields;
>-  rtc::split(line.substr(kLinePrefixLength),
>-                   kSdpDelimiterSpace, &fields);
>+  rtc::split(line.substr(kLinePrefixLength), kSdpDelimiterSpace, &fields);
>   // RFC 4568
>   // a=crypto:<tag> <crypto-suite> <key-params> [<session-params>]
>   const size_t expected_min_fields = 3;
>@@ -3096,8 +3160,8 @@ bool ParseCryptoAttribute(const std::string& line,
>   if (fields.size() > 3) {
>     session_params = fields[3];
>   }
>-  media_desc->AddCrypto(CryptoParams(tag, crypto_suite, key_params,
>-                                     session_params));
>+  media_desc->AddCrypto(
>+      CryptoParams(tag, crypto_suite, key_params, session_params));
>   return true;
> }
> 
>@@ -3136,13 +3200,12 @@ void UpdateCodec(int payload_type,
> }
> 
> bool ParseRtpmapAttribute(const std::string& line,
>-                          const MediaType media_type,
>+                          const cricket::MediaType media_type,
>                           const std::vector<int>& payload_types,
>                           MediaContentDescription* media_desc,
>                           SdpParseError* error) {
>   std::vector<std::string> fields;
>-  rtc::split(line.substr(kLinePrefixLength),
>-                   kSdpDelimiterSpace, &fields);
>+  rtc::split(line.substr(kLinePrefixLength), kSdpDelimiterSpace, &fields);
>   // RFC 4566
>   // a=rtpmap:<payload type> <encoding name>/<clock rate>[/<encodingparameters>]
>   const size_t expected_min_fields = 2;
>@@ -3162,7 +3225,8 @@ bool ParseRtpmapAttribute(const std::string& line,
>   if (std::find(payload_types.begin(), payload_types.end(), payload_type) ==
>       payload_types.end()) {
>     RTC_LOG(LS_WARNING) << "Ignore rtpmap line that did not appear in the "
>-                        << "<fmt> of the m-line: " << line;
>+                           "<fmt> of the m-line: "
>+                        << line;
>     return true;
>   }
>   const std::string& encoder = fields[1];
>@@ -3183,8 +3247,7 @@ bool ParseRtpmapAttribute(const std::string& line,
>   }
>   if (media_type == cricket::MEDIA_TYPE_VIDEO) {
>     VideoContentDescription* video_desc = media_desc->as_video();
>-    UpdateCodec(payload_type, encoding_name,
>-                video_desc);
>+    UpdateCodec(payload_type, encoding_name, video_desc);
>   } else if (media_type == cricket::MEDIA_TYPE_AUDIO) {
>     // RFC 4566
>     // For audio streams, <encoding parameters> indicates the number
>@@ -3207,8 +3270,10 @@ bool ParseRtpmapAttribute(const std::string& line,
>   return true;
> }
> 
>-bool ParseFmtpParam(const std::string& line, std::string* parameter,
>-                    std::string* value, SdpParseError* error) {
>+bool ParseFmtpParam(const std::string& line,
>+                    std::string* parameter,
>+                    std::string* value,
>+                    SdpParseError* error) {
>   if (!rtc::tokenize_first(line, kSdpDelimiterEqual, parameter, value)) {
>     ParseFailed(line, "Unable to parse fmtp parameter. \'=\' missing.", error);
>     return false;
>@@ -3217,7 +3282,8 @@ bool ParseFmtpParam(const std::string& line, std::string* parameter,
>   return true;
> }
> 
>-bool ParseFmtpAttributes(const std::string& line, const MediaType media_type,
>+bool ParseFmtpAttributes(const std::string& line,
>+                         const cricket::MediaType media_type,
>                          MediaContentDescription* media_desc,
>                          SdpParseError* error) {
>   if (media_type != cricket::MEDIA_TYPE_AUDIO &&
>@@ -3280,7 +3346,8 @@ bool ParseFmtpAttributes(const std::string& line, const MediaType media_type,
>   return true;
> }
> 
>-bool ParseRtcpFbAttribute(const std::string& line, const MediaType media_type,
>+bool ParseRtcpFbAttribute(const std::string& line,
>+                          const cricket::MediaType media_type,
>                           MediaContentDescription* media_desc,
>                           SdpParseError* error) {
>   if (media_type != cricket::MEDIA_TYPE_AUDIO &&
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/webrtcsdp.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/webrtcsdp.h
>index 83cc7688a35..2d842a1d5d4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/webrtcsdp.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/webrtcsdp.h
>@@ -38,8 +38,7 @@ struct SdpParseError;
> // jdesc - The JsepSessionDescription object to be serialized.
> // unified_plan_sdp - If set to true, include "a=msid" lines where appropriate.
> // return - SDP string serialized from the arguments.
>-std::string SdpSerialize(const JsepSessionDescription& jdesc,
>-                         bool unified_plan_sdp);
>+std::string SdpSerialize(const JsepSessionDescription& jdesc);
> 
> // Serializes the passed in IceCandidateInterface to a SDP string.
> // candidate - The candidate to be serialized.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/webrtcsdp_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/webrtcsdp_unittest.cc
>index 831d23655af..bfae1b9a12d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/webrtcsdp_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/webrtcsdp_unittest.cc
>@@ -16,8 +16,8 @@
> #include "api/jsepsessiondescription.h"
> #include "media/base/mediaconstants.h"
> #include "media/engine/webrtcvideoengine.h"
>-#include "p2p/base/port.h"
> #include "p2p/base/p2pconstants.h"
>+#include "p2p/base/port.h"
> #include "pc/mediasession.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/gunit.h"
>@@ -86,7 +86,8 @@ static const char kAttributeCryptoVoice[] =
> static const char kAttributeCryptoVideo[] =
>     "a=crypto:1 AES_CM_128_HMAC_SHA1_80 "
>     "inline:d0RmdmcmVCspeEc3QGZiNWpVLFJhQX1cfHAwJSoj|2^20|1:32\r\n";
>-static const char kFingerprint[] = "a=fingerprint:sha-1 "
>+static const char kFingerprint[] =
>+    "a=fingerprint:sha-1 "
>     "4A:AD:B9:B1:3F:82:18:3B:54:02:12:DF:3E:5D:49:6B:19:E5:7C:AB\r\n";
> static const int kExtmapId = 1;
> static const char kExtmapUri[] = "http://example.com/082005/ext.htm#ttime";
>@@ -480,14 +481,15 @@ static const char kPlanBSdpFullString[] =
>     "a=ssrc:6 mslabel:local_stream_2\r\n"
>     "a=ssrc:6 label:video_track_id_3\r\n";
> 
>-// Plan B SDP reference string, with 2 streams, 2 audio tracks and 3 video
>-// tracks, but with the unified plan "a=msid" attribute.
>-static const char kPlanBSdpFullStringWithMsid[] =
>+// Unified Plan SDP reference string, with 2 streams, 2 audio tracks and 3 video
>+// tracks.
>+static const char kUnifiedPlanSdpFullString[] =
>     "v=0\r\n"
>     "o=- 18446744069414584320 18446462598732840960 IN IP4 127.0.0.1\r\n"
>     "s=-\r\n"
>     "t=0 0\r\n"
>-    "a=msid-semantic: WMS local_stream_1 local_stream_2\r\n"
>+    "a=msid-semantic: WMS local_stream_1\r\n"
>+    // Audio track 1, stream 1 (with candidates).
>     "m=audio 2345 RTP/SAVPF 111 103 104\r\n"
>     "c=IN IP4 74.125.127.126\r\n"
>     "a=rtcp:2347 IN IP4 74.125.127.126\r\n"
>@@ -518,13 +520,7 @@ static const char kPlanBSdpFullStringWithMsid[] =
>     "a=rtpmap:103 ISAC/16000\r\n"
>     "a=rtpmap:104 ISAC/32000\r\n"
>     "a=ssrc:1 cname:stream_1_cname\r\n"
>-    "a=ssrc:1 msid:local_stream_1 audio_track_id_1\r\n"
>-    "a=ssrc:1 mslabel:local_stream_1\r\n"
>-    "a=ssrc:1 label:audio_track_id_1\r\n"
>-    "a=ssrc:4 cname:stream_2_cname\r\n"
>-    "a=ssrc:4 msid:local_stream_2 audio_track_id_2\r\n"
>-    "a=ssrc:4 mslabel:local_stream_2\r\n"
>-    "a=ssrc:4 label:audio_track_id_2\r\n"
>+    // Video track 1, stream 1 (with candidates).
>     "m=video 3457 RTP/SAVPF 120\r\n"
>     "c=IN IP4 74.125.224.39\r\n"
>     "a=rtcp:3456 IN IP4 74.125.224.39\r\n"
>@@ -549,31 +545,63 @@ static const char kPlanBSdpFullStringWithMsid[] =
>     "a=rtpmap:120 VP8/90000\r\n"
>     "a=ssrc-group:FEC 2 3\r\n"
>     "a=ssrc:2 cname:stream_1_cname\r\n"
>-    "a=ssrc:2 msid:local_stream_1 video_track_id_1\r\n"
>-    "a=ssrc:2 mslabel:local_stream_1\r\n"
>-    "a=ssrc:2 label:video_track_id_1\r\n"
>     "a=ssrc:3 cname:stream_1_cname\r\n"
>-    "a=ssrc:3 msid:local_stream_1 video_track_id_1\r\n"
>-    "a=ssrc:3 mslabel:local_stream_1\r\n"
>-    "a=ssrc:3 label:video_track_id_1\r\n"
>+    // Audio track 2, stream 2.
>+    "m=audio 9 RTP/SAVPF 111 103 104\r\n"
>+    "c=IN IP4 0.0.0.0\r\n"
>+    "a=rtcp:9 IN IP4 0.0.0.0\r\n"
>+    "a=ice-ufrag:ufrag_voice_2\r\na=ice-pwd:pwd_voice_2\r\n"
>+    "a=mid:audio_content_name_2\r\n"
>+    "a=msid:local_stream_2 audio_track_id_2\r\n"
>+    "a=sendrecv\r\n"
>+    "a=rtcp-mux\r\n"
>+    "a=rtcp-rsize\r\n"
>+    "a=crypto:1 AES_CM_128_HMAC_SHA1_32 "
>+    "inline:NzB4d1BINUAvLEw6UzF3WSJ+PSdFcGdUJShpX1Zj|2^20|1:32 "
>+    "dummy_session_params\r\n"
>+    "a=rtpmap:111 opus/48000/2\r\n"
>+    "a=rtpmap:103 ISAC/16000\r\n"
>+    "a=rtpmap:104 ISAC/32000\r\n"
>+    "a=ssrc:4 cname:stream_2_cname\r\n"
>+    // Video track 2, stream 2.
>+    "m=video 9 RTP/SAVPF 120\r\n"
>+    "c=IN IP4 0.0.0.0\r\n"
>+    "a=rtcp:9 IN IP4 0.0.0.0\r\n"
>+    "a=ice-ufrag:ufrag_video_2\r\na=ice-pwd:pwd_video_2\r\n"
>+    "a=mid:video_content_name_2\r\n"
>+    "a=msid:local_stream_2 video_track_id_2\r\n"
>+    "a=sendrecv\r\n"
>+    "a=crypto:1 AES_CM_128_HMAC_SHA1_80 "
>+    "inline:d0RmdmcmVCspeEc3QGZiNWpVLFJhQX1cfHAwJSoj|2^20|1:32\r\n"
>+    "a=rtpmap:120 VP8/90000\r\n"
>     "a=ssrc:5 cname:stream_2_cname\r\n"
>-    "a=ssrc:5 msid:local_stream_2 video_track_id_2\r\n"
>-    "a=ssrc:5 mslabel:local_stream_2\r\n"
>-    "a=ssrc:5 label:video_track_id_2\r\n"
>-    "a=ssrc:6 cname:stream_2_cname\r\n"
>-    "a=ssrc:6 msid:local_stream_2 video_track_id_3\r\n"
>-    "a=ssrc:6 mslabel:local_stream_2\r\n"
>-    "a=ssrc:6 label:video_track_id_3\r\n";
>+    // Video track 3, stream 2.
>+    "m=video 9 RTP/SAVPF 120\r\n"
>+    "c=IN IP4 0.0.0.0\r\n"
>+    "a=rtcp:9 IN IP4 0.0.0.0\r\n"
>+    "a=ice-ufrag:ufrag_video_3\r\na=ice-pwd:pwd_video_3\r\n"
>+    "a=mid:video_content_name_3\r\n"
>+    "a=msid:local_stream_2 video_track_id_3\r\n"
>+    "a=sendrecv\r\n"
>+    "a=crypto:1 AES_CM_128_HMAC_SHA1_80 "
>+    "inline:d0RmdmcmVCspeEc3QGZiNWpVLFJhQX1cfHAwJSoj|2^20|1:32\r\n"
>+    "a=rtpmap:120 VP8/90000\r\n"
>+    "a=ssrc:6 cname:stream_2_cname\r\n";
> 
>-// Unified Plan SDP reference string, with 2 streams, 2 audio tracks and 3 video
>-// tracks.
>-static const char kUnifiedPlanSdpFullString[] =
>+// Unified Plan SDP reference string:
>+// - audio track 1 has 1 a=msid lines
>+// - audio track 2 has 2 a=msid lines
>+// - audio track 3 has 1 a=msid line with the special "-" marker signifying that
>+//   there are 0 media stream ids.
>+// This Unified Plan SDP represents a SDP that signals the msid using both
>+// a=msid and a=ssrc msid semantics.
>+static const char kUnifiedPlanSdpFullStringWithSpecialMsid[] =
>     "v=0\r\n"
>     "o=- 18446744069414584320 18446462598732840960 IN IP4 127.0.0.1\r\n"
>     "s=-\r\n"
>     "t=0 0\r\n"
>     "a=msid-semantic: WMS local_stream_1\r\n"
>-    // Audio track 1, stream 1 (with candidates).
>+    // Audio track 1, with 1 stream id.
>     "m=audio 2345 RTP/SAVPF 111 103 104\r\n"
>     "c=IN IP4 74.125.127.126\r\n"
>     "a=rtcp:2347 IN IP4 74.125.127.126\r\n"
>@@ -604,38 +632,16 @@ static const char kUnifiedPlanSdpFullString[] =
>     "a=rtpmap:103 ISAC/16000\r\n"
>     "a=rtpmap:104 ISAC/32000\r\n"
>     "a=ssrc:1 cname:stream_1_cname\r\n"
>-    // Video track 1, stream 1 (with candidates).
>-    "m=video 3457 RTP/SAVPF 120\r\n"
>-    "c=IN IP4 74.125.224.39\r\n"
>-    "a=rtcp:3456 IN IP4 74.125.224.39\r\n"
>-    "a=candidate:a0+B/1 2 udp 2130706432 192.168.1.5 1236 typ host "
>-    "generation 2\r\n"
>-    "a=candidate:a0+B/1 1 udp 2130706432 192.168.1.5 1237 typ host "
>-    "generation 2\r\n"
>-    "a=candidate:a0+B/2 2 udp 2130706432 ::1 1240 typ host "
>-    "generation 2\r\n"
>-    "a=candidate:a0+B/2 1 udp 2130706432 ::1 1241 typ host "
>-    "generation 2\r\n"
>-    "a=candidate:a0+B/4 2 udp 2130706432 74.125.224.39 3456 typ relay "
>-    "generation 2\r\n"
>-    "a=candidate:a0+B/4 1 udp 2130706432 74.125.224.39 3457 typ relay "
>-    "generation 2\r\n"
>-    "a=ice-ufrag:ufrag_video\r\na=ice-pwd:pwd_video\r\n"
>-    "a=mid:video_content_name\r\n"
>-    "a=msid:local_stream_1 video_track_id_1\r\n"
>-    "a=sendrecv\r\n"
>-    "a=crypto:1 AES_CM_128_HMAC_SHA1_80 "
>-    "inline:d0RmdmcmVCspeEc3QGZiNWpVLFJhQX1cfHAwJSoj|2^20|1:32\r\n"
>-    "a=rtpmap:120 VP8/90000\r\n"
>-    "a=ssrc-group:FEC 2 3\r\n"
>-    "a=ssrc:2 cname:stream_1_cname\r\n"
>-    "a=ssrc:3 cname:stream_1_cname\r\n"
>-    // Audio track 2, stream 2.
>+    "a=ssrc:1 msid:local_stream_1 audio_track_id_1\r\n"
>+    "a=ssrc:1 mslabel:local_stream_1\r\n"
>+    "a=ssrc:1 label:audio_track_id_1\r\n"
>+    // Audio track 2, with two stream ids.
>     "m=audio 9 RTP/SAVPF 111 103 104\r\n"
>     "c=IN IP4 0.0.0.0\r\n"
>     "a=rtcp:9 IN IP4 0.0.0.0\r\n"
>     "a=ice-ufrag:ufrag_voice_2\r\na=ice-pwd:pwd_voice_2\r\n"
>     "a=mid:audio_content_name_2\r\n"
>+    "a=msid:local_stream_1 audio_track_id_2\r\n"
>     "a=msid:local_stream_2 audio_track_id_2\r\n"
>     "a=sendrecv\r\n"
>     "a=rtcp-mux\r\n"
>@@ -646,31 +652,29 @@ static const char kUnifiedPlanSdpFullString[] =
>     "a=rtpmap:111 opus/48000/2\r\n"
>     "a=rtpmap:103 ISAC/16000\r\n"
>     "a=rtpmap:104 ISAC/32000\r\n"
>-    "a=ssrc:4 cname:stream_2_cname\r\n"
>-    // Video track 2, stream 2.
>-    "m=video 9 RTP/SAVPF 120\r\n"
>-    "c=IN IP4 0.0.0.0\r\n"
>-    "a=rtcp:9 IN IP4 0.0.0.0\r\n"
>-    "a=ice-ufrag:ufrag_video_2\r\na=ice-pwd:pwd_video_2\r\n"
>-    "a=mid:video_content_name_2\r\n"
>-    "a=msid:local_stream_2 video_track_id_2\r\n"
>-    "a=sendrecv\r\n"
>-    "a=crypto:1 AES_CM_128_HMAC_SHA1_80 "
>-    "inline:d0RmdmcmVCspeEc3QGZiNWpVLFJhQX1cfHAwJSoj|2^20|1:32\r\n"
>-    "a=rtpmap:120 VP8/90000\r\n"
>-    "a=ssrc:5 cname:stream_2_cname\r\n"
>-    // Video track 3, stream 2.
>-    "m=video 9 RTP/SAVPF 120\r\n"
>+    "a=ssrc:4 cname:stream_1_cname\r\n"
>+    // The support for Plan B msid signaling only includes the
>+    // first media stream id "local_stream_1."
>+    "a=ssrc:4 msid:local_stream_1 audio_track_id_2\r\n"
>+    "a=ssrc:4 mslabel:local_stream_1\r\n"
>+    "a=ssrc:4 label:audio_track_id_2\r\n"
>+    // Audio track 3, with no stream ids.
>+    "m=audio 9 RTP/SAVPF 111 103 104\r\n"
>     "c=IN IP4 0.0.0.0\r\n"
>     "a=rtcp:9 IN IP4 0.0.0.0\r\n"
>-    "a=ice-ufrag:ufrag_video_3\r\na=ice-pwd:pwd_video_3\r\n"
>-    "a=mid:video_content_name_3\r\n"
>-    "a=msid:local_stream_2 video_track_id_3\r\n"
>+    "a=ice-ufrag:ufrag_voice_3\r\na=ice-pwd:pwd_voice_3\r\n"
>+    "a=mid:audio_content_name_3\r\n"
>+    "a=msid:- audio_track_id_3\r\n"
>     "a=sendrecv\r\n"
>-    "a=crypto:1 AES_CM_128_HMAC_SHA1_80 "
>-    "inline:d0RmdmcmVCspeEc3QGZiNWpVLFJhQX1cfHAwJSoj|2^20|1:32\r\n"
>-    "a=rtpmap:120 VP8/90000\r\n"
>-    "a=ssrc:6 cname:stream_2_cname\r\n";
>+    "a=rtcp-mux\r\n"
>+    "a=rtcp-rsize\r\n"
>+    "a=crypto:1 AES_CM_128_HMAC_SHA1_32 "
>+    "inline:NzB4d1BINUAvLEw6UzF3WSJ+PSdFcGdUJShpX1Zj|2^20|1:32 "
>+    "dummy_session_params\r\n"
>+    "a=rtpmap:111 opus/48000/2\r\n"
>+    "a=rtpmap:103 ISAC/16000\r\n"
>+    "a=rtpmap:104 ISAC/32000\r\n"
>+    "a=ssrc:7 cname:stream_2_cname\r\n";
> 
> // One candidate reference string as per W3c spec.
> // candidate:<blah> not a=candidate:<blah>CRLF
>@@ -704,6 +708,9 @@ static const char kSdpOneCandidateWithUfragPwd[] =
>     "a=candidate:a0+B/1 1 udp 2130706432 192.168.1.5 1234 typ host network_name"
>     " eth0 ufrag user_rtp pwd password_rtp generation 2\r\n";
> 
>+static const char kRawHostnameCandidate[] =
>+    "candidate:a0+B/1 1 udp 2130706432 a.test 1234 typ host generation 2";
>+
> // Session id and version
> static const char kSessionId[] = "18446744069414584320";
> static const char kSessionVersion[] = "18446462598732840960";
>@@ -724,6 +731,8 @@ static const char kPwdData[] = "pwd_data";
> // Extra ufrags/passwords for extra unified plan m= sections.
> static const char kUfragVoice2[] = "ufrag_voice_2";
> static const char kPwdVoice2[] = "pwd_voice_2";
>+static const char kUfragVoice3[] = "ufrag_voice_3";
>+static const char kPwdVoice3[] = "pwd_voice_3";
> static const char kUfragVideo2[] = "ufrag_video_2";
> static const char kPwdVideo2[] = "pwd_video_2";
> static const char kUfragVideo3[] = "ufrag_video_3";
>@@ -736,11 +745,12 @@ static const char kDataContentName[] = "data_content_name";
> 
> // Extra content names for extra unified plan m= sections.
> static const char kAudioContentName2[] = "audio_content_name_2";
>+static const char kAudioContentName3[] = "audio_content_name_3";
> static const char kVideoContentName2[] = "video_content_name_2";
> static const char kVideoContentName3[] = "video_content_name_3";
> 
> // MediaStream 1
>-static const char kStreamLabel1[] = "local_stream_1";
>+static const char kStreamId1[] = "local_stream_1";
> static const char kStream1Cname[] = "stream_1_cname";
> static const char kAudioTrackId1[] = "audio_track_id_1";
> static const uint32_t kAudioTrack1Ssrc = 1;
>@@ -749,7 +759,7 @@ static const uint32_t kVideoTrack1Ssrc1 = 2;
> static const uint32_t kVideoTrack1Ssrc2 = 3;
> 
> // MediaStream 2
>-static const char kStreamLabel2[] = "local_stream_2";
>+static const char kStreamId2[] = "local_stream_2";
> static const char kStream2Cname[] = "stream_2_cname";
> static const char kAudioTrackId2[] = "audio_track_id_2";
> static const uint32_t kAudioTrack2Ssrc = 4;
>@@ -757,6 +767,8 @@ static const char kVideoTrackId2[] = "video_track_id_2";
> static const uint32_t kVideoTrack2Ssrc = 5;
> static const char kVideoTrackId3[] = "video_track_id_3";
> static const uint32_t kVideoTrack3Ssrc = 6;
>+static const char kAudioTrackId3[] = "audio_track_id_3";
>+static const uint32_t kAudioTrack3Ssrc = 7;
> 
> // DataChannel
> static const char kDataChannelLabel[] = "data_channel";
>@@ -788,15 +800,15 @@ static void InjectAfter(const std::string& line,
>                         const std::string& newlines,
>                         std::string* message) {
>   const std::string tmp = line + newlines;
>-  rtc::replace_substrs(line.c_str(), line.length(),
>-                             tmp.c_str(), tmp.length(), message);
>+  rtc::replace_substrs(line.c_str(), line.length(), tmp.c_str(), tmp.length(),
>+                       message);
> }
> 
> static void Replace(const std::string& line,
>                     const std::string& newlines,
>                     std::string* message) {
>-  rtc::replace_substrs(line.c_str(), line.length(),
>-                             newlines.c_str(), newlines.length(), message);
>+  rtc::replace_substrs(line.c_str(), line.length(), newlines.c_str(),
>+                       newlines.length(), message);
> }
> 
> // Expect fail to parase |bad_sdp| and expect |bad_part| be part of the error
>@@ -847,7 +859,8 @@ static void ReplaceDirection(RtpTransceiverDirection direction,
>   Replace("a=sendrecv", new_direction, message);
> }
> 
>-static void ReplaceRejected(bool audio_rejected, bool video_rejected,
>+static void ReplaceRejected(bool audio_rejected,
>+                            bool video_rejected,
>                             std::string* message) {
>   if (audio_rejected) {
>     Replace("m=audio 9", "m=audio 0", message);
>@@ -874,7 +887,7 @@ class WebRtcSdpTest : public testing::Test {
>     StreamParams audio_stream;
>     audio_stream.id = kAudioTrackId1;
>     audio_stream.cname = kStream1Cname;
>-    audio_stream.sync_label = kStreamLabel1;
>+    audio_stream.set_stream_ids({kStreamId1});
>     audio_stream.ssrcs.push_back(kAudioTrack1Ssrc);
>     audio_desc_->AddStream(audio_stream);
>     rtc::SocketAddress audio_addr("74.125.127.126", 2345);
>@@ -886,7 +899,7 @@ class WebRtcSdpTest : public testing::Test {
>     StreamParams video_stream;
>     video_stream.id = kVideoTrackId1;
>     video_stream.cname = kStream1Cname;
>-    video_stream.sync_label = kStreamLabel1;
>+    video_stream.set_stream_ids({kStreamId1});
>     video_stream.ssrcs.push_back(kVideoTrack1Ssrc1);
>     video_stream.ssrcs.push_back(kVideoTrack1Ssrc2);
>     cricket::SsrcGroup ssrc_group(kFecSsrcGroupSemantics, video_stream.ssrcs);
>@@ -990,32 +1003,25 @@ class WebRtcSdpTest : public testing::Test {
>     candidates_.push_back(candidate11);
>     candidates_.push_back(candidate12);
> 
>-    jcandidate_.reset(new JsepIceCandidate(std::string("audio_content_name"),
>-                                           0, candidate1));
>+    jcandidate_.reset(
>+        new JsepIceCandidate(std::string("audio_content_name"), 0, candidate1));
> 
>     // Set up JsepSessionDescription.
>     jdesc_.Initialize(desc_.Copy(), kSessionId, kSessionVersion);
>     std::string mline_id;
>     int mline_index = 0;
>-    for (size_t i = 0; i< candidates_.size(); ++i) {
>+    for (size_t i = 0; i < candidates_.size(); ++i) {
>       // In this test, the audio m line index will be 0, and the video m line
>       // will be 1.
>       bool is_video = (i > 5);
>       mline_id = is_video ? "video_content_name" : "audio_content_name";
>       mline_index = is_video ? 1 : 0;
>-      JsepIceCandidate jice(mline_id,
>-                            mline_index,
>-                            candidates_.at(i));
>+      JsepIceCandidate jice(mline_id, mline_index, candidates_.at(i));
>       jdesc_.AddCandidate(&jice);
>     }
>   }
> 
>-  // Turns the existing reference description into a description using
>-  // a=bundle-only. This means no transport attributes and a 0 port value on
>-  // the m= sections not associated with the BUNDLE-tag.
>-  void MakeBundleOnlyDescription() {
>-    // Remove video candidates. JsepSessionDescription doesn't make it
>-    // simple.
>+  void RemoveVideoCandidates() {
>     const IceCandidateCollection* video_candidates_collection =
>         jdesc_.candidates(1);
>     ASSERT_NE(nullptr, video_candidates_collection);
>@@ -1026,6 +1032,13 @@ class WebRtcSdpTest : public testing::Test {
>       video_candidates.push_back(c);
>     }
>     jdesc_.RemoveCandidates(video_candidates);
>+  }
>+
>+  // Turns the existing reference description into a description using
>+  // a=bundle-only. This means no transport attributes and a 0 port value on
>+  // the m= sections not associated with the BUNDLE-tag.
>+  void MakeBundleOnlyDescription() {
>+    RemoveVideoCandidates();
> 
>     // And the rest of the transport attributes.
>     desc_.transport_infos()[1].description.ice_ufrag.clear();
>@@ -1055,21 +1068,21 @@ class WebRtcSdpTest : public testing::Test {
>     StreamParams audio_track_2;
>     audio_track_2.id = kAudioTrackId2;
>     audio_track_2.cname = kStream2Cname;
>-    audio_track_2.sync_label = kStreamLabel2;
>+    audio_track_2.set_stream_ids({kStreamId2});
>     audio_track_2.ssrcs.push_back(kAudioTrack2Ssrc);
>     audio_desc_->AddStream(audio_track_2);
> 
>     StreamParams video_track_2;
>     video_track_2.id = kVideoTrackId2;
>     video_track_2.cname = kStream2Cname;
>-    video_track_2.sync_label = kStreamLabel2;
>+    video_track_2.set_stream_ids({kStreamId2});
>     video_track_2.ssrcs.push_back(kVideoTrack2Ssrc);
>     video_desc_->AddStream(video_track_2);
> 
>     StreamParams video_track_3;
>     video_track_3.id = kVideoTrackId3;
>     video_track_3.cname = kStream2Cname;
>-    video_track_3.sync_label = kStreamLabel2;
>+    video_track_3.set_stream_ids({kStreamId2});
>     video_track_3.ssrcs.push_back(kVideoTrack3Ssrc);
>     video_desc_->AddStream(video_track_3);
> 
>@@ -1090,7 +1103,7 @@ class WebRtcSdpTest : public testing::Test {
>     StreamParams audio_track_2;
>     audio_track_2.id = kAudioTrackId2;
>     audio_track_2.cname = kStream2Cname;
>-    audio_track_2.sync_label = kStreamLabel2;
>+    audio_track_2.set_stream_ids({kStreamId2});
>     audio_track_2.ssrcs.push_back(kAudioTrack2Ssrc);
>     audio_desc_2->AddStream(audio_track_2);
>     desc_.AddContent(kAudioContentName2, MediaProtocolType::kRtp, audio_desc_2);
>@@ -1101,7 +1114,7 @@ class WebRtcSdpTest : public testing::Test {
>     StreamParams video_track_2;
>     video_track_2.id = kVideoTrackId2;
>     video_track_2.cname = kStream2Cname;
>-    video_track_2.sync_label = kStreamLabel2;
>+    video_track_2.set_stream_ids({kStreamId2});
>     video_track_2.ssrcs.push_back(kVideoTrack2Ssrc);
>     video_desc_2->AddStream(video_track_2);
>     desc_.AddContent(kVideoContentName2, MediaProtocolType::kRtp, video_desc_2);
>@@ -1113,12 +1126,13 @@ class WebRtcSdpTest : public testing::Test {
>     StreamParams video_track_3;
>     video_track_3.id = kVideoTrackId3;
>     video_track_3.cname = kStream2Cname;
>-    video_track_3.sync_label = kStreamLabel2;
>+    video_track_3.set_stream_ids({kStreamId2});
>     video_track_3.ssrcs.push_back(kVideoTrack3Ssrc);
>     video_desc_3->AddStream(video_track_3);
>     desc_.AddContent(kVideoContentName3, MediaProtocolType::kRtp, video_desc_3);
>     EXPECT_TRUE(desc_.AddTransportInfo(TransportInfo(
>         kVideoContentName3, TransportDescription(kUfragVideo3, kPwdVideo3))));
>+    desc_.set_msid_signaling(cricket::kMsidSignalingMediaSection);
> 
>     ASSERT_TRUE(jdesc_.Initialize(desc_.Copy(), jdesc_.session_id(),
>                                   jdesc_.session_version()));
>@@ -1130,7 +1144,8 @@ class WebRtcSdpTest : public testing::Test {
>     AudioContentDescription* audio = new AudioContentDescription();
>     audio->set_rtcp_mux(true);
>     audio->set_rtcp_reduced_size(true);
>-    audio->AddCrypto(CryptoParams(1, "AES_CM_128_HMAC_SHA1_32",
>+    audio->AddCrypto(CryptoParams(
>+        1, "AES_CM_128_HMAC_SHA1_32",
>         "inline:NzB4d1BINUAvLEw6UzF3WSJ+PSdFcGdUJShpX1Zj|2^20|1:32",
>         "dummy_session_params"));
>     audio->set_protocol(cricket::kMediaProtocolSavpf);
>@@ -1141,6 +1156,67 @@ class WebRtcSdpTest : public testing::Test {
>     return audio;
>   }
> 
>+  // Turns the existing reference description into a unified plan description,
>+  // with 3 audio MediaContentDescriptions with special StreamParams that
>+  // contain 0 or multiple stream ids: - audio track 1 has 1 media stream id -
>+  // audio track 2 has 2 media stream ids - audio track 3 has 0 media stream ids
>+  void MakeUnifiedPlanDescriptionMultipleStreamIds() {
>+    desc_.RemoveContentByName(kVideoContentName);
>+    desc_.RemoveTransportInfoByName(kVideoContentName);
>+    RemoveVideoCandidates();
>+
>+    // Audio track 2 has 2 media stream ids.
>+    AudioContentDescription* audio_desc_2 = CreateAudioContentDescription();
>+    StreamParams audio_track_2;
>+    audio_track_2.id = kAudioTrackId2;
>+    audio_track_2.cname = kStream1Cname;
>+    audio_track_2.set_stream_ids({kStreamId1, kStreamId2});
>+    audio_track_2.ssrcs.push_back(kAudioTrack2Ssrc);
>+    audio_desc_2->AddStream(audio_track_2);
>+    desc_.AddContent(kAudioContentName2, MediaProtocolType::kRtp, audio_desc_2);
>+    EXPECT_TRUE(desc_.AddTransportInfo(TransportInfo(
>+        kAudioContentName2, TransportDescription(kUfragVoice2, kPwdVoice2))));
>+
>+    // Audio track 3 has no stream ids.
>+    AudioContentDescription* audio_desc_3 = CreateAudioContentDescription();
>+    StreamParams audio_track_3;
>+    audio_track_3.id = kAudioTrackId3;
>+    audio_track_3.cname = kStream2Cname;
>+    audio_track_3.set_stream_ids({});
>+    audio_track_3.ssrcs.push_back(kAudioTrack3Ssrc);
>+    audio_desc_3->AddStream(audio_track_3);
>+    desc_.AddContent(kAudioContentName3, MediaProtocolType::kRtp, audio_desc_3);
>+    EXPECT_TRUE(desc_.AddTransportInfo(TransportInfo(
>+        kAudioContentName3, TransportDescription(kUfragVoice3, kPwdVoice3))));
>+    // Make sure to create both a=msid lines.
>+    desc_.set_msid_signaling(cricket::kMsidSignalingMediaSection |
>+                             cricket::kMsidSignalingSsrcAttribute);
>+    ASSERT_TRUE(jdesc_.Initialize(desc_.Copy(), jdesc_.session_id(),
>+                                  jdesc_.session_version()));
>+  }
>+
>+  // Turns the existing reference description into a unified plan description
>+  // with one audio MediaContentDescription that contains one StreamParams with
>+  // 0 ssrcs.
>+  void MakeUnifiedPlanDescriptionNoSsrcSignaling() {
>+    desc_.RemoveContentByName(kVideoContentName);
>+    desc_.RemoveContentByName(kAudioContentName);
>+    desc_.RemoveTransportInfoByName(kVideoContentName);
>+    RemoveVideoCandidates();
>+
>+    AudioContentDescription* audio_desc = CreateAudioContentDescription();
>+    StreamParams audio_track;
>+    audio_track.id = kAudioTrackId1;
>+    audio_track.set_stream_ids({kStreamId1});
>+    audio_desc->AddStream(audio_track);
>+    desc_.AddContent(kAudioContentName, MediaProtocolType::kRtp, audio_desc);
>+
>+    // Enable signaling a=msid lines.
>+    desc_.set_msid_signaling(cricket::kMsidSignalingMediaSection);
>+    ASSERT_TRUE(jdesc_.Initialize(desc_.Copy(), jdesc_.session_id(),
>+                                  jdesc_.session_version()));
>+  }
>+
>   // Creates a video content description with no streams, and some default
>   // configuration.
>   VideoContentDescription* CreateVideoContentDescription() {
>@@ -1155,8 +1231,7 @@ class WebRtcSdpTest : public testing::Test {
>   }
> 
>   template <class MCD>
>-  void CompareMediaContentDescription(const MCD* cd1,
>-                                      const MCD* cd2) {
>+  void CompareMediaContentDescription(const MCD* cd1, const MCD* cd2) {
>     // type
>     EXPECT_EQ(cd1->type(), cd1->type());
> 
>@@ -1175,7 +1250,7 @@ class WebRtcSdpTest : public testing::Test {
>       ADD_FAILURE();
>       return;
>     }
>-    for (size_t i = 0; i< cd1->cryptos().size(); ++i) {
>+    for (size_t i = 0; i < cd1->cryptos().size(); ++i) {
>       const CryptoParams c1 = cd1->cryptos().at(i);
>       const CryptoParams c2 = cd2->cryptos().at(i);
>       EXPECT_TRUE(c1.Matches(c2));
>@@ -1186,13 +1261,13 @@ class WebRtcSdpTest : public testing::Test {
>     // protocol
>     // Use an equivalence class here, for old and new versions of the
>     // protocol description.
>-    if (cd1->protocol() == cricket::kMediaProtocolDtlsSctp
>-        || cd1->protocol() == cricket::kMediaProtocolUdpDtlsSctp
>-        || cd1->protocol() == cricket::kMediaProtocolTcpDtlsSctp) {
>+    if (cd1->protocol() == cricket::kMediaProtocolDtlsSctp ||
>+        cd1->protocol() == cricket::kMediaProtocolUdpDtlsSctp ||
>+        cd1->protocol() == cricket::kMediaProtocolTcpDtlsSctp) {
>       const bool cd2_is_also_dtls_sctp =
>-        cd2->protocol() == cricket::kMediaProtocolDtlsSctp
>-        || cd2->protocol() == cricket::kMediaProtocolUdpDtlsSctp
>-        || cd2->protocol() == cricket::kMediaProtocolTcpDtlsSctp;
>+          cd2->protocol() == cricket::kMediaProtocolDtlsSctp ||
>+          cd2->protocol() == cricket::kMediaProtocolUdpDtlsSctp ||
>+          cd2->protocol() == cricket::kMediaProtocolTcpDtlsSctp;
>       EXPECT_TRUE(cd2_is_also_dtls_sctp);
>     } else {
>       EXPECT_EQ(cd1->protocol(), cd2->protocol());
>@@ -1210,7 +1285,7 @@ class WebRtcSdpTest : public testing::Test {
>     // extmap
>     ASSERT_EQ(cd1->rtp_header_extensions().size(),
>               cd2->rtp_header_extensions().size());
>-    for (size_t i = 0; i< cd1->rtp_header_extensions().size(); ++i) {
>+    for (size_t i = 0; i < cd1->rtp_header_extensions().size(); ++i) {
>       const RtpExtension ext1 = cd1->rtp_header_extensions().at(i);
>       const RtpExtension ext2 = cd2->rtp_header_extensions().at(i);
>       EXPECT_EQ(ext1.uri, ext2.uri);
>@@ -1232,7 +1307,7 @@ class WebRtcSdpTest : public testing::Test {
>       ADD_FAILURE();
>       return;
>     }
>-    for (size_t i = 0 ; i < desc1.contents().size(); ++i) {
>+    for (size_t i = 0; i < desc1.contents().size(); ++i) {
>       const cricket::ContentInfo& c1 = desc1.contents().at(i);
>       const cricket::ContentInfo& c2 = desc2.contents().at(i);
>       // ContentInfo properties.
>@@ -1307,8 +1382,9 @@ class WebRtcSdpTest : public testing::Test {
>       EXPECT_EQ(transport1.content_name, transport2.content_name);
>       EXPECT_EQ(transport1.description.ice_ufrag,
>                 transport2.description.ice_ufrag);
>-      EXPECT_EQ(transport1.description.ice_pwd,
>-                transport2.description.ice_pwd);
>+      EXPECT_EQ(transport1.description.ice_pwd, transport2.description.ice_pwd);
>+      EXPECT_EQ(transport1.description.ice_mode,
>+                transport2.description.ice_mode);
>       if (transport1.description.identity_fingerprint) {
>         EXPECT_EQ(*transport1.description.identity_fingerprint,
>                   *transport2.description.identity_fingerprint);
>@@ -1324,9 +1400,8 @@ class WebRtcSdpTest : public testing::Test {
>     EXPECT_EQ(desc1.msid_supported(), desc2.msid_supported());
>   }
> 
>-  bool CompareSessionDescription(
>-      const JsepSessionDescription& desc1,
>-      const JsepSessionDescription& desc2) {
>+  bool CompareSessionDescription(const JsepSessionDescription& desc1,
>+                                 const JsepSessionDescription& desc2) {
>     EXPECT_EQ(desc1.session_id(), desc2.session_id());
>     EXPECT_EQ(desc1.session_version(), desc2.session_version());
>     CompareSessionDescription(*desc1.description(), *desc2.description());
>@@ -1357,16 +1432,18 @@ class WebRtcSdpTest : public testing::Test {
>     const char ice_ufragx[] = "a=xice-ufrag";
>     const char ice_pwd[] = "a=ice-pwd";
>     const char ice_pwdx[] = "a=xice-pwd";
>-    rtc::replace_substrs(ice_ufrag, strlen(ice_ufrag),
>-        ice_ufragx, strlen(ice_ufragx), sdp);
>-    rtc::replace_substrs(ice_pwd, strlen(ice_pwd),
>-        ice_pwdx, strlen(ice_pwdx), sdp);
>+    rtc::replace_substrs(ice_ufrag, strlen(ice_ufrag), ice_ufragx,
>+                         strlen(ice_ufragx), sdp);
>+    rtc::replace_substrs(ice_pwd, strlen(ice_pwd), ice_pwdx, strlen(ice_pwdx),
>+                         sdp);
>     return true;
>   }
> 
>   // Update the candidates in |jdesc| to use the given |ufrag| and |pwd|.
>-  bool UpdateCandidateUfragPwd(JsepSessionDescription* jdesc, int mline_index,
>-      const std::string& ufrag, const std::string& pwd) {
>+  bool UpdateCandidateUfragPwd(JsepSessionDescription* jdesc,
>+                               int mline_index,
>+                               const std::string& ufrag,
>+                               const std::string& pwd) {
>     std::string content_name;
>     if (mline_index == 0) {
>       content_name = kAudioContentName;
>@@ -1375,8 +1452,8 @@ class WebRtcSdpTest : public testing::Test {
>     } else {
>       RTC_NOTREACHED();
>     }
>-    TransportInfo transport_info(
>-        content_name, TransportDescription(ufrag, pwd));
>+    TransportInfo transport_info(content_name,
>+                                 TransportDescription(ufrag, pwd));
>     SessionDescription* desc =
>         const_cast<SessionDescription*>(jdesc->description());
>     desc->RemoveTransportInfoByName(content_name);
>@@ -1385,10 +1462,8 @@ class WebRtcSdpTest : public testing::Test {
>       const IceCandidateCollection* cc = jdesc_.candidates(i);
>       for (size_t j = 0; j < cc->count(); ++j) {
>         if (cc->at(j)->sdp_mline_index() == mline_index) {
>-          const_cast<Candidate&>(cc->at(j)->candidate()).set_username(
>-              ufrag);
>-          const_cast<Candidate&>(cc->at(j)->candidate()).set_password(
>-              pwd);
>+          const_cast<Candidate&>(cc->at(j)->candidate()).set_username(ufrag);
>+          const_cast<Candidate&>(cc->at(j)->candidate()).set_password(pwd);
>         }
>       }
>     }
>@@ -1420,9 +1495,8 @@ class WebRtcSdpTest : public testing::Test {
>   void AddFingerprint() {
>     desc_.RemoveTransportInfoByName(kAudioContentName);
>     desc_.RemoveTransportInfoByName(kVideoContentName);
>-    rtc::SSLFingerprint fingerprint(rtc::DIGEST_SHA_1,
>-                                          kIdentityDigest,
>-                                          sizeof(kIdentityDigest));
>+    rtc::SSLFingerprint fingerprint(rtc::DIGEST_SHA_1, kIdentityDigest,
>+                                    sizeof(kIdentityDigest));
>     EXPECT_TRUE(desc_.AddTransportInfo(TransportInfo(
>         kAudioContentName,
>         TransportDescription(std::vector<std::string>(), kUfragVoice, kPwdVoice,
>@@ -1453,18 +1527,41 @@ class WebRtcSdpTest : public testing::Test {
>     video_desc_->set_cryptos(std::vector<CryptoParams>());
>   }
> 
>+  // Removes everything in StreamParams from the session description that is
>+  // used for a=ssrc lines.
>+  void RemoveSsrcSignalingFromStreamParams() {
>+    for (cricket::ContentInfo content_info : jdesc_.description()->contents()) {
>+      // With Unified Plan there should be one StreamParams per m= section.
>+      StreamParams& stream =
>+          content_info.media_description()->mutable_streams()[0];
>+      stream.ssrcs.clear();
>+      stream.ssrc_groups.clear();
>+      stream.cname.clear();
>+    }
>+  }
>+
>+  // Removes all a=ssrc lines from the SDP string.
>+  void RemoveSsrcLinesFromSdpString(std::string* sdp_string) {
>+    const char kAttributeSsrc[] = "a=ssrc";
>+    while (sdp_string->find(kAttributeSsrc) != std::string::npos) {
>+      size_t pos_ssrc_attribute = sdp_string->find(kAttributeSsrc);
>+      size_t beg_line_pos = sdp_string->rfind('\n', pos_ssrc_attribute);
>+      size_t end_line_pos = sdp_string->find('\n', pos_ssrc_attribute);
>+      sdp_string->erase(beg_line_pos, end_line_pos - beg_line_pos);
>+    }
>+  }
>+
>   bool TestSerializeDirection(RtpTransceiverDirection direction) {
>     audio_desc_->set_direction(direction);
>     video_desc_->set_direction(direction);
>     std::string new_sdp = kSdpFullString;
>     ReplaceDirection(direction, &new_sdp);
> 
>-    if (!jdesc_.Initialize(desc_.Copy(),
>-                           jdesc_.session_id(),
>+    if (!jdesc_.Initialize(desc_.Copy(), jdesc_.session_id(),
>                            jdesc_.session_version())) {
>       return false;
>     }
>-    std::string message = webrtc::SdpSerialize(jdesc_, false);
>+    std::string message = webrtc::SdpSerialize(jdesc_);
>     EXPECT_EQ(new_sdp, message);
>     return true;
>   }
>@@ -1489,7 +1586,7 @@ class WebRtcSdpTest : public testing::Test {
> 
>     JsepSessionDescription jdesc_no_candidates(kDummyType);
>     MakeDescriptionWithoutCandidates(&jdesc_no_candidates);
>-    std::string message = webrtc::SdpSerialize(jdesc_no_candidates, false);
>+    std::string message = webrtc::SdpSerialize(jdesc_no_candidates);
>     EXPECT_EQ(new_sdp, message);
>     return true;
>   }
>@@ -1517,12 +1614,12 @@ class WebRtcSdpTest : public testing::Test {
>     StreamParams data_stream;
>     data_stream.id = kDataChannelMsid;
>     data_stream.cname = kDataChannelCname;
>-    data_stream.sync_label = kDataChannelLabel;
>+    data_stream.set_stream_ids({kDataChannelLabel});
>     data_stream.ssrcs.push_back(kDataChannelSsrc);
>     data_desc_->AddStream(data_stream);
>-    data_desc_->AddCrypto(CryptoParams(
>-        1, "AES_CM_128_HMAC_SHA1_80",
>-        "inline:FvLcvU2P3ZWmQxgPAgcDu7Zl9vftYElFOjEzhWs5", ""));
>+    data_desc_->AddCrypto(
>+        CryptoParams(1, "AES_CM_128_HMAC_SHA1_80",
>+                     "inline:FvLcvU2P3ZWmQxgPAgcDu7Zl9vftYElFOjEzhWs5", ""));
>     data_desc_->set_protocol(cricket::kMediaProtocolSavpf);
>     desc_.AddContent(kDataContentName, MediaProtocolType::kRtp, data.release());
>     EXPECT_TRUE(desc_.AddTransportInfo(TransportInfo(
>@@ -1538,8 +1635,7 @@ class WebRtcSdpTest : public testing::Test {
> 
>     audio_desc_->set_direction(direction);
>     video_desc_->set_direction(direction);
>-    if (!jdesc_.Initialize(desc_.Copy(),
>-                           jdesc_.session_id(),
>+    if (!jdesc_.Initialize(desc_.Copy(), jdesc_.session_id(),
>                            jdesc_.session_version())) {
>       return false;
>     }
>@@ -1574,12 +1670,12 @@ class WebRtcSdpTest : public testing::Test {
>     return true;
>   }
> 
>-  void TestDeserializeExtmap(bool session_level, bool media_level,
>-      bool encrypted) {
>+  void TestDeserializeExtmap(bool session_level,
>+                             bool media_level,
>+                             bool encrypted) {
>     AddExtmap(encrypted);
>     JsepSessionDescription new_jdesc(SdpType::kOffer);
>-    ASSERT_TRUE(new_jdesc.Initialize(desc_.Copy(),
>-                                     jdesc_.session_id(),
>+    ASSERT_TRUE(new_jdesc.Initialize(desc_.Copy(), jdesc_.session_id(),
>                                      jdesc_.session_version()));
>     JsepSessionDescription jdesc_with_extmap(SdpType::kOffer);
>     std::string sdp_with_extmap = kSdpString;
>@@ -1603,8 +1699,8 @@ class WebRtcSdpTest : public testing::Test {
>     // media level.
>     if (session_level && media_level) {
>       SdpParseError error;
>-      EXPECT_FALSE(webrtc::SdpDeserialize(sdp_with_extmap,
>-                   &jdesc_with_extmap, &error));
>+      EXPECT_FALSE(
>+          webrtc::SdpDeserialize(sdp_with_extmap, &jdesc_with_extmap, &error));
>       EXPECT_NE(std::string::npos, error.description.find("a=extmap"));
>     } else {
>       EXPECT_TRUE(SdpDeserialize(sdp_with_extmap, &jdesc_with_extmap));
>@@ -1613,10 +1709,11 @@ class WebRtcSdpTest : public testing::Test {
>   }
> 
>   void VerifyCodecParameter(const cricket::CodecParameterMap& params,
>-      const std::string& name, int expected_value) {
>+                            const std::string& name,
>+                            int expected_value) {
>     cricket::CodecParameterMap::const_iterator found = params.find(name);
>     ASSERT_TRUE(found != params.end());
>-    EXPECT_EQ(found->second, rtc::ToString<int>(expected_value));
>+    EXPECT_EQ(found->second, rtc::ToString(expected_value));
>   }
> 
>   void TestDeserializeCodecParams(const CodecParams& params,
>@@ -1714,9 +1811,9 @@ class WebRtcSdpTest : public testing::Test {
>         "a=rtcp-fb:101 goog-remb\r\n";
>     std::ostringstream os;
>     os << sdp_session_and_audio;
>-    os << "a=rtcp-fb:" << (use_wildcard ? "*" : "111") <<  " nack\r\n";
>+    os << "a=rtcp-fb:" << (use_wildcard ? "*" : "111") << " nack\r\n";
>     os << sdp_video;
>-    os << "a=rtcp-fb:" << (use_wildcard ? "*" : "101") <<  " ccm fir\r\n";
>+    os << "a=rtcp-fb:" << (use_wildcard ? "*" : "101") << " ccm fir\r\n";
>     std::string sdp = os.str();
>     // Deserialize
>     SdpParseError error;
>@@ -1727,9 +1824,8 @@ class WebRtcSdpTest : public testing::Test {
>     ASSERT_FALSE(acd->codecs().empty());
>     cricket::AudioCodec opus = acd->codecs()[0];
>     EXPECT_EQ(111, opus.id);
>-    EXPECT_TRUE(opus.HasFeedbackParam(
>-        cricket::FeedbackParam(cricket::kRtcpFbParamNack,
>-                               cricket::kParamValueEmpty)));
>+    EXPECT_TRUE(opus.HasFeedbackParam(cricket::FeedbackParam(
>+        cricket::kRtcpFbParamNack, cricket::kParamValueEmpty)));
> 
>     const VideoContentDescription* vcd =
>         GetFirstVideoContentDescription(jdesc_output->description());
>@@ -1739,18 +1835,14 @@ class WebRtcSdpTest : public testing::Test {
>     EXPECT_STREQ(webrtc::JsepSessionDescription::kDefaultVideoCodecName,
>                  vp8.name.c_str());
>     EXPECT_EQ(101, vp8.id);
>-    EXPECT_TRUE(vp8.HasFeedbackParam(
>-        cricket::FeedbackParam(cricket::kRtcpFbParamNack,
>-                               cricket::kParamValueEmpty)));
>-    EXPECT_TRUE(vp8.HasFeedbackParam(
>-        cricket::FeedbackParam(cricket::kRtcpFbParamNack,
>-                               cricket::kRtcpFbNackParamPli)));
>-    EXPECT_TRUE(vp8.HasFeedbackParam(
>-        cricket::FeedbackParam(cricket::kRtcpFbParamRemb,
>-                               cricket::kParamValueEmpty)));
>-    EXPECT_TRUE(vp8.HasFeedbackParam(
>-        cricket::FeedbackParam(cricket::kRtcpFbParamCcm,
>-                               cricket::kRtcpFbCcmParamFir)));
>+    EXPECT_TRUE(vp8.HasFeedbackParam(cricket::FeedbackParam(
>+        cricket::kRtcpFbParamNack, cricket::kParamValueEmpty)));
>+    EXPECT_TRUE(vp8.HasFeedbackParam(cricket::FeedbackParam(
>+        cricket::kRtcpFbParamNack, cricket::kRtcpFbNackParamPli)));
>+    EXPECT_TRUE(vp8.HasFeedbackParam(cricket::FeedbackParam(
>+        cricket::kRtcpFbParamRemb, cricket::kParamValueEmpty)));
>+    EXPECT_TRUE(vp8.HasFeedbackParam(cricket::FeedbackParam(
>+        cricket::kRtcpFbParamCcm, cricket::kRtcpFbCcmParamFir)));
>   }
> 
>   // Two SDP messages can mean the same thing but be different strings, e.g.
>@@ -1759,9 +1851,8 @@ class WebRtcSdpTest : public testing::Test {
>   // no order. If deserializer has already been tested, serializing then
>   // deserializing and comparing JsepSessionDescription will test
>   // the serializer sufficiently.
>-  void TestSerialize(const JsepSessionDescription& jdesc,
>-                     bool unified_plan_sdp) {
>-    std::string message = webrtc::SdpSerialize(jdesc, unified_plan_sdp);
>+  void TestSerialize(const JsepSessionDescription& jdesc) {
>+    std::string message = webrtc::SdpSerialize(jdesc);
>     JsepSessionDescription jdesc_output_des(kDummyType);
>     SdpParseError error;
>     EXPECT_TRUE(webrtc::SdpDeserialize(message, &jdesc_output_des, &error));
>@@ -1806,13 +1897,13 @@ void TestMismatch(const std::string& string1, const std::string& string2) {
> 
> TEST_F(WebRtcSdpTest, SerializeSessionDescription) {
>   // SessionDescription with desc and candidates.
>-  std::string message = webrtc::SdpSerialize(jdesc_, false);
>+  std::string message = webrtc::SdpSerialize(jdesc_);
>   TestMismatch(std::string(kSdpFullString), message);
> }
> 
> TEST_F(WebRtcSdpTest, SerializeSessionDescriptionEmpty) {
>   JsepSessionDescription jdesc_empty(kDummyType);
>-  EXPECT_EQ("", webrtc::SdpSerialize(jdesc_empty, false));
>+  EXPECT_EQ("", webrtc::SdpSerialize(jdesc_empty));
> }
> 
> // This tests serialization of SDP with a=crypto and a=fingerprint, as would be
>@@ -1821,13 +1912,11 @@ TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithFingerprint) {
>   AddFingerprint();
>   JsepSessionDescription jdesc_with_fingerprint(kDummyType);
>   MakeDescriptionWithoutCandidates(&jdesc_with_fingerprint);
>-  std::string message = webrtc::SdpSerialize(jdesc_with_fingerprint, false);
>+  std::string message = webrtc::SdpSerialize(jdesc_with_fingerprint);
> 
>   std::string sdp_with_fingerprint = kSdpString;
>-  InjectAfter(kAttributeIcePwdVoice,
>-              kFingerprint, &sdp_with_fingerprint);
>-  InjectAfter(kAttributeIcePwdVideo,
>-              kFingerprint, &sdp_with_fingerprint);
>+  InjectAfter(kAttributeIcePwdVoice, kFingerprint, &sdp_with_fingerprint);
>+  InjectAfter(kAttributeIcePwdVideo, kFingerprint, &sdp_with_fingerprint);
> 
>   EXPECT_EQ(sdp_with_fingerprint, message);
> }
>@@ -1839,15 +1928,13 @@ TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithFingerprintNoCryptos) {
>   RemoveCryptos();
>   JsepSessionDescription jdesc_with_fingerprint(kDummyType);
>   MakeDescriptionWithoutCandidates(&jdesc_with_fingerprint);
>-  std::string message = webrtc::SdpSerialize(jdesc_with_fingerprint, false);
>+  std::string message = webrtc::SdpSerialize(jdesc_with_fingerprint);
> 
>   std::string sdp_with_fingerprint = kSdpString;
>   Replace(kAttributeCryptoVoice, "", &sdp_with_fingerprint);
>   Replace(kAttributeCryptoVideo, "", &sdp_with_fingerprint);
>-  InjectAfter(kAttributeIcePwdVoice,
>-              kFingerprint, &sdp_with_fingerprint);
>-  InjectAfter(kAttributeIcePwdVideo,
>-              kFingerprint, &sdp_with_fingerprint);
>+  InjectAfter(kAttributeIcePwdVoice, kFingerprint, &sdp_with_fingerprint);
>+  InjectAfter(kAttributeIcePwdVideo, kFingerprint, &sdp_with_fingerprint);
> 
>   EXPECT_EQ(sdp_with_fingerprint, message);
> }
>@@ -1856,7 +1943,7 @@ TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithoutCandidates) {
>   // JsepSessionDescription with desc but without candidates.
>   JsepSessionDescription jdesc_no_candidates(kDummyType);
>   MakeDescriptionWithoutCandidates(&jdesc_no_candidates);
>-  std::string message = webrtc::SdpSerialize(jdesc_no_candidates, false);
>+  std::string message = webrtc::SdpSerialize(jdesc_no_candidates);
>   EXPECT_EQ(std::string(kSdpString), message);
> }
> 
>@@ -1865,10 +1952,9 @@ TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithBundle) {
>   group.AddContentName(kAudioContentName);
>   group.AddContentName(kVideoContentName);
>   desc_.AddGroup(group);
>-  ASSERT_TRUE(jdesc_.Initialize(desc_.Copy(),
>-                                jdesc_.session_id(),
>+  ASSERT_TRUE(jdesc_.Initialize(desc_.Copy(), jdesc_.session_id(),
>                                 jdesc_.session_version()));
>-  std::string message = webrtc::SdpSerialize(jdesc_, false);
>+  std::string message = webrtc::SdpSerialize(jdesc_);
>   std::string sdp_with_bundle = kSdpFullString;
>   InjectAfter(kSessionTime,
>               "a=group:BUNDLE audio_content_name video_content_name\r\n",
>@@ -1881,16 +1967,13 @@ TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithBandwidth) {
>   vcd->set_bandwidth(100 * 1000);
>   AudioContentDescription* acd = GetFirstAudioContentDescription(&desc_);
>   acd->set_bandwidth(50 * 1000);
>-  ASSERT_TRUE(jdesc_.Initialize(desc_.Copy(),
>-                                jdesc_.session_id(),
>+  ASSERT_TRUE(jdesc_.Initialize(desc_.Copy(), jdesc_.session_id(),
>                                 jdesc_.session_version()));
>-  std::string message = webrtc::SdpSerialize(jdesc_, false);
>+  std::string message = webrtc::SdpSerialize(jdesc_);
>   std::string sdp_with_bandwidth = kSdpFullString;
>-  InjectAfter("c=IN IP4 74.125.224.39\r\n",
>-              "b=AS:100\r\n",
>+  InjectAfter("c=IN IP4 74.125.224.39\r\n", "b=AS:100\r\n",
>               &sdp_with_bandwidth);
>-  InjectAfter("c=IN IP4 74.125.127.126\r\n",
>-              "b=AS:50\r\n",
>+  InjectAfter("c=IN IP4 74.125.127.126\r\n", "b=AS:50\r\n",
>               &sdp_with_bandwidth);
>   EXPECT_EQ(sdp_with_bandwidth, message);
> }
>@@ -1904,16 +1987,13 @@ TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithIceOptions) {
>   transport_options.push_back(kIceOption2);
>   transport_options.push_back(kIceOption3);
>   AddIceOptions(kVideoContentName, transport_options);
>-  ASSERT_TRUE(jdesc_.Initialize(desc_.Copy(),
>-                                jdesc_.session_id(),
>+  ASSERT_TRUE(jdesc_.Initialize(desc_.Copy(), jdesc_.session_id(),
>                                 jdesc_.session_version()));
>-  std::string message = webrtc::SdpSerialize(jdesc_, false);
>+  std::string message = webrtc::SdpSerialize(jdesc_);
>   std::string sdp_with_ice_options = kSdpFullString;
>-  InjectAfter(kAttributeIcePwdVoice,
>-              "a=ice-options:iceoption1 iceoption3\r\n",
>+  InjectAfter(kAttributeIcePwdVoice, "a=ice-options:iceoption1 iceoption3\r\n",
>               &sdp_with_ice_options);
>-  InjectAfter(kAttributeIcePwdVideo,
>-              "a=ice-options:iceoption2 iceoption3\r\n",
>+  InjectAfter(kAttributeIcePwdVideo, "a=ice-options:iceoption2 iceoption3\r\n",
>               &sdp_with_ice_options);
>   EXPECT_EQ(sdp_with_ice_options, message);
> }
>@@ -1947,7 +2027,7 @@ TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithRtpDataChannel) {
>   JsepSessionDescription jsep_desc(kDummyType);
> 
>   MakeDescriptionWithoutCandidates(&jsep_desc);
>-  std::string message = webrtc::SdpSerialize(jsep_desc, false);
>+  std::string message = webrtc::SdpSerialize(jsep_desc);
> 
>   std::string expected_sdp = kSdpString;
>   expected_sdp.append(kSdpRtpDataChannelString);
>@@ -1960,7 +2040,7 @@ TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithSctpDataChannel) {
>   JsepSessionDescription jsep_desc(kDummyType);
> 
>   MakeDescriptionWithoutCandidates(&jsep_desc);
>-  std::string message = webrtc::SdpSerialize(jsep_desc, false);
>+  std::string message = webrtc::SdpSerialize(jsep_desc);
> 
>   std::string expected_sdp = kSdpString;
>   expected_sdp.append(kSdpSctpDataChannelString);
>@@ -1983,7 +2063,7 @@ TEST_F(WebRtcSdpTest, SerializeWithSctpDataChannelAndNewPort) {
>   codec.SetParam(cricket::kCodecParamPort, kNewPort);
>   dcdesc->AddOrReplaceCodec(codec);
> 
>-  std::string message = webrtc::SdpSerialize(jsep_desc, false);
>+  std::string message = webrtc::SdpSerialize(jsep_desc);
> 
>   std::string expected_sdp = kSdpString;
>   expected_sdp.append(kSdpSctpDataChannelString);
>@@ -1991,11 +2071,10 @@ TEST_F(WebRtcSdpTest, SerializeWithSctpDataChannelAndNewPort) {
>   char default_portstr[16];
>   char new_portstr[16];
>   rtc::sprintfn(default_portstr, sizeof(default_portstr), "%d",
>-                      kDefaultSctpPort);
>+                kDefaultSctpPort);
>   rtc::sprintfn(new_portstr, sizeof(new_portstr), "%d", kNewPort);
>-  rtc::replace_substrs(default_portstr, strlen(default_portstr),
>-                             new_portstr, strlen(new_portstr),
>-                             &expected_sdp);
>+  rtc::replace_substrs(default_portstr, strlen(default_portstr), new_portstr,
>+                       strlen(new_portstr), &expected_sdp);
> 
>   EXPECT_EQ(expected_sdp, message);
> }
>@@ -2003,16 +2082,15 @@ TEST_F(WebRtcSdpTest, SerializeWithSctpDataChannelAndNewPort) {
> TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithDataChannelAndBandwidth) {
>   JsepSessionDescription jsep_desc(kDummyType);
>   AddRtpDataChannel();
>-  data_desc_->set_bandwidth(100*1000);
>+  data_desc_->set_bandwidth(100 * 1000);
>   MakeDescriptionWithoutCandidates(&jsep_desc);
>-  std::string message = webrtc::SdpSerialize(jsep_desc, false);
>+  std::string message = webrtc::SdpSerialize(jsep_desc);
> 
>   std::string expected_sdp = kSdpString;
>   expected_sdp.append(kSdpRtpDataChannelString);
>   // Serializing data content shouldn't ignore bandwidth settings.
>   InjectAfter("m=application 9 RTP/SAVPF 101\r\nc=IN IP4 0.0.0.0\r\n",
>-              "b=AS:100\r\n",
>-              &expected_sdp);
>+              "b=AS:100\r\n", &expected_sdp);
>   EXPECT_EQ(expected_sdp, message);
> }
> 
>@@ -2021,13 +2099,11 @@ TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithExtmap) {
>   AddExtmap(encrypted);
>   JsepSessionDescription desc_with_extmap(kDummyType);
>   MakeDescriptionWithoutCandidates(&desc_with_extmap);
>-  std::string message = webrtc::SdpSerialize(desc_with_extmap, false);
>+  std::string message = webrtc::SdpSerialize(desc_with_extmap);
> 
>   std::string sdp_with_extmap = kSdpString;
>-  InjectAfter("a=mid:audio_content_name\r\n",
>-              kExtmap, &sdp_with_extmap);
>-  InjectAfter("a=mid:video_content_name\r\n",
>-              kExtmap, &sdp_with_extmap);
>+  InjectAfter("a=mid:audio_content_name\r\n", kExtmap, &sdp_with_extmap);
>+  InjectAfter("a=mid:video_content_name\r\n", kExtmap, &sdp_with_extmap);
> 
>   EXPECT_EQ(sdp_with_extmap, message);
> }
>@@ -2036,9 +2112,9 @@ TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithExtmapEncrypted) {
>   bool encrypted = true;
>   AddExtmap(encrypted);
>   JsepSessionDescription desc_with_extmap(kDummyType);
>-  ASSERT_TRUE(desc_with_extmap.Initialize(desc_.Copy(),
>-                                          kSessionId, kSessionVersion));
>-  TestSerialize(desc_with_extmap, false);
>+  ASSERT_TRUE(
>+      desc_with_extmap.Initialize(desc_.Copy(), kSessionId, kSessionVersion));
>+  TestSerialize(desc_with_extmap);
> }
> 
> TEST_F(WebRtcSdpTest, SerializeCandidates) {
>@@ -2066,6 +2142,16 @@ TEST_F(WebRtcSdpTest, SerializeCandidates) {
>             message);
> }
> 
>+TEST_F(WebRtcSdpTest, SerializeHostnameCandidate) {
>+  rtc::SocketAddress address("a.test", 1234);
>+  cricket::Candidate candidate(
>+      cricket::ICE_CANDIDATE_COMPONENT_RTP, "udp", address, kCandidatePriority,
>+      "", "", LOCAL_PORT_TYPE, kCandidateGeneration, kCandidateFoundation1);
>+  JsepIceCandidate jcandidate(std::string("audio_content_name"), 0, candidate);
>+  std::string message = webrtc::SdpSerializeCandidate(jcandidate);
>+  EXPECT_EQ(std::string(kRawHostnameCandidate), message);
>+}
>+
> // TODO(mallinath) : Enable this test once WebRTCSdp capable of parsing
> // RFC 6544.
> TEST_F(WebRtcSdpTest, SerializeTcpCandidates) {
>@@ -2090,7 +2176,7 @@ TEST_F(WebRtcSdpTest, SerializeSessionDescriptionWithH264) {
> 
>   jdesc_.Initialize(desc_.Copy(), kSessionId, kSessionVersion);
> 
>-  std::string message = webrtc::SdpSerialize(jdesc_, false);
>+  std::string message = webrtc::SdpSerialize(jdesc_);
>   size_t after_pt = message.find(" H264/90000");
>   ASSERT_NE(after_pt, std::string::npos);
>   size_t before_pt = message.rfind("a=rtpmap:", after_pt);
>@@ -2123,11 +2209,11 @@ TEST_F(WebRtcSdpTest, DeserializeSessionDescription) {
> TEST_F(WebRtcSdpTest, DeserializeSessionDescriptionWithoutMline) {
>   JsepSessionDescription jdesc(kDummyType);
>   const char kSdpWithoutMline[] =
>-    "v=0\r\n"
>-    "o=- 18446744069414584320 18446462598732840960 IN IP4 127.0.0.1\r\n"
>-    "s=-\r\n"
>-    "t=0 0\r\n"
>-    "a=msid-semantic: WMS local_stream_1 local_stream_2\r\n";
>+      "v=0\r\n"
>+      "o=- 18446744069414584320 18446462598732840960 IN IP4 127.0.0.1\r\n"
>+      "s=-\r\n"
>+      "t=0 0\r\n"
>+      "a=msid-semantic: WMS local_stream_1 local_stream_2\r\n";
>   // Deserialize
>   EXPECT_TRUE(SdpDeserialize(kSdpWithoutMline, &jdesc));
>   EXPECT_EQ(0u, jdesc.description()->contents().size());
>@@ -2146,8 +2232,8 @@ TEST_F(WebRtcSdpTest, DeserializeSessionDescriptionWithoutCarriageReturn) {
> TEST_F(WebRtcSdpTest, DeserializeSessionDescriptionWithoutCandidates) {
>   // SessionDescription with desc but without candidates.
>   JsepSessionDescription jdesc_no_candidates(kDummyType);
>-  ASSERT_TRUE(jdesc_no_candidates.Initialize(desc_.Copy(),
>-                                             kSessionId, kSessionVersion));
>+  ASSERT_TRUE(jdesc_no_candidates.Initialize(desc_.Copy(), kSessionId,
>+                                             kSessionVersion));
>   JsepSessionDescription new_jdesc(kDummyType);
>   EXPECT_TRUE(SdpDeserialize(kSdpString, &new_jdesc));
>   EXPECT_TRUE(CompareSessionDescription(jdesc_no_candidates, new_jdesc));
>@@ -2202,8 +2288,7 @@ TEST_F(WebRtcSdpTest, DeserializeSessionDescriptionWithoutRtpmapButWithFmtp) {
>   EXPECT_EQ("G729", g729.name);
>   EXPECT_EQ(8000, g729.clockrate);
>   EXPECT_EQ(18, g729.id);
>-  cricket::CodecParameterMap::iterator found =
>-      g729.params.find("annexb");
>+  cricket::CodecParameterMap::iterator found = g729.params.find("annexb");
>   ASSERT_TRUE(found != g729.params.end());
>   EXPECT_EQ(found->second, "yes");
> 
>@@ -2218,8 +2303,7 @@ TEST_F(WebRtcSdpTest, DeserializeJsepSessionDescriptionWithFingerprint) {
>   // Add a DTLS a=fingerprint attribute to our session description.
>   AddFingerprint();
>   JsepSessionDescription new_jdesc(kDummyType);
>-  ASSERT_TRUE(new_jdesc.Initialize(desc_.Copy(),
>-                                   jdesc_.session_id(),
>+  ASSERT_TRUE(new_jdesc.Initialize(desc_.Copy(), jdesc_.session_id(),
>                                    jdesc_.session_version()));
> 
>   JsepSessionDescription jdesc_with_fingerprint(kDummyType);
>@@ -2241,8 +2325,7 @@ TEST_F(WebRtcSdpTest, DeserializeSessionDescriptionWithBundle) {
>   group.AddContentName(kAudioContentName);
>   group.AddContentName(kVideoContentName);
>   desc_.AddGroup(group);
>-  ASSERT_TRUE(jdesc_.Initialize(desc_.Copy(),
>-                                jdesc_.session_id(),
>+  ASSERT_TRUE(jdesc_.Initialize(desc_.Copy(), jdesc_.session_id(),
>                                 jdesc_.session_version()));
>   EXPECT_TRUE(CompareSessionDescription(jdesc_, jdesc_with_bundle));
> }
>@@ -2250,20 +2333,16 @@ TEST_F(WebRtcSdpTest, DeserializeSessionDescriptionWithBundle) {
> TEST_F(WebRtcSdpTest, DeserializeSessionDescriptionWithBandwidth) {
>   JsepSessionDescription jdesc_with_bandwidth(kDummyType);
>   std::string sdp_with_bandwidth = kSdpFullString;
>-  InjectAfter("a=mid:video_content_name\r\na=sendrecv\r\n",
>-              "b=AS:100\r\n",
>+  InjectAfter("a=mid:video_content_name\r\na=sendrecv\r\n", "b=AS:100\r\n",
>               &sdp_with_bandwidth);
>-  InjectAfter("a=mid:audio_content_name\r\na=sendrecv\r\n",
>-              "b=AS:50\r\n",
>+  InjectAfter("a=mid:audio_content_name\r\na=sendrecv\r\n", "b=AS:50\r\n",
>               &sdp_with_bandwidth);
>-  EXPECT_TRUE(
>-      SdpDeserialize(sdp_with_bandwidth, &jdesc_with_bandwidth));
>+  EXPECT_TRUE(SdpDeserialize(sdp_with_bandwidth, &jdesc_with_bandwidth));
>   VideoContentDescription* vcd = GetFirstVideoContentDescription(&desc_);
>   vcd->set_bandwidth(100 * 1000);
>   AudioContentDescription* acd = GetFirstAudioContentDescription(&desc_);
>   acd->set_bandwidth(50 * 1000);
>-  ASSERT_TRUE(jdesc_.Initialize(desc_.Copy(),
>-                                jdesc_.session_id(),
>+  ASSERT_TRUE(jdesc_.Initialize(desc_.Copy(), jdesc_.session_id(),
>                                 jdesc_.session_version()));
>   EXPECT_TRUE(CompareSessionDescription(jdesc_, jdesc_with_bandwidth));
> }
>@@ -2271,14 +2350,11 @@ TEST_F(WebRtcSdpTest, DeserializeSessionDescriptionWithBandwidth) {
> TEST_F(WebRtcSdpTest, DeserializeSessionDescriptionWithIceOptions) {
>   JsepSessionDescription jdesc_with_ice_options(kDummyType);
>   std::string sdp_with_ice_options = kSdpFullString;
>-  InjectAfter(kSessionTime,
>-              "a=ice-options:iceoption3\r\n",
>+  InjectAfter(kSessionTime, "a=ice-options:iceoption3\r\n",
>               &sdp_with_ice_options);
>-  InjectAfter(kAttributeIcePwdVoice,
>-              "a=ice-options:iceoption1\r\n",
>+  InjectAfter(kAttributeIcePwdVoice, "a=ice-options:iceoption1\r\n",
>               &sdp_with_ice_options);
>-  InjectAfter(kAttributeIcePwdVideo,
>-              "a=ice-options:iceoption2\r\n",
>+  InjectAfter(kAttributeIcePwdVideo, "a=ice-options:iceoption2\r\n",
>               &sdp_with_ice_options);
>   EXPECT_TRUE(SdpDeserialize(sdp_with_ice_options, &jdesc_with_ice_options));
>   std::vector<std::string> transport_options;
>@@ -2289,8 +2365,7 @@ TEST_F(WebRtcSdpTest, DeserializeSessionDescriptionWithIceOptions) {
>   transport_options.push_back(kIceOption3);
>   transport_options.push_back(kIceOption2);
>   AddIceOptions(kVideoContentName, transport_options);
>-  ASSERT_TRUE(jdesc_.Initialize(desc_.Copy(),
>-                                jdesc_.session_id(),
>+  ASSERT_TRUE(jdesc_.Initialize(desc_.Copy(), jdesc_.session_id(),
>                                 jdesc_.session_version()));
>   EXPECT_TRUE(CompareSessionDescription(jdesc_, jdesc_with_ice_options));
> }
>@@ -2302,18 +2377,19 @@ TEST_F(WebRtcSdpTest, DeserializeSessionDescriptionWithUfragPwd) {
>   EXPECT_TRUE(RemoveCandidateUfragPwd(&sdp_with_ufrag_pwd));
>   // Add session level ufrag and pwd
>   InjectAfter(kSessionTime,
>-      "a=ice-pwd:session+level+icepwd\r\n"
>-      "a=ice-ufrag:session+level+iceufrag\r\n",
>-      &sdp_with_ufrag_pwd);
>+              "a=ice-pwd:session+level+icepwd\r\n"
>+              "a=ice-ufrag:session+level+iceufrag\r\n",
>+              &sdp_with_ufrag_pwd);
>   // Add media level ufrag and pwd for audio
>-  InjectAfter("a=mid:audio_content_name\r\n",
>+  InjectAfter(
>+      "a=mid:audio_content_name\r\n",
>       "a=ice-pwd:media+level+icepwd\r\na=ice-ufrag:media+level+iceufrag\r\n",
>       &sdp_with_ufrag_pwd);
>   // Update the candidate ufrag and pwd to the expected ones.
>-  EXPECT_TRUE(UpdateCandidateUfragPwd(&jdesc_, 0,
>-      "media+level+iceufrag", "media+level+icepwd"));
>-  EXPECT_TRUE(UpdateCandidateUfragPwd(&jdesc_, 1,
>-      "session+level+iceufrag", "session+level+icepwd"));
>+  EXPECT_TRUE(UpdateCandidateUfragPwd(&jdesc_, 0, "media+level+iceufrag",
>+                                      "media+level+icepwd"));
>+  EXPECT_TRUE(UpdateCandidateUfragPwd(&jdesc_, 1, "session+level+iceufrag",
>+                                      "session+level+icepwd"));
>   EXPECT_TRUE(SdpDeserialize(sdp_with_ufrag_pwd, &jdesc_with_ufrag_pwd));
>   EXPECT_TRUE(CompareSessionDescription(jdesc_, jdesc_with_ufrag_pwd));
> }
>@@ -2400,8 +2476,8 @@ TEST_F(WebRtcSdpTest, DeserializeCandidate) {
>                       kCandidateFoundation1);
>   std::unique_ptr<IceCandidateInterface> jcandidate_template(
>       new JsepIceCandidate(std::string("audio_content_name"), 0, candidate));
>-  EXPECT_TRUE(jcandidate.candidate().IsEquivalent(
>-                    jcandidate_template->candidate()));
>+  EXPECT_TRUE(
>+      jcandidate.candidate().IsEquivalent(jcandidate_template->candidate()));
>   sdp = kSdpTcpPassiveCandidate;
>   EXPECT_TRUE(SdpDeserializeCandidate(sdp, &jcandidate));
>   sdp = kSdpTcpSOCandidate;
>@@ -2439,12 +2515,15 @@ TEST_F(WebRtcSdpTest, DeserializeRawCandidateAttribute) {
> 
>   // Candidate line with IPV6 address.
>   EXPECT_TRUE(SdpDeserializeCandidate(kRawIPV6Candidate, &jcandidate));
>+
>+  // Candidate line with hostname address.
>+  EXPECT_TRUE(SdpDeserializeCandidate(kRawHostnameCandidate, &jcandidate));
> }
> 
> // This test verifies that the deserialization of an invalid candidate string
> // fails.
> TEST_F(WebRtcSdpTest, DeserializeInvalidCandidiate) {
>-    JsepIceCandidate jcandidate(kDummyMid, kDummyIndex);
>+  JsepIceCandidate jcandidate(kDummyMid, kDummyIndex);
> 
>   std::string candidate_attribute = kRawCandidate;
>   candidate_attribute.replace(0, 1, "x");
>@@ -2492,14 +2571,14 @@ TEST_F(WebRtcSdpTest, DeserializeSdpWithSctpDataChannels) {
>   EXPECT_TRUE(CompareSessionDescription(jdesc, jdesc_output));
> 
>   // Verify with UDP/DTLS/SCTP.
>-  sdp_with_data.replace(sdp_with_data.find(kDtlsSctp),
>-                        strlen(kDtlsSctp), kUdpDtlsSctp);
>+  sdp_with_data.replace(sdp_with_data.find(kDtlsSctp), strlen(kDtlsSctp),
>+                        kUdpDtlsSctp);
>   EXPECT_TRUE(SdpDeserialize(sdp_with_data, &jdesc_output));
>   EXPECT_TRUE(CompareSessionDescription(jdesc, jdesc_output));
> 
>   // Verify with TCP/DTLS/SCTP.
>-  sdp_with_data.replace(sdp_with_data.find(kUdpDtlsSctp),
>-                        strlen(kUdpDtlsSctp), kTcpDtlsSctp);
>+  sdp_with_data.replace(sdp_with_data.find(kUdpDtlsSctp), strlen(kUdpDtlsSctp),
>+                        kTcpDtlsSctp);
>   EXPECT_TRUE(SdpDeserialize(sdp_with_data, &jdesc_output));
>   EXPECT_TRUE(CompareSessionDescription(jdesc, jdesc_output));
> }
>@@ -2519,14 +2598,14 @@ TEST_F(WebRtcSdpTest, DeserializeSdpWithSctpDataChannelsWithSctpPort) {
>   EXPECT_TRUE(CompareSessionDescription(jdesc, jdesc_output));
> 
>   // Verify with UDP/DTLS/SCTP.
>-  sdp_with_data.replace(sdp_with_data.find(kDtlsSctp),
>-                        strlen(kDtlsSctp), kUdpDtlsSctp);
>+  sdp_with_data.replace(sdp_with_data.find(kDtlsSctp), strlen(kDtlsSctp),
>+                        kUdpDtlsSctp);
>   EXPECT_TRUE(SdpDeserialize(sdp_with_data, &jdesc_output));
>   EXPECT_TRUE(CompareSessionDescription(jdesc, jdesc_output));
> 
>   // Verify with TCP/DTLS/SCTP.
>-  sdp_with_data.replace(sdp_with_data.find(kUdpDtlsSctp),
>-                        strlen(kUdpDtlsSctp), kTcpDtlsSctp);
>+  sdp_with_data.replace(sdp_with_data.find(kUdpDtlsSctp), strlen(kUdpDtlsSctp),
>+                        kTcpDtlsSctp);
>   EXPECT_TRUE(SdpDeserialize(sdp_with_data, &jdesc_output));
>   EXPECT_TRUE(CompareSessionDescription(jdesc, jdesc_output));
> }
>@@ -2546,14 +2625,14 @@ TEST_F(WebRtcSdpTest, DeserializeSdpWithSctpDataChannelsWithSctpColonPort) {
>   EXPECT_TRUE(CompareSessionDescription(jdesc, jdesc_output));
> 
>   // Verify with UDP/DTLS/SCTP.
>-  sdp_with_data.replace(sdp_with_data.find(kDtlsSctp),
>-                        strlen(kDtlsSctp), kUdpDtlsSctp);
>+  sdp_with_data.replace(sdp_with_data.find(kDtlsSctp), strlen(kDtlsSctp),
>+                        kUdpDtlsSctp);
>   EXPECT_TRUE(SdpDeserialize(sdp_with_data, &jdesc_output));
>   EXPECT_TRUE(CompareSessionDescription(jdesc, jdesc_output));
> 
>   // Verify with TCP/DTLS/SCTP.
>-  sdp_with_data.replace(sdp_with_data.find(kUdpDtlsSctp),
>-                        strlen(kUdpDtlsSctp), kTcpDtlsSctp);
>+  sdp_with_data.replace(sdp_with_data.find(kUdpDtlsSctp), strlen(kUdpDtlsSctp),
>+                        kTcpDtlsSctp);
>   EXPECT_TRUE(SdpDeserialize(sdp_with_data, &jdesc_output));
>   EXPECT_TRUE(CompareSessionDescription(jdesc, jdesc_output));
> }
>@@ -2656,8 +2735,7 @@ TEST_F(WebRtcSdpTest, DeserializeSdpWithRtpDataChannelsAndBandwidth) {
>   // See: https://code.google.com/p/chromium/issues/detail?id=280726
>   std::string sdp_with_bandwidth = kSdpString;
>   sdp_with_bandwidth.append(kSdpRtpDataChannelString);
>-  InjectAfter("a=mid:data_content_name\r\n",
>-              "b=AS:100\r\n",
>+  InjectAfter("a=mid:data_content_name\r\n", "b=AS:100\r\n",
>               &sdp_with_bandwidth);
>   JsepSessionDescription jdesc_with_bandwidth(kDummyType);
> 
>@@ -2674,8 +2752,7 @@ TEST_F(WebRtcSdpTest, DeserializeSdpWithSctpDataChannelsAndBandwidth) {
> 
>   std::string sdp_with_bandwidth = kSdpString;
>   sdp_with_bandwidth.append(kSdpSctpDataChannelString);
>-  InjectAfter("a=mid:data_content_name\r\n",
>-              "b=AS:100\r\n",
>+  InjectAfter("a=mid:data_content_name\r\n", "b=AS:100\r\n",
>               &sdp_with_bandwidth);
>   JsepSessionDescription jdesc_with_bandwidth(kDummyType);
> 
>@@ -2685,24 +2762,21 @@ TEST_F(WebRtcSdpTest, DeserializeSdpWithSctpDataChannelsAndBandwidth) {
>   EXPECT_TRUE(CompareSessionDescription(jdesc, jdesc_with_bandwidth));
> }
> 
>-class WebRtcSdpExtmapTest
>-  : public WebRtcSdpTest, public testing::WithParamInterface<bool> {
>-};
>+class WebRtcSdpExtmapTest : public WebRtcSdpTest,
>+                            public testing::WithParamInterface<bool> {};
> 
> TEST_P(WebRtcSdpExtmapTest,
>-    DeserializeSessionDescriptionWithSessionLevelExtmap) {
>+       DeserializeSessionDescriptionWithSessionLevelExtmap) {
>   bool encrypted = GetParam();
>   TestDeserializeExtmap(true, false, encrypted);
> }
> 
>-TEST_P(WebRtcSdpExtmapTest,
>-    DeserializeSessionDescriptionWithMediaLevelExtmap) {
>+TEST_P(WebRtcSdpExtmapTest, DeserializeSessionDescriptionWithMediaLevelExtmap) {
>   bool encrypted = GetParam();
>   TestDeserializeExtmap(false, true, encrypted);
> }
> 
>-TEST_P(WebRtcSdpExtmapTest,
>-    DeserializeSessionDescriptionWithInvalidExtmap) {
>+TEST_P(WebRtcSdpExtmapTest, DeserializeSessionDescriptionWithInvalidExtmap) {
>   bool encrypted = GetParam();
>   TestDeserializeExtmap(true, true, encrypted);
> }
>@@ -2774,7 +2848,7 @@ TEST_F(WebRtcSdpTest, SerializeSdpWithConferenceFlag) {
>   // We tested deserialization already above, so just test that if we serialize
>   // and deserialize the flag doesn't disappear.
>   EXPECT_TRUE(SdpDeserialize(kSdpConferenceString, &jdesc));
>-  std::string reserialized = webrtc::SdpSerialize(jdesc, false);
>+  std::string reserialized = webrtc::SdpSerialize(jdesc);
>   EXPECT_TRUE(SdpDeserialize(reserialized, &jdesc));
> 
>   // Verify.
>@@ -2800,14 +2874,18 @@ TEST_F(WebRtcSdpTest, DeserializeBrokenSdp) {
>   const char kSdpEmptyLine[] = "";
>   const char kSdpMissingValue[] = "a=";
> 
>-  const char kSdpBrokenFingerprint[] = "a=fingerprint:sha-1 "
>+  const char kSdpBrokenFingerprint[] =
>+      "a=fingerprint:sha-1 "
>       "4AAD:B9:B1:3F:82:18:3B:54:02:12:DF:3E:5D:49:6B:19:E5:7C:AB";
>-  const char kSdpExtraField[] = "a=fingerprint:sha-1 "
>+  const char kSdpExtraField[] =
>+      "a=fingerprint:sha-1 "
>       "4A:AD:B9:B1:3F:82:18:3B:54:02:12:DF:3E:5D:49:6B:19:E5:7C:AB XXX";
>-  const char kSdpMissingSpace[] = "a=fingerprint:sha-1"
>+  const char kSdpMissingSpace[] =
>+      "a=fingerprint:sha-1"
>       "4A:AD:B9:B1:3F:82:18:3B:54:02:12:DF:3E:5D:49:6B:19:E5:7C:AB";
>   // MD5 is not allowed in fingerprints.
>-  const char kSdpMd5[] = "a=fingerprint:md5 "
>+  const char kSdpMd5[] =
>+      "a=fingerprint:md5 "
>       "4A:AD:B9:B1:3F:82:18:3B:54:02:12:DF:3E:5D:49:6B";
> 
>   // Broken session description
>@@ -2863,8 +2941,7 @@ TEST_F(WebRtcSdpTest, DeserializeSdpWithInvalidAttributeValue) {
> 
>   // bandwidth
>   ExpectParseFailureWithNewLines("a=mid:video_content_name\r\n",
>-                                 "b=AS:badvalue\r\n",
>-                                 "b=AS:badvalue");
>+                                 "b=AS:badvalue\r\n", "b=AS:badvalue");
>   // rtcp-fb
>   ExpectParseFailureWithNewLines("a=mid:video_content_name\r\n",
>                                  "a=rtcp-fb:badvalue nack\r\n",
>@@ -2884,8 +2961,8 @@ TEST_F(WebRtcSdpTest, DeserializeSdpWithReorderedPltypes) {
>       "s=-\r\n"
>       "t=0 0\r\n"
>       "m=audio 9 RTP/SAVPF 104 103\r\n"  // Pl type 104 preferred.
>-      "a=rtpmap:111 opus/48000/2\r\n"  // Pltype 111 listed before 103 and 104
>-                                       // in the map.
>+      "a=rtpmap:111 opus/48000/2\r\n"    // Pltype 111 listed before 103 and 104
>+                                         // in the map.
>       "a=rtpmap:103 ISAC/16000\r\n"  // Pltype 103 listed before 104 in the map.
>       "a=rtpmap:104 ISAC/32000\r\n";
> 
>@@ -2912,21 +2989,21 @@ TEST_F(WebRtcSdpTest, DeserializeSerializeCodecParams) {
>   params.useinband = 1;
>   params.maxaveragebitrate = 128000;
>   TestDeserializeCodecParams(params, &jdesc_output);
>-  TestSerialize(jdesc_output, false);
>+  TestSerialize(jdesc_output);
> }
> 
> TEST_F(WebRtcSdpTest, DeserializeSerializeRtcpFb) {
>   const bool kUseWildcard = false;
>   JsepSessionDescription jdesc_output(kDummyType);
>   TestDeserializeRtcpFb(&jdesc_output, kUseWildcard);
>-  TestSerialize(jdesc_output, false);
>+  TestSerialize(jdesc_output);
> }
> 
> TEST_F(WebRtcSdpTest, DeserializeSerializeRtcpFbWildcard) {
>   const bool kUseWildcard = true;
>   JsepSessionDescription jdesc_output(kDummyType);
>   TestDeserializeRtcpFb(&jdesc_output, kUseWildcard);
>-  TestSerialize(jdesc_output, false);
>+  TestSerialize(jdesc_output);
> }
> 
> TEST_F(WebRtcSdpTest, DeserializeVideoFmtp) {
>@@ -3010,8 +3087,8 @@ TEST_F(WebRtcSdpTest, DeserializeVideoFmtpWithSpace) {
> 
>   // Deserialize
>   SdpParseError error;
>-  EXPECT_TRUE(webrtc::SdpDeserialize(kSdpWithFmtpString, &jdesc_output,
>-                                     &error));
>+  EXPECT_TRUE(
>+      webrtc::SdpDeserialize(kSdpWithFmtpString, &jdesc_output, &error));
> 
>   const VideoContentDescription* vcd =
>       GetFirstVideoContentDescription(jdesc_output.description());
>@@ -3036,10 +3113,9 @@ TEST_F(WebRtcSdpTest, SerializeAudioFmtpWithUnknownParameter) {
>   codecs[0].params["unknown-future-parameter"] = "SomeFutureValue";
>   acd->set_codecs(codecs);
> 
>-  ASSERT_TRUE(jdesc_.Initialize(desc_.Copy(),
>-                                jdesc_.session_id(),
>+  ASSERT_TRUE(jdesc_.Initialize(desc_.Copy(), jdesc_.session_id(),
>                                 jdesc_.session_version()));
>-  std::string message = webrtc::SdpSerialize(jdesc_, false);
>+  std::string message = webrtc::SdpSerialize(jdesc_);
>   std::string sdp_with_fmtp = kSdpFullString;
>   InjectAfter("a=rtpmap:111 opus/48000/2\r\n",
>               "a=fmtp:111 unknown-future-parameter=SomeFutureValue\r\n",
>@@ -3054,13 +3130,11 @@ TEST_F(WebRtcSdpTest, SerializeAudioFmtpWithKnownFmtpParameter) {
>   codecs[0].params["stereo"] = "1";
>   acd->set_codecs(codecs);
> 
>-  ASSERT_TRUE(jdesc_.Initialize(desc_.Copy(),
>-                                jdesc_.session_id(),
>+  ASSERT_TRUE(jdesc_.Initialize(desc_.Copy(), jdesc_.session_id(),
>                                 jdesc_.session_version()));
>-  std::string message = webrtc::SdpSerialize(jdesc_, false);
>+  std::string message = webrtc::SdpSerialize(jdesc_);
>   std::string sdp_with_fmtp = kSdpFullString;
>-  InjectAfter("a=rtpmap:111 opus/48000/2\r\n",
>-              "a=fmtp:111 stereo=1\r\n",
>+  InjectAfter("a=rtpmap:111 opus/48000/2\r\n", "a=fmtp:111 stereo=1\r\n",
>               &sdp_with_fmtp);
>   EXPECT_EQ(sdp_with_fmtp, message);
> }
>@@ -3073,10 +3147,9 @@ TEST_F(WebRtcSdpTest, SerializeAudioFmtpWithPTimeAndMaxPTime) {
>   codecs[0].params["maxptime"] = "120";
>   acd->set_codecs(codecs);
> 
>-  ASSERT_TRUE(jdesc_.Initialize(desc_.Copy(),
>-                                jdesc_.session_id(),
>+  ASSERT_TRUE(jdesc_.Initialize(desc_.Copy(), jdesc_.session_id(),
>                                 jdesc_.session_version()));
>-  std::string message = webrtc::SdpSerialize(jdesc_, false);
>+  std::string message = webrtc::SdpSerialize(jdesc_);
>   std::string sdp_with_fmtp = kSdpFullString;
>   InjectAfter("a=rtpmap:104 ISAC/32000\r\n",
>               "a=maxptime:120\r\n"  // No comma here. String merging!
>@@ -3092,18 +3165,17 @@ TEST_F(WebRtcSdpTest, SerializeVideoFmtp) {
>   codecs[0].params["x-google-min-bitrate"] = "10";
>   vcd->set_codecs(codecs);
> 
>-  ASSERT_TRUE(jdesc_.Initialize(desc_.Copy(),
>-                                jdesc_.session_id(),
>+  ASSERT_TRUE(jdesc_.Initialize(desc_.Copy(), jdesc_.session_id(),
>                                 jdesc_.session_version()));
>-  std::string message = webrtc::SdpSerialize(jdesc_, false);
>+  std::string message = webrtc::SdpSerialize(jdesc_);
>   std::string sdp_with_fmtp = kSdpFullString;
>   InjectAfter("a=rtpmap:120 VP8/90000\r\n",
>-              "a=fmtp:120 x-google-min-bitrate=10\r\n",
>-              &sdp_with_fmtp);
>+              "a=fmtp:120 x-google-min-bitrate=10\r\n", &sdp_with_fmtp);
>   EXPECT_EQ(sdp_with_fmtp, message);
> }
> 
>-TEST_F(WebRtcSdpTest, DeserializeSdpWithIceLite) {
>+TEST_F(WebRtcSdpTest, DeserializeAndSerializeSdpWithIceLite) {
>+  // Deserialize the baseline description, making sure it's ICE full.
>   JsepSessionDescription jdesc_with_icelite(kDummyType);
>   std::string sdp_with_icelite = kSdpFullString;
>   EXPECT_TRUE(SdpDeserialize(sdp_with_icelite, &jdesc_with_icelite));
>@@ -3114,17 +3186,21 @@ TEST_F(WebRtcSdpTest, DeserializeSdpWithIceLite) {
>   const cricket::TransportInfo* tinfo2 =
>       desc->GetTransportInfoByName("video_content_name");
>   EXPECT_EQ(cricket::ICEMODE_FULL, tinfo2->description.ice_mode);
>-  InjectAfter(kSessionTime,
>-              "a=ice-lite\r\n",
>-              &sdp_with_icelite);
>+
>+  // Add "a=ice-lite" and deserialize, making sure it's ICE lite.
>+  InjectAfter(kSessionTime, "a=ice-lite\r\n", &sdp_with_icelite);
>   EXPECT_TRUE(SdpDeserialize(sdp_with_icelite, &jdesc_with_icelite));
>   desc = jdesc_with_icelite.description();
>   const cricket::TransportInfo* atinfo =
>       desc->GetTransportInfoByName("audio_content_name");
>   EXPECT_EQ(cricket::ICEMODE_LITE, atinfo->description.ice_mode);
>   const cricket::TransportInfo* vtinfo =
>-        desc->GetTransportInfoByName("video_content_name");
>+      desc->GetTransportInfoByName("video_content_name");
>   EXPECT_EQ(cricket::ICEMODE_LITE, vtinfo->description.ice_mode);
>+
>+  // Now that we know deserialization works, we can use TestSerialize to test
>+  // serialization.
>+  TestSerialize(jdesc_with_icelite);
> }
> 
> // Verifies that the candidates in the input SDP are parsed and serialized
>@@ -3135,7 +3211,7 @@ TEST_F(WebRtcSdpTest, RoundTripSdpWithSctpDataChannelsWithCandidates) {
>   JsepSessionDescription jdesc_output(kDummyType);
> 
>   EXPECT_TRUE(SdpDeserialize(sdp_with_data, &jdesc_output));
>-  EXPECT_EQ(sdp_with_data, webrtc::SdpSerialize(jdesc_output, false));
>+  EXPECT_EQ(sdp_with_data, webrtc::SdpSerialize(jdesc_output));
> }
> 
> TEST_F(WebRtcSdpTest, SerializeDtlsSetupAttribute) {
>@@ -3145,14 +3221,14 @@ TEST_F(WebRtcSdpTest, SerializeDtlsSetupAttribute) {
>   EXPECT_EQ(cricket::CONNECTIONROLE_NONE,
>             audio_transport_info.description.connection_role);
>   audio_transport_info.description.connection_role =
>-        cricket::CONNECTIONROLE_ACTIVE;
>+      cricket::CONNECTIONROLE_ACTIVE;
> 
>   TransportInfo video_transport_info =
>       *(desc_.GetTransportInfoByName(kVideoContentName));
>   EXPECT_EQ(cricket::CONNECTIONROLE_NONE,
>             video_transport_info.description.connection_role);
>   video_transport_info.description.connection_role =
>-        cricket::CONNECTIONROLE_ACTIVE;
>+      cricket::CONNECTIONROLE_ACTIVE;
> 
>   desc_.RemoveTransportInfoByName(kAudioContentName);
>   desc_.RemoveTransportInfoByName(kVideoContentName);
>@@ -3160,29 +3236,23 @@ TEST_F(WebRtcSdpTest, SerializeDtlsSetupAttribute) {
>   desc_.AddTransportInfo(audio_transport_info);
>   desc_.AddTransportInfo(video_transport_info);
> 
>-  ASSERT_TRUE(jdesc_.Initialize(desc_.Copy(),
>-                                jdesc_.session_id(),
>+  ASSERT_TRUE(jdesc_.Initialize(desc_.Copy(), jdesc_.session_id(),
>                                 jdesc_.session_version()));
>-  std::string message = webrtc::SdpSerialize(jdesc_, false);
>+  std::string message = webrtc::SdpSerialize(jdesc_);
>   std::string sdp_with_dtlssetup = kSdpFullString;
> 
>   // Fingerprint attribute is necessary to add DTLS setup attribute.
>-  InjectAfter(kAttributeIcePwdVoice,
>-              kFingerprint, &sdp_with_dtlssetup);
>-  InjectAfter(kAttributeIcePwdVideo,
>-              kFingerprint, &sdp_with_dtlssetup);
>+  InjectAfter(kAttributeIcePwdVoice, kFingerprint, &sdp_with_dtlssetup);
>+  InjectAfter(kAttributeIcePwdVideo, kFingerprint, &sdp_with_dtlssetup);
>   // Now adding |setup| attribute.
>-  InjectAfter(kFingerprint,
>-              "a=setup:active\r\n", &sdp_with_dtlssetup);
>+  InjectAfter(kFingerprint, "a=setup:active\r\n", &sdp_with_dtlssetup);
>   EXPECT_EQ(sdp_with_dtlssetup, message);
> }
> 
> TEST_F(WebRtcSdpTest, DeserializeDtlsSetupAttribute) {
>   JsepSessionDescription jdesc_with_dtlssetup(kDummyType);
>   std::string sdp_with_dtlssetup = kSdpFullString;
>-  InjectAfter(kSessionTime,
>-              "a=setup:actpass\r\n",
>-              &sdp_with_dtlssetup);
>+  InjectAfter(kSessionTime, "a=setup:actpass\r\n", &sdp_with_dtlssetup);
>   EXPECT_TRUE(SdpDeserialize(sdp_with_dtlssetup, &jdesc_with_dtlssetup));
>   cricket::SessionDescription* desc = jdesc_with_dtlssetup.description();
>   const cricket::TransportInfo* atinfo =
>@@ -3190,7 +3260,7 @@ TEST_F(WebRtcSdpTest, DeserializeDtlsSetupAttribute) {
>   EXPECT_EQ(cricket::CONNECTIONROLE_ACTPASS,
>             atinfo->description.connection_role);
>   const cricket::TransportInfo* vtinfo =
>-        desc->GetTransportInfoByName("video_content_name");
>+      desc->GetTransportInfoByName("video_content_name");
>   EXPECT_EQ(cricket::CONNECTIONROLE_ACTPASS,
>             vtinfo->description.connection_role);
> }
>@@ -3199,16 +3269,11 @@ TEST_F(WebRtcSdpTest, DeserializeDtlsSetupAttribute) {
> // ContentInfo in SessionDescription, and vise versa for deserialization.
> TEST_F(WebRtcSdpTest, MediaContentOrderMaintainedRoundTrip) {
>   JsepSessionDescription jdesc(kDummyType);
>-  const std::string media_content_sdps[3] = {
>-    kSdpAudioString,
>-    kSdpVideoString,
>-    kSdpSctpDataChannelString
>-  };
>-  const cricket::MediaType media_types[3] = {
>-    cricket::MEDIA_TYPE_AUDIO,
>-    cricket::MEDIA_TYPE_VIDEO,
>-    cricket::MEDIA_TYPE_DATA
>-  };
>+  const std::string media_content_sdps[3] = {kSdpAudioString, kSdpVideoString,
>+                                             kSdpSctpDataChannelString};
>+  const cricket::MediaType media_types[3] = {cricket::MEDIA_TYPE_AUDIO,
>+                                             cricket::MEDIA_TYPE_VIDEO,
>+                                             cricket::MEDIA_TYPE_DATA};
> 
>   // Verifies all 6 permutations.
>   for (size_t i = 0; i < 6; ++i) {
>@@ -3234,7 +3299,7 @@ TEST_F(WebRtcSdpTest, MediaContentOrderMaintainedRoundTrip) {
>       EXPECT_EQ(media_types[media_content_in_sdp[i]], mdesc->type());
>     }
> 
>-    std::string serialized_sdp = webrtc::SdpSerialize(jdesc, false);
>+    std::string serialized_sdp = webrtc::SdpSerialize(jdesc);
>     EXPECT_EQ(sdp_string, serialized_sdp);
>   }
> }
>@@ -3264,7 +3329,7 @@ TEST_F(WebRtcSdpTest, IgnoreBundleOnlyWithNonzeroPort) {
> 
> TEST_F(WebRtcSdpTest, SerializeBundleOnlyAttribute) {
>   MakeBundleOnlyDescription();
>-  TestSerialize(jdesc_, false);
>+  TestSerialize(jdesc_);
> }
> 
> TEST_F(WebRtcSdpTest, DeserializePlanBSessionDescription) {
>@@ -3278,36 +3343,125 @@ TEST_F(WebRtcSdpTest, DeserializePlanBSessionDescription) {
> 
> TEST_F(WebRtcSdpTest, SerializePlanBSessionDescription) {
>   MakePlanBDescription();
>-  TestSerialize(jdesc_, false);
>+  TestSerialize(jdesc_);
> }
> 
>-// Some WebRTC endpoints include the msid in both the Plan B and Unified Plan
>-// ways, to make SDP that's compatible with both Plan B and Unified Plan (to
>-// some extent). If we parse this, the Plan B msid attribute (which is more
>-// specific, since it's at the SSRC level) should take priority.
>-TEST_F(WebRtcSdpTest, DeserializePlanBSessionDescriptionWithMsid) {
>-  MakePlanBDescription();
>+TEST_F(WebRtcSdpTest, DeserializeUnifiedPlanSessionDescription) {
>+  MakeUnifiedPlanDescription();
> 
>   JsepSessionDescription deserialized_description(kDummyType);
>   EXPECT_TRUE(
>-      SdpDeserialize(kPlanBSdpFullStringWithMsid, &deserialized_description));
>+      SdpDeserialize(kUnifiedPlanSdpFullString, &deserialized_description));
> 
>   EXPECT_TRUE(CompareSessionDescription(jdesc_, deserialized_description));
> }
> 
>-TEST_F(WebRtcSdpTest, DeserializeUnifiedPlanSessionDescription) {
>+TEST_F(WebRtcSdpTest, SerializeUnifiedPlanSessionDescription) {
>   MakeUnifiedPlanDescription();
>+  TestSerialize(jdesc_);
>+}
>+
>+// This tests deserializing a Unified Plan SDP that is compatible with both
>+// Unified Plan and Plan B style SDP. It tests the case for audio/video tracks
>+// with no stream ids and multiple stream ids. For parsing this, the Unified
>+// Plan a=msid lines should take priority, because the Plan B style a=ssrc msid
>+// lines do not support multiple stream ids and no stream ids.
>+TEST_F(WebRtcSdpTest, DeserializeUnifiedPlanSessionDescriptionSpecialMsid) {
>+  MakeUnifiedPlanDescriptionMultipleStreamIds();
> 
>   JsepSessionDescription deserialized_description(kDummyType);
>-  EXPECT_TRUE(
>-      SdpDeserialize(kUnifiedPlanSdpFullString, &deserialized_description));
>+  EXPECT_TRUE(SdpDeserialize(kUnifiedPlanSdpFullStringWithSpecialMsid,
>+                             &deserialized_description));
> 
>   EXPECT_TRUE(CompareSessionDescription(jdesc_, deserialized_description));
> }
> 
>-TEST_F(WebRtcSdpTest, SerializeUnifiedPlanSessionDescription) {
>+TEST_F(WebRtcSdpTest, SerializeUnifiedPlanSessionDescriptionSpecialMsid) {
>+  MakeUnifiedPlanDescriptionMultipleStreamIds();
>+  TestSerialize(jdesc_);
>+}
>+
>+// This tests that a Unified Plan SDP with no a=ssrc lines is
>+// serialized/deserialized appropriately. In this case the
>+// MediaContentDescription will contain a StreamParams object that doesn't have
>+// any SSRCs. Vice versa, this will be created upon deserializing an SDP with no
>+// SSRC lines.
>+TEST_F(WebRtcSdpTest, DeserializeUnifiedPlanSessionDescriptionNoSsrcSignaling) {
>+  MakeUnifiedPlanDescription();
>+  RemoveSsrcSignalingFromStreamParams();
>+  std::string unified_plan_sdp_string = kUnifiedPlanSdpFullString;
>+  RemoveSsrcLinesFromSdpString(&unified_plan_sdp_string);
>+
>+  JsepSessionDescription deserialized_description(kDummyType);
>+  EXPECT_TRUE(
>+      SdpDeserialize(unified_plan_sdp_string, &deserialized_description));
>+  EXPECT_TRUE(CompareSessionDescription(jdesc_, deserialized_description));
>+}
>+
>+TEST_F(WebRtcSdpTest, SerializeUnifiedPlanSessionDescriptionNoSsrcSignaling) {
>   MakeUnifiedPlanDescription();
>-  TestSerialize(jdesc_, true);
>+  RemoveSsrcSignalingFromStreamParams();
>+
>+  TestSerialize(jdesc_);
>+}
>+
>+TEST_F(WebRtcSdpTest, EmptyDescriptionHasNoMsidSignaling) {
>+  JsepSessionDescription jsep_desc(kDummyType);
>+  ASSERT_TRUE(SdpDeserialize(kSdpSessionString, &jsep_desc));
>+  EXPECT_EQ(0, jsep_desc.description()->msid_signaling());
>+}
>+
>+TEST_F(WebRtcSdpTest, DataChannelOnlyHasNoMsidSignaling) {
>+  JsepSessionDescription jsep_desc(kDummyType);
>+  std::string sdp = kSdpSessionString;
>+  sdp += kSdpSctpDataChannelString;
>+  ASSERT_TRUE(SdpDeserialize(sdp, &jsep_desc));
>+  EXPECT_EQ(0, jsep_desc.description()->msid_signaling());
>+}
>+
>+TEST_F(WebRtcSdpTest, PlanBHasSsrcAttributeMsidSignaling) {
>+  JsepSessionDescription jsep_desc(kDummyType);
>+  ASSERT_TRUE(SdpDeserialize(kPlanBSdpFullString, &jsep_desc));
>+  EXPECT_EQ(cricket::kMsidSignalingSsrcAttribute,
>+            jsep_desc.description()->msid_signaling());
>+}
>+
>+TEST_F(WebRtcSdpTest, UnifiedPlanHasMediaSectionMsidSignaling) {
>+  JsepSessionDescription jsep_desc(kDummyType);
>+  ASSERT_TRUE(SdpDeserialize(kUnifiedPlanSdpFullString, &jsep_desc));
>+  EXPECT_EQ(cricket::kMsidSignalingMediaSection,
>+            jsep_desc.description()->msid_signaling());
>+}
>+
>+const char kMediaSectionMsidLine[] = "a=msid:local_stream_1 audio_track_id_1";
>+const char kSsrcAttributeMsidLine[] =
>+    "a=ssrc:1 msid:local_stream_1 audio_track_id_1";
>+
>+TEST_F(WebRtcSdpTest, SerializeOnlyMediaSectionMsid) {
>+  jdesc_.description()->set_msid_signaling(cricket::kMsidSignalingMediaSection);
>+  std::string sdp = webrtc::SdpSerialize(jdesc_);
>+
>+  EXPECT_NE(std::string::npos, sdp.find(kMediaSectionMsidLine));
>+  EXPECT_EQ(std::string::npos, sdp.find(kSsrcAttributeMsidLine));
>+}
>+
>+TEST_F(WebRtcSdpTest, SerializeOnlySsrcAttributeMsid) {
>+  jdesc_.description()->set_msid_signaling(
>+      cricket::kMsidSignalingSsrcAttribute);
>+  std::string sdp = webrtc::SdpSerialize(jdesc_);
>+
>+  EXPECT_EQ(std::string::npos, sdp.find(kMediaSectionMsidLine));
>+  EXPECT_NE(std::string::npos, sdp.find(kSsrcAttributeMsidLine));
>+}
>+
>+TEST_F(WebRtcSdpTest, SerializeBothMediaSectionAndSsrcAttributeMsid) {
>+  jdesc_.description()->set_msid_signaling(
>+      cricket::kMsidSignalingMediaSection |
>+      cricket::kMsidSignalingSsrcAttribute);
>+  std::string sdp = webrtc::SdpSerialize(jdesc_);
>+
>+  EXPECT_NE(std::string::npos, sdp.find(kMediaSectionMsidLine));
>+  EXPECT_NE(std::string::npos, sdp.find(kSsrcAttributeMsidLine));
> }
> 
> // Regression test for heap overflow bug:
>@@ -3406,7 +3560,7 @@ TEST_F(WebRtcSdpTest, IceCredentialsInCandidateStringIgnored) {
>       SdpDeserialize(kSdpWithIceCredentialsInCandidateString, &jdesc_output));
>   const IceCandidateCollection* candidates = jdesc_output.candidates(0);
>   ASSERT_NE(nullptr, candidates);
>-  ASSERT_EQ(1, candidates->count());
>+  ASSERT_EQ(1U, candidates->count());
>   cricket::Candidate c = candidates->at(0)->candidate();
>   EXPECT_EQ("ufrag_voice", c.username());
>   EXPECT_EQ("pwd_voice", c.password());
>@@ -3559,7 +3713,7 @@ TEST_F(WebRtcSdpTest, SerializeAndDeserializeWithConnectionAddress) {
>   JsepSessionDescription expected_jsep(kDummyType);
>   MakeDescriptionWithoutCandidates(&expected_jsep);
>   // Serialization.
>-  std::string message = webrtc::SdpSerialize(expected_jsep, false);
>+  std::string message = webrtc::SdpSerialize(expected_jsep);
>   // Deserialization.
>   JsepSessionDescription jdesc(kDummyType);
>   EXPECT_TRUE(SdpDeserialize(message, &jdesc));
>@@ -3574,3 +3728,12 @@ TEST_F(WebRtcSdpTest, SerializeAndDeserializeWithConnectionAddress) {
>   EXPECT_EQ(video_desc_->connection_address().ToString(),
>             video_desc->connection_address().ToString());
> }
>+
>+// RFC4566 says "If a session has no meaningful name, the value "s= " SHOULD be
>+// used (i.e., a single space as the session name)." So we should accept that.
>+TEST_F(WebRtcSdpTest, DeserializeEmptySessionName) {
>+  JsepSessionDescription jsep_desc(kDummyType);
>+  std::string sdp = kSdpString;
>+  Replace("s=-\r\n", "s= \r\n", &sdp);
>+  EXPECT_TRUE(SdpDeserialize(sdp, &jsep_desc));
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/webrtcsessiondescriptionfactory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/webrtcsessiondescriptionfactory.cc
>index feec1fdd41f..22551437e3d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/webrtcsessiondescriptionfactory.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/webrtcsessiondescriptionfactory.cc
>@@ -15,12 +15,11 @@
> #include <utility>
> #include <vector>
> 
>+#include "absl/memory/memory.h"
> #include "api/jsep.h"
> #include "api/jsepsessiondescription.h"
> #include "api/mediaconstraintsinterface.h"
>-#include "pc/peerconnection.h"
> #include "rtc_base/checks.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/sslidentity.h"
> 
> using cricket::MediaSessionOptions;
>@@ -68,12 +67,12 @@ enum {
> 
> struct CreateSessionDescriptionMsg : public rtc::MessageData {
>   explicit CreateSessionDescriptionMsg(
>-      webrtc::CreateSessionDescriptionObserver* observer)
>-      : observer(observer) {
>-  }
>+      webrtc::CreateSessionDescriptionObserver* observer,
>+      RTCError error_in)
>+      : observer(observer), error(std::move(error_in)) {}
> 
>   rtc::scoped_refptr<webrtc::CreateSessionDescriptionObserver> observer;
>-  std::string error;
>+  RTCError error;
>   std::unique_ptr<webrtc::SessionDescriptionInterface> description;
> };
> }  // namespace
>@@ -122,7 +121,7 @@ void WebRtcSessionDescriptionFactory::CopyCandidatesFromSessionDescription(
> WebRtcSessionDescriptionFactory::WebRtcSessionDescriptionFactory(
>     rtc::Thread* signaling_thread,
>     cricket::ChannelManager* channel_manager,
>-    PeerConnection* pc,
>+    PeerConnectionInternal* pc,
>     const std::string& session_id,
>     std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
>     const rtc::scoped_refptr<rtc::RTCCertificate>& certificate)
>@@ -171,12 +170,12 @@ WebRtcSessionDescriptionFactory::WebRtcSessionDescriptionFactory(
> 
>     rtc::KeyParams key_params = rtc::KeyParams();
>     RTC_LOG(LS_VERBOSE)
>-        << "DTLS-SRTP enabled; sending DTLS identity request (key "
>-        << "type: " << key_params.type() << ").";
>+        << "DTLS-SRTP enabled; sending DTLS identity request (key type: "
>+        << key_params.type() << ").";
> 
>     // Request certificate. This happens asynchronously, so that the caller gets
>     // a chance to connect to |SignalCertificateReady|.
>-    cert_generator_->GenerateCertificateAsync(key_params, rtc::nullopt,
>+    cert_generator_->GenerateCertificateAsync(key_params, absl::nullopt,
>                                               callback);
>   }
> }
>@@ -298,7 +297,7 @@ void WebRtcSessionDescriptionFactory::OnMessage(rtc::Message* msg) {
>     case MSG_CREATE_SESSIONDESCRIPTION_FAILED: {
>       CreateSessionDescriptionMsg* param =
>           static_cast<CreateSessionDescriptionMsg*>(msg->pdata);
>-      param->observer->OnFailure(param->error);
>+      param->observer->OnFailure(std::move(param->error));
>       delete param;
>       break;
>     }
>@@ -344,7 +343,7 @@ void WebRtcSessionDescriptionFactory::InternalCreateOffer(
>   // is created regardless if it's identical to the previous one or not.
>   // The |session_version_| is a uint64_t, the wrap around should not happen.
>   RTC_DCHECK(session_version_ + 1 > session_version_);
>-  auto offer = rtc::MakeUnique<JsepSessionDescription>(SdpType::kOffer);
>+  auto offer = absl::make_unique<JsepSessionDescription>(SdpType::kOffer);
>   if (!offer->Initialize(desc, session_id_,
>                          rtc::ToString(session_version_++))) {
>     PostCreateSessionDescriptionFailed(request.observer,
>@@ -397,7 +396,7 @@ void WebRtcSessionDescriptionFactory::InternalCreateAnswer(
>   // Get a new version number by increasing the |session_version_answer_|.
>   // The |session_version_| is a uint64_t, the wrap around should not happen.
>   RTC_DCHECK(session_version_ + 1 > session_version_);
>-  auto answer = rtc::MakeUnique<JsepSessionDescription>(SdpType::kAnswer);
>+  auto answer = absl::make_unique<JsepSessionDescription>(SdpType::kAnswer);
>   if (!answer->Initialize(desc, session_id_,
>                           rtc::ToString(session_version_++))) {
>     PostCreateSessionDescriptionFailed(request.observer,
>@@ -424,17 +423,21 @@ void WebRtcSessionDescriptionFactory::FailPendingRequests(
>   while (!create_session_description_requests_.empty()) {
>     const CreateSessionDescriptionRequest& request =
>         create_session_description_requests_.front();
>-    PostCreateSessionDescriptionFailed(request.observer,
>-        ((request.type == CreateSessionDescriptionRequest::kOffer) ?
>-            "CreateOffer" : "CreateAnswer") + reason);
>+    PostCreateSessionDescriptionFailed(
>+        request.observer,
>+        ((request.type == CreateSessionDescriptionRequest::kOffer)
>+             ? "CreateOffer"
>+             : "CreateAnswer") +
>+            reason);
>     create_session_description_requests_.pop();
>   }
> }
> 
> void WebRtcSessionDescriptionFactory::PostCreateSessionDescriptionFailed(
>-    CreateSessionDescriptionObserver* observer, const std::string& error) {
>-  CreateSessionDescriptionMsg* msg = new CreateSessionDescriptionMsg(observer);
>-  msg->error = error;
>+    CreateSessionDescriptionObserver* observer,
>+    const std::string& error) {
>+  CreateSessionDescriptionMsg* msg = new CreateSessionDescriptionMsg(
>+      observer, RTCError(RTCErrorType::INTERNAL_ERROR, std::string(error)));
>   signaling_thread_->Post(RTC_FROM_HERE, this,
>                           MSG_CREATE_SESSIONDESCRIPTION_FAILED, msg);
>   RTC_LOG(LS_ERROR) << "Create SDP failed: " << error;
>@@ -443,7 +446,8 @@ void WebRtcSessionDescriptionFactory::PostCreateSessionDescriptionFailed(
> void WebRtcSessionDescriptionFactory::PostCreateSessionDescriptionSucceeded(
>     CreateSessionDescriptionObserver* observer,
>     std::unique_ptr<SessionDescriptionInterface> description) {
>-  CreateSessionDescriptionMsg* msg = new CreateSessionDescriptionMsg(observer);
>+  CreateSessionDescriptionMsg* msg =
>+      new CreateSessionDescriptionMsg(observer, RTCError::OK());
>   msg->description = std::move(description);
>   signaling_thread_->Post(RTC_FROM_HERE, this,
>                           MSG_CREATE_SESSIONDESCRIPTION_SUCCESS, msg);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/webrtcsessiondescriptionfactory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/webrtcsessiondescriptionfactory.h
>index 12d1cd215dc..08d01228585 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/webrtcsessiondescriptionfactory.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/webrtcsessiondescriptionfactory.h
>@@ -15,25 +15,15 @@
> #include <queue>
> #include <string>
> 
>-#include "api/peerconnectioninterface.h"
> #include "p2p/base/transportdescriptionfactory.h"
> #include "pc/mediasession.h"
>+#include "pc/peerconnectioninternal.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/messagehandler.h"
> #include "rtc_base/rtccertificate.h"
> #include "rtc_base/rtccertificategenerator.h"
> 
>-namespace cricket {
>-class ChannelManager;
>-class TransportDescriptionFactory;
>-}  // namespace cricket
>-
> namespace webrtc {
>-class CreateSessionDescriptionObserver;
>-class MediaConstraintsInterface;
>-class PeerConnection;
>-class SessionDescriptionInterface;
>-class WebRtcSession;
> 
> // DTLS certificate request callback class.
> class WebRtcCertificateGeneratorCallback
>@@ -56,13 +46,10 @@ struct CreateSessionDescriptionRequest {
>     kAnswer,
>   };
> 
>-  CreateSessionDescriptionRequest(
>-      Type type,
>-      CreateSessionDescriptionObserver* observer,
>-      const cricket::MediaSessionOptions& options)
>-      : type(type),
>-        observer(observer),
>-        options(options) {}
>+  CreateSessionDescriptionRequest(Type type,
>+                                  CreateSessionDescriptionObserver* observer,
>+                                  const cricket::MediaSessionOptions& options)
>+      : type(type), observer(observer), options(options) {}
> 
>   Type type;
>   rtc::scoped_refptr<CreateSessionDescriptionObserver> observer;
>@@ -84,7 +71,7 @@ class WebRtcSessionDescriptionFactory : public rtc::MessageHandler,
>   WebRtcSessionDescriptionFactory(
>       rtc::Thread* signaling_thread,
>       cricket::ChannelManager* channel_manager,
>-      PeerConnection* pc,
>+      PeerConnectionInternal* pc,
>       const std::string& session_id,
>       std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
>       const rtc::scoped_refptr<rtc::RTCCertificate>& certificate);
>@@ -152,7 +139,7 @@ class WebRtcSessionDescriptionFactory : public rtc::MessageHandler,
>   const std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator_;
>   // TODO(jiayl): remove the dependency on peer connection once bug 2264 is
>   // fixed.
>-  PeerConnection* const pc_;
>+  PeerConnectionInternal* const pc_;
>   const std::string session_id_;
>   CertificateRequestState certificate_request_state_;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/presubmit_test.py b/Source/ThirdParty/libwebrtc/Source/webrtc/presubmit_test.py
>old mode 100644
>new mode 100755
>index 9f97df901e0..d6e512954a0
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/presubmit_test.py
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/presubmit_test.py
>@@ -86,7 +86,8 @@ class CheckNewlineAtTheEndOfProtoFilesTest(unittest.TestCase):
>     self._GenerateProtoWithoutNewlineAtTheEnd()
>     self.input_api.files = [MockFile(self.proto_file_path)]
>     errors = PRESUBMIT.CheckNewlineAtTheEndOfProtoFiles(self.input_api,
>-                                                        self.output_api)
>+                                                        self.output_api,
>+                                                        lambda x: True)
>     self.assertEqual(1, len(errors))
>     self.assertEqual(
>         'File %s must end with exactly one newline.' % self.proto_file_path,
>@@ -96,7 +97,8 @@ class CheckNewlineAtTheEndOfProtoFilesTest(unittest.TestCase):
>     self._GenerateProtoWithNewlineAtTheEnd()
>     self.input_api.files = [MockFile(self.proto_file_path)]
>     errors = PRESUBMIT.CheckNewlineAtTheEndOfProtoFiles(self.input_api,
>-                                                        self.output_api)
>+                                                        self.output_api,
>+                                                        lambda x: True)
>     self.assertEqual(0, len(errors))
> 
>   def _GenerateProtoWithNewlineAtTheEnd(self):
>@@ -235,7 +237,7 @@ class CheckNoMixingSourcesTest(unittest.TestCase):
>     self.assertTrue('bar.c' in str(errors[0]))
> 
>   def _AssertNumberOfErrorsWithSources(self, number_of_errors, sources):
>-    assert 3 == len(sources), 'This function accepts a list of 3 source files'
>+    assert len(sources) == 3, 'This function accepts a list of 3 source files'
>     self._GenerateBuildFile(textwrap.dedent("""
>       rtc_static_library("bar_foo") {
>         sources = [
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/presubmit_test_mocks.py b/Source/ThirdParty/libwebrtc/Source/webrtc/presubmit_test_mocks.py
>index 4b2b3c75228..510a553158b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/presubmit_test_mocks.py
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/presubmit_test_mocks.py
>@@ -9,6 +9,9 @@
> # This file is inspired to [1].
> # [1] - https://cs.chromium.org/chromium/src/PRESUBMIT_test_mocks.py
> 
>+import os.path
>+import re
>+
> 
> class MockInputApi(object):
>   """Mock class for the InputApi class.
>@@ -20,11 +23,23 @@ class MockInputApi(object):
>   def __init__(self):
>     self.change = MockChange([], [])
>     self.files = []
>+    self.presubmit_local_path = os.path.dirname(__file__)
> 
>   def AffectedSourceFiles(self, file_filter=None):
>+    return self.AffectedFiles(file_filter=file_filter)
>+
>+  def AffectedFiles(self, file_filter=None, include_deletes=False):
>     # pylint: disable=unused-argument
>     return self.files
> 
>+  @classmethod
>+  def FilterSourceFile(cls, affected_file, white_list=(), black_list=()):
>+    # pylint: disable=unused-argument
>+    return True
>+
>+  def PresubmitLocalPath(self):
>+    return self.presubmit_local_path
>+
>   def ReadFile(self, affected_file, mode='rU'):
>     filename = affected_file.AbsoluteLocalPath()
>     for f in self.files:
>@@ -64,13 +79,20 @@ class MockChange(object):
>   current change.
>   """
> 
>-  def __init__(self, changed_files, bugs_from_description):
>+  def __init__(self, changed_files, bugs_from_description, tags=None):
>     self._changed_files = changed_files
>     self._bugs_from_description = bugs_from_description
>+    self.tags = dict() if not tags else tags
> 
>   def BugsFromDescription(self):
>     return self._bugs_from_description
> 
>+  def __getattr__(self, attr):
>+    """Return tags directly as attributes on the object."""
>+    if not re.match(r"^[A-Z_]*$", attr):
>+      raise AttributeError(self, attr)
>+    return self.tags.get(attr)
>+
> 
> class MockFile(object):
>   """Mock class for the File class.
>@@ -79,11 +101,30 @@ class MockFile(object):
>   MockInputApi for presubmit unittests.
>   """
> 
>-  def __init__(self, local_path):
>+  def __init__(self, local_path, new_contents=None, old_contents=None,
>+      action='A'):
>+    if new_contents is None:
>+      new_contents = ["Data"]
>     self._local_path = local_path
>+    self._new_contents = new_contents
>+    self._changed_contents = [(i + 1, l) for i, l in enumerate(new_contents)]
>+    self._action = action
>+    self._old_contents = old_contents
>+
>+  def Action(self):
>+    return self._action
>+
>+  def ChangedContents(self):
>+    return self._changed_contents
>+
>+  def NewContents(self):
>+    return self._new_contents
> 
>   def LocalPath(self):
>     return self._local_path
> 
>   def AbsoluteLocalPath(self):
>     return self._local_path
>+
>+  def OldContents(self):
>+    return self._old_contents
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pylintrc b/Source/ThirdParty/libwebrtc/Source/webrtc/pylintrc
>index 6d40f783ca8..9809ebe998f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pylintrc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pylintrc
>@@ -20,6 +20,7 @@ disable=
>   I0010,
>   I0011,
>   W0232,
>+  C0413,
>   bad-continuation,
>   broad-except,
>   duplicate-code,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/.gitignore b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/.gitignore
>index 334fc452d60..1f729ae4579 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/.gitignore
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/.gitignore
>@@ -19,3 +19,5 @@
> **/*.BIT*
> **/*.INP
> **/*.OUT*
>+
>+!**/*.sha1
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/.gitignore.rej b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/.gitignore.rej
>deleted file mode 100644
>index a8e70ca352c..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/.gitignore.rej
>+++ /dev/null
>@@ -1,23 +0,0 @@
>-diff a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/.gitignore b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/.gitignore	(rejected hunks)
>-@@ -0,0 +1,21 @@
>-+**/*.aecdump
>-+**/*.bin
>-+**/*.byte
>-+**/*.chn
>-+**/*.dat
>-+**/*.frames
>-+**/*.gai
>-+**/*.jpg
>-+**/*.pb
>-+**/*.pcap
>-+**/*.pcm
>-+**/*.rtp
>-+**/*.rtpdump
>-+**/*.rx
>-+**/*.txt
>-+**/*.wav
>-+**/*.y4m
>-+**/*.yuv
>-+**/*.BIT*
>-+**/*.INP
>-+**/*.OUT*
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/FourPeople_1280x720_30.yuv.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/FourPeople_1280x720_30.yuv.sha1
>new file mode 100644
>index 00000000000..dcfd1971b70
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/FourPeople_1280x720_30.yuv.sha1
>@@ -0,0 +1 @@
>+3baa38557757bb770f7cbeaa390e3a8a1e0c254c
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F00.BIT20.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F00.BIT20.sha1
>new file mode 100644
>index 00000000000..dcbaa095162
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F00.BIT20.sha1
>@@ -0,0 +1 @@
>+27e30ed5e0d449db1cd729c382e18455ad3834fe
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F00.BIT30.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F00.BIT30.sha1
>new file mode 100644
>index 00000000000..874ce9db192
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F00.BIT30.sha1
>@@ -0,0 +1 @@
>+6bafbd8667365939c327eef627fa0f94a75e5fa9
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F00.OUT20.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F00.OUT20.sha1
>new file mode 100644
>index 00000000000..9b9f4bdd46a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F00.OUT20.sha1
>@@ -0,0 +1 @@
>+bb80d3c48c42bb14c8e613a8576cb77283cda3a7
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F00.OUT30.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F00.OUT30.sha1
>new file mode 100644
>index 00000000000..cd62333757f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F00.OUT30.sha1
>@@ -0,0 +1 @@
>+450b538234feff02f35b3e57fbeea5d93a41abe0
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F00_tlm10.OUT20.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F00_tlm10.OUT20.sha1
>new file mode 100644
>index 00000000000..5cb8e52cb25
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F00_tlm10.OUT20.sha1
>@@ -0,0 +1 @@
>+96fb5327ff7a1fe87bd4512773ce7347b4d72888
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F00_tlm10.OUT30.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F00_tlm10.OUT30.sha1
>new file mode 100644
>index 00000000000..1470013b72a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F00_tlm10.OUT30.sha1
>@@ -0,0 +1 @@
>+ea527e8e61241ea73265abba5765793511c42291
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F01.BIT20.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F01.BIT20.sha1
>new file mode 100644
>index 00000000000..7a267240c7f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F01.BIT20.sha1
>@@ -0,0 +1 @@
>+a6230cfa6786ba3d81510aabcaa68bd9b1610d1a
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F01.BIT30.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F01.BIT30.sha1
>new file mode 100644
>index 00000000000..e5a0e5a3563
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F01.BIT30.sha1
>@@ -0,0 +1 @@
>+4f706d94506e0a4dc56f63940caa588e2531a455
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F01.OUT20.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F01.OUT20.sha1
>new file mode 100644
>index 00000000000..db6100bc735
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F01.OUT20.sha1
>@@ -0,0 +1 @@
>+1ffcead0db4293f5f36abfdc7cd959d6733358d4
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F01.OUT30.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F01.OUT30.sha1
>new file mode 100644
>index 00000000000..2c8e141a067
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F01.OUT30.sha1
>@@ -0,0 +1 @@
>+37b5e29f294c3612d4425775b422d7dc9b346965
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F01_tlm10.OUT20.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F01_tlm10.OUT20.sha1
>new file mode 100644
>index 00000000000..d33eb7e1c74
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F01_tlm10.OUT20.sha1
>@@ -0,0 +1 @@
>+a5e8c268936d7c8d03edd708c675254474aed944
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F01_tlm10.OUT30.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F01_tlm10.OUT30.sha1
>new file mode 100644
>index 00000000000..953656fe01c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F01_tlm10.OUT30.sha1
>@@ -0,0 +1 @@
>+9893504bdaa560a4ddb67a99e6701e8d4244896a
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F02.BIT20.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F02.BIT20.sha1
>new file mode 100644
>index 00000000000..98f5fc5d13f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F02.BIT20.sha1
>@@ -0,0 +1 @@
>+ea44732065b09eec558af1957da21c9061c19c08
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F02.BIT30.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F02.BIT30.sha1
>new file mode 100644
>index 00000000000..6e645690f33
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F02.BIT30.sha1
>@@ -0,0 +1 @@
>+40a5e2850e9e838429911cc3dbb1ff6c9bf38387
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F02.OUT20.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F02.OUT20.sha1
>new file mode 100644
>index 00000000000..8f9fb5a3346
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F02.OUT20.sha1
>@@ -0,0 +1 @@
>+13bb10aca91ddd8af71751944e18243735bdda98
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F02.OUT30.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F02.OUT30.sha1
>new file mode 100644
>index 00000000000..5427662ae7c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F02.OUT30.sha1
>@@ -0,0 +1 @@
>+21764cf8bb5e635e842eaf92403b22ce04732805
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F02_tlm10.OUT20.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F02_tlm10.OUT20.sha1
>new file mode 100644
>index 00000000000..b8a52534b3a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F02_tlm10.OUT20.sha1
>@@ -0,0 +1 @@
>+5e0053a0de7b4708402fcc0a1b5ae024815938a3
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F02_tlm10.OUT30.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F02_tlm10.OUT30.sha1
>new file mode 100644
>index 00000000000..d3bbc9866d1
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F02_tlm10.OUT30.sha1
>@@ -0,0 +1 @@
>+d57986d3dc65c1cadc41419a77ed1472b73050b4
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F03.BIT20.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F03.BIT20.sha1
>new file mode 100644
>index 00000000000..0ccdc7d90f1
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F03.BIT20.sha1
>@@ -0,0 +1 @@
>+c145a1a211103145fbb21cf1c78ab90abce83228
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F03.BIT30.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F03.BIT30.sha1
>new file mode 100644
>index 00000000000..565c4c67c1d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F03.BIT30.sha1
>@@ -0,0 +1 @@
>+2ccd6fe240e41906c9f06e05cbde486c1f2bebf7
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F03.OUT20.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F03.OUT20.sha1
>new file mode 100644
>index 00000000000..210e4f69a06
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F03.OUT20.sha1
>@@ -0,0 +1 @@
>+db4e9d9e09b55c3fe5536628c1f2f07f4bad068b
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F03.OUT30.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F03.OUT30.sha1
>new file mode 100644
>index 00000000000..c8a733ce68b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F03.OUT30.sha1
>@@ -0,0 +1 @@
>+f48b05dce8cb0370cda4b93da45b4edc3fdd42d0
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F04.BIT20.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F04.BIT20.sha1
>new file mode 100644
>index 00000000000..a304fa41811
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F04.BIT20.sha1
>@@ -0,0 +1 @@
>+8cc1f82ee0d1b7df39840895d3e99756920b85ad
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F04.BIT30.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F04.BIT30.sha1
>new file mode 100644
>index 00000000000..c6cbe13afb4
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F04.BIT30.sha1
>@@ -0,0 +1 @@
>+85ca3d9a5ca61c1bb9d798ba6d3551945ab7667d
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F04.OUT20.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F04.OUT20.sha1
>new file mode 100644
>index 00000000000..fd27f041e09
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F04.OUT20.sha1
>@@ -0,0 +1 @@
>+985e5ffa0b98718afadfb1f02b5595fb24a9476c
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F04.OUT30.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F04.OUT30.sha1
>new file mode 100644
>index 00000000000..3ccec4effef
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F04.OUT30.sha1
>@@ -0,0 +1 @@
>+8b035fb0f9ce3bdc2ae6c983e232c257a515a781
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F05.BIT20.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F05.BIT20.sha1
>new file mode 100644
>index 00000000000..39d82e6f521
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F05.BIT20.sha1
>@@ -0,0 +1 @@
>+9069fecc3cd13bf901781af4ece8b374b5285984
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F05.BIT30.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F05.BIT30.sha1
>new file mode 100644
>index 00000000000..513c9362a89
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F05.BIT30.sha1
>@@ -0,0 +1 @@
>+44a04e133c15e013f6b4424877528a4445e28bef
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F05.OUT20.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F05.OUT20.sha1
>new file mode 100644
>index 00000000000..df514526521
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F05.OUT20.sha1
>@@ -0,0 +1 @@
>+260f51f8320d92d3c8b834c1f38879baa12a7a5a
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F05.OUT30.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F05.OUT30.sha1
>new file mode 100644
>index 00000000000..bcc318c96f4
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F05.OUT30.sha1
>@@ -0,0 +1 @@
>+72abb4c1d84a4cff1efed4d3e01280d839a9f691
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F06.BIT20.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F06.BIT20.sha1
>new file mode 100644
>index 00000000000..6b6e3dfb684
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F06.BIT20.sha1
>@@ -0,0 +1 @@
>+b6530247acdfd7ae7b3fd80c677d71045351ad07
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F06.BIT30.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F06.BIT30.sha1
>new file mode 100644
>index 00000000000..f9f7975730e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F06.BIT30.sha1
>@@ -0,0 +1 @@
>+0318de4dc48422bfe125fd6f827e6e7114dc61bc
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F06.OUT20.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F06.OUT20.sha1
>new file mode 100644
>index 00000000000..7d50df8258d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F06.OUT20.sha1
>@@ -0,0 +1 @@
>+c7bc71abc973440123a517f61e3dbcb96089ee3f
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F06.OUT30.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F06.OUT30.sha1
>new file mode 100644
>index 00000000000..ad9219dd0ad
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_coding/F06.OUT30.sha1
>@@ -0,0 +1 @@
>+5fde2defd2350b5ca7d3b8f8ceef75d2b7fd7f03
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_processing/agc2/rnn_vad/band_energies.dat.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_processing/agc2/rnn_vad/band_energies.dat.sha1
>new file mode 100644
>index 00000000000..aac4b9307ea
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_processing/agc2/rnn_vad/band_energies.dat.sha1
>@@ -0,0 +1 @@
>+52fad3366911c238929585daad02c8db11f99eae
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_processing/agc2/rnn_vad/fft.dat.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_processing/agc2/rnn_vad/fft.dat.sha1
>new file mode 100644
>index 00000000000..ebd5124d593
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_processing/agc2/rnn_vad/fft.dat.sha1
>@@ -0,0 +1 @@
>+e62364d35abd123663bfc800fa233071d6d7fffd
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_processing/agc2/rnn_vad/pitch_buf_24k.dat.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_processing/agc2/rnn_vad/pitch_buf_24k.dat.sha1
>new file mode 100644
>index 00000000000..1bcf6397e14
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_processing/agc2/rnn_vad/pitch_buf_24k.dat.sha1
>@@ -0,0 +1 @@
>+ea598d4ef3f4e34bce4c4c5d0791a588517582b9
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_processing/agc2/rnn_vad/pitch_lp_res.dat.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_processing/agc2/rnn_vad/pitch_lp_res.dat.sha1
>new file mode 100644
>index 00000000000..48533190abd
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_processing/agc2/rnn_vad/pitch_lp_res.dat.sha1
>@@ -0,0 +1 @@
>+03069dd7d5cf7ba63c0b6c90f7f7a283e7488ea0
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_processing/agc2/rnn_vad/pitch_search_int.dat.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_processing/agc2/rnn_vad/pitch_search_int.dat.sha1
>new file mode 100644
>index 00000000000..fc4a38956e6
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_processing/agc2/rnn_vad/pitch_search_int.dat.sha1
>@@ -0,0 +1 @@
>+d8d99ae84461254662686000b6d3d49b08dbef3e
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_processing/agc2/rnn_vad/samples.pcm.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_processing/agc2/rnn_vad/samples.pcm.sha1
>new file mode 100644
>index 00000000000..975819d3257
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_processing/agc2/rnn_vad/samples.pcm.sha1
>@@ -0,0 +1 @@
>+2417ed3d330eb118f6b16962438189468180b642
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_processing/agc2/rnn_vad/sil_features.dat.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_processing/agc2/rnn_vad/sil_features.dat.sha1
>new file mode 100644
>index 00000000000..bc591e9d6cb
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_processing/agc2/rnn_vad/sil_features.dat.sha1
>@@ -0,0 +1 @@
>+e0a92782c2903be9da10385d924d34e8bf212d5e
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_processing/agc2/rnn_vad/vad_prob.dat.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_processing/agc2/rnn_vad/vad_prob.dat.sha1
>new file mode 100644
>index 00000000000..1aa3bd0d832
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_processing/agc2/rnn_vad/vad_prob.dat.sha1
>@@ -0,0 +1 @@
>+05735ede0b457318e307d12f5acfd11bbbbd0afd
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_processing/output_data_fixed.pb.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_processing/output_data_fixed.pb.sha1
>index c9274a57cc7..1e2d75af914 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_processing/output_data_fixed.pb.sha1
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/audio_processing/output_data_fixed.pb.sha1
>@@ -1 +1 @@
>-7d9a02619aa4a3095ee8d48697bffef8437e6fe0
>\ No newline at end of file
>+75a835fd9e641531989b7ee1de20e84b553d3bc9
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/foreman_240x136.yuv.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/foreman_240x136.yuv.sha1
>new file mode 100644
>index 00000000000..bc162e16b9a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/foreman_240x136.yuv.sha1
>@@ -0,0 +1 @@
>+117c0301ed19d83f0e112ea3e831dd389db68570
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/resources/foreman_480x272.yuv.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/foreman_480x272.yuv.sha1
>new file mode 100644
>index 00000000000..a5fded85192
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/resources/foreman_480x272.yuv.sha1
>@@ -0,0 +1 @@
>+65717a7be6dc7ce5d88afedc73e46838aa0a3abc
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/BUILD.gn
>index 74bb964aca3..6b34ade8d4b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/BUILD.gn
>@@ -14,22 +14,6 @@ if (is_android) {
>   import("//build/config/android/config.gni")
>   import("//build/config/android/rules.gni")
> }
>-if (is_win) {
>-  import("//build/config/clang/clang.gni")
>-}
>-
>-group("base") {
>-  public_deps = [
>-    ":rtc_base",
>-    ":rtc_base_approved",
>-    ":rtc_task_queue",
>-    ":sequenced_task_checker",
>-    ":weak_ptr",
>-  ]
>-  if (is_android) {
>-    public_deps += [ ":base_java" ]
>-  }
>-}
> 
> config("rtc_base_chromium_config") {
>   defines = [ "NO_MAIN_THREAD_WRAPPING" ]
>@@ -39,6 +23,7 @@ config("rtc_base_all_dependent_config") {
>   if (is_ios) {
>     libs = [
>       "CFNetwork.framework",
>+      "Foundation.framework",
>       "Security.framework",
>       "SystemConfiguration.framework",
>       "UIKit.framework",
>@@ -55,7 +40,7 @@ config("rtc_base_all_dependent_config") {
>   }
> }
> 
>-if (!rtc_build_ssl && !build_with_mozilla) {
>+if (!rtc_build_ssl) {
>   config("external_ssl_library") {
>     assert(rtc_ssl_root != "",
>            "You must specify rtc_ssl_root when rtc_build_ssl==0.")
>@@ -63,19 +48,20 @@ if (!rtc_build_ssl && !build_with_mozilla) {
>   }
> }
> 
>-source_set("protobuf_utils") {
>+rtc_source_set("protobuf_utils") {
>   visibility = [ "*" ]
>   sources = [
>     "protobuf_utils.h",
>   ]
>   if (rtc_enable_protobuf) {
>-    public_deps = [
>+    public_configs = [ "//third_party/protobuf:protobuf_config" ]
>+    deps = [
>       "//third_party/protobuf:protobuf_lite",
>     ]
>   }
> }
> 
>-source_set("compile_assert_c") {
>+rtc_source_set("compile_assert_c") {
>   sources = [
>     "compile_assert_c.h",
>   ]
>@@ -84,21 +70,186 @@ source_set("compile_assert_c") {
> rtc_source_set("rtc_base_approved") {
>   visibility = [ "*" ]
>   public_deps = [
>+    ":atomicops",
>+    ":criticalsection",
>+    ":logging",
>+    ":macromagic",
>+    ":platform_thread",
>+    ":platform_thread_types",
>+    ":ptr_util",
>+    ":refcount",
>     ":rtc_base_approved_generic",
>+    ":rtc_event",
>+    ":safe_conversions",
>+    ":stringutils",
>+    ":thread_checker",
>+    ":timeutils",
>   ]
>   if (is_mac && !build_with_chromium) {
>     public_deps += [ ":rtc_base_approved_objc" ]
>   }
> }
> 
>+rtc_source_set("macromagic") {
>+  # TODO(bugs.webrtc.org/9606): This should not be public.
>+  visibility = [ "*" ]
>+  sources = [
>+    "arraysize.h",
>+    "constructormagic.h",
>+    "format_macros.h",
>+    "stringize_macros.h",
>+    "thread_annotations.h",
>+  ]
>+  deps = [
>+    "system:arch",
>+  ]
>+}
>+
>+rtc_source_set("platform_thread_types") {
>+  sources = [
>+    "platform_thread_types.cc",
>+    "platform_thread_types.h",
>+  ]
>+}
>+
>+rtc_source_set("ptr_util") {
>+  visibility = [ "*" ]
>+  sources = [
>+    "scoped_ref_ptr.h",
>+  ]
>+}
>+
>+rtc_source_set("refcount") {
>+  visibility = [ "*" ]
>+  sources = [
>+    "refcount.h",
>+    "refcountedobject.h",
>+    "refcounter.h",
>+  ]
>+  deps = [
>+    ":atomicops",
>+    ":macromagic",
>+  ]
>+}
>+
>+rtc_source_set("criticalsection") {
>+  sources = [
>+    "criticalsection.cc",
>+    "criticalsection.h",
>+  ]
>+  deps = [
>+    ":atomicops",
>+    ":checks",
>+    ":macromagic",
>+    ":platform_thread_types",
>+    "system:unused",
>+  ]
>+}
>+
>+rtc_source_set("platform_thread") {
>+  visibility = [
>+    ":rtc_base_approved",
>+    ":rtc_base_approved_generic",
>+    ":rtc_task_queue_libevent",
>+    ":rtc_task_queue_win",
>+    ":sequenced_task_checker",
>+  ]
>+  sources = [
>+    "platform_thread.cc",
>+    "platform_thread.h",
>+  ]
>+  deps = [
>+    ":atomicops",
>+    ":checks",
>+    ":macromagic",
>+    ":platform_thread_types",
>+    ":rtc_event",
>+    ":thread_checker",
>+    ":timeutils",
>+  ]
>+}
>+
>+rtc_source_set("rtc_event") {
>+  deps = [
>+    ":checks",
>+    ":macromagic",
>+  ]
>+
>+  if (build_with_chromium) {
>+    # Dependency on chromium's waitable_event (in //base).
>+    deps += [ "//base:base" ]
>+    sources = [
>+      "../../webrtc_overrides/rtc_base/event.cc",
>+      "../../webrtc_overrides/rtc_base/event.h",
>+    ]
>+  } else {
>+    sources = [
>+      "event.cc",
>+      "event.h",
>+    ]
>+  }
>+}
>+
>+rtc_source_set("logging") {
>+  visibility = [ "*" ]
>+  deps = [
>+    ":criticalsection",
>+    ":macromagic",
>+    ":platform_thread_types",
>+    ":stringutils",
>+    ":timeutils",
>+  ]
>+
>+  if (build_with_chromium) {
>+    # Dependency on chromium's logging (in //base).
>+    deps += [ "//base:base" ]
>+    sources = [
>+      "../../webrtc_overrides/rtc_base/logging.cc",
>+      "../../webrtc_overrides/rtc_base/logging.h",
>+    ]
>+  } else {
>+    sources = [
>+      "logging.cc",
>+      "logging.h",
>+    ]
>+    deps += [ "system:inline" ]
>+
>+    # logging.h needs the deprecation header while downstream projects are
>+    # removing code that depends on logging implementation details.
>+    deps += [ ":deprecation" ]
>+  }
>+}
>+
>+rtc_source_set("thread_checker") {
>+  sources = [
>+    "thread_checker.h",
>+    "thread_checker_impl.cc",
>+    "thread_checker_impl.h",
>+  ]
>+  deps = [
>+    ":checks",
>+    ":criticalsection",
>+    ":macromagic",
>+    ":platform_thread_types",
>+  ]
>+}
>+
>+rtc_source_set("atomicops") {
>+  sources = [
>+    "atomicops.h",
>+  ]
>+}
>+
> rtc_source_set("checks") {
>+  # TODO(bugs.webrtc.org/9607): This should not be public.
>+  visibility = [ "*" ]
>   sources = [
>     "checks.cc",
>     "checks.h",
>   ]
>   deps = [
>     ":safe_compare",
>-    "..:typedefs",
>+    "system:inline",
>   ]
> }
> 
>@@ -128,13 +279,67 @@ rtc_source_set("safe_compare") {
>   ]
> }
> 
>+rtc_source_set("safe_minmax") {
>+  sources = [
>+    "numerics/safe_minmax.h",
>+  ]
>+  deps = [
>+    ":checks",
>+    ":safe_compare",
>+    ":type_traits",
>+  ]
>+}
>+
>+rtc_source_set("safe_conversions") {
>+  sources = [
>+    "numerics/safe_conversions.h",
>+    "numerics/safe_conversions_impl.h",
>+  ]
>+  deps = [
>+    ":checks",
>+  ]
>+}
>+
>+rtc_source_set("timeutils") {
>+  visibility = [ "*" ]
>+  sources = [
>+    "timeutils.cc",
>+    "timeutils.h",
>+  ]
>+  deps = [
>+    ":checks",
>+    ":safe_conversions",
>+  ]
>+}
>+
> rtc_source_set("stringutils") {
>   sources = [
>+    "string_to_number.cc",
>+    "string_to_number.h",
>+    "stringencode.cc",
>+    "stringencode.h",
>+    "strings/string_builder.cc",
>+    "strings/string_builder.h",
>     "stringutils.cc",
>     "stringutils.h",
>   ]
>   deps = [
>     ":checks",
>+    ":macromagic",
>+    ":safe_minmax",
>+    "../api:array_view",
>+    "//third_party/abseil-cpp/absl/types:optional",
>+  ]
>+}
>+
>+rtc_source_set("audio_format_to_string") {
>+  sources = [
>+    "strings/audio_format_to_string.cc",
>+    "strings/audio_format_to_string.h",
>+  ]
>+  deps = [
>+    ":stringutils",
>+    "../api/audio_codecs:audio_codecs_api",
>   ]
> }
> 
>@@ -154,27 +359,37 @@ rtc_source_set("deprecation") {
> rtc_source_set("rtc_base_approved_generic") {
>   visibility = [
>     ":rtc_base_approved",
>-    ":rtc_base_approved_objc",
>     ":weak_ptr_unittests",
>   ]
> 
>   cflags = []
>   defines = []
>   libs = []
>+  data_deps = []
>   deps = [
>+    ":atomicops",
>     ":checks",
>+    ":criticalsection",
>+    ":logging",
>+    ":macromagic",
>+    ":platform_thread",
>+    ":platform_thread_types",
>+    ":ptr_util",
>+    ":refcount",
>+    ":rtc_event",
>+    ":rtc_task_queue",
>     ":safe_compare",
>+    ":safe_conversions",
>     ":stringutils",
>+    ":thread_checker",
>+    ":timeutils",
>     ":type_traits",
>-    "../:typedefs",
>+    "system:arch",
>+    "system:unused",
>+    "third_party/base64",
>   ]
> 
>   sources = [
>-    "arraysize.h",
>-    "atomicops.h",
>-    "base64.cc",
>-    "base64.h",
>-    "basictypes.h",
>     "bind.h",
>     "bitbuffer.cc",
>     "bitbuffer.h",
>@@ -186,18 +401,14 @@ rtc_source_set("rtc_base_approved_generic") {
>     "bytebuffer.cc",
>     "bytebuffer.h",
>     "byteorder.h",
>-    "constructormagic.h",
>     "copyonwritebuffer.cc",
>     "copyonwritebuffer.h",
>-    "criticalsection.cc",
>-    "criticalsection.h",
>     "event_tracer.cc",
>     "event_tracer.h",
>     "file.cc",
>     "file.h",
>     "flags.cc",
>     "flags.h",
>-    "format_macros.h",
>     "function_view.h",
>     "ignore_wundef.h",
>     "location.cc",
>@@ -206,18 +417,13 @@ rtc_source_set("rtc_base_approved_generic") {
>     "numerics/histogram_percentile_counter.h",
>     "numerics/mod_ops.h",
>     "numerics/moving_max_counter.h",
>-    "numerics/safe_conversions.h",
>-    "numerics/safe_conversions_impl.h",
>-    "numerics/safe_minmax.h",
>+    "numerics/sample_counter.cc",
>+    "numerics/sample_counter.h",
>     "onetimeevent.h",
>     "pathutils.cc",
>     "pathutils.h",
>     "platform_file.cc",
>     "platform_file.h",
>-    "platform_thread.cc",
>-    "platform_thread.h",
>-    "platform_thread_types.h",
>-    "ptr_util.h",
>     "race_checker.cc",
>     "race_checker.h",
>     "random.cc",
>@@ -226,135 +432,38 @@ rtc_source_set("rtc_base_approved_generic") {
>     "rate_statistics.h",
>     "ratetracker.cc",
>     "ratetracker.h",
>-    "refcount.h",
>-    "refcountedobject.h",
>-    "refcounter.h",
>-    "scoped_ref_ptr.h",
>-    "string_to_number.cc",
>-    "string_to_number.h",
>-    "stringencode.cc",
>-    "stringencode.h",
>-    "stringize_macros.h",
>     "swap_queue.h",
>     "template_util.h",
>-    "thread_annotations.h",
>-    "thread_checker.h",
>-    "thread_checker_impl.cc",
>-    "thread_checker_impl.h",
>     "timestampaligner.cc",
>     "timestampaligner.h",
>-    "timeutils.cc",
>-    "timeutils.h",
>     "trace_event.h",
>+    "zero_memory.cc",
>+    "zero_memory.h",
>   ]
> 
>   deps += [
>     "..:webrtc_common",
>     "../api:array_view",
>-    "../api:optional",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> 
>   if (is_android) {
>     libs += [ "log" ]
>   }
> 
>-  if (is_posix) {
>+  if (is_posix || is_fuchsia) {
>     sources += [ "file_posix.cc" ]
>   }
> 
>   if (is_win) {
>-    sources += [ "file_win.cc" ]
>-  }
>-
>-  if (build_with_chromium) {
>-    # Dependency on chromium's logging (in //base).
>-    deps += [ "//base:base" ]
>-    sources += [
>-      "../../webrtc_overrides/rtc_base/event.cc",
>-      "../../webrtc_overrides/rtc_base/event.h",
>-      "../../webrtc_overrides/rtc_base/logging.cc",
>-      "../../webrtc_overrides/rtc_base/logging.h",
>-    ]
>-  } else {
>     sources += [
>-      "event.cc",
>-      "event.h",
>-      "logging.cc",
>-      "logging.h",
>+      "file_win.cc",
>+      "win/windows_version.cc",
>+      "win/windows_version.h",
>     ]
>+    data_deps += [ "//build/win:runtime_libs" ]
>   }
>-  if (is_component_build && is_win) {
>-    # Copy the VS runtime DLLs into the isolate so that they don't have to be
>-    # preinstalled on the target machine. The debug runtimes have a "d" at
>-    # the end.
>-    # This is a copy of https://codereview.chromium.org/1783973002.
>-    # TODO(ehmaldonado): We'd like Chromium to make this changes easier to use,
>-    # so we don't have to copy their changes and risk breakages.
>-    # See http://crbug.com/653569
>-    if (is_debug) {
>-      vcrt_suffix = "d"
>-    } else {
>-      vcrt_suffix = ""
>-    }
> 
>-    # These runtime files are copied to the output directory by the
>-    # vs_toolchain script that runs as part of toolchain configuration.
>-    data = [
>-      "$root_out_dir/msvcp140${vcrt_suffix}.dll",
>-      "$root_out_dir/vccorlib140${vcrt_suffix}.dll",
>-      "$root_out_dir/vcruntime140${vcrt_suffix}.dll",
>-
>-      # Universal Windows 10 CRT files
>-      "$root_out_dir/api-ms-win-core-console-l1-1-0.dll",
>-      "$root_out_dir/api-ms-win-core-datetime-l1-1-0.dll",
>-      "$root_out_dir/api-ms-win-core-debug-l1-1-0.dll",
>-      "$root_out_dir/api-ms-win-core-errorhandling-l1-1-0.dll",
>-      "$root_out_dir/api-ms-win-core-file-l1-1-0.dll",
>-      "$root_out_dir/api-ms-win-core-file-l1-2-0.dll",
>-      "$root_out_dir/api-ms-win-core-file-l2-1-0.dll",
>-      "$root_out_dir/api-ms-win-core-handle-l1-1-0.dll",
>-      "$root_out_dir/api-ms-win-core-heap-l1-1-0.dll",
>-      "$root_out_dir/api-ms-win-core-interlocked-l1-1-0.dll",
>-      "$root_out_dir/api-ms-win-core-libraryloader-l1-1-0.dll",
>-      "$root_out_dir/api-ms-win-core-localization-l1-2-0.dll",
>-      "$root_out_dir/api-ms-win-core-memory-l1-1-0.dll",
>-      "$root_out_dir/api-ms-win-core-namedpipe-l1-1-0.dll",
>-      "$root_out_dir/api-ms-win-core-processenvironment-l1-1-0.dll",
>-      "$root_out_dir/api-ms-win-core-processthreads-l1-1-0.dll",
>-      "$root_out_dir/api-ms-win-core-processthreads-l1-1-1.dll",
>-      "$root_out_dir/api-ms-win-core-profile-l1-1-0.dll",
>-      "$root_out_dir/api-ms-win-core-rtlsupport-l1-1-0.dll",
>-      "$root_out_dir/api-ms-win-core-string-l1-1-0.dll",
>-      "$root_out_dir/api-ms-win-core-synch-l1-1-0.dll",
>-      "$root_out_dir/api-ms-win-core-synch-l1-2-0.dll",
>-      "$root_out_dir/api-ms-win-core-sysinfo-l1-1-0.dll",
>-      "$root_out_dir/api-ms-win-core-timezone-l1-1-0.dll",
>-      "$root_out_dir/api-ms-win-core-util-l1-1-0.dll",
>-      "$root_out_dir/api-ms-win-crt-conio-l1-1-0.dll",
>-      "$root_out_dir/api-ms-win-crt-convert-l1-1-0.dll",
>-      "$root_out_dir/api-ms-win-crt-environment-l1-1-0.dll",
>-      "$root_out_dir/api-ms-win-crt-filesystem-l1-1-0.dll",
>-      "$root_out_dir/api-ms-win-crt-heap-l1-1-0.dll",
>-      "$root_out_dir/api-ms-win-crt-locale-l1-1-0.dll",
>-      "$root_out_dir/api-ms-win-crt-math-l1-1-0.dll",
>-      "$root_out_dir/api-ms-win-crt-multibyte-l1-1-0.dll",
>-      "$root_out_dir/api-ms-win-crt-private-l1-1-0.dll",
>-      "$root_out_dir/api-ms-win-crt-process-l1-1-0.dll",
>-      "$root_out_dir/api-ms-win-crt-runtime-l1-1-0.dll",
>-      "$root_out_dir/api-ms-win-crt-stdio-l1-1-0.dll",
>-      "$root_out_dir/api-ms-win-crt-string-l1-1-0.dll",
>-      "$root_out_dir/api-ms-win-crt-time-l1-1-0.dll",
>-      "$root_out_dir/api-ms-win-crt-utility-l1-1-0.dll",
>-      "$root_out_dir/ucrtbase${vcrt_suffix}.dll",
>-    ]
>-    if (is_asan) {
>-      if (current_cpu == "x64") {
>-        data += [ "$clang_base_path/lib/clang/$clang_version/lib/windows/clang_rt.asan_dynamic-x86_64.dll" ]
>-      } else {
>-        data += [ "$clang_base_path/lib/clang/$clang_version/lib/windows/clang_rt.asan_dynamic-i386.dll" ]
>-      }
>-    }
>-  }
>   if (is_nacl) {
>     deps += [ "//native_client_sdk/src/libraries/nacl_io" ]
>   }
>@@ -373,16 +482,14 @@ if (is_mac && !build_with_chromium) {
>       "logging_mac.mm",
>     ]
>     deps = [
>-      ":rtc_base_approved_generic",
>+      ":logging",
>     ]
>   }
> }
> 
> rtc_source_set("rtc_task_queue") {
>   visibility = [ "*" ]
>-  deps = [
>-    ":rtc_base_approved",
>-  ]
>+  deps = []
>   public_deps = [
>     ":rtc_task_queue_api",
>   ]
>@@ -399,47 +506,111 @@ rtc_source_set("rtc_task_queue") {
> # provided implementation should be used. An external implementation should
> # depend on rtc_task_queue_api.
> rtc_source_set("rtc_task_queue_api") {
>+  # The visibility list is commented out so that we won't break external
>+  # implementations, but left here to manually test as well as for sake of what
>+  # targets we expect to depend on rtc_task_queue_api.
>+  # visibility = [
>+  #   ":rtc_task_queue",
>+  #   ":rtc_task_queue_impl",
>+  #   ":sequenced_task_checker",
>+  # ]
>   sources = [
>     "task_queue.h",
>   ]
>   deps = [
>-    ":rtc_base_approved",
>+    ":macromagic",
>+    ":ptr_util",
>+    "//third_party/abseil-cpp/absl/memory",
>   ]
> }
> 
>-rtc_source_set("rtc_task_queue_impl") {
>-  visibility = [ "*" ]
>-  deps = [
>-    ":checks",
>-    ":rtc_base_approved",
>-    ":rtc_task_queue_api",
>-  ]
>-  if (rtc_build_libevent) {
>-    deps += [ "//base/third_party/libevent" ]
>-  }
>-  if (rtc_enable_libevent) {
>+if (rtc_enable_libevent) {
>+  rtc_source_set("rtc_task_queue_libevent") {
>+    visibility = [ ":rtc_task_queue_impl" ]
>     sources = [
>       "task_queue_libevent.cc",
>       "task_queue_posix.cc",
>       "task_queue_posix.h",
>     ]
>+    deps = [
>+      ":checks",
>+      ":criticalsection",
>+      ":logging",
>+      ":platform_thread",
>+      ":ptr_util",
>+      ":refcount",
>+      ":rtc_task_queue_api",
>+      ":safe_conversions",
>+      ":timeutils",
>+      "system:unused",
>+    ]
>+    if (rtc_build_libevent) {
>+      deps += [ "//base/third_party/libevent" ]
>+    }
>+  }
>+}
>+
>+if (is_mac || is_ios) {
>+  rtc_source_set("rtc_task_queue_gcd") {
>+    visibility = [ ":rtc_task_queue_impl" ]
>+    sources = [
>+      "task_queue_gcd.cc",
>+      "task_queue_posix.cc",
>+      "task_queue_posix.h",
>+    ]
>+    deps = [
>+      ":checks",
>+      ":logging",
>+      ":ptr_util",
>+      ":refcount",
>+      ":rtc_task_queue_api",
>+    ]
>+  }
>+}
>+
>+if (is_win) {
>+  rtc_source_set("rtc_task_queue_win") {
>+    visibility = [ ":rtc_task_queue_impl" ]
>+    sources = [
>+      "task_queue_win.cc",
>+    ]
>+    deps = [
>+      ":checks",
>+      ":criticalsection",
>+      ":logging",
>+      ":macromagic",
>+      ":platform_thread",
>+      ":ptr_util",
>+      ":refcount",
>+      ":rtc_event",
>+      ":rtc_task_queue_api",
>+      ":safe_conversions",
>+      ":timeutils",
>+    ]
>+  }
>+}
>+
>+rtc_source_set("rtc_task_queue_impl") {
>+  visibility = [ "*" ]
>+  if (rtc_enable_libevent) {
>+    deps = [
>+      ":rtc_task_queue_libevent",
>+    ]
>   } else {
>     if (is_mac || is_ios) {
>-      sources = [
>-        "task_queue_gcd.cc",
>-        "task_queue_posix.cc",
>-        "task_queue_posix.h",
>+      deps = [
>+        ":rtc_task_queue_gcd",
>       ]
>     }
>     if (is_win) {
>-      sources = [
>-        "task_queue_win.cc",
>+      deps = [
>+        ":rtc_task_queue_win",
>       ]
>     }
>   }
> }
> 
>-rtc_static_library("sequenced_task_checker") {
>+rtc_source_set("sequenced_task_checker") {
>   sources = [
>     "sequenced_task_checker.h",
>     "sequenced_task_checker_impl.cc",
>@@ -447,8 +618,10 @@ rtc_static_library("sequenced_task_checker") {
>   ]
>   deps = [
>     ":checks",
>-    ":rtc_base_approved",
>+    ":criticalsection",
>+    ":macromagic",
>     ":rtc_task_queue",
>+    ":thread_checker",
>   ]
> }
> 
>@@ -458,7 +631,8 @@ rtc_static_library("weak_ptr") {
>     "weak_ptr.h",
>   ]
>   deps = [
>-    ":rtc_base_approved",
>+    ":ptr_util",
>+    ":refcount",
>     ":sequenced_task_checker",
>   ]
> }
>@@ -475,27 +649,18 @@ rtc_static_library("rtc_numerics") {
>     ":checks",
>     ":rtc_base_approved",
>     ":safe_compare",
>-    "../api:optional",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
> 
>-config("rtc_base_warnings_config") {
>-  if (is_win && is_clang) {
>-    cflags = [
>-      # Disable warnings failing when compiling with Clang on Windows.
>-      # https://bugs.chromium.org/p/webrtc/issues/detail?id=5366
>-      "-Wno-sign-compare",
>-      "-Wno-missing-braces",
>-    ]
>-  }
>-}
>-
> rtc_source_set("rtc_json") {
>+  testonly = true
>   defines = []
>   sources = [
>     "json.cc",
>     "json.h",
>   ]
>+  all_dependent_configs = [ "//third_party/jsoncpp:jsoncpp_config" ]
>   if (rtc_build_json) {
>     public_deps = [
>       "//third_party/jsoncpp",
>@@ -549,7 +714,11 @@ rtc_static_library("rtc_base_generic") {
>     ":checks",
>     ":stringutils",
>     "..:webrtc_common",
>-    "../api:optional",
>+    "../api:array_view",
>+    "third_party/base64",
>+    "third_party/sigslot",
>+    "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
>   public_deps = [
>     ":rtc_base_approved",
>@@ -576,6 +745,8 @@ rtc_static_library("rtc_base_generic") {
>     "crc32.h",
>     "cryptstring.cc",
>     "cryptstring.h",
>+    "data_rate_limiter.cc",
>+    "data_rate_limiter.h",
>     "dscp.h",
>     "filerotatingstream.cc",
>     "filerotatingstream.h",
>@@ -611,27 +782,31 @@ rtc_static_library("rtc_base_generic") {
>     "openssl.h",
>     "openssladapter.cc",
>     "openssladapter.h",
>+    "opensslcertificate.cc",
>+    "opensslcertificate.h",
>     "openssldigest.cc",
>     "openssldigest.h",
>     "opensslidentity.cc",
>     "opensslidentity.h",
>+    "opensslsessioncache.cc",
>+    "opensslsessioncache.h",
>     "opensslstreamadapter.cc",
>     "opensslstreamadapter.h",
>+    "opensslutility.cc",
>+    "opensslutility.h",
>     "physicalsocketserver.cc",
>     "physicalsocketserver.h",
>     "proxyinfo.cc",
>     "proxyinfo.h",
>-    "ratelimiter.cc",
>-    "ratelimiter.h",
>     "rtccertificate.cc",
>     "rtccertificate.h",
>     "rtccertificategenerator.cc",
>     "rtccertificategenerator.h",
>     "signalthread.cc",
>     "signalthread.h",
>-    "sigslot.cc",
>     "sigslot.h",
>     "sigslotrepeater.h",
>+    "socket.cc",
>     "socket.h",
>     "socketadapters.cc",
>     "socketadapters.h",
>@@ -645,6 +820,8 @@ rtc_static_library("rtc_base_generic") {
>     "socketstream.h",
>     "ssladapter.cc",
>     "ssladapter.h",
>+    "sslcertificate.cc",
>+    "sslcertificate.h",
>     "sslfingerprint.cc",
>     "sslfingerprint.h",
>     "sslidentity.cc",
>@@ -662,14 +839,6 @@ rtc_static_library("rtc_base_generic") {
>     ":rtc_base_objc",
>   ]
> 
>-  # TODO(henrike): issue 3307, make rtc_base build with the Chromium default
>-  # compiler settings.
>-  suppressed_configs += [ "//build/config/compiler:chromium_code" ]
>-  configs += [ "//build/config/compiler:no_chromium_code" ]
>-  if (!is_win) {
>-    cflags += [ "-Wno-uninitialized" ]
>-  }
>-
>   if (build_with_chromium) {
>     if (is_win) {
>       sources += [ "../../webrtc_overrides/rtc_base/win32socketinit.cc" ]
>@@ -677,7 +846,6 @@ rtc_static_library("rtc_base_generic") {
>     include_dirs = [ "../../boringssl/src/include" ]
>     public_configs += [ ":rtc_base_chromium_config" ]
>   } else {
>-    configs += [ ":rtc_base_warnings_config" ]
>     sources += [
>       "callback.h",
>       "logsinks.cc",
>@@ -687,8 +855,6 @@ rtc_static_library("rtc_base_generic") {
>       "optionsfile.h",
>       "rollingaccumulator.h",
>       "sslroots.h",
>-      "transformadapter.cc",
>-      "transformadapter.h",
>     ]
> 
>     if (is_win) {
>@@ -703,7 +869,7 @@ rtc_static_library("rtc_base_generic") {
> 
>   if (rtc_build_ssl) {
>     deps += [ "//third_party/boringssl" ]
>-  } else if (!build_with_mozilla) {
>+  } else {
>     configs += [ ":external_ssl_library" ]
>   }
> 
>@@ -758,7 +924,6 @@ rtc_static_library("rtc_base_generic") {
>       "win32.h",
>       "win32filesystem.cc",
>       "win32filesystem.h",
>-      "win32securityerrors.cc",
>       "win32window.cc",
>       "win32window.h",
>     ]
>@@ -769,16 +934,10 @@ rtc_static_library("rtc_base_generic") {
>       "secur32.lib",
>     ]
> 
>-    cflags += [
>-      # Suppress warnings about WIN32_LEAN_AND_MEAN.
>-      "/wd4005",
>-      "/wd4703",
>-    ]
>-
>     defines += [ "_CRT_NONSTDC_NO_DEPRECATE" ]
>   }
> 
>-  if (is_posix) {
>+  if (is_posix || is_fuchsia) {
>     sources += [
>       "ifaddrs_converter.cc",
>       "ifaddrs_converter.h",
>@@ -801,20 +960,6 @@ rtc_source_set("gtest_prod") {
>   ]
> }
> 
>-config("rtc_base_tests_utils_exported_config") {
>-  defines = [ "GTEST_RELATIVE_PATH" ]
>-}
>-
>-config("rtc_base_tests_utils_warnings_config") {
>-  if (is_win && is_clang) {
>-    cflags = [
>-      # See https://bugs.chromium.org/p/webrtc/issues/detail?id=6270
>-      "-Wno-reorder",
>-      "-Wno-sign-compare",
>-    ]
>-  }
>-}
>-
> rtc_source_set("rtc_base_tests_utils") {
>   testonly = true
>   sources = [
>@@ -829,15 +974,12 @@ rtc_source_set("rtc_base_tests_utils") {
>     "fakesslidentity.h",
>     "firewallsocketserver.cc",
>     "firewallsocketserver.h",
>+    "gunit.cc",
>     "gunit.h",
>     "httpbase.cc",
>     "httpbase.h",
>     "httpserver.cc",
>     "httpserver.h",
>-    "md5.cc",
>-    "md5.h",
>-    "md5digest.cc",
>-    "md5digest.h",
>     "memory_usage.cc",
>     "memory_usage.h",
>     "natserver.cc",
>@@ -848,56 +990,80 @@ rtc_source_set("rtc_base_tests_utils") {
>     "nattypes.h",
>     "proxyserver.cc",
>     "proxyserver.h",
>-    "refcount.h",
>-    "sha1.cc",
>-    "sha1.h",
>-    "sha1digest.cc",
>-    "sha1digest.h",
>     "sigslottester.h",
>     "sigslottester.h.pump",
>     "testbase64.h",
>+    "testcertificateverifier.h",
>     "testclient.cc",
>     "testclient.h",
>     "testechoserver.cc",
>     "testechoserver.h",
>     "testutils.cc",
>     "testutils.h",
>-    "timedelta.h",
>     "virtualsocketserver.cc",
>     "virtualsocketserver.h",
>   ]
>-  configs += [ ":rtc_base_tests_utils_warnings_config" ]
>-  public_configs = [ ":rtc_base_tests_utils_exported_config" ]
>   deps = [
>     ":checks",
>     ":rtc_base",
>     ":stringutils",
>+    "../api/units:time_delta",
>     "../test:field_trial",
>     "../test:test_support",
>+    "system:fallthrough",
>+    "third_party/sigslot",
>+    "//third_party/abseil-cpp/absl/memory",
>   ]
>   public_deps = [
>-    "//testing/gmock",
>     "//testing/gtest",
>   ]
> }
> 
>+rtc_source_set("rtc_task_queue_for_test") {
>+  visibility = [ "*" ]
>+  testonly = true
>+
>+  sources = [
>+    "task_queue_for_test.cc",
>+    "task_queue_for_test.h",
>+  ]
>+  deps = [
>+    ":checks",
>+    ":rtc_base_approved",
>+    ":rtc_task_queue",
>+  ]
>+}
>+
> if (rtc_include_tests) {
>+  rtc_source_set("sigslot_unittest") {
>+    testonly = true
>+    sources = [
>+      "sigslot_unittest.cc",
>+    ]
>+    deps = [
>+      ":rtc_base",
>+      ":rtc_base_tests_utils",
>+      "third_party/sigslot",
>+    ]
>+  }
>+
>   rtc_source_set("rtc_base_tests_main") {
>     testonly = true
>     sources = [
>       "unittest_main.cc",
>     ]
>-    public_configs = [ ":rtc_base_tests_utils_exported_config" ]
>     deps = [
>       ":rtc_base",
>       ":rtc_base_approved",
>       ":rtc_base_tests_utils",
>+      "../system_wrappers:field_trial_default",
>+      "../system_wrappers:metrics_default",
>       "../test:field_trial",
>+      "../test:fileutils",
>       "../test:test_support",
>     ]
> 
>     public_deps = [
>-      "//testing/gmock",
>       "//testing/gtest",
>     ]
>   }
>@@ -920,8 +1086,10 @@ if (rtc_include_tests) {
>       ":rtc_base_tests_main",
>       ":rtc_base_tests_utils",
>       "../system_wrappers:system_wrappers",
>+      "../test:fileutils",
>       "../test:test_support",
>       "//testing/gtest",
>+      "//third_party/abseil-cpp/absl/memory",
>     ]
>     if (is_win) {
>       sources += [ "win32socketserver_unittest.cc" ]
>@@ -930,11 +1098,12 @@ if (rtc_include_tests) {
> 
>   rtc_source_set("rtc_base_approved_unittests") {
>     testonly = true
>-
>+    if (is_msan) {
>+      cflags = [ "-fsanitize=memory" ]
>+    }
>     sources = [
>       "atomicops_unittest.cc",
>       "base64_unittest.cc",
>-      "basictypes_unittest.cc",
>       "bind_unittest.cc",
>       "bitbuffer_unittest.cc",
>       "bitrateallocationstrategy_unittest.cc",
>@@ -949,23 +1118,26 @@ if (rtc_include_tests) {
>       "file_unittest.cc",
>       "function_view_unittest.cc",
>       "logging_unittest.cc",
>-      "md5digest_unittest.cc",
>       "numerics/histogram_percentile_counter_unittest.cc",
>       "numerics/mod_ops_unittest.cc",
>       "numerics/moving_max_counter_unittest.cc",
>       "numerics/safe_compare_unittest.cc",
>       "numerics/safe_minmax_unittest.cc",
>+      "numerics/sample_counter_unittest.cc",
>       "onetimeevent_unittest.cc",
>       "pathutils_unittest.cc",
>+      "platform_file_unittest.cc",
>       "platform_thread_unittest.cc",
>       "random_unittest.cc",
>       "rate_limiter_unittest.cc",
>       "rate_statistics_unittest.cc",
>       "ratetracker_unittest.cc",
>       "refcountedobject_unittest.cc",
>+      "sanitizer_unittest.cc",
>       "string_to_number_unittest.cc",
>       "stringencode_unittest.cc",
>       "stringize_macros_unittest.cc",
>+      "strings/string_builder_unittest.cc",
>       "stringutils_unittest.cc",
>       "swap_queue_unittest.cc",
>       "thread_annotations_unittest.cc",
>@@ -973,7 +1145,11 @@ if (rtc_include_tests) {
>       "timestampaligner_unittest.cc",
>       "timeutils_unittest.cc",
>       "virtualsocket_unittest.cc",
>+      "zero_memory_unittest.cc",
>     ]
>+    if (is_win) {
>+      sources += [ "win/windows_version_unittest.cc" ]
>+    }
>     deps = [
>       ":checks",
>       ":rate_limiter",
>@@ -983,10 +1159,16 @@ if (rtc_include_tests) {
>       ":rtc_base_tests_utils",
>       ":rtc_task_queue",
>       ":safe_compare",
>+      ":safe_minmax",
>+      ":sanitizer",
>       ":stringutils",
>       "../api:array_view",
>       "../system_wrappers:system_wrappers",
>+      "../test:fileutils",
>       "../test:test_support",
>+      "memory:unittests",
>+      "third_party/base64",
>+      "//third_party/abseil-cpp/absl/memory",
>     ]
>   }
> 
>@@ -1002,6 +1184,7 @@ if (rtc_include_tests) {
>       ":rtc_base_tests_main",
>       ":rtc_base_tests_utils",
>       ":rtc_task_queue",
>+      ":rtc_task_queue_for_test",
>       "../test:test_support",
>     ]
>   }
>@@ -1032,6 +1215,7 @@ if (rtc_include_tests) {
>       ":rtc_base_approved_generic",
>       ":rtc_base_tests_main",
>       ":rtc_base_tests_utils",
>+      ":rtc_event",
>       ":rtc_task_queue",
>       ":weak_ptr",
>       "../test:test_support",
>@@ -1055,17 +1239,14 @@ if (rtc_include_tests) {
>     ]
>   }
> 
>-  config("rtc_base_unittests_config") {
>-    if (is_clang) {
>-      cflags = [ "-Wno-unused-const-variable" ]
>-    }
>-  }
>   rtc_source_set("rtc_base_unittests") {
>     testonly = true
>+    defines = []
> 
>     sources = [
>       "callback_unittest.cc",
>       "crc32_unittest.cc",
>+      "data_rate_limiter_unittest.cc",
>       "helpers_unittest.cc",
>       "httpbase_unittest.cc",
>       "httpcommon_unittest.cc",
>@@ -1078,14 +1259,10 @@ if (rtc_include_tests) {
>       "network_unittest.cc",
>       "optionsfile_unittest.cc",
>       "proxy_unittest.cc",
>-      "ptr_util_unittest.cc",
>-      "ratelimiter_unittest.cc",
>       "rollingaccumulator_unittest.cc",
>       "rtccertificate_unittest.cc",
>       "rtccertificategenerator_unittest.cc",
>-      "sha1digest_unittest.cc",
>       "signalthread_unittest.cc",
>-      "sigslot_unittest.cc",
>       "sigslottester_unittest.cc",
>       "stream_unittest.cc",
>       "testclient_unittest.cc",
>@@ -1097,9 +1274,11 @@ if (rtc_include_tests) {
>         "win32window_unittest.cc",
>       ]
>     }
>-    if (is_posix) {
>+    if (is_posix || is_fuchsia) {
>       sources += [
>         "openssladapter_unittest.cc",
>+        "opensslsessioncache_unittest.cc",
>+        "opensslutility_unittest.cc",
>         "ssladapter_unittest.cc",
>         "sslidentity_unittest.cc",
>         "sslstreamadapter_unittest.cc",
>@@ -1110,13 +1289,16 @@ if (rtc_include_tests) {
>       ":rtc_base_tests_main",
>       ":rtc_base_tests_utils",
>       ":stringutils",
>-      "../api:optional",
>+      "../api:array_view",
>+      "../test:fileutils",
>       "../test:test_support",
>+      "third_party/sigslot",
>+      "//third_party/abseil-cpp/absl/memory",
>+      "//third_party/abseil-cpp/absl/types:optional",
>     ]
>     public_deps = [
>       ":rtc_base",
>     ]
>-    configs += [ ":rtc_base_unittests_config" ]
>     if (build_with_chromium) {
>       include_dirs = [ "../../boringssl/src/include" ]
>     }
>@@ -1132,6 +1314,7 @@ if (is_android) {
>   rtc_android_library("base_java") {
>     java_files = [
>       "java/src/org/webrtc/ContextUtils.java",
>+      "java/src/org/webrtc/Loggable.java",
>       "java/src/org/webrtc/Logging.java",
>       "java/src/org/webrtc/Size.java",
>       "java/src/org/webrtc/ThreadUtils.java",
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/OWNERS
>index f7e8d2a5a4d..69ce253087f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/OWNERS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/OWNERS
>@@ -1,14 +1,13 @@
>-henrika@webrtc.org
>-henrikg@webrtc.org
> hta@webrtc.org
> juberti@webrtc.org
>+kwiberg@webrtc.org
> mflodman@webrtc.org
> perkj@webrtc.org
> pthatcher@webrtc.org
>+qingsi@webrtc.org
> sergeyu@chromium.org
>+steveanton@webrtc.org
> tommi@webrtc.org
>-deadbeef@webrtc.org
>-kwiberg@webrtc.org
> 
> # These are for the common case of adding or renaming files. If you're doing
> # structural changes, please get a review from a reviewer in this file.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/arraysize.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/arraysize.h
>index f7845b5f497..bf8e6d8810b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/arraysize.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/arraysize.h
>@@ -24,7 +24,8 @@
> // This template function declaration is used in defining arraysize.
> // Note that the function doesn't need an implementation, as we only
> // use its type.
>-template <typename T, size_t N> char (&ArraySizeHelper(T (&array)[N]))[N];
>+template <typename T, size_t N>
>+char (&ArraySizeHelper(T (&array)[N]))[N];
> 
> #define arraysize(array) (sizeof(ArraySizeHelper(array)))
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncinvoker-inl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncinvoker-inl.h
>index 073d5bffac9..0dadc0f3b34 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncinvoker-inl.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncinvoker-inl.h
>@@ -17,7 +17,7 @@
> #include "rtc_base/messagehandler.h"
> #include "rtc_base/refcountedobject.h"
> #include "rtc_base/scoped_ref_ptr.h"
>-#include "rtc_base/sigslot.h"
>+#include "rtc_base/third_party/sigslot/sigslot.h"
> #include "rtc_base/thread.h"
> #include "rtc_base/thread_annotations.h"
> 
>@@ -48,14 +48,12 @@ class AsyncClosure {
> template <class FunctorT>
> class FireAndForgetAsyncClosure : public AsyncClosure {
>  public:
>-  explicit FireAndForgetAsyncClosure(AsyncInvoker* invoker,
>-                                     const FunctorT& functor)
>-      : AsyncClosure(invoker), functor_(functor) {}
>-  virtual void Execute() {
>-    functor_();
>-  }
>+  explicit FireAndForgetAsyncClosure(AsyncInvoker* invoker, FunctorT&& functor)
>+      : AsyncClosure(invoker), functor_(std::forward<FunctorT>(functor)) {}
>+  virtual void Execute() { functor_(); }
>+
>  private:
>-  FunctorT functor_;
>+  typename std::decay<FunctorT>::type functor_;
> };
> 
> }  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncinvoker.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncinvoker.cc
>index 7f65dfc35fc..e255fb98fcc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncinvoker.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncinvoker.cc
>@@ -67,6 +67,10 @@ void AsyncInvoker::Flush(Thread* thread, uint32_t id /*= MQID_ANY*/) {
>   }
> }
> 
>+void AsyncInvoker::Clear() {
>+  MessageQueueManager::Clear(this);
>+}
>+
> void AsyncInvoker::DoInvoke(const Location& posted_from,
>                             Thread* thread,
>                             std::unique_ptr<AsyncClosure> closure,
>@@ -102,8 +106,7 @@ GuardedAsyncInvoker::GuardedAsyncInvoker() : thread_(Thread::Current()) {
>                                         &GuardedAsyncInvoker::ThreadDestroyed);
> }
> 
>-GuardedAsyncInvoker::~GuardedAsyncInvoker() {
>-}
>+GuardedAsyncInvoker::~GuardedAsyncInvoker() {}
> 
> bool GuardedAsyncInvoker::Flush(uint32_t id) {
>   CritScope cs(&crit_);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncinvoker.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncinvoker.h
>index 3f0c4f1eb58..474ec7c335e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncinvoker.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncinvoker.h
>@@ -21,7 +21,7 @@
> #include "rtc_base/event.h"
> #include "rtc_base/refcountedobject.h"
> #include "rtc_base/scoped_ref_ptr.h"
>-#include "rtc_base/sigslot.h"
>+#include "rtc_base/third_party/sigslot/sigslot.h"
> #include "rtc_base/thread.h"
> 
> namespace rtc {
>@@ -97,10 +97,11 @@ class AsyncInvoker : public MessageHandler {
>   template <class ReturnT, class FunctorT>
>   void AsyncInvoke(const Location& posted_from,
>                    Thread* thread,
>-                   const FunctorT& functor,
>+                   FunctorT&& functor,
>                    uint32_t id = 0) {
>     std::unique_ptr<AsyncClosure> closure(
>-        new FireAndForgetAsyncClosure<FunctorT>(this, functor));
>+        new FireAndForgetAsyncClosure<FunctorT>(
>+            this, std::forward<FunctorT>(functor)));
>     DoInvoke(posted_from, thread, std::move(closure), id);
>   }
> 
>@@ -109,11 +110,12 @@ class AsyncInvoker : public MessageHandler {
>   template <class ReturnT, class FunctorT>
>   void AsyncInvokeDelayed(const Location& posted_from,
>                           Thread* thread,
>-                          const FunctorT& functor,
>+                          FunctorT&& functor,
>                           uint32_t delay_ms,
>                           uint32_t id = 0) {
>     std::unique_ptr<AsyncClosure> closure(
>-        new FireAndForgetAsyncClosure<FunctorT>(this, functor));
>+        new FireAndForgetAsyncClosure<FunctorT>(
>+            this, std::forward<FunctorT>(functor)));
>     DoInvokeDelayed(posted_from, thread, std::move(closure), delay_ms, id);
>   }
> 
>@@ -124,6 +126,11 @@ class AsyncInvoker : public MessageHandler {
>   // behavior is desired, call Flush() before destroying this object.
>   void Flush(Thread* thread, uint32_t id = MQID_ANY);
> 
>+  // Cancels any outstanding calls we own that are pending on any thread, and
>+  // which have not yet started to execute. This does not wait for any calls
>+  // that have already started executing to complete.
>+  void Clear();
>+
>  private:
>   void OnMessage(Message* msg) override;
>   void DoInvoke(const Location& posted_from,
>@@ -183,12 +190,13 @@ class GuardedAsyncInvoker : public sigslot::has_slots<> {
>   // immediately. Returns false if the thread has died.
>   template <class ReturnT, class FunctorT>
>   bool AsyncInvoke(const Location& posted_from,
>-                   const FunctorT& functor,
>+                   FunctorT&& functor,
>                    uint32_t id = 0) {
>     CritScope cs(&crit_);
>     if (thread_ == nullptr)
>       return false;
>-    invoker_.AsyncInvoke<ReturnT, FunctorT>(posted_from, thread_, functor, id);
>+    invoker_.AsyncInvoke<ReturnT, FunctorT>(
>+        posted_from, thread_, std::forward<FunctorT>(functor), id);
>     return true;
>   }
> 
>@@ -196,14 +204,14 @@ class GuardedAsyncInvoker : public sigslot::has_slots<> {
>   // completion. Returns immediately. Returns false if the thread has died.
>   template <class ReturnT, class FunctorT>
>   bool AsyncInvokeDelayed(const Location& posted_from,
>-                          const FunctorT& functor,
>+                          FunctorT&& functor,
>                           uint32_t delay_ms,
>                           uint32_t id = 0) {
>     CritScope cs(&crit_);
>     if (thread_ == nullptr)
>       return false;
>-    invoker_.AsyncInvokeDelayed<ReturnT, FunctorT>(posted_from, thread_,
>-                                                   functor, delay_ms, id);
>+    invoker_.AsyncInvokeDelayed<ReturnT, FunctorT>(
>+        posted_from, thread_, std::forward<FunctorT>(functor), delay_ms, id);
>     return true;
>   }
> 
>@@ -212,7 +220,7 @@ class GuardedAsyncInvoker : public sigslot::has_slots<> {
>   template <class ReturnT, class FunctorT, class HostT>
>   bool AsyncInvoke(const Location& posted_from,
>                    const Location& callback_posted_from,
>-                   const FunctorT& functor,
>+                   FunctorT&& functor,
>                    void (HostT::*callback)(ReturnT),
>                    HostT* callback_host,
>                    uint32_t id = 0) {
>@@ -220,8 +228,8 @@ class GuardedAsyncInvoker : public sigslot::has_slots<> {
>     if (thread_ == nullptr)
>       return false;
>     invoker_.AsyncInvoke<ReturnT, FunctorT, HostT>(
>-        posted_from, callback_posted_from, thread_, functor, callback,
>-        callback_host, id);
>+        posted_from, callback_posted_from, thread_,
>+        std::forward<FunctorT>(functor), callback, callback_host, id);
>     return true;
>   }
> 
>@@ -230,7 +238,7 @@ class GuardedAsyncInvoker : public sigslot::has_slots<> {
>   template <class ReturnT, class FunctorT, class HostT>
>   bool AsyncInvoke(const Location& posted_from,
>                    const Location& callback_posted_from,
>-                   const FunctorT& functor,
>+                   FunctorT&& functor,
>                    void (HostT::*callback)(),
>                    HostT* callback_host,
>                    uint32_t id = 0) {
>@@ -238,8 +246,8 @@ class GuardedAsyncInvoker : public sigslot::has_slots<> {
>     if (thread_ == nullptr)
>       return false;
>     invoker_.AsyncInvoke<ReturnT, FunctorT, HostT>(
>-        posted_from, callback_posted_from, thread_, functor, callback,
>-        callback_host, id);
>+        posted_from, callback_posted_from, thread_,
>+        std::forward<FunctorT>(functor), callback, callback_host, id);
>     return true;
>   }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncpacketsocket.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncpacketsocket.cc
>index d9450398a51..e1b1eae6a31 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncpacketsocket.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncpacketsocket.cc
>@@ -12,18 +12,31 @@
> 
> namespace rtc {
> 
>-PacketTimeUpdateParams::PacketTimeUpdateParams()
>-    : rtp_sendtime_extension_id(-1),
>-      srtp_auth_tag_len(-1),
>-      srtp_packet_index(-1) {
>-}
>+PacketTimeUpdateParams::PacketTimeUpdateParams() = default;
>+
>+PacketTimeUpdateParams::PacketTimeUpdateParams(
>+    const PacketTimeUpdateParams& other) = default;
> 
> PacketTimeUpdateParams::~PacketTimeUpdateParams() = default;
> 
>-AsyncPacketSocket::AsyncPacketSocket() {
>-}
>+PacketOptions::PacketOptions() = default;
>+PacketOptions::PacketOptions(DiffServCodePoint dscp) : dscp(dscp) {}
>+PacketOptions::PacketOptions(const PacketOptions& other) = default;
>+PacketOptions::~PacketOptions() = default;
>+
>+AsyncPacketSocket::AsyncPacketSocket() = default;
>+
>+AsyncPacketSocket::~AsyncPacketSocket() = default;
> 
>-AsyncPacketSocket::~AsyncPacketSocket() {
>+void CopySocketInformationToPacketInfo(size_t packet_size_bytes,
>+                                       const AsyncPacketSocket& socket_from,
>+                                       bool is_connectionless,
>+                                       rtc::PacketInfo* info) {
>+  info->packet_size_bytes = packet_size_bytes;
>+  info->local_socket_address = socket_from.GetLocalAddress();
>+  if (!is_connectionless) {
>+    info->remote_socket_address = socket_from.GetRemoteAddress();
>+  }
> }
> 
> };  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncpacketsocket.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncpacketsocket.h
>index 16f4de08229..bb0b3bcc048 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncpacketsocket.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncpacketsocket.h
>@@ -13,8 +13,8 @@
> 
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/dscp.h"
>-#include "rtc_base/sigslot.h"
> #include "rtc_base/socket.h"
>+#include "rtc_base/third_party/sigslot/sigslot.h"
> #include "rtc_base/timeutils.h"
> 
> namespace rtc {
>@@ -24,23 +24,30 @@ namespace rtc {
> // after changing the value.
> struct PacketTimeUpdateParams {
>   PacketTimeUpdateParams();
>+  PacketTimeUpdateParams(const PacketTimeUpdateParams& other);
>   ~PacketTimeUpdateParams();
> 
>-  int rtp_sendtime_extension_id;    // extension header id present in packet.
>-  std::vector<char> srtp_auth_key;  // Authentication key.
>-  int srtp_auth_tag_len;            // Authentication tag length.
>-  int64_t srtp_packet_index;        // Required for Rtp Packet authentication.
>+  int rtp_sendtime_extension_id = -1;  // extension header id present in packet.
>+  std::vector<char> srtp_auth_key;     // Authentication key.
>+  int srtp_auth_tag_len = -1;          // Authentication tag length.
>+  int64_t srtp_packet_index = -1;  // Required for Rtp Packet authentication.
> };
> 
> // This structure holds meta information for the packet which is about to send
> // over network.
> struct PacketOptions {
>-  PacketOptions() : dscp(DSCP_NO_CHANGE), packet_id(-1) {}
>-  explicit PacketOptions(DiffServCodePoint dscp) : dscp(dscp), packet_id(-1) {}
>-
>-  DiffServCodePoint dscp;
>-  int packet_id;  // 16 bits, -1 represents "not set".
>+  PacketOptions();
>+  explicit PacketOptions(DiffServCodePoint dscp);
>+  PacketOptions(const PacketOptions& other);
>+  ~PacketOptions();
>+
>+  DiffServCodePoint dscp = DSCP_NO_CHANGE;
>+  // When used with RTP packets (for example, webrtc::PacketOptions), the value
>+  // should be 16 bits. A value of -1 represents "not set".
>+  int64_t packet_id = -1;
>   PacketTimeUpdateParams packet_time_params;
>+  // PacketInfo is passed to SentPacket when signaling this packet is sent.
>+  PacketInfo info_signaled_after_sent;
> };
> 
> // This structure will have the information about when packet is actually
>@@ -50,7 +57,7 @@ struct PacketTime {
>   PacketTime(int64_t timestamp, int64_t not_before)
>       : timestamp(timestamp), not_before(not_before) {}
> 
>-  int64_t timestamp;   // Receive time after socket delivers the data.
>+  int64_t timestamp;  // Receive time after socket delivers the data.
> 
>   // Earliest possible time the data could have arrived, indicating the
>   // potential error in the |timestamp| value, in case the system, is busy. For
>@@ -86,8 +93,10 @@ class AsyncPacketSocket : public sigslot::has_slots<> {
>   virtual SocketAddress GetRemoteAddress() const = 0;
> 
>   // Send a packet.
>-  virtual int Send(const void *pv, size_t cb, const PacketOptions& options) = 0;
>-  virtual int SendTo(const void *pv, size_t cb, const SocketAddress& addr,
>+  virtual int Send(const void* pv, size_t cb, const PacketOptions& options) = 0;
>+  virtual int SendTo(const void* pv,
>+                     size_t cb,
>+                     const SocketAddress& addr,
>                      const PacketOptions& options) = 0;
> 
>   // Close the socket.
>@@ -107,9 +116,12 @@ class AsyncPacketSocket : public sigslot::has_slots<> {
> 
>   // Emitted each time a packet is read. Used only for UDP and
>   // connected TCP sockets.
>-  sigslot::signal5<AsyncPacketSocket*, const char*, size_t,
>+  sigslot::signal5<AsyncPacketSocket*,
>+                   const char*,
>+                   size_t,
>                    const SocketAddress&,
>-                   const PacketTime&> SignalReadPacket;
>+                   const PacketTime&>
>+      SignalReadPacket;
> 
>   // Emitted each time a packet is sent.
>   sigslot::signal2<AsyncPacketSocket*, const SentPacket&> SignalSentPacket;
>@@ -138,6 +150,11 @@ class AsyncPacketSocket : public sigslot::has_slots<> {
>   RTC_DISALLOW_COPY_AND_ASSIGN(AsyncPacketSocket);
> };
> 
>+void CopySocketInformationToPacketInfo(size_t packet_size_bytes,
>+                                       const AsyncPacketSocket& socket_from,
>+                                       bool is_connectionless,
>+                                       rtc::PacketInfo* info);
>+
> }  // namespace rtc
> 
> #endif  // RTC_BASE_ASYNCPACKETSOCKET_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncresolverinterface.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncresolverinterface.cc
>index 62dd36ac8d9..b2880f28782 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncresolverinterface.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncresolverinterface.cc
>@@ -12,8 +12,7 @@
> 
> namespace rtc {
> 
>-AsyncResolverInterface::AsyncResolverInterface() {
>-}
>+AsyncResolverInterface::AsyncResolverInterface() {}
> 
> AsyncResolverInterface::~AsyncResolverInterface() = default;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncresolverinterface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncresolverinterface.h
>index 96b5ec18b90..5b2303f4d80 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncresolverinterface.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncresolverinterface.h
>@@ -11,8 +11,8 @@
> #ifndef RTC_BASE_ASYNCRESOLVERINTERFACE_H_
> #define RTC_BASE_ASYNCRESOLVERINTERFACE_H_
> 
>-#include "rtc_base/sigslot.h"
> #include "rtc_base/socketaddress.h"
>+#include "rtc_base/third_party/sigslot/sigslot.h"
> 
> namespace rtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncsocket.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncsocket.cc
>index b28b2f9c443..acd5415eb5a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncsocket.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncsocket.cc
>@@ -13,11 +13,9 @@
> 
> namespace rtc {
> 
>-AsyncSocket::AsyncSocket() {
>-}
>+AsyncSocket::AsyncSocket() {}
> 
>-AsyncSocket::~AsyncSocket() {
>-}
>+AsyncSocket::~AsyncSocket() {}
> 
> AsyncSocketAdapter::AsyncSocketAdapter(AsyncSocket* socket) : socket_(nullptr) {
>   Attach(socket);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncsocket.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncsocket.h
>index c018c23f077..0abdc27750e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncsocket.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncsocket.h
>@@ -11,8 +11,8 @@
> #ifndef RTC_BASE_ASYNCSOCKET_H_
> #define RTC_BASE_ASYNCSOCKET_H_
> 
>-#include "rtc_base/sigslot.h"
> #include "rtc_base/socket.h"
>+#include "rtc_base/third_party/sigslot/sigslot.h"
> 
> namespace rtc {
> 
>@@ -31,11 +31,10 @@ class AsyncSocket : public Socket {
>   // For example SignalReadEvent::connect will be called in AsyncUDPSocket ctor
>   // but at the same time the SocketDispatcher maybe signaling the read event.
>   // ready to read
>-  sigslot::signal1<AsyncSocket*,
>-                   sigslot::multi_threaded_local> SignalReadEvent;
>+  sigslot::signal1<AsyncSocket*, sigslot::multi_threaded_local> SignalReadEvent;
>   // ready to write
>-  sigslot::signal1<AsyncSocket*,
>-                   sigslot::multi_threaded_local> SignalWriteEvent;
>+  sigslot::signal1<AsyncSocket*, sigslot::multi_threaded_local>
>+      SignalWriteEvent;
>   sigslot::signal1<AsyncSocket*> SignalConnectEvent;     // connected
>   sigslot::signal2<AsyncSocket*, int> SignalCloseEvent;  // closed
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asynctcpsocket.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asynctcpsocket.cc
>index 9e0589c194e..3d68a2a46a6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asynctcpsocket.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asynctcpsocket.cc
>@@ -56,7 +56,8 @@ AsyncSocket* AsyncTCPSocketBase::ConnectSocket(
>   return owned_socket.release();
> }
> 
>-AsyncTCPSocketBase::AsyncTCPSocketBase(AsyncSocket* socket, bool listen,
>+AsyncTCPSocketBase::AsyncTCPSocketBase(AsyncSocket* socket,
>+                                       bool listen,
>                                        size_t max_packet_size)
>     : socket_(socket),
>       listen_(listen),
>@@ -68,8 +69,8 @@ AsyncTCPSocketBase::AsyncTCPSocketBase(AsyncSocket* socket, bool listen,
>   }
> 
>   RTC_DCHECK(socket_.get() != nullptr);
>-  socket_->SignalConnectEvent.connect(
>-      this, &AsyncTCPSocketBase::OnConnectEvent);
>+  socket_->SignalConnectEvent.connect(this,
>+                                      &AsyncTCPSocketBase::OnConnectEvent);
>   socket_->SignalReadEvent.connect(this, &AsyncTCPSocketBase::OnReadEvent);
>   socket_->SignalWriteEvent.connect(this, &AsyncTCPSocketBase::OnWriteEvent);
>   socket_->SignalCloseEvent.connect(this, &AsyncTCPSocketBase::OnCloseEvent);
>@@ -129,7 +130,8 @@ void AsyncTCPSocketBase::SetError(int error) {
>   return socket_->SetError(error);
> }
> 
>-int AsyncTCPSocketBase::SendTo(const void *pv, size_t cb,
>+int AsyncTCPSocketBase::SendTo(const void* pv,
>+                               size_t cb,
>                                const SocketAddress& addr,
>                                const rtc::PacketOptions& options) {
>   const SocketAddress& remote_address = GetRemoteAddress();
>@@ -141,7 +143,7 @@ int AsyncTCPSocketBase::SendTo(const void *pv, size_t cb,
>   return -1;
> }
> 
>-int AsyncTCPSocketBase::SendRaw(const void * pv, size_t cb) {
>+int AsyncTCPSocketBase::SendRaw(const void* pv, size_t cb) {
>   if (outbuf_.size() + cb > max_outsize_) {
>     socket_->SetError(EMSGSIZE);
>     return -1;
>@@ -263,19 +265,19 @@ void AsyncTCPSocketBase::OnCloseEvent(AsyncSocket* socket, int error) {
> // Binds and connects |socket| and creates AsyncTCPSocket for
> // it. Takes ownership of |socket|. Returns null if bind() or
> // connect() fail (|socket| is destroyed in that case).
>-AsyncTCPSocket* AsyncTCPSocket::Create(
>-    AsyncSocket* socket,
>-    const SocketAddress& bind_address,
>-    const SocketAddress& remote_address) {
>-  return new AsyncTCPSocket(AsyncTCPSocketBase::ConnectSocket(
>-      socket, bind_address, remote_address), false);
>+AsyncTCPSocket* AsyncTCPSocket::Create(AsyncSocket* socket,
>+                                       const SocketAddress& bind_address,
>+                                       const SocketAddress& remote_address) {
>+  return new AsyncTCPSocket(
>+      AsyncTCPSocketBase::ConnectSocket(socket, bind_address, remote_address),
>+      false);
> }
> 
> AsyncTCPSocket::AsyncTCPSocket(AsyncSocket* socket, bool listen)
>-    : AsyncTCPSocketBase(socket, listen, kBufSize) {
>-}
>+    : AsyncTCPSocketBase(socket, listen, kBufSize) {}
> 
>-int AsyncTCPSocket::Send(const void *pv, size_t cb,
>+int AsyncTCPSocket::Send(const void* pv,
>+                         size_t cb,
>                          const rtc::PacketOptions& options) {
>   if (cb > kBufSize) {
>     SetError(EMSGSIZE);
>@@ -297,14 +299,16 @@ int AsyncTCPSocket::Send(const void *pv, size_t cb,
>     return res;
>   }
> 
>-  rtc::SentPacket sent_packet(options.packet_id, rtc::TimeMillis());
>+  rtc::SentPacket sent_packet(options.packet_id, rtc::TimeMillis(),
>+                              options.info_signaled_after_sent);
>+  CopySocketInformationToPacketInfo(cb, *this, false, &sent_packet.info);
>   SignalSentPacket(this, sent_packet);
> 
>   // We claim to have sent the whole thing, even if we only sent partial
>   return static_cast<int>(cb);
> }
> 
>-void AsyncTCPSocket::ProcessInput(char * data, size_t* len) {
>+void AsyncTCPSocket::ProcessInput(char* data, size_t* len) {
>   SocketAddress remote_addr(GetRemoteAddress());
> 
>   while (true) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asynctcpsocket.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asynctcpsocket.h
>index 0a548d04cb3..943e010f4da 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asynctcpsocket.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asynctcpsocket.h
>@@ -29,8 +29,9 @@ class AsyncTCPSocketBase : public AsyncPacketSocket {
>   ~AsyncTCPSocketBase() override;
> 
>   // Pure virtual methods to send and recv data.
>-  int Send(const void *pv, size_t cb,
>-                   const rtc::PacketOptions& options) override = 0;
>+  int Send(const void* pv,
>+           size_t cb,
>+           const rtc::PacketOptions& options) override = 0;
>   virtual void ProcessInput(char* data, size_t* len) = 0;
>   // Signals incoming connection.
>   virtual void HandleIncomingConnection(AsyncSocket* socket) = 0;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asynctcpsocket_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asynctcpsocket_unittest.cc
>index 70814119303..e8fd96cb8bb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asynctcpsocket_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asynctcpsocket_unittest.cc
>@@ -17,9 +17,7 @@
> 
> namespace rtc {
> 
>-class AsyncTCPSocketTest
>-    : public testing::Test,
>-      public sigslot::has_slots<> {
>+class AsyncTCPSocketTest : public testing::Test, public sigslot::has_slots<> {
>  public:
>   AsyncTCPSocketTest()
>       : vss_(new rtc::VirtualSocketServer()),
>@@ -30,9 +28,7 @@ class AsyncTCPSocketTest
>                                            &AsyncTCPSocketTest::OnReadyToSend);
>   }
> 
>-  void OnReadyToSend(rtc::AsyncPacketSocket* socket) {
>-    ready_to_send_ = true;
>-  }
>+  void OnReadyToSend(rtc::AsyncPacketSocket* socket) { ready_to_send_ = true; }
> 
>  protected:
>   std::unique_ptr<VirtualSocketServer> vss_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncudpsocket.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncudpsocket.cc
>index 0896e50f496..ba5fa88aadb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncudpsocket.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncudpsocket.cc
>@@ -16,9 +16,8 @@ namespace rtc {
> 
> static const int BUF_SIZE = 64 * 1024;
> 
>-AsyncUDPSocket* AsyncUDPSocket::Create(
>-    AsyncSocket* socket,
>-    const SocketAddress& bind_address) {
>+AsyncUDPSocket* AsyncUDPSocket::Create(AsyncSocket* socket,
>+                                       const SocketAddress& bind_address) {
>   std::unique_ptr<AsyncSocket> owned_socket(socket);
>   if (socket->Bind(bind_address) < 0) {
>     RTC_LOG(LS_ERROR) << "Bind() failed with error " << socket->GetError();
>@@ -36,8 +35,7 @@ AsyncUDPSocket* AsyncUDPSocket::Create(SocketFactory* factory,
>   return Create(socket, bind_address);
> }
> 
>-AsyncUDPSocket::AsyncUDPSocket(AsyncSocket* socket)
>-    : socket_(socket) {
>+AsyncUDPSocket::AsyncUDPSocket(AsyncSocket* socket) : socket_(socket) {
>   size_ = BUF_SIZE;
>   buf_ = new char[size_];
> 
>@@ -47,7 +45,7 @@ AsyncUDPSocket::AsyncUDPSocket(AsyncSocket* socket)
> }
> 
> AsyncUDPSocket::~AsyncUDPSocket() {
>-  delete [] buf_;
>+  delete[] buf_;
> }
> 
> SocketAddress AsyncUDPSocket::GetLocalAddress() const {
>@@ -58,18 +56,25 @@ SocketAddress AsyncUDPSocket::GetRemoteAddress() const {
>   return socket_->GetRemoteAddress();
> }
> 
>-int AsyncUDPSocket::Send(const void *pv, size_t cb,
>+int AsyncUDPSocket::Send(const void* pv,
>+                         size_t cb,
>                          const rtc::PacketOptions& options) {
>-  rtc::SentPacket sent_packet(options.packet_id, rtc::TimeMillis());
>+  rtc::SentPacket sent_packet(options.packet_id, rtc::TimeMillis(),
>+                              options.info_signaled_after_sent);
>+  CopySocketInformationToPacketInfo(cb, *this, false, &sent_packet.info);
>   int ret = socket_->Send(pv, cb);
>   SignalSentPacket(this, sent_packet);
>   return ret;
> }
> 
>-int AsyncUDPSocket::SendTo(const void *pv, size_t cb,
>+int AsyncUDPSocket::SendTo(const void* pv,
>+                           size_t cb,
>                            const SocketAddress& addr,
>                            const rtc::PacketOptions& options) {
>-  rtc::SentPacket sent_packet(options.packet_id, rtc::TimeMillis());
>+  rtc::SentPacket sent_packet(options.packet_id, rtc::TimeMillis(),
>+                              options.info_signaled_after_sent);
>+  CopySocketInformationToPacketInfo(cb, *this, true, &sent_packet.info);
>+  sent_packet.info.remote_socket_address = addr;
>   int ret = socket_->SendTo(pv, cb, addr);
>   SignalSentPacket(this, sent_packet);
>   return ret;
>@@ -112,8 +117,7 @@ void AsyncUDPSocket::OnReadEvent(AsyncSocket* socket) {
>     // TODO: Do something better like forwarding the error to the user.
>     SocketAddress local_addr = socket_->GetLocalAddress();
>     RTC_LOG(LS_INFO) << "AsyncUDPSocket[" << local_addr.ToSensitiveString()
>-                     << "] "
>-                     << "receive failed with error " << socket_->GetError();
>+                     << "] receive failed with error " << socket_->GetError();
>     return;
>   }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncudpsocket_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncudpsocket_unittest.cc
>index af7cc20c6a0..33b3d136717 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncudpsocket_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/asyncudpsocket_unittest.cc
>@@ -18,9 +18,7 @@
> 
> namespace rtc {
> 
>-class AsyncUdpSocketTest
>-    : public testing::Test,
>-      public sigslot::has_slots<> {
>+class AsyncUdpSocketTest : public testing::Test, public sigslot::has_slots<> {
>  public:
>   AsyncUdpSocketTest()
>       : pss_(new rtc::PhysicalSocketServer),
>@@ -32,9 +30,7 @@ class AsyncUdpSocketTest
>                                            &AsyncUdpSocketTest::OnReadyToSend);
>   }
> 
>-  void OnReadyToSend(rtc::AsyncPacketSocket* socket) {
>-    ready_to_send_ = true;
>-  }
>+  void OnReadyToSend(rtc::AsyncPacketSocket* socket) { ready_to_send_ = true; }
> 
>  protected:
>   std::unique_ptr<PhysicalSocketServer> pss_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/atomicops.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/atomicops.h
>index c0ff1a616b1..16fa603a4ed 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/atomicops.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/atomicops.h
>@@ -12,12 +12,15 @@
> #define RTC_BASE_ATOMICOPS_H_
> 
> #if defined(WEBRTC_WIN)
>+// clang-format off
>+// clang formating would change include order.
>+
> // Include winsock2.h before including <windows.h> to maintain consistency with
>-// win32.h.  We can't include win32.h directly here since it pulls in
>-// headers such as basictypes.h which causes problems in Chromium where webrtc
>-// exists as two separate projects, webrtc and libjingle.
>+// win32.h. To include win32.h directly, it must be broken out into its own
>+// build target.
> #include <winsock2.h>
> #include <windows.h>
>+// clang-format on
> #endif  // defined(WEBRTC_WIN)
> 
> namespace rtc {
>@@ -31,16 +34,11 @@ class AtomicOps {
>   static int Decrement(volatile int* i) {
>     return ::InterlockedDecrement(reinterpret_cast<volatile LONG*>(i));
>   }
>-  static int AcquireLoad(volatile const int* i) {
>-    return *i;
>-  }
>-  static void ReleaseStore(volatile int* i, int value) {
>-    *i = value;
>-  }
>+  static int AcquireLoad(volatile const int* i) { return *i; }
>+  static void ReleaseStore(volatile int* i, int value) { *i = value; }
>   static int CompareAndSwap(volatile int* i, int old_value, int new_value) {
>     return ::InterlockedCompareExchange(reinterpret_cast<volatile LONG*>(i),
>-                                        new_value,
>-                                        old_value);
>+                                        new_value, old_value);
>   }
>   // Pointer variants.
>   template <typename T>
>@@ -53,12 +51,8 @@ class AtomicOps {
>         reinterpret_cast<PVOID volatile*>(ptr), new_value, old_value));
>   }
> #else
>-  static int Increment(volatile int* i) {
>-    return __sync_add_and_fetch(i, 1);
>-  }
>-  static int Decrement(volatile int* i) {
>-    return __sync_sub_and_fetch(i, 1);
>-  }
>+  static int Increment(volatile int* i) { return __sync_add_and_fetch(i, 1); }
>+  static int Decrement(volatile int* i) { return __sync_sub_and_fetch(i, 1); }
>   static int AcquireLoad(volatile const int* i) {
>     return __atomic_load_n(i, __ATOMIC_ACQUIRE);
>   }
>@@ -80,8 +74,6 @@ class AtomicOps {
> #endif
> };
> 
>-
>-
>-}
>+}  // namespace rtc
> 
> #endif  // RTC_BASE_ATOMICOPS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/base64_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/base64_unittest.cc
>index 0f7c80da447..4b857f1aa94 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/base64_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/base64_unittest.cc
>@@ -8,7 +8,7 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "rtc_base/base64.h"
>+#include "rtc_base/third_party/base64/base64.h"
> #include "rtc_base/gunit.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/stringutils.h"
>@@ -24,257 +24,257 @@ static struct {
>   const char* cyphertext;
> } base64_tests[] = {
> 
>-  // Basic bit patterns;
>-  // values obtained with "echo -n '...' | uuencode -m test"
>-
>-  { 1, "\000", "AA==" },
>-  { 1, "\001", "AQ==" },
>-  { 1, "\002", "Ag==" },
>-  { 1, "\004", "BA==" },
>-  { 1, "\010", "CA==" },
>-  { 1, "\020", "EA==" },
>-  { 1, "\040", "IA==" },
>-  { 1, "\100", "QA==" },
>-  { 1, "\200", "gA==" },
>-
>-  { 1, "\377", "/w==" },
>-  { 1, "\376", "/g==" },
>-  { 1, "\375", "/Q==" },
>-  { 1, "\373", "+w==" },
>-  { 1, "\367", "9w==" },
>-  { 1, "\357", "7w==" },
>-  { 1, "\337", "3w==" },
>-  { 1, "\277", "vw==" },
>-  { 1, "\177", "fw==" },
>-  { 2, "\000\000", "AAA=" },
>-  { 2, "\000\001", "AAE=" },
>-  { 2, "\000\002", "AAI=" },
>-  { 2, "\000\004", "AAQ=" },
>-  { 2, "\000\010", "AAg=" },
>-  { 2, "\000\020", "ABA=" },
>-  { 2, "\000\040", "ACA=" },
>-  { 2, "\000\100", "AEA=" },
>-  { 2, "\000\200", "AIA=" },
>-  { 2, "\001\000", "AQA=" },
>-  { 2, "\002\000", "AgA=" },
>-  { 2, "\004\000", "BAA=" },
>-  { 2, "\010\000", "CAA=" },
>-  { 2, "\020\000", "EAA=" },
>-  { 2, "\040\000", "IAA=" },
>-  { 2, "\100\000", "QAA=" },
>-  { 2, "\200\000", "gAA=" },
>-
>-  { 2, "\377\377", "//8=" },
>-  { 2, "\377\376", "//4=" },
>-  { 2, "\377\375", "//0=" },
>-  { 2, "\377\373", "//s=" },
>-  { 2, "\377\367", "//c=" },
>-  { 2, "\377\357", "/+8=" },
>-  { 2, "\377\337", "/98=" },
>-  { 2, "\377\277", "/78=" },
>-  { 2, "\377\177", "/38=" },
>-  { 2, "\376\377", "/v8=" },
>-  { 2, "\375\377", "/f8=" },
>-  { 2, "\373\377", "+/8=" },
>-  { 2, "\367\377", "9/8=" },
>-  { 2, "\357\377", "7/8=" },
>-  { 2, "\337\377", "3/8=" },
>-  { 2, "\277\377", "v/8=" },
>-  { 2, "\177\377", "f/8=" },
>-
>-  { 3, "\000\000\000", "AAAA" },
>-  { 3, "\000\000\001", "AAAB" },
>-  { 3, "\000\000\002", "AAAC" },
>-  { 3, "\000\000\004", "AAAE" },
>-  { 3, "\000\000\010", "AAAI" },
>-  { 3, "\000\000\020", "AAAQ" },
>-  { 3, "\000\000\040", "AAAg" },
>-  { 3, "\000\000\100", "AABA" },
>-  { 3, "\000\000\200", "AACA" },
>-  { 3, "\000\001\000", "AAEA" },
>-  { 3, "\000\002\000", "AAIA" },
>-  { 3, "\000\004\000", "AAQA" },
>-  { 3, "\000\010\000", "AAgA" },
>-  { 3, "\000\020\000", "ABAA" },
>-  { 3, "\000\040\000", "ACAA" },
>-  { 3, "\000\100\000", "AEAA" },
>-  { 3, "\000\200\000", "AIAA" },
>-  { 3, "\001\000\000", "AQAA" },
>-  { 3, "\002\000\000", "AgAA" },
>-  { 3, "\004\000\000", "BAAA" },
>-  { 3, "\010\000\000", "CAAA" },
>-  { 3, "\020\000\000", "EAAA" },
>-  { 3, "\040\000\000", "IAAA" },
>-  { 3, "\100\000\000", "QAAA" },
>-  { 3, "\200\000\000", "gAAA" },
>-
>-  { 3, "\377\377\377", "////" },
>-  { 3, "\377\377\376", "///+" },
>-  { 3, "\377\377\375", "///9" },
>-  { 3, "\377\377\373", "///7" },
>-  { 3, "\377\377\367", "///3" },
>-  { 3, "\377\377\357", "///v" },
>-  { 3, "\377\377\337", "///f" },
>-  { 3, "\377\377\277", "//+/" },
>-  { 3, "\377\377\177", "//9/" },
>-  { 3, "\377\376\377", "//7/" },
>-  { 3, "\377\375\377", "//3/" },
>-  { 3, "\377\373\377", "//v/" },
>-  { 3, "\377\367\377", "//f/" },
>-  { 3, "\377\357\377", "/+//" },
>-  { 3, "\377\337\377", "/9//" },
>-  { 3, "\377\277\377", "/7//" },
>-  { 3, "\377\177\377", "/3//" },
>-  { 3, "\376\377\377", "/v//" },
>-  { 3, "\375\377\377", "/f//" },
>-  { 3, "\373\377\377", "+///" },
>-  { 3, "\367\377\377", "9///" },
>-  { 3, "\357\377\377", "7///" },
>-  { 3, "\337\377\377", "3///" },
>-  { 3, "\277\377\377", "v///" },
>-  { 3, "\177\377\377", "f///" },
>-
>-  // Random numbers: values obtained with
>-  //
>-  //  #! /bin/bash
>-  //  dd bs=$1 count=1 if=/dev/random of=/tmp/bar.random
>-  //  od -N $1 -t o1 /tmp/bar.random
>-  //  uuencode -m test < /tmp/bar.random
>-  //
>-  // where $1 is the number of bytes (2, 3)
>-
>-  { 2, "\243\361", "o/E=" },
>-  { 2, "\024\167", "FHc=" },
>-  { 2, "\313\252", "y6o=" },
>-  { 2, "\046\041", "JiE=" },
>-  { 2, "\145\236", "ZZ4=" },
>-  { 2, "\254\325", "rNU=" },
>-  { 2, "\061\330", "Mdg=" },
>-  { 2, "\245\032", "pRo=" },
>-  { 2, "\006\000", "BgA=" },
>-  { 2, "\375\131", "/Vk=" },
>-  { 2, "\303\210", "w4g=" },
>-  { 2, "\040\037", "IB8=" },
>-  { 2, "\261\372", "sfo=" },
>-  { 2, "\335\014", "3Qw=" },
>-  { 2, "\233\217", "m48=" },
>-  { 2, "\373\056", "+y4=" },
>-  { 2, "\247\232", "p5o=" },
>-  { 2, "\107\053", "Rys=" },
>-  { 2, "\204\077", "hD8=" },
>-  { 2, "\276\211", "vok=" },
>-  { 2, "\313\110", "y0g=" },
>-  { 2, "\363\376", "8/4=" },
>-  { 2, "\251\234", "qZw=" },
>-  { 2, "\103\262", "Q7I=" },
>-  { 2, "\142\312", "Yso=" },
>-  { 2, "\067\211", "N4k=" },
>-  { 2, "\220\001", "kAE=" },
>-  { 2, "\152\240", "aqA=" },
>-  { 2, "\367\061", "9zE=" },
>-  { 2, "\133\255", "W60=" },
>-  { 2, "\176\035", "fh0=" },
>-  { 2, "\032\231", "Gpk=" },
>-
>-  { 3, "\013\007\144", "Cwdk" },
>-  { 3, "\030\112\106", "GEpG" },
>-  { 3, "\047\325\046", "J9Um" },
>-  { 3, "\310\160\022", "yHAS" },
>-  { 3, "\131\100\237", "WUCf" },
>-  { 3, "\064\342\134", "NOJc" },
>-  { 3, "\010\177\004", "CH8E" },
>-  { 3, "\345\147\205", "5WeF" },
>-  { 3, "\300\343\360", "wOPw" },
>-  { 3, "\061\240\201", "MaCB" },
>-  { 3, "\225\333\044", "ldsk" },
>-  { 3, "\215\137\352", "jV/q" },
>-  { 3, "\371\147\160", "+Wdw" },
>-  { 3, "\030\320\051", "GNAp" },
>-  { 3, "\044\174\241", "JHyh" },
>-  { 3, "\260\127\037", "sFcf" },
>-  { 3, "\111\045\033", "SSUb" },
>-  { 3, "\202\114\107", "gkxH" },
>-  { 3, "\057\371\042", "L/ki" },
>-  { 3, "\223\247\244", "k6ek" },
>-  { 3, "\047\216\144", "J45k" },
>-  { 3, "\203\070\327", "gzjX" },
>-  { 3, "\247\140\072", "p2A6" },
>-  { 3, "\124\115\116", "VE1O" },
>-  { 3, "\157\162\050", "b3Io" },
>-  { 3, "\357\223\004", "75ME" },
>-  { 3, "\052\117\156", "Kk9u" },
>-  { 3, "\347\154\000", "52wA" },
>-  { 3, "\303\012\142", "wwpi" },
>-  { 3, "\060\035\362", "MB3y" },
>-  { 3, "\130\226\361", "WJbx" },
>-  { 3, "\173\013\071", "ews5" },
>-  { 3, "\336\004\027", "3gQX" },
>-  { 3, "\357\366\234", "7/ac" },
>-  { 3, "\353\304\111", "68RJ" },
>-  { 3, "\024\264\131", "FLRZ" },
>-  { 3, "\075\114\251", "PUyp" },
>-  { 3, "\315\031\225", "zRmV" },
>-  { 3, "\154\201\276", "bIG+" },
>-  { 3, "\200\066\072", "gDY6" },
>-  { 3, "\142\350\267", "Yui3" },
>-  { 3, "\033\000\166", "GwB2" },
>-  { 3, "\210\055\077", "iC0/" },
>-  { 3, "\341\037\124", "4R9U" },
>-  { 3, "\161\103\152", "cUNq" },
>-  { 3, "\270\142\131", "uGJZ" },
>-  { 3, "\337\076\074", "3z48" },
>-  { 3, "\375\106\362", "/Uby" },
>-  { 3, "\227\301\127", "l8FX" },
>-  { 3, "\340\002\234", "4AKc" },
>-  { 3, "\121\064\033", "UTQb" },
>-  { 3, "\157\134\143", "b1xj" },
>-  { 3, "\247\055\327", "py3X" },
>-  { 3, "\340\142\005", "4GIF" },
>-  { 3, "\060\260\143", "MLBj" },
>-  { 3, "\075\203\170", "PYN4" },
>-  { 3, "\143\160\016", "Y3AO" },
>-  { 3, "\313\013\063", "ywsz" },
>-  { 3, "\174\236\135", "fJ5d" },
>-  { 3, "\103\047\026", "QycW" },
>-  { 3, "\365\005\343", "9QXj" },
>-  { 3, "\271\160\223", "uXCT" },
>-  { 3, "\362\255\172", "8q16" },
>-  { 3, "\113\012\015", "SwoN" },
>-
>-  // various lengths, generated by this python script:
>-  //
>-  // from string import lowercase as lc
>-  // for i in range(27):
>-  //   print '{ %2d, "%s",%s "%s" },' % (i, lc[:i], ' ' * (26-i),
>-  //                                     lc[:i].encode('base64').strip())
>-
>-  {  0, "abcdefghijklmnopqrstuvwxyz", "" },
>-  {  1, "abcdefghijklmnopqrstuvwxyz", "YQ==" },
>-  {  2, "abcdefghijklmnopqrstuvwxyz", "YWI=" },
>-  {  3, "abcdefghijklmnopqrstuvwxyz", "YWJj" },
>-  {  4, "abcdefghijklmnopqrstuvwxyz", "YWJjZA==" },
>-  {  5, "abcdefghijklmnopqrstuvwxyz", "YWJjZGU=" },
>-  {  6, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVm" },
>-  {  7, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZw==" },
>-  {  8, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2g=" },
>-  {  9, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2hp" },
>-  { 10, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2hpag==" },
>-  { 11, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2hpams=" },
>-  { 12, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2hpamts" },
>-  { 13, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2hpamtsbQ==" },
>-  { 14, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2hpamtsbW4=" },
>-  { 15, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2hpamtsbW5v" },
>-  { 16, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2hpamtsbW5vcA==" },
>-  { 17, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2hpamtsbW5vcHE=" },
>-  { 18, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2hpamtsbW5vcHFy" },
>-  { 19, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2hpamtsbW5vcHFycw==" },
>-  { 20, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2hpamtsbW5vcHFyc3Q=" },
>-  { 21, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2hpamtsbW5vcHFyc3R1" },
>-  { 22, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2hpamtsbW5vcHFyc3R1dg==" },
>-  { 23, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2hpamtsbW5vcHFyc3R1dnc=" },
>-  { 24, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2hpamtsbW5vcHFyc3R1dnd4" },
>-  { 25, "abcdefghijklmnopqrstuvwxy",  "YWJjZGVmZ2hpamtsbW5vcHFyc3R1dnd4eQ==" },
>-  { 26, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2hpamtsbW5vcHFyc3R1dnd4eXo=" },
>+    // Basic bit patterns;
>+    // values obtained with "echo -n '...' | uuencode -m test"
>+
>+    {1, "\000", "AA=="},
>+    {1, "\001", "AQ=="},
>+    {1, "\002", "Ag=="},
>+    {1, "\004", "BA=="},
>+    {1, "\010", "CA=="},
>+    {1, "\020", "EA=="},
>+    {1, "\040", "IA=="},
>+    {1, "\100", "QA=="},
>+    {1, "\200", "gA=="},
>+
>+    {1, "\377", "/w=="},
>+    {1, "\376", "/g=="},
>+    {1, "\375", "/Q=="},
>+    {1, "\373", "+w=="},
>+    {1, "\367", "9w=="},
>+    {1, "\357", "7w=="},
>+    {1, "\337", "3w=="},
>+    {1, "\277", "vw=="},
>+    {1, "\177", "fw=="},
>+    {2, "\000\000", "AAA="},
>+    {2, "\000\001", "AAE="},
>+    {2, "\000\002", "AAI="},
>+    {2, "\000\004", "AAQ="},
>+    {2, "\000\010", "AAg="},
>+    {2, "\000\020", "ABA="},
>+    {2, "\000\040", "ACA="},
>+    {2, "\000\100", "AEA="},
>+    {2, "\000\200", "AIA="},
>+    {2, "\001\000", "AQA="},
>+    {2, "\002\000", "AgA="},
>+    {2, "\004\000", "BAA="},
>+    {2, "\010\000", "CAA="},
>+    {2, "\020\000", "EAA="},
>+    {2, "\040\000", "IAA="},
>+    {2, "\100\000", "QAA="},
>+    {2, "\200\000", "gAA="},
>+
>+    {2, "\377\377", "//8="},
>+    {2, "\377\376", "//4="},
>+    {2, "\377\375", "//0="},
>+    {2, "\377\373", "//s="},
>+    {2, "\377\367", "//c="},
>+    {2, "\377\357", "/+8="},
>+    {2, "\377\337", "/98="},
>+    {2, "\377\277", "/78="},
>+    {2, "\377\177", "/38="},
>+    {2, "\376\377", "/v8="},
>+    {2, "\375\377", "/f8="},
>+    {2, "\373\377", "+/8="},
>+    {2, "\367\377", "9/8="},
>+    {2, "\357\377", "7/8="},
>+    {2, "\337\377", "3/8="},
>+    {2, "\277\377", "v/8="},
>+    {2, "\177\377", "f/8="},
>+
>+    {3, "\000\000\000", "AAAA"},
>+    {3, "\000\000\001", "AAAB"},
>+    {3, "\000\000\002", "AAAC"},
>+    {3, "\000\000\004", "AAAE"},
>+    {3, "\000\000\010", "AAAI"},
>+    {3, "\000\000\020", "AAAQ"},
>+    {3, "\000\000\040", "AAAg"},
>+    {3, "\000\000\100", "AABA"},
>+    {3, "\000\000\200", "AACA"},
>+    {3, "\000\001\000", "AAEA"},
>+    {3, "\000\002\000", "AAIA"},
>+    {3, "\000\004\000", "AAQA"},
>+    {3, "\000\010\000", "AAgA"},
>+    {3, "\000\020\000", "ABAA"},
>+    {3, "\000\040\000", "ACAA"},
>+    {3, "\000\100\000", "AEAA"},
>+    {3, "\000\200\000", "AIAA"},
>+    {3, "\001\000\000", "AQAA"},
>+    {3, "\002\000\000", "AgAA"},
>+    {3, "\004\000\000", "BAAA"},
>+    {3, "\010\000\000", "CAAA"},
>+    {3, "\020\000\000", "EAAA"},
>+    {3, "\040\000\000", "IAAA"},
>+    {3, "\100\000\000", "QAAA"},
>+    {3, "\200\000\000", "gAAA"},
>+
>+    {3, "\377\377\377", "////"},
>+    {3, "\377\377\376", "///+"},
>+    {3, "\377\377\375", "///9"},
>+    {3, "\377\377\373", "///7"},
>+    {3, "\377\377\367", "///3"},
>+    {3, "\377\377\357", "///v"},
>+    {3, "\377\377\337", "///f"},
>+    {3, "\377\377\277", "//+/"},
>+    {3, "\377\377\177", "//9/"},
>+    {3, "\377\376\377", "//7/"},
>+    {3, "\377\375\377", "//3/"},
>+    {3, "\377\373\377", "//v/"},
>+    {3, "\377\367\377", "//f/"},
>+    {3, "\377\357\377", "/+//"},
>+    {3, "\377\337\377", "/9//"},
>+    {3, "\377\277\377", "/7//"},
>+    {3, "\377\177\377", "/3//"},
>+    {3, "\376\377\377", "/v//"},
>+    {3, "\375\377\377", "/f//"},
>+    {3, "\373\377\377", "+///"},
>+    {3, "\367\377\377", "9///"},
>+    {3, "\357\377\377", "7///"},
>+    {3, "\337\377\377", "3///"},
>+    {3, "\277\377\377", "v///"},
>+    {3, "\177\377\377", "f///"},
>+
>+    // Random numbers: values obtained with
>+    //
>+    //  #! /bin/bash
>+    //  dd bs=$1 count=1 if=/dev/random of=/tmp/bar.random
>+    //  od -N $1 -t o1 /tmp/bar.random
>+    //  uuencode -m test < /tmp/bar.random
>+    //
>+    // where $1 is the number of bytes (2, 3)
>+
>+    {2, "\243\361", "o/E="},
>+    {2, "\024\167", "FHc="},
>+    {2, "\313\252", "y6o="},
>+    {2, "\046\041", "JiE="},
>+    {2, "\145\236", "ZZ4="},
>+    {2, "\254\325", "rNU="},
>+    {2, "\061\330", "Mdg="},
>+    {2, "\245\032", "pRo="},
>+    {2, "\006\000", "BgA="},
>+    {2, "\375\131", "/Vk="},
>+    {2, "\303\210", "w4g="},
>+    {2, "\040\037", "IB8="},
>+    {2, "\261\372", "sfo="},
>+    {2, "\335\014", "3Qw="},
>+    {2, "\233\217", "m48="},
>+    {2, "\373\056", "+y4="},
>+    {2, "\247\232", "p5o="},
>+    {2, "\107\053", "Rys="},
>+    {2, "\204\077", "hD8="},
>+    {2, "\276\211", "vok="},
>+    {2, "\313\110", "y0g="},
>+    {2, "\363\376", "8/4="},
>+    {2, "\251\234", "qZw="},
>+    {2, "\103\262", "Q7I="},
>+    {2, "\142\312", "Yso="},
>+    {2, "\067\211", "N4k="},
>+    {2, "\220\001", "kAE="},
>+    {2, "\152\240", "aqA="},
>+    {2, "\367\061", "9zE="},
>+    {2, "\133\255", "W60="},
>+    {2, "\176\035", "fh0="},
>+    {2, "\032\231", "Gpk="},
>+
>+    {3, "\013\007\144", "Cwdk"},
>+    {3, "\030\112\106", "GEpG"},
>+    {3, "\047\325\046", "J9Um"},
>+    {3, "\310\160\022", "yHAS"},
>+    {3, "\131\100\237", "WUCf"},
>+    {3, "\064\342\134", "NOJc"},
>+    {3, "\010\177\004", "CH8E"},
>+    {3, "\345\147\205", "5WeF"},
>+    {3, "\300\343\360", "wOPw"},
>+    {3, "\061\240\201", "MaCB"},
>+    {3, "\225\333\044", "ldsk"},
>+    {3, "\215\137\352", "jV/q"},
>+    {3, "\371\147\160", "+Wdw"},
>+    {3, "\030\320\051", "GNAp"},
>+    {3, "\044\174\241", "JHyh"},
>+    {3, "\260\127\037", "sFcf"},
>+    {3, "\111\045\033", "SSUb"},
>+    {3, "\202\114\107", "gkxH"},
>+    {3, "\057\371\042", "L/ki"},
>+    {3, "\223\247\244", "k6ek"},
>+    {3, "\047\216\144", "J45k"},
>+    {3, "\203\070\327", "gzjX"},
>+    {3, "\247\140\072", "p2A6"},
>+    {3, "\124\115\116", "VE1O"},
>+    {3, "\157\162\050", "b3Io"},
>+    {3, "\357\223\004", "75ME"},
>+    {3, "\052\117\156", "Kk9u"},
>+    {3, "\347\154\000", "52wA"},
>+    {3, "\303\012\142", "wwpi"},
>+    {3, "\060\035\362", "MB3y"},
>+    {3, "\130\226\361", "WJbx"},
>+    {3, "\173\013\071", "ews5"},
>+    {3, "\336\004\027", "3gQX"},
>+    {3, "\357\366\234", "7/ac"},
>+    {3, "\353\304\111", "68RJ"},
>+    {3, "\024\264\131", "FLRZ"},
>+    {3, "\075\114\251", "PUyp"},
>+    {3, "\315\031\225", "zRmV"},
>+    {3, "\154\201\276", "bIG+"},
>+    {3, "\200\066\072", "gDY6"},
>+    {3, "\142\350\267", "Yui3"},
>+    {3, "\033\000\166", "GwB2"},
>+    {3, "\210\055\077", "iC0/"},
>+    {3, "\341\037\124", "4R9U"},
>+    {3, "\161\103\152", "cUNq"},
>+    {3, "\270\142\131", "uGJZ"},
>+    {3, "\337\076\074", "3z48"},
>+    {3, "\375\106\362", "/Uby"},
>+    {3, "\227\301\127", "l8FX"},
>+    {3, "\340\002\234", "4AKc"},
>+    {3, "\121\064\033", "UTQb"},
>+    {3, "\157\134\143", "b1xj"},
>+    {3, "\247\055\327", "py3X"},
>+    {3, "\340\142\005", "4GIF"},
>+    {3, "\060\260\143", "MLBj"},
>+    {3, "\075\203\170", "PYN4"},
>+    {3, "\143\160\016", "Y3AO"},
>+    {3, "\313\013\063", "ywsz"},
>+    {3, "\174\236\135", "fJ5d"},
>+    {3, "\103\047\026", "QycW"},
>+    {3, "\365\005\343", "9QXj"},
>+    {3, "\271\160\223", "uXCT"},
>+    {3, "\362\255\172", "8q16"},
>+    {3, "\113\012\015", "SwoN"},
>+
>+    // various lengths, generated by this python script:
>+    //
>+    // from string import lowercase as lc
>+    // for i in range(27):
>+    //   print '{ %2d, "%s",%s "%s" },' % (i, lc[:i], ' ' * (26-i),
>+    //                                     lc[:i].encode('base64').strip())
>+
>+    {0, "abcdefghijklmnopqrstuvwxyz", ""},
>+    {1, "abcdefghijklmnopqrstuvwxyz", "YQ=="},
>+    {2, "abcdefghijklmnopqrstuvwxyz", "YWI="},
>+    {3, "abcdefghijklmnopqrstuvwxyz", "YWJj"},
>+    {4, "abcdefghijklmnopqrstuvwxyz", "YWJjZA=="},
>+    {5, "abcdefghijklmnopqrstuvwxyz", "YWJjZGU="},
>+    {6, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVm"},
>+    {7, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZw=="},
>+    {8, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2g="},
>+    {9, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2hp"},
>+    {10, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2hpag=="},
>+    {11, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2hpams="},
>+    {12, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2hpamts"},
>+    {13, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2hpamtsbQ=="},
>+    {14, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2hpamtsbW4="},
>+    {15, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2hpamtsbW5v"},
>+    {16, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2hpamtsbW5vcA=="},
>+    {17, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2hpamtsbW5vcHE="},
>+    {18, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2hpamtsbW5vcHFy"},
>+    {19, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2hpamtsbW5vcHFycw=="},
>+    {20, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2hpamtsbW5vcHFyc3Q="},
>+    {21, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2hpamtsbW5vcHFyc3R1"},
>+    {22, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2hpamtsbW5vcHFyc3R1dg=="},
>+    {23, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2hpamtsbW5vcHFyc3R1dnc="},
>+    {24, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2hpamtsbW5vcHFyc3R1dnd4"},
>+    {25, "abcdefghijklmnopqrstuvwxy", "YWJjZGVmZ2hpamtsbW5vcHFyc3R1dnd4eQ=="},
>+    {26, "abcdefghijklmnopqrstuvwxyz", "YWJjZGVmZ2hpamtsbW5vcHFyc3R1dnd4eXo="},
> };
> #if 0
> static struct {
>@@ -328,15 +328,19 @@ static struct {
>     }                                                          \
>   }
> 
>-size_t Base64Escape(const unsigned char *src, size_t szsrc, char *dest,
>+size_t Base64Escape(const unsigned char* src,
>+                    size_t szsrc,
>+                    char* dest,
>                     size_t szdest) {
>   std::string escaped;
>-  Base64::EncodeFromArray((const char *)src, szsrc, &escaped);
>+  Base64::EncodeFromArray((const char*)src, szsrc, &escaped);
>   memcpy(dest, escaped.data(), min(escaped.size(), szdest));
>   return escaped.size();
> }
> 
>-size_t Base64Unescape(const char *src, size_t szsrc, char *dest,
>+size_t Base64Unescape(const char* src,
>+                      size_t szsrc,
>+                      char* dest,
>                       size_t szdest) {
>   std::string unescaped;
>   EXPECT_TRUE(
>@@ -345,7 +349,7 @@ size_t Base64Unescape(const char *src, size_t szsrc, char *dest,
>   return unescaped.size();
> }
> 
>-size_t Base64Unescape(const char *src, size_t szsrc, std::string *s) {
>+size_t Base64Unescape(const char* src, size_t szsrc, std::string* s) {
>   EXPECT_TRUE(Base64::DecodeFromArray(src, szsrc, Base64::DO_LAX, s, nullptr));
>   return s->size();
> }
>@@ -356,7 +360,7 @@ TEST(Base64, EncodeDecodeBattery) {
>   size_t i;
> 
>   // Check the short strings; this tests the math (and boundaries)
>-  for( i = 0; i < sizeof(base64_tests) / sizeof(base64_tests[0]); ++i ) {
>+  for (i = 0; i < sizeof(base64_tests) / sizeof(base64_tests[0]); ++i) {
>     char encode_buffer[100];
>     size_t encode_length;
>     char decode_buffer[100];
>@@ -366,16 +370,15 @@ TEST(Base64, EncodeDecodeBattery) {
>     RTC_LOG(LS_VERBOSE) << "B64: " << base64_tests[i].cyphertext;
> 
>     const unsigned char* unsigned_plaintext =
>-      reinterpret_cast<const unsigned char*>(base64_tests[i].plaintext);
>+        reinterpret_cast<const unsigned char*>(base64_tests[i].plaintext);
> 
>     cypher_length = strlen(base64_tests[i].cyphertext);
> 
>     // The basic escape function:
>     memset(encode_buffer, 0, sizeof(encode_buffer));
>-    encode_length = Base64Escape(unsigned_plaintext,
>-                                 base64_tests[i].plain_length,
>-                                 encode_buffer,
>-                                 sizeof(encode_buffer));
>+    encode_length =
>+        Base64Escape(unsigned_plaintext, base64_tests[i].plain_length,
>+                     encode_buffer, sizeof(encode_buffer));
>     //    Is it of the expected length?
>     EXPECT_EQ(encode_length, cypher_length);
> 
>@@ -384,10 +387,9 @@ TEST(Base64, EncodeDecodeBattery) {
> 
>     // If we encode it into a buffer of exactly the right length...
>     memset(encode_buffer, 0, sizeof(encode_buffer));
>-    encode_length = Base64Escape(unsigned_plaintext,
>-                                 base64_tests[i].plain_length,
>-                                 encode_buffer,
>-                                 cypher_length);
>+    encode_length =
>+        Base64Escape(unsigned_plaintext, base64_tests[i].plain_length,
>+                     encode_buffer, cypher_length);
>     //    Is it still of the expected length?
>     EXPECT_EQ(encode_length, cypher_length);
> 
>@@ -396,16 +398,15 @@ TEST(Base64, EncodeDecodeBattery) {
> 
>     // If we decode it back:
>     memset(decode_buffer, 0, sizeof(decode_buffer));
>-    decode_length = Base64Unescape(encode_buffer,
>-                                   cypher_length,
>-                                   decode_buffer,
>+    decode_length = Base64Unescape(encode_buffer, cypher_length, decode_buffer,
>                                    sizeof(decode_buffer));
> 
>     //    Is it of the expected length?
>     EXPECT_EQ(decode_length, base64_tests[i].plain_length);
> 
>     //    Is it the expected decoded value?
>-    EXPECT_EQ(0,  memcmp(decode_buffer, base64_tests[i].plaintext, decode_length));
>+    EXPECT_EQ(0,
>+              memcmp(decode_buffer, base64_tests[i].plaintext, decode_length));
> 
>     // Our decoder treats the padding '=' characters at the end as
>     // optional.  If encode_buffer has any, run some additional
>@@ -413,16 +414,17 @@ TEST(Base64, EncodeDecodeBattery) {
>     char* first_equals = strchr(encode_buffer, '=');
>     if (first_equals) {
>       // How many equals signs does the string start with?
>-      int equals = (*(first_equals+1) == '=') ? 2 : 1;
>+      int equals = (*(first_equals + 1) == '=') ? 2 : 1;
> 
>       // Try chopping off the equals sign(s) entirely.  The decoder
>       // should still be okay with this.
>       std::string decoded2("this junk should also be ignored");
>       *first_equals = '\0';
>-      EXPECT_NE(0U, Base64Unescape(encode_buffer, first_equals-encode_buffer,
>-                           &decoded2));
>+      EXPECT_NE(0U, Base64Unescape(encode_buffer, first_equals - encode_buffer,
>+                                   &decoded2));
>       EXPECT_EQ(decoded2.size(), base64_tests[i].plain_length);
>-      EXPECT_EQ_ARRAY(decoded2.size(), decoded2.data(), base64_tests[i].plaintext, i);
>+      EXPECT_EQ_ARRAY(decoded2.size(), decoded2.data(),
>+                      base64_tests[i].plaintext, i);
> 
>       size_t len;
> 
>@@ -445,453 +447,901 @@ TEST(Base64, EncodeDecodeBattery) {
> // here's a weird case: a giant base64 encoded stream which broke our base64
> // decoding.  Let's test it explicitly.
> const char SpecificTest[] =
>-  "/9j/4AAQSkZJRgABAgEASABIAAD/4Q0HRXhpZgAATU0AKgAAAAgADAEOAAIAAAAgAAAAngEPAAI\n"
>-  "AAAAFAAAAvgEQAAIAAAAJAAAAwwESAAMAAAABAAEAAAEaAAUAAAABAAAAzAEbAAUAAAABAAAA1A\n"
>-  "EoAAMAAAABAAIAAAExAAIAAAAUAAAA3AEyAAIAAAAUAAAA8AE8AAIAAAAQAAABBAITAAMAAAABA\n"
>-  "AIAAIdpAAQAAAABAAABFAAAAsQgICAgICAgICAgICAgICAgICAgICAgICAgICAgICAgAFNPTlkA\n"
>-  "RFNDLVAyMDAAAAAASAAAAAEAAABIAAAAAUFkb2JlIFBob3Rvc2hvcCA3LjAAMjAwNzowMTozMCA\n"
>-  "yMzoxMDowNABNYWMgT1MgWCAxMC40LjgAAByCmgAFAAAAAQAAAmqCnQAFAAAAAQAAAnKIIgADAA\n"
>-  "AAAQACAACIJwADAAAAAQBkAACQAAAHAAAABDAyMjCQAwACAAAAFAAAAnqQBAACAAAAFAAAAo6RA\n"
>-  "QAHAAAABAECAwCRAgAFAAAAAQAAAqKSBAAKAAAAAQAAAqqSBQAFAAAAAQAAArKSBwADAAAAAQAF\n"
>-  "AACSCAADAAAAAQAAAACSCQADAAAAAQAPAACSCgAFAAAAAQAAArqgAAAHAAAABDAxMDCgAQADAAA\n"
>-  "AAf//AACgAgAEAAAAAQAAAGSgAwAEAAAAAQAAAGSjAAAHAAAAAQMAAACjAQAHAAAAAQEAAACkAQ\n"
>-  "ADAAAAAQAAAACkAgADAAAAAQAAAACkAwADAAAAAQAAAACkBgADAAAAAQAAAACkCAADAAAAAQAAA\n"
>-  "ACkCQADAAAAAQAAAACkCgADAAAAAQAAAAAAAAAAAAAACgAAAZAAAAAcAAAACjIwMDc6MDE6MjAg\n"
>-  "MjM6MDU6NTIAMjAwNzowMToyMCAyMzowNTo1MgAAAAAIAAAAAQAAAAAAAAAKAAAAMAAAABAAAAB\n"
>-  "PAAAACgAAAAYBAwADAAAAAQAGAAABGgAFAAAAAQAAAxIBGwAFAAAAAQAAAxoBKAADAAAAAQACAA\n"
>-  "ACAQAEAAAAAQAAAyICAgAEAAAAAQAACd0AAAAAAAAASAAAAAEAAABIAAAAAf/Y/+AAEEpGSUYAA\n"
>-  "QIBAEgASAAA/+0ADEFkb2JlX0NNAAL/7gAOQWRvYmUAZIAAAAAB/9sAhAAMCAgICQgMCQkMEQsK\n"
>-  "CxEVDwwMDxUYExMVExMYEQwMDAwMDBEMDAwMDAwMDAwMDAwMDAwMDAwMDAwMDAwMDAwMAQ0LCw0\n"
>-  "ODRAODhAUDg4OFBQODg4OFBEMDAwMDBERDAwMDAwMEQwMDAwMDAwMDAwMDAwMDAwMDAwMDAwMDA\n"
>-  "wMDAz/wAARCABkAGQDASIAAhEBAxEB/90ABAAH/8QBPwAAAQUBAQEBAQEAAAAAAAAAAwABAgQFB\n"
>-  "gcICQoLAQABBQEBAQEBAQAAAAAAAAABAAIDBAUGBwgJCgsQAAEEAQMCBAIFBwYIBQMMMwEAAhED\n"
>-  "BCESMQVBUWETInGBMgYUkaGxQiMkFVLBYjM0coLRQwclklPw4fFjczUWorKDJkSTVGRFwqN0Nhf\n"
>-  "SVeJl8rOEw9N14/NGJ5SkhbSVxNTk9KW1xdXl9VZmdoaWprbG1ub2N0dXZ3eHl6e3x9fn9xEAAg\n"
>-  "IBAgQEAwQFBgcHBgU1AQACEQMhMRIEQVFhcSITBTKBkRShsUIjwVLR8DMkYuFygpJDUxVjczTxJ\n"
>-  "QYWorKDByY1wtJEk1SjF2RFVTZ0ZeLys4TD03Xj80aUpIW0lcTU5PSltcXV5fVWZnaGlqa2xtbm\n"
>-  "9ic3R1dnd4eXp7fH/9oADAMBAAIRAxEAPwDy7bKNTUXNLz9EaJPDWMjxH4ozhtpYwaACT8ShaaW\n"
>-  "bW0uEc9/JFfjj0Q4Hk/PRDxwX7y47W9z/AN9Cv4+O3ILK2DcRqT2CaSvEbcl1Jbz37KG1dBldLo\n"
>-  "qaS4l9xGjG9v6yoDAdYIaIjUk+AREgo4y5sapirb8Yl0NHHdKvBNm4yA1o5Pc+SPEFvCWqB3HZF\n"
>-  "Hj2SbWQ/afGFP0bHP8ATY0uc4w1o1JPkkimGiS2KvqlnmBkOZQTyydzgPMM9v8A0lp4v1Nx9gF1\n"
>-  "tpdqJaGtH/S3I0i3lISXW/8AMqnd/O2bfg2eUkqVYf/Q8zuncO4Bj7lZ+n7f5Mj5KsJcY8NUZ4d\n"
>-  "uEDVo1HkeU0rg3Om4H2rabCWUN7DQuK1n5FWKW4uCwG92gDRJBS6exhxmMboQI+Cv4WFTQ42Bs2\n"
>-  "fvnkkqEmy2YxoMMbpVzaz6jt+RbpHZs8lzkHqrasKkYOKP0jgDfZ4N/wDM1tNrcWfSPmRyq9uNV\n"
>-  "DnFg2s97i7UkjxKVrq0eVz3spZsja+ASDzwsh9jnOk/JFzb3XZD3v1c4yT8UACTCniKDUnKz5Nj\n"
>-  "G33XV1DV73BrT8dF23SejV4zg9g33cOsPb+SxVvqv9ViwNy8vS0iWs/daf8A0Y5dpTi1sADGxCR\n"
>-  "K1o0YBEmInlXWYbDBcDLdPJXa8f71Yrx2jnUoAqLnfZK5hJaW2vdwEk5a/wD/0fN6Ia/e76IiVf\n"
>-  "xavUL7CPpnT4LNbYXAVjuQt/AqDmNYO/Kjnoy4hr5J8SwMhrRMaeSvbsxrfUazcOw4UX0Cisem2\n"
>-  "SBoD4+Kz8nC6llbSLCRrubJA8kwUWbUDa29X1PMa7aQWjuDC0MXMdbDbhI7eazBiUfZ6GOYRe1s\n"
>-  "WvGgJ8Vbw2+m4Bx9s6JpNHuuGo1FF53r/SHYua61gLse0lzXeBP5rkvqx0o5vVWz7WY49QkiQSP\n"
>-  "oN/tLoevW/ogxv0HA7tJ0AnhT+pdDGYVl/wCdcTPkGn2NU0JWNWvlgAbHV6fEqdu2gR/r2WlWwt\n"
>-  "AA5VXAEsLXTqJafArQY5rRr9LiPBJiZsZCI1pJjxCi0j4oncSICSkWwzwkjeaSch//0vO7sP7Lm\n"
>-  "enO9ogtd5FbPT3Q5pCpZVc4ld3Lmn3O8j9EI2BYdunKjOobMQIyI+rusc2wx4d0eutwGnHh/uQc\n"
>-  "Ha7ladj6mVANGvcqOgz0Go7HJ12/GEHcwvB/dPY6ImbbaMaASGuIBjkN7qofs9Ubg9g7OI9p/t/\n"
>-  "RTSmhTHr0v6eSz6UgCPP2/wAVu9Ex2V49dVY2iACB4BZeVXQ/AJ3gzGnnOi2+kACpru8flUsNmt\n"
>-  "zHRf6xfWCnoeAfTh2ZaQKazx/Ke7+QxcKz61fWA2uuObaC4zGhaPJrXBL64ZFmR124O09ENraPK\n"
>-  "N3/AH5GqxIrZVUyp2K2vfdkENsDnxuex9m4Ox9n82xSgNd9D+p/XR1npgseR9ppOy4Dx/NfH/CL\n"
>-  "oQJGunmvMv8AFq3KHVcq3HkYQbD2nuSf0I/rMavSg6TLjLigQhJ7Z58v9QkmlsTOqSCn/9PzL7R\n"
>-  "d6Qq3n0wZ2zotXpT9xLfFYvkr/S7jXeB8E0jRkhKpC3q8LcJ/kmCrTnkuAPCq4do9Q/ytVbuAeY\n"
>-  "Gg5lQybQK+82GBqEQUA1kOHPYf3LLsoyN36G5w8iUfHxepbXE2l0cApALgLHzBq9UxhTXU5hMC1\n"
>-  "ktnSCup6S4Ctk+C5XqVGcaHPfuiuHkeTTuWz0+9zaKiH6CC0/yXBSQ2a/MxojV57634rq+v2PLY\n"
>-  "be1r2nsYG13/AFKxbfCBMcr0brGAzrGEwCG31ncx0SfBzf7S4+zoHUWWsJq3hz9oLfcBH77R9H+\n"
>-  "0pA13u/qPgDp/Q6ri39JlfpXkDx+h/msWn1L6wdO6bSbcrIbU2Q0xLnSe21kuVejJspbVS5+4bd\n"
>-  "ocBAkD/orG+tP1ar67Wy7GtZTm1SCXfRsb+a18fRe38x6SG3/44H1Z3f0y2I+l6DoSXD/8xPrDs\n"
>-  "3enVu3bdnqN3R+//USSVo//1PLohhce+gRWS0Nsby3lRgFkKxQyW7SgUh3em5Tbq2uB9wWw1wey\n"
>-  "J1XGV2XYdm5k7e4WzidXY9oMwo5RZ4T6Hd1ixwfp96PWbAJBVTHzK7O6Ky5oJB1HZMqmUEFlkGy\n"
>-  "xpa4zI1Hkq31dy7bMN9BAc3HeWAnnbyxEycmuup1jiAGglZ31PyrmZ9tQg1WtNj54EHR3/S2qTH\n"
>-  "1Yc5GgD1FFtzPdWGkd2AyflogZmRmsz6PSrbXbdo+txOrP337f3fzVo15DK2uyrTtqpBOnBKx6b\n"
>-  "7MjJsz7tHWOAYP3WD6LU6cqGjFCNl1MmvLcxv6YtDTLSAqP27LrdtYHXFnJZI+Tp3MWg68OpDPv\n"
>-  "UMUM2lkQBoouKQ6swjE9Nml+1sz1PW+z6xt27zuj+skrX2ZvqR5z8kkuOfdPt43/1fMm/grFG6f\n"
>-  "Lss9JA7JG7tnZs/SfJUrfS3foJ9TvHCopJsV8nWx/t24bJn8Fo/5TjWJXMJIS+i+G36TsZ/7Q9P\n"
>-  "8ATfzfeOFofVSZv2/zvt+O3X/v65dJPjt/BiyfN1/wn0zre79nVej/ADG8ep4x2/6Srjd6TdviF\n"
>-  "52ko8m6/Ht9X1KnftEo+POwxzK8mSTF46vrH6T1/OEl5Okkl//Z/+0uHFBob3Rvc2hvcCAzLjAA\n"
>-  "OEJJTQQEAAAAAAArHAIAAAIAAhwCeAAfICAgICAgICAgICAgICAgICAgICAgICAgICAgICAgIAA\n"
>-  "4QklNBCUAAAAAABD7Caa9B0wqNp2P4sxXqayFOEJJTQPqAAAAAB2wPD94bWwgdmVyc2lvbj0iMS\n"
>-  "4wIiBlbmNvZGluZz0iVVRGLTgiPz4KPCFET0NUWVBFIHBsaXN0IFBVQkxJQyAiLS8vQXBwbGUgQ\n"
>-  "29tcHV0ZXIvL0RURCBQTElTVCAxLjAvL0VOIiAiaHR0cDovL3d3dy5hcHBsZS5jb20vRFREcy9Q\n"
>-  "cm9wZXJ0eUxpc3QtMS4wLmR0ZCI+CjxwbGlzdCB2ZXJzaW9uPSIxLjAiPgo8ZGljdD4KCTxrZXk\n"
>-  "+Y29tLmFwcGxlLnByaW50LlBhZ2VGb3JtYXQuUE1Ib3Jpem9udGFsUmVzPC9rZXk+Cgk8ZGljdD\n"
>-  "4KCQk8a2V5PmNvbS5hcHBsZS5wcmludC50aWNrZXQuY3JlYXRvcjwva2V5PgoJCTxzdHJpbmc+Y\n"
>-  "29tLmFwcGxlLnByaW50aW5nbWFuYWdlcjwvc3RyaW5nPgoJCTxrZXk+Y29tLmFwcGxlLnByaW50\n"
>-  "LnRpY2tldC5pdGVtQXJyYXk8L2tleT4KCQk8YXJyYXk+CgkJCTxkaWN0PgoJCQkJPGtleT5jb20\n"
>-  "uYXBwbGUucHJpbnQuUGFnZUZvcm1hdC5QTUhvcml6b250YWxSZXM8L2tleT4KCQkJCTxyZWFsPj\n"
>-  "cyPC9yZWFsPgoJCQkJPGtleT5jb20uYXBwbGUucHJpbnQudGlja2V0LmNsaWVudDwva2V5PgoJC\n"
>-  "QkJPHN0cmluZz5jb20uYXBwbGUucHJpbnRpbmdtYW5hZ2VyPC9zdHJpbmc+CgkJCQk8a2V5PmNv\n"
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>+    "0fBapxheVh9ocV+nviqY2\n"
>+    "/qQJDew/"
>+    "bioWHiuQ8m0bbvaPKGtQ6jaxSo9JloCK75gZI0Xb4sgkHo8MouoAvP94BsRmGY7uWJU\n"
>+    "gzbypOQpNOvIdK4Nw2WCE2tXulTkjEEbdafgclxMhFBas93dwyQzsWDghlJFONKHJCZtjOFBJf"
>+    "y\n"
>+    "j1y9vPL9zpbIs0WkXL2sUjA8hDXlGCRXtt07ZuYvL5KJeo6bfajbkzWkcToR8dqshZ6in2fhNK"
>+    "/\n"
>+    "PDTUlXmHVvMdr5o0v9H2kdrqGpfu7m0nkY87Uf7tkKAU4/"
>+    "s03ynLkEBbfihx7dGT6va67LbRMNR\n"
>+    "aKOBuUTKgIBXoK1BOYR1M3aQ0mOt9yxUeZNdtJhFapLqMluSXkg5oxJrUMW5KevQ9MmNXXNqOi"
>+    "H\n"
>+    "Rr/Hmv8A1r9I/oj95w+r+j9Yf1+NP5+nXtTD+dF8tkfkOlv/0vC3ph7f0/"
>+    "alcVTbS4A8QibuKb5\n"
>+    "RI05EBYRFpdX3ly79a2qYCavH/"
>+    "EY7TCYyMD5PSdD8+wXUSn1ArDqOhBzFlipz4ZwWbaV5htbsgF\n"
>+    "qg9crMXKErGyYwajFGzxyHlGSePbbwyqg5UZlCaxrFpaWU95LIqrEjMAT4Dp9OShGy1ZslBhv/"
>+    "A\n"
>+    "Dj9rd/a+aL+xUK+m38L3d0HrxRo2HFtu5D8c27y8t30raarbWkU+u6g4gsNORn+EcUaSh2Pc0/"
>+    "4\n"
>+    "lgtAjezzbT9SutY1i782al8Nxdyotqh6xWybIg+jc5q8s+I27bFDgFPQp9RE+nrag70+"
>+    "L6crrZu\n"
>+    "4jajokdv6LW/Dii1Wo61PXKQN3KPK0L+h4/rnD/K5V78a5LhXxd3/0/"
>+    "DMXXtwxVNtL9Xkaf3f7N\n"
>+    "etfbKMjdjtkZ9D6ufrlK0+HpX8coF9HJ26sXvfqXrf7i/U+uften/d/"
>+    "wCyrmQL6uOav0pvpP8Ai\n"
>+    "b1F+rV59+vH6a5XLhcjH4nRmY/xpxHP0/UptWvT6Mx/RbmjxWK+aP8AFf1M/"
>+    "pCv1Kvxen9inavf\n"
>+    "MrFwXtzcLUeLXq5Mv/I3nz1b0v8AjofuKVry9KrUpTanOlf9jmQ68va/zH9b/COn/o7/"
>+    "AI431mP\n"
>+    "65SvLh+zWvbl9rMfNfC34K4kmj9T6lD6FKclp/DNYXZx5srsPrHor6nXvkgxTPS/"
>+    "U+rv6dPU5mt\n"
>+    "fngFN5ulv+l/pL/Lp/scerHo//2Q==\n";
> 
> static std::string gCommandLine;
> 
>@@ -901,87 +1351,85 @@ TEST(Base64, LargeSample) {
>   char unescaped[64 * 1024];
> 
>   // unescape that massive blob above
>-  size_t size = Base64Unescape(SpecificTest,
>-                            sizeof(SpecificTest),
>-                            unescaped,
>-                            sizeof(unescaped));
>+  size_t size = Base64Unescape(SpecificTest, sizeof(SpecificTest), unescaped,
>+                               sizeof(unescaped));
> 
>   EXPECT_EQ(size, sizeof(testbase64));
>   EXPECT_EQ(0, memcmp(testbase64, unescaped, sizeof(testbase64)));
> }
> 
>-bool DecodeTest(const char* encoded, size_t expect_unparsed,
>-                const char* decoded, Base64::DecodeFlags flags)
>-{
>+bool DecodeTest(const char* encoded,
>+                size_t expect_unparsed,
>+                const char* decoded,
>+                Base64::DecodeFlags flags) {
>   std::string result;
>   size_t consumed = 0, encoded_len = strlen(encoded);
>-  bool success = Base64::DecodeFromArray(encoded, encoded_len, flags,
>-                                         &result, &consumed);
>+  bool success =
>+      Base64::DecodeFromArray(encoded, encoded_len, flags, &result, &consumed);
>   size_t unparsed = encoded_len - consumed;
>-  EXPECT_EQ(expect_unparsed, unparsed) << "\"" << encoded
>-                                       << "\" -> \"" << decoded
>-                                       << "\"";
>+  EXPECT_EQ(expect_unparsed, unparsed)
>+      << "\"" << encoded << "\" -> \"" << decoded << "\"";
>   EXPECT_STREQ(decoded, result.c_str());
>   return success;
> }
> 
>-#define Flags(x,y,z) \
>+#define Flags(x, y, z) \
>   Base64::DO_PARSE_##x | Base64::DO_PAD_##y | Base64::DO_TERM_##z
> 
> TEST(Base64, DecodeParseOptions) {
>   // Trailing whitespace
>-  EXPECT_TRUE (DecodeTest("YWJjZA== ", 1, "abcd", Flags(STRICT, YES, CHAR)));
>-  EXPECT_TRUE (DecodeTest("YWJjZA== ", 0, "abcd", Flags(WHITE,  YES, CHAR)));
>-  EXPECT_TRUE (DecodeTest("YWJjZA== ", 0, "abcd", Flags(ANY,    YES, CHAR)));
>+  EXPECT_TRUE(DecodeTest("YWJjZA== ", 1, "abcd", Flags(STRICT, YES, CHAR)));
>+  EXPECT_TRUE(DecodeTest("YWJjZA== ", 0, "abcd", Flags(WHITE, YES, CHAR)));
>+  EXPECT_TRUE(DecodeTest("YWJjZA== ", 0, "abcd", Flags(ANY, YES, CHAR)));
> 
>   // Embedded whitespace
>   EXPECT_FALSE(DecodeTest("YWJjZA= =", 3, "abcd", Flags(STRICT, YES, CHAR)));
>-  EXPECT_TRUE (DecodeTest("YWJjZA= =", 0, "abcd", Flags(WHITE,  YES, CHAR)));
>-  EXPECT_TRUE (DecodeTest("YWJjZA= =", 0, "abcd", Flags(ANY,    YES, CHAR)));
>+  EXPECT_TRUE(DecodeTest("YWJjZA= =", 0, "abcd", Flags(WHITE, YES, CHAR)));
>+  EXPECT_TRUE(DecodeTest("YWJjZA= =", 0, "abcd", Flags(ANY, YES, CHAR)));
> 
>   // Embedded non-base64 characters
>   EXPECT_FALSE(DecodeTest("YWJjZA=*=", 3, "abcd", Flags(STRICT, YES, CHAR)));
>-  EXPECT_FALSE(DecodeTest("YWJjZA=*=", 3, "abcd", Flags(WHITE,  YES, CHAR)));
>-  EXPECT_TRUE (DecodeTest("YWJjZA=*=", 0, "abcd", Flags(ANY,    YES, CHAR)));
>+  EXPECT_FALSE(DecodeTest("YWJjZA=*=", 3, "abcd", Flags(WHITE, YES, CHAR)));
>+  EXPECT_TRUE(DecodeTest("YWJjZA=*=", 0, "abcd", Flags(ANY, YES, CHAR)));
> 
>   // Unexpected padding characters
>-  EXPECT_FALSE(DecodeTest("YW=JjZA==", 7, "a",    Flags(STRICT, YES, CHAR)));
>-  EXPECT_FALSE(DecodeTest("YW=JjZA==", 7, "a",    Flags(WHITE,  YES, CHAR)));
>-  EXPECT_TRUE (DecodeTest("YW=JjZA==", 0, "abcd", Flags(ANY,    YES, CHAR)));
>+  EXPECT_FALSE(DecodeTest("YW=JjZA==", 7, "a", Flags(STRICT, YES, CHAR)));
>+  EXPECT_FALSE(DecodeTest("YW=JjZA==", 7, "a", Flags(WHITE, YES, CHAR)));
>+  EXPECT_TRUE(DecodeTest("YW=JjZA==", 0, "abcd", Flags(ANY, YES, CHAR)));
> }
> 
> TEST(Base64, DecodePadOptions) {
>   // Padding
>-  EXPECT_TRUE (DecodeTest("YWJjZA==",  0, "abcd", Flags(STRICT, YES, CHAR)));
>-  EXPECT_TRUE (DecodeTest("YWJjZA==",  0, "abcd", Flags(STRICT, ANY, CHAR)));
>-  EXPECT_TRUE (DecodeTest("YWJjZA==",  2, "abcd", Flags(STRICT, NO,  CHAR)));
>+  EXPECT_TRUE(DecodeTest("YWJjZA==", 0, "abcd", Flags(STRICT, YES, CHAR)));
>+  EXPECT_TRUE(DecodeTest("YWJjZA==", 0, "abcd", Flags(STRICT, ANY, CHAR)));
>+  EXPECT_TRUE(DecodeTest("YWJjZA==", 2, "abcd", Flags(STRICT, NO, CHAR)));
> 
>   // Incomplete padding
>-  EXPECT_FALSE(DecodeTest("YWJjZA=",   1, "abcd", Flags(STRICT, YES, CHAR)));
>-  EXPECT_TRUE (DecodeTest("YWJjZA=",   1, "abcd", Flags(STRICT, ANY, CHAR)));
>-  EXPECT_TRUE (DecodeTest("YWJjZA=",   1, "abcd", Flags(STRICT, NO,  CHAR)));
>+  EXPECT_FALSE(DecodeTest("YWJjZA=", 1, "abcd", Flags(STRICT, YES, CHAR)));
>+  EXPECT_TRUE(DecodeTest("YWJjZA=", 1, "abcd", Flags(STRICT, ANY, CHAR)));
>+  EXPECT_TRUE(DecodeTest("YWJjZA=", 1, "abcd", Flags(STRICT, NO, CHAR)));
> 
>   // No padding
>-  EXPECT_FALSE(DecodeTest("YWJjZA",    0, "abcd", Flags(STRICT, YES, CHAR)));
>-  EXPECT_TRUE (DecodeTest("YWJjZA",    0, "abcd", Flags(STRICT, ANY, CHAR)));
>-  EXPECT_TRUE (DecodeTest("YWJjZA",    0, "abcd", Flags(STRICT, NO,  CHAR)));
>+  EXPECT_FALSE(DecodeTest("YWJjZA", 0, "abcd", Flags(STRICT, YES, CHAR)));
>+  EXPECT_TRUE(DecodeTest("YWJjZA", 0, "abcd", Flags(STRICT, ANY, CHAR)));
>+  EXPECT_TRUE(DecodeTest("YWJjZA", 0, "abcd", Flags(STRICT, NO, CHAR)));
> }
> 
> TEST(Base64, DecodeTerminateOptions) {
>   // Complete quantum
>-  EXPECT_TRUE (DecodeTest("YWJj",      0, "abc",  Flags(STRICT, NO,  BUFFER)));
>-  EXPECT_TRUE (DecodeTest("YWJj",      0, "abc",  Flags(STRICT, NO,  CHAR)));
>-  EXPECT_TRUE (DecodeTest("YWJj",      0, "abc",  Flags(STRICT, NO,  ANY)));
>+  EXPECT_TRUE(DecodeTest("YWJj", 0, "abc", Flags(STRICT, NO, BUFFER)));
>+  EXPECT_TRUE(DecodeTest("YWJj", 0, "abc", Flags(STRICT, NO, CHAR)));
>+  EXPECT_TRUE(DecodeTest("YWJj", 0, "abc", Flags(STRICT, NO, ANY)));
> 
>   // Complete quantum with trailing data
>-  EXPECT_FALSE(DecodeTest("YWJj*",     1, "abc",  Flags(STRICT, NO,  BUFFER)));
>-  EXPECT_TRUE (DecodeTest("YWJj*",     1, "abc",  Flags(STRICT, NO,  CHAR)));
>-  EXPECT_TRUE (DecodeTest("YWJj*",     1, "abc",  Flags(STRICT, NO,  ANY)));
>+  EXPECT_FALSE(DecodeTest("YWJj*", 1, "abc", Flags(STRICT, NO, BUFFER)));
>+  EXPECT_TRUE(DecodeTest("YWJj*", 1, "abc", Flags(STRICT, NO, CHAR)));
>+  EXPECT_TRUE(DecodeTest("YWJj*", 1, "abc", Flags(STRICT, NO, ANY)));
> 
>   // Incomplete quantum
>-  EXPECT_FALSE(DecodeTest("YWJ",       0, "ab",   Flags(STRICT, NO,  BUFFER)));
>-  EXPECT_FALSE(DecodeTest("YWJ",       0, "ab",   Flags(STRICT, NO,  CHAR)));
>-  EXPECT_TRUE (DecodeTest("YWJ",       0, "ab",   Flags(STRICT, NO,  ANY)));
>+  EXPECT_FALSE(DecodeTest("YWJ", 0, "ab", Flags(STRICT, NO, BUFFER)));
>+  EXPECT_FALSE(DecodeTest("YWJ", 0, "ab", Flags(STRICT, NO, CHAR)));
>+  EXPECT_TRUE(DecodeTest("YWJ", 0, "ab", Flags(STRICT, NO, ANY)));
> }
> 
> TEST(Base64, GetNextBase64Char) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/basictypes.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/basictypes.h
>deleted file mode 100644
>index 42226e79b28..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/basictypes.h
>+++ /dev/null
>@@ -1,70 +0,0 @@
>-/*
>- *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef RTC_BASE_BASICTYPES_H_
>-#define RTC_BASE_BASICTYPES_H_
>-
>-#include <stddef.h>  // for NULL, size_t
>-#include <stdint.h>  // for uintptr_t and (u)int_t types.
>-
>-// Detect compiler is for x86 or x64.
>-#if defined(__x86_64__) || defined(_M_X64) || \
>-    defined(__i386__) || defined(_M_IX86)
>-#define CPU_X86 1
>-#endif
>-
>-// Detect compiler is for arm.
>-#if defined(__arm__) || defined(_M_ARM)
>-#define CPU_ARM 1
>-#endif
>-
>-#if defined(CPU_X86) && defined(CPU_ARM)
>-#error CPU_X86 and CPU_ARM both defined.
>-#endif
>-
>-#if !defined(RTC_ARCH_CPU_BIG_ENDIAN) && !defined(RTC_ARCH_CPU_LITTLE_ENDIAN)
>-// x86, arm or GCC provided __BYTE_ORDER__ macros
>-#if defined(CPU_X86) || defined(CPU_ARM) ||                             \
>-  (defined(__BYTE_ORDER__) && __BYTE_ORDER__ == __ORDER_LITTLE_ENDIAN__)
>-#define RTC_ARCH_CPU_LITTLE_ENDIAN
>-#elif defined(__BYTE_ORDER__) && __BYTE_ORDER__ == __ORDER_BIG_ENDIAN__
>-#define RTC_ARCH_CPU_BIG_ENDIAN
>-#else
>-#error RTC_ARCH_CPU_BIG_ENDIAN or RTC_ARCH_CPU_LITTLE_ENDIAN should be defined.
>-#endif
>-#endif
>-
>-#if defined(RTC_ARCH_CPU_BIG_ENDIAN) && defined(RTC_ARCH_CPU_LITTLE_ENDIAN)
>-#error RTC_ARCH_CPU_BIG_ENDIAN and RTC_ARCH_CPU_LITTLE_ENDIAN both defined.
>-#endif
>-
>-#if defined(WEBRTC_WIN)
>-typedef int socklen_t;
>-#endif
>-
>-// The following only works for C++
>-#ifdef __cplusplus
>-
>-#ifndef ALIGNP
>-#define ALIGNP(p, t)                                             \
>-  (reinterpret_cast<uint8_t*>(((reinterpret_cast<uintptr_t>(p) + \
>-  ((t) - 1)) & ~((t) - 1))))
>-#endif
>-
>-#define RTC_IS_ALIGNED(p, a) (!((uintptr_t)(p) & ((a) - 1)))
>-
>-// Use these to declare and define a static local variable that gets leaked so
>-// that its destructors are not called at exit.
>-#define RTC_DEFINE_STATIC_LOCAL(type, name, arguments) \
>-  static type& name = *new type arguments
>-
>-#endif  // __cplusplus
>-
>-#endif  // RTC_BASE_BASICTYPES_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/basictypes_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/basictypes_unittest.cc
>deleted file mode 100644
>index a8b05339675..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/basictypes_unittest.cc
>+++ /dev/null
>@@ -1,48 +0,0 @@
>-/*
>- *  Copyright 2012 The WebRTC Project Authors. All rights reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "rtc_base/basictypes.h"
>-
>-#include "rtc_base/gunit.h"
>-
>-namespace rtc {
>-
>-TEST(BasicTypesTest, Endian) {
>-  uint16_t v16 = 0x1234u;
>-  uint8_t first_byte = *reinterpret_cast<uint8_t*>(&v16);
>-#if defined(RTC_ARCH_CPU_LITTLE_ENDIAN)
>-  EXPECT_EQ(0x34u, first_byte);
>-#elif defined(RTC_ARCH_CPU_BIG_ENDIAN)
>-  EXPECT_EQ(0x12u, first_byte);
>-#endif
>-}
>-
>-TEST(BasicTypesTest, SizeOfConstants) {
>-  EXPECT_EQ(8u, sizeof(INT64_C(0)));
>-  EXPECT_EQ(8u, sizeof(UINT64_C(0)));
>-  EXPECT_EQ(8u, sizeof(INT64_C(0x1234567887654321)));
>-  EXPECT_EQ(8u, sizeof(UINT64_C(0x8765432112345678)));
>-}
>-
>-// Test CPU_ macros
>-#if !defined(CPU_ARM) && defined(__arm__)
>-#error expected CPU_ARM to be defined.
>-#endif
>-#if !defined(CPU_X86) && (defined(WEBRTC_WIN) || defined(WEBRTC_MAC) && !defined(WEBRTC_IOS))
>-#error expected CPU_X86 to be defined.
>-#endif
>-#if !defined(RTC_ARCH_CPU_LITTLE_ENDIAN) && \
>-  (defined(WEBRTC_WIN) || defined(WEBRTC_MAC) && !defined(WEBRTC_IOS) || defined(CPU_X86))
>-#error expected RTC_ARCH_CPU_LITTLE_ENDIAN to be defined.
>-#endif
>-
>-// TODO(fbarchard): Test all macros in basictypes.h
>-
>-}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bind.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bind.h
>index b9f98b9dfe9..9f225f1cae7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bind.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bind.h
>@@ -77,7 +77,10 @@ namespace detail {
> // matches to the function prototype, but the parameters to bind have
> // references stripped. This trick allows the compiler to dictate the Bind
> // parameter types rather than deduce them.
>-template <class T> struct identity { typedef T type; };
>+template <class T>
>+struct identity {
>+  typedef T type;
>+};
> 
> // IsRefCounted<T>::value will be true for types that can be used in
> // rtc::scoped_refptr<T>, i.e. types that implements nullary functions AddRef()
>@@ -88,8 +91,12 @@ class IsRefCounted {
>   // This is a complex implementation detail done with SFINAE.
> 
>   // Define types such that sizeof(Yes) != sizeof(No).
>-  struct Yes { char dummy[1]; };
>-  struct No { char dummy[2]; };
>+  struct Yes {
>+    char dummy[1];
>+  };
>+  struct No {
>+    char dummy[2];
>+  };
>   // Define two overloaded template functions with return types of different
>   // size. This way, we can use sizeof() on the return type to determine which
>   // function the compiler would have chosen. One function will be preferred
>@@ -98,9 +105,10 @@ class IsRefCounted {
>   // preferred function.
>   template <typename R>
>   static Yes test(R* r, decltype(r->AddRef(), r->Release(), 42));
>-  template <typename C> static No test(...);
>+  template <typename C>
>+  static No test(...);
> 
>-public:
>+ public:
>   // Trick the compiler to tell if it's possible to call AddRef() and Release().
>   static const bool value = sizeof(test<T>((T*)nullptr, 42)) == sizeof(Yes);
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bind_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bind_unittest.cc
>index 8703be4adf6..ed0913fdd3d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bind_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bind_unittest.cc
>@@ -24,10 +24,20 @@ struct LifeTimeCheck;
> 
> struct MethodBindTester {
>   void NullaryVoid() { ++call_count; }
>-  int NullaryInt() { ++call_count; return 1; }
>-  int NullaryConst() const { ++call_count; return 2; }
>+  int NullaryInt() {
>+    ++call_count;
>+    return 1;
>+  }
>+  int NullaryConst() const {
>+    ++call_count;
>+    return 2;
>+  }
>   void UnaryVoid(int dummy) { ++call_count; }
>-  template <class T> T Identity(T value) { ++call_count; return value; }
>+  template <class T>
>+  T Identity(T value) {
>+    ++call_count;
>+    return value;
>+  }
>   int UnaryByPointer(int* value) const {
>     ++call_count;
>     return ++(*value);
>@@ -36,7 +46,10 @@ struct MethodBindTester {
>     ++call_count;
>     return ++const_cast<int&>(value);
>   }
>-  int Multiply(int a, int b) const { ++call_count; return a * b; }
>+  int Multiply(int a, int b) const {
>+    ++call_count;
>+    return a * b;
>+  }
>   void RefArgument(const scoped_refptr<LifeTimeCheck>& object) {
>     EXPECT_TRUE(object.get() != nullptr);
>   }
>@@ -44,13 +57,17 @@ struct MethodBindTester {
>   mutable int call_count;
> };
> 
>-struct A { int dummy; };
>-struct B: public RefCountInterface { int dummy; };
>-struct C: public A, B {};
>+struct A {
>+  int dummy;
>+};
>+struct B : public RefCountInterface {
>+  int dummy;
>+};
>+struct C : public A, B {};
> struct D {
>   int AddRef();
> };
>-struct E: public D {
>+struct E : public D {
>   int Release();
> };
> struct F {
>@@ -66,9 +83,15 @@ struct LifeTimeCheck {
>   int ref_count_;
> };
> 
>-int Return42() { return 42; }
>-int Negate(int a) { return -a; }
>-int Multiply(int a, int b) { return a * b; }
>+int Return42() {
>+  return 42;
>+}
>+int Negate(int a) {
>+  return -a;
>+}
>+int Multiply(int a, int b) {
>+  return a * b;
>+}
> 
> }  // namespace
> 
>@@ -76,7 +99,8 @@ int Multiply(int a, int b) { return a * b; }
> // compile time.
> #define EXPECT_IS_CAPTURED_AS_PTR(T)                              \
>   static_assert(is_same<detail::PointerType<T>::type, T*>::value, \
>-                "PointerType")
>+                "PointerTyp"                                      \
>+                "e")
> #define EXPECT_IS_CAPTURED_AS_SCOPED_REFPTR(T)                        \
>   static_assert(                                                      \
>       is_same<detail::PointerType<T>::type, scoped_refptr<T>>::value, \
>@@ -206,7 +230,9 @@ TEST(BindTest, ScopedRefPointerArgument) {
> 
> namespace {
> 
>-const int* Ref(const int& a) { return &a; }
>+const int* Ref(const int& a) {
>+  return &a;
>+}
> 
> }  // anonymous namespace
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bitbuffer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bitbuffer.cc
>index 86247af6978..025cfe1516a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bitbuffer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bitbuffer.cc
>@@ -200,8 +200,8 @@ bool BitBuffer::ReadSignedExponentialGolomb(int32_t* val) {
>   return true;
> }
> 
>-void BitBuffer::GetCurrentOffset(
>-    size_t* out_byte_offset, size_t* out_bit_offset) {
>+void BitBuffer::GetCurrentOffset(size_t* out_byte_offset,
>+                                 size_t* out_bit_offset) {
>   RTC_CHECK(out_byte_offset != nullptr);
>   RTC_CHECK(out_bit_offset != nullptr);
>   *out_byte_offset = byte_offset_;
>@@ -219,8 +219,7 @@ bool BitBuffer::Seek(size_t byte_offset, size_t bit_offset) {
> }
> 
> BitBufferWriter::BitBufferWriter(uint8_t* bytes, size_t byte_count)
>-    : BitBuffer(bytes, byte_count), writable_bytes_(bytes) {
>-}
>+    : BitBuffer(bytes, byte_count), writable_bytes_(bytes) {}
> 
> bool BitBufferWriter::WriteUInt8(uint8_t val) {
>   return WriteBits(val, sizeof(uint8_t) * 8);
>@@ -251,8 +250,8 @@ bool BitBufferWriter::WriteBits(uint64_t val, size_t bit_count) {
>   size_t remaining_bits_in_current_byte = 8 - bit_offset_;
>   size_t bits_in_first_byte =
>       std::min(bit_count, remaining_bits_in_current_byte);
>-  *bytes = WritePartialByte(
>-      HighestByte(val), bits_in_first_byte, *bytes, bit_offset_);
>+  *bytes = WritePartialByte(HighestByte(val), bits_in_first_byte, *bytes,
>+                            bit_offset_);
>   if (bit_count <= remaining_bits_in_current_byte) {
>     // Nothing left to write, so quit early.
>     return ConsumeBits(total_bits);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bitbuffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bitbuffer.h
>index 85194148073..146577a15af 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bitbuffer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bitbuffer.h
>@@ -11,8 +11,8 @@
> #ifndef RTC_BASE_BITBUFFER_H_
> #define RTC_BASE_BITBUFFER_H_
> 
>-#include <stdint.h>  // For integer types.
> #include <stddef.h>  // For size_t.
>+#include <stdint.h>  // For integer types.
> 
> #include "rtc_base/constructormagic.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bitbuffer_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bitbuffer_unittest.cc
>index abf723250a2..a3f140f4994 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bitbuffer_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bitbuffer_unittest.cc
>@@ -165,8 +165,8 @@ TEST(BitBufferTest, SetOffsetValues) {
>   EXPECT_EQ(0u, bit_offset);
>   EXPECT_FALSE(buffer.Seek(4, 1));
> 
>-  // Disable death test on Android because it relies on fork() and doesn't play
>-  // nicely.
>+// Disable death test on Android because it relies on fork() and doesn't play
>+// nicely.
> #if GTEST_HAS_DEATH_TEST
> #if !defined(WEBRTC_ANDROID)
>   // Passing a null out parameter is death.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/buffer.h
>index eff3e40f0dd..22f89372deb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/buffer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/buffer.h
>@@ -20,6 +20,7 @@
> #include "api/array_view.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/type_traits.h"
>+#include "rtc_base/zero_memory.h"
> 
> namespace rtc {
> 
>@@ -44,7 +45,10 @@ struct BufferCompat {
> 
> // Basic buffer class, can be grown and shrunk dynamically.
> // Unlike std::string/vector, does not initialize data when increasing size.
>-template <typename T>
>+// If "ZeroOnFree" is true, any memory is explicitly cleared before releasing.
>+// The type alias "ZeroOnFreeBuffer" below should be used instead of setting
>+// "ZeroOnFree" in the template manually to "true".
>+template <typename T, bool ZeroOnFree = false>
> class BufferT {
>   // We want T's destructor and default constructor to be trivial, i.e. perform
>   // no action, so that we don't have to touch the memory we allocate and
>@@ -83,7 +87,7 @@ class BufferT {
>   BufferT(size_t size, size_t capacity)
>       : size_(size),
>         capacity_(std::max(size, capacity)),
>-        data_(new T[capacity_]) {
>+        data_(capacity_ > 0 ? new T[capacity_] : nullptr) {
>     RTC_DCHECK(IsConsistent());
>   }
> 
>@@ -108,6 +112,8 @@ class BufferT {
>                 internal::BufferCompat<T, U>::value>::type* = nullptr>
>   BufferT(U (&array)[N]) : BufferT(array, N) {}
> 
>+  ~BufferT() { MaybeZeroCompleteBuffer(); }
>+
>   // Get a pointer to the data. Just .data() will give you a (const) T*, but if
>   // T is a byte-sized integer, you may also use .data<U>() for any other
>   // byte-sized integer U.
>@@ -195,8 +201,12 @@ class BufferT {
>                 internal::BufferCompat<T, U>::value>::type* = nullptr>
>   void SetData(const U* data, size_t size) {
>     RTC_DCHECK(IsConsistent());
>+    const size_t old_size = size_;
>     size_ = 0;
>     AppendData(data, size);
>+    if (ZeroOnFree && size_ < old_size) {
>+      ZeroTrailingData(old_size - size_);
>+    }
>   }
> 
>   template <typename U,
>@@ -229,8 +239,13 @@ class BufferT {
>                 internal::BufferCompat<T, U>::value>::type* = nullptr>
>   size_t SetData(size_t max_elements, F&& setter) {
>     RTC_DCHECK(IsConsistent());
>+    const size_t old_size = size_;
>     size_ = 0;
>-    return AppendData<U>(max_elements, std::forward<F>(setter));
>+    const size_t written = AppendData<U>(max_elements, std::forward<F>(setter));
>+    if (ZeroOnFree && size_ < old_size) {
>+      ZeroTrailingData(old_size - size_);
>+    }
>+    return written;
>   }
> 
>   // The AppendData functions add data to the end of the buffer. They accept
>@@ -301,8 +316,12 @@ class BufferT {
>   // the existing contents will be kept and the new space will be
>   // uninitialized.
>   void SetSize(size_t size) {
>+    const size_t old_size = size_;
>     EnsureCapacityWithHeadroom(size, true);
>     size_ = size;
>+    if (ZeroOnFree && size_ < old_size) {
>+      ZeroTrailingData(old_size - size_);
>+    }
>   }
> 
>   // Ensure that the buffer size can be increased to at least capacity without
>@@ -317,6 +336,7 @@ class BufferT {
>   // Resets the buffer to zero size without altering capacity. Works even if the
>   // buffer has been moved from.
>   void Clear() {
>+    MaybeZeroCompleteBuffer();
>     size_ = 0;
>     RTC_DCHECK(IsConsistent());
>   }
>@@ -346,11 +366,29 @@ class BufferT {
> 
>     std::unique_ptr<T[]> new_data(new T[new_capacity]);
>     std::memcpy(new_data.get(), data_.get(), size_ * sizeof(T));
>+    MaybeZeroCompleteBuffer();
>     data_ = std::move(new_data);
>     capacity_ = new_capacity;
>     RTC_DCHECK(IsConsistent());
>   }
> 
>+  // Zero the complete buffer if template argument "ZeroOnFree" is true.
>+  void MaybeZeroCompleteBuffer() {
>+    if (ZeroOnFree && capacity_) {
>+      // It would be sufficient to only zero "size_" elements, as all other
>+      // methods already ensure that the unused capacity contains no sensitive
>+      // data - but better safe than sorry.
>+      ExplicitZeroMemory(data_.get(), capacity_ * sizeof(T));
>+    }
>+  }
>+
>+  // Zero the first "count" elements of unused capacity.
>+  void ZeroTrailingData(size_t count) {
>+    RTC_DCHECK(IsConsistent());
>+    RTC_DCHECK_LE(count, capacity_ - size_);
>+    ExplicitZeroMemory(data_.get() + size_, count * sizeof(T));
>+  }
>+
>   // Precondition for all methods except Clear and the destructor.
>   // Postcondition for all methods except move construction and move
>   // assignment, which leave the moved-from object in a possibly inconsistent
>@@ -382,6 +420,10 @@ class BufferT {
> // By far the most common sort of buffer.
> using Buffer = BufferT<uint8_t>;
> 
>+// A buffer that zeros memory before releasing it.
>+template <typename T>
>+using ZeroOnFreeBuffer = BufferT<T, true>;
>+
> }  // namespace rtc
> 
> #endif  // RTC_BASE_BUFFER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/buffer_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/buffer_unittest.cc
>index 5cd18891b91..ae976f13943 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/buffer_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/buffer_unittest.cc
>@@ -208,16 +208,16 @@ TEST(BufferTest, TestClear) {
>   EXPECT_EQ(buf.size(), 15u);
>   EXPECT_EQ(buf.capacity(), 15u);
>   EXPECT_FALSE(buf.empty());
>-  const char *data = buf.data<char>();
>+  const char* data = buf.data<char>();
>   buf.Clear();
>   EXPECT_EQ(buf.size(), 0u);
>-  EXPECT_EQ(buf.capacity(), 15u);  // Hasn't shrunk.
>-  EXPECT_EQ(buf.data<char>(), data); // No reallocation.
>+  EXPECT_EQ(buf.capacity(), 15u);     // Hasn't shrunk.
>+  EXPECT_EQ(buf.data<char>(), data);  // No reallocation.
>   EXPECT_TRUE(buf.empty());
> }
> 
> TEST(BufferTest, TestLambdaSetAppend) {
>-  auto setter = [] (rtc::ArrayView<uint8_t> av) {
>+  auto setter = [](rtc::ArrayView<uint8_t> av) {
>     for (int i = 0; i != 15; ++i)
>       av[i] = kTestData[i];
>     return 15;
>@@ -237,7 +237,7 @@ TEST(BufferTest, TestLambdaSetAppend) {
> }
> 
> TEST(BufferTest, TestLambdaSetAppendSigned) {
>-  auto setter = [] (rtc::ArrayView<int8_t> av) {
>+  auto setter = [](rtc::ArrayView<int8_t> av) {
>     for (int i = 0; i != 15; ++i)
>       av[i] = kTestData[i];
>     return 15;
>@@ -257,7 +257,7 @@ TEST(BufferTest, TestLambdaSetAppendSigned) {
> }
> 
> TEST(BufferTest, TestLambdaAppendEmpty) {
>-  auto setter = [] (rtc::ArrayView<uint8_t> av) {
>+  auto setter = [](rtc::ArrayView<uint8_t> av) {
>     for (int i = 0; i != 15; ++i)
>       av[i] = kTestData[i];
>     return 15;
>@@ -275,7 +275,7 @@ TEST(BufferTest, TestLambdaAppendEmpty) {
> }
> 
> TEST(BufferTest, TestLambdaAppendPartial) {
>-  auto setter = [] (rtc::ArrayView<uint8_t> av) {
>+  auto setter = [](rtc::ArrayView<uint8_t> av) {
>     for (int i = 0; i != 7; ++i)
>       av[i] = kTestData[i];
>     return 7;
>@@ -283,15 +283,15 @@ TEST(BufferTest, TestLambdaAppendPartial) {
> 
>   Buffer buf;
>   EXPECT_EQ(buf.AppendData(15, setter), 7u);
>-  EXPECT_EQ(buf.size(), 7u);            // Size is exactly what we wrote.
>-  EXPECT_GE(buf.capacity(), 7u);        // Capacity is valid.
>-  EXPECT_NE(buf.data<char>(), nullptr); // Data is actually stored.
>+  EXPECT_EQ(buf.size(), 7u);             // Size is exactly what we wrote.
>+  EXPECT_GE(buf.capacity(), 7u);         // Capacity is valid.
>+  EXPECT_NE(buf.data<char>(), nullptr);  // Data is actually stored.
>   EXPECT_FALSE(buf.empty());
> }
> 
> TEST(BufferTest, TestMutableLambdaSetAppend) {
>   uint8_t magic_number = 17;
>-  auto setter = [magic_number] (rtc::ArrayView<uint8_t> av) mutable {
>+  auto setter = [magic_number](rtc::ArrayView<uint8_t> av) mutable {
>     for (int i = 0; i != 15; ++i) {
>       av[i] = magic_number;
>       ++magic_number;
>@@ -304,9 +304,9 @@ TEST(BufferTest, TestMutableLambdaSetAppend) {
>   Buffer buf;
>   EXPECT_EQ(buf.SetData(15, setter), 15u);
>   EXPECT_EQ(buf.AppendData(15, setter), 15u);
>-  EXPECT_EQ(buf.size(), 30u);           // Size is exactly what we wrote.
>-  EXPECT_GE(buf.capacity(), 30u);       // Capacity is valid.
>-  EXPECT_NE(buf.data<char>(), nullptr); // Data is actually stored.
>+  EXPECT_EQ(buf.size(), 30u);            // Size is exactly what we wrote.
>+  EXPECT_GE(buf.capacity(), 30u);        // Capacity is valid.
>+  EXPECT_NE(buf.data<char>(), nullptr);  // Data is actually stored.
>   EXPECT_FALSE(buf.empty());
> 
>   for (uint8_t i = 0; i != buf.size(); ++i) {
>@@ -434,4 +434,81 @@ TEST(BufferTest, TestStruct) {
>   EXPECT_EQ(kObsidian, buf[2].stone);
> }
> 
>+TEST(ZeroOnFreeBufferTest, TestZeroOnSetData) {
>+  ZeroOnFreeBuffer<uint8_t> buf(kTestData, 7);
>+  const uint8_t* old_data = buf.data();
>+  const size_t old_capacity = buf.capacity();
>+  const size_t old_size = buf.size();
>+  constexpr size_t offset = 1;
>+  buf.SetData(kTestData + offset, 2);
>+  // Sanity checks to make sure the underlying heap memory was not reallocated.
>+  EXPECT_EQ(old_data, buf.data());
>+  EXPECT_EQ(old_capacity, buf.capacity());
>+  // The first two elements have been overwritten, and the remaining five have
>+  // been zeroed.
>+  EXPECT_EQ(kTestData[offset], buf[0]);
>+  EXPECT_EQ(kTestData[offset + 1], buf[1]);
>+  for (size_t i = 2; i < old_size; i++) {
>+    EXPECT_EQ(0, old_data[i]);
>+  }
>+}
>+
>+TEST(ZeroOnFreeBufferTest, TestZeroOnSetDataFromSetter) {
>+  static constexpr size_t offset = 1;
>+  const auto setter = [](rtc::ArrayView<uint8_t> av) {
>+    for (int i = 0; i != 2; ++i)
>+      av[i] = kTestData[offset + i];
>+    return 2;
>+  };
>+
>+  ZeroOnFreeBuffer<uint8_t> buf(kTestData, 7);
>+  const uint8_t* old_data = buf.data();
>+  const size_t old_capacity = buf.capacity();
>+  const size_t old_size = buf.size();
>+  buf.SetData(2, setter);
>+  // Sanity checks to make sure the underlying heap memory was not reallocated.
>+  EXPECT_EQ(old_data, buf.data());
>+  EXPECT_EQ(old_capacity, buf.capacity());
>+  // The first two elements have been overwritten, and the remaining five have
>+  // been zeroed.
>+  EXPECT_EQ(kTestData[offset], buf[0]);
>+  EXPECT_EQ(kTestData[offset + 1], buf[1]);
>+  for (size_t i = 2; i < old_size; i++) {
>+    EXPECT_EQ(0, old_data[i]);
>+  }
>+}
>+
>+TEST(ZeroOnFreeBufferTest, TestZeroOnSetSize) {
>+  ZeroOnFreeBuffer<uint8_t> buf(kTestData, 7);
>+  const uint8_t* old_data = buf.data();
>+  const size_t old_capacity = buf.capacity();
>+  const size_t old_size = buf.size();
>+  buf.SetSize(2);
>+  // Sanity checks to make sure the underlying heap memory was not reallocated.
>+  EXPECT_EQ(old_data, buf.data());
>+  EXPECT_EQ(old_capacity, buf.capacity());
>+  // The first two elements have not been modified and the remaining five have
>+  // been zeroed.
>+  EXPECT_EQ(kTestData[0], buf[0]);
>+  EXPECT_EQ(kTestData[1], buf[1]);
>+  for (size_t i = 2; i < old_size; i++) {
>+    EXPECT_EQ(0, old_data[i]);
>+  }
>+}
>+
>+TEST(ZeroOnFreeBufferTest, TestZeroOnClear) {
>+  ZeroOnFreeBuffer<uint8_t> buf(kTestData, 7);
>+  const uint8_t* old_data = buf.data();
>+  const size_t old_capacity = buf.capacity();
>+  const size_t old_size = buf.size();
>+  buf.Clear();
>+  // Sanity checks to make sure the underlying heap memory was not reallocated.
>+  EXPECT_EQ(old_data, buf.data());
>+  EXPECT_EQ(old_capacity, buf.capacity());
>+  // The underlying memory was not released but cleared.
>+  for (size_t i = 0; i < old_size; i++) {
>+    EXPECT_EQ(0, old_data[i]);
>+  }
>+}
>+
> }  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bufferqueue.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bufferqueue.cc
>index 3b082a5d25b..48ff2e6d317 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bufferqueue.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bufferqueue.cc
>@@ -15,8 +15,7 @@
> namespace rtc {
> 
> BufferQueue::BufferQueue(size_t capacity, size_t default_size)
>-    : capacity_(capacity), default_size_(default_size) {
>-}
>+    : capacity_(capacity), default_size_(default_size) {}
> 
> BufferQueue::~BufferQueue() {
>   CritScope cs(&crit_);
>@@ -64,7 +63,8 @@ bool BufferQueue::ReadFront(void* buffer, size_t bytes, size_t* bytes_read) {
>   return true;
> }
> 
>-bool BufferQueue::WriteBack(const void* buffer, size_t bytes,
>+bool BufferQueue::WriteBack(const void* buffer,
>+                            size_t bytes,
>                             size_t* bytes_written) {
>   CritScope cs(&crit_);
>   if (queue_.size() == capacity_) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bytebuffer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bytebuffer.cc
>index ea4de3a6965..94fc6acf080 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bytebuffer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bytebuffer.cc
>@@ -14,134 +14,28 @@
> 
> #include <algorithm>
> 
>-#include "rtc_base/basictypes.h"
>-#include "rtc_base/byteorder.h"
>-
> namespace rtc {
> 
>-static const int DEFAULT_SIZE = 4096;
>-
>-ByteBufferWriter::ByteBufferWriter()
>-    : ByteBuffer(ORDER_NETWORK) {
>-  Construct(nullptr, DEFAULT_SIZE);
>-}
>+ByteBufferWriter::ByteBufferWriter() : ByteBufferWriterT() {}
> 
> ByteBufferWriter::ByteBufferWriter(ByteOrder byte_order)
>-    : ByteBuffer(byte_order) {
>-  Construct(nullptr, DEFAULT_SIZE);
>-}
>+    : ByteBufferWriterT(byte_order) {}
> 
> ByteBufferWriter::ByteBufferWriter(const char* bytes, size_t len)
>-    : ByteBuffer(ORDER_NETWORK) {
>-  Construct(bytes, len);
>-}
>+    : ByteBufferWriterT(bytes, len) {}
> 
>-ByteBufferWriter::ByteBufferWriter(const char* bytes, size_t len,
>+ByteBufferWriter::ByteBufferWriter(const char* bytes,
>+                                   size_t len,
>                                    ByteOrder byte_order)
>-    : ByteBuffer(byte_order) {
>-  Construct(bytes, len);
>-}
>-
>-void ByteBufferWriter::Construct(const char* bytes, size_t len) {
>-  size_ = len;
>-  bytes_ = new char[size_];
>-
>-  if (bytes) {
>-    end_ = len;
>-    memcpy(bytes_, bytes, end_);
>-  } else {
>-    end_ = 0;
>-  }
>-}
>-
>-ByteBufferWriter::~ByteBufferWriter() {
>-  delete[] bytes_;
>-}
>-
>-void ByteBufferWriter::WriteUInt8(uint8_t val) {
>-  WriteBytes(reinterpret_cast<const char*>(&val), 1);
>-}
>-
>-void ByteBufferWriter::WriteUInt16(uint16_t val) {
>-  uint16_t v = (Order() == ORDER_NETWORK) ? HostToNetwork16(val) : val;
>-  WriteBytes(reinterpret_cast<const char*>(&v), 2);
>-}
>-
>-void ByteBufferWriter::WriteUInt24(uint32_t val) {
>-  uint32_t v = (Order() == ORDER_NETWORK) ? HostToNetwork32(val) : val;
>-  char* start = reinterpret_cast<char*>(&v);
>-  if (Order() == ORDER_NETWORK || IsHostBigEndian()) {
>-    ++start;
>-  }
>-  WriteBytes(start, 3);
>-}
>-
>-void ByteBufferWriter::WriteUInt32(uint32_t val) {
>-  uint32_t v = (Order() == ORDER_NETWORK) ? HostToNetwork32(val) : val;
>-  WriteBytes(reinterpret_cast<const char*>(&v), 4);
>-}
>-
>-void ByteBufferWriter::WriteUInt64(uint64_t val) {
>-  uint64_t v = (Order() == ORDER_NETWORK) ? HostToNetwork64(val) : val;
>-  WriteBytes(reinterpret_cast<const char*>(&v), 8);
>-}
>-
>-// Serializes an unsigned varint in the format described by
>-// https://developers.google.com/protocol-buffers/docs/encoding#varints
>-// with the caveat that integers are 64-bit, not 128-bit.
>-void ByteBufferWriter::WriteUVarint(uint64_t val) {
>-  while (val >= 0x80) {
>-    // Write 7 bits at a time, then set the msb to a continuation byte (msb=1).
>-    char byte = static_cast<char>(val) | 0x80;
>-    WriteBytes(&byte, 1);
>-    val >>= 7;
>-  }
>-  char last_byte = static_cast<char>(val);
>-  WriteBytes(&last_byte, 1);
>-}
>-
>-void ByteBufferWriter::WriteString(const std::string& val) {
>-  WriteBytes(val.c_str(), val.size());
>-}
>-
>-void ByteBufferWriter::WriteBytes(const char* val, size_t len) {
>-  memcpy(ReserveWriteBuffer(len), val, len);
>-}
>-
>-char* ByteBufferWriter::ReserveWriteBuffer(size_t len) {
>-  if (Length() + len > Capacity())
>-    Resize(Length() + len);
>-
>-  char* start = bytes_ + end_;
>-  end_ += len;
>-  return start;
>-}
>-
>-void ByteBufferWriter::Resize(size_t size) {
>-  size_t len = std::min(end_, size);
>-  if (size > size_) {
>-    // Reallocate a larger buffer.
>-    size_ = std::max(size, 3 * size_ / 2);
>-    char* new_bytes = new char[size_];
>-    memcpy(new_bytes, bytes_, len);
>-    delete [] bytes_;
>-    bytes_ = new_bytes;
>-  }
>-  end_ = len;
>-}
>-
>-void ByteBufferWriter::Clear() {
>-  memset(bytes_, 0, size_);
>-  end_ = 0;
>-}
>-
>+    : ByteBufferWriterT(bytes, len, byte_order) {}
> 
> ByteBufferReader::ByteBufferReader(const char* bytes, size_t len)
>     : ByteBuffer(ORDER_NETWORK) {
>   Construct(bytes, len);
> }
> 
>-ByteBufferReader::ByteBufferReader(const char* bytes, size_t len,
>+ByteBufferReader::ByteBufferReader(const char* bytes,
>+                                   size_t len,
>                                    ByteOrder byte_order)
>     : ByteBuffer(byte_order) {
>   Construct(bytes, len);
>@@ -170,13 +64,15 @@ void ByteBufferReader::Construct(const char* bytes, size_t len) {
> }
> 
> bool ByteBufferReader::ReadUInt8(uint8_t* val) {
>-  if (!val) return false;
>+  if (!val)
>+    return false;
> 
>   return ReadBytes(reinterpret_cast<char*>(val), 1);
> }
> 
> bool ByteBufferReader::ReadUInt16(uint16_t* val) {
>-  if (!val) return false;
>+  if (!val)
>+    return false;
> 
>   uint16_t v;
>   if (!ReadBytes(reinterpret_cast<char*>(&v), 2)) {
>@@ -188,7 +84,8 @@ bool ByteBufferReader::ReadUInt16(uint16_t* val) {
> }
> 
> bool ByteBufferReader::ReadUInt24(uint32_t* val) {
>-  if (!val) return false;
>+  if (!val)
>+    return false;
> 
>   uint32_t v = 0;
>   char* read_into = reinterpret_cast<char*>(&v);
>@@ -205,7 +102,8 @@ bool ByteBufferReader::ReadUInt24(uint32_t* val) {
> }
> 
> bool ByteBufferReader::ReadUInt32(uint32_t* val) {
>-  if (!val) return false;
>+  if (!val)
>+    return false;
> 
>   uint32_t v;
>   if (!ReadBytes(reinterpret_cast<char*>(&v), 4)) {
>@@ -217,7 +115,8 @@ bool ByteBufferReader::ReadUInt32(uint32_t* val) {
> }
> 
> bool ByteBufferReader::ReadUInt64(uint64_t* val) {
>-  if (!val) return false;
>+  if (!val)
>+    return false;
> 
>   uint64_t v;
>   if (!ReadBytes(reinterpret_cast<char*>(&v), 8)) {
>@@ -252,7 +151,8 @@ bool ByteBufferReader::ReadUVarint(uint64_t* val) {
> }
> 
> bool ByteBufferReader::ReadString(std::string* val, size_t len) {
>-  if (!val) return false;
>+  if (!val)
>+    return false;
> 
>   if (len > Length()) {
>     return false;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bytebuffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bytebuffer.h
>index 2062fbb5edb..9036bcde6fd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bytebuffer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bytebuffer.h
>@@ -13,8 +13,8 @@
> 
> #include <string>
> 
>-#include "rtc_base/basictypes.h"
> #include "rtc_base/buffer.h"
>+#include "rtc_base/byteorder.h"
> #include "rtc_base/constructormagic.h"
> 
> namespace rtc {
>@@ -36,51 +36,113 @@ class ByteBuffer {
>   RTC_DISALLOW_COPY_AND_ASSIGN(ByteBuffer);
> };
> 
>-class ByteBufferWriter : public ByteBuffer {
>+template <class BufferClassT>
>+class ByteBufferWriterT : public ByteBuffer {
>  public:
>   // |byte_order| defines order of bytes in the buffer.
>-  ByteBufferWriter();
>-  explicit ByteBufferWriter(ByteOrder byte_order);
>-  ByteBufferWriter(const char* bytes, size_t len);
>-  ByteBufferWriter(const char* bytes, size_t len, ByteOrder byte_order);
>-
>-  ~ByteBufferWriter();
>-
>-  const char* Data() const { return bytes_; }
>-  size_t Length() const { return end_; }
>-  size_t Capacity() const { return size_; }
>+  ByteBufferWriterT() : ByteBuffer(ORDER_NETWORK) {
>+    Construct(nullptr, kDefaultCapacity);
>+  }
>+  explicit ByteBufferWriterT(ByteOrder byte_order) : ByteBuffer(byte_order) {
>+    Construct(nullptr, kDefaultCapacity);
>+  }
>+  ByteBufferWriterT(const char* bytes, size_t len) : ByteBuffer(ORDER_NETWORK) {
>+    Construct(bytes, len);
>+  }
>+  ByteBufferWriterT(const char* bytes, size_t len, ByteOrder byte_order)
>+      : ByteBuffer(byte_order) {
>+    Construct(bytes, len);
>+  }
>+
>+  const char* Data() const { return buffer_.data(); }
>+  size_t Length() const { return buffer_.size(); }
>+  size_t Capacity() const { return buffer_.capacity(); }
> 
>   // Write value to the buffer. Resizes the buffer when it is
>   // neccessary.
>-  void WriteUInt8(uint8_t val);
>-  void WriteUInt16(uint16_t val);
>-  void WriteUInt24(uint32_t val);
>-  void WriteUInt32(uint32_t val);
>-  void WriteUInt64(uint64_t val);
>-  void WriteUVarint(uint64_t val);
>-  void WriteString(const std::string& val);
>-  void WriteBytes(const char* val, size_t len);
>+  void WriteUInt8(uint8_t val) {
>+    WriteBytes(reinterpret_cast<const char*>(&val), 1);
>+  }
>+  void WriteUInt16(uint16_t val) {
>+    uint16_t v = (Order() == ORDER_NETWORK) ? HostToNetwork16(val) : val;
>+    WriteBytes(reinterpret_cast<const char*>(&v), 2);
>+  }
>+  void WriteUInt24(uint32_t val) {
>+    uint32_t v = (Order() == ORDER_NETWORK) ? HostToNetwork32(val) : val;
>+    char* start = reinterpret_cast<char*>(&v);
>+    if (Order() == ORDER_NETWORK || IsHostBigEndian()) {
>+      ++start;
>+    }
>+    WriteBytes(start, 3);
>+  }
>+  void WriteUInt32(uint32_t val) {
>+    uint32_t v = (Order() == ORDER_NETWORK) ? HostToNetwork32(val) : val;
>+    WriteBytes(reinterpret_cast<const char*>(&v), 4);
>+  }
>+  void WriteUInt64(uint64_t val) {
>+    uint64_t v = (Order() == ORDER_NETWORK) ? HostToNetwork64(val) : val;
>+    WriteBytes(reinterpret_cast<const char*>(&v), 8);
>+  }
>+  // Serializes an unsigned varint in the format described by
>+  // https://developers.google.com/protocol-buffers/docs/encoding#varints
>+  // with the caveat that integers are 64-bit, not 128-bit.
>+  void WriteUVarint(uint64_t val) {
>+    while (val >= 0x80) {
>+      // Write 7 bits at a time, then set the msb to a continuation byte
>+      // (msb=1).
>+      char byte = static_cast<char>(val) | 0x80;
>+      WriteBytes(&byte, 1);
>+      val >>= 7;
>+    }
>+    char last_byte = static_cast<char>(val);
>+    WriteBytes(&last_byte, 1);
>+  }
>+  void WriteString(const std::string& val) {
>+    WriteBytes(val.c_str(), val.size());
>+  }
>+  void WriteBytes(const char* val, size_t len) { buffer_.AppendData(val, len); }
> 
>   // Reserves the given number of bytes and returns a char* that can be written
>   // into. Useful for functions that require a char* buffer and not a
>   // ByteBufferWriter.
>-  char* ReserveWriteBuffer(size_t len);
>+  char* ReserveWriteBuffer(size_t len) {
>+    buffer_.SetSize(buffer_.size() + len);
>+    return buffer_.data();
>+  }
> 
>   // Resize the buffer to the specified |size|.
>-  void Resize(size_t size);
>+  void Resize(size_t size) { buffer_.SetSize(size); }
> 
>   // Clears the contents of the buffer. After this, Length() will be 0.
>-  void Clear();
>+  void Clear() { buffer_.Clear(); }
> 
>  private:
>-  void Construct(const char* bytes, size_t size);
>+  static constexpr size_t kDefaultCapacity = 4096;
> 
>-  char* bytes_;
>-  size_t size_;
>-  size_t end_;
>+  void Construct(const char* bytes, size_t size) {
>+    if (bytes) {
>+      buffer_.AppendData(bytes, size);
>+    } else {
>+      buffer_.EnsureCapacity(size);
>+    }
>+  }
>+
>+  BufferClassT buffer_;
> 
>   // There are sensible ways to define these, but they aren't needed in our code
>   // base.
>+  RTC_DISALLOW_COPY_AND_ASSIGN(ByteBufferWriterT);
>+};
>+
>+class ByteBufferWriter : public ByteBufferWriterT<BufferT<char>> {
>+ public:
>+  // |byte_order| defines order of bytes in the buffer.
>+  ByteBufferWriter();
>+  explicit ByteBufferWriter(ByteOrder byte_order);
>+  ByteBufferWriter(const char* bytes, size_t len);
>+  ByteBufferWriter(const char* bytes, size_t len, ByteOrder byte_order);
>+
>+ private:
>   RTC_DISALLOW_COPY_AND_ASSIGN(ByteBufferWriter);
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bytebuffer_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bytebuffer_unittest.cc
>index 6140e9fbfca..718f35c8d89 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bytebuffer_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/bytebuffer_unittest.cc
>@@ -79,8 +79,8 @@ TEST(ByteBufferTest, TestBufferLength) {
> }
> 
> TEST(ByteBufferTest, TestReadWriteBuffer) {
>-  ByteBufferWriter::ByteOrder orders[2] = { ByteBufferWriter::ORDER_HOST,
>-                                            ByteBufferWriter::ORDER_NETWORK };
>+  ByteBufferWriter::ByteOrder orders[2] = {ByteBufferWriter::ORDER_HOST,
>+                                           ByteBufferWriter::ORDER_NETWORK};
>   for (size_t i = 0; i < arraysize(orders); i++) {
>     ByteBufferWriter buffer(orders[i]);
>     EXPECT_EQ(orders[i], buffer.Order());
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/byteorder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/byteorder.h
>index 85f0cc544aa..86546a5ca35 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/byteorder.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/byteorder.h
>@@ -11,11 +11,13 @@
> #ifndef RTC_BASE_BYTEORDER_H_
> #define RTC_BASE_BYTEORDER_H_
> 
>+#include <stdint.h>
>+
> #if defined(WEBRTC_POSIX) && !defined(__native_client__)
> #include <arpa/inet.h>
> #endif
> 
>-#include "rtc_base/basictypes.h"
>+#include "rtc_base/system/arch.h"
> 
> #if defined(WEBRTC_MAC)
> #include <libkern/OSByteOrder.h>
>@@ -51,7 +53,7 @@
> #define be64toh(v) ntohll(v)
> #endif
> 
>-#if defined(RTC_ARCH_CPU_LITTLE_ENDIAN)
>+#if defined(WEBRTC_ARCH_LITTLE_ENDIAN)
> #define htole16(v) (v)
> #define htole32(v) (v)
> #define htole64(v) (v)
>@@ -62,7 +64,7 @@
> #define htobe64(v) __builtin_bswap64(v)
> #define be64toh(v) __builtin_bswap64(v)
> #endif
>-#elif defined(RTC_ARCH_CPU_BIG_ENDIAN)
>+#elif defined(WEBRTC_ARCH_BIG_ENDIAN)
> #define htole16(v) __builtin_bswap16(v)
> #define htole32(v) __builtin_bswap32(v)
> #define htole64(v) __builtin_bswap64(v)
>@@ -74,8 +76,8 @@
> #define be64toh(v) (v)
> #endif
> #else
>-#error RTC_ARCH_CPU_BIG_ENDIAN or RTC_ARCH_CPU_LITTLE_ENDIAN must be defined.
>-#endif  // defined(RTC_ARCH_CPU_LITTLE_ENDIAN)
>+#error WEBRTC_ARCH_BIG_ENDIAN or WEBRTC_ARCH_LITTLE_ENDIAN must be defined.
>+#endif  // defined(WEBRTC_ARCH_LITTLE_ENDIAN)
> #elif defined(WEBRTC_POSIX)
> #include <endian.h>
> #endif
>@@ -142,7 +144,7 @@ inline uint64_t GetLE64(const void* memory) {
> 
> // Check if the current host is big endian.
> inline bool IsHostBigEndian() {
>-#if defined(RTC_ARCH_CPU_BIG_ENDIAN)
>+#if defined(WEBRTC_ARCH_BIG_ENDIAN)
>   return true;
> #else
>   return false;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/byteorder_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/byteorder_unittest.cc
>index 30dc5faaf8e..75509de9c2b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/byteorder_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/byteorder_unittest.cc
>@@ -80,4 +80,3 @@ TEST(ByteOrderTest, TestGet) {
> }
> 
> }  // namespace rtc
>-
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/callback.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/callback.h
>index 0e035ad6a02..65b7c9cdf35 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/callback.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/callback.h
>@@ -73,8 +73,9 @@ class Callback0 {
>  public:
>   // Default copy operations are appropriate for this class.
>   Callback0() {}
>-  template <class T> Callback0(const T& functor)
>-      : helper_(new RefCountedObject< HelperImpl<T> >(functor)) {}
>+  template <class T>
>+  Callback0(const T& functor)
>+      : helper_(new RefCountedObject<HelperImpl<T> >(functor)) {}
>   R operator()() {
>     if (empty())
>       return R();
>@@ -87,24 +88,23 @@ class Callback0 {
>     virtual ~Helper() {}
>     virtual R Run() = 0;
>   };
>-  template <class T> struct HelperImpl : Helper {
>+  template <class T>
>+  struct HelperImpl : Helper {
>     explicit HelperImpl(const T& functor) : functor_(functor) {}
>-    virtual R Run() {
>-      return functor_();
>-    }
>+    virtual R Run() { return functor_(); }
>     T functor_;
>   };
>   scoped_refptr<Helper> helper_;
> };
> 
>-template <class R,
>-          class P1>
>+template <class R, class P1>
> class Callback1 {
>  public:
>   // Default copy operations are appropriate for this class.
>   Callback1() {}
>-  template <class T> Callback1(const T& functor)
>-      : helper_(new RefCountedObject< HelperImpl<T> >(functor)) {}
>+  template <class T>
>+  Callback1(const T& functor)
>+      : helper_(new RefCountedObject<HelperImpl<T> >(functor)) {}
>   R operator()(P1 p1) {
>     if (empty())
>       return R();
>@@ -117,25 +117,23 @@ class Callback1 {
>     virtual ~Helper() {}
>     virtual R Run(P1 p1) = 0;
>   };
>-  template <class T> struct HelperImpl : Helper {
>+  template <class T>
>+  struct HelperImpl : Helper {
>     explicit HelperImpl(const T& functor) : functor_(functor) {}
>-    virtual R Run(P1 p1) {
>-      return functor_(p1);
>-    }
>+    virtual R Run(P1 p1) { return functor_(p1); }
>     T functor_;
>   };
>   scoped_refptr<Helper> helper_;
> };
> 
>-template <class R,
>-          class P1,
>-          class P2>
>+template <class R, class P1, class P2>
> class Callback2 {
>  public:
>   // Default copy operations are appropriate for this class.
>   Callback2() {}
>-  template <class T> Callback2(const T& functor)
>-      : helper_(new RefCountedObject< HelperImpl<T> >(functor)) {}
>+  template <class T>
>+  Callback2(const T& functor)
>+      : helper_(new RefCountedObject<HelperImpl<T> >(functor)) {}
>   R operator()(P1 p1, P2 p2) {
>     if (empty())
>       return R();
>@@ -148,26 +146,23 @@ class Callback2 {
>     virtual ~Helper() {}
>     virtual R Run(P1 p1, P2 p2) = 0;
>   };
>-  template <class T> struct HelperImpl : Helper {
>+  template <class T>
>+  struct HelperImpl : Helper {
>     explicit HelperImpl(const T& functor) : functor_(functor) {}
>-    virtual R Run(P1 p1, P2 p2) {
>-      return functor_(p1, p2);
>-    }
>+    virtual R Run(P1 p1, P2 p2) { return functor_(p1, p2); }
>     T functor_;
>   };
>   scoped_refptr<Helper> helper_;
> };
> 
>-template <class R,
>-          class P1,
>-          class P2,
>-          class P3>
>+template <class R, class P1, class P2, class P3>
> class Callback3 {
>  public:
>   // Default copy operations are appropriate for this class.
>   Callback3() {}
>-  template <class T> Callback3(const T& functor)
>-      : helper_(new RefCountedObject< HelperImpl<T> >(functor)) {}
>+  template <class T>
>+  Callback3(const T& functor)
>+      : helper_(new RefCountedObject<HelperImpl<T> >(functor)) {}
>   R operator()(P1 p1, P2 p2, P3 p3) {
>     if (empty())
>       return R();
>@@ -180,27 +175,23 @@ class Callback3 {
>     virtual ~Helper() {}
>     virtual R Run(P1 p1, P2 p2, P3 p3) = 0;
>   };
>-  template <class T> struct HelperImpl : Helper {
>+  template <class T>
>+  struct HelperImpl : Helper {
>     explicit HelperImpl(const T& functor) : functor_(functor) {}
>-    virtual R Run(P1 p1, P2 p2, P3 p3) {
>-      return functor_(p1, p2, p3);
>-    }
>+    virtual R Run(P1 p1, P2 p2, P3 p3) { return functor_(p1, p2, p3); }
>     T functor_;
>   };
>   scoped_refptr<Helper> helper_;
> };
> 
>-template <class R,
>-          class P1,
>-          class P2,
>-          class P3,
>-          class P4>
>+template <class R, class P1, class P2, class P3, class P4>
> class Callback4 {
>  public:
>   // Default copy operations are appropriate for this class.
>   Callback4() {}
>-  template <class T> Callback4(const T& functor)
>-      : helper_(new RefCountedObject< HelperImpl<T> >(functor)) {}
>+  template <class T>
>+  Callback4(const T& functor)
>+      : helper_(new RefCountedObject<HelperImpl<T> >(functor)) {}
>   R operator()(P1 p1, P2 p2, P3 p3, P4 p4) {
>     if (empty())
>       return R();
>@@ -213,7 +204,8 @@ class Callback4 {
>     virtual ~Helper() {}
>     virtual R Run(P1 p1, P2 p2, P3 p3, P4 p4) = 0;
>   };
>-  template <class T> struct HelperImpl : Helper {
>+  template <class T>
>+  struct HelperImpl : Helper {
>     explicit HelperImpl(const T& functor) : functor_(functor) {}
>     virtual R Run(P1 p1, P2 p2, P3 p3, P4 p4) {
>       return functor_(p1, p2, p3, p4);
>@@ -223,18 +215,14 @@ class Callback4 {
>   scoped_refptr<Helper> helper_;
> };
> 
>-template <class R,
>-          class P1,
>-          class P2,
>-          class P3,
>-          class P4,
>-          class P5>
>+template <class R, class P1, class P2, class P3, class P4, class P5>
> class Callback5 {
>  public:
>   // Default copy operations are appropriate for this class.
>   Callback5() {}
>-  template <class T> Callback5(const T& functor)
>-      : helper_(new RefCountedObject< HelperImpl<T> >(functor)) {}
>+  template <class T>
>+  Callback5(const T& functor)
>+      : helper_(new RefCountedObject<HelperImpl<T> >(functor)) {}
>   R operator()(P1 p1, P2 p2, P3 p3, P4 p4, P5 p5) {
>     if (empty())
>       return R();
>@@ -247,7 +235,8 @@ class Callback5 {
>     virtual ~Helper() {}
>     virtual R Run(P1 p1, P2 p2, P3 p3, P4 p4, P5 p5) = 0;
>   };
>-  template <class T> struct HelperImpl : Helper {
>+  template <class T>
>+  struct HelperImpl : Helper {
>     explicit HelperImpl(const T& functor) : functor_(functor) {}
>     virtual R Run(P1 p1, P2 p2, P3 p3, P4 p4, P5 p5) {
>       return functor_(p1, p2, p3, p4, p5);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/callback_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/callback_unittest.cc
>index 26bfd5daf45..ae86a4cd062 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/callback_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/callback_unittest.cc
>@@ -19,9 +19,15 @@ namespace rtc {
> namespace {
> 
> void f() {}
>-int g() { return 42; }
>-int h(int x) { return x * x; }
>-void i(int& x) { x *= x; }  // NOLINT: Testing refs
>+int g() {
>+  return 42;
>+}
>+int h(int x) {
>+  return x * x;
>+}
>+void i(int& x) {
>+  x *= x;
>+}  // NOLINT: Testing refs
> 
> struct BindTester {
>   int a() { return 24; }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/checks.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/checks.cc
>index ea3c712f1f0..32ac8d6e8b8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/checks.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/checks.cc
>@@ -11,16 +11,10 @@
> // Most of this was borrowed (with minor modifications) from V8's and Chromium's
> // src/base/logging.cc.
> 
>-// Use the C++ version to provide __GLIBCXX__.
> #include <cstdarg>
> #include <cstdio>
> #include <cstdlib>
> 
>-#if defined(__GLIBCXX__) && !defined(__UCLIBC__)
>-#include <cxxabi.h>
>-#include <execinfo.h>
>-#endif
>-
> #if defined(WEBRTC_ANDROID)
> #define RTC_LOG_TAG_ANDROID "rtc"
> #include <android/log.h>  // NOLINT
>@@ -35,123 +29,116 @@
> #elif defined(__native_client__) && __native_client__
> #define LAST_SYSTEM_ERROR (0)
> #elif defined(WEBRTC_POSIX)
>+#include <errno.h>
> #define LAST_SYSTEM_ERROR (errno)
> #endif  // WEBRTC_WIN
> 
> #include "rtc_base/checks.h"
> 
>-#if defined(_MSC_VER)
>-// Warning C4722: destructor never returns, potential memory leak.
>-// FatalMessage's dtor very intentionally aborts.
>-#pragma warning(disable:4722)
>-#endif
>-
> namespace rtc {
>-namespace {
>-
>-void VPrintError(const char* format, va_list args) {
>-#if defined(WEBRTC_ANDROID)
>-  __android_log_vprint(ANDROID_LOG_ERROR, RTC_LOG_TAG_ANDROID, format, args);
>-#else
>-  vfprintf(stderr, format, args);
>-#endif
>+namespace webrtc_checks_impl {
>+
>+// Reads one argument from args, appends it to s and advances fmt.
>+// Returns true iff an argument was sucessfully parsed.
>+bool ParseArg(va_list* args,
>+              const CheckArgType** fmt,
>+              std::ostream& s) {  // no-presubmit-check TODO(webrtc:8982)
>+  if (**fmt == CheckArgType::kEnd)
>+    return false;
>+
>+  switch (**fmt) {
>+    case CheckArgType::kInt:
>+      s << va_arg(*args, int);
>+      break;
>+    case CheckArgType::kLong:
>+      s << va_arg(*args, long);
>+      break;
>+    case CheckArgType::kLongLong:
>+      s << va_arg(*args, long long);
>+      break;
>+    case CheckArgType::kUInt:
>+      s << va_arg(*args, unsigned);
>+      break;
>+    case CheckArgType::kULong:
>+      s << va_arg(*args, unsigned long);
>+      break;
>+    case CheckArgType::kULongLong:
>+      s << va_arg(*args, unsigned long long);
>+      break;
>+    case CheckArgType::kDouble:
>+      s << va_arg(*args, double);
>+      break;
>+    case CheckArgType::kLongDouble:
>+      s << va_arg(*args, long double);
>+      break;
>+    case CheckArgType::kCharP:
>+      s << va_arg(*args, const char*);
>+      break;
>+    case CheckArgType::kStdString:
>+      s << *va_arg(*args, const std::string*);
>+      break;
>+    case CheckArgType::kVoidP:
>+      s << reinterpret_cast<std::uintptr_t>(va_arg(*args, const void*));
>+      break;
>+    default:
>+      s << "[Invalid CheckArgType:" << static_cast<int8_t>(**fmt) << "]";
>+      return false;
>+  }
>+  (*fmt)++;
>+  return true;
> }
> 
>-#if defined(__GNUC__)
>-void PrintError(const char* format, ...)
>-    __attribute__((__format__(__printf__, 1, 2)));
>-#endif
>-
>-void PrintError(const char* format, ...) {
>+RTC_NORETURN void FatalLog(const char* file,
>+                           int line,
>+                           const char* message,
>+                           const CheckArgType* fmt,
>+                           ...) {
>   va_list args;
>-  va_start(args, format);
>-  VPrintError(format, args);
>-  va_end(args);
>-}
>+  va_start(args, fmt);
>+
>+  std::ostringstream ss;  // no-presubmit-check TODO(webrtc:8982)
>+  ss << "\n\n#\n# Fatal error in: " << file << ", line " << line
>+     << "\n# last system error: " << LAST_SYSTEM_ERROR << "\n# Check failed: ";
>+
>+  if (*fmt == CheckArgType::kCheckOp) {
>+    // This log message was generated by RTC_CHECK_OP, so we have to complete
>+    // the error message using the operands that have been passed as the first
>+    // two arguments.
>+    fmt++;
> 
>-// TODO(ajm): This works on Mac (although the parsing fails) but I don't seem
>-// to get usable symbols on Linux. This is copied from V8. Chromium has a more
>-// advanced stace trace system; also more difficult to copy.
>-void DumpBacktrace() {
>-#if defined(__GLIBCXX__) && !defined(__UCLIBC__)
>-  void* trace[100];
>-  int size = backtrace(trace, sizeof(trace) / sizeof(*trace));
>-  char** symbols = backtrace_symbols(trace, size);
>-  PrintError("\n==== C stack trace ===============================\n\n");
>-  if (size == 0) {
>-    PrintError("(empty)\n");
>-  } else if (symbols == nullptr) {
>-    PrintError("(no symbols)\n");
>+    std::ostringstream s1, s2;  // no-presubmit-check TODO(webrtc:8982)
>+    if (ParseArg(&args, &fmt, s1) && ParseArg(&args, &fmt, s2))
>+      ss << message << " (" << s1.str() << " vs. " << s2.str() << ")\n# ";
>   } else {
>-    for (int i = 1; i < size; ++i) {
>-      char mangled[201];
>-      if (sscanf(symbols[i], "%*[^(]%*[(]%200[^)+]", mangled) == 1) {  // NOLINT
>-        PrintError("%2d: ", i);
>-        int status;
>-        size_t length;
>-        char* demangled =
>-            abi::__cxa_demangle(mangled, nullptr, &length, &status);
>-        PrintError("%s\n", demangled != nullptr ? demangled : mangled);
>-        free(demangled);
>-      } else {
>-        // If parsing failed, at least print the unparsed symbol.
>-        PrintError("%s\n", symbols[i]);
>-      }
>-    }
>+    ss << message << "\n# ";
>   }
>-  free(symbols);
>-#endif
>-}
> 
>-}  // namespace
>+  // Append all the user-supplied arguments to the message.
>+  while (ParseArg(&args, &fmt, ss))
>+    ;
> 
>-FatalMessage::FatalMessage(const char* file, int line) {
>-  Init(file, line);
>-}
>+  va_end(args);
> 
>-FatalMessage::FatalMessage(const char* file, int line, std::string* result) {
>-  Init(file, line);
>-  stream_ << "Check failed: " << *result << std::endl << "# ";
>-  delete result;
>-}
>+  std::string s = ss.str();
>+  const char* output = s.c_str();
>+
>+#if defined(WEBRTC_ANDROID)
>+  __android_log_print(ANDROID_LOG_ERROR, RTC_LOG_TAG_ANDROID, "%s\n", output);
>+#endif
> 
>-NO_RETURN FatalMessage::~FatalMessage() {
>   fflush(stdout);
>-  fflush(stderr);
>-  stream_ << std::endl << "#" << std::endl;
>-  PrintError("%s", stream_.str().c_str());
>-  DumpBacktrace();
>+  fprintf(stderr, "%s", output);
>   fflush(stderr);
>   abort();
> }
> 
>-void FatalMessage::Init(const char* file, int line) {
>-  stream_ << std::endl
>-          << std::endl
>-          << "#" << std::endl
>-          << "# Fatal error in " << file << ", line " << line << std::endl
>-          << "# last system error: " << LAST_SYSTEM_ERROR << std::endl
>-          << "# ";
>-}
>-
>-// MSVC doesn't like complex extern templates and DLLs.
>-#if !defined(COMPILER_MSVC)
>-// Explicit instantiations for commonly used comparisons.
>-template std::string* MakeCheckOpString<int, int>(
>-    const int&, const int&, const char* names);
>-template std::string* MakeCheckOpString<unsigned long, unsigned long>(
>-    const unsigned long&, const unsigned long&, const char* names);
>-template std::string* MakeCheckOpString<unsigned long, unsigned int>(
>-    const unsigned long&, const unsigned int&, const char* names);
>-template std::string* MakeCheckOpString<unsigned int, unsigned long>(
>-    const unsigned int&, const unsigned long&, const char* names);
>-template std::string* MakeCheckOpString<std::string, std::string>(
>-    const std::string&, const std::string&, const char* name);
>-#endif
>-
>+}  // namespace webrtc_checks_impl
> }  // namespace rtc
> 
> // Function to call from the C version of the RTC_CHECK and RTC_DCHECK macros.
>-NO_RETURN void rtc_FatalMessage(const char* file, int line, const char* msg) {
>-  rtc::FatalMessage(file, line).stream() << msg;
>+RTC_NORETURN void rtc_FatalMessage(const char* file, int line,
>+                                   const char* msg) {
>+  static constexpr rtc::webrtc_checks_impl::CheckArgType t[] = {
>+      rtc::webrtc_checks_impl::CheckArgType::kEnd};
>+  FatalLog(file, line, msg, t);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/checks.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/checks.h
>index 4fb40624ae1..c7cf36e4b4d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/checks.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/checks.h
>@@ -11,8 +11,6 @@
> #ifndef RTC_BASE_CHECKS_H_
> #define RTC_BASE_CHECKS_H_
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>-
> // If you for some reson need to know if DCHECKs are on, test the value of
> // RTC_DCHECK_IS_ON. (Test its value, not if it's defined; it'll always be
> // defined, to either a true or a false value.)
>@@ -22,10 +20,19 @@
> #define RTC_DCHECK_IS_ON 0
> #endif
> 
>+// Annotate a function that will not return control flow to the caller.
>+#if defined(_MSC_VER)
>+#define RTC_NORETURN __declspec(noreturn)
>+#elif defined(__GNUC__)
>+#define RTC_NORETURN __attribute__ ((__noreturn__))
>+#else
>+#define RTC_NORETURN
>+#endif
>+
> #ifdef __cplusplus
> extern "C" {
> #endif
>-NO_RETURN void rtc_FatalMessage(const char* file, int line, const char* msg);
>+RTC_NORETURN void rtc_FatalMessage(const char* file, int line, const char* msg);
> #ifdef __cplusplus
> }  // extern "C"
> #endif
>@@ -33,10 +40,11 @@ NO_RETURN void rtc_FatalMessage(const char* file, int line, const char* msg);
> #ifdef __cplusplus
> // C++ version.
> 
>-#include <sstream>
>+#include <sstream>  // no-presubmit-check TODO(webrtc:8982)
> #include <string>
> 
> #include "rtc_base/numerics/safe_compare.h"
>+#include "rtc_base/system/inline.h"
> 
> // The macros here print a message to stderr and abort under various
> // conditions. All will accept additional stream messages. For example:
>@@ -79,21 +87,209 @@ NO_RETURN void rtc_FatalMessage(const char* file, int line, const char* msg);
> // consolidation with system_wrappers/logging.h should happen first.
> 
> namespace rtc {
>+namespace webrtc_checks_impl {
>+enum class CheckArgType : int8_t {
>+  kEnd = 0,
>+  kInt,
>+  kLong,
>+  kLongLong,
>+  kUInt,
>+  kULong,
>+  kULongLong,
>+  kDouble,
>+  kLongDouble,
>+  kCharP,
>+  kStdString,
>+  kVoidP,
>+
>+  // kCheckOp doesn't represent an argument type. Instead, it is sent as the
>+  // first argument from RTC_CHECK_OP to make FatalLog use the next two
>+  // arguments to build the special CHECK_OP error message
>+  // (the "a == b (1 vs. 2)" bit).
>+  kCheckOp,
>+};
>+
>+RTC_NORETURN void FatalLog(const char* file,
>+                           int line,
>+                           const char* message,
>+                           const CheckArgType* fmt,
>+                           ...);
>+
>+// Wrapper for log arguments. Only ever make values of this type with the
>+// MakeVal() functions.
>+template <CheckArgType N, typename T>
>+struct Val {
>+  static constexpr CheckArgType Type() { return N; }
>+  T GetVal() const { return val; }
>+  T val;
>+};
>+
>+inline Val<CheckArgType::kInt, int> MakeVal(int x) {
>+  return {x};
>+}
>+inline Val<CheckArgType::kLong, long> MakeVal(long x) {
>+  return {x};
>+}
>+inline Val<CheckArgType::kLongLong, long long> MakeVal(long long x) {
>+  return {x};
>+}
>+inline Val<CheckArgType::kUInt, unsigned int> MakeVal(unsigned int x) {
>+  return {x};
>+}
>+inline Val<CheckArgType::kULong, unsigned long> MakeVal(unsigned long x) {
>+  return {x};
>+}
>+inline Val<CheckArgType::kULongLong, unsigned long long> MakeVal(
>+    unsigned long long x) {
>+  return {x};
>+}
>+
>+inline Val<CheckArgType::kDouble, double> MakeVal(double x) {
>+  return {x};
>+}
>+inline Val<CheckArgType::kLongDouble, long double> MakeVal(long double x) {
>+  return {x};
>+}
> 
>-// Helper macro which avoids evaluating the arguments to a stream if
>-// the condition doesn't hold.
>-#define RTC_LAZY_STREAM(stream, condition)                                    \
>-  !(condition) ? static_cast<void>(0) : rtc::FatalMessageVoidify() & (stream)
>+inline Val<CheckArgType::kCharP, const char*> MakeVal(const char* x) {
>+  return {x};
>+}
>+inline Val<CheckArgType::kStdString, const std::string*> MakeVal(
>+    const std::string& x) {
>+  return {&x};
>+}
>+
>+inline Val<CheckArgType::kVoidP, const void*> MakeVal(const void* x) {
>+  return {x};
>+}
>+
>+// Ephemeral type that represents the result of the logging << operator.
>+template <typename... Ts>
>+class LogStreamer;
>+
>+// Base case: Before the first << argument.
>+template <>
>+class LogStreamer<> final {
>+ public:
>+  template <
>+      typename U,
>+      typename std::enable_if<std::is_arithmetic<U>::value>::type* = nullptr>
>+  RTC_FORCE_INLINE LogStreamer<decltype(MakeVal(std::declval<U>()))> operator<<(
>+      U arg) const {
>+    return LogStreamer<decltype(MakeVal(std::declval<U>()))>(MakeVal(arg),
>+                                                             this);
>+  }
>+
>+  template <
>+      typename U,
>+      typename std::enable_if<!std::is_arithmetic<U>::value>::type* = nullptr>
>+  RTC_FORCE_INLINE LogStreamer<decltype(MakeVal(std::declval<U>()))> operator<<(
>+      const U& arg) const {
>+    return LogStreamer<decltype(MakeVal(std::declval<U>()))>(MakeVal(arg),
>+                                                             this);
>+  }
>+
>+  template <typename... Us>
>+  RTC_NORETURN RTC_FORCE_INLINE static void Call(const char* file,
>+                                                 const int line,
>+                                                 const char* message,
>+                                                 const Us&... args) {
>+    static constexpr CheckArgType t[] = {Us::Type()..., CheckArgType::kEnd};
>+    FatalLog(file, line, message, t, args.GetVal()...);
>+  }
>+
>+  template <typename... Us>
>+  RTC_NORETURN RTC_FORCE_INLINE static void CallCheckOp(const char* file,
>+                                                        const int line,
>+                                                        const char* message,
>+                                                        const Us&... args) {
>+    static constexpr CheckArgType t[] = {CheckArgType::kCheckOp, Us::Type()...,
>+                                         CheckArgType::kEnd};
>+    FatalLog(file, line, message, t, args.GetVal()...);
>+  }
>+};
>+
>+// Inductive case: We've already seen at least one << argument. The most recent
>+// one had type `T`, and the earlier ones had types `Ts`.
>+template <typename T, typename... Ts>
>+class LogStreamer<T, Ts...> final {
>+ public:
>+  RTC_FORCE_INLINE LogStreamer(T arg, const LogStreamer<Ts...>* prior)
>+      : arg_(arg), prior_(prior) {}
>+
>+  template <
>+      typename U,
>+      typename std::enable_if<std::is_arithmetic<U>::value>::type* = nullptr>
>+  RTC_FORCE_INLINE LogStreamer<decltype(MakeVal(std::declval<U>())), T, Ts...>
>+  operator<<(U arg) const {
>+    return LogStreamer<decltype(MakeVal(std::declval<U>())), T, Ts...>(
>+        MakeVal(arg), this);
>+  }
>+
>+  template <
>+      typename U,
>+      typename std::enable_if<!std::is_arithmetic<U>::value>::type* = nullptr>
>+  RTC_FORCE_INLINE LogStreamer<decltype(MakeVal(std::declval<U>())), T, Ts...>
>+  operator<<(const U& arg) const {
>+    return LogStreamer<decltype(MakeVal(std::declval<U>())), T, Ts...>(
>+        MakeVal(arg), this);
>+  }
>+
>+  template <typename... Us>
>+  RTC_NORETURN RTC_FORCE_INLINE void Call(const char* file,
>+                                          const int line,
>+                                          const char* message,
>+                                          const Us&... args) const {
>+    prior_->Call(file, line, message, arg_, args...);
>+  }
>+
>+  template <typename... Us>
>+  RTC_NORETURN RTC_FORCE_INLINE void CallCheckOp(const char* file,
>+                                                 const int line,
>+                                                 const char* message,
>+                                                 const Us&... args) const {
>+    prior_->CallCheckOp(file, line, message, arg_, args...);
>+  }
>+
>+ private:
>+  // The most recent argument.
>+  T arg_;
>+
>+  // Earlier arguments.
>+  const LogStreamer<Ts...>* prior_;
>+};
>+
>+template <bool isCheckOp>
>+class FatalLogCall final {
>+ public:
>+  FatalLogCall(const char* file, int line, const char* message)
>+      : file_(file), line_(line), message_(message) {}
>+
>+  // This can be any binary operator with precedence lower than <<.
>+  template <typename... Ts>
>+  RTC_NORETURN RTC_FORCE_INLINE void operator&(
>+      const LogStreamer<Ts...>& streamer) {
>+    isCheckOp ? streamer.CallCheckOp(file_, line_, message_)
>+              : streamer.Call(file_, line_, message_);
>+  }
>+
>+ private:
>+  const char* file_;
>+  int line_;
>+  const char* message_;
>+};
>+}  // namespace webrtc_checks_impl
> 
> // The actual stream used isn't important. We reference |ignored| in the code
> // but don't evaluate it; this is to avoid "unused variable" warnings (we do so
> // in a particularly convoluted way with an extra ?: because that appears to be
> // the simplest construct that keeps Visual Studio from complaining about
> // condition being unused).
>-#define RTC_EAT_STREAM_PARAMETERS(ignored) \
>-  (true ? true : ((void)(ignored), true))  \
>-      ? static_cast<void>(0)               \
>-      : rtc::FatalMessageVoidify() & rtc::FatalMessage("", 0).stream()
>+#define RTC_EAT_STREAM_PARAMETERS(ignored)                        \
>+  (true ? true : ((void)(ignored), true))                         \
>+      ? static_cast<void>(0)                                      \
>+      : rtc::webrtc_checks_impl::FatalLogCall<false>("", 0, "") & \
>+            rtc::webrtc_checks_impl::LogStreamer<>()
> 
> // Call RTC_EAT_STREAM_PARAMETERS with an argument that fails to compile if
> // values of the same types as |a| and |b| can't be compared with the given
>@@ -105,81 +301,21 @@ namespace rtc {
> // controlled by NDEBUG or anything else, so the check will be executed
> // regardless of compilation mode.
> //
>-// We make sure RTC_CHECK et al. always evaluates their arguments, as
>+// We make sure RTC_CHECK et al. always evaluates |condition|, as
> // doing RTC_CHECK(FunctionWithSideEffect()) is a common idiom.
>-#define RTC_CHECK(condition)                                      \
>-  RTC_LAZY_STREAM(rtc::FatalMessage(__FILE__, __LINE__).stream(), \
>-                  !(condition))                                   \
>-      << "Check failed: " #condition << std::endl << "# "
>+#define RTC_CHECK(condition)                                       \
>+  while (!(condition))                                             \
>+  rtc::webrtc_checks_impl::FatalLogCall<false>(__FILE__, __LINE__, \
>+                                               #condition) &       \
>+      rtc::webrtc_checks_impl::LogStreamer<>()
> 
> // Helper macro for binary operators.
> // Don't use this macro directly in your code, use RTC_CHECK_EQ et al below.
>-//
>-// TODO(akalin): Rewrite this so that constructs like if (...)
>-// RTC_CHECK_EQ(...) else { ... } work properly.
>-#define RTC_CHECK_OP(name, op, val1, val2)                                 \
>-  if (std::string* _result =                                               \
>-          rtc::Check##name##Impl((val1), (val2), #val1 " " #op " " #val2)) \
>-    rtc::FatalMessage(__FILE__, __LINE__, _result).stream()
>-
>-// Build the error message string.  This is separate from the "Impl"
>-// function template because it is not performance critical and so can
>-// be out of line, while the "Impl" code should be inline.  Caller
>-// takes ownership of the returned string.
>-template<class t1, class t2>
>-std::string* MakeCheckOpString(const t1& v1, const t2& v2, const char* names) {
>-  std::ostringstream ss;
>-  ss << names << " (" << v1 << " vs. " << v2 << ")";
>-  std::string* msg = new std::string(ss.str());
>-  return msg;
>-}
>-
>-// MSVC doesn't like complex extern templates and DLLs. WebKit build system neither.
>-#if 0 //!defined(COMPILER_MSVC)
>-// Commonly used instantiations of MakeCheckOpString<>. Explicitly instantiated
>-// in logging.cc.
>-extern template std::string* MakeCheckOpString<int, int>(
>-    const int&, const int&, const char* names);
>-extern template
>-std::string* MakeCheckOpString<unsigned long, unsigned long>(
>-    const unsigned long&, const unsigned long&, const char* names);
>-extern template
>-std::string* MakeCheckOpString<unsigned long, unsigned int>(
>-    const unsigned long&, const unsigned int&, const char* names);
>-extern template
>-std::string* MakeCheckOpString<unsigned int, unsigned long>(
>-    const unsigned int&, const unsigned long&, const char* names);
>-extern template
>-std::string* MakeCheckOpString<std::string, std::string>(
>-    const std::string&, const std::string&, const char* name);
>-#endif
>-
>-// Helper functions for RTC_CHECK_OP macro.
>-// The (int, int) specialization works around the issue that the compiler
>-// will not instantiate the template version of the function on values of
>-// unnamed enum type - see comment below.
>-#define DEFINE_RTC_CHECK_OP_IMPL(name)                                       \
>-  template <class t1, class t2>                                              \
>-  inline std::string* Check##name##Impl(const t1& v1, const t2& v2,          \
>-                                        const char* names) {                 \
>-    if (rtc::Safe##name(v1, v2))                                             \
>-      return nullptr;                                                        \
>-    else                                                                     \
>-      return rtc::MakeCheckOpString(v1, v2, names);                          \
>-  }                                                                          \
>-  inline std::string* Check##name##Impl(int v1, int v2, const char* names) { \
>-    if (rtc::Safe##name(v1, v2))                                             \
>-      return nullptr;                                                        \
>-    else                                                                     \
>-      return rtc::MakeCheckOpString(v1, v2, names);                          \
>-  }
>-DEFINE_RTC_CHECK_OP_IMPL(Eq)
>-DEFINE_RTC_CHECK_OP_IMPL(Ne)
>-DEFINE_RTC_CHECK_OP_IMPL(Le)
>-DEFINE_RTC_CHECK_OP_IMPL(Lt)
>-DEFINE_RTC_CHECK_OP_IMPL(Ge)
>-DEFINE_RTC_CHECK_OP_IMPL(Gt)
>-#undef DEFINE_RTC_CHECK_OP_IMPL
>+#define RTC_CHECK_OP(name, op, val1, val2)                               \
>+  while (!rtc::Safe##name((val1), (val2)))                               \
>+  rtc::webrtc_checks_impl::FatalLogCall<true>(__FILE__, __LINE__,        \
>+                                              #val1 " " #op " " #val2) & \
>+      rtc::webrtc_checks_impl::LogStreamer<>() << (val1) << (val2)
> 
> #define RTC_CHECK_EQ(val1, val2) RTC_CHECK_OP(Eq, ==, val1, val2)
> #define RTC_CHECK_NE(val1, val2) RTC_CHECK_OP(Ne, !=, val1, val2)
>@@ -209,39 +345,14 @@ DEFINE_RTC_CHECK_OP_IMPL(Gt)
> #define RTC_DCHECK_GT(v1, v2) RTC_EAT_STREAM_PARAMETERS_OP(Gt, v1, v2)
> #endif
> 
>-// This is identical to LogMessageVoidify but in name.
>-class FatalMessageVoidify {
>- public:
>-  FatalMessageVoidify() { }
>-  // This has to be an operator with a precedence lower than << but
>-  // higher than ?:
>-  void operator&(std::ostream&) { }
>-};
>-
> #define RTC_UNREACHABLE_CODE_HIT false
> #define RTC_NOTREACHED() RTC_DCHECK(RTC_UNREACHABLE_CODE_HIT)
> 
> // TODO(bugs.webrtc.org/8454): Add an RTC_ prefix or rename differently.
>-#define RTC_FATAL() rtc::FatalMessage(__FILE__, __LINE__).stream()
>-// TODO(ajm): Consider adding RTC_NOTIMPLEMENTED macro when
>-// base/logging.h and system_wrappers/logging.h are consolidated such that we
>-// can match the Chromium behavior.
>-
>-// Like a stripped-down LogMessage from logging.h, except that it aborts.
>-class FatalMessage {
>- public:
>-  FatalMessage(const char* file, int line);
>-  // Used for RTC_CHECK_EQ(), etc. Takes ownership of the given string.
>-  FatalMessage(const char* file, int line, std::string* result);
>-  NO_RETURN ~FatalMessage();
>-
>-  std::ostream& stream() { return stream_; }
>-
>- private:
>-  void Init(const char* file, int line);
>-
>-  std::ostringstream stream_;
>-};
>+#define RTC_FATAL()                                                    \
>+  rtc::webrtc_checks_impl::FatalLogCall<false>(__FILE__, __LINE__, \
>+                                               "FATAL()") &        \
>+      rtc::webrtc_checks_impl::LogStreamer<>()
> 
> // Performs the integer division a/b and returns the result. CHECKs that the
> // remainder is zero.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/compile_assert_c.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/compile_assert_c.h
>index c83d31495d0..db2e4a81169 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/compile_assert_c.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/compile_assert_c.h
>@@ -16,6 +16,10 @@
> // Example:
> //   RTC_COMPILE_ASSERT(sizeof(foo) < 128);
> // Note: In C++, use static_assert instead!
>-#define RTC_COMPILE_ASSERT(expression) switch (0) {case 0: case expression:;}
>+#define RTC_COMPILE_ASSERT(expression) \
>+  switch (0) {                         \
>+    case 0:                            \
>+    case expression:;                  \
>+  }
> 
> #endif  // RTC_BASE_COMPILE_ASSERT_C_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/constructormagic.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/constructormagic.h
>index 8a953aac563..646a058ce3e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/constructormagic.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/constructormagic.h
>@@ -12,8 +12,7 @@
> #define RTC_BASE_CONSTRUCTORMAGIC_H_
> 
> // Put this in the declarations for a class to be unassignable.
>-#define RTC_DISALLOW_ASSIGN(TypeName) \
>-  void operator=(const TypeName&) = delete
>+#define RTC_DISALLOW_ASSIGN(TypeName) void operator=(const TypeName&) = delete
> 
> // A macro to disallow the copy constructor and operator= functions. This should
> // be used in the declarations for a class.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/copyonwritebuffer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/copyonwritebuffer.cc
>index 579dd460ba2..6c48d52f171 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/copyonwritebuffer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/copyonwritebuffer.cc
>@@ -17,12 +17,13 @@ CopyOnWriteBuffer::CopyOnWriteBuffer() {
> }
> 
> CopyOnWriteBuffer::CopyOnWriteBuffer(const CopyOnWriteBuffer& buf)
>-    : buffer_(buf.buffer_) {
>-}
>+    : buffer_(buf.buffer_) {}
> 
> CopyOnWriteBuffer::CopyOnWriteBuffer(CopyOnWriteBuffer&& buf)
>-    : buffer_(std::move(buf.buffer_)) {
>-}
>+    : buffer_(std::move(buf.buffer_)) {}
>+
>+CopyOnWriteBuffer::CopyOnWriteBuffer(const std::string& s)
>+    : CopyOnWriteBuffer(s.data(), s.length()) {}
> 
> CopyOnWriteBuffer::CopyOnWriteBuffer(size_t size)
>     : buffer_(size > 0 ? new RefCountedObject<Buffer>(size) : nullptr) {
>@@ -31,8 +32,8 @@ CopyOnWriteBuffer::CopyOnWriteBuffer(size_t size)
> 
> CopyOnWriteBuffer::CopyOnWriteBuffer(size_t size, size_t capacity)
>     : buffer_(size > 0 || capacity > 0
>-          ? new RefCountedObject<Buffer>(size, capacity)
>-          : nullptr) {
>+                  ? new RefCountedObject<Buffer>(size, capacity)
>+                  : nullptr) {
>   RTC_DCHECK(IsConsistent());
> }
> 
>@@ -43,8 +44,8 @@ bool CopyOnWriteBuffer::operator==(const CopyOnWriteBuffer& buf) const {
>   RTC_DCHECK(IsConsistent());
>   RTC_DCHECK(buf.IsConsistent());
>   return buffer_.get() == buf.buffer_.get() ||
>-      (buffer_.get() && buf.buffer_.get() &&
>-      *buffer_.get() == *buf.buffer_.get());
>+         (buffer_.get() && buf.buffer_.get() &&
>+          *buffer_.get() == *buf.buffer_.get());
> }
> 
> void CopyOnWriteBuffer::SetSize(size_t size) {
>@@ -59,10 +60,9 @@ void CopyOnWriteBuffer::SetSize(size_t size) {
> 
>   // Clone data if referenced.
>   if (!buffer_->HasOneRef()) {
>-    buffer_ = new RefCountedObject<Buffer>(
>-        buffer_->data(),
>-        std::min(buffer_->size(), size),
>-        std::max(buffer_->capacity(), size));
>+    buffer_ = new RefCountedObject<Buffer>(buffer_->data(),
>+                                           std::min(buffer_->size(), size),
>+                                           std::max(buffer_->capacity(), size));
>   }
>   buffer_->SetSize(size);
>   RTC_DCHECK(IsConsistent());
>@@ -103,10 +103,8 @@ void CopyOnWriteBuffer::CloneDataIfReferenced(size_t new_capacity) {
>   }
> 
>   buffer_ = new RefCountedObject<Buffer>(buffer_->data(), buffer_->size(),
>-      new_capacity);
>+                                         new_capacity);
>   RTC_DCHECK(IsConsistent());
> }
> 
>-
>-
> }  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/copyonwritebuffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/copyonwritebuffer.h
>index c4bba874ad7..0514e2fe661 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/copyonwritebuffer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/copyonwritebuffer.h
>@@ -31,6 +31,9 @@ class CopyOnWriteBuffer {
>   // Move contents from an existing buffer.
>   CopyOnWriteBuffer(CopyOnWriteBuffer&& buf);
> 
>+  // Construct a buffer from a string, convenient for unittests.
>+  CopyOnWriteBuffer(const std::string& s);
>+
>   // Construct a buffer with the specified number of uninitialized bytes.
>   explicit CopyOnWriteBuffer(size_t size);
>   CopyOnWriteBuffer(size_t size, size_t capacity);
>@@ -185,8 +188,8 @@ class CopyOnWriteBuffer {
>       return;
>     }
> 
>-    CloneDataIfReferenced(std::max(buffer_->capacity(),
>-        buffer_->size() + size));
>+    CloneDataIfReferenced(
>+        std::max(buffer_->capacity(), buffer_->size() + size));
>     buffer_->AppendData(data, size);
>     RTC_DCHECK(IsConsistent());
>   }
>@@ -229,9 +232,7 @@ class CopyOnWriteBuffer {
>   void CloneDataIfReferenced(size_t new_capacity);
> 
>   // Pre- and postcondition of all methods.
>-  bool IsConsistent() const {
>-    return (!buffer_ || buffer_->capacity() > 0);
>-  }
>+  bool IsConsistent() const { return (!buffer_ || buffer_->capacity() > 0); }
> 
>   // buffer_ is either null, or points to an rtc::Buffer with capacity > 0.
>   scoped_refptr<RefCountedObject<Buffer>> buffer_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/cpu_time.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/cpu_time.cc
>index 020cc72ec8a..f25b50662ea 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/cpu_time.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/cpu_time.cc
>@@ -15,12 +15,12 @@
> #if defined(WEBRTC_LINUX)
> #include <time.h>
> #elif defined(WEBRTC_MAC)
>-#include <sys/resource.h>
>-#include <sys/types.h>
>-#include <sys/times.h>
>+#include <mach/mach_init.h>
> #include <mach/thread_act.h>
> #include <mach/thread_info.h>
>-#include <mach/mach_port.h>
>+#include <sys/resource.h>
>+#include <sys/times.h>
>+#include <sys/types.h>
> #include <unistd.h>
> #elif defined(WEBRTC_WIN)
> #include <windows.h>
>@@ -83,10 +83,8 @@ int64_t GetThreadCpuTimeNanos() {
> #elif defined(WEBRTC_MAC)
>   thread_basic_info_data_t info;
>   mach_msg_type_number_t count = THREAD_BASIC_INFO_COUNT;
>-  auto port = mach_thread_self();
>-  auto result = thread_info(port, THREAD_BASIC_INFO, (thread_info_t)&info, &count);
>-  mach_port_deallocate(mach_task_self(), port);
>-  if (result == KERN_SUCCESS) {
>+  if (thread_info(mach_thread_self(), THREAD_BASIC_INFO, (thread_info_t)&info,
>+                  &count) == KERN_SUCCESS) {
>     return info.user_time.seconds * kNumNanosecsPerSec +
>            info.user_time.microseconds * kNumNanosecsPerMicrosec;
>   } else {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/cpu_time_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/cpu_time_unittest.cc
>index ba973788abd..f4730690dcf 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/cpu_time_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/cpu_time_unittest.cc
>@@ -86,10 +86,10 @@ TEST(CpuTimeTest, MAYBE_TEST(TwoThreads)) {
>             kAllowedErrorMillisecs * kNumNanosecsPerMillisec);
>   // Total process time is at least twice working threads' CPU time.
>   // Therefore process and thread times are correctly related.
>-  EXPECT_GE(
>-      process_duration_nanos,
>-      kWorkingThreads * (kProcessingTimeMillisecs - kAllowedErrorMillisecs)
>-      * kNumNanosecsPerMillisec);
>+  EXPECT_GE(process_duration_nanos,
>+            kWorkingThreads *
>+                (kProcessingTimeMillisecs - kAllowedErrorMillisecs) *
>+                kNumNanosecsPerMillisec);
> }
> 
> TEST(CpuTimeTest, MAYBE_TEST(Sleeping)) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/crc32.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/crc32.cc
>index c214f38cc73..c70f5fef71f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/crc32.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/crc32.cc
>@@ -49,4 +49,3 @@ uint32_t UpdateCrc32(uint32_t start, const void* buf, size_t len) {
> }
> 
> }  // namespace rtc
>-
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/crc32.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/crc32.h
>index a0ce4323e71..79c765f64fa 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/crc32.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/crc32.h
>@@ -11,9 +11,10 @@
> #ifndef RTC_BASE_CRC32_H_
> #define RTC_BASE_CRC32_H_
> 
>-#include <string>
>+#include <stddef.h>
>+#include <stdint.h>
> 
>-#include "rtc_base/basictypes.h"
>+#include <string>
> 
> namespace rtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/crc32_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/crc32_unittest.cc
>index 576b424bd09..752bc9fff59 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/crc32_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/crc32_unittest.cc
>@@ -18,7 +18,8 @@ namespace rtc {
> TEST(Crc32Test, TestBasic) {
>   EXPECT_EQ(0U, ComputeCrc32(""));
>   EXPECT_EQ(0x352441C2U, ComputeCrc32("abc"));
>-  EXPECT_EQ(0x171A3F5FU,
>+  EXPECT_EQ(
>+      0x171A3F5FU,
>       ComputeCrc32("abcdbcdecdefdefgefghfghighijhijkijkljklmklmnlmnomnopnopq"));
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/criticalsection.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/criticalsection.cc
>index d92b478b9b2..d8a5b48c051 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/criticalsection.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/criticalsection.cc
>@@ -10,8 +10,10 @@
> 
> #include "rtc_base/criticalsection.h"
> 
>+#include "rtc_base/atomicops.h"
> #include "rtc_base/checks.h"
>-#include "rtc_base/platform_thread.h"
>+#include "rtc_base/platform_thread_types.h"
>+#include "rtc_base/system/unused.h"
> 
> // TODO(tommi): Split this file up to per-platform implementation files.
> 
>@@ -21,24 +23,24 @@ CriticalSection::CriticalSection() {
> #if defined(WEBRTC_WIN)
>   InitializeCriticalSection(&crit_);
> #elif defined(WEBRTC_POSIX)
>-# if defined(WEBRTC_MAC) && !USE_NATIVE_MUTEX_ON_MAC
>+#if defined(WEBRTC_MAC) && !USE_NATIVE_MUTEX_ON_MAC
>   lock_queue_ = 0;
>   owning_thread_ = 0;
>   recursion_ = 0;
>   semaphore_ = dispatch_semaphore_create(0);
>-# else
>+#else
>   pthread_mutexattr_t mutex_attribute;
>   pthread_mutexattr_init(&mutex_attribute);
>   pthread_mutexattr_settype(&mutex_attribute, PTHREAD_MUTEX_RECURSIVE);
>   pthread_mutex_init(&mutex_, &mutex_attribute);
>   pthread_mutexattr_destroy(&mutex_attribute);
>-# endif
>+#endif
>   CS_DEBUG_CODE(thread_ = 0);
>   CS_DEBUG_CODE(recursion_count_ = 0);
>   RTC_UNUSED(thread_);
>   RTC_UNUSED(recursion_count_);
> #else
>-# error Unsupported platform.
>+#error Unsupported platform.
> #endif
> }
> 
>@@ -46,13 +48,13 @@ CriticalSection::~CriticalSection() {
> #if defined(WEBRTC_WIN)
>   DeleteCriticalSection(&crit_);
> #elif defined(WEBRTC_POSIX)
>-# if defined(WEBRTC_MAC) && !USE_NATIVE_MUTEX_ON_MAC
>+#if defined(WEBRTC_MAC) && !USE_NATIVE_MUTEX_ON_MAC
>   dispatch_release(semaphore_);
>-# else
>+#else
>   pthread_mutex_destroy(&mutex_);
>-# endif
>+#endif
> #else
>-# error Unsupported platform.
>+#error Unsupported platform.
> #endif
> }
> 
>@@ -60,7 +62,7 @@ void CriticalSection::Enter() const RTC_EXCLUSIVE_LOCK_FUNCTION() {
> #if defined(WEBRTC_WIN)
>   EnterCriticalSection(&crit_);
> #elif defined(WEBRTC_POSIX)
>-# if defined(WEBRTC_MAC) && !USE_NATIVE_MUTEX_ON_MAC
>+#if defined(WEBRTC_MAC) && !USE_NATIVE_MUTEX_ON_MAC
>   int spin = 3000;
>   PlatformThreadRef self = CurrentThreadRef();
>   bool have_lock = false;
>@@ -97,11 +99,11 @@ void CriticalSection::Enter() const RTC_EXCLUSIVE_LOCK_FUNCTION() {
>   owning_thread_ = self;
>   ++recursion_;
> 
>-# else
>+#else
>   pthread_mutex_lock(&mutex_);
>-# endif
>+#endif
> 
>-# if CS_DEBUG_CHECKS
>+#if CS_DEBUG_CHECKS
>   if (!recursion_count_) {
>     RTC_DCHECK(!thread_);
>     thread_ = CurrentThreadRef();
>@@ -109,9 +111,9 @@ void CriticalSection::Enter() const RTC_EXCLUSIVE_LOCK_FUNCTION() {
>     RTC_DCHECK(CurrentThreadIsOwner());
>   }
>   ++recursion_count_;
>-# endif
>+#endif
> #else
>-# error Unsupported platform.
>+#error Unsupported platform.
> #endif
> }
> 
>@@ -119,7 +121,7 @@ bool CriticalSection::TryEnter() const RTC_EXCLUSIVE_TRYLOCK_FUNCTION(true) {
> #if defined(WEBRTC_WIN)
>   return TryEnterCriticalSection(&crit_) != FALSE;
> #elif defined(WEBRTC_POSIX)
>-# if defined(WEBRTC_MAC) && !USE_NATIVE_MUTEX_ON_MAC
>+#if defined(WEBRTC_MAC) && !USE_NATIVE_MUTEX_ON_MAC
>   if (!IsThreadRefEqual(owning_thread_, CurrentThreadRef())) {
>     if (AtomicOps::CompareAndSwap(&lock_queue_, 0, 1) != 0)
>       return false;
>@@ -129,11 +131,11 @@ bool CriticalSection::TryEnter() const RTC_EXCLUSIVE_TRYLOCK_FUNCTION(true) {
>     AtomicOps::Increment(&lock_queue_);
>   }
>   ++recursion_;
>-# else
>+#else
>   if (pthread_mutex_trylock(&mutex_) != 0)
>     return false;
>-# endif
>-# if CS_DEBUG_CHECKS
>+#endif
>+#if CS_DEBUG_CHECKS
>   if (!recursion_count_) {
>     RTC_DCHECK(!thread_);
>     thread_ = CurrentThreadRef();
>@@ -141,10 +143,10 @@ bool CriticalSection::TryEnter() const RTC_EXCLUSIVE_TRYLOCK_FUNCTION(true) {
>     RTC_DCHECK(CurrentThreadIsOwner());
>   }
>   ++recursion_count_;
>-# endif
>+#endif
>   return true;
> #else
>-# error Unsupported platform.
>+#error Unsupported platform.
> #endif
> }
> 
>@@ -153,13 +155,13 @@ void CriticalSection::Leave() const RTC_UNLOCK_FUNCTION() {
> #if defined(WEBRTC_WIN)
>   LeaveCriticalSection(&crit_);
> #elif defined(WEBRTC_POSIX)
>-# if CS_DEBUG_CHECKS
>+#if CS_DEBUG_CHECKS
>   --recursion_count_;
>   RTC_DCHECK(recursion_count_ >= 0);
>   if (!recursion_count_)
>     thread_ = 0;
>-# endif
>-# if defined(WEBRTC_MAC) && !USE_NATIVE_MUTEX_ON_MAC
>+#endif
>+#if defined(WEBRTC_MAC) && !USE_NATIVE_MUTEX_ON_MAC
>   RTC_DCHECK(IsThreadRefEqual(owning_thread_, CurrentThreadRef()));
>   RTC_DCHECK_GE(recursion_, 0);
>   --recursion_;
>@@ -168,11 +170,11 @@ void CriticalSection::Leave() const RTC_UNLOCK_FUNCTION() {
> 
>   if (AtomicOps::Decrement(&lock_queue_) > 0 && !recursion_)
>     dispatch_semaphore_signal(semaphore_);
>-# else
>+#else
>   pthread_mutex_unlock(&mutex_);
>-# endif
>+#endif
> #else
>-# error Unsupported platform.
>+#error Unsupported platform.
> #endif
> }
> 
>@@ -185,18 +187,22 @@ bool CriticalSection::CurrentThreadIsOwner() const {
>   return crit_.OwningThread ==
>          reinterpret_cast<HANDLE>(static_cast<size_t>(GetCurrentThreadId()));
> #elif defined(WEBRTC_POSIX)
>-# if CS_DEBUG_CHECKS
>+#if CS_DEBUG_CHECKS
>   return IsThreadRefEqual(thread_, CurrentThreadRef());
>-# else
>+#else
>   return true;
>-# endif  // CS_DEBUG_CHECKS
>+#endif  // CS_DEBUG_CHECKS
> #else
>-# error Unsupported platform.
>+#error Unsupported platform.
> #endif
> }
> 
>-CritScope::CritScope(const CriticalSection* cs) : cs_(cs) { cs_->Enter(); }
>-CritScope::~CritScope() { cs_->Leave(); }
>+CritScope::CritScope(const CriticalSection* cs) : cs_(cs) {
>+  cs_->Enter();
>+}
>+CritScope::~CritScope() {
>+  cs_->Leave();
>+}
> 
> TryCritScope::TryCritScope(const CriticalSection* cs)
>     : cs_(cs), locked_(cs->TryEnter()) {
>@@ -240,8 +246,7 @@ GlobalLock::GlobalLock() {
>   lock_acquired = 0;
> }
> 
>-GlobalLockScope::GlobalLockScope(GlobalLockPod* lock)
>-    : lock_(lock) {
>+GlobalLockScope::GlobalLockScope(GlobalLockPod* lock) : lock_(lock) {
>   lock_->Lock();
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/criticalsection.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/criticalsection.h
>index 1ef9634efe4..f25e7d0c10a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/criticalsection.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/criticalsection.h
>@@ -11,21 +11,22 @@
> #ifndef RTC_BASE_CRITICALSECTION_H_
> #define RTC_BASE_CRITICALSECTION_H_
> 
>-#include "rtc_base/atomicops.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/platform_thread_types.h"
> #include "rtc_base/thread_annotations.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> #if defined(WEBRTC_WIN)
>+// clang-format off
>+// clang formating would change include order.
>+
> // Include winsock2.h before including <windows.h> to maintain consistency with
>-// win32.h.  We can't include win32.h directly here since it pulls in
>-// headers such as basictypes.h which causes problems in Chromium where webrtc
>-// exists as two separate projects, webrtc and libjingle.
>+// win32.h. To include win32.h directly, it must be broken out into its own
>+// build target.
> #include <winsock2.h>
> #include <windows.h>
> #include <sal.h>  // must come after windows headers.
>+// clang-format on
> #endif  // defined(WEBRTC_WIN)
> 
> #if defined(WEBRTC_POSIX)
>@@ -68,7 +69,7 @@ class RTC_LOCKABLE CriticalSection {
> #if defined(WEBRTC_WIN)
>   mutable CRITICAL_SECTION crit_;
> #elif defined(WEBRTC_POSIX)
>-# if defined(WEBRTC_MAC) && !USE_NATIVE_MUTEX_ON_MAC
>+#if defined(WEBRTC_MAC) && !USE_NATIVE_MUTEX_ON_MAC
>   // Number of times the lock has been locked + number of threads waiting.
>   // TODO(tommi): We could use this number and subtract the recursion count
>   // to find places where we have multiple threads contending on the same lock.
>@@ -80,13 +81,13 @@ class RTC_LOCKABLE CriticalSection {
>   mutable dispatch_semaphore_t semaphore_;
>   // The thread that currently holds the lock. Required to handle recursion.
>   mutable PlatformThreadRef owning_thread_;
>-# else
>+#else
>   mutable pthread_mutex_t mutex_;
>-# endif
>+#endif
>   mutable PlatformThreadRef thread_;  // Only used by RTC_DCHECKs.
>   mutable int recursion_count_;       // Only used by RTC_DCHECKs.
> #else  // !defined(WEBRTC_WIN) && !defined(WEBRTC_POSIX)
>-# error Unsupported platform.
>+#error Unsupported platform.
> #endif
> };
> 
>@@ -115,9 +116,9 @@ class TryCritScope {
> #if defined(WEBRTC_WIN)
>   _Check_return_ bool locked() const;
> #elif defined(WEBRTC_POSIX)
>-  bool locked() const __attribute__ ((__warn_unused_result__));
>+  bool locked() const __attribute__((__warn_unused_result__));
> #else  // !defined(WEBRTC_WIN) && !defined(WEBRTC_POSIX)
>-# error Unsupported platform.
>+#error Unsupported platform.
> #endif
>  private:
>   const CriticalSection* const cs_;
>@@ -154,6 +155,6 @@ class RTC_SCOPED_LOCKABLE GlobalLockScope {
>   RTC_DISALLOW_COPY_AND_ASSIGN(GlobalLockScope);
> };
> 
>-} // namespace rtc
>+}  // namespace rtc
> 
>-#endif // RTC_BASE_CRITICALSECTION_H_
>+#endif  // RTC_BASE_CRITICALSECTION_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/criticalsection_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/criticalsection_unittest.cc
>index f5d6957fcf6..db4f9e7452f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/criticalsection_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/criticalsection_unittest.cc
>@@ -13,6 +13,7 @@
> #include <vector>
> 
> #include "rtc_base/arraysize.h"
>+#include "rtc_base/atomicops.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/event.h"
>@@ -62,9 +63,8 @@ class CompareAndSwapVerifier {
>     }
>   }
> 
>-  void Finalize() {
>-    EXPECT_EQ(1, zero_count_);
>-  }
>+  void Finalize() { EXPECT_EQ(1, zero_count_); }
>+
>  private:
>   int zero_count_;
> };
>@@ -85,18 +85,14 @@ class RunnerBase : public MessageHandler {
>     return done_event_.Wait(kLongTime);
>   }
> 
>-  void SetExpectedThreadCount(int count) {
>-    threads_active_ = count;
>-  }
>+  void SetExpectedThreadCount(int count) { threads_active_ = count; }
> 
>   int shared_value() const { return shared_value_; }
> 
>  protected:
>   // Derived classes must override OnMessage, and call BeforeStart and AfterEnd
>   // at the beginning and the end of OnMessage respectively.
>-  void BeforeStart() {
>-    ASSERT_TRUE(start_event_.Wait(kLongTime));
>-  }
>+  void BeforeStart() { ASSERT_TRUE(start_event_.Wait(kLongTime)); }
> 
>   // Returns true if all threads have finished.
>   bool AfterEnd() {
>@@ -167,7 +163,7 @@ class AtomicOpRunner : public RunnerBase {
>       values.push_back(Op::AtomicOp(&shared_value_));
>     }
> 
>-    { // Add them all to the set.
>+    {  // Add them all to the set.
>       CritScope cs(&all_values_crit_);
>       verifier_.Verify(values);
>     }
>@@ -254,8 +250,8 @@ TEST(AtomicOpsTest, Increment) {
> 
> TEST(AtomicOpsTest, Decrement) {
>   // Create and start lots of threads.
>-  AtomicOpRunner<DecrementOp, UniqueValueVerifier> runner(
>-      kOperationsToRun * kNumThreads);
>+  AtomicOpRunner<DecrementOp, UniqueValueVerifier> runner(kOperationsToRun *
>+                                                          kNumThreads);
>   std::vector<std::unique_ptr<Thread>> threads;
>   StartThreads(&threads, &runner);
>   runner.SetExpectedThreadCount(kNumThreads);
>@@ -304,8 +300,10 @@ TEST(CriticalSectionTest, Basic) {
> class PerfTestData {
>  public:
>   PerfTestData(int expected_count, Event* event)
>-      : cache_line_barrier_1_(), cache_line_barrier_2_(),
>-        expected_count_(expected_count), event_(event) {
>+      : cache_line_barrier_1_(),
>+        cache_line_barrier_2_(),
>+        expected_count_(expected_count),
>+        event_(event) {
>     cache_line_barrier_1_[0]++;  // Avoid 'is not used'.
>     cache_line_barrier_2_[0]++;  // Avoid 'is not used'.
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/cryptstring.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/cryptstring.cc
>index 421710c7c8d..2c7c0c72af7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/cryptstring.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/cryptstring.cc
>@@ -34,15 +34,12 @@ void EmptyCryptStringImpl::CopyRawTo(std::vector<unsigned char>* dest) const {
>   dest->clear();
> }
> 
>-CryptString::CryptString() : impl_(new EmptyCryptStringImpl()) {
>-}
>+CryptString::CryptString() : impl_(new EmptyCryptStringImpl()) {}
> 
> CryptString::CryptString(const CryptString& other)
>-    : impl_(other.impl_->Copy()) {
>-}
>+    : impl_(other.impl_->Copy()) {}
> 
>-CryptString::CryptString(const CryptStringImpl& impl) : impl_(impl.Copy()) {
>-}
>+CryptString::CryptString(const CryptStringImpl& impl) : impl_(impl.Copy()) {}
> 
> CryptString::~CryptString() = default;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/cryptstring.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/cryptstring.h
>index 03b38c462f9..d0e36cb1692 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/cryptstring.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/cryptstring.h
>@@ -20,17 +20,17 @@
> namespace rtc {
> 
> class CryptStringImpl {
>-public:
>+ public:
>   virtual ~CryptStringImpl() {}
>   virtual size_t GetLength() const = 0;
>-  virtual void CopyTo(char * dest, bool nullterminate) const = 0;
>+  virtual void CopyTo(char* dest, bool nullterminate) const = 0;
>   virtual std::string UrlEncode() const = 0;
>-  virtual CryptStringImpl * Copy() const = 0;
>-  virtual void CopyRawTo(std::vector<unsigned char> * dest) const = 0;
>+  virtual CryptStringImpl* Copy() const = 0;
>+  virtual void CopyRawTo(std::vector<unsigned char>* dest) const = 0;
> };
> 
> class EmptyCryptStringImpl : public CryptStringImpl {
>-public:
>+ public:
>   ~EmptyCryptStringImpl() override {}
>   size_t GetLength() const override;
>   void CopyTo(char* dest, bool nullterminate) const override;
>@@ -43,11 +43,13 @@ class CryptString {
>  public:
>   CryptString();
>   size_t GetLength() const { return impl_->GetLength(); }
>-  void CopyTo(char * dest, bool nullterminate) const { impl_->CopyTo(dest, nullterminate); }
>+  void CopyTo(char* dest, bool nullterminate) const {
>+    impl_->CopyTo(dest, nullterminate);
>+  }
>   CryptString(const CryptString& other);
>   explicit CryptString(const CryptStringImpl& impl);
>   ~CryptString();
>-  CryptString & operator=(const CryptString & other) {
>+  CryptString& operator=(const CryptString& other) {
>     if (this != &other) {
>       impl_.reset(other.impl_->Copy());
>     }
>@@ -55,7 +57,7 @@ class CryptString {
>   }
>   void Clear() { impl_.reset(new EmptyCryptStringImpl()); }
>   std::string UrlEncode() const { return impl_->UrlEncode(); }
>-  void CopyRawTo(std::vector<unsigned char> * dest) const {
>+  void CopyRawTo(std::vector<unsigned char>* dest) const {
>     return impl_->CopyRawTo(dest);
>   }
> 
>@@ -63,89 +65,6 @@ class CryptString {
>   std::unique_ptr<const CryptStringImpl> impl_;
> };
> 
>-
>-// Used for constructing strings where a password is involved and we
>-// need to ensure that we zero memory afterwards
>-class FormatCryptString {
>-public:
>-  FormatCryptString() {
>-    storage_ = new char[32];
>-    capacity_ = 32;
>-    length_ = 0;
>-    storage_[0] = 0;
>-  }
>-
>-  void Append(const std::string & text) {
>-    Append(text.data(), text.length());
>-  }
>-
>-  void Append(const char * data, size_t length) {
>-    EnsureStorage(length_ + length + 1);
>-    memcpy(storage_ + length_, data, length);
>-    length_ += length;
>-    storage_[length_] = '\0';
>-  }
>-
>-  void Append(const CryptString * password) {
>-    size_t len = password->GetLength();
>-    EnsureStorage(length_ + len + 1);
>-    password->CopyTo(storage_ + length_, true);
>-    length_ += len;
>-  }
>-
>-  size_t GetLength() {
>-    return length_;
>-  }
>-
>-  const char * GetData() {
>-    return storage_;
>-  }
>-
>-
>-  // Ensures storage of at least n bytes
>-  void EnsureStorage(size_t n) {
>-    if (capacity_ >= n) {
>-      return;
>-    }
>-
>-    size_t old_capacity = capacity_;
>-    char * old_storage = storage_;
>-
>-    for (;;) {
>-      capacity_ *= 2;
>-      if (capacity_ >= n)
>-        break;
>-    }
>-
>-    storage_ = new char[capacity_];
>-
>-    if (old_capacity) {
>-      memcpy(storage_, old_storage, length_);
>-
>-      // zero memory in a way that an optimizer won't optimize it out
>-      old_storage[0] = 0;
>-      for (size_t i = 1; i < old_capacity; i++) {
>-        old_storage[i] = old_storage[i - 1];
>-      }
>-      delete[] old_storage;
>-    }
>-  }
>-
>-  ~FormatCryptString() {
>-    if (capacity_) {
>-      storage_[0] = 0;
>-      for (size_t i = 1; i < capacity_; i++) {
>-        storage_[i] = storage_[i - 1];
>-      }
>-    }
>-    delete[] storage_;
>-  }
>-private:
>-  char * storage_;
>-  size_t capacity_;
>-  size_t length_;
>-};
>-
> class InsecureCryptStringImpl : public CryptStringImpl {
>  public:
>   std::string& password() { return password_; }
>@@ -162,6 +81,6 @@ class InsecureCryptStringImpl : public CryptStringImpl {
>   std::string password_;
> };
> 
>-}
>+}  // namespace rtc
> 
> #endif  // RTC_BASE_CRYPTSTRING_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ratelimiter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/data_rate_limiter.cc
>similarity index 82%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ratelimiter.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/data_rate_limiter.cc
>index 842538d8047..7288257d769 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ratelimiter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/data_rate_limiter.cc
>@@ -8,16 +8,16 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "rtc_base/ratelimiter.h"
>+#include "rtc_base/data_rate_limiter.h"
> 
> namespace rtc {
> 
>-bool RateLimiter::CanUse(size_t desired, double time) {
>+bool DataRateLimiter::CanUse(size_t desired, double time) {
>   return ((time > period_end_ && desired <= max_per_period_) ||
>           (used_in_period_ + desired) <= max_per_period_);
> }
> 
>-void RateLimiter::Use(size_t used, double time) {
>+void DataRateLimiter::Use(size_t used, double time) {
>   if (time > period_end_) {
>     period_start_ = time;
>     period_end_ = time + period_length_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ratelimiter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/data_rate_limiter.h
>similarity index 79%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ratelimiter.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/data_rate_limiter.h
>index 85706d3dde3..d290816a2c5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ratelimiter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/data_rate_limiter.h
>@@ -8,8 +8,8 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#ifndef RTC_BASE_RATELIMITER_H_
>-#define RTC_BASE_RATELIMITER_H_
>+#ifndef RTC_BASE_DATA_RATE_LIMITER_H_
>+#define RTC_BASE_DATA_RATE_LIMITER_H_
> 
> #include <stddef.h>
> 
>@@ -21,17 +21,16 @@ namespace rtc {
> // It's implemented like a diet plan: You have so many calories per
> // day.  If you hit the limit, you can't eat any more until the next
> // day.
>-class RateLimiter {
>+class DataRateLimiter {
>  public:
>   // For example, 100kb per second.
>-  RateLimiter(size_t max, double period)
>+  DataRateLimiter(size_t max, double period)
>       : max_per_period_(max),
>         period_length_(period),
>         used_in_period_(0),
>         period_start_(0.0),
>-        period_end_(period) {
>-  }
>-  virtual ~RateLimiter() {}
>+        period_end_(period) {}
>+  virtual ~DataRateLimiter() {}
> 
>   // Returns true if if the desired quantity is available in the
>   // current period (< (max - used)).  Once the given time passes the
>@@ -41,13 +40,9 @@ class RateLimiter {
>   // period, a new period is started.
>   void Use(size_t used, double time);
> 
>-  size_t used_in_period() const {
>-    return used_in_period_;
>-  }
>+  size_t used_in_period() const { return used_in_period_; }
> 
>-  size_t max_per_period() const {
>-    return max_per_period_;
>-  }
>+  size_t max_per_period() const { return max_per_period_; }
> 
>  private:
>   size_t max_per_period_;
>@@ -56,7 +51,6 @@ class RateLimiter {
>   double period_start_;
>   double period_end_;
> };
>-
> }  // namespace rtc
> 
>-#endif  // RTC_BASE_RATELIMITER_H_
>+#endif  // RTC_BASE_DATA_RATE_LIMITER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ratelimiter_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/data_rate_limiter_unittest.cc
>similarity index 94%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ratelimiter_unittest.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/data_rate_limiter_unittest.cc
>index f0931e4466a..8c410fe1e56 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ratelimiter_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/data_rate_limiter_unittest.cc
>@@ -8,14 +8,14 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "rtc_base/ratelimiter.h"
>+#include "rtc_base/data_rate_limiter.h"
> #include "rtc_base/gunit.h"
> 
> namespace rtc {
> 
> TEST(RateLimiterTest, TestCanUse) {
>   // Diet: Can eat 2,000 calories per day.
>-  RateLimiter limiter = RateLimiter(2000, 1.0);
>+  DataRateLimiter limiter = DataRateLimiter(2000, 1.0);
> 
>   double monday = 1.0;
>   double tuesday = 2.0;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/deprecation.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/deprecation.h
>index af776d53ff1..f285ab04bb4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/deprecation.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/deprecation.h
>@@ -37,7 +37,7 @@
> // (https://bugs.chromium.org/p/webrtc/issues/detail?id=5368).
> #define RTC_DEPRECATED __declspec(deprecated)
> #elif defined(__GNUC__)
>-#define RTC_DEPRECATED __attribute__ ((__deprecated__))
>+#define RTC_DEPRECATED __attribute__((__deprecated__))
> #else
> #define RTC_DEPRECATED
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/dscp.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/dscp.h
>index bdce466044c..3c39ca66da7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/dscp.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/dscp.h
>@@ -17,29 +17,29 @@ namespace rtc {
> enum DiffServCodePoint {
>   DSCP_NO_CHANGE = -1,
>   DSCP_DEFAULT = 0,  // Same as DSCP_CS0
>-  DSCP_CS0  = 0,   // The default
>-  DSCP_CS1  = 8,   // Bulk/background traffic
>+  DSCP_CS0 = 0,      // The default
>+  DSCP_CS1 = 8,      // Bulk/background traffic
>   DSCP_AF11 = 10,
>   DSCP_AF12 = 12,
>   DSCP_AF13 = 14,
>-  DSCP_CS2  = 16,
>+  DSCP_CS2 = 16,
>   DSCP_AF21 = 18,
>   DSCP_AF22 = 20,
>   DSCP_AF23 = 22,
>-  DSCP_CS3  = 24,
>+  DSCP_CS3 = 24,
>   DSCP_AF31 = 26,
>   DSCP_AF32 = 28,
>   DSCP_AF33 = 30,
>-  DSCP_CS4  = 32,
>+  DSCP_CS4 = 32,
>   DSCP_AF41 = 34,  // Video
>   DSCP_AF42 = 36,  // Video
>   DSCP_AF43 = 38,  // Video
>-  DSCP_CS5  = 40,  // Video
>-  DSCP_EF   = 46,  // Voice
>-  DSCP_CS6  = 48,  // Voice
>-  DSCP_CS7  = 56,  // Control messages
>+  DSCP_CS5 = 40,   // Video
>+  DSCP_EF = 46,    // Voice
>+  DSCP_CS6 = 48,   // Voice
>+  DSCP_CS7 = 56,   // Control messages
> };
> 
> }  // namespace rtc
> 
>- #endif  // RTC_BASE_DSCP_H_
>+#endif  // RTC_BASE_DSCP_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/event.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/event.cc
>index ff4faadd677..6c9639b4fec 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/event.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/event.cc
>@@ -52,11 +52,33 @@ bool Event::Wait(int milliseconds) {
> 
> #elif defined(WEBRTC_POSIX)
> 
>+// On MacOS, clock_gettime is available from version 10.12, and on
>+// iOS, from version 10.0. So we can't use it yet.
>+#if defined(WEBRTC_MAC) || defined(WEBRTC_IOS)
>+#define USE_CLOCK_GETTIME 0
>+#define USE_PTHREAD_COND_TIMEDWAIT_MONOTONIC_NP 0
>+// On Android, pthread_condattr_setclock is available from version 21. By
>+// default, we target a new enough version for 64-bit platforms but not for
>+// 32-bit platforms. For older versions, use
>+// pthread_cond_timedwait_monotonic_np.
>+#elif defined(WEBRTC_ANDROID) && (__ANDROID_API__ < 21)
>+#define USE_CLOCK_GETTIME 1
>+#define USE_PTHREAD_COND_TIMEDWAIT_MONOTONIC_NP 1
>+#else
>+#define USE_CLOCK_GETTIME 1
>+#define USE_PTHREAD_COND_TIMEDWAIT_MONOTONIC_NP 0
>+#endif
>+
> Event::Event(bool manual_reset, bool initially_signaled)
>-    : is_manual_reset_(manual_reset),
>-      event_status_(initially_signaled) {
>+    : is_manual_reset_(manual_reset), event_status_(initially_signaled) {
>   RTC_CHECK(pthread_mutex_init(&event_mutex_, nullptr) == 0);
>-  RTC_CHECK(pthread_cond_init(&event_cond_, nullptr) == 0);
>+  pthread_condattr_t cond_attr;
>+  RTC_CHECK(pthread_condattr_init(&cond_attr) == 0);
>+#if USE_CLOCK_GETTIME && !USE_PTHREAD_COND_TIMEDWAIT_MONOTONIC_NP
>+  RTC_CHECK(pthread_condattr_setclock(&cond_attr, CLOCK_MONOTONIC) == 0);
>+#endif
>+  RTC_CHECK(pthread_cond_init(&event_cond_, &cond_attr) == 0);
>+  pthread_condattr_destroy(&cond_attr);
> }
> 
> Event::~Event() {
>@@ -82,14 +104,17 @@ bool Event::Wait(int milliseconds) {
> 
>   struct timespec ts;
>   if (milliseconds != kForever) {
>-    // Converting from seconds and microseconds (1e-6) plus
>-    // milliseconds (1e-3) to seconds and nanoseconds (1e-9).
>-
>+#if USE_CLOCK_GETTIME
>+    clock_gettime(CLOCK_MONOTONIC, &ts);
>+#else
>     struct timeval tv;
>     gettimeofday(&tv, nullptr);
>+    ts.tv_sec = tv.tv_sec;
>+    ts.tv_nsec = tv.tv_usec * 1000;
>+#endif
> 
>-    ts.tv_sec = tv.tv_sec + (milliseconds / 1000);
>-    ts.tv_nsec = tv.tv_usec * 1000 + (milliseconds % 1000) * 1000000;
>+    ts.tv_sec += (milliseconds / 1000);
>+    ts.tv_nsec += (milliseconds % 1000) * 1000000;
> 
>     // Handle overflow.
>     if (ts.tv_nsec >= 1000000000) {
>@@ -101,7 +126,12 @@ bool Event::Wait(int milliseconds) {
>   pthread_mutex_lock(&event_mutex_);
>   if (milliseconds != kForever) {
>     while (!event_status_ && error == 0) {
>+#if USE_PTHREAD_COND_TIMEDWAIT_MONOTONIC_NP
>+      error =
>+          pthread_cond_timedwait_monotonic_np(&event_cond_, &event_mutex_, &ts);
>+#else
>       error = pthread_cond_timedwait(&event_cond_, &event_mutex_, &ts);
>+#endif
>     }
>   } else {
>     while (!event_status_ && error == 0)
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/event.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/event.h
>index ce2666fb3ff..7e61c4c8a68 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/event.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/event.h
>@@ -49,6 +49,20 @@ class Event {
>   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(Event);
> };
> 
>+// This class is provided for compatibility with Chromium.
>+// The rtc::Event implementation is overriden inside of Chromium for the
>+// purposes of detecting when threads are blocked that shouldn't be as well as
>+// to use the more accurate event implementation that's there than is provided
>+// by default on some platforms (e.g. Windows).
>+// When building with standalone WebRTC, this class is a noop.
>+// For further information, please see the ScopedAllowBaseSyncPrimitives class
>+// in Chromium.
>+class ScopedAllowBaseSyncPrimitives {
>+ public:
>+  ScopedAllowBaseSyncPrimitives() {}
>+  ~ScopedAllowBaseSyncPrimitives() {}
>+};
>+
> }  // namespace rtc
> 
> #endif  // RTC_BASE_EVENT_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/event_tracer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/event_tracer.cc
>index 9cbda972e3b..31f4271d31e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/event_tracer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/event_tracer.cc
>@@ -14,6 +14,7 @@
> #include <string>
> #include <vector>
> 
>+#include "rtc_base/atomicops.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/event.h"
>@@ -62,15 +63,8 @@ void EventTracer::AddTraceEvent(char phase,
>                                 const unsigned long long* arg_values,
>                                 unsigned char flags) {
>   if (g_add_trace_event_ptr) {
>-    g_add_trace_event_ptr(phase,
>-                          category_enabled,
>-                          name,
>-                          id,
>-                          num_args,
>-                          arg_names,
>-                          arg_types,
>-                          arg_values,
>-                          flags);
>+    g_add_trace_event_ptr(phase, category_enabled, name, id, num_args,
>+                          arg_names, arg_types, arg_values, flags);
>   }
> }
> 
>@@ -127,8 +121,8 @@ class EventLogger final {
>         {name, category_enabled, phase, args, timestamp, 1, thread_id});
>   }
> 
>-// The TraceEvent format is documented here:
>-// https://docs.google.com/document/d/1CvAClvFfyA5R-PhYUmn5OOQtYMH4h6I0nSsKchNAySU/preview
>+  // The TraceEvent format is documented here:
>+  // https://docs.google.com/document/d/1CvAClvFfyA5R-PhYUmn5OOQtYMH4h6I0nSsKchNAySU/preview
>   void Log() {
>     RTC_DCHECK(output_file_);
>     static const int kLoggingIntervalMs = 100;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/event_tracer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/event_tracer.h
>index fc7ad046d45..4bbda579bc7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/event_tracer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/event_tracer.h
>@@ -45,27 +45,24 @@ typedef void (*AddTraceEventPtr)(char phase,
> //
> // This method must be called before any WebRTC methods. Functions
> // provided should be thread-safe.
>-void SetupEventTracer(
>-    GetCategoryEnabledPtr get_category_enabled_ptr,
>-    AddTraceEventPtr add_trace_event_ptr);
>+void SetupEventTracer(GetCategoryEnabledPtr get_category_enabled_ptr,
>+                      AddTraceEventPtr add_trace_event_ptr);
> 
> // This class defines interface for the event tracing system to call
> // internally. Do not call these methods directly.
> class EventTracer {
>  public:
>-  static const unsigned char* GetCategoryEnabled(
>-      const char* name);
>+  static const unsigned char* GetCategoryEnabled(const char* name);
> 
>-  static void AddTraceEvent(
>-      char phase,
>-      const unsigned char* category_enabled,
>-      const char* name,
>-      unsigned long long id,
>-      int num_args,
>-      const char** arg_names,
>-      const unsigned char* arg_types,
>-      const unsigned long long* arg_values,
>-      unsigned char flags);
>+  static void AddTraceEvent(char phase,
>+                            const unsigned char* category_enabled,
>+                            const char* name,
>+                            unsigned long long id,
>+                            int num_args,
>+                            const char** arg_names,
>+                            const unsigned char* arg_types,
>+                            const unsigned long long* arg_values,
>+                            unsigned char flags);
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/event_tracer_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/event_tracer_unittest.cc
>index 25b300b271d..27248e75474 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/event_tracer_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/event_tracer_unittest.cc
>@@ -17,16 +17,11 @@ namespace {
> 
> class TestStatistics {
>  public:
>-  TestStatistics() : events_logged_(0) {
>-  }
>+  TestStatistics() : events_logged_(0) {}
> 
>-  void Reset() {
>-    events_logged_ = 0;
>-  }
>+  void Reset() { events_logged_ = 0; }
> 
>-  void Increment() {
>-    ++events_logged_;
>-  }
>+  void Increment() { ++events_logged_; }
> 
>   int Count() const { return events_logged_; }
> 
>@@ -62,18 +57,14 @@ static void AddTraceEventHandler(char phase,
> namespace webrtc {
> 
> TEST(EventTracerTest, EventTracerDisabled) {
>-  {
>-    TRACE_EVENT0("test", "EventTracerDisabled");
>-  }
>+  { TRACE_EVENT0("test", "EventTracerDisabled"); }
>   EXPECT_FALSE(TestStatistics::Get()->Count());
>   TestStatistics::Get()->Reset();
> }
> 
> TEST(EventTracerTest, ScopedTraceEvent) {
>   SetupEventTracer(&GetCategoryEnabledHandler, &AddTraceEventHandler);
>-  {
>-    TRACE_EVENT0("test", "ScopedTraceEvent");
>-  }
>+  { TRACE_EVENT0("test", "ScopedTraceEvent"); }
>   EXPECT_EQ(2, TestStatistics::Get()->Count());
>   TestStatistics::Get()->Reset();
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/event_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/event_unittest.cc
>index 5bb6f77b805..050619e8495 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/event_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/event_unittest.cc
>@@ -41,7 +41,7 @@ TEST(EventTest, AutoReset) {
> }
> 
> class SignalerThread {
>-public:
>+ public:
>   SignalerThread() : thread_(&ThreadFn, this, "EventPerf") {}
>   void Start(Event* writer, Event* reader) {
>     writer_ = writer;
>@@ -52,9 +52,9 @@ public:
>     stop_event_.Set();
>     thread_.Stop();
>   }
>-  static void ThreadFn(void *param) {
>+  static void ThreadFn(void* param) {
>     auto* me = static_cast<SignalerThread*>(param);
>-    while(!me->stop_event_.Wait(0)) {
>+    while (!me->stop_event_.Wait(0)) {
>       me->writer_->Set();
>       me->reader_->Wait(Event::kForever);
>     }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/BUILD.gn
>index 305ad24a589..a95e2587897 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/BUILD.gn
>@@ -15,7 +15,72 @@ rtc_static_library("alr_experiment") {
>   ]
>   deps = [
>     "../:rtc_base_approved",
>-    "../../api:optional",
>     "../../system_wrappers:field_trial_api",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
>+
>+rtc_static_library("field_trial_parser") {
>+  sources = [
>+    "field_trial_parser.cc",
>+    "field_trial_parser.h",
>+    "field_trial_units.cc",
>+    "field_trial_units.h",
>+  ]
>+  deps = [
>+    "../:rtc_base_approved",
>+    "../../api/units:data_rate",
>+    "../../api/units:data_size",
>+    "../../api/units:time_delta",
>+    "../../system_wrappers:field_trial_api",
>+    "//third_party/abseil-cpp/absl/types:optional",
>+  ]
>+}
>+
>+rtc_static_library("congestion_controller_experiment") {
>+  sources = [
>+    "congestion_controller_experiment.cc",
>+    "congestion_controller_experiment.h",
>+  ]
>+  deps = [
>+    "../:rtc_base_approved",
>+    "../../system_wrappers:field_trial_api",
>+    "//third_party/abseil-cpp/absl/types:optional",
>+  ]
>+}
>+
>+rtc_static_library("quality_scaling_experiment") {
>+  sources = [
>+    "quality_scaling_experiment.cc",
>+    "quality_scaling_experiment.h",
>+  ]
>+  deps = [
>+    "../:rtc_base_approved",
>+    "../..:webrtc_common",
>+    "../../api/video_codecs:video_codecs_api",
>+    "../../system_wrappers:field_trial_api",
>+    "//third_party/abseil-cpp/absl/types:optional",
>+  ]
>+}
>+
>+if (rtc_include_tests) {
>+  rtc_source_set("experiments_unittests") {
>+    testonly = true
>+
>+    sources = [
>+      "congestion_controller_experiment_unittest.cc",
>+      "field_trial_parser_unittest.cc",
>+      "field_trial_units_unittest.cc",
>+      "quality_scaling_experiment_unittest.cc",
>+    ]
>+    deps = [
>+      ":congestion_controller_experiment",
>+      ":field_trial_parser",
>+      ":quality_scaling_experiment",
>+      "../:rtc_base_tests_main",
>+      "../:rtc_base_tests_utils",
>+      "../../system_wrappers:field_trial_api",
>+      "../../test:field_trial",
>+    ]
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/alr_experiment.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/alr_experiment.cc
>index c69caed787f..dff5acecc8b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/alr_experiment.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/alr_experiment.cc
>@@ -31,9 +31,9 @@ bool AlrExperimentSettings::MaxOneFieldTrialEnabled() {
>              .empty();
> }
> 
>-rtc::Optional<AlrExperimentSettings>
>+absl::optional<AlrExperimentSettings>
> AlrExperimentSettings::CreateFromFieldTrial(const char* experiment_name) {
>-  rtc::Optional<AlrExperimentSettings> ret;
>+  absl::optional<AlrExperimentSettings> ret;
>   std::string group_name = field_trial::FindFullName(experiment_name);
> 
>   const std::string kIgnoredSuffix = "_Dogfood";
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/alr_experiment.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/alr_experiment.h
>index a9c483d6326..4d9fd001304 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/alr_experiment.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/alr_experiment.h
>@@ -11,7 +11,7 @@
> #ifndef RTC_BASE_EXPERIMENTS_ALR_EXPERIMENT_H_
> #define RTC_BASE_EXPERIMENTS_ALR_EXPERIMENT_H_
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> 
> namespace webrtc {
> struct AlrExperimentSettings {
>@@ -28,7 +28,7 @@ struct AlrExperimentSettings {
> 
>   static const char kScreenshareProbingBweExperimentName[];
>   static const char kStrictPacingAndProbingExperimentName[];
>-  static rtc::Optional<AlrExperimentSettings> CreateFromFieldTrial(
>+  static absl::optional<AlrExperimentSettings> CreateFromFieldTrial(
>       const char* experiment_name);
>   static bool MaxOneFieldTrialEnabled();
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/congestion_controller_experiment.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/congestion_controller_experiment.cc
>new file mode 100644
>index 00000000000..2b7d7766628
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/congestion_controller_experiment.cc
>@@ -0,0 +1,33 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#include "rtc_base/experiments/congestion_controller_experiment.h"
>+
>+#include <string>
>+
>+#include "system_wrappers/include/field_trial.h"
>+
>+namespace webrtc {
>+namespace {
>+
>+const char kControllerExperiment[] = "WebRTC-BweCongestionController";
>+}  // namespace
>+
>+bool CongestionControllerExperiment::BbrControllerEnabled() {
>+  std::string trial_string =
>+      webrtc::field_trial::FindFullName(kControllerExperiment);
>+  return trial_string.find("Enabled,BBR") == 0;
>+}
>+
>+bool CongestionControllerExperiment::InjectedControllerEnabled() {
>+  std::string trial_string =
>+      webrtc::field_trial::FindFullName(kControllerExperiment);
>+  return trial_string.find("Enabled,Injected") == 0;
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/congestion_controller_experiment.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/congestion_controller_experiment.h
>new file mode 100644
>index 00000000000..1f2d8ae70cd
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/congestion_controller_experiment.h
>@@ -0,0 +1,22 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#ifndef RTC_BASE_EXPERIMENTS_CONGESTION_CONTROLLER_EXPERIMENT_H_
>+#define RTC_BASE_EXPERIMENTS_CONGESTION_CONTROLLER_EXPERIMENT_H_
>+
>+namespace webrtc {
>+class CongestionControllerExperiment {
>+ public:
>+  static bool BbrControllerEnabled();
>+  static bool InjectedControllerEnabled();
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // RTC_BASE_EXPERIMENTS_CONGESTION_CONTROLLER_EXPERIMENT_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/congestion_controller_experiment_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/congestion_controller_experiment_unittest.cc
>new file mode 100644
>index 00000000000..7f929e64f9c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/congestion_controller_experiment_unittest.cc
>@@ -0,0 +1,26 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "rtc_base/experiments/congestion_controller_experiment.h"
>+#include "rtc_base/gunit.h"
>+#include "test/field_trial.h"
>+
>+namespace webrtc {
>+TEST(CongestionControllerExperimentTest, BbrDisabledByDefault) {
>+  webrtc::test::ScopedFieldTrials field_trials("");
>+  EXPECT_FALSE(CongestionControllerExperiment::BbrControllerEnabled());
>+}
>+
>+TEST(CongestionControllerExperimentTest, BbrEnabledByFieldTrial) {
>+  webrtc::test::ScopedFieldTrials field_trials(
>+      "WebRTC-BweCongestionController/Enabled,BBR/");
>+  EXPECT_TRUE(CongestionControllerExperiment::BbrControllerEnabled());
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/field_trial_parser.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/field_trial_parser.cc
>new file mode 100644
>index 00000000000..55299adbb61
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/field_trial_parser.cc
>@@ -0,0 +1,165 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#include "rtc_base/experiments/field_trial_parser.h"
>+
>+#include <algorithm>
>+#include <map>
>+#include <type_traits>
>+#include <utility>
>+
>+#include "rtc_base/logging.h"
>+
>+namespace webrtc {
>+namespace {
>+
>+int FindOrEnd(std::string str, size_t start, char delimiter) {
>+  size_t pos = str.find(delimiter, start);
>+  pos = (pos == std::string::npos) ? str.length() : pos;
>+  return static_cast<int>(pos);
>+}
>+}  // namespace
>+
>+FieldTrialParameterInterface::FieldTrialParameterInterface(std::string key)
>+    : key_(key) {}
>+FieldTrialParameterInterface::~FieldTrialParameterInterface() = default;
>+std::string FieldTrialParameterInterface::Key() const {
>+  return key_;
>+}
>+
>+void ParseFieldTrial(
>+    std::initializer_list<FieldTrialParameterInterface*> fields,
>+    std::string trial_string) {
>+  std::map<std::string, FieldTrialParameterInterface*> field_map;
>+  for (FieldTrialParameterInterface* field : fields) {
>+    field_map[field->Key()] = field;
>+  }
>+  size_t i = 0;
>+  while (i < trial_string.length()) {
>+    int val_end = FindOrEnd(trial_string, i, ',');
>+    int colon_pos = FindOrEnd(trial_string, i, ':');
>+    int key_end = std::min(val_end, colon_pos);
>+    int val_begin = key_end + 1;
>+    std::string key = trial_string.substr(i, key_end - i);
>+    absl::optional<std::string> opt_value;
>+    if (val_end >= val_begin)
>+      opt_value = trial_string.substr(val_begin, val_end - val_begin);
>+    i = val_end + 1;
>+    auto field = field_map.find(key);
>+    if (field != field_map.end()) {
>+      if (!field->second->Parse(std::move(opt_value))) {
>+        RTC_LOG(LS_WARNING) << "Failed to read field with key: '" << key
>+                            << "' in trial: \"" << trial_string << "\"";
>+      }
>+    } else {
>+      RTC_LOG(LS_INFO) << "No field with key: '" << key
>+                       << "' (found in trial: \"" << trial_string << "\")";
>+    }
>+  }
>+}
>+
>+template <>
>+absl::optional<bool> ParseTypedParameter<bool>(std::string str) {
>+  if (str == "true" || str == "1") {
>+    return true;
>+  } else if (str == "false" || str == "0") {
>+    return false;
>+  }
>+  return absl::nullopt;
>+}
>+
>+template <>
>+absl::optional<double> ParseTypedParameter<double>(std::string str) {
>+  double value;
>+  if (sscanf(str.c_str(), "%lf", &value) == 1) {
>+    return value;
>+  } else {
>+    return absl::nullopt;
>+  }
>+}
>+
>+template <>
>+absl::optional<int> ParseTypedParameter<int>(std::string str) {
>+  int value;
>+  if (sscanf(str.c_str(), "%i", &value) == 1) {
>+    return value;
>+  } else {
>+    return absl::nullopt;
>+  }
>+}
>+
>+template <>
>+absl::optional<std::string> ParseTypedParameter<std::string>(std::string str) {
>+  return std::move(str);
>+}
>+
>+FieldTrialFlag::FieldTrialFlag(std::string key) : FieldTrialFlag(key, false) {}
>+
>+FieldTrialFlag::FieldTrialFlag(std::string key, bool default_value)
>+    : FieldTrialParameterInterface(key), value_(default_value) {}
>+
>+bool FieldTrialFlag::Get() const {
>+  return value_;
>+}
>+
>+bool FieldTrialFlag::Parse(absl::optional<std::string> str_value) {
>+  // Only set the flag if there is no argument provided.
>+  if (str_value) {
>+    absl::optional<bool> opt_value = ParseTypedParameter<bool>(*str_value);
>+    if (!opt_value)
>+      return false;
>+    value_ = *opt_value;
>+  } else {
>+    value_ = true;
>+  }
>+  return true;
>+}
>+
>+AbstractFieldTrialEnum::AbstractFieldTrialEnum(
>+    std::string key,
>+    int default_value,
>+    std::map<std::string, int> mapping)
>+    : FieldTrialParameterInterface(key),
>+      value_(default_value),
>+      enum_mapping_(mapping) {
>+  for (auto& key_val : enum_mapping_)
>+    valid_values_.insert(key_val.second);
>+}
>+AbstractFieldTrialEnum::AbstractFieldTrialEnum(const AbstractFieldTrialEnum&) =
>+    default;
>+AbstractFieldTrialEnum::~AbstractFieldTrialEnum() = default;
>+
>+bool AbstractFieldTrialEnum::Parse(absl::optional<std::string> str_value) {
>+  if (str_value) {
>+    auto it = enum_mapping_.find(*str_value);
>+    if (it != enum_mapping_.end()) {
>+      value_ = it->second;
>+      return true;
>+    }
>+    absl::optional<int> value = ParseTypedParameter<int>(*str_value);
>+    if (value.has_value() &&
>+        (valid_values_.find(*value) != valid_values_.end())) {
>+      value_ = *value;
>+      return true;
>+    }
>+  }
>+  return false;
>+}
>+
>+template class FieldTrialParameter<bool>;
>+template class FieldTrialParameter<double>;
>+template class FieldTrialParameter<int>;
>+template class FieldTrialParameter<std::string>;
>+
>+template class FieldTrialOptional<double>;
>+template class FieldTrialOptional<int>;
>+template class FieldTrialOptional<bool>;
>+template class FieldTrialOptional<std::string>;
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/field_trial_parser.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/field_trial_parser.h
>new file mode 100644
>index 00000000000..a385ccfc167
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/field_trial_parser.h
>@@ -0,0 +1,189 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#ifndef RTC_BASE_EXPERIMENTS_FIELD_TRIAL_PARSER_H_
>+#define RTC_BASE_EXPERIMENTS_FIELD_TRIAL_PARSER_H_
>+
>+#include <stdint.h>
>+#include <initializer_list>
>+#include <map>
>+#include <set>
>+#include <string>
>+#include "absl/types/optional.h"
>+
>+// Field trial parser functionality. Provides funcitonality to parse field trial
>+// argument strings in key:value format. Each parameter is described using
>+// key:value, parameters are separated with a ,. Values can't include the comma
>+// character, since there's no quote facility. For most types, white space is
>+// ignored. Parameters are declared with a given type for which an
>+// implementation of ParseTypedParameter should be provided. The
>+// ParseTypedParameter implementation is given whatever is between the : and the
>+// ,. FieldTrialOptional will use nullopt if the key is provided without :.
>+
>+// Example string: "my_optional,my_int:3,my_string:hello"
>+
>+// For further description of usage and behavior, see the examples in the unit
>+// tests.
>+
>+namespace webrtc {
>+class FieldTrialParameterInterface {
>+ public:
>+  virtual ~FieldTrialParameterInterface();
>+
>+ protected:
>+  explicit FieldTrialParameterInterface(std::string key);
>+  friend void ParseFieldTrial(
>+      std::initializer_list<FieldTrialParameterInterface*> fields,
>+      std::string raw_string);
>+  virtual bool Parse(absl::optional<std::string> str_value) = 0;
>+  std::string Key() const;
>+
>+ private:
>+  const std::string key_;
>+};
>+
>+// ParseFieldTrial function parses the given string and fills the given fields
>+// with extrated values if available.
>+void ParseFieldTrial(
>+    std::initializer_list<FieldTrialParameterInterface*> fields,
>+    std::string raw_string);
>+
>+// Specialize this in code file for custom types. Should return absl::nullopt if
>+// the given string cannot be properly parsed.
>+template <typename T>
>+absl::optional<T> ParseTypedParameter(std::string);
>+
>+// This class uses the ParseTypedParameter function to implement a parameter
>+// implementation with an enforced default value.
>+template <typename T>
>+class FieldTrialParameter : public FieldTrialParameterInterface {
>+ public:
>+  FieldTrialParameter(std::string key, T default_value)
>+      : FieldTrialParameterInterface(key), value_(default_value) {}
>+  T Get() const { return value_; }
>+  operator T() const { return Get(); }
>+
>+ protected:
>+  bool Parse(absl::optional<std::string> str_value) override {
>+    if (str_value) {
>+      absl::optional<T> value = ParseTypedParameter<T>(*str_value);
>+      if (value.has_value()) {
>+        value_ = value.value();
>+        return true;
>+      }
>+    }
>+    return false;
>+  }
>+
>+ private:
>+  T value_;
>+};
>+
>+class AbstractFieldTrialEnum : public FieldTrialParameterInterface {
>+ public:
>+  AbstractFieldTrialEnum(std::string key,
>+                         int default_value,
>+                         std::map<std::string, int> mapping);
>+  ~AbstractFieldTrialEnum() override;
>+  AbstractFieldTrialEnum(const AbstractFieldTrialEnum&);
>+
>+ protected:
>+  bool Parse(absl::optional<std::string> str_value) override;
>+
>+ protected:
>+  int value_;
>+  std::map<std::string, int> enum_mapping_;
>+  std::set<int> valid_values_;
>+};
>+
>+// The FieldTrialEnum class can be used to quickly define a parser for a
>+// specific enum. It handles values provided as integers and as strings if a
>+// mapping is provided.
>+template <typename T>
>+class FieldTrialEnum : public AbstractFieldTrialEnum {
>+ public:
>+  FieldTrialEnum(std::string key,
>+                 T default_value,
>+                 std::map<std::string, T> mapping)
>+      : AbstractFieldTrialEnum(key,
>+                               static_cast<int>(default_value),
>+                               ToIntMap(mapping)) {}
>+  T Get() const { return static_cast<T>(value_); }
>+  operator T() const { return Get(); }
>+
>+ private:
>+  static std::map<std::string, int> ToIntMap(std::map<std::string, T> mapping) {
>+    std::map<std::string, int> res;
>+    for (const auto& it : mapping)
>+      res[it.first] = static_cast<int>(it.second);
>+    return res;
>+  }
>+};
>+
>+// This class uses the ParseTypedParameter function to implement an optional
>+// parameter implementation that can default to absl::nullopt.
>+template <typename T>
>+class FieldTrialOptional : public FieldTrialParameterInterface {
>+ public:
>+  explicit FieldTrialOptional(std::string key)
>+      : FieldTrialParameterInterface(key) {}
>+  FieldTrialOptional(std::string key, absl::optional<T> default_value)
>+      : FieldTrialParameterInterface(key), value_(default_value) {}
>+  absl::optional<T> Get() const { return value_; }
>+
>+ protected:
>+  bool Parse(absl::optional<std::string> str_value) override {
>+    if (str_value) {
>+      absl::optional<T> value = ParseTypedParameter<T>(*str_value);
>+      if (!value.has_value())
>+        return false;
>+      value_ = value.value();
>+    } else {
>+      value_ = absl::nullopt;
>+    }
>+    return true;
>+  }
>+
>+ private:
>+  absl::optional<T> value_;
>+};
>+
>+// Equivalent to a FieldTrialParameter<bool> in the case that both key and value
>+// are present. If key is missing, evaluates to false. If key is present, but no
>+// explicit value is provided, the flag evaluates to true.
>+class FieldTrialFlag : public FieldTrialParameterInterface {
>+ public:
>+  explicit FieldTrialFlag(std::string key);
>+  FieldTrialFlag(std::string key, bool default_value);
>+  bool Get() const;
>+
>+ protected:
>+  bool Parse(absl::optional<std::string> str_value) override;
>+
>+ private:
>+  bool value_;
>+};
>+
>+// Accepts true, false, else parsed with sscanf %i, true if != 0.
>+extern template class FieldTrialParameter<bool>;
>+// Interpreted using sscanf %lf.
>+extern template class FieldTrialParameter<double>;
>+// Interpreted using sscanf %i.
>+extern template class FieldTrialParameter<int>;
>+// Using the given value as is.
>+extern template class FieldTrialParameter<std::string>;
>+
>+extern template class FieldTrialOptional<double>;
>+extern template class FieldTrialOptional<int>;
>+extern template class FieldTrialOptional<bool>;
>+extern template class FieldTrialOptional<std::string>;
>+
>+}  // namespace webrtc
>+
>+#endif  // RTC_BASE_EXPERIMENTS_FIELD_TRIAL_PARSER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/field_trial_parser_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/field_trial_parser_unittest.cc
>new file mode 100644
>index 00000000000..69f35bd6471
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/field_trial_parser_unittest.cc
>@@ -0,0 +1,125 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#include "rtc_base/experiments/field_trial_parser.h"
>+#include "rtc_base/gunit.h"
>+#include "system_wrappers/include/field_trial.h"
>+#include "test/field_trial.h"
>+
>+namespace webrtc {
>+namespace {
>+const char kDummyExperiment[] = "WebRTC-DummyExperiment";
>+
>+struct DummyExperiment {
>+  FieldTrialFlag enabled = FieldTrialFlag("Enabled");
>+  FieldTrialParameter<double> factor = FieldTrialParameter<double>("f", 0.5);
>+  FieldTrialParameter<int> retries = FieldTrialParameter<int>("r", 5);
>+  FieldTrialParameter<bool> ping = FieldTrialParameter<bool>("p", 0);
>+  FieldTrialParameter<std::string> hash =
>+      FieldTrialParameter<std::string>("h", "a80");
>+
>+  explicit DummyExperiment(std::string field_trial) {
>+    ParseFieldTrial({&enabled, &factor, &retries, &ping, &hash}, field_trial);
>+  }
>+  DummyExperiment() {
>+    std::string trial_string = field_trial::FindFullName(kDummyExperiment);
>+    ParseFieldTrial({&enabled, &factor, &retries, &ping, &hash}, trial_string);
>+  }
>+};
>+
>+enum class CustomEnum {
>+  kDefault = 0,
>+  kRed = 1,
>+  kBlue = 2,
>+};
>+}  // namespace
>+
>+TEST(FieldTrialParserTest, ParsesValidParameters) {
>+  DummyExperiment exp("Enabled,f:-1.7,r:2,p:1,h:x7c");
>+  EXPECT_TRUE(exp.enabled.Get());
>+  EXPECT_EQ(exp.factor.Get(), -1.7);
>+  EXPECT_EQ(exp.retries.Get(), 2);
>+  EXPECT_EQ(exp.ping.Get(), true);
>+  EXPECT_EQ(exp.hash.Get(), "x7c");
>+}
>+TEST(FieldTrialParserTest, InitializesFromFieldTrial) {
>+  test::ScopedFieldTrials field_trials(
>+      "WebRTC-OtherExperiment/Disabled/"
>+      "WebRTC-DummyExperiment/Enabled,f:-1.7,r:2,p:1,h:x7c/"
>+      "WebRTC-AnotherExperiment/Enabled,f:-3.1,otherstuff:beef/");
>+  DummyExperiment exp;
>+  EXPECT_TRUE(exp.enabled.Get());
>+  EXPECT_EQ(exp.factor.Get(), -1.7);
>+  EXPECT_EQ(exp.retries.Get(), 2);
>+  EXPECT_EQ(exp.ping.Get(), true);
>+  EXPECT_EQ(exp.hash.Get(), "x7c");
>+}
>+TEST(FieldTrialParserTest, UsesDefaults) {
>+  DummyExperiment exp("");
>+  EXPECT_FALSE(exp.enabled.Get());
>+  EXPECT_EQ(exp.factor.Get(), 0.5);
>+  EXPECT_EQ(exp.retries.Get(), 5);
>+  EXPECT_EQ(exp.ping.Get(), false);
>+  EXPECT_EQ(exp.hash.Get(), "a80");
>+}
>+TEST(FieldTrialParserTest, CanHandleMixedInput) {
>+  DummyExperiment exp("p:true,h:,Enabled");
>+  EXPECT_TRUE(exp.enabled.Get());
>+  EXPECT_EQ(exp.factor.Get(), 0.5);
>+  EXPECT_EQ(exp.retries.Get(), 5);
>+  EXPECT_EQ(exp.ping.Get(), true);
>+  EXPECT_EQ(exp.hash.Get(), "");
>+}
>+TEST(FieldTrialParserTest, IgnoresNewKey) {
>+  DummyExperiment exp("Disabled,r:-11,foo");
>+  EXPECT_FALSE(exp.enabled.Get());
>+  EXPECT_EQ(exp.factor.Get(), 0.5);
>+  EXPECT_EQ(exp.retries.Get(), -11);
>+}
>+TEST(FieldTrialParserTest, IgnoresInvalid) {
>+  DummyExperiment exp("Enabled,f,p:,r:%,,:foo,h");
>+  EXPECT_TRUE(exp.enabled.Get());
>+  EXPECT_EQ(exp.factor.Get(), 0.5);
>+  EXPECT_EQ(exp.retries.Get(), 5);
>+  EXPECT_EQ(exp.ping.Get(), false);
>+  EXPECT_EQ(exp.hash.Get(), "a80");
>+}
>+TEST(FieldTrialParserTest, ParsesOptionalParameters) {
>+  FieldTrialOptional<int> max_count("c", absl::nullopt);
>+  ParseFieldTrial({&max_count}, "");
>+  EXPECT_FALSE(max_count.Get().has_value());
>+  ParseFieldTrial({&max_count}, "c:10");
>+  EXPECT_EQ(max_count.Get().value(), 10);
>+  ParseFieldTrial({&max_count}, "c");
>+  EXPECT_FALSE(max_count.Get().has_value());
>+  ParseFieldTrial({&max_count}, "c:20");
>+  EXPECT_EQ(max_count.Get().value(), 20);
>+  ParseFieldTrial({&max_count}, "c:");
>+  EXPECT_EQ(max_count.Get().value(), 20);
>+  FieldTrialOptional<std::string> optional_string("s", std::string("ab"));
>+  ParseFieldTrial({&optional_string}, "s:");
>+  EXPECT_EQ(optional_string.Get().value(), "");
>+  ParseFieldTrial({&optional_string}, "s");
>+  EXPECT_FALSE(optional_string.Get().has_value());
>+}
>+TEST(FieldTrialParserTest, ParsesCustomEnumParameter) {
>+  FieldTrialEnum<CustomEnum> my_enum("e", CustomEnum::kDefault,
>+                                     {{"default", CustomEnum::kDefault},
>+                                      {"red", CustomEnum::kRed},
>+                                      {"blue", CustomEnum::kBlue}});
>+  ParseFieldTrial({&my_enum}, "");
>+  EXPECT_EQ(my_enum.Get(), CustomEnum::kDefault);
>+  ParseFieldTrial({&my_enum}, "e:red");
>+  EXPECT_EQ(my_enum.Get(), CustomEnum::kRed);
>+  ParseFieldTrial({&my_enum}, "e:2");
>+  EXPECT_EQ(my_enum.Get(), CustomEnum::kBlue);
>+  ParseFieldTrial({&my_enum}, "e:5");
>+  EXPECT_EQ(my_enum.Get(), CustomEnum::kBlue);
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/field_trial_units.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/field_trial_units.cc
>new file mode 100644
>index 00000000000..f53978b2115
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/field_trial_units.cc
>@@ -0,0 +1,92 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#include "rtc_base/experiments/field_trial_units.h"
>+
>+#include <limits>
>+#include <string>
>+
>+#include "absl/types/optional.h"
>+
>+// Large enough to fit "seconds", the longest supported unit name.
>+#define RTC_TRIAL_UNIT_LENGTH_STR "7"
>+#define RTC_TRIAL_UNIT_SIZE 8
>+
>+namespace webrtc {
>+namespace {
>+
>+struct ValueWithUnit {
>+  double value;
>+  std::string unit;
>+};
>+
>+absl::optional<ValueWithUnit> ParseValueWithUnit(std::string str) {
>+  if (str == "inf") {
>+    return ValueWithUnit{std::numeric_limits<double>::infinity(), ""};
>+  } else if (str == "-inf") {
>+    return ValueWithUnit{-std::numeric_limits<double>::infinity(), ""};
>+  } else {
>+    double double_val;
>+    char unit_char[RTC_TRIAL_UNIT_SIZE];
>+    unit_char[0] = 0;
>+    if (sscanf(str.c_str(), "%lf%" RTC_TRIAL_UNIT_LENGTH_STR "s", &double_val,
>+               unit_char) >= 1) {
>+      return ValueWithUnit{double_val, unit_char};
>+    }
>+  }
>+  return absl::nullopt;
>+}
>+}  // namespace
>+
>+template <>
>+absl::optional<DataRate> ParseTypedParameter<DataRate>(std::string str) {
>+  absl::optional<ValueWithUnit> result = ParseValueWithUnit(str);
>+  if (result) {
>+    if (result->unit.empty() || result->unit == "kbps") {
>+      return DataRate::kbps(result->value);
>+    } else if (result->unit == "bps") {
>+      return DataRate::bps(result->value);
>+    }
>+  }
>+  return absl::nullopt;
>+}
>+
>+template <>
>+absl::optional<DataSize> ParseTypedParameter<DataSize>(std::string str) {
>+  absl::optional<ValueWithUnit> result = ParseValueWithUnit(str);
>+  if (result) {
>+    if (result->unit.empty() || result->unit == "bytes")
>+      return DataSize::bytes(result->value);
>+  }
>+  return absl::nullopt;
>+}
>+
>+template <>
>+absl::optional<TimeDelta> ParseTypedParameter<TimeDelta>(std::string str) {
>+  absl::optional<ValueWithUnit> result = ParseValueWithUnit(str);
>+  if (result) {
>+    if (result->unit == "s" || result->unit == "seconds") {
>+      return TimeDelta::seconds(result->value);
>+    } else if (result->unit == "us") {
>+      return TimeDelta::us(result->value);
>+    } else if (result->unit.empty() || result->unit == "ms") {
>+      return TimeDelta::ms(result->value);
>+    }
>+  }
>+  return absl::nullopt;
>+}
>+
>+template class FieldTrialParameter<DataRate>;
>+template class FieldTrialParameter<DataSize>;
>+template class FieldTrialParameter<TimeDelta>;
>+
>+template class FieldTrialOptional<DataRate>;
>+template class FieldTrialOptional<DataSize>;
>+template class FieldTrialOptional<TimeDelta>;
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/field_trial_units.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/field_trial_units.h
>new file mode 100644
>index 00000000000..932c5cb4500
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/field_trial_units.h
>@@ -0,0 +1,29 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#ifndef RTC_BASE_EXPERIMENTS_FIELD_TRIAL_UNITS_H_
>+#define RTC_BASE_EXPERIMENTS_FIELD_TRIAL_UNITS_H_
>+
>+#include "rtc_base/experiments/field_trial_parser.h"
>+
>+#include "api/units/data_rate.h"
>+#include "api/units/data_size.h"
>+#include "api/units/time_delta.h"
>+
>+namespace webrtc {
>+extern template class FieldTrialParameter<DataRate>;
>+extern template class FieldTrialParameter<DataSize>;
>+extern template class FieldTrialParameter<TimeDelta>;
>+
>+extern template class FieldTrialOptional<DataRate>;
>+extern template class FieldTrialOptional<DataSize>;
>+extern template class FieldTrialOptional<TimeDelta>;
>+}  // namespace webrtc
>+
>+#endif  // RTC_BASE_EXPERIMENTS_FIELD_TRIAL_UNITS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/field_trial_units_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/field_trial_units_unittest.cc
>new file mode 100644
>index 00000000000..1a2b75d6468
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/field_trial_units_unittest.cc
>@@ -0,0 +1,62 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#include <string>
>+
>+#include "rtc_base/experiments/field_trial_parser.h"
>+#include "rtc_base/experiments/field_trial_units.h"
>+#include "rtc_base/gunit.h"
>+
>+namespace webrtc {
>+namespace {
>+struct DummyExperiment {
>+  FieldTrialParameter<DataRate> target_rate =
>+      FieldTrialParameter<DataRate>("t", DataRate::kbps(100));
>+  FieldTrialParameter<TimeDelta> period =
>+      FieldTrialParameter<TimeDelta>("p", TimeDelta::ms(100));
>+  FieldTrialOptional<DataSize> max_buffer =
>+      FieldTrialOptional<DataSize>("b", absl::nullopt);
>+
>+  explicit DummyExperiment(std::string field_trial) {
>+    ParseFieldTrial({&target_rate, &max_buffer, &period}, field_trial);
>+  }
>+};
>+}  // namespace
>+
>+TEST(FieldTrialParserUnitsTest, FallsBackToDefaults) {
>+  DummyExperiment exp("");
>+  EXPECT_EQ(exp.target_rate.Get(), DataRate::kbps(100));
>+  EXPECT_FALSE(exp.max_buffer.Get().has_value());
>+  EXPECT_EQ(exp.period.Get(), TimeDelta::ms(100));
>+}
>+TEST(FieldTrialParserUnitsTest, ParsesUnitParameters) {
>+  DummyExperiment exp("t:300kbps,b:5bytes,p:300ms");
>+  EXPECT_EQ(exp.target_rate.Get(), DataRate::kbps(300));
>+  EXPECT_EQ(*exp.max_buffer.Get(), DataSize::bytes(5));
>+  EXPECT_EQ(exp.period.Get(), TimeDelta::ms(300));
>+}
>+TEST(FieldTrialParserUnitsTest, ParsesDefaultUnitParameters) {
>+  DummyExperiment exp("t:300,b:5,p:300");
>+  EXPECT_EQ(exp.target_rate.Get(), DataRate::kbps(300));
>+  EXPECT_EQ(*exp.max_buffer.Get(), DataSize::bytes(5));
>+  EXPECT_EQ(exp.period.Get(), TimeDelta::ms(300));
>+}
>+TEST(FieldTrialParserUnitsTest, ParsesInfinityParameter) {
>+  DummyExperiment exp("t:inf,p:inf");
>+  EXPECT_EQ(exp.target_rate.Get(), DataRate::Infinity());
>+  EXPECT_EQ(exp.period.Get(), TimeDelta::PlusInfinity());
>+}
>+TEST(FieldTrialParserUnitsTest, ParsesOtherUnitParameters) {
>+  DummyExperiment exp("t:300bps,p:0.3 seconds,b:8 bytes");
>+  EXPECT_EQ(exp.target_rate.Get(), DataRate::bps(300));
>+  EXPECT_EQ(*exp.max_buffer.Get(), DataSize::bytes(8));
>+  EXPECT_EQ(exp.period.Get(), TimeDelta::ms(300));
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/quality_scaling_experiment.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/quality_scaling_experiment.cc
>new file mode 100644
>index 00000000000..c13de3f46bf
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/quality_scaling_experiment.cc
>@@ -0,0 +1,97 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#include "rtc_base/experiments/quality_scaling_experiment.h"
>+
>+#include <string>
>+
>+#include "rtc_base/logging.h"
>+#include "system_wrappers/include/field_trial.h"
>+
>+namespace webrtc {
>+namespace {
>+constexpr char kFieldTrial[] = "WebRTC-Video-QualityScaling";
>+constexpr int kMinQp = 1;
>+constexpr int kMaxVp8Qp = 127;
>+constexpr int kMaxVp9Qp = 255;
>+constexpr int kMaxH264Qp = 51;
>+constexpr int kMaxGenericQp = 255;
>+
>+absl::optional<VideoEncoder::QpThresholds> GetThresholds(int low,
>+                                                         int high,
>+                                                         int max) {
>+  if (low < kMinQp || high > max || high < low)
>+    return absl::nullopt;
>+
>+  RTC_LOG(LS_INFO) << "QP thresholds: low: " << low << ", high: " << high;
>+  return absl::optional<VideoEncoder::QpThresholds>(
>+      VideoEncoder::QpThresholds(low, high));
>+}
>+}  // namespace
>+
>+bool QualityScalingExperiment::Enabled() {
>+  return webrtc::field_trial::IsEnabled(kFieldTrial);
>+}
>+
>+absl::optional<QualityScalingExperiment::Settings>
>+QualityScalingExperiment::ParseSettings() {
>+  const std::string group = webrtc::field_trial::FindFullName(kFieldTrial);
>+  if (group.empty())
>+    return absl::nullopt;
>+
>+  Settings s;
>+  if (sscanf(group.c_str(), "Enabled-%d,%d,%d,%d,%d,%d,%d,%d,%f,%f,%d",
>+             &s.vp8_low, &s.vp8_high, &s.vp9_low, &s.vp9_high, &s.h264_low,
>+             &s.h264_high, &s.generic_low, &s.generic_high, &s.alpha_high,
>+             &s.alpha_low, &s.drop) != 11) {
>+    RTC_LOG(LS_WARNING) << "Invalid number of parameters provided.";
>+    return absl::nullopt;
>+  }
>+  return s;
>+}
>+
>+absl::optional<VideoEncoder::QpThresholds>
>+QualityScalingExperiment::GetQpThresholds(VideoCodecType codec_type) {
>+  const auto settings = ParseSettings();
>+  if (!settings)
>+    return absl::nullopt;
>+
>+  switch (codec_type) {
>+    case kVideoCodecVP8:
>+      return GetThresholds(settings->vp8_low, settings->vp8_high, kMaxVp8Qp);
>+    case kVideoCodecVP9:
>+      return GetThresholds(settings->vp9_low, settings->vp9_high, kMaxVp9Qp);
>+    case kVideoCodecH264:
>+      return GetThresholds(settings->h264_low, settings->h264_high, kMaxH264Qp);
>+    case kVideoCodecGeneric:
>+      return GetThresholds(settings->generic_low, settings->generic_high,
>+                           kMaxGenericQp);
>+    default:
>+      return absl::nullopt;
>+  }
>+}
>+
>+QualityScalingExperiment::Config QualityScalingExperiment::GetConfig() {
>+  const auto settings = ParseSettings();
>+  if (!settings)
>+    return Config();
>+
>+  Config config;
>+  config.use_all_drop_reasons = settings->drop > 0;
>+
>+  if (settings->alpha_high < 0 || settings->alpha_low < settings->alpha_high) {
>+    RTC_LOG(LS_WARNING) << "Invalid alpha value provided, using default.";
>+    return config;
>+  }
>+  config.alpha_high = settings->alpha_high;
>+  config.alpha_low = settings->alpha_low;
>+  return config;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/quality_scaling_experiment.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/quality_scaling_experiment.h
>new file mode 100644
>index 00000000000..80a25ef1eee
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/quality_scaling_experiment.h
>@@ -0,0 +1,59 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#ifndef RTC_BASE_EXPERIMENTS_QUALITY_SCALING_EXPERIMENT_H_
>+#define RTC_BASE_EXPERIMENTS_QUALITY_SCALING_EXPERIMENT_H_
>+
>+#include "absl/types/optional.h"
>+#include "api/video_codecs/video_encoder.h"
>+#include "common_types.h"  // NOLINT(build/include)
>+
>+namespace webrtc {
>+class QualityScalingExperiment {
>+ public:
>+  struct Settings {
>+    int vp8_low;       // VP8: low QP threshold.
>+    int vp8_high;      // VP8: high QP threshold.
>+    int vp9_low;       // VP9: low QP threshold.
>+    int vp9_high;      // VP9: high QP threshold.
>+    int h264_low;      // H264: low QP threshold.
>+    int h264_high;     // H264: high QP threshold.
>+    int generic_low;   // Generic: low QP threshold.
>+    int generic_high;  // Generic: high QP threshold.
>+    float alpha_high;  // |alpha_| for ExpFilter used when checking high QP.
>+    float alpha_low;   // |alpha_| for ExpFilter used when checking low QP.
>+    int drop;          // >0 sets |use_all_drop_reasons| to true.
>+  };
>+
>+  // Used by QualityScaler.
>+  struct Config {
>+    float alpha_high = 0.9995f;
>+    float alpha_low = 0.9999f;
>+    // If set, all type of dropped frames are used.
>+    // Otherwise only dropped frames by MediaOptimization are used.
>+    bool use_all_drop_reasons = false;
>+  };
>+
>+  // Returns true if the experiment is enabled.
>+  static bool Enabled();
>+
>+  // Returns settings from field trial.
>+  static absl::optional<Settings> ParseSettings();
>+
>+  // Returns QpThresholds for the |codec_type|.
>+  static absl::optional<VideoEncoder::QpThresholds> GetQpThresholds(
>+      VideoCodecType codec_type);
>+
>+  // Returns parsed values. If the parsing fails, default values are returned.
>+  static Config GetConfig();
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // RTC_BASE_EXPERIMENTS_QUALITY_SCALING_EXPERIMENT_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/quality_scaling_experiment_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/quality_scaling_experiment_unittest.cc
>new file mode 100644
>index 00000000000..15aadd6bb1e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/experiments/quality_scaling_experiment_unittest.cc
>@@ -0,0 +1,169 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "rtc_base/experiments/quality_scaling_experiment.h"
>+
>+#include "rtc_base/gunit.h"
>+#include "test/field_trial.h"
>+
>+namespace webrtc {
>+namespace {
>+void ExpectEqualSettings(QualityScalingExperiment::Settings a,
>+                         QualityScalingExperiment::Settings b) {
>+  EXPECT_EQ(a.vp8_low, b.vp8_low);
>+  EXPECT_EQ(a.vp8_high, b.vp8_high);
>+  EXPECT_EQ(a.vp9_low, b.vp9_low);
>+  EXPECT_EQ(a.vp9_high, b.vp9_high);
>+  EXPECT_EQ(a.h264_low, b.h264_low);
>+  EXPECT_EQ(a.h264_high, b.h264_high);
>+  EXPECT_EQ(a.generic_low, b.generic_low);
>+  EXPECT_EQ(a.generic_high, b.generic_high);
>+  EXPECT_EQ(a.alpha_high, b.alpha_high);
>+  EXPECT_EQ(a.alpha_low, b.alpha_low);
>+  EXPECT_EQ(a.drop, b.drop);
>+}
>+
>+void ExpectEqualConfig(QualityScalingExperiment::Config a,
>+                       QualityScalingExperiment::Config b) {
>+  EXPECT_EQ(a.alpha_high, b.alpha_high);
>+  EXPECT_EQ(a.alpha_low, b.alpha_low);
>+  EXPECT_EQ(a.use_all_drop_reasons, b.use_all_drop_reasons);
>+}
>+}  // namespace
>+
>+TEST(QualityScalingExperimentTest, DisabledWithoutFieldTrial) {
>+  webrtc::test::ScopedFieldTrials field_trials("");
>+  EXPECT_FALSE(QualityScalingExperiment::Enabled());
>+}
>+
>+TEST(QualityScalingExperimentTest, EnabledWithFieldTrial) {
>+  webrtc::test::ScopedFieldTrials field_trials(
>+      "WebRTC-Video-QualityScaling/Enabled/");
>+  EXPECT_TRUE(QualityScalingExperiment::Enabled());
>+}
>+
>+TEST(QualityScalingExperimentTest, ParseSettings) {
>+  const QualityScalingExperiment::Settings kExpected = {1, 2, 3,    4,     5, 6,
>+                                                        7, 8, 0.9f, 0.99f, 1};
>+  webrtc::test::ScopedFieldTrials field_trials(
>+      "WebRTC-Video-QualityScaling/Enabled-1,2,3,4,5,6,7,8,0.9,0.99,1/");
>+  const auto settings = QualityScalingExperiment::ParseSettings();
>+  EXPECT_TRUE(settings);
>+  ExpectEqualSettings(kExpected, *settings);
>+}
>+
>+TEST(QualityScalingExperimentTest, ParseSettingsFailsWithoutFieldTrial) {
>+  webrtc::test::ScopedFieldTrials field_trials("");
>+  EXPECT_FALSE(QualityScalingExperiment::ParseSettings());
>+}
>+
>+TEST(QualityScalingExperimentTest, ParseSettingsFailsWithInvalidFieldTrial) {
>+  webrtc::test::ScopedFieldTrials field_trials(
>+      "WebRTC-Video-QualityScaling/Enabled-invalid/");
>+  EXPECT_FALSE(QualityScalingExperiment::ParseSettings());
>+}
>+
>+TEST(QualityScalingExperimentTest, GetConfig) {
>+  webrtc::test::ScopedFieldTrials field_trials(
>+      "WebRTC-Video-QualityScaling/Enabled-1,2,3,4,5,6,7,8,0.9,0.99,0/");
>+  const auto config = QualityScalingExperiment::GetConfig();
>+  EXPECT_EQ(0.9f, config.alpha_high);
>+  EXPECT_EQ(0.99f, config.alpha_low);
>+  EXPECT_FALSE(config.use_all_drop_reasons);
>+}
>+
>+TEST(QualityScalingExperimentTest, GetsDefaultConfigForInvalidFieldTrial) {
>+  webrtc::test::ScopedFieldTrials field_trials(
>+      "WebRTC-Video-QualityScaling/Enabled-invalid/");
>+  const auto config = QualityScalingExperiment::GetConfig();
>+  ExpectEqualConfig(config, QualityScalingExperiment::Config());
>+}
>+
>+TEST(QualityScalingExperimentTest, GetsDefaultAlphaForInvalidValue) {
>+  QualityScalingExperiment::Config expected_config;
>+  expected_config.use_all_drop_reasons = true;
>+  webrtc::test::ScopedFieldTrials field_trials(
>+      "WebRTC-Video-QualityScaling/Enabled-1,2,3,4,5,6,7,8,0.99,0.9,1/");
>+  const auto config = QualityScalingExperiment::GetConfig();
>+  ExpectEqualConfig(config, expected_config);
>+}
>+
>+TEST(QualityScalingExperimentTest, GetVp8Thresholds) {
>+  webrtc::test::ScopedFieldTrials field_trials(
>+      "WebRTC-Video-QualityScaling/Enabled-1,2,3,4,5,6,0,0,0.9,0.99,1/");
>+  const auto thresholds =
>+      QualityScalingExperiment::GetQpThresholds(kVideoCodecVP8);
>+  EXPECT_TRUE(thresholds);
>+  EXPECT_EQ(1, thresholds->low);
>+  EXPECT_EQ(2, thresholds->high);
>+}
>+
>+TEST(QualityScalingExperimentTest, GetThresholdsFailsForInvalidVp8Value) {
>+  webrtc::test::ScopedFieldTrials field_trials(
>+      "WebRTC-Video-QualityScaling/Enabled-0,0,3,4,5,6,7,8,0.9,0.99,1/");
>+  const auto thresholds =
>+      QualityScalingExperiment::GetQpThresholds(kVideoCodecVP8);
>+  EXPECT_FALSE(thresholds);
>+}
>+
>+TEST(QualityScalingExperimentTest, GetVp9Thresholds) {
>+  webrtc::test::ScopedFieldTrials field_trials(
>+      "WebRTC-Video-QualityScaling/Enabled-1,2,3,4,5,6,0,0,0.9,0.99,1/");
>+  const auto thresholds =
>+      QualityScalingExperiment::GetQpThresholds(kVideoCodecVP9);
>+  EXPECT_TRUE(thresholds);
>+  EXPECT_EQ(3, thresholds->low);
>+  EXPECT_EQ(4, thresholds->high);
>+}
>+
>+TEST(QualityScalingExperimentTest, GetThresholdsFailsForInvalidVp9Value) {
>+  webrtc::test::ScopedFieldTrials field_trials(
>+      "WebRTC-Video-QualityScaling/Enabled-1,2,0,0,5,6,7,8,0.9,0.99,1/");
>+  const auto thresholds =
>+      QualityScalingExperiment::GetQpThresholds(kVideoCodecVP9);
>+  EXPECT_FALSE(thresholds);
>+}
>+
>+TEST(QualityScalingExperimentTest, GetH264Thresholds) {
>+  webrtc::test::ScopedFieldTrials field_trials(
>+      "WebRTC-Video-QualityScaling/Enabled-1,2,3,4,5,6,0,0,0.9,0.99,1/");
>+  const auto thresholds =
>+      QualityScalingExperiment::GetQpThresholds(kVideoCodecH264);
>+  EXPECT_TRUE(thresholds);
>+  EXPECT_EQ(5, thresholds->low);
>+  EXPECT_EQ(6, thresholds->high);
>+}
>+
>+TEST(QualityScalingExperimentTest, GetThresholdsFailsForInvalidH264Value) {
>+  webrtc::test::ScopedFieldTrials field_trials(
>+      "WebRTC-Video-QualityScaling/Enabled-1,2,3,4,0,0,7,8,0.9,0.99,1/");
>+  const auto thresholds =
>+      QualityScalingExperiment::GetQpThresholds(kVideoCodecH264);
>+  EXPECT_FALSE(thresholds);
>+}
>+
>+TEST(QualityScalingExperimentTest, GetGenericThresholds) {
>+  webrtc::test::ScopedFieldTrials field_trials(
>+      "WebRTC-Video-QualityScaling/Enabled-1,2,3,4,0,0,7,8,0.9,0.99,1/");
>+  const auto thresholds =
>+      QualityScalingExperiment::GetQpThresholds(kVideoCodecGeneric);
>+  EXPECT_TRUE(thresholds);
>+  EXPECT_EQ(7, thresholds->low);
>+  EXPECT_EQ(8, thresholds->high);
>+}
>+
>+TEST(QualityScalingExperimentTest, GetThresholdsFailsForInvalidGenericValue) {
>+  webrtc::test::ScopedFieldTrials field_trials(
>+      "WebRTC-Video-QualityScaling/Enabled-1,2,3,4,5,6,0,0,0.9,0.99,1/");
>+  const auto thresholds =
>+      QualityScalingExperiment::GetQpThresholds(kVideoCodecGeneric);
>+  EXPECT_FALSE(thresholds);
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/fakeclock.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/fakeclock.cc
>index 721f6997119..ade920893f8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/fakeclock.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/fakeclock.cc
>@@ -31,10 +31,10 @@ void FakeClock::SetTimeNanos(int64_t nanos) {
>   MessageQueueManager::ProcessAllMessageQueues();
> }
> 
>-void FakeClock::AdvanceTime(TimeDelta delta) {
>+void FakeClock::AdvanceTime(webrtc::TimeDelta delta) {
>   {
>     CritScope cs(&lock_);
>-    time_ += delta.ToNanoseconds();
>+    time_ += delta.ns();
>   }
>   MessageQueueManager::ProcessAllMessageQueues();
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/fakeclock.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/fakeclock.h
>index 1b1ee715c0d..2da8c616ccb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/fakeclock.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/fakeclock.h
>@@ -11,8 +11,8 @@
> #ifndef RTC_BASE_FAKECLOCK_H_
> #define RTC_BASE_FAKECLOCK_H_
> 
>+#include "api/units/time_delta.h"
> #include "rtc_base/criticalsection.h"
>-#include "rtc_base/timedelta.h"
> #include "rtc_base/timeutils.h"
> 
> namespace rtc {
>@@ -36,10 +36,11 @@ class FakeClock : public ClockInterface {
>     SetTimeNanos(kNumNanosecsPerMicrosec * micros);
>   }
> 
>-  void AdvanceTime(TimeDelta delta);
>+  void AdvanceTime(webrtc::TimeDelta delta);
>   void AdvanceTimeMicros(int64_t micros) {
>-    AdvanceTime(rtc::TimeDelta::FromMicroseconds(micros));
>+    AdvanceTime(webrtc::TimeDelta::us(micros));
>   }
>+
>  private:
>   CriticalSection lock_;
>   int64_t time_ RTC_GUARDED_BY(lock_) = 0;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/fakenetwork.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/fakenetwork.h
>index fd4eb615bc7..8ed11646cde 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/fakenetwork.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/fakenetwork.h
>@@ -28,8 +28,7 @@ const int kFakeIPv4NetworkPrefixLength = 24;
> const int kFakeIPv6NetworkPrefixLength = 64;
> 
> // Fake network manager that allows us to manually specify the IPs to use.
>-class FakeNetworkManager : public NetworkManagerBase,
>-                           public MessageHandler {
>+class FakeNetworkManager : public NetworkManagerBase, public MessageHandler {
>  public:
>   FakeNetworkManager() {}
> 
>@@ -54,8 +53,7 @@ class FakeNetworkManager : public NetworkManagerBase,
>   }
> 
>   void RemoveInterface(const SocketAddress& iface) {
>-    for (IfaceList::iterator it = ifaces_.begin();
>-         it != ifaces_.end(); ++it) {
>+    for (IfaceList::iterator it = ifaces_.begin(); it != ifaces_.end(); ++it) {
>       if (it->first.EqualIPs(iface)) {
>         ifaces_.erase(it);
>         break;
>@@ -79,9 +77,7 @@ class FakeNetworkManager : public NetworkManagerBase,
>   virtual void StopUpdating() { --start_count_; }
> 
>   // MessageHandler interface.
>-  virtual void OnMessage(Message* msg) {
>-    DoUpdateNetworks();
>-  }
>+  virtual void OnMessage(Message* msg) { DoUpdateNetworks(); }
> 
>   using NetworkManagerBase::set_enumeration_permission;
>   using NetworkManagerBase::set_default_local_addresses;
>@@ -91,8 +87,7 @@ class FakeNetworkManager : public NetworkManagerBase,
>     if (start_count_ == 0)
>       return;
>     std::vector<Network*> networks;
>-    for (IfaceList::iterator it = ifaces_.begin();
>-         it != ifaces_.end(); ++it) {
>+    for (IfaceList::iterator it = ifaces_.begin(); it != ifaces_.end(); ++it) {
>       int prefix_length = 0;
>       if (it->first.ipaddr().family() == AF_INET) {
>         prefix_length = kFakeIPv4NetworkPrefixLength;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/fakesslidentity.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/fakesslidentity.cc
>index 296f09c9181..80a3e788878 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/fakesslidentity.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/fakesslidentity.cc
>@@ -14,25 +14,16 @@
> #include <string>
> #include <utility>
> 
>+#include "absl/memory/memory.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/messagedigest.h"
>-#include "rtc_base/ptr_util.h"
> 
> namespace rtc {
> 
>-FakeSSLCertificate::FakeSSLCertificate(const std::string& data)
>-    : data_(data), digest_algorithm_(DIGEST_SHA_1), expiration_time_(-1) {}
>-
>-FakeSSLCertificate::FakeSSLCertificate(const std::vector<std::string>& certs)
>-    : data_(certs.front()),
>+FakeSSLCertificate::FakeSSLCertificate(const std::string& pem_string)
>+    : pem_string_(pem_string),
>       digest_algorithm_(DIGEST_SHA_1),
>-      expiration_time_(-1) {
>-  std::vector<std::string>::const_iterator it;
>-  // Skip certs[0].
>-  for (it = certs.begin() + 1; it != certs.end(); ++it) {
>-    certs_.push_back(FakeSSLCertificate(*it));
>-  }
>-}
>+      expiration_time_(-1) {}
> 
> FakeSSLCertificate::FakeSSLCertificate(const FakeSSLCertificate&) = default;
> 
>@@ -43,12 +34,13 @@ FakeSSLCertificate* FakeSSLCertificate::GetReference() const {
> }
> 
> std::string FakeSSLCertificate::ToPEMString() const {
>-  return data_;
>+  return pem_string_;
> }
> 
> void FakeSSLCertificate::ToDER(Buffer* der_buffer) const {
>   std::string der_string;
>-  RTC_CHECK(SSLIdentity::PemToDer(kPemTypeCertificate, data_, &der_string));
>+  RTC_CHECK(
>+      SSLIdentity::PemToDer(kPemTypeCertificate, pem_string_, &der_string));
>   der_buffer->SetData(der_string.c_str(), der_string.size());
> }
> 
>@@ -74,30 +66,40 @@ bool FakeSSLCertificate::ComputeDigest(const std::string& algorithm,
>                                        unsigned char* digest,
>                                        size_t size,
>                                        size_t* length) const {
>-  *length =
>-      rtc::ComputeDigest(algorithm, data_.c_str(), data_.size(), digest, size);
>+  *length = rtc::ComputeDigest(algorithm, pem_string_.c_str(),
>+                               pem_string_.size(), digest, size);
>   return (*length != 0);
> }
> 
>-std::unique_ptr<SSLCertChain> FakeSSLCertificate::GetChain() const {
>-  if (certs_.empty())
>-    return nullptr;
>-  std::vector<std::unique_ptr<SSLCertificate>> new_certs(certs_.size());
>-  std::transform(certs_.begin(), certs_.end(), new_certs.begin(), DupCert);
>-  return MakeUnique<SSLCertChain>(std::move(new_certs));
>-}
>+FakeSSLIdentity::FakeSSLIdentity(const std::string& pem_string)
>+    : FakeSSLIdentity(FakeSSLCertificate(pem_string)) {}
> 
>-FakeSSLIdentity::FakeSSLIdentity(const std::string& data) : cert_(data) {}
>+FakeSSLIdentity::FakeSSLIdentity(const std::vector<std::string>& pem_strings) {
>+  std::vector<std::unique_ptr<SSLCertificate>> certs;
>+  for (const std::string& pem_string : pem_strings) {
>+    certs.push_back(absl::make_unique<FakeSSLCertificate>(pem_string));
>+  }
>+  cert_chain_ = absl::make_unique<SSLCertChain>(std::move(certs));
>+}
> 
> FakeSSLIdentity::FakeSSLIdentity(const FakeSSLCertificate& cert)
>-    : cert_(cert) {}
>+    : cert_chain_(absl::make_unique<SSLCertChain>(&cert)) {}
>+
>+FakeSSLIdentity::FakeSSLIdentity(const FakeSSLIdentity& o)
>+    : cert_chain_(o.cert_chain_->UniqueCopy()) {}
>+
>+FakeSSLIdentity::~FakeSSLIdentity() = default;
> 
> FakeSSLIdentity* FakeSSLIdentity::GetReference() const {
>   return new FakeSSLIdentity(*this);
> }
> 
>-const FakeSSLCertificate& FakeSSLIdentity::certificate() const {
>-  return cert_;
>+const SSLCertificate& FakeSSLIdentity::certificate() const {
>+  return cert_chain_->Get(0);
>+}
>+
>+const SSLCertChain& FakeSSLIdentity::cert_chain() const {
>+  return *cert_chain_.get();
> }
> 
> std::string FakeSSLIdentity::PrivateKeyToPEMString() const {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/fakesslidentity.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/fakesslidentity.h
>index 52aaf055582..4494a524efa 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/fakesslidentity.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/fakesslidentity.h
>@@ -18,13 +18,11 @@
> 
> namespace rtc {
> 
>-class FakeSSLCertificate : public rtc::SSLCertificate {
>+class FakeSSLCertificate : public SSLCertificate {
>  public:
>   // SHA-1 is the default digest algorithm because it is available in all build
>   // configurations used for unit testing.
>-  explicit FakeSSLCertificate(const std::string& data);
>-
>-  explicit FakeSSLCertificate(const std::vector<std::string>& certs);
>+  explicit FakeSSLCertificate(const std::string& pem_string);
> 
>   FakeSSLCertificate(const FakeSSLCertificate&);
>   ~FakeSSLCertificate() override;
>@@ -39,32 +37,33 @@ class FakeSSLCertificate : public rtc::SSLCertificate {
>                      unsigned char* digest,
>                      size_t size,
>                      size_t* length) const override;
>-  std::unique_ptr<SSLCertChain> GetChain() const override;
> 
>   void SetCertificateExpirationTime(int64_t expiration_time);
> 
>   void set_digest_algorithm(const std::string& algorithm);
> 
>  private:
>-  static std::unique_ptr<SSLCertificate> DupCert(FakeSSLCertificate cert) {
>-    return cert.GetUniqueReference();
>-  }
>-  static void DeleteCert(SSLCertificate* cert) { delete cert; }
>-  std::string data_;
>-  std::vector<FakeSSLCertificate> certs_;
>+  std::string pem_string_;
>   std::string digest_algorithm_;
>   // Expiration time in seconds relative to epoch, 1970-01-01T00:00:00Z (UTC).
>   int64_t expiration_time_;
> };
> 
>-class FakeSSLIdentity : public rtc::SSLIdentity {
>+class FakeSSLIdentity : public SSLIdentity {
>  public:
>-  explicit FakeSSLIdentity(const std::string& data);
>+  explicit FakeSSLIdentity(const std::string& pem_string);
>+  // For a certificate chain.
>+  explicit FakeSSLIdentity(const std::vector<std::string>& pem_strings);
>   explicit FakeSSLIdentity(const FakeSSLCertificate& cert);
> 
>+  explicit FakeSSLIdentity(const FakeSSLIdentity& o);
>+
>+  ~FakeSSLIdentity() override;
>+
>   // SSLIdentity implementation.
>   FakeSSLIdentity* GetReference() const override;
>-  const FakeSSLCertificate& certificate() const override;
>+  const SSLCertificate& certificate() const override;
>+  const SSLCertChain& cert_chain() const override;
>   // Not implemented.
>   std::string PrivateKeyToPEMString() const override;
>   // Not implemented.
>@@ -73,7 +72,7 @@ class FakeSSLIdentity : public rtc::SSLIdentity {
>   virtual bool operator==(const SSLIdentity& other) const;
> 
>  private:
>-  FakeSSLCertificate cert_;
>+  std::unique_ptr<SSLCertChain> cert_chain_;
> };
> 
> }  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/file.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/file.cc
>index a6ee2aa99c5..62024119fcc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/file.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/file.cc
>@@ -14,15 +14,6 @@
> 
> namespace rtc {
> 
>-namespace {
>-
>-std::string NormalizePathname(Pathname&& path) {
>-  path.Normalize();
>-  return path.pathname();
>-}
>-
>-}  // namespace
>-
> File::File(PlatformFile file) : file_(file) {}
> 
> File::File() : file_(kInvalidPlatformFileValue) {}
>@@ -36,46 +27,16 @@ File File::Open(const std::string& path) {
>   return File(OpenPlatformFile(path));
> }
> 
>-// static
>-File File::Open(Pathname&& path) {
>-  return Open(NormalizePathname(std::move(path)));
>-}
>-
>-// static
>-File File::Open(const Pathname& path) {
>-  return Open(Pathname(path));
>-}
>-
> // static
> File File::Create(const std::string& path) {
>   return File(CreatePlatformFile(path));
> }
> 
>-// static
>-File File::Create(Pathname&& path) {
>-  return Create(NormalizePathname(std::move(path)));
>-}
>-
>-// static
>-File File::Create(const Pathname& path) {
>-  return Create(Pathname(path));
>-}
>-
> // static
> bool File::Remove(const std::string& path) {
>   return RemoveFile(path);
> }
> 
>-// static
>-bool File::Remove(Pathname&& path) {
>-  return Remove(NormalizePathname(std::move(path)));
>-}
>-
>-// static
>-bool File::Remove(const Pathname& path) {
>-  return Remove(Pathname(path));
>-}
>-
> File::File(File&& other) : file_(other.file_) {
>   other.file_ = kInvalidPlatformFileValue;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/file.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/file.h
>index f87d9ce186d..75fd93d1310 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/file.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/file.h
>@@ -16,7 +16,6 @@
> #include <string>
> 
> #include "rtc_base/constructormagic.h"
>-#include "rtc_base/pathutils.h"
> #include "rtc_base/platform_file.h"
> 
> namespace rtc {
>@@ -40,17 +39,11 @@ class File {
> 
>   // Open and Create give files with both reading and writing enabled.
>   static File Open(const std::string& path);
>-  static File Open(Pathname&& path);
>-  static File Open(const Pathname& path);
>   // If the file already exists it will be overwritten.
>   static File Create(const std::string& path);
>-  static File Create(Pathname&& path);
>-  static File Create(const Pathname& path);
> 
>   // Remove a file in the file system.
>   static bool Remove(const std::string& path);
>-  static bool Remove(Pathname&& path);
>-  static bool Remove(const Pathname& path);
> 
>   size_t Write(const uint8_t* data, size_t length);
>   size_t Read(uint8_t* buffer, size_t length);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/file_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/file_unittest.cc
>index a8e39dd98ec..5a72e5d1a0a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/file_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/file_unittest.cc
>@@ -163,39 +163,13 @@ TEST_F(FileTest, RandomAccessReadWrite) {
>   EXPECT_TRUE(VerifyBuffer(out, 2, 0));
> }
> 
>-TEST_F(FileTest, OpenFromPathname) {
>-  {
>-    File file = File::Open(Pathname(path_));
>-    ASSERT_TRUE(file.IsOpen()) << "Error: " << LastError();
>-  }
>-
>-  {
>-    Pathname path(path_);
>-    File file = File::Open(path);
>-    ASSERT_TRUE(file.IsOpen()) << "Error: " << LastError();
>-  }
>-}
>-
>-TEST_F(FileTest, CreateFromPathname) {
>-  {
>-    File file = File::Create(Pathname(path_));
>-    ASSERT_TRUE(file.IsOpen()) << "Error: " << LastError();
>-  }
>-
>-  {
>-    Pathname path(path_);
>-    File file = File::Create(path);
>-    ASSERT_TRUE(file.IsOpen()) << "Error: " << LastError();
>-  }
>-}
>-
> TEST_F(FileTest, ShouldBeAbleToRemoveFile) {
>   {
>-    File file = File::Open(Pathname(path_));
>+    File file = File::Open(path_);
>     ASSERT_TRUE(file.IsOpen()) << "Error: " << LastError();
>   }
> 
>-  ASSERT_TRUE(File::Remove(Pathname(path_))) << "Error: " << LastError();
>+  ASSERT_TRUE(File::Remove(path_)) << "Error: " << LastError();
> }
> 
> }  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/filerotatingstream.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/filerotatingstream.cc
>index 4e3aa732452..c9a663aafff 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/filerotatingstream.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/filerotatingstream.cc
>@@ -11,22 +11,22 @@
> #include "rtc_base/filerotatingstream.h"
> 
> #include <algorithm>
>-#include <iostream>
>+#include <cstdio>
> #include <string>
> 
> #include "rtc_base/checks.h"
> #include "rtc_base/fileutils.h"
> #include "rtc_base/pathutils.h"
>+#include "rtc_base/strings/string_builder.h"
> 
>-// Note: We use std::cerr for logging in the write paths of this stream to avoid
>+// Note: We use fprintf for logging in the write paths of this stream to avoid
> // infinite loops when logging.
> 
> namespace rtc {
> 
> FileRotatingStream::FileRotatingStream(const std::string& dir_path,
>                                        const std::string& file_prefix)
>-    : FileRotatingStream(dir_path, file_prefix, 0, 0, kRead) {
>-}
>+    : FileRotatingStream(dir_path, file_prefix, 0, 0, kRead) {}
> 
> FileRotatingStream::FileRotatingStream(const std::string& dir_path,
>                                        const std::string& file_prefix,
>@@ -77,8 +77,7 @@ FileRotatingStream::FileRotatingStream(const std::string& dir_path,
>   }
> }
> 
>-FileRotatingStream::~FileRotatingStream() {
>-}
>+FileRotatingStream::~FileRotatingStream() {}
> 
> StreamState FileRotatingStream::GetState() const {
>   if (mode_ == kRead && current_file_index_ < file_names_.size()) {
>@@ -149,7 +148,7 @@ StreamResult FileRotatingStream::Write(const void* data,
>     return SR_EOS;
>   }
>   if (!file_stream_) {
>-    std::cerr << "Open() must be called before Write." << std::endl;
>+    std::fprintf(stderr, "Open() must be called before Write.\n");
>     return SR_ERROR;
>   }
>   // Write as much as will fit in to the current file.
>@@ -213,7 +212,7 @@ bool FileRotatingStream::Open() {
>       std::vector<std::string> matching_files = GetFilesWithPrefix();
>       for (auto matching_file : matching_files) {
>         if (!Filesystem::DeleteFile(matching_file)) {
>-          std::cerr << "Failed to delete: " << matching_file << std::endl;
>+          std::fprintf(stderr, "Failed to delete: %s\n", matching_file.c_str());
>         }
>       }
>       return OpenCurrentFile();
>@@ -225,8 +224,7 @@ bool FileRotatingStream::Open() {
> bool FileRotatingStream::DisableBuffering() {
>   disable_buffering_ = true;
>   if (!file_stream_) {
>-    std::cerr << "Open() must be called before DisableBuffering()."
>-              << std::endl;
>+    std::fprintf(stderr, "Open() must be called before DisableBuffering().\n");
>     return false;
>   }
>   return file_stream_->DisableBuffering();
>@@ -257,8 +255,8 @@ bool FileRotatingStream::OpenCurrentFile() {
>   }
>   int error = 0;
>   if (!file_stream_->Open(file_path, mode, &error)) {
>-    std::cerr << "Failed to open: " << file_path << "Error: " << error
>-              << std::endl;
>+    std::fprintf(stderr, "Failed to open: %s Error: %i\n", file_path.c_str(),
>+                 error);
>     file_stream_.reset();
>     return false;
>   }
>@@ -286,7 +284,7 @@ void FileRotatingStream::RotateFiles() {
>   std::string file_to_delete = file_names_[rotation_index_];
>   if (Filesystem::IsFile(file_to_delete)) {
>     if (!Filesystem::DeleteFile(file_to_delete)) {
>-      std::cerr << "Failed to delete: " << file_to_delete << std::endl;
>+      std::fprintf(stderr, "Failed to delete: %s\n", file_to_delete.c_str());
>     }
>   }
>   for (auto i = rotation_index_; i > 0; --i) {
>@@ -294,8 +292,8 @@ void FileRotatingStream::RotateFiles() {
>     std::string unrotated_name = file_names_[i - 1];
>     if (Filesystem::IsFile(unrotated_name)) {
>       if (!Filesystem::MoveFile(unrotated_name, rotated_name)) {
>-        std::cerr << "Failed to move: " << unrotated_name << " to "
>-                  << rotated_name << std::endl;
>+        std::fprintf(stderr, "Failed to move: %s to %s\n",
>+                     unrotated_name.c_str(), rotated_name.c_str());
>       }
>     }
>   }
>@@ -327,21 +325,15 @@ std::vector<std::string> FileRotatingStream::GetFilesWithPrefix() const {
> std::string FileRotatingStream::GetFilePath(size_t index,
>                                             size_t num_files) const {
>   RTC_DCHECK_LT(index, num_files);
>-  std::ostringstream file_name;
>-  // The format will be "_%<num_digits>zu". We want to zero pad the index so
>-  // that it will sort nicely.
>-  size_t max_digits = ((num_files - 1) / 10) + 1;
>-  size_t num_digits = (index / 10) + 1;
>-  RTC_DCHECK_LE(num_digits, max_digits);
>-  size_t padding = max_digits - num_digits;
>-
>-  file_name << file_prefix_ << "_";
>-  for (size_t i = 0; i < padding; ++i) {
>-    file_name << "0";
>-  }
>-  file_name << index;
> 
>-  Pathname file_path(dir_path_, file_name.str());
>+  const size_t buffer_size = 32;
>+  char file_postfix[buffer_size];
>+  // We want to zero pad the index so that it will sort nicely.
>+  const int max_digits = std::snprintf(nullptr, 0, "%zu", num_files - 1);
>+  RTC_DCHECK_LT(1 + max_digits, buffer_size);
>+  std::snprintf(file_postfix, buffer_size, "_%0*zu", max_digits, index);
>+
>+  Pathname file_path(dir_path_, file_prefix_ + file_postfix);
>   return file_path.pathname();
> }
> 
>@@ -349,8 +341,7 @@ CallSessionFileRotatingStream::CallSessionFileRotatingStream(
>     const std::string& dir_path)
>     : FileRotatingStream(dir_path, kLogPrefix),
>       max_total_log_size_(0),
>-      num_rotations_(0) {
>-}
>+      num_rotations_(0) {}
> 
> CallSessionFileRotatingStream::CallSessionFileRotatingStream(
>     const std::string& dir_path,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/filerotatingstream_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/filerotatingstream_unittest.cc
>index 16db280d213..19055162fd0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/filerotatingstream_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/filerotatingstream_unittest.cc
>@@ -31,7 +31,7 @@ void CleanupLogDirectory(const FileRotatingStream& stream) {
> 
> }  // namespace
> 
>-#if defined (WEBRTC_ANDROID)
>+#if defined(WEBRTC_ANDROID)
> // Fails on Android: https://bugs.chromium.org/p/webrtc/issues/detail?id=4364.
> #define MAYBE_FileRotatingStreamTest DISABLED_FileRotatingStreamTest
> #else
>@@ -195,20 +195,20 @@ TEST_F(MAYBE_FileRotatingStreamTest, WriteOverflowAndRead) {
> // Tests that the returned file paths have the right folder and prefix.
> TEST_F(MAYBE_FileRotatingStreamTest, GetFilePath) {
>   Init("FileRotatingStreamTestGetFilePath", kFilePrefix, kMaxFileSize, 20);
>+  // dir_path_ includes a trailing delimiter.
>+  const std::string prefix = dir_path_ + kFilePrefix;
>   for (auto i = 0; i < 20; ++i) {
>-    Pathname path(stream_->GetFilePath(i));
>-    EXPECT_EQ(0, path.folder().compare(dir_path_));
>-    EXPECT_EQ(0, path.filename().compare(0, strlen(kFilePrefix), kFilePrefix));
>+    EXPECT_EQ(0, stream_->GetFilePath(i).compare(0, prefix.size(), prefix));
>   }
> }
> 
>-#if defined (WEBRTC_ANDROID)
>+#if defined(WEBRTC_ANDROID)
> // Fails on Android: https://bugs.chromium.org/p/webrtc/issues/detail?id=4364.
> #define MAYBE_CallSessionFileRotatingStreamTest \
>-    DISABLED_CallSessionFileRotatingStreamTest
>+  DISABLED_CallSessionFileRotatingStreamTest
> #else
> #define MAYBE_CallSessionFileRotatingStreamTest \
>-    CallSessionFileRotatingStreamTest
>+  CallSessionFileRotatingStreamTest
> #endif
> 
> class MAYBE_CallSessionFileRotatingStreamTest : public ::testing::Test {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/fileutils.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/fileutils.cc
>index 40ec86f7a93..7d83f97fae4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/fileutils.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/fileutils.cc
>@@ -35,7 +35,7 @@ namespace rtc {
> // to the first file in the directory, and can be advanecd with Next(). This
> // allows you to get information about each file.
> 
>-  // Constructor
>+// Constructor
> DirectoryIterator::DirectoryIterator()
> #ifdef WEBRTC_WIN
>     : handle_(INVALID_HANDLE_VALUE) {
>@@ -45,7 +45,7 @@ DirectoryIterator::DirectoryIterator()
> #endif
> }
> 
>-  // Destructor
>+// Destructor
> DirectoryIterator::~DirectoryIterator() {
> #if defined(WEBRTC_WIN)
>   if (handle_ != INVALID_HANDLE_VALUE)
>@@ -56,10 +56,10 @@ DirectoryIterator::~DirectoryIterator() {
> #endif
> }
> 
>-  // Starts traversing a directory.
>-  // dir is the directory to traverse
>-  // returns true if the directory exists and is valid
>-bool DirectoryIterator::Iterate(const Pathname &dir) {
>+// Starts traversing a directory.
>+// dir is the directory to traverse
>+// returns true if the directory exists and is valid
>+bool DirectoryIterator::Iterate(const Pathname& dir) {
>   directory_ = dir.pathname();
> #if defined(WEBRTC_WIN)
>   if (handle_ != INVALID_HANDLE_VALUE)
>@@ -84,8 +84,8 @@ bool DirectoryIterator::Iterate(const Pathname &dir) {
>   return true;
> }
> 
>-  // Advances to the next file
>-  // returns true if there were more files in the directory.
>+// Advances to the next file
>+// returns true if there were more files in the directory.
> bool DirectoryIterator::Next() {
> #if defined(WEBRTC_WIN)
>   return ::FindNextFile(handle_, &data_) == TRUE;
>@@ -98,7 +98,7 @@ bool DirectoryIterator::Next() {
> #endif
> }
> 
>-  // returns true if the file currently pointed to is a directory
>+// returns true if the file currently pointed to is a directory
> bool DirectoryIterator::IsDirectory() const {
> #if defined(WEBRTC_WIN)
>   return (data_.dwFileAttributes & FILE_ATTRIBUTE_DIRECTORY) != FALSE;
>@@ -107,7 +107,7 @@ bool DirectoryIterator::IsDirectory() const {
> #endif
> }
> 
>-  // returns the name of the file currently pointed to
>+// returns the name of the file currently pointed to
> std::string DirectoryIterator::Name() const {
> #if defined(WEBRTC_WIN)
>   return ToUtf8(data_.cFileName);
>@@ -117,17 +117,14 @@ std::string DirectoryIterator::Name() const {
> #endif
> }
> 
>-FilesystemInterface* Filesystem::default_filesystem_ = nullptr;
>-
>-FilesystemInterface *Filesystem::EnsureDefaultFilesystem() {
>-  if (!default_filesystem_) {
>+FilesystemInterface* Filesystem::GetFilesystem() {
> #if defined(WEBRTC_WIN)
>-    default_filesystem_ = new Win32Filesystem();
>+  static FilesystemInterface* const filesystem = new Win32Filesystem();
> #else
>-    default_filesystem_ = new UnixFilesystem();
>+  static FilesystemInterface* const filesystem = new UnixFilesystem();
> #endif
>-  }
>-  return default_filesystem_;
>+
>+  return filesystem;
> }
> 
> }  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/fileutils.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/fileutils.h
>index b9e8a05df74..132fd884401 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/fileutils.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/fileutils.h
>@@ -42,6 +42,7 @@ class Pathname;
> 
> class DirectoryIterator {
>   friend class Filesystem;
>+
>  public:
>   // Constructor
>   DirectoryIterator();
>@@ -52,7 +53,7 @@ class DirectoryIterator {
>   // dir is the directory to traverse
>   // returns true if the directory exists and is valid
>   // The iterator will point to the first entry in the directory
>-  virtual bool Iterate(const Pathname &path);
>+  virtual bool Iterate(const Pathname& path);
> 
>   // Advances to the next file
>   // returns true if there were more files in the directory.
>@@ -70,8 +71,8 @@ class DirectoryIterator {
>   WIN32_FIND_DATA data_;
>   HANDLE handle_;
> #else
>-  DIR *dir_;
>-  struct dirent *dirent_;
>+  DIR* dir_;
>+  struct dirent* dirent_;
>   struct stat stat_;
> #endif
> };
>@@ -83,12 +84,12 @@ class FilesystemInterface {
>   // This will attempt to delete the path located at filename.
>   // It DCHECKs and returns false if the path points to a folder or a
>   // non-existent file.
>-  virtual bool DeleteFile(const Pathname &filename) = 0;
>+  virtual bool DeleteFile(const Pathname& filename) = 0;
> 
>   // This moves a file from old_path to new_path, where "old_path" is a
>   // plain file. This DCHECKs and returns false if old_path points to a
>   // directory, and returns true if the function succeeds.
>-  virtual bool MoveFile(const Pathname &old_path, const Pathname &new_path) = 0;
>+  virtual bool MoveFile(const Pathname& old_path, const Pathname& new_path) = 0;
> 
>   // Returns true if pathname refers to a directory
>   virtual bool IsFolder(const Pathname& pathname) = 0;
>@@ -102,46 +103,28 @@ class FilesystemInterface {
> 
> class Filesystem {
>  public:
>-  static FilesystemInterface *default_filesystem() {
>-    RTC_DCHECK(default_filesystem_);
>-    return default_filesystem_;
>-  }
>-
>-  static void set_default_filesystem(FilesystemInterface *filesystem) {
>-    default_filesystem_ = filesystem;
>-  }
>-
>-  static FilesystemInterface *swap_default_filesystem(
>-      FilesystemInterface *filesystem) {
>-    FilesystemInterface *cur = default_filesystem_;
>-    default_filesystem_ = filesystem;
>-    return cur;
>-  }
>-
>-  static bool DeleteFile(const Pathname &filename) {
>-    return EnsureDefaultFilesystem()->DeleteFile(filename);
>+  static bool DeleteFile(const Pathname& filename) {
>+    return GetFilesystem()->DeleteFile(filename);
>   }
> 
>-  static bool MoveFile(const Pathname &old_path, const Pathname &new_path) {
>-    return EnsureDefaultFilesystem()->MoveFile(old_path, new_path);
>+  static bool MoveFile(const Pathname& old_path, const Pathname& new_path) {
>+    return GetFilesystem()->MoveFile(old_path, new_path);
>   }
> 
>   static bool IsFolder(const Pathname& pathname) {
>-    return EnsureDefaultFilesystem()->IsFolder(pathname);
>+    return GetFilesystem()->IsFolder(pathname);
>   }
> 
>-  static bool IsFile(const Pathname &pathname) {
>-    return EnsureDefaultFilesystem()->IsFile(pathname);
>+  static bool IsFile(const Pathname& pathname) {
>+    return GetFilesystem()->IsFile(pathname);
>   }
> 
>   static bool GetFileSize(const Pathname& path, size_t* size) {
>-    return EnsureDefaultFilesystem()->GetFileSize(path, size);
>+    return GetFilesystem()->GetFileSize(path, size);
>   }
> 
>  private:
>-  static FilesystemInterface* default_filesystem_;
>-
>-  static FilesystemInterface *EnsureDefaultFilesystem();
>+  static FilesystemInterface* GetFilesystem();
>   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(Filesystem);
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/firewallsocketserver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/firewallsocketserver.cc
>index 60f45ed9d21..2a28124d236 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/firewallsocketserver.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/firewallsocketserver.cc
>@@ -21,8 +21,7 @@ namespace rtc {
> class FirewallSocket : public AsyncSocketAdapter {
>  public:
>   FirewallSocket(FirewallSocketServer* server, AsyncSocket* socket, int type)
>-    : AsyncSocketAdapter(socket), server_(server), type_(type) {
>-  }
>+      : AsyncSocketAdapter(socket), server_(server), type_(type) {}
> 
>   int Bind(const SocketAddress& addr) override {
>     if (!server_->IsBindableIp(addr.ipaddr())) {
>@@ -117,9 +116,11 @@ class FirewallSocket : public AsyncSocketAdapter {
> FirewallSocketServer::FirewallSocketServer(SocketServer* server,
>                                            FirewallManager* manager,
>                                            bool should_delete_server)
>-    : server_(server), manager_(manager),
>+    : server_(server),
>+      manager_(manager),
>       should_delete_server_(should_delete_server),
>-      udp_sockets_enabled_(true), tcp_sockets_enabled_(true),
>+      udp_sockets_enabled_(true),
>+      tcp_sockets_enabled_(true),
>       tcp_listen_enabled_(true) {
>   if (manager_)
>     manager_->AddServer(this);
>@@ -135,7 +136,8 @@ FirewallSocketServer::~FirewallSocketServer() {
>   }
> }
> 
>-void FirewallSocketServer::AddRule(bool allow, FirewallProtocol p,
>+void FirewallSocketServer::AddRule(bool allow,
>+                                   FirewallProtocol p,
>                                    FirewallDirection d,
>                                    const SocketAddress& addr) {
>   SocketAddress any;
>@@ -147,8 +149,8 @@ void FirewallSocketServer::AddRule(bool allow, FirewallProtocol p,
>   }
> }
> 
>-
>-void FirewallSocketServer::AddRule(bool allow, FirewallProtocol p,
>+void FirewallSocketServer::AddRule(bool allow,
>+                                   FirewallProtocol p,
>                                    const SocketAddress& src,
>                                    const SocketAddress& dst) {
>   Rule r;
>@@ -196,18 +198,10 @@ bool FirewallSocketServer::IsBindableIp(const rtc::IPAddress& ip) {
>          unbindable_ips_.end();
> }
> 
>-Socket* FirewallSocketServer::CreateSocket(int type) {
>-  return CreateSocket(AF_INET, type);
>-}
>-
> Socket* FirewallSocketServer::CreateSocket(int family, int type) {
>   return WrapSocket(server_->CreateAsyncSocket(family, type), type);
> }
> 
>-AsyncSocket* FirewallSocketServer::CreateAsyncSocket(int type) {
>-  return CreateAsyncSocket(AF_INET, type);
>-}
>-
> AsyncSocket* FirewallSocketServer::CreateAsyncSocket(int family, int type) {
>   return WrapSocket(server_->CreateAsyncSocket(family, type), type);
> }
>@@ -225,8 +219,7 @@ void FirewallSocketServer::WakeUp() {
> }
> 
> AsyncSocket* FirewallSocketServer::WrapSocket(AsyncSocket* sock, int type) {
>-  if (!sock ||
>-      (type == SOCK_STREAM && !tcp_sockets_enabled_) ||
>+  if (!sock || (type == SOCK_STREAM && !tcp_sockets_enabled_) ||
>       (type == SOCK_DGRAM && !udp_sockets_enabled_)) {
>     RTC_LOG(LS_VERBOSE) << "FirewallSocketServer socket creation denied";
>     delete sock;
>@@ -235,8 +228,7 @@ AsyncSocket* FirewallSocketServer::WrapSocket(AsyncSocket* sock, int type) {
>   return new FirewallSocket(this, sock, type);
> }
> 
>-FirewallManager::FirewallManager() {
>-}
>+FirewallManager::FirewallManager() {}
> 
> FirewallManager::~FirewallManager() {
>   RTC_DCHECK(servers_.empty());
>@@ -253,19 +245,21 @@ void FirewallManager::RemoveServer(FirewallSocketServer* server) {
>                  servers_.end());
> }
> 
>-void FirewallManager::AddRule(bool allow, FirewallProtocol p,
>-                              FirewallDirection d, const SocketAddress& addr) {
>+void FirewallManager::AddRule(bool allow,
>+                              FirewallProtocol p,
>+                              FirewallDirection d,
>+                              const SocketAddress& addr) {
>   CritScope scope(&crit_);
>-  for (std::vector<FirewallSocketServer*>::const_iterator it =
>-      servers_.begin(); it != servers_.end(); ++it) {
>+  for (std::vector<FirewallSocketServer*>::const_iterator it = servers_.begin();
>+       it != servers_.end(); ++it) {
>     (*it)->AddRule(allow, p, d, addr);
>   }
> }
> 
> void FirewallManager::ClearRules() {
>   CritScope scope(&crit_);
>-  for (std::vector<FirewallSocketServer*>::const_iterator it =
>-      servers_.begin(); it != servers_.end(); ++it) {
>+  for (std::vector<FirewallSocketServer*>::const_iterator it = servers_.begin();
>+       it != servers_.end(); ++it) {
>     (*it)->ClearRules();
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/firewallsocketserver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/firewallsocketserver.h
>index 9f049465727..0e96823ff66 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/firewallsocketserver.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/firewallsocketserver.h
>@@ -48,15 +48,19 @@ class FirewallSocketServer : public SocketServer {
>   void set_tcp_listen_enabled(bool enabled) { tcp_listen_enabled_ = enabled; }
> 
>   // Rules govern the behavior of Connect/Accept/Send/Recv attempts.
>-  void AddRule(bool allow, FirewallProtocol p = FP_ANY,
>+  void AddRule(bool allow,
>+               FirewallProtocol p = FP_ANY,
>                FirewallDirection d = FD_ANY,
>                const SocketAddress& addr = SocketAddress());
>-  void AddRule(bool allow, FirewallProtocol p,
>-               const SocketAddress& src, const SocketAddress& dst);
>+  void AddRule(bool allow,
>+               FirewallProtocol p,
>+               const SocketAddress& src,
>+               const SocketAddress& dst);
>   void ClearRules();
> 
>   bool Check(FirewallProtocol p,
>-             const SocketAddress& src, const SocketAddress& dst);
>+             const SocketAddress& src,
>+             const SocketAddress& dst);
> 
>   // Set the IP addresses for which Bind will fail. By default this list is
>   // empty. This can be used to simulate a real OS that refuses to bind to
>@@ -68,22 +72,19 @@ class FirewallSocketServer : public SocketServer {
>   void SetUnbindableIps(const std::vector<rtc::IPAddress>& unbindable_ips);
>   bool IsBindableIp(const rtc::IPAddress& ip);
> 
>-  Socket* CreateSocket(int type) override;
>   Socket* CreateSocket(int family, int type) override;
>-
>-  AsyncSocket* CreateAsyncSocket(int type) override;
>   AsyncSocket* CreateAsyncSocket(int family, int type) override;
> 
>   void SetMessageQueue(MessageQueue* queue) override;
>   bool Wait(int cms, bool process_io) override;
>   void WakeUp() override;
> 
>-  Socket * WrapSocket(Socket * sock, int type);
>-  AsyncSocket * WrapSocket(AsyncSocket * sock, int type);
>+  Socket* WrapSocket(Socket* sock, int type);
>+  AsyncSocket* WrapSocket(AsyncSocket* sock, int type);
> 
>  private:
>-  SocketServer * server_;
>-  FirewallManager * manager_;
>+  SocketServer* server_;
>+  FirewallManager* manager_;
>   CriticalSection crit_;
>   struct Rule {
>     bool allow;
>@@ -107,17 +108,18 @@ class FirewallManager {
>   FirewallManager();
>   ~FirewallManager();
> 
>-  void AddServer(FirewallSocketServer * server);
>-  void RemoveServer(FirewallSocketServer * server);
>+  void AddServer(FirewallSocketServer* server);
>+  void RemoveServer(FirewallSocketServer* server);
> 
>-  void AddRule(bool allow, FirewallProtocol p = FP_ANY,
>+  void AddRule(bool allow,
>+               FirewallProtocol p = FP_ANY,
>                FirewallDirection d = FD_ANY,
>                const SocketAddress& addr = SocketAddress());
>   void ClearRules();
> 
>  private:
>   CriticalSection crit_;
>-  std::vector<FirewallSocketServer *> servers_;
>+  std::vector<FirewallSocketServer*> servers_;
> };
> 
> }  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/flags.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/flags.cc
>index ac92658e976..4f294594bd4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/flags.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/flags.cc
>@@ -18,16 +18,39 @@
> #include "rtc_base/stringutils.h"
> 
> #if defined(WEBRTC_WIN)
>+// clang-format off
>+// clang formating would change include order.
> #include <windows.h>
>-#include <shellapi.h>
>+#include <shellapi.h> // must come after windows.h
>+// clang-format on
> #endif
> 
>+namespace {
>+bool FlagEq(const char* arg, const char* flag) {
>+  // Compare two flags for equality.
>+  // 'arg' is the name of a flag passed via the command line and 'flag' is the
>+  // name of a flag defined with the DEFINE_* macros.
>+  // We compare the flags for equality, considering hyphens (-) and
>+  // underscores (_) to be equivalent, so that --flag-name and --flag_name both
>+  // match with --flag_name.
>+  while (*arg != '\0' && (*arg == *flag || (*arg == '-' && *flag == '_'))) {
>+    ++arg;
>+    ++flag;
>+  }
>+  return *arg == '\0' && *flag == '\0';
>+}
>+}  // namespace
>+
> namespace rtc {
> // -----------------------------------------------------------------------------
> // Implementation of Flag
> 
>-Flag::Flag(const char* file, const char* name, const char* comment,
>-           Type type, void* variable, FlagValue default__)
>+Flag::Flag(const char* file,
>+           const char* name,
>+           const char* comment,
>+           Type type,
>+           void* variable,
>+           FlagValue default__)
>     : file_(file),
>       name_(name),
>       comment_(comment),
>@@ -37,7 +60,6 @@ Flag::Flag(const char* file, const char* name, const char* comment,
>   FlagList::Register(this);
> }
> 
>-
> void Flag::SetToDefault() {
>   // Note that we cannot simply do '*variable_ = default_;' since
>   // flag variables are not really of type FlagValue and thus may
>@@ -61,18 +83,20 @@ void Flag::SetToDefault() {
>   RTC_FATAL() << "unreachable code";
> }
> 
>-
> static const char* Type2String(Flag::Type type) {
>   switch (type) {
>-    case Flag::BOOL: return "bool";
>-    case Flag::INT: return "int";
>-    case Flag::FLOAT: return "float";
>-    case Flag::STRING: return "string";
>+    case Flag::BOOL:
>+      return "bool";
>+    case Flag::INT:
>+      return "int";
>+    case Flag::FLOAT:
>+      return "float";
>+    case Flag::STRING:
>+      return "string";
>   }
>   RTC_FATAL() << "unreachable code";
> }
> 
>-
> static void PrintFlagValue(Flag::Type type, FlagValue* p) {
>   switch (type) {
>     case Flag::BOOL:
>@@ -91,10 +115,9 @@ static void PrintFlagValue(Flag::Type type, FlagValue* p) {
>   RTC_FATAL() << "unreachable code";
> }
> 
>-
> void Flag::Print(bool print_current_value) {
>   printf("  --%s (%s)  type: %s  default: ", name_, comment_,
>-          Type2String(type_));
>+         Type2String(type_));
>   PrintFlagValue(type_, &default_);
>   if (print_current_value) {
>     printf("  current value: ");
>@@ -103,7 +126,6 @@ void Flag::Print(bool print_current_value) {
>   printf("\n");
> }
> 
>-
> // -----------------------------------------------------------------------------
> // Implementation of FlagList
> 
>@@ -128,18 +150,18 @@ void FlagList::Print(const char* file, bool print_current_value) {
>   }
> }
> 
>-
> Flag* FlagList::Lookup(const char* name) {
>   Flag* f = list_;
>-  while (f != nullptr && strcmp(name, f->name()) != 0)
>+  while (f != nullptr && !FlagEq(name, f->name()))
>     f = f->next();
>   return f;
> }
> 
>-
> void FlagList::SplitArgument(const char* arg,
>-                             char* buffer, int buffer_size,
>-                             const char** name, const char** value,
>+                             char* buffer,
>+                             int buffer_size,
>+                             const char** name,
>+                             const char** value,
>                              bool* is_bool) {
>   *name = nullptr;
>   *value = nullptr;
>@@ -174,8 +196,8 @@ void FlagList::SplitArgument(const char* arg,
>   }
> }
> 
>-
>-int FlagList::SetFlagsFromCommandLine(int* argc, const char** argv,
>+int FlagList::SetFlagsFromCommandLine(int* argc,
>+                                      const char** argv,
>                                       bool remove_flags) {
>   // parse arguments
>   for (int i = 1; i < *argc; /* see below */) {
>@@ -202,14 +224,14 @@ int FlagList::SetFlagsFromCommandLine(int* argc, const char** argv,
>         if (i < *argc) {
>           value = argv[i++];
>         } else {
>-          fprintf(stderr, "Error: missing value for flag %s of type %s\n",
>-            arg, Type2String(flag->type()));
>+          fprintf(stderr, "Error: missing value for flag %s of type %s\n", arg,
>+                  Type2String(flag->type()));
>           return j;
>         }
>       }
> 
>       // set the flag
>-      char empty[] = { '\0' };
>+      char empty[] = {'\0'};
>       char* endp = empty;
>       switch (flag->type()) {
>         case Flag::BOOL:
>@@ -229,8 +251,8 @@ int FlagList::SetFlagsFromCommandLine(int* argc, const char** argv,
>       // handle errors
>       if ((flag->type() == Flag::BOOL && value != nullptr) ||
>           (flag->type() != Flag::BOOL && is_bool) || *endp != '\0') {
>-        fprintf(stderr, "Error: illegal value for flag %s of type %s\n",
>-          arg, Type2String(flag->type()));
>+        fprintf(stderr, "Error: illegal value for flag %s of type %s\n", arg,
>+                Type2String(flag->type()));
>         return j;
>       }
> 
>@@ -270,15 +292,15 @@ void FlagList::Register(Flag* flag) {
> WindowsCommandLineArguments::WindowsCommandLineArguments() {
>   // start by getting the command line.
>   LPTSTR command_line = ::GetCommandLine();
>-   // now, convert it to a list of wide char strings.
>-  LPWSTR *wide_argv = ::CommandLineToArgvW(command_line, &argc_);
>+  // now, convert it to a list of wide char strings.
>+  LPWSTR* wide_argv = ::CommandLineToArgvW(command_line, &argc_);
>   // now allocate an array big enough to hold that many string pointers.
>   argv_ = new char*[argc_];
> 
>   // iterate over the returned wide strings;
>-  for(int i = 0; i < argc_; ++i) {
>+  for (int i = 0; i < argc_; ++i) {
>     std::string s = rtc::ToUtf8(wide_argv[i], wcslen(wide_argv[i]));
>-    char *buffer = new char[s.length() + 1];
>+    char* buffer = new char[s.length() + 1];
>     rtc::strcpyn(buffer, s.length() + 1, s.c_str());
> 
>     // make sure the argv array has the right string at this point.
>@@ -289,7 +311,7 @@ WindowsCommandLineArguments::WindowsCommandLineArguments() {
> 
> WindowsCommandLineArguments::~WindowsCommandLineArguments() {
>   // need to free each string in the array, and then the array.
>-  for(int i = 0; i < argc_; i++) {
>+  for (int i = 0; i < argc_; i++) {
>     delete[] argv_[i];
>   }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/flags.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/flags.h
>index 0515d1140e9..33f6e5bda1d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/flags.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/flags.h
>@@ -8,7 +8,6 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-
> // Originally comes from shared/commandlineflags/flags.h
> 
> // Flags are defined and declared using DEFINE_xxx and DECLARE_xxx macros,
>@@ -65,23 +64,26 @@ union FlagValue {
>   const char* s;
> };
> 
>-
> // Each flag can be accessed programmatically via a Flag object.
> class Flag {
>  public:
>   enum Type { BOOL, INT, FLOAT, STRING };
> 
>   // Internal use only.
>-  Flag(const char* file, const char* name, const char* comment,
>-       Type type, void* variable, FlagValue default_);
>+  Flag(const char* file,
>+       const char* name,
>+       const char* comment,
>+       Type type,
>+       void* variable,
>+       FlagValue default_);
> 
>   // General flag information
>-  const char* file() const  { return file_; }
>-  const char* name() const  { return name_; }
>-  const char* comment() const  { return comment_; }
>+  const char* file() const { return file_; }
>+  const char* name() const { return name_; }
>+  const char* comment() const { return comment_; }
> 
>   // Flag type
>-  Type type() const  { return type_; }
>+  Type type() const { return type_; }
> 
>   // Flag variables
>   bool* bool_variable() const {
>@@ -129,7 +131,7 @@ class Flag {
>   void SetToDefault();
> 
>   // Iteration support
>-  Flag* next() const  { return next_; }
>+  Flag* next() const { return next_; }
> 
>   // Prints flag information. The current flag value is only printed
>   // if print_current_value is set.
>@@ -149,28 +151,19 @@ class Flag {
>   friend class FlagList;  // accesses next_
> };
> 
>-
> // Internal use only.
>-
>-#define DEFINE_FLAG(type, c_type, name, default, comment)    \
>-  static std::pair<std::reference_wrapper<c_type>, std::reference_wrapper<rtc::Flag>> name() { \
>-    /* define and initialize the flag */                     \
>-    static c_type FLAG_##name = (default);                   \
>-    /* register the flag */                                  \
>-    static rtc::Flag Flag_##name(__FILE__, #name, (comment), \
>-      rtc::Flag::type, &FLAG_##name,                         \
>-      rtc::FlagValue::New_##type(default));                  \
>-    return std::make_pair<std::reference_wrapper<c_type>, std::reference_wrapper<rtc::Flag>>(FLAG_##name, Flag_##name); \
>-  }                       \
>-  c_type& FLAG_##name() { \
>-    return name().first;  \
>-  }
>+#define DEFINE_FLAG(type, c_type, name, default, comment)                   \
>+  /* define and initialize the flag */                                      \
>+  c_type FLAG_##name = (default);                                           \
>+  /* register the flag */                                                   \
>+  static rtc::Flag Flag_##name(__FILE__, #name, (comment), rtc::Flag::type, \
>+                               &FLAG_##name,                                \
>+                               rtc::FlagValue::New_##type(default))
> 
> // Internal use only.
>-#define DECLARE_FLAG(c_type, name)              \
>-  /* declare the flag getter */                 \
>-  c_type& FLAG_##name();
>-
>+#define DECLARE_FLAG(c_type, name) \
>+  /* declare the external flag */  \
>+  extern c_type FLAG_##name
> 
> // Use the following macros to define a new flag:
> #define DEFINE_bool(name, default, comment) \
>@@ -182,13 +175,11 @@ class Flag {
> #define DEFINE_string(name, default, comment) \
>   DEFINE_FLAG(STRING, const char*, name, default, comment)
> 
>-
> // Use the following macros to declare a flag defined elsewhere:
>-#define DECLARE_bool(name)  DECLARE_FLAG(bool, name)
>-#define DECLARE_int(name)  DECLARE_FLAG(int, name)
>-#define DECLARE_float(name)  DECLARE_FLAG(double, name)
>-#define DECLARE_string(name)  DECLARE_FLAG(const char*, name)
>-
>+#define DECLARE_bool(name) DECLARE_FLAG(bool, name)
>+#define DECLARE_int(name) DECLARE_FLAG(int, name)
>+#define DECLARE_float(name) DECLARE_FLAG(double, name)
>+#define DECLARE_string(name) DECLARE_FLAG(const char*, name)
> 
> // The global list of all flags.
> class FlagList {
>@@ -196,7 +187,7 @@ class FlagList {
>   FlagList();
> 
>   // The null-terminated list of all flags. Traverse with Flag::next().
>-  static Flag* list()  { return list_; }
>+  static Flag* list() { return list_; }
> 
>   // If file != nullptr, prints information for all flags defined in file;
>   // otherwise prints information for all flags in all files. The current flag
>@@ -211,8 +202,10 @@ class FlagList {
>   // if the arg started with "-no" or "--no". The buffer may be used to NUL-
>   // terminate the name, it must be large enough to hold any possible name.
>   static void SplitArgument(const char* arg,
>-                            char* buffer, int buffer_size,
>-                            const char** name, const char** value,
>+                            char* buffer,
>+                            int buffer_size,
>+                            const char** name,
>+                            const char** value,
>                             bool* is_bool);
> 
>   // Set the flag values by parsing the command line. If remove_flags
>@@ -259,10 +252,11 @@ class WindowsCommandLineArguments {
>   ~WindowsCommandLineArguments();
> 
>   int argc() { return argc_; }
>-  char **argv() { return argv_; }
>+  char** argv() { return argv_; }
>+
>  private:
>   int argc_;
>-  char **argv_;
>+  char** argv_;
> 
>  private:
>   RTC_DISALLOW_COPY_AND_ASSIGN(WindowsCommandLineArguments);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/format_macros.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/format_macros.h
>index d252a94ccaf..7252f2ea741 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/format_macros.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/format_macros.h
>@@ -24,8 +24,6 @@
> //   printf("xyz: %" PRIuS, size);
> // The "u" in the macro corresponds to %u, and S is for "size".
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>-
> #if defined(WEBRTC_POSIX)
> 
> #if (defined(_INTTYPES_H) || defined(_INTTYPES_H_)) && !defined(PRId64)
>@@ -39,6 +37,8 @@
> 
> #include <inttypes.h>
> 
>+#include "rtc_base/system/arch.h"
>+
> #if !defined(PRIuS)
> #define PRIuS "zu"
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/function_view_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/function_view_unittest.cc
>index 98f78d6680e..d0cfeb30883 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/function_view_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/function_view_unittest.cc
>@@ -73,7 +73,7 @@ TEST(FunctionViewTest, Null) {
>   EXPECT_FALSE(rtc::FunctionView<int()>(nullptr));
> 
>   // This calls the constructor for function pointers.
>-  EXPECT_FALSE(rtc::FunctionView<int()>(reinterpret_cast<int(*)()>(0)));
>+  EXPECT_FALSE(rtc::FunctionView<int()>(reinterpret_cast<int (*)()>(0)));
> }
> 
> // Ensure that FunctionView handles move-only arguments and return values.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/gtest_prod_util.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/gtest_prod_util.h
>index dc9679f49e9..0661cd7bd93 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/gtest_prod_util.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/gtest_prod_util.h
>@@ -13,8 +13,8 @@
> 
> // Define our own version of FRIEND_TEST here rather than including
> // gtest_prod.h to avoid depending on any part of GTest in production code.
>-#define FRIEND_TEST_WEBRTC(test_case_name, test_name)\
>-friend class test_case_name##_##test_name##_Test
>+#define FRIEND_TEST_WEBRTC(test_case_name, test_name) \
>+  friend class test_case_name##_##test_name##_Test
> 
> // This file is a plain copy of Chromium's base/gtest_prod_util.h.
> //
>@@ -30,9 +30,9 @@ friend class test_case_name##_##test_name##_Test
> //   FRIEND_TEST_ALL_PREFIXES(MyClassTest, MyMethod);
> // };
> #define FRIEND_TEST_ALL_PREFIXES(test_case_name, test_name) \
>-  FRIEND_TEST_WEBRTC(test_case_name, test_name); \
>+  FRIEND_TEST_WEBRTC(test_case_name, test_name);            \
>   FRIEND_TEST_WEBRTC(test_case_name, DISABLED_##test_name); \
>-  FRIEND_TEST_WEBRTC(test_case_name, FLAKY_##test_name); \
>+  FRIEND_TEST_WEBRTC(test_case_name, FLAKY_##test_name);    \
>   FRIEND_TEST_WEBRTC(test_case_name, FAILS_##test_name)
> 
> #endif  // RTC_BASE_GTEST_PROD_UTIL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/gunit.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/gunit.cc
>new file mode 100644
>index 00000000000..0dd8f1230d3
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/gunit.cc
>@@ -0,0 +1,41 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "rtc_base/gunit.h"
>+
>+#include <string>
>+
>+#include "rtc_base/stringutils.h"
>+
>+::testing::AssertionResult AssertStartsWith(const char* str_expr,
>+                                            const char* prefix_expr,
>+                                            const std::string& str,
>+                                            const std::string& prefix) {
>+  if (rtc::starts_with(str.c_str(), prefix.c_str())) {
>+    return ::testing::AssertionSuccess();
>+  } else {
>+    return ::testing::AssertionFailure()
>+           << str_expr << "\nwhich is\n\"" << str << "\"\ndoes not start with\n"
>+           << prefix_expr << "\nwhich is\n\"" << prefix << "\"";
>+  }
>+}
>+
>+::testing::AssertionResult AssertStringContains(const char* str_expr,
>+                                                const char* substr_expr,
>+                                                const std::string& str,
>+                                                const std::string& substr) {
>+  if (str.find(substr) != std::string::npos) {
>+    return ::testing::AssertionSuccess();
>+  } else {
>+    return ::testing::AssertionFailure()
>+           << str_expr << "\nwhich is\n\"" << str << "\"\ndoes not contain\n"
>+           << substr_expr << "\nwhich is\n\"" << substr << "\"";
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/gunit.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/gunit.h
>index 9ee3c72ad5f..910fbf39bae 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/gunit.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/gunit.h
>@@ -14,11 +14,7 @@
> #include "rtc_base/fakeclock.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/thread.h"
>-#if defined(GTEST_RELATIVE_PATH)
> #include "test/gtest.h"
>-#else
>-#include "testing/base/public/gunit.h"
>-#endif
> 
> // Wait until "ex" is true, or "timeout" expires.
> #define WAIT(ex, timeout)                                       \
>@@ -101,23 +97,23 @@
> // Wait until "ex" is true, or "timeout" expires, using fake clock where
> // messages are processed every millisecond.
> // TODO(pthatcher): Allow tests to control how many milliseconds to advance.
>-#define SIMULATED_WAIT(ex, timeout, clock)                    \
>-  for (int64_t start = rtc::TimeMillis();                     \
>-       !(ex) && rtc::TimeMillis() < start + (timeout);) {     \
>-    (clock).AdvanceTime(rtc::TimeDelta::FromMilliseconds(1)); \
>+#define SIMULATED_WAIT(ex, timeout, clock)                \
>+  for (int64_t start = rtc::TimeMillis();                 \
>+       !(ex) && rtc::TimeMillis() < start + (timeout);) { \
>+    (clock).AdvanceTime(webrtc::TimeDelta::ms(1));        \
>   }
> 
> // This returns the result of the test in res, so that we don't re-evaluate
> // the expression in the XXXX_WAIT macros below, since that causes problems
> // when the expression is only true the first time you check it.
>-#define SIMULATED_WAIT_(ex, timeout, res, clock)                \
>-  do {                                                          \
>-    int64_t start = rtc::TimeMillis();                          \
>-    res = (ex);                                                 \
>-    while (!res && rtc::TimeMillis() < start + (timeout)) {     \
>-      (clock).AdvanceTime(rtc::TimeDelta::FromMilliseconds(1)); \
>-      res = (ex);                                               \
>-    }                                                           \
>+#define SIMULATED_WAIT_(ex, timeout, res, clock)            \
>+  do {                                                      \
>+    int64_t start = rtc::TimeMillis();                      \
>+    res = (ex);                                             \
>+    while (!res && rtc::TimeMillis() < start + (timeout)) { \
>+      (clock).AdvanceTime(webrtc::TimeDelta::ms(1));        \
>+      res = (ex);                                           \
>+    }                                                       \
>   } while (0)
> 
> // The typical EXPECT_XXXX, but done until true or a timeout with a fake clock.
>@@ -157,4 +153,16 @@
>   } else                                                 \
>     GTEST_CONCAT_TOKEN_(gunit_label_, __LINE__) : ASSERT_EQ(v1, v2)
> 
>+// Usage: EXPECT_PRED_FORMAT2(AssertStartsWith, str, "prefix");
>+testing::AssertionResult AssertStartsWith(const char* str_expr,
>+                                          const char* prefix_expr,
>+                                          const std::string& str,
>+                                          const std::string& prefix);
>+
>+// Usage: EXPECT_PRED_FORMAT2(AssertStringContains, str, "substring");
>+testing::AssertionResult AssertStringContains(const char* str_expr,
>+                                              const char* substr_expr,
>+                                              const std::string& str,
>+                                              const std::string& substr);
>+
> #endif  // RTC_BASE_GUNIT_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/gunit_prod.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/gunit_prod.h
>index ae4157d3383..dae2084d155 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/gunit_prod.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/gunit_prod.h
>@@ -15,10 +15,8 @@
> // Android doesn't use gtest at all, so anything that relies on gtest should
> // check this define first.
> #define NO_GTEST
>-#elif defined (GTEST_RELATIVE_PATH)
>-#include "gtest/gtest_prod.h"
> #else
>-#include "testing/base/gunit_prod.h"
>+#include "gtest/gtest_prod.h"
> #endif
> 
> #endif  // RTC_BASE_GUNIT_PROD_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/helpers.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/helpers.cc
>index b0c88566154..b9372b99749 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/helpers.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/helpers.cc
>@@ -15,10 +15,9 @@
> 
> #include <openssl/rand.h>
> 
>-#include "rtc_base/base64.h"
>-#include "rtc_base/basictypes.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>+#include "rtc_base/third_party/base64/base64.h"
> #include "rtc_base/timeutils.h"
> 
> // Protect against max macro inclusion.
>@@ -48,10 +47,8 @@ class SecureRandomGenerator : public RandomGenerator {
> // A test random generator, for predictable output.
> class TestRandomGenerator : public RandomGenerator {
>  public:
>-  TestRandomGenerator() : seed_(7) {
>-  }
>-  ~TestRandomGenerator() override {
>-  }
>+  TestRandomGenerator() : seed_(7) {}
>+  ~TestRandomGenerator() override {}
>   bool Init(const void* seed, size_t len) override { return true; }
>   bool Generate(void* buf, size_t len) override {
>     for (size_t i = 0; i < len; ++i) {
>@@ -85,8 +82,9 @@ static const char kUuidDigit17[4] = {'8', '9', 'a', 'b'};
> // This round about way of creating a global RNG is to safe-guard against
> // indeterminant static initialization order.
> std::unique_ptr<RandomGenerator>& GetGlobalRng() {
>-  RTC_DEFINE_STATIC_LOCAL(std::unique_ptr<RandomGenerator>, global_rng,
>-                          (new SecureRandomGenerator()));
>+  static std::unique_ptr<RandomGenerator>& global_rng =
>+      *new std::unique_ptr<RandomGenerator>(new SecureRandomGenerator());
>+
>   return global_rng;
> }
> 
>@@ -123,8 +121,9 @@ std::string CreateRandomString(size_t len) {
> }
> 
> static bool CreateRandomString(size_t len,
>-                        const char* table, int table_size,
>-                        std::string* str) {
>+                               const char* table,
>+                               int table_size,
>+                               std::string* str) {
>   str->clear();
>   // Avoid biased modulo division below.
>   if (256 % table_size) {
>@@ -147,10 +146,11 @@ bool CreateRandomString(size_t len, std::string* str) {
>   return CreateRandomString(len, kBase64, 64, str);
> }
> 
>-bool CreateRandomString(size_t len, const std::string& table,
>+bool CreateRandomString(size_t len,
>+                        const std::string& table,
>                         std::string* str) {
>-  return CreateRandomString(len, table.c_str(),
>-                            static_cast<int>(table.size()), str);
>+  return CreateRandomString(len, table.c_str(), static_cast<int>(table.size()),
>+                            str);
> }
> 
> bool CreateRandomData(size_t length, std::string* data) {
>@@ -215,4 +215,8 @@ double CreateRandomDouble() {
>                              std::numeric_limits<double>::epsilon());
> }
> 
>+double GetNextMovingAverage(double prev_average, double cur, double ratio) {
>+  return (ratio * prev_average + cur) / (ratio + 1);
>+}
>+
> }  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/helpers.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/helpers.h
>index e1d2ebb2aa2..d3b09cfd48f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/helpers.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/helpers.h
>@@ -12,7 +12,6 @@
> #define RTC_BASE_HELPERS_H_
> 
> #include <string>
>-#include "rtc_base/basictypes.h"
> 
> namespace rtc {
> 
>@@ -37,7 +36,8 @@ bool CreateRandomString(size_t length, std::string* str);
> // number generator failed.
> // For ease of implementation, the function requires that the table
> // size evenly divide 256; otherwise, it returns false.
>-bool CreateRandomString(size_t length, const std::string& table,
>+bool CreateRandomString(size_t length,
>+                        const std::string& table,
>                         std::string* str);
> 
> // Generates (cryptographically) random data of the given length.
>@@ -59,6 +59,10 @@ uint32_t CreateRandomNonZeroId();
> // Generates a random double between 0.0 (inclusive) and 1.0 (exclusive).
> double CreateRandomDouble();
> 
>+// Compute moving average with the given ratio between the previous average
>+// value and the current value.
>+double GetNextMovingAverage(double prev_average, double cur, double ratio);
>+
> }  // namespace rtc
> 
> #endif  // RTC_BASE_HELPERS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/helpers_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/helpers_unittest.cc
>index 9fa16a5f468..3e8729ce1f5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/helpers_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/helpers_unittest.cc
>@@ -81,10 +81,12 @@ TEST_F(RandomTest, TestCreateRandomForTest) {
>   std::string random;
>   EXPECT_TRUE(CreateRandomData(kRandomDataLength, &random));
>   EXPECT_EQ(kRandomDataLength, random.size());
>-  Buffer expected("\xbd\x52\x2a\x4b\x97\x93\x2f\x1c"
>+  Buffer expected(
>+      "\xbd\x52\x2a\x4b\x97\x93\x2f\x1c"
>       "\xc4\x72\xab\xa2\x88\x68\x3e\xcc"
>       "\xa3\x8d\xaf\x13\x3b\xbc\x83\xbb"
>-      "\x16\xf1\xcf\x56\x0c\xf5\x4a\x8b", kRandomDataLength);
>+      "\x16\xf1\xcf\x56\x0c\xf5\x4a\x8b",
>+      kRandomDataLength);
>   EXPECT_EQ(0, memcmp(expected.data(), random.data(), kRandomDataLength));
> 
>   // Reset and make sure we get the same output.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/httpbase.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/httpbase.cc
>index d28d800c421..8f2869d73f7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/httpbase.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/httpbase.cc
>@@ -21,6 +21,7 @@
> #include "rtc_base/logging.h"
> #include "rtc_base/socket.h"
> #include "rtc_base/stringutils.h"
>+#include "rtc_base/system/fallthrough.h"
> #include "rtc_base/thread.h"
> 
> namespace rtc {
>@@ -35,9 +36,7 @@ bool MatchHeader(const char* str, size_t len, HttpHeader header) {
>   return (len == header_len) && (_strnicmp(str, header_str, header_len) == 0);
> }
> 
>-enum {
>-  MSG_READ
>-};
>+enum { MSG_READ };
> 
> //////////////////////////////////////////////////////////////////////
> // HttpParser
>@@ -47,19 +46,18 @@ HttpParser::HttpParser() {
>   reset();
> }
> 
>-HttpParser::~HttpParser() {
>-}
>+HttpParser::~HttpParser() {}
> 
>-void
>-HttpParser::reset() {
>+void HttpParser::reset() {
>   state_ = ST_LEADER;
>   chunked_ = false;
>   data_size_ = SIZE_UNKNOWN;
> }
> 
>-HttpParser::ProcessResult
>-HttpParser::Process(const char* buffer, size_t len, size_t* processed,
>-                    HttpError* error) {
>+HttpParser::ProcessResult HttpParser::Process(const char* buffer,
>+                                              size_t len,
>+                                              size_t* processed,
>+                                              HttpError* error) {
>   *processed = 0;
>   *error = HE_NONE;
> 
>@@ -80,7 +78,7 @@ HttpParser::Process(const char* buffer, size_t len, size_t* processed,
>       const char* line = buffer + *processed;
>       size_t len = (pos - *processed);
>       *processed = pos + 1;
>-      while ((len > 0) && isspace(static_cast<unsigned char>(line[len-1]))) {
>+      while ((len > 0) && isspace(static_cast<unsigned char>(line[len - 1]))) {
>         len -= 1;
>       }
>       ProcessResult result = ProcessLine(line, len, error);
>@@ -98,14 +96,14 @@ HttpParser::Process(const char* buffer, size_t len, size_t* processed,
>     } else {
>       size_t available = len - *processed;
>       if (available <= 0) {
>-        break; // no more data
>+        break;  // no more data
>       }
>       if ((data_size_ != SIZE_UNKNOWN) && (available > data_size_)) {
>         available = data_size_;
>       }
>       size_t read = 0;
>-      ProcessResult result = ProcessData(buffer + *processed, available, read,
>-                                         error);
>+      ProcessResult result =
>+          ProcessData(buffer + *processed, available, read, error);
>       RTC_LOG(LS_VERBOSE) << "Processed data, result: " << result
>                           << " read: " << read << " err: " << error;
> 
>@@ -122,262 +120,125 @@ HttpParser::Process(const char* buffer, size_t len, size_t* processed,
>   return PR_CONTINUE;
> }
> 
>-HttpParser::ProcessResult
>-HttpParser::ProcessLine(const char* line, size_t len, HttpError* error) {
>+HttpParser::ProcessResult HttpParser::ProcessLine(const char* line,
>+                                                  size_t len,
>+                                                  HttpError* error) {
>   RTC_LOG_F(LS_VERBOSE) << " state: " << state_
>                         << " line: " << std::string(line, len)
>                         << " len: " << len << " err: " << error;
> 
>   switch (state_) {
>-  case ST_LEADER:
>-    state_ = ST_HEADERS;
>-    return ProcessLeader(line, len, error);
>-
>-  case ST_HEADERS:
>-    if (len > 0) {
>-      const char* value = strchrn(line, len, ':');
>-      if (!value) {
>-        *error = HE_PROTOCOL;
>-        return PR_COMPLETE;
>-      }
>-      size_t nlen = (value - line);
>-      const char* eol = line + len;
>-      do {
>-        value += 1;
>-      } while ((value < eol) && isspace(static_cast<unsigned char>(*value)));
>-      size_t vlen = eol - value;
>-      if (MatchHeader(line, nlen, HH_CONTENT_LENGTH)) {
>-        // sscanf isn't safe with strings that aren't null-terminated, and there
>-        // is no guarantee that |value| is.
>-        // Create a local copy that is null-terminated.
>-        std::string value_str(value, vlen);
>-        unsigned int temp_size;
>-        if (sscanf(value_str.c_str(), "%u", &temp_size) != 1) {
>+    case ST_LEADER:
>+      state_ = ST_HEADERS;
>+      return ProcessLeader(line, len, error);
>+
>+    case ST_HEADERS:
>+      if (len > 0) {
>+        const char* value = strchrn(line, len, ':');
>+        if (!value) {
>           *error = HE_PROTOCOL;
>           return PR_COMPLETE;
>         }
>-        data_size_ = static_cast<size_t>(temp_size);
>-      } else if (MatchHeader(line, nlen, HH_TRANSFER_ENCODING)) {
>-        if ((vlen == 7) && (_strnicmp(value, "chunked", 7) == 0)) {
>-          chunked_ = true;
>-        } else if ((vlen == 8) && (_strnicmp(value, "identity", 8) == 0)) {
>-          chunked_ = false;
>-        } else {
>+        size_t nlen = (value - line);
>+        const char* eol = line + len;
>+        do {
>+          value += 1;
>+        } while ((value < eol) && isspace(static_cast<unsigned char>(*value)));
>+        size_t vlen = eol - value;
>+        if (MatchHeader(line, nlen, HH_CONTENT_LENGTH)) {
>+          // sscanf isn't safe with strings that aren't null-terminated, and
>+          // there is no guarantee that |value| is. Create a local copy that is
>+          // null-terminated.
>+          std::string value_str(value, vlen);
>+          unsigned int temp_size;
>+          if (sscanf(value_str.c_str(), "%u", &temp_size) != 1) {
>+            *error = HE_PROTOCOL;
>+            return PR_COMPLETE;
>+          }
>+          data_size_ = static_cast<size_t>(temp_size);
>+        } else if (MatchHeader(line, nlen, HH_TRANSFER_ENCODING)) {
>+          if ((vlen == 7) && (_strnicmp(value, "chunked", 7) == 0)) {
>+            chunked_ = true;
>+          } else if ((vlen == 8) && (_strnicmp(value, "identity", 8) == 0)) {
>+            chunked_ = false;
>+          } else {
>+            *error = HE_PROTOCOL;
>+            return PR_COMPLETE;
>+          }
>+        }
>+        return ProcessHeader(line, nlen, value, vlen, error);
>+      } else {
>+        state_ = chunked_ ? ST_CHUNKSIZE : ST_DATA;
>+        return ProcessHeaderComplete(chunked_, data_size_, error);
>+      }
>+      break;
>+
>+    case ST_CHUNKSIZE:
>+      if (len > 0) {
>+        char* ptr = nullptr;
>+        data_size_ = strtoul(line, &ptr, 16);
>+        if (ptr != line + len) {
>           *error = HE_PROTOCOL;
>           return PR_COMPLETE;
>         }
>+        state_ = (data_size_ == 0) ? ST_TRAILERS : ST_DATA;
>+      } else {
>+        *error = HE_PROTOCOL;
>+        return PR_COMPLETE;
>       }
>-      return ProcessHeader(line, nlen, value, vlen, error);
>-    } else {
>-      state_ = chunked_ ? ST_CHUNKSIZE : ST_DATA;
>-      return ProcessHeaderComplete(chunked_, data_size_, error);
>-    }
>-    break;
>+      break;
> 
>-  case ST_CHUNKSIZE:
>-    if (len > 0) {
>-      char* ptr = nullptr;
>-      data_size_ = strtoul(line, &ptr, 16);
>-      if (ptr != line + len) {
>+    case ST_CHUNKTERM:
>+      if (len > 0) {
>         *error = HE_PROTOCOL;
>         return PR_COMPLETE;
>+      } else {
>+        state_ = chunked_ ? ST_CHUNKSIZE : ST_DATA;
>       }
>-      state_ = (data_size_ == 0) ? ST_TRAILERS : ST_DATA;
>-    } else {
>-      *error = HE_PROTOCOL;
>-      return PR_COMPLETE;
>-    }
>-    break;
>-
>-  case ST_CHUNKTERM:
>-    if (len > 0) {
>-      *error = HE_PROTOCOL;
>-      return PR_COMPLETE;
>-    } else {
>-      state_ = chunked_ ? ST_CHUNKSIZE : ST_DATA;
>-    }
>-    break;
>+      break;
> 
>-  case ST_TRAILERS:
>-    if (len == 0) {
>-      return PR_COMPLETE;
>-    }
>-    // *error = onHttpRecvTrailer();
>-    break;
>+    case ST_TRAILERS:
>+      if (len == 0) {
>+        return PR_COMPLETE;
>+      }
>+      // *error = onHttpRecvTrailer();
>+      break;
> 
>-  default:
>-    RTC_NOTREACHED();
>-    break;
>+    default:
>+      RTC_NOTREACHED();
>+      break;
>   }
> 
>   return PR_CONTINUE;
> }
> 
>-bool
>-HttpParser::is_valid_end_of_input() const {
>+bool HttpParser::is_valid_end_of_input() const {
>   return (state_ == ST_DATA) && (data_size_ == SIZE_UNKNOWN);
> }
> 
>-void
>-HttpParser::complete(HttpError error) {
>+void HttpParser::complete(HttpError error) {
>   if (state_ < ST_COMPLETE) {
>     state_ = ST_COMPLETE;
>     OnComplete(error);
>   }
> }
> 
>-//////////////////////////////////////////////////////////////////////
>-// HttpBase::DocumentStream
>-//////////////////////////////////////////////////////////////////////
>-
>-class BlockingMemoryStream : public ExternalMemoryStream {
>-public:
>-  BlockingMemoryStream(char* buffer, size_t size)
>-  : ExternalMemoryStream(buffer, size) { }
>-
>-  StreamResult DoReserve(size_t size, int* error) override {
>-    return (buffer_length_ >= size) ? SR_SUCCESS : SR_BLOCK;
>-  }
>-};
>-
>-class HttpBase::DocumentStream : public StreamInterface {
>-public:
>-  DocumentStream(HttpBase* base) : base_(base), error_(HE_DEFAULT) { }
>-
>-  StreamState GetState() const override {
>-    if (nullptr == base_)
>-      return SS_CLOSED;
>-    if (HM_RECV == base_->mode_)
>-      return SS_OPEN;
>-    return SS_OPENING;
>-  }
>-
>-  StreamResult Read(void* buffer,
>-                    size_t buffer_len,
>-                    size_t* read,
>-                    int* error) override {
>-    if (!base_) {
>-      if (error) *error = error_;
>-      return (HE_NONE == error_) ? SR_EOS : SR_ERROR;
>-    }
>-
>-    if (HM_RECV != base_->mode_) {
>-      return SR_BLOCK;
>-    }
>-
>-    // DoReceiveLoop writes http document data to the StreamInterface* document
>-    // member of HttpData.  In this case, we want this data to be written
>-    // directly to our buffer.  To accomplish this, we wrap our buffer with a
>-    // StreamInterface, and replace the existing document with our wrapper.
>-    // When the method returns, we restore the old document.  Ideally, we would
>-    // pass our StreamInterface* to DoReceiveLoop, but due to the callbacks
>-    // of HttpParser, we would still need to store the pointer temporarily.
>-    std::unique_ptr<StreamInterface> stream(
>-        new BlockingMemoryStream(reinterpret_cast<char*>(buffer), buffer_len));
>-
>-    // Replace the existing document with our wrapped buffer.
>-    base_->data_->document.swap(stream);
>-
>-    // Pump the I/O loop.  DoReceiveLoop is guaranteed not to attempt to
>-    // complete the I/O process, which means that our wrapper is not in danger
>-    // of being deleted.  To ensure this, DoReceiveLoop returns true when it
>-    // wants complete to be called.  We make sure to uninstall our wrapper
>-    // before calling complete().
>-    HttpError http_error;
>-    bool complete = base_->DoReceiveLoop(&http_error);
>-
>-    // Reinstall the original output document.
>-    base_->data_->document.swap(stream);
>-
>-    // If we reach the end of the receive stream, we disconnect our stream
>-    // adapter from the HttpBase, and further calls to read will either return
>-    // EOS or ERROR, appropriately.  Finally, we call complete().
>-    StreamResult result = SR_BLOCK;
>-    if (complete) {
>-      HttpBase* base = Disconnect(http_error);
>-      if (error) *error = error_;
>-      result = (HE_NONE == error_) ? SR_EOS : SR_ERROR;
>-      base->complete(http_error);
>-    }
>-
>-    // Even if we are complete, if some data was read we must return SUCCESS.
>-    // Future Reads will return EOS or ERROR based on the error_ variable.
>-    size_t position;
>-    stream->GetPosition(&position);
>-    if (position > 0) {
>-      if (read) *read = position;
>-      result = SR_SUCCESS;
>-    }
>-    return result;
>-  }
>-
>-  StreamResult Write(const void* data,
>-                     size_t data_len,
>-                     size_t* written,
>-                     int* error) override {
>-    if (error) *error = -1;
>-    return SR_ERROR;
>-  }
>-
>-  void Close() override {
>-    if (base_) {
>-      HttpBase* base = Disconnect(HE_NONE);
>-      if (HM_RECV == base->mode_ && base->http_stream_) {
>-        // Read I/O could have been stalled on the user of this DocumentStream,
>-        // so restart the I/O process now that we've removed ourselves.
>-        base->http_stream_->PostEvent(SE_READ, 0);
>-      }
>-    }
>-  }
>-
>-  bool GetAvailable(size_t* size) const override {
>-    if (!base_ || HM_RECV != base_->mode_)
>-      return false;
>-    size_t data_size = base_->GetDataRemaining();
>-    if (SIZE_UNKNOWN == data_size)
>-      return false;
>-    if (size)
>-      *size = data_size;
>-    return true;
>-  }
>-
>-  HttpBase* Disconnect(HttpError error) {
>-    RTC_DCHECK(nullptr != base_);
>-    RTC_DCHECK(nullptr != base_->doc_stream_);
>-    HttpBase* base = base_;
>-    base_->doc_stream_ = nullptr;
>-    base_ = nullptr;
>-    error_ = error;
>-    return base;
>-  }
>-
>-private:
>-  HttpBase* base_;
>-  HttpError error_;
>-};
>-
> //////////////////////////////////////////////////////////////////////
> // HttpBase
> //////////////////////////////////////////////////////////////////////
> 
> HttpBase::HttpBase()
>-    : mode_(HM_NONE),
>-      data_(nullptr),
>-      notify_(nullptr),
>-      http_stream_(nullptr),
>-      doc_stream_(nullptr) {}
>+    : mode_(HM_NONE), data_(nullptr), notify_(nullptr), http_stream_(nullptr) {}
> 
> HttpBase::~HttpBase() {
>   RTC_DCHECK(HM_NONE == mode_);
> }
> 
>-bool
>-HttpBase::isConnected() const {
>+bool HttpBase::isConnected() const {
>   return (http_stream_ != nullptr) && (http_stream_->GetState() == SS_OPEN);
> }
> 
>-bool
>-HttpBase::attach(StreamInterface* stream) {
>+bool HttpBase::attach(StreamInterface* stream) {
>   if ((mode_ != HM_NONE) || (http_stream_ != nullptr) || (stream == nullptr)) {
>     RTC_NOTREACHED();
>     return false;
>@@ -388,8 +249,7 @@ HttpBase::attach(StreamInterface* stream) {
>   return true;
> }
> 
>-StreamInterface*
>-HttpBase::detach() {
>+StreamInterface* HttpBase::detach() {
>   RTC_DCHECK(HM_NONE == mode_);
>   if (mode_ != HM_NONE) {
>     return nullptr;
>@@ -402,8 +262,7 @@ HttpBase::detach() {
>   return stream;
> }
> 
>-void
>-HttpBase::send(HttpData* data) {
>+void HttpBase::send(HttpData* data) {
>   RTC_DCHECK(HM_NONE == mode_);
>   if (mode_ != HM_NONE) {
>     return;
>@@ -422,8 +281,8 @@ HttpBase::send(HttpData* data) {
>   }
> 
>   std::string encoding;
>-  if (data_->hasHeader(HH_TRANSFER_ENCODING, &encoding)
>-      && (encoding == "chunked")) {
>+  if (data_->hasHeader(HH_TRANSFER_ENCODING, &encoding) &&
>+      (encoding == "chunked")) {
>     chunk_data_ = true;
>   }
> 
>@@ -439,8 +298,7 @@ HttpBase::send(HttpData* data) {
>   flush_data();
> }
> 
>-void
>-HttpBase::recv(HttpData* data) {
>+void HttpBase::recv(HttpData* data) {
>   RTC_DCHECK(HM_NONE == mode_);
>   if (mode_ != HM_NONE) {
>     return;
>@@ -455,15 +313,10 @@ HttpBase::recv(HttpData* data) {
>   ignore_data_ = chunk_data_ = false;
> 
>   reset();
>-  if (doc_stream_) {
>-    doc_stream_->SignalEvent(doc_stream_, SE_OPEN | SE_READ, 0);
>-  } else {
>-    read_and_process_data();
>-  }
>+  read_and_process_data();
> }
> 
>-void
>-HttpBase::abort(HttpError err) {
>+void HttpBase::abort(HttpError err) {
>   if (mode_ != HM_NONE) {
>     if (http_stream_ != nullptr) {
>       http_stream_->Close();
>@@ -472,13 +325,6 @@ HttpBase::abort(HttpError err) {
>   }
> }
> 
>-StreamInterface* HttpBase::GetDocumentStream() {
>-  if (doc_stream_)
>-    return nullptr;
>-  doc_stream_ = new DocumentStream(this);
>-  return doc_stream_;
>-}
>-
> HttpError HttpBase::HandleStreamClose(int error) {
>   if (http_stream_ != nullptr) {
>     http_stream_->Close();
>@@ -519,28 +365,27 @@ bool HttpBase::DoReceiveLoop(HttpError* error) {
>       // Attempt to buffer more data.
>       size_t read;
>       int read_error;
>-      StreamResult read_result = http_stream_->Read(buffer_ + len_,
>-                                                    sizeof(buffer_) - len_,
>-                                                    &read, &read_error);
>+      StreamResult read_result = http_stream_->Read(
>+          buffer_ + len_, sizeof(buffer_) - len_, &read, &read_error);
>       switch (read_result) {
>-      case SR_SUCCESS:
>-        RTC_DCHECK(len_ + read <= sizeof(buffer_));
>-        len_ += read;
>-        break;
>-      case SR_BLOCK:
>-        if (process_requires_more_data) {
>-          // We're can't make progress until more data is available.
>-          return false;
>-        }
>-        // Attempt to process the data already in our buffer.
>-        break;
>-      case SR_EOS:
>-        // Clean close, with no error.
>-        read_error = 0;
>-        FALLTHROUGH();  // Fall through to HandleStreamClose.
>-      case SR_ERROR:
>-        *error = HandleStreamClose(read_error);
>-        return true;
>+        case SR_SUCCESS:
>+          RTC_DCHECK(len_ + read <= sizeof(buffer_));
>+          len_ += read;
>+          break;
>+        case SR_BLOCK:
>+          if (process_requires_more_data) {
>+            // We're can't make progress until more data is available.
>+            return false;
>+          }
>+          // Attempt to process the data already in our buffer.
>+          break;
>+        case SR_EOS:
>+          // Clean close, with no error.
>+          read_error = 0;
>+          RTC_FALLTHROUGH();  // Fall through to HandleStreamClose.
>+        case SR_ERROR:
>+          *error = HandleStreamClose(read_error);
>+          return true;
>       }
>     } else if (process_requires_more_data) {
>       // We have too much unprocessed data in our buffer.  This should only
>@@ -557,22 +402,21 @@ bool HttpBase::DoReceiveLoop(HttpError* error) {
>     // necessary to call Process with an empty buffer, since the state machine
>     // may have interrupted state transitions to complete.
>     size_t processed;
>-    ProcessResult process_result = Process(buffer_, len_, &processed,
>-                                            error);
>+    ProcessResult process_result = Process(buffer_, len_, &processed, error);
>     RTC_DCHECK(processed <= len_);
>     len_ -= processed;
>     memmove(buffer_, buffer_ + processed, len_);
>     switch (process_result) {
>-    case PR_CONTINUE:
>-      // We need more data to make progress.
>-      process_requires_more_data = true;
>-      break;
>-    case PR_BLOCK:
>-      // We're stalled on writing the processed data.
>-      return false;
>-    case PR_COMPLETE:
>-      // *error already contains the correct code.
>-      return true;
>+      case PR_CONTINUE:
>+        // We need more data to make progress.
>+        process_requires_more_data = true;
>+        break;
>+      case PR_BLOCK:
>+        // We're stalled on writing the processed data.
>+        return false;
>+      case PR_COMPLETE:
>+        // *error already contains the correct code.
>+        return true;
>     }
>   } while (++loop_count <= kMaxReadCount);
> 
>@@ -580,16 +424,14 @@ bool HttpBase::DoReceiveLoop(HttpError* error) {
>   return false;
> }
> 
>-void
>-HttpBase::read_and_process_data() {
>+void HttpBase::read_and_process_data() {
>   HttpError error;
>   if (DoReceiveLoop(&error)) {
>     complete(error);
>   }
> }
> 
>-void
>-HttpBase::flush_data() {
>+void HttpBase::flush_data() {
>   RTC_DCHECK(HM_SEND == mode_);
> 
>   // When send_required is true, no more buffering can occur without a network
>@@ -631,17 +473,16 @@ HttpBase::flush_data() {
>       } else {
>         size_t read;
>         int error;
>-        StreamResult result = data_->document->Read(buffer_ + offset,
>-                                                    sizeof(buffer_) - reserve,
>-                                                    &read, &error);
>+        StreamResult result = data_->document->Read(
>+            buffer_ + offset, sizeof(buffer_) - reserve, &read, &error);
>         if (result == SR_SUCCESS) {
>           RTC_DCHECK(reserve + read <= sizeof(buffer_));
>           if (chunk_data_) {
>             // Prepend the chunk length in hex.
>             // Note: sprintfn appends a null terminator, which is why we can't
>             // combine it with the line terminator.
>-            sprintfn(buffer_ + len_, kChunkDigits + 1, "%.*x",
>-                     kChunkDigits, read);
>+            sprintfn(buffer_ + len_, kChunkDigits + 1, "%.*x", kChunkDigits,
>+                     read);
>             // Add line terminator to the chunk length.
>             memcpy(buffer_ + len_ + kChunkDigits, "\r\n", 2);
>             // Add line terminator to the end of the chunk.
>@@ -708,14 +549,13 @@ HttpBase::flush_data() {
>   RTC_NOTREACHED();
> }
> 
>-bool
>-HttpBase::queue_headers() {
>+bool HttpBase::queue_headers() {
>   RTC_DCHECK(HM_SEND == mode_);
>   while (header_ != data_->end()) {
>-    size_t len = sprintfn(buffer_ + len_, sizeof(buffer_) - len_,
>-                          "%.*s: %.*s\r\n",
>-                          header_->first.size(), header_->first.data(),
>-                          header_->second.size(), header_->second.data());
>+    size_t len =
>+        sprintfn(buffer_ + len_, sizeof(buffer_) - len_, "%.*s: %.*s\r\n",
>+                 header_->first.size(), header_->first.data(),
>+                 header_->second.size(), header_->second.data());
>     if (len_ + len < sizeof(buffer_) - 3) {
>       len_ += len;
>       ++header_;
>@@ -733,8 +573,7 @@ HttpBase::queue_headers() {
>   return false;
> }
> 
>-void
>-HttpBase::do_complete(HttpError err) {
>+void HttpBase::do_complete(HttpError err) {
>   RTC_DCHECK(mode_ != HM_NONE);
>   HttpMode mode = mode_;
>   mode_ = HM_NONE;
>@@ -742,13 +581,6 @@ HttpBase::do_complete(HttpError err) {
>     data_->document->SignalEvent.disconnect(this);
>   }
>   data_ = nullptr;
>-  if ((HM_RECV == mode) && doc_stream_) {
>-    RTC_DCHECK(HE_NONE !=
>-               err);  // We should have Disconnected doc_stream_ already.
>-    DocumentStream* ds = doc_stream_;
>-    ds->Disconnect(err);
>-    ds->SignalEvent(ds, SE_CLOSE, err);
>-  }
>   if (notify_) {
>     notify_->onHttpComplete(mode, err);
>   }
>@@ -758,8 +590,9 @@ HttpBase::do_complete(HttpError err) {
> // Stream Signals
> //
> 
>-void
>-HttpBase::OnHttpStreamEvent(StreamInterface* stream, int events, int error) {
>+void HttpBase::OnHttpStreamEvent(StreamInterface* stream,
>+                                 int events,
>+                                 int error) {
>   RTC_DCHECK(stream == http_stream_);
>   if ((events & SE_OPEN) && (mode_ == HM_CONNECT)) {
>     do_complete();
>@@ -772,11 +605,7 @@ HttpBase::OnHttpStreamEvent(StreamInterface* stream, int events, int error) {
>   }
> 
>   if ((events & SE_READ) && (mode_ == HM_RECV)) {
>-    if (doc_stream_) {
>-      doc_stream_->SignalEvent(doc_stream_, SE_READ, 0);
>-    } else {
>-      read_and_process_data();
>-    }
>+    read_and_process_data();
>     return;
>   }
> 
>@@ -793,8 +622,7 @@ HttpBase::OnHttpStreamEvent(StreamInterface* stream, int events, int error) {
>   }
> }
> 
>-void
>-HttpBase::OnDocumentEvent(StreamInterface* stream, int events, int error) {
>+void HttpBase::OnDocumentEvent(StreamInterface* stream, int events, int error) {
>   RTC_DCHECK(stream == data_->document.get());
>   if ((events & SE_WRITE) && (mode_ == HM_RECV)) {
>     read_and_process_data();
>@@ -817,24 +645,26 @@ HttpBase::OnDocumentEvent(StreamInterface* stream, int events, int error) {
> // HttpParser Implementation
> //
> 
>-HttpParser::ProcessResult
>-HttpBase::ProcessLeader(const char* line, size_t len, HttpError* error) {
>+HttpParser::ProcessResult HttpBase::ProcessLeader(const char* line,
>+                                                  size_t len,
>+                                                  HttpError* error) {
>   *error = data_->parseLeader(line, len);
>   return (HE_NONE == *error) ? PR_CONTINUE : PR_COMPLETE;
> }
> 
>-HttpParser::ProcessResult
>-HttpBase::ProcessHeader(const char* name, size_t nlen, const char* value,
>-                        size_t vlen, HttpError* error) {
>+HttpParser::ProcessResult HttpBase::ProcessHeader(const char* name,
>+                                                  size_t nlen,
>+                                                  const char* value,
>+                                                  size_t vlen,
>+                                                  HttpError* error) {
>   std::string sname(name, nlen), svalue(value, vlen);
>   data_->addHeader(sname, svalue);
>   return PR_CONTINUE;
> }
> 
>-HttpParser::ProcessResult
>-HttpBase::ProcessHeaderComplete(bool chunked, size_t& data_size,
>-                                HttpError* error) {
>-  StreamInterface* old_docstream = doc_stream_;
>+HttpParser::ProcessResult HttpBase::ProcessHeaderComplete(bool chunked,
>+                                                          size_t& data_size,
>+                                                          HttpError* error) {
>   if (notify_) {
>     *error = notify_->onHttpHeaderComplete(chunked, data_size);
>     // The request must not be aborted as a result of this callback.
>@@ -846,42 +676,38 @@ HttpBase::ProcessHeaderComplete(bool chunked, size_t& data_size,
>   if (HE_NONE != *error) {
>     return PR_COMPLETE;
>   }
>-  if (old_docstream != doc_stream_) {
>-    // Break out of Process loop, since our I/O model just changed.
>-    return PR_BLOCK;
>-  }
>   return PR_CONTINUE;
> }
> 
>-HttpParser::ProcessResult
>-HttpBase::ProcessData(const char* data, size_t len, size_t& read,
>-                      HttpError* error) {
>+HttpParser::ProcessResult HttpBase::ProcessData(const char* data,
>+                                                size_t len,
>+                                                size_t& read,
>+                                                HttpError* error) {
>   if (ignore_data_ || !data_->document) {
>     read = len;
>     return PR_CONTINUE;
>   }
>   int write_error = 0;
>   switch (data_->document->Write(data, len, &read, &write_error)) {
>-  case SR_SUCCESS:
>-    return PR_CONTINUE;
>-  case SR_BLOCK:
>-    return PR_BLOCK;
>-  case SR_EOS:
>-    RTC_LOG_F(LS_ERROR) << "Unexpected EOS";
>-    *error = HE_STREAM;
>-    return PR_COMPLETE;
>-  case SR_ERROR:
>-  default:
>-    RTC_LOG_F(LS_ERROR) << "Write error: " << write_error;
>-    *error = HE_STREAM;
>-    return PR_COMPLETE;
>+    case SR_SUCCESS:
>+      return PR_CONTINUE;
>+    case SR_BLOCK:
>+      return PR_BLOCK;
>+    case SR_EOS:
>+      RTC_LOG_F(LS_ERROR) << "Unexpected EOS";
>+      *error = HE_STREAM;
>+      return PR_COMPLETE;
>+    case SR_ERROR:
>+    default:
>+      RTC_LOG_F(LS_ERROR) << "Write error: " << write_error;
>+      *error = HE_STREAM;
>+      return PR_COMPLETE;
>   }
> }
> 
>-void
>-HttpBase::OnComplete(HttpError err) {
>+void HttpBase::OnComplete(HttpError err) {
>   RTC_LOG_F(LS_VERBOSE);
>   do_complete(err);
> }
> 
>-} // namespace rtc
>+}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/httpbase.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/httpbase.h
>index 5ca0134c34e..b0e24254be2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/httpbase.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/httpbase.h
>@@ -8,7 +8,6 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-
> #ifndef RTC_BASE_HTTPBASE_H_
> #define RTC_BASE_HTTPBASE_H_
> 
>@@ -26,39 +25,51 @@ class StreamInterface;
> ///////////////////////////////////////////////////////////////////////////////
> 
> class HttpParser {
>-public:
>+ public:
>   enum ProcessResult { PR_CONTINUE, PR_BLOCK, PR_COMPLETE };
>   HttpParser();
>   virtual ~HttpParser();
> 
>   void reset();
>-  ProcessResult Process(const char* buffer, size_t len, size_t* processed,
>+  ProcessResult Process(const char* buffer,
>+                        size_t len,
>+                        size_t* processed,
>                         HttpError* error);
>   bool is_valid_end_of_input() const;
>   void complete(HttpError err);
> 
>   size_t GetDataRemaining() const { return data_size_; }
> 
>-protected:
>+ protected:
>   ProcessResult ProcessLine(const char* line, size_t len, HttpError* error);
> 
>   // HttpParser Interface
>-  virtual ProcessResult ProcessLeader(const char* line, size_t len,
>+  virtual ProcessResult ProcessLeader(const char* line,
>+                                      size_t len,
>                                       HttpError* error) = 0;
>-  virtual ProcessResult ProcessHeader(const char* name, size_t nlen,
>-                                      const char* value, size_t vlen,
>+  virtual ProcessResult ProcessHeader(const char* name,
>+                                      size_t nlen,
>+                                      const char* value,
>+                                      size_t vlen,
>                                       HttpError* error) = 0;
>-  virtual ProcessResult ProcessHeaderComplete(bool chunked, size_t& data_size,
>+  virtual ProcessResult ProcessHeaderComplete(bool chunked,
>+                                              size_t& data_size,
>                                               HttpError* error) = 0;
>-  virtual ProcessResult ProcessData(const char* data, size_t len, size_t& read,
>+  virtual ProcessResult ProcessData(const char* data,
>+                                    size_t len,
>+                                    size_t& read,
>                                     HttpError* error) = 0;
>   virtual void OnComplete(HttpError err) = 0;
> 
>-private:
>+ private:
>   enum State {
>-    ST_LEADER, ST_HEADERS,
>-    ST_CHUNKSIZE, ST_CHUNKTERM, ST_TRAILERS,
>-    ST_DATA, ST_COMPLETE
>+    ST_LEADER,
>+    ST_HEADERS,
>+    ST_CHUNKSIZE,
>+    ST_CHUNKTERM,
>+    ST_TRAILERS,
>+    ST_DATA,
>+    ST_COMPLETE
>   } state_;
>   bool chunked_;
>   size_t data_size_;
>@@ -71,7 +82,7 @@ private:
> enum HttpMode { HM_NONE, HM_CONNECT, HM_RECV, HM_SEND };
> 
> class IHttpNotify {
>-public:
>+ public:
>   virtual ~IHttpNotify() {}
>   virtual HttpError onHttpHeaderComplete(bool chunked, size_t& data_size) = 0;
>   virtual void onHttpComplete(HttpMode mode, HttpError err) = 0;
>@@ -88,11 +99,8 @@ public:
> // stream interface drives I/O via calls to Read().
> ///////////////////////////////////////////////////////////////////////////////
> 
>-class HttpBase
>-: private HttpParser,
>-  public sigslot::has_slots<>
>-{
>-public:
>+class HttpBase : private HttpParser, public sigslot::has_slots<> {
>+ public:
>   HttpBase();
>   ~HttpBase() override;
> 
>@@ -111,12 +119,7 @@ public:
>   void set_ignore_data(bool ignore) { ignore_data_ = ignore; }
>   bool ignore_data() const { return ignore_data_; }
> 
>-  // Obtaining this stream puts HttpBase into stream mode until the stream
>-  // is closed.  HttpBase can only expose one open stream interface at a time.
>-  // Further calls will return null.
>-  StreamInterface* GetDocumentStream();
>-
>-protected:
>+ protected:
>   // Do cleanup when the http stream closes (error may be 0 for a clean
>   // shutdown), and return the error code to signal.
>   HttpError HandleStreamClose(int error);
>@@ -162,7 +165,7 @@ protected:
>                             HttpError* error) override;
>   void OnComplete(HttpError err) override;
> 
>-private:
>+ private:
>   class DocumentStream;
>   friend class DocumentStream;
> 
>@@ -172,7 +175,6 @@ private:
>   HttpData* data_;
>   IHttpNotify* notify_;
>   StreamInterface* http_stream_;
>-  DocumentStream* doc_stream_;
>   char buffer_[kBufferSize];
>   size_t len_;
> 
>@@ -182,6 +184,6 @@ private:
> 
> ///////////////////////////////////////////////////////////////////////////////
> 
>-} // namespace rtc
>+}  // namespace rtc
> 
>-#endif // RTC_BASE_HTTPBASE_H_
>+#endif  // RTC_BASE_HTTPBASE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/httpbase_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/httpbase_unittest.cc
>index 1b7ab7f8e97..35321da243e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/httpbase_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/httpbase_unittest.cc
>@@ -17,35 +17,25 @@
> namespace rtc {
> 
> const char* const kHttpResponse =
>-  "HTTP/1.1 200\r\n"
>-  "Connection: Keep-Alive\r\n"
>-  "Content-Type: text/plain\r\n"
>-  "Proxy-Authorization: 42\r\n"
>-  "Transfer-Encoding: chunked\r\n"
>-  "\r\n"
>-  "00000008\r\n"
>-  "Goodbye!\r\n"
>-  "0\r\n\r\n";
>+    "HTTP/1.1 200\r\n"
>+    "Connection: Keep-Alive\r\n"
>+    "Content-Type: text/plain\r\n"
>+    "Proxy-Authorization: 42\r\n"
>+    "Transfer-Encoding: chunked\r\n"
>+    "\r\n"
>+    "00000008\r\n"
>+    "Goodbye!\r\n"
>+    "0\r\n\r\n";
> 
> const char* const kHttpEmptyResponse =
>-  "HTTP/1.1 200\r\n"
>-  "Connection: Keep-Alive\r\n"
>-  "Content-Length: 0\r\n"
>-  "Proxy-Authorization: 42\r\n"
>-  "\r\n";
>-
>-const char* const kHttpResponsePrefix =
>-  "HTTP/1.1 200\r\n"
>-  "Connection: Keep-Alive\r\n"
>-  "Content-Type: text/plain\r\n"
>-  "Proxy-Authorization: 42\r\n"
>-  "Transfer-Encoding: chunked\r\n"
>-  "\r\n"
>-  "8\r\n"
>-  "Goodbye!\r\n";
>+    "HTTP/1.1 200\r\n"
>+    "Connection: Keep-Alive\r\n"
>+    "Content-Length: 0\r\n"
>+    "Proxy-Authorization: 42\r\n"
>+    "\r\n";
> 
> class HttpBaseTest : public testing::Test, public IHttpNotify {
>-public:
>+ public:
>   enum EventType { E_HEADER_COMPLETE, E_COMPLETE, E_CLOSED };
>   struct Event {
>     EventType event;
>@@ -54,7 +44,7 @@ public:
>     HttpMode mode;
>     HttpError err;
>   };
>-  HttpBaseTest() : mem(nullptr), obtain_stream(false), http_stream(nullptr) {}
>+  HttpBaseTest() : mem(nullptr), http_stream(nullptr) {}
> 
>   void TearDown() override {
>     delete http_stream;
>@@ -64,21 +54,18 @@ public:
> 
>   HttpError onHttpHeaderComplete(bool chunked, size_t& data_size) override {
>     RTC_LOG_F(LS_VERBOSE) << "chunked: " << chunked << " size: " << data_size;
>-    Event e = { E_HEADER_COMPLETE, chunked, data_size, HM_NONE, HE_NONE};
>+    Event e = {E_HEADER_COMPLETE, chunked, data_size, HM_NONE, HE_NONE};
>     events.push_back(e);
>-    if (obtain_stream) {
>-      ObtainDocumentStream();
>-    }
>     return HE_NONE;
>   }
>   void onHttpComplete(HttpMode mode, HttpError err) override {
>     RTC_LOG_F(LS_VERBOSE) << "mode: " << mode << " err: " << err;
>-    Event e = { E_COMPLETE, false, 0, mode, err };
>+    Event e = {E_COMPLETE, false, 0, mode, err};
>     events.push_back(e);
>   }
>   void onHttpClosed(HttpError err) override {
>     RTC_LOG_F(LS_VERBOSE) << "err: " << err;
>-    Event e = { E_CLOSED, false, 0, HM_NONE, err };
>+    Event e = {E_CLOSED, false, 0, HM_NONE, err};
>     events.push_back(e);
>   }
> 
>@@ -88,7 +75,6 @@ public:
>   void VerifyDocumentContents(const char* expected_data,
>                               size_t expected_length = SIZE_UNKNOWN);
> 
>-  void ObtainDocumentStream();
>   void VerifyDocumentStreamIsOpening();
>   void VerifyDocumentStreamOpenEvent();
>   void ReadDocumentStreamData(const char* expected_data);
>@@ -108,8 +94,7 @@ public:
>   webrtc::testing::StreamSource src;
>   std::vector<Event> events;
> 
>-  // Document stream, and stream events
>-  bool obtain_stream;
>+  // Stream events
>   StreamInterface* http_stream;
>   webrtc::testing::StreamSink sink;
> };
>@@ -184,15 +169,6 @@ void HttpBaseTest::VerifyDocumentContents(const char* expected_data,
>   RTC_LOG_F(LS_VERBOSE) << "Exit";
> }
> 
>-void HttpBaseTest::ObtainDocumentStream() {
>-  RTC_LOG_F(LS_VERBOSE) << "Enter";
>-  EXPECT_FALSE(http_stream);
>-  http_stream = base.GetDocumentStream();
>-  ASSERT_TRUE(nullptr != http_stream);
>-  sink.Monitor(http_stream);
>-  RTC_LOG_F(LS_VERBOSE) << "Exit";
>-}
>-
> void HttpBaseTest::VerifyDocumentStreamIsOpening() {
>   RTC_LOG_F(LS_VERBOSE) << "Enter";
>   ASSERT_TRUE(nullptr != http_stream);
>@@ -200,7 +176,7 @@ void HttpBaseTest::VerifyDocumentStreamIsOpening() {
>   EXPECT_EQ(SS_OPENING, http_stream->GetState());
> 
>   size_t read = 0;
>-  char buffer[5] = { 0 };
>+  char buffer[5] = {0};
>   EXPECT_EQ(SR_BLOCK,
>             http_stream->Read(buffer, sizeof(buffer), &read, nullptr));
>   RTC_LOG_F(LS_VERBOSE) << "Exit";
>@@ -230,7 +206,7 @@ void HttpBaseTest::ReadDocumentStreamData(const char* expected_data) {
>   const size_t expected_length = strlen(expected_data);
>   while (verified_length < expected_length) {
>     size_t read = 0;
>-    char buffer[5] = { 0 };
>+    char buffer[5] = {0};
>     size_t amt_to_read =
>         std::min(expected_length - verified_length, sizeof(buffer));
>     EXPECT_EQ(SR_SUCCESS,
>@@ -247,7 +223,7 @@ void HttpBaseTest::VerifyDocumentStreamIsEOS() {
> 
>   ASSERT_TRUE(nullptr != http_stream);
>   size_t read = 0;
>-  char buffer[5] = { 0 };
>+  char buffer[5] = {0};
>   EXPECT_EQ(SR_EOS, http_stream->Read(buffer, sizeof(buffer), &read, nullptr));
>   EXPECT_EQ(SS_CLOSED, http_stream->GetState());
> 
>@@ -379,147 +355,4 @@ TEST_F(HttpBaseTest, SupportsReceiveViaDocumentPush) {
>   VerifyDocumentContents("Goodbye!");
> }
> 
>-TEST_F(HttpBaseTest, SupportsReceiveViaStreamPull) {
>-  // Switch to pull mode
>-  ObtainDocumentStream();
>-  VerifyDocumentStreamIsOpening();
>-
>-  // Queue response document
>-  SetupSource(kHttpResponse);
>-  VerifyDocumentStreamIsOpening();
>-
>-  // Begin receive
>-  base.recv(&data);
>-
>-  // Pull document data
>-  VerifyDocumentStreamOpenEvent();
>-  ReadDocumentStreamData("Goodbye!");
>-  VerifyDocumentStreamIsEOS();
>-
>-  // Document completed successfully
>-  VerifyHeaderComplete(2, false);
>-  VerifyTransferComplete(HM_RECV, HE_NONE);
>-  VerifyDocumentContents("");
>-}
>-
>-TEST_F(HttpBaseTest, DISABLED_AllowsCloseStreamBeforeDocumentIsComplete) {
>-
>-  // TODO: Remove extra logging once test failure is understood
>-  LoggingSeverity old_sev = rtc::LogMessage::GetLogToDebug();
>-  rtc::LogMessage::LogToDebug(LS_VERBOSE);
>-
>-
>-  // Switch to pull mode
>-  ObtainDocumentStream();
>-  VerifyDocumentStreamIsOpening();
>-
>-  // Queue response document
>-  SetupSource(kHttpResponse);
>-  VerifyDocumentStreamIsOpening();
>-
>-  // Begin receive
>-  base.recv(&data);
>-
>-  // Pull some of the data
>-  VerifyDocumentStreamOpenEvent();
>-  ReadDocumentStreamData("Goodb");
>-
>-  // We've seen the header by now
>-  VerifyHeaderComplete(1, false);
>-
>-  // Close the pull stream, this will transition back to push I/O.
>-  http_stream->Close();
>-  Thread::Current()->ProcessMessages(0);
>-
>-  // Remainder of document completed successfully
>-  VerifyTransferComplete(HM_RECV, HE_NONE);
>-  VerifyDocumentContents("ye!");
>-
>-  rtc::LogMessage::LogToDebug(old_sev);
>-}
>-
>-TEST_F(HttpBaseTest, AllowsGetDocumentStreamInResponseToHttpHeader) {
>-  // Queue response document
>-  SetupSource(kHttpResponse);
>-
>-  // Switch to pull mode in response to header arrival
>-  obtain_stream = true;
>-
>-  // Begin receive
>-  base.recv(&data);
>-
>-  // We've already seen the header, but not data has arrived
>-  VerifyHeaderComplete(1, false);
>-  VerifyDocumentContents("");
>-
>-  // Pull the document data
>-  ReadDocumentStreamData("Goodbye!");
>-  VerifyDocumentStreamIsEOS();
>-
>-  // Document completed successfully
>-  VerifyTransferComplete(HM_RECV, HE_NONE);
>-  VerifyDocumentContents("");
>-}
>-
>-TEST_F(HttpBaseTest, AllowsGetDocumentStreamWithEmptyDocumentBody) {
>-  // Queue empty response document
>-  SetupSource(kHttpEmptyResponse);
>-
>-  // Switch to pull mode in response to header arrival
>-  obtain_stream = true;
>-
>-  // Begin receive
>-  base.recv(&data);
>-
>-  // We've already seen the header, but not data has arrived
>-  VerifyHeaderComplete(1, true);
>-  VerifyDocumentContents("");
>-
>-  // The document is still open, until we attempt to read
>-  ASSERT_TRUE(nullptr != http_stream);
>-  EXPECT_EQ(SS_OPEN, http_stream->GetState());
>-
>-  // Attempt to read data, and discover EOS
>-  VerifyDocumentStreamIsEOS();
>-
>-  // Document completed successfully
>-  VerifyTransferComplete(HM_RECV, HE_NONE);
>-  VerifyDocumentContents("");
>-}
>-
>-TEST_F(HttpBaseTest, SignalsDocumentStreamCloseOnUnexpectedClose) {
>-  // Switch to pull mode
>-  ObtainDocumentStream();
>-  VerifyDocumentStreamIsOpening();
>-
>-  // Queue response document
>-  SetupSource(kHttpResponsePrefix);
>-  VerifyDocumentStreamIsOpening();
>-
>-  // Begin receive
>-  base.recv(&data);
>-
>-  // Pull document data
>-  VerifyDocumentStreamOpenEvent();
>-  ReadDocumentStreamData("Goodbye!");
>-
>-  // Simulate unexpected close
>-  src.SetState(SS_CLOSED);
>-
>-  // Observe error event on document stream
>-  EXPECT_EQ(webrtc::testing::SSE_ERROR, sink.Events(http_stream));
>-
>-  // Future reads give an error
>-  int error = 0;
>-  char buffer[5] = { 0 };
>-  EXPECT_EQ(SR_ERROR,
>-            http_stream->Read(buffer, sizeof(buffer), nullptr, &error));
>-  EXPECT_EQ(HE_DISCONNECTED, error);
>-
>-  // Document completed with error
>-  VerifyHeaderComplete(2, false);
>-  VerifyTransferComplete(HM_RECV, HE_DISCONNECTED);
>-  VerifyDocumentContents("");
>-}
>-
>-} // namespace rtc
>+}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/httpcommon-inl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/httpcommon-inl.h
>index 70263daf88c..bb72357c801 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/httpcommon-inl.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/httpcommon-inl.h
>@@ -21,13 +21,15 @@ namespace rtc {
> // Url
> ///////////////////////////////////////////////////////////////////////////////
> 
>-template<class CTYPE>
>+template <class CTYPE>
> void Url<CTYPE>::do_set_url(const CTYPE* val, size_t len) {
>   if (ascnicmp(val, "http://", 7) == 0) {
>-    val += 7; len -= 7;
>+    val += 7;
>+    len -= 7;
>     secure_ = false;
>   } else if (ascnicmp(val, "https://", 8) == 0) {
>-    val += 8; len -= 8;
>+    val += 8;
>+    len -= 8;
>     secure_ = true;
>   } else {
>     clear();
>@@ -42,7 +44,7 @@ void Url<CTYPE>::do_set_url(const CTYPE* val, size_t len) {
>   do_set_full_path(path, len - address_length);
> }
> 
>-template<class CTYPE>
>+template <class CTYPE>
> void Url<CTYPE>::do_set_address(const CTYPE* val, size_t len) {
>   if (const CTYPE* at = strchrn(val, len, static_cast<CTYPE>('@'))) {
>     // Everything before the @ is a user:password combo, so skip it.
>@@ -61,7 +63,7 @@ void Url<CTYPE>::do_set_address(const CTYPE* val, size_t len) {
>   }
> }
> 
>-template<class CTYPE>
>+template <class CTYPE>
> void Url<CTYPE>::do_set_full_path(const CTYPE* val, size_t len) {
>   const CTYPE* query = strchrn(val, len, static_cast<CTYPE>('?'));
>   if (!query) {
>@@ -78,7 +80,7 @@ void Url<CTYPE>::do_set_full_path(const CTYPE* val, size_t len) {
>   query_.assign(query, len - path_length);
> }
> 
>-template<class CTYPE>
>+template <class CTYPE>
> void Url<CTYPE>::do_get_url(string* val) const {
>   CTYPE protocol[9];
>   asccpyn(protocol, arraysize(protocol), secure_ ? "https://" : "http://");
>@@ -87,7 +89,7 @@ void Url<CTYPE>::do_get_url(string* val) const {
>   do_get_full_path(val);
> }
> 
>-template<class CTYPE>
>+template <class CTYPE>
> void Url<CTYPE>::do_get_address(string* val) const {
>   val->append(host_);
>   if (port_ != HttpDefaultPort(secure_)) {
>@@ -98,13 +100,13 @@ void Url<CTYPE>::do_get_address(string* val) const {
>   }
> }
> 
>-template<class CTYPE>
>+template <class CTYPE>
> void Url<CTYPE>::do_get_full_path(string* val) const {
>   val->append(path_);
>   val->append(query_);
> }
> 
>-template<class CTYPE>
>+template <class CTYPE>
> bool Url<CTYPE>::get_attribute(const string& name, string* value) const {
>   if (query_.empty())
>     return false;
>@@ -114,7 +116,7 @@ bool Url<CTYPE>::get_attribute(const string& name, string* value) const {
>     return false;
> 
>   pos += name.length() + 1;
>-  if ((pos > query_.length()) || (static_cast<CTYPE>('=') != query_[pos-1]))
>+  if ((pos > query_.length()) || (static_cast<CTYPE>('=') != query_[pos - 1]))
>     return false;
> 
>   std::string::size_type end = query_.find(static_cast<CTYPE>('&'), pos);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/httpcommon.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/httpcommon.cc
>index 5f2112ac1a4..4ecb393545b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/httpcommon.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/httpcommon.cc
>@@ -11,7 +11,6 @@
> #include <time.h>
> 
> #if defined(WEBRTC_WIN)
>-#define WIN32_LEAN_AND_MEAN
> #include <windows.h>
> #include <winsock2.h>
> #include <ws2tcpip.h>
>@@ -22,27 +21,104 @@
> #include <algorithm>
> 
> #include "rtc_base/arraysize.h"
>-#include "rtc_base/base64.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/cryptstring.h"
> #include "rtc_base/httpcommon-inl.h"
> #include "rtc_base/httpcommon.h"
> #include "rtc_base/messagedigest.h"
> #include "rtc_base/socketaddress.h"
>+#include "rtc_base/third_party/base64/base64.h"
>+#include "rtc_base/zero_memory.h"
> 
> namespace rtc {
>-
>+namespace {
> #if defined(WEBRTC_WIN)
>-extern const ConstantLabel SECURITY_ERRORS[];
>-#endif
>+///////////////////////////////////////////////////////////////////////////////
>+// ConstantToLabel can be used to easily generate string names from constant
>+// values.  This can be useful for logging descriptive names of error messages.
>+// Usage:
>+//   const ConstantToLabel LIBRARY_ERRORS[] = {
>+//     KLABEL(SOME_ERROR),
>+//     KLABEL(SOME_OTHER_ERROR),
>+//     ...
>+//     LASTLABEL
>+//   }
>+//
>+//   int err = LibraryFunc();
>+//   LOG(LS_ERROR) << "LibraryFunc returned: "
>+//                 << GetErrorName(err, LIBRARY_ERRORS);
>+struct ConstantToLabel {
>+  int value;
>+  const char* label;
>+};
>+
>+const char* LookupLabel(int value, const ConstantToLabel entries[]) {
>+  for (int i = 0; entries[i].label; ++i) {
>+    if (value == entries[i].value) {
>+      return entries[i].label;
>+    }
>+  }
>+  return 0;
>+}
>+
>+std::string GetErrorName(int err, const ConstantToLabel* err_table) {
>+  if (err == 0)
>+    return "No error";
>+
>+  if (err_table != 0) {
>+    if (const char* value = LookupLabel(err, err_table))
>+      return value;
>+  }
>+
>+  char buffer[16];
>+  snprintf(buffer, sizeof(buffer), "0x%08x", err);
>+  return buffer;
>+}
>+
>+#define KLABEL(x) \
>+  { x, #x }
>+#define LASTLABEL \
>+  { 0, 0 }
>+
>+const ConstantToLabel SECURITY_ERRORS[] = {
>+    KLABEL(SEC_I_COMPLETE_AND_CONTINUE),
>+    KLABEL(SEC_I_COMPLETE_NEEDED),
>+    KLABEL(SEC_I_CONTEXT_EXPIRED),
>+    KLABEL(SEC_I_CONTINUE_NEEDED),
>+    KLABEL(SEC_I_INCOMPLETE_CREDENTIALS),
>+    KLABEL(SEC_I_RENEGOTIATE),
>+    KLABEL(SEC_E_CERT_EXPIRED),
>+    KLABEL(SEC_E_INCOMPLETE_MESSAGE),
>+    KLABEL(SEC_E_INSUFFICIENT_MEMORY),
>+    KLABEL(SEC_E_INTERNAL_ERROR),
>+    KLABEL(SEC_E_INVALID_HANDLE),
>+    KLABEL(SEC_E_INVALID_TOKEN),
>+    KLABEL(SEC_E_LOGON_DENIED),
>+    KLABEL(SEC_E_NO_AUTHENTICATING_AUTHORITY),
>+    KLABEL(SEC_E_NO_CREDENTIALS),
>+    KLABEL(SEC_E_NOT_OWNER),
>+    KLABEL(SEC_E_OK),
>+    KLABEL(SEC_E_SECPKG_NOT_FOUND),
>+    KLABEL(SEC_E_TARGET_UNKNOWN),
>+    KLABEL(SEC_E_UNKNOWN_CREDENTIALS),
>+    KLABEL(SEC_E_UNSUPPORTED_FUNCTION),
>+    KLABEL(SEC_E_UNTRUSTED_ROOT),
>+    KLABEL(SEC_E_WRONG_PRINCIPAL),
>+    LASTLABEL};
>+#undef KLABEL
>+#undef LASTLABEL
>+#endif  // defined(WEBRTC_WIN)
>+}  // namespace
> 
> //////////////////////////////////////////////////////////////////////
> // Enum - TODO: expose globally later?
> //////////////////////////////////////////////////////////////////////
> 
>-bool find_string(size_t& index, const std::string& needle,
>-                 const char* const haystack[], size_t max_index) {
>-  for (index=0; index<max_index; ++index) {
>+bool find_string(size_t& index,
>+                 const std::string& needle,
>+                 const char* const haystack[],
>+                 size_t max_index) {
>+  for (index = 0; index < max_index; ++index) {
>     if (_stricmp(needle.c_str(), haystack[index]) == 0) {
>       return true;
>     }
>@@ -50,7 +126,7 @@ bool find_string(size_t& index, const std::string& needle,
>   return false;
> }
> 
>-template<class E>
>+template <class E>
> struct Enum {
>   static const char** Names;
>   static size_t Size;
>@@ -67,60 +143,61 @@ struct Enum {
>   E val;
> 
>   inline operator E&() { return val; }
>-  inline Enum& operator=(E rhs) { val = rhs; return *this; }
>+  inline Enum& operator=(E rhs) {
>+    val = rhs;
>+    return *this;
>+  }
> 
>   inline const char* name() const { return Name(val); }
>   inline bool assign(const std::string& name) { return Parse(val, name); }
>-  inline Enum& operator=(const std::string& rhs) { assign(rhs); return *this; }
>+  inline Enum& operator=(const std::string& rhs) {
>+    assign(rhs);
>+    return *this;
>+  }
> };
> 
>-#define ENUM(e,n) \
>-  template<> const char** Enum<e>::Names = n; \
>-  template<> size_t Enum<e>::Size = sizeof(n)/sizeof(n[0])
>+#define ENUM(e, n)                 \
>+  template <>                      \
>+  const char** Enum<e>::Names = n; \
>+  template <>                      \
>+  size_t Enum<e>::Size = sizeof(n) / sizeof(n[0])
> 
> //////////////////////////////////////////////////////////////////////
> // HttpCommon
> //////////////////////////////////////////////////////////////////////
> 
>-static const char* kHttpVersions[HVER_LAST+1] = {
>-  "1.0", "1.1", "Unknown"
>-};
>+static const char* kHttpVersions[HVER_LAST + 1] = {"1.0", "1.1", "Unknown"};
> ENUM(HttpVersion, kHttpVersions);
> 
>-static const char* kHttpVerbs[HV_LAST+1] = {
>-  "GET", "POST", "PUT", "DELETE", "CONNECT", "HEAD"
>-};
>-ENUM(HttpVerb, kHttpVerbs);
>-
>-static const char* kHttpHeaders[HH_LAST+1] = {
>-  "Age",
>-  "Cache-Control",
>-  "Connection",
>-  "Content-Disposition",
>-  "Content-Length",
>-  "Content-Range",
>-  "Content-Type",
>-  "Cookie",
>-  "Date",
>-  "ETag",
>-  "Expires",
>-  "Host",
>-  "If-Modified-Since",
>-  "If-None-Match",
>-  "Keep-Alive",
>-  "Last-Modified",
>-  "Location",
>-  "Proxy-Authenticate",
>-  "Proxy-Authorization",
>-  "Proxy-Connection",
>-  "Range",
>-  "Set-Cookie",
>-  "TE",
>-  "Trailers",
>-  "Transfer-Encoding",
>-  "Upgrade",
>-  "User-Agent",
>-  "WWW-Authenticate",
>+static const char* kHttpHeaders[HH_LAST + 1] = {
>+    "Age",
>+    "Cache-Control",
>+    "Connection",
>+    "Content-Disposition",
>+    "Content-Length",
>+    "Content-Range",
>+    "Content-Type",
>+    "Cookie",
>+    "Date",
>+    "ETag",
>+    "Expires",
>+    "Host",
>+    "If-Modified-Since",
>+    "If-None-Match",
>+    "Keep-Alive",
>+    "Last-Modified",
>+    "Location",
>+    "Proxy-Authenticate",
>+    "Proxy-Authorization",
>+    "Proxy-Connection",
>+    "Range",
>+    "Set-Cookie",
>+    "TE",
>+    "Trailers",
>+    "Transfer-Encoding",
>+    "Upgrade",
>+    "User-Agent",
>+    "WWW-Authenticate",
> };
> ENUM(HttpHeader, kHttpHeaders);
> 
>@@ -132,14 +209,6 @@ bool FromString(HttpVersion& version, const std::string& str) {
>   return Enum<HttpVersion>::Parse(version, str);
> }
> 
>-const char* ToString(HttpVerb verb) {
>-  return Enum<HttpVerb>::Name(verb);
>-}
>-
>-bool FromString(HttpVerb& verb, const std::string& str) {
>-  return Enum<HttpVerb>::Parse(verb, str);
>-}
>-
> const char* ToString(HttpHeader header) {
>   return Enum<HttpHeader>::Name(header);
> }
>@@ -148,57 +217,39 @@ bool FromString(HttpHeader& header, const std::string& str) {
>   return Enum<HttpHeader>::Parse(header, str);
> }
> 
>-bool HttpCodeHasBody(uint32_t code) {
>-  return !HttpCodeIsInformational(code)
>-         && (code != HC_NO_CONTENT) && (code != HC_NOT_MODIFIED);
>-}
>-
>-bool HttpCodeIsCacheable(uint32_t code) {
>-  switch (code) {
>-  case HC_OK:
>-  case HC_NON_AUTHORITATIVE:
>-  case HC_PARTIAL_CONTENT:
>-  case HC_MULTIPLE_CHOICES:
>-  case HC_MOVED_PERMANENTLY:
>-  case HC_GONE:
>-    return true;
>-  default:
>-    return false;
>-  }
>-}
>-
> bool HttpHeaderIsEndToEnd(HttpHeader header) {
>   switch (header) {
>-  case HH_CONNECTION:
>-  case HH_KEEP_ALIVE:
>-  case HH_PROXY_AUTHENTICATE:
>-  case HH_PROXY_AUTHORIZATION:
>-  case HH_PROXY_CONNECTION:  // Note part of RFC... this is non-standard header
>-  case HH_TE:
>-  case HH_TRAILERS:
>-  case HH_TRANSFER_ENCODING:
>-  case HH_UPGRADE:
>-    return false;
>-  default:
>-    return true;
>+    case HH_CONNECTION:
>+    case HH_KEEP_ALIVE:
>+    case HH_PROXY_AUTHENTICATE:
>+    case HH_PROXY_AUTHORIZATION:
>+    case HH_PROXY_CONNECTION:  // Note part of RFC... this is non-standard
>+                               // header
>+    case HH_TE:
>+    case HH_TRAILERS:
>+    case HH_TRANSFER_ENCODING:
>+    case HH_UPGRADE:
>+      return false;
>+    default:
>+      return true;
>   }
> }
> 
> bool HttpHeaderIsCollapsible(HttpHeader header) {
>   switch (header) {
>-  case HH_SET_COOKIE:
>-  case HH_PROXY_AUTHENTICATE:
>-  case HH_WWW_AUTHENTICATE:
>-    return false;
>-  default:
>-    return true;
>+    case HH_SET_COOKIE:
>+    case HH_PROXY_AUTHENTICATE:
>+    case HH_WWW_AUTHENTICATE:
>+      return false;
>+    default:
>+      return true;
>   }
> }
> 
> bool HttpShouldKeepAlive(const HttpData& data) {
>   std::string connection;
>-  if ((data.hasHeader(HH_PROXY_CONNECTION, &connection)
>-      || data.hasHeader(HH_CONNECTION, &connection))) {
>+  if ((data.hasHeader(HH_PROXY_CONNECTION, &connection) ||
>+       data.hasHeader(HH_CONNECTION, &connection))) {
>     return (_stricmp(connection.c_str(), "Keep-Alive") == 0);
>   }
>   return (data.version >= HVER_1_1);
>@@ -206,16 +257,16 @@ bool HttpShouldKeepAlive(const HttpData& data) {
> 
> namespace {
> 
>-inline bool IsEndOfAttributeName(size_t pos, size_t len, const char * data) {
>+inline bool IsEndOfAttributeName(size_t pos, size_t len, const char* data) {
>   if (pos >= len)
>     return true;
>   if (isspace(static_cast<unsigned char>(data[pos])))
>     return true;
>   // The reason for this complexity is that some attributes may contain trailing
>   // equal signs (like base64 tokens in Negotiate auth headers)
>-  if ((pos+1 < len) && (data[pos] == '=') &&
>-      !isspace(static_cast<unsigned char>(data[pos+1])) &&
>-      (data[pos+1] != '=')) {
>+  if ((pos + 1 < len) && (data[pos] == '=') &&
>+      !isspace(static_cast<unsigned char>(data[pos + 1])) &&
>+      (data[pos + 1] != '=')) {
>     return true;
>   }
>   return false;
>@@ -223,7 +274,8 @@ inline bool IsEndOfAttributeName(size_t pos, size_t len, const char * data) {
> 
> }  // anonymous namespace
> 
>-void HttpParseAttributes(const char * data, size_t len,
>+void HttpParseAttributes(const char* data,
>+                         size_t len,
>                          HttpAttributeList& attributes) {
>   size_t pos = 0;
>   while (true) {
>@@ -247,7 +299,7 @@ void HttpParseAttributes(const char * data, size_t len,
> 
>     // Attribute has value?
>     if ((pos < len) && (data[pos] == '=')) {
>-      ++pos; // Skip '='
>+      ++pos;  // Skip '='
>       // Check if quoted value
>       if ((pos < len) && (data[pos] == '"')) {
>         while (++pos < len) {
>@@ -260,16 +312,16 @@ void HttpParseAttributes(const char * data, size_t len,
>           attribute.second.append(1, data[pos]);
>         }
>       } else {
>-        while ((pos < len) &&
>-            !isspace(static_cast<unsigned char>(data[pos])) &&
>-            (data[pos] != ',')) {
>+        while ((pos < len) && !isspace(static_cast<unsigned char>(data[pos])) &&
>+               (data[pos] != ',')) {
>           attribute.second.append(1, data[pos++]);
>         }
>       }
>     }
> 
>     attributes.push_back(attribute);
>-    if ((pos < len) && (data[pos] == ',')) ++pos; // Skip ','
>+    if ((pos < len) && (data[pos] == ','))
>+      ++pos;  // Skip ','
>   }
> }
> 
>@@ -304,15 +356,13 @@ bool HttpHasNthAttribute(HttpAttributeList& attributes,
> 
> bool HttpDateToSeconds(const std::string& date, time_t* seconds) {
>   const char* const kTimeZones[] = {
>-    "UT", "GMT", "EST", "EDT", "CST", "CDT", "MST", "MDT", "PST", "PDT",
>-    "A", "B", "C", "D", "E", "F", "G", "H", "I", "K", "L", "M",
>-    "N", "O", "P", "Q", "R", "S", "T", "U", "V", "W", "X", "Y"
>-  };
>+      "UT",  "GMT", "EST", "EDT", "CST", "CDT", "MST", "MDT", "PST",
>+      "PDT", "A",   "B",   "C",   "D",   "E",   "F",   "G",   "H",
>+      "I",   "K",   "L",   "M",   "N",   "O",   "P",   "Q",   "R",
>+      "S",   "T",   "U",   "V",   "W",   "X",   "Y"};
>   const int kTimeZoneOffsets[] = {
>-     0,  0, -5, -4, -6, -5, -7, -6, -8, -7,
>-    -1, -2, -3, -4, -5, -6, -7, -8, -9, -10, -11, -12,
>-     1,  2,  3,  4,  5,  6,  7,  8,  9,  10,  11,  12
>-  };
>+      0,  0,  -5,  -4,  -6,  -5, -7, -6, -8, -7, -1, -2, -3, -4, -5, -6, -7,
>+      -8, -9, -10, -11, -12, 1,  2,  3,  4,  5,  6,  7,  8,  9,  10, 11, 12};
> 
>   RTC_DCHECK(nullptr != seconds);
>   struct tm tval;
>@@ -321,28 +371,48 @@ bool HttpDateToSeconds(const std::string& date, time_t* seconds) {
>   memset(month, 0, sizeof(month));
>   memset(zone, 0, sizeof(zone));
> 
>-  if (7 != sscanf(date.c_str(), "%*3s, %d %3s %d %d:%d:%d %5c",
>-                  &tval.tm_mday, month, &tval.tm_year,
>-                  &tval.tm_hour, &tval.tm_min, &tval.tm_sec, zone)) {
>+  if (7 != sscanf(date.c_str(), "%*3s, %d %3s %d %d:%d:%d %5c", &tval.tm_mday,
>+                  month, &tval.tm_year, &tval.tm_hour, &tval.tm_min,
>+                  &tval.tm_sec, zone)) {
>     return false;
>   }
>   switch (toupper(month[2])) {
>-  case 'N': tval.tm_mon = (month[1] == 'A') ? 0 : 5; break;
>-  case 'B': tval.tm_mon = 1; break;
>-  case 'R': tval.tm_mon = (month[0] == 'M') ? 2 : 3; break;
>-  case 'Y': tval.tm_mon = 4; break;
>-  case 'L': tval.tm_mon = 6; break;
>-  case 'G': tval.tm_mon = 7; break;
>-  case 'P': tval.tm_mon = 8; break;
>-  case 'T': tval.tm_mon = 9; break;
>-  case 'V': tval.tm_mon = 10; break;
>-  case 'C': tval.tm_mon = 11; break;
>+    case 'N':
>+      tval.tm_mon = (month[1] == 'A') ? 0 : 5;
>+      break;
>+    case 'B':
>+      tval.tm_mon = 1;
>+      break;
>+    case 'R':
>+      tval.tm_mon = (month[0] == 'M') ? 2 : 3;
>+      break;
>+    case 'Y':
>+      tval.tm_mon = 4;
>+      break;
>+    case 'L':
>+      tval.tm_mon = 6;
>+      break;
>+    case 'G':
>+      tval.tm_mon = 7;
>+      break;
>+    case 'P':
>+      tval.tm_mon = 8;
>+      break;
>+    case 'T':
>+      tval.tm_mon = 9;
>+      break;
>+    case 'V':
>+      tval.tm_mon = 10;
>+      break;
>+    case 'C':
>+      tval.tm_mon = 11;
>+      break;
>   }
>   tval.tm_year -= 1900;
>   time_t gmt, non_gmt = mktime(&tval);
>   if ((zone[0] == '+') || (zone[0] == '-')) {
>-    if (!isdigit(zone[1]) || !isdigit(zone[2])
>-        || !isdigit(zone[3]) || !isdigit(zone[4])) {
>+    if (!isdigit(zone[1]) || !isdigit(zone[2]) || !isdigit(zone[3]) ||
>+        !isdigit(zone[4])) {
>       return false;
>     }
>     int hours = (zone[1] - '0') * 10 + (zone[2] - '0');
>@@ -356,9 +426,10 @@ bool HttpDateToSeconds(const std::string& date, time_t* seconds) {
>     }
>     gmt = non_gmt + kTimeZoneOffsets[zindex] * 60 * 60;
>   }
>-  // TODO: Android should support timezone, see b/2441195
>-#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS) || defined(WEBRTC_ANDROID) || defined(BSD)
>-  tm *tm_for_timezone = localtime(&gmt);
>+// TODO: Android should support timezone, see b/2441195
>+#if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS) || defined(WEBRTC_ANDROID) || \
>+    defined(BSD)
>+  tm* tm_for_timezone = localtime(&gmt);
>   *seconds = gmt + tm_for_timezone->tm_gmtoff;
> #else
> #if defined(_MSC_VER) && _MSC_VER >= 1900
>@@ -371,21 +442,19 @@ bool HttpDateToSeconds(const std::string& date, time_t* seconds) {
> }
> 
> std::string HttpAddress(const SocketAddress& address, bool secure) {
>-  return (address.port() == HttpDefaultPort(secure))
>-          ? address.hostname() : address.ToString();
>+  return (address.port() == HttpDefaultPort(secure)) ? address.hostname()
>+                                                     : address.ToString();
> }
> 
> //////////////////////////////////////////////////////////////////////
> // HttpData
> //////////////////////////////////////////////////////////////////////
> 
>-HttpData::HttpData() : version(HVER_1_1) {
>-}
>+HttpData::HttpData() : version(HVER_1_1) {}
> 
> HttpData::~HttpData() = default;
> 
>-void
>-HttpData::clear(bool release_document) {
>+void HttpData::clear(bool release_document) {
>   // Clear headers first, since releasing a document may have far-reaching
>   // effects.
>   headers_.clear();
>@@ -394,19 +463,15 @@ HttpData::clear(bool release_document) {
>   }
> }
> 
>-void
>-HttpData::copy(const HttpData& src) {
>-  headers_ = src.headers_;
>-}
>-
>-void
>-HttpData::changeHeader(const std::string& name, const std::string& value,
>-                       HeaderCombine combine) {
>+void HttpData::changeHeader(const std::string& name,
>+                            const std::string& value,
>+                            HeaderCombine combine) {
>   if (combine == HC_AUTO) {
>     HttpHeader header;
>     // Unrecognized headers are collapsible
>     combine = !FromString(header, name) || HttpHeaderIsCollapsible(header)
>-              ? HC_YES : HC_NO;
>+                  ? HC_YES
>+                  : HC_NO;
>   } else if (combine == HC_REPLACE) {
>     headers_.erase(name);
>     combine = HC_NO;
>@@ -435,8 +500,7 @@ HttpData::iterator HttpData::clearHeader(iterator header) {
>   return header;
> }
> 
>-bool
>-HttpData::hasHeader(const std::string& name, std::string* value) const {
>+bool HttpData::hasHeader(const std::string& name, std::string* value) const {
>   HeaderMap::const_iterator it = headers_.find(name);
>   if (it == headers_.end()) {
>     return false;
>@@ -472,38 +536,27 @@ void HttpData::setDocumentAndLength(StreamInterface* document) {
> // HttpRequestData
> //
> 
>-void
>-HttpRequestData::clear(bool release_document) {
>-  verb = HV_GET;
>+void HttpRequestData::clear(bool release_document) {
>   path.clear();
>   HttpData::clear(release_document);
> }
> 
>-void
>-HttpRequestData::copy(const HttpRequestData& src) {
>-  verb = src.verb;
>-  path = src.path;
>-  HttpData::copy(src);
>-}
>-
>-size_t
>-HttpRequestData::formatLeader(char* buffer, size_t size) const {
>+size_t HttpRequestData::formatLeader(char* buffer, size_t size) const {
>   RTC_DCHECK(path.find(' ') == std::string::npos);
>-  return sprintfn(buffer, size, "%s %.*s HTTP/%s", ToString(verb), path.size(),
>-                  path.data(), ToString(version));
>+  return sprintfn(buffer, size, "GET %.*s HTTP/%s", path.size(), path.data(),
>+                  ToString(version));
> }
> 
>-HttpError
>-HttpRequestData::parseLeader(const char* line, size_t len) {
>+HttpError HttpRequestData::parseLeader(const char* line, size_t len) {
>   unsigned int vmajor, vminor;
>   int vend, dstart, dend;
>   // sscanf isn't safe with strings that aren't null-terminated, and there is
>   // no guarantee that |line| is. Create a local copy that is null-terminated.
>   std::string line_str(line, len);
>   line = line_str.c_str();
>-  if ((sscanf(line, "%*s%n %n%*s%n HTTP/%u.%u",
>-              &vend, &dstart, &dend, &vmajor, &vminor) != 2)
>-      || (vmajor != 1)) {
>+  if ((sscanf(line, "%*s%n %n%*s%n HTTP/%u.%u", &vend, &dstart, &dend, &vmajor,
>+              &vminor) != 2) ||
>+      (vmajor != 1)) {
>     return HE_PROTOCOL;
>   }
>   if (vminor == 0) {
>@@ -513,17 +566,14 @@ HttpRequestData::parseLeader(const char* line, size_t len) {
>   } else {
>     return HE_PROTOCOL;
>   }
>-  std::string sverb(line, vend);
>-  if (!FromString(verb, sverb.c_str())) {
>-    return HE_PROTOCOL; // !?! HC_METHOD_NOT_SUPPORTED?
>+  if (vend != 3 || memcmp(line, "GET", 3)) {
>+    return HE_PROTOCOL;  // !?! HC_METHOD_NOT_SUPPORTED?
>   }
>   path.assign(line + dstart, line + dend);
>   return HE_NONE;
> }
> 
> bool HttpRequestData::getAbsoluteUri(std::string* uri) const {
>-  if (HV_CONNECT == verb)
>-    return false;
>   Url<char> url(path);
>   if (url.valid()) {
>     uri->assign(path);
>@@ -539,10 +589,7 @@ bool HttpRequestData::getAbsoluteUri(std::string* uri) const {
> }
> 
> bool HttpRequestData::getRelativeUri(std::string* host,
>-                                     std::string* path) const
>-{
>-  if (HV_CONNECT == verb)
>-    return false;
>+                                     std::string* path) const {
>   Url<char> url(this->path);
>   if (url.valid()) {
>     host->assign(url.address());
>@@ -559,60 +606,34 @@ bool HttpRequestData::getRelativeUri(std::string* host,
> // HttpResponseData
> //
> 
>-void
>-HttpResponseData::clear(bool release_document) {
>+void HttpResponseData::clear(bool release_document) {
>   scode = HC_INTERNAL_SERVER_ERROR;
>   message.clear();
>   HttpData::clear(release_document);
> }
> 
>-void
>-HttpResponseData::copy(const HttpResponseData& src) {
>-  scode = src.scode;
>-  message = src.message;
>-  HttpData::copy(src);
>-}
>-
> void HttpResponseData::set_success(uint32_t scode) {
>   this->scode = scode;
>   message.clear();
>   setHeader(HH_CONTENT_LENGTH, "0", false);
> }
> 
>-void HttpResponseData::set_success(const std::string& content_type,
>-                                   StreamInterface* document,
>-                                   uint32_t scode) {
>-  this->scode = scode;
>-  message.erase(message.begin(), message.end());
>-  setContent(content_type, document);
>-}
>-
>-void HttpResponseData::set_redirect(const std::string& location,
>-                                    uint32_t scode) {
>-  this->scode = scode;
>-  message.clear();
>-  setHeader(HH_LOCATION, location);
>-  setHeader(HH_CONTENT_LENGTH, "0", false);
>-}
>-
> void HttpResponseData::set_error(uint32_t scode) {
>   this->scode = scode;
>   message.clear();
>   setHeader(HH_CONTENT_LENGTH, "0", false);
> }
> 
>-size_t
>-HttpResponseData::formatLeader(char* buffer, size_t size) const {
>+size_t HttpResponseData::formatLeader(char* buffer, size_t size) const {
>   size_t len = sprintfn(buffer, size, "HTTP/%s %lu", ToString(version), scode);
>   if (!message.empty()) {
>-    len += sprintfn(buffer + len, size - len, " %.*s",
>-                    message.size(), message.data());
>+    len += sprintfn(buffer + len, size - len, " %.*s", message.size(),
>+                    message.data());
>   }
>   return len;
> }
> 
>-HttpError
>-HttpResponseData::parseLeader(const char* line, size_t len) {
>+HttpError HttpResponseData::parseLeader(const char* line, size_t len) {
>   size_t pos = 0;
>   unsigned int vmajor, vminor, temp_scode;
>   int temp_pos;
>@@ -620,16 +641,15 @@ HttpResponseData::parseLeader(const char* line, size_t len) {
>   // no guarantee that |line| is. Create a local copy that is null-terminated.
>   std::string line_str(line, len);
>   line = line_str.c_str();
>-  if (sscanf(line, "HTTP %u%n",
>-             &temp_scode, &temp_pos) == 1) {
>+  if (sscanf(line, "HTTP %u%n", &temp_scode, &temp_pos) == 1) {
>     // This server's response has no version. :( NOTE: This happens for every
>     // response to requests made from Chrome plugins, regardless of the server's
>     // behaviour.
>     RTC_LOG(LS_VERBOSE) << "HTTP version missing from response";
>     version = HVER_UNKNOWN;
>-  } else if ((sscanf(line, "HTTP/%u.%u %u%n",
>-                     &vmajor, &vminor, &temp_scode, &temp_pos) == 3)
>-             && (vmajor == 1)) {
>+  } else if ((sscanf(line, "HTTP/%u.%u %u%n", &vmajor, &vminor, &temp_scode,
>+                     &temp_pos) == 3) &&
>+             (vmajor == 1)) {
>     // This server's response does have a version.
>     if (vminor == 0) {
>       version = HVER_1_0;
>@@ -643,7 +663,8 @@ HttpResponseData::parseLeader(const char* line, size_t len) {
>   }
>   scode = temp_scode;
>   pos = static_cast<size_t>(temp_pos);
>-  while ((pos < len) && isspace(static_cast<unsigned char>(line[pos]))) ++pos;
>+  while ((pos < len) && isspace(static_cast<unsigned char>(line[pos])))
>+    ++pos;
>   message.assign(line + pos, len - pos);
>   return HE_NONE;
> }
>@@ -655,7 +676,7 @@ HttpResponseData::parseLeader(const char* line, size_t len) {
> std::string quote(const std::string& str) {
>   std::string result;
>   result.push_back('"');
>-  for (size_t i=0; i<str.size(); ++i) {
>+  for (size_t i = 0; i < str.size(); ++i) {
>     if ((str[i] == '"') || (str[i] == '\\'))
>       result.push_back('\\');
>     result.push_back(str[i]);
>@@ -672,24 +693,29 @@ struct NegotiateAuthContext : public HttpAuthContext {
>   bool specified_credentials;
> 
>   NegotiateAuthContext(const std::string& auth, CredHandle c1, CtxtHandle c2)
>-  : HttpAuthContext(auth), cred(c1), ctx(c2), steps(0),
>-    specified_credentials(false)
>-  { }
>+      : HttpAuthContext(auth),
>+        cred(c1),
>+        ctx(c2),
>+        steps(0),
>+        specified_credentials(false) {}
> 
>   ~NegotiateAuthContext() override {
>     DeleteSecurityContext(&ctx);
>     FreeCredentialsHandle(&cred);
>   }
> };
>-#endif // WEBRTC_WIN
>-
>-HttpAuthResult HttpAuthenticate(
>-  const char * challenge, size_t len,
>-  const SocketAddress& server,
>-  const std::string& method, const std::string& uri,
>-  const std::string& username, const CryptString& password,
>-  HttpAuthContext *& context, std::string& response, std::string& auth_method)
>-{
>+#endif  // WEBRTC_WIN
>+
>+HttpAuthResult HttpAuthenticate(const char* challenge,
>+                                size_t len,
>+                                const SocketAddress& server,
>+                                const std::string& method,
>+                                const std::string& uri,
>+                                const std::string& username,
>+                                const CryptString& password,
>+                                HttpAuthContext*& context,
>+                                std::string& response,
>+                                std::string& auth_method) {
>   HttpAttributeList args;
>   HttpParseAttributes(challenge, len, args);
>   HttpHasNthAttribute(args, 0, &auth_method, nullptr);
>@@ -700,16 +726,18 @@ HttpAuthResult HttpAuthenticate(
>   // BASIC
>   if (_stricmp(auth_method.c_str(), "basic") == 0) {
>     if (context)
>-      return HAR_CREDENTIALS; // Bad credentials
>+      return HAR_CREDENTIALS;  // Bad credentials
>     if (username.empty())
>-      return HAR_CREDENTIALS; // Missing credentials
>+      return HAR_CREDENTIALS;  // Missing credentials
> 
>     context = new HttpAuthContext(auth_method);
> 
>-    // TODO: convert sensitive to a secure buffer that gets securely deleted
>-    //std::string decoded = username + ":" + password;
>+    // TODO(bugs.webrtc.org/8905): Convert sensitive to a CryptString and also
>+    // return response as CryptString so contents get securely deleted
>+    // automatically.
>+    // std::string decoded = username + ":" + password;
>     size_t len = username.size() + password.GetLength() + 2;
>-    char * sensitive = new char[len];
>+    char* sensitive = new char[len];
>     size_t pos = strcpyn(sensitive, len, username.data(), username.size());
>     pos += strcpyn(sensitive + pos, len - pos, ":");
>     password.CopyTo(sensitive + pos, true);
>@@ -718,17 +746,17 @@ HttpAuthResult HttpAuthenticate(
>     response.append(" ");
>     // TODO: create a sensitive-source version of Base64::encode
>     response.append(Base64::Encode(sensitive));
>-    memset(sensitive, 0, len);
>-    delete [] sensitive;
>+    ExplicitZeroMemory(sensitive, len);
>+    delete[] sensitive;
>     return HAR_RESPONSE;
>   }
> 
>   // DIGEST
>   if (_stricmp(auth_method.c_str(), "digest") == 0) {
>     if (context)
>-      return HAR_CREDENTIALS; // Bad credentials
>+      return HAR_CREDENTIALS;  // Bad credentials
>     if (username.empty())
>-      return HAR_CREDENTIALS; // Missing credentials
>+      return HAR_CREDENTIALS;  // Missing credentials
> 
>     context = new HttpAuthContext(auth_method);
> 
>@@ -744,10 +772,12 @@ HttpAuthResult HttpAuthenticate(
>     bool has_qop = HttpHasAttribute(args, "qop", &qop);
>     bool has_opaque = HttpHasAttribute(args, "opaque", &opaque);
> 
>-    // TODO: convert sensitive to be secure buffer
>-    //std::string A1 = username + ":" + realm + ":" + password;
>+    // TODO(bugs.webrtc.org/8905): Convert sensitive to a CryptString and also
>+    // return response as CryptString so contents get securely deleted
>+    // automatically.
>+    // std::string A1 = username + ":" + realm + ":" + password;
>     size_t len = username.size() + realm.size() + password.GetLength() + 3;
>-    char * sensitive = new char[len];  // A1
>+    char* sensitive = new char[len];  // A1
>     size_t pos = strcpyn(sensitive, len, username.data(), username.size());
>     pos += strcpyn(sensitive + pos, len - pos, ":");
>     pos += strcpyn(sensitive + pos, len - pos, realm.c_str());
>@@ -763,8 +793,8 @@ HttpAuthResult HttpAuthenticate(
>       middle = nonce;
>     }
>     std::string HA1 = MD5(sensitive);
>-    memset(sensitive, 0, len);
>-    delete [] sensitive;
>+    ExplicitZeroMemory(sensitive, len);
>+    delete[] sensitive;
>     std::string HA2 = MD5(A2);
>     std::string dig_response = MD5(HA1 + ":" + middle + ":" + HA2);
> 
>@@ -776,7 +806,7 @@ HttpAuthResult HttpAuthenticate(
>     ss << ", uri=" << quote(uri);
>     if (has_qop) {
>       ss << ", qop=" << qop;
>-      ss << ", nc="  << ncount;
>+      ss << ", nc=" << ncount;
>       ss << ", cnonce=" << quote(cnonce);
>     }
>     ss << ", response=\"" << dig_response << "\"";
>@@ -796,7 +826,7 @@ HttpAuthResult HttpAuthenticate(
>     const size_t MAX_MESSAGE = 12000, MAX_SPN = 256;
>     char out_buf[MAX_MESSAGE], spn[MAX_SPN];
> 
>-#if 0 // Requires funky windows versions
>+#if 0  // Requires funky windows versions
>     DWORD len = MAX_SPN;
>     if (DsMakeSpn("HTTP", server.HostAsURIString().c_str(), nullptr,
>                   server.port(),
>@@ -809,25 +839,24 @@ HttpAuthResult HttpAuthenticate(
> #endif
> 
>     SecBuffer out_sec;
>-    out_sec.pvBuffer   = out_buf;
>-    out_sec.cbBuffer   = sizeof(out_buf);
>+    out_sec.pvBuffer = out_buf;
>+    out_sec.cbBuffer = sizeof(out_buf);
>     out_sec.BufferType = SECBUFFER_TOKEN;
> 
>     SecBufferDesc out_buf_desc;
>     out_buf_desc.ulVersion = 0;
>-    out_buf_desc.cBuffers  = 1;
>-    out_buf_desc.pBuffers  = &out_sec;
>+    out_buf_desc.cBuffers = 1;
>+    out_buf_desc.pBuffers = &out_sec;
> 
>     const ULONG NEG_FLAGS_DEFAULT =
>-      //ISC_REQ_ALLOCATE_MEMORY
>-      ISC_REQ_CONFIDENTIALITY
>-      //| ISC_REQ_EXTENDED_ERROR
>-      //| ISC_REQ_INTEGRITY
>-      | ISC_REQ_REPLAY_DETECT
>-      | ISC_REQ_SEQUENCE_DETECT
>-      //| ISC_REQ_STREAM
>-      //| ISC_REQ_USE_SUPPLIED_CREDS
>-      ;
>+        // ISC_REQ_ALLOCATE_MEMORY
>+        ISC_REQ_CONFIDENTIALITY
>+        //| ISC_REQ_EXTENDED_ERROR
>+        //| ISC_REQ_INTEGRITY
>+        | ISC_REQ_REPLAY_DETECT | ISC_REQ_SEQUENCE_DETECT
>+        //| ISC_REQ_STREAM
>+        //| ISC_REQ_USE_SUPPLIED_CREDS
>+        ;
> 
>     ::TimeStamp lifetime;
>     SECURITY_STATUS ret = S_OK;
>@@ -838,7 +867,7 @@ HttpAuthResult HttpAuthenticate(
> 
>     // uint32_t now = Time();
> 
>-    NegotiateAuthContext * neg = static_cast<NegotiateAuthContext *>(context);
>+    NegotiateAuthContext* neg = static_cast<NegotiateAuthContext*>(context);
>     if (neg) {
>       const size_t max_steps = 10;
>       if (++neg->steps >= max_steps) {
>@@ -853,21 +882,21 @@ HttpAuthResult HttpAuthenticate(
>           Base64::Decode(challenge, Base64::DO_STRICT, &decoded_challenge,
>                          nullptr)) {
>         SecBuffer in_sec;
>-        in_sec.pvBuffer   = const_cast<char *>(decoded_challenge.data());
>-        in_sec.cbBuffer   = static_cast<unsigned long>(decoded_challenge.size());
>+        in_sec.pvBuffer = const_cast<char*>(decoded_challenge.data());
>+        in_sec.cbBuffer = static_cast<unsigned long>(decoded_challenge.size());
>         in_sec.BufferType = SECBUFFER_TOKEN;
> 
>         SecBufferDesc in_buf_desc;
>         in_buf_desc.ulVersion = 0;
>-        in_buf_desc.cBuffers  = 1;
>-        in_buf_desc.pBuffers  = &in_sec;
>+        in_buf_desc.cBuffers = 1;
>+        in_buf_desc.pBuffers = &in_sec;
> 
>-        ret = InitializeSecurityContextA(&neg->cred, &neg->ctx, spn, flags, 0, SECURITY_NATIVE_DREP, &in_buf_desc, 0, &neg->ctx, &out_buf_desc, &ret_flags, &lifetime);
>-        // RTC_LOG(INFO) << "$$$ InitializeSecurityContext @ " <<
>-        // TimeSince(now);
>+        ret = InitializeSecurityContextA(
>+            &neg->cred, &neg->ctx, spn, flags, 0, SECURITY_NATIVE_DREP,
>+            &in_buf_desc, 0, &neg->ctx, &out_buf_desc, &ret_flags, &lifetime);
>         if (FAILED(ret)) {
>           RTC_LOG(LS_ERROR) << "InitializeSecurityContext returned: "
>-                            << ErrorName(ret, SECURITY_ERRORS);
>+                            << GetErrorName(ret, SECURITY_ERRORS);
>           return HAR_ERROR;
>         }
>       } else if (neg->specified_credentials) {
>@@ -882,11 +911,11 @@ HttpAuthResult HttpAuthenticate(
> 
>     if (!neg) {
>       unsigned char userbuf[256], passbuf[256], domainbuf[16];
>-      SEC_WINNT_AUTH_IDENTITY_A auth_id, * pauth_id = 0;
>+      SEC_WINNT_AUTH_IDENTITY_A auth_id, *pauth_id = 0;
>       if (specify_credentials) {
>         memset(&auth_id, 0, sizeof(auth_id));
>-        size_t len = password.GetLength()+1;
>-        char * sensitive = new char[len];
>+        size_t len = password.GetLength() + 1;
>+        char* sensitive = new char[len];
>         password.CopyTo(sensitive, true);
>         std::string::size_type pos = username.find('\\');
>         if (pos == std::string::npos) {
>@@ -914,8 +943,8 @@ HttpAuthResult HttpAuthenticate(
>           memcpy(passbuf, sensitive, auth_id.PasswordLength);
>           passbuf[auth_id.PasswordLength] = 0;
>         }
>-        memset(sensitive, 0, len);
>-        delete [] sensitive;
>+        ExplicitZeroMemory(sensitive, len);
>+        delete[] sensitive;
>         auth_id.User = userbuf;
>         auth_id.Domain = domainbuf;
>         auth_id.Password = passbuf;
>@@ -931,21 +960,21 @@ HttpAuthResult HttpAuthenticate(
>       ret = AcquireCredentialsHandleA(
>           0, const_cast<char*>(want_negotiate ? NEGOSSP_NAME_A : NTLMSP_NAME_A),
>           SECPKG_CRED_OUTBOUND, 0, pauth_id, 0, 0, &cred, &lifetime);
>-      // RTC_LOG(INFO) << "$$$ AcquireCredentialsHandle @ " << TimeSince(now);
>       if (ret != SEC_E_OK) {
>         RTC_LOG(LS_ERROR) << "AcquireCredentialsHandle error: "
>-                          << ErrorName(ret, SECURITY_ERRORS);
>+                          << GetErrorName(ret, SECURITY_ERRORS);
>         return HAR_IGNORE;
>       }
> 
>-      //CSecBufferBundle<5, CSecBufferBase::FreeSSPI> sb_out;
>+      // CSecBufferBundle<5, CSecBufferBase::FreeSSPI> sb_out;
> 
>       CtxtHandle ctx;
>-      ret = InitializeSecurityContextA(&cred, 0, spn, flags, 0, SECURITY_NATIVE_DREP, 0, 0, &ctx, &out_buf_desc, &ret_flags, &lifetime);
>-      // RTC_LOG(INFO) << "$$$ InitializeSecurityContext @ " << TimeSince(now);
>+      ret = InitializeSecurityContextA(&cred, 0, spn, flags, 0,
>+                                       SECURITY_NATIVE_DREP, 0, 0, &ctx,
>+                                       &out_buf_desc, &ret_flags, &lifetime);
>       if (FAILED(ret)) {
>         RTC_LOG(LS_ERROR) << "InitializeSecurityContext returned: "
>-                          << ErrorName(ret, SECURITY_ERRORS);
>+                          << GetErrorName(ret, SECURITY_ERRORS);
>         FreeCredentialsHandle(&cred);
>         return HAR_IGNORE;
>       }
>@@ -956,18 +985,16 @@ HttpAuthResult HttpAuthenticate(
>       neg->steps = steps;
>     }
> 
>-    if ((ret == SEC_I_COMPLETE_NEEDED) || (ret == SEC_I_COMPLETE_AND_CONTINUE)) {
>+    if ((ret == SEC_I_COMPLETE_NEEDED) ||
>+        (ret == SEC_I_COMPLETE_AND_CONTINUE)) {
>       ret = CompleteAuthToken(&neg->ctx, &out_buf_desc);
>-      // RTC_LOG(INFO) << "$$$ CompleteAuthToken @ " << TimeSince(now);
>       RTC_LOG(LS_VERBOSE) << "CompleteAuthToken returned: "
>-                          << ErrorName(ret, SECURITY_ERRORS);
>+                          << GetErrorName(ret, SECURITY_ERRORS);
>       if (FAILED(ret)) {
>         return HAR_ERROR;
>       }
>     }
> 
>-    // RTC_LOG(INFO) << "$$$ NEGOTIATE took " << TimeSince(now) << "ms";
>-
>     std::string decoded(out_buf, out_buf + out_sec.cbBuffer);
>     response = auth_method;
>     response.append(" ");
>@@ -975,11 +1002,11 @@ HttpAuthResult HttpAuthenticate(
>     return HAR_RESPONSE;
>   }
> #endif
>-#endif // WEBRTC_WIN
>+#endif  // WEBRTC_WIN
> 
>   return HAR_IGNORE;
> }
> 
> //////////////////////////////////////////////////////////////////////
> 
>-} // namespace rtc
>+}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/httpcommon.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/httpcommon.h
>index 2897c5a5a66..11c01caf105 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/httpcommon.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/httpcommon.h
>@@ -15,7 +15,7 @@
> #include <memory>
> #include <string>
> #include <vector>
>-#include "rtc_base/basictypes.h"
>+
> #include "rtc_base/checks.h"
> #include "rtc_base/stream.h"
> #include "rtc_base/stringutils.h"
>@@ -31,52 +31,24 @@ class SocketAddress;
> 
> enum HttpCode {
>   HC_OK = 200,
>-  HC_NON_AUTHORITATIVE = 203,
>-  HC_NO_CONTENT = 204,
>-  HC_PARTIAL_CONTENT = 206,
>-
>-  HC_MULTIPLE_CHOICES = 300,
>-  HC_MOVED_PERMANENTLY = 301,
>-  HC_FOUND = 302,
>-  HC_SEE_OTHER = 303,
>-  HC_NOT_MODIFIED = 304,
>-  HC_MOVED_TEMPORARILY = 307,
>-
>-  HC_BAD_REQUEST = 400,
>-  HC_UNAUTHORIZED = 401,
>-  HC_FORBIDDEN = 403,
>-  HC_NOT_FOUND = 404,
>-  HC_PROXY_AUTHENTICATION_REQUIRED = 407,
>-  HC_GONE = 410,
>-
>   HC_INTERNAL_SERVER_ERROR = 500,
>-  HC_NOT_IMPLEMENTED = 501,
>-  HC_SERVICE_UNAVAILABLE = 503,
> };
> 
>-enum HttpVersion {
>-  HVER_1_0, HVER_1_1, HVER_UNKNOWN,
>-  HVER_LAST = HVER_UNKNOWN
>-};
>-
>-enum HttpVerb {
>-  HV_GET, HV_POST, HV_PUT, HV_DELETE, HV_CONNECT, HV_HEAD,
>-  HV_LAST = HV_HEAD
>-};
>+enum HttpVersion { HVER_1_0, HVER_1_1, HVER_UNKNOWN, HVER_LAST = HVER_UNKNOWN };
> 
> enum HttpError {
>   HE_NONE,
>-  HE_PROTOCOL,            // Received non-valid HTTP data
>-  HE_DISCONNECTED,        // Connection closed unexpectedly
>-  HE_OVERFLOW,            // Received too much data for internal buffers
>-  HE_CONNECT_FAILED,      // The socket failed to connect.
>-  HE_SOCKET_ERROR,        // An error occurred on a connected socket
>-  HE_SHUTDOWN,            // Http object is being destroyed
>-  HE_OPERATION_CANCELLED, // Connection aborted locally
>-  HE_AUTH,                // Proxy Authentication Required
>-  HE_CERTIFICATE_EXPIRED, // During SSL negotiation
>-  HE_STREAM,              // Problem reading or writing to the document
>-  HE_CACHE,               // Problem reading from cache
>+  HE_PROTOCOL,             // Received non-valid HTTP data
>+  HE_DISCONNECTED,         // Connection closed unexpectedly
>+  HE_OVERFLOW,             // Received too much data for internal buffers
>+  HE_CONNECT_FAILED,       // The socket failed to connect.
>+  HE_SOCKET_ERROR,         // An error occurred on a connected socket
>+  HE_SHUTDOWN,             // Http object is being destroyed
>+  HE_OPERATION_CANCELLED,  // Connection aborted locally
>+  HE_AUTH,                 // Proxy Authentication Required
>+  HE_CERTIFICATE_EXPIRED,  // During SSL negotiation
>+  HE_STREAM,               // Problem reading or writing to the document
>+  HE_CACHE,                // Problem reading from cache
>   HE_DEFAULT
> };
> 
>@@ -119,37 +91,12 @@ const uint16_t HTTP_SECURE_PORT = 443;
> // Utility Functions
> //////////////////////////////////////////////////////////////////////
> 
>-inline HttpError mkerr(HttpError err, HttpError def_err = HE_DEFAULT) {
>-  return (err != HE_NONE) ? err : def_err;
>-}
>-
> const char* ToString(HttpVersion version);
> bool FromString(HttpVersion& version, const std::string& str);
> 
>-const char* ToString(HttpVerb verb);
>-bool FromString(HttpVerb& verb, const std::string& str);
>-
> const char* ToString(HttpHeader header);
> bool FromString(HttpHeader& header, const std::string& str);
> 
>-inline bool HttpCodeIsInformational(uint32_t code) {
>-  return ((code / 100) == 1);
>-}
>-inline bool HttpCodeIsSuccessful(uint32_t code) {
>-  return ((code / 100) == 2);
>-}
>-inline bool HttpCodeIsRedirection(uint32_t code) {
>-  return ((code / 100) == 3);
>-}
>-inline bool HttpCodeIsClientError(uint32_t code) {
>-  return ((code / 100) == 4);
>-}
>-inline bool HttpCodeIsServerError(uint32_t code) {
>-  return ((code / 100) == 5);
>-}
>-
>-bool HttpCodeHasBody(uint32_t code);
>-bool HttpCodeIsCacheable(uint32_t code);
> bool HttpHeaderIsEndToEnd(HttpHeader header);
> bool HttpHeaderIsCollapsible(HttpHeader header);
> 
>@@ -158,7 +105,8 @@ bool HttpShouldKeepAlive(const HttpData& data);
> 
> typedef std::pair<std::string, std::string> HttpAttribute;
> typedef std::vector<HttpAttribute> HttpAttributeList;
>-void HttpParseAttributes(const char * data, size_t len,
>+void HttpParseAttributes(const char* data,
>+                         size_t len,
>                          HttpAttributeList& attributes);
> bool HttpHasAttribute(const HttpAttributeList& attributes,
>                       const std::string& name,
>@@ -192,9 +140,9 @@ std::string quote(const std::string& str);
> // Url
> //////////////////////////////////////////////////////////////////////
> 
>-template<class CTYPE>
>+template <class CTYPE>
> class Url {
>-public:
>+ public:
>   typedef typename Traits<CTYPE>::string string;
> 
>   // TODO: Implement Encode/Decode
>@@ -218,25 +166,29 @@ public:
>     query_.clear();
>   }
> 
>-  void set_url(const string& val) {
>-    do_set_url(val.c_str(), val.size());
>-  }
>+  void set_url(const string& val) { do_set_url(val.c_str(), val.size()); }
>   string url() const {
>-    string val; do_get_url(&val); return val;
>+    string val;
>+    do_get_url(&val);
>+    return val;
>   }
> 
>   void set_address(const string& val) {
>     do_set_address(val.c_str(), val.size());
>   }
>   string address() const {
>-    string val; do_get_address(&val); return val;
>+    string val;
>+    do_get_address(&val);
>+    return val;
>   }
> 
>   void set_full_path(const string& val) {
>     do_set_full_path(val.c_str(), val.size());
>   }
>   string full_path() const {
>-    string val; do_get_full_path(&val); return val;
>+    string val;
>+    do_get_full_path(&val);
>+    return val;
>   }
> 
>   void set_host(const string& val) { host_ = val; }
>@@ -266,7 +218,7 @@ public:
> 
>   bool get_attribute(const string& name, string* value) const;
> 
>-private:
>+ private:
>   void do_set_url(const CTYPE* val, size_t len);
>   void do_set_address(const CTYPE* val, size_t len);
>   void do_set_full_path(const CTYPE* val, size_t len);
>@@ -295,13 +247,16 @@ struct HttpData {
>   HttpData();
> 
>   enum HeaderCombine { HC_YES, HC_NO, HC_AUTO, HC_REPLACE, HC_NEW };
>-  void changeHeader(const std::string& name, const std::string& value,
>+  void changeHeader(const std::string& name,
>+                    const std::string& value,
>                     HeaderCombine combine);
>-  inline void addHeader(const std::string& name, const std::string& value,
>+  inline void addHeader(const std::string& name,
>+                        const std::string& value,
>                         bool append = true) {
>     changeHeader(name, value, append ? HC_AUTO : HC_NO);
>   }
>-  inline void setHeader(const std::string& name, const std::string& value,
>+  inline void setHeader(const std::string& name,
>+                        const std::string& value,
>                         bool overwrite = true) {
>     changeHeader(name, value, overwrite ? HC_REPLACE : HC_NEW);
>   }
>@@ -313,18 +268,10 @@ struct HttpData {
>   // keep in mind, this may not do what you want in the face of multiple headers
>   bool hasHeader(const std::string& name, std::string* value) const;
> 
>-  inline const_iterator begin() const {
>-    return headers_.begin();
>-  }
>-  inline const_iterator end() const {
>-    return headers_.end();
>-  }
>-  inline iterator begin() {
>-    return headers_.begin();
>-  }
>-  inline iterator end() {
>-    return headers_.end();
>-  }
>+  inline const_iterator begin() const { return headers_.begin(); }
>+  inline const_iterator end() const { return headers_.end(); }
>+  inline iterator begin() { return headers_.begin(); }
>+  inline iterator end() { return headers_.end(); }
>   inline const_iterator begin(const std::string& name) const {
>     return headers_.lower_bound(name);
>   }
>@@ -339,21 +286,22 @@ struct HttpData {
>   }
> 
>   // Convenience methods using HttpHeader
>-  inline void changeHeader(HttpHeader header, const std::string& value,
>+  inline void changeHeader(HttpHeader header,
>+                           const std::string& value,
>                            HeaderCombine combine) {
>     changeHeader(ToString(header), value, combine);
>   }
>-  inline void addHeader(HttpHeader header, const std::string& value,
>+  inline void addHeader(HttpHeader header,
>+                        const std::string& value,
>                         bool append = true) {
>     addHeader(ToString(header), value, append);
>   }
>-  inline void setHeader(HttpHeader header, const std::string& value,
>+  inline void setHeader(HttpHeader header,
>+                        const std::string& value,
>                         bool overwrite = true) {
>     setHeader(ToString(header), value, overwrite);
>   }
>-  inline void clearHeader(HttpHeader header) {
>-    clearHeader(ToString(header));
>-  }
>+  inline void clearHeader(HttpHeader header) { clearHeader(ToString(header)); }
>   inline bool hasHeader(HttpHeader header, std::string* value) const {
>     return hasHeader(ToString(header), value);
>   }
>@@ -376,23 +324,20 @@ struct HttpData {
>   virtual size_t formatLeader(char* buffer, size_t size) const = 0;
>   virtual HttpError parseLeader(const char* line, size_t len) = 0;
> 
>-protected:
>- virtual ~HttpData();
>+ protected:
>+  virtual ~HttpData();
>   void clear(bool release_document);
>-  void copy(const HttpData& src);
> 
>-private:
>+ private:
>   HeaderMap headers_;
> };
> 
> struct HttpRequestData : public HttpData {
>-  HttpVerb verb;
>   std::string path;
> 
>-  HttpRequestData() : verb(HV_GET) { }
>+  HttpRequestData() {}
> 
>   void clear(bool release_document);
>-  void copy(const HttpRequestData& src);
> 
>   size_t formatLeader(char* buffer, size_t size) const override;
>   HttpError parseLeader(const char* line, size_t len) override;
>@@ -405,17 +350,11 @@ struct HttpResponseData : public HttpData {
>   uint32_t scode;
>   std::string message;
> 
>-  HttpResponseData() : scode(HC_INTERNAL_SERVER_ERROR) { }
>+  HttpResponseData() : scode(HC_INTERNAL_SERVER_ERROR) {}
>   void clear(bool release_document);
>-  void copy(const HttpResponseData& src);
> 
>   // Convenience methods
>   void set_success(uint32_t scode = HC_OK);
>-  void set_success(const std::string& content_type,
>-                   StreamInterface* document,
>-                   uint32_t scode = HC_OK);
>-  void set_redirect(const std::string& location,
>-                    uint32_t scode = HC_MOVED_TEMPORARILY);
>   void set_error(uint32_t scode);
> 
>   size_t formatLeader(char* buffer, size_t size) const override;
>@@ -433,8 +372,8 @@ struct HttpTransaction {
> 
> struct HttpAuthContext {
>   std::string auth_method;
>-  HttpAuthContext(const std::string& auth) : auth_method(auth) { }
>-  virtual ~HttpAuthContext() { }
>+  HttpAuthContext(const std::string& auth) : auth_method(auth) {}
>+  virtual ~HttpAuthContext() {}
> };
> 
> enum HttpAuthResult { HAR_RESPONSE, HAR_IGNORE, HAR_CREDENTIALS, HAR_ERROR };
>@@ -442,15 +381,20 @@ enum HttpAuthResult { HAR_RESPONSE, HAR_IGNORE, HAR_CREDENTIALS, HAR_ERROR };
> // 'context' is used by this function to record information between calls.
> // Start by passing a null pointer, then pass the same pointer each additional
> // call.  When the authentication attempt is finished, delete the context.
>-HttpAuthResult HttpAuthenticate(
>-  const char * challenge, size_t len,
>-  const SocketAddress& server,
>-  const std::string& method, const std::string& uri,
>-  const std::string& username, const CryptString& password,
>-  HttpAuthContext *& context, std::string& response, std::string& auth_method);
>+// TODO(bugs.webrtc.org/8905): Change "response" to "ZeroOnFreeBuffer".
>+HttpAuthResult HttpAuthenticate(const char* challenge,
>+                                size_t len,
>+                                const SocketAddress& server,
>+                                const std::string& method,
>+                                const std::string& uri,
>+                                const std::string& username,
>+                                const CryptString& password,
>+                                HttpAuthContext*& context,
>+                                std::string& response,
>+                                std::string& auth_method);
> 
> //////////////////////////////////////////////////////////////////////
> 
>-} // namespace rtc
>+}  // namespace rtc
> 
>-#endif // RTC_BASE_HTTPCOMMON_H_
>+#endif  // RTC_BASE_HTTPCOMMON_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/httpcommon_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/httpcommon_unittest.cc
>index 997d6e31e1f..ed7f242d3fc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/httpcommon_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/httpcommon_unittest.cc
>@@ -122,8 +122,8 @@ TEST(Url, SkipsUser) {
> TEST(HttpResponseData, parseLeaderHttp1_0) {
>   static const char kResponseString[] = "HTTP/1.0 200 OK";
>   HttpResponseData response;
>-  EXPECT_EQ(HE_NONE, response.parseLeader(kResponseString,
>-                                          sizeof(kResponseString) - 1));
>+  EXPECT_EQ(HE_NONE,
>+            response.parseLeader(kResponseString, sizeof(kResponseString) - 1));
>   EXPECT_EQ(HVER_1_0, response.version);
>   EXPECT_EQ(200U, response.scode);
> }
>@@ -131,8 +131,8 @@ TEST(HttpResponseData, parseLeaderHttp1_0) {
> TEST(HttpResponseData, parseLeaderHttp1_1) {
>   static const char kResponseString[] = "HTTP/1.1 200 OK";
>   HttpResponseData response;
>-  EXPECT_EQ(HE_NONE, response.parseLeader(kResponseString,
>-                                          sizeof(kResponseString) - 1));
>+  EXPECT_EQ(HE_NONE,
>+            response.parseLeader(kResponseString, sizeof(kResponseString) - 1));
>   EXPECT_EQ(HVER_1_1, response.version);
>   EXPECT_EQ(200U, response.scode);
> }
>@@ -140,8 +140,8 @@ TEST(HttpResponseData, parseLeaderHttp1_1) {
> TEST(HttpResponseData, parseLeaderHttpUnknown) {
>   static const char kResponseString[] = "HTTP 200 OK";
>   HttpResponseData response;
>-  EXPECT_EQ(HE_NONE, response.parseLeader(kResponseString,
>-                                          sizeof(kResponseString) - 1));
>+  EXPECT_EQ(HE_NONE,
>+            response.parseLeader(kResponseString, sizeof(kResponseString) - 1));
>   EXPECT_EQ(HVER_UNKNOWN, response.version);
>   EXPECT_EQ(200U, response.scode);
> }
>@@ -149,8 +149,8 @@ TEST(HttpResponseData, parseLeaderHttpUnknown) {
> TEST(HttpResponseData, parseLeaderHttpFailure) {
>   static const char kResponseString[] = "HTTP/1.1 503 Service Unavailable";
>   HttpResponseData response;
>-  EXPECT_EQ(HE_NONE, response.parseLeader(kResponseString,
>-                                          sizeof(kResponseString) - 1));
>+  EXPECT_EQ(HE_NONE,
>+            response.parseLeader(kResponseString, sizeof(kResponseString) - 1));
>   EXPECT_EQ(HVER_1_1, response.version);
>   EXPECT_EQ(503U, response.scode);
> }
>@@ -158,8 +158,8 @@ TEST(HttpResponseData, parseLeaderHttpFailure) {
> TEST(HttpResponseData, parseLeaderHttpInvalid) {
>   static const char kResponseString[] = "Durrrrr, what's HTTP?";
>   HttpResponseData response;
>-  EXPECT_EQ(HE_PROTOCOL, response.parseLeader(kResponseString,
>-                                              sizeof(kResponseString) - 1));
>+  EXPECT_EQ(HE_PROTOCOL,
>+            response.parseLeader(kResponseString, sizeof(kResponseString) - 1));
> }
> 
>-} // namespace rtc
>+}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/httpserver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/httpserver.cc
>index c36b4327a6a..0e536c3bc96 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/httpserver.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/httpserver.cc
>@@ -25,24 +25,21 @@ namespace rtc {
> // HttpServer
> ///////////////////////////////////////////////////////////////////////////////
> 
>-HttpServer::HttpServer() : next_connection_id_(1), closing_(false) {
>-}
>+HttpServer::HttpServer() : next_connection_id_(1), closing_(false) {}
> 
> HttpServer::~HttpServer() {
>   if (closing_) {
>     RTC_LOG(LS_WARNING) << "HttpServer::CloseAll has not completed";
>   }
>   for (ConnectionMap::iterator it = connections_.begin();
>-       it != connections_.end();
>-       ++it) {
>+       it != connections_.end(); ++it) {
>     StreamInterface* stream = it->second->EndProcess();
>     delete stream;
>     delete it->second;
>   }
> }
> 
>-int
>-HttpServer::HandleConnection(StreamInterface* stream) {
>+int HttpServer::HandleConnection(StreamInterface* stream) {
>   int connection_id = next_connection_id_++;
>   RTC_DCHECK(connection_id != HTTP_INVALID_CONNECTION_ID);
>   Connection* connection = new Connection(connection_id, this);
>@@ -51,8 +48,7 @@ HttpServer::HandleConnection(StreamInterface* stream) {
>   return connection_id;
> }
> 
>-void
>-HttpServer::Respond(HttpServerTransaction* transaction) {
>+void HttpServer::Respond(HttpServerTransaction* transaction) {
>   int connection_id = transaction->connection_id();
>   if (Connection* connection = Find(connection_id)) {
>     connection->Respond(transaction);
>@@ -63,15 +59,13 @@ HttpServer::Respond(HttpServerTransaction* transaction) {
>   }
> }
> 
>-void
>-HttpServer::Close(int connection_id, bool force) {
>+void HttpServer::Close(int connection_id, bool force) {
>   if (Connection* connection = Find(connection_id)) {
>     connection->InitiateClose(force);
>   }
> }
> 
>-void
>-HttpServer::CloseAll(bool force) {
>+void HttpServer::CloseAll(bool force) {
>   if (connections_.empty()) {
>     SignalCloseAllComplete(this);
>     return;
>@@ -83,21 +77,19 @@ HttpServer::CloseAll(bool force) {
>     connections.push_back(it->second);
>   }
>   for (std::list<Connection*>::const_iterator it = connections.begin();
>-      it != connections.end(); ++it) {
>+       it != connections.end(); ++it) {
>     (*it)->InitiateClose(force);
>   }
> }
> 
>-HttpServer::Connection*
>-HttpServer::Find(int connection_id) {
>+HttpServer::Connection* HttpServer::Find(int connection_id) {
>   ConnectionMap::iterator it = connections_.find(connection_id);
>   if (it == connections_.end())
>     return nullptr;
>   return it->second;
> }
> 
>-void
>-HttpServer::Remove(int connection_id) {
>+void HttpServer::Remove(int connection_id) {
>   ConnectionMap::iterator it = connections_.find(connection_id);
>   if (it == connections_.end()) {
>     RTC_NOTREACHED();
>@@ -130,8 +122,7 @@ HttpServer::Connection::~Connection() {
>   Thread::Current()->Dispose(current_);
> }
> 
>-void
>-HttpServer::Connection::BeginProcess(StreamInterface* stream) {
>+void HttpServer::Connection::BeginProcess(StreamInterface* stream) {
>   base_.notify(this);
>   base_.attach(stream);
>   current_ = new HttpServerTransaction(connection_id_);
>@@ -139,15 +130,13 @@ HttpServer::Connection::BeginProcess(StreamInterface* stream) {
>     base_.recv(&current_->request);
> }
> 
>-StreamInterface*
>-HttpServer::Connection::EndProcess() {
>+StreamInterface* HttpServer::Connection::EndProcess() {
>   base_.notify(nullptr);
>   base_.abort(HE_DISCONNECTED);
>   return base_.detach();
> }
> 
>-void
>-HttpServer::Connection::Respond(HttpServerTransaction* transaction) {
>+void HttpServer::Connection::Respond(HttpServerTransaction* transaction) {
>   RTC_DCHECK(current_ == nullptr);
>   current_ = transaction;
>   if (current_->response.begin() == current_->response.end()) {
>@@ -155,14 +144,12 @@ HttpServer::Connection::Respond(HttpServerTransaction* transaction) {
>   }
>   bool keep_alive = HttpShouldKeepAlive(current_->request);
>   current_->response.setHeader(HH_CONNECTION,
>-                               keep_alive ? "Keep-Alive" : "Close",
>-                               false);
>+                               keep_alive ? "Keep-Alive" : "Close", false);
>   close_ = !HttpShouldKeepAlive(current_->response);
>   base_.send(&current_->response);
> }
> 
>-void
>-HttpServer::Connection::InitiateClose(bool force) {
>+void HttpServer::Connection::InitiateClose(bool force) {
>   bool request_in_progress = (HM_SEND == base_.mode()) || (nullptr == current_);
>   if (!signalling_ && (force || !request_in_progress)) {
>     server_->Remove(connection_id_);
>@@ -175,8 +162,8 @@ HttpServer::Connection::InitiateClose(bool force) {
> // IHttpNotify Implementation
> //
> 
>-HttpError
>-HttpServer::Connection::onHttpHeaderComplete(bool chunked, size_t& data_size) {
>+HttpError HttpServer::Connection::onHttpHeaderComplete(bool chunked,
>+                                                       size_t& data_size) {
>   if (data_size == SIZE_UNKNOWN) {
>     data_size = 0;
>   }
>@@ -189,8 +176,7 @@ HttpServer::Connection::onHttpHeaderComplete(bool chunked, size_t& data_size) {
>   return HE_NONE;
> }
> 
>-void
>-HttpServer::Connection::onHttpComplete(HttpMode mode, HttpError err) {
>+void HttpServer::Connection::onHttpComplete(HttpMode mode, HttpError err) {
>   if (mode == HM_SEND) {
>     RTC_DCHECK(current_ != nullptr);
>     signalling_ = true;
>@@ -208,7 +194,7 @@ HttpServer::Connection::onHttpComplete(HttpMode mode, HttpError err) {
>   } else if (mode == HM_RECV) {
>     RTC_DCHECK(current_ != nullptr);
>     // TODO: do we need this?
>-    //request_.document_->rewind();
>+    // request_.document_->rewind();
>     HttpServerTransaction* transaction = current_;
>     current_ = nullptr;
>     server_->SignalHttpRequest(server_, transaction);
>@@ -222,8 +208,7 @@ HttpServer::Connection::onHttpComplete(HttpMode mode, HttpError err) {
>   }
> }
> 
>-void
>-HttpServer::Connection::onHttpClosed(HttpError err) {
>+void HttpServer::Connection::onHttpClosed(HttpError err) {
>   server_->Remove(connection_id_);
> }
> 
>@@ -235,13 +220,11 @@ HttpListenServer::HttpListenServer() {
>   SignalConnectionClosed.connect(this, &HttpListenServer::OnConnectionClosed);
> }
> 
>-HttpListenServer::~HttpListenServer() {
>-}
>+HttpListenServer::~HttpListenServer() {}
> 
> int HttpListenServer::Listen(const SocketAddress& address) {
>-  AsyncSocket* sock =
>-      Thread::Current()->socketserver()->CreateAsyncSocket(address.family(),
>-                                                           SOCK_STREAM);
>+  AsyncSocket* sock = Thread::Current()->socketserver()->CreateAsyncSocket(
>+      address.family(), SOCK_STREAM);
>   if (!sock) {
>     return SOCKET_ERROR;
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/httpserver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/httpserver.h
>index 61442a93f34..e4d94445ede 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/httpserver.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/httpserver.h
>@@ -29,16 +29,16 @@ class SocketAddress;
> const int HTTP_INVALID_CONNECTION_ID = 0;
> 
> struct HttpServerTransaction : public HttpTransaction {
>-public:
>-  HttpServerTransaction(int id) : connection_id_(id) { }
>+ public:
>+  HttpServerTransaction(int id) : connection_id_(id) {}
>   int connection_id() const { return connection_id_; }
> 
>-private:
>+ private:
>   int connection_id_;
> };
> 
> class HttpServer {
>-public:
>+ public:
>   HttpServer();
>   virtual ~HttpServer();
> 
>@@ -54,7 +54,7 @@ public:
>   // the document can be set to null.  Note that the transaction object is still
>   // owened by the HttpServer at this point.
>   sigslot::signal3<HttpServer*, HttpServerTransaction*, bool*>
>-    SignalHttpRequestHeader;
>+      SignalHttpRequestHeader;
> 
>   // An HTTP request has been made, and is available in the transaction object.
>   // Populate the transaction's response, and then return the object via the
>@@ -66,7 +66,7 @@ public:
> 
>   // If you want to know when a request completes, listen to this event.
>   sigslot::signal3<HttpServer*, HttpServerTransaction*, int>
>-    SignalHttpRequestComplete;
>+      SignalHttpRequestComplete;
> 
>   // Stop processing the connection indicated by connection_id.
>   // Unless force is true, the server will complete sending a response that is
>@@ -78,9 +78,9 @@ public:
>   // outstanding connections have closed.
>   sigslot::signal1<HttpServer*> SignalCloseAllComplete;
> 
>-private:
>+ private:
>   class Connection : private IHttpNotify {
>-  public:
>+   public:
>     Connection(int connection_id, HttpServer* server);
>     ~Connection() override;
> 
>@@ -106,7 +106,7 @@ private:
>   void Remove(int connection_id);
> 
>   friend class Connection;
>-  typedef std::map<int,Connection*> ConnectionMap;
>+  typedef std::map<int, Connection*> ConnectionMap;
> 
>   ConnectionMap connections_;
>   int next_connection_id_;
>@@ -116,7 +116,7 @@ private:
> //////////////////////////////////////////////////////////////////////
> 
> class HttpListenServer : public HttpServer, public sigslot::has_slots<> {
>-public:
>+ public:
>   HttpListenServer();
>   ~HttpListenServer() override;
> 
>@@ -124,9 +124,10 @@ public:
>   bool GetAddress(SocketAddress* address) const;
>   void StopListening();
> 
>-private:
>+ private:
>   void OnReadEvent(AsyncSocket* socket);
>-  void OnConnectionClosed(HttpServer* server, int connection_id,
>+  void OnConnectionClosed(HttpServer* server,
>+                          int connection_id,
>                           StreamInterface* stream);
> 
>   std::unique_ptr<AsyncSocket> listener_;
>@@ -136,4 +137,4 @@ private:
> 
> }  // namespace rtc
> 
>-#endif // RTC_BASE_HTTPSERVER_H_
>+#endif  // RTC_BASE_HTTPSERVER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/httpserver_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/httpserver_unittest.cc
>index 5e86c885dfa..0dde6c5bde5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/httpserver_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/httpserver_unittest.cc
>@@ -17,59 +17,56 @@ using namespace webrtc::testing;
> namespace rtc {
> 
> namespace {
>-  const char* const kRequest =
>+const char* const kRequest =
>     "GET /index.html HTTP/1.1\r\n"
>     "Host: localhost\r\n"
>     "\r\n";
> 
>-  struct HttpServerMonitor : public sigslot::has_slots<> {
>-    HttpServerTransaction* transaction;
>-    bool server_closed, connection_closed;
>+struct HttpServerMonitor : public sigslot::has_slots<> {
>+  HttpServerTransaction* transaction;
>+  bool server_closed, connection_closed;
> 
>-    HttpServerMonitor(HttpServer* server)
>-        : transaction(nullptr), server_closed(false), connection_closed(false) {
>-      server->SignalCloseAllComplete.connect(this,
>-        &HttpServerMonitor::OnClosed);
>-      server->SignalHttpRequest.connect(this, &HttpServerMonitor::OnRequest);
>-      server->SignalHttpRequestComplete.connect(this,
>-        &HttpServerMonitor::OnRequestComplete);
>-      server->SignalConnectionClosed.connect(this,
>-        &HttpServerMonitor::OnConnectionClosed);
>-    }
>-    void OnRequest(HttpServer*, HttpServerTransaction* t) {
>-      ASSERT_FALSE(transaction);
>-      transaction = t;
>-      transaction->response.set_success();
>-      transaction->response.setHeader(HH_CONNECTION, "Close");
>-    }
>-    void OnRequestComplete(HttpServer*, HttpServerTransaction* t, int) {
>-      ASSERT_EQ(transaction, t);
>-      transaction = nullptr;
>-    }
>-    void OnClosed(HttpServer*) {
>-      server_closed = true;
>-    }
>-    void OnConnectionClosed(HttpServer*, int, StreamInterface* stream) {
>-      connection_closed = true;
>-      delete stream;
>-    }
>-  };
>+  HttpServerMonitor(HttpServer* server)
>+      : transaction(nullptr), server_closed(false), connection_closed(false) {
>+    server->SignalCloseAllComplete.connect(this, &HttpServerMonitor::OnClosed);
>+    server->SignalHttpRequest.connect(this, &HttpServerMonitor::OnRequest);
>+    server->SignalHttpRequestComplete.connect(
>+        this, &HttpServerMonitor::OnRequestComplete);
>+    server->SignalConnectionClosed.connect(
>+        this, &HttpServerMonitor::OnConnectionClosed);
>+  }
>+  void OnRequest(HttpServer*, HttpServerTransaction* t) {
>+    ASSERT_FALSE(transaction);
>+    transaction = t;
>+    transaction->response.set_success();
>+    transaction->response.setHeader(HH_CONNECTION, "Close");
>+  }
>+  void OnRequestComplete(HttpServer*, HttpServerTransaction* t, int) {
>+    ASSERT_EQ(transaction, t);
>+    transaction = nullptr;
>+  }
>+  void OnClosed(HttpServer*) { server_closed = true; }
>+  void OnConnectionClosed(HttpServer*, int, StreamInterface* stream) {
>+    connection_closed = true;
>+    delete stream;
>+  }
>+};
> 
>-  void CreateClientConnection(HttpServer& server,
>-                              HttpServerMonitor& monitor,
>-                              bool send_request) {
>-    StreamSource* client = new StreamSource;
>-    client->SetState(SS_OPEN);
>-    server.HandleConnection(client);
>-    EXPECT_FALSE(monitor.server_closed);
>-    EXPECT_FALSE(monitor.transaction);
>+void CreateClientConnection(HttpServer& server,
>+                            HttpServerMonitor& monitor,
>+                            bool send_request) {
>+  StreamSource* client = new StreamSource;
>+  client->SetState(SS_OPEN);
>+  server.HandleConnection(client);
>+  EXPECT_FALSE(monitor.server_closed);
>+  EXPECT_FALSE(monitor.transaction);
> 
>-    if (send_request) {
>-      // Simulate a request
>-      client->QueueString(kRequest);
>-      EXPECT_FALSE(monitor.server_closed);
>-    }
>+  if (send_request) {
>+    // Simulate a request
>+    client->QueueString(kRequest);
>+    EXPECT_FALSE(monitor.server_closed);
>   }
>+}
> }  // anonymous namespace
> 
> TEST(HttpServer, DoesNotSignalCloseUnlessCloseAllIsCalled) {
>@@ -127,4 +124,4 @@ TEST(HttpServer, SignalsCloseAfterForcedCloseAll) {
>   EXPECT_TRUE(monitor.connection_closed);
> }
> 
>-} // namespace rtc
>+}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ifaddrs-android.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ifaddrs-android.cc
>index 85a4497fdd9..cc6cdab12c4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ifaddrs-android.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ifaddrs-android.cc
>@@ -64,7 +64,9 @@ int set_flags(struct ifaddrs* ifaddr) {
>   return 0;
> }
> 
>-int set_addresses(struct ifaddrs* ifaddr, ifaddrmsg* msg, void* data,
>+int set_addresses(struct ifaddrs* ifaddr,
>+                  ifaddrmsg* msg,
>+                  void* data,
>                   size_t len) {
>   if (msg->ifa_family == AF_INET) {
>     sockaddr_in* sa = new sockaddr_in;
>@@ -115,7 +117,9 @@ int make_prefixes(struct ifaddrs* ifaddr, int family, int prefixlen) {
>   return 0;
> }
> 
>-int populate_ifaddrs(struct ifaddrs* ifaddr, ifaddrmsg* msg, void* bytes,
>+int populate_ifaddrs(struct ifaddrs* ifaddr,
>+                     ifaddrmsg* msg,
>+                     void* bytes,
>                      size_t len) {
>   if (set_ifname(ifaddr, msg->ifa_index) != 0) {
>     return -1;
>@@ -156,8 +160,8 @@ int getifaddrs(struct ifaddrs** result) {
>   while (amount_read > 0) {
>     nlmsghdr* header = reinterpret_cast<nlmsghdr*>(&buf[0]);
>     size_t header_size = static_cast<size_t>(amount_read);
>-    for ( ; NLMSG_OK(header, header_size);
>-          header = NLMSG_NEXT(header, header_size)) {
>+    for (; NLMSG_OK(header, header_size);
>+         header = NLMSG_NEXT(header, header_size)) {
>       switch (header->nlmsg_type) {
>         case NLMSG_DONE:
>           // Success. Return.
>@@ -174,24 +178,24 @@ int getifaddrs(struct ifaddrs** result) {
>           rtattr* rta = IFA_RTA(address_msg);
>           ssize_t payload_len = IFA_PAYLOAD(header);
>           while (RTA_OK(rta, payload_len)) {
>-            if (rta->rta_type == IFA_ADDRESS) {
>-              int family = address_msg->ifa_family;
>-              if (family == AF_INET || family == AF_INET6) {
>-                ifaddrs* newest = new ifaddrs;
>-                memset(newest, 0, sizeof(ifaddrs));
>-                if (current) {
>-                  current->ifa_next = newest;
>-                } else {
>-                  start = newest;
>-                }
>-                if (populate_ifaddrs(newest, address_msg, RTA_DATA(rta),
>-                                     RTA_PAYLOAD(rta)) != 0) {
>-                  freeifaddrs(start);
>-                  *result = nullptr;
>-                  return -1;
>-                }
>-                current = newest;
>+            if ((address_msg->ifa_family == AF_INET &&
>+                 rta->rta_type == IFA_LOCAL) ||
>+                (address_msg->ifa_family == AF_INET6 &&
>+                 rta->rta_type == IFA_ADDRESS)) {
>+              ifaddrs* newest = new ifaddrs;
>+              memset(newest, 0, sizeof(ifaddrs));
>+              if (current) {
>+                current->ifa_next = newest;
>+              } else {
>+                start = newest;
>               }
>+              if (populate_ifaddrs(newest, address_msg, RTA_DATA(rta),
>+                                   RTA_PAYLOAD(rta)) != 0) {
>+                freeifaddrs(start);
>+                *result = nullptr;
>+                return -1;
>+              }
>+              current = newest;
>             }
>             rta = RTA_NEXT(rta, payload_len);
>           }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ifaddrs-android.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ifaddrs-android.h
>index 82b4cb3006c..1a5b662f069 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ifaddrs-android.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ifaddrs-android.h
>@@ -14,7 +14,6 @@
> #include <stdio.h>
> #include <sys/socket.h>
> 
>-
> // Implementation of getifaddrs for Android.
> // Fills out a list of ifaddr structs (see below) which contain information
> // about every network interface available on the host.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ifaddrs_converter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ifaddrs_converter.cc
>index 2db99efd6c9..586b4e9edf9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ifaddrs_converter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ifaddrs_converter.cc
>@@ -22,8 +22,8 @@ bool IfAddrsConverter::ConvertIfAddrsToIPAddress(
>     IPAddress* mask) {
>   switch (interface->ifa_addr->sa_family) {
>     case AF_INET: {
>-      *ip = IPAddress(
>-          reinterpret_cast<sockaddr_in*>(interface->ifa_addr)->sin_addr);
>+      *ip = InterfaceAddress(IPAddress(
>+          reinterpret_cast<sockaddr_in*>(interface->ifa_addr)->sin_addr));
>       *mask = IPAddress(
>           reinterpret_cast<sockaddr_in*>(interface->ifa_netmask)->sin_addr);
>       return true;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ipaddress.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ipaddress.cc
>index ac0de91e041..c52c9a4addd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ipaddress.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ipaddress.cc
>@@ -9,9 +9,9 @@
>  */
> 
> #if defined(WEBRTC_POSIX)
>-#include <sys/types.h>
>-#include <sys/socket.h>
> #include <netinet/in.h>
>+#include <sys/socket.h>
>+#include <sys/types.h>
> #ifdef OPENBSD
> #include <netinet/in_systm.h>
> #endif
>@@ -31,19 +31,25 @@
> #include "rtc_base/logging.h"
> #include "rtc_base/nethelpers.h"
> #include "rtc_base/stringutils.h"
>+
>+#if defined(WEBRTC_WIN)
> #include "rtc_base/win32.h"
>+#endif  // WEBRTC_WIN
> 
> namespace rtc {
> 
> // Prefixes used for categorizing IPv6 addresses.
>-static const in6_addr kV4MappedPrefix = {{{0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
>-                                           0xFF, 0xFF, 0}}};
>+static const in6_addr kV4MappedPrefix = {
>+    {{0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0xFF, 0xFF, 0}}};
> static const in6_addr k6To4Prefix = {{{0x20, 0x02, 0}}};
> static const in6_addr kTeredoPrefix = {{{0x20, 0x01, 0x00, 0x00}}};
> static const in6_addr kV4CompatibilityPrefix = {{{0}}};
> static const in6_addr k6BonePrefix = {{{0x3f, 0xfe, 0}}};
>+static const in6_addr kPrivateNetworkPrefix = {{{0xFD}}};
> 
>-static bool IsPrivateV4(uint32_t ip);
>+static bool IPIsHelper(const IPAddress& ip,
>+                       const in6_addr& tomatch,
>+                       int length);
> static in_addr ExtractMappedAddress(const in6_addr& addr);
> 
> uint32_t IPAddress::v4AddressAsHostOrderInteger() const {
>@@ -68,8 +74,7 @@ size_t IPAddress::Size() const {
>   return 0;
> }
> 
>-
>-bool IPAddress::operator==(const IPAddress &other) const {
>+bool IPAddress::operator==(const IPAddress& other) const {
>   if (family_ != other.family_) {
>     return false;
>   }
>@@ -82,15 +87,15 @@ bool IPAddress::operator==(const IPAddress &other) const {
>   return family_ == AF_UNSPEC;
> }
> 
>-bool IPAddress::operator!=(const IPAddress &other) const {
>+bool IPAddress::operator!=(const IPAddress& other) const {
>   return !((*this) == other);
> }
> 
>-bool IPAddress::operator >(const IPAddress &other) const {
>+bool IPAddress::operator>(const IPAddress& other) const {
>   return (*this) != other && !((*this) < other);
> }
> 
>-bool IPAddress::operator <(const IPAddress &other) const {
>+bool IPAddress::operator<(const IPAddress& other) const {
>   // IPv4 is 'less than' IPv6
>   if (family_ != other.family_) {
>     if (family_ == AF_UNSPEC) {
>@@ -105,7 +110,7 @@ bool IPAddress::operator <(const IPAddress &other) const {
>   switch (family_) {
>     case AF_INET: {
>       return NetworkToHost32(u_.ip4.s_addr) <
>-          NetworkToHost32(other.u_.ip4.s_addr);
>+             NetworkToHost32(other.u_.ip4.s_addr);
>     }
>     case AF_INET6: {
>       return memcmp(&u_.ip6.s6_addr, &other.u_.ip6.s6_addr, 16) < 0;
>@@ -115,11 +120,6 @@ bool IPAddress::operator <(const IPAddress &other) const {
>   return false;
> }
> 
>-std::ostream& operator<<(std::ostream& os, const IPAddress& ip) {
>-  os << ip.ToString();
>-  return os;
>-}
>-
> in6_addr IPAddress::ipv6_address() const {
>   return u_.ip6;
> }
>@@ -195,37 +195,52 @@ IPAddress IPAddress::AsIPv6Address() const {
>   return IPAddress(v6addr);
> }
> 
>-bool InterfaceAddress::operator==(const InterfaceAddress &other) const {
>+bool InterfaceAddress::operator==(const InterfaceAddress& other) const {
>   return ipv6_flags_ == other.ipv6_flags() &&
>-    static_cast<const IPAddress&>(*this) == other;
>+         static_cast<const IPAddress&>(*this) == other;
> }
> 
>-bool InterfaceAddress::operator!=(const InterfaceAddress &other) const {
>+bool InterfaceAddress::operator!=(const InterfaceAddress& other) const {
>   return !((*this) == other);
> }
> 
> const InterfaceAddress& InterfaceAddress::operator=(
>-  const InterfaceAddress& other) {
>+    const InterfaceAddress& other) {
>   ipv6_flags_ = other.ipv6_flags_;
>   static_cast<IPAddress&>(*this) = other;
>   return *this;
> }
> 
>-std::ostream& operator<<(std::ostream& os, const InterfaceAddress& ip) {
>-  os << static_cast<const IPAddress&>(ip);
>+std::string InterfaceAddress::ToString() const {
>+  std::string result = IPAddress::ToString();
>+
>+  if (family() == AF_INET6)
>+    result += "|flags:0x" + rtc::ToHex(ipv6_flags());
>+
>+  return result;
>+}
> 
>-  if (ip.family() == AF_INET6)
>-    os << "|flags:0x" << std::hex << ip.ipv6_flags();
>+static bool IPIsPrivateNetworkV4(const IPAddress& ip) {
>+  uint32_t ip_in_host_order = ip.v4AddressAsHostOrderInteger();
>+  return ((ip_in_host_order >> 24) == 10) ||
>+         ((ip_in_host_order >> 20) == ((172 << 4) | 1)) ||
>+         ((ip_in_host_order >> 16) == ((192 << 8) | 168));
>+}
> 
>-  return os;
>+static bool IPIsPrivateNetworkV6(const IPAddress& ip) {
>+  return IPIsHelper(ip, kPrivateNetworkPrefix, 8);
> }
> 
>-bool IsPrivateV4(uint32_t ip_in_host_order) {
>-  return ((ip_in_host_order >> 24) == 127) ||
>-      ((ip_in_host_order >> 24) == 10) ||
>-      ((ip_in_host_order >> 20) == ((172 << 4) | 1)) ||
>-      ((ip_in_host_order >> 16) == ((192 << 8) | 168)) ||
>-      ((ip_in_host_order >> 16) == ((169 << 8) | 254));
>+bool IPIsPrivateNetwork(const IPAddress& ip) {
>+  switch (ip.family()) {
>+    case AF_INET: {
>+      return IPIsPrivateNetworkV4(ip);
>+    }
>+    case AF_INET6: {
>+      return IPIsPrivateNetworkV6(ip);
>+    }
>+  }
>+  return false;
> }
> 
> in_addr ExtractMappedAddress(const in6_addr& in6) {
>@@ -268,8 +283,7 @@ bool IPFromString(const std::string& str, IPAddress* out) {
>   return true;
> }
> 
>-bool IPFromString(const std::string& str, int flags,
>-                  InterfaceAddress* out) {
>+bool IPFromString(const std::string& str, int flags, InterfaceAddress* out) {
>   IPAddress ip;
>   if (!IPFromString(str, &ip)) {
>     return false;
>@@ -291,28 +305,29 @@ bool IPIsAny(const IPAddress& ip) {
>   return false;
> }
> 
>+static bool IPIsLoopbackV4(const IPAddress& ip) {
>+  uint32_t ip_in_host_order = ip.v4AddressAsHostOrderInteger();
>+  return ((ip_in_host_order >> 24) == 127);
>+}
>+
>+static bool IPIsLoopbackV6(const IPAddress& ip) {
>+  return ip == IPAddress(in6addr_loopback);
>+}
>+
> bool IPIsLoopback(const IPAddress& ip) {
>   switch (ip.family()) {
>     case AF_INET: {
>-      return (ip.v4AddressAsHostOrderInteger() >> 24) == 127;
>+      return IPIsLoopbackV4(ip);
>     }
>     case AF_INET6: {
>-      return ip == IPAddress(in6addr_loopback);
>+      return IPIsLoopbackV6(ip);
>     }
>   }
>   return false;
> }
> 
> bool IPIsPrivate(const IPAddress& ip) {
>-  switch (ip.family()) {
>-    case AF_INET: {
>-      return IsPrivateV4(ip.v4AddressAsHostOrderInteger());
>-    }
>-    case AF_INET6: {
>-      return IPIsLinkLocal(ip) || IPIsLoopback(ip);
>-    }
>-  }
>-  return false;
>+  return IPIsLinkLocal(ip) || IPIsLoopback(ip) || IPIsPrivateNetwork(ip);
> }
> 
> bool IPIsUnspec(const IPAddress& ip) {
>@@ -400,9 +415,7 @@ int CountIPMaskBits(IPAddress mask) {
>       bits = (i * 32);
>       break;
>     }
>-    default: {
>-      return 0;
>-    }
>+    default: { return 0; }
>   }
>   if (word_to_count == 0) {
>     return bits;
>@@ -415,12 +428,18 @@ int CountIPMaskBits(IPAddress mask) {
>   // This could also be written word_to_count &= -word_to_count, but
>   // MSVC emits warning C4146 when negating an unsigned number.
>   word_to_count &= ~word_to_count + 1;  // Isolate lowest set bit.
>-  if (word_to_count) zeroes--;
>-  if (word_to_count & 0x0000FFFF) zeroes -= 16;
>-  if (word_to_count & 0x00FF00FF) zeroes -= 8;
>-  if (word_to_count & 0x0F0F0F0F) zeroes -= 4;
>-  if (word_to_count & 0x33333333) zeroes -= 2;
>-  if (word_to_count & 0x55555555) zeroes -= 1;
>+  if (word_to_count)
>+    zeroes--;
>+  if (word_to_count & 0x0000FFFF)
>+    zeroes -= 16;
>+  if (word_to_count & 0x00FF00FF)
>+    zeroes -= 8;
>+  if (word_to_count & 0x0F0F0F0F)
>+    zeroes -= 4;
>+  if (word_to_count & 0x33333333)
>+    zeroes -= 2;
>+  if (word_to_count & 0x55555555)
>+    zeroes -= 1;
> 
>   return bits + (32 - zeroes);
> }
>@@ -440,10 +459,27 @@ bool IPIs6To4(const IPAddress& ip) {
>   return IPIsHelper(ip, k6To4Prefix, 16);
> }
> 
>-bool IPIsLinkLocal(const IPAddress& ip) {
>+static bool IPIsLinkLocalV4(const IPAddress& ip) {
>+  uint32_t ip_in_host_order = ip.v4AddressAsHostOrderInteger();
>+  return ((ip_in_host_order >> 16) == ((169 << 8) | 254));
>+}
>+
>+static bool IPIsLinkLocalV6(const IPAddress& ip) {
>   // Can't use the helper because the prefix is 10 bits.
>   in6_addr addr = ip.ipv6_address();
>-  return addr.s6_addr[0] == 0xFE && addr.s6_addr[1] == 0x80;
>+  return (addr.s6_addr[0] == 0xFE) && ((addr.s6_addr[1] & 0xC0) == 0x80);
>+}
>+
>+bool IPIsLinkLocal(const IPAddress& ip) {
>+  switch (ip.family()) {
>+    case AF_INET: {
>+      return IPIsLinkLocalV4(ip);
>+    }
>+    case AF_INET6: {
>+      return IPIsLinkLocalV6(ip);
>+    }
>+  }
>+  return false;
> }
> 
> // According to http://www.ietf.org/rfc/rfc2373.txt, Appendix A, page 19.  An
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ipaddress.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ipaddress.h
>index f681d79dd6d..614fd8f575a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ipaddress.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ipaddress.h
>@@ -12,10 +12,10 @@
> #define RTC_BASE_IPADDRESS_H_
> 
> #if defined(WEBRTC_POSIX)
>-#include <netinet/in.h>
>-#include <sys/socket.h>
> #include <arpa/inet.h>
> #include <netdb.h>
>+#include <netinet/in.h>
>+#include <sys/socket.h>
> #endif
> #if defined(WEBRTC_WIN)
> #include <winsock2.h>
>@@ -25,7 +25,6 @@
> #include <string>
> #include <vector>
> 
>-#include "rtc_base/basictypes.h"
> #include "rtc_base/byteorder.h"
> #if defined(WEBRTC_WIN)
> #include "rtc_base/win32.h"
>@@ -34,33 +33,29 @@
> namespace rtc {
> 
> enum IPv6AddressFlag {
>-  IPV6_ADDRESS_FLAG_NONE =           0x00,
>+  IPV6_ADDRESS_FLAG_NONE = 0x00,
> 
>   // Temporary address is dynamic by nature and will not carry MAC
>   // address.
>-  IPV6_ADDRESS_FLAG_TEMPORARY =      1 << 0,
>+  IPV6_ADDRESS_FLAG_TEMPORARY = 1 << 0,
> 
>   // Temporary address could become deprecated once the preferred
>   // lifetime is reached. It is still valid but just shouldn't be used
>   // to create new connection.
>-  IPV6_ADDRESS_FLAG_DEPRECATED =     1 << 1,
>+  IPV6_ADDRESS_FLAG_DEPRECATED = 1 << 1,
> };
> 
> // Version-agnostic IP address class, wraps a union of in_addr and in6_addr.
> class IPAddress {
>  public:
>-  IPAddress() : family_(AF_UNSPEC) {
>-    ::memset(&u_, 0, sizeof(u_));
>-  }
>+  IPAddress() : family_(AF_UNSPEC) { ::memset(&u_, 0, sizeof(u_)); }
> 
>   explicit IPAddress(const in_addr& ip4) : family_(AF_INET) {
>     memset(&u_, 0, sizeof(u_));
>     u_.ip4 = ip4;
>   }
> 
>-  explicit IPAddress(const in6_addr& ip6) : family_(AF_INET6) {
>-    u_.ip6 = ip6;
>-  }
>+  explicit IPAddress(const in6_addr& ip6) : family_(AF_INET6) { u_.ip6 = ip6; }
> 
>   explicit IPAddress(uint32_t ip_in_host_byte_order) : family_(AF_INET) {
>     memset(&u_, 0, sizeof(u_));
>@@ -73,7 +68,7 @@ class IPAddress {
> 
>   virtual ~IPAddress() {}
> 
>-  const IPAddress & operator=(const IPAddress& other) {
>+  const IPAddress& operator=(const IPAddress& other) {
>     family_ = other.family_;
>     ::memcpy(&u_, &other.u_, sizeof(u_));
>     return *this;
>@@ -81,9 +76,15 @@ class IPAddress {
> 
>   bool operator==(const IPAddress& other) const;
>   bool operator!=(const IPAddress& other) const;
>-  bool operator <(const IPAddress& other) const;
>-  bool operator >(const IPAddress& other) const;
>-  friend std::ostream& operator<<(std::ostream& os, const IPAddress& addr);
>+  bool operator<(const IPAddress& other) const;
>+  bool operator>(const IPAddress& other) const;
>+
>+#ifdef UNIT_TEST
>+  inline std::ostream& operator<<(  // no-presubmit-check TODO(webrtc:8982)
>+      std::ostream& os) {           // no-presubmit-check TODO(webrtc:8982)
>+    return os << ToString();
>+  }
>+#endif  // UNIT_TEST
> 
>   int family() const { return family_; }
>   in_addr ipv4_address() const;
>@@ -126,23 +127,23 @@ class InterfaceAddress : public IPAddress {
>  public:
>   InterfaceAddress() : ipv6_flags_(IPV6_ADDRESS_FLAG_NONE) {}
> 
>-  InterfaceAddress(IPAddress ip)
>-    : IPAddress(ip), ipv6_flags_(IPV6_ADDRESS_FLAG_NONE) {}
>+  explicit InterfaceAddress(IPAddress ip)
>+      : IPAddress(ip), ipv6_flags_(IPV6_ADDRESS_FLAG_NONE) {}
> 
>   InterfaceAddress(IPAddress addr, int ipv6_flags)
>-    : IPAddress(addr), ipv6_flags_(ipv6_flags) {}
>+      : IPAddress(addr), ipv6_flags_(ipv6_flags) {}
> 
>   InterfaceAddress(const in6_addr& ip6, int ipv6_flags)
>-    : IPAddress(ip6), ipv6_flags_(ipv6_flags) {}
>+      : IPAddress(ip6), ipv6_flags_(ipv6_flags) {}
> 
>-  const InterfaceAddress & operator=(const InterfaceAddress& other);
>+  const InterfaceAddress& operator=(const InterfaceAddress& other);
> 
>   bool operator==(const InterfaceAddress& other) const;
>   bool operator!=(const InterfaceAddress& other) const;
> 
>   int ipv6_flags() const { return ipv6_flags_; }
>-  friend std::ostream& operator<<(std::ostream& os,
>-                                  const InterfaceAddress& addr);
>+
>+  std::string ToString() const;
> 
>  private:
>   int ipv6_flags_;
>@@ -150,10 +151,15 @@ class InterfaceAddress : public IPAddress {
> 
> bool IPFromAddrInfo(struct addrinfo* info, IPAddress* out);
> bool IPFromString(const std::string& str, IPAddress* out);
>-bool IPFromString(const std::string& str, int flags,
>-                  InterfaceAddress* out);
>+bool IPFromString(const std::string& str, int flags, InterfaceAddress* out);
> bool IPIsAny(const IPAddress& ip);
> bool IPIsLoopback(const IPAddress& ip);
>+bool IPIsLinkLocal(const IPAddress& ip);
>+// Identify a private network address like "192.168.111.222"
>+// (see https://en.wikipedia.org/wiki/Private_network )
>+bool IPIsPrivateNetwork(const IPAddress& ip);
>+// Identify if an IP is "private", that is a loopback
>+// or an address belonging to a link-local or a private network.
> bool IPIsPrivate(const IPAddress& ip);
> bool IPIsUnspec(const IPAddress& ip);
> size_t HashIP(const IPAddress& ip);
>@@ -161,7 +167,6 @@ size_t HashIP(const IPAddress& ip);
> // These are only really applicable for IPv6 addresses.
> bool IPIs6Bone(const IPAddress& ip);
> bool IPIs6To4(const IPAddress& ip);
>-bool IPIsLinkLocal(const IPAddress& ip);
> bool IPIsMacBased(const IPAddress& ip);
> bool IPIsSiteLocal(const IPAddress& ip);
> bool IPIsTeredo(const IPAddress& ip);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ipaddress_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ipaddress_unittest.cc
>index 5d7f2a95cf9..30312d06773 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ipaddress_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ipaddress_unittest.cc
>@@ -17,30 +17,25 @@ static const unsigned int kIPv4AddrSize = 4;
> static const unsigned int kIPv6AddrSize = 16;
> static const unsigned int kIPv4RFC1918Addr = 0xC0A80701;
> static const unsigned int kIPv4PublicAddr = 0x01020304;
>-static const in6_addr kIPv6LinkLocalAddr = {{{0xfe, 0x80, 0x00, 0x00,
>-                                              0x00, 0x00, 0x00, 0x00,
>-                                              0xbe, 0x30, 0x5b, 0xff,
>-                                              0xfe, 0xe5, 0x00, 0xc3}}};
>-static const in6_addr kIPv6PublicAddr = {{{0x24, 0x01, 0xfa, 0x00,
>-                                           0x00, 0x04, 0x10, 0x00,
>-                                           0xbe, 0x30, 0x5b, 0xff,
>-                                           0xfe, 0xe5, 0x00, 0xc3}}};
>-static const in6_addr kIPv6PublicAddr2 = {{{0x24, 0x01, 0x00, 0x00,
>-                                            0x00, 0x00, 0x10, 0x00,
>-                                            0xbe, 0x30, 0x5b, 0xff,
>-                                            0xfe, 0xe5, 0x00, 0xc3}}};
>-static const in6_addr kIPv4MappedAnyAddr = {{{0x00, 0x00, 0x00, 0x00,
>-                                              0x00, 0x00, 0x00, 0x00,
>-                                              0x00, 0x00, 0xff, 0xff,
>-                                              0x00, 0x00, 0x00, 0x00}}};
>-static const in6_addr kIPv4MappedRFC1918Addr = {{{0x00, 0x00, 0x00, 0x00,
>-                                                  0x00, 0x00, 0x00, 0x00,
>-                                                  0x00, 0x00, 0xff, 0xff,
>-                                                  0xc0, 0xa8, 0x07, 0x01}}};
>-static const in6_addr kIPv4MappedPublicAddr = {{{0x00, 0x00, 0x00, 0x00,
>-                                                 0x00, 0x00, 0x00, 0x00,
>-                                                 0x00, 0x00, 0xff, 0xff,
>-                                                 0x01, 0x02, 0x03, 0x04}}};
>+static const unsigned int kIPv4LinkLocalAddr = 0xA9FE10C1;  // 169.254.16.193
>+static const in6_addr kIPv6LinkLocalAddr = {
>+    {{0xfe, 0x80, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0xbe, 0x30, 0x5b, 0xff,
>+      0xfe, 0xe5, 0x00, 0xc3}}};
>+static const in6_addr kIPv6PublicAddr = {
>+    {{0x24, 0x01, 0xfa, 0x00, 0x00, 0x04, 0x10, 0x00, 0xbe, 0x30, 0x5b, 0xff,
>+      0xfe, 0xe5, 0x00, 0xc3}}};
>+static const in6_addr kIPv6PublicAddr2 = {
>+    {{0x24, 0x01, 0x00, 0x00, 0x00, 0x00, 0x10, 0x00, 0xbe, 0x30, 0x5b, 0xff,
>+      0xfe, 0xe5, 0x00, 0xc3}}};
>+static const in6_addr kIPv4MappedAnyAddr = {
>+    {{0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0xff, 0xff,
>+      0x00, 0x00, 0x00, 0x00}}};
>+static const in6_addr kIPv4MappedRFC1918Addr = {
>+    {{0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0xff, 0xff,
>+      0xc0, 0xa8, 0x07, 0x01}}};
>+static const in6_addr kIPv4MappedPublicAddr = {
>+    {{0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0xff, 0xff,
>+      0x01, 0x02, 0x03, 0x04}}};
> 
> static const std::string kIPv4AnyAddrString = "0.0.0.0";
> static const std::string kIPv4LoopbackAddrString = "127.0.0.1";
>@@ -79,10 +74,8 @@ static const std::string kIPv6BrokenString2 =
>     "2401:fa00:4:1000:be30:5bff:fee5:c3:1";
> static const std::string kIPv6BrokenString3 =
>     "[2401:fa00:4:1000:be30:5bff:fee5:c3]:1";
>-static const std::string kIPv6BrokenString4 =
>-    "2401::4::be30";
>-static const std::string kIPv6BrokenString5 =
>-    "2401:::4:fee5:be30";
>+static const std::string kIPv6BrokenString4 = "2401::4::be30";
>+static const std::string kIPv6BrokenString5 = "2401:::4:fee5:be30";
> static const std::string kIPv6BrokenString6 =
>     "2401f:fa00:4:1000:be30:5bff:fee5:c3";
> static const std::string kIPv6BrokenString7 =
>@@ -91,24 +84,18 @@ static const std::string kIPv6BrokenString8 =
>     "2401:fa000:4:1000:be30:5bff:fee5:c3";
> static const std::string kIPv6BrokenString9 =
>     "2401:fal0:4:1000:be30:5bff:fee5:c3";
>-static const std::string kIPv6BrokenString10 =
>-    "::ffff:192.168.7.";
>-static const std::string kIPv6BrokenString11 =
>-    "::ffff:192.168.7.1.1.1";
>-static const std::string kIPv6BrokenString12 =
>-    "::fffe:192.168.7.1";
>-static const std::string kIPv6BrokenString13 =
>-    "::ffff:192.168.7.ff";
>+static const std::string kIPv6BrokenString10 = "::ffff:192.168.7.";
>+static const std::string kIPv6BrokenString11 = "::ffff:192.168.7.1.1.1";
>+static const std::string kIPv6BrokenString12 = "::fffe:192.168.7.1";
>+static const std::string kIPv6BrokenString13 = "::ffff:192.168.7.ff";
> static const std::string kIPv6BrokenString14 =
>     "0x2401:fa00:4:1000:be30:5bff:fee5:c3";
> 
>-bool AreEqual(const IPAddress& addr,
>-              const IPAddress& addr2) {
>+bool AreEqual(const IPAddress& addr, const IPAddress& addr2) {
>   if ((IPIsAny(addr) != IPIsAny(addr2)) ||
>       (IPIsLoopback(addr) != IPIsLoopback(addr2)) ||
>       (IPIsPrivate(addr) != IPIsPrivate(addr2)) ||
>-      (HashIP(addr) != HashIP(addr2)) ||
>-      (addr.Size() != addr2.Size()) ||
>+      (HashIP(addr) != HashIP(addr2)) || (addr.Size() != addr2.Size()) ||
>       (addr.family() != addr2.family()) ||
>       (addr.ToString() != addr2.ToString())) {
>     return false;
>@@ -129,7 +116,7 @@ bool AreEqual(const IPAddress& addr,
> }
> 
> bool BrokenIPStringFails(const std::string& broken) {
>-  IPAddress addr(0);   // Intentionally make it v4.
>+  IPAddress addr(0);  // Intentionally make it v4.
>   if (IPFromString(kIPv4BrokenString1, &addr)) {
>     return false;
>   }
>@@ -139,7 +126,7 @@ bool BrokenIPStringFails(const std::string& broken) {
> bool CheckMaskCount(const std::string& mask, int expected_length) {
>   IPAddress addr;
>   return IPFromString(mask, &addr) &&
>-      (expected_length == CountIPMaskBits(addr));
>+         (expected_length == CountIPMaskBits(addr));
> }
> 
> bool TryInvalidMaskCount(const std::string& mask) {
>@@ -153,7 +140,8 @@ bool TryInvalidMaskCount(const std::string& mask) {
>   return true;
> }
> 
>-bool CheckTruncateIP(const std::string& initial, int truncate_length,
>+bool CheckTruncateIP(const std::string& initial,
>+                     int truncate_length,
>                      const std::string& expected_result) {
>   IPAddress addr, expected;
>   IPFromString(initial, &addr);
>@@ -287,39 +275,39 @@ TEST(IPAddressTest, TestCopyCtor) {
>   IPAddress addr(v4addr);
>   IPAddress addr2(addr);
> 
>-  EXPECT_PRED2(AreEqual, addr, addr2);
>+  EXPECT_TRUE(AreEqual(addr, addr2));
> 
>   addr = IPAddress(INADDR_ANY);
>   addr2 = IPAddress(addr);
>-  EXPECT_PRED2(AreEqual, addr, addr2);
>+  EXPECT_TRUE(AreEqual(addr, addr2));
> 
>   addr = IPAddress(INADDR_LOOPBACK);
>   addr2 = IPAddress(addr);
>-  EXPECT_PRED2(AreEqual, addr, addr2);
>+  EXPECT_TRUE(AreEqual(addr, addr2));
> 
>   addr = IPAddress(kIPv4PublicAddr);
>   addr2 = IPAddress(addr);
>-  EXPECT_PRED2(AreEqual, addr, addr2);
>+  EXPECT_TRUE(AreEqual(addr, addr2));
> 
>   addr = IPAddress(kIPv4RFC1918Addr);
>   addr2 = IPAddress(addr);
>-  EXPECT_PRED2(AreEqual, addr, addr2);
>+  EXPECT_TRUE(AreEqual(addr, addr2));
> 
>   addr = IPAddress(in6addr_any);
>   addr2 = IPAddress(addr);
>-  EXPECT_PRED2(AreEqual, addr, addr2);
>+  EXPECT_TRUE(AreEqual(addr, addr2));
> 
>   addr = IPAddress(in6addr_loopback);
>   addr2 = IPAddress(addr);
>-  EXPECT_PRED2(AreEqual, addr, addr2);
>+  EXPECT_TRUE(AreEqual(addr, addr2));
> 
>   addr = IPAddress(kIPv6LinkLocalAddr);
>   addr2 = IPAddress(addr);
>-  EXPECT_PRED2(AreEqual, addr, addr2);
>+  EXPECT_TRUE(AreEqual(addr, addr2));
> 
>   addr = IPAddress(kIPv6PublicAddr);
>   addr2 = IPAddress(addr);
>-  EXPECT_PRED2(AreEqual, addr, addr2);
>+  EXPECT_TRUE(AreEqual(addr, addr2));
> }
> 
> TEST(IPAddressTest, TestEquality) {
>@@ -389,14 +377,14 @@ TEST(IPAddressTest, TestEquality) {
> TEST(IPAddressTest, TestComparison) {
>   // Defined in 'ascending' order.
>   // v6 > v4, and intra-family sorting is purely numerical
>-  IPAddress addr0;  // AF_UNSPEC
>-  IPAddress addr1(INADDR_ANY);  // 0.0.0.0
>-  IPAddress addr2(kIPv4PublicAddr);  // 1.2.3.4
>-  IPAddress addr3(INADDR_LOOPBACK);  // 127.0.0.1
>-  IPAddress addr4(kIPv4RFC1918Addr);  // 192.168.7.1.
>-  IPAddress addr5(in6addr_any);  // ::
>-  IPAddress addr6(in6addr_loopback);  // ::1
>-  IPAddress addr7(kIPv6PublicAddr);  // 2401....
>+  IPAddress addr0;                      // AF_UNSPEC
>+  IPAddress addr1(INADDR_ANY);          // 0.0.0.0
>+  IPAddress addr2(kIPv4PublicAddr);     // 1.2.3.4
>+  IPAddress addr3(INADDR_LOOPBACK);     // 127.0.0.1
>+  IPAddress addr4(kIPv4RFC1918Addr);    // 192.168.7.1.
>+  IPAddress addr5(in6addr_any);         // ::
>+  IPAddress addr6(in6addr_loopback);    // ::1
>+  IPAddress addr7(kIPv6PublicAddr);     // 2401....
>   IPAddress addr8(kIPv6LinkLocalAddr);  // fe80....
> 
>   EXPECT_TRUE(addr0 < addr1);
>@@ -445,46 +433,46 @@ TEST(IPAddressTest, TestFromString) {
> 
>   EXPECT_TRUE(IPFromString(kIPv4AnyAddrString, &addr));
>   EXPECT_EQ(addr.ToString(), kIPv4AnyAddrString);
>-  EXPECT_PRED2(AreEqual, addr, addr2);
>+  EXPECT_TRUE(AreEqual(addr, addr2));
> 
>   addr2 = IPAddress(INADDR_LOOPBACK);
>   EXPECT_TRUE(IPFromString(kIPv4LoopbackAddrString, &addr));
>   EXPECT_EQ(addr.ToString(), kIPv4LoopbackAddrString);
>-  EXPECT_PRED2(AreEqual, addr, addr2);
>+  EXPECT_TRUE(AreEqual(addr, addr2));
> 
>   addr2 = IPAddress(kIPv4RFC1918Addr);
>   EXPECT_TRUE(IPFromString(kIPv4RFC1918AddrString, &addr));
>   EXPECT_EQ(addr.ToString(), kIPv4RFC1918AddrString);
>-  EXPECT_PRED2(AreEqual, addr, addr2);
>+  EXPECT_TRUE(AreEqual(addr, addr2));
> 
>   addr2 = IPAddress(kIPv4PublicAddr);
>   EXPECT_TRUE(IPFromString(kIPv4PublicAddrString, &addr));
>   EXPECT_EQ(addr.ToString(), kIPv4PublicAddrString);
>-  EXPECT_PRED2(AreEqual, addr, addr2);
>+  EXPECT_TRUE(AreEqual(addr, addr2));
> 
>   addr2 = IPAddress(in6addr_any);
>   EXPECT_TRUE(IPFromString(kIPv6AnyAddrString, &addr));
>   EXPECT_EQ(addr.ToString(), kIPv6AnyAddrString);
>-  EXPECT_PRED2(AreEqual, addr, addr2);
>+  EXPECT_TRUE(AreEqual(addr, addr2));
> 
>   addr2 = IPAddress(in6addr_loopback);
>   EXPECT_TRUE(IPFromString(kIPv6LoopbackAddrString, &addr));
>   EXPECT_EQ(addr.ToString(), kIPv6LoopbackAddrString);
>-  EXPECT_PRED2(AreEqual, addr, addr2);
>+  EXPECT_TRUE(AreEqual(addr, addr2));
> 
>   addr2 = IPAddress(kIPv6LinkLocalAddr);
>   EXPECT_TRUE(IPFromString(kIPv6LinkLocalAddrString, &addr));
>   EXPECT_EQ(addr.ToString(), kIPv6LinkLocalAddrString);
>-  EXPECT_PRED2(AreEqual, addr, addr2);
>+  EXPECT_TRUE(AreEqual(addr, addr2));
> 
>   addr2 = IPAddress(kIPv6PublicAddr);
>   EXPECT_TRUE(IPFromString(kIPv6PublicAddrString, &addr));
>   EXPECT_EQ(addr.ToString(), kIPv6PublicAddrString);
>-  EXPECT_PRED2(AreEqual, addr, addr2);
>+  EXPECT_TRUE(AreEqual(addr, addr2));
> 
>   addr2 = IPAddress(kIPv4MappedRFC1918Addr);
>   EXPECT_TRUE(IPFromString(kIPv4MappedV4StyleAddrString, &addr));
>-  EXPECT_PRED2(AreEqual, addr, addr2);
>+  EXPECT_TRUE(AreEqual(addr, addr2));
> 
>   // Broken cases, should set addr to AF_UNSPEC.
>   EXPECT_PRED1(BrokenIPStringFails, kIPv4BrokenString1);
>@@ -581,6 +569,28 @@ TEST(IPAddressTest, TestIsLoopback) {
>   EXPECT_TRUE(IPIsLoopback(IPAddress(in6addr_loopback)));
> }
> 
>+TEST(IPAddressTest, TestIsLinkLocal) {
>+  // "any" addresses
>+  EXPECT_FALSE(IPIsLinkLocal(IPAddress(INADDR_ANY)));
>+  EXPECT_FALSE(IPIsLinkLocal(IPAddress(in6addr_any)));
>+  // loopback addresses
>+  EXPECT_FALSE(IPIsLinkLocal(IPAddress(INADDR_LOOPBACK)));
>+  EXPECT_FALSE(IPIsLinkLocal(IPAddress(in6addr_loopback)));
>+  // public addresses
>+  EXPECT_FALSE(IPIsLinkLocal(IPAddress(kIPv4PublicAddr)));
>+  EXPECT_FALSE(IPIsLinkLocal(IPAddress(kIPv6PublicAddr)));
>+  // private network addresses
>+  EXPECT_FALSE(IPIsLinkLocal(IPAddress(kIPv4RFC1918Addr)));
>+  // mapped addresses
>+  EXPECT_FALSE(IPIsLinkLocal(IPAddress(kIPv4MappedAnyAddr)));
>+  EXPECT_FALSE(IPIsLinkLocal(IPAddress(kIPv4MappedPublicAddr)));
>+  EXPECT_FALSE(IPIsLinkLocal(IPAddress(kIPv4MappedRFC1918Addr)));
>+
>+  // link-local network addresses
>+  EXPECT_TRUE(IPIsLinkLocal(IPAddress(kIPv4LinkLocalAddr)));
>+  EXPECT_TRUE(IPIsLinkLocal(IPAddress(kIPv6LinkLocalAddr)));
>+}
>+
> // Verify that IPIsAny catches all cases of "any" address.
> TEST(IPAddressTest, TestIsAny) {
>   IPAddress addr;
>@@ -918,8 +928,7 @@ TEST(IPAddressTest, TestToSensitiveString) {
> 
> TEST(IPAddressTest, TestInterfaceAddress) {
>   in6_addr addr;
>-  InterfaceAddress addr1(kIPv6PublicAddr,
>-                         IPV6_ADDRESS_FLAG_TEMPORARY);
>+  InterfaceAddress addr1(kIPv6PublicAddr, IPV6_ADDRESS_FLAG_TEMPORARY);
>   EXPECT_EQ(addr1.ipv6_flags(), IPV6_ADDRESS_FLAG_TEMPORARY);
>   EXPECT_EQ(addr1.family(), AF_INET6);
> 
>@@ -938,18 +947,16 @@ TEST(IPAddressTest, TestInterfaceAddress) {
>   addr = addr3.ipv6_address();
>   EXPECT_TRUE(IN6_ARE_ADDR_EQUAL(&addr, &kIPv6PublicAddr));
> 
>-  InterfaceAddress addr4(kIPv6PublicAddr,
>-                         IPV6_ADDRESS_FLAG_DEPRECATED);
>+  InterfaceAddress addr4(kIPv6PublicAddr, IPV6_ADDRESS_FLAG_DEPRECATED);
>   EXPECT_NE(addr1, addr4);
> 
>   // When you compare them as IPAddress, since operator==
>   // is not virtual, it'll be equal.
>-  IPAddress *paddr1 = &addr1;
>-  IPAddress *paddr4 = &addr4;
>+  IPAddress* paddr1 = &addr1;
>+  IPAddress* paddr4 = &addr4;
>   EXPECT_EQ(*paddr1, *paddr4);
> 
>-  InterfaceAddress addr5(kIPv6LinkLocalAddr,
>-                         IPV6_ADDRESS_FLAG_TEMPORARY);
>+  InterfaceAddress addr5(kIPv6LinkLocalAddr, IPV6_ADDRESS_FLAG_TEMPORARY);
>   EXPECT_NE(addr1, addr5);
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/java/src/org/webrtc/ContextUtils.java b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/java/src/org/webrtc/ContextUtils.java
>index a53059a9cce..f0e6ef07361 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/java/src/org/webrtc/ContextUtils.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/java/src/org/webrtc/ContextUtils.java
>@@ -36,7 +36,10 @@ public class ContextUtils {
> 
>   /**
>    * Returns the stored application context.
>+   *
>+   * @deprecated crbug.com/webrtc/8937
>    */
>+  @Deprecated
>   public static Context getApplicationContext() {
>     return applicationContext;
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/java/src/org/webrtc/Loggable.java b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/java/src/org/webrtc/Loggable.java
>new file mode 100644
>index 00000000000..cd66aa12146
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/java/src/org/webrtc/Loggable.java
>@@ -0,0 +1,22 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc;
>+
>+import org.webrtc.Logging.Severity;
>+
>+/**
>+ * Java interface for WebRTC logging. The default implementation uses webrtc.Logging.
>+ *
>+ * When injected, the Loggable will receive logging from both Java and native.
>+ */
>+public interface Loggable {
>+  public void onLogMessage(String message, Severity severity, String tag);
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/java/src/org/webrtc/Logging.java b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/java/src/org/webrtc/Logging.java
>index f143d2f8b2c..aafdbe8023a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/java/src/org/webrtc/Logging.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/java/src/org/webrtc/Logging.java
>@@ -15,21 +15,35 @@ import java.io.StringWriter;
> import java.util.EnumSet;
> import java.util.logging.Level;
> import java.util.logging.Logger;
>+import javax.annotation.Nullable;
>+import org.webrtc.Loggable;
> 
> /**
>- * Java wrapper for WebRTC logging. Logging defaults to java.util.logging.Logger, but will switch to
>- * native logging (rtc::LogMessage) if one of the following static functions are called from the
>- * app:
>+ * Java wrapper for WebRTC logging. Logging defaults to java.util.logging.Logger, but a custom
>+ * logger implementing the Loggable interface can be injected along with a Severity. All subsequent
>+ * log messages will then be redirected to the injected Loggable, except those with a severity lower
>+ * than the specified severity, which will be discarded.
>+ *
>+ * It is also possible to switch to native logging (rtc::LogMessage) if one of the following static
>+ * functions are called from the app:
>  * - Logging.enableLogThreads
>  * - Logging.enableLogTimeStamps
>  * - Logging.enableLogToDebugOutput
>  *
>- * Using these APIs requires that the native library is loaded, using
>- * PeerConnectionFactory.initialize.
>+ * The priority goes:
>+ * 1. Injected loggable
>+ * 2. Native logging
>+ * 3. Fallback logging.
>+ * Only one method will be used at a time.
>+ *
>+ * Injecting a Loggable or using any of the enable... methods requires that the native library is
>+ * loaded, using PeerConnectionFactory.initialize.
>  */
> public class Logging {
>   private static final Logger fallbackLogger = createFallbackLogger();
>   private static volatile boolean loggingEnabled;
>+  @Nullable private static Loggable loggable;
>+  private static Severity loggableSeverity;
> 
>   private static Logger createFallbackLogger() {
>     final Logger fallbackLogger = Logger.getLogger("org.webrtc.Logging");
>@@ -37,6 +51,17 @@ public class Logging {
>     return fallbackLogger;
>   }
> 
>+  static void injectLoggable(Loggable injectedLoggable, Severity severity) {
>+    if (injectedLoggable != null) {
>+      loggable = injectedLoggable;
>+      loggableSeverity = severity;
>+    }
>+  }
>+
>+  static void deleteInjectedLoggable() {
>+    loggable = null;
>+  }
>+
>   // TODO(solenberg): Remove once dependent projects updated.
>   @Deprecated
>   public enum TraceLevel {
>@@ -83,11 +108,29 @@ public class Logging {
>   // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
>   @SuppressWarnings("NoSynchronizedMethodCheck")
>   public static synchronized void enableLogToDebugOutput(Severity severity) {
>+    if (loggable != null) {
>+      throw new IllegalStateException(
>+          "Logging to native debug output not supported while Loggable is injected. "
>+          + "Delete the Loggable before calling this method.");
>+    }
>     nativeEnableLogToDebugOutput(severity.ordinal());
>     loggingEnabled = true;
>   }
> 
>   public static void log(Severity severity, String tag, String message) {
>+    if (tag == null || message == null) {
>+      throw new IllegalArgumentException("Logging tag or message may not be null.");
>+    }
>+    if (loggable != null) {
>+      // Filter log messages below loggableSeverity.
>+      if (severity.ordinal() < loggableSeverity.ordinal()) {
>+        return;
>+      }
>+      loggable.onLogMessage(message, severity, tag);
>+      return;
>+    }
>+
>+    // Try native logging if no loggable is injected.
>     if (loggingEnabled) {
>       nativeLog(severity.ordinal(), tag, message);
>       return;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/java/src/org/webrtc/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/java/src/org/webrtc/OWNERS
>index 4d31ffb663f..299e8b20ec0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/java/src/org/webrtc/OWNERS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/java/src/org/webrtc/OWNERS
>@@ -1 +1,2 @@
> magjed@webrtc.org
>+sakal@webrtc.org
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/java/src/org/webrtc/ThreadUtils.java b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/java/src/org/webrtc/ThreadUtils.java
>index f7c7b81a789..a4038709065 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/java/src/org/webrtc/ThreadUtils.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/java/src/org/webrtc/ThreadUtils.java
>@@ -16,13 +16,14 @@ import android.os.SystemClock;
> import java.util.concurrent.Callable;
> import java.util.concurrent.CountDownLatch;
> import java.util.concurrent.TimeUnit;
>+import javax.annotation.Nullable;
> 
> public class ThreadUtils {
>   /**
>    * Utility class to be used for checking that a method is called on the correct thread.
>    */
>   public static class ThreadChecker {
>-    private Thread thread = Thread.currentThread();
>+    @Nullable private Thread thread = Thread.currentThread();
> 
>     public void checkIsOnValidThread() {
>       if (thread == null) {
>@@ -143,17 +144,6 @@ public class ThreadUtils {
>     return result;
>   }
> 
>-  // TODO(sakal): This method is broken. It should be removed: crbug.com/webrtc/8456
>-  @SuppressWarnings("WaitNotInLoop")
>-  public static void waitUninterruptibly(final Object object) {
>-    executeUninterruptibly(new BlockingOperation() {
>-      @Override
>-      public void run() throws InterruptedException {
>-        object.wait();
>-      }
>-    });
>-  }
>-
>   /**
>    * Post |callable| to |handler| and wait for the result.
>    */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/json.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/json.cc
>index b8071a99bd6..cc40bd678de 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/json.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/json.cc
>@@ -52,8 +52,8 @@ bool GetIntFromJson(const Json::Value& in, int* out) {
>     char* end_ptr;
>     errno = 0;
>     val = strtol(c_str, &end_ptr, 10);  // NOLINT
>-    ret = (end_ptr != c_str && *end_ptr == '\0' && !errno &&
>-           val >= INT_MIN && val <= INT_MAX);
>+    ret = (end_ptr != c_str && *end_ptr == '\0' && !errno && val >= INT_MIN &&
>+           val <= INT_MAX);
>     *out = val;
>   }
>   return ret;
>@@ -72,8 +72,7 @@ bool GetUIntFromJson(const Json::Value& in, unsigned int* out) {
>     char* end_ptr;
>     errno = 0;
>     val = strtoul(c_str, &end_ptr, 10);  // NOLINT
>-    ret = (end_ptr != c_str && *end_ptr == '\0' && !errno &&
>-           val <= UINT_MAX);
>+    ret = (end_ptr != c_str && *end_ptr == '\0' && !errno && val <= UINT_MAX);
>     *out = val;
>   }
>   return ret;
>@@ -120,10 +119,10 @@ bool GetDoubleFromJson(const Json::Value& in, double* out) {
> }
> 
> namespace {
>-template<typename T>
>+template <typename T>
> bool JsonArrayToVector(const Json::Value& value,
>                        bool (*getter)(const Json::Value& in, T* out),
>-                       std::vector<T> *vec) {
>+                       std::vector<T>* vec) {
>   vec->clear();
>   if (!value.isArray()) {
>     return false;
>@@ -151,8 +150,7 @@ bool JsonArrayToValueVector(const Json::Value& in,
>   return JsonArrayToVector(in, GetValueFromJson, out);
> }
> 
>-bool JsonArrayToIntVector(const Json::Value& in,
>-                          std::vector<int>* out) {
>+bool JsonArrayToIntVector(const Json::Value& in, std::vector<int>* out) {
>   return JsonArrayToVector(in, GetIntFromJson, out);
> }
> 
>@@ -166,18 +164,16 @@ bool JsonArrayToStringVector(const Json::Value& in,
>   return JsonArrayToVector(in, GetStringFromJson, out);
> }
> 
>-bool JsonArrayToBoolVector(const Json::Value& in,
>-                           std::vector<bool>* out) {
>+bool JsonArrayToBoolVector(const Json::Value& in, std::vector<bool>* out) {
>   return JsonArrayToVector(in, GetBoolFromJson, out);
> }
> 
>-bool JsonArrayToDoubleVector(const Json::Value& in,
>-                             std::vector<double>* out) {
>+bool JsonArrayToDoubleVector(const Json::Value& in, std::vector<double>* out) {
>   return JsonArrayToVector(in, GetDoubleFromJson, out);
> }
> 
> namespace {
>-template<typename T>
>+template <typename T>
> Json::Value VectorToJsonArray(const std::vector<T>& vec) {
>   Json::Value result(Json::arrayValue);
>   for (size_t i = 0; i < vec.size(); ++i) {
>@@ -211,8 +207,7 @@ Json::Value DoubleVectorToJsonArray(const std::vector<double>& in) {
>   return VectorToJsonArray(in);
> }
> 
>-bool GetValueFromJsonArray(const Json::Value& in, size_t n,
>-                           Json::Value* out) {
>+bool GetValueFromJsonArray(const Json::Value& in, size_t n, Json::Value* out) {
>   if (!in.isArray() || !in.isValidIndex(static_cast<int>(n))) {
>     return false;
>   }
>@@ -221,37 +216,33 @@ bool GetValueFromJsonArray(const Json::Value& in, size_t n,
>   return true;
> }
> 
>-bool GetIntFromJsonArray(const Json::Value& in, size_t n,
>-                         int* out) {
>+bool GetIntFromJsonArray(const Json::Value& in, size_t n, int* out) {
>   Json::Value x;
>   return GetValueFromJsonArray(in, n, &x) && GetIntFromJson(x, out);
> }
> 
>-bool GetUIntFromJsonArray(const Json::Value& in, size_t n,
>-                          unsigned int* out)  {
>+bool GetUIntFromJsonArray(const Json::Value& in, size_t n, unsigned int* out) {
>   Json::Value x;
>   return GetValueFromJsonArray(in, n, &x) && GetUIntFromJson(x, out);
> }
> 
>-bool GetStringFromJsonArray(const Json::Value& in, size_t n,
>-                            std::string* out) {
>+bool GetStringFromJsonArray(const Json::Value& in, size_t n, std::string* out) {
>   Json::Value x;
>   return GetValueFromJsonArray(in, n, &x) && GetStringFromJson(x, out);
> }
> 
>-bool GetBoolFromJsonArray(const Json::Value& in, size_t n,
>-                          bool* out) {
>+bool GetBoolFromJsonArray(const Json::Value& in, size_t n, bool* out) {
>   Json::Value x;
>   return GetValueFromJsonArray(in, n, &x) && GetBoolFromJson(x, out);
> }
> 
>-bool GetDoubleFromJsonArray(const Json::Value& in, size_t n,
>-                            double* out) {
>+bool GetDoubleFromJsonArray(const Json::Value& in, size_t n, double* out) {
>   Json::Value x;
>   return GetValueFromJsonArray(in, n, &x) && GetDoubleFromJson(x, out);
> }
> 
>-bool GetValueFromJsonObject(const Json::Value& in, const std::string& k,
>+bool GetValueFromJsonObject(const Json::Value& in,
>+                            const std::string& k,
>                             Json::Value* out) {
>   if (!in.isObject() || !in.isMember(k)) {
>     return false;
>@@ -261,31 +252,36 @@ bool GetValueFromJsonObject(const Json::Value& in, const std::string& k,
>   return true;
> }
> 
>-bool GetIntFromJsonObject(const Json::Value& in, const std::string& k,
>+bool GetIntFromJsonObject(const Json::Value& in,
>+                          const std::string& k,
>                           int* out) {
>   Json::Value x;
>   return GetValueFromJsonObject(in, k, &x) && GetIntFromJson(x, out);
> }
> 
>-bool GetUIntFromJsonObject(const Json::Value& in, const std::string& k,
>-                           unsigned int* out)  {
>+bool GetUIntFromJsonObject(const Json::Value& in,
>+                           const std::string& k,
>+                           unsigned int* out) {
>   Json::Value x;
>   return GetValueFromJsonObject(in, k, &x) && GetUIntFromJson(x, out);
> }
> 
>-bool GetStringFromJsonObject(const Json::Value& in, const std::string& k,
>-                             std::string* out)  {
>+bool GetStringFromJsonObject(const Json::Value& in,
>+                             const std::string& k,
>+                             std::string* out) {
>   Json::Value x;
>   return GetValueFromJsonObject(in, k, &x) && GetStringFromJson(x, out);
> }
> 
>-bool GetBoolFromJsonObject(const Json::Value& in, const std::string& k,
>+bool GetBoolFromJsonObject(const Json::Value& in,
>+                           const std::string& k,
>                            bool* out) {
>   Json::Value x;
>   return GetValueFromJsonObject(in, k, &x) && GetBoolFromJson(x, out);
> }
> 
>-bool GetDoubleFromJsonObject(const Json::Value& in, const std::string& k,
>+bool GetDoubleFromJsonObject(const Json::Value& in,
>+                             const std::string& k,
>                              double* out) {
>   Json::Value x;
>   return GetValueFromJsonObject(in, k, &x) && GetDoubleFromJson(x, out);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/json.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/json.h
>index 5db8bd6c90d..b8e6d959186 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/json.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/json.h
>@@ -34,32 +34,23 @@ bool GetBoolFromJson(const Json::Value& in, bool* out);
> bool GetDoubleFromJson(const Json::Value& in, double* out);
> 
> // Pull values out of a JSON array.
>-bool GetValueFromJsonArray(const Json::Value& in, size_t n,
>-                           Json::Value* out);
>-bool GetIntFromJsonArray(const Json::Value& in, size_t n,
>-                         int* out);
>-bool GetUIntFromJsonArray(const Json::Value& in, size_t n,
>-                          unsigned int* out);
>-bool GetStringFromJsonArray(const Json::Value& in, size_t n,
>-                            std::string* out);
>-bool GetBoolFromJsonArray(const Json::Value& in, size_t n,
>-                          bool* out);
>-bool GetDoubleFromJsonArray(const Json::Value& in, size_t n,
>-                            double* out);
>+bool GetValueFromJsonArray(const Json::Value& in, size_t n, Json::Value* out);
>+bool GetIntFromJsonArray(const Json::Value& in, size_t n, int* out);
>+bool GetUIntFromJsonArray(const Json::Value& in, size_t n, unsigned int* out);
>+bool GetStringFromJsonArray(const Json::Value& in, size_t n, std::string* out);
>+bool GetBoolFromJsonArray(const Json::Value& in, size_t n, bool* out);
>+bool GetDoubleFromJsonArray(const Json::Value& in, size_t n, double* out);
> 
> // Convert json arrays to std::vector
> bool JsonArrayToValueVector(const Json::Value& in,
>                             std::vector<Json::Value>* out);
>-bool JsonArrayToIntVector(const Json::Value& in,
>-                          std::vector<int>* out);
>+bool JsonArrayToIntVector(const Json::Value& in, std::vector<int>* out);
> bool JsonArrayToUIntVector(const Json::Value& in,
>                            std::vector<unsigned int>* out);
> bool JsonArrayToStringVector(const Json::Value& in,
>                              std::vector<std::string>* out);
>-bool JsonArrayToBoolVector(const Json::Value& in,
>-                           std::vector<bool>* out);
>-bool JsonArrayToDoubleVector(const Json::Value& in,
>-                             std::vector<double>* out);
>+bool JsonArrayToBoolVector(const Json::Value& in, std::vector<bool>* out);
>+bool JsonArrayToDoubleVector(const Json::Value& in, std::vector<double>* out);
> 
> // Convert std::vector to json array
> Json::Value ValueVectorToJsonArray(const std::vector<Json::Value>& in);
>@@ -70,17 +61,23 @@ Json::Value BoolVectorToJsonArray(const std::vector<bool>& in);
> Json::Value DoubleVectorToJsonArray(const std::vector<double>& in);
> 
> // Pull values out of a JSON object.
>-bool GetValueFromJsonObject(const Json::Value& in, const std::string& k,
>+bool GetValueFromJsonObject(const Json::Value& in,
>+                            const std::string& k,
>                             Json::Value* out);
>-bool GetIntFromJsonObject(const Json::Value& in, const std::string& k,
>+bool GetIntFromJsonObject(const Json::Value& in,
>+                          const std::string& k,
>                           int* out);
>-bool GetUIntFromJsonObject(const Json::Value& in, const std::string& k,
>+bool GetUIntFromJsonObject(const Json::Value& in,
>+                           const std::string& k,
>                            unsigned int* out);
>-bool GetStringFromJsonObject(const Json::Value& in, const std::string& k,
>+bool GetStringFromJsonObject(const Json::Value& in,
>+                             const std::string& k,
>                              std::string* out);
>-bool GetBoolFromJsonObject(const Json::Value& in, const std::string& k,
>+bool GetBoolFromJsonObject(const Json::Value& in,
>+                           const std::string& k,
>                            bool* out);
>-bool GetDoubleFromJsonObject(const Json::Value& in, const std::string& k,
>+bool GetDoubleFromJsonObject(const Json::Value& in,
>+                             const std::string& k,
>                              double* out);
> 
> // Writes out a Json value as a string.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/keep_ref_until_done.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/keep_ref_until_done.h
>index 979415d1314..9ba0e879eba 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/keep_ref_until_done.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/keep_ref_until_done.h
>@@ -39,5 +39,4 @@ static inline Callback0<void> KeepRefUntilDone(
> 
> }  // namespace rtc
> 
>-
> #endif  // RTC_BASE_KEEP_REF_UNTIL_DONE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging.cc
>index b95f646f24c..c386600abba 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging.cc
>@@ -9,9 +9,6 @@
>  */
> 
> #if defined(WEBRTC_WIN)
>-#if !defined(WIN32_LEAN_AND_MEAN)
>-#define WIN32_LEAN_AND_MEAN
>-#endif
> #include <windows.h>
> #if _MSC_VER < 1900
> #define snprintf _snprintf
>@@ -29,27 +26,33 @@
> static const int kMaxLogLineSize = 1024 - 60;
> #endif  // WEBRTC_MAC && !defined(WEBRTC_IOS) || WEBRTC_ANDROID
> 
>-static const char kLibjingle[] = "libjingle";
>-
>-#include <time.h>
> #include <limits.h>
>+#include <time.h>
> 
> #include <algorithm>
>+#include <cstdarg>
> #include <iomanip>
>-#include <mutex>
> #include <ostream>
> #include <vector>
>-#include <type_traits>
> 
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/platform_thread.h"
>+#include "rtc_base/platform_thread_types.h"
> #include "rtc_base/stringencode.h"
>+#include "rtc_base/strings/string_builder.h"
> #include "rtc_base/stringutils.h"
> #include "rtc_base/timeutils.h"
> 
> namespace rtc {
> namespace {
>+// By default, release builds don't log, debug builds at info level
>+#if !defined(NDEBUG)
>+static LoggingSeverity g_min_sev = LS_INFO;
>+static LoggingSeverity g_dbg_sev = LS_INFO;
>+#else
>+static LoggingSeverity g_min_sev = LS_NONE;
>+static LoggingSeverity g_dbg_sev = LS_NONE;
>+#endif
> 
> // Return the filename portion of the string (that following the last slash).
> const char* FilenameFromPath(const char* file) {
>@@ -61,89 +64,54 @@ const char* FilenameFromPath(const char* file) {
>     return (end1 > end2) ? end1 + 1 : end2 + 1;
> }
> 
>-}  // namespace
>-
>-/////////////////////////////////////////////////////////////////////////////
>-// Constant Labels
>-/////////////////////////////////////////////////////////////////////////////
>-
>-const char* FindLabel(int value, const ConstantLabel entries[]) {
>-  for (int i = 0; entries[i].label; ++i) {
>-    if (value == entries[i].value) {
>-      return entries[i].label;
>-    }
>-  }
>-  return 0;
>+std::ostream& GetNoopStream() {
>+  class NoopStreamBuf : public std::streambuf {
>+   public:
>+    int overflow(int c) override { return c; }
>+  };
>+  static NoopStreamBuf noop_buffer;
>+  static std::ostream noop_stream(&noop_buffer);
>+  return noop_stream;
> }
> 
>-std::string ErrorName(int err, const ConstantLabel* err_table) {
>-  if (err == 0)
>-    return "No error";
>-
>-  if (err_table != 0) {
>-    if (const char* value = FindLabel(err, err_table))
>-      return value;
>-  }
>+// Global lock for log subsystem, only needed to serialize access to streams_.
>+CriticalSection g_log_crit;
>+}  // namespace
> 
>-  char buffer[16];
>-  snprintf(buffer, sizeof(buffer), "0x%08x", err);
>-  return buffer;
>+// Inefficient default implementation, override is recommended.
>+void LogSink::OnLogMessage(const std::string& msg,
>+                           LoggingSeverity severity,
>+                           const char* tag) {
>+  OnLogMessage(tag + (": " + msg));
> }
> 
> /////////////////////////////////////////////////////////////////////////////
> // LogMessage
> /////////////////////////////////////////////////////////////////////////////
> 
>-// By default, release builds don't log, debug builds at info level
>-#if !defined(NDEBUG)
>-LoggingSeverity LogMessage::min_sev_ = LS_INFO;
>-LoggingSeverity LogMessage::dbg_sev_ = LS_INFO;
>-#else
>-LoggingSeverity LogMessage::min_sev_ = LS_NONE;
>-LoggingSeverity LogMessage::dbg_sev_ = LS_NONE;
>-#endif
> bool LogMessage::log_to_stderr_ = true;
> 
>-namespace {
>-
>-// Global lock for log subsystem, only needed to serialize access to streamList().
>-CriticalSection& logCriticalScope() {
>-  static std::aligned_storage<sizeof(CriticalSection), std::alignment_of<CriticalSection>::value>::type g_log_crit_storage;
>-  static CriticalSection* g_log_crit = new (&g_log_crit_storage) CriticalSection;
>-  return *g_log_crit;
>-}
>-
>-}  // namespace
>-
> // The list of logging streams currently configured.
> // Note: we explicitly do not clean this up, because of the uncertain ordering
> // of destructors at program exit.  Let the person who sets the stream trigger
> // cleanup by setting to null, or let it leak (safe at program exit).
>-
>-typedef std::pair<LogSink*, LoggingSeverity> StreamAndSeverity;
>-typedef std::list<StreamAndSeverity> StreamList;
>-
>-// The output streams and their associated severities
>-StreamList& streamList()
>-    RTC_EXCLUSIVE_LOCKS_REQUIRED(logCriticalScope()) {
>-  static std::aligned_storage<sizeof(StreamList), std::alignment_of<StreamList>::value>::type stream_list_storage;
>-  static StreamList* stream_list = new (&stream_list_storage) StreamList;
>-  return *stream_list;
>-}
>+LogMessage::StreamList LogMessage::streams_ RTC_GUARDED_BY(g_log_crit);
> 
> // Boolean options default to false (0)
> bool LogMessage::thread_, LogMessage::timestamp_;
> 
>+LogMessage::LogMessage(const char* file, int line, LoggingSeverity sev)
>+    : LogMessage(file, line, sev, ERRCTX_NONE, 0) {}
>+
> LogMessage::LogMessage(const char* file,
>                        int line,
>                        LoggingSeverity sev,
>                        LogErrorContext err_ctx,
>-                       int err,
>-                       const char* module)
>-    : severity_(sev), tag_(kLibjingle) {
>-
>-  static std::once_flag callLogCriticalScopeOnce;
>-  std::call_once(callLogCriticalScopeOnce,[] { logCriticalScope(); });
>+                       int err)
>+    : severity_(sev), is_noop_(IsNoop(sev)) {
>+  // If there's no need to do any work, let's not :)
>+  if (is_noop_)
>+    return;
> 
>   if (timestamp_) {
>     // Use SystemTimeMillis so that even if tests use fake clocks, the timestamp
>@@ -162,12 +130,19 @@ LogMessage::LogMessage(const char* file,
>     print_stream_ << "[" << std::dec << id << "] ";
>   }
> 
>-  if (file != nullptr)
>-    print_stream_ << "(" << FilenameFromPath(file)  << ":" << line << "): ";
>+  if (file != nullptr) {
>+#if defined(WEBRTC_ANDROID)
>+    tag_ = FilenameFromPath(file);
>+    print_stream_ << "(line " << line << "): ";
>+#else
>+    print_stream_ << "(" << FilenameFromPath(file) << ":" << line << "): ";
>+#endif
>+  }
> 
>   if (err_ctx != ERRCTX_NONE) {
>-    std::ostringstream tmp;
>-    tmp << "[0x" << std::setfill('0') << std::hex << std::setw(8) << err << "]";
>+    char tmp_buf[1024];
>+    SimpleStringBuilder tmp(tmp_buf);
>+    tmp.AppendFormat("[0x%08X]", err);
>     switch (err_ctx) {
>       case ERRCTX_ERRNO:
>         tmp << " " << strerror(err);
>@@ -175,15 +150,13 @@ LogMessage::LogMessage(const char* file,
> #ifdef WEBRTC_WIN
>       case ERRCTX_HRESULT: {
>         char msgbuf[256];
>-        DWORD flags = FORMAT_MESSAGE_FROM_SYSTEM;
>-        HMODULE hmod = GetModuleHandleA(module);
>-        if (hmod)
>-          flags |= FORMAT_MESSAGE_FROM_HMODULE;
>+        DWORD flags =
>+            FORMAT_MESSAGE_FROM_SYSTEM | FORMAT_MESSAGE_IGNORE_INSERTS;
>         if (DWORD len = FormatMessageA(
>-                flags, hmod, err, MAKELANGID(LANG_NEUTRAL, SUBLANG_DEFAULT),
>+                flags, nullptr, err, MAKELANGID(LANG_NEUTRAL, SUBLANG_DEFAULT),
>                 msgbuf, sizeof(msgbuf) / sizeof(msgbuf[0]), nullptr)) {
>           while ((len > 0) &&
>-              isspace(static_cast<unsigned char>(msgbuf[len-1]))) {
>+                 isspace(static_cast<unsigned char>(msgbuf[len - 1]))) {
>             msgbuf[--len] = 0;
>           }
>           tmp << " " << msgbuf;
>@@ -205,38 +178,86 @@ LogMessage::LogMessage(const char* file,
>   }
> }
> 
>+#if defined(WEBRTC_ANDROID)
>+LogMessage::LogMessage(const char* file,
>+                       int line,
>+                       LoggingSeverity sev,
>+                       const char* tag)
>+    : LogMessage(file, line, sev, ERRCTX_NONE, 0 /* err */) {
>+  if (!is_noop_) {
>+    tag_ = tag;
>+    print_stream_ << tag << ": ";
>+  }
>+}
>+#endif
>+
>+// DEPRECATED. Currently only used by downstream projects that use
>+// implementation details of logging.h. Work is ongoing to remove those
>+// dependencies.
> LogMessage::LogMessage(const char* file,
>                        int line,
>                        LoggingSeverity sev,
>                        const std::string& tag)
>-    : LogMessage(file,
>-                 line,
>-                 sev,
>-                 ERRCTX_NONE,
>-                 0 /* err */,
>-                 nullptr /* module */) {
>-  tag_ = tag;
>-  print_stream_ << tag << ": ";
>+    : LogMessage(file, line, sev) {
>+  if (!is_noop_)
>+    print_stream_ << tag << ": ";
> }
> 
> LogMessage::~LogMessage() {
>-  if (!extra_.empty())
>-    print_stream_ << " : " << extra_;
>-  print_stream_ << std::endl;
>+  if (is_noop_)
>+    return;
> 
>-  const std::string& str = print_stream_.str();
>-  if (severity_ >= dbg_sev_) {
>+  FinishPrintStream();
>+
>+  // TODO(tommi): Unfortunately |ostringstream::str()| always returns a copy
>+  // of the constructed string. This means that we always end up creating
>+  // two copies here (one owned by the stream, one by the return value of
>+  // |str()|). It would be nice to switch to something else.
>+  const std::string str = print_stream_.str();
>+
>+  if (severity_ >= g_dbg_sev) {
>+#if defined(WEBRTC_ANDROID)
>     OutputToDebug(str, severity_, tag_);
>+#else
>+    OutputToDebug(str, severity_);
>+#endif
>   }
> 
>-  CritScope cs(&logCriticalScope());
>-  for (auto& kv : streamList()) {
>+  CritScope cs(&g_log_crit);
>+  for (auto& kv : streams_) {
>     if (severity_ >= kv.second) {
>+#if defined(WEBRTC_ANDROID)
>+      kv.first->OnLogMessage(str, severity_, tag_);
>+#else
>       kv.first->OnLogMessage(str);
>+#endif
>     }
>   }
> }
> 
>+void LogMessage::AddTag(const char* tag) {
>+#ifdef WEBRTC_ANDROID
>+  if (!is_noop_) {
>+    tag_ = tag;
>+  }
>+#endif
>+}
>+
>+std::ostream& LogMessage::stream() {
>+  return is_noop_ ? GetNoopStream() : print_stream_;
>+}
>+
>+bool LogMessage::Loggable(LoggingSeverity sev) {
>+  return sev >= g_min_sev;
>+}
>+
>+int LogMessage::GetMinLogSeverity() {
>+  return g_min_sev;
>+}
>+
>+LoggingSeverity LogMessage::GetLogToDebug() {
>+  return g_dbg_sev;
>+}
> int64_t LogMessage::LogStartTime() {
>   static const int64_t g_start = SystemTimeMillis();
>   return g_start;
>@@ -256,8 +277,8 @@ void LogMessage::LogTimestamps(bool on) {
> }
> 
> void LogMessage::LogToDebug(LoggingSeverity min_sev) {
>-  dbg_sev_ = min_sev;
>-  CritScope cs(&logCriticalScope());
>+  g_dbg_sev = min_sev;
>+  CritScope cs(&g_log_crit);
>   UpdateMinLogSeverity();
> }
> 
>@@ -266,9 +287,9 @@ void LogMessage::SetLogToStderr(bool log_to_stderr) {
> }
> 
> int LogMessage::GetLogToStream(LogSink* stream) {
>-  CritScope cs(&logCriticalScope());
>+  CritScope cs(&g_log_crit);
>   LoggingSeverity sev = LS_NONE;
>-  for (auto& kv : streamList()) {
>+  for (auto& kv : streams_) {
>     if (!stream || stream == kv.first) {
>       sev = std::min(sev, kv.second);
>     }
>@@ -277,17 +298,16 @@ int LogMessage::GetLogToStream(LogSink* stream) {
> }
> 
> void LogMessage::AddLogToStream(LogSink* stream, LoggingSeverity min_sev) {
>-  CritScope cs(&logCriticalScope());
>-  streamList().push_back(std::make_pair(stream, min_sev));
>+  CritScope cs(&g_log_crit);
>+  streams_.push_back(std::make_pair(stream, min_sev));
>   UpdateMinLogSeverity();
> }
> 
> void LogMessage::RemoveLogToStream(LogSink* stream) {
>-  CritScope cs(&logCriticalScope());
>-  auto& streams = streamList();
>-  for (auto it = streams.begin(); it != streams.end(); ++it) {
>+  CritScope cs(&g_log_crit);
>+  for (StreamList::iterator it = streams_.begin(); it != streams_.end(); ++it) {
>     if (stream == it->first) {
>-      streams.erase(it);
>+      streams_.erase(it);
>       break;
>     }
>   }
>@@ -311,7 +331,7 @@ void LogMessage::ConfigureLogging(const char* params) {
>     } else if (token == "thread") {
>       LogThreads();
> 
>-    // Logging levels
>+      // Logging levels
>     } else if (token == "sensitive") {
>       current_level = LS_SENSITIVE;
>     } else if (token == "verbose") {
>@@ -325,7 +345,7 @@ void LogMessage::ConfigureLogging(const char* params) {
>     } else if (token == "none") {
>       current_level = LS_NONE;
> 
>-    // Logging targets
>+      // Logging targets
>     } else if (token == "debug") {
>       debug_level = current_level;
>     }
>@@ -337,19 +357,8 @@ void LogMessage::ConfigureLogging(const char* params) {
>     // from the command line, we'll see the output there.  Otherwise, create
>     // our own console window.
>     // Note: These methods fail if a console already exists, which is fine.
>-    bool success = false;
>-    typedef BOOL (WINAPI* PFN_AttachConsole)(DWORD);
>-    if (HINSTANCE kernel32 = ::LoadLibrary(L"kernel32.dll")) {
>-      // AttachConsole is defined on WinXP+.
>-      if (PFN_AttachConsole attach_console = reinterpret_cast<PFN_AttachConsole>
>-            (::GetProcAddress(kernel32, "AttachConsole"))) {
>-        success = (FALSE != attach_console(ATTACH_PARENT_PROCESS));
>-      }
>-      ::FreeLibrary(kernel32);
>-    }
>-    if (!success) {
>+    if (!AttachConsole(ATTACH_PARENT_PROCESS))
>       ::AllocConsole();
>-    }
>   }
> #endif  // WEBRTC_WIN
> 
>@@ -357,25 +366,30 @@ void LogMessage::ConfigureLogging(const char* params) {
> }
> 
> void LogMessage::UpdateMinLogSeverity()
>-    RTC_EXCLUSIVE_LOCKS_REQUIRED(logCriticalScope()) {
>-  LoggingSeverity min_sev = dbg_sev_;
>-  for (auto& kv : streamList()) {
>-    min_sev = std::min(dbg_sev_, kv.second);
>+    RTC_EXCLUSIVE_LOCKS_REQUIRED(g_log_crit) {
>+  LoggingSeverity min_sev = g_dbg_sev;
>+  for (const auto& kv : streams_) {
>+    const LoggingSeverity sev = kv.second;
>+    min_sev = std::min(min_sev, sev);
>   }
>-  min_sev_ = min_sev;
>+  g_min_sev = min_sev;
> }
> 
>+#if defined(WEBRTC_ANDROID)
> void LogMessage::OutputToDebug(const std::string& str,
>                                LoggingSeverity severity,
>-                               const std::string& tag) {
>+                               const char* tag) {
>+#else
>+void LogMessage::OutputToDebug(const std::string& str,
>+                               LoggingSeverity severity) {
>+#endif
>   bool log_to_stderr = log_to_stderr_;
> #if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS) && defined(NDEBUG)
>   // On the Mac, all stderr output goes to the Console log and causes clutter.
>   // So in opt builds, don't log to stderr unless the user specifically sets
>   // a preference to do so.
>-  CFStringRef key = CFStringCreateWithCString(kCFAllocatorDefault,
>-                                              "logToStdErr",
>-                                              kCFStringEncodingUTF8);
>+  CFStringRef key = CFStringCreateWithCString(
>+      kCFAllocatorDefault, "logToStdErr", kCFStringEncodingUTF8);
>   CFStringRef domain = CFBundleGetIdentifier(CFBundleGetMainBundle());
>   if (key != nullptr && domain != nullptr) {
>     Boolean exists_and_is_valid;
>@@ -388,7 +402,8 @@ void LogMessage::OutputToDebug(const std::string& str,
>   if (key != nullptr) {
>     CFRelease(key);
>   }
>-#endif
>+#endif  // defined(WEBRTC_MAC) && !defined(WEBRTC_IOS) && defined(NDEBUG)
>+
> #if defined(WEBRTC_WIN)
>   // Always log to the debugger.
>   // Perhaps stderr should be controlled by a preference, as on Mac?
>@@ -403,6 +418,7 @@ void LogMessage::OutputToDebug(const std::string& str,
>     }
>   }
> #endif  // WEBRTC_WIN
>+
> #if defined(WEBRTC_ANDROID)
>   // Android's logging facility uses severity to log messages but we
>   // need to map libjingle's severity levels to Android ones first.
>@@ -411,7 +427,7 @@ void LogMessage::OutputToDebug(const std::string& str,
>   int prio;
>   switch (severity) {
>     case LS_SENSITIVE:
>-      __android_log_write(ANDROID_LOG_INFO, tag.c_str(), "SENSITIVE");
>+      __android_log_write(ANDROID_LOG_INFO, tag, "SENSITIVE");
>       if (log_to_stderr) {
>         fprintf(stderr, "SENSITIVE");
>         fflush(stderr);
>@@ -438,15 +454,14 @@ void LogMessage::OutputToDebug(const std::string& str,
>   int idx = 0;
>   const int max_lines = size / kMaxLogLineSize + 1;
>   if (max_lines == 1) {
>-    __android_log_print(prio, tag.c_str(), "%.*s", size, str.c_str());
>+    __android_log_print(prio, tag, "%.*s", size, str.c_str());
>   } else {
>     while (size > 0) {
>       const int len = std::min(size, kMaxLogLineSize);
>       // Use the size of the string in the format (str may have \0 in the
>       // middle).
>-      __android_log_print(prio, tag.c_str(), "[%d/%d] %.*s",
>-                          line + 1, max_lines,
>-                          len, str.c_str() + idx);
>+      __android_log_print(prio, tag, "[%d/%d] %.*s", line + 1, max_lines, len,
>+                          str.c_str() + idx);
>       idx += len;
>       size -= len;
>       ++line;
>@@ -459,6 +474,107 @@ void LogMessage::OutputToDebug(const std::string& str,
>   }
> }
> 
>-//////////////////////////////////////////////////////////////////////
>+// static
>+bool LogMessage::IsNoop(LoggingSeverity severity) {
>+  if (severity >= g_dbg_sev)
>+    return false;
>+
>+  // TODO(tommi): We're grabbing this lock for every LogMessage instance that
>+  // is going to be logged. This introduces unnecessary synchronization for
>+  // a feature that's mostly used for testing.
>+  CritScope cs(&g_log_crit);
>+  return streams_.size() == 0;
>+}
>+
>+void LogMessage::FinishPrintStream() {
>+  if (is_noop_)
>+    return;
>+  if (!extra_.empty())
>+    print_stream_ << " : " << extra_;
>+  print_stream_ << std::endl;
>+}
>+
>+namespace webrtc_logging_impl {
>+
>+void Log(const LogArgType* fmt, ...) {
>+  va_list args;
>+  va_start(args, fmt);
>+
>+  LogMetadataErr meta;
>+  const char* tag = nullptr;
>+  switch (*fmt) {
>+    case LogArgType::kLogMetadata: {
>+      meta = {va_arg(args, LogMetadata), ERRCTX_NONE, 0};
>+      break;
>+    }
>+    case LogArgType::kLogMetadataErr: {
>+      meta = va_arg(args, LogMetadataErr);
>+      break;
>+    }
>+#ifdef WEBRTC_ANDROID
>+    case LogArgType::kLogMetadataTag: {
>+      const LogMetadataTag tag_meta = va_arg(args, LogMetadataTag);
>+      meta = {{nullptr, 0, tag_meta.severity}, ERRCTX_NONE, 0};
>+      tag = tag_meta.tag;
>+      break;
>+    }
>+#endif
>+    default: {
>+      RTC_NOTREACHED();
>+      va_end(args);
>+      return;
>+    }
>+  }
>+  LogMessage log_message(meta.meta.File(), meta.meta.Line(),
>+                         meta.meta.Severity(), meta.err_ctx, meta.err);
>+  if (tag) {
>+    log_message.AddTag(tag);
>+  }
>+
>+  for (++fmt; *fmt != LogArgType::kEnd; ++fmt) {
>+    switch (*fmt) {
>+      case LogArgType::kInt:
>+        log_message.stream() << va_arg(args, int);
>+        break;
>+      case LogArgType::kLong:
>+        log_message.stream() << va_arg(args, long);
>+        break;
>+      case LogArgType::kLongLong:
>+        log_message.stream() << va_arg(args, long long);
>+        break;
>+      case LogArgType::kUInt:
>+        log_message.stream() << va_arg(args, unsigned);
>+        break;
>+      case LogArgType::kULong:
>+        log_message.stream() << va_arg(args, unsigned long);
>+        break;
>+      case LogArgType::kULongLong:
>+        log_message.stream() << va_arg(args, unsigned long long);
>+        break;
>+      case LogArgType::kDouble:
>+        log_message.stream() << va_arg(args, double);
>+        break;
>+      case LogArgType::kLongDouble:
>+        log_message.stream() << va_arg(args, long double);
>+        break;
>+      case LogArgType::kCharP:
>+        log_message.stream() << va_arg(args, const char*);
>+        break;
>+      case LogArgType::kStdString:
>+        log_message.stream() << *va_arg(args, const std::string*);
>+        break;
>+      case LogArgType::kVoidP:
>+        log_message.stream() << va_arg(args, const void*);
>+        break;
>+      default:
>+        RTC_NOTREACHED();
>+        va_end(args);
>+        return;
>+    }
>+  }
>+
>+  va_end(args);
>+}
> 
>+}  // namespace webrtc_logging_impl
> }  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging.h
>index 8a012080135..b5af95928b5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging.h
>@@ -26,9 +26,8 @@
> // RTC_LOG_F(sev) Like RTC_LOG(), but includes the name of the current function.
> // RTC_LOG_T(sev) Like RTC_LOG(), but includes the this pointer.
> // RTC_LOG_T_F(sev) Like RTC_LOG_F(), but includes the this pointer.
>-// RTC_LOG_GLE(M)(sev [, mod]) attempt to add a string description of the
>-//     HRESULT returned by GetLastError.  The "M" variant allows searching of a
>-//     DLL's string table for the error description.
>+// RTC_LOG_GLE(sev [, mod]) attempt to add a string description of the
>+//     HRESULT returned by GetLastError.
> // RTC_LOG_ERRNO(sev) attempts to add a string description of an errno-derived
> //     error. errno and associated facilities exist on both Windows and POSIX,
> //     but on Windows they only apply to the C/C++ runtime.
>@@ -41,7 +40,6 @@
> // RTC_LOG_CHECK_LEVEL(sev) (and RTC_LOG_CHECK_LEVEL_V(sev)) can be used as a
> //     test before performing expensive or sensitive operations whose sole
> //     purpose is to output logging data at the desired level.
>-// Lastly, RTC_PLOG(sev, err) is an alias for RTC_LOG_ERR_EX.
> 
> #ifndef RTC_BASE_LOGGING_H_
> #define RTC_BASE_LOGGING_H_
>@@ -57,8 +55,9 @@
> #include <CoreServices/CoreServices.h>
> #endif
> 
>-#include "rtc_base/basictypes.h"
> #include "rtc_base/constructormagic.h"
>+#include "rtc_base/deprecation.h"
>+#include "rtc_base/system/inline.h"
> #include "rtc_base/thread_annotations.h"
> 
> #if !defined(NDEBUG) || defined(DLOG_ALWAYS_ON)
>@@ -69,29 +68,6 @@
> 
> namespace rtc {
> 
>-///////////////////////////////////////////////////////////////////////////////
>-// ConstantLabel can be used to easily generate string names from constant
>-// values.  This can be useful for logging descriptive names of error messages.
>-// Usage:
>-//   const ConstantLabel LIBRARY_ERRORS[] = {
>-//     KLABEL(SOME_ERROR),
>-//     KLABEL(SOME_OTHER_ERROR),
>-//     ...
>-//     LASTLABEL
>-//   }
>-//
>-//   int err = LibraryFunc();
>-//   LOG(LS_ERROR) << "LibraryFunc returned: "
>-//                 << ErrorName(err, LIBRARY_ERRORS);
>-
>-struct ConstantLabel { int value; const char * label; };
>-#define KLABEL(x) { x, #x }
>-#define TLABEL(x, y) { x, y }
>-#define LASTLABEL { 0, 0 }
>-
>-const char* FindLabel(int value, const ConstantLabel entries[]);
>-std::string ErrorName(int err, const ConstantLabel* err_table);
>-
> #if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
> // Returns a UTF8 description from an OS X Status error.
> std::string DescriptionFromOSStatus(OSStatus err);
>@@ -140,18 +116,288 @@ class LogSink {
>  public:
>   LogSink() {}
>   virtual ~LogSink() {}
>+  virtual void OnLogMessage(const std::string& msg,
>+                            LoggingSeverity severity,
>+                            const char* tag);
>   virtual void OnLogMessage(const std::string& message) = 0;
> };
> 
>+namespace webrtc_logging_impl {
>+
>+class LogMetadata {
>+ public:
>+  LogMetadata(const char* file, int line, LoggingSeverity severity)
>+      : file_(file),
>+        line_and_sev_(static_cast<uint32_t>(line) << 3 | severity) {}
>+  LogMetadata() = default;
>+
>+  const char* File() const { return file_; }
>+  int Line() const { return line_and_sev_ >> 3; }
>+  LoggingSeverity Severity() const {
>+    return static_cast<LoggingSeverity>(line_and_sev_ & 0x7);
>+  }
>+
>+ private:
>+  const char* file_;
>+
>+  // Line number and severity, the former in the most significant 29 bits, the
>+  // latter in the least significant 3 bits. (This is an optimization; since
>+  // both numbers are usually compile-time constants, this way we can load them
>+  // both with a single instruction.)
>+  uint32_t line_and_sev_;
>+};
>+static_assert(std::is_trivial<LogMetadata>::value, "");
>+
>+struct LogMetadataErr {
>+  LogMetadata meta;
>+  LogErrorContext err_ctx;
>+  int err;
>+};
>+
>+#ifdef WEBRTC_ANDROID
>+struct LogMetadataTag {
>+  LoggingSeverity severity;
>+  const char* tag;
>+};
>+#endif
>+
>+enum class LogArgType : int8_t {
>+  kEnd = 0,
>+  kInt,
>+  kLong,
>+  kLongLong,
>+  kUInt,
>+  kULong,
>+  kULongLong,
>+  kDouble,
>+  kLongDouble,
>+  kCharP,
>+  kStdString,
>+  // TODO(kwiberg): Add absl::StringView.
>+  kVoidP,
>+  kLogMetadata,
>+  kLogMetadataErr,
>+#ifdef WEBRTC_ANDROID
>+  kLogMetadataTag,
>+#endif
>+};
>+
>+// Wrapper for log arguments. Only ever make values of this type with the
>+// MakeVal() functions.
>+template <LogArgType N, typename T>
>+struct Val {
>+  static constexpr LogArgType Type() { return N; }
>+  T GetVal() const { return val; }
>+  T val;
>+};
>+
>+// TODO(bugs.webrtc.org/9278): Get rid of this specialization when callers
>+// don't need it anymore. No in-tree caller does, but some external callers
>+// still do.
>+template <>
>+struct Val<LogArgType::kStdString, std::string> {
>+  static constexpr LogArgType Type() { return LogArgType::kStdString; }
>+  const std::string* GetVal() const { return &val; }
>+  std::string val;
>+};
>+
>+inline Val<LogArgType::kInt, int> MakeVal(int x) {
>+  return {x};
>+}
>+inline Val<LogArgType::kLong, long> MakeVal(long x) {
>+  return {x};
>+}
>+inline Val<LogArgType::kLongLong, long long> MakeVal(long long x) {
>+  return {x};
>+}
>+inline Val<LogArgType::kUInt, unsigned int> MakeVal(unsigned int x) {
>+  return {x};
>+}
>+inline Val<LogArgType::kULong, unsigned long> MakeVal(unsigned long x) {
>+  return {x};
>+}
>+inline Val<LogArgType::kULongLong, unsigned long long> MakeVal(
>+    unsigned long long x) {
>+  return {x};
>+}
>+
>+inline Val<LogArgType::kDouble, double> MakeVal(double x) {
>+  return {x};
>+}
>+inline Val<LogArgType::kLongDouble, long double> MakeVal(long double x) {
>+  return {x};
>+}
>+
>+inline Val<LogArgType::kCharP, const char*> MakeVal(const char* x) {
>+  return {x};
>+}
>+inline Val<LogArgType::kStdString, const std::string*> MakeVal(
>+    const std::string& x) {
>+  return {&x};
>+}
>+// TODO(kwiberg): Add absl::string_view
>+
>+inline Val<LogArgType::kVoidP, const void*> MakeVal(const void* x) {
>+  return {x};
>+}
>+
>+inline Val<LogArgType::kLogMetadata, LogMetadata> MakeVal(
>+    const LogMetadata& x) {
>+  return {x};
>+}
>+inline Val<LogArgType::kLogMetadataErr, LogMetadataErr> MakeVal(
>+    const LogMetadataErr& x) {
>+  return {x};
>+}
>+
>+#ifdef WEBRTC_ANDROID
>+inline Val<LogArgType::kLogMetadataTag, LogMetadataTag> MakeVal(
>+    const LogMetadataTag& x) {
>+  return {x};
>+}
>+#endif
>+
>+// Handle arbitrary types other than the above by falling back to stringstream.
>+// TODO(bugs.webrtc.org/9278): Get rid of this overload when callers don't need
>+// it anymore. No in-tree caller does, but some external callers still do.
>+template <
>+    typename T,
>+    typename T1 =
>+        typename std::remove_cv<typename std::remove_reference<T>::type>::type,
>+    typename std::enable_if<
>+        std::is_class<T1>::value && !std::is_same<T1, std::string>::value &&
>+        !std::is_same<T1, LogMetadata>::value &&
>+#ifdef WEBRTC_ANDROID
>+        !std::is_same<T1, LogMetadataTag>::value &&
>+#endif
>+        !std::is_same<T1, LogMetadataErr>::value>::type* = nullptr>
>+Val<LogArgType::kStdString, std::string> MakeVal(const T& x) {
>+  std::ostringstream os;  // no-presubmit-check TODO(webrtc:8982)
>+  os << x;
>+  return {os.str()};
>+}
>+
>+void Log(const LogArgType* fmt, ...);
>+
>+// Ephemeral type that represents the result of the logging << operator.
>+template <typename... Ts>
>+class LogStreamer;
>+
>+// Base case: Before the first << argument.
>+template <>
>+class LogStreamer<> final {
>+ public:
>+  template <
>+      typename U,
>+      typename std::enable_if<std::is_arithmetic<U>::value>::type* = nullptr>
>+  RTC_FORCE_INLINE LogStreamer<decltype(MakeVal(std::declval<U>()))> operator<<(
>+      U arg) const {
>+    return LogStreamer<decltype(MakeVal(std::declval<U>()))>(MakeVal(arg),
>+                                                             this);
>+  }
>+
>+  template <
>+      typename U,
>+      typename std::enable_if<!std::is_arithmetic<U>::value>::type* = nullptr>
>+  RTC_FORCE_INLINE LogStreamer<decltype(MakeVal(std::declval<U>()))> operator<<(
>+      const U& arg) const {
>+    return LogStreamer<decltype(MakeVal(std::declval<U>()))>(MakeVal(arg),
>+                                                             this);
>+  }
>+
>+  template <typename... Us>
>+  RTC_FORCE_INLINE static void Call(const Us&... args) {
>+    static constexpr LogArgType t[] = {Us::Type()..., LogArgType::kEnd};
>+    Log(t, args.GetVal()...);
>+  }
>+};
>+
>+// Inductive case: We've already seen at least one << argument. The most recent
>+// one had type `T`, and the earlier ones had types `Ts`.
>+template <typename T, typename... Ts>
>+class LogStreamer<T, Ts...> final {
>+ public:
>+  RTC_FORCE_INLINE LogStreamer(T arg, const LogStreamer<Ts...>* prior)
>+      : arg_(arg), prior_(prior) {}
>+
>+  template <
>+      typename U,
>+      typename std::enable_if<std::is_arithmetic<U>::value>::type* = nullptr>
>+  RTC_FORCE_INLINE LogStreamer<decltype(MakeVal(std::declval<U>())), T, Ts...>
>+  operator<<(U arg) const {
>+    return LogStreamer<decltype(MakeVal(std::declval<U>())), T, Ts...>(
>+        MakeVal(arg), this);
>+  }
>+
>+  template <
>+      typename U,
>+      typename std::enable_if<!std::is_arithmetic<U>::value>::type* = nullptr>
>+  RTC_FORCE_INLINE LogStreamer<decltype(MakeVal(std::declval<U>())), T, Ts...>
>+  operator<<(const U& arg) const {
>+    return LogStreamer<decltype(MakeVal(std::declval<U>())), T, Ts...>(
>+        MakeVal(arg), this);
>+  }
>+
>+  template <typename... Us>
>+  RTC_FORCE_INLINE void Call(const Us&... args) const {
>+    prior_->Call(arg_, args...);
>+  }
>+
>+ private:
>+  // The most recent argument.
>+  T arg_;
>+
>+  // Earlier arguments.
>+  const LogStreamer<Ts...>* prior_;
>+};
>+
>+class LogCall final {
>+ public:
>+  // This can be any binary operator with precedence lower than <<.
>+  template <typename... Ts>
>+  RTC_FORCE_INLINE void operator&(const LogStreamer<Ts...>& streamer) {
>+    streamer.Call();
>+  }
>+};
>+
>+// TODO(bugs.webrtc.org/9278): Remove this once it's no longer used.
>+struct LogMessageVoidify {
>+  void operator&(std::ostream&) {}  // no-presubmit-check TODO(webrtc:8982)
>+};
>+
>+}  // namespace webrtc_logging_impl
>+
>+// Direct use of this class is deprecated; please use the logging macros
>+// instead.
>+// TODO(bugs.webrtc.org/9278): Move this class to an unnamed namespace in the
>+// .cc file.
> class LogMessage {
>  public:
>+  LogMessage(const char* file, int line, LoggingSeverity sev);
>+
>+  // Same as the above, but using a compile-time constant for the logging
>+  // severity. This saves space at the call site, since passing an empty struct
>+  // is generally the same as not passing an argument at all.
>+  template <LoggingSeverity S>
>+  RTC_NO_INLINE LogMessage(const char* file,
>+                           int line,
>+                           std::integral_constant<LoggingSeverity, S>)
>+      : LogMessage(file, line, S) {}
>+
>   LogMessage(const char* file,
>              int line,
>              LoggingSeverity sev,
>-             LogErrorContext err_ctx = ERRCTX_NONE,
>-             int err = 0,
>-             const char* module = nullptr);
>+             LogErrorContext err_ctx,
>+             int err);
>+
>+#if defined(WEBRTC_ANDROID)
>+  LogMessage(const char* file, int line, LoggingSeverity sev, const char* tag);
>+#endif
> 
>+  // DEPRECATED - DO NOT USE - PLEASE USE THE MACROS INSTEAD OF THE CLASS.
>+  // Android code should use the 'const char*' version since tags are static
>+  // and we want to avoid allocating a std::string copy per log line.
>+  RTC_DEPRECATED
>   LogMessage(const char* file,
>              int line,
>              LoggingSeverity sev,
>@@ -159,8 +405,20 @@ class LogMessage {
> 
>   ~LogMessage();
> 
>-  static inline bool Loggable(LoggingSeverity sev) { return (sev >= min_sev_); }
>-  std::ostream& stream() { return print_stream_; }
>+  void AddTag(const char* tag);
>+
>+  static bool Loggable(LoggingSeverity sev);
>+
>+  // Same as the above, but using a template argument instead of a function
>+  // argument. (When the logging severity is statically known, passing it as a
>+  // template argument instead of as a function argument saves space at the
>+  // call site.)
>+  template <LoggingSeverity S>
>+  RTC_NO_INLINE static bool Loggable() {
>+    return Loggable(S);
>+  }
>+
>+  std::ostream& stream();
> 
>   // Returns the time at which this function was called for the first time.
>   // The time will be used as the logging start time.
>@@ -182,7 +440,7 @@ class LogMessage {
>   // These are the available logging channels
>   //  Debug: Debug console on Windows, otherwise stderr
>   static void LogToDebug(LoggingSeverity min_sev);
>-  static LoggingSeverity GetLogToDebug() { return dbg_sev_; }
>+  static LoggingSeverity GetLogToDebug();
> 
>   // Sets whether logs will be directed to stderr in debug mode.
>   static void SetLogToStderr(bool log_to_stderr);
>@@ -200,20 +458,38 @@ class LogMessage {
> 
>   // Testing against MinLogSeverity allows code to avoid potentially expensive
>   // logging operations by pre-checking the logging level.
>-  static int GetMinLogSeverity() { return min_sev_; }
>+  static int GetMinLogSeverity();
> 
>   // Parses the provided parameter stream to configure the options above.
>   // Useful for configuring logging from the command line.
>   static void ConfigureLogging(const char* params);
> 
>  private:
>+  friend class LogMessageForTesting;
>+  typedef std::pair<LogSink*, LoggingSeverity> StreamAndSeverity;
>+  typedef std::list<StreamAndSeverity> StreamList;
>+
>   // Updates min_sev_ appropriately when debug sinks change.
>   static void UpdateMinLogSeverity();
> 
>-  // These write out the actual log messages.
>+// These write out the actual log messages.
>+#if defined(WEBRTC_ANDROID)
>   static void OutputToDebug(const std::string& msg,
>                             LoggingSeverity severity,
>-                            const std::string& tag);
>+                            const char* tag);
>+#else
>+  static void OutputToDebug(const std::string& msg, LoggingSeverity severity);
>+#endif
>+
>+  // Checks the current global debug severity and if the |streams_| collection
>+  // is empty. If |severity| is smaller than the global severity and if the
>+  // |streams_| collection is empty, the LogMessage will be considered a noop
>+  // LogMessage.
>+  static bool IsNoop(LoggingSeverity severity);
>+
>+  // Called from the dtor (or from a test) to append optional extra error
>+  // information to the log stream and a newline character.
>+  void FinishPrintStream();
> 
>   // The ostream that buffers the formatted message before output
>   std::ostringstream print_stream_;
>@@ -221,19 +497,19 @@ class LogMessage {
>   // The severity level of this message
>   LoggingSeverity severity_;
> 
>-  // The Android debug output tag.
>-  std::string tag_;
>+#if defined(WEBRTC_ANDROID)
>+  // The default Android debug output tag.
>+  const char* tag_ = "libjingle";
>+#endif
> 
>   // String data generated in the constructor, that should be appended to
>   // the message before output.
>   std::string extra_;
> 
>-  // dbg_sev_ is the thresholds for those output targets
>-  // min_sev_ is the minimum (most verbose) of those levels, and is used
>-  //  as a short-circuit in the logging macros to identify messages that won't
>-  //  be logged.
>-  // ctx_sev_ is the minimum level at which file context is displayed
>-  static LoggingSeverity min_sev_, dbg_sev_, ctx_sev_;
>+  const bool is_noop_;
>+
>+  // The output streams and their associated severities
>+  static StreamList streams_;
> 
>   // Flags for formatting options
>   static bool thread_, timestamp_;
>@@ -248,97 +524,98 @@ class LogMessage {
> // Logging Helpers
> //////////////////////////////////////////////////////////////////////
> 
>-// The following non-obvious technique for implementation of a
>-// conditional log stream was stolen from google3/base/logging.h.
>-
>-// This class is used to explicitly ignore values in the conditional
>-// logging macros.  This avoids compiler warnings like "value computed
>-// is not used" and "statement has no effect".
>-
>-class LogMessageVoidify {
>- public:
>-  LogMessageVoidify() { }
>-  // This has to be an operator with a precedence lower than << but
>-  // higher than ?:
>-  void operator&(std::ostream&) { }
>-};
>-
>+// DEPRECATED.
>+// TODO(bugs.webrtc.org/9278): Remove once there are no more users.
> #define RTC_LOG_SEVERITY_PRECONDITION(sev) \
>-  !(rtc::LogMessage::Loggable(sev)) \
>-    ? (void) 0 \
>-    : rtc::LogMessageVoidify() &
>-
>-#define RTC_LOG(sev) \
>-  RTC_LOG_SEVERITY_PRECONDITION(rtc::sev) \
>-    rtc::LogMessage(__FILE__, __LINE__, rtc::sev).stream()
>-
>-// The _V version is for when a variable is passed in.  It doesn't do the
>-// namespace concatination.
>-#define RTC_LOG_V(sev) \
>-  RTC_LOG_SEVERITY_PRECONDITION(sev) \
>-    rtc::LogMessage(__FILE__, __LINE__, sev).stream()
>+  !(rtc::LogMessage::Loggable(sev))        \
>+      ? static_cast<void>(0)               \
>+      : rtc::webrtc_logging_impl::LogMessageVoidify()&
>+
>+#define RTC_LOG(sev)                                                   \
>+  for (bool do_log = rtc::LogMessage::Loggable<rtc::sev>(); do_log;    \
>+       do_log = false)                                                 \
>+  rtc::webrtc_logging_impl::LogCall() &                                \
>+      rtc::webrtc_logging_impl::LogStreamer<>()                        \
>+          << rtc::webrtc_logging_impl::LogMetadata(__FILE__, __LINE__, \
>+                                                   rtc::sev)
>+
>+// The _V version is for when a variable is passed in.
>+#define RTC_LOG_V(sev)                                                       \
>+  for (bool do_log = rtc::LogMessage::Loggable(sev); do_log; do_log = false) \
>+  rtc::webrtc_logging_impl::LogCall() &                                      \
>+      rtc::webrtc_logging_impl::LogStreamer<>()                              \
>+          << rtc::webrtc_logging_impl::LogMetadata(__FILE__, __LINE__, sev)
> 
> // The _F version prefixes the message with the current function name.
> #if (defined(__GNUC__) && !defined(NDEBUG)) || defined(WANT_PRETTY_LOG_F)
> #define RTC_LOG_F(sev) RTC_LOG(sev) << __PRETTY_FUNCTION__ << ": "
>-#define RTC_LOG_T_F(sev) RTC_LOG(sev) << this << ": " \
>-  << __PRETTY_FUNCTION__ << ": "
>+#define RTC_LOG_T_F(sev) \
>+  RTC_LOG(sev) << this << ": " << __PRETTY_FUNCTION__ << ": "
> #else
> #define RTC_LOG_F(sev) RTC_LOG(sev) << __FUNCTION__ << ": "
> #define RTC_LOG_T_F(sev) RTC_LOG(sev) << this << ": " << __FUNCTION__ << ": "
> #endif
> 
>-#define RTC_LOG_CHECK_LEVEL(sev) \
>-  rtc::LogCheckLevel(rtc::sev)
>-#define RTC_LOG_CHECK_LEVEL_V(sev) \
>-  rtc::LogCheckLevel(sev)
>+#define RTC_LOG_CHECK_LEVEL(sev) rtc::LogCheckLevel(rtc::sev)
>+#define RTC_LOG_CHECK_LEVEL_V(sev) rtc::LogCheckLevel(sev)
> 
> inline bool LogCheckLevel(LoggingSeverity sev) {
>   return (LogMessage::GetMinLogSeverity() <= sev);
> }
> 
>-#define RTC_LOG_E(sev, ctx, err, ...) \
>-  RTC_LOG_SEVERITY_PRECONDITION(rtc::sev) \
>-    rtc::LogMessage(__FILE__, __LINE__, rtc::sev, \
>-                    rtc::ERRCTX_ ## ctx, err , ##__VA_ARGS__)   \
>-        .stream()
>+#define RTC_LOG_E(sev, ctx, err)                                    \
>+  for (bool do_log = rtc::LogMessage::Loggable<rtc::sev>(); do_log; \
>+       do_log = false)                                              \
>+    rtc::webrtc_logging_impl::LogCall() &                           \
>+        rtc::webrtc_logging_impl::LogStreamer<>()                   \
>+            << rtc::webrtc_logging_impl::LogMetadataErr {           \
>+      {__FILE__, __LINE__, rtc::sev}, rtc::ERRCTX_##ctx, (err)      \
>+    }
> 
> #define RTC_LOG_T(sev) RTC_LOG(sev) << this << ": "
> 
>-#define RTC_LOG_ERRNO_EX(sev, err) \
>-  RTC_LOG_E(sev, ERRNO, err)
>-#define RTC_LOG_ERRNO(sev) \
>-  RTC_LOG_ERRNO_EX(sev, errno)
>+#define RTC_LOG_ERRNO_EX(sev, err) RTC_LOG_E(sev, ERRNO, err)
>+#define RTC_LOG_ERRNO(sev) RTC_LOG_ERRNO_EX(sev, errno)
> 
> #if defined(WEBRTC_WIN)
>-#define RTC_LOG_GLE_EX(sev, err) \
>-  RTC_LOG_E(sev, HRESULT, err)
>-#define RTC_LOG_GLE(sev) \
>-  RTC_LOG_GLE_EX(sev, GetLastError())
>-#define RTC_LOG_GLEM(sev, mod) \
>-  RTC_LOG_E(sev, HRESULT, GetLastError(), mod)
>-#define RTC_LOG_ERR_EX(sev, err) \
>-  RTC_LOG_GLE_EX(sev, err)
>-#define RTC_LOG_ERR(sev) \
>-  RTC_LOG_GLE(sev)
>+#define RTC_LOG_GLE_EX(sev, err) RTC_LOG_E(sev, HRESULT, err)
>+#define RTC_LOG_GLE(sev) RTC_LOG_GLE_EX(sev, static_cast<int>(GetLastError()))
>+#define RTC_LOG_ERR_EX(sev, err) RTC_LOG_GLE_EX(sev, err)
>+#define RTC_LOG_ERR(sev) RTC_LOG_GLE(sev)
> #elif defined(__native_client__) && __native_client__
>-#define RTC_LOG_ERR_EX(sev, err) \
>-  RTC_LOG(sev)
>-#define RTC_LOG_ERR(sev) \
>-  RTC_LOG(sev)
>+#define RTC_LOG_ERR_EX(sev, err) RTC_LOG(sev)
>+#define RTC_LOG_ERR(sev) RTC_LOG(sev)
> #elif defined(WEBRTC_POSIX)
>-#define RTC_LOG_ERR_EX(sev, err) \
>-  RTC_LOG_ERRNO_EX(sev, err)
>-#define RTC_LOG_ERR(sev) \
>-  RTC_LOG_ERRNO(sev)
>+#define RTC_LOG_ERR_EX(sev, err) RTC_LOG_ERRNO_EX(sev, err)
>+#define RTC_LOG_ERR(sev) RTC_LOG_ERRNO(sev)
> #endif  // WEBRTC_WIN
> 
>-#define RTC_LOG_TAG(sev, tag)        \
>-  RTC_LOG_SEVERITY_PRECONDITION(sev) \
>-  rtc::LogMessage(nullptr, 0, sev, tag).stream()
>+#ifdef WEBRTC_ANDROID
>+
>+namespace webrtc_logging_impl {
>+// TODO(kwiberg): Replace these with absl::string_view.
>+inline const char* AdaptString(const char* str) {
>+  return str;
>+}
>+inline const char* AdaptString(const std::string& str) {
>+  return str.c_str();
>+}
>+}  // namespace webrtc_logging_impl
> 
>-#define RTC_PLOG(sev, err) \
>-  RTC_LOG_ERR_EX(sev, err)
>+#define RTC_LOG_TAG(sev, tag)                                                \
>+  for (bool do_log = rtc::LogMessage::Loggable(sev); do_log; do_log = false) \
>+    rtc::webrtc_logging_impl::LogCall() &                                    \
>+        rtc::webrtc_logging_impl::LogStreamer<>()                            \
>+            << rtc::webrtc_logging_impl::LogMetadataTag {                    \
>+      sev, rtc::webrtc_logging_impl::AdaptString(tag)                        \
>+    }
>+
>+#else
>+
>+// DEPRECATED. This macro is only intended for Android.
>+#define RTC_LOG_TAG(sev, tag) RTC_LOG_V(sev)
>+
>+#endif
> 
> // The RTC_DLOG macros are equivalent to their RTC_LOG counterparts except that
> // they only generate code in debug builds.
>@@ -347,14 +624,12 @@ inline bool LogCheckLevel(LoggingSeverity sev) {
> #define RTC_DLOG_V(sev) RTC_LOG_V(sev)
> #define RTC_DLOG_F(sev) RTC_LOG_F(sev)
> #else
>-#define RTC_DLOG_EAT_STREAM_PARAMS(sev) \
>-  (true ? true : ((void)(rtc::sev), true)) \
>-      ? static_cast<void>(0)          \
>-      : rtc::LogMessageVoidify() &    \
>-        rtc::LogMessage(__FILE__, __LINE__, rtc::sev).stream()
>-#define RTC_DLOG(sev) RTC_DLOG_EAT_STREAM_PARAMS(sev)
>-#define RTC_DLOG_V(sev) RTC_DLOG_EAT_STREAM_PARAMS(sev)
>-#define RTC_DLOG_F(sev) RTC_DLOG_EAT_STREAM_PARAMS(sev)
>+#define RTC_DLOG_EAT_STREAM_PARAMS() \
>+  while (false)                      \
>+  rtc::webrtc_logging_impl::LogStreamer<>()
>+#define RTC_DLOG(sev) RTC_DLOG_EAT_STREAM_PARAMS()
>+#define RTC_DLOG_V(sev) RTC_DLOG_EAT_STREAM_PARAMS()
>+#define RTC_DLOG_F(sev) RTC_DLOG_EAT_STREAM_PARAMS()
> #endif
> 
> }  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging_unittest.cc
>index a2d35fdd3f2..7c411926440 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logging_unittest.cc
>@@ -8,32 +8,171 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "rtc_base/gunit.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/nullsocketserver.h"
>+#include "rtc_base/arraysize.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/event.h"
>+#include "rtc_base/gunit.h"
>+#include "rtc_base/platform_thread.h"
> #include "rtc_base/stream.h"
>-#include "rtc_base/thread.h"
> #include "test/testsupport/fileutils.h"
> 
> namespace rtc {
> 
>+namespace {
>+
>+class StringStream : public StreamInterface {
>+ public:
>+  explicit StringStream(std::string* str);
>+  explicit StringStream(const std::string& str);
>+
>+  StreamState GetState() const override;
>+  StreamResult Read(void* buffer,
>+                    size_t buffer_len,
>+                    size_t* read,
>+                    int* error) override;
>+  StreamResult Write(const void* data,
>+                     size_t data_len,
>+                     size_t* written,
>+                     int* error) override;
>+  void Close() override;
>+  bool SetPosition(size_t position) override;
>+  bool GetPosition(size_t* position) const override;
>+  bool GetSize(size_t* size) const override;
>+  bool GetAvailable(size_t* size) const override;
>+  bool ReserveSize(size_t size) override;
>+
>+ private:
>+  std::string& str_;
>+  size_t read_pos_;
>+  bool read_only_;
>+};
>+
>+StringStream::StringStream(std::string* str)
>+    : str_(*str), read_pos_(0), read_only_(false) {}
>+
>+StringStream::StringStream(const std::string& str)
>+    : str_(const_cast<std::string&>(str)), read_pos_(0), read_only_(true) {}
>+
>+StreamState StringStream::GetState() const {
>+  return SS_OPEN;
>+}
>+
>+StreamResult StringStream::Read(void* buffer,
>+                                size_t buffer_len,
>+                                size_t* read,
>+                                int* error) {
>+  size_t available = std::min(buffer_len, str_.size() - read_pos_);
>+  if (!available)
>+    return SR_EOS;
>+  memcpy(buffer, str_.data() + read_pos_, available);
>+  read_pos_ += available;
>+  if (read)
>+    *read = available;
>+  return SR_SUCCESS;
>+}
>+
>+StreamResult StringStream::Write(const void* data,
>+                                 size_t data_len,
>+                                 size_t* written,
>+                                 int* error) {
>+  if (read_only_) {
>+    if (error) {
>+      *error = -1;
>+    }
>+    return SR_ERROR;
>+  }
>+  str_.append(static_cast<const char*>(data),
>+              static_cast<const char*>(data) + data_len);
>+  if (written)
>+    *written = data_len;
>+  return SR_SUCCESS;
>+}
>+
>+void StringStream::Close() {}
>+
>+bool StringStream::SetPosition(size_t position) {
>+  if (position > str_.size())
>+    return false;
>+  read_pos_ = position;
>+  return true;
>+}
>+
>+bool StringStream::GetPosition(size_t* position) const {
>+  if (position)
>+    *position = read_pos_;
>+  return true;
>+}
>+
>+bool StringStream::GetSize(size_t* size) const {
>+  if (size)
>+    *size = str_.size();
>+  return true;
>+}
>+
>+bool StringStream::GetAvailable(size_t* size) const {
>+  if (size)
>+    *size = str_.size() - read_pos_;
>+  return true;
>+}
>+
>+bool StringStream::ReserveSize(size_t size) {
>+  if (read_only_)
>+    return false;
>+  str_.reserve(size);
>+  return true;
>+}
>+
>+}  // namespace
>+
> template <typename Base>
>-class LogSinkImpl
>-    : public LogSink,
>-      public Base {
>+class LogSinkImpl : public LogSink, public Base {
>  public:
>   LogSinkImpl() {}
> 
>-  template<typename P>
>+  template <typename P>
>   explicit LogSinkImpl(P* p) : Base(p) {}
> 
>  private:
>   void OnLogMessage(const std::string& message) override {
>-    static_cast<Base*>(this)->WriteAll(
>-        message.data(), message.size(), nullptr, nullptr);
>+    static_cast<Base*>(this)->WriteAll(message.data(), message.size(), nullptr,
>+                                       nullptr);
>   }
> };
> 
>+class LogMessageForTesting : public LogMessage {
>+ public:
>+  LogMessageForTesting(const char* file,
>+                       int line,
>+                       LoggingSeverity sev,
>+                       LogErrorContext err_ctx = ERRCTX_NONE,
>+                       int err = 0)
>+      : LogMessage(file, line, sev, err_ctx, err) {}
>+
>+  const std::string& get_extra() const { return extra_; }
>+  bool is_noop() const { return is_noop_; }
>+#if defined(WEBRTC_ANDROID)
>+  const char* get_tag() const { return tag_; }
>+#endif
>+
>+  // Returns the contents of the internal log stream.
>+  // Note that parts of the stream won't (as is) be available until *after* the
>+  // dtor of the parent class has run. So, as is, this only represents a
>+  // partially built stream.
>+  std::string GetPrintStream() {
>+    RTC_DCHECK(!is_finished_);
>+    is_finished_ = true;
>+    FinishPrintStream();
>+    std::string ret = print_stream_.str();
>+    // Just to make an error even more clear if the stream gets used after this.
>+    print_stream_.clear();
>+    return ret;
>+  }
>+
>+ private:
>+  bool is_finished_ = false;
>+};
>+
> // Test basic logging operation. We should get the INFO log but not the VERBOSE.
> // We should restore the correct global state at the end.
> TEST(LogTest, SingleStream) {
>@@ -55,6 +194,34 @@ TEST(LogTest, SingleStream) {
>   EXPECT_EQ(sev, LogMessage::GetLogToStream(nullptr));
> }
> 
>+#if GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
>+TEST(LogTest, Checks) {
>+  EXPECT_DEATH(RTC_FATAL() << "message",
>+               "\n\n#\n"
>+               "# Fatal error in: \\S+, line \\w+\n"
>+               "# last system error: \\w+\n"
>+               "# Check failed: FATAL\\(\\)\n"
>+               "# message");
>+
>+  int a = 1, b = 2;
>+  EXPECT_DEATH(RTC_CHECK_EQ(a, b) << 1 << 2u,
>+               "\n\n#\n"
>+               "# Fatal error in: \\S+, line \\w+\n"
>+               "# last system error: \\w+\n"
>+               "# Check failed: a == b \\(1 vs. 2\\)\n"
>+               "# 12");
>+  RTC_CHECK_EQ(5, 5);
>+
>+  RTC_CHECK(true) << "Shouldn't crash" << 1;
>+  EXPECT_DEATH(RTC_CHECK(false) << "Hi there!",
>+               "\n\n#\n"
>+               "# Fatal error in: \\S+, line \\w+\n"
>+               "# last system error: \\w+\n"
>+               "# Check failed: false\n"
>+               "# Hi there!");
>+}
>+#endif
>+
> // Test using multiple log streams. The INFO stream should get the INFO message,
> // the VERBOSE stream should get the INFO and the VERBOSE.
> // We should restore the correct global state at the end.
>@@ -84,23 +251,27 @@ TEST(LogTest, MultipleStreams) {
>   EXPECT_EQ(sev, LogMessage::GetLogToStream(nullptr));
> }
> 
>-// Ensure we don't crash when adding/removing streams while threads are going.
>-// We should restore the correct global state at the end.
>-class LogThread : public Thread {
>+class LogThread {
>  public:
>-  LogThread() : Thread(std::unique_ptr<SocketServer>(new NullSocketServer())) {}
>+  LogThread() : thread_(&ThreadEntry, this, "LogThread") {}
>+  ~LogThread() { thread_.Stop(); }
> 
>-  ~LogThread() override {
>-    Stop();
>-  }
>+  void Start() { thread_.Start(); }
> 
>  private:
>-  void Run() override {
>-    // LS_SENSITIVE to avoid cluttering up any real logging going on
>+  void Run() {
>+    // LS_SENSITIVE by default to avoid cluttering up any real logging going on.
>     RTC_LOG(LS_SENSITIVE) << "RTC_LOG";
>   }
>+
>+  static void ThreadEntry(void* p) { static_cast<LogThread*>(p)->Run(); }
>+
>+  PlatformThread thread_;
>+  Event event_{false, false};
> };
> 
>+// Ensure we don't crash when adding/removing streams while threads are going.
>+// We should restore the correct global state at the end.
> TEST(LogTest, MultipleThreads) {
>   int sev = LogMessage::GetLogToStream(nullptr);
> 
>@@ -109,7 +280,8 @@ TEST(LogTest, MultipleThreads) {
>   thread2.Start();
>   thread3.Start();
> 
>-  LogSinkImpl<NullStream> stream1, stream2, stream3;
>+  std::string s1, s2, s3;
>+  LogSinkImpl<StringStream> stream1(&s1), stream2(&s2), stream3(&s3);
>   for (int i = 0; i < 1000; ++i) {
>     LogMessage::AddLogToStream(&stream1, LS_INFO);
>     LogMessage::AddLogToStream(&stream2, LS_VERBOSE);
>@@ -122,42 +294,100 @@ TEST(LogTest, MultipleThreads) {
>   EXPECT_EQ(sev, LogMessage::GetLogToStream(nullptr));
> }
> 
>-
> TEST(LogTest, WallClockStartTime) {
>   uint32_t time = LogMessage::WallClockStartTime();
>   // Expect the time to be in a sensible range, e.g. > 2012-01-01.
>   EXPECT_GT(time, 1325376000u);
> }
> 
>-// Test the time required to write 1000 80-character logs to an unbuffered file.
>-#if defined (WEBRTC_ANDROID)
>-// Fails on Android: https://bugs.chromium.org/p/webrtc/issues/detail?id=4364.
>-#define MAYBE_Perf DISABLED_Perf
>+TEST(LogTest, CheckExtraErrorField) {
>+  LogMessageForTesting log_msg("some/path/myfile.cc", 100, LS_WARNING,
>+                               ERRCTX_ERRNO, 0xD);
>+  ASSERT_FALSE(log_msg.is_noop());
>+  log_msg.stream() << "This gets added at dtor time";
>+
>+  const std::string& extra = log_msg.get_extra();
>+  const size_t length_to_check = arraysize("[0x12345678]") - 1;
>+  ASSERT_GE(extra.length(), length_to_check);
>+  EXPECT_EQ(std::string("[0x0000000D]"), extra.substr(0, length_to_check));
>+}
>+
>+TEST(LogTest, CheckFilePathParsed) {
>+  LogMessageForTesting log_msg("some/path/myfile.cc", 100, LS_INFO);
>+  ASSERT_FALSE(log_msg.is_noop());
>+  log_msg.stream() << "<- Does this look right?";
>+
>+  const std::string stream = log_msg.GetPrintStream();
>+#if defined(WEBRTC_ANDROID)
>+  const char* tag = log_msg.get_tag();
>+  EXPECT_NE(nullptr, strstr(tag, "myfile.cc"));
>+  EXPECT_NE(std::string::npos, stream.find("100"));
> #else
>-#define MAYBE_Perf Perf
>+  EXPECT_NE(std::string::npos, stream.find("(myfile.cc:100)"));
> #endif
>+}
> 
>-TEST(LogTest, MAYBE_Perf) {
>-  std::string path =
>-      webrtc::test::TempFilename(webrtc::test::OutputPath(), "ut");
>+#if defined(WEBRTC_ANDROID)
>+TEST(LogTest, CheckTagAddedToStringInDefaultOnLogMessageAndroid) {
>+  std::string str;
>+  LogSinkImpl<StringStream> stream(&str);
>+  LogMessage::AddLogToStream(&stream, LS_INFO);
>+  EXPECT_EQ(LS_INFO, LogMessage::GetLogToStream(&stream));
>+
>+  RTC_LOG_TAG(LS_INFO, "my_tag") << "INFO";
>+  EXPECT_NE(std::string::npos, str.find("INFO"));
>+  EXPECT_NE(std::string::npos, str.find("my_tag"));
>+}
>+#endif
>+
>+TEST(LogTest, CheckNoopLogEntry) {
>+  if (LogMessage::GetLogToDebug() <= LS_SENSITIVE) {
>+    printf("CheckNoopLogEntry: skipping. Global severity is being overridden.");
>+    return;
>+  }
> 
>-  LogSinkImpl<FileStream> stream;
>-  EXPECT_TRUE(stream.Open(path, "wb", nullptr));
>-  stream.DisableBuffering();
>+  // Logging at LS_SENSITIVE severity, is by default turned off, so this should
>+  // be treated as a noop message.
>+  LogMessageForTesting log_msg("some/path/myfile.cc", 100, LS_SENSITIVE);
>+  log_msg.stream() << "Should be logged to nowhere.";
>+  EXPECT_TRUE(log_msg.is_noop());
>+  const std::string stream = log_msg.GetPrintStream();
>+  EXPECT_TRUE(stream.empty());
>+}
>+
>+// Test the time required to write 1000 80-character logs to a string.
>+TEST(LogTest, Perf) {
>+  std::string str;
>+  LogSinkImpl<StringStream> stream(&str);
>   LogMessage::AddLogToStream(&stream, LS_SENSITIVE);
> 
>+  const std::string message(80, 'X');
>+  {
>+    // Just to be sure that we're not measuring the performance of logging
>+    // noop log messages.
>+    LogMessageForTesting sanity_check_msg(__FILE__, __LINE__, LS_SENSITIVE);
>+    ASSERT_FALSE(sanity_check_msg.is_noop());
>+  }
>+
>+  // We now know how many bytes the logging framework will tag onto every msg.
>+  const size_t logging_overhead = str.size();
>+  // Reset the stream to 0 size.
>+  str.clear();
>+  str.reserve(120000);
>+  static const int kRepetitions = 1000;
>+
>   int64_t start = TimeMillis(), finish;
>-  std::string message('X', 80);
>-  for (int i = 0; i < 1000; ++i) {
>-    RTC_LOG(LS_SENSITIVE) << message;
>+  for (int i = 0; i < kRepetitions; ++i) {
>+    LogMessageForTesting(__FILE__, __LINE__, LS_SENSITIVE).stream() << message;
>   }
>   finish = TimeMillis();
> 
>   LogMessage::RemoveLogToStream(&stream);
>   stream.Close();
>-  webrtc::test::RemoveFile(path);
> 
>-  RTC_LOG(LS_INFO) << "Average log time: " << TimeDiff(finish, start) << " ms";
>+  EXPECT_EQ(str.size(), (message.size() + logging_overhead) * kRepetitions);
>+  RTC_LOG(LS_INFO) << "Total log time: " << TimeDiff(finish, start) << " ms "
>+                   << " total bytes logged: " << str.size();
> }
> 
> }  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logsinks.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logsinks.cc
>index 5f64f62aa32..662b1f20de9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logsinks.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logsinks.cc
>@@ -10,7 +10,7 @@
> 
> #include "rtc_base/logsinks.h"
> 
>-#include <iostream>
>+#include <cstdio>
> #include <string>
> 
> #include "rtc_base/checks.h"
>@@ -24,25 +24,35 @@ FileRotatingLogSink::FileRotatingLogSink(const std::string& log_dir_path,
>     : FileRotatingLogSink(new FileRotatingStream(log_dir_path,
>                                                  log_prefix,
>                                                  max_log_size,
>-                                                 num_log_files)) {
>-}
>+                                                 num_log_files)) {}
> 
> FileRotatingLogSink::FileRotatingLogSink(FileRotatingStream* stream)
>     : stream_(stream) {
>   RTC_DCHECK(stream);
> }
> 
>-FileRotatingLogSink::~FileRotatingLogSink() {
>-}
>+FileRotatingLogSink::~FileRotatingLogSink() {}
> 
> void FileRotatingLogSink::OnLogMessage(const std::string& message) {
>   if (stream_->GetState() != SS_OPEN) {
>-    std::cerr << "Init() must be called before adding this sink." << std::endl;
>+    std::fprintf(stderr, "Init() must be called before adding this sink.\n");
>     return;
>   }
>   stream_->WriteAll(message.c_str(), message.size(), nullptr, nullptr);
> }
> 
>+void FileRotatingLogSink::OnLogMessage(const std::string& message,
>+                                       LoggingSeverity sev,
>+                                       const char* tag) {
>+  if (stream_->GetState() != SS_OPEN) {
>+    std::fprintf(stderr, "Init() must be called before adding this sink.\n");
>+    return;
>+  }
>+  stream_->WriteAll(tag, strlen(tag), nullptr, nullptr);
>+  stream_->WriteAll(": ", 2, nullptr, nullptr);
>+  stream_->WriteAll(message.c_str(), message.size(), nullptr, nullptr);
>+}
>+
> bool FileRotatingLogSink::Init() {
>   return stream_->Open();
> }
>@@ -58,7 +68,6 @@ CallSessionFileRotatingLogSink::CallSessionFileRotatingLogSink(
>           new CallSessionFileRotatingStream(log_dir_path, max_total_log_size)) {
> }
> 
>-CallSessionFileRotatingLogSink::~CallSessionFileRotatingLogSink() {
>-}
>+CallSessionFileRotatingLogSink::~CallSessionFileRotatingLogSink() {}
> 
> }  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logsinks.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logsinks.h
>index 315ef9698be..caf4a5f728a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logsinks.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/logsinks.h
>@@ -35,6 +35,9 @@ class FileRotatingLogSink : public LogSink {
>   // Writes the message to the current file. It will spill over to the next
>   // file if needed.
>   void OnLogMessage(const std::string& message) override;
>+  void OnLogMessage(const std::string& message,
>+                    LoggingSeverity sev,
>+                    const char* tag) override;
> 
>   // Deletes any existing files in the directory and creates a new log file.
>   virtual bool Init();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/macutils.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/macutils.cc
>index d3f2919abf6..2b8700f4d60 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/macutils.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/macutils.cc
>@@ -10,7 +10,6 @@
> 
> #include <cstring>
> #include <memory>
>-#include <sstream>
> 
> #include <sys/utsname.h>
> 
>@@ -26,10 +25,11 @@ bool ToUtf8(const CFStringRef str16, std::string* str8) {
>     return false;
>   }
>   size_t maxlen = CFStringGetMaximumSizeForEncoding(CFStringGetLength(str16),
>-                                                    kCFStringEncodingUTF8) + 1;
>+                                                    kCFStringEncodingUTF8) +
>+                  1;
>   std::unique_ptr<char[]> buffer(new char[maxlen]);
>-  if (!buffer || !CFStringGetCString(str16, buffer.get(), maxlen,
>-                                     kCFStringEncodingUTF8)) {
>+  if (!buffer ||
>+      !CFStringGetCString(str16, buffer.get(), maxlen, kCFStringEncodingUTF8)) {
>     return false;
>   }
>   str8->assign(buffer.get());
>@@ -42,8 +42,7 @@ bool ToUtf16(const std::string& str8, CFStringRef* str16) {
>   }
>   *str16 = CFStringCreateWithBytes(kCFAllocatorDefault,
>                                    reinterpret_cast<const UInt8*>(str8.data()),
>-                                   str8.length(), kCFStringEncodingUTF8,
>-                                   false);
>+                                   str8.length(), kCFStringEncodingUTF8, false);
>   return nullptr != *str16;
> }
> }  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/md5.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/md5.cc
>deleted file mode 100644
>index 86cf892dc52..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/md5.cc
>+++ /dev/null
>@@ -1,222 +0,0 @@
>-/*
>- * This code implements the MD5 message-digest algorithm.
>- * The algorithm is due to Ron Rivest.  This code was
>- * written by Colin Plumb in 1993, no copyright is claimed.
>- * This code is in the public domain; do with it what you wish.
>- *
>- * Equivalent code is available from RSA Data Security, Inc.
>- * This code has been tested against that, and is equivalent,
>- * except that you don't need to include two pages of legalese
>- * with every copy.
>- *
>- * To compute the message digest of a chunk of bytes, declare an
>- * MD5Context structure, pass it to MD5Init, call MD5Update as
>- * needed on buffers full of bytes, and then call MD5Final, which
>- * will fill a supplied 16-byte array with the digest.
>- */
>-
>-// Changes from original C code:
>-// Ported to C++, type casting, Google code style.
>-
>-#include "rtc_base/md5.h"
>-
>-// TODO: Avoid memcmpy - hash directly from memory.
>-#include <string.h>  // for memcpy().
>-
>-#include "rtc_base/byteorder.h"  // for RTC_ARCH_CPU_LITTLE_ENDIAN.
>-
>-namespace rtc {
>-
>-#ifdef RTC_ARCH_CPU_LITTLE_ENDIAN
>-#define ByteReverse(buf, len)  // Nothing.
>-#else  // RTC_ARCH_CPU_BIG_ENDIAN
>-static void ByteReverse(uint32_t* buf, int len) {
>-  for (int i = 0; i < len; ++i) {
>-    buf[i] = rtc::GetLE32(&buf[i]);
>-  }
>-}
>-#endif
>-
>-// Start MD5 accumulation.  Set bit count to 0 and buffer to mysterious
>-// initialization constants.
>-void MD5Init(MD5Context* ctx) {
>-  ctx->buf[0] = 0x67452301;
>-  ctx->buf[1] = 0xefcdab89;
>-  ctx->buf[2] = 0x98badcfe;
>-  ctx->buf[3] = 0x10325476;
>-  ctx->bits[0] = 0;
>-  ctx->bits[1] = 0;
>-}
>-
>-// Update context to reflect the concatenation of another buffer full of bytes.
>-void MD5Update(MD5Context* ctx, const uint8_t* buf, size_t len) {
>-  // Update bitcount.
>-  uint32_t t = ctx->bits[0];
>-  if ((ctx->bits[0] = t + (static_cast<uint32_t>(len) << 3)) < t) {
>-    ctx->bits[1]++;  // Carry from low to high.
>-  }
>-  ctx->bits[1] += static_cast<uint32_t>(len >> 29);
>-  t = (t >> 3) & 0x3f;  // Bytes already in shsInfo->data.
>-
>-  // Handle any leading odd-sized chunks.
>-  if (t) {
>-    uint8_t* p = reinterpret_cast<uint8_t*>(ctx->in) + t;
>-
>-    t = 64-t;
>-    if (len < t) {
>-      memcpy(p, buf, len);
>-      return;
>-    }
>-    memcpy(p, buf, t);
>-    ByteReverse(ctx->in, 16);
>-    MD5Transform(ctx->buf, ctx->in);
>-    buf += t;
>-    len -= t;
>-  }
>-
>-  // Process data in 64-byte chunks.
>-  while (len >= 64) {
>-    memcpy(ctx->in, buf, 64);
>-    ByteReverse(ctx->in, 16);
>-    MD5Transform(ctx->buf, ctx->in);
>-    buf += 64;
>-    len -= 64;
>-  }
>-
>-  // Handle any remaining bytes of data.
>-  memcpy(ctx->in, buf, len);
>-}
>-
>-// Final wrapup - pad to 64-byte boundary with the bit pattern.
>-// 1 0* (64-bit count of bits processed, MSB-first)
>-void MD5Final(MD5Context* ctx, uint8_t digest[16]) {
>-  // Compute number of bytes mod 64.
>-  uint32_t count = (ctx->bits[0] >> 3) & 0x3F;
>-
>-  // Set the first char of padding to 0x80.  This is safe since there is
>-  // always at least one byte free.
>-  uint8_t* p = reinterpret_cast<uint8_t*>(ctx->in) + count;
>-  *p++ = 0x80;
>-
>-  // Bytes of padding needed to make 64 bytes.
>-  count = 64 - 1 - count;
>-
>-  // Pad out to 56 mod 64.
>-  if (count < 8) {
>-    // Two lots of padding:  Pad the first block to 64 bytes.
>-    memset(p, 0, count);
>-    ByteReverse(ctx->in, 16);
>-    MD5Transform(ctx->buf, ctx->in);
>-
>-    // Now fill the next block with 56 bytes.
>-    memset(ctx->in, 0, 56);
>-  } else {
>-    // Pad block to 56 bytes.
>-    memset(p, 0, count - 8);
>-  }
>-  ByteReverse(ctx->in, 14);
>-
>-  // Append length in bits and transform.
>-  ctx->in[14] = ctx->bits[0];
>-  ctx->in[15] = ctx->bits[1];
>-
>-  MD5Transform(ctx->buf, ctx->in);
>-  ByteReverse(ctx->buf, 4);
>-  memcpy(digest, ctx->buf, 16);
>-  memset(ctx, 0, sizeof(*ctx));  // In case it's sensitive.
>-}
>-
>-// The four core functions - F1 is optimized somewhat.
>-// #define F1(x, y, z) (x & y | ~x & z)
>-#define F1(x, y, z) (z ^ (x & (y ^ z)))
>-#define F2(x, y, z) F1(z, x, y)
>-#define F3(x, y, z) (x ^ y ^ z)
>-#define F4(x, y, z) (y ^ (x | ~z))
>-
>-// This is the central step in the MD5 algorithm.
>-#define MD5STEP(f, w, x, y, z, data, s) \
>-    (w += f(x, y, z) + data, w = w << s | w >> (32 - s), w += x)
>-
>-// The core of the MD5 algorithm, this alters an existing MD5 hash to
>-// reflect the addition of 16 longwords of new data.  MD5Update blocks
>-// the data and converts bytes into longwords for this routine.
>-void MD5Transform(uint32_t buf[4], const uint32_t in[16]) {
>-  uint32_t a = buf[0];
>-  uint32_t b = buf[1];
>-  uint32_t c = buf[2];
>-  uint32_t d = buf[3];
>-
>-  MD5STEP(F1, a, b, c, d, in[ 0] + 0xd76aa478, 7);
>-  MD5STEP(F1, d, a, b, c, in[ 1] + 0xe8c7b756, 12);
>-  MD5STEP(F1, c, d, a, b, in[ 2] + 0x242070db, 17);
>-  MD5STEP(F1, b, c, d, a, in[ 3] + 0xc1bdceee, 22);
>-  MD5STEP(F1, a, b, c, d, in[ 4] + 0xf57c0faf, 7);
>-  MD5STEP(F1, d, a, b, c, in[ 5] + 0x4787c62a, 12);
>-  MD5STEP(F1, c, d, a, b, in[ 6] + 0xa8304613, 17);
>-  MD5STEP(F1, b, c, d, a, in[ 7] + 0xfd469501, 22);
>-  MD5STEP(F1, a, b, c, d, in[ 8] + 0x698098d8, 7);
>-  MD5STEP(F1, d, a, b, c, in[ 9] + 0x8b44f7af, 12);
>-  MD5STEP(F1, c, d, a, b, in[10] + 0xffff5bb1, 17);
>-  MD5STEP(F1, b, c, d, a, in[11] + 0x895cd7be, 22);
>-  MD5STEP(F1, a, b, c, d, in[12] + 0x6b901122, 7);
>-  MD5STEP(F1, d, a, b, c, in[13] + 0xfd987193, 12);
>-  MD5STEP(F1, c, d, a, b, in[14] + 0xa679438e, 17);
>-  MD5STEP(F1, b, c, d, a, in[15] + 0x49b40821, 22);
>-
>-  MD5STEP(F2, a, b, c, d, in[ 1] + 0xf61e2562, 5);
>-  MD5STEP(F2, d, a, b, c, in[ 6] + 0xc040b340, 9);
>-  MD5STEP(F2, c, d, a, b, in[11] + 0x265e5a51, 14);
>-  MD5STEP(F2, b, c, d, a, in[ 0] + 0xe9b6c7aa, 20);
>-  MD5STEP(F2, a, b, c, d, in[ 5] + 0xd62f105d, 5);
>-  MD5STEP(F2, d, a, b, c, in[10] + 0x02441453, 9);
>-  MD5STEP(F2, c, d, a, b, in[15] + 0xd8a1e681, 14);
>-  MD5STEP(F2, b, c, d, a, in[ 4] + 0xe7d3fbc8, 20);
>-  MD5STEP(F2, a, b, c, d, in[ 9] + 0x21e1cde6, 5);
>-  MD5STEP(F2, d, a, b, c, in[14] + 0xc33707d6, 9);
>-  MD5STEP(F2, c, d, a, b, in[ 3] + 0xf4d50d87, 14);
>-  MD5STEP(F2, b, c, d, a, in[ 8] + 0x455a14ed, 20);
>-  MD5STEP(F2, a, b, c, d, in[13] + 0xa9e3e905, 5);
>-  MD5STEP(F2, d, a, b, c, in[ 2] + 0xfcefa3f8, 9);
>-  MD5STEP(F2, c, d, a, b, in[ 7] + 0x676f02d9, 14);
>-  MD5STEP(F2, b, c, d, a, in[12] + 0x8d2a4c8a, 20);
>-
>-  MD5STEP(F3, a, b, c, d, in[ 5] + 0xfffa3942, 4);
>-  MD5STEP(F3, d, a, b, c, in[ 8] + 0x8771f681, 11);
>-  MD5STEP(F3, c, d, a, b, in[11] + 0x6d9d6122, 16);
>-  MD5STEP(F3, b, c, d, a, in[14] + 0xfde5380c, 23);
>-  MD5STEP(F3, a, b, c, d, in[ 1] + 0xa4beea44, 4);
>-  MD5STEP(F3, d, a, b, c, in[ 4] + 0x4bdecfa9, 11);
>-  MD5STEP(F3, c, d, a, b, in[ 7] + 0xf6bb4b60, 16);
>-  MD5STEP(F3, b, c, d, a, in[10] + 0xbebfbc70, 23);
>-  MD5STEP(F3, a, b, c, d, in[13] + 0x289b7ec6, 4);
>-  MD5STEP(F3, d, a, b, c, in[ 0] + 0xeaa127fa, 11);
>-  MD5STEP(F3, c, d, a, b, in[ 3] + 0xd4ef3085, 16);
>-  MD5STEP(F3, b, c, d, a, in[ 6] + 0x04881d05, 23);
>-  MD5STEP(F3, a, b, c, d, in[ 9] + 0xd9d4d039, 4);
>-  MD5STEP(F3, d, a, b, c, in[12] + 0xe6db99e5, 11);
>-  MD5STEP(F3, c, d, a, b, in[15] + 0x1fa27cf8, 16);
>-  MD5STEP(F3, b, c, d, a, in[ 2] + 0xc4ac5665, 23);
>-
>-  MD5STEP(F4, a, b, c, d, in[ 0] + 0xf4292244, 6);
>-  MD5STEP(F4, d, a, b, c, in[ 7] + 0x432aff97, 10);
>-  MD5STEP(F4, c, d, a, b, in[14] + 0xab9423a7, 15);
>-  MD5STEP(F4, b, c, d, a, in[ 5] + 0xfc93a039, 21);
>-  MD5STEP(F4, a, b, c, d, in[12] + 0x655b59c3, 6);
>-  MD5STEP(F4, d, a, b, c, in[ 3] + 0x8f0ccc92, 10);
>-  MD5STEP(F4, c, d, a, b, in[10] + 0xffeff47d, 15);
>-  MD5STEP(F4, b, c, d, a, in[ 1] + 0x85845dd1, 21);
>-  MD5STEP(F4, a, b, c, d, in[ 8] + 0x6fa87e4f, 6);
>-  MD5STEP(F4, d, a, b, c, in[15] + 0xfe2ce6e0, 10);
>-  MD5STEP(F4, c, d, a, b, in[ 6] + 0xa3014314, 15);
>-  MD5STEP(F4, b, c, d, a, in[13] + 0x4e0811a1, 21);
>-  MD5STEP(F4, a, b, c, d, in[ 4] + 0xf7537e82, 6);
>-  MD5STEP(F4, d, a, b, c, in[11] + 0xbd3af235, 10);
>-  MD5STEP(F4, c, d, a, b, in[ 2] + 0x2ad7d2bb, 15);
>-  MD5STEP(F4, b, c, d, a, in[ 9] + 0xeb86d391, 21);
>-  buf[0] += a;
>-  buf[1] += b;
>-  buf[2] += c;
>-  buf[3] += d;
>-}
>-
>-}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/md5.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/md5.h
>deleted file mode 100644
>index dc0713c3519..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/md5.h
>+++ /dev/null
>@@ -1,44 +0,0 @@
>-/*
>- * This is the header file for the MD5 message-digest algorithm.
>- * The algorithm is due to Ron Rivest.  This code was
>- * written by Colin Plumb in 1993, no copyright is claimed.
>- * This code is in the public domain; do with it what you wish.
>- *
>- * Equivalent code is available from RSA Data Security, Inc.
>- * This code has been tested against that, and is equivalent,
>- * except that you don't need to include two pages of legalese
>- * with every copy.
>- * To compute the message digest of a chunk of bytes, declare an
>- * MD5Context structure, pass it to MD5Init, call MD5Update as
>- * needed on buffers full of bytes, and then call MD5Final, which
>- * will fill a supplied 16-byte array with the digest.
>- *
>- */
>-
>-// Changes(fbarchard): Ported to C++ and Google style guide.
>-// Made context first parameter in MD5Final for consistency with Sha1.
>-// Changes(hellner): added rtc namespace
>-// Changes(pbos): Reverted types back to uint32(8)_t with _t suffix.
>-
>-#ifndef RTC_BASE_MD5_H_
>-#define RTC_BASE_MD5_H_
>-
>-#include <stdint.h>
>-#include <stdlib.h>
>-
>-namespace rtc {
>-
>-struct MD5Context {
>-  uint32_t buf[4];
>-  uint32_t bits[2];
>-  uint32_t in[16];
>-};
>-
>-void MD5Init(MD5Context* context);
>-void MD5Update(MD5Context* context, const uint8_t* data, size_t len);
>-void MD5Final(MD5Context* context, uint8_t digest[16]);
>-void MD5Transform(uint32_t buf[4], const uint32_t in[16]);
>-
>-}  // namespace rtc
>-
>-#endif  // RTC_BASE_MD5_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/md5digest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/md5digest.cc
>deleted file mode 100644
>index 928dbe375a4..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/md5digest.cc
>+++ /dev/null
>@@ -1,32 +0,0 @@
>-/*
>- *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "rtc_base/md5digest.h"
>-
>-namespace rtc {
>-
>-size_t Md5Digest::Size() const {
>-  return kSize;
>-}
>-
>-void Md5Digest::Update(const void* buf, size_t len) {
>-  MD5Update(&ctx_, static_cast<const uint8_t*>(buf), len);
>-}
>-
>-size_t Md5Digest::Finish(void* buf, size_t len) {
>-  if (len < kSize) {
>-    return 0;
>-  }
>-  MD5Final(&ctx_, static_cast<uint8_t*>(buf));
>-  MD5Init(&ctx_);  // Reset for next use.
>-  return kSize;
>-}
>-
>-};  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/md5digest.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/md5digest.h
>deleted file mode 100644
>index a6558ef673c..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/md5digest.h
>+++ /dev/null
>@@ -1,36 +0,0 @@
>-/*
>- *  Copyright 2012 The WebRTC Project Authors. All rights reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef RTC_BASE_MD5DIGEST_H_
>-#define RTC_BASE_MD5DIGEST_H_
>-
>-#include "rtc_base/md5.h"
>-#include "rtc_base/messagedigest.h"
>-
>-namespace rtc {
>-
>-// A simple wrapper for our MD5 implementation.
>-class Md5Digest : public MessageDigest {
>- public:
>-  enum { kSize = 16 };
>-  Md5Digest() {
>-    MD5Init(&ctx_);
>-  }
>-  size_t Size() const override;
>-  void Update(const void* buf, size_t len) override;
>-  size_t Finish(void* buf, size_t len) override;
>-
>- private:
>-  MD5Context ctx_;
>-};
>-
>-}  // namespace rtc
>-
>-#endif  // RTC_BASE_MD5DIGEST_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/md5digest_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/md5digest_unittest.cc
>deleted file mode 100644
>index dcdc19836f3..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/md5digest_unittest.cc
>+++ /dev/null
>@@ -1,79 +0,0 @@
>-/*
>- *  Copyright 2012 The WebRTC Project Authors. All rights reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "rtc_base/md5digest.h"
>-#include "rtc_base/gunit.h"
>-#include "rtc_base/stringencode.h"
>-
>-namespace rtc {
>-
>-std::string Md5(const std::string& input) {
>-  Md5Digest md5;
>-  return ComputeDigest(&md5, input);
>-}
>-
>-TEST(Md5DigestTest, TestSize) {
>-  Md5Digest md5;
>-  EXPECT_EQ(16, static_cast<int>(Md5Digest::kSize));
>-  EXPECT_EQ(16U, md5.Size());
>-}
>-
>-TEST(Md5DigestTest, TestBasic) {
>-  // These are the standard MD5 test vectors from RFC 1321.
>-  EXPECT_EQ("d41d8cd98f00b204e9800998ecf8427e", Md5(""));
>-  EXPECT_EQ("0cc175b9c0f1b6a831c399e269772661", Md5("a"));
>-  EXPECT_EQ("900150983cd24fb0d6963f7d28e17f72", Md5("abc"));
>-  EXPECT_EQ("f96b697d7cb7938d525a2f31aaf161d0", Md5("message digest"));
>-  EXPECT_EQ("c3fcd3d76192e4007dfb496cca67e13b",
>-            Md5("abcdefghijklmnopqrstuvwxyz"));
>-}
>-
>-TEST(Md5DigestTest, TestMultipleUpdates) {
>-  Md5Digest md5;
>-  std::string input = "abcdefghijklmnopqrstuvwxyz";
>-  char output[Md5Digest::kSize];
>-  for (size_t i = 0; i < input.size(); ++i) {
>-    md5.Update(&input[i], 1);
>-  }
>-  EXPECT_EQ(md5.Size(), md5.Finish(output, sizeof(output)));
>-  EXPECT_EQ("c3fcd3d76192e4007dfb496cca67e13b",
>-            hex_encode(output, sizeof(output)));
>-}
>-
>-TEST(Md5DigestTest, TestReuse) {
>-  Md5Digest md5;
>-  std::string input = "message digest";
>-  EXPECT_EQ("f96b697d7cb7938d525a2f31aaf161d0", ComputeDigest(&md5, input));
>-  input = "abcdefghijklmnopqrstuvwxyz";
>-  EXPECT_EQ("c3fcd3d76192e4007dfb496cca67e13b", ComputeDigest(&md5, input));
>-}
>-
>-TEST(Md5DigestTest, TestBufferTooSmall) {
>-  Md5Digest md5;
>-  std::string input = "abcdefghijklmnopqrstuvwxyz";
>-  char output[Md5Digest::kSize - 1];
>-  md5.Update(input.c_str(), input.size());
>-  EXPECT_EQ(0U, md5.Finish(output, sizeof(output)));
>-}
>-
>-TEST(Md5DigestTest, TestBufferConst) {
>-  Md5Digest md5;
>-  const int kLongSize = 1000000;
>-  std::string input(kLongSize, '\0');
>-  for (int i = 0; i < kLongSize; ++i) {
>-    input[i] = static_cast<char>(i);
>-  }
>-  md5.Update(input.c_str(), input.size());
>-  for (int i = 0; i < kLongSize; ++i) {
>-    EXPECT_EQ(static_cast<char>(i), input[i]);
>-  }
>-}
>-
>-}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/memory/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/memory/BUILD.gn
>new file mode 100644
>index 00000000000..05fd7ababcd
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/memory/BUILD.gn
>@@ -0,0 +1,44 @@
>+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+#
>+# Use of this source code is governed by a BSD-style license
>+# that can be found in the LICENSE file in the root of the source
>+# tree. An additional intellectual property rights grant can be found
>+# in the file PATENTS.  All contributing project authors may
>+# be found in the AUTHORS file in the root of the source tree.
>+
>+import("../../webrtc.gni")
>+if (is_android) {
>+  import("//build/config/android/config.gni")
>+  import("//build/config/android/rules.gni")
>+}
>+
>+rtc_source_set("aligned_array") {
>+  sources = [
>+    "aligned_array.h",
>+  ]
>+  deps = [
>+    ":aligned_malloc",
>+    "..:checks",
>+  ]
>+}
>+
>+rtc_source_set("aligned_malloc") {
>+  sources = [
>+    "aligned_malloc.cc",
>+    "aligned_malloc.h",
>+  ]
>+  deps = []
>+}
>+
>+rtc_source_set("unittests") {
>+  testonly = true
>+  sources = [
>+    "aligned_array_unittest.cc",
>+    "aligned_malloc_unittest.cc",
>+  ]
>+  deps = [
>+    ":aligned_array",
>+    ":aligned_malloc",
>+    "../../test:test_support",
>+  ]
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/aligned_array.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/memory/aligned_array.h
>similarity index 89%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/aligned_array.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/memory/aligned_array.h
>index 793c7858207..dcdef1267e8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/aligned_array.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/memory/aligned_array.h
>@@ -8,11 +8,11 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#ifndef WEBRTC_SYSTEM_WRAPPERS_INCLUDE_ALIGNED_ARRAY_
>-#define WEBRTC_SYSTEM_WRAPPERS_INCLUDE_ALIGNED_ARRAY_
>+#ifndef RTC_BASE_MEMORY_ALIGNED_ARRAY_H_
>+#define RTC_BASE_MEMORY_ALIGNED_ARRAY_H_
> 
> #include "rtc_base/checks.h"
>-#include "system_wrappers/include/aligned_malloc.h"
>+#include "rtc_base/memory/aligned_malloc.h"
> 
> namespace webrtc {
> 
>@@ -75,4 +75,4 @@ class AlignedArray {
> 
> }  // namespace webrtc
> 
>-#endif  // WEBRTC_SYSTEM_WRAPPERS_INCLUDE_ALIGNED_ARRAY_
>+#endif  // RTC_BASE_MEMORY_ALIGNED_ARRAY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/aligned_array_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/memory/aligned_array_unittest.cc
>similarity index 96%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/aligned_array_unittest.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/memory/aligned_array_unittest.cc
>index e5a3c18fcc6..81fd468a926 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/aligned_array_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/memory/aligned_array_unittest.cc
>@@ -8,7 +8,7 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "system_wrappers/include/aligned_array.h"
>+#include "rtc_base/memory/aligned_array.h"
> 
> #include <stdint.h>
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/aligned_malloc.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/memory/aligned_malloc.cc
>similarity index 96%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/aligned_malloc.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/memory/aligned_malloc.cc
>index 43ece9e681f..2943a64208d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/aligned_malloc.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/memory/aligned_malloc.cc
>@@ -8,7 +8,7 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "system_wrappers/include/aligned_malloc.h"
>+#include "rtc_base/memory/aligned_malloc.h"
> 
> #include <memory.h>
> #include <stdlib.h>
>@@ -19,8 +19,6 @@
> #include <stdint.h>
> #endif
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>-
> // Reference on memory alignment:
> // http://stackoverflow.com/questions/227897/solve-the-memory-alignment-in-c-interview-question-that-stumped-me
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/aligned_malloc.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/memory/aligned_malloc.h
>similarity index 92%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/aligned_malloc.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/memory/aligned_malloc.h
>index 33b23d2aece..42a6daaef6d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/aligned_malloc.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/memory/aligned_malloc.h
>@@ -8,8 +8,8 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#ifndef SYSTEM_WRAPPERS_INCLUDE_ALIGNED_MALLOC_H_
>-#define SYSTEM_WRAPPERS_INCLUDE_ALIGNED_MALLOC_H_
>+#ifndef RTC_BASE_MEMORY_ALIGNED_MALLOC_H_
>+#define RTC_BASE_MEMORY_ALIGNED_MALLOC_H_
> 
> // The functions declared here
> // 1) Allocates block of aligned memory.
>@@ -54,4 +54,4 @@ struct AlignedFreeDeleter {
> 
> }  // namespace webrtc
> 
>-#endif  // SYSTEM_WRAPPERS_INCLUDE_ALIGNED_MALLOC_H_
>+#endif  // RTC_BASE_MEMORY_ALIGNED_MALLOC_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/aligned_malloc_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/memory/aligned_malloc_unittest.cc
>similarity index 95%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/aligned_malloc_unittest.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/memory/aligned_malloc_unittest.cc
>index 7afbf6d1bba..572af5a8bb6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/aligned_malloc_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/memory/aligned_malloc_unittest.cc
>@@ -8,7 +8,7 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "system_wrappers/include/aligned_malloc.h"
>+#include "rtc_base/memory/aligned_malloc.h"
> 
> #include <memory>
> 
>@@ -19,7 +19,6 @@
> #endif
> 
> #include "test/gtest.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/memory_usage.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/memory_usage.cc
>index 41b27ededaf..a70c54753fc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/memory_usage.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/memory_usage.cc
>@@ -12,14 +12,17 @@
> 
> #if defined(WEBRTC_LINUX)
> #include <unistd.h>
>-#include <cstdlib>
> #include <cstdio>
>+#include <cstdlib>
> #include <cstring>
> #elif defined(WEBRTC_MAC)
> #include <mach/mach.h>
> #elif defined(WEBRTC_WIN)
>+// clang-format off
>+// clang formating would change include order.
> #include <windows.h>
>-#include <psapi.h>
>+#include <psapi.h> // must come after windows.h
>+// clang-format on
> #endif
> 
> #include "rtc_base/logging.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/memory_usage_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/memory_usage_unittest.cc
>index 87cdcef89bb..9c82f98034e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/memory_usage_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/memory_usage_unittest.cc
>@@ -20,4 +20,3 @@ TEST(GetMemoryUsage, SimpleTest) {
> }
> 
> }  // namespace rtc
>-
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagedigest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagedigest.cc
>index bf8d25dcbf8..9c10bcd486a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagedigest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagedigest.cc
>@@ -14,15 +14,14 @@
> 
> #include <string.h>
> 
>-#include "rtc_base/basictypes.h"
> #include "rtc_base/openssldigest.h"
> #include "rtc_base/stringencode.h"
> 
> namespace rtc {
> 
> // From RFC 4572.
>-const char DIGEST_MD5[]     = "md5";
>-const char DIGEST_SHA_1[]   = "sha-1";
>+const char DIGEST_MD5[] = "md5";
>+const char DIGEST_SHA_1[] = "sha-1";
> const char DIGEST_SHA_224[] = "sha-224";
> const char DIGEST_SHA_256[] = "sha-256";
> const char DIGEST_SHA_384[] = "sha-384";
>@@ -45,35 +44,39 @@ bool IsFips180DigestAlgorithm(const std::string& alg) {
>   // consideration) MUST use one of the FIPS 180 algorithms (SHA-1,
>   // SHA-224, SHA-256, SHA-384, or SHA-512) as their signature algorithm,
>   // and thus also MUST use it to calculate certificate fingerprints."
>-  return alg == DIGEST_SHA_1 ||
>-         alg == DIGEST_SHA_224 ||
>-         alg == DIGEST_SHA_256 ||
>-         alg == DIGEST_SHA_384 ||
>+  return alg == DIGEST_SHA_1 || alg == DIGEST_SHA_224 ||
>+         alg == DIGEST_SHA_256 || alg == DIGEST_SHA_384 ||
>          alg == DIGEST_SHA_512;
> }
> 
>-size_t ComputeDigest(MessageDigest* digest, const void* input, size_t in_len,
>-                     void* output, size_t out_len) {
>+size_t ComputeDigest(MessageDigest* digest,
>+                     const void* input,
>+                     size_t in_len,
>+                     void* output,
>+                     size_t out_len) {
>   digest->Update(input, in_len);
>   return digest->Finish(output, out_len);
> }
> 
>-size_t ComputeDigest(const std::string& alg, const void* input, size_t in_len,
>-                     void* output, size_t out_len) {
>+size_t ComputeDigest(const std::string& alg,
>+                     const void* input,
>+                     size_t in_len,
>+                     void* output,
>+                     size_t out_len) {
>   std::unique_ptr<MessageDigest> digest(MessageDigestFactory::Create(alg));
>-  return (digest) ?
>-      ComputeDigest(digest.get(), input, in_len, output, out_len) :
>-      0;
>+  return (digest) ? ComputeDigest(digest.get(), input, in_len, output, out_len)
>+                  : 0;
> }
> 
> std::string ComputeDigest(MessageDigest* digest, const std::string& input) {
>   std::unique_ptr<char[]> output(new char[digest->Size()]);
>-  ComputeDigest(digest, input.data(), input.size(),
>-                output.get(), digest->Size());
>+  ComputeDigest(digest, input.data(), input.size(), output.get(),
>+                digest->Size());
>   return hex_encode(output.get(), digest->Size());
> }
> 
>-bool ComputeDigest(const std::string& alg, const std::string& input,
>+bool ComputeDigest(const std::string& alg,
>+                   const std::string& input,
>                    std::string* output) {
>   std::unique_ptr<MessageDigest> digest(MessageDigestFactory::Create(alg));
>   if (!digest) {
>@@ -91,9 +94,12 @@ std::string ComputeDigest(const std::string& alg, const std::string& input) {
> 
> // Compute a RFC 2104 HMAC: H(K XOR opad, H(K XOR ipad, text))
> size_t ComputeHmac(MessageDigest* digest,
>-                   const void* key, size_t key_len,
>-                   const void* input, size_t in_len,
>-                   void* output, size_t out_len) {
>+                   const void* key,
>+                   size_t key_len,
>+                   const void* input,
>+                   size_t in_len,
>+                   void* output,
>+                   size_t out_len) {
>   // We only handle algorithms with a 64-byte blocksize.
>   // TODO: Add BlockSize() method to MessageDigest.
>   size_t block_len = kBlockSize;
>@@ -128,27 +134,34 @@ size_t ComputeHmac(MessageDigest* digest,
>   return digest->Finish(output, out_len);
> }
> 
>-size_t ComputeHmac(const std::string& alg, const void* key, size_t key_len,
>-                   const void* input, size_t in_len,
>-                   void* output, size_t out_len) {
>+size_t ComputeHmac(const std::string& alg,
>+                   const void* key,
>+                   size_t key_len,
>+                   const void* input,
>+                   size_t in_len,
>+                   void* output,
>+                   size_t out_len) {
>   std::unique_ptr<MessageDigest> digest(MessageDigestFactory::Create(alg));
>   if (!digest) {
>     return 0;
>   }
>-  return ComputeHmac(digest.get(), key, key_len,
>-                     input, in_len, output, out_len);
>+  return ComputeHmac(digest.get(), key, key_len, input, in_len, output,
>+                     out_len);
> }
> 
>-std::string ComputeHmac(MessageDigest* digest, const std::string& key,
>+std::string ComputeHmac(MessageDigest* digest,
>+                        const std::string& key,
>                         const std::string& input) {
>   std::unique_ptr<char[]> output(new char[digest->Size()]);
>-  ComputeHmac(digest, key.data(), key.size(),
>-              input.data(), input.size(), output.get(), digest->Size());
>+  ComputeHmac(digest, key.data(), key.size(), input.data(), input.size(),
>+              output.get(), digest->Size());
>   return hex_encode(output.get(), digest->Size());
> }
> 
>-bool ComputeHmac(const std::string& alg, const std::string& key,
>-                 const std::string& input, std::string* output) {
>+bool ComputeHmac(const std::string& alg,
>+                 const std::string& key,
>+                 const std::string& input,
>+                 std::string* output) {
>   std::unique_ptr<MessageDigest> digest(MessageDigestFactory::Create(alg));
>   if (!digest) {
>     return false;
>@@ -157,7 +170,8 @@ bool ComputeHmac(const std::string& alg, const std::string& key,
>   return true;
> }
> 
>-std::string ComputeHmac(const std::string& alg, const std::string& key,
>+std::string ComputeHmac(const std::string& alg,
>+                        const std::string& key,
>                         const std::string& input) {
>   std::string output;
>   ComputeHmac(alg, key, input, &output);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagedigest.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagedigest.h
>index f80dd7aa53d..fc820885436 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagedigest.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagedigest.h
>@@ -52,13 +52,19 @@ bool IsFips180DigestAlgorithm(const std::string& alg);
> // implementation, and outputs the hash to the buffer |output|, which is
> // |out_len| bytes long. Returns the number of bytes written to |output| if
> // successful, or 0 if |out_len| was too small.
>-size_t ComputeDigest(MessageDigest* digest, const void* input, size_t in_len,
>-                     void* output, size_t out_len);
>+size_t ComputeDigest(MessageDigest* digest,
>+                     const void* input,
>+                     size_t in_len,
>+                     void* output,
>+                     size_t out_len);
> // Like the previous function, but creates a digest implementation based on
> // the desired digest name |alg|, e.g. DIGEST_SHA_1. Returns 0 if there is no
> // digest with the given name.
>-size_t ComputeDigest(const std::string& alg, const void* input, size_t in_len,
>-                     void* output, size_t out_len);
>+size_t ComputeDigest(const std::string& alg,
>+                     const void* input,
>+                     size_t in_len,
>+                     void* output,
>+                     size_t out_len);
> // Computes the hash of |input| using the |digest| hash implementation, and
> // returns it as a hex-encoded string.
> std::string ComputeDigest(MessageDigest* digest, const std::string& input);
>@@ -67,7 +73,8 @@ std::string ComputeDigest(MessageDigest* digest, const std::string& input);
> // there is no digest with the given name.
> std::string ComputeDigest(const std::string& alg, const std::string& input);
> // Like the previous function, but returns an explicit result code.
>-bool ComputeDigest(const std::string& alg, const std::string& input,
>+bool ComputeDigest(const std::string& alg,
>+                   const std::string& input,
>                    std::string* output);
> 
> // Shorthand way to compute a hex-encoded hash using MD5.
>@@ -82,27 +89,39 @@ inline std::string MD5(const std::string& input) {
> // the HMAC to the buffer |output|, which is |out_len| bytes long. Returns the
> // number of bytes written to |output| if successful, or 0 if |out_len| was too
> // small.
>-size_t ComputeHmac(MessageDigest* digest, const void* key, size_t key_len,
>-                   const void* input, size_t in_len,
>-                   void* output, size_t out_len);
>+size_t ComputeHmac(MessageDigest* digest,
>+                   const void* key,
>+                   size_t key_len,
>+                   const void* input,
>+                   size_t in_len,
>+                   void* output,
>+                   size_t out_len);
> // Like the previous function, but creates a digest implementation based on
> // the desired digest name |alg|, e.g. DIGEST_SHA_1. Returns 0 if there is no
> // digest with the given name.
>-size_t ComputeHmac(const std::string& alg, const void* key, size_t key_len,
>-                   const void* input, size_t in_len,
>-                   void* output, size_t out_len);
>+size_t ComputeHmac(const std::string& alg,
>+                   const void* key,
>+                   size_t key_len,
>+                   const void* input,
>+                   size_t in_len,
>+                   void* output,
>+                   size_t out_len);
> // Computes the HMAC of |input| using the |digest| hash implementation and |key|
> // to key the HMAC, and returns it as a hex-encoded string.
>-std::string ComputeHmac(MessageDigest* digest, const std::string& key,
>+std::string ComputeHmac(MessageDigest* digest,
>+                        const std::string& key,
>                         const std::string& input);
> // Like the previous function, but creates a digest implementation based on
> // the desired digest name |alg|, e.g. DIGEST_SHA_1. Returns empty string if
> // there is no digest with the given name.
>-std::string ComputeHmac(const std::string& alg, const std::string& key,
>+std::string ComputeHmac(const std::string& alg,
>+                        const std::string& key,
>                         const std::string& input);
> // Like the previous function, but returns an explicit result code.
>-bool ComputeHmac(const std::string& alg, const std::string& key,
>-                 const std::string& input, std::string* output);
>+bool ComputeHmac(const std::string& alg,
>+                 const std::string& key,
>+                 const std::string& input,
>+                 std::string* output);
> 
> }  // namespace rtc
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagedigest_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagedigest_unittest.cc
>index 4d7c338fa8b..5524b18b996 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagedigest_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagedigest_unittest.cc
>@@ -17,42 +17,42 @@ namespace rtc {
> // Test vectors from RFC 1321.
> TEST(MessageDigestTest, TestMd5Digest) {
>   // Test the string versions of the APIs.
>-  EXPECT_EQ("d41d8cd98f00b204e9800998ecf8427e",
>-      ComputeDigest(DIGEST_MD5, ""));
>+  EXPECT_EQ("d41d8cd98f00b204e9800998ecf8427e", ComputeDigest(DIGEST_MD5, ""));
>   EXPECT_EQ("900150983cd24fb0d6963f7d28e17f72",
>-      ComputeDigest(DIGEST_MD5, "abc"));
>+            ComputeDigest(DIGEST_MD5, "abc"));
>   EXPECT_EQ("c3fcd3d76192e4007dfb496cca67e13b",
>-      ComputeDigest(DIGEST_MD5, "abcdefghijklmnopqrstuvwxyz"));
>+            ComputeDigest(DIGEST_MD5, "abcdefghijklmnopqrstuvwxyz"));
> 
>   // Test the raw buffer versions of the APIs; also check output buffer size.
>   char output[16];
>   EXPECT_EQ(sizeof(output),
>-      ComputeDigest(DIGEST_MD5, "abc", 3, output, sizeof(output)));
>+            ComputeDigest(DIGEST_MD5, "abc", 3, output, sizeof(output)));
>   EXPECT_EQ("900150983cd24fb0d6963f7d28e17f72",
>-      hex_encode(output, sizeof(output)));
>+            hex_encode(output, sizeof(output)));
>   EXPECT_EQ(0U,
>-      ComputeDigest(DIGEST_MD5, "abc", 3, output, sizeof(output) - 1));
>+            ComputeDigest(DIGEST_MD5, "abc", 3, output, sizeof(output) - 1));
> }
> 
> // Test vectors from RFC 3174.
> TEST(MessageDigestTest, TestSha1Digest) {
>   // Test the string versions of the APIs.
>   EXPECT_EQ("da39a3ee5e6b4b0d3255bfef95601890afd80709",
>-      ComputeDigest(DIGEST_SHA_1, ""));
>+            ComputeDigest(DIGEST_SHA_1, ""));
>   EXPECT_EQ("a9993e364706816aba3e25717850c26c9cd0d89d",
>-      ComputeDigest(DIGEST_SHA_1, "abc"));
>+            ComputeDigest(DIGEST_SHA_1, "abc"));
>   EXPECT_EQ("84983e441c3bd26ebaae4aa1f95129e5e54670f1",
>-      ComputeDigest(DIGEST_SHA_1,
>-          "abcdbcdecdefdefgefghfghighijhijkijkljklmklmnlmnomnopnopq"));
>+            ComputeDigest(
>+                DIGEST_SHA_1,
>+                "abcdbcdecdefdefgefghfghighijhijkijkljklmklmnlmnomnopnopq"));
> 
>   // Test the raw buffer versions of the APIs; also check output buffer size.
>   char output[20];
>   EXPECT_EQ(sizeof(output),
>-      ComputeDigest(DIGEST_SHA_1, "abc", 3, output, sizeof(output)));
>+            ComputeDigest(DIGEST_SHA_1, "abc", 3, output, sizeof(output)));
>   EXPECT_EQ("a9993e364706816aba3e25717850c26c9cd0d89d",
>-      hex_encode(output, sizeof(output)));
>+            hex_encode(output, sizeof(output)));
>   EXPECT_EQ(0U,
>-      ComputeDigest(DIGEST_SHA_1, "abc", 3, output, sizeof(output) - 1));
>+            ComputeDigest(DIGEST_SHA_1, "abc", 3, output, sizeof(output) - 1));
> }
> 
> // Test that we fail properly if a bad digest algorithm is specified.
>@@ -66,80 +66,83 @@ TEST(MessageDigestTest, TestBadDigest) {
> TEST(MessageDigestTest, TestMd5Hmac) {
>   // Test the string versions of the APIs.
>   EXPECT_EQ("9294727a3638bb1c13f48ef8158bfc9d",
>-      ComputeHmac(DIGEST_MD5, std::string(16, '\x0b'), "Hi There"));
>+            ComputeHmac(DIGEST_MD5, std::string(16, '\x0b'), "Hi There"));
>   EXPECT_EQ("750c783e6ab0b503eaa86e310a5db738",
>-      ComputeHmac(DIGEST_MD5, "Jefe", "what do ya want for nothing?"));
>+            ComputeHmac(DIGEST_MD5, "Jefe", "what do ya want for nothing?"));
>   EXPECT_EQ("56be34521d144c88dbb8c733f0e8b3f6",
>-      ComputeHmac(DIGEST_MD5, std::string(16, '\xaa'),
>-          std::string(50, '\xdd')));
>-  EXPECT_EQ("697eaf0aca3a3aea3a75164746ffaa79",
>+            ComputeHmac(DIGEST_MD5, std::string(16, '\xaa'),
>+                        std::string(50, '\xdd')));
>+  EXPECT_EQ(
>+      "697eaf0aca3a3aea3a75164746ffaa79",
>       ComputeHmac(DIGEST_MD5,
>-          "\x01\x02\x03\x04\x05\x06\x07\x08\x09\x0a\x0b\x0c\x0d\x0e\x0f"
>-          "\x10\x11\x12\x13\x14\x15\x16\x17\x18\x19",
>-          std::string(50, '\xcd')));
>-  EXPECT_EQ("56461ef2342edc00f9bab995690efd4c",
>-      ComputeHmac(DIGEST_MD5, std::string(16, '\x0c'),
>-          "Test With Truncation"));
>-  EXPECT_EQ("6b1ab7fe4bd7bf8f0b62e6ce61b9d0cd",
>+                  "\x01\x02\x03\x04\x05\x06\x07\x08\x09\x0a\x0b\x0c\x0d\x0e\x0f"
>+                  "\x10\x11\x12\x13\x14\x15\x16\x17\x18\x19",
>+                  std::string(50, '\xcd')));
>+  EXPECT_EQ(
>+      "56461ef2342edc00f9bab995690efd4c",
>+      ComputeHmac(DIGEST_MD5, std::string(16, '\x0c'), "Test With Truncation"));
>+  EXPECT_EQ(
>+      "6b1ab7fe4bd7bf8f0b62e6ce61b9d0cd",
>       ComputeHmac(DIGEST_MD5, std::string(80, '\xaa'),
>-          "Test Using Larger Than Block-Size Key - Hash Key First"));
>+                  "Test Using Larger Than Block-Size Key - Hash Key First"));
>   EXPECT_EQ("6f630fad67cda0ee1fb1f562db3aa53e",
>-      ComputeHmac(DIGEST_MD5, std::string(80, '\xaa'),
>-          "Test Using Larger Than Block-Size Key and Larger "
>-          "Than One Block-Size Data"));
>+            ComputeHmac(DIGEST_MD5, std::string(80, '\xaa'),
>+                        "Test Using Larger Than Block-Size Key and Larger "
>+                        "Than One Block-Size Data"));
> 
>   // Test the raw buffer versions of the APIs; also check output buffer size.
>   std::string key(16, '\x0b');
>   std::string input("Hi There");
>   char output[16];
>   EXPECT_EQ(sizeof(output),
>-      ComputeHmac(DIGEST_MD5, key.c_str(), key.size(),
>-          input.c_str(), input.size(), output, sizeof(output)));
>+            ComputeHmac(DIGEST_MD5, key.c_str(), key.size(), input.c_str(),
>+                        input.size(), output, sizeof(output)));
>   EXPECT_EQ("9294727a3638bb1c13f48ef8158bfc9d",
>-      hex_encode(output, sizeof(output)));
>-  EXPECT_EQ(0U,
>-      ComputeHmac(DIGEST_MD5, key.c_str(), key.size(),
>-          input.c_str(), input.size(), output, sizeof(output) - 1));
>+            hex_encode(output, sizeof(output)));
>+  EXPECT_EQ(0U, ComputeHmac(DIGEST_MD5, key.c_str(), key.size(), input.c_str(),
>+                            input.size(), output, sizeof(output) - 1));
> }
> 
> // Test vectors from RFC 2202.
> TEST(MessageDigestTest, TestSha1Hmac) {
>   // Test the string versions of the APIs.
>   EXPECT_EQ("b617318655057264e28bc0b6fb378c8ef146be00",
>-      ComputeHmac(DIGEST_SHA_1, std::string(20, '\x0b'), "Hi There"));
>+            ComputeHmac(DIGEST_SHA_1, std::string(20, '\x0b'), "Hi There"));
>   EXPECT_EQ("effcdf6ae5eb2fa2d27416d5f184df9c259a7c79",
>-      ComputeHmac(DIGEST_SHA_1, "Jefe", "what do ya want for nothing?"));
>+            ComputeHmac(DIGEST_SHA_1, "Jefe", "what do ya want for nothing?"));
>   EXPECT_EQ("125d7342b9ac11cd91a39af48aa17b4f63f175d3",
>-      ComputeHmac(DIGEST_SHA_1, std::string(20, '\xaa'),
>-          std::string(50, '\xdd')));
>-  EXPECT_EQ("4c9007f4026250c6bc8414f9bf50c86c2d7235da",
>+            ComputeHmac(DIGEST_SHA_1, std::string(20, '\xaa'),
>+                        std::string(50, '\xdd')));
>+  EXPECT_EQ(
>+      "4c9007f4026250c6bc8414f9bf50c86c2d7235da",
>       ComputeHmac(DIGEST_SHA_1,
>-          "\x01\x02\x03\x04\x05\x06\x07\x08\x09\x0a\x0b\x0c\x0d\x0e\x0f"
>-          "\x10\x11\x12\x13\x14\x15\x16\x17\x18\x19",
>-          std::string(50, '\xcd')));
>+                  "\x01\x02\x03\x04\x05\x06\x07\x08\x09\x0a\x0b\x0c\x0d\x0e\x0f"
>+                  "\x10\x11\x12\x13\x14\x15\x16\x17\x18\x19",
>+                  std::string(50, '\xcd')));
>   EXPECT_EQ("4c1a03424b55e07fe7f27be1d58bb9324a9a5a04",
>-      ComputeHmac(DIGEST_SHA_1, std::string(20, '\x0c'),
>-          "Test With Truncation"));
>-  EXPECT_EQ("aa4ae5e15272d00e95705637ce8a3b55ed402112",
>+            ComputeHmac(DIGEST_SHA_1, std::string(20, '\x0c'),
>+                        "Test With Truncation"));
>+  EXPECT_EQ(
>+      "aa4ae5e15272d00e95705637ce8a3b55ed402112",
>       ComputeHmac(DIGEST_SHA_1, std::string(80, '\xaa'),
>-          "Test Using Larger Than Block-Size Key - Hash Key First"));
>+                  "Test Using Larger Than Block-Size Key - Hash Key First"));
>   EXPECT_EQ("e8e99d0f45237d786d6bbaa7965c7808bbff1a91",
>-      ComputeHmac(DIGEST_SHA_1, std::string(80, '\xaa'),
>-          "Test Using Larger Than Block-Size Key and Larger "
>-          "Than One Block-Size Data"));
>+            ComputeHmac(DIGEST_SHA_1, std::string(80, '\xaa'),
>+                        "Test Using Larger Than Block-Size Key and Larger "
>+                        "Than One Block-Size Data"));
> 
>   // Test the raw buffer versions of the APIs; also check output buffer size.
>   std::string key(20, '\x0b');
>   std::string input("Hi There");
>   char output[20];
>   EXPECT_EQ(sizeof(output),
>-      ComputeHmac(DIGEST_SHA_1, key.c_str(), key.size(),
>-          input.c_str(), input.size(), output, sizeof(output)));
>+            ComputeHmac(DIGEST_SHA_1, key.c_str(), key.size(), input.c_str(),
>+                        input.size(), output, sizeof(output)));
>   EXPECT_EQ("b617318655057264e28bc0b6fb378c8ef146be00",
>-      hex_encode(output, sizeof(output)));
>+            hex_encode(output, sizeof(output)));
>   EXPECT_EQ(0U,
>-      ComputeHmac(DIGEST_SHA_1, key.c_str(), key.size(),
>-          input.c_str(), input.size(), output, sizeof(output) - 1));
>+            ComputeHmac(DIGEST_SHA_1, key.c_str(), key.size(), input.c_str(),
>+                        input.size(), output, sizeof(output) - 1));
> }
> 
> TEST(MessageDigestTest, TestBadHmac) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagehandler.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagehandler.cc
>index 2f580cc703b..7376defaf71 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagehandler.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagehandler.cc
>@@ -17,4 +17,4 @@ MessageHandler::~MessageHandler() {
>   MessageQueueManager::Clear(this);
> }
> 
>-} // namespace rtc
>+}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagehandler.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagehandler.h
>index dec3b8b86a0..df2d1ada8dd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagehandler.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagehandler.h
>@@ -40,9 +40,7 @@ class FunctorMessageHandler : public MessageHandler {
>  public:
>   explicit FunctorMessageHandler(FunctorT&& functor)
>       : functor_(std::forward<FunctorT>(functor)) {}
>-  virtual void OnMessage(Message*) {
>-    result_ = functor_();
>-  }
>+  virtual void OnMessage(Message* msg) { result_ = functor_(); }
>   const ReturnT& result() const { return result_; }
> 
>   // Returns moved result. Should not call result() or MoveResult() again
>@@ -58,11 +56,8 @@ class FunctorMessageHandler : public MessageHandler {
> template <class FunctorT>
> class FunctorMessageHandler<void, FunctorT> : public MessageHandler {
>  public:
>-  explicit FunctorMessageHandler(const FunctorT& functor)
>-      : functor_(functor) {}
>-  virtual void OnMessage(Message*) {
>-    functor_();
>-  }
>+  explicit FunctorMessageHandler(const FunctorT& functor) : functor_(functor) {}
>+  virtual void OnMessage(Message* msg) { functor_(); }
>   void result() const {}
>   void MoveResult() {}
> 
>@@ -70,6 +65,6 @@ class FunctorMessageHandler<void, FunctorT> : public MessageHandler {
>   FunctorT functor_;
> };
> 
>-} // namespace rtc
>+}  // namespace rtc
> 
>-#endif // RTC_BASE_MESSAGEHANDLER_H_
>+#endif  // RTC_BASE_MESSAGEHANDLER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagequeue.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagequeue.cc
>index 001d3ed1484..035ff070669 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagequeue.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagequeue.cc
>@@ -20,7 +20,7 @@
> namespace rtc {
> namespace {
> 
>-const int kMaxMsgLatency = 150;  // 150 ms
>+const int kMaxMsgLatency = 150;                // 150 ms
> const int kSlowDispatchLoggingThreshold = 50;  // 50 ms
> 
> class RTC_SCOPED_LOCKABLE MarkProcessingCritScope {
>@@ -64,26 +64,26 @@ bool MessageQueueManager::IsInitialized() {
> 
> MessageQueueManager::MessageQueueManager() : processing_(0) {}
> 
>-MessageQueueManager::~MessageQueueManager() {
>-}
>+MessageQueueManager::~MessageQueueManager() {}
> 
>-void MessageQueueManager::Add(MessageQueue *message_queue) {
>+void MessageQueueManager::Add(MessageQueue* message_queue) {
>   return Instance()->AddInternal(message_queue);
> }
>-void MessageQueueManager::AddInternal(MessageQueue *message_queue) {
>+void MessageQueueManager::AddInternal(MessageQueue* message_queue) {
>   CritScope cs(&crit_);
>   // Prevent changes while the list of message queues is processed.
>   RTC_DCHECK_EQ(processing_, 0);
>   message_queues_.push_back(message_queue);
> }
> 
>-void MessageQueueManager::Remove(MessageQueue *message_queue) {
>+void MessageQueueManager::Remove(MessageQueue* message_queue) {
>   // If there isn't a message queue manager instance, then there isn't a queue
>   // to remove.
>-  if (!instance_) return;
>+  if (!instance_)
>+    return;
>   return Instance()->RemoveInternal(message_queue);
> }
>-void MessageQueueManager::RemoveInternal(MessageQueue *message_queue) {
>+void MessageQueueManager::RemoveInternal(MessageQueue* message_queue) {
>   // If this is the last MessageQueue, destroy the manager as well so that
>   // we don't leak this object at program shutdown. As mentioned above, this is
>   // not thread-safe, but this should only happen at program termination (when
>@@ -93,7 +93,7 @@ void MessageQueueManager::RemoveInternal(MessageQueue *message_queue) {
>     CritScope cs(&crit_);
>     // Prevent changes while the list of message queues is processed.
>     RTC_DCHECK_EQ(processing_, 0);
>-    std::vector<MessageQueue *>::iterator iter;
>+    std::vector<MessageQueue*>::iterator iter;
>     iter = std::find(message_queues_.begin(), message_queues_.end(),
>                      message_queue);
>     if (iter != message_queues_.end()) {
>@@ -107,18 +107,18 @@ void MessageQueueManager::RemoveInternal(MessageQueue *message_queue) {
>   }
> }
> 
>-void MessageQueueManager::Clear(MessageHandler *handler) {
>+void MessageQueueManager::Clear(MessageHandler* handler) {
>   // If there isn't a message queue manager instance, then there aren't any
>   // queues to remove this handler from.
>-  if (!instance_) return;
>+  if (!instance_)
>+    return;
>   return Instance()->ClearInternal(handler);
> }
>-void MessageQueueManager::ClearInternal(MessageHandler *handler) {
>+void MessageQueueManager::ClearInternal(MessageHandler* handler) {
>   // Deleted objects may cause re-entrant calls to ClearInternal. This is
>   // allowed as the list of message queues does not change while queues are
>   // cleared.
>   MarkProcessingCritScope cs(&crit_, &processing_);
>-  std::vector<MessageQueue *>::iterator iter;
>   for (MessageQueue* queue : message_queues_) {
>     queue->Clear(handler);
>   }
>@@ -253,7 +253,7 @@ void MessageQueue::Restart() {
>   AtomicOps::ReleaseStore(&stop_, 0);
> }
> 
>-bool MessageQueue::Peek(Message *pmsg, int cmsWait) {
>+bool MessageQueue::Peek(Message* pmsg, int cmsWait) {
>   if (fPeekKeep_) {
>     *pmsg = msgPeek_;
>     return true;
>@@ -265,7 +265,7 @@ bool MessageQueue::Peek(Message *pmsg, int cmsWait) {
>   return true;
> }
> 
>-bool MessageQueue::Get(Message *pmsg, int cmsWait, bool process_io) {
>+bool MessageQueue::Get(Message* pmsg, int cmsWait, bool process_io) {
>   // Return and clear peek if present
>   // Always return the peek if it exists so there is Peek/Get symmetry
> 
>@@ -367,8 +367,7 @@ bool MessageQueue::Get(Message *pmsg, int cmsWait, bool process_io) {
>   return false;
> }
> 
>-void MessageQueue::ReceiveSends() {
>-}
>+void MessageQueue::ReceiveSends() {}
> 
> void MessageQueue::Post(const Location& posted_from,
>                         MessageHandler* phandler,
>@@ -523,7 +522,7 @@ void MessageQueue::Clear(MessageHandler* phandler,
>   dmsgq_.reheap();
> }
> 
>-void MessageQueue::Dispatch(Message *pmsg) {
>+void MessageQueue::Dispatch(Message* pmsg) {
>   TRACE_EVENT2("webrtc", "MessageQueue::Dispatch", "src_file_and_line",
>                pmsg->posted_from.file_and_line(), "src_func",
>                pmsg->posted_from.function_name());
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagequeue.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagequeue.h
>index 70556438e34..fe64c9c4326 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagequeue.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagequeue.h
>@@ -20,14 +20,13 @@
> #include <utility>
> #include <vector>
> 
>-#include "rtc_base/basictypes.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/location.h"
> #include "rtc_base/messagehandler.h"
> #include "rtc_base/scoped_ref_ptr.h"
>-#include "rtc_base/sigslot.h"
> #include "rtc_base/socketserver.h"
>+#include "rtc_base/third_party/sigslot/sigslot.h"
> #include "rtc_base/thread_annotations.h"
> #include "rtc_base/timeutils.h"
> 
>@@ -40,9 +39,9 @@ class MessageQueue;
> 
> class MessageQueueManager {
>  public:
>-  static void Add(MessageQueue *message_queue);
>-  static void Remove(MessageQueue *message_queue);
>-  static void Clear(MessageHandler *handler);
>+  static void Add(MessageQueue* message_queue);
>+  static void Remove(MessageQueue* message_queue);
>+  static void Clear(MessageHandler* handler);
> 
>   // For testing purposes, we expose whether or not the MessageQueueManager
>   // instance has been initialized. It has no other use relative to the rest of
>@@ -61,9 +60,9 @@ class MessageQueueManager {
>   MessageQueueManager();
>   ~MessageQueueManager();
> 
>-  void AddInternal(MessageQueue *message_queue);
>-  void RemoveInternal(MessageQueue *message_queue);
>-  void ClearInternal(MessageHandler *handler);
>+  void AddInternal(MessageQueue* message_queue);
>+  void RemoveInternal(MessageQueue* message_queue);
>+  void ClearInternal(MessageHandler* handler);
>   void ProcessAllMessageQueuesInternal();
> 
>   static MessageQueueManager* instance_;
>@@ -89,9 +88,10 @@ class MessageData {
> template <class T>
> class TypedMessageData : public MessageData {
>  public:
>-  explicit TypedMessageData(const T& data) : data_(data) { }
>+  explicit TypedMessageData(const T& data) : data_(data) {}
>   const T& data() const { return data_; }
>   T& data() { return data_; }
>+
>  private:
>   T data_;
> };
>@@ -123,28 +123,30 @@ class ScopedMessageData : public MessageData {
> template <class T>
> class ScopedRefMessageData : public MessageData {
>  public:
>-  explicit ScopedRefMessageData(T* data) : data_(data) { }
>+  explicit ScopedRefMessageData(T* data) : data_(data) {}
>   const scoped_refptr<T>& data() const { return data_; }
>   scoped_refptr<T>& data() { return data_; }
>+
>  private:
>   scoped_refptr<T> data_;
> };
> 
>-template<class T>
>+template <class T>
> inline MessageData* WrapMessageData(const T& data) {
>   return new TypedMessageData<T>(data);
> }
> 
>-template<class T>
>+template <class T>
> inline const T& UseMessageData(MessageData* data) {
>-  return static_cast< TypedMessageData<T>* >(data)->data();
>+  return static_cast<TypedMessageData<T>*>(data)->data();
> }
> 
>-template<class T>
>+template <class T>
> class DisposeData : public MessageData {
>  public:
>-  explicit DisposeData(T* data) : data_(data) { }
>+  explicit DisposeData(T* data) : data_(data) {}
>   virtual ~DisposeData() { delete data_; }
>+
>  private:
>   T* data_;
> };
>@@ -162,9 +164,9 @@ struct Message {
>            (id == MQID_ANY || id == message_id);
>   }
>   Location posted_from;
>-  MessageHandler *phandler;
>+  MessageHandler* phandler;
>   uint32_t message_id;
>-  MessageData *pdata;
>+  MessageData* pdata;
>   int64_t ts_sensitive;
> };
> 
>@@ -181,9 +183,9 @@ class DelayedMessage {
>                  const Message& msg)
>       : cmsDelay_(delay), msTrigger_(trigger), num_(num), msg_(msg) {}
> 
>-  bool operator< (const DelayedMessage& dmsg) const {
>-    return (dmsg.msTrigger_ < msTrigger_)
>-           || ((dmsg.msTrigger_ == msTrigger_) && (dmsg.num_ < num_));
>+  bool operator<(const DelayedMessage& dmsg) const {
>+    return (dmsg.msTrigger_ < msTrigger_) ||
>+           ((dmsg.msTrigger_ == msTrigger_) && (dmsg.num_ < num_));
>   }
> 
>   int64_t cmsDelay_;  // for debugging
>@@ -230,9 +232,10 @@ class MessageQueue {
>   //  1) A message is available (returns true)
>   //  2) cmsWait seconds have elapsed (returns false)
>   //  3) Stop() is called (returns false)
>-  virtual bool Get(Message *pmsg, int cmsWait = kForever,
>+  virtual bool Get(Message* pmsg,
>+                   int cmsWait = kForever,
>                    bool process_io = true);
>-  virtual bool Peek(Message *pmsg, int cmsWait = 0);
>+  virtual bool Peek(Message* pmsg, int cmsWait = 0);
>   virtual void Post(const Location& posted_from,
>                     MessageHandler* phandler,
>                     uint32_t id = 0,
>@@ -257,7 +260,7 @@ class MessageQueue {
>   virtual void Clear(MessageHandler* phandler,
>                      uint32_t id = MQID_ANY,
>                      MessageList* removed = nullptr);
>-  virtual void Dispatch(Message *pmsg);
>+  virtual void Dispatch(Message* pmsg);
>   virtual void ReceiveSends();
> 
>   // Amount of time until the next message can be retrieved
>@@ -270,7 +273,8 @@ class MessageQueue {
>   }
> 
>   // Internally posts a message which causes the doomed object to be deleted
>-  template<class T> void Dispose(T* doomed) {
>+  template <class T>
>+  void Dispose(T* doomed) {
>     if (doomed) {
>       Post(RTC_FROM_HERE, nullptr, MQID_DISPOSE, new DisposeData<T>(doomed));
>     }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagequeue_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagequeue_unittest.cc
>index 9e1ba63cf32..1018a62f9f4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagequeue_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagequeue_unittest.cc
>@@ -25,7 +25,7 @@
> 
> using namespace rtc;
> 
>-class MessageQueueTest: public testing::Test, public MessageQueue {
>+class MessageQueueTest : public testing::Test, public MessageQueue {
>  public:
>   MessageQueueTest() : MessageQueue(SocketServer::CreateDefault(), true) {}
>   bool IsLocked_Worker() {
>@@ -47,7 +47,7 @@ class MessageQueueTest: public testing::Test, public MessageQueue {
> 
> struct DeletedLockChecker {
>   DeletedLockChecker(MessageQueueTest* test, bool* was_locked, bool* deleted)
>-      : test(test), was_locked(was_locked), deleted(deleted) { }
>+      : test(test), was_locked(was_locked), deleted(deleted) {}
>   ~DeletedLockChecker() {
>     *deleted = true;
>     *was_locked = test->IsLocked();
>@@ -68,7 +68,7 @@ static void DelayedPostsWithIdenticalTimesAreProcessedInFifoOrder(
>   q->PostAt(RTC_FROM_HERE, now - 1, nullptr, 2);
> 
>   Message msg;
>-  for (size_t i=0; i<5; ++i) {
>+  for (size_t i = 0; i < 5; ++i) {
>     memset(&msg, 0, sizeof(msg));
>     EXPECT_TRUE(q->Get(&msg, 0));
>     EXPECT_EQ(i, msg.message_id);
>@@ -100,7 +100,7 @@ TEST_F(MessageQueueTest, DisposeNotLocked) {
> 
> class DeletedMessageHandler : public MessageHandler {
>  public:
>-  explicit DeletedMessageHandler(bool* deleted) : deleted_(deleted) { }
>+  explicit DeletedMessageHandler(bool* deleted) : deleted_(deleted) {}
>   ~DeletedMessageHandler() override { *deleted_ = true; }
>   void OnMessage(Message* msg) override {}
> 
>@@ -110,7 +110,7 @@ class DeletedMessageHandler : public MessageHandler {
> 
> TEST_F(MessageQueueTest, DiposeHandlerWithPostedMessagePending) {
>   bool deleted = false;
>-  DeletedMessageHandler *handler = new DeletedMessageHandler(&deleted);
>+  DeletedMessageHandler* handler = new DeletedMessageHandler(&deleted);
>   // First, post a dispose.
>   Dispose(handler);
>   // Now, post a message, which should *not* be returned by Get().
>@@ -122,11 +122,14 @@ TEST_F(MessageQueueTest, DiposeHandlerWithPostedMessagePending) {
> 
> struct UnwrapMainThreadScope {
>   UnwrapMainThreadScope() : rewrap_(Thread::Current() != nullptr) {
>-    if (rewrap_) ThreadManager::Instance()->UnwrapCurrentThread();
>+    if (rewrap_)
>+      ThreadManager::Instance()->UnwrapCurrentThread();
>   }
>   ~UnwrapMainThreadScope() {
>-    if (rewrap_) ThreadManager::Instance()->WrapCurrentThread();
>+    if (rewrap_)
>+      ThreadManager::Instance()->WrapCurrentThread();
>   }
>+
>  private:
>   bool rewrap_;
> };
>@@ -218,9 +221,7 @@ TEST(MessageQueueManager, ProcessAllMessageQueuesWithClearedQueue) {
>   MessageQueueManager::ProcessAllMessageQueues();
> }
> 
>-class RefCountedHandler
>-  : public MessageHandler,
>-    public rtc::RefCountInterface {
>+class RefCountedHandler : public MessageHandler, public rtc::RefCountInterface {
>  public:
>   void OnMessage(Message* msg) override {}
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/nat_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/nat_unittest.cc
>index d68df1d1508..ee944272b70 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/nat_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/nat_unittest.cc
>@@ -12,6 +12,7 @@
> #include <memory>
> #include <string>
> 
>+#include "absl/memory/memory.h"
> #include "rtc_base/asynctcpsocket.h"
> #include "rtc_base/gunit.h"
> #include "rtc_base/logging.h"
>@@ -20,36 +21,38 @@
> #include "rtc_base/nethelpers.h"
> #include "rtc_base/network.h"
> #include "rtc_base/physicalsocketserver.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/testclient.h"
> #include "rtc_base/virtualsocketserver.h"
> 
> using namespace rtc;
> 
>-bool CheckReceive(
>-    TestClient* client, bool should_receive, const char* buf, size_t size) {
>-  return (should_receive) ?
>-      client->CheckNextPacket(buf, size, 0) :
>-      client->CheckNoPacket();
>+bool CheckReceive(TestClient* client,
>+                  bool should_receive,
>+                  const char* buf,
>+                  size_t size) {
>+  return (should_receive) ? client->CheckNextPacket(buf, size, 0)
>+                          : client->CheckNoPacket();
> }
> 
>-TestClient* CreateTestClient(
>-      SocketFactory* factory, const SocketAddress& local_addr) {
>+TestClient* CreateTestClient(SocketFactory* factory,
>+                             const SocketAddress& local_addr) {
>   return new TestClient(
>-      WrapUnique(AsyncUDPSocket::Create(factory, local_addr)));
>+      absl::WrapUnique(AsyncUDPSocket::Create(factory, local_addr)));
> }
> 
> TestClient* CreateTCPTestClient(AsyncSocket* socket) {
>-  return new TestClient(MakeUnique<AsyncTCPSocket>(socket, false));
>+  return new TestClient(absl::make_unique<AsyncTCPSocket>(socket, false));
> }
> 
> // Tests that when sending from internal_addr to external_addrs through the
> // NAT type specified by nat_type, all external addrs receive the sent packet
> // and, if exp_same is true, all use the same mapped-address on the NAT.
>-void TestSend(
>-      SocketServer* internal, const SocketAddress& internal_addr,
>-      SocketServer* external, const SocketAddress external_addrs[4],
>-      NATType nat_type, bool exp_same) {
>+void TestSend(SocketServer* internal,
>+              const SocketAddress& internal_addr,
>+              SocketServer* external,
>+              const SocketAddress external_addrs[4],
>+              NATType nat_type,
>+              bool exp_same) {
>   Thread th_int(internal);
>   Thread th_ext(external);
> 
>@@ -57,9 +60,8 @@ void TestSend(
>   server_addr.SetPort(0);  // Auto-select a port
>   NATServer* nat = new NATServer(nat_type, internal, server_addr, server_addr,
>                                  external, external_addrs[0]);
>-  NATSocketFactory* natsf = new NATSocketFactory(internal,
>-                                                 nat->internal_udp_address(),
>-                                                 nat->internal_tcp_address());
>+  NATSocketFactory* natsf = new NATSocketFactory(
>+      internal, nat->internal_udp_address(), nat->internal_tcp_address());
> 
>   TestClient* in = CreateTestClient(natsf, internal_addr);
>   TestClient* out[4];
>@@ -98,10 +100,13 @@ void TestSend(
> 
> // Tests that when sending from external_addrs to internal_addr, the packet
> // is delivered according to the specified filter_ip and filter_port rules.
>-void TestRecv(
>-      SocketServer* internal, const SocketAddress& internal_addr,
>-      SocketServer* external, const SocketAddress external_addrs[4],
>-      NATType nat_type, bool filter_ip, bool filter_port) {
>+void TestRecv(SocketServer* internal,
>+              const SocketAddress& internal_addr,
>+              SocketServer* external,
>+              const SocketAddress external_addrs[4],
>+              NATType nat_type,
>+              bool filter_ip,
>+              bool filter_port) {
>   Thread th_int(internal);
>   Thread th_ext(external);
> 
>@@ -109,9 +114,8 @@ void TestRecv(
>   server_addr.SetPort(0);  // Auto-select a port
>   NATServer* nat = new NATServer(nat_type, internal, server_addr, server_addr,
>                                  external, external_addrs[0]);
>-  NATSocketFactory* natsf = new NATSocketFactory(internal,
>-                                                 nat->internal_udp_address(),
>-                                                 nat->internal_tcp_address());
>+  NATSocketFactory* natsf = new NATSocketFactory(
>+      internal, nat->internal_udp_address(), nat->internal_tcp_address());
> 
>   TestClient* in = CreateTestClient(natsf, internal_addr);
>   TestClient* out[4];
>@@ -148,31 +152,33 @@ void TestRecv(
> }
> 
> // Tests that NATServer allocates bindings properly.
>-void TestBindings(
>-    SocketServer* internal, const SocketAddress& internal_addr,
>-    SocketServer* external, const SocketAddress external_addrs[4]) {
>-  TestSend(internal, internal_addr, external, external_addrs,
>-           NAT_OPEN_CONE, true);
>+void TestBindings(SocketServer* internal,
>+                  const SocketAddress& internal_addr,
>+                  SocketServer* external,
>+                  const SocketAddress external_addrs[4]) {
>+  TestSend(internal, internal_addr, external, external_addrs, NAT_OPEN_CONE,
>+           true);
>   TestSend(internal, internal_addr, external, external_addrs,
>            NAT_ADDR_RESTRICTED, true);
>   TestSend(internal, internal_addr, external, external_addrs,
>            NAT_PORT_RESTRICTED, true);
>-  TestSend(internal, internal_addr, external, external_addrs,
>-           NAT_SYMMETRIC, false);
>+  TestSend(internal, internal_addr, external, external_addrs, NAT_SYMMETRIC,
>+           false);
> }
> 
> // Tests that NATServer filters packets properly.
>-void TestFilters(
>-    SocketServer* internal, const SocketAddress& internal_addr,
>-    SocketServer* external, const SocketAddress external_addrs[4]) {
>-  TestRecv(internal, internal_addr, external, external_addrs,
>-           NAT_OPEN_CONE, false, false);
>+void TestFilters(SocketServer* internal,
>+                 const SocketAddress& internal_addr,
>+                 SocketServer* external,
>+                 const SocketAddress external_addrs[4]) {
>+  TestRecv(internal, internal_addr, external, external_addrs, NAT_OPEN_CONE,
>+           false, false);
>   TestRecv(internal, internal_addr, external, external_addrs,
>            NAT_ADDR_RESTRICTED, true, false);
>   TestRecv(internal, internal_addr, external, external_addrs,
>            NAT_PORT_RESTRICTED, true, true);
>-  TestRecv(internal, internal_addr, external, external_addrs,
>-           NAT_SYMMETRIC, true, true);
>+  TestRecv(internal, internal_addr, external, external_addrs, NAT_SYMMETRIC,
>+           true, true);
> }
> 
> bool TestConnectivity(const SocketAddress& src, const IPAddress& dst) {
>@@ -224,7 +230,7 @@ void TestPhysicalInternal(const SocketAddress& int_addr) {
>   // Find an available IP with matching family. The test breaks if int_addr
>   // can't talk to ip, so check for connectivity as well.
>   for (std::vector<Network*>::iterator it = networks.begin();
>-      it != networks.end(); ++it) {
>+       it != networks.end(); ++it) {
>     const IPAddress& ip = (*it)->GetBestIP();
>     if (ip.family() == int_addr.family() && TestConnectivity(int_addr, ip)) {
>       ext_addr2.SetIP(ip);
>@@ -232,18 +238,16 @@ void TestPhysicalInternal(const SocketAddress& int_addr) {
>     }
>   }
>   if (ext_addr2.IsNil()) {
>-    RTC_LOG(LS_WARNING) << "No available IP of same family as " << int_addr;
>+    RTC_LOG(LS_WARNING) << "No available IP of same family as "
>+                        << int_addr.ToString();
>     return;
>   }
> 
>-  RTC_LOG(LS_INFO) << "selected ip " << ext_addr2.ipaddr();
>+  RTC_LOG(LS_INFO) << "selected ip " << ext_addr2.ipaddr().ToString();
> 
>   SocketAddress ext_addrs[4] = {
>-      SocketAddress(ext_addr1),
>-      SocketAddress(ext_addr2),
>-      SocketAddress(ext_addr1),
>-      SocketAddress(ext_addr2)
>-  };
>+      SocketAddress(ext_addr1), SocketAddress(ext_addr2),
>+      SocketAddress(ext_addr1), SocketAddress(ext_addr2)};
> 
>   std::unique_ptr<PhysicalSocketServer> int_pss(new PhysicalSocketServer());
>   std::unique_ptr<PhysicalSocketServer> ext_pss(new PhysicalSocketServer());
>@@ -327,16 +331,13 @@ class NatTcpTest : public testing::Test, public sigslot::has_slots<> {
>     ext_thread_->Start();
>   }
> 
>-  void OnConnectEvent(AsyncSocket* socket) {
>-    connected_ = true;
>-  }
>+  void OnConnectEvent(AsyncSocket* socket) { connected_ = true; }
> 
>   void OnAcceptEvent(AsyncSocket* socket) {
>     accepted_.reset(server_->Accept(nullptr));
>   }
> 
>-  void OnCloseEvent(AsyncSocket* socket, int error) {
>-  }
>+  void OnCloseEvent(AsyncSocket* socket, int error) {}
> 
>   void ConnectEvents() {
>     server_->SignalReadEvent.connect(this, &NatTcpTest::OnAcceptEvent);
>@@ -358,11 +359,11 @@ class NatTcpTest : public testing::Test, public sigslot::has_slots<> {
> };
> 
> TEST_F(NatTcpTest, DISABLED_TestConnectOut) {
>-  server_.reset(ext_vss_->CreateAsyncSocket(SOCK_STREAM));
>+  server_.reset(ext_vss_->CreateAsyncSocket(AF_INET, SOCK_STREAM));
>   server_->Bind(ext_addr_);
>   server_->Listen(5);
> 
>-  client_.reset(natsf_->CreateAsyncSocket(SOCK_STREAM));
>+  client_.reset(natsf_->CreateAsyncSocket(AF_INET, SOCK_STREAM));
>   EXPECT_GE(0, client_->Bind(int_addr_));
>   EXPECT_GE(0, client_->Connect(server_->GetLocalAddress()));
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/natserver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/natserver.cc
>index bf983fedbff..81193761a63 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/natserver.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/natserver.cc
>@@ -18,8 +18,7 @@
> 
> namespace rtc {
> 
>-RouteCmp::RouteCmp(NAT* nat) : symmetric(nat->IsSymmetric()) {
>-}
>+RouteCmp::RouteCmp(NAT* nat) : symmetric(nat->IsSymmetric()) {}
> 
> size_t RouteCmp::operator()(const SocketAddressPair& r) const {
>   size_t h = r.source().Hash();
>@@ -28,8 +27,8 @@ size_t RouteCmp::operator()(const SocketAddressPair& r) const {
>   return h;
> }
> 
>-bool RouteCmp::operator()(
>-      const SocketAddressPair& r1, const SocketAddressPair& r2) const {
>+bool RouteCmp::operator()(const SocketAddressPair& r1,
>+                          const SocketAddressPair& r2) const {
>   if (r1.source() < r2.source())
>     return true;
>   if (r2.source() < r1.source())
>@@ -42,8 +41,7 @@ bool RouteCmp::operator()(
> }
> 
> AddrCmp::AddrCmp(NAT* nat)
>-    : use_ip(nat->FiltersIP()), use_port(nat->FiltersPort()) {
>-}
>+    : use_ip(nat->FiltersIP()), use_port(nat->FiltersPort()) {}
> 
> size_t AddrCmp::operator()(const SocketAddress& a) const {
>   size_t h = 0;
>@@ -54,8 +52,8 @@ size_t AddrCmp::operator()(const SocketAddress& a) const {
>   return h;
> }
> 
>-bool AddrCmp::operator()(
>-      const SocketAddress& a1, const SocketAddress& a2) const {
>+bool AddrCmp::operator()(const SocketAddress& a1,
>+                         const SocketAddress& a2) const {
>   if (use_ip && (a1.ipaddr() < a2.ipaddr()))
>     return true;
>   if (use_ip && (a2.ipaddr() < a1.ipaddr()))
>@@ -109,15 +107,15 @@ class NATProxyServerSocket : public AsyncProxyServerSocket {
>       SignalReadEvent(this);
>     }
>   }
>-
> };
> 
> class NATProxyServer : public ProxyServer {
>  public:
>-  NATProxyServer(SocketFactory* int_factory, const SocketAddress& int_addr,
>-                 SocketFactory* ext_factory, const SocketAddress& ext_ip)
>-      : ProxyServer(int_factory, int_addr, ext_factory, ext_ip) {
>-  }
>+  NATProxyServer(SocketFactory* int_factory,
>+                 const SocketAddress& int_addr,
>+                 SocketFactory* ext_factory,
>+                 const SocketAddress& ext_ip)
>+      : ProxyServer(int_factory, int_addr, ext_factory, ext_ip) {}
> 
>  protected:
>   AsyncProxyServerSocket* WrapSocket(AsyncSocket* socket) override {
>@@ -125,27 +123,27 @@ class NATProxyServer : public ProxyServer {
>   }
> };
> 
>-NATServer::NATServer(
>-    NATType type, SocketFactory* internal,
>-    const SocketAddress& internal_udp_addr,
>-    const SocketAddress& internal_tcp_addr,
>-    SocketFactory* external, const SocketAddress& external_ip)
>+NATServer::NATServer(NATType type,
>+                     SocketFactory* internal,
>+                     const SocketAddress& internal_udp_addr,
>+                     const SocketAddress& internal_tcp_addr,
>+                     SocketFactory* external,
>+                     const SocketAddress& external_ip)
>     : external_(external), external_ip_(external_ip.ipaddr(), 0) {
>   nat_ = NAT::Create(type);
> 
>   udp_server_socket_ = AsyncUDPSocket::Create(internal, internal_udp_addr);
>   udp_server_socket_->SignalReadPacket.connect(this,
>                                                &NATServer::OnInternalUDPPacket);
>-  tcp_proxy_server_ = new NATProxyServer(internal, internal_tcp_addr, external,
>-                                         external_ip);
>+  tcp_proxy_server_ =
>+      new NATProxyServer(internal, internal_tcp_addr, external, external_ip);
> 
>   int_map_ = new InternalMap(RouteCmp(nat_));
>   ext_map_ = new ExternalMap();
> }
> 
> NATServer::~NATServer() {
>-  for (InternalMap::iterator iter = int_map_->begin();
>-       iter != int_map_->end();
>+  for (InternalMap::iterator iter = int_map_->begin(); iter != int_map_->end();
>        iter++)
>     delete iter->second;
> 
>@@ -156,9 +154,11 @@ NATServer::~NATServer() {
>   delete ext_map_;
> }
> 
>-void NATServer::OnInternalUDPPacket(
>-    AsyncPacketSocket* socket, const char* buf, size_t size,
>-    const SocketAddress& addr, const PacketTime& packet_time) {
>+void NATServer::OnInternalUDPPacket(AsyncPacketSocket* socket,
>+                                    const char* buf,
>+                                    size_t size,
>+                                    const SocketAddress& addr,
>+                                    const PacketTime& packet_time) {
>   // Read the intended destination from the wire.
>   SocketAddress dest_addr;
>   size_t length = UnpackAddressFromNAT(buf, size, &dest_addr);
>@@ -180,9 +180,11 @@ void NATServer::OnInternalUDPPacket(
>   iter->second->socket->SendTo(buf + length, size - length, dest_addr, options);
> }
> 
>-void NATServer::OnExternalUDPPacket(
>-    AsyncPacketSocket* socket, const char* buf, size_t size,
>-    const SocketAddress& remote_addr, const PacketTime& packet_time) {
>+void NATServer::OnExternalUDPPacket(AsyncPacketSocket* socket,
>+                                    const char* buf,
>+                                    size_t size,
>+                                    const SocketAddress& remote_addr,
>+                                    const PacketTime& packet_time) {
>   SocketAddress local_addr = socket->GetLocalAddress();
> 
>   // Find the translation for this addresses.
>@@ -199,9 +201,8 @@ void NATServer::OnExternalUDPPacket(
>   // Forward this packet to the internal address.
>   // First prepend the address in a quasi-STUN format.
>   std::unique_ptr<char[]> real_buf(new char[size + kNATEncodedIPv6AddressSize]);
>-  size_t addrlength = PackAddressForNAT(real_buf.get(),
>-                                        size + kNATEncodedIPv6AddressSize,
>-                                        remote_addr);
>+  size_t addrlength = PackAddressForNAT(
>+      real_buf.get(), size + kNATEncodedIPv6AddressSize, remote_addr);
>   // Copy the data part after the address.
>   rtc::PacketOptions options;
>   memcpy(real_buf.get() + addrlength, buf, size);
>@@ -228,8 +229,9 @@ bool NATServer::ShouldFilterOut(TransEntry* entry,
>   return entry->WhitelistContains(ext_addr);
> }
> 
>-NATServer::TransEntry::TransEntry(
>-    const SocketAddressPair& r, AsyncUDPSocket* s, NAT* nat)
>+NATServer::TransEntry::TransEntry(const SocketAddressPair& r,
>+                                  AsyncUDPSocket* s,
>+                                  NAT* nat)
>     : route(r), socket(s) {
>   whitelist = new AddressSet(AddrCmp(nat));
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/natserver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/natserver.h
>index a7b4d62520f..f4cabcf0d9b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/natserver.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/natserver.h
>@@ -30,8 +30,8 @@ namespace rtc {
> struct RouteCmp {
>   explicit RouteCmp(NAT* nat);
>   size_t operator()(const SocketAddressPair& r) const;
>-  bool operator()(
>-      const SocketAddressPair& r1, const SocketAddressPair& r2) const;
>+  bool operator()(const SocketAddressPair& r1,
>+                  const SocketAddressPair& r2) const;
> 
>   bool symmetric;
> };
>@@ -60,11 +60,12 @@ const int NAT_SERVER_TCP_PORT = 4238;
> 
> class NATServer : public sigslot::has_slots<> {
>  public:
>-  NATServer(
>-      NATType type, SocketFactory* internal,
>-      const SocketAddress& internal_udp_addr,
>-      const SocketAddress& internal_tcp_addr,
>-      SocketFactory* external, const SocketAddress& external_ip);
>+  NATServer(NATType type,
>+            SocketFactory* internal,
>+            const SocketAddress& internal_udp_addr,
>+            const SocketAddress& internal_tcp_addr,
>+            SocketFactory* external,
>+            const SocketAddress& external_ip);
>   ~NATServer() override;
> 
>   SocketAddress internal_udp_address() const {
>@@ -76,11 +77,15 @@ class NATServer : public sigslot::has_slots<> {
>   }
> 
>   // Packets received on one of the networks.
>-  void OnInternalUDPPacket(AsyncPacketSocket* socket, const char* buf,
>-                           size_t size, const SocketAddress& addr,
>+  void OnInternalUDPPacket(AsyncPacketSocket* socket,
>+                           const char* buf,
>+                           size_t size,
>+                           const SocketAddress& addr,
>                            const PacketTime& packet_time);
>-  void OnExternalUDPPacket(AsyncPacketSocket* socket, const char* buf,
>-                           size_t size, const SocketAddress& remote_addr,
>+  void OnExternalUDPPacket(AsyncPacketSocket* socket,
>+                           const char* buf,
>+                           size_t size,
>+                           const SocketAddress& remote_addr,
>                            const PacketTime& packet_time);
> 
>  private:
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/natsocketfactory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/natsocketfactory.cc
>index dd4c030bcbc..a707ce02598 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/natsocketfactory.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/natsocketfactory.cc
>@@ -21,7 +21,8 @@ namespace rtc {
> // Packs the given socketaddress into the buffer in buf, in the quasi-STUN
> // format that the natserver uses.
> // Returns 0 if an invalid address is passed.
>-size_t PackAddressForNAT(char* buf, size_t buf_size,
>+size_t PackAddressForNAT(char* buf,
>+                         size_t buf_size,
>                          const SocketAddress& remote_addr) {
>   const IPAddress& ip = remote_addr.ipaddr();
>   int family = ip.family();
>@@ -46,7 +47,8 @@ size_t PackAddressForNAT(char* buf, size_t buf_size,
> // Decodes the remote address from a packet that has been encoded with the nat's
> // quasi-STUN format. Returns the length of the address (i.e., the offset into
> // data where the original packet starts).
>-size_t UnpackAddressFromNAT(const char* buf, size_t buf_size,
>+size_t UnpackAddressFromNAT(const char* buf,
>+                            size_t buf_size,
>                             SocketAddress* remote_addr) {
>   RTC_DCHECK(buf_size >= 8);
>   RTC_DCHECK(buf[0] == 0);
>@@ -66,7 +68,6 @@ size_t UnpackAddressFromNAT(const char* buf, size_t buf_size,
>   return 0U;
> }
> 
>-
> // NATSocket
> class NATSocket : public AsyncSocket, public sigslot::has_slots<> {
>  public:
>@@ -138,9 +139,8 @@ class NATSocket : public AsyncSocket, public sigslot::has_slots<> {
>     }
>     // This array will be too large for IPv4 packets, but only by 12 bytes.
>     std::unique_ptr<char[]> buf(new char[size + kNATEncodedIPv6AddressSize]);
>-    size_t addrlength = PackAddressForNAT(buf.get(),
>-                                          size + kNATEncodedIPv6AddressSize,
>-                                          addr);
>+    size_t addrlength =
>+        PackAddressForNAT(buf.get(), size + kNATEncodedIPv6AddressSize, addr);
>     size_t encoded_size = size + addrlength;
>     memcpy(buf.get() + addrlength, data, size);
>     int result = socket_->SendTo(buf.get(), encoded_size, server_addr_);
>@@ -331,28 +331,23 @@ class NATSocket : public AsyncSocket, public sigslot::has_slots<> {
> NATSocketFactory::NATSocketFactory(SocketFactory* factory,
>                                    const SocketAddress& nat_udp_addr,
>                                    const SocketAddress& nat_tcp_addr)
>-    : factory_(factory), nat_udp_addr_(nat_udp_addr),
>-      nat_tcp_addr_(nat_tcp_addr) {
>-}
>-
>-Socket* NATSocketFactory::CreateSocket(int type) {
>-  return CreateSocket(AF_INET, type);
>-}
>+    : factory_(factory),
>+      nat_udp_addr_(nat_udp_addr),
>+      nat_tcp_addr_(nat_tcp_addr) {}
> 
> Socket* NATSocketFactory::CreateSocket(int family, int type) {
>   return new NATSocket(this, family, type);
> }
> 
>-AsyncSocket* NATSocketFactory::CreateAsyncSocket(int type) {
>-  return CreateAsyncSocket(AF_INET, type);
>-}
>-
> AsyncSocket* NATSocketFactory::CreateAsyncSocket(int family, int type) {
>   return new NATSocket(this, family, type);
> }
> 
>-AsyncSocket* NATSocketFactory::CreateInternalSocket(int family, int type,
>-    const SocketAddress& local_addr, SocketAddress* nat_addr) {
>+AsyncSocket* NATSocketFactory::CreateInternalSocket(
>+    int family,
>+    int type,
>+    const SocketAddress& local_addr,
>+    SocketAddress* nat_addr) {
>   if (type == SOCK_STREAM) {
>     *nat_addr = nat_tcp_addr_;
>   } else {
>@@ -371,7 +366,9 @@ NATSocketServer::Translator* NATSocketServer::GetTranslator(
> }
> 
> NATSocketServer::Translator* NATSocketServer::AddTranslator(
>-    const SocketAddress& ext_ip, const SocketAddress& int_ip, NATType type) {
>+    const SocketAddress& ext_ip,
>+    const SocketAddress& int_ip,
>+    NATType type) {
>   // Fail if a translator already exists with this extternal address.
>   if (nats_.Get(ext_ip))
>     return nullptr;
>@@ -379,23 +376,14 @@ NATSocketServer::Translator* NATSocketServer::AddTranslator(
>   return nats_.Add(ext_ip, new Translator(this, type, int_ip, server_, ext_ip));
> }
> 
>-void NATSocketServer::RemoveTranslator(
>-    const SocketAddress& ext_ip) {
>+void NATSocketServer::RemoveTranslator(const SocketAddress& ext_ip) {
>   nats_.Remove(ext_ip);
> }
> 
>-Socket* NATSocketServer::CreateSocket(int type) {
>-  return CreateSocket(AF_INET, type);
>-}
>-
> Socket* NATSocketServer::CreateSocket(int family, int type) {
>   return new NATSocket(this, family, type);
> }
> 
>-AsyncSocket* NATSocketServer::CreateAsyncSocket(int type) {
>-  return CreateAsyncSocket(AF_INET, type);
>-}
>-
> AsyncSocket* NATSocketServer::CreateAsyncSocket(int family, int type) {
>   return new NATSocket(this, family, type);
> }
>@@ -413,14 +401,17 @@ void NATSocketServer::WakeUp() {
>   server_->WakeUp();
> }
> 
>-AsyncSocket* NATSocketServer::CreateInternalSocket(int family, int type,
>-    const SocketAddress& local_addr, SocketAddress* nat_addr) {
>+AsyncSocket* NATSocketServer::CreateInternalSocket(
>+    int family,
>+    int type,
>+    const SocketAddress& local_addr,
>+    SocketAddress* nat_addr) {
>   AsyncSocket* socket = nullptr;
>   Translator* nat = nats_.FindClient(local_addr);
>   if (nat) {
>     socket = nat->internal_factory()->CreateAsyncSocket(family, type);
>-    *nat_addr = (type == SOCK_STREAM) ?
>-        nat->internal_tcp_address() : nat->internal_udp_address();
>+    *nat_addr = (type == SOCK_STREAM) ? nat->internal_tcp_address()
>+                                      : nat->internal_udp_address();
>   } else {
>     socket = server_->CreateAsyncSocket(family, type);
>   }
>@@ -428,9 +419,11 @@ AsyncSocket* NATSocketServer::CreateInternalSocket(int family, int type,
> }
> 
> // NATSocketServer::Translator
>-NATSocketServer::Translator::Translator(
>-    NATSocketServer* server, NATType type, const SocketAddress& int_ip,
>-    SocketFactory* ext_factory, const SocketAddress& ext_ip)
>+NATSocketServer::Translator::Translator(NATSocketServer* server,
>+                                        NATType type,
>+                                        const SocketAddress& int_ip,
>+                                        SocketFactory* ext_factory,
>+                                        const SocketAddress& ext_ip)
>     : server_(server) {
>   // Create a new private network, and a NATServer running on the private
>   // network that bridges to the external network. Also tell the private
>@@ -450,7 +443,9 @@ NATSocketServer::Translator* NATSocketServer::Translator::GetTranslator(
> }
> 
> NATSocketServer::Translator* NATSocketServer::Translator::AddTranslator(
>-    const SocketAddress& ext_ip, const SocketAddress& int_ip, NATType type) {
>+    const SocketAddress& ext_ip,
>+    const SocketAddress& int_ip,
>+    NATType type) {
>   // Fail if a translator already exists with this extternal address.
>   if (nats_.Get(ext_ip))
>     return nullptr;
>@@ -465,8 +460,7 @@ void NATSocketServer::Translator::RemoveTranslator(
>   RemoveClient(ext_ip);
> }
> 
>-bool NATSocketServer::Translator::AddClient(
>-    const SocketAddress& int_ip) {
>+bool NATSocketServer::Translator::AddClient(const SocketAddress& int_ip) {
>   // Fail if a client already exists with this internal address.
>   if (clients_.find(int_ip) != clients_.end())
>     return false;
>@@ -475,8 +469,7 @@ bool NATSocketServer::Translator::AddClient(
>   return true;
> }
> 
>-void NATSocketServer::Translator::RemoveClient(
>-    const SocketAddress& int_ip) {
>+void NATSocketServer::Translator::RemoveClient(const SocketAddress& int_ip) {
>   std::set<SocketAddress>::iterator it = clients_.find(int_ip);
>   if (it != clients_.end()) {
>     clients_.erase(it);
>@@ -486,8 +479,8 @@ void NATSocketServer::Translator::RemoveClient(
> NATSocketServer::Translator* NATSocketServer::Translator::FindClient(
>     const SocketAddress& int_ip) {
>   // See if we have the requested IP, or any of our children do.
>-  return (clients_.find(int_ip) != clients_.end()) ?
>-      this : nats_.FindClient(int_ip);
>+  return (clients_.find(int_ip) != clients_.end()) ? this
>+                                                   : nats_.FindClient(int_ip);
> }
> 
> // NATSocketServer::TranslatorMap
>@@ -504,13 +497,13 @@ NATSocketServer::Translator* NATSocketServer::TranslatorMap::Get(
> }
> 
> NATSocketServer::Translator* NATSocketServer::TranslatorMap::Add(
>-    const SocketAddress& ext_ip, Translator* nat) {
>+    const SocketAddress& ext_ip,
>+    Translator* nat) {
>   (*this)[ext_ip] = nat;
>   return nat;
> }
> 
>-void NATSocketServer::TranslatorMap::Remove(
>-    const SocketAddress& ext_ip) {
>+void NATSocketServer::TranslatorMap::Remove(const SocketAddress& ext_ip) {
>   TranslatorMap::iterator it = find(ext_ip);
>   if (it != end()) {
>     delete it->second;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/natsocketfactory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/natsocketfactory.h
>index 319545c1a50..97961d4f9ec 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/natsocketfactory.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/natsocketfactory.h
>@@ -11,10 +11,10 @@
> #ifndef RTC_BASE_NATSOCKETFACTORY_H_
> #define RTC_BASE_NATSOCKETFACTORY_H_
> 
>-#include <string>
> #include <map>
> #include <memory>
> #include <set>
>+#include <string>
> 
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/natserver.h"
>@@ -30,8 +30,10 @@ const size_t kNATEncodedIPv6AddressSize = 20U;
> class NATInternalSocketFactory {
>  public:
>   virtual ~NATInternalSocketFactory() {}
>-  virtual AsyncSocket* CreateInternalSocket(int family, int type,
>-      const SocketAddress& local_addr, SocketAddress* nat_addr) = 0;
>+  virtual AsyncSocket* CreateInternalSocket(int family,
>+                                            int type,
>+                                            const SocketAddress& local_addr,
>+                                            SocketAddress* nat_addr) = 0;
> };
> 
> // Creates sockets that will send all traffic through a NAT, using an existing
>@@ -39,13 +41,12 @@ class NATInternalSocketFactory {
> // from a socket factory, given to the constructor.
> class NATSocketFactory : public SocketFactory, public NATInternalSocketFactory {
>  public:
>-  NATSocketFactory(SocketFactory* factory, const SocketAddress& nat_udp_addr,
>+  NATSocketFactory(SocketFactory* factory,
>+                   const SocketAddress& nat_udp_addr,
>                    const SocketAddress& nat_tcp_addr);
> 
>   // SocketFactory implementation
>-  Socket* CreateSocket(int type) override;
>   Socket* CreateSocket(int family, int type) override;
>-  AsyncSocket* CreateAsyncSocket(int type) override;
>   AsyncSocket* CreateAsyncSocket(int family, int type) override;
> 
>   // NATInternalSocketFactory implementation
>@@ -92,8 +93,10 @@ class NATSocketServer : public SocketServer, public NATInternalSocketFactory {
>   // a specific NAT
>   class Translator {
>    public:
>-    Translator(NATSocketServer* server, NATType type,
>-               const SocketAddress& int_addr, SocketFactory* ext_factory,
>+    Translator(NATSocketServer* server,
>+               NATType type,
>+               const SocketAddress& int_addr,
>+               SocketFactory* ext_factory,
>                const SocketAddress& ext_addr);
>     ~Translator();
> 
>@@ -107,7 +110,8 @@ class NATSocketServer : public SocketServer, public NATInternalSocketFactory {
> 
>     Translator* GetTranslator(const SocketAddress& ext_ip);
>     Translator* AddTranslator(const SocketAddress& ext_ip,
>-                              const SocketAddress& int_ip, NATType type);
>+                              const SocketAddress& int_ip,
>+                              NATType type);
>     void RemoveTranslator(const SocketAddress& ext_ip);
> 
>     bool AddClient(const SocketAddress& int_ip);
>@@ -131,14 +135,12 @@ class NATSocketServer : public SocketServer, public NATInternalSocketFactory {
> 
>   Translator* GetTranslator(const SocketAddress& ext_ip);
>   Translator* AddTranslator(const SocketAddress& ext_ip,
>-                            const SocketAddress& int_ip, NATType type);
>+                            const SocketAddress& int_ip,
>+                            NATType type);
>   void RemoveTranslator(const SocketAddress& ext_ip);
> 
>   // SocketServer implementation
>-  Socket* CreateSocket(int type) override;
>   Socket* CreateSocket(int family, int type) override;
>-
>-  AsyncSocket* CreateAsyncSocket(int type) override;
>   AsyncSocket* CreateAsyncSocket(int family, int type) override;
> 
>   void SetMessageQueue(MessageQueue* queue) override;
>@@ -159,9 +161,11 @@ class NATSocketServer : public SocketServer, public NATInternalSocketFactory {
> };
> 
> // Free-standing NAT helper functions.
>-size_t PackAddressForNAT(char* buf, size_t buf_size,
>+size_t PackAddressForNAT(char* buf,
>+                         size_t buf_size,
>                          const SocketAddress& remote_addr);
>-size_t UnpackAddressFromNAT(const char* buf, size_t buf_size,
>+size_t UnpackAddressFromNAT(const char* buf,
>+                            size_t buf_size,
>                             SocketAddress* remote_addr);
> }  // namespace rtc
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/nattypes.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/nattypes.cc
>index 29936ad848b..a7af57d8c57 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/nattypes.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/nattypes.cc
>@@ -15,31 +15,31 @@
> namespace rtc {
> 
> class SymmetricNAT : public NAT {
>-public:
>- bool IsSymmetric() override { return true; }
>- bool FiltersIP() override { return true; }
>- bool FiltersPort() override { return true; }
>+ public:
>+  bool IsSymmetric() override { return true; }
>+  bool FiltersIP() override { return true; }
>+  bool FiltersPort() override { return true; }
> };
> 
> class OpenConeNAT : public NAT {
>-public:
>- bool IsSymmetric() override { return false; }
>- bool FiltersIP() override { return false; }
>- bool FiltersPort() override { return false; }
>+ public:
>+  bool IsSymmetric() override { return false; }
>+  bool FiltersIP() override { return false; }
>+  bool FiltersPort() override { return false; }
> };
> 
> class AddressRestrictedNAT : public NAT {
>-public:
>- bool IsSymmetric() override { return false; }
>- bool FiltersIP() override { return true; }
>- bool FiltersPort() override { return false; }
>+ public:
>+  bool IsSymmetric() override { return false; }
>+  bool FiltersIP() override { return true; }
>+  bool FiltersPort() override { return false; }
> };
> 
> class PortRestrictedNAT : public NAT {
>-public:
>- bool IsSymmetric() override { return false; }
>- bool FiltersIP() override { return true; }
>- bool FiltersPort() override { return true; }
>+ public:
>+  bool IsSymmetric() override { return false; }
>+  bool FiltersIP() override { return true; }
>+  bool FiltersPort() override { return true; }
> };
> 
> NAT* NAT::Create(NATType type) {
>@@ -58,4 +58,4 @@ NAT* NAT::Create(NATType type) {
>   }
> }
> 
>-} // namespace rtc
>+}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/nattypes.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/nattypes.h
>index 64b36d3f851..1d816ac57e4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/nattypes.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/nattypes.h
>@@ -23,8 +23,8 @@ enum NATType {
> 
> // Implements the rules for each specific type of NAT.
> class NAT {
>-public:
>-  virtual ~NAT() { }
>+ public:
>+  virtual ~NAT() {}
> 
>   // Determines whether this NAT uses both source and destination address when
>   // checking whether a mapping already exists.
>@@ -42,6 +42,6 @@ public:
>   static NAT* Create(NATType type);
> };
> 
>-} // namespace rtc
>+}  // namespace rtc
> 
>-#endif // RTC_BASE_NATTYPES_H_
>+#endif  // RTC_BASE_NATTYPES_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/nethelpers.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/nethelpers.cc
>index 4702de71e4d..b1221c3d001 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/nethelpers.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/nethelpers.cc
>@@ -32,13 +32,14 @@
> 
> namespace rtc {
> 
>-int ResolveHostname(const std::string& hostname, int family,
>+int ResolveHostname(const std::string& hostname,
>+                    int family,
>                     std::vector<IPAddress>* addresses) {
> #ifdef __native_client__
>   RTC_NOTREACHED();
>   RTC_LOG(LS_WARNING) << "ResolveHostname() is not implemented for NaCl";
>   return -1;
>-#else  // __native_client__
>+#else   // __native_client__
>   if (!addresses) {
>     return -1;
>   }
>@@ -83,8 +84,7 @@ int ResolveHostname(const std::string& hostname, int family,
> }
> 
> // AsyncResolver
>-AsyncResolver::AsyncResolver()
>-    : SignalThread(false /* use_socket_server */), error_(-1) {}
>+AsyncResolver::AsyncResolver() : SignalThread(), error_(-1) {}
> 
> AsyncResolver::~AsyncResolver() = default;
> 
>@@ -117,15 +117,15 @@ void AsyncResolver::Destroy(bool wait) {
> }
> 
> void AsyncResolver::DoWork() {
>-  error_ = ResolveHostname(addr_.hostname().c_str(), addr_.family(),
>-                           &addresses_);
>+  error_ =
>+      ResolveHostname(addr_.hostname().c_str(), addr_.family(), &addresses_);
> }
> 
> void AsyncResolver::OnWorkDone() {
>   SignalDone(this);
> }
> 
>-const char* inet_ntop(int af, const void *src, char* dst, socklen_t size) {
>+const char* inet_ntop(int af, const void* src, char* dst, socklen_t size) {
> #if defined(WEBRTC_WIN)
>   return win32_inet_ntop(af, src, dst, size);
> #else
>@@ -133,7 +133,7 @@ const char* inet_ntop(int af, const void *src, char* dst, socklen_t size) {
> #endif
> }
> 
>-int inet_pton(int af, const char* src, void *dst) {
>+int inet_pton(int af, const char* src, void* dst) {
> #if defined(WEBRTC_WIN)
>   return win32_inet_pton(af, src, dst);
> #else
>@@ -182,8 +182,8 @@ bool HasIPv6Enabled() {
>   do {
>     protocols.reset(new char[protbuff_size]);
>     protocol_infos = reinterpret_cast<LPWSAPROTOCOL_INFOW>(protocols.get());
>-    ret = WSCEnumProtocols(requested_protocols, protocol_infos,
>-                           &protbuff_size, &err);
>+    ret = WSCEnumProtocols(requested_protocols, protocol_infos, &protbuff_size,
>+                           &err);
>   } while (ret == SOCKET_ERROR && err == WSAENOBUFS);
> 
>   if (ret == SOCKET_ERROR) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/nethelpers.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/nethelpers.h
>index e118b27dbc0..f9d188f61d0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/nethelpers.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/nethelpers.h
>@@ -22,13 +22,11 @@
> 
> #include "rtc_base/asyncresolverinterface.h"
> #include "rtc_base/signalthread.h"
>-#include "rtc_base/sigslot.h"
> #include "rtc_base/socketaddress.h"
>+#include "rtc_base/third_party/sigslot/sigslot.h"
> 
> namespace rtc {
> 
>-class AsyncResolverTest;
>-
> // AsyncResolver will perform async DNS resolution, signaling the result on
> // the SignalDone from AsyncResolverInterface when the operation completes.
> class AsyncResolver : public SignalThread, public AsyncResolverInterface {
>@@ -56,8 +54,8 @@ class AsyncResolver : public SignalThread, public AsyncResolverInterface {
> 
> // rtc namespaced wrappers for inet_ntop and inet_pton so we can avoid
> // the windows-native versions of these.
>-const char* inet_ntop(int af, const void *src, char* dst, socklen_t size);
>-int inet_pton(int af, const char* src, void *dst);
>+const char* inet_ntop(int af, const void* src, char* dst, socklen_t size);
>+int inet_pton(int af, const char* src, void* dst);
> 
> bool HasIPv4Enabled();
> bool HasIPv6Enabled();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/network.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/network.cc
>index 56d729952f7..34da828477c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/network.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/network.cc
>@@ -23,8 +23,8 @@
> #endif  // WEBRTC_POSIX
> 
> #if defined(WEBRTC_WIN)
>+#include <iphlpapi.h>
> #include "rtc_base/win32.h"
>-#include <Iphlpapi.h>
> #elif !defined(__native_client__)
> #include "rtc_base/ifaddrs_converter.h"
> #endif
>@@ -40,6 +40,7 @@
> #include "rtc_base/socket.h"  // includes something that makes windows happy
> #include "rtc_base/stream.h"
> #include "rtc_base/stringencode.h"
>+#include "rtc_base/stringutils.h"
> #include "rtc_base/thread.h"
> 
> namespace rtc {
>@@ -93,6 +94,8 @@ bool SortNetworks(const Network* a, const Network* b) {
> 
> std::string AdapterTypeToString(AdapterType type) {
>   switch (type) {
>+    case ADAPTER_TYPE_ANY:
>+      return "Wildcard";
>     case ADAPTER_TYPE_UNKNOWN:
>       return "Unknown";
>     case ADAPTER_TYPE_ETHERNET:
>@@ -111,6 +114,35 @@ std::string AdapterTypeToString(AdapterType type) {
>   }
> }
> 
>+uint16_t ComputeNetworkCostByType(int type) {
>+  switch (type) {
>+    case rtc::ADAPTER_TYPE_ETHERNET:
>+    case rtc::ADAPTER_TYPE_LOOPBACK:
>+      return kNetworkCostMin;
>+    case rtc::ADAPTER_TYPE_WIFI:
>+      return kNetworkCostLow;
>+    case rtc::ADAPTER_TYPE_CELLULAR:
>+      return kNetworkCostHigh;
>+    case rtc::ADAPTER_TYPE_ANY:
>+      // Candidates gathered from the any-address/wildcard ports, as backups,
>+      // are given the maximum cost so that if there are other candidates with
>+      // known interface types, we would not select candidate pairs using these
>+      // backup candidates if other selection criteria with higher precedence
>+      // (network conditions over the route) are the same. Note that setting the
>+      // cost to kNetworkCostUnknown would be problematic since
>+      // ADAPTER_TYPE_CELLULAR would then have a higher cost. See
>+      // P2PTransportChannel::SortConnectionsAndUpdateState for how we rank and
>+      // select candidate pairs, where the network cost is among the criteria.
>+      return kNetworkCostMax;
>+    case rtc::ADAPTER_TYPE_VPN:
>+      // The cost of a VPN should be computed using its underlying network type.
>+      RTC_NOTREACHED();
>+      return kNetworkCostUnknown;
>+    default:
>+      return kNetworkCostUnknown;
>+  }
>+}
>+
> #if !defined(__native_client__)
> bool IsIgnoredIPv6(const InterfaceAddress& ip) {
>   if (ip.family() != AF_INET6) {
>@@ -146,19 +178,78 @@ const char kPublicIPv4Host[] = "8.8.8.8";
> const char kPublicIPv6Host[] = "2001:4860:4860::8888";
> const int kPublicPort = 53;  // DNS port.
> 
>-std::string MakeNetworkKey(const std::string& name, const IPAddress& prefix,
>+std::string MakeNetworkKey(const std::string& name,
>+                           const IPAddress& prefix,
>                            int prefix_length) {
>   std::ostringstream ost;
>   ost << name << "%" << prefix.ToString() << "/" << prefix_length;
>   return ost.str();
> }
>-
>-NetworkManager::NetworkManager() {
>+// Test if the network name matches the type<number> pattern, e.g. eth0. The
>+// matching is case-sensitive.
>+bool MatchTypeNameWithIndexPattern(const std::string& network_name,
>+                                   const std::string& type_name) {
>+  if (network_name.find(type_name) != 0) {
>+    return false;
>+  }
>+  return std::find_if(network_name.begin() + type_name.size(),
>+                      network_name.end(),
>+                      [](char c) { return !isdigit(c); }) == network_name.end();
> }
> 
>-NetworkManager::~NetworkManager() {
>+// A cautious note that this method may not provide an accurate adapter type
>+// based on the string matching. Incorrect type of adapters can affect the
>+// result of the downstream network filtering, see e.g.
>+// BasicPortAllocatorSession::GetNetworks when
>+// PORTALLOCATOR_DISABLE_COSTLY_NETWORKS is turned on.
>+AdapterType GetAdapterTypeFromName(const char* network_name) {
>+  if (MatchTypeNameWithIndexPattern(network_name, "lo")) {
>+    // Note that we have a more robust way to determine if a network interface
>+    // is a loopback interface by checking the flag IFF_LOOPBACK in ifa_flags of
>+    // an ifaddr struct. See ConvertIfAddrs in this file.
>+    return ADAPTER_TYPE_LOOPBACK;
>+  }
>+  if (MatchTypeNameWithIndexPattern(network_name, "eth")) {
>+    return ADAPTER_TYPE_ETHERNET;
>+  }
>+
>+  if (MatchTypeNameWithIndexPattern(network_name, "ipsec") ||
>+      MatchTypeNameWithIndexPattern(network_name, "tun") ||
>+      MatchTypeNameWithIndexPattern(network_name, "utun") ||
>+      MatchTypeNameWithIndexPattern(network_name, "tap")) {
>+    return ADAPTER_TYPE_VPN;
>+  }
>+#if defined(WEBRTC_IOS)
>+  // Cell networks are pdp_ipN on iOS.
>+  if (MatchTypeNameWithIndexPattern(network_name, "pdp_ip")) {
>+    return ADAPTER_TYPE_CELLULAR;
>+  }
>+  if (MatchTypeNameWithIndexPattern(network_name, "en")) {
>+    // This may not be most accurate because sometimes Ethernet interface
>+    // name also starts with "en" but it is better than showing it as
>+    // "unknown" type.
>+    // TODO(honghaiz): Write a proper IOS network manager.
>+    return ADAPTER_TYPE_WIFI;
>+  }
>+#elif defined(WEBRTC_ANDROID)
>+  if (MatchTypeNameWithIndexPattern(network_name, "rmnet") ||
>+      MatchTypeNameWithIndexPattern(network_name, "rmnet_data") ||
>+      MatchTypeNameWithIndexPattern(network_name, "v4-rmnet") ||
>+      MatchTypeNameWithIndexPattern(network_name, "v4-rmnet_data")) {
>+    return ADAPTER_TYPE_CELLULAR;
>+  }
>+  if (MatchTypeNameWithIndexPattern(network_name, "wlan")) {
>+    return ADAPTER_TYPE_WIFI;
>+  }
>+#endif
>+
>+  return ADAPTER_TYPE_UNKNOWN;
> }
> 
>+NetworkManager::NetworkManager() {}
>+
>+NetworkManager::~NetworkManager() {}
>+
> NetworkManager::EnumerationPermission NetworkManager::enumeration_permission()
>     const {
>   return ENUMERATION_ALLOWED;
>@@ -170,8 +261,7 @@ bool NetworkManager::GetDefaultLocalAddress(int family, IPAddress* addr) const {
> 
> NetworkManagerBase::NetworkManagerBase()
>     : enumeration_permission_(NetworkManager::ENUMERATION_ALLOWED),
>-      ipv6_enabled_(true) {
>-}
>+      ipv6_enabled_(true) {}
> 
> NetworkManagerBase::~NetworkManagerBase() {
>   for (const auto& kv : networks_map_) {
>@@ -188,7 +278,7 @@ void NetworkManagerBase::GetAnyAddressNetworks(NetworkList* networks) {
>   if (!ipv4_any_address_network_) {
>     const rtc::IPAddress ipv4_any_address(INADDR_ANY);
>     ipv4_any_address_network_.reset(
>-        new rtc::Network("any", "any", ipv4_any_address, 0));
>+        new rtc::Network("any", "any", ipv4_any_address, 0, ADAPTER_TYPE_ANY));
>     ipv4_any_address_network_->set_default_local_address_provider(this);
>     ipv4_any_address_network_->AddIP(ipv4_any_address);
>   }
>@@ -197,8 +287,8 @@ void NetworkManagerBase::GetAnyAddressNetworks(NetworkList* networks) {
>   if (ipv6_enabled()) {
>     if (!ipv6_any_address_network_) {
>       const rtc::IPAddress ipv6_any_address(in6addr_any);
>-      ipv6_any_address_network_.reset(
>-          new rtc::Network("any", "any", ipv6_any_address, 0));
>+      ipv6_any_address_network_.reset(new rtc::Network(
>+          "any", "any", ipv6_any_address, 0, ADAPTER_TYPE_ANY));
>       ipv6_any_address_network_->set_default_local_address_provider(this);
>       ipv6_any_address_network_->AddIP(ipv6_any_address);
>     }
>@@ -229,8 +319,7 @@ void NetworkManagerBase::MergeNetworkList(const NetworkList& new_networks,
>   // First, build a set of network-keys to the ipaddresses.
>   for (Network* network : list) {
>     bool might_add_to_merged_list = false;
>-    std::string key = MakeNetworkKey(network->name(),
>-                                     network->prefix(),
>+    std::string key = MakeNetworkKey(network->name(), network->prefix(),
>                                      network->prefix_length());
>     if (consolidated_address_list.find(key) ==
>         consolidated_address_list.end()) {
>@@ -381,8 +470,7 @@ BasicNetworkManager::BasicNetworkManager()
>       start_count_(0),
>       ignore_non_default_routes_(false) {}
> 
>-BasicNetworkManager::~BasicNetworkManager() {
>-}
>+BasicNetworkManager::~BasicNetworkManager() {}
> 
> void BasicNetworkManager::OnNetworksChanged() {
>   RTC_LOG(LS_INFO) << "Network change was observed";
>@@ -444,15 +532,28 @@ void BasicNetworkManager::ConvertIfAddrs(struct ifaddrs* interfaces,
>     }
> 
>     AdapterType adapter_type = ADAPTER_TYPE_UNKNOWN;
>+    AdapterType vpn_underlying_adapter_type = ADAPTER_TYPE_UNKNOWN;
>     if (cursor->ifa_flags & IFF_LOOPBACK) {
>       adapter_type = ADAPTER_TYPE_LOOPBACK;
>     } else {
>-      adapter_type = GetAdapterTypeFromName(cursor->ifa_name);
>+      // If there is a network_monitor, use it to get the adapter type.
>+      // Otherwise, get the adapter type based on a few name matching rules.
>+      if (network_monitor_) {
>+        adapter_type = network_monitor_->GetAdapterType(cursor->ifa_name);
>+      }
>+      if (adapter_type == ADAPTER_TYPE_UNKNOWN) {
>+        adapter_type = GetAdapterTypeFromName(cursor->ifa_name);
>+      }
>+    }
>+
>+    if (adapter_type == ADAPTER_TYPE_VPN && network_monitor_) {
>+      vpn_underlying_adapter_type =
>+          network_monitor_->GetVpnUnderlyingAdapterType(cursor->ifa_name);
>     }
>     int prefix_length = CountIPMaskBits(mask);
>     prefix = TruncateIP(ip, prefix_length);
>-    std::string key = MakeNetworkKey(std::string(cursor->ifa_name),
>-                                     prefix, prefix_length);
>+    std::string key =
>+        MakeNetworkKey(std::string(cursor->ifa_name), prefix, prefix_length);
>     auto iter = current_networks.find(key);
>     if (iter == current_networks.end()) {
>       // TODO(phoglund): Need to recognize other types as well.
>@@ -463,6 +564,7 @@ void BasicNetworkManager::ConvertIfAddrs(struct ifaddrs* interfaces,
>       network->set_scope_id(scope_id);
>       network->AddIP(ip);
>       network->set_ignored(IsIgnoredNetwork(*network));
>+      network->set_underlying_type_for_vpn(vpn_underlying_adapter_type);
>       if (include_ignored || !network->ignored()) {
>         current_networks[key] = network.get();
>         networks->push_back(network.release());
>@@ -472,6 +574,8 @@ void BasicNetworkManager::ConvertIfAddrs(struct ifaddrs* interfaces,
>       existing_network->AddIP(ip);
>       if (adapter_type != ADAPTER_TYPE_UNKNOWN) {
>         existing_network->set_type(adapter_type);
>+        existing_network->set_underlying_type_for_vpn(
>+            vpn_underlying_adapter_type);
>       }
>     }
>   }
>@@ -498,7 +602,8 @@ bool BasicNetworkManager::CreateNetworks(bool include_ignored,
> #elif defined(WEBRTC_WIN)
> 
> unsigned int GetPrefix(PIP_ADAPTER_PREFIX prefixlist,
>-              const IPAddress& ip, IPAddress* prefix) {
>+                       const IPAddress& ip,
>+                       IPAddress* prefix) {
>   IPAddress current_prefix;
>   IPAddress best_prefix;
>   unsigned int best_length = 0;
>@@ -517,10 +622,10 @@ unsigned int GetPrefix(PIP_ADAPTER_PREFIX prefixlist,
>         break;
>       }
>       case AF_INET6: {
>-          sockaddr_in6* v6_addr =
>-              reinterpret_cast<sockaddr_in6*>(prefixlist->Address.lpSockaddr);
>-          current_prefix = IPAddress(v6_addr->sin6_addr);
>-          break;
>+        sockaddr_in6* v6_addr =
>+            reinterpret_cast<sockaddr_in6*>(prefixlist->Address.lpSockaddr);
>+        current_prefix = IPAddress(v6_addr->sin6_addr);
>+        break;
>       }
>       default: {
>         prefixlist = prefixlist->Next;
>@@ -552,8 +657,7 @@ bool BasicNetworkManager::CreateNetworks(bool include_ignored,
>   do {
>     adapter_info.reset(new char[buffer_size]);
>     adapter_addrs = reinterpret_cast<PIP_ADAPTER_ADDRESSES>(adapter_info.get());
>-    ret = GetAdaptersAddresses(AF_UNSPEC, adapter_flags,
>-                               0, adapter_addrs,
>+    ret = GetAdaptersAddresses(AF_UNSPEC, adapter_flags, 0, adapter_addrs,
>                                reinterpret_cast<PULONG>(&buffer_size));
>   } while (ret == ERROR_BUFFER_OVERFLOW);
>   if (ret != ERROR_SUCCESS) {
>@@ -594,7 +698,7 @@ bool BasicNetworkManager::CreateNetworks(bool include_ignored,
>               scope_id = v6_addr->sin6_scope_id;
>               ip = IPAddress(v6_addr->sin6_addr);
> 
>-              if (IsIgnoredIPv6(ip)) {
>+              if (IsIgnoredIPv6(InterfaceAddress(ip))) {
>                 continue;
>               }
> 
>@@ -603,9 +707,7 @@ bool BasicNetworkManager::CreateNetworks(bool include_ignored,
>               continue;
>             }
>           }
>-          default: {
>-            continue;
>-          }
>+          default: { continue; }
>         }
> 
>         IPAddress prefix;
>@@ -614,9 +716,29 @@ bool BasicNetworkManager::CreateNetworks(bool include_ignored,
>         auto existing_network = current_networks.find(key);
>         if (existing_network == current_networks.end()) {
>           AdapterType adapter_type = ADAPTER_TYPE_UNKNOWN;
>-          if (adapter_addrs->IfType == IF_TYPE_SOFTWARE_LOOPBACK) {
>-            // TODO(phoglund): Need to recognize other types as well.
>-            adapter_type = ADAPTER_TYPE_LOOPBACK;
>+          switch (adapter_addrs->IfType) {
>+            case IF_TYPE_SOFTWARE_LOOPBACK:
>+              adapter_type = ADAPTER_TYPE_LOOPBACK;
>+              break;
>+            case IF_TYPE_ETHERNET_CSMACD:
>+            case IF_TYPE_ETHERNET_3MBIT:
>+            case IF_TYPE_IEEE80212:
>+            case IF_TYPE_FASTETHER:
>+            case IF_TYPE_FASTETHER_FX:
>+            case IF_TYPE_GIGABITETHERNET:
>+              adapter_type = ADAPTER_TYPE_ETHERNET;
>+              break;
>+            case IF_TYPE_IEEE80211:
>+              adapter_type = ADAPTER_TYPE_WIFI;
>+              break;
>+            case IF_TYPE_WWANPP:
>+            case IF_TYPE_WWANPP2:
>+              adapter_type = ADAPTER_TYPE_CELLULAR;
>+              break;
>+            default:
>+              // TODO(phoglund): Need to recognize other types as well.
>+              adapter_type = ADAPTER_TYPE_UNKNOWN;
>+              break;
>           }
>           std::unique_ptr<Network> network(new Network(
>               name, description, prefix, prefix_length, adapter_type));
>@@ -656,12 +778,9 @@ bool IsDefaultRoute(const std::string& network_name) {
>     while (fs.ReadLine(&line) == SR_SUCCESS) {
>       char iface_name[256];
>       unsigned int iface_ip, iface_gw, iface_mask, iface_flags;
>-      if (sscanf(line.c_str(),
>-                 "%255s %8X %8X %4X %*d %*u %*d %8X",
>-                 iface_name, &iface_ip, &iface_gw,
>-                 &iface_flags, &iface_mask) == 5 &&
>-          network_name == iface_name &&
>-          iface_mask == 0 &&
>+      if (sscanf(line.c_str(), "%255s %8X %8X %4X %*d %*u %*d %8X", iface_name,
>+                 &iface_ip, &iface_gw, &iface_flags, &iface_mask) == 5 &&
>+          network_name == iface_name && iface_mask == 0 &&
>           (iface_flags & (RTF_UP | RTF_HOST)) == RTF_UP) {
>         return true;
>       }
>@@ -768,7 +887,7 @@ void BasicNetworkManager::OnMessage(Message* msg) {
>       UpdateNetworksContinually();
>       break;
>     }
>-    case kSignalNetworksMessage:  {
>+    case kSignalNetworksMessage: {
>       SignalNetworksChanged();
>       break;
>     }
>@@ -777,44 +896,6 @@ void BasicNetworkManager::OnMessage(Message* msg) {
>   }
> }
> 
>-AdapterType BasicNetworkManager::GetAdapterTypeFromName(
>-    const char* network_name) const {
>-  // If there is a network_monitor, use it to get the adapter type.
>-  // Otherwise, get the adapter type based on a few name matching rules.
>-  if (network_monitor_) {
>-    AdapterType type = network_monitor_->GetAdapterType(network_name);
>-    if (type != ADAPTER_TYPE_UNKNOWN) {
>-      return type;
>-    }
>-  }
>-  if (strncmp(network_name, "ipsec", 5) == 0) {
>-    return ADAPTER_TYPE_VPN;
>-  }
>-#if defined(WEBRTC_IOS)
>-  // Cell networks are pdp_ipN on iOS.
>-  if (strncmp(network_name, "pdp_ip", 6) == 0) {
>-    return ADAPTER_TYPE_CELLULAR;
>-  }
>-  if (strncmp(network_name, "en", 2) == 0) {
>-    // This may not be most accurate because sometimes Ethernet interface
>-    // name also starts with "en" but it is better than showing it as
>-    // "unknown" type.
>-    // TODO(honghaiz): Write a proper IOS network manager.
>-    return ADAPTER_TYPE_WIFI;
>-  }
>-#elif defined(WEBRTC_ANDROID)
>-  if (strncmp(network_name, "rmnet", 5) == 0 ||
>-      strncmp(network_name, "v4-rmnet", 8) == 0) {
>-    return ADAPTER_TYPE_CELLULAR;
>-  }
>-  if (strncmp(network_name, "wlan", 4) == 0) {
>-    return ADAPTER_TYPE_WIFI;
>-  }
>-#endif
>-
>-  return ADAPTER_TYPE_UNKNOWN;
>-}
>-
> IPAddress BasicNetworkManager::QueryDefaultLocalAddress(int family) const {
>   RTC_DCHECK(thread_ == Thread::Current());
>   RTC_DCHECK(thread_->socketserver() != nullptr);
>@@ -830,8 +911,8 @@ IPAddress BasicNetworkManager::QueryDefaultLocalAddress(int family) const {
>   if (socket->Connect(SocketAddress(
>           family == AF_INET ? kPublicIPv4Host : kPublicIPv6Host, kPublicPort)) <
>       0) {
>-    if (socket->GetError() != ENETUNREACH
>-        && socket->GetError() != EHOSTUNREACH) {
>+    if (socket->GetError() != ENETUNREACH &&
>+        socket->GetError() != EHOSTUNREACH) {
>       // Ignore the expected case of "host/net unreachable" - which happens if
>       // the network is V4- or V6-only.
>       RTC_LOG(LS_INFO) << "Connect failed with " << socket->GetError();
>@@ -970,13 +1051,22 @@ IPAddress Network::GetBestIP() const {
>   return static_cast<IPAddress>(selected_ip);
> }
> 
>+uint16_t Network::GetCost() const {
>+  AdapterType type = IsVpn() ? underlying_type_for_vpn_ : type_;
>+  return ComputeNetworkCostByType(type);
>+}
>+
> std::string Network::ToString() const {
>   std::stringstream ss;
>   // Print out the first space-terminated token of the network desc, plus
>   // the IP address.
>-  ss << "Net[" << description_.substr(0, description_.find(' '))
>-     << ":" << prefix_.ToSensitiveString() << "/" << prefix_length_
>-     << ":" << AdapterTypeToString(type_) << "]";
>+  ss << "Net[" << description_.substr(0, description_.find(' ')) << ":"
>+     << prefix_.ToSensitiveString() << "/" << prefix_length_ << ":"
>+     << AdapterTypeToString(type_);
>+  if (IsVpn()) {
>+    ss << "/" << AdapterTypeToString(underlying_type_for_vpn_);
>+  }
>+  ss << ":id=" << id_ << "]";
>   return ss.str();
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/network.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/network.h
>index a1b5c4fa68a..ad932d05ac3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/network.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/network.h
>@@ -22,7 +22,7 @@
> #include "rtc_base/ipaddress.h"
> #include "rtc_base/messagehandler.h"
> #include "rtc_base/networkmonitor.h"
>-#include "rtc_base/sigslot.h"
>+#include "rtc_base/third_party/sigslot/sigslot.h"
> 
> #if defined(WEBRTC_POSIX)
> struct ifaddrs;
>@@ -44,9 +44,15 @@ const int kDefaultNetworkIgnoreMask = ADAPTER_TYPE_LOOPBACK;
> // Makes a string key for this network. Used in the network manager's maps.
> // Network objects are keyed on interface name, network prefix and the
> // length of that prefix.
>-std::string MakeNetworkKey(const std::string& name, const IPAddress& prefix,
>+std::string MakeNetworkKey(const std::string& name,
>+                           const IPAddress& prefix,
>                            int prefix_length);
> 
>+// Utility function that attempts to determine an adapter type by an interface
>+// name (e.g., "wlan0"). Can be used by NetworkManager subclasses when other
>+// mechanisms fail to determine the type.
>+AdapterType GetAdapterTypeFromName(const char* network_name);
>+
> class DefaultLocalAddressProvider {
>  public:
>   virtual ~DefaultLocalAddressProvider() = default;
>@@ -117,7 +123,7 @@ class NetworkManager : public DefaultLocalAddressProvider {
>   // can optionally be called after GetNetworks.
>   //
>   // TODO(guoweis): remove this body when chromium implements this.
>-  virtual void GetAnyAddressNetworks(NetworkList*) {}
>+  virtual void GetAnyAddressNetworks(NetworkList* networks) {}
> 
>   // Dumps the current list of networks in the network manager.
>   virtual void DumpNetworks() {}
>@@ -265,8 +271,6 @@ class BasicNetworkManager : public NetworkManagerBase,
>   // Only updates the networks; does not reschedule the next update.
>   void UpdateNetworksOnce();
> 
>-  AdapterType GetAdapterTypeFromName(const char* network_name) const;
>-
>   Thread* thread_;
>   bool sent_first_update_;
>   int start_count_;
>@@ -290,7 +294,7 @@ class Network {
>           AdapterType type);
>   Network(const Network&);
>   ~Network();
>-
>+  // This signal is fired whenever type() or underlying_type_for_vpn() changes.
>   sigslot::signal1<const Network*> SignalTypeChanged;
> 
>   const DefaultLocalAddressProvider* default_local_address_provider() {
>@@ -343,12 +347,13 @@ class Network {
> 
>   // Adds an active IP address to this network. Does not check for duplicates.
>   void AddIP(const InterfaceAddress& ip) { ips_.push_back(ip); }
>+  void AddIP(const IPAddress& ip) { ips_.push_back(rtc::InterfaceAddress(ip)); }
> 
>   // Sets the network's IP address list. Returns true if new IP addresses were
>   // detected. Passing true to already_changed skips this check.
>   bool SetIPs(const std::vector<InterfaceAddress>& ips, bool already_changed);
>   // Get the list of IP Addresses associated with this network.
>-  const std::vector<InterfaceAddress>& GetIPs() const { return ips_;}
>+  const std::vector<InterfaceAddress>& GetIPs() const { return ips_; }
>   // Clear the network's list of addresses.
>   void ClearIPs() { ips_.clear(); }
> 
>@@ -363,28 +368,37 @@ class Network {
>   void set_ignored(bool ignored) { ignored_ = ignored; }
> 
>   AdapterType type() const { return type_; }
>+  // When type() is ADAPTER_TYPE_VPN, this returns the type of the underlying
>+  // network interface used by the VPN, typically the preferred network type
>+  // (see for example, the method setUnderlyingNetworks(android.net.Network[])
>+  // on https://developer.android.com/reference/android/net/VpnService.html).
>+  // When this information is unavailable from the OS, ADAPTER_TYPE_UNKNOWN is
>+  // returned.
>+  AdapterType underlying_type_for_vpn() const {
>+    return underlying_type_for_vpn_;
>+  }
>   void set_type(AdapterType type) {
>     if (type_ == type) {
>       return;
>     }
>     type_ = type;
>+    if (type != ADAPTER_TYPE_VPN) {
>+      underlying_type_for_vpn_ = ADAPTER_TYPE_UNKNOWN;
>+    }
>     SignalTypeChanged(this);
>   }
> 
>-  uint16_t GetCost() const {
>-    switch (type_) {
>-      case rtc::ADAPTER_TYPE_ETHERNET:
>-      case rtc::ADAPTER_TYPE_LOOPBACK:
>-        return kNetworkCostMin;
>-      case rtc::ADAPTER_TYPE_WIFI:
>-      case rtc::ADAPTER_TYPE_VPN:
>-        return kNetworkCostLow;
>-      case rtc::ADAPTER_TYPE_CELLULAR:
>-        return kNetworkCostHigh;
>-      default:
>-        return kNetworkCostUnknown;
>+  void set_underlying_type_for_vpn(AdapterType type) {
>+    if (underlying_type_for_vpn_ == type) {
>+      return;
>     }
>+    underlying_type_for_vpn_ = type;
>+    SignalTypeChanged(this);
>   }
>+
>+  bool IsVpn() const { return type_ == ADAPTER_TYPE_VPN; }
>+
>+  uint16_t GetCost() const;
>   // A unique id assigned by the network manager, which may be signaled
>   // to the remote side in the candidate.
>   uint16_t id() const { return id_; }
>@@ -417,6 +431,7 @@ class Network {
>   int scope_id_;
>   bool ignored_;
>   AdapterType type_;
>+  AdapterType underlying_type_for_vpn_ = ADAPTER_TYPE_UNKNOWN;
>   int preference_;
>   bool active_ = true;
>   uint16_t id_ = 0;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/network_constants.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/network_constants.h
>index b4c8beaf319..efb2c83455b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/network_constants.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/network_constants.h
>@@ -28,7 +28,13 @@ enum AdapterType {
>   ADAPTER_TYPE_WIFI = 1 << 1,
>   ADAPTER_TYPE_CELLULAR = 1 << 2,
>   ADAPTER_TYPE_VPN = 1 << 3,
>-  ADAPTER_TYPE_LOOPBACK = 1 << 4
>+  ADAPTER_TYPE_LOOPBACK = 1 << 4,
>+  // ADAPTER_TYPE_ANY is used for a network, which only contains a single "any
>+  // address" IP address (INADDR_ANY for IPv4 or in6addr_any for IPv6), and can
>+  // use any/all network interfaces. Whereas ADAPTER_TYPE_UNKNOWN is used
>+  // when the network uses a specific interface/IP, but its interface type can
>+  // not be determined or not fit in this enum.
>+  ADAPTER_TYPE_ANY = 1 << 5,
> };
> 
> }  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/network_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/network_unittest.cc
>index bf09c2456d7..cf1ceccaecf 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/network_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/network_unittest.cc
>@@ -19,8 +19,8 @@
> #include "rtc_base/nethelpers.h"
> #include "rtc_base/networkmonitor.h"
> #if defined(WEBRTC_POSIX)
>-#include <sys/types.h>
> #include <net/if.h>
>+#include <sys/types.h>
> #include "rtc_base/ifaddrs_converter.h"
> #endif  // defined(WEBRTC_POSIX)
> #include "rtc_base/gunit.h"
>@@ -61,15 +61,25 @@ class FakeNetworkMonitorFactory : public NetworkMonitorFactory {
>   }
> };
> 
>+bool SameNameAndPrefix(const rtc::Network& a, const rtc::Network& b) {
>+  if (a.name() != b.name()) {
>+    RTC_LOG(INFO) << "Different interface names.";
>+    return false;
>+  }
>+  if (a.prefix_length() != b.prefix_length() || a.prefix() != b.prefix()) {
>+    RTC_LOG(INFO) << "Different IP prefixes.";
>+    return false;
>+  }
>+  return true;
>+}
>+
> }  // namespace
> 
>-class NetworkTest : public testing::Test, public sigslot::has_slots<>  {
>+class NetworkTest : public testing::Test, public sigslot::has_slots<> {
>  public:
>   NetworkTest() : callback_called_(false) {}
> 
>-  void OnNetworksChanged() {
>-    callback_called_ = true;
>-  }
>+  void OnNetworksChanged() { callback_called_ = true; }
> 
>   NetworkManager::Stats MergeNetworkList(
>       BasicNetworkManager& network_manager,
>@@ -86,7 +96,8 @@ class NetworkTest : public testing::Test, public sigslot::has_slots<>  {
>   }
> 
>   NetworkManager::NetworkList GetNetworks(
>-      const BasicNetworkManager& network_manager, bool include_ignored) {
>+      const BasicNetworkManager& network_manager,
>+      bool include_ignored) {
>     NetworkManager::NetworkList list;
>     network_manager.CreateNetworks(include_ignored, &list);
>     return list;
>@@ -125,8 +136,8 @@ class NetworkTest : public testing::Test, public sigslot::has_slots<>  {
> 
>   struct sockaddr_in6* CreateIpv6Addr(const std::string& ip_string,
>                                       uint32_t scope_id) {
>-    struct sockaddr_in6* ipv6_addr = static_cast<struct sockaddr_in6*>(
>-        malloc(sizeof(struct sockaddr_in6)));
>+    struct sockaddr_in6* ipv6_addr =
>+        static_cast<struct sockaddr_in6*>(malloc(sizeof(struct sockaddr_in6)));
>     memset(ipv6_addr, 0, sizeof(struct sockaddr_in6));
>     ipv6_addr->sin6_family = AF_INET6;
>     ipv6_addr->sin6_scope_id = scope_id;
>@@ -213,10 +224,10 @@ TEST_F(NetworkTest, TestIsIgnoredNetworkIgnoresIPsStartingWith0) {
> 
> // TODO(phoglund): Remove when ignore list goes away.
> TEST_F(NetworkTest, TestIgnoreList) {
>-  Network ignore_me("ignore_me", "Ignore me please!",
>-                    IPAddress(0x12345600U), 24);
>-  Network include_me("include_me", "Include me please!",
>-                     IPAddress(0x12345600U), 24);
>+  Network ignore_me("ignore_me", "Ignore me please!", IPAddress(0x12345600U),
>+                    24);
>+  Network include_me("include_me", "Include me please!", IPAddress(0x12345600U),
>+                     24);
>   BasicNetworkManager network_manager;
>   EXPECT_FALSE(IsIgnoredNetwork(network_manager, ignore_me));
>   EXPECT_FALSE(IsIgnoredNetwork(network_manager, include_me));
>@@ -233,9 +244,7 @@ TEST_F(NetworkTest, DISABLED_TestCreateNetworks) {
>   NetworkManager::NetworkList result = GetNetworks(manager, true);
>   // We should be able to bind to any addresses we find.
>   NetworkManager::NetworkList::iterator it;
>-  for (it = result.begin();
>-       it != result.end();
>-       ++it) {
>+  for (it = result.begin(); it != result.end(); ++it) {
>     sockaddr_storage storage;
>     memset(&storage, 0, sizeof(storage));
>     IPAddress ip = (*it)->GetBestIP();
>@@ -246,8 +255,7 @@ TEST_F(NetworkTest, DISABLED_TestCreateNetworks) {
>     if (fd > 0) {
>       size_t ipsize = bindaddress.ToSockAddrStorage(&storage);
>       EXPECT_GE(ipsize, 0U);
>-      int success = ::bind(fd,
>-                           reinterpret_cast<sockaddr*>(&storage),
>+      int success = ::bind(fd, reinterpret_cast<sockaddr*>(&storage),
>                            static_cast<int>(ipsize));
> #if defined(WEBRTC_WIN)
>       if (success)
>@@ -268,8 +276,8 @@ TEST_F(NetworkTest, DISABLED_TestCreateNetworks) {
> // ALLOWED.
> TEST_F(NetworkTest, TestUpdateNetworks) {
>   BasicNetworkManager manager;
>-  manager.SignalNetworksChanged.connect(
>-      static_cast<NetworkTest*>(this), &NetworkTest::OnNetworksChanged);
>+  manager.SignalNetworksChanged.connect(static_cast<NetworkTest*>(this),
>+                                        &NetworkTest::OnNetworksChanged);
>   EXPECT_EQ(NetworkManager::ENUMERATION_ALLOWED,
>             manager.enumeration_permission());
>   manager.StartUpdating();
>@@ -320,7 +328,7 @@ TEST_F(NetworkTest, TestBasicMergeNetworkList) {
> 
>   manager.GetNetworks(&list);
>   EXPECT_EQ(1U, list.size());
>-  EXPECT_EQ(ipv4_network1.ToString(), list[0]->ToString());
>+  EXPECT_TRUE(SameNameAndPrefix(ipv4_network1, *list[0]));
>   Network* net1 = list[0];
>   uint16_t net_id1 = net1->id();
>   EXPECT_EQ(1, net_id1);
>@@ -336,7 +344,7 @@ TEST_F(NetworkTest, TestBasicMergeNetworkList) {
> 
>   manager.GetNetworks(&list);
>   EXPECT_EQ(1U, list.size());
>-  EXPECT_EQ(ipv4_network2.ToString(), list[0]->ToString());
>+  EXPECT_TRUE(SameNameAndPrefix(ipv4_network2, *list[0]));
>   Network* net2 = list[0];
>   uint16_t net_id2 = net2->id();
>   // Network id will increase.
>@@ -416,8 +424,8 @@ void SetupNetworks(NetworkManager::NetworkList* list) {
> // Test that the basic network merging case works.
> TEST_F(NetworkTest, TestIPv6MergeNetworkList) {
>   BasicNetworkManager manager;
>-  manager.SignalNetworksChanged.connect(
>-      static_cast<NetworkTest*>(this), &NetworkTest::OnNetworksChanged);
>+  manager.SignalNetworksChanged.connect(static_cast<NetworkTest*>(this),
>+                                        &NetworkTest::OnNetworksChanged);
>   NetworkManager::NetworkList original_list;
>   SetupNetworks(&original_list);
>   bool changed = false;
>@@ -441,8 +449,8 @@ TEST_F(NetworkTest, TestIPv6MergeNetworkList) {
> // objects remain in the result list.
> TEST_F(NetworkTest, TestNoChangeMerge) {
>   BasicNetworkManager manager;
>-  manager.SignalNetworksChanged.connect(
>-      static_cast<NetworkTest*>(this), &NetworkTest::OnNetworksChanged);
>+  manager.SignalNetworksChanged.connect(static_cast<NetworkTest*>(this),
>+                                        &NetworkTest::OnNetworksChanged);
>   NetworkManager::NetworkList original_list;
>   SetupNetworks(&original_list);
>   bool changed = false;
>@@ -476,17 +484,16 @@ TEST_F(NetworkTest, TestNoChangeMerge) {
> // IP changed.
> TEST_F(NetworkTest, MergeWithChangedIP) {
>   BasicNetworkManager manager;
>-  manager.SignalNetworksChanged.connect(
>-      static_cast<NetworkTest*>(this), &NetworkTest::OnNetworksChanged);
>+  manager.SignalNetworksChanged.connect(static_cast<NetworkTest*>(this),
>+                                        &NetworkTest::OnNetworksChanged);
>   NetworkManager::NetworkList original_list;
>   SetupNetworks(&original_list);
>   // Make a network that we're going to change.
>   IPAddress ip;
>   EXPECT_TRUE(IPFromString("2401:fa01:4:1000:be30:faa:fee:faa", &ip));
>   IPAddress prefix = TruncateIP(ip, 64);
>-  Network* network_to_change = new Network("test_eth0",
>-                                          "Test Network Adapter 1",
>-                                          prefix, 64);
>+  Network* network_to_change =
>+      new Network("test_eth0", "Test Network Adapter 1", prefix, 64);
>   Network* changed_network = new Network(*network_to_change);
>   network_to_change->AddIP(ip);
>   IPAddress changed_ip;
>@@ -505,8 +512,7 @@ TEST_F(NetworkTest, MergeWithChangedIP) {
>   manager.GetNetworks(&list);
>   EXPECT_EQ(original_list.size(), list.size());
>   // Make sure the original network is still in the merged list.
>-  EXPECT_NE(list.end(),
>-            std::find(list.begin(), list.end(), network_to_change));
>+  EXPECT_NE(list.end(), std::find(list.begin(), list.end(), network_to_change));
>   EXPECT_EQ(changed_ip, network_to_change->GetIPs().at(0));
> }
> 
>@@ -514,8 +520,8 @@ TEST_F(NetworkTest, MergeWithChangedIP) {
> // with additional IPs (not just a replacement).
> TEST_F(NetworkTest, TestMultipleIPMergeNetworkList) {
>   BasicNetworkManager manager;
>-  manager.SignalNetworksChanged.connect(
>-      static_cast<NetworkTest*>(this), &NetworkTest::OnNetworksChanged);
>+  manager.SignalNetworksChanged.connect(static_cast<NetworkTest*>(this),
>+                                        &NetworkTest::OnNetworksChanged);
>   NetworkManager::NetworkList original_list;
>   SetupNetworks(&original_list);
>   bool changed = false;
>@@ -544,7 +550,7 @@ TEST_F(NetworkTest, TestMultipleIPMergeNetworkList) {
>   int matchcount = 0;
>   for (NetworkManager::NetworkList::iterator it = list.begin();
>        it != list.end(); ++it) {
>-    if ((*it)->ToString() == original_list[2]->ToString()) {
>+    if (SameNameAndPrefix(**it, *original_list[2])) {
>       ++matchcount;
>       EXPECT_EQ(1, matchcount);
>       // This should be the same network object as before.
>@@ -552,19 +558,16 @@ TEST_F(NetworkTest, TestMultipleIPMergeNetworkList) {
>       // But with two addresses now.
>       EXPECT_EQ(2U, (*it)->GetIPs().size());
>       EXPECT_NE((*it)->GetIPs().end(),
>-                std::find((*it)->GetIPs().begin(),
>-                          (*it)->GetIPs().end(),
>-                          check_ip));
>+                std::find((*it)->GetIPs().begin(), (*it)->GetIPs().end(),
>+                          InterfaceAddress(check_ip)));
>       EXPECT_NE((*it)->GetIPs().end(),
>-                std::find((*it)->GetIPs().begin(),
>-                          (*it)->GetIPs().end(),
>-                          ip));
>+                std::find((*it)->GetIPs().begin(), (*it)->GetIPs().end(),
>+                          InterfaceAddress(ip)));
>     } else {
>       // Check the IP didn't get added anywhere it wasn't supposed to.
>       EXPECT_EQ((*it)->GetIPs().end(),
>-                std::find((*it)->GetIPs().begin(),
>-                          (*it)->GetIPs().end(),
>-                          ip));
>+                std::find((*it)->GetIPs().begin(), (*it)->GetIPs().end(),
>+                          InterfaceAddress(ip)));
>     }
>   }
> }
>@@ -572,8 +575,8 @@ TEST_F(NetworkTest, TestMultipleIPMergeNetworkList) {
> // Test that merge correctly distinguishes multiple networks on an interface.
> TEST_F(NetworkTest, TestMultiplePublicNetworksOnOneInterfaceMerge) {
>   BasicNetworkManager manager;
>-  manager.SignalNetworksChanged.connect(
>-      static_cast<NetworkTest*>(this), &NetworkTest::OnNetworksChanged);
>+  manager.SignalNetworksChanged.connect(static_cast<NetworkTest*>(this),
>+                                        &NetworkTest::OnNetworksChanged);
>   NetworkManager::NetworkList original_list;
>   SetupNetworks(&original_list);
>   bool changed = false;
>@@ -606,9 +609,8 @@ TEST_F(NetworkTest, TestMultiplePublicNetworksOnOneInterfaceMerge) {
>     } else {
>       // Check the IP didn't get added anywhere it wasn't supposed to.
>       EXPECT_EQ((*it)->GetIPs().end(),
>-                std::find((*it)->GetIPs().begin(),
>-                          (*it)->GetIPs().end(),
>-                          ip));
>+                std::find((*it)->GetIPs().begin(), (*it)->GetIPs().end(),
>+                          InterfaceAddress(ip)));
>     }
>   }
> }
>@@ -841,6 +843,18 @@ TEST_F(NetworkTest, TestGetAdapterTypeFromNameMatching) {
>   ClearNetworks(manager);
>   ReleaseIfAddrs(addr_list);
> 
>+  strcpy(if_name, "lo0");
>+  addr_list = InstallIpv6Network(if_name, ipv6_address1, ipv6_mask, manager);
>+  EXPECT_EQ(ADAPTER_TYPE_LOOPBACK, GetAdapterType(manager));
>+  ClearNetworks(manager);
>+  ReleaseIfAddrs(addr_list);
>+
>+  strcpy(if_name, "eth0");
>+  addr_list = InstallIpv6Network(if_name, ipv6_address1, ipv6_mask, manager);
>+  EXPECT_EQ(ADAPTER_TYPE_ETHERNET, GetAdapterType(manager));
>+  ClearNetworks(manager);
>+  ReleaseIfAddrs(addr_list);
>+
> #if defined(WEBRTC_IOS)
>   strcpy(if_name, "pdp_ip0");
>   addr_list = InstallIpv6Network(if_name, ipv6_address1, ipv6_mask, manager);
>@@ -962,8 +976,8 @@ TEST_F(NetworkTest, TestMergeNetworkList) {
>   // IPAddresses.
>   EXPECT_EQ(list2.size(), 1uL);
>   EXPECT_EQ(list2[0]->GetIPs().size(), 2uL);
>-  EXPECT_EQ(list2[0]->GetIPs()[0], ip1);
>-  EXPECT_EQ(list2[0]->GetIPs()[1], ip2);
>+  EXPECT_EQ(list2[0]->GetIPs()[0], InterfaceAddress(ip1));
>+  EXPECT_EQ(list2[0]->GetIPs()[1], InterfaceAddress(ip2));
> }
> 
> // Test that MergeNetworkList successfully detects the change if
>@@ -1018,8 +1032,8 @@ TEST_F(NetworkTest, TestIPv6Selection) {
>   ASSERT_TRUE(IPFromString(ipstr, IPV6_ADDRESS_FLAG_DEPRECATED, &ip));
> 
>   // Create a network with this prefix.
>-  Network ipv6_network(
>-      "test_eth0", "Test NetworkAdapter", TruncateIP(ip, 64), 64);
>+  Network ipv6_network("test_eth0", "Test NetworkAdapter", TruncateIP(ip, 64),
>+                       64);
> 
>   // When there is no address added, it should return an unspecified
>   // address.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/networkmonitor.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/networkmonitor.cc
>index 02729513517..ad6805acec7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/networkmonitor.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/networkmonitor.cc
>@@ -39,6 +39,11 @@ void NetworkMonitorBase::OnMessage(Message* msg) {
>   SignalNetworksChanged();
> }
> 
>+AdapterType NetworkMonitorBase::GetVpnUnderlyingAdapterType(
>+    const std::string& interface_name) {
>+  return ADAPTER_TYPE_UNKNOWN;
>+}
>+
> NetworkMonitorFactory::NetworkMonitorFactory() {}
> NetworkMonitorFactory::~NetworkMonitorFactory() {}
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/networkmonitor.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/networkmonitor.h
>index 254b22575a5..a84a30aed2b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/networkmonitor.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/networkmonitor.h
>@@ -13,7 +13,7 @@
> 
> #include "rtc_base/logging.h"
> #include "rtc_base/network_constants.h"
>-#include "rtc_base/sigslot.h"
>+#include "rtc_base/third_party/sigslot/sigslot.h"
> #include "rtc_base/thread.h"
> 
> namespace rtc {
>@@ -74,6 +74,8 @@ class NetworkMonitorInterface {
>   virtual void OnNetworksChanged() = 0;
> 
>   virtual AdapterType GetAdapterType(const std::string& interface_name) = 0;
>+  virtual AdapterType GetVpnUnderlyingAdapterType(
>+      const std::string& interface_name) = 0;
> };
> 
> class NetworkMonitorBase : public NetworkMonitorInterface,
>@@ -87,6 +89,9 @@ class NetworkMonitorBase : public NetworkMonitorInterface,
> 
>   void OnMessage(Message* msg) override;
> 
>+  AdapterType GetVpnUnderlyingAdapterType(
>+      const std::string& interface_name) override;
>+
>  protected:
>   Thread* worker_thread() { return worker_thread_; }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/networkroute.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/networkroute.h
>index 07cba6324cc..31c083135c5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/networkroute.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/networkroute.h
>@@ -11,6 +11,8 @@
> #ifndef RTC_BASE_NETWORKROUTE_H_
> #define RTC_BASE_NETWORKROUTE_H_
> 
>+#include <stdint.h>
>+
> // TODO(honghaiz): Make a directory that describes the interfaces and structs
> // the media code can rely on and the network code can implement, and both can
> // depend on that, but not depend on each other. Then, move this file to that
>@@ -18,30 +20,13 @@
> namespace rtc {
> 
> struct NetworkRoute {
>-  bool connected;
>-  uint16_t local_network_id;
>-  uint16_t remote_network_id;
>-  int last_sent_packet_id;  // Last packet id sent on the PREVIOUS route.
>-  int packet_overhead;      // The overhead in bytes from IP layer and above.
>-
>-  NetworkRoute()
>-      : connected(false),
>-        local_network_id(0),
>-        remote_network_id(0),
>-        last_sent_packet_id(-1),
>-        packet_overhead(0) {}
>-
>-  // The route is connected if the local and remote network ids are provided.
>-  // TODO(zhihuang): Remove this and let the caller set the fields explicitly.
>-  NetworkRoute(bool connected,
>-               uint16_t local_net_id,
>-               uint16_t remote_net_id,
>-               int last_packet_id)
>-      : connected(connected),
>-        local_network_id(local_net_id),
>-        remote_network_id(remote_net_id),
>-        last_sent_packet_id(last_packet_id),
>-        packet_overhead(0) {}
>+  bool connected = false;
>+  uint16_t local_network_id = 0;
>+  uint16_t remote_network_id = 0;
>+  // Last packet id sent on the PREVIOUS route.
>+  int last_sent_packet_id = -1;
>+  // The overhead in bytes from IP layer and above.
>+  int packet_overhead = 0;
> 
>   // |last_sent_packet_id| and |packet_overhead| do not affect the NetworkRoute
>   // comparison.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/nullsocketserver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/nullsocketserver.cc
>index 68b67fa4c8e..c890c6f3dcf 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/nullsocketserver.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/nullsocketserver.cc
>@@ -25,21 +25,11 @@ void NullSocketServer::WakeUp() {
>   event_.Set();
> }
> 
>-rtc::Socket* NullSocketServer::CreateSocket(int /* type */) {
>-  RTC_NOTREACHED();
>-  return nullptr;
>-}
>-
> rtc::Socket* NullSocketServer::CreateSocket(int /* family */, int /* type */) {
>   RTC_NOTREACHED();
>   return nullptr;
> }
> 
>-rtc::AsyncSocket* NullSocketServer::CreateAsyncSocket(int /* type */) {
>-  RTC_NOTREACHED();
>-  return nullptr;
>-}
>-
> rtc::AsyncSocket* NullSocketServer::CreateAsyncSocket(int /* family */,
>                                                       int /* type */) {
>   RTC_NOTREACHED();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/nullsocketserver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/nullsocketserver.h
>index 7715c5cb7f8..408bcd1b2f5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/nullsocketserver.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/nullsocketserver.h
>@@ -24,9 +24,7 @@ class NullSocketServer : public SocketServer {
>   bool Wait(int cms, bool process_io) override;
>   void WakeUp() override;
> 
>-  Socket* CreateSocket(int type) override;
>   Socket* CreateSocket(int family, int type) override;
>-  AsyncSocket* CreateAsyncSocket(int type) override;
>   AsyncSocket* CreateAsyncSocket(int family, int type) override;
> 
>  private:
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/nullsocketserver_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/nullsocketserver_unittest.cc
>index e3d9952f77d..f1d65eab6a5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/nullsocketserver_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/nullsocketserver_unittest.cc
>@@ -15,9 +15,7 @@ namespace rtc {
> 
> static const uint32_t kTimeout = 5000U;
> 
>-class NullSocketServerTest
>-    : public testing::Test,
>-      public MessageHandler {
>+class NullSocketServerTest : public testing::Test, public MessageHandler {
>  protected:
>   void OnMessage(Message* message) override { ss_.WakeUp(); }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/histogram_percentile_counter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/histogram_percentile_counter.cc
>index 87ebd53d476..4bc8cb0dec2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/histogram_percentile_counter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/histogram_percentile_counter.cc
>@@ -48,12 +48,12 @@ void HistogramPercentileCounter::Add(uint32_t value) {
>   Add(value, 1);
> }
> 
>-rtc::Optional<uint32_t> HistogramPercentileCounter::GetPercentile(
>+absl::optional<uint32_t> HistogramPercentileCounter::GetPercentile(
>     float fraction) {
>   RTC_CHECK_LE(fraction, 1.0);
>   RTC_CHECK_GE(fraction, 0.0);
>   if (total_elements_ == 0)
>-    return rtc::nullopt;
>+    return absl::nullopt;
>   size_t elements_to_skip = static_cast<size_t>(
>       std::max(0.0f, std::ceil(total_elements_ * fraction) - 1));
>   if (elements_to_skip >= total_elements_)
>@@ -73,7 +73,7 @@ rtc::Optional<uint32_t> HistogramPercentileCounter::GetPercentile(
>     }
>   }
>   RTC_NOTREACHED();
>-  return rtc::nullopt;
>+  return absl::nullopt;
> }
> 
> }  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/histogram_percentile_counter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/histogram_percentile_counter.h
>index 4ad2e5377d4..7e5743fc628 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/histogram_percentile_counter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/histogram_percentile_counter.h
>@@ -15,7 +15,7 @@
> #include <map>
> #include <vector>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> 
> namespace rtc {
> // Calculates percentiles on the stream of data. Use |Add| methods to add new
>@@ -30,7 +30,7 @@ class HistogramPercentileCounter {
>   void Add(uint32_t value, size_t count);
>   void Add(const HistogramPercentileCounter& other);
>   // Argument should be from 0 to 1.
>-  rtc::Optional<uint32_t> GetPercentile(float fraction);
>+  absl::optional<uint32_t> GetPercentile(float fraction);
> 
>  private:
>   std::vector<size_t> histogram_low_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/moving_max_counter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/moving_max_counter.h
>index 4595cf35510..6c6286dfd4b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/moving_max_counter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/moving_max_counter.h
>@@ -17,7 +17,7 @@
> #include <limits>
> #include <utility>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/constructormagic.h"
> 
>@@ -40,7 +40,7 @@ class MovingMaxCounter {
>   // Advances the current time, and returns the maximum sample in the time
>   // window ending at the current time. The new current time must be at least as
>   // large as the old current time.
>-  rtc::Optional<T> Max(int64_t current_time_ms);
>+  absl::optional<T> Max(int64_t current_time_ms);
>   void Reset();
> 
>  private:
>@@ -83,9 +83,9 @@ void MovingMaxCounter<T>::Add(const T& sample, int64_t current_time_ms) {
> }
> 
> template <class T>
>-rtc::Optional<T> MovingMaxCounter<T>::Max(int64_t current_time_ms) {
>+absl::optional<T> MovingMaxCounter<T>::Max(int64_t current_time_ms) {
>   RollWindow(current_time_ms);
>-  rtc::Optional<T> res;
>+  absl::optional<T> res;
>   if (!samples_.empty()) {
>     res.emplace(samples_.front().second);
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/safe_conversions_impl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/safe_conversions_impl.h
>index 0174a83d2f5..9b4f1c6483d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/safe_conversions_impl.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/safe_conversions_impl.h
>@@ -101,7 +101,7 @@ struct RangeCheckImpl {};
> // Dst range always contains the result: nothing to check.
> template <typename Dst, typename Src, DstSign IsDstSigned, SrcSign IsSrcSigned>
> struct RangeCheckImpl<Dst, Src, IsDstSigned, IsSrcSigned, CONTAINS_RANGE> {
>-  static RangeCheckResult Check(Src) { return TYPE_VALID; }
>+  static RangeCheckResult Check(Src value) { return TYPE_VALID; }
> };
> 
> // Signed to signed narrowing.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/sample_counter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/sample_counter.cc
>new file mode 100644
>index 00000000000..d9244c385ad
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/sample_counter.cc
>@@ -0,0 +1,82 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <limits>
>+
>+#include "rtc_base/checks.h"
>+#include "rtc_base/numerics/safe_conversions.h"
>+#include "rtc_base/numerics/sample_counter.h"
>+
>+namespace rtc {
>+
>+SampleCounter::SampleCounter() = default;
>+SampleCounter::~SampleCounter() = default;
>+
>+void SampleCounter::Add(int sample) {
>+  if (sum_ > 0) {
>+    RTC_DCHECK_LE(sample, std::numeric_limits<int64_t>::max() - sum_);
>+  } else {
>+    RTC_DCHECK_GE(sample, std::numeric_limits<int64_t>::min() - sum_);
>+  }
>+  sum_ += sample;
>+  // Prevent overflow in squaring.
>+  RTC_DCHECK_GT(sample, std::numeric_limits<int32_t>::min());
>+  RTC_DCHECK_LE(int64_t{sample} * sample,
>+                std::numeric_limits<int64_t>::max() - sum_squared_);
>+  sum_squared_ += int64_t{sample} * sample;
>+  ++num_samples_;
>+  if (!max_ || sample > *max_) {
>+    max_ = sample;
>+  }
>+}
>+
>+void SampleCounter::Add(const SampleCounter& other) {
>+  if (sum_ > 0) {
>+    RTC_DCHECK_LE(other.sum_, std::numeric_limits<int64_t>::max() - sum_);
>+  } else {
>+    RTC_DCHECK_GE(other.sum_, std::numeric_limits<int64_t>::min() - sum_);
>+  }
>+  sum_ += other.sum_;
>+  RTC_DCHECK_LE(other.sum_squared_,
>+                std::numeric_limits<int64_t>::max() - sum_squared_);
>+  sum_squared_ += other.sum_squared_;
>+  RTC_DCHECK_LE(other.num_samples_,
>+                std::numeric_limits<int64_t>::max() - num_samples_);
>+  num_samples_ += other.num_samples_;
>+  if (other.max_ && (!max_ || *max_ < *other.max_))
>+    max_ = other.max_;
>+}
>+
>+absl::optional<int> SampleCounter::Avg(int64_t min_required_samples) const {
>+  RTC_DCHECK_GT(min_required_samples, 0);
>+  if (num_samples_ < min_required_samples)
>+    return absl::nullopt;
>+  return rtc::dchecked_cast<int>(sum_ / num_samples_);
>+}
>+
>+absl::optional<int64_t> SampleCounter::Variance(
>+    int64_t min_required_samples) const {
>+  RTC_DCHECK_GT(min_required_samples, 0);
>+  if (num_samples_ < min_required_samples)
>+    return absl::nullopt;
>+  // E[(x-mean)^2] = E[x^2] - mean^2
>+  int64_t mean = sum_ / num_samples_;
>+  return sum_squared_ / num_samples_ - mean * mean;
>+}
>+
>+absl::optional<int> SampleCounter::Max() const {
>+  return max_;
>+}
>+
>+void SampleCounter::Reset() {
>+  *this = {};
>+}
>+
>+}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/sample_counter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/sample_counter.h
>new file mode 100644
>index 00000000000..643754e8f94
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/sample_counter.h
>@@ -0,0 +1,41 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef RTC_BASE_NUMERICS_SAMPLE_COUNTER_H_
>+#define RTC_BASE_NUMERICS_SAMPLE_COUNTER_H_
>+
>+#include "absl/types/optional.h"
>+
>+namespace rtc {
>+
>+// Simple utility class for counting basic statistics (max./avg./variance) on
>+// stream of samples.
>+class SampleCounter {
>+ public:
>+  SampleCounter();
>+  ~SampleCounter();
>+  void Add(int sample);
>+  absl::optional<int> Avg(int64_t min_required_samples) const;
>+  absl::optional<int64_t> Variance(int64_t min_required_samples) const;
>+  absl::optional<int> Max() const;
>+  void Reset();
>+  // Adds all the samples from the |other| SampleCounter as if they were all
>+  // individually added using |Add(int)| method.
>+  void Add(const SampleCounter& other);
>+
>+ private:
>+  int64_t sum_ = 0;
>+  int64_t sum_squared_ = 0;
>+  int64_t num_samples_ = 0;
>+  absl::optional<int> max_;
>+};
>+
>+}  // namespace rtc
>+#endif  // RTC_BASE_NUMERICS_SAMPLE_COUNTER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/sample_counter_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/sample_counter_unittest.cc
>new file mode 100644
>index 00000000000..898f00cc62e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/sample_counter_unittest.cc
>@@ -0,0 +1,53 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "rtc_base/numerics/sample_counter.h"
>+
>+#include <utility>
>+#include <vector>
>+
>+#include "test/gmock.h"
>+#include "test/gtest.h"
>+
>+using testing::Eq;
>+
>+namespace rtc {
>+
>+TEST(SampleCounterTest, ProcessesNoSamples) {
>+  constexpr int kMinSamples = 1;
>+  SampleCounter counter;
>+  EXPECT_THAT(counter.Avg(kMinSamples), Eq(absl::nullopt));
>+  EXPECT_THAT(counter.Avg(kMinSamples), Eq(absl::nullopt));
>+  EXPECT_THAT(counter.Max(), Eq(absl::nullopt));
>+}
>+
>+TEST(SampleCounterTest, NotEnoughSamples) {
>+  constexpr int kMinSamples = 6;
>+  SampleCounter counter;
>+  for (int value : {1, 2, 3, 4, 5}) {
>+    counter.Add(value);
>+  }
>+  EXPECT_THAT(counter.Avg(kMinSamples), Eq(absl::nullopt));
>+  EXPECT_THAT(counter.Avg(kMinSamples), Eq(absl::nullopt));
>+  EXPECT_THAT(counter.Max(), Eq(5));
>+}
>+
>+TEST(SampleCounterTest, EnoughSamples) {
>+  constexpr int kMinSamples = 5;
>+  SampleCounter counter;
>+  for (int value : {1, 2, 3, 4, 5}) {
>+    counter.Add(value);
>+  }
>+  EXPECT_THAT(counter.Avg(kMinSamples), Eq(3));
>+  EXPECT_THAT(counter.Variance(kMinSamples), Eq(2));
>+  EXPECT_THAT(counter.Max(), Eq(5));
>+}
>+
>+}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/sequence_number_util.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/sequence_number_util.h
>index 9e4b8446f5f..c55513a3e47 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/sequence_number_util.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/sequence_number_util.h
>@@ -14,7 +14,7 @@
> #include <limits>
> #include <type_traits>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> #include "rtc_base/numerics/mod_ops.h"
> #include "rtc_base/numerics/safe_compare.h"
> 
>@@ -120,7 +120,7 @@ class SeqNumUnwrapper {
> 
>  private:
>   uint64_t last_unwrapped_;
>-  rtc::Optional<T> last_value_;
>+  absl::optional<T> last_value_;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/onetimeevent.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/onetimeevent.h
>index 8c55e2672d7..f649f15ded7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/onetimeevent.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/onetimeevent.h
>@@ -12,7 +12,6 @@
> #define RTC_BASE_ONETIMEEVENT_H_
> 
> #include "rtc_base/criticalsection.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> // Provides a simple way to perform an operation (such as logging) one
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/openssl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/openssl.h
>index a0325c972cd..eeed373c43e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/openssl.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/openssl.h
>@@ -11,10 +11,15 @@
> #ifndef RTC_BASE_OPENSSL_H_
> #define RTC_BASE_OPENSSL_H_
> 
>+#if defined(WEBRTC_WIN)
>+// Must be included first before openssl headers.
>+#include "rtc_base/win32.h"  // NOLINT
>+#endif                       // WEBRTC_WIN
>+
> #include <openssl/ssl.h>
> 
>-#if (OPENSSL_VERSION_NUMBER < 0x10000000L)
>-#error OpenSSL is older than 1.0.0, which is the minimum supported version.
>+#if (OPENSSL_VERSION_NUMBER < 0x10100000L)
>+#error OpenSSL is older than 1.1.0, which is the minimum supported version.
> #endif
> 
> #endif  // RTC_BASE_OPENSSL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/openssladapter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/openssladapter.cc
>index a064596881d..50284a6719f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/openssladapter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/openssladapter.cc
>@@ -14,9 +14,6 @@
> #include <unistd.h>
> #endif
> 
>-// Must be included first before openssl headers.
>-#include "rtc_base/win32.h"  // NOLINT
>-
> #include <openssl/bio.h>
> #include <openssl/crypto.h>
> #include <openssl/err.h>
>@@ -24,37 +21,36 @@
> #include <openssl/rand.h>
> #include <openssl/x509.h>
> #include <openssl/x509v3.h>
>+#include "rtc_base/openssl.h"
> 
>-#include "rtc_base/arraysize.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/numerics/safe_conversions.h"
>-#include "rtc_base/openssl.h"
>-#include "rtc_base/sslroots.h"
>+#include "rtc_base/opensslutility.h"
> #include "rtc_base/stringencode.h"
> #include "rtc_base/stringutils.h"
> #include "rtc_base/thread.h"
> 
> #ifndef OPENSSL_IS_BORINGSSL
> 
>-// TODO: Use a nicer abstraction for mutex.
>+// TODO(benwright): Use a nicer abstraction for mutex.
> 
> #if defined(WEBRTC_WIN)
>-  #define MUTEX_TYPE HANDLE
>+#define MUTEX_TYPE HANDLE
> #define MUTEX_SETUP(x) (x) = CreateMutex(nullptr, FALSE, nullptr)
> #define MUTEX_CLEANUP(x) CloseHandle(x)
> #define MUTEX_LOCK(x) WaitForSingleObject((x), INFINITE)
> #define MUTEX_UNLOCK(x) ReleaseMutex(x)
> #define THREAD_ID GetCurrentThreadId()
> #elif defined(WEBRTC_POSIX)
>-  #define MUTEX_TYPE pthread_mutex_t
>-  #define MUTEX_SETUP(x) pthread_mutex_init(&(x), nullptr)
>-  #define MUTEX_CLEANUP(x) pthread_mutex_destroy(&(x))
>-  #define MUTEX_LOCK(x) pthread_mutex_lock(&(x))
>-  #define MUTEX_UNLOCK(x) pthread_mutex_unlock(&(x))
>-  #define THREAD_ID pthread_self()
>+#define MUTEX_TYPE pthread_mutex_t
>+#define MUTEX_SETUP(x) pthread_mutex_init(&(x), nullptr)
>+#define MUTEX_CLEANUP(x) pthread_mutex_destroy(&(x))
>+#define MUTEX_LOCK(x) pthread_mutex_lock(&(x))
>+#define MUTEX_UNLOCK(x) pthread_mutex_unlock(&(x))
>+#define THREAD_ID pthread_self()
> #else
>-  #error You must define mutex operations appropriate for your platform!
>+#error You must define mutex operations appropriate for your platform!
> #endif
> 
> struct CRYPTO_dynlock_value {
>@@ -70,32 +66,37 @@ struct CRYPTO_dynlock_value {
> static int socket_write(BIO* h, const char* buf, int num);
> static int socket_read(BIO* h, char* buf, int size);
> static int socket_puts(BIO* h, const char* str);
>-static long socket_ctrl(BIO* h, int cmd, long arg1, void* arg2);
>+static long socket_ctrl(BIO* h, int cmd, long arg1, void* arg2);  // NOLINT
> static int socket_new(BIO* h);
> static int socket_free(BIO* data);
> 
>-// TODO(davidben): This should be const once BoringSSL is assumed.
>-static BIO_METHOD methods_socket = {
>-    BIO_TYPE_BIO, "socket",   socket_write, socket_read, socket_puts, 0,
>-    socket_ctrl,  socket_new, socket_free,  nullptr,
>-};
>-
>-static BIO_METHOD* BIO_s_socket2() { return(&methods_socket); }
>+static BIO_METHOD* BIO_socket_method() {
>+  static BIO_METHOD* methods = [] {
>+    BIO_METHOD* methods = BIO_meth_new(BIO_TYPE_BIO, "socket");
>+    BIO_meth_set_write(methods, socket_write);
>+    BIO_meth_set_read(methods, socket_read);
>+    BIO_meth_set_puts(methods, socket_puts);
>+    BIO_meth_set_ctrl(methods, socket_ctrl);
>+    BIO_meth_set_create(methods, socket_new);
>+    BIO_meth_set_destroy(methods, socket_free);
>+    return methods;
>+  }();
>+  return methods;
>+}
> 
> static BIO* BIO_new_socket(rtc::AsyncSocket* socket) {
>-  BIO* ret = BIO_new(BIO_s_socket2());
>+  BIO* ret = BIO_new(BIO_socket_method());
>   if (ret == nullptr) {
>     return nullptr;
>   }
>-  ret->ptr = socket;
>+  BIO_set_data(ret, socket);
>   return ret;
> }
> 
> static int socket_new(BIO* b) {
>-  b->shutdown = 0;
>-  b->init = 1;
>-  b->num = 0; // 1 means socket closed
>-  b->ptr = 0;
>+  BIO_set_shutdown(b, 0);
>+  BIO_set_init(b, 1);
>+  BIO_set_data(b, 0);
>   return 1;
> }
> 
>@@ -108,13 +109,11 @@ static int socket_free(BIO* b) {
> static int socket_read(BIO* b, char* out, int outl) {
>   if (!out)
>     return -1;
>-  rtc::AsyncSocket* socket = static_cast<rtc::AsyncSocket*>(b->ptr);
>+  rtc::AsyncSocket* socket = static_cast<rtc::AsyncSocket*>(BIO_get_data(b));
>   BIO_clear_retry_flags(b);
>   int result = socket->Recv(out, outl, nullptr);
>   if (result > 0) {
>     return result;
>-  } else if (result == 0) {
>-    b->num = 1;
>   } else if (socket->IsBlocking()) {
>     BIO_set_retry_read(b);
>   }
>@@ -124,7 +123,7 @@ static int socket_read(BIO* b, char* out, int outl) {
> static int socket_write(BIO* b, const char* in, int inl) {
>   if (!in)
>     return -1;
>-  rtc::AsyncSocket* socket = static_cast<rtc::AsyncSocket*>(b->ptr);
>+  rtc::AsyncSocket* socket = static_cast<rtc::AsyncSocket*>(BIO_get_data(b));
>   BIO_clear_retry_flags(b);
>   int result = socket->Send(in, inl);
>   if (result > 0) {
>@@ -139,19 +138,22 @@ static int socket_puts(BIO* b, const char* str) {
>   return socket_write(b, str, rtc::checked_cast<int>(strlen(str)));
> }
> 
>-static long socket_ctrl(BIO* b, int cmd, long num, void* ptr) {
>+static long socket_ctrl(BIO* b, int cmd, long num, void* ptr) {  // NOLINT
>   switch (cmd) {
>-  case BIO_CTRL_RESET:
>-    return 0;
>-  case BIO_CTRL_EOF:
>-    return b->num;
>-  case BIO_CTRL_WPENDING:
>-  case BIO_CTRL_PENDING:
>-    return 0;
>-  case BIO_CTRL_FLUSH:
>-    return 1;
>-  default:
>-    return 0;
>+    case BIO_CTRL_RESET:
>+      return 0;
>+    case BIO_CTRL_EOF: {
>+      rtc::AsyncSocket* socket = static_cast<rtc::AsyncSocket*>(ptr);
>+      // 1 means socket closed.
>+      return (socket->GetState() == rtc::AsyncSocket::CS_CLOSED) ? 1 : 0;
>+    }
>+    case BIO_CTRL_WPENDING:
>+    case BIO_CTRL_PENDING:
>+      return 0;
>+    case BIO_CTRL_FLUSH:
>+      return 1;
>+    default:
>+      return 0;
>   }
> }
> 
>@@ -176,56 +178,9 @@ static void LogSslError() {
> 
> namespace rtc {
> 
>-#ifndef OPENSSL_IS_BORINGSSL
>-
>-// This array will store all of the mutexes available to OpenSSL.
>-static MUTEX_TYPE* mutex_buf = nullptr;
>-
>-static void locking_function(int mode, int n, const char * file, int line) {
>-  if (mode & CRYPTO_LOCK) {
>-    MUTEX_LOCK(mutex_buf[n]);
>-  } else {
>-    MUTEX_UNLOCK(mutex_buf[n]);
>-  }
>-}
>-
>-static unsigned long id_function() {  // NOLINT
>-  // Use old-style C cast because THREAD_ID's type varies with the platform,
>-  // in some cases requiring static_cast, and in others requiring
>-  // reinterpret_cast.
>-  return (unsigned long)THREAD_ID; // NOLINT
>-}
>-
>-static CRYPTO_dynlock_value* dyn_create_function(const char* file, int line) {
>-  CRYPTO_dynlock_value* value = new CRYPTO_dynlock_value;
>-  if (!value)
>-    return nullptr;
>-  MUTEX_SETUP(value->mutex);
>-  return value;
>-}
>-
>-static void dyn_lock_function(int mode, CRYPTO_dynlock_value* l,
>-                              const char* file, int line) {
>-  if (mode & CRYPTO_LOCK) {
>-    MUTEX_LOCK(l->mutex);
>-  } else {
>-    MUTEX_UNLOCK(l->mutex);
>-  }
>-}
>-
>-static void dyn_destroy_function(CRYPTO_dynlock_value* l,
>-                                 const char* file, int line) {
>-  MUTEX_CLEANUP(l->mutex);
>-  delete l;
>-}
>-
>-#endif  // #ifndef OPENSSL_IS_BORINGSSL
>-
>-VerificationCallback OpenSSLAdapter::custom_verify_callback_ = nullptr;
>-
>-bool OpenSSLAdapter::InitializeSSL(VerificationCallback callback) {
>-  if (!InitializeSSLThread() || !SSL_library_init())
>-      return false;
>+bool OpenSSLAdapter::InitializeSSL() {
>+  if (!SSL_library_init())
>+    return false;
> #if !defined(ADDRESS_SANITIZER) || !defined(WEBRTC_MAC) || defined(WEBRTC_IOS)
>   // Loading the error strings crashes mac_asan.  Omit this debugging aid there.
>   SSL_load_error_strings();
>@@ -233,52 +188,19 @@ bool OpenSSLAdapter::InitializeSSL(VerificationCallback callback) {
>   ERR_load_BIO_strings();
>   OpenSSL_add_all_algorithms();
>   RAND_poll();
>-  custom_verify_callback_ = callback;
>-  return true;
>-}
>-
>-bool OpenSSLAdapter::InitializeSSLThread() {
>-  // BoringSSL is doing the locking internally, so the callbacks are not used
>-  // in this case (and are no-ops anyways).
>-#ifndef OPENSSL_IS_BORINGSSL
>-  mutex_buf = new MUTEX_TYPE[CRYPTO_num_locks()];
>-  if (!mutex_buf)
>-    return false;
>-  for (int i = 0; i < CRYPTO_num_locks(); ++i)
>-    MUTEX_SETUP(mutex_buf[i]);
>-
>-  // we need to cast our id_function to return an unsigned long -- pthread_t is
>-  // a pointer
>-  CRYPTO_set_id_callback(id_function);
>-  CRYPTO_set_locking_callback(locking_function);
>-  CRYPTO_set_dynlock_create_callback(dyn_create_function);
>-  CRYPTO_set_dynlock_lock_callback(dyn_lock_function);
>-  CRYPTO_set_dynlock_destroy_callback(dyn_destroy_function);
>-#endif  // #ifndef OPENSSL_IS_BORINGSSL
>   return true;
> }
> 
> bool OpenSSLAdapter::CleanupSSL() {
>-#ifndef OPENSSL_IS_BORINGSSL
>-  if (!mutex_buf)
>-    return false;
>-  CRYPTO_set_id_callback(nullptr);
>-  CRYPTO_set_locking_callback(nullptr);
>-  CRYPTO_set_dynlock_create_callback(nullptr);
>-  CRYPTO_set_dynlock_lock_callback(nullptr);
>-  CRYPTO_set_dynlock_destroy_callback(nullptr);
>-  for (int i = 0; i < CRYPTO_num_locks(); ++i)
>-    MUTEX_CLEANUP(mutex_buf[i]);
>-  delete [] mutex_buf;
>-  mutex_buf = nullptr;
>-#endif  // #ifndef OPENSSL_IS_BORINGSSL
>   return true;
> }
> 
> OpenSSLAdapter::OpenSSLAdapter(AsyncSocket* socket,
>-                               OpenSSLAdapterFactory* factory)
>+                               OpenSSLSessionCache* ssl_session_cache,
>+                               SSLCertificateVerifier* ssl_cert_verifier)
>     : SSLAdapter(socket),
>-      factory_(factory),
>+      ssl_session_cache_(ssl_session_cache),
>+      ssl_cert_verifier_(ssl_cert_verifier),
>       state_(SSL_NONE),
>       role_(SSL_CLIENT),
>       ssl_read_needs_write_(false),
>@@ -288,12 +210,12 @@ OpenSSLAdapter::OpenSSLAdapter(AsyncSocket* socket,
>       ssl_ctx_(nullptr),
>       ssl_mode_(SSL_MODE_TLS),
>       ignore_bad_cert_(false),
>-      custom_verification_succeeded_(false) {
>+      custom_cert_verifier_status_(false) {
>   // If a factory is used, take a reference on the factory's SSL_CTX.
>   // Otherwise, we'll create our own later.
>   // Either way, we'll release our reference via SSL_CTX_free() in Cleanup().
>-  if (factory_) {
>-    ssl_ctx_ = factory_->ssl_ctx();
>+  if (ssl_session_cache_ != nullptr) {
>+    ssl_ctx_ = ssl_session_cache_->GetSSLContext();
>     RTC_DCHECK(ssl_ctx_);
>     // Note: if using OpenSSL, requires version 1.1.0 or later.
>     SSL_CTX_up_ref(ssl_ctx_);
>@@ -322,6 +244,12 @@ void OpenSSLAdapter::SetMode(SSLMode mode) {
>   ssl_mode_ = mode;
> }
> 
>+void OpenSSLAdapter::SetCertVerifier(
>+    SSLCertificateVerifier* ssl_cert_verifier) {
>+  RTC_DCHECK(!ssl_ctx_);
>+  ssl_cert_verifier_ = ssl_cert_verifier;
>+}
>+
> void OpenSSLAdapter::SetIdentity(SSLIdentity* identity) {
>   RTC_DCHECK(!identity_);
>   identity_.reset(static_cast<OpenSSLIdentity*>(identity));
>@@ -377,10 +305,11 @@ int OpenSSLAdapter::BeginSSL() {
>   // First set up the context. We should either have a factory, with its own
>   // pre-existing context, or be running standalone, in which case we will
>   // need to create one, and specify |false| to disable session caching.
>-  if (!factory_) {
>+  if (ssl_session_cache_ == nullptr) {
>     RTC_DCHECK(!ssl_ctx_);
>     ssl_ctx_ = CreateContext(ssl_mode_, false);
>   }
>+
>   if (!ssl_ctx_) {
>     err = -1;
>     goto ssl_error;
>@@ -415,15 +344,15 @@ int OpenSSLAdapter::BeginSSL() {
>   // appear Send handles partial writes properly, though maybe we never notice
>   // since we never send more than 16KB at once..
>   SSL_set_mode(ssl_, SSL_MODE_ENABLE_PARTIAL_WRITE |
>-                     SSL_MODE_ACCEPT_MOVING_WRITE_BUFFER);
>+                         SSL_MODE_ACCEPT_MOVING_WRITE_BUFFER);
> 
>   // Enable SNI, if a hostname is supplied.
>   if (!ssl_host_name_.empty()) {
>     SSL_set_tlsext_host_name(ssl_, ssl_host_name_.c_str());
> 
>     // Enable session caching, if configured and a hostname is supplied.
>-    if (factory_) {
>-      SSL_SESSION* cached = factory_->LookupSession(ssl_host_name_);
>+    if (ssl_session_cache_ != nullptr) {
>+      SSL_SESSION* cached = ssl_session_cache_->LookupSession(ssl_host_name_);
>       if (cached) {
>         if (SSL_set_session(ssl_, cached) == 0) {
>           RTC_LOG(LS_WARNING) << "Failed to apply SSL session from cache";
>@@ -437,16 +366,18 @@ int OpenSSLAdapter::BeginSSL() {
>     }
>   }
> 
>+#ifdef OPENSSL_IS_BORINGSSL
>   // Set a couple common TLS extensions; even though we don't use them yet.
>   SSL_enable_ocsp_stapling(ssl_);
>   SSL_enable_signed_cert_timestamps(ssl_);
>+#endif
> 
>   if (!alpn_protocols_.empty()) {
>     std::string tls_alpn_string = TransformAlpnProtocols(alpn_protocols_);
>     if (!tls_alpn_string.empty()) {
>       SSL_set_alpn_protos(
>           ssl_, reinterpret_cast<const unsigned char*>(tls_alpn_string.data()),
>-          tls_alpn_string.size());
>+          rtc::dchecked_cast<unsigned>(tls_alpn_string.size()));
>     }
>   }
> 
>@@ -482,18 +413,18 @@ int OpenSSLAdapter::ContinueSSL() {
> 
>   int code = (role_ == SSL_CLIENT) ? SSL_connect(ssl_) : SSL_accept(ssl_);
>   switch (SSL_get_error(ssl_, code)) {
>-  case SSL_ERROR_NONE:
>-    if (!SSLPostConnectionCheck(ssl_, ssl_host_name_.c_str())) {
>-      RTC_LOG(LS_ERROR) << "TLS post connection check failed";
>-      // make sure we close the socket
>-      Cleanup();
>-      // The connect failed so return -1 to shut down the socket
>-      return -1;
>-    }
>+    case SSL_ERROR_NONE:
>+      if (!SSLPostConnectionCheck(ssl_, ssl_host_name_)) {
>+        RTC_LOG(LS_ERROR) << "TLS post connection check failed";
>+        // make sure we close the socket
>+        Cleanup();
>+        // The connect failed so return -1 to shut down the socket
>+        return -1;
>+      }
> 
>-    state_ = SSL_CONNECTED;
>-    AsyncSocketAdapter::OnConnectEvent(this);
>-#if 0  // TODO: worry about this
>+      state_ = SSL_CONNECTED;
>+      AsyncSocketAdapter::OnConnectEvent(this);
>+#if 0  // TODO(benwright): worry about this
>     // Don't let ourselves go away during the callbacks
>     PRefPtr<OpenSSLAdapter> lock(this);
>     RTC_LOG(LS_INFO) << " -- onStreamReadable";
>@@ -501,26 +432,26 @@ int OpenSSLAdapter::ContinueSSL() {
>     RTC_LOG(LS_INFO) << " -- onStreamWriteable";
>     AsyncSocketAdapter::OnWriteEvent(this);
> #endif
>-    break;
>+      break;
> 
>-  case SSL_ERROR_WANT_READ:
>-    RTC_LOG(LS_VERBOSE) << " -- error want read";
>-    struct timeval timeout;
>-    if (DTLSv1_get_timeout(ssl_, &timeout)) {
>-      int delay = timeout.tv_sec * 1000 + timeout.tv_usec/1000;
>+    case SSL_ERROR_WANT_READ:
>+      RTC_LOG(LS_VERBOSE) << " -- error want read";
>+      struct timeval timeout;
>+      if (DTLSv1_get_timeout(ssl_, &timeout)) {
>+        int delay = timeout.tv_sec * 1000 + timeout.tv_usec / 1000;
> 
>-      Thread::Current()->PostDelayed(RTC_FROM_HERE, delay, this, MSG_TIMEOUT,
>-                                     0);
>-    }
>-    break;
>+        Thread::Current()->PostDelayed(RTC_FROM_HERE, delay, this, MSG_TIMEOUT,
>+                                       0);
>+      }
>+      break;
> 
>-  case SSL_ERROR_WANT_WRITE:
>-    break;
>+    case SSL_ERROR_WANT_WRITE:
>+      break;
> 
>-  case SSL_ERROR_ZERO_RETURN:
>-  default:
>-    RTC_LOG(LS_WARNING) << "ContinueSSL -- error " << code;
>-    return (code != 0) ? code : -1;
>+    case SSL_ERROR_ZERO_RETURN:
>+    default:
>+      RTC_LOG(LS_WARNING) << "ContinueSSL -- error " << code;
>+      return (code != 0) ? code : -1;
>   }
> 
>   return 0;
>@@ -541,7 +472,7 @@ void OpenSSLAdapter::Cleanup() {
>   state_ = SSL_NONE;
>   ssl_read_needs_write_ = false;
>   ssl_write_needs_read_ = false;
>-  custom_verification_succeeded_ = false;
>+  custom_cert_verifier_status_ = false;
>   pending_data_.Clear();
> 
>   if (ssl_) {
>@@ -582,7 +513,6 @@ int OpenSSLAdapter::DoSslWrite(const void* pv, size_t cb, int* error) {
>       SetError(EWOULDBLOCK);
>       break;
>     case SSL_ERROR_ZERO_RETURN:
>-      // RTC_LOG(LS_INFO) << " -- remote side closed";
>       SetError(EWOULDBLOCK);
>       // do we need to signal closure?
>       break;
>@@ -591,7 +521,6 @@ int OpenSSLAdapter::DoSslWrite(const void* pv, size_t cb, int* error) {
>       Error("SSL_write", ret ? ret : -1, false);
>       break;
>     default:
>-      RTC_LOG(LS_WARNING) << "Unknown error from SSL_write: " << *error;
>       Error("SSL_write", ret ? ret : -1, false);
>       break;
>   }
>@@ -604,23 +533,21 @@ int OpenSSLAdapter::DoSslWrite(const void* pv, size_t cb, int* error) {
> //
> 
> int OpenSSLAdapter::Send(const void* pv, size_t cb) {
>-  // RTC_LOG(LS_INFO) << "OpenSSLAdapter::Send(" << cb << ")";
>-
>   switch (state_) {
>-  case SSL_NONE:
>-    return AsyncSocketAdapter::Send(pv, cb);
>+    case SSL_NONE:
>+      return AsyncSocketAdapter::Send(pv, cb);
> 
>-  case SSL_WAIT:
>-  case SSL_CONNECTING:
>-    SetError(ENOTCONN);
>-    return SOCKET_ERROR;
>+    case SSL_WAIT:
>+    case SSL_CONNECTING:
>+      SetError(ENOTCONN);
>+      return SOCKET_ERROR;
> 
>-  case SSL_CONNECTED:
>-    break;
>+    case SSL_CONNECTED:
>+      break;
> 
>-  case SSL_ERROR:
>-  default:
>-    return SOCKET_ERROR;
>+    case SSL_ERROR:
>+    default:
>+      return SOCKET_ERROR;
>   }
> 
>   int ret;
>@@ -666,7 +593,7 @@ int OpenSSLAdapter::Send(const void* pv, size_t cb) {
>     pending_data_.SetData(static_cast<const uint8_t*>(pv), cb);
>     // Since we're taking responsibility for sending this data, return its full
>     // size. The user of this class can consider it sent.
>-    return cb;
>+    return rtc::dchecked_cast<int>(cb);
>   }
> 
>   return ret;
>@@ -686,23 +613,21 @@ int OpenSSLAdapter::SendTo(const void* pv,
> }
> 
> int OpenSSLAdapter::Recv(void* pv, size_t cb, int64_t* timestamp) {
>-  // RTC_LOG(LS_INFO) << "OpenSSLAdapter::Recv(" << cb << ")";
>   switch (state_) {
>+    case SSL_NONE:
>+      return AsyncSocketAdapter::Recv(pv, cb, timestamp);
> 
>-  case SSL_NONE:
>-    return AsyncSocketAdapter::Recv(pv, cb, timestamp);
>-
>-  case SSL_WAIT:
>-  case SSL_CONNECTING:
>-    SetError(ENOTCONN);
>-    return SOCKET_ERROR;
>+    case SSL_WAIT:
>+    case SSL_CONNECTING:
>+      SetError(ENOTCONN);
>+      return SOCKET_ERROR;
> 
>-  case SSL_CONNECTED:
>-    break;
>+    case SSL_CONNECTED:
>+      break;
> 
>-  case SSL_ERROR:
>-  default:
>-    return SOCKET_ERROR;
>+    case SSL_ERROR:
>+    default:
>+      return SOCKET_ERROR;
>   }
> 
>   // Don't trust OpenSSL with zero byte reads
>@@ -715,19 +640,15 @@ int OpenSSLAdapter::Recv(void* pv, size_t cb, int64_t* timestamp) {
>   int error = SSL_get_error(ssl_, code);
>   switch (error) {
>     case SSL_ERROR_NONE:
>-      // RTC_LOG(LS_INFO) << " -- success";
>       return code;
>     case SSL_ERROR_WANT_READ:
>-      // RTC_LOG(LS_INFO) << " -- error want read";
>       SetError(EWOULDBLOCK);
>       break;
>     case SSL_ERROR_WANT_WRITE:
>-      // RTC_LOG(LS_INFO) << " -- error want write";
>       ssl_read_needs_write_ = true;
>       SetError(EWOULDBLOCK);
>       break;
>     case SSL_ERROR_ZERO_RETURN:
>-      // RTC_LOG(LS_INFO) << " -- remote side closed";
>       SetError(EWOULDBLOCK);
>       // do we need to signal closure?
>       break;
>@@ -736,7 +657,6 @@ int OpenSSLAdapter::Recv(void* pv, size_t cb, int64_t* timestamp) {
>       Error("SSL_read", (code ? code : -1), false);
>       break;
>     default:
>-      RTC_LOG(LS_WARNING) << "Unknown error from SSL_read: " << error;
>       Error("SSL_read", (code ? code : -1), false);
>       break;
>   }
>@@ -768,11 +688,11 @@ int OpenSSLAdapter::Close() {
> }
> 
> Socket::ConnState OpenSSLAdapter::GetState() const {
>-  //if (signal_close_)
>+  // if (signal_close_)
>   //  return CS_CONNECTED;
>   ConnState state = socket_->GetState();
>-  if ((state == CS_CONNECTED)
>-      && ((state_ == SSL_WAIT) || (state_ == SSL_CONNECTING)))
>+  if ((state == CS_CONNECTED) &&
>+      ((state_ == SSL_WAIT) || (state_ == SSL_CONNECTING)))
>     state = CS_CONNECTING;
>   return state;
> }
>@@ -804,8 +724,6 @@ void OpenSSLAdapter::OnConnectEvent(AsyncSocket* socket) {
> }
> 
> void OpenSSLAdapter::OnReadEvent(AsyncSocket* socket) {
>-  // RTC_LOG(LS_INFO) << "OpenSSLAdapter::OnReadEvent";
>-
>   if (state_ == SSL_NONE) {
>     AsyncSocketAdapter::OnReadEvent(socket);
>     return;
>@@ -822,19 +740,15 @@ void OpenSSLAdapter::OnReadEvent(AsyncSocket* socket) {
>     return;
> 
>   // Don't let ourselves go away during the callbacks
>-  //PRefPtr<OpenSSLAdapter> lock(this); // TODO: fix this
>-  if (ssl_write_needs_read_)  {
>-    // RTC_LOG(LS_INFO) << " -- onStreamWriteable";
>+  // PRefPtr<OpenSSLAdapter> lock(this); // TODO(benwright): fix this
>+  if (ssl_write_needs_read_) {
>     AsyncSocketAdapter::OnWriteEvent(socket);
>   }
> 
>-  // RTC_LOG(LS_INFO) << " -- onStreamReadable";
>   AsyncSocketAdapter::OnReadEvent(socket);
> }
> 
> void OpenSSLAdapter::OnWriteEvent(AsyncSocket* socket) {
>-  // RTC_LOG(LS_INFO) << "OpenSSLAdapter::OnWriteEvent";
>-
>   if (state_ == SSL_NONE) {
>     AsyncSocketAdapter::OnWriteEvent(socket);
>     return;
>@@ -851,10 +765,9 @@ void OpenSSLAdapter::OnWriteEvent(AsyncSocket* socket) {
>     return;
> 
>   // Don't let ourselves go away during the callbacks
>-  //PRefPtr<OpenSSLAdapter> lock(this); // TODO: fix this
>+  // PRefPtr<OpenSSLAdapter> lock(this); // TODO(benwright): fix this
> 
>-  if (ssl_read_needs_write_)  {
>-    // RTC_LOG(LS_INFO) << " -- onStreamReadable";
>+  if (ssl_read_needs_write_) {
>     AsyncSocketAdapter::OnReadEvent(socket);
>   }
> 
>@@ -868,7 +781,6 @@ void OpenSSLAdapter::OnWriteEvent(AsyncSocket* socket) {
>     }
>   }
> 
>-  // RTC_LOG(LS_INFO) << " -- onStreamWriteable";
>   AsyncSocketAdapter::OnWriteEvent(socket);
> }
> 
>@@ -877,95 +789,18 @@ void OpenSSLAdapter::OnCloseEvent(AsyncSocket* socket, int err) {
>   AsyncSocketAdapter::OnCloseEvent(socket, err);
> }
> 
>-bool OpenSSLAdapter::VerifyServerName(SSL* ssl, const char* host,
>-                                      bool ignore_bad_cert) {
>-  if (!host)
>-    return false;
>+bool OpenSSLAdapter::SSLPostConnectionCheck(SSL* ssl, const std::string& host) {
>+  bool is_valid_cert_name =
>+      openssl::VerifyPeerCertMatchesHost(ssl, host) &&
>+      (SSL_get_verify_result(ssl) == X509_V_OK || custom_cert_verifier_status_);
> 
>-  // Checking the return from SSL_get_peer_certificate here is not strictly
>-  // necessary.  With our setup, it is not possible for it to return
>-  // null.  However, it is good form to check the return.
>-  X509* certificate = SSL_get_peer_certificate(ssl);
>-  if (!certificate)
>-    return false;
>-
>-  // Logging certificates is extremely verbose. So it is disabled by default.
>-#ifdef LOG_CERTIFICATES
>-  {
>-    RTC_LOG(LS_INFO) << "Certificate from server:";
>-    BIO* mem = BIO_new(BIO_s_mem());
>-    X509_print_ex(mem, certificate, XN_FLAG_SEP_CPLUS_SPC, X509_FLAG_NO_HEADER);
>-    BIO_write(mem, "\0", 1);
>-    char* buffer;
>-    BIO_get_mem_data(mem, &buffer);
>-    RTC_LOG(LS_INFO) << buffer;
>-    BIO_free(mem);
>-
>-    char* cipher_description =
>-        SSL_CIPHER_description(SSL_get_current_cipher(ssl), nullptr, 128);
>-    RTC_LOG(LS_INFO) << "Cipher: " << cipher_description;
>-    OPENSSL_free(cipher_description);
>-  }
>-#endif
>-
>-  bool ok = false;
>-  GENERAL_NAMES* names = reinterpret_cast<GENERAL_NAMES*>(
>-      X509_get_ext_d2i(certificate, NID_subject_alt_name, nullptr, nullptr));
>-  if (names) {
>-    for (size_t i = 0; i < sk_GENERAL_NAME_num(names); i++) {
>-      const GENERAL_NAME* name = sk_GENERAL_NAME_value(names, i);
>-      if (name->type != GEN_DNS)
>-        continue;
>-      std::string value(
>-          reinterpret_cast<const char*>(ASN1_STRING_data(name->d.dNSName)),
>-          ASN1_STRING_length(name->d.dNSName));
>-      // string_match takes NUL-terminated strings, so check for embedded NULs.
>-      if (value.find('\0') != std::string::npos)
>-        continue;
>-      if (string_match(host, value.c_str())) {
>-        ok = true;
>-        break;
>-      }
>-    }
>-    GENERAL_NAMES_free(names);
>-  }
>-
>-  char data[256];
>-  X509_NAME* subject;
>-  if (!ok && ((subject = X509_get_subject_name(certificate)) != nullptr) &&
>-      (X509_NAME_get_text_by_NID(subject, NID_commonName, data, sizeof(data)) >
>-       0)) {
>-    data[sizeof(data)-1] = 0;
>-    if (_stricmp(data, host) == 0)
>-      ok = true;
>+  if (!is_valid_cert_name && ignore_bad_cert_) {
>+    RTC_DLOG(LS_WARNING) << "Other TLS post connection checks failed. "
>+                            "ignore_bad_cert_ set to true. Overriding name "
>+                            "verification failure!";
>+    is_valid_cert_name = true;
>   }
>-
>-  X509_free(certificate);
>-
>-  // This should only ever be turned on for debugging and development.
>-  if (!ok && ignore_bad_cert) {
>-    RTC_LOG(LS_WARNING) << "TLS certificate check FAILED.  "
>-                        << "Allowing connection anyway.";
>-    ok = true;
>-  }
>-
>-  return ok;
>-}
>-
>-bool OpenSSLAdapter::SSLPostConnectionCheck(SSL* ssl, const char* host) {
>-  bool ok = VerifyServerName(ssl, host, ignore_bad_cert_);
>-
>-  if (ok) {
>-    ok = (SSL_get_verify_result(ssl) == X509_V_OK ||
>-          custom_verification_succeeded_);
>-  }
>-
>-  if (!ok && ignore_bad_cert_) {
>-    RTC_LOG(LS_INFO) << "Other TLS post connection checks failed.";
>-    ok = true;
>-  }
>-
>-  return ok;
>+  return is_valid_cert_name;
> }
> 
> #if !defined(NDEBUG)
>@@ -981,17 +816,17 @@ void OpenSSLAdapter::SSLInfoCallback(const SSL* s, int where, int ret) {
>     str = "SSL_accept";
>   }
>   if (where & SSL_CB_LOOP) {
>-    RTC_LOG(LS_INFO) << str << ":" << SSL_state_string_long(s);
>+    RTC_DLOG(LS_INFO) << str << ":" << SSL_state_string_long(s);
>   } else if (where & SSL_CB_ALERT) {
>     str = (where & SSL_CB_READ) ? "read" : "write";
>-    RTC_LOG(LS_INFO) << "SSL3 alert " << str << ":"
>-                     << SSL_alert_type_string_long(ret) << ":"
>-                     << SSL_alert_desc_string_long(ret);
>+    RTC_DLOG(LS_INFO) << "SSL3 alert " << str << ":"
>+                      << SSL_alert_type_string_long(ret) << ":"
>+                      << SSL_alert_desc_string_long(ret);
>   } else if (where & SSL_CB_EXIT) {
>     if (ret == 0) {
>-      RTC_LOG(LS_INFO) << str << ":failed in " << SSL_state_string_long(s);
>+      RTC_DLOG(LS_INFO) << str << ":failed in " << SSL_state_string_long(s);
>     } else if (ret < 0) {
>-      RTC_LOG(LS_INFO) << str << ":error in " << SSL_state_string_long(s);
>+      RTC_DLOG(LS_INFO) << str << ":error in " << SSL_state_string_long(s);
>     }
>   }
> }
>@@ -1006,37 +841,37 @@ int OpenSSLAdapter::SSLVerifyCallback(int ok, X509_STORE_CTX* store) {
>     int depth = X509_STORE_CTX_get_error_depth(store);
>     int err = X509_STORE_CTX_get_error(store);
> 
>-    RTC_LOG(LS_INFO) << "Error with certificate at depth: " << depth;
>+    RTC_DLOG(LS_INFO) << "Error with certificate at depth: " << depth;
>     X509_NAME_oneline(X509_get_issuer_name(cert), data, sizeof(data));
>-    RTC_LOG(LS_INFO) << "  issuer  = " << data;
>+    RTC_DLOG(LS_INFO) << "  issuer  = " << data;
>     X509_NAME_oneline(X509_get_subject_name(cert), data, sizeof(data));
>-    RTC_LOG(LS_INFO) << "  subject = " << data;
>-    RTC_LOG(LS_INFO) << "  err     = " << err << ":"
>-                     << X509_verify_cert_error_string(err);
>+    RTC_DLOG(LS_INFO) << "  subject = " << data;
>+    RTC_DLOG(LS_INFO) << "  err     = " << err << ":"
>+                      << X509_verify_cert_error_string(err);
>   }
> #endif
>-
>   // Get our stream pointer from the store
>   SSL* ssl = reinterpret_cast<SSL*>(
>-                X509_STORE_CTX_get_ex_data(store,
>-                  SSL_get_ex_data_X509_STORE_CTX_idx()));
>+      X509_STORE_CTX_get_ex_data(store, SSL_get_ex_data_X509_STORE_CTX_idx()));
> 
>   OpenSSLAdapter* stream =
>-    reinterpret_cast<OpenSSLAdapter*>(SSL_get_app_data(ssl));
>-
>-  if (!ok && custom_verify_callback_) {
>-    void* cert =
>-        reinterpret_cast<void*>(X509_STORE_CTX_get_current_cert(store));
>-    if (custom_verify_callback_(cert)) {
>-      stream->custom_verification_succeeded_ = true;
>-      RTC_LOG(LS_INFO) << "validated certificate using custom callback";
>+      reinterpret_cast<OpenSSLAdapter*>(SSL_get_app_data(ssl));
>+
>+  if (!ok && stream->ssl_cert_verifier_ != nullptr) {
>+    RTC_LOG(LS_INFO) << "Invoking SSL Verify Callback.";
>+    const OpenSSLCertificate cert(X509_STORE_CTX_get_current_cert(store));
>+    if (stream->ssl_cert_verifier_->Verify(cert)) {
>+      stream->custom_cert_verifier_status_ = true;
>+      RTC_LOG(LS_INFO) << "Validated certificate using custom callback";
>       ok = true;
>+    } else {
>+      RTC_LOG(LS_INFO) << "Failed to verify certificate using custom callback";
>     }
>   }
> 
>   // Should only be used for debugging and development.
>   if (!ok && stream->ignore_bad_cert_) {
>-    RTC_LOG(LS_WARNING) << "Ignoring cert error while verifying cert chain";
>+    RTC_DLOG(LS_WARNING) << "Ignoring cert error while verifying cert chain";
>     ok = 1;
>   }
> 
>@@ -1046,33 +881,12 @@ int OpenSSLAdapter::SSLVerifyCallback(int ok, X509_STORE_CTX* store) {
> int OpenSSLAdapter::NewSSLSessionCallback(SSL* ssl, SSL_SESSION* session) {
>   OpenSSLAdapter* stream =
>       reinterpret_cast<OpenSSLAdapter*>(SSL_get_app_data(ssl));
>-  RTC_DCHECK(stream->factory_);
>+  RTC_DCHECK(stream->ssl_session_cache_);
>   RTC_LOG(LS_INFO) << "Caching SSL session for " << stream->ssl_host_name_;
>-  stream->factory_->AddSession(stream->ssl_host_name_, session);
>+  stream->ssl_session_cache_->AddSession(stream->ssl_host_name_, session);
>   return 1;  // We've taken ownership of the session; OpenSSL shouldn't free it.
> }
> 
>-bool OpenSSLAdapter::ConfigureTrustedRootCertificates(SSL_CTX* ctx) {
>-  // Add the root cert that we care about to the SSL context
>-  int count_of_added_certs = 0;
>-  for (size_t i = 0; i < arraysize(kSSLCertCertificateList); i++) {
>-    const unsigned char* cert_buffer = kSSLCertCertificateList[i];
>-    size_t cert_buffer_len = kSSLCertCertificateSizeList[i];
>-    X509* cert =
>-        d2i_X509(nullptr, &cert_buffer, checked_cast<long>(cert_buffer_len));
>-    if (cert) {
>-      int return_value = X509_STORE_add_cert(SSL_CTX_get_cert_store(ctx), cert);
>-      if (return_value == 0) {
>-        RTC_LOG(LS_WARNING) << "Unable to add certificate.";
>-      } else {
>-        count_of_added_certs++;
>-      }
>-      X509_free(cert);
>-    }
>-  }
>-  return count_of_added_certs > 0;
>-}
>-
> SSL_CTX* OpenSSLAdapter::CreateContext(SSLMode mode, bool enable_cache) {
>   // Use (D)TLS 1.2.
>   // Note: BoringSSL supports a range of versions by setting max/min version
>@@ -1092,10 +906,15 @@ SSL_CTX* OpenSSLAdapter::CreateContext(SSLMode mode, bool enable_cache) {
>                         << "(error=" << error << ')';
>     return nullptr;
>   }
>-  if (!ConfigureTrustedRootCertificates(ctx)) {
>+
>+#ifndef WEBRTC_EXCLUDE_BUILT_IN_SSL_ROOT_CERTS
>+  if (!openssl::LoadBuiltinSSLRootCertificates(ctx)) {
>+    RTC_LOG(LS_ERROR) << "SSL_CTX creation failed: Failed to load any trusted "
>+                         "ssl root certificates.";
>     SSL_CTX_free(ctx);
>     return nullptr;
>   }
>+#endif  // WEBRTC_EXCLUDE_BUILT_IN_SSL_ROOT_CERTS
> 
> #if !defined(NDEBUG)
>   SSL_CTX_set_info_callback(ctx, SSLInfoCallback);
>@@ -1147,43 +966,34 @@ std::string TransformAlpnProtocols(
> // OpenSSLAdapterFactory
> //////////////////////////////////////////////////////////////////////
> 
>-OpenSSLAdapterFactory::OpenSSLAdapterFactory()
>-    : ssl_mode_(SSL_MODE_TLS), ssl_ctx_(nullptr) {}
>+OpenSSLAdapterFactory::OpenSSLAdapterFactory() = default;
> 
>-OpenSSLAdapterFactory::~OpenSSLAdapterFactory() {
>-  for (auto it : sessions_) {
>-    SSL_SESSION_free(it.second);
>-  }
>-  SSL_CTX_free(ssl_ctx_);
>-}
>+OpenSSLAdapterFactory::~OpenSSLAdapterFactory() = default;
> 
> void OpenSSLAdapterFactory::SetMode(SSLMode mode) {
>-  RTC_DCHECK(!ssl_ctx_);
>+  RTC_DCHECK(!ssl_session_cache_);
>   ssl_mode_ = mode;
> }
> 
>+void OpenSSLAdapterFactory::SetCertVerifier(
>+    SSLCertificateVerifier* ssl_cert_verifier) {
>+  RTC_DCHECK(!ssl_session_cache_);
>+  ssl_cert_verifier_ = ssl_cert_verifier;
>+}
>+
> OpenSSLAdapter* OpenSSLAdapterFactory::CreateAdapter(AsyncSocket* socket) {
>-  if (!ssl_ctx_) {
>-    bool enable_cache = true;
>-    ssl_ctx_ = OpenSSLAdapter::CreateContext(ssl_mode_, enable_cache);
>-    if (!ssl_ctx_) {
>+  if (ssl_session_cache_ == nullptr) {
>+    SSL_CTX* ssl_ctx = OpenSSLAdapter::CreateContext(ssl_mode_, true);
>+    if (ssl_ctx == nullptr) {
>       return nullptr;
>     }
>+    // The OpenSSLSessionCache will upref the ssl_ctx.
>+    ssl_session_cache_ =
>+        absl::make_unique<OpenSSLSessionCache>(ssl_mode_, ssl_ctx);
>+    SSL_CTX_free(ssl_ctx);
>   }
>-
>-  return new OpenSSLAdapter(socket, this);
>-}
>-
>-SSL_SESSION* OpenSSLAdapterFactory::LookupSession(const std::string& hostname) {
>-  auto it = sessions_.find(hostname);
>-  return (it != sessions_.end()) ? it->second : nullptr;
>-}
>-
>-void OpenSSLAdapterFactory::AddSession(const std::string& hostname,
>-                                       SSL_SESSION* new_session) {
>-  SSL_SESSION* old_session = LookupSession(hostname);
>-  SSL_SESSION_free(old_session);
>-  sessions_[hostname] = new_session;
>+  return new OpenSSLAdapter(socket, ssl_session_cache_.get(),
>+                            ssl_cert_verifier_);
> }
> 
>-} // namespace rtc
>+}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/openssladapter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/openssladapter.h
>index 7148503d89d..50a7c0845ca 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/openssladapter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/openssladapter.h
>@@ -11,38 +11,45 @@
> #ifndef RTC_BASE_OPENSSLADAPTER_H_
> #define RTC_BASE_OPENSSLADAPTER_H_
> 
>+#include <openssl/ossl_typ.h>
>+
> #include <map>
>+#include <memory>
> #include <string>
>+#include <vector>
>+
>+#include "absl/memory/memory.h"
> #include "rtc_base/buffer.h"
> #include "rtc_base/messagehandler.h"
> #include "rtc_base/messagequeue.h"
>+#include "rtc_base/opensslcertificate.h"
> #include "rtc_base/opensslidentity.h"
>+#include "rtc_base/opensslsessioncache.h"
> #include "rtc_base/ssladapter.h"
> 
>-typedef struct ssl_st SSL;
>-typedef struct ssl_ctx_st SSL_CTX;
>-typedef struct x509_store_ctx_st X509_STORE_CTX;
>-typedef struct ssl_session_st SSL_SESSION;
>-
> namespace rtc {
> 
>-class OpenSSLAdapterFactory;
>-
> class OpenSSLAdapter : public SSLAdapter, public MessageHandler {
>  public:
>-  static bool InitializeSSL(VerificationCallback callback);
>-  static bool InitializeSSLThread();
>+  static bool InitializeSSL();
>   static bool CleanupSSL();
> 
>+  // Creating an OpenSSLAdapter requires a socket to bind to, an optional
>+  // session cache if you wish to improve performance by caching sessions for
>+  // hostnames you have previously connected to and an optional
>+  // SSLCertificateVerifier which can override any existing trusted roots to
>+  // validate a peer certificate. The cache and verifier are effectively
>+  // immutable after the the SSL connection starts.
>   explicit OpenSSLAdapter(AsyncSocket* socket,
>-                          OpenSSLAdapterFactory* factory = nullptr);
>+                          OpenSSLSessionCache* ssl_session_cache = nullptr,
>+                          SSLCertificateVerifier* ssl_cert_verifier = nullptr);
>   ~OpenSSLAdapter() override;
> 
>   void SetIgnoreBadCert(bool ignore) override;
>   void SetAlpnProtocols(const std::vector<std::string>& protos) override;
>   void SetEllipticCurves(const std::vector<std::string>& curves) override;
>-
>   void SetMode(SSLMode mode) override;
>+  void SetCertVerifier(SSLCertificateVerifier* ssl_cert_verifier) override;
>   void SetIdentity(SSLIdentity* identity) override;
>   void SetRole(SSLRole role) override;
>   AsyncSocket* Accept(SocketAddress* paddr) override;
>@@ -55,11 +62,9 @@ class OpenSSLAdapter : public SSLAdapter, public MessageHandler {
>                SocketAddress* paddr,
>                int64_t* timestamp) override;
>   int Close() override;
>-
>   // Note that the socket returns ST_CONNECTING while SSL is being negotiated.
>   ConnState GetState() const override;
>   bool IsResumedSession() override;
>-
>   // Creates a new SSL_CTX object, configured for client-to-server usage
>   // with SSLMode |mode|, and if |enable_cache| is true, with support for
>   // storing successful sessions so that they can be later resumed.
>@@ -76,7 +81,11 @@ class OpenSSLAdapter : public SSLAdapter, public MessageHandler {
> 
>  private:
>   enum SSLState {
>-    SSL_NONE, SSL_WAIT, SSL_CONNECTING, SSL_CONNECTED, SSL_ERROR
>+    SSL_NONE,
>+    SSL_WAIT,
>+    SSL_CONNECTING,
>+    SSL_CONNECTED,
>+    SSL_ERROR
>   };
> 
>   enum { MSG_TIMEOUT };
>@@ -89,18 +98,14 @@ class OpenSSLAdapter : public SSLAdapter, public MessageHandler {
>   // Return value and arguments have the same meanings as for Send; |error| is
>   // an output parameter filled with the result of SSL_get_error.
>   int DoSslWrite(const void* pv, size_t cb, int* error);
>-
>   void OnMessage(Message* msg) override;
>+  bool SSLPostConnectionCheck(SSL* ssl, const std::string& host);
> 
>-  static bool VerifyServerName(SSL* ssl, const char* host,
>-                               bool ignore_bad_cert);
>-  bool SSLPostConnectionCheck(SSL* ssl, const char* host);
> #if !defined(NDEBUG)
>   // In debug builds, logs info about the state of the SSL connection.
>   static void SSLInfoCallback(const SSL* ssl, int where, int ret);
> #endif
>   static int SSLVerifyCallback(int ok, X509_STORE_CTX* store);
>-  static VerificationCallback custom_verify_callback_;
>   friend class OpenSSLStreamAdapter;  // for custom_verify_callback_;
> 
>   // If the SSL_CTX was created with |enable_cache| set to true, this callback
>@@ -108,30 +113,30 @@ class OpenSSLAdapter : public SSLAdapter, public MessageHandler {
>   // to allow its SSL_SESSION* to be cached for later resumption.
>   static int NewSSLSessionCallback(SSL* ssl, SSL_SESSION* session);
> 
>-  static bool ConfigureTrustedRootCertificates(SSL_CTX* ctx);
>-
>-  // Parent object that maintains shared state.
>-  // Can be null if state sharing is not needed.
>-  OpenSSLAdapterFactory* factory_;
>-
>+  // Optional SSL Shared session cache to improve performance.
>+  OpenSSLSessionCache* ssl_session_cache_ = nullptr;
>+  // Optional SSL Certificate verifier which can be set by a third party.
>+  SSLCertificateVerifier* ssl_cert_verifier_ = nullptr;
>+  // The current connection state of the (d)TLS connection.
>   SSLState state_;
>   std::unique_ptr<OpenSSLIdentity> identity_;
>+  // Indicates whethere this is a client or a server.
>   SSLRole role_;
>   bool ssl_read_needs_write_;
>   bool ssl_write_needs_read_;
>   // If true, socket will retain SSL configuration after Close.
>   // TODO(juberti): Remove this unused flag.
>   bool restartable_;
>-
>   // This buffer is used if SSL_write fails with SSL_ERROR_WANT_WRITE, which
>   // means we need to keep retrying with *the same exact data* until it
>   // succeeds. Afterwards it will be cleared.
>   Buffer pending_data_;
>-
>   SSL* ssl_;
>+  // Holds the SSL context, which may be shared if an session cache is provided.
>   SSL_CTX* ssl_ctx_;
>+  // Hostname of server that is being connected, used for SNI.
>   std::string ssl_host_name_;
>-  // Do DTLS or not
>+  // Set the adapter to DTLS or TLS mode before creating the context.
>   SSLMode ssl_mode_;
>   // If true, the server certificate need not match the configured hostname.
>   bool ignore_bad_cert_;
>@@ -139,41 +144,46 @@ class OpenSSLAdapter : public SSLAdapter, public MessageHandler {
>   std::vector<std::string> alpn_protocols_;
>   // List of elliptic curves to be used in the TLS elliptic curves extension.
>   std::vector<std::string> elliptic_curves_;
>-
>-  bool custom_verification_succeeded_;
>+  // Holds the result of the call to run of the ssl_cert_verify_->Verify()
>+  bool custom_cert_verifier_status_;
> };
> 
>-std::string TransformAlpnProtocols(const std::vector<std::string>& protos);
>-
>-/////////////////////////////////////////////////////////////////////////////
>+// The OpenSSLAdapterFactory is responsbile for creating multiple new
>+// OpenSSLAdapters with a shared SSL_CTX and a shared SSL_SESSION cache. The
>+// SSL_SESSION cache allows existing SSL_SESSIONS to be reused instead of
>+// recreating them leading to a significant performance improvement.
> class OpenSSLAdapterFactory : public SSLAdapterFactory {
>  public:
>   OpenSSLAdapterFactory();
>   ~OpenSSLAdapterFactory() override;
>-
>+  // Set the SSL Mode to use with this factory. This should only be set before
>+  // the first adapter is created with the factory. If it is called after it
>+  // will DCHECK.
>   void SetMode(SSLMode mode) override;
>+  // Set a custom certificate verifier to be passed down to each instance
>+  // created with this factory. This should only ever be set before the first
>+  // call to the factory and cannot be changed after the fact.
>+  void SetCertVerifier(SSLCertificateVerifier* ssl_cert_verifier) override;
>+  // Constructs a new socket using the shared OpenSSLSessionCache. This means
>+  // existing SSLSessions already in the cache will be reused instead of
>+  // re-created for improved performance.
>   OpenSSLAdapter* CreateAdapter(AsyncSocket* socket) override;
> 
>-  static OpenSSLAdapterFactory* Create();
>-
>  private:
>-  SSL_CTX* ssl_ctx() { return ssl_ctx_; }
>-  // Looks up a session by hostname. The returned SSL_SESSION is not up_refed.
>-  SSL_SESSION* LookupSession(const std::string& hostname);
>-  // Adds a session to the cache, and up_refs it. Any existing session with the
>-  // same hostname is replaced.
>-  void AddSession(const std::string& hostname, SSL_SESSION* session);
>+  // Holds the SSLMode (DTLS,TLS) that will be used to set the session cache.
>+  SSLMode ssl_mode_ = SSL_MODE_TLS;
>+  // Holds a cache of existing SSL Sessions.
>+  std::unique_ptr<OpenSSLSessionCache> ssl_session_cache_;
>+  // Provides an optional custom callback for verifying SSL certificates, this
>+  // in currently only used for TLS-TURN connections.
>+  SSLCertificateVerifier* ssl_cert_verifier_ = nullptr;
>+  // TODO(benwright): Remove this when context is moved to OpenSSLCommon.
>+  // Hold a friend class to the OpenSSLAdapter to retrieve the context.
>   friend class OpenSSLAdapter;
>-
>-  SSLMode ssl_mode_;
>-  // Holds the shared SSL_CTX for all created adapters.
>-  SSL_CTX* ssl_ctx_;
>-  // Map of hostnames to SSL_SESSIONs; holds references to the SSL_SESSIONs,
>-  // which are cleaned up when the factory is destroyed.
>-  // TODO(juberti): Add LRU eviction to keep the cache from growing forever.
>-  std::map<std::string, SSL_SESSION*> sessions_;
> };
> 
>+std::string TransformAlpnProtocols(const std::vector<std::string>& protos);
>+
> }  // namespace rtc
> 
> #endif  // RTC_BASE_OPENSSLADAPTER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/openssladapter_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/openssladapter_unittest.cc
>index d043353ac1f..4db00e561bd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/openssladapter_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/openssladapter_unittest.cc
>@@ -12,10 +12,46 @@
> #include <string>
> #include <vector>
> 
>+#include "absl/memory/memory.h"
>+#include "rtc_base/asyncsocket.h"
> #include "rtc_base/gunit.h"
> #include "rtc_base/openssladapter.h"
>+#include "test/gmock.h"
> 
> namespace rtc {
>+namespace {
>+
>+class MockAsyncSocket : public AsyncSocket {
>+ public:
>+  virtual ~MockAsyncSocket() = default;
>+  MOCK_METHOD1(Accept, AsyncSocket*(SocketAddress*));
>+  MOCK_CONST_METHOD0(GetLocalAddress, SocketAddress());
>+  MOCK_CONST_METHOD0(GetRemoteAddress, SocketAddress());
>+  MOCK_METHOD1(Bind, int(const SocketAddress&));
>+  MOCK_METHOD1(Connect, int(const SocketAddress&));
>+  MOCK_METHOD2(Send, int(const void*, size_t));
>+  MOCK_METHOD3(SendTo, int(const void*, size_t, const SocketAddress&));
>+  MOCK_METHOD3(Recv, int(void*, size_t, int64_t*));
>+  MOCK_METHOD4(RecvFrom, int(void*, size_t, SocketAddress*, int64_t*));
>+  MOCK_METHOD1(Listen, int(int));
>+  MOCK_METHOD0(Close, int());
>+  MOCK_CONST_METHOD0(GetError, int());
>+  MOCK_METHOD1(SetError, void(int));
>+  MOCK_CONST_METHOD0(GetState, ConnState());
>+  MOCK_METHOD2(GetOption, int(Option, int*));
>+  MOCK_METHOD2(SetOption, int(Option, int));
>+};
>+
>+class MockCertVerifier : public SSLCertificateVerifier {
>+ public:
>+  virtual ~MockCertVerifier() = default;
>+  MOCK_METHOD1(Verify, bool(const SSLCertificate&));
>+};
>+
>+}  // namespace
>+
>+using ::testing::_;
>+using ::testing::Return;
> 
> TEST(OpenSSLAdapterTest, TestTransformAlpnProtocols) {
>   EXPECT_EQ("", TransformAlpnProtocols(std::vector<std::string>()));
>@@ -38,4 +74,36 @@ TEST(OpenSSLAdapterTest, TestTransformAlpnProtocols) {
>   EXPECT_EQ(expected_response.str(), TransformAlpnProtocols(alpn_protos));
> }
> 
>+// Verifies that SSLStart works when OpenSSLAdapter is started in standalone
>+// mode.
>+TEST(OpenSSLAdapterTest, TestBeginSSLBeforeConnection) {
>+  AsyncSocket* async_socket = new MockAsyncSocket();
>+  OpenSSLAdapter adapter(async_socket);
>+  EXPECT_EQ(adapter.StartSSL("webrtc.org", false), 0);
>+}
>+
>+// Verifies that the adapter factory can create new adapters.
>+TEST(OpenSSLAdapterFactoryTest, CreateSingleOpenSSLAdapter) {
>+  OpenSSLAdapterFactory adapter_factory;
>+  AsyncSocket* async_socket = new MockAsyncSocket();
>+  auto simple_adapter = std::unique_ptr<OpenSSLAdapter>(
>+      adapter_factory.CreateAdapter(async_socket));
>+  EXPECT_NE(simple_adapter, nullptr);
>+}
>+
>+// Verifies that setting a custom verifier still allows for adapters to be
>+// created.
>+TEST(OpenSSLAdapterFactoryTest, CreateWorksWithCustomVerifier) {
>+  MockCertVerifier* mock_verifier = new MockCertVerifier();
>+  EXPECT_CALL(*mock_verifier, Verify(_)).WillRepeatedly(Return(true));
>+  auto cert_verifier = std::unique_ptr<SSLCertificateVerifier>(mock_verifier);
>+
>+  OpenSSLAdapterFactory adapter_factory;
>+  adapter_factory.SetCertVerifier(cert_verifier.get());
>+  AsyncSocket* async_socket = new MockAsyncSocket();
>+  auto simple_adapter = std::unique_ptr<OpenSSLAdapter>(
>+      adapter_factory.CreateAdapter(async_socket));
>+  EXPECT_NE(simple_adapter, nullptr);
>+}
>+
> }  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslcertificate.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslcertificate.cc
>new file mode 100644
>index 00000000000..236bef55c8a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslcertificate.cc
>@@ -0,0 +1,321 @@
>+/*
>+ *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "rtc_base/opensslcertificate.h"
>+
>+#include <memory>
>+#include <utility>
>+#include <vector>
>+
>+#if defined(WEBRTC_WIN)
>+// Must be included first before openssl headers.
>+#include "rtc_base/win32.h"  // NOLINT
>+#endif                       // WEBRTC_WIN
>+
>+#include <openssl/bio.h>
>+#include <openssl/bn.h>
>+#include <openssl/crypto.h>
>+#include <openssl/err.h>
>+#include <openssl/pem.h>
>+#include <openssl/rsa.h>
>+
>+#include "absl/memory/memory.h"
>+#include "rtc_base/arraysize.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/helpers.h"
>+#include "rtc_base/logging.h"
>+#include "rtc_base/numerics/safe_conversions.h"
>+#include "rtc_base/openssl.h"
>+#include "rtc_base/openssldigest.h"
>+#include "rtc_base/opensslidentity.h"
>+#include "rtc_base/opensslutility.h"
>+#ifndef WEBRTC_EXCLUDE_BUILT_IN_SSL_ROOT_CERTS
>+#include "rtc_base/sslroots.h"
>+#endif  // WEBRTC_EXCLUDE_BUILT_IN_SSL_ROOT_CERTS
>+
>+namespace rtc {
>+
>+//////////////////////////////////////////////////////////////////////
>+// OpenSSLCertificate
>+//////////////////////////////////////////////////////////////////////
>+
>+// We could have exposed a myriad of parameters for the crypto stuff,
>+// but keeping it simple seems best.
>+
>+// Random bits for certificate serial number
>+static const int SERIAL_RAND_BITS = 64;
>+
>+// Generate a self-signed certificate, with the public key from the
>+// given key pair. Caller is responsible for freeing the returned object.
>+static X509* MakeCertificate(EVP_PKEY* pkey, const SSLIdentityParams& params) {
>+  RTC_LOG(LS_INFO) << "Making certificate for " << params.common_name;
>+  X509* x509 = nullptr;
>+  BIGNUM* serial_number = nullptr;
>+  X509_NAME* name = nullptr;
>+  time_t epoch_off = 0;  // Time offset since epoch.
>+
>+  if ((x509 = X509_new()) == nullptr)
>+    goto error;
>+
>+  if (!X509_set_pubkey(x509, pkey))
>+    goto error;
>+
>+  // serial number
>+  // temporary reference to serial number inside x509 struct
>+  ASN1_INTEGER* asn1_serial_number;
>+  if ((serial_number = BN_new()) == nullptr ||
>+      !BN_pseudo_rand(serial_number, SERIAL_RAND_BITS, 0, 0) ||
>+      (asn1_serial_number = X509_get_serialNumber(x509)) == nullptr ||
>+      !BN_to_ASN1_INTEGER(serial_number, asn1_serial_number))
>+    goto error;
>+
>+  if (!X509_set_version(x509, 2L))  // version 3
>+    goto error;
>+
>+  // There are a lot of possible components for the name entries. In
>+  // our P2P SSL mode however, the certificates are pre-exchanged
>+  // (through the secure XMPP channel), and so the certificate
>+  // identification is arbitrary. It can't be empty, so we set some
>+  // arbitrary common_name. Note that this certificate goes out in
>+  // clear during SSL negotiation, so there may be a privacy issue in
>+  // putting anything recognizable here.
>+  if ((name = X509_NAME_new()) == nullptr ||
>+      !X509_NAME_add_entry_by_NID(name, NID_commonName, MBSTRING_UTF8,
>+                                  (unsigned char*)params.common_name.c_str(),
>+                                  -1, -1, 0) ||
>+      !X509_set_subject_name(x509, name) || !X509_set_issuer_name(x509, name))
>+    goto error;
>+
>+  if (!X509_time_adj(X509_get_notBefore(x509), params.not_before, &epoch_off) ||
>+      !X509_time_adj(X509_get_notAfter(x509), params.not_after, &epoch_off))
>+    goto error;
>+
>+  if (!X509_sign(x509, pkey, EVP_sha256()))
>+    goto error;
>+
>+  BN_free(serial_number);
>+  X509_NAME_free(name);
>+  RTC_LOG(LS_INFO) << "Returning certificate";
>+  return x509;
>+
>+error:
>+  BN_free(serial_number);
>+  X509_NAME_free(name);
>+  X509_free(x509);
>+  return nullptr;
>+}
>+
>+#if !defined(NDEBUG)
>+// Print a certificate to the log, for debugging.
>+static void PrintCert(X509* x509) {
>+  BIO* temp_memory_bio = BIO_new(BIO_s_mem());
>+  if (!temp_memory_bio) {
>+    RTC_DLOG_F(LS_ERROR) << "Failed to allocate temporary memory bio";
>+    return;
>+  }
>+  X509_print_ex(temp_memory_bio, x509, XN_FLAG_SEP_CPLUS_SPC, 0);
>+  BIO_write(temp_memory_bio, "\0", 1);
>+  char* buffer;
>+  BIO_get_mem_data(temp_memory_bio, &buffer);
>+  RTC_DLOG(LS_VERBOSE) << buffer;
>+  BIO_free(temp_memory_bio);
>+}
>+#endif
>+
>+OpenSSLCertificate::OpenSSLCertificate(X509* x509) : x509_(x509) {
>+  AddReference();
>+}
>+
>+OpenSSLCertificate* OpenSSLCertificate::Generate(
>+    OpenSSLKeyPair* key_pair,
>+    const SSLIdentityParams& params) {
>+  SSLIdentityParams actual_params(params);
>+  if (actual_params.common_name.empty()) {
>+    // Use a random string, arbitrarily 8chars long.
>+    actual_params.common_name = CreateRandomString(8);
>+  }
>+  X509* x509 = MakeCertificate(key_pair->pkey(), actual_params);
>+  if (!x509) {
>+    openssl::LogSSLErrors("Generating certificate");
>+    return nullptr;
>+  }
>+#if !defined(NDEBUG)
>+  PrintCert(x509);
>+#endif
>+  OpenSSLCertificate* ret = new OpenSSLCertificate(x509);
>+  X509_free(x509);
>+  return ret;
>+}
>+
>+OpenSSLCertificate* OpenSSLCertificate::FromPEMString(
>+    const std::string& pem_string) {
>+  BIO* bio = BIO_new_mem_buf(const_cast<char*>(pem_string.c_str()), -1);
>+  if (!bio)
>+    return nullptr;
>+  BIO_set_mem_eof_return(bio, 0);
>+  X509* x509 =
>+      PEM_read_bio_X509(bio, nullptr, nullptr, const_cast<char*>("\0"));
>+  BIO_free(bio);  // Frees the BIO, but not the pointed-to string.
>+
>+  if (!x509)
>+    return nullptr;
>+
>+  OpenSSLCertificate* ret = new OpenSSLCertificate(x509);
>+  X509_free(x509);
>+  return ret;
>+}
>+
>+// NOTE: This implementation only functions correctly after InitializeSSL
>+// and before CleanupSSL.
>+bool OpenSSLCertificate::GetSignatureDigestAlgorithm(
>+    std::string* algorithm) const {
>+  int nid = X509_get_signature_nid(x509_);
>+  switch (nid) {
>+    case NID_md5WithRSA:
>+    case NID_md5WithRSAEncryption:
>+      *algorithm = DIGEST_MD5;
>+      break;
>+    case NID_ecdsa_with_SHA1:
>+    case NID_dsaWithSHA1:
>+    case NID_dsaWithSHA1_2:
>+    case NID_sha1WithRSA:
>+    case NID_sha1WithRSAEncryption:
>+      *algorithm = DIGEST_SHA_1;
>+      break;
>+    case NID_ecdsa_with_SHA224:
>+    case NID_sha224WithRSAEncryption:
>+    case NID_dsa_with_SHA224:
>+      *algorithm = DIGEST_SHA_224;
>+      break;
>+    case NID_ecdsa_with_SHA256:
>+    case NID_sha256WithRSAEncryption:
>+    case NID_dsa_with_SHA256:
>+      *algorithm = DIGEST_SHA_256;
>+      break;
>+    case NID_ecdsa_with_SHA384:
>+    case NID_sha384WithRSAEncryption:
>+      *algorithm = DIGEST_SHA_384;
>+      break;
>+    case NID_ecdsa_with_SHA512:
>+    case NID_sha512WithRSAEncryption:
>+      *algorithm = DIGEST_SHA_512;
>+      break;
>+    default:
>+      // Unknown algorithm.  There are several unhandled options that are less
>+      // common and more complex.
>+      RTC_LOG(LS_ERROR) << "Unknown signature algorithm NID: " << nid;
>+      algorithm->clear();
>+      return false;
>+  }
>+  return true;
>+}
>+
>+bool OpenSSLCertificate::ComputeDigest(const std::string& algorithm,
>+                                       unsigned char* digest,
>+                                       size_t size,
>+                                       size_t* length) const {
>+  return ComputeDigest(x509_, algorithm, digest, size, length);
>+}
>+
>+bool OpenSSLCertificate::ComputeDigest(const X509* x509,
>+                                       const std::string& algorithm,
>+                                       unsigned char* digest,
>+                                       size_t size,
>+                                       size_t* length) {
>+  const EVP_MD* md;
>+  unsigned int n;
>+
>+  if (!OpenSSLDigest::GetDigestEVP(algorithm, &md))
>+    return false;
>+
>+  if (size < static_cast<size_t>(EVP_MD_size(md)))
>+    return false;
>+
>+  X509_digest(x509, md, digest, &n);
>+
>+  *length = n;
>+
>+  return true;
>+}
>+
>+OpenSSLCertificate::~OpenSSLCertificate() {
>+  X509_free(x509_);
>+}
>+
>+OpenSSLCertificate* OpenSSLCertificate::GetReference() const {
>+  return new OpenSSLCertificate(x509_);
>+}
>+
>+std::string OpenSSLCertificate::ToPEMString() const {
>+  BIO* bio = BIO_new(BIO_s_mem());
>+  if (!bio) {
>+    RTC_FATAL() << "unreachable code";
>+  }
>+  if (!PEM_write_bio_X509(bio, x509_)) {
>+    BIO_free(bio);
>+    RTC_FATAL() << "unreachable code";
>+  }
>+  BIO_write(bio, "\0", 1);
>+  char* buffer;
>+  BIO_get_mem_data(bio, &buffer);
>+  std::string ret(buffer);
>+  BIO_free(bio);
>+  return ret;
>+}
>+
>+void OpenSSLCertificate::ToDER(Buffer* der_buffer) const {
>+  // In case of failure, make sure to leave the buffer empty.
>+  der_buffer->SetSize(0);
>+
>+  // Calculates the DER representation of the certificate, from scratch.
>+  BIO* bio = BIO_new(BIO_s_mem());
>+  if (!bio) {
>+    RTC_FATAL() << "unreachable code";
>+  }
>+  if (!i2d_X509_bio(bio, x509_)) {
>+    BIO_free(bio);
>+    RTC_FATAL() << "unreachable code";
>+  }
>+  char* data;
>+  size_t length = BIO_get_mem_data(bio, &data);
>+  der_buffer->SetData(data, length);
>+  BIO_free(bio);
>+}
>+
>+void OpenSSLCertificate::AddReference() const {
>+  RTC_DCHECK(x509_ != nullptr);
>+  X509_up_ref(x509_);
>+}
>+
>+bool OpenSSLCertificate::operator==(const OpenSSLCertificate& other) const {
>+  return X509_cmp(x509_, other.x509_) == 0;
>+}
>+
>+bool OpenSSLCertificate::operator!=(const OpenSSLCertificate& other) const {
>+  return !(*this == other);
>+}
>+
>+// Documented in sslidentity.h.
>+int64_t OpenSSLCertificate::CertificateExpirationTime() const {
>+  ASN1_TIME* expire_time = X509_get_notAfter(x509_);
>+  bool long_format;
>+
>+  if (expire_time->type == V_ASN1_UTCTIME) {
>+    long_format = false;
>+  } else if (expire_time->type == V_ASN1_GENERALIZEDTIME) {
>+    long_format = true;
>+  } else {
>+    return -1;
>+  }
>+
>+  return ASN1TimeToSec(expire_time->data, expire_time->length, long_format);
>+}
>+
>+}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslcertificate.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslcertificate.h
>new file mode 100644
>index 00000000000..c730ffd0dc9
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslcertificate.h
>@@ -0,0 +1,80 @@
>+/*
>+ *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef RTC_BASE_OPENSSLCERTIFICATE_H_
>+#define RTC_BASE_OPENSSLCERTIFICATE_H_
>+
>+#include <openssl/evp.h>
>+#include <openssl/x509.h>
>+
>+#include <memory>
>+#include <string>
>+
>+#include "rtc_base/checks.h"
>+#include "rtc_base/constructormagic.h"
>+#include "rtc_base/sslcertificate.h"
>+#include "rtc_base/sslidentity.h"
>+
>+typedef struct ssl_ctx_st SSL_CTX;
>+
>+namespace rtc {
>+
>+class OpenSSLKeyPair;
>+
>+// OpenSSLCertificate encapsulates an OpenSSL X509* certificate object,
>+// which is also reference counted inside the OpenSSL library.
>+class OpenSSLCertificate : public SSLCertificate {
>+ public:
>+  // X509 object has its reference count incremented. So the caller and
>+  // OpenSSLCertificate share ownership.
>+  explicit OpenSSLCertificate(X509* x509);
>+
>+  static OpenSSLCertificate* Generate(OpenSSLKeyPair* key_pair,
>+                                      const SSLIdentityParams& params);
>+  static OpenSSLCertificate* FromPEMString(const std::string& pem_string);
>+
>+  ~OpenSSLCertificate() override;
>+
>+  OpenSSLCertificate* GetReference() const override;
>+
>+  X509* x509() const { return x509_; }
>+
>+  std::string ToPEMString() const override;
>+  void ToDER(Buffer* der_buffer) const override;
>+  bool operator==(const OpenSSLCertificate& other) const;
>+  bool operator!=(const OpenSSLCertificate& other) const;
>+
>+  // Compute the digest of the certificate given algorithm
>+  bool ComputeDigest(const std::string& algorithm,
>+                     unsigned char* digest,
>+                     size_t size,
>+                     size_t* length) const override;
>+
>+  // Compute the digest of a certificate as an X509 *
>+  static bool ComputeDigest(const X509* x509,
>+                            const std::string& algorithm,
>+                            unsigned char* digest,
>+                            size_t size,
>+                            size_t* length);
>+
>+  bool GetSignatureDigestAlgorithm(std::string* algorithm) const override;
>+
>+  int64_t CertificateExpirationTime() const override;
>+
>+ private:
>+  void AddReference() const;
>+
>+  X509* x509_;  // NOT OWNED
>+  RTC_DISALLOW_COPY_AND_ASSIGN(OpenSSLCertificate);
>+};
>+
>+}  // namespace rtc
>+
>+#endif  // RTC_BASE_OPENSSLCERTIFICATE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/openssldigest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/openssldigest.cc
>index 3be82e61d38..9b644c460b5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/openssldigest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/openssldigest.cc
>@@ -16,16 +16,18 @@
> namespace rtc {
> 
> OpenSSLDigest::OpenSSLDigest(const std::string& algorithm) {
>-  EVP_MD_CTX_init(&ctx_);
>+  ctx_ = EVP_MD_CTX_new();
>+  RTC_CHECK(ctx_ != nullptr);
>+  EVP_MD_CTX_init(ctx_);
>   if (GetDigestEVP(algorithm, &md_)) {
>-    EVP_DigestInit_ex(&ctx_, md_, nullptr);
>+    EVP_DigestInit_ex(ctx_, md_, nullptr);
>   } else {
>     md_ = nullptr;
>   }
> }
> 
> OpenSSLDigest::~OpenSSLDigest() {
>-  EVP_MD_CTX_cleanup(&ctx_);
>+  EVP_MD_CTX_destroy(ctx_);
> }
> 
> size_t OpenSSLDigest::Size() const {
>@@ -39,7 +41,7 @@ void OpenSSLDigest::Update(const void* buf, size_t len) {
>   if (!md_) {
>     return;
>   }
>-  EVP_DigestUpdate(&ctx_, buf, len);
>+  EVP_DigestUpdate(ctx_, buf, len);
> }
> 
> size_t OpenSSLDigest::Finish(void* buf, size_t len) {
>@@ -47,8 +49,8 @@ size_t OpenSSLDigest::Finish(void* buf, size_t len) {
>     return 0;
>   }
>   unsigned int md_len;
>-  EVP_DigestFinal_ex(&ctx_, static_cast<unsigned char*>(buf), &md_len);
>-  EVP_DigestInit_ex(&ctx_, md_, nullptr);  // prepare for future Update()s
>+  EVP_DigestFinal_ex(ctx_, static_cast<unsigned char*>(buf), &md_len);
>+  EVP_DigestInit_ex(ctx_, md_, nullptr);  // prepare for future Update()s
>   RTC_DCHECK(md_len == Size());
>   return md_len;
> }
>@@ -78,8 +80,7 @@ bool OpenSSLDigest::GetDigestEVP(const std::string& algorithm,
>   return true;
> }
> 
>-bool OpenSSLDigest::GetDigestName(const EVP_MD* md,
>-                                  std::string* algorithm) {
>+bool OpenSSLDigest::GetDigestName(const EVP_MD* md, std::string* algorithm) {
>   RTC_DCHECK(md != nullptr);
>   RTC_DCHECK(algorithm != nullptr);
> 
>@@ -106,7 +107,7 @@ bool OpenSSLDigest::GetDigestName(const EVP_MD* md,
> 
> bool OpenSSLDigest::GetDigestSize(const std::string& algorithm,
>                                   size_t* length) {
>-  const EVP_MD *md;
>+  const EVP_MD* md;
>   if (!GetDigestEVP(algorithm, &md))
>     return false;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/openssldigest.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/openssldigest.h
>index d7966656751..c4cd1e0314e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/openssldigest.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/openssldigest.h
>@@ -31,17 +31,14 @@ class OpenSSLDigest : public MessageDigest {
>   size_t Finish(void* buf, size_t len) override;
> 
>   // Helper function to look up a digest's EVP by name.
>-  static bool GetDigestEVP(const std::string &algorithm,
>-                           const EVP_MD** md);
>+  static bool GetDigestEVP(const std::string& algorithm, const EVP_MD** md);
>   // Helper function to look up a digest's name by EVP.
>-  static bool GetDigestName(const EVP_MD* md,
>-                            std::string* algorithm);
>+  static bool GetDigestName(const EVP_MD* md, std::string* algorithm);
>   // Helper function to get the length of a digest.
>-  static bool GetDigestSize(const std::string &algorithm,
>-                            size_t* len);
>+  static bool GetDigestSize(const std::string& algorithm, size_t* len);
> 
>  private:
>-  EVP_MD_CTX ctx_;
>+  EVP_MD_CTX* ctx_ = nullptr;
>   const EVP_MD* md_;
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslidentity.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslidentity.cc
>index 7bc72638750..a8c6919779b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslidentity.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslidentity.cc
>@@ -11,9 +11,13 @@
> #include "rtc_base/opensslidentity.h"
> 
> #include <memory>
>+#include <utility>
>+#include <vector>
> 
>+#if defined(WEBRTC_WIN)
> // Must be included first before openssl headers.
> #include "rtc_base/win32.h"  // NOLINT
>+#endif                       // WEBRTC_WIN
> 
> #include <openssl/bio.h>
> #include <openssl/bn.h>
>@@ -22,21 +26,20 @@
> #include <openssl/pem.h>
> #include <openssl/rsa.h>
> 
>+#include "absl/memory/memory.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/helpers.h"
> #include "rtc_base/logging.h"
>+#include "rtc_base/numerics/safe_conversions.h"
> #include "rtc_base/openssl.h"
> #include "rtc_base/openssldigest.h"
>-#include "rtc_base/ptr_util.h"
>+#include "rtc_base/opensslutility.h"
> 
> namespace rtc {
> 
> // We could have exposed a myriad of parameters for the crypto stuff,
> // but keeping it simple seems best.
> 
>-// Random bits for certificate serial number
>-static const int SERIAL_RAND_BITS = 64;
>-
> // Generate a key pair. Caller is responsible for freeing the returned object.
> static EVP_PKEY* MakeKey(const KeyParams& key_params) {
>   RTC_LOG(LS_INFO) << "Making key pair";
>@@ -91,81 +94,10 @@ static EVP_PKEY* MakeKey(const KeyParams& key_params) {
>   return pkey;
> }
> 
>-// Generate a self-signed certificate, with the public key from the
>-// given key pair. Caller is responsible for freeing the returned object.
>-static X509* MakeCertificate(EVP_PKEY* pkey, const SSLIdentityParams& params) {
>-  RTC_LOG(LS_INFO) << "Making certificate for " << params.common_name;
>-  X509* x509 = nullptr;
>-  BIGNUM* serial_number = nullptr;
>-  X509_NAME* name = nullptr;
>-  time_t epoch_off = 0;  // Time offset since epoch.
>-
>-  if ((x509 = X509_new()) == nullptr)
>-    goto error;
>-
>-  if (!X509_set_pubkey(x509, pkey))
>-    goto error;
>-
>-  // serial number
>-  // temporary reference to serial number inside x509 struct
>-  ASN1_INTEGER* asn1_serial_number;
>-  if ((serial_number = BN_new()) == nullptr ||
>-      !BN_pseudo_rand(serial_number, SERIAL_RAND_BITS, 0, 0) ||
>-      (asn1_serial_number = X509_get_serialNumber(x509)) == nullptr ||
>-      !BN_to_ASN1_INTEGER(serial_number, asn1_serial_number))
>-    goto error;
>-
>-  if (!X509_set_version(x509, 2L))  // version 3
>-    goto error;
>-
>-  // There are a lot of possible components for the name entries. In
>-  // our P2P SSL mode however, the certificates are pre-exchanged
>-  // (through the secure XMPP channel), and so the certificate
>-  // identification is arbitrary. It can't be empty, so we set some
>-  // arbitrary common_name. Note that this certificate goes out in
>-  // clear during SSL negotiation, so there may be a privacy issue in
>-  // putting anything recognizable here.
>-  if ((name = X509_NAME_new()) == nullptr ||
>-      !X509_NAME_add_entry_by_NID(name, NID_commonName, MBSTRING_UTF8,
>-                                  (unsigned char*)params.common_name.c_str(),
>-                                  -1, -1, 0) ||
>-      !X509_set_subject_name(x509, name) || !X509_set_issuer_name(x509, name))
>-    goto error;
>-
>-  if (!X509_time_adj(X509_get_notBefore(x509), params.not_before, &epoch_off) ||
>-      !X509_time_adj(X509_get_notAfter(x509), params.not_after, &epoch_off))
>-    goto error;
>-
>-  if (!X509_sign(x509, pkey, EVP_sha256()))
>-    goto error;
>-
>-  BN_free(serial_number);
>-  X509_NAME_free(name);
>-  RTC_LOG(LS_INFO) << "Returning certificate";
>-  return x509;
>-
>-error:
>-  BN_free(serial_number);
>-  X509_NAME_free(name);
>-  X509_free(x509);
>-  return nullptr;
>-}
>-
>-// This dumps the SSL error stack to the log.
>-static void LogSSLErrors(const std::string& prefix) {
>-  char error_buf[200];
>-  unsigned long err;
>-
>-  while ((err = ERR_get_error()) != 0) {
>-    ERR_error_string_n(err, error_buf, sizeof(error_buf));
>-    RTC_LOG(LS_ERROR) << prefix << ": " << error_buf << "\n";
>-  }
>-}
>-
> OpenSSLKeyPair* OpenSSLKeyPair::Generate(const KeyParams& key_params) {
>   EVP_PKEY* pkey = MakeKey(key_params);
>   if (!pkey) {
>-    LogSSLErrors("Generating key pair");
>+    openssl::LogSSLErrors("Generating key pair");
>     return nullptr;
>   }
>   return new OpenSSLKeyPair(pkey);
>@@ -205,11 +137,7 @@ OpenSSLKeyPair* OpenSSLKeyPair::GetReference() {
> }
> 
> void OpenSSLKeyPair::AddReference() {
>-#if defined(OPENSSL_IS_BORINGSSL)
>   EVP_PKEY_up_ref(pkey_);
>-#else
>-  CRYPTO_add(&pkey_->references, 1, CRYPTO_LOCK_EVP_PKEY);
>-#endif
> }
> 
> std::string OpenSSLKeyPair::PrivateKeyToPEMString() const {
>@@ -263,220 +191,6 @@ bool OpenSSLKeyPair::operator!=(const OpenSSLKeyPair& other) const {
>   return !(*this == other);
> }
> 
>-#if !defined(NDEBUG)
>-// Print a certificate to the log, for debugging.
>-static void PrintCert(X509* x509) {
>-  BIO* temp_memory_bio = BIO_new(BIO_s_mem());
>-  if (!temp_memory_bio) {
>-    RTC_LOG_F(LS_ERROR) << "Failed to allocate temporary memory bio";
>-    return;
>-  }
>-  X509_print_ex(temp_memory_bio, x509, XN_FLAG_SEP_CPLUS_SPC, 0);
>-  BIO_write(temp_memory_bio, "\0", 1);
>-  char* buffer;
>-  BIO_get_mem_data(temp_memory_bio, &buffer);
>-  RTC_LOG(LS_VERBOSE) << buffer;
>-  BIO_free(temp_memory_bio);
>-}
>-#endif
>-
>-OpenSSLCertificate::OpenSSLCertificate(X509* x509) : x509_(x509) {
>-  AddReference();
>-}
>-
>-OpenSSLCertificate* OpenSSLCertificate::Generate(
>-    OpenSSLKeyPair* key_pair,
>-    const SSLIdentityParams& params) {
>-  SSLIdentityParams actual_params(params);
>-  if (actual_params.common_name.empty()) {
>-    // Use a random string, arbitrarily 8chars long.
>-    actual_params.common_name = CreateRandomString(8);
>-  }
>-  X509* x509 = MakeCertificate(key_pair->pkey(), actual_params);
>-  if (!x509) {
>-    LogSSLErrors("Generating certificate");
>-    return nullptr;
>-  }
>-#if !defined(NDEBUG)
>-  PrintCert(x509);
>-#endif
>-  OpenSSLCertificate* ret = new OpenSSLCertificate(x509);
>-  X509_free(x509);
>-  return ret;
>-}
>-
>-OpenSSLCertificate* OpenSSLCertificate::FromPEMString(
>-    const std::string& pem_string) {
>-  BIO* bio = BIO_new_mem_buf(const_cast<char*>(pem_string.c_str()), -1);
>-  if (!bio)
>-    return nullptr;
>-  BIO_set_mem_eof_return(bio, 0);
>-  X509* x509 =
>-      PEM_read_bio_X509(bio, nullptr, nullptr, const_cast<char*>("\0"));
>-  BIO_free(bio);  // Frees the BIO, but not the pointed-to string.
>-
>-  if (!x509)
>-    return nullptr;
>-
>-  OpenSSLCertificate* ret = new OpenSSLCertificate(x509);
>-  X509_free(x509);
>-  return ret;
>-}
>-
>-// NOTE: This implementation only functions correctly after InitializeSSL
>-// and before CleanupSSL.
>-bool OpenSSLCertificate::GetSignatureDigestAlgorithm(
>-    std::string* algorithm) const {
>-  int nid = OBJ_obj2nid(x509_->sig_alg->algorithm);
>-  switch (nid) {
>-    case NID_md5WithRSA:
>-    case NID_md5WithRSAEncryption:
>-      *algorithm = DIGEST_MD5;
>-      break;
>-    case NID_ecdsa_with_SHA1:
>-    case NID_dsaWithSHA1:
>-    case NID_dsaWithSHA1_2:
>-    case NID_sha1WithRSA:
>-    case NID_sha1WithRSAEncryption:
>-      *algorithm = DIGEST_SHA_1;
>-      break;
>-    case NID_ecdsa_with_SHA224:
>-    case NID_sha224WithRSAEncryption:
>-    case NID_dsa_with_SHA224:
>-      *algorithm = DIGEST_SHA_224;
>-      break;
>-    case NID_ecdsa_with_SHA256:
>-    case NID_sha256WithRSAEncryption:
>-    case NID_dsa_with_SHA256:
>-      *algorithm = DIGEST_SHA_256;
>-      break;
>-    case NID_ecdsa_with_SHA384:
>-    case NID_sha384WithRSAEncryption:
>-      *algorithm = DIGEST_SHA_384;
>-      break;
>-    case NID_ecdsa_with_SHA512:
>-    case NID_sha512WithRSAEncryption:
>-      *algorithm = DIGEST_SHA_512;
>-      break;
>-    default:
>-      // Unknown algorithm.  There are several unhandled options that are less
>-      // common and more complex.
>-      RTC_LOG(LS_ERROR) << "Unknown signature algorithm NID: " << nid;
>-      algorithm->clear();
>-      return false;
>-  }
>-  return true;
>-}
>-
>-std::unique_ptr<SSLCertChain> OpenSSLCertificate::GetChain() const {
>-  return nullptr;
>-}
>-
>-bool OpenSSLCertificate::ComputeDigest(const std::string& algorithm,
>-                                       unsigned char* digest,
>-                                       size_t size,
>-                                       size_t* length) const {
>-  return ComputeDigest(x509_, algorithm, digest, size, length);
>-}
>-
>-bool OpenSSLCertificate::ComputeDigest(const X509* x509,
>-                                       const std::string& algorithm,
>-                                       unsigned char* digest,
>-                                       size_t size,
>-                                       size_t* length) {
>-  const EVP_MD* md;
>-  unsigned int n;
>-
>-  if (!OpenSSLDigest::GetDigestEVP(algorithm, &md))
>-    return false;
>-
>-  if (size < static_cast<size_t>(EVP_MD_size(md)))
>-    return false;
>-
>-  X509_digest(x509, md, digest, &n);
>-
>-  *length = n;
>-
>-  return true;
>-}
>-
>-OpenSSLCertificate::~OpenSSLCertificate() {
>-  X509_free(x509_);
>-}
>-
>-OpenSSLCertificate* OpenSSLCertificate::GetReference() const {
>-  return new OpenSSLCertificate(x509_);
>-}
>-
>-std::string OpenSSLCertificate::ToPEMString() const {
>-  BIO* bio = BIO_new(BIO_s_mem());
>-  if (!bio) {
>-    RTC_FATAL() << "unreachable code";
>-  }
>-  if (!PEM_write_bio_X509(bio, x509_)) {
>-    BIO_free(bio);
>-    RTC_FATAL() << "unreachable code";
>-  }
>-  BIO_write(bio, "\0", 1);
>-  char* buffer;
>-  BIO_get_mem_data(bio, &buffer);
>-  std::string ret(buffer);
>-  BIO_free(bio);
>-  return ret;
>-}
>-
>-void OpenSSLCertificate::ToDER(Buffer* der_buffer) const {
>-  // In case of failure, make sure to leave the buffer empty.
>-  der_buffer->SetSize(0);
>-
>-  // Calculates the DER representation of the certificate, from scratch.
>-  BIO* bio = BIO_new(BIO_s_mem());
>-  if (!bio) {
>-    RTC_FATAL() << "unreachable code";
>-  }
>-  if (!i2d_X509_bio(bio, x509_)) {
>-    BIO_free(bio);
>-    RTC_FATAL() << "unreachable code";
>-  }
>-  char* data;
>-  size_t length = BIO_get_mem_data(bio, &data);
>-  der_buffer->SetData(data, length);
>-  BIO_free(bio);
>-}
>-
>-void OpenSSLCertificate::AddReference() const {
>-  RTC_DCHECK(x509_ != nullptr);
>-#if defined(OPENSSL_IS_BORINGSSL)
>-  X509_up_ref(x509_);
>-#else
>-  CRYPTO_add(&x509_->references, 1, CRYPTO_LOCK_X509);
>-#endif
>-}
>-
>-bool OpenSSLCertificate::operator==(const OpenSSLCertificate& other) const {
>-  return X509_cmp(x509_, other.x509_) == 0;
>-}
>-
>-bool OpenSSLCertificate::operator!=(const OpenSSLCertificate& other) const {
>-  return !(*this == other);
>-}
>-
>-// Documented in sslidentity.h.
>-int64_t OpenSSLCertificate::CertificateExpirationTime() const {
>-  ASN1_TIME* expire_time = X509_get_notAfter(x509_);
>-  bool long_format;
>-
>-  if (expire_time->type == V_ASN1_UTCTIME) {
>-    long_format = false;
>-  } else if (expire_time->type == V_ASN1_GENERALIZEDTIME) {
>-    long_format = true;
>-  } else {
>-    return -1;
>-  }
>-
>-  return ASN1TimeToSec(expire_time->data, expire_time->length, long_format);
>-}
>-
> OpenSSLIdentity::OpenSSLIdentity(
>     std::unique_ptr<OpenSSLKeyPair> key_pair,
>     std::unique_ptr<OpenSSLCertificate> certificate)
>@@ -553,8 +267,8 @@ SSLIdentity* OpenSSLIdentity::FromPEMStrings(const std::string& private_key,
> SSLIdentity* OpenSSLIdentity::FromPEMChainStrings(
>     const std::string& private_key,
>     const std::string& certificate_chain) {
>-  BIO* bio =
>-      BIO_new_mem_buf(certificate_chain.data(), certificate_chain.size());
>+  BIO* bio = BIO_new_mem_buf(certificate_chain.data(),
>+                             rtc::dchecked_cast<int>(certificate_chain.size()));
>   if (!bio)
>     return nullptr;
>   BIO_set_mem_eof_return(bio, 0);
>@@ -589,16 +303,20 @@ SSLIdentity* OpenSSLIdentity::FromPEMChainStrings(
>   }
> 
>   return new OpenSSLIdentity(std::move(key_pair),
>-                             MakeUnique<SSLCertChain>(std::move(certs)));
>+                             absl::make_unique<SSLCertChain>(std::move(certs)));
> }
> 
> const OpenSSLCertificate& OpenSSLIdentity::certificate() const {
>   return *static_cast<const OpenSSLCertificate*>(&cert_chain_->Get(0));
> }
> 
>+const SSLCertChain& OpenSSLIdentity::cert_chain() const {
>+  return *cert_chain_.get();
>+}
>+
> OpenSSLIdentity* OpenSSLIdentity::GetReference() const {
>-  return new OpenSSLIdentity(WrapUnique(key_pair_->GetReference()),
>-                             WrapUnique(cert_chain_->Copy()));
>+  return new OpenSSLIdentity(absl::WrapUnique(key_pair_->GetReference()),
>+                             absl::WrapUnique(cert_chain_->Copy()));
> }
> 
> bool OpenSSLIdentity::ConfigureIdentity(SSL_CTX* ctx) {
>@@ -606,14 +324,14 @@ bool OpenSSLIdentity::ConfigureIdentity(SSL_CTX* ctx) {
>   const OpenSSLCertificate* cert = &certificate();
>   if (SSL_CTX_use_certificate(ctx, cert->x509()) != 1 ||
>       SSL_CTX_use_PrivateKey(ctx, key_pair_->pkey()) != 1) {
>-    LogSSLErrors("Configuring key and certificate");
>+    openssl::LogSSLErrors("Configuring key and certificate");
>     return false;
>   }
>   // If a chain is available, use it.
>   for (size_t i = 1; i < cert_chain_->GetSize(); ++i) {
>     cert = static_cast<const OpenSSLCertificate*>(&cert_chain_->Get(i));
>     if (SSL_CTX_add1_chain_cert(ctx, cert->x509()) != 1) {
>-      LogSSLErrors("Configuring intermediate certificate");
>+      openssl::LogSSLErrors("Configuring intermediate certificate");
>       return false;
>     }
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslidentity.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslidentity.h
>index a700a1d107c..34044276ca7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslidentity.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslidentity.h
>@@ -19,6 +19,7 @@
> 
> #include "rtc_base/checks.h"
> #include "rtc_base/constructormagic.h"
>+#include "rtc_base/opensslcertificate.h"
> #include "rtc_base/sslidentity.h"
> 
> typedef struct ssl_ctx_st SSL_CTX;
>@@ -56,53 +57,6 @@ class OpenSSLKeyPair {
>   RTC_DISALLOW_COPY_AND_ASSIGN(OpenSSLKeyPair);
> };
> 
>-// OpenSSLCertificate encapsulates an OpenSSL X509* certificate object,
>-// which is also reference counted inside the OpenSSL library.
>-class OpenSSLCertificate : public SSLCertificate {
>- public:
>-  // Caller retains ownership of the X509 object.
>-  explicit OpenSSLCertificate(X509* x509);
>-
>-  static OpenSSLCertificate* Generate(OpenSSLKeyPair* key_pair,
>-                                      const SSLIdentityParams& params);
>-  static OpenSSLCertificate* FromPEMString(const std::string& pem_string);
>-
>-  ~OpenSSLCertificate() override;
>-
>-  OpenSSLCertificate* GetReference() const override;
>-
>-  X509* x509() const { return x509_; }
>-
>-  std::string ToPEMString() const override;
>-  void ToDER(Buffer* der_buffer) const override;
>-  bool operator==(const OpenSSLCertificate& other) const;
>-  bool operator!=(const OpenSSLCertificate& other) const;
>-
>-  // Compute the digest of the certificate given algorithm
>-  bool ComputeDigest(const std::string& algorithm,
>-                     unsigned char* digest,
>-                     size_t size,
>-                     size_t* length) const override;
>-
>-  // Compute the digest of a certificate as an X509 *
>-  static bool ComputeDigest(const X509* x509,
>-                            const std::string& algorithm,
>-                            unsigned char* digest,
>-                            size_t size,
>-                            size_t* length);
>-
>-  bool GetSignatureDigestAlgorithm(std::string* algorithm) const override;
>-  std::unique_ptr<SSLCertChain> GetChain() const override;
>-
>-  int64_t CertificateExpirationTime() const override;
>-
>- private:
>-  void AddReference() const;
>-
>-  X509* x509_;
>-  RTC_DISALLOW_COPY_AND_ASSIGN(OpenSSLCertificate);
>-};
>-
> // Holds a keypair and certificate together, and a method to generate
> // them consistently.
> class OpenSSLIdentity : public SSLIdentity {
>@@ -118,6 +72,7 @@ class OpenSSLIdentity : public SSLIdentity {
>   ~OpenSSLIdentity() override;
> 
>   const OpenSSLCertificate& certificate() const override;
>+  const SSLCertChain& cert_chain() const override;
>   OpenSSLIdentity* GetReference() const override;
> 
>   // Configure an SSL context object to use our key and certificate.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslsessioncache.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslsessioncache.cc
>new file mode 100644
>index 00000000000..2e37d55deb1
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslsessioncache.cc
>@@ -0,0 +1,52 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "rtc_base/opensslsessioncache.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/openssl.h"
>+
>+namespace rtc {
>+
>+OpenSSLSessionCache::OpenSSLSessionCache(SSLMode ssl_mode, SSL_CTX* ssl_ctx)
>+    : ssl_mode_(ssl_mode), ssl_ctx_(ssl_ctx) {
>+  // It is invalid to pass in a null context.
>+  RTC_DCHECK(ssl_ctx != nullptr);
>+  SSL_CTX_up_ref(ssl_ctx);
>+}
>+
>+OpenSSLSessionCache::~OpenSSLSessionCache() {
>+  for (auto it : sessions_) {
>+    SSL_SESSION_free(it.second);
>+  }
>+  SSL_CTX_free(ssl_ctx_);
>+}
>+
>+SSL_SESSION* OpenSSLSessionCache::LookupSession(
>+    const std::string& hostname) const {
>+  auto it = sessions_.find(hostname);
>+  return (it != sessions_.end()) ? it->second : nullptr;
>+}
>+
>+void OpenSSLSessionCache::AddSession(const std::string& hostname,
>+                                     SSL_SESSION* new_session) {
>+  SSL_SESSION* old_session = LookupSession(hostname);
>+  SSL_SESSION_free(old_session);
>+  sessions_[hostname] = new_session;
>+}
>+
>+SSL_CTX* OpenSSLSessionCache::GetSSLContext() const {
>+  return ssl_ctx_;
>+}
>+
>+SSLMode OpenSSLSessionCache::GetSSLMode() const {
>+  return ssl_mode_;
>+}
>+
>+}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslsessioncache.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslsessioncache.h
>new file mode 100644
>index 00000000000..c74b7bf8204
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslsessioncache.h
>@@ -0,0 +1,67 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef RTC_BASE_OPENSSLSESSIONCACHE_H_
>+#define RTC_BASE_OPENSSLSESSIONCACHE_H_
>+
>+#include <openssl/ossl_typ.h>
>+#include <map>
>+#include <string>
>+
>+#include "rtc_base/constructormagic.h"
>+#include "rtc_base/sslstreamadapter.h"
>+
>+#ifndef OPENSSL_IS_BORINGSSL
>+typedef struct ssl_session_st SSL_SESSION;
>+#endif
>+
>+namespace rtc {
>+
>+// The OpenSSLSessionCache maps hostnames to SSL_SESSIONS. This cache is
>+// owned by the OpenSSLAdapterFactory and is passed down to each OpenSSLAdapter
>+// created with the factory.
>+class OpenSSLSessionCache final {
>+ public:
>+  // Creates a new OpenSSLSessionCache using the provided the SSL_CTX and
>+  // the ssl_mode. The SSL_CTX will be up_refed. ssl_ctx cannot be nullptr,
>+  // the constructor immediately dchecks this.
>+  OpenSSLSessionCache(SSLMode ssl_mode, SSL_CTX* ssl_ctx);
>+  // Frees the cached SSL_SESSIONS and then frees the SSL_CTX.
>+  ~OpenSSLSessionCache();
>+  // Looks up a session by hostname. The returned SSL_SESSION is not up_refed.
>+  SSL_SESSION* LookupSession(const std::string& hostname) const;
>+  // Adds a session to the cache, and up_refs it. Any existing session with the
>+  // same hostname is replaced.
>+  void AddSession(const std::string& hostname, SSL_SESSION* session);
>+  // Returns the true underlying SSL Context that holds these cached sessions.
>+  SSL_CTX* GetSSLContext() const;
>+  // The SSL Mode tht the OpenSSLSessionCache was constructed with. This cannot
>+  // be changed after launch.
>+  SSLMode GetSSLMode() const;
>+
>+ private:
>+  // Holds the SSL Mode that the OpenSSLCache was initialized with. This is
>+  // immutable after creation and cannot change.
>+  const SSLMode ssl_mode_;
>+  /// SSL Context for all shared cached sessions. This SSL_CTX is initialized
>+  //  with SSL_CTX_set_session_cache_mode(ctx, SSL_SESS_CACHE_CLIENT); Meaning
>+  //  all client sessions will be added to the cache internal to the context.
>+  SSL_CTX* ssl_ctx_ = nullptr;
>+  // Map of hostnames to SSL_SESSIONs; holds references to the SSL_SESSIONs,
>+  // which are cleaned up when the factory is destroyed.
>+  // TODO(juberti): Add LRU eviction to keep the cache from growing forever.
>+  std::map<std::string, SSL_SESSION*> sessions_;
>+  // The cache should never be copied or assigned directly.
>+  RTC_DISALLOW_COPY_AND_ASSIGN(OpenSSLSessionCache);
>+};
>+
>+}  // namespace rtc
>+
>+#endif  // RTC_BASE_OPENSSLSESSIONCACHE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslsessioncache_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslsessioncache_unittest.cc
>new file mode 100644
>index 00000000000..6489b2bc2f5
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslsessioncache_unittest.cc
>@@ -0,0 +1,85 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <openssl/ssl.h>
>+#include <stdlib.h>
>+
>+#include <map>
>+#include <memory>
>+
>+#include "rtc_base/gunit.h"
>+#include "rtc_base/openssl.h"
>+#include "rtc_base/opensslsessioncache.h"
>+
>+namespace rtc {
>+
>+TEST(OpenSSLSessionCache, DTLSModeSetCorrectly) {
>+  SSL_CTX* ssl_ctx = SSL_CTX_new(DTLSv1_2_client_method());
>+
>+  OpenSSLSessionCache session_cache(SSL_MODE_DTLS, ssl_ctx);
>+  EXPECT_EQ(session_cache.GetSSLMode(), SSL_MODE_DTLS);
>+
>+  SSL_CTX_free(ssl_ctx);
>+}
>+
>+TEST(OpenSSLSessionCache, TLSModeSetCorrectly) {
>+  SSL_CTX* ssl_ctx = SSL_CTX_new(TLSv1_2_client_method());
>+
>+  OpenSSLSessionCache session_cache(SSL_MODE_TLS, ssl_ctx);
>+  EXPECT_EQ(session_cache.GetSSLMode(), SSL_MODE_TLS);
>+
>+  SSL_CTX_free(ssl_ctx);
>+}
>+
>+TEST(OpenSSLSessionCache, SSLContextSetCorrectly) {
>+  SSL_CTX* ssl_ctx = SSL_CTX_new(DTLSv1_2_client_method());
>+
>+  OpenSSLSessionCache session_cache(SSL_MODE_DTLS, ssl_ctx);
>+  EXPECT_EQ(session_cache.GetSSLContext(), ssl_ctx);
>+
>+  SSL_CTX_free(ssl_ctx);
>+}
>+
>+TEST(OpenSSLSessionCache, InvalidLookupReturnsNullptr) {
>+  SSL_CTX* ssl_ctx = SSL_CTX_new(DTLSv1_2_client_method());
>+
>+  OpenSSLSessionCache session_cache(SSL_MODE_DTLS, ssl_ctx);
>+  EXPECT_EQ(session_cache.LookupSession("Invalid"), nullptr);
>+  EXPECT_EQ(session_cache.LookupSession(""), nullptr);
>+  EXPECT_EQ(session_cache.LookupSession("."), nullptr);
>+
>+  SSL_CTX_free(ssl_ctx);
>+}
>+
>+TEST(OpenSSLSessionCache, SimpleValidSessionLookup) {
>+  SSL_CTX* ssl_ctx = SSL_CTX_new(DTLSv1_2_client_method());
>+  SSL_SESSION* ssl_session = SSL_SESSION_new(ssl_ctx);
>+
>+  OpenSSLSessionCache session_cache(SSL_MODE_DTLS, ssl_ctx);
>+  session_cache.AddSession("webrtc.org", ssl_session);
>+  EXPECT_EQ(session_cache.LookupSession("webrtc.org"), ssl_session);
>+
>+  SSL_CTX_free(ssl_ctx);
>+}
>+
>+TEST(OpenSSLSessionCache, AddToExistingReplacesPrevious) {
>+  SSL_CTX* ssl_ctx = SSL_CTX_new(DTLSv1_2_client_method());
>+  SSL_SESSION* ssl_session_1 = SSL_SESSION_new(ssl_ctx);
>+  SSL_SESSION* ssl_session_2 = SSL_SESSION_new(ssl_ctx);
>+
>+  OpenSSLSessionCache session_cache(SSL_MODE_DTLS, ssl_ctx);
>+  session_cache.AddSession("webrtc.org", ssl_session_1);
>+  session_cache.AddSession("webrtc.org", ssl_session_2);
>+  EXPECT_EQ(session_cache.LookupSession("webrtc.org"), ssl_session_2);
>+
>+  SSL_CTX_free(ssl_ctx);
>+}
>+
>+}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslstreamadapter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslstreamadapter.cc
>index 52a0b776231..fd54a082a7d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslstreamadapter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslstreamadapter.cc
>@@ -37,13 +37,13 @@
> #include "rtc_base/timeutils.h"
> 
> namespace {
>-  bool g_use_time_callback_for_testing = false;
>+bool g_use_time_callback_for_testing = false;
> }
> 
> namespace rtc {
> 
>-#if (OPENSSL_VERSION_NUMBER < 0x10001000L)
>-#error "webrtc requires at least OpenSSL version 1.0.1, to support DTLS-SRTP"
>+#if (OPENSSL_VERSION_NUMBER < 0x10100000L)
>+#error "webrtc requires at least OpenSSL version 1.1.0, to support DTLS-SRTP"
> #endif
> 
> // SRTP cipher suite table. |internal_name| is used to construct a
>@@ -77,8 +77,10 @@ struct SslCipherMapEntry {
>   const char* rfc_name;
> };
> 
>-#define DEFINE_CIPHER_ENTRY_SSL3(name)  {SSL3_CK_##name, "TLS_"#name}
>-#define DEFINE_CIPHER_ENTRY_TLS1(name)  {TLS1_CK_##name, "TLS_"#name}
>+#define DEFINE_CIPHER_ENTRY_SSL3(name) \
>+  { SSL3_CK_##name, "TLS_" #name }
>+#define DEFINE_CIPHER_ENTRY_TLS1(name) \
>+  { TLS1_CK_##name, "TLS_" #name }
> 
> // The "SSL_CIPHER_standard_name" function is only available in OpenSSL when
> // compiled with tracing, so we need to define the mapping manually here.
>@@ -162,27 +164,34 @@ static long stream_ctrl(BIO* h, int cmd, long arg1, void* arg2);
> static int stream_new(BIO* h);
> static int stream_free(BIO* data);
> 
>-static const BIO_METHOD methods_stream = {
>-    BIO_TYPE_BIO, "stream",   stream_write, stream_read, stream_puts, 0,
>-    stream_ctrl,  stream_new, stream_free,  nullptr,
>-};
>+static BIO_METHOD* BIO_stream_method() {
>+  static BIO_METHOD* method = [] {
>+    BIO_METHOD* method = BIO_meth_new(BIO_TYPE_BIO, "stream");
>+    BIO_meth_set_write(method, stream_write);
>+    BIO_meth_set_read(method, stream_read);
>+    BIO_meth_set_puts(method, stream_puts);
>+    BIO_meth_set_ctrl(method, stream_ctrl);
>+    BIO_meth_set_create(method, stream_new);
>+    BIO_meth_set_destroy(method, stream_free);
>+    return method;
>+  }();
>+  return method;
>+}
> 
> static BIO* BIO_new_stream(StreamInterface* stream) {
>-  // TODO(davidben): Remove the const_cast when BoringSSL is assumed.
>-  BIO* ret = BIO_new(const_cast<BIO_METHOD*>(&methods_stream));
>+  BIO* ret = BIO_new(BIO_stream_method());
>   if (ret == nullptr)
>     return nullptr;
>-  ret->ptr = stream;
>+  BIO_set_data(ret, stream);
>   return ret;
> }
> 
> // bio methods return 1 (or at least non-zero) on success and 0 on failure.
> 
> static int stream_new(BIO* b) {
>-  b->shutdown = 0;
>-  b->init = 1;
>-  b->num = 0;  // 1 means end-of-stream
>-  b->ptr = 0;
>+  BIO_set_shutdown(b, 0);
>+  BIO_set_init(b, 1);
>+  BIO_set_data(b, 0);
>   return 1;
> }
> 
>@@ -195,15 +204,13 @@ static int stream_free(BIO* b) {
> static int stream_read(BIO* b, char* out, int outl) {
>   if (!out)
>     return -1;
>-  StreamInterface* stream = static_cast<StreamInterface*>(b->ptr);
>+  StreamInterface* stream = static_cast<StreamInterface*>(BIO_get_data(b));
>   BIO_clear_retry_flags(b);
>   size_t read;
>   int error;
>   StreamResult result = stream->Read(out, outl, &read, &error);
>   if (result == SR_SUCCESS) {
>     return checked_cast<int>(read);
>-  } else if (result == SR_EOS) {
>-    b->num = 1;
>   } else if (result == SR_BLOCK) {
>     BIO_set_retry_read(b);
>   }
>@@ -213,7 +220,7 @@ static int stream_read(BIO* b, char* out, int outl) {
> static int stream_write(BIO* b, const char* in, int inl) {
>   if (!in)
>     return -1;
>-  StreamInterface* stream = static_cast<StreamInterface*>(b->ptr);
>+  StreamInterface* stream = static_cast<StreamInterface*>(BIO_get_data(b));
>   BIO_clear_retry_flags(b);
>   size_t written;
>   int error;
>@@ -234,8 +241,11 @@ static long stream_ctrl(BIO* b, int cmd, long num, void* ptr) {
>   switch (cmd) {
>     case BIO_CTRL_RESET:
>       return 0;
>-    case BIO_CTRL_EOF:
>-      return b->num;
>+    case BIO_CTRL_EOF: {
>+      StreamInterface* stream = static_cast<StreamInterface*>(ptr);
>+      // 1 means end-of-stream.
>+      return (stream->GetState() == SS_CLOSED) ? 1 : 0;
>+    }
>     case BIO_CTRL_WPENDING:
>     case BIO_CTRL_PENDING:
>       return 0;
>@@ -280,13 +290,6 @@ void OpenSSLStreamAdapter::SetServerRole(SSLRole role) {
>   role_ = role;
> }
> 
>-std::unique_ptr<SSLCertificate> OpenSSLStreamAdapter::GetPeerCertificate()
>-    const {
>-  return peer_certificate_ ? std::unique_ptr<SSLCertificate>(
>-                                 peer_certificate_->GetReference())
>-                           : nullptr;
>-}
>-
> bool OpenSSLStreamAdapter::SetPeerCertificateDigest(
>     const std::string& digest_alg,
>     const unsigned char* digest_val,
>@@ -316,7 +319,7 @@ bool OpenSSLStreamAdapter::SetPeerCertificateDigest(
>   peer_certificate_digest_value_.SetData(digest_val, digest_len);
>   peer_certificate_digest_algorithm_ = digest_alg;
> 
>-  if (!peer_certificate_) {
>+  if (!peer_cert_chain_) {
>     // Normal case, where the digest is set before we obtain the certificate
>     // from the handshake.
>     return true;
>@@ -451,7 +454,7 @@ bool OpenSSLStreamAdapter::GetDtlsSrtpCryptoSuite(int* crypto_suite) {
>   if (state_ != SSL_CONNECTED)
>     return false;
> 
>-  const SRTP_PROTECTION_PROFILE *srtp_profile =
>+  const SRTP_PROTECTION_PROFILE* srtp_profile =
>       SSL_get_selected_srtp_profile(ssl_);
> 
>   if (!srtp_profile)
>@@ -496,8 +499,7 @@ void OpenSSLStreamAdapter::SetMaxProtocolVersion(SSLProtocolVersion version) {
>   ssl_max_version_ = version;
> }
> 
>-void OpenSSLStreamAdapter::SetInitialRetransmissionTimeout(
>-    int timeout_ms) {
>+void OpenSSLStreamAdapter::SetInitialRetransmissionTimeout(int timeout_ms) {
>   RTC_DCHECK(ssl_ctx_ == nullptr);
>   dtls_handshake_timeout_ms_ = timeout_ms;
> }
>@@ -506,31 +508,33 @@ void OpenSSLStreamAdapter::SetInitialRetransmissionTimeout(
> // StreamInterface Implementation
> //
> 
>-StreamResult OpenSSLStreamAdapter::Write(const void* data, size_t data_len,
>-                                         size_t* written, int* error) {
>+StreamResult OpenSSLStreamAdapter::Write(const void* data,
>+                                         size_t data_len,
>+                                         size_t* written,
>+                                         int* error) {
>   RTC_LOG(LS_VERBOSE) << "OpenSSLStreamAdapter::Write(" << data_len << ")";
> 
>   switch (state_) {
>-  case SSL_NONE:
>-    // pass-through in clear text
>-    return StreamAdapterInterface::Write(data, data_len, written, error);
>-
>-  case SSL_WAIT:
>-  case SSL_CONNECTING:
>-    return SR_BLOCK;
>+    case SSL_NONE:
>+      // pass-through in clear text
>+      return StreamAdapterInterface::Write(data, data_len, written, error);
> 
>-  case SSL_CONNECTED:
>-    if (waiting_to_verify_peer_certificate()) {
>+    case SSL_WAIT:
>+    case SSL_CONNECTING:
>       return SR_BLOCK;
>-    }
>-    break;
>-
>-  case SSL_ERROR:
>-  case SSL_CLOSED:
>-  default:
>-    if (error)
>-      *error = ssl_error_code_;
>-    return SR_ERROR;
>+
>+    case SSL_CONNECTED:
>+      if (waiting_to_verify_peer_certificate()) {
>+        return SR_BLOCK;
>+      }
>+      break;
>+
>+    case SSL_ERROR:
>+    case SSL_CLOSED:
>+    default:
>+      if (error)
>+        *error = ssl_error_code_;
>+      return SR_ERROR;
>   }
> 
>   // OpenSSL will return an error if we try to write zero bytes
>@@ -545,33 +549,35 @@ StreamResult OpenSSLStreamAdapter::Write(const void* data, size_t data_len,
>   int code = SSL_write(ssl_, data, checked_cast<int>(data_len));
>   int ssl_error = SSL_get_error(ssl_, code);
>   switch (ssl_error) {
>-  case SSL_ERROR_NONE:
>-    RTC_LOG(LS_VERBOSE) << " -- success";
>-    RTC_DCHECK_GT(code, 0);
>-    RTC_DCHECK_LE(code, data_len);
>-    if (written)
>-      *written = code;
>-    return SR_SUCCESS;
>-  case SSL_ERROR_WANT_READ:
>-    RTC_LOG(LS_VERBOSE) << " -- error want read";
>-    ssl_write_needs_read_ = true;
>-    return SR_BLOCK;
>-  case SSL_ERROR_WANT_WRITE:
>-    RTC_LOG(LS_VERBOSE) << " -- error want write";
>-    return SR_BLOCK;
>-
>-  case SSL_ERROR_ZERO_RETURN:
>-  default:
>-    Error("SSL_write", (ssl_error ? ssl_error : -1), 0, false);
>-    if (error)
>-      *error = ssl_error_code_;
>-    return SR_ERROR;
>+    case SSL_ERROR_NONE:
>+      RTC_LOG(LS_VERBOSE) << " -- success";
>+      RTC_DCHECK_GT(code, 0);
>+      RTC_DCHECK_LE(code, data_len);
>+      if (written)
>+        *written = code;
>+      return SR_SUCCESS;
>+    case SSL_ERROR_WANT_READ:
>+      RTC_LOG(LS_VERBOSE) << " -- error want read";
>+      ssl_write_needs_read_ = true;
>+      return SR_BLOCK;
>+    case SSL_ERROR_WANT_WRITE:
>+      RTC_LOG(LS_VERBOSE) << " -- error want write";
>+      return SR_BLOCK;
>+
>+    case SSL_ERROR_ZERO_RETURN:
>+    default:
>+      Error("SSL_write", (ssl_error ? ssl_error : -1), 0, false);
>+      if (error)
>+        *error = ssl_error_code_;
>+      return SR_ERROR;
>   }
>   // not reached
> }
> 
>-StreamResult OpenSSLStreamAdapter::Read(void* data, size_t data_len,
>-                                        size_t* read, int* error) {
>+StreamResult OpenSSLStreamAdapter::Read(void* data,
>+                                        size_t data_len,
>+                                        size_t* read,
>+                                        int* error) {
>   RTC_LOG(LS_VERBOSE) << "OpenSSLStreamAdapter::Read(" << data_len << ")";
>   switch (state_) {
>     case SSL_NONE:
>@@ -643,7 +649,6 @@ StreamResult OpenSSLStreamAdapter::Read(void* data, size_t data_len,
>       return SR_EOS;
>       break;
>     default:
>-      RTC_LOG(LS_VERBOSE) << " -- error " << code;
>       Error("SSL_read", (ssl_error ? ssl_error : -1), 0, false);
>       if (error)
>         *error = ssl_error_code_;
>@@ -664,7 +669,7 @@ void OpenSSLStreamAdapter::FlushInput(unsigned int left) {
>     RTC_DCHECK(ssl_error == SSL_ERROR_NONE);
> 
>     if (ssl_error != SSL_ERROR_NONE) {
>-      RTC_LOG(LS_VERBOSE) << " -- error " << code;
>+      RTC_DLOG(LS_VERBOSE) << " -- error " << code;
>       Error("SSL_read", (ssl_error ? ssl_error : -1), 0, false);
>       return;
>     }
>@@ -699,7 +704,8 @@ StreamState OpenSSLStreamAdapter::GetState() const {
>   // not reached
> }
> 
>-void OpenSSLStreamAdapter::OnEvent(StreamInterface* stream, int events,
>+void OpenSSLStreamAdapter::OnEvent(StreamInterface* stream,
>+                                   int events,
>                                    int err) {
>   int events_to_signal = 0;
>   int signal_error = 0;
>@@ -717,12 +723,12 @@ void OpenSSLStreamAdapter::OnEvent(StreamInterface* stream, int events,
>       }
>     }
>   }
>-  if ((events & (SE_READ|SE_WRITE))) {
>+  if ((events & (SE_READ | SE_WRITE))) {
>     RTC_LOG(LS_VERBOSE) << "OpenSSLStreamAdapter::OnEvent"
>                         << ((events & SE_READ) ? " SE_READ" : "")
>                         << ((events & SE_WRITE) ? " SE_WRITE" : "");
>     if (state_ == SSL_NONE) {
>-      events_to_signal |= events & (SE_READ|SE_WRITE);
>+      events_to_signal |= events & (SE_READ | SE_WRITE);
>     } else if (state_ == SSL_CONNECTING) {
>       if (int err = ContinueSSL()) {
>         Error("ContinueSSL", err, 0, true);
>@@ -791,7 +797,7 @@ int OpenSSLStreamAdapter::BeginSSL() {
>   }
> 
>   SSL_set_mode(ssl_, SSL_MODE_ENABLE_PARTIAL_WRITE |
>-               SSL_MODE_ACCEPT_MOVING_WRITE_BUFFER);
>+                         SSL_MODE_ACCEPT_MOVING_WRITE_BUFFER);
> 
> #if !defined(OPENSSL_IS_BORINGSSL)
>   // Specify an ECDH group for ECDHE ciphers, otherwise OpenSSL cannot
>@@ -824,7 +830,7 @@ int OpenSSLStreamAdapter::ContinueSSL() {
>       RTC_LOG(LS_VERBOSE) << " -- success";
>       // By this point, OpenSSL should have given us a certificate, or errored
>       // out if one was missing.
>-      RTC_DCHECK(peer_certificate_ || !client_auth_enabled());
>+      RTC_DCHECK(peer_cert_chain_ || !client_auth_enabled());
> 
>       state_ = SSL_CONNECTED;
>       if (!waiting_to_verify_peer_certificate()) {
>@@ -849,9 +855,8 @@ int OpenSSLStreamAdapter::ContinueSSL() {
> 
>         Thread::Current()->PostDelayed(RTC_FROM_HERE, delay, this, MSG_TIMEOUT,
>                                        0);
>-        }
>       }
>-      break;
>+    } break;
> 
>     case SSL_ERROR_WANT_WRITE:
>       RTC_LOG(LS_VERBOSE) << " -- error want write";
>@@ -921,13 +926,12 @@ void OpenSSLStreamAdapter::Cleanup(uint8_t alert) {
>     ssl_ctx_ = nullptr;
>   }
>   identity_.reset();
>-  peer_certificate_.reset();
>+  peer_cert_chain_.reset();
> 
>   // Clear the DTLS timer
>   Thread::Current()->Clear(this, MSG_TIMEOUT);
> }
> 
>-
> void OpenSSLStreamAdapter::OnMessage(Message* msg) {
>   // Process our own messages and then pass others to the superclass
>   if (MSG_TIMEOUT == msg->message_id) {
>@@ -943,9 +947,8 @@ SSL_CTX* OpenSSLStreamAdapter::SetupSSLContext() {
>   SSL_CTX* ctx = nullptr;
> 
> #ifdef OPENSSL_IS_BORINGSSL
>-    ctx = SSL_CTX_new(ssl_mode_ == SSL_MODE_DTLS ?
>-        DTLS_method() : TLS_method());
>-    // Version limiting for BoringSSL will be done below.
>+  ctx = SSL_CTX_new(ssl_mode_ == SSL_MODE_DTLS ? DTLS_method() : TLS_method());
>+// Version limiting for BoringSSL will be done below.
> #else
>   const SSL_METHOD* method;
>   switch (ssl_max_version_) {
>@@ -970,35 +973,17 @@ SSL_CTX* OpenSSLStreamAdapter::SetupSSLContext() {
>     case SSL_PROTOCOL_TLS_12:
>     default:
>       if (ssl_mode_ == SSL_MODE_DTLS) {
>-#if (OPENSSL_VERSION_NUMBER >= 0x10002000L)
>-        // DTLS 1.2 only available starting from OpenSSL 1.0.2
>         if (role_ == SSL_CLIENT) {
>           method = DTLS_client_method();
>         } else {
>           method = DTLS_server_method();
>         }
>-#else
>-        if (role_ == SSL_CLIENT) {
>-          method = DTLSv1_client_method();
>-        } else {
>-          method = DTLSv1_server_method();
>-        }
>-#endif
>       } else {
>-#if (OPENSSL_VERSION_NUMBER >= 0x10100000L)
>-        // New API only available starting from OpenSSL 1.1.0
>         if (role_ == SSL_CLIENT) {
>           method = TLS_client_method();
>         } else {
>           method = TLS_server_method();
>         }
>-#else
>-        if (role_ == SSL_CLIENT) {
>-          method = SSLv23_client_method();
>-        } else {
>-          method = SSLv23_server_method();
>-        }
>-#endif
>       }
>       break;
>   }
>@@ -1009,21 +994,21 @@ SSL_CTX* OpenSSLStreamAdapter::SetupSSLContext() {
>     return nullptr;
> 
> #ifdef OPENSSL_IS_BORINGSSL
>-  SSL_CTX_set_min_proto_version(ctx, ssl_mode_ == SSL_MODE_DTLS ?
>-      DTLS1_VERSION : TLS1_VERSION);
>+  SSL_CTX_set_min_proto_version(
>+      ctx, ssl_mode_ == SSL_MODE_DTLS ? DTLS1_VERSION : TLS1_VERSION);
>   switch (ssl_max_version_) {
>     case SSL_PROTOCOL_TLS_10:
>-      SSL_CTX_set_max_proto_version(ctx, ssl_mode_ == SSL_MODE_DTLS ?
>-          DTLS1_VERSION : TLS1_VERSION);
>+      SSL_CTX_set_max_proto_version(
>+          ctx, ssl_mode_ == SSL_MODE_DTLS ? DTLS1_VERSION : TLS1_VERSION);
>       break;
>     case SSL_PROTOCOL_TLS_11:
>-      SSL_CTX_set_max_proto_version(ctx, ssl_mode_ == SSL_MODE_DTLS ?
>-          DTLS1_VERSION : TLS1_1_VERSION);
>+      SSL_CTX_set_max_proto_version(
>+          ctx, ssl_mode_ == SSL_MODE_DTLS ? DTLS1_VERSION : TLS1_1_VERSION);
>       break;
>     case SSL_PROTOCOL_TLS_12:
>     default:
>-      SSL_CTX_set_max_proto_version(ctx, ssl_mode_ == SSL_MODE_DTLS ?
>-          DTLS1_2_VERSION : TLS1_2_VERSION);
>+      SSL_CTX_set_max_proto_version(
>+          ctx, ssl_mode_ == SSL_MODE_DTLS ? DTLS1_2_VERSION : TLS1_2_VERSION);
>       break;
>   }
>   if (g_use_time_callback_for_testing) {
>@@ -1073,15 +1058,18 @@ SSL_CTX* OpenSSLStreamAdapter::SetupSSLContext() {
> }
> 
> bool OpenSSLStreamAdapter::VerifyPeerCertificate() {
>-  if (!has_peer_certificate_digest() || !peer_certificate_) {
>+  if (!has_peer_certificate_digest() || !peer_cert_chain_ ||
>+      !peer_cert_chain_->GetSize()) {
>     RTC_LOG(LS_WARNING) << "Missing digest or peer certificate.";
>     return false;
>   }
>+  const OpenSSLCertificate* leaf_cert =
>+      static_cast<const OpenSSLCertificate*>(&peer_cert_chain_->Get(0));
> 
>   unsigned char digest[EVP_MAX_MD_SIZE];
>   size_t digest_length;
>   if (!OpenSSLCertificate::ComputeDigest(
>-          peer_certificate_->x509(), peer_certificate_digest_algorithm_, digest,
>+          leaf_cert->x509(), peer_certificate_digest_algorithm_, digest,
>           sizeof(digest), &digest_length)) {
>     RTC_LOG(LS_WARNING) << "Failed to compute peer cert digest.";
>     return false;
>@@ -1103,7 +1091,7 @@ bool OpenSSLStreamAdapter::VerifyPeerCertificate() {
> 
> std::unique_ptr<SSLCertChain> OpenSSLStreamAdapter::GetPeerSSLCertChain()
>     const {
>-  return std::unique_ptr<SSLCertChain>(peer_cert_chain_->Copy());
>+  return peer_cert_chain_ ? peer_cert_chain_->UniqueCopy() : nullptr;
> }
> 
> int OpenSSLStreamAdapter::SSLVerifyCallback(X509_STORE_CTX* store, void* arg) {
>@@ -1115,9 +1103,6 @@ int OpenSSLStreamAdapter::SSLVerifyCallback(X509_STORE_CTX* store, void* arg) {
> 
> #if defined(OPENSSL_IS_BORINGSSL)
>   STACK_OF(X509)* chain = SSL_get_peer_full_cert_chain(ssl);
>-  // Creates certificate.
>-  stream->peer_certificate_.reset(
>-      new OpenSSLCertificate(sk_X509_value(chain, 0)));
>   // Creates certificate chain.
>   std::vector<std::unique_ptr<SSLCertificate>> cert_chain;
>   for (X509* cert : chain) {
>@@ -1126,9 +1111,9 @@ int OpenSSLStreamAdapter::SSLVerifyCallback(X509_STORE_CTX* store, void* arg) {
>   stream->peer_cert_chain_.reset(new SSLCertChain(std::move(cert_chain)));
> #else
>   // Record the peer's certificate.
>-  X509* cert = SSL_get_peer_certificate(ssl);
>-  stream->peer_certificate_.reset(new OpenSSLCertificate(cert));
>-  X509_free(cert);
>+  X509* cert = X509_STORE_CTX_get0_cert(store);
>+  stream->peer_cert_chain_.reset(
>+      new SSLCertChain(new OpenSSLCertificate(cert)));
> #endif
> 
>   // If the peer certificate digest isn't known yet, we'll wait to verify
>@@ -1164,26 +1149,26 @@ struct cipher_list {
> 
> // TODO(torbjorng): Perhaps add more cipher suites to these lists.
> static const cipher_list OK_RSA_ciphers[] = {
>-  CDEF(ECDHE_RSA_WITH_AES_128_CBC_SHA),
>-  CDEF(ECDHE_RSA_WITH_AES_256_CBC_SHA),
>-  CDEF(ECDHE_RSA_WITH_AES_128_GCM_SHA256),
>+    CDEF(ECDHE_RSA_WITH_AES_128_CBC_SHA),
>+    CDEF(ECDHE_RSA_WITH_AES_256_CBC_SHA),
>+    CDEF(ECDHE_RSA_WITH_AES_128_GCM_SHA256),
> #ifdef TLS1_CK_ECDHE_RSA_WITH_AES_256_GCM_SHA256
>-  CDEF(ECDHE_RSA_WITH_AES_256_GCM_SHA256),
>+    CDEF(ECDHE_RSA_WITH_AES_256_GCM_SHA256),
> #endif
> #ifdef TLS1_CK_ECDHE_RSA_WITH_CHACHA20_POLY1305_SHA256
>-  CDEF(ECDHE_RSA_WITH_CHACHA20_POLY1305_SHA256),
>+    CDEF(ECDHE_RSA_WITH_CHACHA20_POLY1305_SHA256),
> #endif
> };
> 
> static const cipher_list OK_ECDSA_ciphers[] = {
>-  CDEF(ECDHE_ECDSA_WITH_AES_128_CBC_SHA),
>-  CDEF(ECDHE_ECDSA_WITH_AES_256_CBC_SHA),
>-  CDEF(ECDHE_ECDSA_WITH_AES_128_GCM_SHA256),
>+    CDEF(ECDHE_ECDSA_WITH_AES_128_CBC_SHA),
>+    CDEF(ECDHE_ECDSA_WITH_AES_256_CBC_SHA),
>+    CDEF(ECDHE_ECDSA_WITH_AES_128_GCM_SHA256),
> #ifdef TLS1_CK_ECDHE_ECDSA_WITH_AES_256_GCM_SHA256
>-  CDEF(ECDHE_ECDSA_WITH_AES_256_GCM_SHA256),
>+    CDEF(ECDHE_ECDSA_WITH_AES_256_GCM_SHA256),
> #endif
> #ifdef TLS1_CK_ECDHE_ECDSA_WITH_CHACHA20_POLY1305_SHA256
>-  CDEF(ECDHE_ECDSA_WITH_CHACHA20_POLY1305_SHA256),
>+    CDEF(ECDHE_ECDSA_WITH_CHACHA20_POLY1305_SHA256),
> #endif
> };
> #undef CDEF
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslstreamadapter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslstreamadapter.h
>index b43dcc7e018..61ffc3dbfe2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslstreamadapter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslstreamadapter.h
>@@ -11,19 +11,16 @@
> #ifndef RTC_BASE_OPENSSLSTREAMADAPTER_H_
> #define RTC_BASE_OPENSSLSTREAMADAPTER_H_
> 
>-#include <string>
>+#include <openssl/ossl_typ.h>
>+
> #include <memory>
>+#include <string>
> #include <vector>
> 
> #include "rtc_base/buffer.h"
> #include "rtc_base/opensslidentity.h"
> #include "rtc_base/sslstreamadapter.h"
> 
>-typedef struct ssl_st SSL;
>-typedef struct ssl_ctx_st SSL_CTX;
>-typedef struct ssl_cipher_st SSL_CIPHER;
>-typedef struct x509_store_ctx_st X509_STORE_CTX;
>-
> namespace rtc {
> 
> // This class was written with OpenSSLAdapter (a socket adapter) as a
>@@ -69,8 +66,6 @@ class OpenSSLStreamAdapter : public SSLStreamAdapter {
>       size_t digest_len,
>       SSLPeerCertificateDigestError* error = nullptr) override;
> 
>-  std::unique_ptr<SSLCertificate> GetPeerCertificate() const override;
>-
>   std::unique_ptr<SSLCertChain> GetPeerSSLCertChain() const override;
> 
>   // Goes from state SSL_NONE to either SSL_CONNECTING or SSL_WAIT, depending
>@@ -130,14 +125,14 @@ class OpenSSLStreamAdapter : public SSLStreamAdapter {
>     // Before calling one of the StartSSL methods, data flows
>     // in clear text.
>     SSL_NONE,
>-    SSL_WAIT,  // waiting for the stream to open to start SSL negotiation
>+    SSL_WAIT,        // waiting for the stream to open to start SSL negotiation
>     SSL_CONNECTING,  // SSL negotiation in progress
>-    SSL_CONNECTED,  // SSL stream successfully established
>-    SSL_ERROR,  // some SSL error occurred, stream is closed
>-    SSL_CLOSED  // Clean close
>+    SSL_CONNECTED,   // SSL stream successfully established
>+    SSL_ERROR,       // some SSL error occurred, stream is closed
>+    SSL_CLOSED       // Clean close
>   };
> 
>-  enum { MSG_TIMEOUT = MSG_MAX+1};
>+  enum { MSG_TIMEOUT = MSG_MAX + 1 };
> 
>   // The following three methods return 0 on success and a negative
>   // error code on failure. The error code may be from OpenSSL or -1
>@@ -197,9 +192,8 @@ class OpenSSLStreamAdapter : public SSLStreamAdapter {
> 
>   // Our key and certificate.
>   std::unique_ptr<OpenSSLIdentity> identity_;
>-  // The certificate that the peer presented. Initially null, until the
>+  // The certificate chain that the peer presented. Initially null, until the
>   // connection is established.
>-  std::unique_ptr<OpenSSLCertificate> peer_certificate_;
>   std::unique_ptr<SSLCertChain> peer_cert_chain_;
>   bool peer_certificate_verified_ = false;
>   // The digest of the certificate that the peer must present.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslutility.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslutility.cc
>new file mode 100644
>index 00000000000..46f4547436e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslutility.cc
>@@ -0,0 +1,136 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "rtc_base/opensslutility.h"
>+
>+#include <memory>
>+
>+#if defined(WEBRTC_POSIX)
>+#include <unistd.h>
>+#endif
>+
>+#if defined(WEBRTC_WIN)
>+// Must be included first before openssl headers.
>+#include "rtc_base/win32.h"  // NOLINT
>+#endif                       // WEBRTC_WIN
>+
>+#include <openssl/bio.h>
>+#include <openssl/crypto.h>
>+#include <openssl/err.h>
>+#include <openssl/x509.h>
>+#include <openssl/x509v3.h>
>+
>+#include "rtc_base/arraysize.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/logging.h"
>+#include "rtc_base/numerics/safe_conversions.h"
>+#include "rtc_base/openssl.h"
>+#include "rtc_base/opensslcertificate.h"
>+#ifndef WEBRTC_EXCLUDE_BUILT_IN_SSL_ROOT_CERTS
>+#include "rtc_base/sslroots.h"
>+#endif  // WEBRTC_EXCLUDE_BUILT_IN_SSL_ROOT_CERTS
>+
>+namespace rtc {
>+namespace openssl {
>+
>+// Holds various helper methods.
>+namespace {
>+void LogCertificates(SSL* ssl, X509* certificate) {
>+// Logging certificates is extremely verbose. So it is disabled by default.
>+#ifdef LOG_CERTIFICATES
>+  BIO* mem = BIO_new(BIO_s_mem());
>+  if (mem == nullptr) {
>+    RTC_DLOG(LS_ERROR) << "BIO_new() failed to allocate memory.";
>+    return;
>+  }
>+
>+  RTC_DLOG(LS_INFO) << "Certificate from server:";
>+  X509_print_ex(mem, certificate, XN_FLAG_SEP_CPLUS_SPC, X509_FLAG_NO_HEADER);
>+  BIO_write(mem, "\0", 1);
>+
>+  char* buffer = nullptr;
>+  BIO_get_mem_data(mem, &buffer);
>+  if (buffer != nullptr) {
>+    RTC_DLOG(LS_INFO) << buffer;
>+  } else {
>+    RTC_DLOG(LS_ERROR) << "BIO_get_mem_data() failed to get buffer.";
>+  }
>+  BIO_free(mem);
>+
>+  const char* cipher_name = SSL_CIPHER_get_name(SSL_get_current_cipher(ssl));
>+  if (cipher_name != nullptr) {
>+    RTC_DLOG(LS_INFO) << "Cipher: " << cipher_name;
>+  } else {
>+    RTC_DLOG(LS_ERROR) << "SSL_CIPHER_DESCRIPTION() failed to get cipher_name.";
>+  }
>+#endif
>+}
>+}  // namespace
>+
>+bool VerifyPeerCertMatchesHost(SSL* ssl, const std::string& host) {
>+  if (host.empty()) {
>+    RTC_DLOG(LS_ERROR) << "Hostname is empty. Cannot verify peer certificate.";
>+    return false;
>+  }
>+
>+  if (ssl == nullptr) {
>+    RTC_DLOG(LS_ERROR) << "SSL is nullptr. Cannot verify peer certificate.";
>+    return false;
>+  }
>+
>+  X509* certificate = SSL_get_peer_certificate(ssl);
>+  if (certificate == nullptr) {
>+    RTC_DLOG(LS_ERROR)
>+        << "SSL_get_peer_certificate failed. This should never happen.";
>+    return false;
>+  }
>+
>+  LogCertificates(ssl, certificate);
>+
>+  bool is_valid_cert_name =
>+      X509_check_host(certificate, host.c_str(), host.size(), 0, nullptr) == 1;
>+  X509_free(certificate);
>+  return is_valid_cert_name;
>+}
>+
>+void LogSSLErrors(const std::string& prefix) {
>+  char error_buf[200];
>+  unsigned long err;  // NOLINT
>+
>+  while ((err = ERR_get_error()) != 0) {
>+    ERR_error_string_n(err, error_buf, sizeof(error_buf));
>+    RTC_LOG(LS_ERROR) << prefix << ": " << error_buf << "\n";
>+  }
>+}
>+
>+#ifndef WEBRTC_EXCLUDE_BUILT_IN_SSL_ROOT_CERTS
>+bool LoadBuiltinSSLRootCertificates(SSL_CTX* ctx) {
>+  int count_of_added_certs = 0;
>+  for (size_t i = 0; i < arraysize(kSSLCertCertificateList); i++) {
>+    const unsigned char* cert_buffer = kSSLCertCertificateList[i];
>+    size_t cert_buffer_len = kSSLCertCertificateSizeList[i];
>+    X509* cert = d2i_X509(nullptr, &cert_buffer,
>+                          checked_cast<long>(cert_buffer_len));  // NOLINT
>+    if (cert) {
>+      int return_value = X509_STORE_add_cert(SSL_CTX_get_cert_store(ctx), cert);
>+      if (return_value == 0) {
>+        RTC_LOG(LS_WARNING) << "Unable to add certificate.";
>+      } else {
>+        count_of_added_certs++;
>+      }
>+      X509_free(cert);
>+    }
>+  }
>+  return count_of_added_certs > 0;
>+}
>+#endif  // WEBRTC_EXCLUDE_BUILT_IN_SSL_ROOT_CERTS
>+
>+}  // namespace openssl
>+}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslutility.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslutility.h
>new file mode 100644
>index 00000000000..7cb38b5b527
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslutility.h
>@@ -0,0 +1,41 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef RTC_BASE_OPENSSLUTILITY_H_
>+#define RTC_BASE_OPENSSLUTILITY_H_
>+
>+#include <openssl/ossl_typ.h>
>+#include <string>
>+#include "rtc_base/sslcertificate.h"
>+
>+namespace rtc {
>+// The openssl namespace holds static helper methods. All methods related
>+// to OpenSSL that are commonly used and don't require global state should be
>+// placed here.
>+namespace openssl {
>+// Verifies that the hostname provided matches that in the peer certificate
>+// attached to this SSL state.
>+bool VerifyPeerCertMatchesHost(SSL* ssl, const std::string& host);
>+
>+// Logs all the errors in the OpenSSL errror queue from the current thread. A
>+// prefix can be provided for context.
>+void LogSSLErrors(const std::string& prefix);
>+
>+#ifndef WEBRTC_EXCLUDE_BUILT_IN_SSL_ROOT_CERTS
>+// Attempt to add the certificates from the loader into the SSL_CTX. False is
>+// returned only if there are no certificates returned from the loader or none
>+// of them can be added to the TrustStore for the provided context.
>+bool LoadBuiltinSSLRootCertificates(SSL_CTX* ssl_ctx);
>+#endif  // WEBRTC_EXCLUDE_BUILT_IN_SSL_ROOT_CERTS
>+
>+}  // namespace openssl
>+}  // namespace rtc
>+
>+#endif  // RTC_BASE_OPENSSLUTILITY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslutility_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslutility_unittest.cc
>new file mode 100644
>index 00000000000..2f952ae4b27
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslutility_unittest.cc
>@@ -0,0 +1,277 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <string>
>+#include <vector>
>+
>+#if defined(WEBRTC_POSIX)
>+#include <unistd.h>
>+#endif
>+
>+#if defined(WEBRTC_WIN)
>+// Must be included first before openssl headers.
>+#include "rtc_base/win32.h"  // NOLINT
>+#endif                       // WEBRTC_WIN
>+
>+#include <openssl/bio.h>
>+#include <openssl/crypto.h>
>+#include <openssl/evp.h>
>+#include <openssl/ssl.h>
>+#include <openssl/x509.h>
>+#include <openssl/x509v3.h>
>+
>+#include "rtc_base/arraysize.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/gunit.h"
>+#include "rtc_base/numerics/safe_conversions.h"
>+#include "rtc_base/openssl.h"
>+#include "rtc_base/opensslutility.h"
>+#include "rtc_base/sslroots.h"
>+#include "test/gmock.h"
>+
>+namespace rtc {
>+namespace {
>+// Fake P-256 key for use with the test certificates below.
>+const unsigned char kFakeSSLPrivateKey[] = {
>+    0x30, 0x81, 0x87, 0x02, 0x01, 0x00, 0x30, 0x13, 0x06, 0x07, 0x2a, 0x86,
>+    0x48, 0xce, 0x3d, 0x02, 0x01, 0x06, 0x08, 0x2a, 0x86, 0x48, 0xce, 0x3d,
>+    0x03, 0x01, 0x07, 0x04, 0x6d, 0x30, 0x6b, 0x02, 0x01, 0x01, 0x04, 0x20,
>+    0x07, 0x0f, 0x08, 0x72, 0x7a, 0xd4, 0xa0, 0x4a, 0x9c, 0xdd, 0x59, 0xc9,
>+    0x4d, 0x89, 0x68, 0x77, 0x08, 0xb5, 0x6f, 0xc9, 0x5d, 0x30, 0x77, 0x0e,
>+    0xe8, 0xd1, 0xc9, 0xce, 0x0a, 0x8b, 0xb4, 0x6a, 0xa1, 0x44, 0x03, 0x42,
>+    0x00, 0x04, 0xe6, 0x2b, 0x69, 0xe2, 0xbf, 0x65, 0x9f, 0x97, 0xbe, 0x2f,
>+    0x1e, 0x0d, 0x94, 0x8a, 0x4c, 0xd5, 0x97, 0x6b, 0xb7, 0xa9, 0x1e, 0x0d,
>+    0x46, 0xfb, 0xdd, 0xa9, 0xa9, 0x1e, 0x9d, 0xdc, 0xba, 0x5a, 0x01, 0xe7,
>+    0xd6, 0x97, 0xa8, 0x0a, 0x18, 0xf9, 0xc3, 0xc4, 0xa3, 0x1e, 0x56, 0xe2,
>+    0x7c, 0x83, 0x48, 0xdb, 0x16, 0x1a, 0x1c, 0xf5, 0x1d, 0x7e, 0xf1, 0x94,
>+    0x2d, 0x4b, 0xcf, 0x72, 0x22, 0xc1};
>+
>+// A self-signed certificate with CN *.webrtc.org and SANs foo.test, *.bar.test,
>+// and test.webrtc.org.
>+const unsigned char kFakeSSLCertificate[] = {
>+    0x30, 0x82, 0x02, 0x9e, 0x30, 0x82, 0x02, 0x42, 0xa0, 0x03, 0x02, 0x01,
>+    0x02, 0x02, 0x09, 0x00, 0xc8, 0x83, 0x59, 0x4d, 0x90, 0xc3, 0x5f, 0xc8,
>+    0x30, 0x0c, 0x06, 0x08, 0x2a, 0x86, 0x48, 0xce, 0x3d, 0x04, 0x03, 0x02,
>+    0x05, 0x00, 0x30, 0x81, 0x8d, 0x31, 0x0b, 0x30, 0x09, 0x06, 0x03, 0x55,
>+    0x04, 0x06, 0x13, 0x02, 0x55, 0x53, 0x31, 0x0b, 0x30, 0x09, 0x06, 0x03,
>+    0x55, 0x04, 0x08, 0x0c, 0x02, 0x57, 0x41, 0x31, 0x2c, 0x30, 0x2a, 0x06,
>+    0x03, 0x55, 0x04, 0x0a, 0x0c, 0x23, 0x46, 0x61, 0x6b, 0x65, 0x20, 0x57,
>+    0x65, 0x62, 0x52, 0x54, 0x43, 0x20, 0x43, 0x65, 0x72, 0x74, 0x69, 0x66,
>+    0x69, 0x63, 0x61, 0x74, 0x65, 0x20, 0x46, 0x6f, 0x72, 0x20, 0x54, 0x65,
>+    0x73, 0x74, 0x69, 0x6e, 0x67, 0x31, 0x2c, 0x30, 0x2a, 0x06, 0x03, 0x55,
>+    0x04, 0x0b, 0x0c, 0x23, 0x46, 0x61, 0x6b, 0x65, 0x20, 0x57, 0x65, 0x62,
>+    0x52, 0x54, 0x43, 0x20, 0x43, 0x65, 0x72, 0x74, 0x69, 0x66, 0x69, 0x63,
>+    0x61, 0x74, 0x65, 0x20, 0x46, 0x6f, 0x72, 0x20, 0x54, 0x65, 0x73, 0x74,
>+    0x69, 0x6e, 0x67, 0x31, 0x15, 0x30, 0x13, 0x06, 0x03, 0x55, 0x04, 0x03,
>+    0x0c, 0x0c, 0x2a, 0x2e, 0x77, 0x65, 0x62, 0x72, 0x74, 0x63, 0x2e, 0x6f,
>+    0x72, 0x67, 0x30, 0x1e, 0x17, 0x0d, 0x31, 0x38, 0x30, 0x34, 0x30, 0x33,
>+    0x32, 0x31, 0x35, 0x34, 0x30, 0x38, 0x5a, 0x17, 0x0d, 0x31, 0x39, 0x30,
>+    0x34, 0x30, 0x33, 0x32, 0x31, 0x35, 0x34, 0x30, 0x38, 0x5a, 0x30, 0x81,
>+    0x8d, 0x31, 0x0b, 0x30, 0x09, 0x06, 0x03, 0x55, 0x04, 0x06, 0x13, 0x02,
>+    0x55, 0x53, 0x31, 0x0b, 0x30, 0x09, 0x06, 0x03, 0x55, 0x04, 0x08, 0x0c,
>+    0x02, 0x57, 0x41, 0x31, 0x2c, 0x30, 0x2a, 0x06, 0x03, 0x55, 0x04, 0x0a,
>+    0x0c, 0x23, 0x46, 0x61, 0x6b, 0x65, 0x20, 0x57, 0x65, 0x62, 0x52, 0x54,
>+    0x43, 0x20, 0x43, 0x65, 0x72, 0x74, 0x69, 0x66, 0x69, 0x63, 0x61, 0x74,
>+    0x65, 0x20, 0x46, 0x6f, 0x72, 0x20, 0x54, 0x65, 0x73, 0x74, 0x69, 0x6e,
>+    0x67, 0x31, 0x2c, 0x30, 0x2a, 0x06, 0x03, 0x55, 0x04, 0x0b, 0x0c, 0x23,
>+    0x46, 0x61, 0x6b, 0x65, 0x20, 0x57, 0x65, 0x62, 0x52, 0x54, 0x43, 0x20,
>+    0x43, 0x65, 0x72, 0x74, 0x69, 0x66, 0x69, 0x63, 0x61, 0x74, 0x65, 0x20,
>+    0x46, 0x6f, 0x72, 0x20, 0x54, 0x65, 0x73, 0x74, 0x69, 0x6e, 0x67, 0x31,
>+    0x15, 0x30, 0x13, 0x06, 0x03, 0x55, 0x04, 0x03, 0x0c, 0x0c, 0x2a, 0x2e,
>+    0x77, 0x65, 0x62, 0x72, 0x74, 0x63, 0x2e, 0x6f, 0x72, 0x67, 0x30, 0x59,
>+    0x30, 0x13, 0x06, 0x07, 0x2a, 0x86, 0x48, 0xce, 0x3d, 0x02, 0x01, 0x06,
>+    0x08, 0x2a, 0x86, 0x48, 0xce, 0x3d, 0x03, 0x01, 0x07, 0x03, 0x42, 0x00,
>+    0x04, 0xe6, 0x2b, 0x69, 0xe2, 0xbf, 0x65, 0x9f, 0x97, 0xbe, 0x2f, 0x1e,
>+    0x0d, 0x94, 0x8a, 0x4c, 0xd5, 0x97, 0x6b, 0xb7, 0xa9, 0x1e, 0x0d, 0x46,
>+    0xfb, 0xdd, 0xa9, 0xa9, 0x1e, 0x9d, 0xdc, 0xba, 0x5a, 0x01, 0xe7, 0xd6,
>+    0x97, 0xa8, 0x0a, 0x18, 0xf9, 0xc3, 0xc4, 0xa3, 0x1e, 0x56, 0xe2, 0x7c,
>+    0x83, 0x48, 0xdb, 0x16, 0x1a, 0x1c, 0xf5, 0x1d, 0x7e, 0xf1, 0x94, 0x2d,
>+    0x4b, 0xcf, 0x72, 0x22, 0xc1, 0xa3, 0x81, 0x86, 0x30, 0x81, 0x83, 0x30,
>+    0x1d, 0x06, 0x03, 0x55, 0x1d, 0x0e, 0x04, 0x16, 0x04, 0x14, 0xb7, 0xc0,
>+    0x9a, 0xa7, 0x22, 0xaf, 0xf8, 0x7d, 0xff, 0x68, 0xdb, 0x80, 0xac, 0x0a,
>+    0xb6, 0xdc, 0x64, 0x89, 0xdb, 0xd4, 0x30, 0x1f, 0x06, 0x03, 0x55, 0x1d,
>+    0x23, 0x04, 0x18, 0x30, 0x16, 0x80, 0x14, 0xb7, 0xc0, 0x9a, 0xa7, 0x22,
>+    0xaf, 0xf8, 0x7d, 0xff, 0x68, 0xdb, 0x80, 0xac, 0x0a, 0xb6, 0xdc, 0x64,
>+    0x89, 0xdb, 0xd4, 0x30, 0x0f, 0x06, 0x03, 0x55, 0x1d, 0x13, 0x01, 0x01,
>+    0xff, 0x04, 0x05, 0x30, 0x03, 0x01, 0x01, 0xff, 0x30, 0x30, 0x06, 0x03,
>+    0x55, 0x1d, 0x11, 0x04, 0x29, 0x30, 0x27, 0x82, 0x08, 0x66, 0x6f, 0x6f,
>+    0x2e, 0x74, 0x65, 0x73, 0x74, 0x82, 0x0a, 0x2a, 0x2e, 0x62, 0x61, 0x72,
>+    0x2e, 0x74, 0x65, 0x73, 0x74, 0x82, 0x0f, 0x74, 0x65, 0x73, 0x74, 0x2e,
>+    0x77, 0x65, 0x62, 0x72, 0x74, 0x63, 0x2e, 0x6f, 0x72, 0x67, 0x30, 0x0c,
>+    0x06, 0x08, 0x2a, 0x86, 0x48, 0xce, 0x3d, 0x04, 0x03, 0x02, 0x05, 0x00,
>+    0x03, 0x48, 0x00, 0x30, 0x45, 0x02, 0x21, 0x00, 0x81, 0xcb, 0xe2, 0xf9,
>+    0x04, 0xba, 0xf7, 0xfd, 0x3f, 0x0d, 0x56, 0x37, 0xdb, 0x65, 0x68, 0x07,
>+    0x28, 0x8d, 0xc5, 0xe1, 0x73, 0xb7, 0xce, 0xa5, 0x20, 0x65, 0x15, 0xb2,
>+    0xc6, 0x37, 0x8c, 0x5a, 0x02, 0x20, 0x24, 0x62, 0x74, 0xe8, 0xd9, 0x80,
>+    0x78, 0x2a, 0xbb, 0x87, 0xff, 0x49, 0x99, 0xdb, 0x94, 0xab, 0x06, 0x91,
>+    0xc0, 0x7a, 0xa4, 0x62, 0x61, 0x98, 0x97, 0x47, 0xb7, 0x64, 0x2b, 0x99,
>+    0xc3, 0x71};
>+
>+// A self-signed SSL certificate with only the legacy CN field *.webrtc.org.
>+const unsigned char kFakeSSLCertificateLegacy[] = {
>+    0x30, 0x82, 0x02, 0x6a, 0x30, 0x82, 0x02, 0x0e, 0xa0, 0x03, 0x02, 0x01,
>+    0x02, 0x02, 0x09, 0x00, 0xc8, 0x83, 0x59, 0x4d, 0x90, 0xc3, 0x5f, 0xc8,
>+    0x30, 0x0c, 0x06, 0x08, 0x2a, 0x86, 0x48, 0xce, 0x3d, 0x04, 0x03, 0x02,
>+    0x05, 0x00, 0x30, 0x81, 0x8d, 0x31, 0x0b, 0x30, 0x09, 0x06, 0x03, 0x55,
>+    0x04, 0x06, 0x13, 0x02, 0x55, 0x53, 0x31, 0x0b, 0x30, 0x09, 0x06, 0x03,
>+    0x55, 0x04, 0x08, 0x0c, 0x02, 0x57, 0x41, 0x31, 0x2c, 0x30, 0x2a, 0x06,
>+    0x03, 0x55, 0x04, 0x0a, 0x0c, 0x23, 0x46, 0x61, 0x6b, 0x65, 0x20, 0x57,
>+    0x65, 0x62, 0x52, 0x54, 0x43, 0x20, 0x43, 0x65, 0x72, 0x74, 0x69, 0x66,
>+    0x69, 0x63, 0x61, 0x74, 0x65, 0x20, 0x46, 0x6f, 0x72, 0x20, 0x54, 0x65,
>+    0x73, 0x74, 0x69, 0x6e, 0x67, 0x31, 0x2c, 0x30, 0x2a, 0x06, 0x03, 0x55,
>+    0x04, 0x0b, 0x0c, 0x23, 0x46, 0x61, 0x6b, 0x65, 0x20, 0x57, 0x65, 0x62,
>+    0x52, 0x54, 0x43, 0x20, 0x43, 0x65, 0x72, 0x74, 0x69, 0x66, 0x69, 0x63,
>+    0x61, 0x74, 0x65, 0x20, 0x46, 0x6f, 0x72, 0x20, 0x54, 0x65, 0x73, 0x74,
>+    0x69, 0x6e, 0x67, 0x31, 0x15, 0x30, 0x13, 0x06, 0x03, 0x55, 0x04, 0x03,
>+    0x0c, 0x0c, 0x2a, 0x2e, 0x77, 0x65, 0x62, 0x72, 0x74, 0x63, 0x2e, 0x6f,
>+    0x72, 0x67, 0x30, 0x1e, 0x17, 0x0d, 0x31, 0x38, 0x30, 0x34, 0x30, 0x33,
>+    0x32, 0x31, 0x35, 0x34, 0x30, 0x38, 0x5a, 0x17, 0x0d, 0x31, 0x39, 0x30,
>+    0x34, 0x30, 0x33, 0x32, 0x31, 0x35, 0x34, 0x30, 0x38, 0x5a, 0x30, 0x81,
>+    0x8d, 0x31, 0x0b, 0x30, 0x09, 0x06, 0x03, 0x55, 0x04, 0x06, 0x13, 0x02,
>+    0x55, 0x53, 0x31, 0x0b, 0x30, 0x09, 0x06, 0x03, 0x55, 0x04, 0x08, 0x0c,
>+    0x02, 0x57, 0x41, 0x31, 0x2c, 0x30, 0x2a, 0x06, 0x03, 0x55, 0x04, 0x0a,
>+    0x0c, 0x23, 0x46, 0x61, 0x6b, 0x65, 0x20, 0x57, 0x65, 0x62, 0x52, 0x54,
>+    0x43, 0x20, 0x43, 0x65, 0x72, 0x74, 0x69, 0x66, 0x69, 0x63, 0x61, 0x74,
>+    0x65, 0x20, 0x46, 0x6f, 0x72, 0x20, 0x54, 0x65, 0x73, 0x74, 0x69, 0x6e,
>+    0x67, 0x31, 0x2c, 0x30, 0x2a, 0x06, 0x03, 0x55, 0x04, 0x0b, 0x0c, 0x23,
>+    0x46, 0x61, 0x6b, 0x65, 0x20, 0x57, 0x65, 0x62, 0x52, 0x54, 0x43, 0x20,
>+    0x43, 0x65, 0x72, 0x74, 0x69, 0x66, 0x69, 0x63, 0x61, 0x74, 0x65, 0x20,
>+    0x46, 0x6f, 0x72, 0x20, 0x54, 0x65, 0x73, 0x74, 0x69, 0x6e, 0x67, 0x31,
>+    0x15, 0x30, 0x13, 0x06, 0x03, 0x55, 0x04, 0x03, 0x0c, 0x0c, 0x2a, 0x2e,
>+    0x77, 0x65, 0x62, 0x72, 0x74, 0x63, 0x2e, 0x6f, 0x72, 0x67, 0x30, 0x59,
>+    0x30, 0x13, 0x06, 0x07, 0x2a, 0x86, 0x48, 0xce, 0x3d, 0x02, 0x01, 0x06,
>+    0x08, 0x2a, 0x86, 0x48, 0xce, 0x3d, 0x03, 0x01, 0x07, 0x03, 0x42, 0x00,
>+    0x04, 0xe6, 0x2b, 0x69, 0xe2, 0xbf, 0x65, 0x9f, 0x97, 0xbe, 0x2f, 0x1e,
>+    0x0d, 0x94, 0x8a, 0x4c, 0xd5, 0x97, 0x6b, 0xb7, 0xa9, 0x1e, 0x0d, 0x46,
>+    0xfb, 0xdd, 0xa9, 0xa9, 0x1e, 0x9d, 0xdc, 0xba, 0x5a, 0x01, 0xe7, 0xd6,
>+    0x97, 0xa8, 0x0a, 0x18, 0xf9, 0xc3, 0xc4, 0xa3, 0x1e, 0x56, 0xe2, 0x7c,
>+    0x83, 0x48, 0xdb, 0x16, 0x1a, 0x1c, 0xf5, 0x1d, 0x7e, 0xf1, 0x94, 0x2d,
>+    0x4b, 0xcf, 0x72, 0x22, 0xc1, 0xa3, 0x53, 0x30, 0x51, 0x30, 0x1d, 0x06,
>+    0x03, 0x55, 0x1d, 0x0e, 0x04, 0x16, 0x04, 0x14, 0xb7, 0xc0, 0x9a, 0xa7,
>+    0x22, 0xaf, 0xf8, 0x7d, 0xff, 0x68, 0xdb, 0x80, 0xac, 0x0a, 0xb6, 0xdc,
>+    0x64, 0x89, 0xdb, 0xd4, 0x30, 0x1f, 0x06, 0x03, 0x55, 0x1d, 0x23, 0x04,
>+    0x18, 0x30, 0x16, 0x80, 0x14, 0xb7, 0xc0, 0x9a, 0xa7, 0x22, 0xaf, 0xf8,
>+    0x7d, 0xff, 0x68, 0xdb, 0x80, 0xac, 0x0a, 0xb6, 0xdc, 0x64, 0x89, 0xdb,
>+    0xd4, 0x30, 0x0f, 0x06, 0x03, 0x55, 0x1d, 0x13, 0x01, 0x01, 0xff, 0x04,
>+    0x05, 0x30, 0x03, 0x01, 0x01, 0xff, 0x30, 0x0c, 0x06, 0x08, 0x2a, 0x86,
>+    0x48, 0xce, 0x3d, 0x04, 0x03, 0x02, 0x05, 0x00, 0x03, 0x48, 0x00, 0x30,
>+    0x45, 0x02, 0x21, 0x00, 0xae, 0x51, 0xbc, 0x0f, 0x28, 0x29, 0xd9, 0x35,
>+    0x95, 0xcc, 0x68, 0xf1, 0xc6, 0x3e, 0xfe, 0x56, 0xfd, 0x7f, 0xd2, 0x03,
>+    0x6d, 0x09, 0xc7, 0x9b, 0x83, 0x93, 0xd6, 0xd0, 0xfe, 0x45, 0x34, 0x7c,
>+    0x02, 0x20, 0x6b, 0xaa, 0x95, 0x8c, 0xfc, 0x29, 0x5e, 0x5e, 0xc9, 0xf5,
>+    0x84, 0x0b, 0xc7, 0x15, 0x86, 0xc3, 0xfc, 0x48, 0x55, 0xb5, 0x81, 0x94,
>+    0x73, 0xbd, 0x18, 0xcd, 0x9d, 0x92, 0x47, 0xaa, 0xfd, 0x18};
>+
>+// Creates a client SSL that has completed handshaking with a server that uses
>+// the specified certificate (which must have private key kFakeSSLPrivateKey).
>+// The server is deallocated. This client will have a peer certificate available
>+// and is thus suitable for testing VerifyPeerCertMatchesHost.
>+SSL* CreateSSLWithPeerCertificate(const unsigned char* cert, size_t cert_len) {
>+  X509* x509 =
>+      d2i_X509(nullptr, &cert, checked_cast<long>(cert_len));  // NOLINT
>+  RTC_CHECK(x509);
>+
>+  const unsigned char* key_ptr = kFakeSSLPrivateKey;
>+  EVP_PKEY* key = d2i_PrivateKey(
>+      EVP_PKEY_EC, nullptr, &key_ptr,
>+      checked_cast<long>(arraysize(kFakeSSLPrivateKey)));  // NOLINT
>+  RTC_CHECK(key);
>+
>+  SSL_CTX* ctx = SSL_CTX_new(SSLv23_method());
>+  SSL* client = SSL_new(ctx);
>+  SSL* server = SSL_new(ctx);
>+  SSL_set_connect_state(client);
>+  SSL_set_accept_state(server);
>+
>+  RTC_CHECK(SSL_use_certificate(server, x509));
>+  RTC_CHECK(SSL_use_PrivateKey(server, key));
>+
>+  BIO* bio1;
>+  BIO* bio2;
>+  BIO_new_bio_pair(&bio1, 0, &bio2, 0);
>+  // SSL_set_bio takes ownership of the BIOs.
>+  SSL_set_bio(client, bio1, bio1);
>+  SSL_set_bio(server, bio2, bio2);
>+
>+  for (;;) {
>+    int client_ret = SSL_do_handshake(client);
>+    int client_err = SSL_get_error(client, client_ret);
>+    RTC_CHECK(client_err == SSL_ERROR_NONE ||
>+              client_err == SSL_ERROR_WANT_READ ||
>+              client_err == SSL_ERROR_WANT_WRITE);
>+
>+    int server_ret = SSL_do_handshake(server);
>+    int server_err = SSL_get_error(server, server_ret);
>+    RTC_CHECK(server_err == SSL_ERROR_NONE ||
>+              server_err == SSL_ERROR_WANT_READ ||
>+              server_err == SSL_ERROR_WANT_WRITE);
>+
>+    if (client_ret == 1 && server_ret == 1) {
>+      break;
>+    }
>+  }
>+
>+  SSL_free(server);
>+  SSL_CTX_free(ctx);
>+  EVP_PKEY_free(key);
>+  X509_free(x509);
>+  return client;
>+}
>+}  // namespace
>+
>+TEST(OpenSSLUtilityTest, VerifyPeerCertMatchesHostFailsOnNoPeerCertificate) {
>+  SSL_CTX* ssl_ctx = SSL_CTX_new(DTLSv1_2_client_method());
>+  SSL* ssl = SSL_new(ssl_ctx);
>+
>+  EXPECT_FALSE(openssl::VerifyPeerCertMatchesHost(ssl, "webrtc.org"));
>+
>+  SSL_free(ssl);
>+  SSL_CTX_free(ssl_ctx);
>+}
>+
>+TEST(OpenSSLUtilityTest, VerifyPeerCertMatchesHost) {
>+  SSL* ssl = CreateSSLWithPeerCertificate(kFakeSSLCertificate,
>+                                          arraysize(kFakeSSLCertificate));
>+
>+  // Each of the names in the SAN list is valid.
>+  EXPECT_TRUE(openssl::VerifyPeerCertMatchesHost(ssl, "foo.test"));
>+  EXPECT_TRUE(openssl::VerifyPeerCertMatchesHost(ssl, "a.bar.test"));
>+  EXPECT_TRUE(openssl::VerifyPeerCertMatchesHost(ssl, "b.bar.test"));
>+  EXPECT_TRUE(openssl::VerifyPeerCertMatchesHost(ssl, "test.webrtc.org"));
>+
>+  // If the SAN list is present, the CN is not checked for hosts.
>+  EXPECT_FALSE(openssl::VerifyPeerCertMatchesHost(ssl, "www.webrtc.org"));
>+
>+  // Additional cases around wildcards.
>+  EXPECT_FALSE(openssl::VerifyPeerCertMatchesHost(ssl, "a.b.bar.test"));
>+  EXPECT_FALSE(openssl::VerifyPeerCertMatchesHost(ssl, "notbar.test"));
>+  EXPECT_FALSE(openssl::VerifyPeerCertMatchesHost(ssl, "bar.test"));
>+
>+  SSL_free(ssl);
>+}
>+
>+TEST(OpenSSLUtilityTest, VerifyPeerCertMatchesHostLegacy) {
>+  SSL* ssl = CreateSSLWithPeerCertificate(kFakeSSLCertificateLegacy,
>+                                          arraysize(kFakeSSLCertificateLegacy));
>+
>+  // If there is no SAN list, WebRTC still implements the legacy mechanism which
>+  // checks the CN, no longer supported by modern browsers.
>+  EXPECT_TRUE(openssl::VerifyPeerCertMatchesHost(ssl, "www.webrtc.org"));
>+  EXPECT_TRUE(openssl::VerifyPeerCertMatchesHost(ssl, "alice.webrtc.org"));
>+  EXPECT_TRUE(openssl::VerifyPeerCertMatchesHost(ssl, "bob.webrtc.org"));
>+
>+  EXPECT_FALSE(openssl::VerifyPeerCertMatchesHost(ssl, "a.b.webrtc.org"));
>+  EXPECT_FALSE(openssl::VerifyPeerCertMatchesHost(ssl, "notwebrtc.org"));
>+  EXPECT_FALSE(openssl::VerifyPeerCertMatchesHost(ssl, "webrtc.org"));
>+
>+  SSL_free(ssl);
>+}
>+
>+}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/optionsfile.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/optionsfile.cc
>index c3b6a6ab93d..535859b4cc0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/optionsfile.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/optionsfile.cc
>@@ -18,8 +18,7 @@
> 
> namespace rtc {
> 
>-OptionsFile::OptionsFile(const std::string &path) : path_(path) {
>-}
>+OptionsFile::OptionsFile(const std::string& path) : path_(path) {}
> 
> OptionsFile::~OptionsFile() = default;
> 
>@@ -75,8 +74,8 @@ bool OptionsFile::Save() {
>   int error;
>   for (OptionsMap::const_iterator i = options_.begin(); i != options_.end();
>        ++i) {
>-    res = stream.WriteAll(i->first.c_str(), i->first.length(), &written,
>-        &error);
>+    res =
>+        stream.WriteAll(i->first.c_str(), i->first.length(), &written, &error);
>     if (res != SR_SUCCESS) {
>       break;
>     }
>@@ -85,7 +84,7 @@ bool OptionsFile::Save() {
>       break;
>     }
>     res = stream.WriteAll(i->second.c_str(), i->second.length(), &written,
>-        &error);
>+                          &error);
>     if (res != SR_SUCCESS) {
>       break;
>     }
>@@ -102,7 +101,7 @@ bool OptionsFile::Save() {
>   }
> }
> 
>-bool OptionsFile::IsLegalName(const std::string &name) {
>+bool OptionsFile::IsLegalName(const std::string& name) {
>   for (size_t pos = 0; pos < name.length(); ++pos) {
>     if (name[pos] == '\n' || name[pos] == '\\' || name[pos] == '=') {
>       // Illegal character.
>@@ -113,7 +112,7 @@ bool OptionsFile::IsLegalName(const std::string &name) {
>   return true;
> }
> 
>-bool OptionsFile::IsLegalValue(const std::string &value) {
>+bool OptionsFile::IsLegalValue(const std::string& value) {
>   for (size_t pos = 0; pos < value.length(); ++pos) {
>     if (value[pos] == '\n' || value[pos] == '\\') {
>       // Illegal character.
>@@ -125,7 +124,7 @@ bool OptionsFile::IsLegalValue(const std::string &value) {
> }
> 
> bool OptionsFile::GetStringValue(const std::string& option,
>-                                 std::string *out_val) const {
>+                                 std::string* out_val) const {
>   RTC_LOG(LS_VERBOSE) << "OptionsFile::GetStringValue " << option;
>   if (!IsLegalName(option)) {
>     return false;
>@@ -138,8 +137,7 @@ bool OptionsFile::GetStringValue(const std::string& option,
>   return true;
> }
> 
>-bool OptionsFile::GetIntValue(const std::string& option,
>-                              int *out_val) const {
>+bool OptionsFile::GetIntValue(const std::string& option, int* out_val) const {
>   RTC_LOG(LS_VERBOSE) << "OptionsFile::GetIntValue " << option;
>   if (!IsLegalName(option)) {
>     return false;
>@@ -162,13 +160,13 @@ bool OptionsFile::SetStringValue(const std::string& option,
>   return true;
> }
> 
>-bool OptionsFile::SetIntValue(const std::string& option,
>-                              int value) {
>+bool OptionsFile::SetIntValue(const std::string& option, int value) {
>   RTC_LOG(LS_VERBOSE) << "OptionsFile::SetIntValue " << option << ":" << value;
>   if (!IsLegalName(option)) {
>     return false;
>   }
>-  return ToString(value, &options_[option]);
>+  options_[option] = ToString(value);
>+  return true;
> }
> 
> bool OptionsFile::RemoveValue(const std::string& option) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/optionsfile.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/optionsfile.h
>index 90976ac5795..55660ff2598 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/optionsfile.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/optionsfile.h
>@@ -21,7 +21,7 @@ namespace rtc {
> // first-class options storage system.
> class OptionsFile {
>  public:
>-  OptionsFile(const std::string &path);
>+  OptionsFile(const std::string& path);
>   ~OptionsFile();
> 
>   // Loads the file from disk, overwriting the in-memory values.
>@@ -38,8 +38,8 @@ class OptionsFile {
>  private:
>   typedef std::map<std::string, std::string> OptionsMap;
> 
>-  static bool IsLegalName(const std::string &name);
>-  static bool IsLegalValue(const std::string &value);
>+  static bool IsLegalName(const std::string& name);
>+  static bool IsLegalValue(const std::string& value);
> 
>   std::string path_;
>   OptionsMap options_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/optionsfile_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/optionsfile_unittest.cc
>index d2e3cbd8e3d..fc5bc823ff4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/optionsfile_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/optionsfile_unittest.cc
>@@ -26,16 +26,18 @@ static const std::string kOptionWithNewline = "foo\nbar";
> static const std::string kValueWithEquals = "baz=quux";
> static const std::string kValueWithNewline = "baz\nquux";
> static const std::string kEmptyString = "";
>-static const char kOptionWithUtf8[] = {'O', 'p', 't', '\302', '\256', 'i', 'o',
>-    'n', '\342', '\204', '\242', '\0'};  // Opt(R)io(TM).
>-static const char kValueWithUtf8[] = {'V', 'a', 'l', '\302', '\256', 'v', 'e',
>-    '\342', '\204', '\242', '\0'};  // Val(R)ue(TM).
>+static const char kOptionWithUtf8[] = {'O',    'p', 't', '\302', '\256',
>+                                       'i',    'o', 'n', '\342', '\204',
>+                                       '\242', '\0'};  // Opt(R)io(TM).
>+static const char kValueWithUtf8[] = {
>+    'V', 'a',    'l',    '\302', '\256', 'v',
>+    'e', '\342', '\204', '\242', '\0'};  // Val(R)ue(TM).
> static int kTestInt1 = 12345;
> static int kTestInt2 = 67890;
> static int kNegInt = -634;
> static int kZero = 0;
> 
>-#if defined (WEBRTC_ANDROID)
>+#if defined(WEBRTC_ANDROID)
> // Fails on Android: https://bugs.chromium.org/p/webrtc/issues/detail?id=4364.
> #define MAYBE_OptionsFileTest DISABLED_OptionsFileTest
> #else
>@@ -50,14 +52,10 @@ class MAYBE_OptionsFileTest : public testing::Test {
>     OpenStore();
>   }
> 
>-  ~MAYBE_OptionsFileTest() override {
>-    webrtc::test::RemoveFile(test_file_);
>-  }
>+  ~MAYBE_OptionsFileTest() override { webrtc::test::RemoveFile(test_file_); }
> 
>  protected:
>-  void OpenStore() {
>-    store_.reset(new OptionsFile(test_file_));
>-  }
>+  void OpenStore() { store_.reset(new OptionsFile(test_file_)); }
> 
>   std::unique_ptr<OptionsFile> store_;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/pathutils.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/pathutils.cc
>index b85d14fc08b..594deb7f0fc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/pathutils.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/pathutils.cc
>@@ -69,24 +69,6 @@ Pathname::Pathname(const std::string& folder, const std::string& filename)
> Pathname& Pathname::operator=(const Pathname&) = default;
> Pathname& Pathname::operator=(Pathname&&) = default;
> 
>-void Pathname::Normalize() {
>-  for (size_t i=0; i<folder_.length(); ++i) {
>-    if (IsFolderDelimiter(folder_[i])) {
>-      folder_[i] = folder_delimiter_;
>-    }
>-  }
>-}
>-
>-void Pathname::clear() {
>-  folder_.clear();
>-  basename_.clear();
>-  extension_.clear();
>-}
>-
>-bool Pathname::empty() const {
>-  return folder_.empty() && basename_.empty() && extension_.empty();
>-}
>-
> std::string Pathname::pathname() const {
>   std::string pathname(folder_);
>   pathname.append(basename_);
>@@ -116,22 +98,6 @@ void Pathname::SetPathname(const std::string& folder,
>   SetFilename(filename);
> }
> 
>-std::string Pathname::folder() const {
>-  return folder_;
>-}
>-
>-std::string Pathname::parent_folder() const {
>-  std::string::size_type pos = std::string::npos;
>-  if (folder_.size() >= 2) {
>-    pos = folder_.find_last_of(FOLDER_DELIMS, folder_.length() - 2);
>-  }
>-  if (pos != std::string::npos) {
>-    return folder_.substr(0, pos + 1);
>-  } else {
>-    return EMPTY_STR;
>-  }
>-}
>-
> void Pathname::SetFolder(const std::string& folder) {
>   folder_.assign(folder);
>   // Ensure folder ends in a path delimiter
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/pathutils.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/pathutils.h
>index b66cf185e39..9543be073f2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/pathutils.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/pathutils.h
>@@ -38,7 +38,7 @@ namespace rtc {
> ///////////////////////////////////////////////////////////////////////////////
> 
> class Pathname {
>-public:
>+ public:
>   // Folder delimiters are slash and backslash
>   static bool IsFolderDelimiter(char ch);
>   static char DefaultFolderDelimiter();
>@@ -52,16 +52,6 @@ public:
>   Pathname& operator=(const Pathname&);
>   Pathname& operator=(Pathname&&);
> 
>-  // Normalize changes all folder delimiters to folder_delimiter()
>-  void Normalize();
>-
>-  // Reset to the empty pathname
>-  void clear();
>-
>-  // Returns true if the pathname is empty.  Note: this->pathname().empty()
>-  // is always false.
>-  bool empty() const;
>-
>   // Returns the folder and filename components.  If the pathname is empty,
>   // returns a string representing the current directory (as a relative path,
>   // i.e., ".").
>@@ -69,8 +59,6 @@ public:
>   void SetPathname(const std::string& pathname);
>   void SetPathname(const std::string& folder, const std::string& filename);
> 
>-  std::string folder() const;
>-  std::string parent_folder() const;
>   // SetFolder and AppendFolder will append a folder delimiter, if needed.
>   void SetFolder(const std::string& folder);
>   void AppendFolder(const std::string& folder);
>@@ -83,11 +71,11 @@ public:
>   std::string filename() const;
>   bool SetFilename(const std::string& filename);
> 
>-private:
>+ private:
>   std::string folder_, basename_, extension_;
>   char folder_delimiter_;
> };
> 
> }  // namespace rtc
> 
>-#endif // RTC_BASE_PATHUTILS_H_
>+#endif  // RTC_BASE_PATHUTILS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/pathutils_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/pathutils_unittest.cc
>index cbd33ed9e19..fae4f0aba59 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/pathutils_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/pathutils_unittest.cc
>@@ -16,29 +16,21 @@ TEST(Pathname, ReturnsDotForEmptyPathname) {
>       std::string(".") + rtc::Pathname::DefaultFolderDelimiter();
> 
>   rtc::Pathname path("/", "");
>-  EXPECT_FALSE(path.empty());
>-  EXPECT_FALSE(path.folder().empty());
>   EXPECT_TRUE (path.filename().empty());
>   EXPECT_FALSE(path.pathname().empty());
>   EXPECT_EQ(std::string("/"), path.pathname());
> 
>   path.SetPathname("", "foo");
>-  EXPECT_FALSE(path.empty());
>-  EXPECT_TRUE (path.folder().empty());
>   EXPECT_FALSE(path.filename().empty());
>   EXPECT_FALSE(path.pathname().empty());
>   EXPECT_EQ(std::string("foo"), path.pathname());
> 
>   path.SetPathname("", "");
>-  EXPECT_TRUE (path.empty());
>-  EXPECT_TRUE (path.folder().empty());
>   EXPECT_TRUE (path.filename().empty());
>   EXPECT_FALSE(path.pathname().empty());
>   EXPECT_EQ(kCWD, path.pathname());
> 
>   path.SetPathname(kCWD, "");
>-  EXPECT_FALSE(path.empty());
>-  EXPECT_FALSE(path.folder().empty());
>   EXPECT_TRUE (path.filename().empty());
>   EXPECT_FALSE(path.pathname().empty());
>   EXPECT_EQ(kCWD, path.pathname());
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/physicalsocketserver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/physicalsocketserver.cc
>index bba921b6692..0e9a17a6af5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/physicalsocketserver.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/physicalsocketserver.cc
>@@ -10,7 +10,7 @@
> #include "rtc_base/physicalsocketserver.h"
> 
> #if defined(_MSC_VER) && _MSC_VER < 1300
>-#pragma warning(disable:4786)
>+#pragma warning(disable : 4786)
> #endif
> 
> #ifdef MEMORY_SANITIZER
>@@ -18,21 +18,20 @@
> #endif
> 
> #if defined(WEBRTC_POSIX)
>-#include <string.h>
> #include <fcntl.h>
>+#include <string.h>
> #if defined(WEBRTC_USE_EPOLL)
> // "poll" will be used to wait for the signal dispatcher.
> #include <poll.h>
> #endif
>+#include <signal.h>
> #include <sys/ioctl.h>
>-#include <sys/time.h>
> #include <sys/select.h>
>+#include <sys/time.h>
> #include <unistd.h>
>-#include <signal.h>
> #endif
> 
> #if defined(WEBRTC_WIN)
>-#define WIN32_LEAN_AND_MEAN
> #include <windows.h>
> #include <winsock2.h>
> #include <ws2tcpip.h>
>@@ -45,7 +44,6 @@
> #include <map>
> 
> #include "rtc_base/arraysize.h"
>-#include "rtc_base/basictypes.h"
> #include "rtc_base/byteorder.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>@@ -64,7 +62,7 @@
> 
> #if defined(WEBRTC_POSIX)
> #include <netinet/tcp.h>  // for TCP_NODELAY
>-#define IP_MTU 14 // Until this is integrated from linux/in.h to netinet/in.h
>+#define IP_MTU 14  // Until this is integrated from linux/in.h to netinet/in.h
> typedef void* SockOptArg;
> 
> #endif  // WEBRTC_POSIX
>@@ -113,40 +111,12 @@ std::unique_ptr<SocketServer> SocketServer::CreateDefault() {
> #endif
> }
> 
>-#if defined(WEBRTC_WIN)
>-// Standard MTUs, from RFC 1191
>-const uint16_t PACKET_MAXIMUMS[] = {
>-    65535,  // Theoretical maximum, Hyperchannel
>-    32000,  // Nothing
>-    17914,  // 16Mb IBM Token Ring
>-    8166,   // IEEE 802.4
>-    // 4464,   // IEEE 802.5 (4Mb max)
>-    4352,   // FDDI
>-    // 2048,   // Wideband Network
>-    2002,   // IEEE 802.5 (4Mb recommended)
>-    // 1536,   // Expermental Ethernet Networks
>-    // 1500,   // Ethernet, Point-to-Point (default)
>-    1492,   // IEEE 802.3
>-    1006,   // SLIP, ARPANET
>-    // 576,    // X.25 Networks
>-    // 544,    // DEC IP Portal
>-    // 512,    // NETBIOS
>-    508,    // IEEE 802/Source-Rt Bridge, ARCNET
>-    296,    // Point-to-Point (low delay)
>-    68,     // Official minimum
>-    0,      // End of list marker
>-};
>-
>-static const int IP_HEADER_SIZE = 20u;
>-static const int IPV6_HEADER_SIZE = 40u;
>-static const int ICMP_HEADER_SIZE = 8u;
>-static const int ICMP_PING_TIMEOUT_MILLIS = 10000u;
>-#endif
>-
> PhysicalSocket::PhysicalSocket(PhysicalSocketServer* ss, SOCKET s)
>-  : ss_(ss), s_(s), error_(0),
>-    state_((s == INVALID_SOCKET) ? CS_CLOSED : CS_CONNECTED),
>-    resolver_(nullptr) {
>+    : ss_(ss),
>+      s_(s),
>+      error_(0),
>+      state_((s == INVALID_SOCKET) ? CS_CLOSED : CS_CONNECTED),
>+      resolver_(nullptr) {
> #if defined(WEBRTC_WIN)
>   // EnsureWinsockInit() ensures that winsock is initialized. The default
>   // version of this function doesn't do anything because winsock is
>@@ -281,8 +251,7 @@ int PhysicalSocket::Connect(const SocketAddress& addr) {
> }
> 
> int PhysicalSocket::DoConnect(const SocketAddress& connect_addr) {
>-  if ((s_ == INVALID_SOCKET) &&
>-      !Create(connect_addr.family(), SOCK_STREAM)) {
>+  if ((s_ == INVALID_SOCKET) && !Create(connect_addr.family(), SOCK_STREAM)) {
>     return SOCKET_ERROR;
>   }
>   sockaddr_storage addr_storage;
>@@ -347,8 +316,8 @@ int PhysicalSocket::SetOption(Option opt, int value) {
> }
> 
> int PhysicalSocket::Send(const void* pv, size_t cb) {
>-  int sent = DoSend(s_, reinterpret_cast<const char *>(pv),
>-      static_cast<int>(cb),
>+  int sent = DoSend(
>+      s_, reinterpret_cast<const char*>(pv), static_cast<int>(cb),
> #if defined(WEBRTC_LINUX) && !defined(WEBRTC_ANDROID)
>       // Suppress SIGPIPE. Without this, attempting to send on a socket whose
>       // other end is closed will result in a SIGPIPE signal being raised to
>@@ -376,15 +345,15 @@ int PhysicalSocket::SendTo(const void* buffer,
>                            const SocketAddress& addr) {
>   sockaddr_storage saddr;
>   size_t len = addr.ToSockAddrStorage(&saddr);
>-  int sent = DoSendTo(
>-      s_, static_cast<const char *>(buffer), static_cast<int>(length),
>+  int sent =
>+      DoSendTo(s_, static_cast<const char*>(buffer), static_cast<int>(length),
> #if defined(WEBRTC_LINUX) && !defined(WEBRTC_ANDROID)
>-      // Suppress SIGPIPE. See above for explanation.
>-      MSG_NOSIGNAL,
>+               // Suppress SIGPIPE. See above for explanation.
>+               MSG_NOSIGNAL,
> #else
>-      0,
>+               0,
> #endif
>-      reinterpret_cast<sockaddr*>(&saddr), static_cast<int>(len));
>+               reinterpret_cast<sockaddr*>(&saddr), static_cast<int>(len));
>   UpdateLastError();
>   MaybeRemapSendError();
>   // We have seen minidumps where this may be false.
>@@ -397,8 +366,8 @@ int PhysicalSocket::SendTo(const void* buffer,
> }
> 
> int PhysicalSocket::Recv(void* buffer, size_t length, int64_t* timestamp) {
>-  int received = ::recv(s_, static_cast<char*>(buffer),
>-                        static_cast<int>(length), 0);
>+  int received =
>+      ::recv(s_, static_cast<char*>(buffer), static_cast<int>(length), 0);
>   if ((received == 0) && (length != 0)) {
>     // Note: on graceful shutdown, recv can return 0.  In this case, we
>     // pretend it is blocking, and then signal close, so that simplifying
>@@ -601,20 +570,24 @@ int PhysicalSocket::TranslateOption(Option opt, int* slevel, int* sopt) {
>   return 0;
> }
> 
>-SocketDispatcher::SocketDispatcher(PhysicalSocketServer *ss)
>+SocketDispatcher::SocketDispatcher(PhysicalSocketServer* ss)
> #if defined(WEBRTC_WIN)
>-  : PhysicalSocket(ss), id_(0), signal_close_(false)
>+    : PhysicalSocket(ss),
>+      id_(0),
>+      signal_close_(false)
> #else
>-  : PhysicalSocket(ss)
>+    : PhysicalSocket(ss)
> #endif
> {
> }
> 
>-SocketDispatcher::SocketDispatcher(SOCKET s, PhysicalSocketServer *ss)
>+SocketDispatcher::SocketDispatcher(SOCKET s, PhysicalSocketServer* ss)
> #if defined(WEBRTC_WIN)
>-  : PhysicalSocket(ss, s), id_(0), signal_close_(false)
>+    : PhysicalSocket(ss, s),
>+      id_(0),
>+      signal_close_(false)
> #else
>-  : PhysicalSocket(ss, s)
>+    : PhysicalSocket(ss, s)
> #endif
> {
> }
>@@ -625,7 +598,7 @@ SocketDispatcher::~SocketDispatcher() {
> 
> bool SocketDispatcher::Initialize() {
>   RTC_DCHECK(s_ != INVALID_SOCKET);
>-  // Must be a non-blocking
>+// Must be a non-blocking
> #if defined(WEBRTC_WIN)
>   u_long argp = 1;
>   ioctlsocket(s_, FIONBIO, &argp);
>@@ -658,7 +631,9 @@ bool SocketDispatcher::Create(int family, int type) {
>     return false;
> 
> #if defined(WEBRTC_WIN)
>-  do { id_ = ++next_id_; } while (id_ == 0);
>+  do {
>+    id_ = ++next_id_;
>+  } while (id_ == 0);
> #endif
>   return true;
> }
>@@ -742,7 +717,7 @@ bool SocketDispatcher::IsDescriptorClosed() {
>   }
> }
> 
>-#endif // WEBRTC_POSIX
>+#endif  // WEBRTC_POSIX
> 
> uint32_t SocketDispatcher::GetRequestedEvents() {
>   return enabled_events();
>@@ -753,7 +728,7 @@ void SocketDispatcher::OnPreEvent(uint32_t ff) {
>     state_ = CS_CONNECTED;
> 
> #if defined(WEBRTC_WIN)
>-  // We set CS_CLOSED from CheckSignalClose.
>+// We set CS_CLOSED from CheckSignalClose.
> #elif defined(WEBRTC_POSIX)
>   if ((ff & DE_CLOSE) != 0)
>     state_ = CS_CLOSED;
>@@ -833,7 +808,7 @@ void SocketDispatcher::OnEvent(uint32_t ff, int err) {
> #endif
> }
> 
>-#endif // WEBRTC_POSIX
>+#endif  // WEBRTC_POSIX
> 
> #if defined(WEBRTC_USE_EPOLL)
> 
>@@ -946,7 +921,7 @@ class EventDispatcher : public Dispatcher {
>   bool IsDescriptorClosed() override { return false; }
> 
>  private:
>-  PhysicalSocketServer *ss_;
>+  PhysicalSocketServer* ss_;
>   int afd_[2];
>   bool fSignaled_;
>   CriticalSection crit_;
>@@ -968,8 +943,8 @@ class PosixSignalHandler {
>   // sort of user-defined void * parameter, so they can't access anything that
>   // isn't global.)
>   static PosixSignalHandler* Instance() {
>-    RTC_DEFINE_STATIC_LOCAL(PosixSignalHandler, instance, ());
>-    return &instance;
>+    static PosixSignalHandler* const instance = new PosixSignalHandler();
>+    return instance;
>   }
> 
>   // Returns true if the given signal number is set.
>@@ -991,9 +966,7 @@ class PosixSignalHandler {
>   }
> 
>   // Returns the file descriptor to monitor for signal events.
>-  int GetDescriptor() const {
>-    return afd_[0];
>-  }
>+  int GetDescriptor() const { return afd_[0]; }
> 
>   // This is called directly from our real signal handler, so it must be
>   // signal-handler-safe. That means it cannot assume anything about the
>@@ -1030,8 +1003,7 @@ class PosixSignalHandler {
>     if (fcntl(afd_[1], F_SETFL, O_NONBLOCK) < 0) {
>       RTC_LOG_ERR(LS_WARNING) << "fcntl #2 failed";
>     }
>-    memset(const_cast<void *>(static_cast<volatile void *>(received_signal_)),
>-           0,
>+    memset(const_cast<void*>(static_cast<volatile void*>(received_signal_)), 0,
>            sizeof(received_signal_));
>   }
> 
>@@ -1069,13 +1041,11 @@ class PosixSignalHandler {
> 
> class PosixSignalDispatcher : public Dispatcher {
>  public:
>-  PosixSignalDispatcher(PhysicalSocketServer *owner) : owner_(owner) {
>+  PosixSignalDispatcher(PhysicalSocketServer* owner) : owner_(owner) {
>     owner_->Add(this);
>   }
> 
>-  ~PosixSignalDispatcher() override {
>-    owner_->Remove(this);
>-  }
>+  ~PosixSignalDispatcher() override { owner_->Remove(this); }
> 
>   uint32_t GetRequestedEvents() override { return DE_READ; }
> 
>@@ -1120,23 +1090,19 @@ class PosixSignalDispatcher : public Dispatcher {
>     handlers_[signum] = handler;
>   }
> 
>-  void ClearHandler(int signum) {
>-    handlers_.erase(signum);
>-  }
>+  void ClearHandler(int signum) { handlers_.erase(signum); }
> 
>-  bool HasHandlers() {
>-    return !handlers_.empty();
>-  }
>+  bool HasHandlers() { return !handlers_.empty(); }
> 
>  private:
>   typedef std::map<int, void (*)(int)> HandlerMap;
> 
>   HandlerMap handlers_;
>   // Our owner.
>-  PhysicalSocketServer *owner_;
>+  PhysicalSocketServer* owner_;
> };
> 
>-#endif // WEBRTC_POSIX
>+#endif  // WEBRTC_POSIX
> 
> #if defined(WEBRTC_WIN)
> static uint32_t FlagsToEvents(uint32_t events) {
>@@ -1154,7 +1120,7 @@ static uint32_t FlagsToEvents(uint32_t events) {
> 
> class EventDispatcher : public Dispatcher {
>  public:
>-  EventDispatcher(PhysicalSocketServer *ss) : ss_(ss) {
>+  EventDispatcher(PhysicalSocketServer* ss) : ss_(ss) {
>     hev_ = WSACreateEvent();
>     if (hev_) {
>       ss_->Add(this);
>@@ -1195,10 +1161,8 @@ class EventDispatcher : public Dispatcher {
> // Sets the value of a boolean value to false when signaled.
> class Signaler : public EventDispatcher {
>  public:
>-  Signaler(PhysicalSocketServer* ss, bool* pf)
>-      : EventDispatcher(ss), pf_(pf) {
>-  }
>-  ~Signaler() override { }
>+  Signaler(PhysicalSocketServer* ss, bool* pf) : EventDispatcher(ss), pf_(pf) {}
>+  ~Signaler() override {}
> 
>   void OnEvent(uint32_t ff, int err) override {
>     if (pf_)
>@@ -1206,11 +1170,10 @@ class Signaler : public EventDispatcher {
>   }
> 
>  private:
>-  bool *pf_;
>+  bool* pf_;
> };
> 
>-PhysicalSocketServer::PhysicalSocketServer()
>-    : fWait_(false) {
>+PhysicalSocketServer::PhysicalSocketServer() : fWait_(false) {
> #if defined(WEBRTC_USE_EPOLL)
>   // Since Linux 2.6.8, the size argument is ignored, but must be greater than
>   // zero. Before that the size served as hint to the kernel for the amount of
>@@ -1248,10 +1211,6 @@ void PhysicalSocketServer::WakeUp() {
>   signal_wakeup_->Signal();
> }
> 
>-Socket* PhysicalSocketServer::CreateSocket(int type) {
>-  return CreateSocket(AF_INET, type);
>-}
>-
> Socket* PhysicalSocketServer::CreateSocket(int family, int type) {
>   PhysicalSocket* socket = new PhysicalSocket(this);
>   if (socket->Create(family, type)) {
>@@ -1262,10 +1221,6 @@ Socket* PhysicalSocketServer::CreateSocket(int family, int type) {
>   }
> }
> 
>-AsyncSocket* PhysicalSocketServer::CreateAsyncSocket(int type) {
>-  return CreateAsyncSocket(AF_INET, type);
>-}
>-
> AsyncSocket* PhysicalSocketServer::CreateAsyncSocket(int family, int type) {
>   SocketDispatcher* dispatcher = new SocketDispatcher(this);
>   if (dispatcher->Create(family, type)) {
>@@ -1286,7 +1241,7 @@ AsyncSocket* PhysicalSocketServer::WrapSocket(SOCKET s) {
>   }
> }
> 
>-void PhysicalSocketServer::Add(Dispatcher *pdispatcher) {
>+void PhysicalSocketServer::Add(Dispatcher* pdispatcher) {
>   CritScope cs(&crit_);
>   if (processing_dispatchers_) {
>     // A dispatcher is being added while a "Wait" call is processing the
>@@ -1305,7 +1260,7 @@ void PhysicalSocketServer::Add(Dispatcher *pdispatcher) {
> #endif  // WEBRTC_USE_EPOLL
> }
> 
>-void PhysicalSocketServer::Remove(Dispatcher *pdispatcher) {
>+void PhysicalSocketServer::Remove(Dispatcher* pdispatcher) {
>   CritScope cs(&crit_);
>   if (processing_dispatchers_) {
>     // A dispatcher is being removed while a "Wait" call is processing the
>@@ -1459,9 +1414,9 @@ bool PhysicalSocketServer::WaitSelect(int cmsWait, bool process_io) {
>   FD_ZERO(&fdsRead);
>   fd_set fdsWrite;
>   FD_ZERO(&fdsWrite);
>-  // Explicitly unpoison these FDs on MemorySanitizer which doesn't handle the
>-  // inline assembly in FD_ZERO.
>-  // http://crbug.com/344505
>+// Explicitly unpoison these FDs on MemorySanitizer which doesn't handle the
>+// inline assembly in FD_ZERO.
>+// http://crbug.com/344505
> #ifdef MEMORY_SANITIZER
>   __msan_unpoison(&fdsRead, sizeof(fdsRead));
>   __msan_unpoison(&fdsWrite, sizeof(fdsWrite));
>@@ -1548,9 +1503,8 @@ bool PhysicalSocketServer::WaitSelect(int cmsWait, bool process_io) {
>       ptvWait->tv_usec = 0;
>       struct timeval tvT;
>       gettimeofday(&tvT, nullptr);
>-      if ((tvStop.tv_sec > tvT.tv_sec)
>-          || ((tvStop.tv_sec == tvT.tv_sec)
>-              && (tvStop.tv_usec > tvT.tv_usec))) {
>+      if ((tvStop.tv_sec > tvT.tv_sec) ||
>+          ((tvStop.tv_sec == tvT.tv_sec) && (tvStop.tv_usec > tvT.tv_usec))) {
>         ptvWait->tv_sec = tvStop.tv_sec - tvT.tv_sec;
>         ptvWait->tv_usec = tvStop.tv_usec - tvT.tv_usec;
>         if (ptvWait->tv_usec < 0) {
>@@ -1840,7 +1794,7 @@ bool PhysicalSocketServer::Wait(int cmsWait, bool process_io) {
>   fWait_ = true;
>   while (fWait_) {
>     std::vector<WSAEVENT> events;
>-    std::vector<Dispatcher *> event_owners;
>+    std::vector<Dispatcher*> event_owners;
> 
>     events.push_back(socket_ev_);
> 
>@@ -1859,8 +1813,7 @@ bool PhysicalSocketServer::Wait(int cmsWait, bool process_io) {
>         if (disp->CheckSignalClose()) {
>           // We just signalled close, don't poll this socket
>         } else if (s != INVALID_SOCKET) {
>-          WSAEventSelect(s,
>-                         events[0],
>+          WSAEventSelect(s, events[0],
>                          FlagsToEvents(disp->GetRequestedEvents()));
>         } else {
>           events.push_back(disp->GetWSAEvent());
>@@ -1884,11 +1837,9 @@ bool PhysicalSocketServer::Wait(int cmsWait, bool process_io) {
>     }
> 
>     // Wait for one of the events to signal
>-    DWORD dw = WSAWaitForMultipleEvents(static_cast<DWORD>(events.size()),
>-                                        &events[0],
>-                                        false,
>-                                        static_cast<DWORD>(cmsNext),
>-                                        false);
>+    DWORD dw =
>+        WSAWaitForMultipleEvents(static_cast<DWORD>(events.size()), &events[0],
>+                                 false, static_cast<DWORD>(cmsNext), false);
> 
>     if (dw == WSA_WAIT_FAILED) {
>       // Failed?
>@@ -1904,7 +1855,7 @@ bool PhysicalSocketServer::Wait(int cmsWait, bool process_io) {
>       CritScope cr(&crit_);
>       int index = dw - WSA_WAIT_EVENT_0;
>       if (index > 0) {
>-        --index; // The first event is the socket event
>+        --index;  // The first event is the socket event
>         Dispatcher* disp = event_owners[index];
>         // The dispatcher could have been removed while waiting for events.
>         if (dispatchers_.find(disp) != dispatchers_.end()) {
>@@ -1995,7 +1946,7 @@ bool PhysicalSocketServer::Wait(int cmsWait, bool process_io) {
>       break;
>     cmsElapsed = TimeSince(msStart);
>     if ((cmsWait != kForever) && (cmsElapsed >= cmsWait)) {
>-       break;
>+      break;
>     }
>   }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/physicalsocketserver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/physicalsocketserver.h
>index f8167744d0d..ee4f936e81f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/physicalsocketserver.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/physicalsocketserver.h
>@@ -26,17 +26,17 @@
> 
> #if defined(WEBRTC_POSIX)
> typedef int SOCKET;
>-#endif // WEBRTC_POSIX
>+#endif  // WEBRTC_POSIX
> 
> namespace rtc {
> 
> // Event constants for the Dispatcher class.
> enum DispatcherEvent {
>-  DE_READ    = 0x0001,
>-  DE_WRITE   = 0x0002,
>+  DE_READ = 0x0001,
>+  DE_WRITE = 0x0002,
>   DE_CONNECT = 0x0004,
>-  DE_CLOSE   = 0x0008,
>-  DE_ACCEPT  = 0x0010,
>+  DE_CLOSE = 0x0008,
>+  DE_ACCEPT = 0x0010,
> };
> 
> class Signaler;
>@@ -67,10 +67,7 @@ class PhysicalSocketServer : public SocketServer {
>   ~PhysicalSocketServer() override;
> 
>   // SocketFactory:
>-  Socket* CreateSocket(int type) override;
>   Socket* CreateSocket(int family, int type) override;
>-
>-  AsyncSocket* CreateAsyncSocket(int type) override;
>   AsyncSocket* CreateAsyncSocket(int family, int type) override;
> 
>   // Internal Factory for Accept (virtual so it can be overwritten in tests).
>@@ -184,8 +181,12 @@ class PhysicalSocket : public AsyncSocket, public sigslot::has_slots<> {
>   virtual int DoSend(SOCKET socket, const char* buf, int len, int flags);
> 
>   // Make virtual so ::sendto can be overwritten in tests.
>-  virtual int DoSendTo(SOCKET socket, const char* buf, int len, int flags,
>-                       const struct sockaddr* dest_addr, socklen_t addrlen);
>+  virtual int DoSendTo(SOCKET socket,
>+                       const char* buf,
>+                       int len,
>+                       int flags,
>+                       const struct sockaddr* dest_addr,
>+                       socklen_t addrlen);
> 
>   void OnResolveResult(AsyncResolverInterface* resolver);
> 
>@@ -217,8 +218,8 @@ class PhysicalSocket : public AsyncSocket, public sigslot::has_slots<> {
> 
> class SocketDispatcher : public Dispatcher, public PhysicalSocket {
>  public:
>-  explicit SocketDispatcher(PhysicalSocketServer *ss);
>-  SocketDispatcher(SOCKET s, PhysicalSocketServer *ss);
>+  explicit SocketDispatcher(PhysicalSocketServer* ss);
>+  SocketDispatcher(SOCKET s, PhysicalSocketServer* ss);
>   ~SocketDispatcher() override;
> 
>   bool Initialize();
>@@ -257,7 +258,7 @@ class SocketDispatcher : public Dispatcher, public PhysicalSocket {
>   int id_;
>   bool signal_close_;
>   int signal_err_;
>-#endif // WEBRTC_WIN
>+#endif  // WEBRTC_WIN
> #if defined(WEBRTC_USE_EPOLL)
>   void MaybeUpdateDispatcher(uint8_t old_events);
> 
>@@ -265,6 +266,6 @@ class SocketDispatcher : public Dispatcher, public PhysicalSocket {
> #endif
> };
> 
>-} // namespace rtc
>+}  // namespace rtc
> 
>-#endif // RTC_BASE_PHYSICALSOCKETSERVER_H_
>+#endif  // RTC_BASE_PHYSICALSOCKETSERVER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/physicalsocketserver_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/physicalsocketserver_unittest.cc
>index d09385bbb3b..1e046c057e5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/physicalsocketserver_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/physicalsocketserver_unittest.cc
>@@ -8,9 +8,9 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include <memory>
> #include <signal.h>
> #include <stdarg.h>
>+#include <memory>
> 
> #include "rtc_base/gunit.h"
> #include "rtc_base/logging.h"
>@@ -39,34 +39,25 @@ class PhysicalSocketTest;
> class FakeSocketDispatcher : public SocketDispatcher {
>  public:
>   explicit FakeSocketDispatcher(PhysicalSocketServer* ss)
>-    : SocketDispatcher(ss) {
>-  }
>+      : SocketDispatcher(ss) {}
> 
>   FakeSocketDispatcher(SOCKET s, PhysicalSocketServer* ss)
>-    : SocketDispatcher(s, ss) {
>-  }
>+      : SocketDispatcher(s, ss) {}
> 
>  protected:
>   SOCKET DoAccept(SOCKET socket, sockaddr* addr, socklen_t* addrlen) override;
>   int DoSend(SOCKET socket, const char* buf, int len, int flags) override;
>-  int DoSendTo(SOCKET socket, const char* buf, int len, int flags,
>-               const struct sockaddr* dest_addr, socklen_t addrlen) override;
>+  int DoSendTo(SOCKET socket,
>+               const char* buf,
>+               int len,
>+               int flags,
>+               const struct sockaddr* dest_addr,
>+               socklen_t addrlen) override;
> };
> 
> class FakePhysicalSocketServer : public PhysicalSocketServer {
>  public:
>-  explicit FakePhysicalSocketServer(PhysicalSocketTest* test)
>-    : test_(test) {
>-  }
>-
>-  AsyncSocket* CreateAsyncSocket(int type) override {
>-    SocketDispatcher* dispatcher = new FakeSocketDispatcher(this);
>-    if (!dispatcher->Create(type)) {
>-      delete dispatcher;
>-      return nullptr;
>-    }
>-    return dispatcher;
>-  }
>+  explicit FakePhysicalSocketServer(PhysicalSocketTest* test) : test_(test) {}
> 
>   AsyncSocket* CreateAsyncSocket(int family, int type) override {
>     SocketDispatcher* dispatcher = new FakeSocketDispatcher(this);
>@@ -120,10 +111,10 @@ class PhysicalSocketTest : public SocketTest {
> 
>  protected:
>   PhysicalSocketTest()
>-    : server_(new FakePhysicalSocketServer(this)),
>-      thread_(server_.get()),
>-      fail_accept_(false),
>-      max_send_size_(-1) {}
>+      : server_(new FakePhysicalSocketServer(this)),
>+        thread_(server_.get()),
>+        fail_accept_(false),
>+        max_send_size_(-1) {}
> 
>   void ConnectInternalAcceptError(const IPAddress& loopback);
>   void WritableAfterPartialWrite(const IPAddress& loopback);
>@@ -146,8 +137,10 @@ SOCKET FakeSocketDispatcher::DoAccept(SOCKET socket,
>   return SocketDispatcher::DoAccept(socket, addr, addrlen);
> }
> 
>-int FakeSocketDispatcher::DoSend(SOCKET socket, const char* buf, int len,
>-    int flags) {
>+int FakeSocketDispatcher::DoSend(SOCKET socket,
>+                                 const char* buf,
>+                                 int len,
>+                                 int flags) {
>   FakePhysicalSocketServer* ss =
>       static_cast<FakePhysicalSocketServer*>(socketserver());
>   if (ss->GetTest()->MaxSendSize() >= 0) {
>@@ -157,8 +150,12 @@ int FakeSocketDispatcher::DoSend(SOCKET socket, const char* buf, int len,
>   return SocketDispatcher::DoSend(socket, buf, len, flags);
> }
> 
>-int FakeSocketDispatcher::DoSendTo(SOCKET socket, const char* buf, int len,
>-    int flags, const struct sockaddr* dest_addr, socklen_t addrlen) {
>+int FakeSocketDispatcher::DoSendTo(SOCKET socket,
>+                                   const char* buf,
>+                                   int len,
>+                                   int flags,
>+                                   const struct sockaddr* dest_addr,
>+                                   socklen_t addrlen) {
>   FakePhysicalSocketServer* ss =
>       static_cast<FakePhysicalSocketServer*>(socketserver());
>   if (ss->GetTest()->MaxSendSize() >= 0) {
>@@ -166,7 +163,7 @@ int FakeSocketDispatcher::DoSendTo(SOCKET socket, const char* buf, int len,
>   }
> 
>   return SocketDispatcher::DoSendTo(socket, buf, len, flags, dest_addr,
>-      addrlen);
>+                                    addrlen);
> }
> 
> TEST_F(PhysicalSocketTest, TestConnectIPv4) {
>@@ -201,13 +198,13 @@ void PhysicalSocketTest::ConnectInternalAcceptError(const IPAddress& loopback) {
>       server_->CreateAsyncSocket(loopback.family(), SOCK_STREAM));
>   sink.Monitor(client1.get());
>   EXPECT_EQ(AsyncSocket::CS_CLOSED, client1->GetState());
>-  EXPECT_PRED1(IsUnspecOrEmptyIP, client1->GetLocalAddress().ipaddr());
>+  EXPECT_TRUE(IsUnspecOrEmptyIP(client1->GetLocalAddress().ipaddr()));
> 
>   std::unique_ptr<AsyncSocket> client2(
>       server_->CreateAsyncSocket(loopback.family(), SOCK_STREAM));
>   sink.Monitor(client2.get());
>   EXPECT_EQ(AsyncSocket::CS_CLOSED, client2->GetState());
>-  EXPECT_PRED1(IsUnspecOrEmptyIP, client2->GetLocalAddress().ipaddr());
>+  EXPECT_TRUE(IsUnspecOrEmptyIP(client2->GetLocalAddress().ipaddr()));
> 
>   // Create server and listen.
>   std::unique_ptr<AsyncSocket> server(
>@@ -289,9 +286,11 @@ void PhysicalSocketTest::WritableAfterPartialWrite(const IPAddress& loopback) {
> 
> // https://bugs.chromium.org/p/webrtc/issues/detail?id=6167
> #if defined(WEBRTC_WIN)
>-#define MAYBE_TestWritableAfterPartialWriteIPv4 DISABLED_TestWritableAfterPartialWriteIPv4
>+#define MAYBE_TestWritableAfterPartialWriteIPv4 \
>+  DISABLED_TestWritableAfterPartialWriteIPv4
> #else
>-#define MAYBE_TestWritableAfterPartialWriteIPv4 TestWritableAfterPartialWriteIPv4
>+#define MAYBE_TestWritableAfterPartialWriteIPv4 \
>+  TestWritableAfterPartialWriteIPv4
> #endif
> TEST_F(PhysicalSocketTest, MAYBE_TestWritableAfterPartialWriteIPv4) {
>   MAYBE_SKIP_IPV4;
>@@ -300,9 +299,11 @@ TEST_F(PhysicalSocketTest, MAYBE_TestWritableAfterPartialWriteIPv4) {
> 
> // https://bugs.chromium.org/p/webrtc/issues/detail?id=6167
> #if defined(WEBRTC_WIN)
>-#define MAYBE_TestWritableAfterPartialWriteIPv6 DISABLED_TestWritableAfterPartialWriteIPv6
>+#define MAYBE_TestWritableAfterPartialWriteIPv6 \
>+  DISABLED_TestWritableAfterPartialWriteIPv6
> #else
>-#define MAYBE_TestWritableAfterPartialWriteIPv6 TestWritableAfterPartialWriteIPv6
>+#define MAYBE_TestWritableAfterPartialWriteIPv6 \
>+  TestWritableAfterPartialWriteIPv6
> #endif
> TEST_F(PhysicalSocketTest, MAYBE_TestWritableAfterPartialWriteIPv6) {
>   MAYBE_SKIP_IPV6;
>@@ -322,7 +323,6 @@ TEST_F(PhysicalSocketTest, TestConnectWithDnsLookupFailIPv6) {
>   SocketTest::TestConnectWithDnsLookupFailIPv6();
> }
> 
>-
> TEST_F(PhysicalSocketTest, TestConnectWithClosedSocketIPv4) {
>   MAYBE_SKIP_IPV4;
>   SocketTest::TestConnectWithClosedSocketIPv4();
>@@ -472,8 +472,7 @@ TEST_F(PhysicalSocketTest,
> }
> 
> // Network binder shouldn't be used if the socket is bound to the "any" IP.
>-TEST_F(PhysicalSocketTest,
>-       NetworkBinderIsNotUsedForAnyIp) {
>+TEST_F(PhysicalSocketTest, NetworkBinderIsNotUsedForAnyIp) {
>   MAYBE_SKIP_IPV4;
>   FakeNetworkBinder fake_network_binder;
>   server_->set_network_binder(&fake_network_binder);
>@@ -538,7 +537,7 @@ class PosixSignalDeliveryTest : public testing::Test {
>   }
> 
>   static std::vector<int> signals_received_;
>-  static Thread *signaled_thread_;
>+  static Thread* signaled_thread_;
> 
>   std::unique_ptr<PhysicalSocketServer> ss_;
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/platform_file.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/platform_file.cc
>index 4f0222d44f0..a4c906a1ed0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/platform_file.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/platform_file.cc
>@@ -23,17 +23,21 @@
> 
> namespace rtc {
> 
>+FILE* FdopenPlatformFileForWriting(PlatformFile file) {
>+  return FdopenPlatformFile(file, "w");
>+}
>+
> #if defined(WEBRTC_WIN)
> const PlatformFile kInvalidPlatformFileValue = INVALID_HANDLE_VALUE;
> 
>-FILE* FdopenPlatformFileForWriting(PlatformFile file) {
>+FILE* FdopenPlatformFile(PlatformFile file, const char* modes) {
>   if (file == kInvalidPlatformFileValue)
>     return nullptr;
>   int fd = _open_osfhandle(reinterpret_cast<intptr_t>(file), 0);
>   if (fd < 0)
>     return nullptr;
> 
>-  return _fdopen(fd, "w");
>+  return _fdopen(fd, modes);
> }
> 
> bool ClosePlatformFile(PlatformFile file) {
>@@ -49,6 +53,11 @@ PlatformFile OpenPlatformFile(const std::string& path) {
>                       nullptr, OPEN_EXISTING, FILE_ATTRIBUTE_NORMAL, nullptr);
> }
> 
>+PlatformFile OpenPlatformFileReadOnly(const std::string& path) {
>+  return ::CreateFile(ToUtf16(path).c_str(), GENERIC_READ, FILE_SHARE_READ,
>+                      nullptr, OPEN_EXISTING, FILE_ATTRIBUTE_NORMAL, nullptr);
>+}
>+
> PlatformFile CreatePlatformFile(const std::string& path) {
>   return ::CreateFile(ToUtf16(path).c_str(), GENERIC_READ | GENERIC_WRITE, 0,
>                       nullptr, CREATE_ALWAYS, FILE_ATTRIBUTE_NORMAL, nullptr);
>@@ -58,12 +67,12 @@ PlatformFile CreatePlatformFile(const std::string& path) {
> 
> const PlatformFile kInvalidPlatformFileValue = -1;
> 
>-FILE* FdopenPlatformFileForWriting(PlatformFile file) {
>-  return fdopen(file, "w");
>+FILE* FdopenPlatformFile(PlatformFile file, const char* modes) {
>+  return fdopen(file, modes);
> }
> 
> bool ClosePlatformFile(PlatformFile file) {
>-  return close(file);
>+  return close(file) == 0;
> }
> 
> bool RemoveFile(const std::string& path) {
>@@ -74,6 +83,10 @@ PlatformFile OpenPlatformFile(const std::string& path) {
>   return ::open(path.c_str(), O_RDWR);
> }
> 
>+PlatformFile OpenPlatformFileReadOnly(const std::string& path) {
>+  return ::open(path.c_str(), O_RDONLY);
>+}
>+
> PlatformFile CreatePlatformFile(const std::string& path) {
>   return ::open(path.c_str(), O_CREAT | O_TRUNC | O_RDWR, S_IRUSR | S_IWUSR);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/platform_file.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/platform_file.h
>index 0e18ff24877..52fbaff1155 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/platform_file.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/platform_file.h
>@@ -35,7 +35,12 @@ extern const PlatformFile kInvalidPlatformFileValue;
> // the PlatformFile should no longer be used.
> FILE* FdopenPlatformFileForWriting(PlatformFile file);
> 
>-// Closes a PlatformFile.
>+// Associates a standard FILE stream with an existing PlatformFile.
>+// Note that after this function has returned a valid FILE stream,
>+// the PlatformFile should no longer be used.
>+FILE* FdopenPlatformFile(PlatformFile file, const char* modes);
>+
>+// Closes a PlatformFile. Returns true on success, false on failure.
> // Don't use ClosePlatformFile to close a file opened with FdopenPlatformFile.
> // Use fclose instead.
> bool ClosePlatformFile(PlatformFile file);
>@@ -47,6 +52,10 @@ bool RemoveFile(const std::string& path);
> // instead.
> PlatformFile OpenPlatformFile(const std::string& path);
> 
>+// Opens a file for reading only. You might want to use base/file.h
>+// instead.
>+PlatformFile OpenPlatformFileReadOnly(const std::string& path);
>+
> // Creates a new file for reading and writing. If the file already exists it
> // will be overwritten. You might want to use base/file.h instead.
> PlatformFile CreatePlatformFile(const std::string& path);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/platform_file_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/platform_file_unittest.cc
>new file mode 100644
>index 00000000000..4e72400da2a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/platform_file_unittest.cc
>@@ -0,0 +1,77 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "rtc_base/platform_file.h"
>+#include "test/gtest.h"
>+#include "test/testsupport/fileutils.h"
>+
>+namespace rtc {
>+
>+void FillWithDummyDataAndClose(FILE* const file, const std::string& filename) {
>+  EXPECT_GT(fprintf(file, "%s", "Dummy data"), 0)
>+      << "Failed to write to file: " << filename;
>+  fclose(file);
>+}
>+
>+TEST(PlatformFileTest, CreateWriteAndDelete) {
>+  const std::string filename = webrtc::test::GenerateTempFilename(
>+      webrtc::test::OutputPath(), ".testfile");
>+  const PlatformFile fd = rtc::CreatePlatformFile(filename);
>+  ASSERT_NE(fd, rtc::kInvalidPlatformFileValue)
>+      << "Failed to create file descriptor for file: " << filename;
>+  FILE* const file = rtc::FdopenPlatformFile(fd, "w");
>+  ASSERT_TRUE(file != nullptr) << "Failed to open file: " << filename;
>+  FillWithDummyDataAndClose(file, filename);
>+  webrtc::test::RemoveFile(filename);
>+}
>+
>+TEST(PlatformFileTest, OpenExistingWriteAndDelete) {
>+  const std::string filename = webrtc::test::GenerateTempFilename(
>+      webrtc::test::OutputPath(), ".testfile");
>+
>+  // Create file with dummy data.
>+  FILE* file = fopen(filename.c_str(), "wb");
>+  ASSERT_TRUE(file != nullptr) << "Failed to open file: " << filename;
>+  FillWithDummyDataAndClose(file, filename);
>+
>+  // Open it for write, write and delete.
>+  const PlatformFile fd = rtc::OpenPlatformFile(filename);
>+  ASSERT_NE(fd, rtc::kInvalidPlatformFileValue)
>+      << "Failed to open file descriptor for file: " << filename;
>+  file = rtc::FdopenPlatformFile(fd, "w");
>+  ASSERT_TRUE(file != nullptr) << "Failed to open file: " << filename;
>+  FillWithDummyDataAndClose(file, filename);
>+  webrtc::test::RemoveFile(filename);
>+}
>+
>+TEST(PlatformFileTest, OpenExistingReadOnlyAndDelete) {
>+  const std::string filename = webrtc::test::GenerateTempFilename(
>+      webrtc::test::OutputPath(), ".testfile");
>+
>+  // Create file with dummy data.
>+  FILE* file = fopen(filename.c_str(), "wb");
>+  ASSERT_TRUE(file != nullptr) << "Failed to open file: " << filename;
>+  FillWithDummyDataAndClose(file, filename);
>+
>+  // Open it for read, read and delete.
>+  const PlatformFile fd = rtc::OpenPlatformFileReadOnly(filename);
>+  ASSERT_NE(fd, rtc::kInvalidPlatformFileValue)
>+      << "Failed to open file descriptor for file: " << filename;
>+  file = rtc::FdopenPlatformFile(fd, "r");
>+  ASSERT_TRUE(file != nullptr) << "Failed to open file: " << filename;
>+
>+  int buf[]{0};
>+  ASSERT_GT(fread(&buf, 1, 1, file), 0u)
>+      << "Failed to read from file: " << filename;
>+  fclose(file);
>+  webrtc::test::RemoveFile(filename);
>+}
>+
>+}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/platform_thread.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/platform_thread.cc
>index d31d77710f4..79d9d53aeee 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/platform_thread.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/platform_thread.cc
>@@ -13,78 +13,13 @@
> #include "rtc_base/atomicops.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/timeutils.h"
>-#include "rtc_base/trace_event.h"
> 
> #if defined(WEBRTC_LINUX)
> #include <sys/prctl.h>
> #include <sys/syscall.h>
> #endif
> 
>-#if defined(WEBRTC_FUCHSIA)
>-#include <zircon/process.h>
>-#endif
>-
> namespace rtc {
>-
>-PlatformThreadId CurrentThreadId() {
>-  PlatformThreadId ret;
>-#if defined(WEBRTC_WIN)
>-  ret = GetCurrentThreadId();
>-#elif defined(WEBRTC_POSIX)
>-#if defined(WEBRTC_MAC) || defined(WEBRTC_IOS)
>-  ret = pthread_mach_thread_np(pthread_self());
>-#elif defined(WEBRTC_ANDROID)
>-  ret = gettid();
>-#elif defined(WEBRTC_FUCHSIA)
>-  ret = zx_thread_self();
>-#elif defined(WEBRTC_LINUX)
>-  ret = syscall(__NR_gettid);
>-#else
>-  // Default implementation for nacl and solaris.
>-  ret = reinterpret_cast<pid_t>(pthread_self());
>-#endif
>-#endif  // defined(WEBRTC_POSIX)
>-  RTC_DCHECK(ret);
>-  return ret;
>-}
>-
>-PlatformThreadRef CurrentThreadRef() {
>-#if defined(WEBRTC_WIN)
>-  return GetCurrentThreadId();
>-#elif defined(WEBRTC_POSIX)
>-  return pthread_self();
>-#endif
>-}
>-
>-bool IsThreadRefEqual(const PlatformThreadRef& a, const PlatformThreadRef& b) {
>-#if defined(WEBRTC_WIN)
>-  return a == b;
>-#elif defined(WEBRTC_POSIX)
>-  return pthread_equal(a, b);
>-#endif
>-}
>-
>-void SetCurrentThreadName(const char* name) {
>-#if defined(WEBRTC_WIN)
>-  struct {
>-    DWORD dwType;
>-    LPCSTR szName;
>-    DWORD dwThreadID;
>-    DWORD dwFlags;
>-  } threadname_info = {0x1000, name, static_cast<DWORD>(-1), 0};
>-
>-  __try {
>-    ::RaiseException(0x406D1388, 0, sizeof(threadname_info) / sizeof(DWORD),
>-                     reinterpret_cast<ULONG_PTR*>(&threadname_info));
>-  } __except (EXCEPTION_EXECUTE_HANDLER) {
>-  }
>-#elif defined(WEBRTC_LINUX) || defined(WEBRTC_ANDROID)
>-  prctl(PR_SET_NAME, reinterpret_cast<unsigned long>(name));
>-#elif defined(WEBRTC_MAC) || defined(WEBRTC_IOS)
>-  pthread_setname_np(name);
>-#endif
>-}
>-
> namespace {
> #if defined(WEBRTC_WIN)
> void CALLBACK RaiseFlag(ULONG_PTR param) {
>@@ -268,7 +203,7 @@ void PlatformThread::Run() {
>     SleepEx(0, true);
>   } while (!stop_);
> #else
>-#if defined(WEBRTC_MAC)
>+#if defined(WEBRTC_MAC) || defined(WEBRTC_ANDROID)
>     sched_yield();
> #else
>     static const struct timespec ts_null = {0};
>@@ -295,8 +230,8 @@ bool PlatformThread::SetPriority(ThreadPriority priority) {
> 
> #if defined(WEBRTC_WIN)
>   return SetThreadPriority(thread_, priority) != FALSE;
>-#elif defined(__native_client__)
>-  // Setting thread priorities is not supported in NaCl.
>+#elif defined(__native_client__) || defined(WEBRTC_FUCHSIA)
>+  // Setting thread priorities is not supported in NaCl or Fuchsia.
>   return true;
> #elif defined(WEBRTC_CHROMIUM_BUILD) && defined(WEBRTC_LINUX)
>   // TODO(tommi): Switch to the same mechanism as Chromium uses for changing
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/platform_thread.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/platform_thread.h
>index 25fa0369085..33921c209da 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/platform_thread.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/platform_thread.h
>@@ -20,15 +20,6 @@
> 
> namespace rtc {
> 
>-PlatformThreadId CurrentThreadId();
>-PlatformThreadRef CurrentThreadRef();
>-
>-// Compares two thread identifiers for equality.
>-bool IsThreadRefEqual(const PlatformThreadRef& a, const PlatformThreadRef& b);
>-
>-// Sets the current thread name.
>-void SetCurrentThreadName(const char* name);
>-
> // Callback function that the spawned thread will enter once spawned.
> // A return value of false is interpreted as that the function has no
> // more work to do and that the thread can be released.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/platform_thread_types.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/platform_thread_types.cc
>new file mode 100644
>index 00000000000..15d48618bd3
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/platform_thread_types.cc
>@@ -0,0 +1,78 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "rtc_base/platform_thread_types.h"
>+
>+#if defined(WEBRTC_LINUX)
>+#include <sys/prctl.h>
>+#include <sys/syscall.h>
>+#endif
>+
>+namespace rtc {
>+
>+PlatformThreadId CurrentThreadId() {
>+#if defined(WEBRTC_WIN)
>+  return GetCurrentThreadId();
>+#elif defined(WEBRTC_POSIX)
>+#if defined(WEBRTC_MAC) || defined(WEBRTC_IOS)
>+  return pthread_mach_thread_np(pthread_self());
>+#elif defined(WEBRTC_ANDROID)
>+  return gettid();
>+#elif defined(WEBRTC_FUCHSIA)
>+  return zx_thread_self();
>+#elif defined(WEBRTC_LINUX)
>+  return syscall(__NR_gettid);
>+#else
>+  // Default implementation for nacl and solaris.
>+  return reinterpret_cast<pid_t>(pthread_self());
>+#endif
>+#endif  // defined(WEBRTC_POSIX)
>+}
>+
>+PlatformThreadRef CurrentThreadRef() {
>+#if defined(WEBRTC_WIN)
>+  return GetCurrentThreadId();
>+#elif defined(WEBRTC_FUCHSIA)
>+  return zx_thread_self();
>+#elif defined(WEBRTC_POSIX)
>+  return pthread_self();
>+#endif
>+}
>+
>+bool IsThreadRefEqual(const PlatformThreadRef& a, const PlatformThreadRef& b) {
>+#if defined(WEBRTC_WIN) || defined(WEBRTC_FUCHSIA)
>+  return a == b;
>+#elif defined(WEBRTC_POSIX)
>+  return pthread_equal(a, b);
>+#endif
>+}
>+
>+void SetCurrentThreadName(const char* name) {
>+#if defined(WEBRTC_WIN)
>+  struct {
>+    DWORD dwType;
>+    LPCSTR szName;
>+    DWORD dwThreadID;
>+    DWORD dwFlags;
>+  } threadname_info = {0x1000, name, static_cast<DWORD>(-1), 0};
>+
>+  __try {
>+    ::RaiseException(0x406D1388, 0, sizeof(threadname_info) / sizeof(DWORD),
>+                     reinterpret_cast<ULONG_PTR*>(&threadname_info));
>+  } __except (EXCEPTION_EXECUTE_HANDLER) {  // NOLINT
>+  }
>+#elif defined(WEBRTC_LINUX) || defined(WEBRTC_ANDROID)
>+  prctl(PR_SET_NAME, reinterpret_cast<unsigned long>(name));  // NOLINT
>+#elif defined(WEBRTC_MAC) || defined(WEBRTC_IOS)
>+  pthread_setname_np(name);
>+#endif
>+}
>+
>+}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/platform_thread_types.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/platform_thread_types.h
>index 10c5de83970..0bc42eb2c2f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/platform_thread_types.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/platform_thread_types.h
>@@ -11,15 +11,22 @@
> #ifndef RTC_BASE_PLATFORM_THREAD_TYPES_H_
> #define RTC_BASE_PLATFORM_THREAD_TYPES_H_
> 
>+// clang-format off
>+// clang formating would change include order.
> #if defined(WEBRTC_WIN)
>+// Include winsock2.h before including <windows.h> to maintain consistency with
>+// win32.h. To include win32.h directly, it must be broken out into its own
>+// build target.
> #include <winsock2.h>
> #include <windows.h>
> #elif defined(WEBRTC_FUCHSIA)
> #include <zircon/types.h>
>+#include <zircon/process.h>
> #elif defined(WEBRTC_POSIX)
> #include <pthread.h>
> #include <unistd.h>
> #endif
>+// clang-format on
> 
> namespace rtc {
> #if defined(WEBRTC_WIN)
>@@ -27,11 +34,26 @@ typedef DWORD PlatformThreadId;
> typedef DWORD PlatformThreadRef;
> #elif defined(WEBRTC_FUCHSIA)
> typedef zx_handle_t PlatformThreadId;
>-typedef pthread_t PlatformThreadRef;
>+typedef zx_handle_t PlatformThreadRef;
> #elif defined(WEBRTC_POSIX)
> typedef pid_t PlatformThreadId;
> typedef pthread_t PlatformThreadRef;
> #endif
>+
>+// Retrieve the ID of the current thread.
>+PlatformThreadId CurrentThreadId();
>+
>+// Retrieves a reference to the current thread. On Windows, this is the same
>+// as CurrentThreadId. On other platforms it's the pthread_t returned by
>+// pthread_self().
>+PlatformThreadRef CurrentThreadRef();
>+
>+// Compares two thread identifiers for equality.
>+bool IsThreadRefEqual(const PlatformThreadRef& a, const PlatformThreadRef& b);
>+
>+// Sets the current thread name.
>+void SetCurrentThreadName(const char* name);
>+
> }  // namespace rtc
> 
> #endif  // RTC_BASE_PLATFORM_THREAD_TYPES_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/platform_thread_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/platform_thread_unittest.cc
>index d8c8995507c..1661b6a999a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/platform_thread_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/platform_thread_unittest.cc
>@@ -125,4 +125,4 @@ TEST(PlatformThreadTest, DISABLED_TooBusyDeprecated) {
>   thread.Stop();
> }
> 
>-}  // rtc
>+}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/proxy_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/proxy_unittest.cc
>index 7d7b6f89131..d81b3125cb4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/proxy_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/proxy_unittest.cc
>@@ -32,8 +32,8 @@ static const SocketAddress kBogusProxyIntAddr("1.2.3.4", 999);
> class ProxyTest : public testing::Test {
>  public:
>   ProxyTest() : ss_(new rtc::VirtualSocketServer()), thread_(ss_.get()) {
>-    socks_.reset(new rtc::SocksProxyServer(
>-        ss_.get(), kSocksProxyIntAddr, ss_.get(), kSocksProxyExtAddr));
>+    socks_.reset(new rtc::SocksProxyServer(ss_.get(), kSocksProxyIntAddr,
>+                                           ss_.get(), kSocksProxyExtAddr));
>     https_.reset(new rtc::HttpListenServer());
>     https_->Listen(kHttpsProxyIntAddr);
>   }
>@@ -52,13 +52,11 @@ class ProxyTest : public testing::Test {
> TEST_F(ProxyTest, TestSocks5Connect) {
>   rtc::AsyncSocket* socket =
>       ss()->CreateAsyncSocket(kSocksProxyIntAddr.family(), SOCK_STREAM);
>-  rtc::AsyncSocksProxySocket* proxy_socket =
>-      new rtc::AsyncSocksProxySocket(socket, kSocksProxyIntAddr,
>-                                           "", rtc::CryptString());
>+  rtc::AsyncSocksProxySocket* proxy_socket = new rtc::AsyncSocksProxySocket(
>+      socket, kSocksProxyIntAddr, "", rtc::CryptString());
>   // TODO: IPv6-ize these tests when proxy supports IPv6.
> 
>-  rtc::TestEchoServer server(Thread::Current(),
>-                                   SocketAddress(INADDR_ANY, 0));
>+  rtc::TestEchoServer server(Thread::Current(), SocketAddress(INADDR_ANY, 0));
> 
>   std::unique_ptr<rtc::AsyncTCPSocket> packet_socket(
>       rtc::AsyncTCPSocket::Create(proxy_socket, SocketAddress(INADDR_ANY, 0),
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/proxyinfo.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/proxyinfo.cc
>index a165dcad5e9..c394487c1a6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/proxyinfo.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/proxyinfo.cc
>@@ -12,13 +12,12 @@
> 
> namespace rtc {
> 
>-const char * ProxyToString(ProxyType proxy) {
>-  const char * const PROXY_NAMES[] = { "none", "https", "socks5", "unknown" };
>+const char* ProxyToString(ProxyType proxy) {
>+  const char* const PROXY_NAMES[] = {"none", "https", "socks5", "unknown"};
>   return PROXY_NAMES[proxy];
> }
> 
>-ProxyInfo::ProxyInfo() : type(PROXY_NONE), autodetect(false) {
>-}
>+ProxyInfo::ProxyInfo() : type(PROXY_NONE), autodetect(false) {}
> ProxyInfo::~ProxyInfo() = default;
> 
>-} // namespace rtc
>+}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/proxyinfo.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/proxyinfo.h
>index 5affcd86c38..19476696a24 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/proxyinfo.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/proxyinfo.h
>@@ -17,13 +17,8 @@
> 
> namespace rtc {
> 
>-enum ProxyType {
>-  PROXY_NONE,
>-  PROXY_HTTPS,
>-  PROXY_SOCKS5,
>-  PROXY_UNKNOWN
>-};
>-const char * ProxyToString(ProxyType proxy);
>+enum ProxyType { PROXY_NONE, PROXY_HTTPS, PROXY_SOCKS5, PROXY_UNKNOWN };
>+const char* ProxyToString(ProxyType proxy);
> 
> struct ProxyInfo {
>   ProxyType type;
>@@ -38,6 +33,6 @@ struct ProxyInfo {
>   ~ProxyInfo();
> };
> 
>-} // namespace rtc
>+}  // namespace rtc
> 
>-#endif // RTC_BASE_PROXYINFO_H_
>+#endif  // RTC_BASE_PROXYINFO_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/proxyserver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/proxyserver.cc
>index 5ab7943c2e5..55cab80975c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/proxyserver.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/proxyserver.cc
>@@ -18,12 +18,14 @@
> namespace rtc {
> 
> // ProxyServer
>-ProxyServer::ProxyServer(
>-    SocketFactory* int_factory, const SocketAddress& int_addr,
>-    SocketFactory* ext_factory, const SocketAddress& ext_ip)
>-    : ext_factory_(ext_factory), ext_ip_(ext_ip.ipaddr(), 0),  // strip off port
>-      server_socket_(int_factory->CreateAsyncSocket(int_addr.family(),
>-                                                    SOCK_STREAM)) {
>+ProxyServer::ProxyServer(SocketFactory* int_factory,
>+                         const SocketAddress& int_addr,
>+                         SocketFactory* ext_factory,
>+                         const SocketAddress& ext_ip)
>+    : ext_factory_(ext_factory),
>+      ext_ip_(ext_ip.ipaddr(), 0),  // strip off port
>+      server_socket_(
>+          int_factory->CreateAsyncSocket(int_addr.family(), SOCK_STREAM)) {
>   RTC_DCHECK(server_socket_.get() != nullptr);
>   RTC_DCHECK(int_addr.family() == AF_INET || int_addr.family() == AF_INET6);
>   server_socket_->Bind(int_addr);
>@@ -32,8 +34,8 @@ ProxyServer::ProxyServer(
> }
> 
> ProxyServer::~ProxyServer() {
>-  for (BindingList::iterator it = bindings_.begin();
>-       it != bindings_.end(); ++it) {
>+  for (BindingList::iterator it = bindings_.begin(); it != bindings_.end();
>+       ++it) {
>     delete (*it);
>   }
> }
>@@ -47,8 +49,8 @@ void ProxyServer::OnAcceptEvent(AsyncSocket* socket) {
>   RTC_DCHECK_EQ(socket, server_socket_.get());
>   AsyncSocket* int_socket = socket->Accept(nullptr);
>   AsyncProxyServerSocket* wrapped_socket = WrapSocket(int_socket);
>-  AsyncSocket* ext_socket = ext_factory_->CreateAsyncSocket(ext_ip_.family(),
>-                                                            SOCK_STREAM);
>+  AsyncSocket* ext_socket =
>+      ext_factory_->CreateAsyncSocket(ext_ip_.family(), SOCK_STREAM);
>   if (ext_socket) {
>     ext_socket->Bind(ext_ip_);
>     bindings_.push_back(new ProxyBinding(wrapped_socket, ext_socket));
>@@ -68,8 +70,11 @@ void ProxyServer::OnBindingDestroyed(ProxyBinding* binding) {
> // ProxyBinding
> ProxyBinding::ProxyBinding(AsyncProxyServerSocket* int_socket,
>                            AsyncSocket* ext_socket)
>-    : int_socket_(int_socket), ext_socket_(ext_socket), connected_(false),
>-      out_buffer_(kBufferSize), in_buffer_(kBufferSize) {
>+    : int_socket_(int_socket),
>+      ext_socket_(ext_socket),
>+      connected_(false),
>+      out_buffer_(kBufferSize),
>+      in_buffer_(kBufferSize) {
>   int_socket_->SignalConnectRequest.connect(this,
>                                             &ProxyBinding::OnConnectRequest);
>   int_socket_->SignalReadEvent.connect(this, &ProxyBinding::OnInternalRead);
>@@ -85,7 +90,7 @@ ProxyBinding::ProxyBinding(AsyncProxyServerSocket* int_socket,
> ProxyBinding::~ProxyBinding() = default;
> 
> void ProxyBinding::OnConnectRequest(AsyncProxyServerSocket* socket,
>-                                   const SocketAddress& addr) {
>+                                    const SocketAddress& addr) {
>   RTC_DCHECK(!connected_);
>   RTC_DCHECK(ext_socket_);
>   ext_socket_->Connect(addr);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/proxyserver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/proxyserver.h
>index d90b09171c4..37cadd06497 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/proxyserver.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/proxyserver.h
>@@ -62,8 +62,10 @@ class ProxyBinding : public sigslot::has_slots<> {
> 
> class ProxyServer : public sigslot::has_slots<> {
>  public:
>-  ProxyServer(SocketFactory* int_factory, const SocketAddress& int_addr,
>-              SocketFactory* ext_factory, const SocketAddress& ext_ip);
>+  ProxyServer(SocketFactory* int_factory,
>+              const SocketAddress& int_addr,
>+              SocketFactory* ext_factory,
>+              const SocketAddress& ext_ip);
>   ~ProxyServer() override;
> 
>   // Returns the address to which the proxy server is bound
>@@ -86,10 +88,12 @@ class ProxyServer : public sigslot::has_slots<> {
> // SocksProxyServer is a simple extension of ProxyServer to implement SOCKS.
> class SocksProxyServer : public ProxyServer {
>  public:
>-  SocksProxyServer(SocketFactory* int_factory, const SocketAddress& int_addr,
>-                   SocketFactory* ext_factory, const SocketAddress& ext_ip)
>-      : ProxyServer(int_factory, int_addr, ext_factory, ext_ip) {
>-  }
>+  SocksProxyServer(SocketFactory* int_factory,
>+                   const SocketAddress& int_addr,
>+                   SocketFactory* ext_factory,
>+                   const SocketAddress& ext_ip)
>+      : ProxyServer(int_factory, int_addr, ext_factory, ext_ip) {}
>+
>  protected:
>   AsyncProxyServerSocket* WrapSocket(AsyncSocket* socket) override;
>   RTC_DISALLOW_COPY_AND_ASSIGN(SocksProxyServer);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ptr_util.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ptr_util.h
>deleted file mode 100644
>index 156df841b5e..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ptr_util.h
>+++ /dev/null
>@@ -1,82 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC Project Authors. All rights reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-// This implementation is borrowed from chromium.
>-
>-#ifndef RTC_BASE_PTR_UTIL_H_
>-#define RTC_BASE_PTR_UTIL_H_
>-
>-#include <memory>
>-#include <utility>
>-
>-namespace rtc {
>-
>-// Helper to transfer ownership of a raw pointer to a std::unique_ptr<T>.
>-// Note that std::unique_ptr<T> has very different semantics from
>-// std::unique_ptr<T[]>: do not use this helper for array allocations.
>-template <typename T>
>-std::unique_ptr<T> WrapUnique(T* ptr) {
>-  return std::unique_ptr<T>(ptr);
>-}
>-
>-namespace internal {
>-
>-template <typename T>
>-struct MakeUniqueResult {
>-  using Scalar = std::unique_ptr<T>;
>-};
>-
>-template <typename T>
>-struct MakeUniqueResult<T[]> {
>-  using Array = std::unique_ptr<T[]>;
>-};
>-
>-template <typename T, size_t N>
>-struct MakeUniqueResult<T[N]> {
>-  using Invalid = void;
>-};
>-
>-}  // namespace internal
>-
>-// Helper to construct an object wrapped in a std::unique_ptr. This is an
>-// implementation of C++14's std::make_unique that can be used in Chrome.
>-//
>-// MakeUnique<T>(args) should be preferred over WrapUnique(new T(args)): bare
>-// calls to `new` should be treated with scrutiny.
>-//
>-// Usage:
>-//   // ptr is a std::unique_ptr<std::string>
>-//   auto ptr = MakeUnique<std::string>("hello world!");
>-//
>-//   // arr is a std::unique_ptr<int[]>
>-//   auto arr = MakeUnique<int[]>(5);
>-
>-// Overload for non-array types. Arguments are forwarded to T's constructor.
>-template <typename T, typename... Args>
>-typename internal::MakeUniqueResult<T>::Scalar MakeUnique(Args&&... args) {
>-  return std::unique_ptr<T>(new T(std::forward<Args>(args)...));
>-}
>-
>-// Overload for array types of unknown bound, e.g. T[]. The array is allocated
>-// with `new T[n]()` and value-initialized: note that this is distinct from
>-// `new T[n]`, which default-initializes.
>-template <typename T>
>-typename internal::MakeUniqueResult<T>::Array MakeUnique(size_t size) {
>-  return std::unique_ptr<T>(new typename std::remove_extent<T>::type[size]());
>-}
>-
>-// Overload to reject array types of known bound, e.g. T[n].
>-template <typename T, typename... Args>
>-typename internal::MakeUniqueResult<T>::Invalid MakeUnique(Args&&... args) =
>-    delete;
>-
>-}  // namespace rtc
>-
>-#endif  // RTC_BASE_PTR_UTIL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ptr_util_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ptr_util_unittest.cc
>deleted file mode 100644
>index 6497fda525c..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ptr_util_unittest.cc
>+++ /dev/null
>@@ -1,69 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC Project Authors. All rights reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "rtc_base/ptr_util.h"
>-
>-#include <stddef.h>
>-#include <string>
>-
>-#include "rtc_base/gunit.h"
>-
>-namespace rtc {
>-
>-namespace {
>-
>-class DeleteCounter {
>- public:
>-  DeleteCounter() { ++count_; }
>-  ~DeleteCounter() { --count_; }
>-
>-  static size_t count() { return count_; }
>-
>- private:
>-  static size_t count_;
>-};
>-
>-size_t DeleteCounter::count_ = 0;
>-
>-}  // namespace
>-
>-TEST(PtrUtilTest, WrapUnique) {
>-  EXPECT_EQ(0u, DeleteCounter::count());
>-  DeleteCounter* counter = new DeleteCounter;
>-  EXPECT_EQ(1u, DeleteCounter::count());
>-  std::unique_ptr<DeleteCounter> owned_counter = WrapUnique(counter);
>-  EXPECT_EQ(1u, DeleteCounter::count());
>-  owned_counter.reset();
>-  EXPECT_EQ(0u, DeleteCounter::count());
>-}
>-
>-TEST(PtrUtilTest, MakeUniqueScalar) {
>-  auto s = MakeUnique<std::string>();
>-  EXPECT_EQ("", *s);
>-
>-  auto s2 = MakeUnique<std::string>("test");
>-  EXPECT_EQ("test", *s2);
>-}
>-
>-TEST(PtrUtilTest, MakeUniqueScalarWithMoveOnlyType) {
>-  using MoveOnly = std::unique_ptr<std::string>;
>-  auto p = MakeUnique<MoveOnly>(MakeUnique<std::string>("test"));
>-  EXPECT_EQ("test", **p);
>-}
>-
>-TEST(PtrUtilTest, MakeUniqueArray) {
>-  EXPECT_EQ(0u, DeleteCounter::count());
>-  auto a = MakeUnique<DeleteCounter[]>(5);
>-  EXPECT_EQ(5u, DeleteCounter::count());
>-  a.reset();
>-  EXPECT_EQ(0u, DeleteCounter::count());
>-}
>-
>-}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/random.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/random.h
>index 7c103ccfa75..2faa9856dc6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/random.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/random.h
>@@ -15,7 +15,6 @@
> 
> #include "rtc_base/checks.h"
> #include "rtc_base/constructormagic.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rate_limiter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rate_limiter.cc
>index 34e0a4064d0..0343f256b29 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rate_limiter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rate_limiter.cc
>@@ -29,7 +29,7 @@ RateLimiter::~RateLimiter() {}
> bool RateLimiter::TryUseRate(size_t packet_size_bytes) {
>   rtc::CritScope cs(&lock_);
>   int64_t now_ms = clock_->TimeInMilliseconds();
>-  rtc::Optional<uint32_t> current_rate = current_rate_.Rate(now_ms);
>+  absl::optional<uint32_t> current_rate = current_rate_.Rate(now_ms);
>   if (current_rate) {
>     // If there is a current rate, check if adding bytes would cause maximum
>     // bitrate target to be exceeded. If there is NOT a valid current rate,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rate_statistics.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rate_statistics.cc
>index e2d6b113df6..9b1ff8fc9a8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rate_statistics.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rate_statistics.cc
>@@ -61,7 +61,7 @@ void RateStatistics::Update(size_t count, int64_t now_ms) {
>   ++num_samples_;
> }
> 
>-rtc::Optional<uint32_t> RateStatistics::Rate(int64_t now_ms) const {
>+absl::optional<uint32_t> RateStatistics::Rate(int64_t now_ms) const {
>   // Yeah, this const_cast ain't pretty, but the alternative is to declare most
>   // of the members as mutable...
>   const_cast<RateStatistics*>(this)->EraseOld(now_ms);
>@@ -71,7 +71,7 @@ rtc::Optional<uint32_t> RateStatistics::Rate(int64_t now_ms) const {
>   int64_t active_window_size = now_ms - oldest_time_ + 1;
>   if (num_samples_ == 0 || active_window_size <= 1 ||
>       (num_samples_ <= 1 && active_window_size < current_window_size_ms_)) {
>-    return rtc::nullopt;
>+    return absl::nullopt;
>   }
> 
>   float scale = scale_ / active_window_size;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rate_statistics.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rate_statistics.h
>index aa285298a62..89bfddfa552 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rate_statistics.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rate_statistics.h
>@@ -13,8 +13,7 @@
> 
> #include <memory>
> 
>-#include "api/optional.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include "absl/types/optional.h"
> 
> namespace webrtc {
> 
>@@ -42,7 +41,7 @@ class RateStatistics {
>   // from a monotonic clock. Ie, it doesn't matter if this call moves the
>   // window, since any subsequent call to Update or Rate would still have moved
>   // the window as much or more.
>-  rtc::Optional<uint32_t> Rate(int64_t now_ms) const;
>+  absl::optional<uint32_t> Rate(int64_t now_ms) const;
> 
>   // Update the size of the averaging window. The maximum allowed value for
>   // window_size_ms is max_window_size_ms as supplied in the constructor.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rate_statistics_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rate_statistics_unittest.cc
>index 51249b14eae..62f1486d508 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rate_statistics_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rate_statistics_unittest.cc
>@@ -53,7 +53,7 @@ TEST_F(RateStatisticsTest, TestStrictMode) {
>     // Approximately 1200 kbps expected. Not exact since when packets
>     // are removed we will jump 10 ms to the next packet.
>     if (i > kInterval) {
>-      rtc::Optional<uint32_t> rate = stats_.Rate(now_ms);
>+      absl::optional<uint32_t> rate = stats_.Rate(now_ms);
>       EXPECT_TRUE(static_cast<bool>(rate));
>       uint32_t samples = i / kInterval + 1;
>       uint64_t total_bits = samples * kPacketSize * 8;
>@@ -78,7 +78,7 @@ TEST_F(RateStatisticsTest, IncreasingThenDecreasingBitrate) {
>   const uint32_t kExpectedBitrate = 8000000;
>   // 1000 bytes per millisecond until plateau is reached.
>   int prev_error = kExpectedBitrate;
>-  rtc::Optional<uint32_t> bitrate;
>+  absl::optional<uint32_t> bitrate;
>   while (++now_ms < 10000) {
>     stats_.Update(1000, now_ms);
>     bitrate = stats_.Rate(now_ms);
>@@ -102,7 +102,7 @@ TEST_F(RateStatisticsTest, IncreasingThenDecreasingBitrate) {
>   // Zero bytes per millisecond until 0 is reached.
>   while (++now_ms < 20000) {
>     stats_.Update(0, now_ms);
>-    rtc::Optional<uint32_t> new_bitrate = stats_.Rate(now_ms);
>+    absl::optional<uint32_t> new_bitrate = stats_.Rate(now_ms);
>     if (static_cast<bool>(new_bitrate) && *new_bitrate != *bitrate) {
>       // New bitrate must be lower than previous one.
>       EXPECT_LT(*new_bitrate, *bitrate);
>@@ -130,7 +130,7 @@ TEST_F(RateStatisticsTest, ResetAfterSilence) {
>   const uint32_t kExpectedBitrate = 8000000;
>   // 1000 bytes per millisecond until the window has been filled.
>   int prev_error = kExpectedBitrate;
>-  rtc::Optional<uint32_t> bitrate;
>+  absl::optional<uint32_t> bitrate;
>   while (++now_ms < 10000) {
>     stats_.Update(1000, now_ms);
>     bitrate = stats_.Rate(now_ms);
>@@ -213,7 +213,7 @@ TEST_F(RateStatisticsTest, RespectsWindowSizeEdges) {
> 
>   // Window size should be full, and the single data point should be accepted.
>   ++now_ms;
>-  rtc::Optional<uint32_t> bitrate = stats_.Rate(now_ms);
>+  absl::optional<uint32_t> bitrate = stats_.Rate(now_ms);
>   EXPECT_TRUE(static_cast<bool>(bitrate));
>   EXPECT_EQ(1000 * 8u, *bitrate);
> 
>@@ -239,7 +239,7 @@ TEST_F(RateStatisticsTest, HandlesZeroCounts) {
>   stats_.Update(kWindowMs, now_ms);
>   now_ms += kWindowMs - 1;
>   stats_.Update(0, now_ms);
>-  rtc::Optional<uint32_t> bitrate = stats_.Rate(now_ms);
>+  absl::optional<uint32_t> bitrate = stats_.Rate(now_ms);
>   EXPECT_TRUE(static_cast<bool>(bitrate));
>   EXPECT_EQ(1000 * 8u, *bitrate);
> 
>@@ -262,7 +262,7 @@ TEST_F(RateStatisticsTest, HandlesQuietPeriods) {
> 
>   stats_.Update(0, now_ms);
>   now_ms += kWindowMs - 1;
>-  rtc::Optional<uint32_t> bitrate = stats_.Rate(now_ms);
>+  absl::optional<uint32_t> bitrate = stats_.Rate(now_ms);
>   EXPECT_TRUE(static_cast<bool>(bitrate));
>   EXPECT_EQ(0u, *bitrate);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ratetracker.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ratetracker.cc
>index 68bd8a1518c..e31d26627ab 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ratetracker.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ratetracker.cc
>@@ -79,13 +79,12 @@ double RateTracker::ComputeRateForInterval(
>   // Only count a portion of the first bucket according to how much of the
>   // first bucket is within the current interval.
>   size_t total_samples = ((sample_buckets_[start_bucket] *
>-      (bucket_milliseconds_ - milliseconds_to_skip)) +
>-      (bucket_milliseconds_ >> 1)) /
>-      bucket_milliseconds_;
>+                           (bucket_milliseconds_ - milliseconds_to_skip)) +
>+                          (bucket_milliseconds_ >> 1)) /
>+                         bucket_milliseconds_;
>   // All other buckets in the interval are counted in their entirety.
>   for (size_t i = NextBucketIndex(start_bucket);
>-      i != NextBucketIndex(current_bucket_);
>-      i = NextBucketIndex(i)) {
>+       i != NextBucketIndex(current_bucket_); i = NextBucketIndex(i)) {
>     total_samples += sample_buckets_[i];
>   }
>   // Convert to samples per second.
>@@ -125,7 +124,8 @@ void RateTracker::AddSamples(size_t sample_count) {
>   }
>   // Ensure that bucket_start_time_milliseconds_ is updated appropriately if
>   // the entire buffer of samples has been expired.
>-  bucket_start_time_milliseconds_ += bucket_milliseconds_ *
>+  bucket_start_time_milliseconds_ +=
>+      bucket_milliseconds_ *
>       ((current_time - bucket_start_time_milliseconds_) / bucket_milliseconds_);
>   // Add all samples in the bucket that includes the current time.
>   sample_buckets_[current_bucket_] += sample_count;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ratetracker.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ratetracker.h
>index 05eccffce08..ac9b88a5d1a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ratetracker.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ratetracker.h
>@@ -11,8 +11,8 @@
> #ifndef RTC_BASE_RATETRACKER_H_
> #define RTC_BASE_RATETRACKER_H_
> 
>+#include <stdint.h>
> #include <stdlib.h>
>-#include "rtc_base/basictypes.h"
> 
> namespace rtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ratetracker_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ratetracker_unittest.cc
>index cd288a336a5..58102dfaae3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ratetracker_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ratetracker_unittest.cc
>@@ -13,7 +13,7 @@
> 
> namespace rtc {
> namespace {
>-  const uint32_t kBucketIntervalMs = 100;
>+const uint32_t kBucketIntervalMs = 100;
> }  // namespace
> 
> class RateTrackerForTest : public RateTracker {
>@@ -107,7 +107,7 @@ TEST(RateTrackerTest, TestRateTrackerBasics) {
>   EXPECT_DOUBLE_EQ(9876.0 * 10.0, tracker.ComputeRate());
>   EXPECT_EQ(1234U * 2 + 9876U * 55, tracker.TotalSampleCount());
>   EXPECT_DOUBLE_EQ((1234.0 * 2.0 + 9876.0 * 55.0) / 7.5,
>-      tracker.ComputeTotalRate());
>+                   tracker.ComputeTotalRate());
> 
>   // Advance the clock by 500 ms. Since we sent nothing over this half-second,
>   // the reported rate should be reduced by half.
>@@ -116,7 +116,7 @@ TEST(RateTrackerTest, TestRateTrackerBasics) {
>   EXPECT_DOUBLE_EQ(9876.0 * 5.0, tracker.ComputeRate());
>   EXPECT_EQ(1234U * 2 + 9876U * 55, tracker.TotalSampleCount());
>   EXPECT_DOUBLE_EQ((1234.0 * 2.0 + 9876.0 * 55.0) / 8.0,
>-      tracker.ComputeTotalRate());
>+                   tracker.ComputeTotalRate());
> 
>   // Rate over the last half second should be zero.
>   EXPECT_DOUBLE_EQ(0.0, tracker.ComputeRateForInterval(500));
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/refcounter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/refcounter.h
>index baf72ae2a01..3906529c25a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/refcounter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/refcounter.h
>@@ -28,8 +28,8 @@ class RefCounter {
>   // the reference counter can be deleted.
>   rtc::RefCountReleaseStatus DecRef() {
>     return (rtc::AtomicOps::Decrement(&ref_count_) == 0)
>-        ? rtc::RefCountReleaseStatus::kDroppedLastRef
>-        : rtc::RefCountReleaseStatus::kOtherRefsRemained;
>+               ? rtc::RefCountReleaseStatus::kDroppedLastRef
>+               : rtc::RefCountReleaseStatus::kOtherRefsRemained;
>   }
> 
>   // Return whether the reference count is one. If the reference count is used
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rollingaccumulator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rollingaccumulator.h
>index e7d5b06226c..f6215c5407a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rollingaccumulator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rollingaccumulator.h
>@@ -23,23 +23,17 @@ namespace rtc {
> // over N most recent samples.
> //
> // T is assumed to be an int, long, double or float.
>-template<typename T>
>+template <typename T>
> class RollingAccumulator {
>  public:
>-  explicit RollingAccumulator(size_t max_count)
>-    : samples_(max_count) {
>+  explicit RollingAccumulator(size_t max_count) : samples_(max_count) {
>     Reset();
>   }
>-  ~RollingAccumulator() {
>-  }
>+  ~RollingAccumulator() {}
> 
>-  size_t max_count() const {
>-    return samples_.size();
>-  }
>+  size_t max_count() const { return samples_.size(); }
> 
>-  size_t count() const {
>-    return count_;
>-  }
>+  size_t count() const { return count_; }
> 
>   void Reset() {
>     count_ = 0U;
>@@ -84,9 +78,7 @@ class RollingAccumulator {
>     next_index_ = (next_index_ + 1) % max_count();
>   }
> 
>-  T ComputeSum() const {
>-    return static_cast<T>(sum_);
>-  }
>+  T ComputeSum() const { return static_cast<T>(sum_); }
> 
>   double ComputeMean() const {
>     if (count_ == 0) {
>@@ -97,8 +89,8 @@ class RollingAccumulator {
> 
>   T ComputeMax() const {
>     if (max_stale_) {
>-      RTC_DCHECK(count_ > 0) <<
>-                 "It shouldn't be possible for max_stale_ && count_ == 0";
>+      RTC_DCHECK(count_ > 0)
>+          << "It shouldn't be possible for max_stale_ && count_ == 0";
>       max_ = samples_[next_index_];
>       for (size_t i = 1u; i < count_; i++) {
>         max_ = std::max(max_, samples_[(next_index_ + i) % max_count()]);
>@@ -110,8 +102,8 @@ class RollingAccumulator {
> 
>   T ComputeMin() const {
>     if (min_stale_) {
>-      RTC_DCHECK(count_ > 0) <<
>-                 "It shouldn't be possible for min_stale_ && count_ == 0";
>+      RTC_DCHECK(count_ > 0)
>+          << "It shouldn't be possible for min_stale_ && count_ == 0";
>       min_ = samples_[next_index_];
>       for (size_t i = 1u; i < count_; i++) {
>         min_ = std::min(min_, samples_[(next_index_ + i) % max_count()]);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rtccertificate.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rtccertificate.cc
>index dd6f40a6afc..7f027ba5f80 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rtccertificate.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rtccertificate.cc
>@@ -22,13 +22,11 @@ scoped_refptr<RTCCertificate> RTCCertificate::Create(
>   return new RefCountedObject<RTCCertificate>(identity.release());
> }
> 
>-RTCCertificate::RTCCertificate(SSLIdentity* identity)
>-    : identity_(identity) {
>+RTCCertificate::RTCCertificate(SSLIdentity* identity) : identity_(identity) {
>   RTC_DCHECK(identity_);
> }
> 
>-RTCCertificate::~RTCCertificate() {
>-}
>+RTCCertificate::~RTCCertificate() {}
> 
> uint64_t RTCCertificate::Expires() const {
>   int64_t expires = ssl_certificate().CertificateExpirationTime();
>@@ -46,6 +44,10 @@ const SSLCertificate& RTCCertificate::ssl_certificate() const {
>   return identity_->certificate();
> }
> 
>+const SSLCertChain& RTCCertificate::ssl_cert_chain() const {
>+  return identity_->cert_chain();
>+}
>+
> RTCCertificatePEM RTCCertificate::ToPEM() const {
>   return RTCCertificatePEM(identity_->PrivateKeyToPEMString(),
>                            ssl_certificate().ToPEMString());
>@@ -53,8 +55,8 @@ RTCCertificatePEM RTCCertificate::ToPEM() const {
> 
> scoped_refptr<RTCCertificate> RTCCertificate::FromPEM(
>     const RTCCertificatePEM& pem) {
>-  std::unique_ptr<SSLIdentity> identity(SSLIdentity::FromPEMStrings(
>-      pem.private_key(), pem.certificate()));
>+  std::unique_ptr<SSLIdentity> identity(
>+      SSLIdentity::FromPEMStrings(pem.private_key(), pem.certificate()));
>   if (!identity)
>     return nullptr;
>   return new RefCountedObject<RTCCertificate>(identity.release());
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rtccertificate.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rtccertificate.h
>index 47f0e0c9984..d5422f8b4a4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rtccertificate.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rtccertificate.h
>@@ -29,11 +29,9 @@ namespace rtc {
> // the string representations used by OpenSSL.
> class RTCCertificatePEM {
>  public:
>-  RTCCertificatePEM(
>-      const std::string& private_key,
>-      const std::string& certificate)
>-      : private_key_(private_key),
>-        certificate_(certificate) {}
>+  RTCCertificatePEM(const std::string& private_key,
>+                    const std::string& certificate)
>+      : private_key_(private_key), certificate_(certificate) {}
> 
>   const std::string& private_key() const { return private_key_; }
>   const std::string& certificate() const { return certificate_; }
>@@ -58,6 +56,7 @@ class RTCCertificate : public RefCountInterface {
>   // relative to epoch, 1970-01-01T00:00:00Z.
>   bool HasExpired(uint64_t now) const;
>   const SSLCertificate& ssl_certificate() const;
>+  const SSLCertChain& ssl_cert_chain() const;
> 
>   // TODO(hbos): If possible, remove once RTCCertificate and its
>   // ssl_certificate() is used in all relevant places. Should not pass around
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rtccertificate_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rtccertificate_unittest.cc
>index 7252a040b47..bb88d8a4d9f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rtccertificate_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rtccertificate_unittest.cc
>@@ -47,9 +47,7 @@ class RTCCertificateTest : public testing::Test {
>   //   As a result, ExpiresSeconds and HasExpiredSeconds are used instead of
>   // RTCCertificate::Expires and ::HasExpired for ms -> s conversion.
> 
>-  uint64_t NowSeconds() const {
>-    return TimeNanos() / kNumNanosecsPerSec;
>-  }
>+  uint64_t NowSeconds() const { return TimeNanos() / kNumNanosecsPerSec; }
> 
>   uint64_t ExpiresSeconds(const scoped_refptr<RTCCertificate>& cert) const {
>     uint64_t exp_ms = cert->Expires();
>@@ -94,7 +92,7 @@ TEST_F(RTCCertificateTest, NewCertificateNotExpired) {
>   EXPECT_FALSE(HasExpiredSeconds(certificate, now));
>   // Even without specifying the expiration time we would expect it to be valid
>   // for at least half an hour.
>-  EXPECT_FALSE(HasExpiredSeconds(certificate, now + 30*60));
>+  EXPECT_FALSE(HasExpiredSeconds(certificate, now + 30 * 60));
> }
> 
> TEST_F(RTCCertificateTest, UsesExpiresAskedFor) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rtccertificategenerator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rtccertificategenerator.cc
>index cacff61e35d..0b51c611bad 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rtccertificategenerator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rtccertificategenerator.cc
>@@ -42,7 +42,7 @@ class RTCCertificateGenerationTask : public RefCountInterface,
>       Thread* signaling_thread,
>       Thread* worker_thread,
>       const KeyParams& key_params,
>-      const Optional<uint64_t>& expires_ms,
>+      const absl::optional<uint64_t>& expires_ms,
>       const scoped_refptr<RTCCertificateGeneratorCallback>& callback)
>       : signaling_thread_(signaling_thread),
>         worker_thread_(worker_thread),
>@@ -94,7 +94,7 @@ class RTCCertificateGenerationTask : public RefCountInterface,
>   Thread* const signaling_thread_;
>   Thread* const worker_thread_;
>   const KeyParams key_params_;
>-  const Optional<uint64_t> expires_ms_;
>+  const absl::optional<uint64_t> expires_ms_;
>   const scoped_refptr<RTCCertificateGeneratorCallback> callback_;
>   scoped_refptr<RTCCertificate> certificate_;
> };
>@@ -102,10 +102,9 @@ class RTCCertificateGenerationTask : public RefCountInterface,
> }  // namespace
> 
> // static
>-scoped_refptr<RTCCertificate>
>-RTCCertificateGenerator::GenerateCertificate(
>+scoped_refptr<RTCCertificate> RTCCertificateGenerator::GenerateCertificate(
>     const KeyParams& key_params,
>-    const Optional<uint64_t>& expires_ms) {
>+    const absl::optional<uint64_t>& expires_ms) {
>   if (!key_params.IsValid())
>     return nullptr;
>   SSLIdentity* identity;
>@@ -122,8 +121,8 @@ RTCCertificateGenerator::GenerateCertificate(
>     // |SSLIdentity::Generate| should stop relying on |time_t|.
>     // See bugs.webrtc.org/5720.
>     time_t cert_lifetime_s = static_cast<time_t>(expires_s);
>-    identity = SSLIdentity::GenerateWithExpiration(
>-        kIdentityName, key_params, cert_lifetime_s);
>+    identity = SSLIdentity::GenerateWithExpiration(kIdentityName, key_params,
>+                                                   cert_lifetime_s);
>   }
>   if (!identity)
>     return nullptr;
>@@ -131,17 +130,16 @@ RTCCertificateGenerator::GenerateCertificate(
>   return RTCCertificate::Create(std::move(identity_sptr));
> }
> 
>-RTCCertificateGenerator::RTCCertificateGenerator(
>-    Thread* signaling_thread, Thread* worker_thread)
>-    : signaling_thread_(signaling_thread),
>-      worker_thread_(worker_thread) {
>+RTCCertificateGenerator::RTCCertificateGenerator(Thread* signaling_thread,
>+                                                 Thread* worker_thread)
>+    : signaling_thread_(signaling_thread), worker_thread_(worker_thread) {
>   RTC_DCHECK(signaling_thread_);
>   RTC_DCHECK(worker_thread_);
> }
> 
> void RTCCertificateGenerator::GenerateCertificateAsync(
>     const KeyParams& key_params,
>-    const Optional<uint64_t>& expires_ms,
>+    const absl::optional<uint64_t>& expires_ms,
>     const scoped_refptr<RTCCertificateGeneratorCallback>& callback) {
>   RTC_DCHECK(signaling_thread_->IsCurrent());
>   RTC_DCHECK(callback);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rtccertificategenerator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rtccertificategenerator.h
>index 1f854667d5e..a6c503a9f29 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rtccertificategenerator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rtccertificategenerator.h
>@@ -11,7 +11,7 @@
> #ifndef RTC_BASE_RTCCERTIFICATEGENERATOR_H_
> #define RTC_BASE_RTCCERTIFICATEGENERATOR_H_
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> #include "rtc_base/refcount.h"
> #include "rtc_base/rtccertificate.h"
> #include "rtc_base/scoped_ref_ptr.h"
>@@ -23,8 +23,7 @@ namespace rtc {
> // See |RTCCertificateGeneratorInterface::GenerateCertificateAsync|.
> class RTCCertificateGeneratorCallback : public RefCountInterface {
>  public:
>-  virtual void OnSuccess(
>-    const scoped_refptr<RTCCertificate>& certificate) = 0;
>+  virtual void OnSuccess(const scoped_refptr<RTCCertificate>& certificate) = 0;
>   virtual void OnFailure() = 0;
> 
>  protected:
>@@ -44,7 +43,7 @@ class RTCCertificateGeneratorInterface {
>   // its own restrictions on the expiration time.
>   virtual void GenerateCertificateAsync(
>       const KeyParams& key_params,
>-      const Optional<uint64_t>& expires_ms,
>+      const absl::optional<uint64_t>& expires_ms,
>       const scoped_refptr<RTCCertificateGeneratorCallback>& callback) = 0;
> };
> 
>@@ -61,7 +60,7 @@ class RTCCertificateGenerator : public RTCCertificateGeneratorInterface {
>   // specified, a default expiration time is used.
>   static scoped_refptr<RTCCertificate> GenerateCertificate(
>       const KeyParams& key_params,
>-      const Optional<uint64_t>& expires_ms);
>+      const absl::optional<uint64_t>& expires_ms);
> 
>   RTCCertificateGenerator(Thread* signaling_thread, Thread* worker_thread);
>   ~RTCCertificateGenerator() override {}
>@@ -73,7 +72,7 @@ class RTCCertificateGenerator : public RTCCertificateGeneratorInterface {
>   // specified, a default expiration time is used.
>   void GenerateCertificateAsync(
>       const KeyParams& key_params,
>-      const Optional<uint64_t>& expires_ms,
>+      const absl::optional<uint64_t>& expires_ms,
>       const scoped_refptr<RTCCertificateGeneratorCallback>& callback) override;
> 
>  private:
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rtccertificategenerator_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rtccertificategenerator_unittest.cc
>index 9a0ad0cb8d8..3929128960f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rtccertificategenerator_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/rtccertificategenerator_unittest.cc
>@@ -12,7 +12,7 @@
> 
> #include <memory>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/gunit.h"
> #include "rtc_base/logging.h"
>@@ -30,7 +30,7 @@ class RTCCertificateGeneratorFixture : public RTCCertificateGeneratorCallback {
>     RTC_CHECK(signaling_thread_);
>     RTC_CHECK(worker_thread_->Start());
>     generator_.reset(
>-      new RTCCertificateGenerator(signaling_thread_, worker_thread_.get()));
>+        new RTCCertificateGenerator(signaling_thread_, worker_thread_.get()));
>   }
>   ~RTCCertificateGeneratorFixture() override {}
> 
>@@ -67,8 +67,7 @@ class RTCCertificateGeneratorFixture : public RTCCertificateGeneratorCallback {
>   bool generate_async_completed_;
> };
> 
>-class RTCCertificateGeneratorTest
>-    : public testing::Test {
>+class RTCCertificateGeneratorTest : public testing::Test {
>  public:
>   RTCCertificateGeneratorTest()
>       : fixture_(new RefCountedObject<RTCCertificateGeneratorFixture>()) {}
>@@ -80,23 +79,19 @@ class RTCCertificateGeneratorTest
> };
> 
> TEST_F(RTCCertificateGeneratorTest, GenerateECDSA) {
>-  EXPECT_TRUE(RTCCertificateGenerator::GenerateCertificate(
>-      KeyParams::ECDSA(),
>-      Optional<uint64_t>()));
>+  EXPECT_TRUE(RTCCertificateGenerator::GenerateCertificate(KeyParams::ECDSA(),
>+                                                           absl::nullopt));
> }
> 
> TEST_F(RTCCertificateGeneratorTest, GenerateRSA) {
>-  EXPECT_TRUE(RTCCertificateGenerator::GenerateCertificate(
>-      KeyParams::RSA(),
>-      Optional<uint64_t>()));
>+  EXPECT_TRUE(RTCCertificateGenerator::GenerateCertificate(KeyParams::RSA(),
>+                                                           absl::nullopt));
> }
> 
> TEST_F(RTCCertificateGeneratorTest, GenerateAsyncECDSA) {
>   EXPECT_FALSE(fixture_->certificate());
>-  fixture_->generator()->GenerateCertificateAsync(
>-      KeyParams::ECDSA(),
>-      Optional<uint64_t>(),
>-      fixture_);
>+  fixture_->generator()->GenerateCertificateAsync(KeyParams::ECDSA(),
>+                                                  absl::nullopt, fixture_);
>   // Until generation has completed, the certificate is null. Since this is an
>   // async call, generation must not have completed until we process messages
>   // posted to this thread (which is done by |EXPECT_TRUE_WAIT|).
>@@ -115,15 +110,14 @@ TEST_F(RTCCertificateGeneratorTest, GenerateWithExpires) {
> 
>   // Generate a certificate that expires immediately.
>   scoped_refptr<RTCCertificate> cert_a =
>-      RTCCertificateGenerator::GenerateCertificate(
>-          KeyParams::ECDSA(), Optional<uint64_t>(0));
>+      RTCCertificateGenerator::GenerateCertificate(KeyParams::ECDSA(), 0);
>   EXPECT_TRUE(cert_a);
> 
>   // Generate a certificate that expires in one minute.
>   const uint64_t kExpiresMs = 60000;
>   scoped_refptr<RTCCertificate> cert_b =
>-      RTCCertificateGenerator::GenerateCertificate(
>-          KeyParams::ECDSA(), Optional<uint64_t>(kExpiresMs));
>+      RTCCertificateGenerator::GenerateCertificate(KeyParams::ECDSA(),
>+                                                   kExpiresMs);
>   EXPECT_TRUE(cert_b);
> 
>   // Verify that |cert_b| expires approximately |kExpiresMs| after |cert_a|
>@@ -131,20 +125,18 @@ TEST_F(RTCCertificateGeneratorTest, GenerateWithExpires) {
>   EXPECT_GT(cert_b->Expires(), cert_a->Expires());
>   uint64_t expires_diff = cert_b->Expires() - cert_a->Expires();
>   EXPECT_GE(expires_diff, kExpiresMs);
>-  EXPECT_LE(expires_diff, kExpiresMs + 2*kGenerationTimeoutMs + 1000);
>+  EXPECT_LE(expires_diff, kExpiresMs + 2 * kGenerationTimeoutMs + 1000);
> }
> 
> TEST_F(RTCCertificateGeneratorTest, GenerateWithInvalidParamsShouldFail) {
>   KeyParams invalid_params = KeyParams::RSA(0, 0);
>   EXPECT_FALSE(invalid_params.IsValid());
> 
>-  EXPECT_FALSE(RTCCertificateGenerator::GenerateCertificate(
>-      invalid_params, Optional<uint64_t>()));
>+  EXPECT_FALSE(RTCCertificateGenerator::GenerateCertificate(invalid_params,
>+                                                            absl::nullopt));
> 
>-  fixture_->generator()->GenerateCertificateAsync(
>-      invalid_params,
>-      Optional<uint64_t>(),
>-      fixture_);
>+  fixture_->generator()->GenerateCertificateAsync(invalid_params, absl::nullopt,
>+                                                  fixture_);
>   EXPECT_TRUE_WAIT(fixture_->GenerateAsyncCompleted(), kGenerationTimeoutMs);
>   EXPECT_FALSE(fixture_->certificate());
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sanitizer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sanitizer.h
>index 96d024883fa..23a748f84fc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sanitizer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sanitizer.h
>@@ -11,7 +11,11 @@
> #ifndef RTC_BASE_SANITIZER_H_
> #define RTC_BASE_SANITIZER_H_
> 
>-#include <stddef.h>  // for size_t
>+#include <stddef.h>  // For size_t.
>+
>+#ifdef __cplusplus
>+#include <type_traits>
>+#endif
> 
> #if defined(__has_feature)
> #if __has_feature(address_sanitizer)
>@@ -52,10 +56,6 @@ static inline void rtc_AsanPoison(const volatile void* ptr,
>                                   size_t num_elements) {
> #if RTC_HAS_ASAN
>   ASAN_POISON_MEMORY_REGION(ptr, element_size * num_elements);
>-#else
>-  (void)ptr;
>-  (void)element_size;
>-  (void)num_elements;
> #endif
> }
> 
>@@ -67,10 +67,6 @@ static inline void rtc_AsanUnpoison(const volatile void* ptr,
>                                     size_t num_elements) {
> #if RTC_HAS_ASAN
>   ASAN_UNPOISON_MEMORY_REGION(ptr, element_size * num_elements);
>-#else
>-  (void)ptr;
>-  (void)element_size;
>-  (void)num_elements;
> #endif
> }
> 
>@@ -81,10 +77,6 @@ static inline void rtc_MsanMarkUninitialized(const volatile void* ptr,
>                                              size_t num_elements) {
> #if RTC_HAS_MSAN
>   __msan_poison(ptr, element_size * num_elements);
>-#else
>-  (void)ptr;
>-  (void)element_size;
>-  (void)num_elements;
> #endif
> }
> 
>@@ -96,16 +88,23 @@ static inline void rtc_MsanCheckInitialized(const volatile void* ptr,
>                                             size_t num_elements) {
> #if RTC_HAS_MSAN
>   __msan_check_mem_is_initialized(ptr, element_size * num_elements);
>-#else
>-  (void)ptr;
>-  (void)element_size;
>-  (void)num_elements;
> #endif
> }
> 
> #ifdef __cplusplus
> 
> namespace rtc {
>+namespace sanitizer_impl {
>+
>+template <typename T>
>+constexpr bool IsTriviallyCopyable() {
>+  return static_cast<bool>(std::is_trivially_copy_constructible<T>::value &&
>+                           (std::is_trivially_copy_assignable<T>::value ||
>+                            !std::is_copy_assignable<T>::value) &&
>+                           std::is_trivially_destructible<T>::value);
>+}
>+
>+}  // namespace sanitizer_impl
> 
> template <typename T>
> inline void AsanPoison(const T& mem) {
>@@ -122,6 +121,15 @@ inline void MsanMarkUninitialized(const T& mem) {
>   rtc_MsanMarkUninitialized(mem.data(), sizeof(mem.data()[0]), mem.size());
> }
> 
>+template <typename T>
>+inline T MsanUninitialized(T t) {
>+  // TODO(bugs.webrtc.org/8762): Switch to std::is_trivially_copyable when it
>+  // becomes available in downstream projects.
>+  static_assert(sanitizer_impl::IsTriviallyCopyable<T>(), "");
>+  rtc_MsanMarkUninitialized(&t, sizeof(T), 1);
>+  return t;
>+}
>+
> template <typename T>
> inline void MsanCheckInitialized(const T& mem) {
>   rtc_MsanCheckInitialized(mem.data(), sizeof(mem.data()[0]), mem.size());
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sanitizer_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sanitizer_unittest.cc
>new file mode 100644
>index 00000000000..8dc606802fe
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sanitizer_unittest.cc
>@@ -0,0 +1,150 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "rtc_base/sanitizer.h"
>+
>+#include "rtc_base/gunit.h"
>+#include "rtc_base/logging.h"
>+
>+#if RTC_HAS_MSAN
>+#include <sanitizer/msan_interface.h>
>+#endif
>+
>+namespace rtc {
>+namespace {
>+
>+// Test sanitizer_impl::IsTriviallyCopyable (at compile time).
>+
>+// Trivially copyable.
>+
>+struct TrTrTr {
>+  TrTrTr(const TrTrTr&) = default;
>+  TrTrTr& operator=(const TrTrTr&) = default;
>+  ~TrTrTr() = default;
>+};
>+static_assert(sanitizer_impl::IsTriviallyCopyable<TrTrTr>(), "");
>+
>+struct TrDeTr {
>+  TrDeTr(const TrDeTr&) = default;
>+  TrDeTr& operator=(const TrDeTr&) = delete;
>+  ~TrDeTr() = default;
>+};
>+static_assert(sanitizer_impl::IsTriviallyCopyable<TrDeTr>(), "");
>+
>+// Non trivially copyable.
>+
>+struct TrTrNt {
>+  TrTrNt(const TrTrNt&) = default;
>+  TrTrNt& operator=(const TrTrNt&) = default;
>+  ~TrTrNt();
>+};
>+static_assert(!sanitizer_impl::IsTriviallyCopyable<TrTrNt>(), "");
>+
>+struct TrNtTr {
>+  TrNtTr(const TrNtTr&) = default;
>+  TrNtTr& operator=(const TrNtTr&);
>+  ~TrNtTr() = default;
>+};
>+static_assert(!sanitizer_impl::IsTriviallyCopyable<TrNtTr>(), "");
>+
>+struct TrNtNt {
>+  TrNtNt(const TrNtNt&) = default;
>+  TrNtNt& operator=(const TrNtNt&);
>+  ~TrNtNt();
>+};
>+static_assert(!sanitizer_impl::IsTriviallyCopyable<TrNtNt>(), "");
>+
>+struct TrDeNt {
>+  TrDeNt(const TrDeNt&) = default;
>+  TrDeNt& operator=(const TrDeNt&) = delete;
>+  ~TrDeNt();
>+};
>+static_assert(!sanitizer_impl::IsTriviallyCopyable<TrDeNt>(), "");
>+
>+struct NtTrTr {
>+  NtTrTr(const NtTrTr&);
>+  NtTrTr& operator=(const NtTrTr&) = default;
>+  ~NtTrTr() = default;
>+};
>+static_assert(!sanitizer_impl::IsTriviallyCopyable<NtTrTr>(), "");
>+
>+struct NtTrNt {
>+  NtTrNt(const NtTrNt&);
>+  NtTrNt& operator=(const NtTrNt&) = default;
>+  ~NtTrNt();
>+};
>+static_assert(!sanitizer_impl::IsTriviallyCopyable<NtTrNt>(), "");
>+
>+struct NtNtTr {
>+  NtNtTr(const NtNtTr&);
>+  NtNtTr& operator=(const NtNtTr&);
>+  ~NtNtTr() = default;
>+};
>+static_assert(!sanitizer_impl::IsTriviallyCopyable<NtNtTr>(), "");
>+
>+struct NtNtNt {
>+  NtNtNt(const NtNtNt&);
>+  NtNtNt& operator=(const NtNtNt&);
>+  ~NtNtNt();
>+};
>+static_assert(!sanitizer_impl::IsTriviallyCopyable<NtNtNt>(), "");
>+
>+struct NtDeTr {
>+  NtDeTr(const NtDeTr&);
>+  NtDeTr& operator=(const NtDeTr&) = delete;
>+  ~NtDeTr() = default;
>+};
>+static_assert(!sanitizer_impl::IsTriviallyCopyable<NtDeTr>(), "");
>+
>+struct NtDeNt {
>+  NtDeNt(const NtDeNt&);
>+  NtDeNt& operator=(const NtDeNt&) = delete;
>+  ~NtDeNt();
>+};
>+static_assert(!sanitizer_impl::IsTriviallyCopyable<NtDeNt>(), "");
>+
>+// Trivially copyable types.
>+
>+struct Foo {
>+  uint32_t field1;
>+  uint16_t field2;
>+};
>+
>+struct Bar {
>+  uint32_t ID;
>+  Foo foo;
>+};
>+
>+// Run the callback, and expect a crash if it *doesn't* make an uninitialized
>+// memory read. If MSan isn't on, just run the callback.
>+template <typename F>
>+void MsanExpectUninitializedRead(F&& f) {
>+#if RTC_HAS_MSAN
>+  EXPECT_DEATH(f(), "");
>+#else
>+  f();
>+#endif
>+}
>+
>+}  // namespace
>+
>+TEST(SanitizerTest, MsanUninitialized) {
>+  Bar bar = MsanUninitialized<Bar>({});
>+  // Check that a read after initialization is OK.
>+  bar.ID = 1;
>+  EXPECT_EQ(1u, bar.ID);
>+  RTC_LOG(LS_INFO) << "read after init passed";
>+  // Check that other fields are uninitialized and equal to zero.
>+  MsanExpectUninitializedRead([&] { EXPECT_EQ(0u, bar.foo.field1); });
>+  MsanExpectUninitializedRead([&] { EXPECT_EQ(0u, bar.foo.field2); });
>+  RTC_LOG(LS_INFO) << "read with no init passed";
>+}
>+
>+}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/scoped_ref_ptr.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/scoped_ref_ptr.h
>index 0f4698a2561..a583aa99b53 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/scoped_ref_ptr.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/scoped_ref_ptr.h
>@@ -103,11 +103,11 @@ class scoped_refptr {
>   operator T*() const { return ptr_; }
>   T* operator->() const { return ptr_; }
> 
>-  // Release a pointer.
>-  // The return value is the current pointer held by this object.
>-  // If this object holds a null pointer, the return value is null.
>-  // After this operation, this object will hold a null pointer,
>-  // and will not own the object any more.
>+  // Returns the (possibly null) raw pointer, and makes the scoped_refptr hold a
>+  // null pointer, all without touching the reference count of the underlying
>+  // pointed-to object. The object is still reference counted, and the caller of
>+  // release() is now the proud owner of one reference, so it is responsible for
>+  // calling Release() once on the object when no longer using it.
>   T* release() {
>     T* retVal = ptr_;
>     ptr_ = nullptr;
>@@ -118,8 +118,8 @@ class scoped_refptr {
>     // AddRef first so that self assignment should work
>     if (p)
>       p->AddRef();
>-    if (ptr_ )
>-      ptr_ ->Release();
>+    if (ptr_)
>+      ptr_->Release();
>     ptr_ = p;
>     return *this;
>   }
>@@ -150,9 +150,7 @@ class scoped_refptr {
>     *pp = p;
>   }
> 
>-  void swap(scoped_refptr<T>& r) {
>-    swap(&r.ptr_);
>-  }
>+  void swap(scoped_refptr<T>& r) { swap(&r.ptr_); }
> 
>  protected:
>   T* ptr_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sequenced_task_checker.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sequenced_task_checker.h
>index cc369be68ea..eb151985de3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sequenced_task_checker.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sequenced_task_checker.h
>@@ -22,6 +22,12 @@
> #include "rtc_base/thread_annotations.h"
> 
> namespace rtc {
>+namespace internal {
>+// Forward declaration of the internal implementation of RTC_GUARDED_BY().
>+// SequencedTaskChecker grants this class access to call its IsCurrent() method.
>+// See thread_checker.h for more details.
>+class AnnounceOnThread;
>+}  // namespace internal
> 
> // Do nothing implementation, for use in release mode.
> //
>@@ -30,8 +36,11 @@ namespace rtc {
> class SequencedTaskCheckerDoNothing {
>  public:
>   bool CalledSequentially() const { return true; }
>-
>   void Detach() {}
>+
>+ private:
>+  friend class internal::AnnounceOnThread;
>+  bool IsCurrent() const { return CalledSequentially(); }
> };
> 
> // SequencedTaskChecker is a helper class used to help verify that some methods
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sequenced_task_checker_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sequenced_task_checker_impl.cc
>index d7f46ead7df..16069c25f7a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sequenced_task_checker_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sequenced_task_checker_impl.cc
>@@ -14,7 +14,6 @@
> #include <dispatch/dispatch.h>
> #endif
> 
>-#include "rtc_base/platform_thread.h"
> #include "rtc_base/sequenced_task_checker.h"
> #include "rtc_base/task_queue.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sequenced_task_checker_impl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sequenced_task_checker_impl.h
>index 0a5464efe89..293b1ac297b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sequenced_task_checker_impl.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sequenced_task_checker_impl.h
>@@ -11,6 +11,7 @@
> #ifndef RTC_BASE_SEQUENCED_TASK_CHECKER_IMPL_H_
> #define RTC_BASE_SEQUENCED_TASK_CHECKER_IMPL_H_
> 
>+#include "rtc_base/criticalsection.h"
> #include "rtc_base/thread_checker.h"
> 
> namespace rtc {
>@@ -34,6 +35,9 @@ class SequencedTaskCheckerImpl {
>   void Detach();
> 
>  private:
>+  friend class internal::AnnounceOnThread;
>+  bool IsCurrent() const { return CalledSequentially(); }
>+
>   typedef const void* QueueId;
>   CriticalSection lock_;
>   ThreadChecker thread_checker_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sequenced_task_checker_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sequenced_task_checker_unittest.cc
>index d77546289dc..96e655baad4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sequenced_task_checker_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sequenced_task_checker_unittest.cc
>@@ -9,15 +9,34 @@
>  */
> 
> #include "rtc_base/sequenced_task_checker.h"
>+
> #include "rtc_base/checks.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/platform_thread.h"
> #include "rtc_base/task_queue.h"
>+#include "rtc_base/thread_checker.h"
> #include "test/gtest.h"
> 
> namespace rtc {
> 
> namespace {
>+
>+// This class is dead code, but its purpose is to make sure that
>+// SequencedTaskChecker is compatible with the RTC_GUARDED_BY and RTC_RUN_ON
>+// attributes that are checked at compile-time.
>+class CompileTimeTestForGuardedBy {
>+ public:
>+  int CalledOnSequence() RTC_RUN_ON(sequence_checker_) { return guarded_; }
>+
>+  void CallMeFromSequence() {
>+    RTC_DCHECK_RUN_ON(&sequence_checker_) << "Should be called on sequence";
>+  }
>+
>+ private:
>+  int guarded_ RTC_GUARDED_BY(sequence_checker_);
>+  rtc::SequencedTaskChecker sequence_checker_;
>+};
>+
> // Calls SequencedTaskChecker::CalledSequentially on another thread.
> class CallCalledSequentiallyOnThread {
>  public:
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sha1.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sha1.cc
>deleted file mode 100644
>index 5b41f053705..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sha1.cc
>+++ /dev/null
>@@ -1,298 +0,0 @@
>-/*
>- * SHA-1 in C
>- * By Steve Reid <sreid@sea-to-sky.net>
>- * 100% Public Domain
>- *
>- * -----------------
>- * Modified 7/98
>- * By James H. Brown <jbrown@burgoyne.com>
>- * Still 100% Public Domain
>- *
>- * Corrected a problem which generated improper hash values on 16 bit machines
>- * Routine SHA1Update changed from
>- *   void SHA1Update(SHA1_CTX* context, unsigned char* data, unsigned int
>- * len)
>- * to
>- *   void SHA1Update(SHA1_CTX* context, unsigned char* data, unsigned
>- * long len)
>- *
>- * The 'len' parameter was declared an int which works fine on 32 bit machines.
>- * However, on 16 bit machines an int is too small for the shifts being done
>- * against
>- * it.  This caused the hash function to generate incorrect values if len was
>- * greater than 8191 (8K - 1) due to the 'len << 3' on line 3 of SHA1Update().
>- *
>- * Since the file IO in main() reads 16K at a time, any file 8K or larger would
>- * be guaranteed to generate the wrong hash (e.g. Test Vector #3, a million
>- * "a"s).
>- *
>- * I also changed the declaration of variables i & j in SHA1Update to
>- * unsigned long from unsigned int for the same reason.
>- *
>- * These changes should make no difference to any 32 bit implementations since
>- * an
>- * int and a long are the same size in those environments.
>- *
>- * --
>- * I also corrected a few compiler warnings generated by Borland C.
>- * 1. Added #include <process.h> for exit() prototype
>- * 2. Removed unused variable 'j' in SHA1Final
>- * 3. Changed exit(0) to return(0) at end of main.
>- *
>- * ALL changes I made can be located by searching for comments containing 'JHB'
>- * -----------------
>- * Modified 8/98
>- * By Steve Reid <sreid@sea-to-sky.net>
>- * Still 100% public domain
>- *
>- * 1- Removed #include <process.h> and used return() instead of exit()
>- * 2- Fixed overwriting of finalcount in SHA1Final() (discovered by Chris Hall)
>- * 3- Changed email address from steve@edmweb.com to sreid@sea-to-sky.net
>- *
>- * -----------------
>- * Modified 4/01
>- * By Saul Kravitz <Saul.Kravitz@celera.com>
>- * Still 100% PD
>- * Modified to run on Compaq Alpha hardware.
>- *
>- * -----------------
>- * Modified 07/2002
>- * By Ralph Giles <giles@ghostscript.com>
>- * Still 100% public domain
>- * modified for use with stdint types, autoconf
>- * code cleanup, removed attribution comments
>- * switched SHA1Final() argument order for consistency
>- * use SHA1_ prefix for public api
>- * move public api to sha1.h
>- *
>- * -----------------
>- * Modified 02/2012
>- * By Justin Uberti <juberti@google.com>
>- * Remove underscore from SHA1 prefix to avoid conflict with OpenSSL
>- * Remove test code
>- * Untabify
>- *
>- * -----------------
>- * Modified 03/2012
>- * By Ronghua Wu <ronghuawu@google.com>
>- * Change the typedef of uint32(8)_t to uint32(8). We need this because in the
>- * chromium android build, the stdio.h will include stdint.h which already
>- * defined uint32(8)_t.
>- *
>- * -----------------
>- * Modified 04/2012
>- * By Frank Barchard <fbarchard@google.com>
>- * Ported to C++, Google style, change len to size_t, enable SHA1HANDSOFF
>- *
>- * Test Vectors (from FIPS PUB 180-1)
>- * "abc"
>- *   A9993E36 4706816A BA3E2571 7850C26C 9CD0D89D
>- * "abcdbcdecdefdefgefghfghighijhijkijkljklmklmnlmnomnopnopq"
>- *   84983E44 1C3BD26E BAAE4AA1 F95129E5 E54670F1
>- * A million repetitions of "a"
>- *   34AA973C D4C4DAA4 F61EEB2B DBAD2731 6534016F
>- *
>- * -----------------
>- * Modified 05/2015
>- * By Sergey Ulanov <sergeyu@chromium.org>
>- * Removed static buffer to make computation thread-safe.
>- *
>- * -----------------
>- * Modified 10/2015
>- * By Peter BostrÃ¶m <pbos@webrtc.org>
>- * Change uint32(8) back to uint32(8)_t (undoes (03/2012) change).
>- */
>-
>-// Enabling SHA1HANDSOFF preserves the caller's data buffer.
>-// Disabling SHA1HANDSOFF the buffer will be modified (end swapped).
>-#define SHA1HANDSOFF
>-
>-#include "rtc_base/sha1.h"
>-
>-#include <stdio.h>
>-#include <string.h>
>-
>-namespace rtc {
>-
>-namespace {
>-
>-#define rol(value, bits) (((value) << (bits)) | ((value) >> (32 - (bits))))
>-
>-// blk0() and blk() perform the initial expand.
>-// I got the idea of expanding during the round function from SSLeay
>-// FIXME: can we do this in an endian-proof way?
>-#ifdef RTC_ARCH_CPU_BIG_ENDIAN
>-#define blk0(i) block->l[i]
>-#else
>-#define blk0(i) (block->l[i] = (rol(block->l[i], 24) & 0xFF00FF00) | \
>-    (rol(block->l[i], 8) & 0x00FF00FF))
>-#endif
>-#define blk(i) (block->l[i & 15] = rol(block->l[(i + 13) & 15] ^ \
>-    block->l[(i + 8) & 15] ^ block->l[(i + 2) & 15] ^ block->l[i & 15], 1))
>-
>-// (R0+R1), R2, R3, R4 are the different operations used in SHA1.
>-#define R0(v, w, x, y, z, i) \
>-    z += ((w & (x ^ y)) ^ y) + blk0(i) + 0x5A827999 + rol(v, 5); \
>-    w = rol(w, 30);
>-#define R1(v, w, x, y, z, i) \
>-    z += ((w & (x ^ y)) ^ y) + blk(i) + 0x5A827999 + rol(v, 5); \
>-    w = rol(w, 30);
>-#define R2(v, w, x, y, z, i) \
>-    z += (w ^ x ^ y) + blk(i) + 0x6ED9EBA1 + rol(v, 5);\
>-    w = rol(w, 30);
>-#define R3(v, w, x, y, z, i) \
>-    z += (((w | x) & y) | (w & x)) + blk(i) + 0x8F1BBCDC + rol(v, 5); \
>-    w = rol(w, 30);
>-#define R4(v, w, x, y, z, i) \
>-    z += (w ^ x ^ y) + blk(i) + 0xCA62C1D6 + rol(v, 5); \
>-    w = rol(w, 30);
>-
>-#ifdef VERBOSE  // SAK
>-void SHAPrintContext(SHA1_CTX *context, char *msg) {
>-  printf("%s (%d,%d) %x %x %x %x %x\n",
>-         msg,
>-         context->count[0], context->count[1],
>-         context->state[0],
>-         context->state[1],
>-         context->state[2],
>-         context->state[3],
>-         context->state[4]);
>-}
>-#endif /* VERBOSE */
>-
>-// Hash a single 512-bit block. This is the core of the algorithm.
>-void SHA1Transform(uint32_t state[5], const uint8_t buffer[64]) {
>-  union CHAR64LONG16 {
>-    uint8_t c[64];
>-    uint32_t l[16];
>-  };
>-#ifdef SHA1HANDSOFF
>-  uint8_t workspace[64];
>-  memcpy(workspace, buffer, 64);
>-  CHAR64LONG16* block = reinterpret_cast<CHAR64LONG16*>(workspace);
>-#else
>-  // Note(fbarchard): This option does modify the user's data buffer.
>-  CHAR64LONG16* block = const_cast<CHAR64LONG16*>(
>-      reinterpret_cast<const CHAR64LONG16*>(buffer));
>-#endif
>-
>-  // Copy context->state[] to working vars.
>-  uint32_t a = state[0];
>-  uint32_t b = state[1];
>-  uint32_t c = state[2];
>-  uint32_t d = state[3];
>-  uint32_t e = state[4];
>-
>-  // 4 rounds of 20 operations each. Loop unrolled.
>-  // Note(fbarchard): The following has lint warnings for multiple ; on
>-  // a line and no space after , but is left as-is to be similar to the
>-  // original code.
>-  R0(a,b,c,d,e,0); R0(e,a,b,c,d,1); R0(d,e,a,b,c,2); R0(c,d,e,a,b,3);
>-  R0(b,c,d,e,a,4); R0(a,b,c,d,e,5); R0(e,a,b,c,d,6); R0(d,e,a,b,c,7);
>-  R0(c,d,e,a,b,8); R0(b,c,d,e,a,9); R0(a,b,c,d,e,10); R0(e,a,b,c,d,11);
>-  R0(d,e,a,b,c,12); R0(c,d,e,a,b,13); R0(b,c,d,e,a,14); R0(a,b,c,d,e,15);
>-  R1(e,a,b,c,d,16); R1(d,e,a,b,c,17); R1(c,d,e,a,b,18); R1(b,c,d,e,a,19);
>-  R2(a,b,c,d,e,20); R2(e,a,b,c,d,21); R2(d,e,a,b,c,22); R2(c,d,e,a,b,23);
>-  R2(b,c,d,e,a,24); R2(a,b,c,d,e,25); R2(e,a,b,c,d,26); R2(d,e,a,b,c,27);
>-  R2(c,d,e,a,b,28); R2(b,c,d,e,a,29); R2(a,b,c,d,e,30); R2(e,a,b,c,d,31);
>-  R2(d,e,a,b,c,32); R2(c,d,e,a,b,33); R2(b,c,d,e,a,34); R2(a,b,c,d,e,35);
>-  R2(e,a,b,c,d,36); R2(d,e,a,b,c,37); R2(c,d,e,a,b,38); R2(b,c,d,e,a,39);
>-  R3(a,b,c,d,e,40); R3(e,a,b,c,d,41); R3(d,e,a,b,c,42); R3(c,d,e,a,b,43);
>-  R3(b,c,d,e,a,44); R3(a,b,c,d,e,45); R3(e,a,b,c,d,46); R3(d,e,a,b,c,47);
>-  R3(c,d,e,a,b,48); R3(b,c,d,e,a,49); R3(a,b,c,d,e,50); R3(e,a,b,c,d,51);
>-  R3(d,e,a,b,c,52); R3(c,d,e,a,b,53); R3(b,c,d,e,a,54); R3(a,b,c,d,e,55);
>-  R3(e,a,b,c,d,56); R3(d,e,a,b,c,57); R3(c,d,e,a,b,58); R3(b,c,d,e,a,59);
>-  R4(a,b,c,d,e,60); R4(e,a,b,c,d,61); R4(d,e,a,b,c,62); R4(c,d,e,a,b,63);
>-  R4(b,c,d,e,a,64); R4(a,b,c,d,e,65); R4(e,a,b,c,d,66); R4(d,e,a,b,c,67);
>-  R4(c,d,e,a,b,68); R4(b,c,d,e,a,69); R4(a,b,c,d,e,70); R4(e,a,b,c,d,71);
>-  R4(d,e,a,b,c,72); R4(c,d,e,a,b,73); R4(b,c,d,e,a,74); R4(a,b,c,d,e,75);
>-  R4(e,a,b,c,d,76); R4(d,e,a,b,c,77); R4(c,d,e,a,b,78); R4(b,c,d,e,a,79);
>-
>-  // Add the working vars back into context.state[].
>-  state[0] += a;
>-  state[1] += b;
>-  state[2] += c;
>-  state[3] += d;
>-  state[4] += e;
>-}
>-
>-}  // namespace
>-
>-// SHA1Init - Initialize new context.
>-void SHA1Init(SHA1_CTX* context) {
>-  // SHA1 initialization constants.
>-  context->state[0] = 0x67452301;
>-  context->state[1] = 0xEFCDAB89;
>-  context->state[2] = 0x98BADCFE;
>-  context->state[3] = 0x10325476;
>-  context->state[4] = 0xC3D2E1F0;
>-  context->count[0] = context->count[1] = 0;
>-}
>-
>-// Run your data through this.
>-void SHA1Update(SHA1_CTX* context, const uint8_t* data, size_t input_len) {
>-  size_t i = 0;
>-
>-#ifdef VERBOSE
>-  SHAPrintContext(context, "before");
>-#endif
>-
>-  // Compute number of bytes mod 64.
>-  size_t index = (context->count[0] >> 3) & 63;
>-
>-  // Update number of bits.
>-  // TODO: Use uint64_t instead of 2 uint32_t for count.
>-  // count[0] has low 29 bits for byte count + 3 pad 0's making 32 bits for
>-  // bit count.
>-  // Add bit count to low uint32_t
>-  context->count[0] += static_cast<uint32_t>(input_len << 3);
>-  if (context->count[0] < static_cast<uint32_t>(input_len << 3)) {
>-    ++context->count[1];  // if overlow (carry), add one to high word
>-  }
>-  context->count[1] += static_cast<uint32_t>(input_len >> 29);
>-  if ((index + input_len) > 63) {
>-    i = 64 - index;
>-    memcpy(&context->buffer[index], data, i);
>-    SHA1Transform(context->state, context->buffer);
>-    for (; i + 63 < input_len; i += 64) {
>-      SHA1Transform(context->state, data + i);
>-    }
>-    index = 0;
>-  }
>-  memcpy(&context->buffer[index], &data[i], input_len - i);
>-
>-#ifdef VERBOSE
>-  SHAPrintContext(context, "after ");
>-#endif
>-}
>-
>-// Add padding and return the message digest.
>-void SHA1Final(SHA1_CTX* context, uint8_t digest[SHA1_DIGEST_SIZE]) {
>-  uint8_t finalcount[8];
>-  for (int i = 0; i < 8; ++i) {
>-    // Endian independent
>-    finalcount[i] = static_cast<uint8_t>(
>-        (context->count[(i >= 4 ? 0 : 1)] >> ((3 - (i & 3)) * 8)) & 255);
>-  }
>-  SHA1Update(context, reinterpret_cast<const uint8_t*>("\200"), 1);
>-  while ((context->count[0] & 504) != 448) {
>-    SHA1Update(context, reinterpret_cast<const uint8_t*>("\0"), 1);
>-  }
>-  SHA1Update(context, finalcount, 8);  // Should cause a SHA1Transform().
>-  for (int i = 0; i < SHA1_DIGEST_SIZE; ++i) {
>-    digest[i] = static_cast<uint8_t>(
>-        (context->state[i >> 2] >> ((3 - (i & 3)) * 8)) & 255);
>-  }
>-
>-  // Wipe variables.
>-  memset(context->buffer, 0, 64);
>-  memset(context->state, 0, 20);
>-  memset(context->count, 0, 8);
>-  memset(finalcount, 0, 8);   // SWR
>-
>-#ifdef SHA1HANDSOFF  // Make SHA1Transform overwrite its own static vars.
>-  SHA1Transform(context->state, context->buffer);
>-#endif
>-}
>-
>-}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sha1.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sha1.h
>deleted file mode 100644
>index f3242b34989..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sha1.h
>+++ /dev/null
>@@ -1,33 +0,0 @@
>-/*
>- * SHA-1 in C
>- * By Steve Reid <sreid@sea-to-sky.net>
>- * 100% Public Domain
>- *
>-*/
>-
>-// Ported to C++, Google style, under namespace rtc.
>-
>-#ifndef RTC_BASE_SHA1_H_
>-#define RTC_BASE_SHA1_H_
>-
>-#include <stdint.h>
>-#include <stdlib.h>
>-
>-namespace rtc {
>-
>-struct SHA1_CTX {
>-  uint32_t state[5];
>-  // TODO: Change bit count to uint64_t.
>-  uint32_t count[2];  // Bit count of input.
>-  uint8_t buffer[64];
>-};
>-
>-#define SHA1_DIGEST_SIZE 20
>-
>-void SHA1Init(SHA1_CTX* context);
>-void SHA1Update(SHA1_CTX* context, const uint8_t* data, size_t len);
>-void SHA1Final(SHA1_CTX* context, uint8_t digest[SHA1_DIGEST_SIZE]);
>-
>-}  // namespace rtc
>-
>-#endif  // RTC_BASE_SHA1_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sha1digest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sha1digest.cc
>deleted file mode 100644
>index 7f7190a3f1a..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sha1digest.cc
>+++ /dev/null
>@@ -1,32 +0,0 @@
>-/*
>- *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "rtc_base/sha1digest.h"
>-
>-namespace rtc {
>-
>-size_t Sha1Digest::Size() const {
>-  return kSize;
>-}
>-
>-void Sha1Digest::Update(const void* buf, size_t len) {
>-  SHA1Update(&ctx_, static_cast<const uint8_t*>(buf), len);
>-}
>-
>-size_t Sha1Digest::Finish(void* buf, size_t len) {
>-  if (len < kSize) {
>-    return 0;
>-  }
>-  SHA1Final(&ctx_, static_cast<uint8_t*>(buf));
>-  SHA1Init(&ctx_);  // Reset for next use.
>-  return kSize;
>-}
>-
>-}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sha1digest.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sha1digest.h
>deleted file mode 100644
>index 322e22c833b..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sha1digest.h
>+++ /dev/null
>@@ -1,36 +0,0 @@
>-/*
>- *  Copyright 2012 The WebRTC Project Authors. All rights reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef RTC_BASE_SHA1DIGEST_H_
>-#define RTC_BASE_SHA1DIGEST_H_
>-
>-#include "rtc_base/messagedigest.h"
>-#include "rtc_base/sha1.h"
>-
>-namespace rtc {
>-
>-// A simple wrapper for our SHA-1 implementation.
>-class Sha1Digest : public MessageDigest {
>- public:
>-  enum { kSize = SHA1_DIGEST_SIZE };
>-  Sha1Digest() {
>-    SHA1Init(&ctx_);
>-  }
>-  size_t Size() const override;
>-  void Update(const void* buf, size_t len) override;
>-  size_t Finish(void* buf, size_t len) override;
>-
>- private:
>-  SHA1_CTX ctx_;
>-};
>-
>-}  // namespace rtc
>-
>-#endif  // RTC_BASE_SHA1DIGEST_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sha1digest_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sha1digest_unittest.cc
>deleted file mode 100644
>index b3dd8323da8..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sha1digest_unittest.cc
>+++ /dev/null
>@@ -1,82 +0,0 @@
>-/*
>- *  Copyright 2012 The WebRTC Project Authors. All rights reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "rtc_base/sha1digest.h"
>-#include "rtc_base/gunit.h"
>-#include "rtc_base/stringencode.h"
>-
>-namespace rtc {
>-
>-std::string Sha1(const std::string& input) {
>-  Sha1Digest sha1;
>-  return ComputeDigest(&sha1, input);
>-}
>-
>-TEST(Sha1DigestTest, TestSize) {
>-  Sha1Digest sha1;
>-  EXPECT_EQ(20, static_cast<int>(Sha1Digest::kSize));
>-  EXPECT_EQ(20U, sha1.Size());
>-}
>-
>-TEST(Sha1DigestTest, TestBasic) {
>-  // Test vectors from sha1.c.
>-  EXPECT_EQ("da39a3ee5e6b4b0d3255bfef95601890afd80709", Sha1(""));
>-  EXPECT_EQ("a9993e364706816aba3e25717850c26c9cd0d89d", Sha1("abc"));
>-  EXPECT_EQ("84983e441c3bd26ebaae4aa1f95129e5e54670f1",
>-            Sha1("abcdbcdecdefdefgefghfghighijhijkijkljklmklmnlmnomnopnopq"));
>-  std::string a_million_as(1000000, 'a');
>-  EXPECT_EQ("34aa973cd4c4daa4f61eeb2bdbad27316534016f", Sha1(a_million_as));
>-}
>-
>-TEST(Sha1DigestTest, TestMultipleUpdates) {
>-  Sha1Digest sha1;
>-  std::string input =
>-      "abcdbcdecdefdefgefghfghighijhijkijkljklmklmnlmnomnopnopq";
>-  char output[Sha1Digest::kSize];
>-  for (size_t i = 0; i < input.size(); ++i) {
>-    sha1.Update(&input[i], 1);
>-  }
>-  EXPECT_EQ(sha1.Size(), sha1.Finish(output, sizeof(output)));
>-  EXPECT_EQ("84983e441c3bd26ebaae4aa1f95129e5e54670f1",
>-            hex_encode(output, sizeof(output)));
>-}
>-
>-TEST(Sha1DigestTest, TestReuse) {
>-  Sha1Digest sha1;
>-  std::string input = "abc";
>-  EXPECT_EQ("a9993e364706816aba3e25717850c26c9cd0d89d",
>-            ComputeDigest(&sha1, input));
>-  input = "abcdbcdecdefdefgefghfghighijhijkijkljklmklmnlmnomnopnopq";
>-  EXPECT_EQ("84983e441c3bd26ebaae4aa1f95129e5e54670f1",
>-            ComputeDigest(&sha1, input));
>-}
>-
>-TEST(Sha1DigestTest, TestBufferTooSmall) {
>-  Sha1Digest sha1;
>-  std::string input = "abcdefghijklmnopqrstuvwxyz";
>-  char output[Sha1Digest::kSize - 1];
>-  sha1.Update(input.c_str(), input.size());
>-  EXPECT_EQ(0U, sha1.Finish(output, sizeof(output)));
>-}
>-
>-TEST(Sha1DigestTest, TestBufferConst) {
>-  Sha1Digest sha1;
>-  const int kLongSize = 1000000;
>-  std::string input(kLongSize, '\0');
>-  for (int i = 0; i < kLongSize; ++i) {
>-    input[i] = static_cast<char>(i);
>-  }
>-  sha1.Update(input.c_str(), input.size());
>-  for (int i = 0; i < kLongSize; ++i) {
>-    EXPECT_EQ(static_cast<char>(i), input[i]);
>-  }
>-}
>-
>-}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/signalthread.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/signalthread.cc
>index 3856dac99e2..58f8761b808 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/signalthread.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/signalthread.cc
>@@ -10,6 +10,7 @@
> 
> #include "rtc_base/signalthread.h"
> 
>+#include "absl/memory/memory.h"
> #include "rtc_base/checks.h"
> 
> namespace rtc {
>@@ -18,11 +19,8 @@ namespace rtc {
> // SignalThread
> ///////////////////////////////////////////////////////////////////////////////
> 
>-SignalThread::SignalThread(bool use_socket_server)
>-    : main_(Thread::Current()),
>-      worker_(this, use_socket_server),
>-      state_(kInit),
>-      refcount_(1) {
>+SignalThread::SignalThread()
>+    : main_(Thread::Current()), worker_(this), state_(kInit), refcount_(1) {
>   main_->SignalQueueDestroyed.connect(this,
>                                       &SignalThread::OnMainThreadDestroyed);
>   worker_.SetName("SignalThread", this);
>@@ -93,7 +91,7 @@ bool SignalThread::ContinueWork() {
>   return worker_.ProcessMessages(0);
> }
> 
>-void SignalThread::OnMessage(Message *msg) {
>+void SignalThread::OnMessage(Message* msg) {
>   EnterExit ee(this);
>   if (ST_MSG_WORKER_DONE == msg->message_id) {
>     RTC_DCHECK(main_->IsCurrent());
>@@ -124,6 +122,12 @@ void SignalThread::OnMessage(Message *msg) {
>   }
> }
> 
>+SignalThread::Worker::Worker(SignalThread* parent)
>+    : Thread(absl::make_unique<NullSocketServer>(), /*do_init=*/false),
>+      parent_(parent) {
>+  DoInit();
>+}
>+
> SignalThread::Worker::~Worker() {
>   Stop();
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/signalthread.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/signalthread.h
>index 68b7f84085a..be54d9cb4a0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/signalthread.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/signalthread.h
>@@ -16,7 +16,7 @@
> #include "rtc_base/checks.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/nullsocketserver.h"
>-#include "rtc_base/sigslot.h"
>+#include "rtc_base/third_party/sigslot/sigslot.h"
> #include "rtc_base/thread.h"
> 
> namespace rtc {
>@@ -38,11 +38,9 @@ namespace rtc {
> //   tasks in the context of the main thread.
> ///////////////////////////////////////////////////////////////////////////////
> 
>-class SignalThread
>-    : public sigslot::has_slots<>,
>-      protected MessageHandler {
>+class SignalThread : public sigslot::has_slots<>, protected MessageHandler {
>  public:
>-  explicit SignalThread(bool use_socket_server = true);
>+  SignalThread();
> 
>   // Context: Main Thread.  Call before Start to change the worker's name.
>   bool SetName(const std::string& name, const void* obj);
>@@ -63,7 +61,7 @@ class SignalThread
>   void Release();
> 
>   // Context: Main Thread.  Signalled when work is complete.
>-  sigslot::signal1<SignalThread *> SignalWorkDone;
>+  sigslot::signal1<SignalThread*> SignalWorkDone;
> 
>   enum { ST_MSG_WORKER_DONE, ST_MSG_FIRST_AVAILABLE };
> 
>@@ -73,7 +71,7 @@ class SignalThread
>   Thread* worker() { return &worker_; }
> 
>   // Context: Main Thread.  Subclass should override to do pre-work setup.
>-  virtual void OnWorkStart() { }
>+  virtual void OnWorkStart() {}
> 
>   // Context: Worker Thread.  Subclass should override to do work.
>   virtual void DoWork() = 0;
>@@ -84,10 +82,10 @@ class SignalThread
> 
>   // Context: Worker Thread.  Subclass should override when extra work is
>   // needed to abort the worker thread.
>-  virtual void OnWorkStop() { }
>+  virtual void OnWorkStop() {}
> 
>   // Context: Main Thread.  Subclass should override to do post-work cleanup.
>-  virtual void OnWorkDone() { }
>+  virtual void OnWorkDone() {}
> 
>   // Context: Any Thread.  If subclass overrides, be sure to call the base
>   // implementation.  Do not use (message_id < ST_MSG_FIRST_AVAILABLE)
>@@ -95,20 +93,16 @@ class SignalThread
> 
>  private:
>   enum State {
>-    kInit,            // Initialized, but not started
>-    kRunning,         // Started and doing work
>-    kReleasing,       // Same as running, but to be deleted when work is done
>-    kComplete,        // Work is done
>-    kStopping,        // Work is being interrupted
>+    kInit,       // Initialized, but not started
>+    kRunning,    // Started and doing work
>+    kReleasing,  // Same as running, but to be deleted when work is done
>+    kComplete,   // Work is done
>+    kStopping,   // Work is being interrupted
>   };
> 
>   class Worker : public Thread {
>    public:
>-    explicit Worker(SignalThread* parent, bool use_socket_server)
>-        : Thread(use_socket_server
>-                     ? SocketServer::CreateDefault()
>-                     : std::unique_ptr<SocketServer>(new NullSocketServer())),
>-          parent_(parent) {}
>+    explicit Worker(SignalThread* parent);
>     ~Worker() override;
>     void Run() override;
>     bool IsProcessingMessages() override;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sigslot.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sigslot.h
>index 318aca3cf3c..51ab9d758cf 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sigslot.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sigslot.h
>@@ -1,647 +1,14 @@
>-// sigslot.h: Signal/Slot classes
>-//
>-// Written by Sarah Thompson (sarah@telergy.com) 2002.
>-//
>-// License: Public domain. You are free to use this code however you like, with
>-// the proviso that the author takes on no responsibility or liability for any
>-// use.
>-//
>-// QUICK DOCUMENTATION
>-//
>-//        (see also the full documentation at http://sigslot.sourceforge.net/)
>-//
>-//    #define switches
>-//      SIGSLOT_PURE_ISO:
>-//        Define this to force ISO C++ compliance. This also disables all of
>-//        the thread safety support on platforms where it is available.
>-//
>-//      SIGSLOT_USE_POSIX_THREADS:
>-//        Force use of Posix threads when using a C++ compiler other than gcc
>-//        on a platform that supports Posix threads. (When using gcc, this is
>-//        the default - use SIGSLOT_PURE_ISO to disable this if necessary)
>-//
>-//      SIGSLOT_DEFAULT_MT_POLICY:
>-//        Where thread support is enabled, this defaults to
>-//        multi_threaded_global. Otherwise, the default is single_threaded.
>-//        #define this yourself to override the default. In pure ISO mode,
>-//        anything other than single_threaded will cause a compiler error.
>-//
>-//    PLATFORM NOTES
>-//
>-//      Win32:
>-//        On Win32, the WEBRTC_WIN symbol must be #defined. Most mainstream
>-//        compilers do this by default, but you may need to define it yourself
>-//        if your build environment is less standard. This causes the Win32
>-//        thread support to be compiled in and used automatically.
>-//
>-//      Unix/Linux/BSD, etc.:
>-//        If you're using gcc, it is assumed that you have Posix threads
>-//        available, so they are used automatically. You can override this (as
>-//        under Windows) with the SIGSLOT_PURE_ISO switch. If you're using
>-//        something other than gcc but still want to use Posix threads, you
>-//        need to #define SIGSLOT_USE_POSIX_THREADS.
>-//
>-//      ISO C++:
>-//        If none of the supported platforms are detected, or if
>-//        SIGSLOT_PURE_ISO is defined, all multithreading support is turned
>-//        off, along with any code that might cause a pure ISO C++ environment
>-//        to complain. Before you ask, gcc -ansi -pedantic won't compile this
>-//        library, but gcc -ansi is fine. Pedantic mode seems to throw a lot of
>-//        errors that aren't really there. If you feel like investigating this,
>-//        please contact the author.
>-//
>-//
>-//    THREADING MODES
>-//
>-//      single_threaded:
>-//        Your program is assumed to be single threaded from the point of view
>-//        of signal/slot usage (i.e. all objects using signals and slots are
>-//        created and destroyed from a single thread). Behaviour if objects are
>-//        destroyed concurrently is undefined (i.e. you'll get the occasional
>-//        segmentation fault/memory exception).
>-//
>-//      multi_threaded_global:
>-//        Your program is assumed to be multi threaded. Objects using signals
>-//        and slots can be safely created and destroyed from any thread, even
>-//        when connections exist. In multi_threaded_global mode, this is
>-//        achieved by a single global mutex (actually a critical section on
>-//        Windows because they are faster). This option uses less OS resources,
>-//        but results in more opportunities for contention, possibly resulting
>-//        in more context switches than are strictly necessary.
>-//
>-//      multi_threaded_local:
>-//        Behaviour in this mode is essentially the same as
>-//        multi_threaded_global, except that each signal, and each object that
>-//        inherits has_slots, all have their own mutex/critical section. In
>-//        practice, this means that mutex collisions (and hence context
>-//        switches) only happen if they are absolutely essential. However, on
>-//        some platforms, creating a lot of mutexes can slow down the whole OS,
>-//        so use this option with care.
>-//
>-//    USING THE LIBRARY
>-//
>-//      See the full documentation at http://sigslot.sourceforge.net/
>-//
>-// Libjingle specific:
>-//
>-// This file has been modified such that has_slots and signalx do not have to be
>-// using the same threading requirements. E.g. it is possible to connect a
>-// has_slots<single_threaded> and signal0<multi_threaded_local> or
>-// has_slots<multi_threaded_local> and signal0<single_threaded>.
>-// If has_slots is single threaded the user must ensure that it is not trying
>-// to connect or disconnect to signalx concurrently or data race may occur.
>-// If signalx is single threaded the user must ensure that disconnect, connect
>-// or signal is not happening concurrently or data race may occur.
>-
>-#ifndef RTC_BASE_SIGSLOT_H_
>-#define RTC_BASE_SIGSLOT_H_
>-
>-#include <stdlib.h>
>-#include <cstring>
>-#include <list>
>-#include <set>
>-
>-// On our copy of sigslot.h, we set single threading as default.
>-#define SIGSLOT_DEFAULT_MT_POLICY single_threaded
>-
>-#if defined(SIGSLOT_PURE_ISO) ||                   \
>-    (!defined(WEBRTC_WIN) && !defined(__GNUG__) && \
>-     !defined(SIGSLOT_USE_POSIX_THREADS))
>-#define _SIGSLOT_SINGLE_THREADED
>-#elif defined(WEBRTC_WIN)
>-#define _SIGSLOT_HAS_WIN32_THREADS
>-#if !defined(WIN32_LEAN_AND_MEAN)
>-#define WIN32_LEAN_AND_MEAN
>-#endif
>-#include "rtc_base/win32.h"
>-#elif defined(__GNUG__) || defined(SIGSLOT_USE_POSIX_THREADS)
>-#define _SIGSLOT_HAS_POSIX_THREADS
>-#include <pthread.h>
>-#else
>-#define _SIGSLOT_SINGLE_THREADED
>-#endif
>-
>-#ifndef SIGSLOT_DEFAULT_MT_POLICY
>-#ifdef _SIGSLOT_SINGLE_THREADED
>-#define SIGSLOT_DEFAULT_MT_POLICY single_threaded
>-#else
>-#define SIGSLOT_DEFAULT_MT_POLICY multi_threaded_local
>-#endif
>-#endif
>-
>-// TODO: change this namespace to rtc?
>-namespace sigslot {
>-
>-class single_threaded {
>- public:
>-  void lock() {}
>-  void unlock() {}
>-};
>-
>-#ifdef _SIGSLOT_HAS_WIN32_THREADS
>-// The multi threading policies only get compiled in if they are enabled.
>-class multi_threaded_global {
>- public:
>-  multi_threaded_global() {
>-    static bool isinitialised = false;
>-
>-    if (!isinitialised) {
>-      InitializeCriticalSection(get_critsec());
>-      isinitialised = true;
>-    }
>-  }
>-
>-  void lock() { EnterCriticalSection(get_critsec()); }
>-
>-  void unlock() { LeaveCriticalSection(get_critsec()); }
>-
>- private:
>-  CRITICAL_SECTION* get_critsec() {
>-    static CRITICAL_SECTION g_critsec;
>-    return &g_critsec;
>-  }
>-};
>-
>-class multi_threaded_local {
>- public:
>-  multi_threaded_local() { InitializeCriticalSection(&m_critsec); }
>-
>-  multi_threaded_local(const multi_threaded_local&) {
>-    InitializeCriticalSection(&m_critsec);
>-  }
>-
>-  ~multi_threaded_local() { DeleteCriticalSection(&m_critsec); }
>-
>-  void lock() { EnterCriticalSection(&m_critsec); }
>-
>-  void unlock() { LeaveCriticalSection(&m_critsec); }
>-
>- private:
>-  CRITICAL_SECTION m_critsec;
>-};
>-#endif  // _SIGSLOT_HAS_WIN32_THREADS
>-
>-#ifdef _SIGSLOT_HAS_POSIX_THREADS
>-// The multi threading policies only get compiled in if they are enabled.
>-class multi_threaded_global {
>- public:
>-  void lock() { pthread_mutex_lock(get_mutex()); }
>-  void unlock() { pthread_mutex_unlock(get_mutex()); }
>-
>- private:
>-  static pthread_mutex_t* get_mutex();
>-};
>-
>-class multi_threaded_local {
>- public:
>-  multi_threaded_local() { pthread_mutex_init(&m_mutex, nullptr); }
>-  multi_threaded_local(const multi_threaded_local&) {
>-    pthread_mutex_init(&m_mutex, nullptr);
>-  }
>-  ~multi_threaded_local() { pthread_mutex_destroy(&m_mutex); }
>-  void lock() { pthread_mutex_lock(&m_mutex); }
>-  void unlock() { pthread_mutex_unlock(&m_mutex); }
>-
>- private:
>-  pthread_mutex_t m_mutex;
>-};
>-#endif  // _SIGSLOT_HAS_POSIX_THREADS
>-
>-template <class mt_policy>
>-class lock_block {
>- public:
>-  mt_policy* m_mutex;
>-
>-  lock_block(mt_policy* mtx) : m_mutex(mtx) { m_mutex->lock(); }
>-
>-  ~lock_block() { m_mutex->unlock(); }
>-};
>-
>-class _signal_base_interface;
>-
>-class has_slots_interface {
>- private:
>-  typedef void (*signal_connect_t)(has_slots_interface* self,
>-                                   _signal_base_interface* sender);
>-  typedef void (*signal_disconnect_t)(has_slots_interface* self,
>-                                      _signal_base_interface* sender);
>-  typedef void (*disconnect_all_t)(has_slots_interface* self);
>-
>-  const signal_connect_t m_signal_connect;
>-  const signal_disconnect_t m_signal_disconnect;
>-  const disconnect_all_t m_disconnect_all;
>-
>- protected:
>-  has_slots_interface(signal_connect_t conn,
>-                      signal_disconnect_t disc,
>-                      disconnect_all_t disc_all)
>-      : m_signal_connect(conn),
>-        m_signal_disconnect(disc),
>-        m_disconnect_all(disc_all) {}
>-
>-  // Doesn't really need to be virtual, but is for backwards compatibility
>-  // (it was virtual in a previous version of sigslot).
>-  virtual ~has_slots_interface() {}
>-
>- public:
>-  void signal_connect(_signal_base_interface* sender) {
>-    m_signal_connect(this, sender);
>-  }
>-
>-  void signal_disconnect(_signal_base_interface* sender) {
>-    m_signal_disconnect(this, sender);
>-  }
>-
>-  void disconnect_all() { m_disconnect_all(this); }
>-};
>-
>-class _signal_base_interface {
>- private:
>-  typedef void (*slot_disconnect_t)(_signal_base_interface* self,
>-                                    has_slots_interface* pslot);
>-  typedef void (*slot_duplicate_t)(_signal_base_interface* self,
>-                                   const has_slots_interface* poldslot,
>-                                   has_slots_interface* pnewslot);
>-
>-  const slot_disconnect_t m_slot_disconnect;
>-  const slot_duplicate_t m_slot_duplicate;
>-
>- protected:
>-  _signal_base_interface(slot_disconnect_t disc, slot_duplicate_t dupl)
>-      : m_slot_disconnect(disc), m_slot_duplicate(dupl) {}
>-
>-  ~_signal_base_interface() {}
>-
>- public:
>-  void slot_disconnect(has_slots_interface* pslot) {
>-    m_slot_disconnect(this, pslot);
>-  }
>-
>-  void slot_duplicate(const has_slots_interface* poldslot,
>-                      has_slots_interface* pnewslot) {
>-    m_slot_duplicate(this, poldslot, pnewslot);
>-  }
>-};
>-
>-class _opaque_connection {
>- private:
>-  typedef void (*emit_t)(const _opaque_connection*);
>-  template <typename FromT, typename ToT>
>-  union union_caster {
>-    FromT from;
>-    ToT to;
>-  };
>-
>-  emit_t pemit;
>-  has_slots_interface* pdest;
>-  // Pointers to member functions may be up to 16 bytes for virtual classes,
>-  // so make sure we have enough space to store it.
>-  unsigned char pmethod[16];
>-
>- public:
>-  template <typename DestT, typename... Args>
>-  _opaque_connection(DestT* pd, void (DestT::*pm)(Args...)) : pdest(pd) {
>-    typedef void (DestT::*pm_t)(Args...);
>-    static_assert(sizeof(pm_t) <= sizeof(pmethod),
>-                  "Size of slot function pointer too large.");
>-
>-    std::memcpy(pmethod, &pm, sizeof(pm_t));
>-
>-    typedef void (*em_t)(const _opaque_connection* self, Args...);
>-    union_caster<em_t, emit_t> caster2;
>-    caster2.from = &_opaque_connection::emitter<DestT, Args...>;
>-    pemit = caster2.to;
>-  }
>-
>-  has_slots_interface* getdest() const { return pdest; }
>-
>-  _opaque_connection duplicate(has_slots_interface* newtarget) const {
>-    _opaque_connection res = *this;
>-    res.pdest = newtarget;
>-    return res;
>-  }
>-
>-  // Just calls the stored "emitter" function pointer stored at construction
>-  // time.
>-  template <typename... Args>
>-  void emit(Args... args) const {
>-    typedef void (*em_t)(const _opaque_connection*, Args...);
>-    union_caster<emit_t, em_t> caster;
>-    caster.from = pemit;
>-    (caster.to)(this, args...);
>-  }
>-
>- private:
>-  template <typename DestT, typename... Args>
>-  static void emitter(const _opaque_connection* self, Args... args) {
>-    typedef void (DestT::*pm_t)(Args...);
>-    pm_t pm;
>-    std::memcpy(&pm, self->pmethod, sizeof(pm_t));
>-    (static_cast<DestT*>(self->pdest)->*(pm))(args...);
>-  }
>-};
>-
>-template <class mt_policy>
>-class _signal_base : public _signal_base_interface, public mt_policy {
>- protected:
>-  typedef std::list<_opaque_connection> connections_list;
>-
>-  _signal_base()
>-      : _signal_base_interface(&_signal_base::do_slot_disconnect,
>-                               &_signal_base::do_slot_duplicate),
>-        m_current_iterator(m_connected_slots.end()) {}
>-
>-  ~_signal_base() { disconnect_all(); }
>-
>- private:
>-  _signal_base& operator=(_signal_base const& that);
>-
>- public:
>-  _signal_base(const _signal_base& o)
>-      : _signal_base_interface(&_signal_base::do_slot_disconnect,
>-                               &_signal_base::do_slot_duplicate),
>-        m_current_iterator(m_connected_slots.end()) {
>-    lock_block<mt_policy> lock(this);
>-    for (const auto& connection : o.m_connected_slots) {
>-      connection.getdest()->signal_connect(this);
>-      m_connected_slots.push_back(connection);
>-    }
>-  }
>-
>-  bool is_empty() {
>-    lock_block<mt_policy> lock(this);
>-    return m_connected_slots.empty();
>-  }
>-
>-  void disconnect_all() {
>-    lock_block<mt_policy> lock(this);
>-
>-    while (!m_connected_slots.empty()) {
>-      has_slots_interface* pdest = m_connected_slots.front().getdest();
>-      m_connected_slots.pop_front();
>-      pdest->signal_disconnect(static_cast<_signal_base_interface*>(this));
>-    }
>-    // If disconnect_all is called while the signal is firing, advance the
>-    // current slot iterator to the end to avoid an invalidated iterator from
>-    // being dereferenced.
>-    m_current_iterator = m_connected_slots.end();
>-  }
>-
>-#if !defined(NDEBUG)
>-  bool connected(has_slots_interface* pclass) {
>-    lock_block<mt_policy> lock(this);
>-    connections_list::const_iterator it = m_connected_slots.begin();
>-    connections_list::const_iterator itEnd = m_connected_slots.end();
>-    while (it != itEnd) {
>-      if (it->getdest() == pclass)
>-        return true;
>-      ++it;
>-    }
>-    return false;
>-  }
>-#endif
>-
>-  void disconnect(has_slots_interface* pclass) {
>-    lock_block<mt_policy> lock(this);
>-    connections_list::iterator it = m_connected_slots.begin();
>-    connections_list::iterator itEnd = m_connected_slots.end();
>-
>-    while (it != itEnd) {
>-      if (it->getdest() == pclass) {
>-        // If we're currently using this iterator because the signal is firing,
>-        // advance it to avoid it being invalidated.
>-        if (m_current_iterator == it) {
>-          m_current_iterator = m_connected_slots.erase(it);
>-        } else {
>-          m_connected_slots.erase(it);
>-        }
>-        pclass->signal_disconnect(static_cast<_signal_base_interface*>(this));
>-        return;
>-      }
>-      ++it;
>-    }
>-  }
>-
>- private:
>-  static void do_slot_disconnect(_signal_base_interface* p,
>-                                 has_slots_interface* pslot) {
>-    _signal_base* const self = static_cast<_signal_base*>(p);
>-    lock_block<mt_policy> lock(self);
>-    connections_list::iterator it = self->m_connected_slots.begin();
>-    connections_list::iterator itEnd = self->m_connected_slots.end();
>-
>-    while (it != itEnd) {
>-      connections_list::iterator itNext = it;
>-      ++itNext;
>-
>-      if (it->getdest() == pslot) {
>-        // If we're currently using this iterator because the signal is firing,
>-        // advance it to avoid it being invalidated.
>-        if (self->m_current_iterator == it) {
>-          self->m_current_iterator = self->m_connected_slots.erase(it);
>-        } else {
>-          self->m_connected_slots.erase(it);
>-        }
>-      }
>-
>-      it = itNext;
>-    }
>-  }
>-
>-  static void do_slot_duplicate(_signal_base_interface* p,
>-                                const has_slots_interface* oldtarget,
>-                                has_slots_interface* newtarget) {
>-    _signal_base* const self = static_cast<_signal_base*>(p);
>-    lock_block<mt_policy> lock(self);
>-    connections_list::iterator it = self->m_connected_slots.begin();
>-    connections_list::iterator itEnd = self->m_connected_slots.end();
>-
>-    while (it != itEnd) {
>-      if (it->getdest() == oldtarget) {
>-        self->m_connected_slots.push_back(it->duplicate(newtarget));
>-      }
>-
>-      ++it;
>-    }
>-  }
>-
>- protected:
>-  connections_list m_connected_slots;
>-
>-  // Used to handle a slot being disconnected while a signal is
>-  // firing (iterating m_connected_slots).
>-  connections_list::iterator m_current_iterator;
>-  bool m_erase_current_iterator = false;
>-};
>-
>-template <class mt_policy = SIGSLOT_DEFAULT_MT_POLICY>
>-class has_slots : public has_slots_interface, public mt_policy {
>- private:
>-  typedef std::set<_signal_base_interface*> sender_set;
>-  typedef sender_set::const_iterator const_iterator;
>-
>- public:
>-  has_slots()
>-      : has_slots_interface(&has_slots::do_signal_connect,
>-                            &has_slots::do_signal_disconnect,
>-                            &has_slots::do_disconnect_all) {}
>-
>-  has_slots(has_slots const& o)
>-      : has_slots_interface(&has_slots::do_signal_connect,
>-                            &has_slots::do_signal_disconnect,
>-                            &has_slots::do_disconnect_all) {
>-    lock_block<mt_policy> lock(this);
>-    for (auto* sender : o.m_senders) {
>-      sender->slot_duplicate(&o, this);
>-      m_senders.insert(sender);
>-    }
>-  }
>-
>-  ~has_slots() { this->disconnect_all(); }
>-
>- private:
>-  has_slots& operator=(has_slots const&);
>-
>-  static void do_signal_connect(has_slots_interface* p,
>-                                _signal_base_interface* sender) {
>-    has_slots* const self = static_cast<has_slots*>(p);
>-    lock_block<mt_policy> lock(self);
>-    self->m_senders.insert(sender);
>-  }
>-
>-  static void do_signal_disconnect(has_slots_interface* p,
>-                                   _signal_base_interface* sender) {
>-    has_slots* const self = static_cast<has_slots*>(p);
>-    lock_block<mt_policy> lock(self);
>-    self->m_senders.erase(sender);
>-  }
>-
>-  static void do_disconnect_all(has_slots_interface* p) {
>-    has_slots* const self = static_cast<has_slots*>(p);
>-    lock_block<mt_policy> lock(self);
>-    while (!self->m_senders.empty()) {
>-      std::set<_signal_base_interface*> senders;
>-      senders.swap(self->m_senders);
>-      const_iterator it = senders.begin();
>-      const_iterator itEnd = senders.end();
>-
>-      while (it != itEnd) {
>-        _signal_base_interface* s = *it;
>-        ++it;
>-        s->slot_disconnect(p);
>-      }
>-    }
>-  }
>-
>- private:
>-  sender_set m_senders;
>-};
>-
>-template <class mt_policy, typename... Args>
>-class signal_with_thread_policy : public _signal_base<mt_policy> {
>- private:
>-  typedef _signal_base<mt_policy> base;
>-
>- protected:
>-  typedef typename base::connections_list connections_list;
>-
>- public:
>-  signal_with_thread_policy() {}
>-
>-  template <class desttype>
>-  void connect(desttype* pclass, void (desttype::*pmemfun)(Args...)) {
>-    lock_block<mt_policy> lock(this);
>-    this->m_connected_slots.push_back(_opaque_connection(pclass, pmemfun));
>-    pclass->signal_connect(static_cast<_signal_base_interface*>(this));
>-  }
>-
>-  void emit(Args... args) {
>-    lock_block<mt_policy> lock(this);
>-    this->m_current_iterator = this->m_connected_slots.begin();
>-    while (this->m_current_iterator != this->m_connected_slots.end()) {
>-      _opaque_connection const& conn = *this->m_current_iterator;
>-      ++(this->m_current_iterator);
>-      conn.emit<Args...>(args...);
>-    }
>-  }
>-
>-  void operator()(Args... args) { emit(args...); }
>-};
>-
>-// Alias with default thread policy. Needed because both default arguments
>-// and variadic template arguments must go at the end of the list, so we
>-// can't have both at once.
>-template <typename... Args>
>-using signal = signal_with_thread_policy<SIGSLOT_DEFAULT_MT_POLICY, Args...>;
>-
>-// The previous verion of sigslot didn't use variadic templates, so you would
>-// need to write "sigslot::signal2<Arg1, Arg2>", for example.
>-// Now you can just write "sigslot::signal<Arg1, Arg2>", but these aliases
>-// exist for backwards compatibility.
>-template <typename mt_policy = SIGSLOT_DEFAULT_MT_POLICY>
>-using signal0 = signal_with_thread_policy<mt_policy>;
>-
>-template <typename A1, typename mt_policy = SIGSLOT_DEFAULT_MT_POLICY>
>-using signal1 = signal_with_thread_policy<mt_policy, A1>;
>-
>-template <typename A1,
>-          typename A2,
>-          typename mt_policy = SIGSLOT_DEFAULT_MT_POLICY>
>-using signal2 = signal_with_thread_policy<mt_policy, A1, A2>;
>-
>-template <typename A1,
>-          typename A2,
>-          typename A3,
>-          typename mt_policy = SIGSLOT_DEFAULT_MT_POLICY>
>-using signal3 = signal_with_thread_policy<mt_policy, A1, A2, A3>;
>-
>-template <typename A1,
>-          typename A2,
>-          typename A3,
>-          typename A4,
>-          typename mt_policy = SIGSLOT_DEFAULT_MT_POLICY>
>-using signal4 = signal_with_thread_policy<mt_policy, A1, A2, A3, A4>;
>-
>-template <typename A1,
>-          typename A2,
>-          typename A3,
>-          typename A4,
>-          typename A5,
>-          typename mt_policy = SIGSLOT_DEFAULT_MT_POLICY>
>-using signal5 = signal_with_thread_policy<mt_policy, A1, A2, A3, A4, A5>;
>-
>-template <typename A1,
>-          typename A2,
>-          typename A3,
>-          typename A4,
>-          typename A5,
>-          typename A6,
>-          typename mt_policy = SIGSLOT_DEFAULT_MT_POLICY>
>-using signal6 = signal_with_thread_policy<mt_policy, A1, A2, A3, A4, A5, A6>;
>-
>-template <typename A1,
>-          typename A2,
>-          typename A3,
>-          typename A4,
>-          typename A5,
>-          typename A6,
>-          typename A7,
>-          typename mt_policy = SIGSLOT_DEFAULT_MT_POLICY>
>-using signal7 =
>-    signal_with_thread_policy<mt_policy, A1, A2, A3, A4, A5, A6, A7>;
>-
>-template <typename A1,
>-          typename A2,
>-          typename A3,
>-          typename A4,
>-          typename A5,
>-          typename A6,
>-          typename A7,
>-          typename A8,
>-          typename mt_policy = SIGSLOT_DEFAULT_MT_POLICY>
>-using signal8 =
>-    signal_with_thread_policy<mt_policy, A1, A2, A3, A4, A5, A6, A7, A8>;
>-
>-}  // namespace sigslot
>-
>-#endif  // RTC_BASE_SIGSLOT_H_
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+// This is a temporary header file to move origin header from this location
>+// to specified in include. Don't use this header directly, use included one
>+// instead.
>+#include "rtc_base/third_party/sigslot/sigslot.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sigslot_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sigslot_unittest.cc
>index 234bf4547f7..2f3de40ce3e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sigslot_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sigslot_unittest.cc
>@@ -8,7 +8,7 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "rtc_base/sigslot.h"
>+#include "rtc_base/third_party/sigslot/sigslot.h"
> #include "rtc_base/gunit.h"
> #include "rtc_base/sigslotrepeater.h"
> 
>@@ -28,33 +28,32 @@ class SigslotDefault : public testing::Test, public sigslot::has_slots<> {
>   sigslot::signal0<> signal_;
> };
> 
>-template<class slot_policy = sigslot::single_threaded,
>-         class signal_policy = sigslot::single_threaded>
>+template <class slot_policy = sigslot::single_threaded,
>+          class signal_policy = sigslot::single_threaded>
> class SigslotReceiver : public sigslot::has_slots<slot_policy> {
>  public:
>   SigslotReceiver() : signal_(nullptr), signal_count_(0) {}
>-  ~SigslotReceiver() {
>-  }
>+  ~SigslotReceiver() {}
> 
>   // Provide copy constructor so that tests can exercise the has_slots copy
>   // constructor.
>   SigslotReceiver(const SigslotReceiver&) = default;
> 
>   void Connect(sigslot::signal0<signal_policy>* signal) {
>-    if (!signal) return;
>+    if (!signal)
>+      return;
>     Disconnect();
>     signal_ = signal;
>     signal->connect(this,
>                     &SigslotReceiver<slot_policy, signal_policy>::OnSignal);
>   }
>   void Disconnect() {
>-    if (!signal_) return;
>+    if (!signal_)
>+      return;
>     signal_->disconnect(this);
>     signal_ = nullptr;
>   }
>-  void OnSignal() {
>-    ++signal_count_;
>-  }
>+  void OnSignal() { ++signal_count_; }
>   int signal_count() { return signal_count_; }
> 
>  private:
>@@ -62,8 +61,8 @@ class SigslotReceiver : public sigslot::has_slots<slot_policy> {
>   int signal_count_;
> };
> 
>-template<class slot_policy = sigslot::single_threaded,
>-         class mt_signal_policy = sigslot::multi_threaded_local>
>+template <class slot_policy = sigslot::single_threaded,
>+          class mt_signal_policy = sigslot::multi_threaded_local>
> class SigslotSlotTest : public testing::Test {
>  protected:
>   SigslotSlotTest() {
>@@ -71,12 +70,8 @@ class SigslotSlotTest : public testing::Test {
>     TemplateIsMT(&mt_policy);
>   }
> 
>-  virtual void SetUp() {
>-    Connect();
>-  }
>-  virtual void TearDown() {
>-    Disconnect();
>-  }
>+  virtual void SetUp() { Connect(); }
>+  virtual void TearDown() { Disconnect(); }
> 
>   void Disconnect() {
>     st_receiver_.Disconnect();
>@@ -99,12 +94,12 @@ class SigslotSlotTest : public testing::Test {
> 
> typedef SigslotSlotTest<> SigslotSTSlotTest;
> typedef SigslotSlotTest<sigslot::multi_threaded_local,
>-                        sigslot::multi_threaded_local> SigslotMTSlotTest;
>+                        sigslot::multi_threaded_local>
>+    SigslotMTSlotTest;
> 
> class multi_threaded_local_fake : public sigslot::multi_threaded_local {
>  public:
>-  multi_threaded_local_fake() : lock_count_(0), unlock_count_(0) {
>-  }
>+  multi_threaded_local_fake() : lock_count_(0), unlock_count_(0) {}
> 
>   void lock() { ++lock_count_; }
>   void unlock() { ++unlock_count_; }
>@@ -118,8 +113,8 @@ class multi_threaded_local_fake : public sigslot::multi_threaded_local {
>   int unlock_count_;
> };
> 
>-typedef SigslotSlotTest<multi_threaded_local_fake,
>-                        multi_threaded_local_fake> SigslotMTLockBase;
>+typedef SigslotSlotTest<multi_threaded_local_fake, multi_threaded_local_fake>
>+    SigslotMTLockBase;
> 
> class SigslotMTLockTest : public SigslotMTLockBase {
>  protected:
>@@ -329,9 +324,7 @@ class Disconnector2 : public sigslot::has_slots<> {
>   }
> 
>  private:
>-  void Disconnect() {
>-    signal_->disconnect_all();
>-  }
>+  void Disconnect() { signal_->disconnect_all(); }
> 
>   sigslot::signal<>* signal_;
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sigslotrepeater.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sigslotrepeater.h
>index ca4485457e6..724e624a7c9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sigslotrepeater.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sigslotrepeater.h
>@@ -21,7 +21,7 @@
> // TODO(deadbeef): Actually use this, after we decide on some style points on
> // using signals, so it doesn't get deleted again.
> 
>-#include "rtc_base/sigslot.h"
>+#include "rtc_base/third_party/sigslot/sigslot.h"
> 
> namespace sigslot {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sigslottester.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sigslottester.h
>index 04c6302eec1..73f4365a346 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sigslottester.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sigslottester.h
>@@ -18,7 +18,6 @@
> // To generate sigslottester.h from sigslottester.h.pump, execute:
> // /home/build/google3/third_party/gtest/scripts/pump.py sigslottester.h.pump
> 
>-
> // SigslotTester(s) are utility classes to check if signals owned by an
> // object are being invoked at the right time and with the right arguments.
> // They are meant to be used in tests. Tests must provide "capture" pointers
>@@ -40,7 +39,7 @@
> //   /* See unit-tests for more examples */
> 
> #include "rtc_base/constructormagic.h"
>-#include "rtc_base/sigslot.h"
>+#include "rtc_base/third_party/sigslot/sigslot.h"
> 
> namespace rtc {
> 
>@@ -70,10 +69,8 @@ class SigslotTester0 : public sigslot::has_slots<> {
> template <class A1, class C1>
> class SigslotTester1 : public sigslot::has_slots<> {
>  public:
>-  SigslotTester1(sigslot::signal1<A1>* signal,
>-                C1* capture1)
>-      : callback_count_(0),
>-      capture1_(capture1) {
>+  SigslotTester1(sigslot::signal1<A1>* signal, C1* capture1)
>+      : callback_count_(0), capture1_(capture1) {
>     signal->connect(this, &SigslotTester1::OnSignalCallback);
>   }
> 
>@@ -94,10 +91,8 @@ class SigslotTester1 : public sigslot::has_slots<> {
> template <class A1, class A2, class C1, class C2>
> class SigslotTester2 : public sigslot::has_slots<> {
>  public:
>-  SigslotTester2(sigslot::signal2<A1, A2>* signal,
>-                C1* capture1, C2* capture2)
>-      : callback_count_(0),
>-      capture1_(capture1), capture2_(capture2) {
>+  SigslotTester2(sigslot::signal2<A1, A2>* signal, C1* capture1, C2* capture2)
>+      : callback_count_(0), capture1_(capture1), capture2_(capture2) {
>     signal->connect(this, &SigslotTester2::OnSignalCallback);
>   }
> 
>@@ -121,9 +116,13 @@ template <class A1, class A2, class A3, class C1, class C2, class C3>
> class SigslotTester3 : public sigslot::has_slots<> {
>  public:
>   SigslotTester3(sigslot::signal3<A1, A2, A3>* signal,
>-                C1* capture1, C2* capture2, C3* capture3)
>+                 C1* capture1,
>+                 C2* capture2,
>+                 C3* capture3)
>       : callback_count_(0),
>-      capture1_(capture1), capture2_(capture2), capture3_(capture3) {
>+        capture1_(capture1),
>+        capture2_(capture2),
>+        capture3_(capture3) {
>     signal->connect(this, &SigslotTester3::OnSignalCallback);
>   }
> 
>@@ -145,15 +144,26 @@ class SigslotTester3 : public sigslot::has_slots<> {
>   RTC_DISALLOW_COPY_AND_ASSIGN(SigslotTester3);
> };
> 
>-template <class A1, class A2, class A3, class A4, class C1, class C2, class C3,
>-    class C4>
>+template <class A1,
>+          class A2,
>+          class A3,
>+          class A4,
>+          class C1,
>+          class C2,
>+          class C3,
>+          class C4>
> class SigslotTester4 : public sigslot::has_slots<> {
>  public:
>   SigslotTester4(sigslot::signal4<A1, A2, A3, A4>* signal,
>-                C1* capture1, C2* capture2, C3* capture3, C4* capture4)
>+                 C1* capture1,
>+                 C2* capture2,
>+                 C3* capture3,
>+                 C4* capture4)
>       : callback_count_(0),
>-      capture1_(capture1), capture2_(capture2), capture3_(capture3),
>-          capture4_(capture4) {
>+        capture1_(capture1),
>+        capture2_(capture2),
>+        capture3_(capture3),
>+        capture4_(capture4) {
>     signal->connect(this, &SigslotTester4::OnSignalCallback);
>   }
> 
>@@ -177,16 +187,30 @@ class SigslotTester4 : public sigslot::has_slots<> {
>   RTC_DISALLOW_COPY_AND_ASSIGN(SigslotTester4);
> };
> 
>-template <class A1, class A2, class A3, class A4, class A5, class C1, class C2,
>-    class C3, class C4, class C5>
>+template <class A1,
>+          class A2,
>+          class A3,
>+          class A4,
>+          class A5,
>+          class C1,
>+          class C2,
>+          class C3,
>+          class C4,
>+          class C5>
> class SigslotTester5 : public sigslot::has_slots<> {
>  public:
>   SigslotTester5(sigslot::signal5<A1, A2, A3, A4, A5>* signal,
>-                C1* capture1, C2* capture2, C3* capture3, C4* capture4,
>-                    C5* capture5)
>+                 C1* capture1,
>+                 C2* capture2,
>+                 C3* capture3,
>+                 C4* capture4,
>+                 C5* capture5)
>       : callback_count_(0),
>-      capture1_(capture1), capture2_(capture2), capture3_(capture3),
>-          capture4_(capture4), capture5_(capture5) {
>+        capture1_(capture1),
>+        capture2_(capture2),
>+        capture3_(capture3),
>+        capture4_(capture4),
>+        capture5_(capture5) {
>     signal->connect(this, &SigslotTester5::OnSignalCallback);
>   }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sigslottester.h.pump b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sigslottester.h.pump
>old mode 100644
>new mode 100755
>index 1029a0f4a74..0c7c693c61c
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sigslottester.h.pump
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sigslottester.h.pump
>@@ -36,7 +36,7 @@
> //   /* See unit-tests for more examples */
> 
> #include "rtc_base/constructormagic.h"
>-#include "rtc_base/sigslot.h"
>+#include "rtc_base/third_party/sigslot/sigslot.h"
> 
> namespace rtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sigslottester_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sigslottester_unittest.cc
>index c8e87e5a86c..8fde268d2d3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sigslottester_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sigslottester_unittest.cc
>@@ -13,7 +13,7 @@
> #include <string>
> 
> #include "rtc_base/gunit.h"
>-#include "rtc_base/sigslot.h"
>+#include "rtc_base/third_party/sigslot/sigslot.h"
> 
> namespace rtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socket.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socket.cc
>new file mode 100644
>index 00000000000..a9749a473ee
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socket.cc
>@@ -0,0 +1,27 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "rtc_base/socket.h"
>+
>+namespace rtc {
>+
>+PacketInfo::PacketInfo() = default;
>+PacketInfo::PacketInfo(const PacketInfo& info) = default;
>+PacketInfo::~PacketInfo() = default;
>+
>+SentPacket::SentPacket() = default;
>+SentPacket::SentPacket(int64_t packet_id, int64_t send_time_ms)
>+    : packet_id(packet_id), send_time_ms(send_time_ms) {}
>+SentPacket::SentPacket(int64_t packet_id,
>+                       int64_t send_time_ms,
>+                       const rtc::PacketInfo& info)
>+    : packet_id(packet_id), send_time_ms(send_time_ms), info(info) {}
>+
>+}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socket.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socket.h
>index c78963b7688..b8290bb139b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socket.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socket.h
>@@ -14,10 +14,10 @@
> #include <errno.h>
> 
> #if defined(WEBRTC_POSIX)
>-#include <sys/types.h>
>-#include <sys/socket.h>
> #include <arpa/inet.h>
> #include <netinet/in.h>
>+#include <sys/socket.h>
>+#include <sys/types.h>
> #define SOCKET_EACCES EACCES
> #endif
> 
>@@ -25,7 +25,7 @@
> #include "rtc_base/win32.h"
> #endif
> 
>-#include "rtc_base/basictypes.h"
>+#include "absl/types/optional.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/socketaddress.h"
> 
>@@ -108,7 +108,6 @@
> #define ESTALE WSAESTALE
> #undef EREMOTE
> #define EREMOTE WSAEREMOTE
>-#undef EACCES
> #define SOCKET_EACCES WSAEACCES
> #endif  // WEBRTC_WIN
> 
>@@ -124,13 +123,48 @@ inline bool IsBlockingError(int e) {
>   return (e == EWOULDBLOCK) || (e == EAGAIN) || (e == EINPROGRESS);
> }
> 
>-struct SentPacket {
>-  SentPacket() : packet_id(-1), send_time_ms(-1) {}
>-  SentPacket(int packet_id, int64_t send_time_ms)
>-      : packet_id(packet_id), send_time_ms(send_time_ms) {}
>+enum class PacketType {
>+  kUnknown,
>+  kData,
>+  kIceConnectivityCheck,
>+  kIceConnectivityCheckResponse,
>+  kStunMessage,
>+  kTurnMessage,
>+};
> 
>-  int packet_id;
>-  int64_t send_time_ms;
>+enum class PacketInfoProtocolType {
>+  kUnknown,
>+  kUdp,
>+  kTcp,
>+  kSsltcp,
>+  kTls,
>+};
>+
>+struct PacketInfo {
>+  PacketInfo();
>+  PacketInfo(const PacketInfo& info);
>+  ~PacketInfo();
>+
>+  PacketType packet_type = PacketType::kUnknown;
>+  PacketInfoProtocolType protocol = PacketInfoProtocolType::kUnknown;
>+  // A unique id assigned by the network manager, and absl::nullopt if not set.
>+  absl::optional<uint16_t> network_id;
>+  size_t packet_size_bytes = 0;
>+  size_t turn_overhead_bytes = 0;
>+  SocketAddress local_socket_address;
>+  SocketAddress remote_socket_address;
>+};
>+
>+struct SentPacket {
>+  SentPacket();
>+  SentPacket(int64_t packet_id, int64_t send_time_ms);
>+  SentPacket(int64_t packet_id,
>+             int64_t send_time_ms,
>+             const rtc::PacketInfo& info);
>+
>+  int64_t packet_id = -1;
>+  int64_t send_time_ms = -1;
>+  rtc::PacketInfo info;
> };
> 
> // General interface for the socket implementations of various networks.  The
>@@ -149,8 +183,8 @@ class Socket {
> 
>   virtual int Bind(const SocketAddress& addr) = 0;
>   virtual int Connect(const SocketAddress& addr) = 0;
>-  virtual int Send(const void *pv, size_t cb) = 0;
>-  virtual int SendTo(const void *pv, size_t cb, const SocketAddress& addr) = 0;
>+  virtual int Send(const void* pv, size_t cb) = 0;
>+  virtual int SendTo(const void* pv, size_t cb, const SocketAddress& addr) = 0;
>   // |timestamp| is in units of microseconds.
>   virtual int Recv(void* pv, size_t cb, int64_t* timestamp) = 0;
>   virtual int RecvFrom(void* pv,
>@@ -158,26 +192,22 @@ class Socket {
>                        SocketAddress* paddr,
>                        int64_t* timestamp) = 0;
>   virtual int Listen(int backlog) = 0;
>-  virtual Socket *Accept(SocketAddress *paddr) = 0;
>+  virtual Socket* Accept(SocketAddress* paddr) = 0;
>   virtual int Close() = 0;
>   virtual int GetError() const = 0;
>   virtual void SetError(int error) = 0;
>   inline bool IsBlocking() const { return IsBlockingError(GetError()); }
> 
>-  enum ConnState {
>-    CS_CLOSED,
>-    CS_CONNECTING,
>-    CS_CONNECTED
>-  };
>+  enum ConnState { CS_CLOSED, CS_CONNECTING, CS_CONNECTED };
>   virtual ConnState GetState() const = 0;
> 
>   enum Option {
>     OPT_DONTFRAGMENT,
>-    OPT_RCVBUF,      // receive buffer size
>-    OPT_SNDBUF,      // send buffer size
>-    OPT_NODELAY,     // whether Nagle algorithm is enabled
>-    OPT_IPV6_V6ONLY, // Whether the socket is IPv6 only.
>-    OPT_DSCP,        // DSCP code
>+    OPT_RCVBUF,                // receive buffer size
>+    OPT_SNDBUF,                // send buffer size
>+    OPT_NODELAY,               // whether Nagle algorithm is enabled
>+    OPT_IPV6_V6ONLY,           // Whether the socket is IPv6 only.
>+    OPT_DSCP,                  // DSCP code
>     OPT_RTP_SENDTIME_EXTN_ID,  // This is a non-traditional socket option param.
>                                // This is specific to libjingle and will be used
>                                // if SendTime option is needed at socket level.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socket_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socket_unittest.cc
>index a31cc021662..cbc5eb58b53 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socket_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socket_unittest.cc
>@@ -12,12 +12,12 @@
> 
> #include "rtc_base/socket_unittest.h"
> 
>+#include "absl/memory/memory.h"
> #include "rtc_base/arraysize.h"
> #include "rtc_base/asyncudpsocket.h"
> #include "rtc_base/buffer.h"
> #include "rtc_base/gunit.h"
> #include "rtc_base/nethelpers.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/socketserver.h"
> #include "rtc_base/testclient.h"
> #include "rtc_base/testutils.h"
>@@ -224,7 +224,7 @@ void SocketTest::ConnectInternal(const IPAddress& loopback) {
>       ss_->CreateAsyncSocket(loopback.family(), SOCK_STREAM));
>   sink.Monitor(client.get());
>   EXPECT_EQ(AsyncSocket::CS_CLOSED, client->GetState());
>-  EXPECT_PRED1(IsUnspecOrEmptyIP, client->GetLocalAddress().ipaddr());
>+  EXPECT_TRUE(IsUnspecOrEmptyIP(client->GetLocalAddress().ipaddr()));
> 
>   // Create server and listen.
>   std::unique_ptr<AsyncSocket> server(
>@@ -291,7 +291,7 @@ void SocketTest::ConnectWithDnsLookupInternal(const IPAddress& loopback,
>   dns_addr.SetIP(host);
>   EXPECT_EQ(0, client->Connect(dns_addr));
>   // TODO: Bind when doing DNS lookup.
>-  //EXPECT_NE(kEmptyAddr, client->GetLocalAddress());  // Implicit Bind
>+  // EXPECT_NE(kEmptyAddr, client->GetLocalAddress());  // Implicit Bind
> 
>   // Client is connecting, outcome not yet determined.
>   EXPECT_EQ(AsyncSocket::CS_CONNECTING, client->GetState());
>@@ -444,9 +444,8 @@ void SocketTest::ConnectWhileNotClosedInternal(const IPAddress& loopback) {
> 
>   // Try to connect again, to an unresolved hostname.
>   // Shouldn't break anything.
>-  EXPECT_EQ(SOCKET_ERROR,
>-            client->Connect(SocketAddress("localhost",
>-                                          server->GetLocalAddress().port())));
>+  EXPECT_EQ(SOCKET_ERROR, client->Connect(SocketAddress(
>+                              "localhost", server->GetLocalAddress().port())));
>   EXPECT_EQ(AsyncSocket::CS_CONNECTED, accepted->GetState());
>   EXPECT_EQ(AsyncSocket::CS_CONNECTED, client->GetState());
>   EXPECT_EQ(client->GetRemoteAddress(), server->GetLocalAddress());
>@@ -694,8 +693,9 @@ void SocketTest::SocketServerWaitInternal(const IPAddress& loopback) {
>   EXPECT_LT(0, accepted->Recv(buf, 1024, nullptr));
> }
> 
>-void SocketTest::TcpInternal(const IPAddress& loopback, size_t data_size,
>-    ptrdiff_t max_send_size) {
>+void SocketTest::TcpInternal(const IPAddress& loopback,
>+                             size_t data_size,
>+                             ptrdiff_t max_send_size) {
>   StreamSink sink;
>   SocketAddress accept_addr;
> 
>@@ -792,7 +792,7 @@ void SocketTest::TcpInternal(const IPAddress& loopback, size_t data_size,
>       }
>       if (recved_size >= 0) {
>         EXPECT_LE(static_cast<size_t>(recved_size),
>-            sent_size - recv_buffer.size());
>+                  sent_size - recv_buffer.size());
>         recv_buffer.AppendData(recved_data.data(), recved_size);
>       } else {
>         ASSERT_TRUE(receiver->IsBlocking());
>@@ -858,7 +858,8 @@ void SocketTest::SingleFlowControlCallbackInternal(const IPAddress& loopback) {
>   // Fill the socket buffer.
>   char buf[1024 * 16] = {0};
>   int sends = 0;
>-  while (++sends && accepted->Send(&buf, arraysize(buf)) != -1) {}
>+  while (++sends && accepted->Send(&buf, arraysize(buf)) != -1) {
>+  }
>   EXPECT_TRUE(accepted->IsBlocking());
> 
>   // Wait until data is available.
>@@ -892,8 +893,7 @@ void SocketTest::SingleFlowControlCallbackInternal(const IPAddress& loopback) {
> void SocketTest::UdpInternal(const IPAddress& loopback) {
>   SocketAddress empty = EmptySocketAddressWithFamily(loopback.family());
>   // Test basic bind and connect behavior.
>-  AsyncSocket* socket =
>-      ss_->CreateAsyncSocket(loopback.family(), SOCK_DGRAM);
>+  AsyncSocket* socket = ss_->CreateAsyncSocket(loopback.family(), SOCK_DGRAM);
>   EXPECT_EQ(AsyncSocket::CS_CLOSED, socket->GetState());
>   EXPECT_EQ(0, socket->Bind(SocketAddress(loopback, 0)));
>   SocketAddress addr1 = socket->GetLocalAddress();
>@@ -905,9 +905,9 @@ void SocketTest::UdpInternal(const IPAddress& loopback) {
> 
>   // Test send/receive behavior.
>   std::unique_ptr<TestClient> client1(
>-      new TestClient(WrapUnique(AsyncUDPSocket::Create(ss_, addr1))));
>+      new TestClient(absl::WrapUnique(AsyncUDPSocket::Create(ss_, addr1))));
>   std::unique_ptr<TestClient> client2(
>-      new TestClient(WrapUnique(AsyncUDPSocket::Create(ss_, empty))));
>+      new TestClient(absl::WrapUnique(AsyncUDPSocket::Create(ss_, empty))));
> 
>   SocketAddress addr2;
>   EXPECT_EQ(3, client2->SendTo("foo", 3, addr1));
>@@ -920,7 +920,7 @@ void SocketTest::UdpInternal(const IPAddress& loopback) {
>   // TODO: figure out what the intent is here
>   for (int i = 0; i < 10; ++i) {
>     client2.reset(
>-        new TestClient(WrapUnique(AsyncUDPSocket::Create(ss_, empty))));
>+        new TestClient(absl::WrapUnique(AsyncUDPSocket::Create(ss_, empty))));
> 
>     SocketAddress addr4;
>     EXPECT_EQ(3, client2->SendTo("foo", 3, addr1));
>@@ -941,13 +941,13 @@ void SocketTest::UdpReadyToSend(const IPAddress& loopback) {
>   // RFC 5737 - The blocks 192.0.2.0/24 (TEST-NET-1) ... are provided for use in
>   // documentation.
>   // RFC 3849 - 2001:DB8::/32 as a documentation-only prefix.
>-  std::string dest = (loopback.family() == AF_INET6) ?
>-      "2001:db8::1" : "192.0.2.0";
>+  std::string dest =
>+      (loopback.family() == AF_INET6) ? "2001:db8::1" : "192.0.2.0";
>   SocketAddress test_addr(dest, 2345);
> 
>   // Test send
>   std::unique_ptr<TestClient> client(
>-      new TestClient(WrapUnique(AsyncUDPSocket::Create(ss_, empty))));
>+      new TestClient(absl::WrapUnique(AsyncUDPSocket::Create(ss_, empty))));
>   int test_packet_size = 1200;
>   std::unique_ptr<char[]> test_packet(new char[test_packet_size]);
>   // Init the test packet just to avoid memcheck warning.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socket_unittest.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socket_unittest.h
>index 55df219eea2..db226c40d3a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socket_unittest.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socket_unittest.h
>@@ -21,9 +21,10 @@ namespace rtc {
> // socketserver, and call the SocketTest test methods.
> class SocketTest : public testing::Test {
>  protected:
>-  SocketTest() : kIPv4Loopback(INADDR_LOOPBACK),
>-                 kIPv6Loopback(in6addr_loopback),
>-                 ss_(nullptr) {}
>+  SocketTest()
>+      : kIPv4Loopback(INADDR_LOOPBACK),
>+        kIPv6Loopback(in6addr_loopback),
>+        ss_(nullptr) {}
>   void SetUp() override;
>   void TestConnectIPv4();
>   void TestConnectIPv6();
>@@ -65,8 +66,9 @@ class SocketTest : public testing::Test {
>   const IPAddress kIPv6Loopback;
> 
>  protected:
>-  void TcpInternal(const IPAddress& loopback, size_t data_size,
>-      ptrdiff_t max_send_size);
>+  void TcpInternal(const IPAddress& loopback,
>+                   size_t data_size,
>+                   ptrdiff_t max_send_size);
> 
>  private:
>   void ConnectInternal(const IPAddress& loopback);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketadapters.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketadapters.cc
>index 4cde6907824..8095894420c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketadapters.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketadapters.cc
>@@ -9,14 +9,13 @@
>  */
> 
> #if defined(_MSC_VER) && _MSC_VER < 1300
>-#pragma warning(disable:4786)
>+#pragma warning(disable : 4786)
> #endif
> 
>-#include <time.h>
> #include <errno.h>
>+#include <time.h>
> 
> #if defined(WEBRTC_WIN)
>-#define WIN32_LEAN_AND_MEAN
> #include <windows.h>
> #include <winsock2.h>
> #include <ws2tcpip.h>
>@@ -33,20 +32,23 @@
> #include "rtc_base/socketadapters.h"
> #include "rtc_base/stringencode.h"
> #include "rtc_base/stringutils.h"
>+#include "rtc_base/zero_memory.h"
> 
> namespace rtc {
> 
> BufferedReadAdapter::BufferedReadAdapter(AsyncSocket* socket, size_t size)
>-    : AsyncSocketAdapter(socket), buffer_size_(size),
>-      data_len_(0), buffering_(false) {
>+    : AsyncSocketAdapter(socket),
>+      buffer_size_(size),
>+      data_len_(0),
>+      buffering_(false) {
>   buffer_ = new char[buffer_size_];
> }
> 
> BufferedReadAdapter::~BufferedReadAdapter() {
>-  delete [] buffer_;
>+  delete[] buffer_;
> }
> 
>-int BufferedReadAdapter::Send(const void *pv, size_t cb) {
>+int BufferedReadAdapter::Send(const void* pv, size_t cb) {
>   if (buffering_) {
>     // TODO: Spoof error better; Signal Writeable
>     socket_->SetError(EWOULDBLOCK);
>@@ -70,7 +72,7 @@ int BufferedReadAdapter::Recv(void* pv, size_t cb, int64_t* timestamp) {
>     if (data_len_ > 0) {
>       memmove(buffer_, buffer_ + read, data_len_);
>     }
>-    pv = static_cast<char *>(pv) + read;
>+    pv = static_cast<char*>(pv) + read;
>     cb -= read;
>   }
> 
>@@ -95,7 +97,7 @@ void BufferedReadAdapter::BufferInput(bool on) {
>   buffering_ = on;
> }
> 
>-void BufferedReadAdapter::OnReadEvent(AsyncSocket * socket) {
>+void BufferedReadAdapter::OnReadEvent(AsyncSocket* socket) {
>   RTC_DCHECK(socket == socket_);
> 
>   if (!buffering_) {
>@@ -104,7 +106,7 @@ void BufferedReadAdapter::OnReadEvent(AsyncSocket * socket) {
>   }
> 
>   if (data_len_ >= buffer_size_) {
>-    RTC_LOG(INFO) << "Input buffer overflow";
>+    RTC_LOG(LS_ERROR) << "Input buffer overflow";
>     RTC_NOTREACHED();
>     data_len_ = 0;
>   }
>@@ -124,8 +126,7 @@ void BufferedReadAdapter::OnReadEvent(AsyncSocket * socket) {
> 
> AsyncProxyServerSocket::AsyncProxyServerSocket(AsyncSocket* socket,
>                                                size_t buffer_size)
>-    : BufferedReadAdapter(socket, buffer_size) {
>-}
>+    : BufferedReadAdapter(socket, buffer_size) {}
> 
> AsyncProxyServerSocket::~AsyncProxyServerSocket() = default;
> 
>@@ -172,8 +173,7 @@ static const uint8_t kSslServerHello[] = {
> };
> 
> AsyncSSLSocket::AsyncSSLSocket(AsyncSocket* socket)
>-    : BufferedReadAdapter(socket, 1024) {
>-}
>+    : BufferedReadAdapter(socket, 1024) {}
> 
> int AsyncSSLSocket::Connect(const SocketAddress& addr) {
>   // Begin buffering before we connect, so that there isn't a race condition
>@@ -182,7 +182,7 @@ int AsyncSSLSocket::Connect(const SocketAddress& addr) {
>   return BufferedReadAdapter::Connect(addr);
> }
> 
>-void AsyncSSLSocket::OnConnectEvent(AsyncSocket * socket) {
>+void AsyncSSLSocket::OnConnectEvent(AsyncSocket* socket) {
>   RTC_DCHECK(socket == socket_);
>   // TODO: we could buffer output too...
>   const int res = DirectSend(kSslClientHello, sizeof(kSslClientHello));
>@@ -214,7 +214,7 @@ void AsyncSSLSocket::ProcessInput(char* data, size_t* len) {
> }
> 
> AsyncSSLServerSocket::AsyncSSLServerSocket(AsyncSocket* socket)
>-     : BufferedReadAdapter(socket, 1024) {
>+    : BufferedReadAdapter(socket, 1024) {
>   BufferInput(true);
> }
> 
>@@ -249,10 +249,14 @@ AsyncHttpsProxySocket::AsyncHttpsProxySocket(AsyncSocket* socket,
>                                              const SocketAddress& proxy,
>                                              const std::string& username,
>                                              const CryptString& password)
>-  : BufferedReadAdapter(socket, 1024), proxy_(proxy), agent_(user_agent),
>-    user_(username), pass_(password), force_connect_(false), state_(PS_ERROR),
>-    context_(0) {
>-}
>+    : BufferedReadAdapter(socket, 1024),
>+      proxy_(proxy),
>+      agent_(user_agent),
>+      user_(username),
>+      pass_(password),
>+      force_connect_(false),
>+      state_(PS_ERROR),
>+      context_(0) {}
> 
> AsyncHttpsProxySocket::~AsyncHttpsProxySocket() {
>   delete context_;
>@@ -295,7 +299,7 @@ Socket::ConnState AsyncHttpsProxySocket::GetState() const {
>   }
> }
> 
>-void AsyncHttpsProxySocket::OnConnectEvent(AsyncSocket * socket) {
>+void AsyncHttpsProxySocket::OnConnectEvent(AsyncSocket* socket) {
>   RTC_LOG(LS_VERBOSE) << "AsyncHttpsProxySocket::OnConnectEvent";
>   if (!ShouldIssueConnect()) {
>     state_ = PS_TUNNEL;
>@@ -305,7 +309,7 @@ void AsyncHttpsProxySocket::OnConnectEvent(AsyncSocket * socket) {
>   SendRequest();
> }
> 
>-void AsyncHttpsProxySocket::OnCloseEvent(AsyncSocket * socket, int err) {
>+void AsyncHttpsProxySocket::OnCloseEvent(AsyncSocket* socket, int err) {
>   RTC_LOG(LS_VERBOSE) << "AsyncHttpsProxySocket::OnCloseEvent(" << err << ")";
>   if ((state_ == PS_WAIT_CLOSE) && (err == 0)) {
>     state_ = PS_ERROR;
>@@ -383,7 +387,7 @@ void AsyncHttpsProxySocket::SendRequest() {
>   RTC_LOG(LS_VERBOSE) << "AsyncHttpsProxySocket >> " << str;
> }
> 
>-void AsyncHttpsProxySocket::ProcessLine(char * data, size_t len) {
>+void AsyncHttpsProxySocket::ProcessLine(char* data, size_t len) {
>   RTC_LOG(LS_VERBOSE) << "AsyncHttpsProxySocket << " << data;
> 
>   if (len == 0) {
>@@ -404,15 +408,18 @@ void AsyncHttpsProxySocket::ProcessLine(char * data, size_t len) {
>       if (!unknown_mechanisms_.empty() && !report) {
>         report = true;
>         std::string msg(
>-          "Unable to connect to the Google Talk service due to an incompatibility "
>-          "with your proxy.\r\nPlease help us resolve this issue by submitting the "
>-          "following information to us using our technical issue submission form "
>-          "at:\r\n\r\n"
>-          "http://www.google.com/support/talk/bin/request.py\r\n\r\n"
>-          "We apologize for the inconvenience.\r\n\r\n"
>-          "Information to submit to Google: "
>-          );
>-        //std::string msg("Please report the following information to foo@bar.com:\r\nUnknown methods: ");
>+            "Unable to connect to the Google Talk service due to an "
>+            "incompatibility "
>+            "with your proxy.\r\nPlease help us resolve this issue by "
>+            "submitting the "
>+            "following information to us using our technical issue submission "
>+            "form "
>+            "at:\r\n\r\n"
>+            "http://www.google.com/support/talk/bin/request.py\r\n\r\n"
>+            "We apologize for the inconvenience.\r\n\r\n"
>+            "Information to submit to Google: ");
>+        // std::string msg("Please report the following information to
>+        // foo@bar.com:\r\nUnknown methods: ");
>         msg.append(unknown_mechanisms_);
> #if defined(WEBRTC_WIN)
>         MessageBoxA(0, msg.c_str(), "Oops!", MB_OK);
>@@ -433,50 +440,49 @@ void AsyncHttpsProxySocket::ProcessLine(char * data, size_t len) {
>       return;
>     }
>     switch (code) {
>-    case 200:
>-      // connection good!
>-      state_ = PS_TUNNEL_HEADERS;
>-      return;
>+      case 200:
>+        // connection good!
>+        state_ = PS_TUNNEL_HEADERS;
>+        return;
> #if defined(HTTP_STATUS_PROXY_AUTH_REQ) && (HTTP_STATUS_PROXY_AUTH_REQ != 407)
> #error Wrong code for HTTP_STATUS_PROXY_AUTH_REQ
> #endif
>-    case 407:  // HTTP_STATUS_PROXY_AUTH_REQ
>-      state_ = PS_AUTHENTICATE;
>-      return;
>-    default:
>-      defer_error_ = 0;
>-      state_ = PS_ERROR_HEADERS;
>-      return;
>+      case 407:  // HTTP_STATUS_PROXY_AUTH_REQ
>+        state_ = PS_AUTHENTICATE;
>+        return;
>+      default:
>+        defer_error_ = 0;
>+        state_ = PS_ERROR_HEADERS;
>+        return;
>     }
>-  } else if ((state_ == PS_AUTHENTICATE)
>-             && (_strnicmp(data, "Proxy-Authenticate:", 19) == 0)) {
>+  } else if ((state_ == PS_AUTHENTICATE) &&
>+             (_strnicmp(data, "Proxy-Authenticate:", 19) == 0)) {
>     std::string response, auth_method;
>-    switch (HttpAuthenticate(data + 19, len - 19,
>-                             proxy_, "CONNECT", "/",
>-                             user_, pass_, context_, response, auth_method)) {
>-    case HAR_IGNORE:
>-      RTC_LOG(LS_VERBOSE) << "Ignoring Proxy-Authenticate: " << auth_method;
>-      if (!unknown_mechanisms_.empty())
>-        unknown_mechanisms_.append(", ");
>-      unknown_mechanisms_.append(auth_method);
>-      break;
>-    case HAR_RESPONSE:
>-      headers_ = "Proxy-Authorization: ";
>-      headers_.append(response);
>-      headers_.append("\r\n");
>-      state_ = PS_SKIP_HEADERS;
>-      unknown_mechanisms_.clear();
>-      break;
>-    case HAR_CREDENTIALS:
>-      defer_error_ = SOCKET_EACCES;
>-      state_ = PS_ERROR_HEADERS;
>-      unknown_mechanisms_.clear();
>-      break;
>-    case HAR_ERROR:
>-      defer_error_ = 0;
>-      state_ = PS_ERROR_HEADERS;
>-      unknown_mechanisms_.clear();
>-      break;
>+    switch (HttpAuthenticate(data + 19, len - 19, proxy_, "CONNECT", "/", user_,
>+                             pass_, context_, response, auth_method)) {
>+      case HAR_IGNORE:
>+        RTC_LOG(LS_VERBOSE) << "Ignoring Proxy-Authenticate: " << auth_method;
>+        if (!unknown_mechanisms_.empty())
>+          unknown_mechanisms_.append(", ");
>+        unknown_mechanisms_.append(auth_method);
>+        break;
>+      case HAR_RESPONSE:
>+        headers_ = "Proxy-Authorization: ";
>+        headers_.append(response);
>+        headers_.append("\r\n");
>+        state_ = PS_SKIP_HEADERS;
>+        unknown_mechanisms_.clear();
>+        break;
>+      case HAR_CREDENTIALS:
>+        defer_error_ = SOCKET_EACCES;
>+        state_ = PS_ERROR_HEADERS;
>+        unknown_mechanisms_.clear();
>+        break;
>+      case HAR_ERROR:
>+        defer_error_ = 0;
>+        state_ = PS_ERROR_HEADERS;
>+        unknown_mechanisms_.clear();
>+        break;
>     }
>   } else if (_strnicmp(data, "Content-Length:", 15) == 0) {
>     content_length_ = strtoul(data + 15, 0, 0);
>@@ -515,9 +521,11 @@ AsyncSocksProxySocket::AsyncSocksProxySocket(AsyncSocket* socket,
>                                              const SocketAddress& proxy,
>                                              const std::string& username,
>                                              const CryptString& password)
>-    : BufferedReadAdapter(socket, 1024), state_(SS_ERROR), proxy_(proxy),
>-      user_(username), pass_(password) {
>-}
>+    : BufferedReadAdapter(socket, 1024),
>+      state_(SS_ERROR),
>+      proxy_(proxy),
>+      user_(username),
>+      pass_(password) {}
> 
> AsyncSocksProxySocket::~AsyncSocksProxySocket() = default;
> 
>@@ -562,8 +570,7 @@ void AsyncSocksProxySocket::ProcessInput(char* data, size_t* len) {
> 
>   if (state_ == SS_HELLO) {
>     uint8_t ver, method;
>-    if (!response.ReadUInt8(&ver) ||
>-        !response.ReadUInt8(&method))
>+    if (!response.ReadUInt8(&ver) || !response.ReadUInt8(&method))
>       return;
> 
>     if (ver != 5) {
>@@ -581,8 +588,7 @@ void AsyncSocksProxySocket::ProcessInput(char* data, size_t* len) {
>     }
>   } else if (state_ == SS_AUTH) {
>     uint8_t ver, status;
>-    if (!response.ReadUInt8(&ver) ||
>-        !response.ReadUInt8(&status))
>+    if (!response.ReadUInt8(&ver) || !response.ReadUInt8(&status))
>       return;
> 
>     if ((ver != 1) || (status != 0)) {
>@@ -593,10 +599,8 @@ void AsyncSocksProxySocket::ProcessInput(char* data, size_t* len) {
>     SendConnect();
>   } else if (state_ == SS_CONNECT) {
>     uint8_t ver, rep, rsv, atyp;
>-    if (!response.ReadUInt8(&ver) ||
>-        !response.ReadUInt8(&rep) ||
>-        !response.ReadUInt8(&rsv) ||
>-        !response.ReadUInt8(&atyp))
>+    if (!response.ReadUInt8(&ver) || !response.ReadUInt8(&rep) ||
>+        !response.ReadUInt8(&rsv) || !response.ReadUInt8(&atyp))
>       return;
> 
>     if ((ver != 5) || (rep != 0)) {
>@@ -607,22 +611,19 @@ void AsyncSocksProxySocket::ProcessInput(char* data, size_t* len) {
>     uint16_t port;
>     if (atyp == 1) {
>       uint32_t addr;
>-      if (!response.ReadUInt32(&addr) ||
>-          !response.ReadUInt16(&port))
>+      if (!response.ReadUInt32(&addr) || !response.ReadUInt16(&port))
>         return;
>       RTC_LOG(LS_VERBOSE) << "Bound on " << addr << ":" << port;
>     } else if (atyp == 3) {
>       uint8_t len;
>       std::string addr;
>-      if (!response.ReadUInt8(&len) ||
>-          !response.ReadString(&addr, len) ||
>+      if (!response.ReadUInt8(&len) || !response.ReadString(&addr, len) ||
>           !response.ReadUInt16(&port))
>         return;
>       RTC_LOG(LS_VERBOSE) << "Bound on " << addr << ":" << port;
>     } else if (atyp == 4) {
>       std::string addr;
>-      if (!response.ReadString(&addr, 16) ||
>-          !response.ReadUInt16(&port))
>+      if (!response.ReadString(&addr, 16) || !response.ReadUInt16(&port))
>         return;
>       RTC_LOG(LS_VERBOSE) << "Bound on <IPV6>:" << port;
>     } else {
>@@ -651,7 +652,7 @@ void AsyncSocksProxySocket::ProcessInput(char* data, size_t* len) {
> 
> void AsyncSocksProxySocket::SendHello() {
>   ByteBufferWriter request;
>-  request.WriteUInt8(5);    // Socks Version
>+  request.WriteUInt8(5);  // Socks Version
>   if (user_.empty()) {
>     request.WriteUInt8(1);  // Authentication Mechanisms
>     request.WriteUInt8(0);  // No authentication
>@@ -665,31 +666,31 @@ void AsyncSocksProxySocket::SendHello() {
> }
> 
> void AsyncSocksProxySocket::SendAuth() {
>-  ByteBufferWriter request;
>-  request.WriteUInt8(1);           // Negotiation Version
>+  ByteBufferWriterT<ZeroOnFreeBuffer<char>> request;
>+  request.WriteUInt8(1);  // Negotiation Version
>   request.WriteUInt8(static_cast<uint8_t>(user_.size()));
>-  request.WriteString(user_);      // Username
>+  request.WriteString(user_);  // Username
>   request.WriteUInt8(static_cast<uint8_t>(pass_.GetLength()));
>   size_t len = pass_.GetLength() + 1;
>-  char * sensitive = new char[len];
>+  char* sensitive = new char[len];
>   pass_.CopyTo(sensitive, true);
>-  request.WriteString(sensitive);  // Password
>-  memset(sensitive, 0, len);
>-  delete [] sensitive;
>+  request.WriteBytes(sensitive, pass_.GetLength());  // Password
>+  ExplicitZeroMemory(sensitive, len);
>+  delete[] sensitive;
>   DirectSend(request.Data(), request.Length());
>   state_ = SS_AUTH;
> }
> 
> void AsyncSocksProxySocket::SendConnect() {
>   ByteBufferWriter request;
>-  request.WriteUInt8(5);              // Socks Version
>-  request.WriteUInt8(1);              // CONNECT
>-  request.WriteUInt8(0);              // Reserved
>+  request.WriteUInt8(5);  // Socks Version
>+  request.WriteUInt8(1);  // CONNECT
>+  request.WriteUInt8(0);  // Reserved
>   if (dest_.IsUnresolvedIP()) {
>     std::string hostname = dest_.hostname();
>-    request.WriteUInt8(3);            // DOMAINNAME
>+    request.WriteUInt8(3);  // DOMAINNAME
>     request.WriteUInt8(static_cast<uint8_t>(hostname.size()));
>-    request.WriteString(hostname);    // Destination Hostname
>+    request.WriteString(hostname);  // Destination Hostname
>   } else {
>     request.WriteUInt8(1);            // IPV4
>     request.WriteUInt32(dest_.ip());  // Destination IP
>@@ -736,8 +737,7 @@ void AsyncSocksProxyServerSocket::DirectSend(const ByteBufferWriter& buf) {
> 
> void AsyncSocksProxyServerSocket::HandleHello(ByteBufferReader* request) {
>   uint8_t ver, num_methods;
>-  if (!request->ReadUInt8(&ver) ||
>-      !request->ReadUInt8(&num_methods)) {
>+  if (!request->ReadUInt8(&ver) || !request->ReadUInt8(&num_methods)) {
>     Error(0);
>     return;
>   }
>@@ -767,7 +767,7 @@ void AsyncSocksProxyServerSocket::HandleHello(ByteBufferReader* request) {
> 
> void AsyncSocksProxyServerSocket::SendHelloReply(uint8_t method) {
>   ByteBufferWriter response;
>-  response.WriteUInt8(5);  // Socks Version
>+  response.WriteUInt8(5);       // Socks Version
>   response.WriteUInt8(method);  // Auth method
>   DirectSend(response);
> }
>@@ -775,10 +775,8 @@ void AsyncSocksProxyServerSocket::SendHelloReply(uint8_t method) {
> void AsyncSocksProxyServerSocket::HandleAuth(ByteBufferReader* request) {
>   uint8_t ver, user_len, pass_len;
>   std::string user, pass;
>-  if (!request->ReadUInt8(&ver) ||
>-      !request->ReadUInt8(&user_len) ||
>-      !request->ReadString(&user, user_len) ||
>-      !request->ReadUInt8(&pass_len) ||
>+  if (!request->ReadUInt8(&ver) || !request->ReadUInt8(&user_len) ||
>+      !request->ReadString(&user, user_len) || !request->ReadUInt8(&pass_len) ||
>       !request->ReadString(&pass, pass_len)) {
>     Error(0);
>     return;
>@@ -800,20 +798,16 @@ void AsyncSocksProxyServerSocket::HandleConnect(ByteBufferReader* request) {
>   uint8_t ver, command, reserved, addr_type;
>   uint32_t ip;
>   uint16_t port;
>-  if (!request->ReadUInt8(&ver) ||
>-      !request->ReadUInt8(&command) ||
>-      !request->ReadUInt8(&reserved) ||
>-      !request->ReadUInt8(&addr_type) ||
>-      !request->ReadUInt32(&ip) ||
>-      !request->ReadUInt16(&port)) {
>-      Error(0);
>-      return;
>+  if (!request->ReadUInt8(&ver) || !request->ReadUInt8(&command) ||
>+      !request->ReadUInt8(&reserved) || !request->ReadUInt8(&addr_type) ||
>+      !request->ReadUInt32(&ip) || !request->ReadUInt16(&port)) {
>+    Error(0);
>+    return;
>   }
> 
>-  if (ver != 5 || command != 1 ||
>-      reserved != 0 || addr_type != 1) {
>-      Error(0);
>-      return;
>+  if (ver != 5 || command != 1 || reserved != 0 || addr_type != 1) {
>+    Error(0);
>+    return;
>   }
> 
>   SignalConnectRequest(this, SocketAddress(ip, port));
>@@ -826,10 +820,10 @@ void AsyncSocksProxyServerSocket::SendConnectResult(int result,
>     return;
> 
>   ByteBufferWriter response;
>-  response.WriteUInt8(5);  // Socks version
>+  response.WriteUInt8(5);              // Socks version
>   response.WriteUInt8((result != 0));  // 0x01 is generic error
>-  response.WriteUInt8(0);  // reserved
>-  response.WriteUInt8(1);  // IPv4 address
>+  response.WriteUInt8(0);              // reserved
>+  response.WriteUInt8(1);              // IPv4 address
>   response.WriteUInt32(addr.ip());
>   response.WriteUInt16(addr.port());
>   DirectSend(response);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketadapters.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketadapters.h
>index c5d1bfcabb7..ad88fe6b6fb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketadapters.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketadapters.h
>@@ -49,7 +49,7 @@ class BufferedReadAdapter : public AsyncSocketAdapter {
>   void OnReadEvent(AsyncSocket* socket) override;
> 
>  private:
>-  char * buffer_;
>+  char* buffer_;
>   size_t buffer_size_, data_len_;
>   bool buffering_;
>   RTC_DISALLOW_COPY_AND_ASSIGN(BufferedReadAdapter);
>@@ -62,8 +62,8 @@ class AsyncProxyServerSocket : public BufferedReadAdapter {
>  public:
>   AsyncProxyServerSocket(AsyncSocket* socket, size_t buffer_size);
>   ~AsyncProxyServerSocket() override;
>-  sigslot::signal2<AsyncProxyServerSocket*,
>-                   const SocketAddress&>  SignalConnectRequest;
>+  sigslot::signal2<AsyncProxyServerSocket*, const SocketAddress&>
>+      SignalConnectRequest;
>   virtual void SendConnectResult(int err, const SocketAddress& addr) = 0;
> };
> 
>@@ -99,9 +99,11 @@ class AsyncSSLServerSocket : public BufferedReadAdapter {
> // Implements a socket adapter that speaks the HTTP/S proxy protocol.
> class AsyncHttpsProxySocket : public BufferedReadAdapter {
>  public:
>-  AsyncHttpsProxySocket(AsyncSocket* socket, const std::string& user_agent,
>-    const SocketAddress& proxy,
>-    const std::string& username, const CryptString& password);
>+  AsyncHttpsProxySocket(AsyncSocket* socket,
>+                        const std::string& user_agent,
>+                        const SocketAddress& proxy,
>+                        const std::string& username,
>+                        const CryptString& password);
>   ~AsyncHttpsProxySocket() override;
> 
>   // If connect is forced, the adapter will always issue an HTTP CONNECT to the
>@@ -134,10 +136,18 @@ class AsyncHttpsProxySocket : public BufferedReadAdapter {
>   int defer_error_;
>   bool expect_close_;
>   enum ProxyState {
>-    PS_INIT, PS_LEADER, PS_AUTHENTICATE, PS_SKIP_HEADERS, PS_ERROR_HEADERS,
>-    PS_TUNNEL_HEADERS, PS_SKIP_BODY, PS_TUNNEL, PS_WAIT_CLOSE, PS_ERROR
>+    PS_INIT,
>+    PS_LEADER,
>+    PS_AUTHENTICATE,
>+    PS_SKIP_HEADERS,
>+    PS_ERROR_HEADERS,
>+    PS_TUNNEL_HEADERS,
>+    PS_SKIP_BODY,
>+    PS_TUNNEL,
>+    PS_WAIT_CLOSE,
>+    PS_ERROR
>   } state_;
>-  HttpAuthContext * context_;
>+  HttpAuthContext* context_;
>   std::string unknown_mechanisms_;
>   RTC_DISALLOW_COPY_AND_ASSIGN(AsyncHttpsProxySocket);
> };
>@@ -147,8 +157,10 @@ class AsyncHttpsProxySocket : public BufferedReadAdapter {
> // Implements a socket adapter that speaks the SOCKS proxy protocol.
> class AsyncSocksProxySocket : public BufferedReadAdapter {
>  public:
>-  AsyncSocksProxySocket(AsyncSocket* socket, const SocketAddress& proxy,
>-    const std::string& username, const CryptString& password);
>+  AsyncSocksProxySocket(AsyncSocket* socket,
>+                        const SocketAddress& proxy,
>+                        const std::string& username,
>+                        const CryptString& password);
>   ~AsyncSocksProxySocket() override;
> 
>   int Connect(const SocketAddress& addr) override;
>@@ -166,9 +178,7 @@ class AsyncSocksProxySocket : public BufferedReadAdapter {
>   void Error(int error);
> 
>  private:
>-  enum State {
>-    SS_INIT, SS_HELLO, SS_AUTH, SS_CONNECT, SS_TUNNEL, SS_ERROR
>-  };
>+  enum State { SS_INIT, SS_HELLO, SS_AUTH, SS_CONNECT, SS_TUNNEL, SS_ERROR };
>   State state_;
>   SocketAddress proxy_, dest_;
>   std::string user_;
>@@ -196,7 +206,12 @@ class AsyncSocksProxyServerSocket : public AsyncProxyServerSocket {
> 
>   static const int kBufferSize = 1024;
>   enum State {
>-    SS_HELLO, SS_AUTH, SS_CONNECT, SS_CONNECT_PENDING, SS_TUNNEL, SS_ERROR
>+    SS_HELLO,
>+    SS_AUTH,
>+    SS_CONNECT,
>+    SS_CONNECT_PENDING,
>+    SS_TUNNEL,
>+    SS_ERROR
>   };
>   State state_;
>   RTC_DISALLOW_COPY_AND_ASSIGN(AsyncSocksProxyServerSocket);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketaddress.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketaddress.cc
>index 54a41d420e6..c91846d1bad 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketaddress.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketaddress.cc
>@@ -12,9 +12,9 @@
> #include "rtc_base/numerics/safe_conversions.h"
> 
> #if defined(WEBRTC_POSIX)
>-#include <sys/types.h>
>-#include <sys/socket.h>
> #include <netinet/in.h>
>+#include <sys/socket.h>
>+#include <sys/types.h>
> #if defined(OPENBSD)
> #include <netinet/in_systm.h>
> #endif
>@@ -26,12 +26,11 @@
> #include <unistd.h>
> #endif
> 
>-#include <sstream>
>-
> #include "rtc_base/byteorder.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/nethelpers.h"
>+#include "rtc_base/strings/string_builder.h"
> 
> #if defined(WEBRTC_WIN)
> #include "rtc_base/win32.h"
>@@ -161,21 +160,21 @@ std::string SocketAddress::HostAsSensitiveURIString() const {
> }
> 
> std::string SocketAddress::PortAsString() const {
>-  std::ostringstream ost;
>-  ost << port_;
>-  return ost.str();
>+  return std::to_string(port_);
> }
> 
> std::string SocketAddress::ToString() const {
>-  std::ostringstream ost;
>-  ost << *this;
>-  return ost.str();
>+  char buf[1024];
>+  rtc::SimpleStringBuilder sb(buf);
>+  sb << HostAsURIString() << ":" << port();
>+  return sb.str();
> }
> 
> std::string SocketAddress::ToSensitiveString() const {
>-  std::ostringstream ost;
>-  ost << HostAsSensitiveURIString() << ":" << port();
>-  return ost.str();
>+  char buf[1024];
>+  rtc::SimpleStringBuilder sb(buf);
>+  sb << HostAsSensitiveURIString() << ":" << port();
>+  return sb.str();
> }
> 
> bool SocketAddress::FromString(const std::string& str) {
>@@ -200,18 +199,13 @@ bool SocketAddress::FromString(const std::string& str) {
>   return true;
> }
> 
>-std::ostream& operator<<(std::ostream& os, const SocketAddress& addr) {
>-  os << addr.HostAsURIString() << ":" << addr.port();
>-  return os;
>-}
>-
> bool SocketAddress::IsAnyIP() const {
>   return IPIsAny(ip_);
> }
> 
> bool SocketAddress::IsLoopbackIP() const {
>-  return IPIsLoopback(ip_) || (IPIsAny(ip_) &&
>-                               0 == strcmp(hostname_.c_str(), "localhost"));
>+  return IPIsLoopback(ip_) ||
>+         (IPIsAny(ip_) && 0 == strcmp(hostname_.c_str(), "localhost"));
> }
> 
> bool SocketAddress::IsPrivateIP() const {
>@@ -240,7 +234,7 @@ bool SocketAddress::operator<(const SocketAddress& addr) const {
> 
> bool SocketAddress::EqualIPs(const SocketAddress& addr) const {
>   return (ip_ == addr.ip_) &&
>-      ((!IPIsAny(ip_) && !IPIsUnspec(ip_)) || (hostname_ == addr.hostname_));
>+         ((!IPIsAny(ip_) && !IPIsUnspec(ip_)) || (hostname_ == addr.hostname_));
> }
> 
> bool SocketAddress::EqualPorts(const SocketAddress& addr) const {
>@@ -299,7 +293,7 @@ static size_t ToSockAddrStorageHelper(sockaddr_storage* addr,
>   return 0;
> }
> 
>-size_t SocketAddress::ToDualStackSockAddrStorage(sockaddr_storage *addr) const {
>+size_t SocketAddress::ToDualStackSockAddrStorage(sockaddr_storage* addr) const {
>   return ToSockAddrStorageHelper(addr, ip_.AsIPv6Address(), port_, scope_id_);
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketaddress.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketaddress.h
>index d58eed84288..7095be189db 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketaddress.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketaddress.h
>@@ -13,8 +13,10 @@
> 
> #include <iosfwd>
> #include <string>
>+#ifdef UNIT_TEST
>+#include <ostream>  // no-presubmit-check TODO(webrtc:8982)
>+#endif              // UNIT_TEST
> #include <vector>
>-#include "rtc_base/basictypes.h"
> #include "rtc_base/ipaddress.h"
> 
> #undef SetPort
>@@ -92,7 +94,7 @@ class SocketAddress {
> 
>   const IPAddress& ipaddr() const;
> 
>-  int family() const {return ip_.family(); }
>+  int family() const { return ip_.family(); }
> 
>   // Returns the port part of this address.
>   uint16_t port() const;
>@@ -102,7 +104,7 @@ class SocketAddress {
>   // interfaces having different scope-ids for their link-local addresses.
>   // IPv4 address do not have scope_ids and sockaddr_in structures do not have
>   // a field for them.
>-  int scope_id() const {return scope_id_; }
>+  int scope_id() const { return scope_id_; }
>   void SetScopeID(int id) { scope_id_ = id; }
> 
>   // Returns the 'host' portion of the address (hostname or IP) in a form
>@@ -126,7 +128,12 @@ class SocketAddress {
>   // Parses hostname:port and [hostname]:port.
>   bool FromString(const std::string& str);
> 
>-  friend std::ostream& operator<<(std::ostream& os, const SocketAddress& addr);
>+#ifdef UNIT_TEST
>+  inline std::ostream& operator<<(  // no-presubmit-check TODO(webrtc:8982)
>+      std::ostream& os) {           // no-presubmit-check TODO(webrtc:8982)
>+    return os << HostAsURIString() << ":" << port();
>+  }
>+#endif  // UNIT_TEST
> 
>   // Determines whether this represents a missing / any IP address.
>   // That is, 0.0.0.0 or ::.
>@@ -147,13 +154,13 @@ class SocketAddress {
>   bool IsUnresolvedIP() const;
> 
>   // Determines whether this address is identical to the given one.
>-  bool operator ==(const SocketAddress& addr) const;
>-  inline bool operator !=(const SocketAddress& addr) const {
>-    return !this->operator ==(addr);
>+  bool operator==(const SocketAddress& addr) const;
>+  inline bool operator!=(const SocketAddress& addr) const {
>+    return !this->operator==(addr);
>   }
> 
>   // Compares based on IP and then port.
>-  bool operator <(const SocketAddress& addr) const;
>+  bool operator<(const SocketAddress& addr) const;
> 
>   // Determines whether this address has the same IP as the one given.
>   bool EqualIPs(const SocketAddress& addr) const;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketaddress_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketaddress_unittest.cc
>index 0d168dfcf4d..b7a2d92d31a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketaddress_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketaddress_unittest.cc
>@@ -18,14 +18,12 @@
> 
> namespace rtc {
> 
>-const in6_addr kTestV6Addr =  { { {0x20, 0x01, 0x0d, 0xb8,
>-                                   0x10, 0x20, 0x30, 0x40,
>-                                   0x50, 0x60, 0x70, 0x80,
>-                                   0x90, 0xA0, 0xB0, 0xC0} } };
>-const in6_addr kMappedV4Addr = { { {0x00, 0x00, 0x00, 0x00,
>-                                    0x00, 0x00, 0x00, 0x00,
>-                                    0x00, 0x00, 0xFF, 0xFF,
>-                                    0x01, 0x02, 0x03, 0x04} } };
>+const in6_addr kTestV6Addr = {
>+    {{0x20, 0x01, 0x0d, 0xb8, 0x10, 0x20, 0x30, 0x40, 0x50, 0x60, 0x70, 0x80,
>+      0x90, 0xA0, 0xB0, 0xC0}}};
>+const in6_addr kMappedV4Addr = {
>+    {{0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0xFF, 0xFF,
>+      0x01, 0x02, 0x03, 0x04}}};
> const std::string kTestV6AddrString = "2001:db8:1020:3040:5060:7080:90a0:b0c0";
> const std::string kTestV6AddrAnonymizedString = "2001:db8:1020:x:x:x:x:x";
> const std::string kTestV6AddrFullString =
>@@ -240,55 +238,51 @@ TEST(SocketAddressTest, TestToFromSockAddrStorage) {
>   EXPECT_FALSE(SocketAddressFromSockAddrStorage(addr_storage, nullptr));
> }
> 
>-bool AreEqual(const SocketAddress& addr1,
>-              const SocketAddress& addr2) {
>-  return addr1 == addr2 && addr2 == addr1 &&
>-      !(addr1 != addr2) && !(addr2 != addr1);
>+bool AreEqual(const SocketAddress& addr1, const SocketAddress& addr2) {
>+  return addr1 == addr2 && addr2 == addr1 && !(addr1 != addr2) &&
>+         !(addr2 != addr1);
> }
> 
>-bool AreUnequal(const SocketAddress& addr1,
>-                const SocketAddress& addr2) {
>-  return !(addr1 == addr2) && !(addr2 == addr1) &&
>-      addr1 != addr2 && addr2 != addr1;
>+bool AreUnequal(const SocketAddress& addr1, const SocketAddress& addr2) {
>+  return !(addr1 == addr2) && !(addr2 == addr1) && addr1 != addr2 &&
>+         addr2 != addr1;
> }
> 
> TEST(SocketAddressTest, TestEqualityOperators) {
>   SocketAddress addr1("1.2.3.4", 5678);
>   SocketAddress addr2("1.2.3.4", 5678);
>-  EXPECT_PRED2(AreEqual, addr1, addr2);
>+  EXPECT_TRUE(AreEqual(addr1, addr2));
> 
>   addr2 = SocketAddress("0.0.0.1", 5678);
>-  EXPECT_PRED2(AreUnequal, addr1, addr2);
>+  EXPECT_TRUE(AreUnequal(addr1, addr2));
> 
>   addr2 = SocketAddress("1.2.3.4", 1234);
>-  EXPECT_PRED2(AreUnequal, addr1, addr2);
>+  EXPECT_TRUE(AreUnequal(addr1, addr2));
> 
>   addr2 = SocketAddress(kTestV6AddrString, 5678);
>-  EXPECT_PRED2(AreUnequal, addr1, addr2);
>+  EXPECT_TRUE(AreUnequal(addr1, addr2));
> 
>   addr1 = SocketAddress(kTestV6AddrString, 5678);
>-  EXPECT_PRED2(AreEqual, addr1, addr2);
>+  EXPECT_TRUE(AreEqual(addr1, addr2));
> 
>   addr2 = SocketAddress(kTestV6AddrString, 1234);
>-  EXPECT_PRED2(AreUnequal, addr1, addr2);
>+  EXPECT_TRUE(AreUnequal(addr1, addr2));
> 
>   addr2 = SocketAddress("fe80::1", 5678);
>-  EXPECT_PRED2(AreUnequal, addr1, addr2);
>+  EXPECT_TRUE(AreUnequal(addr1, addr2));
> 
>   SocketAddress addr3("a.b.c.d", 1);
>   SocketAddress addr4("b.b.c.d", 1);
>-  EXPECT_PRED2(AreUnequal, addr3, addr4);
>-  EXPECT_PRED2(AreEqual, addr3, addr3);
>+  EXPECT_TRUE(AreUnequal(addr3, addr4));
>+  EXPECT_TRUE(AreEqual(addr3, addr3));
> 
>   addr3.SetIP(addr1.ip());
>   addr4.SetIP(addr1.ip());
>-  EXPECT_PRED2(AreEqual,addr3, addr4);
>+  EXPECT_TRUE(AreEqual(addr3, addr4));
> }
> 
> bool IsLessThan(const SocketAddress& addr1, const SocketAddress& addr2) {
>-  return addr1 < addr2 &&
>-      !(addr2 < addr1) &&
>-      !(addr1 == addr2);
>+  return addr1 < addr2 && !(addr2 < addr1) && !(addr1 == addr2);
> }
> 
> TEST(SocketAddressTest, TestComparisonOperator) {
>@@ -299,19 +293,19 @@ TEST(SocketAddressTest, TestComparisonOperator) {
>   EXPECT_FALSE(addr2 < addr1);
> 
>   addr2 = SocketAddress("1.2.3.4", 5679);
>-  EXPECT_PRED2(IsLessThan, addr1, addr2);
>+  EXPECT_TRUE(IsLessThan(addr1, addr2));
> 
>   addr2 = SocketAddress("2.2.3.4", 49152);
>-  EXPECT_PRED2(IsLessThan, addr1, addr2);
>+  EXPECT_TRUE(IsLessThan(addr1, addr2));
> 
>   addr2 = SocketAddress(kTestV6AddrString, 5678);
>-  EXPECT_PRED2(IsLessThan, addr1, addr2);
>+  EXPECT_TRUE(IsLessThan(addr1, addr2));
> 
>   addr1 = SocketAddress("fe80::1", 5678);
>-  EXPECT_PRED2(IsLessThan, addr2, addr1);
>+  EXPECT_TRUE(IsLessThan(addr2, addr1));
> 
>   addr2 = SocketAddress("fe80::1", 5679);
>-  EXPECT_PRED2(IsLessThan, addr1, addr2);
>+  EXPECT_TRUE(IsLessThan(addr1, addr2));
> 
>   addr2 = SocketAddress("fe80::1", 5678);
>   EXPECT_FALSE(addr1 < addr2);
>@@ -319,7 +313,7 @@ TEST(SocketAddressTest, TestComparisonOperator) {
> 
>   SocketAddress addr3("a.b.c.d", 1);
>   SocketAddress addr4("b.b.c.d", 1);
>-  EXPECT_PRED2(IsLessThan, addr3, addr4);
>+  EXPECT_TRUE(IsLessThan(addr3, addr4));
> }
> 
> TEST(SocketAddressTest, TestToSensitiveString) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketaddresspair.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketaddresspair.cc
>index 3e4748f4212..914ffd9916f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketaddresspair.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketaddresspair.cc
>@@ -12,17 +12,15 @@
> 
> namespace rtc {
> 
>-SocketAddressPair::SocketAddressPair(
>-    const SocketAddress& src, const SocketAddress& dest)
>-    : src_(src), dest_(dest) {
>-}
>-
>+SocketAddressPair::SocketAddressPair(const SocketAddress& src,
>+                                     const SocketAddress& dest)
>+    : src_(src), dest_(dest) {}
> 
>-bool SocketAddressPair::operator ==(const SocketAddressPair& p) const {
>+bool SocketAddressPair::operator==(const SocketAddressPair& p) const {
>   return (src_ == p.src_) && (dest_ == p.dest_);
> }
> 
>-bool SocketAddressPair::operator <(const SocketAddressPair& p) const {
>+bool SocketAddressPair::operator<(const SocketAddressPair& p) const {
>   if (src_ < p.src_)
>     return true;
>   if (p.src_ < src_)
>@@ -38,4 +36,4 @@ size_t SocketAddressPair::Hash() const {
>   return src_.Hash() ^ dest_.Hash();
> }
> 
>-} // namespace rtc
>+}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketaddresspair.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketaddresspair.h
>index 8ff0ee6c48e..5ff148a75d8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketaddresspair.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketaddresspair.h
>@@ -19,23 +19,23 @@ namespace rtc {
> // identify a connection between two machines.  (For UDP, this "connection" is
> // not maintained explicitly in a socket.)
> class SocketAddressPair {
>-public:
>+ public:
>   SocketAddressPair() {}
>   SocketAddressPair(const SocketAddress& srs, const SocketAddress& dest);
> 
>   const SocketAddress& source() const { return src_; }
>   const SocketAddress& destination() const { return dest_; }
> 
>-  bool operator ==(const SocketAddressPair& r) const;
>-  bool operator <(const SocketAddressPair& r) const;
>+  bool operator==(const SocketAddressPair& r) const;
>+  bool operator<(const SocketAddressPair& r) const;
> 
>   size_t Hash() const;
> 
>-private:
>+ private:
>   SocketAddress src_;
>   SocketAddress dest_;
> };
> 
>-} // namespace rtc
>+}  // namespace rtc
> 
>-#endif // RTC_BASE_SOCKETADDRESSPAIR_H_
>+#endif  // RTC_BASE_SOCKETADDRESSPAIR_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketfactory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketfactory.h
>index 58bc0de3990..5a5cd04333b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketfactory.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketfactory.h
>@@ -17,22 +17,17 @@
> namespace rtc {
> 
> class SocketFactory {
>-public:
>+ public:
>   virtual ~SocketFactory() {}
> 
>   // Returns a new socket for blocking communication.  The type can be
>   // SOCK_DGRAM and SOCK_STREAM.
>-  // TODO: C++ inheritance rules mean that all users must have both
>-  // CreateSocket(int) and CreateSocket(int,int). Will remove CreateSocket(int)
>-  // (and CreateAsyncSocket(int) when all callers are changed.
>-  virtual Socket* CreateSocket(int type) = 0;
>   virtual Socket* CreateSocket(int family, int type) = 0;
>   // Returns a new socket for nonblocking communication.  The type can be
>   // SOCK_DGRAM and SOCK_STREAM.
>-  virtual AsyncSocket* CreateAsyncSocket(int type) = 0;
>   virtual AsyncSocket* CreateAsyncSocket(int family, int type) = 0;
> };
> 
>-} // namespace rtc
>+}  // namespace rtc
> 
>-#endif // RTC_BASE_SOCKETFACTORY_H_
>+#endif  // RTC_BASE_SOCKETFACTORY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketserver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketserver.h
>index 38543693d37..e29884e6b3c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketserver.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketserver.h
>@@ -35,7 +35,7 @@ class SocketServer : public SocketFactory {
>   // When the socket server is installed into a Thread, this function is
>   // called to allow the socket server to use the thread's message queue for
>   // any messaging that it might need to perform.
>-  virtual void SetMessageQueue(MessageQueue*) {}
>+  virtual void SetMessageQueue(MessageQueue* queue) {}
> 
>   // Sleeps until:
>   //  1) cms milliseconds have elapsed (unless cms == kForever)
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketstream.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketstream.cc
>index 8b4c513ea8c..2ea1cec0ac7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketstream.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketstream.cc
>@@ -28,9 +28,9 @@ void SocketStream::Attach(AsyncSocket* socket) {
>   socket_ = socket;
>   if (socket_) {
>     socket_->SignalConnectEvent.connect(this, &SocketStream::OnConnectEvent);
>-    socket_->SignalReadEvent.connect(this,    &SocketStream::OnReadEvent);
>-    socket_->SignalWriteEvent.connect(this,   &SocketStream::OnWriteEvent);
>-    socket_->SignalCloseEvent.connect(this,   &SocketStream::OnCloseEvent);
>+    socket_->SignalReadEvent.connect(this, &SocketStream::OnReadEvent);
>+    socket_->SignalWriteEvent.connect(this, &SocketStream::OnWriteEvent);
>+    socket_->SignalCloseEvent.connect(this, &SocketStream::OnCloseEvent);
>   }
> }
> 
>@@ -59,8 +59,10 @@ StreamState SocketStream::GetState() const {
>   }
> }
> 
>-StreamResult SocketStream::Read(void* buffer, size_t buffer_len,
>-                                size_t* read, int* error) {
>+StreamResult SocketStream::Read(void* buffer,
>+                                size_t buffer_len,
>+                                size_t* read,
>+                                int* error) {
>   RTC_DCHECK(socket_ != nullptr);
>   int result = socket_->Recv(buffer, buffer_len, nullptr);
>   if (result < 0) {
>@@ -78,8 +80,10 @@ StreamResult SocketStream::Read(void* buffer, size_t buffer_len,
>   return SR_EOS;
> }
> 
>-StreamResult SocketStream::Write(const void* data, size_t data_len,
>-                                 size_t* written, int* error) {
>+StreamResult SocketStream::Write(const void* data,
>+                                 size_t data_len,
>+                                 size_t* written,
>+                                 int* error) {
>   RTC_DCHECK(socket_ != nullptr);
>   int result = socket_->Send(data, data_len);
>   if (result < 0) {
>@@ -119,5 +123,4 @@ void SocketStream::OnCloseEvent(AsyncSocket* socket, int err) {
>   SignalEvent(this, SE_CLOSE, err);
> }
> 
>-
> }  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ssladapter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ssladapter.cc
>index 4f56cf7f027..e091f005200 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ssladapter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ssladapter.cc
>@@ -26,12 +26,8 @@ SSLAdapter* SSLAdapter::Create(AsyncSocket* socket) {
> 
> ///////////////////////////////////////////////////////////////////////////////
> 
>-bool InitializeSSL(VerificationCallback callback) {
>-  return OpenSSLAdapter::InitializeSSL(callback);
>-}
>-
>-bool InitializeSSLThread() {
>-  return OpenSSLAdapter::InitializeSSLThread();
>+bool InitializeSSL() {
>+  return OpenSSLAdapter::InitializeSSL();
> }
> 
> bool CleanupSSL() {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ssladapter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ssladapter.h
>index 5a17015cdc3..4843d264fcc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ssladapter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ssladapter.h
>@@ -11,7 +11,11 @@
> #ifndef RTC_BASE_SSLADAPTER_H_
> #define RTC_BASE_SSLADAPTER_H_
> 
>+#include <string>
>+#include <vector>
>+
> #include "rtc_base/asyncsocket.h"
>+#include "rtc_base/sslcertificate.h"
> #include "rtc_base/sslstreamadapter.h"
> 
> namespace rtc {
>@@ -26,8 +30,13 @@ class SSLAdapter;
> class SSLAdapterFactory {
>  public:
>   virtual ~SSLAdapterFactory() {}
>+
>   // Specifies whether TLS or DTLS is to be used for the SSL adapters.
>   virtual void SetMode(SSLMode mode) = 0;
>+
>+  // Specify a custom certificate verifier for SSL.
>+  virtual void SetCertVerifier(SSLCertificateVerifier* ssl_cert_verifier) = 0;
>+
>   // Creates a new SSL adapter, but from a shared context.
>   virtual SSLAdapter* CreateAdapter(AsyncSocket* socket) = 0;
> 
>@@ -54,6 +63,8 @@ class SSLAdapter : public AsyncSocketAdapter {
> 
>   // Do DTLS or TLS (default is TLS, if unspecified)
>   virtual void SetMode(SSLMode mode) = 0;
>+  // Specify a custom certificate verifier for SSL.
>+  virtual void SetCertVerifier(SSLCertificateVerifier* ssl_cert_verifier) = 0;
> 
>   // Set the certificate this socket will present to incoming clients.
>   virtual void SetIdentity(SSLIdentity* identity) = 0;
>@@ -82,14 +93,9 @@ class SSLAdapter : public AsyncSocketAdapter {
> 
> ///////////////////////////////////////////////////////////////////////////////
> 
>-typedef bool (*VerificationCallback)(void* cert);
>-
> // Call this on the main thread, before using SSL.
>-// Call CleanupSSLThread when finished with SSL.
>-bool InitializeSSL(VerificationCallback callback = nullptr);
>-
>-// Call to initialize additional threads.
>-bool InitializeSSLThread();
>+// Call CleanupSSL when finished with SSL.
>+bool InitializeSSL();
> 
> // Call to cleanup additional threads, and also the main thread.
> bool CleanupSSL();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ssladapter_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ssladapter_unittest.cc
>index 0996b0105cb..ec532b1c442 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ssladapter_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/ssladapter_unittest.cc
>@@ -10,7 +10,9 @@
> 
> #include <memory>
> #include <string>
>+#include <utility>
> 
>+#include "absl/memory/memory.h"
> #include "rtc_base/gunit.h"
> #include "rtc_base/ipaddress.h"
> #include "rtc_base/socketstream.h"
>@@ -20,16 +22,20 @@
> #include "rtc_base/stream.h"
> #include "rtc_base/stringencode.h"
> #include "rtc_base/virtualsocketserver.h"
>+#include "test/gmock.h"
>+
>+using ::testing::_;
>+using ::testing::Return;
> 
> static const int kTimeout = 5000;
> 
> static rtc::AsyncSocket* CreateSocket(const rtc::SSLMode& ssl_mode) {
>   rtc::SocketAddress address(rtc::IPAddress(INADDR_ANY), 0);
> 
>-  rtc::AsyncSocket* socket = rtc::Thread::Current()->
>-      socketserver()->CreateAsyncSocket(
>-      address.family(), (ssl_mode == rtc::SSL_MODE_DTLS) ?
>-      SOCK_DGRAM : SOCK_STREAM);
>+  rtc::AsyncSocket* socket =
>+      rtc::Thread::Current()->socketserver()->CreateAsyncSocket(
>+          address.family(),
>+          (ssl_mode == rtc::SSL_MODE_DTLS) ? SOCK_DGRAM : SOCK_STREAM);
>   socket->Bind(address);
> 
>   return socket;
>@@ -39,6 +45,15 @@ static std::string GetSSLProtocolName(const rtc::SSLMode& ssl_mode) {
>   return (ssl_mode == rtc::SSL_MODE_DTLS) ? "DTLS" : "TLS";
> }
> 
>+// Simple mock for the certificate verifier.
>+class MockCertVerifier : public rtc::SSLCertificateVerifier {
>+ public:
>+  virtual ~MockCertVerifier() = default;
>+  MOCK_METHOD1(Verify, bool(const rtc::SSLCertificate&));
>+};
>+
>+// TODO(benwright) - Move to using INSTANTIATE_TEST_CASE_P instead of using
>+// duplicate test cases for simple parameter changes.
> class SSLAdapterTestDummyClient : public sigslot::has_slots<> {
>  public:
>   explicit SSLAdapterTestDummyClient(const rtc::SSLMode& ssl_mode)
>@@ -54,10 +69,18 @@ class SSLAdapterTestDummyClient : public sigslot::has_slots<> {
>     // NEVER USE THIS IN PRODUCTION CODE!
>     ssl_adapter_->SetIgnoreBadCert(true);
> 
>-    ssl_adapter_->SignalReadEvent.connect(this,
>-        &SSLAdapterTestDummyClient::OnSSLAdapterReadEvent);
>-    ssl_adapter_->SignalCloseEvent.connect(this,
>-        &SSLAdapterTestDummyClient::OnSSLAdapterCloseEvent);
>+    ssl_adapter_->SignalReadEvent.connect(
>+        this, &SSLAdapterTestDummyClient::OnSSLAdapterReadEvent);
>+    ssl_adapter_->SignalCloseEvent.connect(
>+        this, &SSLAdapterTestDummyClient::OnSSLAdapterCloseEvent);
>+  }
>+
>+  void SetIgnoreBadCert(bool ignore_bad_cert) {
>+    ssl_adapter_->SetIgnoreBadCert(ignore_bad_cert);
>+  }
>+
>+  void SetCertVerifier(rtc::SSLCertificateVerifier* ssl_cert_verifier) {
>+    ssl_adapter_->SetCertVerifier(ssl_cert_verifier);
>   }
> 
>   void SetAlpnProtocols(const std::vector<std::string>& protos) {
>@@ -76,12 +99,10 @@ class SSLAdapterTestDummyClient : public sigslot::has_slots<> {
>     return ssl_adapter_->GetState();
>   }
> 
>-  const std::string& GetReceivedData() const {
>-    return data_;
>-  }
>+  const std::string& GetReceivedData() const { return data_; }
> 
>   int Connect(const std::string& hostname, const rtc::SocketAddress& address) {
>-    RTC_LOG(LS_INFO) << "Initiating connection with " << address;
>+    RTC_LOG(LS_INFO) << "Initiating connection with " << address.ToString();
> 
>     int rv = ssl_adapter_->Connect(address);
> 
>@@ -97,9 +118,7 @@ class SSLAdapterTestDummyClient : public sigslot::has_slots<> {
>     return rv;
>   }
> 
>-  int Close() {
>-    return ssl_adapter_->Close();
>-  }
>+  int Close() { return ssl_adapter_->Close(); }
> 
>   int Send(const std::string& message) {
>     RTC_LOG(LS_INFO) << "Client sending '" << message << "'";
>@@ -149,15 +168,15 @@ class SSLAdapterTestDummyServer : public sigslot::has_slots<> {
>     server_socket_.reset(CreateSocket(ssl_mode_));
> 
>     if (ssl_mode_ == rtc::SSL_MODE_TLS) {
>-      server_socket_->SignalReadEvent.connect(this,
>-          &SSLAdapterTestDummyServer::OnServerSocketReadEvent);
>+      server_socket_->SignalReadEvent.connect(
>+          this, &SSLAdapterTestDummyServer::OnServerSocketReadEvent);
> 
>       server_socket_->Listen(1);
>     }
> 
>     RTC_LOG(LS_INFO) << ((ssl_mode_ == rtc::SSL_MODE_DTLS) ? "UDP" : "TCP")
>                      << " server listening on "
>-                     << server_socket_->GetLocalAddress();
>+                     << server_socket_->GetLocalAddress().ToString();
>   }
> 
>   rtc::SocketAddress GetAddress() const {
>@@ -170,9 +189,7 @@ class SSLAdapterTestDummyServer : public sigslot::has_slots<> {
>     return "example.com";
>   }
> 
>-  const std::string& GetReceivedData() const {
>-    return data_;
>-  }
>+  const std::string& GetReceivedData() const { return data_; }
> 
>   int Send(const std::string& message) {
>     if (ssl_stream_adapter_ == nullptr ||
>@@ -186,8 +203,8 @@ class SSLAdapterTestDummyServer : public sigslot::has_slots<> {
>     size_t written;
>     int error;
> 
>-    rtc::StreamResult r = ssl_stream_adapter_->Write(message.data(),
>-        message.length(), &written, &error);
>+    rtc::StreamResult r = ssl_stream_adapter_->Write(
>+        message.data(), message.length(), &written, &error);
>     if (r == rtc::SR_SUCCESS) {
>       return written;
>     } else {
>@@ -257,12 +274,12 @@ class SSLAdapterTestDummyServer : public sigslot::has_slots<> {
>     unsigned char digest[20];
>     size_t digest_len = sizeof(digest);
>     ssl_stream_adapter_->SetPeerCertificateDigest(rtc::DIGEST_SHA_1, digest,
>-        digest_len);
>+                                                  digest_len);
> 
>     ssl_stream_adapter_->StartSSL();
> 
>-    ssl_stream_adapter_->SignalEvent.connect(this,
>-        &SSLAdapterTestDummyServer::OnSSLStreamAdapterEvent);
>+    ssl_stream_adapter_->SignalEvent.connect(
>+        this, &SSLAdapterTestDummyServer::OnSSLStreamAdapterEvent);
>   }
> 
>   const rtc::SSLMode ssl_mode_;
>@@ -275,8 +292,7 @@ class SSLAdapterTestDummyServer : public sigslot::has_slots<> {
>   std::string data_;
> };
> 
>-class SSLAdapterTestBase : public testing::Test,
>-                           public sigslot::has_slots<> {
>+class SSLAdapterTestBase : public testing::Test, public sigslot::has_slots<> {
>  public:
>   explicit SSLAdapterTestBase(const rtc::SSLMode& ssl_mode,
>                               const rtc::KeyParams& key_params)
>@@ -287,8 +303,14 @@ class SSLAdapterTestBase : public testing::Test,
>         client_(new SSLAdapterTestDummyClient(ssl_mode_)),
>         handshake_wait_(kTimeout) {}
> 
>-  void SetHandshakeWait(int wait) {
>-    handshake_wait_ = wait;
>+  void SetHandshakeWait(int wait) { handshake_wait_ = wait; }
>+
>+  void SetIgnoreBadCert(bool ignore_bad_cert) {
>+    client_->SetIgnoreBadCert(ignore_bad_cert);
>+  }
>+
>+  void SetCertVerifier(rtc::SSLCertificateVerifier* ssl_cert_verifier) {
>+    client_->SetCertVerifier(ssl_cert_verifier);
>   }
> 
>   void SetAlpnProtocols(const std::vector<std::string>& protos) {
>@@ -299,6 +321,16 @@ class SSLAdapterTestBase : public testing::Test,
>     client_->SetEllipticCurves(curves);
>   }
> 
>+  void SetMockCertVerifier(bool return_value) {
>+    auto mock_verifier = absl::make_unique<MockCertVerifier>();
>+    EXPECT_CALL(*mock_verifier, Verify(_)).WillRepeatedly(Return(return_value));
>+    cert_verifier_ =
>+        std::unique_ptr<rtc::SSLCertificateVerifier>(std::move(mock_verifier));
>+
>+    SetIgnoreBadCert(false);
>+    SetCertVerifier(cert_verifier_.get());
>+  }
>+
>   void TestHandshake(bool expect_success) {
>     int rv;
> 
>@@ -320,7 +352,7 @@ class SSLAdapterTestBase : public testing::Test,
>       // If expecting success, the client should end up in the CS_CONNECTED
>       // state after handshake.
>       EXPECT_EQ_WAIT(rtc::AsyncSocket::CS_CONNECTED, client_->GetState(),
>-          handshake_wait_);
>+                     handshake_wait_);
> 
>       RTC_LOG(LS_INFO) << GetSSLProtocolName(ssl_mode_)
>                        << " handshake complete.";
>@@ -328,7 +360,7 @@ class SSLAdapterTestBase : public testing::Test,
>     } else {
>       // On handshake failure the client should end up in the CS_CLOSED state.
>       EXPECT_EQ_WAIT(rtc::AsyncSocket::CS_CLOSED, client_->GetState(),
>-          handshake_wait_);
>+                     handshake_wait_);
> 
>       RTC_LOG(LS_INFO) << GetSSLProtocolName(ssl_mode_) << " handshake failed.";
>     }
>@@ -359,6 +391,7 @@ class SSLAdapterTestBase : public testing::Test,
>   rtc::AutoSocketServerThread thread_;
>   std::unique_ptr<SSLAdapterTestDummyServer> server_;
>   std::unique_ptr<SSLAdapterTestDummyClient> client_;
>+  std::unique_ptr<rtc::SSLCertificateVerifier> cert_verifier_;
> 
>   int handshake_wait_;
> };
>@@ -394,9 +427,34 @@ TEST_F(SSLAdapterTestTLS_RSA, TestTLSConnect) {
>   TestHandshake(true);
> }
> 
>+// Test that handshake works with a custom verifier that returns true. RSA.
>+TEST_F(SSLAdapterTestTLS_RSA, TestTLSConnectCustomCertVerifierSucceeds) {
>+  SetMockCertVerifier(/*return_value=*/true);
>+  TestHandshake(/*expect_success=*/true);
>+}
>+
>+// Test that handshake fails with a custom verifier that returns false. RSA.
>+TEST_F(SSLAdapterTestTLS_RSA, TestTLSConnectCustomCertVerifierFails) {
>+  SetMockCertVerifier(/*return_value=*/false);
>+  TestHandshake(/*expect_success=*/false);
>+}
>+
> // Test that handshake works, using ECDSA
> TEST_F(SSLAdapterTestTLS_ECDSA, TestTLSConnect) {
>-  TestHandshake(true);
>+  SetMockCertVerifier(/*return_value=*/true);
>+  TestHandshake(/*expect_success=*/true);
>+}
>+
>+// Test that handshake works with a custom verifier that returns true. ECDSA.
>+TEST_F(SSLAdapterTestTLS_ECDSA, TestTLSConnectCustomCertVerifierSucceeds) {
>+  SetMockCertVerifier(/*return_value=*/true);
>+  TestHandshake(/*expect_success=*/true);
>+}
>+
>+// Test that handshake fails with a custom verifier that returns false. ECDSA.
>+TEST_F(SSLAdapterTestTLS_ECDSA, TestTLSConnectCustomCertVerifierFails) {
>+  SetMockCertVerifier(/*return_value=*/false);
>+  TestHandshake(/*expect_success=*/false);
> }
> 
> // Test transfer between client and server, using RSA
>@@ -405,6 +463,13 @@ TEST_F(SSLAdapterTestTLS_RSA, TestTLSTransfer) {
>   TestTransfer("Hello, world!");
> }
> 
>+// Test transfer between client and server, using RSA with custom cert verifier.
>+TEST_F(SSLAdapterTestTLS_RSA, TestTLSTransferCustomCertVerifier) {
>+  SetMockCertVerifier(/*return_value=*/true);
>+  TestHandshake(/*expect_success=*/true);
>+  TestTransfer("Hello, world!");
>+}
>+
> TEST_F(SSLAdapterTestTLS_RSA, TestTLSTransferWithBlockedSocket) {
>   TestHandshake(true);
> 
>@@ -452,6 +517,14 @@ TEST_F(SSLAdapterTestTLS_ECDSA, TestTLSTransfer) {
>   TestTransfer("Hello, world!");
> }
> 
>+// Test transfer between client and server, using ECDSA with custom cert
>+// verifier.
>+TEST_F(SSLAdapterTestTLS_ECDSA, TestTLSTransferCustomCertVerifier) {
>+  SetMockCertVerifier(/*return_value=*/true);
>+  TestHandshake(/*expect_success=*/true);
>+  TestTransfer("Hello, world!");
>+}
>+
> // Test transfer using ALPN with protos as h2 and http/1.1
> TEST_F(SSLAdapterTestTLS_ECDSA, TestTLSALPN) {
>   std::vector<std::string> alpn_protos{"h2", "http/1.1"};
>@@ -475,19 +548,61 @@ TEST_F(SSLAdapterTestDTLS_RSA, TestDTLSConnect) {
>   TestHandshake(true);
> }
> 
>+// Test that handshake works with a custom verifier that returns true. DTLS_RSA.
>+TEST_F(SSLAdapterTestDTLS_RSA, TestDTLSConnectCustomCertVerifierSucceeds) {
>+  SetMockCertVerifier(/*return_value=*/true);
>+  TestHandshake(/*expect_success=*/true);
>+}
>+
>+// Test that handshake fails with a custom verifier that returns false.
>+// DTLS_RSA.
>+TEST_F(SSLAdapterTestDTLS_RSA, TestTLSConnectCustomCertVerifierFails) {
>+  SetMockCertVerifier(/*return_value=*/false);
>+  TestHandshake(/*expect_success=*/false);
>+}
>+
> // Test that handshake works, using ECDSA
> TEST_F(SSLAdapterTestDTLS_ECDSA, TestDTLSConnect) {
>   TestHandshake(true);
> }
> 
>+// Test that handshake works with a custom verifier that returns true.
>+// DTLS_ECDSA.
>+TEST_F(SSLAdapterTestDTLS_ECDSA, TestDTLSConnectCustomCertVerifierSucceeds) {
>+  SetMockCertVerifier(/*return_value=*/true);
>+  TestHandshake(/*expect_success=*/true);
>+}
>+
>+// Test that handshake fails with a custom verifier that returns false.
>+// DTLS_ECDSA.
>+TEST_F(SSLAdapterTestDTLS_ECDSA, TestTLSConnectCustomCertVerifierFails) {
>+  SetMockCertVerifier(/*return_value=*/false);
>+  TestHandshake(/*expect_success=*/false);
>+}
>+
> // Test transfer between client and server, using RSA
> TEST_F(SSLAdapterTestDTLS_RSA, TestDTLSTransfer) {
>   TestHandshake(true);
>   TestTransfer("Hello, world!");
> }
> 
>+// Test transfer between client and server, using RSA with custom cert verifier.
>+TEST_F(SSLAdapterTestDTLS_RSA, TestDTLSTransferCustomCertVerifier) {
>+  SetMockCertVerifier(/*return_value=*/true);
>+  TestHandshake(/*expect_success=*/true);
>+  TestTransfer("Hello, world!");
>+}
>+
> // Test transfer between client and server, using ECDSA
> TEST_F(SSLAdapterTestDTLS_ECDSA, TestDTLSTransfer) {
>   TestHandshake(true);
>   TestTransfer("Hello, world!");
> }
>+
>+// Test transfer between client and server, using ECDSA with custom cert
>+// verifier.
>+TEST_F(SSLAdapterTestDTLS_ECDSA, TestDTLSTransferCustomCertVerifier) {
>+  SetMockCertVerifier(/*return_value=*/true);
>+  TestHandshake(/*expect_success=*/true);
>+  TestTransfer("Hello, world!");
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslcertificate.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslcertificate.cc
>new file mode 100644
>index 00000000000..9a38fc0384b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslcertificate.cc
>@@ -0,0 +1,142 @@
>+/*
>+ *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "rtc_base/sslcertificate.h"
>+
>+#include <ctime>
>+#include <string>
>+#include <utility>
>+
>+#include "absl/memory/memory.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/logging.h"
>+#include "rtc_base/opensslcertificate.h"
>+#include "rtc_base/sslfingerprint.h"
>+#include "rtc_base/third_party/base64/base64.h"
>+
>+namespace rtc {
>+
>+//////////////////////////////////////////////////////////////////////
>+// SSLCertificateStats
>+//////////////////////////////////////////////////////////////////////
>+
>+SSLCertificateStats::SSLCertificateStats(
>+    std::string&& fingerprint,
>+    std::string&& fingerprint_algorithm,
>+    std::string&& base64_certificate,
>+    std::unique_ptr<SSLCertificateStats>&& issuer)
>+    : fingerprint(std::move(fingerprint)),
>+      fingerprint_algorithm(std::move(fingerprint_algorithm)),
>+      base64_certificate(std::move(base64_certificate)),
>+      issuer(std::move(issuer)) {}
>+
>+SSLCertificateStats::~SSLCertificateStats() {}
>+
>+//////////////////////////////////////////////////////////////////////
>+// SSLCertificate
>+//////////////////////////////////////////////////////////////////////
>+
>+std::unique_ptr<SSLCertificateStats> SSLCertificate::GetStats() const {
>+  // TODO(bemasc): Move this computation to a helper class that caches these
>+  // values to reduce CPU use in |StatsCollector::GetStats|. This will require
>+  // adding a fast |SSLCertificate::Equals| to detect certificate changes.
>+  std::string digest_algorithm;
>+  if (!GetSignatureDigestAlgorithm(&digest_algorithm))
>+    return nullptr;
>+
>+  // |SSLFingerprint::Create| can fail if the algorithm returned by
>+  // |SSLCertificate::GetSignatureDigestAlgorithm| is not supported by the
>+  // implementation of |SSLCertificate::ComputeDigest|. This currently happens
>+  // with MD5- and SHA-224-signed certificates when linked to libNSS.
>+  std::unique_ptr<SSLFingerprint> ssl_fingerprint(
>+      SSLFingerprint::Create(digest_algorithm, this));
>+  if (!ssl_fingerprint)
>+    return nullptr;
>+  std::string fingerprint = ssl_fingerprint->GetRfc4572Fingerprint();
>+
>+  Buffer der_buffer;
>+  ToDER(&der_buffer);
>+  std::string der_base64;
>+  Base64::EncodeFromArray(der_buffer.data(), der_buffer.size(), &der_base64);
>+
>+  return absl::make_unique<SSLCertificateStats>(std::move(fingerprint),
>+                                                std::move(digest_algorithm),
>+                                                std::move(der_base64), nullptr);
>+}
>+
>+std::unique_ptr<SSLCertificate> SSLCertificate::GetUniqueReference() const {
>+  return absl::WrapUnique(GetReference());
>+}
>+
>+//////////////////////////////////////////////////////////////////////
>+// SSLCertChain
>+//////////////////////////////////////////////////////////////////////
>+
>+SSLCertChain::SSLCertChain(std::vector<std::unique_ptr<SSLCertificate>> certs)
>+    : certs_(std::move(certs)) {}
>+
>+SSLCertChain::SSLCertChain(const std::vector<SSLCertificate*>& certs) {
>+  RTC_DCHECK(!certs.empty());
>+  certs_.resize(certs.size());
>+  std::transform(
>+      certs.begin(), certs.end(), certs_.begin(),
>+      [](const SSLCertificate* cert) -> std::unique_ptr<SSLCertificate> {
>+        return cert->GetUniqueReference();
>+      });
>+}
>+
>+SSLCertChain::SSLCertChain(const SSLCertificate* cert) {
>+  certs_.push_back(cert->GetUniqueReference());
>+}
>+
>+SSLCertChain::SSLCertChain(SSLCertChain&& rhs) = default;
>+
>+SSLCertChain& SSLCertChain::operator=(SSLCertChain&&) = default;
>+
>+SSLCertChain::~SSLCertChain() {}
>+
>+SSLCertChain* SSLCertChain::Copy() const {
>+  std::vector<std::unique_ptr<SSLCertificate>> new_certs(certs_.size());
>+  std::transform(certs_.begin(), certs_.end(), new_certs.begin(),
>+                 [](const std::unique_ptr<SSLCertificate>& cert)
>+                     -> std::unique_ptr<SSLCertificate> {
>+                   return cert->GetUniqueReference();
>+                 });
>+  return new SSLCertChain(std::move(new_certs));
>+}
>+
>+std::unique_ptr<SSLCertChain> SSLCertChain::UniqueCopy() const {
>+  return absl::WrapUnique(Copy());
>+}
>+
>+std::unique_ptr<SSLCertificateStats> SSLCertChain::GetStats() const {
>+  // We have a linked list of certificates, starting with the first element of
>+  // |certs_| and ending with the last element of |certs_|. The "issuer" of a
>+  // certificate is the next certificate in the chain. Stats are produced for
>+  // each certificate in the list. Here, the "issuer" is the issuer's stats.
>+  std::unique_ptr<SSLCertificateStats> issuer;
>+  // The loop runs in reverse so that the |issuer| is known before the
>+  // certificate issued by |issuer|.
>+  for (ptrdiff_t i = certs_.size() - 1; i >= 0; --i) {
>+    std::unique_ptr<SSLCertificateStats> new_stats = certs_[i]->GetStats();
>+    if (new_stats) {
>+      new_stats->issuer = std::move(issuer);
>+    }
>+    issuer = std::move(new_stats);
>+  }
>+  return issuer;
>+}
>+
>+// static
>+SSLCertificate* SSLCertificate::FromPEMString(const std::string& pem_string) {
>+  return OpenSSLCertificate::FromPEMString(pem_string);
>+}
>+
>+}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslcertificate.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslcertificate.h
>new file mode 100644
>index 00000000000..29c4db58ef5
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslcertificate.h
>@@ -0,0 +1,147 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+// Generic interface for SSL Certificates, used in both the SSLAdapter
>+// for TLS TURN connections and the SSLStreamAdapter for DTLS Peer to Peer
>+// Connections for SRTP Key negotiation and SCTP encryption.
>+
>+#ifndef RTC_BASE_SSLCERTIFICATE_H_
>+#define RTC_BASE_SSLCERTIFICATE_H_
>+
>+#include <algorithm>
>+#include <memory>
>+#include <string>
>+#include <vector>
>+
>+#include "rtc_base/buffer.h"
>+#include "rtc_base/constructormagic.h"
>+#include "rtc_base/messagedigest.h"
>+#include "rtc_base/timeutils.h"
>+
>+namespace rtc {
>+
>+struct SSLCertificateStats {
>+  SSLCertificateStats(std::string&& fingerprint,
>+                      std::string&& fingerprint_algorithm,
>+                      std::string&& base64_certificate,
>+                      std::unique_ptr<SSLCertificateStats>&& issuer);
>+  ~SSLCertificateStats();
>+  std::string fingerprint;
>+  std::string fingerprint_algorithm;
>+  std::string base64_certificate;
>+  std::unique_ptr<SSLCertificateStats> issuer;
>+};
>+
>+// Abstract interface overridden by SSL library specific
>+// implementations.
>+
>+// A somewhat opaque type used to encapsulate a certificate.
>+// Wraps the SSL library's notion of a certificate, with reference counting.
>+// The SSLCertificate object is pretty much immutable once created.
>+// (The OpenSSL implementation only does reference counting and
>+// possibly caching of intermediate results.)
>+class SSLCertificate {
>+ public:
>+  // Parses and builds a certificate from a PEM encoded string.
>+  // Returns null on failure.
>+  // The length of the string representation of the certificate is
>+  // stored in *pem_length if it is non-null, and only if
>+  // parsing was successful.
>+  // Caller is responsible for freeing the returned object.
>+  static SSLCertificate* FromPEMString(const std::string& pem_string);
>+  virtual ~SSLCertificate() {}
>+
>+  // Returns a new SSLCertificate object instance wrapping the same
>+  // underlying certificate, including its chain if present.  Caller is
>+  // responsible for freeing the returned object. Use GetUniqueReference
>+  // instead.
>+  virtual SSLCertificate* GetReference() const = 0;
>+
>+  std::unique_ptr<SSLCertificate> GetUniqueReference() const;
>+
>+  // Returns a PEM encoded string representation of the certificate.
>+  virtual std::string ToPEMString() const = 0;
>+
>+  // Provides a DER encoded binary representation of the certificate.
>+  virtual void ToDER(Buffer* der_buffer) const = 0;
>+
>+  // Gets the name of the digest algorithm that was used to compute this
>+  // certificate's signature.
>+  virtual bool GetSignatureDigestAlgorithm(std::string* algorithm) const = 0;
>+
>+  // Compute the digest of the certificate given algorithm
>+  virtual bool ComputeDigest(const std::string& algorithm,
>+                             unsigned char* digest,
>+                             size_t size,
>+                             size_t* length) const = 0;
>+
>+  // Returns the time in seconds relative to epoch, 1970-01-01T00:00:00Z (UTC),
>+  // or -1 if an expiration time could not be retrieved.
>+  virtual int64_t CertificateExpirationTime() const = 0;
>+
>+  // Gets information (fingerprint, etc.) about this certificate. This is used
>+  // for certificate stats, see
>+  // https://w3c.github.io/webrtc-stats/#certificatestats-dict*.
>+  std::unique_ptr<SSLCertificateStats> GetStats() const;
>+};
>+
>+// SSLCertChain is a simple wrapper for a vector of SSLCertificates. It serves
>+// primarily to ensure proper memory management (especially deletion) of the
>+// SSLCertificate pointers.
>+class SSLCertChain {
>+ public:
>+  explicit SSLCertChain(std::vector<std::unique_ptr<SSLCertificate>> certs);
>+  // These constructors copy the provided SSLCertificate(s), so the caller
>+  // retains ownership.
>+  explicit SSLCertChain(const std::vector<SSLCertificate*>& certs);
>+  explicit SSLCertChain(const SSLCertificate* cert);
>+  // Allow move semantics for the object.
>+  SSLCertChain(SSLCertChain&&);
>+  SSLCertChain& operator=(SSLCertChain&&);
>+
>+  ~SSLCertChain();
>+
>+  // Vector access methods.
>+  size_t GetSize() const { return certs_.size(); }
>+
>+  // Returns a temporary reference, only valid until the chain is destroyed.
>+  const SSLCertificate& Get(size_t pos) const { return *(certs_[pos]); }
>+
>+  // Returns a new SSLCertChain object instance wrapping the same underlying
>+  // certificate chain.  Caller is responsible for freeing the returned object.
>+  SSLCertChain* Copy() const;
>+  // Same as above, but returning a unique_ptr for convenience.
>+  std::unique_ptr<SSLCertChain> UniqueCopy() const;
>+
>+  // Gets information (fingerprint, etc.) about this certificate chain. This is
>+  // used for certificate stats, see
>+  // https://w3c.github.io/webrtc-stats/#certificatestats-dict*.
>+  std::unique_ptr<SSLCertificateStats> GetStats() const;
>+
>+ private:
>+  std::vector<std::unique_ptr<SSLCertificate>> certs_;
>+
>+  RTC_DISALLOW_COPY_AND_ASSIGN(SSLCertChain);
>+};
>+
>+// SSLCertificateVerifier provides a simple interface to allow third parties to
>+// define their own certificate verification code. It is completely independent
>+// from the underlying SSL implementation.
>+class SSLCertificateVerifier {
>+ public:
>+  virtual ~SSLCertificateVerifier() = default;
>+  // Returns true if the certificate is valid, else false. It is up to the
>+  // implementer to define what a valid certificate looks like.
>+  virtual bool Verify(const SSLCertificate& certificate) = 0;
>+};
>+
>+}  // namespace rtc
>+
>+#endif  // RTC_BASE_SSLCERTIFICATE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslfingerprint.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslfingerprint.cc
>index dda46f18bc1..b651a3d6285 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslfingerprint.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslfingerprint.cc
>@@ -20,8 +20,8 @@
> 
> namespace rtc {
> 
>-SSLFingerprint* SSLFingerprint::Create(
>-    const std::string& algorithm, const rtc::SSLIdentity* identity) {
>+SSLFingerprint* SSLFingerprint::Create(const std::string& algorithm,
>+                                       const rtc::SSLIdentity* identity) {
>   if (!identity) {
>     return nullptr;
>   }
>@@ -29,12 +29,12 @@ SSLFingerprint* SSLFingerprint::Create(
>   return Create(algorithm, &(identity->certificate()));
> }
> 
>-SSLFingerprint* SSLFingerprint::Create(
>-    const std::string& algorithm, const rtc::SSLCertificate* cert) {
>+SSLFingerprint* SSLFingerprint::Create(const std::string& algorithm,
>+                                       const rtc::SSLCertificate* cert) {
>   uint8_t digest_val[64];
>   size_t digest_len;
>-  bool ret = cert->ComputeDigest(
>-      algorithm, digest_val, sizeof(digest_val), &digest_len);
>+  bool ret = cert->ComputeDigest(algorithm, digest_val, sizeof(digest_val),
>+                                 &digest_len);
>   if (!ret) {
>     return nullptr;
>   }
>@@ -43,7 +43,8 @@ SSLFingerprint* SSLFingerprint::Create(
> }
> 
> SSLFingerprint* SSLFingerprint::CreateFromRfc4572(
>-    const std::string& algorithm, const std::string& fingerprint) {
>+    const std::string& algorithm,
>+    const std::string& fingerprint) {
>   if (algorithm.empty() || !rtc::IsFips180DigestAlgorithm(algorithm))
>     return nullptr;
> 
>@@ -52,10 +53,8 @@ SSLFingerprint* SSLFingerprint::CreateFromRfc4572(
> 
>   size_t value_len;
>   char value[rtc::MessageDigest::kMaxSize];
>-  value_len = rtc::hex_decode_with_delimiter(value, sizeof(value),
>-                                                   fingerprint.c_str(),
>-                                                   fingerprint.length(),
>-                                                   ':');
>+  value_len = rtc::hex_decode_with_delimiter(
>+      value, sizeof(value), fingerprint.c_str(), fingerprint.length(), ':');
>   if (!value_len)
>     return nullptr;
> 
>@@ -91,15 +90,14 @@ SSLFingerprint::SSLFingerprint(const SSLFingerprint& from)
>     : algorithm(from.algorithm), digest(from.digest) {}
> 
> bool SSLFingerprint::operator==(const SSLFingerprint& other) const {
>-  return algorithm == other.algorithm &&
>-         digest == other.digest;
>+  return algorithm == other.algorithm && digest == other.digest;
> }
> 
> std::string SSLFingerprint::GetRfc4572Fingerprint() const {
>   std::string fingerprint =
>       rtc::hex_encode_with_delimiter(digest.data<char>(), digest.size(), ':');
>-  std::transform(fingerprint.begin(), fingerprint.end(),
>-                 fingerprint.begin(), ::toupper);
>+  std::transform(fingerprint.begin(), fingerprint.end(), fingerprint.begin(),
>+                 ::toupper);
>   return fingerprint;
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslfingerprint.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslfingerprint.h
>index b5e9b72bb4c..b204bc77f30 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslfingerprint.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslfingerprint.h
>@@ -13,7 +13,6 @@
> 
> #include <string>
> 
>-#include "rtc_base/basictypes.h"
> #include "rtc_base/copyonwritebuffer.h"
> #include "rtc_base/rtccertificate.h"
> #include "rtc_base/sslidentity.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslidentity.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslidentity.cc
>index e035d9ec1bf..f286884f9f1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslidentity.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslidentity.cc
>@@ -15,87 +15,23 @@
> #include <string>
> #include <utility>
> 
>-#include "rtc_base/base64.h"
>+#include "absl/memory/memory.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/opensslidentity.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/sslfingerprint.h"
>+#include "rtc_base/third_party/base64/base64.h"
> 
> namespace rtc {
> 
>+//////////////////////////////////////////////////////////////////////
>+// KeyParams
>+//////////////////////////////////////////////////////////////////////
>+
> const char kPemTypeCertificate[] = "CERTIFICATE";
> const char kPemTypeRsaPrivateKey[] = "RSA PRIVATE KEY";
> const char kPemTypeEcPrivateKey[] = "EC PRIVATE KEY";
> 
>-SSLCertificateStats::SSLCertificateStats(
>-    std::string&& fingerprint,
>-    std::string&& fingerprint_algorithm,
>-    std::string&& base64_certificate,
>-    std::unique_ptr<SSLCertificateStats>&& issuer)
>-    : fingerprint(std::move(fingerprint)),
>-      fingerprint_algorithm(std::move(fingerprint_algorithm)),
>-      base64_certificate(std::move(base64_certificate)),
>-      issuer(std::move(issuer)) {
>-}
>-
>-SSLCertificateStats::~SSLCertificateStats() {
>-}
>-
>-std::unique_ptr<SSLCertificateStats> SSLCertificate::GetStats() const {
>-  // We have a certificate and optionally a chain of certificates. This forms a
>-  // linked list, starting with |this|, then the first element of |chain| and
>-  // ending with the last element of |chain|. The "issuer" of a certificate is
>-  // the next certificate in the chain. Stats are produced for each certificate
>-  // in the list. Here, the "issuer" is the issuer's stats.
>-  std::unique_ptr<SSLCertChain> chain = GetChain();
>-  std::unique_ptr<SSLCertificateStats> issuer;
>-  if (chain) {
>-    // The loop runs in reverse so that the |issuer| is known before the
>-    // |cert|'s stats.
>-    for (ptrdiff_t i = chain->GetSize() - 1; i >= 0; --i) {
>-      const SSLCertificate* cert = &chain->Get(i);
>-      issuer = cert->GetStats(std::move(issuer));
>-    }
>-  }
>-  return GetStats(std::move(issuer));
>-}
>-
>-std::unique_ptr<SSLCertificate> SSLCertificate::GetUniqueReference() const {
>-  return WrapUnique(GetReference());
>-}
>-
>-std::unique_ptr<SSLCertificateStats> SSLCertificate::GetStats(
>-    std::unique_ptr<SSLCertificateStats> issuer) const {
>-  // TODO(bemasc): Move this computation to a helper class that caches these
>-  // values to reduce CPU use in |StatsCollector::GetStats|. This will require
>-  // adding a fast |SSLCertificate::Equals| to detect certificate changes.
>-  std::string digest_algorithm;
>-  if (!GetSignatureDigestAlgorithm(&digest_algorithm))
>-    return nullptr;
>-
>-  // |SSLFingerprint::Create| can fail if the algorithm returned by
>-  // |SSLCertificate::GetSignatureDigestAlgorithm| is not supported by the
>-  // implementation of |SSLCertificate::ComputeDigest|. This currently happens
>-  // with MD5- and SHA-224-signed certificates when linked to libNSS.
>-  std::unique_ptr<SSLFingerprint> ssl_fingerprint(
>-      SSLFingerprint::Create(digest_algorithm, this));
>-  if (!ssl_fingerprint)
>-    return nullptr;
>-  std::string fingerprint = ssl_fingerprint->GetRfc4572Fingerprint();
>-
>-  Buffer der_buffer;
>-  ToDER(&der_buffer);
>-  std::string der_base64;
>-  Base64::EncodeFromArray(der_buffer.data(), der_buffer.size(), &der_base64);
>-
>-  return std::unique_ptr<SSLCertificateStats>(new SSLCertificateStats(
>-      std::move(fingerprint),
>-      std::move(digest_algorithm),
>-      std::move(der_base64),
>-      std::move(issuer)));
>-}
>-
> KeyParams::KeyParams(KeyType key_type) {
>   if (key_type == KT_ECDSA) {
>     type_ = KT_ECDSA;
>@@ -149,6 +85,10 @@ KeyType IntKeyTypeFamilyToKeyType(int key_type_family) {
>   return static_cast<KeyType>(key_type_family);
> }
> 
>+//////////////////////////////////////////////////////////////////////
>+// SSLIdentity
>+//////////////////////////////////////////////////////////////////////
>+
> bool SSLIdentity::PemToDer(const std::string& pem_type,
>                            const std::string& pem_string,
>                            std::string* der) {
>@@ -168,9 +108,8 @@ bool SSLIdentity::PemToDer(const std::string& pem_type,
> 
>   std::string inner = pem_string.substr(body + 1, trailer - (body + 1));
> 
>-  *der = Base64::Decode(inner, Base64::DO_PARSE_WHITE |
>-                        Base64::DO_PAD_ANY |
>-                        Base64::DO_TERM_BUFFER);
>+  *der = Base64::Decode(inner, Base64::DO_PARSE_WHITE | Base64::DO_PAD_ANY |
>+                                   Base64::DO_TERM_BUFFER);
>   return true;
> }
> 
>@@ -199,40 +138,6 @@ std::string SSLIdentity::DerToPem(const std::string& pem_type,
>   return result.str();
> }
> 
>-SSLCertChain::SSLCertChain(std::vector<std::unique_ptr<SSLCertificate>> certs)
>-    : certs_(std::move(certs)) {}
>-
>-SSLCertChain::SSLCertChain(const std::vector<SSLCertificate*>& certs) {
>-  RTC_DCHECK(!certs.empty());
>-  certs_.resize(certs.size());
>-  std::transform(
>-      certs.begin(), certs.end(), certs_.begin(),
>-      [](const SSLCertificate* cert) -> std::unique_ptr<SSLCertificate> {
>-        return cert->GetUniqueReference();
>-      });
>-}
>-
>-SSLCertChain::SSLCertChain(const SSLCertificate* cert) {
>-  certs_.push_back(cert->GetUniqueReference());
>-}
>-
>-SSLCertChain::~SSLCertChain() {}
>-
>-SSLCertChain* SSLCertChain::Copy() const {
>-  std::vector<std::unique_ptr<SSLCertificate>> new_certs(certs_.size());
>-  std::transform(certs_.begin(), certs_.end(), new_certs.begin(),
>-                 [](const std::unique_ptr<SSLCertificate>& cert)
>-                     -> std::unique_ptr<SSLCertificate> {
>-                   return cert->GetUniqueReference();
>-                 });
>-  return new SSLCertChain(std::move(new_certs));
>-}
>-
>-// static
>-SSLCertificate* SSLCertificate::FromPEMString(const std::string& pem_string) {
>-  return OpenSSLCertificate::FromPEMString(pem_string);
>-}
>-
> // static
> SSLIdentity* SSLIdentity::GenerateWithExpiration(const std::string& common_name,
>                                                  const KeyParams& key_params,
>@@ -280,6 +185,10 @@ bool operator!=(const SSLIdentity& a, const SSLIdentity& b) {
>   return !(a == b);
> }
> 
>+//////////////////////////////////////////////////////////////////////
>+// Helper Functions
>+//////////////////////////////////////////////////////////////////////
>+
> // Read |n| bytes from ASN1 number string at *|pp| and return the numeric value.
> // Update *|pp| and *|np| to reflect number of read bytes.
> static inline int ASN1ReadInt(const unsigned char** pp, size_t* np, size_t n) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslidentity.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslidentity.h
>index 952e2ab8c22..1379d733be5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslidentity.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslidentity.h
>@@ -21,114 +21,11 @@
> #include "rtc_base/buffer.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/messagedigest.h"
>+#include "rtc_base/sslcertificate.h"
> #include "rtc_base/timeutils.h"
> 
> namespace rtc {
> 
>-// Forward declaration due to circular dependency with SSLCertificate.
>-class SSLCertChain;
>-
>-struct SSLCertificateStats {
>-  SSLCertificateStats(std::string&& fingerprint,
>-                      std::string&& fingerprint_algorithm,
>-                      std::string&& base64_certificate,
>-                      std::unique_ptr<SSLCertificateStats>&& issuer);
>-  ~SSLCertificateStats();
>-  std::string fingerprint;
>-  std::string fingerprint_algorithm;
>-  std::string base64_certificate;
>-  std::unique_ptr<SSLCertificateStats> issuer;
>-};
>-
>-// Abstract interface overridden by SSL library specific
>-// implementations.
>-
>-// A somewhat opaque type used to encapsulate a certificate.
>-// Wraps the SSL library's notion of a certificate, with reference counting.
>-// The SSLCertificate object is pretty much immutable once created.
>-// (The OpenSSL implementation only does reference counting and
>-// possibly caching of intermediate results.)
>-class SSLCertificate {
>- public:
>-  // Parses and builds a certificate from a PEM encoded string.
>-  // Returns null on failure.
>-  // The length of the string representation of the certificate is
>-  // stored in *pem_length if it is non-null, and only if
>-  // parsing was successful.
>-  // Caller is responsible for freeing the returned object.
>-  static SSLCertificate* FromPEMString(const std::string& pem_string);
>-  virtual ~SSLCertificate() {}
>-
>-  // Returns a new SSLCertificate object instance wrapping the same
>-  // underlying certificate, including its chain if present.  Caller is
>-  // responsible for freeing the returned object. Use GetUniqueReference
>-  // instead.
>-  virtual SSLCertificate* GetReference() const = 0;
>-
>-  std::unique_ptr<SSLCertificate> GetUniqueReference() const;
>-
>-  // Returns null. This is deprecated. Please use
>-  // SSLStreamAdapter::GetPeerSSLCertChain
>-  virtual std::unique_ptr<SSLCertChain> GetChain() const = 0;
>-
>-  // Returns a PEM encoded string representation of the certificate.
>-  virtual std::string ToPEMString() const = 0;
>-
>-  // Provides a DER encoded binary representation of the certificate.
>-  virtual void ToDER(Buffer* der_buffer) const = 0;
>-
>-  // Gets the name of the digest algorithm that was used to compute this
>-  // certificate's signature.
>-  virtual bool GetSignatureDigestAlgorithm(std::string* algorithm) const = 0;
>-
>-  // Compute the digest of the certificate given algorithm
>-  virtual bool ComputeDigest(const std::string& algorithm,
>-                             unsigned char* digest,
>-                             size_t size,
>-                             size_t* length) const = 0;
>-
>-  // Returns the time in seconds relative to epoch, 1970-01-01T00:00:00Z (UTC),
>-  // or -1 if an expiration time could not be retrieved.
>-  virtual int64_t CertificateExpirationTime() const = 0;
>-
>-  // Gets information (fingerprint, etc.) about this certificate and its chain
>-  // (if it has a certificate chain). This is used for certificate stats, see
>-  // https://w3c.github.io/webrtc-stats/#certificatestats-dict*.
>-  std::unique_ptr<SSLCertificateStats> GetStats() const;
>-
>- private:
>-  std::unique_ptr<SSLCertificateStats> GetStats(
>-      std::unique_ptr<SSLCertificateStats> issuer) const;
>-};
>-
>-// SSLCertChain is a simple wrapper for a vector of SSLCertificates. It serves
>-// primarily to ensure proper memory management (especially deletion) of the
>-// SSLCertificate pointers.
>-class SSLCertChain {
>- public:
>-  explicit SSLCertChain(std::vector<std::unique_ptr<SSLCertificate>> certs);
>-  // These constructors copy the provided SSLCertificate(s), so the caller
>-  // retains ownership.
>-  explicit SSLCertChain(const std::vector<SSLCertificate*>& certs);
>-  explicit SSLCertChain(const SSLCertificate* cert);
>-  ~SSLCertChain();
>-
>-  // Vector access methods.
>-  size_t GetSize() const { return certs_.size(); }
>-
>-  // Returns a temporary reference, only valid until the chain is destroyed.
>-  const SSLCertificate& Get(size_t pos) const { return *(certs_[pos]); }
>-
>-  // Returns a new SSLCertChain object instance wrapping the same underlying
>-  // certificate chain.  Caller is responsible for freeing the returned object.
>-  SSLCertChain* Copy() const;
>-
>- private:
>-  std::vector<std::unique_ptr<SSLCertificate>> certs_;
>-
>-  RTC_DISALLOW_COPY_AND_ASSIGN(SSLCertChain);
>-};
>-
> // KT_LAST is intended for vector declarations and loops over all key types;
> // it does not represent any key type in itself.
> // KT_DEFAULT is used as the default KeyType for KeyParams.
>@@ -241,8 +138,10 @@ class SSLIdentity {
>   // TODO(hbos,torbjorng): Rename to a less confusing name.
>   virtual SSLIdentity* GetReference() const = 0;
> 
>-  // Returns a temporary reference to the certificate.
>+  // Returns a temporary reference to the end-entity (leaf) certificate.
>   virtual const SSLCertificate& certificate() const = 0;
>+  // Returns a temporary reference to the entire certificate chain.
>+  virtual const SSLCertChain& cert_chain() const = 0;
>   virtual std::string PrivateKeyToPEMString() const = 0;
>   virtual std::string PublicKeyToPEMString() const = 0;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslidentity_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslidentity_unittest.cc
>index c26d8d73f91..132e240c2f5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslidentity_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslidentity_unittest.cc
>@@ -21,7 +21,8 @@
> 
> using rtc::SSLIdentity;
> 
>-const char kTestCertificate[] = "-----BEGIN CERTIFICATE-----\n"
>+const char kTestCertificate[] =
>+    "-----BEGIN CERTIFICATE-----\n"
>     "MIIB6TCCAVICAQYwDQYJKoZIhvcNAQEEBQAwWzELMAkGA1UEBhMCQVUxEzARBgNV\n"
>     "BAgTClF1ZWVuc2xhbmQxGjAYBgNVBAoTEUNyeXB0U29mdCBQdHkgTHRkMRswGQYD\n"
>     "VQQDExJUZXN0IENBICgxMDI0IGJpdCkwHhcNMDAxMDE2MjIzMTAzWhcNMDMwMTE0\n"
>@@ -35,36 +36,29 @@ const char kTestCertificate[] = "-----BEGIN CERTIFICATE-----\n"
>     "itAE+OjGF+PFKbwX8Q==\n"
>     "-----END CERTIFICATE-----\n";
> 
>-const unsigned char kTestCertSha1[] = {
>-    0xA6, 0xC8, 0x59, 0xEA, 0xC3, 0x7E, 0x6D, 0x33,
>-    0xCF, 0xE2, 0x69, 0x9D, 0x74, 0xE6, 0xF6, 0x8A,
>-    0x9E, 0x47, 0xA7, 0xCA};
>+const unsigned char kTestCertSha1[] = {0xA6, 0xC8, 0x59, 0xEA, 0xC3, 0x7E, 0x6D,
>+                                       0x33, 0xCF, 0xE2, 0x69, 0x9D, 0x74, 0xE6,
>+                                       0xF6, 0x8A, 0x9E, 0x47, 0xA7, 0xCA};
> const unsigned char kTestCertSha224[] = {
>-    0xd4, 0xce, 0xc6, 0xcf, 0x28, 0xcb, 0xe9, 0x77,
>-    0x38, 0x36, 0xcf, 0xb1, 0x3b, 0x4a, 0xd7, 0xbd,
>-    0xae, 0x24, 0x21, 0x08, 0xcf, 0x6a, 0x44, 0x0d,
>-    0x3f, 0x94, 0x2a, 0x5b};
>+    0xd4, 0xce, 0xc6, 0xcf, 0x28, 0xcb, 0xe9, 0x77, 0x38, 0x36,
>+    0xcf, 0xb1, 0x3b, 0x4a, 0xd7, 0xbd, 0xae, 0x24, 0x21, 0x08,
>+    0xcf, 0x6a, 0x44, 0x0d, 0x3f, 0x94, 0x2a, 0x5b};
> const unsigned char kTestCertSha256[] = {
>-    0x41, 0x6b, 0xb4, 0x93, 0x47, 0x79, 0x77, 0x24,
>-    0x77, 0x0b, 0x8b, 0x2e, 0xa6, 0x2b, 0xe0, 0xf9,
>-    0x0a, 0xed, 0x1f, 0x31, 0xa6, 0xf7, 0x5c, 0xa1,
>-    0x5a, 0xc4, 0xb0, 0xa2, 0xa4, 0x78, 0xb9, 0x76};
>+    0x41, 0x6b, 0xb4, 0x93, 0x47, 0x79, 0x77, 0x24, 0x77, 0x0b, 0x8b,
>+    0x2e, 0xa6, 0x2b, 0xe0, 0xf9, 0x0a, 0xed, 0x1f, 0x31, 0xa6, 0xf7,
>+    0x5c, 0xa1, 0x5a, 0xc4, 0xb0, 0xa2, 0xa4, 0x78, 0xb9, 0x76};
> const unsigned char kTestCertSha384[] = {
>-    0x42, 0x31, 0x9a, 0x79, 0x1d, 0xd6, 0x08, 0xbf,
>-    0x3b, 0xba, 0x36, 0xd8, 0x37, 0x4a, 0x9a, 0x75,
>-    0xd3, 0x25, 0x6e, 0x28, 0x92, 0xbe, 0x06, 0xb7,
>-    0xc5, 0xa0, 0x83, 0xe3, 0x86, 0xb1, 0x03, 0xfc,
>-    0x64, 0x47, 0xd6, 0xd8, 0xaa, 0xd9, 0x36, 0x60,
>-    0x04, 0xcc, 0xbe, 0x7d, 0x6a, 0xe8, 0x34, 0x49};
>+    0x42, 0x31, 0x9a, 0x79, 0x1d, 0xd6, 0x08, 0xbf, 0x3b, 0xba, 0x36, 0xd8,
>+    0x37, 0x4a, 0x9a, 0x75, 0xd3, 0x25, 0x6e, 0x28, 0x92, 0xbe, 0x06, 0xb7,
>+    0xc5, 0xa0, 0x83, 0xe3, 0x86, 0xb1, 0x03, 0xfc, 0x64, 0x47, 0xd6, 0xd8,
>+    0xaa, 0xd9, 0x36, 0x60, 0x04, 0xcc, 0xbe, 0x7d, 0x6a, 0xe8, 0x34, 0x49};
> const unsigned char kTestCertSha512[] = {
>-    0x51, 0x1d, 0xec, 0x02, 0x3d, 0x51, 0x45, 0xd3,
>-    0xd8, 0x1d, 0xa4, 0x9d, 0x43, 0xc9, 0xee, 0x32,
>-    0x6f, 0x4f, 0x37, 0xee, 0xab, 0x3f, 0x25, 0xdf,
>-    0x72, 0xfc, 0x61, 0x1a, 0xd5, 0x92, 0xff, 0x6b,
>-    0x28, 0x71, 0x58, 0xb3, 0xe1, 0x8a, 0x18, 0xcf,
>-    0x61, 0x33, 0x0e, 0x14, 0xc3, 0x04, 0xaa, 0x07,
>-    0xf6, 0xa5, 0xda, 0xdc, 0x42, 0x42, 0x22, 0x35,
>-    0xce, 0x26, 0x58, 0x4a, 0x33, 0x6d, 0xbc, 0xb6};
>+    0x51, 0x1d, 0xec, 0x02, 0x3d, 0x51, 0x45, 0xd3, 0xd8, 0x1d, 0xa4,
>+    0x9d, 0x43, 0xc9, 0xee, 0x32, 0x6f, 0x4f, 0x37, 0xee, 0xab, 0x3f,
>+    0x25, 0xdf, 0x72, 0xfc, 0x61, 0x1a, 0xd5, 0x92, 0xff, 0x6b, 0x28,
>+    0x71, 0x58, 0xb3, 0xe1, 0x8a, 0x18, 0xcf, 0x61, 0x33, 0x0e, 0x14,
>+    0xc3, 0x04, 0xaa, 0x07, 0xf6, 0xa5, 0xda, 0xdc, 0x42, 0x42, 0x22,
>+    0x35, 0xce, 0x26, 0x58, 0x4a, 0x33, 0x6d, 0xbc, 0xb6};
> 
> // These PEM strings were created by generating an identity with
> // |SSLIdentity::Generate| and invoking |identity->PrivateKeyToPEMString()|,
>@@ -112,8 +106,7 @@ static const char kRSA_CERT_PEM[] =
> static const char kRSA_FINGERPRINT[] =
>     "3C:E8:B2:70:09:CF:A9:09:5A:F4:EF:8F:8D:8A:32:FF:EA:04:91:BA:6E:D4:17:78:16"
>     ":2A:EE:F9:9A:DD:E2:2B";
>-static const char kRSA_FINGERPRINT_ALGORITHM[] =
>-    "sha-256";
>+static const char kRSA_FINGERPRINT_ALGORITHM[] = "sha-256";
> static const char kRSA_BASE64_CERTIFICATE[] =
>     "MIIBnDCCAQWgAwIBAgIJAOEHLgeWYwrpMA0GCSqGSIb3DQEBCwUAMBAxDjAMBgNVBAMMBXRlc3"
>     "QxMB4XDTE2MDQyNDE4MTAyMloXDTE2MDUyNTE4MTAyMlowEDEOMAwGA1UEAwwFdGVzdDEwgZ8w"
>@@ -147,8 +140,7 @@ static const char kECDSA_CERT_PEM[] =
> static const char kECDSA_FINGERPRINT[] =
>     "9F:47:FA:88:76:3D:18:B8:00:A0:59:9D:C3:5D:34:0B:1F:B8:99:9E:68:DA:F3:A5:DA"
>     ":50:33:A9:FF:4D:31:89";
>-static const char kECDSA_FINGERPRINT_ALGORITHM[] =
>-    "sha-256";
>+static const char kECDSA_FINGERPRINT_ALGORITHM[] = "sha-256";
> static const char kECDSA_BASE64_CERTIFICATE[] =
>     "MIIBFDCBu6ADAgECAgkArpkxjw62sW4wCgYIKoZIzj0EAwIwEDEOMAwGA1UEAwwFdGVzdDMwHh"
>     "cNMTYwNDI0MTgxNDM4WhcNMTYwNTI1MTgxNDM4WjAQMQ4wDAYDVQQDDAV0ZXN0MzBZMBMGByqG"
>@@ -171,35 +163,25 @@ IdentityAndInfo CreateFakeIdentityAndInfoFromDers(
>   info.ders = ders;
>   for (const std::string& der : ders) {
>     info.pems.push_back(rtc::SSLIdentity::DerToPem(
>-        "CERTIFICATE",
>-        reinterpret_cast<const unsigned char*>(der.c_str()),
>+        "CERTIFICATE", reinterpret_cast<const unsigned char*>(der.c_str()),
>         der.length()));
>   }
>-  info.identity.reset(
>-      new rtc::FakeSSLIdentity(rtc::FakeSSLCertificate(info.pems)));
>+  info.identity.reset(new rtc::FakeSSLIdentity(info.pems));
>   // Strip header/footer and newline characters of PEM strings.
>   for (size_t i = 0; i < info.pems.size(); ++i) {
>-    rtc::replace_substrs("-----BEGIN CERTIFICATE-----", 27,
>-                         "", 0, &info.pems[i]);
>-    rtc::replace_substrs("-----END CERTIFICATE-----", 25,
>-                         "", 0, &info.pems[i]);
>-    rtc::replace_substrs("\n", 1,
>-                         "", 0, &info.pems[i]);
>+    rtc::replace_substrs("-----BEGIN CERTIFICATE-----", 27, "", 0,
>+                         &info.pems[i]);
>+    rtc::replace_substrs("-----END CERTIFICATE-----", 25, "", 0, &info.pems[i]);
>+    rtc::replace_substrs("\n", 1, "", 0, &info.pems[i]);
>   }
>-  // Fingerprint of leaf certificate.
>-  std::unique_ptr<rtc::SSLFingerprint> fp(
>-      rtc::SSLFingerprint::Create("sha-1", &info.identity->certificate()));
>-  EXPECT_TRUE(fp);
>-  info.fingerprints.push_back(fp->GetRfc4572Fingerprint());
>-  // Fingerprints of the rest of the chain.
>-  std::unique_ptr<rtc::SSLCertChain> chain =
>-      info.identity->certificate().GetChain();
>-  if (chain) {
>-    for (size_t i = 0; i < chain->GetSize(); i++) {
>-      fp.reset(rtc::SSLFingerprint::Create("sha-1", &chain->Get(i)));
>-      EXPECT_TRUE(fp);
>-      info.fingerprints.push_back(fp->GetRfc4572Fingerprint());
>-    }
>+  // Fingerprints for the whole certificate chain, starting with leaf
>+  // certificate.
>+  const rtc::SSLCertChain& chain = info.identity->cert_chain();
>+  std::unique_ptr<rtc::SSLFingerprint> fp;
>+  for (size_t i = 0; i < chain.GetSize(); i++) {
>+    fp.reset(rtc::SSLFingerprint::Create("sha-1", &chain.Get(i)));
>+    EXPECT_TRUE(fp);
>+    info.fingerprints.push_back(fp->GetRfc4572Fingerprint());
>   }
>   EXPECT_EQ(info.ders.size(), info.fingerprints.size());
>   return info;
>@@ -434,9 +416,11 @@ TEST_F(SSLIdentityTest, PemDerConversion) {
>   std::string der;
>   EXPECT_TRUE(SSLIdentity::PemToDer("CERTIFICATE", kTestCertificate, &der));
> 
>-  EXPECT_EQ(kTestCertificate, SSLIdentity::DerToPem(
>-      "CERTIFICATE",
>-      reinterpret_cast<const unsigned char*>(der.data()), der.length()));
>+  EXPECT_EQ(
>+      kTestCertificate,
>+      SSLIdentity::DerToPem("CERTIFICATE",
>+                            reinterpret_cast<const unsigned char*>(der.data()),
>+                            der.length()));
> }
> 
> TEST_F(SSLIdentityTest, GetSignatureDigestAlgorithm) {
>@@ -477,7 +461,7 @@ TEST_F(SSLIdentityTest, SSLCertificateGetStatsWithChain) {
>   EXPECT_EQ(info.fingerprints.size(), info.ders.size());
> 
>   std::unique_ptr<rtc::SSLCertificateStats> first_stats =
>-      info.identity->certificate().GetStats();
>+      info.identity->cert_chain().GetStats();
>   rtc::SSLCertificateStats* cert_stats = first_stats.get();
>   for (size_t i = 0; i < info.ders.size(); ++i) {
>     EXPECT_EQ(cert_stats->fingerprint, info.fingerprints[i]);
>@@ -505,6 +489,9 @@ class SSLIdentityExpirationTest : public testing::Test {
>       bool long_format;
>       int64_t want;
>     } static const data[] = {
>+        // clang-format off
>+        // clang formatting breaks this nice alignment
>+
>       // Valid examples.
>       {"19700101000000Z",  true,  0},
>       {"700101000000Z",    false, 0},
>@@ -553,6 +540,8 @@ class SSLIdentityExpirationTest : public testing::Test {
>       {"500101000000Z",     false, -1},  // but too old for epoch
>       {"691231235959Z",     false, -1},  // too old for epoch
>       {"19611118043000Z",   false, -1},  // way too old for epoch
>+
>+        // clang-format off
>     };
> 
>     unsigned char buf[20];
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslroots.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslroots.h
>index 7309a0544a7..0dbd19b4a91 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslroots.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslroots.h
>@@ -18,6 +18,10 @@
> // > python tools/sslroots/generate_sslroots.py
> //    https://pki.google.com/roots.pem
> 
>+// clang-format off
>+// Don't bother formatting generated code,
>+// also it would breaks subject/issuer lines.
>+
> /* subject:/C=BE/O=GlobalSign nv-sa/OU=Root CA/CN=GlobalSign Root CA */
> /* issuer :/C=BE/O=GlobalSign nv-sa/OU=Root CA/CN=GlobalSign Root CA */
> 
>@@ -4277,4 +4281,6 @@ const size_t kSSLCertCertificateSizeList[] = {
>   1122,
> };
> 
>+// clang-format on
>+
> #endif  // RTC_BASE_SSLROOTS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslstreamadapter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslstreamadapter.cc
>index b09c1440a46..746ebd50063 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslstreamadapter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslstreamadapter.cc
>@@ -25,16 +25,16 @@ const char CS_AEAD_AES_256_GCM[] = "AEAD_AES_256_GCM";
> 
> std::string SrtpCryptoSuiteToName(int crypto_suite) {
>   switch (crypto_suite) {
>-  case SRTP_AES128_CM_SHA1_32:
>-    return CS_AES_CM_128_HMAC_SHA1_32;
>-  case SRTP_AES128_CM_SHA1_80:
>-    return CS_AES_CM_128_HMAC_SHA1_80;
>-  case SRTP_AEAD_AES_128_GCM:
>-    return CS_AEAD_AES_128_GCM;
>-  case SRTP_AEAD_AES_256_GCM:
>-    return CS_AEAD_AES_256_GCM;
>-  default:
>-    return std::string();
>+    case SRTP_AES128_CM_SHA1_32:
>+      return CS_AES_CM_128_HMAC_SHA1_32;
>+    case SRTP_AES128_CM_SHA1_80:
>+      return CS_AES_CM_128_HMAC_SHA1_80;
>+    case SRTP_AEAD_AES_128_GCM:
>+      return CS_AEAD_AES_128_GCM;
>+    case SRTP_AEAD_AES_256_GCM:
>+      return CS_AEAD_AES_256_GCM;
>+    default:
>+      return std::string();
>   }
> }
> 
>@@ -50,30 +50,31 @@ int SrtpCryptoSuiteFromName(const std::string& crypto_suite) {
>   return SRTP_INVALID_CRYPTO_SUITE;
> }
> 
>-bool GetSrtpKeyAndSaltLengths(int crypto_suite, int *key_length,
>-    int *salt_length) {
>+bool GetSrtpKeyAndSaltLengths(int crypto_suite,
>+                              int* key_length,
>+                              int* salt_length) {
>   switch (crypto_suite) {
>-  case SRTP_AES128_CM_SHA1_32:
>-  case SRTP_AES128_CM_SHA1_80:
>-    // SRTP_AES128_CM_HMAC_SHA1_32 and SRTP_AES128_CM_HMAC_SHA1_80 are defined
>-    // in RFC 5764 to use a 128 bits key and 112 bits salt for the cipher.
>-    *key_length = 16;
>-    *salt_length = 14;
>-    break;
>-  case SRTP_AEAD_AES_128_GCM:
>-    // SRTP_AEAD_AES_128_GCM is defined in RFC 7714 to use a 128 bits key and
>-    // a 96 bits salt for the cipher.
>-    *key_length = 16;
>-    *salt_length = 12;
>-    break;
>-  case SRTP_AEAD_AES_256_GCM:
>-    // SRTP_AEAD_AES_256_GCM is defined in RFC 7714 to use a 256 bits key and
>-    // a 96 bits salt for the cipher.
>-    *key_length = 32;
>-    *salt_length = 12;
>-    break;
>-  default:
>-    return false;
>+    case SRTP_AES128_CM_SHA1_32:
>+    case SRTP_AES128_CM_SHA1_80:
>+      // SRTP_AES128_CM_HMAC_SHA1_32 and SRTP_AES128_CM_HMAC_SHA1_80 are defined
>+      // in RFC 5764 to use a 128 bits key and 112 bits salt for the cipher.
>+      *key_length = 16;
>+      *salt_length = 14;
>+      break;
>+    case SRTP_AEAD_AES_128_GCM:
>+      // SRTP_AEAD_AES_128_GCM is defined in RFC 7714 to use a 128 bits key and
>+      // a 96 bits salt for the cipher.
>+      *key_length = 16;
>+      *salt_length = 12;
>+      break;
>+    case SRTP_AEAD_AES_256_GCM:
>+      // SRTP_AEAD_AES_256_GCM is defined in RFC 7714 to use a 256 bits key and
>+      // a 96 bits salt for the cipher.
>+      *key_length = 32;
>+      *salt_length = 12;
>+      break;
>+    default:
>+      return false;
>   }
>   return true;
> }
>@@ -105,7 +106,11 @@ std::vector<int> GetSupportedDtlsSrtpCryptoSuites(
>   // Note: SRTP_AES128_CM_SHA1_80 is what is required to be supported (by
>   // draft-ietf-rtcweb-security-arch), but SRTP_AES128_CM_SHA1_32 is allowed as
>   // well, and saves a few bytes per packet if it ends up selected.
>-  crypto_suites.push_back(rtc::SRTP_AES128_CM_SHA1_32);
>+  // As the cipher suite is potentially insecure, it will only be used if
>+  // enabled by both peers.
>+  if (crypto_options.enable_aes128_sha1_32_crypto_cipher) {
>+    crypto_suites.push_back(rtc::SRTP_AES128_CM_SHA1_32);
>+  }
>   crypto_suites.push_back(rtc::SRTP_AES128_CM_SHA1_80);
>   return crypto_suites;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslstreamadapter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslstreamadapter.h
>index 560dd591c0e..2d4e19f9b50 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslstreamadapter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslstreamadapter.h
>@@ -22,6 +22,7 @@ namespace rtc {
> 
> // Constants for SSL profile.
> const int TLS_NULL_WITH_NULL_NULL = 0;
>+const int SSL_CIPHER_SUITE_MAX_VALUE = 0xFFFF;
> 
> // Constants for SRTP profiles.
> const int SRTP_INVALID_CRYPTO_SUITE = 0;
>@@ -37,6 +38,7 @@ const int SRTP_AEAD_AES_128_GCM = 0x0007;
> #ifndef SRTP_AEAD_AES_256_GCM
> const int SRTP_AEAD_AES_256_GCM = 0x0008;
> #endif
>+const int SRTP_CRYPTO_SUITE_MAX_VALUE = 0xFFFF;
> 
> // Names of SRTP profiles listed above.
> // 128-bit AES with 80-bit SHA-1 HMAC.
>@@ -80,6 +82,12 @@ struct CryptoOptions {
>   // if both sides enable it.
>   bool enable_gcm_crypto_suites = false;
> 
>+  // If set to true, the (potentially insecure) crypto cipher
>+  // SRTP_AES128_CM_SHA1_32 will be included in the list of supported ciphers
>+  // during negotiation. It will only be used if both peers support it and no
>+  // other ciphers get preferred.
>+  bool enable_aes128_sha1_32_crypto_cipher = false;
>+
>   // If set to true, encrypted RTP header extensions as defined in RFC 6904
>   // will be negotiated. They will only be used if both peers support them.
>   bool enable_encrypted_rtp_header_extensions = false;
>@@ -201,11 +209,7 @@ class SSLStreamAdapter : public StreamAdapterInterface {
>       size_t digest_len,
>       SSLPeerCertificateDigestError* error = nullptr) = 0;
> 
>-  // Retrieves the peer's X.509 certificate, if a connection has been
>-  // established.
>-  virtual std::unique_ptr<SSLCertificate> GetPeerCertificate() const = 0;
>-
>-  // Retrieves the peer's certificate chain including leaf, if a
>+  // Retrieves the peer's certificate chain including leaf certificate, if a
>   // connection has been established.
>   virtual std::unique_ptr<SSLCertChain> GetPeerSSLCertChain() const = 0;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslstreamadapter_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslstreamadapter_unittest.cc
>index 41ff09b62f6..3403bdbe727 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslstreamadapter_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sslstreamadapter_unittest.cc
>@@ -8,7 +8,6 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-
> #include <algorithm>
> #include <memory>
> #include <set>
>@@ -135,22 +134,20 @@ class SSLDummyStreamBase : public rtc::StreamInterface,
>                            public sigslot::has_slots<> {
>  public:
>   SSLDummyStreamBase(SSLStreamAdapterTestBase* test,
>-                     const std::string &side,
>+                     const std::string& side,
>                      rtc::StreamInterface* in,
>-                     rtc::StreamInterface* out) :
>-      test_base_(test),
>-      side_(side),
>-      in_(in),
>-      out_(out),
>-      first_packet_(true) {
>+                     rtc::StreamInterface* out)
>+      : test_base_(test), side_(side), in_(in), out_(out), first_packet_(true) {
>     in_->SignalEvent.connect(this, &SSLDummyStreamBase::OnEventIn);
>     out_->SignalEvent.connect(this, &SSLDummyStreamBase::OnEventOut);
>   }
> 
>   rtc::StreamState GetState() const override { return rtc::SS_OPEN; }
> 
>-  rtc::StreamResult Read(void* buffer, size_t buffer_len,
>-                         size_t* read, int* error) override {
>+  rtc::StreamResult Read(void* buffer,
>+                         size_t buffer_len,
>+                         size_t* read,
>+                         int* error) override {
>     rtc::StreamResult r;
> 
>     r = in_->Read(buffer, buffer_len, read, error);
>@@ -189,13 +186,17 @@ class SSLDummyStreamBase : public rtc::StreamInterface,
>   }
> 
>   // Write to the outgoing FifoBuffer
>-  rtc::StreamResult WriteData(const void* data, size_t data_len,
>-                              size_t* written, int* error) {
>+  rtc::StreamResult WriteData(const void* data,
>+                              size_t data_len,
>+                              size_t* written,
>+                              int* error) {
>     return out_->Write(data, data_len, written, error);
>   }
> 
>-  rtc::StreamResult Write(const void* data, size_t data_len,
>-                          size_t* written, int* error) override;
>+  rtc::StreamResult Write(const void* data,
>+                          size_t data_len,
>+                          size_t* written,
>+                          int* error) override;
> 
>   void Close() override {
>     RTC_LOG(LS_INFO) << "Closing outbound stream";
>@@ -215,17 +216,14 @@ class SSLDummyStreamTLS : public SSLDummyStreamBase {
>   SSLDummyStreamTLS(SSLStreamAdapterTestBase* test,
>                     const std::string& side,
>                     rtc::FifoBuffer* in,
>-                    rtc::FifoBuffer* out) :
>-      SSLDummyStreamBase(test, side, in, out) {
>-  }
>+                    rtc::FifoBuffer* out)
>+      : SSLDummyStreamBase(test, side, in, out) {}
> };
> 
>-class BufferQueueStream : public rtc::BufferQueue,
>-                          public rtc::StreamInterface {
>+class BufferQueueStream : public rtc::BufferQueue, public rtc::StreamInterface {
>  public:
>   BufferQueueStream(size_t capacity, size_t default_size)
>-      : rtc::BufferQueue(capacity, default_size) {
>-  }
>+      : rtc::BufferQueue(capacity, default_size) {}
> 
>   // Implementation of abstract StreamInterface methods.
> 
>@@ -233,8 +231,10 @@ class BufferQueueStream : public rtc::BufferQueue,
>   rtc::StreamState GetState() const override { return rtc::SS_OPEN; }
> 
>   // Reading a buffer queue stream will either succeed or block.
>-  rtc::StreamResult Read(void* buffer, size_t buffer_len,
>-                         size_t* read, int* error) override {
>+  rtc::StreamResult Read(void* buffer,
>+                         size_t buffer_len,
>+                         size_t* read,
>+                         int* error) override {
>     if (!ReadFront(buffer, buffer_len, read)) {
>       return rtc::SR_BLOCK;
>     }
>@@ -242,8 +242,10 @@ class BufferQueueStream : public rtc::BufferQueue,
>   }
> 
>   // Writing to a buffer queue stream will either succeed or block.
>-  rtc::StreamResult Write(const void* data, size_t data_len,
>-                          size_t* written, int* error) override {
>+  rtc::StreamResult Write(const void* data,
>+                          size_t data_len,
>+                          size_t* written,
>+                          int* error) override {
>     if (!WriteBack(data, data_len, written)) {
>       return rtc::SR_BLOCK;
>     }
>@@ -254,13 +256,9 @@ class BufferQueueStream : public rtc::BufferQueue,
>   void Close() override {}
> 
>  protected:
>-  void NotifyReadableForTest() override {
>-    PostEvent(rtc::SE_READ, 0);
>-  }
>+  void NotifyReadableForTest() override { PostEvent(rtc::SE_READ, 0); }
> 
>-  void NotifyWritableForTest() override {
>-    PostEvent(rtc::SE_WRITE, 0);
>-  }
>+  void NotifyWritableForTest() override { PostEvent(rtc::SE_WRITE, 0); }
> };
> 
> class SSLDummyStreamDTLS : public SSLDummyStreamBase {
>@@ -268,9 +266,8 @@ class SSLDummyStreamDTLS : public SSLDummyStreamBase {
>   SSLDummyStreamDTLS(SSLStreamAdapterTestBase* test,
>                      const std::string& side,
>                      BufferQueueStream* in,
>-                     BufferQueueStream* out) :
>-      SSLDummyStreamBase(test, side, in, out) {
>-  }
>+                     BufferQueueStream* out)
>+      : SSLDummyStreamBase(test, side, in, out) {}
> };
> 
> static const int kFifoBufferSize = 4096;
>@@ -372,7 +369,7 @@ class SSLStreamAdapterTestBase : public testing::Test,
>     server_ssl_->SetIdentity(server_identity_);
>   }
> 
>-  virtual void OnEvent(rtc::StreamInterface *stream, int sig, int err) {
>+  virtual void OnEvent(rtc::StreamInterface* stream, int sig, int err) {
>     RTC_LOG(LS_VERBOSE) << "SSLStreamAdapterTestBase::OnEvent sig=" << sig;
> 
>     if (sig & rtc::SE_READ) {
>@@ -433,10 +430,8 @@ class SSLStreamAdapterTestBase : public testing::Test,
>   }
> 
>   void TestHandshake(bool expect_success = true) {
>-    server_ssl_->SetMode(dtls_ ? rtc::SSL_MODE_DTLS :
>-                         rtc::SSL_MODE_TLS);
>-    client_ssl_->SetMode(dtls_ ? rtc::SSL_MODE_DTLS :
>-                         rtc::SSL_MODE_TLS);
>+    server_ssl_->SetMode(dtls_ ? rtc::SSL_MODE_DTLS : rtc::SSL_MODE_TLS);
>+    client_ssl_->SetMode(dtls_ ? rtc::SSL_MODE_DTLS : rtc::SSL_MODE_TLS);
> 
>     if (!dtls_) {
>       // Make sure we simulate a reliable network for TLS.
>@@ -462,8 +457,8 @@ class SSLStreamAdapterTestBase : public testing::Test,
> 
>     // Now run the handshake
>     if (expect_success) {
>-      EXPECT_TRUE_WAIT((client_ssl_->GetState() == rtc::SS_OPEN)
>-                       && (server_ssl_->GetState() == rtc::SS_OPEN),
>+      EXPECT_TRUE_WAIT((client_ssl_->GetState() == rtc::SS_OPEN) &&
>+                           (server_ssl_->GetState() == rtc::SS_OPEN),
>                        handshake_wait_);
>     } else {
>       EXPECT_TRUE_WAIT(client_ssl_->GetState() == rtc::SS_CLOSED,
>@@ -524,9 +519,11 @@ class SSLStreamAdapterTestBase : public testing::Test,
>     }
>   }
> 
>-  rtc::StreamResult DataWritten(SSLDummyStreamBase *from, const void *data,
>-                                size_t data_len, size_t *written,
>-                                int *error) {
>+  rtc::StreamResult DataWritten(SSLDummyStreamBase* from,
>+                                const void* data,
>+                                size_t data_len,
>+                                size_t* written,
>+                                int* error) {
>     // Randomly drop loss_ percent of packets
>     if (rtc::CreateRandomId() % 100 < static_cast<uint32_t>(loss_)) {
>       RTC_LOG(LS_VERBOSE) << "Randomly dropping packet, size=" << data_len;
>@@ -542,7 +539,7 @@ class SSLStreamAdapterTestBase : public testing::Test,
>     // Optionally damage application data (type 23). Note that we don't damage
>     // handshake packets and we damage the last byte to keep the header
>     // intact but break the MAC.
>-    if (damage_ && (*static_cast<const unsigned char *>(data) == 23)) {
>+    if (damage_ && (*static_cast<const unsigned char*>(data) == 23)) {
>       std::vector<char> buf(data_len);
> 
>       RTC_LOG(LS_VERBOSE) << "Damaging packet";
>@@ -556,31 +553,19 @@ class SSLStreamAdapterTestBase : public testing::Test,
>     return from->WriteData(data, data_len, written, error);
>   }
> 
>-  void SetDelay(int delay) {
>-    delay_ = delay;
>-  }
>+  void SetDelay(int delay) { delay_ = delay; }
>   int GetDelay() { return delay_; }
> 
>-  void SetLoseFirstPacket(bool lose) {
>-    lose_first_packet_ = lose;
>-  }
>+  void SetLoseFirstPacket(bool lose) { lose_first_packet_ = lose; }
>   bool GetLoseFirstPacket() { return lose_first_packet_; }
> 
>-  void SetLoss(int percent) {
>-    loss_ = percent;
>-  }
>+  void SetLoss(int percent) { loss_ = percent; }
> 
>-  void SetDamage() {
>-    damage_ = true;
>-  }
>+  void SetDamage() { damage_ = true; }
> 
>-  void SetMtu(size_t mtu) {
>-    mtu_ = mtu;
>-  }
>+  void SetMtu(size_t mtu) { mtu_ = mtu; }
> 
>-  void SetHandshakeWait(int wait) {
>-    handshake_wait_ = wait;
>-  }
>+  void SetHandshakeWait(int wait) { handshake_wait_ = wait; }
> 
>   void SetDtlsSrtpCryptoSuites(const std::vector<int>& ciphers, bool client) {
>     if (client)
>@@ -597,10 +582,13 @@ class SSLStreamAdapterTestBase : public testing::Test,
>   }
> 
>   std::unique_ptr<rtc::SSLCertificate> GetPeerCertificate(bool client) {
>+    std::unique_ptr<rtc::SSLCertChain> chain;
>     if (client)
>-      return client_ssl_->GetPeerCertificate();
>+      chain = client_ssl_->GetPeerSSLCertChain();
>     else
>-      return server_ssl_->GetPeerCertificate();
>+      chain = server_ssl_->GetPeerSSLCertChain();
>+    return (chain && chain->GetSize()) ? chain->Get(0).GetUniqueReference()
>+                                       : nullptr;
>   }
> 
>   bool GetSslCipherSuite(bool client, int* retval) {
>@@ -617,28 +605,24 @@ class SSLStreamAdapterTestBase : public testing::Test,
>       return server_ssl_->GetSslVersion();
>   }
> 
>-  bool ExportKeyingMaterial(const char *label,
>-                            const unsigned char *context,
>+  bool ExportKeyingMaterial(const char* label,
>+                            const unsigned char* context,
>                             size_t context_len,
>                             bool use_context,
>                             bool client,
>-                            unsigned char *result,
>+                            unsigned char* result,
>                             size_t result_len) {
>     if (client)
>-      return client_ssl_->ExportKeyingMaterial(label,
>-                                               context, context_len,
>-                                               use_context,
>-                                               result, result_len);
>+      return client_ssl_->ExportKeyingMaterial(label, context, context_len,
>+                                               use_context, result, result_len);
>     else
>-      return server_ssl_->ExportKeyingMaterial(label,
>-                                               context, context_len,
>-                                               use_context,
>-                                               result, result_len);
>+      return server_ssl_->ExportKeyingMaterial(label, context, context_len,
>+                                               use_context, result, result_len);
>   }
> 
>   // To be implemented by subclasses.
>   virtual void WriteData() = 0;
>-  virtual void ReadData(rtc::StreamInterface *stream) = 0;
>+  virtual void ReadData(rtc::StreamInterface* stream) = 0;
>   virtual void TestTransfer(int size) = 0;
> 
>  protected:
>@@ -646,12 +630,12 @@ class SSLStreamAdapterTestBase : public testing::Test,
>   std::string client_private_key_pem_;
>   rtc::KeyParams client_key_type_;
>   rtc::KeyParams server_key_type_;
>-  SSLDummyStreamBase *client_stream_;  // freed by client_ssl_ destructor
>-  SSLDummyStreamBase *server_stream_;  // freed by server_ssl_ destructor
>+  SSLDummyStreamBase* client_stream_;  // freed by client_ssl_ destructor
>+  SSLDummyStreamBase* server_stream_;  // freed by server_ssl_ destructor
>   std::unique_ptr<rtc::SSLStreamAdapter> client_ssl_;
>   std::unique_ptr<rtc::SSLStreamAdapter> server_ssl_;
>-  rtc::SSLIdentity *client_identity_;  // freed by client_ssl_ destructor
>-  rtc::SSLIdentity *server_identity_;  // freed by server_ssl_ destructor
>+  rtc::SSLIdentity* client_identity_;  // freed by client_ssl_ destructor
>+  rtc::SSLIdentity* server_identity_;  // freed by server_ssl_ destructor
>   int delay_;
>   size_t mtu_;
>   int loss_;
>@@ -673,8 +657,7 @@ class SSLStreamAdapterTestTLS
>                                  ::testing::get<0>(GetParam()),
>                                  ::testing::get<1>(GetParam())),
>         client_buffer_(kFifoBufferSize),
>-        server_buffer_(kFifoBufferSize) {
>-  }
>+        server_buffer_(kFifoBufferSize) {}
> 
>   void CreateStreams() override {
>     client_stream_ =
>@@ -709,8 +692,8 @@ class SSLStreamAdapterTestTLS
>     recv_stream_.GetSize(&received);
> 
>     EXPECT_EQ(static_cast<size_t>(size), received);
>-    EXPECT_EQ(0, memcmp(send_stream_.GetBuffer(),
>-                        recv_stream_.GetBuffer(), size));
>+    EXPECT_EQ(0,
>+              memcmp(send_stream_.GetBuffer(), recv_stream_.GetBuffer(), size));
>   }
> 
>   void WriteData() override {
>@@ -749,7 +732,7 @@ class SSLStreamAdapterTestTLS
>     }
>   };
> 
>-  void ReadData(rtc::StreamInterface *stream) override {
>+  void ReadData(rtc::StreamInterface* stream) override {
>     char buffer[1600];
>     size_t bread;
>     int err2;
>@@ -799,12 +782,13 @@ class SSLStreamAdapterTestDTLS
>         sent_(0) {}
> 
>   SSLStreamAdapterTestDTLS(const std::string& cert_pem,
>-                           const std::string& private_key_pem) :
>-      SSLStreamAdapterTestBase(cert_pem, private_key_pem, true),
>-      client_buffer_(kBufferCapacity, kDefaultBufferSize),
>-      server_buffer_(kBufferCapacity, kDefaultBufferSize),
>-      packet_size_(1000), count_(0), sent_(0) {
>-  }
>+                           const std::string& private_key_pem)
>+      : SSLStreamAdapterTestBase(cert_pem, private_key_pem, true),
>+        client_buffer_(kBufferCapacity, kDefaultBufferSize),
>+        server_buffer_(kBufferCapacity, kDefaultBufferSize),
>+        packet_size_(1000),
>+        count_(0),
>+        sent_(0) {}
> 
>   void CreateStreams() override {
>     client_stream_ =
>@@ -814,7 +798,7 @@ class SSLStreamAdapterTestDTLS
>   }
> 
>   void WriteData() override {
>-    unsigned char *packet = new unsigned char[1600];
>+    unsigned char* packet = new unsigned char[1600];
> 
>     while (sent_ < count_) {
>       unsigned int rand_state = sent_;
>@@ -839,10 +823,10 @@ class SSLStreamAdapterTestDTLS
>       }
>     }
> 
>-    delete [] packet;
>+    delete[] packet;
>   }
> 
>-  void ReadData(rtc::StreamInterface *stream) override {
>+  void ReadData(rtc::StreamInterface* stream) override {
>     unsigned char buffer[2000];
>     size_t bread;
>     int err2;
>@@ -890,7 +874,7 @@ class SSLStreamAdapterTestDTLS
>       WAIT(false, 2000);
>       EXPECT_EQ(0U, received_.size());
>     } else if (loss_ == 0) {
>-        EXPECT_EQ_WAIT(static_cast<size_t>(sent_), received_.size(), 1000);
>+      EXPECT_EQ_WAIT(static_cast<size_t>(sent_), received_.size(), 1000);
>     } else {
>       RTC_LOG(LS_INFO) << "Sent " << sent_ << " packets; received "
>                        << received_.size();
>@@ -906,9 +890,10 @@ class SSLStreamAdapterTestDTLS
>   std::set<int> received_;
> };
> 
>-
>-rtc::StreamResult SSLDummyStreamBase::Write(const void* data, size_t data_len,
>-                                              size_t* written, int* error) {
>+rtc::StreamResult SSLDummyStreamBase::Write(const void* data,
>+                                            size_t data_len,
>+                                            size_t* written,
>+                                            int* error) {
>   RTC_LOG(LS_VERBOSE) << "Writing to loopback " << data_len;
> 
>   if (first_packet_) {
>@@ -925,9 +910,8 @@ rtc::StreamResult SSLDummyStreamBase::Write(const void* data, size_t data_len,
> 
> class SSLStreamAdapterTestDTLSFromPEMStrings : public SSLStreamAdapterTestDTLS {
>  public:
>-  SSLStreamAdapterTestDTLSFromPEMStrings() :
>-      SSLStreamAdapterTestDTLS(kCERT_PEM, kRSA_PRIVATE_KEY_PEM) {
>-  }
>+  SSLStreamAdapterTestDTLSFromPEMStrings()
>+      : SSLStreamAdapterTestDTLS(kCERT_PEM, kRSA_PRIVATE_KEY_PEM) {}
> };
> 
> // Test fixture for certificate chaining. Server will push more than one
>@@ -971,11 +955,10 @@ TEST_P(SSLStreamAdapterTestTLS, GetPeerCertChainWithOneCertificate) {
>   TestHandshake();
>   std::unique_ptr<rtc::SSLCertChain> cert_chain =
>       client_ssl_->GetPeerSSLCertChain();
>-  std::unique_ptr<rtc::SSLCertificate> certificate =
>-      client_ssl_->GetPeerCertificate();
>   ASSERT_NE(nullptr, cert_chain);
>   EXPECT_EQ(1u, cert_chain->GetSize());
>-  EXPECT_EQ(cert_chain->Get(0).ToPEMString(), certificate->ToPEMString());
>+  EXPECT_EQ(cert_chain->Get(0).ToPEMString(),
>+            server_identity_->certificate().ToPEMString());
> }
> 
> TEST_F(SSLStreamAdapterTestDTLSCertChain, TwoCertHandshake) {
>@@ -1303,28 +1286,28 @@ TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpKeyAndSaltLengths) {
>   int key_len;
>   int salt_len;
> 
>-  ASSERT_FALSE(rtc::GetSrtpKeyAndSaltLengths(
>-      rtc::SRTP_INVALID_CRYPTO_SUITE, &key_len, &salt_len));
>+  ASSERT_FALSE(rtc::GetSrtpKeyAndSaltLengths(rtc::SRTP_INVALID_CRYPTO_SUITE,
>+                                             &key_len, &salt_len));
> 
>-  ASSERT_TRUE(rtc::GetSrtpKeyAndSaltLengths(
>-      rtc::SRTP_AES128_CM_SHA1_32, &key_len, &salt_len));
>-  ASSERT_EQ(128/8, key_len);
>-  ASSERT_EQ(112/8, salt_len);
>+  ASSERT_TRUE(rtc::GetSrtpKeyAndSaltLengths(rtc::SRTP_AES128_CM_SHA1_32,
>+                                            &key_len, &salt_len));
>+  ASSERT_EQ(128 / 8, key_len);
>+  ASSERT_EQ(112 / 8, salt_len);
> 
>-  ASSERT_TRUE(rtc::GetSrtpKeyAndSaltLengths(
>-      rtc::SRTP_AES128_CM_SHA1_80, &key_len, &salt_len));
>-  ASSERT_EQ(128/8, key_len);
>-  ASSERT_EQ(112/8, salt_len);
>+  ASSERT_TRUE(rtc::GetSrtpKeyAndSaltLengths(rtc::SRTP_AES128_CM_SHA1_80,
>+                                            &key_len, &salt_len));
>+  ASSERT_EQ(128 / 8, key_len);
>+  ASSERT_EQ(112 / 8, salt_len);
> 
>-  ASSERT_TRUE(rtc::GetSrtpKeyAndSaltLengths(
>-      rtc::SRTP_AEAD_AES_128_GCM, &key_len, &salt_len));
>-  ASSERT_EQ(128/8, key_len);
>-  ASSERT_EQ(96/8, salt_len);
>+  ASSERT_TRUE(rtc::GetSrtpKeyAndSaltLengths(rtc::SRTP_AEAD_AES_128_GCM,
>+                                            &key_len, &salt_len));
>+  ASSERT_EQ(128 / 8, key_len);
>+  ASSERT_EQ(96 / 8, salt_len);
> 
>-  ASSERT_TRUE(rtc::GetSrtpKeyAndSaltLengths(
>-      rtc::SRTP_AEAD_AES_256_GCM, &key_len, &salt_len));
>-  ASSERT_EQ(256/8, key_len);
>-  ASSERT_EQ(96/8, salt_len);
>+  ASSERT_TRUE(rtc::GetSrtpKeyAndSaltLengths(rtc::SRTP_AEAD_AES_256_GCM,
>+                                            &key_len, &salt_len));
>+  ASSERT_EQ(256 / 8, key_len);
>+  ASSERT_EQ(96 / 8, salt_len);
> };
> 
> // Test an exporter
>@@ -1334,16 +1317,14 @@ TEST_P(SSLStreamAdapterTestDTLS, TestDTLSExporter) {
>   unsigned char server_out[20];
> 
>   bool result;
>-  result = ExportKeyingMaterial(kExporterLabel,
>-                                kExporterContext, kExporterContextLen,
>-                                true, true,
>-                                client_out, sizeof(client_out));
>+  result = ExportKeyingMaterial(kExporterLabel, kExporterContext,
>+                                kExporterContextLen, true, true, client_out,
>+                                sizeof(client_out));
>   ASSERT_TRUE(result);
> 
>-  result = ExportKeyingMaterial(kExporterLabel,
>-                                kExporterContext, kExporterContextLen,
>-                                true, false,
>-                                server_out, sizeof(server_out));
>+  result = ExportKeyingMaterial(kExporterLabel, kExporterContext,
>+                                kExporterContextLen, true, false, server_out,
>+                                sizeof(server_out));
>   ASSERT_TRUE(result);
> 
>   ASSERT_TRUE(!memcmp(client_out, server_out, sizeof(client_out)));
>@@ -1388,9 +1369,6 @@ TEST_F(SSLStreamAdapterTestDTLSFromPEMStrings, TestDTLSGetPeerCertificate) {
>   std::string client_peer_string = client_peer_cert->ToPEMString();
>   ASSERT_NE(kCERT_PEM, client_peer_string);
> 
>-  // It must not have a chain, because the test certs are self-signed.
>-  ASSERT_FALSE(client_peer_cert->GetChain());
>-
>   // The server should have a peer certificate after the handshake.
>   std::unique_ptr<rtc::SSLCertificate> server_peer_cert =
>       GetPeerCertificate(false);
>@@ -1398,9 +1376,6 @@ TEST_F(SSLStreamAdapterTestDTLSFromPEMStrings, TestDTLSGetPeerCertificate) {
> 
>   // It's kCERT_PEM
>   ASSERT_EQ(kCERT_PEM, server_peer_cert->ToPEMString());
>-
>-  // It must not have a chain, because the test certs are self-signed.
>-  ASSERT_FALSE(server_peer_cert->GetChain());
> }
> 
> // Test getting the used DTLS ciphers.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stream.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stream.cc
>index b75490bf80b..3f4d3dc93b4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stream.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stream.cc
>@@ -11,14 +11,13 @@
> #if defined(WEBRTC_POSIX)
> #include <sys/file.h>
> #endif  // WEBRTC_POSIX
>-#include <sys/types.h>
>-#include <sys/stat.h>
> #include <errno.h>
>+#include <sys/stat.h>
>+#include <sys/types.h>
> 
> #include <algorithm>
> #include <string>
> 
>-#include "rtc_base/basictypes.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/messagequeue.h"
>@@ -38,11 +37,12 @@ namespace rtc {
> ///////////////////////////////////////////////////////////////////////////////
> // StreamInterface
> ///////////////////////////////////////////////////////////////////////////////
>-StreamInterface::~StreamInterface() {
>-}
>+StreamInterface::~StreamInterface() {}
> 
>-StreamResult StreamInterface::WriteAll(const void* data, size_t data_len,
>-                                       size_t* written, int* error) {
>+StreamResult StreamInterface::WriteAll(const void* data,
>+                                       size_t data_len,
>+                                       size_t* written,
>+                                       int* error) {
>   StreamResult result = SR_SUCCESS;
>   size_t total_written = 0, current_written;
>   while (total_written < data_len) {
>@@ -57,8 +57,10 @@ StreamResult StreamInterface::WriteAll(const void* data, size_t data_len,
>   return result;
> }
> 
>-StreamResult StreamInterface::ReadAll(void* buffer, size_t buffer_len,
>-                                      size_t* read, int* error) {
>+StreamResult StreamInterface::ReadAll(void* buffer,
>+                                      size_t buffer_len,
>+                                      size_t* read,
>+                                      int* error) {
>   StreamResult result = SR_SUCCESS;
>   size_t total_read = 0, current_read;
>   while (total_read < buffer_len) {
>@@ -138,8 +140,7 @@ bool StreamInterface::ReserveSize(size_t size) {
>   return true;
> }
> 
>-StreamInterface::StreamInterface() {
>-}
>+StreamInterface::StreamInterface() {}
> 
> void StreamInterface::OnMessage(Message* msg) {
>   if (MSG_POST_EVENT == msg->message_id) {
>@@ -237,90 +238,6 @@ void StreamAdapterInterface::OnEvent(StreamInterface* stream,
>   SignalEvent(this, events, err);
> }
> 
>-///////////////////////////////////////////////////////////////////////////////
>-// StreamTap
>-///////////////////////////////////////////////////////////////////////////////
>-
>-StreamTap::StreamTap(StreamInterface* stream, StreamInterface* tap)
>-    : StreamAdapterInterface(stream), tap_(), tap_result_(SR_SUCCESS),
>-        tap_error_(0) {
>-  AttachTap(tap);
>-}
>-
>-StreamTap::~StreamTap() = default;
>-
>-void StreamTap::AttachTap(StreamInterface* tap) {
>-  tap_.reset(tap);
>-}
>-
>-StreamInterface* StreamTap::DetachTap() {
>-  return tap_.release();
>-}
>-
>-StreamResult StreamTap::GetTapResult(int* error) {
>-  if (error) {
>-    *error = tap_error_;
>-  }
>-  return tap_result_;
>-}
>-
>-StreamResult StreamTap::Read(void* buffer, size_t buffer_len,
>-                             size_t* read, int* error) {
>-  size_t backup_read;
>-  if (!read) {
>-    read = &backup_read;
>-  }
>-  StreamResult res = StreamAdapterInterface::Read(buffer, buffer_len,
>-                                                  read, error);
>-  if ((res == SR_SUCCESS) && (tap_result_ == SR_SUCCESS)) {
>-    tap_result_ = tap_->WriteAll(buffer, *read, nullptr, &tap_error_);
>-  }
>-  return res;
>-}
>-
>-StreamResult StreamTap::Write(const void* data, size_t data_len,
>-                              size_t* written, int* error) {
>-  size_t backup_written;
>-  if (!written) {
>-    written = &backup_written;
>-  }
>-  StreamResult res = StreamAdapterInterface::Write(data, data_len,
>-                                                   written, error);
>-  if ((res == SR_SUCCESS) && (tap_result_ == SR_SUCCESS)) {
>-    tap_result_ = tap_->WriteAll(data, *written, nullptr, &tap_error_);
>-  }
>-  return res;
>-}
>-
>-///////////////////////////////////////////////////////////////////////////////
>-// NullStream
>-///////////////////////////////////////////////////////////////////////////////
>-
>-NullStream::NullStream() {
>-}
>-
>-NullStream::~NullStream() {
>-}
>-
>-StreamState NullStream::GetState() const {
>-  return SS_OPEN;
>-}
>-
>-StreamResult NullStream::Read(void* buffer, size_t buffer_len,
>-                              size_t* read, int* error) {
>-  if (error) *error = -1;
>-  return SR_ERROR;
>-}
>-
>-StreamResult NullStream::Write(const void* data, size_t data_len,
>-                               size_t* written, int* error) {
>-  if (written) *written = data_len;
>-  return SR_SUCCESS;
>-}
>-
>-void NullStream::Close() {
>-}
>-
> ///////////////////////////////////////////////////////////////////////////////
> // FileStream
> ///////////////////////////////////////////////////////////////////////////////
>@@ -331,7 +248,8 @@ FileStream::~FileStream() {
>   FileStream::Close();
> }
> 
>-bool FileStream::Open(const std::string& filename, const char* mode,
>+bool FileStream::Open(const std::string& filename,
>+                      const char* mode,
>                       int* error) {
>   Close();
> #if defined(WEBRTC_WIN)
>@@ -353,8 +271,10 @@ bool FileStream::Open(const std::string& filename, const char* mode,
>   return (file_ != nullptr);
> }
> 
>-bool FileStream::OpenShare(const std::string& filename, const char* mode,
>-                           int shflag, int* error) {
>+bool FileStream::OpenShare(const std::string& filename,
>+                           const char* mode,
>+                           int shflag,
>+                           int* error) {
>   Close();
> #if defined(WEBRTC_WIN)
>   std::wstring wfilename;
>@@ -386,8 +306,10 @@ StreamState FileStream::GetState() const {
>   return (file_ == nullptr) ? SS_CLOSED : SS_OPEN;
> }
> 
>-StreamResult FileStream::Read(void* buffer, size_t buffer_len,
>-                              size_t* read, int* error) {
>+StreamResult FileStream::Read(void* buffer,
>+                              size_t buffer_len,
>+                              size_t* read,
>+                              int* error) {
>   if (!file_)
>     return SR_EOS;
>   size_t result = fread(buffer, 1, buffer_len, file_);
>@@ -403,8 +325,10 @@ StreamResult FileStream::Read(void* buffer, size_t buffer_len,
>   return SR_SUCCESS;
> }
> 
>-StreamResult FileStream::Write(const void* data, size_t data_len,
>-                               size_t* written, int* error) {
>+StreamResult FileStream::Write(const void* data,
>+                               size_t data_len,
>+                               size_t* written,
>+                               int* error) {
>   if (!file_)
>     return SR_EOS;
>   size_t result = fwrite(data, 1, data_len, file_);
>@@ -481,30 +405,6 @@ bool FileStream::Flush() {
>   return false;
> }
> 
>-#if defined(WEBRTC_POSIX) && !defined(__native_client__)
>-
>-bool FileStream::TryLock() {
>-  if (file_ == nullptr) {
>-    // Stream not open.
>-    RTC_NOTREACHED();
>-    return false;
>-  }
>-
>-  return flock(fileno(file_), LOCK_EX|LOCK_NB) == 0;
>-}
>-
>-bool FileStream::Unlock() {
>-  if (file_ == nullptr) {
>-    // Stream not open.
>-    RTC_NOTREACHED();
>-    return false;
>-  }
>-
>-  return flock(fileno(file_), LOCK_UN) == 0;
>-}
>-
>-#endif
>-
> void FileStream::DoClose() {
>   fclose(file_);
> }
>@@ -520,8 +420,10 @@ StreamState MemoryStreamBase::GetState() const {
>   return SS_OPEN;
> }
> 
>-StreamResult MemoryStreamBase::Read(void* buffer, size_t bytes,
>-                                    size_t* bytes_read, int* error) {
>+StreamResult MemoryStreamBase::Read(void* buffer,
>+                                    size_t bytes,
>+                                    size_t* bytes_read,
>+                                    int* error) {
>   if (seek_position_ >= data_length_) {
>     return SR_EOS;
>   }
>@@ -538,8 +440,10 @@ StreamResult MemoryStreamBase::Read(void* buffer, size_t bytes,
>   return SR_SUCCESS;
> }
> 
>-StreamResult MemoryStreamBase::Write(const void* buffer, size_t bytes,
>-                                     size_t* bytes_written, int* error) {
>+StreamResult MemoryStreamBase::Write(const void* buffer,
>+                                     size_t bytes,
>+                                     size_t* bytes_written,
>+                                     int* error) {
>   size_t available = buffer_length_ - seek_position_;
>   if (0 == available) {
>     // Increase buffer size to the larger of:
>@@ -608,26 +512,24 @@ StreamResult MemoryStreamBase::DoReserve(size_t size, int* error) {
> 
> ///////////////////////////////////////////////////////////////////////////////
> 
>-MemoryStream::MemoryStream() : buffer_alloc_(nullptr) {}
>+MemoryStream::MemoryStream() {}
> 
>-MemoryStream::MemoryStream(const char* data) : buffer_alloc_(nullptr) {
>+MemoryStream::MemoryStream(const char* data) {
>   SetData(data, strlen(data));
> }
> 
>-MemoryStream::MemoryStream(const void* data, size_t length)
>-    : buffer_alloc_(nullptr) {
>+MemoryStream::MemoryStream(const void* data, size_t length) {
>   SetData(data, length);
> }
> 
> MemoryStream::~MemoryStream() {
>-  delete [] buffer_alloc_;
>+  delete[] buffer_;
> }
> 
> void MemoryStream::SetData(const void* data, size_t length) {
>   data_length_ = buffer_length_ = length;
>-  delete [] buffer_alloc_;
>-  buffer_alloc_ = new char[buffer_length_ + kAlignment];
>-  buffer_ = reinterpret_cast<char*>(ALIGNP(buffer_alloc_, kAlignment));
>+  delete[] buffer_;
>+  buffer_ = new char[buffer_length_];
>   memcpy(buffer_, data, data_length_);
>   seek_position_ = 0;
> }
>@@ -636,12 +538,9 @@ StreamResult MemoryStream::DoReserve(size_t size, int* error) {
>   if (buffer_length_ >= size)
>     return SR_SUCCESS;
> 
>-  if (char* new_buffer_alloc = new char[size + kAlignment]) {
>-    char* new_buffer = reinterpret_cast<char*>(
>-        ALIGNP(new_buffer_alloc, kAlignment));
>+  if (char* new_buffer = new char[size]) {
>     memcpy(new_buffer, buffer_, data_length_);
>-    delete [] buffer_alloc_;
>-    buffer_alloc_ = new_buffer_alloc;
>+    delete[] buffer_;
>     buffer_ = new_buffer;
>     buffer_length_ = size;
>     return SR_SUCCESS;
>@@ -653,42 +552,31 @@ StreamResult MemoryStream::DoReserve(size_t size, int* error) {
>   return SR_ERROR;
> }
> 
>-///////////////////////////////////////////////////////////////////////////////
>-
>-ExternalMemoryStream::ExternalMemoryStream() {
>-}
>-
>-ExternalMemoryStream::ExternalMemoryStream(void* data, size_t length) {
>-  SetData(data, length);
>-}
>-
>-ExternalMemoryStream::~ExternalMemoryStream() {
>-}
>-
>-void ExternalMemoryStream::SetData(void* data, size_t length) {
>-  data_length_ = buffer_length_ = length;
>-  buffer_ = static_cast<char*>(data);
>-  seek_position_ = 0;
>-}
>-
> ///////////////////////////////////////////////////////////////////////////////
> // FifoBuffer
> ///////////////////////////////////////////////////////////////////////////////
> 
> FifoBuffer::FifoBuffer(size_t size)
>-    : state_(SS_OPEN), buffer_(new char[size]), buffer_length_(size),
>-      data_length_(0), read_position_(0), owner_(Thread::Current()) {
>+    : state_(SS_OPEN),
>+      buffer_(new char[size]),
>+      buffer_length_(size),
>+      data_length_(0),
>+      read_position_(0),
>+      owner_(Thread::Current()) {
>   // all events are done on the owner_ thread
> }
> 
> FifoBuffer::FifoBuffer(size_t size, Thread* owner)
>-    : state_(SS_OPEN), buffer_(new char[size]), buffer_length_(size),
>-      data_length_(0), read_position_(0), owner_(owner) {
>+    : state_(SS_OPEN),
>+      buffer_(new char[size]),
>+      buffer_length_(size),
>+      data_length_(0),
>+      read_position_(0),
>+      owner_(owner) {
>   // all events are done on the owner_ thread
> }
> 
>-FifoBuffer::~FifoBuffer() {
>-}
>+FifoBuffer::~FifoBuffer() {}
> 
> bool FifoBuffer::GetBuffered(size_t* size) const {
>   CritScope cs(&crit_);
>@@ -715,14 +603,18 @@ bool FifoBuffer::SetCapacity(size_t size) {
>   return true;
> }
> 
>-StreamResult FifoBuffer::ReadOffset(void* buffer, size_t bytes,
>-                                    size_t offset, size_t* bytes_read) {
>+StreamResult FifoBuffer::ReadOffset(void* buffer,
>+                                    size_t bytes,
>+                                    size_t offset,
>+                                    size_t* bytes_read) {
>   CritScope cs(&crit_);
>   return ReadOffsetLocked(buffer, bytes, offset, bytes_read);
> }
> 
>-StreamResult FifoBuffer::WriteOffset(const void* buffer, size_t bytes,
>-                                     size_t offset, size_t* bytes_written) {
>+StreamResult FifoBuffer::WriteOffset(const void* buffer,
>+                                     size_t bytes,
>+                                     size_t offset,
>+                                     size_t* bytes_written) {
>   CritScope cs(&crit_);
>   return WriteOffsetLocked(buffer, bytes, offset, bytes_written);
> }
>@@ -732,8 +624,10 @@ StreamState FifoBuffer::GetState() const {
>   return state_;
> }
> 
>-StreamResult FifoBuffer::Read(void* buffer, size_t bytes,
>-                              size_t* bytes_read, int* error) {
>+StreamResult FifoBuffer::Read(void* buffer,
>+                              size_t bytes,
>+                              size_t* bytes_read,
>+                              int* error) {
>   CritScope cs(&crit_);
>   const bool was_writable = data_length_ < buffer_length_;
>   size_t copy = 0;
>@@ -756,8 +650,10 @@ StreamResult FifoBuffer::Read(void* buffer, size_t bytes,
>   return result;
> }
> 
>-StreamResult FifoBuffer::Write(const void* buffer, size_t bytes,
>-                               size_t* bytes_written, int* error) {
>+StreamResult FifoBuffer::Write(const void* buffer,
>+                               size_t bytes,
>+                               size_t* bytes_written,
>+                               int* error) {
>   CritScope cs(&crit_);
> 
>   const bool was_readable = (data_length_ > 0);
>@@ -786,8 +682,9 @@ void FifoBuffer::Close() {
> 
> const void* FifoBuffer::GetReadData(size_t* size) {
>   CritScope cs(&crit_);
>-  *size = (read_position_ + data_length_ <= buffer_length_) ?
>-      data_length_ : buffer_length_ - read_position_;
>+  *size = (read_position_ + data_length_ <= buffer_length_)
>+              ? data_length_
>+              : buffer_length_ - read_position_;
>   return &buffer_[read_position_];
> }
> 
>@@ -814,10 +711,11 @@ void* FifoBuffer::GetWriteBuffer(size_t* size) {
>     read_position_ = 0;
>   }
> 
>-  const size_t write_position = (read_position_ + data_length_)
>-      % buffer_length_;
>-  *size = (write_position > read_position_ || data_length_ == 0) ?
>-      buffer_length_ - write_position : read_position_ - write_position;
>+  const size_t write_position =
>+      (read_position_ + data_length_) % buffer_length_;
>+  *size = (write_position > read_position_ || data_length_ == 0)
>+              ? buffer_length_ - write_position
>+              : read_position_ - write_position;
>   return &buffer_[write_position];
> }
> 
>@@ -872,8 +770,8 @@ StreamResult FifoBuffer::WriteOffsetLocked(const void* buffer,
>   }
> 
>   const size_t available = buffer_length_ - data_length_ - offset;
>-  const size_t write_position = (read_position_ + data_length_ + offset)
>-      % buffer_length_;
>+  const size_t write_position =
>+      (read_position_ + data_length_ + offset) % buffer_length_;
>   const size_t copy = std::min(bytes, available);
>   const size_t tail_copy = std::min(copy, buffer_length_ - write_position);
>   const char* const p = static_cast<const char*>(buffer);
>@@ -887,109 +785,6 @@ StreamResult FifoBuffer::WriteOffsetLocked(const void* buffer,
> }
> 
> 
>-///////////////////////////////////////////////////////////////////////////////
>-// StringStream - Reads/Writes to an external std::string
>-///////////////////////////////////////////////////////////////////////////////
>-
>-StringStream::StringStream(std::string* str)
>-    : str_(*str), read_pos_(0), read_only_(false) {
>-}
>-
>-StringStream::StringStream(const std::string& str)
>-    : str_(const_cast<std::string&>(str)), read_pos_(0), read_only_(true) {
>-}
>-
>-StreamState StringStream::GetState() const {
>-  return SS_OPEN;
>-}
>-
>-StreamResult StringStream::Read(void* buffer, size_t buffer_len,
>-                                      size_t* read, int* error) {
>-  size_t available = std::min(buffer_len, str_.size() - read_pos_);
>-  if (!available)
>-    return SR_EOS;
>-  memcpy(buffer, str_.data() + read_pos_, available);
>-  read_pos_ += available;
>-  if (read)
>-    *read = available;
>-  return SR_SUCCESS;
>-}
>-
>-StreamResult StringStream::Write(const void* data, size_t data_len,
>-                                      size_t* written, int* error) {
>-  if (read_only_) {
>-    if (error) {
>-      *error = -1;
>-    }
>-    return SR_ERROR;
>-  }
>-  str_.append(static_cast<const char*>(data),
>-              static_cast<const char*>(data) + data_len);
>-  if (written)
>-    *written = data_len;
>-  return SR_SUCCESS;
>-}
>-
>-void StringStream::Close() {
>-}
>-
>-bool StringStream::SetPosition(size_t position) {
>-  if (position > str_.size())
>-    return false;
>-  read_pos_ = position;
>-  return true;
>-}
>-
>-bool StringStream::GetPosition(size_t* position) const {
>-  if (position)
>-    *position = read_pos_;
>-  return true;
>-}
>-
>-bool StringStream::GetSize(size_t* size) const {
>-  if (size)
>-    *size = str_.size();
>-  return true;
>-}
>-
>-bool StringStream::GetAvailable(size_t* size) const {
>-  if (size)
>-    *size = str_.size() - read_pos_;
>-  return true;
>-}
>-
>-bool StringStream::ReserveSize(size_t size) {
>-  if (read_only_)
>-    return false;
>-  str_.reserve(size);
>-  return true;
>-}
>-
>-///////////////////////////////////////////////////////////////////////////////
>-// StreamReference
>-///////////////////////////////////////////////////////////////////////////////
>-
>-StreamReference::StreamReference(StreamInterface* stream)
>-    : StreamAdapterInterface(stream, false) {
>-  // owner set to false so the destructor does not free the stream.
>-  stream_ref_count_ = new StreamRefCount(stream);
>-}
>-
>-StreamInterface* StreamReference::NewReference() {
>-  stream_ref_count_->AddReference();
>-  return new StreamReference(stream_ref_count_, stream());
>-}
>-
>-StreamReference::~StreamReference() {
>-  stream_ref_count_->Release();
>-}
>-
>-StreamReference::StreamReference(StreamRefCount* stream_ref_count,
>-                                 StreamInterface* stream)
>-    : StreamAdapterInterface(stream, false),
>-      stream_ref_count_(stream_ref_count) {
>-}
>-
> ///////////////////////////////////////////////////////////////////////////////
> 
> StreamResult Flow(StreamInterface* source,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stream.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stream.h
>index dd250079fb8..ac8193b3d57 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stream.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stream.h
>@@ -21,7 +21,7 @@
> #include "rtc_base/logging.h"
> #include "rtc_base/messagehandler.h"
> #include "rtc_base/messagequeue.h"
>-#include "rtc_base/sigslot.h"
>+#include "rtc_base/third_party/sigslot/sigslot.h"
> 
> namespace rtc {
> 
>@@ -55,14 +55,12 @@ class Thread;
> 
> struct StreamEventData : public MessageData {
>   int events, error;
>-  StreamEventData(int ev, int er) : events(ev), error(er) { }
>+  StreamEventData(int ev, int er) : events(ev), error(er) {}
> };
> 
> class StreamInterface : public MessageHandler {
>  public:
>-  enum {
>-    MSG_POST_EVENT = 0xF1F1, MSG_MAX = MSG_POST_EVENT
>-  };
>+  enum { MSG_POST_EVENT = 0xF1F1, MSG_MAX = MSG_POST_EVENT };
> 
>   ~StreamInterface() override;
> 
>@@ -81,10 +79,14 @@ class StreamInterface : public MessageHandler {
>   //    block, or the stream is in SS_OPENING state.
>   //  SR_EOS: the end-of-stream has been reached, or the stream is in the
>   //    SS_CLOSED state.
>-  virtual StreamResult Read(void* buffer, size_t buffer_len,
>-                            size_t* read, int* error) = 0;
>-  virtual StreamResult Write(const void* data, size_t data_len,
>-                             size_t* written, int* error) = 0;
>+  virtual StreamResult Read(void* buffer,
>+                            size_t buffer_len,
>+                            size_t* read,
>+                            int* error) = 0;
>+  virtual StreamResult Write(const void* data,
>+                             size_t data_len,
>+                             size_t* written,
>+                             int* error) = 0;
>   // Attempt to transition to the SS_CLOSED state.  SE_CLOSE will not be
>   // signalled as a result of this call.
>   virtual void Close() = 0;
>@@ -132,7 +134,7 @@ class StreamInterface : public MessageHandler {
>   // processed.  Read and ConsumeReadData invalidate the buffer returned by
>   // GetReadData.
>   virtual const void* GetReadData(size_t* data_len);
>-  virtual void ConsumeReadData(size_t) {}
>+  virtual void ConsumeReadData(size_t used) {}
> 
>   // GetWriteBuffer returns a pointer to a buffer which is owned by the stream.
>   // The buffer has a capacity of buf_len bytes.  null is returned if there is
>@@ -146,7 +148,7 @@ class StreamInterface : public MessageHandler {
>   // when it is available.  If the requested amount is too large, return an
>   // error.
>   virtual void* GetWriteBuffer(size_t* buf_len);
>-  virtual void ConsumeWriteBuffer(size_t) {}
>+  virtual void ConsumeWriteBuffer(size_t used) {}
> 
>   // Write data_len bytes found in data, circumventing any throttling which
>   // would could cause SR_BLOCK to be returned.  Returns true if all the data
>@@ -158,7 +160,7 @@ class StreamInterface : public MessageHandler {
>   // NOTE: This interface is being considered experimentally at the moment.  It
>   // would be used by JUDP and BandwidthStream as a way to circumvent certain
>   // soft limits in writing.
>-  //virtual bool ForceWrite(const void* data, size_t data_len, int* error) {
>+  // virtual bool ForceWrite(const void* data, size_t data_len, int* error) {
>   //  if (error) *error = -1;
>   //  return false;
>   //}
>@@ -208,13 +210,17 @@ class StreamInterface : public MessageHandler {
>   // unlike Write, the argument 'written' is always set, and may be non-zero
>   // on results other than SR_SUCCESS.  The remaining arguments have the
>   // same semantics as Write.
>-  StreamResult WriteAll(const void* data, size_t data_len,
>-                        size_t* written, int* error);
>+  StreamResult WriteAll(const void* data,
>+                        size_t data_len,
>+                        size_t* written,
>+                        int* error);
> 
>   // Similar to ReadAll.  Calls Read until buffer_len bytes have been read, or
>   // until a non-SR_SUCCESS result is returned.  'read' is always set.
>-  StreamResult ReadAll(void* buffer, size_t buffer_len,
>-                       size_t* read, int* error);
>+  StreamResult ReadAll(void* buffer,
>+                       size_t buffer_len,
>+                       size_t* read,
>+                       int* error);
> 
>   // ReadLine is a helper function which repeatedly calls Read until it hits
>   // the end-of-line character, or something other than SR_SUCCESS.
>@@ -309,60 +315,6 @@ class StreamAdapterInterface : public StreamInterface,
>   RTC_DISALLOW_COPY_AND_ASSIGN(StreamAdapterInterface);
> };
> 
>-///////////////////////////////////////////////////////////////////////////////
>-// StreamTap is a non-modifying, pass-through adapter, which copies all data
>-// in either direction to the tap.  Note that errors or blocking on writing to
>-// the tap will prevent further tap writes from occurring.
>-///////////////////////////////////////////////////////////////////////////////
>-
>-class StreamTap : public StreamAdapterInterface {
>- public:
>-  explicit StreamTap(StreamInterface* stream, StreamInterface* tap);
>-  ~StreamTap() override;
>-
>-  void AttachTap(StreamInterface* tap);
>-  StreamInterface* DetachTap();
>-  StreamResult GetTapResult(int* error);
>-
>-  // StreamAdapterInterface Interface
>-  StreamResult Read(void* buffer,
>-                    size_t buffer_len,
>-                    size_t* read,
>-                    int* error) override;
>-  StreamResult Write(const void* data,
>-                     size_t data_len,
>-                     size_t* written,
>-                     int* error) override;
>-
>- private:
>-  std::unique_ptr<StreamInterface> tap_;
>-  StreamResult tap_result_;
>-  int tap_error_;
>-  RTC_DISALLOW_COPY_AND_ASSIGN(StreamTap);
>-};
>-
>-///////////////////////////////////////////////////////////////////////////////
>-// NullStream gives errors on read, and silently discards all written data.
>-///////////////////////////////////////////////////////////////////////////////
>-
>-class NullStream : public StreamInterface {
>- public:
>-  NullStream();
>-  ~NullStream() override;
>-
>-  // StreamInterface Interface
>-  StreamState GetState() const override;
>-  StreamResult Read(void* buffer,
>-                    size_t buffer_len,
>-                    size_t* read,
>-                    int* error) override;
>-  StreamResult Write(const void* data,
>-                     size_t data_len,
>-                     size_t* written,
>-                     int* error) override;
>-  void Close() override;
>-};
>-
> ///////////////////////////////////////////////////////////////////////////////
> // FileStream is a simple implementation of a StreamInterface, which does not
> // support asynchronous notification.
>@@ -375,8 +327,10 @@ class FileStream : public StreamInterface {
> 
>   // The semantics of filename and mode are the same as stdio's fopen
>   virtual bool Open(const std::string& filename, const char* mode, int* error);
>-  virtual bool OpenShare(const std::string& filename, const char* mode,
>-                         int shflag, int* error);
>+  virtual bool OpenShare(const std::string& filename,
>+                         const char* mode,
>+                         int shflag,
>+                         int* error);
> 
>   // By default, reads and writes are buffered for efficiency.  Disabling
>   // buffering causes writes to block until the bytes on disk are updated.
>@@ -400,13 +354,6 @@ class FileStream : public StreamInterface {
> 
>   bool Flush() override;
> 
>-#if defined(WEBRTC_POSIX) && !defined(__native_client__)
>-  // Tries to aquire an exclusive lock on the file.
>-  // Use OpenShare(...) on win32 to get similar functionality.
>-  bool TryLock();
>-  bool Unlock();
>-#endif
>-
>  protected:
>   virtual void DoClose();
> 
>@@ -472,21 +419,6 @@ class MemoryStream : public MemoryStreamBase {
> 
>  protected:
>   StreamResult DoReserve(size_t size, int* error) override;
>-  // Memory Streams are aligned for efficiency.
>-  static const int kAlignment = 16;
>-  char* buffer_alloc_;
>-};
>-
>-// ExternalMemoryStream adapts an external memory buffer, so writes which would
>-// extend past the end of the buffer will return end-of-stream.
>-
>-class ExternalMemoryStream : public MemoryStreamBase {
>- public:
>-  ExternalMemoryStream();
>-  ExternalMemoryStream(void* data, size_t length);
>-  ~ExternalMemoryStream() override;
>-
>-  void SetData(void* data, size_t length);
> };
> 
> // FifoBuffer allows for efficient, thread-safe buffering of data between
>@@ -509,14 +441,18 @@ class FifoBuffer : public StreamInterface {
>   // is specified in number of bytes.
>   // This method doesn't adjust read position nor the number of available
>   // bytes, user has to call ConsumeReadData() to do this.
>-  StreamResult ReadOffset(void* buffer, size_t bytes, size_t offset,
>+  StreamResult ReadOffset(void* buffer,
>+                          size_t bytes,
>+                          size_t offset,
>                           size_t* bytes_read);
> 
>   // Write |buffer| with an offset from the current write position, offset is
>   // specified in number of bytes.
>   // This method doesn't adjust the number of buffered bytes, user has to call
>   // ConsumeWriteBuffer() to do this.
>-  StreamResult WriteOffset(const void* buffer, size_t bytes, size_t offset,
>+  StreamResult WriteOffset(const void* buffer,
>+                           size_t bytes,
>+                           size_t offset,
>                            size_t* bytes_written);
> 
>   // StreamInterface methods
>@@ -570,97 +506,6 @@ class FifoBuffer : public StreamInterface {
>   RTC_DISALLOW_COPY_AND_ASSIGN(FifoBuffer);
> };
> 
>-///////////////////////////////////////////////////////////////////////////////
>-// StringStream - Reads/Writes to an external std::string
>-///////////////////////////////////////////////////////////////////////////////
>-
>-class StringStream : public StreamInterface {
>- public:
>-  explicit StringStream(std::string* str);
>-  explicit StringStream(const std::string& str);
>-
>-  StreamState GetState() const override;
>-  StreamResult Read(void* buffer,
>-                    size_t buffer_len,
>-                    size_t* read,
>-                    int* error) override;
>-  StreamResult Write(const void* data,
>-                     size_t data_len,
>-                     size_t* written,
>-                     int* error) override;
>-  void Close() override;
>-  bool SetPosition(size_t position) override;
>-  bool GetPosition(size_t* position) const override;
>-  bool GetSize(size_t* size) const override;
>-  bool GetAvailable(size_t* size) const override;
>-  bool ReserveSize(size_t size) override;
>-
>- private:
>-  std::string& str_;
>-  size_t read_pos_;
>-  bool read_only_;
>-};
>-
>-///////////////////////////////////////////////////////////////////////////////
>-// StreamReference - A reference counting stream adapter
>-///////////////////////////////////////////////////////////////////////////////
>-
>-// Keep in mind that the streams and adapters defined in this file are
>-// not thread-safe, so this has limited uses.
>-
>-// A StreamRefCount holds the reference count and a pointer to the
>-// wrapped stream. It deletes the wrapped stream when there are no
>-// more references. We can then have multiple StreamReference
>-// instances pointing to one StreamRefCount, all wrapping the same
>-// stream.
>-
>-class StreamReference : public StreamAdapterInterface {
>-  class StreamRefCount;
>- public:
>-  // Constructor for the first reference to a stream
>-  // Note: get more references through NewReference(). Use this
>-  // constructor only once on a given stream.
>-  explicit StreamReference(StreamInterface* stream);
>-  StreamInterface* GetStream() { return stream(); }
>-  StreamInterface* NewReference();
>-  ~StreamReference() override;
>-
>- private:
>-  class StreamRefCount {
>-   public:
>-    explicit StreamRefCount(StreamInterface* stream)
>-        : stream_(stream), ref_count_(1) {
>-    }
>-    void AddReference() {
>-      CritScope lock(&cs_);
>-      ++ref_count_;
>-    }
>-    void Release() {
>-      int ref_count;
>-      {  // Atomic ops would have been a better fit here.
>-        CritScope lock(&cs_);
>-        ref_count = --ref_count_;
>-      }
>-      if (ref_count == 0) {
>-        delete stream_;
>-        delete this;
>-      }
>-    }
>-   private:
>-    StreamInterface* stream_;
>-    int ref_count_;
>-    CriticalSection cs_;
>-    RTC_DISALLOW_COPY_AND_ASSIGN(StreamRefCount);
>-  };
>-
>-  // Constructor for adding references
>-  explicit StreamReference(StreamRefCount* stream_ref_count,
>-                           StreamInterface* stream);
>-
>-  StreamRefCount* stream_ref_count_;
>-  RTC_DISALLOW_COPY_AND_ASSIGN(StreamReference);
>-};
>-
> ///////////////////////////////////////////////////////////////////////////////
> 
> // Flow attempts to move bytes from source to sink via buffer of size
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stream_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stream_unittest.cc
>index 8c305c5a197..11e0be0422b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stream_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stream_unittest.cc
>@@ -19,7 +19,7 @@ namespace rtc {
> 
> class TestStream : public StreamInterface {
>  public:
>-  TestStream() : pos_(0) { }
>+  TestStream() : pos_(0) {}
> 
>   StreamState GetState() const override { return SS_OPEN; }
> 
>@@ -53,7 +53,8 @@ class TestStream : public StreamInterface {
>   }
> 
>   bool GetPosition(size_t* position) const override {
>-    if (position) *position = pos_;
>+    if (position)
>+      *position = pos_;
>     return true;
>   }
> 
>@@ -65,8 +66,7 @@ class TestStream : public StreamInterface {
>   size_t pos_;
> };
> 
>-bool VerifyTestBuffer(unsigned char* buffer, size_t len,
>-                      unsigned char value) {
>+bool VerifyTestBuffer(unsigned char* buffer, size_t len, unsigned char value) {
>   bool passed = true;
>   for (size_t i = 0; i < len; ++i) {
>     if (buffer[i] != value++) {
>@@ -81,7 +81,7 @@ bool VerifyTestBuffer(unsigned char* buffer, size_t len,
> 
> void SeekTest(StreamInterface* stream, const unsigned char value) {
>   size_t bytes;
>-  unsigned char buffer[13] = { 0 };
>+  unsigned char buffer[13] = {0};
>   const size_t kBufSize = sizeof(buffer);
> 
>   EXPECT_EQ(stream->Read(buffer, kBufSize, &bytes, nullptr), SR_SUCCESS);
>@@ -176,15 +176,15 @@ TEST(FifoBufferTest, TestAll) {
>   EXPECT_EQ(SR_SUCCESS, stream->Write(in, kSize / 2, &bytes, nullptr));
>   EXPECT_EQ(kSize / 2, bytes);
>   EXPECT_EQ(SR_SUCCESS, stream->Read(out, kSize / 4, &bytes, nullptr));
>-  EXPECT_EQ(kSize / 4 , bytes);
>+  EXPECT_EQ(kSize / 4, bytes);
>   EXPECT_EQ(0, memcmp(in + kSize / 2, out, kSize / 4));
>   EXPECT_EQ(SR_SUCCESS, stream->Write(in, kSize / 2, &bytes, nullptr));
>   EXPECT_EQ(kSize / 2, bytes);
>   EXPECT_EQ(SR_SUCCESS, stream->Read(out, kSize / 2, &bytes, nullptr));
>-  EXPECT_EQ(kSize / 2 , bytes);
>+  EXPECT_EQ(kSize / 2, bytes);
>   EXPECT_EQ(0, memcmp(in, out, kSize / 2));
>   EXPECT_EQ(SR_SUCCESS, stream->Read(out, kSize / 2, &bytes, nullptr));
>-  EXPECT_EQ(kSize / 2 , bytes);
>+  EXPECT_EQ(kSize / 2, bytes);
>   EXPECT_EQ(0, memcmp(in, out, kSize / 2));
> 
>   // Use GetWriteBuffer to reset the read_position for the next tests
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/string_to_number.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/string_to_number.cc
>index ad49d646d63..9201242d5f5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/string_to_number.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/string_to_number.cc
>@@ -8,15 +8,17 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>+#include "rtc_base/string_to_number.h"
>+
> #include <cerrno>
> #include <cstdlib>
> 
>-#include "rtc_base/string_to_number.h"
>+#include "rtc_base/checks.h"
> 
> namespace rtc {
> namespace string_to_number_internal {
> 
>-rtc::Optional<signed_type> ParseSigned(const char* str, int base) {
>+absl::optional<signed_type> ParseSigned(const char* str, int base) {
>   RTC_DCHECK(str);
>   if (isdigit(str[0]) || str[0] == '-') {
>     char* end = nullptr;
>@@ -26,10 +28,10 @@ rtc::Optional<signed_type> ParseSigned(const char* str, int base) {
>       return value;
>     }
>   }
>-  return rtc::nullopt;
>+  return absl::nullopt;
> }
> 
>-rtc::Optional<unsigned_type> ParseUnsigned(const char* str, int base) {
>+absl::optional<unsigned_type> ParseUnsigned(const char* str, int base) {
>   RTC_DCHECK(str);
>   if (isdigit(str[0]) || str[0] == '-') {
>     // Explicitly discard negative values. std::strtoull parsing causes unsigned
>@@ -43,8 +45,44 @@ rtc::Optional<unsigned_type> ParseUnsigned(const char* str, int base) {
>       return value;
>     }
>   }
>-  return rtc::nullopt;
>+  return absl::nullopt;
>+}
>+
>+template <typename T>
>+T StrToT(const char* str, char** str_end);
>+
>+template <>
>+inline float StrToT(const char* str, char** str_end) {
>+  return std::strtof(str, str_end);
>+}
>+
>+template <>
>+inline double StrToT(const char* str, char** str_end) {
>+  return std::strtod(str, str_end);
> }
> 
>+template <>
>+inline long double StrToT(const char* str, char** str_end) {
>+  return std::strtold(str, str_end);
>+}
>+
>+template <typename T>
>+absl::optional<T> ParseFloatingPoint(const char* str) {
>+  RTC_DCHECK(str);
>+  if (*str == '\0')
>+    return absl::nullopt;
>+  char* end = nullptr;
>+  errno = 0;
>+  const T value = StrToT<T>(str, &end);
>+  if (end && *end == '\0' && errno == 0) {
>+    return value;
>+  }
>+  return absl::nullopt;
>+}
>+
>+template absl::optional<float> ParseFloatingPoint(const char* str);
>+template absl::optional<double> ParseFloatingPoint(const char* str);
>+template absl::optional<long double> ParseFloatingPoint(const char* str);
>+
> }  // namespace string_to_number_internal
> }  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/string_to_number.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/string_to_number.h
>index 4a8fb8f8e10..7ea9f251ed4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/string_to_number.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/string_to_number.h
>@@ -11,10 +11,10 @@
> #ifndef RTC_BASE_STRING_TO_NUMBER_H_
> #define RTC_BASE_STRING_TO_NUMBER_H_
> 
>-#include <string>
> #include <limits>
>+#include <string>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> 
> namespace rtc {
> 
>@@ -25,8 +25,8 @@ namespace rtc {
> // are disabled in WebRTC.
> //
> // Integers are parsed using one of the following functions:
>-//   rtc::Optional<int-type> StringToNumber(const char* str, int base = 10);
>-//   rtc::Optional<int-type> StringToNumber(const std::string& str,
>+//   absl::optional<int-type> StringToNumber(const char* str, int base = 10);
>+//   absl::optional<int-type> StringToNumber(const std::string& str,
> //                                          int base = 10);
> //
> // These functions parse a value from the beginning of a string into one of the
>@@ -44,15 +44,18 @@ namespace rtc {
> namespace string_to_number_internal {
> // These must be (unsigned) long long, to match the signature of strto(u)ll.
> using unsigned_type = unsigned long long;  // NOLINT(runtime/int)
>-using signed_type = long long;  // NOLINT(runtime/int)
>+using signed_type = long long;             // NOLINT(runtime/int)
>+
>+absl::optional<signed_type> ParseSigned(const char* str, int base);
>+absl::optional<unsigned_type> ParseUnsigned(const char* str, int base);
> 
>-rtc::Optional<signed_type> ParseSigned(const char* str, int base);
>-rtc::Optional<unsigned_type> ParseUnsigned(const char* str, int base);
>+template <typename T>
>+absl::optional<T> ParseFloatingPoint(const char* str);
> }  // namespace string_to_number_internal
> 
> template <typename T>
> typename std::enable_if<std::is_integral<T>::value && std::is_signed<T>::value,
>-                        rtc::Optional<T>>::type
>+                        absl::optional<T>>::type
> StringToNumber(const char* str, int base = 10) {
>   using string_to_number_internal::signed_type;
>   static_assert(
>@@ -61,31 +64,42 @@ StringToNumber(const char* str, int base = 10) {
>           std::numeric_limits<T>::lowest() >=
>               std::numeric_limits<signed_type>::lowest(),
>       "StringToNumber only supports signed integers as large as long long int");
>-  rtc::Optional<signed_type> value =
>+  absl::optional<signed_type> value =
>       string_to_number_internal::ParseSigned(str, base);
>   if (value && *value >= std::numeric_limits<T>::lowest() &&
>       *value <= std::numeric_limits<T>::max()) {
>     return static_cast<T>(*value);
>   }
>-  return rtc::nullopt;
>+  return absl::nullopt;
> }
> 
> template <typename T>
> typename std::enable_if<std::is_integral<T>::value &&
>                             std::is_unsigned<T>::value,
>-                        rtc::Optional<T>>::type
>+                        absl::optional<T>>::type
> StringToNumber(const char* str, int base = 10) {
>   using string_to_number_internal::unsigned_type;
>   static_assert(std::numeric_limits<T>::max() <=
>                     std::numeric_limits<unsigned_type>::max(),
>                 "StringToNumber only supports unsigned integers as large as "
>                 "unsigned long long int");
>-  rtc::Optional<unsigned_type> value =
>+  absl::optional<unsigned_type> value =
>       string_to_number_internal::ParseUnsigned(str, base);
>   if (value && *value <= std::numeric_limits<T>::max()) {
>     return static_cast<T>(*value);
>   }
>-  return rtc::nullopt;
>+  return absl::nullopt;
>+}
>+
>+template <typename T>
>+typename std::enable_if<std::is_floating_point<T>::value,
>+                        absl::optional<T>>::type
>+StringToNumber(const char* str, int base = 10) {
>+  static_assert(
>+      std::numeric_limits<T>::max() <= std::numeric_limits<long double>::max(),
>+      "StringToNumber only supports floating-point numbers as large "
>+      "as long double");
>+  return string_to_number_internal::ParseFloatingPoint<T>(str);
> }
> 
> // The std::string overloads only exists if there is a matching const char*
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/string_to_number_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/string_to_number_unittest.cc
>index f5e5b573f1a..f7bf48421c7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/string_to_number_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/string_to_number_unittest.cc
>@@ -10,9 +10,9 @@
> 
> #include "rtc_base/string_to_number.h"
> 
>+#include <limits>
> #include <string>
> #include <type_traits>
>-#include <limits>
> 
> #include "rtc_base/gunit.h"
> 
>@@ -42,15 +42,16 @@ TYPED_TEST_P(BasicNumberTest, TestValidNumbers) {
>   using T = TypeParam;
>   constexpr T min_value = std::numeric_limits<T>::lowest();
>   constexpr T max_value = std::numeric_limits<T>::max();
>+  constexpr T zero_value = 0;
>   const std::string min_string = std::to_string(min_value);
>   const std::string max_string = std::to_string(max_value);
>   EXPECT_EQ(min_value, StringToNumber<T>(min_string));
>   EXPECT_EQ(min_value, StringToNumber<T>(min_string.c_str()));
>   EXPECT_EQ(max_value, StringToNumber<T>(max_string));
>   EXPECT_EQ(max_value, StringToNumber<T>(max_string.c_str()));
>-  EXPECT_EQ(0, StringToNumber<T>("0"));
>-  EXPECT_EQ(0, StringToNumber<T>("-0"));
>-  EXPECT_EQ(0, StringToNumber<T>(std::string("-0000000000000")));
>+  EXPECT_EQ(zero_value, StringToNumber<T>("0"));
>+  EXPECT_EQ(zero_value, StringToNumber<T>("-0"));
>+  EXPECT_EQ(zero_value, StringToNumber<T>(std::string("-0000000000000")));
> }
> 
> TYPED_TEST_P(BasicNumberTest, TestInvalidNumbers) {
>@@ -67,10 +68,10 @@ TYPED_TEST_P(BasicNumberTest, TestInvalidNumbers) {
>       (min_value == 0) ? "-2" : (std::to_string(min_value) + "1");
>   // Make the large value approximately ten times larger than the maximum.
>   const std::string too_large_string = std::to_string(max_value) + "1";
>-  EXPECT_EQ(rtc::nullopt, StringToNumber<T>(too_low_string));
>-  EXPECT_EQ(rtc::nullopt, StringToNumber<T>(too_low_string.c_str()));
>-  EXPECT_EQ(rtc::nullopt, StringToNumber<T>(too_large_string));
>-  EXPECT_EQ(rtc::nullopt, StringToNumber<T>(too_large_string.c_str()));
>+  EXPECT_EQ(absl::nullopt, StringToNumber<T>(too_low_string));
>+  EXPECT_EQ(absl::nullopt, StringToNumber<T>(too_low_string.c_str()));
>+  EXPECT_EQ(absl::nullopt, StringToNumber<T>(too_large_string));
>+  EXPECT_EQ(absl::nullopt, StringToNumber<T>(too_large_string.c_str()));
> }
> 
> TYPED_TEST_P(BasicNumberTest, TestInvalidInputs) {
>@@ -78,18 +79,18 @@ TYPED_TEST_P(BasicNumberTest, TestInvalidInputs) {
>   const char kInvalidCharArray[] = "Invalid string containing 47";
>   const char kPlusMinusCharArray[] = "+-100";
>   const char kNumberFollowedByCruft[] = "640x480";
>-  EXPECT_EQ(rtc::nullopt, StringToNumber<T>(kInvalidCharArray));
>-  EXPECT_EQ(rtc::nullopt, StringToNumber<T>(std::string(kInvalidCharArray)));
>-  EXPECT_EQ(rtc::nullopt, StringToNumber<T>(kPlusMinusCharArray));
>-  EXPECT_EQ(rtc::nullopt, StringToNumber<T>(std::string(kPlusMinusCharArray)));
>-  EXPECT_EQ(rtc::nullopt, StringToNumber<T>(kNumberFollowedByCruft));
>-  EXPECT_EQ(rtc::nullopt,
>+  EXPECT_EQ(absl::nullopt, StringToNumber<T>(kInvalidCharArray));
>+  EXPECT_EQ(absl::nullopt, StringToNumber<T>(std::string(kInvalidCharArray)));
>+  EXPECT_EQ(absl::nullopt, StringToNumber<T>(kPlusMinusCharArray));
>+  EXPECT_EQ(absl::nullopt, StringToNumber<T>(std::string(kPlusMinusCharArray)));
>+  EXPECT_EQ(absl::nullopt, StringToNumber<T>(kNumberFollowedByCruft));
>+  EXPECT_EQ(absl::nullopt,
>             StringToNumber<T>(std::string(kNumberFollowedByCruft)));
>-  EXPECT_EQ(rtc::nullopt, StringToNumber<T>(" 5"));
>-  EXPECT_EQ(rtc::nullopt, StringToNumber<T>(" - 5"));
>-  EXPECT_EQ(rtc::nullopt, StringToNumber<T>("- 5"));
>-  EXPECT_EQ(rtc::nullopt, StringToNumber<T>(" -5"));
>-  EXPECT_EQ(rtc::nullopt, StringToNumber<T>("5 "));
>+  EXPECT_EQ(absl::nullopt, StringToNumber<T>(" 5"));
>+  EXPECT_EQ(absl::nullopt, StringToNumber<T>(" - 5"));
>+  EXPECT_EQ(absl::nullopt, StringToNumber<T>("- 5"));
>+  EXPECT_EQ(absl::nullopt, StringToNumber<T>(" -5"));
>+  EXPECT_EQ(absl::nullopt, StringToNumber<T>("5 "));
> }
> 
> REGISTER_TYPED_TEST_CASE_P(BasicNumberTest,
>@@ -104,10 +105,10 @@ INSTANTIATE_TYPED_TEST_CASE_P(StringToNumberTest_Integers,
>                               IntegerTypes);
> 
> TEST(StringToNumberTest, TestSpecificValues) {
>-  EXPECT_EQ(rtc::nullopt, StringToNumber<uint8_t>("256"));
>-  EXPECT_EQ(rtc::nullopt, StringToNumber<uint8_t>("-256"));
>-  EXPECT_EQ(rtc::nullopt, StringToNumber<int8_t>("256"));
>-  EXPECT_EQ(rtc::nullopt, StringToNumber<int8_t>("-256"));
>+  EXPECT_EQ(absl::nullopt, StringToNumber<uint8_t>("256"));
>+  EXPECT_EQ(absl::nullopt, StringToNumber<uint8_t>("-256"));
>+  EXPECT_EQ(absl::nullopt, StringToNumber<int8_t>("256"));
>+  EXPECT_EQ(absl::nullopt, StringToNumber<int8_t>("-256"));
> }
> 
> }  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringencode.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringencode.cc
>index 755cb2ca026..b4104772eab 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringencode.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringencode.cc
>@@ -13,6 +13,7 @@
> #include <stdio.h>
> #include <stdlib.h>
> 
>+#include "rtc_base/arraysize.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/stringutils.h"
> 
>@@ -22,8 +23,10 @@ namespace rtc {
> // String Encoding Utilities
> /////////////////////////////////////////////////////////////////////////////
> 
>-size_t url_decode(char * buffer, size_t buflen,
>-                  const char * source, size_t srclen) {
>+size_t url_decode(char* buffer,
>+                  size_t buflen,
>+                  const char* source,
>+                  size_t srclen) {
>   if (nullptr == buffer)
>     return srclen + 1;
>   if (buflen <= 0)
>@@ -35,11 +38,9 @@ size_t url_decode(char * buffer, size_t buflen,
>     unsigned char ch = source[srcpos++];
>     if (ch == '+') {
>       buffer[bufpos++] = ' ';
>-    } else if ((ch == '%')
>-               && (srcpos + 1 < srclen)
>-               && hex_decode(source[srcpos], &h1)
>-               && hex_decode(source[srcpos+1], &h2))
>-    {
>+    } else if ((ch == '%') && (srcpos + 1 < srclen) &&
>+               hex_decode(source[srcpos], &h1) &&
>+               hex_decode(source[srcpos + 1], &h2)) {
>       buffer[bufpos++] = (h1 << 4) | h2;
>       srcpos += 2;
>     } else {
>@@ -52,7 +53,7 @@ size_t url_decode(char * buffer, size_t buflen,
> 
> size_t utf8_decode(const char* source, size_t srclen, unsigned long* value) {
>   const unsigned char* s = reinterpret_cast<const unsigned char*>(source);
>-  if ((s[0] & 0x80) == 0x00) {                    // Check s[0] == 0xxxxxxx
>+  if ((s[0] & 0x80) == 0x00) {  // Check s[0] == 0xxxxxxx
>     *value = s[0];
>     return 1;
>   }
>@@ -62,7 +63,7 @@ size_t utf8_decode(const char* source, size_t srclen, unsigned long* value) {
>   // Accumulate the trailer byte values in value16, and combine it with the
>   // relevant bits from s[0], once we've determined the sequence length.
>   unsigned long value16 = (s[1] & 0x3F);
>-  if ((s[0] & 0xE0) == 0xC0) {                    // Check s[0] == 110xxxxx
>+  if ((s[0] & 0xE0) == 0xC0) {  // Check s[0] == 110xxxxx
>     *value = ((s[0] & 0x1F) << 6) | value16;
>     return 2;
>   }
>@@ -70,7 +71,7 @@ size_t utf8_decode(const char* source, size_t srclen, unsigned long* value) {
>     return 0;
>   }
>   value16 = (value16 << 6) | (s[2] & 0x3F);
>-  if ((s[0] & 0xF0) == 0xE0) {                    // Check s[0] == 1110xxxx
>+  if ((s[0] & 0xF0) == 0xE0) {  // Check s[0] == 1110xxxx
>     *value = ((s[0] & 0x0F) << 12) | value16;
>     return 3;
>   }
>@@ -78,7 +79,7 @@ size_t utf8_decode(const char* source, size_t srclen, unsigned long* value) {
>     return 0;
>   }
>   value16 = (value16 << 6) | (s[3] & 0x3F);
>-  if ((s[0] & 0xF8) == 0xF0) {                    // Check s[0] == 11110xxx
>+  if ((s[0] & 0xF8) == 0xF0) {  // Check s[0] == 11110xxx
>     *value = ((s[0] & 0x07) << 18) | value16;
>     return 4;
>   }
>@@ -131,15 +132,19 @@ bool hex_decode(char ch, unsigned char* val) {
>   return true;
> }
> 
>-size_t hex_encode(char* buffer, size_t buflen,
>-                  const char* csource, size_t srclen) {
>+size_t hex_encode(char* buffer,
>+                  size_t buflen,
>+                  const char* csource,
>+                  size_t srclen) {
>   return hex_encode_with_delimiter(buffer, buflen, csource, srclen, 0);
> }
> 
>-size_t hex_encode_with_delimiter(char* buffer, size_t buflen,
>-                                 const char* csource, size_t srclen,
>+size_t hex_encode_with_delimiter(char* buffer,
>+                                 size_t buflen,
>+                                 const char* csource,
>+                                 size_t srclen,
>                                  char delimiter) {
>-  RTC_DCHECK(buffer);  // TODO(grunell): estimate output size
>+  RTC_DCHECK(buffer);  // TODO(kwiberg): estimate output size
>   if (buflen == 0)
>     return 0;
> 
>@@ -153,8 +158,8 @@ size_t hex_encode_with_delimiter(char* buffer, size_t buflen,
> 
>   while (srcpos < srclen) {
>     unsigned char ch = bsource[srcpos++];
>-    buffer[bufpos  ] = hex_encode((ch >> 4) & 0xF);
>-    buffer[bufpos+1] = hex_encode((ch     ) & 0xF);
>+    buffer[bufpos] = hex_encode((ch >> 4) & 0xF);
>+    buffer[bufpos + 1] = hex_encode((ch)&0xF);
>     bufpos += 2;
> 
>     // Don't write a delimiter after the last byte.
>@@ -177,25 +182,30 @@ std::string hex_encode(const char* source, size_t srclen) {
>   return hex_encode_with_delimiter(source, srclen, 0);
> }
> 
>-std::string hex_encode_with_delimiter(const char* source, size_t srclen,
>+std::string hex_encode_with_delimiter(const char* source,
>+                                      size_t srclen,
>                                       char delimiter) {
>   const size_t kBufferSize = srclen * 3;
>   char* buffer = STACK_ARRAY(char, kBufferSize);
>-  size_t length = hex_encode_with_delimiter(buffer, kBufferSize,
>-                                            source, srclen, delimiter);
>+  size_t length =
>+      hex_encode_with_delimiter(buffer, kBufferSize, source, srclen, delimiter);
>   RTC_DCHECK(srclen == 0 || length > 0);
>   return std::string(buffer, length);
> }
> 
>-size_t hex_decode(char * cbuffer, size_t buflen,
>-                  const char * source, size_t srclen) {
>+size_t hex_decode(char* cbuffer,
>+                  size_t buflen,
>+                  const char* source,
>+                  size_t srclen) {
>   return hex_decode_with_delimiter(cbuffer, buflen, source, srclen, 0);
> }
> 
>-size_t hex_decode_with_delimiter(char* cbuffer, size_t buflen,
>-                                 const char* source, size_t srclen,
>+size_t hex_decode_with_delimiter(char* cbuffer,
>+                                 size_t buflen,
>+                                 const char* source,
>+                                 size_t srclen,
>                                  char delimiter) {
>-  RTC_DCHECK(cbuffer);  // TODO(grunell): estimate output size
>+  RTC_DCHECK(cbuffer);  // TODO(kwiberg): estimate output size
>   if (buflen == 0)
>     return 0;
> 
>@@ -234,13 +244,17 @@ size_t hex_decode_with_delimiter(char* cbuffer, size_t buflen,
> size_t hex_decode(char* buffer, size_t buflen, const std::string& source) {
>   return hex_decode_with_delimiter(buffer, buflen, source, 0);
> }
>-size_t hex_decode_with_delimiter(char* buffer, size_t buflen,
>-                                 const std::string& source, char delimiter) {
>-  return hex_decode_with_delimiter(buffer, buflen,
>-                                   source.c_str(), source.length(), delimiter);
>+size_t hex_decode_with_delimiter(char* buffer,
>+                                 size_t buflen,
>+                                 const std::string& source,
>+                                 char delimiter) {
>+  return hex_decode_with_delimiter(buffer, buflen, source.c_str(),
>+                                   source.length(), delimiter);
> }
> 
>-size_t transform(std::string& value, size_t maxlen, const std::string& source,
>+size_t transform(std::string& value,
>+                 size_t maxlen,
>+                 const std::string& source,
>                  Transform t) {
>   char* buffer = STACK_ARRAY(char, maxlen + 1);
>   size_t length = t(buffer, maxlen + 1, source.data(), source.length());
>@@ -249,15 +263,17 @@ size_t transform(std::string& value, size_t maxlen, const std::string& source,
> }
> 
> std::string s_transform(const std::string& source, Transform t) {
>-  // Ask transformation function to approximate the destination size (returns upper bound)
>+  // Ask transformation function to approximate the destination size (returns
>+  // upper bound)
>   size_t maxlen = t(nullptr, 0, source.data(), source.length());
>-  char * buffer = STACK_ARRAY(char, maxlen);
>+  char* buffer = STACK_ARRAY(char, maxlen);
>   size_t len = t(buffer, maxlen, source.data(), source.length());
>   std::string result(buffer, len);
>   return result;
> }
> 
>-size_t tokenize(const std::string& source, char delimiter,
>+size_t tokenize(const std::string& source,
>+                char delimiter,
>                 std::vector<std::string>* fields) {
>   fields->clear();
>   size_t last = 0;
>@@ -290,9 +306,11 @@ size_t tokenize_with_empty_tokens(const std::string& source,
>   return fields->size();
> }
> 
>-size_t tokenize_append(const std::string& source, char delimiter,
>+size_t tokenize_append(const std::string& source,
>+                       char delimiter,
>                        std::vector<std::string>* fields) {
>-  if (!fields) return 0;
>+  if (!fields)
>+    return 0;
> 
>   std::vector<std::string> new_fields;
>   tokenize(source, delimiter, &new_fields);
>@@ -300,15 +318,20 @@ size_t tokenize_append(const std::string& source, char delimiter,
>   return fields->size();
> }
> 
>-size_t tokenize(const std::string& source, char delimiter, char start_mark,
>-                char end_mark, std::vector<std::string>* fields) {
>-  if (!fields) return 0;
>+size_t tokenize(const std::string& source,
>+                char delimiter,
>+                char start_mark,
>+                char end_mark,
>+                std::vector<std::string>* fields) {
>+  if (!fields)
>+    return 0;
>   fields->clear();
> 
>   std::string remain_source = source;
>   while (!remain_source.empty()) {
>     size_t start_pos = remain_source.find(start_mark);
>-    if (std::string::npos == start_pos) break;
>+    if (std::string::npos == start_pos)
>+      break;
>     std::string pre_mark;
>     if (start_pos > 0) {
>       pre_mark = remain_source.substr(0, start_pos - 1);
>@@ -316,7 +339,8 @@ size_t tokenize(const std::string& source, char delimiter, char start_mark,
> 
>     ++start_pos;
>     size_t end_pos = remain_source.find(end_mark, start_pos);
>-    if (std::string::npos == end_pos) break;
>+    if (std::string::npos == end_pos)
>+      break;
> 
>     // We have found the matching marks. First tokenize the pre-mask. Then add
>     // the marked part as a single field. Finally, loop back for the post-mark.
>@@ -371,7 +395,8 @@ std::string join(const std::vector<std::string>& source, char delimiter) {
>   return joined_string;
> }
> 
>-size_t split(const std::string& source, char delimiter,
>+size_t split(const std::string& source,
>+             char delimiter,
>              std::vector<std::string>* fields) {
>   RTC_DCHECK(fields);
>   fields->clear();
>@@ -386,4 +411,90 @@ size_t split(const std::string& source, char delimiter,
>   return fields->size();
> }
> 
>+std::string ToString(const bool b) {
>+  return b ? "true" : "false";
>+}
>+
>+std::string ToString(const char* const s) {
>+  return std::string(s);
>+}
>+std::string ToString(const std::string s) {
>+  return s;
>+}
>+
>+std::string ToString(const short s) {
>+  char buf[32];
>+  const int len = std::snprintf(&buf[0], arraysize(buf), "%hd", s);
>+  RTC_DCHECK_LE(len, arraysize(buf));
>+  return std::string(&buf[0], len);
>+}
>+std::string ToString(const unsigned short s) {
>+  char buf[32];
>+  const int len = std::snprintf(&buf[0], arraysize(buf), "%hu", s);
>+  RTC_DCHECK_LE(len, arraysize(buf));
>+  return std::string(&buf[0], len);
>+}
>+std::string ToString(const int s) {
>+  char buf[32];
>+  const int len = std::snprintf(&buf[0], arraysize(buf), "%d", s);
>+  RTC_DCHECK_LE(len, arraysize(buf));
>+  return std::string(&buf[0], len);
>+}
>+std::string ToString(const unsigned int s) {
>+  char buf[32];
>+  const int len = std::snprintf(&buf[0], arraysize(buf), "%u", s);
>+  RTC_DCHECK_LE(len, arraysize(buf));
>+  return std::string(&buf[0], len);
>+}
>+std::string ToString(const long int s) {
>+  char buf[32];
>+  const int len = std::snprintf(&buf[0], arraysize(buf), "%ld", s);
>+  RTC_DCHECK_LE(len, arraysize(buf));
>+  return std::string(&buf[0], len);
>+}
>+std::string ToString(const unsigned long int s) {
>+  char buf[32];
>+  const int len = std::snprintf(&buf[0], arraysize(buf), "%lu", s);
>+  RTC_DCHECK_LE(len, arraysize(buf));
>+  return std::string(&buf[0], len);
>+}
>+std::string ToString(const long long int s) {
>+  char buf[32];
>+  const int len = std::snprintf(&buf[0], arraysize(buf), "%lld", s);
>+  RTC_DCHECK_LE(len, arraysize(buf));
>+  return std::string(&buf[0], len);
>+}
>+std::string ToString(const unsigned long long int s) {
>+  char buf[32];
>+  const int len = std::snprintf(&buf[0], arraysize(buf), "%llu", s);
>+  RTC_DCHECK_LE(len, arraysize(buf));
>+  return std::string(&buf[0], len);
>+}
>+
>+std::string ToString(const double d) {
>+  char buf[32];
>+  const int len = std::snprintf(&buf[0], arraysize(buf), "%g", d);
>+  RTC_DCHECK_LE(len, arraysize(buf));
>+  return std::string(&buf[0], len);
>+}
>+
>+std::string ToString(const void* const p) {
>+  char buf[32];
>+  const int len = std::snprintf(&buf[0], arraysize(buf), "%p", p);
>+  RTC_DCHECK_LE(len, arraysize(buf));
>+  return std::string(&buf[0], len);
>+}
>+
>+bool FromString(const std::string& s, bool* b) {
>+  if (s == "false") {
>+    *b = false;
>+    return true;
>+  }
>+  if (s == "true") {
>+    *b = true;
>+    return true;
>+  }
>+  return false;
>+}
>+
> }  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringencode.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringencode.h
>index e5395b7edf2..830ea77c7b9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringencode.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringencode.h
>@@ -16,6 +16,7 @@
> #include <vector>
> 
> #include "rtc_base/checks.h"
>+#include "rtc_base/string_to_number.h"
> 
> namespace rtc {
> 
>@@ -24,8 +25,10 @@ namespace rtc {
> //////////////////////////////////////////////////////////////////////
> 
> // Note: in-place decoding (buffer == source) is allowed.
>-size_t url_decode(char * buffer, size_t buflen,
>-                  const char * source, size_t srclen);
>+size_t url_decode(char* buffer,
>+                  size_t buflen,
>+                  const char* source,
>+                  size_t srclen);
> 
> // Convert an unsigned value from 0 to 15 to the hex character equivalent...
> char hex_encode(unsigned char val);
>@@ -33,45 +36,60 @@ char hex_encode(unsigned char val);
> bool hex_decode(char ch, unsigned char* val);
> 
> // hex_encode shows the hex representation of binary data in ascii.
>-size_t hex_encode(char* buffer, size_t buflen,
>-                  const char* source, size_t srclen);
>+size_t hex_encode(char* buffer,
>+                  size_t buflen,
>+                  const char* source,
>+                  size_t srclen);
> 
> // hex_encode, but separate each byte representation with a delimiter.
> // |delimiter| == 0 means no delimiter
> // If the buffer is too short, we return 0
>-size_t hex_encode_with_delimiter(char* buffer, size_t buflen,
>-                                 const char* source, size_t srclen,
>+size_t hex_encode_with_delimiter(char* buffer,
>+                                 size_t buflen,
>+                                 const char* source,
>+                                 size_t srclen,
>                                  char delimiter);
> 
> // Helper functions for hex_encode.
> std::string hex_encode(const std::string& str);
> std::string hex_encode(const char* source, size_t srclen);
>-std::string hex_encode_with_delimiter(const char* source, size_t srclen,
>+std::string hex_encode_with_delimiter(const char* source,
>+                                      size_t srclen,
>                                       char delimiter);
> 
> // hex_decode converts ascii hex to binary.
>-size_t hex_decode(char* buffer, size_t buflen,
>-                  const char* source, size_t srclen);
>+size_t hex_decode(char* buffer,
>+                  size_t buflen,
>+                  const char* source,
>+                  size_t srclen);
> 
> // hex_decode, assuming that there is a delimiter between every byte
> // pair.
> // |delimiter| == 0 means no delimiter
> // If the buffer is too short or the data is invalid, we return 0.
>-size_t hex_decode_with_delimiter(char* buffer, size_t buflen,
>-                                 const char* source, size_t srclen,
>+size_t hex_decode_with_delimiter(char* buffer,
>+                                 size_t buflen,
>+                                 const char* source,
>+                                 size_t srclen,
>                                  char delimiter);
> 
> // Helper functions for hex_decode.
> size_t hex_decode(char* buffer, size_t buflen, const std::string& source);
>-size_t hex_decode_with_delimiter(char* buffer, size_t buflen,
>-                                 const std::string& source, char delimiter);
>+size_t hex_decode_with_delimiter(char* buffer,
>+                                 size_t buflen,
>+                                 const std::string& source,
>+                                 char delimiter);
> 
> // Apply any suitable string transform (including the ones above) to an STL
> // string.  Stack-allocated temporary space is used for the transformation,
> // so value and source may refer to the same string.
>-typedef size_t (*Transform)(char * buffer, size_t buflen,
>-                            const char * source, size_t srclen);
>-size_t transform(std::string& value, size_t maxlen, const std::string& source,
>+typedef size_t (*Transform)(char* buffer,
>+                            size_t buflen,
>+                            const char* source,
>+                            size_t srclen);
>+size_t transform(std::string& value,
>+                 size_t maxlen,
>+                 const std::string& source,
>                  Transform t);
> 
> // Return the result of applying transform t to source.
>@@ -88,12 +106,14 @@ std::string join(const std::vector<std::string>& source, char delimiter);
> 
> // Splits the source string into multiple fields separated by delimiter,
> // with duplicates of delimiter creating empty fields.
>-size_t split(const std::string& source, char delimiter,
>+size_t split(const std::string& source,
>+             char delimiter,
>              std::vector<std::string>* fields);
> 
> // Splits the source string into multiple fields separated by delimiter,
> // with duplicates of delimiter ignored.  Trailing delimiter ignored.
>-size_t tokenize(const std::string& source, char delimiter,
>+size_t tokenize(const std::string& source,
>+                char delimiter,
>                 std::vector<std::string>* fields);
> 
> // Tokenize, including the empty tokens.
>@@ -102,7 +122,8 @@ size_t tokenize_with_empty_tokens(const std::string& source,
>                                   std::vector<std::string>* fields);
> 
> // Tokenize and append the tokens to fields. Return the new size of fields.
>-size_t tokenize_append(const std::string& source, char delimiter,
>+size_t tokenize_append(const std::string& source,
>+                       char delimiter,
>                        std::vector<std::string>* fields);
> 
> // Splits the source string into multiple fields separated by delimiter, with
>@@ -112,8 +133,11 @@ size_t tokenize_append(const std::string& source, char delimiter,
> // \"/Library/Application Support/media content.txt\"", delimiter is ' ', and
> // the start_mark and end_mark are '"', this method returns two fields:
> // "filename" and "/Library/Application Support/media content.txt".
>-size_t tokenize(const std::string& source, char delimiter, char start_mark,
>-                char end_mark, std::vector<std::string>* fields);
>+size_t tokenize(const std::string& source,
>+                char delimiter,
>+                char start_mark,
>+                char end_mark,
>+                std::vector<std::string>* fields);
> 
> // Extract the first token from source as separated by delimiter, with
> // duplicates of delimiter ignored. Return false if the delimiter could not be
>@@ -124,39 +148,46 @@ bool tokenize_first(const std::string& source,
>                     std::string* rest);
> 
> // Convert arbitrary values to/from a string.
>+// TODO(jonasolsson): Remove these when absl::StrCat becomes available.
>+std::string ToString(bool b);
> 
>-template <class T>
>-static bool ToString(const T &t, std::string* s) {
>-  RTC_DCHECK(s);
>-  std::ostringstream oss;
>-  oss << std::boolalpha << t;
>-  *s = oss.str();
>-  return !oss.fail();
>-}
>+std::string ToString(const char* s);
>+std::string ToString(std::string t);
>+
>+std::string ToString(short s);
>+std::string ToString(unsigned short s);
>+std::string ToString(int s);
>+std::string ToString(unsigned int s);
>+std::string ToString(long int s);
>+std::string ToString(unsigned long int s);
>+std::string ToString(long long int s);
>+std::string ToString(unsigned long long int s);
>+
>+std::string ToString(double t);
> 
>-template <class T>
>+std::string ToString(const void* p);
>+
>+template <typename T,
>+          typename std::enable_if<std::is_arithmetic<T>::value &&
>+                                      !std::is_same<T, bool>::value,
>+                                  int>::type = 0>
> static bool FromString(const std::string& s, T* t) {
>   RTC_DCHECK(t);
>-  std::istringstream iss(s);
>-  iss >> std::boolalpha >> *t;
>-  return !iss.fail();
>-}
>+  absl::optional<T> result = StringToNumber<T>(s);
> 
>-// Inline versions of the string conversion routines.
>+  if (result)
>+    *t = *result;
> 
>-template<typename T>
>-static inline std::string ToString(const T& val) {
>-  std::string str; ToString(val, &str); return str;
>+  return result.has_value();
> }
> 
>-template<typename T>
>-static inline T FromString(const std::string& str) {
>-  T val; FromString(str, &val); return val;
>-}
>+bool FromString(const std::string& s, bool* b);
> 
>-template<typename T>
>-static inline T FromString(const T& defaultValue, const std::string& str) {
>-  T val(defaultValue); FromString(str, &val); return val;
>+template <typename T>
>+static inline T FromString(const std::string& str) {
>+  T val;
>+  FromString(str, &val);
>+  return val;
> }
> 
> //////////////////////////////////////////////////////////////////////
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringencode_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringencode_unittest.cc
>index 63d8290276e..9bdc592c29a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringencode_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringencode_unittest.cc
>@@ -13,6 +13,8 @@
> #include "rtc_base/gunit.h"
> #include "rtc_base/stringutils.h"
> 
>+#include <sstream>  // no-presubmit-check TODO(webrtc:8982)
>+
> namespace rtc {
> 
> class HexEncodeTest : public testing::Test {
>@@ -43,23 +45,23 @@ TEST_F(HexEncodeTest, TestWithNoDelimiter) {
> 
> // Test that we can convert to/from hex with a colon delimiter.
> TEST_F(HexEncodeTest, TestWithDelimiter) {
>-  enc_res_ = hex_encode_with_delimiter(encoded_, sizeof(encoded_),
>-                                       data_, sizeof(data_), ':');
>+  enc_res_ = hex_encode_with_delimiter(encoded_, sizeof(encoded_), data_,
>+                                       sizeof(data_), ':');
>   ASSERT_EQ(sizeof(data_) * 3 - 1, enc_res_);
>   ASSERT_STREQ("80:81:82:83:84:85:86:87:88:89", encoded_);
>-  dec_res_ = hex_decode_with_delimiter(decoded_, sizeof(decoded_),
>-                                       encoded_, enc_res_, ':');
>+  dec_res_ = hex_decode_with_delimiter(decoded_, sizeof(decoded_), encoded_,
>+                                       enc_res_, ':');
>   ASSERT_EQ(sizeof(data_), dec_res_);
>   ASSERT_EQ(0, memcmp(data_, decoded_, dec_res_));
> }
> 
> // Test that encoding with one delimiter and decoding with another fails.
> TEST_F(HexEncodeTest, TestWithWrongDelimiter) {
>-  enc_res_ = hex_encode_with_delimiter(encoded_, sizeof(encoded_),
>-                                       data_, sizeof(data_), ':');
>+  enc_res_ = hex_encode_with_delimiter(encoded_, sizeof(encoded_), data_,
>+                                       sizeof(data_), ':');
>   ASSERT_EQ(sizeof(data_) * 3 - 1, enc_res_);
>-  dec_res_ = hex_decode_with_delimiter(decoded_, sizeof(decoded_),
>-                                       encoded_, enc_res_, '/');
>+  dec_res_ = hex_decode_with_delimiter(decoded_, sizeof(decoded_), encoded_,
>+                                       enc_res_, '/');
>   ASSERT_EQ(0U, dec_res_);
> }
> 
>@@ -67,15 +69,15 @@ TEST_F(HexEncodeTest, TestWithWrongDelimiter) {
> TEST_F(HexEncodeTest, TestExpectedDelimiter) {
>   enc_res_ = hex_encode(encoded_, sizeof(encoded_), data_, sizeof(data_));
>   ASSERT_EQ(sizeof(data_) * 2, enc_res_);
>-  dec_res_ = hex_decode_with_delimiter(decoded_, sizeof(decoded_),
>-                                       encoded_, enc_res_, ':');
>+  dec_res_ = hex_decode_with_delimiter(decoded_, sizeof(decoded_), encoded_,
>+                                       enc_res_, ':');
>   ASSERT_EQ(0U, dec_res_);
> }
> 
> // Test that encoding with a delimiter and decoding without one fails.
> TEST_F(HexEncodeTest, TestExpectedNoDelimiter) {
>-  enc_res_ = hex_encode_with_delimiter(encoded_, sizeof(encoded_),
>-                                       data_, sizeof(data_), ':');
>+  enc_res_ = hex_encode_with_delimiter(encoded_, sizeof(encoded_), data_,
>+                                       sizeof(data_), ':');
>   ASSERT_EQ(sizeof(data_) * 3 - 1, enc_res_);
>   dec_res_ = hex_decode(decoded_, sizeof(decoded_), encoded_, enc_res_);
>   ASSERT_EQ(0U, dec_res_);
>@@ -93,8 +95,8 @@ TEST_F(HexEncodeTest, TestZeroLengthNoDelimiter) {
> TEST_F(HexEncodeTest, TestZeroLengthWithDelimiter) {
>   enc_res_ = hex_encode_with_delimiter(encoded_, sizeof(encoded_), "", 0, ':');
>   ASSERT_EQ(0U, enc_res_);
>-  dec_res_ = hex_decode_with_delimiter(decoded_, sizeof(decoded_),
>-                                       encoded_, enc_res_, ':');
>+  dec_res_ = hex_decode_with_delimiter(decoded_, sizeof(decoded_), encoded_,
>+                                       enc_res_, ':');
>   ASSERT_EQ(0U, dec_res_);
> }
> 
>@@ -118,15 +120,15 @@ TEST_F(HexEncodeTest, TestHelpersWithDelimiter) {
> 
> // Test that encoding into a too-small output buffer (without delimiter) fails.
> TEST_F(HexEncodeTest, TestEncodeTooShort) {
>-  enc_res_ = hex_encode_with_delimiter(encoded_, sizeof(data_) * 2,
>-                                       data_, sizeof(data_), 0);
>+  enc_res_ = hex_encode_with_delimiter(encoded_, sizeof(data_) * 2, data_,
>+                                       sizeof(data_), 0);
>   ASSERT_EQ(0U, enc_res_);
> }
> 
> // Test that encoding into a too-small output buffer (with delimiter) fails.
> TEST_F(HexEncodeTest, TestEncodeWithDelimiterTooShort) {
>-  enc_res_ = hex_encode_with_delimiter(encoded_, sizeof(data_) * 3 - 1,
>-                                       data_, sizeof(data_), ':');
>+  enc_res_ = hex_encode_with_delimiter(encoded_, sizeof(data_) * 3 - 1, data_,
>+                                       sizeof(data_), ':');
>   ASSERT_EQ(0U, enc_res_);
> }
> 
>@@ -351,53 +353,79 @@ TEST(SplitTest, CompareSubstrings) {
>   ASSERT_STREQ("", fields.at(0).c_str());
> }
> 
>-TEST(BoolTest, DecodeValid) {
>-  bool value;
>-  EXPECT_TRUE(FromString("true", &value));
>-  EXPECT_TRUE(value);
>-  EXPECT_TRUE(FromString("true,", &value));
>-  EXPECT_TRUE(value);
>-  EXPECT_TRUE(FromString("true , true", &value));
>-  EXPECT_TRUE(value);
>-  EXPECT_TRUE(FromString("true ,\n false", &value));
>-  EXPECT_TRUE(value);
>-  EXPECT_TRUE(FromString("  true  \n", &value));
>-  EXPECT_TRUE(value);
>-
>-  EXPECT_TRUE(FromString("false", &value));
>-  EXPECT_FALSE(value);
>-  EXPECT_TRUE(FromString("  false ", &value));
>-  EXPECT_FALSE(value);
>-  EXPECT_TRUE(FromString("  false, ", &value));
>-  EXPECT_FALSE(value);
>-
>-  EXPECT_TRUE(FromString<bool>("true\n"));
>-  EXPECT_FALSE(FromString<bool>("false\n"));
>-}
>-
>-TEST(BoolTest, DecodeInvalid) {
>-  bool value;
>-  EXPECT_FALSE(FromString("True", &value));
>-  EXPECT_FALSE(FromString("TRUE", &value));
>-  EXPECT_FALSE(FromString("False", &value));
>-  EXPECT_FALSE(FromString("FALSE", &value));
>-  EXPECT_FALSE(FromString("0", &value));
>-  EXPECT_FALSE(FromString("1", &value));
>-  EXPECT_FALSE(FromString("0,", &value));
>-  EXPECT_FALSE(FromString("1,", &value));
>-  EXPECT_FALSE(FromString("1,0", &value));
>-  EXPECT_FALSE(FromString("1.", &value));
>-  EXPECT_FALSE(FromString("1.0", &value));
>-  EXPECT_FALSE(FromString("", &value));
>-  EXPECT_FALSE(FromString<bool>("false\nfalse"));
>-}
>-
>-TEST(BoolTest, RoundTrip) {
>-  bool value;
>-  EXPECT_TRUE(FromString(ToString(true), &value));
>-  EXPECT_TRUE(value);
>-  EXPECT_TRUE(FromString(ToString(false), &value));
>-  EXPECT_FALSE(value);
>+TEST(ToString, SanityCheck) {
>+  EXPECT_EQ(ToString(true), "true");
>+  EXPECT_EQ(ToString(false), "false");
>+
>+  const char* c = "message";
>+  EXPECT_EQ(ToString(c), c);
>+  EXPECT_EQ(ToString(std::string(c)), c);
>+
>+  EXPECT_EQ(ToString(short{-123}), "-123");
>+  EXPECT_EQ(ToString((unsigned short)123), "123");
>+  EXPECT_EQ(ToString(int{-123}), "-123");
>+  EXPECT_EQ(ToString((unsigned int)123), "123");
>+  EXPECT_EQ(ToString((long int)-123), "-123");
>+  EXPECT_EQ(ToString((unsigned long int)123), "123");
>+  EXPECT_EQ(ToString((long long int)-123), "-123");
>+  EXPECT_EQ(ToString((unsigned long long int)123), "123");
>+
>+  int i = 10;
>+  int* p = &i;
>+  std::ostringstream s;  // no-presubmit-check TODO(webrtc:8982)
>+  s << p;
>+  EXPECT_EQ(s.str(), ToString(p));
>+
>+  EXPECT_EQ(ToString(0.5), "0.5");
>+}
>+
>+template <typename T>
>+void ParsesTo(std::string s, T t) {
>+  T value;
>+  EXPECT_TRUE(FromString(s, &value));
>+  EXPECT_EQ(value, t);
>+}
>+
>+TEST(FromString, DecodeValid) {
>+  ParsesTo("true", true);
>+  ParsesTo("false", false);
>+
>+  ParsesTo("105", 105);
>+  ParsesTo("0.25", 0.25);
>+}
>+
>+template <typename T>
>+void FailsToParse(std::string s) {
>+  T value;
>+  EXPECT_FALSE(FromString(s, &value)) << "[" << s << "]";
>+}
>+
>+TEST(FromString, DecodeInvalid) {
>+  FailsToParse<bool>("True");
>+  FailsToParse<bool>("0");
>+  FailsToParse<bool>("yes");
>+
>+  FailsToParse<int>("0.5");
>+  FailsToParse<int>("XIV");
>+  FailsToParse<double>("");
>+  FailsToParse<double>("  ");
>+  FailsToParse<int>("1 2");
>+}
>+
>+template <typename T>
>+void RoundTrip(T t) {
>+  std::string s = ToString(t);
>+  T value;
>+  EXPECT_TRUE(FromString(s, &value));
>+  EXPECT_EQ(value, t);
>+}
>+
>+TEST(FromString, RoundTrip) {
>+  RoundTrip<int>(123);
>+  RoundTrip(false);
>+  RoundTrip(true);
>+  RoundTrip(0.5);
>+  RoundTrip(-15l);
> }
> 
> }  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringize_macros.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringize_macros.h
>index acf0ba404ca..c94a90de1ba 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringize_macros.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringize_macros.h
>@@ -21,7 +21,7 @@
> // This is not very useful as it does not expand defined symbols if
> // called directly. Use its counterpart without the _NO_EXPANSION
> // suffix, below.
>-#define STRINGIZE_NO_EXPANSION(x) #x
>+#define RTC_STRINGIZE_NO_EXPANSION(x) #x
> 
> // Use this to quote the provided parameter, first expanding it if it
> // is a preprocessor symbol.
>@@ -33,6 +33,6 @@
> // Then:
> //   STRINGIZE(A) produces "FOO"
> //   STRINGIZE(B(y)) produces "myobj->FunctionCall(y)"
>-#define RTC_STRINGIZE(x) STRINGIZE_NO_EXPANSION(x)
>+#define RTC_STRINGIZE(x) RTC_STRINGIZE_NO_EXPANSION(x)
> 
> #endif  // RTC_BASE_STRINGIZE_MACROS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringize_macros_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringize_macros_unittest.cc
>index d1dea5e2bfe..78e6b55b2d2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringize_macros_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringize_macros_unittest.cc
>@@ -18,15 +18,12 @@
> #define PREPROCESSOR_UTIL_UNITTEST_C "foo"
> 
> TEST(StringizeTest, Ansi) {
>-  EXPECT_STREQ(
>-      "PREPROCESSOR_UTIL_UNITTEST_A",
>-      STRINGIZE_NO_EXPANSION(PREPROCESSOR_UTIL_UNITTEST_A));
>-  EXPECT_STREQ(
>-      "PREPROCESSOR_UTIL_UNITTEST_B(y)",
>-      STRINGIZE_NO_EXPANSION(PREPROCESSOR_UTIL_UNITTEST_B(y)));
>-  EXPECT_STREQ(
>-      "PREPROCESSOR_UTIL_UNITTEST_C",
>-      STRINGIZE_NO_EXPANSION(PREPROCESSOR_UTIL_UNITTEST_C));
>+  EXPECT_STREQ("PREPROCESSOR_UTIL_UNITTEST_A",
>+               STRINGIZE_NO_EXPANSION(PREPROCESSOR_UTIL_UNITTEST_A));
>+  EXPECT_STREQ("PREPROCESSOR_UTIL_UNITTEST_B(y)",
>+               STRINGIZE_NO_EXPANSION(PREPROCESSOR_UTIL_UNITTEST_B(y)));
>+  EXPECT_STREQ("PREPROCESSOR_UTIL_UNITTEST_C",
>+               STRINGIZE_NO_EXPANSION(PREPROCESSOR_UTIL_UNITTEST_C));
> 
>   EXPECT_STREQ("FOO", STRINGIZE(PREPROCESSOR_UTIL_UNITTEST_A));
>   EXPECT_STREQ("myobj->FunctionCall(y)",
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/strings/audio_format_to_string.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/strings/audio_format_to_string.cc
>new file mode 100644
>index 00000000000..a1493441985
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/strings/audio_format_to_string.cc
>@@ -0,0 +1,52 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "rtc_base/strings/audio_format_to_string.h"
>+
>+#include "rtc_base/strings/string_builder.h"
>+
>+namespace rtc {
>+std::string ToString(const webrtc::SdpAudioFormat& saf) {
>+  char sb_buf[1024];
>+  rtc::SimpleStringBuilder sb(sb_buf);
>+  sb << "{name: " << saf.name;
>+  sb << ", clockrate_hz: " << saf.clockrate_hz;
>+  sb << ", num_channels: " << saf.num_channels;
>+  sb << ", parameters: {";
>+  const char* sep = "";
>+  for (const auto& kv : saf.parameters) {
>+    sb << sep << kv.first << ": " << kv.second;
>+    sep = ", ";
>+  }
>+  sb << "}}";
>+  return sb.str();
>+}
>+std::string ToString(const webrtc::AudioCodecInfo& aci) {
>+  char sb_buf[1024];
>+  rtc::SimpleStringBuilder sb(sb_buf);
>+  sb << "{sample_rate_hz: " << aci.sample_rate_hz;
>+  sb << ", num_channels: " << aci.num_channels;
>+  sb << ", default_bitrate_bps: " << aci.default_bitrate_bps;
>+  sb << ", min_bitrate_bps: " << aci.min_bitrate_bps;
>+  sb << ", max_bitrate_bps: " << aci.max_bitrate_bps;
>+  sb << ", allow_comfort_noise: " << aci.allow_comfort_noise;
>+  sb << ", supports_network_adaption: " << aci.supports_network_adaption;
>+  sb << "}";
>+  return sb.str();
>+}
>+std::string ToString(const webrtc::AudioCodecSpec& acs) {
>+  char sb_buf[1024];
>+  rtc::SimpleStringBuilder sb(sb_buf);
>+  sb << "{format: " << ToString(acs.format);
>+  sb << ", info: " << ToString(acs.info);
>+  sb << "}";
>+  return sb.str();
>+}
>+}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/strings/audio_format_to_string.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/strings/audio_format_to_string.h
>new file mode 100644
>index 00000000000..6a4535c9394
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/strings/audio_format_to_string.h
>@@ -0,0 +1,24 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef RTC_BASE_STRINGS_AUDIO_FORMAT_TO_STRING_H_
>+#define RTC_BASE_STRINGS_AUDIO_FORMAT_TO_STRING_H_
>+
>+#include <string>
>+
>+#include "api/audio_codecs/audio_format.h"
>+
>+namespace rtc {
>+std::string ToString(const webrtc::SdpAudioFormat& saf);
>+std::string ToString(const webrtc::AudioCodecInfo& saf);
>+std::string ToString(const webrtc::AudioCodecSpec& acs);
>+}  // namespace rtc
>+
>+#endif  // RTC_BASE_STRINGS_AUDIO_FORMAT_TO_STRING_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/strings/string_builder.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/strings/string_builder.cc
>new file mode 100644
>index 00000000000..70a14da4ca6
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/strings/string_builder.cc
>@@ -0,0 +1,112 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "rtc_base/strings/string_builder.h"
>+
>+namespace rtc {
>+
>+SimpleStringBuilder::SimpleStringBuilder(rtc::ArrayView<char> buffer)
>+    : buffer_(buffer) {
>+  buffer_[0] = '\0';
>+  RTC_DCHECK(IsConsistent());
>+}
>+
>+SimpleStringBuilder& SimpleStringBuilder::operator<<(const char* str) {
>+  return Append(str);
>+}
>+
>+SimpleStringBuilder& SimpleStringBuilder::operator<<(char ch) {
>+  return Append(&ch, 1);
>+}
>+
>+SimpleStringBuilder& SimpleStringBuilder::operator<<(const std::string& str) {
>+  return Append(str.c_str(), str.length());
>+}
>+
>+// Numeric conversion routines.
>+//
>+// We use std::[v]snprintf instead of std::to_string because:
>+// * std::to_string relies on the current locale for formatting purposes,
>+//   and therefore concurrent calls to std::to_string from multiple threads
>+//   may result in partial serialization of calls
>+// * snprintf allows us to print the number directly into our buffer.
>+// * avoid allocating a std::string (potential heap alloc).
>+// TODO(tommi): Switch to std::to_chars in C++17.
>+
>+SimpleStringBuilder& SimpleStringBuilder::operator<<(int i) {
>+  return AppendFormat("%d", i);
>+}
>+
>+SimpleStringBuilder& SimpleStringBuilder::operator<<(unsigned i) {
>+  return AppendFormat("%u", i);
>+}
>+
>+SimpleStringBuilder& SimpleStringBuilder::operator<<(long i) {  // NOLINT
>+  return AppendFormat("%ld", i);
>+}
>+
>+SimpleStringBuilder& SimpleStringBuilder::operator<<(long long i) {  // NOLINT
>+  return AppendFormat("%lld", i);
>+}
>+
>+SimpleStringBuilder& SimpleStringBuilder::operator<<(
>+    unsigned long i) {  // NOLINT
>+  return AppendFormat("%lu", i);
>+}
>+
>+SimpleStringBuilder& SimpleStringBuilder::operator<<(
>+    unsigned long long i) {  // NOLINT
>+  return AppendFormat("%llu", i);
>+}
>+
>+SimpleStringBuilder& SimpleStringBuilder::operator<<(float f) {
>+  return AppendFormat("%g", f);
>+}
>+
>+SimpleStringBuilder& SimpleStringBuilder::operator<<(double f) {
>+  return AppendFormat("%g", f);
>+}
>+
>+SimpleStringBuilder& SimpleStringBuilder::operator<<(long double f) {
>+  return AppendFormat("%Lg", f);
>+}
>+
>+SimpleStringBuilder& SimpleStringBuilder::AppendFormat(const char* fmt, ...) {
>+  va_list args;
>+  va_start(args, fmt);
>+  const int len =
>+      std::vsnprintf(&buffer_[size_], buffer_.size() - size_, fmt, args);
>+  if (len >= 0) {
>+    const size_t chars_added = rtc::SafeMin(len, buffer_.size() - 1 - size_);
>+    size_ += chars_added;
>+    RTC_DCHECK_EQ(len, chars_added) << "Buffer size was insufficient";
>+  } else {
>+    // This should never happen, but we're paranoid, so re-write the
>+    // terminator in case vsnprintf() overwrote it.
>+    RTC_NOTREACHED();
>+    buffer_[size_] = '\0';
>+  }
>+  va_end(args);
>+  RTC_DCHECK(IsConsistent());
>+  return *this;
>+}
>+
>+SimpleStringBuilder& SimpleStringBuilder::Append(const char* str,
>+                                                 size_t length) {
>+  const size_t chars_added =
>+      rtc::strcpyn(&buffer_[size_], buffer_.size() - size_, str, length);
>+  size_ += chars_added;
>+  RTC_DCHECK_EQ(chars_added, length == SIZE_UNKNOWN ? std::strlen(str) : length)
>+      << "Buffer size was insufficient";
>+  RTC_DCHECK(IsConsistent());
>+  return *this;
>+}
>+
>+}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/strings/string_builder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/strings/string_builder.h
>new file mode 100644
>index 00000000000..093954b3619
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/strings/string_builder.h
>@@ -0,0 +1,87 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef RTC_BASE_STRINGS_STRING_BUILDER_H_
>+#define RTC_BASE_STRINGS_STRING_BUILDER_H_
>+
>+#include <cstdio>
>+#include <cstring>
>+#include <string>
>+
>+#include "api/array_view.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/numerics/safe_minmax.h"
>+#include "rtc_base/stringutils.h"
>+
>+namespace rtc {
>+
>+// This is a minimalistic string builder class meant to cover the most cases of
>+// when you might otherwise be tempted to use a stringstream (discouraged for
>+// anything except logging). It uses a fixed-size buffer provided by the caller
>+// and concatenates strings and numbers into it, allowing the results to be
>+// read via |str()|.
>+class SimpleStringBuilder {
>+ public:
>+  explicit SimpleStringBuilder(rtc::ArrayView<char> buffer);
>+  SimpleStringBuilder(const SimpleStringBuilder&) = delete;
>+  SimpleStringBuilder& operator=(const SimpleStringBuilder&) = delete;
>+
>+  SimpleStringBuilder& operator<<(const char* str);
>+  SimpleStringBuilder& operator<<(char ch);
>+  SimpleStringBuilder& operator<<(const std::string& str);
>+  SimpleStringBuilder& operator<<(int i);
>+  SimpleStringBuilder& operator<<(unsigned i);
>+  SimpleStringBuilder& operator<<(long i);                // NOLINT
>+  SimpleStringBuilder& operator<<(long long i);           // NOLINT
>+  SimpleStringBuilder& operator<<(unsigned long i);       // NOLINT
>+  SimpleStringBuilder& operator<<(unsigned long long i);  // NOLINT
>+  SimpleStringBuilder& operator<<(float f);
>+  SimpleStringBuilder& operator<<(double f);
>+  SimpleStringBuilder& operator<<(long double f);
>+
>+  // Returns a pointer to the built string. The name |str()| is borrowed for
>+  // compatibility reasons as we replace usage of stringstream throughout the
>+  // code base.
>+  const char* str() const { return buffer_.data(); }
>+
>+  // Returns the length of the string. The name |size()| is picked for STL
>+  // compatibility reasons.
>+  size_t size() const { return size_; }
>+
>+// Allows appending a printf style formatted string.
>+#if defined(__GNUC__)
>+  __attribute__((__format__(__printf__, 2, 3)))
>+#endif
>+  SimpleStringBuilder&
>+  AppendFormat(const char* fmt, ...);
>+
>+  // An alternate way from operator<<() to append a string. This variant is
>+  // slightly more efficient when the length of the string to append, is known.
>+  SimpleStringBuilder& Append(const char* str, size_t length = SIZE_UNKNOWN);
>+
>+ private:
>+  bool IsConsistent() const {
>+    return size_ <= buffer_.size() - 1 && buffer_[size_] == '\0';
>+  }
>+
>+  // An always-zero-terminated fixed-size buffer that we write to. The fixed
>+  // size allows the buffer to be stack allocated, which helps performance.
>+  // Having a fixed size is furthermore useful to avoid unnecessary resizing
>+  // while building it.
>+  const rtc::ArrayView<char> buffer_;
>+
>+  // Represents the number of characters written to the buffer.
>+  // This does not include the terminating '\0'.
>+  size_t size_ = 0;
>+};
>+
>+}  // namespace rtc
>+
>+#endif  // RTC_BASE_STRINGS_STRING_BUILDER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/strings/string_builder_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/strings/string_builder_unittest.cc
>new file mode 100644
>index 00000000000..aa570ebbb8a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/strings/string_builder_unittest.cc
>@@ -0,0 +1,143 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "rtc_base/strings/string_builder.h"
>+
>+#include "rtc_base/checks.h"
>+#include "rtc_base/stringutils.h"
>+#include "test/gmock.h"
>+#include "test/gtest.h"
>+
>+namespace rtc {
>+
>+TEST(SimpleStringBuilder, Limit) {
>+  char sb_buf[10];
>+  SimpleStringBuilder sb(sb_buf);
>+  EXPECT_EQ(0u, strlen(sb.str()));
>+
>+  // Test that for a SSB with a buffer size of 10, that we can write 9 chars
>+  // into it.
>+  sb << "012345678";  // 9 characters + '\0'.
>+  EXPECT_EQ(0, strcmp(sb.str(), "012345678"));
>+}
>+
>+TEST(SimpleStringBuilder, NumbersAndChars) {
>+  char sb_buf[100];
>+  SimpleStringBuilder sb(sb_buf);
>+  sb << 1 << ':' << 2.1 << ":" << 2.2f << ':' << 78187493520ll << ':'
>+     << 78187493520ul;
>+  EXPECT_EQ(0, strcmp(sb.str(), "1:2.1:2.2:78187493520:78187493520"));
>+}
>+
>+TEST(SimpleStringBuilder, Format) {
>+  char sb_buf[100];
>+  SimpleStringBuilder sb(sb_buf);
>+  sb << "Here we go - ";
>+  sb.AppendFormat("This is a hex formatted value: 0x%08llx", 3735928559ULL);
>+  EXPECT_EQ(0,
>+            strcmp(sb.str(),
>+                   "Here we go - This is a hex formatted value: 0xdeadbeef"));
>+}
>+
>+TEST(SimpleStringBuilder, StdString) {
>+  char sb_buf[100];
>+  SimpleStringBuilder sb(sb_buf);
>+  std::string str = "does this work?";
>+  sb << str;
>+  EXPECT_EQ(str, sb.str());
>+}
>+
>+// These tests are safe to run if we have death test support or if DCHECKs are
>+// off.
>+#if (GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)) || !RTC_DCHECK_IS_ON
>+
>+TEST(SimpleStringBuilder, BufferOverrunConstCharP) {
>+  char sb_buf[4];
>+  SimpleStringBuilder sb(sb_buf);
>+  const char* const msg = "This is just too much";
>+#if RTC_DCHECK_IS_ON
>+  EXPECT_DEATH(sb << msg, "");
>+#else
>+  sb << msg;
>+  EXPECT_THAT(sb.str(), testing::StrEq("Thi"));
>+#endif
>+}
>+
>+TEST(SimpleStringBuilder, BufferOverrunStdString) {
>+  char sb_buf[4];
>+  SimpleStringBuilder sb(sb_buf);
>+  sb << 12;
>+  const std::string msg = "Aw, come on!";
>+#if RTC_DCHECK_IS_ON
>+  EXPECT_DEATH(sb << msg, "");
>+#else
>+  sb << msg;
>+  EXPECT_THAT(sb.str(), testing::StrEq("12A"));
>+#endif
>+}
>+
>+TEST(SimpleStringBuilder, BufferOverrunInt) {
>+  char sb_buf[4];
>+  SimpleStringBuilder sb(sb_buf);
>+  constexpr int num = -12345;
>+#if RTC_DCHECK_IS_ON
>+  EXPECT_DEATH(sb << num, "");
>+#else
>+  sb << num;
>+  // If we run into the end of the buffer, resonable results are either that
>+  // the append has no effect or that it's truncated at the point where the
>+  // buffer ends.
>+  EXPECT_THAT(sb.str(),
>+              testing::AnyOf(testing::StrEq(""), testing::StrEq("-12")));
>+#endif
>+}
>+
>+TEST(SimpleStringBuilder, BufferOverrunDouble) {
>+  char sb_buf[5];
>+  SimpleStringBuilder sb(sb_buf);
>+  constexpr double num = 123.456;
>+#if RTC_DCHECK_IS_ON
>+  EXPECT_DEATH(sb << num, "");
>+#else
>+  sb << num;
>+  EXPECT_THAT(sb.str(),
>+              testing::AnyOf(testing::StrEq(""), testing::StrEq("123.")));
>+#endif
>+}
>+
>+TEST(SimpleStringBuilder, BufferOverrunConstCharPAlreadyFull) {
>+  char sb_buf[4];
>+  SimpleStringBuilder sb(sb_buf);
>+  sb << 123;
>+  const char* const msg = "This is just too much";
>+#if RTC_DCHECK_IS_ON
>+  EXPECT_DEATH(sb << msg, "");
>+#else
>+  sb << msg;
>+  EXPECT_THAT(sb.str(), testing::StrEq("123"));
>+#endif
>+}
>+
>+TEST(SimpleStringBuilder, BufferOverrunIntAlreadyFull) {
>+  char sb_buf[4];
>+  SimpleStringBuilder sb(sb_buf);
>+  sb << "xyz";
>+  constexpr int num = -12345;
>+#if RTC_DCHECK_IS_ON
>+  EXPECT_DEATH(sb << num, "");
>+#else
>+  sb << num;
>+  EXPECT_THAT(sb.str(), testing::StrEq("xyz"));
>+#endif
>+}
>+
>+#endif
>+
>+}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringutils.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringutils.cc
>index 8671b52c5d2..bef128c2295 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringutils.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringutils.cc
>@@ -7,9 +7,11 @@
>  *  in the file PATENTS.  All contributing project authors may
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
>+#include <algorithm>
>+#include <cstdio>
> 
>-#include "rtc_base/stringutils.h"
> #include "rtc_base/checks.h"
>+#include "rtc_base/stringutils.h"
> 
> namespace rtc {
> 
>@@ -31,8 +33,8 @@ bool string_match(const char* target, const char* pattern) {
>         return true;
>       }
>       while (*target) {
>-        if ((toupper(*pattern) == toupper(*target))
>-            && string_match(target + 1, pattern + 1)) {
>+        if ((toupper(*pattern) == toupper(*target)) &&
>+            string_match(target + 1, pattern + 1)) {
>           return true;
>         }
>         ++target;
>@@ -50,25 +52,32 @@ bool string_match(const char* target, const char* pattern) {
> }
> 
> #if defined(WEBRTC_WIN)
>-int ascii_string_compare(const wchar_t* s1, const char* s2, size_t n,
>+int ascii_string_compare(const wchar_t* s1,
>+                         const char* s2,
>+                         size_t n,
>                          CharacterTransformation transformation) {
>   wchar_t c1, c2;
>   while (true) {
>-    if (n-- == 0) return 0;
>+    if (n-- == 0)
>+      return 0;
>     c1 = transformation(*s1);
>     // Double check that characters are not UTF-8
>     RTC_DCHECK_LT(*s2, 128);
>     // Note: *s2 gets implicitly promoted to wchar_t
>     c2 = transformation(*s2);
>-    if (c1 != c2) return (c1 < c2) ? -1 : 1;
>-    if (!c1) return 0;
>+    if (c1 != c2)
>+      return (c1 < c2) ? -1 : 1;
>+    if (!c1)
>+      return 0;
>     ++s1;
>     ++s2;
>   }
> }
> 
>-size_t asccpyn(wchar_t* buffer, size_t buflen,
>-               const char* source, size_t srclen) {
>+size_t asccpyn(wchar_t* buffer,
>+               size_t buflen,
>+               const char* source,
>+               size_t srclen) {
>   if (buflen <= 0)
>     return 0;
> 
>@@ -89,11 +98,11 @@ size_t asccpyn(wchar_t* buffer, size_t buflen,
> 
> #endif  // WEBRTC_WIN
> 
>-void replace_substrs(const char *search,
>+void replace_substrs(const char* search,
>                      size_t search_len,
>-                     const char *replace,
>+                     const char* replace,
>                      size_t replace_len,
>-                     std::string *s) {
>+                     std::string* s) {
>   size_t pos = 0;
>   while ((pos = s->find(search, pos, search_len)) != std::string::npos) {
>     s->replace(pos, search_len, replace, replace_len);
>@@ -101,11 +110,11 @@ void replace_substrs(const char *search,
>   }
> }
> 
>-bool starts_with(const char *s1, const char *s2) {
>+bool starts_with(const char* s1, const char* s2) {
>   return strncmp(s1, s2, strlen(s2)) == 0;
> }
> 
>-bool ends_with(const char *s1, const char *s2) {
>+bool ends_with(const char* s1, const char* s2) {
>   size_t s1_length = strlen(s1);
>   size_t s2_length = strlen(s2);
> 
>@@ -121,7 +130,7 @@ static const char kWhitespace[] = " \n\r\t";
> 
> std::string string_trim(const std::string& s) {
>   std::string::size_type first = s.find_first_not_of(kWhitespace);
>-  std::string::size_type last  = s.find_last_not_of(kWhitespace);
>+  std::string::size_type last = s.find_last_not_of(kWhitespace);
> 
>   if (first == std::string::npos || last == std::string::npos) {
>     return std::string("");
>@@ -130,4 +139,11 @@ std::string string_trim(const std::string& s) {
>   return s.substr(first, last - first + 1);
> }
> 
>+std::string ToHex(const int i) {
>+  char buffer[50];
>+  snprintf(buffer, sizeof(buffer), "%x", i);
>+
>+  return std::string(buffer);
>+}
>+
> }  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringutils.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringutils.h
>index 686402c1f12..d92ba0260f9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringutils.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringutils.h
>@@ -37,7 +37,8 @@
> // Generic string/memory utilities
> ///////////////////////////////////////////////////////////////////////////////
> 
>-#define STACK_ARRAY(TYPE, LEN) static_cast<TYPE*>(::alloca((LEN)*sizeof(TYPE)))
>+#define STACK_ARRAY(TYPE, LEN) \
>+  static_cast<TYPE*>(::alloca((LEN) * sizeof(TYPE)))
> 
> namespace rtc {
> 
>@@ -78,7 +79,7 @@ inline int _strnicmp(const char* s1, const char* s2, size_t n) {
>   return strncasecmp(s1, s2, n);
> }
> 
>-#endif // WEBRTC_POSIX
>+#endif  // WEBRTC_POSIX
> 
> ///////////////////////////////////////////////////////////////////////////////
> // Traits simplifies porting string functions to be CTYPE-agnostic
>@@ -88,12 +89,12 @@ namespace rtc {
> 
> const size_t SIZE_UNKNOWN = static_cast<size_t>(-1);
> 
>-template<class CTYPE>
>+template <class CTYPE>
> struct Traits {
>   // STL string type
>-  //typedef XXX string;
>+  // typedef XXX string;
>   // Null-terminated string
>-  //inline static const CTYPE* empty_str();
>+  // inline static const CTYPE* empty_str();
> };
> 
> ///////////////////////////////////////////////////////////////////////////////
>@@ -105,10 +106,10 @@ inline const CTYPE* nonnull(const CTYPE* str, const CTYPE* def_str = nullptr) {
>   return str ? str : (def_str ? def_str : Traits<CTYPE>::empty_str());
> }
> 
>-template<class CTYPE>
>+template <class CTYPE>
> const CTYPE* strchr(const CTYPE* str, const CTYPE* chs) {
>-  for (size_t i=0; str[i]; ++i) {
>-    for (size_t j=0; chs[j]; ++j) {
>+  for (size_t i = 0; str[i]; ++i) {
>+    for (size_t j = 0; chs[j]; ++j) {
>       if (str[i] == chs[j]) {
>         return str + i;
>       }
>@@ -117,9 +118,9 @@ const CTYPE* strchr(const CTYPE* str, const CTYPE* chs) {
>   return 0;
> }
> 
>-template<class CTYPE>
>+template <class CTYPE>
> const CTYPE* strchrn(const CTYPE* str, size_t slen, CTYPE ch) {
>-  for (size_t i=0; i<slen && str[i]; ++i) {
>+  for (size_t i = 0; i < slen && str[i]; ++i) {
>     if (str[i] == ch) {
>       return str + i;
>     }
>@@ -127,7 +128,7 @@ const CTYPE* strchrn(const CTYPE* str, size_t slen, CTYPE ch) {
>   return 0;
> }
> 
>-template<class CTYPE>
>+template <class CTYPE>
> size_t strlenn(const CTYPE* buffer, size_t buflen) {
>   size_t bufpos = 0;
>   while (buffer[bufpos] && (bufpos < buflen)) {
>@@ -139,9 +140,11 @@ size_t strlenn(const CTYPE* buffer, size_t buflen) {
> // Safe versions of strncpy, strncat, snprintf and vsnprintf that always
> // null-terminate.
> 
>-template<class CTYPE>
>-size_t strcpyn(CTYPE* buffer, size_t buflen,
>-               const CTYPE* source, size_t srclen = SIZE_UNKNOWN) {
>+template <class CTYPE>
>+size_t strcpyn(CTYPE* buffer,
>+               size_t buflen,
>+               const CTYPE* source,
>+               size_t srclen = SIZE_UNKNOWN) {
>   if (buflen <= 0)
>     return 0;
> 
>@@ -155,9 +158,11 @@ size_t strcpyn(CTYPE* buffer, size_t buflen,
>   return srclen;
> }
> 
>-template<class CTYPE>
>-size_t strcatn(CTYPE* buffer, size_t buflen,
>-               const CTYPE* source, size_t srclen = SIZE_UNKNOWN) {
>+template <class CTYPE>
>+size_t strcatn(CTYPE* buffer,
>+               size_t buflen,
>+               const CTYPE* source,
>+               size_t srclen = SIZE_UNKNOWN) {
>   if (buflen <= 0)
>     return 0;
> 
>@@ -167,8 +172,10 @@ size_t strcatn(CTYPE* buffer, size_t buflen,
> 
> // Some compilers (clang specifically) require vsprintfn be defined before
> // sprintfn.
>-template<class CTYPE>
>-size_t vsprintfn(CTYPE* buffer, size_t buflen, const CTYPE* format,
>+template <class CTYPE>
>+size_t vsprintfn(CTYPE* buffer,
>+                 size_t buflen,
>+                 const CTYPE* format,
>                  va_list args) {
>   int len = vsnprintf(buffer, buflen, format, args);
>   if ((len < 0) || (static_cast<size_t>(len) >= buflen)) {
>@@ -178,9 +185,9 @@ size_t vsprintfn(CTYPE* buffer, size_t buflen, const CTYPE* format,
>   return len;
> }
> 
>-template<class CTYPE>
>+template <class CTYPE>
> size_t sprintfn(CTYPE* buffer, size_t buflen, const CTYPE* format, ...);
>-template<class CTYPE>
>+template <class CTYPE>
> size_t sprintfn(CTYPE* buffer, size_t buflen, const CTYPE* format, ...) {
>   va_list args;
>   va_start(args, format);
>@@ -206,16 +213,22 @@ inline int ascncmp(const char* s1, const char* s2, size_t n) {
> inline int ascnicmp(const char* s1, const char* s2, size_t n) {
>   return _strnicmp(s1, s2, n);
> }
>-inline size_t asccpyn(char* buffer, size_t buflen,
>-                      const char* source, size_t srclen = SIZE_UNKNOWN) {
>+inline size_t asccpyn(char* buffer,
>+                      size_t buflen,
>+                      const char* source,
>+                      size_t srclen = SIZE_UNKNOWN) {
>   return strcpyn(buffer, buflen, source, srclen);
> }
> 
> #if defined(WEBRTC_WIN)
> 
>-typedef wchar_t(*CharacterTransformation)(wchar_t);
>-inline wchar_t identity(wchar_t c) { return c; }
>-int ascii_string_compare(const wchar_t* s1, const char* s2, size_t n,
>+typedef wchar_t (*CharacterTransformation)(wchar_t);
>+inline wchar_t identity(wchar_t c) {
>+  return c;
>+}
>+int ascii_string_compare(const wchar_t* s1,
>+                         const char* s2,
>+                         size_t n,
>                          CharacterTransformation transformation);
> 
> inline int asccmp(const wchar_t* s1, const char* s2) {
>@@ -230,8 +243,10 @@ inline int ascncmp(const wchar_t* s1, const char* s2, size_t n) {
> inline int ascnicmp(const wchar_t* s1, const char* s2, size_t n) {
>   return ascii_string_compare(s1, s2, n, tolowercase);
> }
>-size_t asccpyn(wchar_t* buffer, size_t buflen,
>-               const char* source, size_t srclen = SIZE_UNKNOWN);
>+size_t asccpyn(wchar_t* buffer,
>+               size_t buflen,
>+               const char* source,
>+               size_t srclen = SIZE_UNKNOWN);
> 
> #endif  // WEBRTC_WIN
> 
>@@ -239,7 +254,7 @@ size_t asccpyn(wchar_t* buffer, size_t buflen,
> // Traits<char> specializations
> ///////////////////////////////////////////////////////////////////////////////
> 
>-template<>
>+template <>
> struct Traits<char> {
>   typedef std::string string;
>   inline static const char* empty_str() { return ""; }
>@@ -251,7 +266,7 @@ struct Traits<char> {
> 
> #if defined(WEBRTC_WIN)
> 
>-template<>
>+template <>
> struct Traits<wchar_t> {
>   typedef std::wstring string;
>   inline static const wchar_t* empty_str() { return L""; }
>@@ -297,21 +312,23 @@ inline std::string ToUtf8(const std::wstring& wstr) {
> #endif  // WEBRTC_WIN
> 
> // Replaces all occurrences of "search" with "replace".
>-void replace_substrs(const char *search,
>+void replace_substrs(const char* search,
>                      size_t search_len,
>-                     const char *replace,
>+                     const char* replace,
>                      size_t replace_len,
>-                     std::string *s);
>+                     std::string* s);
> 
> // True iff s1 starts with s2.
>-bool starts_with(const char *s1, const char *s2);
>+bool starts_with(const char* s1, const char* s2);
> 
> // True iff s1 ends with s2.
>-bool ends_with(const char *s1, const char *s2);
>+bool ends_with(const char* s1, const char* s2);
> 
> // Remove leading and trailing whitespaces.
> std::string string_trim(const std::string& s);
> 
>+// TODO(jonasolsson): replace with absl::Hex when that becomes available.
>+std::string ToHex(const int i);
> }  // namespace rtc
> 
>-#endif // RTC_BASE_STRINGUTILS_H_
>+#endif  // RTC_BASE_STRINGUTILS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringutils_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringutils_unittest.cc
>index 85d0c3cdba4..a6e64680bb7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringutils_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringutils_unittest.cc
>@@ -16,12 +16,12 @@ namespace rtc {
> // Tests for string_match().
> 
> TEST(string_matchTest, Matches) {
>-  EXPECT_TRUE( string_match("A.B.C.D", "a.b.c.d"));
>-  EXPECT_TRUE( string_match("www.TEST.GOOGLE.COM", "www.*.com"));
>-  EXPECT_TRUE( string_match("127.0.0.1",  "12*.0.*1"));
>-  EXPECT_TRUE( string_match("127.1.0.21", "12*.0.*1"));
>-  EXPECT_FALSE(string_match("127.0.0.0",  "12*.0.*1"));
>-  EXPECT_FALSE(string_match("127.0.0.0",  "12*.0.*1"));
>+  EXPECT_TRUE(string_match("A.B.C.D", "a.b.c.d"));
>+  EXPECT_TRUE(string_match("www.TEST.GOOGLE.COM", "www.*.com"));
>+  EXPECT_TRUE(string_match("127.0.0.1", "12*.0.*1"));
>+  EXPECT_TRUE(string_match("127.1.0.21", "12*.0.*1"));
>+  EXPECT_FALSE(string_match("127.0.0.0", "12*.0.*1"));
>+  EXPECT_FALSE(string_match("127.0.0.0", "12*.0.*1"));
>   EXPECT_FALSE(string_match("127.1.1.21", "12*.0.*1"));
> }
> 
>@@ -75,7 +75,7 @@ TEST(ascii_string_compareTest, GreaterThan) {
>   EXPECT_EQ(1, ascii_string_compare(L"xyz", "xy", 5, identity));
>   EXPECT_EQ(1, ascii_string_compare(L"abc", "ABB", 5, tolowercase));
> }
>-#endif  // WEBRTC_WIN 
>+#endif  // WEBRTC_WIN
> 
> TEST(string_trim_Test, Trimming) {
>   EXPECT_EQ("temp", string_trim("\n\r\t temp \n\r\t"));
>@@ -105,4 +105,10 @@ TEST(string_endsTest, EndsWith) {
>   EXPECT_FALSE(ends_with("", "f"));
> }
> 
>-} // namespace rtc
>+TEST(string_toHexTest, ToHex) {
>+  EXPECT_EQ(ToHex(0), "0");
>+  EXPECT_EQ(ToHex(0X1243E), "1243e");
>+  EXPECT_EQ(ToHex(-20), "ffffffec");
>+}
>+
>+}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/swap_queue.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/swap_queue.h
>index f794ad91f92..172f2f5f8f1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/swap_queue.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/swap_queue.h
>@@ -18,6 +18,7 @@
> #include "rtc_base/checks.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/criticalsection.h"
>+#include "rtc_base/system/unused.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/synchronization/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/synchronization/BUILD.gn
>new file mode 100644
>index 00000000000..447be38b8ae
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/synchronization/BUILD.gn
>@@ -0,0 +1,37 @@
>+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+#
>+# Use of this source code is governed by a BSD-style license
>+# that can be found in the LICENSE file in the root of the source
>+# tree. An additional intellectual property rights grant can be found
>+# in the file PATENTS.  All contributing project authors may
>+# be found in the AUTHORS file in the root of the source tree.
>+
>+import("../../webrtc.gni")
>+if (is_android) {
>+  import("//build/config/android/config.gni")
>+  import("//build/config/android/rules.gni")
>+}
>+
>+rtc_source_set("rw_lock_wrapper") {
>+  public = [
>+    "rw_lock_wrapper.h",
>+  ]
>+  sources = [
>+    "rw_lock_wrapper.cc",
>+  ]
>+  deps = [
>+    "..:macromagic",
>+  ]
>+  if (is_win) {
>+    sources += [
>+      "rw_lock_win.cc",
>+      "rw_lock_win.h",
>+    ]
>+    deps += [ "..:logging" ]
>+  } else {
>+    sources += [
>+      "rw_lock_posix.cc",
>+      "rw_lock_posix.h",
>+    ]
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/rw_lock_posix.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/synchronization/rw_lock_posix.cc
>similarity index 95%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/rw_lock_posix.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/synchronization/rw_lock_posix.cc
>index 412873c770b..7e37dc924fd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/rw_lock_posix.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/synchronization/rw_lock_posix.cc
>@@ -8,7 +8,7 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "system_wrappers/source/rw_lock_posix.h"
>+#include "rtc_base/synchronization/rw_lock_posix.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/rw_lock_posix.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/synchronization/rw_lock_posix.h
>similarity index 76%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/rw_lock_posix.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/synchronization/rw_lock_posix.h
>index d15682b9286..a103fe77146 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/rw_lock_posix.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/synchronization/rw_lock_posix.h
>@@ -8,14 +8,13 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#ifndef SYSTEM_WRAPPERS_SOURCE_RW_LOCK_POSIX_H_
>-#define SYSTEM_WRAPPERS_SOURCE_RW_LOCK_POSIX_H_
>-
>-#include "system_wrappers/include/rw_lock_wrapper.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>+#ifndef RTC_BASE_SYNCHRONIZATION_RW_LOCK_POSIX_H_
>+#define RTC_BASE_SYNCHRONIZATION_RW_LOCK_POSIX_H_
> 
> #include <pthread.h>
> 
>+#include "rtc_base/synchronization/rw_lock_wrapper.h"
>+
> namespace webrtc {
> 
> class RWLockPosix : public RWLockWrapper {
>@@ -38,4 +37,4 @@ class RWLockPosix : public RWLockWrapper {
> 
> }  // namespace webrtc
> 
>-#endif  // SYSTEM_WRAPPERS_SOURCE_RW_LOCK_POSIX_H_
>+#endif  // RTC_BASE_SYNCHRONIZATION_RW_LOCK_POSIX_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/rw_lock_win.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/synchronization/rw_lock_win.cc
>similarity index 98%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/rw_lock_win.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/synchronization/rw_lock_win.cc
>index 23df15a2752..44cc0a75b3d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/rw_lock_win.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/synchronization/rw_lock_win.cc
>@@ -8,7 +8,7 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "system_wrappers/source/rw_lock_win.h"
>+#include "rtc_base/synchronization/rw_lock_win.h"
> 
> #include "rtc_base/logging.h"
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/rw_lock_win.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/synchronization/rw_lock_win.h
>similarity index 63%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/rw_lock_win.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/synchronization/rw_lock_win.h
>index 41537ba10ba..52ad9bbe40c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/rw_lock_win.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/synchronization/rw_lock_win.h
>@@ -8,10 +8,10 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#ifndef SYSTEM_WRAPPERS_SOURCE_RW_LOCK_WIN_H_
>-#define SYSTEM_WRAPPERS_SOURCE_RW_LOCK_WIN_H_
>+#ifndef RTC_BASE_SYNCHRONIZATION_RW_LOCK_WIN_H_
>+#define RTC_BASE_SYNCHRONIZATION_RW_LOCK_WIN_H_
> 
>-#include "system_wrappers/include/rw_lock_wrapper.h"
>+#include "rtc_base/synchronization/rw_lock_wrapper.h"
> 
> #include <Windows.h>
> 
>@@ -20,13 +20,12 @@ namespace webrtc {
> class RWLockWin : public RWLockWrapper {
>  public:
>   static RWLockWin* Create();
>-  ~RWLockWin() {}
> 
>-  virtual void AcquireLockExclusive();
>-  virtual void ReleaseLockExclusive();
>+  void AcquireLockExclusive() override;
>+  void ReleaseLockExclusive() override;
> 
>-  virtual void AcquireLockShared();
>-  virtual void ReleaseLockShared();
>+  void AcquireLockShared() override;
>+  void ReleaseLockShared() override;
> 
>  private:
>   RWLockWin();
>@@ -37,4 +36,4 @@ class RWLockWin : public RWLockWrapper {
> 
> }  // namespace webrtc
> 
>-#endif  // SYSTEM_WRAPPERS_SOURCE_RW_LOCK_WIN_H_
>+#endif  // RTC_BASE_SYNCHRONIZATION_RW_LOCK_WIN_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/rw_lock.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/synchronization/rw_lock_wrapper.cc
>similarity index 80%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/rw_lock.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/synchronization/rw_lock_wrapper.cc
>index c38c44ad759..c8cd17edb8e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/rw_lock.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/synchronization/rw_lock_wrapper.cc
>@@ -8,14 +8,14 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "system_wrappers/include/rw_lock_wrapper.h"
>+#include "rtc_base/synchronization/rw_lock_wrapper.h"
> 
> #include <assert.h>
> 
> #if defined(_WIN32)
>-#include "system_wrappers/source/rw_lock_win.h"
>+#include "rtc_base/synchronization/rw_lock_win.h"
> #else
>-#include "system_wrappers/source/rw_lock_posix.h"
>+#include "rtc_base/synchronization/rw_lock_posix.h"
> #endif
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/rw_lock_wrapper.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/synchronization/rw_lock_wrapper.h
>similarity index 83%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/rw_lock_wrapper.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/synchronization/rw_lock_wrapper.h
>index a22b6ab9b9a..39f52fca35b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/rw_lock_wrapper.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/synchronization/rw_lock_wrapper.h
>@@ -8,8 +8,8 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#ifndef SYSTEM_WRAPPERS_INCLUDE_RW_LOCK_WRAPPER_H_
>-#define SYSTEM_WRAPPERS_INCLUDE_RW_LOCK_WRAPPER_H_
>+#ifndef RTC_BASE_SYNCHRONIZATION_RW_LOCK_WRAPPER_H_
>+#define RTC_BASE_SYNCHRONIZATION_RW_LOCK_WRAPPER_H_
> 
> #include "rtc_base/thread_annotations.h"
> 
>@@ -35,7 +35,8 @@ class RTC_LOCKABLE RWLockWrapper {
> // provides more compact locking syntax.
> class RTC_SCOPED_LOCKABLE ReadLockScoped {
>  public:
>-  ReadLockScoped(RWLockWrapper& rw_lock) RTC_SHARED_LOCK_FUNCTION(rw_lock)
>+  explicit ReadLockScoped(RWLockWrapper& rw_lock)
>+      RTC_SHARED_LOCK_FUNCTION(rw_lock)
>       : rw_lock_(rw_lock) {
>     rw_lock_.AcquireLockShared();
>   }
>@@ -48,7 +49,8 @@ class RTC_SCOPED_LOCKABLE ReadLockScoped {
> 
> class RTC_SCOPED_LOCKABLE WriteLockScoped {
>  public:
>-  WriteLockScoped(RWLockWrapper& rw_lock) RTC_EXCLUSIVE_LOCK_FUNCTION(rw_lock)
>+  explicit WriteLockScoped(RWLockWrapper& rw_lock)
>+      RTC_EXCLUSIVE_LOCK_FUNCTION(rw_lock)
>       : rw_lock_(rw_lock) {
>     rw_lock_.AcquireLockExclusive();
>   }
>@@ -61,4 +63,4 @@ class RTC_SCOPED_LOCKABLE WriteLockScoped {
> 
> }  // namespace webrtc
> 
>-#endif  // SYSTEM_WRAPPERS_INCLUDE_RW_LOCK_WRAPPER_H_
>+#endif  // RTC_BASE_SYNCHRONIZATION_RW_LOCK_WRAPPER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/system/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/system/BUILD.gn
>new file mode 100644
>index 00000000000..64df84f6635
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/system/BUILD.gn
>@@ -0,0 +1,61 @@
>+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+#
>+# Use of this source code is governed by a BSD-style license
>+# that can be found in the LICENSE file in the root of the source
>+# tree. An additional intellectual property rights grant can be found
>+# in the file PATENTS.  All contributing project authors may
>+# be found in the AUTHORS file in the root of the source tree.
>+
>+import("../../webrtc.gni")
>+if (is_android) {
>+  import("//build/config/android/config.gni")
>+  import("//build/config/android/rules.gni")
>+}
>+
>+rtc_source_set("arch") {
>+  sources = [
>+    "arch.h",
>+  ]
>+}
>+
>+rtc_source_set("asm_defines") {
>+  sources = [
>+    "asm_defines.h",
>+  ]
>+}
>+
>+rtc_source_set("fallthrough") {
>+  sources = [
>+    "fallthrough.h",
>+  ]
>+}
>+
>+rtc_source_set("file_wrapper") {
>+  sources = [
>+    "file_wrapper.cc",
>+    "file_wrapper.h",
>+  ]
>+  deps = [
>+    "..:checks",
>+    "..:criticalsection",
>+    "../..:webrtc_common",
>+  ]
>+}
>+
>+rtc_source_set("ignore_warnings") {
>+  sources = [
>+    "ignore_warnings.h",
>+  ]
>+}
>+
>+rtc_source_set("inline") {
>+  sources = [
>+    "inline.h",
>+  ]
>+}
>+
>+rtc_source_set("unused") {
>+  sources = [
>+    "unused.h",
>+  ]
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/system/arch.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/system/arch.h
>new file mode 100644
>index 00000000000..a2a11806445
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/system/arch.h
>@@ -0,0 +1,58 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+// This file contains platform-specific typedefs and defines.
>+// Much of it is derived from Chromium's build/build_config.h.
>+
>+#ifndef RTC_BASE_SYSTEM_ARCH_H_
>+#define RTC_BASE_SYSTEM_ARCH_H_
>+
>+// Processor architecture detection.  For more info on what's defined, see:
>+//   http://msdn.microsoft.com/en-us/library/b0084kay.aspx
>+//   http://www.agner.org/optimize/calling_conventions.pdf
>+//   or with gcc, run: "echo | gcc -E -dM -"
>+#if defined(_M_X64) || defined(__x86_64__)
>+#define WEBRTC_ARCH_X86_FAMILY
>+#define WEBRTC_ARCH_X86_64
>+#define WEBRTC_ARCH_64_BITS
>+#define WEBRTC_ARCH_LITTLE_ENDIAN
>+#elif defined(__aarch64__)
>+#define WEBRTC_ARCH_ARM_FAMILY
>+#define WEBRTC_ARCH_64_BITS
>+#define WEBRTC_ARCH_LITTLE_ENDIAN
>+#elif defined(_M_IX86) || defined(__i386__)
>+#define WEBRTC_ARCH_X86_FAMILY
>+#define WEBRTC_ARCH_X86
>+#define WEBRTC_ARCH_32_BITS
>+#define WEBRTC_ARCH_LITTLE_ENDIAN
>+#elif defined(__ARMEL__)
>+#define WEBRTC_ARCH_ARM_FAMILY
>+#define WEBRTC_ARCH_32_BITS
>+#define WEBRTC_ARCH_LITTLE_ENDIAN
>+#elif defined(__MIPSEL__)
>+#define WEBRTC_ARCH_MIPS_FAMILY
>+#if defined(__LP64__)
>+#define WEBRTC_ARCH_64_BITS
>+#else
>+#define WEBRTC_ARCH_32_BITS
>+#endif
>+#define WEBRTC_ARCH_LITTLE_ENDIAN
>+#elif defined(__pnacl__)
>+#define WEBRTC_ARCH_32_BITS
>+#define WEBRTC_ARCH_LITTLE_ENDIAN
>+#else
>+#error Please add support for your architecture in typedefs.h
>+#endif
>+
>+#if !(defined(WEBRTC_ARCH_LITTLE_ENDIAN) ^ defined(WEBRTC_ARCH_BIG_ENDIAN))
>+#error Define either WEBRTC_ARCH_LITTLE_ENDIAN or WEBRTC_ARCH_BIG_ENDIAN
>+#endif
>+
>+#endif  // RTC_BASE_SYSTEM_ARCH_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/asm_defines.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/system/asm_defines.h
>similarity index 84%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/asm_defines.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/system/asm_defines.h
>index 7f4c80ef2c1..a7f6aad272f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/asm_defines.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/system/asm_defines.h
>@@ -8,8 +8,12 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#ifndef SYSTEM_WRAPPERS_INCLUDE_ASM_DEFINES_H_
>-#define SYSTEM_WRAPPERS_INCLUDE_ASM_DEFINES_H_
>+#ifndef RTC_BASE_SYSTEM_ASM_DEFINES_H_
>+#define RTC_BASE_SYSTEM_ASM_DEFINES_H_
>+
>+// clang-format off
>+// clang formatting breaks everything here, e.g. concatenating directives,
>+// due to absence of context via asm keyword.
> 
> #if defined(__linux__) && defined(__ELF__)
> .section .note.GNU-stack,"",%progbits
>@@ -63,4 +67,6 @@ strheq \reg1, \reg2, \num
> 
> .text
> 
>-#endif  // SYSTEM_WRAPPERS_INCLUDE_ASM_DEFINES_H_
>+// clang-format on
>+
>+#endif  // RTC_BASE_SYSTEM_ASM_DEFINES_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/system/fallthrough.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/system/fallthrough.h
>new file mode 100644
>index 00000000000..2bf0feac93a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/system/fallthrough.h
>@@ -0,0 +1,31 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef RTC_BASE_SYSTEM_FALLTHROUGH_H_
>+#define RTC_BASE_SYSTEM_FALLTHROUGH_H_
>+
>+// Macro to be used for switch-case fallthrough (required for enabling
>+// -Wimplicit-fallthrough warning on Clang).
>+
>+// This macro definition must not be included from public headers! Because
>+// clang's diagnostic checks if there's a macro expanding to
>+// [[clang::fallthrough]] defined, and if so it suggests the first macro
>+// expanding to it. So if this macro is included in a public header, clang may
>+// suggest it instead of the client's own macro, which can cause confusion.
>+
>+#ifdef __clang__
>+#define RTC_FALLTHROUGH() [[clang::fallthrough]]
>+#else
>+#define RTC_FALLTHROUGH() \
>+  do {                    \
>+  } while (0)
>+#endif
>+
>+#endif  // RTC_BASE_SYSTEM_FALLTHROUGH_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/file_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/system/file_wrapper.cc
>similarity index 98%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/file_impl.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/system/file_wrapper.cc
>index 350aaeb15bd..72f5f259dbd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/file_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/system/file_wrapper.cc
>@@ -8,7 +8,7 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "system_wrappers/include/file_wrapper.h"
>+#include "rtc_base/system/file_wrapper.h"
> 
> #ifdef _WIN32
> #include <Windows.h>
>@@ -17,6 +17,8 @@
> #include <string.h>
> #endif
> 
>+#include <utility>
>+
> #include "rtc_base/checks.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/file_wrapper.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/system/file_wrapper.h
>similarity index 78%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/file_wrapper.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/system/file_wrapper.h
>index 143da13ac4b..5411b04f699 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/file_wrapper.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/system/file_wrapper.h
>@@ -8,23 +8,21 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#ifndef SYSTEM_WRAPPERS_INCLUDE_FILE_WRAPPER_H_
>-#define SYSTEM_WRAPPERS_INCLUDE_FILE_WRAPPER_H_
>+#ifndef RTC_BASE_SYSTEM_FILE_WRAPPER_H_
>+#define RTC_BASE_SYSTEM_FILE_WRAPPER_H_
> 
> #include <stddef.h>
> #include <stdio.h>
> 
> #include "common_types.h"  // NOLINT(build/include)
> #include "rtc_base/criticalsection.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
>-// Implementation of an InStream and OutStream that can read (exclusive) or
>-// write from/to a file.
>+// Implementation that can read (exclusive) or write from/to a file.
> 
> namespace webrtc {
> 
>-// TODO(tommi): Remove the base classes, rename to rtc::File and move to base.
>-class FileWrapper : public InStream, public OutStream {
>+// TODO(tommi): Rename to rtc::File and move to base.
>+class FileWrapper final {
>  public:
>   static const size_t kMaxFileNameSize = 1024;
> 
>@@ -34,7 +32,7 @@ class FileWrapper : public InStream, public OutStream {
>   static FileWrapper Open(const char* file_name_utf8, bool read_only);
> 
>   FileWrapper(FILE* file, size_t max_size);
>-  ~FileWrapper() override;
>+  ~FileWrapper();
> 
>   // Support for move semantics.
>   FileWrapper(FileWrapper&& other);
>@@ -61,9 +59,9 @@ class FileWrapper : public InStream, public OutStream {
>   int Flush();
> 
>   // Rewinds the file to the start.
>-  int Rewind() override;
>-  int Read(void* buf, size_t length) override;
>-  bool Write(const void* buf, size_t length) override;
>+  int Rewind();
>+  int Read(void* buf, size_t length);
>+  bool Write(const void* buf, size_t length);
> 
>  private:
>   FileWrapper();
>@@ -85,4 +83,4 @@ class FileWrapper : public InStream, public OutStream {
> 
> }  // namespace webrtc
> 
>-#endif  // SYSTEM_WRAPPERS_INCLUDE_FILE_WRAPPER_H_
>+#endif  // RTC_BASE_SYSTEM_FILE_WRAPPER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/system/ignore_warnings.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/system/ignore_warnings.h
>new file mode 100644
>index 00000000000..e891c508f84
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/system/ignore_warnings.h
>@@ -0,0 +1,29 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef RTC_BASE_SYSTEM_IGNORE_WARNINGS_H_
>+#define RTC_BASE_SYSTEM_IGNORE_WARNINGS_H_
>+
>+#ifdef __clang__
>+#define RTC_PUSH_IGNORING_WFRAME_LARGER_THAN() \
>+  _Pragma("clang diagnostic push")             \
>+      _Pragma("clang diagnostic ignored \"-Wframe-larger-than=\"")
>+#define RTC_POP_IGNORING_WFRAME_LARGER_THAN() _Pragma("clang diagnostic pop")
>+#elif __GNUC__
>+#define RTC_PUSH_IGNORING_WFRAME_LARGER_THAN() \
>+  _Pragma("GCC diagnostic push")               \
>+      _Pragma("GCC diagnostic ignored \"-Wframe-larger-than=\"")
>+#define RTC_POP_IGNORING_WFRAME_LARGER_THAN() _Pragma("GCC diagnostic pop")
>+#else
>+#define RTC_PUSH_IGNORING_WFRAME_LARGER_THAN()
>+#define RTC_POP_IGNORING_WFRAME_LARGER_THAN()
>+#endif
>+
>+#endif  // RTC_BASE_SYSTEM_IGNORE_WARNINGS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/system/inline.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/system/inline.h
>new file mode 100644
>index 00000000000..f585d34ded3
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/system/inline.h
>@@ -0,0 +1,31 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef RTC_BASE_SYSTEM_INLINE_H_
>+#define RTC_BASE_SYSTEM_INLINE_H_
>+
>+#if defined(_MSC_VER)
>+
>+#define RTC_FORCE_INLINE __forceinline
>+#define RTC_NO_INLINE __declspec(noinline)
>+
>+#elif defined(__GNUC__)
>+
>+#define RTC_FORCE_INLINE __attribute__((__always_inline__))
>+#define RTC_NO_INLINE __attribute__((__noinline__))
>+
>+#else
>+
>+#define RTC_FORCE_INLINE
>+#define RTC_NO_INLINE
>+
>+#endif
>+
>+#endif  // RTC_BASE_SYSTEM_INLINE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/system/unused.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/system/unused.h
>new file mode 100644
>index 00000000000..a0add4ee292
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/system/unused.h
>@@ -0,0 +1,39 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef RTC_BASE_SYSTEM_UNUSED_H_
>+#define RTC_BASE_SYSTEM_UNUSED_H_
>+
>+// Annotate a function indicating the caller must examine the return value.
>+// Use like:
>+//   int foo() RTC_WARN_UNUSED_RESULT;
>+// To explicitly ignore a result, cast to void.
>+// TODO(kwiberg): Remove when we can use [[nodiscard]] from C++17.
>+#if defined(__clang__)
>+#define RTC_WARN_UNUSED_RESULT __attribute__((__warn_unused_result__))
>+#elif defined(__GNUC__)
>+// gcc has a __warn_unused_result__ attribute, but you can't quiet it by
>+// casting to void, so we don't use it.
>+#define RTC_WARN_UNUSED_RESULT
>+#else
>+#define RTC_WARN_UNUSED_RESULT
>+#endif
>+
>+// Prevent the compiler from warning about an unused variable. For example:
>+//   int result = DoSomething();
>+//   assert(result == 17);
>+//   RTC_UNUSED(result);
>+// Note: In most cases it is better to remove the unused variable rather than
>+// suppressing the compiler warning.
>+#ifndef RTC_UNUSED
>+#define RTC_UNUSED(x) static_cast<void>(x)
>+#endif  // RTC_UNUSED
>+
>+#endif  // RTC_BASE_SYSTEM_UNUSED_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue.h
>index 2f247e6038b..b8b307ca6a4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue.h
>@@ -11,16 +11,14 @@
> #ifndef RTC_BASE_TASK_QUEUE_H_
> #define RTC_BASE_TASK_QUEUE_H_
> 
>-#include <list>
> #include <memory>
>-#include <queue>
> #include <type_traits>
> #include <utility>
> 
>+#include "absl/memory/memory.h"
> #include "rtc_base/constructormagic.h"
>-#include "rtc_base/criticalsection.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/scoped_ref_ptr.h"
>+#include "rtc_base/thread_annotations.h"
> 
> namespace rtc {
> 
>@@ -81,13 +79,14 @@ class ClosureTaskWithCleanup : public ClosureTask<Closure> {
> // based parameters.
> template <class Closure>
> static std::unique_ptr<QueuedTask> NewClosure(Closure&& closure) {
>-  return rtc::MakeUnique<ClosureTask<Closure>>(std::forward<Closure>(closure));
>+  return absl::make_unique<ClosureTask<Closure>>(
>+      std::forward<Closure>(closure));
> }
> 
> template <class Closure, class Cleanup>
> static std::unique_ptr<QueuedTask> NewClosure(Closure&& closure,
>                                               Cleanup&& cleanup) {
>-  return rtc::MakeUnique<ClosureTaskWithCleanup<Closure, Cleanup>>(
>+  return absl::make_unique<ClosureTaskWithCleanup<Closure, Cleanup>>(
>       std::forward<Closure>(closure), std::forward<Cleanup>(cleanup));
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_for_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_for_test.cc
>new file mode 100644
>index 00000000000..5d99df59597
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_for_test.cc
>@@ -0,0 +1,19 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "rtc_base/task_queue_for_test.h"
>+
>+namespace rtc {
>+namespace test {
>+TaskQueueForTest::TaskQueueForTest(const char* queue_name, Priority priority)
>+    : TaskQueue(queue_name, priority) {}
>+TaskQueueForTest::~TaskQueueForTest() {}
>+}  // namespace test
>+}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_for_test.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_for_test.h
>new file mode 100644
>index 00000000000..70c58fb516c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_for_test.h
>@@ -0,0 +1,61 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef RTC_BASE_TASK_QUEUE_FOR_TEST_H_
>+#define RTC_BASE_TASK_QUEUE_FOR_TEST_H_
>+
>+#include <utility>
>+
>+#include "rtc_base/checks.h"
>+#include "rtc_base/event.h"
>+#include "rtc_base/task_queue.h"
>+
>+namespace rtc {
>+namespace test {
>+class RTC_LOCKABLE TaskQueueForTest : public TaskQueue {
>+ public:
>+  explicit TaskQueueForTest(const char* queue_name,
>+                            Priority priority = Priority::NORMAL);
>+  ~TaskQueueForTest();
>+
>+  // A convenience, test-only method that blocks the current thread while
>+  // a task executes on the task queue.
>+  // This variant is specifically for posting custom QueuedTask derived
>+  // implementations that tests do not want to pass ownership of over to the
>+  // task queue (i.e. the Run() method always returns |false|.).
>+  template <class Closure>
>+  void SendTask(Closure* task) {
>+    RTC_DCHECK(!IsCurrent());
>+    rtc::Event event(false, false);
>+    PostTask(rtc::NewClosure(
>+        [&task]() {
>+          RTC_CHECK_EQ(false, static_cast<QueuedTask*>(task)->Run());
>+        },
>+        [&event]() { event.Set(); }));
>+    event.Wait(rtc::Event::kForever);
>+  }
>+
>+  // A convenience, test-only method that blocks the current thread while
>+  // a task executes on the task queue.
>+  template <class Closure>
>+  void SendTask(Closure&& task) {
>+    RTC_DCHECK(!IsCurrent());
>+    rtc::Event event(false, false);
>+    PostTask(rtc::NewClosure(std::move(task), [&event]() { event.Set(); }));
>+    event.Wait(rtc::Event::kForever);
>+  }
>+
>+ private:
>+  RTC_DISALLOW_COPY_AND_ASSIGN(TaskQueueForTest);
>+};
>+}  // namespace test
>+}  // namespace rtc
>+
>+#endif  // RTC_BASE_TASK_QUEUE_FOR_TEST_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_gcd.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_gcd.cc
>index 675b8005722..f7ba4d1db80 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_gcd.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_gcd.cc
>@@ -39,7 +39,7 @@ int TaskQueuePriorityToGCD(Priority priority) {
>       return DISPATCH_QUEUE_PRIORITY_LOW;
>   }
> }
>-}
>+}  // namespace
> 
> using internal::GetQueuePtrTls;
> using internal::AutoSetCurrentQueuePtr;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_libevent.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_libevent.cc
>index 1b4014c97ac..d3b1a7c5cb1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_libevent.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_libevent.cc
>@@ -14,21 +14,17 @@
> #include <signal.h>
> #include <string.h>
> #include <unistd.h>
>+#include <list>
> 
>-#if defined(WEBRTC_LINUX)
>-#include <event2/event.h>
>-#include <event2/event_compat.h>
>-#include <event2/event_struct.h>
>-#else
> #include "base/third_party/libevent/event.h"
>-#endif
>-
> #include "rtc_base/checks.h"
>+#include "rtc_base/criticalsection.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/numerics/safe_conversions.h"
> #include "rtc_base/platform_thread.h"
> #include "rtc_base/refcount.h"
> #include "rtc_base/refcountedobject.h"
>+#include "rtc_base/system/unused.h"
> #include "rtc_base/task_queue.h"
> #include "rtc_base/task_queue_posix.h"
> #include "rtc_base/timeutils.h"
>@@ -238,7 +234,7 @@ class TaskQueue::Impl::PostAndReplyTask : public QueuedTask {
>     // PostAndReplyTask(), the reply task may or may not actually run.
>     // In either case, it will be deleted.
>     char message = kRunReplyTask;
>-    write(reply_pipe_, &message, sizeof(message));
>+    RTC_UNUSED(write(reply_pipe_, &message, sizeof(message)));
>   }
> 
>  private:
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_unittest.cc
>index d70a5fb090d..cdf0d59b046 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_unittest.cc
>@@ -21,10 +21,13 @@
> #include "rtc_base/bind.h"
> #include "rtc_base/event.h"
> #include "rtc_base/gunit.h"
>-#include "rtc_base/task_queue.h"
>+#include "rtc_base/task_queue_for_test.h"
> #include "rtc_base/timeutils.h"
> 
>+using rtc::test::TaskQueueForTest;
>+
> namespace rtc {
>+
> namespace {
> // Noop on all platforms except Windows, where it turns on high precision
> // multimedia timers which increases the precision of TimeMillis() while in
>@@ -44,9 +47,7 @@ class EnableHighResTimers {
>   const bool enabled_;
> #endif
> };
>-}
> 
>-namespace {
> void CheckCurrent(Event* signal, TaskQueue* queue) {
>   EXPECT_TRUE(queue->IsCurrent());
>   if (signal)
>@@ -76,34 +77,31 @@ TEST(TaskQueueTest, PostAndCheckCurrent) {
> 
> TEST(TaskQueueTest, PostCustomTask) {
>   static const char kQueueName[] = "PostCustomImplementation";
>-  Event event(false, false);
>-  TaskQueue queue(kQueueName);
>+  TaskQueueForTest queue(kQueueName);
> 
>   class CustomTask : public QueuedTask {
>    public:
>-    explicit CustomTask(Event* event) : event_(event) {}
>+    CustomTask() {}
>+    bool ran() const { return ran_; }
> 
>    private:
>     bool Run() override {
>-      event_->Set();
>-      return false;  // Never allows the task to be deleted by the queue.
>+      ran_ = true;
>+      return false;  // Never allow the task to be deleted by the queue.
>     }
> 
>-    Event* const event_;
>-  } my_task(&event);
>+    bool ran_ = false;
>+  } my_task;
> 
>-  // Please don't do this in production code! :)
>-  queue.PostTask(std::unique_ptr<QueuedTask>(&my_task));
>-  EXPECT_TRUE(event.Wait(1000));
>+  queue.SendTask(&my_task);
>+  EXPECT_TRUE(my_task.ran());
> }
> 
> TEST(TaskQueueTest, PostLambda) {
>-  static const char kQueueName[] = "PostLambda";
>-  Event event(false, false);
>-  TaskQueue queue(kQueueName);
>-
>-  queue.PostTask([&event]() { event.Set(); });
>-  EXPECT_TRUE(event.Wait(1000));
>+  TaskQueueForTest queue("PostLambda");
>+  bool ran = false;
>+  queue.SendTask([&ran]() { ran = true; });
>+  EXPECT_TRUE(ran);
> }
> 
> TEST(TaskQueueTest, PostDelayedZero) {
>@@ -168,8 +166,7 @@ TEST(TaskQueueTest, PostMultipleDelayed) {
>   std::vector<std::unique_ptr<Event>> events;
>   for (int i = 0; i < 100; ++i) {
>     events.push_back(std::unique_ptr<Event>(new Event(false, false)));
>-    queue.PostDelayedTask(
>-        Bind(&CheckCurrent, events.back().get(), &queue), i);
>+    queue.PostDelayedTask(Bind(&CheckCurrent, events.back().get(), &queue), i);
>   }
> 
>   for (const auto& e : events)
>@@ -193,9 +190,9 @@ TEST(TaskQueueTest, PostAndReply) {
>   TaskQueue post_queue(kPostQueue);
>   TaskQueue reply_queue(kReplyQueue);
> 
>-  post_queue.PostTaskAndReply(
>-      Bind(&CheckCurrent, nullptr, &post_queue),
>-      Bind(&CheckCurrent, &event, &reply_queue), &reply_queue);
>+  post_queue.PostTaskAndReply(Bind(&CheckCurrent, nullptr, &post_queue),
>+                              Bind(&CheckCurrent, &event, &reply_queue),
>+                              &reply_queue);
>   EXPECT_TRUE(event.Wait(1000));
> }
> 
>@@ -374,12 +371,10 @@ TEST(TaskQueueTest, PostAndReplyDeadlock) {
>   EXPECT_TRUE(event.Wait(1000));
> }
> 
>-void TestPostTaskAndReply(TaskQueue* work_queue,
>-                          Event* event) {
>+void TestPostTaskAndReply(TaskQueue* work_queue, Event* event) {
>   ASSERT_FALSE(work_queue->IsCurrent());
>-  work_queue->PostTaskAndReply(
>-      Bind(&CheckCurrent, nullptr, work_queue),
>-      NewClosure([event]() { event->Set(); }));
>+  work_queue->PostTaskAndReply(Bind(&CheckCurrent, nullptr, work_queue),
>+                               NewClosure([event]() { event->Set(); }));
> }
> 
> // Does a PostTaskAndReply from within a task to post and reply to the current
>@@ -391,8 +386,7 @@ TEST(TaskQueueTest, PostAndReply2) {
>   TaskQueue queue(kQueueName);
>   TaskQueue work_queue(kWorkQueueName);
> 
>-  queue.PostTask(
>-      Bind(&TestPostTaskAndReply, &work_queue, &event));
>+  queue.PostTask(Bind(&TestPostTaskAndReply, &work_queue, &event));
>   EXPECT_TRUE(event.Wait(1000));
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_win.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_win.cc
>index 8a626e92997..c1e7c4621c8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_win.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_win.cc
>@@ -10,7 +10,17 @@
> 
> #include "rtc_base/task_queue.h"
> 
>-#include <mmsystem.h>
>+// clang-format off
>+// clang formating would change include order.
>+
>+// Include winsock2.h before including <windows.h> to maintain consistency with
>+// win32.h. To include win32.h directly, it must be broken out into its own
>+// build target.
>+#include <winsock2.h>
>+#include <windows.h>
>+#include <sal.h>       // Must come after windows headers.
>+#include <mmsystem.h>  // Must come after windows headers.
>+// clang-format on
> #include <string.h>
> 
> #include <algorithm>
>@@ -19,6 +29,7 @@
> 
> #include "rtc_base/arraysize.h"
> #include "rtc_base/checks.h"
>+#include "rtc_base/criticalsection.h"
> #include "rtc_base/event.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/numerics/safe_conversions.h"
>@@ -352,7 +363,7 @@ void TaskQueue::Impl::ThreadMain(void* context) {
> }
> 
> void TaskQueue::Impl::ThreadState::RunThreadMain() {
>-  HANDLE handles[2] = { *timer_.event_for_wait(), in_queue_ };
>+  HANDLE handles[2] = {*timer_.event_for_wait(), in_queue_};
>   while (true) {
>     // Make sure we do an alertable wait as that's required to allow APCs to run
>     // (e.g. required for InitializeQueueThread and stopping the thread in
>@@ -366,8 +377,9 @@ void TaskQueue::Impl::ThreadState::RunThreadMain() {
>         break;
>     }
> 
>-    if (result == WAIT_OBJECT_0 || (!timer_tasks_.empty() &&
>-        ::WaitForSingleObject(*timer_.event_for_wait(), 0) == WAIT_OBJECT_0)) {
>+    if (result == WAIT_OBJECT_0 ||
>+        (!timer_tasks_.empty() &&
>+         ::WaitForSingleObject(*timer_.event_for_wait(), 0) == WAIT_OBJECT_0)) {
>       // The multimedia timer was signaled.
>       timer_.Cancel();
>       RunDueTasks();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/template_util.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/template_util.h
>index 04e5e37fc05..3c04a860d8f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/template_util.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/template_util.h
>@@ -19,34 +19,47 @@ namespace rtc {
> 
> // Template definitions from tr1.
> 
>-template<class T, T v>
>+template <class T, T v>
> struct integral_constant {
>   static const T value = v;
>   typedef T value_type;
>   typedef integral_constant<T, v> type;
> };
> 
>-template <class T, T v> const T integral_constant<T, v>::value;
>+template <class T, T v>
>+const T integral_constant<T, v>::value;
> 
> typedef integral_constant<bool, true> true_type;
> typedef integral_constant<bool, false> false_type;
> 
>-template <class T> struct is_pointer : false_type {};
>-template <class T> struct is_pointer<T*> : true_type {};
>-
>-template <class T, class U> struct is_same : public false_type {};
>-template <class T> struct is_same<T, T> : true_type {};
>-
>-template<class> struct is_array : public false_type {};
>-template<class T, size_t n> struct is_array<T[n]> : public true_type {};
>-template<class T> struct is_array<T[]> : public true_type {};
>-
>-template <class T> struct is_non_const_reference : false_type {};
>-template <class T> struct is_non_const_reference<T&> : true_type {};
>-template <class T> struct is_non_const_reference<const T&> : false_type {};
>-
>-template <class T> struct is_void : false_type {};
>-template <> struct is_void<void> : true_type {};
>+template <class T>
>+struct is_pointer : false_type {};
>+template <class T>
>+struct is_pointer<T*> : true_type {};
>+
>+template <class T, class U>
>+struct is_same : public false_type {};
>+template <class T>
>+struct is_same<T, T> : true_type {};
>+
>+template <class>
>+struct is_array : public false_type {};
>+template <class T, size_t n>
>+struct is_array<T[n]> : public true_type {};
>+template <class T>
>+struct is_array<T[]> : public true_type {};
>+
>+template <class T>
>+struct is_non_const_reference : false_type {};
>+template <class T>
>+struct is_non_const_reference<T&> : true_type {};
>+template <class T>
>+struct is_non_const_reference<const T&> : false_type {};
>+
>+template <class T>
>+struct is_void : false_type {};
>+template <>
>+struct is_void<void> : true_type {};
> 
> // Helper useful for converting a tuple to variadic template function
> // arguments.
>@@ -95,7 +108,7 @@ struct ConvertHelper {
> // is_class type_trait implementation.
> struct IsClassHelper {
>   template <typename C>
>-  static YesType Test(void(C::*)(void));
>+  static YesType Test(void (C::*)(void));
> 
>   template <typename C>
>   static NoType Test(...);
>@@ -111,16 +124,14 @@ template <typename From, typename To>
> struct is_convertible
>     : integral_constant<bool,
>                         sizeof(internal::ConvertHelper::Test<To>(
>-                                   internal::ConvertHelper::Create<From>())) ==
>-                        sizeof(internal::YesType)> {
>-};
>+                            internal::ConvertHelper::Create<From>())) ==
>+                            sizeof(internal::YesType)> {};
> 
> template <typename T>
> struct is_class
>     : integral_constant<bool,
>                         sizeof(internal::IsClassHelper::Test<T>(0)) ==
>-                            sizeof(internal::YesType)> {
>-};
>+                            sizeof(internal::YesType)> {};
> 
> }  // namespace rtc
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/testbase64.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/testbase64.h
>index dbbdec9fe43..5dc94b6cbc6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/testbase64.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/testbase64.h
>@@ -14,7 +14,2098 @@
> /* This file was generated by googleclient/talk/binary2header.sh */
> 
> static unsigned char testbase64[] = {
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>+    0xac, 0x7a, 0x2b, 0xea, 0x75, 0xef, 0x92, 0x0c, 0x53, 0x3d, 0x2f, 0xd4,
>+    0xfa, 0xbb, 0xfa, 0x74, 0xf5, 0x39, 0x9a, 0xd7, 0xe7, 0x80, 0x53, 0x79,
>+    0xba, 0x5b, 0xfe, 0x97, 0xfa, 0x4b, 0xfc, 0xba, 0x7f, 0xb1, 0xc7, 0xab,
>+    0x1e, 0x8f, 0xff, 0xd9};
> 
> #endif  // RTC_BASE_TESTBASE64_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/testcertificateverifier.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/testcertificateverifier.h
>new file mode 100644
>index 00000000000..8ad6e4d736c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/testcertificateverifier.h
>@@ -0,0 +1,34 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef RTC_BASE_TESTCERTIFICATEVERIFIER_H_
>+#define RTC_BASE_TESTCERTIFICATEVERIFIER_H_
>+
>+#include "rtc_base/sslcertificate.h"
>+
>+namespace rtc {
>+
>+class TestCertificateVerifier : public SSLCertificateVerifier {
>+ public:
>+  TestCertificateVerifier() = default;
>+  ~TestCertificateVerifier() override = default;
>+
>+  bool Verify(const SSLCertificate& certificate) override {
>+    call_count_++;
>+    return verify_certificate_;
>+  }
>+
>+  size_t call_count_ = 0;
>+  bool verify_certificate_ = true;
>+};
>+
>+}  // namespace rtc
>+
>+#endif  // RTC_BASE_TESTCERTIFICATEVERIFIER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/testclient.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/testclient.cc
>index 585db77785f..7c151c75101 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/testclient.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/testclient.cc
>@@ -10,8 +10,8 @@
> 
> #include "rtc_base/testclient.h"
> 
>+#include "absl/memory/memory.h"
> #include "rtc_base/gunit.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/thread.h"
> #include "rtc_base/timeutils.h"
> 
>@@ -49,7 +49,8 @@ int TestClient::Send(const char* buf, size_t size) {
>   return socket_->Send(buf, size, options);
> }
> 
>-int TestClient::SendTo(const char* buf, size_t size,
>+int TestClient::SendTo(const char* buf,
>+                       size_t size,
>                        const SocketAddress& dest) {
>   rtc::PacketOptions options;
>   return socket_->SendTo(buf, size, dest, options);
>@@ -89,7 +90,8 @@ std::unique_ptr<TestClient::Packet> TestClient::NextPacket(int timeout_ms) {
>   return packet;
> }
> 
>-bool TestClient::CheckNextPacket(const char* buf, size_t size,
>+bool TestClient::CheckNextPacket(const char* buf,
>+                                 size_t size,
>                                  SocketAddress* addr) {
>   bool res = false;
>   std::unique_ptr<Packet> packet = NextPacket(kTimeoutMs);
>@@ -138,11 +140,14 @@ int TestClient::SetOption(Socket::Option opt, int value) {
>   return socket_->SetOption(opt, value);
> }
> 
>-void TestClient::OnPacket(AsyncPacketSocket* socket, const char* buf,
>-                          size_t size, const SocketAddress& remote_addr,
>+void TestClient::OnPacket(AsyncPacketSocket* socket,
>+                          const char* buf,
>+                          size_t size,
>+                          const SocketAddress& remote_addr,
>                           const PacketTime& packet_time) {
>   CritScope cs(&crit_);
>-  packets_.push_back(MakeUnique<Packet>(remote_addr, buf, size, packet_time));
>+  packets_.push_back(
>+      absl::make_unique<Packet>(remote_addr, buf, size, packet_time));
> }
> 
> void TestClient::OnReadyToSend(AsyncPacketSocket* socket) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/testclient.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/testclient.h
>index c0dbe65bc8d..a0d98b30d16 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/testclient.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/testclient.h
>@@ -34,7 +34,7 @@ class TestClient : public sigslot::has_slots<> {
>     virtual ~Packet();
> 
>     SocketAddress addr;
>-    char*  buf;
>+    char* buf;
>     size_t size;
>     PacketTime packet_time;
>   };
>@@ -93,7 +93,9 @@ class TestClient : public sigslot::has_slots<> {
>   // Workaround for the fact that AsyncPacketSocket::GetConnState doesn't exist.
>   Socket::ConnState GetState();
>   // Slot for packets read on the socket.
>-  void OnPacket(AsyncPacketSocket* socket, const char* buf, size_t len,
>+  void OnPacket(AsyncPacketSocket* socket,
>+                const char* buf,
>+                size_t len,
>                 const SocketAddress& remote_addr,
>                 const PacketTime& packet_time);
>   void OnReadyToSend(AsyncPacketSocket* socket);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/testclient_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/testclient_unittest.cc
>index 1d1d3f21e98..b3a6c805638 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/testclient_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/testclient_unittest.cc
>@@ -9,10 +9,10 @@
>  */
> 
> #include "rtc_base/testclient.h"
>+#include "absl/memory/memory.h"
> #include "rtc_base/gunit.h"
> #include "rtc_base/nethelpers.h"
> #include "rtc_base/physicalsocketserver.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/testechoserver.h"
> #include "rtc_base/thread.h"
> 
>@@ -31,12 +31,12 @@ using namespace rtc;
>   }
> 
> void TestUdpInternal(const SocketAddress& loopback) {
>-  Thread *main = Thread::Current();
>-  AsyncSocket* socket = main->socketserver()
>-      ->CreateAsyncSocket(loopback.family(), SOCK_DGRAM);
>+  Thread* main = Thread::Current();
>+  AsyncSocket* socket =
>+      main->socketserver()->CreateAsyncSocket(loopback.family(), SOCK_DGRAM);
>   socket->Bind(loopback);
> 
>-  TestClient client(MakeUnique<AsyncUDPSocket>(socket));
>+  TestClient client(absl::make_unique<AsyncUDPSocket>(socket));
>   SocketAddress addr = client.address(), from;
>   EXPECT_EQ(3, client.SendTo("foo", 3, addr));
>   EXPECT_TRUE(client.CheckNextPacket("foo", 3, &from));
>@@ -45,13 +45,13 @@ void TestUdpInternal(const SocketAddress& loopback) {
> }
> 
> void TestTcpInternal(const SocketAddress& loopback) {
>-  Thread *main = Thread::Current();
>+  Thread* main = Thread::Current();
>   TestEchoServer server(main, loopback);
> 
>-  AsyncSocket* socket = main->socketserver()
>-      ->CreateAsyncSocket(loopback.family(), SOCK_STREAM);
>-  std::unique_ptr<AsyncTCPSocket> tcp_socket =
>-      WrapUnique(AsyncTCPSocket::Create(socket, loopback, server.address()));
>+  AsyncSocket* socket =
>+      main->socketserver()->CreateAsyncSocket(loopback.family(), SOCK_STREAM);
>+  std::unique_ptr<AsyncTCPSocket> tcp_socket = absl::WrapUnique(
>+      AsyncTCPSocket::Create(socket, loopback, server.address()));
>   ASSERT_TRUE(tcp_socket != nullptr);
> 
>   TestClient client(std::move(tcp_socket));
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/testechoserver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/testechoserver.h
>index 672dda0f18d..5e714eb63dd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/testechoserver.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/testechoserver.h
>@@ -40,7 +40,9 @@ class TestEchoServer : public sigslot::has_slots<> {
>       client_sockets_.push_back(packet_socket);
>     }
>   }
>-  void OnPacket(AsyncPacketSocket* socket, const char* buf, size_t size,
>+  void OnPacket(AsyncPacketSocket* socket,
>+                const char* buf,
>+                size_t size,
>                 const SocketAddress& remote_addr,
>                 const PacketTime& packet_time) {
>     rtc::PacketOptions options;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/testutils.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/testutils.h
>index b0003847cb9..2ab4a35956a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/testutils.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/testutils.h
>@@ -37,8 +37,8 @@ using namespace rtc;
> // event.
> 
> enum StreamSinkEvent {
>-  SSE_OPEN  = SE_OPEN,
>-  SSE_READ  = SE_READ,
>+  SSE_OPEN = SE_OPEN,
>+  SSE_READ = SE_READ,
>   SSE_WRITE = SE_WRITE,
>   SSE_CLOSE = SE_CLOSE,
>   SSE_ERROR = 16
>@@ -50,15 +50,17 @@ class StreamSink : public sigslot::has_slots<> {
>   ~StreamSink() override;
> 
>   void Monitor(StreamInterface* stream) {
>-   stream->SignalEvent.connect(this, &StreamSink::OnEvent);
>-   events_.erase(stream);
>+    stream->SignalEvent.connect(this, &StreamSink::OnEvent);
>+    events_.erase(stream);
>   }
>   void Unmonitor(StreamInterface* stream) {
>-   stream->SignalEvent.disconnect(this);
>-   // In case you forgot to unmonitor a previous object with this address
>-   events_.erase(stream);
>+    stream->SignalEvent.disconnect(this);
>+    // In case you forgot to unmonitor a previous object with this address
>+    events_.erase(stream);
>   }
>-  bool Check(StreamInterface* stream, StreamSinkEvent event, bool reset = true) {
>+  bool Check(StreamInterface* stream,
>+             StreamSinkEvent event,
>+             bool reset = true) {
>     return DoCheck(stream, event, reset);
>   }
>   int Events(StreamInterface* stream, bool reset = true) {
>@@ -66,19 +68,19 @@ class StreamSink : public sigslot::has_slots<> {
>   }
> 
>   void Monitor(AsyncSocket* socket) {
>-   socket->SignalConnectEvent.connect(this, &StreamSink::OnConnectEvent);
>-   socket->SignalReadEvent.connect(this, &StreamSink::OnReadEvent);
>-   socket->SignalWriteEvent.connect(this, &StreamSink::OnWriteEvent);
>-   socket->SignalCloseEvent.connect(this, &StreamSink::OnCloseEvent);
>-   // In case you forgot to unmonitor a previous object with this address
>-   events_.erase(socket);
>+    socket->SignalConnectEvent.connect(this, &StreamSink::OnConnectEvent);
>+    socket->SignalReadEvent.connect(this, &StreamSink::OnReadEvent);
>+    socket->SignalWriteEvent.connect(this, &StreamSink::OnWriteEvent);
>+    socket->SignalCloseEvent.connect(this, &StreamSink::OnCloseEvent);
>+    // In case you forgot to unmonitor a previous object with this address
>+    events_.erase(socket);
>   }
>   void Unmonitor(AsyncSocket* socket) {
>-   socket->SignalConnectEvent.disconnect(this);
>-   socket->SignalReadEvent.disconnect(this);
>-   socket->SignalWriteEvent.disconnect(this);
>-   socket->SignalCloseEvent.disconnect(this);
>-   events_.erase(socket);
>+    socket->SignalConnectEvent.disconnect(this);
>+    socket->SignalReadEvent.disconnect(this);
>+    socket->SignalWriteEvent.disconnect(this);
>+    socket->SignalCloseEvent.disconnect(this);
>+    events_.erase(socket);
>   }
>   bool Check(AsyncSocket* socket, StreamSinkEvent event, bool reset = true) {
>     return DoCheck(socket, event, reset);
>@@ -88,7 +90,7 @@ class StreamSink : public sigslot::has_slots<> {
>   }
> 
>  private:
>-  typedef std::map<void*,int> EventMap;
>+  typedef std::map<void*, int> EventMap;
> 
>   void OnEvent(StreamInterface* stream, int events, int error) {
>     if (error) {
>@@ -96,15 +98,9 @@ class StreamSink : public sigslot::has_slots<> {
>     }
>     AddEvents(stream, events);
>   }
>-  void OnConnectEvent(AsyncSocket* socket) {
>-    AddEvents(socket, SSE_OPEN);
>-  }
>-  void OnReadEvent(AsyncSocket* socket) {
>-    AddEvents(socket, SSE_READ);
>-  }
>-  void OnWriteEvent(AsyncSocket* socket) {
>-    AddEvents(socket, SSE_WRITE);
>-  }
>+  void OnConnectEvent(AsyncSocket* socket) { AddEvents(socket, SSE_OPEN); }
>+  void OnReadEvent(AsyncSocket* socket) { AddEvents(socket, SSE_READ); }
>+  void OnWriteEvent(AsyncSocket* socket) { AddEvents(socket, SSE_WRITE); }
>   void OnCloseEvent(AsyncSocket* socket, int error) {
>     AddEvents(socket, (0 == error) ? SSE_CLOSE : SSE_ERROR);
>   }
>@@ -147,25 +143,24 @@ class StreamSink : public sigslot::has_slots<> {
> ///////////////////////////////////////////////////////////////////////////////
> 
> class StreamSource : public StreamInterface {
>-public:
>- StreamSource();
>- ~StreamSource() override;
>+ public:
>+  StreamSource();
>+  ~StreamSource() override;
> 
>- void Clear() {
>-   readable_data_.clear();
>-   written_data_.clear();
>-   state_ = SS_CLOSED;
>-   read_block_ = 0;
>-   write_block_ = SIZE_UNKNOWN;
>-  }
>-  void QueueString(const char* data) {
>-    QueueData(data, strlen(data));
>+  void Clear() {
>+    readable_data_.clear();
>+    written_data_.clear();
>+    state_ = SS_CLOSED;
>+    read_block_ = 0;
>+    write_block_ = SIZE_UNKNOWN;
>   }
>+  void QueueString(const char* data) { QueueData(data, strlen(data)); }
> #if defined(__GNUC__)
>   // Note: Implicit |this| argument counts as the first argument.
>   __attribute__((__format__(__printf__, 2, 3)))
> #endif
>-  void QueueStringF(const char* format, ...) {
>+  void
>+  QueueStringF(const char* format, ...) {
>     va_list args;
>     va_start(args, format);
>     char buffer[1024];
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/third_party/base64/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/third_party/base64/BUILD.gn
>new file mode 100644
>index 00000000000..dca47e02b7e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/third_party/base64/BUILD.gn
>@@ -0,0 +1,17 @@
>+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+#
>+# Use of this source code is governed by a BSD-style license
>+# that can be found in the ../../../LICENSE file in the root of the source
>+# tree. An additional intellectual property rights grant can be found
>+# in the file PATENTS.  All contributing project authors may
>+# be found in the AUTHORS file in the root of the source tree.
>+
>+import("../../../webrtc.gni")
>+
>+rtc_source_set("base64") {
>+  visibility = [ "*" ]
>+  sources = [
>+    "base64.cc",
>+    "base64.h",
>+  ]
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/third_party/base64/LICENSE b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/third_party/base64/LICENSE
>new file mode 100644
>index 00000000000..4ad21d1b717
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/third_party/base64/LICENSE
>@@ -0,0 +1,14 @@
>+//*********************************************************************
>+//* Base64 - a simple base64 encoder and decoder.
>+//*
>+//*     Copyright (c) 1999, Bob Withers - bwit@pobox.com
>+//*
>+//* This code may be freely used for any purpose, either personal
>+//* or commercial, provided the authors copyright notice remains
>+//* intact.
>+//*
>+//* Enhancements by Stanley Yamane:
>+//*     o reverse lookup table for the decode function
>+//*     o reserve string buffer space in advance
>+//*
>+//*********************************************************************
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/third_party/base64/README.chromium b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/third_party/base64/README.chromium
>new file mode 100644
>index 00000000000..92ba0d36d7f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/third_party/base64/README.chromium
>@@ -0,0 +1,11 @@
>+Name: A simple base64 encoder and decoder
>+Short Name: base64
>+URL:
>+Version: 0
>+Date: 2018-06-20
>+License: Custom license
>+License File: LICENSE
>+Security Critical: yes
>+
>+Description:
>+A simple base64 encoder and decoder
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/base64.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/third_party/base64/base64.cc
>similarity index 96%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/base64.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/third_party/base64/base64.cc
>index 0ccec0e68c7..53ff6b9d543 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/base64.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/third_party/base64/base64.cc
>@@ -14,12 +14,11 @@
> //*
> //*********************************************************************
> 
>-#include "rtc_base/base64.h"
>+#include "rtc_base/third_party/base64/base64.h"
> 
>+#include <assert.h>
> #include <string.h>
> 
>-#include "rtc_base/checks.h"
>-
> using std::vector;
> 
> namespace rtc {
>@@ -96,7 +95,7 @@ bool Base64::IsBase64Encoded(const std::string& str) {
> void Base64::EncodeFromArray(const void* data,
>                              size_t len,
>                              std::string* result) {
>-  RTC_DCHECK(nullptr != result);
>+  assert(nullptr != result);
>   result->clear();
>   result->resize(((len + 2) / 3) * 4);
>   const unsigned char* byte_data = static_cast<const unsigned char*>(data);
>@@ -224,15 +223,15 @@ bool Base64::DecodeFromArrayTemplate(const char* data,
>                                      DecodeFlags flags,
>                                      T* result,
>                                      size_t* data_used) {
>-  RTC_DCHECK(nullptr != result);
>-  RTC_DCHECK(flags <= (DO_PARSE_MASK | DO_PAD_MASK | DO_TERM_MASK));
>+  assert(nullptr != result);
>+  assert(flags <= (DO_PARSE_MASK | DO_PAD_MASK | DO_TERM_MASK));
> 
>   const DecodeFlags parse_flags = flags & DO_PARSE_MASK;
>   const DecodeFlags pad_flags = flags & DO_PAD_MASK;
>   const DecodeFlags term_flags = flags & DO_TERM_MASK;
>-  RTC_DCHECK(0 != parse_flags);
>-  RTC_DCHECK(0 != pad_flags);
>-  RTC_DCHECK(0 != term_flags);
>+  assert(0 != parse_flags);
>+  assert(0 != pad_flags);
>+  assert(0 != term_flags);
> 
>   result->clear();
>   result->reserve(len);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/base64.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/third_party/base64/base64.h
>similarity index 96%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/base64.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/third_party/base64/base64.h
>index bfe2fe6a947..bbc93c8e736 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/base64.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/third_party/base64/base64.h
>@@ -9,8 +9,8 @@
> //* intact.
> //*********************************************************************
> 
>-#ifndef RTC_BASE_BASE64_H_
>-#define RTC_BASE_BASE64_H_
>+#ifndef RTC_BASE_THIRD_PARTY_BASE64_BASE64_H_
>+#define RTC_BASE_THIRD_PARTY_BASE64_BASE64_H_
> 
> #include <string>
> #include <vector>
>@@ -120,4 +120,4 @@ class Base64 {
> 
> }  // namespace rtc
> 
>-#endif  // RTC_BASE_BASE64_H_
>+#endif  /* RTC_BASE_THIRD_PARTY_BASE64_BASE64_H_ */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/third_party/sigslot/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/third_party/sigslot/BUILD.gn
>new file mode 100644
>index 00000000000..aea748e1bdd
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/third_party/sigslot/BUILD.gn
>@@ -0,0 +1,17 @@
>+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+#
>+# Use of this source code is governed by a BSD-style license
>+# that can be found in the ../../../LICENSE file in the root of the source
>+# tree. An additional intellectual property rights grant can be found
>+# in the file PATENTS.  All contributing project authors may
>+# be found in the AUTHORS file in the root of the source tree.
>+
>+import("../../../webrtc.gni")
>+
>+rtc_source_set("sigslot") {
>+  visibility = [ "*" ]
>+  sources = [
>+    "sigslot.cc",
>+    "sigslot.h",
>+  ]
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/third_party/sigslot/LICENSE b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/third_party/sigslot/LICENSE
>new file mode 100644
>index 00000000000..d1c66c7335f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/third_party/sigslot/LICENSE
>@@ -0,0 +1,7 @@
>+// sigslot.h: Signal/Slot classes
>+//
>+// Written by Sarah Thompson (sarah@telergy.com) 2002.
>+//
>+// License: Public domain. You are free to use this code however you like, with
>+// the proviso that the author takes on no responsibility or liability for any
>+// use.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/third_party/sigslot/README.chromium b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/third_party/sigslot/README.chromium
>new file mode 100644
>index 00000000000..e0575cf16ad
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/third_party/sigslot/README.chromium
>@@ -0,0 +1,20 @@
>+Name: C++ Signal/Slot Library
>+Short Name: sigslot
>+URL: http://sigslot.sourceforge.net/
>+Version: 0
>+Date: 2018-07-09
>+License: Custom license
>+License File: LICENSE
>+Security Critical: yes
>+
>+Description:
>+C++ Signal/Slot Library
>+
>+This file has been modified such that has_slots and signalx do not have to be
>+using the same threading requirements. E.g. it is possible to connect a
>+has_slots<single_threaded> and signal0<multi_threaded_local> or
>+has_slots<multi_threaded_local> and signal0<single_threaded>.
>+If has_slots is single threaded the user must ensure that it is not trying
>+to connect or disconnect to signalx concurrently or data race may occur.
>+If signalx is single threaded the user must ensure that disconnect, connect
>+or signal is not happening concurrently or data race may occur.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sigslot.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/third_party/sigslot/sigslot.cc
>similarity index 91%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sigslot.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/third_party/sigslot/sigslot.cc
>index 9d792cbbcd6..d852a2b5367 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sigslot.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/third_party/sigslot/sigslot.cc
>@@ -6,7 +6,7 @@
> // the proviso that the author takes on no responsibility or liability for any
> // use.
> 
>-#include "rtc_base/sigslot.h"
>+#include "rtc_base/third_party/sigslot/sigslot.h"
> 
> namespace sigslot {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/third_party/sigslot/sigslot.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/third_party/sigslot/sigslot.h
>new file mode 100644
>index 00000000000..c77e4e675df
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/third_party/sigslot/sigslot.h
>@@ -0,0 +1,644 @@
>+// sigslot.h: Signal/Slot classes
>+//
>+// Written by Sarah Thompson (sarah@telergy.com) 2002.
>+//
>+// License: Public domain. You are free to use this code however you like, with
>+// the proviso that the author takes on no responsibility or liability for any
>+// use.
>+//
>+// QUICK DOCUMENTATION
>+//
>+//        (see also the full documentation at http://sigslot.sourceforge.net/)
>+//
>+//    #define switches
>+//      SIGSLOT_PURE_ISO:
>+//        Define this to force ISO C++ compliance. This also disables all of
>+//        the thread safety support on platforms where it is available.
>+//
>+//      SIGSLOT_USE_POSIX_THREADS:
>+//        Force use of Posix threads when using a C++ compiler other than gcc
>+//        on a platform that supports Posix threads. (When using gcc, this is
>+//        the default - use SIGSLOT_PURE_ISO to disable this if necessary)
>+//
>+//      SIGSLOT_DEFAULT_MT_POLICY:
>+//        Where thread support is enabled, this defaults to
>+//        multi_threaded_global. Otherwise, the default is single_threaded.
>+//        #define this yourself to override the default. In pure ISO mode,
>+//        anything other than single_threaded will cause a compiler error.
>+//
>+//    PLATFORM NOTES
>+//
>+//      Win32:
>+//        On Win32, the WEBRTC_WIN symbol must be #defined. Most mainstream
>+//        compilers do this by default, but you may need to define it yourself
>+//        if your build environment is less standard. This causes the Win32
>+//        thread support to be compiled in and used automatically.
>+//
>+//      Unix/Linux/BSD, etc.:
>+//        If you're using gcc, it is assumed that you have Posix threads
>+//        available, so they are used automatically. You can override this (as
>+//        under Windows) with the SIGSLOT_PURE_ISO switch. If you're using
>+//        something other than gcc but still want to use Posix threads, you
>+//        need to #define SIGSLOT_USE_POSIX_THREADS.
>+//
>+//      ISO C++:
>+//        If none of the supported platforms are detected, or if
>+//        SIGSLOT_PURE_ISO is defined, all multithreading support is turned
>+//        off, along with any code that might cause a pure ISO C++ environment
>+//        to complain. Before you ask, gcc -ansi -pedantic won't compile this
>+//        library, but gcc -ansi is fine. Pedantic mode seems to throw a lot of
>+//        errors that aren't really there. If you feel like investigating this,
>+//        please contact the author.
>+//
>+//
>+//    THREADING MODES
>+//
>+//      single_threaded:
>+//        Your program is assumed to be single threaded from the point of view
>+//        of signal/slot usage (i.e. all objects using signals and slots are
>+//        created and destroyed from a single thread). Behaviour if objects are
>+//        destroyed concurrently is undefined (i.e. you'll get the occasional
>+//        segmentation fault/memory exception).
>+//
>+//      multi_threaded_global:
>+//        Your program is assumed to be multi threaded. Objects using signals
>+//        and slots can be safely created and destroyed from any thread, even
>+//        when connections exist. In multi_threaded_global mode, this is
>+//        achieved by a single global mutex (actually a critical section on
>+//        Windows because they are faster). This option uses less OS resources,
>+//        but results in more opportunities for contention, possibly resulting
>+//        in more context switches than are strictly necessary.
>+//
>+//      multi_threaded_local:
>+//        Behaviour in this mode is essentially the same as
>+//        multi_threaded_global, except that each signal, and each object that
>+//        inherits has_slots, all have their own mutex/critical section. In
>+//        practice, this means that mutex collisions (and hence context
>+//        switches) only happen if they are absolutely essential. However, on
>+//        some platforms, creating a lot of mutexes can slow down the whole OS,
>+//        so use this option with care.
>+//
>+//    USING THE LIBRARY
>+//
>+//      See the full documentation at http://sigslot.sourceforge.net/
>+//
>+// Libjingle specific:
>+//
>+// This file has been modified such that has_slots and signalx do not have to be
>+// using the same threading requirements. E.g. it is possible to connect a
>+// has_slots<single_threaded> and signal0<multi_threaded_local> or
>+// has_slots<multi_threaded_local> and signal0<single_threaded>.
>+// If has_slots is single threaded the user must ensure that it is not trying
>+// to connect or disconnect to signalx concurrently or data race may occur.
>+// If signalx is single threaded the user must ensure that disconnect, connect
>+// or signal is not happening concurrently or data race may occur.
>+
>+#ifndef RTC_BASE_THIRD_PARTY_SIGSLOT_SIGSLOT_H_
>+#define RTC_BASE_THIRD_PARTY_SIGSLOT_SIGSLOT_H_
>+
>+#include <stdlib.h>
>+#include <cstring>
>+#include <list>
>+#include <set>
>+
>+// On our copy of sigslot.h, we set single threading as default.
>+#define SIGSLOT_DEFAULT_MT_POLICY single_threaded
>+
>+#if defined(SIGSLOT_PURE_ISO) ||                   \
>+    (!defined(WEBRTC_WIN) && !defined(__GNUG__) && \
>+     !defined(SIGSLOT_USE_POSIX_THREADS))
>+#define _SIGSLOT_SINGLE_THREADED
>+#elif defined(WEBRTC_WIN)
>+#define _SIGSLOT_HAS_WIN32_THREADS
>+#include "windows.h"
>+#elif defined(__GNUG__) || defined(SIGSLOT_USE_POSIX_THREADS)
>+#define _SIGSLOT_HAS_POSIX_THREADS
>+#include <pthread.h>
>+#else
>+#define _SIGSLOT_SINGLE_THREADED
>+#endif
>+
>+#ifndef SIGSLOT_DEFAULT_MT_POLICY
>+#ifdef _SIGSLOT_SINGLE_THREADED
>+#define SIGSLOT_DEFAULT_MT_POLICY single_threaded
>+#else
>+#define SIGSLOT_DEFAULT_MT_POLICY multi_threaded_local
>+#endif
>+#endif
>+
>+// TODO: change this namespace to rtc?
>+namespace sigslot {
>+
>+class single_threaded {
>+ public:
>+  void lock() {}
>+  void unlock() {}
>+};
>+
>+#ifdef _SIGSLOT_HAS_WIN32_THREADS
>+// The multi threading policies only get compiled in if they are enabled.
>+class multi_threaded_global {
>+ public:
>+  multi_threaded_global() {
>+    static bool isinitialised = false;
>+
>+    if (!isinitialised) {
>+      InitializeCriticalSection(get_critsec());
>+      isinitialised = true;
>+    }
>+  }
>+
>+  void lock() { EnterCriticalSection(get_critsec()); }
>+
>+  void unlock() { LeaveCriticalSection(get_critsec()); }
>+
>+ private:
>+  CRITICAL_SECTION* get_critsec() {
>+    static CRITICAL_SECTION g_critsec;
>+    return &g_critsec;
>+  }
>+};
>+
>+class multi_threaded_local {
>+ public:
>+  multi_threaded_local() { InitializeCriticalSection(&m_critsec); }
>+
>+  multi_threaded_local(const multi_threaded_local&) {
>+    InitializeCriticalSection(&m_critsec);
>+  }
>+
>+  ~multi_threaded_local() { DeleteCriticalSection(&m_critsec); }
>+
>+  void lock() { EnterCriticalSection(&m_critsec); }
>+
>+  void unlock() { LeaveCriticalSection(&m_critsec); }
>+
>+ private:
>+  CRITICAL_SECTION m_critsec;
>+};
>+#endif  // _SIGSLOT_HAS_WIN32_THREADS
>+
>+#ifdef _SIGSLOT_HAS_POSIX_THREADS
>+// The multi threading policies only get compiled in if they are enabled.
>+class multi_threaded_global {
>+ public:
>+  void lock() { pthread_mutex_lock(get_mutex()); }
>+  void unlock() { pthread_mutex_unlock(get_mutex()); }
>+
>+ private:
>+  static pthread_mutex_t* get_mutex();
>+};
>+
>+class multi_threaded_local {
>+ public:
>+  multi_threaded_local() { pthread_mutex_init(&m_mutex, nullptr); }
>+  multi_threaded_local(const multi_threaded_local&) {
>+    pthread_mutex_init(&m_mutex, nullptr);
>+  }
>+  ~multi_threaded_local() { pthread_mutex_destroy(&m_mutex); }
>+  void lock() { pthread_mutex_lock(&m_mutex); }
>+  void unlock() { pthread_mutex_unlock(&m_mutex); }
>+
>+ private:
>+  pthread_mutex_t m_mutex;
>+};
>+#endif  // _SIGSLOT_HAS_POSIX_THREADS
>+
>+template <class mt_policy>
>+class lock_block {
>+ public:
>+  mt_policy* m_mutex;
>+
>+  lock_block(mt_policy* mtx) : m_mutex(mtx) { m_mutex->lock(); }
>+
>+  ~lock_block() { m_mutex->unlock(); }
>+};
>+
>+class _signal_base_interface;
>+
>+class has_slots_interface {
>+ private:
>+  typedef void (*signal_connect_t)(has_slots_interface* self,
>+                                   _signal_base_interface* sender);
>+  typedef void (*signal_disconnect_t)(has_slots_interface* self,
>+                                      _signal_base_interface* sender);
>+  typedef void (*disconnect_all_t)(has_slots_interface* self);
>+
>+  const signal_connect_t m_signal_connect;
>+  const signal_disconnect_t m_signal_disconnect;
>+  const disconnect_all_t m_disconnect_all;
>+
>+ protected:
>+  has_slots_interface(signal_connect_t conn,
>+                      signal_disconnect_t disc,
>+                      disconnect_all_t disc_all)
>+      : m_signal_connect(conn),
>+        m_signal_disconnect(disc),
>+        m_disconnect_all(disc_all) {}
>+
>+  // Doesn't really need to be virtual, but is for backwards compatibility
>+  // (it was virtual in a previous version of sigslot).
>+  virtual ~has_slots_interface() {}
>+
>+ public:
>+  void signal_connect(_signal_base_interface* sender) {
>+    m_signal_connect(this, sender);
>+  }
>+
>+  void signal_disconnect(_signal_base_interface* sender) {
>+    m_signal_disconnect(this, sender);
>+  }
>+
>+  void disconnect_all() { m_disconnect_all(this); }
>+};
>+
>+class _signal_base_interface {
>+ private:
>+  typedef void (*slot_disconnect_t)(_signal_base_interface* self,
>+                                    has_slots_interface* pslot);
>+  typedef void (*slot_duplicate_t)(_signal_base_interface* self,
>+                                   const has_slots_interface* poldslot,
>+                                   has_slots_interface* pnewslot);
>+
>+  const slot_disconnect_t m_slot_disconnect;
>+  const slot_duplicate_t m_slot_duplicate;
>+
>+ protected:
>+  _signal_base_interface(slot_disconnect_t disc, slot_duplicate_t dupl)
>+      : m_slot_disconnect(disc), m_slot_duplicate(dupl) {}
>+
>+  ~_signal_base_interface() {}
>+
>+ public:
>+  void slot_disconnect(has_slots_interface* pslot) {
>+    m_slot_disconnect(this, pslot);
>+  }
>+
>+  void slot_duplicate(const has_slots_interface* poldslot,
>+                      has_slots_interface* pnewslot) {
>+    m_slot_duplicate(this, poldslot, pnewslot);
>+  }
>+};
>+
>+class _opaque_connection {
>+ private:
>+  typedef void (*emit_t)(const _opaque_connection*);
>+  template <typename FromT, typename ToT>
>+  union union_caster {
>+    FromT from;
>+    ToT to;
>+  };
>+
>+  emit_t pemit;
>+  has_slots_interface* pdest;
>+  // Pointers to member functions may be up to 16 bytes for virtual classes,
>+  // so make sure we have enough space to store it.
>+  unsigned char pmethod[16];
>+
>+ public:
>+  template <typename DestT, typename... Args>
>+  _opaque_connection(DestT* pd, void (DestT::*pm)(Args...)) : pdest(pd) {
>+    typedef void (DestT::*pm_t)(Args...);
>+    static_assert(sizeof(pm_t) <= sizeof(pmethod),
>+                  "Size of slot function pointer too large.");
>+
>+    std::memcpy(pmethod, &pm, sizeof(pm_t));
>+
>+    typedef void (*em_t)(const _opaque_connection* self, Args...);
>+    union_caster<em_t, emit_t> caster2;
>+    caster2.from = &_opaque_connection::emitter<DestT, Args...>;
>+    pemit = caster2.to;
>+  }
>+
>+  has_slots_interface* getdest() const { return pdest; }
>+
>+  _opaque_connection duplicate(has_slots_interface* newtarget) const {
>+    _opaque_connection res = *this;
>+    res.pdest = newtarget;
>+    return res;
>+  }
>+
>+  // Just calls the stored "emitter" function pointer stored at construction
>+  // time.
>+  template <typename... Args>
>+  void emit(Args... args) const {
>+    typedef void (*em_t)(const _opaque_connection*, Args...);
>+    union_caster<emit_t, em_t> caster;
>+    caster.from = pemit;
>+    (caster.to)(this, args...);
>+  }
>+
>+ private:
>+  template <typename DestT, typename... Args>
>+  static void emitter(const _opaque_connection* self, Args... args) {
>+    typedef void (DestT::*pm_t)(Args...);
>+    pm_t pm;
>+    std::memcpy(&pm, self->pmethod, sizeof(pm_t));
>+    (static_cast<DestT*>(self->pdest)->*(pm))(args...);
>+  }
>+};
>+
>+template <class mt_policy>
>+class _signal_base : public _signal_base_interface, public mt_policy {
>+ protected:
>+  typedef std::list<_opaque_connection> connections_list;
>+
>+  _signal_base()
>+      : _signal_base_interface(&_signal_base::do_slot_disconnect,
>+                               &_signal_base::do_slot_duplicate),
>+        m_current_iterator(m_connected_slots.end()) {}
>+
>+  ~_signal_base() { disconnect_all(); }
>+
>+ private:
>+  _signal_base& operator=(_signal_base const& that);
>+
>+ public:
>+  _signal_base(const _signal_base& o)
>+      : _signal_base_interface(&_signal_base::do_slot_disconnect,
>+                               &_signal_base::do_slot_duplicate),
>+        m_current_iterator(m_connected_slots.end()) {
>+    lock_block<mt_policy> lock(this);
>+    for (const auto& connection : o.m_connected_slots) {
>+      connection.getdest()->signal_connect(this);
>+      m_connected_slots.push_back(connection);
>+    }
>+  }
>+
>+  bool is_empty() {
>+    lock_block<mt_policy> lock(this);
>+    return m_connected_slots.empty();
>+  }
>+
>+  void disconnect_all() {
>+    lock_block<mt_policy> lock(this);
>+
>+    while (!m_connected_slots.empty()) {
>+      has_slots_interface* pdest = m_connected_slots.front().getdest();
>+      m_connected_slots.pop_front();
>+      pdest->signal_disconnect(static_cast<_signal_base_interface*>(this));
>+    }
>+    // If disconnect_all is called while the signal is firing, advance the
>+    // current slot iterator to the end to avoid an invalidated iterator from
>+    // being dereferenced.
>+    m_current_iterator = m_connected_slots.end();
>+  }
>+
>+#if !defined(NDEBUG)
>+  bool connected(has_slots_interface* pclass) {
>+    lock_block<mt_policy> lock(this);
>+    connections_list::const_iterator it = m_connected_slots.begin();
>+    connections_list::const_iterator itEnd = m_connected_slots.end();
>+    while (it != itEnd) {
>+      if (it->getdest() == pclass)
>+        return true;
>+      ++it;
>+    }
>+    return false;
>+  }
>+#endif
>+
>+  void disconnect(has_slots_interface* pclass) {
>+    lock_block<mt_policy> lock(this);
>+    connections_list::iterator it = m_connected_slots.begin();
>+    connections_list::iterator itEnd = m_connected_slots.end();
>+
>+    while (it != itEnd) {
>+      if (it->getdest() == pclass) {
>+        // If we're currently using this iterator because the signal is firing,
>+        // advance it to avoid it being invalidated.
>+        if (m_current_iterator == it) {
>+          m_current_iterator = m_connected_slots.erase(it);
>+        } else {
>+          m_connected_slots.erase(it);
>+        }
>+        pclass->signal_disconnect(static_cast<_signal_base_interface*>(this));
>+        return;
>+      }
>+      ++it;
>+    }
>+  }
>+
>+ private:
>+  static void do_slot_disconnect(_signal_base_interface* p,
>+                                 has_slots_interface* pslot) {
>+    _signal_base* const self = static_cast<_signal_base*>(p);
>+    lock_block<mt_policy> lock(self);
>+    connections_list::iterator it = self->m_connected_slots.begin();
>+    connections_list::iterator itEnd = self->m_connected_slots.end();
>+
>+    while (it != itEnd) {
>+      connections_list::iterator itNext = it;
>+      ++itNext;
>+
>+      if (it->getdest() == pslot) {
>+        // If we're currently using this iterator because the signal is firing,
>+        // advance it to avoid it being invalidated.
>+        if (self->m_current_iterator == it) {
>+          self->m_current_iterator = self->m_connected_slots.erase(it);
>+        } else {
>+          self->m_connected_slots.erase(it);
>+        }
>+      }
>+
>+      it = itNext;
>+    }
>+  }
>+
>+  static void do_slot_duplicate(_signal_base_interface* p,
>+                                const has_slots_interface* oldtarget,
>+                                has_slots_interface* newtarget) {
>+    _signal_base* const self = static_cast<_signal_base*>(p);
>+    lock_block<mt_policy> lock(self);
>+    connections_list::iterator it = self->m_connected_slots.begin();
>+    connections_list::iterator itEnd = self->m_connected_slots.end();
>+
>+    while (it != itEnd) {
>+      if (it->getdest() == oldtarget) {
>+        self->m_connected_slots.push_back(it->duplicate(newtarget));
>+      }
>+
>+      ++it;
>+    }
>+  }
>+
>+ protected:
>+  connections_list m_connected_slots;
>+
>+  // Used to handle a slot being disconnected while a signal is
>+  // firing (iterating m_connected_slots).
>+  connections_list::iterator m_current_iterator;
>+  bool m_erase_current_iterator = false;
>+};
>+
>+template <class mt_policy = SIGSLOT_DEFAULT_MT_POLICY>
>+class has_slots : public has_slots_interface, public mt_policy {
>+ private:
>+  typedef std::set<_signal_base_interface*> sender_set;
>+  typedef sender_set::const_iterator const_iterator;
>+
>+ public:
>+  has_slots()
>+      : has_slots_interface(&has_slots::do_signal_connect,
>+                            &has_slots::do_signal_disconnect,
>+                            &has_slots::do_disconnect_all) {}
>+
>+  has_slots(has_slots const& o)
>+      : has_slots_interface(&has_slots::do_signal_connect,
>+                            &has_slots::do_signal_disconnect,
>+                            &has_slots::do_disconnect_all) {
>+    lock_block<mt_policy> lock(this);
>+    for (auto* sender : o.m_senders) {
>+      sender->slot_duplicate(&o, this);
>+      m_senders.insert(sender);
>+    }
>+  }
>+
>+  ~has_slots() { this->disconnect_all(); }
>+
>+ private:
>+  has_slots& operator=(has_slots const&);
>+
>+  static void do_signal_connect(has_slots_interface* p,
>+                                _signal_base_interface* sender) {
>+    has_slots* const self = static_cast<has_slots*>(p);
>+    lock_block<mt_policy> lock(self);
>+    self->m_senders.insert(sender);
>+  }
>+
>+  static void do_signal_disconnect(has_slots_interface* p,
>+                                   _signal_base_interface* sender) {
>+    has_slots* const self = static_cast<has_slots*>(p);
>+    lock_block<mt_policy> lock(self);
>+    self->m_senders.erase(sender);
>+  }
>+
>+  static void do_disconnect_all(has_slots_interface* p) {
>+    has_slots* const self = static_cast<has_slots*>(p);
>+    lock_block<mt_policy> lock(self);
>+    while (!self->m_senders.empty()) {
>+      std::set<_signal_base_interface*> senders;
>+      senders.swap(self->m_senders);
>+      const_iterator it = senders.begin();
>+      const_iterator itEnd = senders.end();
>+
>+      while (it != itEnd) {
>+        _signal_base_interface* s = *it;
>+        ++it;
>+        s->slot_disconnect(p);
>+      }
>+    }
>+  }
>+
>+ private:
>+  sender_set m_senders;
>+};
>+
>+template <class mt_policy, typename... Args>
>+class signal_with_thread_policy : public _signal_base<mt_policy> {
>+ private:
>+  typedef _signal_base<mt_policy> base;
>+
>+ protected:
>+  typedef typename base::connections_list connections_list;
>+
>+ public:
>+  signal_with_thread_policy() {}
>+
>+  template <class desttype>
>+  void connect(desttype* pclass, void (desttype::*pmemfun)(Args...)) {
>+    lock_block<mt_policy> lock(this);
>+    this->m_connected_slots.push_back(_opaque_connection(pclass, pmemfun));
>+    pclass->signal_connect(static_cast<_signal_base_interface*>(this));
>+  }
>+
>+  void emit(Args... args) {
>+    lock_block<mt_policy> lock(this);
>+    this->m_current_iterator = this->m_connected_slots.begin();
>+    while (this->m_current_iterator != this->m_connected_slots.end()) {
>+      _opaque_connection const& conn = *this->m_current_iterator;
>+      ++(this->m_current_iterator);
>+      conn.emit<Args...>(args...);
>+    }
>+  }
>+
>+  void operator()(Args... args) { emit(args...); }
>+};
>+
>+// Alias with default thread policy. Needed because both default arguments
>+// and variadic template arguments must go at the end of the list, so we
>+// can't have both at once.
>+template <typename... Args>
>+using signal = signal_with_thread_policy<SIGSLOT_DEFAULT_MT_POLICY, Args...>;
>+
>+// The previous verion of sigslot didn't use variadic templates, so you would
>+// need to write "sigslot::signal2<Arg1, Arg2>", for example.
>+// Now you can just write "sigslot::signal<Arg1, Arg2>", but these aliases
>+// exist for backwards compatibility.
>+template <typename mt_policy = SIGSLOT_DEFAULT_MT_POLICY>
>+using signal0 = signal_with_thread_policy<mt_policy>;
>+
>+template <typename A1, typename mt_policy = SIGSLOT_DEFAULT_MT_POLICY>
>+using signal1 = signal_with_thread_policy<mt_policy, A1>;
>+
>+template <typename A1,
>+          typename A2,
>+          typename mt_policy = SIGSLOT_DEFAULT_MT_POLICY>
>+using signal2 = signal_with_thread_policy<mt_policy, A1, A2>;
>+
>+template <typename A1,
>+          typename A2,
>+          typename A3,
>+          typename mt_policy = SIGSLOT_DEFAULT_MT_POLICY>
>+using signal3 = signal_with_thread_policy<mt_policy, A1, A2, A3>;
>+
>+template <typename A1,
>+          typename A2,
>+          typename A3,
>+          typename A4,
>+          typename mt_policy = SIGSLOT_DEFAULT_MT_POLICY>
>+using signal4 = signal_with_thread_policy<mt_policy, A1, A2, A3, A4>;
>+
>+template <typename A1,
>+          typename A2,
>+          typename A3,
>+          typename A4,
>+          typename A5,
>+          typename mt_policy = SIGSLOT_DEFAULT_MT_POLICY>
>+using signal5 = signal_with_thread_policy<mt_policy, A1, A2, A3, A4, A5>;
>+
>+template <typename A1,
>+          typename A2,
>+          typename A3,
>+          typename A4,
>+          typename A5,
>+          typename A6,
>+          typename mt_policy = SIGSLOT_DEFAULT_MT_POLICY>
>+using signal6 = signal_with_thread_policy<mt_policy, A1, A2, A3, A4, A5, A6>;
>+
>+template <typename A1,
>+          typename A2,
>+          typename A3,
>+          typename A4,
>+          typename A5,
>+          typename A6,
>+          typename A7,
>+          typename mt_policy = SIGSLOT_DEFAULT_MT_POLICY>
>+using signal7 =
>+    signal_with_thread_policy<mt_policy, A1, A2, A3, A4, A5, A6, A7>;
>+
>+template <typename A1,
>+          typename A2,
>+          typename A3,
>+          typename A4,
>+          typename A5,
>+          typename A6,
>+          typename A7,
>+          typename A8,
>+          typename mt_policy = SIGSLOT_DEFAULT_MT_POLICY>
>+using signal8 =
>+    signal_with_thread_policy<mt_policy, A1, A2, A3, A4, A5, A6, A7, A8>;
>+
>+}  // namespace sigslot
>+
>+#endif  /* RTC_BASE_THIRD_PARTY_SIGSLOT_SIGSLOT_H_ */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/thread.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/thread.cc
>index 749334125f7..373fa3952de 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/thread.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/thread.cc
>@@ -18,6 +18,12 @@
> #error "Either WEBRTC_WIN or WEBRTC_POSIX needs to be defined."
> #endif
> 
>+#if defined(WEBRTC_WIN)
>+// Disable warning that we don't care about:
>+// warning C4722: destructor never returns, potential memory leak
>+#pragma warning(disable : 4722)
>+#endif
>+
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/nullsocketserver.h"
>@@ -29,8 +35,8 @@
> namespace rtc {
> 
> ThreadManager* ThreadManager::Instance() {
>-  RTC_DEFINE_STATIC_LOCAL(ThreadManager, thread_manager, ());
>-  return &thread_manager;
>+  static ThreadManager* const thread_manager = new ThreadManager();
>+  return thread_manager;
> }
> 
> ThreadManager::~ThreadManager() {
>@@ -61,8 +67,8 @@ ThreadManager::ThreadManager() : main_thread_ref_(CurrentThreadRef()) {
> }
> #endif
> 
>-Thread *ThreadManager::CurrentThread() {
>-  return static_cast<Thread *>(pthread_getspecific(key_));
>+Thread* ThreadManager::CurrentThread() {
>+  return static_cast<Thread*>(pthread_getspecific(key_));
> }
> 
> void ThreadManager::SetCurrentThread(Thread* thread) {
>@@ -77,20 +83,19 @@ void ThreadManager::SetCurrentThread(Thread* thread) {
> 
> #if defined(WEBRTC_WIN)
> ThreadManager::ThreadManager()
>-    : key_(TlsAlloc()), main_thread_ref_(CurrentThreadRef()) {
>-}
>+    : key_(TlsAlloc()), main_thread_ref_(CurrentThreadRef()) {}
> 
>-Thread *ThreadManager::CurrentThread() {
>-  return static_cast<Thread *>(TlsGetValue(key_));
>+Thread* ThreadManager::CurrentThread() {
>+  return static_cast<Thread*>(TlsGetValue(key_));
> }
> 
>-void ThreadManager::SetCurrentThread(Thread *thread) {
>+void ThreadManager::SetCurrentThread(Thread* thread) {
>   RTC_DCHECK(!CurrentThread() || !thread);
>   TlsSetValue(key_, thread);
> }
> #endif
> 
>-Thread *ThreadManager::WrapCurrentThread() {
>+Thread* ThreadManager::WrapCurrentThread() {
>   Thread* result = CurrentThread();
>   if (nullptr == result) {
>     result = new Thread(SocketServer::CreateDefault());
>@@ -112,9 +117,8 @@ bool ThreadManager::IsMainThread() {
> }
> 
> Thread::ScopedDisallowBlockingCalls::ScopedDisallowBlockingCalls()
>-  : thread_(Thread::Current()),
>-    previous_state_(thread_->SetAllowBlockingCalls(false)) {
>-}
>+    : thread_(Thread::Current()),
>+      previous_state_(thread_->SetAllowBlockingCalls(false)) {}
> 
> Thread::ScopedDisallowBlockingCalls::~ScopedDisallowBlockingCalls() {
>   RTC_DCHECK(thread_->IsCurrent());
>@@ -124,15 +128,25 @@ Thread::ScopedDisallowBlockingCalls::~ScopedDisallowBlockingCalls() {
> // DEPRECATED.
> Thread::Thread() : Thread(SocketServer::CreateDefault()) {}
> 
>-Thread::Thread(SocketServer* ss) : MessageQueue(ss, false) {
>+Thread::Thread(SocketServer* ss) : Thread(ss, /*do_init=*/true) {}
>+
>+Thread::Thread(std::unique_ptr<SocketServer> ss)
>+    : Thread(std::move(ss), /*do_init=*/true) {}
>+
>+Thread::Thread(SocketServer* ss, bool do_init)
>+    : MessageQueue(ss, /*do_init=*/false) {
>   SetName("Thread", this);  // default name
>-  DoInit();
>+  if (do_init) {
>+    DoInit();
>+  }
> }
> 
>-Thread::Thread(std::unique_ptr<SocketServer> ss)
>+Thread::Thread(std::unique_ptr<SocketServer> ss, bool do_init)
>     : MessageQueue(std::move(ss), false) {
>   SetName("Thread", this);  // default name
>-  DoInit();
>+  if (do_init) {
>+    DoInit();
>+  }
> }
> 
> Thread::~Thread() {
>@@ -347,7 +361,7 @@ void Thread::Send(const Location& posted_from,
>   AssertBlockingIsAllowedOnCurrentThread();
> 
>   AutoThread thread;
>-  Thread *current_thread = Thread::Current();
>+  Thread* current_thread = Thread::Current();
>   RTC_DCHECK(current_thread != nullptr);  // AutoThread ensures this
> 
>   bool ready = false;
>@@ -408,7 +422,7 @@ void Thread::ReceiveSendsFromThread(const Thread* source) {
>   while (PopSendMessageFromThread(source, &smsg)) {
>     crit_.Leave();
> 
>-    smsg.msg.phandler->OnMessage(&smsg.msg);
>+    Dispatch(&smsg.msg);
> 
>     crit_.Enter();
>     *smsg.ready = true;
>@@ -524,7 +538,9 @@ bool Thread::IsRunning() {
> #endif
> }
> 
>-AutoThread::AutoThread() : Thread(SocketServer::CreateDefault()) {
>+AutoThread::AutoThread()
>+    : Thread(SocketServer::CreateDefault(), /*do_init=*/false) {
>+  DoInit();
>   if (!ThreadManager::Instance()->CurrentThread()) {
>     ThreadManager::Instance()->SetCurrentThread(this);
>   }
>@@ -539,7 +555,8 @@ AutoThread::~AutoThread() {
> }
> 
> AutoSocketServerThread::AutoSocketServerThread(SocketServer* ss)
>-    : Thread(ss) {
>+    : Thread(ss, /*do_init=*/false) {
>+  DoInit();
>   old_thread_ = ThreadManager::Instance()->CurrentThread();
>   // Temporarily set the current thread to nullptr so that we can keep checks
>   // around that catch unintentional pointer overwrites.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/thread.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/thread.h
>index ab6c7b1b501..0408a0d4d09 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/thread.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/thread.h
>@@ -56,7 +56,7 @@ class ThreadManager {
>   // shame to break it.  It is also conceivable on Win32 that we won't even
>   // be able to get synchronization privileges, in which case the result
>   // will have a null handle.
>-  Thread *WrapCurrentThread();
>+  Thread* WrapCurrentThread();
>   void UnwrapCurrentThread();
> 
>   bool IsMainThread();
>@@ -81,9 +81,9 @@ class ThreadManager {
> 
> struct _SendMessage {
>   _SendMessage() {}
>-  Thread *thread;
>+  Thread* thread;
>   Message msg;
>-  bool *ready;
>+  bool* ready;
> };
> 
> class Runnable {
>@@ -111,6 +111,11 @@ class RTC_LOCKABLE Thread : public MessageQueue {
> 
>   explicit Thread(SocketServer* ss);
>   explicit Thread(std::unique_ptr<SocketServer> ss);
>+  // Constructors meant for subclasses; they should call DoInit themselves and
>+  // pass false for |do_init|, so that DoInit is called only on the fully
>+  // instantiated class, which avoids a vptr data race.
>+  Thread(SocketServer* ss, bool do_init);
>+  Thread(std::unique_ptr<SocketServer> ss, bool do_init);
> 
>   // NOTE: ALL SUBCLASSES OF Thread MUST CALL Stop() IN THEIR DESTRUCTORS (or
>   // guarantee Stop() is explicitly called before the subclass is destroyed).
>@@ -130,6 +135,7 @@ class RTC_LOCKABLE Thread : public MessageQueue {
>    public:
>     ScopedDisallowBlockingCalls();
>     ~ScopedDisallowBlockingCalls();
>+
>    private:
>     Thread* const thread_;
>     const bool previous_state_;
>@@ -244,7 +250,7 @@ class RTC_LOCKABLE Thread : public MessageQueue {
> #if defined(WEBRTC_WIN)
>   static DWORD WINAPI PreRun(LPVOID context);
> #else
>-  static void *PreRun(void *pv);
>+  static void* PreRun(void* pv);
> #endif
> 
>   // ThreadManager calls this instead WrapCurrent() because
>@@ -273,8 +279,8 @@ class RTC_LOCKABLE Thread : public MessageQueue {
>   std::list<_SendMessage> sendlist_;
>   std::string name_;
> 
>-  // TODO(tommi): Add thread checks for proper use of control methods.
>-  // Ideally we should be able to just use PlatformThread.
>+// TODO(tommi): Add thread checks for proper use of control methods.
>+// Ideally we should be able to just use PlatformThread.
> 
> #if defined(WEBRTC_POSIX)
>   pthread_t thread_ = 0;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/thread_annotations_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/thread_annotations_unittest.cc
>index d8a4af1ac35..949f042afa3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/thread_annotations_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/thread_annotations_unittest.cc
>@@ -32,13 +32,9 @@ class RTC_SCOPED_LOCKABLE ScopeLock {
> 
> class ThreadSafe {
>  public:
>-  ThreadSafe() {
>-    pt_protected_by_lock_ = new int;
>-  }
>+  ThreadSafe() { pt_protected_by_lock_ = new int; }
> 
>-  ~ThreadSafe() {
>-    delete pt_protected_by_lock_;
>-  }
>+  ~ThreadSafe() { delete pt_protected_by_lock_; }
> 
>   void LockInOrder() {
>     beforelock_.EnterWrite();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/thread_checker.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/thread_checker.h
>index 028e9bf367e..ad0450871a1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/thread_checker.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/thread_checker.h
>@@ -32,9 +32,7 @@ namespace rtc {
> // right version for your build configuration.
> class ThreadCheckerDoNothing {
>  public:
>-  bool CalledOnValidThread() const {
>-    return true;
>-  }
>+  bool CalledOnValidThread() const { return true; }
> 
>   void DetachFromThread() {}
> };
>@@ -86,7 +84,7 @@ class RTC_SCOPED_LOCKABLE AnnounceOnThread {
>       RTC_EXCLUSIVE_LOCK_FUNCTION(thread_like_object) {}
>   ~AnnounceOnThread() RTC_UNLOCK_FUNCTION() {}
> 
>-  template<typename ThreadLikeObject>
>+  template <typename ThreadLikeObject>
>   static bool IsCurrent(const ThreadLikeObject* thread_like_object) {
>     return thread_like_object->IsCurrent();
>   }
>@@ -101,7 +99,7 @@ class RTC_SCOPED_LOCKABLE AnnounceOnThread {
> }  // namespace internal
> }  // namespace rtc
> 
>-// RTC_RUN_ON/RTC_ACCESS_ON/RTC_DCHECK_RUN_ON macros allows to annotate
>+// RTC_RUN_ON/RTC_GUARDED_BY/RTC_DCHECK_RUN_ON macros allows to annotate
> // variables are accessed from same thread/task queue.
> // Using tools designed to check mutexes, it checks at compile time everywhere
> // variable is access, there is a run-time dcheck thread/task queue is correct.
>@@ -115,7 +113,7 @@ class RTC_SCOPED_LOCKABLE AnnounceOnThread {
> //
> //  private:
> //   rtc::Thread* network_thread_;
>-//   int transport_ RTC_ACCESS_ON(network_thread_);
>+//   int transport_ RTC_GUARDED_BY(network_thread_);
> // };
> //
> // class ExampleThreadChecker {
>@@ -131,7 +129,7 @@ class RTC_SCOPED_LOCKABLE AnnounceOnThread {
> //   }
> //
> //  private:
>-//   int pacer_var_ RTC_ACCESS_ON(pacer_thread_checker_);
>+//   int pacer_var_ RTC_GUARDED_BY(pacer_thread_checker_);
> //   rtc::ThreadChecker pacer_thread_checker_;
> // };
> //
>@@ -147,7 +145,7 @@ class RTC_SCOPED_LOCKABLE AnnounceOnThread {
> //
> //    private:
> //     rtc::TaskQueue* const encoder_queue_;
>-//     Frame var_ RTC_ACCESS_ON(encoder_queue_);
>+//     Frame var_ RTC_GUARDED_BY(encoder_queue_);
> //   };
> //
> //   void Encode() {
>@@ -162,14 +160,11 @@ class RTC_SCOPED_LOCKABLE AnnounceOnThread {
> //   rtc::scoped_ref_ptr<Encoder> encoder_;
> // }
> 
>-// Document if a variable/field is not shared and should be accessed from
>-// same thread/task queue.
>-#define RTC_ACCESS_ON(x) RTC_THREAD_ANNOTATION_ATTRIBUTE__(guarded_by(x))
>-
> // Document if a function expected to be called from same thread/task queue.
>-#define RTC_RUN_ON(x) RTC_THREAD_ANNOTATION_ATTRIBUTE__(exclusive_locks_required(x))
>+#define RTC_RUN_ON(x) \
>+  RTC_THREAD_ANNOTATION_ATTRIBUTE__(exclusive_locks_required(x))
> 
>-#define RTC_DCHECK_RUN_ON(thread_like_object) \
>+#define RTC_DCHECK_RUN_ON(thread_like_object)                           \
>   rtc::internal::AnnounceOnThread thread_announcer(thread_like_object); \
>   RTC_DCHECK(rtc::internal::AnnounceOnThread::IsCurrent(thread_like_object))
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/thread_checker_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/thread_checker_impl.cc
>index 6ec5c91d2e8..a9e87c67a01 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/thread_checker_impl.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/thread_checker_impl.cc
>@@ -12,15 +12,11 @@
> 
> #include "rtc_base/thread_checker_impl.h"
> 
>-#include "rtc_base/platform_thread.h"
>-
> namespace rtc {
> 
>-ThreadCheckerImpl::ThreadCheckerImpl() : valid_thread_(CurrentThreadRef()) {
>-}
>+ThreadCheckerImpl::ThreadCheckerImpl() : valid_thread_(CurrentThreadRef()) {}
> 
>-ThreadCheckerImpl::~ThreadCheckerImpl() {
>-}
>+ThreadCheckerImpl::~ThreadCheckerImpl() {}
> 
> bool ThreadCheckerImpl::CalledOnValidThread() const {
>   const PlatformThreadRef current_thread = CurrentThreadRef();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/thread_checker_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/thread_checker_unittest.cc
>index 071815d1360..9c0942f884d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/thread_checker_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/thread_checker_unittest.cc
>@@ -38,9 +38,7 @@ class ThreadCheckerClass : public ThreadChecker {
>   // Verifies that it was called on the same thread as the constructor.
>   void DoStuff() { RTC_DCHECK(CalledOnValidThread()); }
> 
>-  void DetachFromThread() {
>-    ThreadChecker::DetachFromThread();
>-  }
>+  void DetachFromThread() { ThreadChecker::DetachFromThread(); }
> 
>   static void MethodOnDifferentThreadImpl();
>   static void DetachThenCallFromDifferentThreadImpl();
>@@ -62,9 +60,7 @@ class CallDoStuffOnThread : public Thread {
> 
>   // New method. Needed since Thread::Join is protected, and it is called by
>   // the TEST.
>-  void Join() {
>-    Thread::Join();
>-  }
>+  void Join() { Thread::Join(); }
> 
>  private:
>   ThreadCheckerClass* thread_checker_class_;
>@@ -86,9 +82,7 @@ class DeleteThreadCheckerClassOnThread : public Thread {
> 
>   // New method. Needed since Thread::Join is protected, and it is called by
>   // the TEST.
>-  void Join() {
>-    Thread::Join();
>-  }
>+  void Join() { Thread::Join(); }
> 
>   bool has_been_deleted() const { return !thread_checker_class_; }
> 
>@@ -157,9 +151,7 @@ void ThreadCheckerClass::MethodOnDifferentThreadImpl() {
> 
> #if ENABLE_THREAD_CHECKER
> TEST(ThreadCheckerDeathTest, MethodNotAllowedOnDifferentThreadInDebug) {
>-  ASSERT_DEATH({
>-      ThreadCheckerClass::MethodOnDifferentThreadImpl();
>-    }, "");
>+  ASSERT_DEATH({ ThreadCheckerClass::MethodOnDifferentThreadImpl(); }, "");
> }
> #else
> TEST(ThreadCheckerTest, MethodAllowedOnDifferentThreadInRelease) {
>@@ -186,9 +178,8 @@ void ThreadCheckerClass::DetachThenCallFromDifferentThreadImpl() {
> 
> #if ENABLE_THREAD_CHECKER
> TEST(ThreadCheckerDeathTest, DetachFromThreadInDebug) {
>-  ASSERT_DEATH({
>-    ThreadCheckerClass::DetachThenCallFromDifferentThreadImpl();
>-    }, "");
>+  ASSERT_DEATH({ ThreadCheckerClass::DetachThenCallFromDifferentThreadImpl(); },
>+               "");
> }
> #else
> TEST(ThreadCheckerTest, DetachFromThreadInRelease) {
>@@ -204,12 +195,12 @@ class ThreadAnnotateTest {
>   // specific T).
>   // TODO(danilchap): Find a way to test they do not compile when thread
>   // annotation checks enabled.
>-  template<typename T>
>+  template <typename T>
>   void access_var_no_annotate() {
>     var_thread_ = 42;
>   }
> 
>-  template<typename T>
>+  template <typename T>
>   void access_fun_no_annotate() {
>     function();
>   }
>@@ -248,9 +239,9 @@ class ThreadAnnotateTest {
>   rtc::ThreadChecker checker_;
>   rtc::TaskQueue* queue_;
> 
>-  int var_thread_ RTC_ACCESS_ON(thread_);
>+  int var_thread_ RTC_GUARDED_BY(thread_);
>   int var_checker_ RTC_GUARDED_BY(checker_);
>-  int var_queue_ RTC_ACCESS_ON(queue_);
>+  int var_queue_ RTC_GUARDED_BY(queue_);
> };
> 
> // Just in case we ever get lumped together with other compilation units.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/thread_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/thread_unittest.cc
>index 5f00064eb17..d5c53f8a7cc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/thread_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/thread_unittest.cc
>@@ -16,8 +16,8 @@
> #include "rtc_base/gunit.h"
> #include "rtc_base/nullsocketserver.h"
> #include "rtc_base/physicalsocketserver.h"
>-#include "rtc_base/sigslot.h"
> #include "rtc_base/socketaddress.h"
>+#include "rtc_base/third_party/sigslot/sigslot.h"
> #include "rtc_base/thread.h"
> 
> #if defined(WEBRTC_WIN)
>@@ -51,8 +51,10 @@ struct TestMessage : public MessageData {
> // Receives on a socket and sends by posting messages.
> class SocketClient : public TestGenerator, public sigslot::has_slots<> {
>  public:
>-  SocketClient(AsyncSocket* socket, const SocketAddress& addr,
>-               Thread* post_thread, MessageHandler* phandler)
>+  SocketClient(AsyncSocket* socket,
>+               const SocketAddress& addr,
>+               Thread* post_thread,
>+               MessageHandler* phandler)
>       : socket_(AsyncUDPSocket::Create(socket, addr)),
>         post_thread_(post_thread),
>         post_handler_(phandler) {
>@@ -63,7 +65,9 @@ class SocketClient : public TestGenerator, public sigslot::has_slots<> {
> 
>   SocketAddress address() const { return socket_->GetLocalAddress(); }
> 
>-  void OnPacket(AsyncPacketSocket* socket, const char* buf, size_t size,
>+  void OnPacket(AsyncPacketSocket* socket,
>+                const char* buf,
>+                size_t size,
>                 const SocketAddress& remote_addr,
>                 const PacketTime& packet_time) {
>     EXPECT_EQ(size, sizeof(uint32_t));
>@@ -83,9 +87,7 @@ class SocketClient : public TestGenerator, public sigslot::has_slots<> {
> // Receives messages and sends on a socket.
> class MessageClient : public MessageHandler, public TestGenerator {
>  public:
>-  MessageClient(Thread* pth, Socket* socket)
>-      : socket_(socket) {
>-  }
>+  MessageClient(Thread* pth, Socket* socket) : socket_(socket) {}
> 
>   ~MessageClient() override { delete socket_; }
> 
>@@ -107,15 +109,10 @@ class CustomThread : public rtc::Thread {
>   ~CustomThread() override { Stop(); }
>   bool Start() { return false; }
> 
>-  bool WrapCurrent() {
>-    return Thread::WrapCurrent();
>-  }
>-  void UnwrapCurrent() {
>-    Thread::UnwrapCurrent();
>-  }
>+  bool WrapCurrent() { return Thread::WrapCurrent(); }
>+  void UnwrapCurrent() { Thread::UnwrapCurrent(); }
> };
> 
>-
> // A thread that does nothing when it runs and signals an event
> // when it is destroyed.
> class SignalWhenDestroyedThread : public Thread {
>@@ -176,7 +173,11 @@ struct FunctorA {
> class FunctorB {
>  public:
>   explicit FunctorB(AtomicBool* flag) : flag_(flag) {}
>-  void operator()() { if (flag_) *flag_ = true; }
>+  void operator()() {
>+    if (flag_)
>+      *flag_ = true;
>+  }
>+
>  private:
>   AtomicBool* flag_;
> };
>@@ -186,6 +187,20 @@ struct FunctorC {
>     return 24;
>   }
> };
>+struct FunctorD {
>+ public:
>+  explicit FunctorD(AtomicBool* flag) : flag_(flag) {}
>+  FunctorD(FunctorD&&) = default;
>+  FunctorD& operator=(FunctorD&&) = default;
>+  void operator()() {
>+    if (flag_)
>+      *flag_ = true;
>+  }
>+
>+ private:
>+  AtomicBool* flag_;
>+  RTC_DISALLOW_COPY_AND_ASSIGN(FunctorD);
>+};
> 
> // See: https://code.google.com/p/webrtc/issues/detail?id=2409
> TEST(ThreadTest, DISABLED_Main) {
>@@ -421,9 +436,7 @@ class AsyncInvokeTest : public testing::Test {
> 
>  protected:
>   enum { kWaitTimeout = 1000 };
>-  AsyncInvokeTest()
>-      : int_value_(0),
>-        expected_thread_(nullptr) {}
>+  AsyncInvokeTest() : int_value_(0), expected_thread_(nullptr) {}
> 
>   int int_value_;
>   Thread* expected_thread_;
>@@ -441,6 +454,18 @@ TEST_F(AsyncInvokeTest, FireAndForget) {
>   thread->Stop();
> }
> 
>+TEST_F(AsyncInvokeTest, NonCopyableFunctor) {
>+  AsyncInvoker invoker;
>+  // Create and start the thread.
>+  auto thread = Thread::CreateWithSocketServer();
>+  thread->Start();
>+  // Try calling functor.
>+  AtomicBool called;
>+  invoker.AsyncInvoke<void>(RTC_FROM_HERE, thread.get(), FunctorD(&called));
>+  EXPECT_TRUE_WAIT(called.get(), kWaitTimeout);
>+  thread->Stop();
>+}
>+
> TEST_F(AsyncInvokeTest, KillInvokerDuringExecute) {
>   // Use these events to get in a state where the functor is in the middle of
>   // executing, and then to wait for it to finish, ensuring the "EXPECT_FALSE"
>@@ -558,9 +583,7 @@ class GuardedAsyncInvokeTest : public testing::Test {
> 
>  protected:
>   const static int kWaitTimeout = 1000;
>-  GuardedAsyncInvokeTest()
>-      : int_value_(0),
>-        expected_thread_(nullptr) {}
>+  GuardedAsyncInvokeTest() : int_value_(0), expected_thread_(nullptr) {}
> 
>   int int_value_;
>   Thread* expected_thread_;
>@@ -602,6 +625,14 @@ TEST_F(GuardedAsyncInvokeTest, FireAndForget) {
>   EXPECT_TRUE_WAIT(called.get(), kWaitTimeout);
> }
> 
>+TEST_F(GuardedAsyncInvokeTest, NonCopyableFunctor) {
>+  GuardedAsyncInvoker invoker;
>+  // Try calling functor.
>+  AtomicBool called;
>+  EXPECT_TRUE(invoker.AsyncInvoke<void>(RTC_FROM_HERE, FunctorD(&called)));
>+  EXPECT_TRUE_WAIT(called.get(), kWaitTimeout);
>+}
>+
> TEST_F(GuardedAsyncInvokeTest, Flush) {
>   GuardedAsyncInvoker invoker;
>   AtomicBool flag1;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/time/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/time/BUILD.gn
>new file mode 100644
>index 00000000000..7b298ce1170
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/time/BUILD.gn
>@@ -0,0 +1,23 @@
>+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+#
>+# Use of this source code is governed by a BSD-style license
>+# that can be found in the LICENSE file in the root of the source
>+# tree. An additional intellectual property rights grant can be found
>+# in the file PATENTS.  All contributing project authors may
>+# be found in the AUTHORS file in the root of the source tree.
>+
>+import("../../webrtc.gni")
>+if (is_android) {
>+  import("//build/config/android/config.gni")
>+  import("//build/config/android/rules.gni")
>+}
>+
>+rtc_source_set("timestamp_extrapolator") {
>+  sources = [
>+    "timestamp_extrapolator.cc",
>+    "timestamp_extrapolator.h",
>+  ]
>+  deps = [
>+    "../synchronization:rw_lock_wrapper",
>+  ]
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/timestamp_extrapolator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/time/timestamp_extrapolator.cc
>similarity index 92%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/timestamp_extrapolator.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/time/timestamp_extrapolator.cc
>index b8c6ba0934e..bf9f726c425 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/timestamp_extrapolator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/time/timestamp_extrapolator.cc
>@@ -8,7 +8,7 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "system_wrappers/include/timestamp_extrapolator.h"
>+#include "rtc_base/time/timestamp_extrapolator.h"
> 
> #include <algorithm>
> 
>@@ -178,12 +178,14 @@ void TimestampExtrapolator::CheckForWrapArounds(uint32_t ts90khz) {
>       // Forward wrap around
>       _wrapArounds++;
>     }
>-  }
>-  // This difference will probably be less than -2^31 if we have had a backward
>-  // wrap around. Since it is casted to a Word32, it should be positive.
>-  else if (static_cast<int32_t>(_prevWrapTimestamp - ts90khz) > 0) {
>-    // Backward wrap around
>-    _wrapArounds--;
>+  } else {
>+    // This difference will probably be less than -2^31 if we have had a
>+    // backward wrap around. Since it is casted to a Word32, it should be
>+    // positive.
>+    if (static_cast<int32_t>(_prevWrapTimestamp - ts90khz) > 0) {
>+      // Backward wrap around
>+      _wrapArounds--;
>+    }
>   }
>   _prevWrapTimestamp = ts90khz;
> }
>@@ -193,9 +195,9 @@ bool TimestampExtrapolator::DelayChangeDetection(double error) {
>   error = (error > 0) ? std::min(error, _accMaxError)
>                       : std::max(error, -_accMaxError);
>   _detectorAccumulatorPos =
>-      std::max(_detectorAccumulatorPos + error - _accDrift, (double)0);
>+      std::max(_detectorAccumulatorPos + error - _accDrift, double{0});
>   _detectorAccumulatorNeg =
>-      std::min(_detectorAccumulatorNeg + error + _accDrift, (double)0);
>+      std::min(_detectorAccumulatorNeg + error + _accDrift, double{0});
>   if (_detectorAccumulatorPos > _alarmThreshold ||
>       _detectorAccumulatorNeg < -_alarmThreshold) {
>     // Alarm
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/timestamp_extrapolator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/time/timestamp_extrapolator.h
>similarity index 82%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/timestamp_extrapolator.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/time/timestamp_extrapolator.h
>index 94181003733..63af57b2275 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/timestamp_extrapolator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/time/timestamp_extrapolator.h
>@@ -8,11 +8,12 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#ifndef SYSTEM_WRAPPERS_INCLUDE_TIMESTAMP_EXTRAPOLATOR_H_
>-#define SYSTEM_WRAPPERS_INCLUDE_TIMESTAMP_EXTRAPOLATOR_H_
>+#ifndef RTC_BASE_TIME_TIMESTAMP_EXTRAPOLATOR_H_
>+#define RTC_BASE_TIME_TIMESTAMP_EXTRAPOLATOR_H_
> 
>-#include "system_wrappers/include/rw_lock_wrapper.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include <stdint.h>
>+
>+#include "rtc_base/synchronization/rw_lock_wrapper.h"
> 
> namespace webrtc {
> 
>@@ -51,4 +52,4 @@ class TimestampExtrapolator {
> 
> }  // namespace webrtc
> 
>-#endif  // SYSTEM_WRAPPERS_INCLUDE_TIMESTAMP_EXTRAPOLATOR_H_
>+#endif  // RTC_BASE_TIME_TIMESTAMP_EXTRAPOLATOR_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/timedelta.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/timedelta.h
>deleted file mode 100644
>index c8dcf03bd3a..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/timedelta.h
>+++ /dev/null
>@@ -1,129 +0,0 @@
>-/*
>- *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef RTC_BASE_TIMEDELTA_H_
>-#define RTC_BASE_TIMEDELTA_H_
>-
>-#include <stdint.h>
>-
>-#include "rtc_base/timeutils.h"
>-
>-// Convenience class to convert between different units of relative time.
>-// Stores time to precision of nanoseconds, as int64_t internally.
>-// Doesn't check for overflow/underflow.
>-//
>-// Based on TimeDelta in:
>-// https://code.google.com/p/chromium/codesearch#chromium/src/base/time/time.h
>-namespace rtc {
>-
>-class TimeDelta {
>- public:
>-  TimeDelta() : delta_(0) {}
>-
>-  // Converts units of time to TimeDeltas.
>-  static constexpr TimeDelta FromSeconds(int64_t secs) {
>-    return TimeDelta(secs * kNumNanosecsPerSec);
>-  }
>-  static constexpr TimeDelta FromMilliseconds(int64_t ms) {
>-    return TimeDelta(ms * kNumNanosecsPerMillisec);
>-  }
>-  static constexpr TimeDelta FromMicroseconds(int64_t us) {
>-    return TimeDelta(us * kNumNanosecsPerMicrosec);
>-  }
>-  static constexpr TimeDelta FromNanoseconds(int64_t ns) {
>-    return TimeDelta(ns);
>-  }
>-
>-  // Returns true if the time delta is zero.
>-  bool is_zero() const { return delta_ == 0; }
>-
>-  // Converts TimeDelta to units of time.
>-  int64_t ToSeconds() const { return delta_ / kNumNanosecsPerSec; }
>-  int64_t ToMilliseconds() const { return delta_ / kNumNanosecsPerMillisec; }
>-  int64_t ToMicroseconds() const { return delta_ / kNumNanosecsPerMicrosec; }
>-  int64_t ToNanoseconds() const { return delta_; }
>-
>-  TimeDelta& operator=(TimeDelta other) {
>-    delta_ = other.delta_;
>-    return *this;
>-  }
>-
>-  // Computations with other deltas.
>-  TimeDelta operator+(TimeDelta other) const {
>-    return TimeDelta(delta_ + other.delta_);
>-  }
>-  TimeDelta operator-(TimeDelta other) const {
>-    return TimeDelta(delta_ + other.delta_);
>-  }
>-
>-  TimeDelta& operator+=(TimeDelta other) { return *this = (*this + other); }
>-  TimeDelta& operator-=(TimeDelta other) { return *this = (*this - other); }
>-  TimeDelta operator-() const { return TimeDelta(-delta_); }
>-
>-  // Computations with numeric types.
>-  template <typename T>
>-  TimeDelta operator*(T a) const {
>-    return TimeDelta(delta_ * a);
>-  }
>-  template <typename T>
>-  TimeDelta operator/(T a) const {
>-    return TimeDelta(delta_ / a);
>-  }
>-  template <typename T>
>-  TimeDelta& operator*=(T a) {
>-    return *this = (*this * a);
>-  }
>-  template <typename T>
>-  TimeDelta& operator/=(T a) {
>-    return *this = (*this / a);
>-  }
>-
>-  TimeDelta operator%(TimeDelta a) const {
>-    return TimeDelta(delta_ % a.delta_);
>-  }
>-
>-  // Comparison operators.
>-  constexpr bool operator==(TimeDelta other) const {
>-    return delta_ == other.delta_;
>-  }
>-  constexpr bool operator!=(TimeDelta other) const {
>-    return delta_ != other.delta_;
>-  }
>-  constexpr bool operator<(TimeDelta other) const {
>-    return delta_ < other.delta_;
>-  }
>-  constexpr bool operator<=(TimeDelta other) const {
>-    return delta_ <= other.delta_;
>-  }
>-  constexpr bool operator>(TimeDelta other) const {
>-    return delta_ > other.delta_;
>-  }
>-  constexpr bool operator>=(TimeDelta other) const {
>-    return delta_ >= other.delta_;
>-  }
>-
>- private:
>-  // Constructs a delta given the duration in nanoseconds. This is private
>-  // to avoid confusion by callers with an integer constructor. Use
>-  // FromSeconds, FromMilliseconds, etc. instead.
>-  constexpr explicit TimeDelta(int64_t delta_ns) : delta_(delta_ns) {}
>-
>-  // Delta in nanoseconds.
>-  int64_t delta_;
>-};
>-
>-template <typename T>
>-inline TimeDelta operator*(T a, TimeDelta td) {
>-  return td * a;
>-}
>-
>-}  // namespace rtc
>-
>-#endif  // RTC_BASE_TIMEDELTA_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/timeutils.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/timeutils.cc
>index f924d5fddc4..de8fc3259c2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/timeutils.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/timeutils.cc
>@@ -18,15 +18,17 @@
> #endif
> 
> #if defined(WEBRTC_WIN)
>-#ifndef WIN32_LEAN_AND_MEAN
>-#define WIN32_LEAN_AND_MEAN
>-#endif
>+// clang-format off
>+// clang formatting would put <windows.h> last,
>+// which leads to compilation failure.
> #include <windows.h>
> #include <mmsystem.h>
> #include <sys/timeb.h>
>+// clang-format on
> #endif
> 
> #include "rtc_base/checks.h"
>+#include "rtc_base/numerics/safe_conversions.h"
> #include "rtc_base/timeutils.h"
> 
> namespace rtc {
>@@ -55,7 +57,13 @@ int64_t SystemTimeNanos() {
>     }
>   }
>   // Use timebase to convert absolute time tick units into nanoseconds.
>-  ticks = mach_absolute_time() * timebase.numer / timebase.denom;
>+  const auto mul = [](uint64_t a, uint32_t b) -> int64_t {
>+    RTC_DCHECK_NE(b, 0);
>+    RTC_DCHECK_LE(a, std::numeric_limits<int64_t>::max() / b)
>+        << "The multiplication " << a << " * " << b << " overflows";
>+    return rtc::dchecked_cast<int64_t>(a * b);
>+  };
>+  ticks = mul(mach_absolute_time(), timebase.numer) / timebase.denom;
> #elif defined(WEBRTC_POSIX)
>   struct timespec ts;
>   // TODO(deadbeef): Do we need to handle the case when CLOCK_MONOTONIC is not
>@@ -157,8 +165,8 @@ int64_t TmToSeconds(const std::tm& tm) {
>   int min = tm.tm_min;
>   int sec = tm.tm_sec;
> 
>-  bool expiry_in_leap_year = (year % 4 == 0 &&
>-                              (year % 100 != 0 || year % 400 == 0));
>+  bool expiry_in_leap_year =
>+      (year % 4 == 0 && (year % 100 != 0 || year % 400 == 0));
> 
>   if (year < 1970)
>     return -1;
>@@ -181,16 +189,21 @@ int64_t TmToSeconds(const std::tm& tm) {
> 
>   // We will have added one day too much above if expiration is during a leap
>   // year, and expiration is in January or February.
>-  if (expiry_in_leap_year && month <= 2 - 1) // |month| is zero based.
>+  if (expiry_in_leap_year && month <= 2 - 1)  // |month| is zero based.
>     day -= 1;
> 
>   // Combine all variables into seconds from 1970-01-01 00:00 (except |month|
>   // which was accumulated into |day| above).
>-  return (((static_cast<int64_t>
>-            (year - 1970) * 365 + day) * 24 + hour) * 60 + min) * 60 + sec;
>+  return (((static_cast<int64_t>(year - 1970) * 365 + day) * 24 + hour) * 60 +
>+          min) *
>+             60 +
>+         sec;
> }
> 
> int64_t TimeUTCMicros() {
>+  if (g_clock) {
>+    return g_clock->TimeNanos() / kNumNanosecsPerMicrosec;
>+  }
> #if defined(WEBRTC_POSIX)
>   struct timeval time;
>   gettimeofday(&time, nullptr);
>@@ -207,4 +220,8 @@ int64_t TimeUTCMicros() {
> #endif
> }
> 
>-} // namespace rtc
>+int64_t TimeUTCMillis() {
>+  return TimeUTCMicros() / kNumMicrosecsPerMillisec;
>+}
>+
>+}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/timeutils.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/timeutils.h
>index f602d483d5a..20b1dac2c10 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/timeutils.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/timeutils.h
>@@ -80,7 +80,6 @@ int64_t TimeMicros();
> // Returns the current time in nanoseconds.
> int64_t TimeNanos();
> 
>-
> // Returns a future timestamp, 'elapsed' milliseconds from now.
> int64_t TimeAfter(int64_t elapsed);
> 
>@@ -127,6 +126,10 @@ int64_t TmToSeconds(const std::tm& tm);
> // measuring time intervals and timeouts.
> int64_t TimeUTCMicros();
> 
>+// Return the number of milliseconds since January 1, 1970, UTC.
>+// See above.
>+int64_t TimeUTCMillis();
>+
> // Interval of time from the range [min, max] inclusive.
> class IntervalRange {
>  public:
>@@ -148,6 +151,8 @@ class IntervalRange {
>     return min_ == o.min_ && max_ == o.max_;
>   }
> 
>+  bool operator!=(const IntervalRange& o) const { return !operator==(o); }
>+
>  private:
>   int min_;
>   int max_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/timeutils_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/timeutils_unittest.cc
>index 592b7f82b96..0e1949aee31 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/timeutils_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/timeutils_unittest.cc
>@@ -33,7 +33,7 @@ TEST(TimeTest, Intervals) {
> 
>   // We can't depend on ts_later and ts_earlier to be exactly 500 apart
>   // since time elapses between the calls to TimeMillis() and TimeAfter(500)
>-  EXPECT_LE(500,  TimeDiff(ts_later, ts_earlier));
>+  EXPECT_LE(500, TimeDiff(ts_later, ts_earlier));
>   EXPECT_GE(-500, TimeDiff(ts_earlier, ts_later));
> 
>   // Time has elapsed since ts_earlier
>@@ -125,7 +125,6 @@ class TmToSeconds : public testing::Test {
>   void TestTmToSeconds(int times) {
>     static char mdays[12] = {31, 28, 31, 30, 31, 30, 31, 31, 30, 31, 30, 31};
>     for (int i = 0; i < times; i++) {
>-
>       // First generate something correct and check that TmToSeconds is happy.
>       int year = rtc::CreateRandomId() % 400 + 1970;
> 
>@@ -206,96 +205,6 @@ TEST_F(TmToSeconds, TestTmToSeconds) {
>   TestTmToSeconds(100000);
> }
> 
>-TEST(TimeDelta, FromAndTo) {
>-  EXPECT_TRUE(TimeDelta::FromSeconds(2) == TimeDelta::FromMilliseconds(2000));
>-  EXPECT_TRUE(TimeDelta::FromMilliseconds(3) ==
>-              TimeDelta::FromMicroseconds(3000));
>-  EXPECT_TRUE(TimeDelta::FromMicroseconds(4) ==
>-              TimeDelta::FromNanoseconds(4000));
>-  EXPECT_EQ(13, TimeDelta::FromSeconds(13).ToSeconds());
>-  EXPECT_EQ(13, TimeDelta::FromMilliseconds(13).ToMilliseconds());
>-  EXPECT_EQ(13, TimeDelta::FromMicroseconds(13).ToMicroseconds());
>-  EXPECT_EQ(13, TimeDelta::FromNanoseconds(13).ToNanoseconds());
>-}
>-
>-TEST(TimeDelta, ComparisonOperators) {
>-  EXPECT_LT(TimeDelta::FromSeconds(1), TimeDelta::FromSeconds(2));
>-  EXPECT_EQ(TimeDelta::FromSeconds(3), TimeDelta::FromSeconds(3));
>-  EXPECT_GT(TimeDelta::FromSeconds(5), TimeDelta::FromSeconds(4));
>-}
>-
>-TEST(TimeDelta, NumericOperators) {
>-  double d = 0.5;
>-  EXPECT_EQ(TimeDelta::FromMilliseconds(500),
>-            TimeDelta::FromMilliseconds(1000) * d);
>-  EXPECT_EQ(TimeDelta::FromMilliseconds(2000),
>-            TimeDelta::FromMilliseconds(1000) / d);
>-  EXPECT_EQ(TimeDelta::FromMilliseconds(500),
>-            TimeDelta::FromMilliseconds(1000) *= d);
>-  EXPECT_EQ(TimeDelta::FromMilliseconds(2000),
>-            TimeDelta::FromMilliseconds(1000) /= d);
>-  EXPECT_EQ(TimeDelta::FromMilliseconds(500),
>-            d * TimeDelta::FromMilliseconds(1000));
>-
>-  float f = 0.5;
>-  EXPECT_EQ(TimeDelta::FromMilliseconds(500),
>-            TimeDelta::FromMilliseconds(1000) * f);
>-  EXPECT_EQ(TimeDelta::FromMilliseconds(2000),
>-            TimeDelta::FromMilliseconds(1000) / f);
>-  EXPECT_EQ(TimeDelta::FromMilliseconds(500),
>-            TimeDelta::FromMilliseconds(1000) *= f);
>-  EXPECT_EQ(TimeDelta::FromMilliseconds(2000),
>-            TimeDelta::FromMilliseconds(1000) /= f);
>-  EXPECT_EQ(TimeDelta::FromMilliseconds(500),
>-            f * TimeDelta::FromMilliseconds(1000));
>-
>-  int i = 2;
>-  EXPECT_EQ(TimeDelta::FromMilliseconds(2000),
>-            TimeDelta::FromMilliseconds(1000) * i);
>-  EXPECT_EQ(TimeDelta::FromMilliseconds(500),
>-            TimeDelta::FromMilliseconds(1000) / i);
>-  EXPECT_EQ(TimeDelta::FromMilliseconds(2000),
>-            TimeDelta::FromMilliseconds(1000) *= i);
>-  EXPECT_EQ(TimeDelta::FromMilliseconds(500),
>-            TimeDelta::FromMilliseconds(1000) /= i);
>-  EXPECT_EQ(TimeDelta::FromMilliseconds(2000),
>-            i * TimeDelta::FromMilliseconds(1000));
>-
>-  int64_t i64 = 2;
>-  EXPECT_EQ(TimeDelta::FromMilliseconds(2000),
>-            TimeDelta::FromMilliseconds(1000) * i64);
>-  EXPECT_EQ(TimeDelta::FromMilliseconds(500),
>-            TimeDelta::FromMilliseconds(1000) / i64);
>-  EXPECT_EQ(TimeDelta::FromMilliseconds(2000),
>-            TimeDelta::FromMilliseconds(1000) *= i64);
>-  EXPECT_EQ(TimeDelta::FromMilliseconds(500),
>-            TimeDelta::FromMilliseconds(1000) /= i64);
>-  EXPECT_EQ(TimeDelta::FromMilliseconds(2000),
>-            i64 * TimeDelta::FromMilliseconds(1000));
>-
>-  EXPECT_EQ(TimeDelta::FromMilliseconds(500),
>-            TimeDelta::FromMilliseconds(1000) * 0.5);
>-  EXPECT_EQ(TimeDelta::FromMilliseconds(2000),
>-            TimeDelta::FromMilliseconds(1000) / 0.5);
>-  EXPECT_EQ(TimeDelta::FromMilliseconds(500),
>-            TimeDelta::FromMilliseconds(1000) *= 0.5);
>-  EXPECT_EQ(TimeDelta::FromMilliseconds(2000),
>-            TimeDelta::FromMilliseconds(1000) /= 0.5);
>-  EXPECT_EQ(TimeDelta::FromMilliseconds(500),
>-            0.5 * TimeDelta::FromMilliseconds(1000));
>-
>-  EXPECT_EQ(TimeDelta::FromMilliseconds(2000),
>-            TimeDelta::FromMilliseconds(1000) * 2);
>-  EXPECT_EQ(TimeDelta::FromMilliseconds(500),
>-            TimeDelta::FromMilliseconds(1000) / 2);
>-  EXPECT_EQ(TimeDelta::FromMilliseconds(2000),
>-            TimeDelta::FromMilliseconds(1000) *= 2);
>-  EXPECT_EQ(TimeDelta::FromMilliseconds(500),
>-            TimeDelta::FromMilliseconds(1000) /= 2);
>-  EXPECT_EQ(TimeDelta::FromMilliseconds(2000),
>-            2 * TimeDelta::FromMilliseconds(1000));
>-}
>-
> // Test that all the time functions exposed by TimeUtils get time from the
> // fake clock when it's set.
> TEST(FakeClock, TimeFunctionsUseFakeClock) {
>@@ -329,14 +238,14 @@ TEST(FakeClock, SetTimeNanos) {
> 
> TEST(FakeClock, AdvanceTime) {
>   FakeClock clock;
>-  clock.AdvanceTime(TimeDelta::FromNanoseconds(1111u));
>-  EXPECT_EQ(1111, clock.TimeNanos());
>-  clock.AdvanceTime(TimeDelta::FromMicroseconds(2222u));
>-  EXPECT_EQ(2223111, clock.TimeNanos());
>-  clock.AdvanceTime(TimeDelta::FromMilliseconds(3333u));
>-  EXPECT_EQ(3335223111, clock.TimeNanos());
>-  clock.AdvanceTime(TimeDelta::FromSeconds(4444u));
>-  EXPECT_EQ(4447335223111, clock.TimeNanos());
>+  clock.AdvanceTime(webrtc::TimeDelta::us(1u));
>+  EXPECT_EQ(1000, clock.TimeNanos());
>+  clock.AdvanceTime(webrtc::TimeDelta::us(2222u));
>+  EXPECT_EQ(2223000, clock.TimeNanos());
>+  clock.AdvanceTime(webrtc::TimeDelta::ms(3333u));
>+  EXPECT_EQ(3335223000, clock.TimeNanos());
>+  clock.AdvanceTime(webrtc::TimeDelta::seconds(4444u));
>+  EXPECT_EQ(4447335223000, clock.TimeNanos());
> }
> 
> // When the clock is advanced, threads that are waiting in a socket select
>@@ -367,7 +276,7 @@ TEST(FakeClock, SettingTimeWakesThreads) {
> 
>   // Advance the fake clock, expecting the worker thread to wake up
>   // and dispatch the message instantly.
>-  clock.AdvanceTime(TimeDelta::FromSeconds(60u));
>+  clock.AdvanceTime(webrtc::TimeDelta::seconds(60u));
>   EXPECT_TRUE(message_handler_dispatched.Wait(0));
>   worker->Stop();
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/trace_event.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/trace_event.h
>index 7a9f2ddf73e..fee9a7f7665 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/trace_event.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/trace_event.h
>@@ -168,35 +168,6 @@
>     INTERNAL_TRACE_EVENT_ADD_SCOPED(category, name, arg1_name, arg1_val, \
>         arg2_name, arg2_val)
> 
>-// Same as TRACE_EVENT except that they are not included in official builds.
>-#ifdef OFFICIAL_BUILD
>-#define UNSHIPPED_TRACE_EVENT0(category, name) (void)0
>-#define UNSHIPPED_TRACE_EVENT1(category, name, arg1_name, arg1_val) (void)0
>-#define UNSHIPPED_TRACE_EVENT2(category, name, arg1_name, arg1_val, \
>-                               arg2_name, arg2_val) (void)0
>-#define UNSHIPPED_TRACE_EVENT_INSTANT0(category, name) (void)0
>-#define UNSHIPPED_TRACE_EVENT_INSTANT1(category, name, arg1_name, arg1_val) \
>-    (void)0
>-#define UNSHIPPED_TRACE_EVENT_INSTANT2(category, name, arg1_name, arg1_val, \
>-                                       arg2_name, arg2_val) (void)0
>-#else
>-#define UNSHIPPED_TRACE_EVENT0(category, name) \
>-    TRACE_EVENT0(category, name)
>-#define UNSHIPPED_TRACE_EVENT1(category, name, arg1_name, arg1_val) \
>-    TRACE_EVENT1(category, name, arg1_name, arg1_val)
>-#define UNSHIPPED_TRACE_EVENT2(category, name, arg1_name, arg1_val, \
>-                               arg2_name, arg2_val) \
>-    TRACE_EVENT2(category, name, arg1_name, arg1_val, arg2_name, arg2_val)
>-#define UNSHIPPED_TRACE_EVENT_INSTANT0(category, name) \
>-    TRACE_EVENT_INSTANT0(category, name)
>-#define UNSHIPPED_TRACE_EVENT_INSTANT1(category, name, arg1_name, arg1_val) \
>-    TRACE_EVENT_INSTANT1(category, name, arg1_name, arg1_val)
>-#define UNSHIPPED_TRACE_EVENT_INSTANT2(category, name, arg1_name, arg1_val, \
>-                                       arg2_name, arg2_val) \
>-    TRACE_EVENT_INSTANT2(category, name, arg1_name, arg1_val, \
>-                         arg2_name, arg2_val)
>-#endif
>-
> // Records a single event called "name" immediately, with 0, 1 or 2
> // associated arguments. If the category is not enabled, then this
> // does nothing.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/transformadapter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/transformadapter.cc
>deleted file mode 100644
>index 943a5b96727..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/transformadapter.cc
>+++ /dev/null
>@@ -1,197 +0,0 @@
>-/*
>- *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "rtc_base/transformadapter.h"
>-
>-#include <string.h>
>-
>-#include "rtc_base/checks.h"
>-
>-namespace rtc {
>-
>-///////////////////////////////////////////////////////////////////////////////
>-
>-TransformAdapter::TransformAdapter(StreamInterface * stream,
>-                                   TransformInterface * transform,
>-                                   bool direction_read)
>-    : StreamAdapterInterface(stream), transform_(transform),
>-      direction_read_(direction_read), state_(ST_PROCESSING), len_(0) {
>-}
>-
>-TransformAdapter::~TransformAdapter() {
>-  TransformAdapter::Close();
>-  delete transform_;
>-}
>-
>-StreamResult
>-TransformAdapter::Read(void * buffer, size_t buffer_len,
>-                       size_t * read, int * error) {
>-  if (!direction_read_)
>-    return SR_EOS;
>-
>-  while (state_ != ST_ERROR) {
>-    if (state_ == ST_COMPLETE)
>-      return SR_EOS;
>-
>-    // Buffer more data
>-    if ((state_ == ST_PROCESSING) && (len_ < sizeof(buffer_))) {
>-      size_t subread;
>-      StreamResult result = StreamAdapterInterface::Read(
>-                              buffer_ + len_,
>-                              sizeof(buffer_) - len_,
>-                              &subread,
>-                              &error_);
>-      if (result == SR_BLOCK) {
>-        return SR_BLOCK;
>-      } else if (result == SR_ERROR) {
>-        state_ = ST_ERROR;
>-        break;
>-      } else if (result == SR_EOS) {
>-        state_ = ST_FLUSHING;
>-      } else {
>-        len_ += subread;
>-      }
>-    }
>-
>-    // Process buffered data
>-    size_t in_len = len_;
>-    size_t out_len = buffer_len;
>-    StreamResult result = transform_->Transform(buffer_, &in_len,
>-                                                buffer, &out_len,
>-                                                (state_ == ST_FLUSHING));
>-    RTC_DCHECK(result != SR_BLOCK);
>-    if (result == SR_EOS) {
>-      // Note: Don't signal SR_EOS this iteration, unless out_len is zero
>-      state_ = ST_COMPLETE;
>-    } else if (result == SR_ERROR) {
>-      state_ = ST_ERROR;
>-      error_ = -1; // TODO: propagate error
>-      break;
>-    } else if ((out_len == 0) && (state_ == ST_FLUSHING)) {
>-      // If there is no output AND no more input, then something is wrong
>-      state_ = ST_ERROR;
>-      error_ = -1; // TODO: better error code?
>-      break;
>-    }
>-
>-    len_ -= in_len;
>-    if (len_ > 0)
>-      memmove(buffer_, buffer_ + in_len, len_);
>-
>-    if (out_len == 0)
>-      continue;
>-
>-    if (read)
>-      *read = out_len;
>-    return SR_SUCCESS;
>-  }
>-
>-  if (error)
>-    *error = error_;
>-  return SR_ERROR;
>-}
>-
>-StreamResult
>-TransformAdapter::Write(const void * data, size_t data_len,
>-                        size_t * written, int * error) {
>-  if (direction_read_)
>-    return SR_EOS;
>-
>-  size_t bytes_written = 0;
>-  while (state_ != ST_ERROR) {
>-    if (state_ == ST_COMPLETE)
>-      return SR_EOS;
>-
>-    if (len_ < sizeof(buffer_)) {
>-      // Process buffered data
>-      size_t in_len = data_len;
>-      size_t out_len = sizeof(buffer_) - len_;
>-      StreamResult result = transform_->Transform(data, &in_len,
>-                                                  buffer_ + len_, &out_len,
>-                                                  (state_ == ST_FLUSHING));
>-
>-      RTC_DCHECK(result != SR_BLOCK);
>-      if (result == SR_EOS) {
>-        // Note: Don't signal SR_EOS this iteration, unless no data written
>-        state_ = ST_COMPLETE;
>-      } else if (result == SR_ERROR) {
>-        RTC_NOTREACHED();  // When this happens, think about what should be done
>-        state_ = ST_ERROR;
>-        error_ = -1; // TODO: propagate error
>-        break;
>-      }
>-
>-      len_ = out_len;
>-      bytes_written = in_len;
>-    }
>-
>-    size_t pos = 0;
>-    while (pos < len_) {
>-      size_t subwritten;
>-      StreamResult result = StreamAdapterInterface::Write(buffer_ + pos,
>-                                                          len_ - pos,
>-                                                          &subwritten,
>-                                                          &error_);
>-      if (result == SR_BLOCK) {
>-        RTC_NOTREACHED();  // We should handle this
>-        return SR_BLOCK;
>-      } else if (result == SR_ERROR) {
>-        state_ = ST_ERROR;
>-        break;
>-      } else if (result == SR_EOS) {
>-        state_ = ST_COMPLETE;
>-        break;
>-      }
>-
>-      pos += subwritten;
>-    }
>-
>-    len_ -= pos;
>-    if (len_ > 0)
>-      memmove(buffer_, buffer_ + pos, len_);
>-
>-    if (bytes_written == 0)
>-      continue;
>-
>-    if (written)
>-      *written = bytes_written;
>-    return SR_SUCCESS;
>-  }
>-
>-  if (error)
>-    *error = error_;
>-  return SR_ERROR;
>-}
>-
>-void
>-TransformAdapter::Close() {
>-  if (!direction_read_ && (state_ == ST_PROCESSING)) {
>-    state_ = ST_FLUSHING;
>-    do {
>-      Write(0, 0, nullptr, nullptr);
>-    } while (state_ == ST_FLUSHING);
>-  }
>-  state_ = ST_COMPLETE;
>-  StreamAdapterInterface::Close();
>-}
>-
>-bool TransformAdapter::GetAvailable(size_t* size) const {
>-  return false;
>-}
>-
>-bool TransformAdapter::ReserveSize(size_t size) {
>-  return true;
>-}
>-
>-bool TransformAdapter::Rewind() {
>-  return false;
>-}
>-
>-} // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/transformadapter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/transformadapter.h
>deleted file mode 100644
>index 5e8aa9a4aad..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/transformadapter.h
>+++ /dev/null
>@@ -1,84 +0,0 @@
>-/*
>- *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef RTC_BASE_TRANSFORMADAPTER_H_
>-#define RTC_BASE_TRANSFORMADAPTER_H_
>-
>-#include "rtc_base/stream.h"
>-
>-namespace rtc {
>-///////////////////////////////////////////////////////////////////////////////
>-
>-class TransformInterface {
>-public:
>-  virtual ~TransformInterface() { }
>-
>-  // Transform should convert the in_len bytes of input into the out_len-sized
>-  // output buffer.  If flush is true, there will be no more data following
>-  // input.
>-  // After the transformation, in_len contains the number of bytes consumed, and
>-  // out_len contains the number of bytes ready in output.
>-  // Note: Transform should not return SR_BLOCK, as there is no asynchronous
>-  // notification available.
>-  virtual StreamResult Transform(const void * input, size_t * in_len,
>-                                 void * output, size_t * out_len,
>-                                 bool flush) = 0;
>-};
>-
>-///////////////////////////////////////////////////////////////////////////////
>-
>-// TransformAdapter causes all data passed through to be transformed by the
>-// supplied TransformInterface object, which may apply compression, encryption,
>-// etc.
>-
>-class TransformAdapter : public StreamAdapterInterface {
>-public:
>-  // Note that the transformation is unidirectional, in the direction specified
>-  // by the constructor.  Operations in the opposite direction result in SR_EOS.
>-  TransformAdapter(StreamInterface * stream,
>-                   TransformInterface * transform,
>-                   bool direction_read);
>-  ~TransformAdapter() override;
>-
>-  StreamResult Read(void* buffer,
>-                    size_t buffer_len,
>-                    size_t* read,
>-                    int* error) override;
>-  StreamResult Write(const void* data,
>-                     size_t data_len,
>-                     size_t* written,
>-                     int* error) override;
>-  void Close() override;
>-
>-  // Apriori, we can't tell what the transformation does to the stream length.
>-  bool GetAvailable(size_t* size) const override;
>-  bool ReserveSize(size_t size) override;
>-
>-  // Transformations might not be restartable
>-  virtual bool Rewind();
>-
>-private:
>-  enum State { ST_PROCESSING, ST_FLUSHING, ST_COMPLETE, ST_ERROR };
>-  enum { BUFFER_SIZE = 1024 };
>-
>-  TransformInterface * transform_;
>-  bool direction_read_;
>-  State state_;
>-  int error_;
>-
>-  char buffer_[BUFFER_SIZE];
>-  size_t len_;
>-};
>-
>-///////////////////////////////////////////////////////////////////////////////
>-
>-} // namespace rtc
>-
>-#endif // RTC_BASE_TRANSFORMADAPTER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/type_traits.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/type_traits.h
>index 4f004cda8be..0cb899c47f1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/type_traits.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/type_traits.h
>@@ -133,7 +133,7 @@ static_assert(IsIntlike<int>::value, "");
> static_assert(!IsIntlike<float>::value, "");
> static_assert(!IsIntlike<S>::value, "");
> 
>-}  // test_enum_intlike
>+}  // namespace test_enum_intlike
> 
> }  // namespace rtc
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/unittest_main.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/unittest_main.cc
>index 330bd3f813f..0b5a39da555 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/unittest_main.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/unittest_main.cc
>@@ -19,6 +19,8 @@
> #include "rtc_base/logging.h"
> #include "rtc_base/ssladapter.h"
> #include "rtc_base/sslstreamadapter.h"
>+#include "system_wrappers/include/field_trial_default.h"
>+#include "system_wrappers/include/metrics_default.h"
> #include "test/field_trial.h"
> #include "test/testsupport/fileutils.h"
> 
>@@ -36,7 +38,8 @@ DEFINE_string(
>     " will assign the group Enable to field trial WebRTC-FooFeature.");
> #if defined(WEBRTC_WIN)
> DEFINE_int(crt_break_alloc, -1, "memory allocation to break on");
>-DEFINE_bool(default_error_handlers, false,
>+DEFINE_bool(default_error_handlers,
>+            false,
>             "leave the default exception/dbg handler functions in place");
> 
> void TestInvalidParameterHandler(const wchar_t* expression,
>@@ -44,11 +47,10 @@ void TestInvalidParameterHandler(const wchar_t* expression,
>                                  const wchar_t* file,
>                                  unsigned int line,
>                                  uintptr_t pReserved) {
>+  // In order to log `expression`, `function`, and `file` here, we would have
>+  // to convert them to const char*. std::wcsrtombs can do that, but it's
>+  // locale dependent.
>   RTC_LOG(LS_ERROR) << "InvalidParameter Handler called.  Exiting.";
>-  RTC_LOG(LS_ERROR) << expression << std::endl
>-                    << function << std::endl
>-                    << file << std::endl
>-                    << line;
>   exit(1);
> }
> void TestPureCallHandler() {
>@@ -76,7 +78,11 @@ int main(int argc, char* argv[]) {
>   }
> 
>   webrtc::test::SetExecutablePath(argv[0]);
>-  webrtc::test::InitFieldTrialsFromString(FLAG_force_fieldtrials);
>+  webrtc::test::ValidateFieldTrialsStringOrDie(FLAG_force_fieldtrials);
>+  // InitFieldTrialsFromString stores the char*, so the char array must outlive
>+  // the application.
>+  webrtc::field_trial::InitFieldTrialsFromString(FLAG_force_fieldtrials);
>+  webrtc::metrics::Enable();
> 
> #if defined(WEBRTC_WIN)
>   if (!FLAG_default_error_handlers) {
>@@ -87,7 +93,7 @@ int main(int argc, char* argv[]) {
>   }
> 
> #if !defined(NDEBUG)  // Turn on memory leak checking on Windows.
>-  _CrtSetDbgFlag(_CRTDBG_ALLOC_MEM_DF |_CRTDBG_LEAK_CHECK_DF);
>+  _CrtSetDbgFlag(_CRTDBG_ALLOC_MEM_DF | _CRTDBG_LEAK_CHECK_DF);
>   if (FLAG_crt_break_alloc >= 0) {
>     _crtBreakAlloc = FLAG_crt_break_alloc;
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/unixfilesystem.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/unixfilesystem.cc
>index a48aca1a33d..023c34c8c94 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/unixfilesystem.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/unixfilesystem.cc
>@@ -57,7 +57,7 @@ UnixFilesystem::UnixFilesystem() {}
> 
> UnixFilesystem::~UnixFilesystem() {}
> 
>-bool UnixFilesystem::DeleteFile(const Pathname &filename) {
>+bool UnixFilesystem::DeleteFile(const Pathname& filename) {
>   RTC_LOG(LS_INFO) << "Deleting file:" << filename.pathname();
> 
>   if (!IsFile(filename)) {
>@@ -67,8 +67,8 @@ bool UnixFilesystem::DeleteFile(const Pathname &filename) {
>   return ::unlink(filename.pathname().c_str()) == 0;
> }
> 
>-bool UnixFilesystem::MoveFile(const Pathname &old_path,
>-                              const Pathname &new_path) {
>+bool UnixFilesystem::MoveFile(const Pathname& old_path,
>+                              const Pathname& new_path) {
>   if (!IsFile(old_path)) {
>     RTC_DCHECK(IsFile(old_path));
>     return false;
>@@ -81,7 +81,7 @@ bool UnixFilesystem::MoveFile(const Pathname &old_path,
>   return true;
> }
> 
>-bool UnixFilesystem::IsFolder(const Pathname &path) {
>+bool UnixFilesystem::IsFolder(const Pathname& path) {
>   struct stat st;
>   if (stat(path.pathname().c_str(), &st) < 0)
>     return false;
>@@ -95,7 +95,7 @@ bool UnixFilesystem::IsFile(const Pathname& pathname) {
>   return res == 0 && !S_ISDIR(st.st_mode);
> }
> 
>-bool UnixFilesystem::GetFileSize(const Pathname& pathname, size_t *size) {
>+bool UnixFilesystem::GetFileSize(const Pathname& pathname, size_t* size) {
>   struct stat st;
>   if (::stat(pathname.pathname().c_str(), &st) != 0)
>     return false;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/virtualsocket_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/virtualsocket_unittest.cc
>index a6a4ee1e311..d2bff546883 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/virtualsocket_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/virtualsocket_unittest.cc
>@@ -16,11 +16,11 @@
> 
> #include <memory>
> 
>+#include "absl/memory/memory.h"
> #include "rtc_base/arraysize.h"
> #include "rtc_base/fakeclock.h"
> #include "rtc_base/gunit.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/testclient.h"
> #include "rtc_base/testutils.h"
> #include "rtc_base/thread.h"
>@@ -40,7 +40,7 @@ using webrtc::testing::StreamSink;
> struct Sender : public MessageHandler {
>   Sender(Thread* th, AsyncSocket* s, uint32_t rt)
>       : thread(th),
>-        socket(MakeUnique<AsyncUDPSocket>(s)),
>+        socket(absl::make_unique<AsyncUDPSocket>(s)),
>         done(false),
>         rate(rt),
>         count(0) {
>@@ -86,7 +86,7 @@ struct Sender : public MessageHandler {
> struct Receiver : public MessageHandler, public sigslot::has_slots<> {
>   Receiver(Thread* th, AsyncSocket* s, uint32_t bw)
>       : thread(th),
>-        socket(MakeUnique<AsyncUDPSocket>(s)),
>+        socket(absl::make_unique<AsyncUDPSocket>(s)),
>         bandwidth(bw),
>         done(false),
>         count(0),
>@@ -100,7 +100,9 @@ struct Receiver : public MessageHandler, public sigslot::has_slots<> {
> 
>   ~Receiver() override { thread->Clear(this); }
> 
>-  void OnReadPacket(AsyncPacketSocket* s, const char* data, size_t size,
>+  void OnReadPacket(AsyncPacketSocket* s,
>+                    const char* data,
>+                    size_t size,
>                     const SocketAddress& remote_addr,
>                     const PacketTime& packet_time) {
>     ASSERT_EQ(socket.get(), s);
>@@ -182,8 +184,8 @@ class VirtualSocketServerTest : public testing::Test {
>     socket->Bind(EmptySocketAddressWithFamily(default_route.family()));
>     SocketAddress client1_any_addr = socket->GetLocalAddress();
>     EXPECT_TRUE(client1_any_addr.IsAnyIP());
>-    auto client1 = MakeUnique<TestClient>(MakeUnique<AsyncUDPSocket>(socket),
>-                                          &fake_clock_);
>+    auto client1 = absl::make_unique<TestClient>(
>+        absl::make_unique<AsyncUDPSocket>(socket), &fake_clock_);
> 
>     // Create client2 bound to the default route.
>     AsyncSocket* socket2 =
>@@ -191,8 +193,8 @@ class VirtualSocketServerTest : public testing::Test {
>     socket2->Bind(SocketAddress(default_route, 0));
>     SocketAddress client2_addr = socket2->GetLocalAddress();
>     EXPECT_FALSE(client2_addr.IsAnyIP());
>-    auto client2 = MakeUnique<TestClient>(MakeUnique<AsyncUDPSocket>(socket2),
>-                                          &fake_clock_);
>+    auto client2 = absl::make_unique<TestClient>(
>+        absl::make_unique<AsyncUDPSocket>(socket2), &fake_clock_);
> 
>     // Client1 sends to client2, client2 should see the default route as
>     // client1's address.
>@@ -215,12 +217,12 @@ class VirtualSocketServerTest : public testing::Test {
>     // Make sure VSS didn't switch families on us.
>     EXPECT_EQ(server_addr.family(), initial_addr.family());
> 
>-    auto client1 = MakeUnique<TestClient>(MakeUnique<AsyncUDPSocket>(socket),
>-                                          &fake_clock_);
>+    auto client1 = absl::make_unique<TestClient>(
>+        absl::make_unique<AsyncUDPSocket>(socket), &fake_clock_);
>     AsyncSocket* socket2 =
>         ss_.CreateAsyncSocket(initial_addr.family(), SOCK_DGRAM);
>-    auto client2 = MakeUnique<TestClient>(MakeUnique<AsyncUDPSocket>(socket2),
>-                                          &fake_clock_);
>+    auto client2 = absl::make_unique<TestClient>(
>+        absl::make_unique<AsyncUDPSocket>(socket2), &fake_clock_);
> 
>     SocketAddress client2_addr;
>     EXPECT_EQ(3, client2->SendTo("foo", 3, server_addr));
>@@ -233,8 +235,8 @@ class VirtualSocketServerTest : public testing::Test {
> 
>     SocketAddress empty = EmptySocketAddressWithFamily(initial_addr.family());
>     for (int i = 0; i < 10; i++) {
>-      client2 = MakeUnique<TestClient>(
>-          WrapUnique(AsyncUDPSocket::Create(&ss_, empty)), &fake_clock_);
>+      client2 = absl::make_unique<TestClient>(
>+          absl::WrapUnique(AsyncUDPSocket::Create(&ss_, empty)), &fake_clock_);
> 
>       SocketAddress next_client2_addr;
>       EXPECT_EQ(3, client2->SendTo("foo", 3, server_addr));
>@@ -259,15 +261,15 @@ class VirtualSocketServerTest : public testing::Test {
>         EmptySocketAddressWithFamily(initial_addr.family());
> 
>     // Create client
>-    std::unique_ptr<AsyncSocket> client =
>-        WrapUnique(ss_.CreateAsyncSocket(initial_addr.family(), SOCK_STREAM));
>+    std::unique_ptr<AsyncSocket> client = absl::WrapUnique(
>+        ss_.CreateAsyncSocket(initial_addr.family(), SOCK_STREAM));
>     sink.Monitor(client.get());
>     EXPECT_EQ(client->GetState(), AsyncSocket::CS_CLOSED);
>     EXPECT_TRUE(client->GetLocalAddress().IsNil());
> 
>     // Create server
>-    std::unique_ptr<AsyncSocket> server =
>-        WrapUnique(ss_.CreateAsyncSocket(initial_addr.family(), SOCK_STREAM));
>+    std::unique_ptr<AsyncSocket> server = absl::WrapUnique(
>+        ss_.CreateAsyncSocket(initial_addr.family(), SOCK_STREAM));
>     sink.Monitor(server.get());
>     EXPECT_NE(0, server->Listen(5));  // Bind required
>     EXPECT_EQ(0, server->Bind(initial_addr));
>@@ -282,7 +284,7 @@ class VirtualSocketServerTest : public testing::Test {
> 
>     // Attempt connect to listening socket
>     EXPECT_EQ(0, client->Connect(server->GetLocalAddress()));
>-    EXPECT_NE(client->GetLocalAddress(), kEmptyAddr);  // Implicit Bind
>+    EXPECT_NE(client->GetLocalAddress(), kEmptyAddr);          // Implicit Bind
>     EXPECT_NE(AF_UNSPEC, client->GetLocalAddress().family());  // Implicit Bind
>     EXPECT_NE(client->GetLocalAddress(), server->GetLocalAddress());
> 
>@@ -300,7 +302,8 @@ class VirtualSocketServerTest : public testing::Test {
> 
>     // Server has pending connection
>     EXPECT_TRUE(sink.Check(server.get(), SSE_READ));
>-    std::unique_ptr<Socket> accepted = WrapUnique(server->Accept(&accept_addr));
>+    std::unique_ptr<Socket> accepted =
>+        absl::WrapUnique(server->Accept(&accept_addr));
>     EXPECT_TRUE(nullptr != accepted);
>     EXPECT_NE(accept_addr, kEmptyAddr);
>     EXPECT_EQ(accepted->GetRemoteAddress(), accept_addr);
>@@ -327,13 +330,13 @@ class VirtualSocketServerTest : public testing::Test {
>         EmptySocketAddressWithFamily(initial_addr.family());
> 
>     // Create client
>-    std::unique_ptr<AsyncSocket> client =
>-        WrapUnique(ss_.CreateAsyncSocket(initial_addr.family(), SOCK_STREAM));
>+    std::unique_ptr<AsyncSocket> client = absl::WrapUnique(
>+        ss_.CreateAsyncSocket(initial_addr.family(), SOCK_STREAM));
>     sink.Monitor(client.get());
> 
>     // Create server
>-    std::unique_ptr<AsyncSocket> server =
>-        WrapUnique(ss_.CreateAsyncSocket(initial_addr.family(), SOCK_STREAM));
>+    std::unique_ptr<AsyncSocket> server = absl::WrapUnique(
>+        ss_.CreateAsyncSocket(initial_addr.family(), SOCK_STREAM));
>     sink.Monitor(server.get());
>     EXPECT_EQ(0, server->Bind(initial_addr));
>     EXPECT_EQ(server->GetLocalAddress().family(), initial_addr.family());
>@@ -446,14 +449,14 @@ class VirtualSocketServerTest : public testing::Test {
>     const SocketAddress kEmptyAddr;
> 
>     // Create clients
>-    std::unique_ptr<AsyncSocket> a =
>-        WrapUnique(ss_.CreateAsyncSocket(initial_addr.family(), SOCK_STREAM));
>+    std::unique_ptr<AsyncSocket> a = absl::WrapUnique(
>+        ss_.CreateAsyncSocket(initial_addr.family(), SOCK_STREAM));
>     sink.Monitor(a.get());
>     a->Bind(initial_addr);
>     EXPECT_EQ(a->GetLocalAddress().family(), initial_addr.family());
> 
>-    std::unique_ptr<AsyncSocket> b =
>-        WrapUnique(ss_.CreateAsyncSocket(initial_addr.family(), SOCK_STREAM));
>+    std::unique_ptr<AsyncSocket> b = absl::WrapUnique(
>+        ss_.CreateAsyncSocket(initial_addr.family(), SOCK_STREAM));
>     sink.Monitor(b.get());
>     b->Bind(initial_addr);
>     EXPECT_EQ(b->GetLocalAddress().family(), initial_addr.family());
>@@ -496,14 +499,14 @@ class VirtualSocketServerTest : public testing::Test {
>     const SocketAddress kEmptyAddr;
> 
>     // Connect two sockets
>-    std::unique_ptr<AsyncSocket> a =
>-        WrapUnique(ss_.CreateAsyncSocket(initial_addr.family(), SOCK_STREAM));
>+    std::unique_ptr<AsyncSocket> a = absl::WrapUnique(
>+        ss_.CreateAsyncSocket(initial_addr.family(), SOCK_STREAM));
>     sink.Monitor(a.get());
>     a->Bind(initial_addr);
>     EXPECT_EQ(a->GetLocalAddress().family(), initial_addr.family());
> 
>-    std::unique_ptr<AsyncSocket> b =
>-        WrapUnique(ss_.CreateAsyncSocket(initial_addr.family(), SOCK_STREAM));
>+    std::unique_ptr<AsyncSocket> b = absl::WrapUnique(
>+        ss_.CreateAsyncSocket(initial_addr.family(), SOCK_STREAM));
>     sink.Monitor(b.get());
>     b->Bind(initial_addr);
>     EXPECT_EQ(b->GetLocalAddress().family(), initial_addr.family());
>@@ -618,10 +621,10 @@ class VirtualSocketServerTest : public testing::Test {
>     const SocketAddress kEmptyAddr;
> 
>     // Connect two sockets
>-    std::unique_ptr<AsyncSocket> a =
>-        WrapUnique(ss_.CreateAsyncSocket(initial_addr.family(), SOCK_STREAM));
>-    std::unique_ptr<AsyncSocket> b =
>-        WrapUnique(ss_.CreateAsyncSocket(initial_addr.family(), SOCK_STREAM));
>+    std::unique_ptr<AsyncSocket> a = absl::WrapUnique(
>+        ss_.CreateAsyncSocket(initial_addr.family(), SOCK_STREAM));
>+    std::unique_ptr<AsyncSocket> b = absl::WrapUnique(
>+        ss_.CreateAsyncSocket(initial_addr.family(), SOCK_STREAM));
>     a->Bind(initial_addr);
>     EXPECT_EQ(a->GetLocalAddress().family(), initial_addr.family());
> 
>@@ -633,7 +636,7 @@ class VirtualSocketServerTest : public testing::Test {
>     ss_.ProcessMessagesUntilIdle();
> 
>     // First, deliver all packets in 0 ms.
>-    char buffer[2] = { 0, 0 };
>+    char buffer[2] = {0, 0};
>     const char cNumPackets = 10;
>     for (char i = 0; i < cNumPackets; ++i) {
>       buffer[0] = '0' + i;
>@@ -769,8 +772,8 @@ class VirtualSocketServerTest : public testing::Test {
>     const SocketAddress kEmptyAddr;
> 
>     // Client gets a IPv4 address
>-    std::unique_ptr<AsyncSocket> client =
>-        WrapUnique(ss_.CreateAsyncSocket(client_addr.family(), SOCK_STREAM));
>+    std::unique_ptr<AsyncSocket> client = absl::WrapUnique(
>+        ss_.CreateAsyncSocket(client_addr.family(), SOCK_STREAM));
>     sink.Monitor(client.get());
>     EXPECT_EQ(client->GetState(), AsyncSocket::CS_CLOSED);
>     EXPECT_EQ(client->GetLocalAddress(), kEmptyAddr);
>@@ -778,8 +781,8 @@ class VirtualSocketServerTest : public testing::Test {
> 
>     // Server gets a non-mapped non-any IPv6 address.
>     // IPv4 sockets should not be able to connect to this.
>-    std::unique_ptr<AsyncSocket> server =
>-        WrapUnique(ss_.CreateAsyncSocket(server_addr.family(), SOCK_STREAM));
>+    std::unique_ptr<AsyncSocket> server = absl::WrapUnique(
>+        ss_.CreateAsyncSocket(server_addr.family(), SOCK_STREAM));
>     sink.Monitor(server.get());
>     server->Bind(server_addr);
>     server->Listen(5);
>@@ -789,7 +792,7 @@ class VirtualSocketServerTest : public testing::Test {
>       ss_.ProcessMessagesUntilIdle();
>       EXPECT_TRUE(sink.Check(server.get(), SSE_READ));
>       std::unique_ptr<Socket> accepted =
>-          WrapUnique(server->Accept(&accept_address));
>+          absl::WrapUnique(server->Accept(&accept_address));
>       EXPECT_TRUE(nullptr != accepted);
>       EXPECT_NE(kEmptyAddr, accept_address);
>       ss_.ProcessMessagesUntilIdle();
>@@ -815,16 +818,16 @@ class VirtualSocketServerTest : public testing::Test {
>   void CrossFamilyDatagramTest(const SocketAddress& client_addr,
>                                const SocketAddress& server_addr,
>                                bool shouldSucceed) {
>-    AsyncSocket* socket = ss_.CreateAsyncSocket(SOCK_DGRAM);
>+    AsyncSocket* socket = ss_.CreateAsyncSocket(AF_INET, SOCK_DGRAM);
>     socket->Bind(server_addr);
>     SocketAddress bound_server_addr = socket->GetLocalAddress();
>-    auto client1 = MakeUnique<TestClient>(MakeUnique<AsyncUDPSocket>(socket),
>-                                          &fake_clock_);
>+    auto client1 = absl::make_unique<TestClient>(
>+        absl::make_unique<AsyncUDPSocket>(socket), &fake_clock_);
> 
>-    AsyncSocket* socket2 = ss_.CreateAsyncSocket(SOCK_DGRAM);
>+    AsyncSocket* socket2 = ss_.CreateAsyncSocket(AF_INET, SOCK_DGRAM);
>     socket2->Bind(client_addr);
>-    auto client2 = MakeUnique<TestClient>(MakeUnique<AsyncUDPSocket>(socket2),
>-                                          &fake_clock_);
>+    auto client2 = absl::make_unique<TestClient>(
>+        absl::make_unique<AsyncUDPSocket>(socket2), &fake_clock_);
>     SocketAddress client2_addr;
> 
>     if (shouldSucceed) {
>@@ -937,116 +940,100 @@ TEST_F(VirtualSocketServerTest, delay_v6) {
> // Works, receiving socket sees 127.0.0.2.
> TEST_F(VirtualSocketServerTest, CanConnectFromMappedIPv6ToIPv4Any) {
>   CrossFamilyConnectionTest(SocketAddress("::ffff:127.0.0.2", 0),
>-                            SocketAddress("0.0.0.0", 5000),
>-                            true);
>+                            SocketAddress("0.0.0.0", 5000), true);
> }
> 
> // Fails.
> TEST_F(VirtualSocketServerTest, CantConnectFromUnMappedIPv6ToIPv4Any) {
>   CrossFamilyConnectionTest(SocketAddress("::2", 0),
>-                            SocketAddress("0.0.0.0", 5000),
>-                            false);
>+                            SocketAddress("0.0.0.0", 5000), false);
> }
> 
> // Fails.
> TEST_F(VirtualSocketServerTest, CantConnectFromUnMappedIPv6ToMappedIPv6) {
>   CrossFamilyConnectionTest(SocketAddress("::2", 0),
>-                            SocketAddress("::ffff:127.0.0.1", 5000),
>-                            false);
>+                            SocketAddress("::ffff:127.0.0.1", 5000), false);
> }
> 
> // Works. receiving socket sees ::ffff:127.0.0.2.
> TEST_F(VirtualSocketServerTest, CanConnectFromIPv4ToIPv6Any) {
>   CrossFamilyConnectionTest(SocketAddress("127.0.0.2", 0),
>-                            SocketAddress("::", 5000),
>-                            true);
>+                            SocketAddress("::", 5000), true);
> }
> 
> // Fails.
> TEST_F(VirtualSocketServerTest, CantConnectFromIPv4ToUnMappedIPv6) {
>   CrossFamilyConnectionTest(SocketAddress("127.0.0.2", 0),
>-                            SocketAddress("::1", 5000),
>-                            false);
>+                            SocketAddress("::1", 5000), false);
> }
> 
> // Works. Receiving socket sees ::ffff:127.0.0.1.
> TEST_F(VirtualSocketServerTest, CanConnectFromIPv4ToMappedIPv6) {
>   CrossFamilyConnectionTest(SocketAddress("127.0.0.1", 0),
>-                            SocketAddress("::ffff:127.0.0.2", 5000),
>-                            true);
>+                            SocketAddress("::ffff:127.0.0.2", 5000), true);
> }
> 
> // Works, receiving socket sees a result from GetNextIP.
> TEST_F(VirtualSocketServerTest, CanConnectFromUnboundIPv6ToIPv4Any) {
>   CrossFamilyConnectionTest(SocketAddress("::", 0),
>-                            SocketAddress("0.0.0.0", 5000),
>-                            true);
>+                            SocketAddress("0.0.0.0", 5000), true);
> }
> 
> // Works, receiving socket sees whatever GetNextIP gave the client.
> TEST_F(VirtualSocketServerTest, CanConnectFromUnboundIPv4ToIPv6Any) {
>   CrossFamilyConnectionTest(SocketAddress("0.0.0.0", 0),
>-                            SocketAddress("::", 5000),
>-                            true);
>+                            SocketAddress("::", 5000), true);
> }
> 
> TEST_F(VirtualSocketServerTest, CanSendDatagramFromUnboundIPv4ToIPv6Any) {
>   CrossFamilyDatagramTest(SocketAddress("0.0.0.0", 0),
>-                          SocketAddress("::", 5000),
>-                          true);
>+                          SocketAddress("::", 5000), true);
> }
> 
> TEST_F(VirtualSocketServerTest, CanSendDatagramFromMappedIPv6ToIPv4Any) {
>   CrossFamilyDatagramTest(SocketAddress("::ffff:127.0.0.1", 0),
>-                          SocketAddress("0.0.0.0", 5000),
>-                          true);
>+                          SocketAddress("0.0.0.0", 5000), true);
> }
> 
> TEST_F(VirtualSocketServerTest, CantSendDatagramFromUnMappedIPv6ToIPv4Any) {
>   CrossFamilyDatagramTest(SocketAddress("::2", 0),
>-                          SocketAddress("0.0.0.0", 5000),
>-                          false);
>+                          SocketAddress("0.0.0.0", 5000), false);
> }
> 
> TEST_F(VirtualSocketServerTest, CantSendDatagramFromUnMappedIPv6ToMappedIPv6) {
>   CrossFamilyDatagramTest(SocketAddress("::2", 0),
>-                          SocketAddress("::ffff:127.0.0.1", 5000),
>-                          false);
>+                          SocketAddress("::ffff:127.0.0.1", 5000), false);
> }
> 
> TEST_F(VirtualSocketServerTest, CanSendDatagramFromIPv4ToIPv6Any) {
>   CrossFamilyDatagramTest(SocketAddress("127.0.0.2", 0),
>-                          SocketAddress("::", 5000),
>-                          true);
>+                          SocketAddress("::", 5000), true);
> }
> 
> TEST_F(VirtualSocketServerTest, CantSendDatagramFromIPv4ToUnMappedIPv6) {
>   CrossFamilyDatagramTest(SocketAddress("127.0.0.2", 0),
>-                          SocketAddress("::1", 5000),
>-                          false);
>+                          SocketAddress("::1", 5000), false);
> }
> 
> TEST_F(VirtualSocketServerTest, CanSendDatagramFromIPv4ToMappedIPv6) {
>   CrossFamilyDatagramTest(SocketAddress("127.0.0.1", 0),
>-                          SocketAddress("::ffff:127.0.0.2", 5000),
>-                          true);
>+                          SocketAddress("::ffff:127.0.0.2", 5000), true);
> }
> 
> TEST_F(VirtualSocketServerTest, CanSendDatagramFromUnboundIPv6ToIPv4Any) {
>   CrossFamilyDatagramTest(SocketAddress("::", 0),
>-                          SocketAddress("0.0.0.0", 5000),
>-                          true);
>+                          SocketAddress("0.0.0.0", 5000), true);
> }
> 
> TEST_F(VirtualSocketServerTest, SetSendingBlockedWithUdpSocket) {
>   AsyncSocket* socket1 =
>       ss_.CreateAsyncSocket(kIPv4AnyAddress.family(), SOCK_DGRAM);
>-  std::unique_ptr<AsyncSocket> socket2 =
>-      WrapUnique(ss_.CreateAsyncSocket(kIPv4AnyAddress.family(), SOCK_DGRAM));
>+  std::unique_ptr<AsyncSocket> socket2 = absl::WrapUnique(
>+      ss_.CreateAsyncSocket(kIPv4AnyAddress.family(), SOCK_DGRAM));
>   socket1->Bind(kIPv4AnyAddress);
>   socket2->Bind(kIPv4AnyAddress);
>-  auto client1 =
>-      MakeUnique<TestClient>(MakeUnique<AsyncUDPSocket>(socket1), &fake_clock_);
>+  auto client1 = absl::make_unique<TestClient>(
>+      absl::make_unique<AsyncUDPSocket>(socket1), &fake_clock_);
> 
>   ss_.SetSendingBlocked(true);
>   EXPECT_EQ(-1, client1->SendTo("foo", 3, socket2->GetLocalAddress()));
>@@ -1064,10 +1051,10 @@ TEST_F(VirtualSocketServerTest, SetSendingBlockedWithTcpSocket) {
>   ss_.set_recv_buffer_capacity(kBufferSize);
> 
>   StreamSink sink;
>-  std::unique_ptr<AsyncSocket> socket1 =
>-      WrapUnique(ss_.CreateAsyncSocket(kIPv4AnyAddress.family(), SOCK_STREAM));
>-  std::unique_ptr<AsyncSocket> socket2 =
>-      WrapUnique(ss_.CreateAsyncSocket(kIPv4AnyAddress.family(), SOCK_STREAM));
>+  std::unique_ptr<AsyncSocket> socket1 = absl::WrapUnique(
>+      ss_.CreateAsyncSocket(kIPv4AnyAddress.family(), SOCK_STREAM));
>+  std::unique_ptr<AsyncSocket> socket2 = absl::WrapUnique(
>+      ss_.CreateAsyncSocket(kIPv4AnyAddress.family(), SOCK_STREAM));
>   sink.Monitor(socket1.get());
>   sink.Monitor(socket2.get());
>   socket1->Bind(kIPv4AnyAddress);
>@@ -1104,7 +1091,7 @@ TEST_F(VirtualSocketServerTest, SetSendingBlockedWithTcpSocket) {
> 
> TEST_F(VirtualSocketServerTest, CreatesStandardDistribution) {
>   const uint32_t kTestMean[] = {10, 100, 333, 1000};
>-  const double kTestDev[] = { 0.25, 0.1, 0.01 };
>+  const double kTestDev[] = {0.25, 0.1, 0.01};
>   // TODO(deadbeef): The current code only works for 1000 data points or more.
>   const uint32_t kTestSamples[] = {/*10, 100,*/ 1000};
>   for (size_t midx = 0; midx < arraysize(kTestMean); ++midx) {
>@@ -1114,8 +1101,7 @@ TEST_F(VirtualSocketServerTest, CreatesStandardDistribution) {
>         const uint32_t kStdDev =
>             static_cast<uint32_t>(kTestDev[didx] * kTestMean[midx]);
>         VirtualSocketServer::Function* f =
>-            VirtualSocketServer::CreateDistribution(kTestMean[midx],
>-                                                    kStdDev,
>+            VirtualSocketServer::CreateDistribution(kTestMean[midx], kStdDev,
>                                                     kTestSamples[sidx]);
>         ASSERT_TRUE(nullptr != f);
>         ASSERT_EQ(kTestSamples[sidx], f->size());
>@@ -1131,13 +1117,11 @@ TEST_F(VirtualSocketServerTest, CreatesStandardDistribution) {
>         }
>         const double stddev = sqrt(sum_sq_dev / f->size());
>         EXPECT_NEAR(kTestMean[midx], mean, 0.1 * kTestMean[midx])
>-          << "M=" << kTestMean[midx]
>-          << " SD=" << kStdDev
>-          << " N=" << kTestSamples[sidx];
>+            << "M=" << kTestMean[midx] << " SD=" << kStdDev
>+            << " N=" << kTestSamples[sidx];
>         EXPECT_NEAR(kStdDev, stddev, 0.1 * kStdDev)
>-          << "M=" << kTestMean[midx]
>-          << " SD=" << kStdDev
>-          << " N=" << kTestSamples[sidx];
>+            << "M=" << kTestMean[midx] << " SD=" << kStdDev
>+            << " N=" << kTestSamples[sidx];
>         delete f;
>       }
>     }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/virtualsocketserver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/virtualsocketserver.cc
>index 1b12cd49e6e..53dad46fced 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/virtualsocketserver.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/virtualsocketserver.cc
>@@ -28,15 +28,14 @@
> 
> namespace rtc {
> #if defined(WEBRTC_WIN)
>-const in_addr kInitialNextIPv4 = { { { 0x01, 0, 0, 0 } } };
>+const in_addr kInitialNextIPv4 = {{{0x01, 0, 0, 0}}};
> #else
> // This value is entirely arbitrary, hence the lack of concern about endianness.
>-const in_addr kInitialNextIPv4 = { 0x01000000 };
>+const in_addr kInitialNextIPv4 = {0x01000000};
> #endif
> // Starts at ::2 so as to not cause confusion with ::1.
>-const in6_addr kInitialNextIPv6 = { { {
>-      0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 2
>-    } } };
>+const in6_addr kInitialNextIPv6 = {
>+    {{0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 2}}};
> 
> const uint16_t kFirstEphemeralPort = 49152;
> const uint16_t kLastEphemeralPort = 65535;
>@@ -65,15 +64,13 @@ enum {
> class Packet : public MessageData {
>  public:
>   Packet(const char* data, size_t size, const SocketAddress& from)
>-        : size_(size), consumed_(0), from_(from) {
>+      : size_(size), consumed_(0), from_(from) {
>     RTC_DCHECK(nullptr != data);
>     data_ = new char[size_];
>     memcpy(data_, data, size_);
>   }
> 
>-  ~Packet() override {
>-    delete[] data_;
>-  }
>+  ~Packet() override { delete[] data_; }
> 
>   const char* data() const { return data_ + consumed_; }
>   size_t size() const { return size_ - consumed_; }
>@@ -92,7 +89,7 @@ class Packet : public MessageData {
> };
> 
> struct MessageAddress : public MessageData {
>-  explicit MessageAddress(const SocketAddress& a) : addr(a) { }
>+  explicit MessageAddress(const SocketAddress& a) : addr(a) {}
>   SocketAddress addr;
> };
> 
>@@ -236,15 +233,15 @@ int VirtualSocket::Close() {
> }
> 
> int VirtualSocket::Send(const void* pv, size_t cb) {
>-    if (CS_CONNECTED != state_) {
>-      error_ = ENOTCONN;
>-      return -1;
>-    }
>-    if (SOCK_DGRAM == type_) {
>-      return SendUdp(pv, cb, remote_addr_);
>-    } else {
>-      return SendTcp(pv, cb);
>-    }
>+  if (CS_CONNECTED != state_) {
>+    error_ = ENOTCONN;
>+    return -1;
>+  }
>+  if (SOCK_DGRAM == type_) {
>+    return SendUdp(pv, cb, remote_addr_);
>+  } else {
>+    return SendTcp(pv, cb);
>+  }
> }
> 
> int VirtualSocket::SendTo(const void* pv,
>@@ -409,7 +406,8 @@ void VirtualSocket::OnMessage(Message* pmsg) {
>     } else if ((SOCK_STREAM == type_) && (CS_CONNECTING == state_)) {
>       CompleteConnect(data->addr, true);
>     } else {
>-      RTC_LOG(LS_VERBOSE) << "Socket at " << local_addr_ << " is not listening";
>+      RTC_LOG(LS_VERBOSE) << "Socket at " << local_addr_.ToString()
>+                          << " is not listening";
>       server_->Disconnect(server_->LookupBinding(data->addr));
>     }
>     delete data;
>@@ -554,8 +552,7 @@ VirtualSocketServer::~VirtualSocketServer() {
> IPAddress VirtualSocketServer::GetNextIP(int family) {
>   if (family == AF_INET) {
>     IPAddress next_ip(next_ipv4_);
>-    next_ipv4_.s_addr =
>-        HostToNetwork32(NetworkToHost32(next_ipv4_.s_addr) + 1);
>+    next_ipv4_.s_addr = HostToNetwork32(NetworkToHost32(next_ipv4_.s_addr) + 1);
>     return next_ip;
>   } else if (family == AF_INET6) {
>     IPAddress next_ip(next_ipv6_);
>@@ -589,18 +586,10 @@ void VirtualSocketServer::SetSendingBlocked(bool blocked) {
>   }
> }
> 
>-Socket* VirtualSocketServer::CreateSocket(int type) {
>-  return CreateSocket(AF_INET, type);
>-}
>-
> Socket* VirtualSocketServer::CreateSocket(int family, int type) {
>   return CreateSocketInternal(family, type);
> }
> 
>-AsyncSocket* VirtualSocketServer::CreateAsyncSocket(int type) {
>-  return CreateAsyncSocket(AF_INET, type);
>-}
>-
> AsyncSocket* VirtualSocketServer::CreateAsyncSocket(int family, int type) {
>   return CreateSocketInternal(family, type);
> }
>@@ -614,8 +603,8 @@ VirtualSocket* VirtualSocketServer::CreateSocketInternal(int family, int type) {
> void VirtualSocketServer::SetMessageQueue(MessageQueue* msg_queue) {
>   msg_queue_ = msg_queue;
>   if (msg_queue_) {
>-    msg_queue_->SignalQueueDestroyed.connect(this,
>-        &VirtualSocketServer::OnMessageQueueDestroyed);
>+    msg_queue_->SignalQueueDestroyed.connect(
>+        this, &VirtualSocketServer::OnMessageQueueDestroyed);
>   }
> }
> 
>@@ -649,7 +638,7 @@ bool VirtualSocketServer::ProcessMessagesUntilIdle() {
>     if (fake_clock_) {
>       // If using a fake clock, advance it in millisecond increments until the
>       // queue is empty.
>-      fake_clock_->AdvanceTime(rtc::TimeDelta::FromMilliseconds(1));
>+      fake_clock_->AdvanceTime(webrtc::TimeDelta::ms(1));
>     } else {
>       // Otherwise, run a normal message loop.
>       Message msg;
>@@ -728,8 +717,7 @@ int VirtualSocketServer::Bind(VirtualSocket* socket, SocketAddress* addr) {
> }
> 
> VirtualSocket* VirtualSocketServer::LookupBinding(const SocketAddress& addr) {
>-  SocketAddress normalized(addr.ipaddr().Normalized(),
>-                           addr.port());
>+  SocketAddress normalized(addr.ipaddr().Normalized(), addr.port());
>   AddressMap::iterator it = bindings_->find(normalized);
>   if (it != bindings_->end()) {
>     return it->second;
>@@ -751,8 +739,7 @@ VirtualSocket* VirtualSocketServer::LookupBinding(const SocketAddress& addr) {
> 
> int VirtualSocketServer::Unbind(const SocketAddress& addr,
>                                 VirtualSocket* socket) {
>-  SocketAddress normalized(addr.ipaddr().Normalized(),
>-                           addr.port());
>+  SocketAddress normalized(addr.ipaddr().Normalized(), addr.port());
>   RTC_DCHECK((*bindings_)[normalized] == socket);
>   bindings_->erase(bindings_->find(normalized));
>   return 0;
>@@ -763,22 +750,18 @@ void VirtualSocketServer::AddConnection(const SocketAddress& local,
>                                         VirtualSocket* remote_socket) {
>   // Add this socket pair to our routing table. This will allow
>   // multiple clients to connect to the same server address.
>-  SocketAddress local_normalized(local.ipaddr().Normalized(),
>-                                 local.port());
>-  SocketAddress remote_normalized(remote.ipaddr().Normalized(),
>-                                  remote.port());
>+  SocketAddress local_normalized(local.ipaddr().Normalized(), local.port());
>+  SocketAddress remote_normalized(remote.ipaddr().Normalized(), remote.port());
>   SocketAddressPair address_pair(local_normalized, remote_normalized);
>-  connections_->insert(std::pair<SocketAddressPair,
>-                       VirtualSocket*>(address_pair, remote_socket));
>+  connections_->insert(std::pair<SocketAddressPair, VirtualSocket*>(
>+      address_pair, remote_socket));
> }
> 
> VirtualSocket* VirtualSocketServer::LookupConnection(
>     const SocketAddress& local,
>     const SocketAddress& remote) {
>-  SocketAddress local_normalized(local.ipaddr().Normalized(),
>-                                 local.port());
>-  SocketAddress remote_normalized(remote.ipaddr().Normalized(),
>-                                  remote.port());
>+  SocketAddress local_normalized(local.ipaddr().Normalized(), local.port());
>+  SocketAddress remote_normalized(remote.ipaddr().Normalized(), remote.port());
>   SocketAddressPair address_pair(local_normalized, remote_normalized);
>   ConnectionMap::iterator it = connections_->find(address_pair);
>   return (connections_->end() != it) ? it->second : nullptr;
>@@ -786,10 +769,8 @@ VirtualSocket* VirtualSocketServer::LookupConnection(
> 
> void VirtualSocketServer::RemoveConnection(const SocketAddress& local,
>                                            const SocketAddress& remote) {
>-  SocketAddress local_normalized(local.ipaddr().Normalized(),
>-                                local.port());
>-  SocketAddress remote_normalized(remote.ipaddr().Normalized(),
>-                                 remote.port());
>+  SocketAddress local_normalized(local.ipaddr().Normalized(), local.port());
>+  SocketAddress remote_normalized(remote.ipaddr().Normalized(), remote.port());
>   SocketAddressPair address_pair(local_normalized, remote_normalized);
>   connections_->erase(address_pair);
> }
>@@ -805,7 +786,8 @@ int VirtualSocketServer::Connect(VirtualSocket* socket,
>   VirtualSocket* remote = LookupBinding(remote_addr);
>   if (!CanInteractWith(socket, remote)) {
>     RTC_LOG(LS_INFO) << "Address family mismatch between "
>-                     << socket->GetLocalAddress() << " and " << remote_addr;
>+                     << socket->GetLocalAddress().ToString() << " and "
>+                     << remote_addr.ToString();
>     return -1;
>   }
>   if (remote != nullptr) {
>@@ -813,7 +795,7 @@ int VirtualSocketServer::Connect(VirtualSocket* socket,
>     msg_queue_->PostDelayed(RTC_FROM_HERE, delay, remote, MSG_ID_CONNECT,
>                             new MessageAddress(addr));
>   } else {
>-    RTC_LOG(LS_INFO) << "No one listening at " << remote_addr;
>+    RTC_LOG(LS_INFO) << "No one listening at " << remote_addr.ToString();
>     msg_queue_->PostDelayed(RTC_FROM_HERE, delay, socket, MSG_ID_DISCONNECT);
>   }
>   return 0;
>@@ -832,7 +814,8 @@ bool VirtualSocketServer::Disconnect(VirtualSocket* socket) {
> }
> 
> int VirtualSocketServer::SendUdp(VirtualSocket* socket,
>-                                 const char* data, size_t data_size,
>+                                 const char* data,
>+                                 size_t data_size,
>                                  const SocketAddress& remote_addr) {
>   ++sent_packets_;
>   if (sending_blocked_) {
>@@ -856,17 +839,18 @@ int VirtualSocketServer::SendUdp(VirtualSocket* socket,
>     dummy_socket->SetLocalAddress(remote_addr);
>     if (!CanInteractWith(socket, dummy_socket.get())) {
>       RTC_LOG(LS_VERBOSE) << "Incompatible address families: "
>-                          << socket->GetLocalAddress() << " and "
>-                          << remote_addr;
>+                          << socket->GetLocalAddress().ToString() << " and "
>+                          << remote_addr.ToString();
>       return -1;
>     }
>-    RTC_LOG(LS_VERBOSE) << "No one listening at " << remote_addr;
>+    RTC_LOG(LS_VERBOSE) << "No one listening at " << remote_addr.ToString();
>     return static_cast<int>(data_size);
>   }
> 
>   if (!CanInteractWith(socket, recipient)) {
>     RTC_LOG(LS_VERBOSE) << "Incompatible address families: "
>-                        << socket->GetLocalAddress() << " and " << remote_addr;
>+                        << socket->GetLocalAddress().ToString() << " and "
>+                        << remote_addr.ToString();
>     return -1;
>   }
> 
>@@ -912,8 +896,8 @@ void VirtualSocketServer::SendTcp(VirtualSocket* socket) {
>   // is read from the recv_buffer.
> 
>   // Lookup the local/remote pair in the connections table.
>-  VirtualSocket* recipient = LookupConnection(socket->local_addr_,
>-                                              socket->remote_addr_);
>+  VirtualSocket* recipient =
>+      LookupConnection(socket->local_addr_, socket->remote_addr_);
>   if (!recipient) {
>     RTC_LOG(LS_VERBOSE) << "Sending data to no one.";
>     return;
>@@ -1034,8 +1018,8 @@ void PrintFunction(std::vector<std::pair<double, double> >* f) {
> #endif  // <unused>
> 
> void VirtualSocketServer::UpdateDelayDistribution() {
>-  Function* dist = CreateDistribution(delay_mean_, delay_stddev_,
>-                                      delay_samples_);
>+  Function* dist =
>+      CreateDistribution(delay_mean_, delay_stddev_, delay_samples_);
>   // We take a lock just to make sure we don't leak memory.
>   {
>     CritScope cs(&delay_crit_);
>@@ -1043,9 +1027,11 @@ void VirtualSocketServer::UpdateDelayDistribution() {
>   }
> }
> 
>+static double PI = 4 * atan(1.0);
>+
> static double Normal(double x, double mean, double stddev) {
>   double a = (x - mean) * (x - mean) / (2 * stddev * stddev);
>-  return exp(-a) / (stddev * sqrt(2 * M_PI));
>+  return exp(-a) / (stddev * sqrt(2 * PI));
> }
> 
> #if 0  // static unused gives a warning
>@@ -1095,7 +1081,7 @@ uint32_t VirtualSocketServer::GetTransitDelay(Socket* socket) {
> 
> struct FunctionDomainCmp {
>   bool operator()(const VirtualSocketServer::Point& p1,
>-                   const VirtualSocketServer::Point& p2) {
>+                  const VirtualSocketServer::Point& p2) {
>     return p1.first < p2.first;
>   }
>   bool operator()(double v1, const VirtualSocketServer::Point& p2) {
>@@ -1115,7 +1101,7 @@ VirtualSocketServer::Function* VirtualSocketServer::Accumulate(Function* f) {
>     (*f)[i].second = v;
>     v = v + dx * avgy;
>   }
>-  (*f)[f->size()-1].second = v;
>+  (*f)[f->size() - 1].second = v;
>   return f;
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/virtualsocketserver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/virtualsocketserver.h
>index e17caaec3c9..4fa7d77a875 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/virtualsocketserver.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/virtualsocketserver.h
>@@ -103,10 +103,7 @@ class VirtualSocketServer : public SocketServer, public sigslot::has_slots<> {
>   void SetSendingBlocked(bool blocked);
> 
>   // SocketFactory:
>-  Socket* CreateSocket(int type) override;
>   Socket* CreateSocket(int family, int type) override;
>-
>-  AsyncSocket* CreateAsyncSocket(int type) override;
>   AsyncSocket* CreateAsyncSocket(int family, int type) override;
> 
>   // SocketServer:
>@@ -186,14 +183,17 @@ class VirtualSocketServer : public SocketServer, public sigslot::has_slots<> {
>                         const SocketAddress& server);
> 
>   // Connects the given socket to the socket at the given address
>-  int Connect(VirtualSocket* socket, const SocketAddress& remote_addr,
>+  int Connect(VirtualSocket* socket,
>+              const SocketAddress& remote_addr,
>               bool use_delay);
> 
>   // Sends a disconnect message to the socket at the given address
>   bool Disconnect(VirtualSocket* socket);
> 
>   // Sends the given packet to the socket at the given address (if one exists).
>-  int SendUdp(VirtualSocket* socket, const char* data, size_t data_size,
>+  int SendUdp(VirtualSocket* socket,
>+              const char* data,
>+              size_t data_size,
>               const SocketAddress& remote_addr);
> 
>   // Moves as much data as possible from the sender's buffer to the network
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/weak_ptr.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/weak_ptr.cc
>index c76256a0929..ad98c246119 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/weak_ptr.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/weak_ptr.cc
>@@ -51,8 +51,7 @@ bool WeakReference::is_valid() const {
>   return flag_.get() && flag_->IsValid();
> }
> 
>-WeakReferenceOwner::WeakReferenceOwner() {
>-}
>+WeakReferenceOwner::WeakReferenceOwner() {}
> 
> WeakReferenceOwner::~WeakReferenceOwner() {
>   Invalidate();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/weak_ptr_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/weak_ptr_unittest.cc
>index 6b0b45289cb..954171834e6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/weak_ptr_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/weak_ptr_unittest.cc
>@@ -225,7 +225,7 @@ TEST(WeakPtrTest, WeakPtrInitiateAndUseOnDifferentThreads) {
>   // Test that it is OK to create a WeakPtr on one thread, but use it on
>   // another. This tests that we do not trip runtime checks that ensure that a
>   // WeakPtr is not used by multiple threads.
>-  auto target = rtc::MakeUnique<TargetWithFactory>();
>+  auto target = absl::make_unique<TargetWithFactory>();
>   // Create weak ptr on main thread
>   WeakPtr<Target> weak_ptr = target->factory.GetWeakPtr();
>   rtc::TaskQueue queue("queue");
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win/windows_version.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win/windows_version.cc
>new file mode 100644
>index 00000000000..9ccde4c510b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win/windows_version.cc
>@@ -0,0 +1,397 @@
>+/*
>+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "rtc_base/win/windows_version.h"
>+
>+#include <windows.h>
>+#include <memory>
>+
>+#include "rtc_base/checks.h"
>+#include "rtc_base/stringutils.h"
>+
>+#if !defined(__clang__) && _MSC_FULL_VER < 191125507
>+#error VS 2017 Update 3.2 or higher is required
>+#endif
>+
>+#if !defined(NTDDI_WIN10_RS2)
>+// Windows 10 Creators Update SDK is required to build Chrome. It is important
>+// to install the 10.0.15063.468 version, released June 2017, because earlier
>+// versions had bugs and could not build Chrome. See this link for details:
>+// https://developercommunity.visualstudio.com/content/problem/42961/15063-sdk-is-broken-bitsh-indirectly-references-no.html
>+#error Creators Update SDK (10.0.15063.468) required.
>+#endif
>+
>+namespace {
>+
>+typedef BOOL(WINAPI* GetProductInfoPtr)(DWORD, DWORD, DWORD, DWORD, PDWORD);
>+
>+// Mask to pull WOW64 access flags out of REGSAM access.
>+const REGSAM kWow64AccessMask = KEY_WOW64_32KEY | KEY_WOW64_64KEY;
>+
>+// Utility class to read, write and manipulate the Windows Registry.
>+// Registry vocabulary primer: a "key" is like a folder, in which there
>+// are "values", which are <name, data> pairs, with an associated data type.
>+// Based on base::win::RegKey but only implements a small fraction of it.
>+class RegKey {
>+ public:
>+  RegKey() : key_(nullptr), wow64access_(0) {}
>+
>+  RegKey(HKEY rootkey, const wchar_t* subkey, REGSAM access)
>+      : key_(nullptr), wow64access_(0) {
>+    if (rootkey) {
>+      if (access & (KEY_SET_VALUE | KEY_CREATE_SUB_KEY | KEY_CREATE_LINK))
>+        Create(rootkey, subkey, access);
>+      else
>+        Open(rootkey, subkey, access);
>+    } else {
>+      RTC_DCHECK(!subkey);
>+      wow64access_ = access & kWow64AccessMask;
>+    }
>+  }
>+
>+  ~RegKey() { Close(); }
>+
>+  LONG Create(HKEY rootkey, const wchar_t* subkey, REGSAM access) {
>+    DWORD disposition_value;
>+    return CreateWithDisposition(rootkey, subkey, &disposition_value, access);
>+  }
>+
>+  LONG CreateWithDisposition(HKEY rootkey,
>+                             const wchar_t* subkey,
>+                             DWORD* disposition,
>+                             REGSAM access) {
>+    RTC_DCHECK(rootkey && subkey && access && disposition);
>+    HKEY subhkey = NULL;
>+    LONG result =
>+        ::RegCreateKeyEx(rootkey, subkey, 0, NULL, REG_OPTION_NON_VOLATILE,
>+                         access, NULL, &subhkey, disposition);
>+    if (result == ERROR_SUCCESS) {
>+      Close();
>+      key_ = subhkey;
>+      wow64access_ = access & kWow64AccessMask;
>+    }
>+
>+    return result;
>+  }
>+
>+  // Opens an existing reg key.
>+  LONG Open(HKEY rootkey, const wchar_t* subkey, REGSAM access) {
>+    RTC_DCHECK(rootkey && subkey && access);
>+    HKEY subhkey = NULL;
>+
>+    LONG result = ::RegOpenKeyEx(rootkey, subkey, 0, access, &subhkey);
>+    if (result == ERROR_SUCCESS) {
>+      Close();
>+      key_ = subhkey;
>+      wow64access_ = access & kWow64AccessMask;
>+    }
>+
>+    return result;
>+  }
>+
>+  // Closes this reg key.
>+  void Close() {
>+    if (key_) {
>+      ::RegCloseKey(key_);
>+      key_ = nullptr;
>+    }
>+  }
>+
>+  // Reads a REG_DWORD (uint32_t) into |out_value|. If |name| is null or empty,
>+  // reads the key's default value, if any.
>+  LONG ReadValueDW(const wchar_t* name, DWORD* out_value) const {
>+    RTC_DCHECK(out_value);
>+    DWORD type = REG_DWORD;
>+    DWORD size = sizeof(DWORD);
>+    DWORD local_value = 0;
>+    LONG result = ReadValue(name, &local_value, &size, &type);
>+    if (result == ERROR_SUCCESS) {
>+      if ((type == REG_DWORD || type == REG_BINARY) && size == sizeof(DWORD))
>+        *out_value = local_value;
>+      else
>+        result = ERROR_CANTREAD;
>+    }
>+
>+    return result;
>+  }
>+
>+  // Reads a string into |out_value|. If |name| is null or empty, reads
>+  // the key's default value, if any.
>+  LONG ReadValue(const wchar_t* name, std::wstring* out_value) const {
>+    RTC_DCHECK(out_value);
>+    const size_t kMaxStringLength = 1024;  // This is after expansion.
>+    // Use the one of the other forms of ReadValue if 1024 is too small for you.
>+    wchar_t raw_value[kMaxStringLength];
>+    DWORD type = REG_SZ, size = sizeof(raw_value);
>+    LONG result = ReadValue(name, raw_value, &size, &type);
>+    if (result == ERROR_SUCCESS) {
>+      if (type == REG_SZ) {
>+        *out_value = raw_value;
>+      } else if (type == REG_EXPAND_SZ) {
>+        wchar_t expanded[kMaxStringLength];
>+        size =
>+            ::ExpandEnvironmentStrings(raw_value, expanded, kMaxStringLength);
>+        // Success: returns the number of wchar_t's copied
>+        // Fail: buffer too small, returns the size required
>+        // Fail: other, returns 0
>+        if (size == 0 || size > kMaxStringLength) {
>+          result = ERROR_MORE_DATA;
>+        } else {
>+          *out_value = expanded;
>+        }
>+      } else {
>+        // Not a string. Oops.
>+        result = ERROR_CANTREAD;
>+      }
>+    }
>+
>+    return result;
>+  }
>+
>+  LONG ReadValue(const wchar_t* name,
>+                 void* data,
>+                 DWORD* dsize,
>+                 DWORD* dtype) const {
>+    LONG result = RegQueryValueEx(key_, name, 0, dtype,
>+                                  reinterpret_cast<LPBYTE>(data), dsize);
>+    return result;
>+  }
>+
>+ private:
>+  HKEY key_;
>+  REGSAM wow64access_;
>+};
>+
>+}  // namespace
>+
>+namespace rtc {
>+namespace rtc_win {
>+namespace {
>+
>+// Helper to map a major.minor.x.build version (e.g. 6.1) to a Windows release.
>+Version MajorMinorBuildToVersion(int major, int minor, int build) {
>+  if ((major == 5) && (minor > 0)) {
>+    // Treat XP Pro x64, Home Server, and Server 2003 R2 as Server 2003.
>+    return (minor == 1) ? VERSION_XP : VERSION_SERVER_2003;
>+  } else if (major == 6) {
>+    switch (minor) {
>+      case 0:
>+        // Treat Windows Server 2008 the same as Windows Vista.
>+        return VERSION_VISTA;
>+      case 1:
>+        // Treat Windows Server 2008 R2 the same as Windows 7.
>+        return VERSION_WIN7;
>+      case 2:
>+        // Treat Windows Server 2012 the same as Windows 8.
>+        return VERSION_WIN8;
>+      default:
>+        RTC_DCHECK_EQ(minor, 3);
>+        return VERSION_WIN8_1;
>+    }
>+  } else if (major == 10) {
>+    if (build < 10586) {
>+      return VERSION_WIN10;
>+    } else if (build < 14393) {
>+      return VERSION_WIN10_TH2;
>+    } else if (build < 15063) {
>+      return VERSION_WIN10_RS1;
>+    } else if (build < 16299) {
>+      return VERSION_WIN10_RS2;
>+    } else if (build < 17134) {
>+      return VERSION_WIN10_RS3;
>+    } else {
>+      return VERSION_WIN10_RS4;
>+    }
>+  } else if (major > 6) {
>+    RTC_NOTREACHED();
>+    return VERSION_WIN_LAST;
>+  }
>+
>+  return VERSION_PRE_XP;
>+}
>+
>+// Returns the the "UBR" value from the registry. Introduced in Windows 10,
>+// this undocumented value appears to be similar to a patch number.
>+// Returns 0 if the value does not exist or it could not be read.
>+int GetUBR() {
>+  // The values under the CurrentVersion registry hive are mirrored under
>+  // the corresponding Wow6432 hive.
>+  static constexpr wchar_t kRegKeyWindowsNTCurrentVersion[] =
>+      L"SOFTWARE\\Microsoft\\Windows NT\\CurrentVersion";
>+
>+  RegKey key;
>+  if (key.Open(HKEY_LOCAL_MACHINE, kRegKeyWindowsNTCurrentVersion,
>+               KEY_QUERY_VALUE) != ERROR_SUCCESS) {
>+    return 0;
>+  }
>+
>+  DWORD ubr = 0;
>+  key.ReadValueDW(L"UBR", &ubr);
>+
>+  return static_cast<int>(ubr);
>+}
>+
>+}  // namespace
>+
>+// static
>+OSInfo* OSInfo::GetInstance() {
>+  // Note: we don't use the Singleton class because it depends on AtExitManager,
>+  // and it's convenient for other modules to use this class without it. This
>+  // pattern is copied from gurl.cc.
>+  static OSInfo* info;
>+  if (!info) {
>+    OSInfo* new_info = new OSInfo();
>+    if (InterlockedCompareExchangePointer(reinterpret_cast<PVOID*>(&info),
>+                                          new_info, NULL)) {
>+      delete new_info;
>+    }
>+  }
>+  return info;
>+}
>+
>+OSInfo::OSInfo()
>+    : version_(VERSION_PRE_XP),
>+      architecture_(OTHER_ARCHITECTURE),
>+      wow64_status_(GetWOW64StatusForProcess(GetCurrentProcess())) {
>+  OSVERSIONINFOEX version_info = {sizeof version_info};
>+  ::GetVersionEx(reinterpret_cast<OSVERSIONINFO*>(&version_info));
>+  version_number_.major = version_info.dwMajorVersion;
>+  version_number_.minor = version_info.dwMinorVersion;
>+  version_number_.build = version_info.dwBuildNumber;
>+  version_number_.patch = GetUBR();
>+  version_ = MajorMinorBuildToVersion(
>+      version_number_.major, version_number_.minor, version_number_.build);
>+  service_pack_.major = version_info.wServicePackMajor;
>+  service_pack_.minor = version_info.wServicePackMinor;
>+  service_pack_str_ = rtc::ToUtf8(version_info.szCSDVersion);
>+
>+  SYSTEM_INFO system_info = {};
>+  ::GetNativeSystemInfo(&system_info);
>+  switch (system_info.wProcessorArchitecture) {
>+    case PROCESSOR_ARCHITECTURE_INTEL:
>+      architecture_ = X86_ARCHITECTURE;
>+      break;
>+    case PROCESSOR_ARCHITECTURE_AMD64:
>+      architecture_ = X64_ARCHITECTURE;
>+      break;
>+    case PROCESSOR_ARCHITECTURE_IA64:
>+      architecture_ = IA64_ARCHITECTURE;
>+      break;
>+  }
>+  processors_ = system_info.dwNumberOfProcessors;
>+  allocation_granularity_ = system_info.dwAllocationGranularity;
>+
>+  GetProductInfoPtr get_product_info;
>+  DWORD os_type;
>+
>+  if (version_info.dwMajorVersion == 6 || version_info.dwMajorVersion == 10) {
>+    // Only present on Vista+.
>+    get_product_info = reinterpret_cast<GetProductInfoPtr>(
>+        ::GetProcAddress(::GetModuleHandle(L"kernel32.dll"), "GetProductInfo"));
>+
>+    get_product_info(version_info.dwMajorVersion, version_info.dwMinorVersion,
>+                     0, 0, &os_type);
>+    switch (os_type) {
>+      case PRODUCT_CLUSTER_SERVER:
>+      case PRODUCT_DATACENTER_SERVER:
>+      case PRODUCT_DATACENTER_SERVER_CORE:
>+      case PRODUCT_ENTERPRISE_SERVER:
>+      case PRODUCT_ENTERPRISE_SERVER_CORE:
>+      case PRODUCT_ENTERPRISE_SERVER_IA64:
>+      case PRODUCT_SMALLBUSINESS_SERVER:
>+      case PRODUCT_SMALLBUSINESS_SERVER_PREMIUM:
>+      case PRODUCT_STANDARD_SERVER:
>+      case PRODUCT_STANDARD_SERVER_CORE:
>+      case PRODUCT_WEB_SERVER:
>+        version_type_ = SUITE_SERVER;
>+        break;
>+      case PRODUCT_PROFESSIONAL:
>+      case PRODUCT_ULTIMATE:
>+        version_type_ = SUITE_PROFESSIONAL;
>+        break;
>+      case PRODUCT_ENTERPRISE:
>+      case PRODUCT_ENTERPRISE_E:
>+      case PRODUCT_ENTERPRISE_EVALUATION:
>+      case PRODUCT_ENTERPRISE_N:
>+      case PRODUCT_ENTERPRISE_N_EVALUATION:
>+      case PRODUCT_ENTERPRISE_S:
>+      case PRODUCT_ENTERPRISE_S_EVALUATION:
>+      case PRODUCT_ENTERPRISE_S_N:
>+      case PRODUCT_ENTERPRISE_S_N_EVALUATION:
>+      case PRODUCT_BUSINESS:
>+      case PRODUCT_BUSINESS_N:
>+        version_type_ = SUITE_ENTERPRISE;
>+        break;
>+      case PRODUCT_EDUCATION:
>+      case PRODUCT_EDUCATION_N:
>+        version_type_ = SUITE_EDUCATION;
>+        break;
>+      case PRODUCT_HOME_BASIC:
>+      case PRODUCT_HOME_PREMIUM:
>+      case PRODUCT_STARTER:
>+      default:
>+        version_type_ = SUITE_HOME;
>+        break;
>+    }
>+  } else if (version_info.dwMajorVersion == 5 &&
>+             version_info.dwMinorVersion == 2) {
>+    if (version_info.wProductType == VER_NT_WORKSTATION &&
>+        system_info.wProcessorArchitecture == PROCESSOR_ARCHITECTURE_AMD64) {
>+      version_type_ = SUITE_PROFESSIONAL;
>+    } else if (version_info.wSuiteMask & VER_SUITE_WH_SERVER) {
>+      version_type_ = SUITE_HOME;
>+    } else {
>+      version_type_ = SUITE_SERVER;
>+    }
>+  } else if (version_info.dwMajorVersion == 5 &&
>+             version_info.dwMinorVersion == 1) {
>+    if (version_info.wSuiteMask & VER_SUITE_PERSONAL)
>+      version_type_ = SUITE_HOME;
>+    else
>+      version_type_ = SUITE_PROFESSIONAL;
>+  } else {
>+    // Windows is pre XP so we don't care but pick a safe default.
>+    version_type_ = SUITE_HOME;
>+  }
>+}
>+
>+OSInfo::~OSInfo() {}
>+
>+std::string OSInfo::processor_model_name() {
>+  if (processor_model_name_.empty()) {
>+    const wchar_t kProcessorNameString[] =
>+        L"HARDWARE\\DESCRIPTION\\System\\CentralProcessor\\0";
>+    RegKey key(HKEY_LOCAL_MACHINE, kProcessorNameString, KEY_READ);
>+    std::wstring value;
>+    key.ReadValue(L"ProcessorNameString", &value);
>+    processor_model_name_ = rtc::ToUtf8(value);
>+  }
>+  return processor_model_name_;
>+}
>+
>+// static
>+OSInfo::WOW64Status OSInfo::GetWOW64StatusForProcess(HANDLE process_handle) {
>+  typedef BOOL(WINAPI * IsWow64ProcessFunc)(HANDLE, PBOOL);
>+  IsWow64ProcessFunc is_wow64_process = reinterpret_cast<IsWow64ProcessFunc>(
>+      GetProcAddress(GetModuleHandle(L"kernel32.dll"), "IsWow64Process"));
>+  if (!is_wow64_process)
>+    return WOW64_DISABLED;
>+  BOOL is_wow64 = FALSE;
>+  if (!(*is_wow64_process)(process_handle, &is_wow64))
>+    return WOW64_UNKNOWN;
>+  return is_wow64 ? WOW64_ENABLED : WOW64_DISABLED;
>+}
>+
>+Version GetVersion() {
>+  return OSInfo::GetInstance()->version();
>+}
>+
>+}  // namespace rtc_win
>+}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win/windows_version.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win/windows_version.h
>new file mode 100644
>index 00000000000..39d333e4d7d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win/windows_version.h
>@@ -0,0 +1,149 @@
>+/*
>+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef RTC_BASE_WIN_WINDOWS_VERSION_H_
>+#define RTC_BASE_WIN_WINDOWS_VERSION_H_
>+
>+#include <stddef.h>
>+#include <string>
>+
>+#include "rtc_base/constructormagic.h"
>+
>+typedef void* HANDLE;
>+
>+namespace rtc {
>+namespace rtc_win {
>+
>+// The running version of Windows.  This is declared outside OSInfo for
>+// syntactic sugar reasons; see the declaration of GetVersion() below.
>+// NOTE: Keep these in order so callers can do things like
>+// "if (rtc_win::GetVersion() >= rtc_win::VERSION_VISTA) ...".
>+//
>+// This enum is used in metrics histograms, so they shouldn't be reordered or
>+// removed. New values can be added before VERSION_WIN_LAST.
>+enum Version {
>+  VERSION_PRE_XP = 0,  // Not supported.
>+  VERSION_XP = 1,
>+  VERSION_SERVER_2003 = 2,  // Also includes XP Pro x64 and Server 2003 R2.
>+  VERSION_VISTA = 3,        // Also includes Windows Server 2008.
>+  VERSION_WIN7 = 4,         // Also includes Windows Server 2008 R2.
>+  VERSION_WIN8 = 5,         // Also includes Windows Server 2012.
>+  VERSION_WIN8_1 = 6,       // Also includes Windows Server 2012 R2.
>+  VERSION_WIN10 = 7,        // Threshold 1: Version 1507, Build 10240.
>+  VERSION_WIN10_TH2 = 8,    // Threshold 2: Version 1511, Build 10586.
>+  VERSION_WIN10_RS1 = 9,    // Redstone 1: Version 1607, Build 14393.
>+  VERSION_WIN10_RS2 = 10,   // Redstone 2: Version 1703, Build 15063.
>+  VERSION_WIN10_RS3 = 11,   // Redstone 3: Version 1709, Build 16299.
>+  VERSION_WIN10_RS4 = 12,   // Redstone 4: Version 1803, Build 17134.
>+  // On edit, update tools\metrics\histograms\enums.xml "WindowsVersion" and
>+  // "GpuBlacklistFeatureTestResultsWindows2".
>+  VERSION_WIN_LAST,  // Indicates error condition.
>+};
>+
>+// A rough bucketing of the available types of versions of Windows. This is used
>+// to distinguish enterprise enabled versions from home versions and potentially
>+// server versions. Keep these values in the same order, since they are used as
>+// is for metrics histogram ids.
>+enum VersionType {
>+  SUITE_HOME = 0,
>+  SUITE_PROFESSIONAL,
>+  SUITE_SERVER,
>+  SUITE_ENTERPRISE,
>+  SUITE_EDUCATION,
>+  SUITE_LAST,
>+};
>+
>+// A singleton that can be used to query various pieces of information about the
>+// OS and process state. Note that this doesn't use the base Singleton class, so
>+// it can be used without an AtExitManager.
>+class OSInfo {
>+ public:
>+  struct VersionNumber {
>+    int major;
>+    int minor;
>+    int build;
>+    int patch;
>+  };
>+
>+  struct ServicePack {
>+    int major;
>+    int minor;
>+  };
>+
>+  // The processor architecture this copy of Windows natively uses.  For
>+  // example, given an x64-capable processor, we have three possibilities:
>+  //   32-bit Chrome running on 32-bit Windows:           X86_ARCHITECTURE
>+  //   32-bit Chrome running on 64-bit Windows via WOW64: X64_ARCHITECTURE
>+  //   64-bit Chrome running on 64-bit Windows:           X64_ARCHITECTURE
>+  enum WindowsArchitecture {
>+    X86_ARCHITECTURE,
>+    X64_ARCHITECTURE,
>+    IA64_ARCHITECTURE,
>+    OTHER_ARCHITECTURE,
>+  };
>+
>+  // Whether a process is running under WOW64 (the wrapper that allows 32-bit
>+  // processes to run on 64-bit versions of Windows).  This will return
>+  // WOW64_DISABLED for both "32-bit Chrome on 32-bit Windows" and "64-bit
>+  // Chrome on 64-bit Windows".  WOW64_UNKNOWN means "an error occurred", e.g.
>+  // the process does not have sufficient access rights to determine this.
>+  enum WOW64Status {
>+    WOW64_DISABLED,
>+    WOW64_ENABLED,
>+    WOW64_UNKNOWN,
>+  };
>+
>+  static OSInfo* GetInstance();
>+
>+  Version version() const { return version_; }
>+  VersionNumber version_number() const { return version_number_; }
>+  VersionType version_type() const { return version_type_; }
>+  ServicePack service_pack() const { return service_pack_; }
>+  std::string service_pack_str() const { return service_pack_str_; }
>+  WindowsArchitecture architecture() const { return architecture_; }
>+  int processors() const { return processors_; }
>+  size_t allocation_granularity() const { return allocation_granularity_; }
>+  WOW64Status wow64_status() const { return wow64_status_; }
>+  std::string processor_model_name();
>+
>+  // Like wow64_status(), but for the supplied handle instead of the current
>+  // process.  This doesn't touch member state, so you can bypass the singleton.
>+  static WOW64Status GetWOW64StatusForProcess(HANDLE process_handle);
>+
>+ private:
>+  OSInfo();
>+  ~OSInfo();
>+
>+  Version version_;
>+  VersionNumber version_number_;
>+  VersionType version_type_;
>+  ServicePack service_pack_;
>+
>+  // A string, such as "Service Pack 3", that indicates the latest Service Pack
>+  // installed on the system. If no Service Pack has been installed, the string
>+  // is empty.
>+  std::string service_pack_str_;
>+  WindowsArchitecture architecture_;
>+  int processors_;
>+  size_t allocation_granularity_;
>+  WOW64Status wow64_status_;
>+  std::string processor_model_name_;
>+
>+  RTC_DISALLOW_COPY_AND_ASSIGN(OSInfo);
>+};
>+
>+// Because this is by far the most commonly-requested value from the above
>+// singleton, we add a global-scope accessor here as syntactic sugar.
>+Version GetVersion();
>+
>+}  // namespace rtc_win
>+}  // namespace rtc
>+
>+#endif  // RTC_BASE_WIN_WINDOWS_VERSION_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win/windows_version_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win/windows_version_unittest.cc
>new file mode 100644
>index 00000000000..9e582e549f9
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win/windows_version_unittest.cc
>@@ -0,0 +1,47 @@
>+/*
>+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "rtc_base/win/windows_version.h"
>+
>+#include "rtc_base/gunit.h"
>+#include "rtc_base/logging.h"
>+
>+namespace rtc {
>+namespace rtc_win {
>+namespace {
>+
>+void MethodSupportedOnWin10AndLater() {
>+  RTC_DLOG(INFO) << "MethodSupportedOnWin10AndLater";
>+}
>+
>+void MethodNotSupportedOnWin10AndLater() {
>+  RTC_DLOG(INFO) << "MethodNotSupportedOnWin10AndLater";
>+}
>+
>+// Use global GetVersion() and use it in a way a user would typically use it
>+// when checking for support of a certain API:
>+// "if (rtc_win::GetVersion() < VERSION_WIN10) ...".
>+TEST(WindowsVersion, GetVersionGlobalScopeAccessor) {
>+  if (GetVersion() < VERSION_WIN10) {
>+    MethodNotSupportedOnWin10AndLater();
>+  } else {
>+    MethodSupportedOnWin10AndLater();
>+  }
>+}
>+
>+TEST(WindowsVersion, ProcessorModelName) {
>+  std::string name = OSInfo::GetInstance()->processor_model_name();
>+  EXPECT_FALSE(name.empty());
>+  RTC_DLOG(INFO) << "processor_model_name: " << name;
>+}
>+
>+}  // namespace
>+}  // namespace rtc_win
>+}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win32.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win32.cc
>index cbf6fbb057a..d81d685f16c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win32.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win32.cc
>@@ -15,10 +15,10 @@
> #include <algorithm>
> 
> #include "rtc_base/arraysize.h"
>-#include "rtc_base/basictypes.h"
> #include "rtc_base/byteorder.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>+#include "rtc_base/stringutils.h"
> 
> namespace rtc {
> 
>@@ -32,8 +32,10 @@ static int inet_pton_v6(const char* src, void* dst);
> // ip address). XP doesn't have its own inet_ntop, and
> // WSAAddressToString requires both IPv6 to be  installed and for Winsock
> // to be initialized.
>-const char* win32_inet_ntop(int af, const void *src,
>-                            char* dst, socklen_t size) {
>+const char* win32_inet_ntop(int af,
>+                            const void* src,
>+                            char* dst,
>+                            socklen_t size) {
>   if (!src || !dst) {
>     return nullptr;
>   }
>@@ -70,11 +72,9 @@ const char* inet_ntop_v4(const void* src, char* dst, socklen_t size) {
>   }
>   const struct in_addr* as_in_addr =
>       reinterpret_cast<const struct in_addr*>(src);
>-  rtc::sprintfn(dst, size, "%d.%d.%d.%d",
>-                      as_in_addr->S_un.S_un_b.s_b1,
>-                      as_in_addr->S_un.S_un_b.s_b2,
>-                      as_in_addr->S_un.S_un_b.s_b3,
>-                      as_in_addr->S_un.S_un_b.s_b4);
>+  rtc::sprintfn(dst, size, "%d.%d.%d.%d", as_in_addr->S_un.S_un_b.s_b1,
>+                as_in_addr->S_un.S_un_b.s_b2, as_in_addr->S_un.S_un_b.s_b3,
>+                as_in_addr->S_un.S_un_b.s_b4);
>   return dst;
> }
> 
>@@ -122,8 +122,8 @@ const char* inet_ntop_v6(const void* src, char* dst, socklen_t size) {
>   // Print IPv4 compatible and IPv4 mapped addresses using the IPv4 helper.
>   // These addresses have an initial run of either eight zero-bytes followed
>   // by 0xFFFF, or an initial run of ten zero-bytes.
>-  if (runpos[0] == 1 && (maxpos == 5 ||
>-                         (maxpos == 4 && as_shorts[5] == 0xFFFF))) {
>+  if (runpos[0] == 1 &&
>+      (maxpos == 5 || (maxpos == 4 && as_shorts[5] == 0xFFFF))) {
>     *cursor++ = ':';
>     *cursor++ = ':';
>     if (maxpos == 4) {
>@@ -136,9 +136,8 @@ const char* inet_ntop_v6(const void* src, char* dst, socklen_t size) {
>   } else {
>     for (int i = 0; i < run_array_size; ++i) {
>       if (runpos[i] == -1) {
>-        cursor += rtc::sprintfn(cursor,
>-                                      INET6_ADDRSTRLEN - (cursor - dst),
>-                                      "%x", NetworkToHost16(as_shorts[i]));
>+        cursor += rtc::sprintfn(cursor, INET6_ADDRSTRLEN - (cursor - dst), "%x",
>+                                NetworkToHost16(as_shorts[i]));
>         if (i != 7 && runpos[i + 1] != 1) {
>           *cursor++ = ':';
>         }
>@@ -221,7 +220,7 @@ int inet_pton_v6(const char* src, void* dst) {
>   // Addresses that start with "::" (i.e., a run of initial zeros) or
>   // "::ffff:" can potentially be IPv4 mapped or compatibility addresses.
>   // These have dotted-style IPv4 addresses on the end (e.g. "::192.168.7.1").
>-  if (*readcursor == ':' && *(readcursor+1) == ':' &&
>+  if (*readcursor == ':' && *(readcursor + 1) == ':' &&
>       *(readcursor + 2) != 0) {
>     // Check for periods, which we'll take as a sign of v4 addresses.
>     const char* addrstart = readcursor + 2;
>@@ -326,8 +325,8 @@ bool Utf8ToWindowsFilename(const std::string& utf8, std::wstring* filename) {
>   }
>   wchar_t* wfilename = STACK_ARRAY(wchar_t, wlen);
>   if (0 == ::MultiByteToWideChar(CP_UTF8, 0, utf8.c_str(),
>-                                 static_cast<int>(utf8.length() + 1),
>-                                 wfilename, wlen)) {
>+                                 static_cast<int>(utf8.length() + 1), wfilename,
>+                                 wlen)) {
>     return false;
>   }
>   // Replace forward slashes with backslashes
>@@ -368,9 +367,12 @@ bool GetOsVersion(int* major, int* minor, int* build) {
>   OSVERSIONINFO info = {0};
>   info.dwOSVersionInfoSize = sizeof(info);
>   if (GetVersionEx(&info)) {
>-    if (major) *major = info.dwMajorVersion;
>-    if (minor) *minor = info.dwMinorVersion;
>-    if (build) *build = info.dwBuildNumber;
>+    if (major)
>+      *major = info.dwMajorVersion;
>+    if (minor)
>+      *minor = info.dwMinorVersion;
>+    if (build)
>+      *build = info.dwBuildNumber;
>     return true;
>   }
>   return false;
>@@ -387,7 +389,6 @@ bool GetCurrentProcessIntegrityLevel(int* level) {
>       TOKEN_MANDATORY_LABEL* til =
>           reinterpret_cast<TOKEN_MANDATORY_LABEL*>(buf);
>       if (GetTokenInformation(token, TokenIntegrityLevel, til, size, &size)) {
>-
>         DWORD count = *GetSidSubAuthorityCount(til->Label.Sid);
>         *level = *GetSidSubAuthority(til->Label.Sid, count - 1);
>         ret = true;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win32.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win32.h
>index 0cbc412c0a8..a5fd541f3f4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win32.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win32.h
>@@ -11,10 +11,8 @@
> #ifndef RTC_BASE_WIN32_H_
> #define RTC_BASE_WIN32_H_
> 
>-#if defined(WEBRTC_WIN)
>-
>-#ifndef WIN32_LEAN_AND_MEAN
>-#define WIN32_LEAN_AND_MEAN
>+#ifndef WEBRTC_WIN
>+#error "Only #include this header in Windows builds"
> #endif
> 
> // Make sure we don't get min/max macros
>@@ -22,15 +20,20 @@
> #define NOMINMAX
> #endif
> 
>-#include <winsock2.h>
>+// clang-format off
>+// clang formating would change include order.
>+#include <winsock2.h> // must come first
> #include <windows.h>
>+// clang-format on
>+
>+typedef int socklen_t;
> 
> #ifndef SECURITY_MANDATORY_LABEL_AUTHORITY
> // Add defines that we use if we are compiling against older sdks
>-#define SECURITY_MANDATORY_MEDIUM_RID               (0x00002000L)
>+#define SECURITY_MANDATORY_MEDIUM_RID (0x00002000L)
> #define TokenIntegrityLevel static_cast<TOKEN_INFORMATION_CLASS>(0x19)
> typedef struct _TOKEN_MANDATORY_LABEL {
>-    SID_AND_ATTRIBUTES Label;
>+  SID_AND_ATTRIBUTES Label;
> } TOKEN_MANDATORY_LABEL, *PTOKEN_MANDATORY_LABEL;
> #endif  // SECURITY_MANDATORY_LABEL_AUTHORITY
> 
>@@ -38,13 +41,10 @@ typedef struct _TOKEN_MANDATORY_LABEL {
> 
> #include <string>
> 
>-#include "rtc_base/basictypes.h"
>-#include "rtc_base/stringutils.h"
>-
> namespace rtc {
> 
>-const char* win32_inet_ntop(int af, const void *src, char* dst, socklen_t size);
>-int win32_inet_pton(int af, const char* src, void *dst);
>+const char* win32_inet_ntop(int af, const void* src, char* dst, socklen_t size);
>+int win32_inet_pton(int af, const char* src, void* dst);
> 
> // Convert a Utf8 path representation to a non-length-limited Unicode pathname.
> bool Utf8ToWindowsFilename(const std::string& utf8, std::wstring* filename);
>@@ -52,6 +52,7 @@ bool Utf8ToWindowsFilename(const std::string& utf8, std::wstring* filename);
> enum WindowsMajorVersions {
>   kWindows2000 = 5,
>   kWindowsVista = 6,
>+  kWindows10 = 10,
> };
> bool GetOsVersion(int* major, int* minor, int* build);
> 
>@@ -72,16 +73,20 @@ inline bool IsWindows8OrLater() {
>           (major > kWindowsVista || (major == kWindowsVista && minor >= 2)));
> }
> 
>+inline bool IsWindows10OrLater() {
>+  int major;
>+  return (GetOsVersion(&major, nullptr, nullptr) && (major >= kWindows10));
>+}
>+
> // Determine the current integrity level of the process.
> bool GetCurrentProcessIntegrityLevel(int* level);
> 
> inline bool IsCurrentProcessLowIntegrity() {
>   int level;
>   return (GetCurrentProcessIntegrityLevel(&level) &&
>-      level < SECURITY_MANDATORY_MEDIUM_RID);
>+          level < SECURITY_MANDATORY_MEDIUM_RID);
> }
> 
> }  // namespace rtc
> 
>-#endif  // WEBRTC_WIN
> #endif  // RTC_BASE_WIN32_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win32_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win32_unittest.cc
>index 5ee3ff44cd4..5691be46c86 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win32_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win32_unittest.cc
>@@ -22,8 +22,7 @@ namespace rtc {
> 
> class Win32Test : public testing::Test {
>  public:
>-  Win32Test() {
>-  }
>+  Win32Test() {}
> };
> 
> TEST_F(Win32Test, IPv6AddressCompression) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win32filesystem.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win32filesystem.cc
>index 8ca84c34bb1..b500e5e629b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win32filesystem.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win32filesystem.cc
>@@ -33,7 +33,7 @@
> 
> namespace rtc {
> 
>-bool Win32Filesystem::DeleteFile(const Pathname &filename) {
>+bool Win32Filesystem::DeleteFile(const Pathname& filename) {
>   RTC_LOG(LS_INFO) << "Deleting file " << filename.pathname();
>   if (!IsFile(filename)) {
>     RTC_DCHECK(IsFile(filename));
>@@ -42,8 +42,8 @@ bool Win32Filesystem::DeleteFile(const Pathname &filename) {
>   return ::DeleteFile(ToUtf16(filename.pathname()).c_str()) != 0;
> }
> 
>-bool Win32Filesystem::MoveFile(const Pathname &old_path,
>-                               const Pathname &new_path) {
>+bool Win32Filesystem::MoveFile(const Pathname& old_path,
>+                               const Pathname& new_path) {
>   if (!IsFile(old_path)) {
>     RTC_DCHECK(IsFile(old_path));
>     return false;
>@@ -54,16 +54,16 @@ bool Win32Filesystem::MoveFile(const Pathname &old_path,
>                     ToUtf16(new_path.pathname()).c_str()) != 0;
> }
> 
>-bool Win32Filesystem::IsFolder(const Pathname &path) {
>+bool Win32Filesystem::IsFolder(const Pathname& path) {
>   WIN32_FILE_ATTRIBUTE_DATA data = {0};
>   if (0 == ::GetFileAttributesEx(ToUtf16(path.pathname()).c_str(),
>                                  GetFileExInfoStandard, &data))
>     return false;
>   return (data.dwFileAttributes & FILE_ATTRIBUTE_DIRECTORY) ==
>-      FILE_ATTRIBUTE_DIRECTORY;
>+         FILE_ATTRIBUTE_DIRECTORY;
> }
> 
>-bool Win32Filesystem::IsFile(const Pathname &path) {
>+bool Win32Filesystem::IsFile(const Pathname& path) {
>   WIN32_FILE_ATTRIBUTE_DATA data = {0};
>   if (0 == ::GetFileAttributesEx(ToUtf16(path.pathname()).c_str(),
>                                  GetFileExInfoStandard, &data))
>@@ -71,11 +71,11 @@ bool Win32Filesystem::IsFile(const Pathname &path) {
>   return (data.dwFileAttributes & FILE_ATTRIBUTE_DIRECTORY) == 0;
> }
> 
>-bool Win32Filesystem::GetFileSize(const Pathname &pathname, size_t *size) {
>+bool Win32Filesystem::GetFileSize(const Pathname& pathname, size_t* size) {
>   WIN32_FILE_ATTRIBUTE_DATA data = {0};
>   if (::GetFileAttributesEx(ToUtf16(pathname.pathname()).c_str(),
>                             GetFileExInfoStandard, &data) == 0)
>-  return false;
>+    return false;
>   *size = data.nFileSizeLow;
>   return true;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win32securityerrors.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win32securityerrors.cc
>deleted file mode 100644
>index 2f8530a6bc7..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win32securityerrors.cc
>+++ /dev/null
>@@ -1,49 +0,0 @@
>-/*
>- *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "rtc_base/logging.h"
>-#include "rtc_base/win32.h"
>-
>-namespace rtc {
>-
>-///////////////////////////////////////////////////////////////////////////////
>-
>-extern const ConstantLabel SECURITY_ERRORS[];
>-
>-const ConstantLabel SECURITY_ERRORS[] = {
>-  KLABEL(SEC_I_COMPLETE_AND_CONTINUE),
>-  KLABEL(SEC_I_COMPLETE_NEEDED),
>-  KLABEL(SEC_I_CONTEXT_EXPIRED),
>-  KLABEL(SEC_I_CONTINUE_NEEDED),
>-  KLABEL(SEC_I_INCOMPLETE_CREDENTIALS),
>-  KLABEL(SEC_I_RENEGOTIATE),
>-  KLABEL(SEC_E_CERT_EXPIRED),
>-  KLABEL(SEC_E_INCOMPLETE_MESSAGE),
>-  KLABEL(SEC_E_INSUFFICIENT_MEMORY),
>-  KLABEL(SEC_E_INTERNAL_ERROR),
>-  KLABEL(SEC_E_INVALID_HANDLE),
>-  KLABEL(SEC_E_INVALID_TOKEN),
>-  KLABEL(SEC_E_LOGON_DENIED),
>-  KLABEL(SEC_E_NO_AUTHENTICATING_AUTHORITY),
>-  KLABEL(SEC_E_NO_CREDENTIALS),
>-  KLABEL(SEC_E_NOT_OWNER),
>-  KLABEL(SEC_E_OK),
>-  KLABEL(SEC_E_SECPKG_NOT_FOUND),
>-  KLABEL(SEC_E_TARGET_UNKNOWN),
>-  KLABEL(SEC_E_UNKNOWN_CREDENTIALS),
>-  KLABEL(SEC_E_UNSUPPORTED_FUNCTION),
>-  KLABEL(SEC_E_UNTRUSTED_ROOT),
>-  KLABEL(SEC_E_WRONG_PRINCIPAL),
>-  LASTLABEL
>-};
>-
>-///////////////////////////////////////////////////////////////////////////////
>-
>-}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win32socketinit.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win32socketinit.cc
>index 5bf65468fa2..ea0aae60542 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win32socketinit.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win32socketinit.cc
>@@ -34,9 +34,8 @@ class WinsockInitializer {
>     if (!err_)
>       WSACleanup();
>   }
>-  int error() {
>-    return err_;
>-  }
>+  int error() { return err_; }
>+
>  private:
>   int err_;
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win32socketserver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win32socketserver.cc
>index d79a1b356f3..cab751a6317 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win32socketserver.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win32socketserver.cc
>@@ -10,8 +10,8 @@
> 
> #include "rtc_base/win32socketserver.h"
> 
>-#include <algorithm>
> #include <ws2tcpip.h>  // NOLINT
>+#include <algorithm>
> 
> #include "rtc_base/byteorder.h"
> #include "rtc_base/checks.h"
>@@ -24,38 +24,9 @@ namespace rtc {
> // Win32Socket
> ///////////////////////////////////////////////////////////////////////////////
> 
>-// TODO: Move this to a common place where PhysicalSocketServer can
>-// share it.
>-// Standard MTUs
>-static const uint16_t PACKET_MAXIMUMS[] = {
>-    65535,  // Theoretical maximum, Hyperchannel
>-    32000,  // Nothing
>-    17914,  // 16Mb IBM Token Ring
>-    8166,   // IEEE 802.4
>-    // 4464   // IEEE 802.5 (4Mb max)
>-    4352,   // FDDI
>-    // 2048,  // Wideband Network
>-    2002,   // IEEE 802.5 (4Mb recommended)
>-    // 1536,  // Expermental Ethernet Networks
>-    // 1500,  // Ethernet, Point-to-Point (default)
>-    1492,   // IEEE 802.3
>-    1006,   // SLIP, ARPANET
>-    // 576,   // X.25 Networks
>-    // 544,   // DEC IP Portal
>-    // 512,   // NETBIOS
>-    508,    // IEEE 802/Source-Rt Bridge, ARCNET
>-    296,    // Point-to-Point (low delay)
>-    68,     // Official minimum
>-    0,      // End of list marker
>-};
>-
>-static const int IP_HEADER_SIZE = 20u;
>-static const int ICMP_HEADER_SIZE = 8u;
>-static const int ICMP_PING_TIMEOUT_MILLIS = 10000u;
>-
> // TODO: Enable for production builds also? Use FormatMessage?
> #if !defined(NDEBUG)
>-LPCSTR WSAErrorToString(int error, LPCSTR *description_result) {
>+LPCSTR WSAErrorToString(int error, LPCSTR* description_result) {
>   LPCSTR string = "Unspecified";
>   LPCSTR description = "Unspecified description";
>   switch (error) {
>@@ -136,8 +107,8 @@ void ReportWSAError(LPCSTR context, int error, const SocketAddress& address) {}
> // Win32Socket::EventSink
> /////////////////////////////////////////////////////////////////////////////
> 
>-#define WM_SOCKETNOTIFY  (WM_USER + 50)
>-#define WM_DNSNOTIFY     (WM_USER + 51)
>+#define WM_SOCKETNOTIFY (WM_USER + 50)
>+#define WM_DNSNOTIFY (WM_USER + 51)
> 
> struct Win32Socket::DnsLookup {
>   HANDLE handle;
>@@ -147,7 +118,7 @@ struct Win32Socket::DnsLookup {
> 
> class Win32Socket::EventSink : public Win32Window {
>  public:
>-  explicit EventSink(Win32Socket * parent) : parent_(parent) { }
>+  explicit EventSink(Win32Socket* parent) : parent_(parent) {}
> 
>   void Dispose();
> 
>@@ -161,7 +132,7 @@ class Win32Socket::EventSink : public Win32Window {
>   bool OnSocketNotify(UINT uMsg, WPARAM wParam, LPARAM lParam, LRESULT& result);
>   bool OnDnsNotify(WPARAM wParam, LPARAM lParam, LRESULT& result);
> 
>-  Win32Socket * parent_;
>+  Win32Socket* parent_;
> };
> 
> void Win32Socket::EventSink::Dispose() {
>@@ -173,20 +144,24 @@ void Win32Socket::EventSink::Dispose() {
>   }
> }
> 
>-bool Win32Socket::EventSink::OnMessage(UINT uMsg, WPARAM wParam,
>-                                       LPARAM lParam, LRESULT& result) {
>+bool Win32Socket::EventSink::OnMessage(UINT uMsg,
>+                                       WPARAM wParam,
>+                                       LPARAM lParam,
>+                                       LRESULT& result) {
>   switch (uMsg) {
>-  case WM_SOCKETNOTIFY:
>-  case WM_TIMER:
>-    return OnSocketNotify(uMsg, wParam, lParam, result);
>-  case WM_DNSNOTIFY:
>-    return OnDnsNotify(wParam, lParam, result);
>+    case WM_SOCKETNOTIFY:
>+    case WM_TIMER:
>+      return OnSocketNotify(uMsg, wParam, lParam, result);
>+    case WM_DNSNOTIFY:
>+      return OnDnsNotify(wParam, lParam, result);
>   }
>   return false;
> }
> 
>-bool Win32Socket::EventSink::OnSocketNotify(UINT uMsg, WPARAM wParam,
>-                                            LPARAM lParam, LRESULT& result) {
>+bool Win32Socket::EventSink::OnSocketNotify(UINT uMsg,
>+                                            WPARAM wParam,
>+                                            LPARAM lParam,
>+                                            LRESULT& result) {
>   result = 0;
> 
>   int wsa_event = WSAGETSELECTEVENT(lParam);
>@@ -203,7 +178,8 @@ bool Win32Socket::EventSink::OnSocketNotify(UINT uMsg, WPARAM wParam,
>   return true;
> }
> 
>-bool Win32Socket::EventSink::OnDnsNotify(WPARAM wParam, LPARAM lParam,
>+bool Win32Socket::EventSink::OnDnsNotify(WPARAM wParam,
>+                                         LPARAM lParam,
>                                          LRESULT& result) {
>   result = 0;
> 
>@@ -280,8 +256,8 @@ void Win32Socket::SetTimeout(int ms) {
> SocketAddress Win32Socket::GetLocalAddress() const {
>   sockaddr_storage addr = {0};
>   socklen_t addrlen = sizeof(addr);
>-  int result = ::getsockname(socket_, reinterpret_cast<sockaddr*>(&addr),
>-                             &addrlen);
>+  int result =
>+      ::getsockname(socket_, reinterpret_cast<sockaddr*>(&addr), &addrlen);
>   SocketAddress address;
>   if (result >= 0) {
>     SocketAddressFromSockAddrStorage(addr, &address);
>@@ -295,8 +271,8 @@ SocketAddress Win32Socket::GetLocalAddress() const {
> SocketAddress Win32Socket::GetRemoteAddress() const {
>   sockaddr_storage addr = {0};
>   socklen_t addrlen = sizeof(addr);
>-  int result = ::getpeername(socket_, reinterpret_cast<sockaddr*>(&addr),
>-                             &addrlen);
>+  int result =
>+      ::getpeername(socket_, reinterpret_cast<sockaddr*>(&addr), &addrlen);
>   SocketAddress address;
>   if (result >= 0) {
>     SocketAddressFromSockAddrStorage(addr, &address);
>@@ -314,8 +290,7 @@ int Win32Socket::Bind(const SocketAddress& addr) {
> 
>   sockaddr_storage saddr;
>   size_t len = addr.ToSockAddrStorage(&saddr);
>-  int err = ::bind(socket_,
>-                   reinterpret_cast<sockaddr*>(&saddr),
>+  int err = ::bind(socket_, reinterpret_cast<sockaddr*>(&saddr),
>                    static_cast<int>(len));
>   UpdateLastError();
>   return err;
>@@ -332,7 +307,7 @@ int Win32Socket::Connect(const SocketAddress& addr) {
>   }
> 
>   RTC_LOG_F(LS_INFO) << "async dns lookup (" << addr.hostname() << ")";
>-  DnsLookup * dns = new DnsLookup;
>+  DnsLookup* dns = new DnsLookup;
>   if (!sink_) {
>     // Explicitly create the sink ourselves here; we can't rely on SetAsync
>     // because we don't have a socket_ yet.
>@@ -368,8 +343,7 @@ int Win32Socket::DoConnect(const SocketAddress& addr) {
>   sockaddr_storage saddr = {0};
>   size_t len = addr.ToSockAddrStorage(&saddr);
>   connect_time_ = Time();
>-  int result = connect(socket_,
>-                       reinterpret_cast<SOCKADDR*>(&saddr),
>+  int result = connect(socket_, reinterpret_cast<SOCKADDR*>(&saddr),
>                        static_cast<int>(len));
>   if (result != SOCKET_ERROR) {
>     state_ = CS_CONNECTED;
>@@ -423,22 +397,20 @@ int Win32Socket::SetOption(Option opt, int value) {
> }
> 
> int Win32Socket::Send(const void* buffer, size_t length) {
>-  int sent = ::send(socket_,
>-                    reinterpret_cast<const char*>(buffer),
>-                    static_cast<int>(length),
>-                    0);
>+  int sent = ::send(socket_, reinterpret_cast<const char*>(buffer),
>+                    static_cast<int>(length), 0);
>   UpdateLastError();
>   return sent;
> }
> 
>-int Win32Socket::SendTo(const void* buffer, size_t length,
>+int Win32Socket::SendTo(const void* buffer,
>+                        size_t length,
>                         const SocketAddress& addr) {
>   sockaddr_storage saddr;
>   size_t addr_len = addr.ToSockAddrStorage(&saddr);
>-  int sent = ::sendto(socket_, reinterpret_cast<const char*>(buffer),
>-                      static_cast<int>(length), 0,
>-                      reinterpret_cast<sockaddr*>(&saddr),
>-                      static_cast<int>(addr_len));
>+  int sent = ::sendto(
>+      socket_, reinterpret_cast<const char*>(buffer), static_cast<int>(length),
>+      0, reinterpret_cast<sockaddr*>(&saddr), static_cast<int>(addr_len));
>   UpdateLastError();
>   return sent;
> }
>@@ -447,8 +419,8 @@ int Win32Socket::Recv(void* buffer, size_t length, int64_t* timestamp) {
>   if (timestamp) {
>     *timestamp = -1;
>   }
>-  int received = ::recv(socket_, static_cast<char*>(buffer),
>-                        static_cast<int>(length), 0);
>+  int received =
>+      ::recv(socket_, static_cast<char*>(buffer), static_cast<int>(length), 0);
>   UpdateLastError();
>   if (closing_ && received <= static_cast<int>(length))
>     PostClosed();
>@@ -464,9 +436,9 @@ int Win32Socket::RecvFrom(void* buffer,
>   }
>   sockaddr_storage saddr;
>   socklen_t addr_len = sizeof(saddr);
>-  int received = ::recvfrom(socket_, static_cast<char*>(buffer),
>-                            static_cast<int>(length), 0,
>-                            reinterpret_cast<sockaddr*>(&saddr), &addr_len);
>+  int received =
>+      ::recvfrom(socket_, static_cast<char*>(buffer), static_cast<int>(length),
>+                 0, reinterpret_cast<sockaddr*>(&saddr), &addr_len);
>   UpdateLastError();
>   if (received != SOCKET_ERROR)
>     SocketAddressFromSockAddrStorage(saddr, out_addr);
>@@ -540,8 +512,8 @@ bool Win32Socket::SetAsync(int events) {
>   }
> 
>   // start the async select
>-  if (WSAAsyncSelect(socket_, sink_->handle(), WM_SOCKETNOTIFY, events)
>-      == SOCKET_ERROR) {
>+  if (WSAAsyncSelect(socket_, sink_->handle(), WM_SOCKETNOTIFY, events) ==
>+      SOCKET_ERROR) {
>     UpdateLastError();
>     Close();
>     return false;
>@@ -562,8 +534,8 @@ bool Win32Socket::HandleClosed(int close_error) {
> void Win32Socket::PostClosed() {
>   // If we see that the buffer is indeed drained, then send the close.
>   closing_ = false;
>-  ::PostMessage(sink_->handle(), WM_SOCKETNOTIFY,
>-                socket_, WSAMAKESELECTREPLY(FD_CLOSE, close_error_));
>+  ::PostMessage(sink_->handle(), WM_SOCKETNOTIFY, socket_,
>+                WSAMAKESELECTREPLY(FD_CLOSE, close_error_));
> }
> 
> void Win32Socket::UpdateLastError() {
>@@ -696,9 +668,7 @@ static UINT s_wm_wakeup_id = 0;
> const TCHAR Win32SocketServer::kWindowName[] = L"libjingle Message Window";
> 
> Win32SocketServer::Win32SocketServer()
>-    : wnd_(this),
>-      posted_(false),
>-      hdlg_(nullptr) {
>+    : wnd_(this), posted_(false), hdlg_(nullptr) {
>   if (s_wm_wakeup_id == 0)
>     s_wm_wakeup_id = RegisterWindowMessage(L"WM_WAKEUP");
>   if (!wnd_.Create(nullptr, kWindowName, 0, 0, 0, 0, 0, 0)) {
>@@ -713,18 +683,10 @@ Win32SocketServer::~Win32SocketServer() {
>   }
> }
> 
>-Socket* Win32SocketServer::CreateSocket(int type) {
>-  return CreateSocket(AF_INET, type);
>-}
>-
> Socket* Win32SocketServer::CreateSocket(int family, int type) {
>   return CreateAsyncSocket(family, type);
> }
> 
>-AsyncSocket* Win32SocketServer::CreateAsyncSocket(int type) {
>-  return CreateAsyncSocket(AF_INET, type);
>-}
>-
> AsyncSocket* Win32SocketServer::CreateAsyncSocket(int family, int type) {
>   Win32Socket* socket = new Win32Socket;
>   if (socket->CreateT(family, type)) {
>@@ -755,7 +717,7 @@ bool Win32SocketServer::Wait(int cms, bool process_io) {
>       if (b == -1) {
>         RTC_LOG_GLE(LS_ERROR) << "GetMessage failed.";
>         return false;
>-      } else if(b) {
>+      } else if (b) {
>         if (!hdlg_ || !IsDialogMessage(hdlg_, &msg)) {
>           TranslateMessage(&msg);
>           DispatchMessage(&msg);
>@@ -821,8 +783,10 @@ void Win32SocketServer::Pump() {
>   }
> }
> 
>-bool Win32SocketServer::MessageWindow::OnMessage(UINT wm, WPARAM wp,
>-                                                 LPARAM lp, LRESULT& lr) {
>+bool Win32SocketServer::MessageWindow::OnMessage(UINT wm,
>+                                                 WPARAM wp,
>+                                                 LPARAM lp,
>+                                                 LRESULT& lr) {
>   bool handled = false;
>   if (wm == s_wm_wakeup_id || (wm == WM_TIMER && wp == 1)) {
>     ss_->Pump();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win32socketserver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win32socketserver.h
>index ce29bc50031..0b11d644ece 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win32socketserver.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win32socketserver.h
>@@ -75,17 +75,17 @@ class Win32Socket : public AsyncSocket {
>   SOCKET socket_;
>   int error_;
>   ConnState state_;
>-  SocketAddress addr_;         // address that we connected to (see DoConnect)
>+  SocketAddress addr_;  // address that we connected to (see DoConnect)
>   uint32_t connect_time_;
>   bool closing_;
>   int close_error_;
> 
>   class EventSink;
>   friend class EventSink;
>-  EventSink * sink_;
>+  EventSink* sink_;
> 
>   struct DnsLookup;
>-  DnsLookup * dns_;
>+  DnsLookup* dns_;
> };
> 
> ///////////////////////////////////////////////////////////////////////////////
>@@ -97,15 +97,10 @@ class Win32SocketServer : public SocketServer {
>   Win32SocketServer();
>   ~Win32SocketServer() override;
> 
>-  void set_modeless_dialog(HWND hdlg) {
>-    hdlg_ = hdlg;
>-  }
>+  void set_modeless_dialog(HWND hdlg) { hdlg_ = hdlg; }
> 
>   // SocketServer Interface
>-  Socket* CreateSocket(int type) override;
>   Socket* CreateSocket(int family, int type) override;
>-
>-  AsyncSocket* CreateAsyncSocket(int type) override;
>   AsyncSocket* CreateAsyncSocket(int family, int type) override;
> 
>   void SetMessageQueue(MessageQueue* queue) override;
>@@ -120,13 +115,14 @@ class Win32SocketServer : public SocketServer {
>   class MessageWindow : public Win32Window {
>    public:
>     explicit MessageWindow(Win32SocketServer* ss) : ss_(ss) {}
>+
>    private:
>     bool OnMessage(UINT msg, WPARAM wp, LPARAM lp, LRESULT& result) override;
>     Win32SocketServer* ss_;
>   };
> 
>   static const TCHAR kWindowName[];
>-  MessageQueue *message_queue_;
>+  MessageQueue* message_queue_;
>   MessageWindow wnd_;
>   CriticalSection cs_;
>   bool posted_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win32window.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win32window.cc
>index d71c68e12b2..4290894d266 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win32window.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win32window.cc
>@@ -28,8 +28,14 @@ Win32Window::~Win32Window() {
>   RTC_DCHECK(nullptr == wnd_);
> }
> 
>-bool Win32Window::Create(HWND parent, const wchar_t* title, DWORD style,
>-                         DWORD exstyle, int x, int y, int cx, int cy) {
>+bool Win32Window::Create(HWND parent,
>+                         const wchar_t* title,
>+                         DWORD style,
>+                         DWORD exstyle,
>+                         int x,
>+                         int y,
>+                         int cx,
>+                         int cy) {
>   if (wnd_) {
>     // Window already exists.
>     return false;
>@@ -37,7 +43,7 @@ bool Win32Window::Create(HWND parent, const wchar_t* title, DWORD style,
> 
>   if (!window_class_) {
>     if (!GetModuleHandleEx(GET_MODULE_HANDLE_EX_FLAG_FROM_ADDRESS |
>-                           GET_MODULE_HANDLE_EX_FLAG_UNCHANGED_REFCOUNT,
>+                               GET_MODULE_HANDLE_EX_FLAG_UNCHANGED_REFCOUNT,
>                            reinterpret_cast<LPCWSTR>(&Win32Window::WndProc),
>                            &instance_)) {
>       RTC_LOG_GLE(LS_ERROR) << "GetModuleHandleEx failed";
>@@ -74,15 +80,17 @@ void Win32Window::Shutdown() {
>   }
> }
> 
>-bool Win32Window::OnMessage(UINT uMsg, WPARAM wParam, LPARAM lParam,
>+bool Win32Window::OnMessage(UINT uMsg,
>+                            WPARAM wParam,
>+                            LPARAM lParam,
>                             LRESULT& result) {
>   switch (uMsg) {
>-  case WM_CLOSE:
>-    if (!OnClose()) {
>-      result = 0;
>-      return true;
>-    }
>-    break;
>+    case WM_CLOSE:
>+      if (!OnClose()) {
>+        result = 0;
>+        return true;
>+      }
>+      break;
>   }
>   return false;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win32window.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win32window.h
>index fa026d118ae..e96aa350fb4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win32window.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/win32window.h
>@@ -28,22 +28,32 @@ class Win32Window {
> 
>   HWND handle() const { return wnd_; }
> 
>-  bool Create(HWND parent, const wchar_t* title, DWORD style, DWORD exstyle,
>-              int x, int y, int cx, int cy);
>+  bool Create(HWND parent,
>+              const wchar_t* title,
>+              DWORD style,
>+              DWORD exstyle,
>+              int x,
>+              int y,
>+              int cx,
>+              int cy);
>   void Destroy();
> 
>   // Call this when your DLL unloads.
>   static void Shutdown();
> 
>  protected:
>-  virtual bool OnMessage(UINT uMsg, WPARAM wParam, LPARAM lParam,
>+  virtual bool OnMessage(UINT uMsg,
>+                         WPARAM wParam,
>+                         LPARAM lParam,
>                          LRESULT& result);
> 
>   virtual bool OnClose();
>   virtual void OnNcDestroy();
> 
>  private:
>-  static LRESULT CALLBACK WndProc(HWND hwnd, UINT uMsg, WPARAM wParam,
>+  static LRESULT CALLBACK WndProc(HWND hwnd,
>+                                  UINT uMsg,
>+                                  WPARAM wParam,
>                                   LPARAM lParam);
> 
>   HWND wnd_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/zero_memory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/zero_memory.cc
>new file mode 100644
>index 00000000000..b9c5b380ac9
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/zero_memory.cc
>@@ -0,0 +1,38 @@
>+/*
>+ *  Copyright 2017 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#if defined(WEBRTC_WIN)
>+#include <windows.h>
>+#else
>+#include <string.h>
>+#endif
>+
>+#include "rtc_base/checks.h"
>+#include "rtc_base/zero_memory.h"
>+
>+namespace rtc {
>+
>+// Code and comment taken from "OPENSSL_cleanse" of BoringSSL.
>+void ExplicitZeroMemory(void* ptr, size_t len) {
>+  RTC_DCHECK(ptr || !len);
>+#if defined(WEBRTC_WIN)
>+  SecureZeroMemory(ptr, len);
>+#else
>+  memset(ptr, 0, len);
>+#if !defined(__pnacl__)
>+  /* As best as we can tell, this is sufficient to break any optimisations that
>+     might try to eliminate "superfluous" memsets. If there's an easy way to
>+     detect memset_s, it would be better to use that. */
>+  __asm__ __volatile__("" : : "r"(ptr) : "memory");  // NOLINT
>+#endif
>+#endif  // !WEBRTC_WIN
>+}
>+
>+}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/zero_memory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/zero_memory.h
>new file mode 100644
>index 00000000000..cb4646c1fe4
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/zero_memory.h
>@@ -0,0 +1,33 @@
>+/*
>+ *  Copyright 2017 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef RTC_BASE_ZERO_MEMORY_H_
>+#define RTC_BASE_ZERO_MEMORY_H_
>+
>+#include <type_traits>
>+
>+#include "api/array_view.h"
>+
>+namespace rtc {
>+
>+// Fill memory with zeros in a way that the compiler doesn't optimize it away
>+// even if the pointer is not used afterwards.
>+void ExplicitZeroMemory(void* ptr, size_t len);
>+
>+template <typename T,
>+          typename std::enable_if<!std::is_const<T>::value &&
>+                                  std::is_trivial<T>::value>::type* = nullptr>
>+void ExplicitZeroMemory(rtc::ArrayView<T> a) {
>+  ExplicitZeroMemory(a.data(), a.size());
>+}
>+
>+}  // namespace rtc
>+
>+#endif  // RTC_BASE_ZERO_MEMORY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/zero_memory_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/zero_memory_unittest.cc
>new file mode 100644
>index 00000000000..0f496705e7d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/zero_memory_unittest.cc
>@@ -0,0 +1,52 @@
>+/*
>+ *  Copyright 2017 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "rtc_base/zero_memory.h"
>+
>+#include "api/array_view.h"
>+#include "rtc_base/buffer.h"
>+#include "rtc_base/gunit.h"
>+
>+namespace rtc {
>+
>+TEST(ZeroMemoryTest, TestZeroMemory) {
>+  static const size_t kBufferSize = 32;
>+  uint8_t buffer[kBufferSize];
>+  for (size_t i = 0; i < kBufferSize; i++) {
>+    buffer[i] = static_cast<uint8_t>(i + 1);
>+  }
>+  ExplicitZeroMemory(buffer, sizeof(buffer));
>+  for (size_t i = 0; i < kBufferSize; i++) {
>+    EXPECT_EQ(buffer[i], 0);
>+  }
>+}
>+
>+TEST(ZeroMemoryTest, TestZeroArrayView) {
>+  static const size_t kBufferSize = 32;
>+  uint8_t buffer[kBufferSize];
>+  for (size_t i = 0; i < kBufferSize; i++) {
>+    buffer[i] = static_cast<uint8_t>(i + 1);
>+  }
>+  ExplicitZeroMemory(rtc::ArrayView<uint8_t>(buffer, sizeof(buffer)));
>+  for (size_t i = 0; i < kBufferSize; i++) {
>+    EXPECT_EQ(buffer[i], 0);
>+  }
>+}
>+
>+// While this test doesn't actually test anything, it can be used to check
>+// the compiler output to make sure the call to "ExplicitZeroMemory" is not
>+// optimized away.
>+TEST(ZeroMemoryTest, TestZeroMemoryUnused) {
>+  static const size_t kBufferSize = 32;
>+  uint8_t buffer[kBufferSize];
>+  ExplicitZeroMemory(buffer, sizeof(buffer));
>+}
>+
>+}  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/BUILD.gn
>index 59ee4eac813..99f1ae6b4f9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/BUILD.gn
>@@ -13,34 +13,35 @@ group("rtc_tools") {
>   # This target shall build all targets in tools/.
>   testonly = true
> 
>-  public_deps = [
>+  deps = [
>     ":command_line_parser",
>     ":frame_analyzer",
>     ":video_quality_analysis",
>   ]
>   if (!build_with_chromium) {
>-    public_deps += [
>+    deps += [
>       ":frame_editor",
>       ":psnr_ssim_analyzer",
>       ":rgba_to_i420_converter",
>     ]
>     if (rtc_include_internal_audio_device) {
>-      public_deps += [ ":force_mic_volume_max" ]
>+      deps += [ ":force_mic_volume_max" ]
>     }
>     if (rtc_enable_protobuf) {
>-      public_deps += [ ":chart_proto" ]
>+      deps += [ ":chart_proto" ]
>     }
>   }
> 
>   if (rtc_include_tests) {
>-    public_deps += [
>+    deps += [
>       ":activity_metric",
>       ":tools_unittests",
>     ]
>     if (rtc_enable_protobuf) {
>-      public_deps += [
>+      deps += [
>         ":event_log_visualizer",
>         ":rtp_analyzer",
>+        ":unpack_aecdump",
>         "network_tester",
>       ]
>     }
>@@ -63,13 +64,15 @@ rtc_static_library("video_quality_analysis") {
>     "frame_analyzer/video_quality_analysis.cc",
>     "frame_analyzer/video_quality_analysis.h",
>   ]
>-
>-  public_deps = [
>+  deps = [
>     "../common_video",
>+    "../test:perf_test",
>+    "//third_party/libyuv",
>   ]
> }
> 
> rtc_executable("frame_analyzer") {
>+  visibility = [ "*" ]
>   sources = [
>     "frame_analyzer/frame_analyzer.cc",
>   ]
>@@ -77,6 +80,7 @@ rtc_executable("frame_analyzer") {
>   deps = [
>     ":command_line_parser",
>     ":video_quality_analysis",
>+    "../test:perf_test",
>     "//build/win:default_exe_manifest",
>   ]
> }
>@@ -120,6 +124,7 @@ if (!build_with_chromium) {
>   }
> 
>   rtc_executable("rgba_to_i420_converter") {
>+    visibility = [ "*" ]
>     sources = [
>       "converter/converter.cc",
>       "converter/converter.h",
>@@ -130,6 +135,7 @@ if (!build_with_chromium) {
>       ":command_line_parser",
>       "../common_video",
>       "//build/win:default_exe_manifest",
>+      "//third_party/libyuv",
>     ]
>   }
> 
>@@ -139,14 +145,6 @@ if (!build_with_chromium) {
>       "frame_editing/frame_editing_lib.h",
>     ]
> 
>-    # TODO(jschuh): Bug 1348: fix this warning.
>-    configs += [ "//build/config/compiler:no_size_t_to_int_warning" ]
>-
>-    if (!build_with_chromium && is_clang) {
>-      # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>-      suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-    }
>-
>     deps = [
>       "..:webrtc_common",
>       "../common_video",
>@@ -172,13 +170,9 @@ if (!build_with_chromium) {
>         "force_mic_volume_max/force_mic_volume_max.cc",
>       ]
> 
>-      if (!build_with_chromium && is_clang) {
>-        # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>-        suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-      }
>-
>       deps = [
>         "../modules/audio_device",
>+        "../modules/audio_device:audio_device_impl",
>         "../system_wrappers:system_wrappers_default",
>         "//build/win:default_exe_manifest",
>       ]
>@@ -194,6 +188,8 @@ if (!build_with_chromium) {
>     }
> 
>     rtc_static_library("event_log_visualizer_utils") {
>+      visibility = [ "*" ]
>+      allow_poison = [ "audio_codecs" ]  # TODO(bugs.webrtc.org/8396): Remove.
>       sources = [
>         "event_log_visualizer/analyzer.cc",
>         "event_log_visualizer/analyzer.h",
>@@ -203,6 +199,7 @@ if (!build_with_chromium) {
>         "event_log_visualizer/plot_protobuf.h",
>         "event_log_visualizer/plot_python.cc",
>         "event_log_visualizer/plot_python.h",
>+        "event_log_visualizer/triage_notifications.h",
>       ]
>       if (!build_with_chromium && is_clang) {
>         # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>@@ -210,27 +207,33 @@ if (!build_with_chromium) {
>       }
>       defines = [ "ENABLE_RTC_EVENT_LOG" ]
>       deps = [
>+        ":chart_proto",
>+        "../:webrtc_common",
>         "../call:call_interfaces",
>         "../call:video_stream_api",
>-        "../logging:rtc_event_log_impl",
>+        "../logging:rtc_event_log_api",
>+        "../logging:rtc_event_log_impl_base",
>         "../logging:rtc_event_log_parser",
>+        "../logging:rtc_stream_config",
>         "../modules:module_api",
>         "../modules/audio_coding:ana_debug_dump_proto",
>+        "../modules/audio_coding:audio_network_adaptor",
>         "../modules/audio_coding:neteq_tools",
>+        "../modules/rtp_rtcp:rtp_rtcp_format",
>+        "../rtc_base:checks",
>         "../rtc_base:rtc_base_approved",
>         "../rtc_base:rtc_numerics",
>+        "../rtc_base:stringutils",
> 
>         # TODO(kwiberg): Remove this dependency.
>         "../api/audio_codecs:audio_codecs_api",
>         "../modules/congestion_controller",
>+        "../modules/congestion_controller/goog_cc:delay_based_bwe",
>+        "../modules/congestion_controller/goog_cc:estimators",
>         "../modules/pacing",
>         "../modules/rtp_rtcp",
>         "../system_wrappers:system_wrappers_default",
>-        "//build/config:exe_and_shlib_deps",
>-      ]
>-      public_deps = [
>-        ":chart_proto",
>-        "../logging:rtc_event_log_parser",
>+        "//third_party/abseil-cpp/absl/memory",
>       ]
>     }
>   }
>@@ -252,8 +255,12 @@ if (rtc_include_tests) {
>       defines = [ "ENABLE_RTC_EVENT_LOG" ]
>       deps = [
>         ":event_log_visualizer_utils",
>+        "../logging:rtc_event_log_parser",
>+        "../rtc_base:protobuf_utils",
>         "../rtc_base:rtc_base_approved",
>+        "../system_wrappers:field_trial_default",
>         "../test:field_trial",
>+        "../test:fileutils",
>         "../test:test_support",
>       ]
>     }
>@@ -265,15 +272,12 @@ if (rtc_include_tests) {
>       "agc/activity_metric.cc",
>     ]
> 
>-    if (!build_with_chromium && is_clang) {
>-      # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>-      suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-    }
>-
>     deps = [
>-      "../modules:module_api",
>-      "../modules/audio_processing",
>+      "../api/audio:audio_frame_api",
>+      "../modules/audio_processing/agc:level_estimation",
>+      "../modules/audio_processing/vad",
>       "../rtc_base:rtc_base_approved",
>+      "../rtc_base:safe_minmax",
>       "../system_wrappers:metrics_default",
>       "../test:test_support",
>       "//build/win:default_exe_manifest",
>@@ -308,9 +312,6 @@ if (rtc_include_tests) {
>       "simple_command_line_parser_unittest.cc",
>     ]
> 
>-    # TODO(jschuh): Bug 1348: fix this warning.
>-    configs += [ "//build/config/compiler:no_size_t_to_int_warning" ]
>-
>     if (!build_with_chromium && is_clang) {
>       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>@@ -321,9 +322,13 @@ if (rtc_include_tests) {
>       ":frame_editing_lib",
>       ":reference_less_video_analysis_lib",
>       ":video_quality_analysis",
>+      "../common_video:common_video",
>       "../rtc_base",
>+      "../rtc_base:checks",
>+      "../test:fileutils",
>       "../test:test_main",
>       "//testing/gtest",
>+      "//third_party/abseil-cpp/absl/memory",
>     ]
> 
>     if (rtc_enable_protobuf) {
>@@ -354,6 +359,27 @@ if (rtc_include_tests) {
>       deps = [
>         "../logging:rtc_event_log_proto",
>       ]
>-    }
>+    }  # rtp_analyzer
>+
>+    rtc_executable("unpack_aecdump") {
>+      visibility = [ "*" ]
>+      testonly = true
>+      sources = [
>+        "unpack_aecdump/unpack.cc",
>+      ]
>+
>+      deps = [
>+        "..:webrtc_common",
>+        "../common_audio",
>+        "../modules/audio_processing",
>+        "../modules/audio_processing:audioproc_debug_proto",
>+        "../modules/audio_processing:audioproc_debug_proto",
>+        "../modules/audio_processing:audioproc_protobuf_utils",
>+        "../modules/audio_processing:audioproc_test_utils",
>+        "../rtc_base:protobuf_utils",
>+        "../rtc_base:rtc_base_approved",
>+        "../system_wrappers:system_wrappers_default",
>+      ]
>+    }  # unpack_aecdump
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/DEPS b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/DEPS
>index c21ca8f2548..d9b162cd15f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/DEPS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/DEPS
>@@ -12,5 +12,6 @@ include_rules = [
>   "+modules/rtp_rtcp",
>   "+system_wrappers",
>   "+p2p",
>+  "+third_party/libyuv",
> ]
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/agc/activity_metric.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/agc/activity_metric.cc
>index 3c65d02b4ae..b4ed3facb35 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/agc/activity_metric.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/agc/activity_metric.cc
>@@ -8,7 +8,6 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-
> #include <math.h>
> #include <stdio.h>
> #include <stdlib.h>
>@@ -16,6 +15,7 @@
> #include <algorithm>
> #include <memory>
> 
>+#include "api/audio/audio_frame.h"
> #include "modules/audio_processing/agc/agc.h"
> #include "modules/audio_processing/agc/loudness_histogram.h"
> #include "modules/audio_processing/agc/utility.h"
>@@ -23,7 +23,6 @@
> #include "modules/audio_processing/vad/pitch_based_vad.h"
> #include "modules/audio_processing/vad/standalone_vad.h"
> #include "modules/audio_processing/vad/vad_audio_proc.h"
>-#include "modules/include/module_common_types.h"
> #include "rtc_base/flags.h"
> #include "rtc_base/numerics/safe_minmax.h"
> #include "test/gtest.h"
>@@ -32,21 +31,30 @@ static const int kAgcAnalWindowSamples = 100;
> static const float kDefaultActivityThreshold = 0.3f;
> 
> DEFINE_bool(standalone_vad, true, "enable stand-alone VAD");
>-DEFINE_string(true_vad, "", "name of a file containing true VAD in 'int'"
>+DEFINE_string(true_vad,
>+              "",
>+              "name of a file containing true VAD in 'int'"
>               " format");
>-DEFINE_string(video_vad, "", "name of a file containing video VAD (activity"
>+DEFINE_string(video_vad,
>+              "",
>+              "name of a file containing video VAD (activity"
>               " probabilities) in double format. One activity per 10ms is"
>               " required. If no file is given the video information is not"
>               " incorporated. Negative activity is interpreted as video is"
>               " not adapted and the statistics are not computed during"
>               " the learning phase. Note that the negative video activities"
>               " are ONLY allowed at the beginning.");
>-DEFINE_string(result, "", "name of a file to write the results. The results"
>+DEFINE_string(result,
>+              "",
>+              "name of a file to write the results. The results"
>               " will be appended to the end of the file. This is optional.");
>-DEFINE_string(audio_content, "", "name of a file where audio content is written"
>+DEFINE_string(audio_content,
>+              "",
>+              "name of a file where audio content is written"
>               " to, in double format.");
>-DEFINE_float(activity_threshold, kDefaultActivityThreshold,
>-              "Activity threshold");
>+DEFINE_float(activity_threshold,
>+             kDefaultActivityThreshold,
>+             "Activity threshold");
> DEFINE_bool(help, false, "prints this message");
> 
> namespace webrtc {
>@@ -61,8 +69,8 @@ namespace webrtc {
> static void DitherSilence(AudioFrame* frame) {
>   ASSERT_EQ(1u, frame->num_channels_);
>   const double kRmsSilence = 5;
>-  const double sum_squared_silence = kRmsSilence * kRmsSilence *
>-      frame->samples_per_channel_;
>+  const double sum_squared_silence =
>+      kRmsSilence * kRmsSilence * frame->samples_per_channel_;
>   double sum_squared = 0;
>   int16_t* frame_data = frame->mutable_data();
>   for (size_t n = 0; n < frame->samples_per_channel_; n++)
>@@ -97,21 +105,18 @@ class AgcStat {
>     audio_content_fid_ = audio_content_fid;
>   }
> 
>-  int AddAudio(const AudioFrame& frame, double p_video,
>-               int* combined_vad) {
>+  int AddAudio(const AudioFrame& frame, double p_video, int* combined_vad) {
>     if (frame.num_channels_ != 1 ||
>-        frame.samples_per_channel_ !=
>-            kSampleRateHz / 100 ||
>-            frame.sample_rate_hz_ != kSampleRateHz)
>+        frame.samples_per_channel_ != kSampleRateHz / 100 ||
>+        frame.sample_rate_hz_ != kSampleRateHz)
>       return -1;
>     video_vad_[video_index_++] = p_video;
>     AudioFeatures features;
>     const int16_t* frame_data = frame.data();
>-    audio_processing_->ExtractFeatures(
>-        frame_data, frame.samples_per_channel_, &features);
>+    audio_processing_->ExtractFeatures(frame_data, frame.samples_per_channel_,
>+                                       &features);
>     if (FLAG_standalone_vad) {
>-      standalone_vad_->AddAudio(frame_data,
>-                                frame.samples_per_channel_);
>+      standalone_vad_->AddAudio(frame_data, frame.samples_per_channel_);
>     }
>     if (features.num_frames > 0) {
>       double p[kMaxNumFrames] = {0.5, 0.5, 0.5, 0.5};
>@@ -145,9 +150,7 @@ class AgcStat {
>     return static_cast<int>(features.num_frames);
>   }
> 
>-  void Reset() {
>-    audio_content_->Reset();
>-  }
>+  void Reset() { audio_content_->Reset(); }
> 
>   void SetActivityThreshold(double activity_threshold) {
>     activity_threshold_ = activity_threshold;
>@@ -165,7 +168,6 @@ class AgcStat {
>   FILE* audio_content_fid_;
> };
> 
>-
> void void_main(int argc, char* argv[]) {
>   webrtc::AgcStat agc_stat;
> 
>@@ -178,10 +180,10 @@ void void_main(int argc, char* argv[]) {
> 
>   FILE* true_vad_fid = NULL;
>   ASSERT_GT(strlen(FLAG_true_vad), 0u) << "Specify the file containing true "
>-      "VADs using --true_vad flag.";
>+                                          "VADs using --true_vad flag.";
>   true_vad_fid = fopen(FLAG_true_vad, "rb");
>-  ASSERT_TRUE(true_vad_fid != NULL) << "Cannot open the active list " <<
>-      FLAG_true_vad;
>+  ASSERT_TRUE(true_vad_fid != NULL)
>+      << "Cannot open the active list " << FLAG_true_vad;
> 
>   FILE* results_fid = NULL;
>   if (strlen(FLAG_result) > 0) {
>@@ -199,11 +201,12 @@ void void_main(int argc, char* argv[]) {
>     }
>     // Open in append mode.
>     results_fid = fopen(FLAG_result, "a");
>-    ASSERT_TRUE(results_fid != NULL) << "Cannot open the file, " <<
>-              FLAG_result << ", to write the results.";
>+    ASSERT_TRUE(results_fid != NULL)
>+        << "Cannot open the file, " << FLAG_result << ", to write the results.";
>     // Write the header if required.
>     if (write_header) {
>-      fprintf(results_fid, "%% Total Active,  Misdetection,  "
>+      fprintf(results_fid,
>+              "%% Total Active,  Misdetection,  "
>               "Total inactive,  False Positive,  On-sets,  Missed segments,  "
>               "Average response\n");
>     }
>@@ -212,8 +215,9 @@ void void_main(int argc, char* argv[]) {
>   FILE* video_vad_fid = NULL;
>   if (strlen(FLAG_video_vad) > 0) {
>     video_vad_fid = fopen(FLAG_video_vad, "rb");
>-    ASSERT_TRUE(video_vad_fid != NULL) <<  "Cannot open the file, " <<
>-              FLAG_video_vad << " to read video-based VAD decisions.\n";
>+    ASSERT_TRUE(video_vad_fid != NULL)
>+        << "Cannot open the file, " << FLAG_video_vad
>+        << " to read video-based VAD decisions.\n";
>   }
> 
>   // AgsStat will be the owner of this file and will close it at its
>@@ -221,8 +225,9 @@ void void_main(int argc, char* argv[]) {
>   FILE* audio_content_fid = NULL;
>   if (strlen(FLAG_audio_content) > 0) {
>     audio_content_fid = fopen(FLAG_audio_content, "wb");
>-    ASSERT_TRUE(audio_content_fid != NULL) << "Cannot open file, " <<
>-              FLAG_audio_content << " to write audio-content.\n";
>+    ASSERT_TRUE(audio_content_fid != NULL)
>+        << "Cannot open file, " << FLAG_audio_content
>+        << " to write audio-content.\n";
>     agc_stat.set_audio_content_file(audio_content_fid);
>   }
> 
>@@ -230,8 +235,8 @@ void void_main(int argc, char* argv[]) {
>   frame.num_channels_ = 1;
>   frame.sample_rate_hz_ = 16000;
>   frame.samples_per_channel_ = frame.sample_rate_hz_ / 100;
>-  const size_t kSamplesToRead = frame.num_channels_ *
>-      frame.samples_per_channel_;
>+  const size_t kSamplesToRead =
>+      frame.num_channels_ * frame.samples_per_channel_;
> 
>   agc_stat.SetActivityThreshold(FLAG_activity_threshold);
> 
>@@ -260,16 +265,17 @@ void void_main(int argc, char* argv[]) {
>                         true_vad_fid))
>         << "Size mismatch between True-VAD and the PCM file.\n";
>     if (video_vad_fid != NULL) {
>-      ASSERT_EQ(1u, fread(&p_video, sizeof(p_video), 1, video_vad_fid)) <<
>-          "Not enough video-based VAD probabilities.";
>+      ASSERT_EQ(1u, fread(&p_video, sizeof(p_video), 1, video_vad_fid))
>+          << "Not enough video-based VAD probabilities.";
>     }
> 
>     // Negative video activity indicates that the video-based VAD is not yet
>     // adapted. Disregards the learning phase in statistics.
>     if (p_video < 0) {
>       if (video_adapted) {
>-        fprintf(stderr, "Negative video probabilities ONLY allowed at the "
>-            "beginning of the sequence, not in the middle.\n");
>+        fprintf(stderr,
>+                "Negative video probabilities ONLY allowed at the "
>+                "beginning of the sequence, not in the middle.\n");
>         exit(1);
>       }
>       continue;
>@@ -337,23 +343,15 @@ void void_main(int argc, char* argv[]) {
> 
>   if (results_fid != NULL) {
>     fprintf(results_fid, "%4d  %4d  %4d  %4d  %4d  %4d  %4.0f %4.0f\n",
>-            total_active,
>-            total_missed_detection,
>-            total_passive,
>-            total_false_positive,
>-            num_onsets,
>-            num_not_adapted,
>+            total_active, total_missed_detection, total_passive,
>+            total_false_positive, num_onsets, num_not_adapted,
>             static_cast<float>(onset_adaptation) / (num_onsets + 1e-12),
>             static_cast<float>(total_false_positive_duration) /
>-            (total_passive + 1e-12));
>+                (total_passive + 1e-12));
>   }
>-  fprintf(stdout, "%4d %4d %4d %4d %4d %4d %4.0f %4.0f\n",
>-          total_active,
>-          total_missed_detection,
>-          total_passive,
>-          total_false_positive,
>-          num_onsets,
>-          num_not_adapted,
>+  fprintf(stdout, "%4d %4d %4d %4d %4d %4d %4.0f %4.0f\n", total_active,
>+          total_missed_detection, total_passive, total_false_positive,
>+          num_onsets, num_not_adapted,
>           static_cast<float>(onset_adaptation) / (num_onsets + 1e-12),
>           static_cast<float>(total_false_positive_duration) /
>               (total_passive + 1e-12));
>@@ -373,16 +371,18 @@ void void_main(int argc, char* argv[]) {
> int main(int argc, char* argv[]) {
>   if (argc == 1) {
>     // Print usage information.
>-    std::cout <<
>-      "\nCompute the number of misdetected and false-positive frames. Not\n"
>-      " that for each frame of audio (10 ms) there should be one true\n"
>-      " activity. If any video-based activity is given, there should also be\n"
>-      " one probability per frame.\n"
>-      "Run with --help for more details on available flags.\n"
>-      "\nUsage:\n\n"
>-      "activity_metric input_pcm [options]\n"
>-      "where 'input_pcm' is the input audio sampled at 16 kHz in 16 bits "
>-      "format.\n\n";
>+    std::cout
>+        << "\nCompute the number of misdetected and false-positive frames. "
>+           "Not\n"
>+           " that for each frame of audio (10 ms) there should be one true\n"
>+           " activity. If any video-based activity is given, there should also "
>+           "be\n"
>+           " one probability per frame.\n"
>+           "Run with --help for more details on available flags.\n"
>+           "\nUsage:\n\n"
>+           "activity_metric input_pcm [options]\n"
>+           "where 'input_pcm' is the input audio sampled at 16 kHz in 16 bits "
>+           "format.\n\n";
>     return 0;
>   }
>   rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/author_line_count.sh b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/author_line_count.sh
>old mode 100644
>new mode 100755
>index 8d67a6d8eae..f99b3c0f57c
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/author_line_count.sh
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/author_line_count.sh
>@@ -16,7 +16,7 @@ git log "$@" --pretty=format:%ae --shortstat \
>   | gawk '
> /^[^ ]/ {
>   /* Some author "email addresses" have a trailing @svn-id, strip that out. */
>-  author = gensub(/^([^@]*@[^@]*).*/, "\\1", "", $1);
>+  author = gensub(/^([^@]*@[^@]*).*/, "\\1", "g", $1);
> }
> /^ .*insertion/ { total[author] += $1 }
> /^ .*deletion/ { total[author] -= $1 }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/barcode_tools/barcode_decoder.py b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/barcode_tools/barcode_decoder.py
>old mode 100644
>new mode 100755
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/barcode_tools/barcode_encoder.py b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/barcode_tools/barcode_encoder.py
>old mode 100644
>new mode 100755
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/barcode_tools/build_zxing.py b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/barcode_tools/build_zxing.py
>old mode 100644
>new mode 100755
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/barcode_tools/yuv_cropper.py b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/barcode_tools/yuv_cropper.py
>old mode 100644
>new mode 100755
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/class_usage.sh b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/class_usage.sh
>old mode 100644
>new mode 100755
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/compare_videos.py b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/compare_videos.py
>old mode 100644
>new mode 100755
>index ce5bff30c13..40a2aabb327
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/compare_videos.py
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/compare_videos.py
>@@ -62,6 +62,8 @@ def _ParseArgs():
>                     help='Width of the YUV file\'s frames. Default: %default')
>   parser.add_option('--yuv_frame_height', type='int', default=480,
>                     help='Height of the YUV file\'s frames. Default: %default')
>+  parser.add_option('--chartjson_result_file', type='str', default=None,
>+                    help='Where to store perf results in chartjson format.')
>   options, _ = parser.parse_args()
> 
>   if options.stats_file:
>@@ -161,6 +163,8 @@ def main():
>     '--width=%d' % options.yuv_frame_width,
>     '--height=%d' % options.yuv_frame_height,
>   ]
>+  if options.chartjson_result_file:
>+    cmd.append('--chartjson_result_file=%s' % options.chartjson_result_file)
>   frame_analyzer = subprocess.Popen(cmd, stdin=_DevNull(),
>                                     stdout=sys.stdout, stderr=sys.stderr)
>   frame_analyzer.wait()
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/converter/converter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/converter/converter.cc
>index ee1316efc24..e36d4054518 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/converter/converter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/converter/converter.cc
>@@ -27,10 +27,7 @@
> namespace webrtc {
> namespace test {
> 
>-Converter::Converter(int width, int height)
>-    : width_(width),
>-      height_(height) {
>-}
>+Converter::Converter(int width, int height) : width_(width), height_(height) {}
> 
> bool Converter::ConvertRGBAToI420Video(std::string frames_dir,
>                                        std::string output_file_name,
>@@ -53,7 +50,7 @@ bool Converter::ConvertRGBAToI420Video(std::string frames_dir,
>   int v_plane_size = VPlaneSize();
>   uint8_t* dst_v = new uint8_t[v_plane_size];
> 
>-  int counter = 0;  // Counter to form frame names.
>+  int counter = 0;       // Counter to form frame names.
>   bool success = false;  // Is conversion successful.
> 
>   while (true) {
>@@ -79,16 +76,13 @@ bool Converter::ConvertRGBAToI420Video(std::string frames_dir,
>     }
> 
>     // Convert to I420 frame.
>-    libyuv::ABGRToI420(rgba_buffer, SrcStrideFrame(),
>-                       dst_y, DstStrideY(),
>-                       dst_u, DstStrideU(),
>-                       dst_v, DstStrideV(),
>-                       width_, height_);
>+    libyuv::ABGRToI420(rgba_buffer, SrcStrideFrame(), dst_y, DstStrideY(),
>+                       dst_u, DstStrideU(), dst_v, DstStrideV(), width_,
>+                       height_);
> 
>     // Add the I420 frame to the YUV video file.
>-    success = AddYUVToFile(dst_y, y_plane_size, dst_u, u_plane_size,
>-                           dst_v, v_plane_size, output_file);
>-
>+    success = AddYUVToFile(dst_y, y_plane_size, dst_u, u_plane_size, dst_v,
>+                           v_plane_size, output_file);
> 
>     if (!success) {
>       fprintf(stderr, "LibYUV error during RGBA to I420 frame conversion\n");
>@@ -125,14 +119,17 @@ bool Converter::AddYUVPlaneToFile(uint8_t* yuv_plane,
>   size_t bytes_written = fwrite(yuv_plane, 1, yuv_plane_size, file);
> 
>   if (bytes_written != static_cast<size_t>(yuv_plane_size)) {
>-    fprintf(stderr, "Number of bytes written (%d) doesn't match size of y plane"
>-            " (%d)\n", static_cast<int>(bytes_written), yuv_plane_size);
>+    fprintf(stderr,
>+            "Number of bytes written (%d) doesn't match size of y plane"
>+            " (%d)\n",
>+            static_cast<int>(bytes_written), yuv_plane_size);
>     return false;
>   }
>   return true;
> }
> 
>-bool Converter::ReadRGBAFrame(const char* input_file_name, int input_frame_size,
>+bool Converter::ReadRGBAFrame(const char* input_file_name,
>+                              int input_frame_size,
>                               unsigned char* buffer) {
>   FILE* input_file = fopen(input_file_name, "rb");
>   if (input_file == NULL) {
>@@ -157,7 +154,7 @@ std::string Converter::FindFullFileName(std::string dir_name,
>   return dir_name + SEPARATOR + file_name;
> }
> 
>-bool Converter:: FileExists(std::string file_name_to_check) {
>+bool Converter::FileExists(std::string file_name_to_check) {
>   struct STAT file_info;
>   int result = STAT(file_name_to_check.c_str(), &file_info);
>   return (result == 0);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/converter/converter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/converter/converter.h
>index e8bee0414e3..e3b3ea3f4cf 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/converter/converter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/converter/converter.h
>@@ -13,8 +13,8 @@
> 
> #include <string>
> 
>-#include "libyuv/compare.h"  // NOLINT
>-#include "libyuv/convert.h"  // NOLINT
>+#include "third_party/libyuv/include/libyuv/compare.h"
>+#include "third_party/libyuv/include/libyuv/convert.h"
> 
> namespace webrtc {
> namespace test {
>@@ -27,51 +27,36 @@ class Converter {
>   // Converts RGBA to YUV video. If the delete_frames argument is true, the
>   // method will delete the input frames after conversion.
>   bool ConvertRGBAToI420Video(std::string frames_dir,
>-                              std::string output_file_name, bool delete_frames);
>+                              std::string output_file_name,
>+                              bool delete_frames);
> 
>  private:
>-  int width_;  // Width of the video (respectively of the RGBA frames).
>+  int width_;   // Width of the video (respectively of the RGBA frames).
>   int height_;  // Height of the video (respectively of the RGBA frames).
> 
>   // Returns the size of the Y plane in bytes.
>-  int YPlaneSize() const {
>-    return width_*height_;
>-  }
>+  int YPlaneSize() const { return width_ * height_; }
> 
>   // Returns the size of the U plane in bytes.
>-  int UPlaneSize() const {
>-    return ((width_+1)/2)*((height_)/2);
>-  }
>+  int UPlaneSize() const { return ((width_ + 1) / 2) * ((height_) / 2); }
> 
>   // Returns the size of the V plane in bytes.
>-  int VPlaneSize() const {
>-    return ((width_+1)/2)*((height_)/2);
>-  }
>+  int VPlaneSize() const { return ((width_ + 1) / 2) * ((height_) / 2); }
> 
>   // Returns the number of bytes per row in the RGBA frame.
>-  int SrcStrideFrame() const {
>-    return width_*4;
>-  }
>+  int SrcStrideFrame() const { return width_ * 4; }
> 
>   // Returns the number of bytes in the Y plane.
>-  int DstStrideY() const {
>-    return width_;
>-  }
>+  int DstStrideY() const { return width_; }
> 
>   // Returns the number of bytes in the U plane.
>-  int DstStrideU() const {
>-    return (width_+1)/2;
>-  }
>+  int DstStrideU() const { return (width_ + 1) / 2; }
> 
>   // Returns the number of bytes in the V plane.
>-  int DstStrideV() const {
>-    return (width_+1)/2;
>-  }
>+  int DstStrideV() const { return (width_ + 1) / 2; }
> 
>   // Returns the size in bytes of the input RGBA frames.
>-  int InputFrameSize() const {
>-    return width_*height_*4;
>-  }
>+  int InputFrameSize() const { return width_ * height_ * 4; }
> 
>   // Writes the Y, U and V (in this order) planes to the file, thus adding a
>   // raw YUV frame to the file.
>@@ -88,7 +73,8 @@ class Converter {
> 
>   // Reads a RGBA frame from input_file_name with input_frame_size size in bytes
>   // into the buffer.
>-  bool ReadRGBAFrame(const char* input_file_name, int input_frame_size,
>+  bool ReadRGBAFrame(const char* input_file_name,
>+                     int input_frame_size,
>                      unsigned char* buffer);
> 
>   // Finds the full path name of the file - concatenates the directory and file
>@@ -99,7 +85,7 @@ class Converter {
>   bool FileExists(std::string file_name_to_check);
> 
>   // Returns the name of the file in the form frame_<number>, where <number> is
>-    // 4 zero padded (i.e. frame_0000, frame_0001, etc.).
>+  // 4 zero padded (i.e. frame_0000, frame_0001, etc.).
>   std::string FormFrameName(int width, int number);
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/converter/rgba_to_i420_converter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/converter/rgba_to_i420_converter.cc
>index 23342402a8c..9fe5141d517 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/converter/rgba_to_i420_converter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/converter/rgba_to_i420_converter.cc
>@@ -26,24 +26,27 @@
>  * --output_file=<output_yuv_file> --width=<width_of_input_frames>
>  * --height=<height_of_input_frames>
>  */
>-int main(int argc, char** argv) {
>+int main(int argc, char* argv[]) {
>   std::string program_name = argv[0];
>-  std::string usage = "Converts RGBA raw image files to I420 frames for YUV.\n"
>-    "Example usage:\n" + program_name +
>-    " --frames_dir=. --output_file=output.yuv --width=320 --height=240\n"
>-    "IMPORTANT: If you pass the --delete_frames command line parameter, the "
>-    "tool will delete the input frames after conversion.\n"
>-    "Command line flags:\n"
>-    "  - width(int): Width in pixels of the frames in the input file."
>-    " Default: -1\n"
>-    "  - height(int): Height in pixels of the frames in the input file."
>-    " Default: -1\n"
>-    "  - frames_dir(string): The path to the directory where the frames reside."
>-    " Default: .\n"
>-    "  - output_file(string): The output file to which frames are written."
>-    " Default: output.yuv\n"
>-    "  - delete_frames(bool): Whether or not to delete the input frames after"
>-    " the conversion. Default: false.\n";
>+  std::string usage =
>+      "Converts RGBA raw image files to I420 frames for YUV.\n"
>+      "Example usage:\n" +
>+      program_name +
>+      " --frames_dir=. --output_file=output.yuv --width=320 --height=240\n"
>+      "IMPORTANT: If you pass the --delete_frames command line parameter, the "
>+      "tool will delete the input frames after conversion.\n"
>+      "Command line flags:\n"
>+      "  - width(int): Width in pixels of the frames in the input file."
>+      " Default: -1\n"
>+      "  - height(int): Height in pixels of the frames in the input file."
>+      " Default: -1\n"
>+      "  - frames_dir(string): The path to the directory where the frames "
>+      "reside."
>+      " Default: .\n"
>+      "  - output_file(string): The output file to which frames are written."
>+      " Default: output.yuv\n"
>+      "  - delete_frames(bool): Whether or not to delete the input frames after"
>+      " the conversion. Default: false.\n";
> 
>   webrtc::test::CommandLineParser parser;
> 
>@@ -76,9 +79,8 @@ int main(int argc, char** argv) {
>   bool del_frames = (parser.GetFlag("delete_frames") == "true") ? true : false;
> 
>   webrtc::test::Converter converter(width, height);
>-  bool success = converter.ConvertRGBAToI420Video(parser.GetFlag("frames_dir"),
>-                                                  parser.GetFlag("output_file"),
>-                                                  del_frames);
>+  bool success = converter.ConvertRGBAToI420Video(
>+      parser.GetFlag("frames_dir"), parser.GetFlag("output_file"), del_frames);
> 
>   if (success) {
>     fprintf(stdout, "Successful conversion of RGBA frames to YUV video!\n");
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/event_log_visualizer/analyzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/event_log_visualizer/analyzer.cc
>index bbbde3f4c13..56ea1c7fe31 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/event_log_visualizer/analyzer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/event_log_visualizer/analyzer.cc
>@@ -14,10 +14,10 @@
> #include <cmath>
> #include <limits>
> #include <map>
>-#include <sstream>
> #include <string>
> #include <utility>
> 
>+#include "absl/memory/memory.h"
> #include "call/audio_receive_stream.h"
> #include "call/audio_send_stream.h"
> #include "call/call.h"
>@@ -25,20 +25,23 @@
> #include "call/video_send_stream.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "logging/rtc_event_log/rtc_stream_config.h"
>+#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
> #include "modules/audio_coding/neteq/tools/audio_sink.h"
> #include "modules/audio_coding/neteq/tools/fake_decode_from_file.h"
> #include "modules/audio_coding/neteq/tools/neteq_delay_analyzer.h"
> #include "modules/audio_coding/neteq/tools/neteq_replacement_input.h"
> #include "modules/audio_coding/neteq/tools/neteq_test.h"
> #include "modules/audio_coding/neteq/tools/resample_input_audio_file.h"
>-#include "modules/congestion_controller/acknowledged_bitrate_estimator.h"
>-#include "modules/congestion_controller/bitrate_estimator.h"
>+#include "modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h"
>+#include "modules/congestion_controller/goog_cc/bitrate_estimator.h"
>+#include "modules/congestion_controller/goog_cc/delay_based_bwe.h"
> #include "modules/congestion_controller/include/receive_side_congestion_controller.h"
> #include "modules/congestion_controller/include/send_side_congestion_controller.h"
> #include "modules/include/module_common_types.h"
> #include "modules/pacing/packet_router.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>+#include "modules/rtp_rtcp/source/rtcp_packet.h"
> #include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
> #include "modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
> #include "modules/rtp_rtcp/source/rtcp_packet/remb.h"
>@@ -48,9 +51,9 @@
> #include "modules/rtp_rtcp/source/rtp_utility.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/format_macros.h"
>+#include "rtc_base/function_view.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/numerics/sequence_number_util.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/rate_statistics.h"
> 
> #ifndef BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
>@@ -58,10 +61,11 @@
> #endif  // BWE_TEST_LOGGING_COMPILE_TIME_ENABLE
> 
> namespace webrtc {
>-namespace plotting {
> 
> namespace {
> 
>+const int kNumMicrosecsPerSec = 1000000;
>+
> void SortPacketFeedbackVector(std::vector<PacketFeedback>* vec) {
>   auto pred = [](const PacketFeedback& packet_feedback) {
>     return packet_feedback.arrival_time_ms == PacketFeedback::kNotReceived;
>@@ -88,9 +92,10 @@ bool MatchingSsrc(uint32_t ssrc, const std::vector<uint32_t>& desired_ssrc) {
> double AbsSendTimeToMicroseconds(int64_t abs_send_time) {
>   // The timestamp is a fixed point representation with 6 bits for seconds
>   // and 18 bits for fractions of a second. Thus, we divide by 2^18 to get the
>-  // time in seconds and then multiply by 1000000 to convert to microseconds.
>+  // time in seconds and then multiply by kNumMicrosecsPerSec to convert to
>+  // microseconds.
>   static constexpr double kTimestampToMicroSec =
>-      1000000.0 / static_cast<double>(1ul << 18);
>+      static_cast<double>(kNumMicrosecsPerSec) / static_cast<double>(1ul << 18);
>   return abs_send_time * kTimestampToMicroSec;
> }
> 
>@@ -123,29 +128,45 @@ int64_t WrappingDifference(uint32_t later, uint32_t earlier, int64_t modulus) {
>   return difference;
> }
> 
>-// Return default values for header extensions, to use on streams without stored
>-// mapping data. Currently this only applies to audio streams, since the mapping
>-// is not stored in the event log.
>-// TODO(ivoc): Remove this once this mapping is stored in the event log for
>-//             audio streams. Tracking bug: webrtc:6399
>-webrtc::RtpHeaderExtensionMap GetDefaultHeaderExtensionMap() {
>-  webrtc::RtpHeaderExtensionMap default_map;
>-  default_map.Register<AudioLevel>(webrtc::RtpExtension::kAudioLevelDefaultId);
>-  default_map.Register<TransmissionOffset>(
>-      webrtc::RtpExtension::kTimestampOffsetDefaultId);
>-  default_map.Register<AbsoluteSendTime>(
>-      webrtc::RtpExtension::kAbsSendTimeDefaultId);
>-  default_map.Register<VideoOrientation>(
>-      webrtc::RtpExtension::kVideoRotationDefaultId);
>-  default_map.Register<VideoContentTypeExtension>(
>-      webrtc::RtpExtension::kVideoContentTypeDefaultId);
>-  default_map.Register<VideoTimingExtension>(
>-      webrtc::RtpExtension::kVideoTimingDefaultId);
>-  default_map.Register<TransportSequenceNumber>(
>-      webrtc::RtpExtension::kTransportSequenceNumberDefaultId);
>-  default_map.Register<PlayoutDelayLimits>(
>-      webrtc::RtpExtension::kPlayoutDelayDefaultId);
>-  return default_map;
>+// This is much more reliable for outgoing streams than for incoming streams.
>+template <typename RtpPacketContainer>
>+absl::optional<uint32_t> EstimateRtpClockFrequency(
>+    const RtpPacketContainer& packets,
>+    int64_t end_time_us) {
>+  RTC_CHECK(packets.size() >= 2);
>+  SeqNumUnwrapper<uint32_t> unwrapper;
>+  uint64_t first_rtp_timestamp =
>+      unwrapper.Unwrap(packets[0].rtp.header.timestamp);
>+  int64_t first_log_timestamp = packets[0].log_time_us();
>+  uint64_t last_rtp_timestamp = first_rtp_timestamp;
>+  int64_t last_log_timestamp = first_log_timestamp;
>+  for (size_t i = 1; i < packets.size(); i++) {
>+    if (packets[i].log_time_us() > end_time_us)
>+      break;
>+    last_rtp_timestamp = unwrapper.Unwrap(packets[i].rtp.header.timestamp);
>+    last_log_timestamp = packets[i].log_time_us();
>+  }
>+  if (last_log_timestamp - first_log_timestamp < kNumMicrosecsPerSec) {
>+    RTC_LOG(LS_WARNING)
>+        << "Failed to estimate RTP clock frequency: Stream too short. ("
>+        << packets.size() << " packets, "
>+        << last_log_timestamp - first_log_timestamp << " us)";
>+    return absl::nullopt;
>+  }
>+  double duration =
>+      static_cast<double>(last_log_timestamp - first_log_timestamp) /
>+      kNumMicrosecsPerSec;
>+  double estimated_frequency =
>+      (last_rtp_timestamp - first_rtp_timestamp) / duration;
>+  for (uint32_t f : {8000, 16000, 32000, 48000, 90000}) {
>+    if (std::fabs(estimated_frequency - f) < 0.05 * f) {
>+      return f;
>+    }
>+  }
>+  RTC_LOG(LS_WARNING) << "Failed to estimate RTP clock frequency: Estimate "
>+                      << estimated_frequency
>+                      << "not close to any stardard RTP frequency.";
>+  return absl::nullopt;
> }
> 
> constexpr float kLeftMargin = 0.01f;
>@@ -153,7 +174,7 @@ constexpr float kRightMargin = 0.02f;
> constexpr float kBottomMargin = 0.02f;
> constexpr float kTopMargin = 0.05f;
> 
>-rtc::Optional<double> NetworkDelayDiff_AbsSendTime(
>+absl::optional<double> NetworkDelayDiff_AbsSendTime(
>     const LoggedRtpPacket& old_packet,
>     const LoggedRtpPacket& new_packet) {
>   if (old_packet.header.extension.hasAbsoluteSendTime &&
>@@ -161,21 +182,22 @@ rtc::Optional<double> NetworkDelayDiff_AbsSendTime(
>     int64_t send_time_diff = WrappingDifference(
>         new_packet.header.extension.absoluteSendTime,
>         old_packet.header.extension.absoluteSendTime, 1ul << 24);
>-    int64_t recv_time_diff = new_packet.timestamp - old_packet.timestamp;
>+    int64_t recv_time_diff =
>+        new_packet.log_time_us() - old_packet.log_time_us();
>     double delay_change_us =
>         recv_time_diff - AbsSendTimeToMicroseconds(send_time_diff);
>     return delay_change_us / 1000;
>   } else {
>-    return rtc::nullopt;
>+    return absl::nullopt;
>   }
> }
> 
>-rtc::Optional<double> NetworkDelayDiff_CaptureTime(
>+absl::optional<double> NetworkDelayDiff_CaptureTime(
>     const LoggedRtpPacket& old_packet,
>     const LoggedRtpPacket& new_packet) {
>   int64_t send_time_diff = WrappingDifference(
>       new_packet.header.timestamp, old_packet.header.timestamp, 1ull << 32);
>-  int64_t recv_time_diff = new_packet.timestamp - old_packet.timestamp;
>+  int64_t recv_time_diff = new_packet.log_time_us() - old_packet.log_time_us();
> 
>   const double kVideoSampleRate = 90000;
>   // TODO(terelius): We treat all streams as video for now, even though
>@@ -189,11 +211,13 @@ rtc::Optional<double> NetworkDelayDiff_CaptureTime(
>   if (delay_change < -10000 || 10000 < delay_change) {
>     RTC_LOG(LS_WARNING) << "Very large delay change. Timestamps correct?";
>     RTC_LOG(LS_WARNING) << "Old capture time " << old_packet.header.timestamp
>-                        << ", received time " << old_packet.timestamp;
>+                        << ", received time " << old_packet.log_time_us();
>     RTC_LOG(LS_WARNING) << "New capture time " << new_packet.header.timestamp
>-                        << ", received time " << new_packet.timestamp;
>+                        << ", received time " << new_packet.log_time_us();
>     RTC_LOG(LS_WARNING) << "Receive time difference " << recv_time_diff << " = "
>-                        << static_cast<double>(recv_time_diff) / 1000000 << "s";
>+                        << static_cast<double>(recv_time_diff) /
>+                               kNumMicrosecsPerSec
>+                        << "s";
>     RTC_LOG(LS_WARNING) << "Send time difference " << send_time_diff << " = "
>                         << static_cast<double>(send_time_diff) /
>                                kVideoSampleRate
>@@ -202,73 +226,54 @@ rtc::Optional<double> NetworkDelayDiff_CaptureTime(
>   return delay_change;
> }
> 
>-// For each element in data, use |get_y()| to extract a y-coordinate and
>+// For each element in data_view, use |f()| to extract a y-coordinate and
> // store the result in a TimeSeries.
>-template <typename DataType>
>-void ProcessPoints(
>-    rtc::FunctionView<rtc::Optional<float>(const DataType&)> get_y,
>-    const std::vector<DataType>& data,
>-    uint64_t begin_time,
>-    TimeSeries* result) {
>-  for (size_t i = 0; i < data.size(); i++) {
>-    float x = static_cast<float>(data[i].timestamp - begin_time) / 1000000;
>-    rtc::Optional<float> y = get_y(data[i]);
>+template <typename DataType, typename IterableType>
>+void ProcessPoints(rtc::FunctionView<float(const DataType&)> fx,
>+                   rtc::FunctionView<absl::optional<float>(const DataType&)> fy,
>+                   const IterableType& data_view,
>+                   TimeSeries* result) {
>+  for (size_t i = 0; i < data_view.size(); i++) {
>+    const DataType& elem = data_view[i];
>+    float x = fx(elem);
>+    absl::optional<float> y = fy(elem);
>     if (y)
>       result->points.emplace_back(x, *y);
>   }
> }
> 
>-// For each pair of adjacent elements in |data|, use |get_y| to extract a
>+// For each pair of adjacent elements in |data|, use |f()| to extract a
> // y-coordinate and store the result in a TimeSeries. Note that the x-coordinate
> // will be the time of the second element in the pair.
>-template <typename DataType, typename ResultType>
>+template <typename DataType, typename ResultType, typename IterableType>
> void ProcessPairs(
>-    rtc::FunctionView<rtc::Optional<ResultType>(const DataType&,
>-                                                const DataType&)> get_y,
>-    const std::vector<DataType>& data,
>-    uint64_t begin_time,
>+    rtc::FunctionView<float(const DataType&)> fx,
>+    rtc::FunctionView<absl::optional<ResultType>(const DataType&,
>+                                                 const DataType&)> fy,
>+    const IterableType& data,
>     TimeSeries* result) {
>   for (size_t i = 1; i < data.size(); i++) {
>-    float x = static_cast<float>(data[i].timestamp - begin_time) / 1000000;
>-    rtc::Optional<ResultType> y = get_y(data[i - 1], data[i]);
>+    float x = fx(data[i]);
>+    absl::optional<ResultType> y = fy(data[i - 1], data[i]);
>     if (y)
>       result->points.emplace_back(x, static_cast<float>(*y));
>   }
> }
> 
>-// For each element in data, use |extract()| to extract a y-coordinate and
>-// store the result in a TimeSeries.
>-template <typename DataType, typename ResultType>
>-void AccumulatePoints(
>-    rtc::FunctionView<rtc::Optional<ResultType>(const DataType&)> extract,
>-    const std::vector<DataType>& data,
>-    uint64_t begin_time,
>-    TimeSeries* result) {
>-  ResultType sum = 0;
>-  for (size_t i = 0; i < data.size(); i++) {
>-    float x = static_cast<float>(data[i].timestamp - begin_time) / 1000000;
>-    rtc::Optional<ResultType> y = extract(data[i]);
>-    if (y) {
>-      sum += *y;
>-      result->points.emplace_back(x, static_cast<float>(sum));
>-    }
>-  }
>-}
>-
>-// For each pair of adjacent elements in |data|, use |extract()| to extract a
>+// For each pair of adjacent elements in |data|, use |f()| to extract a
> // y-coordinate and store the result in a TimeSeries. Note that the x-coordinate
> // will be the time of the second element in the pair.
>-template <typename DataType, typename ResultType>
>+template <typename DataType, typename ResultType, typename IterableType>
> void AccumulatePairs(
>-    rtc::FunctionView<rtc::Optional<ResultType>(const DataType&,
>-                                                const DataType&)> extract,
>-    const std::vector<DataType>& data,
>-    uint64_t begin_time,
>+    rtc::FunctionView<float(const DataType&)> fx,
>+    rtc::FunctionView<absl::optional<ResultType>(const DataType&,
>+                                                 const DataType&)> fy,
>+    const IterableType& data,
>     TimeSeries* result) {
>   ResultType sum = 0;
>   for (size_t i = 1; i < data.size(); i++) {
>-    float x = static_cast<float>(data[i].timestamp - begin_time) / 1000000;
>-    rtc::Optional<ResultType> y = extract(data[i - 1], data[i]);
>+    float x = fx(data[i]);
>+    absl::optional<ResultType> y = fy(data[i - 1], data[i]);
>     if (y)
>       sum += *y;
>     result->points.emplace_back(x, static_cast<float>(sum));
>@@ -279,275 +284,215 @@ void AccumulatePairs(
> // A data point is generated every |step| microseconds from |begin_time|
> // to |end_time|. The value of each data point is the average of the data
> // during the preceeding |window_duration_us| microseconds.
>-template <typename DataType, typename ResultType>
>+template <typename DataType, typename ResultType, typename IterableType>
> void MovingAverage(
>-    rtc::FunctionView<rtc::Optional<ResultType>(const DataType&)> extract,
>-    const std::vector<DataType>& data,
>-    uint64_t begin_time,
>-    uint64_t end_time,
>-    uint64_t window_duration_us,
>-    uint64_t step,
>-    webrtc::plotting::TimeSeries* result) {
>+    rtc::FunctionView<float(int64_t)> fx,
>+    rtc::FunctionView<absl::optional<ResultType>(const DataType&)> fy,
>+    const IterableType& data_view,
>+    int64_t begin_time,
>+    int64_t end_time,
>+    int64_t window_duration_us,
>+    int64_t step,
>+    TimeSeries* result) {
>   size_t window_index_begin = 0;
>   size_t window_index_end = 0;
>   ResultType sum_in_window = 0;
> 
>-  for (uint64_t t = begin_time; t < end_time + step; t += step) {
>-    while (window_index_end < data.size() &&
>-           data[window_index_end].timestamp < t) {
>-      rtc::Optional<ResultType> value = extract(data[window_index_end]);
>+  for (int64_t t = begin_time; t < end_time + step; t += step) {
>+    while (window_index_end < data_view.size() &&
>+           data_view[window_index_end].log_time_us() < t) {
>+      absl::optional<ResultType> value = fy(data_view[window_index_end]);
>       if (value)
>         sum_in_window += *value;
>       ++window_index_end;
>     }
>-    while (window_index_begin < data.size() &&
>-           data[window_index_begin].timestamp < t - window_duration_us) {
>-      rtc::Optional<ResultType> value = extract(data[window_index_begin]);
>+    while (window_index_begin < data_view.size() &&
>+           data_view[window_index_begin].log_time_us() <
>+               t - window_duration_us) {
>+      absl::optional<ResultType> value = fy(data_view[window_index_begin]);
>       if (value)
>         sum_in_window -= *value;
>       ++window_index_begin;
>     }
>-    float window_duration_s = static_cast<float>(window_duration_us) / 1000000;
>-    float x = static_cast<float>(t - begin_time) / 1000000;
>+    float window_duration_s =
>+        static_cast<float>(window_duration_us) / kNumMicrosecsPerSec;
>+    float x = fx(t);
>     float y = sum_in_window / window_duration_s;
>     result->points.emplace_back(x, y);
>   }
> }
> 
>-}  // namespace
>+const char kUnknownEnumValue[] = "unknown";
>+
>+const char kIceCandidateTypeLocal[] = "local";
>+const char kIceCandidateTypeStun[] = "stun";
>+const char kIceCandidateTypePrflx[] = "prflx";
>+const char kIceCandidateTypeRelay[] = "relay";
>+
>+const char kProtocolUdp[] = "udp";
>+const char kProtocolTcp[] = "tcp";
>+const char kProtocolSsltcp[] = "ssltcp";
>+const char kProtocolTls[] = "tls";
>+
>+const char kAddressFamilyIpv4[] = "ipv4";
>+const char kAddressFamilyIpv6[] = "ipv6";
>+
>+const char kNetworkTypeEthernet[] = "ethernet";
>+const char kNetworkTypeLoopback[] = "loopback";
>+const char kNetworkTypeWifi[] = "wifi";
>+const char kNetworkTypeVpn[] = "vpn";
>+const char kNetworkTypeCellular[] = "cellular";
>+
>+std::string GetIceCandidateTypeAsString(webrtc::IceCandidateType type) {
>+  switch (type) {
>+    case webrtc::IceCandidateType::kLocal:
>+      return kIceCandidateTypeLocal;
>+    case webrtc::IceCandidateType::kStun:
>+      return kIceCandidateTypeStun;
>+    case webrtc::IceCandidateType::kPrflx:
>+      return kIceCandidateTypePrflx;
>+    case webrtc::IceCandidateType::kRelay:
>+      return kIceCandidateTypeRelay;
>+    default:
>+      return kUnknownEnumValue;
>+  }
>+}
> 
>-EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLog& log)
>-    : parsed_log_(log), window_duration_(250000), step_(10000) {
>-  uint64_t first_timestamp = std::numeric_limits<uint64_t>::max();
>-  uint64_t last_timestamp = std::numeric_limits<uint64_t>::min();
>-
>-  PacketDirection direction;
>-  uint8_t header[IP_PACKET_SIZE];
>-  size_t header_length;
>-  size_t total_length;
>-
>-  uint8_t last_incoming_rtcp_packet[IP_PACKET_SIZE];
>-  uint8_t last_incoming_rtcp_packet_length = 0;
>-
>-  // Make a default extension map for streams without configuration information.
>-  // TODO(ivoc): Once configuration of audio streams is stored in the event log,
>-  //             this can be removed. Tracking bug: webrtc:6399
>-  RtpHeaderExtensionMap default_extension_map = GetDefaultHeaderExtensionMap();
>-
>-  rtc::Optional<uint64_t> last_log_start;
>-
>-  for (size_t i = 0; i < parsed_log_.GetNumberOfEvents(); i++) {
>-    ParsedRtcEventLog::EventType event_type = parsed_log_.GetEventType(i);
>-    if (event_type != ParsedRtcEventLog::VIDEO_RECEIVER_CONFIG_EVENT &&
>-        event_type != ParsedRtcEventLog::VIDEO_SENDER_CONFIG_EVENT &&
>-        event_type != ParsedRtcEventLog::AUDIO_RECEIVER_CONFIG_EVENT &&
>-        event_type != ParsedRtcEventLog::AUDIO_SENDER_CONFIG_EVENT &&
>-        event_type != ParsedRtcEventLog::LOG_START &&
>-        event_type != ParsedRtcEventLog::LOG_END) {
>-      uint64_t timestamp = parsed_log_.GetTimestamp(i);
>-      first_timestamp = std::min(first_timestamp, timestamp);
>-      last_timestamp = std::max(last_timestamp, timestamp);
>-    }
>+std::string GetProtocolAsString(webrtc::IceCandidatePairProtocol protocol) {
>+  switch (protocol) {
>+    case webrtc::IceCandidatePairProtocol::kUdp:
>+      return kProtocolUdp;
>+    case webrtc::IceCandidatePairProtocol::kTcp:
>+      return kProtocolTcp;
>+    case webrtc::IceCandidatePairProtocol::kSsltcp:
>+      return kProtocolSsltcp;
>+    case webrtc::IceCandidatePairProtocol::kTls:
>+      return kProtocolTls;
>+    default:
>+      return kUnknownEnumValue;
>+  }
>+}
> 
>-    switch (parsed_log_.GetEventType(i)) {
>-      case ParsedRtcEventLog::VIDEO_RECEIVER_CONFIG_EVENT: {
>-        rtclog::StreamConfig config = parsed_log_.GetVideoReceiveConfig(i);
>-        StreamId stream(config.remote_ssrc, kIncomingPacket);
>-        video_ssrcs_.insert(stream);
>-        StreamId rtx_stream(config.rtx_ssrc, kIncomingPacket);
>-        video_ssrcs_.insert(rtx_stream);
>-        rtx_ssrcs_.insert(rtx_stream);
>-        break;
>-      }
>-      case ParsedRtcEventLog::VIDEO_SENDER_CONFIG_EVENT: {
>-        std::vector<rtclog::StreamConfig> configs =
>-            parsed_log_.GetVideoSendConfig(i);
>-        for (const auto& config : configs) {
>-          StreamId stream(config.local_ssrc, kOutgoingPacket);
>-          video_ssrcs_.insert(stream);
>-          StreamId rtx_stream(config.rtx_ssrc, kOutgoingPacket);
>-          video_ssrcs_.insert(rtx_stream);
>-          rtx_ssrcs_.insert(rtx_stream);
>-        }
>-        break;
>-      }
>-      case ParsedRtcEventLog::AUDIO_RECEIVER_CONFIG_EVENT: {
>-        rtclog::StreamConfig config = parsed_log_.GetAudioReceiveConfig(i);
>-        StreamId stream(config.remote_ssrc, kIncomingPacket);
>-        audio_ssrcs_.insert(stream);
>-        break;
>-      }
>-      case ParsedRtcEventLog::AUDIO_SENDER_CONFIG_EVENT: {
>-        rtclog::StreamConfig config = parsed_log_.GetAudioSendConfig(i);
>-        StreamId stream(config.local_ssrc, kOutgoingPacket);
>-        audio_ssrcs_.insert(stream);
>-        break;
>-      }
>-      case ParsedRtcEventLog::RTP_EVENT: {
>-        RtpHeaderExtensionMap* extension_map = parsed_log_.GetRtpHeader(
>-            i, &direction, header, &header_length, &total_length, nullptr);
>-        RtpUtility::RtpHeaderParser rtp_parser(header, header_length);
>-        RTPHeader parsed_header;
>-        if (extension_map != nullptr) {
>-          rtp_parser.Parse(&parsed_header, extension_map);
>-        } else {
>-          // Use the default extension map.
>-          // TODO(ivoc): Once configuration of audio streams is stored in the
>-          //             event log, this can be removed.
>-          //             Tracking bug: webrtc:6399
>-          rtp_parser.Parse(&parsed_header, &default_extension_map);
>-        }
>-        uint64_t timestamp = parsed_log_.GetTimestamp(i);
>-        StreamId stream(parsed_header.ssrc, direction);
>-        rtp_packets_[stream].push_back(
>-            LoggedRtpPacket(timestamp, parsed_header, total_length));
>-        break;
>-      }
>-      case ParsedRtcEventLog::RTCP_EVENT: {
>-        uint8_t packet[IP_PACKET_SIZE];
>-        parsed_log_.GetRtcpPacket(i, &direction, packet, &total_length);
>-        // Currently incoming RTCP packets are logged twice, both for audio and
>-        // video. Only act on one of them. Compare against the previous parsed
>-        // incoming RTCP packet.
>-        if (direction == webrtc::kIncomingPacket) {
>-          RTC_CHECK_LE(total_length, IP_PACKET_SIZE);
>-          if (total_length == last_incoming_rtcp_packet_length &&
>-              memcmp(last_incoming_rtcp_packet, packet, total_length) == 0) {
>-            continue;
>-          } else {
>-            memcpy(last_incoming_rtcp_packet, packet, total_length);
>-            last_incoming_rtcp_packet_length = total_length;
>-          }
>-        }
>-        rtcp::CommonHeader header;
>-        const uint8_t* packet_end = packet + total_length;
>-        for (const uint8_t* block = packet; block < packet_end;
>-             block = header.NextPacket()) {
>-          RTC_CHECK(header.Parse(block, packet_end - block));
>-          if (header.type() == rtcp::TransportFeedback::kPacketType &&
>-              header.fmt() == rtcp::TransportFeedback::kFeedbackMessageType) {
>-            std::unique_ptr<rtcp::TransportFeedback> rtcp_packet(
>-                rtc::MakeUnique<rtcp::TransportFeedback>());
>-            if (rtcp_packet->Parse(header)) {
>-              uint32_t ssrc = rtcp_packet->sender_ssrc();
>-              StreamId stream(ssrc, direction);
>-              uint64_t timestamp = parsed_log_.GetTimestamp(i);
>-              rtcp_packets_[stream].push_back(LoggedRtcpPacket(
>-                  timestamp, kRtcpTransportFeedback, std::move(rtcp_packet)));
>-            }
>-          } else if (header.type() == rtcp::SenderReport::kPacketType) {
>-            std::unique_ptr<rtcp::SenderReport> rtcp_packet(
>-                rtc::MakeUnique<rtcp::SenderReport>());
>-            if (rtcp_packet->Parse(header)) {
>-              uint32_t ssrc = rtcp_packet->sender_ssrc();
>-              StreamId stream(ssrc, direction);
>-              uint64_t timestamp = parsed_log_.GetTimestamp(i);
>-              rtcp_packets_[stream].push_back(
>-                  LoggedRtcpPacket(timestamp, kRtcpSr, std::move(rtcp_packet)));
>-            }
>-          } else if (header.type() == rtcp::ReceiverReport::kPacketType) {
>-            std::unique_ptr<rtcp::ReceiverReport> rtcp_packet(
>-                rtc::MakeUnique<rtcp::ReceiverReport>());
>-            if (rtcp_packet->Parse(header)) {
>-              uint32_t ssrc = rtcp_packet->sender_ssrc();
>-              StreamId stream(ssrc, direction);
>-              uint64_t timestamp = parsed_log_.GetTimestamp(i);
>-              rtcp_packets_[stream].push_back(
>-                  LoggedRtcpPacket(timestamp, kRtcpRr, std::move(rtcp_packet)));
>-            }
>-          } else if (header.type() == rtcp::Remb::kPacketType &&
>-                     header.fmt() == rtcp::Remb::kFeedbackMessageType) {
>-            std::unique_ptr<rtcp::Remb> rtcp_packet(
>-                rtc::MakeUnique<rtcp::Remb>());
>-            if (rtcp_packet->Parse(header)) {
>-              uint32_t ssrc = rtcp_packet->sender_ssrc();
>-              StreamId stream(ssrc, direction);
>-              uint64_t timestamp = parsed_log_.GetTimestamp(i);
>-              rtcp_packets_[stream].push_back(LoggedRtcpPacket(
>-                  timestamp, kRtcpRemb, std::move(rtcp_packet)));
>-            }
>-          }
>-        }
>-        break;
>-      }
>-      case ParsedRtcEventLog::LOG_START: {
>-        if (last_log_start) {
>-          // A LOG_END event was missing. Use last_timestamp.
>-          RTC_DCHECK_GE(last_timestamp, *last_log_start);
>-          log_segments_.push_back(
>-            std::make_pair(*last_log_start, last_timestamp));
>-        }
>-        last_log_start = parsed_log_.GetTimestamp(i);
>-        break;
>-      }
>-      case ParsedRtcEventLog::LOG_END: {
>-        RTC_DCHECK(last_log_start);
>-        log_segments_.push_back(
>-            std::make_pair(*last_log_start, parsed_log_.GetTimestamp(i)));
>-        last_log_start.reset();
>-        break;
>-      }
>-      case ParsedRtcEventLog::AUDIO_PLAYOUT_EVENT: {
>-        uint32_t this_ssrc;
>-        parsed_log_.GetAudioPlayout(i, &this_ssrc);
>-        audio_playout_events_[this_ssrc].push_back(parsed_log_.GetTimestamp(i));
>-        break;
>-      }
>-      case ParsedRtcEventLog::LOSS_BASED_BWE_UPDATE: {
>-        LossBasedBweUpdate bwe_update;
>-        bwe_update.timestamp = parsed_log_.GetTimestamp(i);
>-        parsed_log_.GetLossBasedBweUpdate(i, &bwe_update.new_bitrate,
>-                                          &bwe_update.fraction_loss,
>-                                          &bwe_update.expected_packets);
>-        bwe_loss_updates_.push_back(bwe_update);
>-        break;
>-      }
>-      case ParsedRtcEventLog::DELAY_BASED_BWE_UPDATE: {
>-        bwe_delay_updates_.push_back(parsed_log_.GetDelayBasedBweUpdate(i));
>-        break;
>-      }
>-      case ParsedRtcEventLog::AUDIO_NETWORK_ADAPTATION_EVENT: {
>-        AudioNetworkAdaptationEvent ana_event;
>-        ana_event.timestamp = parsed_log_.GetTimestamp(i);
>-        parsed_log_.GetAudioNetworkAdaptation(i, &ana_event.config);
>-        audio_network_adaptation_events_.push_back(ana_event);
>-        break;
>-      }
>-      case ParsedRtcEventLog::BWE_PROBE_CLUSTER_CREATED_EVENT: {
>-        bwe_probe_cluster_created_events_.push_back(
>-            parsed_log_.GetBweProbeClusterCreated(i));
>-        break;
>-      }
>-      case ParsedRtcEventLog::BWE_PROBE_RESULT_EVENT: {
>-        bwe_probe_result_events_.push_back(parsed_log_.GetBweProbeResult(i));
>-        break;
>-      }
>-      case ParsedRtcEventLog::UNKNOWN_EVENT: {
>-        break;
>-      }
>-    }
>+std::string GetAddressFamilyAsString(
>+    webrtc::IceCandidatePairAddressFamily family) {
>+  switch (family) {
>+    case webrtc::IceCandidatePairAddressFamily::kIpv4:
>+      return kAddressFamilyIpv4;
>+    case webrtc::IceCandidatePairAddressFamily::kIpv6:
>+      return kAddressFamilyIpv6;
>+    default:
>+      return kUnknownEnumValue;
>   }
>+}
> 
>-  if (last_timestamp < first_timestamp) {
>-    // No useful events in the log.
>-    first_timestamp = last_timestamp = 0;
>+std::string GetNetworkTypeAsString(webrtc::IceCandidateNetworkType type) {
>+  switch (type) {
>+    case webrtc::IceCandidateNetworkType::kEthernet:
>+      return kNetworkTypeEthernet;
>+    case webrtc::IceCandidateNetworkType::kLoopback:
>+      return kNetworkTypeLoopback;
>+    case webrtc::IceCandidateNetworkType::kWifi:
>+      return kNetworkTypeWifi;
>+    case webrtc::IceCandidateNetworkType::kVpn:
>+      return kNetworkTypeVpn;
>+    case webrtc::IceCandidateNetworkType::kCellular:
>+      return kNetworkTypeCellular;
>+    default:
>+      return kUnknownEnumValue;
>   }
>-  begin_time_ = first_timestamp;
>-  end_time_ = last_timestamp;
>-  call_duration_s_ = static_cast<float>(end_time_ - begin_time_) / 1000000;
>-  if (last_log_start) {
>-    // The log was missing the last LOG_END event. Fake it.
>-    log_segments_.push_back(std::make_pair(*last_log_start, end_time_));
>+}
>+
>+std::string GetCandidatePairLogDescriptionAsString(
>+    const LoggedIceCandidatePairConfig& config) {
>+  // Example: stun:wifi->relay(tcp):cellular@udp:ipv4
>+  // represents a pair of a local server-reflexive candidate on a WiFi network
>+  // and a remote relay candidate using TCP as the relay protocol on a cell
>+  // network, when the candidate pair communicates over UDP using IPv4.
>+  std::stringstream ss;
>+  std::string local_candidate_type =
>+      GetIceCandidateTypeAsString(config.local_candidate_type);
>+  std::string remote_candidate_type =
>+      GetIceCandidateTypeAsString(config.remote_candidate_type);
>+  if (config.local_candidate_type == webrtc::IceCandidateType::kRelay) {
>+    local_candidate_type +=
>+        "(" + GetProtocolAsString(config.local_relay_protocol) + ")";
>+  }
>+  ss << local_candidate_type << ":"
>+     << GetNetworkTypeAsString(config.local_network_type) << ":"
>+     << GetAddressFamilyAsString(config.local_address_family) << "->"
>+     << remote_candidate_type << ":"
>+     << GetAddressFamilyAsString(config.remote_address_family) << "@"
>+     << GetProtocolAsString(config.candidate_pair_protocol);
>+  return ss.str();
>+}
>+
>+std::string GetDirectionAsString(PacketDirection direction) {
>+  if (direction == kIncomingPacket) {
>+    return "Incoming";
>+  } else {
>+    return "Outgoing";
>   }
> }
> 
>-class BitrateObserver : public SendSideCongestionController::Observer,
>+std::string GetDirectionAsShortString(PacketDirection direction) {
>+  if (direction == kIncomingPacket) {
>+    return "In";
>+  } else {
>+    return "Out";
>+  }
>+}
>+
>+}  // namespace
>+
>+EventLogAnalyzer::EventLogAnalyzer(const ParsedRtcEventLogNew& log,
>+                                   bool normalize_time)
>+    : parsed_log_(log),
>+      window_duration_(250000),
>+      step_(10000),
>+      normalize_time_(normalize_time) {
>+  begin_time_ = parsed_log_.first_timestamp();
>+  end_time_ = parsed_log_.last_timestamp();
>+  if (end_time_ < begin_time_) {
>+    RTC_LOG(LS_WARNING) << "No useful events in the log.";
>+    begin_time_ = end_time_ = 0;
>+  }
>+  call_duration_s_ = ToCallTimeSec(end_time_);
>+
>+  const auto& log_start_events = parsed_log_.start_log_events();
>+  const auto& log_end_events = parsed_log_.stop_log_events();
>+  auto start_iter = log_start_events.begin();
>+  auto end_iter = log_end_events.begin();
>+  while (start_iter != log_start_events.end()) {
>+    int64_t start = start_iter->log_time_us();
>+    ++start_iter;
>+    absl::optional<int64_t> next_start;
>+    if (start_iter != log_start_events.end())
>+      next_start.emplace(start_iter->log_time_us());
>+    if (end_iter != log_end_events.end() &&
>+        end_iter->log_time_us() <=
>+            next_start.value_or(std::numeric_limits<int64_t>::max())) {
>+      int64_t end = end_iter->log_time_us();
>+      RTC_DCHECK_LE(start, end);
>+      log_segments_.push_back(std::make_pair(start, end));
>+      ++end_iter;
>+    } else {
>+      // we're missing an end event. Assume that it occurred just before the
>+      // next start.
>+      log_segments_.push_back(
>+          std::make_pair(start, next_start.value_or(end_time_)));
>+    }
>+  }
>+  RTC_LOG(LS_INFO) << "Found " << log_segments_.size()
>+                   << " (LOG_START, LOG_END) segments in log.";
>+}
>+
>+class BitrateObserver : public NetworkChangedObserver,
>                         public RemoteBitrateObserver {
>  public:
>   BitrateObserver() : last_bitrate_bps_(0), bitrate_updated_(false) {}
> 
>   void OnNetworkChanged(uint32_t bitrate_bps,
>-                        uint8_t fraction_loss,
>+                        uint8_t fraction_lost,
>                         int64_t rtt_ms,
>                         int64_t probing_interval_ms) override {
>     last_bitrate_bps_ = bitrate_bps;
>@@ -569,296 +514,212 @@ class BitrateObserver : public SendSideCongestionController::Observer,
>   bool bitrate_updated_;
> };
> 
>-bool EventLogAnalyzer::IsRtxSsrc(StreamId stream_id) const {
>-  return rtx_ssrcs_.count(stream_id) == 1;
>-}
>-
>-bool EventLogAnalyzer::IsVideoSsrc(StreamId stream_id) const {
>-  return video_ssrcs_.count(stream_id) == 1;
>-}
>-
>-bool EventLogAnalyzer::IsAudioSsrc(StreamId stream_id) const {
>-  return audio_ssrcs_.count(stream_id) == 1;
>-}
>-
>-std::string EventLogAnalyzer::GetStreamName(StreamId stream_id) const {
>-  std::stringstream name;
>-  if (IsAudioSsrc(stream_id)) {
>-    name << "Audio ";
>-  } else if (IsVideoSsrc(stream_id)) {
>-    name << "Video ";
>-  } else {
>-    name << "Unknown ";
>+int64_t EventLogAnalyzer::ToCallTimeUs(int64_t timestamp) const {
>+  int64_t begin_time = 0;
>+  if (normalize_time_) {
>+    begin_time = begin_time_;
>   }
>-  if (IsRtxSsrc(stream_id))
>-    name << "RTX ";
>-  if (stream_id.GetDirection() == kIncomingPacket) {
>-    name << "(In) ";
>-  } else {
>-    name << "(Out) ";
>-  }
>-  name << SsrcToString(stream_id.GetSsrc());
>-  return name.str();
>+  return timestamp - begin_time;
> }
> 
>-// This is much more reliable for outgoing streams than for incoming streams.
>-rtc::Optional<uint32_t> EventLogAnalyzer::EstimateRtpClockFrequency(
>-    const std::vector<LoggedRtpPacket>& packets) const {
>-  RTC_CHECK(packets.size() >= 2);
>-  uint64_t end_time_us = log_segments_.empty()
>-                             ? std::numeric_limits<uint64_t>::max()
>-                             : log_segments_.front().second;
>-  SeqNumUnwrapper<uint32_t> unwrapper;
>-  uint64_t first_rtp_timestamp = unwrapper.Unwrap(packets[0].header.timestamp);
>-  uint64_t first_log_timestamp = packets[0].timestamp;
>-  uint64_t last_rtp_timestamp = first_rtp_timestamp;
>-  uint64_t last_log_timestamp = first_log_timestamp;
>-  for (size_t i = 1; i < packets.size(); i++) {
>-    if (packets[i].timestamp > end_time_us)
>-      break;
>-    last_rtp_timestamp = unwrapper.Unwrap(packets[i].header.timestamp);
>-    last_log_timestamp = packets[i].timestamp;
>-  }
>-  if (last_log_timestamp - first_log_timestamp < 1000000) {
>-    RTC_LOG(LS_WARNING)
>-        << "Failed to estimate RTP clock frequency: Stream too short. ("
>-        << packets.size() << " packets, "
>-        << last_log_timestamp - first_log_timestamp << " us)";
>-    return rtc::nullopt;
>-  }
>-  double duration =
>-      static_cast<double>(last_log_timestamp - first_log_timestamp) / 1000000;
>-  double estimated_frequency =
>-      (last_rtp_timestamp - first_rtp_timestamp) / duration;
>-  for (uint32_t f : {8000, 16000, 32000, 48000, 90000}) {
>-    if (std::fabs(estimated_frequency - f) < 0.05 * f) {
>-      return f;
>-    }
>-  }
>-  RTC_LOG(LS_WARNING) << "Failed to estimate RTP clock frequency: Estimate "
>-                      << estimated_frequency
>-                      << "not close to any stardard RTP frequency.";
>-  return rtc::nullopt;
>+float EventLogAnalyzer::ToCallTimeSec(int64_t timestamp) const {
>+  return static_cast<float>(ToCallTimeUs(timestamp)) / kNumMicrosecsPerSec;
> }
> 
>-void EventLogAnalyzer::CreatePacketGraph(PacketDirection desired_direction,
>+void EventLogAnalyzer::CreatePacketGraph(PacketDirection direction,
>                                          Plot* plot) {
>-  for (auto& kv : rtp_packets_) {
>-    StreamId stream_id = kv.first;
>-    const std::vector<LoggedRtpPacket>& packet_stream = kv.second;
>-    // Filter on direction and SSRC.
>-    if (stream_id.GetDirection() != desired_direction ||
>-        !MatchingSsrc(stream_id.GetSsrc(), desired_ssrc_)) {
>+  for (const auto& stream : parsed_log_.rtp_packets_by_ssrc(direction)) {
>+    // Filter on SSRC.
>+    if (!MatchingSsrc(stream.ssrc, desired_ssrc_)) {
>       continue;
>     }
> 
>-    TimeSeries time_series(GetStreamName(stream_id), LineStyle::kBar);
>-    ProcessPoints<LoggedRtpPacket>(
>-        [](const LoggedRtpPacket& packet) {
>-          return rtc::Optional<float>(packet.total_length);
>-        },
>-        packet_stream, begin_time_, &time_series);
>+    TimeSeries time_series(GetStreamName(direction, stream.ssrc),
>+                           LineStyle::kBar);
>+    auto GetPacketSize = [](const LoggedRtpPacket& packet) {
>+      return absl::optional<float>(packet.total_length);
>+    };
>+    auto ToCallTime = [this](const LoggedRtpPacket& packet) {
>+      return this->ToCallTimeSec(packet.log_time_us());
>+    };
>+    ProcessPoints<LoggedRtpPacket>(ToCallTime, GetPacketSize,
>+                                   stream.packet_view, &time_series);
>     plot->AppendTimeSeries(std::move(time_series));
>   }
> 
>-  plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
>+  plot->SetXAxis(ToCallTimeSec(begin_time_), call_duration_s_, "Time (s)",
>+                 kLeftMargin, kRightMargin);
>   plot->SetSuggestedYAxis(0, 1, "Packet size (bytes)", kBottomMargin,
>                           kTopMargin);
>-  if (desired_direction == webrtc::PacketDirection::kIncomingPacket) {
>-    plot->SetTitle("Incoming RTP packets");
>-  } else if (desired_direction == webrtc::PacketDirection::kOutgoingPacket) {
>-    plot->SetTitle("Outgoing RTP packets");
>-  }
>+  plot->SetTitle(GetDirectionAsString(direction) + " RTP packets");
> }
> 
>-template <typename T>
>+template <typename IterableType>
> void EventLogAnalyzer::CreateAccumulatedPacketsTimeSeries(
>-    PacketDirection desired_direction,
>     Plot* plot,
>-    const std::map<StreamId, std::vector<T>>& packets,
>-    const std::string& label_prefix) {
>-  for (auto& kv : packets) {
>-    StreamId stream_id = kv.first;
>-    const std::vector<T>& packet_stream = kv.second;
>-    // Filter on direction and SSRC.
>-    if (stream_id.GetDirection() != desired_direction ||
>-        !MatchingSsrc(stream_id.GetSsrc(), desired_ssrc_)) {
>-      continue;
>-    }
>-
>-    std::string label = label_prefix + " " + GetStreamName(stream_id);
>-    TimeSeries time_series(label, LineStyle::kStep);
>-    for (size_t i = 0; i < packet_stream.size(); i++) {
>-      float x = static_cast<float>(packet_stream[i].timestamp - begin_time_) /
>-                1000000;
>-      time_series.points.emplace_back(x, i + 1);
>-    }
>-
>-    plot->AppendTimeSeries(std::move(time_series));
>+    const IterableType& packets,
>+    const std::string& label) {
>+  TimeSeries time_series(label, LineStyle::kStep);
>+  for (size_t i = 0; i < packets.size(); i++) {
>+    float x = ToCallTimeSec(packets[i].log_time_us());
>+    time_series.points.emplace_back(x, i + 1);
>   }
>+  plot->AppendTimeSeries(std::move(time_series));
> }
> 
>-void EventLogAnalyzer::CreateAccumulatedPacketsGraph(
>-    PacketDirection desired_direction,
>-    Plot* plot) {
>-  CreateAccumulatedPacketsTimeSeries(desired_direction, plot, rtp_packets_,
>-                                     "RTP");
>-  CreateAccumulatedPacketsTimeSeries(desired_direction, plot, rtcp_packets_,
>-                                     "RTCP");
>+void EventLogAnalyzer::CreateAccumulatedPacketsGraph(PacketDirection direction,
>+                                                     Plot* plot) {
>+  for (const auto& stream : parsed_log_.rtp_packets_by_ssrc(direction)) {
>+    if (!MatchingSsrc(stream.ssrc, desired_ssrc_))
>+      continue;
>+    std::string label =
>+        std::string("RTP ") + GetStreamName(direction, stream.ssrc);
>+    CreateAccumulatedPacketsTimeSeries(plot, stream.packet_view, label);
>+  }
>+  std::string label =
>+      std::string("RTCP ") + "(" + GetDirectionAsShortString(direction) + ")";
>+  if (direction == kIncomingPacket) {
>+    CreateAccumulatedPacketsTimeSeries(
>+        plot, parsed_log_.incoming_rtcp_packets(), label);
>+  } else {
>+    CreateAccumulatedPacketsTimeSeries(
>+        plot, parsed_log_.outgoing_rtcp_packets(), label);
>+  }
> 
>-  plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
>+  plot->SetXAxis(ToCallTimeSec(begin_time_), call_duration_s_, "Time (s)",
>+                 kLeftMargin, kRightMargin);
>   plot->SetSuggestedYAxis(0, 1, "Received Packets", kBottomMargin, kTopMargin);
>-  if (desired_direction == webrtc::PacketDirection::kIncomingPacket) {
>-    plot->SetTitle("Accumulated Incoming RTP/RTCP packets");
>-  } else if (desired_direction == webrtc::PacketDirection::kOutgoingPacket) {
>-    plot->SetTitle("Accumulated Outgoing RTP/RTCP packets");
>-  }
>+  plot->SetTitle(std::string("Accumulated ") + GetDirectionAsString(direction) +
>+                 " RTP/RTCP packets");
> }
> 
> // For each SSRC, plot the time between the consecutive playouts.
> void EventLogAnalyzer::CreatePlayoutGraph(Plot* plot) {
>-  std::map<uint32_t, TimeSeries> time_series;
>-  std::map<uint32_t, uint64_t> last_playout;
>-
>-  uint32_t ssrc;
>-
>-  for (size_t i = 0; i < parsed_log_.GetNumberOfEvents(); i++) {
>-    ParsedRtcEventLog::EventType event_type = parsed_log_.GetEventType(i);
>-    if (event_type == ParsedRtcEventLog::AUDIO_PLAYOUT_EVENT) {
>-      parsed_log_.GetAudioPlayout(i, &ssrc);
>-      uint64_t timestamp = parsed_log_.GetTimestamp(i);
>-      if (MatchingSsrc(ssrc, desired_ssrc_)) {
>-        float x = static_cast<float>(timestamp - begin_time_) / 1000000;
>-        float y = static_cast<float>(timestamp - last_playout[ssrc]) / 1000;
>-        if (time_series[ssrc].points.size() == 0) {
>-          // There were no previusly logged playout for this SSRC.
>-          // Generate a point, but place it on the x-axis.
>-          y = 0;
>-        }
>-        time_series[ssrc].points.push_back(TimeSeriesPoint(x, y));
>-        last_playout[ssrc] = timestamp;
>-      }
>+  for (const auto& playout_stream : parsed_log_.audio_playout_events()) {
>+    uint32_t ssrc = playout_stream.first;
>+    if (!MatchingSsrc(ssrc, desired_ssrc_))
>+      continue;
>+    absl::optional<int64_t> last_playout_ms;
>+    TimeSeries time_series(SsrcToString(ssrc), LineStyle::kBar);
>+    for (const auto& playout_event : playout_stream.second) {
>+      float x = ToCallTimeSec(playout_event.log_time_us());
>+      int64_t playout_time_ms = playout_event.log_time_ms();
>+      // If there were no previous playouts, place the point on the x-axis.
>+      float y = playout_time_ms - last_playout_ms.value_or(playout_time_ms);
>+      time_series.points.push_back(TimeSeriesPoint(x, y));
>+      last_playout_ms.emplace(playout_time_ms);
>     }
>+    plot->AppendTimeSeries(std::move(time_series));
>   }
> 
>-  // Set labels and put in graph.
>-  for (auto& kv : time_series) {
>-    kv.second.label = SsrcToString(kv.first);
>-    kv.second.line_style = LineStyle::kBar;
>-    plot->AppendTimeSeries(std::move(kv.second));
>-  }
>-
>-  plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
>+  plot->SetXAxis(ToCallTimeSec(begin_time_), call_duration_s_, "Time (s)",
>+                 kLeftMargin, kRightMargin);
>   plot->SetSuggestedYAxis(0, 1, "Time since last playout (ms)", kBottomMargin,
>                           kTopMargin);
>   plot->SetTitle("Audio playout");
> }
> 
> // For audio SSRCs, plot the audio level.
>-void EventLogAnalyzer::CreateAudioLevelGraph(Plot* plot) {
>-  std::map<StreamId, TimeSeries> time_series;
>-
>-  for (auto& kv : rtp_packets_) {
>-    StreamId stream_id = kv.first;
>-    const std::vector<LoggedRtpPacket>& packet_stream = kv.second;
>-    // TODO(ivoc): When audio send/receive configs are stored in the event
>-    //             log, a check should be added here to only process audio
>-    //             streams. Tracking bug: webrtc:6399
>-    for (auto& packet : packet_stream) {
>+void EventLogAnalyzer::CreateAudioLevelGraph(PacketDirection direction,
>+                                             Plot* plot) {
>+  for (const auto& stream : parsed_log_.rtp_packets_by_ssrc(direction)) {
>+    if (!IsAudioSsrc(direction, stream.ssrc))
>+      continue;
>+    TimeSeries time_series(GetStreamName(direction, stream.ssrc),
>+                           LineStyle::kLine);
>+    for (auto& packet : stream.packet_view) {
>       if (packet.header.extension.hasAudioLevel) {
>-        float x = static_cast<float>(packet.timestamp - begin_time_) / 1000000;
>+        float x = ToCallTimeSec(packet.log_time_us());
>         // The audio level is stored in -dBov (so e.g. -10 dBov is stored as 10)
>         // Here we convert it to dBov.
>         float y = static_cast<float>(-packet.header.extension.audioLevel);
>-        time_series[stream_id].points.emplace_back(TimeSeriesPoint(x, y));
>+        time_series.points.emplace_back(TimeSeriesPoint(x, y));
>       }
>     }
>+    plot->AppendTimeSeries(std::move(time_series));
>   }
> 
>-  for (auto& series : time_series) {
>-    series.second.label = GetStreamName(series.first);
>-    series.second.line_style = LineStyle::kLine;
>-    plot->AppendTimeSeries(std::move(series.second));
>-  }
>-
>-  plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
>-  plot->SetYAxis(-127, 0, "Audio level (dBov)", kBottomMargin,
>-                 kTopMargin);
>-  plot->SetTitle("Audio level");
>+  plot->SetXAxis(ToCallTimeSec(begin_time_), call_duration_s_, "Time (s)",
>+                 kLeftMargin, kRightMargin);
>+  plot->SetYAxis(-127, 0, "Audio level (dBov)", kBottomMargin, kTopMargin);
>+  plot->SetTitle(GetDirectionAsString(direction) + " audio level");
> }
> 
> // For each SSRC, plot the time between the consecutive playouts.
> void EventLogAnalyzer::CreateSequenceNumberGraph(Plot* plot) {
>-  for (auto& kv : rtp_packets_) {
>-    StreamId stream_id = kv.first;
>-    const std::vector<LoggedRtpPacket>& packet_stream = kv.second;
>-    // Filter on direction and SSRC.
>-    if (stream_id.GetDirection() != kIncomingPacket ||
>-        !MatchingSsrc(stream_id.GetSsrc(), desired_ssrc_)) {
>+  for (const auto& stream : parsed_log_.incoming_rtp_packets_by_ssrc()) {
>+    // Filter on SSRC.
>+    if (!MatchingSsrc(stream.ssrc, desired_ssrc_)) {
>       continue;
>     }
> 
>-    TimeSeries time_series(GetStreamName(stream_id), LineStyle::kBar);
>-    ProcessPairs<LoggedRtpPacket, float>(
>-        [](const LoggedRtpPacket& old_packet,
>-           const LoggedRtpPacket& new_packet) {
>-          int64_t diff =
>-              WrappingDifference(new_packet.header.sequenceNumber,
>-                                 old_packet.header.sequenceNumber, 1ul << 16);
>-          return diff;
>-        },
>-        packet_stream, begin_time_, &time_series);
>+    TimeSeries time_series(GetStreamName(kIncomingPacket, stream.ssrc),
>+                           LineStyle::kBar);
>+    auto GetSequenceNumberDiff = [](const LoggedRtpPacketIncoming& old_packet,
>+                                    const LoggedRtpPacketIncoming& new_packet) {
>+      int64_t diff =
>+          WrappingDifference(new_packet.rtp.header.sequenceNumber,
>+                             old_packet.rtp.header.sequenceNumber, 1ul << 16);
>+      return diff;
>+    };
>+    auto ToCallTime = [this](const LoggedRtpPacketIncoming& packet) {
>+      return this->ToCallTimeSec(packet.log_time_us());
>+    };
>+    ProcessPairs<LoggedRtpPacketIncoming, float>(
>+        ToCallTime, GetSequenceNumberDiff, stream.incoming_packets,
>+        &time_series);
>     plot->AppendTimeSeries(std::move(time_series));
>   }
> 
>-  plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
>+  plot->SetXAxis(ToCallTimeSec(begin_time_), call_duration_s_, "Time (s)",
>+                 kLeftMargin, kRightMargin);
>   plot->SetSuggestedYAxis(0, 1, "Difference since last packet", kBottomMargin,
>                           kTopMargin);
>   plot->SetTitle("Sequence number");
> }
> 
> void EventLogAnalyzer::CreateIncomingPacketLossGraph(Plot* plot) {
>-  for (auto& kv : rtp_packets_) {
>-    StreamId stream_id = kv.first;
>-    const std::vector<LoggedRtpPacket>& packet_stream = kv.second;
>-    // Filter on direction and SSRC.
>-    if (stream_id.GetDirection() != kIncomingPacket ||
>-        !MatchingSsrc(stream_id.GetSsrc(), desired_ssrc_) ||
>-        packet_stream.size() == 0) {
>+  for (const auto& stream : parsed_log_.incoming_rtp_packets_by_ssrc()) {
>+    const std::vector<LoggedRtpPacketIncoming>& packets =
>+        stream.incoming_packets;
>+    // Filter on SSRC.
>+    if (!MatchingSsrc(stream.ssrc, desired_ssrc_) || packets.size() == 0) {
>       continue;
>     }
> 
>-    TimeSeries time_series(GetStreamName(stream_id), LineStyle::kLine,
>-                           PointStyle::kHighlight);
>-    const uint64_t kWindowUs = 1000000;
>-    const uint64_t kStep = 1000000;
>-    SequenceNumberUnwrapper unwrapper_;
>-    SequenceNumberUnwrapper prior_unwrapper_;
>+    TimeSeries time_series(GetStreamName(kIncomingPacket, stream.ssrc),
>+                           LineStyle::kLine, PointStyle::kHighlight);
>+    // TODO(terelius): Should the window and step size be read from the class
>+    // instead?
>+    const int64_t kWindowUs = 1000000;
>+    const int64_t kStep = 1000000;
>+    SeqNumUnwrapper<uint16_t> unwrapper_;
>+    SeqNumUnwrapper<uint16_t> prior_unwrapper_;
>     size_t window_index_begin = 0;
>     size_t window_index_end = 0;
>-    int64_t highest_seq_number =
>-        unwrapper_.Unwrap(packet_stream[0].header.sequenceNumber) - 1;
>-    int64_t highest_prior_seq_number =
>-        prior_unwrapper_.Unwrap(packet_stream[0].header.sequenceNumber) - 1;
>-
>-    for (uint64_t t = begin_time_; t < end_time_ + kStep; t += kStep) {
>-      while (window_index_end < packet_stream.size() &&
>-             packet_stream[window_index_end].timestamp < t) {
>-        int64_t sequence_number = unwrapper_.Unwrap(
>-            packet_stream[window_index_end].header.sequenceNumber);
>+    uint64_t highest_seq_number =
>+        unwrapper_.Unwrap(packets[0].rtp.header.sequenceNumber) - 1;
>+    uint64_t highest_prior_seq_number =
>+        prior_unwrapper_.Unwrap(packets[0].rtp.header.sequenceNumber) - 1;
>+
>+    for (int64_t t = begin_time_; t < end_time_ + kStep; t += kStep) {
>+      while (window_index_end < packets.size() &&
>+             packets[window_index_end].rtp.log_time_us() < t) {
>+        uint64_t sequence_number = unwrapper_.Unwrap(
>+            packets[window_index_end].rtp.header.sequenceNumber);
>         highest_seq_number = std::max(highest_seq_number, sequence_number);
>         ++window_index_end;
>       }
>-      while (window_index_begin < packet_stream.size() &&
>-             packet_stream[window_index_begin].timestamp < t - kWindowUs) {
>-        int64_t sequence_number = prior_unwrapper_.Unwrap(
>-            packet_stream[window_index_begin].header.sequenceNumber);
>+      while (window_index_begin < packets.size() &&
>+             packets[window_index_begin].rtp.log_time_us() < t - kWindowUs) {
>+        uint64_t sequence_number = prior_unwrapper_.Unwrap(
>+            packets[window_index_begin].rtp.header.sequenceNumber);
>         highest_prior_seq_number =
>             std::max(highest_prior_seq_number, sequence_number);
>         ++window_index_begin;
>       }
>-      float x = static_cast<float>(t - begin_time_) / 1000000;
>-      int64_t expected_packets = highest_seq_number - highest_prior_seq_number;
>+      float x = ToCallTimeSec(t);
>+      uint64_t expected_packets = highest_seq_number - highest_prior_seq_number;
>       if (expected_packets > 0) {
>         int64_t received_packets = window_index_end - window_index_begin;
>         int64_t lost_packets = expected_packets - received_packets;
>@@ -869,73 +730,82 @@ void EventLogAnalyzer::CreateIncomingPacketLossGraph(Plot* plot) {
>     plot->AppendTimeSeries(std::move(time_series));
>   }
> 
>-  plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
>+  plot->SetXAxis(ToCallTimeSec(begin_time_), call_duration_s_, "Time (s)",
>+                 kLeftMargin, kRightMargin);
>   plot->SetSuggestedYAxis(0, 1, "Estimated loss rate (%)", kBottomMargin,
>                           kTopMargin);
>   plot->SetTitle("Estimated incoming loss rate");
> }
> 
> void EventLogAnalyzer::CreateIncomingDelayDeltaGraph(Plot* plot) {
>-  for (auto& kv : rtp_packets_) {
>-    StreamId stream_id = kv.first;
>-    const std::vector<LoggedRtpPacket>& packet_stream = kv.second;
>-    // Filter on direction and SSRC.
>-    if (stream_id.GetDirection() != kIncomingPacket ||
>-        !MatchingSsrc(stream_id.GetSsrc(), desired_ssrc_) ||
>-        IsAudioSsrc(stream_id) || !IsVideoSsrc(stream_id) ||
>-        IsRtxSsrc(stream_id)) {
>+  for (const auto& stream : parsed_log_.rtp_packets_by_ssrc(kIncomingPacket)) {
>+    // Filter on SSRC.
>+    if (!MatchingSsrc(stream.ssrc, desired_ssrc_) ||
>+        IsAudioSsrc(kIncomingPacket, stream.ssrc) ||
>+        !IsVideoSsrc(kIncomingPacket, stream.ssrc) ||
>+        IsRtxSsrc(kIncomingPacket, stream.ssrc)) {
>       continue;
>     }
> 
>-    TimeSeries capture_time_data(GetStreamName(stream_id) + " capture-time",
>-                                 LineStyle::kBar);
>-    ProcessPairs<LoggedRtpPacket, double>(NetworkDelayDiff_CaptureTime,
>-                                          packet_stream, begin_time_,
>-                                          &capture_time_data);
>+    TimeSeries capture_time_data(
>+        GetStreamName(kIncomingPacket, stream.ssrc) + " capture-time",
>+        LineStyle::kBar);
>+    auto ToCallTime = [this](const LoggedRtpPacket& packet) {
>+      return this->ToCallTimeSec(packet.log_time_us());
>+    };
>+    ProcessPairs<LoggedRtpPacket, double>(
>+        ToCallTime, NetworkDelayDiff_CaptureTime, stream.packet_view,
>+        &capture_time_data);
>     plot->AppendTimeSeries(std::move(capture_time_data));
> 
>-    TimeSeries send_time_data(GetStreamName(stream_id) + " abs-send-time",
>-                              LineStyle::kBar);
>-    ProcessPairs<LoggedRtpPacket, double>(NetworkDelayDiff_AbsSendTime,
>-                                          packet_stream, begin_time_,
>-                                          &send_time_data);
>+    TimeSeries send_time_data(
>+        GetStreamName(kIncomingPacket, stream.ssrc) + " abs-send-time",
>+        LineStyle::kBar);
>+    ProcessPairs<LoggedRtpPacket, double>(ToCallTime,
>+                                          NetworkDelayDiff_AbsSendTime,
>+                                          stream.packet_view, &send_time_data);
>     plot->AppendTimeSeries(std::move(send_time_data));
>   }
> 
>-  plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
>+  plot->SetXAxis(ToCallTimeSec(begin_time_), call_duration_s_, "Time (s)",
>+                 kLeftMargin, kRightMargin);
>   plot->SetSuggestedYAxis(0, 1, "Latency change (ms)", kBottomMargin,
>                           kTopMargin);
>   plot->SetTitle("Network latency difference between consecutive packets");
> }
> 
> void EventLogAnalyzer::CreateIncomingDelayGraph(Plot* plot) {
>-  for (auto& kv : rtp_packets_) {
>-    StreamId stream_id = kv.first;
>-    const std::vector<LoggedRtpPacket>& packet_stream = kv.second;
>-    // Filter on direction and SSRC.
>-    if (stream_id.GetDirection() != kIncomingPacket ||
>-        !MatchingSsrc(stream_id.GetSsrc(), desired_ssrc_) ||
>-        IsAudioSsrc(stream_id) || !IsVideoSsrc(stream_id) ||
>-        IsRtxSsrc(stream_id)) {
>+  for (const auto& stream : parsed_log_.rtp_packets_by_ssrc(kIncomingPacket)) {
>+    // Filter on SSRC.
>+    if (!MatchingSsrc(stream.ssrc, desired_ssrc_) ||
>+        IsAudioSsrc(kIncomingPacket, stream.ssrc) ||
>+        !IsVideoSsrc(kIncomingPacket, stream.ssrc) ||
>+        IsRtxSsrc(kIncomingPacket, stream.ssrc)) {
>       continue;
>     }
> 
>-    TimeSeries capture_time_data(GetStreamName(stream_id) + " capture-time",
>-                                 LineStyle::kLine);
>-    AccumulatePairs<LoggedRtpPacket, double>(NetworkDelayDiff_CaptureTime,
>-                                             packet_stream, begin_time_,
>-                                             &capture_time_data);
>+    TimeSeries capture_time_data(
>+        GetStreamName(kIncomingPacket, stream.ssrc) + " capture-time",
>+        LineStyle::kLine);
>+    auto ToCallTime = [this](const LoggedRtpPacket& packet) {
>+      return this->ToCallTimeSec(packet.log_time_us());
>+    };
>+    AccumulatePairs<LoggedRtpPacket, double>(
>+        ToCallTime, NetworkDelayDiff_CaptureTime, stream.packet_view,
>+        &capture_time_data);
>     plot->AppendTimeSeries(std::move(capture_time_data));
> 
>-    TimeSeries send_time_data(GetStreamName(stream_id) + " abs-send-time",
>-                              LineStyle::kLine);
>-    AccumulatePairs<LoggedRtpPacket, double>(NetworkDelayDiff_AbsSendTime,
>-                                             packet_stream, begin_time_,
>-                                             &send_time_data);
>+    TimeSeries send_time_data(
>+        GetStreamName(kIncomingPacket, stream.ssrc) + " abs-send-time",
>+        LineStyle::kLine);
>+    AccumulatePairs<LoggedRtpPacket, double>(
>+        ToCallTime, NetworkDelayDiff_AbsSendTime, stream.packet_view,
>+        &send_time_data);
>     plot->AppendTimeSeries(std::move(send_time_data));
>   }
> 
>-  plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
>+  plot->SetXAxis(ToCallTimeSec(begin_time_), call_duration_s_, "Time (s)",
>+                 kLeftMargin, kRightMargin);
>   plot->SetSuggestedYAxis(0, 1, "Latency change (ms)", kBottomMargin,
>                           kTopMargin);
>   plot->SetTitle("Network latency (relative to first packet)");
>@@ -945,241 +815,266 @@ void EventLogAnalyzer::CreateIncomingDelayGraph(Plot* plot) {
> void EventLogAnalyzer::CreateFractionLossGraph(Plot* plot) {
>   TimeSeries time_series("Fraction lost", LineStyle::kLine,
>                          PointStyle::kHighlight);
>-  for (auto& bwe_update : bwe_loss_updates_) {
>-    float x = static_cast<float>(bwe_update.timestamp - begin_time_) / 1000000;
>-    float y = static_cast<float>(bwe_update.fraction_loss) / 255 * 100;
>+  for (auto& bwe_update : parsed_log_.bwe_loss_updates()) {
>+    float x = ToCallTimeSec(bwe_update.log_time_us());
>+    float y = static_cast<float>(bwe_update.fraction_lost) / 255 * 100;
>     time_series.points.emplace_back(x, y);
>   }
> 
>   plot->AppendTimeSeries(std::move(time_series));
>-  plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
>+  plot->SetXAxis(ToCallTimeSec(begin_time_), call_duration_s_, "Time (s)",
>+                 kLeftMargin, kRightMargin);
>   plot->SetSuggestedYAxis(0, 10, "Percent lost packets", kBottomMargin,
>                           kTopMargin);
>   plot->SetTitle("Reported packet loss");
> }
> 
> // Plot the total bandwidth used by all RTP streams.
>-void EventLogAnalyzer::CreateTotalBitrateGraph(
>-    PacketDirection desired_direction,
>-    Plot* plot,
>-    bool show_detector_state) {
>-  struct TimestampSize {
>-    TimestampSize(uint64_t t, size_t s) : timestamp(t), size(s) {}
>-    uint64_t timestamp;
>-    size_t size;
>-  };
>-  std::vector<TimestampSize> packets;
>-
>-  PacketDirection direction;
>-  size_t total_length;
>-
>-  // Extract timestamps and sizes for the relevant packets.
>-  for (size_t i = 0; i < parsed_log_.GetNumberOfEvents(); i++) {
>-    ParsedRtcEventLog::EventType event_type = parsed_log_.GetEventType(i);
>-    if (event_type == ParsedRtcEventLog::RTP_EVENT) {
>-      parsed_log_.GetRtpHeader(i, &direction, nullptr, nullptr, &total_length,
>-                               nullptr);
>-      if (direction == desired_direction) {
>-        uint64_t timestamp = parsed_log_.GetTimestamp(i);
>-        packets.push_back(TimestampSize(timestamp, total_length));
>-      }
>-    }
>+void EventLogAnalyzer::CreateTotalIncomingBitrateGraph(Plot* plot) {
>+  // TODO(terelius): This could be provided by the parser.
>+  std::multimap<int64_t, size_t> packets_in_order;
>+  for (const auto& stream : parsed_log_.incoming_rtp_packets_by_ssrc()) {
>+    for (const LoggedRtpPacketIncoming& packet : stream.incoming_packets)
>+      packets_in_order.insert(
>+          std::make_pair(packet.rtp.log_time_us(), packet.rtp.total_length));
>   }
> 
>-  size_t window_index_begin = 0;
>-  size_t window_index_end = 0;
>+  auto window_begin = packets_in_order.begin();
>+  auto window_end = packets_in_order.begin();
>   size_t bytes_in_window = 0;
> 
>   // Calculate a moving average of the bitrate and store in a TimeSeries.
>   TimeSeries bitrate_series("Bitrate", LineStyle::kLine);
>-  for (uint64_t time = begin_time_; time < end_time_ + step_; time += step_) {
>-    while (window_index_end < packets.size() &&
>-           packets[window_index_end].timestamp < time) {
>-      bytes_in_window += packets[window_index_end].size;
>-      ++window_index_end;
>+  for (int64_t time = begin_time_; time < end_time_ + step_; time += step_) {
>+    while (window_end != packets_in_order.end() && window_end->first < time) {
>+      bytes_in_window += window_end->second;
>+      ++window_end;
>     }
>-    while (window_index_begin < packets.size() &&
>-           packets[window_index_begin].timestamp < time - window_duration_) {
>-      RTC_DCHECK_LE(packets[window_index_begin].size, bytes_in_window);
>-      bytes_in_window -= packets[window_index_begin].size;
>-      ++window_index_begin;
>+    while (window_begin != packets_in_order.end() &&
>+           window_begin->first < time - window_duration_) {
>+      RTC_DCHECK_LE(window_begin->second, bytes_in_window);
>+      bytes_in_window -= window_begin->second;
>+      ++window_begin;
>     }
>     float window_duration_in_seconds =
>-        static_cast<float>(window_duration_) / 1000000;
>-    float x = static_cast<float>(time - begin_time_) / 1000000;
>+        static_cast<float>(window_duration_) / kNumMicrosecsPerSec;
>+    float x = ToCallTimeSec(time);
>     float y = bytes_in_window * 8 / window_duration_in_seconds / 1000;
>     bitrate_series.points.emplace_back(x, y);
>   }
>   plot->AppendTimeSeries(std::move(bitrate_series));
> 
>-  // Overlay the send-side bandwidth estimate over the outgoing bitrate.
>-  if (desired_direction == kOutgoingPacket) {
>-    TimeSeries loss_series("Loss-based estimate", LineStyle::kStep);
>-    for (auto& loss_update : bwe_loss_updates_) {
>-      float x =
>-          static_cast<float>(loss_update.timestamp - begin_time_) / 1000000;
>-      float y = static_cast<float>(loss_update.new_bitrate) / 1000;
>-      loss_series.points.emplace_back(x, y);
>-    }
>-
>-    TimeSeries delay_series("Delay-based estimate", LineStyle::kStep);
>-    IntervalSeries overusing_series("Overusing", "#ff8e82",
>-                                    IntervalSeries::kHorizontal);
>-    IntervalSeries underusing_series("Underusing", "#5092fc",
>-                                     IntervalSeries::kHorizontal);
>-    IntervalSeries normal_series("Normal", "#c4ffc4",
>-                                 IntervalSeries::kHorizontal);
>-    IntervalSeries* last_series = &normal_series;
>-    double last_detector_switch = 0.0;
>-
>-    BandwidthUsage last_detector_state = BandwidthUsage::kBwNormal;
>-
>-    for (auto& delay_update : bwe_delay_updates_) {
>-      float x =
>-          static_cast<float>(delay_update.timestamp - begin_time_) / 1000000;
>-      float y = static_cast<float>(delay_update.bitrate_bps) / 1000;
>-
>-      if (last_detector_state != delay_update.detector_state) {
>-        last_series->intervals.emplace_back(last_detector_switch, x);
>-        last_detector_state = delay_update.detector_state;
>-        last_detector_switch = x;
>-
>-        switch (delay_update.detector_state) {
>-          case BandwidthUsage::kBwNormal:
>-            last_series = &normal_series;
>-            break;
>-          case BandwidthUsage::kBwUnderusing:
>-            last_series = &underusing_series;
>-            break;
>-          case BandwidthUsage::kBwOverusing:
>-            last_series = &overusing_series;
>-            break;
>-          case BandwidthUsage::kLast:
>-            RTC_NOTREACHED();
>-        }
>-      }
>+  // Overlay the outgoing REMB over incoming bitrate.
>+  TimeSeries remb_series("Remb", LineStyle::kStep);
>+  for (const auto& rtcp : parsed_log_.rembs(kOutgoingPacket)) {
>+    float x = ToCallTimeSec(rtcp.log_time_us());
>+    float y = static_cast<float>(rtcp.remb.bitrate_bps()) / 1000;
>+    remb_series.points.emplace_back(x, y);
>+  }
>+  plot->AppendTimeSeriesIfNotEmpty(std::move(remb_series));
> 
>-      delay_series.points.emplace_back(x, y);
>-    }
>+  plot->SetXAxis(ToCallTimeSec(begin_time_), call_duration_s_, "Time (s)",
>+                 kLeftMargin, kRightMargin);
>+  plot->SetSuggestedYAxis(0, 1, "Bitrate (kbps)", kBottomMargin, kTopMargin);
>+  plot->SetTitle("Incoming RTP bitrate");
>+}
> 
>-    RTC_CHECK(last_series);
>-    last_series->intervals.emplace_back(last_detector_switch, end_time_);
>+// Plot the total bandwidth used by all RTP streams.
>+void EventLogAnalyzer::CreateTotalOutgoingBitrateGraph(Plot* plot,
>+                                                       bool show_detector_state,
>+                                                       bool show_alr_state) {
>+  // TODO(terelius): This could be provided by the parser.
>+  std::multimap<int64_t, size_t> packets_in_order;
>+  for (const auto& stream : parsed_log_.outgoing_rtp_packets_by_ssrc()) {
>+    for (const LoggedRtpPacketOutgoing& packet : stream.outgoing_packets)
>+      packets_in_order.insert(
>+          std::make_pair(packet.rtp.log_time_us(), packet.rtp.total_length));
>+  }
>+
>+  auto window_begin = packets_in_order.begin();
>+  auto window_end = packets_in_order.begin();
>+  size_t bytes_in_window = 0;
> 
>-    TimeSeries created_series("Probe cluster created.", LineStyle::kNone,
>-                              PointStyle::kHighlight);
>-    for (auto& cluster : bwe_probe_cluster_created_events_) {
>-      float x = static_cast<float>(cluster.timestamp - begin_time_) / 1000000;
>-      float y = static_cast<float>(cluster.bitrate_bps) / 1000;
>-      created_series.points.emplace_back(x, y);
>+  // Calculate a moving average of the bitrate and store in a TimeSeries.
>+  TimeSeries bitrate_series("Bitrate", LineStyle::kLine);
>+  for (int64_t time = begin_time_; time < end_time_ + step_; time += step_) {
>+    while (window_end != packets_in_order.end() && window_end->first < time) {
>+      bytes_in_window += window_end->second;
>+      ++window_end;
>     }
>+    while (window_begin != packets_in_order.end() &&
>+           window_begin->first < time - window_duration_) {
>+      RTC_DCHECK_LE(window_begin->second, bytes_in_window);
>+      bytes_in_window -= window_begin->second;
>+      ++window_begin;
>+    }
>+    float window_duration_in_seconds =
>+        static_cast<float>(window_duration_) / kNumMicrosecsPerSec;
>+    float x = ToCallTimeSec(time);
>+    float y = bytes_in_window * 8 / window_duration_in_seconds / 1000;
>+    bitrate_series.points.emplace_back(x, y);
>+  }
>+  plot->AppendTimeSeries(std::move(bitrate_series));
> 
>-    TimeSeries result_series("Probing results.", LineStyle::kNone,
>-                             PointStyle::kHighlight);
>-    for (auto& result : bwe_probe_result_events_) {
>-      if (result.bitrate_bps) {
>-        float x = static_cast<float>(result.timestamp - begin_time_) / 1000000;
>-        float y = static_cast<float>(*result.bitrate_bps) / 1000;
>-        result_series.points.emplace_back(x, y);
>+  // Overlay the send-side bandwidth estimate over the outgoing bitrate.
>+  TimeSeries loss_series("Loss-based estimate", LineStyle::kStep);
>+  for (auto& loss_update : parsed_log_.bwe_loss_updates()) {
>+    float x = ToCallTimeSec(loss_update.log_time_us());
>+    float y = static_cast<float>(loss_update.bitrate_bps) / 1000;
>+    loss_series.points.emplace_back(x, y);
>+  }
>+
>+  TimeSeries delay_series("Delay-based estimate", LineStyle::kStep);
>+  IntervalSeries overusing_series("Overusing", "#ff8e82",
>+                                  IntervalSeries::kHorizontal);
>+  IntervalSeries underusing_series("Underusing", "#5092fc",
>+                                   IntervalSeries::kHorizontal);
>+  IntervalSeries normal_series("Normal", "#c4ffc4",
>+                               IntervalSeries::kHorizontal);
>+  IntervalSeries* last_series = &normal_series;
>+  double last_detector_switch = 0.0;
>+
>+  BandwidthUsage last_detector_state = BandwidthUsage::kBwNormal;
>+
>+  for (auto& delay_update : parsed_log_.bwe_delay_updates()) {
>+    float x = ToCallTimeSec(delay_update.log_time_us());
>+    float y = static_cast<float>(delay_update.bitrate_bps) / 1000;
>+
>+    if (last_detector_state != delay_update.detector_state) {
>+      last_series->intervals.emplace_back(last_detector_switch, x);
>+      last_detector_state = delay_update.detector_state;
>+      last_detector_switch = x;
>+
>+      switch (delay_update.detector_state) {
>+        case BandwidthUsage::kBwNormal:
>+          last_series = &normal_series;
>+          break;
>+        case BandwidthUsage::kBwUnderusing:
>+          last_series = &underusing_series;
>+          break;
>+        case BandwidthUsage::kBwOverusing:
>+          last_series = &overusing_series;
>+          break;
>+        case BandwidthUsage::kLast:
>+          RTC_NOTREACHED();
>       }
>     }
> 
>-    if (show_detector_state) {
>-      plot->AppendIntervalSeries(std::move(overusing_series));
>-      plot->AppendIntervalSeries(std::move(underusing_series));
>-      plot->AppendIntervalSeries(std::move(normal_series));
>-    }
>+    delay_series.points.emplace_back(x, y);
>+  }
>+
>+  RTC_CHECK(last_series);
>+  last_series->intervals.emplace_back(last_detector_switch, end_time_);
> 
>-    plot->AppendTimeSeries(std::move(loss_series));
>-    plot->AppendTimeSeries(std::move(delay_series));
>-    plot->AppendTimeSeries(std::move(created_series));
>-    plot->AppendTimeSeries(std::move(result_series));
>+  TimeSeries created_series("Probe cluster created.", LineStyle::kNone,
>+                            PointStyle::kHighlight);
>+  for (auto& cluster : parsed_log_.bwe_probe_cluster_created_events()) {
>+    float x = ToCallTimeSec(cluster.log_time_us());
>+    float y = static_cast<float>(cluster.bitrate_bps) / 1000;
>+    created_series.points.emplace_back(x, y);
>   }
> 
>-  // Overlay the incoming REMB over the outgoing bitrate
>-  // and outgoing REMB over incoming bitrate.
>-  PacketDirection remb_direction =
>-      desired_direction == kOutgoingPacket ? kIncomingPacket : kOutgoingPacket;
>-  TimeSeries remb_series("Remb", LineStyle::kStep);
>-  std::multimap<uint64_t, const LoggedRtcpPacket*> remb_packets;
>-  for (const auto& kv : rtcp_packets_) {
>-    if (kv.first.GetDirection() == remb_direction) {
>-      for (const LoggedRtcpPacket& rtcp_packet : kv.second) {
>-        if (rtcp_packet.type == kRtcpRemb) {
>-          remb_packets.insert(
>-              std::make_pair(rtcp_packet.timestamp, &rtcp_packet));
>-        }
>-      }
>+  TimeSeries result_series("Probing results.", LineStyle::kNone,
>+                           PointStyle::kHighlight);
>+  for (auto& result : parsed_log_.bwe_probe_success_events()) {
>+    float x = ToCallTimeSec(result.log_time_us());
>+    float y = static_cast<float>(result.bitrate_bps) / 1000;
>+    result_series.points.emplace_back(x, y);
>+  }
>+
>+  IntervalSeries alr_state("ALR", "#555555", IntervalSeries::kHorizontal);
>+  bool previously_in_alr = false;
>+  int64_t alr_start = 0;
>+  for (auto& alr : parsed_log_.alr_state_events()) {
>+    float y = ToCallTimeSec(alr.log_time_us());
>+    if (!previously_in_alr && alr.in_alr) {
>+      alr_start = alr.log_time_us();
>+      previously_in_alr = true;
>+    } else if (previously_in_alr && !alr.in_alr) {
>+      float x = ToCallTimeSec(alr_start);
>+      alr_state.intervals.emplace_back(x, y);
>+      previously_in_alr = false;
>     }
>   }
> 
>-  for (const auto& kv : remb_packets) {
>-    const LoggedRtcpPacket* const rtcp = kv.second;
>-    const rtcp::Remb* const remb = static_cast<rtcp::Remb*>(rtcp->packet.get());
>-    float x = static_cast<float>(rtcp->timestamp - begin_time_) / 1000000;
>-    float y = static_cast<float>(remb->bitrate_bps()) / 1000;
>+  if (previously_in_alr) {
>+    float x = ToCallTimeSec(alr_start);
>+    float y = ToCallTimeSec(end_time_);
>+    alr_state.intervals.emplace_back(x, y);
>+  }
>+
>+  if (show_detector_state) {
>+    plot->AppendIntervalSeries(std::move(overusing_series));
>+    plot->AppendIntervalSeries(std::move(underusing_series));
>+    plot->AppendIntervalSeries(std::move(normal_series));
>+  }
>+
>+  if (show_alr_state) {
>+    plot->AppendIntervalSeries(std::move(alr_state));
>+  }
>+  plot->AppendTimeSeries(std::move(loss_series));
>+  plot->AppendTimeSeries(std::move(delay_series));
>+  plot->AppendTimeSeries(std::move(created_series));
>+  plot->AppendTimeSeries(std::move(result_series));
>+
>+  // Overlay the incoming REMB over the outgoing bitrate.
>+  TimeSeries remb_series("Remb", LineStyle::kStep);
>+  for (const auto& rtcp : parsed_log_.rembs(kIncomingPacket)) {
>+    float x = ToCallTimeSec(rtcp.log_time_us());
>+    float y = static_cast<float>(rtcp.remb.bitrate_bps()) / 1000;
>     remb_series.points.emplace_back(x, y);
>   }
>   plot->AppendTimeSeriesIfNotEmpty(std::move(remb_series));
> 
>-  plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
>+  plot->SetXAxis(ToCallTimeSec(begin_time_), call_duration_s_, "Time (s)",
>+                 kLeftMargin, kRightMargin);
>   plot->SetSuggestedYAxis(0, 1, "Bitrate (kbps)", kBottomMargin, kTopMargin);
>-  if (desired_direction == webrtc::PacketDirection::kIncomingPacket) {
>-    plot->SetTitle("Incoming RTP bitrate");
>-  } else if (desired_direction == webrtc::PacketDirection::kOutgoingPacket) {
>-    plot->SetTitle("Outgoing RTP bitrate");
>-  }
>+  plot->SetTitle("Outgoing RTP bitrate");
> }
> 
> // For each SSRC, plot the bandwidth used by that stream.
>-void EventLogAnalyzer::CreateStreamBitrateGraph(
>-    PacketDirection desired_direction,
>-    Plot* plot) {
>-  for (auto& kv : rtp_packets_) {
>-    StreamId stream_id = kv.first;
>-    const std::vector<LoggedRtpPacket>& packet_stream = kv.second;
>-    // Filter on direction and SSRC.
>-    if (stream_id.GetDirection() != desired_direction ||
>-        !MatchingSsrc(stream_id.GetSsrc(), desired_ssrc_)) {
>+void EventLogAnalyzer::CreateStreamBitrateGraph(PacketDirection direction,
>+                                                Plot* plot) {
>+  for (const auto& stream : parsed_log_.rtp_packets_by_ssrc(direction)) {
>+    // Filter on SSRC.
>+    if (!MatchingSsrc(stream.ssrc, desired_ssrc_)) {
>       continue;
>     }
> 
>-    TimeSeries time_series(GetStreamName(stream_id), LineStyle::kLine);
>+    TimeSeries time_series(GetStreamName(direction, stream.ssrc),
>+                           LineStyle::kLine);
>+    auto GetPacketSizeKilobits = [](const LoggedRtpPacket& packet) {
>+      return packet.total_length * 8.0 / 1000.0;
>+    };
>+    auto ToCallTime = [this](int64_t time) {
>+      return this->ToCallTimeSec(time);
>+    };
>     MovingAverage<LoggedRtpPacket, double>(
>-        [](const LoggedRtpPacket& packet) {
>-          return packet.total_length * 8.0 / 1000.0;
>-        },
>-        packet_stream, begin_time_, end_time_, window_duration_, step_,
>-        &time_series);
>+        ToCallTime, GetPacketSizeKilobits, stream.packet_view, begin_time_,
>+        end_time_, window_duration_, step_, &time_series);
>     plot->AppendTimeSeries(std::move(time_series));
>   }
> 
>-  plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
>+  plot->SetXAxis(ToCallTimeSec(begin_time_), call_duration_s_, "Time (s)",
>+                 kLeftMargin, kRightMargin);
>   plot->SetSuggestedYAxis(0, 1, "Bitrate (kbps)", kBottomMargin, kTopMargin);
>-  if (desired_direction == webrtc::PacketDirection::kIncomingPacket) {
>-    plot->SetTitle("Incoming bitrate per stream");
>-  } else if (desired_direction == webrtc::PacketDirection::kOutgoingPacket) {
>-    plot->SetTitle("Outgoing bitrate per stream");
>-  }
>+  plot->SetTitle(GetDirectionAsString(direction) + " bitrate per stream");
> }
> 
> void EventLogAnalyzer::CreateSendSideBweSimulationGraph(Plot* plot) {
>-  std::multimap<uint64_t, const LoggedRtpPacket*> outgoing_rtp;
>-  std::multimap<uint64_t, const LoggedRtcpPacket*> incoming_rtcp;
>+  using RtpPacketType = LoggedRtpPacketOutgoing;
>+  using TransportFeedbackType = LoggedRtcpPacketTransportFeedback;
> 
>-  for (const auto& kv : rtp_packets_) {
>-    if (kv.first.GetDirection() == PacketDirection::kOutgoingPacket) {
>-      for (const LoggedRtpPacket& rtp_packet : kv.second)
>-        outgoing_rtp.insert(std::make_pair(rtp_packet.timestamp, &rtp_packet));
>-    }
>+  // TODO(terelius): This could be provided by the parser.
>+  std::multimap<int64_t, const RtpPacketType*> outgoing_rtp;
>+  for (const auto& stream : parsed_log_.outgoing_rtp_packets_by_ssrc()) {
>+    for (const RtpPacketType& rtp_packet : stream.outgoing_packets)
>+      outgoing_rtp.insert(
>+          std::make_pair(rtp_packet.rtp.log_time_us(), &rtp_packet));
>   }
> 
>-  for (const auto& kv : rtcp_packets_) {
>-    if (kv.first.GetDirection() == PacketDirection::kIncomingPacket) {
>-      for (const LoggedRtcpPacket& rtcp_packet : kv.second)
>-        incoming_rtcp.insert(
>-            std::make_pair(rtcp_packet.timestamp, &rtcp_packet));
>-    }
>-  }
>+  const std::vector<TransportFeedbackType>& incoming_rtcp =
>+      parsed_log_.transport_feedbacks(kIncomingPacket);
> 
>   SimulatedClock clock(0);
>   BitrateObserver observer;
>@@ -1209,7 +1104,7 @@ void EventLogAnalyzer::CreateSendSideBweSimulationGraph(Plot* plot) {
> 
>   auto NextRtcpTime = [&]() {
>     if (rtcp_iterator != incoming_rtcp.end())
>-      return static_cast<int64_t>(rtcp_iterator->first);
>+      return static_cast<int64_t>(rtcp_iterator->log_time_us());
>     return std::numeric_limits<int64_t>::max();
>   };
> 
>@@ -1232,7 +1127,7 @@ void EventLogAnalyzer::CreateSendSideBweSimulationGraph(Plot* plot) {
>   // we don't instantiate the AcknowledgedBitrateEstimator both here and in
>   // SendSideCongestionController since that would lead to duplicate outputs.
>   AcknowledgedBitrateEstimator acknowledged_bitrate_estimator(
>-      rtc::MakeUnique<BitrateEstimator>());
>+      absl::make_unique<BitrateEstimator>());
> #endif  // !(BWE_TEST_LOGGING_COMPILE_TIME_ENABLE)
>   int64_t time_us = std::min(NextRtpTime(), NextRtcpTime());
>   int64_t last_update_us = 0;
>@@ -1240,42 +1135,38 @@ void EventLogAnalyzer::CreateSendSideBweSimulationGraph(Plot* plot) {
>     clock.AdvanceTimeMicroseconds(time_us - clock.TimeInMicroseconds());
>     if (clock.TimeInMicroseconds() >= NextRtcpTime()) {
>       RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextRtcpTime());
>-      const LoggedRtcpPacket& rtcp = *rtcp_iterator->second;
>-      if (rtcp.type == kRtcpTransportFeedback) {
>-        cc.OnTransportFeedback(
>-            *static_cast<rtcp::TransportFeedback*>(rtcp.packet.get()));
>-        std::vector<PacketFeedback> feedback = cc.GetTransportFeedbackVector();
>-        SortPacketFeedbackVector(&feedback);
>-        rtc::Optional<uint32_t> bitrate_bps;
>-        if (!feedback.empty()) {
>+      cc.OnTransportFeedback(rtcp_iterator->transport_feedback);
>+      std::vector<PacketFeedback> feedback = cc.GetTransportFeedbackVector();
>+      SortPacketFeedbackVector(&feedback);
>+      absl::optional<uint32_t> bitrate_bps;
>+      if (!feedback.empty()) {
> #if !(BWE_TEST_LOGGING_COMPILE_TIME_ENABLE)
>-          acknowledged_bitrate_estimator.IncomingPacketFeedbackVector(feedback);
>+        acknowledged_bitrate_estimator.IncomingPacketFeedbackVector(feedback);
> #endif  // !(BWE_TEST_LOGGING_COMPILE_TIME_ENABLE)
>-          for (const PacketFeedback& packet : feedback)
>-            acked_bitrate.Update(packet.payload_size, packet.arrival_time_ms);
>-          bitrate_bps = acked_bitrate.Rate(feedback.back().arrival_time_ms);
>-        }
>-        float x = static_cast<float>(clock.TimeInMicroseconds() - begin_time_) /
>-                  1000000;
>-        float y = bitrate_bps.value_or(0) / 1000;
>-        acked_time_series.points.emplace_back(x, y);
>+        for (const PacketFeedback& packet : feedback)
>+          acked_bitrate.Update(packet.payload_size, packet.arrival_time_ms);
>+        bitrate_bps = acked_bitrate.Rate(feedback.back().arrival_time_ms);
>+      }
>+      float x = ToCallTimeSec(clock.TimeInMicroseconds());
>+      float y = bitrate_bps.value_or(0) / 1000;
>+      acked_time_series.points.emplace_back(x, y);
> #if !(BWE_TEST_LOGGING_COMPILE_TIME_ENABLE)
>-        y = acknowledged_bitrate_estimator.bitrate_bps().value_or(0) / 1000;
>-        acked_estimate_time_series.points.emplace_back(x, y);
>+      y = acknowledged_bitrate_estimator.bitrate_bps().value_or(0) / 1000;
>+      acked_estimate_time_series.points.emplace_back(x, y);
> #endif  // !(BWE_TEST_LOGGING_COMPILE_TIME_ENABLE)
>-      }
>       ++rtcp_iterator;
>     }
>     if (clock.TimeInMicroseconds() >= NextRtpTime()) {
>       RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextRtpTime());
>-      const LoggedRtpPacket& rtp = *rtp_iterator->second;
>-      if (rtp.header.extension.hasTransportSequenceNumber) {
>-        RTC_DCHECK(rtp.header.extension.hasTransportSequenceNumber);
>-        cc.AddPacket(rtp.header.ssrc,
>-                     rtp.header.extension.transportSequenceNumber,
>-                     rtp.total_length, PacedPacketInfo());
>+      const RtpPacketType& rtp_packet = *rtp_iterator->second;
>+      if (rtp_packet.rtp.header.extension.hasTransportSequenceNumber) {
>+        RTC_DCHECK(rtp_packet.rtp.header.extension.hasTransportSequenceNumber);
>+        cc.AddPacket(rtp_packet.rtp.header.ssrc,
>+                     rtp_packet.rtp.header.extension.transportSequenceNumber,
>+                     rtp_packet.rtp.total_length, PacedPacketInfo());
>         rtc::SentPacket sent_packet(
>-            rtp.header.extension.transportSequenceNumber, rtp.timestamp / 1000);
>+            rtp_packet.rtp.header.extension.transportSequenceNumber,
>+            rtp_packet.rtp.log_time_us() / 1000);
>         cc.OnSentPacket(sent_packet);
>       }
>       ++rtp_iterator;
>@@ -1287,8 +1178,7 @@ void EventLogAnalyzer::CreateSendSideBweSimulationGraph(Plot* plot) {
>     if (observer.GetAndResetBitrateUpdated() ||
>         time_us - last_update_us >= 1e6) {
>       uint32_t y = observer.last_bitrate_bps() / 1000;
>-      float x = static_cast<float>(clock.TimeInMicroseconds() - begin_time_) /
>-                1000000;
>+      float x = ToCallTimeSec(clock.TimeInMicroseconds());
>       time_series.points.emplace_back(x, y);
>       last_update_us = time_us;
>     }
>@@ -1299,12 +1189,14 @@ void EventLogAnalyzer::CreateSendSideBweSimulationGraph(Plot* plot) {
>   plot->AppendTimeSeries(std::move(acked_time_series));
>   plot->AppendTimeSeriesIfNotEmpty(std::move(acked_estimate_time_series));
> 
>-  plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
>+  plot->SetXAxis(ToCallTimeSec(begin_time_), call_duration_s_, "Time (s)",
>+                 kLeftMargin, kRightMargin);
>   plot->SetSuggestedYAxis(0, 10, "Bitrate (kbps)", kBottomMargin, kTopMargin);
>   plot->SetTitle("Simulated send-side BWE behavior");
> }
> 
> void EventLogAnalyzer::CreateReceiveSideBweSimulationGraph(Plot* plot) {
>+  using RtpPacketType = LoggedRtpPacketIncoming;
>   class RembInterceptingPacketRouter : public PacketRouter {
>    public:
>     void OnReceiveBitrateChanged(const std::vector<uint32_t>& ssrcs,
>@@ -1325,19 +1217,19 @@ void EventLogAnalyzer::CreateReceiveSideBweSimulationGraph(Plot* plot) {
>     bool bitrate_updated_;
>   };
> 
>-  std::multimap<uint64_t, const LoggedRtpPacket*> incoming_rtp;
>+  std::multimap<int64_t, const RtpPacketType*> incoming_rtp;
> 
>-  for (const auto& kv : rtp_packets_) {
>-    if (kv.first.GetDirection() == PacketDirection::kIncomingPacket &&
>-        IsVideoSsrc(kv.first)) {
>-      for (const LoggedRtpPacket& rtp_packet : kv.second)
>-        incoming_rtp.insert(std::make_pair(rtp_packet.timestamp, &rtp_packet));
>+  for (const auto& stream : parsed_log_.incoming_rtp_packets_by_ssrc()) {
>+    if (IsVideoSsrc(kIncomingPacket, stream.ssrc)) {
>+      for (const auto& rtp_packet : stream.incoming_packets)
>+        incoming_rtp.insert(
>+            std::make_pair(rtp_packet.rtp.log_time_us(), &rtp_packet));
>     }
>   }
> 
>   SimulatedClock clock(0);
>   RembInterceptingPacketRouter packet_router;
>-  // TODO(terelius): The PacketRrouter is the used as the RemoteBitrateObserver.
>+  // TODO(terelius): The PacketRouter is used as the RemoteBitrateObserver.
>   // Is this intentional?
>   ReceiveSideCongestionController rscc(&clock, &packet_router);
>   // TODO(holmer): Log the call config and use that here instead.
>@@ -1351,25 +1243,23 @@ void EventLogAnalyzer::CreateReceiveSideBweSimulationGraph(Plot* plot) {
>   RateStatistics acked_bitrate(250, 8000);
>   int64_t last_update_us = 0;
>   for (const auto& kv : incoming_rtp) {
>-    const LoggedRtpPacket& packet = *kv.second;
>-    int64_t arrival_time_ms = packet.timestamp / 1000;
>-    size_t payload = packet.total_length; /*Should subtract header?*/
>-    clock.AdvanceTimeMicroseconds(packet.timestamp -
>+    const RtpPacketType& packet = *kv.second;
>+    int64_t arrival_time_ms = packet.rtp.log_time_us() / 1000;
>+    size_t payload = packet.rtp.total_length; /*Should subtract header?*/
>+    clock.AdvanceTimeMicroseconds(packet.rtp.log_time_us() -
>                                   clock.TimeInMicroseconds());
>-    rscc.OnReceivedPacket(arrival_time_ms, payload, packet.header);
>+    rscc.OnReceivedPacket(arrival_time_ms, payload, packet.rtp.header);
>     acked_bitrate.Update(payload, arrival_time_ms);
>-    rtc::Optional<uint32_t> bitrate_bps = acked_bitrate.Rate(arrival_time_ms);
>+    absl::optional<uint32_t> bitrate_bps = acked_bitrate.Rate(arrival_time_ms);
>     if (bitrate_bps) {
>       uint32_t y = *bitrate_bps / 1000;
>-      float x = static_cast<float>(clock.TimeInMicroseconds() - begin_time_) /
>-                1000000;
>+      float x = ToCallTimeSec(clock.TimeInMicroseconds());
>       acked_time_series.points.emplace_back(x, y);
>     }
>     if (packet_router.GetAndResetBitrateUpdated() ||
>         clock.TimeInMicroseconds() - last_update_us >= 1e6) {
>       uint32_t y = packet_router.last_bitrate_bps() / 1000;
>-      float x = static_cast<float>(clock.TimeInMicroseconds() - begin_time_) /
>-                1000000;
>+      float x = ToCallTimeSec(clock.TimeInMicroseconds());
>       time_series.points.emplace_back(x, y);
>       last_update_us = clock.TimeInMicroseconds();
>     }
>@@ -1378,29 +1268,26 @@ void EventLogAnalyzer::CreateReceiveSideBweSimulationGraph(Plot* plot) {
>   plot->AppendTimeSeries(std::move(time_series));
>   plot->AppendTimeSeries(std::move(acked_time_series));
> 
>-  plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
>+  plot->SetXAxis(ToCallTimeSec(begin_time_), call_duration_s_, "Time (s)",
>+                 kLeftMargin, kRightMargin);
>   plot->SetSuggestedYAxis(0, 10, "Bitrate (kbps)", kBottomMargin, kTopMargin);
>   plot->SetTitle("Simulated receive-side BWE behavior");
> }
> 
> void EventLogAnalyzer::CreateNetworkDelayFeedbackGraph(Plot* plot) {
>-  std::multimap<uint64_t, const LoggedRtpPacket*> outgoing_rtp;
>-  std::multimap<uint64_t, const LoggedRtcpPacket*> incoming_rtcp;
>+  using RtpPacketType = LoggedRtpPacketOutgoing;
>+  using TransportFeedbackType = LoggedRtcpPacketTransportFeedback;
> 
>-  for (const auto& kv : rtp_packets_) {
>-    if (kv.first.GetDirection() == PacketDirection::kOutgoingPacket) {
>-      for (const LoggedRtpPacket& rtp_packet : kv.second)
>-        outgoing_rtp.insert(std::make_pair(rtp_packet.timestamp, &rtp_packet));
>-    }
>+  // TODO(terelius): This could be provided by the parser.
>+  std::multimap<int64_t, const RtpPacketType*> outgoing_rtp;
>+  for (const auto& stream : parsed_log_.outgoing_rtp_packets_by_ssrc()) {
>+    for (const RtpPacketType& rtp_packet : stream.outgoing_packets)
>+      outgoing_rtp.insert(
>+          std::make_pair(rtp_packet.rtp.log_time_us(), &rtp_packet));
>   }
> 
>-  for (const auto& kv : rtcp_packets_) {
>-    if (kv.first.GetDirection() == PacketDirection::kIncomingPacket) {
>-      for (const LoggedRtcpPacket& rtcp_packet : kv.second)
>-        incoming_rtcp.insert(
>-            std::make_pair(rtcp_packet.timestamp, &rtcp_packet));
>-    }
>-  }
>+  const std::vector<TransportFeedbackType>& incoming_rtcp =
>+      parsed_log_.transport_feedbacks(kIncomingPacket);
> 
>   SimulatedClock clock(0);
>   TransportFeedbackAdapter feedback_adapter(&clock);
>@@ -1422,7 +1309,7 @@ void EventLogAnalyzer::CreateNetworkDelayFeedbackGraph(Plot* plot) {
> 
>   auto NextRtcpTime = [&]() {
>     if (rtcp_iterator != incoming_rtcp.end())
>-      return static_cast<int64_t>(rtcp_iterator->first);
>+      return static_cast<int64_t>(rtcp_iterator->log_time_us());
>     return std::numeric_limits<int64_t>::max();
>   };
> 
>@@ -1432,39 +1319,34 @@ void EventLogAnalyzer::CreateNetworkDelayFeedbackGraph(Plot* plot) {
>     clock.AdvanceTimeMicroseconds(time_us - clock.TimeInMicroseconds());
>     if (clock.TimeInMicroseconds() >= NextRtcpTime()) {
>       RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextRtcpTime());
>-      const LoggedRtcpPacket& rtcp = *rtcp_iterator->second;
>-      if (rtcp.type == kRtcpTransportFeedback) {
>-        feedback_adapter.OnTransportFeedback(
>-            *static_cast<rtcp::TransportFeedback*>(rtcp.packet.get()));
>-        std::vector<PacketFeedback> feedback =
>-            feedback_adapter.GetTransportFeedbackVector();
>-        SortPacketFeedbackVector(&feedback);
>-        for (const PacketFeedback& packet : feedback) {
>-          float x =
>-              static_cast<float>(clock.TimeInMicroseconds() - begin_time_) /
>-              1000000;
>-          if (packet.send_time_ms == -1) {
>-            late_feedback_series.points.emplace_back(x, prev_y);
>-            continue;
>-          }
>-          int64_t y = packet.arrival_time_ms - packet.send_time_ms;
>-          prev_y = y;
>-          estimated_base_delay_ms = std::min(y, estimated_base_delay_ms);
>-          time_series.points.emplace_back(x, y);
>+      feedback_adapter.OnTransportFeedback(rtcp_iterator->transport_feedback);
>+      std::vector<PacketFeedback> feedback =
>+          feedback_adapter.GetTransportFeedbackVector();
>+      SortPacketFeedbackVector(&feedback);
>+      for (const PacketFeedback& packet : feedback) {
>+        float x = ToCallTimeSec(clock.TimeInMicroseconds());
>+        if (packet.send_time_ms == PacketFeedback::kNoSendTime) {
>+          late_feedback_series.points.emplace_back(x, prev_y);
>+          continue;
>         }
>+        int64_t y = packet.arrival_time_ms - packet.send_time_ms;
>+        prev_y = y;
>+        estimated_base_delay_ms = std::min(y, estimated_base_delay_ms);
>+        time_series.points.emplace_back(x, y);
>       }
>       ++rtcp_iterator;
>     }
>     if (clock.TimeInMicroseconds() >= NextRtpTime()) {
>       RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextRtpTime());
>-      const LoggedRtpPacket& rtp = *rtp_iterator->second;
>-      if (rtp.header.extension.hasTransportSequenceNumber) {
>-        RTC_DCHECK(rtp.header.extension.hasTransportSequenceNumber);
>-        feedback_adapter.AddPacket(rtp.header.ssrc,
>-                                   rtp.header.extension.transportSequenceNumber,
>-                                   rtp.total_length, PacedPacketInfo());
>+      const RtpPacketType& rtp_packet = *rtp_iterator->second;
>+      if (rtp_packet.rtp.header.extension.hasTransportSequenceNumber) {
>+        feedback_adapter.AddPacket(
>+            rtp_packet.rtp.header.ssrc,
>+            rtp_packet.rtp.header.extension.transportSequenceNumber,
>+            rtp_packet.rtp.total_length, PacedPacketInfo());
>         feedback_adapter.OnSentPacket(
>-            rtp.header.extension.transportSequenceNumber, rtp.timestamp / 1000);
>+            rtp_packet.rtp.header.extension.transportSequenceNumber,
>+            rtp_packet.rtp.log_time_us() / 1000);
>       }
>       ++rtp_iterator;
>     }
>@@ -1480,45 +1362,16 @@ void EventLogAnalyzer::CreateNetworkDelayFeedbackGraph(Plot* plot) {
>   plot->AppendTimeSeriesIfNotEmpty(std::move(time_series));
>   plot->AppendTimeSeriesIfNotEmpty(std::move(late_feedback_series));
> 
>-  plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
>+  plot->SetXAxis(ToCallTimeSec(begin_time_), call_duration_s_, "Time (s)",
>+                 kLeftMargin, kRightMargin);
>   plot->SetSuggestedYAxis(0, 10, "Delay (ms)", kBottomMargin, kTopMargin);
>   plot->SetTitle("Network Delay Change.");
> }
> 
>-std::vector<std::pair<int64_t, int64_t>> EventLogAnalyzer::GetFrameTimestamps()
>-    const {
>-  std::vector<std::pair<int64_t, int64_t>> timestamps;
>-  size_t largest_stream_size = 0;
>-  const std::vector<LoggedRtpPacket>* largest_video_stream = nullptr;
>-  // Find the incoming video stream with the most number of packets that is
>-  // not rtx.
>-  for (const auto& kv : rtp_packets_) {
>-    if (kv.first.GetDirection() == kIncomingPacket &&
>-        video_ssrcs_.find(kv.first) != video_ssrcs_.end() &&
>-        rtx_ssrcs_.find(kv.first) == rtx_ssrcs_.end() &&
>-        kv.second.size() > largest_stream_size) {
>-      largest_stream_size = kv.second.size();
>-      largest_video_stream = &kv.second;
>-    }
>-  }
>-  if (largest_video_stream == nullptr) {
>-    for (auto& packet : *largest_video_stream) {
>-      if (packet.header.markerBit) {
>-        int64_t capture_ms = packet.header.timestamp / 90.0;
>-        int64_t arrival_ms = packet.timestamp / 1000.0;
>-        timestamps.push_back(std::make_pair(capture_ms, arrival_ms));
>-      }
>-    }
>-  }
>-  return timestamps;
>-}
>-
> void EventLogAnalyzer::CreatePacerDelayGraph(Plot* plot) {
>-  for (const auto& kv : rtp_packets_) {
>-    const std::vector<LoggedRtpPacket>& packets = kv.second;
>-    StreamId stream_id = kv.first;
>-    if (stream_id.GetDirection() == kIncomingPacket)
>-      continue;
>+  for (const auto& stream : parsed_log_.outgoing_rtp_packets_by_ssrc()) {
>+    const std::vector<LoggedRtpPacketOutgoing>& packets =
>+        stream.outgoing_packets;
> 
>     if (packets.size() < 2) {
>       RTC_LOG(LS_WARNING)
>@@ -1526,11 +1379,15 @@ void EventLogAnalyzer::CreatePacerDelayGraph(Plot* plot) {
>              "pacer delay with less than 2 packets in the stream";
>       continue;
>     }
>-    rtc::Optional<uint32_t> estimated_frequency =
>-        EstimateRtpClockFrequency(packets);
>+    int64_t end_time_us = log_segments_.empty()
>+                              ? std::numeric_limits<int64_t>::max()
>+                              : log_segments_.front().second;
>+    absl::optional<uint32_t> estimated_frequency =
>+        EstimateRtpClockFrequency(packets, end_time_us);
>     if (!estimated_frequency)
>       continue;
>-    if (IsVideoSsrc(stream_id) && *estimated_frequency != 90000) {
>+    if (IsVideoSsrc(kOutgoingPacket, stream.ssrc) &&
>+        *estimated_frequency != 90000) {
>       RTC_LOG(LS_WARNING)
>           << "Video stream should use a 90 kHz clock but appears to use "
>           << *estimated_frequency / 1000 << ". Discarding.";
>@@ -1538,87 +1395,143 @@ void EventLogAnalyzer::CreatePacerDelayGraph(Plot* plot) {
>     }
> 
>     TimeSeries pacer_delay_series(
>-        GetStreamName(stream_id) + "(" +
>+        GetStreamName(kOutgoingPacket, stream.ssrc) + "(" +
>             std::to_string(*estimated_frequency / 1000) + " kHz)",
>         LineStyle::kLine, PointStyle::kHighlight);
>     SeqNumUnwrapper<uint32_t> timestamp_unwrapper;
>     uint64_t first_capture_timestamp =
>-        timestamp_unwrapper.Unwrap(packets.front().header.timestamp);
>-    uint64_t first_send_timestamp = packets.front().timestamp;
>-    for (LoggedRtpPacket packet : packets) {
>+        timestamp_unwrapper.Unwrap(packets.front().rtp.header.timestamp);
>+    uint64_t first_send_timestamp = packets.front().rtp.log_time_us();
>+    for (const auto& packet : packets) {
>       double capture_time_ms = (static_cast<double>(timestamp_unwrapper.Unwrap(
>-                                    packet.header.timestamp)) -
>+                                    packet.rtp.header.timestamp)) -
>                                 first_capture_timestamp) /
>                                *estimated_frequency * 1000;
>       double send_time_ms =
>-          static_cast<double>(packet.timestamp - first_send_timestamp) / 1000;
>-      float x = static_cast<float>(packet.timestamp - begin_time_) / 1000000;
>+          static_cast<double>(packet.rtp.log_time_us() - first_send_timestamp) /
>+          1000;
>+      float x = ToCallTimeSec(packet.rtp.log_time_us());
>       float y = send_time_ms - capture_time_ms;
>       pacer_delay_series.points.emplace_back(x, y);
>     }
>     plot->AppendTimeSeries(std::move(pacer_delay_series));
>   }
> 
>-  plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
>+  plot->SetXAxis(ToCallTimeSec(begin_time_), call_duration_s_, "Time (s)",
>+                 kLeftMargin, kRightMargin);
>   plot->SetSuggestedYAxis(0, 10, "Pacer delay (ms)", kBottomMargin, kTopMargin);
>   plot->SetTitle(
>       "Delay from capture to send time. (First packet normalized to 0.)");
> }
> 
>-void EventLogAnalyzer::CreateTimestampGraph(Plot* plot) {
>-  for (const auto& kv : rtp_packets_) {
>-    const std::vector<LoggedRtpPacket>& rtp_packets = kv.second;
>-    StreamId stream_id = kv.first;
>-
>-    {
>-      TimeSeries timestamp_data(GetStreamName(stream_id) + " capture-time",
>-                                LineStyle::kLine, PointStyle::kHighlight);
>-      for (LoggedRtpPacket packet : rtp_packets) {
>-        float x = static_cast<float>(packet.timestamp - begin_time_) / 1000000;
>-        float y = packet.header.timestamp;
>-        timestamp_data.points.emplace_back(x, y);
>+void EventLogAnalyzer::CreateTimestampGraph(PacketDirection direction,
>+                                            Plot* plot) {
>+  for (const auto& stream : parsed_log_.rtp_packets_by_ssrc(direction)) {
>+    TimeSeries rtp_timestamps(
>+        GetStreamName(direction, stream.ssrc) + " capture-time",
>+        LineStyle::kLine, PointStyle::kHighlight);
>+    for (const auto& packet : stream.packet_view) {
>+      float x = ToCallTimeSec(packet.log_time_us());
>+      float y = packet.header.timestamp;
>+      rtp_timestamps.points.emplace_back(x, y);
>+    }
>+    plot->AppendTimeSeries(std::move(rtp_timestamps));
>+
>+    TimeSeries rtcp_timestamps(
>+        GetStreamName(direction, stream.ssrc) + " rtcp capture-time",
>+        LineStyle::kLine, PointStyle::kHighlight);
>+    // TODO(terelius): Why only sender reports?
>+    const auto& sender_reports = parsed_log_.sender_reports(direction);
>+    for (const auto& rtcp : sender_reports) {
>+      if (rtcp.sr.sender_ssrc() != stream.ssrc)
>+        continue;
>+      float x = ToCallTimeSec(rtcp.log_time_us());
>+      float y = rtcp.sr.rtp_timestamp();
>+      rtcp_timestamps.points.emplace_back(x, y);
>+    }
>+    plot->AppendTimeSeriesIfNotEmpty(std::move(rtcp_timestamps));
>+  }
>+
>+  plot->SetXAxis(ToCallTimeSec(begin_time_), call_duration_s_, "Time (s)",
>+                 kLeftMargin, kRightMargin);
>+  plot->SetSuggestedYAxis(0, 1, "RTP timestamp", kBottomMargin, kTopMargin);
>+  plot->SetTitle(GetDirectionAsString(direction) + " timestamps");
>+}
>+
>+void EventLogAnalyzer::CreateSenderAndReceiverReportPlot(
>+    PacketDirection direction,
>+    rtc::FunctionView<float(const rtcp::ReportBlock&)> fy,
>+    std::string title,
>+    std::string yaxis_label,
>+    Plot* plot) {
>+  std::map<uint32_t, TimeSeries> sr_reports_by_ssrc;
>+  const auto& sender_reports = parsed_log_.sender_reports(direction);
>+  for (const auto& rtcp : sender_reports) {
>+    float x = ToCallTimeSec(rtcp.log_time_us());
>+    uint32_t ssrc = rtcp.sr.sender_ssrc();
>+    for (const auto& block : rtcp.sr.report_blocks()) {
>+      float y = fy(block);
>+      auto sr_report_it = sr_reports_by_ssrc.find(ssrc);
>+      bool inserted;
>+      if (sr_report_it == sr_reports_by_ssrc.end()) {
>+        std::tie(sr_report_it, inserted) = sr_reports_by_ssrc.emplace(
>+            ssrc, TimeSeries(GetStreamName(direction, ssrc) + " Sender Reports",
>+                             LineStyle::kLine, PointStyle::kHighlight));
>       }
>-      plot->AppendTimeSeries(std::move(timestamp_data));
>+      sr_report_it->second.points.emplace_back(x, y);
>     }
>+  }
>+  for (auto& kv : sr_reports_by_ssrc) {
>+    plot->AppendTimeSeries(std::move(kv.second));
>+  }
> 
>-    {
>-      auto kv = rtcp_packets_.find(stream_id);
>-      if (kv != rtcp_packets_.end()) {
>-        const auto& packets = kv->second;
>-        TimeSeries timestamp_data(
>-            GetStreamName(stream_id) + " rtcp capture-time", LineStyle::kLine,
>-            PointStyle::kHighlight);
>-        for (const LoggedRtcpPacket& rtcp : packets) {
>-          if (rtcp.type != kRtcpSr)
>-            continue;
>-          rtcp::SenderReport* sr;
>-          sr = static_cast<rtcp::SenderReport*>(rtcp.packet.get());
>-          float x = static_cast<float>(rtcp.timestamp - begin_time_) / 1000000;
>-          float y = sr->rtp_timestamp();
>-          timestamp_data.points.emplace_back(x, y);
>-        }
>-        plot->AppendTimeSeries(std::move(timestamp_data));
>+  std::map<uint32_t, TimeSeries> rr_reports_by_ssrc;
>+  const auto& receiver_reports = parsed_log_.receiver_reports(direction);
>+  for (const auto& rtcp : receiver_reports) {
>+    float x = ToCallTimeSec(rtcp.log_time_us());
>+    uint32_t ssrc = rtcp.rr.sender_ssrc();
>+    for (const auto& block : rtcp.rr.report_blocks()) {
>+      float y = fy(block);
>+      auto rr_report_it = rr_reports_by_ssrc.find(ssrc);
>+      bool inserted;
>+      if (rr_report_it == rr_reports_by_ssrc.end()) {
>+        std::tie(rr_report_it, inserted) = rr_reports_by_ssrc.emplace(
>+            ssrc,
>+            TimeSeries(GetStreamName(direction, ssrc) + " Receiver Reports",
>+                       LineStyle::kLine, PointStyle::kHighlight));
>       }
>+      rr_report_it->second.points.emplace_back(x, y);
>     }
>   }
>+  for (auto& kv : rr_reports_by_ssrc) {
>+    plot->AppendTimeSeries(std::move(kv.second));
>+  }
> 
>-  plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
>-  plot->SetSuggestedYAxis(0, 1, "Timestamp (90khz)", kBottomMargin, kTopMargin);
>-  plot->SetTitle("Timestamps");
>+  plot->SetXAxis(ToCallTimeSec(begin_time_), call_duration_s_, "Time (s)",
>+                 kLeftMargin, kRightMargin);
>+  plot->SetSuggestedYAxis(0, 1, yaxis_label, kBottomMargin, kTopMargin);
>+  plot->SetTitle(title);
> }
> 
> void EventLogAnalyzer::CreateAudioEncoderTargetBitrateGraph(Plot* plot) {
>   TimeSeries time_series("Audio encoder target bitrate", LineStyle::kLine,
>                          PointStyle::kHighlight);
>-  ProcessPoints<AudioNetworkAdaptationEvent>(
>-      [](const AudioNetworkAdaptationEvent& ana_event) -> rtc::Optional<float> {
>-        if (ana_event.config.bitrate_bps)
>-          return static_cast<float>(*ana_event.config.bitrate_bps);
>-        return rtc::nullopt;
>-      },
>-      audio_network_adaptation_events_, begin_time_, &time_series);
>+  auto GetAnaBitrateBps = [](const LoggedAudioNetworkAdaptationEvent& ana_event)
>+      -> absl::optional<float> {
>+    if (ana_event.config.bitrate_bps)
>+      return absl::optional<float>(
>+          static_cast<float>(*ana_event.config.bitrate_bps));
>+    return absl::nullopt;
>+  };
>+  auto ToCallTime = [this](const LoggedAudioNetworkAdaptationEvent& packet) {
>+    return this->ToCallTimeSec(packet.log_time_us());
>+  };
>+  ProcessPoints<LoggedAudioNetworkAdaptationEvent>(
>+      ToCallTime, GetAnaBitrateBps,
>+      parsed_log_.audio_network_adaptation_events(), &time_series);
>   plot->AppendTimeSeries(std::move(time_series));
>-  plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
>+  plot->SetXAxis(ToCallTimeSec(begin_time_), call_duration_s_, "Time (s)",
>+                 kLeftMargin, kRightMargin);
>   plot->SetSuggestedYAxis(0, 1, "Bitrate (bps)", kBottomMargin, kTopMargin);
>   plot->SetTitle("Reported audio encoder target bitrate");
> }
>@@ -1626,16 +1539,22 @@ void EventLogAnalyzer::CreateAudioEncoderTargetBitrateGraph(Plot* plot) {
> void EventLogAnalyzer::CreateAudioEncoderFrameLengthGraph(Plot* plot) {
>   TimeSeries time_series("Audio encoder frame length", LineStyle::kLine,
>                          PointStyle::kHighlight);
>-  ProcessPoints<AudioNetworkAdaptationEvent>(
>-      [](const AudioNetworkAdaptationEvent& ana_event) {
>+  auto GetAnaFrameLengthMs =
>+      [](const LoggedAudioNetworkAdaptationEvent& ana_event) {
>         if (ana_event.config.frame_length_ms)
>-          return rtc::Optional<float>(
>+          return absl::optional<float>(
>               static_cast<float>(*ana_event.config.frame_length_ms));
>-        return rtc::Optional<float>();
>-      },
>-      audio_network_adaptation_events_, begin_time_, &time_series);
>+        return absl::optional<float>();
>+      };
>+  auto ToCallTime = [this](const LoggedAudioNetworkAdaptationEvent& packet) {
>+    return this->ToCallTimeSec(packet.log_time_us());
>+  };
>+  ProcessPoints<LoggedAudioNetworkAdaptationEvent>(
>+      ToCallTime, GetAnaFrameLengthMs,
>+      parsed_log_.audio_network_adaptation_events(), &time_series);
>   plot->AppendTimeSeries(std::move(time_series));
>-  plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
>+  plot->SetXAxis(ToCallTimeSec(begin_time_), call_duration_s_, "Time (s)",
>+                 kLeftMargin, kRightMargin);
>   plot->SetSuggestedYAxis(0, 1, "Frame length (ms)", kBottomMargin, kTopMargin);
>   plot->SetTitle("Reported audio encoder frame length");
> }
>@@ -1643,16 +1562,22 @@ void EventLogAnalyzer::CreateAudioEncoderFrameLengthGraph(Plot* plot) {
> void EventLogAnalyzer::CreateAudioEncoderPacketLossGraph(Plot* plot) {
>   TimeSeries time_series("Audio encoder uplink packet loss fraction",
>                          LineStyle::kLine, PointStyle::kHighlight);
>-  ProcessPoints<AudioNetworkAdaptationEvent>(
>-      [](const AudioNetworkAdaptationEvent& ana_event) {
>+  auto GetAnaPacketLoss =
>+      [](const LoggedAudioNetworkAdaptationEvent& ana_event) {
>         if (ana_event.config.uplink_packet_loss_fraction)
>-          return rtc::Optional<float>(static_cast<float>(
>+          return absl::optional<float>(static_cast<float>(
>               *ana_event.config.uplink_packet_loss_fraction));
>-        return rtc::Optional<float>();
>-      },
>-      audio_network_adaptation_events_, begin_time_, &time_series);
>+        return absl::optional<float>();
>+      };
>+  auto ToCallTime = [this](const LoggedAudioNetworkAdaptationEvent& packet) {
>+    return this->ToCallTimeSec(packet.log_time_us());
>+  };
>+  ProcessPoints<LoggedAudioNetworkAdaptationEvent>(
>+      ToCallTime, GetAnaPacketLoss,
>+      parsed_log_.audio_network_adaptation_events(), &time_series);
>   plot->AppendTimeSeries(std::move(time_series));
>-  plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
>+  plot->SetXAxis(ToCallTimeSec(begin_time_), call_duration_s_, "Time (s)",
>+                 kLeftMargin, kRightMargin);
>   plot->SetSuggestedYAxis(0, 10, "Percent lost packets", kBottomMargin,
>                           kTopMargin);
>   plot->SetTitle("Reported audio encoder lost packets");
>@@ -1661,16 +1586,22 @@ void EventLogAnalyzer::CreateAudioEncoderPacketLossGraph(Plot* plot) {
> void EventLogAnalyzer::CreateAudioEncoderEnableFecGraph(Plot* plot) {
>   TimeSeries time_series("Audio encoder FEC", LineStyle::kLine,
>                          PointStyle::kHighlight);
>-  ProcessPoints<AudioNetworkAdaptationEvent>(
>-      [](const AudioNetworkAdaptationEvent& ana_event) {
>+  auto GetAnaFecEnabled =
>+      [](const LoggedAudioNetworkAdaptationEvent& ana_event) {
>         if (ana_event.config.enable_fec)
>-          return rtc::Optional<float>(
>+          return absl::optional<float>(
>               static_cast<float>(*ana_event.config.enable_fec));
>-        return rtc::Optional<float>();
>-      },
>-      audio_network_adaptation_events_, begin_time_, &time_series);
>+        return absl::optional<float>();
>+      };
>+  auto ToCallTime = [this](const LoggedAudioNetworkAdaptationEvent& packet) {
>+    return this->ToCallTimeSec(packet.log_time_us());
>+  };
>+  ProcessPoints<LoggedAudioNetworkAdaptationEvent>(
>+      ToCallTime, GetAnaFecEnabled,
>+      parsed_log_.audio_network_adaptation_events(), &time_series);
>   plot->AppendTimeSeries(std::move(time_series));
>-  plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
>+  plot->SetXAxis(ToCallTimeSec(begin_time_), call_duration_s_, "Time (s)",
>+                 kLeftMargin, kRightMargin);
>   plot->SetSuggestedYAxis(0, 1, "FEC (false/true)", kBottomMargin, kTopMargin);
>   plot->SetTitle("Reported audio encoder FEC");
> }
>@@ -1678,16 +1609,22 @@ void EventLogAnalyzer::CreateAudioEncoderEnableFecGraph(Plot* plot) {
> void EventLogAnalyzer::CreateAudioEncoderEnableDtxGraph(Plot* plot) {
>   TimeSeries time_series("Audio encoder DTX", LineStyle::kLine,
>                          PointStyle::kHighlight);
>-  ProcessPoints<AudioNetworkAdaptationEvent>(
>-      [](const AudioNetworkAdaptationEvent& ana_event) {
>+  auto GetAnaDtxEnabled =
>+      [](const LoggedAudioNetworkAdaptationEvent& ana_event) {
>         if (ana_event.config.enable_dtx)
>-          return rtc::Optional<float>(
>+          return absl::optional<float>(
>               static_cast<float>(*ana_event.config.enable_dtx));
>-        return rtc::Optional<float>();
>-      },
>-      audio_network_adaptation_events_, begin_time_, &time_series);
>+        return absl::optional<float>();
>+      };
>+  auto ToCallTime = [this](const LoggedAudioNetworkAdaptationEvent& packet) {
>+    return this->ToCallTimeSec(packet.log_time_us());
>+  };
>+  ProcessPoints<LoggedAudioNetworkAdaptationEvent>(
>+      ToCallTime, GetAnaDtxEnabled,
>+      parsed_log_.audio_network_adaptation_events(), &time_series);
>   plot->AppendTimeSeries(std::move(time_series));
>-  plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
>+  plot->SetXAxis(ToCallTimeSec(begin_time_), call_duration_s_, "Time (s)",
>+                 kLeftMargin, kRightMargin);
>   plot->SetSuggestedYAxis(0, 1, "DTX (false/true)", kBottomMargin, kTopMargin);
>   plot->SetTitle("Reported audio encoder DTX");
> }
>@@ -1695,16 +1632,22 @@ void EventLogAnalyzer::CreateAudioEncoderEnableDtxGraph(Plot* plot) {
> void EventLogAnalyzer::CreateAudioEncoderNumChannelsGraph(Plot* plot) {
>   TimeSeries time_series("Audio encoder number of channels", LineStyle::kLine,
>                          PointStyle::kHighlight);
>-  ProcessPoints<AudioNetworkAdaptationEvent>(
>-      [](const AudioNetworkAdaptationEvent& ana_event) {
>+  auto GetAnaNumChannels =
>+      [](const LoggedAudioNetworkAdaptationEvent& ana_event) {
>         if (ana_event.config.num_channels)
>-          return rtc::Optional<float>(
>+          return absl::optional<float>(
>               static_cast<float>(*ana_event.config.num_channels));
>-        return rtc::Optional<float>();
>-      },
>-      audio_network_adaptation_events_, begin_time_, &time_series);
>+        return absl::optional<float>();
>+      };
>+  auto ToCallTime = [this](const LoggedAudioNetworkAdaptationEvent& packet) {
>+    return this->ToCallTimeSec(packet.log_time_us());
>+  };
>+  ProcessPoints<LoggedAudioNetworkAdaptationEvent>(
>+      ToCallTime, GetAnaNumChannels,
>+      parsed_log_.audio_network_adaptation_events(), &time_series);
>   plot->AppendTimeSeries(std::move(time_series));
>-  plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
>+  plot->SetXAxis(ToCallTimeSec(begin_time_), call_duration_s_, "Time (s)",
>+                 kLeftMargin, kRightMargin);
>   plot->SetSuggestedYAxis(0, 1, "Number of channels (1 (mono)/2 (stereo))",
>                           kBottomMargin, kTopMargin);
>   plot->SetTitle("Reported audio encoder number of channels");
>@@ -1714,38 +1657,36 @@ class NetEqStreamInput : public test::NetEqInput {
>  public:
>   // Does not take any ownership, and all pointers must refer to valid objects
>   // that outlive the one constructed.
>-  NetEqStreamInput(const std::vector<LoggedRtpPacket>* packet_stream,
>-                   const std::vector<uint64_t>* output_events_us,
>-                   rtc::Optional<uint64_t> end_time_us)
>+  NetEqStreamInput(const std::vector<LoggedRtpPacketIncoming>* packet_stream,
>+                   const std::vector<LoggedAudioPlayoutEvent>* output_events,
>+                   absl::optional<int64_t> end_time_ms)
>       : packet_stream_(*packet_stream),
>         packet_stream_it_(packet_stream_.begin()),
>-        output_events_us_it_(output_events_us->begin()),
>-        output_events_us_end_(output_events_us->end()),
>-        end_time_us_(end_time_us) {
>+        output_events_it_(output_events->begin()),
>+        output_events_end_(output_events->end()),
>+        end_time_ms_(end_time_ms) {
>     RTC_DCHECK(packet_stream);
>-    RTC_DCHECK(output_events_us);
>+    RTC_DCHECK(output_events);
>   }
> 
>-  rtc::Optional<int64_t> NextPacketTime() const override {
>+  absl::optional<int64_t> NextPacketTime() const override {
>     if (packet_stream_it_ == packet_stream_.end()) {
>-      return rtc::nullopt;
>+      return absl::nullopt;
>     }
>-    if (end_time_us_ && packet_stream_it_->timestamp > *end_time_us_) {
>-      return rtc::nullopt;
>+    if (end_time_ms_ && packet_stream_it_->rtp.log_time_ms() > *end_time_ms_) {
>+      return absl::nullopt;
>     }
>-    // Convert from us to ms.
>-    return packet_stream_it_->timestamp / 1000;
>+    return packet_stream_it_->rtp.log_time_ms();
>   }
> 
>-  rtc::Optional<int64_t> NextOutputEventTime() const override {
>-    if (output_events_us_it_ == output_events_us_end_) {
>-      return rtc::nullopt;
>+  absl::optional<int64_t> NextOutputEventTime() const override {
>+    if (output_events_it_ == output_events_end_) {
>+      return absl::nullopt;
>     }
>-    if (end_time_us_ && *output_events_us_it_ > *end_time_us_) {
>-      return rtc::nullopt;
>+    if (end_time_ms_ && output_events_it_->log_time_ms() > *end_time_ms_) {
>+      return absl::nullopt;
>     }
>-    // Convert from us to ms.
>-    return rtc::checked_cast<int64_t>(*output_events_us_it_ / 1000);
>+    return output_events_it_->log_time_ms();
>   }
> 
>   std::unique_ptr<PacketData> PopPacket() override {
>@@ -1753,13 +1694,13 @@ class NetEqStreamInput : public test::NetEqInput {
>       return std::unique_ptr<PacketData>();
>     }
>     std::unique_ptr<PacketData> packet_data(new PacketData());
>-    packet_data->header = packet_stream_it_->header;
>-    // Convert from us to ms.
>-    packet_data->time_ms = packet_stream_it_->timestamp / 1000.0;
>+    packet_data->header = packet_stream_it_->rtp.header;
>+    packet_data->time_ms = packet_stream_it_->rtp.log_time_ms();
> 
>     // This is a header-only "dummy" packet. Set the payload to all zeros, with
>     // length according to the virtual length.
>-    packet_data->payload.SetSize(packet_stream_it_->total_length);
>+    packet_data->payload.SetSize(packet_stream_it_->rtp.total_length -
>+                                 packet_stream_it_->rtp.header_length);
>     std::fill_n(packet_data->payload.data(), packet_data->payload.size(), 0);
> 
>     ++packet_stream_it_;
>@@ -1767,40 +1708,40 @@ class NetEqStreamInput : public test::NetEqInput {
>   }
> 
>   void AdvanceOutputEvent() override {
>-    if (output_events_us_it_ != output_events_us_end_) {
>-      ++output_events_us_it_;
>+    if (output_events_it_ != output_events_end_) {
>+      ++output_events_it_;
>     }
>   }
> 
>   bool ended() const override { return !NextEventTime(); }
> 
>-  rtc::Optional<RTPHeader> NextHeader() const override {
>+  absl::optional<RTPHeader> NextHeader() const override {
>     if (packet_stream_it_ == packet_stream_.end()) {
>-      return rtc::nullopt;
>+      return absl::nullopt;
>     }
>-    return packet_stream_it_->header;
>+    return packet_stream_it_->rtp.header;
>   }
> 
>  private:
>-  const std::vector<LoggedRtpPacket>& packet_stream_;
>-  std::vector<LoggedRtpPacket>::const_iterator packet_stream_it_;
>-  std::vector<uint64_t>::const_iterator output_events_us_it_;
>-  const std::vector<uint64_t>::const_iterator output_events_us_end_;
>-  const rtc::Optional<uint64_t> end_time_us_;
>+  const std::vector<LoggedRtpPacketIncoming>& packet_stream_;
>+  std::vector<LoggedRtpPacketIncoming>::const_iterator packet_stream_it_;
>+  std::vector<LoggedAudioPlayoutEvent>::const_iterator output_events_it_;
>+  const std::vector<LoggedAudioPlayoutEvent>::const_iterator output_events_end_;
>+  const absl::optional<int64_t> end_time_ms_;
> };
> 
> namespace {
> // Creates a NetEq test object and all necessary input and output helpers. Runs
> // the test and returns the NetEqDelayAnalyzer object that was used to
> // instrument the test.
>-std::unique_ptr<test::NetEqDelayAnalyzer> CreateNetEqTestAndRun(
>-    const std::vector<LoggedRtpPacket>* packet_stream,
>-    const std::vector<uint64_t>* output_events_us,
>-    rtc::Optional<uint64_t> end_time_us,
>+std::unique_ptr<test::NetEqStatsGetter> CreateNetEqTestAndRun(
>+    const std::vector<LoggedRtpPacketIncoming>* packet_stream,
>+    const std::vector<LoggedAudioPlayoutEvent>* output_events,
>+    absl::optional<int64_t> end_time_ms,
>     const std::string& replacement_file_name,
>     int file_sample_rate_hz) {
>   std::unique_ptr<test::NetEqInput> input(
>-      new NetEqStreamInput(packet_stream, output_events_us, end_time_us));
>+      new NetEqStreamInput(packet_stream, output_events, end_time_ms));
> 
>   constexpr int kReplacementPt = 127;
>   std::set<uint8_t> cn_types;
>@@ -1831,124 +1772,426 @@ std::unique_ptr<test::NetEqDelayAnalyzer> CreateNetEqTestAndRun(
> 
>   std::unique_ptr<test::NetEqDelayAnalyzer> delay_cb(
>       new test::NetEqDelayAnalyzer);
>+  std::unique_ptr<test::NetEqStatsGetter> neteq_stats_getter(
>+      new test::NetEqStatsGetter(std::move(delay_cb)));
>   test::DefaultNetEqTestErrorCallback error_cb;
>   test::NetEqTest::Callbacks callbacks;
>   callbacks.error_callback = &error_cb;
>-  callbacks.post_insert_packet = delay_cb.get();
>-  callbacks.get_audio_callback = delay_cb.get();
>+  callbacks.post_insert_packet = neteq_stats_getter->delay_analyzer();
>+  callbacks.get_audio_callback = neteq_stats_getter.get();
> 
>   test::NetEqTest test(config, codecs, ext_codecs, std::move(input),
>                        std::move(output), callbacks);
>   test.Run();
>-  return delay_cb;
>+  return neteq_stats_getter;
> }
> }  // namespace
> 
>-// Plots the jitter buffer delay profile. This will plot only for the first
>-// incoming audio SSRC. If the stream contains more than one incoming audio
>-// SSRC, all but the first will be ignored.
>-void EventLogAnalyzer::CreateAudioJitterBufferGraph(
>+EventLogAnalyzer::NetEqStatsGetterMap EventLogAnalyzer::SimulateNetEq(
>     const std::string& replacement_file_name,
>-    int file_sample_rate_hz,
>-    Plot* plot) {
>-  const auto& incoming_audio_kv = std::find_if(
>-      rtp_packets_.begin(), rtp_packets_.end(),
>-      [this](std::pair<StreamId, std::vector<LoggedRtpPacket>> kv) {
>-        return kv.first.GetDirection() == kIncomingPacket &&
>-               this->IsAudioSsrc(kv.first);
>-      });
>-  if (incoming_audio_kv == rtp_packets_.end()) {
>-    // No incoming audio stream found.
>-    return;
>-  }
>-
>-  const uint32_t ssrc = incoming_audio_kv->first.GetSsrc();
>-
>-  std::map<uint32_t, std::vector<uint64_t>>::const_iterator output_events_it =
>-      audio_playout_events_.find(ssrc);
>-  if (output_events_it == audio_playout_events_.end()) {
>-    // Could not find output events with SSRC matching the input audio stream.
>-    // Using the first available stream of output events.
>-    output_events_it = audio_playout_events_.cbegin();
>-  }
>-
>-  rtc::Optional<uint64_t> end_time_us =
>-      log_segments_.empty()
>-          ? rtc::nullopt
>-          : rtc::Optional<uint64_t>(log_segments_.front().second);
>-
>-  auto delay_cb = CreateNetEqTestAndRun(
>-      &incoming_audio_kv->second, &output_events_it->second, end_time_us,
>-      replacement_file_name, file_sample_rate_hz);
>-
>-  std::vector<float> send_times_s;
>-  std::vector<float> arrival_delay_ms;
>-  std::vector<float> corrected_arrival_delay_ms;
>-  std::vector<rtc::Optional<float>> playout_delay_ms;
>-  std::vector<rtc::Optional<float>> target_delay_ms;
>-  delay_cb->CreateGraphs(&send_times_s, &arrival_delay_ms,
>-                         &corrected_arrival_delay_ms, &playout_delay_ms,
>-                         &target_delay_ms);
>-  RTC_DCHECK_EQ(send_times_s.size(), arrival_delay_ms.size());
>-  RTC_DCHECK_EQ(send_times_s.size(), corrected_arrival_delay_ms.size());
>-  RTC_DCHECK_EQ(send_times_s.size(), playout_delay_ms.size());
>-  RTC_DCHECK_EQ(send_times_s.size(), target_delay_ms.size());
>-
>-  std::map<StreamId, TimeSeries> time_series_packet_arrival;
>-  std::map<StreamId, TimeSeries> time_series_relative_packet_arrival;
>-  std::map<StreamId, TimeSeries> time_series_play_time;
>-  std::map<StreamId, TimeSeries> time_series_target_time;
>-  float min_y_axis = 0.f;
>-  float max_y_axis = 0.f;
>-  const StreamId stream_id = incoming_audio_kv->first;
>-  for (size_t i = 0; i < send_times_s.size(); ++i) {
>-    time_series_packet_arrival[stream_id].points.emplace_back(
>-        TimeSeriesPoint(send_times_s[i], arrival_delay_ms[i]));
>-    time_series_relative_packet_arrival[stream_id].points.emplace_back(
>-        TimeSeriesPoint(send_times_s[i], corrected_arrival_delay_ms[i]));
>-    min_y_axis = std::min(min_y_axis, corrected_arrival_delay_ms[i]);
>-    max_y_axis = std::max(max_y_axis, corrected_arrival_delay_ms[i]);
>-    if (playout_delay_ms[i]) {
>-      time_series_play_time[stream_id].points.emplace_back(
>-          TimeSeriesPoint(send_times_s[i], *playout_delay_ms[i]));
>-      min_y_axis = std::min(min_y_axis, *playout_delay_ms[i]);
>-      max_y_axis = std::max(max_y_axis, *playout_delay_ms[i]);
>+    int file_sample_rate_hz) const {
>+  NetEqStatsGetterMap neteq_stats;
>+
>+  for (const auto& stream : parsed_log_.incoming_rtp_packets_by_ssrc()) {
>+    const uint32_t ssrc = stream.ssrc;
>+    if (!IsAudioSsrc(kIncomingPacket, ssrc))
>+      continue;
>+    const std::vector<LoggedRtpPacketIncoming>* audio_packets =
>+        &stream.incoming_packets;
>+    if (audio_packets == nullptr) {
>+      // No incoming audio stream found.
>+      continue;
>+    }
>+
>+    RTC_DCHECK(neteq_stats.find(ssrc) == neteq_stats.end());
>+
>+    std::map<uint32_t, std::vector<LoggedAudioPlayoutEvent>>::const_iterator
>+        output_events_it = parsed_log_.audio_playout_events().find(ssrc);
>+    if (output_events_it == parsed_log_.audio_playout_events().end()) {
>+      // Could not find output events with SSRC matching the input audio stream.
>+      // Using the first available stream of output events.
>+      output_events_it = parsed_log_.audio_playout_events().cbegin();
>     }
>-    if (target_delay_ms[i]) {
>-      time_series_target_time[stream_id].points.emplace_back(
>-          TimeSeriesPoint(send_times_s[i], *target_delay_ms[i]));
>-      min_y_axis = std::min(min_y_axis, *target_delay_ms[i]);
>-      max_y_axis = std::max(max_y_axis, *target_delay_ms[i]);
>+
>+    absl::optional<int64_t> end_time_ms =
>+        log_segments_.empty()
>+            ? absl::nullopt
>+            : absl::optional<int64_t>(log_segments_.front().second / 1000);
>+
>+    neteq_stats[ssrc] = CreateNetEqTestAndRun(
>+        audio_packets, &output_events_it->second, end_time_ms,
>+        replacement_file_name, file_sample_rate_hz);
>+  }
>+
>+  return neteq_stats;
>+}
>+
>+// Given a NetEqStatsGetter and the SSRC that the NetEqStatsGetter was created
>+// for, this method generates a plot for the jitter buffer delay profile.
>+void EventLogAnalyzer::CreateAudioJitterBufferGraph(
>+    uint32_t ssrc,
>+    const test::NetEqStatsGetter* stats_getter,
>+    Plot* plot) const {
>+  test::NetEqDelayAnalyzer::Delays arrival_delay_ms;
>+  test::NetEqDelayAnalyzer::Delays corrected_arrival_delay_ms;
>+  test::NetEqDelayAnalyzer::Delays playout_delay_ms;
>+  test::NetEqDelayAnalyzer::Delays target_delay_ms;
>+
>+  stats_getter->delay_analyzer()->CreateGraphs(
>+      &arrival_delay_ms, &corrected_arrival_delay_ms, &playout_delay_ms,
>+      &target_delay_ms);
>+
>+  TimeSeries time_series_packet_arrival("packet arrival delay",
>+                                        LineStyle::kLine);
>+  TimeSeries time_series_relative_packet_arrival(
>+      "Relative packet arrival delay", LineStyle::kLine);
>+  TimeSeries time_series_play_time("Playout delay", LineStyle::kLine);
>+  TimeSeries time_series_target_time("Target delay", LineStyle::kLine,
>+                                     PointStyle::kHighlight);
>+
>+  for (const auto& data : arrival_delay_ms) {
>+    const float x = ToCallTimeSec(data.first * 1000);  // ms to us.
>+    const float y = data.second;
>+    time_series_packet_arrival.points.emplace_back(TimeSeriesPoint(x, y));
>+  }
>+  for (const auto& data : corrected_arrival_delay_ms) {
>+    const float x = ToCallTimeSec(data.first * 1000);  // ms to us.
>+    const float y = data.second;
>+    time_series_relative_packet_arrival.points.emplace_back(
>+        TimeSeriesPoint(x, y));
>+  }
>+  for (const auto& data : playout_delay_ms) {
>+    const float x = ToCallTimeSec(data.first * 1000);  // ms to us.
>+    const float y = data.second;
>+    time_series_play_time.points.emplace_back(TimeSeriesPoint(x, y));
>+  }
>+  for (const auto& data : target_delay_ms) {
>+    const float x = ToCallTimeSec(data.first * 1000);  // ms to us.
>+    const float y = data.second;
>+    time_series_target_time.points.emplace_back(TimeSeriesPoint(x, y));
>+  }
>+
>+  plot->AppendTimeSeries(std::move(time_series_packet_arrival));
>+  plot->AppendTimeSeries(std::move(time_series_relative_packet_arrival));
>+  plot->AppendTimeSeries(std::move(time_series_play_time));
>+  plot->AppendTimeSeries(std::move(time_series_target_time));
>+
>+  plot->SetXAxis(ToCallTimeSec(begin_time_), call_duration_s_, "Time (s)",
>+                 kLeftMargin, kRightMargin);
>+  plot->SetSuggestedYAxis(0, 1, "Relative delay (ms)", kBottomMargin,
>+                          kTopMargin);
>+  plot->SetTitle("NetEq timing for " + GetStreamName(kIncomingPacket, ssrc));
>+}
>+
>+template <typename NetEqStatsType>
>+void EventLogAnalyzer::CreateNetEqStatsGraphInternal(
>+    const NetEqStatsGetterMap& neteq_stats,
>+    rtc::FunctionView<const std::vector<std::pair<int64_t, NetEqStatsType>>*(
>+        const test::NetEqStatsGetter*)> data_extractor,
>+    rtc::FunctionView<float(const NetEqStatsType&)> stats_extractor,
>+    const std::string& plot_name,
>+    Plot* plot) const {
>+  std::map<uint32_t, TimeSeries> time_series;
>+
>+  for (const auto& st : neteq_stats) {
>+    const uint32_t ssrc = st.first;
>+    const std::vector<std::pair<int64_t, NetEqStatsType>>* data_vector =
>+        data_extractor(st.second.get());
>+    for (const auto& data : *data_vector) {
>+      const float time = ToCallTimeSec(data.first * 1000);  // ms to us.
>+      const float value = stats_extractor(data.second);
>+      time_series[ssrc].points.emplace_back(TimeSeriesPoint(time, value));
>     }
>   }
> 
>-  // This code is adapted for a single stream. The creation of the streams above
>-  // guarantee that no more than one steam is included. If multiple streams are
>-  // to be plotted, they should likely be given distinct labels below.
>-  RTC_DCHECK_EQ(time_series_relative_packet_arrival.size(), 1);
>-  for (auto& series : time_series_relative_packet_arrival) {
>-    series.second.label = "Relative packet arrival delay";
>+  for (auto& series : time_series) {
>+    series.second.label = GetStreamName(kIncomingPacket, series.first);
>     series.second.line_style = LineStyle::kLine;
>     plot->AppendTimeSeries(std::move(series.second));
>   }
>-  RTC_DCHECK_EQ(time_series_play_time.size(), 1);
>-  for (auto& series : time_series_play_time) {
>-    series.second.label = "Playout delay";
>-    series.second.line_style = LineStyle::kLine;
>-    plot->AppendTimeSeries(std::move(series.second));
>+
>+  plot->SetXAxis(ToCallTimeSec(begin_time_), call_duration_s_, "Time (s)",
>+                 kLeftMargin, kRightMargin);
>+  plot->SetSuggestedYAxis(0, 1, plot_name, kBottomMargin, kTopMargin);
>+  plot->SetTitle(plot_name);
>+}
>+
>+void EventLogAnalyzer::CreateNetEqNetworkStatsGraph(
>+    const NetEqStatsGetterMap& neteq_stats,
>+    rtc::FunctionView<float(const NetEqNetworkStatistics&)> stats_extractor,
>+    const std::string& plot_name,
>+    Plot* plot) const {
>+  CreateNetEqStatsGraphInternal<NetEqNetworkStatistics>(
>+      neteq_stats,
>+      [](const test::NetEqStatsGetter* stats_getter) {
>+        return stats_getter->stats();
>+      },
>+      stats_extractor, plot_name, plot);
>+}
>+
>+void EventLogAnalyzer::CreateNetEqLifetimeStatsGraph(
>+    const NetEqStatsGetterMap& neteq_stats,
>+    rtc::FunctionView<float(const NetEqLifetimeStatistics&)> stats_extractor,
>+    const std::string& plot_name,
>+    Plot* plot) const {
>+  CreateNetEqStatsGraphInternal<NetEqLifetimeStatistics>(
>+      neteq_stats,
>+      [](const test::NetEqStatsGetter* stats_getter) {
>+        return stats_getter->lifetime_stats();
>+      },
>+      stats_extractor, plot_name, plot);
>+}
>+
>+void EventLogAnalyzer::CreateIceCandidatePairConfigGraph(Plot* plot) {
>+  std::map<uint32_t, TimeSeries> configs_by_cp_id;
>+  for (const auto& config : parsed_log_.ice_candidate_pair_configs()) {
>+    if (configs_by_cp_id.find(config.candidate_pair_id) ==
>+        configs_by_cp_id.end()) {
>+      const std::string candidate_pair_desc =
>+          GetCandidatePairLogDescriptionAsString(config);
>+      configs_by_cp_id[config.candidate_pair_id] =
>+          TimeSeries("[" + std::to_string(config.candidate_pair_id) + "]" +
>+                         candidate_pair_desc,
>+                     LineStyle::kNone, PointStyle::kHighlight);
>+      candidate_pair_desc_by_id_[config.candidate_pair_id] =
>+          candidate_pair_desc;
>+    }
>+    float x = ToCallTimeSec(config.log_time_us());
>+    float y = static_cast<float>(config.type);
>+    configs_by_cp_id[config.candidate_pair_id].points.emplace_back(x, y);
>   }
>-  RTC_DCHECK_EQ(time_series_target_time.size(), 1);
>-  for (auto& series : time_series_target_time) {
>-    series.second.label = "Target delay";
>-    series.second.line_style = LineStyle::kLine;
>-    series.second.point_style = PointStyle::kHighlight;
>-    plot->AppendTimeSeries(std::move(series.second));
>+
>+  // TODO(qingsi): There can be a large number of candidate pairs generated by
>+  // certain calls and the frontend cannot render the chart in this case due to
>+  // the failure of generating a palette with the same number of colors.
>+  for (auto& kv : configs_by_cp_id) {
>+    plot->AppendTimeSeries(std::move(kv.second));
>+  }
>+
>+  plot->SetXAxis(ToCallTimeSec(begin_time_), call_duration_s_, "Time (s)",
>+                 kLeftMargin, kRightMargin);
>+  plot->SetSuggestedYAxis(0, 3, "Numeric Config Type", kBottomMargin,
>+                          kTopMargin);
>+  plot->SetTitle("[IceEventLog] ICE candidate pair configs");
>+}
>+
>+std::string EventLogAnalyzer::GetCandidatePairLogDescriptionFromId(
>+    uint32_t candidate_pair_id) {
>+  if (candidate_pair_desc_by_id_.find(candidate_pair_id) !=
>+      candidate_pair_desc_by_id_.end()) {
>+    return candidate_pair_desc_by_id_[candidate_pair_id];
>+  }
>+  for (const auto& config : parsed_log_.ice_candidate_pair_configs()) {
>+    // TODO(qingsi): Add the handling of the "Updated" config event after the
>+    // visualization of property change for candidate pairs is introduced.
>+    if (candidate_pair_desc_by_id_.find(config.candidate_pair_id) ==
>+        candidate_pair_desc_by_id_.end()) {
>+      const std::string candidate_pair_desc =
>+          GetCandidatePairLogDescriptionAsString(config);
>+      candidate_pair_desc_by_id_[config.candidate_pair_id] =
>+          candidate_pair_desc;
>+    }
>+  }
>+  return candidate_pair_desc_by_id_[candidate_pair_id];
>+}
>+
>+void EventLogAnalyzer::CreateIceConnectivityCheckGraph(Plot* plot) {
>+  std::map<uint32_t, TimeSeries> checks_by_cp_id;
>+  for (const auto& event : parsed_log_.ice_candidate_pair_events()) {
>+    if (checks_by_cp_id.find(event.candidate_pair_id) ==
>+        checks_by_cp_id.end()) {
>+      checks_by_cp_id[event.candidate_pair_id] = TimeSeries(
>+          "[" + std::to_string(event.candidate_pair_id) + "]" +
>+              GetCandidatePairLogDescriptionFromId(event.candidate_pair_id),
>+          LineStyle::kNone, PointStyle::kHighlight);
>+    }
>+    float x = ToCallTimeSec(event.log_time_us());
>+    float y = static_cast<float>(event.type);
>+    checks_by_cp_id[event.candidate_pair_id].points.emplace_back(x, y);
>   }
> 
>-  plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
>-  plot->SetYAxis(min_y_axis, max_y_axis, "Relative delay (ms)", kBottomMargin,
>-                 kTopMargin);
>-  plot->SetTitle("NetEq timing");
>+  // TODO(qingsi): The same issue as in CreateIceCandidatePairConfigGraph.
>+  for (auto& kv : checks_by_cp_id) {
>+    plot->AppendTimeSeries(std::move(kv.second));
>+  }
>+
>+  plot->SetXAxis(ToCallTimeSec(begin_time_), call_duration_s_, "Time (s)",
>+                 kLeftMargin, kRightMargin);
>+  plot->SetSuggestedYAxis(0, 4, "Numeric Connectivity State", kBottomMargin,
>+                          kTopMargin);
>+  plot->SetTitle("[IceEventLog] ICE connectivity checks");
> }
>-}  // namespace plotting
>+
>+void EventLogAnalyzer::PrintNotifications(FILE* file) {
>+  fprintf(file, "========== TRIAGE NOTIFICATIONS ==========\n");
>+  for (const auto& alert : incoming_rtp_recv_time_gaps_) {
>+    fprintf(file, "%3.3lf s : %s\n", alert.Time(), alert.ToString().c_str());
>+  }
>+  for (const auto& alert : incoming_rtcp_recv_time_gaps_) {
>+    fprintf(file, "%3.3lf s : %s\n", alert.Time(), alert.ToString().c_str());
>+  }
>+  for (const auto& alert : outgoing_rtp_send_time_gaps_) {
>+    fprintf(file, "%3.3lf s : %s\n", alert.Time(), alert.ToString().c_str());
>+  }
>+  for (const auto& alert : outgoing_rtcp_send_time_gaps_) {
>+    fprintf(file, "%3.3lf s : %s\n", alert.Time(), alert.ToString().c_str());
>+  }
>+  for (const auto& alert : incoming_seq_num_jumps_) {
>+    fprintf(file, "%3.3lf s : %s\n", alert.Time(), alert.ToString().c_str());
>+  }
>+  for (const auto& alert : incoming_capture_time_jumps_) {
>+    fprintf(file, "%3.3lf s : %s\n", alert.Time(), alert.ToString().c_str());
>+  }
>+  for (const auto& alert : outgoing_seq_num_jumps_) {
>+    fprintf(file, "%3.3lf s : %s\n", alert.Time(), alert.ToString().c_str());
>+  }
>+  for (const auto& alert : outgoing_capture_time_jumps_) {
>+    fprintf(file, "%3.3lf s : %s\n", alert.Time(), alert.ToString().c_str());
>+  }
>+  for (const auto& alert : outgoing_high_loss_alerts_) {
>+    fprintf(file, "          : %s\n", alert.ToString().c_str());
>+  }
>+  fprintf(file, "========== END TRIAGE NOTIFICATIONS ==========\n");
>+}
>+
>+void EventLogAnalyzer::CreateStreamGapAlerts(PacketDirection direction) {
>+  // With 100 packets/s (~800kbps), false positives would require 10 s without
>+  // data.
>+  constexpr int64_t kMaxSeqNumJump = 1000;
>+  // With a 90 kHz clock, false positives would require 10 s without data.
>+  constexpr int64_t kMaxCaptureTimeJump = 900000;
>+
>+  int64_t end_time_us = log_segments_.empty()
>+                            ? std::numeric_limits<int64_t>::max()
>+                            : log_segments_.front().second;
>+
>+  SeqNumUnwrapper<uint16_t> seq_num_unwrapper;
>+  absl::optional<int64_t> last_seq_num;
>+  SeqNumUnwrapper<uint32_t> capture_time_unwrapper;
>+  absl::optional<int64_t> last_capture_time;
>+  // Check for gaps in sequence numbers and capture timestamps.
>+  for (const auto& stream : parsed_log_.rtp_packets_by_ssrc(direction)) {
>+    for (const auto& packet : stream.packet_view) {
>+      if (packet.log_time_us() > end_time_us) {
>+        // Only process the first (LOG_START, LOG_END) segment.
>+        break;
>+      }
>+
>+      int64_t seq_num = seq_num_unwrapper.Unwrap(packet.header.sequenceNumber);
>+      if (last_seq_num.has_value() &&
>+          std::abs(seq_num - last_seq_num.value()) > kMaxSeqNumJump) {
>+        Alert_SeqNumJump(direction, ToCallTimeSec(packet.log_time_us()),
>+                         packet.header.ssrc);
>+      }
>+      last_seq_num.emplace(seq_num);
>+
>+      int64_t capture_time =
>+          capture_time_unwrapper.Unwrap(packet.header.timestamp);
>+      if (last_capture_time.has_value() &&
>+          std::abs(capture_time - last_capture_time.value()) >
>+              kMaxCaptureTimeJump) {
>+        Alert_CaptureTimeJump(direction, ToCallTimeSec(packet.log_time_us()),
>+                              packet.header.ssrc);
>+      }
>+      last_capture_time.emplace(capture_time);
>+    }
>+  }
>+}
>+
>+void EventLogAnalyzer::CreateTransmissionGapAlerts(PacketDirection direction) {
>+  constexpr int64_t kMaxRtpTransmissionGap = 500000;
>+  constexpr int64_t kMaxRtcpTransmissionGap = 2000000;
>+  int64_t end_time_us = log_segments_.empty()
>+                            ? std::numeric_limits<int64_t>::max()
>+                            : log_segments_.front().second;
>+
>+  // TODO(terelius): The parser could provide a list of all packets, ordered
>+  // by time, for each direction.
>+  std::multimap<int64_t, const LoggedRtpPacket*> rtp_in_direction;
>+  for (const auto& stream : parsed_log_.rtp_packets_by_ssrc(direction)) {
>+    for (const LoggedRtpPacket& rtp_packet : stream.packet_view)
>+      rtp_in_direction.emplace(rtp_packet.log_time_us(), &rtp_packet);
>+  }
>+  absl::optional<int64_t> last_rtp_time;
>+  for (const auto& kv : rtp_in_direction) {
>+    int64_t timestamp = kv.first;
>+    if (timestamp > end_time_us) {
>+      // Only process the first (LOG_START, LOG_END) segment.
>+      break;
>+    }
>+    int64_t duration = timestamp - last_rtp_time.value_or(0);
>+    if (last_rtp_time.has_value() && duration > kMaxRtpTransmissionGap) {
>+      // No packet sent/received for more than 500 ms.
>+      Alert_RtpLogTimeGap(direction, ToCallTimeSec(timestamp), duration / 1000);
>+    }
>+    last_rtp_time.emplace(timestamp);
>+  }
>+
>+  absl::optional<int64_t> last_rtcp_time;
>+  if (direction == kIncomingPacket) {
>+    for (const auto& rtcp : parsed_log_.incoming_rtcp_packets()) {
>+      if (rtcp.log_time_us() > end_time_us) {
>+        // Only process the first (LOG_START, LOG_END) segment.
>+        break;
>+      }
>+      int64_t duration = rtcp.log_time_us() - last_rtcp_time.value_or(0);
>+      if (last_rtcp_time.has_value() && duration > kMaxRtcpTransmissionGap) {
>+        // No feedback sent/received for more than 2000 ms.
>+        Alert_RtcpLogTimeGap(direction, ToCallTimeSec(rtcp.log_time_us()),
>+                             duration / 1000);
>+      }
>+      last_rtcp_time.emplace(rtcp.log_time_us());
>+    }
>+  } else {
>+    for (const auto& rtcp : parsed_log_.outgoing_rtcp_packets()) {
>+      if (rtcp.log_time_us() > end_time_us) {
>+        // Only process the first (LOG_START, LOG_END) segment.
>+        break;
>+      }
>+      int64_t duration = rtcp.log_time_us() - last_rtcp_time.value_or(0);
>+      if (last_rtcp_time.has_value() && duration > kMaxRtcpTransmissionGap) {
>+        // No feedback sent/received for more than 2000 ms.
>+        Alert_RtcpLogTimeGap(direction, ToCallTimeSec(rtcp.log_time_us()),
>+                             duration / 1000);
>+      }
>+      last_rtcp_time.emplace(rtcp.log_time_us());
>+    }
>+  }
>+}
>+
>+// TODO(terelius): Notifications could possibly be generated by the same code
>+// that produces the graphs. There is some code duplication that could be
>+// avoided, but that might be solved anyway when we move functionality from the
>+// analyzer to the parser.
>+void EventLogAnalyzer::CreateTriageNotifications() {
>+  CreateStreamGapAlerts(kIncomingPacket);
>+  CreateStreamGapAlerts(kOutgoingPacket);
>+  CreateTransmissionGapAlerts(kIncomingPacket);
>+  CreateTransmissionGapAlerts(kOutgoingPacket);
>+
>+  int64_t end_time_us = log_segments_.empty()
>+                            ? std::numeric_limits<int64_t>::max()
>+                            : log_segments_.front().second;
>+
>+  constexpr double kMaxLossFraction = 0.05;
>+  // Loss feedback
>+  int64_t total_lost_packets = 0;
>+  int64_t total_expected_packets = 0;
>+  for (auto& bwe_update : parsed_log_.bwe_loss_updates()) {
>+    if (bwe_update.log_time_us() > end_time_us) {
>+      // Only process the first (LOG_START, LOG_END) segment.
>+      break;
>+    }
>+    int64_t lost_packets = static_cast<double>(bwe_update.fraction_lost) / 255 *
>+                           bwe_update.expected_packets;
>+    total_lost_packets += lost_packets;
>+    total_expected_packets += bwe_update.expected_packets;
>+  }
>+  double avg_outgoing_loss =
>+      static_cast<double>(total_lost_packets) / total_expected_packets;
>+  if (avg_outgoing_loss > kMaxLossFraction) {
>+    Alert_OutgoingHighLoss(avg_outgoing_loss);
>+  }
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/event_log_visualizer/analyzer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/event_log_visualizer/analyzer.h
>index 15f1fa7ea47..4b8c27c9517 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/event_log_visualizer/analyzer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/event_log_visualizer/analyzer.h
>@@ -18,62 +18,28 @@
> #include <utility>
> #include <vector>
> 
>-#include "logging/rtc_event_log/rtc_event_log_parser.h"
>-#include "modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
>-#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>-#include "modules/rtp_rtcp/source/rtcp_packet.h"
>-#include "rtc_base/function_view.h"
>+#include "logging/rtc_event_log/rtc_event_log_parser_new.h"
>+#include "modules/audio_coding/neteq/tools/neteq_stats_getter.h"
>+#include "rtc_base/strings/string_builder.h"
> #include "rtc_tools/event_log_visualizer/plot_base.h"
>+#include "rtc_tools/event_log_visualizer/triage_notifications.h"
> 
> namespace webrtc {
>-namespace plotting {
>-
>-struct LoggedRtpPacket {
>-  LoggedRtpPacket(uint64_t timestamp, RTPHeader header, size_t total_length)
>-      : timestamp(timestamp), header(header), total_length(total_length) {}
>-  uint64_t timestamp;
>-  // TODO(terelius): This allocates space for 15 CSRCs even if none are used.
>-  RTPHeader header;
>-  size_t total_length;
>-};
>-
>-struct LoggedRtcpPacket {
>-  LoggedRtcpPacket(uint64_t timestamp,
>-                   RTCPPacketType rtcp_type,
>-                   std::unique_ptr<rtcp::RtcpPacket> rtcp_packet)
>-      : timestamp(timestamp), type(rtcp_type), packet(std::move(rtcp_packet)) {}
>-  uint64_t timestamp;
>-  RTCPPacketType type;
>-  std::unique_ptr<rtcp::RtcpPacket> packet;
>-};
>-
>-struct LossBasedBweUpdate {
>-  uint64_t timestamp;
>-  int32_t new_bitrate;
>-  uint8_t fraction_loss;
>-  int32_t expected_packets;
>-};
>-
>-struct AudioNetworkAdaptationEvent {
>-  uint64_t timestamp;
>-  AudioEncoderRuntimeConfig config;
>-};
> 
> class EventLogAnalyzer {
>  public:
>-  // The EventLogAnalyzer keeps a reference to the ParsedRtcEventLog for the
>-  // duration of its lifetime. The ParsedRtcEventLog must not be destroyed or
>+  // The EventLogAnalyzer keeps a reference to the ParsedRtcEventLogNew for the
>+  // duration of its lifetime. The ParsedRtcEventLogNew must not be destroyed or
>   // modified while the EventLogAnalyzer is being used.
>-  explicit EventLogAnalyzer(const ParsedRtcEventLog& log);
>+  EventLogAnalyzer(const ParsedRtcEventLogNew& log, bool normalize_time);
> 
>-  void CreatePacketGraph(PacketDirection desired_direction, Plot* plot);
>+  void CreatePacketGraph(PacketDirection direction, Plot* plot);
> 
>-  void CreateAccumulatedPacketsGraph(PacketDirection desired_direction,
>-                                     Plot* plot);
>+  void CreateAccumulatedPacketsGraph(PacketDirection direction, Plot* plot);
> 
>   void CreatePlayoutGraph(Plot* plot);
> 
>-  void CreateAudioLevelGraph(Plot* plot);
>+  void CreateAudioLevelGraph(PacketDirection direction, Plot* plot);
> 
>   void CreateSequenceNumberGraph(Plot* plot);
> 
>@@ -84,18 +50,26 @@ class EventLogAnalyzer {
> 
>   void CreateFractionLossGraph(Plot* plot);
> 
>-  void CreateTotalBitrateGraph(PacketDirection desired_direction,
>-                               Plot* plot,
>-                               bool show_detector_state = false);
>+  void CreateTotalIncomingBitrateGraph(Plot* plot);
>+  void CreateTotalOutgoingBitrateGraph(Plot* plot,
>+                                       bool show_detector_state = false,
>+                                       bool show_alr_state = false);
> 
>-  void CreateStreamBitrateGraph(PacketDirection desired_direction, Plot* plot);
>+  void CreateStreamBitrateGraph(PacketDirection direction, Plot* plot);
> 
>   void CreateSendSideBweSimulationGraph(Plot* plot);
>   void CreateReceiveSideBweSimulationGraph(Plot* plot);
> 
>   void CreateNetworkDelayFeedbackGraph(Plot* plot);
>   void CreatePacerDelayGraph(Plot* plot);
>-  void CreateTimestampGraph(Plot* plot);
>+
>+  void CreateTimestampGraph(PacketDirection direction, Plot* plot);
>+  void CreateSenderAndReceiverReportPlot(
>+      PacketDirection direction,
>+      rtc::FunctionView<float(const rtcp::ReportBlock&)> fy,
>+      std::string title,
>+      std::string yaxis_label,
>+      Plot* plot);
> 
>   void CreateAudioEncoderTargetBitrateGraph(Plot* plot);
>   void CreateAudioEncoderFrameLengthGraph(Plot* plot);
>@@ -103,110 +77,188 @@ class EventLogAnalyzer {
>   void CreateAudioEncoderEnableFecGraph(Plot* plot);
>   void CreateAudioEncoderEnableDtxGraph(Plot* plot);
>   void CreateAudioEncoderNumChannelsGraph(Plot* plot);
>-  void CreateAudioJitterBufferGraph(const std::string& replacement_file_name,
>-                                    int file_sample_rate_hz,
>-                                    Plot* plot);
> 
>-  // Returns a vector of capture and arrival timestamps for the video frames
>-  // of the stream with the most number of frames.
>-  std::vector<std::pair<int64_t, int64_t>> GetFrameTimestamps() const;
>+  using NetEqStatsGetterMap =
>+      std::map<uint32_t, std::unique_ptr<test::NetEqStatsGetter>>;
>+  NetEqStatsGetterMap SimulateNetEq(const std::string& replacement_file_name,
>+                                    int file_sample_rate_hz) const;
>+
>+  void CreateAudioJitterBufferGraph(uint32_t ssrc,
>+                                    const test::NetEqStatsGetter* stats_getter,
>+                                    Plot* plot) const;
>+  void CreateNetEqNetworkStatsGraph(
>+      const NetEqStatsGetterMap& neteq_stats_getters,
>+      rtc::FunctionView<float(const NetEqNetworkStatistics&)> stats_extractor,
>+      const std::string& plot_name,
>+      Plot* plot) const;
>+  void CreateNetEqLifetimeStatsGraph(
>+      const NetEqStatsGetterMap& neteq_stats_getters,
>+      rtc::FunctionView<float(const NetEqLifetimeStatistics&)> stats_extractor,
>+      const std::string& plot_name,
>+      Plot* plot) const;
>+
>+  void CreateIceCandidatePairConfigGraph(Plot* plot);
>+  void CreateIceConnectivityCheckGraph(Plot* plot);
>+
>+  void CreateTriageNotifications();
>+  void PrintNotifications(FILE* file);
> 
>  private:
>-  class StreamId {
>-   public:
>-    StreamId(uint32_t ssrc, webrtc::PacketDirection direction)
>-        : ssrc_(ssrc), direction_(direction) {}
>-    bool operator<(const StreamId& other) const {
>-      return std::tie(ssrc_, direction_) <
>-             std::tie(other.ssrc_, other.direction_);
>+  bool IsRtxSsrc(PacketDirection direction, uint32_t ssrc) const {
>+    if (direction == kIncomingPacket) {
>+      return parsed_log_.incoming_rtx_ssrcs().find(ssrc) !=
>+             parsed_log_.incoming_rtx_ssrcs().end();
>+    } else {
>+      return parsed_log_.outgoing_rtx_ssrcs().find(ssrc) !=
>+             parsed_log_.outgoing_rtx_ssrcs().end();
>     }
>-    bool operator==(const StreamId& other) const {
>-      return std::tie(ssrc_, direction_) ==
>-             std::tie(other.ssrc_, other.direction_);
>+  }
>+
>+  bool IsVideoSsrc(PacketDirection direction, uint32_t ssrc) const {
>+    if (direction == kIncomingPacket) {
>+      return parsed_log_.incoming_video_ssrcs().find(ssrc) !=
>+             parsed_log_.incoming_video_ssrcs().end();
>+    } else {
>+      return parsed_log_.outgoing_video_ssrcs().find(ssrc) !=
>+             parsed_log_.outgoing_video_ssrcs().end();
>     }
>-    uint32_t GetSsrc() const { return ssrc_; }
>-    webrtc::PacketDirection GetDirection() const { return direction_; }
>-
>-   private:
>-    uint32_t ssrc_;
>-    webrtc::PacketDirection direction_;
>-  };
>-
>-  template <typename T>
>-  void CreateAccumulatedPacketsTimeSeries(
>-      PacketDirection desired_direction,
>-      Plot* plot,
>-      const std::map<StreamId, std::vector<T>>& packets,
>-      const std::string& label_prefix);
>-
>-  bool IsRtxSsrc(StreamId stream_id) const;
>-
>-  bool IsVideoSsrc(StreamId stream_id) const;
>-
>-  bool IsAudioSsrc(StreamId stream_id) const;
>+  }
>+
>+  bool IsAudioSsrc(PacketDirection direction, uint32_t ssrc) const {
>+    if (direction == kIncomingPacket) {
>+      return parsed_log_.incoming_audio_ssrcs().find(ssrc) !=
>+             parsed_log_.incoming_audio_ssrcs().end();
>+    } else {
>+      return parsed_log_.outgoing_audio_ssrcs().find(ssrc) !=
>+             parsed_log_.outgoing_audio_ssrcs().end();
>+    }
>+  }
>+
>+  template <typename NetEqStatsType>
>+  void CreateNetEqStatsGraphInternal(
>+      const NetEqStatsGetterMap& neteq_stats,
>+      rtc::FunctionView<const std::vector<std::pair<int64_t, NetEqStatsType>>*(
>+          const test::NetEqStatsGetter*)> data_extractor,
>+      rtc::FunctionView<float(const NetEqStatsType&)> stats_extractor,
>+      const std::string& plot_name,
>+      Plot* plot) const;
>+
>+  template <typename IterableType>
>+  void CreateAccumulatedPacketsTimeSeries(Plot* plot,
>+                                          const IterableType& packets,
>+                                          const std::string& label);
>+
>+  void CreateStreamGapAlerts(PacketDirection direction);
>+  void CreateTransmissionGapAlerts(PacketDirection direction);
>+
>+  std::string GetStreamName(PacketDirection direction, uint32_t ssrc) const {
>+    char buffer[200];
>+    rtc::SimpleStringBuilder name(buffer);
>+    if (IsAudioSsrc(direction, ssrc)) {
>+      name << "Audio ";
>+    } else if (IsVideoSsrc(direction, ssrc)) {
>+      name << "Video ";
>+    } else {
>+      name << "Unknown ";
>+    }
>+    if (IsRtxSsrc(direction, ssrc)) {
>+      name << "RTX ";
>+    }
>+    if (direction == kIncomingPacket)
>+      name << "(In) ";
>+    else
>+      name << "(Out) ";
>+    name << "SSRC " << ssrc;
>+    return name.str();
>+  }
>+
>+  int64_t ToCallTimeUs(int64_t timestamp) const;
>+  float ToCallTimeSec(int64_t timestamp) const;
>+
>+  void Alert_RtpLogTimeGap(PacketDirection direction,
>+                           float time_seconds,
>+                           int64_t duration) {
>+    if (direction == kIncomingPacket) {
>+      incoming_rtp_recv_time_gaps_.emplace_back(time_seconds, duration);
>+    } else {
>+      outgoing_rtp_send_time_gaps_.emplace_back(time_seconds, duration);
>+    }
>+  }
>+
>+  void Alert_RtcpLogTimeGap(PacketDirection direction,
>+                            float time_seconds,
>+                            int64_t duration) {
>+    if (direction == kIncomingPacket) {
>+      incoming_rtcp_recv_time_gaps_.emplace_back(time_seconds, duration);
>+    } else {
>+      outgoing_rtcp_send_time_gaps_.emplace_back(time_seconds, duration);
>+    }
>+  }
>+
>+  void Alert_SeqNumJump(PacketDirection direction,
>+                        float time_seconds,
>+                        uint32_t ssrc) {
>+    if (direction == kIncomingPacket) {
>+      incoming_seq_num_jumps_.emplace_back(time_seconds, ssrc);
>+    } else {
>+      outgoing_seq_num_jumps_.emplace_back(time_seconds, ssrc);
>+    }
>+  }
>+
>+  void Alert_CaptureTimeJump(PacketDirection direction,
>+                             float time_seconds,
>+                             uint32_t ssrc) {
>+    if (direction == kIncomingPacket) {
>+      incoming_capture_time_jumps_.emplace_back(time_seconds, ssrc);
>+    } else {
>+      outgoing_capture_time_jumps_.emplace_back(time_seconds, ssrc);
>+    }
>+  }
> 
>-  std::string GetStreamName(StreamId stream_id) const;
>+  void Alert_OutgoingHighLoss(double avg_loss_fraction) {
>+    outgoing_high_loss_alerts_.emplace_back(avg_loss_fraction);
>+  }
> 
>-  rtc::Optional<uint32_t> EstimateRtpClockFrequency(
>-      const std::vector<LoggedRtpPacket>& packets) const;
>+  std::string GetCandidatePairLogDescriptionFromId(uint32_t candidate_pair_id);
> 
>-  const ParsedRtcEventLog& parsed_log_;
>+  const ParsedRtcEventLogNew& parsed_log_;
> 
>   // A list of SSRCs we are interested in analysing.
>   // If left empty, all SSRCs will be considered relevant.
>   std::vector<uint32_t> desired_ssrc_;
> 
>-  // Tracks what each stream is configured for. Note that a single SSRC can be
>-  // in several sets. For example, the SSRC used for sending video over RTX
>-  // will appear in both video_ssrcs_ and rtx_ssrcs_. In the unlikely case that
>-  // an SSRC is reconfigured to a different media type mid-call, it will also
>-  // appear in multiple sets.
>-  std::set<StreamId> rtx_ssrcs_;
>-  std::set<StreamId> video_ssrcs_;
>-  std::set<StreamId> audio_ssrcs_;
>-
>-  // Maps a stream identifier consisting of ssrc and direction to the parsed
>-  // RTP headers in that stream. Header extensions are parsed if the stream
>-  // has been configured.
>-  std::map<StreamId, std::vector<LoggedRtpPacket>> rtp_packets_;
>-
>-  std::map<StreamId, std::vector<LoggedRtcpPacket>> rtcp_packets_;
>-
>-  // Maps an SSRC to the timestamps of parsed audio playout events.
>-  std::map<uint32_t, std::vector<uint64_t>> audio_playout_events_;
>-
>   // Stores the timestamps for all log segments, in the form of associated start
>   // and end events.
>-  std::vector<std::pair<uint64_t, uint64_t>> log_segments_;
>-
>-  // A list of all updates from the send-side loss-based bandwidth estimator.
>-  std::vector<LossBasedBweUpdate> bwe_loss_updates_;
>-
>-  std::vector<AudioNetworkAdaptationEvent> audio_network_adaptation_events_;
>-
>-  std::vector<ParsedRtcEventLog::BweProbeClusterCreatedEvent>
>-      bwe_probe_cluster_created_events_;
>+  std::vector<std::pair<int64_t, int64_t>> log_segments_;
> 
>-  std::vector<ParsedRtcEventLog::BweProbeResultEvent> bwe_probe_result_events_;
>+  std::vector<IncomingRtpReceiveTimeGap> incoming_rtp_recv_time_gaps_;
>+  std::vector<IncomingRtcpReceiveTimeGap> incoming_rtcp_recv_time_gaps_;
>+  std::vector<OutgoingRtpSendTimeGap> outgoing_rtp_send_time_gaps_;
>+  std::vector<OutgoingRtcpSendTimeGap> outgoing_rtcp_send_time_gaps_;
>+  std::vector<IncomingSeqNumJump> incoming_seq_num_jumps_;
>+  std::vector<IncomingCaptureTimeJump> incoming_capture_time_jumps_;
>+  std::vector<OutgoingSeqNoJump> outgoing_seq_num_jumps_;
>+  std::vector<OutgoingCaptureTimeJump> outgoing_capture_time_jumps_;
>+  std::vector<OutgoingHighLoss> outgoing_high_loss_alerts_;
> 
>-  std::vector<ParsedRtcEventLog::BweDelayBasedUpdate> bwe_delay_updates_;
>+  std::map<uint32_t, std::string> candidate_pair_desc_by_id_;
> 
>   // Window and step size used for calculating moving averages, e.g. bitrate.
>   // The generated data points will be |step_| microseconds apart.
>   // Only events occuring at most |window_duration_| microseconds before the
>   // current data point will be part of the average.
>-  uint64_t window_duration_;
>-  uint64_t step_;
>+  int64_t window_duration_;
>+  int64_t step_;
> 
>   // First and last events of the log.
>-  uint64_t begin_time_;
>-  uint64_t end_time_;
>+  int64_t begin_time_;
>+  int64_t end_time_;
>+  const bool normalize_time_;
> 
>   // Duration (in seconds) of log file.
>   float call_duration_s_;
> };
> 
>-}  // namespace plotting
> }  // namespace webrtc
> 
> #endif  // RTC_TOOLS_EVENT_LOG_VISUALIZER_ANALYZER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/event_log_visualizer/main.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/event_log_visualizer/main.cc
>index e028a0f8e1e..fb497db5776 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/event_log_visualizer/main.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/event_log_visualizer/main.cc
>@@ -10,11 +10,12 @@
> 
> #include <iostream>
> 
>-#include "logging/rtc_event_log/rtc_event_log_parser.h"
>+#include "logging/rtc_event_log/rtc_event_log_parser_new.h"
> #include "rtc_base/flags.h"
> #include "rtc_tools/event_log_visualizer/analyzer.h"
> #include "rtc_tools/event_log_visualizer/plot_base.h"
> #include "rtc_tools/event_log_visualizer/plot_python.h"
>+#include "system_wrappers/include/field_trial_default.h"
> #include "test/field_trial.h"
> #include "test/testsupport/fileutils.h"
> 
>@@ -96,6 +97,12 @@ DEFINE_bool(plot_pacer_delay,
> DEFINE_bool(plot_timestamps,
>             false,
>             "Plot the rtp timestamps of all rtp and rtcp packets over time.");
>+DEFINE_bool(plot_rtcp_details,
>+            false,
>+            "Plot the contents of all report blocks in all sender and receiver "
>+            "reports. This includes fraction lost, cumulative number of lost "
>+            "packets, extended highest sequence number and time since last "
>+            "received SR.");
> DEFINE_bool(plot_audio_encoder_bitrate_bps,
>             false,
>             "Plot the audio encoder target bitrate.");
>@@ -111,9 +118,13 @@ DEFINE_bool(plot_audio_encoder_dtx, false, "Plot the audio encoder DTX.");
> DEFINE_bool(plot_audio_encoder_num_channels,
>             false,
>             "Plot the audio encoder number of channels.");
>-DEFINE_bool(plot_audio_jitter_buffer,
>+DEFINE_bool(plot_neteq_stats, false, "Plot the NetEq statistics.");
>+DEFINE_bool(plot_ice_candidate_pair_config,
>+            false,
>+            "Plot the ICE candidate pair config events.");
>+DEFINE_bool(plot_ice_connectivity_check,
>             false,
>-            "Plot the audio jitter buffer delay profile.");
>+            "Plot the ICE candidate pair connectivity checks.");
> 
> DEFINE_string(
>     force_fieldtrials,
>@@ -132,8 +143,25 @@ DEFINE_bool(show_detector_state,
>             "Show the state of the delay based BWE detector on the total "
>             "bitrate graph");
> 
>-void SetAllPlotFlags(bool setting);
>+DEFINE_bool(show_alr_state,
>+            false,
>+            "Show the state ALR state on the total bitrate graph");
>+
>+DEFINE_bool(parse_unconfigured_header_extensions,
>+            true,
>+            "Attempt to parse unconfigured header extensions using the default "
>+            "WebRTC mapping. This can give very misleading results if the "
>+            "application negotiates a different mapping.");
> 
>+DEFINE_bool(print_triage_alerts,
>+            false,
>+            "Print triage alerts, i.e. a list of potential problems.");
>+
>+DEFINE_bool(normalize_time,
>+            true,
>+            "Normalize the log timestamps so that the call starts at time 0.");
>+
>+void SetAllPlotFlags(bool setting);
> 
> int main(int argc, char* argv[]) {
>   std::string program_name = argv[0];
>@@ -186,11 +214,20 @@ int main(int argc, char* argv[]) {
>   }
> 
>   webrtc::test::SetExecutablePath(argv[0]);
>-  webrtc::test::InitFieldTrialsFromString(FLAG_force_fieldtrials);
>+  webrtc::test::ValidateFieldTrialsStringOrDie(FLAG_force_fieldtrials);
>+  // InitFieldTrialsFromString stores the char*, so the char array must outlive
>+  // the application.
>+  webrtc::field_trial::InitFieldTrialsFromString(FLAG_force_fieldtrials);
> 
>   std::string filename = argv[1];
> 
>-  webrtc::ParsedRtcEventLog parsed_log;
>+  webrtc::ParsedRtcEventLogNew::UnconfiguredHeaderExtensions header_extensions =
>+      webrtc::ParsedRtcEventLogNew::UnconfiguredHeaderExtensions::kDontParse;
>+  if (FLAG_parse_unconfigured_header_extensions) {
>+    header_extensions = webrtc::ParsedRtcEventLogNew::
>+        UnconfiguredHeaderExtensions::kAttemptWebrtcDefaultConfig;
>+  }
>+  webrtc::ParsedRtcEventLogNew parsed_log(header_extensions);
> 
>   if (!parsed_log.ParseFile(filename)) {
>     std::cerr << "Could not parse the entire log file." << std::endl;
>@@ -199,31 +236,34 @@ int main(int argc, char* argv[]) {
>               << std::endl;
>   }
> 
>-  webrtc::plotting::EventLogAnalyzer analyzer(parsed_log);
>-  std::unique_ptr<webrtc::plotting::PlotCollection> collection(
>-      new webrtc::plotting::PythonPlotCollection());
>+  webrtc::EventLogAnalyzer analyzer(parsed_log, FLAG_normalize_time);
>+  std::unique_ptr<webrtc::PlotCollection> collection(
>+      new webrtc::PythonPlotCollection());
> 
>   if (FLAG_plot_incoming_packet_sizes) {
>-    analyzer.CreatePacketGraph(webrtc::PacketDirection::kIncomingPacket,
>+    analyzer.CreatePacketGraph(webrtc::kIncomingPacket,
>                                collection->AppendNewPlot());
>   }
>   if (FLAG_plot_outgoing_packet_sizes) {
>-    analyzer.CreatePacketGraph(webrtc::PacketDirection::kOutgoingPacket,
>+    analyzer.CreatePacketGraph(webrtc::kOutgoingPacket,
>                                collection->AppendNewPlot());
>   }
>   if (FLAG_plot_incoming_packet_count) {
>-    analyzer.CreateAccumulatedPacketsGraph(
>-        webrtc::PacketDirection::kIncomingPacket, collection->AppendNewPlot());
>+    analyzer.CreateAccumulatedPacketsGraph(webrtc::kIncomingPacket,
>+                                           collection->AppendNewPlot());
>   }
>   if (FLAG_plot_outgoing_packet_count) {
>-    analyzer.CreateAccumulatedPacketsGraph(
>-        webrtc::PacketDirection::kOutgoingPacket, collection->AppendNewPlot());
>+    analyzer.CreateAccumulatedPacketsGraph(webrtc::kOutgoingPacket,
>+                                           collection->AppendNewPlot());
>   }
>   if (FLAG_plot_audio_playout) {
>     analyzer.CreatePlayoutGraph(collection->AppendNewPlot());
>   }
>   if (FLAG_plot_audio_level) {
>-    analyzer.CreateAudioLevelGraph(collection->AppendNewPlot());
>+    analyzer.CreateAudioLevelGraph(webrtc::kIncomingPacket,
>+                                   collection->AppendNewPlot());
>+    analyzer.CreateAudioLevelGraph(webrtc::kOutgoingPacket,
>+                                   collection->AppendNewPlot());
>   }
>   if (FLAG_plot_incoming_sequence_number_delta) {
>     analyzer.CreateSequenceNumberGraph(collection->AppendNewPlot());
>@@ -238,21 +278,19 @@ int main(int argc, char* argv[]) {
>     analyzer.CreateIncomingPacketLossGraph(collection->AppendNewPlot());
>   }
>   if (FLAG_plot_incoming_bitrate) {
>-    analyzer.CreateTotalBitrateGraph(webrtc::PacketDirection::kIncomingPacket,
>-                                     collection->AppendNewPlot(),
>-                                     FLAG_show_detector_state);
>+    analyzer.CreateTotalIncomingBitrateGraph(collection->AppendNewPlot());
>   }
>   if (FLAG_plot_outgoing_bitrate) {
>-    analyzer.CreateTotalBitrateGraph(webrtc::PacketDirection::kOutgoingPacket,
>-                                     collection->AppendNewPlot(),
>-                                     FLAG_show_detector_state);
>+    analyzer.CreateTotalOutgoingBitrateGraph(collection->AppendNewPlot(),
>+                                             FLAG_show_detector_state,
>+                                             FLAG_show_alr_state);
>   }
>   if (FLAG_plot_incoming_stream_bitrate) {
>-    analyzer.CreateStreamBitrateGraph(webrtc::PacketDirection::kIncomingPacket,
>+    analyzer.CreateStreamBitrateGraph(webrtc::kIncomingPacket,
>                                       collection->AppendNewPlot());
>   }
>   if (FLAG_plot_outgoing_stream_bitrate) {
>-    analyzer.CreateStreamBitrateGraph(webrtc::PacketDirection::kOutgoingPacket,
>+    analyzer.CreateStreamBitrateGraph(webrtc::kOutgoingPacket,
>                                       collection->AppendNewPlot());
>   }
>   if (FLAG_plot_simulated_receiveside_bwe) {
>@@ -268,8 +306,64 @@ int main(int argc, char* argv[]) {
>     analyzer.CreateFractionLossGraph(collection->AppendNewPlot());
>   }
>   if (FLAG_plot_timestamps) {
>-    analyzer.CreateTimestampGraph(collection->AppendNewPlot());
>+    analyzer.CreateTimestampGraph(webrtc::kIncomingPacket,
>+                                  collection->AppendNewPlot());
>+    analyzer.CreateTimestampGraph(webrtc::kOutgoingPacket,
>+                                  collection->AppendNewPlot());
>   }
>+  if (FLAG_plot_rtcp_details) {
>+    auto GetFractionLost = [](const webrtc::rtcp::ReportBlock& block) -> float {
>+      return static_cast<double>(block.fraction_lost()) / 256 * 100;
>+    };
>+    analyzer.CreateSenderAndReceiverReportPlot(
>+        webrtc::kIncomingPacket, GetFractionLost,
>+        "Fraction lost (incoming RTCP)", "Loss rate (percent)",
>+        collection->AppendNewPlot());
>+    analyzer.CreateSenderAndReceiverReportPlot(
>+        webrtc::kOutgoingPacket, GetFractionLost,
>+        "Fraction lost (outgoing RTCP)", "Loss rate (percent)",
>+        collection->AppendNewPlot());
>+
>+    auto GetCumulativeLost =
>+        [](const webrtc::rtcp::ReportBlock& block) -> float {
>+      return block.cumulative_lost_signed();
>+    };
>+    analyzer.CreateSenderAndReceiverReportPlot(
>+        webrtc::kIncomingPacket, GetCumulativeLost,
>+        "Cumulative lost packets (incoming RTCP)", "Packets",
>+        collection->AppendNewPlot());
>+    analyzer.CreateSenderAndReceiverReportPlot(
>+        webrtc::kOutgoingPacket, GetCumulativeLost,
>+        "Cumulative lost packets (outgoing RTCP)", "Packets",
>+        collection->AppendNewPlot());
>+
>+    auto GetHighestSeqNumber =
>+        [](const webrtc::rtcp::ReportBlock& block) -> float {
>+      return block.extended_high_seq_num();
>+    };
>+    analyzer.CreateSenderAndReceiverReportPlot(
>+        webrtc::kIncomingPacket, GetHighestSeqNumber,
>+        "Highest sequence number (incoming RTCP)", "Seqence number",
>+        collection->AppendNewPlot());
>+    analyzer.CreateSenderAndReceiverReportPlot(
>+        webrtc::kOutgoingPacket, GetHighestSeqNumber,
>+        "Highest sequence number (outgoing RTCP)", "Seqence number",
>+        collection->AppendNewPlot());
>+
>+    auto DelaySinceLastSr =
>+        [](const webrtc::rtcp::ReportBlock& block) -> float {
>+      return static_cast<double>(block.delay_since_last_sr()) / 65536;
>+    };
>+    analyzer.CreateSenderAndReceiverReportPlot(
>+        webrtc::kIncomingPacket, DelaySinceLastSr,
>+        "Delay since last received sender report (incoming RTCP)", "Time (s)",
>+        collection->AppendNewPlot());
>+    analyzer.CreateSenderAndReceiverReportPlot(
>+        webrtc::kOutgoingPacket, DelaySinceLastSr,
>+        "Delay since last received sender report (outgoing RTCP)", "Time (s)",
>+        collection->AppendNewPlot());
>+  }
>+
>   if (FLAG_plot_pacer_delay) {
>     analyzer.CreatePacerDelayGraph(collection->AppendNewPlot());
>   }
>@@ -291,7 +385,7 @@ int main(int argc, char* argv[]) {
>   if (FLAG_plot_audio_encoder_num_channels) {
>     analyzer.CreateAudioEncoderNumChannelsGraph(collection->AppendNewPlot());
>   }
>-  if (FLAG_plot_audio_jitter_buffer) {
>+  if (FLAG_plot_neteq_stats) {
>     std::string wav_path;
>     if (FLAG_wav_filename[0] != '\0') {
>       wav_path = FLAG_wav_filename;
>@@ -299,16 +393,62 @@ int main(int argc, char* argv[]) {
>       wav_path = webrtc::test::ResourcePath(
>           "audio_processing/conversational_speech/EN_script2_F_sp2_B1", "wav");
>     }
>-    analyzer.CreateAudioJitterBufferGraph(wav_path, 48000,
>-                                          collection->AppendNewPlot());
>+    auto neteq_stats = analyzer.SimulateNetEq(wav_path, 48000);
>+    for (webrtc::EventLogAnalyzer::NetEqStatsGetterMap::const_iterator it =
>+             neteq_stats.cbegin();
>+         it != neteq_stats.cend(); ++it) {
>+      analyzer.CreateAudioJitterBufferGraph(it->first, it->second.get(),
>+                                            collection->AppendNewPlot());
>+    }
>+    analyzer.CreateNetEqNetworkStatsGraph(
>+        neteq_stats,
>+        [](const webrtc::NetEqNetworkStatistics& stats) {
>+          return stats.expand_rate / 16384.f;
>+        },
>+        "Expand rate", collection->AppendNewPlot());
>+    analyzer.CreateNetEqNetworkStatsGraph(
>+        neteq_stats,
>+        [](const webrtc::NetEqNetworkStatistics& stats) {
>+          return stats.speech_expand_rate / 16384.f;
>+        },
>+        "Speech expand rate", collection->AppendNewPlot());
>+    analyzer.CreateNetEqNetworkStatsGraph(
>+        neteq_stats,
>+        [](const webrtc::NetEqNetworkStatistics& stats) {
>+          return stats.accelerate_rate / 16384.f;
>+        },
>+        "Accelerate rate", collection->AppendNewPlot());
>+    analyzer.CreateNetEqNetworkStatsGraph(
>+        neteq_stats,
>+        [](const webrtc::NetEqNetworkStatistics& stats) {
>+          return stats.packet_loss_rate / 16384.f;
>+        },
>+        "Packet loss rate", collection->AppendNewPlot());
>+    analyzer.CreateNetEqLifetimeStatsGraph(
>+        neteq_stats,
>+        [](const webrtc::NetEqLifetimeStatistics& stats) {
>+          return static_cast<float>(stats.concealment_events);
>+        },
>+        "Concealment events", collection->AppendNewPlot());
>+  }
>+
>+  if (FLAG_plot_ice_candidate_pair_config) {
>+    analyzer.CreateIceCandidatePairConfigGraph(collection->AppendNewPlot());
>+  }
>+  if (FLAG_plot_ice_connectivity_check) {
>+    analyzer.CreateIceConnectivityCheckGraph(collection->AppendNewPlot());
>   }
> 
>   collection->Draw();
> 
>+  if (FLAG_print_triage_alerts) {
>+    analyzer.CreateTriageNotifications();
>+    analyzer.PrintNotifications(stderr);
>+  }
>+
>   return 0;
> }
> 
>-
> void SetAllPlotFlags(bool setting) {
>   FLAG_plot_incoming_packet_sizes = setting;
>   FLAG_plot_outgoing_packet_sizes = setting;
>@@ -329,11 +469,14 @@ void SetAllPlotFlags(bool setting) {
>   FLAG_plot_network_delay_feedback = setting;
>   FLAG_plot_fraction_loss_feedback = setting;
>   FLAG_plot_timestamps = setting;
>+  FLAG_plot_rtcp_details = setting;
>   FLAG_plot_audio_encoder_bitrate_bps = setting;
>   FLAG_plot_audio_encoder_frame_length_ms = setting;
>   FLAG_plot_audio_encoder_packet_loss = setting;
>   FLAG_plot_audio_encoder_fec = setting;
>   FLAG_plot_audio_encoder_dtx = setting;
>   FLAG_plot_audio_encoder_num_channels = setting;
>-  FLAG_plot_audio_jitter_buffer = setting;
>+  FLAG_plot_neteq_stats = setting;
>+  FLAG_plot_ice_candidate_pair_config = setting;
>+  FLAG_plot_ice_connectivity_check = setting;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/event_log_visualizer/plot_base.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/event_log_visualizer/plot_base.cc
>index 7ff4ef9c2cd..0847cfe3903 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/event_log_visualizer/plot_base.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/event_log_visualizer/plot_base.cc
>@@ -11,11 +11,11 @@
> #include "rtc_tools/event_log_visualizer/plot_base.h"
> 
> #include <algorithm>
>+#include <limits>
> 
> #include "rtc_base/checks.h"
> 
> namespace webrtc {
>-namespace plotting {
> 
> void Plot::SetXAxis(float min_value,
>                     float max_value,
>@@ -85,5 +85,4 @@ void Plot::AppendTimeSeriesIfNotEmpty(TimeSeries&& time_series) {
>   }
> }
> 
>-}  // namespace plotting
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/event_log_visualizer/plot_base.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/event_log_visualizer/plot_base.h
>index 700ffbf02dd..e73f004937c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/event_log_visualizer/plot_base.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/event_log_visualizer/plot_base.h
>@@ -16,7 +16,6 @@
> #include <vector>
> 
> namespace webrtc {
>-namespace plotting {
> 
> enum class LineStyle {
>   kNone,  // No line connecting the points. Used to create scatter plots.
>@@ -173,7 +172,6 @@ class PlotCollection {
>   std::vector<std::unique_ptr<Plot> > plots_;
> };
> 
>-}  // namespace plotting
> }  // namespace webrtc
> 
> #endif  // RTC_TOOLS_EVENT_LOG_VISUALIZER_PLOT_BASE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/event_log_visualizer/plot_protobuf.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/event_log_visualizer/plot_protobuf.cc
>index e5e0a8b0941..0c4c709f833 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/event_log_visualizer/plot_protobuf.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/event_log_visualizer/plot_protobuf.cc
>@@ -13,7 +13,6 @@
> #include <memory>
> 
> namespace webrtc {
>-namespace plotting {
> 
> ProtobufPlot::ProtobufPlot() {}
> 
>@@ -70,8 +69,8 @@ void ProtobufPlotCollection::ExportProtobuf(
>     // TODO(terelius): Ensure that there is no way to insert plots other than
>     // ProtobufPlots in a ProtobufPlotCollection. Needed to safely static_cast
>     // here.
>-    webrtc::analytics::Chart* protobuf_representation
>-        = collection->add_charts();
>+    webrtc::analytics::Chart* protobuf_representation =
>+        collection->add_charts();
>     static_cast<ProtobufPlot*>(plot.get())
>         ->ExportProtobuf(protobuf_representation);
>   }
>@@ -83,5 +82,4 @@ Plot* ProtobufPlotCollection::AppendNewPlot() {
>   return plot;
> }
> 
>-}  // namespace plotting
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/event_log_visualizer/plot_protobuf.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/event_log_visualizer/plot_protobuf.h
>index 5c5cce1242b..f59d303f1e4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/event_log_visualizer/plot_protobuf.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/event_log_visualizer/plot_protobuf.h
>@@ -17,7 +17,6 @@ RTC_POP_IGNORING_WUNDEF()
> #include "rtc_tools/event_log_visualizer/plot_base.h"
> 
> namespace webrtc {
>-namespace plotting {
> 
> class ProtobufPlot final : public Plot {
>  public:
>@@ -36,7 +35,6 @@ class ProtobufPlotCollection final : public PlotCollection {
>   void ExportProtobuf(webrtc::analytics::ChartCollection* collection);
> };
> 
>-}  // namespace plotting
> }  // namespace webrtc
> 
> #endif  // RTC_TOOLS_EVENT_LOG_VISUALIZER_PLOT_PROTOBUF_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/event_log_visualizer/plot_python.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/event_log_visualizer/plot_python.cc
>index 58fad10b72d..3808e876065 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/event_log_visualizer/plot_python.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/event_log_visualizer/plot_python.cc
>@@ -17,7 +17,6 @@
> #include "rtc_base/checks.h"
> 
> namespace webrtc {
>-namespace plotting {
> 
> PythonPlot::PythonPlot() {}
> 
>@@ -39,9 +38,9 @@ void PythonPlot::Draw() {
>       // List x coordinates
>       printf("x%zu = [", i);
>       if (series_list_[i].points.size() > 0)
>-        printf("%G", series_list_[i].points[0].x);
>+        printf("%.3f", series_list_[i].points[0].x);
>       for (size_t j = 1; j < series_list_[i].points.size(); j++)
>-        printf(", %G", series_list_[i].points[j].x);
>+        printf(", %.3f", series_list_[i].points[j].x);
>       printf("]\n");
> 
>       // List y coordinates
>@@ -103,8 +102,8 @@ void PythonPlot::Draw() {
>     }
> 
>     // IntervalSeries
>-    printf("interval_colors = ['#ff8e82','#5092fc','#c4ffc4']\n");
>-    RTC_CHECK_LE(interval_list_.size(), 3);
>+    printf("interval_colors = ['#ff8e82','#5092fc','#c4ffc4','#aaaaaa']\n");
>+    RTC_CHECK_LE(interval_list_.size(), 4);
>     // To get the intervals to show up in the legend we have to create patches
>     // for them.
>     printf("legend_patches = []\n");
>@@ -180,5 +179,4 @@ Plot* PythonPlotCollection::AppendNewPlot() {
>   return plot;
> }
> 
>-}  // namespace plotting
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/event_log_visualizer/plot_python.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/event_log_visualizer/plot_python.h
>index 2a5a66c31ce..61d17a0de35 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/event_log_visualizer/plot_python.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/event_log_visualizer/plot_python.h
>@@ -13,7 +13,6 @@
> #include "rtc_tools/event_log_visualizer/plot_base.h"
> 
> namespace webrtc {
>-namespace plotting {
> 
> class PythonPlot final : public Plot {
>  public:
>@@ -30,7 +29,6 @@ class PythonPlotCollection final : public PlotCollection {
>   Plot* AppendNewPlot() override;
> };
> 
>-}  // namespace plotting
> }  // namespace webrtc
> 
> #endif  // RTC_TOOLS_EVENT_LOG_VISUALIZER_PLOT_PYTHON_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/event_log_visualizer/triage_notifications.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/event_log_visualizer/triage_notifications.h
>new file mode 100644
>index 00000000000..49e06209343
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/event_log_visualizer/triage_notifications.h
>@@ -0,0 +1,158 @@
>+/*
>+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef RTC_TOOLS_EVENT_LOG_VISUALIZER_TRIAGE_NOTIFICATIONS_H_
>+#define RTC_TOOLS_EVENT_LOG_VISUALIZER_TRIAGE_NOTIFICATIONS_H_
>+
>+#include <string>
>+
>+namespace webrtc {
>+
>+class IncomingRtpReceiveTimeGap {
>+ public:
>+  IncomingRtpReceiveTimeGap(float time_seconds, int64_t duration)
>+      : time_seconds_(time_seconds), duration_(duration) {}
>+  float Time() const { return time_seconds_; }
>+  std::string ToString() const {
>+    return std::string("No RTP packets received for ") +
>+           std::to_string(duration_) + std::string(" ms");
>+  }
>+
>+ private:
>+  float time_seconds_;
>+  int64_t duration_;
>+};
>+
>+class IncomingRtcpReceiveTimeGap {
>+ public:
>+  IncomingRtcpReceiveTimeGap(float time_seconds, int64_t duration)
>+      : time_seconds_(time_seconds), duration_(duration) {}
>+  float Time() const { return time_seconds_; }
>+  std::string ToString() const {
>+    return std::string("No RTCP packets received for ") +
>+           std::to_string(duration_) + std::string(" ms");
>+  }
>+
>+ private:
>+  float time_seconds_;
>+  int64_t duration_;
>+};
>+
>+class OutgoingRtpSendTimeGap {
>+ public:
>+  OutgoingRtpSendTimeGap(float time_seconds, int64_t duration)
>+      : time_seconds_(time_seconds), duration_(duration) {}
>+  float Time() const { return time_seconds_; }
>+  std::string ToString() const {
>+    return std::string("No RTP packets sent for ") + std::to_string(duration_) +
>+           std::string(" ms");
>+  }
>+
>+ private:
>+  float time_seconds_;
>+  int64_t duration_;
>+};
>+
>+class OutgoingRtcpSendTimeGap {
>+ public:
>+  OutgoingRtcpSendTimeGap(float time_seconds, int64_t duration)
>+      : time_seconds_(time_seconds), duration_(duration) {}
>+  float Time() const { return time_seconds_; }
>+  std::string ToString() const {
>+    return std::string("No RTCP packets sent for ") +
>+           std::to_string(duration_) + std::string(" ms");
>+  }
>+
>+ private:
>+  float time_seconds_;
>+  int64_t duration_;
>+};
>+
>+class IncomingSeqNumJump {
>+ public:
>+  IncomingSeqNumJump(float time_seconds, uint32_t ssrc)
>+      : time_seconds_(time_seconds), ssrc_(ssrc) {}
>+  float Time() const { return time_seconds_; }
>+  std::string ToString() const {
>+    return std::string("Sequence number jumps on incoming SSRC ") +
>+           std::to_string(ssrc_);
>+  }
>+
>+ private:
>+  float time_seconds_;
>+
>+  uint32_t ssrc_;
>+};
>+
>+class IncomingCaptureTimeJump {
>+ public:
>+  IncomingCaptureTimeJump(float time_seconds, uint32_t ssrc)
>+      : time_seconds_(time_seconds), ssrc_(ssrc) {}
>+  float Time() const { return time_seconds_; }
>+  std::string ToString() const {
>+    return std::string("Capture timestamp jumps on incoming SSRC ") +
>+           std::to_string(ssrc_);
>+  }
>+
>+ private:
>+  float time_seconds_;
>+
>+  uint32_t ssrc_;
>+};
>+
>+class OutgoingSeqNoJump {
>+ public:
>+  OutgoingSeqNoJump(float time_seconds, uint32_t ssrc)
>+      : time_seconds_(time_seconds), ssrc_(ssrc) {}
>+  float Time() const { return time_seconds_; }
>+  std::string ToString() const {
>+    return std::string("Sequence number jumps on outgoing SSRC ") +
>+           std::to_string(ssrc_);
>+  }
>+
>+ private:
>+  float time_seconds_;
>+
>+  uint32_t ssrc_;
>+};
>+
>+class OutgoingCaptureTimeJump {
>+ public:
>+  OutgoingCaptureTimeJump(float time_seconds, uint32_t ssrc)
>+      : time_seconds_(time_seconds), ssrc_(ssrc) {}
>+  float Time() const { return time_seconds_; }
>+  std::string ToString() const {
>+    return std::string("Capture timestamp jumps on outgoing SSRC ") +
>+           std::to_string(ssrc_);
>+  }
>+
>+ private:
>+  float time_seconds_;
>+
>+  uint32_t ssrc_;
>+};
>+
>+class OutgoingHighLoss {
>+ public:
>+  explicit OutgoingHighLoss(double avg_loss_fraction)
>+      : avg_loss_fraction_(avg_loss_fraction) {}
>+  std::string ToString() const {
>+    return std::string("High average loss (") +
>+           std::to_string(avg_loss_fraction_ * 100) +
>+           std::string("%) across the call.");
>+  }
>+
>+ private:
>+  double avg_loss_fraction_;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // RTC_TOOLS_EVENT_LOG_VISUALIZER_TRIAGE_NOTIFICATIONS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/force_mic_volume_max/force_mic_volume_max.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/force_mic_volume_max/force_mic_volume_max.cc
>index 40fb023f183..2cd1b36d904 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/force_mic_volume_max/force_mic_volume_max.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/force_mic_volume_max/force_mic_volume_max.cc
>@@ -22,7 +22,7 @@ using webrtc::AudioDeviceModule;
> #define DEFAULT_INPUT_DEVICE (0)
> #endif
> 
>-int main(int /*argc*/, char** /*argv*/) {
>+int main(int /*argc*/, char* /*argv*/ []) {
>   // Create and initialize the ADM.
>   rtc::scoped_refptr<AudioDeviceModule> adm(
>       AudioDeviceModule::Create(AudioDeviceModule::kPlatformDefaultAudio));
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/frame_analyzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/frame_analyzer.cc
>index 98f47430433..b3d9b94b0a7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/frame_analyzer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/frame_analyzer.cc
>@@ -17,6 +17,7 @@
> 
> #include "rtc_tools/frame_analyzer/video_quality_analysis.h"
> #include "rtc_tools/simple_command_line_parser.h"
>+#include "test/testsupport/perf_test.h"
> 
> /*
>  * A command line tool running PSNR and SSIM on a reference video and a test
>@@ -39,7 +40,7 @@
>  * --stats_file=<name_of_file> --width=<frame_width>
>  * --height=<frame_height>
>  */
>-int main(int argc, char** argv) {
>+int main(int argc, char* argv[]) {
>   std::string program_name = argv[0];
>   std::string usage =
>       "Compares the output video with the initially sent video."
>@@ -62,7 +63,10 @@ int main(int argc, char** argv) {
>       "  - reference_file(string): The reference YUV file to compare against."
>       " Default: ref.yuv\n"
>       "  - test_file(string): The test YUV file to run the analysis for."
>-      " Default: test_file.yuv\n";
>+      " Default: test_file.yuv\n"
>+      "  - chartjson_result_file: Where to store perf result in chartjson"
>+      " format. If not present, no perf result will be stored."
>+      " Default: None\n";
> 
>   webrtc::test::CommandLineParser parser;
> 
>@@ -77,6 +81,7 @@ int main(int argc, char** argv) {
>   parser.SetFlag("stats_file_test", "stats_test.txt");
>   parser.SetFlag("reference_file", "ref.yuv");
>   parser.SetFlag("test_file", "test.yuv");
>+  parser.SetFlag("chartjson_result_file", "");
>   parser.SetFlag("help", "false");
> 
>   parser.ProcessFlags();
>@@ -101,10 +106,16 @@ int main(int argc, char** argv) {
>                             parser.GetFlag("stats_file_ref").c_str(),
>                             parser.GetFlag("stats_file_test").c_str(), width,
>                             height, &results);
>+  webrtc::test::GetMaxRepeatedAndSkippedFrames(
>+      parser.GetFlag("stats_file_ref"), parser.GetFlag("stats_file_test"),
>+      &results);
> 
>-  std::string label = parser.GetFlag("label");
>-  webrtc::test::PrintAnalysisResults(label, &results);
>-  webrtc::test::PrintMaxRepeatedAndSkippedFrames(
>-      label, parser.GetFlag("stats_file_ref"),
>-      parser.GetFlag("stats_file_test"));
>+  webrtc::test::PrintAnalysisResults(parser.GetFlag("label"), &results);
>+
>+  std::string chartjson_result_file = parser.GetFlag("chartjson_result_file");
>+  if (!chartjson_result_file.empty()) {
>+    webrtc::test::WritePerfResults(chartjson_result_file);
>+  }
>+
>+  return 0;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/reference_less_video_analysis.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/reference_less_video_analysis.cc
>index d75b3af731b..3d7fd7fc487 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/reference_less_video_analysis.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/reference_less_video_analysis.cc
>@@ -14,11 +14,13 @@
> #include "rtc_tools/frame_analyzer/reference_less_video_analysis_lib.h"
> #include "rtc_tools/simple_command_line_parser.h"
> 
>-int main(int argc, char** argv) {
>+int main(int argc, char* argv[]) {
>   // This captures the freezing metrics for reference less video analysis.
>   std::string program_name = argv[0];
>-  std::string usage = "Outputs the freezing score by comparing current frame "
>-      "with the previous frame.\nExample usage:\n" + program_name +
>+  std::string usage =
>+      "Outputs the freezing score by comparing current frame "
>+      "with the previous frame.\nExample usage:\n" +
>+      program_name +
>       " --video_file=video_file.y4m\n"
>       "Command line flags:\n"
>       "  - video_file(string): Path of the video "
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/reference_less_video_analysis_lib.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/reference_less_video_analysis_lib.cc
>index 78d10cecf92..9a7535b213a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/reference_less_video_analysis_lib.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/reference_less_video_analysis_lib.cc
>@@ -25,7 +25,9 @@
> #define strtok_r strtok_s
> #endif
> 
>-void get_height_width_fps(int *height, int *width, int *fps,
>+void get_height_width_fps(int* height,
>+                          int* width,
>+                          int* fps,
>                           const std::string& video_file) {
>   // File header looks like :
>   //  YUV4MPEG2 W1280 H720 F25:1 Ip A0:0 C420mpeg2 XYSCSS=420MPEG2.
>@@ -37,8 +39,8 @@ void get_height_width_fps(int *height, int *width, int *fps,
>   frame_header[bytes_read] = '\0';
>   std::string file_header_stats[5];
>   int no_of_stats = 0;
>-  char *save_ptr;
>-  char *token = strtok_r(frame_header, " ", &save_ptr);
>+  char* save_ptr;
>+  char* token = strtok_r(frame_header, " ", &save_ptr);
> 
>   while (token != NULL) {
>     file_header_stats[no_of_stats++] = token;
>@@ -54,27 +56,29 @@ void get_height_width_fps(int *height, int *width, int *fps,
> }
> 
> bool frozen_frame(std::vector<double> psnr_per_frame,
>-                  std::vector<double> ssim_per_frame, size_t frame) {
>+                  std::vector<double> ssim_per_frame,
>+                  size_t frame) {
>   if (psnr_per_frame[frame] >= PSNR_FREEZE_THRESHOLD ||
>       ssim_per_frame[frame] >= SSIM_FREEZE_THRESHOLD)
>     return true;
>   return false;
> }
> 
>-std::vector<int> find_frame_clusters(const std::vector<double>& psnr_per_frame,
>-                                   const std::vector<double>& ssim_per_frame) {
>+std::vector<int> find_frame_clusters(
>+    const std::vector<double>& psnr_per_frame,
>+    const std::vector<double>& ssim_per_frame) {
>   std::vector<int> identical_frame_clusters;
>   int num_frozen = 0;
>   size_t total_no_of_frames = psnr_per_frame.size();
> 
>   for (size_t each_frame = 0; each_frame < total_no_of_frames; each_frame++) {
>-      if (frozen_frame(psnr_per_frame, ssim_per_frame, each_frame)) {
>-        num_frozen++;
>-        } else if (num_frozen > 0) {
>-          // Not frozen anymore.
>-          identical_frame_clusters.push_back(num_frozen);
>-          num_frozen = 0;
>-        }
>+    if (frozen_frame(psnr_per_frame, ssim_per_frame, each_frame)) {
>+      num_frozen++;
>+    } else if (num_frozen > 0) {
>+      // Not frozen anymore.
>+      identical_frame_clusters.push_back(num_frozen);
>+      num_frozen = 0;
>+    }
>   }
>   return identical_frame_clusters;
> }
>@@ -106,8 +110,8 @@ void print_freezing_metrics(const std::vector<double>& psnr_per_frame,
>    *
>    */
>   size_t total_no_of_frames = psnr_per_frame.size();
>-  std::vector<int> identical_frame_clusters = find_frame_clusters(
>-        psnr_per_frame, ssim_per_frame);
>+  std::vector<int> identical_frame_clusters =
>+      find_frame_clusters(psnr_per_frame, ssim_per_frame);
>   int total_identical_frames = std::accumulate(
>       identical_frame_clusters.begin(), identical_frame_clusters.end(), 0);
>   size_t unique_frames = total_no_of_frames - total_identical_frames;
>@@ -126,7 +130,7 @@ void print_freezing_metrics(const std::vector<double>& psnr_per_frame,
> 
>   printf("Print identical frame which appears in clusters : \n");
>   for (int cluster = 0;
>-      cluster < static_cast<int>(identical_frame_clusters.size()); cluster++)
>+       cluster < static_cast<int>(identical_frame_clusters.size()); cluster++)
>     printf("%d ", identical_frame_clusters[cluster]);
>   printf("\n");
> }
>@@ -145,28 +149,22 @@ void compute_metrics(const std::string& video_file_name,
>   uint8_t* next_frame = new uint8_t[size];
> 
>   while (true) {
>-    if (!(webrtc::test::ExtractFrameFromY4mFile (video_file_name.c_str(),
>-                                                 width, height,
>-                                                 no_of_frames,
>-                                                 current_frame))) {
>+    if (!(webrtc::test::ExtractFrameFromY4mFile(video_file_name.c_str(), width,
>+                                                height, no_of_frames,
>+                                                current_frame))) {
>       break;
>     }
> 
>-    if (!(webrtc::test::ExtractFrameFromY4mFile (video_file_name.c_str(),
>-                                                 width, height,
>-                                                 no_of_frames + 1,
>-                                                 next_frame))) {
>+    if (!(webrtc::test::ExtractFrameFromY4mFile(video_file_name.c_str(), width,
>+                                                height, no_of_frames + 1,
>+                                                next_frame))) {
>       break;
>     }
> 
>-    double result_psnr = webrtc::test::CalculateMetrics(webrtc::test::kPSNR,
>-                                                        current_frame,
>-                                                        next_frame,
>-                                                        width, height);
>-    double result_ssim = webrtc::test::CalculateMetrics(webrtc::test::kSSIM,
>-                                                        current_frame,
>-                                                        next_frame,
>-                                                        width, height);
>+    double result_psnr = webrtc::test::CalculateMetrics(
>+        webrtc::test::kPSNR, current_frame, next_frame, width, height);
>+    double result_ssim = webrtc::test::CalculateMetrics(
>+        webrtc::test::kSSIM, current_frame, next_frame, width, height);
> 
>     psnr_per_frame->push_back(result_psnr);
>     ssim_per_frame->push_back(result_ssim);
>@@ -178,7 +176,7 @@ void compute_metrics(const std::string& video_file_name,
> }
> 
> bool check_file_extension(const std::string& video_file_name) {
>-  if (video_file_name.substr(video_file_name.length()-3, 3) != "y4m") {
>+  if (video_file_name.substr(video_file_name.length() - 3, 3) != "y4m") {
>     printf("Only y4m video file format is supported. Given: %s\n",
>            video_file_name.c_str());
>     return false;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/reference_less_video_analysis_lib.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/reference_less_video_analysis_lib.h
>index 1a7b1b1bd57..a1de03baf61 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/reference_less_video_analysis_lib.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/reference_less_video_analysis_lib.h
>@@ -16,13 +16,16 @@
> 
> // Parse the file header to extract height, width and fps
> // for a given video file.
>-void get_height_width_fps(int *height, int *width, int *fps,
>+void get_height_width_fps(int* height,
>+                          int* width,
>+                          int* fps,
>                           const std::string& video_file);
> 
> // Returns true if the frame is frozen based on psnr and ssim freezing
> // threshold values.
> bool frozen_frame(std::vector<double> psnr_per_frame,
>-                     std::vector<double> ssim_per_frame, size_t frame);
>+                  std::vector<double> ssim_per_frame,
>+                  size_t frame);
> 
> // Returns the vector of identical cluster of frames that are frozen
> // and appears continuously.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/reference_less_video_analysis_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/reference_less_video_analysis_unittest.cc
>index 13ccb854226..4e205320002 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/reference_less_video_analysis_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/reference_less_video_analysis_unittest.cc
>@@ -7,8 +7,8 @@
>  *  in the file PATENTS.  All contributing project authors may
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
>-#include <string.h>
> #include <stdio.h>
>+#include <string.h>
> #include <cstring>
> #include <iostream>
> #include <vector>
>@@ -21,7 +21,7 @@ class ReferenceLessVideoAnalysisTest : public ::testing::Test {
>  public:
>   void SetUp() override {
>     video_file =
>-           webrtc::test::ResourcePath("reference_less_video_test_file", "y4m");
>+        webrtc::test::ResourcePath("reference_less_video_test_file", "y4m");
>   }
>   std::string video_file;
>   std::vector<double> psnr_per_frame;
>@@ -62,7 +62,3 @@ TEST_F(ReferenceLessVideoAnalysisTest, CheckFileExtension) {
>       webrtc::test::ResourcePath("video_quality_analysis_frame", "txt");
>   EXPECT_FALSE(check_file_extension(txt_file));
> }
>-
>-
>-
>-
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/video_quality_analysis.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/video_quality_analysis.cc
>index 7dce7344986..60a4a016cca 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/video_quality_analysis.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/video_quality_analysis.cc
>@@ -14,10 +14,12 @@
> #include <stdio.h>
> #include <stdlib.h>
> #include <algorithm>
>-#include <string>
> #include <map>
>+#include <string>
> #include <utility>
> 
>+#include "test/testsupport/perf_test.h"
>+
> #define STATS_LINE_LENGTH 32
> #define Y4M_FILE_HEADER_MAX_SIZE 200
> #define Y4M_FRAME_DELIMITER "FRAME"
>@@ -33,7 +35,7 @@ int GetI420FrameSize(int width, int height) {
>   int half_width = (width + 1) >> 1;
>   int half_height = (height + 1) >> 1;
> 
>-  int y_plane = width * height;  // I420 Y plane.
>+  int y_plane = width * height;            // I420 Y plane.
>   int u_plane = half_width * half_height;  // I420 U plane.
>   int v_plane = half_width * half_height;  // I420 V plane.
> 
>@@ -86,7 +88,7 @@ bool GetNextStatsLine(FILE* stats_file, char* line) {
>     line[chars] = buf;
>     ++chars;
>   }
>-  line[chars-1] = '\0';  // Strip the trailing \n and put end of string.
>+  line[chars - 1] = '\0';  // Strip the trailing \n and put end of string.
>   return true;
> }
> 
>@@ -110,8 +112,7 @@ bool ExtractFrameFromYuvFile(const char* i420_file_name,
>   fseek(input_file, offset, SEEK_SET);
> 
>   size_t bytes_read = fread(result_frame, 1, frame_size, input_file);
>-  if (bytes_read != static_cast<size_t>(frame_size) &&
>-      ferror(input_file)) {
>+  if (bytes_read != static_cast<size_t>(frame_size) && ferror(input_file)) {
>     fprintf(stdout, "Error while reading frame no %d from file %s\n",
>             frame_number, i420_file_name);
>     errors = true;
>@@ -142,8 +143,7 @@ bool ExtractFrameFromY4mFile(const char* y4m_file_name,
>   size_t bytes_read =
>       fread(frame_header, 1, Y4M_FILE_HEADER_MAX_SIZE - 1, input_file);
>   if (bytes_read != static_cast<size_t>(frame_size) && ferror(input_file)) {
>-    fprintf(stdout, "Error while reading frame from file %s\n",
>-        y4m_file_name);
>+    fprintf(stdout, "Error while reading frame from file %s\n", y4m_file_name);
>     fclose(input_file);
>     return false;
>   }
>@@ -152,7 +152,7 @@ bool ExtractFrameFromY4mFile(const char* y4m_file_name,
>   std::size_t found = header_contents.find(Y4M_FRAME_DELIMITER);
>   if (found == std::string::npos) {
>     fprintf(stdout, "Corrupted Y4M header, could not find \"FRAME\" in %s\n",
>-        header_contents.c_str());
>+            header_contents.c_str());
>     fclose(input_file);
>     return false;
>   }
>@@ -204,19 +204,17 @@ double CalculateMetrics(VideoAnalysisMetricsType video_metrics_type,
>   switch (video_metrics_type) {
>     case kPSNR:
>       // In the following: stride is determined by width.
>-      result = libyuv::I420Psnr(src_y_a, width, src_u_a, half_width,
>-                                src_v_a, half_width, src_y_b, width,
>-                                src_u_b, half_width, src_v_b, half_width,
>-                                width, height);
>+      result = libyuv::I420Psnr(src_y_a, width, src_u_a, half_width, src_v_a,
>+                                half_width, src_y_b, width, src_u_b, half_width,
>+                                src_v_b, half_width, width, height);
>       // LibYuv sets the max psnr value to 128, we restrict it to 48.
>       // In case of 0 mse in one frame, 128 can skew the results significantly.
>       result = (result > 48.0) ? 48.0 : result;
>       break;
>     case kSSIM:
>-      result = libyuv::I420Ssim(src_y_a, stride_y, src_u_a, stride_uv,
>-                                src_v_a, stride_uv, src_y_b, stride_y,
>-                                src_u_b, stride_uv, src_v_b, stride_uv,
>-                                width, height);
>+      result = libyuv::I420Ssim(src_y_a, stride_y, src_u_a, stride_uv, src_v_a,
>+                                stride_uv, src_y_b, stride_y, src_u_b,
>+                                stride_uv, src_v_b, stride_uv, width, height);
>       break;
>     default:
>       assert(false);
>@@ -318,13 +316,6 @@ void RunAnalysis(const char* reference_file_name,
>   delete[] reference_frame;
> }
> 
>-void PrintMaxRepeatedAndSkippedFrames(const std::string& label,
>-                                      const std::string& stats_file_ref_name,
>-                                      const std::string& stats_file_test_name) {
>-  PrintMaxRepeatedAndSkippedFrames(stdout, label, stats_file_ref_name,
>-                                   stats_file_test_name);
>-}
>-
> std::vector<std::pair<int, int> > CalculateFrameClusters(
>     FILE* file,
>     int* num_decode_errors) {
>@@ -359,10 +350,9 @@ std::vector<std::pair<int, int> > CalculateFrameClusters(
>   return frame_cnt;
> }
> 
>-void PrintMaxRepeatedAndSkippedFrames(FILE* output,
>-                                      const std::string& label,
>-                                      const std::string& stats_file_ref_name,
>-                                      const std::string& stats_file_test_name) {
>+void GetMaxRepeatedAndSkippedFrames(const std::string& stats_file_ref_name,
>+                                    const std::string& stats_file_test_name,
>+                                    ResultsContainer* results) {
>   FILE* stats_file_ref = fopen(stats_file_ref_name.c_str(), "r");
>   FILE* stats_file_test = fopen(stats_file_test_name.c_str(), "r");
>   if (stats_file_ref == NULL) {
>@@ -460,50 +450,52 @@ void PrintMaxRepeatedAndSkippedFrames(FILE* output,
>       }
>       continue;
>     }
>-    fprintf(output,
>+    fprintf(stdout,
>             "Found barcode %d in test video, which is not in reference video\n",
>             it_test->first);
>     break;
>   }
> 
>-  fprintf(output, "RESULT Max_repeated: %s= %d\n", label.c_str(),
>-          max_repeated_frames);
>-  fprintf(output, "RESULT Max_skipped: %s= %d\n", label.c_str(),
>-          max_skipped_frames);
>-  fprintf(output, "RESULT Total_skipped: %s= %d\n", label.c_str(),
>-          total_skipped_frames);
>-  fprintf(output, "RESULT Decode_errors_reference: %s= %d\n", label.c_str(),
>-          decode_errors_ref);
>-  fprintf(output, "RESULT Decode_errors_test: %s= %d\n", label.c_str(),
>-          decode_errors_test);
>+  results->max_repeated_frames = max_repeated_frames;
>+  results->max_skipped_frames = max_skipped_frames;
>+  results->total_skipped_frames = total_skipped_frames;
>+  results->decode_errors_ref = decode_errors_ref;
>+  results->decode_errors_test = decode_errors_test;
> }
> 
> void PrintAnalysisResults(const std::string& label, ResultsContainer* results) {
>   PrintAnalysisResults(stdout, label, results);
> }
> 
>-void PrintAnalysisResults(FILE* output, const std::string& label,
>+void PrintAnalysisResults(FILE* output,
>+                          const std::string& label,
>                           ResultsContainer* results) {
>-  std::vector<AnalysisResult>::iterator iter;
>-
>-  fprintf(output, "RESULT Unique_frames_count: %s= %u score\n", label.c_str(),
>-          static_cast<unsigned int>(results->frames.size()));
>+  SetPerfResultsOutput(output);
> 
>   if (results->frames.size() > 0u) {
>-    fprintf(output, "RESULT PSNR: %s= [", label.c_str());
>-    for (iter = results->frames.begin(); iter != results->frames.end() - 1;
>-         ++iter) {
>-      fprintf(output, "%f,", iter->psnr_value);
>+    PrintResult("Unique_frames_count", "", label, results->frames.size(),
>+                "score", false);
>+
>+    std::vector<double> psnr_values;
>+    std::vector<double> ssim_values;
>+    for (const auto& frame : results->frames) {
>+      psnr_values.push_back(frame.psnr_value);
>+      ssim_values.push_back(frame.ssim_value);
>     }
>-    fprintf(output, "%f] dB\n", iter->psnr_value);
> 
>-    fprintf(output, "RESULT SSIM: %s= [", label.c_str());
>-    for (iter = results->frames.begin(); iter != results->frames.end() - 1;
>-         ++iter) {
>-      fprintf(output, "%f,", iter->ssim_value);
>-    }
>-    fprintf(output, "%f] score\n", iter->ssim_value);
>+    PrintResultList("PSNR", "", label, psnr_values, "dB", false);
>+    PrintResultList("SSIM", "", label, ssim_values, "score", false);
>   }
>+
>+  PrintResult("Max_repeated", "", label, results->max_repeated_frames, "",
>+              false);
>+  PrintResult("Max_skipped", "", label, results->max_skipped_frames, "", false);
>+  PrintResult("Total_skipped", "", label, results->total_skipped_frames, "",
>+              false);
>+  PrintResult("Decode_errors_reference", "", label, results->decode_errors_ref,
>+              "", false);
>+  PrintResult("Decode_errors_test", "", label, results->decode_errors_test, "",
>+              false);
> }
> 
> }  // namespace test
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/video_quality_analysis.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/video_quality_analysis.h
>index c7a179cc268..dca719d295b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/video_quality_analysis.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/video_quality_analysis.h
>@@ -12,11 +12,11 @@
> #define RTC_TOOLS_FRAME_ANALYZER_VIDEO_QUALITY_ANALYSIS_H_
> 
> #include <string>
>-#include <vector>
> #include <utility>
>+#include <vector>
> 
>-#include "libyuv/compare.h"  // NOLINT
>-#include "libyuv/convert.h"  // NOLINT
>+#include "third_party/libyuv/include/libyuv/compare.h"
>+#include "third_party/libyuv/include/libyuv/convert.h"
> 
> namespace webrtc {
> namespace test {
>@@ -37,9 +37,14 @@ struct ResultsContainer {
>   ~ResultsContainer();
> 
>   std::vector<AnalysisResult> frames;
>+  int max_repeated_frames;
>+  int max_skipped_frames;
>+  int total_skipped_frames;
>+  int decode_errors_ref;
>+  int decode_errors_test;
> };
> 
>-enum VideoAnalysisMetricsType {kPSNR, kSSIM};
>+enum VideoAnalysisMetricsType { kPSNR, kSSIM };
> 
> // A function to run the PSNR and SSIM analysis on the test file. The test file
> // comprises the frames that were captured during the quality measurement test.
>@@ -80,7 +85,8 @@ double CalculateMetrics(VideoAnalysisMetricsType video_metrics_type,
> void PrintAnalysisResults(const std::string& label, ResultsContainer* results);
> 
> // Similar to the above, but will print to the specified file handle.
>-void PrintAnalysisResults(FILE* output, const std::string& label,
>+void PrintAnalysisResults(FILE* output,
>+                          const std::string& label,
>                           ResultsContainer* results);
> 
> // The barcode number that means that the barcode could not be decoded.
>@@ -102,15 +108,9 @@ std::vector<std::pair<int, int> > CalculateFrameClusters(
> 
> // Calculates max repeated and skipped frames and prints them to stdout in a
> // format that is compatible with Chromium performance numbers.
>-void PrintMaxRepeatedAndSkippedFrames(const std::string& label,
>-                                      const std::string& stats_file_ref_name,
>-                                      const std::string& stats_file_test_name);
>-
>-// Similar to the above, but will print to the specified file handle.
>-void PrintMaxRepeatedAndSkippedFrames(FILE* output,
>-                                      const std::string& label,
>-                                      const std::string& stats_file_ref_name,
>-                                      const std::string& stats_file_test_name);
>+void GetMaxRepeatedAndSkippedFrames(const std::string& stats_file_ref_name,
>+                                    const std::string& stats_file_test_name,
>+                                    ResultsContainer* results);
> 
> // Gets the next line from an open stats file.
> bool GetNextStatsLine(FILE* stats_file, char* line);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/video_quality_analysis_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/video_quality_analysis_unittest.cc
>index 0b1258a23ef..d1edb30a198 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/video_quality_analysis_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_analyzer/video_quality_analysis_unittest.cc
>@@ -43,10 +43,10 @@ class VideoQualityAnalysisTest : public ::testing::Test {
> 
> TEST_F(VideoQualityAnalysisTest, MatchExtractedY4mFrame) {
>   std::string video_file =
>-         webrtc::test::ResourcePath("reference_less_video_test_file", "y4m");
>+      webrtc::test::ResourcePath("reference_less_video_test_file", "y4m");
> 
>   std::string extracted_frame_from_video_file =
>-         webrtc::test::ResourcePath("video_quality_analysis_frame", "txt");
>+      webrtc::test::ResourcePath("video_quality_analysis_frame", "txt");
> 
>   int frame_height = 720, frame_width = 1280;
>   int frame_number = 2;
>@@ -57,8 +57,7 @@ TEST_F(VideoQualityAnalysisTest, MatchExtractedY4mFrame) {
>   FILE* input_file = fopen(extracted_frame_from_video_file.c_str(), "rb");
>   fread(expected_frame, 1, size, input_file);
> 
>-  ExtractFrameFromY4mFile(video_file.c_str(),
>-                          frame_width, frame_height,
>+  ExtractFrameFromY4mFile(video_file.c_str(), frame_width, frame_height,
>                           frame_number, result_frame);
> 
>   EXPECT_EQ(*expected_frame, *result_frame);
>@@ -86,24 +85,24 @@ TEST_F(VideoQualityAnalysisTest, PrintAnalysisResultsThreeFrames) {
>   PrintAnalysisResults(logfile_, "ThreeFrames", &result);
> }
> 
>-TEST_F(VideoQualityAnalysisTest, PrintMaxRepeatedAndSkippedFramesInvalidFile) {
>+TEST_F(VideoQualityAnalysisTest, GetMaxRepeatedAndSkippedFramesInvalidFile) {
>+  ResultsContainer result;
>   remove(stats_filename_.c_str());
>-  PrintMaxRepeatedAndSkippedFrames(logfile_, "NonExistingStatsFile",
>-                                   stats_filename_ref_, stats_filename_);
>+  GetMaxRepeatedAndSkippedFrames(stats_filename_ref_, stats_filename_, &result);
> }
> 
>-TEST_F(VideoQualityAnalysisTest,
>-       PrintMaxRepeatedAndSkippedFramesEmptyStatsFile) {
>+TEST_F(VideoQualityAnalysisTest, GetMaxRepeatedAndSkippedFramesEmptyStatsFile) {
>+  ResultsContainer result;
>   std::ofstream stats_file;
>   stats_file.open(stats_filename_ref_.c_str());
>   stats_file.close();
>   stats_file.open(stats_filename_.c_str());
>   stats_file.close();
>-  PrintMaxRepeatedAndSkippedFrames(logfile_, "EmptyStatsFile",
>-                                   stats_filename_ref_, stats_filename_);
>+  GetMaxRepeatedAndSkippedFrames(stats_filename_ref_, stats_filename_, &result);
> }
> 
>-TEST_F(VideoQualityAnalysisTest, PrintMaxRepeatedAndSkippedFramesNormalFile) {
>+TEST_F(VideoQualityAnalysisTest, GetMaxRepeatedAndSkippedFramesNormalFile) {
>+  ResultsContainer result;
>   std::ofstream stats_file;
> 
>   stats_file.open(stats_filename_ref_.c_str());
>@@ -123,8 +122,7 @@ TEST_F(VideoQualityAnalysisTest, PrintMaxRepeatedAndSkippedFramesNormalFile) {
>   stats_file << "frame_0004 0106\n";
>   stats_file.close();
> 
>-  PrintMaxRepeatedAndSkippedFrames(logfile_, "NormalStatsFile",
>-                                   stats_filename_ref_, stats_filename_);
>+  GetMaxRepeatedAndSkippedFrames(stats_filename_ref_, stats_filename_, &result);
> }
> 
> namespace {
>@@ -143,6 +141,7 @@ void VerifyLogOutput(const std::string& log_filename,
> 
> TEST_F(VideoQualityAnalysisTest,
>        PrintMaxRepeatedAndSkippedFramesSkippedFrames) {
>+  ResultsContainer result;
>   std::ofstream stats_file;
> 
>   std::string log_filename =
>@@ -171,21 +170,22 @@ TEST_F(VideoQualityAnalysisTest,
>   stats_file << "frame_0006 0112\n";
>   stats_file.close();
> 
>-  PrintMaxRepeatedAndSkippedFrames(logfile, "NormalStatsFile",
>-                                   stats_filename_ref_, stats_filename_);
>+  GetMaxRepeatedAndSkippedFrames(stats_filename_ref_, stats_filename_, &result);
>+  PrintAnalysisResults(logfile, "NormalStatsFile", &result);
>   ASSERT_EQ(0, fclose(logfile));
> 
>   std::vector<std::string> expected_out = {
>-      "RESULT Max_repeated: NormalStatsFile= 2",
>-      "RESULT Max_skipped: NormalStatsFile= 2",
>-      "RESULT Total_skipped: NormalStatsFile= 3",
>-      "RESULT Decode_errors_reference: NormalStatsFile= 0",
>-      "RESULT Decode_errors_test: NormalStatsFile= 0"};
>+      "RESULT Max_repeated: NormalStatsFile= 2 ",
>+      "RESULT Max_skipped: NormalStatsFile= 2 ",
>+      "RESULT Total_skipped: NormalStatsFile= 3 ",
>+      "RESULT Decode_errors_reference: NormalStatsFile= 0 ",
>+      "RESULT Decode_errors_test: NormalStatsFile= 0 "};
>   VerifyLogOutput(log_filename, expected_out);
> }
> 
> TEST_F(VideoQualityAnalysisTest,
>        PrintMaxRepeatedAndSkippedFramesDecodeErrorInTest) {
>+  ResultsContainer result;
>   std::ofstream stats_file;
> 
>   std::string log_filename =
>@@ -214,16 +214,16 @@ TEST_F(VideoQualityAnalysisTest,
>   stats_file << "frame_0006 0110\n";
>   stats_file.close();
> 
>-  PrintMaxRepeatedAndSkippedFrames(logfile, "NormalStatsFile",
>-                                   stats_filename_ref_, stats_filename_);
>+  GetMaxRepeatedAndSkippedFrames(stats_filename_ref_, stats_filename_, &result);
>+  PrintAnalysisResults(logfile, "NormalStatsFile", &result);
>   ASSERT_EQ(0, fclose(logfile));
> 
>   std::vector<std::string> expected_out = {
>-      "RESULT Max_repeated: NormalStatsFile= 1",
>-      "RESULT Max_skipped: NormalStatsFile= 0",
>-      "RESULT Total_skipped: NormalStatsFile= 0",
>-      "RESULT Decode_errors_reference: NormalStatsFile= 0",
>-      "RESULT Decode_errors_test: NormalStatsFile= 3"};
>+      "RESULT Max_repeated: NormalStatsFile= 1 ",
>+      "RESULT Max_skipped: NormalStatsFile= 0 ",
>+      "RESULT Total_skipped: NormalStatsFile= 0 ",
>+      "RESULT Decode_errors_reference: NormalStatsFile= 0 ",
>+      "RESULT Decode_errors_test: NormalStatsFile= 3 "};
>   VerifyLogOutput(log_filename, expected_out);
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_editing/frame_editing.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_editing/frame_editing.cc
>index 45af7fed418..fc281e1082a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_editing/frame_editing.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_editing/frame_editing.cc
>@@ -17,48 +17,53 @@
> #include "rtc_tools/simple_command_line_parser.h"
> 
> // A command-line tool to edit a YUV-video (I420 sub-sampled).
>-int main(int argc, char** argv) {
>+int main(int argc, char* argv[]) {
>   std::string program_name = argv[0];
>-  std::string usage = "Deletes a series of frames in a yuv file."
>-    " Only I420 is supported!\n"
>-    "Example usage:\n" + program_name +
>-    " --in_path=input.yuv --width=320 --height=240 --f=60 --interval=1 --l=120"
>-    " --out_path=edited_clip.yuv\n"
>-    "Command line flags:\n"
>-    "--in_path(string): Path and filename to the input file\n"
>-    "--width(int): Width in pixels of the frames in the input file."
>-    " Default: -1\n"
>-    "--height(int): Height in pixels of the frames in the input file."
>-    " Default: -1\n"
>-    "--f(int): First frame to process. Default: -1\n"
>-    "--l(int): Last frame to process.  Default: -1\n"
>-    "Frame numbering starts at 1. The set of frames to be processed includes "
>-    "the frame with the number <f> and <l>.\n"
>-    "--interval(int): Interval specifies with what ratio the number of frames "
>-    "should be increased or decreased with.\n"
>-    "If you set <interval> to a positive number, frames between <f> and <l> "
>-    "will be inserted <interval> times."
>-    " If you set <interval> to a negative number then the amount of frames "
>-    "between <f> and <l> will be decreased with a ratio of abs(interval)."
>-    " Set interval=-1 if every frame between <f> and <l> should be "
>-    "deleted. Set interval=-2 if every second frame should be deleted, and so "
>-    "on. Frame numbering between <f> and <l> starts with 1 and frames with"
>-    " number n where (n - 1)  % interval == 0 will be kept.\n"
>-    "Example 1:\n"
>-    "If one clip has 10 frames (1 to 10) and you specify <f>=4, <l>=7 and "
>-    "interval=2, then you will get a clip that contains frame "
>-    "1, 2, 3, 4, 4, 5, 5, 6, 6, 7, 7, 8, 9 and 10.\n"
>-    "Example 2:\n"
>-    "If you specify f=4, l=7 and interval=-1, then you will get a clip that"
>-    " contains frame 1, 2, 3, 8, 9 and 10.\n"
>-    "Example 3:\n"
>-    "If one clip has 10 frames (1 to 10), and you specify f=1, l=10 and "
>-    " interval=-4, then you will get a clip that contains frame "
>-    "1, 5 and 9.\n"
>-    "No interpolation is done when up-sampling."
>-    " Default: -1\n"
>-    "--out_path(string): The output file to which frames are written."
>-    " Default: output.yuv\n";
>+  std::string usage =
>+      "Deletes a series of frames in a yuv file."
>+      " Only I420 is supported!\n"
>+      "Example usage:\n" +
>+      program_name +
>+      " --in_path=input.yuv --width=320 --height=240 --f=60 --interval=1 "
>+      "--l=120"
>+      " --out_path=edited_clip.yuv\n"
>+      "Command line flags:\n"
>+      "--in_path(string): Path and filename to the input file\n"
>+      "--width(int): Width in pixels of the frames in the input file."
>+      " Default: -1\n"
>+      "--height(int): Height in pixels of the frames in the input file."
>+      " Default: -1\n"
>+      "--f(int): First frame to process. Default: -1\n"
>+      "--l(int): Last frame to process.  Default: -1\n"
>+      "Frame numbering starts at 1. The set of frames to be processed includes "
>+      "the frame with the number <f> and <l>.\n"
>+      "--interval(int): Interval specifies with what ratio the number of "
>+      "frames "
>+      "should be increased or decreased with.\n"
>+      "If you set <interval> to a positive number, frames between <f> and <l> "
>+      "will be inserted <interval> times."
>+      " If you set <interval> to a negative number then the amount of frames "
>+      "between <f> and <l> will be decreased with a ratio of abs(interval)."
>+      " Set interval=-1 if every frame between <f> and <l> should be "
>+      "deleted. Set interval=-2 if every second frame should be deleted, and "
>+      "so "
>+      "on. Frame numbering between <f> and <l> starts with 1 and frames with"
>+      " number n where (n - 1)  % interval == 0 will be kept.\n"
>+      "Example 1:\n"
>+      "If one clip has 10 frames (1 to 10) and you specify <f>=4, <l>=7 and "
>+      "interval=2, then you will get a clip that contains frame "
>+      "1, 2, 3, 4, 4, 5, 5, 6, 6, 7, 7, 8, 9 and 10.\n"
>+      "Example 2:\n"
>+      "If you specify f=4, l=7 and interval=-1, then you will get a clip that"
>+      " contains frame 1, 2, 3, 8, 9 and 10.\n"
>+      "Example 3:\n"
>+      "If one clip has 10 frames (1 to 10), and you specify f=1, l=10 and "
>+      " interval=-4, then you will get a clip that contains frame "
>+      "1, 5 and 9.\n"
>+      "No interpolation is done when up-sampling."
>+      " Default: -1\n"
>+      "--out_path(string): The output file to which frames are written."
>+      " Default: output.yuv\n";
> 
>   webrtc::test::CommandLineParser parser;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_editing/frame_editing_lib.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_editing/frame_editing_lib.cc
>index 5af01845678..d052cfc3fde 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_editing/frame_editing_lib.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_editing/frame_editing_lib.cc
>@@ -16,19 +16,22 @@
> 
> #include "common_video/libyuv/include/webrtc_libyuv.h"
> #include "rtc_tools/frame_editing/frame_editing_lib.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>-int EditFrames(const std::string& in_path, int width, int height,
>-               int first_frame_to_process, int interval,
>-               int last_frame_to_process, const std::string& out_path) {
>+int EditFrames(const std::string& in_path,
>+               int width,
>+               int height,
>+               int first_frame_to_process,
>+               int interval,
>+               int last_frame_to_process,
>+               const std::string& out_path) {
>   if (last_frame_to_process < first_frame_to_process) {
>     fprintf(stderr, "The set of frames to cut is empty! (l < f)\n");
>     return -10;
>   }
> 
>-  FILE* in_fid = fopen(in_path.c_str() , "rb");
>+  FILE* in_fid = fopen(in_path.c_str(), "rb");
>   if (!in_fid) {
>     fprintf(stderr, "Could not read input file: %s.\n", in_path.c_str());
>     return -11;
>@@ -51,8 +54,8 @@ int EditFrames(const std::string& in_path, int width, int height,
>   int num_frames_read_between = 0;
>   size_t num_bytes_read;
> 
>-  while ((num_bytes_read = fread(temp_buffer.get(), 1, frame_length, in_fid))
>-      == frame_length) {
>+  while ((num_bytes_read = fread(temp_buffer.get(), 1, frame_length, in_fid)) ==
>+         frame_length) {
>     num_frames_read++;
>     if ((num_frames_read < first_frame_to_process) ||
>         (last_frame_to_process < num_frames_read)) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_editing/frame_editing_lib.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_editing/frame_editing_lib.h
>index a805a117ed2..91eaa3bfec8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_editing/frame_editing_lib.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_editing/frame_editing_lib.h
>@@ -31,9 +31,13 @@ namespace webrtc {
> // 1, 2, 3, 4, 4, 5, 5, 6, 6, 7, 7, 8, 9 and 10.
> // No interpolation is done when up-sampling.
> 
>-int EditFrames(const std::string& in_path, int width, int height,
>-                int first_frame_to_process, int interval,
>-                int last_frame_to_process, const std::string& out_path);
>+int EditFrames(const std::string& in_path,
>+               int width,
>+               int height,
>+               int first_frame_to_process,
>+               int interval,
>+               int last_frame_to_process,
>+               const std::string& out_path);
> }  // namespace webrtc
> 
> #endif  // RTC_TOOLS_FRAME_EDITING_FRAME_EDITING_LIB_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_editing/frame_editing_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_editing/frame_editing_unittest.cc
>index 2cca3086385..e8261b5664e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_editing/frame_editing_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/frame_editing/frame_editing_unittest.cc
>@@ -63,14 +63,13 @@ class FrameEditingTest : public ::testing::Test {
>       if (!feof(ref_video_fid)) {
>         EXPECT_EQ(kFrameSize, num_bytes_read_);
>       }
>-      num_bytes_read_ = fread(test_buffer->get(), 1, kFrameSize,
>-                              test_video_fid);
>+      num_bytes_read_ =
>+          fread(test_buffer->get(), 1, kFrameSize, test_video_fid);
>       if (!feof(test_video_fid)) {
>         EXPECT_EQ(kFrameSize, num_bytes_read_);
>       }
>       if (!feof(test_video_fid) && !feof(test_video_fid)) {
>-        EXPECT_EQ(0, memcmp(ref_buffer->get(), test_buffer->get(),
>-                            kFrameSize));
>+        EXPECT_EQ(0, memcmp(ref_buffer->get(), test_buffer->get(), kFrameSize));
>       }
>     }
>     // There should not be anything left in either stream.
>@@ -91,26 +90,26 @@ TEST_F(FrameEditingTest, ValidInPath) {
>   const int kInterval = -1;
>   const int kLastFrameToProcess = 240;
> 
>-  int result = EditFrames(reference_video_, kWidth, kHeight,
>-                          kFirstFrameToProcess, kInterval, kLastFrameToProcess,
>-                          test_video_);
>+  int result =
>+      EditFrames(reference_video_, kWidth, kHeight, kFirstFrameToProcess,
>+                 kInterval, kLastFrameToProcess, test_video_);
>   EXPECT_EQ(0, result);
> 
>   for (int i = 1; i < kFirstFrameToProcess; ++i) {
>-    num_bytes_read_ = fread(original_buffer_.get(), 1, kFrameSize,
>-                            original_fid_);
>+    num_bytes_read_ =
>+        fread(original_buffer_.get(), 1, kFrameSize, original_fid_);
>     EXPECT_EQ(kFrameSize, num_bytes_read_);
> 
>     num_bytes_read_ = fread(edited_buffer_.get(), 1, kFrameSize, edited_fid_);
>     EXPECT_EQ(kFrameSize, num_bytes_read_);
> 
>-    EXPECT_EQ(0, memcmp(original_buffer_.get(), edited_buffer_.get(),
>-                        kFrameSize));
>+    EXPECT_EQ(0,
>+              memcmp(original_buffer_.get(), edited_buffer_.get(), kFrameSize));
>   }
>   // Do not compare the frames that have been cut.
>   for (int i = kFirstFrameToProcess; i <= kLastFrameToProcess; ++i) {
>-    num_bytes_read_ = fread(original_buffer_.get(), 1, kFrameSize,
>-                            original_fid_);
>+    num_bytes_read_ =
>+        fread(original_buffer_.get(), 1, kFrameSize, original_fid_);
>     EXPECT_EQ(kFrameSize, num_bytes_read_);
>   }
>   CompareToTheEnd(edited_fid_, original_fid_, &original_buffer_,
>@@ -122,9 +121,9 @@ TEST_F(FrameEditingTest, EmptySetToCut) {
>   const int kInterval = -1;
>   const int kLastFrameToProcess = 1;
> 
>-  int result = EditFrames(reference_video_, kWidth, kHeight,
>-                          kFirstFrameToProcess, kInterval, kLastFrameToProcess,
>-                          test_video_);
>+  int result =
>+      EditFrames(reference_video_, kWidth, kHeight, kFirstFrameToProcess,
>+                 kInterval, kLastFrameToProcess, test_video_);
>   EXPECT_EQ(-10, result);
> }
> 
>@@ -145,9 +144,9 @@ TEST_F(FrameEditingTest, DeletingEverySecondFrame) {
>   const int kInterval = -2;
>   const int kLastFrameToProcess = 10000;
>   // Set kLastFrameToProcess to a large value so that all frame are processed.
>-  int result = EditFrames(reference_video_, kWidth, kHeight,
>-                          kFirstFrameToProcess, kInterval, kLastFrameToProcess,
>-                          test_video_);
>+  int result =
>+      EditFrames(reference_video_, kWidth, kHeight, kFirstFrameToProcess,
>+                 kInterval, kLastFrameToProcess, test_video_);
>   EXPECT_EQ(0, result);
> 
>   while (!feof(original_fid_) && !feof(edited_fid_)) {
>@@ -162,14 +161,13 @@ TEST_F(FrameEditingTest, DeletingEverySecondFrame) {
>     // every second frame.
>     // num_frames_read_ - 1 because we have deleted frame number 2, 4 , 6 etc.
>     if ((num_frames_read_ - 1) % kInterval == -1) {
>-      num_bytes_read_ = fread(edited_buffer_.get(), 1, kFrameSize,
>-                              edited_fid_);
>+      num_bytes_read_ = fread(edited_buffer_.get(), 1, kFrameSize, edited_fid_);
>       if (!feof(edited_fid_)) {
>         EXPECT_EQ(kFrameSize, num_bytes_read_);
>       }
>       if (!feof(original_fid_) && !feof(edited_fid_)) {
>-        EXPECT_EQ(0, memcmp(original_buffer_.get(),
>-                            edited_buffer_.get(), kFrameSize));
>+        EXPECT_EQ(0, memcmp(original_buffer_.get(), edited_buffer_.get(),
>+                            kFrameSize));
>       }
>     }
>   }
>@@ -180,33 +178,32 @@ TEST_F(FrameEditingTest, RepeatFrames) {
>   const int kInterval = 2;
>   const int kLastFrameToProcess = 240;
> 
>-  int result = EditFrames(reference_video_, kWidth, kHeight,
>-                          kFirstFrameToProcess, kInterval, kLastFrameToProcess,
>-                          test_video_);
>+  int result =
>+      EditFrames(reference_video_, kWidth, kHeight, kFirstFrameToProcess,
>+                 kInterval, kLastFrameToProcess, test_video_);
>   EXPECT_EQ(0, result);
> 
>   for (int i = 1; i < kFirstFrameToProcess; ++i) {
>-    num_bytes_read_ = fread(original_buffer_.get(), 1, kFrameSize,
>-                            original_fid_);
>+    num_bytes_read_ =
>+        fread(original_buffer_.get(), 1, kFrameSize, original_fid_);
>     EXPECT_EQ(kFrameSize, num_bytes_read_);
> 
>     num_bytes_read_ = fread(edited_buffer_.get(), 1, kFrameSize, edited_fid_);
>     EXPECT_EQ(kFrameSize, num_bytes_read_);
> 
>-    EXPECT_EQ(0, memcmp(original_buffer_.get(), edited_buffer_.get(),
>-                        kFrameSize));
>+    EXPECT_EQ(0,
>+              memcmp(original_buffer_.get(), edited_buffer_.get(), kFrameSize));
>   }
>   // Do not compare the frames that have been repeated.
>   for (int i = kFirstFrameToProcess; i <= kLastFrameToProcess; ++i) {
>-    num_bytes_read_ = fread(original_buffer_.get(), 1, kFrameSize,
>-                            original_fid_);
>+    num_bytes_read_ =
>+        fread(original_buffer_.get(), 1, kFrameSize, original_fid_);
>     EXPECT_EQ(kFrameSize, num_bytes_read_);
>     for (int i = 1; i <= kInterval; ++i) {
>-      num_bytes_read_ = fread(edited_buffer_.get(), 1, kFrameSize,
>-                              edited_fid_);
>+      num_bytes_read_ = fread(edited_buffer_.get(), 1, kFrameSize, edited_fid_);
>       EXPECT_EQ(kFrameSize, num_bytes_read_);
>-      EXPECT_EQ(0, memcmp(original_buffer_.get(), edited_buffer_.get(),
>-                          kFrameSize));
>+      EXPECT_EQ(
>+          0, memcmp(original_buffer_.get(), edited_buffer_.get(), kFrameSize));
>     }
>   }
>   CompareToTheEnd(edited_fid_, original_fid_, &original_buffer_,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/header_usage.sh b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/header_usage.sh
>old mode 100644
>new mode 100755
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/loopback_test/record-test.sh b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/loopback_test/record-test.sh
>old mode 100644
>new mode 100755
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/loopback_test/run-server.sh b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/loopback_test/run-server.sh
>old mode 100644
>new mode 100755
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/BUILD.gn
>index cfc116c3fb0..cf6b9eb4520 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/BUILD.gn
>@@ -41,12 +41,13 @@ if (rtc_enable_protobuf) {
>     deps = [
>       ":network_tester_config_proto",
>       ":network_tester_packet_proto",
>-      "../../api:optional",
>       "../../p2p",
>+      "../../rtc_base:checks",
>       "../../rtc_base:protobuf_utils",
>       "../../rtc_base:rtc_base_approved",
>       "../../rtc_base:rtc_task_queue",
>       "../../rtc_base:sequenced_task_checker",
>+      "//third_party/abseil-cpp/absl/types:optional",
>     ]
> 
>     if (!build_with_chromium && is_clang) {
>@@ -80,6 +81,7 @@ if (rtc_enable_protobuf) {
>     deps = [
>       ":network_tester",
>       "../../rtc_base:rtc_base_tests_utils",
>+      "../../test:fileutils",
>       "../../test:test_support",
>       "//testing/gtest",
>     ]
>@@ -88,10 +90,7 @@ if (rtc_enable_protobuf) {
>       deps += [ ":network_tester_unittests_bundle_data" ]
>     }
> 
>-    defines = [
>-      "GTEST_RELATIVE_PATH",
>-      "WEBRTC_NETWORK_TESTER_TEST_ENABLED",
>-    ]
>+    defines = [ "WEBRTC_NETWORK_TESTER_TEST_ENABLED" ]
> 
>     data = network_tester_unittests_resources
> 
>@@ -127,7 +126,6 @@ if (is_android) {
>       ":NetworkTesterMobile_javalib",
>       ":NetworkTesterMobile_resources",
>       "../../rtc_base:base_java",
>-      "//base:base_java",
>     ]
> 
>     shared_libraries = [ "../../rtc_tools/network_tester:network_tester_so" ]
>@@ -135,7 +133,7 @@ if (is_android) {
> 
>   android_library("NetworkTesterMobile_javalib") {
>     testonly = true
>-    android_manifest = "androidapp/AndroidManifest.xml"
>+    android_manifest_for_lint = "androidapp/AndroidManifest.xml"
> 
>     java_files = [
>       "androidapp/src/com/google/media/networktester/MainActivity.java",
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/androidapp/AndroidManifest.xml b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/androidapp/AndroidManifest.xml
>old mode 100644
>new mode 100755
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/androidapp/res/layout/activity_main.xml b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/androidapp/res/layout/activity_main.xml
>old mode 100644
>new mode 100755
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/network_tester/androidapp/res/mipmap-hdpi/ic_launcher.png b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/androidapp/res/mipmap-hdpi/ic_launcher.png
>similarity index 100%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/tools/network_tester/androidapp/res/mipmap-hdpi/ic_launcher.png
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/androidapp/res/mipmap-hdpi/ic_launcher.png
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/network_tester/androidapp/res/mipmap-mdpi/ic_launcher.png b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/androidapp/res/mipmap-mdpi/ic_launcher.png
>similarity index 100%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/tools/network_tester/androidapp/res/mipmap-mdpi/ic_launcher.png
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/androidapp/res/mipmap-mdpi/ic_launcher.png
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/network_tester/androidapp/res/mipmap-xhdpi/ic_launcher.png b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/androidapp/res/mipmap-xhdpi/ic_launcher.png
>similarity index 100%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/tools/network_tester/androidapp/res/mipmap-xhdpi/ic_launcher.png
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/androidapp/res/mipmap-xhdpi/ic_launcher.png
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/network_tester/androidapp/res/mipmap-xxhdpi/ic_launcher.png b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/androidapp/res/mipmap-xxhdpi/ic_launcher.png
>similarity index 100%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/tools/network_tester/androidapp/res/mipmap-xxhdpi/ic_launcher.png
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/androidapp/res/mipmap-xxhdpi/ic_launcher.png
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/network_tester/androidapp/res/mipmap-xxxhdpi/ic_launcher.png b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/androidapp/res/mipmap-xxxhdpi/ic_launcher.png
>similarity index 100%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/tools/network_tester/androidapp/res/mipmap-xxxhdpi/ic_launcher.png
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/androidapp/res/mipmap-xxxhdpi/ic_launcher.png
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/androidapp/res/values-v17/styles.xml b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/androidapp/res/values-v17/styles.xml
>old mode 100644
>new mode 100755
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/androidapp/res/values-w820dp/dimens.xml b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/androidapp/res/values-w820dp/dimens.xml
>old mode 100644
>new mode 100755
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/androidapp/res/values/colors.xml b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/androidapp/res/values/colors.xml
>old mode 100644
>new mode 100755
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/androidapp/res/values/dimens.xml b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/androidapp/res/values/dimens.xml
>old mode 100644
>new mode 100755
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/androidapp/res/values/strings.xml b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/androidapp/res/values/strings.xml
>old mode 100644
>new mode 100755
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/config_reader.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/config_reader.cc
>index 7e01a6577e2..8ef88c7eda2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/config_reader.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/config_reader.cc
>@@ -12,6 +12,8 @@
> #include <string>
> #include <vector>
> 
>+#include "rtc_base/checks.h"
>+
> namespace webrtc {
> 
> ConfigReader::ConfigReader(const std::string& config_file_path)
>@@ -29,10 +31,10 @@ ConfigReader::ConfigReader(const std::string& config_file_path)
> 
> ConfigReader::~ConfigReader() = default;
> 
>-rtc::Optional<ConfigReader::Config> ConfigReader::GetNextConfig() {
>+absl::optional<ConfigReader::Config> ConfigReader::GetNextConfig() {
> #ifdef WEBRTC_NETWORK_TESTER_PROTO
>   if (proto_config_index_ >= proto_all_configs_.configs_size())
>-    return rtc::Optional<Config>();
>+    return absl::nullopt;
>   auto proto_config = proto_all_configs_.configs(proto_config_index_++);
>   RTC_DCHECK(proto_config.has_packet_send_interval_ms());
>   RTC_DCHECK(proto_config.has_packet_size());
>@@ -41,9 +43,9 @@ rtc::Optional<ConfigReader::Config> ConfigReader::GetNextConfig() {
>   config.packet_send_interval_ms = proto_config.packet_send_interval_ms();
>   config.packet_size = proto_config.packet_size();
>   config.execution_time_ms = proto_config.execution_time_ms();
>-  return rtc::Optional<Config>(config);
>+  return config;
> #else
>-  return rtc::Optional<Config>();
>+  return absl::nullopt;
> #endif  //  WEBRTC_NETWORK_TESTER_PROTO
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/config_reader.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/config_reader.h
>index e03317fe430..deee245eefb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/config_reader.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/config_reader.h
>@@ -14,7 +14,7 @@
> #include <fstream>
> #include <string>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> #include "rtc_base/constructormagic.h"
> 
> #include "rtc_base/ignore_wundef.h"
>@@ -40,7 +40,7 @@ class ConfigReader {
>   explicit ConfigReader(const std::string& config_file_path);
>   ~ConfigReader();
> 
>-  rtc::Optional<Config> GetNextConfig();
>+  absl::optional<Config> GetNextConfig();
> 
>  private:
>   NetworkTesterAllConfigs proto_all_configs_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/network_tester_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/network_tester_unittest.cc
>index b60a712746b..c4c9e0c9a2b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/network_tester_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/network_tester_unittest.cc
>@@ -8,7 +8,6 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-
> #ifdef WEBRTC_NETWORK_TESTER_TEST_ENABLED
> 
> #include "rtc_tools/network_tester/test_controller.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/packet_sender.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/packet_sender.cc
>index 3b252524dd9..2c38b2fc4de 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/packet_sender.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/packet_sender.cc
>@@ -115,7 +115,7 @@ void PacketSender::SendPacket() {
>   packet.set_type(NetworkTesterPacket::TEST_DATA);
>   packet.set_sequence_number(sequence_number_++);
>   packet.set_send_timestamp(rtc::TimeMicros());
>-  test_controller_->SendData(packet, rtc::Optional<size_t>(packet_size_));
>+  test_controller_->SendData(packet, packet_size_);
> }
> 
> int64_t PacketSender::GetSendIntervalMs() const {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/packet_sender.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/packet_sender.h
>index b2f3901e1e8..da3eb293b14 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/packet_sender.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/packet_sender.h
>@@ -49,10 +49,10 @@ class PacketSender {
> 
>  private:
>   rtc::SequencedTaskChecker worker_queue_checker_;
>-  size_t packet_size_ RTC_ACCESS_ON(worker_queue_checker_);
>-  int64_t send_interval_ms_ RTC_ACCESS_ON(worker_queue_checker_);
>-  int64_t sequence_number_ RTC_ACCESS_ON(worker_queue_checker_);
>-  bool sending_ RTC_ACCESS_ON(worker_queue_checker_);
>+  size_t packet_size_ RTC_GUARDED_BY(worker_queue_checker_);
>+  int64_t send_interval_ms_ RTC_GUARDED_BY(worker_queue_checker_);
>+  int64_t sequence_number_ RTC_GUARDED_BY(worker_queue_checker_);
>+  bool sending_ RTC_GUARDED_BY(worker_queue_checker_);
>   const std::string config_file_path_;
>   TestController* const test_controller_;
>   rtc::TaskQueue worker_queue_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/parse_packet_log.py b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/parse_packet_log.py
>old mode 100644
>new mode 100755
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/server.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/server.cc
>index 6edd9f958f6..25ca5a1d378 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/server.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/server.cc
>@@ -10,10 +10,11 @@
> 
> #include "rtc_tools/network_tester/test_controller.h"
> 
>-int main() {
>+int main(int /*argn*/, char* /*argv*/ []) {
>   webrtc::TestController server(9090, 9090, "server_config.dat",
>                                 "server_packet_log.dat");
>   while (!server.IsTestDone()) {
>     server.Run();
>   }
>+  return 0;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/test_controller.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/test_controller.cc
>index 908e6954ced..e5bd92efa93 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/test_controller.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/test_controller.cc
>@@ -37,7 +37,7 @@ void TestController::SendConnectTo(const std::string& hostname, int port) {
>   udp_transport_->SetRemoteAddress(rtc::SocketAddress(hostname, port));
>   NetworkTesterPacket packet;
>   packet.set_type(NetworkTesterPacket::HAND_SHAKING);
>-  SendData(packet, rtc::Optional<size_t>());
>+  SendData(packet, absl::nullopt);
>   rtc::CritScope scoped_lock(&local_test_done_lock_);
>   local_test_done_ = false;
>   remote_test_done_ = false;
>@@ -49,9 +49,9 @@ void TestController::Run() {
> }
> 
> void TestController::SendData(const NetworkTesterPacket& packet,
>-                              rtc::Optional<size_t> data_size) {
>+                              absl::optional<size_t> data_size) {
>   // Can be call from packet_sender or from test_controller thread.
>-  size_t packet_size = packet.ByteSize();
>+  size_t packet_size = packet.ByteSizeLong();
>   send_data_[0] = packet_size;
>   packet_size++;
>   packet.SerializeToArray(&send_data_[1], std::numeric_limits<char>::max());
>@@ -65,7 +65,7 @@ void TestController::OnTestDone() {
>   RTC_DCHECK_CALLED_SEQUENTIALLY(&packet_sender_checker_);
>   NetworkTesterPacket packet;
>   packet.set_type(NetworkTesterPacket::TEST_DONE);
>-  SendData(packet, rtc::Optional<size_t>());
>+  SendData(packet, absl::nullopt);
>   rtc::CritScope scoped_lock(&local_test_done_lock_);
>   local_test_done_ = true;
> }
>@@ -92,7 +92,7 @@ void TestController::OnReadPacket(rtc::AsyncPacketSocket* socket,
>       NetworkTesterPacket packet;
>       packet.set_type(NetworkTesterPacket::TEST_START);
>       udp_transport_->SetRemoteAddress(remote_addr);
>-      SendData(packet, rtc::Optional<size_t>());
>+      SendData(packet, absl::nullopt);
>       packet_sender_.reset(new PacketSender(this, config_file_path_));
>       packet_sender_->StartSending();
>       rtc::CritScope scoped_lock(&local_test_done_lock_);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/test_controller.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/test_controller.h
>index 4c9ede18aee..cf65e174595 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/test_controller.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/network_tester/test_controller.h
>@@ -49,7 +49,7 @@ class TestController : public sigslot::has_slots<> {
>   void SendConnectTo(const std::string& hostname, int port);
> 
>   void SendData(const NetworkTesterPacket& packet,
>-                rtc::Optional<size_t> data_size);
>+                absl::optional<size_t> data_size);
> 
>   void OnTestDone();
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/psnr_ssim_analyzer/psnr_ssim_analyzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/psnr_ssim_analyzer/psnr_ssim_analyzer.cc
>index ac1c8d997a0..dc63aca9d75 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/psnr_ssim_analyzer/psnr_ssim_analyzer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/psnr_ssim_analyzer/psnr_ssim_analyzer.cc
>@@ -21,8 +21,11 @@
> 
> #define MAX_NUM_FRAMES_PER_FILE INT_MAX
> 
>-void CompareFiles(const char* reference_file_name, const char* test_file_name,
>-                  const char* results_file_name, int width, int height) {
>+void CompareFiles(const char* reference_file_name,
>+                  const char* test_file_name,
>+                  const char* results_file_name,
>+                  int width,
>+                  int height) {
>   // Check if the reference_file_name ends with "y4m".
>   bool y4m_mode = false;
>   if (std::string(reference_file_name).find("y4m") != std::string::npos) {
>@@ -39,13 +42,15 @@ void CompareFiles(const char* reference_file_name, const char* test_file_name,
> 
>   bool read_result = true;
>   for (int frame_counter = 0; frame_counter < MAX_NUM_FRAMES_PER_FILE;
>-      ++frame_counter) {
>-    read_result &= (y4m_mode) ? webrtc::test::ExtractFrameFromY4mFile(
>-        reference_file_name, width, height, frame_counter, ref_frame):
>-        webrtc::test::ExtractFrameFromYuvFile(reference_file_name, width,
>-                                              height, frame_counter, ref_frame);
>-    read_result &=  webrtc::test::ExtractFrameFromYuvFile(test_file_name, width,
>-        height, frame_counter, test_frame);
>+       ++frame_counter) {
>+    read_result &=
>+        (y4m_mode)
>+            ? webrtc::test::ExtractFrameFromY4mFile(
>+                  reference_file_name, width, height, frame_counter, ref_frame)
>+            : webrtc::test::ExtractFrameFromYuvFile(
>+                  reference_file_name, width, height, frame_counter, ref_frame);
>+    read_result &= webrtc::test::ExtractFrameFromYuvFile(
>+        test_file_name, width, height, frame_counter, test_frame);
> 
>     if (!read_result)
>       break;
>@@ -80,11 +85,14 @@ void CompareFiles(const char* reference_file_name, const char* test_file_name,
>  * --results_file=<name_of_file> --width=<width_of_frames>
>  * --height=<height_of_frames>
>  */
>-int main(int argc, char** argv) {
>+int main(int argc, char* argv[]) {
>   std::string program_name = argv[0];
>-  std::string usage = "Runs PSNR and SSIM on two I420 videos and write the"
>+  std::string usage =
>+      "Runs PSNR and SSIM on two I420 videos and write the"
>       "results in a file.\n"
>-      "Example usage:\n" + program_name + " --reference_file=ref.yuv "
>+      "Example usage:\n" +
>+      program_name +
>+      " --reference_file=ref.yuv "
>       "--test_file=test.yuv --results_file=results.txt --width=320 "
>       "--height=240\n"
>       "Command line flags:\n"
>@@ -130,4 +138,5 @@ int main(int argc, char** argv) {
>   CompareFiles(parser.GetFlag("reference_file").c_str(),
>                parser.GetFlag("test_file").c_str(),
>                parser.GetFlag("results_file").c_str(), width, height);
>+  return 0;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/py_event_log_analyzer/misc_test.py b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/py_event_log_analyzer/misc_test.py
>old mode 100644
>new mode 100755
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/py_event_log_analyzer/pb_parse.py b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/py_event_log_analyzer/pb_parse.py
>index f4c8a336e48..bc835ae0232 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/py_event_log_analyzer/pb_parse.py
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/py_event_log_analyzer/pb_parse.py
>@@ -10,7 +10,7 @@
> 
> from __future__ import division
> import struct
>-import pyproto.webrtc.logging.rtc_event_log.rtc_event_log_pb2 as rtc_pb
>+import pyproto.logging.rtc_event_log.rtc_event_log_pb2 as rtc_pb
> 
> 
> class DataPoint(object):
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/py_event_log_analyzer/rtp_analyzer.sh b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/py_event_log_analyzer/rtp_analyzer.sh
>old mode 100644
>new mode 100755
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/py_event_log_analyzer/rtp_analyzer_test.py b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/py_event_log_analyzer/rtp_analyzer_test.py
>old mode 100644
>new mode 100755
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/sanitizers_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/sanitizers_unittest.cc
>index 4cdf5fd56f2..077cf7e6995 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/sanitizers_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/sanitizers_unittest.cc
>@@ -12,9 +12,9 @@
> #include <stdio.h>
> #include <random>
> 
>+#include "absl/memory/memory.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/nullsocketserver.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/thread.h"
> #include "test/gtest.h"
> 
>@@ -39,7 +39,7 @@ TEST(SanitizersDeathTest, MemorySanitizer) {
> 
> #if defined(ADDRESS_SANITIZER)
> void HeapUseAfterFree() {
>-  char *buf = new char[2];
>+  char* buf = new char[2];
>   delete[] buf;
>   buf[0] = buf[1];
> }
>@@ -61,8 +61,7 @@ struct Base {
>   virtual void f() {}
>   virtual ~Base() {}
> };
>-struct Derived : public Base {
>-};
>+struct Derived : public Base {};
> 
> void InvalidVptr() {
>   Base b;
>@@ -71,7 +70,12 @@ void InvalidVptr() {
> }
> 
> TEST(SanitizersDeathTest, UndefinedSanitizer) {
>-  EXPECT_DEATH({ SignedIntegerOverflow(); InvalidVptr(); }, "runtime error");
>+  EXPECT_DEATH(
>+      {
>+        SignedIntegerOverflow();
>+        InvalidVptr();
>+      },
>+      "runtime error");
> }
> #endif
> 
>@@ -79,8 +83,7 @@ TEST(SanitizersDeathTest, UndefinedSanitizer) {
> class IncrementThread : public Thread {
>  public:
>   explicit IncrementThread(int* value)
>-      : Thread(rtc::MakeUnique<NullSocketServer>()),
>-        value_(value) {}
>+      : Thread(absl::make_unique<NullSocketServer>()), value_(value) {}
> 
>   void Run() override {
>     ++*value_;
>@@ -88,9 +91,7 @@ class IncrementThread : public Thread {
>   }
> 
>   // Un-protect Thread::Join for the test.
>-  void Join() {
>-    Thread::Join();
>-  }
>+  void Join() { Thread::Join(); }
> 
>  private:
>   int* value_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/simple_command_line_parser.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/simple_command_line_parser.cc
>index b20ba6481f6..c34aa905560 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/simple_command_line_parser.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/simple_command_line_parser.cc
>@@ -22,7 +22,7 @@ CommandLineParser::CommandLineParser() {}
> CommandLineParser::~CommandLineParser() {}
> 
> void CommandLineParser::Init(int argc, char** argv) {
>-  args_ = std::vector<std::string> (argv + 1, argv + argc);
>+  args_ = std::vector<std::string>(argv + 1, argv + argc);
> }
> 
> bool CommandLineParser::IsStandaloneFlag(std::string flag) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/simple_command_line_parser_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/simple_command_line_parser_unittest.cc
>index d98868aa2e2..a94f9921e96 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/simple_command_line_parser_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/simple_command_line_parser_unittest.cc
>@@ -8,8 +8,8 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "test/gtest.h"
> #include "rtc_tools/simple_command_line_parser.h"
>+#include "test/gtest.h"
> 
> namespace webrtc {
> namespace test {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/testing/README.md b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/testing/README.md
>index 528b8703452..f60c65960aa 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/testing/README.md
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/testing/README.md
>@@ -17,8 +17,14 @@ Updating prebuilt_apprtc.zip:
>     - `ln -s "$(pwd)/src/collider" src/src`
>     - `GOPATH="$(pwd)/src" go get -d collidermain`
>     - `rm src/src`
>-- Remove unneeded files:
>-    - `rm -rf .git node_modules browsers`
>+- Install additional components:
>+    - `python temp/google-cloud-sdk/bin/dev_appserver.py out/app_engine`
>+        - Stop it and run the suggestion: `./temp/google-cloud-sdk/bin/gcloud
>+          components install app-engine-python-extras`
>+- Remove largest unneeded files:
>+    - `rm -rf .git node_modules browsers
>+      temp/google-cloud-sdk/.install/.{backup,download}
>+      temp/google-cloud-sdk/platform/google_appengine/lib/django-*`
> - `zip -r prebuilt_apprtc.zip apprtc/`
> - `mv prebuilt_apprtc.zip webrtc/src/rtc_tools/testing/prebuilt_apprtc.zip`
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/testing/build_apprtc.py b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/testing/build_apprtc.py
>old mode 100644
>new mode 100755
>index a390c94f0ec..367a2602d56
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/testing/build_apprtc.py
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/testing/build_apprtc.py
>@@ -16,6 +16,7 @@ This script needs to know the path to the 'src' directory in apprtc, the
> root directory of 'go' and the output_dir.
> """
> 
>+import fileinput
> import os
> import shutil
> import subprocess
>@@ -24,7 +25,15 @@ import sys
> import utils
> 
> 
>-USAGE_STR = "Usage: {} <apprtc_src_dir> <go_dir> <output_dir>"
>+USAGE_STR = "Usage: {} <apprtc_dir> <go_dir> <output_dir>"
>+
>+
>+def _ConfigureApprtcServerToDeveloperMode(app_yaml_path):
>+  for line in fileinput.input(app_yaml_path, inplace=True):
>+    # We can't click past these in browser-based tests, so disable them.
>+    line = line.replace('BYPASS_JOIN_CONFIRMATION: false',
>+                        'BYPASS_JOIN_CONFIRMATION: true')
>+    sys.stdout.write(line)
> 
> 
> def main(argv):
>@@ -35,21 +44,22 @@ def main(argv):
>   go_root_dir = os.path.abspath(argv[2])
>   golang_workspace = os.path.abspath(argv[3])
> 
>-  utils.RemoveDirectory(golang_workspace)
>+  app_yaml_path = os.path.join(apprtc_dir, 'out', 'app_engine', 'app.yaml')
>+  _ConfigureApprtcServerToDeveloperMode(app_yaml_path)
> 
>-  golang_workspace_src = os.path.join(golang_workspace, 'src')
>-
>-  collider_dir = os.path.join(apprtc_dir, 'collider')
>-  shutil.copytree(collider_dir, golang_workspace_src)
>+  utils.RemoveDirectory(golang_workspace)
> 
>-  golang_binary = 'go%s' % ('.exe' if utils.GetPlatform() == 'win' else '')
>-  golang_path = os.path.join(go_root_dir, 'bin', golang_binary)
>+  collider_dir = os.path.join(apprtc_dir, 'src', 'collider')
>+  shutil.copytree(collider_dir, os.path.join(golang_workspace, 'src'))
> 
>+  golang_path = os.path.join(go_root_dir, 'bin',
>+                             'go' + utils.GetExecutableExtension())
>   golang_env = os.environ.copy()
>   golang_env['GOROOT'] = go_root_dir
>   golang_env['GOPATH'] = golang_workspace
>-  collider_exec = os.path.join(golang_workspace, 'collidermain')
>-  subprocess.check_call([golang_path, 'build', '-o', collider_exec,
>+  collider_out = os.path.join(golang_workspace,
>+                              'collidermain' + utils.GetExecutableExtension())
>+  subprocess.check_call([golang_path, 'build', '-o', collider_out,
>                          'collidermain'], env=golang_env)
> 
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/testing/download_apprtc.py b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/testing/download_apprtc.py
>old mode 100644
>new mode 100755
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/testing/prebuilt_apprtc.zip.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/testing/prebuilt_apprtc.zip.sha1
>index 95b41c71587..55fda33d812 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/testing/prebuilt_apprtc.zip.sha1
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/testing/prebuilt_apprtc.zip.sha1
>@@ -1 +1 @@
>-d8d95f55129e0c1a4e0f5d0d3acbf7163395354c
>\ No newline at end of file
>+4a1853f16014b64446d9792088c1088abbe11083
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/testing/setup_apprtc.py b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/testing/setup_apprtc.py
>old mode 100644
>new mode 100755
>index 697d5031d8b..2b463e004dd
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/testing/setup_apprtc.py
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/testing/setup_apprtc.py
>@@ -30,14 +30,15 @@ def main(argv):
>   output_dir = os.path.abspath(argv[1])
> 
>   download_apprtc_path = os.path.join(SCRIPT_DIR, 'download_apprtc.py')
>-  utils.RunSubprocessWithRetry([download_apprtc_path, output_dir])
>+  utils.RunSubprocessWithRetry([sys.executable, download_apprtc_path,
>+                                output_dir])
> 
>   build_apprtc_path = os.path.join(SCRIPT_DIR, 'build_apprtc.py')
>-  apprtc_src_dir = os.path.join(output_dir, 'apprtc', 'src')
>+  apprtc_dir = os.path.join(output_dir, 'apprtc')
>   go_dir = os.path.join(output_dir, 'go')
>   collider_dir = os.path.join(output_dir, 'collider')
>-  utils.RunSubprocessWithRetry([build_apprtc_path, apprtc_src_dir,
>-                                go_dir, collider_dir])
>+  utils.RunSubprocessWithRetry([sys.executable, build_apprtc_path,
>+                                apprtc_dir, go_dir, collider_dir])
> 
> 
> if __name__ == '__main__':
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/testing/utils.py b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/testing/utils.py
>old mode 100644
>new mode 100755
>index 3b82fc88679..e0f679d6b51
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/testing/utils.py
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/testing/utils.py
>@@ -118,3 +118,7 @@ def GetPlatform():
>   if sys.platform.startswith('darwin'):
>     return 'mac'
>   raise Exception("Can't run on platform %s." % sys.platform)
>+
>+
>+def GetExecutableExtension():
>+  return '.exe' if GetPlatform() == 'win' else ''
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/unpack_aecdump/unpack.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/unpack_aecdump/unpack.cc
>new file mode 100644
>index 00000000000..44f8346af01
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/unpack_aecdump/unpack.cc
>@@ -0,0 +1,376 @@
>+/*
>+ *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+// Commandline tool to unpack audioproc debug files.
>+//
>+// The debug files are dumped as protobuf blobs. For analysis, it's necessary
>+// to unpack the file into its component parts: audio and other data.
>+
>+#include <stdio.h>
>+
>+#include <memory>
>+
>+#include "modules/audio_processing/test/protobuf_utils.h"
>+#include "modules/audio_processing/test/test_utils.h"
>+#include "rtc_base/flags.h"
>+#include "rtc_base/format_macros.h"
>+#include "rtc_base/ignore_wundef.h"
>+
>+RTC_PUSH_IGNORING_WUNDEF()
>+#include "modules/audio_processing/debug.pb.h"
>+RTC_POP_IGNORING_WUNDEF()
>+
>+// TODO(andrew): unpack more of the data.
>+DEFINE_string(input_file, "input", "The name of the input stream file.");
>+DEFINE_string(output_file,
>+              "ref_out",
>+              "The name of the reference output stream file.");
>+DEFINE_string(reverse_file,
>+              "reverse",
>+              "The name of the reverse input stream file.");
>+DEFINE_string(delay_file, "delay.int32", "The name of the delay file.");
>+DEFINE_string(drift_file, "drift.int32", "The name of the drift file.");
>+DEFINE_string(level_file, "level.int32", "The name of the level file.");
>+DEFINE_string(keypress_file, "keypress.bool", "The name of the keypress file.");
>+DEFINE_string(callorder_file,
>+              "callorder",
>+              "The name of the render/capture call order file.");
>+DEFINE_string(settings_file, "settings.txt", "The name of the settings file.");
>+DEFINE_bool(full, false, "Unpack the full set of files (normally not needed).");
>+DEFINE_bool(raw, false, "Write raw data instead of a WAV file.");
>+DEFINE_bool(text,
>+            false,
>+            "Write non-audio files as text files instead of binary files.");
>+DEFINE_bool(help, false, "Print this message.");
>+
>+#define PRINT_CONFIG(field_name)                                         \
>+  if (msg.has_##field_name()) {                                          \
>+    fprintf(settings_file, "  " #field_name ": %d\n", msg.field_name()); \
>+  }
>+
>+#define PRINT_CONFIG_FLOAT(field_name)                                   \
>+  if (msg.has_##field_name()) {                                          \
>+    fprintf(settings_file, "  " #field_name ": %f\n", msg.field_name()); \
>+  }
>+
>+namespace webrtc {
>+
>+using audioproc::Event;
>+using audioproc::ReverseStream;
>+using audioproc::Stream;
>+using audioproc::Init;
>+
>+namespace {
>+
>+void WriteData(const void* data,
>+               size_t size,
>+               FILE* file,
>+               const std::string& filename) {
>+  if (fwrite(data, size, 1, file) != 1) {
>+    printf("Error when writing to %s\n", filename.c_str());
>+    exit(1);
>+  }
>+}
>+
>+void WriteCallOrderData(const bool render_call,
>+                        FILE* file,
>+                        const std::string& filename) {
>+  const char call_type = render_call ? 'r' : 'c';
>+  WriteData(&call_type, sizeof(call_type), file, filename.c_str());
>+}
>+
>+}  // namespace
>+
>+int do_main(int argc, char* argv[]) {
>+  std::string program_name = argv[0];
>+  std::string usage =
>+      "Commandline tool to unpack audioproc debug files.\n"
>+      "Example usage:\n" +
>+      program_name + " debug_dump.pb\n";
>+
>+  if (rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true) || FLAG_help ||
>+      argc < 2) {
>+    printf("%s", usage.c_str());
>+    if (FLAG_help) {
>+      rtc::FlagList::Print(nullptr, false);
>+      return 0;
>+    }
>+    return 1;
>+  }
>+
>+  FILE* debug_file = OpenFile(argv[1], "rb");
>+
>+  Event event_msg;
>+  int frame_count = 0;
>+  size_t reverse_samples_per_channel = 0;
>+  size_t input_samples_per_channel = 0;
>+  size_t output_samples_per_channel = 0;
>+  size_t num_reverse_channels = 0;
>+  size_t num_input_channels = 0;
>+  size_t num_output_channels = 0;
>+  std::unique_ptr<WavWriter> reverse_wav_file;
>+  std::unique_ptr<WavWriter> input_wav_file;
>+  std::unique_ptr<WavWriter> output_wav_file;
>+  std::unique_ptr<RawFile> reverse_raw_file;
>+  std::unique_ptr<RawFile> input_raw_file;
>+  std::unique_ptr<RawFile> output_raw_file;
>+
>+  std::stringstream callorder_raw_name;
>+  callorder_raw_name << FLAG_callorder_file << ".char";
>+  FILE* callorder_char_file = OpenFile(callorder_raw_name.str(), "wb");
>+  FILE* settings_file = OpenFile(FLAG_settings_file, "wb");
>+
>+  while (ReadMessageFromFile(debug_file, &event_msg)) {
>+    if (event_msg.type() == Event::REVERSE_STREAM) {
>+      if (!event_msg.has_reverse_stream()) {
>+        printf("Corrupt input file: ReverseStream missing.\n");
>+        return 1;
>+      }
>+
>+      const ReverseStream msg = event_msg.reverse_stream();
>+      if (msg.has_data()) {
>+        if (FLAG_raw && !reverse_raw_file) {
>+          reverse_raw_file.reset(
>+              new RawFile(std::string(FLAG_reverse_file) + ".pcm"));
>+        }
>+        // TODO(aluebs): Replace "num_reverse_channels *
>+        // reverse_samples_per_channel" with "msg.data().size() /
>+        // sizeof(int16_t)" and so on when this fix in audio_processing has made
>+        // it into stable: https://webrtc-codereview.appspot.com/15299004/
>+        WriteIntData(reinterpret_cast<const int16_t*>(msg.data().data()),
>+                     num_reverse_channels * reverse_samples_per_channel,
>+                     reverse_wav_file.get(), reverse_raw_file.get());
>+      } else if (msg.channel_size() > 0) {
>+        if (FLAG_raw && !reverse_raw_file) {
>+          reverse_raw_file.reset(
>+              new RawFile(std::string(FLAG_reverse_file) + ".float"));
>+        }
>+        std::unique_ptr<const float* []> data(
>+            new const float*[num_reverse_channels]);
>+        for (size_t i = 0; i < num_reverse_channels; ++i) {
>+          data[i] = reinterpret_cast<const float*>(msg.channel(i).data());
>+        }
>+        WriteFloatData(data.get(), reverse_samples_per_channel,
>+                       num_reverse_channels, reverse_wav_file.get(),
>+                       reverse_raw_file.get());
>+      }
>+      if (FLAG_full) {
>+        WriteCallOrderData(true /* render_call */, callorder_char_file,
>+                           FLAG_callorder_file);
>+      }
>+    } else if (event_msg.type() == Event::STREAM) {
>+      frame_count++;
>+      if (!event_msg.has_stream()) {
>+        printf("Corrupt input file: Stream missing.\n");
>+        return 1;
>+      }
>+
>+      const Stream msg = event_msg.stream();
>+      if (msg.has_input_data()) {
>+        if (FLAG_raw && !input_raw_file) {
>+          input_raw_file.reset(
>+              new RawFile(std::string(FLAG_input_file) + ".pcm"));
>+        }
>+        WriteIntData(reinterpret_cast<const int16_t*>(msg.input_data().data()),
>+                     num_input_channels * input_samples_per_channel,
>+                     input_wav_file.get(), input_raw_file.get());
>+      } else if (msg.input_channel_size() > 0) {
>+        if (FLAG_raw && !input_raw_file) {
>+          input_raw_file.reset(
>+              new RawFile(std::string(FLAG_input_file) + ".float"));
>+        }
>+        std::unique_ptr<const float* []> data(
>+            new const float*[num_input_channels]);
>+        for (size_t i = 0; i < num_input_channels; ++i) {
>+          data[i] = reinterpret_cast<const float*>(msg.input_channel(i).data());
>+        }
>+        WriteFloatData(data.get(), input_samples_per_channel,
>+                       num_input_channels, input_wav_file.get(),
>+                       input_raw_file.get());
>+      }
>+
>+      if (msg.has_output_data()) {
>+        if (FLAG_raw && !output_raw_file) {
>+          output_raw_file.reset(
>+              new RawFile(std::string(FLAG_output_file) + ".pcm"));
>+        }
>+        WriteIntData(reinterpret_cast<const int16_t*>(msg.output_data().data()),
>+                     num_output_channels * output_samples_per_channel,
>+                     output_wav_file.get(), output_raw_file.get());
>+      } else if (msg.output_channel_size() > 0) {
>+        if (FLAG_raw && !output_raw_file) {
>+          output_raw_file.reset(
>+              new RawFile(std::string(FLAG_output_file) + ".float"));
>+        }
>+        std::unique_ptr<const float* []> data(
>+            new const float*[num_output_channels]);
>+        for (size_t i = 0; i < num_output_channels; ++i) {
>+          data[i] =
>+              reinterpret_cast<const float*>(msg.output_channel(i).data());
>+        }
>+        WriteFloatData(data.get(), output_samples_per_channel,
>+                       num_output_channels, output_wav_file.get(),
>+                       output_raw_file.get());
>+      }
>+
>+      if (FLAG_full) {
>+        WriteCallOrderData(false /* render_call */, callorder_char_file,
>+                           FLAG_callorder_file);
>+        if (msg.has_delay()) {
>+          static FILE* delay_file = OpenFile(FLAG_delay_file, "wb");
>+          int32_t delay = msg.delay();
>+          if (FLAG_text) {
>+            fprintf(delay_file, "%d\n", delay);
>+          } else {
>+            WriteData(&delay, sizeof(delay), delay_file, FLAG_delay_file);
>+          }
>+        }
>+
>+        if (msg.has_drift()) {
>+          static FILE* drift_file = OpenFile(FLAG_drift_file, "wb");
>+          int32_t drift = msg.drift();
>+          if (FLAG_text) {
>+            fprintf(drift_file, "%d\n", drift);
>+          } else {
>+            WriteData(&drift, sizeof(drift), drift_file, FLAG_drift_file);
>+          }
>+        }
>+
>+        if (msg.has_level()) {
>+          static FILE* level_file = OpenFile(FLAG_level_file, "wb");
>+          int32_t level = msg.level();
>+          if (FLAG_text) {
>+            fprintf(level_file, "%d\n", level);
>+          } else {
>+            WriteData(&level, sizeof(level), level_file, FLAG_level_file);
>+          }
>+        }
>+
>+        if (msg.has_keypress()) {
>+          static FILE* keypress_file = OpenFile(FLAG_keypress_file, "wb");
>+          bool keypress = msg.keypress();
>+          if (FLAG_text) {
>+            fprintf(keypress_file, "%d\n", keypress);
>+          } else {
>+            WriteData(&keypress, sizeof(keypress), keypress_file,
>+                      FLAG_keypress_file);
>+          }
>+        }
>+      }
>+    } else if (event_msg.type() == Event::CONFIG) {
>+      if (!event_msg.has_config()) {
>+        printf("Corrupt input file: Config missing.\n");
>+        return 1;
>+      }
>+      const audioproc::Config msg = event_msg.config();
>+
>+      fprintf(settings_file, "APM re-config at frame: %d\n", frame_count);
>+
>+      PRINT_CONFIG(aec_enabled);
>+      PRINT_CONFIG(aec_delay_agnostic_enabled);
>+      PRINT_CONFIG(aec_drift_compensation_enabled);
>+      PRINT_CONFIG(aec_extended_filter_enabled);
>+      PRINT_CONFIG(aec_suppression_level);
>+      PRINT_CONFIG(aecm_enabled);
>+      PRINT_CONFIG(aecm_comfort_noise_enabled);
>+      PRINT_CONFIG(aecm_routing_mode);
>+      PRINT_CONFIG(agc_enabled);
>+      PRINT_CONFIG(agc_mode);
>+      PRINT_CONFIG(agc_limiter_enabled);
>+      PRINT_CONFIG(noise_robust_agc_enabled);
>+      PRINT_CONFIG(hpf_enabled);
>+      PRINT_CONFIG(ns_enabled);
>+      PRINT_CONFIG(ns_level);
>+      PRINT_CONFIG(transient_suppression_enabled);
>+      PRINT_CONFIG(intelligibility_enhancer_enabled);
>+      PRINT_CONFIG(pre_amplifier_enabled);
>+      PRINT_CONFIG_FLOAT(pre_amplifier_fixed_gain_factor);
>+
>+      if (msg.has_experiments_description()) {
>+        fprintf(settings_file, "  experiments_description: %s\n",
>+                msg.experiments_description().c_str());
>+      }
>+    } else if (event_msg.type() == Event::INIT) {
>+      if (!event_msg.has_init()) {
>+        printf("Corrupt input file: Init missing.\n");
>+        return 1;
>+      }
>+
>+      const Init msg = event_msg.init();
>+      // These should print out zeros if they're missing.
>+      fprintf(settings_file, "Init at frame: %d\n", frame_count);
>+      int input_sample_rate = msg.sample_rate();
>+      fprintf(settings_file, "  Input sample rate: %d\n", input_sample_rate);
>+      int output_sample_rate = msg.output_sample_rate();
>+      fprintf(settings_file, "  Output sample rate: %d\n", output_sample_rate);
>+      int reverse_sample_rate = msg.reverse_sample_rate();
>+      fprintf(settings_file, "  Reverse sample rate: %d\n",
>+              reverse_sample_rate);
>+      num_input_channels = msg.num_input_channels();
>+      fprintf(settings_file, "  Input channels: %" PRIuS "\n",
>+              num_input_channels);
>+      num_output_channels = msg.num_output_channels();
>+      fprintf(settings_file, "  Output channels: %" PRIuS "\n",
>+              num_output_channels);
>+      num_reverse_channels = msg.num_reverse_channels();
>+      fprintf(settings_file, "  Reverse channels: %" PRIuS "\n",
>+              num_reverse_channels);
>+      if (msg.has_timestamp_ms()) {
>+        const int64_t timestamp = msg.timestamp_ms();
>+        fprintf(settings_file, "  Timestamp in millisecond: %" PRId64 "\n",
>+                timestamp);
>+      }
>+
>+      fprintf(settings_file, "\n");
>+
>+      if (reverse_sample_rate == 0) {
>+        reverse_sample_rate = input_sample_rate;
>+      }
>+      if (output_sample_rate == 0) {
>+        output_sample_rate = input_sample_rate;
>+      }
>+
>+      reverse_samples_per_channel =
>+          static_cast<size_t>(reverse_sample_rate / 100);
>+      input_samples_per_channel = static_cast<size_t>(input_sample_rate / 100);
>+      output_samples_per_channel =
>+          static_cast<size_t>(output_sample_rate / 100);
>+
>+      if (!FLAG_raw) {
>+        // The WAV files need to be reset every time, because they cant change
>+        // their sample rate or number of channels.
>+        std::stringstream reverse_name;
>+        reverse_name << FLAG_reverse_file << frame_count << ".wav";
>+        reverse_wav_file.reset(new WavWriter(
>+            reverse_name.str(), reverse_sample_rate, num_reverse_channels));
>+        std::stringstream input_name;
>+        input_name << FLAG_input_file << frame_count << ".wav";
>+        input_wav_file.reset(new WavWriter(input_name.str(), input_sample_rate,
>+                                           num_input_channels));
>+        std::stringstream output_name;
>+        output_name << FLAG_output_file << frame_count << ".wav";
>+        output_wav_file.reset(new WavWriter(
>+            output_name.str(), output_sample_rate, num_output_channels));
>+
>+        std::stringstream callorder_name;
>+        callorder_name << FLAG_callorder_file << frame_count << ".char";
>+        callorder_char_file = OpenFile(callorder_name.str(), "wb");
>+      }
>+    }
>+  }
>+
>+  return 0;
>+}
>+
>+}  // namespace webrtc
>+
>+int main(int argc, char* argv[]) {
>+  return webrtc::do_main(argc, argv);
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/video_analysis.py b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/video_analysis.py
>old mode 100644
>new mode 100755
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/video_analysis_test.py b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_tools/video_analysis_test.py
>old mode 100644
>new mode 100755
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/BUILD.gn
>index b587a55ca91..aa62d8b07c9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/BUILD.gn
>@@ -48,7 +48,14 @@ if (is_ios || is_mac) {
>     ]
>   }
> 
>+  config("used_from_extension") {
>+    if (is_ios && rtc_apprtcmobile_broadcast_extension) {
>+      cflags = [ "-fapplication-extension" ]
>+    }
>+  }
>+
>   rtc_static_library("common_objc") {
>+    visibility = [ "*" ]
>     sources = [
>       "objc/Framework/Classes/Common/NSString+StdString.h",
>       "objc/Framework/Classes/Common/NSString+StdString.mm",
>@@ -71,7 +78,10 @@ if (is_ios || is_mac) {
>       "../rtc_base:checks",
>       "../rtc_base:rtc_base",
>     ]
>-    configs += [ "..:common_objc" ]
>+    configs += [
>+      "..:common_objc",
>+      ":used_from_extension",
>+    ]
> 
>     public_configs = [ ":common_config_objc" ]
> 
>@@ -84,80 +94,172 @@ if (is_ios || is_mac) {
> 
>     if (!build_with_chromium) {
>       sources += [
>+        "objc/Framework/Classes/Common/RTCCallbackLogger.mm",
>         "objc/Framework/Classes/Common/RTCFileLogger.mm",
>+        "objc/Framework/Headers/WebRTC/RTCCallbackLogger.h",
>         "objc/Framework/Headers/WebRTC/RTCFileLogger.h",
>       ]
>     }
>   }
> 
>   if (!build_with_chromium) {
>-    rtc_static_library("audio_objc") {
>+    if (is_ios) {
>+      rtc_static_library("native_api_audio_device_module") {
>+        visibility = [ "*" ]
>+
>+        sources = [
>+          "objc/Framework/Native/api/audio_device_module.h",
>+          "objc/Framework/Native/api/audio_device_module.mm",
>+        ]
>+
>+        deps = [
>+          ":audio_device",
>+          "../modules/audio_device:audio_device_api",
>+          "../modules/audio_device:audio_device_generic",
>+          "../rtc_base:checks",
>+          "../rtc_base:rtc_base_approved",
>+          "../system_wrappers",
>+          "../system_wrappers:metrics_api",
>+        ]
>+
>+        if (is_clang) {
>+          # Suppress warnings from the Chromium Clang plugin
>+          # (bugs.webrtc.org/163).
>+          suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>+        }
>+      }
>+
>+      rtc_source_set("audio_session_observer") {
>+        visibility = [ ":*" ]
>+
>+        sources = [
>+          "objc/Framework/Native/src/audio/audio_session_observer.h",
>+        ]
>+
>+        deps = [
>+          "../rtc_base:rtc_base",
>+        ]
>+      }
>+
>+      rtc_static_library("audio_device") {
>+        visibility = [ "*" ]
>+
>+        sources = [
>+          "objc/Framework/Native/src/audio/audio_device_ios.h",
>+          "objc/Framework/Native/src/audio/audio_device_ios.mm",
>+          "objc/Framework/Native/src/audio/audio_device_module_ios.h",
>+          "objc/Framework/Native/src/audio/audio_device_module_ios.mm",
>+          "objc/Framework/Native/src/audio/voice_processing_audio_unit.h",
>+          "objc/Framework/Native/src/audio/voice_processing_audio_unit.mm",
>+        ]
>+
>+        deps = [
>+          ":audio_objc",
>+          ":audio_session_observer",
>+          ":common_objc",
>+          "../api:array_view",
>+          "../modules/audio_device:audio_device_api",
>+          "../modules/audio_device:audio_device_buffer",
>+          "../modules/audio_device:audio_device_generic",
>+          "../rtc_base:checks",
>+          "../rtc_base:rtc_base",
>+          "../rtc_base/system:fallthrough",
>+          "../sdk:common_objc",
>+          "../system_wrappers:metrics_api",
>+        ]
>+
>+        if (is_clang) {
>+          # Suppress warnings from the Chromium Clang plugin
>+          # (bugs.webrtc.org/163).
>+          suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>+        }
>+      }
>+
>+      rtc_static_library("audio_objc") {
>+        sources = [
>+          "objc/Framework/Classes/Audio/RTCAudioSession+Configuration.mm",
>+          "objc/Framework/Classes/Audio/RTCAudioSession+Private.h",
>+          "objc/Framework/Classes/Audio/RTCAudioSession.mm",
>+          "objc/Framework/Classes/Audio/RTCAudioSessionConfiguration.m",
>+          "objc/Framework/Classes/Audio/RTCNativeAudioSessionDelegateAdapter.h",
>+          "objc/Framework/Classes/Audio/RTCNativeAudioSessionDelegateAdapter.mm",
>+          "objc/Framework/Headers/WebRTC/RTCAudioSession.h",
>+          "objc/Framework/Headers/WebRTC/RTCAudioSessionConfiguration.h",
>+        ]
>+        configs += [
>+          "..:common_objc",
>+          ":used_from_extension",
>+        ]
>+
>+        public_configs = [ ":common_config_objc" ]
>+
>+        deps = [
>+          ":audio_session_observer",
>+          ":common_objc",
>+          "../rtc_base:checks",
>+          "../rtc_base:rtc_base",
>+          "../rtc_base:rtc_base_approved",
>+        ]
>+      }
>+    }
>+
>+    rtc_static_library("videosource_objc") {
>       sources = [
>-        "objc/Framework/Classes/Audio/RTCAudioSession+Configuration.mm",
>-        "objc/Framework/Classes/Audio/RTCAudioSession+Private.h",
>-        "objc/Framework/Classes/Audio/RTCAudioSession.mm",
>-        "objc/Framework/Classes/Audio/RTCAudioSessionConfiguration.m",
>-        "objc/Framework/Headers/WebRTC/RTCAudioSession.h",
>-        "objc/Framework/Headers/WebRTC/RTCAudioSessionConfiguration.h",
>+        "objc/Framework/Classes/PeerConnection/RTCVideoSource+Private.h",
>+        "objc/Framework/Classes/PeerConnection/RTCVideoSource.mm",
>+        "objc/Framework/Headers/WebRTC/RTCVideoSource.h",
>       ]
>-      configs += [ "..:common_objc" ]
>-
>-      public_configs = [ ":common_config_objc" ]
> 
>       deps = [
>         ":common_objc",
>+        ":mediasource_objc",
>+        ":native_video",
>+        ":videoframebuffer_objc",
>+        "../api:libjingle_peerconnection_api",
>+        "../api/video:video_frame",
>+        "../api/video:video_frame_i420",
>+        "../common_video",
>+        "../media:rtc_media_base",
>         "../rtc_base:checks",
>-        "../rtc_base:rtc_base_approved",
>+        "../rtc_base:rtc_base",
>+        "//third_party/libyuv",
>       ]
> 
>-      if (is_clang) {
>-        # Suppress warnings from the Chromium Clang plugin
>-        # (bugs.webrtc.org/163).
>-        suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-      }
>+      configs += [
>+        "..:common_objc",
>+        ":used_from_extension",
>+      ]
>     }
> 
>-    # Keep videotracksource related code in a separate target so that we can
>-    # build PeerConnectionFactory without audio/video support when that's called
>-    # for.
>-    rtc_static_library("videotracksource_objc") {
>+    rtc_static_library("videoframebuffer_objc") {
>+      visibility = [ "*" ]
>       sources = [
>+        "objc/Framework/Classes/PeerConnection/RTCVideoFrame.mm",
>         "objc/Framework/Classes/Video/RTCCVPixelBuffer.mm",
>         "objc/Framework/Classes/Video/RTCI420Buffer+Private.h",
>         "objc/Framework/Classes/Video/RTCI420Buffer.mm",
>-        "objc/Framework/Classes/Video/objc_frame_buffer.h",
>-        "objc/Framework/Classes/Video/objc_frame_buffer.mm",
>-        "objc/Framework/Classes/Video/objcvideotracksource.h",
>-        "objc/Framework/Classes/Video/objcvideotracksource.mm",
>+        "objc/Framework/Headers/WebRTC/RTCVideoFrame.h",
>         "objc/Framework/Headers/WebRTC/RTCVideoFrameBuffer.h",
>       ]
>-
>       deps = [
>         ":common_objc",
>-        "../api:libjingle_peerconnection_api",
>-        "../api:video_frame_api",
>-        "../api:video_frame_api_i420",
>-        "../common_video",
>-        "../media:rtc_media_base",
>-        "../rtc_base:checks",
>-        "../rtc_base:rtc_base",
>+        "//api/video:video_frame",
>+        "//api/video:video_frame_i420",
>+        "//common_video",
>+        "//rtc_base:checks",
>+        "//rtc_base:rtc_base_approved",
>         "//third_party/libyuv",
>       ]
>-
>-      configs += [ "..:common_objc" ]
>-      if (!build_with_chromium && is_clang) {
>-        # Suppress warnings from the Chromium Clang plugin
>-        # (bugs.webrtc.org/163).
>-        suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-      }
>+      configs += [
>+        "..:common_objc",
>+        ":used_from_extension",
>+      ]
>     }
> 
>     rtc_static_library("video_objc") {
>       sources = [
>         "objc/Framework/Classes/Video/AVCaptureSession+DevicePosition.h",
>         "objc/Framework/Classes/Video/AVCaptureSession+DevicePosition.mm",
>-        "objc/Framework/Classes/Video/RTCAVFoundationVideoCapturerInternal.h",
>-        "objc/Framework/Classes/Video/RTCAVFoundationVideoCapturerInternal.mm",
>         "objc/Framework/Classes/Video/RTCDefaultShader.h",
>         "objc/Framework/Classes/Video/RTCDefaultShader.mm",
>         "objc/Framework/Classes/Video/RTCI420TextureCache.h",
>@@ -165,10 +267,7 @@ if (is_ios || is_mac) {
>         "objc/Framework/Classes/Video/RTCOpenGLDefines.h",
>         "objc/Framework/Classes/Video/RTCShader.h",
>         "objc/Framework/Classes/Video/RTCShader.mm",
>-        "objc/Framework/Classes/Video/avfoundationformatmapper.h",
>-        "objc/Framework/Classes/Video/avfoundationformatmapper.mm",
>-        "objc/Framework/Classes/Video/avfoundationvideocapturer.h",
>-        "objc/Framework/Classes/Video/avfoundationvideocapturer.mm",
>+        "objc/Framework/Headers/WebRTC/RTCVideoViewShading.h",
>       ]
>       libs = []
>       if (is_ios) {
>@@ -182,8 +281,6 @@ if (is_ios || is_mac) {
>           "QuartzCore.framework",
>         ]
>       } else if (is_mac) {
>-        sources += []
>-
>         libs += [
>           "CoreMedia.framework",
>           "CoreVideo.framework",
>@@ -193,26 +290,31 @@ if (is_ios || is_mac) {
> 
>       deps = [
>         ":common_objc",
>-        ":videotracksource_objc",
>+        ":mediaconstraints_objc",
>+        ":native_video",
>+        ":videoframebuffer_objc",
>+        ":videosource_objc",
>         "../api:libjingle_peerconnection_api",
>-        "../api:optional",
>-        "../api:video_frame_api",
>+        "../api/video:video_frame",
>         "../common_video",
>         "../media:rtc_media_base",
>         "../rtc_base:checks",
>         "../rtc_base:rtc_base",
>+        "//third_party/abseil-cpp/absl/types:optional",
>       ]
> 
>-      configs += [ "..:common_objc" ]
>-      if (!build_with_chromium && is_clang) {
>-        # Suppress warnings from the Chromium Clang plugin
>-        # (bugs.webrtc.org/163).
>-        suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-      }
>+      configs += [
>+        "..:common_objc",
>+        ":used_from_extension",
>+      ]
>     }
> 
>     rtc_static_library("ui_objc") {
>       visibility = [ "*" ]
>+      allow_poison = [
>+        "audio_codecs",  # TODO(bugs.webrtc.org/8396): Remove.
>+        "software_video_codecs",  # TODO(bugs.webrtc.org/7925): Remove.
>+      ]
>       if (is_ios) {
>         sources = [
>           "objc/Framework/Classes/UI/RTCCameraPreviewView.m",
>@@ -230,12 +332,19 @@ if (is_ios || is_mac) {
>       configs += [ "..:common_objc" ]
>       deps = [
>         ":common_objc",
>-        ":peerconnection_objc",
>+        ":video_objc",
>+        ":videocapture_objc",
>+        ":videoframebuffer_objc",
>       ]
>     }
> 
>     if (rtc_use_metal_rendering) {
>       rtc_static_library("metal_objc") {
>+        visibility = [ "*" ]
>+        allow_poison = [
>+          "audio_codecs",  # TODO(bugs.webrtc.org/8396): Remove.
>+          "software_video_codecs",  # TODO(bugs.webrtc.org/7925): Remove.
>+        ]
>         sources = [
>           "objc/Framework/Classes/Metal/RTCMTLI420Renderer.h",
>           "objc/Framework/Classes/Metal/RTCMTLI420Renderer.mm",
>@@ -247,6 +356,8 @@ if (is_ios || is_mac) {
>           sources += [
>             "objc/Framework/Classes/Metal/RTCMTLNV12Renderer.h",
>             "objc/Framework/Classes/Metal/RTCMTLNV12Renderer.mm",
>+            "objc/Framework/Classes/Metal/RTCMTLRGBRenderer.h",
>+            "objc/Framework/Classes/Metal/RTCMTLRGBRenderer.mm",
>             "objc/Framework/Classes/Metal/RTCMTLVideoView.m",
>             "objc/Framework/Headers/WebRTC/RTCMTLVideoView.h",
>           ]
>@@ -263,8 +374,12 @@ if (is_ios || is_mac) {
>           "MetalKit.framework",
>         ]
>         deps = [
>+          ":common_objc",
>+          ":peerconnectionfactory_base_objc",
>           ":video_objc",
>-          "../api:video_frame_api",
>+          ":videoframebuffer_objc",
>+          ":videorenderer_objc",
>+          "../api/video:video_frame",
>           "../rtc_base:checks",
>           "../rtc_base:rtc_base_approved",
>         ]
>@@ -273,11 +388,31 @@ if (is_ios || is_mac) {
>       }
>     }
> 
>-    rtc_static_library("peerconnection_objc") {
>+    rtc_static_library("videocapturebase_objc") {
>       visibility = [ "*" ]
>       sources = [
>-        "objc/Framework/Classes/PeerConnection/RTCAVFoundationVideoSource+Private.h",
>-        "objc/Framework/Classes/PeerConnection/RTCAVFoundationVideoSource.mm",
>+        "objc/Framework/Classes/PeerConnection/RTCVideoCapturer.m",
>+        "objc/Framework/Headers/WebRTC/RTCVideoCapturer.h",
>+      ]
>+      libs = [ "AVFoundation.framework" ]
>+
>+      configs += [ "..:common_objc" ]
>+
>+      public_configs = [ ":common_config_objc" ]
>+
>+      deps = [
>+        ":common_objc",
>+        ":videoframebuffer_objc",
>+      ]
>+    }
>+
>+    rtc_static_library("videocapture_objc") {
>+      visibility = [ "*" ]
>+      allow_poison = [
>+        "audio_codecs",  # TODO(bugs.webrtc.org/8396): Remove.
>+        "software_video_codecs",  # TODO(bugs.webrtc.org/7925): Remove.
>+      ]
>+      sources = [
>         "objc/Framework/Classes/PeerConnection/RTCCameraVideoCapturer.m",
>         "objc/Framework/Headers/WebRTC/RTCCameraVideoCapturer.h",
>       ]
>@@ -289,77 +424,101 @@ if (is_ios || is_mac) {
>       }
>       libs = [ "AVFoundation.framework" ]
> 
>-      if (is_ios) {
>-        libs += [ "CoreGraphics.framework" ]
>-      }
>-
>       configs += [ "..:common_objc" ]
> 
>       public_configs = [ ":common_config_objc" ]
> 
>+      deps = [
>+        ":common_objc",
>+        ":video_objc",
>+        ":videocapturebase_objc",
>+        ":videoframebuffer_objc",
>+      ]
>+    }
>+
>+    rtc_static_library("videocodec_objc") {
>+      visibility = [ "*" ]
>+      sources = [
>+        "objc/Framework/Classes/PeerConnection/RTCEncodedImage.mm",
>+        "objc/Framework/Classes/PeerConnection/RTCH264ProfileLevelId.mm",
>+        "objc/Framework/Classes/PeerConnection/RTCRtpFragmentationHeader.mm",
>+        "objc/Framework/Classes/PeerConnection/RTCVideoCodec+Private.h",
>+        "objc/Framework/Classes/PeerConnection/RTCVideoCodec.mm",
>+        "objc/Framework/Classes/PeerConnection/RTCVideoCodecH264.mm",
>+        "objc/Framework/Classes/PeerConnection/RTCVideoEncoderSettings.mm",
>+        "objc/Framework/Headers/WebRTC/RTCVideoCodec.h",
>+        "objc/Framework/Headers/WebRTC/RTCVideoCodecFactory.h",
>+        "objc/Framework/Headers/WebRTC/RTCVideoCodecH264.h",
>+      ]
>+      if (is_ios) {
>+        sources += [
>+          "objc/Framework/Classes/Video/UIDevice+H264Profile.h",
>+          "objc/Framework/Classes/Video/UIDevice+H264Profile.mm",
>+        ]
>+      }
>       if (!build_with_chromium && is_clang) {
>         # Suppress warnings from the Chromium Clang plugin
>         # (bugs.webrtc.org/163).
>         suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>       }
>-
>+      public_configs = [ ":common_config_objc" ]
>       deps = [
>         ":common_objc",
>-        ":corevideoframebuffer_objc",
>-        ":peerconnectionfactory_objc",
>-        ":video_objc",
>-        "../api:video_frame_api",
>+        "../api/video_codecs:video_codecs_api",
>+        "../common_video",
>+        "../media:rtc_h264_profile_id",
>         "../media:rtc_media_base",
>-        "../pc:libjingle_peerconnection",
>-        "../rtc_base:rtc_base",
>+        "../modules:module_api",
>+        "../modules/video_coding:video_codec_interface",
>+        "../rtc_base:rtc_base_approved",
>+        "../system_wrappers:field_trial_api",
>       ]
>-
>-      if (rtc_use_metal_rendering) {
>-        deps += [ ":metal_objc" ]
>-      }
>     }
> 
>-    rtc_static_library("peerconnectionfactory_objc") {
>+    rtc_static_library("default_codec_factory_objc") {
>       sources = [
>         "objc/Framework/Classes/PeerConnection/RTCDefaultVideoDecoderFactory.m",
>         "objc/Framework/Classes/PeerConnection/RTCDefaultVideoEncoderFactory.m",
>-        "objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactory+Native.h",
>-        "objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactory+Private.h",
>-        "objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactory.mm",
>-        "objc/Framework/Classes/PeerConnection/RTCVideoCodecH264.mm",
>-        "objc/Framework/Classes/PeerConnection/RTCVideoCodecVP8.mm",
>-        "objc/Framework/Classes/PeerConnection/RTCVideoCodecVP9.mm",
>       ]
> 
>-      public_configs = [ ":common_config_objc" ]
>+      deps = [
>+        ":common_objc",
>+        ":native_video",
>+        ":videocodec_objc",
>+        ":vp8",
>+        ":vp9",
>+      ]
>+    }
> 
>-      if (!build_with_chromium && is_clang) {
>-        # Suppress warnings from the Chromium Clang plugin
>-        # (bugs.webrtc.org/163).
>-        suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-      }
>+    rtc_static_library("vp8") {
>+      visibility = [ "*" ]
>+      allow_poison = [ "software_video_codecs" ]
>+      sources = [
>+        "objc/Framework/Classes/PeerConnection/RTCVideoCodecVP8.mm",
>+        "objc/Framework/Headers/WebRTC/RTCVideoDecoderVP8.h",
>+        "objc/Framework/Headers/WebRTC/RTCVideoEncoderVP8.h",
>+      ]
> 
>       deps = [
>-        ":common_objc",
>-        ":corevideoframebuffer_objc",
>-        ":peerconnectionfactory_base_objc",
>-        ":video_objc",
>-        ":videotoolbox_objc",
>-        ":videotracksource_objc",
>-        "../api:libjingle_peerconnection_api",
>-        "../api:peerconnection_and_implicit_call_api",
>-        "../api:video_frame_api",
>-        "../api/audio_codecs:builtin_audio_decoder_factory",
>-        "../api/audio_codecs:builtin_audio_encoder_factory",
>-        "../api/video_codecs:video_codecs_api",
>-        "../media:rtc_audio_video",
>-        "../media:rtc_media_base",
>+        ":wrapped_native_codec_objc",
>         "../modules/video_coding:webrtc_vp8",
>+        "../system_wrappers:metrics_default",
>+      ]
>+    }
>+
>+    rtc_static_library("vp9") {
>+      visibility = [ "*" ]
>+      allow_poison = [ "software_video_codecs" ]
>+      sources = [
>+        "objc/Framework/Classes/PeerConnection/RTCVideoCodecVP9.mm",
>+        "objc/Framework/Headers/WebRTC/RTCVideoDecoderVP9.h",
>+        "objc/Framework/Headers/WebRTC/RTCVideoEncoderVP9.h",
>+      ]
>+
>+      deps = [
>+        ":wrapped_native_codec_objc",
>         "../modules/video_coding:webrtc_vp9",
>-        "../pc:create_pc_factory",
>-        "../pc:peerconnection",
>-        "../rtc_base:rtc_base",
>-        "../system_wrappers:field_trial_api",
>+        "../system_wrappers:metrics_default",
>       ]
>     }
> 
>@@ -369,40 +528,110 @@ if (is_ios || is_mac) {
>     # these objects are just thin wrappers of native C++ interfaces required
>     # when implementing webrtc::PeerConnectionFactoryInterface and
>     # webrtc::PeerConnectionInterface.
>-    # The applications which only use WebRTC DataChannel can depend on this
>-    # target instead of rtc_sdk_objc.
>+    # The applications which only use WebRTC DataChannel can depend on this.
>     rtc_static_library("peerconnectionfactory_no_media_objc") {
>       visibility = [ "*" ]
>+      allow_poison = [
>+        "audio_codecs",  # TODO(bugs.webrtc.org/8396): Remove.
>+        "software_video_codecs",  # TODO(bugs.webrtc.org/7925): Remove.
>+      ]
>       defines = [ "HAVE_NO_MEDIA" ]
> 
>       sources = [
>-        "objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactory+Native.h",
>-        "objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactory+Private.h",
>-        "objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactory.mm",
>+        "objc/Framework/Classes/Common/noop.mm",
>       ]
> 
>       public_configs = [ ":common_config_objc" ]
> 
>-      if (!build_with_chromium && is_clang) {
>-        # Suppress warnings from the Chromium Clang plugin
>-        # (bugs.webrtc.org/163).
>-        suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-      }
>-
>       deps = [
>+        ":native_api",
>+        ":native_video",
>         ":peerconnectionfactory_base_objc",
>         "../api:libjingle_peerconnection_api",
>-        "../api:peerconnection_and_implicit_call_api",
>         "../rtc_base:rtc_base",
>       ]
>     }
> 
>+    rtc_static_library("mediaconstraints_objc") {
>+      sources = [
>+        "objc/Framework/Classes/PeerConnection/RTCMediaConstraints+Private.h",
>+        "objc/Framework/Classes/PeerConnection/RTCMediaConstraints.mm",
>+        "objc/Framework/Headers/WebRTC/RTCMediaConstraints.h",
>+      ]
>+
>+      public_configs = [ ":common_config_objc" ]
>+      deps = [
>+        ":common_objc",
>+        "../api:libjingle_peerconnection_api",
>+      ]
>+    }
>+
>+    rtc_source_set("videorenderer_objc") {
>+      sources = [
>+        "objc/Framework/Headers/WebRTC/RTCVideoRenderer.h",
>+      ]
>+
>+      configs += [ "..:common_objc" ]
>+      visibility = [ "*" ]
>+      public_configs = [ ":common_config_objc" ]
>+    }
>+
>+    rtc_static_library("videorendereradapter_objc") {
>+      visibility = [ "*" ]
>+      allow_poison = [
>+        "audio_codecs",  # TODO(bugs.webrtc.org/8396): Remove.
>+        "software_video_codecs",  # TODO(bugs.webrtc.org/7925): Remove.
>+      ]
>+      sources = [
>+        "objc/Framework/Classes/PeerConnection/RTCVideoRendererAdapter+Private.h",
>+        "objc/Framework/Classes/PeerConnection/RTCVideoRendererAdapter.h",
>+        "objc/Framework/Classes/PeerConnection/RTCVideoRendererAdapter.mm",
>+      ]
>+
>+      configs += [ "..:common_objc" ]
>+      public_configs = [ ":common_config_objc" ]
>+
>+      deps = [
>+        ":common_objc",
>+        ":native_api",
>+        ":videoframebuffer_objc",
>+        ":videorenderer_objc",
>+        "../api:libjingle_peerconnection_api",
>+      ]
>+    }
>+
>+    rtc_static_library("mediasource_objc") {
>+      sources = [
>+        "objc/Framework/Classes/PeerConnection/RTCMediaSource+Private.h",
>+        "objc/Framework/Classes/PeerConnection/RTCMediaSource.mm",
>+        "objc/Framework/Headers/WebRTC/RTCMediaSource.h",
>+      ]
>+
>+      configs += [
>+        "..:common_objc",
>+        ":used_from_extension",
>+      ]
>+      public_configs = [ ":common_config_objc" ]
>+
>+      deps = [
>+        ":common_objc",
>+        "../api:libjingle_peerconnection_api",
>+        "../rtc_base:checks",
>+      ]
>+    }
>+
>     rtc_static_library("peerconnectionfactory_base_objc") {
>+      visibility = [ "*" ]
>+      allow_poison = [
>+        "audio_codecs",  # TODO(bugs.webrtc.org/8396): Remove.
>+        "software_video_codecs",  # TODO(bugs.webrtc.org/7925): Remove.
>+      ]
>       sources = [
>         "objc/Framework/Classes/PeerConnection/RTCAudioSource+Private.h",
>         "objc/Framework/Classes/PeerConnection/RTCAudioSource.mm",
>         "objc/Framework/Classes/PeerConnection/RTCAudioTrack+Private.h",
>         "objc/Framework/Classes/PeerConnection/RTCAudioTrack.mm",
>+        "objc/Framework/Classes/PeerConnection/RTCCertificate.mm",
>         "objc/Framework/Classes/PeerConnection/RTCConfiguration+Native.h",
>         "objc/Framework/Classes/PeerConnection/RTCConfiguration+Private.h",
>         "objc/Framework/Classes/PeerConnection/RTCConfiguration.mm",
>@@ -412,7 +641,6 @@ if (is_ios || is_mac) {
>         "objc/Framework/Classes/PeerConnection/RTCDataChannelConfiguration.mm",
>         "objc/Framework/Classes/PeerConnection/RTCDtmfSender+Private.h",
>         "objc/Framework/Classes/PeerConnection/RTCDtmfSender.mm",
>-        "objc/Framework/Classes/PeerConnection/RTCEncodedImage.mm",
>         "objc/Framework/Classes/PeerConnection/RTCIceCandidate+Private.h",
>         "objc/Framework/Classes/PeerConnection/RTCIceCandidate.mm",
>         "objc/Framework/Classes/PeerConnection/RTCIceServer+Private.h",
>@@ -421,10 +649,6 @@ if (is_ios || is_mac) {
>         "objc/Framework/Classes/PeerConnection/RTCIntervalRange.mm",
>         "objc/Framework/Classes/PeerConnection/RTCLegacyStatsReport+Private.h",
>         "objc/Framework/Classes/PeerConnection/RTCLegacyStatsReport.mm",
>-        "objc/Framework/Classes/PeerConnection/RTCMediaConstraints+Private.h",
>-        "objc/Framework/Classes/PeerConnection/RTCMediaConstraints.mm",
>-        "objc/Framework/Classes/PeerConnection/RTCMediaSource+Private.h",
>-        "objc/Framework/Classes/PeerConnection/RTCMediaSource.mm",
>         "objc/Framework/Classes/PeerConnection/RTCMediaStream+Private.h",
>         "objc/Framework/Classes/PeerConnection/RTCMediaStream.mm",
>         "objc/Framework/Classes/PeerConnection/RTCMediaStreamTrack+Private.h",
>@@ -437,43 +661,40 @@ if (is_ios || is_mac) {
>         "objc/Framework/Classes/PeerConnection/RTCPeerConnection+Private.h",
>         "objc/Framework/Classes/PeerConnection/RTCPeerConnection+Stats.mm",
>         "objc/Framework/Classes/PeerConnection/RTCPeerConnection.mm",
>+        "objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactory+Native.h",
>+        "objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactory+Private.h",
>+        "objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactory.mm",
>+        "objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactoryBuilder+DefaultComponents.h",
>+        "objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactoryBuilder+DefaultComponents.mm",
>+        "objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactoryBuilder.h",
>+        "objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactoryBuilder.mm",
>         "objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactoryOptions+Private.h",
>         "objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactoryOptions.mm",
>+        "objc/Framework/Classes/PeerConnection/RTCRtcpParameters+Private.h",
>+        "objc/Framework/Classes/PeerConnection/RTCRtcpParameters.mm",
>         "objc/Framework/Classes/PeerConnection/RTCRtpCodecParameters+Private.h",
>         "objc/Framework/Classes/PeerConnection/RTCRtpCodecParameters.mm",
>         "objc/Framework/Classes/PeerConnection/RTCRtpEncodingParameters+Private.h",
>         "objc/Framework/Classes/PeerConnection/RTCRtpEncodingParameters.mm",
>-        "objc/Framework/Classes/PeerConnection/RTCRtpFragmentationHeader.mm",
>+        "objc/Framework/Classes/PeerConnection/RTCRtpHeaderExtension+Private.h",
>+        "objc/Framework/Classes/PeerConnection/RTCRtpHeaderExtension.mm",
>         "objc/Framework/Classes/PeerConnection/RTCRtpParameters+Private.h",
>         "objc/Framework/Classes/PeerConnection/RTCRtpParameters.mm",
>         "objc/Framework/Classes/PeerConnection/RTCRtpReceiver+Private.h",
>         "objc/Framework/Classes/PeerConnection/RTCRtpReceiver.mm",
>         "objc/Framework/Classes/PeerConnection/RTCRtpSender+Private.h",
>         "objc/Framework/Classes/PeerConnection/RTCRtpSender.mm",
>+        "objc/Framework/Classes/PeerConnection/RTCRtpTransceiver+Private.h",
>+        "objc/Framework/Classes/PeerConnection/RTCRtpTransceiver.mm",
>         "objc/Framework/Classes/PeerConnection/RTCSSLAdapter.mm",
>         "objc/Framework/Classes/PeerConnection/RTCSessionDescription+Private.h",
>         "objc/Framework/Classes/PeerConnection/RTCSessionDescription.mm",
>         "objc/Framework/Classes/PeerConnection/RTCTracing.mm",
>-        "objc/Framework/Classes/PeerConnection/RTCVideoCapturer.m",
>-        "objc/Framework/Classes/PeerConnection/RTCVideoCodec+Private.h",
>-        "objc/Framework/Classes/PeerConnection/RTCVideoCodec.mm",
>-        "objc/Framework/Classes/PeerConnection/RTCVideoEncoderSettings.mm",
>-        "objc/Framework/Classes/PeerConnection/RTCVideoFrame+Private.h",
>-        "objc/Framework/Classes/PeerConnection/RTCVideoFrame.mm",
>-        "objc/Framework/Classes/PeerConnection/RTCVideoRendererAdapter+Private.h",
>-        "objc/Framework/Classes/PeerConnection/RTCVideoRendererAdapter.h",
>-        "objc/Framework/Classes/PeerConnection/RTCVideoRendererAdapter.mm",
>-        "objc/Framework/Classes/PeerConnection/RTCVideoSource+Private.h",
>-        "objc/Framework/Classes/PeerConnection/RTCVideoSource.mm",
>         "objc/Framework/Classes/PeerConnection/RTCVideoTrack+Private.h",
>         "objc/Framework/Classes/PeerConnection/RTCVideoTrack.mm",
>-        "objc/Framework/Classes/PeerConnection/RTCWrappedNativeVideoDecoder.h",
>-        "objc/Framework/Classes/PeerConnection/RTCWrappedNativeVideoDecoder.mm",
>-        "objc/Framework/Classes/PeerConnection/RTCWrappedNativeVideoEncoder.h",
>-        "objc/Framework/Classes/PeerConnection/RTCWrappedNativeVideoEncoder.mm",
>-        "objc/Framework/Headers/WebRTC/RTCAVFoundationVideoSource.h",
>         "objc/Framework/Headers/WebRTC/RTCAudioSource.h",
>         "objc/Framework/Headers/WebRTC/RTCAudioTrack.h",
>+        "objc/Framework/Headers/WebRTC/RTCCertificate.h",
>         "objc/Framework/Headers/WebRTC/RTCConfiguration.h",
>         "objc/Framework/Headers/WebRTC/RTCDataChannel.h",
>         "objc/Framework/Headers/WebRTC/RTCDataChannelConfiguration.h",
>@@ -482,8 +703,6 @@ if (is_ios || is_mac) {
>         "objc/Framework/Headers/WebRTC/RTCIceServer.h",
>         "objc/Framework/Headers/WebRTC/RTCIntervalRange.h",
>         "objc/Framework/Headers/WebRTC/RTCLegacyStatsReport.h",
>-        "objc/Framework/Headers/WebRTC/RTCMediaConstraints.h",
>-        "objc/Framework/Headers/WebRTC/RTCMediaSource.h",
>         "objc/Framework/Headers/WebRTC/RTCMediaStream.h",
>         "objc/Framework/Headers/WebRTC/RTCMediaStreamTrack.h",
>         "objc/Framework/Headers/WebRTC/RTCMetrics.h",
>@@ -491,49 +710,60 @@ if (is_ios || is_mac) {
>         "objc/Framework/Headers/WebRTC/RTCPeerConnection.h",
>         "objc/Framework/Headers/WebRTC/RTCPeerConnectionFactory.h",
>         "objc/Framework/Headers/WebRTC/RTCPeerConnectionFactoryOptions.h",
>+        "objc/Framework/Headers/WebRTC/RTCRtcpParameters.h",
>         "objc/Framework/Headers/WebRTC/RTCRtpCodecParameters.h",
>         "objc/Framework/Headers/WebRTC/RTCRtpEncodingParameters.h",
>+        "objc/Framework/Headers/WebRTC/RTCRtpHeaderExtension.h",
>         "objc/Framework/Headers/WebRTC/RTCRtpParameters.h",
>         "objc/Framework/Headers/WebRTC/RTCRtpReceiver.h",
>         "objc/Framework/Headers/WebRTC/RTCRtpSender.h",
>+        "objc/Framework/Headers/WebRTC/RTCRtpTransceiver.h",
>         "objc/Framework/Headers/WebRTC/RTCSSLAdapter.h",
>         "objc/Framework/Headers/WebRTC/RTCSessionDescription.h",
>         "objc/Framework/Headers/WebRTC/RTCTracing.h",
>-        "objc/Framework/Headers/WebRTC/RTCVideoCapturer.h",
>-        "objc/Framework/Headers/WebRTC/RTCVideoFrame.h",
>-        "objc/Framework/Headers/WebRTC/RTCVideoFrameBuffer.h",
>-        "objc/Framework/Headers/WebRTC/RTCVideoRenderer.h",
>-        "objc/Framework/Headers/WebRTC/RTCVideoSource.h",
>         "objc/Framework/Headers/WebRTC/RTCVideoTrack.h",
>-        "objc/Framework/Headers/WebRTC/RTCVideoViewShading.h",
>       ]
> 
>-      configs += [ "..:common_objc" ]
>-
>+      configs += [
>+        "..:common_objc",
>+        ":used_from_extension",
>+      ]
>       public_configs = [ ":common_config_objc" ]
> 
>-      if (!build_with_chromium && is_clang) {
>-        # Suppress warnings from the Chromium Clang plugin
>-        # (bugs.webrtc.org/163).
>-        suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-      }
>-
>       deps = [
>         ":common_objc",
>-        ":corevideoframebuffer_objc",
>-        ":videotracksource_objc",
>+        ":mediaconstraints_objc",
>+        ":mediasource_objc",
>+        ":native_api",
>+        ":native_video",
>+        ":video_objc",
>+        ":videocodec_objc",
>+        ":videoframebuffer_objc",
>+        ":videorenderer_objc",
>+        ":videorendereradapter_objc",
>+        ":videosource_objc",
>         "../api:libjingle_peerconnection_api",
>-        "../api:peerconnection_and_implicit_call_api",
>-        "../api:video_frame_api",
>+        "../api/audio_codecs:audio_codecs_api",
>+        "../api/audio_codecs:builtin_audio_decoder_factory",
>+        "../api/audio_codecs:builtin_audio_encoder_factory",
>+        "../api/video:video_frame",
>         "../api/video_codecs:video_codecs_api",
>         "../common_video",
>         "../media:rtc_media_base",
>         "../modules:module_api",
>+        "../modules/audio_device:audio_device_api",
>+        "../modules/audio_processing:audio_processing",
>         "../modules/video_coding:video_codec_interface",
>+        "../pc:create_pc_factory",
>         "../pc:peerconnection",
>         "../rtc_base:checks",
>         "../rtc_base:rtc_base",
>+        "//third_party/abseil-cpp/absl/memory",
>       ]
>+
>+      if (is_ios) {
>+        deps += [ ":native_api_audio_device_module" ]
>+      }
>     }
> 
>     if (rtc_include_tests) {
>@@ -546,43 +776,71 @@ if (is_ios || is_mac) {
>           ]
> 
>           sources = [
>+            "objc/Framework/UnitTests/ObjCVideoTrackSource_xctest.mm",
>+            "objc/Framework/UnitTests/RTCCVPixelBuffer_xctest.mm",
>+            "objc/Framework/UnitTests/RTCCallbackLogger_xctest.m",
>             "objc/Framework/UnitTests/RTCDoNotPutCPlusPlusInFrameworkHeaders_xctest.m",
>             "objc/Framework/UnitTests/RTCFileVideoCapturer_xctest.mm",
>+            "objc/Framework/UnitTests/RTCH264ProfileLevelId_xctest.m",
>+            "objc/Framework/UnitTests/RTCPeerConnectionFactory_xctest.m",
>+            "objc/Framework/UnitTests/frame_buffer_helpers.h",
>+            "objc/Framework/UnitTests/frame_buffer_helpers.mm",
>           ]
> 
>-          if (current_cpu == "arm64" || use_ios_simulator) {
>-            sources += [ "objc/Framework/UnitTests/RTCMTLVideoView_xctest.mm" ]
>+          # TODO(peterhanspers): Reenable these tests on simulator.
>+          # See bugs.webrtc.org/7812
>+          if (!use_ios_simulator) {
>+            sources += [
>+              "objc/Framework/UnitTests/RTCAudioDeviceModule_xctest.mm",
>+              "objc/Framework/UnitTests/RTCAudioDevice_xctest.mm",
>+            ]
>           }
> 
>-          if (use_ios_simulator) {
>-            # Only include this file on simulator, as it's already
>-            # included in device builds.
>-            sources += [ "objc/Framework/Classes/Metal/RTCMTLVideoView.m" ]
>-            libs = [ "CoreVideo.framework" ]
>-          }
>           deps = [
>+            ":audio_device",
>+            ":audio_objc",
>             ":common_objc",
>             ":framework_objc",
>-            ":peerconnection_objc",
>-            ":peerconnectionfactory_objc",
>+            ":native_api",
>+            ":native_api_audio_device_module",
>+            ":native_video",
>+            ":videocapture_objc",
>+            ":videocodec_objc",
>+            ":videoframebuffer_objc",
>+            ":videosource_objc",
>             ":videotoolbox_objc",
>-            ":videotracksource_objc",
>-            "../../system_wrappers:system_wrappers_default",
>+            "../api/video:video_frame_i420",
>+            "../common_video:common_video",
>             "../media:rtc_media_base",
>+            "../media:rtc_media_tests_utils",
>             "../modules:module_api",
>             "../rtc_base:rtc_base",
>             "../rtc_base:rtc_base_tests_utils",
>+            "../system_wrappers:system_wrappers",
>+            "../system_wrappers:system_wrappers_default",
>+            "//third_party/libyuv",
>           ]
>+
>+          if (rtc_use_metal_rendering) {
>+            sources += [ "objc/Framework/UnitTests/RTCMTLVideoView_xctest.m" ]
>+            deps += [ ":metal_objc" ]
>+          }
>+
>           public_deps = [
>             "//build/config/ios:xctest",
>             "//third_party/ocmock",
>           ]
>+
>           include_dirs += [ "$root_out_dir/WebRTC.framework/Headers/" ]
>         }
> 
>         bundle_data("sdk_unittests_bundle_data") {
>-          # Sample video taken from https://media.xiph.org/video/derf/
>           sources = [
>+            "objc/Framework/UnitTests/audio_short16.pcm",
>+            "objc/Framework/UnitTests/audio_short44.pcm",
>+            "objc/Framework/UnitTests/audio_short48.pcm",
>+
>+            # Sample video taken from https://media.xiph.org/video/derf/
>             "objc/Framework/UnitTests/foreman.mp4",
>           ]
>           outputs = [
>@@ -599,6 +857,7 @@ if (is_ios || is_mac) {
>           _bundle_id_suffix = ios_generic_test_bundle_id_suffix
>           extra_substitutions = [ "GTEST_BUNDLE_ID_SUFFIX=$_bundle_id_suffix" ]
>           deps = [
>+            ":peerconnectionfactory_base_objc",
>             ":sdk_unittests_bundle_data",
>             ":sdk_unittests_sources",
>           ]
>@@ -607,28 +866,26 @@ if (is_ios || is_mac) {
>       }
> 
>       # TODO(denicija): once all tests are migrated to xctest remove this source set.
>-      rtc_source_set("sdk_unittests_objc") {
>+      rtc_source_set("rtc_unittests_objc") {
>         testonly = true
> 
>         sources = [
>+          "objc/Framework/UnitTests/RTCCertificateTest.mm",
>           "objc/Framework/UnitTests/RTCConfigurationTest.mm",
>           "objc/Framework/UnitTests/RTCDataChannelConfigurationTest.mm",
>           "objc/Framework/UnitTests/RTCIceCandidateTest.mm",
>           "objc/Framework/UnitTests/RTCIceServerTest.mm",
>           "objc/Framework/UnitTests/RTCIntervalRangeTests.mm",
>           "objc/Framework/UnitTests/RTCMediaConstraintsTest.mm",
>+          "objc/Framework/UnitTests/RTCPeerConnectionFactoryBuilderTest.mm",
>           "objc/Framework/UnitTests/RTCPeerConnectionTest.mm",
>           "objc/Framework/UnitTests/RTCSessionDescriptionTest.mm",
>           "objc/Framework/UnitTests/RTCTracingTest.mm",
>-          "objc/Framework/UnitTests/avformatmappertests.mm",
>           "objc/Framework/UnitTests/objc_video_decoder_factory_tests.mm",
>           "objc/Framework/UnitTests/objc_video_encoder_factory_tests.mm",
>           "objc/Framework/UnitTests/scoped_cftyperef_tests.mm",
>         ]
>-        if (is_ios &&
>-            !(use_ios_simulator &&
>-              # The tests crash on these simulator versions:
>-              (ios_sdk_version == "10.0" || ios_sdk_version == "10.1"))) {
>+        if (is_ios) {
>           sources +=
>               [ "objc/Framework/UnitTests/RTCCameraVideoCapturerTests.mm" ]
>         }
>@@ -638,30 +895,37 @@ if (is_ios || is_mac) {
>         # https://developer.apple.com/library/mac/qa/qa1490/_index.html
>         ldflags = [ "-ObjC" ]
> 
>-        defines = [ "GTEST_RELATIVE_PATH" ]
>         deps = [
>           ":common_objc",
>-          ":peerconnection_objc",
>-          ":peerconnectionfactory_objc",
>+          ":mediaconstraints_objc",
>+          ":native_api",
>+          ":native_video",
>+          ":peerconnectionfactory_base_objc",
>+          ":video_objc",
>+          ":videocapture_objc",
>+          ":videocodec_objc",
>+          ":videoframebuffer_objc",
>+          ":videosource_objc",
>           ":videotoolbox_objc",
>-          ":videotracksource_objc",
>-          "../../system_wrappers:system_wrappers_default",
>+          "../api/audio_codecs:audio_codecs_api",
>+          "../api/audio_codecs:builtin_audio_decoder_factory",
>+          "../api/audio_codecs:builtin_audio_encoder_factory",
>+          "../api/video_codecs:video_codecs_api",
>           "../media:rtc_media_base",
>           "../modules:module_api",
>+          "../modules/audio_device:audio_device_api",
>+          "../modules/audio_processing:audio_processing",
>           "../modules/video_coding:video_codec_interface",
>           "../rtc_base:rtc_base_tests_utils",
>+          "../rtc_base/system:unused",
>+          "../system_wrappers:system_wrappers_default",
>           "//test:test_support",
>           "//third_party/ocmock",
>         ]
> 
>         if (is_ios) {
>           sources += [ "objc/Framework/UnitTests/RTCAudioSessionTest.mm" ]
>-        }
>-
>-        if (!build_with_chromium && is_clang) {
>-          # Suppress warnings from the Chromium Clang plugin
>-          # (bugs.webrtc.org/163).
>-          suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>+          deps += [ ":audio_objc" ]
>         }
>       }
>     }
>@@ -676,11 +940,11 @@ if (is_ios || is_mac) {
>           "objc/Framework/Headers/WebRTC/RTCVideoCodec.h",
>           "objc/Framework/Headers/WebRTC/RTCVideoCodecFactory.h",
>           "objc/Framework/Headers/WebRTC/RTCAudioSessionConfiguration.h",
>-          "objc/Framework/Headers/WebRTC/RTCAVFoundationVideoSource.h",
>           "objc/Framework/Headers/WebRTC/RTCAudioSource.h",
>           "objc/Framework/Headers/WebRTC/RTCAudioTrack.h",
>           "objc/Framework/Headers/WebRTC/RTCCameraVideoCapturer.h",
>           "objc/Framework/Headers/WebRTC/RTCCameraPreviewView.h",
>+          "objc/Framework/Headers/WebRTC/RTCCertificate.h",
>           "objc/Framework/Headers/WebRTC/RTCConfiguration.h",
>           "objc/Framework/Headers/WebRTC/RTCDataChannel.h",
>           "objc/Framework/Headers/WebRTC/RTCDataChannelConfiguration.h",
>@@ -700,24 +964,24 @@ if (is_ios || is_mac) {
>           "objc/Framework/Headers/WebRTC/RTCMediaStreamTrack.h",
>           "objc/Framework/Headers/WebRTC/RTCMetrics.h",
>           "objc/Framework/Headers/WebRTC/RTCMetricsSampleInfo.h",
>+          "objc/Framework/Headers/WebRTC/RTCMTLVideoView.h",
>           "objc/Framework/Headers/WebRTC/RTCPeerConnection.h",
>           "objc/Framework/Headers/WebRTC/RTCPeerConnectionFactory.h",
>           "objc/Framework/Headers/WebRTC/RTCPeerConnectionFactoryOptions.h",
>+          "objc/Framework/Headers/WebRTC/RTCRtcpParameters.h",
>           "objc/Framework/Headers/WebRTC/RTCRtpCodecParameters.h",
>           "objc/Framework/Headers/WebRTC/RTCRtpEncodingParameters.h",
>+          "objc/Framework/Headers/WebRTC/RTCRtpHeaderExtension.h",
>           "objc/Framework/Headers/WebRTC/RTCRtpParameters.h",
>           "objc/Framework/Headers/WebRTC/RTCRtpReceiver.h",
>           "objc/Framework/Headers/WebRTC/RTCRtpSender.h",
>+          "objc/Framework/Headers/WebRTC/RTCRtpTransceiver.h",
>           "objc/Framework/Headers/WebRTC/RTCDtmfSender.h",
>           "objc/Framework/Headers/WebRTC/RTCSSLAdapter.h",
>           "objc/Framework/Headers/WebRTC/RTCSessionDescription.h",
>           "objc/Framework/Headers/WebRTC/RTCTracing.h",
>           "objc/Framework/Headers/WebRTC/RTCVideoCapturer.h",
>           "objc/Framework/Headers/WebRTC/RTCVideoCodecH264.h",
>-          "objc/Framework/Headers/WebRTC/RTCVideoDecoderVP8.h",
>-          "objc/Framework/Headers/WebRTC/RTCVideoDecoderVP9.h",
>-          "objc/Framework/Headers/WebRTC/RTCVideoEncoderVP8.h",
>-          "objc/Framework/Headers/WebRTC/RTCVideoEncoderVP9.h",
>           "objc/Framework/Headers/WebRTC/RTCVideoFrame.h",
>           "objc/Framework/Headers/WebRTC/RTCVideoFrameBuffer.h",
>           "objc/Framework/Headers/WebRTC/RTCVideoRenderer.h",
>@@ -725,19 +989,22 @@ if (is_ios || is_mac) {
>           "objc/Framework/Headers/WebRTC/RTCVideoTrack.h",
>           "objc/Framework/Headers/WebRTC/RTCVideoViewShading.h",
>           "objc/Framework/Headers/WebRTC/UIDevice+RTCDevice.h",
>+          "objc/Framework/Headers/WebRTC/RTCVideoDecoderVP8.h",
>+          "objc/Framework/Headers/WebRTC/RTCVideoDecoderVP9.h",
>+          "objc/Framework/Headers/WebRTC/RTCVideoEncoderVP8.h",
>+          "objc/Framework/Headers/WebRTC/RTCVideoEncoderVP9.h",
>         ]
>-        if (rtc_use_metal_rendering) {
>-          common_objc_headers +=
>-              [ "objc/Framework/Headers/WebRTC/RTCMTLVideoView.h" ]
>-        }
>-        sources = common_objc_headers
>-        public_headers = common_objc_headers
> 
>         if (!build_with_chromium) {
>-          sources += [ "objc/Framework/Headers/WebRTC/RTCFileLogger.h" ]
>-          public_headers += [ "objc/Framework/Headers/WebRTC/RTCFileLogger.h" ]
>+          common_objc_headers += [
>+            "objc/Framework/Headers/WebRTC/RTCCallbackLogger.h",
>+            "objc/Framework/Headers/WebRTC/RTCFileLogger.h",
>+          ]
>         }
> 
>+        sources = common_objc_headers
>+        public_headers = common_objc_headers
>+
>         ldflags = [
>           "-all_load",
>           "-install_name",
>@@ -746,13 +1013,23 @@ if (is_ios || is_mac) {
> 
>         deps = [
>           ":audio_objc",
>-          ":peerconnection_objc",
>+          ":common_objc",
>+          ":default_codec_factory_objc",
>+          ":native_api",
>+          ":native_video",
>+          ":peerconnectionfactory_base_objc",
>           ":ui_objc",
>+          ":videocapture_objc",
>+          ":videocodec_objc",
>+          ":videotoolbox_objc",
>           "../rtc_base:rtc_base_approved",
>           "../system_wrappers:field_trial_default",
>           "../system_wrappers:metrics_default",
>           "../system_wrappers:runtime_enabled_features_default",
>         ]
>+        if (rtc_use_metal_rendering) {
>+          deps += [ ":metal_objc" ]
>+        }
> 
>         libs = [
>           "AVFoundation.framework",
>@@ -761,42 +1038,94 @@ if (is_ios || is_mac) {
>           "GLKit.framework",
>         ]
> 
>-        configs += [ "..:common_objc" ]
>+        configs += [
>+          "..:common_objc",
>+          ":used_from_extension",
>+        ]
> 
>         public_configs = [ ":common_config_objc" ]
>-
>-        if (!build_with_chromium && is_clang) {
>-          # Suppress warnings from the Chromium Clang plugin
>-          # (bugs.webrtc.org/163).
>-          configs -= [ "//build/config/clang:find_bad_constructs" ]
>-        }
>       }
>     }
> 
>-    rtc_static_library("rtc_sdk_objc") {
>-      complete_static_lib = true
>+    rtc_static_library("wrapped_native_codec_objc") {
>+      sources = [
>+        "objc/Framework/Classes/PeerConnection/RTCWrappedNativeVideoDecoder.h",
>+        "objc/Framework/Classes/PeerConnection/RTCWrappedNativeVideoDecoder.mm",
>+        "objc/Framework/Classes/PeerConnection/RTCWrappedNativeVideoEncoder.h",
>+        "objc/Framework/Classes/PeerConnection/RTCWrappedNativeVideoEncoder.mm",
>+      ]
>+
>+      configs += [ "..:common_objc" ]
>+      public_configs = [ ":common_config_objc" ]
>+
>       deps = [
>-        ":peerconnection_objc",
>-        ":ui_objc",
>-        "../system_wrappers:field_trial_default",
>-        "../system_wrappers:metrics_default",
>-        "../system_wrappers:runtime_enabled_features_default",
>+        ":common_objc",
>+        ":videocodec_objc",
>+        "../api/video_codecs:video_codecs_api",
>+        "../media:rtc_media_base",
>       ]
>     }
> 
>-    rtc_static_library("corevideoframebuffer_objc") {
>+    # The native API is currently experimental and may change without notice.
>+    rtc_static_library("native_api") {
>+      visibility = [ "*" ]
>+      allow_poison = [
>+        "audio_codecs",  # TODO(bugs.webrtc.org/8396): Remove.
>+        "software_video_codecs",  # TODO(bugs.webrtc.org/7925): Remove.
>+      ]
>       sources = [
>-        "objc/Framework/Classes/Video/corevideo_frame_buffer.cc",
>-        "objc/Framework/Classes/Video/corevideo_frame_buffer.h",
>+        "objc/Framework/Native/api/video_capturer.h",
>+        "objc/Framework/Native/api/video_capturer.mm",
>+        "objc/Framework/Native/api/video_decoder_factory.h",
>+        "objc/Framework/Native/api/video_decoder_factory.mm",
>+        "objc/Framework/Native/api/video_encoder_factory.h",
>+        "objc/Framework/Native/api/video_encoder_factory.mm",
>+        "objc/Framework/Native/api/video_frame.h",
>+        "objc/Framework/Native/api/video_frame.mm",
>+        "objc/Framework/Native/api/video_frame_buffer.h",
>+        "objc/Framework/Native/api/video_frame_buffer.mm",
>+        "objc/Framework/Native/api/video_renderer.h",
>+        "objc/Framework/Native/api/video_renderer.mm",
>       ]
> 
>+      configs += [ "..:common_objc" ]
>+
>+      public_configs = [ ":common_config_objc" ]
>+
>       deps = [
>-        "../api:video_frame_api_i420",
>+        ":native_video",
>+        ":videocapturebase_objc",
>+        ":videocodec_objc",
>+        ":videoframebuffer_objc",
>+        ":videorenderer_objc",
>+        "../api:libjingle_peerconnection_api",
>+        "../api/video:video_frame",
>+        "../api/video_codecs:video_codecs_api",
>         "../common_video",
>-        "../rtc_base:checks",
>-        "../rtc_base:rtc_base_approved",
>-        "//third_party/libyuv",
>+        "../rtc_base:rtc_base",
>+        "//third_party/abseil-cpp/absl/memory",
>       ]
>+    }
>+
>+    rtc_static_library("native_video") {
>+      sources = [
>+        "objc/Framework/Native/src/objc_frame_buffer.h",
>+        "objc/Framework/Native/src/objc_frame_buffer.mm",
>+        "objc/Framework/Native/src/objc_video_decoder_factory.h",
>+        "objc/Framework/Native/src/objc_video_decoder_factory.mm",
>+        "objc/Framework/Native/src/objc_video_encoder_factory.h",
>+        "objc/Framework/Native/src/objc_video_encoder_factory.mm",
>+        "objc/Framework/Native/src/objc_video_frame.h",
>+        "objc/Framework/Native/src/objc_video_frame.mm",
>+        "objc/Framework/Native/src/objc_video_renderer.h",
>+        "objc/Framework/Native/src/objc_video_renderer.mm",
>+        "objc/Framework/Native/src/objc_video_track_source.h",
>+        "objc/Framework/Native/src/objc_video_track_source.mm",
>+      ]
>+
>+      configs += [ "..:common_objc" ]
>+
>+      public_configs = [ ":common_config_objc" ]
> 
>       if (!build_with_chromium && is_clang) {
>         # Suppress warnings from the Chromium Clang plugin
>@@ -804,7 +1133,25 @@ if (is_ios || is_mac) {
>         suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>       }
> 
>-      libs = [ "CoreVideo.framework" ]
>+      deps = [
>+        ":common_objc",
>+        ":videocapturebase_objc",
>+        ":videocodec_objc",
>+        ":videoframebuffer_objc",
>+        ":videorenderer_objc",
>+        ":wrapped_native_codec_objc",
>+        "../api/video:video_frame",
>+        "../api/video:video_frame_i420",
>+        "../api/video_codecs:video_codecs_api",
>+        "../common_video",
>+        "../media:rtc_audio_video",
>+        "../media:rtc_media_base",
>+        "../modules:module_api",
>+        "../modules/video_coding:video_codec_interface",
>+        "../rtc_base:checks",
>+        "../rtc_base:rtc_base",
>+        "//third_party/abseil-cpp/absl/memory",
>+      ]
>     }
> 
>     rtc_static_library("video_toolbox_cc") {
>@@ -822,45 +1169,47 @@ if (is_ios || is_mac) {
>         "../rtc_base:checks",
>         "../rtc_base:rtc_base_approved",
>       ]
>-      if (!build_with_chromium && is_clang) {
>-        # Suppress warnings from the Chromium Clang plugin
>-        # (bugs.webrtc.org/163).
>-        suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-      }
>     }
> 
>     rtc_static_library("videotoolbox_objc") {
>+      visibility = [ "*" ]
>+      allow_poison = [
>+        "audio_codecs",  # TODO(bugs.webrtc.org/8396): Remove.
>+        "software_video_codecs",  # TODO(bugs.webrtc.org/7925): Remove.
>+      ]
>       sources = [
>         "objc/Framework/Classes/VideoToolbox/RTCVideoDecoderH264.mm",
>         "objc/Framework/Classes/VideoToolbox/RTCVideoEncoderH264.mm",
>-        "objc/Framework/Classes/VideoToolbox/objc_video_decoder_factory.h",
>-        "objc/Framework/Classes/VideoToolbox/objc_video_decoder_factory.mm",
>-        "objc/Framework/Classes/VideoToolbox/objc_video_encoder_factory.h",
>-        "objc/Framework/Classes/VideoToolbox/objc_video_encoder_factory.mm",
>-        "objc/Framework/Headers/WebRTC/RTCVideoFrameBuffer.h",
>       ]
> 
>-      configs += [ "..:common_objc" ]
>+      configs += [
>+        "..:common_objc",
>+        ":used_from_extension",
>+      ]
>+
>+      if (is_ios && rtc_apprtcmobile_broadcast_extension) {
>+        defines = [ "RTC_APPRTCMOBILE_BROADCAST_EXTENSION" ]
>+      }
> 
>       deps = [
>         ":common_objc",
>+        ":native_api",
>         ":video_objc",
>         ":video_toolbox_cc",
>-        ":videotracksource_objc",
>-        "../api:video_frame_api",
>+        ":videocodec_objc",
>+        ":videoframebuffer_objc",
>+        ":videosource_objc",
>+        "../api/video:video_frame",
>         "../api/video_codecs:video_codecs_api",
>         "../common_video",
>         "../media:rtc_audio_video",
>-        "../media:rtc_internal_video_codecs",
>         "../media:rtc_media",
>         "../media:rtc_media_base",
>         "../modules:module_api",
>         "../modules/video_coding:video_codec_interface",
>         "../modules/video_coding:video_coding_utility",
>-        "../modules/video_coding:webrtc_h264",
>         "../rtc_base:checks",
>         "../rtc_base:rtc_base_approved",
>-        "../system_wrappers",
>         "//third_party/libyuv",
>       ]
> 
>@@ -870,12 +1219,6 @@ if (is_ios || is_mac) {
>         "CoreVideo.framework",
>         "VideoToolbox.framework",
>       ]
>-
>-      if (!build_with_chromium && is_clang) {
>-        # Suppress warnings from the Chromium Clang plugin
>-        # (bugs.webrtc.org/163).
>-        suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-      }
>     }
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/VideoProcessingSoftLink.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/VideoProcessingSoftLink.h
>index 4e5019dd4d8..cfa9dd6da96 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/VideoProcessingSoftLink.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/VideoProcessingSoftLink.h
>@@ -38,7 +38,7 @@
> #elif (defined(TARGET_OS_IPHONE)  && TARGET_OS_IPHONE)
> #define ENABLE_VCP_ENCODER 0
> #elif (defined(TARGET_OS_MAC)  && TARGET_OS_MAC)
>-#define ENABLE_VCP_ENCODER (__MAC_OS_X_VERSION_MAX_ALLOWED >= 101304)
>+#define ENABLE_VCP_ENCODER 0 //(__MAC_OS_X_VERSION_MAX_ALLOWED >= 101304)
> #endif
> 
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/VideoToolBoxDecoderFactory.cpp b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/VideoToolBoxDecoderFactory.cpp
>index 37de8f50070..e7307700080 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/VideoToolBoxDecoderFactory.cpp
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/VideoToolBoxDecoderFactory.cpp
>@@ -69,7 +69,7 @@ std::unique_ptr<VideoDecoder> VideoToolboxVideoDecoderFactory::CreateVideoDecode
> }
> 
> static inline SdpVideoFormat CreateH264Format(H264::Profile profile, H264::Level level) {
>-    const rtc::Optional<std::string> profile_string =
>+    const absl::optional<std::string> profile_string =
>     H264::ProfileLevelIdToString(H264::ProfileLevelId(profile, level));
>     RTC_CHECK(profile_string);
>     return SdpVideoFormat(cricket::kH264CodecName,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/VideoToolBoxEncoderFactory.cpp b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/VideoToolBoxEncoderFactory.cpp
>index f45ff01780d..296703f35e0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/VideoToolBoxEncoderFactory.cpp
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/VideoToolBoxEncoderFactory.cpp
>@@ -69,8 +69,7 @@ void VideoToolboxVideoEncoderFactory::setH264HardwareEncoderAllowed(bool allowed
> }
> 
> static inline SdpVideoFormat CreateH264Format(H264::Profile profile, H264::Level level) {
>-    const rtc::Optional<std::string> profile_string =
>-    H264::ProfileLevelIdToString(H264::ProfileLevelId(profile, level));
>+    const absl::optional<std::string> profile_string = H264::ProfileLevelIdToString(H264::ProfileLevelId(profile, level));
>     RTC_CHECK(profile_string);
>     return SdpVideoFormat(cricket::kH264CodecName,
>                           {{cricket::kH264FmtpProfileLevelId, *profile_string},
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/WebKitUtilities.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/WebKitUtilities.h
>index 776da18fbc1..1ad27c85260 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/WebKitUtilities.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/WebKitUtilities.h
>@@ -25,9 +25,10 @@
> 
> #pragma once
> 
>-#include "api/video_codecs/video_encoder_factory.h"
> #include "api/video/video_frame_buffer.h"
> #include "rtc_base/scoped_ref_ptr.h"
>+#include "webrtc/media/engine/webrtcvideodecoderfactory.h"
>+#include "webrtc/media/engine/webrtcvideoencoderfactory.h"
> 
> typedef struct __CVBuffer* CVPixelBufferRef;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/WebKitUtilities.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/WebKitUtilities.mm
>index 5e65230f321..240a174534d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/WebKitUtilities.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/WebKitUtilities.mm
>@@ -24,13 +24,17 @@
>  */
> 
> #include "WebKitUtilities.h"
>+
> #include "Common/RTCUIApplicationStatusObserver.h"
>-#include "VideoToolbox/objc_video_decoder_factory.h"
>-#include "VideoToolbox/objc_video_encoder_factory.h"
> #import "WebRTC/RTCVideoCodecH264.h"
>+
> #include "api/video/video_frame.h"
>-#include <webrtc/sdk/objc/Framework/Classes/Video/objc_frame_buffer.h>
>-#include <webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoFrameBuffer.h>
>+#include "webrtc/sdk/objc/Framework/Native/src/objc_frame_buffer.h"
>+#include "webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoFrame.h"
>+#include "webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoFrameBuffer.h"
>+#include "webrtc/sdk/objc/Framework/Native/api/video_decoder_factory.h"
>+#include "webrtc/sdk/objc/Framework/Native/api/video_encoder_factory.h"
>+
> 
> #if !defined(WEBRTC_IOS)
> __attribute__((objc_runtime_name("WK_RTCUIApplicationStatusObserver")))
>@@ -99,12 +103,12 @@ std::unique_ptr<webrtc::VideoEncoderFactory> createVideoToolboxEncoderFactory()
>         webrtc::VPModuleInitialize();
>     });
> #endif
>-    return std::make_unique<webrtc::ObjCVideoEncoderFactory>([[RTCVideoEncoderFactoryH264 alloc] init]);
>+    return ObjCToNativeVideoEncoderFactory([[RTCVideoEncoderFactoryH264 alloc] init]);
> }
> 
> std::unique_ptr<webrtc::VideoDecoderFactory> createVideoToolboxDecoderFactory()
> {
>-    return std::make_unique<webrtc::ObjCVideoDecoderFactory>([[RTCVideoDecoderFactoryH264 alloc] init]);
>+    return ObjCToNativeVideoDecoderFactory([[RTCVideoDecoderFactoryH264 alloc] init]);
> }
> 
> static bool h264HardwareEncoderAllowed = true;
>@@ -120,17 +124,21 @@ bool isH264HardwareEncoderAllowed()
> 
> rtc::scoped_refptr<webrtc::VideoFrameBuffer> pixelBufferToFrame(CVPixelBufferRef pixelBuffer)
> {
>-    RTCCVPixelBuffer *frameBuffer = [[RTCCVPixelBuffer alloc] initWithPixelBuffer: pixelBuffer];
>-    return new rtc::RefCountedObject<webrtc::ObjCFrameBuffer>(frameBuffer);
>+    RTCCVPixelBuffer *frameBuffer = [[RTCCVPixelBuffer alloc] initWithPixelBuffer:pixelBuffer];
>+    return new rtc::RefCountedObject<ObjCFrameBuffer>(frameBuffer);
> }
> 
> CVPixelBufferRef pixelBufferFromFrame(const VideoFrame& frame)
> {
>-    auto buffer = frame.video_frame_buffer();
>-    auto frameBuffer = static_cast<webrtc::ObjCFrameBuffer&>(*buffer).wrapped_frame_buffer();
>-    if ([frameBuffer isKindOfClass: [RTCCVPixelBuffer class]])
>-        return [(RTCCVPixelBuffer *)frameBuffer pixelBuffer];
>-    return nullptr;
>+    if (frame.video_frame_buffer()->type() != VideoFrameBuffer::Type::kNative)
>+        return nullptr;
>+
>+    auto *frameBuffer = static_cast<ObjCFrameBuffer*>(frame.video_frame_buffer().get())->wrapped_frame_buffer();
>+    if (![frameBuffer isKindOfClass:[RTCCVPixelBuffer class]])
>+        return nullptr;
>+
>+    auto *rtcPixelBuffer = (RTCCVPixelBuffer *)frameBuffer;
>+    return rtcPixelBuffer.pixelBuffer;
> }
> 
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/decoder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/decoder.h
>index e0a7b03ef22..75b57dad79c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/decoder.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/decoder.h
>@@ -37,7 +37,6 @@ class H264VideoToolboxDecoder : public H264Decoder {
> 
>   int Decode(const EncodedImage& input_image,
>              bool missing_frames,
>-             const RTPFragmentationHeader* fragmentation,
>              const CodecSpecificInfo* codec_specific_info,
>              int64_t render_time_ms) override;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/decoder.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/decoder.mm
>index 664a99aae56..ca4586bb7b5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/decoder.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/decoder.mm
>@@ -99,7 +99,6 @@ int H264VideoToolboxDecoder::InitDecode(const VideoCodec* video_codec,
> int H264VideoToolboxDecoder::Decode(
>     const EncodedImage& input_image,
>     bool missing_frames,
>-    const RTPFragmentationHeader* fragmentation,
>     const CodecSpecificInfo* codec_specific_info,
>     int64_t render_time_ms) {
>   RTC_DCHECK(input_image._buffer);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/BUILD.gn
>index c7427da178b..3b735e0be40 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/BUILD.gn
>@@ -6,698 +6,1427 @@
> # in the file PATENTS.  All contributing project authors may
> # be found in the AUTHORS file in the root of the source tree.
> 
>-import("../../webrtc.gni")
>-import("//build/config/android/config.gni")
>-import("//build/config/android/rules.gni")
>-
>-group("android") {
>-  if (!build_with_chromium && is_android) {
>-    public_deps = [
>-      ":libjingle_peerconnection_datachannelonly_so",
>+if (is_android) {
>+  import("../../webrtc.gni")
>+  import("//build/config/android/config.gni")
>+  import("//build/config/android/rules.gni")
>+
>+  group("android") {
>+    if (!build_with_chromium && is_android) {
>+      public_deps = [
>+        ":libjingle_peerconnection_datachannelonly_so",
>+        ":libjingle_peerconnection_jni",
>+        ":libjingle_peerconnection_so",
>+        ":libwebrtc",
>+        ":native_api",
>+      ]
>+    }
>+  }
>+
>+  #####################
>+  # Aggregate targets #
>+  #####################
>+
>+  dist_jar("libwebrtc") {
>+    _target_dir_name = get_label_info(":$target_name", "dir")
>+    output = "${root_out_dir}/lib.java${_target_dir_name}/${target_name}.jar"
>+    direct_deps_only = true
>+    use_unprocessed_jars = true
>+    requires_android = true
>+    no_build_hooks = true
>+
>+    deps = [
>+      ":audio_api_java",
>+      ":base_java",
>+      ":camera_java",
>+      ":default_video_codec_factory_java",
>+      ":filevideo_java",
>+      ":hwcodecs_java",
>+      ":java_audio_device_module_java",
>+      ":libjingle_peerconnection_java",
>+      ":libjingle_peerconnection_metrics_default_java",
>+      ":logging_java",
>+      ":peerconnection_java",
>+      ":screencapturer_java",
>+      ":surfaceviewrenderer_java",
>+      ":swcodecs_java",
>+      ":video_api_java",
>+      ":video_java",
>+      "../../modules/audio_device:audio_device_java",
>+      "../../rtc_base:base_java",
>+    ]
>+  }
>+
>+  # The native API is currently experimental and may change without notice.
>+  group("native_api") {
>+    deps = [
>+      ":native_api_audio_device_module",
>+      ":native_api_base",
>+      ":native_api_codecs",
>+      ":native_api_jni",
>+      ":native_api_peerconnection",
>+      ":native_api_video",
>+    ]
>+  }
>+
>+  # Old target that pulls in everything. This will be going away in the future,
>+  # clients should depend on individual video_java etc. targets instead.
>+  rtc_android_library("libjingle_peerconnection_java") {
>+    java_files = [ "src/java/org/webrtc/Empty.java" ]
>+
>+    deps = [
>+      ":audio_api_java",
>+      ":base_java",
>+      ":camera_java",
>+      ":filevideo_java",
>+      ":hwcodecs_java",
>+      ":java_audio_device_module_java",
>+      ":peerconnection_java",
>+      ":screencapturer_java",
>+      ":surfaceviewrenderer_java",
>+      ":video_api_java",
>+      ":video_java",
>+      "//modules/audio_device:audio_device_java",
>+      "//rtc_base:base_java",
>+    ]
>+  }
>+
>+  rtc_android_library("libjingle_peerconnection_metrics_default_java") {
>+    java_files = [ "api/org/webrtc/Metrics.java" ]
>+
>+    deps = [
>+      ":base_java",
>+      ":libjingle_peerconnection_java",
>+      "../../rtc_base:base_java",
>+    ]
>+  }
>+
>+  rtc_static_library("libjingle_peerconnection_jni") {
>+    visibility = [ "*" ]
>+    allow_poison = [
>+      "audio_codecs",  # TODO(bugs.webrtc.org/8396): Remove.
>+      "software_video_codecs",  # TODO(bugs.webrtc.org/7925): Remove.
>+    ]
>+    public_deps = [  # no-presubmit-check TODO(webrtc:8603)
>+      ":audio_jni",
>+      ":base_jni",
>+      ":java_audio_device_module_jni",
>+      ":media_jni",
>+      ":peerconnection_jni",
>+      ":video_jni",
>+      "../../pc:create_pc_factory",
>+    ]
>+  }
>+
>+  rtc_static_library("libjingle_peerconnection_metrics_default_jni") {
>+    visibility = [ "*" ]
>+
>+    allow_poison = [ "software_video_codecs" ]  # TODO(bugs.webrtc.org/7925): Remove.
>+
>+    sources = [
>+      "src/jni/androidmetrics.cc",
>+    ]
>+
>+    deps = [
>+      ":base_jni",
>+      ":generated_metrics_jni",
>+      ":native_api_jni",
>+      ":peerconnection_jni",
>+      "../../pc:peerconnection",
>+      "../../system_wrappers:field_trial_default",
>+      "../../system_wrappers:metrics_api",
>+      "../../system_wrappers:metrics_default",
>+      "../../system_wrappers:runtime_enabled_features_default",
>+    ]
>+  }
>+
>+  rtc_shared_library("libjingle_peerconnection_so") {
>+    sources = [
>+      "src/jni/jni_onload.cc",
>+    ]
>+
>+    suppressed_configs += [ "//build/config/android:hide_all_but_jni_onload" ]
>+    configs += [ "//build/config/android:hide_all_but_jni" ]
>+
>+    deps = [
>       ":libjingle_peerconnection_jni",
>-      ":libjingle_peerconnection_so",
>-      ":libwebrtc",
>+      ":libjingle_peerconnection_metrics_default_jni",
>+      "../../pc:libjingle_peerconnection",
>+      "../../rtc_base:rtc_base",
>     ]
>+    output_extension = "so"
>   }
>-}
> 
>-config("libjingle_peerconnection_jni_warnings_config") {
>-  # The warnings below are enabled by default. Since GN orders compiler flags
>-  # for a target before flags from configs, the only way to disable such
>-  # warnings is by having them in a separate config, loaded from the target.
>-  if (!is_win) {
>-    cflags = [
>-      "-Wno-sign-compare",
>-      "-Wno-unused-variable",
>+  # The modular build targets can be used to build WebRTC with different
>+  # functionalities. The users can choose either the real implemenation or the
>+  # null implementation of the audio/video modules based on their requirments.
>+  rtc_shared_library("libjingle_peerconnection_datachannelonly_so") {
>+    sources = [
>+      "src/jni/jni_onload.cc",
>     ]
>+
>+    suppressed_configs += [ "//build/config/android:hide_all_but_jni_onload" ]
>+    configs += [ "//build/config/android:hide_all_but_jni" ]
>+
>+    deps = [
>+      ":base_jni",
>+      ":libjingle_peerconnection_metrics_default_jni",
>+      ":null_audio_jni",
>+      ":null_media_jni",
>+      ":null_video_jni",
>+      ":peerconnection_jni",
>+      "../../pc:peerconnection",
>+      "../../rtc_base:rtc_base",
>+      "../../rtc_base:rtc_base_approved",
>+    ]
>+    output_extension = "so"
>   }
>-}
> 
>-generate_jni("generated_base_jni") {
>-  sources = [
>-    "src/java/org/webrtc/WebRtcClassLoader.java",
>-  ]
>-  jni_package = ""
>-  jni_generator_include = "//sdk/android/src/jni/jni_generator_helper.h"
>-}
>+  #######################
>+  # Public Java modules #
>+  #######################
> 
>-generate_jar_jni("generated_external_classes_jni") {
>-  classes = [
>-    "java/lang/Integer.class",
>-    "java/lang/Double.class",
>-    "java/lang/Long.class",
>-    "java/lang/Boolean.class",
>-    "java/math/BigInteger.class",
>-    "java/lang/String.class",
>-  ]
>-  jni_package = ""
>-  jni_generator_include = "//sdk/android/src/jni/jni_generator_helper.h"
>-}
>+  # Core targets.
> 
>-rtc_source_set("base_jni") {
>-  sources = [
>-    "src/jni/androidhistogram_jni.cc",
>-    "src/jni/class_loader.cc",
>-    "src/jni/class_loader.h",
>-    "src/jni/classreferenceholder.cc",
>-    "src/jni/classreferenceholder.h",
>-    "src/jni/jni_common.cc",
>-    "src/jni/jni_generator_helper.cc",
>-    "src/jni/jni_generator_helper.h",
>-    "src/jni/jni_helpers.cc",
>-    "src/jni/jni_helpers.h",
>-    "src/jni/pc/audio_jni.h",
>-    "src/jni/pc/media_jni.h",
>-    "src/jni/pc/video_jni.h",
>-  ]
>-
>-  deps = [
>-    ":generated_base_jni",
>-    ":generated_external_classes_jni",
>-    "../../api:libjingle_peerconnection_api",
>-    "../../api:optional",
>-    "../../rtc_base:rtc_base",
>-    "../../rtc_base:rtc_base_approved",
>-    "../../system_wrappers:metrics_api",
>-  ]
>-
>-  if (is_clang) {
>-    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>-    suppressed_configs += [
>-      "//build/config/clang:extra_warnings",
>-      "//build/config/clang:find_bad_constructs",
>+  # TODO(sakal): Extract files from this target to releveant subtargets, video, audio etc.
>+  rtc_android_library("base_java") {
>+    java_files = [
>+      "api/org/webrtc/RefCounted.java",
>+      "src/java/org/webrtc/CalledByNative.java",
>+      "src/java/org/webrtc/CalledByNativeUnchecked.java",
>+      "src/java/org/webrtc/Histogram.java",
>+      "src/java/org/webrtc/JniCommon.java",
>+      "src/java/org/webrtc/JniHelper.java",
>+      "src/java/org/webrtc/RefCountDelegate.java",
>+      "src/java/org/webrtc/WebRtcClassLoader.java",
>+    ]
>+
>+    deps = [
>+      "//rtc_base:base_java",
>     ]
>   }
>-}
> 
>-rtc_static_library("audio_jni") {
>-  sources = [
>-    "src/jni/pc/audio_jni.cc",
>-    "src/jni/pc/defaultaudioprocessingfactory_jni.cc",
>-  ]
>-
>-  deps = [
>-    ":base_jni",
>-    "../../api/audio_codecs:builtin_audio_decoder_factory",
>-    "../../api/audio_codecs:builtin_audio_encoder_factory",
>-    "../../modules/audio_processing:audio_processing",
>-    "../../rtc_base:rtc_base_approved",
>-    "../../voice_engine:voice_engine",
>-  ]
>-}
>+  rtc_android_library("audio_api_java") {
>+    java_files = [ "api/org/webrtc/audio/AudioDeviceModule.java" ]
> 
>-rtc_static_library("null_audio_jni") {
>-  sources = [
>-    "src/jni/pc/null_audio_jni.cc",
>-  ]
>+    deps = [
>+      ":base_java",
>+      "//rtc_base:base_java",
>+    ]
>+  }
> 
>-  deps = [
>-    ":base_jni",
>-  ]
>-}
>+  rtc_android_library("video_api_java") {
>+    java_files = [
>+      "api/org/webrtc/EncodedImage.java",
>+      "api/org/webrtc/VideoCodecInfo.java",
>+      "api/org/webrtc/CapturerObserver.java",
>+      "api/org/webrtc/VideoCodecStatus.java",
>+      "api/org/webrtc/VideoDecoder.java",
>+      "api/org/webrtc/VideoDecoderFactory.java",
>+      "api/org/webrtc/VideoEncoder.java",
>+      "api/org/webrtc/VideoEncoderFactory.java",
>+      "api/org/webrtc/VideoFrame.java",
>+      "api/org/webrtc/VideoSink.java",
>+    ]
> 
>-generate_jni("generated_video_jni") {
>-  sources = [
>-    "api/org/webrtc/EncodedImage.java",
>-    "api/org/webrtc/SurfaceTextureHelper.java",
>-    "api/org/webrtc/VideoCodecStatus.java",
>-    "api/org/webrtc/VideoDecoder.java",
>-    "api/org/webrtc/VideoEncoder.java",
>-    "api/org/webrtc/VideoFrame.java",
>-    "api/org/webrtc/VideoSink.java",
>-    "src/java/org/webrtc/VideoDecoderWrapper.java",
>-    "src/java/org/webrtc/VideoEncoderWrapper.java",
>-    "src/java/org/webrtc/WrappedNativeVideoDecoder.java",
>-    "src/java/org/webrtc/WrappedNativeVideoEncoder.java",
>-  ]
>-  jni_package = ""
>-  jni_generator_include = "//sdk/android/src/jni/jni_generator_helper.h"
>-}
>+    deps = [
>+      ":base_java",
>+      "//rtc_base:base_java",
>+    ]
>+  }
> 
>-rtc_static_library("video_jni") {
>-  sources = [
>-    "src/jni/androidmediacodeccommon.h",
>-    "src/jni/androidmediadecoder_jni.cc",
>-    "src/jni/androidmediadecoder_jni.h",
>-    "src/jni/androidmediaencoder_jni.cc",
>-    "src/jni/androidmediaencoder_jni.h",
>-    "src/jni/androidvideotracksource.cc",
>-    "src/jni/androidvideotracksource.h",
>-    "src/jni/androidvideotracksource_jni.cc",
>-    "src/jni/encodedimage.cc",
>-    "src/jni/encodedimage.h",
>-    "src/jni/hardwarevideoencoderfactory.cc",
>-    "src/jni/jni_generator_helper.h",
>-    "src/jni/nv12buffer_jni.cc",
>-    "src/jni/nv21buffer_jni.cc",
>-    "src/jni/pc/video_jni.cc",
>-    "src/jni/surfacetexturehelper_jni.cc",
>-    "src/jni/surfacetexturehelper_jni.h",
>-    "src/jni/video_renderer_jni.cc",
>-    "src/jni/videocodecinfo.cc",
>-    "src/jni/videocodecinfo.h",
>-    "src/jni/videocodecstatus.cc",
>-    "src/jni/videocodecstatus.h",
>-    "src/jni/videodecoderfactorywrapper.cc",
>-    "src/jni/videodecoderfactorywrapper.h",
>-    "src/jni/videodecoderfallback.cc",
>-    "src/jni/videodecoderwrapper.cc",
>-    "src/jni/videodecoderwrapper.h",
>-    "src/jni/videoencoderfactorywrapper.cc",
>-    "src/jni/videoencoderfactorywrapper.h",
>-    "src/jni/videoencoderfallback.cc",
>-    "src/jni/videoencoderwrapper.cc",
>-    "src/jni/videoencoderwrapper.h",
>-    "src/jni/videofilerenderer_jni.cc",
>-    "src/jni/videoframe.cc",
>-    "src/jni/videoframe.h",
>-    "src/jni/videotrack_jni.cc",
>-    "src/jni/vp8codec.cc",
>-    "src/jni/vp9codec.cc",
>-    "src/jni/wrapped_native_i420_buffer.cc",
>-    "src/jni/wrapped_native_i420_buffer.h",
>-    "src/jni/wrappednativecodec.cc",
>-    "src/jni/wrappednativecodec.h",
>-    "src/jni/yuvhelper.cc",
>-  ]
>-
>-  configs += [ ":libjingle_peerconnection_jni_warnings_config" ]
>-
>-  if (is_clang) {
>-    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>-    suppressed_configs += [
>-      "//build/config/clang:extra_warnings",
>-      "//build/config/clang:find_bad_constructs",
>-    ]
>-  }
>-
>-  # TODO(jschuh): Bug 1348: fix this warning.
>-  configs += [ "//build/config/compiler:no_size_t_to_int_warning" ]
>-
>-  if (is_win) {
>-    cflags += [
>-      "/wd4245",  # conversion from "int" to "size_t", signed/unsigned mismatch.
>-      "/wd4389",  # signed/unsigned mismatch.
>-    ]
>-  }
>-
>-  deps = [
>-    ":base_jni",
>-    ":generated_video_jni",
>-    ":peerconnection_jni",
>-    "../..:webrtc_common",
>-    "../../api:libjingle_peerconnection_api",
>-    "../../api:video_frame_api",
>-    "../../api/video_codecs:video_codecs_api",
>-    "../../common_video:common_video",
>-    "../../media:rtc_audio_video",
>-    "../../media:rtc_media_base",
>-    "../../modules:module_api",
>-    "../../modules/utility:utility",
>-    "../../modules/video_coding:codec_globals_headers",
>-    "../../modules/video_coding:video_coding_utility",
>-    "../../modules/video_coding:webrtc_vp8",
>-    "../../modules/video_coding:webrtc_vp9",
>-    "../../rtc_base:rtc_base",
>-    "../../rtc_base:rtc_base_approved",
>-    "../../rtc_base:rtc_task_queue",
>-    "../../rtc_base:sequenced_task_checker",
>-    "../../rtc_base:weak_ptr",
>-    "../../system_wrappers:system_wrappers",
>-  ]
>-
>-  if (rtc_build_libyuv) {
>-    deps += [ "$rtc_libyuv_dir" ]
>-    public_deps = [
>-      "$rtc_libyuv_dir",
>-    ]
>-  } else {
>-    # Need to add a directory normally exported by libyuv.
>-    include_dirs = [ "$rtc_libyuv_dir/include" ]
>+  rtc_android_library("video_java") {
>+    java_files = [
>+      "api/org/webrtc/EglBase.java",
>+      "api/org/webrtc/EglRenderer.java",
>+      "api/org/webrtc/GlRectDrawer.java",
>+      "api/org/webrtc/GlShader.java",
>+      "api/org/webrtc/GlTextureFrameBuffer.java",
>+      "api/org/webrtc/GlUtil.java",
>+      "api/org/webrtc/JavaI420Buffer.java",
>+      "api/org/webrtc/RendererCommon.java",
>+      "api/org/webrtc/SurfaceTextureHelper.java",
>+      "api/org/webrtc/TextureBufferImpl.java",
>+      "api/org/webrtc/VideoCapturer.java",
>+      "api/org/webrtc/VideoDecoderFallback.java",
>+      "api/org/webrtc/VideoEncoderFallback.java",
>+      "api/org/webrtc/VideoFrameDrawer.java",
>+      "api/org/webrtc/YuvConverter.java",
>+      "api/org/webrtc/YuvHelper.java",
>+      "src/java/org/webrtc/EglBase10.java",
>+      "src/java/org/webrtc/EglBase14.java",
>+      "src/java/org/webrtc/GlGenericDrawer.java",
>+      "src/java/org/webrtc/H264Utils.java",
>+      "src/java/org/webrtc/NV21Buffer.java",
>+      "src/java/org/webrtc/VideoDecoderWrapper.java",
>+      "src/java/org/webrtc/VideoEncoderWrapper.java",
>+      "src/java/org/webrtc/WrappedNativeI420Buffer.java",
>+      "src/java/org/webrtc/WrappedNativeVideoDecoder.java",
>+      "src/java/org/webrtc/WrappedNativeVideoEncoder.java",
>+    ]
>+
>+    deps = [
>+      ":base_java",
>+      ":video_api_java",
>+      "//rtc_base:base_java",
>+    ]
>   }
>-}
> 
>-rtc_static_library("null_video_jni") {
>-  sources = [
>-    "src/jni/pc/null_video_jni.cc",
>-  ]
>+  rtc_android_library("peerconnection_java") {
>+    java_files = [
>+      "api/org/webrtc/audio/LegacyAudioDeviceModule.java",
>+      "api/org/webrtc/AudioProcessingFactory.java",
>+      "api/org/webrtc/AudioSource.java",
>+      "api/org/webrtc/AudioTrack.java",
>+      "api/org/webrtc/CallSessionFileRotatingLogSink.java",
>+      "api/org/webrtc/DataChannel.java",
>+      "api/org/webrtc/DtmfSender.java",
>+      "api/org/webrtc/FecControllerFactoryFactoryInterface.java",
>+      "api/org/webrtc/IceCandidate.java",
>+      "api/org/webrtc/MediaCodecVideoDecoder.java",
>+      "api/org/webrtc/MediaCodecVideoEncoder.java",
>+      "api/org/webrtc/MediaConstraints.java",
>+      "api/org/webrtc/MediaSource.java",
>+      "api/org/webrtc/MediaStream.java",
>+      "api/org/webrtc/MediaStreamTrack.java",
>+      "api/org/webrtc/NativeLibraryLoader.java",
>+      "api/org/webrtc/NativePeerConnectionFactory.java",
>+      "api/org/webrtc/NetworkMonitor.java",  # TODDO(sakal): Break dependencies and move to base_java.
>+      "api/org/webrtc/NetworkMonitorAutoDetect.java",  # TODDO(sakal): Break dependencies and move to base_java.
>+      "api/org/webrtc/PeerConnection.java",
>+      "api/org/webrtc/PeerConnectionDependencies.java",
>+      "api/org/webrtc/PeerConnectionFactory.java",
>+      "api/org/webrtc/RTCStats.java",
>+      "api/org/webrtc/RTCStatsCollectorCallback.java",
>+      "api/org/webrtc/RTCStatsReport.java",
>+      "api/org/webrtc/RtpParameters.java",
>+      "api/org/webrtc/RtpReceiver.java",
>+      "api/org/webrtc/RtpSender.java",
>+      "api/org/webrtc/RtpTransceiver.java",
>+      "api/org/webrtc/SSLCertificateVerifier.java",
>+      "api/org/webrtc/SdpObserver.java",
>+      "api/org/webrtc/SessionDescription.java",
>+      "api/org/webrtc/StatsObserver.java",
>+      "api/org/webrtc/StatsReport.java",
>+      "api/org/webrtc/TurnCustomizer.java",
>+      "api/org/webrtc/VideoSource.java",
>+      "api/org/webrtc/VideoTrack.java",
>+      "src/java/org/webrtc/NativeLibrary.java",
>+      "src/java/org/webrtc/NativeCapturerObserver.java",
>+    ]
> 
>-  deps = [
>-    ":base_jni",
>-  ]
>-}
>+    deps = [
>+      ":audio_api_java",
>+      ":base_java",
>+      ":default_video_codec_factory_java",
>+      ":logging_java",
>+      ":swcodecs_java",
>+      ":video_api_java",
>+      ":video_java",
>+      "//modules/audio_device:audio_device_java",
>+      "//rtc_base:base_java",
>+    ]
>+  }
> 
>-rtc_static_library("media_jni") {
>-  sources = [
>-    "src/jni/pc/media_jni.cc",
>-  ]
>+  # Modules, in alphabetical order.
> 
>-  deps = [
>-    ":base_jni",
>-    "../../api/video_codecs:video_codecs_api",
>-    "../../call:call_interfaces",
>-    "../../logging:rtc_event_log_api",
>-    "../../media:rtc_audio_video",
>-    "../../modules/audio_device:audio_device",
>-    "../../modules/audio_processing:audio_processing",
>-  ]
>+  rtc_android_library("camera_java") {
>+    java_files = [
>+      "api/org/webrtc/Camera1Capturer.java",
>+      "api/org/webrtc/Camera1Enumerator.java",
>+      "api/org/webrtc/Camera2Capturer.java",
>+      "api/org/webrtc/Camera2Enumerator.java",
>+      "api/org/webrtc/CameraEnumerationAndroid.java",
>+      "api/org/webrtc/CameraEnumerator.java",
>+      "api/org/webrtc/CameraVideoCapturer.java",
>+      "src/java/org/webrtc/Camera1Session.java",
>+      "src/java/org/webrtc/Camera2Session.java",
>+      "src/java/org/webrtc/CameraCapturer.java",
>+      "src/java/org/webrtc/CameraSession.java",
>+    ]
> 
>-  if (is_clang) {
>-    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>-    suppressed_configs += [
>-      "//build/config/clang:extra_warnings",
>-      "//build/config/clang:find_bad_constructs",
>+    deps = [
>+      ":base_java",
>+      ":video_api_java",
>+      ":video_java",
>+      "//rtc_base:base_java",
>     ]
>   }
>-}
> 
>-rtc_static_library("null_media_jni") {
>-  sources = [
>-    "src/jni/pc/null_media_jni.cc",
>-  ]
>+  rtc_android_library("default_video_codec_factory_java") {
>+    java_files = [
>+      "api/org/webrtc/DefaultVideoDecoderFactory.java",
>+      "api/org/webrtc/DefaultVideoEncoderFactory.java",
>+    ]
> 
>-  deps = [
>-    ":base_jni",
>-  ]
>+    deps = [
>+      ":hwcodecs_java",
>+      ":swcodecs_java",
>+      ":video_api_java",
>+      ":video_java",
>+    ]
>+  }
> 
>-  if (is_clang) {
>-    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>-    suppressed_configs += [
>-      "//build/config/clang:extra_warnings",
>-      "//build/config/clang:find_bad_constructs",
>+  rtc_android_library("filevideo_java") {
>+    java_files = [
>+      "api/org/webrtc/VideoFileRenderer.java",
>+      "api/org/webrtc/FileVideoCapturer.java",
>+    ]
>+
>+    deps = [
>+      ":base_java",
>+      ":video_api_java",
>+      ":video_java",
>+      "//rtc_base:base_java",
>     ]
>   }
>-}
> 
>-generate_jni("generated_peerconnection_jni") {
>-  sources = [
>-    "api/org/webrtc/DataChannel.java",
>-    "api/org/webrtc/IceCandidate.java",
>-    "api/org/webrtc/MediaConstraints.java",
>-    "api/org/webrtc/MediaStream.java",
>-    "api/org/webrtc/MediaStreamTrack.java",
>-    "api/org/webrtc/NetworkMonitor.java",
>-    "api/org/webrtc/NetworkMonitorAutoDetect.java",
>-    "api/org/webrtc/RTCStats.java",
>-    "api/org/webrtc/RTCStatsCollectorCallback.java",
>-    "api/org/webrtc/RTCStatsReport.java",
>-    "api/org/webrtc/StatsObserver.java",
>-    "api/org/webrtc/StatsReport.java",
>-  ]
>-  jni_package = ""
>-  jni_generator_include = "//sdk/android/src/jni/jni_generator_helper.h"
>-}
>+  rtc_android_library("hwcodecs_java") {
>+    java_files = [
>+      "api/org/webrtc/HardwareVideoDecoderFactory.java",
>+      "api/org/webrtc/HardwareVideoEncoderFactory.java",
>+      "src/java/org/webrtc/BaseBitrateAdjuster.java",
>+      "src/java/org/webrtc/BitrateAdjuster.java",
>+      "src/java/org/webrtc/DynamicBitrateAdjuster.java",
>+      "src/java/org/webrtc/FramerateBitrateAdjuster.java",
>+      "src/java/org/webrtc/HardwareVideoDecoder.java",
>+      "src/java/org/webrtc/HardwareVideoEncoder.java",
>+      "src/java/org/webrtc/MediaCodecWrapper.java",
>+      "src/java/org/webrtc/MediaCodecWrapperFactory.java",
>+      "src/java/org/webrtc/MediaCodecWrapperFactoryImpl.java",
>+      "src/java/org/webrtc/MediaCodecUtils.java",
>+      "src/java/org/webrtc/NV12Buffer.java",
>+      "src/java/org/webrtc/VideoCodecType.java",
>+    ]
> 
>-rtc_static_library("peerconnection_jni") {
>-  sources = [
>-    "src/jni/androidnetworkmonitor_jni.h",
>-    "src/jni/pc/androidnetworkmonitor_jni.cc",
>-    "src/jni/pc/androidnetworkmonitor_jni.h",
>-    "src/jni/pc/audiotrack_jni.cc",
>-    "src/jni/pc/callsessionfilerotatinglogsink_jni.cc",
>-    "src/jni/pc/datachannel.cc",
>-    "src/jni/pc/datachannel.h",
>-    "src/jni/pc/dtmfsender_jni.cc",
>-    "src/jni/pc/java_native_conversion.cc",
>-    "src/jni/pc/java_native_conversion.h",
>-    "src/jni/pc/logging_jni.cc",
>-    "src/jni/pc/mediaconstraints_jni.cc",
>-    "src/jni/pc/mediaconstraints_jni.h",
>-    "src/jni/pc/mediasource_jni.cc",
>-    "src/jni/pc/mediastream_jni.cc",
>-    "src/jni/pc/mediastreamtrack_jni.cc",
>-    "src/jni/pc/ownedfactoryandthreads.cc",
>-    "src/jni/pc/ownedfactoryandthreads.h",
>-    "src/jni/pc/peerconnection_jni.cc",
>-    "src/jni/pc/peerconnectionfactory_jni.cc",
>-    "src/jni/pc/peerconnectionobserver_jni.cc",
>-    "src/jni/pc/peerconnectionobserver_jni.h",
>-    "src/jni/pc/rtcstatscollectorcallbackwrapper.cc",
>-    "src/jni/pc/rtcstatscollectorcallbackwrapper.h",
>-    "src/jni/pc/rtpreceiver_jni.cc",
>-    "src/jni/pc/rtpreceiverobserver_jni.cc",
>-    "src/jni/pc/rtpreceiverobserver_jni.h",
>-    "src/jni/pc/rtpsender_jni.cc",
>-    "src/jni/pc/sdpobserver_jni.h",
>-    "src/jni/pc/statsobserver_jni.cc",
>-    "src/jni/pc/statsobserver_jni.h",
>-    "src/jni/pc/turncustomizer_jni.cc",
>-  ]
>-
>-  configs += [ ":libjingle_peerconnection_jni_warnings_config" ]
>-
>-  if (is_clang) {
>-    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>-    suppressed_configs += [
>-      "//build/config/clang:extra_warnings",
>-      "//build/config/clang:find_bad_constructs",
>-    ]
>-  }
>-
>-  # TODO(jschuh): Bug 1348: fix this warning.
>-  configs += [ "//build/config/compiler:no_size_t_to_int_warning" ]
>-
>-  if (is_win) {
>-    cflags += [
>-      "/wd4245",  # conversion from "int" to "size_t", signed/unsigned mismatch.
>-      "/wd4389",  # signed/unsigned mismatch.
>-    ]
>-  }
>-
>-  deps = [
>-    ":base_jni",
>-    ":generated_external_classes_jni",
>-    ":generated_peerconnection_jni",
>-    "../..:webrtc_common",
>-    "../../api/video_codecs:video_codecs_api",
>-    "../../media:rtc_data",
>-    "../../media:rtc_media_base",
>-    "../../modules/audio_device:audio_device",
>-    "../../modules/utility:utility",
>-    "../../pc:peerconnection",
>-    "../../rtc_base:rtc_base",
>-    "../../rtc_base:rtc_base_approved",
>-    "../../rtc_base:rtc_task_queue",
>-    "../../system_wrappers:system_wrappers",
>-  ]
>-}
>+    deps = [
>+      ":base_java",
>+      ":video_api_java",
>+      ":video_java",
>+      "//rtc_base:base_java",
>+    ]
>+  }
> 
>-rtc_static_library("libjingle_peerconnection_jni") {
>-  public_deps = [
>-    ":audio_jni",
>-    ":base_jni",
>-    ":media_jni",
>-    ":peerconnection_jni",
>-    ":video_jni",
>-    "../../pc:create_pc_factory",
>-  ]
>-}
>+  rtc_android_library("java_audio_device_module_java") {
>+    java_files = [
>+      "api/org/webrtc/audio/JavaAudioDeviceModule.java",
>+      "src/java/org/webrtc/audio/VolumeLogger.java",
>+      "src/java/org/webrtc/audio/WebRtcAudioEffects.java",
>+      "src/java/org/webrtc/audio/WebRtcAudioManager.java",
>+      "src/java/org/webrtc/audio/WebRtcAudioRecord.java",
>+      "src/java/org/webrtc/audio/WebRtcAudioTrack.java",
>+      "src/java/org/webrtc/audio/WebRtcAudioUtils.java",
>+    ]
> 
>-generate_jni("generated_metrics_jni") {
>-  sources = [
>-    "api/org/webrtc/Metrics.java",
>-  ]
>-  jni_package = ""
>-  jni_generator_include = "//sdk/android/src/jni/jni_generator_helper.h"
>-}
>+    deps = [
>+      ":audio_api_java",
>+      ":base_java",
>+      "//rtc_base:base_java",
>+    ]
>+  }
> 
>-rtc_static_library("libjingle_peerconnection_metrics_default_jni") {
>-  sources = [
>-    "src/jni/androidmetrics_jni.cc",
>-  ]
>-
>-  configs += [ ":libjingle_peerconnection_jni_warnings_config" ]
>-
>-  deps = [
>-    ":base_jni",
>-    ":generated_metrics_jni",
>-    ":peerconnection_jni",
>-    "../../pc:peerconnection",
>-    "../../system_wrappers",
>-    "../../system_wrappers:field_trial_default",
>-    "../../system_wrappers:metrics_default",
>-  ]
>-}
>+  rtc_android_library("screencapturer_java") {
>+    java_files = [ "api/org/webrtc/ScreenCapturerAndroid.java" ]
> 
>-# The modular build targets can be used to build WebRTC with different
>-# functionalities. The users can choose either the real implemenation or the
>-# null implementation of the audio/video modules based on their requirments.
>-rtc_shared_library("libjingle_peerconnection_datachannelonly_so") {
>-  sources = [
>-    "src/jni/jni_onload.cc",
>-  ]
>-
>-  suppressed_configs += [ "//build/config/android:hide_all_but_jni_onload" ]
>-  configs += [ "//build/config/android:hide_all_but_jni" ]
>-
>-  deps = [
>-    ":base_jni",
>-    ":libjingle_peerconnection_metrics_default_jni",
>-    ":null_audio_jni",
>-    ":null_media_jni",
>-    ":null_video_jni",
>-    ":peerconnection_jni",
>-    "../../pc:peerconnection",
>-    "../../rtc_base:rtc_base",
>-    "../../rtc_base:rtc_base_approved",
>-  ]
>-  output_extension = "so"
>-}
>+    deps = [
>+      ":video_api_java",
>+      ":video_java",
>+      "//rtc_base:base_java",
>+    ]
>+  }
> 
>-rtc_shared_library("libjingle_peerconnection_so") {
>-  sources = [
>-    "src/jni/jni_onload.cc",
>-  ]
>-
>-  suppressed_configs += [ "//build/config/android:hide_all_but_jni_onload" ]
>-  configs += [ "//build/config/android:hide_all_but_jni" ]
>-
>-  deps = [
>-    ":libjingle_peerconnection_jni",
>-    ":libjingle_peerconnection_metrics_default_jni",
>-    "../../pc:libjingle_peerconnection",
>-    "../../rtc_base:rtc_base",
>-  ]
>-  output_extension = "so"
>-}
>+  rtc_android_library("surfaceviewrenderer_java") {
>+    java_files = [
>+      "api/org/webrtc/SurfaceEglRenderer.java",
>+      "api/org/webrtc/SurfaceViewRenderer.java",
>+    ]
>+
>+    deps = [
>+      ":video_api_java",
>+      ":video_java",
>+      "//rtc_base:base_java",
>+    ]
>+  }
>+
>+  rtc_android_library("swcodecs_java") {
>+    java_files = [
>+      "api/org/webrtc/SoftwareVideoDecoderFactory.java",
>+      "api/org/webrtc/SoftwareVideoEncoderFactory.java",
>+      "src/java/org/webrtc/VP8Encoder.java",
>+      "src/java/org/webrtc/VP8Decoder.java",
>+      "src/java/org/webrtc/VP9Encoder.java",
>+      "src/java/org/webrtc/VP9Decoder.java",
>+    ]
>+
>+    deps = [
>+      ":base_java",
>+      ":video_api_java",
>+      ":video_java",
>+      "//rtc_base:base_java",
>+    ]
>+  }
>+
>+  ################################
>+  # JNI targets for Java modules #
>+  ################################
>+
>+  # Mirrors the order of targets in the section above.
>+
>+  rtc_source_set("base_jni") {
>+    visibility = [ "*" ]
>+    sources = [
>+      "src/jni/androidhistogram.cc",
>+      "src/jni/androidnetworkmonitor.cc",
>+      "src/jni/androidnetworkmonitor.h",
>+      "src/jni/class_loader.h",
>+      "src/jni/classreferenceholder.h",
>+      "src/jni/jni_common.cc",
>+      "src/jni/jni_helpers.cc",
>+      "src/jni/jni_helpers.h",
>+      "src/jni/pc/audio.h",
>+      "src/jni/pc/logging.cc",
>+      "src/jni/pc/media.h",
>+      "src/jni/pc/video.h",
>+    ]
>+
>+    deps = [
>+      ":generated_base_jni",
>+      ":internal_jni",
>+      ":native_api_jni",
>+      "../../api:libjingle_peerconnection_api",
>+      "../../rtc_base:checks",
>+      "../../rtc_base:rtc_base",
>+      "../../rtc_base:rtc_base_approved",
>+      "../../system_wrappers:metrics_api",
>+      "//third_party/abseil-cpp/absl/memory",
>+      "//third_party/abseil-cpp/absl/types:optional",
>+    ]
>+  }
>+
>+  rtc_static_library("audio_jni") {
>+    sources = [
>+      "src/jni/pc/audio.cc",
>+    ]
>+
>+    deps = [
>+      ":base_jni",
>+      "../../api/audio_codecs:builtin_audio_decoder_factory",
>+      "../../api/audio_codecs:builtin_audio_encoder_factory",
>+      "../../modules/audio_processing:audio_processing",
>+      "../../rtc_base:rtc_base_approved",
>+    ]
>+  }
>+
>+  rtc_static_library("video_jni") {
>+    sources = []
>+    deps = []
>+
>+    sources += [
>+      "src/jni/androidmediacodeccommon.h",
>+      "src/jni/androidmediadecoder.cc",
>+      "src/jni/androidmediaencoder.cc",
>+      "src/jni/androidvideotracksource.cc",
>+      "src/jni/androidvideotracksource.h",
>+      "src/jni/encodedimage.cc",
>+      "src/jni/encodedimage.h",
>+      "src/jni/h264utils.cc",
>+      "src/jni/javai420buffer.cc",
>+      "src/jni/jni_generator_helper.h",
>+      "src/jni/nativecapturerobserver.cc",
>+      "src/jni/nativecapturerobserver.h",
>+      "src/jni/nv12buffer.cc",
>+      "src/jni/nv21buffer.cc",
>+      "src/jni/pc/video.cc",
>+      "src/jni/videocodecinfo.cc",
>+      "src/jni/videocodecinfo.h",
>+      "src/jni/videocodecstatus.cc",
>+      "src/jni/videocodecstatus.h",
>+      "src/jni/videodecoderfactorywrapper.cc",
>+      "src/jni/videodecoderfactorywrapper.h",
>+      "src/jni/videodecoderfallback.cc",
>+      "src/jni/videodecoderwrapper.cc",
>+      "src/jni/videodecoderwrapper.h",
>+      "src/jni/videoencoderfactorywrapper.cc",
>+      "src/jni/videoencoderfactorywrapper.h",
>+      "src/jni/videoencoderfallback.cc",
>+      "src/jni/videoencoderwrapper.cc",
>+      "src/jni/videoencoderwrapper.h",
>+      "src/jni/videosink.cc",
>+      "src/jni/videosink.h",
>+      "src/jni/videosource.cc",
>+      "src/jni/videotrack.cc",
>+      "src/jni/yuvhelper.cc",
>+    ]
>+
>+    deps += [
>+      ":base_jni",
>+      ":generated_video_jni",
>+      ":native_api_jni",
>+      ":videoframe_jni",
>+      ":vp8_jni",  # TODO(bugs.webrtc.org/7925): Remove.
>+      ":vp9_jni",  # TODO(bugs.webrtc.org/7925): Remove.
>+      "../..:webrtc_common",
>+      "../../api:libjingle_peerconnection_api",
>+      "../../api/video:video_frame",
>+      "../../api/video_codecs:builtin_video_decoder_factory",
>+      "../../api/video_codecs:builtin_video_encoder_factory",
>+      "../../api/video_codecs:rtc_software_fallback_wrappers",
>+      "../../api/video_codecs:video_codecs_api",
>+      "../../common_video:common_video",
>+      "../../media:rtc_audio_video",
>+      "../../media:rtc_h264_profile_id",
>+      "../../media:rtc_internal_video_codecs",
>+      "../../media:rtc_media_base",
>+      "../../modules:module_api",
>+      "../../modules/utility:utility",
>+      "../../modules/video_coding:codec_globals_headers",
>+      "../../modules/video_coding:video_codec_interface",
>+      "../../modules/video_coding:video_coding_utility",
>+      "../../rtc_base:checks",
>+      "../../rtc_base:rtc_base",
>+      "../../rtc_base:rtc_base_approved",
>+      "../../rtc_base:rtc_task_queue",
>+      "../../rtc_base:sequenced_task_checker",
>+      "../../rtc_base:weak_ptr",
>+      "../../rtc_base/memory:aligned_malloc",
>+      "../../system_wrappers:field_trial_api",
>+      "//third_party/libyuv",
>+    ]
>+  }
>+
>+  rtc_static_library("peerconnection_jni") {
>+    # Do not depend on this target externally unless you absolute have to. It is
>+    # made public because we don't have a proper NDK yet. Header APIs here are not
>+    # considered public and are subject to change.
>+    visibility = [ "*" ]
>+
>+    sources = [
>+      "src/jni/androidnetworkmonitor_jni.h",
>+      "src/jni/pc/androidnetworkmonitor.h",
>+      "src/jni/pc/audiotrack.cc",
>+      "src/jni/pc/callsessionfilerotatinglogsink.cc",
>+      "src/jni/pc/datachannel.cc",
>+      "src/jni/pc/datachannel.h",
>+      "src/jni/pc/dtmfsender.cc",
>+      "src/jni/pc/icecandidate.cc",
>+      "src/jni/pc/icecandidate.h",
>+      "src/jni/pc/mediaconstraints.cc",
>+      "src/jni/pc/mediaconstraints.h",
>+      "src/jni/pc/mediasource.cc",
>+      "src/jni/pc/mediastream.cc",
>+      "src/jni/pc/mediastream.h",
>+      "src/jni/pc/mediastreamtrack.cc",
>+      "src/jni/pc/mediastreamtrack.h",
>+      "src/jni/pc/ownedfactoryandthreads.cc",
>+      "src/jni/pc/ownedfactoryandthreads.h",
>+      "src/jni/pc/peerconnection.cc",
>+      "src/jni/pc/peerconnection.h",
>+      "src/jni/pc/peerconnectionfactory.cc",
>+      "src/jni/pc/peerconnectionfactory.h",
>+      "src/jni/pc/rtcstatscollectorcallbackwrapper.cc",
>+      "src/jni/pc/rtcstatscollectorcallbackwrapper.h",
>+      "src/jni/pc/rtpparameters.cc",
>+      "src/jni/pc/rtpparameters.h",
>+      "src/jni/pc/rtpreceiver.cc",
>+      "src/jni/pc/rtpreceiver.h",
>+      "src/jni/pc/rtpsender.cc",
>+      "src/jni/pc/rtpsender.h",
>+      "src/jni/pc/rtptransceiver.cc",
>+      "src/jni/pc/rtptransceiver.h",
>+      "src/jni/pc/sdpobserver.cc",
>+      "src/jni/pc/sdpobserver.h",
>+      "src/jni/pc/sessiondescription.cc",
>+      "src/jni/pc/sessiondescription.h",
>+      "src/jni/pc/sslcertificateverifierwrapper.cc",
>+      "src/jni/pc/sslcertificateverifierwrapper.h",
>+      "src/jni/pc/statsobserver.cc",
>+      "src/jni/pc/statsobserver.h",
>+      "src/jni/pc/turncustomizer.cc",
>+      "src/jni/pc/turncustomizer.h",
>+    ]
> 
>-template("java8_dist_jar") {
>-  _deps = []
>-  _output_jars = []
>-  foreach(dep, invoker.deps) {
>-    _dep_name = get_label_info(dep, "name")
>-    _source_jar = get_label_info(dep, "target_gen_dir") + "/" + _dep_name +
>-                  "__compile_java.javac.jar"
>-    _output_jar = "$target_gen_dir/" + _dep_name + ".jar"
>+    deps = [
>+      ":base_jni",
>+      ":generated_external_classes_jni",
>+      ":generated_peerconnection_jni",
>+      ":logging_jni",
>+      ":native_api_jni",
>+      "../..:webrtc_common",
>+      "../../api:libjingle_peerconnection_api",
>+      "../../api/video_codecs:video_codecs_api",
>+      "../../logging:rtc_event_log_api",
>+      "../../logging:rtc_event_log_impl_base",
>+      "../../media:rtc_data",
>+      "../../media:rtc_media_base",
>+      "../../modules/audio_device:audio_device",
>+      "../../modules/utility:utility",
>+      "../../pc:peerconnection",
>+      "../../rtc_base:checks",
>+      "../../rtc_base:rtc_base",
>+      "../../rtc_base:rtc_base_approved",
>+      "../../rtc_base:rtc_task_queue",
>+      "../../rtc_base:stringutils",
>+      "../../system_wrappers:field_trial_api",
>+      "//third_party/abseil-cpp/absl/memory",
>+    ]
>+  }
> 
>-    _copy_dep = dep + "__compile_java__javac"
>-    _copy_target_name = "${target_name}_${_dep_name}_copy"
>+  # JNI target for java_audio_device_module_java
>+  rtc_source_set("java_audio_device_module_jni") {
>+    sources = [
>+      "src/jni/audio_device/java_audio_device_module.cc",
>+    ]
>+
>+    deps = [
>+      ":base_jni",
>+      ":generated_java_audio_jni",
>+      ":java_audio_device_module",
>+    ]
>+  }
> 
>-    copy(_copy_target_name) {
>+  ######################
>+  # Native API targets #
>+  ######################
>+
>+  # Core targets.
>+
>+  # JNI helpers that are also needed from internal JNI code. Cannot depend on any
>+  # other JNI targets than internal_jni.
>+  rtc_static_library("native_api_jni") {
>+    visibility = [ "*" ]
>+    sources = [
>+      "native_api/jni/class_loader.cc",
>+      "native_api/jni/java_types.cc",
>+      "native_api/jni/jvm.cc",
>+      "src/jni/jni_generator_helper.cc",
>+      "src/jni/jni_generator_helper.h",
>+    ]
>+
>+    public = [
>+      "native_api/jni/class_loader.h",
>+      "native_api/jni/java_types.h",
>+      "native_api/jni/jni_int_wrapper.h",
>+      "native_api/jni/jvm.h",
>+      "native_api/jni/scoped_java_ref.h",
>+    ]
>+
>+    deps = [
>+      ":generated_external_classes_jni",
>+      ":generated_native_api_jni",
>+      ":internal_jni",
>+      "//rtc_base:checks",
>+      "//rtc_base:rtc_base_approved",
>+      "//third_party/abseil-cpp/absl/types:optional",
>+    ]
>+  }
>+
>+  rtc_static_library("native_api_base") {
>+    visibility = [ "*" ]
>+    sources = [
>+      "native_api/base/init.cc",
>+      "native_api/base/init.h",
>+      "native_api/base/networkmonitor.cc",
>+      "native_api/base/networkmonitor.h",
>+    ]
>+
>+    deps = [
>+      ":base_jni",
>+      ":native_api_jni",
>+      "//rtc_base:checks",
>+      "//rtc_base:rtc_base",
>+      "//third_party/abseil-cpp/absl/memory",
>+    ]
>+  }
>+
>+  # Modules, in alphabetical order.
>+
>+  rtc_source_set("native_api_audio_device_module") {
>+    visibility = [ "*" ]
>+
>+    sources = [
>+      "native_api/audio_device_module/audio_device_android.cc",
>+      "native_api/audio_device_module/audio_device_android.h",
>+    ]
>+
>+    deps = [
>+      ":base_jni",
>+      ":java_audio_device_module",
>+      ":opensles_audio_device_module",
>+      "../../modules/audio_device:audio_device",
>+      "../../rtc_base:checks",
>+      "../../rtc_base:rtc_base_approved",
>+      "../../system_wrappers",
>+      "../../system_wrappers:metrics_api",
>+      "//third_party/abseil-cpp/absl/memory",
>+    ]
>+    if (rtc_enable_android_aaudio) {
>+      deps += [ ":aaudio_audio_device_module" ]
>+    }
>+  }
>+
>+  # API for wrapping Java VideoDecoderFactory/VideoEncoderFactory classes to C++
>+  # objects.
>+  rtc_static_library("native_api_codecs") {
>+    visibility = [ "*" ]
>+    allow_poison = [
>+      "audio_codecs",  # TODO(bugs.webrtc.org/8396): Remove.
>+      "software_video_codecs",  # TODO(bugs.webrtc.org/7925): Remove.
>+    ]
>+    sources = [
>+      "native_api/codecs/wrapper.cc",
>+      "native_api/codecs/wrapper.h",
>+    ]
>+
>+    deps = [
>+      ":base_jni",
>+      ":native_api_jni",
>+      ":video_jni",
>+      "//api/video_codecs:video_codecs_api",
>+      "//rtc_base:checks",
>+      "//rtc_base:rtc_base_approved",
>+      "//third_party/abseil-cpp/absl/memory",
>+    ]
>+  }
>+
>+  # API for creating Java PeerConnectionFactory from C++ equivalents.
>+  rtc_static_library("native_api_peerconnection") {
>+    visibility = [ "*" ]
>+
>+    allow_poison = [ "software_video_codecs" ]  # TODO(bugs.webrtc.org/7925): Remove.
>+
>+    sources = [
>+      "native_api/peerconnection/peerconnectionfactory.cc",
>+      "native_api/peerconnection/peerconnectionfactory.h",
>+    ]
>+
>+    deps = [
>+      ":base_jni",
>+      ":peerconnection_jni",
>+      "//api:libjingle_peerconnection_api",
>+      "//api/video_codecs:video_codecs_api",
>+      "//rtc_base:rtc_base",
>+    ]
>+  }
>+
>+  # API for creating C++ wrapper implementations of api/mediastreaminterface.h
>+  # video interfaces from their Java equivalents.
>+  rtc_static_library("native_api_video") {
>+    visibility = [ "*" ]
>+    allow_poison = [
>+      "audio_codecs",  # TODO(bugs.webrtc.org/8396): Remove.
>+      "software_video_codecs",  # TODO(bugs.webrtc.org/7925): Remove.
>+    ]
>+    sources = [
>+      "native_api/video/videosource.cc",
>+      "native_api/video/videosource.h",
>+      "native_api/video/wrapper.cc",
>+      "native_api/video/wrapper.h",
>+    ]
>+    deps = [
>+      ":native_api_jni",
>+      ":video_jni",
>+      ":videoframe_jni",
>+      "//api:libjingle_peerconnection_api",
>+      "//api/video:video_frame",
>+      "//rtc_base:rtc_base_approved",
>+      "//third_party/abseil-cpp/absl/memory",
>+    ]
>+  }
>+
>+  ####################
>+  # Internal targets #
>+  ####################
>+
>+  rtc_android_library("logging_java") {
>+    java_files = [ "src/java/org/webrtc/JNILogging.java" ]
>+
>+    deps = [
>+      ":base_java",
>+      "//rtc_base:base_java",
>+    ]
>+  }
>+
>+  rtc_static_library("null_audio_jni") {
>+    sources = [
>+      "src/jni/pc/null_audio.cc",
>+    ]
>+
>+    deps = [
>+      ":base_jni",
>+    ]
>+  }
>+
>+  rtc_static_library("null_video_jni") {
>+    sources = [
>+      "src/jni/pc/null_video.cc",
>+    ]
>+
>+    deps = [
>+      ":base_jni",
>+      "../../api/video_codecs:video_codecs_api",
>+    ]
>+  }
>+
>+  rtc_static_library("null_media_jni") {
>+    sources = [
>+      "src/jni/pc/null_media.cc",
>+    ]
>+
>+    deps = [
>+      ":base_jni",
>+    ]
>+  }
>+
>+  rtc_static_library("media_jni") {
>+    sources = [
>+      "src/jni/pc/media.cc",
>+    ]
>+
>+    deps = [
>+      ":base_jni",
>+      "../../api:callfactory_api",
>+      "../../api/video_codecs:video_codecs_api",
>+      "../../call:call_interfaces",
>+      "../../logging:rtc_event_log_api",
>+      "../../media:rtc_audio_video",
>+      "../../modules/audio_device:audio_device",
>+      "../../modules/audio_processing:audio_processing",
>+    ]
>+  }
>+
>+  # Internal code that is needed by native_api_jni. The code cannot be placed in
>+  # base_jni because native_api_jni depends on the code (and base_jni depends on
>+  # native_api_jni).
>+  rtc_source_set("internal_jni") {
>+    sources = [
>+      "src/jni/jvm.cc",
>+      "src/jni/jvm.h",
>+    ]
>+
>+    deps = [
>+      "../../rtc_base:checks",
>+    ]
>+  }
>+
>+  rtc_static_library("videoframe_jni") {
>+    sources = [
>+      "src/jni/videoframe.cc",
>+      "src/jni/videoframe.h",
>+      "src/jni/wrapped_native_i420_buffer.cc",
>+      "src/jni/wrapped_native_i420_buffer.h",
>+    ]
>+
>+    deps = [
>+      ":base_jni",
>+      ":generated_video_jni",
>+      ":native_api_jni",
>+      "../../api/video:video_frame",
>+      "../../common_video:common_video",
>+      "../../rtc_base:checks",
>+      "../../rtc_base:rtc_base",
>+      "../../rtc_base:rtc_base_approved",
>+    ]
>+  }
>+
>+  rtc_static_library("vp8_jni") {
>+    sources = [
>+      "src/jni/vp8codec.cc",
>+    ]
>+
>+    deps = [
>+      ":base_jni",
>+      ":generated_vp8_jni",
>+      "../../modules/video_coding:webrtc_vp8",
>+    ]
>+  }
>+
>+  rtc_static_library("vp9_jni") {
>+    sources = [
>+      "src/jni/vp9codec.cc",
>+    ]
>+
>+    deps = [
>+      ":base_jni",
>+      ":generated_vp9_jni",
>+      "../../modules/video_coding:webrtc_vp9",
>+    ]
>+  }
>+
>+  rtc_static_library("logging_jni") {
>+    visibility = [ "*" ]
>+    sources = [
>+      "src/jni/logging/logsink.cc",
>+      "src/jni/logging/logsink.h",
>+    ]
>+
>+    deps = [
>+      ":base_jni",
>+      ":generated_logging_jni",
>+      ":native_api_jni",
>+      "../../rtc_base:rtc_base",
>+    ]
>+  }
>+
>+  rtc_source_set("audio_device_module_base") {
>+    visibility = [ "*" ]
>+
>+    sources = [
>+      "src/jni/audio_device/audio_common.h",
>+      "src/jni/audio_device/audio_device_module.cc",
>+      "src/jni/audio_device/audio_device_module.h",
>+    ]
>+
>+    deps = [
>+      ":base_jni",
>+      ":generated_audio_device_module_base_jni",
>+      ":native_api_jni",
>+      "../../modules/audio_device:audio_device_buffer",
>+      "../../rtc_base:checks",
>+      "../../rtc_base:rtc_base_approved",
>+      "../../system_wrappers:metrics_api",
>+      "//third_party/abseil-cpp/absl/types:optional",
>+    ]
>+  }
>+
>+  rtc_source_set("java_audio_device_module") {
>+    visibility = [ "*" ]
>+
>+    sources = [
>+      "src/jni/audio_device/audio_record_jni.cc",
>+      "src/jni/audio_device/audio_record_jni.h",
>+      "src/jni/audio_device/audio_track_jni.cc",
>+      "src/jni/audio_device/audio_track_jni.h",
>+    ]
>+    deps = [
>+      ":audio_device_module_base",
>+      ":base_jni",
>+      ":generated_java_audio_device_module_native_jni",
>+      "../../modules/audio_device:audio_device",
>+      "../../modules/audio_device:audio_device_buffer",
>+      "../../rtc_base:checks",
>+      "../../rtc_base:rtc_base_approved",
>+      "../../system_wrappers:metrics_api",
>+      "//third_party/abseil-cpp/absl/types:optional",
>+    ]
>+  }
>+
>+  if (rtc_enable_android_aaudio) {
>+    rtc_source_set("aaudio_audio_device_module") {
>+      visibility = [ "*" ]
>+      defines = [ "AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO" ]
>       sources = [
>-        _source_jar,
>-      ]
>-      outputs = [
>-        _output_jar,
>+        "src/jni/audio_device/aaudio_player.cc",
>+        "src/jni/audio_device/aaudio_player.h",
>+        "src/jni/audio_device/aaudio_recorder.cc",
>+        "src/jni/audio_device/aaudio_recorder.h",
>+        "src/jni/audio_device/aaudio_wrapper.cc",
>+        "src/jni/audio_device/aaudio_wrapper.h",
>       ]
>+      libs = [ "aaudio" ]
>       deps = [
>-        _copy_dep,
>+        ":audio_device_module_base",
>+        ":base_jni",
>+        "../../api:array_view",
>+        "../../modules/audio_device:audio_device",
>+        "../../modules/audio_device:audio_device_buffer",
>+        "../../rtc_base:checks",
>+        "../../rtc_base:rtc_base",
>+        "../../rtc_base:rtc_base_approved",
>+        "../../system_wrappers",
>+        "//third_party/abseil-cpp/absl/types:optional",
>       ]
>     }
>-    _deps += [ ":" + _copy_target_name ]
>-    _output_jars += [ _output_jar ]
>   }
> 
>-  action(target_name) {
>-    forward_variables_from(invoker, [ "data" ])
>-    script = "//build/android/gyp/create_dist_jar.py"
>-    depfile = "$target_gen_dir/$target_name.d"
>-    deps = _deps
>+  rtc_source_set("opensles_audio_device_module") {
>+    visibility = [ "*" ]
>+    sources = [
>+      "src/jni/audio_device/opensles_common.cc",
>+      "src/jni/audio_device/opensles_common.h",
>+      "src/jni/audio_device/opensles_player.cc",
>+      "src/jni/audio_device/opensles_player.h",
>+      "src/jni/audio_device/opensles_recorder.cc",
>+      "src/jni/audio_device/opensles_recorder.h",
>+    ]
>+    libs = [ "OpenSLES" ]
>+    deps = [
>+      ":audio_device_module_base",
>+      ":base_jni",
>+      "../../api:array_view",
>+      "../../modules/audio_device:audio_device",
>+      "../../modules/audio_device:audio_device_buffer",
>+      "../../rtc_base:checks",
>+      "../../rtc_base:rtc_base_approved",
>+      "//third_party/abseil-cpp/absl/memory",
>+      "//third_party/abseil-cpp/absl/types:optional",
>+    ]
>+  }
>+
>+  #########################
>+  # Generated JNI targets #
>+  #########################
>+
>+  generate_jar_jni("generated_external_classes_jni") {
>+    classes = [
>+      "java/lang/Integer.class",
>+      "java/lang/Double.class",
>+      "java/lang/Long.class",
>+      "java/lang/Iterable.class",
>+      "java/util/Iterator.class",
>+      "java/lang/Boolean.class",
>+      "java/math/BigInteger.class",
>+      "java/util/Map.class",
>+      "java/util/LinkedHashMap.class",
>+      "java/util/ArrayList.class",
>+      "java/lang/Enum.class",
>+    ]
>+    jni_package = ""
>+    jni_generator_include = "//sdk/android/src/jni/jni_generator_helper.h"
>+  }
>+
>+  generate_jni("generated_metrics_jni") {
>+    sources = [
>+      "api/org/webrtc/Metrics.java",
>+    ]
>+    jni_package = ""
>+    namespace = "webrtc::jni"
>+    jni_generator_include = "//sdk/android/src/jni/jni_generator_helper.h"
>+  }
>+
>+  # Generated JNI for public JNI targets, matching order of targets
>+
>+  generate_jni("generated_base_jni") {
>+    sources = [
>+      "api/org/webrtc/NetworkMonitor.java",
>+      "api/org/webrtc/NetworkMonitorAutoDetect.java",
>+      "src/java/org/webrtc/Histogram.java",
>+      "src/java/org/webrtc/JniCommon.java",
>+    ]
>+    jni_package = ""
>+    namespace = "webrtc::jni"
>+    jni_generator_include = "//sdk/android/src/jni/jni_generator_helper.h"
>+  }
>+
>+  generate_jni("generated_video_jni") {
>+    sources = [
>+      "api/org/webrtc/EncodedImage.java",
>+      "api/org/webrtc/JavaI420Buffer.java",
>+      "api/org/webrtc/MediaCodecVideoDecoder.java",
>+      "api/org/webrtc/MediaCodecVideoEncoder.java",
>+      "api/org/webrtc/VideoCodecInfo.java",
>+      "api/org/webrtc/VideoCodecStatus.java",
>+      "api/org/webrtc/VideoDecoder.java",
>+      "api/org/webrtc/VideoDecoderFactory.java",
>+      "api/org/webrtc/VideoDecoderFallback.java",
>+      "api/org/webrtc/VideoEncoder.java",
>+      "api/org/webrtc/VideoEncoderFactory.java",
>+      "api/org/webrtc/VideoEncoderFallback.java",
>+      "api/org/webrtc/VideoFrame.java",
>+      "api/org/webrtc/VideoSink.java",
>+      "api/org/webrtc/VideoSource.java",
>+      "api/org/webrtc/VideoTrack.java",
>+      "api/org/webrtc/YuvHelper.java",
>+      "src/java/org/webrtc/H264Utils.java",
>+      "src/java/org/webrtc/NV12Buffer.java",
>+      "src/java/org/webrtc/NV21Buffer.java",
>+      "src/java/org/webrtc/NativeCapturerObserver.java",
>+      "src/java/org/webrtc/VideoDecoderWrapper.java",
>+      "src/java/org/webrtc/VideoEncoderWrapper.java",
>+      "src/java/org/webrtc/WrappedNativeI420Buffer.java",
>+    ]
>+    jni_package = ""
>+    namespace = "webrtc::jni"
>+    jni_generator_include = "//sdk/android/src/jni/jni_generator_helper.h"
>+  }
>+
>+  generate_jni("generated_vp8_jni") {
>+    sources = [
>+      "src/java/org/webrtc/VP8Decoder.java",
>+      "src/java/org/webrtc/VP8Encoder.java",
>+    ]
>+
>+    jni_package = ""
>+    namespace = "webrtc::jni"
>+    jni_generator_include = "//sdk/android/src/jni/jni_generator_helper.h"
>+  }
>+
>+  generate_jni("generated_vp9_jni") {
>+    sources = [
>+      "src/java/org/webrtc/VP9Decoder.java",
>+      "src/java/org/webrtc/VP9Encoder.java",
>+    ]
>+
>+    jni_package = ""
>+    namespace = "webrtc::jni"
>+    jni_generator_include = "//sdk/android/src/jni/jni_generator_helper.h"
>+  }
>+
>+  generate_jni("generated_peerconnection_jni") {
>+    sources = [
>+      "api/org/webrtc/AudioTrack.java",
>+      "api/org/webrtc/CallSessionFileRotatingLogSink.java",
>+      "api/org/webrtc/DataChannel.java",
>+      "api/org/webrtc/DtmfSender.java",
>+      "api/org/webrtc/IceCandidate.java",
>+      "api/org/webrtc/MediaConstraints.java",
>+      "api/org/webrtc/MediaSource.java",
>+      "api/org/webrtc/MediaStream.java",
>+      "api/org/webrtc/MediaStreamTrack.java",
>+      "api/org/webrtc/PeerConnection.java",
>+      "api/org/webrtc/PeerConnectionFactory.java",
>+      "api/org/webrtc/RTCStats.java",
>+      "api/org/webrtc/RTCStatsCollectorCallback.java",
>+      "api/org/webrtc/RTCStatsReport.java",
>+      "api/org/webrtc/RtpParameters.java",
>+      "api/org/webrtc/RtpReceiver.java",
>+      "api/org/webrtc/RtpSender.java",
>+      "api/org/webrtc/RtpTransceiver.java",
>+      "api/org/webrtc/SSLCertificateVerifier.java",
>+      "api/org/webrtc/SdpObserver.java",
>+      "api/org/webrtc/SessionDescription.java",
>+      "api/org/webrtc/StatsObserver.java",
>+      "api/org/webrtc/StatsReport.java",
>+      "api/org/webrtc/TurnCustomizer.java",
>+    ]
>+    jni_package = ""
>+    namespace = "webrtc::jni"
>+    jni_generator_include = "//sdk/android/src/jni/jni_generator_helper.h"
>+  }
>+
>+  generate_jni("generated_java_audio_jni") {
>+    sources = [
>+      "api/org/webrtc/audio/JavaAudioDeviceModule.java",
>+    ]
>+    jni_package = ""
>+    namespace = "webrtc::jni"
>+    jni_generator_include = "//sdk/android/src/jni/jni_generator_helper.h"
>+  }
>+
>+  # Generated JNI for native API targets
> 
>-    outputs = [
>-      invoker.output,
>+  generate_jni("generated_native_api_jni") {
>+    sources = [
>+      "src/java/org/webrtc/JniHelper.java",
>+      "src/java/org/webrtc/WebRtcClassLoader.java",
>     ]
>+    jni_package = ""
>+    namespace = "webrtc::jni"
>+    jni_generator_include = "//sdk/android/src/jni/jni_generator_helper.h"
>+  }
>+
>+  # Generated JNI for internal targets.
>+
>+  generate_jni("generated_logging_jni") {
>+    sources = [
>+      "src/java/org/webrtc/JNILogging.java",
>+    ]
>+    jni_package = ""
>+    jni_generator_include = "//sdk/android/src/jni/jni_generator_helper.h"
>+  }
> 
>-    args = [
>-      "--depfile",
>-      rebase_path(depfile, root_build_dir),
>-      "--output",
>-      rebase_path(invoker.output, root_build_dir),
>+  generate_jni("generated_audio_device_module_base_jni") {
>+    sources = [
>+      "src/java/org/webrtc/audio/WebRtcAudioManager.java",
>     ]
>+    jni_package = ""
>+    namespace = "webrtc::jni"
>+    jni_generator_include = "//sdk/android/src/jni/jni_generator_helper.h"
>+  }
>+
>+  generate_jni("generated_java_audio_device_module_native_jni") {
>+    sources = [
>+      "src/java/org/webrtc/audio/WebRtcAudioRecord.java",
>+      "src/java/org/webrtc/audio/WebRtcAudioTrack.java",
>+    ]
>+    jni_package = ""
>+    namespace = "webrtc::jni"
>+    jni_generator_include = "//sdk/android/src/jni/jni_generator_helper.h"
>+  }
>+
>+  ################
>+  # Test targets #
>+  ################
>+
>+  if (rtc_include_tests) {
>+    rtc_instrumentation_test_apk("android_instrumentation_test_apk") {
>+      apk_name = "android_instrumentation_test_apk"
>+      android_manifest = "instrumentationtests/AndroidManifest.xml"
>+
>+      java_files = [
>+        "instrumentationtests/src/org/webrtc/Camera1CapturerUsingByteBufferTest.java",
>+        "instrumentationtests/src/org/webrtc/Camera1CapturerUsingTextureTest.java",
>+        "instrumentationtests/src/org/webrtc/Camera2CapturerTest.java",
>+        "instrumentationtests/src/org/webrtc/CameraVideoCapturerTestFixtures.java",
>+        "instrumentationtests/src/org/webrtc/TestConstants.java",
>+        "instrumentationtests/src/org/webrtc/DefaultVideoEncoderFactoryTest.java",
>+        "instrumentationtests/src/org/webrtc/EglRendererTest.java",
>+        "instrumentationtests/src/org/webrtc/FileVideoCapturerTest.java",
>+        "instrumentationtests/src/org/webrtc/GlRectDrawerTest.java",
>+        "instrumentationtests/src/org/webrtc/VideoFrameBufferTest.java",
>+        "instrumentationtests/src/org/webrtc/HardwareVideoDecoderTest.java",
>+        "instrumentationtests/src/org/webrtc/HardwareVideoEncoderTest.java",
>+        "instrumentationtests/src/org/webrtc/LoggableTest.java",
>+        "instrumentationtests/src/org/webrtc/MediaCodecVideoEncoderTest.java",
>+        "instrumentationtests/src/org/webrtc/NetworkMonitorTest.java",
>+        "instrumentationtests/src/org/webrtc/PeerConnectionFactoryTest.java",
>+        "instrumentationtests/src/org/webrtc/PeerConnectionTest.java",
>+        "instrumentationtests/src/org/webrtc/RendererCommonTest.java",
>+        "instrumentationtests/src/org/webrtc/SurfaceTextureHelperTest.java",
>+        "instrumentationtests/src/org/webrtc/SurfaceViewRendererOnMeasureTest.java",
>+        "instrumentationtests/src/org/webrtc/VideoFileRendererTest.java",
>+        "instrumentationtests/src/org/webrtc/WebRtcJniBootTest.java",
>+        "instrumentationtests/src/org/webrtc/YuvHelperTest.java",
>+      ]
> 
>-    foreach(output_jar, _output_jars) {
>-      args += [
>-        "--inputs",
>-        rebase_path(output_jar, root_build_dir),
>+      data = [
>+        "../../sdk/android/instrumentationtests/src/org/webrtc/capturetestvideo.y4m",
>+      ]
>+
>+      deps = [
>+        ":base_java",
>+        ":default_video_codec_factory_java",
>+        ":libjingle_peerconnection_java",
>+        ":libjingle_peerconnection_metrics_default_java",
>+        ":video_api_java",
>+        ":video_java",
>+        "//base:base_java_test_support",
>+        "//rtc_base:base_java",
>+        "//third_party/android_support_test_runner:rules_java",
>+        "//third_party/android_support_test_runner:runner_java",
>+        "//third_party/junit",
>+      ]
>+
>+      shared_libraries = [
>+        "../../sdk/android:libjingle_peerconnection_instrumentationtests_so",
>       ]
>     }
>   }
>-}
> 
>-java8_dist_jar("libwebrtc") {
>-  _target_dir_name = get_label_info(":$target_name", "dir")
>-  output = "${root_out_dir}/lib.java${_target_dir_name}/${target_name}.jar"
>+  rtc_shared_library("libjingle_peerconnection_instrumentationtests_so") {
>+    testonly = true
>+    sources = [
>+      "src/jni/jni_onload.cc",
>+    ]
> 
>-  deps = [
>-    ":libjingle_peerconnection_java",
>-    ":libjingle_peerconnection_metrics_default_java",
>-    "../../modules/audio_device:audio_device_java",
>-    "../../rtc_base:base_java",
>-  ]
>-}
>+    suppressed_configs += [ "//build/config/android:hide_all_but_jni_onload" ]
>+    configs += [ "//build/config/android:hide_all_but_jni" ]
> 
>-rtc_android_library("libjingle_peerconnection_java") {
>-  java_files = [
>-    "src/java/org/webrtc/VideoDecoderWrapper.java",
>-    "api/org/webrtc/AudioProcessingFactory.java",
>-    "api/org/webrtc/AudioSource.java",
>-    "api/org/webrtc/AudioTrack.java",
>-    "api/org/webrtc/CallSessionFileRotatingLogSink.java",
>-    "api/org/webrtc/Camera1Capturer.java",
>-    "api/org/webrtc/Camera1Enumerator.java",
>-    "api/org/webrtc/Camera2Capturer.java",
>-    "api/org/webrtc/Camera2Enumerator.java",
>-    "api/org/webrtc/CameraEnumerationAndroid.java",
>-    "api/org/webrtc/CameraEnumerator.java",
>-    "api/org/webrtc/CameraVideoCapturer.java",
>-    "api/org/webrtc/DataChannel.java",
>-    "api/org/webrtc/DefaultAudioProcessingFactory.java",
>-    "api/org/webrtc/DefaultVideoDecoderFactory.java",
>-    "api/org/webrtc/DefaultVideoEncoderFactory.java",
>-    "api/org/webrtc/DtmfSender.java",
>-    "api/org/webrtc/EglBase.java",
>-    "api/org/webrtc/EglRenderer.java",
>-    "api/org/webrtc/EncodedImage.java",
>-    "api/org/webrtc/FileVideoCapturer.java",
>-    "api/org/webrtc/GlRectDrawer.java",
>-    "api/org/webrtc/GlShader.java",
>-    "api/org/webrtc/GlTextureFrameBuffer.java",
>-    "api/org/webrtc/GlUtil.java",
>-    "api/org/webrtc/HardwareVideoDecoderFactory.java",
>-    "api/org/webrtc/HardwareVideoEncoderFactory.java",
>-    "api/org/webrtc/IceCandidate.java",
>-    "api/org/webrtc/JavaI420Buffer.java",
>-    "api/org/webrtc/MediaCodecVideoDecoder.java",
>-    "api/org/webrtc/MediaCodecVideoEncoder.java",
>-    "api/org/webrtc/MediaConstraints.java",
>-    "api/org/webrtc/MediaSource.java",
>-    "api/org/webrtc/MediaStream.java",
>-    "api/org/webrtc/MediaStreamTrack.java",
>-    "api/org/webrtc/NativeLibraryLoader.java",
>-    "api/org/webrtc/NetworkMonitor.java",
>-    "api/org/webrtc/NetworkMonitorAutoDetect.java",
>-    "api/org/webrtc/PeerConnection.java",
>-    "api/org/webrtc/PeerConnectionFactory.java",
>-    "api/org/webrtc/PostProcessingFactory.java",
>-    "api/org/webrtc/RendererCommon.java",
>-    "api/org/webrtc/RTCStats.java",
>-    "api/org/webrtc/RTCStatsCollectorCallback.java",
>-    "api/org/webrtc/RTCStatsReport.java",
>-    "api/org/webrtc/RtpParameters.java",
>-    "api/org/webrtc/RtpReceiver.java",
>-    "api/org/webrtc/RtpSender.java",
>-    "api/org/webrtc/ScreenCapturerAndroid.java",
>-    "api/org/webrtc/SdpObserver.java",
>-    "api/org/webrtc/SessionDescription.java",
>-    "api/org/webrtc/SoftwareVideoDecoderFactory.java",
>-    "api/org/webrtc/SoftwareVideoEncoderFactory.java",
>-    "api/org/webrtc/StatsObserver.java",
>-    "api/org/webrtc/StatsReport.java",
>-    "api/org/webrtc/SurfaceTextureHelper.java",
>-    "api/org/webrtc/SurfaceEglRenderer.java",
>-    "api/org/webrtc/SurfaceViewRenderer.java",
>-    "api/org/webrtc/TurnCustomizer.java",
>-    "api/org/webrtc/VideoCapturer.java",
>-    "api/org/webrtc/VideoCodecInfo.java",
>-    "api/org/webrtc/VideoCodecStatus.java",
>-    "api/org/webrtc/VideoDecoder.java",
>-    "api/org/webrtc/VideoDecoderFactory.java",
>-    "api/org/webrtc/VideoDecoderFallback.java",
>-    "api/org/webrtc/VideoEncoder.java",
>-    "api/org/webrtc/VideoEncoderFactory.java",
>-    "api/org/webrtc/VideoEncoderFallback.java",
>-    "api/org/webrtc/VideoFileRenderer.java",
>-    "api/org/webrtc/VideoFrame.java",
>-    "api/org/webrtc/VideoFrameDrawer.java",
>-    "api/org/webrtc/VideoRenderer.java",
>-    "api/org/webrtc/VideoSink.java",
>-    "api/org/webrtc/VideoSource.java",
>-    "api/org/webrtc/VideoTrack.java",
>-    "api/org/webrtc/YuvConverter.java",
>-    "api/org/webrtc/YuvHelper.java",
>-    "src/java/org/webrtc/AndroidVideoTrackSourceObserver.java",
>-    "src/java/org/webrtc/BaseBitrateAdjuster.java",
>-    "src/java/org/webrtc/BitrateAdjuster.java",
>-    "src/java/org/webrtc/CalledByNative.java",
>-    "src/java/org/webrtc/Camera1Session.java",
>-    "src/java/org/webrtc/Camera2Session.java",
>-    "src/java/org/webrtc/CameraCapturer.java",
>-    "src/java/org/webrtc/CameraSession.java",
>-    "src/java/org/webrtc/WebRtcClassLoader.java",
>-    "src/java/org/webrtc/DynamicBitrateAdjuster.java",
>-    "src/java/org/webrtc/EglBase10.java",
>-    "src/java/org/webrtc/EglBase14.java",
>-    "src/java/org/webrtc/FramerateBitrateAdjuster.java",
>-    "src/java/org/webrtc/HardwareVideoDecoder.java",
>-    "src/java/org/webrtc/HardwareVideoEncoder.java",
>-    "src/java/org/webrtc/Histogram.java",
>-    "src/java/org/webrtc/JniCommon.java",
>-    "src/java/org/webrtc/MediaCodecUtils.java",
>-    "src/java/org/webrtc/NativeClassQualifiedName.java",
>-    "src/java/org/webrtc/NativeLibrary.java",
>-    "src/java/org/webrtc/NV12Buffer.java",
>-    "src/java/org/webrtc/NV21Buffer.java",
>-    "src/java/org/webrtc/TextureBufferImpl.java",
>-    "src/java/org/webrtc/VideoCodecType.java",
>-    "src/java/org/webrtc/VideoEncoderWrapper.java",
>-    "src/java/org/webrtc/WrappedNativeI420Buffer.java",
>-    "src/java/org/webrtc/WrappedNativeVideoEncoder.java",
>-    "src/java/org/webrtc/WrappedNativeVideoDecoder.java",
>-    "src/java/org/webrtc/VP8Encoder.java",
>-    "src/java/org/webrtc/VP8Decoder.java",
>-    "src/java/org/webrtc/VP9Encoder.java",
>-    "src/java/org/webrtc/VP9Decoder.java",
>-  ]
>-
>-  deps = [
>-    "../../modules/audio_device:audio_device_java",
>-    "../../rtc_base:base_java",
>-  ]
>-}
>+    deps = [
>+      ":instrumentationtests_jni",
>+      ":libjingle_peerconnection_jni",
>+      ":libjingle_peerconnection_metrics_default_jni",
>+      "../../pc:libjingle_peerconnection",
>+      "../../rtc_base:rtc_base",
>+    ]
>+    output_extension = "so"
>+  }
> 
>-rtc_android_library("libjingle_peerconnection_metrics_default_java") {
>-  java_files = [ "api/org/webrtc/Metrics.java" ]
>+  rtc_static_library("instrumentationtests_jni") {
>+    testonly = true
>+    sources = [
>+      "instrumentationtests/loggable_test.cc",
>+    ]
> 
>-  deps = [
>-    ":libjingle_peerconnection_java",
>-    "../../rtc_base:base_java",
>-  ]
>-}
>+    deps = [
>+      ":base_jni",
>+      ":native_api_jni",
>+      "../../rtc_base:rtc_base_approved",
>+    ]
>+  }
> 
>-if (rtc_include_tests) {
>-  rtc_instrumentation_test_apk("libjingle_peerconnection_android_unittest") {
>-    apk_name = "libjingle_peerconnection_android_unittest"
>-    android_manifest = "instrumentationtests/AndroidManifest.xml"
>+  rtc_source_set("native_unittests") {
>+    testonly = true
> 
>-    java_files = [
>-      "instrumentationtests/src/org/webrtc/Camera1CapturerUsingByteBufferTest.java",
>-      "instrumentationtests/src/org/webrtc/Camera1CapturerUsingTextureTest.java",
>-      "instrumentationtests/src/org/webrtc/Camera2CapturerTest.java",
>-      "instrumentationtests/src/org/webrtc/CameraVideoCapturerTestFixtures.java",
>-      "instrumentationtests/src/org/webrtc/DefaultAudioProcessingFactoryTest.java",
>-      "instrumentationtests/src/org/webrtc/DefaultVideoEncoderFactoryTest.java",
>-      "instrumentationtests/src/org/webrtc/EglRendererTest.java",
>-      "instrumentationtests/src/org/webrtc/FileVideoCapturerTest.java",
>-      "instrumentationtests/src/org/webrtc/GlRectDrawerTest.java",
>-      "instrumentationtests/src/org/webrtc/HardwareVideoDecoderTest.java",
>-      "instrumentationtests/src/org/webrtc/HardwareVideoEncoderTest.java",
>-      "instrumentationtests/src/org/webrtc/MediaCodecVideoEncoderTest.java",
>-      "instrumentationtests/src/org/webrtc/NetworkMonitorTest.java",
>-      "instrumentationtests/src/org/webrtc/PeerConnectionFactoryTest.java",
>-      "instrumentationtests/src/org/webrtc/PeerConnectionTest.java",
>-      "instrumentationtests/src/org/webrtc/RendererCommonTest.java",
>-      "instrumentationtests/src/org/webrtc/SurfaceTextureHelperTest.java",
>-      "instrumentationtests/src/org/webrtc/SurfaceViewRendererOnMeasureTest.java",
>-      "instrumentationtests/src/org/webrtc/VideoFileRendererTest.java",
>-      "instrumentationtests/src/org/webrtc/WebRtcJniBootTest.java",
>-      "instrumentationtests/src/org/webrtc/YuvHelperTest.java",
>+    sources = [
>+      "native_unittests/audio_device/audio_device_unittest.cc",
>+      "native_unittests/codecs/wrapper_unittest.cc",
>+      "native_unittests/java_types_unittest.cc",
>+      "native_unittests/peerconnection/peerconnectionfactory_unittest.cc",
>+      "native_unittests/test_jni_onload.cc",
>+      "native_unittests/video/videosource_unittest.cc",
>     ]
> 
>     data = [
>-      "../../sdk/android/instrumentationtests/src/org/webrtc/capturetestvideo.y4m",
>+      "../../resources/audio_device/audio_short44.pcm",
>+      "../../resources/audio_device/audio_short48.pcm",
>     ]
> 
>+    if (is_clang) {
>+      # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>+      suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>+    }
>+
>     deps = [
>+      ":audio_device_module_base",
>+      ":audio_jni",
>+      ":base_jni",
>+      ":generated_native_unittests_jni",
>+      ":media_jni",
>+      ":native_api_audio_device_module",
>+      ":native_api_base",
>+      ":native_api_codecs",
>+      ":native_api_jni",
>+      ":native_api_peerconnection",
>+      ":native_api_video",
>+      ":opensles_audio_device_module",
>+      ":video_jni",
>+      "../../system_wrappers:system_wrappers",
>+      "//api/audio_codecs:builtin_audio_decoder_factory",
>+      "//api/audio_codecs:builtin_audio_encoder_factory",
>+      "//api/video:video_frame",
>+      "//media:rtc_audio_video",
>+      "//media:rtc_internal_video_codecs",
>+      "//media:rtc_media_base",
>+      "//modules/audio_device:audio_device",
>+      "//modules/audio_device:mock_audio_device",
>+      "//modules/audio_processing:audio_processing",
>+      "//modules/utility:utility",
>+      "//pc:libjingle_peerconnection",
>+      "//rtc_base:checks",
>+      "//rtc_base:rtc_base_approved",
>+      "//test:fileutils",
>+      "//test:test_support",
>+      "//testing/gtest",
>+      "//third_party/abseil-cpp/absl/memory",
>+    ]
>+  }
>+
>+  rtc_android_library("native_unittests_java") {
>+    testonly = true
>+
>+    java_files = [
>+      "native_unittests/org/webrtc/ApplicationContextProvider.java",
>+      "native_unittests/org/webrtc/BuildInfo.java",
>+      "native_unittests/org/webrtc/CodecsWrapperTestHelper.java",
>+      "native_unittests/org/webrtc/JavaTypesTestHelper.java",
>+      "native_unittests/org/webrtc/JavaVideoSourceTestHelper.java",
>+      "native_unittests/org/webrtc/PeerConnectionFactoryInitializationHelper.java",
>+    ]
>+
>+    deps = [
>+      ":base_java",
>+      ":java_audio_device_module_java",
>+      ":peerconnection_java",
>+      ":video_api_java",
>+      ":video_java",
>       "../../rtc_base:base_java",
>-      "../../sdk/android:libjingle_peerconnection_java",
>-      "../../sdk/android:libjingle_peerconnection_metrics_default_java",
>-      "//base:base_java",
>-      "//base:base_java_test_support",
>-      "//third_party/android_support_test_runner:rules_java",
>       "//third_party/android_support_test_runner:runner_java",
>-      "//third_party/junit",
>     ]
>+  }
> 
>-    shared_libraries = [ "../../sdk/android:libjingle_peerconnection_so" ]
>+  generate_jni("generated_native_unittests_jni") {
>+    testonly = true
>+
>+    sources = [
>+      "native_unittests/org/webrtc/ApplicationContextProvider.java",
>+      "native_unittests/org/webrtc/BuildInfo.java",
>+      "native_unittests/org/webrtc/CodecsWrapperTestHelper.java",
>+      "native_unittests/org/webrtc/JavaTypesTestHelper.java",
>+      "native_unittests/org/webrtc/JavaVideoSourceTestHelper.java",
>+      "native_unittests/org/webrtc/PeerConnectionFactoryInitializationHelper.java",
>+    ]
>+    jni_package = ""
>+    namespace = "webrtc::jni"
>+    jni_generator_include = "//sdk/android/src/jni/jni_generator_helper.h"
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/OWNERS
>index b77294b3d84..4af75f35bcd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/OWNERS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/OWNERS
>@@ -1,3 +1,4 @@
> glaznev@webrtc.org
> magjed@webrtc.org
> sakal@webrtc.org
>+per-file *Audio*.java=henrika@webrtc.org
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/README b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/README
>index c5aba93e2ff..53bdc9edc58 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/README
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/README
>@@ -2,10 +2,20 @@ This directory holds a Java implementation of the webrtc::PeerConnection API, as
> well as the JNI glue C++ code that lets the Java implementation reuse the C++
> implementation of the same API.
> 
>-To build the Java API and related tests, generate GN projects with:
>---args='target_os="android"'
>+To build the Java API and related tests, make sure you have a WebRTC checkout
>+with Android specific parts. This can be used for linux development as well by
>+configuring gn appropriately, as it is a superset of the webrtc checkout:
>+fetch --nohooks webrtc_android
>+gclient sync
>+
>+You also must generate GN projects with:
>+--args='target_os="android" target_cpu="arm"'
>+
>+More information on getting the code, compiling and running the AppRTCMobile
>+app can be found at:
>+https://webrtc.org/native-code/android/
> 
> To use the Java API, start by looking at the public interface of
> org.webrtc.PeerConnection{,Factory} and the org.webrtc.PeerConnectionTest.
> 
>-To understand the implementation of the API, see the native code in jni/.
>+To understand the implementation of the API, see the native code in src/jni/pc/.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/AudioTrack.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/AudioTrack.java
>index 6b713536104..828e89c5e53 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/AudioTrack.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/AudioTrack.java
>@@ -23,5 +23,5 @@ public class AudioTrack extends MediaStreamTrack {
>     nativeSetVolume(super.nativeTrack, volume);
>   }
> 
>-  private static native void nativeSetVolume(long nativeTrack, double volume);
>+  private static native void nativeSetVolume(long track, double volume);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/CallSessionFileRotatingLogSink.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/CallSessionFileRotatingLogSink.java
>index 132248797d0..f4edb588471 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/CallSessionFileRotatingLogSink.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/CallSessionFileRotatingLogSink.java
>@@ -14,11 +14,17 @@ public class CallSessionFileRotatingLogSink {
>   private long nativeSink;
> 
>   public static byte[] getLogData(String dirPath) {
>+    if (dirPath == null) {
>+      throw new IllegalArgumentException("dirPath may not be null.");
>+    }
>     return nativeGetLogData(dirPath);
>   }
> 
>   public CallSessionFileRotatingLogSink(
>       String dirPath, int maxFileSize, Logging.Severity severity) {
>+    if (dirPath == null) {
>+      throw new IllegalArgumentException("dirPath may not be null.");
>+    }
>     nativeSink = nativeAddSink(dirPath, maxFileSize, severity.ordinal());
>   }
> 
>@@ -30,6 +36,6 @@ public class CallSessionFileRotatingLogSink {
>   }
> 
>   private static native long nativeAddSink(String dirPath, int maxFileSize, int severity);
>-  private static native void nativeDeleteSink(long nativeSink);
>+  private static native void nativeDeleteSink(long sink);
>   private static native byte[] nativeGetLogData(String dirPath);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/Camera1Capturer.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/Camera1Capturer.java
>index 41352abf51b..f178a3d5d3e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/Camera1Capturer.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/Camera1Capturer.java
>@@ -11,7 +11,7 @@
> package org.webrtc;
> 
> import android.content.Context;
>-import android.media.MediaRecorder;
>+import javax.annotation.Nullable;
> 
> public class Camera1Capturer extends CameraCapturer {
>   private final boolean captureToTexture;
>@@ -26,10 +26,10 @@ public class Camera1Capturer extends CameraCapturer {
>   @Override
>   protected void createCameraSession(CameraSession.CreateSessionCallback createSessionCallback,
>       CameraSession.Events events, Context applicationContext,
>-      SurfaceTextureHelper surfaceTextureHelper, MediaRecorder mediaRecorder, String cameraName,
>-      int width, int height, int framerate) {
>-    Camera1Session.create(createSessionCallback, events,
>-        captureToTexture || (mediaRecorder != null), applicationContext, surfaceTextureHelper,
>-        mediaRecorder, Camera1Enumerator.getCameraIndex(cameraName), width, height, framerate);
>+      SurfaceTextureHelper surfaceTextureHelper, String cameraName, int width, int height,
>+      int framerate) {
>+    Camera1Session.create(createSessionCallback, events, captureToTexture, applicationContext,
>+        surfaceTextureHelper, Camera1Enumerator.getCameraIndex(cameraName), width, height,
>+        framerate);
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/Camera1Enumerator.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/Camera1Enumerator.java
>index 429daf5dc35..b11da315463 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/Camera1Enumerator.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/Camera1Enumerator.java
>@@ -13,6 +13,7 @@ package org.webrtc;
> import android.os.SystemClock;
> import java.util.ArrayList;
> import java.util.List;
>+import javax.annotation.Nullable;
> import org.webrtc.CameraEnumerationAndroid.CaptureFormat;
> 
> @SuppressWarnings("deprecation")
>@@ -73,7 +74,7 @@ public class Camera1Enumerator implements CameraEnumerator {
>     return new Camera1Capturer(deviceName, eventsHandler, captureToTexture);
>   }
> 
>-  private static android.hardware.Camera.CameraInfo getCameraInfo(int index) {
>+  private static @Nullable android.hardware.Camera.CameraInfo getCameraInfo(int index) {
>     android.hardware.Camera.CameraInfo info = new android.hardware.Camera.CameraInfo();
>     try {
>       android.hardware.Camera.getCameraInfo(index, info);
>@@ -171,7 +172,7 @@ public class Camera1Enumerator implements CameraEnumerator {
> 
>   // Returns the name of the camera with camera index. Returns null if the
>   // camera can not be used.
>-  static String getDeviceName(int index) {
>+  static @Nullable String getDeviceName(int index) {
>     android.hardware.Camera.CameraInfo info = getCameraInfo(index);
>     if (info == null) {
>       return null;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/Camera2Capturer.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/Camera2Capturer.java
>index 769a94a60c0..3c04cdb10e0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/Camera2Capturer.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/Camera2Capturer.java
>@@ -13,12 +13,12 @@ package org.webrtc;
> import android.annotation.TargetApi;
> import android.content.Context;
> import android.hardware.camera2.CameraManager;
>-import android.media.MediaRecorder;
>+import javax.annotation.Nullable;
> 
> @TargetApi(21)
> public class Camera2Capturer extends CameraCapturer {
>   private final Context context;
>-  private final CameraManager cameraManager;
>+  @Nullable private final CameraManager cameraManager;
> 
>   public Camera2Capturer(Context context, String cameraName, CameraEventsHandler eventsHandler) {
>     super(cameraName, eventsHandler, new Camera2Enumerator(context));
>@@ -30,9 +30,9 @@ public class Camera2Capturer extends CameraCapturer {
>   @Override
>   protected void createCameraSession(CameraSession.CreateSessionCallback createSessionCallback,
>       CameraSession.Events events, Context applicationContext,
>-      SurfaceTextureHelper surfaceTextureHelper, MediaRecorder mediaRecoder, String cameraName,
>-      int width, int height, int framerate) {
>+      SurfaceTextureHelper surfaceTextureHelper, String cameraName, int width, int height,
>+      int framerate) {
>     Camera2Session.create(createSessionCallback, events, applicationContext, cameraManager,
>-        surfaceTextureHelper, mediaRecoder, cameraName, width, height, framerate);
>+        surfaceTextureHelper, cameraName, width, height, framerate);
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/Camera2Enumerator.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/Camera2Enumerator.java
>index fa5c55df8aa..4c3fc9aeac6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/Camera2Enumerator.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/Camera2Enumerator.java
>@@ -10,8 +10,6 @@
> 
> package org.webrtc;
> 
>-import org.webrtc.CameraEnumerationAndroid.CaptureFormat;
>-
> import android.annotation.TargetApi;
> import android.content.Context;
> import android.graphics.Rect;
>@@ -24,11 +22,12 @@ import android.os.Build;
> import android.os.SystemClock;
> import android.util.AndroidException;
> import android.util.Range;
>-
> import java.util.ArrayList;
> import java.util.HashMap;
> import java.util.List;
> import java.util.Map;
>+import javax.annotation.Nullable;
>+import org.webrtc.CameraEnumerationAndroid.CaptureFormat;
> 
> @TargetApi(21)
> public class Camera2Enumerator implements CameraEnumerator {
>@@ -41,7 +40,7 @@ public class Camera2Enumerator implements CameraEnumerator {
>       new HashMap<String, List<CaptureFormat>>();
> 
>   final Context context;
>-  final CameraManager cameraManager;
>+  @Nullable final CameraManager cameraManager;
> 
>   public Camera2Enumerator(Context context) {
>     this.context = context;
>@@ -90,7 +89,7 @@ public class Camera2Enumerator implements CameraEnumerator {
>     return new Camera2Capturer(context, deviceName, eventsHandler);
>   }
> 
>-  private CameraCharacteristics getCameraCharacteristics(String deviceName) {
>+  private @Nullable CameraCharacteristics getCameraCharacteristics(String deviceName) {
>     try {
>       return cameraManager.getCameraCharacteristics(deviceName);
>       // On Android OS pre 4.4.2, a class will not load because of VerifyError if it contains a
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/CameraVideoCapturer.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/CameraVideoCapturer.java
>index be9d669fc4d..2e9c11fe4f6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/CameraVideoCapturer.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/CameraVideoCapturer.java
>@@ -11,6 +11,7 @@
> package org.webrtc;
> 
> import android.media.MediaRecorder;
>+import javax.annotation.Nullable;
> 
> /**
>  * Base interface for camera1 and camera2 implementations. Extends VideoCapturer with a
>@@ -66,6 +67,7 @@ public interface CameraVideoCapturer extends VideoCapturer {
>    * addMediaRecorderToCamera() or removeMediaRecorderFromCamera calls.
>    * The callback may be called on an arbitrary thread.
>    */
>+  @Deprecated
>   public interface MediaRecorderHandler {
>     // Invoked on success.
>     void onMediaRecorderSuccess();
>@@ -79,13 +81,20 @@ public interface CameraVideoCapturer extends VideoCapturer {
>    * Once MediaRecorder is added to camera pipeline camera switch is not allowed.
>    * This function can be called from any thread.
>    */
>-  void addMediaRecorderToCamera(MediaRecorder mediaRecorder, MediaRecorderHandler resultHandler);
>+  @Deprecated
>+  default void addMediaRecorderToCamera(
>+      MediaRecorder mediaRecorder, MediaRecorderHandler resultHandler) {
>+    throw new UnsupportedOperationException("Deprecated and not implemented.");
>+  }
> 
>   /**
>    * Remove MediaRecorder from camera pipeline. This can only be called while the camera is running.
>    * This function can be called from any thread.
>    */
>-  void removeMediaRecorderFromCamera(MediaRecorderHandler resultHandler);
>+  @Deprecated
>+  default void removeMediaRecorderFromCamera(MediaRecorderHandler resultHandler) {
>+    throw new UnsupportedOperationException("Deprecated and not implemented.");
>+  }
> 
>   /**
>    * Helper class to log framerate and detect if the camera freezes. It will run periodic callbacks
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/CapturerObserver.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/CapturerObserver.java
>new file mode 100644
>index 00000000000..382dc15b3a6
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/CapturerObserver.java
>@@ -0,0 +1,27 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc;
>+
>+/**
>+ * Interface for observering a capturer. Passed to {@link VideoCapturer#initialize}. Provided by
>+ * {@link VideoSource#getCapturerObserver}.
>+ *
>+ * All callbacks must be executed on a single thread.
>+ */
>+public interface CapturerObserver {
>+  /** Notify if the capturer have been started successfully or not. */
>+  void onCapturerStarted(boolean success);
>+  /** Notify that the capturer has been stopped. */
>+  void onCapturerStopped();
>+
>+  /** Delivers a captured frame. */
>+  void onFrameCaptured(VideoFrame frame);
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/DataChannel.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/DataChannel.java
>index 883025e960c..6b034096dca 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/DataChannel.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/DataChannel.java
>@@ -113,33 +113,41 @@ public class DataChannel {
>   /** Register |observer|, replacing any previously-registered observer. */
>   public void registerObserver(Observer observer) {
>     if (nativeObserver != 0) {
>-      unregisterObserverNative(nativeObserver);
>+      nativeUnregisterObserver(nativeObserver);
>     }
>-    nativeObserver = registerObserverNative(observer);
>+    nativeObserver = nativeRegisterObserver(observer);
>   }
>-  private native long registerObserverNative(Observer observer);
> 
>   /** Unregister the (only) observer. */
>   public void unregisterObserver() {
>-    unregisterObserverNative(nativeObserver);
>+    nativeUnregisterObserver(nativeObserver);
>   }
>-  private native void unregisterObserverNative(long nativeObserver);
> 
>-  public native String label();
>+  public String label() {
>+    return nativeLabel();
>+  }
> 
>-  public native int id();
>+  public int id() {
>+    return nativeId();
>+  }
> 
>-  public native State state();
>+  public State state() {
>+    return nativeState();
>+  }
> 
>   /**
>    * Return the number of bytes of application data (UTF-8 text and binary data)
>    * that have been queued using SendBuffer but have not yet been transmitted
>    * to the network.
>    */
>-  public native long bufferedAmount();
>+  public long bufferedAmount() {
>+    return nativeBufferedAmount();
>+  }
> 
>   /** Close the channel. */
>-  public native void close();
>+  public void close() {
>+    nativeClose();
>+  }
> 
>   /** Send |data| to the remote peer; return success. */
>   public boolean send(Buffer buffer) {
>@@ -147,9 +155,8 @@ public class DataChannel {
>     // ByteBuffer is direct and/or is backed by an array.
>     byte[] data = new byte[buffer.data.remaining()];
>     buffer.data.get(data);
>-    return sendNative(data, buffer.binary);
>+    return nativeSend(data, buffer.binary);
>   }
>-  private native boolean sendNative(byte[] data, boolean binary);
> 
>   /** Dispose of native resources attached to this channel. */
>   public void dispose() {
>@@ -160,4 +167,13 @@ public class DataChannel {
>   long getNativeDataChannel() {
>     return nativeDataChannel;
>   }
>+
>+  private native long nativeRegisterObserver(Observer observer);
>+  private native void nativeUnregisterObserver(long observer);
>+  private native String nativeLabel();
>+  private native int nativeId();
>+  private native State nativeState();
>+  private native long nativeBufferedAmount();
>+  private native void nativeClose();
>+  private native boolean nativeSend(byte[] data, boolean binary);
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/DefaultAudioProcessingFactory.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/DefaultAudioProcessingFactory.java
>deleted file mode 100644
>index 6fcfe233275..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/DefaultAudioProcessingFactory.java
>+++ /dev/null
>@@ -1,47 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-package org.webrtc;
>-
>-/** Factory for instantiating the default webrtc::AudioProcessing implementation. */
>-public class DefaultAudioProcessingFactory implements AudioProcessingFactory {
>-  public DefaultAudioProcessingFactory() {
>-    this(null /* postProcessingFactory */);
>-  }
>-
>-  /**
>-   * Allows injecting a PostProcessingFactory. A null PostProcessingFactory creates a
>-   * webrtc::AudioProcessing with nullptr webrtc::postProcessing.
>-   */
>-  public DefaultAudioProcessingFactory(PostProcessingFactory postProcessingFactory) {
>-    this.postProcessingFactory = postProcessingFactory;
>-  }
>-
>-  /**
>-   * Creates a default webrtc::AudioProcessing module, which takes ownership of objects created by
>-   * its factories.
>-   */
>-  @Override
>-  public long createNative() {
>-    long nativePostProcessor = 0;
>-    if (postProcessingFactory != null) {
>-      nativePostProcessor = postProcessingFactory.createNative();
>-      if (nativePostProcessor == 0) {
>-        throw new NullPointerException(
>-            "PostProcessingFactory.createNative() may not return 0 (nullptr).");
>-      }
>-    }
>-    return nativeCreateAudioProcessing(nativePostProcessor);
>-  }
>-
>-  private PostProcessingFactory postProcessingFactory;
>-
>-  private static native long nativeCreateAudioProcessing(long nativePostProcessor);
>-}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/DefaultVideoDecoderFactory.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/DefaultVideoDecoderFactory.java
>index 561e7a51ffb..4e5c082fd8b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/DefaultVideoDecoderFactory.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/DefaultVideoDecoderFactory.java
>@@ -10,22 +10,45 @@
> 
> package org.webrtc;
> 
>+import javax.annotation.Nullable;
>+import java.util.Arrays;
>+import java.util.LinkedHashSet;
>+import java.util.List;
>+
>+/** Helper class that combines HW and SW decoders. */
> public class DefaultVideoDecoderFactory implements VideoDecoderFactory {
>-  private final HardwareVideoDecoderFactory hardwareVideoDecoderFactory;
>-  private final SoftwareVideoDecoderFactory softwareVideoDecoderFactory;
>+  private final VideoDecoderFactory hardwareVideoDecoderFactory;
>+  private final VideoDecoderFactory softwareVideoDecoderFactory = new SoftwareVideoDecoderFactory();
> 
>+  /** Create decoder factory using default hardware decoder factory. */
>   public DefaultVideoDecoderFactory(EglBase.Context eglContext) {
>-    hardwareVideoDecoderFactory =
>-        new HardwareVideoDecoderFactory(eglContext, false /* fallbackToSoftware */);
>-    softwareVideoDecoderFactory = new SoftwareVideoDecoderFactory();
>+    this.hardwareVideoDecoderFactory = new HardwareVideoDecoderFactory(eglContext);
>+  }
>+
>+  /** Create decoder factory using explicit hardware decoder factory. */
>+  DefaultVideoDecoderFactory(VideoDecoderFactory hardwareVideoDecoderFactory) {
>+    this.hardwareVideoDecoderFactory = hardwareVideoDecoderFactory;
>   }
> 
>   @Override
>-  public VideoDecoder createDecoder(String codecType) {
>-    VideoDecoder decoder = hardwareVideoDecoderFactory.createDecoder(codecType);
>-    if (decoder != null) {
>-      return decoder;
>+  public @Nullable VideoDecoder createDecoder(VideoCodecInfo codecType) {
>+    final VideoDecoder softwareDecoder = softwareVideoDecoderFactory.createDecoder(codecType);
>+    final VideoDecoder hardwareDecoder = hardwareVideoDecoderFactory.createDecoder(codecType);
>+    if (hardwareDecoder != null && softwareDecoder != null) {
>+      // Both hardware and software supported, wrap it in a software fallback
>+      return new VideoDecoderFallback(
>+          /* fallback= */ softwareDecoder, /* primary= */ hardwareDecoder);
>     }
>-    return softwareVideoDecoderFactory.createDecoder(codecType);
>+    return hardwareDecoder != null ? hardwareDecoder : softwareDecoder;
>+  }
>+
>+  @Override
>+  public VideoCodecInfo[] getSupportedCodecs() {
>+    LinkedHashSet<VideoCodecInfo> supportedCodecInfos = new LinkedHashSet<VideoCodecInfo>();
>+
>+    supportedCodecInfos.addAll(Arrays.asList(softwareVideoDecoderFactory.getSupportedCodecs()));
>+    supportedCodecInfos.addAll(Arrays.asList(hardwareVideoDecoderFactory.getSupportedCodecs()));
>+
>+    return supportedCodecInfos.toArray(new VideoCodecInfo[supportedCodecInfos.size()]);
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/DefaultVideoEncoderFactory.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/DefaultVideoEncoderFactory.java
>index 309ae7432b4..029b2f9bc7e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/DefaultVideoEncoderFactory.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/DefaultVideoEncoderFactory.java
>@@ -10,34 +10,39 @@
> 
> package org.webrtc;
> 
>+import javax.annotation.Nullable;
> import java.util.Arrays;
> import java.util.LinkedHashSet;
> import java.util.List;
> 
>+/** Helper class that combines HW and SW encoders. */
> public class DefaultVideoEncoderFactory implements VideoEncoderFactory {
>   private final VideoEncoderFactory hardwareVideoEncoderFactory;
>-  private final VideoEncoderFactory softwareVideoEncoderFactory;
>+  private final VideoEncoderFactory softwareVideoEncoderFactory = new SoftwareVideoEncoderFactory();
> 
>+  /** Create encoder factory using default hardware encoder factory. */
>   public DefaultVideoEncoderFactory(
>       EglBase.Context eglContext, boolean enableIntelVp8Encoder, boolean enableH264HighProfile) {
>-    hardwareVideoEncoderFactory = new HardwareVideoEncoderFactory(
>-        eglContext, enableIntelVp8Encoder, enableH264HighProfile, false /* fallbackToSoftware */);
>-    softwareVideoEncoderFactory = new SoftwareVideoEncoderFactory();
>+    this.hardwareVideoEncoderFactory =
>+        new HardwareVideoEncoderFactory(eglContext, enableIntelVp8Encoder, enableH264HighProfile);
>   }
> 
>-  /* This is used for testing. */
>+  /** Create encoder factory using explicit hardware encoder factory. */
>   DefaultVideoEncoderFactory(VideoEncoderFactory hardwareVideoEncoderFactory) {
>     this.hardwareVideoEncoderFactory = hardwareVideoEncoderFactory;
>-    softwareVideoEncoderFactory = new SoftwareVideoEncoderFactory();
>   }
> 
>+  @Nullable
>   @Override
>   public VideoEncoder createEncoder(VideoCodecInfo info) {
>-    final VideoEncoder videoEncoder = hardwareVideoEncoderFactory.createEncoder(info);
>-    if (videoEncoder != null) {
>-      return videoEncoder;
>+    final VideoEncoder softwareEncoder = softwareVideoEncoderFactory.createEncoder(info);
>+    final VideoEncoder hardwareEncoder = hardwareVideoEncoderFactory.createEncoder(info);
>+    if (hardwareEncoder != null && softwareEncoder != null) {
>+      // Both hardware and software supported, wrap it in a software fallback
>+      return new VideoEncoderFallback(
>+          /* fallback= */ softwareEncoder, /* primary= */ hardwareEncoder);
>     }
>-    return softwareVideoEncoderFactory.createEncoder(info);
>+    return hardwareEncoder != null ? hardwareEncoder : softwareEncoder;
>   }
> 
>   @Override
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/DtmfSender.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/DtmfSender.java
>index d284313dcb1..9f203f390e2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/DtmfSender.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/DtmfSender.java
>@@ -74,10 +74,10 @@ public class DtmfSender {
>     JniCommon.nativeReleaseRef(nativeDtmfSender);
>   }
> 
>-  private static native boolean nativeCanInsertDtmf(long nativeDtmfSender);
>+  private static native boolean nativeCanInsertDtmf(long dtmfSender);
>   private static native boolean nativeInsertDtmf(
>-      long nativeDtmfSender, String tones, int duration, int interToneGap);
>-  private static native String nativeTones(long nativeDtmfSender);
>-  private static native int nativeDuration(long nativeDtmfSender);
>-  private static native int nativeInterToneGap(long nativeDtmfSender);
>+      long dtmfSender, String tones, int duration, int interToneGap);
>+  private static native String nativeTones(long dtmfSender);
>+  private static native int nativeDuration(long dtmfSender);
>+  private static native int nativeInterToneGap(long dtmfSender);
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/EglBase.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/EglBase.java
>index 23acdefc8d1..31ab6782252 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/EglBase.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/EglBase.java
>@@ -11,15 +11,15 @@
> package org.webrtc;
> 
> import android.graphics.SurfaceTexture;
>+import javax.annotation.Nullable;
> import android.view.Surface;
>-
> import javax.microedition.khronos.egl.EGL10;
> 
> /**
>  * Holds EGL state and utility methods for handling an egl 1.0 EGLContext, an EGLDisplay,
>  * and an EGLSurface.
>  */
>-public abstract class EglBase {
>+public interface EglBase {
>   // EGL wrapper for an actual EGLContext.
>   public interface Context { long getNativeEglContext(); }
> 
>@@ -85,7 +85,7 @@ public abstract class EglBase {
>    * If |sharedContext| is null, a root context is created. This function will try to create an EGL
>    * 1.4 context if possible, and an EGL 1.0 context otherwise.
>    */
>-  public static EglBase create(Context sharedContext, int[] configAttributes) {
>+  public static EglBase create(@Nullable Context sharedContext, int[] configAttributes) {
>     return (EglBase14.isEGL14Supported()
>                && (sharedContext == null || sharedContext instanceof EglBase14.Context))
>         ? new EglBase14((EglBase14.Context) sharedContext, configAttributes)
>@@ -140,34 +140,34 @@ public abstract class EglBase {
>     return new EglBase14(new EglBase14.Context(sharedContext), configAttributes);
>   }
> 
>-  public abstract void createSurface(Surface surface);
>+  void createSurface(Surface surface);
> 
>   // Create EGLSurface from the Android SurfaceTexture.
>-  public abstract void createSurface(SurfaceTexture surfaceTexture);
>+  void createSurface(SurfaceTexture surfaceTexture);
> 
>   // Create dummy 1x1 pixel buffer surface so the context can be made current.
>-  public abstract void createDummyPbufferSurface();
>+  void createDummyPbufferSurface();
> 
>-  public abstract void createPbufferSurface(int width, int height);
>+  void createPbufferSurface(int width, int height);
> 
>-  public abstract Context getEglBaseContext();
>+  Context getEglBaseContext();
> 
>-  public abstract boolean hasSurface();
>+  boolean hasSurface();
> 
>-  public abstract int surfaceWidth();
>+  int surfaceWidth();
> 
>-  public abstract int surfaceHeight();
>+  int surfaceHeight();
> 
>-  public abstract void releaseSurface();
>+  void releaseSurface();
> 
>-  public abstract void release();
>+  void release();
> 
>-  public abstract void makeCurrent();
>+  void makeCurrent();
> 
>   // Detach the current EGL context, so that it can be made current on another thread.
>-  public abstract void detachCurrent();
>+  void detachCurrent();
> 
>-  public abstract void swapBuffers();
>+  void swapBuffers();
> 
>-  public abstract void swapBuffers(long presentationTimeStampNs);
>+  void swapBuffers(long presentationTimeStampNs);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/EglRenderer.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/EglRenderer.java
>index ba39971e5bd..d0a1d98b2bd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/EglRenderer.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/EglRenderer.java
>@@ -19,18 +19,18 @@ import android.os.HandlerThread;
> import android.os.Looper;
> import android.view.Surface;
> import java.nio.ByteBuffer;
>+import java.text.DecimalFormat;
> import java.util.ArrayList;
> import java.util.Iterator;
>-import java.util.Locale;
> import java.util.concurrent.CountDownLatch;
> import java.util.concurrent.TimeUnit;
>+import javax.annotation.Nullable;
> 
> /**
>- * Implements org.webrtc.VideoRenderer.Callbacks by displaying the video stream on an EGL Surface.
>- * This class is intended to be used as a helper class for rendering on SurfaceViews and
>- * TextureViews.
>+ * Implements VideoSink by displaying the video stream on an EGL Surface. This class is intended to
>+ * be used as a helper class for rendering on SurfaceViews and TextureViews.
>  */
>-public class EglRenderer implements VideoRenderer.Callbacks, VideoSink {
>+public class EglRenderer implements VideoSink {
>   private static final String TAG = "EglRenderer";
>   private static final long LOG_INTERVAL_SEC = 4;
> 
>@@ -84,7 +84,7 @@ public class EglRenderer implements VideoRenderer.Callbacks, VideoSink {
>   // |renderThreadHandler| is a handler for communicating with |renderThread|, and is synchronized
>   // on |handlerLock|.
>   private final Object handlerLock = new Object();
>-  private Handler renderThreadHandler;
>+  @Nullable private Handler renderThreadHandler;
> 
>   private final ArrayList<FrameListenerAndParams> frameListeners = new ArrayList<>();
> 
>@@ -98,14 +98,14 @@ public class EglRenderer implements VideoRenderer.Callbacks, VideoSink {
> 
>   // EGL and GL resources for drawing YUV/OES textures. After initilization, these are only accessed
>   // from the render thread.
>-  private EglBase eglBase;
>+  @Nullable private EglBase eglBase;
>   private final VideoFrameDrawer frameDrawer = new VideoFrameDrawer();
>-  private RendererCommon.GlDrawer drawer;
>+  @Nullable private RendererCommon.GlDrawer drawer;
>   private final Matrix drawMatrix = new Matrix();
> 
>   // Pending frame to render. Serves as a queue with size 1. Synchronized on |frameLock|.
>   private final Object frameLock = new Object();
>-  private VideoFrame pendingFrame;
>+  @Nullable private VideoFrame pendingFrame;
> 
>   // These variables are synchronized on |layoutLock|.
>   private final Object layoutLock = new Object();
>@@ -130,7 +130,8 @@ public class EglRenderer implements VideoRenderer.Callbacks, VideoSink {
>   private long renderSwapBufferTimeNs;
> 
>   // Used for bitmap capturing.
>-  private GlTextureFrameBuffer bitmapTextureFramebuffer;
>+  private final GlTextureFrameBuffer bitmapTextureFramebuffer =
>+      new GlTextureFrameBuffer(GLES20.GL_RGBA);
> 
>   private final Runnable logStatisticsRunnable = new Runnable() {
>     @Override
>@@ -162,7 +163,7 @@ public class EglRenderer implements VideoRenderer.Callbacks, VideoSink {
>    * |drawer|. It is allowed to call init() to reinitialize the renderer after a previous
>    * init()/release() cycle.
>    */
>-  public void init(final EglBase.Context sharedContext, final int[] configAttributes,
>+  public void init(@Nullable final EglBase.Context sharedContext, final int[] configAttributes,
>       RendererCommon.GlDrawer drawer) {
>     synchronized (handlerLock) {
>       if (renderThreadHandler != null) {
>@@ -232,10 +233,7 @@ public class EglRenderer implements VideoRenderer.Callbacks, VideoSink {
>           drawer = null;
>         }
>         frameDrawer.release();
>-        if (bitmapTextureFramebuffer != null) {
>-          bitmapTextureFramebuffer.release();
>-          bitmapTextureFramebuffer = null;
>-        }
>+        bitmapTextureFramebuffer.release();
>         if (eglBase != null) {
>           logD("eglBase detach and release.");
>           eglBase.detachCurrent();
>@@ -385,7 +383,7 @@ public class EglRenderer implements VideoRenderer.Callbacks, VideoSink {
>    *                          FPS reduction.
>    */
>   public void addFrameListener(final FrameListener listener, final float scale,
>-      final RendererCommon.GlDrawer drawerParam, final boolean applyFpsReduction) {
>+      @Nullable final RendererCommon.GlDrawer drawerParam, final boolean applyFpsReduction) {
>     postToRenderThread(() -> {
>       final RendererCommon.GlDrawer listenerDrawer = drawerParam == null ? drawer : drawerParam;
>       frameListeners.add(
>@@ -422,14 +420,6 @@ public class EglRenderer implements VideoRenderer.Callbacks, VideoSink {
>     ThreadUtils.awaitUninterruptibly(latch);
>   }
> 
>-  // VideoRenderer.Callbacks interface.
>-  @Override
>-  public void renderFrame(VideoRenderer.I420Frame frame) {
>-    VideoFrame videoFrame = frame.toVideoFrame();
>-    onFrame(videoFrame);
>-    videoFrame.release();
>-  }
>-
>   // VideoSink interface.
>   @Override
>   public void onFrame(VideoFrame frame) {
>@@ -636,9 +626,6 @@ public class EglRenderer implements VideoRenderer.Callbacks, VideoSink {
>         continue;
>       }
> 
>-      if (bitmapTextureFramebuffer == null) {
>-        bitmapTextureFramebuffer = new GlTextureFrameBuffer(GLES20.GL_RGBA);
>-      }
>       bitmapTextureFramebuffer.setSize(scaledWidth, scaledHeight);
> 
>       GLES20.glBindFramebuffer(GLES20.GL_FRAMEBUFFER, bitmapTextureFramebuffer.getFrameBufferId());
>@@ -669,6 +656,7 @@ public class EglRenderer implements VideoRenderer.Callbacks, VideoSink {
>   }
> 
>   private void logStatistics() {
>+    final DecimalFormat fpsFormat = new DecimalFormat("#.0");
>     final long currentTimeNs = System.nanoTime();
>     synchronized (statisticsLock) {
>       final long elapsedTimeNs = currentTimeNs - statisticsStartTimeNs;
>@@ -680,7 +668,7 @@ public class EglRenderer implements VideoRenderer.Callbacks, VideoSink {
>           + " Frames received: " + framesReceived + "."
>           + " Dropped: " + framesDropped + "."
>           + " Rendered: " + framesRendered + "."
>-          + " Render fps: " + String.format(Locale.US, "%.1f", renderFps) + "."
>+          + " Render fps: " + fpsFormat.format(renderFps) + "."
>           + " Average render time: " + averageTimeAsString(renderTimeNs, framesRendered) + "."
>           + " Average swapBuffer time: "
>           + averageTimeAsString(renderSwapBufferTimeNs, framesRendered) + ".");
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/FecControllerFactoryFactoryInterface.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/FecControllerFactoryFactoryInterface.java
>new file mode 100644
>index 00000000000..6d39390f72d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/FecControllerFactoryFactoryInterface.java
>@@ -0,0 +1,22 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc;
>+
>+/**
>+ * Factory for creating webrtc::FecControllerFactory instances.
>+ */
>+public interface FecControllerFactoryFactoryInterface {
>+  /**
>+   * Dynamically allocates a webrtc::FecControllerFactory instance and returns a pointer to it.
>+   * The caller takes ownership of the object.
>+   */
>+  public long createNative();
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/FileVideoCapturer.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/FileVideoCapturer.java
>index 331f9a7736a..4efed36a5f8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/FileVideoCapturer.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/FileVideoCapturer.java
>@@ -15,6 +15,7 @@ import android.os.SystemClock;
> import java.io.IOException;
> import java.io.RandomAccessFile;
> import java.nio.ByteBuffer;
>+import java.nio.channels.FileChannel;
> import java.nio.charset.Charset;
> import java.util.Timer;
> import java.util.TimerTask;
>@@ -29,21 +30,25 @@ public class FileVideoCapturer implements VideoCapturer {
>   /**
>    * Read video data from file for the .y4m container.
>    */
>+  @SuppressWarnings("StringSplitter")
>   private static class VideoReaderY4M implements VideoReader {
>     private static final String TAG = "VideoReaderY4M";
>     private static final String Y4M_FRAME_DELIMETER = "FRAME";
>+    private static final int FRAME_DELIMETER_LENGTH = Y4M_FRAME_DELIMETER.length() + 1;
> 
>     private final int frameWidth;
>     private final int frameHeight;
>     // First char after header
>     private final long videoStart;
>-    private final RandomAccessFile mediaFileStream;
>+    private final RandomAccessFile mediaFile;
>+    private final FileChannel mediaFileChannel;
> 
>     public VideoReaderY4M(String file) throws IOException {
>-      mediaFileStream = new RandomAccessFile(file, "r");
>+      mediaFile = new RandomAccessFile(file, "r");
>+      mediaFileChannel = mediaFile.getChannel();
>       StringBuilder builder = new StringBuilder();
>       for (;;) {
>-        int c = mediaFileStream.read();
>+        int c = mediaFile.read();
>         if (c == -1) {
>           // End of file reached.
>           throw new RuntimeException("Found end of file before end of header for file: " + file);
>@@ -54,7 +59,7 @@ public class FileVideoCapturer implements VideoCapturer {
>         }
>         builder.append((char) c);
>       }
>-      videoStart = mediaFileStream.getFilePointer();
>+      videoStart = mediaFileChannel.position();
>       String header = builder.toString();
>       String[] headerTokens = header.split("[ ]");
>       int w = 0;
>@@ -100,24 +105,24 @@ public class FileVideoCapturer implements VideoCapturer {
>       final int sizeV = chromaHeight * buffer.getStrideV();
> 
>       try {
>-        byte[] frameDelim = new byte[Y4M_FRAME_DELIMETER.length() + 1];
>-        if (mediaFileStream.read(frameDelim) < frameDelim.length) {
>+        ByteBuffer frameDelim = ByteBuffer.allocate(FRAME_DELIMETER_LENGTH);
>+        if (mediaFileChannel.read(frameDelim) < FRAME_DELIMETER_LENGTH) {
>           // We reach end of file, loop
>-          mediaFileStream.seek(videoStart);
>-          if (mediaFileStream.read(frameDelim) < frameDelim.length) {
>+          mediaFileChannel.position(videoStart);
>+          if (mediaFileChannel.read(frameDelim) < FRAME_DELIMETER_LENGTH) {
>             throw new RuntimeException("Error looping video");
>           }
>         }
>-        String frameDelimStr = new String(frameDelim, Charset.forName("US-ASCII"));
>+        String frameDelimStr = new String(frameDelim.array(), Charset.forName("US-ASCII"));
>         if (!frameDelimStr.equals(Y4M_FRAME_DELIMETER + "\n")) {
>           throw new RuntimeException(
>               "Frames should be delimited by FRAME plus newline, found delimter was: '"
>               + frameDelimStr + "'");
>         }
> 
>-        mediaFileStream.readFully(dataY.array(), dataY.arrayOffset(), sizeY);
>-        mediaFileStream.readFully(dataU.array(), dataU.arrayOffset(), sizeU);
>-        mediaFileStream.readFully(dataV.array(), dataV.arrayOffset(), sizeV);
>+        mediaFileChannel.read(dataY);
>+        mediaFileChannel.read(dataU);
>+        mediaFileChannel.read(dataV);
>       } catch (IOException e) {
>         throw new RuntimeException(e);
>       }
>@@ -128,7 +133,8 @@ public class FileVideoCapturer implements VideoCapturer {
>     @Override
>     public void close() {
>       try {
>-        mediaFileStream.close();
>+        // Closing a file also closes the channel.
>+        mediaFile.close();
>       } catch (IOException e) {
>         Logging.e(TAG, "Problem closing file", e);
>       }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/GlRectDrawer.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/GlRectDrawer.java
>index 44787f7e279..d1fbd1b7bce 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/GlRectDrawer.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/GlRectDrawer.java
>@@ -10,201 +10,22 @@
> 
> package org.webrtc;
> 
>-import android.opengl.GLES11Ext;
>-import android.opengl.GLES20;
>-import java.nio.FloatBuffer;
>-import java.util.IdentityHashMap;
>-import java.util.Map;
>-
>-/**
>- * Helper class to draw an opaque quad on the target viewport location. Rotation, mirror, and
>- * cropping is specified using a 4x4 texture coordinate transform matrix. The frame input can either
>- * be an OES texture or YUV textures in I420 format. The GL state must be preserved between draw
>- * calls, this is intentional to maximize performance. The function release() must be called
>- * manually to free the resources held by this object.
>- */
>-public class GlRectDrawer implements RendererCommon.GlDrawer {
>-  // clang-format off
>-  // Simple vertex shader, used for both YUV and OES.
>-  private static final String VERTEX_SHADER_STRING =
>-        "varying vec2 interp_tc;\n"
>-      + "attribute vec4 in_pos;\n"
>-      + "attribute vec4 in_tc;\n"
>-      + "\n"
>-      + "uniform mat4 texMatrix;\n"
>-      + "\n"
>-      + "void main() {\n"
>-      + "    gl_Position = in_pos;\n"
>-      + "    interp_tc = (texMatrix * in_tc).xy;\n"
>-      + "}\n";
>-
>-  private static final String YUV_FRAGMENT_SHADER_STRING =
>-        "precision mediump float;\n"
>-      + "varying vec2 interp_tc;\n"
>-      + "\n"
>-      + "uniform sampler2D y_tex;\n"
>-      + "uniform sampler2D u_tex;\n"
>-      + "uniform sampler2D v_tex;\n"
>-      + "\n"
>-      + "void main() {\n"
>-      // CSC according to http://www.fourcc.org/fccyvrgb.php
>-      + "  float y = texture2D(y_tex, interp_tc).r;\n"
>-      + "  float u = texture2D(u_tex, interp_tc).r - 0.5;\n"
>-      + "  float v = texture2D(v_tex, interp_tc).r - 0.5;\n"
>-      + "  gl_FragColor = vec4(y + 1.403 * v, "
>-      + "                      y - 0.344 * u - 0.714 * v, "
>-      + "                      y + 1.77 * u, 1);\n"
>-      + "}\n";
>-
>-  private static final String RGB_FRAGMENT_SHADER_STRING =
>-        "precision mediump float;\n"
>-      + "varying vec2 interp_tc;\n"
>-      + "\n"
>-      + "uniform sampler2D rgb_tex;\n"
>-      + "\n"
>-      + "void main() {\n"
>-      + "  gl_FragColor = texture2D(rgb_tex, interp_tc);\n"
>+/** Simplest possible GL shader that just draws frames as opaque quads. */
>+public class GlRectDrawer extends GlGenericDrawer {
>+  private static final String FRAGMENT_SHADER = "void main() {\n"
>+      + "  gl_FragColor = sample(tc);\n"
>       + "}\n";
> 
>-  private static final String OES_FRAGMENT_SHADER_STRING =
>-        "#extension GL_OES_EGL_image_external : require\n"
>-      + "precision mediump float;\n"
>-      + "varying vec2 interp_tc;\n"
>-      + "\n"
>-      + "uniform samplerExternalOES oes_tex;\n"
>-      + "\n"
>-      + "void main() {\n"
>-      + "  gl_FragColor = texture2D(oes_tex, interp_tc);\n"
>-      + "}\n";
>-  // clang-format on
>-
>-  // Vertex coordinates in Normalized Device Coordinates, i.e. (-1, -1) is bottom-left and (1, 1) is
>-  // top-right.
>-  private static final FloatBuffer FULL_RECTANGLE_BUF = GlUtil.createFloatBuffer(new float[] {
>-      -1.0f, -1.0f, // Bottom left.
>-      1.0f, -1.0f, // Bottom right.
>-      -1.0f, 1.0f, // Top left.
>-      1.0f, 1.0f, // Top right.
>-  });
>-
>-  // Texture coordinates - (0, 0) is bottom-left and (1, 1) is top-right.
>-  private static final FloatBuffer FULL_RECTANGLE_TEX_BUF = GlUtil.createFloatBuffer(new float[] {
>-      0.0f, 0.0f, // Bottom left.
>-      1.0f, 0.0f, // Bottom right.
>-      0.0f, 1.0f, // Top left.
>-      1.0f, 1.0f // Top right.
>-  });
>-
>-  private static class Shader {
>-    public final GlShader glShader;
>-    public final int texMatrixLocation;
>-
>-    public Shader(String fragmentShader) {
>-      this.glShader = new GlShader(VERTEX_SHADER_STRING, fragmentShader);
>-      this.texMatrixLocation = glShader.getUniformLocation("texMatrix");
>-    }
>-  }
>-
>-  // The keys are one of the fragments shaders above.
>-  private final Map<String, Shader> shaders = new IdentityHashMap<String, Shader>();
>-
>-  /**
>-   * Draw an OES texture frame with specified texture transformation matrix. Required resources are
>-   * allocated at the first call to this function.
>-   */
>-  @Override
>-  public void drawOes(int oesTextureId, float[] texMatrix, int frameWidth, int frameHeight,
>-      int viewportX, int viewportY, int viewportWidth, int viewportHeight) {
>-    prepareShader(OES_FRAGMENT_SHADER_STRING, texMatrix);
>-    GLES20.glActiveTexture(GLES20.GL_TEXTURE0);
>-    // updateTexImage() may be called from another thread in another EGL context, so we need to
>-    // bind/unbind the texture in each draw call so that GLES understads it's a new texture.
>-    GLES20.glBindTexture(GLES11Ext.GL_TEXTURE_EXTERNAL_OES, oesTextureId);
>-    drawRectangle(viewportX, viewportY, viewportWidth, viewportHeight);
>-    GLES20.glBindTexture(GLES11Ext.GL_TEXTURE_EXTERNAL_OES, 0);
>-  }
>-
>-  /**
>-   * Draw a RGB(A) texture frame with specified texture transformation matrix. Required resources
>-   * are allocated at the first call to this function.
>-   */
>-  @Override
>-  public void drawRgb(int textureId, float[] texMatrix, int frameWidth, int frameHeight,
>-      int viewportX, int viewportY, int viewportWidth, int viewportHeight) {
>-    prepareShader(RGB_FRAGMENT_SHADER_STRING, texMatrix);
>-    GLES20.glActiveTexture(GLES20.GL_TEXTURE0);
>-    GLES20.glBindTexture(GLES20.GL_TEXTURE_2D, textureId);
>-    drawRectangle(viewportX, viewportY, viewportWidth, viewportHeight);
>-    // Unbind the texture as a precaution.
>-    GLES20.glBindTexture(GLES20.GL_TEXTURE_2D, 0);
>-  }
>-
>-  /**
>-   * Draw a YUV frame with specified texture transformation matrix. Required resources are
>-   * allocated at the first call to this function.
>-   */
>-  @Override
>-  public void drawYuv(int[] yuvTextures, float[] texMatrix, int frameWidth, int frameHeight,
>-      int viewportX, int viewportY, int viewportWidth, int viewportHeight) {
>-    prepareShader(YUV_FRAGMENT_SHADER_STRING, texMatrix);
>-    // Bind the textures.
>-    for (int i = 0; i < 3; ++i) {
>-      GLES20.glActiveTexture(GLES20.GL_TEXTURE0 + i);
>-      GLES20.glBindTexture(GLES20.GL_TEXTURE_2D, yuvTextures[i]);
>-    }
>-    drawRectangle(viewportX, viewportY, viewportWidth, viewportHeight);
>-    // Unbind the textures as a precaution..
>-    for (int i = 0; i < 3; ++i) {
>-      GLES20.glActiveTexture(GLES20.GL_TEXTURE0 + i);
>-      GLES20.glBindTexture(GLES20.GL_TEXTURE_2D, 0);
>-    }
>-  }
>-
>-  private void drawRectangle(int x, int y, int width, int height) {
>-    // Draw quad.
>-    GLES20.glViewport(x, y, width, height);
>-    GLES20.glDrawArrays(GLES20.GL_TRIANGLE_STRIP, 0, 4);
>-  }
>+  private static class ShaderCallbacks implements GlGenericDrawer.ShaderCallbacks {
>+    @Override
>+    public void onNewShader(GlShader shader) {}
> 
>-  private void prepareShader(String fragmentShader, float[] texMatrix) {
>-    final Shader shader;
>-    if (shaders.containsKey(fragmentShader)) {
>-      shader = shaders.get(fragmentShader);
>-    } else {
>-      // Lazy allocation.
>-      shader = new Shader(fragmentShader);
>-      shaders.put(fragmentShader, shader);
>-      shader.glShader.useProgram();
>-      // Initialize fragment shader uniform values.
>-      if (YUV_FRAGMENT_SHADER_STRING.equals(fragmentShader)) {
>-        GLES20.glUniform1i(shader.glShader.getUniformLocation("y_tex"), 0);
>-        GLES20.glUniform1i(shader.glShader.getUniformLocation("u_tex"), 1);
>-        GLES20.glUniform1i(shader.glShader.getUniformLocation("v_tex"), 2);
>-      } else if (RGB_FRAGMENT_SHADER_STRING.equals(fragmentShader)) {
>-        GLES20.glUniform1i(shader.glShader.getUniformLocation("rgb_tex"), 0);
>-      } else if (OES_FRAGMENT_SHADER_STRING.equals(fragmentShader)) {
>-        GLES20.glUniform1i(shader.glShader.getUniformLocation("oes_tex"), 0);
>-      } else {
>-        throw new IllegalStateException("Unknown fragment shader: " + fragmentShader);
>-      }
>-      GlUtil.checkNoGLES2Error("Initialize fragment shader uniform values.");
>-      // Initialize vertex shader attributes.
>-      shader.glShader.setVertexAttribArray("in_pos", 2, FULL_RECTANGLE_BUF);
>-      shader.glShader.setVertexAttribArray("in_tc", 2, FULL_RECTANGLE_TEX_BUF);
>-    }
>-    shader.glShader.useProgram();
>-    // Copy the texture transformation matrix over.
>-    GLES20.glUniformMatrix4fv(shader.texMatrixLocation, 1, false, texMatrix, 0);
>+    @Override
>+    public void onPrepareShader(GlShader shader, float[] texMatrix, int frameWidth, int frameHeight,
>+        int viewportWidth, int viewportHeight) {}
>   }
> 
>-  /**
>-   * Release all GLES resources. This needs to be done manually, otherwise the resources are leaked.
>-   */
>-  @Override
>-  public void release() {
>-    for (Shader shader : shaders.values()) {
>-      shader.glShader.release();
>-    }
>-    shaders.clear();
>+  public GlRectDrawer() {
>+    super(FRAGMENT_SHADER, new ShaderCallbacks());
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/GlShader.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/GlShader.java
>index c51a0da3147..db5c630663a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/GlShader.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/GlShader.java
>@@ -29,7 +29,7 @@ public class GlShader {
>     GLES20.glGetShaderiv(shader, GLES20.GL_COMPILE_STATUS, compileStatus, 0);
>     if (compileStatus[0] != GLES20.GL_TRUE) {
>       Logging.e(
>-          TAG, "Could not compile shader " + shaderType + ":" + GLES20.glGetShaderInfoLog(shader));
>+          TAG, "Compile error " + GLES20.glGetShaderInfoLog(shader) + " in shader:\n" + source);
>       throw new RuntimeException(GLES20.glGetShaderInfoLog(shader));
>     }
>     GlUtil.checkNoGLES2Error("compileShader");
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/GlTextureFrameBuffer.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/GlTextureFrameBuffer.java
>index 164dff29cb1..b906fe56e03 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/GlTextureFrameBuffer.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/GlTextureFrameBuffer.java
>@@ -15,13 +15,13 @@ import android.opengl.GLES20;
> /**
>  * Helper class for handling OpenGL framebuffer with only color attachment and no depth or stencil
>  * buffer. Intended for simple tasks such as texture copy, texture downscaling, and texture color
>- * conversion.
>+ * conversion. This class is not thread safe and must be used by a thread with an active GL context.
>  */
> // TODO(magjed): Add unittests for this class.
> public class GlTextureFrameBuffer {
>-  private final int frameBufferId;
>-  private final int textureId;
>   private final int pixelFormat;
>+  private int frameBufferId;
>+  private int textureId;
>   private int width;
>   private int height;
> 
>@@ -39,16 +39,8 @@ public class GlTextureFrameBuffer {
>       default:
>         throw new IllegalArgumentException("Invalid pixel format: " + pixelFormat);
>     }
>-
>-    // Create texture.
>-    textureId = GlUtil.generateTexture(GLES20.GL_TEXTURE_2D);
>     this.width = 0;
>     this.height = 0;
>-
>-    // Create framebuffer object.
>-    final int frameBuffers[] = new int[1];
>-    GLES20.glGenFramebuffers(1, frameBuffers, 0);
>-    frameBufferId = frameBuffers[0];
>   }
> 
>   /**
>@@ -57,7 +49,7 @@ public class GlTextureFrameBuffer {
>    * least once before using the framebuffer. May be called multiple times to change size.
>    */
>   public void setSize(int width, int height) {
>-    if (width == 0 || height == 0) {
>+    if (width <= 0 || height <= 0) {
>       throw new IllegalArgumentException("Invalid size: " + width + "x" + height);
>     }
>     if (width == this.width && height == this.height) {
>@@ -65,6 +57,15 @@ public class GlTextureFrameBuffer {
>     }
>     this.width = width;
>     this.height = height;
>+    // Lazy allocation the first time setSize() is called.
>+    if (textureId == 0) {
>+      textureId = GlUtil.generateTexture(GLES20.GL_TEXTURE_2D);
>+    }
>+    if (frameBufferId == 0) {
>+      final int frameBuffers[] = new int[1];
>+      GLES20.glGenFramebuffers(1, frameBuffers, 0);
>+      frameBufferId = frameBuffers[0];
>+    }
> 
>     // Allocate texture.
>     GLES20.glActiveTexture(GLES20.GL_TEXTURE0);
>@@ -96,10 +97,12 @@ public class GlTextureFrameBuffer {
>     return height;
>   }
> 
>+  /** Gets the OpenGL frame buffer id. This value is only valid after setSize() has been called. */
>   public int getFrameBufferId() {
>     return frameBufferId;
>   }
> 
>+  /** Gets the OpenGL texture id. This value is only valid after setSize() has been called. */
>   public int getTextureId() {
>     return textureId;
>   }
>@@ -110,7 +113,9 @@ public class GlTextureFrameBuffer {
>    */
>   public void release() {
>     GLES20.glDeleteTextures(1, new int[] {textureId}, 0);
>+    textureId = 0;
>     GLES20.glDeleteFramebuffers(1, new int[] {frameBufferId}, 0);
>+    frameBufferId = 0;
>     width = 0;
>     height = 0;
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/HardwareVideoDecoderFactory.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/HardwareVideoDecoderFactory.java
>index c14ec5f1d8a..3e885830e4c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/HardwareVideoDecoderFactory.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/HardwareVideoDecoderFactory.java
>@@ -19,6 +19,9 @@ import android.media.MediaCodecInfo;
> import android.media.MediaCodecInfo.CodecCapabilities;
> import android.media.MediaCodecList;
> import android.os.Build;
>+import java.util.ArrayList;
>+import java.util.List;
>+import javax.annotation.Nullable;
> 
> /** Factory for Android hardware VideoDecoders. */
> @SuppressWarnings("deprecation") // API level 16 requires use of deprecated methods.
>@@ -26,7 +29,6 @@ public class HardwareVideoDecoderFactory implements VideoDecoderFactory {
>   private static final String TAG = "HardwareVideoDecoderFactory";
> 
>   private final EglBase.Context sharedContext;
>-  private final boolean fallbackToSoftware;
> 
>   /** Creates a HardwareVideoDecoderFactory that does not use surface textures. */
>   @Deprecated // Not removed yet to avoid breaking callers.
>@@ -39,38 +41,49 @@ public class HardwareVideoDecoderFactory implements VideoDecoderFactory {
>    * shared context.  The context may be null.  If it is null, then surface support is disabled.
>    */
>   public HardwareVideoDecoderFactory(EglBase.Context sharedContext) {
>-    this(sharedContext, true /* fallbackToSoftware */);
>-  }
>-
>-  HardwareVideoDecoderFactory(EglBase.Context sharedContext, boolean fallbackToSoftware) {
>     this.sharedContext = sharedContext;
>-    this.fallbackToSoftware = fallbackToSoftware;
>   }
> 
>+  @Nullable
>   @Override
>-  public VideoDecoder createDecoder(String codecType) {
>-    VideoCodecType type = VideoCodecType.valueOf(codecType);
>+  public VideoDecoder createDecoder(VideoCodecInfo codecType) {
>+    VideoCodecType type = VideoCodecType.valueOf(codecType.getName());
>     MediaCodecInfo info = findCodecForType(type);
> 
>     if (info == null) {
>-      // No hardware support for this type.
>-      // TODO(andersc): This is for backwards compatibility. Remove when clients have migrated to
>-      // new DefaultVideoEncoderFactory.
>-      if (fallbackToSoftware) {
>-        SoftwareVideoDecoderFactory softwareVideoDecoderFactory = new SoftwareVideoDecoderFactory();
>-        return softwareVideoDecoderFactory.createDecoder(codecType);
>-      } else {
>-        return null;
>-      }
>+      return null;
>     }
> 
>     CodecCapabilities capabilities = info.getCapabilitiesForType(type.mimeType());
>-    return new HardwareVideoDecoder(info.getName(), type,
>+    return new HardwareVideoDecoder(new MediaCodecWrapperFactoryImpl(), info.getName(), type,
>         MediaCodecUtils.selectColorFormat(MediaCodecUtils.DECODER_COLOR_FORMATS, capabilities),
>         sharedContext);
>   }
> 
>-  private MediaCodecInfo findCodecForType(VideoCodecType type) {
>+  @Override
>+  public VideoCodecInfo[] getSupportedCodecs() {
>+    List<VideoCodecInfo> supportedCodecInfos = new ArrayList<VideoCodecInfo>();
>+    // Generate a list of supported codecs in order of preference:
>+    // VP8, VP9, H264 (high profile), and H264 (baseline profile).
>+    for (VideoCodecType type :
>+        new VideoCodecType[] {VideoCodecType.VP8, VideoCodecType.VP9, VideoCodecType.H264}) {
>+      MediaCodecInfo codec = findCodecForType(type);
>+      if (codec != null) {
>+        String name = type.name();
>+        if (type == VideoCodecType.H264 && isH264HighProfileSupported(codec)) {
>+          supportedCodecInfos.add(new VideoCodecInfo(
>+              name, MediaCodecUtils.getCodecProperties(type, /* highProfile= */ true)));
>+        }
>+
>+        supportedCodecInfos.add(new VideoCodecInfo(
>+            name, MediaCodecUtils.getCodecProperties(type, /* highProfile= */ false)));
>+      }
>+    }
>+
>+    return supportedCodecInfos.toArray(new VideoCodecInfo[supportedCodecInfos.size()]);
>+  }
>+
>+  private @Nullable MediaCodecInfo findCodecForType(VideoCodecType type) {
>     // HW decoding is not supported on builds before KITKAT.
>     if (Build.VERSION.SDK_INT < Build.VERSION_CODES.KITKAT) {
>       return null;
>@@ -127,4 +140,17 @@ public class HardwareVideoDecoderFactory implements VideoDecoderFactory {
>         return false;
>     }
>   }
>+
>+  private boolean isH264HighProfileSupported(MediaCodecInfo info) {
>+    String name = info.getName();
>+    // Support H.264 HP decoding on QCOM chips for Android L and above.
>+    if (Build.VERSION.SDK_INT >= Build.VERSION_CODES.LOLLIPOP && name.startsWith(QCOM_PREFIX)) {
>+      return true;
>+    }
>+    // Support H.264 HP decoding on Exynos chips for Android M and above.
>+    if (Build.VERSION.SDK_INT >= Build.VERSION_CODES.M && name.startsWith(EXYNOS_PREFIX)) {
>+      return true;
>+    }
>+    return false;
>+  }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/HardwareVideoEncoderFactory.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/HardwareVideoEncoderFactory.java
>index 25b7c77e73d..0664f5ebe12 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/HardwareVideoEncoderFactory.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/HardwareVideoEncoderFactory.java
>@@ -19,9 +19,9 @@ import android.media.MediaCodecList;
> import android.os.Build;
> import java.util.ArrayList;
> import java.util.Arrays;
>-import java.util.HashMap;
> import java.util.List;
> import java.util.Map;
>+import javax.annotation.Nullable;
> 
> /** Factory for android hardware video encoders. */
> @SuppressWarnings("deprecation") // API 16 requires the use of deprecated methods.
>@@ -39,19 +39,12 @@ public class HardwareVideoEncoderFactory implements VideoEncoderFactory {
>   private static final List<String> H264_HW_EXCEPTION_MODELS =
>       Arrays.asList("SAMSUNG-SGH-I337", "Nexus 7", "Nexus 4");
> 
>-  private final EglBase14.Context sharedContext;
>+  @Nullable private final EglBase14.Context sharedContext;
>   private final boolean enableIntelVp8Encoder;
>   private final boolean enableH264HighProfile;
>-  private final boolean fallbackToSoftware;
> 
>   public HardwareVideoEncoderFactory(
>       EglBase.Context sharedContext, boolean enableIntelVp8Encoder, boolean enableH264HighProfile) {
>-    this(
>-        sharedContext, enableIntelVp8Encoder, enableH264HighProfile, true /* fallbackToSoftware */);
>-  }
>-
>-  HardwareVideoEncoderFactory(EglBase.Context sharedContext, boolean enableIntelVp8Encoder,
>-      boolean enableH264HighProfile, boolean fallbackToSoftware) {
>     // Texture mode requires EglBase14.
>     if (sharedContext instanceof EglBase14.Context) {
>       this.sharedContext = (EglBase14.Context) sharedContext;
>@@ -61,7 +54,6 @@ public class HardwareVideoEncoderFactory implements VideoEncoderFactory {
>     }
>     this.enableIntelVp8Encoder = enableIntelVp8Encoder;
>     this.enableH264HighProfile = enableH264HighProfile;
>-    this.fallbackToSoftware = fallbackToSoftware;
>   }
> 
>   @Deprecated
>@@ -69,21 +61,14 @@ public class HardwareVideoEncoderFactory implements VideoEncoderFactory {
>     this(null, enableIntelVp8Encoder, enableH264HighProfile);
>   }
> 
>+  @Nullable
>   @Override
>   public VideoEncoder createEncoder(VideoCodecInfo input) {
>     VideoCodecType type = VideoCodecType.valueOf(input.name);
>     MediaCodecInfo info = findCodecForType(type);
> 
>     if (info == null) {
>-      // No hardware support for this type.
>-      // TODO(andersc): This is for backwards compatibility. Remove when clients have migrated to
>-      // new DefaultVideoEncoderFactory.
>-      if (fallbackToSoftware) {
>-        SoftwareVideoEncoderFactory softwareVideoEncoderFactory = new SoftwareVideoEncoderFactory();
>-        return softwareVideoEncoderFactory.createEncoder(input);
>-      } else {
>-        return null;
>-      }
>+      return null;
>     }
> 
>     String codecName = info.getName();
>@@ -94,18 +79,23 @@ public class HardwareVideoEncoderFactory implements VideoEncoderFactory {
>         MediaCodecUtils.ENCODER_COLOR_FORMATS, info.getCapabilitiesForType(mime));
> 
>     if (type == VideoCodecType.H264) {
>-      boolean isHighProfile = isSameH264Profile(input.params, getCodecProperties(type, true))
>-          && isH264HighProfileSupported(info);
>-      boolean isBaselineProfile = isSameH264Profile(input.params, getCodecProperties(type, false));
>+      boolean isHighProfile = H264Utils.isSameH264Profile(
>+          input.params, MediaCodecUtils.getCodecProperties(type, /* highProfile= */ true));
>+      boolean isBaselineProfile = H264Utils.isSameH264Profile(
>+          input.params, MediaCodecUtils.getCodecProperties(type, /* highProfile= */ false));
> 
>       if (!isHighProfile && !isBaselineProfile) {
>         return null;
>       }
>+      if (isHighProfile && !isH264HighProfileSupported(info)) {
>+        return null;
>+      }
>     }
> 
>-    return new HardwareVideoEncoder(codecName, type, surfaceColorFormat, yuvColorFormat,
>-        input.params, getKeyFrameIntervalSec(type), getForcedKeyFrameIntervalMs(type, codecName),
>-        createBitrateAdjuster(type, codecName), sharedContext);
>+    return new HardwareVideoEncoder(new MediaCodecWrapperFactoryImpl(), codecName, type,
>+        surfaceColorFormat, yuvColorFormat, input.params, getKeyFrameIntervalSec(type),
>+        getForcedKeyFrameIntervalMs(type, codecName), createBitrateAdjuster(type, codecName),
>+        sharedContext);
>   }
> 
>   @Override
>@@ -118,28 +108,22 @@ public class HardwareVideoEncoderFactory implements VideoEncoderFactory {
>       MediaCodecInfo codec = findCodecForType(type);
>       if (codec != null) {
>         String name = type.name();
>+        // TODO(sakal): Always add H264 HP once WebRTC correctly removes codecs that are not
>+        // supported by the decoder.
>         if (type == VideoCodecType.H264 && isH264HighProfileSupported(codec)) {
>-          supportedCodecInfos.add(new VideoCodecInfo(name, getCodecProperties(type, true)));
>+          supportedCodecInfos.add(new VideoCodecInfo(
>+              name, MediaCodecUtils.getCodecProperties(type, /* highProfile= */ true)));
>         }
> 
>-        supportedCodecInfos.add(new VideoCodecInfo(name, getCodecProperties(type, false)));
>-      }
>-    }
>-
>-    // TODO(andersc): This is for backwards compatibility. Remove when clients have migrated to
>-    // new DefaultVideoEncoderFactory.
>-    if (fallbackToSoftware) {
>-      for (VideoCodecInfo info : SoftwareVideoEncoderFactory.supportedCodecs()) {
>-        if (!supportedCodecInfos.contains(info)) {
>-          supportedCodecInfos.add(info);
>-        }
>+        supportedCodecInfos.add(new VideoCodecInfo(
>+            name, MediaCodecUtils.getCodecProperties(type, /* highProfile= */ false)));
>       }
>     }
> 
>     return supportedCodecInfos.toArray(new VideoCodecInfo[supportedCodecInfos.size()]);
>   }
> 
>-  private MediaCodecInfo findCodecForType(VideoCodecType type) {
>+  private @Nullable MediaCodecInfo findCodecForType(VideoCodecType type) {
>     for (int i = 0; i < MediaCodecList.getCodecCount(); ++i) {
>       MediaCodecInfo info = null;
>       try {
>@@ -259,27 +243,7 @@ public class HardwareVideoEncoderFactory implements VideoEncoderFactory {
>   }
> 
>   private boolean isH264HighProfileSupported(MediaCodecInfo info) {
>-    return enableH264HighProfile && info.getName().startsWith(QCOM_PREFIX);
>+    return enableH264HighProfile && Build.VERSION.SDK_INT > Build.VERSION_CODES.M
>+        && info.getName().startsWith(EXYNOS_PREFIX);
>   }
>-
>-  private Map<String, String> getCodecProperties(VideoCodecType type, boolean highProfile) {
>-    switch (type) {
>-      case VP8:
>-      case VP9:
>-        return new HashMap<String, String>();
>-      case H264:
>-        Map<String, String> properties = new HashMap<>();
>-        properties.put(VideoCodecInfo.H264_FMTP_LEVEL_ASYMMETRY_ALLOWED, "1");
>-        properties.put(VideoCodecInfo.H264_FMTP_PACKETIZATION_MODE, "1");
>-        properties.put(VideoCodecInfo.H264_FMTP_PROFILE_LEVEL_ID,
>-            highProfile ? VideoCodecInfo.H264_CONSTRAINED_HIGH_3_1
>-                        : VideoCodecInfo.H264_CONSTRAINED_BASELINE_3_1);
>-        return properties;
>-      default:
>-        throw new IllegalArgumentException("Unsupported codec: " + type);
>-    }
>-  }
>-
>-  private static native boolean isSameH264Profile(
>-      Map<String, String> params1, Map<String, String> params2);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/JavaI420Buffer.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/JavaI420Buffer.java
>index 13ed63bf183..dce6a4c0dfd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/JavaI420Buffer.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/JavaI420Buffer.java
>@@ -11,6 +11,7 @@
> package org.webrtc;
> 
> import java.nio.ByteBuffer;
>+import javax.annotation.Nullable;
> import org.webrtc.VideoFrame.I420Buffer;
> 
> /** Implementation of VideoFrame.I420Buffer backed by Java direct byte buffers. */
>@@ -23,13 +24,10 @@ public class JavaI420Buffer implements VideoFrame.I420Buffer {
>   private final int strideY;
>   private final int strideU;
>   private final int strideV;
>-  private final Runnable releaseCallback;
>-  private final Object refCountLock = new Object();
>-
>-  private int refCount;
>+  private final RefCountDelegate refCountDelegate;
> 
>   private JavaI420Buffer(int width, int height, ByteBuffer dataY, int strideY, ByteBuffer dataU,
>-      int strideU, ByteBuffer dataV, int strideV, Runnable releaseCallback) {
>+      int strideU, ByteBuffer dataV, int strideV, @Nullable Runnable releaseCallback) {
>     this.width = width;
>     this.height = height;
>     this.dataY = dataY;
>@@ -38,14 +36,22 @@ public class JavaI420Buffer implements VideoFrame.I420Buffer {
>     this.strideY = strideY;
>     this.strideU = strideU;
>     this.strideV = strideV;
>-    this.releaseCallback = releaseCallback;
>+    this.refCountDelegate = new RefCountDelegate(releaseCallback);
>+  }
> 
>-    this.refCount = 1;
>+  private static void checkCapacity(ByteBuffer data, int width, int height, int stride) {
>+    // The last row does not necessarily need padding.
>+    final int minCapacity = stride * (height - 1) + width;
>+    if (data.capacity() < minCapacity) {
>+      throw new IllegalArgumentException(
>+          "Buffer must be at least " + minCapacity + " bytes, but was " + data.capacity());
>+    }
>   }
> 
>   /** Wraps existing ByteBuffers into JavaI420Buffer object without copying the contents. */
>   public static JavaI420Buffer wrap(int width, int height, ByteBuffer dataY, int strideY,
>-      ByteBuffer dataU, int strideU, ByteBuffer dataV, int strideV, Runnable releaseCallback) {
>+      ByteBuffer dataU, int strideU, ByteBuffer dataV, int strideV,
>+      @Nullable Runnable releaseCallback) {
>     if (dataY == null || dataU == null || dataV == null) {
>       throw new IllegalArgumentException("Data buffers cannot be null.");
>     }
>@@ -59,19 +65,11 @@ public class JavaI420Buffer implements VideoFrame.I420Buffer {
>     dataU = dataU.slice();
>     dataV = dataV.slice();
> 
>+    final int chromaWidth = (width + 1) / 2;
>     final int chromaHeight = (height + 1) / 2;
>-    final int minCapacityY = strideY * height;
>-    final int minCapacityU = strideU * chromaHeight;
>-    final int minCapacityV = strideV * chromaHeight;
>-    if (dataY.capacity() < minCapacityY) {
>-      throw new IllegalArgumentException("Y-buffer must be at least " + minCapacityY + " bytes.");
>-    }
>-    if (dataU.capacity() < minCapacityU) {
>-      throw new IllegalArgumentException("U-buffer must be at least " + minCapacityU + " bytes.");
>-    }
>-    if (dataV.capacity() < minCapacityV) {
>-      throw new IllegalArgumentException("V-buffer must be at least " + minCapacityV + " bytes.");
>-    }
>+    checkCapacity(dataY, width, height, strideY);
>+    checkCapacity(dataU, chromaWidth, chromaHeight, strideU);
>+    checkCapacity(dataV, chromaWidth, chromaHeight, strideV);
> 
>     return new JavaI420Buffer(
>         width, height, dataY, strideY, dataU, strideU, dataV, strideV, releaseCallback);
>@@ -85,7 +83,8 @@ public class JavaI420Buffer implements VideoFrame.I420Buffer {
>     int uPos = yPos + width * height;
>     int vPos = uPos + strideUV * chromaHeight;
> 
>-    ByteBuffer buffer = ByteBuffer.allocateDirect(width * height + 2 * strideUV * chromaHeight);
>+    ByteBuffer buffer =
>+        JniCommon.nativeAllocateByteBuffer(width * height + 2 * strideUV * chromaHeight);
> 
>     buffer.position(yPos);
>     buffer.limit(uPos);
>@@ -99,8 +98,8 @@ public class JavaI420Buffer implements VideoFrame.I420Buffer {
>     buffer.limit(vPos + strideUV * chromaHeight);
>     ByteBuffer dataV = buffer.slice();
> 
>-    return new JavaI420Buffer(
>-        width, height, dataY, width, dataU, strideUV, dataV, strideUV, null /* releaseCallback */);
>+    return new JavaI420Buffer(width, height, dataY, width, dataU, strideUV, dataV, strideUV,
>+        () -> { JniCommon.nativeFreeByteBuffer(buffer); });
>   }
> 
>   @Override
>@@ -154,24 +153,48 @@ public class JavaI420Buffer implements VideoFrame.I420Buffer {
> 
>   @Override
>   public void retain() {
>-    synchronized (refCountLock) {
>-      ++refCount;
>-    }
>+    refCountDelegate.retain();
>   }
> 
>   @Override
>   public void release() {
>-    synchronized (refCountLock) {
>-      if (--refCount == 0 && releaseCallback != null) {
>-        releaseCallback.run();
>-      }
>-    }
>+    refCountDelegate.release();
>   }
> 
>   @Override
>   public VideoFrame.Buffer cropAndScale(
>       int cropX, int cropY, int cropWidth, int cropHeight, int scaleWidth, int scaleHeight) {
>-    return VideoFrame.cropAndScaleI420(
>-        this, cropX, cropY, cropWidth, cropHeight, scaleWidth, scaleHeight);
>+    return cropAndScaleI420(this, cropX, cropY, cropWidth, cropHeight, scaleWidth, scaleHeight);
>   }
>+
>+  public static VideoFrame.Buffer cropAndScaleI420(final I420Buffer buffer, int cropX, int cropY,
>+      int cropWidth, int cropHeight, int scaleWidth, int scaleHeight) {
>+    if (cropWidth == scaleWidth && cropHeight == scaleHeight) {
>+      // No scaling.
>+      ByteBuffer dataY = buffer.getDataY();
>+      ByteBuffer dataU = buffer.getDataU();
>+      ByteBuffer dataV = buffer.getDataV();
>+
>+      dataY.position(cropX + cropY * buffer.getStrideY());
>+      dataU.position(cropX / 2 + cropY / 2 * buffer.getStrideU());
>+      dataV.position(cropX / 2 + cropY / 2 * buffer.getStrideV());
>+
>+      buffer.retain();
>+      return JavaI420Buffer.wrap(scaleWidth, scaleHeight, dataY.slice(), buffer.getStrideY(),
>+          dataU.slice(), buffer.getStrideU(), dataV.slice(), buffer.getStrideV(), buffer::release);
>+    }
>+
>+    JavaI420Buffer newBuffer = JavaI420Buffer.allocate(scaleWidth, scaleHeight);
>+    nativeCropAndScaleI420(buffer.getDataY(), buffer.getStrideY(), buffer.getDataU(),
>+        buffer.getStrideU(), buffer.getDataV(), buffer.getStrideV(), cropX, cropY, cropWidth,
>+        cropHeight, newBuffer.getDataY(), newBuffer.getStrideY(), newBuffer.getDataU(),
>+        newBuffer.getStrideU(), newBuffer.getDataV(), newBuffer.getStrideV(), scaleWidth,
>+        scaleHeight);
>+    return newBuffer;
>+  }
>+
>+  private static native void nativeCropAndScaleI420(ByteBuffer srcY, int srcStrideY,
>+      ByteBuffer srcU, int srcStrideU, ByteBuffer srcV, int srcStrideV, int cropX, int cropY,
>+      int cropWidth, int cropHeight, ByteBuffer dstY, int dstStrideY, ByteBuffer dstU,
>+      int dstStrideU, ByteBuffer dstV, int dstStrideV, int scaleWidth, int scaleHeight);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/MediaCodecVideoDecoder.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/MediaCodecVideoDecoder.java
>index 82957edd8ef..aa87cdb177b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/MediaCodecVideoDecoder.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/MediaCodecVideoDecoder.java
>@@ -10,6 +10,7 @@
> 
> package org.webrtc;
> 
>+import android.graphics.SurfaceTexture;
> import android.media.MediaCodec;
> import android.media.MediaCodecInfo;
> import android.media.MediaCodecInfo.CodecCapabilities;
>@@ -20,6 +21,8 @@ import android.os.SystemClock;
> import android.view.Surface;
> import java.nio.ByteBuffer;
> import java.util.ArrayDeque;
>+import java.util.HashMap;
>+import java.util.ArrayList;
> import java.util.Arrays;
> import java.util.HashSet;
> import java.util.List;
>@@ -27,10 +30,14 @@ import java.util.Queue;
> import java.util.Set;
> import java.util.concurrent.CountDownLatch;
> import java.util.concurrent.TimeUnit;
>+import javax.annotation.Nullable;
>+import org.webrtc.EglBase;
>+import org.webrtc.VideoFrame;
> 
>-// Java-side of peerconnection_jni.cc:MediaCodecVideoDecoder.
>+// Java-side of peerconnection.cc:MediaCodecVideoDecoder.
> // This class is an implementation detail of the Java PeerConnection API.
> @SuppressWarnings("deprecation")
>+@Deprecated
> public class MediaCodecVideoDecoder {
>   // This class is constructed, operated, and destroyed by its C++ incarnation,
>   // so the class and its methods have non-public visibility.  The API this
>@@ -38,6 +45,86 @@ public class MediaCodecVideoDecoder {
>   // possibly to minimize the amount of translation work necessary.
> 
>   private static final String TAG = "MediaCodecVideoDecoder";
>+
>+  /**
>+   * Create a VideoDecoderFactory that can be injected in the PeerConnectionFactory and replicate
>+   * the old behavior.
>+   */
>+  public static VideoDecoderFactory createFactory() {
>+    return new DefaultVideoDecoderFactory(new HwDecoderFactory());
>+  }
>+
>+  // Factory for creating HW MediaCodecVideoDecoder instances.
>+  static class HwDecoderFactory implements VideoDecoderFactory {
>+    private static boolean isSameCodec(VideoCodecInfo codecA, VideoCodecInfo codecB) {
>+      if (!codecA.name.equalsIgnoreCase(codecB.name)) {
>+        return false;
>+      }
>+      return codecA.name.equalsIgnoreCase("H264")
>+          ? H264Utils.isSameH264Profile(codecA.params, codecB.params)
>+          : true;
>+    }
>+
>+    private static boolean isCodecSupported(
>+        VideoCodecInfo[] supportedCodecs, VideoCodecInfo codec) {
>+      for (VideoCodecInfo supportedCodec : supportedCodecs) {
>+        if (isSameCodec(supportedCodec, codec)) {
>+          return true;
>+        }
>+      }
>+      return false;
>+    }
>+
>+    private static VideoCodecInfo[] getSupportedHardwareCodecs() {
>+      final List<VideoCodecInfo> codecs = new ArrayList<VideoCodecInfo>();
>+
>+      if (isVp8HwSupported()) {
>+        Logging.d(TAG, "VP8 HW Decoder supported.");
>+        codecs.add(new VideoCodecInfo("VP8", new HashMap<>()));
>+      }
>+
>+      if (isVp9HwSupported()) {
>+        Logging.d(TAG, "VP9 HW Decoder supported.");
>+        codecs.add(new VideoCodecInfo("VP9", new HashMap<>()));
>+      }
>+
>+      if (isH264HighProfileHwSupported()) {
>+        Logging.d(TAG, "H.264 High Profile HW Decoder supported.");
>+        codecs.add(H264Utils.DEFAULT_H264_HIGH_PROFILE_CODEC);
>+      }
>+
>+      if (isH264HwSupported()) {
>+        Logging.d(TAG, "H.264 HW Decoder supported.");
>+        codecs.add(H264Utils.DEFAULT_H264_BASELINE_PROFILE_CODEC);
>+      }
>+
>+      return codecs.toArray(new VideoCodecInfo[codecs.size()]);
>+    }
>+
>+    private final VideoCodecInfo[] supportedHardwareCodecs = getSupportedHardwareCodecs();
>+
>+    @Override
>+    public VideoCodecInfo[] getSupportedCodecs() {
>+      return supportedHardwareCodecs;
>+    }
>+
>+    @Nullable
>+    @Override
>+    public VideoDecoder createDecoder(VideoCodecInfo codec) {
>+      if (!isCodecSupported(supportedHardwareCodecs, codec)) {
>+        Logging.d(TAG, "No HW video decoder for codec " + codec.name);
>+        return null;
>+      }
>+      Logging.d(TAG, "Create HW video decoder for " + codec.name);
>+      return new WrappedNativeVideoDecoder() {
>+        @Override
>+        public long createNativeVideoDecoder() {
>+          return nativeCreateDecoder(codec.name, useSurface());
>+        }
>+      };
>+    }
>+  }
>+
>   private static final long MAX_DECODE_TIME_MS = 200;
> 
>   // TODO(magjed): Use MediaFormat constants when part of the public API.
>@@ -49,7 +136,17 @@ public class MediaCodecVideoDecoder {
>   private static final String FORMAT_KEY_CROP_BOTTOM = "crop-bottom";
> 
>   // Tracks webrtc::VideoCodecType.
>-  public enum VideoCodecType { VIDEO_CODEC_VP8, VIDEO_CODEC_VP9, VIDEO_CODEC_H264 }
>+  public enum VideoCodecType {
>+    VIDEO_CODEC_UNKNOWN,
>+    VIDEO_CODEC_VP8,
>+    VIDEO_CODEC_VP9,
>+    VIDEO_CODEC_H264;
>+
>+    @CalledByNative("VideoCodecType")
>+    static VideoCodecType fromNativeIndex(int nativeIndex) {
>+      return values()[nativeIndex];
>+    }
>+  }
> 
>   // Timeout for input buffer dequeue.
>   private static final int DEQUEUE_INPUT_TIMEOUT = 500000;
>@@ -59,30 +156,52 @@ public class MediaCodecVideoDecoder {
>   private static final int MAX_QUEUED_OUTPUTBUFFERS = 3;
>   // Active running decoder instance. Set in initDecode() (called from native code)
>   // and reset to null in release() call.
>-  private static MediaCodecVideoDecoder runningInstance = null;
>-  private static MediaCodecVideoDecoderErrorCallback errorCallback = null;
>+  @Nullable private static MediaCodecVideoDecoder runningInstance = null;
>+  @Nullable private static MediaCodecVideoDecoderErrorCallback errorCallback = null;
>   private static int codecErrors = 0;
>   // List of disabled codec types - can be set from application.
>   private static Set<String> hwDecoderDisabledTypes = new HashSet<String>();
>+  @Nullable private static EglBase eglBase;
> 
>-  private Thread mediaCodecThread;
>-  private MediaCodec mediaCodec;
>+  @Nullable private Thread mediaCodecThread;
>+  @Nullable private MediaCodec mediaCodec;
>   private ByteBuffer[] inputBuffers;
>   private ByteBuffer[] outputBuffers;
>   private static final String VP8_MIME_TYPE = "video/x-vnd.on2.vp8";
>   private static final String VP9_MIME_TYPE = "video/x-vnd.on2.vp9";
>   private static final String H264_MIME_TYPE = "video/avc";
>   // List of supported HW VP8 decoders.
>-  private static final String[] supportedVp8HwCodecPrefixes = {
>-      "OMX.qcom.", "OMX.Nvidia.", "OMX.Exynos.", "OMX.Intel."};
>+  private static final String[] supportedVp8HwCodecPrefixes() {
>+    ArrayList<String> supportedPrefixes = new ArrayList<String>();
>+    supportedPrefixes.add("OMX.qcom.");
>+    supportedPrefixes.add("OMX.Nvidia.");
>+    supportedPrefixes.add("OMX.Exynos.");
>+    supportedPrefixes.add("OMX.Intel.");
>+    if (PeerConnectionFactory.fieldTrialsFindFullName("WebRTC-MediaTekVP8").equals("Enabled")
>+        && Build.VERSION.SDK_INT >= Build.VERSION_CODES.N) {
>+      supportedPrefixes.add("OMX.MTK.");
>+    }
>+    return supportedPrefixes.toArray(new String[supportedPrefixes.size()]);
>+  }
>   // List of supported HW VP9 decoders.
>   private static final String[] supportedVp9HwCodecPrefixes = {"OMX.qcom.", "OMX.Exynos."};
>   // List of supported HW H.264 decoders.
>-  private static final String[] supportedH264HwCodecPrefixes = {
>-      "OMX.qcom.", "OMX.Intel.", "OMX.Exynos."};
>+  private static final String[] supportedH264HwCodecPrefixes() {
>+    ArrayList<String> supportedPrefixes = new ArrayList<String>();
>+    supportedPrefixes.add("OMX.qcom.");
>+    supportedPrefixes.add("OMX.Intel.");
>+    supportedPrefixes.add("OMX.Exynos.");
>+    if (PeerConnectionFactory.fieldTrialsFindFullName("WebRTC-MediaTekH264").equals("Enabled")
>+        && Build.VERSION.SDK_INT >= Build.VERSION_CODES.O_MR1) {
>+      supportedPrefixes.add("OMX.MTK.");
>+    }
>+    return supportedPrefixes.toArray(new String[supportedPrefixes.size()]);
>+  }
>+
>   // List of supported HW H.264 high profile decoders.
>   private static final String supportedQcomH264HighProfileHwCodecPrefix = "OMX.qcom.";
>   private static final String supportedExynosH264HighProfileHwCodecPrefix = "OMX.Exynos.";
>+  private static final String supportedMediaTekH264HighProfileHwCodecPrefix = "OMX.MTK.";
> 
>   // NV12 color format supported by QCOM codec, but not declared in MediaCodec -
>   // see /hardware/qcom/media/mm-core/inc/OMX_QCOMExtns.h
>@@ -105,12 +224,11 @@ public class MediaCodecVideoDecoder {
>   private int sliceHeight;
>   private boolean hasDecodedFirstFrame;
>   private final Queue<TimeStamps> decodeStartTimeMs = new ArrayDeque<TimeStamps>();
>-  private boolean useSurface;
> 
>   // The below variables are only used when decoding to a Surface.
>-  private TextureListener textureListener;
>+  @Nullable private TextureListener textureListener;
>   private int droppedFrames;
>-  private Surface surface = null;
>+  @Nullable private Surface surface = null;
>   private final Queue<DecodedOutputBuffer> dequeuedSurfaceOutputBuffers =
>       new ArrayDeque<DecodedOutputBuffer>();
> 
>@@ -121,6 +239,27 @@ public class MediaCodecVideoDecoder {
>     void onMediaCodecVideoDecoderCriticalError(int codecErrors);
>   }
> 
>+  /** Set EGL context used by HW decoding. The EGL context must be shared with the remote render. */
>+  public static void setEglContext(EglBase.Context eglContext) {
>+    if (eglBase != null) {
>+      Logging.w(TAG, "Egl context already set.");
>+      eglBase.release();
>+    }
>+    eglBase = EglBase.create(eglContext);
>+  }
>+
>+  /** Dispose the EGL context used by HW decoding. */
>+  public static void disposeEglContext() {
>+    if (eglBase != null) {
>+      eglBase.release();
>+      eglBase = null;
>+    }
>+  }
>+
>+  static boolean useSurface() {
>+    return eglBase != null;
>+  }
>+
>   public static void setErrorCallback(MediaCodecVideoDecoderErrorCallback errorCallback) {
>     Logging.d(TAG, "Set error callback");
>     MediaCodecVideoDecoder.errorCallback = errorCallback;
>@@ -146,7 +285,7 @@ public class MediaCodecVideoDecoder {
>   // Functions to query if HW decoding is supported.
>   public static boolean isVp8HwSupported() {
>     return !hwDecoderDisabledTypes.contains(VP8_MIME_TYPE)
>-        && (findDecoder(VP8_MIME_TYPE, supportedVp8HwCodecPrefixes) != null);
>+        && (findDecoder(VP8_MIME_TYPE, supportedVp8HwCodecPrefixes()) != null);
>   }
> 
>   public static boolean isVp9HwSupported() {
>@@ -156,7 +295,7 @@ public class MediaCodecVideoDecoder {
> 
>   public static boolean isH264HwSupported() {
>     return !hwDecoderDisabledTypes.contains(H264_MIME_TYPE)
>-        && (findDecoder(H264_MIME_TYPE, supportedH264HwCodecPrefixes) != null);
>+        && (findDecoder(H264_MIME_TYPE, supportedH264HwCodecPrefixes()) != null);
>   }
> 
>   public static boolean isH264HighProfileHwSupported() {
>@@ -175,6 +314,13 @@ public class MediaCodecVideoDecoder {
>             != null) {
>       return true;
>     }
>+    // Support H.264 HP decoding on MediaTek chips for Android O_MR1 and above
>+    if (PeerConnectionFactory.fieldTrialsFindFullName("WebRTC-MediaTekH264").equals("Enabled")
>+        && Build.VERSION.SDK_INT >= Build.VERSION_CODES.O_MR1
>+        && findDecoder(H264_MIME_TYPE, new String[] {supportedMediaTekH264HighProfileHwCodecPrefix})
>+            != null) {
>+      return true;
>+    }
>     return false;
>   }
> 
>@@ -200,7 +346,8 @@ public class MediaCodecVideoDecoder {
>     public final int colorFormat; // Color format supported by codec.
>   }
> 
>-  private static DecoderProperties findDecoder(String mime, String[] supportedCodecPrefixes) {
>+  private static @Nullable DecoderProperties findDecoder(
>+      String mime, String[] supportedCodecPrefixes) {
>     if (Build.VERSION.SDK_INT < Build.VERSION_CODES.KITKAT) {
>       return null; // MediaCodec.setParameters is missing.
>     }
>@@ -265,6 +412,9 @@ public class MediaCodecVideoDecoder {
>     return null; // No HW decoder.
>   }
> 
>+  @CalledByNative
>+  MediaCodecVideoDecoder() {}
>+
>   private void checkOnMediaCodecThread() throws IllegalStateException {
>     if (mediaCodecThread.getId() != Thread.currentThread().getId()) {
>       throw new IllegalStateException("MediaCodecVideoDecoder previously operated on "
>@@ -272,25 +422,23 @@ public class MediaCodecVideoDecoder {
>     }
>   }
> 
>-  // Pass null in |surfaceTextureHelper| to configure the codec for ByteBuffer output.
>-  private boolean initDecode(
>-      VideoCodecType type, int width, int height, SurfaceTextureHelper surfaceTextureHelper) {
>+  @CalledByNativeUnchecked
>+  private boolean initDecode(VideoCodecType type, int width, int height) {
>     if (mediaCodecThread != null) {
>       throw new RuntimeException("initDecode: Forgot to release()?");
>     }
> 
>     String mime = null;
>-    useSurface = (surfaceTextureHelper != null);
>     String[] supportedCodecPrefixes = null;
>     if (type == VideoCodecType.VIDEO_CODEC_VP8) {
>       mime = VP8_MIME_TYPE;
>-      supportedCodecPrefixes = supportedVp8HwCodecPrefixes;
>+      supportedCodecPrefixes = supportedVp8HwCodecPrefixes();
>     } else if (type == VideoCodecType.VIDEO_CODEC_VP9) {
>       mime = VP9_MIME_TYPE;
>       supportedCodecPrefixes = supportedVp9HwCodecPrefixes;
>     } else if (type == VideoCodecType.VIDEO_CODEC_H264) {
>       mime = H264_MIME_TYPE;
>-      supportedCodecPrefixes = supportedH264HwCodecPrefixes;
>+      supportedCodecPrefixes = supportedH264HwCodecPrefixes();
>     } else {
>       throw new RuntimeException("initDecode: Non-supported codec " + type);
>     }
>@@ -299,8 +447,9 @@ public class MediaCodecVideoDecoder {
>       throw new RuntimeException("Cannot find HW decoder for " + type);
>     }
> 
>-    Logging.d(TAG, "Java initDecode: " + type + " : " + width + " x " + height + ". Color: 0x"
>-            + Integer.toHexString(properties.colorFormat) + ". Use Surface: " + useSurface);
>+    Logging.d(TAG,
>+        "Java initDecode: " + type + " : " + width + " x " + height + ". Color: 0x"
>+            + Integer.toHexString(properties.colorFormat) + ". Use Surface: " + useSurface());
> 
>     runningInstance = this; // Decoder is now running and can be queried for stack traces.
>     mediaCodecThread = Thread.currentThread();
>@@ -310,13 +459,19 @@ public class MediaCodecVideoDecoder {
>       stride = width;
>       sliceHeight = height;
> 
>-      if (useSurface) {
>-        textureListener = new TextureListener(surfaceTextureHelper);
>-        surface = new Surface(surfaceTextureHelper.getSurfaceTexture());
>+      if (useSurface()) {
>+        @Nullable
>+        final SurfaceTextureHelper surfaceTextureHelper = SurfaceTextureHelper.create(
>+            "Decoder SurfaceTextureHelper", eglBase.getEglBaseContext());
>+        if (surfaceTextureHelper != null) {
>+          textureListener = new TextureListener(surfaceTextureHelper);
>+          textureListener.setSize(width, height);
>+          surface = new Surface(surfaceTextureHelper.getSurfaceTexture());
>+        }
>       }
> 
>       MediaFormat format = MediaFormat.createVideoFormat(mime, width, height);
>-      if (!useSurface) {
>+      if (!useSurface()) {
>         format.setInteger(MediaFormat.KEY_COLOR_FORMAT, properties.colorFormat);
>       }
>       Logging.d(TAG, "  Format: " + format);
>@@ -346,6 +501,7 @@ public class MediaCodecVideoDecoder {
> 
>   // Resets the decoder so it can start decoding frames with new resolution.
>   // Flushes MediaCodec and clears decoder output buffers.
>+  @CalledByNativeUnchecked
>   private void reset(int width, int height) {
>     if (mediaCodecThread == null || mediaCodec == null) {
>       throw new RuntimeException("Incorrect reset call for non-initialized decoder.");
>@@ -356,12 +512,16 @@ public class MediaCodecVideoDecoder {
> 
>     this.width = width;
>     this.height = height;
>+    if (textureListener != null) {
>+      textureListener.setSize(width, height);
>+    }
>     decodeStartTimeMs.clear();
>     dequeuedSurfaceOutputBuffers.clear();
>     hasDecodedFirstFrame = false;
>     droppedFrames = 0;
>   }
> 
>+  @CalledByNativeUnchecked
>   private void release() {
>     Logging.d(TAG, "Java releaseDecoder. Total number of dropped frames: " + droppedFrames);
>     checkOnMediaCodecThread();
>@@ -398,7 +558,7 @@ public class MediaCodecVideoDecoder {
>     mediaCodec = null;
>     mediaCodecThread = null;
>     runningInstance = null;
>-    if (useSurface) {
>+    if (useSurface()) {
>       surface.release();
>       surface = null;
>       textureListener.release();
>@@ -408,6 +568,7 @@ public class MediaCodecVideoDecoder {
> 
>   // Dequeue an input buffer and return its index, -1 if no input buffer is
>   // available, or -2 if the codec is no longer operative.
>+  @CalledByNativeUnchecked
>   private int dequeueInputBuffer() {
>     checkOnMediaCodecThread();
>     try {
>@@ -418,6 +579,7 @@ public class MediaCodecVideoDecoder {
>     }
>   }
> 
>+  @CalledByNativeUnchecked
>   private boolean queueInputBuffer(int inputBufferIndex, int size, long presentationTimeStamUs,
>       long timeStampMs, long ntpTimeStamp) {
>     checkOnMediaCodecThread();
>@@ -475,12 +637,46 @@ public class MediaCodecVideoDecoder {
>     private final long decodeTimeMs;
>     // System time when this frame decoding finished.
>     private final long endDecodeTimeMs;
>+
>+    @CalledByNative("DecodedOutputBuffer")
>+    int getIndex() {
>+      return index;
>+    }
>+
>+    @CalledByNative("DecodedOutputBuffer")
>+    int getOffset() {
>+      return offset;
>+    }
>+
>+    @CalledByNative("DecodedOutputBuffer")
>+    int getSize() {
>+      return size;
>+    }
>+
>+    @CalledByNative("DecodedOutputBuffer")
>+    long getPresentationTimestampMs() {
>+      return presentationTimeStampMs;
>+    }
>+
>+    @CalledByNative("DecodedOutputBuffer")
>+    long getTimestampMs() {
>+      return timeStampMs;
>+    }
>+
>+    @CalledByNative("DecodedOutputBuffer")
>+    long getNtpTimestampMs() {
>+      return ntpTimeStampMs;
>+    }
>+
>+    @CalledByNative("DecodedOutputBuffer")
>+    long getDecodeTimeMs() {
>+      return decodeTimeMs;
>+    }
>   }
> 
>   // Helper struct for dequeueTextureBuffer() below.
>   private static class DecodedTextureBuffer {
>-    private final int textureID;
>-    private final float[] transformMatrix;
>+    private final VideoFrame.Buffer videoFrameBuffer;
>     // Presentation timestamp returned in dequeueOutputBuffer call.
>     private final long presentationTimeStampMs;
>     // C++ inputImage._timeStamp value for output frame.
>@@ -497,29 +693,56 @@ public class MediaCodecVideoDecoder {
> 
>     // A DecodedTextureBuffer with zero |textureID| has special meaning and represents a frame
>     // that was dropped.
>-    public DecodedTextureBuffer(int textureID, float[] transformMatrix,
>-        long presentationTimeStampMs, long timeStampMs, long ntpTimeStampMs, long decodeTimeMs,
>-        long frameDelay) {
>-      this.textureID = textureID;
>-      this.transformMatrix = transformMatrix;
>+    public DecodedTextureBuffer(VideoFrame.Buffer videoFrameBuffer, long presentationTimeStampMs,
>+        long timeStampMs, long ntpTimeStampMs, long decodeTimeMs, long frameDelay) {
>+      this.videoFrameBuffer = videoFrameBuffer;
>       this.presentationTimeStampMs = presentationTimeStampMs;
>       this.timeStampMs = timeStampMs;
>       this.ntpTimeStampMs = ntpTimeStampMs;
>       this.decodeTimeMs = decodeTimeMs;
>       this.frameDelayMs = frameDelay;
>     }
>+
>+    @CalledByNative("DecodedTextureBuffer")
>+    VideoFrame.Buffer getVideoFrameBuffer() {
>+      return videoFrameBuffer;
>+    }
>+
>+    @CalledByNative("DecodedTextureBuffer")
>+    long getPresentationTimestampMs() {
>+      return presentationTimeStampMs;
>+    }
>+
>+    @CalledByNative("DecodedTextureBuffer")
>+    long getTimeStampMs() {
>+      return timeStampMs;
>+    }
>+
>+    @CalledByNative("DecodedTextureBuffer")
>+    long getNtpTimestampMs() {
>+      return ntpTimeStampMs;
>+    }
>+
>+    @CalledByNative("DecodedTextureBuffer")
>+    long getDecodeTimeMs() {
>+      return decodeTimeMs;
>+    }
>+
>+    @CalledByNative("DecodedTextureBuffer")
>+    long getFrameDelayMs() {
>+      return frameDelayMs;
>+    }
>   }
> 
>   // Poll based texture listener.
>-  private static class TextureListener
>-      implements SurfaceTextureHelper.OnTextureFrameAvailableListener {
>+  private class TextureListener implements VideoSink {
>     private final SurfaceTextureHelper surfaceTextureHelper;
>     // |newFrameLock| is used to synchronize arrival of new frames with wait()/notifyAll().
>     private final Object newFrameLock = new Object();
>     // |bufferToRender| is non-null when waiting for transition between addBufferToRender() to
>-    // onTextureFrameAvailable().
>-    private DecodedOutputBuffer bufferToRender;
>-    private DecodedTextureBuffer renderedBuffer;
>+    // onFrame().
>+    @Nullable private DecodedOutputBuffer bufferToRender;
>+    @Nullable private DecodedTextureBuffer renderedBuffer;
> 
>     public TextureListener(SurfaceTextureHelper surfaceTextureHelper) {
>       this.surfaceTextureHelper = surfaceTextureHelper;
>@@ -540,20 +763,23 @@ public class MediaCodecVideoDecoder {
>       }
>     }
> 
>+    public void setSize(int width, int height) {
>+      surfaceTextureHelper.setTextureSize(width, height);
>+    }
>+
>     // Callback from |surfaceTextureHelper|. May be called on an arbitrary thread.
>     @Override
>-    public void onTextureFrameAvailable(
>-        int oesTextureId, float[] transformMatrix, long timestampNs) {
>+    public void onFrame(VideoFrame frame) {
>       synchronized (newFrameLock) {
>         if (renderedBuffer != null) {
>-          Logging.e(
>-              TAG, "Unexpected onTextureFrameAvailable() called while already holding a texture.");
>+          Logging.e(TAG, "Unexpected onFrame() called while already holding a texture.");
>           throw new IllegalStateException("Already holding a texture.");
>         }
>         // |timestampNs| is always zero on some Android versions.
>-        renderedBuffer = new DecodedTextureBuffer(oesTextureId, transformMatrix,
>-            bufferToRender.presentationTimeStampMs, bufferToRender.timeStampMs,
>-            bufferToRender.ntpTimeStampMs, bufferToRender.decodeTimeMs,
>+        final VideoFrame.Buffer buffer = frame.getBuffer();
>+        buffer.retain();
>+        renderedBuffer = new DecodedTextureBuffer(buffer, bufferToRender.presentationTimeStampMs,
>+            bufferToRender.timeStampMs, bufferToRender.ntpTimeStampMs, bufferToRender.decodeTimeMs,
>             SystemClock.elapsedRealtime() - bufferToRender.endDecodeTimeMs);
>         bufferToRender = null;
>         newFrameLock.notifyAll();
>@@ -561,6 +787,7 @@ public class MediaCodecVideoDecoder {
>     }
> 
>     // Dequeues and returns a DecodedTextureBuffer if available, or null otherwise.
>+    @Nullable
>     @SuppressWarnings("WaitNotInLoop")
>     public DecodedTextureBuffer dequeueTextureBuffer(int timeoutMs) {
>       synchronized (newFrameLock) {
>@@ -579,16 +806,17 @@ public class MediaCodecVideoDecoder {
>     }
> 
>     public void release() {
>-      // SurfaceTextureHelper.stopListening() will block until any onTextureFrameAvailable() in
>-      // progress is done. Therefore, the call must be outside any synchronized
>-      // statement that is also used in the onTextureFrameAvailable() above to avoid deadlocks.
>+      // SurfaceTextureHelper.stopListening() will block until any onFrame() in progress is done.
>+      // Therefore, the call must be outside any synchronized statement that is also used in the
>+      // onFrame() above to avoid deadlocks.
>       surfaceTextureHelper.stopListening();
>       synchronized (newFrameLock) {
>         if (renderedBuffer != null) {
>-          surfaceTextureHelper.returnTextureFrame();
>+          renderedBuffer.getVideoFrameBuffer().release();
>           renderedBuffer = null;
>         }
>       }
>+      surfaceTextureHelper.dispose();
>     }
>   }
> 
>@@ -596,7 +824,8 @@ public class MediaCodecVideoDecoder {
>   // Throws IllegalStateException if call is made on the wrong thread, if color format changes to an
>   // unsupported format, or if |mediaCodec| is not in the Executing state. Throws CodecException
>   // upon codec error.
>-  private DecodedOutputBuffer dequeueOutputBuffer(int dequeueTimeoutMs) {
>+  @CalledByNativeUnchecked
>+  private @Nullable DecodedOutputBuffer dequeueOutputBuffer(int dequeueTimeoutMs) {
>     checkOnMediaCodecThread();
>     if (decodeStartTimeMs.isEmpty()) {
>       return null;
>@@ -637,8 +866,11 @@ public class MediaCodecVideoDecoder {
>           }
>           width = newWidth;
>           height = newHeight;
>+          if (textureListener != null) {
>+            textureListener.setSize(width, height);
>+          }
> 
>-          if (!useSurface && format.containsKey(MediaFormat.KEY_COLOR_FORMAT)) {
>+          if (!useSurface() && format.containsKey(MediaFormat.KEY_COLOR_FORMAT)) {
>             colorFormat = format.getInteger(MediaFormat.KEY_COLOR_FORMAT);
>             Logging.d(TAG, "Color: 0x" + Integer.toHexString(colorFormat));
>             if (!supportedColorList.contains(colorFormat)) {
>@@ -679,9 +911,10 @@ public class MediaCodecVideoDecoder {
>   // unsupported format, or if |mediaCodec| is not in the Executing state. Throws CodecException
>   // upon codec error. If |dequeueTimeoutMs| > 0, the oldest decoded frame will be dropped if
>   // a frame can't be returned.
>-  private DecodedTextureBuffer dequeueTextureBuffer(int dequeueTimeoutMs) {
>+  @CalledByNativeUnchecked
>+  private @Nullable DecodedTextureBuffer dequeueTextureBuffer(int dequeueTimeoutMs) {
>     checkOnMediaCodecThread();
>-    if (!useSurface) {
>+    if (!useSurface()) {
>       throw new IllegalStateException("dequeueTexture() called for byte buffer decoding.");
>     }
>     DecodedOutputBuffer outputBuffer = dequeueOutputBuffer(dequeueTimeoutMs);
>@@ -718,8 +951,9 @@ public class MediaCodecVideoDecoder {
>       }
> 
>       mediaCodec.releaseOutputBuffer(droppedFrame.index, false /* render */);
>-      return new DecodedTextureBuffer(0, null, droppedFrame.presentationTimeStampMs,
>-          droppedFrame.timeStampMs, droppedFrame.ntpTimeStampMs, droppedFrame.decodeTimeMs,
>+      return new DecodedTextureBuffer(null /* videoFrameBuffer */,
>+          droppedFrame.presentationTimeStampMs, droppedFrame.timeStampMs,
>+          droppedFrame.ntpTimeStampMs, droppedFrame.decodeTimeMs,
>           SystemClock.elapsedRealtime() - droppedFrame.endDecodeTimeMs);
>     }
>     return null;
>@@ -740,12 +974,50 @@ public class MediaCodecVideoDecoder {
>   // Throws IllegalStateException if the call is made on the wrong thread, if codec is configured
>   // for surface decoding, or if |mediaCodec| is not in the Executing state. Throws
>   // MediaCodec.CodecException upon codec error.
>+  @CalledByNativeUnchecked
>   private void returnDecodedOutputBuffer(int index)
>       throws IllegalStateException, MediaCodec.CodecException {
>     checkOnMediaCodecThread();
>-    if (useSurface) {
>+    if (useSurface()) {
>       throw new IllegalStateException("returnDecodedOutputBuffer() called for surface decoding.");
>     }
>     mediaCodec.releaseOutputBuffer(index, false /* render */);
>   }
>+
>+  @CalledByNative
>+  ByteBuffer[] getInputBuffers() {
>+    return inputBuffers;
>+  }
>+
>+  @CalledByNative
>+  ByteBuffer[] getOutputBuffers() {
>+    return outputBuffers;
>+  }
>+
>+  @CalledByNative
>+  int getColorFormat() {
>+    return colorFormat;
>+  }
>+
>+  @CalledByNative
>+  int getWidth() {
>+    return width;
>+  }
>+
>+  @CalledByNative
>+  int getHeight() {
>+    return height;
>+  }
>+
>+  @CalledByNative
>+  int getStride() {
>+    return stride;
>+  }
>+
>+  @CalledByNative
>+  int getSliceHeight() {
>+    return sliceHeight;
>+  }
>+
>+  private static native long nativeCreateDecoder(String codec, boolean useSurface);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/MediaCodecVideoEncoder.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/MediaCodecVideoEncoder.java
>index b009893ba7f..5846e703ee0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/MediaCodecVideoEncoder.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/MediaCodecVideoEncoder.java
>@@ -23,17 +23,23 @@ import android.os.Bundle;
> import android.view.Surface;
> import java.nio.ByteBuffer;
> import java.util.ArrayList;
>+import java.util.HashMap;
> import java.util.Arrays;
> import java.util.HashSet;
> import java.util.List;
> import java.util.Set;
> import java.util.concurrent.CountDownLatch;
> import java.util.concurrent.TimeUnit;
>+import javax.annotation.Nullable;
>+import org.webrtc.EglBase;
>+import org.webrtc.EglBase14;
>+import org.webrtc.VideoFrame;
> 
>-// Java-side of peerconnection_jni.cc:MediaCodecVideoEncoder.
>+// Java-side of peerconnection.cc:MediaCodecVideoEncoder.
> // This class is an implementation detail of the Java PeerConnection API.
> @TargetApi(19)
> @SuppressWarnings("deprecation")
>+@Deprecated
> public class MediaCodecVideoEncoder {
>   // This class is constructed, operated, and destroyed by its C++ incarnation,
>   // so the class and its methods have non-public visibility.  The API this
>@@ -42,8 +48,105 @@ public class MediaCodecVideoEncoder {
> 
>   private static final String TAG = "MediaCodecVideoEncoder";
> 
>+  /**
>+   * Create a VideoEncoderFactory that can be injected in the PeerConnectionFactory and replicate
>+   * the old behavior.
>+   */
>+  public static VideoEncoderFactory createFactory() {
>+    return new DefaultVideoEncoderFactory(new HwEncoderFactory());
>+  }
>+
>+  // Factory for creating HW MediaCodecVideoEncoder instances.
>+  static class HwEncoderFactory implements VideoEncoderFactory {
>+    private static boolean isSameCodec(VideoCodecInfo codecA, VideoCodecInfo codecB) {
>+      if (!codecA.name.equalsIgnoreCase(codecB.name)) {
>+        return false;
>+      }
>+      return codecA.name.equalsIgnoreCase("H264")
>+          ? H264Utils.isSameH264Profile(codecA.params, codecB.params)
>+          : true;
>+    }
>+
>+    private static boolean isCodecSupported(
>+        VideoCodecInfo[] supportedCodecs, VideoCodecInfo codec) {
>+      for (VideoCodecInfo supportedCodec : supportedCodecs) {
>+        if (isSameCodec(supportedCodec, codec)) {
>+          return true;
>+        }
>+      }
>+      return false;
>+    }
>+
>+    private static VideoCodecInfo[] getSupportedHardwareCodecs() {
>+      final List<VideoCodecInfo> codecs = new ArrayList<VideoCodecInfo>();
>+
>+      if (isVp8HwSupported()) {
>+        Logging.d(TAG, "VP8 HW Encoder supported.");
>+        codecs.add(new VideoCodecInfo("VP8", new HashMap<>()));
>+      }
>+
>+      if (isVp9HwSupported()) {
>+        Logging.d(TAG, "VP9 HW Encoder supported.");
>+        codecs.add(new VideoCodecInfo("VP9", new HashMap<>()));
>+      }
>+
>+      // Check if high profile is supported by decoder. If yes, encoder can always
>+      // fall back to baseline profile as a subset as high profile.
>+      if (MediaCodecVideoDecoder.isH264HighProfileHwSupported()) {
>+        Logging.d(TAG, "H.264 High Profile HW Encoder supported.");
>+        codecs.add(H264Utils.DEFAULT_H264_HIGH_PROFILE_CODEC);
>+      }
>+
>+      if (isH264HwSupported()) {
>+        Logging.d(TAG, "H.264 HW Encoder supported.");
>+        codecs.add(H264Utils.DEFAULT_H264_BASELINE_PROFILE_CODEC);
>+      }
>+
>+      return codecs.toArray(new VideoCodecInfo[codecs.size()]);
>+    }
>+
>+    private final VideoCodecInfo[] supportedHardwareCodecs = getSupportedHardwareCodecs();
>+
>+    @Override
>+    public VideoCodecInfo[] getSupportedCodecs() {
>+      return supportedHardwareCodecs;
>+    }
>+
>+    @Nullable
>+    @Override
>+    public VideoEncoder createEncoder(VideoCodecInfo info) {
>+      if (!isCodecSupported(supportedHardwareCodecs, info)) {
>+        Logging.d(TAG, "No HW video encoder for codec " + info.name);
>+        return null;
>+      }
>+      Logging.d(TAG, "Create HW video encoder for " + info.name);
>+      return new WrappedNativeVideoEncoder() {
>+        @Override
>+        public long createNativeVideoEncoder() {
>+          return nativeCreateEncoder(
>+              info, /* hasEgl14Context= */ staticEglBase instanceof EglBase14);
>+        }
>+
>+        @Override
>+        public boolean isHardwareEncoder() {
>+          return true;
>+        }
>+      };
>+    }
>+  }
>+
>   // Tracks webrtc::VideoCodecType.
>-  public enum VideoCodecType { VIDEO_CODEC_VP8, VIDEO_CODEC_VP9, VIDEO_CODEC_H264 }
>+  public enum VideoCodecType {
>+    VIDEO_CODEC_UNKNOWN,
>+    VIDEO_CODEC_VP8,
>+    VIDEO_CODEC_VP9,
>+    VIDEO_CODEC_H264;
>+
>+    @CalledByNative("VideoCodecType")
>+    static VideoCodecType fromNativeIndex(int nativeIndex) {
>+      return values()[nativeIndex];
>+    }
>+  }
> 
>   private static final int MEDIA_CODEC_RELEASE_TIMEOUT_MS = 5000; // Timeout for codec releasing.
>   private static final int DEQUEUE_TIMEOUT = 0; // Non-blocking, no wait.
>@@ -61,21 +164,22 @@ public class MediaCodecVideoEncoder {
> 
>   // Active running encoder instance. Set in initEncode() (called from native code)
>   // and reset to null in release() call.
>-  private static MediaCodecVideoEncoder runningInstance = null;
>-  private static MediaCodecVideoEncoderErrorCallback errorCallback = null;
>+  @Nullable private static MediaCodecVideoEncoder runningInstance = null;
>+  @Nullable private static MediaCodecVideoEncoderErrorCallback errorCallback = null;
>   private static int codecErrors = 0;
>   // List of disabled codec types - can be set from application.
>   private static Set<String> hwEncoderDisabledTypes = new HashSet<String>();
>+  @Nullable private static EglBase staticEglBase;
> 
>-  private Thread mediaCodecThread;
>-  private MediaCodec mediaCodec;
>+  @Nullable private Thread mediaCodecThread;
>+  @Nullable private MediaCodec mediaCodec;
>   private ByteBuffer[] outputBuffers;
>-  private EglBase14 eglBase;
>+  @Nullable private EglBase14 eglBase;
>   private int profile;
>   private int width;
>   private int height;
>-  private Surface inputSurface;
>-  private GlRectDrawer drawer;
>+  @Nullable private Surface inputSurface;
>+  @Nullable private GlRectDrawer drawer;
> 
>   private static final String VP8_MIME_TYPE = "video/x-vnd.on2.vp8";
>   private static final String VP9_MIME_TYPE = "video/x-vnd.on2.vp9";
>@@ -134,6 +238,31 @@ public class MediaCodecVideoEncoder {
>     }
>   }
> 
>+  /**
>+   * Set EGL context used by HW encoding. The EGL context must be shared with the video capturer
>+   * and any local render.
>+   */
>+  public static void setEglContext(EglBase.Context eglContext) {
>+    if (staticEglBase != null) {
>+      Logging.w(TAG, "Egl context already set.");
>+      staticEglBase.release();
>+    }
>+    staticEglBase = EglBase.create(eglContext);
>+  }
>+
>+  /** Dispose the EGL context used by HW encoding. */
>+  public static void disposeEglContext() {
>+    if (staticEglBase != null) {
>+      staticEglBase.release();
>+      staticEglBase = null;
>+    }
>+  }
>+
>+  @Nullable
>+  static EglBase.Context getEglContext() {
>+    return staticEglBase == null ? null : staticEglBase.getEglBaseContext();
>+  }
>+
>   // List of supported HW VP8 encoders.
>   private static final MediaCodecProperties qcomVp8HwProperties = new MediaCodecProperties(
>       "OMX.qcom.", Build.VERSION_CODES.KITKAT, BitrateAdjustmentType.NO_ADJUSTMENT);
>@@ -164,8 +293,17 @@ public class MediaCodecVideoEncoder {
>       "OMX.qcom.", Build.VERSION_CODES.KITKAT, BitrateAdjustmentType.NO_ADJUSTMENT);
>   private static final MediaCodecProperties exynosH264HwProperties = new MediaCodecProperties(
>       "OMX.Exynos.", Build.VERSION_CODES.LOLLIPOP, BitrateAdjustmentType.FRAMERATE_ADJUSTMENT);
>-  private static final MediaCodecProperties[] h264HwList =
>-      new MediaCodecProperties[] {qcomH264HwProperties, exynosH264HwProperties};
>+  private static final MediaCodecProperties mediatekH264HwProperties = new MediaCodecProperties(
>+      "OMX.MTK.", Build.VERSION_CODES.O_MR1, BitrateAdjustmentType.FRAMERATE_ADJUSTMENT);
>+  private static final MediaCodecProperties[] h264HwList() {
>+    final ArrayList<MediaCodecProperties> supported_codecs = new ArrayList<MediaCodecProperties>();
>+    supported_codecs.add(qcomH264HwProperties);
>+    supported_codecs.add(exynosH264HwProperties);
>+    if (PeerConnectionFactory.fieldTrialsFindFullName("WebRTC-MediaTekH264").equals("Enabled")) {
>+      supported_codecs.add(mediatekH264HwProperties);
>+    }
>+    return supported_codecs.toArray(new MediaCodecProperties[supported_codecs.size()]);
>+  }
> 
>   // List of supported HW H.264 high profile encoders.
>   private static final MediaCodecProperties exynosH264HighProfileHwProperties =
>@@ -193,7 +331,7 @@ public class MediaCodecVideoEncoder {
>       COLOR_QCOM_FORMATYUV420PackedSemiPlanar32m};
>   private static final int[] supportedSurfaceColorList = {CodecCapabilities.COLOR_FormatSurface};
>   private VideoCodecType type;
>-  private int colorFormat; // Used by native code.
>+  private int colorFormat;
> 
>   // Variables used for dynamic bitrate adjustment.
>   private BitrateAdjustmentType bitrateAdjustmentType = BitrateAdjustmentType.NO_ADJUSTMENT;
>@@ -210,7 +348,7 @@ public class MediaCodecVideoEncoder {
>   private long lastKeyFrameMs;
> 
>   // SPS and PPS NALs (Config frame) for H.264.
>-  private ByteBuffer configData = null;
>+  @Nullable private ByteBuffer configData = null;
> 
>   // MediaCodec error handler - invoked when critical error happens which may prevent
>   // further use of media codec API. Now it means that one of media codec instances
>@@ -247,7 +385,7 @@ public class MediaCodecVideoEncoder {
>         && (findHwEncoder(VP8_MIME_TYPE, vp8HwList(), supportedColorList) != null);
>   }
> 
>-  public static EncoderProperties vp8HwEncoderProperties() {
>+  public static @Nullable EncoderProperties vp8HwEncoderProperties() {
>     if (hwEncoderDisabledTypes.contains(VP8_MIME_TYPE)) {
>       return null;
>     } else {
>@@ -262,7 +400,7 @@ public class MediaCodecVideoEncoder {
> 
>   public static boolean isH264HwSupported() {
>     return !hwEncoderDisabledTypes.contains(H264_MIME_TYPE)
>-        && (findHwEncoder(H264_MIME_TYPE, h264HwList, supportedColorList) != null);
>+        && (findHwEncoder(H264_MIME_TYPE, h264HwList(), supportedColorList) != null);
>   }
> 
>   public static boolean isH264HighProfileHwSupported() {
>@@ -282,7 +420,7 @@ public class MediaCodecVideoEncoder {
> 
>   public static boolean isH264HwSupportedUsingTextures() {
>     return !hwEncoderDisabledTypes.contains(H264_MIME_TYPE)
>-        && (findHwEncoder(H264_MIME_TYPE, h264HwList, supportedSurfaceColorList) != null);
>+        && (findHwEncoder(H264_MIME_TYPE, h264HwList(), supportedSurfaceColorList) != null);
>   }
> 
>   // Helper struct for findHwEncoder() below.
>@@ -298,7 +436,7 @@ public class MediaCodecVideoEncoder {
>     public final BitrateAdjustmentType bitrateAdjustmentType; // Bitrate adjustment type
>   }
> 
>-  private static EncoderProperties findHwEncoder(
>+  private static @Nullable EncoderProperties findHwEncoder(
>       String mime, MediaCodecProperties[] supportedHwCodecProperties, int[] colorList) {
>     // MediaCodec.setParameters is missing for JB and below, so bitrate
>     // can not be adjusted dynamically.
>@@ -387,6 +525,9 @@ public class MediaCodecVideoEncoder {
>     return null; // No HW encoder.
>   }
> 
>+  @CalledByNative
>+  MediaCodecVideoEncoder() {}
>+
>   private void checkOnMediaCodecThread() {
>     if (mediaCodecThread.getId() != Thread.currentThread().getId()) {
>       throw new RuntimeException("MediaCodecVideoEncoder previously operated on " + mediaCodecThread
>@@ -406,7 +547,7 @@ public class MediaCodecVideoEncoder {
>     }
>   }
> 
>-  static MediaCodec createByCodecName(String codecName) {
>+  static @Nullable MediaCodec createByCodecName(String codecName) {
>     try {
>       // In the L-SDK this call can throw IOException so in order to work in
>       // both cases catch an exception.
>@@ -416,9 +557,9 @@ public class MediaCodecVideoEncoder {
>     }
>   }
> 
>+  @CalledByNativeUnchecked
>   boolean initEncode(VideoCodecType type, int profile, int width, int height, int kbps, int fps,
>-      EglBase14.Context sharedContext) {
>-    final boolean useSurface = sharedContext != null;
>+      boolean useSurface) {
>     Logging.d(TAG,
>         "Java initEncode: " + type + ". Profile: " + profile + " : " + width + " x " + height
>             + ". @ " + kbps + " kbps. Fps: " + fps + ". Encode from texture : " + useSurface);
>@@ -445,8 +586,8 @@ public class MediaCodecVideoEncoder {
>       keyFrameIntervalSec = 100;
>     } else if (type == VideoCodecType.VIDEO_CODEC_H264) {
>       mime = H264_MIME_TYPE;
>-      properties = findHwEncoder(
>-          H264_MIME_TYPE, h264HwList, useSurface ? supportedSurfaceColorList : supportedColorList);
>+      properties = findHwEncoder(H264_MIME_TYPE, h264HwList(),
>+          useSurface ? supportedSurfaceColorList : supportedColorList);
>       if (profile == H264Profile.CONSTRAINED_HIGH.getValue()) {
>         EncoderProperties h264HighProfileProperties = findHwEncoder(H264_MIME_TYPE,
>             h264HighProfileHwList, useSurface ? supportedSurfaceColorList : supportedColorList);
>@@ -458,6 +599,8 @@ public class MediaCodecVideoEncoder {
>         }
>       }
>       keyFrameIntervalSec = 20;
>+    } else {
>+      throw new RuntimeException("initEncode: Non-supported codec " + type);
>     }
>     if (properties == null) {
>       throw new RuntimeException("Can not find HW encoder for " + type);
>@@ -517,7 +660,7 @@ public class MediaCodecVideoEncoder {
>       mediaCodec.configure(format, null, null, MediaCodec.CONFIGURE_FLAG_ENCODE);
> 
>       if (useSurface) {
>-        eglBase = new EglBase14(sharedContext, EglBase.CONFIG_RECORDABLE);
>+        eglBase = new EglBase14((EglBase14.Context) getEglContext(), EglBase.CONFIG_RECORDABLE);
>         // Create an input surface and keep a reference since we must release the surface when done.
>         inputSurface = mediaCodec.createInputSurface();
>         eglBase.createSurface(inputSurface);
>@@ -535,6 +678,7 @@ public class MediaCodecVideoEncoder {
>     return true;
>   }
> 
>+  @CalledByNativeUnchecked
>   ByteBuffer[] getInputBuffers() {
>     ByteBuffer[] inputBuffers = mediaCodec.getInputBuffers();
>     Logging.d(TAG, "Input buffers: " + inputBuffers.length);
>@@ -568,6 +712,7 @@ public class MediaCodecVideoEncoder {
>     }
>   }
> 
>+  @CalledByNativeUnchecked
>   boolean encodeBuffer(
>       boolean isKeyframe, int inputBuffer, int size, long presentationTimestampUs) {
>     checkOnMediaCodecThread();
>@@ -581,32 +726,14 @@ public class MediaCodecVideoEncoder {
>     }
>   }
> 
>-  boolean encodeTexture(boolean isKeyframe, int oesTextureId, float[] transformationMatrix,
>-      long presentationTimestampUs) {
>-    checkOnMediaCodecThread();
>-    try {
>-      checkKeyFrameRequired(isKeyframe, presentationTimestampUs);
>-      eglBase.makeCurrent();
>-      // TODO(perkj): glClear() shouldn't be necessary since every pixel is covered anyway,
>-      // but it's a workaround for bug webrtc:5147.
>-      GLES20.glClear(GLES20.GL_COLOR_BUFFER_BIT);
>-      drawer.drawOes(oesTextureId, transformationMatrix, width, height, 0, 0, width, height);
>-      eglBase.swapBuffers(TimeUnit.MICROSECONDS.toNanos(presentationTimestampUs));
>-      return true;
>-    } catch (RuntimeException e) {
>-      Logging.e(TAG, "encodeTexture failed", e);
>-      return false;
>-    }
>-  }
>-
>   /**
>-   * Encodes a new style VideoFrame. Called by JNI. |bufferIndex| is -1 if we are not encoding in
>-   * surface mode.
>+   * Encodes a new style VideoFrame. |bufferIndex| is -1 if we are not encoding in surface mode.
>    */
>-  boolean encodeFrame(long nativeEncoder, boolean isKeyframe, VideoFrame frame, int bufferIndex) {
>+  @CalledByNativeUnchecked
>+  boolean encodeFrame(long nativeEncoder, boolean isKeyframe, VideoFrame frame, int bufferIndex,
>+      long presentationTimestampUs) {
>     checkOnMediaCodecThread();
>     try {
>-      long presentationTimestampUs = TimeUnit.NANOSECONDS.toMicros(frame.getTimestampNs());
>       checkKeyFrameRequired(isKeyframe, presentationTimestampUs);
> 
>       VideoFrame.Buffer buffer = frame.getBuffer();
>@@ -618,7 +745,7 @@ public class MediaCodecVideoEncoder {
>         GLES20.glClear(GLES20.GL_COLOR_BUFFER_BIT);
>         VideoFrameDrawer.drawTexture(drawer, textureBuffer, new Matrix() /* renderMatrix */, width,
>             height, 0 /* viewportX */, 0 /* viewportY */, width, height);
>-        eglBase.swapBuffers(frame.getTimestampNs());
>+        eglBase.swapBuffers(TimeUnit.MICROSECONDS.toNanos(presentationTimestampUs));
>       } else {
>         VideoFrame.I420Buffer i420Buffer = buffer.toI420();
>         final int chromaHeight = (height + 1) / 2;
>@@ -637,7 +764,7 @@ public class MediaCodecVideoEncoder {
>         if (dataV.capacity() < strideV * chromaHeight) {
>           throw new RuntimeException("V-plane buffer size too small.");
>         }
>-        nativeFillBuffer(
>+        nativeFillInputBuffer(
>             nativeEncoder, bufferIndex, dataY, strideY, dataU, strideU, dataV, strideV);
>         i420Buffer.release();
>         // I420 consists of one full-resolution and two half-resolution planes.
>@@ -652,6 +779,7 @@ public class MediaCodecVideoEncoder {
>     }
>   }
> 
>+  @CalledByNativeUnchecked
>   void release() {
>     Logging.d(TAG, "Java releaseEncoder");
>     checkOnMediaCodecThread();
>@@ -733,6 +861,7 @@ public class MediaCodecVideoEncoder {
>     Logging.d(TAG, "Java releaseEncoder done");
>   }
> 
>+  @CalledByNativeUnchecked
>   private boolean setRates(int kbps, int frameRate) {
>     checkOnMediaCodecThread();
> 
>@@ -775,6 +904,7 @@ public class MediaCodecVideoEncoder {
> 
>   // Dequeue an input buffer and return its index, -1 if no input buffer is
>   // available, or -2 if the codec is no longer operative.
>+  @CalledByNativeUnchecked
>   int dequeueInputBuffer() {
>     checkOnMediaCodecThread();
>     try {
>@@ -799,10 +929,32 @@ public class MediaCodecVideoEncoder {
>     public final ByteBuffer buffer;
>     public final boolean isKeyFrame;
>     public final long presentationTimestampUs;
>+
>+    @CalledByNative("OutputBufferInfo")
>+    int getIndex() {
>+      return index;
>+    }
>+
>+    @CalledByNative("OutputBufferInfo")
>+    ByteBuffer getBuffer() {
>+      return buffer;
>+    }
>+
>+    @CalledByNative("OutputBufferInfo")
>+    boolean isKeyFrame() {
>+      return isKeyFrame;
>+    }
>+
>+    @CalledByNative("OutputBufferInfo")
>+    long getPresentationTimestampUs() {
>+      return presentationTimestampUs;
>+    }
>   }
> 
>   // Dequeue and return an output buffer, or null if no output is ready.  Return
>   // a fake OutputBufferInfo with index -1 if the codec is no longer operable.
>+  @Nullable
>+  @CalledByNativeUnchecked
>   OutputBufferInfo dequeueOutputBuffer() {
>     checkOnMediaCodecThread();
>     try {
>@@ -925,6 +1077,7 @@ public class MediaCodecVideoEncoder {
> 
>   // Release a dequeued output buffer back to the codec for re-use.  Return
>   // false if the codec is no longer operable.
>+  @CalledByNativeUnchecked
>   boolean releaseOutputBuffer(int index) {
>     checkOnMediaCodecThread();
>     try {
>@@ -936,7 +1089,18 @@ public class MediaCodecVideoEncoder {
>     }
>   }
> 
>+  @CalledByNative
>+  int getColorFormat() {
>+    return colorFormat;
>+  }
>+
>+  @CalledByNative
>+  static boolean isTextureBuffer(VideoFrame.Buffer buffer) {
>+    return buffer instanceof VideoFrame.TextureBuffer;
>+  }
>+
>   /** Fills an inputBuffer with the given index with data from the byte buffers. */
>-  private static native void nativeFillBuffer(long nativeEncoder, int inputBuffer, ByteBuffer dataY,
>+  private static native void nativeFillInputBuffer(long encoder, int inputBuffer, ByteBuffer dataY,
>       int strideY, ByteBuffer dataU, int strideU, ByteBuffer dataV, int strideV);
>+  private static native long nativeCreateEncoder(VideoCodecInfo info, boolean hasEgl14Context);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/MediaConstraints.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/MediaConstraints.java
>index fc0c9c2c2d8..76f04836ea5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/MediaConstraints.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/MediaConstraints.java
>@@ -10,6 +10,7 @@
> 
> package org.webrtc;
> 
>+import javax.annotation.Nullable;
> import java.util.ArrayList;
> import java.util.List;
> 
>@@ -44,7 +45,7 @@ public class MediaConstraints {
>     }
> 
>     @Override
>-    public boolean equals(Object other) {
>+    public boolean equals(@Nullable Object other) {
>       if (this == other) {
>         return true;
>       }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/MediaSource.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/MediaSource.java
>index a26e889bd28..3f20dd7353a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/MediaSource.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/MediaSource.java
>@@ -13,7 +13,17 @@ package org.webrtc;
> /** Java wrapper for a C++ MediaSourceInterface. */
> public class MediaSource {
>   /** Tracks MediaSourceInterface.SourceState */
>-  public enum State { INITIALIZING, LIVE, ENDED, MUTED }
>+  public enum State {
>+    INITIALIZING,
>+    LIVE,
>+    ENDED,
>+    MUTED;
>+
>+    @CalledByNative("State")
>+    static State fromNativeIndex(int nativeIndex) {
>+      return values()[nativeIndex];
>+    }
>+  }
> 
>   final long nativeSource; // Package-protected for PeerConnectionFactory.
> 
>@@ -22,12 +32,12 @@ public class MediaSource {
>   }
> 
>   public State state() {
>-    return nativeState(nativeSource);
>+    return nativeGetState(nativeSource);
>   }
> 
>   public void dispose() {
>     JniCommon.nativeReleaseRef(nativeSource);
>   }
> 
>-  private static native State nativeState(long pointer);
>+  private static native State nativeGetState(long pointer);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/MediaStream.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/MediaStream.java
>index 59caf386c9e..7f4a43402d5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/MediaStream.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/MediaStream.java
>@@ -30,7 +30,7 @@ public class MediaStream {
>   }
> 
>   public boolean addTrack(AudioTrack track) {
>-    if (addAudioTrackToNativeStream(nativeStream, track.nativeTrack)) {
>+    if (nativeAddAudioTrackToNativeStream(nativeStream, track.nativeTrack)) {
>       audioTracks.add(track);
>       return true;
>     }
>@@ -38,7 +38,7 @@ public class MediaStream {
>   }
> 
>   public boolean addTrack(VideoTrack track) {
>-    if (addVideoTrackToNativeStream(nativeStream, track.nativeTrack)) {
>+    if (nativeAddVideoTrackToNativeStream(nativeStream, track.nativeTrack)) {
>       videoTracks.add(track);
>       return true;
>     }
>@@ -49,7 +49,7 @@ public class MediaStream {
>   // is called. If video track need to be preserved after MediaStream is destroyed it
>   // should be added to MediaStream using addPreservedTrack() call.
>   public boolean addPreservedTrack(VideoTrack track) {
>-    if (addVideoTrackToNativeStream(nativeStream, track.nativeTrack)) {
>+    if (nativeAddVideoTrackToNativeStream(nativeStream, track.nativeTrack)) {
>       preservedVideoTracks.add(track);
>       return true;
>     }
>@@ -58,13 +58,13 @@ public class MediaStream {
> 
>   public boolean removeTrack(AudioTrack track) {
>     audioTracks.remove(track);
>-    return removeNativeAudioTrack(nativeStream, track.nativeTrack);
>+    return nativeRemoveAudioTrack(nativeStream, track.nativeTrack);
>   }
> 
>   public boolean removeTrack(VideoTrack track) {
>     videoTracks.remove(track);
>     preservedVideoTracks.remove(track);
>-    return removeNativeVideoTrack(nativeStream, track.nativeTrack);
>+    return nativeRemoveVideoTrack(nativeStream, track.nativeTrack);
>   }
> 
>   @CalledByNative
>@@ -84,16 +84,16 @@ public class MediaStream {
>     while (!preservedVideoTracks.isEmpty()) {
>       removeTrack(preservedVideoTracks.get(0 /* index */));
>     }
>-    free(nativeStream);
>+    JniCommon.nativeReleaseRef(nativeStream);
>   }
> 
>-  public String label() {
>-    return getNativeLabel(nativeStream);
>+  public String getId() {
>+    return nativeGetId(nativeStream);
>   }
> 
>   @Override
>   public String toString() {
>-    return "[" + label() + ":A=" + audioTracks.size() + ":V=" + videoTracks.size() + "]";
>+    return "[" + getId() + ":A=" + audioTracks.size() + ":V=" + videoTracks.size() + "]";
>   }
> 
>   @CalledByNative
>@@ -130,17 +130,11 @@ public class MediaStream {
>     Logging.e(TAG, "Couldn't not find track");
>   }
> 
>-  private static native boolean addAudioTrackToNativeStream(
>-      long nativeStream, long nativeAudioTrack);
>-
>-  private static native boolean addVideoTrackToNativeStream(
>-      long nativeStream, long nativeVideoTrack);
>-
>-  private static native boolean removeNativeAudioTrack(long nativeStream, long nativeAudioTrack);
>-
>-  private static native boolean removeNativeVideoTrack(long nativeStream, long nativeVideoTrack);
>-
>-  private static native String getNativeLabel(long nativeStream);
>-
>-  private static native void free(long nativeStream);
>+  private static native boolean nativeAddAudioTrackToNativeStream(
>+      long stream, long nativeAudioTrack);
>+  private static native boolean nativeAddVideoTrackToNativeStream(
>+      long stream, long nativeVideoTrack);
>+  private static native boolean nativeRemoveAudioTrack(long stream, long nativeAudioTrack);
>+  private static native boolean nativeRemoveVideoTrack(long stream, long nativeVideoTrack);
>+  private static native String nativeGetId(long stream);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/MediaStreamTrack.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/MediaStreamTrack.java
>index b13ea9eafc7..198da1e3e8c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/MediaStreamTrack.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/MediaStreamTrack.java
>@@ -10,10 +10,23 @@
> 
> package org.webrtc;
> 
>+import javax.annotation.Nullable;
>+
> /** Java wrapper for a C++ MediaStreamTrackInterface. */
> public class MediaStreamTrack {
>+  public static final String AUDIO_TRACK_KIND = "audio";
>+  public static final String VIDEO_TRACK_KIND = "video";
>+
>   /** Tracks MediaStreamTrackInterface.TrackState */
>-  public enum State { LIVE, ENDED }
>+  public enum State {
>+    LIVE,
>+    ENDED;
>+
>+    @CalledByNative("State")
>+    static State fromNativeIndex(int nativeIndex) {
>+      return values()[nativeIndex];
>+    }
>+  }
> 
>   // Must be kept in sync with cricket::MediaType.
>   public enum MediaType {
>@@ -42,6 +55,21 @@ public class MediaStreamTrack {
>     }
>   }
> 
>+  /** Factory method to create an AudioTrack or VideoTrack subclass. */
>+  static @Nullable MediaStreamTrack createMediaStreamTrack(long nativeTrack) {
>+    if (nativeTrack == 0) {
>+      return null;
>+    }
>+    String trackKind = nativeGetKind(nativeTrack);
>+    if (trackKind.equals(AUDIO_TRACK_KIND)) {
>+      return new AudioTrack(nativeTrack);
>+    } else if (trackKind.equals(VIDEO_TRACK_KIND)) {
>+      return new VideoTrack(nativeTrack);
>+    } else {
>+      return null;
>+    }
>+  }
>+
>   final long nativeTrack;
> 
>   public MediaStreamTrack(long nativeTrack) {
>@@ -49,36 +77,32 @@ public class MediaStreamTrack {
>   }
> 
>   public String id() {
>-    return getNativeId(nativeTrack);
>+    return nativeGetId(nativeTrack);
>   }
> 
>   public String kind() {
>-    return getNativeKind(nativeTrack);
>+    return nativeGetKind(nativeTrack);
>   }
> 
>   public boolean enabled() {
>-    return getNativeEnabled(nativeTrack);
>+    return nativeGetEnabled(nativeTrack);
>   }
> 
>   public boolean setEnabled(boolean enable) {
>-    return setNativeEnabled(nativeTrack, enable);
>+    return nativeSetEnabled(nativeTrack, enable);
>   }
> 
>   public State state() {
>-    return getNativeState(nativeTrack);
>+    return nativeGetState(nativeTrack);
>   }
> 
>   public void dispose() {
>     JniCommon.nativeReleaseRef(nativeTrack);
>   }
> 
>-  private static native String getNativeId(long nativeTrack);
>-
>-  private static native String getNativeKind(long nativeTrack);
>-
>-  private static native boolean getNativeEnabled(long nativeTrack);
>-
>-  private static native boolean setNativeEnabled(long nativeTrack, boolean enabled);
>-
>-  private static native State getNativeState(long nativeTrack);
>+  private static native String nativeGetId(long track);
>+  private static native String nativeGetKind(long track);
>+  private static native boolean nativeGetEnabled(long track);
>+  private static native boolean nativeSetEnabled(long track, boolean enabled);
>+  private static native State nativeGetState(long track);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/Metrics.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/Metrics.java
>index 010e839fe23..3098f7265cb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/Metrics.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/Metrics.java
>@@ -13,7 +13,7 @@ package org.webrtc;
> import java.util.HashMap;
> import java.util.Map;
> 
>-// Java-side of androidmetrics_jni.cc.
>+// Java-side of androidmetrics.cc
> //
> // Rtc histograms can be queried through the API, getAndReset().
> // The returned map holds the name of a histogram and its samples.
>@@ -28,7 +28,6 @@ import java.util.Map;
> // Most histograms are not updated frequently (e.g. most video metrics are an
> // average over the call and recorded when a stream is removed).
> // The metrics can for example be retrieved when a peer connection is closed.
>-
> public class Metrics {
>   private static final String TAG = "Metrics";
> 
>@@ -69,14 +68,14 @@ public class Metrics {
>   // Enables gathering of metrics (which can be fetched with getAndReset()).
>   // Must be called before PeerConnectionFactory is created.
>   public static void enable() {
>-    enableNative();
>+    nativeEnable();
>   }
> 
>   // Gets and clears native histograms.
>   public static Metrics getAndReset() {
>-    return getAndResetNative();
>+    return nativeGetAndReset();
>   }
> 
>-  private static native void enableNative();
>-  private static native Metrics getAndResetNative();
>+  private static native void nativeEnable();
>+  private static native Metrics nativeGetAndReset();
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/PostProcessingFactory.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/NativePeerConnectionFactory.java
>similarity index 66%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/PostProcessingFactory.java
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/NativePeerConnectionFactory.java
>index 11b8fdb1188..aeb91e17508 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/PostProcessingFactory.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/NativePeerConnectionFactory.java
>@@ -10,11 +10,11 @@
> 
> package org.webrtc;
> 
>-/** Factory for creating webrtc::PostProcessing instances. */
>-public interface PostProcessingFactory {
>+/** Factory for creating webrtc::jni::OwnedPeerConnection instances. */
>+public interface NativePeerConnectionFactory {
>   /**
>-   * Dynamically allocates a webrtc::PostProcessing instance and returns a pointer to it.
>+   * Create a new webrtc::jni::OwnedPeerConnection instance and returns a pointer to it.
>    * The caller takes ownership of the object.
>    */
>-  public long createNative();
>+  long createNativePeerConnection();
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/NetworkMonitor.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/NetworkMonitor.java
>index ad070eb61e4..ba6bae0ed83 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/NetworkMonitor.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/NetworkMonitor.java
>@@ -10,27 +10,27 @@
> 
> package org.webrtc;
> 
>-import static org.webrtc.NetworkMonitorAutoDetect.ConnectionType;
> import static org.webrtc.NetworkMonitorAutoDetect.INVALID_NET_ID;
>-import static org.webrtc.NetworkMonitorAutoDetect.NetworkInformation;
> 
>-import android.annotation.SuppressLint;
> import android.content.Context;
> import android.os.Build;
>+import javax.annotation.Nullable;
> import java.util.ArrayList;
> import java.util.List;
>+import org.webrtc.NetworkMonitorAutoDetect.ConnectionType;
>+import org.webrtc.NetworkMonitorAutoDetect.NetworkInformation;
> 
> /**
>- * Borrowed from Chromium's src/net/android/java/src/org/chromium/net/NetworkChangeNotifier.java
>+ * Borrowed from Chromium's
>+ * src/net/android/java/src/org/chromium/net/NetworkChangeNotifier.java
>  *
>- * Triggers updates to the underlying network state from OS networking events.
>+ * <p>Triggers updates to the underlying network state from OS networking events.
>  *
>- * WARNING: This class is not thread-safe.
>+ * <p>This class is thread-safe.
>  */
> public class NetworkMonitor {
>   /**
>-   * Alerted when the connection type of the network changes.
>-   * The alert is fired on the UI thread.
>+   * Alerted when the connection type of the network changes. The alert is fired on the UI thread.
>    */
>   public interface NetworkObserver {
>     public void onConnectionTypeChanged(ConnectionType connectionType);
>@@ -38,37 +38,40 @@ public class NetworkMonitor {
> 
>   private static final String TAG = "NetworkMonitor";
> 
>-  // We are storing application context so it is okay.
>-  private static NetworkMonitor instance;
>+  // Lazy initialization holder class idiom for static fields.
>+  private static class InstanceHolder {
>+    // We are storing application context so it is okay.
>+    static final NetworkMonitor instance = new NetworkMonitor();
>+  }
> 
>   // Native observers of the connection type changes.
>   private final ArrayList<Long> nativeNetworkObservers;
>   // Java observers of the connection type changes.
>   private final ArrayList<NetworkObserver> networkObservers;
> 
>-  // Object that detects the connection type changes.
>-  private NetworkMonitorAutoDetect autoDetector;
>+  private final Object autoDetectLock = new Object();
>+  // Object that detects the connection type changes and brings up mobile networks.
>+  @Nullable private NetworkMonitorAutoDetect autoDetect;
>+  // Also guarded by autoDetectLock.
>+  private int numObservers;
> 
>-  private ConnectionType currentConnectionType = ConnectionType.CONNECTION_UNKNOWN;
>+  private volatile ConnectionType currentConnectionType;
> 
>   private NetworkMonitor() {
>     nativeNetworkObservers = new ArrayList<Long>();
>     networkObservers = new ArrayList<NetworkObserver>();
>+    numObservers = 0;
>+    currentConnectionType = ConnectionType.CONNECTION_UNKNOWN;
>   }
> 
>   // TODO(sakal): Remove once downstream dependencies have been updated.
>   @Deprecated
>   public static void init(Context context) {}
> 
>-  /**
>-   * Returns the singleton instance.
>-   */
>+  /** Returns the singleton instance. This may be called from native or from Java code. */
>   @CalledByNative
>   public static NetworkMonitor getInstance() {
>-    if (instance == null) {
>-      instance = new NetworkMonitor();
>-    }
>-    return instance;
>+    return InstanceHolder.instance;
>   }
> 
>   private static void assertIsTrue(boolean condition) {
>@@ -78,38 +81,73 @@ public class NetworkMonitor {
>   }
> 
>   /**
>-   * Enables auto detection of the current network state based on notifications
>-   * from the system. Note that this requires the embedding app have the
>-   * platform ACCESS_NETWORK_STATE permission.
>+   * Enables auto detection of the network state change and brings up mobile networks for using
>+   * multi-networking. This requires the embedding app have the platform ACCESS_NETWORK_STATE and
>+   * CHANGE_NETWORK_STATE permission.
>+   */
>+  public void startMonitoring(Context applicationContext) {
>+    synchronized (autoDetectLock) {
>+      ++numObservers;
>+      if (autoDetect == null) {
>+        autoDetect = createAutoDetect(applicationContext);
>+      }
>+      currentConnectionType =
>+          NetworkMonitorAutoDetect.getConnectionType(autoDetect.getCurrentNetworkState());
>+    }
>+  }
>+
>+  /** Deprecated, pass in application context in startMonitoring instead. */
>+  @Deprecated
>+  public void startMonitoring() {
>+    startMonitoring(ContextUtils.getApplicationContext());
>+  }
>+
>+  /**
>+   * Enables auto detection of the network state change and brings up mobile networks for using
>+   * multi-networking. This requires the embedding app have the platform ACCESS_NETWORK_STATE and
>+   * CHANGE_NETWORK_STATE permission.
>    */
>   @CalledByNative
>-  private void startMonitoring(long nativeObserver) {
>-    Logging.d(TAG, "Start monitoring from native observer " + nativeObserver);
>-    nativeNetworkObservers.add(nativeObserver);
>-    if (autoDetector == null) {
>-      createAutoDetector();
>-    }
>-    // The observers expect a network list update after they call startMonitoring.
>-    final NetworkMonitorAutoDetect.NetworkState networkState =
>-        autoDetector.getCurrentNetworkState();
>-    updateCurrentConnectionType(NetworkMonitorAutoDetect.getConnectionType(networkState));
>+  private void startMonitoring(@Nullable Context applicationContext, long nativeObserver) {
>+    Logging.d(TAG, "Start monitoring with native observer " + nativeObserver);
>+
>+    startMonitoring(
>+        applicationContext != null ? applicationContext : ContextUtils.getApplicationContext());
>+    // The native observers expect a network list update after they call startMonitoring.
>+    synchronized (nativeNetworkObservers) {
>+      nativeNetworkObservers.add(nativeObserver);
>+    }
>     updateObserverActiveNetworkList(nativeObserver);
>+    // currentConnectionType was updated in startMonitoring().
>+    // Need to notify the native observers here.
>+    notifyObserversOfConnectionTypeChange(currentConnectionType);
>+  }
>+
>+  /** Stop network monitoring. If no one is monitoring networks, destroy and reset autoDetect. */
>+  public void stopMonitoring() {
>+    synchronized (autoDetectLock) {
>+      if (--numObservers == 0) {
>+        autoDetect.destroy();
>+        autoDetect = null;
>+      }
>+    }
>   }
> 
>   @CalledByNative
>   private void stopMonitoring(long nativeObserver) {
>-    Logging.d(TAG, "Stop monitoring from native observer " + nativeObserver);
>-    nativeNetworkObservers.remove(nativeObserver);
>-    if (nativeNetworkObservers.isEmpty()) {
>-      autoDetector.destroy();
>-      autoDetector = null;
>+    Logging.d(TAG, "Stop monitoring with native observer " + nativeObserver);
>+    stopMonitoring();
>+    synchronized (nativeNetworkObservers) {
>+      nativeNetworkObservers.remove(nativeObserver);
>     }
>   }
> 
>   // Returns true if network binding is supported on this platform.
>   @CalledByNative
>   private boolean networkBindingSupported() {
>-    return autoDetector != null && autoDetector.supportNetworkCallback();
>+    synchronized (autoDetectLock) {
>+      return autoDetect != null && autoDetect.supportNetworkCallback();
>+    }
>   }
> 
>   @CalledByNative
>@@ -122,11 +160,13 @@ public class NetworkMonitor {
>   }
> 
>   private long getCurrentDefaultNetId() {
>-    return autoDetector == null ? INVALID_NET_ID : autoDetector.getDefaultNetId();
>+    synchronized (autoDetectLock) {
>+      return autoDetect == null ? INVALID_NET_ID : autoDetect.getDefaultNetId();
>+    }
>   }
> 
>-  private void createAutoDetector() {
>-    autoDetector = new NetworkMonitorAutoDetect(new NetworkMonitorAutoDetect.Observer() {
>+  private NetworkMonitorAutoDetect createAutoDetect(Context appContext) {
>+    return new NetworkMonitorAutoDetect(new NetworkMonitorAutoDetect.Observer() {
> 
>       @Override
>       public void onConnectionTypeChanged(ConnectionType newConnectionType) {
>@@ -142,7 +182,7 @@ public class NetworkMonitor {
>       public void onNetworkDisconnect(long networkHandle) {
>         notifyObserversOfNetworkDisconnect(networkHandle);
>       }
>-    }, ContextUtils.getApplicationContext());
>+    }, appContext);
>   }
> 
>   private void updateCurrentConnectionType(ConnectionType newConnectionType) {
>@@ -150,32 +190,41 @@ public class NetworkMonitor {
>     notifyObserversOfConnectionTypeChange(newConnectionType);
>   }
> 
>-  /**
>-   * Alerts all observers of a connection change.
>-   */
>+  /** Alerts all observers of a connection change. */
>   private void notifyObserversOfConnectionTypeChange(ConnectionType newConnectionType) {
>-    for (long nativeObserver : nativeNetworkObservers) {
>+    List<Long> nativeObservers = getNativeNetworkObserversSync();
>+    for (Long nativeObserver : nativeObservers) {
>       nativeNotifyConnectionTypeChanged(nativeObserver);
>     }
>-    for (NetworkObserver observer : networkObservers) {
>+    // This avoids calling external methods while locking on an object.
>+    List<NetworkObserver> javaObservers;
>+    synchronized (networkObservers) {
>+      javaObservers = new ArrayList<>(networkObservers);
>+    }
>+    for (NetworkObserver observer : javaObservers) {
>       observer.onConnectionTypeChanged(newConnectionType);
>     }
>   }
> 
>   private void notifyObserversOfNetworkConnect(NetworkInformation networkInfo) {
>-    for (long nativeObserver : nativeNetworkObservers) {
>+    List<Long> nativeObservers = getNativeNetworkObserversSync();
>+    for (Long nativeObserver : nativeObservers) {
>       nativeNotifyOfNetworkConnect(nativeObserver, networkInfo);
>     }
>   }
> 
>   private void notifyObserversOfNetworkDisconnect(long networkHandle) {
>-    for (long nativeObserver : nativeNetworkObservers) {
>+    List<Long> nativeObservers = getNativeNetworkObserversSync();
>+    for (Long nativeObserver : nativeObservers) {
>       nativeNotifyOfNetworkDisconnect(nativeObserver, networkHandle);
>     }
>   }
> 
>   private void updateObserverActiveNetworkList(long nativeObserver) {
>-    List<NetworkInformation> networkInfoList = autoDetector.getActiveNetworkList();
>+    List<NetworkInformation> networkInfoList;
>+    synchronized (autoDetectLock) {
>+      networkInfoList = (autoDetect == null) ? null : autoDetect.getActiveNetworkList();
>+    }
>     if (networkInfoList == null || networkInfoList.size() == 0) {
>       return;
>     }
>@@ -185,61 +234,77 @@ public class NetworkMonitor {
>     nativeNotifyOfActiveNetworkList(nativeObserver, networkInfos);
>   }
> 
>+  private List<Long> getNativeNetworkObserversSync() {
>+    synchronized (nativeNetworkObservers) {
>+      return new ArrayList<>(nativeNetworkObservers);
>+    }
>+  }
>+
>   /**
>    * Adds an observer for any connection type changes.
>+   *
>+   * @deprecated Use getInstance(appContext).addObserver instead.
>    */
>+  @Deprecated
>   public static void addNetworkObserver(NetworkObserver observer) {
>-    getInstance().addNetworkObserverInternal(observer);
>+    getInstance().addObserver(observer);
>   }
> 
>-  private void addNetworkObserverInternal(NetworkObserver observer) {
>-    networkObservers.add(observer);
>+  public void addObserver(NetworkObserver observer) {
>+    synchronized (networkObservers) {
>+      networkObservers.add(observer);
>+    }
>   }
> 
>   /**
>    * Removes an observer for any connection type changes.
>+   *
>+   * @deprecated Use getInstance(appContext).removeObserver instead.
>    */
>+  @Deprecated
>   public static void removeNetworkObserver(NetworkObserver observer) {
>-    getInstance().removeNetworkObserverInternal(observer);
>+    getInstance().removeObserver(observer);
>   }
> 
>-  private void removeNetworkObserverInternal(NetworkObserver observer) {
>-    networkObservers.remove(observer);
>+  public void removeObserver(NetworkObserver observer) {
>+    synchronized (networkObservers) {
>+      networkObservers.remove(observer);
>+    }
>   }
> 
>-  /**
>-   * Checks if there currently is connectivity.
>-   */
>+  /** Checks if there currently is connectivity. */
>   public static boolean isOnline() {
>     ConnectionType connectionType = getInstance().getCurrentConnectionType();
>     return connectionType != ConnectionType.CONNECTION_NONE;
>   }
> 
>-  @NativeClassQualifiedName("webrtc::jni::AndroidNetworkMonitor")
>-  private native void nativeNotifyConnectionTypeChanged(long nativePtr);
>-
>-  @NativeClassQualifiedName("webrtc::jni::AndroidNetworkMonitor")
>-  private native void nativeNotifyOfNetworkConnect(long nativePtr, NetworkInformation networkInfo);
>-
>-  @NativeClassQualifiedName("webrtc::jni::AndroidNetworkMonitor")
>-  private native void nativeNotifyOfNetworkDisconnect(long nativePtr, long networkHandle);
>-
>-  @NativeClassQualifiedName("webrtc::jni::AndroidNetworkMonitor")
>+  private native void nativeNotifyConnectionTypeChanged(long nativeAndroidNetworkMonitor);
>+  private native void nativeNotifyOfNetworkConnect(
>+      long nativeAndroidNetworkMonitor, NetworkInformation networkInfo);
>+  private native void nativeNotifyOfNetworkDisconnect(
>+      long nativeAndroidNetworkMonitor, long networkHandle);
>   private native void nativeNotifyOfActiveNetworkList(
>-      long nativePtr, NetworkInformation[] networkInfos);
>+      long nativeAndroidNetworkMonitor, NetworkInformation[] networkInfos);
> 
>   // For testing only.
>-  static void resetInstanceForTests() {
>-    instance = new NetworkMonitor();
>+  @Nullable
>+  NetworkMonitorAutoDetect getNetworkMonitorAutoDetect() {
>+    synchronized (autoDetectLock) {
>+      return autoDetect;
>+    }
>   }
> 
>   // For testing only.
>-  static void createAutoDetectorForTest() {
>-    getInstance().createAutoDetector();
>+  int getNumObservers() {
>+    synchronized (autoDetectLock) {
>+      return numObservers;
>+    }
>   }
> 
>   // For testing only.
>-  static NetworkMonitorAutoDetect getAutoDetectorForTest() {
>-    return getInstance().autoDetector;
>+  static NetworkMonitorAutoDetect createAndSetAutoDetectForTest(Context context) {
>+    NetworkMonitor networkMonitor = getInstance();
>+    NetworkMonitorAutoDetect autoDetect = networkMonitor.createAutoDetect(context);
>+    return networkMonitor.autoDetect = autoDetect;
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/NetworkMonitorAutoDetect.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/NetworkMonitorAutoDetect.java
>index 86eb1c20f69..d8c83440b26 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/NetworkMonitorAutoDetect.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/NetworkMonitorAutoDetect.java
>@@ -35,6 +35,7 @@ import java.net.SocketException;
> import java.util.ArrayList;
> import java.util.Collections;
> import java.util.List;
>+import javax.annotation.Nullable;
> 
> /**
>  * Borrowed from Chromium's
>@@ -54,6 +55,7 @@ public class NetworkMonitorAutoDetect extends BroadcastReceiver {
>     CONNECTION_2G,
>     CONNECTION_UNKNOWN_CELLULAR,
>     CONNECTION_BLUETOOTH,
>+    CONNECTION_VPN,
>     CONNECTION_NONE
>   }
> 
>@@ -73,12 +75,15 @@ public class NetworkMonitorAutoDetect extends BroadcastReceiver {
>   public static class NetworkInformation {
>     public final String name;
>     public final ConnectionType type;
>+    // Used to specify the underlying network type if the type is CONNECTION_VPN.
>+    public final ConnectionType underlyingTypeForVpn;
>     public final long handle;
>     public final IPAddress[] ipAddresses;
>-    public NetworkInformation(
>-        String name, ConnectionType type, long handle, IPAddress[] addresses) {
>+    public NetworkInformation(String name, ConnectionType type, ConnectionType underlyingTypeForVpn,
>+        long handle, IPAddress[] addresses) {
>       this.name = name;
>       this.type = type;
>+      this.underlyingTypeForVpn = underlyingTypeForVpn;
>       this.handle = handle;
>       this.ipAddresses = addresses;
>     }
>@@ -93,6 +98,11 @@ public class NetworkMonitorAutoDetect extends BroadcastReceiver {
>       return type;
>     }
> 
>+    @CalledByNative("NetworkInformation")
>+    private ConnectionType getUnderlyingConnectionTypeForVpn() {
>+      return underlyingTypeForVpn;
>+    }
>+
>     @CalledByNative("NetworkInformation")
>     private long getHandle() {
>       return handle;
>@@ -112,11 +122,18 @@ public class NetworkMonitorAutoDetect extends BroadcastReceiver {
>     // Defined from NetworkInfo.subtype, which is one of the TelephonyManager.NETWORK_TYPE_XXXs.
>     // Will be useful to find the maximum bandwidth.
>     private final int subtype;
>+    // When the type is TYPE_VPN, the following two fields specify the similar type and subtype as
>+    // above for the underlying network that is used by the VPN.
>+    private final int underlyingNetworkTypeForVpn;
>+    private final int underlyingNetworkSubtypeForVpn;
> 
>-    public NetworkState(boolean connected, int type, int subtype) {
>+    public NetworkState(boolean connected, int type, int subtype, int underlyingNetworkTypeForVpn,
>+        int underlyingNetworkSubtypeForVpn) {
>       this.connected = connected;
>       this.type = type;
>       this.subtype = subtype;
>+      this.underlyingNetworkTypeForVpn = underlyingNetworkTypeForVpn;
>+      this.underlyingNetworkSubtypeForVpn = underlyingNetworkSubtypeForVpn;
>     }
> 
>     public boolean isConnected() {
>@@ -130,6 +147,14 @@ public class NetworkMonitorAutoDetect extends BroadcastReceiver {
>     public int getNetworkSubType() {
>       return subtype;
>     }
>+
>+    public int getUnderlyingNetworkTypeForVpn() {
>+      return underlyingNetworkTypeForVpn;
>+    }
>+
>+    public int getUnderlyingNetworkSubtypeForVpn() {
>+      return underlyingNetworkSubtypeForVpn;
>+    }
>   }
>   /**
>    * The methods in this class get called when the network changes if the callback
>@@ -188,7 +213,7 @@ public class NetworkMonitorAutoDetect extends BroadcastReceiver {
>      *  Note: In some rare Android systems connectivityManager is null.  We handle that
>      *  gracefully below.
>      */
>-    private final ConnectivityManager connectivityManager;
>+    @Nullable private final ConnectivityManager connectivityManager;
> 
>     ConnectivityManagerDelegate(Context context) {
>       connectivityManager =
>@@ -207,7 +232,7 @@ public class NetworkMonitorAutoDetect extends BroadcastReceiver {
>      */
>     NetworkState getNetworkState() {
>       if (connectivityManager == null) {
>-        return new NetworkState(false, -1, -1);
>+        return new NetworkState(false, -1, -1, -1, -1);
>       }
>       return getNetworkState(connectivityManager.getActiveNetworkInfo());
>     }
>@@ -217,21 +242,79 @@ public class NetworkMonitorAutoDetect extends BroadcastReceiver {
>      * Only callable on Lollipop and newer releases.
>      */
>     @SuppressLint("NewApi")
>-    NetworkState getNetworkState(Network network) {
>-      if (connectivityManager == null) {
>-        return new NetworkState(false, -1, -1);
>+    NetworkState getNetworkState(@Nullable Network network) {
>+      if (network == null || connectivityManager == null) {
>+        return new NetworkState(false, -1, -1, -1, -1);
>+      }
>+      NetworkInfo networkInfo = connectivityManager.getNetworkInfo(network);
>+      if (networkInfo == null) {
>+        Logging.w(TAG, "Couldn't retrieve information from network " + network.toString());
>+        return new NetworkState(false, -1, -1, -1, -1);
>+      }
>+      // The general logic of handling a VPN in this method is as follows. getNetworkInfo will
>+      // return the info of the network with the same id as in |network| when it is registered via
>+      // ConnectivityManager.registerNetworkAgent in Android. |networkInfo| may or may not indicate
>+      // the type TYPE_VPN if |network| is a VPN. To reliably detect the VPN interface, we need to
>+      // query the network capability as below in the case when networkInfo.getType() is not
>+      // TYPE_VPN. On the other hand when networkInfo.getType() is TYPE_VPN, the only solution so
>+      // far to obtain the underlying network information is to query the active network interface.
>+      // However, the active network interface may not be used for the VPN, for example, if the VPN
>+      // is restricted to WiFi by the implementation but the WiFi interface is currently turned
>+      // off and the active interface is the Cell. Using directly the result from
>+      // getActiveNetworkInfo may thus give the wrong interface information, and one should note
>+      // that getActiveNetworkInfo would return the default network interface if the VPN does not
>+      // specify its underlying networks in the implementation. Therefore, we need further compare
>+      // |network| to the active network. If they are not the same network, we will have to fall
>+      // back to report an unknown network.
>+
>+      if (networkInfo.getType() != ConnectivityManager.TYPE_VPN) {
>+        // Note that getNetworkCapabilities returns null if the network is unknown.
>+        NetworkCapabilities networkCapabilities =
>+            connectivityManager.getNetworkCapabilities(network);
>+        if (networkCapabilities == null
>+            || !networkCapabilities.hasTransport(NetworkCapabilities.TRANSPORT_VPN)) {
>+          return getNetworkState(networkInfo);
>+        }
>+        // When |network| is in fact a VPN after querying its capability but |networkInfo| is not of
>+        // type TYPE_VPN, |networkInfo| contains the info for the underlying network, and we return
>+        // a NetworkState constructed from it.
>+        return new NetworkState(networkInfo.isConnected(), ConnectivityManager.TYPE_VPN, -1,
>+            networkInfo.getType(), networkInfo.getSubtype());
>+      }
>+
>+      // When |networkInfo| is of type TYPE_VPN, which implies |network| is a VPN, we return the
>+      // NetworkState of the active network via getActiveNetworkInfo(), if |network| is the active
>+      // network that supports the VPN. Otherwise, NetworkState of an unknown network with type -1
>+      // will be returned.
>+      //
>+      // Note that getActiveNetwork and getActiveNetworkInfo return null if no default network is
>+      // currently active.
>+      if (networkInfo.getType() == ConnectivityManager.TYPE_VPN
>+          && network.equals(connectivityManager.getActiveNetwork())) {
>+        // If a VPN network is in place, we can find the underlying network type via querying the
>+        // active network info thanks to
>+        // https://android.googlesource.com/platform/frameworks/base/+/d6a7980d
>+        NetworkInfo underlyingActiveNetworkInfo = connectivityManager.getActiveNetworkInfo();
>+        // We use the NetworkInfo of the underlying network if it is not of TYPE_VPN itself.
>+        if (underlyingActiveNetworkInfo.getType() != ConnectivityManager.TYPE_VPN) {
>+          return new NetworkState(networkInfo.isConnected(), ConnectivityManager.TYPE_VPN, -1,
>+              underlyingActiveNetworkInfo.getType(), underlyingActiveNetworkInfo.getSubtype());
>+        }
>       }
>-      return getNetworkState(connectivityManager.getNetworkInfo(network));
>+
>+      return getNetworkState(networkInfo);
>     }
> 
>     /**
>-     * Returns connection type and status information gleaned from networkInfo.
>+     * Returns connection type and status information gleaned from networkInfo. Note that to obtain
>+     * the complete information about a VPN including the type of the underlying network, one should
>+     * use the above method getNetworkState with a Network object.
>      */
>-    NetworkState getNetworkState(NetworkInfo networkInfo) {
>+    private NetworkState getNetworkState(@Nullable NetworkInfo networkInfo) {
>       if (networkInfo == null || !networkInfo.isConnected()) {
>-        return new NetworkState(false, -1, -1);
>+        return new NetworkState(false, -1, -1, -1, -1);
>       }
>-      return new NetworkState(true, networkInfo.getType(), networkInfo.getSubtype());
>+      return new NetworkState(true, networkInfo.getType(), networkInfo.getSubtype(), -1, -1);
>     }
> 
>     /**
>@@ -246,6 +329,7 @@ public class NetworkMonitorAutoDetect extends BroadcastReceiver {
>       return connectivityManager.getAllNetworks();
>     }
> 
>+    @Nullable
>     List<NetworkInformation> getActiveNetworkList() {
>       if (!supportNetworkCallback()) {
>         return null;
>@@ -304,7 +388,10 @@ public class NetworkMonitorAutoDetect extends BroadcastReceiver {
>     }
> 
>     @SuppressLint("NewApi")
>-    private NetworkInformation networkToInfo(Network network) {
>+    private @Nullable NetworkInformation networkToInfo(@Nullable Network network) {
>+      if (network == null || connectivityManager == null) {
>+        return null;
>+      }
>       LinkProperties linkProperties = connectivityManager.getLinkProperties(network);
>       // getLinkProperties will return null if the network is unknown.
>       if (linkProperties == null) {
>@@ -317,12 +404,6 @@ public class NetworkMonitorAutoDetect extends BroadcastReceiver {
>       }
> 
>       NetworkState networkState = getNetworkState(network);
>-      if (networkState.connected && networkState.getNetworkType() == ConnectivityManager.TYPE_VPN) {
>-        // If a VPN network is in place, we can find the underlying network type via querying the
>-        // active network info thanks to
>-        // https://android.googlesource.com/platform/frameworks/base/+/d6a7980d
>-        networkState = getNetworkState();
>-      }
>       ConnectionType connectionType = getConnectionType(networkState);
>       if (connectionType == ConnectionType.CONNECTION_NONE) {
>         // This may not be an error. The OS may signal a network event with connection type
>@@ -339,10 +420,14 @@ public class NetworkMonitorAutoDetect extends BroadcastReceiver {
>                 + " because it has type " + networkState.getNetworkType() + " and subtype "
>                 + networkState.getNetworkSubType());
>       }
>-
>-      NetworkInformation networkInformation =
>-          new NetworkInformation(linkProperties.getInterfaceName(), connectionType,
>-              networkToNetId(network), getIPAddresses(linkProperties));
>+      // ConnectionType.CONNECTION_UNKNOWN if the network is not a VPN or the underlying network is
>+      // unknown.
>+      ConnectionType underlyingConnectionTypeForVpn =
>+          getUnderlyingConnectionTypeForVpn(networkState);
>+
>+      NetworkInformation networkInformation = new NetworkInformation(
>+          linkProperties.getInterfaceName(), connectionType, underlyingConnectionTypeForVpn,
>+          networkToNetId(network), getIPAddresses(linkProperties));
>       return networkInformation;
>     }
> 
>@@ -405,7 +490,7 @@ public class NetworkMonitorAutoDetect extends BroadcastReceiver {
> 
>   /** Queries the WifiManager for SSID of the current Wifi connection. */
>   static class WifiManagerDelegate {
>-    private final Context context;
>+    @Nullable private final Context context;
>     WifiManagerDelegate(Context context) {
>       this.context = context;
>     }
>@@ -441,7 +526,7 @@ public class NetworkMonitorAutoDetect extends BroadcastReceiver {
>     private final Observer observer;
>     // Network information about a WifiP2p (aka WiFi-Direct) network, or null if no such network is
>     // connected.
>-    private NetworkInformation wifiP2pNetworkInfo = null;
>+    @Nullable private NetworkInformation wifiP2pNetworkInfo = null;
> 
>     WifiDirectManagerDelegate(Observer observer, Context context) {
>       this.context = context;
>@@ -479,7 +564,7 @@ public class NetworkMonitorAutoDetect extends BroadcastReceiver {
>     }
> 
>     /** Handle a change notification about the wifi p2p group. */
>-    private void onWifiP2pGroupChange(WifiP2pGroup wifiP2pGroup) {
>+    private void onWifiP2pGroupChange(@Nullable WifiP2pGroup wifiP2pGroup) {
>       if (wifiP2pGroup == null || wifiP2pGroup.getInterface() == null) {
>         return;
>       }
>@@ -499,11 +584,8 @@ public class NetworkMonitorAutoDetect extends BroadcastReceiver {
>       }
> 
>       wifiP2pNetworkInfo =
>-          new NetworkInformation(
>-              wifiP2pGroup.getInterface(),
>-              ConnectionType.CONNECTION_WIFI,
>-              WIFI_P2P_NETWORK_HANDLE,
>-              ipAddresses);
>+          new NetworkInformation(wifiP2pGroup.getInterface(), ConnectionType.CONNECTION_WIFI,
>+              ConnectionType.CONNECTION_NONE, WIFI_P2P_NETWORK_HANDLE, ipAddresses);
>       observer.onNetworkConnect(wifiP2pNetworkInfo);
>     }
> 
>@@ -525,9 +607,9 @@ public class NetworkMonitorAutoDetect extends BroadcastReceiver {
>   private final Context context;
>   // Used to request mobile network. It does not do anything except for keeping
>   // the callback for releasing the request.
>-  private final NetworkCallback mobileNetworkCallback;
>+  @Nullable private final NetworkCallback mobileNetworkCallback;
>   // Used to receive updates on all networks.
>-  private final NetworkCallback allNetworkCallback;
>+  @Nullable private final NetworkCallback allNetworkCallback;
>   // connectivityManagerDelegate and wifiManagerDelegate are only non-final for testing.
>   private ConnectivityManagerDelegate connectivityManagerDelegate;
>   private WifiManagerDelegate wifiManagerDelegate;
>@@ -614,6 +696,7 @@ public class NetworkMonitorAutoDetect extends BroadcastReceiver {
>     return isRegistered;
>   }
> 
>+  @Nullable
>   List<NetworkInformation> getActiveNetworkList() {
>     List<NetworkInformation> connectivityManagerList =
>         connectivityManagerDelegate.getActiveNetworkList();
>@@ -677,12 +760,13 @@ public class NetworkMonitorAutoDetect extends BroadcastReceiver {
>     return connectivityManagerDelegate.getDefaultNetId();
>   }
> 
>-  public static ConnectionType getConnectionType(NetworkState networkState) {
>-    if (!networkState.isConnected()) {
>+  private static ConnectionType getConnectionType(
>+      boolean isConnected, int networkType, int networkSubtype) {
>+    if (!isConnected) {
>       return ConnectionType.CONNECTION_NONE;
>     }
> 
>-    switch (networkState.getNetworkType()) {
>+    switch (networkType) {
>       case ConnectivityManager.TYPE_ETHERNET:
>         return ConnectionType.CONNECTION_ETHERNET;
>       case ConnectivityManager.TYPE_WIFI:
>@@ -693,7 +777,7 @@ public class NetworkMonitorAutoDetect extends BroadcastReceiver {
>         return ConnectionType.CONNECTION_BLUETOOTH;
>       case ConnectivityManager.TYPE_MOBILE:
>         // Use information from TelephonyManager to classify the connection.
>-        switch (networkState.getNetworkSubType()) {
>+        switch (networkSubtype) {
>           case TelephonyManager.NETWORK_TYPE_GPRS:
>           case TelephonyManager.NETWORK_TYPE_EDGE:
>           case TelephonyManager.NETWORK_TYPE_CDMA:
>@@ -715,11 +799,27 @@ public class NetworkMonitorAutoDetect extends BroadcastReceiver {
>           default:
>             return ConnectionType.CONNECTION_UNKNOWN_CELLULAR;
>         }
>+      case ConnectivityManager.TYPE_VPN:
>+        return ConnectionType.CONNECTION_VPN;
>       default:
>         return ConnectionType.CONNECTION_UNKNOWN;
>     }
>   }
> 
>+  public static ConnectionType getConnectionType(NetworkState networkState) {
>+    return getConnectionType(networkState.isConnected(), networkState.getNetworkType(),
>+        networkState.getNetworkSubType());
>+  }
>+
>+  private static ConnectionType getUnderlyingConnectionTypeForVpn(NetworkState networkState) {
>+    if (networkState.getNetworkType() != ConnectivityManager.TYPE_VPN) {
>+      return ConnectionType.CONNECTION_NONE;
>+    }
>+    return getConnectionType(networkState.isConnected(),
>+        networkState.getUnderlyingNetworkTypeForVpn(),
>+        networkState.getUnderlyingNetworkSubtypeForVpn());
>+  }
>+
>   private String getWifiSSID(NetworkState networkState) {
>     if (getConnectionType(networkState) != ConnectionType.CONNECTION_WIFI)
>       return "";
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/PeerConnection.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/PeerConnection.java
>index 9bc4ebf3096..2f9adcf9621 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/PeerConnection.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/PeerConnection.java
>@@ -13,6 +13,7 @@ package org.webrtc;
> import java.util.ArrayList;
> import java.util.Collections;
> import java.util.List;
>+import javax.annotation.Nullable;
> 
> /**
>  * Java-land version of the PeerConnection APIs; wraps the C++ API
>@@ -22,7 +23,16 @@ import java.util.List;
>  */
> public class PeerConnection {
>   /** Tracks PeerConnectionInterface::IceGatheringState */
>-  public enum IceGatheringState { NEW, GATHERING, COMPLETE }
>+  public enum IceGatheringState {
>+    NEW,
>+    GATHERING,
>+    COMPLETE;
>+
>+    @CalledByNative("IceGatheringState")
>+    static IceGatheringState fromNativeIndex(int nativeIndex) {
>+      return values()[nativeIndex];
>+    }
>+  }
> 
>   /** Tracks PeerConnectionInterface::IceConnectionState */
>   public enum IceConnectionState {
>@@ -32,7 +42,12 @@ public class PeerConnection {
>     COMPLETED,
>     FAILED,
>     DISCONNECTED,
>-    CLOSED
>+    CLOSED;
>+
>+    @CalledByNative("IceConnectionState")
>+    static IceConnectionState fromNativeIndex(int nativeIndex) {
>+      return values()[nativeIndex];
>+    }
>   }
> 
>   /** Tracks PeerConnectionInterface::TlsCertPolicy */
>@@ -48,46 +63,58 @@ public class PeerConnection {
>     HAVE_LOCAL_PRANSWER,
>     HAVE_REMOTE_OFFER,
>     HAVE_REMOTE_PRANSWER,
>-    CLOSED
>+    CLOSED;
>+
>+    @CalledByNative("SignalingState")
>+    static SignalingState fromNativeIndex(int nativeIndex) {
>+      return values()[nativeIndex];
>+    }
>   }
> 
>   /** Java version of PeerConnectionObserver. */
>   public static interface Observer {
>     /** Triggered when the SignalingState changes. */
>-    public void onSignalingChange(SignalingState newState);
>+    @CalledByNative("Observer") void onSignalingChange(SignalingState newState);
> 
>     /** Triggered when the IceConnectionState changes. */
>-    public void onIceConnectionChange(IceConnectionState newState);
>+    @CalledByNative("Observer") void onIceConnectionChange(IceConnectionState newState);
> 
>     /** Triggered when the ICE connection receiving status changes. */
>-    public void onIceConnectionReceivingChange(boolean receiving);
>+    @CalledByNative("Observer") void onIceConnectionReceivingChange(boolean receiving);
> 
>     /** Triggered when the IceGatheringState changes. */
>-    public void onIceGatheringChange(IceGatheringState newState);
>+    @CalledByNative("Observer") void onIceGatheringChange(IceGatheringState newState);
> 
>     /** Triggered when a new ICE candidate has been found. */
>-    public void onIceCandidate(IceCandidate candidate);
>+    @CalledByNative("Observer") void onIceCandidate(IceCandidate candidate);
> 
>     /** Triggered when some ICE candidates have been removed. */
>-    public void onIceCandidatesRemoved(IceCandidate[] candidates);
>+    @CalledByNative("Observer") void onIceCandidatesRemoved(IceCandidate[] candidates);
> 
>     /** Triggered when media is received on a new stream from remote peer. */
>-    public void onAddStream(MediaStream stream);
>+    @CalledByNative("Observer") void onAddStream(MediaStream stream);
> 
>     /** Triggered when a remote peer close a stream. */
>-    public void onRemoveStream(MediaStream stream);
>+    @CalledByNative("Observer") void onRemoveStream(MediaStream stream);
> 
>     /** Triggered when a remote peer opens a DataChannel. */
>-    public void onDataChannel(DataChannel dataChannel);
>+    @CalledByNative("Observer") void onDataChannel(DataChannel dataChannel);
> 
>     /** Triggered when renegotiation is necessary. */
>-    public void onRenegotiationNeeded();
>+    @CalledByNative("Observer") void onRenegotiationNeeded();
> 
>     /**
>      * Triggered when a new track is signaled by the remote peer, as a result of
>      * setRemoteDescription.
>      */
>-    public void onAddTrack(RtpReceiver receiver, MediaStream[] mediaStreams);
>+    @CalledByNative("Observer") void onAddTrack(RtpReceiver receiver, MediaStream[] mediaStreams);
>+
>+    /**
>+     * Triggered when the signaling from SetRemoteDescription indicates that a transceiver
>+     * will be receiving media from a remote endpoint. This is only called if UNIFIED_PLAN
>+     * semantics are specified. The transceiver will be disposed automatically.
>+     */
>+    @CalledByNative("Observer") default void onTrack(RtpTransceiver transceiver){};
>   }
> 
>   /** Java version of PeerConnectionInterface.IceServer. */
>@@ -182,7 +209,7 @@ public class PeerConnection {
>     }
> 
>     public static class Builder {
>-      private final List<String> urls;
>+      @Nullable private final List<String> urls;
>       private String username = "";
>       private String password = "";
>       private TlsCertPolicy tlsCertPolicy = TlsCertPolicy.TLS_CERT_POLICY_SECURE;
>@@ -232,6 +259,45 @@ public class PeerConnection {
>             tlsAlpnProtocols, tlsEllipticCurves);
>       }
>     }
>+
>+    @Nullable
>+    @CalledByNative("IceServer")
>+    List<String> getUrls() {
>+      return urls;
>+    }
>+
>+    @Nullable
>+    @CalledByNative("IceServer")
>+    String getUsername() {
>+      return username;
>+    }
>+
>+    @Nullable
>+    @CalledByNative("IceServer")
>+    String getPassword() {
>+      return password;
>+    }
>+
>+    @CalledByNative("IceServer")
>+    TlsCertPolicy getTlsCertPolicy() {
>+      return tlsCertPolicy;
>+    }
>+
>+    @Nullable
>+    @CalledByNative("IceServer")
>+    String getHostname() {
>+      return hostname;
>+    }
>+
>+    @CalledByNative("IceServer")
>+    List<String> getTlsAlpnProtocols() {
>+      return tlsAlpnProtocols;
>+    }
>+
>+    @CalledByNative("IceServer")
>+    List<String> getTlsEllipticCurves() {
>+      return tlsEllipticCurves;
>+    }
>   }
> 
>   /** Java version of PeerConnectionInterface.IceTransportsType */
>@@ -249,6 +315,16 @@ public class PeerConnection {
>   /** Java version of PeerConnectionInterface.CandidateNetworkPolicy */
>   public enum CandidateNetworkPolicy { ALL, LOW_COST }
> 
>+  // Keep in sync with webrtc/rtc_base/network_constants.h.
>+  public enum AdapterType {
>+    UNKNOWN,
>+    ETHERNET,
>+    WIFI,
>+    CELLULAR,
>+    VPN,
>+    LOOPBACK,
>+  }
>+
>   /** Java version of rtc::KeyType */
>   public enum KeyType { RSA, ECDSA }
> 
>@@ -265,16 +341,46 @@ public class PeerConnection {
>       this.max = max;
>     }
> 
>+    @CalledByNative("IntervalRange")
>     public int getMin() {
>       return min;
>     }
> 
>+    @CalledByNative("IntervalRange")
>     public int getMax() {
>       return max;
>     }
>   }
> 
>+  /**
>+   * Java version of webrtc::SdpSemantics.
>+   *
>+   * Configure the SDP semantics used by this PeerConnection. Note that the
>+   * WebRTC 1.0 specification requires UNIFIED_PLAN semantics. The
>+   * RtpTransceiver API is only available with UNIFIED_PLAN semantics.
>+   *
>+   * <p>PLAN_B will cause PeerConnection to create offers and answers with at
>+   * most one audio and one video m= section with multiple RtpSenders and
>+   * RtpReceivers specified as multiple a=ssrc lines within the section. This
>+   * will also cause PeerConnection to ignore all but the first m= section of
>+   * the same media type.
>+   *
>+   * <p>UNIFIED_PLAN will cause PeerConnection to create offers and answers with
>+   * multiple m= sections where each m= section maps to one RtpSender and one
>+   * RtpReceiver (an RtpTransceiver), either both audio or both video. This
>+   * will also cause PeerConnection to ignore all but the first a=ssrc lines
>+   * that form a Plan B stream.
>+   *
>+   * <p>For users who wish to send multiple audio/video streams and need to stay
>+   * interoperable with legacy WebRTC implementations, specify PLAN_B.
>+   *
>+   * <p>For users who wish to send multiple audio/video streams and/or wish to
>+   * use the new RtpTransceiver API, specify UNIFIED_PLAN.
>+   */
>+  public enum SdpSemantics { PLAN_B, UNIFIED_PLAN }
>+
>   /** Java version of PeerConnectionInterface.RTCConfiguration */
>+  // TODO(qingsi): Resolve the naming inconsistency of fields with/without units.
>   public static class RTCConfiguration {
>     public IceTransportsType iceTransportsType;
>     public List<IceServer> iceServers;
>@@ -291,7 +397,39 @@ public class PeerConnection {
>     public int iceCandidatePoolSize;
>     public boolean pruneTurnPorts;
>     public boolean presumeWritableWhenFullyRelayed;
>-    public Integer iceCheckMinInterval;
>+    // The following fields define intervals in milliseconds at which ICE
>+    // connectivity checks are sent.
>+    //
>+    // We consider ICE is "strongly connected" for an agent when there is at
>+    // least one candidate pair that currently succeeds in connectivity check
>+    // from its direction i.e. sending a ping and receives a ping response, AND
>+    // all candidate pairs have sent a minimum number of pings for connectivity
>+    // (this number is implementation-specific). Otherwise, ICE is considered in
>+    // "weak connectivity".
>+    //
>+    // Note that the above notion of strong and weak connectivity is not defined
>+    // in RFC 5245, and they apply to our current ICE implementation only.
>+    //
>+    // 1) iceCheckIntervalStrongConnectivityMs defines the interval applied to
>+    // ALL candidate pairs when ICE is strongly connected,
>+    // 2) iceCheckIntervalWeakConnectivityMs defines the counterpart for ALL
>+    // pairs when ICE is weakly connected, and
>+    // 3) iceCheckMinInterval defines the minimal interval (equivalently the
>+    // maximum rate) that overrides the above two intervals when either of them
>+    // is less.
>+    @Nullable public Integer iceCheckIntervalStrongConnectivityMs;
>+    @Nullable public Integer iceCheckIntervalWeakConnectivityMs;
>+    @Nullable public Integer iceCheckMinInterval;
>+    // The time period in milliseconds for which a candidate pair must wait for response to
>+    // connectivitiy checks before it becomes unwritable.
>+    @Nullable public Integer iceUnwritableTimeMs;
>+    // The minimum number of connectivity checks that a candidate pair must sent without receiving
>+    // response before it becomes unwritable.
>+    @Nullable public Integer iceUnwritableMinChecks;
>+    // The interval in milliseconds at which STUN candidates will resend STUN binding requests
>+    // to keep NAT bindings open.
>+    // The default value in the implementation is used if this field is null.
>+    @Nullable public Integer stunCandidateKeepaliveIntervalMs;
>     public boolean disableIPv6OnWifi;
>     // By default, PeerConnection will use a limited number of IPv6 network
>     // interfaces, in order to avoid too many ICE candidate pairs being created
>@@ -299,10 +437,29 @@ public class PeerConnection {
>     //
>     // Can be set to Integer.MAX_VALUE to effectively disable the limit.
>     public int maxIPv6Networks;
>-    public IntervalRange iceRegatherIntervalRange;
>+    @Nullable public IntervalRange iceRegatherIntervalRange;
>+
>+    // These values will be overridden by MediaStream constraints if deprecated constraints-based
>+    // create peerconnection interface is used.
>+    public boolean disableIpv6;
>+    public boolean enableDscp;
>+    public boolean enableCpuOveruseDetection;
>+    public boolean enableRtpDataChannel;
>+    public boolean suspendBelowMinBitrate;
>+    @Nullable public Integer screencastMinBitrate;
>+    @Nullable public Boolean combinedAudioVideoBwe;
>+    @Nullable public Boolean enableDtlsSrtp;
>+    // Use "Unknown" to represent no preference of adapter types, not the
>+    // preference of adapters of unknown types.
>+    public AdapterType networkPreference;
>+    public SdpSemantics sdpSemantics;
> 
>     // This is an optional wrapper for the C++ webrtc::TurnCustomizer.
>-    public TurnCustomizer turnCustomizer;
>+    @Nullable public TurnCustomizer turnCustomizer;
>+
>+    // Actively reset the SRTP parameters whenever the DTLS transports underneath are reset for
>+    // every offer/answer negotiation.This is only intended to be a workaround for crbug.com/835958
>+    public boolean activeResetSrtpParams;
> 
>     // TODO(deadbeef): Instead of duplicating the defaults here, we should do
>     // something to pick up the defaults from C++. The Objective-C equivalent
>@@ -323,53 +480,291 @@ public class PeerConnection {
>       iceCandidatePoolSize = 0;
>       pruneTurnPorts = false;
>       presumeWritableWhenFullyRelayed = false;
>+      iceCheckIntervalStrongConnectivityMs = null;
>+      iceCheckIntervalWeakConnectivityMs = null;
>       iceCheckMinInterval = null;
>+      iceUnwritableTimeMs = null;
>+      iceUnwritableMinChecks = null;
>+      stunCandidateKeepaliveIntervalMs = null;
>       disableIPv6OnWifi = false;
>       maxIPv6Networks = 5;
>       iceRegatherIntervalRange = null;
>+      disableIpv6 = false;
>+      enableDscp = false;
>+      enableCpuOveruseDetection = true;
>+      enableRtpDataChannel = false;
>+      suspendBelowMinBitrate = false;
>+      screencastMinBitrate = null;
>+      combinedAudioVideoBwe = null;
>+      enableDtlsSrtp = null;
>+      networkPreference = AdapterType.UNKNOWN;
>+      sdpSemantics = SdpSemantics.PLAN_B;
>+      activeResetSrtpParams = false;
>+    }
>+
>+    @CalledByNative("RTCConfiguration")
>+    IceTransportsType getIceTransportsType() {
>+      return iceTransportsType;
>+    }
>+
>+    @CalledByNative("RTCConfiguration")
>+    List<IceServer> getIceServers() {
>+      return iceServers;
>+    }
>+
>+    @CalledByNative("RTCConfiguration")
>+    BundlePolicy getBundlePolicy() {
>+      return bundlePolicy;
>+    }
>+
>+    @CalledByNative("RTCConfiguration")
>+    RtcpMuxPolicy getRtcpMuxPolicy() {
>+      return rtcpMuxPolicy;
>+    }
>+
>+    @CalledByNative("RTCConfiguration")
>+    TcpCandidatePolicy getTcpCandidatePolicy() {
>+      return tcpCandidatePolicy;
>+    }
>+
>+    @CalledByNative("RTCConfiguration")
>+    CandidateNetworkPolicy getCandidateNetworkPolicy() {
>+      return candidateNetworkPolicy;
>+    }
>+
>+    @CalledByNative("RTCConfiguration")
>+    int getAudioJitterBufferMaxPackets() {
>+      return audioJitterBufferMaxPackets;
>+    }
>+
>+    @CalledByNative("RTCConfiguration")
>+    boolean getAudioJitterBufferFastAccelerate() {
>+      return audioJitterBufferFastAccelerate;
>+    }
>+
>+    @CalledByNative("RTCConfiguration")
>+    int getIceConnectionReceivingTimeout() {
>+      return iceConnectionReceivingTimeout;
>+    }
>+
>+    @CalledByNative("RTCConfiguration")
>+    int getIceBackupCandidatePairPingInterval() {
>+      return iceBackupCandidatePairPingInterval;
>+    }
>+
>+    @CalledByNative("RTCConfiguration")
>+    KeyType getKeyType() {
>+      return keyType;
>+    }
>+
>+    @CalledByNative("RTCConfiguration")
>+    ContinualGatheringPolicy getContinualGatheringPolicy() {
>+      return continualGatheringPolicy;
>+    }
>+
>+    @CalledByNative("RTCConfiguration")
>+    int getIceCandidatePoolSize() {
>+      return iceCandidatePoolSize;
>+    }
>+
>+    @CalledByNative("RTCConfiguration")
>+    boolean getPruneTurnPorts() {
>+      return pruneTurnPorts;
>+    }
>+
>+    @CalledByNative("RTCConfiguration")
>+    boolean getPresumeWritableWhenFullyRelayed() {
>+      return presumeWritableWhenFullyRelayed;
>+    }
>+
>+    @Nullable
>+    @CalledByNative("RTCConfiguration")
>+    Integer getIceCheckIntervalStrongConnectivity() {
>+      return iceCheckIntervalStrongConnectivityMs;
>+    }
>+
>+    @Nullable
>+    @CalledByNative("RTCConfiguration")
>+    Integer getIceCheckIntervalWeakConnectivity() {
>+      return iceCheckIntervalWeakConnectivityMs;
>+    }
>+
>+    @Nullable
>+    @CalledByNative("RTCConfiguration")
>+    Integer getIceCheckMinInterval() {
>+      return iceCheckMinInterval;
>+    }
>+
>+    @Nullable
>+    @CalledByNative("RTCConfiguration")
>+    Integer getIceUnwritableTimeout() {
>+      return iceUnwritableTimeMs;
>+    }
>+
>+    @Nullable
>+    @CalledByNative("RTCConfiguration")
>+    Integer getIceUnwritableMinChecks() {
>+      return iceUnwritableMinChecks;
>+    }
>+
>+    @Nullable
>+    @CalledByNative("RTCConfiguration")
>+    Integer getStunCandidateKeepaliveInterval() {
>+      return stunCandidateKeepaliveIntervalMs;
>+    }
>+
>+    @CalledByNative("RTCConfiguration")
>+    boolean getDisableIPv6OnWifi() {
>+      return disableIPv6OnWifi;
>+    }
>+
>+    @CalledByNative("RTCConfiguration")
>+    int getMaxIPv6Networks() {
>+      return maxIPv6Networks;
>+    }
>+
>+    @Nullable
>+    @CalledByNative("RTCConfiguration")
>+    IntervalRange getIceRegatherIntervalRange() {
>+      return iceRegatherIntervalRange;
>+    }
>+
>+    @Nullable
>+    @CalledByNative("RTCConfiguration")
>+    TurnCustomizer getTurnCustomizer() {
>+      return turnCustomizer;
>+    }
>+
>+    @CalledByNative("RTCConfiguration")
>+    boolean getDisableIpv6() {
>+      return disableIpv6;
>+    }
>+
>+    @CalledByNative("RTCConfiguration")
>+    boolean getEnableDscp() {
>+      return enableDscp;
>+    }
>+
>+    @CalledByNative("RTCConfiguration")
>+    boolean getEnableCpuOveruseDetection() {
>+      return enableCpuOveruseDetection;
>+    }
>+
>+    @CalledByNative("RTCConfiguration")
>+    boolean getEnableRtpDataChannel() {
>+      return enableRtpDataChannel;
>+    }
>+
>+    @CalledByNative("RTCConfiguration")
>+    boolean getSuspendBelowMinBitrate() {
>+      return suspendBelowMinBitrate;
>+    }
>+
>+    @Nullable
>+    @CalledByNative("RTCConfiguration")
>+    Integer getScreencastMinBitrate() {
>+      return screencastMinBitrate;
>+    }
>+
>+    @Nullable
>+    @CalledByNative("RTCConfiguration")
>+    Boolean getCombinedAudioVideoBwe() {
>+      return combinedAudioVideoBwe;
>+    }
>+
>+    @Nullable
>+    @CalledByNative("RTCConfiguration")
>+    Boolean getEnableDtlsSrtp() {
>+      return enableDtlsSrtp;
>+    }
>+
>+    @CalledByNative("RTCConfiguration")
>+    AdapterType getNetworkPreference() {
>+      return networkPreference;
>+    }
>+
>+    @CalledByNative("RTCConfiguration")
>+    SdpSemantics getSdpSemantics() {
>+      return sdpSemantics;
>+    }
>+
>+    @CalledByNative("RTCConfiguration")
>+    boolean getActiveResetSrtpParams() {
>+      return activeResetSrtpParams;
>     }
>   };
> 
>   private final List<MediaStream> localStreams = new ArrayList<>();
>   private final long nativePeerConnection;
>-  private final long nativeObserver;
>   private List<RtpSender> senders = new ArrayList<>();
>   private List<RtpReceiver> receivers = new ArrayList<>();
>+  private List<RtpTransceiver> transceivers = new ArrayList<>();
>+
>+  /**
>+   * Wraps a PeerConnection created by the factory. Can be used by clients that want to implement
>+   * their PeerConnection creation in JNI.
>+   */
>+  public PeerConnection(NativePeerConnectionFactory factory) {
>+    this(factory.createNativePeerConnection());
>+  }
> 
>-  PeerConnection(long nativePeerConnection, long nativeObserver) {
>+  PeerConnection(long nativePeerConnection) {
>     this.nativePeerConnection = nativePeerConnection;
>-    this.nativeObserver = nativeObserver;
>   }
> 
>   // JsepInterface.
>-  public native SessionDescription getLocalDescription();
>+  public SessionDescription getLocalDescription() {
>+    return nativeGetLocalDescription();
>+  }
> 
>-  public native SessionDescription getRemoteDescription();
>+  public SessionDescription getRemoteDescription() {
>+    return nativeGetRemoteDescription();
>+  }
> 
>-  public native DataChannel createDataChannel(String label, DataChannel.Init init);
>+  public DataChannel createDataChannel(String label, DataChannel.Init init) {
>+    return nativeCreateDataChannel(label, init);
>+  }
> 
>-  public native void createOffer(SdpObserver observer, MediaConstraints constraints);
>+  public void createOffer(SdpObserver observer, MediaConstraints constraints) {
>+    nativeCreateOffer(observer, constraints);
>+  }
> 
>-  public native void createAnswer(SdpObserver observer, MediaConstraints constraints);
>+  public void createAnswer(SdpObserver observer, MediaConstraints constraints) {
>+    nativeCreateAnswer(observer, constraints);
>+  }
> 
>-  public native void setLocalDescription(SdpObserver observer, SessionDescription sdp);
>+  public void setLocalDescription(SdpObserver observer, SessionDescription sdp) {
>+    nativeSetLocalDescription(observer, sdp);
>+  }
> 
>-  public native void setRemoteDescription(SdpObserver observer, SessionDescription sdp);
>+  public void setRemoteDescription(SdpObserver observer, SessionDescription sdp) {
>+    nativeSetRemoteDescription(observer, sdp);
>+  }
> 
>-  // True if remote audio should be played out. Defaults to true.
>-  // Note that even if playout is enabled, streams will only be played out if
>-  // the appropriate SDP is also applied. The main purpose of this API is to
>-  // be able to control the exact time when audio playout starts.
>-  public native void setAudioPlayout(boolean playout);
>+  /**
>+   * Enables/disables playout of received audio streams. Enabled by default.
>+   *
>+   * Note that even if playout is enabled, streams will only be played out if
>+   * the appropriate SDP is also applied. The main purpose of this API is to
>+   * be able to control the exact time when audio playout starts.
>+   */
>+  public void setAudioPlayout(boolean playout) {
>+    nativeSetAudioPlayout(playout);
>+  }
> 
>-  // True if local audio shall be recorded. Defaults to true.
>-  // Note that even if recording is enabled, streams will only be recorded if
>-  // the appropriate SDP is also applied. The main purpose of this API is to
>-  // be able to control the exact time when audio recording starts.
>-  public native void setAudioRecording(boolean recording);
>+  /**
>+   * Enables/disables recording of transmitted audio streams. Enabled by default.
>+   *
>+   * Note that even if recording is enabled, streams will only be recorded if
>+   * the appropriate SDP is also applied. The main purpose of this API is to
>+   * be able to control the exact time when audio recording starts.
>+   */
>+  public void setAudioRecording(boolean recording) {
>+    nativeSetAudioRecording(recording);
>+  }
> 
>   public boolean setConfiguration(RTCConfiguration config) {
>-    return nativeSetConfiguration(config, nativeObserver);
>+    return nativeSetConfiguration(config);
>   }
> 
>   public boolean addIceCandidate(IceCandidate candidate) {
>@@ -380,6 +775,11 @@ public class PeerConnection {
>     return nativeRemoveIceCandidates(candidates);
>   }
> 
>+  /**
>+   * Adds a new MediaStream to be sent on this peer connection.
>+   * Note: This method is not supported with SdpSemantics.UNIFIED_PLAN. Please
>+   * use addTrack instead.
>+   */
>   public boolean addStream(MediaStream stream) {
>     boolean ret = nativeAddLocalStream(stream.nativeStream);
>     if (!ret) {
>@@ -389,6 +789,11 @@ public class PeerConnection {
>     return true;
>   }
> 
>+  /**
>+   * Removes the given media stream from this peer connection.
>+   * This method is not supported with SdpSemantics.UNIFIED_PLAN. Please use
>+   * removeTrack instead.
>+   */
>   public void removeStream(MediaStream stream) {
>     nativeRemoveLocalStream(stream.nativeStream);
>     localStreams.remove(stream);
>@@ -396,15 +801,15 @@ public class PeerConnection {
> 
>   /**
>    * Creates an RtpSender without a track.
>-   * <p>
>-   * This method allows an application to cause the PeerConnection to negotiate
>+   *
>+   * <p>This method allows an application to cause the PeerConnection to negotiate
>    * sending/receiving a specific media type, but without having a track to
>    * send yet.
>-   * <p>
>-   * When the application does want to begin sending a track, it can call
>+   *
>+   * <p>When the application does want to begin sending a track, it can call
>    * RtpSender.setTrack, which doesn't require any additional SDP negotiation.
>-   * <p>
>-   * Example use:
>+   *
>+   * <p>Example use:
>    * <pre>
>    * {@code
>    * audioSender = pc.createSender("audio", "stream1");
>@@ -416,12 +821,15 @@ public class PeerConnection {
>    * videoSender.setTrack(videoTrack, false);
>    * }
>    * </pre>
>-   * Note: This corresponds most closely to "addTransceiver" in the official
>+   * <p>Note: This corresponds most closely to "addTransceiver" in the official
>    * WebRTC API, in that it creates a sender without a track. It was
>    * implemented before addTransceiver because it provides useful
>    * functionality, and properly implementing transceivers would have required
>    * a great deal more work.
>    *
>+   * <p>Note: This is only available with SdpSemantics.PLAN_B specified. Please use
>+   * addTransceiver instead.
>+   *
>    * @param kind      Corresponds to MediaStreamTrack kinds (must be "audio" or
>    *                  "video").
>    * @param stream_id The ID of the MediaStream that this sender's track will
>@@ -432,15 +840,18 @@ public class PeerConnection {
>    * @return          A new RtpSender object if successful, or null otherwise.
>    */
>   public RtpSender createSender(String kind, String stream_id) {
>-    RtpSender new_sender = nativeCreateSender(kind, stream_id);
>-    if (new_sender != null) {
>-      senders.add(new_sender);
>+    RtpSender newSender = nativeCreateSender(kind, stream_id);
>+    if (newSender != null) {
>+      senders.add(newSender);
>     }
>-    return new_sender;
>+    return newSender;
>   }
> 
>-  // Note that calling getSenders will dispose of the senders previously
>-  // returned (and same goes for getReceivers).
>+  /**
>+   * Gets all RtpSenders associated with this peer connection.
>+   * Note that calling getSenders will dispose of the senders previously
>+   * returned.
>+   */
>   public List<RtpSender> getSenders() {
>     for (RtpSender sender : senders) {
>       sender.dispose();
>@@ -449,6 +860,11 @@ public class PeerConnection {
>     return Collections.unmodifiableList(senders);
>   }
> 
>+  /**
>+   * Gets all RtpReceivers associated with this peer connection.
>+   * Note that calling getReceivers will dispose of the receivers previously
>+   * returned.
>+   */
>   public List<RtpReceiver> getReceivers() {
>     for (RtpReceiver receiver : receivers) {
>       receiver.dispose();
>@@ -457,57 +873,194 @@ public class PeerConnection {
>     return Collections.unmodifiableList(receivers);
>   }
> 
>+  /**
>+   * Gets all RtpTransceivers associated with this peer connection.
>+   * Note that calling getTransceivers will dispose of the transceivers previously
>+   * returned.
>+   * Note: This is only available with SdpSemantics.UNIFIED_PLAN specified.
>+   */
>+  public List<RtpTransceiver> getTransceivers() {
>+    for (RtpTransceiver transceiver : transceivers) {
>+      transceiver.dispose();
>+    }
>+    transceivers = nativeGetTransceivers();
>+    return Collections.unmodifiableList(transceivers);
>+  }
>+
>+  /**
>+   * Adds a new media stream track to be sent on this peer connection, and returns
>+   * the newly created RtpSender. If streamIds are specified, the RtpSender will
>+   * be associated with the streams specified in the streamIds list.
>+   *
>+   * @throws IllegalStateException if an error accors in C++ addTrack.
>+   *         An error can occur if:
>+   *           - A sender already exists for the track.
>+   *           - The peer connection is closed.
>+   */
>+  public RtpSender addTrack(MediaStreamTrack track) {
>+    return addTrack(track, Collections.emptyList());
>+  }
>+
>+  public RtpSender addTrack(MediaStreamTrack track, List<String> streamIds) {
>+    if (track == null || streamIds == null) {
>+      throw new NullPointerException("No MediaStreamTrack specified in addTrack.");
>+    }
>+    RtpSender newSender = nativeAddTrack(track.nativeTrack, streamIds);
>+    if (newSender == null) {
>+      throw new IllegalStateException("C++ addTrack failed.");
>+    }
>+    senders.add(newSender);
>+    return newSender;
>+  }
>+
>+  /**
>+   * Stops sending media from sender. The sender will still appear in getSenders. Future
>+   * calls to createOffer will mark the m section for the corresponding transceiver as
>+   * receive only or inactive, as defined in JSEP. Returns true on success.
>+   */
>+  public boolean removeTrack(RtpSender sender) {
>+    if (sender == null) {
>+      throw new NullPointerException("No RtpSender specified for removeTrack.");
>+    }
>+    return nativeRemoveTrack(sender.nativeRtpSender);
>+  }
>+
>+  /**
>+   * Creates a new RtpTransceiver and adds it to the set of transceivers. Adding a
>+   * transceiver will cause future calls to CreateOffer to add a media description
>+   * for the corresponding transceiver.
>+   *
>+   * <p>The initial value of |mid| in the returned transceiver is null. Setting a
>+   * new session description may change it to a non-null value.
>+   *
>+   * <p>https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnection-addtransceiver
>+   *
>+   * <p>If a MediaStreamTrack is specified then a transceiver will be added with a
>+   * sender set to transmit the given track. The kind
>+   * of the transceiver (and sender/receiver) will be derived from the kind of
>+   * the track.
>+   *
>+   * <p>If MediaType is specified then a transceiver will be added based upon that type.
>+   * This can be either MEDIA_TYPE_AUDIO or MEDIA_TYPE_VIDEO.
>+   *
>+   * <p>Optionally, an RtpTransceiverInit structure can be specified to configure
>+   * the transceiver from construction. If not specified, the transceiver will
>+   * default to having a direction of kSendRecv and not be part of any streams.
>+   *
>+   * <p>Note: These methods are only available with SdpSemantics.UNIFIED_PLAN specified.
>+   * @throws IllegalStateException if an error accors in C++ addTransceiver
>+   */
>+  public RtpTransceiver addTransceiver(MediaStreamTrack track) {
>+    return addTransceiver(track, new RtpTransceiver.RtpTransceiverInit());
>+  }
>+
>+  public RtpTransceiver addTransceiver(
>+      MediaStreamTrack track, @Nullable RtpTransceiver.RtpTransceiverInit init) {
>+    if (track == null) {
>+      throw new NullPointerException("No MediaStreamTrack specified for addTransceiver.");
>+    }
>+    if (init == null) {
>+      init = new RtpTransceiver.RtpTransceiverInit();
>+    }
>+    RtpTransceiver newTransceiver = nativeAddTransceiverWithTrack(track.nativeTrack, init);
>+    if (newTransceiver == null) {
>+      throw new IllegalStateException("C++ addTransceiver failed.");
>+    }
>+    transceivers.add(newTransceiver);
>+    return newTransceiver;
>+  }
>+
>+  public RtpTransceiver addTransceiver(MediaStreamTrack.MediaType mediaType) {
>+    return addTransceiver(mediaType, new RtpTransceiver.RtpTransceiverInit());
>+  }
>+
>+  public RtpTransceiver addTransceiver(
>+      MediaStreamTrack.MediaType mediaType, @Nullable RtpTransceiver.RtpTransceiverInit init) {
>+    if (mediaType == null) {
>+      throw new NullPointerException("No MediaType specified for addTransceiver.");
>+    }
>+    if (init == null) {
>+      init = new RtpTransceiver.RtpTransceiverInit();
>+    }
>+    RtpTransceiver newTransceiver = nativeAddTransceiverOfType(mediaType, init);
>+    if (newTransceiver == null) {
>+      throw new IllegalStateException("C++ addTransceiver failed.");
>+    }
>+    transceivers.add(newTransceiver);
>+    return newTransceiver;
>+  }
>+
>   // Older, non-standard implementation of getStats.
>   @Deprecated
>-  public boolean getStats(StatsObserver observer, MediaStreamTrack track) {
>+  public boolean getStats(StatsObserver observer, @Nullable MediaStreamTrack track) {
>     return nativeOldGetStats(observer, (track == null) ? 0 : track.nativeTrack);
>   }
> 
>-  // Gets stats using the new stats collection API, see webrtc/api/stats/. These
>-  // will replace old stats collection API when the new API has matured enough.
>+  /**
>+   * Gets stats using the new stats collection API, see webrtc/api/stats/. These
>+   * will replace old stats collection API when the new API has matured enough.
>+   */
>   public void getStats(RTCStatsCollectorCallback callback) {
>     nativeNewGetStats(callback);
>   }
> 
>-  // Limits the bandwidth allocated for all RTP streams sent by this
>-  // PeerConnection. Pass null to leave a value unchanged.
>-  public native boolean setBitrate(Integer min, Integer current, Integer max);
>+  /**
>+   * Limits the bandwidth allocated for all RTP streams sent by this
>+   * PeerConnection. Pass null to leave a value unchanged.
>+   */
>+  public boolean setBitrate(Integer min, Integer current, Integer max) {
>+    return nativeSetBitrate(min, current, max);
>+  }
> 
>-  // Starts recording an RTC event log. Ownership of the file is transfered to
>-  // the native code. If an RTC event log is already being recorded, it will be
>-  // stopped and a new one will start using the provided file. Logging will
>-  // continue until the stopRtcEventLog function is called. The max_size_bytes
>-  // argument is ignored, it is added for future use.
>+  /**
>+   * Starts recording an RTC event log.
>+   *
>+   * Ownership of the file is transfered to the native code. If an RTC event
>+   * log is already being recorded, it will be stopped and a new one will start
>+   * using the provided file. Logging will continue until the stopRtcEventLog
>+   * function is called. The max_size_bytes argument is ignored, it is added
>+   * for future use.
>+   */
>   public boolean startRtcEventLog(int file_descriptor, int max_size_bytes) {
>     return nativeStartRtcEventLog(file_descriptor, max_size_bytes);
>   }
> 
>-  // Stops recording an RTC event log. If no RTC event log is currently being
>-  // recorded, this call will have no effect.
>+  /**
>+   * Stops recording an RTC event log. If no RTC event log is currently being
>+   * recorded, this call will have no effect.
>+   */
>   public void stopRtcEventLog() {
>     nativeStopRtcEventLog();
>   }
> 
>   // TODO(fischman): add support for DTMF-related methods once that API
>   // stabilizes.
>-  public native SignalingState signalingState();
>+  public SignalingState signalingState() {
>+    return nativeSignalingState();
>+  }
> 
>-  public native IceConnectionState iceConnectionState();
>+  public IceConnectionState iceConnectionState() {
>+    return nativeIceConnectionState();
>+  }
> 
>-  public native IceGatheringState iceGatheringState();
>+  public IceGatheringState iceGatheringState() {
>+    return nativeIceGatheringState();
>+  }
> 
>-  public native void close();
>+  public void close() {
>+    nativeClose();
>+  }
> 
>   /**
>    * Free native resources associated with this PeerConnection instance.
>-   * <p>
>+   *
>    * This method removes a reference count from the C++ PeerConnection object,
>    * which should result in it being destroyed. It also calls equivalent
>    * "dispose" methods on the Java objects attached to this PeerConnection
>    * (streams, senders, receivers), such that their associated C++ objects
>    * will also be destroyed.
>-   * <p>
>-   * Note that this method cannot be safely called from an observer callback
>+   *
>+   * <p>Note that this method cannot be safely called from an observer callback
>    * (PeerConnection.Observer, DataChannel.Observer, etc.). If you want to, for
>    * example, destroy the PeerConnection after an "ICE failed" callback, you
>    * must do this asynchronously (in other words, unwind the stack first). See
>@@ -528,35 +1081,63 @@ public class PeerConnection {
>     for (RtpReceiver receiver : receivers) {
>       receiver.dispose();
>     }
>+    for (RtpTransceiver transceiver : transceivers) {
>+      transceiver.dispose();
>+    }
>+    transceivers.clear();
>     receivers.clear();
>-    JniCommon.nativeReleaseRef(nativePeerConnection);
>-    freeObserver(nativeObserver);
>+    nativeFreeOwnedPeerConnection(nativePeerConnection);
>+  }
>+
>+  /** Returns a pointer to the native webrtc::PeerConnectionInterface. */
>+  public long getNativePeerConnection() {
>+    return nativeGetNativePeerConnection();
>   }
> 
>-  private static native void freeObserver(long nativeObserver);
>+  @CalledByNative
>+  long getNativeOwnedPeerConnection() {
>+    return nativePeerConnection;
>+  }
> 
>-  public native boolean nativeSetConfiguration(RTCConfiguration config, long nativeObserver);
>+  public static long createNativePeerConnectionObserver(Observer observer) {
>+    return nativeCreatePeerConnectionObserver(observer);
>+  }
> 
>+  private native long nativeGetNativePeerConnection();
>+  private native SessionDescription nativeGetLocalDescription();
>+  private native SessionDescription nativeGetRemoteDescription();
>+  private native DataChannel nativeCreateDataChannel(String label, DataChannel.Init init);
>+  private native void nativeCreateOffer(SdpObserver observer, MediaConstraints constraints);
>+  private native void nativeCreateAnswer(SdpObserver observer, MediaConstraints constraints);
>+  private native void nativeSetLocalDescription(SdpObserver observer, SessionDescription sdp);
>+  private native void nativeSetRemoteDescription(SdpObserver observer, SessionDescription sdp);
>+  private native void nativeSetAudioPlayout(boolean playout);
>+  private native void nativeSetAudioRecording(boolean recording);
>+  private native boolean nativeSetBitrate(Integer min, Integer current, Integer max);
>+  private native SignalingState nativeSignalingState();
>+  private native IceConnectionState nativeIceConnectionState();
>+  private native IceGatheringState nativeIceGatheringState();
>+  private native void nativeClose();
>+  private static native long nativeCreatePeerConnectionObserver(Observer observer);
>+  private static native void nativeFreeOwnedPeerConnection(long ownedPeerConnection);
>+  private native boolean nativeSetConfiguration(RTCConfiguration config);
>   private native boolean nativeAddIceCandidate(
>       String sdpMid, int sdpMLineIndex, String iceCandidateSdp);
>-
>   private native boolean nativeRemoveIceCandidates(final IceCandidate[] candidates);
>-
>-  private native boolean nativeAddLocalStream(long nativeStream);
>-
>-  private native void nativeRemoveLocalStream(long nativeStream);
>-
>+  private native boolean nativeAddLocalStream(long stream);
>+  private native void nativeRemoveLocalStream(long stream);
>   private native boolean nativeOldGetStats(StatsObserver observer, long nativeTrack);
>-
>   private native void nativeNewGetStats(RTCStatsCollectorCallback callback);
>-
>   private native RtpSender nativeCreateSender(String kind, String stream_id);
>-
>   private native List<RtpSender> nativeGetSenders();
>-
>   private native List<RtpReceiver> nativeGetReceivers();
>-
>+  private native List<RtpTransceiver> nativeGetTransceivers();
>+  private native RtpSender nativeAddTrack(long track, List<String> streamIds);
>+  private native boolean nativeRemoveTrack(long sender);
>+  private native RtpTransceiver nativeAddTransceiverWithTrack(
>+      long track, RtpTransceiver.RtpTransceiverInit init);
>+  private native RtpTransceiver nativeAddTransceiverOfType(
>+      MediaStreamTrack.MediaType mediaType, RtpTransceiver.RtpTransceiverInit init);
>   private native boolean nativeStartRtcEventLog(int file_descriptor, int max_size_bytes);
>-
>   private native void nativeStopRtcEventLog();
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/PeerConnectionDependencies.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/PeerConnectionDependencies.java
>new file mode 100644
>index 00000000000..fd765b17d17
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/PeerConnectionDependencies.java
>@@ -0,0 +1,65 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc;
>+
>+import javax.annotation.Nullable;
>+
>+/**
>+ * PeerConnectionDependencies holds all PeerConnection dependencies that are
>+ * applied per PeerConnection. A dependency is distinct from a configuration
>+ * as it defines significant executable code that can be provided by a user of
>+ * the API.
>+ */
>+public final class PeerConnectionDependencies {
>+  // Mandatory dependencies.
>+  private final PeerConnection.Observer observer;
>+
>+  // Optional fields.
>+  private final SSLCertificateVerifier sslCertificateVerifier;
>+
>+  public static class Builder {
>+    private PeerConnection.Observer observer;
>+    private SSLCertificateVerifier sslCertificateVerifier;
>+
>+    private Builder(PeerConnection.Observer observer) {
>+      this.observer = observer;
>+    }
>+
>+    public Builder setSSLCertificateVerifier(SSLCertificateVerifier sslCertificateVerifier) {
>+      this.sslCertificateVerifier = sslCertificateVerifier;
>+      return this;
>+    }
>+
>+    // Observer is a required dependency and so is forced in the construction of the object.
>+    public PeerConnectionDependencies createPeerConnectionDependencies() {
>+      return new PeerConnectionDependencies(observer, sslCertificateVerifier);
>+    }
>+  }
>+
>+  public static Builder builder(PeerConnection.Observer observer) {
>+    return new Builder(observer);
>+  }
>+
>+  PeerConnection.Observer getObserver() {
>+    return observer;
>+  }
>+
>+  @Nullable
>+  SSLCertificateVerifier getSSLCertificateVerifier() {
>+    return sslCertificateVerifier;
>+  }
>+
>+  private PeerConnectionDependencies(
>+      PeerConnection.Observer observer, SSLCertificateVerifier sslCertificateVerifier) {
>+    this.observer = observer;
>+    this.sslCertificateVerifier = sslCertificateVerifier;
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/PeerConnectionFactory.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/PeerConnectionFactory.java
>index cc1d83f003a..acf3a713823 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/PeerConnectionFactory.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/PeerConnectionFactory.java
>@@ -12,6 +12,10 @@ package org.webrtc;
> 
> import android.content.Context;
> import java.util.List;
>+import javax.annotation.Nullable;
>+import org.webrtc.Logging.Severity;
>+import org.webrtc.audio.AudioDeviceModule;
>+import org.webrtc.audio.LegacyAudioDeviceModule;
> 
> /**
>  * Java wrapper for a C++ PeerConnectionFactoryInterface.  Main entry point to
>@@ -19,19 +23,17 @@ import java.util.List;
>  */
> public class PeerConnectionFactory {
>   public static final String TRIAL_ENABLED = "Enabled";
>-  public static final String VIDEO_FRAME_EMIT_TRIAL = "VideoFrameEmit";
>+  @Deprecated public static final String VIDEO_FRAME_EMIT_TRIAL = "VideoFrameEmit";
> 
>   private static final String TAG = "PeerConnectionFactory";
>   private static final String VIDEO_CAPTURER_THREAD_NAME = "VideoCapturerThread";
> 
>   private final long nativeFactory;
>+  private static boolean enableVideoHwAcceleration;
>   private static volatile boolean internalTracerInitialized = false;
>-  private static Context applicationContext;
>-  private static Thread networkThread;
>-  private static Thread workerThread;
>-  private static Thread signalingThread;
>-  private EglBase localEglbase;
>-  private EglBase remoteEglbase;
>+  @Nullable private static Thread networkThread;
>+  @Nullable private static Thread workerThread;
>+  @Nullable private static Thread signalingThread;
> 
>   public static class InitializationOptions {
>     final Context applicationContext;
>@@ -39,15 +41,22 @@ public class PeerConnectionFactory {
>     final boolean enableInternalTracer;
>     final boolean enableVideoHwAcceleration;
>     final NativeLibraryLoader nativeLibraryLoader;
>+    final String nativeLibraryName;
>+    @Nullable Loggable loggable;
>+    @Nullable Severity loggableSeverity;
> 
>     private InitializationOptions(Context applicationContext, String fieldTrials,
>         boolean enableInternalTracer, boolean enableVideoHwAcceleration,
>-        NativeLibraryLoader nativeLibraryLoader) {
>+        NativeLibraryLoader nativeLibraryLoader, String nativeLibraryName,
>+        @Nullable Loggable loggable, @Nullable Severity loggableSeverity) {
>       this.applicationContext = applicationContext;
>       this.fieldTrials = fieldTrials;
>       this.enableInternalTracer = enableInternalTracer;
>       this.enableVideoHwAcceleration = enableVideoHwAcceleration;
>       this.nativeLibraryLoader = nativeLibraryLoader;
>+      this.nativeLibraryName = nativeLibraryName;
>+      this.loggable = loggable;
>+      this.loggableSeverity = loggableSeverity;
>     }
> 
>     public static Builder builder(Context applicationContext) {
>@@ -60,6 +69,9 @@ public class PeerConnectionFactory {
>       private boolean enableInternalTracer = false;
>       private boolean enableVideoHwAcceleration = true;
>       private NativeLibraryLoader nativeLibraryLoader = new NativeLibrary.DefaultLoader();
>+      private String nativeLibraryName = "jingle_peerconnection_so";
>+      @Nullable private Loggable loggable = null;
>+      @Nullable private Severity loggableSeverity = null;
> 
>       Builder(Context applicationContext) {
>         this.applicationContext = applicationContext;
>@@ -75,6 +87,8 @@ public class PeerConnectionFactory {
>         return this;
>       }
> 
>+      // Deprecated, this method only affects the deprecated HW codecs and not the new ones.
>+      @Deprecated
>       public Builder setEnableVideoHwAcceleration(boolean enableVideoHwAcceleration) {
>         this.enableVideoHwAcceleration = enableVideoHwAcceleration;
>         return this;
>@@ -85,25 +99,138 @@ public class PeerConnectionFactory {
>         return this;
>       }
> 
>+      public Builder setNativeLibraryName(String nativeLibraryName) {
>+        this.nativeLibraryName = nativeLibraryName;
>+        return this;
>+      }
>+
>+      public Builder setInjectableLogger(Loggable loggable, Severity severity) {
>+        this.loggable = loggable;
>+        this.loggableSeverity = severity;
>+        return this;
>+      }
>+
>       public PeerConnectionFactory.InitializationOptions createInitializationOptions() {
>         return new PeerConnectionFactory.InitializationOptions(applicationContext, fieldTrials,
>-            enableInternalTracer, enableVideoHwAcceleration, nativeLibraryLoader);
>+            enableInternalTracer, enableVideoHwAcceleration, nativeLibraryLoader, nativeLibraryName,
>+            loggable, loggableSeverity);
>       }
>     }
>   }
> 
>   public static class Options {
>     // Keep in sync with webrtc/rtc_base/network.h!
>+    //
>+    // These bit fields are defined for |networkIgnoreMask| below.
>     static final int ADAPTER_TYPE_UNKNOWN = 0;
>     static final int ADAPTER_TYPE_ETHERNET = 1 << 0;
>     static final int ADAPTER_TYPE_WIFI = 1 << 1;
>     static final int ADAPTER_TYPE_CELLULAR = 1 << 2;
>     static final int ADAPTER_TYPE_VPN = 1 << 3;
>     static final int ADAPTER_TYPE_LOOPBACK = 1 << 4;
>+    static final int ADAPTER_TYPE_ANY = 1 << 5;
> 
>     public int networkIgnoreMask;
>     public boolean disableEncryption;
>     public boolean disableNetworkMonitor;
>+    public boolean enableAes128Sha1_32CryptoCipher;
>+    public boolean enableGcmCryptoSuites;
>+
>+    @CalledByNative("Options")
>+    int getNetworkIgnoreMask() {
>+      return networkIgnoreMask;
>+    }
>+
>+    @CalledByNative("Options")
>+    boolean getDisableEncryption() {
>+      return disableEncryption;
>+    }
>+
>+    @CalledByNative("Options")
>+    boolean getDisableNetworkMonitor() {
>+      return disableNetworkMonitor;
>+    }
>+
>+    @CalledByNative("Options")
>+    boolean getEnableAes128Sha1_32CryptoCipher() {
>+      return enableAes128Sha1_32CryptoCipher;
>+    }
>+
>+    @CalledByNative("Options")
>+    boolean getEnableGcmCryptoSuites() {
>+      return enableGcmCryptoSuites;
>+    }
>+  }
>+
>+  public static class Builder {
>+    private @Nullable Options options;
>+    private @Nullable AudioDeviceModule audioDeviceModule = new LegacyAudioDeviceModule();
>+    private @Nullable VideoEncoderFactory encoderFactory;
>+    private @Nullable VideoDecoderFactory decoderFactory;
>+    private @Nullable AudioProcessingFactory audioProcessingFactory;
>+    private @Nullable FecControllerFactoryFactoryInterface fecControllerFactoryFactory;
>+
>+    private Builder() {}
>+
>+    public Builder setOptions(Options options) {
>+      this.options = options;
>+      return this;
>+    }
>+
>+    public Builder setAudioDeviceModule(AudioDeviceModule audioDeviceModule) {
>+      this.audioDeviceModule = audioDeviceModule;
>+      return this;
>+    }
>+
>+    public Builder setVideoEncoderFactory(VideoEncoderFactory encoderFactory) {
>+      this.encoderFactory = encoderFactory;
>+      return this;
>+    }
>+
>+    public Builder setVideoDecoderFactory(VideoDecoderFactory decoderFactory) {
>+      this.decoderFactory = decoderFactory;
>+      return this;
>+    }
>+
>+    public Builder setAudioProcessingFactory(AudioProcessingFactory audioProcessingFactory) {
>+      if (audioProcessingFactory == null) {
>+        throw new NullPointerException(
>+            "PeerConnectionFactory builder does not accept a null AudioProcessingFactory.");
>+      }
>+      this.audioProcessingFactory = audioProcessingFactory;
>+      return this;
>+    }
>+
>+    public Builder setFecControllerFactoryFactoryInterface(
>+        FecControllerFactoryFactoryInterface fecControllerFactoryFactory) {
>+      this.fecControllerFactoryFactory = fecControllerFactoryFactory;
>+      return this;
>+    }
>+
>+    public PeerConnectionFactory createPeerConnectionFactory() {
>+      VideoEncoderFactory encoderFactory = this.encoderFactory;
>+      VideoDecoderFactory decoderFactory = this.decoderFactory;
>+      // For legacy reasons, we provide implicit built-in codec factories.
>+      // TODO(bugs.webrtc.org/9181): Remove code below. All codec factories should be injected
>+      // explicitly.
>+      if (encoderFactory == null && decoderFactory == null && !enableVideoHwAcceleration) {
>+        encoderFactory = new SoftwareVideoEncoderFactory();
>+        decoderFactory = new SoftwareVideoDecoderFactory();
>+      } else {
>+        if (encoderFactory == null) {
>+          encoderFactory = MediaCodecVideoEncoder.createFactory();
>+        }
>+        if (decoderFactory == null) {
>+          decoderFactory = MediaCodecVideoDecoder.createFactory();
>+        }
>+      }
>+      return new PeerConnectionFactory(options, audioDeviceModule, encoderFactory, decoderFactory,
>+          audioProcessingFactory, fecControllerFactoryFactory);
>+    }
>+  }
>+
>+  public static Builder builder() {
>+    return new Builder();
>   }
> 
>   /**
>@@ -113,12 +240,23 @@ public class PeerConnectionFactory {
>    */
>   public static void initialize(InitializationOptions options) {
>     ContextUtils.initialize(options.applicationContext);
>-    NativeLibrary.initialize(options.nativeLibraryLoader);
>-    nativeInitializeAndroidGlobals(options.applicationContext, options.enableVideoHwAcceleration);
>-    initializeFieldTrials(options.fieldTrials);
>+    NativeLibrary.initialize(options.nativeLibraryLoader, options.nativeLibraryName);
>+    enableVideoHwAcceleration = options.enableVideoHwAcceleration;
>+    nativeInitializeAndroidGlobals();
>+    nativeInitializeFieldTrials(options.fieldTrials);
>     if (options.enableInternalTracer && !internalTracerInitialized) {
>       initializeInternalTracer();
>     }
>+    if (options.loggable != null) {
>+      Logging.injectLoggable(options.loggable, options.loggableSeverity);
>+      nativeInjectLoggable(new JNILogging(options.loggable), options.loggableSeverity.ordinal());
>+    } else {
>+      Logging.d(TAG,
>+          "PeerConnectionFactory was initialized without an injected Loggable. "
>+              + "Any existing Loggable will be deleted.");
>+      Logging.deleteInjectedLoggable();
>+      nativeDeleteLoggable();
>+    }
>   }
> 
>   private void checkInitializeHasBeenCalled() {
>@@ -129,11 +267,6 @@ public class PeerConnectionFactory {
>     }
>   }
> 
>-  // Must be called at least once before creating a PeerConnectionFactory
>-  // (for example, at application startup time).
>-  private static native void nativeInitializeAndroidGlobals(
>-      Context context, boolean videoHwAcceleration);
>-
>   private static void initializeInternalTracer() {
>     internalTracerInitialized = true;
>     nativeInitializeInternalTracer();
>@@ -147,7 +280,11 @@ public class PeerConnectionFactory {
>   // Field trial initialization. Must be called before PeerConnectionFactory
>   // is created.
>   // Deprecated, use PeerConnectionFactory.initialize instead.
>-  @Deprecated public static native void initializeFieldTrials(String fieldTrialsInitString);
>+  @Deprecated
>+  public static void initializeFieldTrials(String fieldTrialsInitString) {
>+    nativeInitializeFieldTrials(fieldTrialsInitString);
>+  }
>+
>   // Wrapper of webrtc::field_trial::FindFullName. Develop the feature with default behaviour off.
>   // Example usage:
>   // if (PeerConnectionFactory.fieldTrialsFindFullName("WebRTCExperiment").equals("Enabled")) {
>@@ -156,90 +293,110 @@ public class PeerConnectionFactory {
>   //   method2();
>   // }
>   public static String fieldTrialsFindFullName(String name) {
>-    return NativeLibrary.isLoaded() ? nativeFieldTrialsFindFullName(name) : "";
>+    return NativeLibrary.isLoaded() ? nativeFindFieldTrialsFullName(name) : "";
>   }
>-  private static native String nativeFieldTrialsFindFullName(String name);
>-  // Internal tracing initialization. Must be called before PeerConnectionFactory is created to
>-  // prevent racing with tracing code.
>-  // Deprecated, use PeerConnectionFactory.initialize instead.
>-  private static native void nativeInitializeInternalTracer();
>-  // Internal tracing shutdown, called to prevent resource leaks. Must be called after
>-  // PeerConnectionFactory is gone to prevent races with code performing tracing.
>-  private static native void nativeShutdownInternalTracer();
>   // Start/stop internal capturing of internal tracing.
>-  public static native boolean startInternalTracingCapture(String tracing_filename);
>-  public static native void stopInternalTracingCapture();
>-
>-  @Deprecated
>-  public PeerConnectionFactory() {
>-    this(null);
>+  public static boolean startInternalTracingCapture(String tracingFilename) {
>+    return nativeStartInternalTracingCapture(tracingFilename);
>   }
> 
>-  // Note: initializeAndroidGlobals must be called at least once before
>-  // constructing a PeerConnectionFactory.
>-  public PeerConnectionFactory(Options options) {
>-    this(options, null /* encoderFactory */, null /* decoderFactory */);
>+  public static void stopInternalTracingCapture() {
>+    nativeStopInternalTracingCapture();
>   }
> 
>-  public PeerConnectionFactory(
>-      Options options, VideoEncoderFactory encoderFactory, VideoDecoderFactory decoderFactory) {
>+  private PeerConnectionFactory(Options options, @Nullable AudioDeviceModule audioDeviceModule,
>+      @Nullable VideoEncoderFactory encoderFactory, @Nullable VideoDecoderFactory decoderFactory,
>+      @Nullable AudioProcessingFactory audioProcessingFactory,
>+      @Nullable FecControllerFactoryFactoryInterface fecControllerFactoryFactory) {
>     checkInitializeHasBeenCalled();
>-    nativeFactory = nativeCreatePeerConnectionFactory(options, encoderFactory, decoderFactory);
>+    nativeFactory = nativeCreatePeerConnectionFactory(ContextUtils.getApplicationContext(), options,
>+        audioDeviceModule == null ? 0 : audioDeviceModule.getNativeAudioDeviceModulePointer(),
>+        encoderFactory, decoderFactory,
>+        audioProcessingFactory == null ? 0 : audioProcessingFactory.createNative(),
>+        fecControllerFactoryFactory == null ? 0 : fecControllerFactoryFactory.createNative());
>     if (nativeFactory == 0) {
>       throw new RuntimeException("Failed to initialize PeerConnectionFactory!");
>     }
>   }
> 
>-  public PeerConnectionFactory(Options options, VideoEncoderFactory encoderFactory,
>-      VideoDecoderFactory decoderFactory, AudioProcessingFactory audioProcessingFactory) {
>+  @CalledByNative
>+  PeerConnectionFactory(long nativeFactory) {
>     checkInitializeHasBeenCalled();
>-    if (audioProcessingFactory == null) {
>-      throw new NullPointerException(
>-          "PeerConnectionFactory constructor does not accept a null AudioProcessingFactory.");
>-    }
>-    nativeFactory = nativeCreatePeerConnectionFactoryWithAudioProcessing(
>-        options, encoderFactory, decoderFactory, audioProcessingFactory.createNative());
>     if (nativeFactory == 0) {
>       throw new RuntimeException("Failed to initialize PeerConnectionFactory!");
>     }
>+    this.nativeFactory = nativeFactory;
>   }
> 
>-  public PeerConnection createPeerConnection(PeerConnection.RTCConfiguration rtcConfig,
>-      MediaConstraints constraints, PeerConnection.Observer observer) {
>-    long nativeObserver = nativeCreateObserver(observer);
>+  /**
>+   * Internal helper function to pass the parameters down into the native JNI bridge.
>+   */
>+  @Nullable
>+  PeerConnection createPeerConnectionInternal(PeerConnection.RTCConfiguration rtcConfig,
>+      MediaConstraints constraints, PeerConnection.Observer observer,
>+      SSLCertificateVerifier sslCertificateVerifier) {
>+    long nativeObserver = PeerConnection.createNativePeerConnectionObserver(observer);
>     if (nativeObserver == 0) {
>       return null;
>     }
>-    long nativePeerConnection =
>-        nativeCreatePeerConnection(nativeFactory, rtcConfig, constraints, nativeObserver);
>+    long nativePeerConnection = nativeCreatePeerConnection(
>+        nativeFactory, rtcConfig, constraints, nativeObserver, sslCertificateVerifier);
>     if (nativePeerConnection == 0) {
>       return null;
>     }
>-    return new PeerConnection(nativePeerConnection, nativeObserver);
>+    return new PeerConnection(nativePeerConnection);
>+  }
>+
>+  /**
>+   * Deprecated. PeerConnection constraints are deprecated. Supply values in rtcConfig struct
>+   * instead and use the method without constraints in the signature.
>+   */
>+  @Nullable
>+  @Deprecated
>+  public PeerConnection createPeerConnection(PeerConnection.RTCConfiguration rtcConfig,
>+      MediaConstraints constraints, PeerConnection.Observer observer) {
>+    return createPeerConnectionInternal(
>+        rtcConfig, constraints, observer, /* sslCertificateVerifier= */ null);
>   }
> 
>+  /**
>+   * Deprecated. PeerConnection constraints are deprecated. Supply values in rtcConfig struct
>+   * instead and use the method without constraints in the signature.
>+   */
>+  @Nullable
>+  @Deprecated
>   public PeerConnection createPeerConnection(List<PeerConnection.IceServer> iceServers,
>       MediaConstraints constraints, PeerConnection.Observer observer) {
>     PeerConnection.RTCConfiguration rtcConfig = new PeerConnection.RTCConfiguration(iceServers);
>     return createPeerConnection(rtcConfig, constraints, observer);
>   }
> 
>+  @Nullable
>+  public PeerConnection createPeerConnection(
>+      List<PeerConnection.IceServer> iceServers, PeerConnection.Observer observer) {
>+    PeerConnection.RTCConfiguration rtcConfig = new PeerConnection.RTCConfiguration(iceServers);
>+    return createPeerConnection(rtcConfig, observer);
>+  }
>+
>+  @Nullable
>+  public PeerConnection createPeerConnection(
>+      PeerConnection.RTCConfiguration rtcConfig, PeerConnection.Observer observer) {
>+    return createPeerConnection(rtcConfig, null /* constraints */, observer);
>+  }
>+
>+  @Nullable
>+  public PeerConnection createPeerConnection(
>+      PeerConnection.RTCConfiguration rtcConfig, PeerConnectionDependencies dependencies) {
>+    return createPeerConnectionInternal(rtcConfig, null /* constraints */,
>+        dependencies.getObserver(), dependencies.getSSLCertificateVerifier());
>+  }
>+
>   public MediaStream createLocalMediaStream(String label) {
>     return new MediaStream(nativeCreateLocalMediaStream(nativeFactory, label));
>   }
> 
>-  public VideoSource createVideoSource(VideoCapturer capturer) {
>-    final EglBase.Context eglContext =
>-        localEglbase == null ? null : localEglbase.getEglBaseContext();
>-    final SurfaceTextureHelper surfaceTextureHelper =
>-        SurfaceTextureHelper.create(VIDEO_CAPTURER_THREAD_NAME, eglContext);
>-    long nativeAndroidVideoTrackSource =
>-        nativeCreateVideoSource(nativeFactory, surfaceTextureHelper, capturer.isScreencast());
>-    VideoCapturer.CapturerObserver capturerObserver =
>-        new AndroidVideoTrackSourceObserver(nativeAndroidVideoTrackSource);
>-    capturer.initialize(
>-        surfaceTextureHelper, ContextUtils.getApplicationContext(), capturerObserver);
>-    return new VideoSource(nativeAndroidVideoTrackSource);
>+  public VideoSource createVideoSource(boolean isScreencast) {
>+    return new VideoSource(nativeCreateVideoSource(nativeFactory, isScreencast));
>   }
> 
>   public VideoTrack createVideoTrack(String id, VideoSource source) {
>@@ -267,31 +424,20 @@ public class PeerConnectionFactory {
>     nativeStopAecDump(nativeFactory);
>   }
> 
>-  @Deprecated
>-  public void setOptions(Options options) {
>-    nativeSetOptions(nativeFactory, options);
>-  }
>-
>-  /** Set the EGL context used by HW Video encoding and decoding.
>+  /**
>+   * Set the EGL context used by HW Video encoding and decoding.
>    *
>    * @param localEglContext   Must be the same as used by VideoCapturerAndroid and any local video
>    *                          renderer.
>    * @param remoteEglContext  Must be the same as used by any remote video renderer.
>+   * @deprecated Use new HW video encoded/decoder instead, and use createVideoSource(boolean
>+   * isScreencast) instead of createVideoSource(VideoCapturer).
>    */
>+  @Deprecated
>   public void setVideoHwAccelerationOptions(
>       EglBase.Context localEglContext, EglBase.Context remoteEglContext) {
>-    if (localEglbase != null) {
>-      Logging.w(TAG, "Egl context already set.");
>-      localEglbase.release();
>-    }
>-    if (remoteEglbase != null) {
>-      Logging.w(TAG, "Egl context already set.");
>-      remoteEglbase.release();
>-    }
>-    localEglbase = EglBase.create(localEglContext);
>-    remoteEglbase = EglBase.create(remoteEglContext);
>-    nativeSetVideoHwAccelerationOptions(
>-        nativeFactory, localEglbase.getEglBaseContext(), remoteEglbase.getEglBaseContext());
>+    MediaCodecVideoEncoder.setEglContext(localEglContext);
>+    MediaCodecVideoDecoder.setEglContext(remoteEglContext);
>   }
> 
>   public void dispose() {
>@@ -299,17 +445,25 @@ public class PeerConnectionFactory {
>     networkThread = null;
>     workerThread = null;
>     signalingThread = null;
>-    if (localEglbase != null)
>-      localEglbase.release();
>-    if (remoteEglbase != null)
>-      remoteEglbase.release();
>+    MediaCodecVideoEncoder.disposeEglContext();
>+    MediaCodecVideoDecoder.disposeEglContext();
>   }
> 
>   public void threadsCallbacks() {
>-    nativeThreadsCallbacks(nativeFactory);
>+    nativeInvokeThreadsCallbacks(nativeFactory);
>+  }
>+
>+  /** Returns a pointer to the native webrtc::PeerConnectionFactoryInterface. */
>+  public long getNativePeerConnectionFactory() {
>+    return nativeGetNativePeerConnectionFactory(nativeFactory);
>   }
> 
>-  private static void printStackTrace(Thread thread, String threadName) {
>+  /** Returns a pointer to the native OwnedFactoryAndThreads object */
>+  public long getNativeOwnedFactoryAndThreads() {
>+    return nativeFactory;
>+  }
>+
>+  private static void printStackTrace(@Nullable Thread thread, String threadName) {
>     if (thread != null) {
>       StackTraceElement[] stackTraces = thread.getStackTrace();
>       if (stackTraces.length > 0) {
>@@ -327,58 +481,54 @@ public class PeerConnectionFactory {
>     printStackTrace(signalingThread, "Signaling thread");
>   }
> 
>+  @CalledByNative
>   private static void onNetworkThreadReady() {
>     networkThread = Thread.currentThread();
>     Logging.d(TAG, "onNetworkThreadReady");
>   }
> 
>+  @CalledByNative
>   private static void onWorkerThreadReady() {
>     workerThread = Thread.currentThread();
>     Logging.d(TAG, "onWorkerThreadReady");
>   }
> 
>+  @CalledByNative
>   private static void onSignalingThreadReady() {
>     signalingThread = Thread.currentThread();
>     Logging.d(TAG, "onSignalingThreadReady");
>   }
> 
>-  private static native long nativeCreatePeerConnectionFactory(
>-      Options options, VideoEncoderFactory encoderFactory, VideoDecoderFactory decoderFactory);
>-
>-  private static native long nativeCreatePeerConnectionFactoryWithAudioProcessing(Options options,
>-      VideoEncoderFactory encoderFactory, VideoDecoderFactory decoderFactory,
>-      long nativeAudioProcessor);
>-
>-  private static native long nativeCreateObserver(PeerConnection.Observer observer);
>-
>-  private static native long nativeCreatePeerConnection(long nativeFactory,
>-      PeerConnection.RTCConfiguration rtcConfig, MediaConstraints constraints, long nativeObserver);
>-
>-  private static native long nativeCreateLocalMediaStream(long nativeFactory, String label);
>-
>-  private static native long nativeCreateVideoSource(
>-      long nativeFactory, SurfaceTextureHelper surfaceTextureHelper, boolean is_screencast);
>-
>+  // Must be called at least once before creating a PeerConnectionFactory
>+  // (for example, at application startup time).
>+  private static native void nativeInitializeAndroidGlobals();
>+  private static native void nativeInitializeFieldTrials(String fieldTrialsInitString);
>+  private static native String nativeFindFieldTrialsFullName(String name);
>+  private static native void nativeInitializeInternalTracer();
>+  // Internal tracing shutdown, called to prevent resource leaks. Must be called after
>+  // PeerConnectionFactory is gone to prevent races with code performing tracing.
>+  private static native void nativeShutdownInternalTracer();
>+  private static native boolean nativeStartInternalTracingCapture(String tracingFilename);
>+  private static native void nativeStopInternalTracingCapture();
>+  private static native long nativeCreatePeerConnectionFactory(Context context, Options options,
>+      long nativeAudioDeviceModule, VideoEncoderFactory encoderFactory,
>+      VideoDecoderFactory decoderFactory, long nativeAudioProcessor,
>+      long nativeFecControllerFactory);
>+  private static native long nativeCreatePeerConnection(long factory,
>+      PeerConnection.RTCConfiguration rtcConfig, MediaConstraints constraints, long nativeObserver,
>+      SSLCertificateVerifier sslCertificateVerifier);
>+  private static native long nativeCreateLocalMediaStream(long factory, String label);
>+  private static native long nativeCreateVideoSource(long factory, boolean is_screencast);
>   private static native long nativeCreateVideoTrack(
>-      long nativeFactory, String id, long nativeVideoSource);
>-
>-  private static native long nativeCreateAudioSource(
>-      long nativeFactory, MediaConstraints constraints);
>-
>-  private static native long nativeCreateAudioTrack(
>-      long nativeFactory, String id, long nativeSource);
>-
>+      long factory, String id, long nativeVideoSource);
>+  private static native long nativeCreateAudioSource(long factory, MediaConstraints constraints);
>+  private static native long nativeCreateAudioTrack(long factory, String id, long nativeSource);
>   private static native boolean nativeStartAecDump(
>-      long nativeFactory, int file_descriptor, int filesize_limit_bytes);
>-
>-  private static native void nativeStopAecDump(long nativeFactory);
>-
>-  @Deprecated public native void nativeSetOptions(long nativeFactory, Options options);
>-
>-  private static native void nativeSetVideoHwAccelerationOptions(
>-      long nativeFactory, Object localEGLContext, Object remoteEGLContext);
>-
>-  private static native void nativeThreadsCallbacks(long nativeFactory);
>-
>-  private static native void nativeFreeFactory(long nativeFactory);
>+      long factory, int file_descriptor, int filesize_limit_bytes);
>+  private static native void nativeStopAecDump(long factory);
>+  private static native void nativeInvokeThreadsCallbacks(long factory);
>+  private static native void nativeFreeFactory(long factory);
>+  private static native long nativeGetNativePeerConnectionFactory(long factory);
>+  private static native void nativeInjectLoggable(JNILogging jniLogging, int severity);
>+  private static native void nativeDeleteLoggable();
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/RefCounted.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/RefCounted.java
>new file mode 100644
>index 00000000000..7741a445d97
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/RefCounted.java
>@@ -0,0 +1,28 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc;
>+
>+/**
>+ * Interface for ref counted objects in WebRTC. These objects have significant resources that need
>+ * to be freed when they are no longer in use. Each objects starts with ref count of one when
>+ * created. If a reference is passed as a parameter to a method, the caller has ownesrship of the
>+ * object by default - calling release is not necessary unless retain is called.
>+ */
>+public interface RefCounted {
>+  /** Increases ref count by one. */
>+  void retain();
>+
>+  /**
>+   * Decreases ref count by one. When the ref count reaches zero, resources related to the object
>+   * will be freed.
>+   */
>+  void release();
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/RendererCommon.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/RendererCommon.java
>index 0554f119c3a..f039a06f29d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/RendererCommon.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/RendererCommon.java
>@@ -31,7 +31,12 @@ public class RendererCommon {
>     public void onFrameResolutionChanged(int videoWidth, int videoHeight, int rotation);
>   }
> 
>-  /** Interface for rendering frames on an EGLSurface. */
>+  /**
>+   * Interface for rendering frames on an EGLSurface with specified viewport location. Rotation,
>+   * mirror, and cropping is specified using a 4x4 texture coordinate transform matrix. The frame
>+   * input can either be an OES texture, RGB texture, or YUV textures in I420 format. The function
>+   * release() must be called manually to free the resources held by this object.
>+   */
>   public static interface GlDrawer {
>     /**
>      * Functions for drawing frames with different sources. The rendering surface target is
>@@ -113,52 +118,6 @@ public class RendererCommon {
>   // The minimum fraction of the frame content that will be shown for |SCALE_ASPECT_BALANCED|.
>   // This limits excessive cropping when adjusting display size.
>   private static float BALANCED_VISIBLE_FRACTION = 0.5625f;
>-  // clang-format off
>-  public static final float[] identityMatrix() {
>-    return new float[] {
>-        1, 0, 0, 0,
>-        0, 1, 0, 0,
>-        0, 0, 1, 0,
>-        0, 0, 0, 1};
>-  }
>-  // Matrix with transform y' = 1 - y.
>-  public static final float[] verticalFlipMatrix() {
>-    return new float[] {
>-        1,  0, 0, 0,
>-        0, -1, 0, 0,
>-        0,  0, 1, 0,
>-        0,  1, 0, 1};
>-  }
>-
>-  // Matrix with transform x' = 1 - x.
>-  public static final float[] horizontalFlipMatrix() {
>-    return new float[] {
>-        -1, 0, 0, 0,
>-         0, 1, 0, 0,
>-         0, 0, 1, 0,
>-         1, 0, 0, 1};
>-  }
>-  // clang-format on
>-
>-  /**
>-   * Returns texture matrix that will have the effect of rotating the frame |rotationDegree|
>-   * clockwise when rendered.
>-   */
>-  public static float[] rotateTextureMatrix(float[] textureMatrix, float rotationDegree) {
>-    final float[] rotationMatrix = new float[16];
>-    Matrix.setRotateM(rotationMatrix, 0, rotationDegree, 0, 0, 1);
>-    adjustOrigin(rotationMatrix);
>-    return multiplyMatrices(textureMatrix, rotationMatrix);
>-  }
>-
>-  /**
>-   * Returns new matrix with the result of a * b.
>-   */
>-  public static float[] multiplyMatrices(float[] a, float[] b) {
>-    final float[] resultMatrix = new float[16];
>-    Matrix.multiplyMM(resultMatrix, 0, a, 0, b, 0);
>-    return resultMatrix;
>-  }
> 
>   /**
>    * Returns layout transformation matrix that applies an optional mirror effect and compensates
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/RtpParameters.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/RtpParameters.java
>index f1cca9d6602..087936e4917 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/RtpParameters.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/RtpParameters.java
>@@ -10,8 +10,11 @@
> 
> package org.webrtc;
> 
>-import java.util.List;
>+import javax.annotation.Nullable;
> import java.util.ArrayList;
>+import java.util.List;
>+import java.util.Map;
>+import org.webrtc.MediaStreamTrack;
> 
> /**
>  * The parameters for an {@code RtpSender}, as defined in
>@@ -30,10 +33,42 @@ public class RtpParameters {
>     // If non-null, this represents the Transport Independent Application
>     // Specific maximum bandwidth defined in RFC3890. If null, there is no
>     // maximum bitrate.
>-    public Integer maxBitrateBps;
>+    @Nullable public Integer maxBitrateBps;
>+    // Not implemented.
>+    @Nullable public Integer minBitrateBps;
>     // SSRC to be used by this encoding.
>     // Can't be changed between getParameters/setParameters.
>     public Long ssrc;
>+
>+    @CalledByNative("Encoding")
>+    Encoding(boolean active, Integer maxBitrateBps, Integer minBitrateBps, Long ssrc) {
>+      this.active = active;
>+      this.maxBitrateBps = maxBitrateBps;
>+      this.minBitrateBps = minBitrateBps;
>+      this.ssrc = ssrc;
>+    }
>+
>+    @CalledByNative("Encoding")
>+    boolean getActive() {
>+      return active;
>+    }
>+
>+    @Nullable
>+    @CalledByNative("Encoding")
>+    Integer getMaxBitrateBps() {
>+      return maxBitrateBps;
>+    }
>+
>+    @Nullable
>+    @CalledByNative("Encoding")
>+    Integer getMinBitrateBps() {
>+      return minBitrateBps;
>+    }
>+
>+    @CalledByNative("Encoding")
>+    Long getSsrc() {
>+      return ssrc;
>+    }
>   }
> 
>   public static class Codec {
>@@ -47,11 +82,149 @@ public class RtpParameters {
>     public Integer clockRate;
>     // The number of audio channels used. Set to null for video codecs.
>     public Integer numChannels;
>+    // The "format specific parameters" field from the "a=fmtp" line in the SDP
>+    public Map<String, String> parameters;
>+
>+    @CalledByNative("Codec")
>+    Codec(int payloadType, String name, MediaStreamTrack.MediaType kind, Integer clockRate,
>+        Integer numChannels, Map<String, String> parameters) {
>+      this.payloadType = payloadType;
>+      this.name = name;
>+      this.kind = kind;
>+      this.clockRate = clockRate;
>+      this.numChannels = numChannels;
>+      this.parameters = parameters;
>+    }
>+
>+    @CalledByNative("Codec")
>+    int getPayloadType() {
>+      return payloadType;
>+    }
>+
>+    @CalledByNative("Codec")
>+    String getName() {
>+      return name;
>+    }
>+
>+    @CalledByNative("Codec")
>+    MediaStreamTrack.MediaType getKind() {
>+      return kind;
>+    }
>+
>+    @CalledByNative("Codec")
>+    Integer getClockRate() {
>+      return clockRate;
>+    }
>+
>+    @CalledByNative("Codec")
>+    Integer getNumChannels() {
>+      return numChannels;
>+    }
>+
>+    @CalledByNative("Codec")
>+    Map getParameters() {
>+      return parameters;
>+    }
>+  }
>+
>+  public static class Rtcp {
>+    /** The Canonical Name used by RTCP */
>+    private final String cname;
>+    /** Whether reduced size RTCP is configured or compound RTCP */
>+    private final boolean reducedSize;
>+
>+    @CalledByNative("Rtcp")
>+    Rtcp(String cname, boolean reducedSize) {
>+      this.cname = cname;
>+      this.reducedSize = reducedSize;
>+    }
>+
>+    @CalledByNative("Rtcp")
>+    public String getCname() {
>+      return cname;
>+    }
>+
>+    @CalledByNative("Rtcp")
>+    public boolean getReducedSize() {
>+      return reducedSize;
>+    }
>+  }
>+
>+  public static class HeaderExtension {
>+    /** The URI of the RTP header extension, as defined in RFC5285. */
>+    private final String uri;
>+    /** The value put in the RTP packet to identify the header extension. */
>+    private final int id;
>+    /** Whether the header extension is encrypted or not. */
>+    private final boolean encrypted;
>+
>+    @CalledByNative("HeaderExtension")
>+    HeaderExtension(String uri, int id, boolean encrypted) {
>+      this.uri = uri;
>+      this.id = id;
>+      this.encrypted = encrypted;
>+    }
>+
>+    @CalledByNative("HeaderExtension")
>+    public String getUri() {
>+      return uri;
>+    }
>+
>+    @CalledByNative("HeaderExtension")
>+    public int getId() {
>+      return id;
>+    }
>+
>+    @CalledByNative("HeaderExtension")
>+    public boolean getEncrypted() {
>+      return encrypted;
>+    }
>   }
> 
>-  public final List<Encoding> encodings = new ArrayList<>();
>+  public final String transactionId;
>+
>+  private final Rtcp rtcp;
>+
>+  private final List<HeaderExtension> headerExtensions;
>+
>+  public final List<Encoding> encodings;
>   // Codec parameters can't currently be changed between getParameters and
>   // setParameters. Though in the future it will be possible to reorder them or
>   // remove them.
>-  public final List<Codec> codecs = new ArrayList<>();
>+  public final List<Codec> codecs;
>+
>+  @CalledByNative
>+  RtpParameters(String transactionId, Rtcp rtcp, List<HeaderExtension> headerExtensions,
>+      List<Encoding> encodings, List<Codec> codecs) {
>+    this.transactionId = transactionId;
>+    this.rtcp = rtcp;
>+    this.headerExtensions = headerExtensions;
>+    this.encodings = encodings;
>+    this.codecs = codecs;
>+  }
>+
>+  @CalledByNative
>+  String getTransactionId() {
>+    return transactionId;
>+  }
>+
>+  @CalledByNative
>+  public Rtcp getRtcp() {
>+    return rtcp;
>+  }
>+
>+  @CalledByNative
>+  public List<HeaderExtension> getHeaderExtensions() {
>+    return headerExtensions;
>+  }
>+
>+  @CalledByNative
>+  List<Encoding> getEncodings() {
>+    return encodings;
>+  }
>+
>+  @CalledByNative
>+  List<Codec> getCodecs() {
>+    return codecs;
>+  }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/RtpReceiver.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/RtpReceiver.java
>index 69dfd9230f7..193820b5ac7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/RtpReceiver.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/RtpReceiver.java
>@@ -10,32 +10,37 @@
> 
> package org.webrtc;
> 
>+import javax.annotation.Nullable;
>+import org.webrtc.MediaStreamTrack;
>+
> /** Java wrapper for a C++ RtpReceiverInterface. */
> public class RtpReceiver {
>   /** Java wrapper for a C++ RtpReceiverObserverInterface*/
>   public static interface Observer {
>     // Called when the first audio or video packet is received.
>+    @CalledByNative("Observer")
>     public void onFirstPacketReceived(MediaStreamTrack.MediaType media_type);
>   }
> 
>   final long nativeRtpReceiver;
>   private long nativeObserver;
> 
>-  private MediaStreamTrack cachedTrack;
>+  @Nullable private MediaStreamTrack cachedTrack;
> 
>+  @CalledByNative
>   public RtpReceiver(long nativeRtpReceiver) {
>     this.nativeRtpReceiver = nativeRtpReceiver;
>-    long track = nativeGetTrack(nativeRtpReceiver);
>-    // We can assume that an RtpReceiver always has an associated track.
>-    cachedTrack = new MediaStreamTrack(track);
>+    long nativeTrack = nativeGetTrack(nativeRtpReceiver);
>+    cachedTrack = MediaStreamTrack.createMediaStreamTrack(nativeTrack);
>   }
> 
>+  @Nullable
>   public MediaStreamTrack track() {
>     return cachedTrack;
>   }
> 
>-  public boolean setParameters(RtpParameters parameters) {
>-    return nativeSetParameters(nativeRtpReceiver, parameters);
>+  public boolean setParameters(@Nullable RtpParameters parameters) {
>+    return parameters == null ? false : nativeSetParameters(nativeRtpReceiver, parameters);
>   }
> 
>   public RtpParameters getParameters() {
>@@ -43,9 +48,10 @@ public class RtpReceiver {
>   }
> 
>   public String id() {
>-    return nativeId(nativeRtpReceiver);
>+    return nativeGetId(nativeRtpReceiver);
>   }
> 
>+  @CalledByNative
>   public void dispose() {
>     cachedTrack.dispose();
>     if (nativeObserver != 0) {
>@@ -65,16 +71,10 @@ public class RtpReceiver {
> 
>   // This should increment the reference count of the track.
>   // Will be released in dispose().
>-  private static native long nativeGetTrack(long nativeRtpReceiver);
>-
>-  private static native boolean nativeSetParameters(
>-      long nativeRtpReceiver, RtpParameters parameters);
>-
>-  private static native RtpParameters nativeGetParameters(long nativeRtpReceiver);
>-
>-  private static native String nativeId(long nativeRtpReceiver);
>-
>-  private static native long nativeSetObserver(long nativeRtpReceiver, Observer observer);
>-
>-  private static native long nativeUnsetObserver(long nativeRtpReceiver, long nativeObserver);
>+  private static native long nativeGetTrack(long rtpReceiver);
>+  private static native boolean nativeSetParameters(long rtpReceiver, RtpParameters parameters);
>+  private static native RtpParameters nativeGetParameters(long rtpReceiver);
>+  private static native String nativeGetId(long rtpReceiver);
>+  private static native long nativeSetObserver(long rtpReceiver, Observer observer);
>+  private static native void nativeUnsetObserver(long rtpReceiver, long nativeObserver);
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/RtpSender.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/RtpSender.java
>index fbb62db853c..4e268f3361c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/RtpSender.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/RtpSender.java
>@@ -10,20 +10,22 @@
> 
> package org.webrtc;
> 
>+import javax.annotation.Nullable;
>+
> /** Java wrapper for a C++ RtpSenderInterface. */
> public class RtpSender {
>   final long nativeRtpSender;
> 
>-  private MediaStreamTrack cachedTrack;
>+  @Nullable private MediaStreamTrack cachedTrack;
>   private boolean ownsTrack = true;
> 
>-  private final DtmfSender dtmfSender;
>+  private final @Nullable DtmfSender dtmfSender;
> 
>+  @CalledByNative
>   public RtpSender(long nativeRtpSender) {
>     this.nativeRtpSender = nativeRtpSender;
>-    long track = nativeGetTrack(nativeRtpSender);
>-    // It may be possible for an RtpSender to be created without a track.
>-    cachedTrack = (track != 0) ? new MediaStreamTrack(track) : null;
>+    long nativeTrack = nativeGetTrack(nativeRtpSender);
>+    cachedTrack = MediaStreamTrack.createMediaStreamTrack(nativeTrack);
> 
>     long nativeDtmfSender = nativeGetDtmfSender(nativeRtpSender);
>     dtmfSender = (nativeDtmfSender != 0) ? new DtmfSender(nativeDtmfSender) : null;
>@@ -43,7 +45,7 @@ public class RtpSender {
>    *                      or a MediaStream.
>    * @return              true on success and false on failure.
>    */
>-  public boolean setTrack(MediaStreamTrack track, boolean takeOwnership) {
>+  public boolean setTrack(@Nullable MediaStreamTrack track, boolean takeOwnership) {
>     if (!nativeSetTrack(nativeRtpSender, (track == null) ? 0 : track.nativeTrack)) {
>       return false;
>     }
>@@ -55,6 +57,7 @@ public class RtpSender {
>     return true;
>   }
> 
>+  @Nullable
>   public MediaStreamTrack track() {
>     return cachedTrack;
>   }
>@@ -68,9 +71,10 @@ public class RtpSender {
>   }
> 
>   public String id() {
>-    return nativeId(nativeRtpSender);
>+    return nativeGetId(nativeRtpSender);
>   }
> 
>+  @Nullable
>   public DtmfSender dtmf() {
>     return dtmfSender;
>   }
>@@ -85,19 +89,19 @@ public class RtpSender {
>     JniCommon.nativeReleaseRef(nativeRtpSender);
>   }
> 
>-  private static native boolean nativeSetTrack(long nativeRtpSender, long nativeTrack);
>+  private static native boolean nativeSetTrack(long rtpSender, long nativeTrack);
> 
>   // This should increment the reference count of the track.
>   // Will be released in dispose() or setTrack().
>-  private static native long nativeGetTrack(long nativeRtpSender);
>+  private static native long nativeGetTrack(long rtpSender);
> 
>   // This should increment the reference count of the DTMF sender.
>   // Will be released in dispose().
>-  private static native long nativeGetDtmfSender(long nativeRtpSender);
>+  private static native long nativeGetDtmfSender(long rtpSender);
> 
>-  private static native boolean nativeSetParameters(long nativeRtpSender, RtpParameters parameters);
>+  private static native boolean nativeSetParameters(long rtpSender, RtpParameters parameters);
> 
>-  private static native RtpParameters nativeGetParameters(long nativeRtpSender);
>+  private static native RtpParameters nativeGetParameters(long rtpSender);
> 
>-  private static native String nativeId(long nativeRtpSender);
>+  private static native String nativeGetId(long rtpSender);
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/RtpTransceiver.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/RtpTransceiver.java
>new file mode 100644
>index 00000000000..93326a41a30
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/RtpTransceiver.java
>@@ -0,0 +1,215 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc;
>+
>+import java.util.ArrayList;
>+import java.util.Collections;
>+import java.util.List;
>+import org.webrtc.RtpParameters.Encoding;
>+
>+/**
>+ * Java wrapper for a C++ RtpTransceiverInterface.
>+ *
>+ * <p>The RTCRtpTransceiver maps to the RTCRtpTransceiver defined by the WebRTC
>+ * specification. A transceiver represents a combination of an RTCRtpSender
>+ * and an RTCRtpReceiver that share a common mid. As defined in JSEP, an
>+ * RTCRtpTransceiver is said to be associated with a media description if its
>+ * mid property is non-nil; otherwise, it is said to be disassociated.
>+ * JSEP: https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24
>+ *
>+ * <p>Note that RTCRtpTransceivers are only supported when using
>+ * RTCPeerConnection with Unified Plan SDP.
>+ *
>+ * <p>WebRTC specification for RTCRtpTransceiver, the JavaScript analog:
>+ * https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver
>+ */
>+public class RtpTransceiver {
>+  /** Java version of webrtc::RtpTransceiverDirection - the ordering must be kept in sync. */
>+  public enum RtpTransceiverDirection {
>+    SEND_RECV(0),
>+    SEND_ONLY(1),
>+    RECV_ONLY(2),
>+    INACTIVE(3);
>+
>+    private final int nativeIndex;
>+
>+    private RtpTransceiverDirection(int nativeIndex) {
>+      this.nativeIndex = nativeIndex;
>+    }
>+
>+    @CalledByNative("RtpTransceiverDirection")
>+    int getNativeIndex() {
>+      return nativeIndex;
>+    }
>+
>+    @CalledByNative("RtpTransceiverDirection")
>+    static RtpTransceiverDirection fromNativeIndex(int nativeIndex) {
>+      for (RtpTransceiverDirection type : RtpTransceiverDirection.values()) {
>+        if (type.getNativeIndex() == nativeIndex) {
>+          return type;
>+        }
>+      }
>+      throw new IllegalArgumentException(
>+          "Uknown native RtpTransceiverDirection type" + nativeIndex);
>+    }
>+  }
>+
>+  /**
>+   * Tracks webrtc::RtpTransceiverInit. https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiverinit
>+   * A structure for initializing an RtpTransceiver in a call to addTransceiver.
>+   * Note: This does not contain a list of encoding parameters, because they are currently
>+   * not being used natively.
>+   */
>+  public static final class RtpTransceiverInit {
>+    private final RtpTransceiverDirection direction;
>+    private final List<String> streamIds;
>+
>+    public RtpTransceiverInit() {
>+      this(RtpTransceiverDirection.SEND_RECV);
>+    }
>+
>+    public RtpTransceiverInit(RtpTransceiverDirection direction) {
>+      this(direction, Collections.emptyList());
>+    }
>+
>+    public RtpTransceiverInit(RtpTransceiverDirection direction, List<String> streamIds) {
>+      this.direction = direction;
>+      this.streamIds = new ArrayList<String>(streamIds);
>+    }
>+
>+    @CalledByNative("RtpTransceiverInit")
>+    int getDirectionNativeIndex() {
>+      return direction.getNativeIndex();
>+    }
>+
>+    @CalledByNative("RtpTransceiverInit")
>+    List<String> getStreamIds() {
>+      return new ArrayList<String>(this.streamIds);
>+    }
>+  }
>+
>+  private final long nativeRtpTransceiver;
>+  private RtpSender cachedSender;
>+  private RtpReceiver cachedReceiver;
>+
>+  @CalledByNative
>+  protected RtpTransceiver(long nativeRtpTransceiver) {
>+    this.nativeRtpTransceiver = nativeRtpTransceiver;
>+    cachedSender = nativeGetSender(nativeRtpTransceiver);
>+    cachedReceiver = nativeGetReceiver(nativeRtpTransceiver);
>+  }
>+
>+  /**
>+   * Media type of the transceiver. Any sender(s)/receiver(s) will have this
>+   * type as well.
>+   */
>+  public MediaStreamTrack.MediaType getMediaType() {
>+    return nativeGetMediaType(nativeRtpTransceiver);
>+  }
>+
>+  /**
>+   * The mid attribute is the mid negotiated and present in the local and
>+   * remote descriptions. Before negotiation is complete, the mid value may be
>+   * null. After rollbacks, the value may change from a non-null value to null.
>+   * https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-mid
>+   */
>+  public String getMid() {
>+    return nativeGetMid(nativeRtpTransceiver);
>+  }
>+
>+  /**
>+   * The sender attribute exposes the RtpSender corresponding to the RTP media
>+   * that may be sent with the transceiver's mid. The sender is always present,
>+   * regardless of the direction of media.
>+   * https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-sender
>+   */
>+  public RtpSender getSender() {
>+    return cachedSender;
>+  }
>+
>+  /**
>+   * The receiver attribute exposes the RtpReceiver corresponding to the RTP
>+   * media that may be received with the transceiver's mid. The receiver is
>+   * always present, regardless of the direction of media.
>+   * https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-receiver
>+   */
>+  public RtpReceiver getReceiver() {
>+    return cachedReceiver;
>+  }
>+
>+  /**
>+   * The stopped attribute indicates that the sender of this transceiver will no
>+   * longer send, and that the receiver will no longer receive. It is true if
>+   * either stop has been called or if setting the local or remote description
>+   * has caused the RtpTransceiver to be stopped.
>+   * https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-stopped
>+   */
>+  public boolean isStopped() {
>+    return nativeStopped(nativeRtpTransceiver);
>+  }
>+
>+  /**
>+   * The direction attribute indicates the preferred direction of this
>+   * transceiver, which will be used in calls to CreateOffer and CreateAnswer.
>+   * https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-direction
>+   */
>+  public RtpTransceiverDirection getDirection() {
>+    return nativeDirection(nativeRtpTransceiver);
>+  }
>+
>+  /**
>+   * The current_direction attribute indicates the current direction negotiated
>+   * for this transceiver. If this transceiver has never been represented in an
>+   * offer/answer exchange, or if the transceiver is stopped, the value is null.
>+   * https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-currentdirection
>+   */
>+  public RtpTransceiverDirection getCurrentDirection() {
>+    return nativeCurrentDirection(nativeRtpTransceiver);
>+  }
>+
>+  /**
>+   * Sets the preferred direction of this transceiver. An update of
>+   * directionality does not take effect immediately. Instead, future calls to
>+   * CreateOffer and CreateAnswer mark the corresponding media descriptions as
>+   * sendrecv, sendonly, recvonly, or inactive.
>+   * https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-direction
>+   */
>+  public void setDirection(RtpTransceiverDirection rtpTransceiverDirection) {
>+    nativeSetDirection(nativeRtpTransceiver, rtpTransceiverDirection);
>+  }
>+
>+  /**
>+   * The Stop method irreversibly stops the RtpTransceiver. The sender of this
>+   * transceiver will no longer send, the receiver will no longer receive.
>+   * https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-stop
>+   */
>+  public void stop() {
>+    nativeStop(nativeRtpTransceiver);
>+  }
>+
>+  @CalledByNative
>+  public void dispose() {
>+    cachedSender.dispose();
>+    cachedReceiver.dispose();
>+    JniCommon.nativeReleaseRef(nativeRtpTransceiver);
>+  }
>+
>+  private static native MediaStreamTrack.MediaType nativeGetMediaType(long rtpTransceiver);
>+  private static native String nativeGetMid(long rtpTransceiver);
>+  private static native RtpSender nativeGetSender(long rtpTransceiver);
>+  private static native RtpReceiver nativeGetReceiver(long rtpTransceiver);
>+  private static native boolean nativeStopped(long rtpTransceiver);
>+  private static native RtpTransceiverDirection nativeDirection(long rtpTransceiver);
>+  private static native RtpTransceiverDirection nativeCurrentDirection(long rtpTransceiver);
>+  private static native void nativeStop(long rtpTransceiver);
>+  private static native void nativeSetDirection(
>+      long rtpTransceiver, RtpTransceiverDirection rtpTransceiverDirection);
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/SSLCertificateVerifier.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/SSLCertificateVerifier.java
>new file mode 100644
>index 00000000000..461cd3b143d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/SSLCertificateVerifier.java
>@@ -0,0 +1,27 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc;
>+
>+/**
>+ * The SSLCertificateVerifier interface allows API users to provide custom
>+ * logic to verify certificates.
>+ */
>+public interface SSLCertificateVerifier {
>+  /**
>+   * Implementations of verify allow applications to provide custom logic for
>+   * verifying certificates. This is not required by default and should be used
>+   * with care.
>+   *
>+   * @param certificate A byte array containing a DER encoded X509 certificate.
>+   * @return True if the certificate is verified and trusted else false.
>+   */
>+  @CalledByNative boolean verify(byte[] certificate);
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/ScreenCapturerAndroid.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/ScreenCapturerAndroid.java
>index 0279dfe721b..a13f871d745 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/ScreenCapturerAndroid.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/ScreenCapturerAndroid.java
>@@ -19,6 +19,7 @@ import android.hardware.display.VirtualDisplay;
> import android.media.projection.MediaProjection;
> import android.media.projection.MediaProjectionManager;
> import android.view.Surface;
>+import javax.annotation.Nullable;
> 
> /**
>  * An implementation of VideoCapturer to capture the screen content as a video stream.
>@@ -26,14 +27,13 @@ import android.view.Surface;
>  * {@code SurfaceTexture} using a {@code SurfaceTextureHelper}.
>  * The {@code SurfaceTextureHelper} is created by the native code and passed to this capturer in
>  * {@code VideoCapturer.initialize()}. On receiving a new frame, this capturer passes it
>- * as a texture to the native code via {@code CapturerObserver.onTextureFrameCaptured()}. This takes
>+ * as a texture to the native code via {@code CapturerObserver.onFrameCaptured()}. This takes
>  * place on the HandlerThread of the given {@code SurfaceTextureHelper}. When done with each frame,
>  * the native code returns the buffer to the  {@code SurfaceTextureHelper} to be used for new
>  * frames. At any time, at most one frame is being processed.
>  */
> @TargetApi(21)
>-public class ScreenCapturerAndroid
>-    implements VideoCapturer, SurfaceTextureHelper.OnTextureFrameAvailableListener {
>+public class ScreenCapturerAndroid implements VideoCapturer, VideoSink {
>   private static final int DISPLAY_FLAGS =
>       DisplayManager.VIRTUAL_DISPLAY_FLAG_PUBLIC | DisplayManager.VIRTUAL_DISPLAY_FLAG_PRESENTATION;
>   // DPI for VirtualDisplay, does not seem to matter for us.
>@@ -44,13 +44,13 @@ public class ScreenCapturerAndroid
> 
>   private int width;
>   private int height;
>-  private VirtualDisplay virtualDisplay;
>-  private SurfaceTextureHelper surfaceTextureHelper;
>-  private CapturerObserver capturerObserver;
>+  @Nullable private VirtualDisplay virtualDisplay;
>+  @Nullable private SurfaceTextureHelper surfaceTextureHelper;
>+  @Nullable private CapturerObserver capturerObserver;
>   private long numCapturedFrames = 0;
>-  private MediaProjection mediaProjection;
>+  @Nullable private MediaProjection mediaProjection;
>   private boolean isDisposed = false;
>-  private MediaProjectionManager mediaProjectionManager;
>+  @Nullable private MediaProjectionManager mediaProjectionManager;
> 
>   /**
>    * Constructs a new Screen Capturer.
>@@ -77,7 +77,7 @@ public class ScreenCapturerAndroid
>   // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
>   @SuppressWarnings("NoSynchronizedMethodCheck")
>   public synchronized void initialize(final SurfaceTextureHelper surfaceTextureHelper,
>-      final Context applicationContext, final VideoCapturer.CapturerObserver capturerObserver) {
>+      final Context applicationContext, final CapturerObserver capturerObserver) {
>     checkNotDisposed();
> 
>     if (capturerObserver == null) {
>@@ -185,7 +185,7 @@ public class ScreenCapturerAndroid
>   }
> 
>   private void createVirtualDisplay() {
>-    surfaceTextureHelper.getSurfaceTexture().setDefaultBufferSize(width, height);
>+    surfaceTextureHelper.setTextureSize(width, height);
>     virtualDisplay = mediaProjection.createVirtualDisplay("WebRTC_ScreenCapture", width, height,
>         VIRTUAL_DISPLAY_DPI, DISPLAY_FLAGS, new Surface(surfaceTextureHelper.getSurfaceTexture()),
>         null /* callback */, null /* callback handler */);
>@@ -193,10 +193,9 @@ public class ScreenCapturerAndroid
> 
>   // This is called on the internal looper thread of {@Code SurfaceTextureHelper}.
>   @Override
>-  public void onTextureFrameAvailable(int oesTextureId, float[] transformMatrix, long timestampNs) {
>+  public void onFrame(VideoFrame frame) {
>     numCapturedFrames++;
>-    capturerObserver.onTextureFrameCaptured(
>-        width, height, oesTextureId, transformMatrix, 0 /* rotation */, timestampNs);
>+    capturerObserver.onFrameCaptured(frame);
>   }
> 
>   @Override
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/SdpObserver.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/SdpObserver.java
>index a9c03f7e770..afa99bc5528 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/SdpObserver.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/SdpObserver.java
>@@ -13,14 +13,14 @@ package org.webrtc;
> /** Interface for observing SDP-related events. */
> public interface SdpObserver {
>   /** Called on success of Create{Offer,Answer}(). */
>-  public void onCreateSuccess(SessionDescription sdp);
>+  @CalledByNative void onCreateSuccess(SessionDescription sdp);
> 
>   /** Called on success of Set{Local,Remote}Description(). */
>-  public void onSetSuccess();
>+  @CalledByNative void onSetSuccess();
> 
>   /** Called on error of Create{Offer,Answer}(). */
>-  public void onCreateFailure(String error);
>+  @CalledByNative void onCreateFailure(String error);
> 
>   /** Called on error of Set{Local,Remote}Description(). */
>-  public void onSetFailure(String error);
>+  @CalledByNative void onSetFailure(String error);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/SessionDescription.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/SessionDescription.java
>index 7367085afef..62601f0bf2a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/SessionDescription.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/SessionDescription.java
>@@ -28,6 +28,7 @@ public class SessionDescription {
>       return name().toLowerCase(Locale.US);
>     }
> 
>+    @CalledByNative("Type")
>     public static Type fromCanonicalForm(String canonical) {
>       return Type.valueOf(Type.class, canonical.toUpperCase(Locale.US));
>     }
>@@ -36,8 +37,19 @@ public class SessionDescription {
>   public final Type type;
>   public final String description;
> 
>+  @CalledByNative
>   public SessionDescription(Type type, String description) {
>     this.type = type;
>     this.description = description;
>   }
>+
>+  @CalledByNative
>+  String getDescription() {
>+    return description;
>+  }
>+
>+  @CalledByNative
>+  String getTypeInCanonicalForm() {
>+    return type.canonicalForm();
>+  }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/SoftwareVideoDecoderFactory.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/SoftwareVideoDecoderFactory.java
>index 996c299c8db..86dfb7a2c6f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/SoftwareVideoDecoderFactory.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/SoftwareVideoDecoderFactory.java
>@@ -10,16 +10,45 @@
> 
> package org.webrtc;
> 
>+import java.util.ArrayList;
>+import java.util.HashMap;
>+import java.util.List;
>+import javax.annotation.Nullable;
>+
> public class SoftwareVideoDecoderFactory implements VideoDecoderFactory {
>+  @Deprecated
>+  @Nullable
>   @Override
>   public VideoDecoder createDecoder(String codecType) {
>-    if (codecType.equalsIgnoreCase("VP8")) {
>+    return createDecoder(new VideoCodecInfo(codecType, new HashMap<>()));
>+  }
>+
>+  @Nullable
>+  @Override
>+  public VideoDecoder createDecoder(VideoCodecInfo codecType) {
>+    if (codecType.getName().equalsIgnoreCase("VP8")) {
>       return new VP8Decoder();
>     }
>-    if (codecType.equalsIgnoreCase("VP9") && VP9Decoder.isSupported()) {
>+    if (codecType.getName().equalsIgnoreCase("VP9") && VP9Decoder.nativeIsSupported()) {
>       return new VP9Decoder();
>     }
> 
>     return null;
>   }
>+
>+  @Override
>+  public VideoCodecInfo[] getSupportedCodecs() {
>+    return supportedCodecs();
>+  }
>+
>+  static VideoCodecInfo[] supportedCodecs() {
>+    List<VideoCodecInfo> codecs = new ArrayList<VideoCodecInfo>();
>+
>+    codecs.add(new VideoCodecInfo("VP8", new HashMap<>()));
>+    if (VP9Decoder.nativeIsSupported()) {
>+      codecs.add(new VideoCodecInfo("VP9", new HashMap<>()));
>+    }
>+
>+    return codecs.toArray(new VideoCodecInfo[codecs.size()]);
>+  }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/SoftwareVideoEncoderFactory.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/SoftwareVideoEncoderFactory.java
>index 7cc3f9c9561..a2ddf36c389 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/SoftwareVideoEncoderFactory.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/SoftwareVideoEncoderFactory.java
>@@ -10,17 +10,19 @@
> 
> package org.webrtc;
> 
>+import javax.annotation.Nullable;
> import java.util.ArrayList;
> import java.util.HashMap;
> import java.util.List;
> 
> public class SoftwareVideoEncoderFactory implements VideoEncoderFactory {
>+  @Nullable
>   @Override
>   public VideoEncoder createEncoder(VideoCodecInfo info) {
>     if (info.name.equalsIgnoreCase("VP8")) {
>       return new VP8Encoder();
>     }
>-    if (info.name.equalsIgnoreCase("VP9") && VP9Encoder.isSupported()) {
>+    if (info.name.equalsIgnoreCase("VP9") && VP9Encoder.nativeIsSupported()) {
>       return new VP9Encoder();
>     }
> 
>@@ -36,7 +38,7 @@ public class SoftwareVideoEncoderFactory implements VideoEncoderFactory {
>     List<VideoCodecInfo> codecs = new ArrayList<VideoCodecInfo>();
> 
>     codecs.add(new VideoCodecInfo("VP8", new HashMap<>()));
>-    if (VP9Encoder.isSupported()) {
>+    if (VP9Encoder.nativeIsSupported()) {
>       codecs.add(new VideoCodecInfo("VP9", new HashMap<>()));
>     }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/SurfaceEglRenderer.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/SurfaceEglRenderer.java
>index 338871c0a12..350a4cb51a8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/SurfaceEglRenderer.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/SurfaceEglRenderer.java
>@@ -97,13 +97,6 @@ public class SurfaceEglRenderer extends EglRenderer implements SurfaceHolder.Cal
>     super.pauseVideo();
>   }
> 
>-  // VideoRenderer.Callbacks interface.
>-  @Override
>-  public void renderFrame(VideoRenderer.I420Frame frame) {
>-    updateFrameDimensionsAndReportEvents(frame);
>-    super.renderFrame(frame);
>-  }
>-
>   // VideoSink interface.
>   @Override
>   public void onFrame(VideoFrame frame) {
>@@ -132,33 +125,6 @@ public class SurfaceEglRenderer extends EglRenderer implements SurfaceHolder.Cal
>     logD("surfaceChanged: format: " + format + " size: " + width + "x" + height);
>   }
> 
>-  // Update frame dimensions and report any changes to |rendererEvents|.
>-  private void updateFrameDimensionsAndReportEvents(VideoRenderer.I420Frame frame) {
>-    synchronized (layoutLock) {
>-      if (isRenderingPaused) {
>-        return;
>-      }
>-      if (!isFirstFrameRendered) {
>-        isFirstFrameRendered = true;
>-        logD("Reporting first rendered frame.");
>-        if (rendererEvents != null) {
>-          rendererEvents.onFirstFrameRendered();
>-        }
>-      }
>-      if (rotatedFrameWidth != frame.rotatedWidth() || rotatedFrameHeight != frame.rotatedHeight()
>-          || frameRotation != frame.rotationDegree) {
>-        logD("Reporting frame resolution changed to " + frame.width + "x" + frame.height
>-            + " with rotation " + frame.rotationDegree);
>-        if (rendererEvents != null) {
>-          rendererEvents.onFrameResolutionChanged(frame.width, frame.height, frame.rotationDegree);
>-        }
>-        rotatedFrameWidth = frame.rotatedWidth();
>-        rotatedFrameHeight = frame.rotatedHeight();
>-        frameRotation = frame.rotationDegree;
>-      }
>-    }
>-  }
>-
>   // Update frame dimensions and report any changes to |rendererEvents|.
>   private void updateFrameDimensionsAndReportEvents(VideoFrame frame) {
>     synchronized (layoutLock) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/SurfaceTextureHelper.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/SurfaceTextureHelper.java
>index fc5eb376618..146eb22437a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/SurfaceTextureHelper.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/SurfaceTextureHelper.java
>@@ -20,18 +20,16 @@ import android.os.Handler;
> import android.os.HandlerThread;
> import java.nio.ByteBuffer;
> import java.util.concurrent.Callable;
>+import javax.annotation.Nullable;
> import org.webrtc.EglBase;
> import org.webrtc.VideoFrame.TextureBuffer;
> 
> /**
>- * Helper class to create and synchronize access to a SurfaceTexture. The caller will get notified
>- * of new frames in onTextureFrameAvailable(), and should call returnTextureFrame() when done with
>- * the frame. Only one texture frame can be in flight at once, so returnTextureFrame() must be
>- * called in order to receive a new frame. Call stopListening() to stop receiveing new frames. Call
>- * dispose to release all resources once the texture frame is returned.
>- * Note that there is a C++ counter part of this class that optionally can be used. It is used for
>- * wrapping texture frames into webrtc::VideoFrames and also handles calling returnTextureFrame()
>- * when the webrtc::VideoFrame is no longer used.
>+ * Helper class for using a SurfaceTexture to create WebRTC VideoFrames. In order to create WebRTC
>+ * VideoFrames, render onto the SurfaceTexture. The frames will be delivered to the listener. Only
>+ * one texture frame can be in flight at once, so the frame must be released in order to receive a
>+ * new frame. Call stopListening() to stop receiveing new frames. Call dispose to release all
>+ * resources once the texture frame is released.
>  */
> public class SurfaceTextureHelper {
>   private static final String TAG = "SurfaceTextureHelper";
>@@ -39,10 +37,12 @@ public class SurfaceTextureHelper {
>    * Callback interface for being notified that a new texture frame is available. The calls will be
>    * made on the SurfaceTextureHelper handler thread, with a bound EGLContext. The callee is not
>    * allowed to make another EGLContext current on the calling thread.
>+   *
>+   * @deprecated Use a VideoSink as listener instead.
>    */
>+  @Deprecated
>   public interface OnTextureFrameAvailableListener {
>-    abstract void onTextureFrameAvailable(
>-        int oesTextureId, float[] transformMatrix, long timestampNs);
>+    void onTextureFrameAvailable(int oesTextureId, float[] transformMatrix, long timestampNs);
>   }
> 
>   /**
>@@ -50,7 +50,6 @@ public class SurfaceTextureHelper {
>    * thread and handler is created for handling the SurfaceTexture. May return null if EGL fails to
>    * initialize a pixel buffer surface and make it current.
>    */
>-  @CalledByNative
>   public static SurfaceTextureHelper create(
>       final String threadName, final EglBase.Context sharedContext) {
>     final HandlerThread thread = new HandlerThread(threadName);
>@@ -62,6 +61,7 @@ public class SurfaceTextureHelper {
>     // Therefore, in order to control the callback thread on API lvl < 21, the SurfaceTextureHelper
>     // is constructed on the |handler| thread.
>     return ThreadUtils.invokeAtFrontUninterruptibly(handler, new Callable<SurfaceTextureHelper>() {
>+      @Nullable
>       @Override
>       public SurfaceTextureHelper call() {
>         try {
>@@ -78,17 +78,23 @@ public class SurfaceTextureHelper {
>   private final EglBase eglBase;
>   private final SurfaceTexture surfaceTexture;
>   private final int oesTextureId;
>-  private YuvConverter yuvConverter;
>+  private final YuvConverter yuvConverter = new YuvConverter();
> 
>   // These variables are only accessed from the |handler| thread.
>-  private OnTextureFrameAvailableListener listener;
>+  // The type of |listener| is either a VideoSink or the deprecated OnTextureFrameAvailableListener.
>+  @Nullable private Object listener;
>   // The possible states of this class.
>   private boolean hasPendingTexture = false;
>   private volatile boolean isTextureInUse = false;
>   private boolean isQuitting = false;
>+  private int frameRotation;
>+  private int textureWidth;
>+  private int textureHeight;
>   // |pendingListener| is set in setListener() and the runnable is posted to the handler thread.
>   // setListener() is not allowed to be called again before stopListening(), so this is thread safe.
>-  private OnTextureFrameAvailableListener pendingListener;
>+  // The type of |pendingListener| is either a VideoSink or the deprecated
>+  // OnTextureFrameAvailableListener.
>+  @Nullable private Object pendingListener;
>   final Runnable setListenerRunnable = new Runnable() {
>     @Override
>     public void run() {
>@@ -147,8 +153,23 @@ public class SurfaceTextureHelper {
>   /**
>    * Start to stream textures to the given |listener|. If you need to change listener, you need to
>    * call stopListening() first.
>+   *
>+   * @deprecated Use a VideoSink as listener instead.
>    */
>+  @Deprecated
>   public void startListening(final OnTextureFrameAvailableListener listener) {
>+    startListeningInternal(listener);
>+  }
>+
>+  /**
>+   * Start to stream textures to the given |listener|. If you need to change listener, you need to
>+   * call stopListening() first.
>+   */
>+  public void startListening(final VideoSink listener) {
>+    startListeningInternal(listener);
>+  }
>+
>+  private void startListeningInternal(Object listener) {
>     if (this.listener != null || this.pendingListener != null) {
>       throw new IllegalStateException("SurfaceTextureHelper listener has already been set.");
>     }
>@@ -163,15 +184,36 @@ public class SurfaceTextureHelper {
>   public void stopListening() {
>     Logging.d(TAG, "stopListening()");
>     handler.removeCallbacks(setListenerRunnable);
>-    ThreadUtils.invokeAtFrontUninterruptibly(handler, new Runnable() {
>-      @Override
>-      public void run() {
>-        listener = null;
>-        pendingListener = null;
>-      }
>+    ThreadUtils.invokeAtFrontUninterruptibly(handler, () -> {
>+      listener = null;
>+      pendingListener = null;
>     });
>   }
> 
>+  /**
>+   * Use this function to set the texture size. Note, do not call setDefaultBufferSize() yourself
>+   * since this class needs to be aware of the texture size.
>+   */
>+  public void setTextureSize(int textureWidth, int textureHeight) {
>+    if (textureWidth <= 0) {
>+      throw new IllegalArgumentException("Texture width must be positive, but was " + textureWidth);
>+    }
>+    if (textureHeight <= 0) {
>+      throw new IllegalArgumentException(
>+          "Texture height must be positive, but was " + textureHeight);
>+    }
>+    surfaceTexture.setDefaultBufferSize(textureWidth, textureHeight);
>+    handler.post(() -> {
>+      this.textureWidth = textureWidth;
>+      this.textureHeight = textureHeight;
>+    });
>+  }
>+
>+  /** Set the rotation of the delivered frames. */
>+  public void setFrameRotation(int rotation) {
>+    handler.post(() -> this.frameRotation = rotation);
>+  }
>+
>   /**
>    * Retrieve the underlying SurfaceTexture. The SurfaceTexture should be passed in to a video
>    * producer such as a camera or decoder.
>@@ -192,18 +234,17 @@ public class SurfaceTextureHelper {
>    * Call this function to signal that you are done with the frame received in
>    * onTextureFrameAvailable(). Only one texture frame can be in flight at once, so you must call
>    * this function in order to receive a new frame.
>+   *
>+   * @deprecated Use a VideoSink as listener instead.
>    */
>-  @CalledByNative
>+  @Deprecated
>   public void returnTextureFrame() {
>-    handler.post(new Runnable() {
>-      @Override
>-      public void run() {
>-        isTextureInUse = false;
>-        if (isQuitting) {
>-          release();
>-        } else {
>-          tryDeliverTextureFrame();
>-        }
>+    handler.post(() -> {
>+      isTextureInUse = false;
>+      if (isQuitting) {
>+        release();
>+      } else {
>+        tryDeliverTextureFrame();
>       }
>     });
>   }
>@@ -217,52 +258,24 @@ public class SurfaceTextureHelper {
>    * stopped when the texture frame has been returned by a call to returnTextureFrame(). You are
>    * guaranteed to not receive any more onTextureFrameAvailable() after this function returns.
>    */
>-  @CalledByNative
>   public void dispose() {
>     Logging.d(TAG, "dispose()");
>-    ThreadUtils.invokeAtFrontUninterruptibly(handler, new Runnable() {
>-      @Override
>-      public void run() {
>-        isQuitting = true;
>-        if (!isTextureInUse) {
>-          release();
>-        }
>-      }
>-    });
>-  }
>-
>-  /** Deprecated, use textureToYuv. */
>-  @Deprecated
>-  @SuppressWarnings("deprecation") // yuvConverter.convert is deprecated
>-  void textureToYUV(final ByteBuffer buf, final int width, final int height, final int stride,
>-      final int textureId, final float[] transformMatrix) {
>-    if (textureId != oesTextureId) {
>-      throw new IllegalStateException("textureToByteBuffer called with unexpected textureId");
>-    }
>-
>-    ThreadUtils.invokeAtFrontUninterruptibly(handler, new Runnable() {
>-      @Override
>-      public void run() {
>-        if (yuvConverter == null) {
>-          yuvConverter = new YuvConverter();
>-        }
>-        yuvConverter.convert(buf, width, height, stride, textureId, transformMatrix);
>+    ThreadUtils.invokeAtFrontUninterruptibly(handler, () -> {
>+      isQuitting = true;
>+      if (!isTextureInUse) {
>+        release();
>       }
>     });
>   }
> 
>   /**
>    * Posts to the correct thread to convert |textureBuffer| to I420.
>+   *
>+   * @deprecated Use toI420() instead.
>    */
>+  @Deprecated
>   public VideoFrame.I420Buffer textureToYuv(final TextureBuffer textureBuffer) {
>-    final VideoFrame.I420Buffer[] result = new VideoFrame.I420Buffer[1];
>-    ThreadUtils.invokeAtFrontUninterruptibly(handler, () -> {
>-      if (yuvConverter == null) {
>-        yuvConverter = new YuvConverter();
>-      }
>-      result[0] = yuvConverter.convert(textureBuffer);
>-    });
>-    return result[0];
>+    return textureBuffer.toI420();
>   }
> 
>   private void updateTexImage() {
>@@ -289,7 +302,19 @@ public class SurfaceTextureHelper {
>     final float[] transformMatrix = new float[16];
>     surfaceTexture.getTransformMatrix(transformMatrix);
>     final long timestampNs = surfaceTexture.getTimestamp();
>-    listener.onTextureFrameAvailable(oesTextureId, transformMatrix, timestampNs);
>+    if (listener instanceof OnTextureFrameAvailableListener) {
>+      ((OnTextureFrameAvailableListener) listener)
>+          .onTextureFrameAvailable(oesTextureId, transformMatrix, timestampNs);
>+    } else if (listener instanceof VideoSink) {
>+      if (textureWidth == 0 || textureHeight == 0) {
>+        throw new RuntimeException("Texture size has not been set.");
>+      }
>+      final VideoFrame.Buffer buffer = createTextureBuffer(textureWidth, textureHeight,
>+          RendererCommon.convertMatrixToAndroidGraphicsMatrix(transformMatrix));
>+      final VideoFrame frame = new VideoFrame(buffer, frameRotation, timestampNs);
>+      ((VideoSink) listener).onFrame(frame);
>+      frame.release();
>+    }
>   }
> 
>   private void release() {
>@@ -299,9 +324,7 @@ public class SurfaceTextureHelper {
>     if (isTextureInUse || !isQuitting) {
>       throw new IllegalStateException("Unexpected release.");
>     }
>-    if (yuvConverter != null) {
>-      yuvConverter.release();
>-    }
>+    yuvConverter.release();
>     GLES20.glDeleteTextures(1, new int[] {oesTextureId}, 0);
>     surfaceTexture.release();
>     eglBase.release();
>@@ -315,14 +338,12 @@ public class SurfaceTextureHelper {
>    *
>    * The returned TextureBuffer holds a reference to the SurfaceTextureHelper that created it. The
>    * buffer calls returnTextureFrame() when it is released.
>+   *
>+   * @deprecated Use a VideoSink as listener instead.
>    */
>-  public TextureBuffer createTextureBuffer(int width, int height, Matrix transformMatrix) {
>-    return new TextureBufferImpl(
>-        width, height, TextureBuffer.Type.OES, oesTextureId, transformMatrix, this, new Runnable() {
>-          @Override
>-          public void run() {
>-            returnTextureFrame();
>-          }
>-        });
>+  @Deprecated
>+  public TextureBufferImpl createTextureBuffer(int width, int height, Matrix transformMatrix) {
>+    return new TextureBufferImpl(width, height, TextureBuffer.Type.OES, oesTextureId,
>+        transformMatrix, handler, yuvConverter, this ::returnTextureFrame);
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/SurfaceViewRenderer.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/SurfaceViewRenderer.java
>index a8eb57b8e3a..c39416c3e14 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/SurfaceViewRenderer.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/SurfaceViewRenderer.java
>@@ -21,9 +21,8 @@ import android.view.SurfaceView;
> /**
>  * Display the video stream on a SurfaceView.
>  */
>-public class SurfaceViewRenderer extends SurfaceView implements SurfaceHolder.Callback,
>-                                                                VideoRenderer.Callbacks, VideoSink,
>-                                                                RendererCommon.RendererEvents {
>+public class SurfaceViewRenderer extends SurfaceView
>+    implements SurfaceHolder.Callback, VideoSink, RendererCommon.RendererEvents {
>   private static final String TAG = "SurfaceViewRenderer";
> 
>   // Cached resource name.
>@@ -180,12 +179,6 @@ public class SurfaceViewRenderer extends SurfaceView implements SurfaceHolder.Ca
>     eglRenderer.pauseVideo();
>   }
> 
>-  // VideoRenderer.Callbacks interface.
>-  @Override
>-  public void renderFrame(VideoRenderer.I420Frame frame) {
>-    eglRenderer.renderFrame(frame);
>-  }
>-
>   // VideoSink interface.
>   @Override
>   public void onFrame(VideoFrame frame) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/TextureBufferImpl.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/TextureBufferImpl.java
>new file mode 100644
>index 00000000000..96d7d4380e7
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/TextureBufferImpl.java
>@@ -0,0 +1,110 @@
>+/*
>+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc;
>+
>+import android.graphics.Matrix;
>+import javax.annotation.Nullable;
>+import android.os.Handler;
>+
>+/**
>+ * Android texture buffer that glues together the necessary information together with a generic
>+ * release callback. ToI420() is implemented by providing a Handler and a YuvConverter.
>+ */
>+public class TextureBufferImpl implements VideoFrame.TextureBuffer {
>+  private final int width;
>+  private final int height;
>+  private final Type type;
>+  private final int id;
>+  private final Matrix transformMatrix;
>+  private final Handler toI420Handler;
>+  private final YuvConverter yuvConverter;
>+  private final RefCountDelegate refCountDelegate;
>+
>+  public TextureBufferImpl(int width, int height, Type type, int id, Matrix transformMatrix,
>+      Handler toI420Handler, YuvConverter yuvConverter, @Nullable Runnable releaseCallback) {
>+    this.width = width;
>+    this.height = height;
>+    this.type = type;
>+    this.id = id;
>+    this.transformMatrix = transformMatrix;
>+    this.toI420Handler = toI420Handler;
>+    this.yuvConverter = yuvConverter;
>+    this.refCountDelegate = new RefCountDelegate(releaseCallback);
>+  }
>+
>+  @Override
>+  public VideoFrame.TextureBuffer.Type getType() {
>+    return type;
>+  }
>+
>+  @Override
>+  public int getTextureId() {
>+    return id;
>+  }
>+
>+  @Override
>+  public Matrix getTransformMatrix() {
>+    return transformMatrix;
>+  }
>+
>+  @Override
>+  public int getWidth() {
>+    return width;
>+  }
>+
>+  @Override
>+  public int getHeight() {
>+    return height;
>+  }
>+
>+  @Override
>+  public VideoFrame.I420Buffer toI420() {
>+    return ThreadUtils.invokeAtFrontUninterruptibly(
>+        toI420Handler, () -> yuvConverter.convert(this));
>+  }
>+
>+  @Override
>+  public void retain() {
>+    refCountDelegate.retain();
>+  }
>+
>+  @Override
>+  public void release() {
>+    refCountDelegate.release();
>+  }
>+
>+  @Override
>+  public VideoFrame.Buffer cropAndScale(
>+      int cropX, int cropY, int cropWidth, int cropHeight, int scaleWidth, int scaleHeight) {
>+    final Matrix cropAndScaleMatrix = new Matrix();
>+    // In WebRTC, Y=0 is the top row, while in OpenGL Y=0 is the bottom row. This means that the Y
>+    // direction is effectively reversed.
>+    final int cropYFromBottom = height - (cropY + cropHeight);
>+    cropAndScaleMatrix.preTranslate(cropX / (float) width, cropYFromBottom / (float) height);
>+    cropAndScaleMatrix.preScale(cropWidth / (float) width, cropHeight / (float) height);
>+
>+    return applyTransformMatrix(cropAndScaleMatrix, scaleWidth, scaleHeight);
>+  }
>+
>+  /**
>+   * Create a new TextureBufferImpl with an applied transform matrix and a new size. The
>+   * existing buffer is unchanged. The given transform matrix is applied first when texture
>+   * coordinates are still in the unmodified [0, 1] range.
>+   */
>+  public TextureBufferImpl applyTransformMatrix(
>+      Matrix transformMatrix, int newWidth, int newHeight) {
>+    final Matrix newMatrix = new Matrix(this.transformMatrix);
>+    newMatrix.preConcat(transformMatrix);
>+    retain();
>+    return new TextureBufferImpl(
>+        newWidth, newHeight, type, id, newMatrix, toI420Handler, yuvConverter, this ::release);
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/TurnCustomizer.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/TurnCustomizer.java
>index 7c68bd1e771..bfc72ea2d09 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/TurnCustomizer.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/TurnCustomizer.java
>@@ -22,6 +22,10 @@ public class TurnCustomizer {
>     nativeFreeTurnCustomizer(nativeTurnCustomizer);
>   }
> 
>-  private static native void nativeFreeTurnCustomizer(
>-      long nativeTurnCustomizer);
>+  private static native void nativeFreeTurnCustomizer(long turnCustomizer);
>+
>+  @CalledByNative
>+  long getNativeTurnCustomizer() {
>+    return nativeTurnCustomizer;
>+  }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoCapturer.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoCapturer.java
>index 804560a3c37..f5684fc5cc6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoCapturer.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoCapturer.java
>@@ -15,40 +15,23 @@ import java.util.List;
> 
> // Base interface for all VideoCapturers to implement.
> public interface VideoCapturer {
>-  // Interface used for providing callbacks to an observer.
>-  public interface CapturerObserver {
>-    // Notify if the camera have been started successfully or not.
>-    // Called on a Java thread owned by VideoCapturer.
>-    void onCapturerStarted(boolean success);
>-    void onCapturerStopped();
>-
>-    // Delivers a captured frame. Called on a Java thread owned by VideoCapturer.
>-    void onByteBufferFrameCaptured(
>-        byte[] data, int width, int height, int rotation, long timeStamp);
>-
>-    // Delivers a captured frame in a texture with id |oesTextureId|. Called on a Java thread
>-    // owned by VideoCapturer.
>-    void onTextureFrameCaptured(int width, int height, int oesTextureId, float[] transformMatrix,
>-        int rotation, long timestamp);
>-
>-    // Delivers a captured frame. Called on a Java thread owned by VideoCapturer.
>-    void onFrameCaptured(VideoFrame frame);
>-  }
>-
>   /**
>    * This function is used to initialize the camera thread, the android application context, and the
>    * capture observer. It will be called only once and before any startCapture() request. The
>    * camera thread is guaranteed to be valid until dispose() is called. If the VideoCapturer wants
>    * to deliver texture frames, it should do this by rendering on the SurfaceTexture in
>-   * |surfaceTextureHelper|, register itself as a listener, and forward the texture frames to
>-   * CapturerObserver.onTextureFrameCaptured().
>+   * {@code surfaceTextureHelper}, register itself as a listener, and forward the frames to
>+   * CapturerObserver.onFrameCaptured(). The caller still has ownership of {@code
>+   * surfaceTextureHelper} and is responsible for making sure surfaceTextureHelper.dispose() is
>+   * called. This also means that the caller can reuse the SurfaceTextureHelper to initialize a new
>+   * VideoCapturer once the previous VideoCapturer has been disposed.
>    */
>   void initialize(SurfaceTextureHelper surfaceTextureHelper, Context applicationContext,
>       CapturerObserver capturerObserver);
> 
>   /**
>-   * Start capturing frames in a format that is as close as possible to |width| x |height| and
>-   * |framerate|.
>+   * Start capturing frames in a format that is as close as possible to {@code width x height} and
>+   * {@code framerate}.
>    */
>   void startCapture(int width, int height, int framerate);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoCodecInfo.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoCodecInfo.java
>index 9f52cf25d05..3ee51f0f1b4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoCodecInfo.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoCodecInfo.java
>@@ -10,6 +10,7 @@
> 
> package org.webrtc;
> 
>+import javax.annotation.Nullable;
> import java.util.Arrays;
> import java.util.Locale;
> import java.util.Map;
>@@ -35,6 +36,7 @@ public class VideoCodecInfo {
>   public final Map<String, String> params;
>   @Deprecated public final int payload;
> 
>+  @CalledByNative
>   public VideoCodecInfo(String name, Map<String, String> params) {
>     this.payload = 0;
>     this.name = name;
>@@ -49,7 +51,7 @@ public class VideoCodecInfo {
>   }
> 
>   @Override
>-  public boolean equals(Object obj) {
>+  public boolean equals(@Nullable Object obj) {
>     if (obj == null)
>       return false;
>     if (obj == this)
>@@ -66,4 +68,14 @@ public class VideoCodecInfo {
>     Object[] values = {name.toUpperCase(Locale.ROOT), params};
>     return Arrays.hashCode(values);
>   }
>+
>+  @CalledByNative
>+  String getName() {
>+    return name;
>+  }
>+
>+  @CalledByNative
>+  Map getParams() {
>+    return params;
>+  }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoDecoder.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoDecoder.java
>index 7bf9ac54d7c..879b694b28f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoDecoder.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoDecoder.java
>@@ -51,6 +51,28 @@ public interface VideoDecoder {
>     void onDecodedFrame(VideoFrame frame, Integer decodeTimeMs, Integer qp);
>   }
> 
>+  /**
>+   * The decoder implementation backing this interface is either 1) a Java
>+   * decoder (e.g., an Android platform decoder), or alternatively 2) a native
>+   * decoder (e.g., a software decoder or a C++ decoder adapter).
>+   *
>+   * For case 1), createNativeVideoDecoder() should return zero.
>+   * In this case, we expect the native library to call the decoder through
>+   * JNI using the Java interface declared below.
>+   *
>+   * For case 2), createNativeVideoDecoder() should return a non-zero value.
>+   * In this case, we expect the native library to treat the returned value as
>+   * a raw pointer of type webrtc::VideoDecoder* (ownership is transferred to
>+   * the caller). The native library should then directly call the
>+   * webrtc::VideoDecoder interface without going through JNI. All calls to
>+   * the Java interface methods declared below should thus throw an
>+   * UnsupportedOperationException.
>+   */
>+  @CalledByNative
>+  default long createNativeVideoDecoder() {
>+    return 0;
>+  }
>+
>   /**
>    * Initializes the decoding process with specified settings. Will be called on the decoding thread
>    * before any decode calls.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoDecoderFactory.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoDecoderFactory.java
>index 1c01503f6de..c54f61d9a3b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoDecoderFactory.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoDecoderFactory.java
>@@ -10,11 +10,32 @@
> 
> package org.webrtc;
> 
>+import javax.annotation.Nullable;
>+
> /** Factory for creating VideoDecoders. */
> public interface VideoDecoderFactory {
>   /**
>    * Creates a VideoDecoder for the given codec. Supports the same codecs supported by
>    * VideoEncoderFactory.
>    */
>-  public VideoDecoder createDecoder(String codecType);
>+  @Deprecated
>+  @Nullable
>+  default VideoDecoder createDecoder(String codecType) {
>+    throw new UnsupportedOperationException("Deprecated and not implemented.");
>+  }
>+
>+  /** Creates a decoder for the given video codec. */
>+  @Nullable
>+  @CalledByNative
>+  default VideoDecoder createDecoder(VideoCodecInfo info) {
>+    return createDecoder(info.getName());
>+  }
>+
>+  /**
>+   * Enumerates the list of supported video codecs.
>+   */
>+  @CalledByNative
>+  default VideoCodecInfo[] getSupportedCodecs() {
>+    return new VideoCodecInfo[0];
>+  }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoDecoderFallback.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoDecoderFallback.java
>index 67e746f83c0..ddfa3ecd40f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoDecoderFallback.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoDecoderFallback.java
>@@ -23,9 +23,9 @@ public class VideoDecoderFallback extends WrappedNativeVideoDecoder {
>   }
> 
>   @Override
>-  long createNativeDecoder() {
>-    return createNativeDecoder(fallback, primary);
>+  public long createNativeVideoDecoder() {
>+    return nativeCreateDecoder(fallback, primary);
>   }
> 
>-  private static native long createNativeDecoder(VideoDecoder fallback, VideoDecoder primary);
>+  private static native long nativeCreateDecoder(VideoDecoder fallback, VideoDecoder primary);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoEncoder.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoEncoder.java
>index 4f97fb0c0d1..b739e99aa6b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoEncoder.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoEncoder.java
>@@ -10,6 +10,7 @@
> 
> package org.webrtc;
> 
>+import javax.annotation.Nullable;
> import org.webrtc.EncodedImage;
> 
> /**
>@@ -93,14 +94,41 @@ public interface VideoEncoder {
>   /** Settings for WebRTC quality based scaling. */
>   public class ScalingSettings {
>     public final boolean on;
>-    public final Integer low;
>-    public final Integer high;
>+    @Nullable public final Integer low;
>+    @Nullable public final Integer high;
> 
>+    /**
>+     * Settings to disable quality based scaling.
>+     */
>+    public static final ScalingSettings OFF = new ScalingSettings();
>+
>+    /**
>+     * Creates settings to enable quality based scaling.
>+     *
>+     * @param low Average QP at which to scale up the resolution.
>+     * @param high Average QP at which to scale down the resolution.
>+     */
>+    public ScalingSettings(int low, int high) {
>+      this.on = true;
>+      this.low = low;
>+      this.high = high;
>+    }
>+
>+    private ScalingSettings() {
>+      this.on = false;
>+      this.low = null;
>+      this.high = null;
>+    }
>+
>+    // TODO(bugs.webrtc.org/8830): Below constructors are deprecated.
>+    // Default thresholds are going away, so thresholds have to be set
>+    // when scaling is on.
>     /**
>      * Creates quality based scaling setting.
>      *
>      * @param on True if quality scaling is turned on.
>      */
>+    @Deprecated
>     public ScalingSettings(boolean on) {
>       this.on = on;
>       this.low = null;
>@@ -114,18 +142,57 @@ public interface VideoEncoder {
>      * @param low Average QP at which to scale up the resolution.
>      * @param high Average QP at which to scale down the resolution.
>      */
>+    @Deprecated
>     public ScalingSettings(boolean on, int low, int high) {
>       this.on = on;
>       this.low = low;
>       this.high = high;
>     }
>+
>+    @Override
>+    public String toString() {
>+      return on ? "[ " + low + ", " + high + " ]" : "OFF";
>+    }
>   }
> 
>   public interface Callback {
>-    /** Call to return an encoded frame. */
>+    /**
>+     * Call to return an encoded frame. It is safe to assume the byte buffer held by |frame| is not
>+     * accessed after the call to this method returns.
>+     */
>     void onEncodedFrame(EncodedImage frame, CodecSpecificInfo info);
>   }
> 
>+  /**
>+   * The encoder implementation backing this interface is either 1) a Java
>+   * encoder (e.g., an Android platform encoder), or alternatively 2) a native
>+   * encoder (e.g., a software encoder or a C++ encoder adapter).
>+   *
>+   * For case 1), createNativeVideoEncoder() should return zero.
>+   * In this case, we expect the native library to call the encoder through
>+   * JNI using the Java interface declared below.
>+   *
>+   * For case 2), createNativeVideoEncoder() should return a non-zero value.
>+   * In this case, we expect the native library to treat the returned value as
>+   * a raw pointer of type webrtc::VideoEncoder* (ownership is transferred to
>+   * the caller). The native library should then directly call the
>+   * webrtc::VideoEncoder interface without going through JNI. All calls to
>+   * the Java interface methods declared below should thus throw an
>+   * UnsupportedOperationException.
>+   */
>+  @CalledByNative
>+  default long createNativeVideoEncoder() {
>+    return 0;
>+  }
>+
>+  /**
>+   * Returns true if the encoder is backed by hardware.
>+   */
>+  @CalledByNative
>+  default boolean isHardwareEncoder() {
>+    return true;
>+  }
>+
>   /**
>    * Initializes the encoding process. Call before any calls to encode.
>    */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoEncoderFactory.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoEncoderFactory.java
>index 4445524a99c..fdf13093b64 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoEncoderFactory.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoEncoderFactory.java
>@@ -10,14 +10,16 @@
> 
> package org.webrtc;
> 
>+import javax.annotation.Nullable;
>+
> /** Factory for creating VideoEncoders. */
> public interface VideoEncoderFactory {
>   /** Creates an encoder for the given video codec. */
>-  public VideoEncoder createEncoder(VideoCodecInfo info);
>+  @Nullable @CalledByNative VideoEncoder createEncoder(VideoCodecInfo info);
> 
>   /**
>    * Enumerates the list of supported video codecs. This method will only be called once and the
>    * result will be cached.
>    */
>-  public VideoCodecInfo[] getSupportedCodecs();
>+  @CalledByNative VideoCodecInfo[] getSupportedCodecs();
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoEncoderFallback.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoEncoderFallback.java
>index 7894da9026b..fa36b7c9894 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoEncoderFallback.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoEncoderFallback.java
>@@ -23,14 +23,14 @@ public class VideoEncoderFallback extends WrappedNativeVideoEncoder {
>   }
> 
>   @Override
>-  long createNativeEncoder() {
>-    return createNativeEncoder(fallback, primary);
>+  public long createNativeVideoEncoder() {
>+    return nativeCreateEncoder(fallback, primary);
>   }
> 
>   @Override
>-  boolean isSoftwareEncoder() {
>-    return isWrappedSoftwareEncoder(primary);
>+  public boolean isHardwareEncoder() {
>+    return primary.isHardwareEncoder();
>   }
> 
>-  private static native long createNativeEncoder(VideoEncoder fallback, VideoEncoder primary);
>+  private static native long nativeCreateEncoder(VideoEncoder fallback, VideoEncoder primary);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoFileRenderer.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoFileRenderer.java
>index 06b433c47d8..f88687150d6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoFileRenderer.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoFileRenderer.java
>@@ -17,17 +17,20 @@ import java.io.IOException;
> import java.nio.ByteBuffer;
> import java.nio.charset.Charset;
> import java.util.ArrayList;
>+import java.util.concurrent.BlockingQueue;
> import java.util.concurrent.CountDownLatch;
>+import java.util.concurrent.LinkedBlockingQueue;
> 
> /**
>  * Can be used to save the video frames to file.
>  */
>-public class VideoFileRenderer implements VideoRenderer.Callbacks {
>+public class VideoFileRenderer implements VideoSink {
>   private static final String TAG = "VideoFileRenderer";
> 
>   private final HandlerThread renderThread;
>-  private final Object handlerLock = new Object();
>   private final Handler renderThreadHandler;
>+  private final HandlerThread fileThread;
>+  private final Handler fileThreadHandler;
>   private final FileOutputStream videoOutFile;
>   private final String outputFileName;
>   private final int outputFileWidth;
>@@ -36,7 +39,7 @@ public class VideoFileRenderer implements VideoRenderer.Callbacks {
>   private final ByteBuffer outputFrameBuffer;
>   private EglBase eglBase;
>   private YuvConverter yuvConverter;
>-  private ArrayList<ByteBuffer> rawFrames = new ArrayList<>();
>+  private int frameCount;
> 
>   public VideoFileRenderer(String outputFile, int outputFileWidth, int outputFileHeight,
>       final EglBase.Context sharedContext) throws IOException {
>@@ -56,10 +59,14 @@ public class VideoFileRenderer implements VideoRenderer.Callbacks {
>         ("YUV4MPEG2 C420 W" + outputFileWidth + " H" + outputFileHeight + " Ip F30:1 A1:1\n")
>             .getBytes(Charset.forName("US-ASCII")));
> 
>-    renderThread = new HandlerThread(TAG);
>+    renderThread = new HandlerThread(TAG + "RenderThread");
>     renderThread.start();
>     renderThreadHandler = new Handler(renderThread.getLooper());
> 
>+    fileThread = new HandlerThread(TAG + "FileThread");
>+    fileThread.start();
>+    fileThreadHandler = new Handler(fileThread.getLooper());
>+
>     ThreadUtils.invokeAtFrontUninterruptibly(renderThreadHandler, new Runnable() {
>       @Override
>       public void run() {
>@@ -72,61 +79,56 @@ public class VideoFileRenderer implements VideoRenderer.Callbacks {
>   }
> 
>   @Override
>-  public void renderFrame(final VideoRenderer.I420Frame frame) {
>-    renderThreadHandler.post(new Runnable() {
>-      @Override
>-      public void run() {
>-        renderFrameOnRenderThread(frame);
>-      }
>-    });
>+  public void onFrame(VideoFrame frame) {
>+    frame.retain();
>+    renderThreadHandler.post(() -> renderFrameOnRenderThread(frame));
>   }
> 
>-  // TODO(sakal): yuvConverter.convert is deprecated. This will be removed once this file is updated
>-  // to implement VideoSink instead of VideoRenderer.Callbacks.
>-  @SuppressWarnings("deprecation")
>-  private void renderFrameOnRenderThread(VideoRenderer.I420Frame frame) {
>-    final float frameAspectRatio = (float) frame.rotatedWidth() / (float) frame.rotatedHeight();
>+  private void renderFrameOnRenderThread(VideoFrame frame) {
>+    final VideoFrame.Buffer buffer = frame.getBuffer();
> 
>-    final float[] rotatedSamplingMatrix =
>-        RendererCommon.rotateTextureMatrix(frame.samplingMatrix, frame.rotationDegree);
>-    final float[] layoutMatrix = RendererCommon.getLayoutMatrix(
>-        false, frameAspectRatio, (float) outputFileWidth / outputFileHeight);
>-    final float[] texMatrix = RendererCommon.multiplyMatrices(rotatedSamplingMatrix, layoutMatrix);
>+    // If the frame is rotated, it will be applied after cropAndScale. Therefore, if the frame is
>+    // rotated by 90 degrees, swap width and height.
>+    final int targetWidth = frame.getRotation() % 180 == 0 ? outputFileWidth : outputFileHeight;
>+    final int targetHeight = frame.getRotation() % 180 == 0 ? outputFileHeight : outputFileWidth;
> 
>-    try {
>-      ByteBuffer buffer = JniCommon.allocateNativeByteBuffer(outputFrameSize);
>-      if (!frame.yuvFrame) {
>-        yuvConverter.convert(outputFrameBuffer, outputFileWidth, outputFileHeight, outputFileWidth,
>-            frame.textureId, texMatrix);
>-
>-        int stride = outputFileWidth;
>-        byte[] data = outputFrameBuffer.array();
>-        int offset = outputFrameBuffer.arrayOffset();
>-
>-        // Write Y
>-        buffer.put(data, offset, outputFileWidth * outputFileHeight);
>-
>-        // Write U
>-        for (int r = outputFileHeight; r < outputFileHeight * 3 / 2; ++r) {
>-          buffer.put(data, offset + r * stride, stride / 2);
>-        }
>-
>-        // Write V
>-        for (int r = outputFileHeight; r < outputFileHeight * 3 / 2; ++r) {
>-          buffer.put(data, offset + r * stride + stride / 2, stride / 2);
>-        }
>-      } else {
>-        nativeI420Scale(frame.yuvPlanes[0], frame.yuvStrides[0], frame.yuvPlanes[1],
>-            frame.yuvStrides[1], frame.yuvPlanes[2], frame.yuvStrides[2], frame.width, frame.height,
>-            outputFrameBuffer, outputFileWidth, outputFileHeight);
>-
>-        buffer.put(outputFrameBuffer.array(), outputFrameBuffer.arrayOffset(), outputFrameSize);
>-      }
>-      buffer.rewind();
>-      rawFrames.add(buffer);
>-    } finally {
>-      VideoRenderer.renderFrameDone(frame);
>+    final float frameAspectRatio = (float) buffer.getWidth() / (float) buffer.getHeight();
>+    final float fileAspectRatio = (float) targetWidth / (float) targetHeight;
>+
>+    // Calculate cropping to equalize the aspect ratio.
>+    int cropWidth = buffer.getWidth();
>+    int cropHeight = buffer.getHeight();
>+    if (fileAspectRatio > frameAspectRatio) {
>+      cropHeight = (int) (cropHeight * (frameAspectRatio / fileAspectRatio));
>+    } else {
>+      cropWidth = (int) (cropWidth * (fileAspectRatio / frameAspectRatio));
>     }
>+
>+    final int cropX = (buffer.getWidth() - cropWidth) / 2;
>+    final int cropY = (buffer.getHeight() - cropHeight) / 2;
>+
>+    final VideoFrame.Buffer scaledBuffer =
>+        buffer.cropAndScale(cropX, cropY, cropWidth, cropHeight, targetWidth, targetHeight);
>+    frame.release();
>+
>+    final VideoFrame.I420Buffer i420 = scaledBuffer.toI420();
>+    scaledBuffer.release();
>+
>+    fileThreadHandler.post(() -> {
>+      YuvHelper.I420Rotate(i420.getDataY(), i420.getStrideY(), i420.getDataU(), i420.getStrideU(),
>+          i420.getDataV(), i420.getStrideV(), outputFrameBuffer, i420.getWidth(), i420.getHeight(),
>+          frame.getRotation());
>+      i420.release();
>+
>+      try {
>+        videoOutFile.write("FRAME\n".getBytes(Charset.forName("US-ASCII")));
>+        videoOutFile.write(
>+            outputFrameBuffer.array(), outputFrameBuffer.arrayOffset(), outputFrameSize);
>+      } catch (IOException e) {
>+        throw new RuntimeException("Error writing video to disk", e);
>+      }
>+      frameCount++;
>+    });
>   }
> 
>   /**
>@@ -134,37 +136,30 @@ public class VideoFileRenderer implements VideoRenderer.Callbacks {
>    */
>   public void release() {
>     final CountDownLatch cleanupBarrier = new CountDownLatch(1);
>-    renderThreadHandler.post(new Runnable() {
>-      @Override
>-      public void run() {
>-        yuvConverter.release();
>-        eglBase.release();
>-        renderThread.quit();
>-        cleanupBarrier.countDown();
>-      }
>+    renderThreadHandler.post(() -> {
>+      yuvConverter.release();
>+      eglBase.release();
>+      renderThread.quit();
>+      cleanupBarrier.countDown();
>     });
>     ThreadUtils.awaitUninterruptibly(cleanupBarrier);
>-    try {
>-      for (ByteBuffer buffer : rawFrames) {
>-        videoOutFile.write("FRAME\n".getBytes(Charset.forName("US-ASCII")));
>-
>-        byte[] data = new byte[outputFrameSize];
>-        buffer.get(data);
>-
>-        videoOutFile.write(data);
>-
>-        JniCommon.freeNativeByteBuffer(buffer);
>+    fileThreadHandler.post(() -> {
>+      try {
>+        videoOutFile.close();
>+        Logging.d(TAG,
>+            "Video written to disk as " + outputFileName + ". The number of frames is " + frameCount
>+                + " and the dimensions of the frames are " + outputFileWidth + "x"
>+                + outputFileHeight + ".");
>+      } catch (IOException e) {
>+        throw new RuntimeException("Error closing output file", e);
>       }
>-      videoOutFile.close();
>-      Logging.d(TAG, "Video written to disk as " + outputFileName + ". Number frames are "
>-              + rawFrames.size() + " and the dimension of the frames are " + outputFileWidth + "x"
>-              + outputFileHeight + ".");
>-    } catch (IOException e) {
>-      Logging.e(TAG, "Error writing video to disk", e);
>+      fileThread.quit();
>+    });
>+    try {
>+      fileThread.join();
>+    } catch (InterruptedException e) {
>+      Thread.currentThread().interrupt();
>+      Logging.e(TAG, "Interrupted while waiting for the write to disk to complete.", e);
>     }
>   }
>-
>-  public static native void nativeI420Scale(ByteBuffer srcY, int strideY, ByteBuffer srcU,
>-      int strideU, ByteBuffer srcV, int strideV, int width, int height, ByteBuffer dst,
>-      int dstWidth, int dstHeight);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoFrame.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoFrame.java
>index 3ee368d2ea8..fb733af7db0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoFrame.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoFrame.java
>@@ -14,6 +14,7 @@ import android.graphics.Matrix;
> import android.opengl.GLES11Ext;
> import android.opengl.GLES20;
> import java.nio.ByteBuffer;
>+import javax.annotation.Nullable;
> 
> /**
>  * Java version of webrtc::VideoFrame and webrtc::VideoFrameBuffer. A difference from the C++
>@@ -25,8 +26,15 @@ import java.nio.ByteBuffer;
>  * format and serves as a fallback for video sinks that can only handle I420, e.g. the internal
>  * WebRTC software encoders.
>  */
>-public class VideoFrame {
>-  public interface Buffer {
>+public class VideoFrame implements RefCounted {
>+  /**
>+   * Implements image storage medium. Might be for example an OpenGL texture or a memory region
>+   * containing I420-data.
>+   *
>+   * <p>Reference counting is needed since a video buffer can be shared between multiple VideoSinks,
>+   * and the buffer needs to be returned to the VideoSource as soon as all references are gone.
>+   */
>+  public interface Buffer extends RefCounted {
>     /**
>      * Resolution of the buffer in pixels.
>      */
>@@ -40,12 +48,8 @@ public class VideoFrame {
>      */
>     @CalledByNative("Buffer") I420Buffer toI420();
> 
>-    /**
>-     * Reference counting is needed since a video buffer can be shared between multiple VideoSinks,
>-     * and the buffer needs to be returned to the VideoSource as soon as all references are gone.
>-     */
>-    @CalledByNative("Buffer") void retain();
>-    @CalledByNative("Buffer") void release();
>+    @Override @CalledByNative("Buffer") void retain();
>+    @Override @CalledByNative("Buffer") void release();
> 
>     /**
>      * Crops a region defined by |cropx|, |cropY|, |cropWidth| and |cropHeight|. Scales it to size
>@@ -114,13 +118,18 @@ public class VideoFrame {
>      * homogeneous coordinates of the form (s, t, 1) with s and t in the inclusive range [0, 1] to
>      * the coordinate that should be used to sample that location from the buffer.
>      */
>-    public Matrix getTransformMatrix();
>+    Matrix getTransformMatrix();
>   }
> 
>   private final Buffer buffer;
>   private final int rotation;
>   private final long timestampNs;
> 
>+  /**
>+   * Constructs a new VideoFrame backed by the given {@code buffer}.
>+   *
>+   * @note Ownership of the buffer object is tranferred to the new VideoFrame.
>+   */
>   @CalledByNative
>   public VideoFrame(Buffer buffer, int rotation, long timestampNs) {
>     if (buffer == null) {
>@@ -169,46 +178,14 @@ public class VideoFrame {
>     return buffer.getWidth();
>   }
> 
>-  /**
>-   * Reference counting of the underlying buffer.
>-   */
>+  @Override
>   public void retain() {
>     buffer.retain();
>   }
> 
>+  @Override
>+  @CalledByNative
>   public void release() {
>     buffer.release();
>   }
>-
>-  public static VideoFrame.Buffer cropAndScaleI420(final I420Buffer buffer, int cropX, int cropY,
>-      int cropWidth, int cropHeight, int scaleWidth, int scaleHeight) {
>-    if (cropWidth == scaleWidth && cropHeight == scaleHeight) {
>-      // No scaling.
>-      ByteBuffer dataY = buffer.getDataY();
>-      ByteBuffer dataU = buffer.getDataU();
>-      ByteBuffer dataV = buffer.getDataV();
>-
>-      dataY.position(cropX + cropY * buffer.getStrideY());
>-      dataU.position(cropX / 2 + cropY / 2 * buffer.getStrideU());
>-      dataV.position(cropX / 2 + cropY / 2 * buffer.getStrideV());
>-
>-      buffer.retain();
>-      return JavaI420Buffer.wrap(buffer.getWidth(), buffer.getHeight(), dataY.slice(),
>-          buffer.getStrideY(), dataU.slice(), buffer.getStrideU(), dataV.slice(),
>-          buffer.getStrideV(), buffer::release);
>-    }
>-
>-    JavaI420Buffer newBuffer = JavaI420Buffer.allocate(scaleWidth, scaleHeight);
>-    cropAndScaleI420Native(buffer.getDataY(), buffer.getStrideY(), buffer.getDataU(),
>-        buffer.getStrideU(), buffer.getDataV(), buffer.getStrideV(), cropX, cropY, cropWidth,
>-        cropHeight, newBuffer.getDataY(), newBuffer.getStrideY(), newBuffer.getDataU(),
>-        newBuffer.getStrideU(), newBuffer.getDataV(), newBuffer.getStrideV(), scaleWidth,
>-        scaleHeight);
>-    return newBuffer;
>-  }
>-
>-  private static native void cropAndScaleI420Native(ByteBuffer srcY, int srcStrideY,
>-      ByteBuffer srcU, int srcStrideU, ByteBuffer srcV, int srcStrideV, int cropX, int cropY,
>-      int cropWidth, int cropHeight, ByteBuffer dstY, int dstStrideY, ByteBuffer dstU,
>-      int dstStrideU, ByteBuffer dstV, int dstStrideV, int scaleWidth, int scaleHeight);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoFrameDrawer.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoFrameDrawer.java
>index 491fd054c26..1e8a04d3f3c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoFrameDrawer.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoFrameDrawer.java
>@@ -14,6 +14,7 @@ import android.graphics.Matrix;
> import android.graphics.Point;
> import android.opengl.GLES20;
> import java.nio.ByteBuffer;
>+import javax.annotation.Nullable;
> 
> /**
>  * Helper class to draw VideoFrames. Calls either drawer.drawOes, drawer.drawRgb, or
>@@ -55,14 +56,15 @@ public class VideoFrameDrawer {
>     // Intermediate copy buffer for uploading yuv frames that are not packed, i.e. stride > width.
>     // TODO(magjed): Investigate when GL_UNPACK_ROW_LENGTH is available, or make a custom shader
>     // that handles stride and compare performance with intermediate copy.
>-    private ByteBuffer copyBuffer;
>-    private int[] yuvTextures;
>+    @Nullable private ByteBuffer copyBuffer;
>+    @Nullable private int[] yuvTextures;
> 
>     /**
>      * Upload |planes| into OpenGL textures, taking stride into consideration.
>      *
>      * @return Array of three texture indices corresponding to Y-, U-, and V-plane respectively.
>      */
>+    @Nullable
>     public int[] uploadYuvData(int width, int height, int[] strides, ByteBuffer[] planes) {
>       final int[] planeWidths = new int[] {width, width / 2, width / 2};
>       final int[] planeHeights = new int[] {height, height / 2, height / 2};
>@@ -95,8 +97,8 @@ public class VideoFrameDrawer {
>           // Input is packed already.
>           packedByteBuffer = planes[i];
>         } else {
>-          VideoRenderer.nativeCopyPlane(
>-              planes[i], planeWidths[i], planeHeights[i], strides[i], copyBuffer, planeWidths[i]);
>+          YuvHelper.copyPlane(
>+              planes[i], strides[i], copyBuffer, planeWidths[i], planeWidths[i], planeHeights[i]);
>           packedByteBuffer = copyBuffer;
>         }
>         GLES20.glTexImage2D(GLES20.GL_TEXTURE_2D, 0, GLES20.GL_LUMINANCE, planeWidths[i],
>@@ -105,12 +107,14 @@ public class VideoFrameDrawer {
>       return yuvTextures;
>     }
> 
>+    @Nullable
>     public int[] uploadFromBuffer(VideoFrame.I420Buffer buffer) {
>       int[] strides = {buffer.getStrideY(), buffer.getStrideU(), buffer.getStrideV()};
>       ByteBuffer[] planes = {buffer.getDataY(), buffer.getDataU(), buffer.getDataV()};
>       return uploadYuvData(buffer.getWidth(), buffer.getHeight(), strides, planes);
>     }
> 
>+    @Nullable
>     public int[] getYuvTextures() {
>       return yuvTextures;
>     }
>@@ -144,7 +148,7 @@ public class VideoFrameDrawer {
>   // |renderWidth| and |renderHeight| to avoid allocations since this function is called for every
>   // frame.
>   private void calculateTransformedRenderSize(
>-      int frameWidth, int frameHeight, Matrix renderMatrix) {
>+      int frameWidth, int frameHeight, @Nullable Matrix renderMatrix) {
>     if (renderMatrix == null) {
>       renderWidth = frameWidth;
>       renderHeight = frameHeight;
>@@ -167,7 +171,7 @@ public class VideoFrameDrawer {
>   private final YuvUploader yuvUploader = new YuvUploader();
>   // This variable will only be used for checking reference equality and is used for caching I420
>   // textures.
>-  private VideoFrame lastI420Frame;
>+  @Nullable private VideoFrame lastI420Frame;
>   private final Matrix renderMatrix = new Matrix();
> 
>   public void drawFrame(VideoFrame frame, RendererCommon.GlDrawer drawer) {
>@@ -181,7 +185,7 @@ public class VideoFrameDrawer {
>   }
> 
>   public void drawFrame(VideoFrame frame, RendererCommon.GlDrawer drawer,
>-      Matrix additionalRenderMatrix, int viewportX, int viewportY, int viewportWidth,
>+      @Nullable Matrix additionalRenderMatrix, int viewportX, int viewportY, int viewportWidth,
>       int viewportHeight) {
>     final int width = frame.getRotatedWidth();
>     final int height = frame.getRotatedHeight();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoRenderer.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoRenderer.java
>deleted file mode 100644
>index 182093c9321..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoRenderer.java
>+++ /dev/null
>@@ -1,223 +0,0 @@
>-/*
>- *  Copyright 2013 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-package org.webrtc;
>-
>-import java.nio.ByteBuffer;
>-
>-/**
>- * Java version of VideoSinkInterface.  In addition to allowing clients to
>- * define their own rendering behavior (by passing in a Callbacks object), this
>- * class also provides a createGui() method for creating a GUI-rendering window
>- * on various platforms.
>- */
>-public class VideoRenderer {
>-  /**
>-   * Java version of webrtc::VideoFrame. Frames are only constructed from native code and test
>-   * code.
>-   */
>-  public static class I420Frame {
>-    public final int width;
>-    public final int height;
>-    public final int[] yuvStrides;
>-    public ByteBuffer[] yuvPlanes;
>-    public final boolean yuvFrame;
>-    // Matrix that transforms standard coordinates to their proper sampling locations in
>-    // the texture. This transform compensates for any properties of the video source that
>-    // cause it to appear different from a normalized texture. This matrix does not take
>-    // |rotationDegree| into account.
>-    public final float[] samplingMatrix;
>-    public int textureId;
>-    // Frame pointer in C++.
>-    private long nativeFramePointer;
>-
>-    // rotationDegree is the degree that the frame must be rotated clockwisely
>-    // to be rendered correctly.
>-    public int rotationDegree;
>-
>-    // If this I420Frame was constructed from VideoFrame.Buffer, this points to
>-    // the backing buffer.
>-    private final VideoFrame.Buffer backingBuffer;
>-
>-    /**
>-     * Construct a frame of the given dimensions with the specified planar data.
>-     */
>-    public I420Frame(int width, int height, int rotationDegree, int[] yuvStrides,
>-        ByteBuffer[] yuvPlanes, long nativeFramePointer) {
>-      this.width = width;
>-      this.height = height;
>-      this.yuvStrides = yuvStrides;
>-      this.yuvPlanes = yuvPlanes;
>-      this.yuvFrame = true;
>-      this.rotationDegree = rotationDegree;
>-      this.nativeFramePointer = nativeFramePointer;
>-      backingBuffer = null;
>-      if (rotationDegree % 90 != 0) {
>-        throw new IllegalArgumentException("Rotation degree not multiple of 90: " + rotationDegree);
>-      }
>-      // The convention in WebRTC is that the first element in a ByteBuffer corresponds to the
>-      // top-left corner of the image, but in glTexImage2D() the first element corresponds to the
>-      // bottom-left corner. This discrepancy is corrected by setting a vertical flip as sampling
>-      // matrix.
>-      samplingMatrix = RendererCommon.verticalFlipMatrix();
>-    }
>-
>-    /**
>-     * Construct a texture frame of the given dimensions with data in SurfaceTexture
>-     */
>-    public I420Frame(int width, int height, int rotationDegree, int textureId,
>-        float[] samplingMatrix, long nativeFramePointer) {
>-      this.width = width;
>-      this.height = height;
>-      this.yuvStrides = null;
>-      this.yuvPlanes = null;
>-      this.samplingMatrix = samplingMatrix;
>-      this.textureId = textureId;
>-      this.yuvFrame = false;
>-      this.rotationDegree = rotationDegree;
>-      this.nativeFramePointer = nativeFramePointer;
>-      backingBuffer = null;
>-      if (rotationDegree % 90 != 0) {
>-        throw new IllegalArgumentException("Rotation degree not multiple of 90: " + rotationDegree);
>-      }
>-    }
>-
>-    /**
>-     * Construct a frame from VideoFrame.Buffer.
>-     */
>-    public I420Frame(int rotationDegree, VideoFrame.Buffer buffer, long nativeFramePointer) {
>-      this.width = buffer.getWidth();
>-      this.height = buffer.getHeight();
>-      this.rotationDegree = rotationDegree;
>-      if (rotationDegree % 90 != 0) {
>-        throw new IllegalArgumentException("Rotation degree not multiple of 90: " + rotationDegree);
>-      }
>-      if (buffer instanceof VideoFrame.TextureBuffer
>-          && ((VideoFrame.TextureBuffer) buffer).getType() == VideoFrame.TextureBuffer.Type.OES) {
>-        VideoFrame.TextureBuffer textureBuffer = (VideoFrame.TextureBuffer) buffer;
>-        this.yuvFrame = false;
>-        this.textureId = textureBuffer.getTextureId();
>-        this.samplingMatrix = RendererCommon.convertMatrixFromAndroidGraphicsMatrix(
>-            textureBuffer.getTransformMatrix());
>-
>-        this.yuvStrides = null;
>-        this.yuvPlanes = null;
>-      } else if (buffer instanceof VideoFrame.I420Buffer) {
>-        VideoFrame.I420Buffer i420Buffer = (VideoFrame.I420Buffer) buffer;
>-        this.yuvFrame = true;
>-        this.yuvStrides =
>-            new int[] {i420Buffer.getStrideY(), i420Buffer.getStrideU(), i420Buffer.getStrideV()};
>-        this.yuvPlanes =
>-            new ByteBuffer[] {i420Buffer.getDataY(), i420Buffer.getDataU(), i420Buffer.getDataV()};
>-        // The convention in WebRTC is that the first element in a ByteBuffer corresponds to the
>-        // top-left corner of the image, but in glTexImage2D() the first element corresponds to the
>-        // bottom-left corner. This discrepancy is corrected by multiplying the sampling matrix with
>-        // a vertical flip matrix.
>-        this.samplingMatrix = RendererCommon.verticalFlipMatrix();
>-
>-        this.textureId = 0;
>-      } else {
>-        this.yuvFrame = false;
>-        this.textureId = 0;
>-        this.samplingMatrix = null;
>-        this.yuvStrides = null;
>-        this.yuvPlanes = null;
>-      }
>-      this.nativeFramePointer = nativeFramePointer;
>-      backingBuffer = buffer;
>-    }
>-
>-    public int rotatedWidth() {
>-      return (rotationDegree % 180 == 0) ? width : height;
>-    }
>-
>-    public int rotatedHeight() {
>-      return (rotationDegree % 180 == 0) ? height : width;
>-    }
>-
>-    @Override
>-    public String toString() {
>-      final String type = yuvFrame
>-          ? "Y: " + yuvStrides[0] + ", U: " + yuvStrides[1] + ", V: " + yuvStrides[2]
>-          : "Texture: " + textureId;
>-      return width + "x" + height + ", " + type;
>-    }
>-
>-    /**
>-     * Convert the frame to VideoFrame. It is no longer safe to use the I420Frame after calling
>-     * this.
>-     */
>-    VideoFrame toVideoFrame() {
>-      final VideoFrame.Buffer buffer;
>-      if (backingBuffer != null) {
>-        // We were construted from a VideoFrame.Buffer, just return it.
>-        // Make sure webrtc::VideoFrame object is released.
>-        backingBuffer.retain();
>-        VideoRenderer.renderFrameDone(this);
>-        buffer = backingBuffer;
>-      } else if (yuvFrame) {
>-        buffer = JavaI420Buffer.wrap(width, height, yuvPlanes[0], yuvStrides[0], yuvPlanes[1],
>-            yuvStrides[1], yuvPlanes[2], yuvStrides[2],
>-            () -> { VideoRenderer.renderFrameDone(this); });
>-      } else {
>-        // Note: surfaceTextureHelper being null means calling toI420 will crash.
>-        buffer = new TextureBufferImpl(width, height, VideoFrame.TextureBuffer.Type.OES, textureId,
>-            RendererCommon.convertMatrixToAndroidGraphicsMatrix(samplingMatrix),
>-            null /* surfaceTextureHelper */, () -> { VideoRenderer.renderFrameDone(this); });
>-      }
>-      return new VideoFrame(buffer, rotationDegree, 0 /* timestampNs */);
>-    }
>-  }
>-
>-  // Helper native function to do a video frame plane copying.
>-  public static native void nativeCopyPlane(
>-      ByteBuffer src, int width, int height, int srcStride, ByteBuffer dst, int dstStride);
>-
>-  /** The real meat of VideoSinkInterface. */
>-  public static interface Callbacks {
>-    // |frame| might have pending rotation and implementation of Callbacks
>-    // should handle that by applying rotation during rendering. The callee
>-    // is responsible for signaling when it is done with |frame| by calling
>-    // renderFrameDone(frame).
>-    public void renderFrame(I420Frame frame);
>-  }
>-
>-  /**
>-   * This must be called after every renderFrame() to release the frame.
>-   */
>-  public static void renderFrameDone(I420Frame frame) {
>-    frame.yuvPlanes = null;
>-    frame.textureId = 0;
>-    if (frame.nativeFramePointer != 0) {
>-      releaseNativeFrame(frame.nativeFramePointer);
>-      frame.nativeFramePointer = 0;
>-    }
>-  }
>-
>-  long nativeVideoRenderer;
>-
>-  public VideoRenderer(Callbacks callbacks) {
>-    nativeVideoRenderer = nativeWrapVideoRenderer(callbacks);
>-  }
>-
>-  public void dispose() {
>-    if (nativeVideoRenderer == 0) {
>-      // Already disposed.
>-      return;
>-    }
>-
>-    freeWrappedVideoRenderer(nativeVideoRenderer);
>-    nativeVideoRenderer = 0;
>-  }
>-
>-  private static native long nativeWrapVideoRenderer(Callbacks callbacks);
>-  private static native void freeWrappedVideoRenderer(long nativeVideoRenderer);
>-  private static native void releaseNativeFrame(long nativeFramePointer);
>-}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoSource.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoSource.java
>index d9531cd9a5f..e0c61675bcb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoSource.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoSource.java
>@@ -10,12 +10,17 @@
> 
> package org.webrtc;
> 
>+import javax.annotation.Nullable;
>+
> /**
>  * Java wrapper of native AndroidVideoTrackSource.
>  */
> public class VideoSource extends MediaSource {
>+  private final NativeCapturerObserver capturerObserver;
>+
>   public VideoSource(long nativeSource) {
>     super(nativeSource);
>+    this.capturerObserver = new NativeCapturerObserver(nativeGetInternalSource(nativeSource));
>   }
> 
>   /**
>@@ -28,6 +33,11 @@ public class VideoSource extends MediaSource {
>     nativeAdaptOutputFormat(nativeSource, width, height, fps);
>   }
> 
>-  private static native void nativeAdaptOutputFormat(
>-      long nativeSource, int width, int height, int fps);
>+  public CapturerObserver getCapturerObserver() {
>+    return capturerObserver;
>+  }
>+
>+  // Returns source->internal() from webrtc::VideoTrackSourceProxy.
>+  private static native long nativeGetInternalSource(long source);
>+  private static native void nativeAdaptOutputFormat(long source, int width, int height, int fps);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoTrack.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoTrack.java
>index 7839a03998f..5495ee82b18 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoTrack.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/VideoTrack.java
>@@ -16,7 +16,6 @@ import java.util.List;
> 
> /** Java version of VideoTrackInterface. */
> public class VideoTrack extends MediaStreamTrack {
>-  private final List<VideoRenderer> renderers = new ArrayList<>();
>   private final IdentityHashMap<VideoSink, Long> sinks = new IdentityHashMap<VideoSink, Long>();
> 
>   public VideoTrack(long nativeTrack) {
>@@ -32,9 +31,16 @@ public class VideoTrack extends MediaStreamTrack {
>    * sources produce VideoFrames.
>    */
>   public void addSink(VideoSink sink) {
>-    final long nativeSink = nativeWrapSink(sink);
>-    sinks.put(sink, nativeSink);
>-    nativeAddSink(nativeTrack, nativeSink);
>+    if (sink == null) {
>+      throw new IllegalArgumentException("The VideoSink is not allowed to be null");
>+    }
>+    // We allow calling addSink() with the same sink multiple times. This is similar to the C++
>+    // VideoTrack::AddOrUpdateSink().
>+    if (!sinks.containsKey(sink)) {
>+      final long nativeSink = nativeWrapSink(sink);
>+      sinks.put(sink, nativeSink);
>+      nativeAddSink(nativeTrack, nativeSink);
>+    }
>   }
> 
>   /**
>@@ -43,33 +49,15 @@ public class VideoTrack extends MediaStreamTrack {
>    * If the VideoSink was not attached to the track, this is a no-op.
>    */
>   public void removeSink(VideoSink sink) {
>-    final long nativeSink = sinks.remove(sink);
>-    if (nativeSink != 0) {
>+    final Long nativeSink = sinks.remove(sink);
>+    if (nativeSink != null) {
>       nativeRemoveSink(nativeTrack, nativeSink);
>       nativeFreeSink(nativeSink);
>     }
>   }
> 
>-  public void addRenderer(VideoRenderer renderer) {
>-    renderers.add(renderer);
>-    nativeAddSink(nativeTrack, renderer.nativeVideoRenderer);
>-  }
>-
>-  public void removeRenderer(VideoRenderer renderer) {
>-    if (!renderers.remove(renderer)) {
>-      return;
>-    }
>-    nativeRemoveSink(nativeTrack, renderer.nativeVideoRenderer);
>-    renderer.dispose();
>-  }
>-
>   @Override
>   public void dispose() {
>-    for (VideoRenderer renderer : renderers) {
>-      nativeRemoveSink(nativeTrack, renderer.nativeVideoRenderer);
>-      renderer.dispose();
>-    }
>-    renderers.clear();
>     for (long nativeSink : sinks.values()) {
>       nativeRemoveSink(nativeTrack, nativeSink);
>       nativeFreeSink(nativeSink);
>@@ -78,9 +66,8 @@ public class VideoTrack extends MediaStreamTrack {
>     super.dispose();
>   }
> 
>-  private static native void nativeAddSink(long nativeTrack, long nativeSink);
>-  private static native void nativeRemoveSink(long nativeTrack, long nativeSink);
>-
>+  private static native void nativeAddSink(long track, long nativeSink);
>+  private static native void nativeRemoveSink(long track, long nativeSink);
>   private static native long nativeWrapSink(VideoSink sink);
>-  private static native void nativeFreeSink(long nativeSink);
>+  private static native void nativeFreeSink(long sink);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/YuvConverter.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/YuvConverter.java
>index 844ab58bfe0..de02c11c3bf 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/YuvConverter.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/YuvConverter.java
>@@ -10,57 +10,21 @@
> 
> package org.webrtc;
> 
>-import android.opengl.GLES11Ext;
>+import android.graphics.Matrix;
> import android.opengl.GLES20;
> import java.nio.ByteBuffer;
>-import java.nio.FloatBuffer;
> import org.webrtc.VideoFrame.I420Buffer;
> import org.webrtc.VideoFrame.TextureBuffer;
> 
> /**
>- * Class for converting OES textures to a YUV ByteBuffer. It should be constructed on a thread with
>- * an active EGL context, and only be used from that thread.
>+ * Class for converting OES textures to a YUV ByteBuffer. It can be constructed on any thread, but
>+ * should only be operated from a single thread with an active EGL context.
>  */
> public class YuvConverter {
>-  // Vertex coordinates in Normalized Device Coordinates, i.e.
>-  // (-1, -1) is bottom-left and (1, 1) is top-right.
>-  private static final FloatBuffer DEVICE_RECTANGLE = GlUtil.createFloatBuffer(new float[] {
>-      -1.0f, -1.0f, // Bottom left.
>-      1.0f, -1.0f, // Bottom right.
>-      -1.0f, 1.0f, // Top left.
>-      1.0f, 1.0f, // Top right.
>-  });
>-
>-  // Texture coordinates - (0, 0) is bottom-left and (1, 1) is top-right.
>-  private static final FloatBuffer TEXTURE_RECTANGLE = GlUtil.createFloatBuffer(new float[] {
>-      0.0f, 0.0f, // Bottom left.
>-      1.0f, 0.0f, // Bottom right.
>-      0.0f, 1.0f, // Top left.
>-      1.0f, 1.0f // Top right.
>-  });
>-
>-  // clang-format off
>-  private static final String VERTEX_SHADER =
>-        "varying vec2 interp_tc;\n"
>-      + "attribute vec4 in_pos;\n"
>-      + "attribute vec4 in_tc;\n"
>-      + "\n"
>-      + "uniform mat4 texMatrix;\n"
>-      + "\n"
>-      + "void main() {\n"
>-      + "    gl_Position = in_pos;\n"
>-      + "    interp_tc = (texMatrix * in_tc).xy;\n"
>-      + "}\n";
>-
>-  private static final String OES_FRAGMENT_SHADER =
>-        "#extension GL_OES_EGL_image_external : require\n"
>-      + "precision mediump float;\n"
>-      + "varying vec2 interp_tc;\n"
>-      + "\n"
>-      + "uniform samplerExternalOES tex;\n"
>+  private static final String FRAGMENT_SHADER =
>       // Difference in texture coordinate corresponding to one
>       // sub-pixel in the x direction.
>-      + "uniform vec2 xUnit;\n"
>+      "uniform vec2 xUnit;\n"
>       // Color conversion coefficients, including constant term
>       + "uniform vec4 coeffs;\n"
>       + "\n"
>@@ -72,152 +36,76 @@ public class YuvConverter {
>       // try to do it as a vec3 x mat3x4, followed by an add in of a
>       // constant vector.
>       + "  gl_FragColor.r = coeffs.a + dot(coeffs.rgb,\n"
>-      + "      texture2D(tex, interp_tc - 1.5 * xUnit).rgb);\n"
>+      + "      sample(tc - 1.5 * xUnit).rgb);\n"
>       + "  gl_FragColor.g = coeffs.a + dot(coeffs.rgb,\n"
>-      + "      texture2D(tex, interp_tc - 0.5 * xUnit).rgb);\n"
>+      + "      sample(tc - 0.5 * xUnit).rgb);\n"
>       + "  gl_FragColor.b = coeffs.a + dot(coeffs.rgb,\n"
>-      + "      texture2D(tex, interp_tc + 0.5 * xUnit).rgb);\n"
>+      + "      sample(tc + 0.5 * xUnit).rgb);\n"
>       + "  gl_FragColor.a = coeffs.a + dot(coeffs.rgb,\n"
>-      + "      texture2D(tex, interp_tc + 1.5 * xUnit).rgb);\n"
>+      + "      sample(tc + 1.5 * xUnit).rgb);\n"
>       + "}\n";
> 
>-  private static final String RGB_FRAGMENT_SHADER =
>-        "precision mediump float;\n"
>-      + "varying vec2 interp_tc;\n"
>-      + "\n"
>-      + "uniform sampler2D tex;\n"
>-      // Difference in texture coordinate corresponding to one
>-      // sub-pixel in the x direction.
>-      + "uniform vec2 xUnit;\n"
>-      // Color conversion coefficients, including constant term
>-      + "uniform vec4 coeffs;\n"
>-      + "\n"
>-      + "void main() {\n"
>-      // Since the alpha read from the texture is always 1, this could
>-      // be written as a mat4 x vec4 multiply. However, that seems to
>-      // give a worse framerate, possibly because the additional
>-      // multiplies by 1.0 consume resources. TODO(nisse): Could also
>-      // try to do it as a vec3 x mat3x4, followed by an add in of a
>-      // constant vector.
>-      + "  gl_FragColor.r = coeffs.a + dot(coeffs.rgb,\n"
>-      + "      texture2D(tex, interp_tc - 1.5 * xUnit).rgb);\n"
>-      + "  gl_FragColor.g = coeffs.a + dot(coeffs.rgb,\n"
>-      + "      texture2D(tex, interp_tc - 0.5 * xUnit).rgb);\n"
>-      + "  gl_FragColor.b = coeffs.a + dot(coeffs.rgb,\n"
>-      + "      texture2D(tex, interp_tc + 0.5 * xUnit).rgb);\n"
>-      + "  gl_FragColor.a = coeffs.a + dot(coeffs.rgb,\n"
>-      + "      texture2D(tex, interp_tc + 1.5 * xUnit).rgb);\n"
>-      + "}\n";
>-  // clang-format on
>-
>-  private final ThreadUtils.ThreadChecker threadChecker = new ThreadUtils.ThreadChecker();
>-  private final GlTextureFrameBuffer textureFrameBuffer;
>-  private TextureBuffer.Type shaderTextureType;
>-  private GlShader shader;
>-  private int texMatrixLoc;
>-  private int xUnitLoc;
>-  private int coeffsLoc;
>-  private boolean released = false;
>-
>-  /**
>-   * This class should be constructed on a thread that has an active EGL context.
>-   */
>-  public YuvConverter() {
>-    threadChecker.checkIsOnValidThread();
>-    textureFrameBuffer = new GlTextureFrameBuffer(GLES20.GL_RGBA);
>-  }
>-
>-  /** Converts the texture buffer to I420. */
>-  public I420Buffer convert(TextureBuffer textureBuffer) {
>-    final int width = textureBuffer.getWidth();
>-    final int height = textureBuffer.getHeight();
>-
>-    // SurfaceTextureHelper requires a stride that is divisible by 8.  Round width up.
>-    // See SurfaceTextureHelper for details on the size and format.
>-    final int stride = ((width + 7) / 8) * 8;
>-    final int uvHeight = (height + 1) / 2;
>-    // Due to the layout used by SurfaceTextureHelper, vPos + stride * uvHeight would overrun the
>-    // buffer.  Add one row at the bottom to compensate for this.  There will never be data in the
>-    // extra row, but now other code does not have to deal with v stride * v height exceeding the
>-    // buffer's capacity.
>-    final int size = stride * (height + uvHeight + 1);
>-    ByteBuffer buffer = JniCommon.allocateNativeByteBuffer(size);
>-    convert(buffer, width, height, stride, textureBuffer.getTextureId(),
>-        RendererCommon.convertMatrixFromAndroidGraphicsMatrix(textureBuffer.getTransformMatrix()),
>-        textureBuffer.getType());
>-
>-    final int yPos = 0;
>-    final int uPos = yPos + stride * height;
>-    // Rows of U and V alternate in the buffer, so V data starts after the first row of U.
>-    final int vPos = uPos + stride / 2;
>+  private static class ShaderCallbacks implements GlGenericDrawer.ShaderCallbacks {
>+    // Y'UV444 to RGB888, see https://en.wikipedia.org/wiki/YUV#Y.27UV444_to_RGB888_conversion. We
>+    // use the ITU-R coefficients for U and V.
>+    private static final float[] yCoeffs = new float[] {0.2987856f, 0.5871095f, 0.1141049f, 0.0f};
>+    private static final float[] uCoeffs =
>+        new float[] {-0.168805420f, -0.3317003f, 0.5005057f, 0.5f};
>+    private static final float[] vCoeffs = new float[] {0.4997964f, -0.4184672f, -0.0813292f, 0.5f};
> 
>-    buffer.position(yPos);
>-    buffer.limit(yPos + stride * height);
>-    ByteBuffer dataY = buffer.slice();
>+    private int xUnitLoc;
>+    private int coeffsLoc;
> 
>-    buffer.position(uPos);
>-    buffer.limit(uPos + stride * uvHeight);
>-    ByteBuffer dataU = buffer.slice();
>+    private float[] coeffs;
>+    private float stepSize;
> 
>-    buffer.position(vPos);
>-    buffer.limit(vPos + stride * uvHeight);
>-    ByteBuffer dataV = buffer.slice();
>+    public void setPlaneY() {
>+      coeffs = yCoeffs;
>+      stepSize = 1.0f;
>+    }
> 
>-    // SurfaceTextureHelper uses the same stride for Y, U, and V data.
>-    return JavaI420Buffer.wrap(width, height, dataY, stride, dataU, stride, dataV, stride,
>-        () -> { JniCommon.freeNativeByteBuffer(buffer); });
>-  }
>+    public void setPlaneU() {
>+      coeffs = uCoeffs;
>+      stepSize = 2.0f;
>+    }
> 
>-  /** Deprecated, use convert(TextureBuffer). */
>-  @Deprecated
>-  void convert(ByteBuffer buf, int width, int height, int stride, int srcTextureId,
>-      float[] transformMatrix) {
>-    convert(buf, width, height, stride, srcTextureId, transformMatrix, TextureBuffer.Type.OES);
>-  }
>+    public void setPlaneV() {
>+      coeffs = vCoeffs;
>+      stepSize = 2.0f;
>+    }
> 
>-  private void initShader(TextureBuffer.Type textureType) {
>-    if (shader != null) {
>-      shader.release();
>+    @Override
>+    public void onNewShader(GlShader shader) {
>+      xUnitLoc = shader.getUniformLocation("xUnit");
>+      coeffsLoc = shader.getUniformLocation("coeffs");
>     }
> 
>-    final String fragmentShader;
>-    switch (textureType) {
>-      case OES:
>-        fragmentShader = OES_FRAGMENT_SHADER;
>-        break;
>-      case RGB:
>-        fragmentShader = RGB_FRAGMENT_SHADER;
>-        break;
>-      default:
>-        throw new IllegalArgumentException("Unsupported texture type.");
>+    @Override
>+    public void onPrepareShader(GlShader shader, float[] texMatrix, int frameWidth, int frameHeight,
>+        int viewportWidth, int viewportHeight) {
>+      GLES20.glUniform4fv(coeffsLoc, /* count= */ 1, coeffs, /* offset= */ 0);
>+      // Matrix * (1;0;0;0) / (width / stepSize). Note that OpenGL uses column major order.
>+      GLES20.glUniform2f(
>+          xUnitLoc, stepSize * texMatrix[0] / frameWidth, stepSize * texMatrix[1] / frameWidth);
>     }
>+  }
>+
>+  private final ThreadUtils.ThreadChecker threadChecker = new ThreadUtils.ThreadChecker();
>+  private final GlTextureFrameBuffer i420TextureFrameBuffer =
>+      new GlTextureFrameBuffer(GLES20.GL_RGBA);
>+  private final ShaderCallbacks shaderCallbacks = new ShaderCallbacks();
>+  private final GlGenericDrawer drawer = new GlGenericDrawer(FRAGMENT_SHADER, shaderCallbacks);
> 
>-    shaderTextureType = textureType;
>-    shader = new GlShader(VERTEX_SHADER, fragmentShader);
>-    shader.useProgram();
>-    texMatrixLoc = shader.getUniformLocation("texMatrix");
>-    xUnitLoc = shader.getUniformLocation("xUnit");
>-    coeffsLoc = shader.getUniformLocation("coeffs");
>-    GLES20.glUniform1i(shader.getUniformLocation("tex"), 0);
>-    GlUtil.checkNoGLES2Error("Initialize fragment shader uniform values.");
>-    // Initialize vertex shader attributes.
>-    shader.setVertexAttribArray("in_pos", 2, DEVICE_RECTANGLE);
>-    // If the width is not a multiple of 4 pixels, the texture
>-    // will be scaled up slightly and clipped at the right border.
>-    shader.setVertexAttribArray("in_tc", 2, TEXTURE_RECTANGLE);
>+  /**
>+   * This class should be constructed on a thread that has an active EGL context.
>+   */
>+  public YuvConverter() {
>+    threadChecker.detachThread();
>   }
> 
>-  private void convert(ByteBuffer buf, int width, int height, int stride, int srcTextureId,
>-      float[] transformMatrix, TextureBuffer.Type textureType) {
>+  /** Converts the texture buffer to I420. */
>+  public I420Buffer convert(TextureBuffer inputTextureBuffer) {
>     threadChecker.checkIsOnValidThread();
>-    if (released) {
>-      throw new IllegalStateException("YuvConverter.convert called on released object");
>-    }
>-    if (textureType != shaderTextureType) {
>-      initShader(textureType);
>-    }
>-    shader.useProgram();
>-
>     // We draw into a buffer laid out like
>     //
>     //    +---------+
>@@ -246,83 +134,84 @@ public class YuvConverter {
>     // Since the V data needs to start on a boundary of such a
>     // larger pixel, it is not sufficient that |stride| is even, it
>     // has to be a multiple of 8 pixels.
>-
>-    if (stride % 8 != 0) {
>-      throw new IllegalArgumentException("Invalid stride, must be a multiple of 8");
>-    }
>-    if (stride < width) {
>-      throw new IllegalArgumentException("Invalid stride, must >= width");
>-    }
>-
>-    int y_width = (width + 3) / 4;
>-    int uv_width = (width + 7) / 8;
>-    int uv_height = (height + 1) / 2;
>-    int total_height = height + uv_height;
>-    int size = stride * total_height;
>-
>-    if (buf.capacity() < size) {
>-      throw new IllegalArgumentException("YuvConverter.convert called with too small buffer");
>-    }
>-    // Produce a frame buffer starting at top-left corner, not
>-    // bottom-left.
>-    transformMatrix =
>-        RendererCommon.multiplyMatrices(transformMatrix, RendererCommon.verticalFlipMatrix());
>-
>-    final int frameBufferWidth = stride / 4;
>-    final int frameBufferHeight = total_height;
>-    textureFrameBuffer.setSize(frameBufferWidth, frameBufferHeight);
>+    final int frameWidth = inputTextureBuffer.getWidth();
>+    final int frameHeight = inputTextureBuffer.getHeight();
>+    final int stride = ((frameWidth + 7) / 8) * 8;
>+    final int uvHeight = (frameHeight + 1) / 2;
>+    // Total height of the combined memory layout.
>+    final int totalHeight = frameHeight + uvHeight;
>+    final ByteBuffer i420ByteBuffer = JniCommon.nativeAllocateByteBuffer(stride * totalHeight);
>+    // Viewport width is divided by four since we are squeezing in four color bytes in each RGBA
>+    // pixel.
>+    final int viewportWidth = stride / 4;
>+
>+    // Produce a frame buffer starting at top-left corner, not bottom-left.
>+    final Matrix renderMatrix = new Matrix();
>+    renderMatrix.preTranslate(0.5f, 0.5f);
>+    renderMatrix.preScale(1f, -1f);
>+    renderMatrix.preTranslate(-0.5f, -0.5f);
>+
>+    i420TextureFrameBuffer.setSize(viewportWidth, totalHeight);
> 
>     // Bind our framebuffer.
>-    GLES20.glBindFramebuffer(GLES20.GL_FRAMEBUFFER, textureFrameBuffer.getFrameBufferId());
>+    GLES20.glBindFramebuffer(GLES20.GL_FRAMEBUFFER, i420TextureFrameBuffer.getFrameBufferId());
>     GlUtil.checkNoGLES2Error("glBindFramebuffer");
> 
>-    GLES20.glActiveTexture(GLES20.GL_TEXTURE0);
>-    GLES20.glBindTexture(textureType.getGlTarget(), srcTextureId);
>-    GLES20.glUniformMatrix4fv(texMatrixLoc, 1, false, transformMatrix, 0);
>+    // Draw Y.
>+    shaderCallbacks.setPlaneY();
>+    VideoFrameDrawer.drawTexture(drawer, inputTextureBuffer, renderMatrix, frameWidth, frameHeight,
>+        /* viewportX= */ 0, /* viewportY= */ 0, viewportWidth,
>+        /* viewportHeight= */ frameHeight);
> 
>-    // Draw Y
>-    GLES20.glViewport(0, 0, y_width, height);
>-    // Matrix * (1;0;0;0) / width. Note that opengl uses column major order.
>-    GLES20.glUniform2f(xUnitLoc, transformMatrix[0] / width, transformMatrix[1] / width);
>-    // Y'UV444 to RGB888, see
>-    // https://en.wikipedia.org/wiki/YUV#Y.27UV444_to_RGB888_conversion.
>-    // We use the ITU-R coefficients for U and V */
>-    GLES20.glUniform4f(coeffsLoc, 0.299f, 0.587f, 0.114f, 0.0f);
>-    GLES20.glDrawArrays(GLES20.GL_TRIANGLE_STRIP, 0, 4);
>+    // Draw U.
>+    shaderCallbacks.setPlaneU();
>+    VideoFrameDrawer.drawTexture(drawer, inputTextureBuffer, renderMatrix, frameWidth, frameHeight,
>+        /* viewportX= */ 0, /* viewportY= */ frameHeight, viewportWidth / 2,
>+        /* viewportHeight= */ uvHeight);
> 
>-    // Draw U
>-    GLES20.glViewport(0, height, uv_width, uv_height);
>-    // Matrix * (1;0;0;0) / (width / 2). Note that opengl uses column major order.
>-    GLES20.glUniform2f(
>-        xUnitLoc, 2.0f * transformMatrix[0] / width, 2.0f * transformMatrix[1] / width);
>-    GLES20.glUniform4f(coeffsLoc, -0.169f, -0.331f, 0.499f, 0.5f);
>-    GLES20.glDrawArrays(GLES20.GL_TRIANGLE_STRIP, 0, 4);
>+    // Draw V.
>+    shaderCallbacks.setPlaneV();
>+    VideoFrameDrawer.drawTexture(drawer, inputTextureBuffer, renderMatrix, frameWidth, frameHeight,
>+        /* viewportX= */ viewportWidth / 2, /* viewportY= */ frameHeight, viewportWidth / 2,
>+        /* viewportHeight= */ uvHeight);
> 
>-    // Draw V
>-    GLES20.glViewport(stride / 8, height, uv_width, uv_height);
>-    GLES20.glUniform4f(coeffsLoc, 0.499f, -0.418f, -0.0813f, 0.5f);
>-    GLES20.glDrawArrays(GLES20.GL_TRIANGLE_STRIP, 0, 4);
>-
>-    GLES20.glReadPixels(
>-        0, 0, frameBufferWidth, frameBufferHeight, GLES20.GL_RGBA, GLES20.GL_UNSIGNED_BYTE, buf);
>+    GLES20.glReadPixels(0, 0, i420TextureFrameBuffer.getWidth(), i420TextureFrameBuffer.getHeight(),
>+        GLES20.GL_RGBA, GLES20.GL_UNSIGNED_BYTE, i420ByteBuffer);
> 
>     GlUtil.checkNoGLES2Error("YuvConverter.convert");
> 
>     // Restore normal framebuffer.
>     GLES20.glBindFramebuffer(GLES20.GL_FRAMEBUFFER, 0);
>-    GLES20.glBindTexture(GLES20.GL_TEXTURE_2D, 0);
> 
>-    // Unbind texture. Reportedly needed on some devices to get
>-    // the texture updated from the camera.
>-    GLES20.glBindTexture(textureType.getGlTarget(), 0);
>+    // Prepare Y, U, and V ByteBuffer slices.
>+    final int yPos = 0;
>+    final int uPos = yPos + stride * frameHeight;
>+    // Rows of U and V alternate in the buffer, so V data starts after the first row of U.
>+    final int vPos = uPos + stride / 2;
>+
>+    i420ByteBuffer.position(yPos);
>+    i420ByteBuffer.limit(yPos + stride * frameHeight);
>+    final ByteBuffer dataY = i420ByteBuffer.slice();
>+
>+    i420ByteBuffer.position(uPos);
>+    // The last row does not have padding.
>+    final int uvSize = stride * (uvHeight - 1) + stride / 2;
>+    i420ByteBuffer.limit(uPos + uvSize);
>+    final ByteBuffer dataU = i420ByteBuffer.slice();
>+
>+    i420ByteBuffer.position(vPos);
>+    i420ByteBuffer.limit(vPos + uvSize);
>+    final ByteBuffer dataV = i420ByteBuffer.slice();
>+
>+    return JavaI420Buffer.wrap(frameWidth, frameHeight, dataY, stride, dataU, stride, dataV, stride,
>+        () -> { JniCommon.nativeFreeByteBuffer(i420ByteBuffer); });
>   }
> 
>   public void release() {
>     threadChecker.checkIsOnValidThread();
>-    released = true;
>-    if (shader != null) {
>-      shader.release();
>-    }
>-    textureFrameBuffer.release();
>+    drawer.release();
>+    i420TextureFrameBuffer.release();
>+    // Allow this class to be reused.
>+    threadChecker.detachThread();
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/YuvHelper.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/YuvHelper.java
>index 344d5d962a3..f82fde880ab 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/YuvHelper.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/YuvHelper.java
>@@ -37,8 +37,8 @@ public class YuvHelper {
>     dst.position(startV);
>     final ByteBuffer dstV = dst.slice();
> 
>-    I420Copy(srcY, srcStrideY, srcU, srcStrideU, srcV, srcStrideV, dstY, width, dstU, chromaWidth,
>-        dstV, chromaWidth, width, height);
>+    nativeI420Copy(srcY, srcStrideY, srcU, srcStrideU, srcV, srcStrideV, dstY, width, dstU,
>+        chromaWidth, dstV, chromaWidth, width, height);
>   }
> 
>   /** Helper method for copying I420 to tightly packed NV12 destination buffer. */
>@@ -61,14 +61,79 @@ public class YuvHelper {
>     dst.position(startUV);
>     final ByteBuffer dstUV = dst.slice();
> 
>-    I420ToNV12(srcY, srcStrideY, srcU, srcStrideU, srcV, srcStrideV, dstY, width, dstUV,
>+    nativeI420ToNV12(srcY, srcStrideY, srcU, srcStrideU, srcV, srcStrideV, dstY, width, dstUV,
>         chromaWidth * 2, width, height);
>   }
> 
>-  public static native void I420Copy(ByteBuffer srcY, int srcStrideY, ByteBuffer srcU,
>+  /** Helper method for rotating I420 to tightly packed destination buffer. */
>+  public static void I420Rotate(ByteBuffer srcY, int srcStrideY, ByteBuffer srcU, int srcStrideU,
>+      ByteBuffer srcV, int srcStrideV, ByteBuffer dst, int srcWidth, int srcHeight,
>+      int rotationMode) {
>+    final int dstWidth = rotationMode % 180 == 0 ? srcWidth : srcHeight;
>+    final int dstHeight = rotationMode % 180 == 0 ? srcHeight : srcWidth;
>+
>+    final int dstChromaHeight = (dstHeight + 1) / 2;
>+    final int dstChromaWidth = (dstWidth + 1) / 2;
>+
>+    final int minSize = dstWidth * dstHeight + dstChromaWidth * dstChromaHeight * 2;
>+    if (dst.capacity() < minSize) {
>+      throw new IllegalArgumentException("Expected destination buffer capacity to be at least "
>+          + minSize + " was " + dst.capacity());
>+    }
>+
>+    final int startY = 0;
>+    final int startU = dstHeight * dstWidth;
>+    final int startV = startU + dstChromaHeight * dstChromaWidth;
>+
>+    dst.position(startY);
>+    final ByteBuffer dstY = dst.slice();
>+    dst.position(startU);
>+    final ByteBuffer dstU = dst.slice();
>+    dst.position(startV);
>+    final ByteBuffer dstV = dst.slice();
>+
>+    nativeI420Rotate(srcY, srcStrideY, srcU, srcStrideU, srcV, srcStrideV, dstY, dstWidth, dstU,
>+        dstChromaWidth, dstV, dstChromaWidth, srcWidth, srcHeight, rotationMode);
>+  }
>+
>+  /** Helper method for copying a single colour plane. */
>+  public static void copyPlane(
>+      ByteBuffer src, int srcStride, ByteBuffer dst, int dstStride, int width, int height) {
>+    nativeCopyPlane(src, srcStride, dst, dstStride, width, height);
>+  }
>+
>+  public static void I420Copy(ByteBuffer srcY, int srcStrideY, ByteBuffer srcU, int srcStrideU,
>+      ByteBuffer srcV, int srcStrideV, ByteBuffer dstY, int dstStrideY, ByteBuffer dstU,
>+      int dstStrideU, ByteBuffer dstV, int dstStrideV, int width, int height) {
>+    nativeI420Copy(srcY, srcStrideY, srcU, srcStrideU, srcV, srcStrideV, dstY, dstStrideY, dstU,
>+        dstStrideU, dstV, dstStrideV, width, height);
>+  }
>+
>+  public static void I420ToNV12(ByteBuffer srcY, int srcStrideY, ByteBuffer srcU, int srcStrideU,
>+      ByteBuffer srcV, int srcStrideV, ByteBuffer dstY, int dstStrideY, ByteBuffer dstUV,
>+      int dstStrideUV, int width, int height) {
>+    nativeI420ToNV12(srcY, srcStrideY, srcU, srcStrideU, srcV, srcStrideV, dstY, dstStrideY, dstUV,
>+        dstStrideUV, width, height);
>+  }
>+
>+  public static void I420Rotate(ByteBuffer srcY, int srcStrideY, ByteBuffer srcU, int srcStrideU,
>+      ByteBuffer srcV, int srcStrideV, ByteBuffer dstY, int dstStrideY, ByteBuffer dstU,
>+      int dstStrideU, ByteBuffer dstV, int dstStrideV, int srcWidth, int srcHeight,
>+      int rotationMode) {
>+    nativeI420Rotate(srcY, srcStrideY, srcU, srcStrideU, srcV, srcStrideV, dstY, dstStrideY, dstU,
>+        dstStrideU, dstV, dstStrideV, srcWidth, srcHeight, rotationMode);
>+  }
>+
>+  private static native void nativeCopyPlane(
>+      ByteBuffer src, int srcStride, ByteBuffer dst, int dstStride, int width, int height);
>+  private static native void nativeI420Copy(ByteBuffer srcY, int srcStrideY, ByteBuffer srcU,
>       int srcStrideU, ByteBuffer srcV, int srcStrideV, ByteBuffer dstY, int dstStrideY,
>       ByteBuffer dstU, int dstStrideU, ByteBuffer dstV, int dstStrideV, int width, int height);
>-  public static native void I420ToNV12(ByteBuffer srcY, int srcStrideY, ByteBuffer srcU,
>+  private static native void nativeI420ToNV12(ByteBuffer srcY, int srcStrideY, ByteBuffer srcU,
>       int srcStrideU, ByteBuffer srcV, int srcStrideV, ByteBuffer dstY, int dstStrideY,
>       ByteBuffer dstUV, int dstStrideUV, int width, int height);
>+  private static native void nativeI420Rotate(ByteBuffer srcY, int srcStrideY, ByteBuffer srcU,
>+      int srcStrideU, ByteBuffer srcV, int srcStrideV, ByteBuffer dstY, int dstStrideY,
>+      ByteBuffer dstU, int dstStrideU, ByteBuffer dstV, int dstStrideV, int srcWidth, int srcHeight,
>+      int rotationMode);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/audio/AudioDeviceModule.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/audio/AudioDeviceModule.java
>new file mode 100644
>index 00000000000..502c68cc9ab
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/audio/AudioDeviceModule.java
>@@ -0,0 +1,38 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc.audio;
>+
>+/**
>+ * This interface is a thin wrapper on top of a native C++ webrtc::AudioDeviceModule (ADM). The
>+ * reason for basing it on a native ADM instead of a pure Java interface is that we have two native
>+ * Android implementations (OpenSLES and AAudio) that does not make sense to wrap through JNI.
>+ *
>+ * <p>Note: This class is still under development and may change without notice.
>+ */
>+public interface AudioDeviceModule {
>+  /**
>+   * Returns a C++ pointer to a webrtc::AudioDeviceModule. Caller does _not_ take ownership and
>+   * lifetime is handled through the release() call.
>+   */
>+  long getNativeAudioDeviceModulePointer();
>+
>+  /**
>+   * Release resources for this AudioDeviceModule, including native resources. The object should not
>+   * be used after this call.
>+   */
>+  void release();
>+
>+  /** Control muting/unmuting the speaker. */
>+  void setSpeakerMute(boolean mute);
>+
>+  /** Control muting/unmuting the microphone. */
>+  void setMicrophoneMute(boolean mute);
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/audio/JavaAudioDeviceModule.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/audio/JavaAudioDeviceModule.java
>new file mode 100644
>index 00000000000..4daa5f06485
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/audio/JavaAudioDeviceModule.java
>@@ -0,0 +1,310 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc.audio;
>+
>+import android.media.AudioManager;
>+import android.content.Context;
>+import org.webrtc.JniCommon;
>+import org.webrtc.Logging;
>+
>+/**
>+ * AudioDeviceModule implemented using android.media.AudioRecord as input and
>+ * android.media.AudioTrack as output.
>+ */
>+public class JavaAudioDeviceModule implements AudioDeviceModule {
>+  private static final String TAG = "JavaAudioDeviceModule";
>+
>+  public static Builder builder(Context context) {
>+    return new Builder(context);
>+  }
>+
>+  public static class Builder {
>+    private final Context context;
>+    private final AudioManager audioManager;
>+    private int sampleRate;
>+    private int audioSource = WebRtcAudioRecord.DEFAULT_AUDIO_SOURCE;
>+    private AudioTrackErrorCallback audioTrackErrorCallback;
>+    private AudioRecordErrorCallback audioRecordErrorCallback;
>+    private SamplesReadyCallback samplesReadyCallback;
>+    private boolean useHardwareAcousticEchoCanceler = isBuiltInAcousticEchoCancelerSupported();
>+    private boolean useHardwareNoiseSuppressor = isBuiltInNoiseSuppressorSupported();
>+    private boolean useStereoInput;
>+    private boolean useStereoOutput;
>+
>+    private Builder(Context context) {
>+      this.context = context;
>+      this.audioManager = (AudioManager) context.getSystemService(Context.AUDIO_SERVICE);
>+      this.sampleRate = WebRtcAudioManager.getSampleRate(audioManager);
>+    }
>+
>+    /**
>+     * Call this method if the default handling of querying the native sample rate shall be
>+     * overridden. Can be useful on some devices where the available Android APIs are known to
>+     * return invalid results.
>+     */
>+    public Builder setSampleRate(int sampleRate) {
>+      Logging.d(TAG, "Sample rate overridden to: " + sampleRate);
>+      this.sampleRate = sampleRate;
>+      return this;
>+    }
>+
>+    /**
>+     * Call this to change the audio source. The argument should be one of the values from
>+     * android.media.MediaRecorder.AudioSource. The default is AudioSource.VOICE_COMMUNICATION.
>+     */
>+    public Builder setAudioSource(int audioSource) {
>+      this.audioSource = audioSource;
>+      return this;
>+    }
>+
>+    /**
>+     * Set a callback to retrieve errors from the AudioTrack.
>+     */
>+    public Builder setAudioTrackErrorCallback(AudioTrackErrorCallback audioTrackErrorCallback) {
>+      this.audioTrackErrorCallback = audioTrackErrorCallback;
>+      return this;
>+    }
>+
>+    /**
>+     * Set a callback to retrieve errors from the AudioRecord.
>+     */
>+    public Builder setAudioRecordErrorCallback(AudioRecordErrorCallback audioRecordErrorCallback) {
>+      this.audioRecordErrorCallback = audioRecordErrorCallback;
>+      return this;
>+    }
>+
>+    /**
>+     * Set a callback to listen to the raw audio input from the AudioRecord.
>+     */
>+    public Builder setSamplesReadyCallback(SamplesReadyCallback samplesReadyCallback) {
>+      this.samplesReadyCallback = samplesReadyCallback;
>+      return this;
>+    }
>+
>+    /**
>+     * Control if the built-in HW noise suppressor should be used or not. The default is on if it is
>+     * supported. It is possible to query support by calling isBuiltInNoiseSuppressorSupported().
>+     */
>+    public Builder setUseHardwareNoiseSuppressor(boolean useHardwareNoiseSuppressor) {
>+      if (useHardwareNoiseSuppressor && !isBuiltInNoiseSuppressorSupported()) {
>+        Logging.e(TAG, "HW NS not supported");
>+        useHardwareNoiseSuppressor = false;
>+      }
>+      this.useHardwareNoiseSuppressor = useHardwareNoiseSuppressor;
>+      return this;
>+    }
>+
>+    /**
>+     * Control if the built-in HW acoustic echo canceler should be used or not. The default is on if
>+     * it is supported. It is possible to query support by calling
>+     * isBuiltInAcousticEchoCancelerSupported().
>+     */
>+    public Builder setUseHardwareAcousticEchoCanceler(boolean useHardwareAcousticEchoCanceler) {
>+      if (useHardwareAcousticEchoCanceler && !isBuiltInAcousticEchoCancelerSupported()) {
>+        Logging.e(TAG, "HW AEC not supported");
>+        useHardwareAcousticEchoCanceler = false;
>+      }
>+      this.useHardwareAcousticEchoCanceler = useHardwareAcousticEchoCanceler;
>+      return this;
>+    }
>+
>+    /**
>+     * Control if stereo input should be used or not. The default is mono.
>+     */
>+    public Builder setUseStereoInput(boolean useStereoInput) {
>+      this.useStereoInput = useStereoInput;
>+      return this;
>+    }
>+
>+    /**
>+     * Control if stereo output should be used or not. The default is mono.
>+     */
>+    public Builder setUseStereoOutput(boolean useStereoOutput) {
>+      this.useStereoOutput = useStereoOutput;
>+      return this;
>+    }
>+
>+    /**
>+     * Construct an AudioDeviceModule based on the supplied arguments. The caller takes ownership
>+     * and is responsible for calling release().
>+     */
>+    public AudioDeviceModule createAudioDeviceModule() {
>+      Logging.d(TAG, "createAudioDeviceModule");
>+      if (useHardwareNoiseSuppressor) {
>+        Logging.d(TAG, "HW NS will be used.");
>+      } else {
>+        if (isBuiltInNoiseSuppressorSupported()) {
>+          Logging.d(TAG, "Overriding default behavior; now using WebRTC NS!");
>+        }
>+        Logging.d(TAG, "HW NS will not be used.");
>+      }
>+      if (useHardwareAcousticEchoCanceler) {
>+        Logging.d(TAG, "HW AEC will be used.");
>+      } else {
>+        if (isBuiltInAcousticEchoCancelerSupported()) {
>+          Logging.d(TAG, "Overriding default behavior; now using WebRTC AEC!");
>+        }
>+        Logging.d(TAG, "HW AEC will not be used.");
>+      }
>+      final WebRtcAudioRecord audioInput =
>+          new WebRtcAudioRecord(context, audioManager, audioSource, audioRecordErrorCallback,
>+              samplesReadyCallback, useHardwareAcousticEchoCanceler, useHardwareNoiseSuppressor);
>+      final WebRtcAudioTrack audioOutput =
>+          new WebRtcAudioTrack(context, audioManager, audioTrackErrorCallback);
>+      return new JavaAudioDeviceModule(context, audioManager, audioInput, audioOutput, sampleRate,
>+          useStereoInput, useStereoOutput);
>+    }
>+  }
>+
>+  /* AudioRecord */
>+  // Audio recording error handler functions.
>+  public enum AudioRecordStartErrorCode {
>+    AUDIO_RECORD_START_EXCEPTION,
>+    AUDIO_RECORD_START_STATE_MISMATCH,
>+  }
>+
>+  public static interface AudioRecordErrorCallback {
>+    void onWebRtcAudioRecordInitError(String errorMessage);
>+    void onWebRtcAudioRecordStartError(AudioRecordStartErrorCode errorCode, String errorMessage);
>+    void onWebRtcAudioRecordError(String errorMessage);
>+  }
>+
>+  /**
>+   * Contains audio sample information.
>+   */
>+  public static class AudioSamples {
>+    /** See {@link AudioRecord#getAudioFormat()} */
>+    private final int audioFormat;
>+    /** See {@link AudioRecord#getChannelCount()} */
>+    private final int channelCount;
>+    /** See {@link AudioRecord#getSampleRate()} */
>+    private final int sampleRate;
>+
>+    private final byte[] data;
>+
>+    public AudioSamples(int audioFormat, int channelCount, int sampleRate, byte[] data) {
>+      this.audioFormat = audioFormat;
>+      this.channelCount = channelCount;
>+      this.sampleRate = sampleRate;
>+      this.data = data;
>+    }
>+
>+    public int getAudioFormat() {
>+      return audioFormat;
>+    }
>+
>+    public int getChannelCount() {
>+      return channelCount;
>+    }
>+
>+    public int getSampleRate() {
>+      return sampleRate;
>+    }
>+
>+    public byte[] getData() {
>+      return data;
>+    }
>+  }
>+
>+  /** Called when new audio samples are ready. This should only be set for debug purposes */
>+  public static interface SamplesReadyCallback {
>+    void onWebRtcAudioRecordSamplesReady(AudioSamples samples);
>+  }
>+
>+  /* AudioTrack */
>+  // Audio playout/track error handler functions.
>+  public enum AudioTrackStartErrorCode {
>+    AUDIO_TRACK_START_EXCEPTION,
>+    AUDIO_TRACK_START_STATE_MISMATCH,
>+  }
>+
>+  public static interface AudioTrackErrorCallback {
>+    void onWebRtcAudioTrackInitError(String errorMessage);
>+    void onWebRtcAudioTrackStartError(AudioTrackStartErrorCode errorCode, String errorMessage);
>+    void onWebRtcAudioTrackError(String errorMessage);
>+  }
>+
>+  /**
>+   * Returns true if the device supports built-in HW AEC, and the UUID is approved (some UUIDs can
>+   * be excluded).
>+   */
>+  public static boolean isBuiltInAcousticEchoCancelerSupported() {
>+    return WebRtcAudioEffects.isAcousticEchoCancelerSupported();
>+  }
>+
>+  /**
>+   * Returns true if the device supports built-in HW NS, and the UUID is approved (some UUIDs can be
>+   * excluded).
>+   */
>+  public static boolean isBuiltInNoiseSuppressorSupported() {
>+    return WebRtcAudioEffects.isNoiseSuppressorSupported();
>+  }
>+
>+  private final Context context;
>+  private final AudioManager audioManager;
>+  private final WebRtcAudioRecord audioInput;
>+  private final WebRtcAudioTrack audioOutput;
>+  private final int sampleRate;
>+  private final boolean useStereoInput;
>+  private final boolean useStereoOutput;
>+
>+  private final Object nativeLock = new Object();
>+  private long nativeAudioDeviceModule;
>+
>+  private JavaAudioDeviceModule(Context context, AudioManager audioManager,
>+      WebRtcAudioRecord audioInput, WebRtcAudioTrack audioOutput, int sampleRate,
>+      boolean useStereoInput, boolean useStereoOutput) {
>+    this.context = context;
>+    this.audioManager = audioManager;
>+    this.audioInput = audioInput;
>+    this.audioOutput = audioOutput;
>+    this.sampleRate = sampleRate;
>+    this.useStereoInput = useStereoInput;
>+    this.useStereoOutput = useStereoOutput;
>+  }
>+
>+  @Override
>+  public long getNativeAudioDeviceModulePointer() {
>+    synchronized (nativeLock) {
>+      if (nativeAudioDeviceModule == 0) {
>+        nativeAudioDeviceModule = nativeCreateAudioDeviceModule(context, audioManager, audioInput,
>+            audioOutput, sampleRate, useStereoInput, useStereoOutput);
>+      }
>+      return nativeAudioDeviceModule;
>+    }
>+  }
>+
>+  @Override
>+  public void release() {
>+    synchronized (nativeLock) {
>+      if (nativeAudioDeviceModule != 0) {
>+        JniCommon.nativeReleaseRef(nativeAudioDeviceModule);
>+        nativeAudioDeviceModule = 0;
>+      }
>+    }
>+  }
>+
>+  @Override
>+  public void setSpeakerMute(boolean mute) {
>+    Logging.d(TAG, "setSpeakerMute: " + mute);
>+    audioOutput.setSpeakerMute(mute);
>+  }
>+
>+  @Override
>+  public void setMicrophoneMute(boolean mute) {
>+    Logging.d(TAG, "setMicrophoneMute: " + mute);
>+    audioInput.setMicrophoneMute(mute);
>+  }
>+
>+  private static native long nativeCreateAudioDeviceModule(Context context,
>+      AudioManager audioManager, WebRtcAudioRecord audioInput, WebRtcAudioTrack audioOutput,
>+      int sampleRate, boolean useStereoInput, boolean useStereoOutput);
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/audio/LegacyAudioDeviceModule.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/audio/LegacyAudioDeviceModule.java
>new file mode 100644
>index 00000000000..de0d0d61f94
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/api/org/webrtc/audio/LegacyAudioDeviceModule.java
>@@ -0,0 +1,46 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc.audio;
>+
>+import org.webrtc.voiceengine.WebRtcAudioRecord;
>+import org.webrtc.voiceengine.WebRtcAudioTrack;
>+
>+/**
>+ * This class represents the legacy AudioDeviceModule that is currently hardcoded into C++ WebRTC.
>+ * It will return a null native AudioDeviceModule pointer, leading to an internal object being
>+ * created inside WebRTC that is controlled by static calls to the classes under the voiceengine
>+ * package. Please use the new JavaAudioDeviceModule instead of this class.
>+ */
>+@Deprecated
>+public class LegacyAudioDeviceModule implements AudioDeviceModule {
>+  @Override
>+  public long getNativeAudioDeviceModulePointer() {
>+    // Returning a null pointer will make WebRTC construct the built-in legacy AudioDeviceModule for
>+    // Android internally.
>+    return 0;
>+  }
>+
>+  @Override
>+  public void release() {
>+    // All control for this ADM goes through static global methods and the C++ object is owned
>+    // internally by WebRTC.
>+  }
>+
>+  @Override
>+  public void setSpeakerMute(boolean mute) {
>+    WebRtcAudioTrack.setSpeakerMute(mute);
>+  }
>+
>+  @Override
>+  public void setMicrophoneMute(boolean mute) {
>+    WebRtcAudioRecord.setMicrophoneMute(mute);
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/mediasource_jni.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/loggable_test.cc
>similarity index 56%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/mediasource_jni.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/loggable_test.cc
>index e7532237787..1a11075216f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/mediasource_jni.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/loggable_test.cc
>@@ -1,5 +1,5 @@
> /*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>  *
>  *  Use of this source code is governed by a BSD-style license
>  *  that can be found in the LICENSE file in the root of the source
>@@ -8,20 +8,23 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "api/mediastreaminterface.h"
>+#include <memory>
>+
>+#include "rtc_base/logging.h"
>+#include "sdk/android/native_api/jni/java_types.h"
> #include "sdk/android/src/jni/jni_helpers.h"
> 
> namespace webrtc {
> namespace jni {
> 
>-JNI_FUNCTION_DECLARATION(jobject,
>-                         MediaSource_nativeState,
>+JNI_FUNCTION_DECLARATION(void,
>+                         LoggableTest_nativeLogInfoTestMessage,
>                          JNIEnv* jni,
>                          jclass,
>-                         jlong j_p) {
>-  rtc::scoped_refptr<MediaSourceInterface> p(
>-      reinterpret_cast<MediaSourceInterface*>(j_p));
>-  return JavaEnumFromIndexAndClassName(jni, "MediaSource$State", p->state());
>+                         jstring j_message) {
>+  std::string message =
>+      JavaToNativeString(jni, JavaParamRef<jstring>(j_message));
>+  RTC_LOG(LS_INFO) << message;
> }
> 
> }  // namespace jni
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/Camera1CapturerUsingByteBufferTest.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/Camera1CapturerUsingByteBufferTest.java
>index eccdd723c03..ef7669102f9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/Camera1CapturerUsingByteBufferTest.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/Camera1CapturerUsingByteBufferTest.java
>@@ -121,12 +121,6 @@ public class Camera1CapturerUsingByteBufferTest {
>     fixtures.cameraEventsInvoked();
>   }
> 
>-  @Test
>-  @MediumTest
>-  public void testUpdateMediaRecorder() throws InterruptedException, IOException {
>-    fixtures.updateMediaRecorder(false /* useSurfaceCapture */);
>-  }
>-
>   // Test what happens when attempting to call e.g. switchCamera() after camera has been stopped.
>   @Test
>   @MediumTest
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/Camera1CapturerUsingTextureTest.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/Camera1CapturerUsingTextureTest.java
>index b067662966f..28fa825ad74 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/Camera1CapturerUsingTextureTest.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/Camera1CapturerUsingTextureTest.java
>@@ -116,12 +116,6 @@ public class Camera1CapturerUsingTextureTest {
>     fixtures.cameraEventsInvoked();
>   }
> 
>-  @Test
>-  @MediumTest
>-  public void testUpdateMediaRecorder() throws InterruptedException, IOException {
>-    fixtures.updateMediaRecorder(false /* useSurfaceCapture */);
>-  }
>-
>   // Test what happens when attempting to call e.g. switchCamera() after camera has been stopped.
>   @Test
>   @MediumTest
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/Camera2CapturerTest.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/Camera2CapturerTest.java
>index 6b6b18d1ed0..77e8d77f0ae 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/Camera2CapturerTest.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/Camera2CapturerTest.java
>@@ -27,6 +27,7 @@ import android.support.test.filters.MediumTest;
> import android.support.test.filters.SmallTest;
> import java.io.IOException;
> import java.util.concurrent.CountDownLatch;
>+import javax.annotation.Nullable;
> import org.chromium.base.test.BaseJUnit4ClassRunner;
> import org.junit.After;
> import org.junit.Before;
>@@ -46,7 +47,7 @@ public class Camera2CapturerTest {
>     final LooperThread looperThread;
>     final CountDownLatch openDoneSignal;
>     final Object cameraDeviceLock;
>-    CameraDevice cameraDevice; // Guarded by cameraDeviceLock
>+    @Nullable CameraDevice cameraDevice; // Guarded by cameraDeviceLock
>     boolean openSucceeded; // Guarded by cameraDeviceLock
> 
>     private class LooperThread extends Thread {
>@@ -245,12 +246,6 @@ public class Camera2CapturerTest {
>     fixtures.cameraEventsInvoked();
>   }
> 
>-  @Test
>-  @MediumTest
>-  public void testUpdateMediaRecorder() throws InterruptedException, IOException {
>-    fixtures.updateMediaRecorder(true /* useSurfaceCapture */);
>-  }
>-
>   // Test what happens when attempting to call e.g. switchCamera() after camera has been stopped.
>   @Test
>   @MediumTest
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/CameraVideoCapturerTestFixtures.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/CameraVideoCapturerTestFixtures.java
>index d8e3d5bf23a..17702c318bc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/CameraVideoCapturerTestFixtures.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/CameraVideoCapturerTestFixtures.java
>@@ -18,17 +18,17 @@ import static org.junit.Assert.fail;
> import android.annotation.TargetApi;
> import android.content.Context;
> import android.media.CamcorderProfile;
>-import android.media.MediaRecorder;
> import android.os.Environment;
> import java.io.File;
> import java.io.IOException;
> import java.util.ArrayList;
> import java.util.List;
> import java.util.concurrent.CountDownLatch;
>+import javax.annotation.Nullable;
> import org.chromium.base.test.BaseJUnit4ClassRunner;
> import org.junit.runner.RunWith;
> import org.webrtc.CameraEnumerationAndroid.CaptureFormat;
>-import org.webrtc.VideoRenderer.I420Frame;
>+import org.webrtc.VideoFrame;
> 
> class CameraVideoCapturerTestFixtures {
>   static final String TAG = "CameraVideoCapturerTestFixtures";
>@@ -37,21 +37,20 @@ class CameraVideoCapturerTestFixtures {
>   static final int DEFAULT_HEIGHT = 480;
>   static final int DEFAULT_FPS = 15;
> 
>-  static private class RendererCallbacks implements VideoRenderer.Callbacks {
>+  static private class RendererCallbacks implements VideoSink {
>     private final Object frameLock = new Object();
>     private int framesRendered = 0;
>     private int width = 0;
>     private int height = 0;
> 
>     @Override
>-    public void renderFrame(I420Frame frame) {
>+    public void onFrame(VideoFrame frame) {
>       synchronized (frameLock) {
>         ++framesRendered;
>-        width = frame.rotatedWidth();
>-        height = frame.rotatedHeight();
>+        width = frame.getRotatedWidth();
>+        height = frame.getRotatedHeight();
>         frameLock.notify();
>       }
>-      VideoRenderer.renderFrameDone(frame);
>     }
> 
>     public int frameWidth() {
>@@ -78,32 +77,33 @@ class CameraVideoCapturerTestFixtures {
>     }
>   }
> 
>-  static private class FakeAsyncRenderer implements VideoRenderer.Callbacks {
>-    private final List<I420Frame> pendingFrames = new ArrayList<I420Frame>();
>+  static private class FakeAsyncRenderer implements VideoSink {
>+    private final List<VideoFrame> pendingFrames = new ArrayList<VideoFrame>();
> 
>     @Override
>-    public void renderFrame(I420Frame frame) {
>+    public void onFrame(VideoFrame frame) {
>       synchronized (pendingFrames) {
>+        frame.retain();
>         pendingFrames.add(frame);
>         pendingFrames.notifyAll();
>       }
>     }
> 
>     // Wait until at least one frame have been received, before returning them.
>-    public List<I420Frame> waitForPendingFrames() throws InterruptedException {
>+    public List<VideoFrame> waitForPendingFrames() throws InterruptedException {
>       Logging.d(TAG, "Waiting for pending frames");
>       synchronized (pendingFrames) {
>         while (pendingFrames.isEmpty()) {
>           pendingFrames.wait();
>         }
>-        return new ArrayList<I420Frame>(pendingFrames);
>+        return new ArrayList<VideoFrame>(pendingFrames);
>       }
>     }
>   }
> 
>-  static private class FakeCapturerObserver implements CameraVideoCapturer.CapturerObserver {
>+  static private class FakeCapturerObserver implements CapturerObserver {
>     private int framesCaptured = 0;
>-    private VideoFrame videoFrame;
>+    private @Nullable VideoFrame videoFrame;
>     final private Object frameLock = new Object();
>     final private Object capturerStartLock = new Object();
>     private Boolean capturerStartResult;
>@@ -124,18 +124,6 @@ class CameraVideoCapturerTestFixtures {
>       Logging.d(TAG, "onCapturerStopped");
>     }
> 
>-    @Override
>-    public void onByteBufferFrameCaptured(
>-        byte[] frame, int width, int height, int rotation, long timeStamp) {
>-      throw new RuntimeException("onByteBufferFrameCaptured called");
>-    }
>-
>-    @Override
>-    public void onTextureFrameCaptured(int width, int height, int oesTextureId,
>-        float[] transformMatrix, int rotation, long timeStamp) {
>-      throw new RuntimeException("onTextureFrameCaptured called");
>-    }
>-
>     @Override
>     public void onFrameCaptured(VideoFrame frame) {
>       synchronized (frameLock) {
>@@ -284,6 +272,7 @@ class CameraVideoCapturerTestFixtures {
>    * is used for testing local rendering from a capturer.
>    */
>   static private class VideoTrackWithRenderer {
>+    public SurfaceTextureHelper surfaceTextureHelper;
>     public VideoSource source;
>     public VideoTrack track;
>     public RendererCallbacks rendererCallbacks;
>@@ -302,7 +291,7 @@ class CameraVideoCapturerTestFixtures {
>       return cameraEnumerator.createCapturer(name, eventsHandler);
>     }
> 
>-    public String getNameOfFrontFacingDevice() {
>+    public @Nullable String getNameOfFrontFacingDevice() {
>       for (String deviceName : cameraEnumerator.getDeviceNames()) {
>         if (cameraEnumerator.isFrontFacing(deviceName)) {
>           return deviceName;
>@@ -312,7 +301,7 @@ class CameraVideoCapturerTestFixtures {
>       return null;
>     }
> 
>-    public String getNameOfBackFacingDevice() {
>+    public @Nullable String getNameOfBackFacingDevice() {
>       for (String deviceName : cameraEnumerator.getDeviceNames()) {
>         if (cameraEnumerator.isBackFacing(deviceName)) {
>           return deviceName;
>@@ -346,11 +335,10 @@ class CameraVideoCapturerTestFixtures {
>   CameraVideoCapturerTestFixtures(TestObjectFactory testObjectFactory) {
>     PeerConnectionFactory.initialize(
>         PeerConnectionFactory.InitializationOptions.builder(testObjectFactory.getAppContext())
>-            .setFieldTrials(PeerConnectionFactory.VIDEO_FRAME_EMIT_TRIAL + "/"
>-                + PeerConnectionFactory.TRIAL_ENABLED + "/")
>+            .setNativeLibraryName(TestConstants.NATIVE_LIBRARY)
>             .createInitializationOptions());
> 
>-    this.peerConnectionFactory = new PeerConnectionFactory(null /* options */);
>+    this.peerConnectionFactory = PeerConnectionFactory.builder().createPeerConnectionFactory();
>     this.testObjectFactory = testObjectFactory;
>   }
> 
>@@ -400,13 +388,18 @@ class CameraVideoCapturerTestFixtures {
>   }
> 
>   private VideoTrackWithRenderer createVideoTrackWithRenderer(
>-      CameraVideoCapturer capturer, VideoRenderer.Callbacks rendererCallbacks) {
>+      CameraVideoCapturer capturer, VideoSink rendererCallbacks) {
>     VideoTrackWithRenderer videoTrackWithRenderer = new VideoTrackWithRenderer();
>-    videoTrackWithRenderer.source = peerConnectionFactory.createVideoSource(capturer);
>+    videoTrackWithRenderer.surfaceTextureHelper = SurfaceTextureHelper.create(
>+        "SurfaceTextureHelper test" /* threadName */, null /* sharedContext */);
>+    videoTrackWithRenderer.source =
>+        peerConnectionFactory.createVideoSource(/* isScreencast= */ false);
>+    capturer.initialize(videoTrackWithRenderer.surfaceTextureHelper,
>+        testObjectFactory.getAppContext(), videoTrackWithRenderer.source.getCapturerObserver());
>     capturer.startCapture(DEFAULT_WIDTH, DEFAULT_HEIGHT, DEFAULT_FPS);
>     videoTrackWithRenderer.track =
>         peerConnectionFactory.createVideoTrack("dummy", videoTrackWithRenderer.source);
>-    videoTrackWithRenderer.track.addRenderer(new VideoRenderer(rendererCallbacks));
>+    videoTrackWithRenderer.track.addSink(rendererCallbacks);
>     return videoTrackWithRenderer;
>   }
> 
>@@ -524,111 +517,6 @@ class CameraVideoCapturerTestFixtures {
>     disposeVideoTrackWithRenderer(videoTrackWithRenderer);
>   }
> 
>-  @TargetApi(21)
>-  private static void prepareMediaRecorderForTests(
>-      MediaRecorder mediaRecorder, File outputFile, boolean useSurfaceCapture) throws IOException {
>-    mediaRecorder.setVideoSource(
>-        useSurfaceCapture ? MediaRecorder.VideoSource.SURFACE : MediaRecorder.VideoSource.CAMERA);
>-    CamcorderProfile profile = CamcorderProfile.get(CamcorderProfile.QUALITY_480P);
>-    profile.videoCodec = MediaRecorder.VideoEncoder.H264;
>-    profile.videoBitRate = 2500000;
>-    profile.videoFrameWidth = 640;
>-    profile.videoFrameHeight = 480;
>-    mediaRecorder.setOutputFormat(MediaRecorder.OutputFormat.MPEG_4);
>-    mediaRecorder.setVideoFrameRate(profile.videoFrameRate);
>-    mediaRecorder.setVideoSize(profile.videoFrameWidth, profile.videoFrameHeight);
>-    mediaRecorder.setVideoEncodingBitRate(profile.videoBitRate);
>-    mediaRecorder.setVideoEncoder(profile.videoCodec);
>-    mediaRecorder.setOutputFile(outputFile.getPath());
>-    mediaRecorder.prepare();
>-  }
>-
>-  @TargetApi(21)
>-  public void updateMediaRecorder(boolean useSurfaceCapture)
>-      throws InterruptedException, IOException {
>-    final CapturerInstance capturerInstance = createCapturer(false /* initialize */);
>-    final VideoTrackWithRenderer videoTrackWithRenderer =
>-        createVideoTrackWithRenderer(capturerInstance.capturer);
>-    // Wait for the camera to start so we can add and remove MediaRecorder.
>-    assertTrue(videoTrackWithRenderer.rendererCallbacks.waitForNextFrameToRender() > 0);
>-
>-    final String videoOutPath = Environment.getExternalStorageDirectory().getPath()
>-        + "/chromium_tests_root/testmediarecorder.mp4";
>-    File outputFile = new File(videoOutPath);
>-
>-    // Create MediaRecorder object
>-    MediaRecorder mediaRecorder = new MediaRecorder();
>-    if (useSurfaceCapture) {
>-      // When using using surface capture, media recorder has to be prepared before adding it to the
>-      // camera.
>-      prepareMediaRecorderForTests(mediaRecorder, outputFile, useSurfaceCapture);
>-    }
>-
>-    // Add MediaRecorder to camera pipeline.
>-    final boolean[] addMediaRecorderSuccessful = new boolean[1];
>-    final CountDownLatch addBarrier = new CountDownLatch(1);
>-    CameraVideoCapturer.MediaRecorderHandler addMediaRecorderHandler =
>-        new CameraVideoCapturer.MediaRecorderHandler() {
>-          @Override
>-          public void onMediaRecorderSuccess() {
>-            addMediaRecorderSuccessful[0] = true;
>-            addBarrier.countDown();
>-          }
>-          @Override
>-          public void onMediaRecorderError(String errorDescription) {
>-            Logging.e(TAG, errorDescription);
>-            addMediaRecorderSuccessful[0] = false;
>-            addBarrier.countDown();
>-          }
>-        };
>-    capturerInstance.capturer.addMediaRecorderToCamera(mediaRecorder, addMediaRecorderHandler);
>-    // Wait until MediaRecoder has been added.
>-    addBarrier.await();
>-    // Check result.
>-    assertTrue(addMediaRecorderSuccessful[0]);
>-
>-    // Start MediaRecorder and wait for a few frames to capture.
>-    if (!useSurfaceCapture) {
>-      // When using using camera capture, media recorder has to be prepared after adding it to the
>-      // camera.
>-      prepareMediaRecorderForTests(mediaRecorder, outputFile, useSurfaceCapture);
>-    }
>-    mediaRecorder.start();
>-    for (int i = 0; i < 5; i++) {
>-      assertTrue(videoTrackWithRenderer.rendererCallbacks.waitForNextFrameToRender() > 0);
>-    }
>-    mediaRecorder.stop();
>-
>-    // Remove MediaRecorder from camera pipeline.
>-    final boolean[] removeMediaRecorderSuccessful = new boolean[1];
>-    final CountDownLatch removeBarrier = new CountDownLatch(1);
>-    CameraVideoCapturer.MediaRecorderHandler removeMediaRecorderHandler =
>-        new CameraVideoCapturer.MediaRecorderHandler() {
>-          @Override
>-          public void onMediaRecorderSuccess() {
>-            removeMediaRecorderSuccessful[0] = true;
>-            removeBarrier.countDown();
>-          }
>-          @Override
>-          public void onMediaRecorderError(String errorDescription) {
>-            removeMediaRecorderSuccessful[0] = false;
>-            removeBarrier.countDown();
>-          }
>-        };
>-    capturerInstance.capturer.removeMediaRecorderFromCamera(removeMediaRecorderHandler);
>-    // Wait until MediaRecoder has been removed.
>-    removeBarrier.await();
>-    // Check result.
>-    assertTrue(removeMediaRecorderSuccessful[0]);
>-    // Ensure that frames are received after removing MediaRecorder.
>-    assertTrue(videoTrackWithRenderer.rendererCallbacks.waitForNextFrameToRender() > 0);
>-    // Check that recorded file contains some data.
>-    assertTrue(outputFile.length() > 0);
>-
>-    disposeCapturer(capturerInstance);
>-    disposeVideoTrackWithRenderer(videoTrackWithRenderer);
>-  }
>-
>   public void cameraEventsInvoked() throws InterruptedException {
>     final CapturerInstance capturerInstance = createCapturer(true /* initialize */);
>     startCapture(capturerInstance);
>@@ -740,13 +628,13 @@ class CameraVideoCapturerTestFixtures {
>     disposeVideoTrackWithRenderer(videoTrackWithRenderer);
> 
>     // Return the frame(s), on a different thread out of spite.
>-    final List<I420Frame> pendingFrames =
>+    final List<VideoFrame> pendingFrames =
>         videoTrackWithRenderer.fakeAsyncRenderer.waitForPendingFrames();
>     final Thread returnThread = new Thread(new Runnable() {
>       @Override
>       public void run() {
>-        for (I420Frame frame : pendingFrames) {
>-          VideoRenderer.renderFrameDone(frame);
>+        for (VideoFrame frame : pendingFrames) {
>+          frame.release();
>         }
>       }
>     });
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/DefaultAudioProcessingFactoryTest.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/DefaultAudioProcessingFactoryTest.java
>deleted file mode 100644
>index 0ed8c35e3cd..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/DefaultAudioProcessingFactoryTest.java
>+++ /dev/null
>@@ -1,47 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-package org.webrtc;
>-
>-import static org.junit.Assert.assertEquals;
>-import static org.junit.Assert.assertNotNull;
>-import static org.junit.Assert.assertTrue;
>-
>-import android.support.test.InstrumentationRegistry;
>-import android.support.test.filters.MediumTest;
>-import android.support.test.filters.SmallTest;
>-import org.chromium.base.test.BaseJUnit4ClassRunner;
>-import org.junit.Before;
>-import org.junit.Test;
>-import org.junit.runner.RunWith;
>-
>-/** Unit tests for {@link DefaultAudioProcessingFactory}. */
>-@RunWith(BaseJUnit4ClassRunner.class)
>-public final class DefaultAudioProcessingFactoryTest {
>-  @Before
>-  public void setUp() {
>-    PeerConnectionFactory.initialize(
>-        PeerConnectionFactory.InitializationOptions.builder(InstrumentationRegistry.getContext())
>-            .createInitializationOptions());
>-  }
>-
>-  /**
>-   * Tests that a PeerConnectionFactory can be initialized with an AudioProcessingFactory without
>-   * crashing.
>-   */
>-  @Test
>-  @MediumTest
>-  public void testInitializePeerConnectionFactory() {
>-    AudioProcessingFactory audioProcessingFactory = new DefaultAudioProcessingFactory();
>-    PeerConnectionFactory peerConnectionFactory = new PeerConnectionFactory(null /* options */,
>-        null /* encoderFactory */, null /* decoderFactory */, audioProcessingFactory);
>-    peerConnectionFactory.dispose();
>-  }
>-}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/DefaultVideoEncoderFactoryTest.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/DefaultVideoEncoderFactoryTest.java
>index d1a3b6629b6..4d2fe5c0643 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/DefaultVideoEncoderFactoryTest.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/DefaultVideoEncoderFactoryTest.java
>@@ -18,6 +18,7 @@ import java.util.ArrayList;
> import java.util.Arrays;
> import java.util.HashMap;
> import java.util.List;
>+import javax.annotation.Nullable;
> import org.chromium.base.test.BaseJUnit4ClassRunner;
> import org.junit.Before;
> import org.junit.Test;
>@@ -51,7 +52,7 @@ public class DefaultVideoEncoderFactoryTest {
>     }
> 
>     @Override
>-    public VideoEncoder createEncoder(VideoCodecInfo info) {
>+    public @Nullable VideoEncoder createEncoder(VideoCodecInfo info) {
>       return null;
>     }
> 
>@@ -63,7 +64,7 @@ public class DefaultVideoEncoderFactoryTest {
> 
>   @Before
>   public void setUp() {
>-    NativeLibrary.initialize(new NativeLibrary.DefaultLoader());
>+    NativeLibrary.initialize(new NativeLibrary.DefaultLoader(), TestConstants.NATIVE_LIBRARY);
>   }
> 
>   @SmallTest
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/EglRendererTest.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/EglRendererTest.java
>index 6ba9168ee17..29d822facef 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/EglRendererTest.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/EglRendererTest.java
>@@ -115,6 +115,7 @@ public class EglRendererTest {
>   public void setUp() throws Exception {
>     PeerConnectionFactory.initialize(PeerConnectionFactory.InitializationOptions
>                                          .builder(InstrumentationRegistry.getTargetContext())
>+                                         .setNativeLibraryName(TestConstants.NATIVE_LIBRARY)
>                                          .createInitializationOptions());
>     eglRenderer = new EglRenderer("TestRenderer: ");
>     eglRenderer.init(null /* sharedContext */, EglBase.CONFIG_RGBA, new GlRectDrawer());
>@@ -224,6 +225,8 @@ public class EglRendererTest {
>   }
> 
>   /** Checks that the bitmap content matches the test frame with the given index. */
>+  // TODO(titovartem) make correct fix during webrtc:9175
>+  @SuppressWarnings("ByteBufferBackingArray")
>   private static void checkBitmapContent(Bitmap bitmap, int frame) {
>     checkBitmap(bitmap, 1f);
> 
>@@ -246,9 +249,12 @@ public class EglRendererTest {
> 
>   /** Tells eglRenderer to render test frame with given index. */
>   private void feedFrame(int i) {
>-    eglRenderer.renderFrame(new VideoRenderer.I420Frame(TEST_FRAME_WIDTH, TEST_FRAME_HEIGHT, 0,
>-        new int[] {TEST_FRAME_WIDTH, TEST_FRAME_WIDTH / 2, TEST_FRAME_WIDTH / 2}, TEST_FRAMES[i],
>-        0));
>+    final VideoFrame.I420Buffer buffer = JavaI420Buffer.wrap(TEST_FRAME_WIDTH, TEST_FRAME_HEIGHT,
>+        TEST_FRAMES[i][0], TEST_FRAME_WIDTH, TEST_FRAMES[i][1], TEST_FRAME_WIDTH / 2,
>+        TEST_FRAMES[i][2], TEST_FRAME_WIDTH / 2, null /* releaseCallback */);
>+    final VideoFrame frame = new VideoFrame(buffer, 0 /* rotation */, 0 /* timestamp */);
>+    eglRenderer.onFrame(frame);
>+    frame.release();
>   }
> 
>   @Test
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/FileVideoCapturerTest.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/FileVideoCapturerTest.java
>index f3b7902eb64..7a2aedf5a00 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/FileVideoCapturerTest.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/FileVideoCapturerTest.java
>@@ -21,12 +21,13 @@ import java.nio.charset.Charset;
> import java.util.ArrayList;
> import java.util.Arrays;
> import org.chromium.base.test.BaseJUnit4ClassRunner;
>+import org.junit.Before;
> import org.junit.Test;
> import org.junit.runner.RunWith;
> 
> @RunWith(BaseJUnit4ClassRunner.class)
> public class FileVideoCapturerTest {
>-  public static class MockCapturerObserver implements VideoCapturer.CapturerObserver {
>+  public static class MockCapturerObserver implements CapturerObserver {
>     private final ArrayList<VideoFrame> frames = new ArrayList<VideoFrame>();
> 
>     @Override
>@@ -39,20 +40,6 @@ public class FileVideoCapturerTest {
>       // Empty on purpose.
>     }
> 
>-    @Override
>-    // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
>-    @SuppressWarnings("NoSynchronizedMethodCheck")
>-    public synchronized void onByteBufferFrameCaptured(
>-        byte[] data, int width, int height, int rotation, long timeStamp) {
>-      // Empty on purpose.
>-    }
>-
>-    @Override
>-    public void onTextureFrameCaptured(int width, int height, int oesTextureId,
>-        float[] transformMatrix, int rotation, long timestamp) {
>-      // Empty on purpose.
>-    }
>-
>     @Override
>     // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
>     @SuppressWarnings("NoSynchronizedMethodCheck")
>@@ -72,6 +59,11 @@ public class FileVideoCapturerTest {
>     }
>   }
> 
>+  @Before
>+  public void setUp() {
>+    NativeLibrary.initialize(new NativeLibrary.DefaultLoader(), TestConstants.NATIVE_LIBRARY);
>+  }
>+
>   @Test
>   @SmallTest
>   public void testVideoCaptureFromFile() throws InterruptedException, IOException {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/GlRectDrawerTest.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/GlRectDrawerTest.java
>index e1853f2da2f..fecddb34fe0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/GlRectDrawerTest.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/GlRectDrawerTest.java
>@@ -34,6 +34,14 @@ public class GlRectDrawerTest {
>   // When comparing pixels, allow some slack for float arithmetic and integer rounding.
>   private static final float MAX_DIFF = 1.5f;
> 
>+  // clang-format off
>+  private static final float[] IDENTITY_MATRIX = {
>+      1, 0, 0, 0,
>+      0, 1, 0, 0,
>+      0, 0, 1, 0,
>+      0, 0, 0, 1};
>+  // clang-format on
>+
>   private static float normalizedByte(byte b) {
>     return (b & 0xFF) / 255.0f;
>   }
>@@ -85,6 +93,8 @@ public class GlRectDrawerTest {
>     return rgbBuffer;
>   }
> 
>+  // TODO(titovartem) make correct fix during webrtc:9175
>+  @SuppressWarnings("ByteBufferBackingArray")
>   @Test
>   @SmallTest
>   public void testRgbRendering() {
>@@ -108,8 +118,8 @@ public class GlRectDrawerTest {
> 
>     // Draw the RGB frame onto the pixel buffer.
>     final GlRectDrawer drawer = new GlRectDrawer();
>-    drawer.drawRgb(rgbTexture, RendererCommon.identityMatrix(), WIDTH, HEIGHT, 0 /* viewportX */,
>-        0 /* viewportY */, WIDTH, HEIGHT);
>+    drawer.drawRgb(rgbTexture, IDENTITY_MATRIX, WIDTH, HEIGHT, 0 /* viewportX */, 0 /* viewportY */,
>+        WIDTH, HEIGHT);
> 
>     // Download the pixels in the pixel buffer as RGBA. Not all platforms support RGB, e.g. Nexus 9.
>     final ByteBuffer rgbaData = ByteBuffer.allocateDirect(WIDTH * HEIGHT * 4);
>@@ -124,6 +134,8 @@ public class GlRectDrawerTest {
>     eglBase.release();
>   }
> 
>+  // TODO(titovartem) make correct fix during webrtc:9175
>+  @SuppressWarnings("ByteBufferBackingArray")
>   @Test
>   @SmallTest
>   public void testYuvRendering() {
>@@ -157,7 +169,7 @@ public class GlRectDrawerTest {
> 
>     // Draw the YUV frame onto the pixel buffer.
>     final GlRectDrawer drawer = new GlRectDrawer();
>-    drawer.drawYuv(yuvTextures, RendererCommon.identityMatrix(), WIDTH, HEIGHT, 0 /* viewportX */,
>+    drawer.drawYuv(yuvTextures, IDENTITY_MATRIX, WIDTH, HEIGHT, 0 /* viewportX */,
>         0 /* viewportY */, WIDTH, HEIGHT);
> 
>     // Download the pixels in the pixel buffer as RGBA. Not all platforms support RGB, e.g. Nexus 9.
>@@ -212,6 +224,8 @@ public class GlRectDrawerTest {
>    *  - Render the OES texture onto the pixel buffer.
>    *  - Read back the pixel buffer and compare it with the known RGB data.
>    */
>+  // TODO(titovartem) make correct fix during webrtc:9175
>+  @SuppressWarnings("ByteBufferBackingArray")
>   @Test
>   @MediumTest
>   public void testOesRendering() throws InterruptedException {
>@@ -223,11 +237,11 @@ public class GlRectDrawerTest {
>       private final GlRectDrawer drawer;
>       private final int rgbTexture;
> 
>-      public StubOesTextureProducer(
>-          EglBase.Context sharedContext, SurfaceTexture surfaceTexture, int width, int height) {
>+      public StubOesTextureProducer(EglBase.Context sharedContext,
>+          SurfaceTextureHelper surfaceTextureHelper, int width, int height) {
>         eglBase = EglBase.create(sharedContext, EglBase.CONFIG_PLAIN);
>-        surfaceTexture.setDefaultBufferSize(width, height);
>-        eglBase.createSurface(surfaceTexture);
>+        surfaceTextureHelper.setTextureSize(width, height);
>+        eglBase.createSurface(surfaceTextureHelper.getSurfaceTexture());
>         assertEquals(eglBase.surfaceWidth(), width);
>         assertEquals(eglBase.surfaceHeight(), height);
> 
>@@ -246,8 +260,8 @@ public class GlRectDrawerTest {
>         GLES20.glTexImage2D(GLES20.GL_TEXTURE_2D, 0, GLES20.GL_RGB, WIDTH, HEIGHT, 0, GLES20.GL_RGB,
>             GLES20.GL_UNSIGNED_BYTE, rgbPlane);
>         // Draw the RGB data onto the SurfaceTexture.
>-        drawer.drawRgb(rgbTexture, RendererCommon.identityMatrix(), WIDTH, HEIGHT,
>-            0 /* viewportX */, 0 /* viewportY */, WIDTH, HEIGHT);
>+        drawer.drawRgb(rgbTexture, IDENTITY_MATRIX, WIDTH, HEIGHT, 0 /* viewportX */,
>+            0 /* viewportY */, WIDTH, HEIGHT);
>         eglBase.swapBuffers();
>       }
> 
>@@ -267,7 +281,7 @@ public class GlRectDrawerTest {
>     final SurfaceTextureHelper surfaceTextureHelper = SurfaceTextureHelper.create(
>         "SurfaceTextureHelper test" /* threadName */, eglBase.getEglBaseContext());
>     final StubOesTextureProducer oesProducer = new StubOesTextureProducer(
>-        eglBase.getEglBaseContext(), surfaceTextureHelper.getSurfaceTexture(), WIDTH, HEIGHT);
>+        eglBase.getEglBaseContext(), surfaceTextureHelper, WIDTH, HEIGHT);
>     final SurfaceTextureHelperTest.MockTextureListener listener =
>         new SurfaceTextureHelperTest.MockTextureListener();
>     surfaceTextureHelper.startListening(listener);
>@@ -279,14 +293,15 @@ public class GlRectDrawerTest {
> 
>     // Draw the frame and block until an OES texture is delivered.
>     oesProducer.draw(rgbPlane);
>-    listener.waitForNewFrame();
>+    final VideoFrame.TextureBuffer textureBuffer = listener.waitForTextureBuffer();
> 
>     // Real test starts here.
>     // Draw the OES texture on the pixel buffer.
>     eglBase.makeCurrent();
>     final GlRectDrawer drawer = new GlRectDrawer();
>-    drawer.drawOes(listener.oesTextureId, listener.transformMatrix, WIDTH, HEIGHT,
>-        0 /* viewportX */, 0 /* viewportY */, WIDTH, HEIGHT);
>+    drawer.drawOes(textureBuffer.getTextureId(),
>+        RendererCommon.convertMatrixFromAndroidGraphicsMatrix(textureBuffer.getTransformMatrix()),
>+        WIDTH, HEIGHT, 0 /* viewportX */, 0 /* viewportY */, WIDTH, HEIGHT);
> 
>     // Download the pixels in the pixel buffer as RGBA. Not all platforms support RGB, e.g. Nexus 9.
>     final ByteBuffer rgbaData = ByteBuffer.allocateDirect(WIDTH * HEIGHT * 4);
>@@ -297,7 +312,7 @@ public class GlRectDrawerTest {
>     assertByteBufferEquals(WIDTH, HEIGHT, stripAlphaChannel(rgbaData), rgbPlane);
> 
>     drawer.release();
>-    surfaceTextureHelper.returnTextureFrame();
>+    textureBuffer.release();
>     oesProducer.release();
>     surfaceTextureHelper.dispose();
>     eglBase.release();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/HardwareVideoDecoderTest.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/HardwareVideoDecoderTest.java
>index f680102a7cd..e641aac06c2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/HardwareVideoDecoderTest.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/HardwareVideoDecoderTest.java
>@@ -21,6 +21,7 @@ import java.util.List;
> import java.util.concurrent.BlockingQueue;
> import java.util.concurrent.LinkedBlockingQueue;
> import java.util.concurrent.TimeUnit;
>+import javax.annotation.Nullable;
> import org.chromium.base.test.params.BaseJUnit4RunnerDelegate;
> import org.chromium.base.test.params.ParameterAnnotations.ClassParameter;
> import org.chromium.base.test.params.ParameterAnnotations.UseRunnerDelegate;
>@@ -39,18 +40,28 @@ public final class HardwareVideoDecoderTest {
> 
>   static {
>     CLASS_PARAMS.add(new ParameterSet()
>-                         .value("VP8" /* codecType */, false /* useEglContext */)
>+                         .value(/* codecName= */ "VP8", false /* useEglContext */)
>                          .name("VP8WithoutEglContext"));
>     CLASS_PARAMS.add(new ParameterSet()
>-                         .value("VP8" /* codecType */, true /* useEglContext */)
>+                         .value(/* codecName= */ "VP8", true /* useEglContext */)
>                          .name("VP8WithEglContext"));
>+    CLASS_PARAMS.add(new ParameterSet()
>+                         .value(/* codecName= */ "H264", false /* useEglContext */)
>+                         .name("H264WithoutEglContext"));
>+    CLASS_PARAMS.add(new ParameterSet()
>+                         .value(/* codecName= */ "H264", true /* useEglContext */)
>+                         .name("H264WithEglContext"));
>   }
> 
>-  private final String codecType;
>+  private final VideoCodecInfo codecType;
>   private final boolean useEglContext;
> 
>-  public HardwareVideoDecoderTest(String codecType, boolean useEglContext) {
>-    this.codecType = codecType;
>+  public HardwareVideoDecoderTest(String codecName, boolean useEglContext) {
>+    if (codecName.equals("H264")) {
>+      this.codecType = H264Utils.DEFAULT_H264_BASELINE_PROFILE_CODEC;
>+    } else {
>+      this.codecType = new VideoCodecInfo(codecName, new HashMap<>());
>+    }
>     this.useEglContext = useEglContext;
>   }
> 
>@@ -59,17 +70,17 @@ public final class HardwareVideoDecoderTest {
>   private static final int TEST_FRAME_COUNT = 10;
>   private static final int TEST_FRAME_WIDTH = 640;
>   private static final int TEST_FRAME_HEIGHT = 360;
>-  private static final VideoFrame.I420Buffer[] TEST_FRAMES = generateTestFrames();
>+  private VideoFrame.I420Buffer[] TEST_FRAMES;
> 
>   private static final boolean ENABLE_INTEL_VP8_ENCODER = true;
>   private static final boolean ENABLE_H264_HIGH_PROFILE = true;
>   private static final VideoEncoder.Settings ENCODER_SETTINGS =
>-      new VideoEncoder.Settings(1 /* core */, 640 /* width */, 480 /* height */, 300 /* kbps */,
>+      new VideoEncoder.Settings(1 /* core */, TEST_FRAME_WIDTH, TEST_FRAME_HEIGHT, 300 /* kbps */,
>           30 /* fps */, true /* automaticResizeOn */);
> 
>   private static final int DECODE_TIMEOUT_MS = 1000;
>   private static final VideoDecoder.Settings SETTINGS =
>-      new VideoDecoder.Settings(1 /* core */, 640 /* width */, 480 /* height */);
>+      new VideoDecoder.Settings(1 /* core */, TEST_FRAME_WIDTH, TEST_FRAME_HEIGHT);
> 
>   private static class MockDecodeCallback implements VideoDecoder.Callback {
>     private BlockingQueue<VideoFrame> frameQueue = new LinkedBlockingQueue<>();
>@@ -118,7 +129,7 @@ public final class HardwareVideoDecoderTest {
>     return new HardwareVideoDecoderFactory(eglContext);
>   }
> 
>-  private VideoDecoder createDecoder() {
>+  private @Nullable VideoDecoder createDecoder() {
>     VideoDecoderFactory factory =
>         createDecoderFactory(useEglContext ? eglBase.getEglBaseContext() : null);
>     return factory.createDecoder(codecType);
>@@ -127,8 +138,7 @@ public final class HardwareVideoDecoderTest {
>   private void encodeTestFrames() {
>     VideoEncoderFactory encoderFactory = new HardwareVideoEncoderFactory(
>         eglBase.getEglBaseContext(), ENABLE_INTEL_VP8_ENCODER, ENABLE_H264_HIGH_PROFILE);
>-    VideoEncoder encoder =
>-        encoderFactory.createEncoder(new VideoCodecInfo(codecType, new HashMap<>()));
>+    VideoEncoder encoder = encoderFactory.createEncoder(codecType);
>     HardwareVideoEncoderTest.MockEncoderCallback encodeCallback =
>         new HardwareVideoEncoderTest.MockEncoderCallback();
>     assertEquals(VideoCodecStatus.OK, encoder.initEncode(ENCODER_SETTINGS, encodeCallback));
>@@ -148,7 +158,9 @@ public final class HardwareVideoDecoderTest {
> 
>   @Before
>   public void setUp() {
>-    NativeLibrary.initialize(new NativeLibrary.DefaultLoader());
>+    NativeLibrary.initialize(new NativeLibrary.DefaultLoader(), TestConstants.NATIVE_LIBRARY);
>+
>+    TEST_FRAMES = generateTestFrames();
> 
>     eglBase = new EglBase14(null, EglBase.CONFIG_PLAIN);
>     eglBase.createDummyPbufferSurface();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/HardwareVideoEncoderTest.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/HardwareVideoEncoderTest.java
>index 5fbf239c447..7782fc26476 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/HardwareVideoEncoderTest.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/HardwareVideoEncoderTest.java
>@@ -26,6 +26,7 @@ import java.util.List;
> import java.util.concurrent.BlockingQueue;
> import java.util.concurrent.LinkedBlockingQueue;
> import java.util.concurrent.TimeUnit;
>+import javax.annotation.Nullable;
> import org.chromium.base.test.params.BaseJUnit4RunnerDelegate;
> import org.chromium.base.test.params.ParameterAnnotations.ClassParameter;
> import org.chromium.base.test.params.ParameterAnnotations.UseRunnerDelegate;
>@@ -87,7 +88,22 @@ public class HardwareVideoEncoderTest {
>     public void onEncodedFrame(EncodedImage frame, VideoEncoder.CodecSpecificInfo info) {
>       assertNotNull(frame);
>       assertNotNull(info);
>-      frameQueue.offer(frame);
>+
>+      // Make a copy because keeping a reference to the buffer is not allowed.
>+      final ByteBuffer bufferCopy = ByteBuffer.allocateDirect(frame.buffer.remaining());
>+      bufferCopy.put(frame.buffer);
>+      bufferCopy.rewind();
>+
>+      frameQueue.offer(EncodedImage.builder()
>+                           .setBuffer(bufferCopy)
>+                           .setEncodedWidth(frame.encodedWidth)
>+                           .setEncodedHeight(frame.encodedHeight)
>+                           .setCaptureTimeNs(frame.captureTimeNs)
>+                           .setFrameType(frame.frameType)
>+                           .setRotation(frame.rotation)
>+                           .setCompleteFrame(frame.completeFrame)
>+                           .setQp(frame.qp)
>+                           .createEncodedImage());
>     }
> 
>     public EncodedImage poll() {
>@@ -196,7 +212,6 @@ public class HardwareVideoEncoderTest {
> 
>     public MockI420Buffer(int width, int height, Runnable releaseCallback) {
>       super(width, height, releaseCallback);
>-      // We never release this but it is not a problem in practice because the release is a no-op.
>       realBuffer = JavaI420Buffer.allocate(width, height);
>     }
> 
>@@ -236,6 +251,18 @@ public class HardwareVideoEncoderTest {
>       return this;
>     }
> 
>+    @Override
>+    public void retain() {
>+      super.retain();
>+      realBuffer.retain();
>+    }
>+
>+    @Override
>+    public void release() {
>+      super.release();
>+      realBuffer.release();
>+    }
>+
>     @Override
>     public VideoFrame.Buffer cropAndScale(
>         int cropX, int cropY, int cropWidth, int cropHeight, int scaleWidth, int scaleHeight) {
>@@ -265,7 +292,7 @@ public class HardwareVideoEncoderTest {
>         eglContext, ENABLE_INTEL_VP8_ENCODER, ENABLE_H264_HIGH_PROFILE);
>   }
> 
>-  private VideoEncoder createEncoder() {
>+  private @Nullable VideoEncoder createEncoder() {
>     VideoEncoderFactory factory =
>         createEncoderFactory(useEglContext ? eglBase.getEglBaseContext() : null);
>     VideoCodecInfo[] supportedCodecs = factory.getSupportedCodecs();
>@@ -327,7 +354,7 @@ public class HardwareVideoEncoderTest {
>   // # Tests
>   @Before
>   public void setUp() {
>-    NativeLibrary.initialize(new NativeLibrary.DefaultLoader());
>+    NativeLibrary.initialize(new NativeLibrary.DefaultLoader(), TestConstants.NATIVE_LIBRARY);
> 
>     eglBase = new EglBase14(null, EglBase.CONFIG_PLAIN);
>     eglBase.createDummyPbufferSurface();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/LoggableTest.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/LoggableTest.java
>new file mode 100644
>index 00000000000..3831345cb2d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/LoggableTest.java
>@@ -0,0 +1,164 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc;
>+
>+import static org.junit.Assert.assertTrue;
>+import static org.junit.Assert.assertFalse;
>+
>+import android.support.test.InstrumentationRegistry;
>+import android.support.test.filters.SmallTest;
>+import android.support.test.runner.AndroidJUnit4;
>+import java.util.ArrayList;
>+import org.junit.Test;
>+import org.junit.runner.RunWith;
>+import org.webrtc.PeerConnectionFactory;
>+import org.webrtc.Logging.Severity;
>+import org.webrtc.Loggable;
>+
>+@RunWith(AndroidJUnit4.class)
>+public class LoggableTest {
>+  private static String TAG = "LoggableTest";
>+  private static String NATIVE_FILENAME_TAG = "loggable_test.cc";
>+
>+  private static class MockLoggable implements Loggable {
>+    private ArrayList<String> messages = new ArrayList<>();
>+    private ArrayList<Severity> sevs = new ArrayList<>();
>+    private ArrayList<String> tags = new ArrayList<>();
>+
>+    @Override
>+    public void onLogMessage(String message, Severity sev, String tag) {
>+      messages.add(message);
>+      sevs.add(sev);
>+      tags.add(tag);
>+    }
>+
>+    public boolean isMessageReceived(String message) {
>+      for (int i = 0; i < messages.size(); i++) {
>+        if (messages.get(i).contains(message)) {
>+          return true;
>+        }
>+      }
>+      return false;
>+    }
>+
>+    public boolean isMessageReceived(String message, Severity sev, String tag) {
>+      for (int i = 0; i < messages.size(); i++) {
>+        if (messages.get(i).contains(message) && sevs.get(i) == sev && tags.get(i).equals(tag)) {
>+          return true;
>+        }
>+      }
>+      return false;
>+    }
>+  }
>+
>+  private final MockLoggable mockLoggable = new MockLoggable();
>+
>+  @Test
>+  @SmallTest
>+  public void testLoggableSetWithoutError() throws InterruptedException {
>+    PeerConnectionFactory.initialize(PeerConnectionFactory.InitializationOptions
>+                                         .builder(InstrumentationRegistry.getTargetContext())
>+                                         .setInjectableLogger(mockLoggable, Severity.LS_INFO)
>+                                         .setNativeLibraryName(TestConstants.NATIVE_LIBRARY)
>+                                         .createInitializationOptions());
>+  }
>+
>+  @Test
>+  @SmallTest
>+  public void testMessageIsLoggedCorrectly() throws InterruptedException {
>+    PeerConnectionFactory.initialize(PeerConnectionFactory.InitializationOptions
>+                                         .builder(InstrumentationRegistry.getTargetContext())
>+                                         .setInjectableLogger(mockLoggable, Severity.LS_INFO)
>+                                         .setNativeLibraryName(TestConstants.NATIVE_LIBRARY)
>+                                         .createInitializationOptions());
>+    String msg = "Message that should be logged";
>+    Logging.d(TAG, msg);
>+    assertTrue(mockLoggable.isMessageReceived(msg, Severity.LS_INFO, TAG));
>+  }
>+
>+  @Test
>+  @SmallTest
>+  public void testLowSeverityIsFiltered() throws InterruptedException {
>+    // Set severity to LS_WARNING to filter out LS_INFO and below.
>+    PeerConnectionFactory.initialize(PeerConnectionFactory.InitializationOptions
>+                                         .builder(InstrumentationRegistry.getTargetContext())
>+                                         .setInjectableLogger(mockLoggable, Severity.LS_WARNING)
>+                                         .setNativeLibraryName(TestConstants.NATIVE_LIBRARY)
>+                                         .createInitializationOptions());
>+    String msg = "Message that should NOT be logged";
>+    Logging.d(TAG, msg);
>+    assertFalse(mockLoggable.isMessageReceived(msg));
>+  }
>+
>+  @Test
>+  @SmallTest
>+  public void testLoggableDoesNotReceiveMessagesAfterUnsetting() {
>+    PeerConnectionFactory.initialize(PeerConnectionFactory.InitializationOptions
>+                                         .builder(InstrumentationRegistry.getTargetContext())
>+                                         .setInjectableLogger(mockLoggable, Severity.LS_INFO)
>+                                         .setNativeLibraryName(TestConstants.NATIVE_LIBRARY)
>+                                         .createInitializationOptions());
>+    // Reinitialize without Loggable
>+    PeerConnectionFactory.initialize(PeerConnectionFactory.InitializationOptions
>+                                         .builder(InstrumentationRegistry.getTargetContext())
>+                                         .setNativeLibraryName(TestConstants.NATIVE_LIBRARY)
>+                                         .createInitializationOptions());
>+    String msg = "Message that should NOT be logged";
>+    Logging.d(TAG, msg);
>+    assertFalse(mockLoggable.isMessageReceived(msg));
>+  }
>+
>+  @Test
>+  @SmallTest
>+  public void testNativeMessageIsLoggedCorrectly() throws InterruptedException {
>+    PeerConnectionFactory.initialize(PeerConnectionFactory.InitializationOptions
>+                                         .builder(InstrumentationRegistry.getTargetContext())
>+                                         .setInjectableLogger(mockLoggable, Severity.LS_INFO)
>+                                         .setNativeLibraryName(TestConstants.NATIVE_LIBRARY)
>+                                         .createInitializationOptions());
>+    String msg = "Message that should be logged";
>+    nativeLogInfoTestMessage(msg);
>+    assertTrue(mockLoggable.isMessageReceived(msg, Severity.LS_INFO, NATIVE_FILENAME_TAG));
>+  }
>+
>+  @Test
>+  @SmallTest
>+  public void testNativeLowSeverityIsFiltered() throws InterruptedException {
>+    PeerConnectionFactory.initialize(PeerConnectionFactory.InitializationOptions
>+                                         .builder(InstrumentationRegistry.getTargetContext())
>+                                         .setInjectableLogger(mockLoggable, Severity.LS_WARNING)
>+                                         .setNativeLibraryName(TestConstants.NATIVE_LIBRARY)
>+                                         .createInitializationOptions());
>+    String msg = "Message that should NOT be logged";
>+    nativeLogInfoTestMessage(msg);
>+    assertFalse(mockLoggable.isMessageReceived(msg));
>+  }
>+
>+  @Test
>+  @SmallTest
>+  public void testNativeLoggableDoesNotReceiveMessagesAfterUnsetting() {
>+    PeerConnectionFactory.initialize(PeerConnectionFactory.InitializationOptions
>+                                         .builder(InstrumentationRegistry.getTargetContext())
>+                                         .setInjectableLogger(mockLoggable, Severity.LS_INFO)
>+                                         .setNativeLibraryName(TestConstants.NATIVE_LIBRARY)
>+                                         .createInitializationOptions());
>+    // Reinitialize without Loggable
>+    PeerConnectionFactory.initialize(PeerConnectionFactory.InitializationOptions
>+                                         .builder(InstrumentationRegistry.getTargetContext())
>+                                         .setNativeLibraryName(TestConstants.NATIVE_LIBRARY)
>+                                         .createInitializationOptions());
>+    String msg = "Message that should NOT be logged";
>+    nativeLogInfoTestMessage(msg);
>+    assertFalse(mockLoggable.isMessageReceived(msg));
>+  }
>+
>+  private static native void nativeLogInfoTestMessage(String message);
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/MediaCodecVideoEncoderTest.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/MediaCodecVideoEncoderTest.java
>index c48bc986868..a91f88e04c8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/MediaCodecVideoEncoderTest.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/MediaCodecVideoEncoderTest.java
>@@ -40,8 +40,8 @@ public class MediaCodecVideoEncoderTest {
>       return;
>     }
>     MediaCodecVideoEncoder encoder = new MediaCodecVideoEncoder();
>-    assertTrue(encoder.initEncode(
>-        MediaCodecVideoEncoder.VideoCodecType.VIDEO_CODEC_VP8, profile, 640, 480, 300, 30, null));
>+    assertTrue(encoder.initEncode(MediaCodecVideoEncoder.VideoCodecType.VIDEO_CODEC_VP8, profile,
>+        640, 480, 300, 30, /* useSurface= */ false));
>     encoder.release();
>   }
> 
>@@ -53,10 +53,12 @@ public class MediaCodecVideoEncoderTest {
>       return;
>     }
>     EglBase14 eglBase = new EglBase14(null, EglBase.CONFIG_PLAIN);
>+    MediaCodecVideoEncoder.setEglContext(eglBase.getEglBaseContext());
>     MediaCodecVideoEncoder encoder = new MediaCodecVideoEncoder();
>     assertTrue(encoder.initEncode(MediaCodecVideoEncoder.VideoCodecType.VIDEO_CODEC_VP8, profile,
>-        640, 480, 300, 30, eglBase.getEglBaseContext()));
>+        640, 480, 300, 30, /* useSurface= */ true));
>     encoder.release();
>+    MediaCodecVideoEncoder.disposeEglContext();
>     eglBase.release();
>   }
> 
>@@ -68,13 +70,15 @@ public class MediaCodecVideoEncoderTest {
>       return;
>     }
>     MediaCodecVideoEncoder encoder = new MediaCodecVideoEncoder();
>-    assertTrue(encoder.initEncode(
>-        MediaCodecVideoEncoder.VideoCodecType.VIDEO_CODEC_VP8, profile, 640, 480, 300, 30, null));
>+    assertTrue(encoder.initEncode(MediaCodecVideoEncoder.VideoCodecType.VIDEO_CODEC_VP8, profile,
>+        640, 480, 300, 30, /* useSurface= */ false));
>     encoder.release();
>     EglBase14 eglBase = new EglBase14(null, EglBase.CONFIG_PLAIN);
>+    MediaCodecVideoEncoder.setEglContext(eglBase.getEglBaseContext());
>     assertTrue(encoder.initEncode(MediaCodecVideoEncoder.VideoCodecType.VIDEO_CODEC_VP8, profile,
>-        640, 480, 300, 30, eglBase.getEglBaseContext()));
>+        640, 480, 300, 30, /* useSurface= */ true));
>     encoder.release();
>+    MediaCodecVideoEncoder.disposeEglContext();
>     eglBase.release();
>   }
> 
>@@ -94,7 +98,7 @@ public class MediaCodecVideoEncoderTest {
>     MediaCodecVideoEncoder encoder = new MediaCodecVideoEncoder();
> 
>     assertTrue(encoder.initEncode(MediaCodecVideoEncoder.VideoCodecType.VIDEO_CODEC_VP8, profile,
>-        width, height, 300, 30, null));
>+        width, height, 300, 30, /* useSurface= */ false));
>     ByteBuffer[] inputBuffers = encoder.getInputBuffers();
>     assertNotNull(inputBuffers);
>     assertTrue(min_size <= inputBuffers[0].capacity());
>@@ -121,50 +125,4 @@ public class MediaCodecVideoEncoderTest {
> 
>     encoder.release();
>   }
>-
>-  @Test
>-  @SmallTest
>-  public void testEncoderUsingTextures() throws InterruptedException {
>-    if (!MediaCodecVideoEncoder.isVp8HwSupportedUsingTextures()) {
>-      Log.i(TAG, "Hardware does not support VP8 encoding, skipping testEncoderUsingTextures");
>-      return;
>-    }
>-
>-    final int width = 640;
>-    final int height = 480;
>-    final long presentationTs = 2;
>-
>-    final EglBase14 eglOesBase = new EglBase14(null, EglBase.CONFIG_PIXEL_BUFFER);
>-    eglOesBase.createDummyPbufferSurface();
>-    eglOesBase.makeCurrent();
>-    int oesTextureId = GlUtil.generateTexture(GLES11Ext.GL_TEXTURE_EXTERNAL_OES);
>-
>-    // TODO(perkj): This test is week since we don't fill the texture with valid data with correct
>-    // width and height and verify the encoded data. Fill the OES texture and figure out a way to
>-    // verify that the output make sense.
>-
>-    MediaCodecVideoEncoder encoder = new MediaCodecVideoEncoder();
>-
>-    assertTrue(encoder.initEncode(MediaCodecVideoEncoder.VideoCodecType.VIDEO_CODEC_VP8, profile,
>-        width, height, 300, 30, eglOesBase.getEglBaseContext()));
>-    assertTrue(
>-        encoder.encodeTexture(true, oesTextureId, RendererCommon.identityMatrix(), presentationTs));
>-    GlUtil.checkNoGLES2Error("encodeTexture");
>-
>-    // It should be Ok to delete the texture after calling encodeTexture.
>-    GLES20.glDeleteTextures(1, new int[] {oesTextureId}, 0);
>-
>-    OutputBufferInfo info = encoder.dequeueOutputBuffer();
>-    while (info == null) {
>-      info = encoder.dequeueOutputBuffer();
>-      Thread.sleep(20);
>-    }
>-    assertTrue(info.index != -1);
>-    assertTrue(info.buffer.capacity() > 0);
>-    assertEquals(presentationTs, info.presentationTimestampUs);
>-    encoder.releaseOutputBuffer(info.index);
>-
>-    encoder.release();
>-    eglOesBase.release();
>-  }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/NetworkMonitorTest.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/NetworkMonitorTest.java
>index 579fba76b67..e4a0e6ae12f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/NetworkMonitorTest.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/NetworkMonitorTest.java
>@@ -10,8 +10,10 @@
> 
> package org.webrtc;
> 
>+import static org.junit.Assert.assertEquals;
> import static org.junit.Assert.assertFalse;
> import static org.junit.Assert.assertNotNull;
>+import static org.junit.Assert.assertNull;
> import static org.junit.Assert.assertTrue;
> import static org.webrtc.NetworkMonitorAutoDetect.ConnectionType;
> import static org.webrtc.NetworkMonitorAutoDetect.ConnectivityManagerDelegate;
>@@ -33,6 +35,7 @@ import android.support.test.annotation.UiThreadTest;
> import android.support.test.filters.MediumTest;
> import android.support.test.filters.SmallTest;
> import android.support.test.rule.UiThreadTestRule;
>+import javax.annotation.Nullable;
> import org.chromium.base.test.BaseJUnit4ClassRunner;
> import org.junit.Before;
> import org.junit.Rule;
>@@ -43,7 +46,7 @@ import org.junit.runner.RunWith;
>  * Tests for org.webrtc.NetworkMonitor.
>  *
>  * TODO(deadbeef): These tests don't cover the interaction between
>- * NetworkManager.java and androidnetworkmonitor_jni.cc, which is how this
>+ * NetworkManager.java and androidnetworkmonitor.cc, which is how this
>  * class is used in practice in WebRTC.
>  */
> @SuppressLint("NewApi")
>@@ -78,10 +81,13 @@ public class NetworkMonitorTest {
>     private boolean activeNetworkExists;
>     private int networkType;
>     private int networkSubtype;
>+    private int underlyingNetworkTypeForVpn;
>+    private int underlyingNetworkSubtypeForVpn;
> 
>     @Override
>     public NetworkState getNetworkState() {
>-      return new NetworkState(activeNetworkExists, networkType, networkSubtype);
>+      return new NetworkState(activeNetworkExists, networkType, networkSubtype,
>+          underlyingNetworkTypeForVpn, underlyingNetworkSubtypeForVpn);
>     }
> 
>     // Dummy implementations to avoid NullPointerExceptions in default implementations:
>@@ -98,7 +104,7 @@ public class NetworkMonitorTest {
> 
>     @Override
>     public NetworkState getNetworkState(Network network) {
>-      return new NetworkState(false, -1, -1);
>+      return new NetworkState(false, -1, -1, -1, -1);
>     }
> 
>     public void setActiveNetworkExists(boolean networkExists) {
>@@ -112,6 +118,14 @@ public class NetworkMonitorTest {
>     public void setNetworkSubtype(int networkSubtype) {
>       this.networkSubtype = networkSubtype;
>     }
>+
>+    public void setUnderlyingNetworkType(int underlyingNetworkTypeForVpn) {
>+      this.underlyingNetworkTypeForVpn = underlyingNetworkTypeForVpn;
>+    }
>+
>+    public void setUnderlyingNetworkSubype(int underlyingNetworkSubtypeForVpn) {
>+      this.underlyingNetworkSubtypeForVpn = underlyingNetworkSubtypeForVpn;
>+    }
>   }
> 
>   /**
>@@ -145,7 +159,7 @@ public class NetworkMonitorTest {
>   }
> 
>   private static final Object lock = new Object();
>-  private static Handler uiThreadHandler = null;
>+  private static @Nullable Handler uiThreadHandler = null;
> 
>   private NetworkMonitorAutoDetect receiver;
>   private MockConnectivityManagerDelegate connectivityDelegate;
>@@ -153,10 +167,11 @@ public class NetworkMonitorTest {
> 
>   private static Handler getUiThreadHandler() {
>     synchronized (lock) {
>-      if (uiThreadHandler == null) {
>-        uiThreadHandler = new Handler(Looper.getMainLooper());
>+      Handler handler = uiThreadHandler;
>+      if (handler != null) {
>+        return handler;
>       }
>-      return uiThreadHandler;
>+      return uiThreadHandler = new Handler(Looper.getMainLooper());
>     }
>   }
> 
>@@ -165,9 +180,7 @@ public class NetworkMonitorTest {
>    */
>   private void createTestMonitor() {
>     Context context = InstrumentationRegistry.getTargetContext();
>-    NetworkMonitor.resetInstanceForTests();
>-    NetworkMonitor.createAutoDetectorForTest();
>-    receiver = NetworkMonitor.getAutoDetectorForTest();
>+    receiver = NetworkMonitor.createAndSetAutoDetectForTest(context);
>     assertNotNull(receiver);
> 
>     connectivityDelegate = new MockConnectivityManagerDelegate();
>@@ -288,4 +301,20 @@ public class NetworkMonitorTest {
>         new NetworkMonitorAutoDetect(observer, InstrumentationRegistry.getTargetContext());
>     ncn.getDefaultNetId();
>   }
>+
>+  /**
>+   * Tests startMonitoring and stopMonitoring correctly set the autoDetect and number of observers.
>+   */
>+  @Test
>+  @SmallTest
>+  public void testStartStopMonitoring() {
>+    NetworkMonitor networkMonitor = NetworkMonitor.getInstance();
>+    Context context = ContextUtils.getApplicationContext();
>+    networkMonitor.startMonitoring(context);
>+    assertEquals(1, networkMonitor.getNumObservers());
>+    assertNotNull(networkMonitor.getNetworkMonitorAutoDetect());
>+    networkMonitor.stopMonitoring();
>+    assertEquals(0, networkMonitor.getNumObservers());
>+    assertNull(networkMonitor.getNetworkMonitorAutoDetect());
>+  }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/PeerConnectionFactoryTest.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/PeerConnectionFactoryTest.java
>index 01e618dabb8..907067ae21f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/PeerConnectionFactoryTest.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/PeerConnectionFactoryTest.java
>@@ -23,6 +23,7 @@ public class PeerConnectionFactoryTest {
>   public void testInitialize() {
>     PeerConnectionFactory.initialize(PeerConnectionFactory.InitializationOptions
>                                          .builder(InstrumentationRegistry.getTargetContext())
>+                                         .setNativeLibraryName(TestConstants.NATIVE_LIBRARY)
>                                          .createInitializationOptions());
>   }
> 
>@@ -31,9 +32,11 @@ public class PeerConnectionFactoryTest {
>   public void testInitializeTwice() {
>     PeerConnectionFactory.initialize(PeerConnectionFactory.InitializationOptions
>                                          .builder(InstrumentationRegistry.getTargetContext())
>+                                         .setNativeLibraryName(TestConstants.NATIVE_LIBRARY)
>                                          .createInitializationOptions());
>     PeerConnectionFactory.initialize(PeerConnectionFactory.InitializationOptions
>                                          .builder(InstrumentationRegistry.getTargetContext())
>+                                         .setNativeLibraryName(TestConstants.NATIVE_LIBRARY)
>                                          .createInitializationOptions());
>   }
> 
>@@ -43,10 +46,12 @@ public class PeerConnectionFactoryTest {
>     PeerConnectionFactory.initialize(PeerConnectionFactory.InitializationOptions
>                                          .builder(InstrumentationRegistry.getTargetContext())
>                                          .setEnableInternalTracer(true)
>+                                         .setNativeLibraryName(TestConstants.NATIVE_LIBRARY)
>                                          .createInitializationOptions());
>     PeerConnectionFactory.initialize(PeerConnectionFactory.InitializationOptions
>                                          .builder(InstrumentationRegistry.getTargetContext())
>                                          .setEnableInternalTracer(true)
>+                                         .setNativeLibraryName(TestConstants.NATIVE_LIBRARY)
>                                          .createInitializationOptions());
>   }
> 
>@@ -56,6 +61,7 @@ public class PeerConnectionFactoryTest {
>     PeerConnectionFactory.initialize(PeerConnectionFactory.InitializationOptions
>                                          .builder(InstrumentationRegistry.getTargetContext())
>                                          .setEnableInternalTracer(true)
>+                                         .setNativeLibraryName(TestConstants.NATIVE_LIBRARY)
>                                          .createInitializationOptions());
>     PeerConnectionFactory.shutdownInternalTracer();
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/PeerConnectionTest.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/PeerConnectionTest.java
>index 2cd3e126e39..d1869794671 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/PeerConnectionTest.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/PeerConnectionTest.java
>@@ -15,6 +15,7 @@ import static org.junit.Assert.assertFalse;
> import static org.junit.Assert.assertNotNull;
> import static org.junit.Assert.assertNull;
> import static org.junit.Assert.assertTrue;
>+import static org.junit.Assert.fail;
> 
> import android.support.test.InstrumentationRegistry;
> import android.support.test.filters.MediumTest;
>@@ -23,17 +24,18 @@ import java.io.File;
> import java.lang.ref.WeakReference;
> import java.nio.ByteBuffer;
> import java.nio.charset.Charset;
>+import java.util.ArrayDeque;
>+import java.util.ArrayList;
> import java.util.Arrays;
> import java.util.HashSet;
> import java.util.IdentityHashMap;
>-import java.util.ArrayList;
> import java.util.List;
>-import java.util.Queue;
>-import java.util.ArrayDeque;
> import java.util.Map;
>+import java.util.Queue;
> import java.util.TreeSet;
> import java.util.concurrent.CountDownLatch;
> import java.util.concurrent.TimeUnit;
>+import javax.annotation.Nullable;
> import org.chromium.base.test.BaseJUnit4ClassRunner;
> import org.chromium.base.test.util.DisabledTest;
> import org.junit.Before;
>@@ -48,18 +50,19 @@ import org.webrtc.PeerConnection.SignalingState;
> @RunWith(BaseJUnit4ClassRunner.class)
> public class PeerConnectionTest {
>   private static final int TIMEOUT_SECONDS = 20;
>-  private TreeSet<String> threadsBeforeTest = null;
>+  private @Nullable TreeSet<String> threadsBeforeTest = null;
> 
>   @Before
>   public void setUp() {
>     PeerConnectionFactory.initialize(PeerConnectionFactory.InitializationOptions
>                                          .builder(InstrumentationRegistry.getTargetContext())
>+                                         .setNativeLibraryName(TestConstants.NATIVE_LIBRARY)
>                                          .createInitializationOptions());
>   }
> 
>   private static class ObserverExpectations
>-      implements PeerConnection.Observer, VideoRenderer.Callbacks, DataChannel.Observer,
>-                 StatsObserver, RTCStatsCollectorCallback, RtpReceiver.Observer {
>+      implements PeerConnection.Observer, VideoSink, DataChannel.Observer, StatsObserver,
>+                 RTCStatsCollectorCallback, RtpReceiver.Observer {
>     private final String name;
>     private int expectedIceCandidates = 0;
>     private int expectedErrors = 0;
>@@ -74,7 +77,7 @@ public class PeerConnectionTest {
>     private Queue<String> expectedAddStreamLabels = new ArrayDeque<>();
>     private Queue<String> expectedRemoveStreamLabels = new ArrayDeque<>();
>     private final List<IceCandidate> gotIceCandidates = new ArrayList<>();
>-    private Map<MediaStream, WeakReference<VideoRenderer>> renderers = new IdentityHashMap<>();
>+    private Map<MediaStream, WeakReference<VideoSink>> videoSinks = new IdentityHashMap<>();
>     private DataChannel dataChannel;
>     private Queue<DataChannel.Buffer> expectedBuffers = new ArrayDeque<>();
>     private Queue<DataChannel.State> expectedStateChanges = new ArrayDeque<>();
>@@ -139,13 +142,12 @@ public class PeerConnectionTest {
>     @Override
>     // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
>     @SuppressWarnings("NoSynchronizedMethodCheck")
>-    public synchronized void renderFrame(VideoRenderer.I420Frame frame) {
>+    public synchronized void onFrame(VideoFrame frame) {
>       assertTrue(expectedWidth > 0);
>       assertTrue(expectedHeight > 0);
>-      assertEquals(expectedWidth, frame.rotatedWidth());
>-      assertEquals(expectedHeight, frame.rotatedHeight());
>+      assertEquals(expectedWidth, frame.getRotatedWidth());
>+      assertEquals(expectedHeight, frame.getRotatedHeight());
>       --expectedFramesDelivered;
>-      VideoRenderer.renderFrameDone(frame);
>     }
> 
>     // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
>@@ -224,15 +226,14 @@ public class PeerConnectionTest {
>     // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
>     @SuppressWarnings("NoSynchronizedMethodCheck")
>     public synchronized void onAddStream(MediaStream stream) {
>-      assertEquals(expectedAddStreamLabels.remove(), stream.label());
>+      assertEquals(expectedAddStreamLabels.remove(), stream.getId());
>       for (AudioTrack track : stream.audioTracks) {
>         assertEquals("audio", track.kind());
>       }
>       for (VideoTrack track : stream.videoTracks) {
>         assertEquals("video", track.kind());
>-        VideoRenderer renderer = createVideoRenderer(this);
>-        track.addRenderer(renderer);
>-        assertNull(renderers.put(stream, new WeakReference<VideoRenderer>(renderer)));
>+        track.addSink(this);
>+        assertNull(videoSinks.put(stream, new WeakReference<VideoSink>(this)));
>       }
>       gotRemoteStreams.add(stream);
>     }
>@@ -247,12 +248,13 @@ public class PeerConnectionTest {
>     // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
>     @SuppressWarnings("NoSynchronizedMethodCheck")
>     public synchronized void onRemoveStream(MediaStream stream) {
>-      assertEquals(expectedRemoveStreamLabels.remove(), stream.label());
>-      WeakReference<VideoRenderer> renderer = renderers.remove(stream);
>-      assertNotNull(renderer);
>-      assertNotNull(renderer.get());
>-      assertEquals(1, stream.videoTracks.size());
>-      stream.videoTracks.get(0).removeRenderer(renderer.get());
>+      assertEquals(expectedRemoveStreamLabels.remove(), stream.getId());
>+      WeakReference<VideoSink> videoSink = videoSinks.remove(stream);
>+      assertNotNull(videoSink);
>+      assertNotNull(videoSink.get());
>+      for (VideoTrack videoTrack : stream.videoTracks) {
>+        videoTrack.removeSink(videoSink.get());
>+      }
>       gotRemoteStreams.remove(stream);
>     }
> 
>@@ -509,7 +511,7 @@ public class PeerConnectionTest {
> 
>   // Sets the expected resolution for an ObserverExpectations once a frame
>   // has been captured.
>-  private static class ExpectedResolutionSetter implements VideoRenderer.Callbacks {
>+  private static class ExpectedResolutionSetter implements VideoSink {
>     private ObserverExpectations observer;
> 
>     public ExpectedResolutionSetter(ObserverExpectations observer) {
>@@ -519,19 +521,20 @@ public class PeerConnectionTest {
>     @Override
>     // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
>     @SuppressWarnings("NoSynchronizedMethodCheck")
>-    public synchronized void renderFrame(VideoRenderer.I420Frame frame) {
>+    public synchronized void onFrame(VideoFrame frame) {
>       // Because different camera devices (fake & physical) produce different
>       // resolutions, we only sanity-check the set sizes,
>-      assertTrue(frame.rotatedWidth() > 0);
>-      assertTrue(frame.rotatedHeight() > 0);
>-      observer.setExpectedResolution(frame.rotatedWidth(), frame.rotatedHeight());
>+      assertTrue(frame.getRotatedWidth() > 0);
>+      assertTrue(frame.getRotatedHeight() > 0);
>+      observer.setExpectedResolution(frame.getRotatedWidth(), frame.getRotatedHeight());
>+      frame.retain();
>     }
>   }
> 
>   private static class SdpObserverLatch implements SdpObserver {
>     private boolean success = false;
>-    private SessionDescription sdp = null;
>-    private String error = null;
>+    private @Nullable SessionDescription sdp = null;
>+    private @Nullable String error = null;
>     private CountDownLatch latch = new CountDownLatch(1);
> 
>     public SdpObserverLatch() {}
>@@ -572,32 +575,27 @@ public class PeerConnectionTest {
>       return success;
>     }
> 
>-    public SessionDescription getSdp() {
>+    public @Nullable SessionDescription getSdp() {
>       return sdp;
>     }
> 
>-    public String getError() {
>+    public @Nullable String getError() {
>       return error;
>     }
>   }
> 
>   static int videoWindowsMapped = -1;
> 
>-  private static VideoRenderer createVideoRenderer(VideoRenderer.Callbacks videoCallbacks) {
>-    return new VideoRenderer(videoCallbacks);
>-  }
>-
>   // Return a weak reference to test that ownership is correctly held by
>   // PeerConnection, not by test code.
>   private static WeakReference<MediaStream> addTracksToPC(PeerConnectionFactory factory,
>       PeerConnection pc, VideoSource videoSource, String streamLabel, String videoTrackId,
>-      String audioTrackId, VideoRenderer.Callbacks videoCallbacks) {
>+      String audioTrackId, VideoSink videoSink) {
>     MediaStream lMS = factory.createLocalMediaStream(streamLabel);
>     VideoTrack videoTrack = factory.createVideoTrack(videoTrackId, videoSource);
>     assertNotNull(videoTrack);
>-    VideoRenderer videoRenderer = createVideoRenderer(videoCallbacks);
>-    assertNotNull(videoRenderer);
>-    videoTrack.addRenderer(videoRenderer);
>+    assertNotNull(videoSink);
>+    videoTrack.addSink(videoSink);
>     lMS.addTrack(videoTrack);
>     // Just for fun, let's remove and re-add the track.
>     lMS.removeTrack(videoTrack);
>@@ -638,7 +636,7 @@ public class PeerConnectionTest {
>   // - audit each place that uses |constraints| for specifying non-trivial
>   //   constraints (and ensure they're honored).
>   // - test error cases
>-  // - ensure reasonable coverage of _jni.cc is achieved.  Coverage is
>+  // - ensure reasonable coverage of jni code is achieved.  Coverage is
>   //   extra-important because of all the free-text (class/method names, etc)
>   //   in JNI-style programming; make sure no typos!
>   // - Test that shutdown mid-interaction is crash-free.
>@@ -647,8 +645,7 @@ public class PeerConnectionTest {
>   @Test
>   @SmallTest
>   public void testCreationWithConfig() throws Exception {
>-    PeerConnectionFactory.Options options = new PeerConnectionFactory.Options();
>-    PeerConnectionFactory factory = new PeerConnectionFactory(options);
>+    PeerConnectionFactory factory = PeerConnectionFactory.builder().createPeerConnectionFactory();
>     List<PeerConnection.IceServer> iceServers = Arrays.asList(
>         PeerConnection.IceServer.builder("stun:stun.l.google.com:19302").createIceServer(),
>         PeerConnection.IceServer.builder("turn:fake.example.com")
>@@ -661,10 +658,8 @@ public class PeerConnectionTest {
>     config.continualGatheringPolicy = PeerConnection.ContinualGatheringPolicy.GATHER_CONTINUALLY;
>     config.iceRegatherIntervalRange = new PeerConnection.IntervalRange(1000, 2000);
> 
>-    MediaConstraints constraints = new MediaConstraints();
>     ObserverExpectations offeringExpectations = new ObserverExpectations("PCTest:offerer");
>-    PeerConnection offeringPC =
>-        factory.createPeerConnection(config, constraints, offeringExpectations);
>+    PeerConnection offeringPC = factory.createPeerConnection(config, offeringExpectations);
>     assertNotNull(offeringPC);
>   }
> 
>@@ -676,10 +671,8 @@ public class PeerConnectionTest {
>     // have those.
>     PeerConnectionFactory.Options options = new PeerConnectionFactory.Options();
>     options.networkIgnoreMask = 0;
>-    PeerConnectionFactory factory = new PeerConnectionFactory(options);
>-
>-    MediaConstraints pcConstraints = new MediaConstraints();
>-    pcConstraints.mandatory.add(new MediaConstraints.KeyValuePair("DtlsSrtpKeyAgreement", "true"));
>+    PeerConnectionFactory factory =
>+        PeerConnectionFactory.builder().setOptions(options).createPeerConnectionFactory();
> 
>     List<PeerConnection.IceServer> iceServers = new ArrayList<>();
>     iceServers.add(
>@@ -688,14 +681,16 @@ public class PeerConnectionTest {
>                        .setUsername("fakeUsername")
>                        .setPassword("fakePassword")
>                        .createIceServer());
>+
>+    PeerConnection.RTCConfiguration rtcConfig = new PeerConnection.RTCConfiguration(iceServers);
>+    rtcConfig.enableDtlsSrtp = true;
>+
>     ObserverExpectations offeringExpectations = new ObserverExpectations("PCTest:offerer");
>-    PeerConnection offeringPC =
>-        factory.createPeerConnection(iceServers, pcConstraints, offeringExpectations);
>+    PeerConnection offeringPC = factory.createPeerConnection(rtcConfig, offeringExpectations);
>     assertNotNull(offeringPC);
> 
>     ObserverExpectations answeringExpectations = new ObserverExpectations("PCTest:answerer");
>-    PeerConnection answeringPC =
>-        factory.createPeerConnection(iceServers, pcConstraints, answeringExpectations);
>+    PeerConnection answeringPC = factory.createPeerConnection(rtcConfig, answeringExpectations);
>     assertNotNull(answeringPC);
> 
>     // We want to use the same camera for offerer & answerer, so create it here
>@@ -703,7 +698,11 @@ public class PeerConnectionTest {
>     final CameraEnumerator enumerator = new Camera1Enumerator(false /* captureToTexture */);
>     final VideoCapturer videoCapturer =
>         enumerator.createCapturer(enumerator.getDeviceNames()[0], null /* eventsHandler */);
>-    final VideoSource videoSource = factory.createVideoSource(videoCapturer);
>+    final SurfaceTextureHelper surfaceTextureHelper =
>+        SurfaceTextureHelper.create("SurfaceTextureHelper", /* sharedContext= */ null);
>+    final VideoSource videoSource = factory.createVideoSource(/* isScreencast= */ false);
>+    videoCapturer.initialize(surfaceTextureHelper, InstrumentationRegistry.getTargetContext(),
>+        videoSource.getCapturerObserver());
>     videoCapturer.startCapture(640, 480, 30);
> 
>     offeringExpectations.expectRenegotiationNeeded();
>@@ -851,8 +850,10 @@ public class PeerConnectionTest {
>     assertNotNull(rtpParameters);
>     assertEquals(1, rtpParameters.encodings.size());
>     assertNull(rtpParameters.encodings.get(0).maxBitrateBps);
>+    assertNull(rtpParameters.encodings.get(0).minBitrateBps);
> 
>     rtpParameters.encodings.get(0).maxBitrateBps = 300000;
>+    rtpParameters.encodings.get(0).minBitrateBps = 100000;
>     assertTrue(videoSender.setParameters(rtpParameters));
> 
>     // Create a DTMF sender.
>@@ -864,6 +865,7 @@ public class PeerConnectionTest {
>     // Verify that we can read back the updated value.
>     rtpParameters = videoSender.getParameters();
>     assertEquals(300000, (int) rtpParameters.encodings.get(0).maxBitrateBps);
>+    assertEquals(100000, (int) rtpParameters.encodings.get(0).minBitrateBps);
> 
>     // Test send & receive UTF-8 text.
>     answeringExpectations.expectMessage(
>@@ -891,6 +893,8 @@ public class PeerConnectionTest {
>     answeringExpectations.expectStateChange(DataChannel.State.CLOSED);
>     answeringExpectations.dataChannel.close();
>     offeringExpectations.dataChannel.close();
>+    assertTrue(offeringExpectations.waitForAllExpectationsToBeSatisfied(TIMEOUT_SECONDS));
>+    assertTrue(answeringExpectations.waitForAllExpectationsToBeSatisfied(TIMEOUT_SECONDS));
> 
>     // Test SetBitrate.
>     assertTrue(offeringPC.setBitrate(100000, 5000000, 500000000));
>@@ -905,6 +909,7 @@ public class PeerConnectionTest {
>     videoCapturer.stopCapture();
>     videoCapturer.dispose();
>     videoSource.dispose();
>+    surfaceTextureHelper.dispose();
>     factory.dispose();
>     System.gc();
>   }
>@@ -916,10 +921,8 @@ public class PeerConnectionTest {
>     // have those.
>     PeerConnectionFactory.Options options = new PeerConnectionFactory.Options();
>     options.networkIgnoreMask = 0;
>-    PeerConnectionFactory factory = new PeerConnectionFactory(options);
>-
>-    MediaConstraints pcConstraints = new MediaConstraints();
>-    pcConstraints.mandatory.add(new MediaConstraints.KeyValuePair("DtlsSrtpKeyAgreement", "true"));
>+    PeerConnectionFactory factory =
>+        PeerConnectionFactory.builder().setOptions(options).createPeerConnectionFactory();
> 
>     List<PeerConnection.IceServer> iceServers = new ArrayList<>();
>     iceServers.add(
>@@ -928,14 +931,16 @@ public class PeerConnectionTest {
>                        .setUsername("fakeUsername")
>                        .setPassword("fakePassword")
>                        .createIceServer());
>+
>+    PeerConnection.RTCConfiguration rtcConfig = new PeerConnection.RTCConfiguration(iceServers);
>+    rtcConfig.enableDtlsSrtp = true;
>+
>     ObserverExpectations offeringExpectations = new ObserverExpectations("PCTest:offerer");
>-    PeerConnection offeringPC =
>-        factory.createPeerConnection(iceServers, pcConstraints, offeringExpectations);
>+    PeerConnection offeringPC = factory.createPeerConnection(rtcConfig, offeringExpectations);
>     assertNotNull(offeringPC);
> 
>     ObserverExpectations answeringExpectations = new ObserverExpectations("PCTest:answerer");
>-    PeerConnection answeringPC =
>-        factory.createPeerConnection(iceServers, pcConstraints, answeringExpectations);
>+    PeerConnection answeringPC = factory.createPeerConnection(rtcConfig, answeringExpectations);
>     assertNotNull(answeringPC);
> 
>     offeringExpectations.expectRenegotiationNeeded();
>@@ -1052,6 +1057,8 @@ public class PeerConnectionTest {
>     answeringExpectations.expectStateChange(DataChannel.State.CLOSED);
>     answeringExpectations.dataChannel.close();
>     offeringExpectations.dataChannel.close();
>+    assertTrue(offeringExpectations.waitForAllExpectationsToBeSatisfied(TIMEOUT_SECONDS));
>+    assertTrue(answeringExpectations.waitForAllExpectationsToBeSatisfied(TIMEOUT_SECONDS));
> 
>     // Free the Java-land objects and collect them.
>     shutdownPC(offeringPC, offeringExpectations);
>@@ -1062,8 +1069,6 @@ public class PeerConnectionTest {
>     System.gc();
>   }
> 
>-  // Flaky on Android. See webrtc:7761
>-  @DisabledTest
>   @Test
>   @MediumTest
>   public void testTrackRemovalAndAddition() throws Exception {
>@@ -1071,23 +1076,22 @@ public class PeerConnectionTest {
>     // have those.
>     PeerConnectionFactory.Options options = new PeerConnectionFactory.Options();
>     options.networkIgnoreMask = 0;
>-    PeerConnectionFactory factory = new PeerConnectionFactory(options);
>-
>-    MediaConstraints pcConstraints = new MediaConstraints();
>-    pcConstraints.mandatory.add(new MediaConstraints.KeyValuePair("DtlsSrtpKeyAgreement", "true"));
>+    PeerConnectionFactory factory =
>+        PeerConnectionFactory.builder().setOptions(options).createPeerConnectionFactory();
> 
>     List<PeerConnection.IceServer> iceServers = new ArrayList<>();
>     iceServers.add(
>         PeerConnection.IceServer.builder("stun:stun.l.google.com:19302").createIceServer());
> 
>+    PeerConnection.RTCConfiguration rtcConfig = new PeerConnection.RTCConfiguration(iceServers);
>+    rtcConfig.enableDtlsSrtp = true;
>+
>     ObserverExpectations offeringExpectations = new ObserverExpectations("PCTest:offerer");
>-    PeerConnection offeringPC =
>-        factory.createPeerConnection(iceServers, pcConstraints, offeringExpectations);
>+    PeerConnection offeringPC = factory.createPeerConnection(rtcConfig, offeringExpectations);
>     assertNotNull(offeringPC);
> 
>     ObserverExpectations answeringExpectations = new ObserverExpectations("PCTest:answerer");
>-    PeerConnection answeringPC =
>-        factory.createPeerConnection(iceServers, pcConstraints, answeringExpectations);
>+    PeerConnection answeringPC = factory.createPeerConnection(rtcConfig, answeringExpectations);
>     assertNotNull(answeringPC);
> 
>     // We want to use the same camera for offerer & answerer, so create it here
>@@ -1095,7 +1099,11 @@ public class PeerConnectionTest {
>     final CameraEnumerator enumerator = new Camera1Enumerator(false /* captureToTexture */);
>     final VideoCapturer videoCapturer =
>         enumerator.createCapturer(enumerator.getDeviceNames()[0], null /* eventsHandler */);
>-    final VideoSource videoSource = factory.createVideoSource(videoCapturer);
>+    final SurfaceTextureHelper surfaceTextureHelper =
>+        SurfaceTextureHelper.create("SurfaceTextureHelper", /* sharedContext= */ null);
>+    final VideoSource videoSource = factory.createVideoSource(/* isScreencast= */ false);
>+    videoCapturer.initialize(surfaceTextureHelper, InstrumentationRegistry.getTargetContext(),
>+        videoSource.getCapturerObserver());
>     videoCapturer.startCapture(640, 480, 30);
> 
>     // Add offerer media stream.
>@@ -1203,19 +1211,17 @@ public class PeerConnectionTest {
>     VideoTrack offererVideoTrack = oLMS.get().videoTracks.get(0);
>     // Note that when we call removeTrack, we regain responsibility for
>     // disposing of the track.
>+    offeringExpectations.expectRenegotiationNeeded();
>     oLMS.get().removeTrack(offererVideoTrack);
>     negotiate(offeringPC, offeringExpectations, answeringPC, answeringExpectations);
> 
>     // Make sure the track was really removed.
>-    // TODO(deadbeef): Currently the expectation is that the video track's
>-    // state will be set to "ended". However, in the future, it's likely that
>-    // the video track will be completely removed from the remote stream
>-    // (as it is on the C++ level).
>     MediaStream aRMS = answeringExpectations.gotRemoteStreams.iterator().next();
>-    assertEquals(aRMS.videoTracks.get(0).state(), MediaStreamTrack.State.ENDED);
>+    assertTrue(aRMS.videoTracks.isEmpty());
> 
>     // Add the video track to test if the answeringPC will create a new track
>     // for the updated remote description.
>+    offeringExpectations.expectRenegotiationNeeded();
>     oLMS.get().addTrack(offererVideoTrack);
>     // The answeringPC sets the updated remote description with a track added.
>     // So the onAddTrack callback is expected to be called once.
>@@ -1226,8 +1232,10 @@ public class PeerConnectionTest {
>     // Finally, remove both the audio and video tracks, which should completely
>     // remove the remote stream. This used to trigger an assert.
>     // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=5128
>+    offeringExpectations.expectRenegotiationNeeded();
>     oLMS.get().removeTrack(offererVideoTrack);
>     AudioTrack offererAudioTrack = oLMS.get().audioTracks.get(0);
>+    offeringExpectations.expectRenegotiationNeeded();
>     oLMS.get().removeTrack(offererAudioTrack);
> 
>     answeringExpectations.expectRemoveStream("offeredMediaStream");
>@@ -1246,6 +1254,7 @@ public class PeerConnectionTest {
>     videoCapturer.stopCapture();
>     videoCapturer.dispose();
>     videoSource.dispose();
>+    surfaceTextureHelper.dispose();
>     factory.dispose();
>     System.gc();
>   }
>@@ -1266,11 +1275,9 @@ public class PeerConnectionTest {
>   @Test
>   @MediumTest
>   public void testRemoteStreamUpdatedWhenTracksAddedOrRemoved() throws Exception {
>-    PeerConnectionFactory.Options options = new PeerConnectionFactory.Options();
>-    PeerConnectionFactory factory = new PeerConnectionFactory(options);
>+    PeerConnectionFactory factory = PeerConnectionFactory.builder().createPeerConnectionFactory();
> 
>-    // This test is fine with default PC constraints and no ICE servers.
>-    MediaConstraints pcConstraints = new MediaConstraints();
>+    // This test is fine with no ICE servers.
>     List<PeerConnection.IceServer> iceServers = new ArrayList<>();
> 
>     // Use OfferToReceiveAudio/Video to ensure every offer has an audio and
>@@ -1285,13 +1292,11 @@ public class PeerConnectionTest {
> 
>     // This PeerConnection will only be used to generate offers.
>     ObserverExpectations offeringExpectations = new ObserverExpectations("offerer");
>-    PeerConnection offeringPC =
>-        factory.createPeerConnection(iceServers, pcConstraints, offeringExpectations);
>+    PeerConnection offeringPC = factory.createPeerConnection(iceServers, offeringExpectations);
>     assertNotNull(offeringPC);
> 
>     ObserverExpectations expectations = new ObserverExpectations("PC under test");
>-    PeerConnection pcUnderTest =
>-        factory.createPeerConnection(iceServers, pcConstraints, expectations);
>+    PeerConnection pcUnderTest = factory.createPeerConnection(iceServers, expectations);
>     assertNotNull(pcUnderTest);
> 
>     // Add offerer media stream with just an audio track.
>@@ -1323,7 +1328,11 @@ public class PeerConnectionTest {
>     final CameraEnumerator enumerator = new Camera1Enumerator(false /* captureToTexture */);
>     final VideoCapturer videoCapturer =
>         enumerator.createCapturer(enumerator.getDeviceNames()[0], null /* eventsHandler */);
>-    final VideoSource videoSource = factory.createVideoSource(videoCapturer);
>+    final SurfaceTextureHelper surfaceTextureHelper =
>+        SurfaceTextureHelper.create("SurfaceTextureHelper", /* sharedContext= */ null);
>+    final VideoSource videoSource = factory.createVideoSource(/* isScreencast= */ false);
>+    videoCapturer.initialize(surfaceTextureHelper, InstrumentationRegistry.getTargetContext(),
>+        videoSource.getCapturerObserver());
>     VideoTrack videoTrack = factory.createVideoTrack("video", videoSource);
>     offeringExpectations.expectRenegotiationNeeded();
>     localStream.addTrack(videoTrack);
>@@ -1374,9 +1383,105 @@ public class PeerConnectionTest {
>     pcUnderTest.dispose();
>     videoCapturer.dispose();
>     videoSource.dispose();
>+    surfaceTextureHelper.dispose();
>     factory.dispose();
>   }
> 
>+  @Test
>+  @MediumTest
>+  public void testAddingNullVideoSink() {
>+    final PeerConnectionFactory factory =
>+        PeerConnectionFactory.builder().createPeerConnectionFactory();
>+    final VideoSource videoSource = factory.createVideoSource(/* isScreencast= */ false);
>+    final VideoTrack videoTrack = factory.createVideoTrack("video", videoSource);
>+    try {
>+      videoTrack.addSink(/* sink= */ null);
>+      fail("Should have thrown an IllegalArgumentException.");
>+    } catch (IllegalArgumentException e) {
>+      // Expected path.
>+    }
>+  }
>+
>+  @Test
>+  @MediumTest
>+  public void testRemovingNullVideoSink() {
>+    final PeerConnectionFactory factory =
>+        PeerConnectionFactory.builder().createPeerConnectionFactory();
>+    final VideoSource videoSource = factory.createVideoSource(/* isScreencast= */ false);
>+    final VideoTrack videoTrack = factory.createVideoTrack("video", videoSource);
>+    videoTrack.removeSink(/* sink= */ null);
>+  }
>+
>+  @Test
>+  @MediumTest
>+  public void testRemovingNonExistantVideoSink() {
>+    final PeerConnectionFactory factory =
>+        PeerConnectionFactory.builder().createPeerConnectionFactory();
>+    final VideoSource videoSource = factory.createVideoSource(/* isScreencast= */ false);
>+    final VideoTrack videoTrack = factory.createVideoTrack("video", videoSource);
>+    final VideoSink videoSink = new VideoSink() {
>+      @Override
>+      public void onFrame(VideoFrame frame) {}
>+    };
>+    videoTrack.removeSink(videoSink);
>+  }
>+
>+  @Test
>+  @MediumTest
>+  public void testAddingSameVideoSinkMultipleTimes() {
>+    final PeerConnectionFactory factory =
>+        PeerConnectionFactory.builder().createPeerConnectionFactory();
>+    final VideoSource videoSource = factory.createVideoSource(/* isScreencast= */ false);
>+    final VideoTrack videoTrack = factory.createVideoTrack("video", videoSource);
>+
>+    class FrameCounter implements VideoSink {
>+      private int count = 0;
>+
>+      public int getCount() {
>+        return count;
>+      }
>+
>+      @Override
>+      public void onFrame(VideoFrame frame) {
>+        count += 1;
>+      }
>+    }
>+    final FrameCounter frameCounter = new FrameCounter();
>+
>+    final VideoFrame videoFrame = new VideoFrame(
>+        JavaI420Buffer.allocate(/* width= */ 32, /* height= */ 32), /* rotation= */ 0,
>+        /* timestampNs= */ 0);
>+
>+    videoTrack.addSink(frameCounter);
>+    videoTrack.addSink(frameCounter);
>+    videoSource.getCapturerObserver().onFrameCaptured(videoFrame);
>+
>+    // Even though we called addSink() multiple times, we should only get one frame out.
>+    assertEquals(1, frameCounter.count);
>+  }
>+
>+  @Test
>+  @MediumTest
>+  public void testAddingAndRemovingVideoSink() {
>+    final PeerConnectionFactory factory =
>+        PeerConnectionFactory.builder().createPeerConnectionFactory();
>+    final VideoSource videoSource = factory.createVideoSource(/* isScreencast= */ false);
>+    final VideoTrack videoTrack = factory.createVideoTrack("video", videoSource);
>+    final VideoFrame videoFrame = new VideoFrame(
>+        JavaI420Buffer.allocate(/* width= */ 32, /* height= */ 32), /* rotation= */ 0,
>+        /* timestampNs= */ 0);
>+
>+    final VideoSink failSink = new VideoSink() {
>+      @Override
>+      public void onFrame(VideoFrame frame) {
>+        fail("onFrame() should not be called on removed sink");
>+      }
>+    };
>+    videoTrack.addSink(failSink);
>+    videoTrack.removeSink(failSink);
>+    videoSource.getCapturerObserver().onFrameCaptured(videoFrame);
>+  }
>+
>   private static void negotiate(PeerConnection offeringPC,
>       ObserverExpectations offeringExpectations, PeerConnection answeringPC,
>       ObserverExpectations answeringExpectations) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/RendererCommonTest.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/RendererCommonTest.java
>index 34d795b9b80..d29fc0b0306 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/RendererCommonTest.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/RendererCommonTest.java
>@@ -15,7 +15,6 @@ import static org.junit.Assert.assertEquals;
> import static org.webrtc.RendererCommon.ScalingType.*;
> import static org.webrtc.RendererCommon.getDisplaySize;
> import static org.webrtc.RendererCommon.getLayoutMatrix;
>-import static org.webrtc.RendererCommon.rotateTextureMatrix;
> 
> import android.graphics.Point;
> import android.support.test.filters.SmallTest;
>@@ -149,52 +148,4 @@ public class RendererCommonTest {
>         0.25, 0, 0, 1}, round(layoutMatrix), 0.0);
>     // clang-format on
>   }
>-
>-  @Test
>-  @SmallTest
>-  public void testRotateTextureMatrixDefault() {
>-    // Test that rotation with 0 degrees returns an identical matrix.
>-    // clang-format off
>-    final float[] matrix = new float[] {
>-        1, 2, 3, 4,
>-        5, 6, 7, 8,
>-        9, 0, 1, 2,
>-        3, 4, 5, 6
>-    };
>-    // clang-format on
>-    final float rotatedMatrix[] = rotateTextureMatrix(matrix, 0);
>-    assertArrayEquals(round(matrix), round(rotatedMatrix), 0.0);
>-  }
>-
>-  @Test
>-  @SmallTest
>-  public void testRotateTextureMatrix90Deg() {
>-    final float samplingMatrix[] = rotateTextureMatrix(RendererCommon.identityMatrix(), 90);
>-    // Assert:
>-    // u' = 1 - v.
>-    // v' = u.
>-    // clang-format off
>-    assertArrayEquals(new double[] {
>-         0, 1, 0, 0,
>-        -1, 0, 0, 0,
>-         0, 0, 1, 0,
>-         1, 0, 0, 1}, round(samplingMatrix), 0.0);
>-    // clang-format on
>-  }
>-
>-  @Test
>-  @SmallTest
>-  public void testRotateTextureMatrix180Deg() {
>-    final float samplingMatrix[] = rotateTextureMatrix(RendererCommon.identityMatrix(), 180);
>-    // Assert:
>-    // u' = 1 - u.
>-    // v' = 1 - v.
>-    // clang-format off
>-    assertArrayEquals(new double[] {
>-        -1,  0, 0, 0,
>-         0, -1, 0, 0,
>-         0,  0, 1, 0,
>-         1,  1, 0, 1}, round(samplingMatrix), 0.0);
>-    // clang-format on
>-  }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/SurfaceTextureHelperTest.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/SurfaceTextureHelperTest.java
>index 563336b984c..3cd08cc8b6b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/SurfaceTextureHelperTest.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/SurfaceTextureHelperTest.java
>@@ -12,6 +12,7 @@ package org.webrtc;
> 
> import static org.junit.Assert.assertEquals;
> import static org.junit.Assert.assertFalse;
>+import static org.junit.Assert.assertTrue;
> import static org.junit.Assert.fail;
> 
> import android.graphics.SurfaceTexture;
>@@ -21,6 +22,7 @@ import android.support.test.filters.MediumTest;
> import android.support.test.filters.SmallTest;
> import java.nio.ByteBuffer;
> import java.util.concurrent.CountDownLatch;
>+import javax.annotation.Nullable;
> import org.chromium.base.test.BaseJUnit4ClassRunner;
> import org.junit.Before;
> import org.junit.Test;
>@@ -31,13 +33,11 @@ public class SurfaceTextureHelperTest {
>   /**
>    * Mock texture listener with blocking wait functionality.
>    */
>-  public static final class MockTextureListener
>-      implements SurfaceTextureHelper.OnTextureFrameAvailableListener {
>-    public int oesTextureId;
>-    public float[] transformMatrix;
>-    private boolean hasNewFrame = false;
>+  public static final class MockTextureListener implements VideoSink {
>+    private final Object lock = new Object();
>+    private @Nullable VideoFrame.TextureBuffer textureBuffer;
>     // Thread where frames are expected to be received on.
>-    private final Thread expectedThread;
>+    private final @Nullable Thread expectedThread;
> 
>     MockTextureListener() {
>       this.expectedThread = null;
>@@ -48,48 +48,43 @@ public class SurfaceTextureHelperTest {
>     }
> 
>     @Override
>-    // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
>-    @SuppressWarnings("NoSynchronizedMethodCheck")
>-    public synchronized void onTextureFrameAvailable(
>-        int oesTextureId, float[] transformMatrix, long timestampNs) {
>+    public void onFrame(VideoFrame frame) {
>       if (expectedThread != null && Thread.currentThread() != expectedThread) {
>         throw new IllegalStateException("onTextureFrameAvailable called on wrong thread.");
>       }
>-      this.oesTextureId = oesTextureId;
>-      this.transformMatrix = transformMatrix;
>-      hasNewFrame = true;
>-      notifyAll();
>+      synchronized (lock) {
>+        this.textureBuffer = (VideoFrame.TextureBuffer) frame.getBuffer();
>+        textureBuffer.retain();
>+        lock.notifyAll();
>+      }
>     }
> 
>-    /**
>-     * Wait indefinitely for a new frame.
>-     */
>-    // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
>-    @SuppressWarnings("NoSynchronizedMethodCheck")
>-    public synchronized void waitForNewFrame() throws InterruptedException {
>-      while (!hasNewFrame) {
>-        wait();
>+    /** Wait indefinitely for a new textureBuffer. */
>+    public VideoFrame.TextureBuffer waitForTextureBuffer() throws InterruptedException {
>+      synchronized (lock) {
>+        while (true) {
>+          final VideoFrame.TextureBuffer textureBufferToReturn = textureBuffer;
>+          if (textureBufferToReturn != null) {
>+            textureBuffer = null;
>+            return textureBufferToReturn;
>+          }
>+          lock.wait();
>+        }
>       }
>-      hasNewFrame = false;
>     }
> 
>-    /**
>-     * Wait for a new frame, or until the specified timeout elapses. Returns true if a new frame was
>-     * received before the timeout.
>-     */
>-    // TODO(bugs.webrtc.org/8491): Remove NoSynchronizedMethodCheck suppression.
>-    @SuppressWarnings("NoSynchronizedMethodCheck")
>-    public synchronized boolean waitForNewFrame(final long timeoutMs) throws InterruptedException {
>+    /** Make sure we get no frame in the specified time period. */
>+    public void assertNoFrameIsDelivered(final long waitPeriodMs) throws InterruptedException {
>       final long startTimeMs = SystemClock.elapsedRealtime();
>-      long timeRemainingMs = timeoutMs;
>-      while (!hasNewFrame && timeRemainingMs > 0) {
>-        wait(timeRemainingMs);
>-        final long elapsedTimeMs = SystemClock.elapsedRealtime() - startTimeMs;
>-        timeRemainingMs = timeoutMs - elapsedTimeMs;
>+      long timeRemainingMs = waitPeriodMs;
>+      synchronized (lock) {
>+        while (textureBuffer == null && timeRemainingMs > 0) {
>+          lock.wait(timeRemainingMs);
>+          final long elapsedTimeMs = SystemClock.elapsedRealtime() - startTimeMs;
>+          timeRemainingMs = waitPeriodMs - elapsedTimeMs;
>+        }
>+        assertTrue(textureBuffer == null);
>       }
>-      final boolean didReceiveFrame = hasNewFrame;
>-      hasNewFrame = false;
>-      return didReceiveFrame;
>     }
>   }
> 
>@@ -105,7 +100,7 @@ public class SurfaceTextureHelperTest {
>   @Before
>   public void setUp() {
>     // Load the JNI library for textureToYuv.
>-    NativeLibrary.initialize(new NativeLibrary.DefaultLoader());
>+    NativeLibrary.initialize(new NativeLibrary.DefaultLoader(), TestConstants.NATIVE_LIBRARY);
>   }
> 
>   /**
>@@ -128,7 +123,7 @@ public class SurfaceTextureHelperTest {
>         "SurfaceTextureHelper test" /* threadName */, eglBase.getEglBaseContext());
>     final MockTextureListener listener = new MockTextureListener();
>     surfaceTextureHelper.startListening(listener);
>-    surfaceTextureHelper.getSurfaceTexture().setDefaultBufferSize(width, height);
>+    surfaceTextureHelper.setTextureSize(width, height);
> 
>     // Create resources for stubbing an OES texture producer. |eglOesBase| has the SurfaceTexture in
>     // |surfaceTextureHelper| as the target EGLSurface.
>@@ -150,12 +145,12 @@ public class SurfaceTextureHelperTest {
>       eglOesBase.swapBuffers();
> 
>       // Wait for an OES texture to arrive and draw it onto the pixel buffer.
>-      listener.waitForNewFrame();
>+      final VideoFrame.TextureBuffer textureBuffer = listener.waitForTextureBuffer();
>       eglBase.makeCurrent();
>-      drawer.drawOes(
>-          listener.oesTextureId, listener.transformMatrix, width, height, 0, 0, width, height);
>-
>-      surfaceTextureHelper.returnTextureFrame();
>+      drawer.drawOes(textureBuffer.getTextureId(),
>+          RendererCommon.convertMatrixFromAndroidGraphicsMatrix(textureBuffer.getTransformMatrix()),
>+          width, height, 0, 0, width, height);
>+      textureBuffer.release();
> 
>       // Download the pixels in the pixel buffer as RGBA. Not all platforms support RGB, e.g.
>       // Nexus 9.
>@@ -196,7 +191,7 @@ public class SurfaceTextureHelperTest {
>         "SurfaceTextureHelper test" /* threadName */, eglBase.getEglBaseContext());
>     final MockTextureListener listener = new MockTextureListener();
>     surfaceTextureHelper.startListening(listener);
>-    surfaceTextureHelper.getSurfaceTexture().setDefaultBufferSize(width, height);
>+    surfaceTextureHelper.setTextureSize(width, height);
> 
>     // Create resources for stubbing an OES texture producer. |eglOesBase| has the SurfaceTexture in
>     // |surfaceTextureHelper| as the target EGLSurface.
>@@ -217,15 +212,16 @@ public class SurfaceTextureHelperTest {
>     eglOesBase.release();
> 
>     // Wait for OES texture frame.
>-    listener.waitForNewFrame();
>+    final VideoFrame.TextureBuffer textureBuffer = listener.waitForTextureBuffer();
>     // Diconnect while holding the frame.
>     surfaceTextureHelper.dispose();
> 
>     // Draw the pending texture frame onto the pixel buffer.
>     eglBase.makeCurrent();
>     final GlRectDrawer drawer = new GlRectDrawer();
>-    drawer.drawOes(
>-        listener.oesTextureId, listener.transformMatrix, width, height, 0, 0, width, height);
>+    drawer.drawOes(textureBuffer.getTextureId(),
>+        RendererCommon.convertMatrixFromAndroidGraphicsMatrix(textureBuffer.getTransformMatrix()),
>+        width, height, 0, 0, width, height);
>     drawer.release();
> 
>     // Download the pixels in the pixel buffer as RGBA. Not all platforms support RGB, e.g. Nexus 9.
>@@ -242,7 +238,7 @@ public class SurfaceTextureHelperTest {
>       assertEquals(rgbaData.get() & 0xFF, 255);
>     }
>     // Late frame return after everything has been disposed and released.
>-    surfaceTextureHelper.returnTextureFrame();
>+    textureBuffer.release();
>   }
> 
>   /**
>@@ -259,16 +255,16 @@ public class SurfaceTextureHelperTest {
>     surfaceTextureHelper.startListening(listener);
>     // Create EglBase with the SurfaceTexture as target EGLSurface.
>     final EglBase eglBase = EglBase.create(null, EglBase.CONFIG_PLAIN);
>+    surfaceTextureHelper.setTextureSize(/* textureWidth= */ 32, /* textureHeight= */ 32);
>     eglBase.createSurface(surfaceTextureHelper.getSurfaceTexture());
>     eglBase.makeCurrent();
>     // Assert no frame has been received yet.
>-    assertFalse(listener.waitForNewFrame(1));
>+    listener.assertNoFrameIsDelivered(/* waitPeriodMs= */ 1);
>     // Draw and wait for one frame.
>     GLES20.glClear(GLES20.GL_COLOR_BUFFER_BIT);
>     // swapBuffers() will ultimately trigger onTextureFrameAvailable().
>     eglBase.swapBuffers();
>-    listener.waitForNewFrame();
>-    surfaceTextureHelper.returnTextureFrame();
>+    listener.waitForTextureBuffer().release();
> 
>     // Dispose - we should not receive any textures after this.
>     surfaceTextureHelper.dispose();
>@@ -278,7 +274,7 @@ public class SurfaceTextureHelperTest {
>     eglBase.swapBuffers();
>     // swapBuffers() should not trigger onTextureFrameAvailable() because disposed has been called.
>     // Assert that no OES texture was delivered.
>-    assertFalse(listener.waitForNewFrame(500));
>+    listener.assertNoFrameIsDelivered(/* waitPeriodMs= */ 500);
> 
>     eglBase.release();
>   }
>@@ -305,6 +301,7 @@ public class SurfaceTextureHelperTest {
>     // Create SurfaceTextureHelper and listener.
>     final SurfaceTextureHelper surfaceTextureHelper =
>         SurfaceTextureHelper.create("SurfaceTextureHelper test" /* threadName */, null);
>+    surfaceTextureHelper.setTextureSize(/* textureWidth= */ 32, /* textureHeight= */ 32);
>     final MockTextureListener listener = new MockTextureListener();
>     surfaceTextureHelper.startListening(listener);
>     // Create EglBase with the SurfaceTexture as target EGLSurface.
>@@ -312,13 +309,12 @@ public class SurfaceTextureHelperTest {
>     eglBase.createSurface(surfaceTextureHelper.getSurfaceTexture());
>     eglBase.makeCurrent();
>     // Assert no frame has been received yet.
>-    assertFalse(listener.waitForNewFrame(1));
>+    listener.assertNoFrameIsDelivered(/* waitPeriodMs= */ 1);
>     // Draw and wait for one frame.
>     GLES20.glClear(GLES20.GL_COLOR_BUFFER_BIT);
>     // swapBuffers() will ultimately trigger onTextureFrameAvailable().
>     eglBase.swapBuffers();
>-    listener.waitForNewFrame();
>-    surfaceTextureHelper.returnTextureFrame();
>+    listener.waitForTextureBuffer().release();
> 
>     // Stop listening - we should not receive any textures after this.
>     surfaceTextureHelper.stopListening();
>@@ -328,7 +324,7 @@ public class SurfaceTextureHelperTest {
>     eglBase.swapBuffers();
>     // swapBuffers() should not trigger onTextureFrameAvailable() because disposed has been called.
>     // Assert that no OES texture was delivered.
>-    assertFalse(listener.waitForNewFrame(500));
>+    listener.assertNoFrameIsDelivered(/* waitPeriodMs= */ 500);
> 
>     surfaceTextureHelper.dispose();
>     eglBase.release();
>@@ -401,6 +397,7 @@ public class SurfaceTextureHelperTest {
>     // Create SurfaceTextureHelper and listener.
>     final SurfaceTextureHelper surfaceTextureHelper =
>         SurfaceTextureHelper.create("SurfaceTextureHelper test" /* threadName */, null);
>+    surfaceTextureHelper.setTextureSize(/* textureWidth= */ 32, /* textureHeight= */ 32);
>     final MockTextureListener listener1 = new MockTextureListener();
>     surfaceTextureHelper.startListening(listener1);
>     // Create EglBase with the SurfaceTexture as target EGLSurface.
>@@ -408,13 +405,12 @@ public class SurfaceTextureHelperTest {
>     eglBase.createSurface(surfaceTextureHelper.getSurfaceTexture());
>     eglBase.makeCurrent();
>     // Assert no frame has been received yet.
>-    assertFalse(listener1.waitForNewFrame(1));
>+    listener1.assertNoFrameIsDelivered(/* waitPeriodMs= */ 1);
>     // Draw and wait for one frame.
>     GLES20.glClear(GLES20.GL_COLOR_BUFFER_BIT);
>     // swapBuffers() will ultimately trigger onTextureFrameAvailable().
>     eglBase.swapBuffers();
>-    listener1.waitForNewFrame();
>-    surfaceTextureHelper.returnTextureFrame();
>+    listener1.waitForTextureBuffer().release();
> 
>     // Stop listening - |listener1| should not receive any textures after this.
>     surfaceTextureHelper.stopListening();
>@@ -423,17 +419,15 @@ public class SurfaceTextureHelperTest {
>     final MockTextureListener listener2 = new MockTextureListener();
>     surfaceTextureHelper.startListening(listener2);
>     // Assert no frame has been received yet.
>-    assertFalse(listener2.waitForNewFrame(1));
>+    listener2.assertNoFrameIsDelivered(/* waitPeriodMs= */ 1);
> 
>     // Draw one frame.
>     GLES20.glClear(GLES20.GL_COLOR_BUFFER_BIT);
>     eglBase.swapBuffers();
> 
>     // Check that |listener2| received the frame, and not |listener1|.
>-    listener2.waitForNewFrame();
>-    assertFalse(listener1.waitForNewFrame(1));
>-
>-    surfaceTextureHelper.returnTextureFrame();
>+    listener2.waitForTextureBuffer().release();
>+    listener1.assertNoFrameIsDelivered(/* waitPeriodMs= */ 1);
> 
>     surfaceTextureHelper.dispose();
>     eglBase.release();
>@@ -452,7 +446,7 @@ public class SurfaceTextureHelperTest {
>         "SurfaceTextureHelper test" /* threadName */, eglBase.getEglBaseContext());
>     final MockTextureListener listener = new MockTextureListener();
>     surfaceTextureHelper.startListening(listener);
>-    surfaceTextureHelper.getSurfaceTexture().setDefaultBufferSize(width, height);
>+    surfaceTextureHelper.setTextureSize(width, height);
> 
>     // Create resources for stubbing an OES texture producer. |eglBase| has the SurfaceTexture in
>     // |surfaceTextureHelper| as the target EGLSurface.
>@@ -479,7 +473,9 @@ public class SurfaceTextureHelperTest {
>       eglBase.swapBuffers();
> 
>       // Wait for an OES texture to arrive.
>-      listener.waitForNewFrame();
>+      final VideoFrame.TextureBuffer textureBuffer = listener.waitForTextureBuffer();
>+      final VideoFrame.I420Buffer i420 = textureBuffer.toI420();
>+      textureBuffer.release();
> 
>       // Memory layout: Lines are 16 bytes. First 16 lines are
>       // the Y data. These are followed by 8 lines with 8 bytes of U
>@@ -495,11 +491,6 @@ public class SurfaceTextureHelperTest {
>       //    ...
>       //    368 UUUUUUUU VVVVVVVV
>       //    384 buffer end
>-      final VideoFrame.I420Buffer i420 =
>-          surfaceTextureHelper.textureToYuv(surfaceTextureHelper.createTextureBuffer(width, height,
>-              RendererCommon.convertMatrixToAndroidGraphicsMatrix(listener.transformMatrix)));
>-
>-      surfaceTextureHelper.returnTextureFrame();
> 
>       // Allow off-by-one differences due to different rounding.
>       final ByteBuffer dataY = i420.getDataY();
>@@ -523,6 +514,7 @@ public class SurfaceTextureHelperTest {
>           assertClose(1, ref_v[i], dataV.get(y * strideV + x) & 0xFF);
>         }
>       }
>+      i420.release();
>     }
> 
>     surfaceTextureHelper.dispose();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/SurfaceViewRendererOnMeasureTest.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/SurfaceViewRendererOnMeasureTest.java
>index ca89d147f4a..57ddd7ef3a2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/SurfaceViewRendererOnMeasureTest.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/SurfaceViewRendererOnMeasureTest.java
>@@ -48,14 +48,17 @@ public class SurfaceViewRendererOnMeasureTest {
>   /**
>    * Returns a dummy YUV frame.
>    */
>-  static VideoRenderer.I420Frame createFrame(int width, int height, int rotationDegree) {
>+  static VideoFrame createFrame(int width, int height, int rotationDegree) {
>     final int[] yuvStrides = new int[] {width, (width + 1) / 2, (width + 1) / 2};
>     final int[] yuvHeights = new int[] {height, (height + 1) / 2, (height + 1) / 2};
>     final ByteBuffer[] yuvPlanes = new ByteBuffer[3];
>     for (int i = 0; i < 3; ++i) {
>       yuvPlanes[i] = ByteBuffer.allocateDirect(yuvStrides[i] * yuvHeights[i]);
>     }
>-    return new VideoRenderer.I420Frame(width, height, rotationDegree, yuvStrides, yuvPlanes, 0);
>+    final VideoFrame.I420Buffer buffer =
>+        JavaI420Buffer.wrap(width, height, yuvPlanes[0], yuvStrides[0], yuvPlanes[1], yuvStrides[1],
>+            yuvPlanes[2], yuvStrides[2], null /* releaseCallback */);
>+    return new VideoFrame(buffer, rotationDegree, 0 /* timestamp */);
>   }
> 
>   /**
>@@ -167,13 +170,13 @@ public class SurfaceViewRendererOnMeasureTest {
>       final int rotatedHeight = 720;
>       final int unrotatedWidth = (rotationDegree % 180 == 0 ? rotatedWidth : rotatedHeight);
>       final int unrotatedHeight = (rotationDegree % 180 == 0 ? rotatedHeight : rotatedWidth);
>-      final VideoRenderer.I420Frame frame =
>-          createFrame(unrotatedWidth, unrotatedHeight, rotationDegree);
>-      assertEquals(rotatedWidth, frame.rotatedWidth());
>-      assertEquals(rotatedHeight, frame.rotatedHeight());
>+      final VideoFrame frame = createFrame(unrotatedWidth, unrotatedHeight, rotationDegree);
>+      assertEquals(rotatedWidth, frame.getRotatedWidth());
>+      assertEquals(rotatedHeight, frame.getRotatedHeight());
>       final String frameDimensions =
>           unrotatedWidth + "x" + unrotatedHeight + " with rotation " + rotationDegree;
>-      surfaceViewRenderer.renderFrame(frame);
>+      surfaceViewRenderer.onFrame(frame);
>+      frame.release();
>       rendererEvents.waitForFrameSize(unrotatedWidth, unrotatedHeight, rotationDegree);
> 
>       // Test forcing to zero size.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/voicechannel.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/TestConstants.java
>similarity index 63%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/pc/voicechannel.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/TestConstants.java
>index 852ef173542..6c7904c9f30 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/voicechannel.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/TestConstants.java
>@@ -1,5 +1,5 @@
> /*
>- *  Copyright 2004 The WebRTC project authors. All Rights Reserved.
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>  *
>  *  Use of this source code is governed by a BSD-style license
>  *  that can be found in the LICENSE file in the root of the source
>@@ -8,9 +8,8 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#ifndef PC_VOICECHANNEL_H_
>-#define PC_VOICECHANNEL_H_
>+package org.webrtc;
> 
>-#include "pc/channel.h"
>-
>-#endif  // PC_VOICECHANNEL_H_
>+class TestConstants {
>+  public static final String NATIVE_LIBRARY = "jingle_peerconnection_instrumentationtests_so";
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/VideoFileRendererTest.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/VideoFileRendererTest.java
>index cfcdd655f7a..02bff40ff93 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/VideoFileRendererTest.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/VideoFileRendererTest.java
>@@ -31,7 +31,7 @@ import org.junit.runner.RunWith;
> public class VideoFileRendererTest {
>   @Before
>   public void setUp() {
>-    NativeLibrary.initialize(new NativeLibrary.DefaultLoader());
>+    NativeLibrary.initialize(new NativeLibrary.DefaultLoader(), TestConstants.NATIVE_LIBRARY);
>   }
> 
>   @Test
>@@ -51,22 +51,25 @@ public class VideoFileRendererTest {
>     for (String frameStr : frames) {
>       int[] planeSizes = {
>           frameWidth * frameWidth, frameWidth * frameHeight / 4, frameWidth * frameHeight / 4};
>+      int[] yuvStrides = {frameWidth, frameWidth / 2, frameWidth / 2};
> 
>-      byte[] frameBytes = frameStr.getBytes(Charset.forName("US-ASCII"));
>       ByteBuffer[] yuvPlanes = new ByteBuffer[3];
>+      byte[] frameBytes = frameStr.getBytes(Charset.forName("US-ASCII"));
>       int pos = 0;
>       for (int i = 0; i < 3; i++) {
>         yuvPlanes[i] = ByteBuffer.allocateDirect(planeSizes[i]);
>         yuvPlanes[i].put(frameBytes, pos, planeSizes[i]);
>+        yuvPlanes[i].rewind();
>         pos += planeSizes[i];
>       }
> 
>-      int[] yuvStrides = {frameWidth, frameWidth / 2, frameWidth / 2};
>-
>-      VideoRenderer.I420Frame frame =
>-          new VideoRenderer.I420Frame(frameWidth, frameHeight, 0, yuvStrides, yuvPlanes, 0);
>+      VideoFrame.I420Buffer buffer =
>+          JavaI420Buffer.wrap(frameWidth, frameHeight, yuvPlanes[0], yuvStrides[0], yuvPlanes[1],
>+              yuvStrides[1], yuvPlanes[2], yuvStrides[2], null /* releaseCallback */);
> 
>-      videoFileRenderer.renderFrame(frame);
>+      VideoFrame frame = new VideoFrame(buffer, 0 /* rotation */, 0 /* timestampNs */);
>+      videoFileRenderer.onFrame(frame);
>+      frame.release();
>     }
>     videoFileRenderer.release();
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/VideoFrameBufferTest.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/VideoFrameBufferTest.java
>new file mode 100644
>index 00000000000..539ba3ddf5b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/VideoFrameBufferTest.java
>@@ -0,0 +1,509 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc;
>+
>+import static org.hamcrest.Matchers.greaterThanOrEqualTo;
>+import static org.hamcrest.Matchers.lessThanOrEqualTo;
>+import static org.junit.Assert.assertEquals;
>+import static org.junit.Assert.assertThat;
>+
>+import android.graphics.Matrix;
>+import android.graphics.SurfaceTexture;
>+import android.opengl.GLES20;
>+import android.os.Handler;
>+import android.os.HandlerThread;
>+import android.support.test.InstrumentationRegistry;
>+import android.support.test.filters.SmallTest;
>+import java.nio.ByteBuffer;
>+import java.util.ArrayList;
>+import java.util.Collections;
>+import java.util.List;
>+import org.chromium.base.test.params.BaseJUnit4RunnerDelegate;
>+import org.chromium.base.test.params.ParameterAnnotations.ClassParameter;
>+import org.chromium.base.test.params.ParameterAnnotations.UseRunnerDelegate;
>+import org.chromium.base.test.params.ParameterSet;
>+import org.chromium.base.test.params.ParameterizedRunner;
>+import org.junit.BeforeClass;
>+import org.junit.Test;
>+import org.junit.runner.RunWith;
>+
>+/**
>+ * Test VideoFrame buffers of different kind of formats: I420, RGB, OES, NV12, NV21, and verify
>+ * toI420() and cropAndScale() behavior. Creating RGB/OES textures involves VideoFrameDrawer and
>+ * GlRectDrawer and we are testing the full chain I420 -> OES/RGB texture -> I420, with and without
>+ * cropping in the middle. Reading textures back to I420 also exercises the YuvConverter code.
>+ */
>+@RunWith(ParameterizedRunner.class)
>+@UseRunnerDelegate(BaseJUnit4RunnerDelegate.class)
>+public class VideoFrameBufferTest {
>+  /**
>+   * These tests are parameterized on this enum which represents the different VideoFrame.Buffers.
>+   */
>+  private static enum BufferType { I420, RGB_TEXTURE, OES_TEXTURE, NV21, NV12 }
>+
>+  @ClassParameter private static List<ParameterSet> CLASS_PARAMS = new ArrayList<>();
>+
>+  static {
>+    for (BufferType bufferType : BufferType.values()) {
>+      // The parameterization framework can not send Enums, so we pass it as a string instead.
>+      CLASS_PARAMS.add(new ParameterSet().value(bufferType.name()).name(bufferType.name()));
>+    }
>+  }
>+
>+  @BeforeClass
>+  public static void setUp() {
>+    // Needed for JniCommon.nativeAllocateByteBuffer() to work, which is used from JavaI420Buffer.
>+    System.loadLibrary(TestConstants.NATIVE_LIBRARY);
>+  }
>+
>+  private final BufferType bufferType;
>+
>+  public VideoFrameBufferTest(String bufferTypeName) {
>+    // Parse the string back to an enum.
>+    bufferType = BufferType.valueOf(bufferTypeName);
>+  }
>+
>+  /**
>+   * Create a VideoFrame.Buffer of the given type with the same pixel content as the given I420
>+   * buffer.
>+   */
>+  private static VideoFrame.Buffer createBufferWithType(
>+      BufferType bufferType, VideoFrame.I420Buffer i420Buffer) {
>+    switch (bufferType) {
>+      case I420:
>+        return i420Buffer.toI420();
>+      case RGB_TEXTURE:
>+        return createRgbTextureBuffer(/* eglContext= */ null, i420Buffer);
>+      case OES_TEXTURE:
>+        return createOesTextureBuffer(/* eglContext= */ null, i420Buffer);
>+      case NV21:
>+        return createNV21Buffer(i420Buffer);
>+      case NV12:
>+        return createNV12Buffer(i420Buffer);
>+      default:
>+        throw new IllegalArgumentException("Unknown buffer type: " + bufferType);
>+    }
>+  }
>+
>+  private VideoFrame.Buffer createBufferToTest(VideoFrame.I420Buffer i420Buffer) {
>+    return createBufferWithType(this.bufferType, i420Buffer);
>+  }
>+
>+  /**
>+   * Creates a 16x16 I420 buffer that varies smoothly and spans all RGB values.
>+   */
>+  public static VideoFrame.I420Buffer createTestI420Buffer() {
>+    final int width = 16;
>+    final int height = 16;
>+    final int[] yData = new int[] {164, 170, 176, 182, 188, 194, 200, 206, 213, 218, 225, 231, 237,
>+        243, 249, 255, 153, 159, 165, 171, 177, 183, 189, 195, 201, 207, 213, 220, 226, 232, 238,
>+        244, 142, 148, 154, 160, 166, 172, 178, 184, 191, 196, 203, 209, 215, 221, 227, 233, 131,
>+        137, 143, 149, 155, 161, 167, 174, 180, 186, 192, 198, 204, 210, 216, 222, 120, 126, 132,
>+        138, 145, 151, 157, 163, 169, 175, 181, 187, 193, 199, 205, 211, 109, 115, 121, 128, 133,
>+        140, 146, 152, 158, 164, 170, 176, 182, 188, 194, 200, 99, 104, 111, 117, 123, 129, 135,
>+        141, 147, 153, 159, 165, 171, 177, 183, 189, 87, 94, 100, 106, 112, 118, 124, 130, 136, 142,
>+        148, 154, 160, 166, 172, 178, 77, 83, 89, 95, 101, 107, 113, 119, 125, 131, 137, 143, 149,
>+        155, 161, 167, 66, 72, 78, 84, 90, 96, 102, 108, 114, 120, 126, 132, 138, 144, 150, 156, 55,
>+        61, 67, 73, 79, 85, 91, 97, 103, 109, 115, 121, 127, 133, 139, 145, 44, 50, 56, 62, 68, 74,
>+        80, 86, 92, 98, 104, 110, 116, 122, 128, 134, 33, 39, 45, 51, 57, 63, 69, 75, 81, 87, 93,
>+        99, 105, 111, 117, 123, 22, 28, 34, 40, 46, 52, 58, 64, 70, 76, 82, 88, 94, 100, 106, 113,
>+        11, 17, 23, 29, 35, 41, 47, 53, 59, 65, 71, 77, 83, 89, 96, 102, 0, 6, 12, 18, 24, 30, 36,
>+        42, 48, 54, 60, 66, 72, 79, 85, 91};
>+    final int[] uData = new int[] {108, 111, 114, 117, 119, 122, 125, 128, 111, 114, 117, 119, 122,
>+        125, 128, 130, 114, 117, 119, 122, 125, 128, 130, 133, 117, 119, 122, 125, 128, 130, 133,
>+        136, 119, 122, 125, 128, 130, 133, 136, 139, 122, 125, 128, 130, 133, 136, 139, 141, 125,
>+        128, 130, 133, 136, 139, 141, 144, 128, 130, 133, 136, 139, 141, 144, 147};
>+    final int[] vData = new int[] {18, 34, 49, 65, 81, 96, 112, 127, 34, 49, 65, 81, 96, 112, 128,
>+        143, 49, 65, 81, 96, 112, 127, 143, 159, 65, 81, 96, 112, 127, 143, 159, 174, 81, 96, 112,
>+        128, 143, 159, 174, 190, 96, 112, 128, 143, 159, 174, 190, 206, 112, 127, 143, 159, 174,
>+        190, 205, 221, 127, 143, 159, 174, 190, 205, 221, 237};
>+    return JavaI420Buffer.wrap(width, height, toByteBuffer(yData),
>+        /* strideY= */ width, toByteBuffer(uData), /* strideU= */ width / 2, toByteBuffer(vData),
>+        /* strideV= */ width / 2,
>+        /* releaseCallback= */ null);
>+  }
>+
>+  /**
>+   * Create an RGB texture buffer available in |eglContext| with the same pixel content as the given
>+   * I420 buffer.
>+   */
>+  public static VideoFrame.TextureBuffer createRgbTextureBuffer(
>+      EglBase.Context eglContext, VideoFrame.I420Buffer i420Buffer) {
>+    final int width = i420Buffer.getWidth();
>+    final int height = i420Buffer.getHeight();
>+
>+    final HandlerThread renderThread = new HandlerThread("RGB texture thread");
>+    renderThread.start();
>+    final Handler renderThreadHandler = new Handler(renderThread.getLooper());
>+    return ThreadUtils.invokeAtFrontUninterruptibly(renderThreadHandler, () -> {
>+      // Create EGL base with a pixel buffer as display output.
>+      final EglBase eglBase = EglBase.create(eglContext, EglBase.CONFIG_PIXEL_BUFFER);
>+      eglBase.createDummyPbufferSurface();
>+      eglBase.makeCurrent();
>+
>+      final GlTextureFrameBuffer textureFrameBuffer = new GlTextureFrameBuffer(GLES20.GL_RGBA);
>+      textureFrameBuffer.setSize(width, height);
>+
>+      GLES20.glBindFramebuffer(GLES20.GL_FRAMEBUFFER, textureFrameBuffer.getFrameBufferId());
>+      drawI420Buffer(i420Buffer);
>+      GLES20.glBindFramebuffer(GLES20.GL_FRAMEBUFFER, 0);
>+
>+      final YuvConverter yuvConverter = new YuvConverter();
>+      return new TextureBufferImpl(width, height, VideoFrame.TextureBuffer.Type.RGB,
>+          textureFrameBuffer.getTextureId(),
>+          /* transformMatrix= */ new Matrix(), renderThreadHandler, yuvConverter,
>+          /* releaseCallback= */ () -> renderThreadHandler.post(() -> {
>+            textureFrameBuffer.release();
>+            yuvConverter.release();
>+            eglBase.release();
>+            renderThread.quit();
>+          }));
>+    });
>+  }
>+
>+  /**
>+   * Create an OES texture buffer available in |eglContext| with the same pixel content as the given
>+   * I420 buffer.
>+   */
>+  public static VideoFrame.TextureBuffer createOesTextureBuffer(
>+      EglBase.Context eglContext, VideoFrame.I420Buffer i420Buffer) {
>+    final int width = i420Buffer.getWidth();
>+    final int height = i420Buffer.getHeight();
>+
>+    // Create resources for generating OES textures.
>+    final SurfaceTextureHelper surfaceTextureHelper =
>+        SurfaceTextureHelper.create("SurfaceTextureHelper test", eglContext);
>+    surfaceTextureHelper.setTextureSize(width, height);
>+
>+    final HandlerThread renderThread = new HandlerThread("OES texture thread");
>+    renderThread.start();
>+    final Handler renderThreadHandler = new Handler(renderThread.getLooper());
>+    final VideoFrame.TextureBuffer oesBuffer =
>+        ThreadUtils.invokeAtFrontUninterruptibly(renderThreadHandler, () -> {
>+          // Create EGL base with the SurfaceTexture as display output.
>+          final EglBase eglBase = EglBase.create(eglContext, EglBase.CONFIG_PLAIN);
>+          eglBase.createSurface(surfaceTextureHelper.getSurfaceTexture());
>+          eglBase.makeCurrent();
>+          assertEquals(width, eglBase.surfaceWidth());
>+          assertEquals(height, eglBase.surfaceHeight());
>+
>+          final SurfaceTextureHelperTest.MockTextureListener listener =
>+              new SurfaceTextureHelperTest.MockTextureListener();
>+          surfaceTextureHelper.startListening(listener);
>+
>+          // Draw the frame and block until an OES texture is delivered.
>+          drawI420Buffer(i420Buffer);
>+          eglBase.swapBuffers();
>+          final VideoFrame.TextureBuffer textureBuffer = listener.waitForTextureBuffer();
>+          surfaceTextureHelper.stopListening();
>+          surfaceTextureHelper.dispose();
>+
>+          return textureBuffer;
>+        });
>+    renderThread.quit();
>+
>+    return oesBuffer;
>+  }
>+
>+  /** Create an NV21Buffer with the same pixel content as the given I420 buffer. */
>+  public static NV21Buffer createNV21Buffer(VideoFrame.I420Buffer i420Buffer) {
>+    final int width = i420Buffer.getWidth();
>+    final int height = i420Buffer.getHeight();
>+    final int chromaStride = width;
>+    final int chromaWidth = (width + 1) / 2;
>+    final int chromaHeight = (height + 1) / 2;
>+    final int ySize = width * height;
>+
>+    final ByteBuffer nv21Buffer = ByteBuffer.allocateDirect(ySize + chromaStride * chromaHeight);
>+    // We don't care what the array offset is since we only want an array that is direct.
>+    @SuppressWarnings("ByteBufferBackingArray") final byte[] nv21Data = nv21Buffer.array();
>+
>+    for (int y = 0; y < height; ++y) {
>+      for (int x = 0; x < width; ++x) {
>+        final byte yValue = i420Buffer.getDataY().get(y * i420Buffer.getStrideY() + x);
>+        nv21Data[y * width + x] = yValue;
>+      }
>+    }
>+    for (int y = 0; y < chromaHeight; ++y) {
>+      for (int x = 0; x < chromaWidth; ++x) {
>+        final byte uValue = i420Buffer.getDataU().get(y * i420Buffer.getStrideU() + x);
>+        final byte vValue = i420Buffer.getDataV().get(y * i420Buffer.getStrideV() + x);
>+        nv21Data[ySize + y * chromaStride + 2 * x + 0] = vValue;
>+        nv21Data[ySize + y * chromaStride + 2 * x + 1] = uValue;
>+      }
>+    }
>+    return new NV21Buffer(nv21Data, width, height, /* releaseCallback= */ null);
>+  }
>+
>+  /** Create an NV12Buffer with the same pixel content as the given I420 buffer. */
>+  public static NV12Buffer createNV12Buffer(VideoFrame.I420Buffer i420Buffer) {
>+    final int width = i420Buffer.getWidth();
>+    final int height = i420Buffer.getHeight();
>+    final int chromaStride = width;
>+    final int chromaWidth = (width + 1) / 2;
>+    final int chromaHeight = (height + 1) / 2;
>+    final int ySize = width * height;
>+
>+    final ByteBuffer nv12Buffer = ByteBuffer.allocateDirect(ySize + chromaStride * chromaHeight);
>+
>+    for (int y = 0; y < height; ++y) {
>+      for (int x = 0; x < width; ++x) {
>+        final byte yValue = i420Buffer.getDataY().get(y * i420Buffer.getStrideY() + x);
>+        nv12Buffer.put(y * width + x, yValue);
>+      }
>+    }
>+    for (int y = 0; y < chromaHeight; ++y) {
>+      for (int x = 0; x < chromaWidth; ++x) {
>+        final byte uValue = i420Buffer.getDataU().get(y * i420Buffer.getStrideU() + x);
>+        final byte vValue = i420Buffer.getDataV().get(y * i420Buffer.getStrideV() + x);
>+        nv12Buffer.put(ySize + y * chromaStride + 2 * x + 0, uValue);
>+        nv12Buffer.put(ySize + y * chromaStride + 2 * x + 1, vValue);
>+      }
>+    }
>+    return new NV12Buffer(width, height, /* stride= */ width, /* sliceHeight= */ height, nv12Buffer,
>+        /* releaseCallback */ null);
>+  }
>+
>+  /** Print the ByteBuffer plane to the StringBuilder. */
>+  private static void printPlane(
>+      StringBuilder stringBuilder, int width, int height, ByteBuffer plane, int stride) {
>+    for (int y = 0; y < height; ++y) {
>+      for (int x = 0; x < width; ++x) {
>+        final int value = plane.get(y * stride + x) & 0xFF;
>+        if (x != 0) {
>+          stringBuilder.append(", ");
>+        }
>+        stringBuilder.append(value);
>+      }
>+      stringBuilder.append("\n");
>+    }
>+  }
>+
>+  /** Convert the pixel content of an I420 buffer to a string representation. */
>+  private static String i420BufferToString(VideoFrame.I420Buffer buffer) {
>+    final StringBuilder stringBuilder = new StringBuilder();
>+    stringBuilder.append(
>+        "I420 buffer with size: " + buffer.getWidth() + "x" + buffer.getHeight() + ".\n");
>+    stringBuilder.append("Y-plane:\n");
>+    printPlane(stringBuilder, buffer.getWidth(), buffer.getHeight(), buffer.getDataY(),
>+        buffer.getStrideY());
>+    final int chromaWidth = (buffer.getWidth() + 1) / 2;
>+    final int chromaHeight = (buffer.getHeight() + 1) / 2;
>+    stringBuilder.append("U-plane:\n");
>+    printPlane(stringBuilder, chromaWidth, chromaHeight, buffer.getDataU(), buffer.getStrideU());
>+    stringBuilder.append("V-plane:\n");
>+    printPlane(stringBuilder, chromaWidth, chromaHeight, buffer.getDataV(), buffer.getStrideV());
>+    return stringBuilder.toString();
>+  }
>+
>+  /**
>+   * Assert that the given I420 buffers are almost identical, allowing for some difference due to
>+   * numerical errors. It has limits for both overall PSNR and maximum individual pixel difference.
>+   */
>+  public static void assertAlmostEqualI420Buffers(
>+      VideoFrame.I420Buffer bufferA, VideoFrame.I420Buffer bufferB) {
>+    final int diff = maxDiff(bufferA, bufferB);
>+    assertThat("Pixel difference too high: " + diff + "."
>+            + "\nBuffer A: " + i420BufferToString(bufferA)
>+            + "Buffer B: " + i420BufferToString(bufferB),
>+        diff, lessThanOrEqualTo(4));
>+    final double psnr = calculatePsnr(bufferA, bufferB);
>+    assertThat("PSNR too low: " + psnr + "."
>+            + "\nBuffer A: " + i420BufferToString(bufferA)
>+            + "Buffer B: " + i420BufferToString(bufferB),
>+        psnr, greaterThanOrEqualTo(50.0));
>+  }
>+
>+  /** Returns a flattened list of pixel differences for two ByteBuffer planes. */
>+  private static List<Integer> getPixelDiffs(
>+      int width, int height, ByteBuffer planeA, int strideA, ByteBuffer planeB, int strideB) {
>+    List<Integer> res = new ArrayList<>();
>+    for (int y = 0; y < height; ++y) {
>+      for (int x = 0; x < width; ++x) {
>+        final int valueA = planeA.get(y * strideA + x) & 0xFF;
>+        final int valueB = planeB.get(y * strideB + x) & 0xFF;
>+        res.add(Math.abs(valueA - valueB));
>+      }
>+    }
>+    return res;
>+  }
>+
>+  /** Returns a flattened list of pixel differences for two I420 buffers. */
>+  private static List<Integer> getPixelDiffs(
>+      VideoFrame.I420Buffer bufferA, VideoFrame.I420Buffer bufferB) {
>+    assertEquals(bufferA.getWidth(), bufferB.getWidth());
>+    assertEquals(bufferA.getHeight(), bufferB.getHeight());
>+    final int width = bufferA.getWidth();
>+    final int height = bufferA.getHeight();
>+    final int chromaWidth = (width + 1) / 2;
>+    final int chromaHeight = (height + 1) / 2;
>+    final List<Integer> diffs = getPixelDiffs(width, height, bufferA.getDataY(),
>+        bufferA.getStrideY(), bufferB.getDataY(), bufferB.getStrideY());
>+    diffs.addAll(getPixelDiffs(chromaWidth, chromaHeight, bufferA.getDataU(), bufferA.getStrideU(),
>+        bufferB.getDataU(), bufferB.getStrideU()));
>+    diffs.addAll(getPixelDiffs(chromaWidth, chromaHeight, bufferA.getDataV(), bufferA.getStrideV(),
>+        bufferB.getDataV(), bufferB.getStrideV()));
>+    return diffs;
>+  }
>+
>+  /** Returns the maximum pixel difference from any of the Y/U/V planes in the given buffers. */
>+  private static int maxDiff(VideoFrame.I420Buffer bufferA, VideoFrame.I420Buffer bufferB) {
>+    return Collections.max(getPixelDiffs(bufferA, bufferB));
>+  }
>+
>+  /**
>+   * Returns the PSNR given a sum of squared error and the number of measurements that were added.
>+   */
>+  private static double sseToPsnr(long sse, int count) {
>+    if (sse == 0) {
>+      return Double.POSITIVE_INFINITY;
>+    }
>+    final double meanSquaredError = (double) sse / (double) count;
>+    final double maxPixelValue = 255.0;
>+    return 10.0 * Math.log10(maxPixelValue * maxPixelValue / meanSquaredError);
>+  }
>+
>+  /** Returns the PSNR of the given I420 buffers. */
>+  private static double calculatePsnr(
>+      VideoFrame.I420Buffer bufferA, VideoFrame.I420Buffer bufferB) {
>+    final List<Integer> pixelDiffs = getPixelDiffs(bufferA, bufferB);
>+    long sse = 0;
>+    for (int pixelDiff : pixelDiffs) {
>+      sse += pixelDiff * pixelDiff;
>+    }
>+    return sseToPsnr(sse, pixelDiffs.size());
>+  }
>+
>+  /**
>+   * Convert an int array to a byte array and make sure the values are within the range [0, 255].
>+   */
>+  private static byte[] toByteArray(int[] array) {
>+    final byte[] res = new byte[array.length];
>+    for (int i = 0; i < array.length; ++i) {
>+      final int value = array[i];
>+      assertThat(value, greaterThanOrEqualTo(0));
>+      assertThat(value, lessThanOrEqualTo(255));
>+      res[i] = (byte) value;
>+    }
>+    return res;
>+  }
>+
>+  /** Convert a byte array to a direct ByteBuffer. */
>+  private static ByteBuffer toByteBuffer(int[] array) {
>+    final ByteBuffer buffer = ByteBuffer.allocateDirect(array.length);
>+    buffer.put(toByteArray(array));
>+    buffer.rewind();
>+    return buffer;
>+  }
>+
>+  /**
>+   * Draw an I420 buffer on the currently bound frame buffer, allocating and releasing any
>+   * resources necessary.
>+   */
>+  private static void drawI420Buffer(VideoFrame.I420Buffer i420Buffer) {
>+    final GlRectDrawer drawer = new GlRectDrawer();
>+    final VideoFrameDrawer videoFrameDrawer = new VideoFrameDrawer();
>+    videoFrameDrawer.drawFrame(
>+        new VideoFrame(i420Buffer, /* rotation= */ 0, /* timestampNs= */ 0), drawer);
>+    videoFrameDrawer.release();
>+    drawer.release();
>+  }
>+
>+  /**
>+   * Helper function that tests cropAndScale() with the given cropping and scaling parameters, and
>+   * compares the pixel content against a reference I420 buffer.
>+   */
>+  private void testCropAndScale(
>+      int cropX, int cropY, int cropWidth, int cropHeight, int scaleWidth, int scaleHeight) {
>+    final VideoFrame.I420Buffer referenceI420Buffer = createTestI420Buffer();
>+    final VideoFrame.Buffer bufferToTest = createBufferToTest(referenceI420Buffer);
>+
>+    final VideoFrame.Buffer croppedReferenceBuffer = referenceI420Buffer.cropAndScale(
>+        cropX, cropY, cropWidth, cropHeight, scaleWidth, scaleHeight);
>+    referenceI420Buffer.release();
>+    final VideoFrame.I420Buffer croppedReferenceI420Buffer = croppedReferenceBuffer.toI420();
>+    croppedReferenceBuffer.release();
>+
>+    final VideoFrame.Buffer croppedBufferToTest =
>+        bufferToTest.cropAndScale(cropX, cropY, cropWidth, cropHeight, scaleWidth, scaleHeight);
>+    bufferToTest.release();
>+
>+    final VideoFrame.I420Buffer croppedOutputI420Buffer = croppedBufferToTest.toI420();
>+    croppedBufferToTest.release();
>+
>+    assertAlmostEqualI420Buffers(croppedReferenceI420Buffer, croppedOutputI420Buffer);
>+    croppedReferenceI420Buffer.release();
>+    croppedOutputI420Buffer.release();
>+  }
>+
>+  @Test
>+  @SmallTest
>+  /** Test calling toI420() and comparing pixel content against I420 reference. */
>+  public void testToI420() {
>+    final VideoFrame.I420Buffer referenceI420Buffer = createTestI420Buffer();
>+    final VideoFrame.Buffer bufferToTest = createBufferToTest(referenceI420Buffer);
>+
>+    final VideoFrame.I420Buffer outputI420Buffer = bufferToTest.toI420();
>+    bufferToTest.release();
>+
>+    assertAlmostEqualI420Buffers(referenceI420Buffer, outputI420Buffer);
>+    referenceI420Buffer.release();
>+    outputI420Buffer.release();
>+  }
>+
>+  @Test
>+  @SmallTest
>+  /** Pure 2x scaling with no cropping. */
>+  public void testScale2x() {
>+    testCropAndScale(0 /* cropX= */, 0 /* cropY= */, /* cropWidth= */ 16, /* cropHeight= */ 16,
>+        /* scaleWidth= */ 8, /* scaleHeight= */ 8);
>+  }
>+
>+  @Test
>+  @SmallTest
>+  /** Test cropping only X direction, with no scaling. */
>+  public void testCropX() {
>+    testCropAndScale(8 /* cropX= */, 0 /* cropY= */, /* cropWidth= */ 8, /* cropHeight= */ 16,
>+        /* scaleWidth= */ 8, /* scaleHeight= */ 16);
>+  }
>+
>+  @Test
>+  @SmallTest
>+  /** Test cropping only Y direction, with no scaling. */
>+  public void testCropY() {
>+    testCropAndScale(0 /* cropX= */, 8 /* cropY= */, /* cropWidth= */ 16, /* cropHeight= */ 8,
>+        /* scaleWidth= */ 16, /* scaleHeight= */ 8);
>+  }
>+
>+  @Test
>+  @SmallTest
>+  /** Test center crop, with no scaling. */
>+  public void testCenterCrop() {
>+    testCropAndScale(4 /* cropX= */, 4 /* cropY= */, /* cropWidth= */ 8, /* cropHeight= */ 8,
>+        /* scaleWidth= */ 8, /* scaleHeight= */ 8);
>+  }
>+
>+  @Test
>+  @SmallTest
>+  /** Test non-center crop for right bottom corner, with no scaling. */
>+  public void testRightBottomCornerCrop() {
>+    testCropAndScale(8 /* cropX= */, 8 /* cropY= */, /* cropWidth= */ 8, /* cropHeight= */ 8,
>+        /* scaleWidth= */ 8, /* scaleHeight= */ 8);
>+  }
>+
>+  @Test
>+  @SmallTest
>+  /** Test combined cropping and scaling. */
>+  public void testCropAndScale() {
>+    testCropAndScale(4 /* cropX= */, 4 /* cropY= */, /* cropWidth= */ 12, /* cropHeight= */ 12,
>+        /* scaleWidth= */ 8, /* scaleHeight= */ 8);
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/WebRtcJniBootTest.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/WebRtcJniBootTest.java
>index ebee38dd64f..e6c4328dd93 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/WebRtcJniBootTest.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/WebRtcJniBootTest.java
>@@ -27,9 +27,8 @@ public class WebRtcJniBootTest {
>   public void testJniLoadsWithoutError() throws InterruptedException {
>     PeerConnectionFactory.initialize(PeerConnectionFactory.InitializationOptions
>                                          .builder(InstrumentationRegistry.getTargetContext())
>+                                         .setNativeLibraryName(TestConstants.NATIVE_LIBRARY)
>                                          .createInitializationOptions());
>-
>-    PeerConnectionFactory.Options options = new PeerConnectionFactory.Options();
>-    new PeerConnectionFactory(options);
>+    PeerConnectionFactory.builder().createPeerConnectionFactory();
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/YuvHelperTest.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/YuvHelperTest.java
>index a6ded5cc7a7..c8a88f0e6e7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/YuvHelperTest.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/instrumentationtests/src/org/webrtc/YuvHelperTest.java
>@@ -55,7 +55,18 @@ public class YuvHelperTest {
> 
>   @Before
>   public void setUp() {
>-    NativeLibrary.initialize(new NativeLibrary.DefaultLoader());
>+    NativeLibrary.initialize(new NativeLibrary.DefaultLoader(), TestConstants.NATIVE_LIBRARY);
>+  }
>+
>+  @SmallTest
>+  @Test
>+  public void testCopyPlane() {
>+    final int dstStride = TEST_WIDTH;
>+    final ByteBuffer dst = ByteBuffer.allocateDirect(TEST_HEIGHT * dstStride);
>+
>+    YuvHelper.copyPlane(TEST_I420_Y, TEST_I420_STRIDE_Y, dst, dstStride, TEST_WIDTH, TEST_HEIGHT);
>+
>+    assertByteBufferContentEquals(new byte[] {1, 2, 3, 4, 5, 6, 7, 8, 9}, dst);
>   }
> 
>   @SmallTest
>@@ -129,4 +140,36 @@ public class YuvHelperTest {
>       assertEquals("Unexpected ByteBuffer contents at index: " + i, expected[i], test.get(i));
>     }
>   }
>+
>+  @SmallTest
>+  @Test
>+  public void testI420Rotate90() {
>+    final int dstStrideY = TEST_HEIGHT;
>+    final int dstStrideU = TEST_CHROMA_HEIGHT;
>+    final int dstStrideV = TEST_CHROMA_HEIGHT;
>+    final ByteBuffer dstY = ByteBuffer.allocateDirect(TEST_WIDTH * dstStrideY);
>+    final ByteBuffer dstU = ByteBuffer.allocateDirect(TEST_CHROMA_WIDTH * dstStrideU);
>+    final ByteBuffer dstV = ByteBuffer.allocateDirect(TEST_CHROMA_WIDTH * dstStrideV);
>+
>+    YuvHelper.I420Rotate(TEST_I420_Y, TEST_I420_STRIDE_Y, TEST_I420_U, TEST_I420_STRIDE_V,
>+        TEST_I420_V, TEST_I420_STRIDE_U, dstY, dstStrideY, dstU, dstStrideU, dstV, dstStrideV,
>+        TEST_WIDTH, TEST_HEIGHT, 90);
>+
>+    assertByteBufferContentEquals(new byte[] {7, 4, 1, 8, 5, 2, 9, 6, 3}, dstY);
>+    assertByteBufferContentEquals(new byte[] {53, 51, 54, 52}, dstU);
>+    assertByteBufferContentEquals(new byte[] {105, 101, 106, 102}, dstV);
>+  }
>+
>+  @SmallTest
>+  @Test
>+  public void testI420Rotate90Tight() {
>+    final ByteBuffer dst = ByteBuffer.allocateDirect(
>+        TEST_WIDTH * TEST_HEIGHT + TEST_CHROMA_WIDTH * TEST_CHROMA_HEIGHT * 2);
>+
>+    YuvHelper.I420Rotate(TEST_I420_Y, TEST_I420_STRIDE_Y, TEST_I420_U, TEST_I420_STRIDE_V,
>+        TEST_I420_V, TEST_I420_STRIDE_U, dst, TEST_WIDTH, TEST_HEIGHT, 90);
>+
>+    assertByteBufferContentEquals(
>+        new byte[] {7, 4, 1, 8, 5, 2, 9, 6, 3, 53, 51, 54, 52, 105, 101, 106, 102}, dst);
>+  }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/DEPS b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/DEPS
>new file mode 100644
>index 00000000000..020e1cbf09e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/DEPS
>@@ -0,0 +1,4 @@
>+include_rules = [
>+  "+modules/audio_device/include/audio_device.h",
>+  "+system_wrappers/include",
>+]
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/audio_device_module/audio_device_android.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/audio_device_module/audio_device_android.cc
>new file mode 100644
>index 00000000000..89ffeb6971d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/audio_device_module/audio_device_android.cc
>@@ -0,0 +1,148 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "sdk/android/native_api/audio_device_module/audio_device_android.h"
>+
>+#include <stdlib.h>
>+#include <utility>
>+
>+#include "absl/memory/memory.h"
>+#include "rtc_base/logging.h"
>+#include "rtc_base/refcount.h"
>+#include "rtc_base/refcountedobject.h"
>+
>+#if defined(AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO)
>+#include "sdk/android/src/jni/audio_device/aaudio_player.h"
>+#include "sdk/android/src/jni/audio_device/aaudio_recorder.h"
>+#endif
>+
>+#include "sdk/android/src/jni/audio_device/audio_record_jni.h"
>+#include "sdk/android/src/jni/audio_device/audio_track_jni.h"
>+#include "sdk/android/src/jni/audio_device/opensles_player.h"
>+#include "sdk/android/src/jni/audio_device/opensles_recorder.h"
>+#include "system_wrappers/include/metrics.h"
>+
>+namespace webrtc {
>+
>+namespace {
>+
>+void GetDefaultAudioParameters(JNIEnv* env,
>+                               jobject application_context,
>+                               AudioParameters* input_parameters,
>+                               AudioParameters* output_parameters) {
>+  const JavaParamRef<jobject> j_context(application_context);
>+  const ScopedJavaLocalRef<jobject> j_audio_manager =
>+      jni::GetAudioManager(env, j_context);
>+  const int sample_rate = jni::GetDefaultSampleRate(env, j_audio_manager);
>+  jni::GetAudioParameters(env, j_context, j_audio_manager, sample_rate,
>+                          false /* use_stereo_input */,
>+                          false /* use_stereo_output */, input_parameters,
>+                          output_parameters);
>+}
>+
>+}  // namespace
>+
>+#if defined(AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO)
>+rtc::scoped_refptr<AudioDeviceModule> CreateAAudioAudioDeviceModule(
>+    JNIEnv* env,
>+    jobject application_context) {
>+  RTC_LOG(INFO) << __FUNCTION__;
>+  // Get default audio input/output parameters.
>+  AudioParameters input_parameters;
>+  AudioParameters output_parameters;
>+  GetDefaultAudioParameters(env, application_context, &input_parameters,
>+                            &output_parameters);
>+  // Create ADM from AAudioRecorder and AAudioPlayer.
>+  return CreateAudioDeviceModuleFromInputAndOutput(
>+      AudioDeviceModule::kAndroidAAudioAudio, false /* use_stereo_input */,
>+      false /* use_stereo_output */,
>+      jni::kLowLatencyModeDelayEstimateInMilliseconds,
>+      absl::make_unique<jni::AAudioRecorder>(input_parameters),
>+      absl::make_unique<jni::AAudioPlayer>(output_parameters));
>+}
>+#endif
>+
>+rtc::scoped_refptr<AudioDeviceModule> CreateJavaAudioDeviceModule(
>+    JNIEnv* env,
>+    jobject application_context) {
>+  RTC_LOG(INFO) << __FUNCTION__;
>+  // Get default audio input/output parameters.
>+  const JavaParamRef<jobject> j_context(application_context);
>+  const ScopedJavaLocalRef<jobject> j_audio_manager =
>+      jni::GetAudioManager(env, j_context);
>+  AudioParameters input_parameters;
>+  AudioParameters output_parameters;
>+  GetDefaultAudioParameters(env, application_context, &input_parameters,
>+                            &output_parameters);
>+  // Create ADM from AudioRecord and AudioTrack.
>+  auto audio_input = absl::make_unique<jni::AudioRecordJni>(
>+      env, input_parameters, jni::kHighLatencyModeDelayEstimateInMilliseconds,
>+      jni::AudioRecordJni::CreateJavaWebRtcAudioRecord(env, j_context,
>+                                                       j_audio_manager));
>+  auto audio_output = absl::make_unique<jni::AudioTrackJni>(
>+      env, output_parameters,
>+      jni::AudioTrackJni::CreateJavaWebRtcAudioTrack(env, j_context,
>+                                                     j_audio_manager));
>+  return CreateAudioDeviceModuleFromInputAndOutput(
>+      AudioDeviceModule::kAndroidJavaAudio, false /* use_stereo_input */,
>+      false /* use_stereo_output */,
>+      jni::kHighLatencyModeDelayEstimateInMilliseconds, std::move(audio_input),
>+      std::move(audio_output));
>+}
>+
>+rtc::scoped_refptr<AudioDeviceModule> CreateOpenSLESAudioDeviceModule(
>+    JNIEnv* env,
>+    jobject application_context) {
>+  RTC_LOG(INFO) << __FUNCTION__;
>+  // Get default audio input/output parameters.
>+  AudioParameters input_parameters;
>+  AudioParameters output_parameters;
>+  GetDefaultAudioParameters(env, application_context, &input_parameters,
>+                            &output_parameters);
>+  // Create ADM from OpenSLESRecorder and OpenSLESPlayer.
>+  auto engine_manager = absl::make_unique<jni::OpenSLEngineManager>();
>+  auto audio_input = absl::make_unique<jni::OpenSLESRecorder>(
>+      input_parameters, engine_manager.get());
>+  auto audio_output = absl::make_unique<jni::OpenSLESPlayer>(
>+      output_parameters, std::move(engine_manager));
>+  return CreateAudioDeviceModuleFromInputAndOutput(
>+      AudioDeviceModule::kAndroidOpenSLESAudio, false /* use_stereo_input */,
>+      false /* use_stereo_output */,
>+      jni::kLowLatencyModeDelayEstimateInMilliseconds, std::move(audio_input),
>+      std::move(audio_output));
>+}
>+
>+rtc::scoped_refptr<AudioDeviceModule>
>+CreateJavaInputAndOpenSLESOutputAudioDeviceModule(JNIEnv* env,
>+                                                  jobject application_context) {
>+  RTC_LOG(INFO) << __FUNCTION__;
>+  // Get default audio input/output parameters.
>+  const JavaParamRef<jobject> j_context(application_context);
>+  const ScopedJavaLocalRef<jobject> j_audio_manager =
>+      jni::GetAudioManager(env, j_context);
>+  AudioParameters input_parameters;
>+  AudioParameters output_parameters;
>+  GetDefaultAudioParameters(env, application_context, &input_parameters,
>+                            &output_parameters);
>+  // Create ADM from AudioRecord and OpenSLESPlayer.
>+  auto audio_input = absl::make_unique<jni::AudioRecordJni>(
>+      env, input_parameters, jni::kLowLatencyModeDelayEstimateInMilliseconds,
>+      jni::AudioRecordJni::CreateJavaWebRtcAudioRecord(env, j_context,
>+                                                       j_audio_manager));
>+  auto audio_output = absl::make_unique<jni::OpenSLESPlayer>(
>+      output_parameters, absl::make_unique<jni::OpenSLEngineManager>());
>+  return CreateAudioDeviceModuleFromInputAndOutput(
>+      AudioDeviceModule::kAndroidJavaInputAndOpenSLESOutputAudio,
>+      false /* use_stereo_input */, false /* use_stereo_output */,
>+      jni::kLowLatencyModeDelayEstimateInMilliseconds, std::move(audio_input),
>+      std::move(audio_output));
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/audio_device_module/audio_device_android.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/audio_device_module/audio_device_android.h
>new file mode 100644
>index 00000000000..ef404fd277e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/audio_device_module/audio_device_android.h
>@@ -0,0 +1,40 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef SDK_ANDROID_NATIVE_API_AUDIO_DEVICE_MODULE_AUDIO_DEVICE_ANDROID_H_
>+#define SDK_ANDROID_NATIVE_API_AUDIO_DEVICE_MODULE_AUDIO_DEVICE_ANDROID_H_
>+
>+#include <jni.h>
>+
>+#include "modules/audio_device/include/audio_device.h"
>+
>+namespace webrtc {
>+
>+#if defined(AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO)
>+rtc::scoped_refptr<AudioDeviceModule> CreateAAudioAudioDeviceModule(
>+    JNIEnv* env,
>+    jobject application_context);
>+#endif
>+
>+rtc::scoped_refptr<AudioDeviceModule> CreateJavaAudioDeviceModule(
>+    JNIEnv* env,
>+    jobject application_context);
>+
>+rtc::scoped_refptr<AudioDeviceModule> CreateOpenSLESAudioDeviceModule(
>+    JNIEnv* env,
>+    jobject application_context);
>+
>+rtc::scoped_refptr<AudioDeviceModule>
>+CreateJavaInputAndOpenSLESOutputAudioDeviceModule(JNIEnv* env,
>+                                                  jobject application_context);
>+
>+}  // namespace webrtc
>+
>+#endif  // SDK_ANDROID_NATIVE_API_AUDIO_DEVICE_MODULE_AUDIO_DEVICE_ANDROID_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/base/init.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/base/init.cc
>new file mode 100644
>index 00000000000..176aa89ecec
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/base/init.cc
>@@ -0,0 +1,24 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "sdk/android/native_api/base/init.h"
>+
>+#include "rtc_base/checks.h"
>+#include "sdk/android/native_api/jni/class_loader.h"
>+#include "sdk/android/src/jni/jni_helpers.h"
>+
>+namespace webrtc {
>+
>+void InitAndroid(JavaVM* jvm) {
>+  RTC_CHECK_GE(jni::InitGlobalJniVariables(jvm), 0);
>+  InitClassLoader(jni::GetEnv());
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/base/init.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/base/init.h
>new file mode 100644
>index 00000000000..d6a0ec15090
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/base/init.h
>@@ -0,0 +1,23 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef SDK_ANDROID_NATIVE_API_BASE_INIT_H_
>+#define SDK_ANDROID_NATIVE_API_BASE_INIT_H_
>+
>+#include <jni.h>
>+
>+namespace webrtc {
>+
>+// Initializes global state needed by WebRTC Android NDK.
>+void InitAndroid(JavaVM* jvm);
>+
>+}  // namespace webrtc
>+
>+#endif  // SDK_ANDROID_NATIVE_API_BASE_INIT_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/base/networkmonitor.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/base/networkmonitor.cc
>new file mode 100644
>index 00000000000..5d33ab781ed
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/base/networkmonitor.cc
>@@ -0,0 +1,32 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "sdk/android/native_api/base/networkmonitor.h"
>+
>+#include <memory>
>+
>+#include "absl/memory/memory.h"
>+#include "sdk/android/src/jni/androidnetworkmonitor.h"
>+
>+namespace webrtc {
>+
>+std::unique_ptr<rtc::NetworkMonitorFactory> CreateAndroidNetworkMonitorFactory(
>+    JNIEnv* env,
>+    jobject application_context) {
>+  return absl::make_unique<jni::AndroidNetworkMonitorFactory>(
>+      env, JavaParamRef<jobject>(application_context));
>+}
>+
>+std::unique_ptr<rtc::NetworkMonitorFactory>
>+CreateAndroidNetworkMonitorFactory() {
>+  return absl::make_unique<jni::AndroidNetworkMonitorFactory>();
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/base/networkmonitor.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/base/networkmonitor.h
>new file mode 100644
>index 00000000000..537a2470c95
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/base/networkmonitor.h
>@@ -0,0 +1,36 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef SDK_ANDROID_NATIVE_API_BASE_NETWORKMONITOR_H_
>+#define SDK_ANDROID_NATIVE_API_BASE_NETWORKMONITOR_H_
>+
>+#include <jni.h>
>+
>+#include <memory>
>+
>+#include "rtc_base/networkmonitor.h"
>+
>+namespace webrtc {
>+
>+// Creates an Android-specific network monitor, which is capable of detecting
>+// network changes as soon as they occur, requesting a cellular interface
>+// (dependent on permissions), and binding sockets to network interfaces (more
>+// reliable than binding to IP addresses on Android).
>+std::unique_ptr<rtc::NetworkMonitorFactory> CreateAndroidNetworkMonitorFactory(
>+    JNIEnv* env,
>+    jobject application_context);
>+
>+// Deprecated. Pass in application context instead.
>+std::unique_ptr<rtc::NetworkMonitorFactory>
>+CreateAndroidNetworkMonitorFactory();
>+
>+}  // namespace webrtc
>+
>+#endif  // SDK_ANDROID_NATIVE_API_BASE_NETWORKMONITOR_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/codecs/wrapper.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/codecs/wrapper.cc
>new file mode 100644
>index 00000000000..3c5fb3d3e17
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/codecs/wrapper.cc
>@@ -0,0 +1,40 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "sdk/android/native_api/codecs/wrapper.h"
>+
>+#include "absl/memory/memory.h"
>+#include "sdk/android/native_api/jni/scoped_java_ref.h"
>+#include "sdk/android/src/jni/videocodecinfo.h"
>+#include "sdk/android/src/jni/videodecoderfactorywrapper.h"
>+#include "sdk/android/src/jni/videoencoderfactorywrapper.h"
>+
>+namespace webrtc {
>+
>+SdpVideoFormat JavaToNativeVideoCodecInfo(JNIEnv* jni, jobject codec_info) {
>+  return jni::VideoCodecInfoToSdpVideoFormat(jni,
>+                                             JavaParamRef<jobject>(codec_info));
>+}
>+
>+std::unique_ptr<VideoDecoderFactory> JavaToNativeVideoDecoderFactory(
>+    JNIEnv* jni,
>+    jobject decoder_factory) {
>+  return absl::make_unique<jni::VideoDecoderFactoryWrapper>(
>+      jni, JavaParamRef<jobject>(decoder_factory));
>+}
>+
>+std::unique_ptr<VideoEncoderFactory> JavaToNativeVideoEncoderFactory(
>+    JNIEnv* jni,
>+    jobject encoder_factory) {
>+  return absl::make_unique<jni::VideoEncoderFactoryWrapper>(
>+      jni, JavaParamRef<jobject>(encoder_factory));
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/codecs/wrapper.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/codecs/wrapper.h
>new file mode 100644
>index 00000000000..b1ff24c6796
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/codecs/wrapper.h
>@@ -0,0 +1,40 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef SDK_ANDROID_NATIVE_API_CODECS_WRAPPER_H_
>+#define SDK_ANDROID_NATIVE_API_CODECS_WRAPPER_H_
>+
>+#include <jni.h>
>+#include <memory>
>+
>+#include "api/video_codecs/sdp_video_format.h"
>+#include "api/video_codecs/video_decoder_factory.h"
>+#include "api/video_codecs/video_encoder_factory.h"
>+
>+namespace webrtc {
>+
>+// Creates an instance of webrtc::SdpVideoFormat from Java VideoCodecInfo.
>+SdpVideoFormat JavaToNativeVideoCodecInfo(JNIEnv* jni, jobject codec_info);
>+
>+// Creates an instance of webrtc::VideoDecoderFactory from Java
>+// VideoDecoderFactory.
>+std::unique_ptr<VideoDecoderFactory> JavaToNativeVideoDecoderFactory(
>+    JNIEnv* jni,
>+    jobject decoder_factory);
>+
>+// Creates an instance of webrtc::VideoEncoderFactory from Java
>+// VideoEncoderFactory.
>+std::unique_ptr<VideoEncoderFactory> JavaToNativeVideoEncoderFactory(
>+    JNIEnv* jni,
>+    jobject encoder_factory);
>+
>+}  // namespace webrtc
>+
>+#endif  // SDK_ANDROID_NATIVE_API_CODECS_WRAPPER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/class_loader.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/jni/class_loader.cc
>similarity index 66%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/class_loader.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/jni/class_loader.cc
>index 84b3e9062a2..2c06460f9eb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/class_loader.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/jni/class_loader.cc
>@@ -8,13 +8,15 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "sdk/android/src/jni/class_loader.h"
>+#include "sdk/android/native_api/jni/class_loader.h"
> 
> #include <algorithm>
> #include <string>
> 
> #include "rtc_base/checks.h"
>-#include "sdk/android/generated_base_jni/jni/WebRtcClassLoader_jni.h"
>+#include "sdk/android/generated_native_api_jni/jni/WebRtcClassLoader_jni.h"
>+#include "sdk/android/native_api/jni/java_types.h"
>+#include "sdk/android/native_api/jni/scoped_java_ref.h"
> 
> // Abort the process if |jni| has a Java exception pending. This macros uses the
> // comma operator to execute ExceptionDescribe and ExceptionClear ignoring their
>@@ -24,13 +26,13 @@
>       << (jni->ExceptionDescribe(), jni->ExceptionClear(), "")
> 
> namespace webrtc {
>-namespace jni {
> 
> namespace {
> 
> class ClassLoader {
>  public:
>-  explicit ClassLoader(JNIEnv* env) {
>+  explicit ClassLoader(JNIEnv* env)
>+      : class_loader_(jni::Java_WebRtcClassLoader_getClassLoader(env)) {
>     class_loader_class_ = reinterpret_cast<jclass>(
>         env->NewGlobalRef(env->FindClass("java/lang/ClassLoader")));
>     CHECK_EXCEPTION(env);
>@@ -38,27 +40,24 @@ class ClassLoader {
>         env->GetMethodID(class_loader_class_, "loadClass",
>                          "(Ljava/lang/String;)Ljava/lang/Class;");
>     CHECK_EXCEPTION(env);
>-    class_loader_ =
>-        env->NewGlobalRef(Java_WebRtcClassLoader_getClassLoader(env));
>-    CHECK_EXCEPTION(env);
>   }
> 
>-  jclass FindClass(JNIEnv* env, const char* c_name) {
>+  ScopedJavaLocalRef<jclass> FindClass(JNIEnv* env, const char* c_name) {
>     // ClassLoader.loadClass expects a classname with components separated by
>     // dots instead of the slashes that JNIEnv::FindClass expects.
>     std::string name(c_name);
>     std::replace(name.begin(), name.end(), '/', '.');
>-    jstring jstr = env->NewStringUTF(name.c_str());
>-    const jclass clazz = static_cast<jclass>(
>-        env->CallObjectMethod(class_loader_, load_class_method_, jstr));
>+    ScopedJavaLocalRef<jstring> j_name = NativeToJavaString(env, name);
>+    const jclass clazz = static_cast<jclass>(env->CallObjectMethod(
>+        class_loader_.obj(), load_class_method_, j_name.obj()));
>     CHECK_EXCEPTION(env);
>-    return clazz;
>+    return ScopedJavaLocalRef<jclass>(env, clazz);
>   }
> 
>  private:
>+  ScopedJavaGlobalRef<jobject> class_loader_;
>   jclass class_loader_class_;
>   jmethodID load_class_method_;
>-  jobject class_loader_;
> };
> 
> static ClassLoader* g_class_loader = nullptr;
>@@ -70,12 +69,12 @@ void InitClassLoader(JNIEnv* env) {
>   g_class_loader = new ClassLoader(env);
> }
> 
>-jclass GetClass(JNIEnv* env, const char* name) {
>+ScopedJavaLocalRef<jclass> GetClass(JNIEnv* env, const char* name) {
>   // The class loader will be null in the JNI code called from the ClassLoader
>   // ctor when we are bootstrapping ourself.
>-  return (g_class_loader == nullptr) ? env->FindClass(name)
>-                                     : g_class_loader->FindClass(env, name);
>+  return (g_class_loader == nullptr)
>+             ? ScopedJavaLocalRef<jclass>(env, env->FindClass(name))
>+             : g_class_loader->FindClass(env, name);
> }
> 
>-}  // namespace jni
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/jni/class_loader.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/jni/class_loader.h
>new file mode 100644
>index 00000000000..2d102fe4a22
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/jni/class_loader.h
>@@ -0,0 +1,40 @@
>+/*
>+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+// Android's FindClass() is tricky because the app-specific ClassLoader is not
>+// consulted when there is no app-specific frame on the stack (i.e. when called
>+// from a thread created from native C++ code). These helper functions provide a
>+// workaround for this.
>+// http://developer.android.com/training/articles/perf-jni.html#faq_FindClass
>+
>+#ifndef SDK_ANDROID_NATIVE_API_JNI_CLASS_LOADER_H_
>+#define SDK_ANDROID_NATIVE_API_JNI_CLASS_LOADER_H_
>+
>+#include <jni.h>
>+
>+#include "sdk/android/native_api/jni/scoped_java_ref.h"
>+
>+namespace webrtc {
>+
>+// This method should be called from JNI_OnLoad and before any calls to
>+// FindClass. This is normally called by InitAndroid.
>+void InitClassLoader(JNIEnv* env);
>+
>+// This function is identical to JNIEnv::FindClass except that it works from any
>+// thread. This function loads and returns a local reference to the class with
>+// the given name. The name argument is a fully-qualified class name. For
>+// example, the fully-qualified class name for the java.lang.String class is:
>+// "java/lang/String". This function will be used from the JNI generated code
>+// and should rarely be used manually.
>+ScopedJavaLocalRef<jclass> GetClass(JNIEnv* env, const char* name);
>+
>+}  // namespace webrtc
>+
>+#endif  // SDK_ANDROID_NATIVE_API_JNI_CLASS_LOADER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/jni/java_types.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/jni/java_types.cc
>new file mode 100644
>index 00000000000..df0a22905e7
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/jni/java_types.cc
>@@ -0,0 +1,294 @@
>+/*
>+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "sdk/android/native_api/jni/java_types.h"
>+
>+#include <string>
>+#include <utility>
>+
>+#include "sdk/android/generated_external_classes_jni/jni/ArrayList_jni.h"
>+#include "sdk/android/generated_external_classes_jni/jni/Boolean_jni.h"
>+#include "sdk/android/generated_external_classes_jni/jni/Double_jni.h"
>+#include "sdk/android/generated_external_classes_jni/jni/Enum_jni.h"
>+#include "sdk/android/generated_external_classes_jni/jni/Integer_jni.h"
>+#include "sdk/android/generated_external_classes_jni/jni/Iterable_jni.h"
>+#include "sdk/android/generated_external_classes_jni/jni/Iterator_jni.h"
>+#include "sdk/android/generated_external_classes_jni/jni/LinkedHashMap_jni.h"
>+#include "sdk/android/generated_external_classes_jni/jni/Long_jni.h"
>+#include "sdk/android/generated_external_classes_jni/jni/Map_jni.h"
>+#include "sdk/android/generated_native_api_jni/jni/JniHelper_jni.h"
>+
>+namespace webrtc {
>+
>+Iterable::Iterable(JNIEnv* jni, const JavaRef<jobject>& iterable)
>+    : jni_(jni), iterable_(jni, iterable) {}
>+
>+Iterable::Iterable(Iterable&& other) = default;
>+
>+Iterable::~Iterable() = default;
>+
>+// Creates an iterator representing the end of any collection.
>+Iterable::Iterator::Iterator() = default;
>+
>+// Creates an iterator pointing to the beginning of the specified collection.
>+Iterable::Iterator::Iterator(JNIEnv* jni, const JavaRef<jobject>& iterable)
>+    : jni_(jni) {
>+  iterator_ = JNI_Iterable::Java_Iterable_iterator(jni, iterable);
>+  RTC_CHECK(!iterator_.is_null());
>+  // Start at the first element in the collection.
>+  ++(*this);
>+}
>+
>+// Move constructor - necessary to be able to return iterator types from
>+// functions.
>+Iterable::Iterator::Iterator(Iterator&& other)
>+    : jni_(std::move(other.jni_)),
>+      iterator_(std::move(other.iterator_)),
>+      value_(std::move(other.value_)),
>+      thread_checker_(std::move(other.thread_checker_)) {}
>+
>+Iterable::Iterator::~Iterator() = default;
>+
>+// Advances the iterator one step.
>+Iterable::Iterator& Iterable::Iterator::operator++() {
>+  RTC_CHECK(thread_checker_.CalledOnValidThread());
>+  if (AtEnd()) {
>+    // Can't move past the end.
>+    return *this;
>+  }
>+  bool has_next = JNI_Iterator::Java_Iterator_hasNext(jni_, iterator_);
>+  if (!has_next) {
>+    iterator_ = nullptr;
>+    value_ = nullptr;
>+    return *this;
>+  }
>+
>+  value_ = JNI_Iterator::Java_Iterator_next(jni_, iterator_);
>+  return *this;
>+}
>+
>+void Iterable::Iterator::Remove() {
>+  JNI_Iterator::Java_Iterator_remove(jni_, iterator_);
>+}
>+
>+// Provides a way to compare the iterator with itself and with the end iterator.
>+// Note: all other comparison results are undefined, just like for C++ input
>+// iterators.
>+bool Iterable::Iterator::operator==(const Iterable::Iterator& other) {
>+  // Two different active iterators should never be compared.
>+  RTC_DCHECK(this == &other || AtEnd() || other.AtEnd());
>+  return AtEnd() == other.AtEnd();
>+}
>+
>+ScopedJavaLocalRef<jobject>& Iterable::Iterator::operator*() {
>+  RTC_CHECK(!AtEnd());
>+  return value_;
>+}
>+
>+bool Iterable::Iterator::AtEnd() const {
>+  RTC_CHECK(thread_checker_.CalledOnValidThread());
>+  return jni_ == nullptr || IsNull(jni_, iterator_);
>+}
>+
>+bool IsNull(JNIEnv* jni, const JavaRef<jobject>& obj) {
>+  return jni->IsSameObject(obj.obj(), nullptr);
>+}
>+
>+std::string GetJavaEnumName(JNIEnv* jni, const JavaRef<jobject>& j_enum) {
>+  return JavaToStdString(jni, JNI_Enum::Java_Enum_name(jni, j_enum));
>+}
>+
>+Iterable GetJavaMapEntrySet(JNIEnv* jni, const JavaRef<jobject>& j_map) {
>+  return Iterable(jni, JNI_Map::Java_Map_entrySet(jni, j_map));
>+}
>+
>+ScopedJavaLocalRef<jobject> GetJavaMapEntryKey(
>+    JNIEnv* jni,
>+    const JavaRef<jobject>& j_entry) {
>+  return jni::Java_JniHelper_getKey(jni, j_entry);
>+}
>+
>+ScopedJavaLocalRef<jobject> GetJavaMapEntryValue(
>+    JNIEnv* jni,
>+    const JavaRef<jobject>& j_entry) {
>+  return jni::Java_JniHelper_getValue(jni, j_entry);
>+}
>+
>+int64_t JavaToNativeLong(JNIEnv* env, const JavaRef<jobject>& j_long) {
>+  return JNI_Long::Java_Long_longValue(env, j_long);
>+}
>+
>+absl::optional<bool> JavaToNativeOptionalBool(JNIEnv* jni,
>+                                              const JavaRef<jobject>& boolean) {
>+  if (IsNull(jni, boolean))
>+    return absl::nullopt;
>+  return JNI_Boolean::Java_Boolean_booleanValue(jni, boolean);
>+}
>+
>+absl::optional<int32_t> JavaToNativeOptionalInt(
>+    JNIEnv* jni,
>+    const JavaRef<jobject>& integer) {
>+  if (IsNull(jni, integer))
>+    return absl::nullopt;
>+  return JNI_Integer::Java_Integer_intValue(jni, integer);
>+}
>+
>+// Given a jstring, reinterprets it to a new native string.
>+std::string JavaToNativeString(JNIEnv* jni, const JavaRef<jstring>& j_string) {
>+  const ScopedJavaLocalRef<jbyteArray> j_byte_array =
>+      jni::Java_JniHelper_getStringBytes(jni, j_string);
>+
>+  const size_t len = jni->GetArrayLength(j_byte_array.obj());
>+  CHECK_EXCEPTION(jni) << "error during GetArrayLength";
>+  std::string str(len, '\0');
>+  jni->GetByteArrayRegion(j_byte_array.obj(), 0, len,
>+                          reinterpret_cast<jbyte*>(&str[0]));
>+  CHECK_EXCEPTION(jni) << "error during GetByteArrayRegion";
>+  return str;
>+}
>+
>+std::map<std::string, std::string> JavaToNativeStringMap(
>+    JNIEnv* jni,
>+    const JavaRef<jobject>& j_map) {
>+  return JavaToNativeMap<std::string, std::string>(
>+      jni, j_map,
>+      [](JNIEnv* env, JavaRef<jobject> const& key,
>+         JavaRef<jobject> const& value) {
>+        return std::make_pair(
>+            JavaToNativeString(env, static_java_ref_cast<jstring>(env, key)),
>+            JavaToNativeString(env, static_java_ref_cast<jstring>(env, value)));
>+      });
>+}
>+
>+ScopedJavaLocalRef<jobject> NativeToJavaBoolean(JNIEnv* env, bool b) {
>+  return JNI_Boolean::Java_Boolean_ConstructorJLB_Z(env, b);
>+}
>+
>+ScopedJavaLocalRef<jobject> NativeToJavaDouble(JNIEnv* env, double d) {
>+  return JNI_Double::Java_Double_ConstructorJLD_D(env, d);
>+}
>+
>+ScopedJavaLocalRef<jobject> NativeToJavaInteger(JNIEnv* jni, int32_t i) {
>+  return JNI_Integer::Java_Integer_ConstructorJLI_I(jni, i);
>+}
>+
>+ScopedJavaLocalRef<jobject> NativeToJavaLong(JNIEnv* env, int64_t u) {
>+  return JNI_Long::Java_Long_ConstructorJLLO_J(env, u);
>+}
>+
>+ScopedJavaLocalRef<jstring> NativeToJavaString(JNIEnv* env, const char* str) {
>+  jstring j_str = env->NewStringUTF(str);
>+  CHECK_EXCEPTION(env) << "error during NewStringUTF";
>+  return ScopedJavaLocalRef<jstring>(env, j_str);
>+}
>+
>+ScopedJavaLocalRef<jstring> NativeToJavaString(JNIEnv* jni,
>+                                               const std::string& str) {
>+  return NativeToJavaString(jni, str.c_str());
>+}
>+
>+ScopedJavaLocalRef<jobject> NativeToJavaInteger(
>+    JNIEnv* jni,
>+    const absl::optional<int32_t>& optional_int) {
>+  return optional_int ? NativeToJavaInteger(jni, *optional_int) : nullptr;
>+}
>+
>+ScopedJavaLocalRef<jstring> NativeToJavaString(
>+    JNIEnv* jni,
>+    const absl::optional<std::string>& str) {
>+  return str ? NativeToJavaString(jni, *str) : nullptr;
>+}
>+
>+ScopedJavaLocalRef<jobjectArray> NativeToJavaBooleanArray(
>+    JNIEnv* env,
>+    const std::vector<bool>& container) {
>+  return NativeToJavaObjectArray(env, container, java_lang_Boolean_clazz(env),
>+                                 &NativeToJavaBoolean);
>+}
>+
>+ScopedJavaLocalRef<jobjectArray> NativeToJavaDoubleArray(
>+    JNIEnv* env,
>+    const std::vector<double>& container) {
>+  return NativeToJavaObjectArray(env, container, java_lang_Double_clazz(env),
>+                                 &NativeToJavaDouble);
>+}
>+
>+ScopedJavaLocalRef<jobjectArray> NativeToJavaIntegerArray(
>+    JNIEnv* env,
>+    const std::vector<int32_t>& container) {
>+  ScopedJavaLocalRef<jobject> (*convert_function)(JNIEnv*, int32_t) =
>+      &NativeToJavaInteger;
>+  return NativeToJavaObjectArray(env, container, java_lang_Integer_clazz(env),
>+                                 convert_function);
>+}
>+
>+ScopedJavaLocalRef<jobjectArray> NativeToJavaLongArray(
>+    JNIEnv* env,
>+    const std::vector<int64_t>& container) {
>+  return NativeToJavaObjectArray(env, container, java_lang_Long_clazz(env),
>+                                 &NativeToJavaLong);
>+}
>+
>+ScopedJavaLocalRef<jobjectArray> NativeToJavaStringArray(
>+    JNIEnv* env,
>+    const std::vector<std::string>& container) {
>+  ScopedJavaLocalRef<jstring> (*convert)(JNIEnv*, const std::string&) =
>+      &NativeToJavaString;
>+  return NativeToJavaObjectArray(
>+      env, container,
>+      static_cast<jclass>(jni::Java_JniHelper_getStringClass(env).obj()),
>+      convert);
>+}
>+
>+JavaListBuilder::JavaListBuilder(JNIEnv* env)
>+    : env_(env), j_list_(JNI_ArrayList::Java_ArrayList_ConstructorJUALI(env)) {}
>+
>+JavaListBuilder::~JavaListBuilder() = default;
>+
>+void JavaListBuilder::add(const JavaRef<jobject>& element) {
>+  JNI_ArrayList::Java_ArrayList_addZ_JUE(env_, j_list_, element);
>+}
>+
>+JavaMapBuilder::JavaMapBuilder(JNIEnv* env)
>+    : env_(env),
>+      j_map_(JNI_LinkedHashMap::Java_LinkedHashMap_ConstructorJULIHM(env)) {}
>+
>+JavaMapBuilder::~JavaMapBuilder() = default;
>+
>+void JavaMapBuilder::put(const JavaRef<jobject>& key,
>+                         const JavaRef<jobject>& value) {
>+  JNI_Map::Java_Map_put(env_, j_map_, key, value);
>+}
>+
>+jlong NativeToJavaPointer(void* ptr) {
>+  static_assert(sizeof(intptr_t) <= sizeof(jlong),
>+                "Time to rethink the use of jlongs");
>+  // Going through intptr_t to be obvious about the definedness of the
>+  // conversion from pointer to integral type.  intptr_t to jlong is a standard
>+  // widening by the static_assert above.
>+  jlong ret = reinterpret_cast<intptr_t>(ptr);
>+  RTC_DCHECK(reinterpret_cast<void*>(ret) == ptr);
>+  return ret;
>+}
>+
>+// Given a list of jstrings, reinterprets it to a new vector of native strings.
>+std::vector<std::string> JavaToStdVectorStrings(JNIEnv* jni,
>+                                                const JavaRef<jobject>& list) {
>+  std::vector<std::string> converted_list;
>+  if (!list.is_null()) {
>+    for (const JavaRef<jobject>& str : Iterable(jni, list)) {
>+      converted_list.push_back(JavaToStdString(
>+          jni, JavaParamRef<jstring>(static_cast<jstring>(str.obj()))));
>+    }
>+  }
>+  return converted_list;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/jni/java_types.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/jni/java_types.h
>new file mode 100644
>index 00000000000..3b857049ccc
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/jni/java_types.h
>@@ -0,0 +1,343 @@
>+/*
>+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+// Android's FindClass() is tricky because the app-specific ClassLoader is not
>+// consulted when there is no app-specific frame on the stack (i.e. when called
>+// from a thread created from native C++ code). These helper functions provide a
>+// workaround for this.
>+// http://developer.android.com/training/articles/perf-jni.html#faq_FindClass
>+
>+#ifndef SDK_ANDROID_NATIVE_API_JNI_JAVA_TYPES_H_
>+#define SDK_ANDROID_NATIVE_API_JNI_JAVA_TYPES_H_
>+
>+#include <jni.h>
>+#include <map>
>+#include <string>
>+#include <vector>
>+
>+#include "absl/types/optional.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/thread_checker.h"
>+#include "sdk/android/native_api/jni/scoped_java_ref.h"
>+
>+// Abort the process if |jni| has a Java exception pending.
>+// This macros uses the comma operator to execute ExceptionDescribe
>+// and ExceptionClear ignoring their return values and sending ""
>+// to the error stream.
>+#define CHECK_EXCEPTION(jni)        \
>+  RTC_CHECK(!jni->ExceptionCheck()) \
>+      << (jni->ExceptionDescribe(), jni->ExceptionClear(), "")
>+
>+namespace webrtc {
>+
>+// ---------------
>+// -- Utilities --
>+// ---------------
>+
>+// Provides a convenient way to iterate over a Java Iterable using the
>+// C++ range-for loop.
>+// E.g. for (jobject value : Iterable(jni, j_iterable)) { ... }
>+// Note: Since Java iterators cannot be duplicated, the iterator class is not
>+// copyable to prevent creating multiple C++ iterators that refer to the same
>+// Java iterator.
>+class Iterable {
>+ public:
>+  Iterable(JNIEnv* jni, const JavaRef<jobject>& iterable);
>+  Iterable(Iterable&& other);
>+
>+  ~Iterable();
>+
>+  class Iterator {
>+   public:
>+    // Creates an iterator representing the end of any collection.
>+    Iterator();
>+    // Creates an iterator pointing to the beginning of the specified
>+    // collection.
>+    Iterator(JNIEnv* jni, const JavaRef<jobject>& iterable);
>+
>+    // Move constructor - necessary to be able to return iterator types from
>+    // functions.
>+    Iterator(Iterator&& other);
>+
>+    ~Iterator();
>+
>+    // Move assignment should not be used.
>+    Iterator& operator=(Iterator&&) = delete;
>+
>+    // Advances the iterator one step.
>+    Iterator& operator++();
>+
>+    // Removes the element the iterator is pointing to. Must still advance the
>+    // iterator afterwards.
>+    void Remove();
>+
>+    // Provides a way to compare the iterator with itself and with the end
>+    // iterator.
>+    // Note: all other comparison results are undefined, just like for C++ input
>+    // iterators.
>+    bool operator==(const Iterator& other);
>+    bool operator!=(const Iterator& other) { return !(*this == other); }
>+    ScopedJavaLocalRef<jobject>& operator*();
>+
>+   private:
>+    bool AtEnd() const;
>+
>+    JNIEnv* jni_ = nullptr;
>+    ScopedJavaLocalRef<jobject> iterator_;
>+    ScopedJavaLocalRef<jobject> value_;
>+    rtc::ThreadChecker thread_checker_;
>+
>+    RTC_DISALLOW_COPY_AND_ASSIGN(Iterator);
>+  };
>+
>+  Iterable::Iterator begin() { return Iterable::Iterator(jni_, iterable_); }
>+  Iterable::Iterator end() { return Iterable::Iterator(); }
>+
>+ private:
>+  JNIEnv* jni_;
>+  ScopedJavaLocalRef<jobject> iterable_;
>+
>+  RTC_DISALLOW_COPY_AND_ASSIGN(Iterable);
>+};
>+
>+// Returns true if |obj| == null in Java.
>+bool IsNull(JNIEnv* jni, const JavaRef<jobject>& obj);
>+
>+// Returns the name of a Java enum.
>+std::string GetJavaEnumName(JNIEnv* jni, const JavaRef<jobject>& j_enum);
>+
>+Iterable GetJavaMapEntrySet(JNIEnv* jni, const JavaRef<jobject>& j_map);
>+ScopedJavaLocalRef<jobject> GetJavaMapEntryKey(JNIEnv* jni,
>+                                               const JavaRef<jobject>& j_entry);
>+ScopedJavaLocalRef<jobject> GetJavaMapEntryValue(
>+    JNIEnv* jni,
>+    const JavaRef<jobject>& j_entry);
>+
>+// --------------------------------------------------------
>+// -- Methods for converting Java types to native types. --
>+// --------------------------------------------------------
>+
>+int64_t JavaToNativeLong(JNIEnv* env, const JavaRef<jobject>& j_long);
>+
>+absl::optional<bool> JavaToNativeOptionalBool(JNIEnv* jni,
>+                                              const JavaRef<jobject>& boolean);
>+absl::optional<int32_t> JavaToNativeOptionalInt(
>+    JNIEnv* jni,
>+    const JavaRef<jobject>& integer);
>+
>+// Given a (UTF-16) jstring return a new UTF-8 native string.
>+std::string JavaToNativeString(JNIEnv* jni, const JavaRef<jstring>& j_string);
>+
>+template <typename T, typename Convert>
>+std::vector<T> JavaToNativeVector(JNIEnv* env,
>+                                  const JavaRef<jobjectArray>& j_container,
>+                                  Convert convert) {
>+  std::vector<T> container;
>+  const size_t size = env->GetArrayLength(j_container.obj());
>+  container.reserve(size);
>+  for (size_t i = 0; i < size; ++i) {
>+    container.emplace_back(convert(
>+        env, ScopedJavaLocalRef<jobject>(
>+                 env, env->GetObjectArrayElement(j_container.obj(), i))));
>+  }
>+  CHECK_EXCEPTION(env) << "Error during JavaToNativeVector";
>+  return container;
>+}
>+
>+template <typename T, typename Java_T = jobject, typename Convert>
>+std::vector<T> JavaListToNativeVector(JNIEnv* env,
>+                                      const JavaRef<jobject>& j_list,
>+                                      Convert convert) {
>+  std::vector<T> native_list;
>+  if (!j_list.is_null()) {
>+    for (ScopedJavaLocalRef<jobject>& j_item : Iterable(env, j_list)) {
>+      native_list.emplace_back(
>+          convert(env, static_java_ref_cast<Java_T>(env, j_item)));
>+    }
>+    CHECK_EXCEPTION(env) << "Error during JavaListToNativeVector";
>+  }
>+  return native_list;
>+}
>+
>+template <typename Key, typename T, typename Convert>
>+std::map<Key, T> JavaToNativeMap(JNIEnv* env,
>+                                 const JavaRef<jobject>& j_map,
>+                                 Convert convert) {
>+  std::map<Key, T> container;
>+  for (auto const& j_entry : GetJavaMapEntrySet(env, j_map)) {
>+    container.emplace(convert(env, GetJavaMapEntryKey(env, j_entry),
>+                              GetJavaMapEntryValue(env, j_entry)));
>+  }
>+  return container;
>+}
>+
>+// Converts Map<String, String> to std::map<std::string, std::string>.
>+std::map<std::string, std::string> JavaToNativeStringMap(
>+    JNIEnv* env,
>+    const JavaRef<jobject>& j_map);
>+
>+// --------------------------------------------------------
>+// -- Methods for converting native types to Java types. --
>+// --------------------------------------------------------
>+
>+ScopedJavaLocalRef<jobject> NativeToJavaBoolean(JNIEnv* env, bool b);
>+ScopedJavaLocalRef<jobject> NativeToJavaDouble(JNIEnv* env, double d);
>+ScopedJavaLocalRef<jobject> NativeToJavaInteger(JNIEnv* jni, int32_t i);
>+ScopedJavaLocalRef<jobject> NativeToJavaLong(JNIEnv* env, int64_t u);
>+ScopedJavaLocalRef<jstring> NativeToJavaString(JNIEnv* jni, const char* str);
>+ScopedJavaLocalRef<jstring> NativeToJavaString(JNIEnv* jni,
>+                                               const std::string& str);
>+
>+ScopedJavaLocalRef<jobject> NativeToJavaInteger(
>+    JNIEnv* jni,
>+    const absl::optional<int32_t>& optional_int);
>+ScopedJavaLocalRef<jstring> NativeToJavaString(
>+    JNIEnv* jni,
>+    const absl::optional<std::string>& str);
>+
>+// Helper function for converting std::vector<T> into a Java array.
>+template <typename T, typename Convert>
>+ScopedJavaLocalRef<jobjectArray> NativeToJavaObjectArray(
>+    JNIEnv* env,
>+    const std::vector<T>& container,
>+    jclass clazz,
>+    Convert convert) {
>+  ScopedJavaLocalRef<jobjectArray> j_container(
>+      env, env->NewObjectArray(container.size(), clazz, nullptr));
>+  int i = 0;
>+  for (const T& element : container) {
>+    env->SetObjectArrayElement(j_container.obj(), i,
>+                               convert(env, element).obj());
>+    ++i;
>+  }
>+  return j_container;
>+}
>+
>+ScopedJavaLocalRef<jobjectArray> NativeToJavaBooleanArray(
>+    JNIEnv* env,
>+    const std::vector<bool>& container);
>+ScopedJavaLocalRef<jobjectArray> NativeToJavaDoubleArray(
>+    JNIEnv* env,
>+    const std::vector<double>& container);
>+ScopedJavaLocalRef<jobjectArray> NativeToJavaIntegerArray(
>+    JNIEnv* env,
>+    const std::vector<int32_t>& container);
>+ScopedJavaLocalRef<jobjectArray> NativeToJavaLongArray(
>+    JNIEnv* env,
>+    const std::vector<int64_t>& container);
>+ScopedJavaLocalRef<jobjectArray> NativeToJavaStringArray(
>+    JNIEnv* env,
>+    const std::vector<std::string>& container);
>+
>+// This is a helper class for NativeToJavaList(). Use that function instead of
>+// using this class directly.
>+class JavaListBuilder {
>+ public:
>+  explicit JavaListBuilder(JNIEnv* env);
>+  ~JavaListBuilder();
>+  void add(const JavaRef<jobject>& element);
>+  ScopedJavaLocalRef<jobject> java_list() { return j_list_; }
>+
>+ private:
>+  JNIEnv* env_;
>+  ScopedJavaLocalRef<jobject> j_list_;
>+};
>+
>+template <typename C, typename Convert>
>+ScopedJavaLocalRef<jobject> NativeToJavaList(JNIEnv* env,
>+                                             const C& container,
>+                                             Convert convert) {
>+  JavaListBuilder builder(env);
>+  for (const auto& e : container)
>+    builder.add(convert(env, e));
>+  return builder.java_list();
>+}
>+
>+// This is a helper class for NativeToJavaMap(). Use that function instead of
>+// using this class directly.
>+class JavaMapBuilder {
>+ public:
>+  explicit JavaMapBuilder(JNIEnv* env);
>+  ~JavaMapBuilder();
>+  void put(const JavaRef<jobject>& key, const JavaRef<jobject>& value);
>+  ScopedJavaLocalRef<jobject> GetJavaMap() { return j_map_; }
>+
>+ private:
>+  JNIEnv* env_;
>+  ScopedJavaLocalRef<jobject> j_map_;
>+};
>+
>+template <typename C, typename Convert>
>+ScopedJavaLocalRef<jobject> NativeToJavaMap(JNIEnv* env,
>+                                            const C& container,
>+                                            Convert convert) {
>+  JavaMapBuilder builder(env);
>+  for (const auto& e : container) {
>+    const auto key_value_pair = convert(env, e);
>+    builder.put(key_value_pair.first, key_value_pair.second);
>+  }
>+  return builder.GetJavaMap();
>+}
>+
>+template <typename C>
>+ScopedJavaLocalRef<jobject> NativeToJavaStringMap(JNIEnv* env,
>+                                                  const C& container) {
>+  JavaMapBuilder builder(env);
>+  for (const auto& e : container) {
>+    const auto key_value_pair = std::make_pair(
>+        NativeToJavaString(env, e.first), NativeToJavaString(env, e.second));
>+    builder.put(key_value_pair.first, key_value_pair.second);
>+  }
>+  return builder.GetJavaMap();
>+}
>+
>+// Return a |jlong| that will correctly convert back to |ptr|.  This is needed
>+// because the alternative (of silently passing a 32-bit pointer to a vararg
>+// function expecting a 64-bit param) picks up garbage in the high 32 bits.
>+jlong NativeToJavaPointer(void* ptr);
>+
>+// ------------------------
>+// -- Deprecated methods --
>+// ------------------------
>+
>+// Deprecated. Use JavaToNativeString.
>+inline std::string JavaToStdString(JNIEnv* jni,
>+                                   const JavaRef<jstring>& j_string) {
>+  return JavaToNativeString(jni, j_string);
>+}
>+
>+// Deprecated. Use scoped jobjects instead.
>+inline std::string JavaToStdString(JNIEnv* jni, jstring j_string) {
>+  return JavaToStdString(jni, JavaParamRef<jstring>(j_string));
>+}
>+
>+// Deprecated. Use JavaListToNativeVector<std::string, jstring> instead.
>+// Given a List of (UTF-16) jstrings
>+// return a new vector of UTF-8 native strings.
>+std::vector<std::string> JavaToStdVectorStrings(JNIEnv* jni,
>+                                                const JavaRef<jobject>& list);
>+
>+// Deprecated. Use JavaToNativeStringMap instead.
>+// Parses Map<String, String> to std::map<std::string, std::string>.
>+inline std::map<std::string, std::string> JavaToStdMapStrings(
>+    JNIEnv* jni,
>+    const JavaRef<jobject>& j_map) {
>+  return JavaToNativeStringMap(jni, j_map);
>+}
>+
>+// Deprecated. Use scoped jobjects instead.
>+inline std::map<std::string, std::string> JavaToStdMapStrings(JNIEnv* jni,
>+                                                              jobject j_map) {
>+  return JavaToStdMapStrings(jni, JavaParamRef<jobject>(j_map));
>+}
>+
>+}  // namespace webrtc
>+
>+#endif  // SDK_ANDROID_NATIVE_API_JNI_JAVA_TYPES_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/jni/jni_int_wrapper.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/jni/jni_int_wrapper.h
>new file mode 100644
>index 00000000000..23da7f2ce49
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/jni/jni_int_wrapper.h
>@@ -0,0 +1,59 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+// Originally this class is from Chromium.
>+// https://cs.chromium.org/chromium/src/base/android/jni_int_wrapper.h.
>+
>+#ifndef SDK_ANDROID_NATIVE_API_JNI_JNI_INT_WRAPPER_H_
>+#define SDK_ANDROID_NATIVE_API_JNI_JNI_INT_WRAPPER_H_
>+
>+// Wrapper used to receive int when calling Java from native. The wrapper
>+// disallows automatic conversion of anything besides int32_t to a jint.
>+// Checking is only done in debugging builds.
>+
>+#ifdef NDEBUG
>+
>+typedef jint JniIntWrapper;
>+
>+// This inline is sufficiently trivial that it does not change the
>+// final code generated by g++.
>+inline jint as_jint(JniIntWrapper wrapper) {
>+  return wrapper;
>+}
>+
>+#else
>+
>+class JniIntWrapper {
>+ public:
>+  JniIntWrapper() : i_(0) {}
>+  JniIntWrapper(int32_t i) : i_(i) {}  // NOLINT(runtime/explicit)
>+  explicit JniIntWrapper(const JniIntWrapper& ji) : i_(ji.i_) {}
>+
>+  jint as_jint() const { return i_; }
>+
>+  // If you get an "invokes a deleted function" error at the lines below it is
>+  // because you used an implicit conversion to convert e.g. a long to an
>+  // int32_t when calling Java. We disallow this. If you want a lossy
>+  // conversion, please use an explicit conversion in your C++ code.
>+  JniIntWrapper(uint32_t) = delete;  // NOLINT(runtime/explicit)
>+  JniIntWrapper(uint64_t) = delete;  // NOLINT(runtime/explicit)
>+  JniIntWrapper(int64_t) = delete;   // NOLINT(runtime/explicit)
>+
>+ private:
>+  const jint i_;
>+};
>+
>+inline jint as_jint(const JniIntWrapper& wrapper) {
>+  return wrapper.as_jint();
>+}
>+
>+#endif  // NDEBUG
>+
>+#endif  // SDK_ANDROID_NATIVE_API_JNI_JNI_INT_WRAPPER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/android_test_initializer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/jni/jvm.cc
>similarity index 56%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/android_test_initializer.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/jni/jvm.cc
>index 2094b2d47f9..3356cbeb6fd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/test/android_test_initializer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/jni/jvm.cc
>@@ -1,5 +1,5 @@
> /*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>  *
>  *  Use of this source code is governed by a BSD-style license
>  *  that can be found in the LICENSE file in the root of the source
>@@ -8,13 +8,14 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#ifndef MODULES_VIDEO_CODING_CODECS_TEST_ANDROID_TEST_INITIALIZER_H_
>-#define MODULES_VIDEO_CODING_CODECS_TEST_ANDROID_TEST_INITIALIZER_H_
>+#include "sdk/android/native_api/jni/jvm.h"
>+
>+#include "sdk/android/src/jni/jvm.h"
> 
> namespace webrtc {
> 
>-void InitializeAndroidObjects();
>+JNIEnv* AttachCurrentThreadIfNeeded() {
>+  return jni::AttachCurrentThreadIfNeeded();
>+}
> 
> }  // namespace webrtc
>-
>-#endif  // MODULES_VIDEO_CODING_CODECS_TEST_ANDROID_TEST_INITIALIZER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/jni/jvm.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/jni/jvm.h
>new file mode 100644
>index 00000000000..00bce6734d8
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/jni/jvm.h
>@@ -0,0 +1,21 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef SDK_ANDROID_NATIVE_API_JNI_JVM_H_
>+#define SDK_ANDROID_NATIVE_API_JNI_JVM_H_
>+
>+#include <jni.h>
>+
>+namespace webrtc {
>+// Returns a JNI environment usable on this thread.
>+JNIEnv* AttachCurrentThreadIfNeeded();
>+}  // namespace webrtc
>+
>+#endif  // SDK_ANDROID_NATIVE_API_JNI_JVM_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/jni/scoped_java_ref.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/jni/scoped_java_ref.h
>new file mode 100644
>index 00000000000..b26ef2c4c57
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/jni/scoped_java_ref.h
>@@ -0,0 +1,210 @@
>+/*
>+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+// Originally these classes are from Chromium.
>+// https://cs.chromium.org/chromium/src/base/android/scoped_java_ref.h.
>+
>+#ifndef SDK_ANDROID_NATIVE_API_JNI_SCOPED_JAVA_REF_H_
>+#define SDK_ANDROID_NATIVE_API_JNI_SCOPED_JAVA_REF_H_
>+
>+#include <jni.h>
>+#include <utility>
>+
>+#include "rtc_base/constructormagic.h"
>+#include "sdk/android/native_api/jni/jvm.h"
>+
>+namespace webrtc {
>+
>+// Generic base class for ScopedJavaLocalRef and ScopedJavaGlobalRef. Useful
>+// for allowing functions to accept a reference without having to mandate
>+// whether it is a local or global type.
>+template <typename T>
>+class JavaRef;
>+
>+// Template specialization of JavaRef, which acts as the base class for all
>+// other JavaRef<> template types. This allows you to e.g. pass JavaRef<jstring>
>+// into a function taking const JavaRef<jobject>&.
>+template <>
>+class JavaRef<jobject> {
>+ public:
>+  jobject obj() const { return obj_; }
>+  bool is_null() const {
>+    // This is not valid for weak references. For weak references you need to
>+    // use env->IsSameObject(objc_, nullptr), but that should be avoided anyway
>+    // since it does not prevent the object from being freed immediately
>+    // thereafter. Consequently, programmers should not use this check on weak
>+    // references anyway and should first make a ScopedJavaLocalRef or
>+    // ScopedJavaGlobalRef before checking if it is null.
>+    return obj_ == nullptr;
>+  }
>+
>+ protected:
>+  constexpr JavaRef() : obj_(nullptr) {}
>+  explicit JavaRef(jobject obj) : obj_(obj) {}
>+  jobject obj_;
>+
>+ private:
>+  RTC_DISALLOW_COPY_AND_ASSIGN(JavaRef);
>+};
>+
>+template <typename T>
>+class JavaRef : public JavaRef<jobject> {
>+ public:
>+  T obj() const { return static_cast<T>(obj_); }
>+
>+ protected:
>+  JavaRef() : JavaRef<jobject>(nullptr) {}
>+  explicit JavaRef(T obj) : JavaRef<jobject>(obj) {}
>+
>+ private:
>+  RTC_DISALLOW_COPY_AND_ASSIGN(JavaRef);
>+};
>+
>+// Holds a local reference to a JNI method parameter.
>+// Method parameters should not be deleted, and so this class exists purely to
>+// wrap them as a JavaRef<T> in the JNI binding generator. Do not create
>+// instances manually.
>+template <typename T>
>+class JavaParamRef : public JavaRef<T> {
>+ public:
>+  // Assumes that |obj| is a parameter passed to a JNI method from Java.
>+  // Does not assume ownership as parameters should not be deleted.
>+  explicit JavaParamRef(T obj) : JavaRef<T>(obj) {}
>+  JavaParamRef(JNIEnv*, T obj) : JavaRef<T>(obj) {}
>+
>+ private:
>+  RTC_DISALLOW_COPY_AND_ASSIGN(JavaParamRef);
>+};
>+
>+// Holds a local reference to a Java object. The local reference is scoped
>+// to the lifetime of this object.
>+// Instances of this class may hold onto any JNIEnv passed into it until
>+// destroyed. Therefore, since a JNIEnv is only suitable for use on a single
>+// thread, objects of this class must be created, used, and destroyed, on a
>+// single thread.
>+// Therefore, this class should only be used as a stack-based object and from a
>+// single thread. If you wish to have the reference outlive the current
>+// callstack (e.g. as a class member) or you wish to pass it across threads,
>+// use a ScopedJavaGlobalRef instead.
>+template <typename T>
>+class ScopedJavaLocalRef : public JavaRef<T> {
>+ public:
>+  ScopedJavaLocalRef() = default;
>+  ScopedJavaLocalRef(std::nullptr_t) {}  // NOLINT(runtime/explicit)
>+
>+  ScopedJavaLocalRef(JNIEnv* env, const JavaRef<T>& other) : env_(env) {
>+    Reset(other.obj(), OwnershipPolicy::RETAIN);
>+  }
>+  // Allow constructing e.g. ScopedJavaLocalRef<jobject> from
>+  // ScopedJavaLocalRef<jstring>.
>+  template <typename G>
>+  ScopedJavaLocalRef(ScopedJavaLocalRef<G>&& other) : env_(other.env()) {
>+    Reset(other.Release(), OwnershipPolicy::ADOPT);
>+  }
>+  ScopedJavaLocalRef(const ScopedJavaLocalRef& other) : env_(other.env_) {
>+    Reset(other.obj(), OwnershipPolicy::RETAIN);
>+  }
>+
>+  // Assumes that |obj| is a reference to a Java object and takes
>+  // ownership  of this  reference. This should preferably not be used
>+  // outside of JNI helper functions.
>+  ScopedJavaLocalRef(JNIEnv* env, T obj) : JavaRef<T>(obj), env_(env) {}
>+
>+  ~ScopedJavaLocalRef() {
>+    if (obj_ != nullptr)
>+      env_->DeleteLocalRef(obj_);
>+  }
>+
>+  void operator=(const ScopedJavaLocalRef& other) {
>+    Reset(other.obj(), OwnershipPolicy::RETAIN);
>+  }
>+  void operator=(ScopedJavaLocalRef&& other) {
>+    Reset(other.Release(), OwnershipPolicy::ADOPT);
>+  }
>+
>+  // Releases the reference to the caller. The caller *must* delete the
>+  // reference when it is done with it. Note that calling a Java method
>+  // is *not* a transfer of ownership and Release() should not be used.
>+  T Release() {
>+    T obj = static_cast<T>(obj_);
>+    obj_ = nullptr;
>+    return obj;
>+  }
>+
>+  JNIEnv* env() const { return env_; }
>+
>+ private:
>+  using JavaRef<T>::obj_;
>+
>+  enum OwnershipPolicy {
>+    // The scoped object takes ownership of an object by taking over an existing
>+    // ownership claim.
>+    ADOPT,
>+    // The scoped object will retain the the object and any initial ownership is
>+    // not changed.
>+    RETAIN
>+  };
>+
>+  void Reset(T obj, OwnershipPolicy policy) {
>+    if (obj_ != nullptr)
>+      env_->DeleteLocalRef(obj_);
>+    obj_ = (obj != nullptr && policy == OwnershipPolicy::RETAIN)
>+               ? env_->NewLocalRef(obj)
>+               : obj;
>+  }
>+
>+  JNIEnv* const env_ = AttachCurrentThreadIfNeeded();
>+};
>+
>+// Holds a global reference to a Java object. The global reference is scoped
>+// to the lifetime of this object. This class does not hold onto any JNIEnv*
>+// passed to it, hence it is safe to use across threads (within the constraints
>+// imposed by the underlying Java object that it references).
>+template <typename T>
>+class ScopedJavaGlobalRef : public JavaRef<T> {
>+ public:
>+  using JavaRef<T>::obj_;
>+
>+  explicit constexpr ScopedJavaGlobalRef(std::nullptr_t) {}
>+  ScopedJavaGlobalRef(JNIEnv* env, const JavaRef<T>& other)
>+      : JavaRef<T>(static_cast<T>(env->NewGlobalRef(other.obj()))) {}
>+  explicit ScopedJavaGlobalRef(const ScopedJavaLocalRef<T>& other)
>+      : ScopedJavaGlobalRef(other.env(), other) {}
>+  ScopedJavaGlobalRef(ScopedJavaGlobalRef&& other)
>+      : JavaRef<T>(other.Release()) {}
>+
>+  ~ScopedJavaGlobalRef() {
>+    if (obj_ != nullptr)
>+      AttachCurrentThreadIfNeeded()->DeleteGlobalRef(obj_);
>+  }
>+
>+  // Releases the reference to the caller. The caller *must* delete the
>+  // reference when it is done with it. Note that calling a Java method
>+  // is *not* a transfer of ownership and Release() should not be used.
>+  T Release() {
>+    T obj = static_cast<T>(obj_);
>+    obj_ = nullptr;
>+    return obj;
>+  }
>+
>+ private:
>+  RTC_DISALLOW_COPY_AND_ASSIGN(ScopedJavaGlobalRef);
>+};
>+
>+template <typename T>
>+inline ScopedJavaLocalRef<T> static_java_ref_cast(JNIEnv* env,
>+                                                  JavaRef<jobject> const& ref) {
>+  ScopedJavaLocalRef<jobject> owned_ref(env, ref);
>+  return ScopedJavaLocalRef<T>(env, static_cast<T>(owned_ref.Release()));
>+}
>+
>+}  // namespace webrtc
>+
>+#endif  // SDK_ANDROID_NATIVE_API_JNI_SCOPED_JAVA_REF_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/peerconnection/peerconnectionfactory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/peerconnection/peerconnectionfactory.cc
>new file mode 100644
>index 00000000000..892905bf0f0
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/peerconnection/peerconnectionfactory.cc
>@@ -0,0 +1,33 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#include "sdk/android/native_api/peerconnection/peerconnectionfactory.h"
>+
>+#include <jni.h>
>+
>+#include <memory>
>+#include <utility>
>+
>+#include "sdk/android/src/jni/pc/peerconnectionfactory.h"
>+
>+namespace webrtc {
>+
>+jobject NativeToJavaPeerConnectionFactory(
>+    JNIEnv* jni,
>+    rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> pcf,
>+    std::unique_ptr<rtc::Thread> network_thread,
>+    std::unique_ptr<rtc::Thread> worker_thread,
>+    std::unique_ptr<rtc::Thread> signaling_thread,
>+    rtc::NetworkMonitorFactory* network_monitor_factory) {
>+  return webrtc::jni::NativeToJavaPeerConnectionFactory(
>+      jni, pcf, std::move(network_thread), std::move(worker_thread),
>+      std::move(signaling_thread), network_monitor_factory);
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/peerconnection/peerconnectionfactory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/peerconnection/peerconnectionfactory.h
>new file mode 100644
>index 00000000000..652be7adc60
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/peerconnection/peerconnectionfactory.h
>@@ -0,0 +1,34 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef SDK_ANDROID_NATIVE_API_PEERCONNECTION_PEERCONNECTIONFACTORY_H_
>+#define SDK_ANDROID_NATIVE_API_PEERCONNECTION_PEERCONNECTIONFACTORY_H_
>+
>+#include <jni.h>
>+
>+#include <memory>
>+
>+#include "api/peerconnectioninterface.h"
>+#include "rtc_base/thread.h"
>+
>+namespace webrtc {
>+
>+// Creates java PeerConnectionFactory with specified |pcf|.
>+jobject NativeToJavaPeerConnectionFactory(
>+    JNIEnv* jni,
>+    rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> pcf,
>+    std::unique_ptr<rtc::Thread> network_thread,
>+    std::unique_ptr<rtc::Thread> worker_thread,
>+    std::unique_ptr<rtc::Thread> signaling_thread,
>+    rtc::NetworkMonitorFactory* network_monitor_factory = nullptr);
>+
>+}  // namespace webrtc
>+
>+#endif  // SDK_ANDROID_NATIVE_API_PEERCONNECTION_PEERCONNECTIONFACTORY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/video/videosource.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/video/videosource.cc
>new file mode 100644
>index 00000000000..4c302da0ed9
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/video/videosource.cc
>@@ -0,0 +1,104 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "sdk/android/native_api/video/videosource.h"
>+
>+#include "sdk/android/src/jni/androidvideotracksource.h"
>+#include "sdk/android/src/jni/nativecapturerobserver.h"
>+
>+namespace webrtc {
>+
>+namespace {
>+
>+// Hides full jni::AndroidVideoTrackSource interface and provides an instance of
>+// NativeCapturerObserver associated with the video source. Does not extend
>+// AndroidVideoTrackSource to avoid diamond inheritance on
>+// VideoTrackSourceInterface.
>+class JavaVideoTrackSourceImpl : public JavaVideoTrackSourceInterface {
>+ public:
>+  JavaVideoTrackSourceImpl(JNIEnv* env,
>+                           rtc::Thread* signaling_thread,
>+                           bool is_screencast)
>+      : android_video_track_source_(
>+            new rtc::RefCountedObject<jni::AndroidVideoTrackSource>(
>+                signaling_thread,
>+                env,
>+                is_screencast)),
>+        native_capturer_observer_(jni::CreateJavaNativeCapturerObserver(
>+            env,
>+            android_video_track_source_)) {}
>+
>+  ScopedJavaLocalRef<jobject> GetJavaVideoCapturerObserver(
>+      JNIEnv* env) override {
>+    return ScopedJavaLocalRef<jobject>(env, native_capturer_observer_);
>+  }
>+
>+  // Delegate VideoTrackSourceInterface methods to android_video_track_source_.
>+  void RegisterObserver(ObserverInterface* observer) override {
>+    android_video_track_source_->RegisterObserver(observer);
>+  }
>+
>+  void UnregisterObserver(ObserverInterface* observer) override {
>+    android_video_track_source_->UnregisterObserver(observer);
>+  }
>+
>+  SourceState state() const override {
>+    return android_video_track_source_->state();
>+  }
>+
>+  bool remote() const override { return android_video_track_source_->remote(); }
>+
>+  void AddOrUpdateSink(rtc::VideoSinkInterface<VideoFrame>* sink,
>+                       const rtc::VideoSinkWants& wants) override {
>+    // The method is defined private in the implementation so we have to access
>+    // it through the interface...
>+    static_cast<VideoTrackSourceInterface*>(android_video_track_source_.get())
>+        ->AddOrUpdateSink(sink, wants);
>+  }
>+
>+  void RemoveSink(rtc::VideoSinkInterface<VideoFrame>* sink) override {
>+    // The method is defined private in the implementation so we have to access
>+    // it through the interface...
>+    static_cast<VideoTrackSourceInterface*>(android_video_track_source_.get())
>+        ->RemoveSink(sink);
>+  }
>+
>+  bool is_screencast() const override {
>+    return android_video_track_source_->is_screencast();
>+  }
>+
>+  absl::optional<bool> needs_denoising() const override {
>+    return android_video_track_source_->needs_denoising();
>+  }
>+
>+  bool GetStats(Stats* stats) override {
>+    // The method is defined private in the implementation so we have to access
>+    // it through the interface...
>+    return static_cast<VideoTrackSourceInterface*>(
>+               android_video_track_source_.get())
>+        ->GetStats(stats);
>+  }
>+
>+ private:
>+  rtc::scoped_refptr<jni::AndroidVideoTrackSource> android_video_track_source_;
>+  ScopedJavaGlobalRef<jobject> native_capturer_observer_;
>+};
>+
>+}  // namespace
>+
>+rtc::scoped_refptr<JavaVideoTrackSourceInterface> CreateJavaVideoSource(
>+    JNIEnv* jni,
>+    rtc::Thread* signaling_thread,
>+    bool is_screencast) {
>+  return new rtc::RefCountedObject<JavaVideoTrackSourceImpl>(
>+      jni, signaling_thread, is_screencast);
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/video/videosource.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/video/videosource.h
>new file mode 100644
>index 00000000000..0c4f33342eb
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/video/videosource.h
>@@ -0,0 +1,39 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef SDK_ANDROID_NATIVE_API_VIDEO_VIDEOSOURCE_H_
>+#define SDK_ANDROID_NATIVE_API_VIDEO_VIDEOSOURCE_H_
>+
>+#include <jni.h>
>+
>+#include "api/mediastreaminterface.h"
>+#include "sdk/android/native_api/jni/scoped_java_ref.h"
>+
>+namespace webrtc {
>+
>+// Interface for class that implements VideoTrackSourceInterface and provides a
>+// Java object that can be used to feed frames to the source.
>+class JavaVideoTrackSourceInterface : public VideoTrackSourceInterface {
>+ public:
>+  // Returns CapturerObserver object that can be used to feed frames to the
>+  // video source.
>+  virtual ScopedJavaLocalRef<jobject> GetJavaVideoCapturerObserver(
>+      JNIEnv* env) = 0;
>+};
>+
>+// Creates an instance of JavaVideoTrackSourceInterface,
>+rtc::scoped_refptr<JavaVideoTrackSourceInterface> CreateJavaVideoSource(
>+    JNIEnv* env,
>+    rtc::Thread* signaling_thread,
>+    bool is_screencast);
>+
>+}  // namespace webrtc
>+
>+#endif  // SDK_ANDROID_NATIVE_API_VIDEO_VIDEOSOURCE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/video/wrapper.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/video/wrapper.cc
>new file mode 100644
>index 00000000000..fe2d45a04aa
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/video/wrapper.cc
>@@ -0,0 +1,32 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "sdk/android/native_api/video/wrapper.h"
>+
>+#include "absl/memory/memory.h"
>+#include "sdk/android/native_api/jni/scoped_java_ref.h"
>+#include "sdk/android/src/jni/videoframe.h"
>+#include "sdk/android/src/jni/videosink.h"
>+
>+namespace webrtc {
>+
>+std::unique_ptr<rtc::VideoSinkInterface<VideoFrame>> JavaToNativeVideoSink(
>+    JNIEnv* jni,
>+    jobject video_sink) {
>+  return absl::make_unique<jni::VideoSinkWrapper>(
>+      jni, JavaParamRef<jobject>(video_sink));
>+}
>+
>+ScopedJavaLocalRef<jobject> NativeToJavaVideoFrame(JNIEnv* jni,
>+                                                   const VideoFrame& frame) {
>+  return jni::NativeToJavaVideoFrame(jni, frame);
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/video/wrapper.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/video/wrapper.h
>new file mode 100644
>index 00000000000..fc670ea9160
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_api/video/wrapper.h
>@@ -0,0 +1,36 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef SDK_ANDROID_NATIVE_API_VIDEO_WRAPPER_H_
>+#define SDK_ANDROID_NATIVE_API_VIDEO_WRAPPER_H_
>+
>+#include <jni.h>
>+#include <memory>
>+
>+#include "api/mediastreaminterface.h"
>+#include "api/video/video_frame.h"
>+#include "sdk/android/native_api/jni/scoped_java_ref.h"
>+
>+namespace webrtc {
>+
>+// Creates an instance of rtc::VideoSinkInterface<VideoFrame> from Java
>+// VideoSink.
>+std::unique_ptr<rtc::VideoSinkInterface<VideoFrame>> JavaToNativeVideoSink(
>+    JNIEnv* jni,
>+    jobject video_sink);
>+
>+// Creates a Java VideoFrame object from a native VideoFrame. The returned
>+// object has to be released by calling release.
>+ScopedJavaLocalRef<jobject> NativeToJavaVideoFrame(JNIEnv* jni,
>+                                                   const VideoFrame& frame);
>+
>+}  // namespace webrtc
>+
>+#endif  // SDK_ANDROID_NATIVE_API_VIDEO_WRAPPER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/DEPS b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/DEPS
>new file mode 100644
>index 00000000000..7825103fb44
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/DEPS
>@@ -0,0 +1,5 @@
>+include_rules = [
>+  "+modules/audio_device/include/audio_device.h",
>+  "+modules/audio_device/include/mock_audio_transport.h",
>+  "+system_wrappers/include",
>+]
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/audio_device/audio_device_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/audio_device/audio_device_unittest.cc
>new file mode 100644
>index 00000000000..5e46052b900
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/audio_device/audio_device_unittest.cc
>@@ -0,0 +1,1170 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <memory>
>+#include <numeric>
>+
>+#include "modules/audio_device/include/audio_device.h"
>+#include "modules/audio_device/include/mock_audio_transport.h"
>+#include "rtc_base/arraysize.h"
>+#include "rtc_base/criticalsection.h"
>+#include "rtc_base/format_macros.h"
>+#include "rtc_base/scoped_ref_ptr.h"
>+#include "rtc_base/timeutils.h"
>+#include "sdk/android/generated_native_unittests_jni/jni/ApplicationContextProvider_jni.h"
>+#include "sdk/android/generated_native_unittests_jni/jni/BuildInfo_jni.h"
>+#include "sdk/android/native_api/audio_device_module/audio_device_android.h"
>+#include "sdk/android/src/jni/audio_device/audio_common.h"
>+#include "sdk/android/src/jni/audio_device/audio_device_module.h"
>+#include "sdk/android/src/jni/audio_device/opensles_common.h"
>+#include "sdk/android/src/jni/jni_helpers.h"
>+#include "system_wrappers/include/event_wrapper.h"
>+#include "test/gmock.h"
>+#include "test/gtest.h"
>+#include "test/testsupport/fileutils.h"
>+
>+using std::cout;
>+using std::endl;
>+using ::testing::_;
>+using ::testing::AtLeast;
>+using ::testing::Gt;
>+using ::testing::Invoke;
>+using ::testing::NiceMock;
>+using ::testing::NotNull;
>+using ::testing::Return;
>+
>+// #define ENABLE_DEBUG_PRINTF
>+#ifdef ENABLE_DEBUG_PRINTF
>+#define PRINTD(...) fprintf(stderr, __VA_ARGS__);
>+#else
>+#define PRINTD(...) ((void)0)
>+#endif
>+#define PRINT(...) fprintf(stderr, __VA_ARGS__);
>+
>+namespace webrtc {
>+
>+namespace jni {
>+
>+// Number of callbacks (input or output) the tests waits for before we set
>+// an event indicating that the test was OK.
>+static const size_t kNumCallbacks = 10;
>+// Max amount of time we wait for an event to be set while counting callbacks.
>+static const int kTestTimeOutInMilliseconds = 10 * 1000;
>+// Average number of audio callbacks per second assuming 10ms packet size.
>+static const size_t kNumCallbacksPerSecond = 100;
>+// Play out a test file during this time (unit is in seconds).
>+static const int kFilePlayTimeInSec = 5;
>+static const size_t kBitsPerSample = 16;
>+static const size_t kBytesPerSample = kBitsPerSample / 8;
>+// Run the full-duplex test during this time (unit is in seconds).
>+// Note that first |kNumIgnoreFirstCallbacks| are ignored.
>+static const int kFullDuplexTimeInSec = 5;
>+// Wait for the callback sequence to stabilize by ignoring this amount of the
>+// initial callbacks (avoids initial FIFO access).
>+// Only used in the RunPlayoutAndRecordingInFullDuplex test.
>+static const size_t kNumIgnoreFirstCallbacks = 50;
>+// Sets the number of impulses per second in the latency test.
>+static const int kImpulseFrequencyInHz = 1;
>+// Length of round-trip latency measurements. Number of transmitted impulses
>+// is kImpulseFrequencyInHz * kMeasureLatencyTimeInSec - 1.
>+static const int kMeasureLatencyTimeInSec = 11;
>+// Utilized in round-trip latency measurements to avoid capturing noise samples.
>+static const int kImpulseThreshold = 1000;
>+static const char kTag[] = "[..........] ";
>+
>+enum TransportType {
>+  kPlayout = 0x1,
>+  kRecording = 0x2,
>+};
>+
>+// Interface for processing the audio stream. Real implementations can e.g.
>+// run audio in loopback, read audio from a file or perform latency
>+// measurements.
>+class AudioStreamInterface {
>+ public:
>+  virtual void Write(const void* source, size_t num_frames) = 0;
>+  virtual void Read(void* destination, size_t num_frames) = 0;
>+
>+ protected:
>+  virtual ~AudioStreamInterface() {}
>+};
>+
>+// Reads audio samples from a PCM file where the file is stored in memory at
>+// construction.
>+class FileAudioStream : public AudioStreamInterface {
>+ public:
>+  FileAudioStream(size_t num_callbacks,
>+                  const std::string& file_name,
>+                  int sample_rate)
>+      : file_size_in_bytes_(0), sample_rate_(sample_rate), file_pos_(0) {
>+    file_size_in_bytes_ = test::GetFileSize(file_name);
>+    sample_rate_ = sample_rate;
>+    EXPECT_GE(file_size_in_callbacks(), num_callbacks)
>+        << "Size of test file is not large enough to last during the test.";
>+    const size_t num_16bit_samples =
>+        test::GetFileSize(file_name) / kBytesPerSample;
>+    file_.reset(new int16_t[num_16bit_samples]);
>+    FILE* audio_file = fopen(file_name.c_str(), "rb");
>+    EXPECT_NE(audio_file, nullptr);
>+    size_t num_samples_read =
>+        fread(file_.get(), sizeof(int16_t), num_16bit_samples, audio_file);
>+    EXPECT_EQ(num_samples_read, num_16bit_samples);
>+    fclose(audio_file);
>+  }
>+
>+  // AudioStreamInterface::Write() is not implemented.
>+  void Write(const void* source, size_t num_frames) override {}
>+
>+  // Read samples from file stored in memory (at construction) and copy
>+  // |num_frames| (<=> 10ms) to the |destination| byte buffer.
>+  void Read(void* destination, size_t num_frames) override {
>+    memcpy(destination, static_cast<int16_t*>(&file_[file_pos_]),
>+           num_frames * sizeof(int16_t));
>+    file_pos_ += num_frames;
>+  }
>+
>+  int file_size_in_seconds() const {
>+    return static_cast<int>(file_size_in_bytes_ /
>+                            (kBytesPerSample * sample_rate_));
>+  }
>+  size_t file_size_in_callbacks() const {
>+    return file_size_in_seconds() * kNumCallbacksPerSecond;
>+  }
>+
>+ private:
>+  size_t file_size_in_bytes_;
>+  int sample_rate_;
>+  std::unique_ptr<int16_t[]> file_;
>+  size_t file_pos_;
>+};
>+
>+// Simple first in first out (FIFO) class that wraps a list of 16-bit audio
>+// buffers of fixed size and allows Write and Read operations. The idea is to
>+// store recorded audio buffers (using Write) and then read (using Read) these
>+// stored buffers with as short delay as possible when the audio layer needs
>+// data to play out. The number of buffers in the FIFO will stabilize under
>+// normal conditions since there will be a balance between Write and Read calls.
>+// The container is a std::list container and access is protected with a lock
>+// since both sides (playout and recording) are driven by its own thread.
>+class FifoAudioStream : public AudioStreamInterface {
>+ public:
>+  explicit FifoAudioStream(size_t frames_per_buffer)
>+      : frames_per_buffer_(frames_per_buffer),
>+        bytes_per_buffer_(frames_per_buffer_ * sizeof(int16_t)),
>+        fifo_(new AudioBufferList),
>+        largest_size_(0),
>+        total_written_elements_(0),
>+        write_count_(0) {
>+    EXPECT_NE(fifo_.get(), nullptr);
>+  }
>+
>+  ~FifoAudioStream() { Flush(); }
>+
>+  // Allocate new memory, copy |num_frames| samples from |source| into memory
>+  // and add pointer to the memory location to end of the list.
>+  // Increases the size of the FIFO by one element.
>+  void Write(const void* source, size_t num_frames) override {
>+    ASSERT_EQ(num_frames, frames_per_buffer_);
>+    PRINTD("+");
>+    if (write_count_++ < kNumIgnoreFirstCallbacks) {
>+      return;
>+    }
>+    int16_t* memory = new int16_t[frames_per_buffer_];
>+    memcpy(static_cast<int16_t*>(&memory[0]), source, bytes_per_buffer_);
>+    rtc::CritScope lock(&lock_);
>+    fifo_->push_back(memory);
>+    const size_t size = fifo_->size();
>+    if (size > largest_size_) {
>+      largest_size_ = size;
>+      PRINTD("(%" PRIuS ")", largest_size_);
>+    }
>+    total_written_elements_ += size;
>+  }
>+
>+  // Read pointer to data buffer from front of list, copy |num_frames| of stored
>+  // data into |destination| and delete the utilized memory allocation.
>+  // Decreases the size of the FIFO by one element.
>+  void Read(void* destination, size_t num_frames) override {
>+    ASSERT_EQ(num_frames, frames_per_buffer_);
>+    PRINTD("-");
>+    rtc::CritScope lock(&lock_);
>+    if (fifo_->empty()) {
>+      memset(destination, 0, bytes_per_buffer_);
>+    } else {
>+      int16_t* memory = fifo_->front();
>+      fifo_->pop_front();
>+      memcpy(destination, static_cast<int16_t*>(&memory[0]), bytes_per_buffer_);
>+      delete memory;
>+    }
>+  }
>+
>+  size_t size() const { return fifo_->size(); }
>+
>+  size_t largest_size() const { return largest_size_; }
>+
>+  size_t average_size() const {
>+    return (total_written_elements_ == 0)
>+               ? 0.0
>+               : 0.5 + static_cast<float>(total_written_elements_) /
>+                           (write_count_ - kNumIgnoreFirstCallbacks);
>+  }
>+
>+ private:
>+  void Flush() {
>+    for (auto it = fifo_->begin(); it != fifo_->end(); ++it) {
>+      delete *it;
>+    }
>+    fifo_->clear();
>+  }
>+
>+  using AudioBufferList = std::list<int16_t*>;
>+  rtc::CriticalSection lock_;
>+  const size_t frames_per_buffer_;
>+  const size_t bytes_per_buffer_;
>+  std::unique_ptr<AudioBufferList> fifo_;
>+  size_t largest_size_;
>+  size_t total_written_elements_;
>+  size_t write_count_;
>+};
>+
>+// Inserts periodic impulses and measures the latency between the time of
>+// transmission and time of receiving the same impulse.
>+// Usage requires a special hardware called Audio Loopback Dongle.
>+// See http://source.android.com/devices/audio/loopback.html for details.
>+class LatencyMeasuringAudioStream : public AudioStreamInterface {
>+ public:
>+  explicit LatencyMeasuringAudioStream(size_t frames_per_buffer)
>+      : frames_per_buffer_(frames_per_buffer),
>+        bytes_per_buffer_(frames_per_buffer_ * sizeof(int16_t)),
>+        play_count_(0),
>+        rec_count_(0),
>+        pulse_time_(0) {}
>+
>+  // Insert periodic impulses in first two samples of |destination|.
>+  void Read(void* destination, size_t num_frames) override {
>+    ASSERT_EQ(num_frames, frames_per_buffer_);
>+    if (play_count_ == 0) {
>+      PRINT("[");
>+    }
>+    play_count_++;
>+    memset(destination, 0, bytes_per_buffer_);
>+    if (play_count_ % (kNumCallbacksPerSecond / kImpulseFrequencyInHz) == 0) {
>+      if (pulse_time_ == 0) {
>+        pulse_time_ = rtc::TimeMillis();
>+      }
>+      PRINT(".");
>+      const int16_t impulse = std::numeric_limits<int16_t>::max();
>+      int16_t* ptr16 = static_cast<int16_t*>(destination);
>+      for (size_t i = 0; i < 2; ++i) {
>+        ptr16[i] = impulse;
>+      }
>+    }
>+  }
>+
>+  // Detect received impulses in |source|, derive time between transmission and
>+  // detection and add the calculated delay to list of latencies.
>+  void Write(const void* source, size_t num_frames) override {
>+    ASSERT_EQ(num_frames, frames_per_buffer_);
>+    rec_count_++;
>+    if (pulse_time_ == 0) {
>+      // Avoid detection of new impulse response until a new impulse has
>+      // been transmitted (sets |pulse_time_| to value larger than zero).
>+      return;
>+    }
>+    const int16_t* ptr16 = static_cast<const int16_t*>(source);
>+    std::vector<int16_t> vec(ptr16, ptr16 + num_frames);
>+    // Find max value in the audio buffer.
>+    int max = *std::max_element(vec.begin(), vec.end());
>+    // Find index (element position in vector) of the max element.
>+    int index_of_max =
>+        std::distance(vec.begin(), std::find(vec.begin(), vec.end(), max));
>+    if (max > kImpulseThreshold) {
>+      PRINTD("(%d,%d)", max, index_of_max);
>+      int64_t now_time = rtc::TimeMillis();
>+      int extra_delay = IndexToMilliseconds(static_cast<double>(index_of_max));
>+      PRINTD("[%d]", static_cast<int>(now_time - pulse_time_));
>+      PRINTD("[%d]", extra_delay);
>+      // Total latency is the difference between transmit time and detection
>+      // tome plus the extra delay within the buffer in which we detected the
>+      // received impulse. It is transmitted at sample 0 but can be received
>+      // at sample N where N > 0. The term |extra_delay| accounts for N and it
>+      // is a value between 0 and 10ms.
>+      latencies_.push_back(now_time - pulse_time_ + extra_delay);
>+      pulse_time_ = 0;
>+    } else {
>+      PRINTD("-");
>+    }
>+  }
>+
>+  size_t num_latency_values() const { return latencies_.size(); }
>+
>+  int min_latency() const {
>+    if (latencies_.empty())
>+      return 0;
>+    return *std::min_element(latencies_.begin(), latencies_.end());
>+  }
>+
>+  int max_latency() const {
>+    if (latencies_.empty())
>+      return 0;
>+    return *std::max_element(latencies_.begin(), latencies_.end());
>+  }
>+
>+  int average_latency() const {
>+    if (latencies_.empty())
>+      return 0;
>+    return 0.5 + static_cast<double>(
>+                     std::accumulate(latencies_.begin(), latencies_.end(), 0)) /
>+                     latencies_.size();
>+  }
>+
>+  void PrintResults() const {
>+    PRINT("] ");
>+    for (auto it = latencies_.begin(); it != latencies_.end(); ++it) {
>+      PRINT("%d ", *it);
>+    }
>+    PRINT("\n");
>+    PRINT("%s[min, max, avg]=[%d, %d, %d] ms\n", kTag, min_latency(),
>+          max_latency(), average_latency());
>+  }
>+
>+  int IndexToMilliseconds(double index) const {
>+    return static_cast<int>(10.0 * (index / frames_per_buffer_) + 0.5);
>+  }
>+
>+ private:
>+  const size_t frames_per_buffer_;
>+  const size_t bytes_per_buffer_;
>+  size_t play_count_;
>+  size_t rec_count_;
>+  int64_t pulse_time_;
>+  std::vector<int> latencies_;
>+};
>+
>+// Mocks the AudioTransport object and proxies actions for the two callbacks
>+// (RecordedDataIsAvailable and NeedMorePlayData) to different implementations
>+// of AudioStreamInterface.
>+class MockAudioTransportAndroid : public test::MockAudioTransport {
>+ public:
>+  explicit MockAudioTransportAndroid(int type)
>+      : num_callbacks_(0),
>+        type_(type),
>+        play_count_(0),
>+        rec_count_(0),
>+        audio_stream_(nullptr) {}
>+
>+  virtual ~MockAudioTransportAndroid() {}
>+
>+  // Set default actions of the mock object. We are delegating to fake
>+  // implementations (of AudioStreamInterface) here.
>+  void HandleCallbacks(EventWrapper* test_is_done,
>+                       AudioStreamInterface* audio_stream,
>+                       int num_callbacks) {
>+    test_is_done_ = test_is_done;
>+    audio_stream_ = audio_stream;
>+    num_callbacks_ = num_callbacks;
>+    if (play_mode()) {
>+      ON_CALL(*this, NeedMorePlayData(_, _, _, _, _, _, _, _))
>+          .WillByDefault(
>+              Invoke(this, &MockAudioTransportAndroid::RealNeedMorePlayData));
>+    }
>+    if (rec_mode()) {
>+      ON_CALL(*this, RecordedDataIsAvailable(_, _, _, _, _, _, _, _, _, _))
>+          .WillByDefault(Invoke(
>+              this, &MockAudioTransportAndroid::RealRecordedDataIsAvailable));
>+    }
>+  }
>+
>+  int32_t RealRecordedDataIsAvailable(const void* audioSamples,
>+                                      const size_t nSamples,
>+                                      const size_t nBytesPerSample,
>+                                      const size_t nChannels,
>+                                      const uint32_t samplesPerSec,
>+                                      const uint32_t totalDelayMS,
>+                                      const int32_t clockDrift,
>+                                      const uint32_t currentMicLevel,
>+                                      const bool keyPressed,
>+                                      const uint32_t& newMicLevel) {
>+    EXPECT_TRUE(rec_mode()) << "No test is expecting these callbacks.";
>+    rec_count_++;
>+    // Process the recorded audio stream if an AudioStreamInterface
>+    // implementation exists.
>+    if (audio_stream_) {
>+      audio_stream_->Write(audioSamples, nSamples);
>+    }
>+    if (ReceivedEnoughCallbacks()) {
>+      test_is_done_->Set();
>+    }
>+    return 0;
>+  }
>+
>+  int32_t RealNeedMorePlayData(const size_t nSamples,
>+                               const size_t nBytesPerSample,
>+                               const size_t nChannels,
>+                               const uint32_t samplesPerSec,
>+                               void* audioSamples,
>+                               size_t& nSamplesOut,  // NOLINT
>+                               int64_t* elapsed_time_ms,
>+                               int64_t* ntp_time_ms) {
>+    EXPECT_TRUE(play_mode()) << "No test is expecting these callbacks.";
>+    play_count_++;
>+    nSamplesOut = nSamples;
>+    // Read (possibly processed) audio stream samples to be played out if an
>+    // AudioStreamInterface implementation exists.
>+    if (audio_stream_) {
>+      audio_stream_->Read(audioSamples, nSamples);
>+    }
>+    if (ReceivedEnoughCallbacks()) {
>+      test_is_done_->Set();
>+    }
>+    return 0;
>+  }
>+
>+  bool ReceivedEnoughCallbacks() {
>+    bool recording_done = false;
>+    if (rec_mode())
>+      recording_done = rec_count_ >= num_callbacks_;
>+    else
>+      recording_done = true;
>+
>+    bool playout_done = false;
>+    if (play_mode())
>+      playout_done = play_count_ >= num_callbacks_;
>+    else
>+      playout_done = true;
>+
>+    return recording_done && playout_done;
>+  }
>+
>+  bool play_mode() const { return type_ & kPlayout; }
>+  bool rec_mode() const { return type_ & kRecording; }
>+
>+ private:
>+  EventWrapper* test_is_done_;
>+  size_t num_callbacks_;
>+  int type_;
>+  size_t play_count_;
>+  size_t rec_count_;
>+  AudioStreamInterface* audio_stream_;
>+  std::unique_ptr<LatencyMeasuringAudioStream> latency_audio_stream_;
>+};
>+
>+// AudioDeviceTest test fixture.
>+class AudioDeviceTest : public ::testing::Test {
>+ protected:
>+  AudioDeviceTest() : test_is_done_(EventWrapper::Create()) {
>+    // One-time initialization of JVM and application context. Ensures that we
>+    // can do calls between C++ and Java. Initializes both Java and OpenSL ES
>+    // implementations.
>+    // Creates an audio device using a default audio layer.
>+    jni_ = AttachCurrentThreadIfNeeded();
>+    audio_device_ = CreateJavaAudioDeviceModule(jni_, context());
>+    EXPECT_NE(audio_device_.get(), nullptr);
>+    EXPECT_EQ(0, audio_device_->Init());
>+    audio_manager_ = GetAudioManager(jni_, context_javaref());
>+    UpdateParameters();
>+  }
>+  virtual ~AudioDeviceTest() { EXPECT_EQ(0, audio_device_->Terminate()); }
>+
>+  int total_delay_ms() const { return 10; }
>+
>+  jobject context() {
>+    return jni::NewGlobalRef(
>+        jni_, Java_ApplicationContextProvider_getApplicationContextForTest(jni_)
>+                  .obj());
>+  }
>+
>+  ScopedJavaLocalRef<jobject> context_javaref() {
>+    return ScopedJavaLocalRef<jobject>(
>+        Java_ApplicationContextProvider_getApplicationContextForTest(jni_));
>+  }
>+
>+  void UpdateParameters() {
>+    int sample_rate = GetDefaultSampleRate(jni_, audio_manager_);
>+    bool stereo_playout_is_available;
>+    bool stereo_record_is_available;
>+    audio_device_->StereoPlayoutIsAvailable(&stereo_playout_is_available);
>+    audio_device_->StereoRecordingIsAvailable(&stereo_record_is_available);
>+    GetAudioParameters(jni_, context_javaref(), audio_manager_, sample_rate,
>+                       stereo_playout_is_available, stereo_record_is_available,
>+                       &input_parameters_, &output_parameters_);
>+  }
>+
>+  void SetActiveAudioLayer(AudioDeviceModule::AudioLayer audio_layer) {
>+    audio_device_ = CreateAudioDevice(audio_layer);
>+    EXPECT_NE(audio_device_.get(), nullptr);
>+    EXPECT_EQ(0, audio_device_->Init());
>+    UpdateParameters();
>+  }
>+
>+  int playout_sample_rate() const { return output_parameters_.sample_rate(); }
>+  int record_sample_rate() const { return input_parameters_.sample_rate(); }
>+  size_t playout_channels() const { return output_parameters_.channels(); }
>+  size_t record_channels() const { return input_parameters_.channels(); }
>+  size_t playout_frames_per_10ms_buffer() const {
>+    return output_parameters_.frames_per_10ms_buffer();
>+  }
>+  size_t record_frames_per_10ms_buffer() const {
>+    return input_parameters_.frames_per_10ms_buffer();
>+  }
>+
>+  rtc::scoped_refptr<AudioDeviceModule> audio_device() const {
>+    return audio_device_;
>+  }
>+
>+  rtc::scoped_refptr<AudioDeviceModule> CreateAudioDevice(
>+      AudioDeviceModule::AudioLayer audio_layer) {
>+#if defined(AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO)
>+    if (audio_layer == AudioDeviceModule::kAndroidAAudioAudio) {
>+      return rtc::scoped_refptr<AudioDeviceModule>(
>+          CreateAAudioAudioDeviceModule(jni_, context()));
>+    }
>+#endif
>+    if (audio_layer == AudioDeviceModule::kAndroidJavaAudio) {
>+      return rtc::scoped_refptr<AudioDeviceModule>(
>+          CreateJavaAudioDeviceModule(jni_, context()));
>+    } else if (audio_layer == AudioDeviceModule::kAndroidOpenSLESAudio) {
>+      return rtc::scoped_refptr<AudioDeviceModule>(
>+          CreateOpenSLESAudioDeviceModule(jni_, context()));
>+    } else if (audio_layer ==
>+               AudioDeviceModule::kAndroidJavaInputAndOpenSLESOutputAudio) {
>+      return rtc::scoped_refptr<AudioDeviceModule>(
>+          CreateJavaInputAndOpenSLESOutputAudioDeviceModule(jni_, context()));
>+    } else {
>+      return nullptr;
>+    }
>+  }
>+
>+  // Returns file name relative to the resource root given a sample rate.
>+  std::string GetFileName(int sample_rate) {
>+    EXPECT_TRUE(sample_rate == 48000 || sample_rate == 44100);
>+    char fname[64];
>+    snprintf(fname, sizeof(fname), "audio_device/audio_short%d",
>+             sample_rate / 1000);
>+    std::string file_name(webrtc::test::ResourcePath(fname, "pcm"));
>+    EXPECT_TRUE(test::FileExists(file_name));
>+#ifdef ENABLE_PRINTF
>+    PRINT("file name: %s\n", file_name.c_str());
>+    const size_t bytes = test::GetFileSize(file_name);
>+    PRINT("file size: %" PRIuS " [bytes]\n", bytes);
>+    PRINT("file size: %" PRIuS " [samples]\n", bytes / kBytesPerSample);
>+    const int seconds =
>+        static_cast<int>(bytes / (sample_rate * kBytesPerSample));
>+    PRINT("file size: %d [secs]\n", seconds);
>+    PRINT("file size: %" PRIuS " [callbacks]\n",
>+          seconds * kNumCallbacksPerSecond);
>+#endif
>+    return file_name;
>+  }
>+
>+  AudioDeviceModule::AudioLayer GetActiveAudioLayer() const {
>+    AudioDeviceModule::AudioLayer audio_layer;
>+    EXPECT_EQ(0, audio_device()->ActiveAudioLayer(&audio_layer));
>+    return audio_layer;
>+  }
>+
>+  int TestDelayOnAudioLayer(
>+      const AudioDeviceModule::AudioLayer& layer_to_test) {
>+    rtc::scoped_refptr<AudioDeviceModule> audio_device;
>+    audio_device = CreateAudioDevice(layer_to_test);
>+    EXPECT_NE(audio_device.get(), nullptr);
>+    uint16_t playout_delay;
>+    EXPECT_EQ(0, audio_device->PlayoutDelay(&playout_delay));
>+    return playout_delay;
>+  }
>+
>+  AudioDeviceModule::AudioLayer TestActiveAudioLayer(
>+      const AudioDeviceModule::AudioLayer& layer_to_test) {
>+    rtc::scoped_refptr<AudioDeviceModule> audio_device;
>+    audio_device = CreateAudioDevice(layer_to_test);
>+    EXPECT_NE(audio_device.get(), nullptr);
>+    AudioDeviceModule::AudioLayer active;
>+    EXPECT_EQ(0, audio_device->ActiveAudioLayer(&active));
>+    return active;
>+  }
>+
>+  // One way to ensure that the engine object is valid is to create an
>+  // SL Engine interface since it exposes creation methods of all the OpenSL ES
>+  // object types and it is only supported on the engine object. This method
>+  // also verifies that the engine interface supports at least one interface.
>+  // Note that, the test below is not a full test of the SLEngineItf object
>+  // but only a simple sanity test to check that the global engine object is OK.
>+  void ValidateSLEngine(SLObjectItf engine_object) {
>+    EXPECT_NE(nullptr, engine_object);
>+    // Get the SL Engine interface which is exposed by the engine object.
>+    SLEngineItf engine;
>+    SLresult result =
>+        (*engine_object)->GetInterface(engine_object, SL_IID_ENGINE, &engine);
>+    EXPECT_EQ(result, SL_RESULT_SUCCESS) << "GetInterface() on engine failed";
>+    // Ensure that the SL Engine interface exposes at least one interface.
>+    SLuint32 object_id = SL_OBJECTID_ENGINE;
>+    SLuint32 num_supported_interfaces = 0;
>+    result = (*engine)->QueryNumSupportedInterfaces(engine, object_id,
>+                                                    &num_supported_interfaces);
>+    EXPECT_EQ(result, SL_RESULT_SUCCESS)
>+        << "QueryNumSupportedInterfaces() failed";
>+    EXPECT_GE(num_supported_interfaces, 1u);
>+  }
>+
>+  // Volume control is currently only supported for the Java output audio layer.
>+  // For OpenSL ES, the internal stream volume is always on max level and there
>+  // is no need for this test to set it to max.
>+  bool AudioLayerSupportsVolumeControl() const {
>+    return GetActiveAudioLayer() == AudioDeviceModule::kAndroidJavaAudio;
>+  }
>+
>+  void SetMaxPlayoutVolume() {
>+    if (!AudioLayerSupportsVolumeControl())
>+      return;
>+    uint32_t max_volume;
>+    EXPECT_EQ(0, audio_device()->MaxSpeakerVolume(&max_volume));
>+    EXPECT_EQ(0, audio_device()->SetSpeakerVolume(max_volume));
>+  }
>+
>+  void DisableBuiltInAECIfAvailable() {
>+    if (audio_device()->BuiltInAECIsAvailable()) {
>+      EXPECT_EQ(0, audio_device()->EnableBuiltInAEC(false));
>+    }
>+  }
>+
>+  void StartPlayout() {
>+    EXPECT_FALSE(audio_device()->PlayoutIsInitialized());
>+    EXPECT_FALSE(audio_device()->Playing());
>+    EXPECT_EQ(0, audio_device()->InitPlayout());
>+    EXPECT_TRUE(audio_device()->PlayoutIsInitialized());
>+    EXPECT_EQ(0, audio_device()->StartPlayout());
>+    EXPECT_TRUE(audio_device()->Playing());
>+  }
>+
>+  void StopPlayout() {
>+    EXPECT_EQ(0, audio_device()->StopPlayout());
>+    EXPECT_FALSE(audio_device()->Playing());
>+    EXPECT_FALSE(audio_device()->PlayoutIsInitialized());
>+  }
>+
>+  void StartRecording() {
>+    EXPECT_FALSE(audio_device()->RecordingIsInitialized());
>+    EXPECT_FALSE(audio_device()->Recording());
>+    EXPECT_EQ(0, audio_device()->InitRecording());
>+    EXPECT_TRUE(audio_device()->RecordingIsInitialized());
>+    EXPECT_EQ(0, audio_device()->StartRecording());
>+    EXPECT_TRUE(audio_device()->Recording());
>+  }
>+
>+  void StopRecording() {
>+    EXPECT_EQ(0, audio_device()->StopRecording());
>+    EXPECT_FALSE(audio_device()->Recording());
>+  }
>+
>+  int GetMaxSpeakerVolume() const {
>+    uint32_t max_volume(0);
>+    EXPECT_EQ(0, audio_device()->MaxSpeakerVolume(&max_volume));
>+    return max_volume;
>+  }
>+
>+  int GetMinSpeakerVolume() const {
>+    uint32_t min_volume(0);
>+    EXPECT_EQ(0, audio_device()->MinSpeakerVolume(&min_volume));
>+    return min_volume;
>+  }
>+
>+  int GetSpeakerVolume() const {
>+    uint32_t volume(0);
>+    EXPECT_EQ(0, audio_device()->SpeakerVolume(&volume));
>+    return volume;
>+  }
>+
>+  JNIEnv* jni_;
>+  std::unique_ptr<JavaParamRef<jobject>> context_;
>+  std::unique_ptr<EventWrapper> test_is_done_;
>+  rtc::scoped_refptr<AudioDeviceModule> audio_device_;
>+  ScopedJavaLocalRef<jobject> audio_manager_;
>+  AudioParameters output_parameters_;
>+  AudioParameters input_parameters_;
>+};
>+
>+TEST_F(AudioDeviceTest, ConstructDestruct) {
>+  // Using the test fixture to create and destruct the audio device module.
>+}
>+
>+// Verify that it is possible to explicitly create the two types of supported
>+// ADMs. These two tests overrides the default selection of native audio layer
>+// by ignoring if the device supports low-latency output or not.
>+TEST_F(AudioDeviceTest, CorrectAudioLayerIsUsedForCombinedJavaOpenSLCombo) {
>+  AudioDeviceModule::AudioLayer expected_layer =
>+      AudioDeviceModule::kAndroidJavaInputAndOpenSLESOutputAudio;
>+  AudioDeviceModule::AudioLayer active_layer =
>+      TestActiveAudioLayer(expected_layer);
>+  EXPECT_EQ(expected_layer, active_layer);
>+}
>+
>+TEST_F(AudioDeviceTest, CorrectAudioLayerIsUsedForJavaInBothDirections) {
>+  AudioDeviceModule::AudioLayer expected_layer =
>+      AudioDeviceModule::kAndroidJavaAudio;
>+  AudioDeviceModule::AudioLayer active_layer =
>+      TestActiveAudioLayer(expected_layer);
>+  EXPECT_EQ(expected_layer, active_layer);
>+}
>+
>+TEST_F(AudioDeviceTest, CorrectAudioLayerIsUsedForOpenSLInBothDirections) {
>+  AudioDeviceModule::AudioLayer expected_layer =
>+      AudioDeviceModule::kAndroidOpenSLESAudio;
>+  AudioDeviceModule::AudioLayer active_layer =
>+      TestActiveAudioLayer(expected_layer);
>+  EXPECT_EQ(expected_layer, active_layer);
>+}
>+
>+// TODO(bugs.webrtc.org/8914)
>+// TODO(phensman): Add test for AAudio/Java combination when this combination
>+// is supported.
>+#if !defined(AUDIO_DEVICE_INCLUDE_ANDROID_AAUDIO)
>+#define MAYBE_CorrectAudioLayerIsUsedForAAudioInBothDirections \
>+  DISABLED_CorrectAudioLayerIsUsedForAAudioInBothDirections
>+#else
>+#define MAYBE_CorrectAudioLayerIsUsedForAAudioInBothDirections \
>+  CorrectAudioLayerIsUsedForAAudioInBothDirections
>+#endif
>+TEST_F(AudioDeviceTest,
>+       MAYBE_CorrectAudioLayerIsUsedForAAudioInBothDirections) {
>+  AudioDeviceModule::AudioLayer expected_layer =
>+      AudioDeviceModule::kAndroidAAudioAudio;
>+  AudioDeviceModule::AudioLayer active_layer =
>+      TestActiveAudioLayer(expected_layer);
>+  EXPECT_EQ(expected_layer, active_layer);
>+}
>+
>+// The Android ADM supports two different delay reporting modes. One for the
>+// low-latency output path (in combination with OpenSL ES), and one for the
>+// high-latency output path (Java backends in both directions). These two tests
>+// verifies that the audio device reports correct delay estimate given the
>+// selected audio layer. Note that, this delay estimate will only be utilized
>+// if the HW AEC is disabled.
>+TEST_F(AudioDeviceTest, UsesCorrectDelayEstimateForHighLatencyOutputPath) {
>+  EXPECT_EQ(kHighLatencyModeDelayEstimateInMilliseconds,
>+            TestDelayOnAudioLayer(AudioDeviceModule::kAndroidJavaAudio));
>+}
>+
>+TEST_F(AudioDeviceTest, UsesCorrectDelayEstimateForLowLatencyOutputPath) {
>+  EXPECT_EQ(kLowLatencyModeDelayEstimateInMilliseconds,
>+            TestDelayOnAudioLayer(
>+                AudioDeviceModule::kAndroidJavaInputAndOpenSLESOutputAudio));
>+}
>+
>+TEST_F(AudioDeviceTest, InitTerminate) {
>+  // Initialization is part of the test fixture.
>+  EXPECT_TRUE(audio_device()->Initialized());
>+  EXPECT_EQ(0, audio_device()->Terminate());
>+  EXPECT_FALSE(audio_device()->Initialized());
>+}
>+
>+TEST_F(AudioDeviceTest, Devices) {
>+  // Device enumeration is not supported. Verify fixed values only.
>+  EXPECT_EQ(1, audio_device()->PlayoutDevices());
>+  EXPECT_EQ(1, audio_device()->RecordingDevices());
>+}
>+
>+TEST_F(AudioDeviceTest, IsAcousticEchoCancelerSupported) {
>+  PRINT("%sAcoustic Echo Canceler support: %s\n", kTag,
>+        audio_device()->BuiltInAECIsAvailable() ? "Yes" : "No");
>+}
>+
>+TEST_F(AudioDeviceTest, IsNoiseSuppressorSupported) {
>+  PRINT("%sNoise Suppressor support: %s\n", kTag,
>+        audio_device()->BuiltInNSIsAvailable() ? "Yes" : "No");
>+}
>+
>+// Verify that playout side is configured for mono by default.
>+TEST_F(AudioDeviceTest, UsesMonoPlayoutByDefault) {
>+  EXPECT_EQ(1u, output_parameters_.channels());
>+}
>+
>+// Verify that recording side is configured for mono by default.
>+TEST_F(AudioDeviceTest, UsesMonoRecordingByDefault) {
>+  EXPECT_EQ(1u, input_parameters_.channels());
>+}
>+
>+TEST_F(AudioDeviceTest, SpeakerVolumeShouldBeAvailable) {
>+  // The OpenSL ES output audio path does not support volume control.
>+  if (!AudioLayerSupportsVolumeControl())
>+    return;
>+  bool available;
>+  EXPECT_EQ(0, audio_device()->SpeakerVolumeIsAvailable(&available));
>+  EXPECT_TRUE(available);
>+}
>+
>+TEST_F(AudioDeviceTest, MaxSpeakerVolumeIsPositive) {
>+  // The OpenSL ES output audio path does not support volume control.
>+  if (!AudioLayerSupportsVolumeControl())
>+    return;
>+  StartPlayout();
>+  EXPECT_GT(GetMaxSpeakerVolume(), 0);
>+  StopPlayout();
>+}
>+
>+TEST_F(AudioDeviceTest, MinSpeakerVolumeIsZero) {
>+  // The OpenSL ES output audio path does not support volume control.
>+  if (!AudioLayerSupportsVolumeControl())
>+    return;
>+  EXPECT_EQ(GetMinSpeakerVolume(), 0);
>+}
>+
>+TEST_F(AudioDeviceTest, DefaultSpeakerVolumeIsWithinMinMax) {
>+  // The OpenSL ES output audio path does not support volume control.
>+  if (!AudioLayerSupportsVolumeControl())
>+    return;
>+  const int default_volume = GetSpeakerVolume();
>+  EXPECT_GE(default_volume, GetMinSpeakerVolume());
>+  EXPECT_LE(default_volume, GetMaxSpeakerVolume());
>+}
>+
>+TEST_F(AudioDeviceTest, SetSpeakerVolumeActuallySetsVolume) {
>+  // The OpenSL ES output audio path does not support volume control.
>+  if (!AudioLayerSupportsVolumeControl())
>+    return;
>+  const int default_volume = GetSpeakerVolume();
>+  const int max_volume = GetMaxSpeakerVolume();
>+  EXPECT_EQ(0, audio_device()->SetSpeakerVolume(max_volume));
>+  int new_volume = GetSpeakerVolume();
>+  EXPECT_EQ(new_volume, max_volume);
>+  EXPECT_EQ(0, audio_device()->SetSpeakerVolume(default_volume));
>+}
>+
>+// Tests that playout can be initiated, started and stopped. No audio callback
>+// is registered in this test.
>+TEST_F(AudioDeviceTest, StartStopPlayout) {
>+  StartPlayout();
>+  StopPlayout();
>+  StartPlayout();
>+  StopPlayout();
>+}
>+
>+// Tests that recording can be initiated, started and stopped. No audio callback
>+// is registered in this test.
>+TEST_F(AudioDeviceTest, StartStopRecording) {
>+  StartRecording();
>+  StopRecording();
>+  StartRecording();
>+  StopRecording();
>+}
>+
>+// Verify that calling StopPlayout() will leave us in an uninitialized state
>+// which will require a new call to InitPlayout(). This test does not call
>+// StartPlayout() while being uninitialized since doing so will hit a
>+// RTC_DCHECK and death tests are not supported on Android.
>+TEST_F(AudioDeviceTest, StopPlayoutRequiresInitToRestart) {
>+  EXPECT_EQ(0, audio_device()->InitPlayout());
>+  EXPECT_EQ(0, audio_device()->StartPlayout());
>+  EXPECT_EQ(0, audio_device()->StopPlayout());
>+  EXPECT_FALSE(audio_device()->PlayoutIsInitialized());
>+}
>+
>+// Verify that calling StopRecording() will leave us in an uninitialized state
>+// which will require a new call to InitRecording(). This test does not call
>+// StartRecording() while being uninitialized since doing so will hit a
>+// RTC_DCHECK and death tests are not supported on Android.
>+TEST_F(AudioDeviceTest, StopRecordingRequiresInitToRestart) {
>+  EXPECT_EQ(0, audio_device()->InitRecording());
>+  EXPECT_EQ(0, audio_device()->StartRecording());
>+  EXPECT_EQ(0, audio_device()->StopRecording());
>+  EXPECT_FALSE(audio_device()->RecordingIsInitialized());
>+}
>+
>+// Start playout and verify that the native audio layer starts asking for real
>+// audio samples to play out using the NeedMorePlayData callback.
>+TEST_F(AudioDeviceTest, StartPlayoutVerifyCallbacks) {
>+  MockAudioTransportAndroid mock(kPlayout);
>+  mock.HandleCallbacks(test_is_done_.get(), nullptr, kNumCallbacks);
>+  EXPECT_CALL(mock, NeedMorePlayData(playout_frames_per_10ms_buffer(),
>+                                     kBytesPerSample, playout_channels(),
>+                                     playout_sample_rate(), NotNull(), _, _, _))
>+      .Times(AtLeast(kNumCallbacks));
>+  EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
>+  StartPlayout();
>+  test_is_done_->Wait(kTestTimeOutInMilliseconds);
>+  StopPlayout();
>+}
>+
>+// Start recording and verify that the native audio layer starts feeding real
>+// audio samples via the RecordedDataIsAvailable callback.
>+TEST_F(AudioDeviceTest, StartRecordingVerifyCallbacks) {
>+  MockAudioTransportAndroid mock(kRecording);
>+  mock.HandleCallbacks(test_is_done_.get(), nullptr, kNumCallbacks);
>+  EXPECT_CALL(
>+      mock, RecordedDataIsAvailable(NotNull(), record_frames_per_10ms_buffer(),
>+                                    kBytesPerSample, record_channels(),
>+                                    record_sample_rate(), _, 0, 0, false, _))
>+      .Times(AtLeast(kNumCallbacks));
>+
>+  EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
>+  StartRecording();
>+  test_is_done_->Wait(kTestTimeOutInMilliseconds);
>+  StopRecording();
>+}
>+
>+// Start playout and recording (full-duplex audio) and verify that audio is
>+// active in both directions.
>+TEST_F(AudioDeviceTest, StartPlayoutAndRecordingVerifyCallbacks) {
>+  MockAudioTransportAndroid mock(kPlayout | kRecording);
>+  mock.HandleCallbacks(test_is_done_.get(), nullptr, kNumCallbacks);
>+  EXPECT_CALL(mock, NeedMorePlayData(playout_frames_per_10ms_buffer(),
>+                                     kBytesPerSample, playout_channels(),
>+                                     playout_sample_rate(), NotNull(), _, _, _))
>+      .Times(AtLeast(kNumCallbacks));
>+  EXPECT_CALL(
>+      mock, RecordedDataIsAvailable(NotNull(), record_frames_per_10ms_buffer(),
>+                                    kBytesPerSample, record_channels(),
>+                                    record_sample_rate(), _, 0, 0, false, _))
>+      .Times(AtLeast(kNumCallbacks));
>+  EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
>+  StartPlayout();
>+  StartRecording();
>+  test_is_done_->Wait(kTestTimeOutInMilliseconds);
>+  StopRecording();
>+  StopPlayout();
>+}
>+
>+// Start playout and read audio from an external PCM file when the audio layer
>+// asks for data to play out. Real audio is played out in this test but it does
>+// not contain any explicit verification that the audio quality is perfect.
>+TEST_F(AudioDeviceTest, RunPlayoutWithFileAsSource) {
>+  // TODO(henrika): extend test when mono output is supported.
>+  EXPECT_EQ(1u, playout_channels());
>+  NiceMock<MockAudioTransportAndroid> mock(kPlayout);
>+  const int num_callbacks = kFilePlayTimeInSec * kNumCallbacksPerSecond;
>+  std::string file_name = GetFileName(playout_sample_rate());
>+  std::unique_ptr<FileAudioStream> file_audio_stream(
>+      new FileAudioStream(num_callbacks, file_name, playout_sample_rate()));
>+  mock.HandleCallbacks(test_is_done_.get(), file_audio_stream.get(),
>+                       num_callbacks);
>+  // SetMaxPlayoutVolume();
>+  EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
>+  StartPlayout();
>+  test_is_done_->Wait(kTestTimeOutInMilliseconds);
>+  StopPlayout();
>+}
>+
>+// It should be possible to create an OpenSL engine object if OpenSL ES based
>+// audio is requested in any direction.
>+TEST_F(AudioDeviceTest, TestCreateOpenSLEngine) {
>+  // Verify that the global (singleton) OpenSL Engine can be acquired.
>+  OpenSLEngineManager engine_manager;
>+  SLObjectItf engine_object = engine_manager.GetOpenSLEngine();
>+  EXPECT_NE(nullptr, engine_object);
>+  // Perform a simple sanity check of the created engine object.
>+  ValidateSLEngine(engine_object);
>+}
>+
>+// The audio device module only suppors the same sample rate in both directions.
>+// In addition, in full-duplex low-latency mode (OpenSL ES), both input and
>+// output must use the same native buffer size to allow for usage of the fast
>+// audio track in Android.
>+TEST_F(AudioDeviceTest, VerifyAudioParameters) {
>+  EXPECT_EQ(output_parameters_.sample_rate(), input_parameters_.sample_rate());
>+  SetActiveAudioLayer(AudioDeviceModule::kAndroidOpenSLESAudio);
>+  EXPECT_EQ(output_parameters_.frames_per_buffer(),
>+            input_parameters_.frames_per_buffer());
>+}
>+
>+TEST_F(AudioDeviceTest, ShowAudioParameterInfo) {
>+  const bool low_latency_out = false;
>+  const bool low_latency_in = false;
>+  PRINT("PLAYOUT:\n");
>+  PRINT("%saudio layer: %s\n", kTag,
>+        low_latency_out ? "Low latency OpenSL" : "Java/JNI based AudioTrack");
>+  PRINT("%ssample rate: %d Hz\n", kTag, output_parameters_.sample_rate());
>+  PRINT("%schannels: %" PRIuS "\n", kTag, output_parameters_.channels());
>+  PRINT("%sframes per buffer: %" PRIuS " <=> %.2f ms\n", kTag,
>+        output_parameters_.frames_per_buffer(),
>+        output_parameters_.GetBufferSizeInMilliseconds());
>+  PRINT("RECORD: \n");
>+  PRINT("%saudio layer: %s\n", kTag,
>+        low_latency_in ? "Low latency OpenSL" : "Java/JNI based AudioRecord");
>+  PRINT("%ssample rate: %d Hz\n", kTag, input_parameters_.sample_rate());
>+  PRINT("%schannels: %" PRIuS "\n", kTag, input_parameters_.channels());
>+  PRINT("%sframes per buffer: %" PRIuS " <=> %.2f ms\n", kTag,
>+        input_parameters_.frames_per_buffer(),
>+        input_parameters_.GetBufferSizeInMilliseconds());
>+}
>+
>+// Add device-specific information to the test for logging purposes.
>+TEST_F(AudioDeviceTest, ShowDeviceInfo) {
>+  std::string model =
>+      JavaToNativeString(jni_, Java_BuildInfo_getDeviceModel(jni_));
>+  std::string brand = JavaToNativeString(jni_, Java_BuildInfo_getBrand(jni_));
>+  std::string manufacturer =
>+      JavaToNativeString(jni_, Java_BuildInfo_getDeviceManufacturer(jni_));
>+
>+  PRINT("%smodel: %s\n", kTag, model.c_str());
>+  PRINT("%sbrand: %s\n", kTag, brand.c_str());
>+  PRINT("%smanufacturer: %s\n", kTag, manufacturer.c_str());
>+}
>+
>+// Add Android build information to the test for logging purposes.
>+TEST_F(AudioDeviceTest, ShowBuildInfo) {
>+  std::string release =
>+      JavaToNativeString(jni_, Java_BuildInfo_getBuildRelease(jni_));
>+  std::string build_id =
>+      JavaToNativeString(jni_, Java_BuildInfo_getAndroidBuildId(jni_));
>+  std::string build_type =
>+      JavaToNativeString(jni_, Java_BuildInfo_getBuildType(jni_));
>+  int sdk = Java_BuildInfo_getSdkVersion(jni_);
>+
>+  PRINT("%sbuild release: %s\n", kTag, release.c_str());
>+  PRINT("%sbuild id: %s\n", kTag, build_id.c_str());
>+  PRINT("%sbuild type: %s\n", kTag, build_type.c_str());
>+  PRINT("%sSDK version: %d\n", kTag, sdk);
>+}
>+
>+// Basic test of the AudioParameters class using default construction where
>+// all members are set to zero.
>+TEST_F(AudioDeviceTest, AudioParametersWithDefaultConstruction) {
>+  AudioParameters params;
>+  EXPECT_FALSE(params.is_valid());
>+  EXPECT_EQ(0, params.sample_rate());
>+  EXPECT_EQ(0U, params.channels());
>+  EXPECT_EQ(0U, params.frames_per_buffer());
>+  EXPECT_EQ(0U, params.frames_per_10ms_buffer());
>+  EXPECT_EQ(0U, params.GetBytesPerFrame());
>+  EXPECT_EQ(0U, params.GetBytesPerBuffer());
>+  EXPECT_EQ(0U, params.GetBytesPer10msBuffer());
>+  EXPECT_EQ(0.0f, params.GetBufferSizeInMilliseconds());
>+}
>+
>+// Basic test of the AudioParameters class using non default construction.
>+TEST_F(AudioDeviceTest, AudioParametersWithNonDefaultConstruction) {
>+  const int kSampleRate = 48000;
>+  const size_t kChannels = 1;
>+  const size_t kFramesPerBuffer = 480;
>+  const size_t kFramesPer10msBuffer = 480;
>+  const size_t kBytesPerFrame = 2;
>+  const float kBufferSizeInMs = 10.0f;
>+  AudioParameters params(kSampleRate, kChannels, kFramesPerBuffer);
>+  EXPECT_TRUE(params.is_valid());
>+  EXPECT_EQ(kSampleRate, params.sample_rate());
>+  EXPECT_EQ(kChannels, params.channels());
>+  EXPECT_EQ(kFramesPerBuffer, params.frames_per_buffer());
>+  EXPECT_EQ(static_cast<size_t>(kSampleRate / 100),
>+            params.frames_per_10ms_buffer());
>+  EXPECT_EQ(kBytesPerFrame, params.GetBytesPerFrame());
>+  EXPECT_EQ(kBytesPerFrame * kFramesPerBuffer, params.GetBytesPerBuffer());
>+  EXPECT_EQ(kBytesPerFrame * kFramesPer10msBuffer,
>+            params.GetBytesPer10msBuffer());
>+  EXPECT_EQ(kBufferSizeInMs, params.GetBufferSizeInMilliseconds());
>+}
>+
>+// Start playout and recording and store recorded data in an intermediate FIFO
>+// buffer from which the playout side then reads its samples in the same order
>+// as they were stored. Under ideal circumstances, a callback sequence would
>+// look like: ...+-+-+-+-+-+-+-..., where '+' means 'packet recorded' and '-'
>+// means 'packet played'. Under such conditions, the FIFO would only contain
>+// one packet on average. However, under more realistic conditions, the size
>+// of the FIFO will vary more due to an unbalance between the two sides.
>+// This test tries to verify that the device maintains a balanced callback-
>+// sequence by running in loopback for kFullDuplexTimeInSec seconds while
>+// measuring the size (max and average) of the FIFO. The size of the FIFO is
>+// increased by the recording side and decreased by the playout side.
>+// TODO(henrika): tune the final test parameters after running tests on several
>+// different devices.
>+// Disabling this test on bots since it is difficult to come up with a robust
>+// test condition that all worked as intended. The main issue is that, when
>+// swarming is used, an initial latency can be built up when the both sides
>+// starts at different times. Hence, the test can fail even if audio works
>+// as intended. Keeping the test so it can be enabled manually.
>+// http://bugs.webrtc.org/7744
>+TEST_F(AudioDeviceTest, DISABLED_RunPlayoutAndRecordingInFullDuplex) {
>+  EXPECT_EQ(record_channels(), playout_channels());
>+  EXPECT_EQ(record_sample_rate(), playout_sample_rate());
>+  NiceMock<MockAudioTransportAndroid> mock(kPlayout | kRecording);
>+  std::unique_ptr<FifoAudioStream> fifo_audio_stream(
>+      new FifoAudioStream(playout_frames_per_10ms_buffer()));
>+  mock.HandleCallbacks(test_is_done_.get(), fifo_audio_stream.get(),
>+                       kFullDuplexTimeInSec * kNumCallbacksPerSecond);
>+  SetMaxPlayoutVolume();
>+  EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
>+  StartRecording();
>+  StartPlayout();
>+  test_is_done_->Wait(
>+      std::max(kTestTimeOutInMilliseconds, 1000 * kFullDuplexTimeInSec));
>+  StopPlayout();
>+  StopRecording();
>+
>+  // These thresholds are set rather high to accomodate differences in hardware
>+  // in several devices, so this test can be used in swarming.
>+  // See http://bugs.webrtc.org/6464
>+  EXPECT_LE(fifo_audio_stream->average_size(), 60u);
>+  EXPECT_LE(fifo_audio_stream->largest_size(), 70u);
>+}
>+
>+// Measures loopback latency and reports the min, max and average values for
>+// a full duplex audio session.
>+// The latency is measured like so:
>+// - Insert impulses periodically on the output side.
>+// - Detect the impulses on the input side.
>+// - Measure the time difference between the transmit time and receive time.
>+// - Store time differences in a vector and calculate min, max and average.
>+// This test requires a special hardware called Audio Loopback Dongle.
>+// See http://source.android.com/devices/audio/loopback.html for details.
>+TEST_F(AudioDeviceTest, DISABLED_MeasureLoopbackLatency) {
>+  EXPECT_EQ(record_channels(), playout_channels());
>+  EXPECT_EQ(record_sample_rate(), playout_sample_rate());
>+  NiceMock<MockAudioTransportAndroid> mock(kPlayout | kRecording);
>+  std::unique_ptr<LatencyMeasuringAudioStream> latency_audio_stream(
>+      new LatencyMeasuringAudioStream(playout_frames_per_10ms_buffer()));
>+  mock.HandleCallbacks(test_is_done_.get(), latency_audio_stream.get(),
>+                       kMeasureLatencyTimeInSec * kNumCallbacksPerSecond);
>+  EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
>+  SetMaxPlayoutVolume();
>+  DisableBuiltInAECIfAvailable();
>+  StartRecording();
>+  StartPlayout();
>+  test_is_done_->Wait(
>+      std::max(kTestTimeOutInMilliseconds, 1000 * kMeasureLatencyTimeInSec));
>+  StopPlayout();
>+  StopRecording();
>+  // Verify that the correct number of transmitted impulses are detected.
>+  EXPECT_EQ(latency_audio_stream->num_latency_values(),
>+            static_cast<size_t>(
>+                kImpulseFrequencyInHz * kMeasureLatencyTimeInSec - 1));
>+  latency_audio_stream->PrintResults();
>+}
>+
>+TEST(JavaAudioDeviceTest, TestRunningTwoAdmsSimultaneously) {
>+  JNIEnv* jni = AttachCurrentThreadIfNeeded();
>+  ScopedJavaLocalRef<jobject> context =
>+      Java_ApplicationContextProvider_getApplicationContextForTest(jni);
>+
>+  // Create and start the first ADM.
>+  rtc::scoped_refptr<AudioDeviceModule> adm_1 =
>+      CreateJavaAudioDeviceModule(jni, context.obj());
>+  EXPECT_EQ(0, adm_1->Init());
>+  EXPECT_EQ(0, adm_1->InitRecording());
>+  EXPECT_EQ(0, adm_1->StartRecording());
>+
>+  // Create and start a second ADM. Expect this to fail due to the microphone
>+  // already being in use.
>+  rtc::scoped_refptr<AudioDeviceModule> adm_2 =
>+      CreateJavaAudioDeviceModule(jni, context.obj());
>+  int32_t err = adm_2->Init();
>+  err |= adm_2->InitRecording();
>+  err |= adm_2->StartRecording();
>+  EXPECT_NE(0, err);
>+
>+  // Stop and terminate second adm.
>+  adm_2->StopRecording();
>+  adm_2->Terminate();
>+
>+  // Stop first ADM.
>+  EXPECT_EQ(0, adm_1->StopRecording());
>+  EXPECT_EQ(0, adm_1->Terminate());
>+}
>+
>+}  // namespace jni
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/codecs/DEPS b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/codecs/DEPS
>new file mode 100644
>index 00000000000..5c7289df4b0
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/codecs/DEPS
>@@ -0,0 +1,3 @@
>+include_rules = [
>+  "+media/base/mediaconstants.h",
>+]
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/codecs/wrapper_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/codecs/wrapper_unittest.cc
>new file mode 100644
>index 00000000000..453b7da2625
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/codecs/wrapper_unittest.cc
>@@ -0,0 +1,35 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "sdk/android/native_api/codecs/wrapper.h"
>+#include "media/base/mediaconstants.h"
>+#include "sdk/android/generated_native_unittests_jni/jni/CodecsWrapperTestHelper_jni.h"
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+namespace test {
>+namespace {
>+TEST(JavaCodecsWrapperTest, JavaToNativeVideoCodecInfo) {
>+  JNIEnv* env = AttachCurrentThreadIfNeeded();
>+  ScopedJavaLocalRef<jobject> j_video_codec_info =
>+      jni::Java_CodecsWrapperTestHelper_createTestVideoCodecInfo(env);
>+
>+  const SdpVideoFormat video_format =
>+      JavaToNativeVideoCodecInfo(env, j_video_codec_info.obj());
>+
>+  EXPECT_EQ(cricket::kH264CodecName, video_format.name);
>+  const auto it =
>+      video_format.parameters.find(cricket::kH264FmtpProfileLevelId);
>+  ASSERT_NE(it, video_format.parameters.end());
>+  EXPECT_EQ(cricket::kH264ProfileLevelConstrainedBaseline, it->second);
>+}
>+}  // namespace
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/java_types_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/java_types_unittest.cc
>new file mode 100644
>index 00000000000..bd769e6eed1
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/java_types_unittest.cc
>@@ -0,0 +1,35 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <memory>
>+#include <vector>
>+
>+#include "sdk/android/generated_native_unittests_jni/jni/JavaTypesTestHelper_jni.h"
>+#include "sdk/android/native_api/jni/java_types.h"
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+namespace test {
>+namespace {
>+TEST(JavaTypesTest, TestJavaToNativeStringMap) {
>+  JNIEnv* env = AttachCurrentThreadIfNeeded();
>+  ScopedJavaLocalRef<jobject> j_map =
>+      jni::Java_JavaTypesTestHelper_createTestStringMap(env);
>+
>+  std::map<std::string, std::string> output = JavaToNativeStringMap(env, j_map);
>+
>+  std::map<std::string, std::string> expected{
>+      {"one", "1"}, {"two", "2"}, {"three", "3"},
>+  };
>+  EXPECT_EQ(expected, output);
>+}
>+}  // namespace
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/org/webrtc/ApplicationContextProvider.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/org/webrtc/ApplicationContextProvider.java
>new file mode 100644
>index 00000000000..e10d34710d3
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/org/webrtc/ApplicationContextProvider.java
>@@ -0,0 +1,20 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc;
>+
>+import android.content.Context;
>+
>+public class ApplicationContextProvider {
>+  @CalledByNative
>+  public static Context getApplicationContextForTest() {
>+    return ContextUtils.getApplicationContext();
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/org/webrtc/BuildInfo.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/org/webrtc/BuildInfo.java
>new file mode 100644
>index 00000000000..0440ae42099
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/org/webrtc/BuildInfo.java
>@@ -0,0 +1,59 @@
>+/*
>+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc;
>+
>+import android.os.Build;
>+import org.webrtc.CalledByNative;
>+
>+public final class BuildInfo {
>+  public static String getDevice() {
>+    return Build.DEVICE;
>+  }
>+
>+  @CalledByNative
>+  public static String getDeviceModel() {
>+    return Build.MODEL;
>+  }
>+
>+  public static String getProduct() {
>+    return Build.PRODUCT;
>+  }
>+
>+  @CalledByNative
>+  public static String getBrand() {
>+    return Build.BRAND;
>+  }
>+
>+  @CalledByNative
>+  public static String getDeviceManufacturer() {
>+    return Build.MANUFACTURER;
>+  }
>+
>+  @CalledByNative
>+  public static String getAndroidBuildId() {
>+    return Build.ID;
>+  }
>+
>+  @CalledByNative
>+  public static String getBuildType() {
>+    return Build.TYPE;
>+  }
>+
>+  @CalledByNative
>+  public static String getBuildRelease() {
>+    return Build.VERSION.RELEASE;
>+  }
>+
>+  @CalledByNative
>+  public static int getSdkVersion() {
>+    return Build.VERSION.SDK_INT;
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/org/webrtc/CodecsWrapperTestHelper.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/org/webrtc/CodecsWrapperTestHelper.java
>new file mode 100644
>index 00000000000..02135748e88
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/org/webrtc/CodecsWrapperTestHelper.java
>@@ -0,0 +1,26 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc;
>+
>+import java.util.HashMap;
>+import java.util.Map;
>+
>+public class CodecsWrapperTestHelper {
>+  @CalledByNative
>+  public static VideoCodecInfo createTestVideoCodecInfo() {
>+    Map<String, String> params = new HashMap<String, String>();
>+    params.put(
>+        VideoCodecInfo.H264_FMTP_PROFILE_LEVEL_ID, VideoCodecInfo.H264_CONSTRAINED_BASELINE_3_1);
>+
>+    VideoCodecInfo codec_info = new VideoCodecInfo("H264", params);
>+    return codec_info;
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/org/webrtc/JavaTypesTestHelper.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/org/webrtc/JavaTypesTestHelper.java
>new file mode 100644
>index 00000000000..6695ef79af0
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/org/webrtc/JavaTypesTestHelper.java
>@@ -0,0 +1,25 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc;
>+
>+import java.util.HashMap;
>+import java.util.Map;
>+
>+public class JavaTypesTestHelper {
>+  @CalledByNative
>+  public static Map createTestStringMap() {
>+    Map<String, String> testMap = new HashMap<String, String>();
>+    testMap.put("one", "1");
>+    testMap.put("two", "2");
>+    testMap.put("three", "3");
>+    return testMap;
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/org/webrtc/JavaVideoSourceTestHelper.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/org/webrtc/JavaVideoSourceTestHelper.java
>new file mode 100644
>index 00000000000..bde34e95cf6
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/org/webrtc/JavaVideoSourceTestHelper.java
>@@ -0,0 +1,31 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc;
>+
>+import org.webrtc.VideoCapturer;
>+
>+public class JavaVideoSourceTestHelper {
>+  @CalledByNative
>+  public static void startCapture(CapturerObserver observer, boolean success) {
>+    observer.onCapturerStarted(success);
>+  }
>+
>+  @CalledByNative
>+  public static void stopCapture(CapturerObserver observer) {
>+    observer.onCapturerStopped();
>+  }
>+
>+  @CalledByNative
>+  public static void deliverFrame(int width, int height, int rotation, CapturerObserver observer) {
>+    observer.onFrameCaptured(
>+        new VideoFrame(JavaI420Buffer.allocate(width, height), rotation, 0 /* timestampNs= */));
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/org/webrtc/PeerConnectionFactoryInitializationHelper.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/org/webrtc/PeerConnectionFactoryInitializationHelper.java
>new file mode 100644
>index 00000000000..445a6733ea5
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/org/webrtc/PeerConnectionFactoryInitializationHelper.java
>@@ -0,0 +1,33 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc;
>+
>+import android.content.Context;
>+import org.webrtc.PeerConnectionFactory.InitializationOptions;
>+
>+public class PeerConnectionFactoryInitializationHelper {
>+  private static class MockLoader implements NativeLibraryLoader {
>+    @Override
>+    public boolean load(String name) {
>+      return true;
>+    }
>+  }
>+
>+  @CalledByNative
>+  public static void initializeFactoryForTests() {
>+    Context ctx = ContextUtils.getApplicationContext();
>+    InitializationOptions options = InitializationOptions.builder(ctx)
>+                                        .setNativeLibraryLoader(new MockLoader())
>+                                        .createInitializationOptions();
>+
>+    PeerConnectionFactory.initialize(options);
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/peerconnection/DEPS b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/peerconnection/DEPS
>new file mode 100644
>index 00000000000..3c47a815226
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/peerconnection/DEPS
>@@ -0,0 +1,5 @@
>+include_rules = [
>+  "+media/base",
>+  "+media/engine",
>+  "+modules/audio_processing/include/audio_processing.h",
>+]
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/peerconnection/peerconnectionfactory_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/peerconnection/peerconnectionfactory_unittest.cc
>new file mode 100644
>index 00000000000..747786090cd
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/peerconnection/peerconnectionfactory_unittest.cc
>@@ -0,0 +1,97 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#include "sdk/android/native_api/peerconnection/peerconnectionfactory.h"
>+
>+#include "absl/memory/memory.h"
>+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
>+#include "api/audio_codecs/builtin_audio_encoder_factory.h"
>+#include "media/base/mediaengine.h"
>+#include "media/engine/internaldecoderfactory.h"
>+#include "media/engine/internalencoderfactory.h"
>+#include "media/engine/webrtcmediaengine.h"
>+#include "modules/audio_processing/include/audio_processing.h"
>+#include "rtc_base/logging.h"
>+#include "sdk/android/generated_native_unittests_jni/jni/PeerConnectionFactoryInitializationHelper_jni.h"
>+#include "sdk/android/native_api/jni/jvm.h"
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+namespace test {
>+namespace {
>+
>+// Create native peer connection factory, that will be wrapped by java one
>+rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> CreateTestPCF(
>+    rtc::Thread* network_thread,
>+    rtc::Thread* worker_thread,
>+    rtc::Thread* signaling_thread) {
>+  // talk/ assumes pretty widely that the current Thread is ThreadManager'd, but
>+  // ThreadManager only WrapCurrentThread()s the thread where it is first
>+  // created.  Since the semantics around when auto-wrapping happens in
>+  // webrtc/rtc_base/ are convoluted, we simply wrap here to avoid having to
>+  // think about ramifications of auto-wrapping there.
>+  rtc::ThreadManager::Instance()->WrapCurrentThread();
>+
>+  std::unique_ptr<cricket::MediaEngineInterface> media_engine =
>+      cricket::WebRtcMediaEngineFactory::Create(
>+          nullptr /* adm */, webrtc::CreateBuiltinAudioEncoderFactory(),
>+          webrtc::CreateBuiltinAudioDecoderFactory(),
>+          absl::make_unique<webrtc::InternalEncoderFactory>(),
>+          absl::make_unique<webrtc::InternalDecoderFactory>(),
>+          nullptr /* audio_mixer */, webrtc::AudioProcessingBuilder().Create());
>+  RTC_LOG(LS_INFO) << "Media engine created: " << media_engine.get();
>+
>+  auto factory = CreateModularPeerConnectionFactory(
>+      network_thread, worker_thread, signaling_thread, std::move(media_engine),
>+      webrtc::CreateCallFactory(), webrtc::CreateRtcEventLogFactory());
>+  RTC_LOG(LS_INFO) << "PeerConnectionFactory created: " << factory;
>+  RTC_CHECK(factory) << "Failed to create the peer connection factory; "
>+                     << "WebRTC/libjingle init likely failed on this device";
>+
>+  return factory;
>+}
>+
>+TEST(PeerConnectionFactoryTest, NativeToJavaPeerConnectionFactory) {
>+  JNIEnv* jni = AttachCurrentThreadIfNeeded();
>+
>+  RTC_LOG(INFO) << "Initializing java peer connection factory.";
>+  jni::Java_PeerConnectionFactoryInitializationHelper_initializeFactoryForTests(
>+      jni);
>+  RTC_LOG(INFO) << "Java peer connection factory initialized.";
>+
>+  // Create threads.
>+  std::unique_ptr<rtc::Thread> network_thread =
>+      rtc::Thread::CreateWithSocketServer();
>+  network_thread->SetName("network_thread", nullptr);
>+  RTC_CHECK(network_thread->Start()) << "Failed to start thread";
>+
>+  std::unique_ptr<rtc::Thread> worker_thread = rtc::Thread::Create();
>+  worker_thread->SetName("worker_thread", nullptr);
>+  RTC_CHECK(worker_thread->Start()) << "Failed to start thread";
>+
>+  std::unique_ptr<rtc::Thread> signaling_thread = rtc::Thread::Create();
>+  signaling_thread->SetName("signaling_thread", NULL);
>+  RTC_CHECK(signaling_thread->Start()) << "Failed to start thread";
>+
>+  rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> factory =
>+      CreateTestPCF(network_thread.get(), worker_thread.get(),
>+                    signaling_thread.get());
>+
>+  jobject java_factory = NativeToJavaPeerConnectionFactory(
>+      jni, factory, std::move(network_thread), std::move(worker_thread),
>+      std::move(signaling_thread), nullptr /* network_monitor_factory */);
>+
>+  RTC_LOG(INFO) << java_factory;
>+
>+  EXPECT_NE(java_factory, nullptr);
>+}
>+
>+}  // namespace
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/test_jni_onload.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/test_jni_onload.cc
>new file mode 100644
>index 00000000000..dafe49c4743
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/test_jni_onload.cc
>@@ -0,0 +1,23 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <jni.h>
>+#undef JNIEXPORT
>+#define JNIEXPORT __attribute__((visibility("default")))
>+
>+#include "rtc_base/checks.h"
>+#include "sdk/android/native_api/base/init.h"
>+#include "sdk/android/native_api/jni/java_types.h"
>+
>+// This is called by the VM when the shared library is first loaded.
>+JNIEXPORT jint JNI_OnLoad(JavaVM* vm, void* reserved) {
>+  webrtc::InitAndroid(vm);
>+  return JNI_VERSION_1_4;
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/video/videosource_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/video/videosource_unittest.cc
>new file mode 100644
>index 00000000000..b58ed1a45a5
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/native_unittests/video/videosource_unittest.cc
>@@ -0,0 +1,140 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <vector>
>+
>+#include "api/video/video_sink_interface.h"
>+#include "sdk/android/generated_native_unittests_jni/jni/JavaVideoSourceTestHelper_jni.h"
>+#include "sdk/android/native_api/video/videosource.h"
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+namespace {
>+class TestVideoSink : public rtc::VideoSinkInterface<VideoFrame> {
>+ public:
>+  void OnFrame(const VideoFrame& frame) { frames_.push_back(frame); }
>+
>+  std::vector<VideoFrame> GetFrames() {
>+    std::vector<VideoFrame> temp = frames_;
>+    frames_.clear();
>+    return temp;
>+  }
>+
>+ private:
>+  std::vector<VideoFrame> frames_;
>+};
>+}  // namespace
>+
>+TEST(JavaVideoSourceTest, CreateJavaVideoSource) {
>+  JNIEnv* env = AttachCurrentThreadIfNeeded();
>+  // Wrap test thread so it can be used as the signaling thread.
>+  rtc::ThreadManager::Instance()->WrapCurrentThread();
>+
>+  rtc::scoped_refptr<JavaVideoTrackSourceInterface> video_track_source =
>+      CreateJavaVideoSource(env,
>+                            rtc::ThreadManager::Instance()->CurrentThread(),
>+                            false /* is_screencast */);
>+
>+  ASSERT_NE(nullptr, video_track_source);
>+  EXPECT_NE(nullptr,
>+            video_track_source->GetJavaVideoCapturerObserver(env).obj());
>+}
>+
>+TEST(JavaVideoSourceTest, OnFrameCapturedFrameIsDeliveredToSink) {
>+  TestVideoSink test_video_sink;
>+
>+  JNIEnv* env = AttachCurrentThreadIfNeeded();
>+  // Wrap test thread so it can be used as the signaling thread.
>+  rtc::ThreadManager::Instance()->WrapCurrentThread();
>+
>+  rtc::scoped_refptr<JavaVideoTrackSourceInterface> video_track_source =
>+      CreateJavaVideoSource(env,
>+                            rtc::ThreadManager::Instance()->CurrentThread(),
>+                            false /* is_screencast */);
>+  video_track_source->AddOrUpdateSink(&test_video_sink, rtc::VideoSinkWants());
>+
>+  jni::Java_JavaVideoSourceTestHelper_startCapture(
>+      env, video_track_source->GetJavaVideoCapturerObserver(env),
>+      true /* success */);
>+  const int width = 20;
>+  const int height = 32;
>+  const int rotation = 180;
>+  jni::Java_JavaVideoSourceTestHelper_deliverFrame(
>+      env, width, height, rotation,
>+      video_track_source->GetJavaVideoCapturerObserver(env));
>+
>+  std::vector<VideoFrame> frames = test_video_sink.GetFrames();
>+  ASSERT_EQ(1u, frames.size());
>+  webrtc::VideoFrame frame = frames[0];
>+  EXPECT_EQ(width, frame.width());
>+  EXPECT_EQ(height, frame.height());
>+  EXPECT_EQ(rotation, frame.rotation());
>+}
>+
>+TEST(JavaVideoSourceTest, CapturerStartedSuccessStateBecomesLive) {
>+  JNIEnv* env = AttachCurrentThreadIfNeeded();
>+  // Wrap test thread so it can be used as the signaling thread.
>+  rtc::ThreadManager::Instance()->WrapCurrentThread();
>+
>+  rtc::scoped_refptr<JavaVideoTrackSourceInterface> video_track_source =
>+      CreateJavaVideoSource(env,
>+                            rtc::ThreadManager::Instance()->CurrentThread(),
>+                            false /* is_screencast */);
>+
>+  jni::Java_JavaVideoSourceTestHelper_startCapture(
>+      env, video_track_source->GetJavaVideoCapturerObserver(env),
>+      true /* success */);
>+
>+  EXPECT_EQ(VideoTrackSourceInterface::SourceState::kLive,
>+            video_track_source->state());
>+}
>+
>+TEST(JavaVideoSourceTest, CapturerStartedFailureStateBecomesEnded) {
>+  JNIEnv* env = AttachCurrentThreadIfNeeded();
>+  // Wrap test thread so it can be used as the signaling thread.
>+  rtc::ThreadManager::Instance()->WrapCurrentThread();
>+
>+  rtc::scoped_refptr<JavaVideoTrackSourceInterface> video_track_source =
>+      CreateJavaVideoSource(env,
>+                            rtc::ThreadManager::Instance()->CurrentThread(),
>+                            false /* is_screencast */);
>+
>+  jni::Java_JavaVideoSourceTestHelper_startCapture(
>+      env, video_track_source->GetJavaVideoCapturerObserver(env),
>+      false /* success */);
>+
>+  EXPECT_EQ(VideoTrackSourceInterface::SourceState::kEnded,
>+            video_track_source->state());
>+}
>+
>+TEST(JavaVideoSourceTest, CapturerStoppedStateBecomesEnded) {
>+  JNIEnv* env = AttachCurrentThreadIfNeeded();
>+  // Wrap test thread so it can be used as the signaling thread.
>+  rtc::ThreadManager::Instance()->WrapCurrentThread();
>+
>+  rtc::scoped_refptr<JavaVideoTrackSourceInterface> video_track_source =
>+      CreateJavaVideoSource(env,
>+                            rtc::ThreadManager::Instance()->CurrentThread(),
>+                            false /* is_screencast */);
>+
>+  jni::Java_JavaVideoSourceTestHelper_startCapture(
>+      env, video_track_source->GetJavaVideoCapturerObserver(env),
>+      true /* success */);
>+  jni::Java_JavaVideoSourceTestHelper_stopCapture(
>+      env, video_track_source->GetJavaVideoCapturerObserver(env));
>+
>+  EXPECT_EQ(VideoTrackSourceInterface::SourceState::kEnded,
>+            video_track_source->state());
>+}
>+
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/AndroidVideoTrackSourceObserver.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/AndroidVideoTrackSourceObserver.java
>deleted file mode 100644
>index 123204791f6..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/AndroidVideoTrackSourceObserver.java
>+++ /dev/null
>@@ -1,60 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-package org.webrtc;
>-
>-/** An implementation of CapturerObserver that forwards all calls from Java to the C layer. */
>-class AndroidVideoTrackSourceObserver implements VideoCapturer.CapturerObserver {
>-  // Pointer to VideoTrackSourceProxy proxying AndroidVideoTrackSource.
>-  private final long nativeSource;
>-
>-  public AndroidVideoTrackSourceObserver(long nativeSource) {
>-    this.nativeSource = nativeSource;
>-  }
>-
>-  @Override
>-  public void onCapturerStarted(boolean success) {
>-    nativeCapturerStarted(nativeSource, success);
>-  }
>-
>-  @Override
>-  public void onCapturerStopped() {
>-    nativeCapturerStopped(nativeSource);
>-  }
>-
>-  @Override
>-  public void onByteBufferFrameCaptured(
>-      byte[] data, int width, int height, int rotation, long timeStamp) {
>-    nativeOnByteBufferFrameCaptured(
>-        nativeSource, data, data.length, width, height, rotation, timeStamp);
>-  }
>-
>-  @Override
>-  public void onTextureFrameCaptured(int width, int height, int oesTextureId,
>-      float[] transformMatrix, int rotation, long timestamp) {
>-    nativeOnTextureFrameCaptured(
>-        nativeSource, width, height, oesTextureId, transformMatrix, rotation, timestamp);
>-  }
>-
>-  @Override
>-  public void onFrameCaptured(VideoFrame frame) {
>-    nativeOnFrameCaptured(nativeSource, frame.getBuffer().getWidth(), frame.getBuffer().getHeight(),
>-        frame.getRotation(), frame.getTimestampNs(), frame.getBuffer());
>-  }
>-
>-  private native void nativeCapturerStarted(long nativeSource, boolean success);
>-  private native void nativeCapturerStopped(long nativeSource);
>-  private native void nativeOnByteBufferFrameCaptured(long nativeSource, byte[] data, int length,
>-      int width, int height, int rotation, long timeStamp);
>-  private native void nativeOnTextureFrameCaptured(long nativeSource, int width, int height,
>-      int oesTextureId, float[] transformMatrix, int rotation, long timestamp);
>-  private native void nativeOnFrameCaptured(long nativeSource, int width, int height, int rotation,
>-      long timestampNs, VideoFrame.Buffer frame);
>-}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/BaseBitrateAdjuster.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/BaseBitrateAdjuster.java
>index ca9ebb52a84..19cf5ea38a4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/BaseBitrateAdjuster.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/BaseBitrateAdjuster.java
>@@ -32,7 +32,7 @@ class BaseBitrateAdjuster implements BitrateAdjuster {
>   }
> 
>   @Override
>-  public int getAdjustedFramerate() {
>+  public int getCodecConfigFramerate() {
>     return targetFps;
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/BitrateAdjuster.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/BitrateAdjuster.java
>index 044a43a1ebd..854ef067e57 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/BitrateAdjuster.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/BitrateAdjuster.java
>@@ -26,6 +26,6 @@ interface BitrateAdjuster {
>   /** Gets the current bitrate. */
>   int getAdjustedBitrateBps();
> 
>-  /** Gets the current framerate. */
>-  int getAdjustedFramerate();
>+  /** Gets the framerate for initial codec configuration. */
>+  int getCodecConfigFramerate();
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/CalledByNative.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/CalledByNative.java
>index 5d6c41f1a25..9b410ceaefb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/CalledByNative.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/CalledByNative.java
>@@ -21,7 +21,7 @@ import java.lang.annotation.Target;
>  */
> @Target({ElementType.CONSTRUCTOR, ElementType.METHOD})
> @Retention(RetentionPolicy.CLASS)
>-@interface CalledByNative {
>+public @interface CalledByNative {
>   /*
>    *  If present, tells which inner class the method belongs to.
>    */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/CalledByNativeUnchecked.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/CalledByNativeUnchecked.java
>new file mode 100644
>index 00000000000..8a00a7fadb8
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/CalledByNativeUnchecked.java
>@@ -0,0 +1,33 @@
>+/*
>+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc;
>+
>+import java.lang.annotation.ElementType;
>+import java.lang.annotation.Retention;
>+import java.lang.annotation.RetentionPolicy;
>+import java.lang.annotation.Target;
>+
>+/**
>+ *  @CalledByNativeUnchecked is used to generate JNI bindings that do not check for exceptions.
>+ *  It only makes sense to use this annotation on methods that declare a throws... spec.
>+ *  However, note that the exception received native side maybe an 'unchecked' (RuntimeExpception)
>+ *  such as NullPointerException, so the native code should differentiate these cases.
>+ *  Usage of this should be very rare; where possible handle exceptions in the Java side and use a
>+ *  return value to indicate success / failure.
>+ */
>+@Target(ElementType.METHOD)
>+@Retention(RetentionPolicy.CLASS)
>+public @interface CalledByNativeUnchecked {
>+  /*
>+   *  If present, tells which inner class the method belongs to.
>+   */
>+  public String value() default "";
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/Camera1Session.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/Camera1Session.java
>index 61a51c940ae..e4c68c48ca3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/Camera1Session.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/Camera1Session.java
>@@ -11,16 +11,16 @@
> package org.webrtc;
> 
> import android.content.Context;
>-import android.media.MediaRecorder;
> import android.os.Handler;
> import android.os.SystemClock;
> import android.view.Surface;
>-import android.view.WindowManager;
> import java.io.IOException;
> import java.nio.ByteBuffer;
> import java.util.List;
> import java.util.concurrent.TimeUnit;
>+import javax.annotation.Nullable;
> import org.webrtc.CameraEnumerationAndroid.CaptureFormat;
>+import android.graphics.Matrix;
> 
> @SuppressWarnings("deprecation")
> class Camera1Session implements CameraSession {
>@@ -36,8 +36,6 @@ class Camera1Session implements CameraSession {
> 
>   private static enum SessionState { RUNNING, STOPPED }
> 
>-  private final boolean videoFrameEmitTrialEnabled;
>-
>   private final Handler cameraThreadHandler;
>   private final Events events;
>   private final boolean captureToTexture;
>@@ -53,10 +51,12 @@ class Camera1Session implements CameraSession {
>   private SessionState state;
>   private boolean firstFrameReported = false;
> 
>+  // TODO(titovartem) make correct fix during webrtc:9175
>+  @SuppressWarnings("ByteBufferBackingArray")
>   public static void create(final CreateSessionCallback callback, final Events events,
>       final boolean captureToTexture, final Context applicationContext,
>-      final SurfaceTextureHelper surfaceTextureHelper, final MediaRecorder mediaRecorder,
>-      final int cameraId, final int width, final int height, final int framerate) {
>+      final SurfaceTextureHelper surfaceTextureHelper, final int cameraId, final int width,
>+      final int height, final int framerate) {
>     final long constructionTimeNs = System.nanoTime();
>     Logging.d(TAG, "Open camera " + cameraId);
>     events.onCameraOpening();
>@@ -69,9 +69,15 @@ class Camera1Session implements CameraSession {
>       return;
>     }
> 
>+    if (camera == null) {
>+      callback.onFailure(FailureType.ERROR,
>+          "android.hardware.Camera.open returned null for camera id = " + cameraId);
>+      return;
>+    }
>+
>     try {
>       camera.setPreviewTexture(surfaceTextureHelper.getSurfaceTexture());
>-    } catch (IOException e) {
>+    } catch (IOException | RuntimeException e) {
>       camera.release();
>       callback.onFailure(FailureType.ERROR, e.getMessage());
>       return;
>@@ -80,12 +86,17 @@ class Camera1Session implements CameraSession {
>     final android.hardware.Camera.CameraInfo info = new android.hardware.Camera.CameraInfo();
>     android.hardware.Camera.getCameraInfo(cameraId, info);
> 
>-    final android.hardware.Camera.Parameters parameters = camera.getParameters();
>-    final CaptureFormat captureFormat =
>-        findClosestCaptureFormat(parameters, width, height, framerate);
>-    final Size pictureSize = findClosestPictureSize(parameters, width, height);
>-
>-    updateCameraParameters(camera, parameters, captureFormat, pictureSize, captureToTexture);
>+    final CaptureFormat captureFormat;
>+    try {
>+      final android.hardware.Camera.Parameters parameters = camera.getParameters();
>+      captureFormat = findClosestCaptureFormat(parameters, width, height, framerate);
>+      final Size pictureSize = findClosestPictureSize(parameters, width, height);
>+      updateCameraParameters(camera, parameters, captureFormat, pictureSize, captureToTexture);
>+    } catch (RuntimeException e) {
>+      camera.release();
>+      callback.onFailure(FailureType.ERROR, e.getMessage());
>+      return;
>+    }
> 
>     if (!captureToTexture) {
>       final int frameSize = captureFormat.frameSize();
>@@ -98,9 +109,8 @@ class Camera1Session implements CameraSession {
>     // Calculate orientation manually and send it as CVO insted.
>     camera.setDisplayOrientation(0 /* degrees */);
> 
>-    callback.onDone(
>-        new Camera1Session(events, captureToTexture, applicationContext, surfaceTextureHelper,
>-            mediaRecorder, cameraId, camera, info, captureFormat, constructionTimeNs));
>+    callback.onDone(new Camera1Session(events, captureToTexture, applicationContext,
>+        surfaceTextureHelper, cameraId, camera, info, captureFormat, constructionTimeNs));
>   }
> 
>   private static void updateCameraParameters(android.hardware.Camera camera,
>@@ -148,13 +158,10 @@ class Camera1Session implements CameraSession {
>   }
> 
>   private Camera1Session(Events events, boolean captureToTexture, Context applicationContext,
>-      SurfaceTextureHelper surfaceTextureHelper, MediaRecorder mediaRecorder, int cameraId,
>-      android.hardware.Camera camera, android.hardware.Camera.CameraInfo info,
>-      CaptureFormat captureFormat, long constructionTimeNs) {
>+      SurfaceTextureHelper surfaceTextureHelper, int cameraId, android.hardware.Camera camera,
>+      android.hardware.Camera.CameraInfo info, CaptureFormat captureFormat,
>+      long constructionTimeNs) {
>     Logging.d(TAG, "Create new camera1 session on camera " + cameraId);
>-    videoFrameEmitTrialEnabled =
>-        PeerConnectionFactory.fieldTrialsFindFullName(PeerConnectionFactory.VIDEO_FRAME_EMIT_TRIAL)
>-            .equals(PeerConnectionFactory.TRIAL_ENABLED);
> 
>     this.cameraThreadHandler = new Handler();
>     this.events = events;
>@@ -167,12 +174,9 @@ class Camera1Session implements CameraSession {
>     this.captureFormat = captureFormat;
>     this.constructionTimeNs = constructionTimeNs;
> 
>-    startCapturing();
>+    surfaceTextureHelper.setTextureSize(captureFormat.width, captureFormat.height);
> 
>-    if (mediaRecorder != null) {
>-      camera.unlock();
>-      mediaRecorder.setCamera(camera);
>-    }
>+    startCapturing();
>   }
> 
>   @Override
>@@ -245,44 +249,31 @@ class Camera1Session implements CameraSession {
>   }
> 
>   private void listenForTextureFrames() {
>-    surfaceTextureHelper.startListening(new SurfaceTextureHelper.OnTextureFrameAvailableListener() {
>-      @Override
>-      public void onTextureFrameAvailable(
>-          int oesTextureId, float[] transformMatrix, long timestampNs) {
>-        checkIsOnCameraThread();
>-
>-        if (state != SessionState.RUNNING) {
>-          Logging.d(TAG, "Texture frame captured but camera is no longer running.");
>-          surfaceTextureHelper.returnTextureFrame();
>-          return;
>-        }
>+    surfaceTextureHelper.startListening((VideoFrame frame) -> {
>+      checkIsOnCameraThread();
> 
>-        if (!firstFrameReported) {
>-          final int startTimeMs =
>-              (int) TimeUnit.NANOSECONDS.toMillis(System.nanoTime() - constructionTimeNs);
>-          camera1StartTimeMsHistogram.addSample(startTimeMs);
>-          firstFrameReported = true;
>-        }
>+      if (state != SessionState.RUNNING) {
>+        Logging.d(TAG, "Texture frame captured but camera is no longer running.");
>+        return;
>+      }
> 
>-        int rotation = getFrameOrientation();
>-        if (info.facing == android.hardware.Camera.CameraInfo.CAMERA_FACING_FRONT) {
>-          // Undo the mirror that the OS "helps" us with.
>-          // http://developer.android.com/reference/android/hardware/Camera.html#setDisplayOrientation(int)
>-          transformMatrix = RendererCommon.multiplyMatrices(
>-              transformMatrix, RendererCommon.horizontalFlipMatrix());
>-        }
>-        if (videoFrameEmitTrialEnabled) {
>-          final VideoFrame.Buffer buffer =
>-              surfaceTextureHelper.createTextureBuffer(captureFormat.width, captureFormat.height,
>-                  RendererCommon.convertMatrixToAndroidGraphicsMatrix(transformMatrix));
>-          final VideoFrame frame = new VideoFrame(buffer, rotation, timestampNs);
>-          events.onFrameCaptured(Camera1Session.this, frame);
>-          frame.release();
>-        } else {
>-          events.onTextureFrameCaptured(Camera1Session.this, captureFormat.width,
>-              captureFormat.height, oesTextureId, transformMatrix, rotation, timestampNs);
>-        }
>+      if (!firstFrameReported) {
>+        final int startTimeMs =
>+            (int) TimeUnit.NANOSECONDS.toMillis(System.nanoTime() - constructionTimeNs);
>+        camera1StartTimeMsHistogram.addSample(startTimeMs);
>+        firstFrameReported = true;
>       }
>+
>+      // Undo the mirror that the OS "helps" us with.
>+      // http://developer.android.com/reference/android/hardware/Camera.html#setDisplayOrientation(int)
>+      final VideoFrame modifiedFrame = new VideoFrame(
>+          CameraSession.createTextureBufferWithModifiedTransformMatrix(
>+              (TextureBufferImpl) frame.getBuffer(),
>+              /* mirror= */ info.facing == android.hardware.Camera.CameraInfo.CAMERA_FACING_FRONT,
>+              /* rotation= */ 0),
>+          /* rotation= */ getFrameOrientation(), frame.getTimestampNs());
>+      events.onFrameCaptured(Camera1Session.this, modifiedFrame);
>+      modifiedFrame.release();
>     });
>   }
> 
>@@ -311,50 +302,21 @@ class Camera1Session implements CameraSession {
>           firstFrameReported = true;
>         }
> 
>-        if (videoFrameEmitTrialEnabled) {
>-          VideoFrame.Buffer frameBuffer = new NV21Buffer(data, captureFormat.width,
>-              captureFormat.height, () -> cameraThreadHandler.post(() -> {
>-                if (state == SessionState.RUNNING) {
>-                  camera.addCallbackBuffer(data);
>-                }
>-              }));
>-          final VideoFrame frame =
>-              new VideoFrame(frameBuffer, getFrameOrientation(), captureTimeNs);
>-          events.onFrameCaptured(Camera1Session.this, frame);
>-          frame.release();
>-        } else {
>-          events.onByteBufferFrameCaptured(Camera1Session.this, data, captureFormat.width,
>-              captureFormat.height, getFrameOrientation(), captureTimeNs);
>-          camera.addCallbackBuffer(data);
>-        }
>+        VideoFrame.Buffer frameBuffer = new NV21Buffer(
>+            data, captureFormat.width, captureFormat.height, () -> cameraThreadHandler.post(() -> {
>+              if (state == SessionState.RUNNING) {
>+                camera.addCallbackBuffer(data);
>+              }
>+            }));
>+        final VideoFrame frame = new VideoFrame(frameBuffer, getFrameOrientation(), captureTimeNs);
>+        events.onFrameCaptured(Camera1Session.this, frame);
>+        frame.release();
>       }
>     });
>   }
> 
>-  private int getDeviceOrientation() {
>-    int orientation = 0;
>-
>-    WindowManager wm = (WindowManager) applicationContext.getSystemService(Context.WINDOW_SERVICE);
>-    switch (wm.getDefaultDisplay().getRotation()) {
>-      case Surface.ROTATION_90:
>-        orientation = 90;
>-        break;
>-      case Surface.ROTATION_180:
>-        orientation = 180;
>-        break;
>-      case Surface.ROTATION_270:
>-        orientation = 270;
>-        break;
>-      case Surface.ROTATION_0:
>-      default:
>-        orientation = 0;
>-        break;
>-    }
>-    return orientation;
>-  }
>-
>   private int getFrameOrientation() {
>-    int rotation = getDeviceOrientation();
>+    int rotation = CameraSession.getDeviceOrientation(applicationContext);
>     if (info.facing == android.hardware.Camera.CameraInfo.CAMERA_FACING_BACK) {
>       rotation = 360 - rotation;
>     }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/Camera2Session.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/Camera2Session.java
>index b6420fcb1e0..fd34ce7ef69 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/Camera2Session.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/Camera2Session.java
>@@ -22,12 +22,11 @@ import android.hardware.camera2.CameraManager;
> import android.hardware.camera2.CameraMetadata;
> import android.hardware.camera2.CaptureFailure;
> import android.hardware.camera2.CaptureRequest;
>-import android.media.MediaRecorder;
> import android.os.Handler;
>+import javax.annotation.Nullable;
> import android.util.Range;
> import android.view.Surface;
>-import android.view.WindowManager;
>-import java.util.ArrayList;
>+import java.util.Arrays;
> import java.util.List;
> import java.util.concurrent.TimeUnit;
> import org.webrtc.CameraEnumerationAndroid.CaptureFormat;
>@@ -45,15 +44,12 @@ class Camera2Session implements CameraSession {
> 
>   private static enum SessionState { RUNNING, STOPPED }
> 
>-  private final boolean videoFrameEmitTrialEnabled;
>-
>   private final Handler cameraThreadHandler;
>   private final CreateSessionCallback callback;
>   private final Events events;
>   private final Context applicationContext;
>   private final CameraManager cameraManager;
>   private final SurfaceTextureHelper surfaceTextureHelper;
>-  private final Surface mediaRecorderSurface;
>   private final String cameraId;
>   private final int width;
>   private final int height;
>@@ -67,11 +63,11 @@ class Camera2Session implements CameraSession {
>   private CaptureFormat captureFormat;
> 
>   // Initialized when camera opens
>-  private CameraDevice cameraDevice;
>-  private Surface surface;
>+  @Nullable private CameraDevice cameraDevice;
>+  @Nullable private Surface surface;
> 
>   // Initialized when capture session is created
>-  private CameraCaptureSession captureSession;
>+  @Nullable private CameraCaptureSession captureSession;
> 
>   // State
>   private SessionState state = SessionState.RUNNING;
>@@ -125,17 +121,11 @@ class Camera2Session implements CameraSession {
>       Logging.d(TAG, "Camera opened.");
>       cameraDevice = camera;
> 
>-      final SurfaceTexture surfaceTexture = surfaceTextureHelper.getSurfaceTexture();
>-      surfaceTexture.setDefaultBufferSize(captureFormat.width, captureFormat.height);
>-      surface = new Surface(surfaceTexture);
>-      List<Surface> surfaces = new ArrayList<Surface>();
>-      surfaces.add(surface);
>-      if (mediaRecorderSurface != null) {
>-        Logging.d(TAG, "Add MediaRecorder surface to capture session.");
>-        surfaces.add(mediaRecorderSurface);
>-      }
>+      surfaceTextureHelper.setTextureSize(captureFormat.width, captureFormat.height);
>+      surface = new Surface(surfaceTextureHelper.getSurfaceTexture());
>       try {
>-        camera.createCaptureSession(surfaces, new CaptureSessionCallback(), cameraThreadHandler);
>+        camera.createCaptureSession(
>+            Arrays.asList(surface), new CaptureSessionCallback(), cameraThreadHandler);
>       } catch (CameraAccessException e) {
>         reportError("Failed to create capture session. " + e);
>         return;
>@@ -185,10 +175,6 @@ class Camera2Session implements CameraSession {
>         chooseFocusMode(captureRequestBuilder);
> 
>         captureRequestBuilder.addTarget(surface);
>-        if (mediaRecorderSurface != null) {
>-          Logging.d(TAG, "Add MediaRecorder surface to CaptureRequest.Builder");
>-          captureRequestBuilder.addTarget(mediaRecorderSurface);
>-        }
>         session.setRepeatingRequest(
>             captureRequestBuilder.build(), new CameraCaptureCallback(), cameraThreadHandler);
>       } catch (CameraAccessException e) {
>@@ -196,51 +182,33 @@ class Camera2Session implements CameraSession {
>         return;
>       }
> 
>-      surfaceTextureHelper.startListening(
>-          new SurfaceTextureHelper.OnTextureFrameAvailableListener() {
>-            @Override
>-            public void onTextureFrameAvailable(
>-                int oesTextureId, float[] transformMatrix, long timestampNs) {
>-              checkIsOnCameraThread();
>-
>-              if (state != SessionState.RUNNING) {
>-                Logging.d(TAG, "Texture frame captured but camera is no longer running.");
>-                surfaceTextureHelper.returnTextureFrame();
>-                return;
>-              }
>-
>-              if (!firstFrameReported) {
>-                firstFrameReported = true;
>-                final int startTimeMs =
>-                    (int) TimeUnit.NANOSECONDS.toMillis(System.nanoTime() - constructionTimeNs);
>-                camera2StartTimeMsHistogram.addSample(startTimeMs);
>-              }
>-
>-              int rotation = getFrameOrientation();
>-              if (isCameraFrontFacing) {
>-                // Undo the mirror that the OS "helps" us with.
>-                // http://developer.android.com/reference/android/hardware/Camera.html#setDisplayOrientation(int)
>-                transformMatrix = RendererCommon.multiplyMatrices(
>-                    transformMatrix, RendererCommon.horizontalFlipMatrix());
>-              }
>-
>-              // Undo camera orientation - we report it as rotation instead.
>-              transformMatrix =
>-                  RendererCommon.rotateTextureMatrix(transformMatrix, -cameraOrientation);
>-
>-              if (videoFrameEmitTrialEnabled) {
>-                VideoFrame.Buffer buffer = surfaceTextureHelper.createTextureBuffer(
>-                    captureFormat.width, captureFormat.height,
>-                    RendererCommon.convertMatrixToAndroidGraphicsMatrix(transformMatrix));
>-                final VideoFrame frame = new VideoFrame(buffer, rotation, timestampNs);
>-                events.onFrameCaptured(Camera2Session.this, frame);
>-                frame.release();
>-              } else {
>-                events.onTextureFrameCaptured(Camera2Session.this, captureFormat.width,
>-                    captureFormat.height, oesTextureId, transformMatrix, rotation, timestampNs);
>-              }
>-            }
>-          });
>+      surfaceTextureHelper.startListening((VideoFrame frame) -> {
>+        checkIsOnCameraThread();
>+
>+        if (state != SessionState.RUNNING) {
>+          Logging.d(TAG, "Texture frame captured but camera is no longer running.");
>+          return;
>+        }
>+
>+        if (!firstFrameReported) {
>+          firstFrameReported = true;
>+          final int startTimeMs =
>+              (int) TimeUnit.NANOSECONDS.toMillis(System.nanoTime() - constructionTimeNs);
>+          camera2StartTimeMsHistogram.addSample(startTimeMs);
>+        }
>+
>+        // Undo the mirror that the OS "helps" us with.
>+        // http://developer.android.com/reference/android/hardware/Camera.html#setDisplayOrientation(int)
>+        // Also, undo camera orientation, we report it as rotation instead.
>+        final VideoFrame modifiedFrame =
>+            new VideoFrame(CameraSession.createTextureBufferWithModifiedTransformMatrix(
>+                               (TextureBufferImpl) frame.getBuffer(),
>+                               /* mirror= */ isCameraFrontFacing,
>+                               /* rotation= */ -cameraOrientation),
>+                /* rotation= */ getFrameOrientation(), frame.getTimestampNs());
>+        events.onFrameCaptured(Camera2Session.this, modifiedFrame);
>+        modifiedFrame.release();
>+      });
>       Logging.d(TAG, "Camera device successfully started.");
>       callback.onDone(Camera2Session.this);
>     }
>@@ -303,19 +271,16 @@ class Camera2Session implements CameraSession {
> 
>   public static void create(CreateSessionCallback callback, Events events,
>       Context applicationContext, CameraManager cameraManager,
>-      SurfaceTextureHelper surfaceTextureHelper, MediaRecorder mediaRecorder, String cameraId,
>-      int width, int height, int framerate) {
>+      SurfaceTextureHelper surfaceTextureHelper, String cameraId, int width, int height,
>+      int framerate) {
>     new Camera2Session(callback, events, applicationContext, cameraManager, surfaceTextureHelper,
>-        mediaRecorder, cameraId, width, height, framerate);
>+        cameraId, width, height, framerate);
>   }
> 
>   private Camera2Session(CreateSessionCallback callback, Events events, Context applicationContext,
>-      CameraManager cameraManager, SurfaceTextureHelper surfaceTextureHelper,
>-      MediaRecorder mediaRecorder, String cameraId, int width, int height, int framerate) {
>+      CameraManager cameraManager, SurfaceTextureHelper surfaceTextureHelper, String cameraId,
>+      int width, int height, int framerate) {
>     Logging.d(TAG, "Create new camera2 session on camera " + cameraId);
>-    videoFrameEmitTrialEnabled =
>-        PeerConnectionFactory.fieldTrialsFindFullName(PeerConnectionFactory.VIDEO_FRAME_EMIT_TRIAL)
>-            .equals(PeerConnectionFactory.TRIAL_ENABLED);
> 
>     constructionTimeNs = System.nanoTime();
> 
>@@ -325,7 +290,6 @@ class Camera2Session implements CameraSession {
>     this.applicationContext = applicationContext;
>     this.cameraManager = cameraManager;
>     this.surfaceTextureHelper = surfaceTextureHelper;
>-    this.mediaRecorderSurface = (mediaRecorder != null) ? mediaRecorder.getSurface() : null;
>     this.cameraId = cameraId;
>     this.width = width;
>     this.height = height;
>@@ -442,30 +406,8 @@ class Camera2Session implements CameraSession {
>     }
>   }
> 
>-  private int getDeviceOrientation() {
>-    int orientation = 0;
>-
>-    WindowManager wm = (WindowManager) applicationContext.getSystemService(Context.WINDOW_SERVICE);
>-    switch (wm.getDefaultDisplay().getRotation()) {
>-      case Surface.ROTATION_90:
>-        orientation = 90;
>-        break;
>-      case Surface.ROTATION_180:
>-        orientation = 180;
>-        break;
>-      case Surface.ROTATION_270:
>-        orientation = 270;
>-        break;
>-      case Surface.ROTATION_0:
>-      default:
>-        orientation = 0;
>-        break;
>-    }
>-    return orientation;
>-  }
>-
>   private int getFrameOrientation() {
>-    int rotation = getDeviceOrientation();
>+    int rotation = CameraSession.getDeviceOrientation(applicationContext);
>     if (!isCameraFrontFacing) {
>       rotation = 360 - rotation;
>     }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/CameraCapturer.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/CameraCapturer.java
>index 3b442fb1637..6addd3664d7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/CameraCapturer.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/CameraCapturer.java
>@@ -11,10 +11,10 @@
> package org.webrtc;
> 
> import android.content.Context;
>-import android.media.MediaRecorder;
> import android.os.Handler;
> import android.os.Looper;
> import java.util.Arrays;
>+import javax.annotation.Nullable;
> 
> @SuppressWarnings("deprecation")
> abstract class CameraCapturer implements CameraVideoCapturer {
>@@ -24,30 +24,22 @@ abstract class CameraCapturer implements CameraVideoCapturer {
>     IN_PROGRESS, // Waiting for new switched capture session to start.
>   }
> 
>-  enum MediaRecorderState {
>-    IDLE, // No media recording update (add or remove) requested.
>-    IDLE_TO_ACTIVE, // Waiting for new capture session with added MediaRecorder surface to start.
>-    ACTIVE_TO_IDLE, // Waiting for new capture session with removed MediaRecorder surface to start.
>-    ACTIVE, // MediaRecorder was successfully added to camera pipeline.
>-  }
>-
>   private static final String TAG = "CameraCapturer";
>   private final static int MAX_OPEN_CAMERA_ATTEMPTS = 3;
>   private final static int OPEN_CAMERA_DELAY_MS = 500;
>   private final static int OPEN_CAMERA_TIMEOUT = 10000;
> 
>   private final CameraEnumerator cameraEnumerator;
>-  private final CameraEventsHandler eventsHandler;
>+  @Nullable private final CameraEventsHandler eventsHandler;
>   private final Handler uiThreadHandler;
> 
>+  @Nullable
>   private final CameraSession.CreateSessionCallback createSessionCallback =
>       new CameraSession.CreateSessionCallback() {
>         @Override
>         public void onDone(CameraSession session) {
>           checkIsOnCameraThread();
>-          Logging.d(TAG,
>-              "Create session done. Switch state: " + switchState
>-                  + ". MediaRecorder state: " + mediaRecorderState);
>+          Logging.d(TAG, "Create session done. Switch state: " + switchState);
>           uiThreadHandler.removeCallbacks(openCameraTimeoutRunnable);
>           synchronized (stateLock) {
>             capturerObserver.onCapturerStarted(true /* success */);
>@@ -58,28 +50,15 @@ abstract class CameraCapturer implements CameraVideoCapturer {
>             stateLock.notifyAll();
> 
>             if (switchState == SwitchState.IN_PROGRESS) {
>+              switchState = SwitchState.IDLE;
>               if (switchEventsHandler != null) {
>                 switchEventsHandler.onCameraSwitchDone(cameraEnumerator.isFrontFacing(cameraName));
>                 switchEventsHandler = null;
>               }
>-              switchState = SwitchState.IDLE;
>             } else if (switchState == SwitchState.PENDING) {
>               switchState = SwitchState.IDLE;
>               switchCameraInternal(switchEventsHandler);
>             }
>-
>-            if (mediaRecorderState == MediaRecorderState.IDLE_TO_ACTIVE
>-                || mediaRecorderState == MediaRecorderState.ACTIVE_TO_IDLE) {
>-              if (mediaRecorderEventsHandler != null) {
>-                mediaRecorderEventsHandler.onMediaRecorderSuccess();
>-                mediaRecorderEventsHandler = null;
>-              }
>-              if (mediaRecorderState == MediaRecorderState.IDLE_TO_ACTIVE) {
>-                mediaRecorderState = MediaRecorderState.ACTIVE;
>-              } else {
>-                mediaRecorderState = MediaRecorderState.IDLE;
>-              }
>-            }
>           }
>         }
> 
>@@ -104,14 +83,6 @@ abstract class CameraCapturer implements CameraVideoCapturer {
>                 switchState = SwitchState.IDLE;
>               }
> 
>-              if (mediaRecorderState != MediaRecorderState.IDLE) {
>-                if (mediaRecorderEventsHandler != null) {
>-                  mediaRecorderEventsHandler.onMediaRecorderError(error);
>-                  mediaRecorderEventsHandler = null;
>-                }
>-                mediaRecorderState = MediaRecorderState.IDLE;
>-              }
>-
>               if (failureType == CameraSession.FailureType.DISCONNECTED) {
>                 eventsHandler.onCameraDisconnected();
>               } else {
>@@ -119,12 +90,13 @@ abstract class CameraCapturer implements CameraVideoCapturer {
>               }
>             } else {
>               Logging.w(TAG, "Opening camera failed, retry: " + error);
>-              createSessionInternal(OPEN_CAMERA_DELAY_MS, null /* mediaRecorder */);
>+              createSessionInternal(OPEN_CAMERA_DELAY_MS);
>             }
>           }
>         }
>       };
> 
>+  @Nullable
>   private final CameraSession.Events cameraSessionEventsHandler = new CameraSession.Events() {
>     @Override
>     public void onCameraOpening() {
>@@ -181,7 +153,7 @@ abstract class CameraCapturer implements CameraVideoCapturer {
>       checkIsOnCameraThread();
>       synchronized (stateLock) {
>         if (session != currentSession) {
>-          Logging.w(TAG, "onTextureFrameCaptured from another session.");
>+          Logging.w(TAG, "onFrameCaptured from another session.");
>           return;
>         }
>         if (!firstFrameObserved) {
>@@ -192,44 +164,6 @@ abstract class CameraCapturer implements CameraVideoCapturer {
>         capturerObserver.onFrameCaptured(frame);
>       }
>     }
>-
>-    @Override
>-    public void onByteBufferFrameCaptured(
>-        CameraSession session, byte[] data, int width, int height, int rotation, long timestamp) {
>-      checkIsOnCameraThread();
>-      synchronized (stateLock) {
>-        if (session != currentSession) {
>-          Logging.w(TAG, "onByteBufferFrameCaptured from another session.");
>-          return;
>-        }
>-        if (!firstFrameObserved) {
>-          eventsHandler.onFirstFrameAvailable();
>-          firstFrameObserved = true;
>-        }
>-        cameraStatistics.addFrame();
>-        capturerObserver.onByteBufferFrameCaptured(data, width, height, rotation, timestamp);
>-      }
>-    }
>-
>-    @Override
>-    public void onTextureFrameCaptured(CameraSession session, int width, int height,
>-        int oesTextureId, float[] transformMatrix, int rotation, long timestamp) {
>-      checkIsOnCameraThread();
>-      synchronized (stateLock) {
>-        if (session != currentSession) {
>-          Logging.w(TAG, "onTextureFrameCaptured from another session.");
>-          surfaceHelper.returnTextureFrame();
>-          return;
>-        }
>-        if (!firstFrameObserved) {
>-          eventsHandler.onFirstFrameAvailable();
>-          firstFrameObserved = true;
>-        }
>-        cameraStatistics.addFrame();
>-        capturerObserver.onTextureFrameCaptured(
>-            width, height, oesTextureId, transformMatrix, rotation, timestamp);
>-      }
>-    }
>   };
> 
>   private final Runnable openCameraTimeoutRunnable = new Runnable() {
>@@ -241,31 +175,27 @@ abstract class CameraCapturer implements CameraVideoCapturer {
> 
>   // Initialized on initialize
>   // -------------------------
>-  private Handler cameraThreadHandler;
>+  @Nullable private Handler cameraThreadHandler;
>   private Context applicationContext;
>-  private CapturerObserver capturerObserver;
>-  private SurfaceTextureHelper surfaceHelper;
>+  private org.webrtc.CapturerObserver capturerObserver;
>+  @Nullable private SurfaceTextureHelper surfaceHelper;
> 
>   private final Object stateLock = new Object();
>   private boolean sessionOpening; /* guarded by stateLock */
>-  private CameraSession currentSession; /* guarded by stateLock */
>+  @Nullable private CameraSession currentSession; /* guarded by stateLock */
>   private String cameraName; /* guarded by stateLock */
>   private int width; /* guarded by stateLock */
>   private int height; /* guarded by stateLock */
>   private int framerate; /* guarded by stateLock */
>   private int openAttemptsRemaining; /* guarded by stateLock */
>   private SwitchState switchState = SwitchState.IDLE; /* guarded by stateLock */
>-  private CameraSwitchHandler switchEventsHandler; /* guarded by stateLock */
>+  @Nullable private CameraSwitchHandler switchEventsHandler; /* guarded by stateLock */
>   // Valid from onDone call until stopCapture, otherwise null.
>-  private CameraStatistics cameraStatistics; /* guarded by stateLock */
>+  @Nullable private CameraStatistics cameraStatistics; /* guarded by stateLock */
>   private boolean firstFrameObserved; /* guarded by stateLock */
> 
>-  // Variables used on camera thread - do not require stateLock synchronization.
>-  private MediaRecorderState mediaRecorderState = MediaRecorderState.IDLE;
>-  private MediaRecorderHandler mediaRecorderEventsHandler;
>-
>-  public CameraCapturer(
>-      String cameraName, CameraEventsHandler eventsHandler, CameraEnumerator cameraEnumerator) {
>+  public CameraCapturer(String cameraName, @Nullable CameraEventsHandler eventsHandler,
>+      CameraEnumerator cameraEnumerator) {
>     if (eventsHandler == null) {
>       eventsHandler = new CameraEventsHandler() {
>         @Override
>@@ -300,8 +230,8 @@ abstract class CameraCapturer implements CameraVideoCapturer {
>   }
> 
>   @Override
>-  public void initialize(SurfaceTextureHelper surfaceTextureHelper, Context applicationContext,
>-      CapturerObserver capturerObserver) {
>+  public void initialize(@Nullable SurfaceTextureHelper surfaceTextureHelper,
>+      Context applicationContext, org.webrtc.CapturerObserver capturerObserver) {
>     this.applicationContext = applicationContext;
>     this.capturerObserver = capturerObserver;
>     this.surfaceHelper = surfaceTextureHelper;
>@@ -328,17 +258,17 @@ abstract class CameraCapturer implements CameraVideoCapturer {
> 
>       sessionOpening = true;
>       openAttemptsRemaining = MAX_OPEN_CAMERA_ATTEMPTS;
>-      createSessionInternal(0, null /* mediaRecorder */);
>+      createSessionInternal(0);
>     }
>   }
> 
>-  private void createSessionInternal(int delayMs, final MediaRecorder mediaRecorder) {
>+  private void createSessionInternal(int delayMs) {
>     uiThreadHandler.postDelayed(openCameraTimeoutRunnable, delayMs + OPEN_CAMERA_TIMEOUT);
>     cameraThreadHandler.postDelayed(new Runnable() {
>       @Override
>       public void run() {
>         createCameraSession(createSessionCallback, cameraSessionEventsHandler, applicationContext,
>-            surfaceHelper, mediaRecorder, cameraName, width, height, framerate);
>+            surfaceHelper, cameraName, width, height, framerate);
>       }
>     }, delayMs);
>   }
>@@ -350,7 +280,13 @@ abstract class CameraCapturer implements CameraVideoCapturer {
>     synchronized (stateLock) {
>       while (sessionOpening) {
>         Logging.d(TAG, "Stop capture: Waiting for session to open");
>-        ThreadUtils.waitUninterruptibly(stateLock);
>+        try {
>+          stateLock.wait();
>+        } catch (InterruptedException e) {
>+          Logging.w(TAG, "Stop capture interrupted while waiting for the session to open.");
>+          Thread.currentThread().interrupt();
>+          return;
>+        }
>       }
> 
>       if (currentSession != null) {
>@@ -400,29 +336,6 @@ abstract class CameraCapturer implements CameraVideoCapturer {
>     });
>   }
> 
>-  @Override
>-  public void addMediaRecorderToCamera(
>-      final MediaRecorder mediaRecorder, final MediaRecorderHandler mediaRecoderEventsHandler) {
>-    Logging.d(TAG, "addMediaRecorderToCamera");
>-    cameraThreadHandler.post(new Runnable() {
>-      @Override
>-      public void run() {
>-        updateMediaRecorderInternal(mediaRecorder, mediaRecoderEventsHandler);
>-      }
>-    });
>-  }
>-
>-  @Override
>-  public void removeMediaRecorderFromCamera(final MediaRecorderHandler mediaRecoderEventsHandler) {
>-    Logging.d(TAG, "removeMediaRecorderFromCamera");
>-    cameraThreadHandler.post(new Runnable() {
>-      @Override
>-      public void run() {
>-        updateMediaRecorderInternal(null /* mediaRecorder */, mediaRecoderEventsHandler);
>-      }
>-    });
>-  }
>-
>   @Override
>   public boolean isScreencast() {
>     return false;
>@@ -444,14 +357,15 @@ abstract class CameraCapturer implements CameraVideoCapturer {
>     }
>   }
> 
>-  private void reportCameraSwitchError(String error, CameraSwitchHandler switchEventsHandler) {
>+  private void reportCameraSwitchError(
>+      String error, @Nullable CameraSwitchHandler switchEventsHandler) {
>     Logging.e(TAG, error);
>     if (switchEventsHandler != null) {
>       switchEventsHandler.onCameraSwitchError(error);
>     }
>   }
> 
>-  private void switchCameraInternal(final CameraSwitchHandler switchEventsHandler) {
>+  private void switchCameraInternal(@Nullable final CameraSwitchHandler switchEventsHandler) {
>     Logging.d(TAG, "switchCamera internal");
> 
>     final String[] deviceNames = cameraEnumerator.getDeviceNames();
>@@ -468,10 +382,6 @@ abstract class CameraCapturer implements CameraVideoCapturer {
>         reportCameraSwitchError("Camera switch already in progress.", switchEventsHandler);
>         return;
>       }
>-      if (mediaRecorderState != MediaRecorderState.IDLE) {
>-        reportCameraSwitchError("switchCamera: media recording is active", switchEventsHandler);
>-        return;
>-      }
>       if (!sessionOpening && currentSession == null) {
>         reportCameraSwitchError("switchCamera: camera is not running.", switchEventsHandler);
>         return;
>@@ -502,74 +412,11 @@ abstract class CameraCapturer implements CameraVideoCapturer {
> 
>       sessionOpening = true;
>       openAttemptsRemaining = 1;
>-      createSessionInternal(0, null /* mediaRecorder */);
>+      createSessionInternal(0);
>     }
>     Logging.d(TAG, "switchCamera done");
>   }
> 
>-  private void reportUpdateMediaRecorderError(
>-      String error, MediaRecorderHandler mediaRecoderEventsHandler) {
>-    checkIsOnCameraThread();
>-    Logging.e(TAG, error);
>-    if (mediaRecoderEventsHandler != null) {
>-      mediaRecoderEventsHandler.onMediaRecorderError(error);
>-    }
>-  }
>-
>-  private void updateMediaRecorderInternal(
>-      MediaRecorder mediaRecorder, MediaRecorderHandler mediaRecoderEventsHandler) {
>-    checkIsOnCameraThread();
>-    boolean addMediaRecorder = (mediaRecorder != null);
>-    Logging.d(TAG,
>-        "updateMediaRecoderInternal internal. State: " + mediaRecorderState
>-            + ". Switch state: " + switchState + ". Add MediaRecorder: " + addMediaRecorder);
>-
>-    synchronized (stateLock) {
>-      if ((addMediaRecorder && mediaRecorderState != MediaRecorderState.IDLE)
>-          || (!addMediaRecorder && mediaRecorderState != MediaRecorderState.ACTIVE)) {
>-        reportUpdateMediaRecorderError(
>-            "Incorrect state for MediaRecorder update.", mediaRecoderEventsHandler);
>-        return;
>-      }
>-      if (switchState != SwitchState.IDLE) {
>-        reportUpdateMediaRecorderError(
>-            "MediaRecorder update while camera is switching.", mediaRecoderEventsHandler);
>-        return;
>-      }
>-      if (currentSession == null) {
>-        reportUpdateMediaRecorderError(
>-            "MediaRecorder update while camera is closed.", mediaRecoderEventsHandler);
>-        return;
>-      }
>-      if (sessionOpening) {
>-        reportUpdateMediaRecorderError(
>-            "MediaRecorder update while camera is still opening.", mediaRecoderEventsHandler);
>-        return;
>-      }
>-
>-      this.mediaRecorderEventsHandler = mediaRecoderEventsHandler;
>-      mediaRecorderState =
>-          addMediaRecorder ? MediaRecorderState.IDLE_TO_ACTIVE : MediaRecorderState.ACTIVE_TO_IDLE;
>-
>-      Logging.d(TAG, "updateMediaRecoder: Stopping session");
>-      cameraStatistics.release();
>-      cameraStatistics = null;
>-      final CameraSession oldSession = currentSession;
>-      cameraThreadHandler.post(new Runnable() {
>-        @Override
>-        public void run() {
>-          oldSession.stop();
>-        }
>-      });
>-      currentSession = null;
>-
>-      sessionOpening = true;
>-      openAttemptsRemaining = 1;
>-      createSessionInternal(0, mediaRecorder);
>-    }
>-    Logging.d(TAG, "updateMediaRecoderInternal done");
>-  }
>-
>   private void checkIsOnCameraThread() {
>     if (Thread.currentThread() != cameraThreadHandler.getLooper().getThread()) {
>       Logging.e(TAG, "Check is on camera thread failed.");
>@@ -585,6 +432,6 @@ abstract class CameraCapturer implements CameraVideoCapturer {
> 
>   abstract protected void createCameraSession(
>       CameraSession.CreateSessionCallback createSessionCallback, CameraSession.Events events,
>-      Context applicationContext, SurfaceTextureHelper surfaceTextureHelper,
>-      MediaRecorder mediaRecoder, String cameraName, int width, int height, int framerate);
>+      Context applicationContext, SurfaceTextureHelper surfaceTextureHelper, String cameraName,
>+      int width, int height, int framerate);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/CameraSession.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/CameraSession.java
>index a560378afd4..8d137854d82 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/CameraSession.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/CameraSession.java
>@@ -10,28 +10,27 @@
> 
> package org.webrtc;
> 
>+import android.content.Context;
>+import android.graphics.Matrix;
>+import android.view.WindowManager;
>+import android.view.Surface;
>+
> interface CameraSession {
>   enum FailureType { ERROR, DISCONNECTED }
> 
>   // Callbacks are fired on the camera thread.
>-  public interface CreateSessionCallback {
>+  interface CreateSessionCallback {
>     void onDone(CameraSession session);
>     void onFailure(FailureType failureType, String error);
>   }
> 
>   // Events are fired on the camera thread.
>-  public interface Events {
>+  interface Events {
>     void onCameraOpening();
>     void onCameraError(CameraSession session, String error);
>     void onCameraDisconnected(CameraSession session);
>     void onCameraClosed(CameraSession session);
>     void onFrameCaptured(CameraSession session, VideoFrame frame);
>-
>-    // The old way of passing frames. Will be removed eventually.
>-    void onByteBufferFrameCaptured(
>-        CameraSession session, byte[] data, int width, int height, int rotation, long timestamp);
>-    void onTextureFrameCaptured(CameraSession session, int width, int height, int oesTextureId,
>-        float[] transformMatrix, int rotation, long timestamp);
>   }
> 
>   /**
>@@ -39,4 +38,35 @@ interface CameraSession {
>    * If waitCameraStop is true, also waits for the camera to stop.
>    */
>   void stop();
>+
>+  static int getDeviceOrientation(Context context) {
>+    final WindowManager wm = (WindowManager) context.getSystemService(Context.WINDOW_SERVICE);
>+    switch (wm.getDefaultDisplay().getRotation()) {
>+      case Surface.ROTATION_90:
>+        return 90;
>+      case Surface.ROTATION_180:
>+        return 180;
>+      case Surface.ROTATION_270:
>+        return 270;
>+      case Surface.ROTATION_0:
>+      default:
>+        return 0;
>+    }
>+  }
>+
>+  static VideoFrame.TextureBuffer createTextureBufferWithModifiedTransformMatrix(
>+      TextureBufferImpl buffer, boolean mirror, int rotation) {
>+    final Matrix transformMatrix = new Matrix();
>+    // Perform mirror and rotation around (0.5, 0.5) since that is the center of the texture.
>+    transformMatrix.preTranslate(/* dx= */ 0.5f, /* dy= */ 0.5f);
>+    if (mirror) {
>+      transformMatrix.preScale(/* sx= */ -1f, /* sy= */ 1f);
>+    }
>+    transformMatrix.preRotate(rotation);
>+    transformMatrix.preTranslate(/* dx= */ -0.5f, /* dy= */ -0.5f);
>+
>+    // The width and height are not affected by rotation since Camera2Session has set them to the
>+    // value they should be after undoing the rotation.
>+    return buffer.applyTransformMatrix(transformMatrix, buffer.getWidth(), buffer.getHeight());
>+  }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/DynamicBitrateAdjuster.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/DynamicBitrateAdjuster.java
>index 742cfad7b62..1f970bbbbdc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/DynamicBitrateAdjuster.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/DynamicBitrateAdjuster.java
>@@ -86,10 +86,13 @@ class DynamicBitrateAdjuster extends BaseBitrateAdjuster {
>     timeSinceLastAdjustmentMs = 0;
>   }
> 
>+  private double getBitrateAdjustmentScale() {
>+    return Math.pow(BITRATE_ADJUSTMENT_MAX_SCALE,
>+        (double) bitrateAdjustmentScaleExp / BITRATE_ADJUSTMENT_STEPS);
>+  }
>+
>   @Override
>   public int getAdjustedBitrateBps() {
>-    return (int) (targetBitrateBps
>-        * Math.pow(BITRATE_ADJUSTMENT_MAX_SCALE,
>-              (double) bitrateAdjustmentScaleExp / BITRATE_ADJUSTMENT_STEPS));
>+    return (int) (targetBitrateBps * getBitrateAdjustmentScale());
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/EglBase10.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/EglBase10.java
>index 8cbe0c0225b..ab840f7f264 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/EglBase10.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/EglBase10.java
>@@ -13,6 +13,7 @@ package org.webrtc;
> import android.graphics.Canvas;
> import android.graphics.Rect;
> import android.graphics.SurfaceTexture;
>+import javax.annotation.Nullable;
> import android.view.Surface;
> import android.view.SurfaceHolder;
> import javax.microedition.khronos.egl.EGL10;
>@@ -25,13 +26,13 @@ import javax.microedition.khronos.egl.EGLSurface;
>  * Holds EGL state and utility methods for handling an egl 1.0 EGLContext, an EGLDisplay,
>  * and an EGLSurface.
>  */
>-class EglBase10 extends EglBase {
>+class EglBase10 implements EglBase {
>   // This constant is taken from EGL14.EGL_CONTEXT_CLIENT_VERSION.
>   private static final int EGL_CONTEXT_CLIENT_VERSION = 0x3098;
> 
>   private final EGL10 egl;
>   private EGLContext eglContext;
>-  private EGLConfig eglConfig;
>+  @Nullable private EGLConfig eglConfig;
>   private EGLDisplay eglDisplay;
>   private EGLSurface eglSurface = EGL10.EGL_NO_SURFACE;
> 
>@@ -102,11 +103,13 @@ class EglBase10 extends EglBase {
>       @Override
>       public void setKeepScreenOn(boolean b) {}
> 
>+      @Nullable
>       @Override
>       public Canvas lockCanvas() {
>         return null;
>       }
> 
>+      @Nullable
>       @Override
>       public Canvas lockCanvas(Rect rect) {
>         return null;
>@@ -115,6 +118,7 @@ class EglBase10 extends EglBase {
>       @Override
>       public void unlockCanvasAndPost(Canvas canvas) {}
> 
>+      @Nullable
>       @Override
>       public Rect getSurfaceFrame() {
>         return null;
>@@ -301,7 +305,7 @@ class EglBase10 extends EglBase {
> 
>   // Return an EGLConfig, or die trying.
>   private EGLContext createEglContext(
>-      Context sharedContext, EGLDisplay eglDisplay, EGLConfig eglConfig) {
>+      @Nullable Context sharedContext, EGLDisplay eglDisplay, EGLConfig eglConfig) {
>     if (sharedContext != null && sharedContext.eglContext == EGL10.EGL_NO_CONTEXT) {
>       throw new RuntimeException("Invalid sharedContext");
>     }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/EglBase14.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/EglBase14.java
>index ed5e9543ba8..404c3a6e29d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/EglBase14.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/EglBase14.java
>@@ -19,7 +19,9 @@ import android.opengl.EGLDisplay;
> import android.opengl.EGLExt;
> import android.opengl.EGLSurface;
> import android.os.Build;
>+import javax.annotation.Nullable;
> import android.view.Surface;
>+import org.webrtc.EglBase;
> 
> /**
>  * Holds EGL state and utility methods for handling an EGL14 EGLContext, an EGLDisplay,
>@@ -27,12 +29,12 @@ import android.view.Surface;
>  */
> @SuppressWarnings("ReferenceEquality") // We want to compare to EGL14 constants.
> @TargetApi(18)
>-class EglBase14 extends EglBase {
>+class EglBase14 implements EglBase {
>   private static final String TAG = "EglBase14";
>   private static final int EGLExt_SDK_VERSION = android.os.Build.VERSION_CODES.JELLY_BEAN_MR2;
>   private static final int CURRENT_SDK_VERSION = android.os.Build.VERSION.SDK_INT;
>   private EGLContext eglContext;
>-  private EGLConfig eglConfig;
>+  @Nullable private EGLConfig eglConfig;
>   private EGLDisplay eglDisplay;
>   private EGLSurface eglSurface = EGL14.EGL_NO_SURFACE;
> 
>@@ -256,7 +258,7 @@ class EglBase14 extends EglBase {
> 
>   // Return an EGLConfig, or die trying.
>   private static EGLContext createEglContext(
>-      EglBase14.Context sharedContext, EGLDisplay eglDisplay, EGLConfig eglConfig) {
>+      @Nullable EglBase14.Context sharedContext, EGLDisplay eglDisplay, EGLConfig eglConfig) {
>     if (sharedContext != null && sharedContext.egl14Context == EGL14.EGL_NO_CONTEXT) {
>       throw new RuntimeException("Invalid sharedContext");
>     }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/Empty.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/Empty.java
>new file mode 100644
>index 00000000000..fe9481e182d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/Empty.java
>@@ -0,0 +1,17 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc;
>+
>+/**
>+ * Empty class for use in libjingle_peerconnection_java because all targets require at least one
>+ * Java file.
>+ */
>+class Empty {}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/FramerateBitrateAdjuster.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/FramerateBitrateAdjuster.java
>index a0a0664bf6e..b9d1af66c02 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/FramerateBitrateAdjuster.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/FramerateBitrateAdjuster.java
>@@ -25,6 +25,11 @@ class FramerateBitrateAdjuster extends BaseBitrateAdjuster {
>     }
>     super.setTargets(targetBitrateBps, targetFps);
> 
>-    this.targetBitrateBps *= INITIAL_FPS / this.targetFps;
>+    this.targetBitrateBps = this.targetBitrateBps * INITIAL_FPS / this.targetFps;
>+  }
>+
>+  @Override
>+  public int getCodecConfigFramerate() {
>+    return INITIAL_FPS;
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/GlGenericDrawer.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/GlGenericDrawer.java
>new file mode 100644
>index 00000000000..cd7b1e8c773
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/GlGenericDrawer.java
>@@ -0,0 +1,279 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc;
>+
>+import android.opengl.GLES11Ext;
>+import android.opengl.GLES20;
>+import java.nio.FloatBuffer;
>+import javax.annotation.Nullable;
>+import org.webrtc.GlShader;
>+import org.webrtc.GlUtil;
>+import org.webrtc.RendererCommon;
>+
>+/**
>+ * Helper class to implement an instance of RendererCommon.GlDrawer that can accept multiple input
>+ * sources (OES, RGB, or YUV) using a generic fragment shader as input. The generic fragment shader
>+ * should sample pixel values from the function "sample" that will be provided by this class and
>+ * provides an abstraction for the input source type (OES, RGB, or YUV). The texture coordinate
>+ * variable name will be "tc" and the texture matrix in the vertex shader will be "tex_mat". The
>+ * simplest possible generic shader that just draws pixel from the frame unmodified looks like:
>+ * void main() {
>+ *   gl_FragColor = sample(tc);
>+ * }
>+ * This class covers the cases for most simple shaders and generates the necessary boiler plate.
>+ * Advanced shaders can always implement RendererCommon.GlDrawer directly.
>+ */
>+class GlGenericDrawer implements RendererCommon.GlDrawer {
>+  /**
>+   * The different shader types representing different input sources. YUV here represents three
>+   * separate Y, U, V textures.
>+   */
>+  public static enum ShaderType { OES, RGB, YUV }
>+
>+  /**
>+   * The shader callbacks is used to customize behavior for a GlDrawer. It provides a hook to set
>+   * uniform variables in the shader before a frame is drawn.
>+   */
>+  public static interface ShaderCallbacks {
>+    /**
>+     * This callback is called when a new shader has been compiled and created. It will be called
>+     * for the first frame as well as when the shader type is changed. This callback can be used to
>+     * do custom initialization of the shader that only needs to happen once.
>+     */
>+    void onNewShader(GlShader shader);
>+
>+    /**
>+     * This callback is called before rendering a frame. It can be used to do custom preparation of
>+     * the shader that needs to happen every frame.
>+     */
>+    void onPrepareShader(GlShader shader, float[] texMatrix, int frameWidth, int frameHeight,
>+        int viewportWidth, int viewportHeight);
>+  }
>+
>+  private static final String INPUT_VERTEX_COORDINATE_NAME = "in_pos";
>+  private static final String INPUT_TEXTURE_COORDINATE_NAME = "in_tc";
>+  private static final String TEXTURE_MATRIX_NAME = "tex_mat";
>+  private static final String DEFAULT_VERTEX_SHADER_STRING = "varying vec2 tc;\n"
>+      + "attribute vec4 in_pos;\n"
>+      + "attribute vec4 in_tc;\n"
>+      + "uniform mat4 tex_mat;\n"
>+      + "void main() {\n"
>+      + "  gl_Position = in_pos;\n"
>+      + "  tc = (tex_mat * in_tc).xy;\n"
>+      + "}\n";
>+
>+  // Vertex coordinates in Normalized Device Coordinates, i.e. (-1, -1) is bottom-left and (1, 1)
>+  // is top-right.
>+  private static final FloatBuffer FULL_RECTANGLE_BUFFER = GlUtil.createFloatBuffer(new float[] {
>+      -1.0f, -1.0f, // Bottom left.
>+      1.0f, -1.0f, // Bottom right.
>+      -1.0f, 1.0f, // Top left.
>+      1.0f, 1.0f, // Top right.
>+  });
>+
>+  // Texture coordinates - (0, 0) is bottom-left and (1, 1) is top-right.
>+  private static final FloatBuffer FULL_RECTANGLE_TEXTURE_BUFFER =
>+      GlUtil.createFloatBuffer(new float[] {
>+          0.0f, 0.0f, // Bottom left.
>+          1.0f, 0.0f, // Bottom right.
>+          0.0f, 1.0f, // Top left.
>+          1.0f, 1.0f, // Top right.
>+      });
>+
>+  static String createFragmentShaderString(String genericFragmentSource, ShaderType shaderType) {
>+    final StringBuilder stringBuilder = new StringBuilder();
>+    if (shaderType == ShaderType.OES) {
>+      stringBuilder.append("#extension GL_OES_EGL_image_external : require\n");
>+    }
>+    stringBuilder.append("precision mediump float;\n");
>+    stringBuilder.append("varying vec2 tc;\n");
>+
>+    if (shaderType == ShaderType.YUV) {
>+      stringBuilder.append("uniform sampler2D y_tex;\n");
>+      stringBuilder.append("uniform sampler2D u_tex;\n");
>+      stringBuilder.append("uniform sampler2D v_tex;\n");
>+
>+      // Add separate function for sampling texture.
>+      stringBuilder.append("vec4 sample(vec2 p) {\n");
>+      stringBuilder.append("  float y = texture2D(y_tex, p).r;\n");
>+      stringBuilder.append("  float u = texture2D(u_tex, p).r - 0.5;\n");
>+      stringBuilder.append("  float v = texture2D(v_tex, p).r - 0.5;\n");
>+      stringBuilder.append(
>+          "  return vec4(y + 1.403 * v, y - 0.344 * u - 0.714 * v, y + 1.77 * u, 1);\n");
>+      stringBuilder.append("}\n");
>+      stringBuilder.append(genericFragmentSource);
>+    } else {
>+      final String samplerName = shaderType == ShaderType.OES ? "samplerExternalOES" : "sampler2D";
>+      stringBuilder.append("uniform ").append(samplerName).append(" tex;\n");
>+
>+      // Update the sampling function in-place.
>+      stringBuilder.append(genericFragmentSource.replace("sample(", "texture2D(tex, "));
>+    }
>+
>+    return stringBuilder.toString();
>+  }
>+
>+  private final String genericFragmentSource;
>+  private final String vertexShader;
>+  private final ShaderCallbacks shaderCallbacks;
>+  @Nullable private ShaderType currentShaderType;
>+  @Nullable private GlShader currentShader;
>+  private int inPosLocation;
>+  private int inTcLocation;
>+  private int texMatrixLocation;
>+
>+  public GlGenericDrawer(String genericFragmentSource, ShaderCallbacks shaderCallbacks) {
>+    this(DEFAULT_VERTEX_SHADER_STRING, genericFragmentSource, shaderCallbacks);
>+  }
>+
>+  public GlGenericDrawer(
>+      String vertexShader, String genericFragmentSource, ShaderCallbacks shaderCallbacks) {
>+    this.vertexShader = vertexShader;
>+    this.genericFragmentSource = genericFragmentSource;
>+    this.shaderCallbacks = shaderCallbacks;
>+  }
>+
>+  // Visible for testing.
>+  GlShader createShader(ShaderType shaderType) {
>+    return new GlShader(
>+        vertexShader, createFragmentShaderString(genericFragmentSource, shaderType));
>+  }
>+
>+  /**
>+   * Draw an OES texture frame with specified texture transformation matrix. Required resources are
>+   * allocated at the first call to this function.
>+   */
>+  @Override
>+  public void drawOes(int oesTextureId, float[] texMatrix, int frameWidth, int frameHeight,
>+      int viewportX, int viewportY, int viewportWidth, int viewportHeight) {
>+    prepareShader(
>+        ShaderType.OES, texMatrix, frameWidth, frameHeight, viewportWidth, viewportHeight);
>+    // Bind the texture.
>+    GLES20.glActiveTexture(GLES20.GL_TEXTURE0);
>+    GLES20.glBindTexture(GLES11Ext.GL_TEXTURE_EXTERNAL_OES, oesTextureId);
>+    // Draw the texture.
>+    GLES20.glViewport(viewportX, viewportY, viewportWidth, viewportHeight);
>+    GLES20.glDrawArrays(GLES20.GL_TRIANGLE_STRIP, 0, 4);
>+    // Unbind the texture as a precaution.
>+    GLES20.glBindTexture(GLES11Ext.GL_TEXTURE_EXTERNAL_OES, 0);
>+  }
>+
>+  /**
>+   * Draw a RGB(A) texture frame with specified texture transformation matrix. Required resources
>+   * are allocated at the first call to this function.
>+   */
>+  @Override
>+  public void drawRgb(int textureId, float[] texMatrix, int frameWidth, int frameHeight,
>+      int viewportX, int viewportY, int viewportWidth, int viewportHeight) {
>+    prepareShader(
>+        ShaderType.RGB, texMatrix, frameWidth, frameHeight, viewportWidth, viewportHeight);
>+    // Bind the texture.
>+    GLES20.glActiveTexture(GLES20.GL_TEXTURE0);
>+    GLES20.glBindTexture(GLES20.GL_TEXTURE_2D, textureId);
>+    // Draw the texture.
>+    GLES20.glViewport(viewportX, viewportY, viewportWidth, viewportHeight);
>+    GLES20.glDrawArrays(GLES20.GL_TRIANGLE_STRIP, 0, 4);
>+    // Unbind the texture as a precaution.
>+    GLES20.glBindTexture(GLES20.GL_TEXTURE_2D, 0);
>+  }
>+
>+  /**
>+   * Draw a YUV frame with specified texture transformation matrix. Required resources are allocated
>+   * at the first call to this function.
>+   */
>+  @Override
>+  public void drawYuv(int[] yuvTextures, float[] texMatrix, int frameWidth, int frameHeight,
>+      int viewportX, int viewportY, int viewportWidth, int viewportHeight) {
>+    prepareShader(
>+        ShaderType.YUV, texMatrix, frameWidth, frameHeight, viewportWidth, viewportHeight);
>+    // Bind the textures.
>+    for (int i = 0; i < 3; ++i) {
>+      GLES20.glActiveTexture(GLES20.GL_TEXTURE0 + i);
>+      GLES20.glBindTexture(GLES20.GL_TEXTURE_2D, yuvTextures[i]);
>+    }
>+    // Draw the textures.
>+    GLES20.glViewport(viewportX, viewportY, viewportWidth, viewportHeight);
>+    GLES20.glDrawArrays(GLES20.GL_TRIANGLE_STRIP, 0, 4);
>+    // Unbind the textures as a precaution.
>+    for (int i = 0; i < 3; ++i) {
>+      GLES20.glActiveTexture(GLES20.GL_TEXTURE0 + i);
>+      GLES20.glBindTexture(GLES20.GL_TEXTURE_2D, 0);
>+    }
>+  }
>+
>+  private void prepareShader(ShaderType shaderType, float[] texMatrix, int frameWidth,
>+      int frameHeight, int viewportWidth, int viewportHeight) {
>+    final GlShader shader;
>+    if (shaderType.equals(currentShaderType)) {
>+      // Same shader type as before, reuse exising shader.
>+      shader = currentShader;
>+    } else {
>+      // Allocate new shader.
>+      currentShaderType = shaderType;
>+      if (currentShader != null) {
>+        currentShader.release();
>+      }
>+      shader = createShader(shaderType);
>+      currentShader = shader;
>+
>+      shader.useProgram();
>+      // Set input texture units.
>+      if (shaderType == ShaderType.YUV) {
>+        GLES20.glUniform1i(shader.getUniformLocation("y_tex"), 0);
>+        GLES20.glUniform1i(shader.getUniformLocation("u_tex"), 1);
>+        GLES20.glUniform1i(shader.getUniformLocation("v_tex"), 2);
>+      } else {
>+        GLES20.glUniform1i(shader.getUniformLocation("tex"), 0);
>+      }
>+
>+      GlUtil.checkNoGLES2Error("Create shader");
>+      shaderCallbacks.onNewShader(shader);
>+      texMatrixLocation = shader.getUniformLocation(TEXTURE_MATRIX_NAME);
>+      inPosLocation = shader.getAttribLocation(INPUT_VERTEX_COORDINATE_NAME);
>+      inTcLocation = shader.getAttribLocation(INPUT_TEXTURE_COORDINATE_NAME);
>+    }
>+
>+    shader.useProgram();
>+
>+    // Upload the vertex coordinates.
>+    GLES20.glEnableVertexAttribArray(inPosLocation);
>+    GLES20.glVertexAttribPointer(inPosLocation, /* size= */ 2,
>+        /* type= */ GLES20.GL_FLOAT, /* normalized= */ false, /* stride= */ 0,
>+        FULL_RECTANGLE_BUFFER);
>+
>+    // Upload the texture coordinates.
>+    GLES20.glEnableVertexAttribArray(inTcLocation);
>+    GLES20.glVertexAttribPointer(inTcLocation, /* size= */ 2,
>+        /* type= */ GLES20.GL_FLOAT, /* normalized= */ false, /* stride= */ 0,
>+        FULL_RECTANGLE_TEXTURE_BUFFER);
>+
>+    // Upload the texture transformation matrix.
>+    GLES20.glUniformMatrix4fv(
>+        texMatrixLocation, 1 /* count= */, false /* transpose= */, texMatrix, 0 /* offset= */);
>+
>+    // Do custom per-frame shader preparation.
>+    shaderCallbacks.onPrepareShader(
>+        shader, texMatrix, frameWidth, frameHeight, viewportWidth, viewportHeight);
>+    GlUtil.checkNoGLES2Error("Prepare shader");
>+  }
>+
>+  /**
>+   * Release all GLES resources. This needs to be done manually, otherwise the resources are leaked.
>+   */
>+  @Override
>+  public void release() {
>+    if (currentShader != null) {
>+      currentShader.release();
>+      currentShader = null;
>+      currentShaderType = null;
>+    }
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/H264Utils.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/H264Utils.java
>new file mode 100644
>index 00000000000..abb79c65825
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/H264Utils.java
>@@ -0,0 +1,52 @@
>+/*
>+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc;
>+
>+import java.util.Map;
>+import java.util.HashMap;
>+
>+/** Container for static helper functions related to dealing with H264 codecs. */
>+class H264Utils {
>+  public static final String H264_FMTP_PROFILE_LEVEL_ID = "profile-level-id";
>+  public static final String H264_FMTP_LEVEL_ASYMMETRY_ALLOWED = "level-asymmetry-allowed";
>+  public static final String H264_FMTP_PACKETIZATION_MODE = "packetization-mode";
>+
>+  public static final String H264_PROFILE_CONSTRAINED_BASELINE = "42e0";
>+  public static final String H264_PROFILE_CONSTRAINED_HIGH = "640c";
>+  public static final String H264_LEVEL_3_1 = "1f"; // 31 in hex.
>+  public static final String H264_CONSTRAINED_HIGH_3_1 =
>+      H264_PROFILE_CONSTRAINED_HIGH + H264_LEVEL_3_1;
>+  public static final String H264_CONSTRAINED_BASELINE_3_1 =
>+      H264_PROFILE_CONSTRAINED_BASELINE + H264_LEVEL_3_1;
>+
>+  public static Map<String, String> getDefaultH264Params(boolean isHighProfile) {
>+    final Map<String, String> params = new HashMap<>();
>+    params.put(VideoCodecInfo.H264_FMTP_LEVEL_ASYMMETRY_ALLOWED, "1");
>+    params.put(VideoCodecInfo.H264_FMTP_PACKETIZATION_MODE, "1");
>+    params.put(VideoCodecInfo.H264_FMTP_PROFILE_LEVEL_ID,
>+        isHighProfile ? VideoCodecInfo.H264_CONSTRAINED_HIGH_3_1
>+                      : VideoCodecInfo.H264_CONSTRAINED_BASELINE_3_1);
>+    return params;
>+  }
>+
>+  public static VideoCodecInfo DEFAULT_H264_BASELINE_PROFILE_CODEC =
>+      new VideoCodecInfo("H264", getDefaultH264Params(/* isHighProfile= */ false));
>+  public static VideoCodecInfo DEFAULT_H264_HIGH_PROFILE_CODEC =
>+      new VideoCodecInfo("H264", getDefaultH264Params(/* isHighProfile= */ true));
>+
>+  public static boolean isSameH264Profile(
>+      Map<String, String> params1, Map<String, String> params2) {
>+    return nativeIsSameH264Profile(params1, params2);
>+  }
>+
>+  private static native boolean nativeIsSameH264Profile(
>+      Map<String, String> params1, Map<String, String> params2);
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/HardwareVideoDecoder.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/HardwareVideoDecoder.java
>index 72d0ccf52ea..6cc5379e252 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/HardwareVideoDecoder.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/HardwareVideoDecoder.java
>@@ -21,13 +21,13 @@ import java.nio.ByteBuffer;
> import java.util.concurrent.BlockingDeque;
> import java.util.concurrent.LinkedBlockingDeque;
> import java.util.concurrent.TimeUnit;
>+import javax.annotation.Nullable;
> import org.webrtc.ThreadUtils.ThreadChecker;
> 
> /** Android hardware video decoder. */
> @TargetApi(16)
> @SuppressWarnings("deprecation") // Cannot support API 16 without using deprecated methods.
>-class HardwareVideoDecoder
>-    implements VideoDecoder, SurfaceTextureHelper.OnTextureFrameAvailableListener {
>+class HardwareVideoDecoder implements VideoDecoder, VideoSink {
>   private static final String TAG = "HardwareVideoDecoder";
> 
>   // TODO(magjed): Use MediaFormat.KEY_* constants when part of the public API.
>@@ -52,6 +52,7 @@ class HardwareVideoDecoder
>   // MediaCodec.
>   private static final int DEQUEUE_OUTPUT_BUFFER_TIMEOUT_US = 100000;
> 
>+  private final MediaCodecWrapperFactory mediaCodecWrapperFactory;
>   private final String codecName;
>   private final VideoCodecType codecType;
> 
>@@ -71,7 +72,7 @@ class HardwareVideoDecoder
>   // Output thread runs a loop which polls MediaCodec for decoded output buffers.  It reformats
>   // those buffers into VideoFrames and delivers them to the callback.  Variable is set on decoder
>   // thread and is immutable while the codec is running.
>-  private Thread outputThread;
>+  @Nullable private Thread outputThread;
> 
>   // Checker that ensures work is run on the output thread.
>   private ThreadChecker outputThreadChecker;
>@@ -81,7 +82,7 @@ class HardwareVideoDecoder
>   private ThreadChecker decoderThreadChecker;
> 
>   private volatile boolean running = false;
>-  private volatile Exception shutdownException = null;
>+  @Nullable private volatile Exception shutdownException = null;
> 
>   // Dimensions (width, height, stride, and sliceHeight) may be accessed by either the decode thread
>   // or the output thread.  Accesses should be protected with this lock.
>@@ -101,21 +102,14 @@ class HardwareVideoDecoder
> 
>   private final EglBase.Context sharedContext;
>   // Valid and immutable while the decoder is running.
>-  private SurfaceTextureHelper surfaceTextureHelper;
>-  private Surface surface = null;
>+  @Nullable private SurfaceTextureHelper surfaceTextureHelper;
>+  @Nullable private Surface surface = null;
> 
>   private static class DecodedTextureMetadata {
>-    final int width;
>-    final int height;
>-    final int rotation;
>     final long presentationTimestampUs;
>     final Integer decodeTimeMs;
> 
>-    DecodedTextureMetadata(
>-        int width, int height, int rotation, long presentationTimestampUs, Integer decodeTimeMs) {
>-      this.width = width;
>-      this.height = height;
>-      this.rotation = rotation;
>+    DecodedTextureMetadata(long presentationTimestampUs, Integer decodeTimeMs) {
>       this.presentationTimestampUs = presentationTimestampUs;
>       this.decodeTimeMs = decodeTimeMs;
>     }
>@@ -123,20 +117,21 @@ class HardwareVideoDecoder
> 
>   // Metadata for the last frame rendered to the texture.
>   private final Object renderedTextureMetadataLock = new Object();
>-  private DecodedTextureMetadata renderedTextureMetadata;
>+  @Nullable private DecodedTextureMetadata renderedTextureMetadata;
> 
>   // Decoding proceeds asynchronously.  This callback returns decoded frames to the caller.  Valid
>   // and immutable while the decoder is running.
>-  private Callback callback;
>+  @Nullable private Callback callback;
> 
>   // Valid and immutable while the decoder is running.
>-  private MediaCodec codec = null;
>+  @Nullable private MediaCodecWrapper codec = null;
> 
>-  HardwareVideoDecoder(
>-      String codecName, VideoCodecType codecType, int colorFormat, EglBase.Context sharedContext) {
>+  HardwareVideoDecoder(MediaCodecWrapperFactory mediaCodecWrapperFactory, String codecName,
>+      VideoCodecType codecType, int colorFormat, EglBase.Context sharedContext) {
>     if (!isSupportedColorFormat(colorFormat)) {
>       throw new IllegalArgumentException("Unsupported color format: " + colorFormat);
>     }
>+    this.mediaCodecWrapperFactory = mediaCodecWrapperFactory;
>     this.codecName = codecName;
>     this.codecType = codecType;
>     this.colorFormat = colorFormat;
>@@ -150,7 +145,7 @@ class HardwareVideoDecoder
> 
>     this.callback = callback;
>     if (sharedContext != null) {
>-      surfaceTextureHelper = SurfaceTextureHelper.create("decoder-texture-thread", sharedContext);
>+      surfaceTextureHelper = createSurfaceTextureHelper();
>       surface = new Surface(surfaceTextureHelper.getSurfaceTexture());
>       surfaceTextureHelper.startListening(this);
>     }
>@@ -163,7 +158,7 @@ class HardwareVideoDecoder
>     Logging.d(TAG, "initDecodeInternal");
>     if (outputThread != null) {
>       Logging.e(TAG, "initDecodeInternal called while the codec is already running");
>-      return VideoCodecStatus.ERROR;
>+      return VideoCodecStatus.FALLBACK_SOFTWARE;
>     }
> 
>     // Note:  it is not necessary to initialize dimensions under the lock, since the output thread
>@@ -177,10 +172,10 @@ class HardwareVideoDecoder
>     keyFrameRequired = true;
> 
>     try {
>-      codec = MediaCodec.createByCodecName(codecName);
>+      codec = mediaCodecWrapperFactory.createByCodecName(codecName);
>     } catch (IOException | IllegalArgumentException e) {
>       Logging.e(TAG, "Cannot create media decoder " + codecName);
>-      return VideoCodecStatus.ERROR;
>+      return VideoCodecStatus.FALLBACK_SOFTWARE;
>     }
>     try {
>       MediaFormat format = MediaFormat.createVideoFormat(codecType.mimeType(), width, height);
>@@ -192,7 +187,7 @@ class HardwareVideoDecoder
>     } catch (IllegalStateException e) {
>       Logging.e(TAG, "initDecode failed", e);
>       release();
>-      return VideoCodecStatus.ERROR;
>+      return VideoCodecStatus.FALLBACK_SOFTWARE;
>     }
>     running = true;
>     outputThread = createOutputThread();
>@@ -206,7 +201,7 @@ class HardwareVideoDecoder
>   public VideoCodecStatus decode(EncodedImage frame, DecodeInfo info) {
>     decoderThreadChecker.checkIsOnValidThread();
>     if (codec == null || callback == null) {
>-      Logging.d(TAG, "decode uninitalized, codec: " + codec + ", callback: " + callback);
>+      Logging.d(TAG, "decode uninitalized, codec: " + (codec != null) + ", callback: " + callback);
>       return VideoCodecStatus.UNINITIALIZED;
>     }
> 
>@@ -222,7 +217,8 @@ class HardwareVideoDecoder
>     }
> 
>     // Load dimensions from shared memory under the dimension lock.
>-    int width, height;
>+    final int width;
>+    final int height;
>     synchronized (dimensionLock) {
>       width = this.width;
>       height = this.height;
>@@ -241,11 +237,11 @@ class HardwareVideoDecoder
>       // Need to process a key frame first.
>       if (frame.frameType != EncodedImage.FrameType.VideoFrameKey) {
>         Logging.e(TAG, "decode() - key frame required first");
>-        return VideoCodecStatus.ERROR;
>+        return VideoCodecStatus.NO_OUTPUT;
>       }
>       if (!frame.completeFrame) {
>         Logging.e(TAG, "decode() - complete frame required first");
>-        return VideoCodecStatus.ERROR;
>+        return VideoCodecStatus.NO_OUTPUT;
>       }
>     }
> 
>@@ -310,7 +306,7 @@ class HardwareVideoDecoder
>     Logging.d(TAG, "release");
>     VideoCodecStatus status = releaseInternal();
>     if (surface != null) {
>-      surface.release();
>+      releaseSurface();
>       surface = null;
>       surfaceTextureHelper.stopListening();
>       surfaceTextureHelper.dispose();
>@@ -374,7 +370,8 @@ class HardwareVideoDecoder
>     };
>   }
> 
>-  private void deliverDecodedFrame() {
>+  // Visible for testing.
>+  protected void deliverDecodedFrame() {
>     outputThreadChecker.checkIsOnValidThread();
>     try {
>       MediaCodec.BufferInfo info = new MediaCodec.BufferInfo();
>@@ -417,7 +414,8 @@ class HardwareVideoDecoder
>   private void deliverTextureFrame(final int index, final MediaCodec.BufferInfo info,
>       final int rotation, final Integer decodeTimeMs) {
>     // Load dimensions from shared memory under the dimension lock.
>-    final int width, height;
>+    final int width;
>+    final int height;
>     synchronized (dimensionLock) {
>       width = this.width;
>       height = this.height;
>@@ -425,34 +423,34 @@ class HardwareVideoDecoder
> 
>     synchronized (renderedTextureMetadataLock) {
>       if (renderedTextureMetadata != null) {
>+        codec.releaseOutputBuffer(index, false);
>         return; // We are still waiting for texture for the previous frame, drop this one.
>       }
>-      renderedTextureMetadata = new DecodedTextureMetadata(
>-          width, height, rotation, info.presentationTimeUs, decodeTimeMs);
>-      codec.releaseOutputBuffer(index, true);
>+      surfaceTextureHelper.setTextureSize(width, height);
>+      surfaceTextureHelper.setFrameRotation(rotation);
>+      renderedTextureMetadata = new DecodedTextureMetadata(info.presentationTimeUs, decodeTimeMs);
>+      codec.releaseOutputBuffer(index, /* render= */ true);
>     }
>   }
> 
>   @Override
>-  public void onTextureFrameAvailable(int oesTextureId, float[] transformMatrix, long timestampNs) {
>-    final VideoFrame frame;
>+  public void onFrame(VideoFrame frame) {
>+    final VideoFrame newFrame;
>     final int decodeTimeMs;
>+    final long timestampNs;
>     synchronized (renderedTextureMetadataLock) {
>       if (renderedTextureMetadata == null) {
>         throw new IllegalStateException(
>             "Rendered texture metadata was null in onTextureFrameAvailable.");
>       }
>-      VideoFrame.TextureBuffer oesBuffer = surfaceTextureHelper.createTextureBuffer(
>-          renderedTextureMetadata.width, renderedTextureMetadata.height,
>-          RendererCommon.convertMatrixToAndroidGraphicsMatrix(transformMatrix));
>-      frame = new VideoFrame(oesBuffer, renderedTextureMetadata.rotation,
>-          renderedTextureMetadata.presentationTimestampUs * 1000);
>+      timestampNs = renderedTextureMetadata.presentationTimestampUs * 1000;
>       decodeTimeMs = renderedTextureMetadata.decodeTimeMs;
>       renderedTextureMetadata = null;
>     }
>-
>-    callback.onDecodedFrame(frame, decodeTimeMs, null /* qp */);
>-    frame.release();
>+    // Change timestamp of frame.
>+    final VideoFrame frameWithModifiedTimeStamp =
>+        new VideoFrame(frame.getBuffer(), frame.getRotation(), timestampNs);
>+    callback.onDecodedFrame(frameWithModifiedTimeStamp, decodeTimeMs, null /* qp */);
>   }
> 
>   private void deliverByteFrame(
>@@ -492,7 +490,7 @@ class HardwareVideoDecoder
>       // All other supported color formats are NV12.
>       frameBuffer = copyNV12ToI420Buffer(buffer, stride, sliceHeight, width, height);
>     }
>-    codec.releaseOutputBuffer(result, false);
>+    codec.releaseOutputBuffer(result, /* render= */ false);
> 
>     long presentationTimeNs = info.presentationTimeUs * 1000;
>     VideoFrame frame = new VideoFrame(frameBuffer, rotation, presentationTimeNs);
>@@ -511,37 +509,57 @@ class HardwareVideoDecoder
> 
>   private VideoFrame.Buffer copyI420Buffer(
>       ByteBuffer buffer, int stride, int sliceHeight, int width, int height) {
>+    if (stride % 2 != 0) {
>+      throw new AssertionError("Stride is not divisible by two: " + stride);
>+    }
>+
>+    // Note that the case with odd |sliceHeight| is handled in a special way.
>+    // The chroma height contained in the payload is rounded down instead of
>+    // up, making it one row less than what we expect in WebRTC. Therefore, we
>+    // have to duplicate the last chroma rows for this case. Also, the offset
>+    // between the Y plane and the U plane is unintuitive for this case. See
>+    // http://bugs.webrtc.org/6651 for more info.
>+    final int chromaWidth = (width + 1) / 2;
>+    final int chromaHeight = (sliceHeight % 2 == 0) ? (height + 1) / 2 : height / 2;
>+
>     final int uvStride = stride / 2;
> 
>     final int yPos = 0;
>+    final int yEnd = yPos + stride * height;
>     final int uPos = yPos + stride * sliceHeight;
>-    final int uEnd = uPos + uvStride * (sliceHeight / 2);
>+    final int uEnd = uPos + uvStride * chromaHeight;
>     final int vPos = uPos + uvStride * sliceHeight / 2;
>-    final int vEnd = vPos + uvStride * (sliceHeight / 2);
>+    final int vEnd = vPos + uvStride * chromaHeight;
> 
>-    VideoFrame.I420Buffer frameBuffer = JavaI420Buffer.allocate(width, height);
>+    VideoFrame.I420Buffer frameBuffer = allocateI420Buffer(width, height);
> 
>-    ByteBuffer dataY = frameBuffer.getDataY();
>+    buffer.limit(yEnd);
>     buffer.position(yPos);
>-    buffer.limit(uPos);
>-    dataY.put(buffer);
>+    copyPlane(
>+        buffer.slice(), stride, frameBuffer.getDataY(), frameBuffer.getStrideY(), width, height);
> 
>-    ByteBuffer dataU = frameBuffer.getDataU();
>-    buffer.position(uPos);
>     buffer.limit(uEnd);
>-    dataU.put(buffer);
>-    if (sliceHeight % 2 != 0) {
>-      buffer.position(uEnd - uvStride); // Repeat the last row.
>-      dataU.put(buffer);
>+    buffer.position(uPos);
>+    copyPlane(buffer.slice(), uvStride, frameBuffer.getDataU(), frameBuffer.getStrideU(),
>+        chromaWidth, chromaHeight);
>+    if (sliceHeight % 2 == 1) {
>+      buffer.position(uPos + uvStride * (chromaHeight - 1)); // Seek to beginning of last full row.
>+
>+      ByteBuffer dataU = frameBuffer.getDataU();
>+      dataU.position(frameBuffer.getStrideU() * chromaHeight); // Seek to beginning of last row.
>+      dataU.put(buffer); // Copy the last row.
>     }
> 
>-    ByteBuffer dataV = frameBuffer.getDataV();
>-    buffer.position(vPos);
>     buffer.limit(vEnd);
>-    dataV.put(buffer);
>-    if (sliceHeight % 2 != 0) {
>-      buffer.position(vEnd - uvStride); // Repeat the last row.
>-      dataV.put(buffer);
>+    buffer.position(vPos);
>+    copyPlane(buffer.slice(), uvStride, frameBuffer.getDataV(), frameBuffer.getStrideV(),
>+        chromaWidth, chromaHeight);
>+    if (sliceHeight % 2 == 1) {
>+      buffer.position(vPos + uvStride * (chromaHeight - 1)); // Seek to beginning of last full row.
>+
>+      ByteBuffer dataV = frameBuffer.getDataV();
>+      dataV.position(frameBuffer.getStrideV() * chromaHeight); // Seek to beginning of last row.
>+      dataV.put(buffer); // Copy the last row.
>     }
> 
>     return frameBuffer;
>@@ -632,4 +650,26 @@ class HardwareVideoDecoder
>     }
>     return false;
>   }
>+
>+  // Visible for testing.
>+  protected SurfaceTextureHelper createSurfaceTextureHelper() {
>+    return SurfaceTextureHelper.create("decoder-texture-thread", sharedContext);
>+  }
>+
>+  // Visible for testing.
>+  // TODO(sakal): Remove once Robolectric commit fa991a0 has been rolled to WebRTC.
>+  protected void releaseSurface() {
>+    surface.release();
>+  }
>+
>+  // Visible for testing.
>+  protected VideoFrame.I420Buffer allocateI420Buffer(int width, int height) {
>+    return JavaI420Buffer.allocate(width, height);
>+  }
>+
>+  // Visible for testing.
>+  protected void copyPlane(
>+      ByteBuffer src, int srcStride, ByteBuffer dst, int dstStride, int width, int height) {
>+    YuvHelper.copyPlane(src, srcStride, dst, dstStride, width, height);
>+  }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/HardwareVideoEncoder.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/HardwareVideoEncoder.java
>index c1e50889f36..c5d6df71bb9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/HardwareVideoEncoder.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/HardwareVideoEncoder.java
>@@ -25,6 +25,7 @@ import java.util.Map;
> import java.util.concurrent.BlockingDeque;
> import java.util.concurrent.LinkedBlockingDeque;
> import java.util.concurrent.TimeUnit;
>+import javax.annotation.Nullable;
> import org.webrtc.ThreadUtils.ThreadChecker;
> 
> /** Android hardware video encoder. */
>@@ -45,13 +46,14 @@ class HardwareVideoEncoder implements VideoEncoder {
> 
>   private static final int MAX_VIDEO_FRAMERATE = 30;
> 
>-  // See MAX_ENCODER_Q_SIZE in androidmediaencoder_jni.cc.
>+  // See MAX_ENCODER_Q_SIZE in androidmediaencoder.cc.
>   private static final int MAX_ENCODER_Q_SIZE = 2;
> 
>   private static final int MEDIA_CODEC_RELEASE_TIMEOUT_MS = 5000;
>   private static final int DEQUEUE_OUTPUT_BUFFER_TIMEOUT_US = 100000;
> 
>   // --- Initialized on construction.
>+  private final MediaCodecWrapperFactory mediaCodecWrapperFactory;
>   private final String codecName;
>   private final VideoCodecType codecType;
>   private final Integer surfaceColorFormat;
>@@ -81,16 +83,16 @@ class HardwareVideoEncoder implements VideoEncoder {
>   private boolean automaticResizeOn;
> 
>   // --- Valid and immutable while an encoding session is running.
>-  private MediaCodec codec;
>+  @Nullable private MediaCodecWrapper codec;
>   // Thread that delivers encoded frames to the user callback.
>-  private Thread outputThread;
>+  @Nullable private Thread outputThread;
> 
>   // EGL base wrapping the shared texture context.  Holds hooks to both the shared context and the
>   // input surface.  Making this base current allows textures from the context to be drawn onto the
>   // surface.
>-  private EglBase14 textureEglBase;
>+  @Nullable private EglBase14 textureEglBase;
>   // Input surface for the codec.  The encoder will draw input textures onto this surface.
>-  private Surface textureInputSurface;
>+  @Nullable private Surface textureInputSurface;
> 
>   private int width;
>   private int height;
>@@ -102,7 +104,7 @@ class HardwareVideoEncoder implements VideoEncoder {
> 
>   // --- Only accessed on the output thread.
>   // Contents of the last observed config frame output by the MediaCodec. Used by H.264.
>-  private ByteBuffer configBuffer = null;
>+  @Nullable private ByteBuffer configBuffer = null;
>   private int adjustedBitrate;
> 
>   // Whether the encoder is running.  Volatile so that the output thread can watch this value and
>@@ -110,7 +112,7 @@ class HardwareVideoEncoder implements VideoEncoder {
>   private volatile boolean running = false;
>   // Any exception thrown during shutdown.  The output thread releases the MediaCodec and uses this
>   // value to send exceptions thrown during release back to the encoder thread.
>-  private volatile Exception shutdownException = null;
>+  @Nullable private volatile Exception shutdownException = null;
> 
>   /**
>    * Creates a new HardwareVideoEncoder with the given codecName, codecType, colorFormat, key frame
>@@ -127,10 +129,11 @@ class HardwareVideoEncoder implements VideoEncoder {
>    *     desired bitrates
>    * @throws IllegalArgumentException if colorFormat is unsupported
>    */
>-  public HardwareVideoEncoder(String codecName, VideoCodecType codecType,
>-      Integer surfaceColorFormat, Integer yuvColorFormat, Map<String, String> params,
>-      int keyFrameIntervalSec, int forceKeyFrameIntervalMs, BitrateAdjuster bitrateAdjuster,
>-      EglBase14.Context sharedContext) {
>+  public HardwareVideoEncoder(MediaCodecWrapperFactory mediaCodecWrapperFactory, String codecName,
>+      VideoCodecType codecType, Integer surfaceColorFormat, Integer yuvColorFormat,
>+      Map<String, String> params, int keyFrameIntervalSec, int forceKeyFrameIntervalMs,
>+      BitrateAdjuster bitrateAdjuster, EglBase14.Context sharedContext) {
>+    this.mediaCodecWrapperFactory = mediaCodecWrapperFactory;
>     this.codecName = codecName;
>     this.codecType = codecType;
>     this.surfaceColorFormat = surfaceColorFormat;
>@@ -173,10 +176,10 @@ class HardwareVideoEncoder implements VideoEncoder {
>     lastKeyFrameNs = -1;
> 
>     try {
>-      codec = MediaCodec.createByCodecName(codecName);
>+      codec = mediaCodecWrapperFactory.createByCodecName(codecName);
>     } catch (IOException | IllegalArgumentException e) {
>       Logging.e(TAG, "Cannot create media encoder " + codecName);
>-      return VideoCodecStatus.ERROR;
>+      return VideoCodecStatus.FALLBACK_SOFTWARE;
>     }
> 
>     final int colorFormat = useSurfaceMode ? surfaceColorFormat : yuvColorFormat;
>@@ -185,7 +188,7 @@ class HardwareVideoEncoder implements VideoEncoder {
>       format.setInteger(MediaFormat.KEY_BIT_RATE, adjustedBitrate);
>       format.setInteger(KEY_BITRATE_MODE, VIDEO_ControlRateConstant);
>       format.setInteger(MediaFormat.KEY_COLOR_FORMAT, colorFormat);
>-      format.setInteger(MediaFormat.KEY_FRAME_RATE, bitrateAdjuster.getAdjustedFramerate());
>+      format.setInteger(MediaFormat.KEY_FRAME_RATE, bitrateAdjuster.getCodecConfigFramerate());
>       format.setInteger(MediaFormat.KEY_I_FRAME_INTERVAL, keyFrameIntervalSec);
>       if (codecType == VideoCodecType.H264) {
>         String profileLevelId = params.get(VideoCodecInfo.H264_FMTP_PROFILE_LEVEL_ID);
>@@ -218,7 +221,7 @@ class HardwareVideoEncoder implements VideoEncoder {
>     } catch (IllegalStateException e) {
>       Logging.e(TAG, "initEncodeInternal failed", e);
>       release();
>-      return VideoCodecStatus.ERROR;
>+      return VideoCodecStatus.FALLBACK_SOFTWARE;
>     }
> 
>     running = true;
>@@ -383,7 +386,7 @@ class HardwareVideoEncoder implements VideoEncoder {
>       Logging.e(TAG, "getInputBuffers failed", e);
>       return VideoCodecStatus.ERROR;
>     }
>-    yuvFormat.fillBuffer(buffer, videoFrameBuffer);
>+    fillInputBuffer(buffer, videoFrameBuffer);
> 
>     try {
>       codec.queueInputBuffer(
>@@ -415,7 +418,18 @@ class HardwareVideoEncoder implements VideoEncoder {
>   @Override
>   public ScalingSettings getScalingSettings() {
>     encodeThreadChecker.checkIsOnValidThread();
>-    return new ScalingSettings(automaticResizeOn);
>+    if (automaticResizeOn) {
>+      if (codecType == VideoCodecType.VP8) {
>+        final int kLowVp8QpThreshold = 29;
>+        final int kHighVp8QpThreshold = 95;
>+        return new ScalingSettings(kLowVp8QpThreshold, kHighVp8QpThreshold);
>+      } else if (codecType == VideoCodecType.H264) {
>+        final int kLowH264QpThreshold = 24;
>+        final int kHighH264QpThreshold = 37;
>+        return new ScalingSettings(kLowH264QpThreshold, kHighH264QpThreshold);
>+      }
>+    }
>+    return ScalingSettings.OFF;
>   }
> 
>   @Override
>@@ -469,7 +483,8 @@ class HardwareVideoEncoder implements VideoEncoder {
>     };
>   }
> 
>-  private void deliverEncodedImage() {
>+  // Visible for testing.
>+  protected void deliverEncodedImage() {
>     outputThreadChecker.checkIsOnValidThread();
>     try {
>       MediaCodec.BufferInfo info = new MediaCodec.BufferInfo();
>@@ -506,11 +521,11 @@ class HardwareVideoEncoder implements VideoEncoder {
>           frameBuffer = ByteBuffer.allocateDirect(info.size + configBuffer.capacity());
>           configBuffer.rewind();
>           frameBuffer.put(configBuffer);
>+          frameBuffer.put(codecOutputBuffer);
>+          frameBuffer.rewind();
>         } else {
>-          frameBuffer = ByteBuffer.allocateDirect(info.size);
>+          frameBuffer = codecOutputBuffer.slice();
>         }
>-        frameBuffer.put(codecOutputBuffer);
>-        frameBuffer.rewind();
> 
>         final EncodedImage.FrameType frameType = isKeyFrame
>             ? EncodedImage.FrameType.VideoFrameKey
>@@ -564,6 +579,11 @@ class HardwareVideoEncoder implements VideoEncoder {
>     return sharedContext != null && surfaceColorFormat != null;
>   }
> 
>+  // Visible for testing.
>+  protected void fillInputBuffer(ByteBuffer buffer, VideoFrame.Buffer videoFrameBuffer) {
>+    yuvFormat.fillBuffer(buffer, videoFrameBuffer);
>+  }
>+
>   /**
>    * Enumeration of supported YUV color formats used for MediaCodec's input.
>    */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/JNILogging.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/JNILogging.java
>new file mode 100644
>index 00000000000..f391db61a13
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/JNILogging.java
>@@ -0,0 +1,28 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc;
>+
>+import org.webrtc.CalledByNative;
>+import org.webrtc.Loggable;
>+import org.webrtc.Logging.Severity;
>+
>+class JNILogging {
>+  private final Loggable loggable;
>+
>+  public JNILogging(Loggable loggable) {
>+    this.loggable = loggable;
>+  }
>+
>+  @CalledByNative
>+  public void logToInjectable(String message, Integer severity, String tag) {
>+    loggable.onLogMessage(message, Severity.values()[severity], tag);
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/JniCommon.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/JniCommon.java
>index 9a8455bd56e..e1b2e513d73 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/JniCommon.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/JniCommon.java
>@@ -13,11 +13,11 @@ package org.webrtc;
> import java.nio.ByteBuffer;
> 
> /** Class with static JNI helper functions that are used in many places. */
>-class JniCommon {
>+public class JniCommon {
>   /** Functions to increment/decrement an rtc::RefCountInterface pointer. */
>-  static native void nativeAddRef(long nativeRefCountedPointer);
>-  static native void nativeReleaseRef(long nativeRefCountedPointer);
>+  public static native void nativeAddRef(long refCountedPointer);
>+  public static native void nativeReleaseRef(long refCountedPointer);
> 
>-  public static native ByteBuffer allocateNativeByteBuffer(int size);
>-  public static native void freeNativeByteBuffer(ByteBuffer buffer);
>+  public static native ByteBuffer nativeAllocateByteBuffer(int size);
>+  public static native void nativeFreeByteBuffer(ByteBuffer buffer);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/JniHelper.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/JniHelper.java
>new file mode 100644
>index 00000000000..0d56d5d92b6
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/JniHelper.java
>@@ -0,0 +1,48 @@
>+/*
>+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc;
>+
>+import java.io.UnsupportedEncodingException;
>+import java.util.Map;
>+
>+/**
>+ * This class is only used from jni_helper.cc to give some Java functionality that were not possible
>+ * to generate in other ways due to bugs.webrtc.org/8606 and bugs.webrtc.org/8632.
>+ */
>+class JniHelper {
>+  // TODO(bugs.webrtc.org/8632): Remove.
>+  @CalledByNative
>+  static byte[] getStringBytes(String s) {
>+    try {
>+      return s.getBytes("ISO-8859-1");
>+    } catch (UnsupportedEncodingException e) {
>+      throw new RuntimeException("ISO-8859-1 is unsupported");
>+    }
>+  }
>+
>+  // TODO(bugs.webrtc.org/8632): Remove.
>+  @CalledByNative
>+  static Object getStringClass() {
>+    return String.class;
>+  }
>+
>+  // TODO(bugs.webrtc.org/8606): Remove.
>+  @CalledByNative
>+  static Object getKey(Map.Entry entry) {
>+    return entry.getKey();
>+  }
>+
>+  // TODO(bugs.webrtc.org/8606): Remove.
>+  @CalledByNative
>+  static Object getValue(Map.Entry entry) {
>+    return entry.getValue();
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/MediaCodecUtils.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/MediaCodecUtils.java
>index 8c829b8d0b6..6221a104dcb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/MediaCodecUtils.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/MediaCodecUtils.java
>@@ -14,6 +14,9 @@ import android.annotation.TargetApi;
> import android.media.MediaCodec;
> import android.media.MediaCodecInfo;
> import android.media.MediaCodecInfo.CodecCapabilities;
>+import java.util.HashMap;
>+import java.util.Map;
>+import javax.annotation.Nullable;
> 
> /** Container class for static constants and helpers used with MediaCodec. */
> @TargetApi(18)
>@@ -54,7 +57,8 @@ class MediaCodecUtils {
>   // Color formats supported by texture mode encoding - in order of preference.
>   static final int[] TEXTURE_COLOR_FORMATS = {MediaCodecInfo.CodecCapabilities.COLOR_FormatSurface};
> 
>-  static Integer selectColorFormat(int[] supportedColorFormats, CodecCapabilities capabilities) {
>+  static @Nullable Integer selectColorFormat(
>+      int[] supportedColorFormats, CodecCapabilities capabilities) {
>     for (int supportedColorFormat : supportedColorFormats) {
>       for (int codecColorFormat : capabilities.colorFormats) {
>         if (codecColorFormat == supportedColorFormat) {
>@@ -74,6 +78,18 @@ class MediaCodecUtils {
>     return false;
>   }
> 
>+  static Map<String, String> getCodecProperties(VideoCodecType type, boolean highProfile) {
>+    switch (type) {
>+      case VP8:
>+      case VP9:
>+        return new HashMap<String, String>();
>+      case H264:
>+        return H264Utils.getDefaultH264Params(highProfile);
>+      default:
>+        throw new IllegalArgumentException("Unsupported codec: " + type);
>+    }
>+  }
>+
>   private MediaCodecUtils() {
>     // This class should not be instantiated.
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/MediaCodecWrapper.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/MediaCodecWrapper.java
>new file mode 100644
>index 00000000000..7e376b8e2d5
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/MediaCodecWrapper.java
>@@ -0,0 +1,53 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc;
>+
>+import android.media.MediaCodec;
>+import android.media.MediaCrypto;
>+import android.media.MediaFormat;
>+import android.os.Bundle;
>+import android.view.Surface;
>+import java.nio.ByteBuffer;
>+
>+/**
>+ * Subset of methods defined in {@link android.media.MediaCodec} needed by
>+ * {@link HardwareVideoEncoder} and {@link HardwareVideoDecoder}. This interface
>+ * exists to allow mocking and using a fake implementation in tests.
>+ */
>+interface MediaCodecWrapper {
>+  void configure(MediaFormat format, Surface surface, MediaCrypto crypto, int flags);
>+
>+  void start();
>+
>+  void flush();
>+
>+  void stop();
>+
>+  void release();
>+
>+  int dequeueInputBuffer(long timeoutUs);
>+
>+  void queueInputBuffer(int index, int offset, int size, long presentationTimeUs, int flags);
>+
>+  int dequeueOutputBuffer(MediaCodec.BufferInfo info, long timeoutUs);
>+
>+  void releaseOutputBuffer(int index, boolean render);
>+
>+  MediaFormat getOutputFormat();
>+
>+  ByteBuffer[] getInputBuffers();
>+
>+  ByteBuffer[] getOutputBuffers();
>+
>+  Surface createInputSurface();
>+
>+  void setParameters(Bundle params);
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/MediaCodecWrapperFactory.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/MediaCodecWrapperFactory.java
>new file mode 100644
>index 00000000000..2962cb62a7f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/MediaCodecWrapperFactory.java
>@@ -0,0 +1,22 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc;
>+
>+import java.io.IOException;
>+
>+interface MediaCodecWrapperFactory {
>+  /**
>+   * Creates a new {@link MediaCodecWrapper} by codec name.
>+   *
>+   * <p>For additional information see {@link android.media.MediaCodec#createByCodecName}.
>+   */
>+  MediaCodecWrapper createByCodecName(String name) throws IOException;
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/MediaCodecWrapperFactoryImpl.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/MediaCodecWrapperFactoryImpl.java
>new file mode 100644
>index 00000000000..544d6ebe4f5
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/MediaCodecWrapperFactoryImpl.java
>@@ -0,0 +1,113 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc;
>+
>+import android.annotation.TargetApi;
>+import android.media.MediaCodec;
>+import android.media.MediaCodec.BufferInfo;
>+import android.media.MediaCrypto;
>+import android.media.MediaFormat;
>+import android.os.Bundle;
>+import android.view.Surface;
>+import java.io.IOException;
>+import java.nio.ByteBuffer;
>+
>+/**
>+ * Implementation of MediaCodecWrapperFactory that returns MediaCodecInterfaces wrapping
>+ * {@link android.media.MediaCodec} objects.
>+ */
>+class MediaCodecWrapperFactoryImpl implements MediaCodecWrapperFactory {
>+  private static class MediaCodecWrapperImpl implements MediaCodecWrapper {
>+    private final MediaCodec mediaCodec;
>+
>+    public MediaCodecWrapperImpl(MediaCodec mediaCodec) {
>+      this.mediaCodec = mediaCodec;
>+    }
>+
>+    @Override
>+    public void configure(MediaFormat format, Surface surface, MediaCrypto crypto, int flags) {
>+      mediaCodec.configure(format, surface, crypto, flags);
>+    }
>+
>+    @Override
>+    public void start() {
>+      mediaCodec.start();
>+    }
>+
>+    @Override
>+    public void flush() {
>+      mediaCodec.flush();
>+    }
>+
>+    @Override
>+    public void stop() {
>+      mediaCodec.stop();
>+    }
>+
>+    @Override
>+    public void release() {
>+      mediaCodec.release();
>+    }
>+
>+    @Override
>+    public int dequeueInputBuffer(long timeoutUs) {
>+      return mediaCodec.dequeueInputBuffer(timeoutUs);
>+    }
>+
>+    @Override
>+    public void queueInputBuffer(
>+        int index, int offset, int size, long presentationTimeUs, int flags) {
>+      mediaCodec.queueInputBuffer(index, offset, size, presentationTimeUs, flags);
>+    }
>+
>+    @Override
>+    public int dequeueOutputBuffer(BufferInfo info, long timeoutUs) {
>+      return mediaCodec.dequeueOutputBuffer(info, timeoutUs);
>+    }
>+
>+    @Override
>+    public void releaseOutputBuffer(int index, boolean render) {
>+      mediaCodec.releaseOutputBuffer(index, render);
>+    }
>+
>+    @Override
>+    public MediaFormat getOutputFormat() {
>+      return mediaCodec.getOutputFormat();
>+    }
>+
>+    @Override
>+    public ByteBuffer[] getInputBuffers() {
>+      return mediaCodec.getInputBuffers();
>+    }
>+
>+    @Override
>+    public ByteBuffer[] getOutputBuffers() {
>+      return mediaCodec.getOutputBuffers();
>+    }
>+
>+    @Override
>+    @TargetApi(18)
>+    public Surface createInputSurface() {
>+      return mediaCodec.createInputSurface();
>+    }
>+
>+    @Override
>+    @TargetApi(19)
>+    public void setParameters(Bundle params) {
>+      mediaCodec.setParameters(params);
>+    }
>+  }
>+
>+  @Override
>+  public MediaCodecWrapper createByCodecName(String name) throws IOException {
>+    return new MediaCodecWrapperImpl(MediaCodec.createByCodecName(name));
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/NV12Buffer.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/NV12Buffer.java
>index 9c5df025621..7fe7f587afa 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/NV12Buffer.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/NV12Buffer.java
>@@ -11,6 +11,7 @@
> package org.webrtc;
> 
> import java.nio.ByteBuffer;
>+import javax.annotation.Nullable;
> 
> public class NV12Buffer implements VideoFrame.Buffer {
>   private final int width;
>@@ -18,21 +19,16 @@ public class NV12Buffer implements VideoFrame.Buffer {
>   private final int stride;
>   private final int sliceHeight;
>   private final ByteBuffer buffer;
>-  private final Runnable releaseCallback;
>-  private final Object refCountLock = new Object();
>-
>-  private int refCount;
>+  private final RefCountDelegate refCountDelegate;
> 
>   public NV12Buffer(int width, int height, int stride, int sliceHeight, ByteBuffer buffer,
>-      Runnable releaseCallback) {
>+      @Nullable Runnable releaseCallback) {
>     this.width = width;
>     this.height = height;
>     this.stride = stride;
>     this.sliceHeight = sliceHeight;
>     this.buffer = buffer;
>-    this.releaseCallback = releaseCallback;
>-
>-    refCount = 1;
>+    this.refCountDelegate = new RefCountDelegate(releaseCallback);
>   }
> 
>   @Override
>@@ -52,18 +48,12 @@ public class NV12Buffer implements VideoFrame.Buffer {
> 
>   @Override
>   public void retain() {
>-    synchronized (refCountLock) {
>-      ++refCount;
>-    }
>+    refCountDelegate.retain();
>   }
> 
>   @Override
>   public void release() {
>-    synchronized (refCountLock) {
>-      if (--refCount == 0 && releaseCallback != null) {
>-        releaseCallback.run();
>-      }
>-    }
>+    refCountDelegate.release();
>   }
> 
>   @Override
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/NV21Buffer.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/NV21Buffer.java
>index 3a528be6df4..a2a77434855 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/NV21Buffer.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/NV21Buffer.java
>@@ -11,21 +11,20 @@
> package org.webrtc;
> 
> import java.nio.ByteBuffer;
>+import java.util.concurrent.atomic.AtomicInteger;
>+import javax.annotation.Nullable;
> 
> public class NV21Buffer implements VideoFrame.Buffer {
>   private final byte[] data;
>   private final int width;
>   private final int height;
>-  private final Runnable releaseCallback;
>-  private final Object refCountLock = new Object();
>+  private final RefCountDelegate refCountDelegate;
> 
>-  private int refCount = 1;
>-
>-  public NV21Buffer(byte[] data, int width, int height, Runnable releaseCallback) {
>+  public NV21Buffer(byte[] data, int width, int height, @Nullable Runnable releaseCallback) {
>     this.data = data;
>     this.width = width;
>     this.height = height;
>-    this.releaseCallback = releaseCallback;
>+    this.refCountDelegate = new RefCountDelegate(releaseCallback);
>   }
> 
>   @Override
>@@ -47,18 +46,12 @@ public class NV21Buffer implements VideoFrame.Buffer {
> 
>   @Override
>   public void retain() {
>-    synchronized (refCountLock) {
>-      ++refCount;
>-    }
>+    refCountDelegate.retain();
>   }
> 
>   @Override
>   public void release() {
>-    synchronized (refCountLock) {
>-      if (--refCount == 0 && releaseCallback != null) {
>-        releaseCallback.run();
>-      }
>-    }
>+    refCountDelegate.release();
>   }
> 
>   @Override
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/NativeCapturerObserver.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/NativeCapturerObserver.java
>new file mode 100644
>index 00000000000..24869818a66
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/NativeCapturerObserver.java
>@@ -0,0 +1,48 @@
>+/*
>+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc;
>+
>+import javax.annotation.Nullable;
>+
>+/**
>+ * Implements VideoCapturer.CapturerObserver and feeds frames to
>+ * webrtc::jni::AndroidVideoTrackSource.
>+ */
>+class NativeCapturerObserver implements CapturerObserver {
>+  // Pointer to webrtc::jni::AndroidVideoTrackSource.
>+  private final long nativeSource;
>+
>+  @CalledByNative
>+  public NativeCapturerObserver(long nativeSource) {
>+    this.nativeSource = nativeSource;
>+  }
>+
>+  @Override
>+  public void onCapturerStarted(boolean success) {
>+    nativeCapturerStarted(nativeSource, success);
>+  }
>+
>+  @Override
>+  public void onCapturerStopped() {
>+    nativeCapturerStopped(nativeSource);
>+  }
>+
>+  @Override
>+  public void onFrameCaptured(VideoFrame frame) {
>+    nativeOnFrameCaptured(nativeSource, frame.getBuffer().getWidth(), frame.getBuffer().getHeight(),
>+        frame.getRotation(), frame.getTimestampNs(), frame.getBuffer());
>+  }
>+
>+  private static native void nativeCapturerStarted(long source, boolean success);
>+  private static native void nativeCapturerStopped(long source);
>+  private static native void nativeOnFrameCaptured(
>+      long source, int width, int height, int rotation, long timestampNs, VideoFrame.Buffer frame);
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/NativeClassQualifiedName.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/NativeClassQualifiedName.java
>deleted file mode 100644
>index 5adce75ce5b..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/NativeClassQualifiedName.java
>+++ /dev/null
>@@ -1,31 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-package org.webrtc;
>-
>-import java.lang.annotation.ElementType;
>-import java.lang.annotation.Retention;
>-import java.lang.annotation.RetentionPolicy;
>-import java.lang.annotation.Target;
>-
>-/**
>- * @NativeClassQualifiedName is used by the JNI generator to create the necessary JNI
>- * bindings to call into the specified native class name.
>- */
>-@Target(ElementType.METHOD)
>-@Retention(RetentionPolicy.RUNTIME)
>-public @interface NativeClassQualifiedName {
>-  /*
>-   * Tells which native class the method is going to be bound to.
>-   * The first parameter of the annotated method must be an int nativePtr pointing to
>-   * an instance of this class.
>-   */
>-  public String value();
>-}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/NativeLibrary.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/NativeLibrary.java
>index 3b88f14b305..91ec4ca3c6b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/NativeLibrary.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/NativeLibrary.java
>@@ -34,14 +34,14 @@ class NativeLibrary {
>    * Loads the native library. Clients should call PeerConnectionFactory.initialize. It will call
>    * this method for them.
>    */
>-  static void initialize(NativeLibraryLoader loader) {
>+  static void initialize(NativeLibraryLoader loader, String libraryName) {
>     synchronized (lock) {
>       if (libraryLoaded) {
>         Logging.d(TAG, "Native library has already been loaded.");
>         return;
>       }
>-      Logging.d(TAG, "Loading native library.");
>-      libraryLoaded = loader.load("jingle_peerconnection_so");
>+      Logging.d(TAG, "Loading native library: " + libraryName);
>+      libraryLoaded = loader.load(libraryName);
>     }
>   }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/RefCountDelegate.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/RefCountDelegate.java
>new file mode 100644
>index 00000000000..6ea88243126
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/RefCountDelegate.java
>@@ -0,0 +1,41 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc;
>+
>+import java.util.concurrent.atomic.AtomicInteger;
>+import javax.annotation.Nullable;
>+
>+/**
>+ * Implementation of RefCounted that executes a Runnable once the ref count reaches zero.
>+ */
>+class RefCountDelegate implements RefCounted {
>+  private final AtomicInteger refCount = new AtomicInteger(1);
>+  private final @Nullable Runnable releaseCallback;
>+
>+  /**
>+   * @param releaseCallback Callback that will be executed once the ref count reaches zero.
>+   */
>+  public RefCountDelegate(@Nullable Runnable releaseCallback) {
>+    this.releaseCallback = releaseCallback;
>+  }
>+
>+  @Override
>+  public void retain() {
>+    refCount.incrementAndGet();
>+  }
>+
>+  @Override
>+  public void release() {
>+    if (refCount.decrementAndGet() == 0 && releaseCallback != null) {
>+      releaseCallback.run();
>+    }
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/TextureBufferImpl.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/TextureBufferImpl.java
>deleted file mode 100644
>index b862f1dd723..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/TextureBufferImpl.java
>+++ /dev/null
>@@ -1,105 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-package org.webrtc;
>-
>-import android.graphics.Matrix;
>-import java.nio.ByteBuffer;
>-
>-/**
>- * Android texture buffer backed by a SurfaceTextureHelper's texture. The buffer calls
>- * |releaseCallback| when it is released.
>- */
>-class TextureBufferImpl implements VideoFrame.TextureBuffer {
>-  private final int width;
>-  private final int height;
>-  private final Type type;
>-  private final int id;
>-  private final Matrix transformMatrix;
>-  private final SurfaceTextureHelper surfaceTextureHelper;
>-  private final Runnable releaseCallback;
>-  private final Object refCountLock = new Object();
>-  private int refCount;
>-
>-  public TextureBufferImpl(int width, int height, Type type, int id, Matrix transformMatrix,
>-      SurfaceTextureHelper surfaceTextureHelper, Runnable releaseCallback) {
>-    this.width = width;
>-    this.height = height;
>-    this.type = type;
>-    this.id = id;
>-    this.transformMatrix = transformMatrix;
>-    this.surfaceTextureHelper = surfaceTextureHelper;
>-    this.releaseCallback = releaseCallback;
>-    this.refCount = 1; // Creator implicitly holds a reference.
>-  }
>-
>-  @Override
>-  public VideoFrame.TextureBuffer.Type getType() {
>-    return type;
>-  }
>-
>-  @Override
>-  public int getTextureId() {
>-    return id;
>-  }
>-
>-  @Override
>-  public Matrix getTransformMatrix() {
>-    return transformMatrix;
>-  }
>-
>-  @Override
>-  public int getWidth() {
>-    return width;
>-  }
>-
>-  @Override
>-  public int getHeight() {
>-    return height;
>-  }
>-
>-  @Override
>-  public VideoFrame.I420Buffer toI420() {
>-    return surfaceTextureHelper.textureToYuv(this);
>-  }
>-
>-  @Override
>-  public void retain() {
>-    synchronized (refCountLock) {
>-      ++refCount;
>-    }
>-  }
>-
>-  @Override
>-  public void release() {
>-    synchronized (refCountLock) {
>-      if (--refCount == 0 && releaseCallback != null) {
>-        releaseCallback.run();
>-      }
>-    }
>-  }
>-
>-  @Override
>-  public VideoFrame.Buffer cropAndScale(
>-      int cropX, int cropY, int cropWidth, int cropHeight, int scaleWidth, int scaleHeight) {
>-    retain();
>-    Matrix newMatrix = new Matrix(transformMatrix);
>-    newMatrix.postScale(cropWidth / (float) width, cropHeight / (float) height);
>-    newMatrix.postTranslate(cropX / (float) width, cropY / (float) height);
>-
>-    return new TextureBufferImpl(
>-        scaleWidth, scaleHeight, type, id, newMatrix, surfaceTextureHelper, new Runnable() {
>-          @Override
>-          public void run() {
>-            release();
>-          }
>-        });
>-  }
>-}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/VP8Decoder.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/VP8Decoder.java
>index ea7ff9730d7..ab29c349c7a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/VP8Decoder.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/VP8Decoder.java
>@@ -11,5 +11,10 @@
> package org.webrtc;
> 
> class VP8Decoder extends WrappedNativeVideoDecoder {
>-  @Override native long createNativeDecoder();
>+  @Override
>+  public long createNativeVideoDecoder() {
>+    return nativeCreateDecoder();
>+  }
>+
>+  static native long nativeCreateDecoder();
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/VP8Encoder.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/VP8Encoder.java
>index 41575686702..c2f8f504177 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/VP8Encoder.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/VP8Encoder.java
>@@ -11,10 +11,15 @@
> package org.webrtc;
> 
> class VP8Encoder extends WrappedNativeVideoEncoder {
>-  @Override native long createNativeEncoder();
>+  @Override
>+  public long createNativeVideoEncoder() {
>+    return nativeCreateEncoder();
>+  }
>+
>+  static native long nativeCreateEncoder();
> 
>   @Override
>-  boolean isSoftwareEncoder() {
>-    return true;
>+  public boolean isHardwareEncoder() {
>+    return false;
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/VP9Decoder.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/VP9Decoder.java
>index 619ba917689..b6fee2d362b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/VP9Decoder.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/VP9Decoder.java
>@@ -11,7 +11,12 @@
> package org.webrtc;
> 
> class VP9Decoder extends WrappedNativeVideoDecoder {
>-  @Override native long createNativeDecoder();
>+  @Override
>+  public long createNativeVideoDecoder() {
>+    return nativeCreateDecoder();
>+  }
> 
>-  static native boolean isSupported();
>+  static native long nativeCreateDecoder();
>+
>+  static native boolean nativeIsSupported();
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/VP9Encoder.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/VP9Encoder.java
>index 7839ab9981d..bc705bd9c60 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/VP9Encoder.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/VP9Encoder.java
>@@ -11,12 +11,17 @@
> package org.webrtc;
> 
> class VP9Encoder extends WrappedNativeVideoEncoder {
>-  @Override native long createNativeEncoder();
>+  @Override
>+  public long createNativeVideoEncoder() {
>+    return nativeCreateEncoder();
>+  }
>+
>+  static native long nativeCreateEncoder();
> 
>   @Override
>-  boolean isSoftwareEncoder() {
>-    return true;
>+  public boolean isHardwareEncoder() {
>+    return false;
>   }
> 
>-  static native boolean isSupported();
>+  static native boolean nativeIsSupported();
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/VideoDecoderWrapper.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/VideoDecoderWrapper.java
>index 93e74e05e4b..2aae0416402 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/VideoDecoderWrapper.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/VideoDecoderWrapper.java
>@@ -22,7 +22,6 @@ class VideoDecoderWrapper {
>                Integer qp) -> nativeOnDecodedFrame(nativeDecoder, frame, decodeTimeMs, qp);
>   }
> 
>-  @NativeClassQualifiedName("webrtc::jni::VideoDecoderWrapper")
>   private static native void nativeOnDecodedFrame(
>-      long nativeDecoder, VideoFrame frame, Integer decodeTimeMs, Integer qp);
>+      long nativeVideoDecoderWrapper, VideoFrame frame, Integer decodeTimeMs, Integer qp);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/VideoEncoderWrapper.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/VideoEncoderWrapper.java
>index bc9fcece6df..207d7d09c90 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/VideoEncoderWrapper.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/VideoEncoderWrapper.java
>@@ -11,6 +11,7 @@
> package org.webrtc;
> 
> // Explicit imports necessary for JNI generation.
>+import javax.annotation.Nullable;
> import org.webrtc.VideoEncoder;
> import java.nio.ByteBuffer;
> 
>@@ -23,11 +24,13 @@ class VideoEncoderWrapper {
>     return scalingSettings.on;
>   }
> 
>+  @Nullable
>   @CalledByNative
>   static Integer getScalingSettingsLow(VideoEncoder.ScalingSettings scalingSettings) {
>     return scalingSettings.low;
>   }
> 
>+  @Nullable
>   @CalledByNative
>   static Integer getScalingSettingsHigh(VideoEncoder.ScalingSettings scalingSettings) {
>     return scalingSettings.high;
>@@ -41,8 +44,7 @@ class VideoEncoderWrapper {
>                    frame.rotation, frame.completeFrame, frame.qp);
>   }
> 
>-  @NativeClassQualifiedName("webrtc::jni::VideoEncoderWrapper")
>-  private static native void nativeOnEncodedFrame(long nativeEncoder, ByteBuffer buffer,
>+  private static native void nativeOnEncodedFrame(long nativeVideoEncoderWrapper, ByteBuffer buffer,
>       int encodedWidth, int encodedHeight, long captureTimeNs, int frameType, int rotation,
>       boolean completeFrame, Integer qp);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/WrappedNativeI420Buffer.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/WrappedNativeI420Buffer.java
>index ba7f386e19c..0461660fcff 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/WrappedNativeI420Buffer.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/WrappedNativeI420Buffer.java
>@@ -26,6 +26,7 @@ class WrappedNativeI420Buffer implements VideoFrame.I420Buffer {
>   private final int strideV;
>   private final long nativeBuffer;
> 
>+  @CalledByNative
>   WrappedNativeI420Buffer(int width, int height, ByteBuffer dataY, int strideY, ByteBuffer dataU,
>       int strideU, ByteBuffer dataV, int strideV, long nativeBuffer) {
>     this.width = width;
>@@ -103,7 +104,7 @@ class WrappedNativeI420Buffer implements VideoFrame.I420Buffer {
>   @Override
>   public VideoFrame.Buffer cropAndScale(
>       int cropX, int cropY, int cropWidth, int cropHeight, int scaleWidth, int scaleHeight) {
>-    return VideoFrame.cropAndScaleI420(
>+    return JavaI420Buffer.cropAndScaleI420(
>         this, cropX, cropY, cropWidth, cropHeight, scaleWidth, scaleHeight);
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/WrappedNativeVideoDecoder.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/WrappedNativeVideoDecoder.java
>index 916d07bd8e1..80070d3ada8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/WrappedNativeVideoDecoder.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/WrappedNativeVideoDecoder.java
>@@ -14,7 +14,7 @@ package org.webrtc;
>  * Wraps a native webrtc::VideoDecoder.
>  */
> abstract class WrappedNativeVideoDecoder implements VideoDecoder {
>-  @CalledByNative abstract long createNativeDecoder();
>+  @Override public abstract long createNativeVideoDecoder();
> 
>   @Override
>   public VideoCodecStatus initDecode(Settings settings, Callback decodeCallback) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/WrappedNativeVideoEncoder.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/WrappedNativeVideoEncoder.java
>index c41b3b02129..07e84286210 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/WrappedNativeVideoEncoder.java
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/WrappedNativeVideoEncoder.java
>@@ -14,9 +14,8 @@ package org.webrtc;
>  * Wraps a native webrtc::VideoEncoder.
>  */
> abstract class WrappedNativeVideoEncoder implements VideoEncoder {
>-  @CalledByNative abstract long createNativeEncoder();
>-
>-  abstract boolean isSoftwareEncoder();
>+  @Override public abstract long createNativeVideoEncoder();
>+  @Override public abstract boolean isHardwareEncoder();
> 
>   @Override
>   public VideoCodecStatus initEncode(Settings settings, Callback encodeCallback) {
>@@ -52,15 +51,4 @@ abstract class WrappedNativeVideoEncoder implements VideoEncoder {
>   public String getImplementationName() {
>     throw new UnsupportedOperationException("Not implemented.");
>   }
>-
>-  @CalledByNative
>-  static boolean isWrappedSoftwareEncoder(VideoEncoder encoder) {
>-    return (encoder instanceof WrappedNativeVideoEncoder)
>-        && ((WrappedNativeVideoEncoder) encoder).isSoftwareEncoder();
>-  }
>-
>-  @CalledByNative
>-  static boolean isInstanceOf(VideoEncoder encoder) {
>-    return encoder instanceof WrappedNativeVideoEncoder;
>-  }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/audio/VolumeLogger.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/audio/VolumeLogger.java
>new file mode 100644
>index 00000000000..859d7c16599
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/audio/VolumeLogger.java
>@@ -0,0 +1,83 @@
>+/*
>+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc.audio;
>+
>+import android.media.AudioManager;
>+import java.util.Timer;
>+import java.util.TimerTask;
>+import javax.annotation.Nullable;
>+import org.webrtc.Logging;
>+
>+// TODO(magjed): Do we really need to spawn a new thread just to log volume? Can we re-use the
>+// AudioTrackThread instead?
>+/**
>+ * Private utility class that periodically checks and logs the volume level of the audio stream that
>+ * is currently controlled by the volume control. A timer triggers logs once every 30 seconds and
>+ * the timer's associated thread is named "WebRtcVolumeLevelLoggerThread".
>+ */
>+class VolumeLogger {
>+  private static final String TAG = "VolumeLogger";
>+  private static final String THREAD_NAME = "WebRtcVolumeLevelLoggerThread";
>+  private static final int TIMER_PERIOD_IN_SECONDS = 30;
>+
>+  private final AudioManager audioManager;
>+  private @Nullable Timer timer;
>+
>+  public VolumeLogger(AudioManager audioManager) {
>+    this.audioManager = audioManager;
>+  }
>+
>+  public void start() {
>+    Logging.d(TAG, "start" + WebRtcAudioUtils.getThreadInfo());
>+    if (timer != null) {
>+      return;
>+    }
>+    Logging.d(TAG, "audio mode is: " + WebRtcAudioUtils.modeToString(audioManager.getMode()));
>+
>+    timer = new Timer(THREAD_NAME);
>+    timer.schedule(new LogVolumeTask(audioManager.getStreamMaxVolume(AudioManager.STREAM_RING),
>+                       audioManager.getStreamMaxVolume(AudioManager.STREAM_VOICE_CALL)),
>+        0, TIMER_PERIOD_IN_SECONDS * 1000);
>+  }
>+
>+  private class LogVolumeTask extends TimerTask {
>+    private final int maxRingVolume;
>+    private final int maxVoiceCallVolume;
>+
>+    LogVolumeTask(int maxRingVolume, int maxVoiceCallVolume) {
>+      this.maxRingVolume = maxRingVolume;
>+      this.maxVoiceCallVolume = maxVoiceCallVolume;
>+    }
>+
>+    @Override
>+    public void run() {
>+      final int mode = audioManager.getMode();
>+      if (mode == AudioManager.MODE_RINGTONE) {
>+        Logging.d(TAG,
>+            "STREAM_RING stream volume: " + audioManager.getStreamVolume(AudioManager.STREAM_RING)
>+                + " (max=" + maxRingVolume + ")");
>+      } else if (mode == AudioManager.MODE_IN_COMMUNICATION) {
>+        Logging.d(TAG,
>+            "VOICE_CALL stream volume: "
>+                + audioManager.getStreamVolume(AudioManager.STREAM_VOICE_CALL)
>+                + " (max=" + maxVoiceCallVolume + ")");
>+      }
>+    }
>+  }
>+
>+  public void stop() {
>+    Logging.d(TAG, "stop" + WebRtcAudioUtils.getThreadInfo());
>+    if (timer != null) {
>+      timer.cancel();
>+      timer = null;
>+    }
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/audio/WebRtcAudioEffects.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/audio/WebRtcAudioEffects.java
>new file mode 100644
>index 00000000000..77aec83bcce
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/audio/WebRtcAudioEffects.java
>@@ -0,0 +1,237 @@
>+/*
>+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc.audio;
>+
>+import android.annotation.TargetApi;
>+import android.media.audiofx.AcousticEchoCanceler;
>+import android.media.audiofx.AudioEffect;
>+import android.media.audiofx.AudioEffect.Descriptor;
>+import android.media.audiofx.AutomaticGainControl;
>+import android.media.audiofx.NoiseSuppressor;
>+import android.os.Build;
>+import java.util.List;
>+import java.util.UUID;
>+import javax.annotation.Nullable;
>+import org.webrtc.Logging;
>+
>+// This class wraps control of three different platform effects. Supported
>+// effects are: AcousticEchoCanceler (AEC) and NoiseSuppressor (NS).
>+// Calling enable() will active all effects that are
>+// supported by the device if the corresponding |shouldEnableXXX| member is set.
>+class WebRtcAudioEffects {
>+  private static final boolean DEBUG = false;
>+
>+  private static final String TAG = "WebRtcAudioEffectsExternal";
>+
>+  // UUIDs for Software Audio Effects that we want to avoid using.
>+  // The implementor field will be set to "The Android Open Source Project".
>+  private static final UUID AOSP_ACOUSTIC_ECHO_CANCELER =
>+      UUID.fromString("bb392ec0-8d4d-11e0-a896-0002a5d5c51b");
>+  private static final UUID AOSP_NOISE_SUPPRESSOR =
>+      UUID.fromString("c06c8400-8e06-11e0-9cb6-0002a5d5c51b");
>+
>+  // Contains the available effect descriptors returned from the
>+  // AudioEffect.getEffects() call. This result is cached to avoid doing the
>+  // slow OS call multiple times.
>+  private static @Nullable Descriptor[] cachedEffects = null;
>+
>+  // Contains the audio effect objects. Created in enable() and destroyed
>+  // in release().
>+  private @Nullable AcousticEchoCanceler aec = null;
>+  private @Nullable NoiseSuppressor ns = null;
>+
>+  // Affects the final state given to the setEnabled() method on each effect.
>+  // The default state is set to "disabled" but each effect can also be enabled
>+  // by calling setAEC() and setNS().
>+  private boolean shouldEnableAec = false;
>+  private boolean shouldEnableNs = false;
>+
>+  // Returns true if all conditions for supporting HW Acoustic Echo Cancellation (AEC) are
>+  // fulfilled.
>+  @TargetApi(18)
>+  public static boolean isAcousticEchoCancelerSupported() {
>+    return isEffectTypeAvailable(AudioEffect.EFFECT_TYPE_AEC, AOSP_ACOUSTIC_ECHO_CANCELER);
>+  }
>+
>+  // Returns true if all conditions for supporting HW Noise Suppression (NS) are fulfilled.
>+  @TargetApi(18)
>+  public static boolean isNoiseSuppressorSupported() {
>+    return isEffectTypeAvailable(AudioEffect.EFFECT_TYPE_NS, AOSP_NOISE_SUPPRESSOR);
>+  }
>+
>+  public WebRtcAudioEffects() {
>+    Logging.d(TAG, "ctor" + WebRtcAudioUtils.getThreadInfo());
>+  }
>+
>+  // Call this method to enable or disable the platform AEC. It modifies
>+  // |shouldEnableAec| which is used in enable() where the actual state
>+  // of the AEC effect is modified. Returns true if HW AEC is supported and
>+  // false otherwise.
>+  public boolean setAEC(boolean enable) {
>+    Logging.d(TAG, "setAEC(" + enable + ")");
>+    if (!isAcousticEchoCancelerSupported()) {
>+      Logging.w(TAG, "Platform AEC is not supported");
>+      shouldEnableAec = false;
>+      return false;
>+    }
>+    if (aec != null && (enable != shouldEnableAec)) {
>+      Logging.e(TAG, "Platform AEC state can't be modified while recording");
>+      return false;
>+    }
>+    shouldEnableAec = enable;
>+    return true;
>+  }
>+
>+  // Call this method to enable or disable the platform NS. It modifies
>+  // |shouldEnableNs| which is used in enable() where the actual state
>+  // of the NS effect is modified. Returns true if HW NS is supported and
>+  // false otherwise.
>+  public boolean setNS(boolean enable) {
>+    Logging.d(TAG, "setNS(" + enable + ")");
>+    if (!isNoiseSuppressorSupported()) {
>+      Logging.w(TAG, "Platform NS is not supported");
>+      shouldEnableNs = false;
>+      return false;
>+    }
>+    if (ns != null && (enable != shouldEnableNs)) {
>+      Logging.e(TAG, "Platform NS state can't be modified while recording");
>+      return false;
>+    }
>+    shouldEnableNs = enable;
>+    return true;
>+  }
>+
>+  public void enable(int audioSession) {
>+    Logging.d(TAG, "enable(audioSession=" + audioSession + ")");
>+    assertTrue(aec == null);
>+    assertTrue(ns == null);
>+
>+    if (DEBUG) {
>+      // Add logging of supported effects but filter out "VoIP effects", i.e.,
>+      // AEC, AEC and NS. Avoid calling AudioEffect.queryEffects() unless the
>+      // DEBUG flag is set since we have seen crashes in this API.
>+      for (Descriptor d : AudioEffect.queryEffects()) {
>+        if (effectTypeIsVoIP(d.type)) {
>+          Logging.d(TAG,
>+              "name: " + d.name + ", "
>+                  + "mode: " + d.connectMode + ", "
>+                  + "implementor: " + d.implementor + ", "
>+                  + "UUID: " + d.uuid);
>+        }
>+      }
>+    }
>+
>+    if (isAcousticEchoCancelerSupported()) {
>+      // Create an AcousticEchoCanceler and attach it to the AudioRecord on
>+      // the specified audio session.
>+      aec = AcousticEchoCanceler.create(audioSession);
>+      if (aec != null) {
>+        boolean enabled = aec.getEnabled();
>+        boolean enable = shouldEnableAec && isAcousticEchoCancelerSupported();
>+        if (aec.setEnabled(enable) != AudioEffect.SUCCESS) {
>+          Logging.e(TAG, "Failed to set the AcousticEchoCanceler state");
>+        }
>+        Logging.d(TAG,
>+            "AcousticEchoCanceler: was " + (enabled ? "enabled" : "disabled") + ", enable: "
>+                + enable + ", is now: " + (aec.getEnabled() ? "enabled" : "disabled"));
>+      } else {
>+        Logging.e(TAG, "Failed to create the AcousticEchoCanceler instance");
>+      }
>+    }
>+
>+    if (isNoiseSuppressorSupported()) {
>+      // Create an NoiseSuppressor and attach it to the AudioRecord on the
>+      // specified audio session.
>+      ns = NoiseSuppressor.create(audioSession);
>+      if (ns != null) {
>+        boolean enabled = ns.getEnabled();
>+        boolean enable = shouldEnableNs && isNoiseSuppressorSupported();
>+        if (ns.setEnabled(enable) != AudioEffect.SUCCESS) {
>+          Logging.e(TAG, "Failed to set the NoiseSuppressor state");
>+        }
>+        Logging.d(TAG,
>+            "NoiseSuppressor: was " + (enabled ? "enabled" : "disabled") + ", enable: " + enable
>+                + ", is now: " + (ns.getEnabled() ? "enabled" : "disabled"));
>+      } else {
>+        Logging.e(TAG, "Failed to create the NoiseSuppressor instance");
>+      }
>+    }
>+  }
>+
>+  // Releases all native audio effect resources. It is a good practice to
>+  // release the effect engine when not in use as control can be returned
>+  // to other applications or the native resources released.
>+  public void release() {
>+    Logging.d(TAG, "release");
>+    if (aec != null) {
>+      aec.release();
>+      aec = null;
>+    }
>+    if (ns != null) {
>+      ns.release();
>+      ns = null;
>+    }
>+  }
>+
>+  // Returns true for effect types in |type| that are of "VoIP" types:
>+  // Acoustic Echo Canceler (AEC) or Automatic Gain Control (AGC) or
>+  // Noise Suppressor (NS). Note that, an extra check for support is needed
>+  // in each comparison since some devices includes effects in the
>+  // AudioEffect.Descriptor array that are actually not available on the device.
>+  // As an example: Samsung Galaxy S6 includes an AGC in the descriptor but
>+  // AutomaticGainControl.isAvailable() returns false.
>+  @TargetApi(18)
>+  private boolean effectTypeIsVoIP(UUID type) {
>+    if (!WebRtcAudioUtils.runningOnJellyBeanMR2OrHigher())
>+      return false;
>+
>+    return (AudioEffect.EFFECT_TYPE_AEC.equals(type) && isAcousticEchoCancelerSupported())
>+        || (AudioEffect.EFFECT_TYPE_NS.equals(type) && isNoiseSuppressorSupported());
>+  }
>+
>+  // Helper method which throws an exception when an assertion has failed.
>+  private static void assertTrue(boolean condition) {
>+    if (!condition) {
>+      throw new AssertionError("Expected condition to be true");
>+    }
>+  }
>+
>+  // Returns the cached copy of the audio effects array, if available, or
>+  // queries the operating system for the list of effects.
>+  private static @Nullable Descriptor[] getAvailableEffects() {
>+    if (cachedEffects != null) {
>+      return cachedEffects;
>+    }
>+    // The caching is best effort only - if this method is called from several
>+    // threads in parallel, they may end up doing the underlying OS call
>+    // multiple times. It's normally only called on one thread so there's no
>+    // real need to optimize for the multiple threads case.
>+    cachedEffects = AudioEffect.queryEffects();
>+    return cachedEffects;
>+  }
>+
>+  // Returns true if an effect of the specified type is available. Functionally
>+  // equivalent to (NoiseSuppressor|AutomaticGainControl|...).isAvailable(), but
>+  // faster as it avoids the expensive OS call to enumerate effects.
>+  @TargetApi(18)
>+  private static boolean isEffectTypeAvailable(UUID effectType, UUID blackListedUuid) {
>+    Descriptor[] effects = getAvailableEffects();
>+    if (effects == null) {
>+      return false;
>+    }
>+    for (Descriptor d : effects) {
>+      if (d.type.equals(effectType)) {
>+        return !d.uuid.equals(blackListedUuid);
>+      }
>+    }
>+    return false;
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/audio/WebRtcAudioManager.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/audio/WebRtcAudioManager.java
>new file mode 100644
>index 00000000000..5b810084cc8
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/audio/WebRtcAudioManager.java
>@@ -0,0 +1,129 @@
>+/*
>+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc.audio;
>+
>+import android.annotation.TargetApi;
>+import android.content.Context;
>+import android.content.pm.PackageManager;
>+import android.media.AudioFormat;
>+import android.media.AudioManager;
>+import android.media.AudioRecord;
>+import android.media.AudioTrack;
>+import org.webrtc.Logging;
>+import org.webrtc.CalledByNative;
>+
>+/**
>+ * This class contains static functions to query sample rate and input/output audio buffer sizes.
>+ */
>+class WebRtcAudioManager {
>+  private static final String TAG = "WebRtcAudioManagerExternal";
>+
>+  private static final int DEFAULT_SAMPLE_RATE_HZ = 16000;
>+
>+  // Default audio data format is PCM 16 bit per sample.
>+  // Guaranteed to be supported by all devices.
>+  private static final int BITS_PER_SAMPLE = 16;
>+
>+  private static final int DEFAULT_FRAME_PER_BUFFER = 256;
>+
>+  @CalledByNative
>+  static AudioManager getAudioManager(Context context) {
>+    return (AudioManager) context.getSystemService(Context.AUDIO_SERVICE);
>+  }
>+
>+  @CalledByNative
>+  static int getOutputBufferSize(
>+      Context context, AudioManager audioManager, int sampleRate, int numberOfOutputChannels) {
>+    return isLowLatencyOutputSupported(context)
>+        ? getLowLatencyFramesPerBuffer(audioManager)
>+        : getMinOutputFrameSize(sampleRate, numberOfOutputChannels);
>+  }
>+
>+  @CalledByNative
>+  static int getInputBufferSize(
>+      Context context, AudioManager audioManager, int sampleRate, int numberOfInputChannels) {
>+    return isLowLatencyInputSupported(context)
>+        ? getLowLatencyFramesPerBuffer(audioManager)
>+        : getMinInputFrameSize(sampleRate, numberOfInputChannels);
>+  }
>+
>+  private static boolean isLowLatencyOutputSupported(Context context) {
>+    return context.getPackageManager().hasSystemFeature(PackageManager.FEATURE_AUDIO_LOW_LATENCY);
>+  }
>+
>+  private static boolean isLowLatencyInputSupported(Context context) {
>+    // TODO(henrika): investigate if some sort of device list is needed here
>+    // as well. The NDK doc states that: "As of API level 21, lower latency
>+    // audio input is supported on select devices. To take advantage of this
>+    // feature, first confirm that lower latency output is available".
>+    return WebRtcAudioUtils.runningOnLollipopOrHigher() && isLowLatencyOutputSupported(context);
>+  }
>+
>+  /**
>+   * Returns the native input/output sample rate for this device's output stream.
>+   */
>+  @CalledByNative
>+  static int getSampleRate(AudioManager audioManager) {
>+    // Override this if we're running on an old emulator image which only
>+    // supports 8 kHz and doesn't support PROPERTY_OUTPUT_SAMPLE_RATE.
>+    if (WebRtcAudioUtils.runningOnEmulator()) {
>+      Logging.d(TAG, "Running emulator, overriding sample rate to 8 kHz.");
>+      return 8000;
>+    }
>+    // Deliver best possible estimate based on default Android AudioManager APIs.
>+    final int sampleRateHz = WebRtcAudioUtils.runningOnJellyBeanMR1OrHigher()
>+        ? getSampleRateOnJellyBeanMR10OrHigher(audioManager)
>+        : DEFAULT_SAMPLE_RATE_HZ;
>+    Logging.d(TAG, "Sample rate is set to " + sampleRateHz + " Hz");
>+    return sampleRateHz;
>+  }
>+
>+  @TargetApi(17)
>+  private static int getSampleRateOnJellyBeanMR10OrHigher(AudioManager audioManager) {
>+    String sampleRateString = audioManager.getProperty(AudioManager.PROPERTY_OUTPUT_SAMPLE_RATE);
>+    return (sampleRateString == null) ? DEFAULT_SAMPLE_RATE_HZ : Integer.parseInt(sampleRateString);
>+  }
>+
>+  // Returns the native output buffer size for low-latency output streams.
>+  @TargetApi(17)
>+  private static int getLowLatencyFramesPerBuffer(AudioManager audioManager) {
>+    if (!WebRtcAudioUtils.runningOnJellyBeanMR1OrHigher()) {
>+      return DEFAULT_FRAME_PER_BUFFER;
>+    }
>+    String framesPerBuffer =
>+        audioManager.getProperty(AudioManager.PROPERTY_OUTPUT_FRAMES_PER_BUFFER);
>+    return framesPerBuffer == null ? DEFAULT_FRAME_PER_BUFFER : Integer.parseInt(framesPerBuffer);
>+  }
>+
>+  // Returns the minimum output buffer size for Java based audio (AudioTrack).
>+  // This size can also be used for OpenSL ES implementations on devices that
>+  // lacks support of low-latency output.
>+  private static int getMinOutputFrameSize(int sampleRateInHz, int numChannels) {
>+    final int bytesPerFrame = numChannels * (BITS_PER_SAMPLE / 8);
>+    final int channelConfig =
>+        (numChannels == 1 ? AudioFormat.CHANNEL_OUT_MONO : AudioFormat.CHANNEL_OUT_STEREO);
>+    return AudioTrack.getMinBufferSize(
>+               sampleRateInHz, channelConfig, AudioFormat.ENCODING_PCM_16BIT)
>+        / bytesPerFrame;
>+  }
>+
>+  // Returns the minimum input buffer size for Java based audio (AudioRecord).
>+  // This size can calso be used for OpenSL ES implementations on devices that
>+  // lacks support of low-latency input.
>+  private static int getMinInputFrameSize(int sampleRateInHz, int numChannels) {
>+    final int bytesPerFrame = numChannels * (BITS_PER_SAMPLE / 8);
>+    final int channelConfig =
>+        (numChannels == 1 ? AudioFormat.CHANNEL_IN_MONO : AudioFormat.CHANNEL_IN_STEREO);
>+    return AudioRecord.getMinBufferSize(
>+               sampleRateInHz, channelConfig, AudioFormat.ENCODING_PCM_16BIT)
>+        / bytesPerFrame;
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/audio/WebRtcAudioRecord.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/audio/WebRtcAudioRecord.java
>new file mode 100644
>index 00000000000..66f619dfbff
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/audio/WebRtcAudioRecord.java
>@@ -0,0 +1,367 @@
>+/*
>+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc.audio;
>+
>+import android.annotation.TargetApi;
>+import android.content.Context;
>+import android.media.AudioFormat;
>+import android.media.AudioManager;
>+import android.media.AudioRecord;
>+import android.media.MediaRecorder.AudioSource;
>+import android.os.Process;
>+import java.lang.System;
>+import java.nio.ByteBuffer;
>+import java.util.Arrays;
>+import java.util.concurrent.TimeUnit;
>+import javax.annotation.Nullable;
>+import org.webrtc.CalledByNative;
>+import org.webrtc.Logging;
>+import org.webrtc.ThreadUtils;
>+import org.webrtc.audio.JavaAudioDeviceModule.AudioRecordErrorCallback;
>+import org.webrtc.audio.JavaAudioDeviceModule.AudioRecordStartErrorCode;
>+import org.webrtc.audio.JavaAudioDeviceModule.SamplesReadyCallback;
>+
>+class WebRtcAudioRecord {
>+  private static final String TAG = "WebRtcAudioRecordExternal";
>+
>+  // Default audio data format is PCM 16 bit per sample.
>+  // Guaranteed to be supported by all devices.
>+  private static final int BITS_PER_SAMPLE = 16;
>+
>+  // Requested size of each recorded buffer provided to the client.
>+  private static final int CALLBACK_BUFFER_SIZE_MS = 10;
>+
>+  // Average number of callbacks per second.
>+  private static final int BUFFERS_PER_SECOND = 1000 / CALLBACK_BUFFER_SIZE_MS;
>+
>+  // We ask for a native buffer size of BUFFER_SIZE_FACTOR * (minimum required
>+  // buffer size). The extra space is allocated to guard against glitches under
>+  // high load.
>+  private static final int BUFFER_SIZE_FACTOR = 2;
>+
>+  // The AudioRecordJavaThread is allowed to wait for successful call to join()
>+  // but the wait times out afther this amount of time.
>+  private static final long AUDIO_RECORD_THREAD_JOIN_TIMEOUT_MS = 2000;
>+
>+  public static final int DEFAULT_AUDIO_SOURCE = AudioSource.VOICE_COMMUNICATION;
>+
>+  private final Context context;
>+  private final AudioManager audioManager;
>+  private final int audioSource;
>+
>+  private long nativeAudioRecord;
>+
>+  private final WebRtcAudioEffects effects = new WebRtcAudioEffects();
>+
>+  private @Nullable ByteBuffer byteBuffer;
>+
>+  private @Nullable AudioRecord audioRecord = null;
>+  private @Nullable AudioRecordThread audioThread = null;
>+
>+  private volatile boolean microphoneMute = false;
>+  private byte[] emptyBytes;
>+
>+  private final @Nullable AudioRecordErrorCallback errorCallback;
>+  private final @Nullable SamplesReadyCallback audioSamplesReadyCallback;
>+  private final boolean isAcousticEchoCancelerSupported;
>+  private final boolean isNoiseSuppressorSupported;
>+
>+  /**
>+   * Audio thread which keeps calling ByteBuffer.read() waiting for audio
>+   * to be recorded. Feeds recorded data to the native counterpart as a
>+   * periodic sequence of callbacks using DataIsRecorded().
>+   * This thread uses a Process.THREAD_PRIORITY_URGENT_AUDIO priority.
>+   */
>+  private class AudioRecordThread extends Thread {
>+    private volatile boolean keepAlive = true;
>+
>+    public AudioRecordThread(String name) {
>+      super(name);
>+    }
>+
>+    @Override
>+    public void run() {
>+      Process.setThreadPriority(Process.THREAD_PRIORITY_URGENT_AUDIO);
>+      Logging.d(TAG, "AudioRecordThread" + WebRtcAudioUtils.getThreadInfo());
>+      assertTrue(audioRecord.getRecordingState() == AudioRecord.RECORDSTATE_RECORDING);
>+
>+      long lastTime = System.nanoTime();
>+      while (keepAlive) {
>+        int bytesRead = audioRecord.read(byteBuffer, byteBuffer.capacity());
>+        if (bytesRead == byteBuffer.capacity()) {
>+          if (microphoneMute) {
>+            byteBuffer.clear();
>+            byteBuffer.put(emptyBytes);
>+          }
>+          // It's possible we've been shut down during the read, and stopRecording() tried and
>+          // failed to join this thread. To be a bit safer, try to avoid calling any native methods
>+          // in case they've been unregistered after stopRecording() returned.
>+          if (keepAlive) {
>+            nativeDataIsRecorded(nativeAudioRecord, bytesRead);
>+          }
>+          if (audioSamplesReadyCallback != null) {
>+            // Copy the entire byte buffer array. The start of the byteBuffer is not necessarily
>+            // at index 0.
>+            byte[] data = Arrays.copyOfRange(byteBuffer.array(), byteBuffer.arrayOffset(),
>+                byteBuffer.capacity() + byteBuffer.arrayOffset());
>+            audioSamplesReadyCallback.onWebRtcAudioRecordSamplesReady(
>+                new JavaAudioDeviceModule.AudioSamples(audioRecord.getAudioFormat(),
>+                    audioRecord.getChannelCount(), audioRecord.getSampleRate(), data));
>+          }
>+        } else {
>+          String errorMessage = "AudioRecord.read failed: " + bytesRead;
>+          Logging.e(TAG, errorMessage);
>+          if (bytesRead == AudioRecord.ERROR_INVALID_OPERATION) {
>+            keepAlive = false;
>+            reportWebRtcAudioRecordError(errorMessage);
>+          }
>+        }
>+      }
>+
>+      try {
>+        if (audioRecord != null) {
>+          audioRecord.stop();
>+        }
>+      } catch (IllegalStateException e) {
>+        Logging.e(TAG, "AudioRecord.stop failed: " + e.getMessage());
>+      }
>+    }
>+
>+    // Stops the inner thread loop and also calls AudioRecord.stop().
>+    // Does not block the calling thread.
>+    public void stopThread() {
>+      Logging.d(TAG, "stopThread");
>+      keepAlive = false;
>+    }
>+  }
>+
>+  @CalledByNative
>+  WebRtcAudioRecord(Context context, AudioManager audioManager) {
>+    this(context, audioManager, DEFAULT_AUDIO_SOURCE, null /* errorCallback */,
>+        null /* audioSamplesReadyCallback */, WebRtcAudioEffects.isAcousticEchoCancelerSupported(),
>+        WebRtcAudioEffects.isNoiseSuppressorSupported());
>+  }
>+
>+  public WebRtcAudioRecord(Context context, AudioManager audioManager, int audioSource,
>+      @Nullable AudioRecordErrorCallback errorCallback,
>+      @Nullable SamplesReadyCallback audioSamplesReadyCallback,
>+      boolean isAcousticEchoCancelerSupported, boolean isNoiseSuppressorSupported) {
>+    if (isAcousticEchoCancelerSupported && !WebRtcAudioEffects.isAcousticEchoCancelerSupported()) {
>+      throw new IllegalArgumentException("HW AEC not supported");
>+    }
>+    if (isNoiseSuppressorSupported && !WebRtcAudioEffects.isNoiseSuppressorSupported()) {
>+      throw new IllegalArgumentException("HW NS not supported");
>+    }
>+    this.context = context;
>+    this.audioManager = audioManager;
>+    this.audioSource = audioSource;
>+    this.errorCallback = errorCallback;
>+    this.audioSamplesReadyCallback = audioSamplesReadyCallback;
>+    this.isAcousticEchoCancelerSupported = isAcousticEchoCancelerSupported;
>+    this.isNoiseSuppressorSupported = isNoiseSuppressorSupported;
>+  }
>+
>+  @CalledByNative
>+  public void setNativeAudioRecord(long nativeAudioRecord) {
>+    this.nativeAudioRecord = nativeAudioRecord;
>+  }
>+
>+  @CalledByNative
>+  boolean isAcousticEchoCancelerSupported() {
>+    return isAcousticEchoCancelerSupported;
>+  }
>+
>+  @CalledByNative
>+  boolean isNoiseSuppressorSupported() {
>+    return isNoiseSuppressorSupported;
>+  }
>+
>+  @CalledByNative
>+  private boolean enableBuiltInAEC(boolean enable) {
>+    Logging.d(TAG, "enableBuiltInAEC(" + enable + ")");
>+    return effects.setAEC(enable);
>+  }
>+
>+  @CalledByNative
>+  private boolean enableBuiltInNS(boolean enable) {
>+    Logging.d(TAG, "enableBuiltInNS(" + enable + ")");
>+    return effects.setNS(enable);
>+  }
>+
>+  @CalledByNative
>+  private int initRecording(int sampleRate, int channels) {
>+    Logging.d(TAG, "initRecording(sampleRate=" + sampleRate + ", channels=" + channels + ")");
>+    if (audioRecord != null) {
>+      reportWebRtcAudioRecordInitError("InitRecording called twice without StopRecording.");
>+      return -1;
>+    }
>+    final int bytesPerFrame = channels * (BITS_PER_SAMPLE / 8);
>+    final int framesPerBuffer = sampleRate / BUFFERS_PER_SECOND;
>+    byteBuffer = ByteBuffer.allocateDirect(bytesPerFrame * framesPerBuffer);
>+    if (!(byteBuffer.hasArray())) {
>+      reportWebRtcAudioRecordInitError("ByteBuffer does not have backing array.");
>+      return -1;
>+    }
>+    Logging.d(TAG, "byteBuffer.capacity: " + byteBuffer.capacity());
>+    emptyBytes = new byte[byteBuffer.capacity()];
>+    // Rather than passing the ByteBuffer with every callback (requiring
>+    // the potentially expensive GetDirectBufferAddress) we simply have the
>+    // the native class cache the address to the memory once.
>+    nativeCacheDirectBufferAddress(nativeAudioRecord, byteBuffer);
>+
>+    // Get the minimum buffer size required for the successful creation of
>+    // an AudioRecord object, in byte units.
>+    // Note that this size doesn't guarantee a smooth recording under load.
>+    final int channelConfig = channelCountToConfiguration(channels);
>+    int minBufferSize =
>+        AudioRecord.getMinBufferSize(sampleRate, channelConfig, AudioFormat.ENCODING_PCM_16BIT);
>+    if (minBufferSize == AudioRecord.ERROR || minBufferSize == AudioRecord.ERROR_BAD_VALUE) {
>+      reportWebRtcAudioRecordInitError("AudioRecord.getMinBufferSize failed: " + minBufferSize);
>+      return -1;
>+    }
>+    Logging.d(TAG, "AudioRecord.getMinBufferSize: " + minBufferSize);
>+
>+    // Use a larger buffer size than the minimum required when creating the
>+    // AudioRecord instance to ensure smooth recording under load. It has been
>+    // verified that it does not increase the actual recording latency.
>+    int bufferSizeInBytes = Math.max(BUFFER_SIZE_FACTOR * minBufferSize, byteBuffer.capacity());
>+    Logging.d(TAG, "bufferSizeInBytes: " + bufferSizeInBytes);
>+    try {
>+      audioRecord = new AudioRecord(audioSource, sampleRate, channelConfig,
>+          AudioFormat.ENCODING_PCM_16BIT, bufferSizeInBytes);
>+    } catch (IllegalArgumentException e) {
>+      reportWebRtcAudioRecordInitError("AudioRecord ctor error: " + e.getMessage());
>+      releaseAudioResources();
>+      return -1;
>+    }
>+    if (audioRecord == null || audioRecord.getState() != AudioRecord.STATE_INITIALIZED) {
>+      reportWebRtcAudioRecordInitError("Failed to create a new AudioRecord instance");
>+      releaseAudioResources();
>+      return -1;
>+    }
>+    effects.enable(audioRecord.getAudioSessionId());
>+    logMainParameters();
>+    logMainParametersExtended();
>+    return framesPerBuffer;
>+  }
>+
>+  @CalledByNative
>+  private boolean startRecording() {
>+    Logging.d(TAG, "startRecording");
>+    assertTrue(audioRecord != null);
>+    assertTrue(audioThread == null);
>+    try {
>+      audioRecord.startRecording();
>+    } catch (IllegalStateException e) {
>+      reportWebRtcAudioRecordStartError(AudioRecordStartErrorCode.AUDIO_RECORD_START_EXCEPTION,
>+          "AudioRecord.startRecording failed: " + e.getMessage());
>+      return false;
>+    }
>+    if (audioRecord.getRecordingState() != AudioRecord.RECORDSTATE_RECORDING) {
>+      reportWebRtcAudioRecordStartError(AudioRecordStartErrorCode.AUDIO_RECORD_START_STATE_MISMATCH,
>+          "AudioRecord.startRecording failed - incorrect state :"
>+              + audioRecord.getRecordingState());
>+      return false;
>+    }
>+    audioThread = new AudioRecordThread("AudioRecordJavaThread");
>+    audioThread.start();
>+    return true;
>+  }
>+
>+  @CalledByNative
>+  private boolean stopRecording() {
>+    Logging.d(TAG, "stopRecording");
>+    assertTrue(audioThread != null);
>+    audioThread.stopThread();
>+    if (!ThreadUtils.joinUninterruptibly(audioThread, AUDIO_RECORD_THREAD_JOIN_TIMEOUT_MS)) {
>+      Logging.e(TAG, "Join of AudioRecordJavaThread timed out");
>+      WebRtcAudioUtils.logAudioState(TAG, context, audioManager);
>+    }
>+    audioThread = null;
>+    effects.release();
>+    releaseAudioResources();
>+    return true;
>+  }
>+
>+  private void logMainParameters() {
>+    Logging.d(TAG,
>+        "AudioRecord: "
>+            + "session ID: " + audioRecord.getAudioSessionId() + ", "
>+            + "channels: " + audioRecord.getChannelCount() + ", "
>+            + "sample rate: " + audioRecord.getSampleRate());
>+  }
>+
>+  @TargetApi(23)
>+  private void logMainParametersExtended() {
>+    if (WebRtcAudioUtils.runningOnMarshmallowOrHigher()) {
>+      Logging.d(TAG,
>+          "AudioRecord: "
>+              // The frame count of the native AudioRecord buffer.
>+              + "buffer size in frames: " + audioRecord.getBufferSizeInFrames());
>+    }
>+  }
>+
>+  // Helper method which throws an exception  when an assertion has failed.
>+  private static void assertTrue(boolean condition) {
>+    if (!condition) {
>+      throw new AssertionError("Expected condition to be true");
>+    }
>+  }
>+
>+  private int channelCountToConfiguration(int channels) {
>+    return (channels == 1 ? AudioFormat.CHANNEL_IN_MONO : AudioFormat.CHANNEL_IN_STEREO);
>+  }
>+
>+  private native void nativeCacheDirectBufferAddress(
>+      long nativeAudioRecordJni, ByteBuffer byteBuffer);
>+  private native void nativeDataIsRecorded(long nativeAudioRecordJni, int bytes);
>+
>+  // Sets all recorded samples to zero if |mute| is true, i.e., ensures that
>+  // the microphone is muted.
>+  public void setMicrophoneMute(boolean mute) {
>+    Logging.w(TAG, "setMicrophoneMute(" + mute + ")");
>+    microphoneMute = mute;
>+  }
>+
>+  // Releases the native AudioRecord resources.
>+  private void releaseAudioResources() {
>+    Logging.d(TAG, "releaseAudioResources");
>+    if (audioRecord != null) {
>+      audioRecord.release();
>+      audioRecord = null;
>+    }
>+  }
>+
>+  private void reportWebRtcAudioRecordInitError(String errorMessage) {
>+    Logging.e(TAG, "Init recording error: " + errorMessage);
>+    WebRtcAudioUtils.logAudioState(TAG, context, audioManager);
>+    if (errorCallback != null) {
>+      errorCallback.onWebRtcAudioRecordInitError(errorMessage);
>+    }
>+  }
>+
>+  private void reportWebRtcAudioRecordStartError(
>+      AudioRecordStartErrorCode errorCode, String errorMessage) {
>+    Logging.e(TAG, "Start recording error: " + errorCode + ". " + errorMessage);
>+    WebRtcAudioUtils.logAudioState(TAG, context, audioManager);
>+    if (errorCallback != null) {
>+      errorCallback.onWebRtcAudioRecordStartError(errorCode, errorMessage);
>+    }
>+  }
>+
>+  private void reportWebRtcAudioRecordError(String errorMessage) {
>+    Logging.e(TAG, "Run-time recording error: " + errorMessage);
>+    WebRtcAudioUtils.logAudioState(TAG, context, audioManager);
>+    if (errorCallback != null) {
>+      errorCallback.onWebRtcAudioRecordError(errorMessage);
>+    }
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/audio/WebRtcAudioTrack.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/audio/WebRtcAudioTrack.java
>new file mode 100644
>index 00000000000..f3a38f0c236
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/audio/WebRtcAudioTrack.java
>@@ -0,0 +1,489 @@
>+/*
>+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc.audio;
>+
>+import android.annotation.SuppressLint;
>+import android.annotation.TargetApi;
>+import android.content.Context;
>+import android.media.AudioAttributes;
>+import android.media.AudioFormat;
>+import android.media.AudioManager;
>+import android.media.AudioTrack;
>+import android.os.Process;
>+import java.lang.Thread;
>+import java.nio.ByteBuffer;
>+import javax.annotation.Nullable;
>+import org.webrtc.Logging;
>+import org.webrtc.ThreadUtils;
>+import org.webrtc.audio.JavaAudioDeviceModule.AudioTrackErrorCallback;
>+import org.webrtc.audio.JavaAudioDeviceModule.AudioTrackStartErrorCode;
>+import org.webrtc.CalledByNative;
>+
>+class WebRtcAudioTrack {
>+  private static final String TAG = "WebRtcAudioTrackExternal";
>+
>+  // Default audio data format is PCM 16 bit per sample.
>+  // Guaranteed to be supported by all devices.
>+  private static final int BITS_PER_SAMPLE = 16;
>+
>+  // Requested size of each recorded buffer provided to the client.
>+  private static final int CALLBACK_BUFFER_SIZE_MS = 10;
>+
>+  // Average number of callbacks per second.
>+  private static final int BUFFERS_PER_SECOND = 1000 / CALLBACK_BUFFER_SIZE_MS;
>+
>+  // The AudioTrackThread is allowed to wait for successful call to join()
>+  // but the wait times out afther this amount of time.
>+  private static final long AUDIO_TRACK_THREAD_JOIN_TIMEOUT_MS = 2000;
>+
>+  // By default, WebRTC creates audio tracks with a usage attribute
>+  // corresponding to voice communications, such as telephony or VoIP.
>+  private static final int DEFAULT_USAGE = getDefaultUsageAttribute();
>+
>+  private static int getDefaultUsageAttribute() {
>+    if (WebRtcAudioUtils.runningOnLollipopOrHigher()) {
>+      return getDefaultUsageAttributeOnLollipopOrHigher();
>+    } else {
>+      // Not used on SDKs lower than L.
>+      return 0;
>+    }
>+  }
>+
>+  @TargetApi(21)
>+  private static int getDefaultUsageAttributeOnLollipopOrHigher() {
>+    return AudioAttributes.USAGE_VOICE_COMMUNICATION;
>+  }
>+
>+  private long nativeAudioTrack;
>+  private final Context context;
>+  private final AudioManager audioManager;
>+  private final ThreadUtils.ThreadChecker threadChecker = new ThreadUtils.ThreadChecker();
>+
>+  private ByteBuffer byteBuffer;
>+
>+  private @Nullable AudioTrack audioTrack = null;
>+  private @Nullable AudioTrackThread audioThread = null;
>+  private final VolumeLogger volumeLogger;
>+
>+  // Samples to be played are replaced by zeros if |speakerMute| is set to true.
>+  // Can be used to ensure that the speaker is fully muted.
>+  private volatile boolean speakerMute = false;
>+  private byte[] emptyBytes;
>+
>+  private final @Nullable AudioTrackErrorCallback errorCallback;
>+
>+  /**
>+   * Audio thread which keeps calling AudioTrack.write() to stream audio.
>+   * Data is periodically acquired from the native WebRTC layer using the
>+   * nativeGetPlayoutData callback function.
>+   * This thread uses a Process.THREAD_PRIORITY_URGENT_AUDIO priority.
>+   */
>+  private class AudioTrackThread extends Thread {
>+    private volatile boolean keepAlive = true;
>+
>+    public AudioTrackThread(String name) {
>+      super(name);
>+    }
>+
>+    @Override
>+    public void run() {
>+      Process.setThreadPriority(Process.THREAD_PRIORITY_URGENT_AUDIO);
>+      Logging.d(TAG, "AudioTrackThread" + WebRtcAudioUtils.getThreadInfo());
>+      assertTrue(audioTrack.getPlayState() == AudioTrack.PLAYSTATE_PLAYING);
>+
>+      // Fixed size in bytes of each 10ms block of audio data that we ask for
>+      // using callbacks to the native WebRTC client.
>+      final int sizeInBytes = byteBuffer.capacity();
>+
>+      while (keepAlive) {
>+        // Get 10ms of PCM data from the native WebRTC client. Audio data is
>+        // written into the common ByteBuffer using the address that was
>+        // cached at construction.
>+        nativeGetPlayoutData(nativeAudioTrack, sizeInBytes);
>+        // Write data until all data has been written to the audio sink.
>+        // Upon return, the buffer position will have been advanced to reflect
>+        // the amount of data that was successfully written to the AudioTrack.
>+        assertTrue(sizeInBytes <= byteBuffer.remaining());
>+        if (speakerMute) {
>+          byteBuffer.clear();
>+          byteBuffer.put(emptyBytes);
>+          byteBuffer.position(0);
>+        }
>+        int bytesWritten = 0;
>+        if (WebRtcAudioUtils.runningOnLollipopOrHigher()) {
>+          bytesWritten = writeOnLollipop(audioTrack, byteBuffer, sizeInBytes);
>+        } else {
>+          bytesWritten = writePreLollipop(audioTrack, byteBuffer, sizeInBytes);
>+        }
>+        if (bytesWritten != sizeInBytes) {
>+          Logging.e(TAG, "AudioTrack.write played invalid number of bytes: " + bytesWritten);
>+          // If a write() returns a negative value, an error has occurred.
>+          // Stop playing and report an error in this case.
>+          if (bytesWritten < 0) {
>+            keepAlive = false;
>+            reportWebRtcAudioTrackError("AudioTrack.write failed: " + bytesWritten);
>+          }
>+        }
>+        // The byte buffer must be rewinded since byteBuffer.position() is
>+        // increased at each call to AudioTrack.write(). If we don't do this,
>+        // next call to AudioTrack.write() will fail.
>+        byteBuffer.rewind();
>+
>+        // TODO(henrika): it is possible to create a delay estimate here by
>+        // counting number of written frames and subtracting the result from
>+        // audioTrack.getPlaybackHeadPosition().
>+      }
>+
>+      // Stops playing the audio data. Since the instance was created in
>+      // MODE_STREAM mode, audio will stop playing after the last buffer that
>+      // was written has been played.
>+      if (audioTrack != null) {
>+        Logging.d(TAG, "Calling AudioTrack.stop...");
>+        try {
>+          audioTrack.stop();
>+          Logging.d(TAG, "AudioTrack.stop is done.");
>+        } catch (IllegalStateException e) {
>+          Logging.e(TAG, "AudioTrack.stop failed: " + e.getMessage());
>+        }
>+      }
>+    }
>+
>+    @TargetApi(21)
>+    private int writeOnLollipop(AudioTrack audioTrack, ByteBuffer byteBuffer, int sizeInBytes) {
>+      return audioTrack.write(byteBuffer, sizeInBytes, AudioTrack.WRITE_BLOCKING);
>+    }
>+
>+    private int writePreLollipop(AudioTrack audioTrack, ByteBuffer byteBuffer, int sizeInBytes) {
>+      return audioTrack.write(byteBuffer.array(), byteBuffer.arrayOffset(), sizeInBytes);
>+    }
>+
>+    // Stops the inner thread loop which results in calling AudioTrack.stop().
>+    // Does not block the calling thread.
>+    public void stopThread() {
>+      Logging.d(TAG, "stopThread");
>+      keepAlive = false;
>+    }
>+  }
>+
>+  @CalledByNative
>+  WebRtcAudioTrack(Context context, AudioManager audioManager) {
>+    this(context, audioManager, null /* errorCallback */);
>+  }
>+
>+  WebRtcAudioTrack(
>+      Context context, AudioManager audioManager, @Nullable AudioTrackErrorCallback errorCallback) {
>+    threadChecker.detachThread();
>+    this.context = context;
>+    this.audioManager = audioManager;
>+    this.errorCallback = errorCallback;
>+    this.volumeLogger = new VolumeLogger(audioManager);
>+  }
>+
>+  @CalledByNative
>+  public void setNativeAudioTrack(long nativeAudioTrack) {
>+    this.nativeAudioTrack = nativeAudioTrack;
>+  }
>+
>+  @CalledByNative
>+  private boolean initPlayout(int sampleRate, int channels) {
>+    threadChecker.checkIsOnValidThread();
>+    Logging.d(TAG, "initPlayout(sampleRate=" + sampleRate + ", channels=" + channels + ")");
>+    final int bytesPerFrame = channels * (BITS_PER_SAMPLE / 8);
>+    byteBuffer = ByteBuffer.allocateDirect(bytesPerFrame * (sampleRate / BUFFERS_PER_SECOND));
>+    Logging.d(TAG, "byteBuffer.capacity: " + byteBuffer.capacity());
>+    emptyBytes = new byte[byteBuffer.capacity()];
>+    // Rather than passing the ByteBuffer with every callback (requiring
>+    // the potentially expensive GetDirectBufferAddress) we simply have the
>+    // the native class cache the address to the memory once.
>+    nativeCacheDirectBufferAddress(nativeAudioTrack, byteBuffer);
>+
>+    // Get the minimum buffer size required for the successful creation of an
>+    // AudioTrack object to be created in the MODE_STREAM mode.
>+    // Note that this size doesn't guarantee a smooth playback under load.
>+    // TODO(henrika): should we extend the buffer size to avoid glitches?
>+    final int channelConfig = channelCountToConfiguration(channels);
>+    final int minBufferSizeInBytes =
>+        AudioTrack.getMinBufferSize(sampleRate, channelConfig, AudioFormat.ENCODING_PCM_16BIT);
>+    Logging.d(TAG, "AudioTrack.getMinBufferSize: " + minBufferSizeInBytes);
>+    // For the streaming mode, data must be written to the audio sink in
>+    // chunks of size (given by byteBuffer.capacity()) less than or equal
>+    // to the total buffer size |minBufferSizeInBytes|. But, we have seen
>+    // reports of "getMinBufferSize(): error querying hardware". Hence, it
>+    // can happen that |minBufferSizeInBytes| contains an invalid value.
>+    if (minBufferSizeInBytes < byteBuffer.capacity()) {
>+      reportWebRtcAudioTrackInitError("AudioTrack.getMinBufferSize returns an invalid value.");
>+      return false;
>+    }
>+
>+    // Ensure that prevision audio session was stopped correctly before trying
>+    // to create a new AudioTrack.
>+    if (audioTrack != null) {
>+      reportWebRtcAudioTrackInitError("Conflict with existing AudioTrack.");
>+      return false;
>+    }
>+    try {
>+      // Create an AudioTrack object and initialize its associated audio buffer.
>+      // The size of this buffer determines how long an AudioTrack can play
>+      // before running out of data.
>+      if (WebRtcAudioUtils.runningOnLollipopOrHigher()) {
>+        // If we are on API level 21 or higher, it is possible to use a special AudioTrack
>+        // constructor that uses AudioAttributes and AudioFormat as input. It allows us to
>+        // supersede the notion of stream types for defining the behavior of audio playback,
>+        // and to allow certain platforms or routing policies to use this information for more
>+        // refined volume or routing decisions.
>+        audioTrack =
>+            createAudioTrackOnLollipopOrHigher(sampleRate, channelConfig, minBufferSizeInBytes);
>+      } else {
>+        // Use default constructor for API levels below 21.
>+        audioTrack =
>+            createAudioTrackOnLowerThanLollipop(sampleRate, channelConfig, minBufferSizeInBytes);
>+      }
>+    } catch (IllegalArgumentException e) {
>+      reportWebRtcAudioTrackInitError(e.getMessage());
>+      releaseAudioResources();
>+      return false;
>+    }
>+
>+    // It can happen that an AudioTrack is created but it was not successfully
>+    // initialized upon creation. Seems to be the case e.g. when the maximum
>+    // number of globally available audio tracks is exceeded.
>+    if (audioTrack == null || audioTrack.getState() != AudioTrack.STATE_INITIALIZED) {
>+      reportWebRtcAudioTrackInitError("Initialization of audio track failed.");
>+      releaseAudioResources();
>+      return false;
>+    }
>+    logMainParameters();
>+    logMainParametersExtended();
>+    return true;
>+  }
>+
>+  @CalledByNative
>+  private boolean startPlayout() {
>+    threadChecker.checkIsOnValidThread();
>+    volumeLogger.start();
>+    Logging.d(TAG, "startPlayout");
>+    assertTrue(audioTrack != null);
>+    assertTrue(audioThread == null);
>+
>+    // Starts playing an audio track.
>+    try {
>+      audioTrack.play();
>+    } catch (IllegalStateException e) {
>+      reportWebRtcAudioTrackStartError(AudioTrackStartErrorCode.AUDIO_TRACK_START_EXCEPTION,
>+          "AudioTrack.play failed: " + e.getMessage());
>+      releaseAudioResources();
>+      return false;
>+    }
>+    if (audioTrack.getPlayState() != AudioTrack.PLAYSTATE_PLAYING) {
>+      reportWebRtcAudioTrackStartError(AudioTrackStartErrorCode.AUDIO_TRACK_START_STATE_MISMATCH,
>+          "AudioTrack.play failed - incorrect state :" + audioTrack.getPlayState());
>+      releaseAudioResources();
>+      return false;
>+    }
>+
>+    // Create and start new high-priority thread which calls AudioTrack.write()
>+    // and where we also call the native nativeGetPlayoutData() callback to
>+    // request decoded audio from WebRTC.
>+    audioThread = new AudioTrackThread("AudioTrackJavaThread");
>+    audioThread.start();
>+    return true;
>+  }
>+
>+  @CalledByNative
>+  private boolean stopPlayout() {
>+    threadChecker.checkIsOnValidThread();
>+    volumeLogger.stop();
>+    Logging.d(TAG, "stopPlayout");
>+    assertTrue(audioThread != null);
>+    logUnderrunCount();
>+    audioThread.stopThread();
>+
>+    Logging.d(TAG, "Stopping the AudioTrackThread...");
>+    audioThread.interrupt();
>+    if (!ThreadUtils.joinUninterruptibly(audioThread, AUDIO_TRACK_THREAD_JOIN_TIMEOUT_MS)) {
>+      Logging.e(TAG, "Join of AudioTrackThread timed out.");
>+      WebRtcAudioUtils.logAudioState(TAG, context, audioManager);
>+    }
>+    Logging.d(TAG, "AudioTrackThread has now been stopped.");
>+    audioThread = null;
>+    releaseAudioResources();
>+    return true;
>+  }
>+
>+  // Get max possible volume index for a phone call audio stream.
>+  @CalledByNative
>+  private int getStreamMaxVolume() {
>+    threadChecker.checkIsOnValidThread();
>+    Logging.d(TAG, "getStreamMaxVolume");
>+    return audioManager.getStreamMaxVolume(AudioManager.STREAM_VOICE_CALL);
>+  }
>+
>+  // Set current volume level for a phone call audio stream.
>+  @CalledByNative
>+  private boolean setStreamVolume(int volume) {
>+    threadChecker.checkIsOnValidThread();
>+    Logging.d(TAG, "setStreamVolume(" + volume + ")");
>+    if (isVolumeFixed()) {
>+      Logging.e(TAG, "The device implements a fixed volume policy.");
>+      return false;
>+    }
>+    audioManager.setStreamVolume(AudioManager.STREAM_VOICE_CALL, volume, 0);
>+    return true;
>+  }
>+
>+  // TODO(bugs.webrtc.org/8580): Call requires API level 21 (current min is 16):
>+  // `android.media.AudioManager#isVolumeFixed`: NewApi [warning]
>+  @SuppressLint("NewApi")
>+  private boolean isVolumeFixed() {
>+    if (!WebRtcAudioUtils.runningOnLollipopOrHigher())
>+      return false;
>+    return audioManager.isVolumeFixed();
>+  }
>+
>+  /** Get current volume level for a phone call audio stream. */
>+  @CalledByNative
>+  private int getStreamVolume() {
>+    threadChecker.checkIsOnValidThread();
>+    Logging.d(TAG, "getStreamVolume");
>+    return audioManager.getStreamVolume(AudioManager.STREAM_VOICE_CALL);
>+  }
>+
>+  private void logMainParameters() {
>+    Logging.d(TAG,
>+        "AudioTrack: "
>+            + "session ID: " + audioTrack.getAudioSessionId() + ", "
>+            + "channels: " + audioTrack.getChannelCount() + ", "
>+            + "sample rate: " + audioTrack.getSampleRate()
>+            + ", "
>+            // Gain (>=1.0) expressed as linear multiplier on sample values.
>+            + "max gain: " + AudioTrack.getMaxVolume());
>+  }
>+
>+  // Creates and AudioTrack instance using AudioAttributes and AudioFormat as input.
>+  // It allows certain platforms or routing policies to use this information for more
>+  // refined volume or routing decisions.
>+  @TargetApi(21)
>+  private static AudioTrack createAudioTrackOnLollipopOrHigher(
>+      int sampleRateInHz, int channelConfig, int bufferSizeInBytes) {
>+    Logging.d(TAG, "createAudioTrackOnLollipopOrHigher");
>+    // TODO(henrika): use setPerformanceMode(int) with PERFORMANCE_MODE_LOW_LATENCY to control
>+    // performance when Android O is supported. Add some logging in the mean time.
>+    final int nativeOutputSampleRate =
>+        AudioTrack.getNativeOutputSampleRate(AudioManager.STREAM_VOICE_CALL);
>+    Logging.d(TAG, "nativeOutputSampleRate: " + nativeOutputSampleRate);
>+    if (sampleRateInHz != nativeOutputSampleRate) {
>+      Logging.w(TAG, "Unable to use fast mode since requested sample rate is not native");
>+    }
>+    // Create an audio track where the audio usage is for VoIP and the content type is speech.
>+    return new AudioTrack(new AudioAttributes.Builder()
>+                              .setUsage(DEFAULT_USAGE)
>+                              .setContentType(AudioAttributes.CONTENT_TYPE_SPEECH)
>+                              .build(),
>+        new AudioFormat.Builder()
>+            .setEncoding(AudioFormat.ENCODING_PCM_16BIT)
>+            .setSampleRate(sampleRateInHz)
>+            .setChannelMask(channelConfig)
>+            .build(),
>+        bufferSizeInBytes, AudioTrack.MODE_STREAM, AudioManager.AUDIO_SESSION_ID_GENERATE);
>+  }
>+
>+  @SuppressWarnings("deprecation") // Deprecated in API level 25.
>+  private static AudioTrack createAudioTrackOnLowerThanLollipop(
>+      int sampleRateInHz, int channelConfig, int bufferSizeInBytes) {
>+    return new AudioTrack(AudioManager.STREAM_VOICE_CALL, sampleRateInHz, channelConfig,
>+        AudioFormat.ENCODING_PCM_16BIT, bufferSizeInBytes, AudioTrack.MODE_STREAM);
>+  }
>+
>+  @TargetApi(24)
>+  private void logMainParametersExtended() {
>+    if (WebRtcAudioUtils.runningOnMarshmallowOrHigher()) {
>+      Logging.d(TAG,
>+          "AudioTrack: "
>+              // The effective size of the AudioTrack buffer that the app writes to.
>+              + "buffer size in frames: " + audioTrack.getBufferSizeInFrames());
>+    }
>+    if (WebRtcAudioUtils.runningOnNougatOrHigher()) {
>+      Logging.d(TAG,
>+          "AudioTrack: "
>+              // Maximum size of the AudioTrack buffer in frames.
>+              + "buffer capacity in frames: " + audioTrack.getBufferCapacityInFrames());
>+    }
>+  }
>+
>+  // Prints the number of underrun occurrences in the application-level write
>+  // buffer since the AudioTrack was created. An underrun occurs if the app does
>+  // not write audio data quickly enough, causing the buffer to underflow and a
>+  // potential audio glitch.
>+  // TODO(henrika): keep track of this value in the field and possibly add new
>+  // UMA stat if needed.
>+  @TargetApi(24)
>+  private void logUnderrunCount() {
>+    if (WebRtcAudioUtils.runningOnNougatOrHigher()) {
>+      Logging.d(TAG, "underrun count: " + audioTrack.getUnderrunCount());
>+    }
>+  }
>+
>+  // Helper method which throws an exception  when an assertion has failed.
>+  private static void assertTrue(boolean condition) {
>+    if (!condition) {
>+      throw new AssertionError("Expected condition to be true");
>+    }
>+  }
>+
>+  private int channelCountToConfiguration(int channels) {
>+    return (channels == 1 ? AudioFormat.CHANNEL_OUT_MONO : AudioFormat.CHANNEL_OUT_STEREO);
>+  }
>+
>+  private static native void nativeCacheDirectBufferAddress(
>+      long nativeAudioTrackJni, ByteBuffer byteBuffer);
>+  private static native void nativeGetPlayoutData(long nativeAudioTrackJni, int bytes);
>+
>+  // Sets all samples to be played out to zero if |mute| is true, i.e.,
>+  // ensures that the speaker is muted.
>+  public void setSpeakerMute(boolean mute) {
>+    Logging.w(TAG, "setSpeakerMute(" + mute + ")");
>+    speakerMute = mute;
>+  }
>+
>+  // Releases the native AudioTrack resources.
>+  private void releaseAudioResources() {
>+    Logging.d(TAG, "releaseAudioResources");
>+    if (audioTrack != null) {
>+      audioTrack.release();
>+      audioTrack = null;
>+    }
>+  }
>+
>+  private void reportWebRtcAudioTrackInitError(String errorMessage) {
>+    Logging.e(TAG, "Init playout error: " + errorMessage);
>+    WebRtcAudioUtils.logAudioState(TAG, context, audioManager);
>+    if (errorCallback != null) {
>+      errorCallback.onWebRtcAudioTrackInitError(errorMessage);
>+    }
>+  }
>+
>+  private void reportWebRtcAudioTrackStartError(
>+      AudioTrackStartErrorCode errorCode, String errorMessage) {
>+    Logging.e(TAG, "Start playout error: " + errorCode + ". " + errorMessage);
>+    WebRtcAudioUtils.logAudioState(TAG, context, audioManager);
>+    if (errorCallback != null) {
>+      errorCallback.onWebRtcAudioTrackStartError(errorCode, errorMessage);
>+    }
>+  }
>+
>+  private void reportWebRtcAudioTrackError(String errorMessage) {
>+    Logging.e(TAG, "Run-time playback error: " + errorMessage);
>+    WebRtcAudioUtils.logAudioState(TAG, context, audioManager);
>+    if (errorCallback != null) {
>+      errorCallback.onWebRtcAudioTrackError(errorMessage);
>+    }
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/audio/WebRtcAudioUtils.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/audio/WebRtcAudioUtils.java
>new file mode 100644
>index 00000000000..052e26c5cb0
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/audio/WebRtcAudioUtils.java
>@@ -0,0 +1,280 @@
>+/*
>+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc.audio;
>+
>+import static android.media.AudioManager.MODE_IN_CALL;
>+import static android.media.AudioManager.MODE_IN_COMMUNICATION;
>+import static android.media.AudioManager.MODE_NORMAL;
>+import static android.media.AudioManager.MODE_RINGTONE;
>+
>+import android.annotation.SuppressLint;
>+import android.annotation.TargetApi;
>+import android.content.Context;
>+import android.content.pm.PackageManager;
>+import android.media.AudioDeviceInfo;
>+import android.media.AudioManager;
>+import android.media.AudioRecordingConfiguration;
>+import android.media.MediaRecorder.AudioSource;
>+import android.os.Build;
>+import android.os.Process;
>+import java.lang.Thread;
>+import java.util.Arrays;
>+import java.util.Iterator;
>+import java.util.List;
>+import org.webrtc.Logging;
>+
>+final class WebRtcAudioUtils {
>+  private static final String TAG = "WebRtcAudioUtilsExternal";
>+
>+  public static boolean runningOnJellyBeanMR1OrHigher() {
>+    // November 2012: Android 4.2. API Level 17.
>+    return Build.VERSION.SDK_INT >= Build.VERSION_CODES.JELLY_BEAN_MR1;
>+  }
>+
>+  public static boolean runningOnJellyBeanMR2OrHigher() {
>+    // July 24, 2013: Android 4.3. API Level 18.
>+    return Build.VERSION.SDK_INT >= Build.VERSION_CODES.JELLY_BEAN_MR2;
>+  }
>+
>+  public static boolean runningOnLollipopOrHigher() {
>+    // API Level 21.
>+    return Build.VERSION.SDK_INT >= Build.VERSION_CODES.LOLLIPOP;
>+  }
>+
>+  public static boolean runningOnMarshmallowOrHigher() {
>+    // API Level 23.
>+    return Build.VERSION.SDK_INT >= Build.VERSION_CODES.M;
>+  }
>+
>+  public static boolean runningOnNougatOrHigher() {
>+    // API Level 24.
>+    return Build.VERSION.SDK_INT >= Build.VERSION_CODES.N;
>+  }
>+
>+  public static boolean runningOnOreoOrHigher() {
>+    // API Level 26.
>+    return Build.VERSION.SDK_INT >= Build.VERSION_CODES.O;
>+  }
>+
>+  public static boolean runningOnOreoMR1OrHigher() {
>+    // API Level 27.
>+    return Build.VERSION.SDK_INT >= Build.VERSION_CODES.O_MR1;
>+  }
>+
>+  // Helper method for building a string of thread information.
>+  public static String getThreadInfo() {
>+    return "@[name=" + Thread.currentThread().getName() + ", id=" + Thread.currentThread().getId()
>+        + "]";
>+  }
>+
>+  // Returns true if we're running on emulator.
>+  public static boolean runningOnEmulator() {
>+    return Build.HARDWARE.equals("goldfish") && Build.BRAND.startsWith("generic_");
>+  }
>+
>+  // Information about the current build, taken from system properties.
>+  static void logDeviceInfo(String tag) {
>+    Logging.d(tag,
>+        "Android SDK: " + Build.VERSION.SDK_INT + ", "
>+            + "Release: " + Build.VERSION.RELEASE + ", "
>+            + "Brand: " + Build.BRAND + ", "
>+            + "Device: " + Build.DEVICE + ", "
>+            + "Id: " + Build.ID + ", "
>+            + "Hardware: " + Build.HARDWARE + ", "
>+            + "Manufacturer: " + Build.MANUFACTURER + ", "
>+            + "Model: " + Build.MODEL + ", "
>+            + "Product: " + Build.PRODUCT);
>+  }
>+
>+  // Logs information about the current audio state. The idea is to call this
>+  // method when errors are detected to log under what conditions the error
>+  // occurred. Hopefully it will provide clues to what might be the root cause.
>+  static void logAudioState(String tag, Context context, AudioManager audioManager) {
>+    logDeviceInfo(tag);
>+    logAudioStateBasic(tag, context, audioManager);
>+    logAudioStateVolume(tag, audioManager);
>+    logAudioDeviceInfo(tag, audioManager);
>+  }
>+
>+  // Reports basic audio statistics.
>+  private static void logAudioStateBasic(String tag, Context context, AudioManager audioManager) {
>+    Logging.d(tag,
>+        "Audio State: "
>+            + "audio mode: " + modeToString(audioManager.getMode()) + ", "
>+            + "has mic: " + hasMicrophone(context) + ", "
>+            + "mic muted: " + audioManager.isMicrophoneMute() + ", "
>+            + "music active: " + audioManager.isMusicActive() + ", "
>+            + "speakerphone: " + audioManager.isSpeakerphoneOn() + ", "
>+            + "BT SCO: " + audioManager.isBluetoothScoOn());
>+  }
>+
>+  // TODO(bugs.webrtc.org/8580): Call requires API level 21 (current min is 16):
>+  // `android.media.AudioManager#isVolumeFixed`: NewApi [warning]
>+  @SuppressLint("NewApi")
>+  // Adds volume information for all possible stream types.
>+  private static void logAudioStateVolume(String tag, AudioManager audioManager) {
>+    final int[] streams = {AudioManager.STREAM_VOICE_CALL, AudioManager.STREAM_MUSIC,
>+        AudioManager.STREAM_RING, AudioManager.STREAM_ALARM, AudioManager.STREAM_NOTIFICATION,
>+        AudioManager.STREAM_SYSTEM};
>+    Logging.d(tag, "Audio State: ");
>+    boolean fixedVolume = false;
>+    if (WebRtcAudioUtils.runningOnLollipopOrHigher()) {
>+      fixedVolume = audioManager.isVolumeFixed();
>+      // Some devices may not have volume controls and might use a fixed volume.
>+      Logging.d(tag, "  fixed volume=" + fixedVolume);
>+    }
>+    if (!fixedVolume) {
>+      for (int stream : streams) {
>+        StringBuilder info = new StringBuilder();
>+        info.append("  " + streamTypeToString(stream) + ": ");
>+        info.append("volume=").append(audioManager.getStreamVolume(stream));
>+        info.append(", max=").append(audioManager.getStreamMaxVolume(stream));
>+        logIsStreamMute(tag, audioManager, stream, info);
>+        Logging.d(tag, info.toString());
>+      }
>+    }
>+  }
>+
>+  @TargetApi(23)
>+  private static void logIsStreamMute(
>+      String tag, AudioManager audioManager, int stream, StringBuilder info) {
>+    if (WebRtcAudioUtils.runningOnMarshmallowOrHigher()) {
>+      info.append(", muted=").append(audioManager.isStreamMute(stream));
>+    }
>+  }
>+
>+  @TargetApi(23)
>+  private static void logAudioDeviceInfo(String tag, AudioManager audioManager) {
>+    if (!WebRtcAudioUtils.runningOnMarshmallowOrHigher()) {
>+      return;
>+    }
>+    final AudioDeviceInfo[] devices = audioManager.getDevices(AudioManager.GET_DEVICES_ALL);
>+    if (devices.length == 0) {
>+      return;
>+    }
>+    Logging.d(tag, "Audio Devices: ");
>+    for (AudioDeviceInfo device : devices) {
>+      StringBuilder info = new StringBuilder();
>+      info.append("  ").append(deviceTypeToString(device.getType()));
>+      info.append(device.isSource() ? "(in): " : "(out): ");
>+      // An empty array indicates that the device supports arbitrary channel counts.
>+      if (device.getChannelCounts().length > 0) {
>+        info.append("channels=").append(Arrays.toString(device.getChannelCounts()));
>+        info.append(", ");
>+      }
>+      if (device.getEncodings().length > 0) {
>+        // Examples: ENCODING_PCM_16BIT = 2, ENCODING_PCM_FLOAT = 4.
>+        info.append("encodings=").append(Arrays.toString(device.getEncodings()));
>+        info.append(", ");
>+      }
>+      if (device.getSampleRates().length > 0) {
>+        info.append("sample rates=").append(Arrays.toString(device.getSampleRates()));
>+        info.append(", ");
>+      }
>+      info.append("id=").append(device.getId());
>+      Logging.d(tag, info.toString());
>+    }
>+  }
>+
>+  // Converts media.AudioManager modes into local string representation.
>+  static String modeToString(int mode) {
>+    switch (mode) {
>+      case MODE_IN_CALL:
>+        return "MODE_IN_CALL";
>+      case MODE_IN_COMMUNICATION:
>+        return "MODE_IN_COMMUNICATION";
>+      case MODE_NORMAL:
>+        return "MODE_NORMAL";
>+      case MODE_RINGTONE:
>+        return "MODE_RINGTONE";
>+      default:
>+        return "MODE_INVALID";
>+    }
>+  }
>+
>+  private static String streamTypeToString(int stream) {
>+    switch (stream) {
>+      case AudioManager.STREAM_VOICE_CALL:
>+        return "STREAM_VOICE_CALL";
>+      case AudioManager.STREAM_MUSIC:
>+        return "STREAM_MUSIC";
>+      case AudioManager.STREAM_RING:
>+        return "STREAM_RING";
>+      case AudioManager.STREAM_ALARM:
>+        return "STREAM_ALARM";
>+      case AudioManager.STREAM_NOTIFICATION:
>+        return "STREAM_NOTIFICATION";
>+      case AudioManager.STREAM_SYSTEM:
>+        return "STREAM_SYSTEM";
>+      default:
>+        return "STREAM_INVALID";
>+    }
>+  }
>+
>+  // Converts AudioDeviceInfo types to local string representation.
>+  private static String deviceTypeToString(int type) {
>+    switch (type) {
>+      case AudioDeviceInfo.TYPE_UNKNOWN:
>+        return "TYPE_UNKNOWN";
>+      case AudioDeviceInfo.TYPE_BUILTIN_EARPIECE:
>+        return "TYPE_BUILTIN_EARPIECE";
>+      case AudioDeviceInfo.TYPE_BUILTIN_SPEAKER:
>+        return "TYPE_BUILTIN_SPEAKER";
>+      case AudioDeviceInfo.TYPE_WIRED_HEADSET:
>+        return "TYPE_WIRED_HEADSET";
>+      case AudioDeviceInfo.TYPE_WIRED_HEADPHONES:
>+        return "TYPE_WIRED_HEADPHONES";
>+      case AudioDeviceInfo.TYPE_LINE_ANALOG:
>+        return "TYPE_LINE_ANALOG";
>+      case AudioDeviceInfo.TYPE_LINE_DIGITAL:
>+        return "TYPE_LINE_DIGITAL";
>+      case AudioDeviceInfo.TYPE_BLUETOOTH_SCO:
>+        return "TYPE_BLUETOOTH_SCO";
>+      case AudioDeviceInfo.TYPE_BLUETOOTH_A2DP:
>+        return "TYPE_BLUETOOTH_A2DP";
>+      case AudioDeviceInfo.TYPE_HDMI:
>+        return "TYPE_HDMI";
>+      case AudioDeviceInfo.TYPE_HDMI_ARC:
>+        return "TYPE_HDMI_ARC";
>+      case AudioDeviceInfo.TYPE_USB_DEVICE:
>+        return "TYPE_USB_DEVICE";
>+      case AudioDeviceInfo.TYPE_USB_ACCESSORY:
>+        return "TYPE_USB_ACCESSORY";
>+      case AudioDeviceInfo.TYPE_DOCK:
>+        return "TYPE_DOCK";
>+      case AudioDeviceInfo.TYPE_FM:
>+        return "TYPE_FM";
>+      case AudioDeviceInfo.TYPE_BUILTIN_MIC:
>+        return "TYPE_BUILTIN_MIC";
>+      case AudioDeviceInfo.TYPE_FM_TUNER:
>+        return "TYPE_FM_TUNER";
>+      case AudioDeviceInfo.TYPE_TV_TUNER:
>+        return "TYPE_TV_TUNER";
>+      case AudioDeviceInfo.TYPE_TELEPHONY:
>+        return "TYPE_TELEPHONY";
>+      case AudioDeviceInfo.TYPE_AUX_LINE:
>+        return "TYPE_AUX_LINE";
>+      case AudioDeviceInfo.TYPE_IP:
>+        return "TYPE_IP";
>+      case AudioDeviceInfo.TYPE_BUS:
>+        return "TYPE_BUS";
>+      case AudioDeviceInfo.TYPE_USB_HEADSET:
>+        return "TYPE_USB_HEADSET";
>+      default:
>+        return "TYPE_UNKNOWN";
>+    }
>+  }
>+
>+  // Returns true if the device can record audio via a microphone.
>+  private static boolean hasMicrophone(Context context) {
>+    return context.getPackageManager().hasSystemFeature(PackageManager.FEATURE_MICROPHONE);
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/DEPS b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/DEPS
>index db95db649f0..1cf4ba19163 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/DEPS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/DEPS
>@@ -12,5 +12,4 @@ include_rules = [
>   "+modules/video_coding",
>   "+pc",
>   "+system_wrappers/include",
>-  "+voice_engine/include/voe_base.h",
> ]
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/OWNERS
>index 4c1d7c2378a..4f2f2421027 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/OWNERS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/OWNERS
>@@ -1,4 +1,4 @@
>-per-file androidhistogram_jni.cc=sakal@webrtc.org
>-per-file androidmetrics_jni.cc=sakal@webrtc.org
>+per-file androidhistogram.cc=sakal@webrtc.org
>+per-file androidmetrics.cc=sakal@webrtc.org
> per-file androidvideotracksource.*=sakal@webrtc.org
>-per-file androidvideotracksource_jni.cc=sakal@webrtc.org
>+per-file androidvideotracksource.cc=sakal@webrtc.org
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidhistogram_jni.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidhistogram.cc
>similarity index 55%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidhistogram_jni.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidhistogram.cc
>index 2b05d8d7b5d..d5f2a026cae 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidhistogram_jni.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidhistogram.cc
>@@ -11,7 +11,8 @@
> #include <map>
> #include <memory>
> 
>-#include "sdk/android/src/jni/classreferenceholder.h"
>+#include "sdk/android/generated_base_jni/jni/Histogram_jni.h"
>+#include "sdk/android/native_api/jni/java_types.h"
> #include "sdk/android/src/jni/jni_helpers.h"
> #include "system_wrappers/include/metrics.h"
> 
>@@ -19,35 +20,30 @@
> namespace webrtc {
> namespace jni {
> 
>-JNI_FUNCTION_DECLARATION(jlong,
>-                         Histogram_nativeCreateCounts,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jstring j_name,
>-                         jint min,
>-                         jint max,
>-                         jint buckets) {
>+static jlong JNI_Histogram_CreateCounts(JNIEnv* jni,
>+                                        const JavaParamRef<jclass>&,
>+                                        const JavaParamRef<jstring>& j_name,
>+                                        jint min,
>+                                        jint max,
>+                                        jint buckets) {
>   std::string name = JavaToStdString(jni, j_name);
>   return jlongFromPointer(
>       metrics::HistogramFactoryGetCounts(name, min, max, buckets));
> }
> 
>-JNI_FUNCTION_DECLARATION(jlong,
>-                         Histogram_nativeCreateEnumeration,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jstring j_name,
>-                         jint max) {
>+static jlong JNI_Histogram_CreateEnumeration(
>+    JNIEnv* jni,
>+    const JavaParamRef<jclass>&,
>+    const JavaParamRef<jstring>& j_name,
>+    jint max) {
>   std::string name = JavaToStdString(jni, j_name);
>   return jlongFromPointer(metrics::HistogramFactoryGetEnumeration(name, max));
> }
> 
>-JNI_FUNCTION_DECLARATION(void,
>-                         Histogram_nativeAddSample,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jlong histogram,
>-                         jint sample) {
>+static void JNI_Histogram_AddSample(JNIEnv* jni,
>+                                    const JavaParamRef<jclass>&,
>+                                    jlong histogram,
>+                                    jint sample) {
>   if (histogram) {
>     HistogramAdd(reinterpret_cast<metrics::Histogram*>(histogram), sample);
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidmediacodeccommon.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidmediacodeccommon.h
>index 40f7f13a50d..8865b875e2e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidmediacodeccommon.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidmediacodeccommon.h
>@@ -16,7 +16,6 @@
> 
> #include "rtc_base/logging.h"
> #include "rtc_base/thread.h"
>-#include "sdk/android/src/jni/classreferenceholder.h"
> #include "sdk/android/src/jni/jni_helpers.h"
> 
> namespace webrtc {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidmediadecoder_jni.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidmediadecoder.cc
>similarity index 52%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidmediadecoder_jni.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidmediadecoder.cc
>index fb8b0993d91..2f5398e0acf 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidmediadecoder_jni.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidmediadecoder.cc
>@@ -13,32 +13,30 @@
> #include <memory>
> #include <vector>
> 
>-// NOTICE: androidmediadecoder_jni.h must be included before
>-// androidmediacodeccommon.h to avoid build errors.
>-#include "sdk/android/src/jni/androidmediadecoder_jni.h"
>-
>+#include "api/video_codecs/sdp_video_format.h"
> #include "common_video/h264/h264_bitstream_parser.h"
> #include "common_video/include/i420_buffer_pool.h"
>+#include "media/base/mediaconstants.h"
> #include "modules/video_coding/include/video_codec_interface.h"
> #include "modules/video_coding/utility/vp8_header_parser.h"
> #include "rtc_base/bind.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>+#include "rtc_base/numerics/safe_conversions.h"
> #include "rtc_base/scoped_ref_ptr.h"
> #include "rtc_base/thread.h"
> #include "rtc_base/timeutils.h"
>+#include "sdk/android/generated_video_jni/jni/MediaCodecVideoDecoder_jni.h"
>+#include "sdk/android/native_api/jni/java_types.h"
> #include "sdk/android/src/jni/androidmediacodeccommon.h"
>-#include "sdk/android/src/jni/classreferenceholder.h"
>-#include "sdk/android/src/jni/surfacetexturehelper_jni.h"
> #include "sdk/android/src/jni/videoframe.h"
> #include "third_party/libyuv/include/libyuv/convert.h"
>-#include "third_party/libyuv/include/libyuv/convert_from.h"
>+#include "third_party/libyuv/include/libyuv/planar_functions.h"
> #include "third_party/libyuv/include/libyuv/video_common.h"
> 
> using rtc::Bind;
> using rtc::Thread;
> using rtc::ThreadManager;
>-
> namespace webrtc {
> namespace jni {
> 
>@@ -58,21 +56,21 @@ enum { kMaxWarningLogFrames = 2 };
> 
> class MediaCodecVideoDecoder : public VideoDecoder, public rtc::MessageHandler {
>  public:
>-  explicit MediaCodecVideoDecoder(
>-      JNIEnv* jni, VideoCodecType codecType, jobject render_egl_context);
>-  virtual ~MediaCodecVideoDecoder();
>+  explicit MediaCodecVideoDecoder(JNIEnv* jni,
>+                                  VideoCodecType codecType,
>+                                  bool use_surface);
>+  ~MediaCodecVideoDecoder() override;
> 
>-  int32_t InitDecode(const VideoCodec* codecSettings, int32_t numberOfCores)
>-      override;
>+  int32_t InitDecode(const VideoCodec* codecSettings,
>+                     int32_t numberOfCores) override;
> 
>-  int32_t Decode(
>-      const EncodedImage& inputImage, bool missingFrames,
>-      const RTPFragmentationHeader* fragmentation,
>-      const CodecSpecificInfo* codecSpecificInfo = NULL,
>-      int64_t renderTimeMs = -1) override;
>+  int32_t Decode(const EncodedImage& inputImage,
>+                 bool missingFrames,
>+                 const CodecSpecificInfo* codecSpecificInfo = NULL,
>+                 int64_t renderTimeMs = -1) override;
> 
>-  int32_t RegisterDecodeCompleteCallback(DecodedImageCallback* callback)
>-      override;
>+  int32_t RegisterDecodeCompleteCallback(
>+      DecodedImageCallback* callback) override;
> 
>   int32_t Release() override;
> 
>@@ -101,171 +99,51 @@ class MediaCodecVideoDecoder : public VideoDecoder, public rtc::MessageHandler {
>   // Type of video codec.
>   VideoCodecType codecType_;
> 
>-  // Render EGL context - owned by factory, should not be allocated/destroyed
>-  // by VideoDecoder.
>-  jobject render_egl_context_;
>-
>   bool key_frame_required_;
>   bool inited_;
>   bool sw_fallback_required_;
>-  bool use_surface_;
>+  const bool use_surface_;
>   VideoCodec codec_;
>   I420BufferPool decoded_frame_pool_;
>-  rtc::scoped_refptr<SurfaceTextureHelper> surface_texture_helper_;
>   DecodedImageCallback* callback_;
>   int frames_received_;  // Number of frames received by decoder.
>-  int frames_decoded_;  // Number of frames decoded by decoder.
>+  int frames_decoded_;   // Number of frames decoded by decoder.
>   // Number of decoded frames for which log information is displayed.
>   int frames_decoded_logged_;
>   int64_t start_time_ms_;  // Start time for statistics.
>   int current_frames_;  // Number of frames in the current statistics interval.
>-  int current_bytes_;  // Encoded bytes in the current statistics interval.
>+  int current_bytes_;   // Encoded bytes in the current statistics interval.
>   int current_decoding_time_ms_;  // Overall decoding time in the current second
>-  int current_delay_time_ms_;  // Overall delay time in the current second.
>-  uint32_t max_pending_frames_;  // Maximum number of pending input frames.
>+  int current_delay_time_ms_;     // Overall delay time in the current second.
>+  int32_t max_pending_frames_;    // Maximum number of pending input frames.
>   H264BitstreamParser h264_bitstream_parser_;
>-  std::deque<rtc::Optional<uint8_t>> pending_frame_qps_;
>+  std::deque<absl::optional<uint8_t>> pending_frame_qps_;
> 
>   // State that is constant for the lifetime of this object once the ctor
>   // returns.
>   std::unique_ptr<Thread>
>       codec_thread_;  // Thread on which to operate MediaCodec.
>-  ScopedGlobalRef<jclass> j_media_codec_video_decoder_class_;
>-  ScopedGlobalRef<jobject> j_media_codec_video_decoder_;
>-  jmethodID j_init_decode_method_;
>-  jmethodID j_reset_method_;
>-  jmethodID j_release_method_;
>-  jmethodID j_dequeue_input_buffer_method_;
>-  jmethodID j_queue_input_buffer_method_;
>-  jmethodID j_dequeue_byte_buffer_method_;
>-  jmethodID j_dequeue_texture_buffer_method_;
>-  jmethodID j_return_decoded_byte_buffer_method_;
>-  // MediaCodecVideoDecoder fields.
>-  jfieldID j_input_buffers_field_;
>-  jfieldID j_output_buffers_field_;
>-  jfieldID j_color_format_field_;
>-  jfieldID j_width_field_;
>-  jfieldID j_height_field_;
>-  jfieldID j_stride_field_;
>-  jfieldID j_slice_height_field_;
>-  // MediaCodecVideoDecoder.DecodedTextureBuffer fields.
>-  jfieldID j_texture_id_field_;
>-  jfieldID j_transform_matrix_field_;
>-  jfieldID j_texture_presentation_timestamp_ms_field_;
>-  jfieldID j_texture_timestamp_ms_field_;
>-  jfieldID j_texture_ntp_timestamp_ms_field_;
>-  jfieldID j_texture_decode_time_ms_field_;
>-  jfieldID j_texture_frame_delay_ms_field_;
>-  // MediaCodecVideoDecoder.DecodedOutputBuffer fields.
>-  jfieldID j_info_index_field_;
>-  jfieldID j_info_offset_field_;
>-  jfieldID j_info_size_field_;
>-  jfieldID j_presentation_timestamp_ms_field_;
>-  jfieldID j_timestamp_ms_field_;
>-  jfieldID j_ntp_timestamp_ms_field_;
>-  jfieldID j_byte_buffer_decode_time_ms_field_;
>+  ScopedJavaGlobalRef<jobject> j_media_codec_video_decoder_;
> 
>   // Global references; must be deleted in Release().
>-  std::vector<jobject> input_buffers_;
>+  std::vector<ScopedJavaGlobalRef<jobject>> input_buffers_;
> };
> 
> MediaCodecVideoDecoder::MediaCodecVideoDecoder(JNIEnv* jni,
>                                                VideoCodecType codecType,
>-                                               jobject render_egl_context)
>+                                               bool use_surface)
>     : codecType_(codecType),
>-      render_egl_context_(render_egl_context),
>       key_frame_required_(true),
>       inited_(false),
>       sw_fallback_required_(false),
>+      use_surface_(use_surface),
>       codec_thread_(Thread::Create()),
>-      j_media_codec_video_decoder_class_(
>-          jni,
>-          FindClass(jni, "org/webrtc/MediaCodecVideoDecoder")),
>       j_media_codec_video_decoder_(
>           jni,
>-          jni->NewObject(*j_media_codec_video_decoder_class_,
>-                         GetMethodID(jni,
>-                                     *j_media_codec_video_decoder_class_,
>-                                     "<init>",
>-                                     "()V"))) {
>+          Java_MediaCodecVideoDecoder_Constructor(jni)) {
>   codec_thread_->SetName("MediaCodecVideoDecoder", NULL);
>   RTC_CHECK(codec_thread_->Start()) << "Failed to start MediaCodecVideoDecoder";
> 
>-  j_init_decode_method_ = GetMethodID(
>-      jni, *j_media_codec_video_decoder_class_, "initDecode",
>-      "(Lorg/webrtc/MediaCodecVideoDecoder$VideoCodecType;"
>-      "IILorg/webrtc/SurfaceTextureHelper;)Z");
>-  j_reset_method_ =
>-      GetMethodID(jni, *j_media_codec_video_decoder_class_, "reset", "(II)V");
>-  j_release_method_ =
>-      GetMethodID(jni, *j_media_codec_video_decoder_class_, "release", "()V");
>-  j_dequeue_input_buffer_method_ = GetMethodID(
>-      jni, *j_media_codec_video_decoder_class_, "dequeueInputBuffer", "()I");
>-  j_queue_input_buffer_method_ = GetMethodID(
>-      jni, *j_media_codec_video_decoder_class_, "queueInputBuffer", "(IIJJJ)Z");
>-  j_dequeue_byte_buffer_method_ = GetMethodID(
>-      jni, *j_media_codec_video_decoder_class_, "dequeueOutputBuffer",
>-      "(I)Lorg/webrtc/MediaCodecVideoDecoder$DecodedOutputBuffer;");
>-  j_dequeue_texture_buffer_method_ = GetMethodID(
>-      jni, *j_media_codec_video_decoder_class_, "dequeueTextureBuffer",
>-      "(I)Lorg/webrtc/MediaCodecVideoDecoder$DecodedTextureBuffer;");
>-  j_return_decoded_byte_buffer_method_ =
>-      GetMethodID(jni, *j_media_codec_video_decoder_class_,
>-                  "returnDecodedOutputBuffer", "(I)V");
>-
>-  j_input_buffers_field_ = GetFieldID(
>-      jni, *j_media_codec_video_decoder_class_,
>-      "inputBuffers", "[Ljava/nio/ByteBuffer;");
>-  j_output_buffers_field_ = GetFieldID(
>-      jni, *j_media_codec_video_decoder_class_,
>-      "outputBuffers", "[Ljava/nio/ByteBuffer;");
>-  j_color_format_field_ = GetFieldID(
>-      jni, *j_media_codec_video_decoder_class_, "colorFormat", "I");
>-  j_width_field_ = GetFieldID(
>-      jni, *j_media_codec_video_decoder_class_, "width", "I");
>-  j_height_field_ = GetFieldID(
>-      jni, *j_media_codec_video_decoder_class_, "height", "I");
>-  j_stride_field_ = GetFieldID(
>-      jni, *j_media_codec_video_decoder_class_, "stride", "I");
>-  j_slice_height_field_ = GetFieldID(
>-      jni, *j_media_codec_video_decoder_class_, "sliceHeight", "I");
>-
>-  jclass j_decoded_texture_buffer_class = FindClass(jni,
>-      "org/webrtc/MediaCodecVideoDecoder$DecodedTextureBuffer");
>-  j_texture_id_field_ = GetFieldID(
>-      jni, j_decoded_texture_buffer_class, "textureID", "I");
>-  j_transform_matrix_field_ = GetFieldID(
>-      jni, j_decoded_texture_buffer_class, "transformMatrix", "[F");
>-  j_texture_presentation_timestamp_ms_field_ = GetFieldID(
>-      jni, j_decoded_texture_buffer_class, "presentationTimeStampMs", "J");
>-  j_texture_timestamp_ms_field_ = GetFieldID(
>-      jni, j_decoded_texture_buffer_class, "timeStampMs", "J");
>-  j_texture_ntp_timestamp_ms_field_ = GetFieldID(
>-      jni, j_decoded_texture_buffer_class, "ntpTimeStampMs", "J");
>-  j_texture_decode_time_ms_field_ = GetFieldID(
>-      jni, j_decoded_texture_buffer_class, "decodeTimeMs", "J");
>-  j_texture_frame_delay_ms_field_ = GetFieldID(
>-      jni, j_decoded_texture_buffer_class, "frameDelayMs", "J");
>-
>-  jclass j_decoded_output_buffer_class = FindClass(jni,
>-      "org/webrtc/MediaCodecVideoDecoder$DecodedOutputBuffer");
>-  j_info_index_field_ = GetFieldID(
>-      jni, j_decoded_output_buffer_class, "index", "I");
>-  j_info_offset_field_ = GetFieldID(
>-      jni, j_decoded_output_buffer_class, "offset", "I");
>-  j_info_size_field_ = GetFieldID(
>-      jni, j_decoded_output_buffer_class, "size", "I");
>-  j_presentation_timestamp_ms_field_ = GetFieldID(
>-      jni, j_decoded_output_buffer_class, "presentationTimeStampMs", "J");
>-  j_timestamp_ms_field_ = GetFieldID(
>-      jni, j_decoded_output_buffer_class, "timeStampMs", "J");
>-  j_ntp_timestamp_ms_field_ = GetFieldID(
>-      jni, j_decoded_output_buffer_class, "ntpTimeStampMs", "J");
>-  j_byte_buffer_decode_time_ms_field_ = GetFieldID(
>-      jni, j_decoded_output_buffer_class, "decodeTimeMs", "J");
>-
>-  CHECK_EXCEPTION(jni) << "MediaCodecVideoDecoder ctor failed";
>-  use_surface_ = (render_egl_context_ != NULL);
>   ALOGD << "MediaCodecVideoDecoder ctor. Use surface: " << use_surface_;
>   memset(&codec_, 0, sizeof(codec_));
>   AllowBlockingCalls();
>@@ -277,7 +155,7 @@ MediaCodecVideoDecoder::~MediaCodecVideoDecoder() {
> }
> 
> int32_t MediaCodecVideoDecoder::InitDecode(const VideoCodec* inst,
>-    int32_t numberOfCores) {
>+                                           int32_t numberOfCores) {
>   ALOGD << "InitDecode.";
>   if (inst == NULL) {
>     ALOGE << "NULL VideoCodec instance";
>@@ -323,9 +201,9 @@ int32_t MediaCodecVideoDecoder::InitDecodeOnCodecThread() {
>   CheckOnCodecThread();
>   JNIEnv* jni = AttachCurrentThreadIfNeeded();
>   ScopedLocalRefFrame local_ref_frame(jni);
>-  ALOGD << "InitDecodeOnCodecThread Type: " << (int)codecType_ << ". "
>-      << codec_.width << " x " << codec_.height << ". Fps: " <<
>-      (int)codec_.maxFramerate;
>+  ALOGD << "InitDecodeOnCodecThread Type: " << static_cast<int>(codecType_)
>+        << ". " << codec_.width << " x " << codec_.height
>+        << ". Fps: " << static_cast<int>(codec_.maxFramerate);
> 
>   // Release previous codec first if it was allocated before.
>   int ret_val = ReleaseOnCodecThread();
>@@ -337,26 +215,11 @@ int32_t MediaCodecVideoDecoder::InitDecodeOnCodecThread() {
> 
>   ResetVariables();
> 
>-  if (use_surface_) {
>-    surface_texture_helper_ = SurfaceTextureHelper::create(
>-        jni, "Decoder SurfaceTextureHelper", render_egl_context_);
>-    if (!surface_texture_helper_) {
>-      ALOGE << "Couldn't create SurfaceTextureHelper - fallback to SW codec";
>-      sw_fallback_required_ = true;
>-      return WEBRTC_VIDEO_CODEC_ERROR;
>-    }
>-  }
>-
>-  jobject j_video_codec_enum = JavaEnumFromIndexAndClassName(
>-      jni, "MediaCodecVideoDecoder$VideoCodecType", codecType_);
>-  bool success = jni->CallBooleanMethod(
>-      *j_media_codec_video_decoder_,
>-      j_init_decode_method_,
>-      j_video_codec_enum,
>-      codec_.width,
>-      codec_.height,
>-      use_surface_ ? surface_texture_helper_->GetJavaSurfaceTextureHelper()
>-                   : nullptr);
>+  ScopedJavaLocalRef<jobject> j_video_codec_enum =
>+      Java_VideoCodecType_fromNativeIndex(jni, codecType_);
>+  bool success = Java_MediaCodecVideoDecoder_initDecode(
>+      jni, j_media_codec_video_decoder_, j_video_codec_enum, codec_.width,
>+      codec_.height);
> 
>   if (CheckException(jni) || !success) {
>     ALOGE << "Codec initialization error - fallback to SW codec.";
>@@ -380,19 +243,13 @@ int32_t MediaCodecVideoDecoder::InitDecodeOnCodecThread() {
>   }
>   ALOGD << "Maximum amount of pending frames: " << max_pending_frames_;
> 
>-  jobjectArray input_buffers = (jobjectArray)GetObjectField(
>-      jni, *j_media_codec_video_decoder_, j_input_buffers_field_);
>-  size_t num_input_buffers = jni->GetArrayLength(input_buffers);
>-  input_buffers_.resize(num_input_buffers);
>-  for (size_t i = 0; i < num_input_buffers; ++i) {
>-    input_buffers_[i] =
>-        jni->NewGlobalRef(jni->GetObjectArrayElement(input_buffers, i));
>-    if (CheckException(jni)) {
>-      ALOGE << "NewGlobalRef error - fallback to SW codec.";
>-      sw_fallback_required_ = true;
>-      return WEBRTC_VIDEO_CODEC_ERROR;
>-    }
>-  }
>+  ScopedJavaLocalRef<jobjectArray> input_buffers =
>+      Java_MediaCodecVideoDecoder_getInputBuffers(jni,
>+                                                  j_media_codec_video_decoder_);
>+  input_buffers_ = JavaToNativeVector<ScopedJavaGlobalRef<jobject>>(
>+      jni, input_buffers, [](JNIEnv* env, const JavaRef<jobject>& o) {
>+        return ScopedJavaGlobalRef<jobject>(env, o);
>+      });
> 
>   codec_thread_->PostDelayed(RTC_FROM_HERE, kMediaCodecPollMs, this);
> 
>@@ -403,20 +260,17 @@ int32_t MediaCodecVideoDecoder::ResetDecodeOnCodecThread() {
>   CheckOnCodecThread();
>   JNIEnv* jni = AttachCurrentThreadIfNeeded();
>   ScopedLocalRefFrame local_ref_frame(jni);
>-  ALOGD << "ResetDecodeOnCodecThread Type: " << (int)codecType_ << ". "
>-      << codec_.width << " x " << codec_.height;
>-  ALOGD << "  Frames received: " << frames_received_ <<
>-      ". Frames decoded: " << frames_decoded_;
>+  ALOGD << "ResetDecodeOnCodecThread Type: " << static_cast<int>(codecType_)
>+        << ". " << codec_.width << " x " << codec_.height;
>+  ALOGD << "  Frames received: " << frames_received_
>+        << ". Frames decoded: " << frames_decoded_;
> 
>   inited_ = false;
>   rtc::MessageQueueManager::Clear(this);
>   ResetVariables();
> 
>-  jni->CallVoidMethod(
>-      *j_media_codec_video_decoder_,
>-      j_reset_method_,
>-      codec_.width,
>-      codec_.height);
>+  Java_MediaCodecVideoDecoder_reset(jni, j_media_codec_video_decoder_,
>+                                    codec_.width, codec_.height);
> 
>   if (CheckException(jni)) {
>     ALOGE << "Soft reset error - fallback to SW codec.";
>@@ -442,15 +296,11 @@ int32_t MediaCodecVideoDecoder::ReleaseOnCodecThread() {
>   }
>   CheckOnCodecThread();
>   JNIEnv* jni = AttachCurrentThreadIfNeeded();
>-  ALOGD << "DecoderReleaseOnCodecThread: Frames received: " <<
>-      frames_received_ << ". Frames decoded: " << frames_decoded_;
>+  ALOGD << "DecoderReleaseOnCodecThread: Frames received: " << frames_received_
>+        << ". Frames decoded: " << frames_decoded_;
>   ScopedLocalRefFrame local_ref_frame(jni);
>-  for (size_t i = 0; i < input_buffers_.size(); i++) {
>-    jni->DeleteGlobalRef(input_buffers_[i]);
>-  }
>   input_buffers_.clear();
>-  jni->CallVoidMethod(*j_media_codec_video_decoder_, j_release_method_);
>-  surface_texture_helper_ = nullptr;
>+  Java_MediaCodecVideoDecoder_release(jni, j_media_codec_video_decoder_);
>   inited_ = false;
>   rtc::MessageQueueManager::Clear(this);
>   if (CheckException(jni)) {
>@@ -468,8 +318,8 @@ void MediaCodecVideoDecoder::CheckOnCodecThread() {
> 
> void MediaCodecVideoDecoder::EnableFrameLogOnWarning() {
>   // Log next 2 output frames.
>-  frames_decoded_logged_ = std::max(
>-      frames_decoded_logged_, frames_decoded_ + kMaxWarningLogFrames);
>+  frames_decoded_logged_ =
>+      std::max(frames_decoded_logged_, frames_decoded_ + kMaxWarningLogFrames);
> }
> 
> int32_t MediaCodecVideoDecoder::ProcessHWErrorOnCodecThread() {
>@@ -501,7 +351,6 @@ int32_t MediaCodecVideoDecoder::ProcessHWErrorOnCodecThread() {
> int32_t MediaCodecVideoDecoder::Decode(
>     const EncodedImage& inputImage,
>     bool missingFrames,
>-    const RTPFragmentationHeader* fragmentation,
>     const CodecSpecificInfo* codecSpecificInfo,
>     int64_t renderTimeMs) {
>   if (sw_fallback_required_) {
>@@ -524,10 +373,10 @@ int32_t MediaCodecVideoDecoder::Decode(
>   // Check if encoded frame dimension has changed.
>   if ((inputImage._encodedWidth * inputImage._encodedHeight > 0) &&
>       (inputImage._encodedWidth != codec_.width ||
>-      inputImage._encodedHeight != codec_.height)) {
>-    ALOGW << "Input resolution changed from " <<
>-        codec_.width << " x " << codec_.height << " to " <<
>-        inputImage._encodedWidth << " x " << inputImage._encodedHeight;
>+       inputImage._encodedHeight != codec_.height)) {
>+    ALOGW << "Input resolution changed from " << codec_.width << " x "
>+          << codec_.height << " to " << inputImage._encodedWidth << " x "
>+          << inputImage._encodedHeight;
>     codec_.width = inputImage._encodedWidth;
>     codec_.height = inputImage._encodedHeight;
>     int32_t ret;
>@@ -580,41 +429,41 @@ int32_t MediaCodecVideoDecoder::DecodeOnCodecThread(
>   if (codecType_ == kVideoCodecH264 &&
>       frames_received_ > frames_decoded_ + max_pending_frames_) {
>     // Print warning for H.264 only - for VP8/VP9 one frame delay is ok.
>-    ALOGW << "Decoder is too far behind. Try to drain. Received: " <<
>-        frames_received_ << ". Decoded: " << frames_decoded_;
>+    ALOGW << "Decoder is too far behind. Try to drain. Received: "
>+          << frames_received_ << ". Decoded: " << frames_decoded_;
>     EnableFrameLogOnWarning();
>   }
>-  const int64 drain_start = rtc::TimeMillis();
>+  const int64_t drain_start = rtc::TimeMillis();
>   while ((frames_received_ > frames_decoded_ + max_pending_frames_) &&
>          (rtc::TimeMillis() - drain_start) < kMediaCodecTimeoutMs) {
>     if (!DeliverPendingOutputs(jni, kMediaCodecPollMs)) {
>-      ALOGE << "DeliverPendingOutputs error. Frames received: " <<
>-          frames_received_ << ". Frames decoded: " << frames_decoded_;
>+      ALOGE << "DeliverPendingOutputs error. Frames received: "
>+            << frames_received_ << ". Frames decoded: " << frames_decoded_;
>       return ProcessHWErrorOnCodecThread();
>     }
>   }
>   if (frames_received_ > frames_decoded_ + max_pending_frames_) {
>-    ALOGE << "Output buffer dequeue timeout. Frames received: " <<
>-        frames_received_ << ". Frames decoded: " << frames_decoded_;
>+    ALOGE << "Output buffer dequeue timeout. Frames received: "
>+          << frames_received_ << ". Frames decoded: " << frames_decoded_;
>     return ProcessHWErrorOnCodecThread();
>   }
> 
>   // Get input buffer.
>-  int j_input_buffer_index = jni->CallIntMethod(
>-      *j_media_codec_video_decoder_, j_dequeue_input_buffer_method_);
>+  int j_input_buffer_index = Java_MediaCodecVideoDecoder_dequeueInputBuffer(
>+      jni, j_media_codec_video_decoder_);
>   if (CheckException(jni) || j_input_buffer_index < 0) {
>-    ALOGE << "dequeueInputBuffer error: " << j_input_buffer_index <<
>-        ". Retry DeliverPendingOutputs.";
>+    ALOGE << "dequeueInputBuffer error: " << j_input_buffer_index
>+          << ". Retry DeliverPendingOutputs.";
>     EnableFrameLogOnWarning();
>     // Try to drain the decoder.
>     if (!DeliverPendingOutputs(jni, kMediaCodecPollMs)) {
>-      ALOGE << "DeliverPendingOutputs error. Frames received: " <<
>-          frames_received_ << ". Frames decoded: " << frames_decoded_;
>+      ALOGE << "DeliverPendingOutputs error. Frames received: "
>+            << frames_received_ << ". Frames decoded: " << frames_decoded_;
>       return ProcessHWErrorOnCodecThread();
>     }
>     // Try dequeue input buffer one last time.
>-    j_input_buffer_index = jni->CallIntMethod(
>-        *j_media_codec_video_decoder_, j_dequeue_input_buffer_method_);
>+    j_input_buffer_index = Java_MediaCodecVideoDecoder_dequeueInputBuffer(
>+        jni, j_media_codec_video_decoder_);
>     if (CheckException(jni) || j_input_buffer_index < 0) {
>       ALOGE << "dequeueInputBuffer critical error: " << j_input_buffer_index;
>       return ProcessHWErrorOnCodecThread();
>@@ -622,14 +471,15 @@ int32_t MediaCodecVideoDecoder::DecodeOnCodecThread(
>   }
> 
>   // Copy encoded data to Java ByteBuffer.
>-  jobject j_input_buffer = input_buffers_[j_input_buffer_index];
>+  jobject j_input_buffer = input_buffers_[j_input_buffer_index].obj();
>   uint8_t* buffer =
>       reinterpret_cast<uint8_t*>(jni->GetDirectBufferAddress(j_input_buffer));
>   RTC_CHECK(buffer) << "Indirect buffer??";
>-  int64_t buffer_capacity = jni->GetDirectBufferCapacity(j_input_buffer);
>+  size_t buffer_capacity =
>+      rtc::dchecked_cast<size_t>(jni->GetDirectBufferCapacity(j_input_buffer));
>   if (CheckException(jni) || buffer_capacity < inputImage._length) {
>-    ALOGE << "Input frame size "<<  inputImage._length <<
>-        " is bigger than buffer size " << buffer_capacity;
>+    ALOGE << "Input frame size " << inputImage._length
>+          << " is bigger than buffer size " << buffer_capacity;
>     return ProcessHWErrorOnCodecThread();
>   }
>   jlong presentation_timestamp_us = static_cast<jlong>(
>@@ -637,17 +487,17 @@ int32_t MediaCodecVideoDecoder::DecodeOnCodecThread(
>   memcpy(buffer, inputImage._buffer, inputImage._length);
> 
>   if (frames_decoded_ < frames_decoded_logged_) {
>-    ALOGD << "Decoder frame in # " << frames_received_ <<
>-        ". Type: " << inputImage._frameType <<
>-        ". Buffer # " << j_input_buffer_index <<
>-        ". TS: " << presentation_timestamp_us / 1000 <<
>-        ". Size: " << inputImage._length;
>+    ALOGD << "Decoder frame in # " << frames_received_
>+          << ". Type: " << inputImage._frameType << ". Buffer # "
>+          << j_input_buffer_index
>+          << ". TS: " << presentation_timestamp_us / 1000
>+          << ". Size: " << inputImage._length;
>   }
> 
>   // Save input image timestamps for later output.
>   frames_received_++;
>   current_bytes_ += inputImage._length;
>-  rtc::Optional<uint8_t> qp;
>+  absl::optional<uint8_t> qp;
>   if (codecType_ == kVideoCodecVP8) {
>     int qp_int;
>     if (vp8::GetQp(inputImage._buffer, inputImage._length, &qp_int)) {
>@@ -664,14 +514,10 @@ int32_t MediaCodecVideoDecoder::DecodeOnCodecThread(
>   pending_frame_qps_.push_back(qp);
> 
>   // Feed input to decoder.
>-  bool success = jni->CallBooleanMethod(
>-      *j_media_codec_video_decoder_,
>-      j_queue_input_buffer_method_,
>-      j_input_buffer_index,
>-      inputImage._length,
>-      presentation_timestamp_us,
>-      static_cast<int64_t> (inputImage._timeStamp),
>-      inputImage.ntp_time_ms_);
>+  bool success = Java_MediaCodecVideoDecoder_queueInputBuffer(
>+      jni, j_media_codec_video_decoder_, j_input_buffer_index,
>+      static_cast<int>(inputImage._length), presentation_timestamp_us,
>+      static_cast<int64_t>(inputImage._timeStamp), inputImage.ntp_time_ms_);
>   if (CheckException(jni) || !success) {
>     ALOGE << "queueInputBuffer error";
>     return ProcessHWErrorOnCodecThread();
>@@ -686,20 +532,18 @@ int32_t MediaCodecVideoDecoder::DecodeOnCodecThread(
>   return WEBRTC_VIDEO_CODEC_OK;
> }
> 
>-bool MediaCodecVideoDecoder::DeliverPendingOutputs(
>-    JNIEnv* jni, int dequeue_timeout_ms) {
>+bool MediaCodecVideoDecoder::DeliverPendingOutputs(JNIEnv* jni,
>+                                                   int dequeue_timeout_ms) {
>   CheckOnCodecThread();
>   if (frames_received_ <= frames_decoded_) {
>     // No need to query for output buffers - decoder is drained.
>     return true;
>   }
>   // Get decoder output.
>-  jobject j_decoder_output_buffer =
>-      jni->CallObjectMethod(*j_media_codec_video_decoder_,
>-          use_surface_ ? j_dequeue_texture_buffer_method_
>-                       : j_dequeue_byte_buffer_method_,
>-          dequeue_timeout_ms);
>-
>+  ScopedJavaLocalRef<jobject> j_decoder_output_buffer =
>+      (use_surface_ ? &Java_MediaCodecVideoDecoder_dequeueTextureBuffer
>+                    : &Java_MediaCodecVideoDecoder_dequeueOutputBuffer)(
>+          jni, j_media_codec_video_decoder_, dequeue_timeout_ms);
>   if (CheckException(jni)) {
>     ALOGE << "dequeueOutputBuffer() error";
>     return false;
>@@ -710,10 +554,12 @@ bool MediaCodecVideoDecoder::DeliverPendingOutputs(
>   }
> 
>   // Get decoded video frame properties.
>-  int color_format = GetIntField(jni, *j_media_codec_video_decoder_,
>-      j_color_format_field_);
>-  int width = GetIntField(jni, *j_media_codec_video_decoder_, j_width_field_);
>-  int height = GetIntField(jni, *j_media_codec_video_decoder_, j_height_field_);
>+  int color_format = Java_MediaCodecVideoDecoder_getColorFormat(
>+      jni, j_media_codec_video_decoder_);
>+  int width =
>+      Java_MediaCodecVideoDecoder_getWidth(jni, j_media_codec_video_decoder_);
>+  int height =
>+      Java_MediaCodecVideoDecoder_getHeight(jni, j_media_codec_video_decoder_);
> 
>   rtc::scoped_refptr<VideoFrameBuffer> frame_buffer;
>   int64_t presentation_timestamps_ms = 0;
>@@ -723,53 +569,50 @@ bool MediaCodecVideoDecoder::DeliverPendingOutputs(
>   int64_t frame_delayed_ms = 0;
>   if (use_surface_) {
>     // Extract data from Java DecodedTextureBuffer.
>-    presentation_timestamps_ms = GetLongField(
>-        jni, j_decoder_output_buffer,
>-        j_texture_presentation_timestamp_ms_field_);
>-    output_timestamps_ms = GetLongField(
>-        jni, j_decoder_output_buffer, j_texture_timestamp_ms_field_);
>-    output_ntp_timestamps_ms = GetLongField(
>-        jni, j_decoder_output_buffer, j_texture_ntp_timestamp_ms_field_);
>-    decode_time_ms = GetLongField(
>-        jni, j_decoder_output_buffer, j_texture_decode_time_ms_field_);
>-
>-    const int texture_id =
>-        GetIntField(jni, j_decoder_output_buffer, j_texture_id_field_);
>-    if (texture_id != 0) {  // |texture_id| == 0 represents a dropped frame.
>-      const jfloatArray j_transform_matrix =
>-          reinterpret_cast<jfloatArray>(GetObjectField(
>-              jni, j_decoder_output_buffer, j_transform_matrix_field_));
>-      frame_delayed_ms = GetLongField(
>-          jni, j_decoder_output_buffer, j_texture_frame_delay_ms_field_);
>-
>-      // Create VideoFrameBuffer with native texture handle.
>-      frame_buffer = surface_texture_helper_->CreateTextureFrame(
>-          width, height, NativeHandleImpl(jni, texture_id, j_transform_matrix));
>+    presentation_timestamps_ms =
>+        Java_DecodedTextureBuffer_getPresentationTimestampMs(
>+            jni, j_decoder_output_buffer);
>+    output_timestamps_ms =
>+        Java_DecodedTextureBuffer_getTimeStampMs(jni, j_decoder_output_buffer);
>+    output_ntp_timestamps_ms = Java_DecodedTextureBuffer_getNtpTimestampMs(
>+        jni, j_decoder_output_buffer);
>+    decode_time_ms =
>+        Java_DecodedTextureBuffer_getDecodeTimeMs(jni, j_decoder_output_buffer);
>+
>+    ScopedJavaLocalRef<jobject> j_video_frame_buffer =
>+        Java_DecodedTextureBuffer_getVideoFrameBuffer(jni,
>+                                                      j_decoder_output_buffer);
>+    // |video_frame_buffer| == null represents a dropped frame.
>+    if (!j_video_frame_buffer.is_null()) {
>+      frame_delayed_ms = Java_DecodedTextureBuffer_getFrameDelayMs(
>+          jni, j_decoder_output_buffer);
>+      frame_buffer = AndroidVideoBuffer::Adopt(jni, j_video_frame_buffer);
>     } else {
>       EnableFrameLogOnWarning();
>     }
>   } else {
>     // Extract data from Java ByteBuffer and create output yuv420 frame -
>     // for non surface decoding only.
>-    int stride =
>-        GetIntField(jni, *j_media_codec_video_decoder_, j_stride_field_);
>-    const int slice_height =
>-        GetIntField(jni, *j_media_codec_video_decoder_, j_slice_height_field_);
>-    const int output_buffer_index = GetIntField(
>-        jni, j_decoder_output_buffer, j_info_index_field_);
>-    const int output_buffer_offset = GetIntField(
>-        jni, j_decoder_output_buffer, j_info_offset_field_);
>-    const int output_buffer_size = GetIntField(
>-        jni, j_decoder_output_buffer, j_info_size_field_);
>-    presentation_timestamps_ms = GetLongField(
>-        jni, j_decoder_output_buffer, j_presentation_timestamp_ms_field_);
>-    output_timestamps_ms = GetLongField(
>-        jni, j_decoder_output_buffer, j_timestamp_ms_field_);
>-    output_ntp_timestamps_ms = GetLongField(
>-        jni, j_decoder_output_buffer, j_ntp_timestamp_ms_field_);
>-
>-    decode_time_ms = GetLongField(jni, j_decoder_output_buffer,
>-                                  j_byte_buffer_decode_time_ms_field_);
>+    int stride = Java_MediaCodecVideoDecoder_getStride(
>+        jni, j_media_codec_video_decoder_);
>+    const int slice_height = Java_MediaCodecVideoDecoder_getSliceHeight(
>+        jni, j_media_codec_video_decoder_);
>+    const int output_buffer_index =
>+        Java_DecodedOutputBuffer_getIndex(jni, j_decoder_output_buffer);
>+    const int output_buffer_offset =
>+        Java_DecodedOutputBuffer_getOffset(jni, j_decoder_output_buffer);
>+    const int output_buffer_size =
>+        Java_DecodedOutputBuffer_getSize(jni, j_decoder_output_buffer);
>+    presentation_timestamps_ms =
>+        Java_DecodedOutputBuffer_getPresentationTimestampMs(
>+            jni, j_decoder_output_buffer);
>+    output_timestamps_ms =
>+        Java_DecodedOutputBuffer_getTimestampMs(jni, j_decoder_output_buffer);
>+    output_ntp_timestamps_ms = Java_DecodedOutputBuffer_getNtpTimestampMs(
>+        jni, j_decoder_output_buffer);
>+
>+    decode_time_ms =
>+        Java_DecodedOutputBuffer_getDecodeTimeMs(jni, j_decoder_output_buffer);
>     RTC_CHECK_GE(slice_height, height);
> 
>     if (output_buffer_size < width * height * 3 / 2) {
>@@ -782,12 +625,13 @@ bool MediaCodecVideoDecoder::DeliverPendingOutputs(
>       // output byte buffer, so actual stride value need to be corrected.
>       stride = output_buffer_size * 2 / (height * 3);
>     }
>-    jobjectArray output_buffers = reinterpret_cast<jobjectArray>(GetObjectField(
>-        jni, *j_media_codec_video_decoder_, j_output_buffers_field_));
>+    ScopedJavaLocalRef<jobjectArray> output_buffers =
>+        Java_MediaCodecVideoDecoder_getOutputBuffers(
>+            jni, j_media_codec_video_decoder_);
>     jobject output_buffer =
>-        jni->GetObjectArrayElement(output_buffers, output_buffer_index);
>-    uint8_t* payload = reinterpret_cast<uint8_t*>(jni->GetDirectBufferAddress(
>-        output_buffer));
>+        jni->GetObjectArrayElement(output_buffers.obj(), output_buffer_index);
>+    uint8_t* payload =
>+        reinterpret_cast<uint8_t*>(jni->GetDirectBufferAddress(output_buffer));
>     if (CheckException(jni)) {
>       return false;
>     }
>@@ -813,15 +657,12 @@ bool MediaCodecVideoDecoder::DeliverPendingOutputs(
>           (slice_height % 2 == 0) ? (height + 1) / 2 : height / 2;
>       const int u_offset = uv_stride * slice_height / 2;
>       const uint8_t* v_ptr = u_ptr + u_offset;
>-      libyuv::CopyPlane(y_ptr, stride,
>-                        i420_buffer->MutableDataY(), i420_buffer->StrideY(),
>-                        width, height);
>-      libyuv::CopyPlane(u_ptr, uv_stride,
>-                        i420_buffer->MutableDataU(), i420_buffer->StrideU(),
>-                        chroma_width, chroma_height);
>-      libyuv::CopyPlane(v_ptr, uv_stride,
>-                        i420_buffer->MutableDataV(), i420_buffer->StrideV(),
>-                        chroma_width, chroma_height);
>+      libyuv::CopyPlane(y_ptr, stride, i420_buffer->MutableDataY(),
>+                        i420_buffer->StrideY(), width, height);
>+      libyuv::CopyPlane(u_ptr, uv_stride, i420_buffer->MutableDataU(),
>+                        i420_buffer->StrideU(), chroma_width, chroma_height);
>+      libyuv::CopyPlane(v_ptr, uv_stride, i420_buffer->MutableDataV(),
>+                        i420_buffer->StrideV(), chroma_width, chroma_height);
>       if (slice_height % 2 == 1) {
>         RTC_CHECK_EQ(height, slice_height);
>         // Duplicate the last chroma rows.
>@@ -847,22 +688,19 @@ bool MediaCodecVideoDecoder::DeliverPendingOutputs(
>     frame_buffer = i420_buffer;
> 
>     // Return output byte buffer back to codec.
>-    jni->CallVoidMethod(
>-        *j_media_codec_video_decoder_,
>-        j_return_decoded_byte_buffer_method_,
>-        output_buffer_index);
>+    Java_MediaCodecVideoDecoder_returnDecodedOutputBuffer(
>+        jni, j_media_codec_video_decoder_, output_buffer_index);
>     if (CheckException(jni)) {
>       ALOGE << "returnDecodedOutputBuffer error";
>       return false;
>     }
>   }
>   if (frames_decoded_ < frames_decoded_logged_) {
>-    ALOGD << "Decoder frame out # " << frames_decoded_ <<
>-        ". " << width << " x " << height <<
>-        ". Color: " << color_format <<
>-        ". TS: " << presentation_timestamps_ms <<
>-        ". DecTime: " << (int)decode_time_ms <<
>-        ". DelayTime: " << (int)frame_delayed_ms;
>+    ALOGD << "Decoder frame out # " << frames_decoded_ << ". " << width << " x "
>+          << height << ". Color: " << color_format
>+          << ". TS: " << presentation_timestamps_ms
>+          << ". DecTime: " << static_cast<int>(decode_time_ms)
>+          << ". DelayTime: " << static_cast<int>(frame_delayed_ms);
>   }
> 
>   // Calculate and print decoding statistics - every 3 seconds.
>@@ -876,13 +714,13 @@ bool MediaCodecVideoDecoder::DeliverPendingOutputs(
>     int current_bitrate = current_bytes_ * 8 / statistic_time_ms;
>     int current_fps =
>         (current_frames_ * 1000 + statistic_time_ms / 2) / statistic_time_ms;
>-    ALOGD << "Frames decoded: " << frames_decoded_ <<
>-        ". Received: " <<  frames_received_ <<
>-        ". Bitrate: " << current_bitrate << " kbps" <<
>-        ". Fps: " << current_fps <<
>-        ". DecTime: " << (current_decoding_time_ms_ / current_frames_) <<
>-        ". DelayTime: " << (current_delay_time_ms_ / current_frames_) <<
>-        " for last " << statistic_time_ms << " ms.";
>+    ALOGD << "Frames decoded: " << frames_decoded_
>+          << ". Received: " << frames_received_
>+          << ". Bitrate: " << current_bitrate << " kbps"
>+          << ". Fps: " << current_fps
>+          << ". DecTime: " << (current_decoding_time_ms_ / current_frames_)
>+          << ". DelayTime: " << (current_delay_time_ms_ / current_frames_)
>+          << " for last " << statistic_time_ms << " ms.";
>     start_time_ms_ = rtc::TimeMillis();
>     current_frames_ = 0;
>     current_bytes_ = 0;
>@@ -896,7 +734,7 @@ bool MediaCodecVideoDecoder::DeliverPendingOutputs(
>     decoded_frame.set_timestamp(output_timestamps_ms);
>     decoded_frame.set_ntp_time_ms(output_ntp_timestamps_ms);
> 
>-    rtc::Optional<uint8_t> qp = pending_frame_qps_.front();
>+    absl::optional<uint8_t> qp = pending_frame_qps_.front();
>     pending_frame_qps_.pop_front();
>     callback_->Decoded(decoded_frame, decode_time_ms, qp);
>   }
>@@ -929,95 +767,20 @@ void MediaCodecVideoDecoder::OnMessage(rtc::Message* msg) {
>   codec_thread_->PostDelayed(RTC_FROM_HERE, kMediaCodecPollMs, this);
> }
> 
>-MediaCodecVideoDecoderFactory::MediaCodecVideoDecoderFactory()
>-    : egl_context_(nullptr) {
>-  ALOGD << "MediaCodecVideoDecoderFactory ctor";
>-  JNIEnv* jni = AttachCurrentThreadIfNeeded();
>-  ScopedLocalRefFrame local_ref_frame(jni);
>-  jclass j_decoder_class = FindClass(jni, "org/webrtc/MediaCodecVideoDecoder");
>-  supported_codec_types_.clear();
>-
>-  bool is_vp8_hw_supported = jni->CallStaticBooleanMethod(
>-      j_decoder_class,
>-      GetStaticMethodID(jni, j_decoder_class, "isVp8HwSupported", "()Z"));
>-  if (CheckException(jni)) {
>-    is_vp8_hw_supported = false;
>-  }
>-  if (is_vp8_hw_supported) {
>-    ALOGD << "VP8 HW Decoder supported.";
>-    supported_codec_types_.push_back(kVideoCodecVP8);
>-  }
>-
>-  bool is_vp9_hw_supported = jni->CallStaticBooleanMethod(
>-      j_decoder_class,
>-      GetStaticMethodID(jni, j_decoder_class, "isVp9HwSupported", "()Z"));
>-  if (CheckException(jni)) {
>-    is_vp9_hw_supported = false;
>-  }
>-  if (is_vp9_hw_supported) {
>-    ALOGD << "VP9 HW Decoder supported.";
>-    supported_codec_types_.push_back(kVideoCodecVP9);
>-  }
>-
>-  bool is_h264_hw_supported = jni->CallStaticBooleanMethod(
>-      j_decoder_class,
>-      GetStaticMethodID(jni, j_decoder_class, "isH264HwSupported", "()Z"));
>-  if (CheckException(jni)) {
>-    is_h264_hw_supported = false;
>-  }
>-  if (is_h264_hw_supported) {
>-    ALOGD << "H264 HW Decoder supported.";
>-    supported_codec_types_.push_back(kVideoCodecH264);
>-  }
>-}
>-
>-MediaCodecVideoDecoderFactory::~MediaCodecVideoDecoderFactory() {
>-  ALOGD << "MediaCodecVideoDecoderFactory dtor";
>-  if (egl_context_) {
>-    JNIEnv* jni = AttachCurrentThreadIfNeeded();
>-    jni->DeleteGlobalRef(egl_context_);
>-  }
>-}
>-
>-void MediaCodecVideoDecoderFactory::SetEGLContext(
>-    JNIEnv* jni, jobject egl_context) {
>-  ALOGD << "MediaCodecVideoDecoderFactory::SetEGLContext";
>-  if (egl_context_) {
>-    jni->DeleteGlobalRef(egl_context_);
>-    egl_context_ = nullptr;
>-  }
>-  egl_context_ = jni->NewGlobalRef(egl_context);
>-  if (CheckException(jni)) {
>-    ALOGE << "error calling NewGlobalRef for EGL Context.";
>-  }
>-}
>-
>-VideoDecoder* MediaCodecVideoDecoderFactory::CreateVideoDecoder(
>-    VideoCodecType type) {
>-  if (supported_codec_types_.empty()) {
>-    ALOGW << "No HW video decoder for type " << (int)type;
>-    return nullptr;
>-  }
>-  for (VideoCodecType codec_type : supported_codec_types_) {
>-    if (codec_type == type) {
>-      ALOGD << "Create HW video decoder for type " << (int)type;
>-      JNIEnv* jni = AttachCurrentThreadIfNeeded();
>-      ScopedLocalRefFrame local_ref_frame(jni);
>-      return new MediaCodecVideoDecoder(jni, type, egl_context_);
>-    }
>-  }
>-  ALOGW << "Can not find HW video decoder for type " << (int)type;
>-  return nullptr;
>-}
>-
>-void MediaCodecVideoDecoderFactory::DestroyVideoDecoder(VideoDecoder* decoder) {
>-  ALOGD << "Destroy video decoder.";
>-  delete decoder;
>-}
>-
> const char* MediaCodecVideoDecoder::ImplementationName() const {
>   return "MediaCodec";
> }
> 
>+static jlong JNI_MediaCodecVideoDecoder_CreateDecoder(
>+    JNIEnv* env,
>+    const JavaParamRef<jclass>&,
>+    const JavaParamRef<jstring>& codec,
>+    jboolean use_surface) {
>+  ScopedLocalRefFrame local_ref_frame(env);
>+  return jlongFromPointer(new MediaCodecVideoDecoder(
>+      env, PayloadStringToCodecType(JavaToNativeString(env, codec)),
>+      use_surface));
>+}
>+
> }  // namespace jni
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidmediadecoder_jni.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidmediadecoder_jni.h
>deleted file mode 100644
>index bcdf9cc0cd6..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidmediadecoder_jni.h
>+++ /dev/null
>@@ -1,42 +0,0 @@
>-/*
>- *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef SDK_ANDROID_SRC_JNI_ANDROIDMEDIADECODER_JNI_H_
>-#define SDK_ANDROID_SRC_JNI_ANDROIDMEDIADECODER_JNI_H_
>-
>-#include "sdk/android/src/jni/jni_helpers.h"
>-#include "media/engine/webrtcvideodecoderfactory.h"
>-
>-namespace webrtc {
>-namespace jni {
>-
>-// Implementation of Android MediaCodec based decoder factory.
>-class MediaCodecVideoDecoderFactory
>-    : public cricket::WebRtcVideoDecoderFactory {
>- public:
>-  MediaCodecVideoDecoderFactory();
>-  virtual ~MediaCodecVideoDecoderFactory();
>-
>-  void SetEGLContext(JNIEnv* jni, jobject render_egl_context);
>-
>-  // WebRtcVideoDecoderFactory implementation.
>-  VideoDecoder* CreateVideoDecoder(VideoCodecType type) override;
>-
>-  void DestroyVideoDecoder(VideoDecoder* decoder) override;
>-
>- private:
>-  jobject egl_context_;
>-  std::vector<VideoCodecType> supported_codec_types_;
>-};
>-
>-}  // namespace jni
>-}  // namespace webrtc
>-
>-#endif  // SDK_ANDROID_SRC_JNI_ANDROIDMEDIADECODER_JNI_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidmediaencoder_jni.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidmediaencoder.cc
>similarity index 67%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidmediaencoder_jni.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidmediaencoder.cc
>index 9b804c64328..0fd6912cbca 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidmediaencoder_jni.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidmediaencoder.cc
>@@ -8,21 +8,20 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-// NOTICE: androidmediaencoder_jni.h must be included before
>-// androidmediacodeccommon.h to avoid build errors.
>-#include "sdk/android/src/jni/androidmediaencoder_jni.h"
>-
> #include <algorithm>
> #include <list>
> #include <memory>
> #include <string>
> #include <utility>
> 
>+#include "api/video_codecs/sdp_video_format.h"
> #include "api/video_codecs/video_encoder.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "common_video/h264/h264_bitstream_parser.h"
> #include "common_video/h264/h264_common.h"
> #include "common_video/h264/profile_level_id.h"
>+#include "media/base/codec.h"
>+#include "media/base/mediaconstants.h"
> #include "media/engine/internalencoderfactory.h"
> #include "modules/video_coding/include/video_codec_interface.h"
> #include "modules/video_coding/utility/quality_scaler.h"
>@@ -31,15 +30,16 @@
> #include "rtc_base/bind.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/random.h"
> #include "rtc_base/sequenced_task_checker.h"
> #include "rtc_base/task_queue.h"
> #include "rtc_base/thread.h"
> #include "rtc_base/timeutils.h"
> #include "rtc_base/weak_ptr.h"
>+#include "sdk/android/generated_video_jni/jni/MediaCodecVideoEncoder_jni.h"
>+#include "sdk/android/native_api/jni/java_types.h"
> #include "sdk/android/src/jni/androidmediacodeccommon.h"
>-#include "sdk/android/src/jni/classreferenceholder.h"
> #include "sdk/android/src/jni/jni_helpers.h"
>+#include "sdk/android/src/jni/videocodecinfo.h"
> #include "sdk/android/src/jni/videoframe.h"
> #include "system_wrappers/include/field_trial.h"
> #include "third_party/libyuv/include/libyuv/convert.h"
>@@ -77,9 +77,8 @@ __android_log_print(ANDROID_LOG_VERBOSE, TAG_ENCODER, __VA_ARGS__)
> 
>     namespace {
>   // Maximum time limit between incoming frames before requesting a key frame.
>-  const size_t kFrameDiffThresholdMs = 350;
>+  const int64_t kFrameDiffThresholdMs = 350;
>   const int kMinKeyFrameInterval = 6;
>-  const char kH264HighProfileFieldTrial[] = "WebRTC-H264HighProfile";
>   const char kCustomQPThresholdsFieldTrial[] = "WebRTC-CustomQPThresholds";
> }  // namespace
> 
>@@ -91,10 +90,10 @@ __android_log_print(ANDROID_LOG_VERBOSE, TAG_ENCODER, __VA_ARGS__)
> // this is the encoder queue from ViE encoder.
> class MediaCodecVideoEncoder : public VideoEncoder {
>  public:
>-  virtual ~MediaCodecVideoEncoder();
>+  ~MediaCodecVideoEncoder() override;
>   MediaCodecVideoEncoder(JNIEnv* jni,
>-                         const cricket::VideoCodec& codec,
>-                         jobject egl_context);
>+                         const SdpVideoFormat& format,
>+                         bool has_egl_context);
> 
>   // VideoEncoder implementation.
>   int32_t InitEncode(const VideoCodec* codec_settings,
>@@ -108,10 +107,10 @@ class MediaCodecVideoEncoder : public VideoEncoder {
>   int32_t Release() override;
>   int32_t SetChannelParameters(uint32_t /* packet_loss */,
>                                int64_t /* rtt */) override;
>-  int32_t SetRateAllocation(const BitrateAllocation& rate_allocation,
>+  int32_t SetRateAllocation(const VideoBitrateAllocation& rate_allocation,
>                             uint32_t frame_rate) override;
> 
>-  bool SupportsNativeHandle() const override { return egl_context_ != nullptr; }
>+  bool SupportsNativeHandle() const override { return has_egl_context_; }
>   const char* ImplementationName() const override;
> 
>   // Fills the input buffer with data from the buffers passed as parameters.
>@@ -170,21 +169,13 @@ class MediaCodecVideoEncoder : public VideoEncoder {
>                         bool key_frame,
>                         const VideoFrame& frame,
>                         int input_buffer_index);
>-  bool EncodeTexture(JNIEnv* jni, bool key_frame, const VideoFrame& frame);
>   // Encodes a new style org.webrtc.VideoFrame. Might be a I420 or a texture
>   // frame.
>   bool EncodeJavaFrame(JNIEnv* jni,
>                        bool key_frame,
>-                       jobject frame,
>+                       const JavaRef<jobject>& frame,
>                        int input_buffer_index);
> 
>-  // Helper accessors for MediaCodecVideoEncoder$OutputBufferInfo members.
>-  int GetOutputBufferInfoIndex(JNIEnv* jni, jobject j_output_buffer_info);
>-  jobject GetOutputBufferInfoBuffer(JNIEnv* jni, jobject j_output_buffer_info);
>-  bool GetOutputBufferInfoIsKeyFrame(JNIEnv* jni, jobject j_output_buffer_info);
>-  jlong GetOutputBufferInfoPresentationTimestampUs(
>-      JNIEnv* jni, jobject j_output_buffer_info);
>-
>   // Deliver any outputs pending in the MediaCodec to our |callback_| and return
>   // true on success.
>   bool DeliverPendingOutputs(JNIEnv* jni);
>@@ -205,32 +196,14 @@ class MediaCodecVideoEncoder : public VideoEncoder {
> #endif
> 
>   // Type of video codec.
>-  const cricket::VideoCodec codec_;
>+  const SdpVideoFormat format_;
> 
>   EncodedImageCallback* callback_;
> 
>   // State that is constant for the lifetime of this object once the ctor
>   // returns.
>   rtc::SequencedTaskChecker encoder_queue_checker_;
>-  ScopedGlobalRef<jclass> j_media_codec_video_encoder_class_;
>-  ScopedGlobalRef<jobject> j_media_codec_video_encoder_;
>-  jmethodID j_init_encode_method_;
>-  jmethodID j_get_input_buffers_method_;
>-  jmethodID j_dequeue_input_buffer_method_;
>-  jmethodID j_encode_buffer_method_;
>-  jmethodID j_encode_texture_method_;
>-  jmethodID j_encode_frame_method_;
>-  jmethodID j_release_method_;
>-  jmethodID j_set_rates_method_;
>-  jmethodID j_dequeue_output_buffer_method_;
>-  jmethodID j_release_output_buffer_method_;
>-  jfieldID j_color_format_field_;
>-  jfieldID j_info_index_field_;
>-  jfieldID j_info_buffer_field_;
>-  jfieldID j_info_is_key_frame_field_;
>-  jfieldID j_info_presentation_timestamp_us_field_;
>-
>-  ScopedGlobalRef<jclass> j_video_frame_texture_buffer_class_;
>+  ScopedJavaGlobalRef<jobject> j_media_codec_video_encoder_;
> 
>   // State that is valid only between InitEncode() and the next Release().
>   int width_;   // Frame width in pixels.
>@@ -238,20 +211,20 @@ class MediaCodecVideoEncoder : public VideoEncoder {
>   bool inited_;
>   bool use_surface_;
>   enum libyuv::FourCC encoder_fourcc_;  // Encoder color space format.
>-  int last_set_bitrate_kbps_;  // Last-requested bitrate in kbps.
>-  int last_set_fps_;  // Last-requested frame rate.
>-  int64_t current_timestamp_us_;  // Current frame timestamps in us.
>-  int frames_received_;  // Number of frames received by encoder.
>-  int frames_encoded_;  // Number of frames encoded by encoder.
>-  int frames_dropped_media_encoder_;  // Number of frames dropped by encoder.
>+  uint32_t last_set_bitrate_kbps_;      // Last-requested bitrate in kbps.
>+  uint32_t last_set_fps_;               // Last-requested frame rate.
>+  int64_t current_timestamp_us_;        // Current frame timestamps in us.
>+  int frames_received_;                 // Number of frames received by encoder.
>+  int frames_encoded_;                  // Number of frames encoded by encoder.
>+  int frames_dropped_media_encoder_;    // Number of frames dropped by encoder.
>   // Number of dropped frames caused by full queue.
>   int consecutive_full_queue_frame_drops_;
>   int64_t stat_start_time_ms_;  // Start time for statistics.
>   int current_frames_;  // Number of frames in the current statistics interval.
>-  int current_bytes_;  // Encoded bytes in the current statistics interval.
>+  int current_bytes_;   // Encoded bytes in the current statistics interval.
>   int current_acc_qp_;  // Accumulated QP in the current statistics interval.
>   int current_encoding_time_ms_;  // Overall encoding time in the current second
>-  int64_t last_input_timestamp_ms_;  // Timestamp of last received yuv frame.
>+  int64_t last_input_timestamp_ms_;   // Timestamp of last received yuv frame.
>   int64_t last_output_timestamp_ms_;  // Timestamp of last encoded frame.
>   // Holds the task while the polling loop is paused.
>   std::unique_ptr<rtc::QueuedTask> encode_task_;
>@@ -289,7 +262,7 @@ class MediaCodecVideoEncoder : public VideoEncoder {
>   bool scale_;
>   H264::Profile profile_;
>   // Global references; must be deleted in Release().
>-  std::vector<jobject> input_buffers_;
>+  std::vector<ScopedJavaGlobalRef<jobject>> input_buffers_;
>   H264BitstreamParser h264_bitstream_parser_;
> 
>   // VP9 variables to populate codec specific structure.
>@@ -297,9 +270,7 @@ class MediaCodecVideoEncoder : public VideoEncoder {
>                     // non-flexible VP9 mode.
>   size_t gof_idx_;
> 
>-  // EGL context - owned by factory, should not be allocated/destroyed
>-  // by MediaCodecVideoEncoder.
>-  jobject egl_context_;
>+  const bool has_egl_context_;
> 
>   // Temporary fix for VP8.
>   // Sends a key frame if frames are largely spaced apart (possibly
>@@ -308,10 +279,6 @@ class MediaCodecVideoEncoder : public VideoEncoder {
>   int frames_received_since_last_key_;
>   VideoCodecMode codec_mode_;
> 
>-  // RTP state.
>-  uint16_t picture_id_;
>-  uint8_t tl0_pic_idx_;
>-
>   bool sw_fallback_required_;
> 
>   // All other member variables should be before WeakPtrFactory. Valid only from
>@@ -327,80 +294,18 @@ MediaCodecVideoEncoder::~MediaCodecVideoEncoder() {
> }
> 
> MediaCodecVideoEncoder::MediaCodecVideoEncoder(JNIEnv* jni,
>-                                               const cricket::VideoCodec& codec,
>-                                               jobject egl_context)
>-    : codec_(codec),
>+                                               const SdpVideoFormat& format,
>+                                               bool has_egl_context)
>+    : format_(format),
>       callback_(NULL),
>-      j_media_codec_video_encoder_class_(
>-          jni,
>-          FindClass(jni, "org/webrtc/MediaCodecVideoEncoder")),
>       j_media_codec_video_encoder_(
>           jni,
>-          jni->NewObject(*j_media_codec_video_encoder_class_,
>-                         GetMethodID(jni,
>-                                     *j_media_codec_video_encoder_class_,
>-                                     "<init>",
>-                                     "()V"))),
>-      j_video_frame_texture_buffer_class_(
>-          jni,
>-          FindClass(jni, "org/webrtc/VideoFrame$TextureBuffer")),
>+          Java_MediaCodecVideoEncoder_Constructor(jni)),
>       inited_(false),
>       use_surface_(false),
>-      egl_context_(egl_context),
>+      has_egl_context_(has_egl_context),
>       sw_fallback_required_(false) {
>   encoder_queue_checker_.Detach();
>-
>-  jclass j_output_buffer_info_class =
>-      FindClass(jni, "org/webrtc/MediaCodecVideoEncoder$OutputBufferInfo");
>-  j_init_encode_method_ =
>-      GetMethodID(jni, *j_media_codec_video_encoder_class_, "initEncode",
>-                  "(Lorg/webrtc/MediaCodecVideoEncoder$VideoCodecType;"
>-                  "IIIIILorg/webrtc/EglBase14$Context;)Z");
>-  j_get_input_buffers_method_ = GetMethodID(
>-      jni,
>-      *j_media_codec_video_encoder_class_,
>-      "getInputBuffers",
>-      "()[Ljava/nio/ByteBuffer;");
>-  j_dequeue_input_buffer_method_ = GetMethodID(
>-      jni, *j_media_codec_video_encoder_class_, "dequeueInputBuffer", "()I");
>-  j_encode_buffer_method_ = GetMethodID(
>-      jni, *j_media_codec_video_encoder_class_, "encodeBuffer", "(ZIIJ)Z");
>-  j_encode_texture_method_ = GetMethodID(
>-        jni, *j_media_codec_video_encoder_class_, "encodeTexture",
>-        "(ZI[FJ)Z");
>-  j_encode_frame_method_ =
>-      GetMethodID(jni, *j_media_codec_video_encoder_class_, "encodeFrame",
>-                  "(JZLorg/webrtc/VideoFrame;I)Z");
>-  j_release_method_ =
>-      GetMethodID(jni, *j_media_codec_video_encoder_class_, "release", "()V");
>-  j_set_rates_method_ = GetMethodID(
>-      jni, *j_media_codec_video_encoder_class_, "setRates", "(II)Z");
>-  j_dequeue_output_buffer_method_ = GetMethodID(
>-      jni,
>-      *j_media_codec_video_encoder_class_,
>-      "dequeueOutputBuffer",
>-      "()Lorg/webrtc/MediaCodecVideoEncoder$OutputBufferInfo;");
>-  j_release_output_buffer_method_ = GetMethodID(
>-      jni, *j_media_codec_video_encoder_class_, "releaseOutputBuffer", "(I)Z");
>-
>-  j_color_format_field_ =
>-      GetFieldID(jni, *j_media_codec_video_encoder_class_, "colorFormat", "I");
>-  j_info_index_field_ =
>-      GetFieldID(jni, j_output_buffer_info_class, "index", "I");
>-  j_info_buffer_field_ = GetFieldID(
>-      jni, j_output_buffer_info_class, "buffer", "Ljava/nio/ByteBuffer;");
>-  j_info_is_key_frame_field_ =
>-      GetFieldID(jni, j_output_buffer_info_class, "isKeyFrame", "Z");
>-  j_info_presentation_timestamp_us_field_ = GetFieldID(
>-      jni, j_output_buffer_info_class, "presentationTimestampUs", "J");
>-  if (CheckException(jni)) {
>-    ALOGW << "MediaCodecVideoEncoder ctor failed.";
>-    ProcessHWError(true /* reset_if_fallback_unavailable */);
>-  }
>-
>-  Random random(rtc::TimeMicros());
>-  picture_id_ = random.Rand<uint16_t>() & 0x7FFF;
>-  tl0_pic_idx_ = random.Rand<uint8_t>();
> }
> 
> int32_t MediaCodecVideoEncoder::InitEncode(const VideoCodec* codec_settings,
>@@ -439,8 +344,8 @@ int32_t MediaCodecVideoEncoder::InitEncode(const VideoCodec* codec_settings,
>   // Check allowed H.264 profile
>   profile_ = H264::Profile::kProfileBaseline;
>   if (codec_type == kVideoCodecH264) {
>-    const rtc::Optional<H264::ProfileLevelId> profile_level_id =
>-        H264::ParseSdpProfileLevelId(codec_.params);
>+    const absl::optional<H264::ProfileLevelId> profile_level_id =
>+        H264::ParseSdpProfileLevelId(format_.parameters);
>     RTC_DCHECK(profile_level_id);
>     profile_ = profile_level_id->profile;
>     ALOGD << "H.264 profile: " << profile_;
>@@ -448,7 +353,8 @@ int32_t MediaCodecVideoEncoder::InitEncode(const VideoCodec* codec_settings,
> 
>   return InitEncodeInternal(
>       init_width, init_height, codec_settings->startBitrate,
>-      codec_settings->maxFramerate, codec_settings->expect_encode_from_texture);
>+      codec_settings->maxFramerate,
>+      codec_settings->expect_encode_from_texture && has_egl_context_);
> }
> 
> int32_t MediaCodecVideoEncoder::SetChannelParameters(uint32_t /* packet_loss */,
>@@ -516,12 +422,14 @@ bool MediaCodecVideoEncoder::EncodeTask::Run() {
>   return false;
> }
> 
>-bool IsFormatSupported(
>-    const std::vector<webrtc::SdpVideoFormat>& supported_formats,
>-    const std::string& name) {
>-  for (const webrtc::SdpVideoFormat& supported_format : supported_formats) {
>-    if (cricket::CodecNamesEq(name, supported_format.name))
>+bool IsFormatSupported(const std::vector<SdpVideoFormat>& supported_formats,
>+                       const SdpVideoFormat& format) {
>+  for (const SdpVideoFormat& supported_format : supported_formats) {
>+    if (cricket::IsSameCodec(format.name, format.parameters,
>+                             supported_format.name,
>+                             supported_format.parameters)) {
>       return true;
>+    }
>   }
>   return false;
> }
>@@ -530,7 +438,7 @@ bool MediaCodecVideoEncoder::ProcessHWError(
>     bool reset_if_fallback_unavailable) {
>   ALOGE << "ProcessHWError";
>   if (IsFormatSupported(InternalEncoderFactory().GetSupportedFormats(),
>-                        codec_.name)) {
>+                        format_)) {
>     ALOGE << "Fallback to SW encoder.";
>     sw_fallback_required_ = true;
>     return false;
>@@ -548,7 +456,7 @@ int32_t MediaCodecVideoEncoder::ProcessHWErrorOnEncode() {
> }
> 
> VideoCodecType MediaCodecVideoEncoder::GetCodecType() const {
>-  return PayloadStringToCodecType(codec_.name);
>+  return PayloadStringToCodecType(format_.name);
> }
> 
> int32_t MediaCodecVideoEncoder::InitEncodeInternal(int width,
>@@ -560,7 +468,7 @@ int32_t MediaCodecVideoEncoder::InitEncodeInternal(int width,
>   if (sw_fallback_required_) {
>     return WEBRTC_VIDEO_CODEC_OK;
>   }
>-  RTC_CHECK(!use_surface || egl_context_ != nullptr) << "EGL context not set.";
>+  RTC_CHECK(!use_surface || has_egl_context_) << "EGL context not set.";
>   JNIEnv* jni = AttachCurrentThreadIfNeeded();
>   ScopedLocalRefFrame local_ref_frame(jni);
> 
>@@ -603,12 +511,12 @@ int32_t MediaCodecVideoEncoder::InitEncodeInternal(int width,
>   frames_received_since_last_key_ = kMinKeyFrameInterval;
> 
>   // We enforce no extra stride/padding in the format creation step.
>-  jobject j_video_codec_enum = JavaEnumFromIndexAndClassName(
>-      jni, "MediaCodecVideoEncoder$VideoCodecType", codec_type);
>-  const bool encode_status = jni->CallBooleanMethod(
>-      *j_media_codec_video_encoder_, j_init_encode_method_, j_video_codec_enum,
>-      profile_, width, height, kbps, fps,
>-      (use_surface ? egl_context_ : nullptr));
>+  ScopedJavaLocalRef<jobject> j_video_codec_enum =
>+      Java_VideoCodecType_fromNativeIndex(jni, codec_type);
>+  const bool encode_status = Java_MediaCodecVideoEncoder_initEncode(
>+      jni, j_media_codec_video_encoder_, j_video_codec_enum, profile_, width,
>+      height, kbps, fps, use_surface);
>+
>   if (!encode_status) {
>     ALOGE << "Failed to configure encoder.";
>     ProcessHWError(false /* reset_if_fallback_unavailable */);
>@@ -621,9 +529,9 @@ int32_t MediaCodecVideoEncoder::InitEncodeInternal(int width,
>   }
> 
>   if (!use_surface) {
>-    jobjectArray input_buffers = reinterpret_cast<jobjectArray>(
>-        jni->CallObjectMethod(*j_media_codec_video_encoder_,
>-            j_get_input_buffers_method_));
>+    ScopedJavaLocalRef<jobjectArray> input_buffers =
>+        Java_MediaCodecVideoEncoder_getInputBuffers(
>+            jni, j_media_codec_video_encoder_);
>     if (CheckException(jni)) {
>       ALOGE << "Exception in get input buffers.";
>       ProcessHWError(false /* reset_if_fallback_unavailable */);
>@@ -635,8 +543,8 @@ int32_t MediaCodecVideoEncoder::InitEncodeInternal(int width,
>       return WEBRTC_VIDEO_CODEC_ERROR;
>     }
> 
>-    switch (GetIntField(jni, *j_media_codec_video_encoder_,
>-        j_color_format_field_)) {
>+    switch (Java_MediaCodecVideoEncoder_getColorFormat(
>+        jni, j_media_codec_video_encoder_)) {
>       case COLOR_FormatYUV420Planar:
>         encoder_fourcc_ = libyuv::FOURCC_YU12;
>         break;
>@@ -650,15 +558,15 @@ int32_t MediaCodecVideoEncoder::InitEncodeInternal(int width,
>         ProcessHWError(false /* reset_if_fallback_unavailable */);
>         return WEBRTC_VIDEO_CODEC_ERROR;
>     }
>-    size_t num_input_buffers = jni->GetArrayLength(input_buffers);
>+
>     RTC_CHECK(input_buffers_.empty())
>         << "Unexpected double InitEncode without Release";
>-    input_buffers_.resize(num_input_buffers);
>-    for (size_t i = 0; i < num_input_buffers; ++i) {
>-      input_buffers_[i] =
>-          jni->NewGlobalRef(jni->GetObjectArrayElement(input_buffers, i));
>-      int64_t yuv_buffer_capacity =
>-          jni->GetDirectBufferCapacity(input_buffers_[i]);
>+    input_buffers_ = JavaToNativeVector<ScopedJavaGlobalRef<jobject>>(
>+        jni, input_buffers, [](JNIEnv* env, const JavaRef<jobject>& o) {
>+          return ScopedJavaGlobalRef<jobject>(env, o);
>+        });
>+    for (const ScopedJavaGlobalRef<jobject>& buffer : input_buffers_) {
>+      int64_t yuv_buffer_capacity = jni->GetDirectBufferCapacity(buffer.obj());
>       if (CheckException(jni)) {
>         ALOGE << "Exception in get direct buffer capacity.";
>         ProcessHWError(false /* reset_if_fallback_unavailable */);
>@@ -696,7 +604,7 @@ int32_t MediaCodecVideoEncoder::Encode(
>   }
> 
>   bool send_key_frame = false;
>-  if (codec_mode_ == kRealtimeVideo) {
>+  if (codec_mode_ == VideoCodecMode::kRealtimeVideo) {
>     ++frames_received_since_last_key_;
>     int64_t now_ms = rtc::TimeMillis();
>     if (last_frame_received_ms_ != -1 &&
>@@ -772,8 +680,8 @@ int32_t MediaCodecVideoEncoder::Encode(
> 
>   int j_input_buffer_index = -1;
>   if (!use_surface_) {
>-    j_input_buffer_index = jni->CallIntMethod(*j_media_codec_video_encoder_,
>-                                              j_dequeue_input_buffer_method_);
>+    j_input_buffer_index = Java_MediaCodecVideoEncoder_dequeueInputBuffer(
>+        jni, j_media_codec_video_encoder_);
>     if (CheckException(jni)) {
>       ALOGE << "Exception in dequeu input buffer.";
>       return ProcessHWErrorOnEncode();
>@@ -801,22 +709,10 @@ int32_t MediaCodecVideoEncoder::Encode(
>     encode_status =
>         EncodeByteBuffer(jni, key_frame, input_frame, j_input_buffer_index);
>   } else {
>-    AndroidVideoFrameBuffer* android_buffer =
>-        static_cast<AndroidVideoFrameBuffer*>(
>-            input_frame.video_frame_buffer().get());
>-    switch (android_buffer->android_type()) {
>-      case AndroidVideoFrameBuffer::AndroidType::kTextureBuffer:
>-        encode_status = EncodeTexture(jni, key_frame, input_frame);
>-        break;
>-      case AndroidVideoFrameBuffer::AndroidType::kJavaBuffer:
>-        encode_status =
>-            EncodeJavaFrame(jni, key_frame, NativeToJavaFrame(jni, input_frame),
>-                            j_input_buffer_index);
>-        break;
>-      default:
>-        RTC_NOTREACHED();
>-        return WEBRTC_VIDEO_CODEC_ERROR;
>-    }
>+    ScopedJavaLocalRef<jobject> j_frame = NativeToJavaVideoFrame(jni, frame);
>+    encode_status =
>+        EncodeJavaFrame(jni, key_frame, j_frame, j_input_buffer_index);
>+    ReleaseJavaVideoFrame(jni, j_frame);
>   }
> 
>   if (!encode_status) {
>@@ -856,16 +752,15 @@ bool MediaCodecVideoEncoder::MaybeReconfigureEncoder(JNIEnv* jni,
>       frame.width() != width_ || frame.height() != height_;
> 
>   if (reconfigure_due_to_format) {
>-      ALOGD << "Reconfigure encoder due to format change. "
>-            << (use_surface_ ?
>-                "Reconfiguring to encode from byte buffer." :
>-                "Reconfiguring to encode from texture.");
>-      LogStatistics(true);
>+    ALOGD << "Reconfigure encoder due to format change. "
>+          << (use_surface_ ? "Reconfiguring to encode from byte buffer."
>+                           : "Reconfiguring to encode from texture.");
>+    LogStatistics(true);
>   }
>   if (reconfigure_due_to_size) {
>     ALOGW << "Reconfigure encoder due to frame resolution change from "
>-        << width_ << " x " << height_ << " to " << frame.width() << " x "
>-        << frame.height();
>+          << width_ << " x " << height_ << " to " << frame.width() << " x "
>+          << frame.height();
>     LogStatistics(true);
>     width_ = frame.width();
>     height_ = frame.height();
>@@ -885,20 +780,9 @@ bool MediaCodecVideoEncoder::IsTextureFrame(JNIEnv* jni,
>   if (frame.video_frame_buffer()->type() != VideoFrameBuffer::Type::kNative) {
>     return false;
>   }
>-
>-  AndroidVideoFrameBuffer* android_buffer =
>-      static_cast<AndroidVideoFrameBuffer*>(frame.video_frame_buffer().get());
>-  switch (android_buffer->android_type()) {
>-    case AndroidVideoFrameBuffer::AndroidType::kTextureBuffer:
>-      return true;
>-    case AndroidVideoFrameBuffer::AndroidType::kJavaBuffer:
>-      return jni->IsInstanceOf(static_cast<AndroidVideoBuffer*>(android_buffer)
>-                                   ->video_frame_buffer(),
>-                               *j_video_frame_texture_buffer_class_);
>-    default:
>-      RTC_NOTREACHED();
>-      return false;
>-  }
>+  return Java_MediaCodecVideoEncoder_isTextureBuffer(
>+      jni, static_cast<AndroidVideoBuffer*>(frame.video_frame_buffer().get())
>+               ->video_frame_buffer());
> }
> 
> bool MediaCodecVideoEncoder::EncodeByteBuffer(JNIEnv* jni,
>@@ -916,9 +800,9 @@ bool MediaCodecVideoEncoder::EncodeByteBuffer(JNIEnv* jni,
>                        i420_buffer->StrideV())) {
>     return false;
>   }
>-  bool encode_status = jni->CallBooleanMethod(
>-      *j_media_codec_video_encoder_, j_encode_buffer_method_, key_frame,
>-      input_buffer_index, yuv_size_, current_timestamp_us_);
>+  bool encode_status = Java_MediaCodecVideoEncoder_encodeBuffer(
>+      jni, j_media_codec_video_encoder_, key_frame, input_buffer_index,
>+      yuv_size_, current_timestamp_us_);
>   if (CheckException(jni)) {
>     ALOGE << "Exception in encode buffer.";
>     ProcessHWError(true /* reset_if_fallback_unavailable */);
>@@ -935,9 +819,8 @@ bool MediaCodecVideoEncoder::FillInputBuffer(JNIEnv* jni,
>                                              int stride_u,
>                                              uint8_t const* buffer_v,
>                                              int stride_v) {
>-  jobject j_input_buffer = input_buffers_[input_buffer_index];
>-  uint8_t* yuv_buffer =
>-      reinterpret_cast<uint8_t*>(jni->GetDirectBufferAddress(j_input_buffer));
>+  uint8_t* yuv_buffer = reinterpret_cast<uint8_t*>(
>+      jni->GetDirectBufferAddress(input_buffers_[input_buffer_index].obj()));
>   if (CheckException(jni)) {
>     ALOGE << "Exception in get direct buffer address.";
>     ProcessHWError(true /* reset_if_fallback_unavailable */);
>@@ -952,34 +835,13 @@ bool MediaCodecVideoEncoder::FillInputBuffer(JNIEnv* jni,
>   return true;
> }
> 
>-bool MediaCodecVideoEncoder::EncodeTexture(JNIEnv* jni,
>-                                           bool key_frame,
>-                                           const VideoFrame& frame) {
>-  RTC_DCHECK_CALLED_SEQUENTIALLY(&encoder_queue_checker_);
>-  RTC_CHECK(use_surface_);
>-  NativeHandleImpl handle =
>-      static_cast<AndroidTextureBuffer*>(frame.video_frame_buffer().get())
>-          ->native_handle_impl();
>-
>-  jfloatArray sampling_matrix = handle.sampling_matrix.ToJava(jni);
>-  bool encode_status = jni->CallBooleanMethod(
>-      *j_media_codec_video_encoder_, j_encode_texture_method_, key_frame,
>-      handle.oes_texture_id, sampling_matrix, current_timestamp_us_);
>-  if (CheckException(jni)) {
>-    ALOGE << "Exception in encode texture.";
>-    ProcessHWError(true /* reset_if_fallback_unavailable */);
>-    return false;
>-  }
>-  return encode_status;
>-}
>-
> bool MediaCodecVideoEncoder::EncodeJavaFrame(JNIEnv* jni,
>                                              bool key_frame,
>-                                             jobject frame,
>+                                             const JavaRef<jobject>& frame,
>                                              int input_buffer_index) {
>-  bool encode_status = jni->CallBooleanMethod(
>-      *j_media_codec_video_encoder_, j_encode_frame_method_,
>-      jlongFromPointer(this), key_frame, frame, input_buffer_index);
>+  bool encode_status = Java_MediaCodecVideoEncoder_encodeFrame(
>+      jni, j_media_codec_video_encoder_, jlongFromPointer(this), key_frame,
>+      frame, input_buffer_index, current_timestamp_us_);
>   if (CheckException(jni)) {
>     ALOGE << "Exception in encode frame.";
>     ProcessHWError(true /* reset_if_fallback_unavailable */);
>@@ -1009,10 +871,8 @@ int32_t MediaCodecVideoEncoder::Release() {
>   encode_task_.reset(nullptr);
>   weak_factory_.reset(nullptr);
>   ScopedLocalRefFrame local_ref_frame(jni);
>-  for (size_t i = 0; i < input_buffers_.size(); ++i)
>-    jni->DeleteGlobalRef(input_buffers_[i]);
>   input_buffers_.clear();
>-  jni->CallVoidMethod(*j_media_codec_video_encoder_, j_release_method_);
>+  Java_MediaCodecVideoEncoder_release(jni, j_media_codec_video_encoder_);
>   if (CheckException(jni)) {
>     ALOGE << "Exception in release.";
>     ProcessHWError(false /* reset_if_fallback_unavailable */);
>@@ -1032,7 +892,7 @@ int32_t MediaCodecVideoEncoder::Release() {
> }
> 
> int32_t MediaCodecVideoEncoder::SetRateAllocation(
>-    const BitrateAllocation& rate_allocation,
>+    const VideoBitrateAllocation& rate_allocation,
>     uint32_t frame_rate) {
>   RTC_DCHECK_CALLED_SEQUENTIALLY(&encoder_queue_checker_);
>   const uint32_t new_bit_rate = rate_allocation.get_sum_kbps();
>@@ -1051,9 +911,10 @@ int32_t MediaCodecVideoEncoder::SetRateAllocation(
>   if (frame_rate > 0) {
>     last_set_fps_ = frame_rate;
>   }
>-  bool ret =
>-      jni->CallBooleanMethod(*j_media_codec_video_encoder_, j_set_rates_method_,
>-                             last_set_bitrate_kbps_, last_set_fps_);
>+  bool ret = Java_MediaCodecVideoEncoder_setRates(
>+      jni, j_media_codec_video_encoder_,
>+      rtc::dchecked_cast<int>(last_set_bitrate_kbps_),
>+      rtc::dchecked_cast<int>(last_set_fps_));
>   if (CheckException(jni) || !ret) {
>     ProcessHWError(true /* reset_if_fallback_unavailable */);
>     return sw_fallback_required_ ? WEBRTC_VIDEO_CODEC_OK
>@@ -1062,37 +923,13 @@ int32_t MediaCodecVideoEncoder::SetRateAllocation(
>   return WEBRTC_VIDEO_CODEC_OK;
> }
> 
>-int MediaCodecVideoEncoder::GetOutputBufferInfoIndex(
>-    JNIEnv* jni,
>-    jobject j_output_buffer_info) {
>-  return GetIntField(jni, j_output_buffer_info, j_info_index_field_);
>-}
>-
>-jobject MediaCodecVideoEncoder::GetOutputBufferInfoBuffer(
>-    JNIEnv* jni,
>-    jobject j_output_buffer_info) {
>-  return GetObjectField(jni, j_output_buffer_info, j_info_buffer_field_);
>-}
>-
>-bool MediaCodecVideoEncoder::GetOutputBufferInfoIsKeyFrame(
>-    JNIEnv* jni,
>-    jobject j_output_buffer_info) {
>-  return GetBooleanField(jni, j_output_buffer_info, j_info_is_key_frame_field_);
>-}
>-
>-jlong MediaCodecVideoEncoder::GetOutputBufferInfoPresentationTimestampUs(
>-    JNIEnv* jni,
>-    jobject j_output_buffer_info) {
>-  return GetLongField(
>-      jni, j_output_buffer_info, j_info_presentation_timestamp_us_field_);
>-}
>-
> bool MediaCodecVideoEncoder::DeliverPendingOutputs(JNIEnv* jni) {
>   RTC_DCHECK_CALLED_SEQUENTIALLY(&encoder_queue_checker_);
> 
>   while (true) {
>-    jobject j_output_buffer_info = jni->CallObjectMethod(
>-        *j_media_codec_video_encoder_, j_dequeue_output_buffer_method_);
>+    ScopedJavaLocalRef<jobject> j_output_buffer_info =
>+        Java_MediaCodecVideoEncoder_dequeueOutputBuffer(
>+            jni, j_media_codec_video_encoder_);
>     if (CheckException(jni)) {
>       ALOGE << "Exception in set dequeue output buffer.";
>       ProcessHWError(true /* reset_if_fallback_unavailable */);
>@@ -1103,22 +940,24 @@ bool MediaCodecVideoEncoder::DeliverPendingOutputs(JNIEnv* jni) {
>     }
> 
>     int output_buffer_index =
>-        GetOutputBufferInfoIndex(jni, j_output_buffer_info);
>+        Java_OutputBufferInfo_getIndex(jni, j_output_buffer_info);
>     if (output_buffer_index == -1) {
>       ProcessHWError(true /* reset_if_fallback_unavailable */);
>       return false;
>     }
> 
>     // Get key and config frame flags.
>-    jobject j_output_buffer =
>-        GetOutputBufferInfoBuffer(jni, j_output_buffer_info);
>-    bool key_frame = GetOutputBufferInfoIsKeyFrame(jni, j_output_buffer_info);
>+    ScopedJavaLocalRef<jobject> j_output_buffer =
>+        Java_OutputBufferInfo_getBuffer(jni, j_output_buffer_info);
>+    bool key_frame =
>+        Java_OutputBufferInfo_isKeyFrame(jni, j_output_buffer_info);
> 
>     // Get frame timestamps from a queue - for non config frames only.
>     int64_t encoding_start_time_ms = 0;
>     int64_t frame_encoding_time_ms = 0;
>     last_output_timestamp_ms_ =
>-        GetOutputBufferInfoPresentationTimestampUs(jni, j_output_buffer_info) /
>+        Java_OutputBufferInfo_getPresentationTimestampUs(jni,
>+                                                         j_output_buffer_info) /
>         rtc::kNumMicrosecsPerMillisec;
>     if (!input_frame_infos_.empty()) {
>       const InputFrameInfo& frame_info = input_frame_infos_.front();
>@@ -1130,9 +969,9 @@ bool MediaCodecVideoEncoder::DeliverPendingOutputs(JNIEnv* jni) {
>     }
> 
>     // Extract payload.
>-    size_t payload_size = jni->GetDirectBufferCapacity(j_output_buffer);
>+    size_t payload_size = jni->GetDirectBufferCapacity(j_output_buffer.obj());
>     uint8_t* payload = reinterpret_cast<uint8_t*>(
>-        jni->GetDirectBufferAddress(j_output_buffer));
>+        jni->GetDirectBufferAddress(j_output_buffer.obj()));
>     if (CheckException(jni)) {
>       ALOGE << "Exception in get direct buffer address.";
>       ProcessHWError(true /* reset_if_fallback_unavailable */);
>@@ -1154,29 +993,25 @@ bool MediaCodecVideoEncoder::DeliverPendingOutputs(JNIEnv* jni) {
>       image->content_type_ = (codec_mode_ == VideoCodecMode::kScreensharing)
>                                  ? VideoContentType::SCREENSHARE
>                                  : VideoContentType::UNSPECIFIED;
>-      image->timing_.flags = TimingFrameFlags::kInvalid;
>+      image->timing_.flags = VideoSendTiming::kInvalid;
>       image->_frameType = (key_frame ? kVideoFrameKey : kVideoFrameDelta);
>       image->_completeFrame = true;
>       CodecSpecificInfo info;
>       memset(&info, 0, sizeof(info));
>       info.codecType = codec_type;
>       if (codec_type == kVideoCodecVP8) {
>-        info.codecSpecific.VP8.pictureId = picture_id_;
>         info.codecSpecific.VP8.nonReference = false;
>         info.codecSpecific.VP8.simulcastIdx = 0;
>         info.codecSpecific.VP8.temporalIdx = kNoTemporalIdx;
>         info.codecSpecific.VP8.layerSync = false;
>-        info.codecSpecific.VP8.tl0PicIdx = kNoTl0PicIdx;
>         info.codecSpecific.VP8.keyIdx = kNoKeyIdx;
>       } else if (codec_type == kVideoCodecVP9) {
>         if (key_frame) {
>           gof_idx_ = 0;
>         }
>-        info.codecSpecific.VP9.picture_id = picture_id_;
>         info.codecSpecific.VP9.inter_pic_predicted = key_frame ? false : true;
>         info.codecSpecific.VP9.flexible_mode = false;
>         info.codecSpecific.VP9.ss_data_available = key_frame ? true : false;
>-        info.codecSpecific.VP9.tl0_pic_idx = tl0_pic_idx_++;
>         info.codecSpecific.VP9.temporal_idx = kNoTemporalIdx;
>         info.codecSpecific.VP9.spatial_idx = kNoSpatialIdx;
>         info.codecSpecific.VP9.temporal_up_switch = true;
>@@ -1184,6 +1019,8 @@ bool MediaCodecVideoEncoder::DeliverPendingOutputs(JNIEnv* jni) {
>         info.codecSpecific.VP9.gof_idx =
>             static_cast<uint8_t>(gof_idx_++ % gof_.num_frames_in_gof);
>         info.codecSpecific.VP9.num_spatial_layers = 1;
>+        info.codecSpecific.VP9.first_frame_in_picture = true;
>+        info.codecSpecific.VP9.end_of_picture = true;
>         info.codecSpecific.VP9.spatial_layer_resolution_present = false;
>         if (info.codecSpecific.VP9.ss_data_available) {
>           info.codecSpecific.VP9.spatial_layer_resolution_present = true;
>@@ -1192,7 +1029,6 @@ bool MediaCodecVideoEncoder::DeliverPendingOutputs(JNIEnv* jni) {
>           info.codecSpecific.VP9.gof.CopyGofInfoVP9(gof_);
>         }
>       }
>-      picture_id_ = (picture_id_ + 1) & 0x7FFF;
> 
>       // Generate a header describing a single fragment.
>       RTPFragmentationHeader header;
>@@ -1228,9 +1064,9 @@ bool MediaCodecVideoEncoder::DeliverPendingOutputs(JNIEnv* jni) {
>             H264::FindNaluIndices(payload, payload_size);
>         if (nalu_idxs.empty()) {
>           ALOGE << "Start code is not found!";
>-          ALOGE << "Data:" <<  image->_buffer[0] << " " << image->_buffer[1]
>-              << " " << image->_buffer[2] << " " << image->_buffer[3]
>-              << " " << image->_buffer[4] << " " << image->_buffer[5];
>+          ALOGE << "Data:" << image->_buffer[0] << " " << image->_buffer[1]
>+                << " " << image->_buffer[2] << " " << image->_buffer[3] << " "
>+                << image->_buffer[4] << " " << image->_buffer[5];
>           ProcessHWError(true /* reset_if_fallback_unavailable */);
>           return false;
>         }
>@@ -1247,9 +1083,8 @@ bool MediaCodecVideoEncoder::DeliverPendingOutputs(JNIEnv* jni) {
>     }
> 
>     // Return output buffer back to the encoder.
>-    bool success = jni->CallBooleanMethod(*j_media_codec_video_encoder_,
>-                                          j_release_output_buffer_method_,
>-                                          output_buffer_index);
>+    bool success = Java_MediaCodecVideoEncoder_releaseOutputBuffer(
>+        jni, j_media_codec_video_encoder_, output_buffer_index);
>     if (CheckException(jni) || !success) {
>       ProcessHWError(true /* reset_if_fallback_unavailable */);
>       return false;
>@@ -1285,21 +1120,21 @@ bool MediaCodecVideoEncoder::DeliverPendingOutputs(JNIEnv* jni) {
> 
> void MediaCodecVideoEncoder::LogStatistics(bool force_log) {
>   int statistic_time_ms = rtc::TimeMillis() - stat_start_time_ms_;
>-  if ((statistic_time_ms >= kMediaCodecStatisticsIntervalMs || force_log)
>-      && statistic_time_ms > 0) {
>+  if ((statistic_time_ms >= kMediaCodecStatisticsIntervalMs || force_log) &&
>+      statistic_time_ms > 0) {
>     // Prevent division by zero.
>     int current_frames_divider = current_frames_ != 0 ? current_frames_ : 1;
> 
>     int current_bitrate = current_bytes_ * 8 / statistic_time_ms;
>     int current_fps =
>         (current_frames_ * 1000 + statistic_time_ms / 2) / statistic_time_ms;
>-    ALOGD << "Encoded frames: " << frames_encoded_ <<
>-        ". Bitrate: " << current_bitrate <<
>-        ", target: " << last_set_bitrate_kbps_ << " kbps" <<
>-        ", fps: " << current_fps <<
>-        ", encTime: " << (current_encoding_time_ms_ / current_frames_divider) <<
>-        ". QP: " << (current_acc_qp_ / current_frames_divider) <<
>-        " for last " << statistic_time_ms << " ms.";
>+    ALOGD << "Encoded frames: " << frames_encoded_
>+          << ". Bitrate: " << current_bitrate
>+          << ", target: " << last_set_bitrate_kbps_ << " kbps"
>+          << ", fps: " << current_fps << ", encTime: "
>+          << (current_encoding_time_ms_ / current_frames_divider)
>+          << ". QP: " << (current_acc_qp_ / current_frames_divider)
>+          << " for last " << statistic_time_ms << " ms.";
>     stat_start_time_ms_ = rtc::TimeMillis();
>     current_frames_ = 0;
>     current_bytes_ = 0;
>@@ -1310,8 +1145,11 @@ void MediaCodecVideoEncoder::LogStatistics(bool force_log) {
> 
> VideoEncoder::ScalingSettings MediaCodecVideoEncoder::GetScalingSettings()
>     const {
>+  if (!scale_)
>+    return VideoEncoder::ScalingSettings::kOff;
>+
>+  const VideoCodecType codec_type = GetCodecType();
>   if (field_trial::IsEnabled(kCustomQPThresholdsFieldTrial)) {
>-    const VideoCodecType codec_type = GetCodecType();
>     std::string experiment_string =
>         field_trial::FindFullName(kCustomQPThresholdsFieldTrial);
>     ALOGD << "QP custom thresholds: " << experiment_string << " for codec "
>@@ -1329,162 +1167,61 @@ VideoEncoder::ScalingSettings MediaCodecVideoEncoder::GetScalingSettings()
>       RTC_CHECK_GT(high_h264_qp_threshold, low_h264_qp_threshold);
>       RTC_CHECK_GT(low_h264_qp_threshold, 0);
>       if (codec_type == kVideoCodecVP8) {
>-        return VideoEncoder::ScalingSettings(scale_, low_vp8_qp_threshold,
>+        return VideoEncoder::ScalingSettings(low_vp8_qp_threshold,
>                                              high_vp8_qp_threshold);
>       } else if (codec_type == kVideoCodecH264) {
>-        return VideoEncoder::ScalingSettings(scale_, low_h264_qp_threshold,
>+        return VideoEncoder::ScalingSettings(low_h264_qp_threshold,
>                                              high_h264_qp_threshold);
>       }
>     }
>   }
>-  return VideoEncoder::ScalingSettings(scale_);
>-}
>-
>-const char* MediaCodecVideoEncoder::ImplementationName() const {
>-  return "MediaCodec";
>-}
>-
>-MediaCodecVideoEncoderFactory::MediaCodecVideoEncoderFactory()
>-    : egl_context_(nullptr) {
>-  JNIEnv* jni = AttachCurrentThreadIfNeeded();
>-  ScopedLocalRefFrame local_ref_frame(jni);
>-  jclass j_encoder_class = FindClass(jni, "org/webrtc/MediaCodecVideoEncoder");
>-  jclass j_decoder_class = FindClass(jni, "org/webrtc/MediaCodecVideoDecoder");
>-  supported_codecs_.clear();
>-
>-  bool is_vp8_hw_supported = jni->CallStaticBooleanMethod(
>-      j_encoder_class,
>-      GetStaticMethodID(jni, j_encoder_class, "isVp8HwSupported", "()Z"));
>-  CHECK_EXCEPTION(jni);
>-  if (is_vp8_hw_supported) {
>-    ALOGD << "VP8 HW Encoder supported.";
>-    supported_codecs_.push_back(cricket::VideoCodec(cricket::kVp8CodecName));
>-  }
>-
>-  bool is_vp9_hw_supported = jni->CallStaticBooleanMethod(
>-      j_encoder_class,
>-      GetStaticMethodID(jni, j_encoder_class, "isVp9HwSupported", "()Z"));
>-  CHECK_EXCEPTION(jni);
>-  if (is_vp9_hw_supported) {
>-    ALOGD << "VP9 HW Encoder supported.";
>-    supported_codecs_.push_back(cricket::VideoCodec(cricket::kVp9CodecName));
>-  }
>-  supported_codecs_with_h264_hp_ = supported_codecs_;
>-
>-  // Check if high profile is supported by decoder. If yes, encoder can always
>-  // fall back to baseline profile as a subset as high profile.
>-  bool is_h264_high_profile_hw_supported = jni->CallStaticBooleanMethod(
>-      j_decoder_class,
>-      GetStaticMethodID(jni, j_decoder_class, "isH264HighProfileHwSupported",
>-                        "()Z"));
>-  CHECK_EXCEPTION(jni);
>-  if (is_h264_high_profile_hw_supported) {
>-    ALOGD << "H.264 High Profile HW Encoder supported.";
>-    // TODO(magjed): Enumerate actual level instead of using hardcoded level
>-    // 3.1. Level 3.1 is 1280x720@30fps which is enough for now.
>-    cricket::VideoCodec constrained_high(cricket::kH264CodecName);
>-    const H264::ProfileLevelId constrained_high_profile(
>-        H264::kProfileConstrainedHigh, H264::kLevel3_1);
>-    constrained_high.SetParam(
>-        cricket::kH264FmtpProfileLevelId,
>-        *H264::ProfileLevelIdToString(constrained_high_profile));
>-    constrained_high.SetParam(cricket::kH264FmtpLevelAsymmetryAllowed, "1");
>-    constrained_high.SetParam(cricket::kH264FmtpPacketizationMode, "1");
>-    supported_codecs_with_h264_hp_.push_back(constrained_high);
>-  }
>-
>-  bool is_h264_hw_supported = jni->CallStaticBooleanMethod(
>-      j_encoder_class,
>-      GetStaticMethodID(jni, j_encoder_class, "isH264HwSupported", "()Z"));
>-  CHECK_EXCEPTION(jni);
>-  if (is_h264_hw_supported) {
>-    ALOGD << "H.264 HW Encoder supported.";
>-    // TODO(magjed): Push Constrained High profile as well when negotiation is
>-    // ready, http://crbug/webrtc/6337. We can negotiate Constrained High
>-    // profile as long as we have decode support for it and still send Baseline
>-    // since Baseline is a subset of the High profile.
>-    cricket::VideoCodec constrained_baseline(cricket::kH264CodecName);
>-    const H264::ProfileLevelId constrained_baseline_profile(
>-        H264::kProfileConstrainedBaseline, H264::kLevel3_1);
>-    constrained_baseline.SetParam(
>-        cricket::kH264FmtpProfileLevelId,
>-        *H264::ProfileLevelIdToString(constrained_baseline_profile));
>-    constrained_baseline.SetParam(cricket::kH264FmtpLevelAsymmetryAllowed, "1");
>-    constrained_baseline.SetParam(cricket::kH264FmtpPacketizationMode, "1");
>-    supported_codecs_.push_back(constrained_baseline);
>-    supported_codecs_with_h264_hp_.push_back(constrained_baseline);
>-  }
>-}
>-
>-MediaCodecVideoEncoderFactory::~MediaCodecVideoEncoderFactory() {
>-  ALOGD << "MediaCodecVideoEncoderFactory dtor";
>-  if (egl_context_) {
>-    JNIEnv* jni = AttachCurrentThreadIfNeeded();
>-    jni->DeleteGlobalRef(egl_context_);
>-  }
>-}
>-
>-void MediaCodecVideoEncoderFactory::SetEGLContext(
>-    JNIEnv* jni, jobject egl_context) {
>-  ALOGD << "MediaCodecVideoEncoderFactory::SetEGLContext";
>-  if (egl_context_) {
>-    jni->DeleteGlobalRef(egl_context_);
>-    egl_context_ = nullptr;
>-  }
>-  egl_context_ = jni->NewGlobalRef(egl_context);
>-  if (CheckException(jni)) {
>-    ALOGE << "error calling NewGlobalRef for EGL Context.";
>-  }
>-}
>-
>-VideoEncoder* MediaCodecVideoEncoderFactory::CreateVideoEncoder(
>-    const cricket::VideoCodec& codec) {
>-  if (supported_codecs().empty()) {
>-    ALOGW << "No HW video encoder for codec " << codec.name;
>-    return nullptr;
>-  }
>-  if (FindMatchingCodec(supported_codecs(), codec)) {
>-    ALOGD << "Create HW video encoder for " << codec.name;
>-    JNIEnv* jni = AttachCurrentThreadIfNeeded();
>-    ScopedLocalRefFrame local_ref_frame(jni);
>-    return new MediaCodecVideoEncoder(jni, codec, egl_context_);
>-  }
>-  ALOGW << "Can not find HW video encoder for type " << codec.name;
>-  return nullptr;
>-}
>+  if (codec_type == kVideoCodecVP8) {
>+    // Same as in vp8_impl.cc.
>+    static const int kLowVp8QpThreshold = 29;
>+    static const int kHighVp8QpThreshold = 95;
> 
>-const std::vector<cricket::VideoCodec>&
>-MediaCodecVideoEncoderFactory::supported_codecs() const {
>-  if (field_trial::IsEnabled(kH264HighProfileFieldTrial)) {
>-    return supported_codecs_with_h264_hp_;
>-  } else {
>-    return supported_codecs_;
>+    return VideoEncoder::ScalingSettings(kLowVp8QpThreshold,
>+                                         kHighVp8QpThreshold);
>+  } else if (codec_type == kVideoCodecVP9) {
>+    // QP is obtained from VP9-bitstream, so the QP corresponds to the bitstream
>+    // range of [0, 255] and not the user-level range of [0,63].
>+    static const int kLowVp9QpThreshold = 96;
>+    static const int kHighVp9QpThreshold = 185;
>+
>+    return VideoEncoder::ScalingSettings(kLowVp9QpThreshold,
>+                                         kHighVp9QpThreshold);
>+  } else if (codec_type == kVideoCodecH264) {
>+    // Same as in h264_encoder_impl.cc.
>+    static const int kLowH264QpThreshold = 24;
>+    static const int kHighH264QpThreshold = 37;
>+
>+    return VideoEncoder::ScalingSettings(kLowH264QpThreshold,
>+                                         kHighH264QpThreshold);
>   }
>+  return VideoEncoder::ScalingSettings::kOff;
> }
> 
>-void MediaCodecVideoEncoderFactory::DestroyVideoEncoder(VideoEncoder* encoder) {
>-  ALOGD << "Destroy video encoder.";
>-  delete encoder;
>+const char* MediaCodecVideoEncoder::ImplementationName() const {
>+  return "MediaCodec";
> }
> 
>-JNI_FUNCTION_DECLARATION(void,
>-                         MediaCodecVideoEncoder_nativeFillBuffer,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jlong native_encoder,
>-                         jint input_buffer,
>-                         jobject j_buffer_y,
>-                         jint stride_y,
>-                         jobject j_buffer_u,
>-                         jint stride_u,
>-                         jobject j_buffer_v,
>-                         jint stride_v) {
>+static void JNI_MediaCodecVideoEncoder_FillInputBuffer(
>+    JNIEnv* jni,
>+    const JavaParamRef<jclass>&,
>+    jlong native_encoder,
>+    jint input_buffer,
>+    const JavaParamRef<jobject>& j_buffer_y,
>+    jint stride_y,
>+    const JavaParamRef<jobject>& j_buffer_u,
>+    jint stride_u,
>+    const JavaParamRef<jobject>& j_buffer_v,
>+    jint stride_v) {
>   uint8_t* buffer_y =
>-      static_cast<uint8_t*>(jni->GetDirectBufferAddress(j_buffer_y));
>+      static_cast<uint8_t*>(jni->GetDirectBufferAddress(j_buffer_y.obj()));
>   uint8_t* buffer_u =
>-      static_cast<uint8_t*>(jni->GetDirectBufferAddress(j_buffer_u));
>+      static_cast<uint8_t*>(jni->GetDirectBufferAddress(j_buffer_u.obj()));
>   uint8_t* buffer_v =
>-      static_cast<uint8_t*>(jni->GetDirectBufferAddress(j_buffer_v));
>+      static_cast<uint8_t*>(jni->GetDirectBufferAddress(j_buffer_v.obj()));
> 
>   RTC_DCHECK(buffer_y) << "GetDirectBufferAddress returned null. Ensure that "
>                           "getDataY returns a direct ByteBuffer.";
>@@ -1498,5 +1235,15 @@ JNI_FUNCTION_DECLARATION(void,
>                         stride_u, buffer_v, stride_v);
> }
> 
>+static jlong JNI_MediaCodecVideoEncoder_CreateEncoder(
>+    JNIEnv* env,
>+    const JavaParamRef<jclass>&,
>+    const JavaParamRef<jobject>& format,
>+    jboolean has_egl_context) {
>+  ScopedLocalRefFrame local_ref_frame(env);
>+  return jlongFromPointer(new MediaCodecVideoEncoder(
>+      env, VideoCodecInfoToSdpVideoFormat(env, format), has_egl_context));
>+}
>+
> }  // namespace jni
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidmediaencoder_jni.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidmediaencoder_jni.h
>deleted file mode 100644
>index 06d1fb48845..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidmediaencoder_jni.h
>+++ /dev/null
>@@ -1,47 +0,0 @@
>-/*
>- *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef SDK_ANDROID_SRC_JNI_ANDROIDMEDIAENCODER_JNI_H_
>-#define SDK_ANDROID_SRC_JNI_ANDROIDMEDIAENCODER_JNI_H_
>-
>-#include <vector>
>-
>-#include "sdk/android/src/jni/jni_helpers.h"
>-#include "media/engine/webrtcvideoencoderfactory.h"
>-
>-namespace webrtc {
>-namespace jni {
>-
>-// Implementation of Android MediaCodec based encoder factory.
>-class MediaCodecVideoEncoderFactory
>-    : public cricket::WebRtcVideoEncoderFactory {
>- public:
>-  MediaCodecVideoEncoderFactory();
>-  virtual ~MediaCodecVideoEncoderFactory();
>-
>-  void SetEGLContext(JNIEnv* jni, jobject egl_context);
>-
>-  // WebRtcVideoEncoderFactory implementation.
>-  VideoEncoder* CreateVideoEncoder(const cricket::VideoCodec& codec) override;
>-  const std::vector<cricket::VideoCodec>& supported_codecs() const override;
>-  void DestroyVideoEncoder(VideoEncoder* encoder) override;
>-
>- private:
>-  jobject egl_context_;
>-
>-  // Empty if platform support is lacking, const after ctor returns.
>-  std::vector<cricket::VideoCodec> supported_codecs_;
>-  std::vector<cricket::VideoCodec> supported_codecs_with_h264_hp_;
>-};
>-
>-}  // namespace jni
>-}  // namespace webrtc
>-
>-#endif  // SDK_ANDROID_SRC_JNI_ANDROIDMEDIAENCODER_JNI_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidmetrics_jni.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidmetrics.cc
>similarity index 73%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidmetrics_jni.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidmetrics.cc
>index 6391e175db5..3f96c0a8324 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidmetrics_jni.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidmetrics.cc
>@@ -12,7 +12,7 @@
> #include <memory>
> 
> #include "sdk/android/generated_metrics_jni/jni/Metrics_jni.h"
>-#include "sdk/android/src/jni/classreferenceholder.h"
>+#include "sdk/android/native_api/jni/java_types.h"
> #include "sdk/android/src/jni/jni_helpers.h"
> #include "system_wrappers/include/metrics.h"
> #include "system_wrappers/include/metrics_default.h"
>@@ -21,32 +21,29 @@
> namespace webrtc {
> namespace jni {
> 
>-JNI_FUNCTION_DECLARATION(void, Metrics_enableNative, JNIEnv* jni, jclass) {
>+static void JNI_Metrics_Enable(JNIEnv* jni, const JavaParamRef<jclass>&) {
>   metrics::Enable();
> }
> 
> // Gets and clears native histograms.
>-JNI_FUNCTION_DECLARATION(jobject,
>-                         Metrics_getAndResetNative,
>-                         JNIEnv* jni,
>-                         jclass) {
>-  jobject j_metrics = Java_Metrics_Constructor(jni);
>+static ScopedJavaLocalRef<jobject> JNI_Metrics_GetAndReset(
>+    JNIEnv* jni,
>+    const JavaParamRef<jclass>& jcaller) {
>+  ScopedJavaLocalRef<jobject> j_metrics = Java_Metrics_Constructor(jni);
> 
>   std::map<std::string, std::unique_ptr<metrics::SampleInfo>> histograms;
>   metrics::GetAndReset(&histograms);
>   for (const auto& kv : histograms) {
>     // Create and add samples to |HistogramInfo|.
>-    jobject j_info = Java_HistogramInfo_Constructor(
>+    ScopedJavaLocalRef<jobject> j_info = Java_HistogramInfo_Constructor(
>         jni, kv.second->min, kv.second->max,
>         static_cast<int>(kv.second->bucket_count));
>     for (const auto& sample : kv.second->samples) {
>       Java_HistogramInfo_addSample(jni, j_info, sample.first, sample.second);
>     }
>     // Add |HistogramInfo| to |Metrics|.
>-    jstring j_name = jni->NewStringUTF(kv.first.c_str());
>+    ScopedJavaLocalRef<jstring> j_name = NativeToJavaString(jni, kv.first);
>     Java_Metrics_add(jni, j_metrics, j_name, j_info);
>-    jni->DeleteLocalRef(j_name);
>-    jni->DeleteLocalRef(j_info);
>   }
>   CHECK_EXCEPTION(jni);
>   return j_metrics;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/androidnetworkmonitor_jni.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidnetworkmonitor.cc
>similarity index 74%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/androidnetworkmonitor_jni.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidnetworkmonitor.cc
>index 5cac1073988..cfd51e851f8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/androidnetworkmonitor_jni.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidnetworkmonitor.cc
>@@ -8,17 +8,20 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "sdk/android/src/jni/pc/androidnetworkmonitor_jni.h"
>+#include "sdk/android/src/jni/androidnetworkmonitor.h"
> 
> #include <dlfcn.h>
>+#ifndef RTLD_NOLOAD
> // This was added in Lollipop to dlfcn.h
> #define RTLD_NOLOAD 4
>+#endif
> 
> #include "rtc_base/bind.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/ipaddress.h"
>-#include "sdk/android/generated_peerconnection_jni/jni/NetworkMonitorAutoDetect_jni.h"
>-#include "sdk/android/generated_peerconnection_jni/jni/NetworkMonitor_jni.h"
>+#include "sdk/android/generated_base_jni/jni/NetworkMonitorAutoDetect_jni.h"
>+#include "sdk/android/generated_base_jni/jni/NetworkMonitor_jni.h"
>+#include "sdk/android/native_api/jni/java_types.h"
> #include "sdk/android/src/jni/jni_helpers.h"
> 
> namespace webrtc {
>@@ -29,7 +32,9 @@ enum AndroidSdkVersion {
>   SDK_VERSION_MARSHMALLOW = 23
> };
> 
>-static NetworkType GetNetworkTypeFromJava(JNIEnv* jni, jobject j_network_type) {
>+static NetworkType GetNetworkTypeFromJava(
>+    JNIEnv* jni,
>+    const JavaRef<jobject>& j_network_type) {
>   std::string enum_name = GetJavaEnumName(jni, j_network_type);
>   if (enum_name == "CONNECTION_UNKNOWN") {
>     return NetworkType::NETWORK_UNKNOWN;
>@@ -55,6 +60,9 @@ static NetworkType GetNetworkTypeFromJava(JNIEnv* jni, jobject j_network_type) {
>   if (enum_name == "CONNECTION_BLUETOOTH") {
>     return NetworkType::NETWORK_BLUETOOTH;
>   }
>+  if (enum_name == "CONNECTION_VPN") {
>+    return NetworkType::NETWORK_VPN;
>+  }
>   if (enum_name == "CONNECTION_NONE") {
>     return NetworkType::NETWORK_NONE;
>   }
>@@ -75,6 +83,7 @@ static rtc::AdapterType AdapterTypeFromNetworkType(NetworkType network_type) {
>     case NETWORK_2G:
>     case NETWORK_UNKNOWN_CELLULAR:
>       return rtc::ADAPTER_TYPE_CELLULAR;
>+    case NETWORK_VPN:
>     case NETWORK_BLUETOOTH:
>       // There is no corresponding mapping for bluetooth networks.
>       // Map it to VPN for now.
>@@ -85,43 +94,32 @@ static rtc::AdapterType AdapterTypeFromNetworkType(NetworkType network_type) {
>   }
> }
> 
>-static rtc::IPAddress GetIPAddressFromJava(JNIEnv* jni, jobject j_ip_address) {
>-  jbyteArray j_addresses = Java_IPAddress_getAddress(jni, j_ip_address);
>-  size_t address_length = jni->GetArrayLength(j_addresses);
>-  jbyte* addr_array = jni->GetByteArrayElements(j_addresses, nullptr);
>-  CHECK_EXCEPTION(jni) << "Error during GetIPAddressFromJava";
>+static rtc::IPAddress JavaToNativeIpAddress(
>+    JNIEnv* jni,
>+    const JavaRef<jobject>& j_ip_address) {
>+  ScopedJavaLocalRef<jbyteArray> j_addresses =
>+      Java_IPAddress_getAddress(jni, j_ip_address);
>+  size_t address_length = jni->GetArrayLength(j_addresses.obj());
>+  jbyte* addr_array = jni->GetByteArrayElements(j_addresses.obj(), nullptr);
>+  CHECK_EXCEPTION(jni) << "Error during JavaToNativeIpAddress";
>   if (address_length == 4) {
>     // IP4
>     struct in_addr ip4_addr;
>     memcpy(&ip4_addr.s_addr, addr_array, 4);
>-    jni->ReleaseByteArrayElements(j_addresses, addr_array, JNI_ABORT);
>+    jni->ReleaseByteArrayElements(j_addresses.obj(), addr_array, JNI_ABORT);
>     return rtc::IPAddress(ip4_addr);
>   }
>   // IP6
>   RTC_CHECK(address_length == 16);
>   struct in6_addr ip6_addr;
>   memcpy(ip6_addr.s6_addr, addr_array, address_length);
>-  jni->ReleaseByteArrayElements(j_addresses, addr_array, JNI_ABORT);
>+  jni->ReleaseByteArrayElements(j_addresses.obj(), addr_array, JNI_ABORT);
>   return rtc::IPAddress(ip6_addr);
> }
> 
>-static void GetIPAddressesFromJava(JNIEnv* jni,
>-                                   jobjectArray j_ip_addresses,
>-                                   std::vector<rtc::IPAddress>* ip_addresses) {
>-  ip_addresses->clear();
>-  size_t num_addresses = jni->GetArrayLength(j_ip_addresses);
>-  CHECK_EXCEPTION(jni) << "Error during GetArrayLength";
>-  for (size_t i = 0; i < num_addresses; ++i) {
>-    jobject j_ip_address = jni->GetObjectArrayElement(j_ip_addresses, i);
>-    CHECK_EXCEPTION(jni) << "Error during GetObjectArrayElement";
>-    rtc::IPAddress ip = GetIPAddressFromJava(jni, j_ip_address);
>-    ip_addresses->push_back(ip);
>-  }
>-}
>-
> static NetworkInformation GetNetworkInformationFromJava(
>     JNIEnv* jni,
>-    jobject j_network_info) {
>+    const JavaRef<jobject>& j_network_info) {
>   NetworkInformation network_info;
>   network_info.interface_name = JavaToStdString(
>       jni, Java_NetworkInformation_getName(jni, j_network_info));
>@@ -129,16 +127,38 @@ static NetworkInformation GetNetworkInformationFromJava(
>       Java_NetworkInformation_getHandle(jni, j_network_info));
>   network_info.type = GetNetworkTypeFromJava(
>       jni, Java_NetworkInformation_getConnectionType(jni, j_network_info));
>-  jobjectArray j_ip_addresses =
>+  network_info.underlying_type_for_vpn = GetNetworkTypeFromJava(
>+      jni, Java_NetworkInformation_getUnderlyingConnectionTypeForVpn(
>+               jni, j_network_info));
>+  ScopedJavaLocalRef<jobjectArray> j_ip_addresses =
>       Java_NetworkInformation_getIpAddresses(jni, j_network_info);
>-  GetIPAddressesFromJava(jni, j_ip_addresses, &network_info.ip_addresses);
>+  network_info.ip_addresses = JavaToNativeVector<rtc::IPAddress>(
>+      jni, j_ip_addresses, &JavaToNativeIpAddress);
>   return network_info;
> }
> 
>+NetworkInformation::NetworkInformation() = default;
>+
>+NetworkInformation::NetworkInformation(const NetworkInformation&) = default;
>+
>+NetworkInformation::NetworkInformation(NetworkInformation&&) = default;
>+
>+NetworkInformation::~NetworkInformation() = default;
>+
>+NetworkInformation& NetworkInformation::operator=(const NetworkInformation&) =
>+    default;
>+
>+NetworkInformation& NetworkInformation::operator=(NetworkInformation&&) =
>+    default;
>+
> std::string NetworkInformation::ToString() const {
>   std::stringstream ss;
>   ss << "NetInfo[name " << interface_name << "; handle " << handle << "; type "
>-     << type << "; address";
>+     << type;
>+  if (type == NETWORK_VPN) {
>+    ss << "; underlying_type_for_vpn " << underlying_type_for_vpn;
>+  }
>+  ss << "; address";
>   for (const rtc::IPAddress address : ip_addresses) {
>     ss << " " << address.ToString();
>   }
>@@ -146,10 +166,15 @@ std::string NetworkInformation::ToString() const {
>   return ss.str();
> }
> 
>-AndroidNetworkMonitor::AndroidNetworkMonitor(JNIEnv* env)
>+AndroidNetworkMonitor::AndroidNetworkMonitor(
>+    JNIEnv* env,
>+    const JavaRef<jobject>& j_application_context)
>     : android_sdk_int_(Java_NetworkMonitor_androidSdkInt(env)),
>+      j_application_context_(env, j_application_context),
>       j_network_monitor_(env, Java_NetworkMonitor_getInstance(env)) {}
> 
>+AndroidNetworkMonitor::~AndroidNetworkMonitor() = default;
>+
> void AndroidNetworkMonitor::Start() {
>   RTC_CHECK(thread_checker_.CalledOnValidThread());
>   if (started_) {
>@@ -163,8 +188,8 @@ void AndroidNetworkMonitor::Start() {
>   worker_thread()->socketserver()->set_network_binder(this);
> 
>   JNIEnv* env = AttachCurrentThreadIfNeeded();
>-  Java_NetworkMonitor_startMonitoring(env, *j_network_monitor_,
>-                                      jlongFromPointer(this));
>+  Java_NetworkMonitor_startMonitoring(
>+      env, j_network_monitor_, j_application_context_, jlongFromPointer(this));
> }
> 
> void AndroidNetworkMonitor::Stop() {
>@@ -181,7 +206,7 @@ void AndroidNetworkMonitor::Stop() {
>   }
> 
>   JNIEnv* env = AttachCurrentThreadIfNeeded();
>-  Java_NetworkMonitor_stopMonitoring(env, *j_network_monitor_,
>+  Java_NetworkMonitor_stopMonitoring(env, j_network_monitor_,
>                                      jlongFromPointer(this));
> 
>   network_handle_by_address_.clear();
>@@ -199,7 +224,7 @@ rtc::NetworkBindingResult AndroidNetworkMonitor::BindSocketToNetwork(
>   // networks. This may also occur if there is no connectivity manager service.
>   JNIEnv* env = AttachCurrentThreadIfNeeded();
>   const bool network_binding_supported =
>-      Java_NetworkMonitor_networkBindingSupported(env, *j_network_monitor_);
>+      Java_NetworkMonitor_networkBindingSupported(env, j_network_monitor_);
>   if (!network_binding_supported) {
>     RTC_LOG(LS_WARNING)
>         << "BindSocketToNetwork is not supported on this platform "
>@@ -302,6 +327,10 @@ void AndroidNetworkMonitor::OnNetworkConnected_w(
>   RTC_LOG(LS_INFO) << "Network connected: " << network_info.ToString();
>   adapter_type_by_name_[network_info.interface_name] =
>       AdapterTypeFromNetworkType(network_info.type);
>+  if (network_info.type == NETWORK_VPN) {
>+    vpn_underlying_adapter_type_by_name_[network_info.interface_name] =
>+        AdapterTypeFromNetworkType(network_info.underlying_type_for_vpn);
>+  }
>   network_info_by_handle_[network_info.handle] = network_info;
>   for (const rtc::IPAddress& address : network_info.ip_addresses) {
>     network_handle_by_address_[address] = network_info.handle;
>@@ -349,41 +378,60 @@ rtc::AdapterType AndroidNetworkMonitor::GetAdapterType(
>   return type;
> }
> 
>+rtc::AdapterType AndroidNetworkMonitor::GetVpnUnderlyingAdapterType(
>+    const std::string& if_name) {
>+  auto iter = vpn_underlying_adapter_type_by_name_.find(if_name);
>+  rtc::AdapterType type = (iter == vpn_underlying_adapter_type_by_name_.end())
>+                              ? rtc::ADAPTER_TYPE_UNKNOWN
>+                              : iter->second;
>+  return type;
>+}
>+
>+AndroidNetworkMonitorFactory::AndroidNetworkMonitorFactory()
>+    : j_application_context_(nullptr) {}
>+
>+AndroidNetworkMonitorFactory::AndroidNetworkMonitorFactory(
>+    JNIEnv* env,
>+    const JavaRef<jobject>& j_application_context)
>+    : j_application_context_(env, j_application_context) {}
>+
>+AndroidNetworkMonitorFactory::~AndroidNetworkMonitorFactory() = default;
>+
> rtc::NetworkMonitorInterface*
> AndroidNetworkMonitorFactory::CreateNetworkMonitor() {
>-  return new AndroidNetworkMonitor(AttachCurrentThreadIfNeeded());
>+  return new AndroidNetworkMonitor(AttachCurrentThreadIfNeeded(),
>+                                   j_application_context_);
> }
> 
>-void AndroidNetworkMonitor::NotifyConnectionTypeChanged(JNIEnv* env,
>-                                                        jobject j_caller) {
>+void AndroidNetworkMonitor::NotifyConnectionTypeChanged(
>+    JNIEnv* env,
>+    const JavaRef<jobject>& j_caller) {
>   OnNetworksChanged();
> }
> 
> void AndroidNetworkMonitor::NotifyOfActiveNetworkList(
>     JNIEnv* env,
>-    jobject j_caller,
>-    jobjectArray j_network_infos) {
>-  std::vector<NetworkInformation> network_infos;
>-  size_t num_networks = env->GetArrayLength(j_network_infos);
>-  for (size_t i = 0; i < num_networks; ++i) {
>-    jobject j_network_info = env->GetObjectArrayElement(j_network_infos, i);
>-    CHECK_EXCEPTION(env) << "Error during GetObjectArrayElement";
>-    network_infos.push_back(GetNetworkInformationFromJava(env, j_network_info));
>-  }
>+    const JavaRef<jobject>& j_caller,
>+    const JavaRef<jobjectArray>& j_network_infos) {
>+  std::vector<NetworkInformation> network_infos =
>+      JavaToNativeVector<NetworkInformation>(env, j_network_infos,
>+                                             &GetNetworkInformationFromJava);
>   SetNetworkInfos(network_infos);
> }
> 
>-void AndroidNetworkMonitor::NotifyOfNetworkConnect(JNIEnv* env,
>-                                                   jobject j_caller,
>-                                                   jobject j_network_info) {
>+void AndroidNetworkMonitor::NotifyOfNetworkConnect(
>+    JNIEnv* env,
>+    const JavaRef<jobject>& j_caller,
>+    const JavaRef<jobject>& j_network_info) {
>   NetworkInformation network_info =
>       GetNetworkInformationFromJava(env, j_network_info);
>   OnNetworkConnected(network_info);
> }
> 
>-void AndroidNetworkMonitor::NotifyOfNetworkDisconnect(JNIEnv* env,
>-                                                      jobject j_caller,
>-                                                      jlong network_handle) {
>+void AndroidNetworkMonitor::NotifyOfNetworkDisconnect(
>+    JNIEnv* env,
>+    const JavaRef<jobject>& j_caller,
>+    jlong network_handle) {
>   OnNetworkDisconnected(static_cast<NetworkHandle>(network_handle));
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/androidnetworkmonitor_jni.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidnetworkmonitor.h
>similarity index 64%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/androidnetworkmonitor_jni.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidnetworkmonitor.h
>index cf8d50fe713..563d5bd5e4f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/androidnetworkmonitor_jni.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidnetworkmonitor.h
>@@ -8,8 +8,8 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#ifndef SDK_ANDROID_SRC_JNI_PC_ANDROIDNETWORKMONITOR_JNI_H_
>-#define SDK_ANDROID_SRC_JNI_PC_ANDROIDNETWORKMONITOR_JNI_H_
>+#ifndef SDK_ANDROID_SRC_JNI_ANDROIDNETWORKMONITOR_H_
>+#define SDK_ANDROID_SRC_JNI_ANDROIDNETWORKMONITOR_H_
> 
> #include <stdint.h>
> #include <map>
>@@ -35,6 +35,7 @@ enum NetworkType {
>   NETWORK_2G,
>   NETWORK_UNKNOWN_CELLULAR,
>   NETWORK_BLUETOOTH,
>+  NETWORK_VPN,
>   NETWORK_NONE
> };
> 
>@@ -44,15 +45,25 @@ struct NetworkInformation {
>   std::string interface_name;
>   NetworkHandle handle;
>   NetworkType type;
>+  NetworkType underlying_type_for_vpn;
>   std::vector<rtc::IPAddress> ip_addresses;
> 
>+  NetworkInformation();
>+  NetworkInformation(const NetworkInformation&);
>+  NetworkInformation(NetworkInformation&&);
>+  ~NetworkInformation();
>+  NetworkInformation& operator=(const NetworkInformation&);
>+  NetworkInformation& operator=(NetworkInformation&&);
>+
>   std::string ToString() const;
> };
> 
> class AndroidNetworkMonitor : public rtc::NetworkMonitorBase,
>                               public rtc::NetworkBinderInterface {
>  public:
>-  explicit AndroidNetworkMonitor(JNIEnv* env);
>+  explicit AndroidNetworkMonitor(JNIEnv* env,
>+                                 const JavaRef<jobject>& j_application_context);
>+  ~AndroidNetworkMonitor() override;
> 
>   // TODO(sakal): Remove once down stream dependencies have been updated.
>   static void SetAndroidContext(JNIEnv* jni, jobject context) {}
>@@ -64,40 +75,54 @@ class AndroidNetworkMonitor : public rtc::NetworkMonitorBase,
>       int socket_fd,
>       const rtc::IPAddress& address) override;
>   rtc::AdapterType GetAdapterType(const std::string& if_name) override;
>+  rtc::AdapterType GetVpnUnderlyingAdapterType(
>+      const std::string& if_name) override;
>   void OnNetworkConnected(const NetworkInformation& network_info);
>   void OnNetworkDisconnected(NetworkHandle network_handle);
>   // Always expected to be called on the network thread.
>   void SetNetworkInfos(const std::vector<NetworkInformation>& network_infos);
> 
>-  void NotifyConnectionTypeChanged(JNIEnv* env, jobject j_caller);
>+  void NotifyConnectionTypeChanged(JNIEnv* env,
>+                                   const JavaRef<jobject>& j_caller);
>   void NotifyOfNetworkConnect(JNIEnv* env,
>-                              jobject j_caller,
>-                              jobject j_network_info);
>+                              const JavaRef<jobject>& j_caller,
>+                              const JavaRef<jobject>& j_network_info);
>   void NotifyOfNetworkDisconnect(JNIEnv* env,
>-                                 jobject j_caller,
>+                                 const JavaRef<jobject>& j_caller,
>                                  jlong network_handle);
>   void NotifyOfActiveNetworkList(JNIEnv* env,
>-                                 jobject j_caller,
>-                                 jobjectArray j_network_infos);
>+                                 const JavaRef<jobject>& j_caller,
>+                                 const JavaRef<jobjectArray>& j_network_infos);
> 
>  private:
>   void OnNetworkConnected_w(const NetworkInformation& network_info);
>   void OnNetworkDisconnected_w(NetworkHandle network_handle);
> 
>   const int android_sdk_int_;
>-  ScopedGlobalRef<jobject> j_network_monitor_;
>+  ScopedJavaGlobalRef<jobject> j_application_context_;
>+  ScopedJavaGlobalRef<jobject> j_network_monitor_;
>   rtc::ThreadChecker thread_checker_;
>   bool started_ = false;
>   std::map<std::string, rtc::AdapterType> adapter_type_by_name_;
>+  std::map<std::string, rtc::AdapterType> vpn_underlying_adapter_type_by_name_;
>   std::map<rtc::IPAddress, NetworkHandle> network_handle_by_address_;
>   std::map<NetworkHandle, NetworkInformation> network_info_by_handle_;
> };
> 
> class AndroidNetworkMonitorFactory : public rtc::NetworkMonitorFactory {
>  public:
>-  AndroidNetworkMonitorFactory() {}
>+  // Deprecated. Pass in application context to this class.
>+  AndroidNetworkMonitorFactory();
>+
>+  AndroidNetworkMonitorFactory(JNIEnv* env,
>+                               const JavaRef<jobject>& j_application_context);
>+
>+  ~AndroidNetworkMonitorFactory() override;
> 
>   rtc::NetworkMonitorInterface* CreateNetworkMonitor() override;
>+
>+ private:
>+  ScopedJavaGlobalRef<jobject> j_application_context_;
> };
> 
> }  // namespace jni
>@@ -111,4 +136,4 @@ using webrtc::jni::AndroidNetworkMonitorFactory;
> 
> }  // namespace webrtc_jni
> 
>-#endif  // SDK_ANDROID_SRC_JNI_PC_ANDROIDNETWORKMONITOR_JNI_H_
>+#endif  // SDK_ANDROID_SRC_JNI_ANDROIDNETWORKMONITOR_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidnetworkmonitor_jni.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidnetworkmonitor_jni.h
>index d8a11aafb14..407e6f68e15 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidnetworkmonitor_jni.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidnetworkmonitor_jni.h
>@@ -11,4 +11,4 @@
> // TODO(deadbeef): Remove this file when clients are updated to new include
> // path.
> 
>-#include "sdk/android/src/jni/pc/androidnetworkmonitor_jni.h"
>+#include "sdk/android/src/jni/pc/androidnetworkmonitor.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidvideotracksource.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidvideotracksource.cc
>index 42430c2804a..41d4278ac9c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidvideotracksource.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidvideotracksource.cc
>@@ -12,31 +12,35 @@
> 
> #include <utility>
> 
>+#include "api/videosourceproxy.h"
> #include "rtc_base/logging.h"
>-#include "sdk/android/src/jni/classreferenceholder.h"
>+
>+namespace webrtc {
>+namespace jni {
> 
> namespace {
> // MediaCodec wants resolution to be divisible by 2.
> const int kRequiredResolutionAlignment = 2;
>-}
>+}  // namespace
> 
>-namespace webrtc {
>-namespace jni {
>-
>-AndroidVideoTrackSource::AndroidVideoTrackSource(
>-    rtc::Thread* signaling_thread,
>-    JNIEnv* jni,
>-    jobject j_surface_texture_helper,
>-    bool is_screencast)
>+AndroidVideoTrackSource::AndroidVideoTrackSource(rtc::Thread* signaling_thread,
>+                                                 JNIEnv* jni,
>+                                                 bool is_screencast)
>     : AdaptedVideoTrackSource(kRequiredResolutionAlignment),
>       signaling_thread_(signaling_thread),
>-      surface_texture_helper_(new rtc::RefCountedObject<SurfaceTextureHelper>(
>-          jni,
>-          j_surface_texture_helper)),
>       is_screencast_(is_screencast) {
>   RTC_LOG(LS_INFO) << "AndroidVideoTrackSource ctor";
>   camera_thread_checker_.DetachFromThread();
> }
>+AndroidVideoTrackSource::~AndroidVideoTrackSource() = default;
>+
>+bool AndroidVideoTrackSource::is_screencast() const {
>+  return is_screencast_;
>+}
>+
>+absl::optional<bool> AndroidVideoTrackSource::needs_denoising() const {
>+  return false;
>+}
> 
> void AndroidVideoTrackSource::SetState(SourceState state) {
>   if (rtc::Thread::Current() != signaling_thread_) {
>@@ -52,112 +56,21 @@ void AndroidVideoTrackSource::SetState(SourceState state) {
>   }
> }
> 
>-void AndroidVideoTrackSource::OnByteBufferFrameCaptured(const void* frame_data,
>-                                                        int length,
>-                                                        int width,
>-                                                        int height,
>-                                                        VideoRotation rotation,
>-                                                        int64_t timestamp_ns) {
>-  RTC_DCHECK(camera_thread_checker_.CalledOnValidThread());
>-
>-  int64_t camera_time_us = timestamp_ns / rtc::kNumNanosecsPerMicrosec;
>-  int64_t translated_camera_time_us =
>-      timestamp_aligner_.TranslateTimestamp(camera_time_us, rtc::TimeMicros());
>-
>-  int adapted_width;
>-  int adapted_height;
>-  int crop_width;
>-  int crop_height;
>-  int crop_x;
>-  int crop_y;
>-
>-  if (!AdaptFrame(width, height, camera_time_us, &adapted_width,
>-                  &adapted_height, &crop_width, &crop_height, &crop_x,
>-                  &crop_y)) {
>-    return;
>-  }
>-
>-  const uint8_t* y_plane = static_cast<const uint8_t*>(frame_data);
>-  const uint8_t* uv_plane = y_plane + width * height;
>-  const int uv_width = (width + 1) / 2;
>-
>-  RTC_CHECK_GE(length, width * height + 2 * uv_width * ((height + 1) / 2));
>-
>-  // Can only crop at even pixels.
>-  crop_x &= ~1;
>-  crop_y &= ~1;
>-  // Crop just by modifying pointers.
>-  y_plane += width * crop_y + crop_x;
>-  uv_plane += uv_width * crop_y + crop_x;
>-
>-  rtc::scoped_refptr<I420Buffer> buffer =
>-      buffer_pool_.CreateBuffer(adapted_width, adapted_height);
>-
>-  nv12toi420_scaler_.NV12ToI420Scale(
>-      y_plane, width, uv_plane, uv_width * 2, crop_width, crop_height,
>-      buffer->MutableDataY(), buffer->StrideY(),
>-      // Swap U and V, since we have NV21, not NV12.
>-      buffer->MutableDataV(), buffer->StrideV(), buffer->MutableDataU(),
>-      buffer->StrideU(), buffer->width(), buffer->height());
>+AndroidVideoTrackSource::SourceState AndroidVideoTrackSource::state() const {
>+  return state_;
>+}
> 
>-  OnFrame(VideoFrame(buffer, rotation, translated_camera_time_us));
>+bool AndroidVideoTrackSource::remote() const {
>+  return false;
> }
> 
>-void AndroidVideoTrackSource::OnTextureFrameCaptured(
>+void AndroidVideoTrackSource::OnFrameCaptured(
>+    JNIEnv* jni,
>     int width,
>     int height,
>-    VideoRotation rotation,
>     int64_t timestamp_ns,
>-    const NativeHandleImpl& handle) {
>-  RTC_DCHECK(camera_thread_checker_.CalledOnValidThread());
>-
>-  int64_t camera_time_us = timestamp_ns / rtc::kNumNanosecsPerMicrosec;
>-  int64_t translated_camera_time_us =
>-      timestamp_aligner_.TranslateTimestamp(camera_time_us, rtc::TimeMicros());
>-
>-  int adapted_width;
>-  int adapted_height;
>-  int crop_width;
>-  int crop_height;
>-  int crop_x;
>-  int crop_y;
>-
>-  if (!AdaptFrame(width, height, camera_time_us, &adapted_width,
>-                  &adapted_height, &crop_width, &crop_height, &crop_x,
>-                  &crop_y)) {
>-    surface_texture_helper_->ReturnTextureFrame();
>-    return;
>-  }
>-
>-  Matrix matrix = handle.sampling_matrix;
>-
>-  matrix.Crop(crop_width / static_cast<float>(width),
>-              crop_height / static_cast<float>(height),
>-              crop_x / static_cast<float>(width),
>-              crop_y / static_cast<float>(height));
>-
>-  // Note that apply_rotation() may change under our feet, so we should only
>-  // check once.
>-  if (apply_rotation()) {
>-    if (rotation == kVideoRotation_90 || rotation == kVideoRotation_270) {
>-      std::swap(adapted_width, adapted_height);
>-    }
>-    matrix.Rotate(rotation);
>-    rotation = kVideoRotation_0;
>-  }
>-
>-  OnFrame(VideoFrame(surface_texture_helper_->CreateTextureFrame(
>-                         adapted_width, adapted_height,
>-                         NativeHandleImpl(handle.oes_texture_id, matrix)),
>-                     rotation, translated_camera_time_us));
>-}
>-
>-void AndroidVideoTrackSource::OnFrameCaptured(JNIEnv* jni,
>-                                              int width,
>-                                              int height,
>-                                              int64_t timestamp_ns,
>-                                              VideoRotation rotation,
>-                                              jobject j_video_frame_buffer) {
>+    VideoRotation rotation,
>+    const JavaRef<jobject>& j_video_frame_buffer) {
>   RTC_DCHECK(camera_thread_checker_.CalledOnValidThread());
> 
>   int64_t camera_time_us = timestamp_ns / rtc::kNumNanosecsPerMicrosec;
>@@ -198,5 +111,5 @@ void AndroidVideoTrackSource::OnOutputFormatRequest(int width,
>   video_adapter()->OnOutputFormatRequest(format);
> }
> 
>-}  // namespace webrtc
>+}  // namespace jni
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidvideotracksource.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidvideotracksource.h
>index 524c36d837c..8f092c1b041 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidvideotracksource.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidvideotracksource.h
>@@ -20,7 +20,6 @@
> #include "rtc_base/checks.h"
> #include "rtc_base/thread_checker.h"
> #include "rtc_base/timestampaligner.h"
>-#include "sdk/android/src/jni/surfacetexturehelper_jni.h"
> #include "sdk/android/src/jni/videoframe.h"
> 
> namespace webrtc {
>@@ -30,58 +29,38 @@ class AndroidVideoTrackSource : public rtc::AdaptedVideoTrackSource {
>  public:
>   AndroidVideoTrackSource(rtc::Thread* signaling_thread,
>                           JNIEnv* jni,
>-                          jobject j_surface_texture_helper,
>                           bool is_screencast = false);
>+  ~AndroidVideoTrackSource() override;
> 
>-  bool is_screencast() const override { return is_screencast_; }
>+  bool is_screencast() const override;
> 
>   // Indicates that the encoder should denoise video before encoding it.
>   // If it is not set, the default configuration is used which is different
>   // depending on video codec.
>-  rtc::Optional<bool> needs_denoising() const override { return false; }
>+  absl::optional<bool> needs_denoising() const override;
> 
>   // Called by the native capture observer
>   void SetState(SourceState state);
> 
>-  SourceState state() const override { return state_; }
>+  SourceState state() const override;
> 
>-  bool remote() const override { return false; }
>-
>-  void OnByteBufferFrameCaptured(const void* frame_data,
>-                                 int length,
>-                                 int width,
>-                                 int height,
>-                                 VideoRotation rotation,
>-                                 int64_t timestamp_ns);
>-
>-  void OnTextureFrameCaptured(int width,
>-                              int height,
>-                              VideoRotation rotation,
>-                              int64_t timestamp_ns,
>-                              const NativeHandleImpl& handle);
>+  bool remote() const override;
> 
>   void OnFrameCaptured(JNIEnv* jni,
>                        int width,
>                        int height,
>                        int64_t timestamp_ns,
>                        VideoRotation rotation,
>-                       jobject j_video_frame_buffer);
>+                       const JavaRef<jobject>& j_video_frame_buffer);
> 
>   void OnOutputFormatRequest(int width, int height, int fps);
> 
>-  rtc::scoped_refptr<SurfaceTextureHelper> surface_texture_helper() {
>-    return surface_texture_helper_;
>-  }
>-
>  private:
>   rtc::Thread* signaling_thread_;
>   rtc::AsyncInvoker invoker_;
>   rtc::ThreadChecker camera_thread_checker_;
>   SourceState state_;
>   rtc::TimestampAligner timestamp_aligner_;
>-  NV12ToI420Scaler nv12toi420_scaler_;
>-  I420BufferPool buffer_pool_;
>-  rtc::scoped_refptr<SurfaceTextureHelper> surface_texture_helper_;
>   const bool is_screencast_;
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidvideotracksource_jni.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidvideotracksource_jni.cc
>deleted file mode 100644
>index 178b099f025..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/androidvideotracksource_jni.cc
>+++ /dev/null
>@@ -1,132 +0,0 @@
>-/*
>- *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "api/video/video_rotation.h"
>-#include "api/videosourceproxy.h"
>-#include "rtc_base/logging.h"
>-#include "sdk/android/src/jni/androidvideotracksource.h"
>-#include "sdk/android/src/jni/classreferenceholder.h"
>-
>-namespace webrtc {
>-
>-namespace {
>-
>-static VideoRotation jintToVideoRotation(jint rotation) {
>-  RTC_DCHECK(rotation == 0 || rotation == 90 || rotation == 180 ||
>-             rotation == 270);
>-  return static_cast<VideoRotation>(rotation);
>-}
>-
>-}  // namespace
>-
>-namespace jni {
>-
>-static AndroidVideoTrackSource* AndroidVideoTrackSourceFromJavaProxy(
>-    jlong j_proxy) {
>-  auto proxy_source = reinterpret_cast<VideoTrackSourceProxy*>(j_proxy);
>-  return reinterpret_cast<AndroidVideoTrackSource*>(proxy_source->internal());
>-}
>-
>-JNI_FUNCTION_DECLARATION(
>-    void,
>-    AndroidVideoTrackSourceObserver_nativeOnByteBufferFrameCaptured,
>-    JNIEnv* jni,
>-    jclass,
>-    jlong j_source,
>-    jbyteArray j_frame,
>-    jint length,
>-    jint width,
>-    jint height,
>-    jint rotation,
>-    jlong timestamp) {
>-  AndroidVideoTrackSource* source =
>-      AndroidVideoTrackSourceFromJavaProxy(j_source);
>-  jbyte* bytes = jni->GetByteArrayElements(j_frame, nullptr);
>-  source->OnByteBufferFrameCaptured(bytes, length, width, height,
>-                                    jintToVideoRotation(rotation), timestamp);
>-  jni->ReleaseByteArrayElements(j_frame, bytes, JNI_ABORT);
>-}
>-
>-JNI_FUNCTION_DECLARATION(
>-    void,
>-    AndroidVideoTrackSourceObserver_nativeOnTextureFrameCaptured,
>-    JNIEnv* jni,
>-    jclass,
>-    jlong j_source,
>-    jint j_width,
>-    jint j_height,
>-    jint j_oes_texture_id,
>-    jfloatArray j_transform_matrix,
>-    jint j_rotation,
>-    jlong j_timestamp) {
>-  AndroidVideoTrackSource* source =
>-      AndroidVideoTrackSourceFromJavaProxy(j_source);
>-  source->OnTextureFrameCaptured(
>-      j_width, j_height, jintToVideoRotation(j_rotation), j_timestamp,
>-      NativeHandleImpl(jni, j_oes_texture_id, j_transform_matrix));
>-}
>-
>-JNI_FUNCTION_DECLARATION(void,
>-                         AndroidVideoTrackSourceObserver_nativeOnFrameCaptured,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jlong j_source,
>-                         jint j_width,
>-                         jint j_height,
>-                         jint j_rotation,
>-                         jlong j_timestamp_ns,
>-                         jobject j_video_frame_buffer) {
>-  AndroidVideoTrackSource* source =
>-      AndroidVideoTrackSourceFromJavaProxy(j_source);
>-  source->OnFrameCaptured(jni, j_width, j_height, j_timestamp_ns,
>-                          jintToVideoRotation(j_rotation),
>-                          j_video_frame_buffer);
>-}
>-
>-JNI_FUNCTION_DECLARATION(void,
>-                         AndroidVideoTrackSourceObserver_nativeCapturerStarted,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jlong j_source,
>-                         jboolean j_success) {
>-  RTC_LOG(LS_INFO) << "AndroidVideoTrackSourceObserve_nativeCapturerStarted";
>-  AndroidVideoTrackSource* source =
>-      AndroidVideoTrackSourceFromJavaProxy(j_source);
>-  source->SetState(j_success ? AndroidVideoTrackSource::SourceState::kLive
>-                             : AndroidVideoTrackSource::SourceState::kEnded);
>-}
>-
>-JNI_FUNCTION_DECLARATION(void,
>-                         AndroidVideoTrackSourceObserver_nativeCapturerStopped,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jlong j_source) {
>-  RTC_LOG(LS_INFO) << "AndroidVideoTrackSourceObserve_nativeCapturerStopped";
>-  AndroidVideoTrackSource* source =
>-      AndroidVideoTrackSourceFromJavaProxy(j_source);
>-  source->SetState(AndroidVideoTrackSource::SourceState::kEnded);
>-}
>-
>-JNI_FUNCTION_DECLARATION(void,
>-                         VideoSource_nativeAdaptOutputFormat,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jlong j_source,
>-                         jint j_width,
>-                         jint j_height,
>-                         jint j_fps) {
>-  RTC_LOG(LS_INFO) << "VideoSource_nativeAdaptOutputFormat";
>-  AndroidVideoTrackSource* source =
>-      AndroidVideoTrackSourceFromJavaProxy(j_source);
>-  source->OnOutputFormatRequest(j_width, j_height, j_fps);
>-}
>-
>-}  // namespace jni
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/DEPS b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/DEPS
>new file mode 100644
>index 00000000000..9a3adee6876
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/DEPS
>@@ -0,0 +1,4 @@
>+include_rules = [
>+  "+base/android/jni_android.h",
>+  "+modules/audio_device",
>+]
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/OWNERS
>new file mode 100644
>index 00000000000..95662c195c0
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/OWNERS
>@@ -0,0 +1 @@
>+henrika@webrtc.org
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/aaudio_player.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/aaudio_player.cc
>new file mode 100644
>index 00000000000..ac45e900d8d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/aaudio_player.cc
>@@ -0,0 +1,246 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "sdk/android/src/jni/audio_device/aaudio_player.h"
>+
>+#include "absl/memory/memory.h"
>+#include "api/array_view.h"
>+#include "modules/audio_device/fine_audio_buffer.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/logging.h"
>+
>+namespace webrtc {
>+
>+namespace jni {
>+
>+enum AudioDeviceMessageType : uint32_t {
>+  kMessageOutputStreamDisconnected,
>+};
>+
>+AAudioPlayer::AAudioPlayer(const AudioParameters& audio_parameters)
>+    : main_thread_(rtc::Thread::Current()),
>+      aaudio_(audio_parameters, AAUDIO_DIRECTION_OUTPUT, this) {
>+  RTC_LOG(INFO) << "ctor";
>+  thread_checker_aaudio_.DetachFromThread();
>+}
>+
>+AAudioPlayer::~AAudioPlayer() {
>+  RTC_LOG(INFO) << "dtor";
>+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
>+  Terminate();
>+  RTC_LOG(INFO) << "#detected underruns: " << underrun_count_;
>+}
>+
>+int AAudioPlayer::Init() {
>+  RTC_LOG(INFO) << "Init";
>+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
>+  if (aaudio_.audio_parameters().channels() == 2) {
>+    RTC_DLOG(LS_WARNING) << "Stereo mode is enabled";
>+  }
>+  return 0;
>+}
>+
>+int AAudioPlayer::Terminate() {
>+  RTC_LOG(INFO) << "Terminate";
>+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
>+  StopPlayout();
>+  return 0;
>+}
>+
>+int AAudioPlayer::InitPlayout() {
>+  RTC_LOG(INFO) << "InitPlayout";
>+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
>+  RTC_DCHECK(!initialized_);
>+  RTC_DCHECK(!playing_);
>+  if (!aaudio_.Init()) {
>+    return -1;
>+  }
>+  initialized_ = true;
>+  return 0;
>+}
>+
>+bool AAudioPlayer::PlayoutIsInitialized() const {
>+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
>+  return initialized_;
>+}
>+
>+int AAudioPlayer::StartPlayout() {
>+  RTC_LOG(INFO) << "StartPlayout";
>+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
>+  RTC_DCHECK(!playing_);
>+  if (!initialized_) {
>+    RTC_DLOG(LS_WARNING)
>+        << "Playout can not start since InitPlayout must succeed first";
>+    return 0;
>+  }
>+  if (fine_audio_buffer_) {
>+    fine_audio_buffer_->ResetPlayout();
>+  }
>+  if (!aaudio_.Start()) {
>+    return -1;
>+  }
>+  underrun_count_ = aaudio_.xrun_count();
>+  first_data_callback_ = true;
>+  playing_ = true;
>+  return 0;
>+}
>+
>+int AAudioPlayer::StopPlayout() {
>+  RTC_LOG(INFO) << "StopPlayout";
>+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
>+  if (!initialized_ || !playing_) {
>+    return 0;
>+  }
>+  if (!aaudio_.Stop()) {
>+    RTC_LOG(LS_ERROR) << "StopPlayout failed";
>+    return -1;
>+  }
>+  thread_checker_aaudio_.DetachFromThread();
>+  initialized_ = false;
>+  playing_ = false;
>+  return 0;
>+}
>+
>+bool AAudioPlayer::Playing() const {
>+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
>+  return playing_;
>+}
>+
>+void AAudioPlayer::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
>+  RTC_DLOG(INFO) << "AttachAudioBuffer";
>+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
>+  audio_device_buffer_ = audioBuffer;
>+  const AudioParameters audio_parameters = aaudio_.audio_parameters();
>+  audio_device_buffer_->SetPlayoutSampleRate(audio_parameters.sample_rate());
>+  audio_device_buffer_->SetPlayoutChannels(audio_parameters.channels());
>+  RTC_CHECK(audio_device_buffer_);
>+  // Create a modified audio buffer class which allows us to ask for any number
>+  // of samples (and not only multiple of 10ms) to match the optimal buffer
>+  // size per callback used by AAudio.
>+  fine_audio_buffer_ = absl::make_unique<FineAudioBuffer>(audio_device_buffer_);
>+}
>+
>+bool AAudioPlayer::SpeakerVolumeIsAvailable() {
>+  return false;
>+}
>+
>+int AAudioPlayer::SetSpeakerVolume(uint32_t volume) {
>+  return -1;
>+}
>+
>+absl::optional<uint32_t> AAudioPlayer::SpeakerVolume() const {
>+  return absl::nullopt;
>+}
>+
>+absl::optional<uint32_t> AAudioPlayer::MaxSpeakerVolume() const {
>+  return absl::nullopt;
>+}
>+
>+absl::optional<uint32_t> AAudioPlayer::MinSpeakerVolume() const {
>+  return absl::nullopt;
>+}
>+
>+void AAudioPlayer::OnErrorCallback(aaudio_result_t error) {
>+  RTC_LOG(LS_ERROR) << "OnErrorCallback: " << AAudio_convertResultToText(error);
>+  // TODO(henrika): investigate if we can use a thread checker here. Initial
>+  // tests shows that this callback can sometimes be called on a unique thread
>+  // but according to the documentation it should be on the same thread as the
>+  // data callback.
>+  // RTC_DCHECK_RUN_ON(&thread_checker_aaudio_);
>+  if (aaudio_.stream_state() == AAUDIO_STREAM_STATE_DISCONNECTED) {
>+    // The stream is disconnected and any attempt to use it will return
>+    // AAUDIO_ERROR_DISCONNECTED.
>+    RTC_LOG(WARNING) << "Output stream disconnected";
>+    // AAudio documentation states: "You should not close or reopen the stream
>+    // from the callback, use another thread instead". A message is therefore
>+    // sent to the main thread to do the restart operation.
>+    RTC_DCHECK(main_thread_);
>+    main_thread_->Post(RTC_FROM_HERE, this, kMessageOutputStreamDisconnected);
>+  }
>+}
>+
>+aaudio_data_callback_result_t AAudioPlayer::OnDataCallback(void* audio_data,
>+                                                           int32_t num_frames) {
>+  RTC_DCHECK_RUN_ON(&thread_checker_aaudio_);
>+  // Log device id in first data callback to ensure that a valid device is
>+  // utilized.
>+  if (first_data_callback_) {
>+    RTC_LOG(INFO) << "--- First output data callback: "
>+                  << "device id=" << aaudio_.device_id();
>+    first_data_callback_ = false;
>+  }
>+
>+  // Check if the underrun count has increased. If it has, increase the buffer
>+  // size by adding the size of a burst. It will reduce the risk of underruns
>+  // at the expense of an increased latency.
>+  // TODO(henrika): enable possibility to disable and/or tune the algorithm.
>+  const int32_t underrun_count = aaudio_.xrun_count();
>+  if (underrun_count > underrun_count_) {
>+    RTC_LOG(LS_ERROR) << "Underrun detected: " << underrun_count;
>+    underrun_count_ = underrun_count;
>+    aaudio_.IncreaseOutputBufferSize();
>+  }
>+
>+  // Estimate latency between writing an audio frame to the output stream and
>+  // the time that same frame is played out on the output audio device.
>+  latency_millis_ = aaudio_.EstimateLatencyMillis();
>+  // TODO(henrika): use for development only.
>+  if (aaudio_.frames_written() % (1000 * aaudio_.frames_per_burst()) == 0) {
>+    RTC_DLOG(INFO) << "output latency: " << latency_millis_
>+                   << ", num_frames: " << num_frames;
>+  }
>+
>+  // Read audio data from the WebRTC source using the FineAudioBuffer object
>+  // and write that data into |audio_data| to be played out by AAudio.
>+  // Prime output with zeros during a short initial phase to avoid distortion.
>+  // TODO(henrika): do more work to figure out of if the initial forced silence
>+  // period is really needed.
>+  if (aaudio_.frames_written() < 50 * aaudio_.frames_per_burst()) {
>+    const size_t num_bytes =
>+        sizeof(int16_t) * aaudio_.samples_per_frame() * num_frames;
>+    memset(audio_data, 0, num_bytes);
>+  } else {
>+    fine_audio_buffer_->GetPlayoutData(
>+        rtc::MakeArrayView(static_cast<int16_t*>(audio_data),
>+                           aaudio_.samples_per_frame() * num_frames),
>+        static_cast<int>(latency_millis_ + 0.5));
>+  }
>+
>+  // TODO(henrika): possibly add trace here to be included in systrace.
>+  // See https://developer.android.com/studio/profile/systrace-commandline.html.
>+  return AAUDIO_CALLBACK_RESULT_CONTINUE;
>+}
>+
>+void AAudioPlayer::OnMessage(rtc::Message* msg) {
>+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
>+  switch (msg->message_id) {
>+    case kMessageOutputStreamDisconnected:
>+      HandleStreamDisconnected();
>+      break;
>+  }
>+}
>+
>+void AAudioPlayer::HandleStreamDisconnected() {
>+  RTC_DCHECK_RUN_ON(&main_thread_checker_);
>+  RTC_DLOG(INFO) << "HandleStreamDisconnected";
>+  if (!initialized_ || !playing_) {
>+    return;
>+  }
>+  // Perform a restart by first closing the disconnected stream and then start
>+  // a new stream; this time using the new (preferred) audio output device.
>+  audio_device_buffer_->NativeAudioPlayoutInterrupted();
>+  StopPlayout();
>+  InitPlayout();
>+  StartPlayout();
>+}
>+
>+}  // namespace jni
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/aaudio_player.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/aaudio_player.h
>new file mode 100644
>index 00000000000..b43b5b3157c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/aaudio_player.h
>@@ -0,0 +1,153 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef SDK_ANDROID_SRC_JNI_AUDIO_DEVICE_AAUDIO_PLAYER_H_
>+#define SDK_ANDROID_SRC_JNI_AUDIO_DEVICE_AAUDIO_PLAYER_H_
>+
>+#include <aaudio/AAudio.h>
>+#include <memory>
>+
>+#include "absl/types/optional.h"
>+#include "modules/audio_device/audio_device_buffer.h"
>+#include "modules/audio_device/include/audio_device_defines.h"
>+#include "rtc_base/messagehandler.h"
>+#include "rtc_base/thread.h"
>+#include "rtc_base/thread_annotations.h"
>+#include "rtc_base/thread_checker.h"
>+#include "sdk/android/src/jni/audio_device/aaudio_wrapper.h"
>+#include "sdk/android/src/jni/audio_device/audio_device_module.h"
>+
>+namespace webrtc {
>+
>+class AudioDeviceBuffer;
>+class FineAudioBuffer;
>+
>+namespace jni {
>+
>+// Implements low-latency 16-bit mono PCM audio output support for Android
>+// using the C based AAudio API.
>+//
>+// An instance must be created and destroyed on one and the same thread.
>+// All public methods must also be called on the same thread. A thread checker
>+// will DCHECK if any method is called on an invalid thread. Audio buffers
>+// are requested on a dedicated high-priority thread owned by AAudio.
>+//
>+// The existing design forces the user to call InitPlayout() after StopPlayout()
>+// to be able to call StartPlayout() again. This is in line with how the Java-
>+// based implementation works.
>+//
>+// An audio stream can be disconnected, e.g. when an audio device is removed.
>+// This implementation will restart the audio stream using the new preferred
>+// device if such an event happens.
>+//
>+// Also supports automatic buffer-size adjustment based on underrun detections
>+// where the internal AAudio buffer can be increased when needed. It will
>+// reduce the risk of underruns (~glitches) at the expense of an increased
>+// latency.
>+class AAudioPlayer final : public AudioOutput,
>+                           public AAudioObserverInterface,
>+                           public rtc::MessageHandler {
>+ public:
>+  explicit AAudioPlayer(const AudioParameters& audio_parameters);
>+  ~AAudioPlayer() override;
>+
>+  int Init() override;
>+  int Terminate() override;
>+
>+  int InitPlayout() override;
>+  bool PlayoutIsInitialized() const override;
>+
>+  int StartPlayout() override;
>+  int StopPlayout() override;
>+  bool Playing() const override;
>+
>+  void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override;
>+
>+  // Not implemented in AAudio.
>+  bool SpeakerVolumeIsAvailable() override;
>+  int SetSpeakerVolume(uint32_t volume) override;
>+  absl::optional<uint32_t> SpeakerVolume() const override;
>+  absl::optional<uint32_t> MaxSpeakerVolume() const override;
>+  absl::optional<uint32_t> MinSpeakerVolume() const override;
>+
>+ protected:
>+  // AAudioObserverInterface implementation.
>+
>+  // For an output stream, this function should render and write |num_frames|
>+  // of data in the streams current data format to the |audio_data| buffer.
>+  // Called on a real-time thread owned by AAudio.
>+  aaudio_data_callback_result_t OnDataCallback(void* audio_data,
>+                                               int32_t num_frames) override;
>+  // AAudio calls this functions if any error occurs on a callback thread.
>+  // Called on a real-time thread owned by AAudio.
>+  void OnErrorCallback(aaudio_result_t error) override;
>+
>+  // rtc::MessageHandler used for restart messages from the error-callback
>+  // thread to the main (creating) thread.
>+  void OnMessage(rtc::Message* msg) override;
>+
>+ private:
>+  // Closes the existing stream and starts a new stream.
>+  void HandleStreamDisconnected();
>+
>+  // Ensures that methods are called from the same thread as this object is
>+  // created on.
>+  rtc::ThreadChecker main_thread_checker_;
>+
>+  // Stores thread ID in first call to AAudioPlayer::OnDataCallback from a
>+  // real-time thread owned by AAudio. Detached during construction of this
>+  // object.
>+  rtc::ThreadChecker thread_checker_aaudio_;
>+
>+  // The thread on which this object is created on.
>+  rtc::Thread* main_thread_;
>+
>+  // Wraps all AAudio resources. Contains an output stream using the default
>+  // output audio device. Can be accessed on both the main thread and the
>+  // real-time thread owned by AAudio. See separate AAudio documentation about
>+  // thread safety.
>+  AAudioWrapper aaudio_;
>+
>+  // FineAudioBuffer takes an AudioDeviceBuffer which delivers audio data
>+  // in chunks of 10ms. It then allows for this data to be pulled in
>+  // a finer or coarser granularity. I.e. interacting with this class instead
>+  // of directly with the AudioDeviceBuffer one can ask for any number of
>+  // audio data samples.
>+  // Example: native buffer size can be 192 audio frames at 48kHz sample rate.
>+  // WebRTC will provide 480 audio frames per 10ms but AAudio asks for 192
>+  // in each callback (once every 4th ms). This class can then ask for 192 and
>+  // the FineAudioBuffer will ask WebRTC for new data approximately only every
>+  // second callback and also cache non-utilized audio.
>+  std::unique_ptr<FineAudioBuffer> fine_audio_buffer_;
>+
>+  // Counts number of detected underrun events reported by AAudio.
>+  int32_t underrun_count_ = 0;
>+
>+  // True only for the first data callback in each audio session.
>+  bool first_data_callback_ = true;
>+
>+  // Raw pointer handle provided to us in AttachAudioBuffer(). Owned by the
>+  // AudioDeviceModuleImpl class and set by AudioDeviceModule::Create().
>+  AudioDeviceBuffer* audio_device_buffer_ RTC_GUARDED_BY(main_thread_checker_) =
>+      nullptr;
>+
>+  bool initialized_ RTC_GUARDED_BY(main_thread_checker_) = false;
>+  bool playing_ RTC_GUARDED_BY(main_thread_checker_) = false;
>+
>+  // Estimated latency between writing an audio frame to the output stream and
>+  // the time that same frame is played out on the output audio device.
>+  double latency_millis_ RTC_GUARDED_BY(thread_checker_aaudio_) = 0;
>+};
>+
>+}  // namespace jni
>+
>+}  // namespace webrtc
>+
>+#endif  // SDK_ANDROID_SRC_JNI_AUDIO_DEVICE_AAUDIO_PLAYER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/aaudio_recorder.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/aaudio_recorder.cc
>new file mode 100644
>index 00000000000..5ba9d3a1867
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/aaudio_recorder.cc
>@@ -0,0 +1,235 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "sdk/android/src/jni/audio_device/aaudio_recorder.h"
>+
>+#include "absl/memory/memory.h"
>+#include "api/array_view.h"
>+#include "modules/audio_device/fine_audio_buffer.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/logging.h"
>+#include "rtc_base/timeutils.h"
>+
>+#include "system_wrappers/include/sleep.h"
>+
>+namespace webrtc {
>+
>+namespace jni {
>+
>+enum AudioDeviceMessageType : uint32_t {
>+  kMessageInputStreamDisconnected,
>+};
>+
>+AAudioRecorder::AAudioRecorder(const AudioParameters& audio_parameters)
>+    : main_thread_(rtc::Thread::Current()),
>+      aaudio_(audio_parameters, AAUDIO_DIRECTION_INPUT, this) {
>+  RTC_LOG(INFO) << "ctor";
>+  thread_checker_aaudio_.DetachFromThread();
>+}
>+
>+AAudioRecorder::~AAudioRecorder() {
>+  RTC_LOG(INFO) << "dtor";
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  Terminate();
>+  RTC_LOG(INFO) << "detected owerflows: " << overflow_count_;
>+}
>+
>+int AAudioRecorder::Init() {
>+  RTC_LOG(INFO) << "Init";
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  if (aaudio_.audio_parameters().channels() == 2) {
>+    RTC_DLOG(LS_WARNING) << "Stereo mode is enabled";
>+  }
>+  return 0;
>+}
>+
>+int AAudioRecorder::Terminate() {
>+  RTC_LOG(INFO) << "Terminate";
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  StopRecording();
>+  return 0;
>+}
>+
>+int AAudioRecorder::InitRecording() {
>+  RTC_LOG(INFO) << "InitRecording";
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  RTC_DCHECK(!initialized_);
>+  RTC_DCHECK(!recording_);
>+  if (!aaudio_.Init()) {
>+    return -1;
>+  }
>+  initialized_ = true;
>+  return 0;
>+}
>+
>+bool AAudioRecorder::RecordingIsInitialized() const {
>+  return initialized_;
>+}
>+
>+int AAudioRecorder::StartRecording() {
>+  RTC_LOG(INFO) << "StartRecording";
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  RTC_DCHECK(initialized_);
>+  RTC_DCHECK(!recording_);
>+  if (fine_audio_buffer_) {
>+    fine_audio_buffer_->ResetPlayout();
>+  }
>+  if (!aaudio_.Start()) {
>+    return -1;
>+  }
>+  overflow_count_ = aaudio_.xrun_count();
>+  first_data_callback_ = true;
>+  recording_ = true;
>+  return 0;
>+}
>+
>+int AAudioRecorder::StopRecording() {
>+  RTC_LOG(INFO) << "StopRecording";
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  if (!initialized_ || !recording_) {
>+    return 0;
>+  }
>+  if (!aaudio_.Stop()) {
>+    return -1;
>+  }
>+  thread_checker_aaudio_.DetachFromThread();
>+  initialized_ = false;
>+  recording_ = false;
>+  return 0;
>+}
>+
>+bool AAudioRecorder::Recording() const {
>+  return recording_;
>+}
>+
>+void AAudioRecorder::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
>+  RTC_LOG(INFO) << "AttachAudioBuffer";
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  audio_device_buffer_ = audioBuffer;
>+  const AudioParameters audio_parameters = aaudio_.audio_parameters();
>+  audio_device_buffer_->SetRecordingSampleRate(audio_parameters.sample_rate());
>+  audio_device_buffer_->SetRecordingChannels(audio_parameters.channels());
>+  RTC_CHECK(audio_device_buffer_);
>+  // Create a modified audio buffer class which allows us to deliver any number
>+  // of samples (and not only multiples of 10ms which WebRTC uses) to match the
>+  // native AAudio buffer size.
>+  fine_audio_buffer_ = absl::make_unique<FineAudioBuffer>(audio_device_buffer_);
>+}
>+
>+bool AAudioRecorder::IsAcousticEchoCancelerSupported() const {
>+  return false;
>+}
>+
>+bool AAudioRecorder::IsNoiseSuppressorSupported() const {
>+  return false;
>+}
>+
>+int AAudioRecorder::EnableBuiltInAEC(bool enable) {
>+  RTC_LOG(INFO) << "EnableBuiltInAEC: " << enable;
>+  RTC_LOG(LS_ERROR) << "Not implemented";
>+  return -1;
>+}
>+
>+int AAudioRecorder::EnableBuiltInNS(bool enable) {
>+  RTC_LOG(INFO) << "EnableBuiltInNS: " << enable;
>+  RTC_LOG(LS_ERROR) << "Not implemented";
>+  return -1;
>+}
>+
>+void AAudioRecorder::OnErrorCallback(aaudio_result_t error) {
>+  RTC_LOG(LS_ERROR) << "OnErrorCallback: " << AAudio_convertResultToText(error);
>+  // RTC_DCHECK(thread_checker_aaudio_.CalledOnValidThread());
>+  if (aaudio_.stream_state() == AAUDIO_STREAM_STATE_DISCONNECTED) {
>+    // The stream is disconnected and any attempt to use it will return
>+    // AAUDIO_ERROR_DISCONNECTED..
>+    RTC_LOG(WARNING) << "Input stream disconnected => restart is required";
>+    // AAudio documentation states: "You should not close or reopen the stream
>+    // from the callback, use another thread instead". A message is therefore
>+    // sent to the main thread to do the restart operation.
>+    RTC_DCHECK(main_thread_);
>+    main_thread_->Post(RTC_FROM_HERE, this, kMessageInputStreamDisconnected);
>+  }
>+}
>+
>+// Read and process |num_frames| of data from the |audio_data| buffer.
>+// TODO(henrika): possibly add trace here to be included in systrace.
>+// See https://developer.android.com/studio/profile/systrace-commandline.html.
>+aaudio_data_callback_result_t AAudioRecorder::OnDataCallback(
>+    void* audio_data,
>+    int32_t num_frames) {
>+  // TODO(henrika): figure out why we sometimes hit this one.
>+  // RTC_DCHECK(thread_checker_aaudio_.CalledOnValidThread());
>+  // RTC_LOG(INFO) << "OnDataCallback: " << num_frames;
>+  // Drain the input buffer at first callback to ensure that it does not
>+  // contain any old data. Will also ensure that the lowest possible latency
>+  // is obtained.
>+  if (first_data_callback_) {
>+    RTC_LOG(INFO) << "--- First input data callback: "
>+                  << "device id=" << aaudio_.device_id();
>+    aaudio_.ClearInputStream(audio_data, num_frames);
>+    first_data_callback_ = false;
>+  }
>+  // Check if the overflow counter has increased and if so log a warning.
>+  // TODO(henrika): possible add UMA stat or capacity extension.
>+  const int32_t overflow_count = aaudio_.xrun_count();
>+  if (overflow_count > overflow_count_) {
>+    RTC_LOG(LS_ERROR) << "Overflow detected: " << overflow_count;
>+    overflow_count_ = overflow_count;
>+  }
>+  // Estimated time between an audio frame was recorded by the input device and
>+  // it can read on the input stream.
>+  latency_millis_ = aaudio_.EstimateLatencyMillis();
>+  // TODO(henrika): use for development only.
>+  if (aaudio_.frames_read() % (1000 * aaudio_.frames_per_burst()) == 0) {
>+    RTC_DLOG(INFO) << "input latency: " << latency_millis_
>+                   << ", num_frames: " << num_frames;
>+  }
>+  // Copy recorded audio in |audio_data| to the WebRTC sink using the
>+  // FineAudioBuffer object.
>+  fine_audio_buffer_->DeliverRecordedData(
>+      rtc::MakeArrayView(static_cast<const int16_t*>(audio_data),
>+                         aaudio_.samples_per_frame() * num_frames),
>+      static_cast<int>(latency_millis_ + 0.5));
>+
>+  return AAUDIO_CALLBACK_RESULT_CONTINUE;
>+}
>+
>+void AAudioRecorder::OnMessage(rtc::Message* msg) {
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  switch (msg->message_id) {
>+    case kMessageInputStreamDisconnected:
>+      HandleStreamDisconnected();
>+      break;
>+    default:
>+      RTC_LOG(LS_ERROR) << "Invalid message id: " << msg->message_id;
>+      break;
>+  }
>+}
>+
>+void AAudioRecorder::HandleStreamDisconnected() {
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  RTC_LOG(INFO) << "HandleStreamDisconnected";
>+  if (!initialized_ || !recording_) {
>+    return;
>+  }
>+  // Perform a restart by first closing the disconnected stream and then start
>+  // a new stream; this time using the new (preferred) audio input device.
>+  // TODO(henrika): resolve issue where a one restart attempt leads to a long
>+  // sequence of new calls to OnErrorCallback().
>+  // See b/73148976 for details.
>+  audio_device_buffer_->NativeAudioRecordingInterrupted();
>+  StopRecording();
>+  InitRecording();
>+  StartRecording();
>+}
>+
>+}  // namespace jni
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/aaudio_recorder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/aaudio_recorder.h
>new file mode 100644
>index 00000000000..ec380830af6
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/aaudio_recorder.h
>@@ -0,0 +1,133 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef SDK_ANDROID_SRC_JNI_AUDIO_DEVICE_AAUDIO_RECORDER_H_
>+#define SDK_ANDROID_SRC_JNI_AUDIO_DEVICE_AAUDIO_RECORDER_H_
>+
>+#include <aaudio/AAudio.h>
>+#include <memory>
>+
>+#include "modules/audio_device/audio_device_buffer.h"
>+#include "modules/audio_device/include/audio_device_defines.h"
>+#include "rtc_base/messagehandler.h"
>+#include "rtc_base/thread.h"
>+#include "rtc_base/thread_checker.h"
>+#include "sdk/android/src/jni/audio_device/aaudio_wrapper.h"
>+#include "sdk/android/src/jni/audio_device/audio_device_module.h"
>+
>+namespace webrtc {
>+
>+class FineAudioBuffer;
>+class AudioDeviceBuffer;
>+
>+namespace jni {
>+
>+// Implements low-latency 16-bit mono PCM audio input support for Android
>+// using the C based AAudio API.
>+//
>+// An instance must be created and destroyed on one and the same thread.
>+// All public methods must also be called on the same thread. A thread checker
>+// will RTC_DCHECK if any method is called on an invalid thread. Audio buffers
>+// are delivered on a dedicated high-priority thread owned by AAudio.
>+//
>+// The existing design forces the user to call InitRecording() after
>+// StopRecording() to be able to call StartRecording() again. This is in line
>+// with how the Java- based implementation works.
>+//
>+// TODO(henrika): add comments about device changes and adaptive buffer
>+// management.
>+class AAudioRecorder : public AudioInput,
>+                       public AAudioObserverInterface,
>+                       public rtc::MessageHandler {
>+ public:
>+  explicit AAudioRecorder(const AudioParameters& audio_parameters);
>+  ~AAudioRecorder() override;
>+
>+  int Init() override;
>+  int Terminate() override;
>+
>+  int InitRecording() override;
>+  bool RecordingIsInitialized() const override;
>+
>+  int StartRecording() override;
>+  int StopRecording() override;
>+  bool Recording() const override;
>+
>+  void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override;
>+
>+  // TODO(henrika): add support using AAudio APIs when available.
>+  bool IsAcousticEchoCancelerSupported() const override;
>+  bool IsNoiseSuppressorSupported() const override;
>+  int EnableBuiltInAEC(bool enable) override;
>+  int EnableBuiltInNS(bool enable) override;
>+
>+ protected:
>+  // AAudioObserverInterface implementation.
>+
>+  // For an input stream, this function should read |num_frames| of recorded
>+  // data, in the stream's current data format, from the |audio_data| buffer.
>+  // Called on a real-time thread owned by AAudio.
>+  aaudio_data_callback_result_t OnDataCallback(void* audio_data,
>+                                               int32_t num_frames) override;
>+
>+  // AAudio calls this function if any error occurs on a callback thread.
>+  // Called on a real-time thread owned by AAudio.
>+  void OnErrorCallback(aaudio_result_t error) override;
>+
>+  // rtc::MessageHandler used for restart messages.
>+  void OnMessage(rtc::Message* msg) override;
>+
>+ private:
>+  // Closes the existing stream and starts a new stream.
>+  void HandleStreamDisconnected();
>+
>+  // Ensures that methods are called from the same thread as this object is
>+  // created on.
>+  rtc::ThreadChecker thread_checker_;
>+
>+  // Stores thread ID in first call to AAudioPlayer::OnDataCallback from a
>+  // real-time thread owned by AAudio. Detached during construction of this
>+  // object.
>+  rtc::ThreadChecker thread_checker_aaudio_;
>+
>+  // The thread on which this object is created on.
>+  rtc::Thread* main_thread_;
>+
>+  // Wraps all AAudio resources. Contains an input stream using the default
>+  // input audio device.
>+  AAudioWrapper aaudio_;
>+
>+  // Raw pointer handle provided to us in AttachAudioBuffer(). Owned by the
>+  // AudioDeviceModuleImpl class and called by AudioDeviceModule::Create().
>+  AudioDeviceBuffer* audio_device_buffer_ = nullptr;
>+
>+  bool initialized_ = false;
>+  bool recording_ = false;
>+
>+  // Consumes audio of native buffer size and feeds the WebRTC layer with 10ms
>+  // chunks of audio.
>+  std::unique_ptr<FineAudioBuffer> fine_audio_buffer_;
>+
>+  // Counts number of detected overflow events reported by AAudio.
>+  int32_t overflow_count_ = 0;
>+
>+  // Estimated time between an audio frame was recorded by the input device and
>+  // it can read on the input stream.
>+  double latency_millis_ = 0;
>+
>+  // True only for the first data callback in each audio session.
>+  bool first_data_callback_ = true;
>+};
>+
>+}  // namespace jni
>+
>+}  // namespace webrtc
>+
>+#endif  // SDK_ANDROID_SRC_JNI_AUDIO_DEVICE_AAUDIO_RECORDER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/aaudio_wrapper.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/aaudio_wrapper.cc
>new file mode 100644
>index 00000000000..a9d647f69ae
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/aaudio_wrapper.cc
>@@ -0,0 +1,501 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "sdk/android/src/jni/audio_device/aaudio_wrapper.h"
>+
>+#include "rtc_base/logging.h"
>+#include "rtc_base/strings/string_builder.h"
>+#include "rtc_base/timeutils.h"
>+
>+#define LOG_ON_ERROR(op)                                                      \
>+  do {                                                                        \
>+    aaudio_result_t result = (op);                                            \
>+    if (result != AAUDIO_OK) {                                                \
>+      RTC_LOG(LS_ERROR) << #op << ": " << AAudio_convertResultToText(result); \
>+    }                                                                         \
>+  } while (0)
>+
>+#define RETURN_ON_ERROR(op, ...)                                              \
>+  do {                                                                        \
>+    aaudio_result_t result = (op);                                            \
>+    if (result != AAUDIO_OK) {                                                \
>+      RTC_LOG(LS_ERROR) << #op << ": " << AAudio_convertResultToText(result); \
>+      return __VA_ARGS__;                                                     \
>+    }                                                                         \
>+  } while (0)
>+
>+namespace webrtc {
>+
>+namespace jni {
>+
>+namespace {
>+
>+const char* DirectionToString(aaudio_direction_t direction) {
>+  switch (direction) {
>+    case AAUDIO_DIRECTION_OUTPUT:
>+      return "OUTPUT";
>+    case AAUDIO_DIRECTION_INPUT:
>+      return "INPUT";
>+    default:
>+      return "UNKNOWN";
>+  }
>+}
>+
>+const char* SharingModeToString(aaudio_sharing_mode_t mode) {
>+  switch (mode) {
>+    case AAUDIO_SHARING_MODE_EXCLUSIVE:
>+      return "EXCLUSIVE";
>+    case AAUDIO_SHARING_MODE_SHARED:
>+      return "SHARED";
>+    default:
>+      return "UNKNOWN";
>+  }
>+}
>+
>+const char* PerformanceModeToString(aaudio_performance_mode_t mode) {
>+  switch (mode) {
>+    case AAUDIO_PERFORMANCE_MODE_NONE:
>+      return "NONE";
>+    case AAUDIO_PERFORMANCE_MODE_POWER_SAVING:
>+      return "POWER_SAVING";
>+    case AAUDIO_PERFORMANCE_MODE_LOW_LATENCY:
>+      return "LOW_LATENCY";
>+    default:
>+      return "UNKNOWN";
>+  }
>+}
>+
>+const char* FormatToString(int32_t id) {
>+  switch (id) {
>+    case AAUDIO_FORMAT_INVALID:
>+      return "INVALID";
>+    case AAUDIO_FORMAT_UNSPECIFIED:
>+      return "UNSPECIFIED";
>+    case AAUDIO_FORMAT_PCM_I16:
>+      return "PCM_I16";
>+    case AAUDIO_FORMAT_PCM_FLOAT:
>+      return "FLOAT";
>+    default:
>+      return "UNKNOWN";
>+  }
>+}
>+
>+void ErrorCallback(AAudioStream* stream,
>+                   void* user_data,
>+                   aaudio_result_t error) {
>+  RTC_DCHECK(user_data);
>+  AAudioWrapper* aaudio_wrapper = reinterpret_cast<AAudioWrapper*>(user_data);
>+  RTC_LOG(WARNING) << "ErrorCallback: "
>+                   << DirectionToString(aaudio_wrapper->direction());
>+  RTC_DCHECK(aaudio_wrapper->observer());
>+  aaudio_wrapper->observer()->OnErrorCallback(error);
>+}
>+
>+aaudio_data_callback_result_t DataCallback(AAudioStream* stream,
>+                                           void* user_data,
>+                                           void* audio_data,
>+                                           int32_t num_frames) {
>+  RTC_DCHECK(user_data);
>+  RTC_DCHECK(audio_data);
>+  AAudioWrapper* aaudio_wrapper = reinterpret_cast<AAudioWrapper*>(user_data);
>+  RTC_DCHECK(aaudio_wrapper->observer());
>+  return aaudio_wrapper->observer()->OnDataCallback(audio_data, num_frames);
>+}
>+
>+// Wraps the stream builder object to ensure that it is released properly when
>+// the stream builder goes out of scope.
>+class ScopedStreamBuilder {
>+ public:
>+  ScopedStreamBuilder() {
>+    LOG_ON_ERROR(AAudio_createStreamBuilder(&builder_));
>+    RTC_DCHECK(builder_);
>+  }
>+  ~ScopedStreamBuilder() {
>+    if (builder_) {
>+      LOG_ON_ERROR(AAudioStreamBuilder_delete(builder_));
>+    }
>+  }
>+
>+  AAudioStreamBuilder* get() const { return builder_; }
>+
>+ private:
>+  AAudioStreamBuilder* builder_ = nullptr;
>+};
>+
>+}  // namespace
>+
>+AAudioWrapper::AAudioWrapper(const AudioParameters& audio_parameters,
>+                             aaudio_direction_t direction,
>+                             AAudioObserverInterface* observer)
>+    : audio_parameters_(audio_parameters),
>+      direction_(direction),
>+      observer_(observer) {
>+  RTC_LOG(INFO) << "ctor";
>+  RTC_DCHECK(observer_);
>+  aaudio_thread_checker_.DetachFromThread();
>+  RTC_LOG(INFO) << audio_parameters_.ToString();
>+}
>+
>+AAudioWrapper::~AAudioWrapper() {
>+  RTC_LOG(INFO) << "dtor";
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  RTC_DCHECK(!stream_);
>+}
>+
>+bool AAudioWrapper::Init() {
>+  RTC_LOG(INFO) << "Init";
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  // Creates a stream builder which can be used to open an audio stream.
>+  ScopedStreamBuilder builder;
>+  // Configures the stream builder using audio parameters given at construction.
>+  SetStreamConfiguration(builder.get());
>+  // Opens a stream based on options in the stream builder.
>+  if (!OpenStream(builder.get())) {
>+    return false;
>+  }
>+  // Ensures that the opened stream could activate the requested settings.
>+  if (!VerifyStreamConfiguration()) {
>+    return false;
>+  }
>+  // Optimizes the buffer scheme for lowest possible latency and creates
>+  // additional buffer logic to match the 10ms buffer size used in WebRTC.
>+  if (!OptimizeBuffers()) {
>+    return false;
>+  }
>+  LogStreamState();
>+  return true;
>+}
>+
>+bool AAudioWrapper::Start() {
>+  RTC_LOG(INFO) << "Start";
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  // TODO(henrika): this state check might not be needed.
>+  aaudio_stream_state_t current_state = AAudioStream_getState(stream_);
>+  if (current_state != AAUDIO_STREAM_STATE_OPEN) {
>+    RTC_LOG(LS_ERROR) << "Invalid state: "
>+                      << AAudio_convertStreamStateToText(current_state);
>+    return false;
>+  }
>+  // Asynchronous request for the stream to start.
>+  RETURN_ON_ERROR(AAudioStream_requestStart(stream_), false);
>+  LogStreamState();
>+  return true;
>+}
>+
>+bool AAudioWrapper::Stop() {
>+  RTC_LOG(INFO) << "Stop: " << DirectionToString(direction());
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  // Asynchronous request for the stream to stop.
>+  RETURN_ON_ERROR(AAudioStream_requestStop(stream_), false);
>+  CloseStream();
>+  aaudio_thread_checker_.DetachFromThread();
>+  return true;
>+}
>+
>+double AAudioWrapper::EstimateLatencyMillis() const {
>+  RTC_DCHECK(stream_);
>+  double latency_millis = 0.0;
>+  if (direction() == AAUDIO_DIRECTION_INPUT) {
>+    // For input streams. Best guess we can do is to use the current burst size
>+    // as delay estimate.
>+    latency_millis = static_cast<double>(frames_per_burst()) / sample_rate() *
>+                     rtc::kNumMillisecsPerSec;
>+  } else {
>+    int64_t existing_frame_index;
>+    int64_t existing_frame_presentation_time;
>+    // Get the time at which a particular frame was presented to audio hardware.
>+    aaudio_result_t result = AAudioStream_getTimestamp(
>+        stream_, CLOCK_MONOTONIC, &existing_frame_index,
>+        &existing_frame_presentation_time);
>+    // Results are only valid when the stream is in AAUDIO_STREAM_STATE_STARTED.
>+    if (result == AAUDIO_OK) {
>+      // Get write index for next audio frame.
>+      int64_t next_frame_index = frames_written();
>+      // Number of frames between next frame and the existing frame.
>+      int64_t frame_index_delta = next_frame_index - existing_frame_index;
>+      // Assume the next frame will be written now.
>+      int64_t next_frame_write_time = rtc::TimeNanos();
>+      // Calculate time when next frame will be presented to the hardware taking
>+      // sample rate into account.
>+      int64_t frame_time_delta =
>+          (frame_index_delta * rtc::kNumNanosecsPerSec) / sample_rate();
>+      int64_t next_frame_presentation_time =
>+          existing_frame_presentation_time + frame_time_delta;
>+      // Derive a latency estimate given results above.
>+      latency_millis = static_cast<double>(next_frame_presentation_time -
>+                                           next_frame_write_time) /
>+                       rtc::kNumNanosecsPerMillisec;
>+    }
>+  }
>+  return latency_millis;
>+}
>+
>+// Returns new buffer size or a negative error value if buffer size could not
>+// be increased.
>+bool AAudioWrapper::IncreaseOutputBufferSize() {
>+  RTC_LOG(INFO) << "IncreaseBufferSize";
>+  RTC_DCHECK(stream_);
>+  RTC_DCHECK(aaudio_thread_checker_.CalledOnValidThread());
>+  RTC_DCHECK_EQ(direction(), AAUDIO_DIRECTION_OUTPUT);
>+  aaudio_result_t buffer_size = AAudioStream_getBufferSizeInFrames(stream_);
>+  // Try to increase size of buffer with one burst to reduce risk of underrun.
>+  buffer_size += frames_per_burst();
>+  // Verify that the new buffer size is not larger than max capacity.
>+  // TODO(henrika): keep track of case when we reach the capacity limit.
>+  const int32_t max_buffer_size = buffer_capacity_in_frames();
>+  if (buffer_size > max_buffer_size) {
>+    RTC_LOG(LS_ERROR) << "Required buffer size (" << buffer_size
>+                      << ") is higher than max: " << max_buffer_size;
>+    return false;
>+  }
>+  RTC_LOG(INFO) << "Updating buffer size to: " << buffer_size
>+                << " (max=" << max_buffer_size << ")";
>+  buffer_size = AAudioStream_setBufferSizeInFrames(stream_, buffer_size);
>+  if (buffer_size < 0) {
>+    RTC_LOG(LS_ERROR) << "Failed to change buffer size: "
>+                      << AAudio_convertResultToText(buffer_size);
>+    return false;
>+  }
>+  RTC_LOG(INFO) << "Buffer size changed to: " << buffer_size;
>+  return true;
>+}
>+
>+void AAudioWrapper::ClearInputStream(void* audio_data, int32_t num_frames) {
>+  RTC_LOG(INFO) << "ClearInputStream";
>+  RTC_DCHECK(stream_);
>+  RTC_DCHECK(aaudio_thread_checker_.CalledOnValidThread());
>+  RTC_DCHECK_EQ(direction(), AAUDIO_DIRECTION_INPUT);
>+  aaudio_result_t cleared_frames = 0;
>+  do {
>+    cleared_frames = AAudioStream_read(stream_, audio_data, num_frames, 0);
>+  } while (cleared_frames > 0);
>+}
>+
>+AAudioObserverInterface* AAudioWrapper::observer() const {
>+  return observer_;
>+}
>+
>+AudioParameters AAudioWrapper::audio_parameters() const {
>+  return audio_parameters_;
>+}
>+
>+int32_t AAudioWrapper::samples_per_frame() const {
>+  RTC_DCHECK(stream_);
>+  return AAudioStream_getSamplesPerFrame(stream_);
>+}
>+
>+int32_t AAudioWrapper::buffer_size_in_frames() const {
>+  RTC_DCHECK(stream_);
>+  return AAudioStream_getBufferSizeInFrames(stream_);
>+}
>+
>+int32_t AAudioWrapper::buffer_capacity_in_frames() const {
>+  RTC_DCHECK(stream_);
>+  return AAudioStream_getBufferCapacityInFrames(stream_);
>+}
>+
>+int32_t AAudioWrapper::device_id() const {
>+  RTC_DCHECK(stream_);
>+  return AAudioStream_getDeviceId(stream_);
>+}
>+
>+int32_t AAudioWrapper::xrun_count() const {
>+  RTC_DCHECK(stream_);
>+  return AAudioStream_getXRunCount(stream_);
>+}
>+
>+int32_t AAudioWrapper::format() const {
>+  RTC_DCHECK(stream_);
>+  return AAudioStream_getFormat(stream_);
>+}
>+
>+int32_t AAudioWrapper::sample_rate() const {
>+  RTC_DCHECK(stream_);
>+  return AAudioStream_getSampleRate(stream_);
>+}
>+
>+int32_t AAudioWrapper::channel_count() const {
>+  RTC_DCHECK(stream_);
>+  return AAudioStream_getChannelCount(stream_);
>+}
>+
>+int32_t AAudioWrapper::frames_per_callback() const {
>+  RTC_DCHECK(stream_);
>+  return AAudioStream_getFramesPerDataCallback(stream_);
>+}
>+
>+aaudio_sharing_mode_t AAudioWrapper::sharing_mode() const {
>+  RTC_DCHECK(stream_);
>+  return AAudioStream_getSharingMode(stream_);
>+}
>+
>+aaudio_performance_mode_t AAudioWrapper::performance_mode() const {
>+  RTC_DCHECK(stream_);
>+  return AAudioStream_getPerformanceMode(stream_);
>+}
>+
>+aaudio_stream_state_t AAudioWrapper::stream_state() const {
>+  RTC_DCHECK(stream_);
>+  return AAudioStream_getState(stream_);
>+}
>+
>+int64_t AAudioWrapper::frames_written() const {
>+  RTC_DCHECK(stream_);
>+  return AAudioStream_getFramesWritten(stream_);
>+}
>+
>+int64_t AAudioWrapper::frames_read() const {
>+  RTC_DCHECK(stream_);
>+  return AAudioStream_getFramesRead(stream_);
>+}
>+
>+void AAudioWrapper::SetStreamConfiguration(AAudioStreamBuilder* builder) {
>+  RTC_LOG(INFO) << "SetStreamConfiguration";
>+  RTC_DCHECK(builder);
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  // Request usage of default primary output/input device.
>+  // TODO(henrika): verify that default device follows Java APIs.
>+  // https://developer.android.com/reference/android/media/AudioDeviceInfo.html.
>+  AAudioStreamBuilder_setDeviceId(builder, AAUDIO_UNSPECIFIED);
>+  // Use preferred sample rate given by the audio parameters.
>+  AAudioStreamBuilder_setSampleRate(builder, audio_parameters().sample_rate());
>+  // Use preferred channel configuration given by the audio parameters.
>+  AAudioStreamBuilder_setChannelCount(builder, audio_parameters().channels());
>+  // Always use 16-bit PCM audio sample format.
>+  AAudioStreamBuilder_setFormat(builder, AAUDIO_FORMAT_PCM_I16);
>+  // TODO(henrika): investigate effect of using AAUDIO_SHARING_MODE_EXCLUSIVE.
>+  // Ask for exclusive mode since this will give us the lowest possible latency.
>+  // If exclusive mode isn't available, shared mode will be used instead.
>+  AAudioStreamBuilder_setSharingMode(builder, AAUDIO_SHARING_MODE_SHARED);
>+  // Use the direction that was given at construction.
>+  AAudioStreamBuilder_setDirection(builder, direction_);
>+  // TODO(henrika): investigate performance using different performance modes.
>+  AAudioStreamBuilder_setPerformanceMode(builder,
>+                                         AAUDIO_PERFORMANCE_MODE_LOW_LATENCY);
>+  // Given that WebRTC applications require low latency, our audio stream uses
>+  // an asynchronous callback function to transfer data to and from the
>+  // application. AAudio executes the callback in a higher-priority thread that
>+  // has better performance.
>+  AAudioStreamBuilder_setDataCallback(builder, DataCallback, this);
>+  // Request that AAudio calls this functions if any error occurs on a callback
>+  // thread.
>+  AAudioStreamBuilder_setErrorCallback(builder, ErrorCallback, this);
>+}
>+
>+bool AAudioWrapper::OpenStream(AAudioStreamBuilder* builder) {
>+  RTC_LOG(INFO) << "OpenStream";
>+  RTC_DCHECK(builder);
>+  AAudioStream* stream = nullptr;
>+  RETURN_ON_ERROR(AAudioStreamBuilder_openStream(builder, &stream), false);
>+  stream_ = stream;
>+  LogStreamConfiguration();
>+  return true;
>+}
>+
>+void AAudioWrapper::CloseStream() {
>+  RTC_LOG(INFO) << "CloseStream";
>+  RTC_DCHECK(stream_);
>+  LOG_ON_ERROR(AAudioStream_close(stream_));
>+  stream_ = nullptr;
>+}
>+
>+void AAudioWrapper::LogStreamConfiguration() {
>+  RTC_DCHECK(stream_);
>+  char ss_buf[1024];
>+  rtc::SimpleStringBuilder ss(ss_buf);
>+  ss << "Stream Configuration: ";
>+  ss << "sample rate=" << sample_rate() << ", channels=" << channel_count();
>+  ss << ", samples per frame=" << samples_per_frame();
>+  ss << ", format=" << FormatToString(format());
>+  ss << ", sharing mode=" << SharingModeToString(sharing_mode());
>+  ss << ", performance mode=" << PerformanceModeToString(performance_mode());
>+  ss << ", direction=" << DirectionToString(direction());
>+  ss << ", device id=" << AAudioStream_getDeviceId(stream_);
>+  ss << ", frames per callback=" << frames_per_callback();
>+  RTC_LOG(INFO) << ss.str();
>+}
>+
>+void AAudioWrapper::LogStreamState() {
>+  RTC_LOG(INFO) << "AAudio stream state: "
>+                << AAudio_convertStreamStateToText(stream_state());
>+}
>+
>+bool AAudioWrapper::VerifyStreamConfiguration() {
>+  RTC_LOG(INFO) << "VerifyStreamConfiguration";
>+  RTC_DCHECK(stream_);
>+  // TODO(henrika): should we verify device ID as well?
>+  if (AAudioStream_getSampleRate(stream_) != audio_parameters().sample_rate()) {
>+    RTC_LOG(LS_ERROR) << "Stream unable to use requested sample rate";
>+    return false;
>+  }
>+  if (AAudioStream_getChannelCount(stream_) !=
>+      static_cast<int32_t>(audio_parameters().channels())) {
>+    RTC_LOG(LS_ERROR) << "Stream unable to use requested channel count";
>+    return false;
>+  }
>+  if (AAudioStream_getFormat(stream_) != AAUDIO_FORMAT_PCM_I16) {
>+    RTC_LOG(LS_ERROR) << "Stream unable to use requested format";
>+    return false;
>+  }
>+  if (AAudioStream_getSharingMode(stream_) != AAUDIO_SHARING_MODE_SHARED) {
>+    RTC_LOG(LS_ERROR) << "Stream unable to use requested sharing mode";
>+    return false;
>+  }
>+  if (AAudioStream_getPerformanceMode(stream_) !=
>+      AAUDIO_PERFORMANCE_MODE_LOW_LATENCY) {
>+    RTC_LOG(LS_ERROR) << "Stream unable to use requested performance mode";
>+    return false;
>+  }
>+  if (AAudioStream_getDirection(stream_) != direction()) {
>+    RTC_LOG(LS_ERROR) << "Stream direction could not be set";
>+    return false;
>+  }
>+  if (AAudioStream_getSamplesPerFrame(stream_) !=
>+      static_cast<int32_t>(audio_parameters().channels())) {
>+    RTC_LOG(LS_ERROR) << "Invalid number of samples per frame";
>+    return false;
>+  }
>+  return true;
>+}
>+
>+bool AAudioWrapper::OptimizeBuffers() {
>+  RTC_LOG(INFO) << "OptimizeBuffers";
>+  RTC_DCHECK(stream_);
>+  // Maximum number of frames that can be filled without blocking.
>+  RTC_LOG(INFO) << "max buffer capacity in frames: "
>+                << buffer_capacity_in_frames();
>+  // Query the number of frames that the application should read or write at
>+  // one time for optimal performance.
>+  int32_t frames_per_burst = AAudioStream_getFramesPerBurst(stream_);
>+  RTC_LOG(INFO) << "frames per burst for optimal performance: "
>+                << frames_per_burst;
>+  frames_per_burst_ = frames_per_burst;
>+  if (direction() == AAUDIO_DIRECTION_INPUT) {
>+    // There is no point in calling setBufferSizeInFrames() for input streams
>+    // since it has no effect on the performance (latency in this case).
>+    return true;
>+  }
>+  // Set buffer size to same as burst size to guarantee lowest possible latency.
>+  // This size might change for output streams if underruns are detected and
>+  // automatic buffer adjustment is enabled.
>+  AAudioStream_setBufferSizeInFrames(stream_, frames_per_burst);
>+  int32_t buffer_size = AAudioStream_getBufferSizeInFrames(stream_);
>+  if (buffer_size != frames_per_burst) {
>+    RTC_LOG(LS_ERROR) << "Failed to use optimal buffer burst size";
>+    return false;
>+  }
>+  // Maximum number of frames that can be filled without blocking.
>+  RTC_LOG(INFO) << "buffer burst size in frames: " << buffer_size;
>+  return true;
>+}
>+
>+}  // namespace jni
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/aaudio_wrapper.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/aaudio_wrapper.h
>new file mode 100644
>index 00000000000..1900ab988c8
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/aaudio_wrapper.h
>@@ -0,0 +1,129 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef SDK_ANDROID_SRC_JNI_AUDIO_DEVICE_AAUDIO_WRAPPER_H_
>+#define SDK_ANDROID_SRC_JNI_AUDIO_DEVICE_AAUDIO_WRAPPER_H_
>+
>+#include <aaudio/AAudio.h>
>+
>+#include "modules/audio_device/include/audio_device_defines.h"
>+#include "rtc_base/thread_checker.h"
>+
>+namespace webrtc {
>+
>+namespace jni {
>+
>+// AAudio callback interface for audio transport to/from the AAudio stream.
>+// The interface also contains an error callback method for notifications of
>+// e.g. device changes.
>+class AAudioObserverInterface {
>+ public:
>+  // Audio data will be passed in our out of this function dependning on the
>+  // direction of the audio stream. This callback function will be called on a
>+  // real-time thread owned by AAudio.
>+  virtual aaudio_data_callback_result_t OnDataCallback(void* audio_data,
>+                                                       int32_t num_frames) = 0;
>+  // AAudio will call this functions if any error occurs on a callback thread.
>+  // In response, this function could signal or launch another thread to reopen
>+  // a stream on another device. Do not reopen the stream in this callback.
>+  virtual void OnErrorCallback(aaudio_result_t error) = 0;
>+
>+ protected:
>+  virtual ~AAudioObserverInterface() {}
>+};
>+
>+// Utility class which wraps the C-based AAudio API into a more handy C++ class
>+// where the underlying resources (AAudioStreamBuilder and AAudioStream) are
>+// encapsulated. User must set the direction (in or out) at construction since
>+// it defines the stream type and the direction of the data flow in the
>+// AAudioObserverInterface.
>+//
>+// AAudio is a new Android C API introduced in the Android O (26) release.
>+// It is designed for high-performance audio applications that require low
>+// latency. Applications communicate with AAudio by reading and writing data
>+// to streams.
>+//
>+// Each stream is attached to a single audio device, where each audio device
>+// has a unique ID. The ID can be used to bind an audio stream to a specific
>+// audio device but this implementation lets AAudio choose the default primary
>+// device instead (device selection takes place in Java). A stream can only
>+// move data in one direction. When a stream is opened, Android checks to
>+// ensure that the audio device and stream direction agree.
>+class AAudioWrapper {
>+ public:
>+  AAudioWrapper(const AudioParameters& audio_parameters,
>+                aaudio_direction_t direction,
>+                AAudioObserverInterface* observer);
>+  ~AAudioWrapper();
>+
>+  bool Init();
>+  bool Start();
>+  bool Stop();
>+
>+  // For output streams: estimates latency between writing an audio frame to
>+  // the output stream and the time that same frame is played out on the output
>+  // audio device.
>+  // For input streams: estimates latency between reading an audio frame from
>+  // the input stream and the time that same frame was recorded on the input
>+  // audio device.
>+  double EstimateLatencyMillis() const;
>+
>+  // Increases the internal buffer size for output streams by one burst size to
>+  // reduce the risk of underruns. Can be used while a stream is active.
>+  bool IncreaseOutputBufferSize();
>+
>+  // Drains the recording stream of any existing data by reading from it until
>+  // it's empty. Can be used to clear out old data before starting a new audio
>+  // session.
>+  void ClearInputStream(void* audio_data, int32_t num_frames);
>+
>+  AAudioObserverInterface* observer() const;
>+  AudioParameters audio_parameters() const;
>+  int32_t samples_per_frame() const;
>+  int32_t buffer_size_in_frames() const;
>+  int32_t buffer_capacity_in_frames() const;
>+  int32_t device_id() const;
>+  int32_t xrun_count() const;
>+  int32_t format() const;
>+  int32_t sample_rate() const;
>+  int32_t channel_count() const;
>+  int32_t frames_per_callback() const;
>+  aaudio_sharing_mode_t sharing_mode() const;
>+  aaudio_performance_mode_t performance_mode() const;
>+  aaudio_stream_state_t stream_state() const;
>+  int64_t frames_written() const;
>+  int64_t frames_read() const;
>+  aaudio_direction_t direction() const { return direction_; }
>+  AAudioStream* stream() const { return stream_; }
>+  int32_t frames_per_burst() const { return frames_per_burst_; }
>+
>+ private:
>+  void SetStreamConfiguration(AAudioStreamBuilder* builder);
>+  bool OpenStream(AAudioStreamBuilder* builder);
>+  void CloseStream();
>+  void LogStreamConfiguration();
>+  void LogStreamState();
>+  bool VerifyStreamConfiguration();
>+  bool OptimizeBuffers();
>+
>+  rtc::ThreadChecker thread_checker_;
>+  rtc::ThreadChecker aaudio_thread_checker_;
>+  const AudioParameters audio_parameters_;
>+  const aaudio_direction_t direction_;
>+  AAudioObserverInterface* observer_ = nullptr;
>+  AAudioStream* stream_ = nullptr;
>+  int32_t frames_per_burst_ = 0;
>+};
>+
>+}  // namespace jni
>+
>+}  // namespace webrtc
>+
>+#endif  // SDK_ANDROID_SRC_JNI_AUDIO_DEVICE_AAUDIO_WRAPPER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/audio_common.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/audio_common.h
>new file mode 100644
>index 00000000000..fdecf384c94
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/audio_common.h
>@@ -0,0 +1,32 @@
>+/*
>+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef SDK_ANDROID_SRC_JNI_AUDIO_DEVICE_AUDIO_COMMON_H_
>+#define SDK_ANDROID_SRC_JNI_AUDIO_DEVICE_AUDIO_COMMON_H_
>+
>+namespace webrtc {
>+
>+namespace jni {
>+
>+const int kDefaultSampleRate = 44100;
>+// Delay estimates for the two different supported modes. These values are based
>+// on real-time round-trip delay estimates on a large set of devices and they
>+// are lower bounds since the filter length is 128 ms, so the AEC works for
>+// delays in the range [50, ~170] ms and [150, ~270] ms. Note that, in most
>+// cases, the lowest delay estimate will not be utilized since devices that
>+// support low-latency output audio often supports HW AEC as well.
>+const int kLowLatencyModeDelayEstimateInMilliseconds = 50;
>+const int kHighLatencyModeDelayEstimateInMilliseconds = 150;
>+
>+}  // namespace jni
>+
>+}  // namespace webrtc
>+
>+#endif  // SDK_ANDROID_SRC_JNI_AUDIO_DEVICE_AUDIO_COMMON_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/audio_device_module.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/audio_device_module.cc
>new file mode 100644
>index 00000000000..dc3a8fe4ba5
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/audio_device_module.cc
>@@ -0,0 +1,649 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "sdk/android/src/jni/audio_device/audio_device_module.h"
>+
>+#include <utility>
>+
>+#include "rtc_base/checks.h"
>+#include "rtc_base/logging.h"
>+#include "rtc_base/refcountedobject.h"
>+#include "rtc_base/thread_checker.h"
>+#include "sdk/android/generated_audio_device_module_base_jni/jni/WebRtcAudioManager_jni.h"
>+#include "system_wrappers/include/metrics.h"
>+
>+namespace webrtc {
>+namespace jni {
>+
>+namespace {
>+
>+// This class combines a generic instance of an AudioInput and a generic
>+// instance of an AudioOutput to create an AudioDeviceModule. This is mostly
>+// done by delegating to the audio input/output with some glue code. This class
>+// also directly implements some of the AudioDeviceModule methods with dummy
>+// implementations.
>+//
>+// An instance can be created on any thread, but must then be used on one and
>+// the same thread. All public methods must also be called on the same thread.
>+// A thread checker will RTC_DCHECK if any method is called on an invalid
>+// thread.
>+// TODO(henrika): it might be useful to also support a scenario where the ADM
>+// is constructed on thread T1, used on thread T2 and destructed on T2 or T3.
>+// If so, care must be taken to ensure that only T2 is a COM thread.
>+class AndroidAudioDeviceModule : public AudioDeviceModule {
>+ public:
>+  // For use with UMA logging. Must be kept in sync with histograms.xml in
>+  // Chrome, located at
>+  // https://cs.chromium.org/chromium/src/tools/metrics/histograms/histograms.xml
>+  enum class InitStatus {
>+    OK = 0,
>+    PLAYOUT_ERROR = 1,
>+    RECORDING_ERROR = 2,
>+    OTHER_ERROR = 3,
>+    NUM_STATUSES = 4
>+  };
>+
>+  AndroidAudioDeviceModule(AudioDeviceModule::AudioLayer audio_layer,
>+                           bool is_stereo_playout_supported,
>+                           bool is_stereo_record_supported,
>+                           uint16_t playout_delay_ms,
>+                           std::unique_ptr<AudioInput> audio_input,
>+                           std::unique_ptr<AudioOutput> audio_output)
>+      : audio_layer_(audio_layer),
>+        is_stereo_playout_supported_(is_stereo_playout_supported),
>+        is_stereo_record_supported_(is_stereo_record_supported),
>+        playout_delay_ms_(playout_delay_ms),
>+        input_(std::move(audio_input)),
>+        output_(std::move(audio_output)),
>+        initialized_(false) {
>+    RTC_CHECK(input_);
>+    RTC_CHECK(output_);
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    thread_checker_.DetachFromThread();
>+  }
>+
>+  ~AndroidAudioDeviceModule() override { RTC_LOG(INFO) << __FUNCTION__; }
>+
>+  int32_t ActiveAudioLayer(
>+      AudioDeviceModule::AudioLayer* audioLayer) const override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    *audioLayer = audio_layer_;
>+    return 0;
>+  }
>+
>+  int32_t RegisterAudioCallback(AudioTransport* audioCallback) override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    return audio_device_buffer_->RegisterAudioCallback(audioCallback);
>+  }
>+
>+  int32_t Init() override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+    audio_device_buffer_ = absl::make_unique<AudioDeviceBuffer>();
>+    AttachAudioBuffer();
>+    if (initialized_) {
>+      return 0;
>+    }
>+    InitStatus status;
>+    if (output_->Init() != 0) {
>+      status = InitStatus::PLAYOUT_ERROR;
>+    } else if (input_->Init() != 0) {
>+      output_->Terminate();
>+      status = InitStatus::RECORDING_ERROR;
>+    } else {
>+      initialized_ = true;
>+      status = InitStatus::OK;
>+    }
>+    RTC_HISTOGRAM_ENUMERATION("WebRTC.Audio.InitializationResult",
>+                              static_cast<int>(status),
>+                              static_cast<int>(InitStatus::NUM_STATUSES));
>+    if (status != InitStatus::OK) {
>+      RTC_LOG(LS_ERROR) << "Audio device initialization failed.";
>+      return -1;
>+    }
>+    return 0;
>+  }
>+
>+  int32_t Terminate() override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    if (!initialized_)
>+      return 0;
>+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+    int32_t err = input_->Terminate();
>+    err |= output_->Terminate();
>+    initialized_ = false;
>+    thread_checker_.DetachFromThread();
>+    audio_device_buffer_.reset(nullptr);
>+    RTC_DCHECK_EQ(err, 0);
>+    return err;
>+  }
>+
>+  bool Initialized() const override {
>+    RTC_LOG(INFO) << __FUNCTION__ << ":" << initialized_;
>+    return initialized_;
>+  }
>+
>+  int16_t PlayoutDevices() override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    RTC_LOG(INFO) << "output: " << 1;
>+    return 1;
>+  }
>+
>+  int16_t RecordingDevices() override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    RTC_LOG(INFO) << "output: " << 1;
>+    return 1;
>+  }
>+
>+  int32_t PlayoutDeviceName(uint16_t index,
>+                            char name[kAdmMaxDeviceNameSize],
>+                            char guid[kAdmMaxGuidSize]) override {
>+    FATAL() << "Should never be called";
>+    return -1;
>+  }
>+
>+  int32_t RecordingDeviceName(uint16_t index,
>+                              char name[kAdmMaxDeviceNameSize],
>+                              char guid[kAdmMaxGuidSize]) override {
>+    FATAL() << "Should never be called";
>+    return -1;
>+  }
>+
>+  int32_t SetPlayoutDevice(uint16_t index) override {
>+    // OK to use but it has no effect currently since device selection is
>+    // done using Andoid APIs instead.
>+    RTC_LOG(INFO) << __FUNCTION__ << "(" << index << ")";
>+    return 0;
>+  }
>+
>+  int32_t SetPlayoutDevice(
>+      AudioDeviceModule::WindowsDeviceType device) override {
>+    FATAL() << "Should never be called";
>+    return -1;
>+  }
>+
>+  int32_t SetRecordingDevice(uint16_t index) override {
>+    // OK to use but it has no effect currently since device selection is
>+    // done using Andoid APIs instead.
>+    RTC_LOG(INFO) << __FUNCTION__ << "(" << index << ")";
>+    return 0;
>+  }
>+
>+  int32_t SetRecordingDevice(
>+      AudioDeviceModule::WindowsDeviceType device) override {
>+    FATAL() << "Should never be called";
>+    return -1;
>+  }
>+
>+  int32_t PlayoutIsAvailable(bool* available) override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    *available = true;
>+    RTC_LOG(INFO) << "output: " << *available;
>+    return 0;
>+  }
>+
>+  int32_t InitPlayout() override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    if (!initialized_)
>+      return -1;
>+    if (PlayoutIsInitialized()) {
>+      return 0;
>+    }
>+    int32_t result = output_->InitPlayout();
>+    RTC_LOG(INFO) << "output: " << result;
>+    RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.InitPlayoutSuccess",
>+                          static_cast<int>(result == 0));
>+    return result;
>+  }
>+
>+  bool PlayoutIsInitialized() const override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    return output_->PlayoutIsInitialized();
>+  }
>+
>+  int32_t RecordingIsAvailable(bool* available) override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    *available = true;
>+    RTC_LOG(INFO) << "output: " << *available;
>+    return 0;
>+  }
>+
>+  int32_t InitRecording() override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    if (!initialized_)
>+      return -1;
>+    if (RecordingIsInitialized()) {
>+      return 0;
>+    }
>+    int32_t result = input_->InitRecording();
>+    RTC_LOG(INFO) << "output: " << result;
>+    RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.InitRecordingSuccess",
>+                          static_cast<int>(result == 0));
>+    return result;
>+  }
>+
>+  bool RecordingIsInitialized() const override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    return input_->RecordingIsInitialized();
>+  }
>+
>+  int32_t StartPlayout() override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    if (!initialized_)
>+      return -1;
>+    if (Playing()) {
>+      return 0;
>+    }
>+    int32_t result = output_->StartPlayout();
>+    RTC_LOG(INFO) << "output: " << result;
>+    RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StartPlayoutSuccess",
>+                          static_cast<int>(result == 0));
>+    if (result == 0) {
>+      // Only start playing the audio device buffer if starting the audio
>+      // output succeeded.
>+      audio_device_buffer_->StartPlayout();
>+    }
>+    return result;
>+  }
>+
>+  int32_t StopPlayout() override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    if (!initialized_)
>+      return -1;
>+    if (!Playing())
>+      return 0;
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    audio_device_buffer_->StopPlayout();
>+    int32_t result = output_->StopPlayout();
>+    RTC_LOG(INFO) << "output: " << result;
>+    RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StopPlayoutSuccess",
>+                          static_cast<int>(result == 0));
>+    return result;
>+  }
>+
>+  bool Playing() const override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    return output_->Playing();
>+  }
>+
>+  int32_t StartRecording() override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    if (!initialized_)
>+      return -1;
>+    if (Recording()) {
>+      return 0;
>+    }
>+    int32_t result = input_->StartRecording();
>+    RTC_LOG(INFO) << "output: " << result;
>+    RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StartRecordingSuccess",
>+                          static_cast<int>(result == 0));
>+    if (result == 0) {
>+      // Only start recording the audio device buffer if starting the audio
>+      // input succeeded.
>+      audio_device_buffer_->StartRecording();
>+    }
>+    return result;
>+  }
>+
>+  int32_t StopRecording() override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    if (!initialized_)
>+      return -1;
>+    if (!Recording())
>+      return 0;
>+    audio_device_buffer_->StopRecording();
>+    int32_t result = input_->StopRecording();
>+    RTC_LOG(INFO) << "output: " << result;
>+    RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StopRecordingSuccess",
>+                          static_cast<int>(result == 0));
>+    return result;
>+  }
>+
>+  bool Recording() const override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    return input_->Recording();
>+  }
>+
>+  int32_t InitSpeaker() override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    return initialized_ ? 0 : -1;
>+  }
>+
>+  bool SpeakerIsInitialized() const override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    return initialized_;
>+  }
>+
>+  int32_t InitMicrophone() override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    return initialized_ ? 0 : -1;
>+  }
>+
>+  bool MicrophoneIsInitialized() const override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    return initialized_;
>+  }
>+
>+  int32_t SpeakerVolumeIsAvailable(bool* available) override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    if (!initialized_)
>+      return -1;
>+    *available = output_->SpeakerVolumeIsAvailable();
>+    RTC_LOG(INFO) << "output: " << *available;
>+    return 0;
>+  }
>+
>+  int32_t SetSpeakerVolume(uint32_t volume) override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    if (!initialized_)
>+      return -1;
>+    return output_->SetSpeakerVolume(volume);
>+  }
>+
>+  int32_t SpeakerVolume(uint32_t* output_volume) const override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    if (!initialized_)
>+      return -1;
>+    absl::optional<uint32_t> volume = output_->SpeakerVolume();
>+    if (!volume)
>+      return -1;
>+    *output_volume = *volume;
>+    RTC_LOG(INFO) << "output: " << *volume;
>+    return 0;
>+  }
>+
>+  int32_t MaxSpeakerVolume(uint32_t* output_max_volume) const override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    if (!initialized_)
>+      return -1;
>+    absl::optional<uint32_t> max_volume = output_->MaxSpeakerVolume();
>+    if (!max_volume)
>+      return -1;
>+    *output_max_volume = *max_volume;
>+    return 0;
>+  }
>+
>+  int32_t MinSpeakerVolume(uint32_t* output_min_volume) const override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    if (!initialized_)
>+      return -1;
>+    absl::optional<uint32_t> min_volume = output_->MinSpeakerVolume();
>+    if (!min_volume)
>+      return -1;
>+    *output_min_volume = *min_volume;
>+    return 0;
>+  }
>+
>+  int32_t MicrophoneVolumeIsAvailable(bool* available) override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    *available = false;
>+    RTC_LOG(INFO) << "output: " << *available;
>+    return -1;
>+  }
>+
>+  int32_t SetMicrophoneVolume(uint32_t volume) override {
>+    RTC_LOG(INFO) << __FUNCTION__ << "(" << volume << ")";
>+    FATAL() << "Should never be called";
>+    return -1;
>+  }
>+
>+  int32_t MicrophoneVolume(uint32_t* volume) const override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    FATAL() << "Should never be called";
>+    return -1;
>+  }
>+
>+  int32_t MaxMicrophoneVolume(uint32_t* maxVolume) const override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    FATAL() << "Should never be called";
>+    return -1;
>+  }
>+
>+  int32_t MinMicrophoneVolume(uint32_t* minVolume) const override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    FATAL() << "Should never be called";
>+    return -1;
>+  }
>+
>+  int32_t SpeakerMuteIsAvailable(bool* available) override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    FATAL() << "Should never be called";
>+    return -1;
>+  }
>+
>+  int32_t SetSpeakerMute(bool enable) override {
>+    RTC_LOG(INFO) << __FUNCTION__ << "(" << enable << ")";
>+    FATAL() << "Should never be called";
>+    return -1;
>+  }
>+
>+  int32_t SpeakerMute(bool* enabled) const override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    FATAL() << "Should never be called";
>+    return -1;
>+  }
>+
>+  int32_t MicrophoneMuteIsAvailable(bool* available) override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    FATAL() << "Not implemented";
>+    return -1;
>+  }
>+
>+  int32_t SetMicrophoneMute(bool enable) override {
>+    RTC_LOG(INFO) << __FUNCTION__ << "(" << enable << ")";
>+    FATAL() << "Not implemented";
>+    return -1;
>+  }
>+
>+  int32_t MicrophoneMute(bool* enabled) const override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    FATAL() << "Not implemented";
>+    return -1;
>+  }
>+
>+  int32_t StereoPlayoutIsAvailable(bool* available) const override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    *available = is_stereo_playout_supported_;
>+    RTC_LOG(INFO) << "output: " << *available;
>+    return 0;
>+  }
>+
>+  int32_t SetStereoPlayout(bool enable) override {
>+    RTC_LOG(INFO) << __FUNCTION__ << "(" << enable << ")";
>+    // Android does not support changes between mono and stero on the fly. The
>+    // use of stereo or mono is determined by the audio layer. It is allowed
>+    // to call this method if that same state is not modified.
>+    bool available = is_stereo_playout_supported_;
>+    if (enable != available) {
>+      RTC_LOG(WARNING) << "changing stereo playout not supported";
>+      return -1;
>+    }
>+    return 0;
>+  }
>+
>+  int32_t StereoPlayout(bool* enabled) const override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    *enabled = is_stereo_playout_supported_;
>+    RTC_LOG(INFO) << "output: " << *enabled;
>+    return 0;
>+  }
>+
>+  int32_t StereoRecordingIsAvailable(bool* available) const override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    *available = is_stereo_record_supported_;
>+    RTC_LOG(INFO) << "output: " << *available;
>+    return 0;
>+  }
>+
>+  int32_t SetStereoRecording(bool enable) override {
>+    RTC_LOG(INFO) << __FUNCTION__ << "(" << enable << ")";
>+    // Android does not support changes between mono and stero on the fly. The
>+    // use of stereo or mono is determined by the audio layer. It is allowed
>+    // to call this method if that same state is not modified.
>+    bool available = is_stereo_record_supported_;
>+    if (enable != available) {
>+      RTC_LOG(WARNING) << "changing stereo recording not supported";
>+      return -1;
>+    }
>+    return 0;
>+  }
>+
>+  int32_t StereoRecording(bool* enabled) const override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    *enabled = is_stereo_record_supported_;
>+    RTC_LOG(INFO) << "output: " << *enabled;
>+    return 0;
>+  }
>+
>+  int32_t PlayoutDelay(uint16_t* delay_ms) const override {
>+    // Best guess we can do is to use half of the estimated total delay.
>+    *delay_ms = playout_delay_ms_;
>+    RTC_DCHECK_GT(*delay_ms, 0);
>+    return 0;
>+  }
>+
>+  // Returns true if the device both supports built in AEC and the device
>+  // is not blacklisted.
>+  // Currently, if OpenSL ES is used in both directions, this method will still
>+  // report the correct value and it has the correct effect. As an example:
>+  // a device supports built in AEC and this method returns true. Libjingle
>+  // will then disable the WebRTC based AEC and that will work for all devices
>+  // (mainly Nexus) even when OpenSL ES is used for input since our current
>+  // implementation will enable built-in AEC by default also for OpenSL ES.
>+  // The only "bad" thing that happens today is that when Libjingle calls
>+  // OpenSLESRecorder::EnableBuiltInAEC() it will not have any real effect and
>+  // a "Not Implemented" log will be filed. This non-perfect state will remain
>+  // until I have added full support for audio effects based on OpenSL ES APIs.
>+  bool BuiltInAECIsAvailable() const override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    if (!initialized_)
>+      return false;
>+    bool isAvailable = input_->IsAcousticEchoCancelerSupported();
>+    RTC_LOG(INFO) << "output: " << isAvailable;
>+    return isAvailable;
>+  }
>+
>+  // Not implemented for any input device on Android.
>+  bool BuiltInAGCIsAvailable() const override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    RTC_LOG(INFO) << "output: " << false;
>+    return false;
>+  }
>+
>+  // Returns true if the device both supports built in NS and the device
>+  // is not blacklisted.
>+  // TODO(henrika): add implementation for OpenSL ES based audio as well.
>+  // In addition, see comments for BuiltInAECIsAvailable().
>+  bool BuiltInNSIsAvailable() const override {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    if (!initialized_)
>+      return false;
>+    bool isAvailable = input_->IsNoiseSuppressorSupported();
>+    RTC_LOG(INFO) << "output: " << isAvailable;
>+    return isAvailable;
>+  }
>+
>+  // TODO(henrika): add implementation for OpenSL ES based audio as well.
>+  int32_t EnableBuiltInAEC(bool enable) override {
>+    RTC_LOG(INFO) << __FUNCTION__ << "(" << enable << ")";
>+    if (!initialized_)
>+      return -1;
>+    RTC_CHECK(BuiltInAECIsAvailable()) << "HW AEC is not available";
>+    int32_t result = input_->EnableBuiltInAEC(enable);
>+    RTC_LOG(INFO) << "output: " << result;
>+    return result;
>+  }
>+
>+  int32_t EnableBuiltInAGC(bool enable) override {
>+    RTC_LOG(INFO) << __FUNCTION__ << "(" << enable << ")";
>+    FATAL() << "HW AGC is not available";
>+    return -1;
>+  }
>+
>+  // TODO(henrika): add implementation for OpenSL ES based audio as well.
>+  int32_t EnableBuiltInNS(bool enable) override {
>+    RTC_LOG(INFO) << __FUNCTION__ << "(" << enable << ")";
>+    if (!initialized_)
>+      return -1;
>+    RTC_CHECK(BuiltInNSIsAvailable()) << "HW NS is not available";
>+    int32_t result = input_->EnableBuiltInNS(enable);
>+    RTC_LOG(INFO) << "output: " << result;
>+    return result;
>+  }
>+
>+  int32_t AttachAudioBuffer() {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    output_->AttachAudioBuffer(audio_device_buffer_.get());
>+    input_->AttachAudioBuffer(audio_device_buffer_.get());
>+    return 0;
>+  }
>+
>+ private:
>+  rtc::ThreadChecker thread_checker_;
>+
>+  const AudioDeviceModule::AudioLayer audio_layer_;
>+  const bool is_stereo_playout_supported_;
>+  const bool is_stereo_record_supported_;
>+  const uint16_t playout_delay_ms_;
>+  const std::unique_ptr<AudioInput> input_;
>+  const std::unique_ptr<AudioOutput> output_;
>+  std::unique_ptr<AudioDeviceBuffer> audio_device_buffer_;
>+
>+  bool initialized_;
>+};
>+
>+}  // namespace
>+
>+ScopedJavaLocalRef<jobject> GetAudioManager(JNIEnv* env,
>+                                            const JavaRef<jobject>& j_context) {
>+  return Java_WebRtcAudioManager_getAudioManager(env, j_context);
>+}
>+
>+int GetDefaultSampleRate(JNIEnv* env, const JavaRef<jobject>& j_audio_manager) {
>+  return Java_WebRtcAudioManager_getSampleRate(env, j_audio_manager);
>+}
>+
>+void GetAudioParameters(JNIEnv* env,
>+                        const JavaRef<jobject>& j_context,
>+                        const JavaRef<jobject>& j_audio_manager,
>+                        int sample_rate,
>+                        bool use_stereo_input,
>+                        bool use_stereo_output,
>+                        AudioParameters* input_parameters,
>+                        AudioParameters* output_parameters) {
>+  const int output_channels = use_stereo_output ? 2 : 1;
>+  const int input_channels = use_stereo_input ? 2 : 1;
>+  const size_t output_buffer_size = Java_WebRtcAudioManager_getOutputBufferSize(
>+      env, j_context, j_audio_manager, sample_rate, output_channels);
>+  const size_t input_buffer_size = Java_WebRtcAudioManager_getInputBufferSize(
>+      env, j_context, j_audio_manager, sample_rate, input_channels);
>+  output_parameters->reset(sample_rate, static_cast<size_t>(output_channels),
>+                           static_cast<size_t>(output_buffer_size));
>+  input_parameters->reset(sample_rate, static_cast<size_t>(input_channels),
>+                          static_cast<size_t>(input_buffer_size));
>+  RTC_CHECK(input_parameters->is_valid());
>+  RTC_CHECK(output_parameters->is_valid());
>+}
>+
>+rtc::scoped_refptr<AudioDeviceModule> CreateAudioDeviceModuleFromInputAndOutput(
>+    AudioDeviceModule::AudioLayer audio_layer,
>+    bool is_stereo_playout_supported,
>+    bool is_stereo_record_supported,
>+    uint16_t playout_delay_ms,
>+    std::unique_ptr<AudioInput> audio_input,
>+    std::unique_ptr<AudioOutput> audio_output) {
>+  RTC_LOG(INFO) << __FUNCTION__;
>+  return new rtc::RefCountedObject<AndroidAudioDeviceModule>(
>+      audio_layer, is_stereo_playout_supported, is_stereo_record_supported,
>+      playout_delay_ms, std::move(audio_input), std::move(audio_output));
>+}
>+
>+}  // namespace jni
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/audio_device_module.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/audio_device_module.h
>new file mode 100644
>index 00000000000..cddd3e06769
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/audio_device_module.h
>@@ -0,0 +1,97 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef SDK_ANDROID_SRC_JNI_AUDIO_DEVICE_AUDIO_DEVICE_MODULE_H_
>+#define SDK_ANDROID_SRC_JNI_AUDIO_DEVICE_AUDIO_DEVICE_MODULE_H_
>+
>+#include <memory>
>+
>+#include "absl/types/optional.h"
>+#include "modules/audio_device/audio_device_buffer.h"
>+#include "sdk/android/native_api/jni/scoped_java_ref.h"
>+
>+namespace webrtc {
>+namespace jni {
>+
>+class AudioInput {
>+ public:
>+  virtual ~AudioInput() {}
>+
>+  virtual int32_t Init() = 0;
>+  virtual int32_t Terminate() = 0;
>+
>+  virtual int32_t InitRecording() = 0;
>+  virtual bool RecordingIsInitialized() const = 0;
>+
>+  virtual int32_t StartRecording() = 0;
>+  virtual int32_t StopRecording() = 0;
>+  virtual bool Recording() const = 0;
>+
>+  virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) = 0;
>+
>+  // Returns true if the audio input supports built-in audio effects for AEC and
>+  // NS.
>+  virtual bool IsAcousticEchoCancelerSupported() const = 0;
>+  virtual bool IsNoiseSuppressorSupported() const = 0;
>+
>+  virtual int32_t EnableBuiltInAEC(bool enable) = 0;
>+  virtual int32_t EnableBuiltInNS(bool enable) = 0;
>+};
>+
>+class AudioOutput {
>+ public:
>+  virtual ~AudioOutput() {}
>+
>+  virtual int32_t Init() = 0;
>+  virtual int32_t Terminate() = 0;
>+  virtual int32_t InitPlayout() = 0;
>+  virtual bool PlayoutIsInitialized() const = 0;
>+  virtual int32_t StartPlayout() = 0;
>+  virtual int32_t StopPlayout() = 0;
>+  virtual bool Playing() const = 0;
>+  virtual bool SpeakerVolumeIsAvailable() = 0;
>+  virtual int SetSpeakerVolume(uint32_t volume) = 0;
>+  virtual absl::optional<uint32_t> SpeakerVolume() const = 0;
>+  virtual absl::optional<uint32_t> MaxSpeakerVolume() const = 0;
>+  virtual absl::optional<uint32_t> MinSpeakerVolume() const = 0;
>+  virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) = 0;
>+};
>+
>+// Extract an android.media.AudioManager from an android.content.Context.
>+ScopedJavaLocalRef<jobject> GetAudioManager(JNIEnv* env,
>+                                            const JavaRef<jobject>& j_context);
>+
>+// Get default audio sample rate by querying an android.media.AudioManager.
>+int GetDefaultSampleRate(JNIEnv* env, const JavaRef<jobject>& j_audio_manager);
>+
>+// Get audio input and output parameters based on a number of settings.
>+void GetAudioParameters(JNIEnv* env,
>+                        const JavaRef<jobject>& j_context,
>+                        const JavaRef<jobject>& j_audio_manager,
>+                        int sample_rate,
>+                        bool use_stereo_input,
>+                        bool use_stereo_output,
>+                        AudioParameters* input_parameters,
>+                        AudioParameters* output_parameters);
>+
>+// Glue together an audio input and audio output to get an AudioDeviceModule.
>+rtc::scoped_refptr<AudioDeviceModule> CreateAudioDeviceModuleFromInputAndOutput(
>+    AudioDeviceModule::AudioLayer audio_layer,
>+    bool is_stereo_playout_supported,
>+    bool is_stereo_record_supported,
>+    uint16_t playout_delay_ms,
>+    std::unique_ptr<AudioInput> audio_input,
>+    std::unique_ptr<AudioOutput> audio_output);
>+
>+}  // namespace jni
>+
>+}  // namespace webrtc
>+
>+#endif  // SDK_ANDROID_SRC_JNI_AUDIO_DEVICE_AUDIO_DEVICE_MODULE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/audio_record_jni.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/audio_record_jni.cc
>new file mode 100644
>index 00000000000..390cec0e9e7
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/audio_record_jni.cc
>@@ -0,0 +1,255 @@
>+/*
>+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "sdk/android/src/jni/audio_device/audio_record_jni.h"
>+
>+#include <string>
>+#include <utility>
>+
>+#include "rtc_base/arraysize.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/format_macros.h"
>+#include "rtc_base/logging.h"
>+#include "rtc_base/platform_thread.h"
>+#include "rtc_base/timeutils.h"
>+#include "sdk/android/generated_java_audio_device_module_native_jni/jni/WebRtcAudioRecord_jni.h"
>+#include "sdk/android/src/jni/audio_device/audio_common.h"
>+#include "sdk/android/src/jni/jni_helpers.h"
>+#include "system_wrappers/include/metrics.h"
>+
>+namespace webrtc {
>+
>+namespace jni {
>+
>+namespace {
>+// Scoped class which logs its time of life as a UMA statistic. It generates
>+// a histogram which measures the time it takes for a method/scope to execute.
>+class ScopedHistogramTimer {
>+ public:
>+  explicit ScopedHistogramTimer(const std::string& name)
>+      : histogram_name_(name), start_time_ms_(rtc::TimeMillis()) {}
>+  ~ScopedHistogramTimer() {
>+    const int64_t life_time_ms = rtc::TimeSince(start_time_ms_);
>+    RTC_HISTOGRAM_COUNTS_1000(histogram_name_, life_time_ms);
>+    RTC_LOG(INFO) << histogram_name_ << ": " << life_time_ms;
>+  }
>+
>+ private:
>+  const std::string histogram_name_;
>+  int64_t start_time_ms_;
>+};
>+
>+}  // namespace
>+
>+ScopedJavaLocalRef<jobject> AudioRecordJni::CreateJavaWebRtcAudioRecord(
>+    JNIEnv* env,
>+    const JavaRef<jobject>& j_context,
>+    const JavaRef<jobject>& j_audio_manager) {
>+  return Java_WebRtcAudioRecord_Constructor(env, j_context, j_audio_manager);
>+}
>+
>+AudioRecordJni::AudioRecordJni(JNIEnv* env,
>+                               const AudioParameters& audio_parameters,
>+                               int total_delay_ms,
>+                               const JavaRef<jobject>& j_audio_record)
>+    : j_audio_record_(env, j_audio_record),
>+      audio_parameters_(audio_parameters),
>+      total_delay_ms_(total_delay_ms),
>+      direct_buffer_address_(nullptr),
>+      direct_buffer_capacity_in_bytes_(0),
>+      frames_per_buffer_(0),
>+      initialized_(false),
>+      recording_(false),
>+      audio_device_buffer_(nullptr) {
>+  RTC_LOG(INFO) << "ctor";
>+  RTC_DCHECK(audio_parameters_.is_valid());
>+  Java_WebRtcAudioRecord_setNativeAudioRecord(env, j_audio_record_,
>+                                              jni::jlongFromPointer(this));
>+  // Detach from this thread since construction is allowed to happen on a
>+  // different thread.
>+  thread_checker_.DetachFromThread();
>+  thread_checker_java_.DetachFromThread();
>+}
>+
>+AudioRecordJni::~AudioRecordJni() {
>+  RTC_LOG(INFO) << "dtor";
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  Terminate();
>+}
>+
>+int32_t AudioRecordJni::Init() {
>+  RTC_LOG(INFO) << "Init";
>+  env_ = AttachCurrentThreadIfNeeded();
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  return 0;
>+}
>+
>+int32_t AudioRecordJni::Terminate() {
>+  RTC_LOG(INFO) << "Terminate";
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  StopRecording();
>+  thread_checker_.DetachFromThread();
>+  return 0;
>+}
>+
>+int32_t AudioRecordJni::InitRecording() {
>+  RTC_LOG(INFO) << "InitRecording";
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  if (initialized_) {
>+    // Already initialized.
>+    return 0;
>+  }
>+  RTC_DCHECK(!recording_);
>+  ScopedHistogramTimer timer("WebRTC.Audio.InitRecordingDurationMs");
>+
>+  int frames_per_buffer = Java_WebRtcAudioRecord_initRecording(
>+      env_, j_audio_record_, audio_parameters_.sample_rate(),
>+      static_cast<int>(audio_parameters_.channels()));
>+  if (frames_per_buffer < 0) {
>+    direct_buffer_address_ = nullptr;
>+    RTC_LOG(LS_ERROR) << "InitRecording failed";
>+    return -1;
>+  }
>+  frames_per_buffer_ = static_cast<size_t>(frames_per_buffer);
>+  RTC_LOG(INFO) << "frames_per_buffer: " << frames_per_buffer_;
>+  const size_t bytes_per_frame = audio_parameters_.channels() * sizeof(int16_t);
>+  RTC_CHECK_EQ(direct_buffer_capacity_in_bytes_,
>+               frames_per_buffer_ * bytes_per_frame);
>+  RTC_CHECK_EQ(frames_per_buffer_, audio_parameters_.frames_per_10ms_buffer());
>+  initialized_ = true;
>+  return 0;
>+}
>+
>+bool AudioRecordJni::RecordingIsInitialized() const {
>+  return initialized_;
>+}
>+
>+int32_t AudioRecordJni::StartRecording() {
>+  RTC_LOG(INFO) << "StartRecording";
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  if (recording_) {
>+    // Already recording.
>+    return 0;
>+  }
>+  if (!initialized_) {
>+    RTC_DLOG(LS_WARNING)
>+        << "Recording can not start since InitRecording must succeed first";
>+    return 0;
>+  }
>+  ScopedHistogramTimer timer("WebRTC.Audio.StartRecordingDurationMs");
>+  if (!Java_WebRtcAudioRecord_startRecording(env_, j_audio_record_)) {
>+    RTC_LOG(LS_ERROR) << "StartRecording failed";
>+    return -1;
>+  }
>+  recording_ = true;
>+  return 0;
>+}
>+
>+int32_t AudioRecordJni::StopRecording() {
>+  RTC_LOG(INFO) << "StopRecording";
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  if (!initialized_ || !recording_) {
>+    return 0;
>+  }
>+  if (!Java_WebRtcAudioRecord_stopRecording(env_, j_audio_record_)) {
>+    RTC_LOG(LS_ERROR) << "StopRecording failed";
>+    return -1;
>+  }
>+  // If we don't detach here, we will hit a RTC_DCHECK in OnDataIsRecorded()
>+  // next time StartRecording() is called since it will create a new Java
>+  // thread.
>+  thread_checker_java_.DetachFromThread();
>+  initialized_ = false;
>+  recording_ = false;
>+  direct_buffer_address_ = nullptr;
>+  return 0;
>+}
>+
>+bool AudioRecordJni::Recording() const {
>+  return recording_;
>+}
>+
>+void AudioRecordJni::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
>+  RTC_LOG(INFO) << "AttachAudioBuffer";
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  audio_device_buffer_ = audioBuffer;
>+  const int sample_rate_hz = audio_parameters_.sample_rate();
>+  RTC_LOG(INFO) << "SetRecordingSampleRate(" << sample_rate_hz << ")";
>+  audio_device_buffer_->SetRecordingSampleRate(sample_rate_hz);
>+  const size_t channels = audio_parameters_.channels();
>+  RTC_LOG(INFO) << "SetRecordingChannels(" << channels << ")";
>+  audio_device_buffer_->SetRecordingChannels(channels);
>+}
>+
>+bool AudioRecordJni::IsAcousticEchoCancelerSupported() const {
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  return Java_WebRtcAudioRecord_isAcousticEchoCancelerSupported(
>+      env_, j_audio_record_);
>+}
>+
>+bool AudioRecordJni::IsNoiseSuppressorSupported() const {
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  return Java_WebRtcAudioRecord_isNoiseSuppressorSupported(env_,
>+                                                           j_audio_record_);
>+}
>+
>+int32_t AudioRecordJni::EnableBuiltInAEC(bool enable) {
>+  RTC_LOG(INFO) << "EnableBuiltInAEC(" << enable << ")";
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  return Java_WebRtcAudioRecord_enableBuiltInAEC(env_, j_audio_record_, enable)
>+             ? 0
>+             : -1;
>+}
>+
>+int32_t AudioRecordJni::EnableBuiltInNS(bool enable) {
>+  RTC_LOG(INFO) << "EnableBuiltInNS(" << enable << ")";
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  return Java_WebRtcAudioRecord_enableBuiltInNS(env_, j_audio_record_, enable)
>+             ? 0
>+             : -1;
>+}
>+
>+void AudioRecordJni::CacheDirectBufferAddress(
>+    JNIEnv* env,
>+    const JavaParamRef<jobject>& j_caller,
>+    const JavaParamRef<jobject>& byte_buffer) {
>+  RTC_LOG(INFO) << "OnCacheDirectBufferAddress";
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  RTC_DCHECK(!direct_buffer_address_);
>+  direct_buffer_address_ = env->GetDirectBufferAddress(byte_buffer.obj());
>+  jlong capacity = env->GetDirectBufferCapacity(byte_buffer.obj());
>+  RTC_LOG(INFO) << "direct buffer capacity: " << capacity;
>+  direct_buffer_capacity_in_bytes_ = static_cast<size_t>(capacity);
>+}
>+
>+// This method is called on a high-priority thread from Java. The name of
>+// the thread is 'AudioRecordThread'.
>+void AudioRecordJni::DataIsRecorded(JNIEnv* env,
>+                                    const JavaParamRef<jobject>& j_caller,
>+                                    int length) {
>+  RTC_DCHECK(thread_checker_java_.CalledOnValidThread());
>+  if (!audio_device_buffer_) {
>+    RTC_LOG(LS_ERROR) << "AttachAudioBuffer has not been called";
>+    return;
>+  }
>+  audio_device_buffer_->SetRecordedBuffer(direct_buffer_address_,
>+                                          frames_per_buffer_);
>+  // We provide one (combined) fixed delay estimate for the APM and use the
>+  // |playDelayMs| parameter only. Components like the AEC only sees the sum
>+  // of |playDelayMs| and |recDelayMs|, hence the distributions does not matter.
>+  audio_device_buffer_->SetVQEData(total_delay_ms_, 0);
>+  if (audio_device_buffer_->DeliverRecordedData() == -1) {
>+    RTC_LOG(INFO) << "AudioDeviceBuffer::DeliverRecordedData failed";
>+  }
>+}
>+
>+}  // namespace jni
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/audio_record_jni.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/audio_record_jni.h
>new file mode 100644
>index 00000000000..7578f839484
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/audio_record_jni.h
>@@ -0,0 +1,138 @@
>+/*
>+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef SDK_ANDROID_SRC_JNI_AUDIO_DEVICE_AUDIO_RECORD_JNI_H_
>+#define SDK_ANDROID_SRC_JNI_AUDIO_DEVICE_AUDIO_RECORD_JNI_H_
>+
>+#include <jni.h>
>+#include <memory>
>+
>+#include "modules/audio_device/audio_device_buffer.h"
>+#include "modules/audio_device/include/audio_device_defines.h"
>+#include "rtc_base/thread_checker.h"
>+#include "sdk/android/src/jni/audio_device/audio_device_module.h"
>+
>+namespace webrtc {
>+
>+namespace jni {
>+
>+// Implements 16-bit mono PCM audio input support for Android using the Java
>+// AudioRecord interface. Most of the work is done by its Java counterpart in
>+// WebRtcAudioRecord.java. This class is created and lives on a thread in
>+// C++-land, but recorded audio buffers are delivered on a high-priority
>+// thread managed by the Java class.
>+//
>+// The Java class makes use of AudioEffect features (mainly AEC) which are
>+// first available in Jelly Bean. If it is instantiated running against earlier
>+// SDKs, the AEC provided by the APM in WebRTC must be used and enabled
>+// separately instead.
>+//
>+// An instance can be created on any thread, but must then be used on one and
>+// the same thread. All public methods must also be called on the same thread. A
>+// thread checker will RTC_DCHECK if any method is called on an invalid thread.
>+//
>+// This class uses AttachCurrentThreadIfNeeded to attach to a Java VM if needed.
>+// Additional thread checking guarantees that no other (possibly non attached)
>+// thread is used.
>+class AudioRecordJni : public AudioInput {
>+ public:
>+  static ScopedJavaLocalRef<jobject> CreateJavaWebRtcAudioRecord(
>+      JNIEnv* env,
>+      const JavaRef<jobject>& j_context,
>+      const JavaRef<jobject>& j_audio_manager);
>+
>+  AudioRecordJni(JNIEnv* env,
>+                 const AudioParameters& audio_parameters,
>+                 int total_delay_ms,
>+                 const JavaRef<jobject>& j_webrtc_audio_record);
>+  ~AudioRecordJni() override;
>+
>+  int32_t Init() override;
>+  int32_t Terminate() override;
>+
>+  int32_t InitRecording() override;
>+  bool RecordingIsInitialized() const override;
>+
>+  int32_t StartRecording() override;
>+  int32_t StopRecording() override;
>+  bool Recording() const override;
>+
>+  void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override;
>+
>+  bool IsAcousticEchoCancelerSupported() const override;
>+  bool IsNoiseSuppressorSupported() const override;
>+
>+  int32_t EnableBuiltInAEC(bool enable) override;
>+  int32_t EnableBuiltInNS(bool enable) override;
>+
>+  // Called from Java side so we can cache the address of the Java-manged
>+  // |byte_buffer| in |direct_buffer_address_|. The size of the buffer
>+  // is also stored in |direct_buffer_capacity_in_bytes_|.
>+  // This method will be called by the WebRtcAudioRecord constructor, i.e.,
>+  // on the same thread that this object is created on.
>+  void CacheDirectBufferAddress(JNIEnv* env,
>+                                const JavaParamRef<jobject>& j_caller,
>+                                const JavaParamRef<jobject>& byte_buffer);
>+
>+  // Called periodically by the Java based WebRtcAudioRecord object when
>+  // recording has started. Each call indicates that there are |length| new
>+  // bytes recorded in the memory area |direct_buffer_address_| and it is
>+  // now time to send these to the consumer.
>+  // This method is called on a high-priority thread from Java. The name of
>+  // the thread is 'AudioRecordThread'.
>+  void DataIsRecorded(JNIEnv* env,
>+                      const JavaParamRef<jobject>& j_caller,
>+                      int length);
>+
>+ private:
>+  // Stores thread ID in constructor.
>+  rtc::ThreadChecker thread_checker_;
>+
>+  // Stores thread ID in first call to OnDataIsRecorded() from high-priority
>+  // thread in Java. Detached during construction of this object.
>+  rtc::ThreadChecker thread_checker_java_;
>+
>+  // Wraps the Java specific parts of the AudioRecordJni class.
>+  JNIEnv* env_ = nullptr;
>+  ScopedJavaGlobalRef<jobject> j_audio_record_;
>+
>+  const AudioParameters audio_parameters_;
>+
>+  // Delay estimate of the total round-trip delay (input + output).
>+  // Fixed value set once in AttachAudioBuffer() and it can take one out of two
>+  // possible values. See audio_common.h for details.
>+  const int total_delay_ms_;
>+
>+  // Cached copy of address to direct audio buffer owned by |j_audio_record_|.
>+  void* direct_buffer_address_;
>+
>+  // Number of bytes in the direct audio buffer owned by |j_audio_record_|.
>+  size_t direct_buffer_capacity_in_bytes_;
>+
>+  // Number audio frames per audio buffer. Each audio frame corresponds to
>+  // one sample of PCM mono data at 16 bits per sample. Hence, each audio
>+  // frame contains 2 bytes (given that the Java layer only supports mono).
>+  // Example: 480 for 48000 Hz or 441 for 44100 Hz.
>+  size_t frames_per_buffer_;
>+
>+  bool initialized_;
>+
>+  bool recording_;
>+
>+  // Raw pointer handle provided to us in AttachAudioBuffer(). Owned by the
>+  // AudioDeviceModuleImpl class and called by AudioDeviceModule::Create().
>+  AudioDeviceBuffer* audio_device_buffer_;
>+};
>+
>+}  // namespace jni
>+
>+}  // namespace webrtc
>+
>+#endif  // SDK_ANDROID_SRC_JNI_AUDIO_DEVICE_AUDIO_RECORD_JNI_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/audio_track_jni.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/audio_track_jni.cc
>new file mode 100644
>index 00000000000..f48673f0480
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/audio_track_jni.cc
>@@ -0,0 +1,228 @@
>+/*
>+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "sdk/android/src/jni/audio_device/audio_track_jni.h"
>+
>+#include <utility>
>+
>+#include "rtc_base/arraysize.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/format_macros.h"
>+#include "rtc_base/logging.h"
>+#include "rtc_base/platform_thread.h"
>+#include "sdk/android/generated_java_audio_device_module_native_jni/jni/WebRtcAudioTrack_jni.h"
>+#include "sdk/android/src/jni/jni_helpers.h"
>+
>+namespace webrtc {
>+
>+namespace jni {
>+
>+ScopedJavaLocalRef<jobject> AudioTrackJni::CreateJavaWebRtcAudioTrack(
>+    JNIEnv* env,
>+    const JavaRef<jobject>& j_context,
>+    const JavaRef<jobject>& j_audio_manager) {
>+  return Java_WebRtcAudioTrack_Constructor(env, j_context, j_audio_manager);
>+}
>+
>+AudioTrackJni::AudioTrackJni(JNIEnv* env,
>+                             const AudioParameters& audio_parameters,
>+                             const JavaRef<jobject>& j_webrtc_audio_track)
>+    : j_audio_track_(env, j_webrtc_audio_track),
>+      audio_parameters_(audio_parameters),
>+      direct_buffer_address_(nullptr),
>+      direct_buffer_capacity_in_bytes_(0),
>+      frames_per_buffer_(0),
>+      initialized_(false),
>+      playing_(false),
>+      audio_device_buffer_(nullptr) {
>+  RTC_LOG(INFO) << "ctor";
>+  RTC_DCHECK(audio_parameters_.is_valid());
>+  Java_WebRtcAudioTrack_setNativeAudioTrack(env, j_audio_track_,
>+                                            jni::jlongFromPointer(this));
>+  // Detach from this thread since construction is allowed to happen on a
>+  // different thread.
>+  thread_checker_.DetachFromThread();
>+  thread_checker_java_.DetachFromThread();
>+}
>+
>+AudioTrackJni::~AudioTrackJni() {
>+  RTC_LOG(INFO) << "dtor";
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  Terminate();
>+}
>+
>+int32_t AudioTrackJni::Init() {
>+  RTC_LOG(INFO) << "Init";
>+  env_ = AttachCurrentThreadIfNeeded();
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  return 0;
>+}
>+
>+int32_t AudioTrackJni::Terminate() {
>+  RTC_LOG(INFO) << "Terminate";
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  StopPlayout();
>+  thread_checker_.DetachFromThread();
>+  return 0;
>+}
>+
>+int32_t AudioTrackJni::InitPlayout() {
>+  RTC_LOG(INFO) << "InitPlayout";
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  if (initialized_) {
>+    // Already initialized.
>+    return 0;
>+  }
>+  RTC_DCHECK(!playing_);
>+  if (!Java_WebRtcAudioTrack_initPlayout(
>+          env_, j_audio_track_, audio_parameters_.sample_rate(),
>+          static_cast<int>(audio_parameters_.channels()))) {
>+    RTC_LOG(LS_ERROR) << "InitPlayout failed";
>+    return -1;
>+  }
>+  initialized_ = true;
>+  return 0;
>+}
>+
>+bool AudioTrackJni::PlayoutIsInitialized() const {
>+  return initialized_;
>+}
>+
>+int32_t AudioTrackJni::StartPlayout() {
>+  RTC_LOG(INFO) << "StartPlayout";
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  if (playing_) {
>+    // Already playing.
>+    return 0;
>+  }
>+  if (!initialized_) {
>+    RTC_DLOG(LS_WARNING)
>+        << "Playout can not start since InitPlayout must succeed first";
>+    return 0;
>+  }
>+  if (!Java_WebRtcAudioTrack_startPlayout(env_, j_audio_track_)) {
>+    RTC_LOG(LS_ERROR) << "StartPlayout failed";
>+    return -1;
>+  }
>+  playing_ = true;
>+  return 0;
>+}
>+
>+int32_t AudioTrackJni::StopPlayout() {
>+  RTC_LOG(INFO) << "StopPlayout";
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  if (!initialized_ || !playing_) {
>+    return 0;
>+  }
>+  if (!Java_WebRtcAudioTrack_stopPlayout(env_, j_audio_track_)) {
>+    RTC_LOG(LS_ERROR) << "StopPlayout failed";
>+    return -1;
>+  }
>+  // If we don't detach here, we will hit a RTC_DCHECK next time StartPlayout()
>+  // is called since it will create a new Java thread.
>+  thread_checker_java_.DetachFromThread();
>+  initialized_ = false;
>+  playing_ = false;
>+  direct_buffer_address_ = nullptr;
>+  return 0;
>+}
>+
>+bool AudioTrackJni::Playing() const {
>+  return playing_;
>+}
>+
>+bool AudioTrackJni::SpeakerVolumeIsAvailable() {
>+  return true;
>+}
>+
>+int AudioTrackJni::SetSpeakerVolume(uint32_t volume) {
>+  RTC_LOG(INFO) << "SetSpeakerVolume(" << volume << ")";
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  return Java_WebRtcAudioTrack_setStreamVolume(env_, j_audio_track_,
>+                                               static_cast<int>(volume))
>+             ? 0
>+             : -1;
>+}
>+
>+absl::optional<uint32_t> AudioTrackJni::MaxSpeakerVolume() const {
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  return Java_WebRtcAudioTrack_getStreamMaxVolume(env_, j_audio_track_);
>+}
>+
>+absl::optional<uint32_t> AudioTrackJni::MinSpeakerVolume() const {
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  return 0;
>+}
>+
>+absl::optional<uint32_t> AudioTrackJni::SpeakerVolume() const {
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  const uint32_t volume =
>+      Java_WebRtcAudioTrack_getStreamVolume(env_, j_audio_track_);
>+  RTC_LOG(INFO) << "SpeakerVolume: " << volume;
>+  return volume;
>+}
>+
>+// TODO(henrika): possibly add stereo support.
>+void AudioTrackJni::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
>+  RTC_LOG(INFO) << "AttachAudioBuffer";
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  audio_device_buffer_ = audioBuffer;
>+  const int sample_rate_hz = audio_parameters_.sample_rate();
>+  RTC_LOG(INFO) << "SetPlayoutSampleRate(" << sample_rate_hz << ")";
>+  audio_device_buffer_->SetPlayoutSampleRate(sample_rate_hz);
>+  const size_t channels = audio_parameters_.channels();
>+  RTC_LOG(INFO) << "SetPlayoutChannels(" << channels << ")";
>+  audio_device_buffer_->SetPlayoutChannels(channels);
>+}
>+
>+void AudioTrackJni::CacheDirectBufferAddress(
>+    JNIEnv* env,
>+    const JavaParamRef<jobject>&,
>+    const JavaParamRef<jobject>& byte_buffer) {
>+  RTC_LOG(INFO) << "OnCacheDirectBufferAddress";
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  RTC_DCHECK(!direct_buffer_address_);
>+  direct_buffer_address_ = env->GetDirectBufferAddress(byte_buffer.obj());
>+  jlong capacity = env->GetDirectBufferCapacity(byte_buffer.obj());
>+  RTC_LOG(INFO) << "direct buffer capacity: " << capacity;
>+  direct_buffer_capacity_in_bytes_ = static_cast<size_t>(capacity);
>+  const size_t bytes_per_frame = audio_parameters_.channels() * sizeof(int16_t);
>+  frames_per_buffer_ = direct_buffer_capacity_in_bytes_ / bytes_per_frame;
>+  RTC_LOG(INFO) << "frames_per_buffer: " << frames_per_buffer_;
>+}
>+
>+// This method is called on a high-priority thread from Java. The name of
>+// the thread is 'AudioRecordTrack'.
>+void AudioTrackJni::GetPlayoutData(JNIEnv* env,
>+                                   const JavaParamRef<jobject>&,
>+                                   size_t length) {
>+  RTC_DCHECK(thread_checker_java_.CalledOnValidThread());
>+  const size_t bytes_per_frame = audio_parameters_.channels() * sizeof(int16_t);
>+  RTC_DCHECK_EQ(frames_per_buffer_, length / bytes_per_frame);
>+  if (!audio_device_buffer_) {
>+    RTC_LOG(LS_ERROR) << "AttachAudioBuffer has not been called";
>+    return;
>+  }
>+  // Pull decoded data (in 16-bit PCM format) from jitter buffer.
>+  int samples = audio_device_buffer_->RequestPlayoutData(frames_per_buffer_);
>+  if (samples <= 0) {
>+    RTC_LOG(LS_ERROR) << "AudioDeviceBuffer::RequestPlayoutData failed";
>+    return;
>+  }
>+  RTC_DCHECK_EQ(samples, frames_per_buffer_);
>+  // Copy decoded data into common byte buffer to ensure that it can be
>+  // written to the Java based audio track.
>+  samples = audio_device_buffer_->GetPlayoutData(direct_buffer_address_);
>+  RTC_DCHECK_EQ(length, bytes_per_frame * samples);
>+}
>+
>+}  // namespace jni
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/audio_track_jni.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/audio_track_jni.h
>new file mode 100644
>index 00000000000..25c6b6f9ff6
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/audio_track_jni.h
>@@ -0,0 +1,130 @@
>+/*
>+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef SDK_ANDROID_SRC_JNI_AUDIO_DEVICE_AUDIO_TRACK_JNI_H_
>+#define SDK_ANDROID_SRC_JNI_AUDIO_DEVICE_AUDIO_TRACK_JNI_H_
>+
>+#include <jni.h>
>+#include <memory>
>+
>+#include "absl/types/optional.h"
>+#include "modules/audio_device/audio_device_buffer.h"
>+#include "modules/audio_device/include/audio_device_defines.h"
>+#include "rtc_base/thread_checker.h"
>+#include "sdk/android/src/jni/audio_device/audio_common.h"
>+#include "sdk/android/src/jni/audio_device/audio_device_module.h"
>+
>+namespace webrtc {
>+
>+namespace jni {
>+
>+// Implements 16-bit mono PCM audio output support for Android using the Java
>+// AudioTrack interface. Most of the work is done by its Java counterpart in
>+// WebRtcAudioTrack.java. This class is created and lives on a thread in
>+// C++-land, but decoded audio buffers are requested on a high-priority
>+// thread managed by the Java class.
>+//
>+// An instance can be created on any thread, but must then be used on one and
>+// the same thread. All public methods must also be called on the same thread. A
>+// thread checker will RTC_DCHECK if any method is called on an invalid thread
>+//
>+// This class uses AttachCurrentThreadIfNeeded to attach to a Java VM if needed.
>+// Additional thread checking guarantees that no other (possibly non attached)
>+// thread is used.
>+class AudioTrackJni : public AudioOutput {
>+ public:
>+  static ScopedJavaLocalRef<jobject> CreateJavaWebRtcAudioTrack(
>+      JNIEnv* env,
>+      const JavaRef<jobject>& j_context,
>+      const JavaRef<jobject>& j_audio_manager);
>+
>+  AudioTrackJni(JNIEnv* env,
>+                const AudioParameters& audio_parameters,
>+                const JavaRef<jobject>& j_webrtc_audio_track);
>+  ~AudioTrackJni() override;
>+
>+  int32_t Init() override;
>+  int32_t Terminate() override;
>+
>+  int32_t InitPlayout() override;
>+  bool PlayoutIsInitialized() const override;
>+
>+  int32_t StartPlayout() override;
>+  int32_t StopPlayout() override;
>+  bool Playing() const override;
>+
>+  bool SpeakerVolumeIsAvailable() override;
>+  int SetSpeakerVolume(uint32_t volume) override;
>+  absl::optional<uint32_t> SpeakerVolume() const override;
>+  absl::optional<uint32_t> MaxSpeakerVolume() const override;
>+  absl::optional<uint32_t> MinSpeakerVolume() const override;
>+
>+  void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override;
>+
>+  // Called from Java side so we can cache the address of the Java-manged
>+  // |byte_buffer| in |direct_buffer_address_|. The size of the buffer
>+  // is also stored in |direct_buffer_capacity_in_bytes_|.
>+  // Called on the same thread as the creating thread.
>+  void CacheDirectBufferAddress(JNIEnv* env,
>+                                const JavaParamRef<jobject>& j_caller,
>+                                const JavaParamRef<jobject>& byte_buffer);
>+  // Called periodically by the Java based WebRtcAudioTrack object when
>+  // playout has started. Each call indicates that |length| new bytes should
>+  // be written to the memory area |direct_buffer_address_| for playout.
>+  // This method is called on a high-priority thread from Java. The name of
>+  // the thread is 'AudioTrackThread'.
>+  void GetPlayoutData(JNIEnv* env,
>+                      const JavaParamRef<jobject>& j_caller,
>+                      size_t length);
>+
>+ private:
>+  // Stores thread ID in constructor.
>+  rtc::ThreadChecker thread_checker_;
>+
>+  // Stores thread ID in first call to OnGetPlayoutData() from high-priority
>+  // thread in Java. Detached during construction of this object.
>+  rtc::ThreadChecker thread_checker_java_;
>+
>+  // Wraps the Java specific parts of the AudioTrackJni class.
>+  JNIEnv* env_ = nullptr;
>+  ScopedJavaGlobalRef<jobject> j_audio_track_;
>+
>+  // Contains audio parameters provided to this class at construction by the
>+  // AudioManager.
>+  const AudioParameters audio_parameters_;
>+
>+  // Cached copy of address to direct audio buffer owned by |j_audio_track_|.
>+  void* direct_buffer_address_;
>+
>+  // Number of bytes in the direct audio buffer owned by |j_audio_track_|.
>+  size_t direct_buffer_capacity_in_bytes_;
>+
>+  // Number of audio frames per audio buffer. Each audio frame corresponds to
>+  // one sample of PCM mono data at 16 bits per sample. Hence, each audio
>+  // frame contains 2 bytes (given that the Java layer only supports mono).
>+  // Example: 480 for 48000 Hz or 441 for 44100 Hz.
>+  size_t frames_per_buffer_;
>+
>+  bool initialized_;
>+
>+  bool playing_;
>+
>+  // Raw pointer handle provided to us in AttachAudioBuffer(). Owned by the
>+  // AudioDeviceModuleImpl class and called by AudioDeviceModule::Create().
>+  // The AudioDeviceBuffer is a member of the AudioDeviceModuleImpl instance
>+  // and therefore outlives this object.
>+  AudioDeviceBuffer* audio_device_buffer_;
>+};
>+
>+}  // namespace jni
>+
>+}  // namespace webrtc
>+
>+#endif  // SDK_ANDROID_SRC_JNI_AUDIO_DEVICE_AUDIO_TRACK_JNI_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/java_audio_device_module.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/java_audio_device_module.cc
>new file mode 100644
>index 00000000000..e4b235220e1
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/java_audio_device_module.cc
>@@ -0,0 +1,48 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "sdk/android/generated_java_audio_jni/jni/JavaAudioDeviceModule_jni.h"
>+#include "sdk/android/src/jni/audio_device/audio_record_jni.h"
>+#include "sdk/android/src/jni/audio_device/audio_track_jni.h"
>+#include "sdk/android/src/jni/jni_helpers.h"
>+
>+namespace webrtc {
>+namespace jni {
>+
>+static jlong JNI_JavaAudioDeviceModule_CreateAudioDeviceModule(
>+    JNIEnv* env,
>+    const JavaParamRef<jclass>& j_caller,
>+    const JavaParamRef<jobject>& j_context,
>+    const JavaParamRef<jobject>& j_audio_manager,
>+    const JavaParamRef<jobject>& j_webrtc_audio_record,
>+    const JavaParamRef<jobject>& j_webrtc_audio_track,
>+    int sample_rate,
>+    jboolean j_use_stereo_input,
>+    jboolean j_use_stereo_output) {
>+  AudioParameters input_parameters;
>+  AudioParameters output_parameters;
>+  GetAudioParameters(env, j_context, j_audio_manager, sample_rate,
>+                     j_use_stereo_input, j_use_stereo_output, &input_parameters,
>+                     &output_parameters);
>+  auto audio_input = absl::make_unique<AudioRecordJni>(
>+      env, input_parameters, kHighLatencyModeDelayEstimateInMilliseconds,
>+      j_webrtc_audio_record);
>+  auto audio_output = absl::make_unique<AudioTrackJni>(env, output_parameters,
>+                                                       j_webrtc_audio_track);
>+  return jlongFromPointer(CreateAudioDeviceModuleFromInputAndOutput(
>+                              AudioDeviceModule::kAndroidJavaAudio,
>+                              j_use_stereo_input, j_use_stereo_output,
>+                              kHighLatencyModeDelayEstimateInMilliseconds,
>+                              std::move(audio_input), std::move(audio_output))
>+                              .release());
>+}
>+
>+}  // namespace jni
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/opensles_common.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/opensles_common.cc
>new file mode 100644
>index 00000000000..ba0713116c5
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/opensles_common.cc
>@@ -0,0 +1,145 @@
>+/*
>+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "sdk/android/src/jni/audio_device/opensles_common.h"
>+
>+#include <SLES/OpenSLES.h>
>+
>+#include "rtc_base/arraysize.h"
>+#include "rtc_base/checks.h"
>+
>+namespace webrtc {
>+
>+namespace jni {
>+
>+// Returns a string representation given an integer SL_RESULT_XXX code.
>+// The mapping can be found in <SLES/OpenSLES.h>.
>+const char* GetSLErrorString(size_t code) {
>+  static const char* sl_error_strings[] = {
>+      "SL_RESULT_SUCCESS",                 // 0
>+      "SL_RESULT_PRECONDITIONS_VIOLATED",  // 1
>+      "SL_RESULT_PARAMETER_INVALID",       // 2
>+      "SL_RESULT_MEMORY_FAILURE",          // 3
>+      "SL_RESULT_RESOURCE_ERROR",          // 4
>+      "SL_RESULT_RESOURCE_LOST",           // 5
>+      "SL_RESULT_IO_ERROR",                // 6
>+      "SL_RESULT_BUFFER_INSUFFICIENT",     // 7
>+      "SL_RESULT_CONTENT_CORRUPTED",       // 8
>+      "SL_RESULT_CONTENT_UNSUPPORTED",     // 9
>+      "SL_RESULT_CONTENT_NOT_FOUND",       // 10
>+      "SL_RESULT_PERMISSION_DENIED",       // 11
>+      "SL_RESULT_FEATURE_UNSUPPORTED",     // 12
>+      "SL_RESULT_INTERNAL_ERROR",          // 13
>+      "SL_RESULT_UNKNOWN_ERROR",           // 14
>+      "SL_RESULT_OPERATION_ABORTED",       // 15
>+      "SL_RESULT_CONTROL_LOST",            // 16
>+  };
>+
>+  if (code >= arraysize(sl_error_strings)) {
>+    return "SL_RESULT_UNKNOWN_ERROR";
>+  }
>+  return sl_error_strings[code];
>+}
>+
>+SLDataFormat_PCM CreatePCMConfiguration(size_t channels,
>+                                        int sample_rate,
>+                                        size_t bits_per_sample) {
>+  RTC_CHECK_EQ(bits_per_sample, SL_PCMSAMPLEFORMAT_FIXED_16);
>+  SLDataFormat_PCM format;
>+  format.formatType = SL_DATAFORMAT_PCM;
>+  format.numChannels = static_cast<SLuint32>(channels);
>+  // Note that, the unit of sample rate is actually in milliHertz and not Hertz.
>+  switch (sample_rate) {
>+    case 8000:
>+      format.samplesPerSec = SL_SAMPLINGRATE_8;
>+      break;
>+    case 16000:
>+      format.samplesPerSec = SL_SAMPLINGRATE_16;
>+      break;
>+    case 22050:
>+      format.samplesPerSec = SL_SAMPLINGRATE_22_05;
>+      break;
>+    case 32000:
>+      format.samplesPerSec = SL_SAMPLINGRATE_32;
>+      break;
>+    case 44100:
>+      format.samplesPerSec = SL_SAMPLINGRATE_44_1;
>+      break;
>+    case 48000:
>+      format.samplesPerSec = SL_SAMPLINGRATE_48;
>+      break;
>+    case 64000:
>+      format.samplesPerSec = SL_SAMPLINGRATE_64;
>+      break;
>+    case 88200:
>+      format.samplesPerSec = SL_SAMPLINGRATE_88_2;
>+      break;
>+    case 96000:
>+      format.samplesPerSec = SL_SAMPLINGRATE_96;
>+      break;
>+    default:
>+      RTC_CHECK(false) << "Unsupported sample rate: " << sample_rate;
>+      break;
>+  }
>+  format.bitsPerSample = SL_PCMSAMPLEFORMAT_FIXED_16;
>+  format.containerSize = SL_PCMSAMPLEFORMAT_FIXED_16;
>+  format.endianness = SL_BYTEORDER_LITTLEENDIAN;
>+  if (format.numChannels == 1) {
>+    format.channelMask = SL_SPEAKER_FRONT_CENTER;
>+  } else if (format.numChannels == 2) {
>+    format.channelMask = SL_SPEAKER_FRONT_LEFT | SL_SPEAKER_FRONT_RIGHT;
>+  } else {
>+    RTC_CHECK(false) << "Unsupported number of channels: "
>+                     << format.numChannels;
>+  }
>+  return format;
>+}
>+
>+OpenSLEngineManager::OpenSLEngineManager() {
>+  thread_checker_.DetachFromThread();
>+}
>+
>+OpenSLEngineManager::~OpenSLEngineManager() = default;
>+
>+SLObjectItf OpenSLEngineManager::GetOpenSLEngine() {
>+  RTC_LOG(INFO) << "GetOpenSLEngine";
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  // OpenSL ES for Android only supports a single engine per application.
>+  // If one already has been created, return existing object instead of
>+  // creating a new.
>+  if (engine_object_.Get() != nullptr) {
>+    RTC_LOG(WARNING) << "The OpenSL ES engine object has already been created";
>+    return engine_object_.Get();
>+  }
>+  // Create the engine object in thread safe mode.
>+  const SLEngineOption option[] = {
>+      {SL_ENGINEOPTION_THREADSAFE, static_cast<SLuint32>(SL_BOOLEAN_TRUE)}};
>+  SLresult result =
>+      slCreateEngine(engine_object_.Receive(), 1, option, 0, NULL, NULL);
>+  if (result != SL_RESULT_SUCCESS) {
>+    RTC_LOG(LS_ERROR) << "slCreateEngine() failed: "
>+                      << GetSLErrorString(result);
>+    engine_object_.Reset();
>+    return nullptr;
>+  }
>+  // Realize the SL Engine in synchronous mode.
>+  result = engine_object_->Realize(engine_object_.Get(), SL_BOOLEAN_FALSE);
>+  if (result != SL_RESULT_SUCCESS) {
>+    RTC_LOG(LS_ERROR) << "Realize() failed: " << GetSLErrorString(result);
>+    engine_object_.Reset();
>+    return nullptr;
>+  }
>+  // Finally return the SLObjectItf interface of the engine object.
>+  return engine_object_.Get();
>+}
>+
>+}  // namespace jni
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/opensles_common.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/opensles_common.h
>new file mode 100644
>index 00000000000..0a8c115accb
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/opensles_common.h
>@@ -0,0 +1,90 @@
>+/*
>+ *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef SDK_ANDROID_SRC_JNI_AUDIO_DEVICE_OPENSLES_COMMON_H_
>+#define SDK_ANDROID_SRC_JNI_AUDIO_DEVICE_OPENSLES_COMMON_H_
>+
>+#include <SLES/OpenSLES.h>
>+#include <stddef.h>
>+
>+#include "rtc_base/checks.h"
>+#include "rtc_base/logging.h"
>+#include "rtc_base/thread_checker.h"
>+
>+namespace webrtc {
>+
>+namespace jni {
>+
>+// Returns a string representation given an integer SL_RESULT_XXX code.
>+// The mapping can be found in <SLES/OpenSLES.h>.
>+const char* GetSLErrorString(size_t code);
>+
>+// Configures an SL_DATAFORMAT_PCM structure based on native audio parameters.
>+SLDataFormat_PCM CreatePCMConfiguration(size_t channels,
>+                                        int sample_rate,
>+                                        size_t bits_per_sample);
>+
>+// Helper class for using SLObjectItf interfaces.
>+template <typename SLType, typename SLDerefType>
>+class ScopedSLObject {
>+ public:
>+  ScopedSLObject() : obj_(nullptr) {}
>+
>+  ~ScopedSLObject() { Reset(); }
>+
>+  SLType* Receive() {
>+    RTC_DCHECK(!obj_);
>+    return &obj_;
>+  }
>+
>+  SLDerefType operator->() { return *obj_; }
>+
>+  SLType Get() const { return obj_; }
>+
>+  void Reset() {
>+    if (obj_) {
>+      (*obj_)->Destroy(obj_);
>+      obj_ = nullptr;
>+    }
>+  }
>+
>+ private:
>+  SLType obj_;
>+};
>+
>+typedef ScopedSLObject<SLObjectItf, const SLObjectItf_*> ScopedSLObjectItf;
>+
>+// Creates and realizes the main (global) Open SL engine object and returns
>+// a reference to it. The engine object is only created at the first call
>+// since OpenSL ES for Android only supports a single engine per application.
>+// Subsequent calls returns the already created engine.
>+// Note: This class must be used single threaded and this is enfored by a thread
>+// checker.
>+class OpenSLEngineManager {
>+ public:
>+  OpenSLEngineManager();
>+  ~OpenSLEngineManager();
>+  SLObjectItf GetOpenSLEngine();
>+
>+ private:
>+  rtc::ThreadChecker thread_checker_;
>+  // This object is the global entry point of the OpenSL ES API.
>+  // After creating the engine object, the application can obtain this objectâs
>+  // SLEngineItf interface. This interface contains creation methods for all
>+  // the other object types in the API. None of these interface are realized
>+  // by this class. It only provides access to the global engine object.
>+  ScopedSLObjectItf engine_object_;
>+};
>+
>+}  // namespace jni
>+
>+}  // namespace webrtc
>+
>+#endif  // SDK_ANDROID_SRC_JNI_AUDIO_DEVICE_OPENSLES_COMMON_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/opensles_player.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/opensles_player.cc
>new file mode 100644
>index 00000000000..00a5f1ecc8e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/opensles_player.cc
>@@ -0,0 +1,446 @@
>+/*
>+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "sdk/android/src/jni/audio_device/opensles_player.h"
>+
>+#include <android/log.h>
>+
>+#include "absl/memory/memory.h"
>+#include "api/array_view.h"
>+#include "modules/audio_device/fine_audio_buffer.h"
>+#include "rtc_base/arraysize.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/format_macros.h"
>+#include "rtc_base/platform_thread.h"
>+#include "rtc_base/timeutils.h"
>+#include "sdk/android/src/jni/audio_device/audio_common.h"
>+
>+#define TAG "OpenSLESPlayer"
>+#define ALOGV(...) __android_log_print(ANDROID_LOG_VERBOSE, TAG, __VA_ARGS__)
>+#define ALOGD(...) __android_log_print(ANDROID_LOG_DEBUG, TAG, __VA_ARGS__)
>+#define ALOGE(...) __android_log_print(ANDROID_LOG_ERROR, TAG, __VA_ARGS__)
>+#define ALOGW(...) __android_log_print(ANDROID_LOG_WARN, TAG, __VA_ARGS__)
>+#define ALOGI(...) __android_log_print(ANDROID_LOG_INFO, TAG, __VA_ARGS__)
>+
>+#define RETURN_ON_ERROR(op, ...)                          \
>+  do {                                                    \
>+    SLresult err = (op);                                  \
>+    if (err != SL_RESULT_SUCCESS) {                       \
>+      ALOGE("%s failed: %s", #op, GetSLErrorString(err)); \
>+      return __VA_ARGS__;                                 \
>+    }                                                     \
>+  } while (0)
>+
>+namespace webrtc {
>+
>+namespace jni {
>+
>+OpenSLESPlayer::OpenSLESPlayer(
>+    const AudioParameters& audio_parameters,
>+    std::unique_ptr<OpenSLEngineManager> engine_manager)
>+    : audio_parameters_(audio_parameters),
>+      audio_device_buffer_(nullptr),
>+      initialized_(false),
>+      playing_(false),
>+      buffer_index_(0),
>+      engine_manager_(std::move(engine_manager)),
>+      engine_(nullptr),
>+      player_(nullptr),
>+      simple_buffer_queue_(nullptr),
>+      volume_(nullptr),
>+      last_play_time_(0) {
>+  ALOGD("ctor[tid=%d]", rtc::CurrentThreadId());
>+  // Use native audio output parameters provided by the audio manager and
>+  // define the PCM format structure.
>+  pcm_format_ = CreatePCMConfiguration(audio_parameters_.channels(),
>+                                       audio_parameters_.sample_rate(),
>+                                       audio_parameters_.bits_per_sample());
>+  // Detach from this thread since we want to use the checker to verify calls
>+  // from the internal  audio thread.
>+  thread_checker_opensles_.DetachFromThread();
>+}
>+
>+OpenSLESPlayer::~OpenSLESPlayer() {
>+  ALOGD("dtor[tid=%d]", rtc::CurrentThreadId());
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  Terminate();
>+  DestroyAudioPlayer();
>+  DestroyMix();
>+  engine_ = nullptr;
>+  RTC_DCHECK(!engine_);
>+  RTC_DCHECK(!output_mix_.Get());
>+  RTC_DCHECK(!player_);
>+  RTC_DCHECK(!simple_buffer_queue_);
>+  RTC_DCHECK(!volume_);
>+}
>+
>+int OpenSLESPlayer::Init() {
>+  ALOGD("Init[tid=%d]", rtc::CurrentThreadId());
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  if (audio_parameters_.channels() == 2) {
>+    ALOGW("Stereo mode is enabled");
>+  }
>+  return 0;
>+}
>+
>+int OpenSLESPlayer::Terminate() {
>+  ALOGD("Terminate[tid=%d]", rtc::CurrentThreadId());
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  StopPlayout();
>+  return 0;
>+}
>+
>+int OpenSLESPlayer::InitPlayout() {
>+  ALOGD("InitPlayout[tid=%d]", rtc::CurrentThreadId());
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  RTC_DCHECK(!initialized_);
>+  RTC_DCHECK(!playing_);
>+  if (!ObtainEngineInterface()) {
>+    ALOGE("Failed to obtain SL Engine interface");
>+    return -1;
>+  }
>+  CreateMix();
>+  initialized_ = true;
>+  buffer_index_ = 0;
>+  return 0;
>+}
>+
>+bool OpenSLESPlayer::PlayoutIsInitialized() const {
>+  return initialized_;
>+}
>+
>+int OpenSLESPlayer::StartPlayout() {
>+  ALOGD("StartPlayout[tid=%d]", rtc::CurrentThreadId());
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  RTC_DCHECK(initialized_);
>+  RTC_DCHECK(!playing_);
>+  if (fine_audio_buffer_) {
>+    fine_audio_buffer_->ResetPlayout();
>+  }
>+  // The number of lower latency audio players is limited, hence we create the
>+  // audio player in Start() and destroy it in Stop().
>+  CreateAudioPlayer();
>+  // Fill up audio buffers to avoid initial glitch and to ensure that playback
>+  // starts when mode is later changed to SL_PLAYSTATE_PLAYING.
>+  // TODO(henrika): we can save some delay by only making one call to
>+  // EnqueuePlayoutData. Most likely not worth the risk of adding a glitch.
>+  last_play_time_ = rtc::Time();
>+  for (int i = 0; i < kNumOfOpenSLESBuffers; ++i) {
>+    EnqueuePlayoutData(true);
>+  }
>+  // Start streaming data by setting the play state to SL_PLAYSTATE_PLAYING.
>+  // For a player object, when the object is in the SL_PLAYSTATE_PLAYING
>+  // state, adding buffers will implicitly start playback.
>+  RETURN_ON_ERROR((*player_)->SetPlayState(player_, SL_PLAYSTATE_PLAYING), -1);
>+  playing_ = (GetPlayState() == SL_PLAYSTATE_PLAYING);
>+  RTC_DCHECK(playing_);
>+  return 0;
>+}
>+
>+int OpenSLESPlayer::StopPlayout() {
>+  ALOGD("StopPlayout[tid=%d]", rtc::CurrentThreadId());
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  if (!initialized_ || !playing_) {
>+    return 0;
>+  }
>+  // Stop playing by setting the play state to SL_PLAYSTATE_STOPPED.
>+  RETURN_ON_ERROR((*player_)->SetPlayState(player_, SL_PLAYSTATE_STOPPED), -1);
>+  // Clear the buffer queue to flush out any remaining data.
>+  RETURN_ON_ERROR((*simple_buffer_queue_)->Clear(simple_buffer_queue_), -1);
>+#if RTC_DCHECK_IS_ON
>+  // Verify that the buffer queue is in fact cleared as it should.
>+  SLAndroidSimpleBufferQueueState buffer_queue_state;
>+  (*simple_buffer_queue_)->GetState(simple_buffer_queue_, &buffer_queue_state);
>+  RTC_DCHECK_EQ(0, buffer_queue_state.count);
>+  RTC_DCHECK_EQ(0, buffer_queue_state.index);
>+#endif
>+  // The number of lower latency audio players is limited, hence we create the
>+  // audio player in Start() and destroy it in Stop().
>+  DestroyAudioPlayer();
>+  thread_checker_opensles_.DetachFromThread();
>+  initialized_ = false;
>+  playing_ = false;
>+  return 0;
>+}
>+
>+bool OpenSLESPlayer::Playing() const {
>+  return playing_;
>+}
>+
>+bool OpenSLESPlayer::SpeakerVolumeIsAvailable() {
>+  return false;
>+}
>+
>+int OpenSLESPlayer::SetSpeakerVolume(uint32_t volume) {
>+  return -1;
>+}
>+
>+absl::optional<uint32_t> OpenSLESPlayer::SpeakerVolume() const {
>+  return absl::nullopt;
>+}
>+
>+absl::optional<uint32_t> OpenSLESPlayer::MaxSpeakerVolume() const {
>+  return absl::nullopt;
>+}
>+
>+absl::optional<uint32_t> OpenSLESPlayer::MinSpeakerVolume() const {
>+  return absl::nullopt;
>+}
>+
>+void OpenSLESPlayer::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
>+  ALOGD("AttachAudioBuffer");
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  audio_device_buffer_ = audioBuffer;
>+  const int sample_rate_hz = audio_parameters_.sample_rate();
>+  ALOGD("SetPlayoutSampleRate(%d)", sample_rate_hz);
>+  audio_device_buffer_->SetPlayoutSampleRate(sample_rate_hz);
>+  const size_t channels = audio_parameters_.channels();
>+  ALOGD("SetPlayoutChannels(%" PRIuS ")", channels);
>+  audio_device_buffer_->SetPlayoutChannels(channels);
>+  RTC_CHECK(audio_device_buffer_);
>+  AllocateDataBuffers();
>+}
>+
>+void OpenSLESPlayer::AllocateDataBuffers() {
>+  ALOGD("AllocateDataBuffers");
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  RTC_DCHECK(!simple_buffer_queue_);
>+  RTC_CHECK(audio_device_buffer_);
>+  // Create a modified audio buffer class which allows us to ask for any number
>+  // of samples (and not only multiple of 10ms) to match the native OpenSL ES
>+  // buffer size. The native buffer size corresponds to the
>+  // PROPERTY_OUTPUT_FRAMES_PER_BUFFER property which is the number of audio
>+  // frames that the HAL (Hardware Abstraction Layer) buffer can hold. It is
>+  // recommended to construct audio buffers so that they contain an exact
>+  // multiple of this number. If so, callbacks will occur at regular intervals,
>+  // which reduces jitter.
>+  const size_t buffer_size_in_samples =
>+      audio_parameters_.frames_per_buffer() * audio_parameters_.channels();
>+  ALOGD("native buffer size: %" PRIuS, buffer_size_in_samples);
>+  ALOGD("native buffer size in ms: %.2f",
>+        audio_parameters_.GetBufferSizeInMilliseconds());
>+  fine_audio_buffer_ = absl::make_unique<FineAudioBuffer>(audio_device_buffer_);
>+  // Allocated memory for audio buffers.
>+  for (int i = 0; i < kNumOfOpenSLESBuffers; ++i) {
>+    audio_buffers_[i].reset(new SLint16[buffer_size_in_samples]);
>+  }
>+}
>+
>+bool OpenSLESPlayer::ObtainEngineInterface() {
>+  ALOGD("ObtainEngineInterface");
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  if (engine_)
>+    return true;
>+  // Get access to (or create if not already existing) the global OpenSL Engine
>+  // object.
>+  SLObjectItf engine_object = engine_manager_->GetOpenSLEngine();
>+  if (engine_object == nullptr) {
>+    ALOGE("Failed to access the global OpenSL engine");
>+    return false;
>+  }
>+  // Get the SL Engine Interface which is implicit.
>+  RETURN_ON_ERROR(
>+      (*engine_object)->GetInterface(engine_object, SL_IID_ENGINE, &engine_),
>+      false);
>+  return true;
>+}
>+
>+bool OpenSLESPlayer::CreateMix() {
>+  ALOGD("CreateMix");
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  RTC_DCHECK(engine_);
>+  if (output_mix_.Get())
>+    return true;
>+
>+  // Create the ouput mix on the engine object. No interfaces will be used.
>+  RETURN_ON_ERROR((*engine_)->CreateOutputMix(engine_, output_mix_.Receive(), 0,
>+                                              nullptr, nullptr),
>+                  false);
>+  RETURN_ON_ERROR(output_mix_->Realize(output_mix_.Get(), SL_BOOLEAN_FALSE),
>+                  false);
>+  return true;
>+}
>+
>+void OpenSLESPlayer::DestroyMix() {
>+  ALOGD("DestroyMix");
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  if (!output_mix_.Get())
>+    return;
>+  output_mix_.Reset();
>+}
>+
>+bool OpenSLESPlayer::CreateAudioPlayer() {
>+  ALOGD("CreateAudioPlayer");
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  RTC_DCHECK(output_mix_.Get());
>+  if (player_object_.Get())
>+    return true;
>+  RTC_DCHECK(!player_);
>+  RTC_DCHECK(!simple_buffer_queue_);
>+  RTC_DCHECK(!volume_);
>+
>+  // source: Android Simple Buffer Queue Data Locator is source.
>+  SLDataLocator_AndroidSimpleBufferQueue simple_buffer_queue = {
>+      SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE,
>+      static_cast<SLuint32>(kNumOfOpenSLESBuffers)};
>+  SLDataSource audio_source = {&simple_buffer_queue, &pcm_format_};
>+
>+  // sink: OutputMix-based data is sink.
>+  SLDataLocator_OutputMix locator_output_mix = {SL_DATALOCATOR_OUTPUTMIX,
>+                                                output_mix_.Get()};
>+  SLDataSink audio_sink = {&locator_output_mix, nullptr};
>+
>+  // Define interfaces that we indend to use and realize.
>+  const SLInterfaceID interface_ids[] = {SL_IID_ANDROIDCONFIGURATION,
>+                                         SL_IID_BUFFERQUEUE, SL_IID_VOLUME};
>+  const SLboolean interface_required[] = {SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE,
>+                                          SL_BOOLEAN_TRUE};
>+
>+  // Create the audio player on the engine interface.
>+  RETURN_ON_ERROR(
>+      (*engine_)->CreateAudioPlayer(
>+          engine_, player_object_.Receive(), &audio_source, &audio_sink,
>+          arraysize(interface_ids), interface_ids, interface_required),
>+      false);
>+
>+  // Use the Android configuration interface to set platform-specific
>+  // parameters. Should be done before player is realized.
>+  SLAndroidConfigurationItf player_config;
>+  RETURN_ON_ERROR(
>+      player_object_->GetInterface(player_object_.Get(),
>+                                   SL_IID_ANDROIDCONFIGURATION, &player_config),
>+      false);
>+  // Set audio player configuration to SL_ANDROID_STREAM_VOICE which
>+  // corresponds to android.media.AudioManager.STREAM_VOICE_CALL.
>+  SLint32 stream_type = SL_ANDROID_STREAM_VOICE;
>+  RETURN_ON_ERROR(
>+      (*player_config)
>+          ->SetConfiguration(player_config, SL_ANDROID_KEY_STREAM_TYPE,
>+                             &stream_type, sizeof(SLint32)),
>+      false);
>+
>+  // Realize the audio player object after configuration has been set.
>+  RETURN_ON_ERROR(
>+      player_object_->Realize(player_object_.Get(), SL_BOOLEAN_FALSE), false);
>+
>+  // Get the SLPlayItf interface on the audio player.
>+  RETURN_ON_ERROR(
>+      player_object_->GetInterface(player_object_.Get(), SL_IID_PLAY, &player_),
>+      false);
>+
>+  // Get the SLAndroidSimpleBufferQueueItf interface on the audio player.
>+  RETURN_ON_ERROR(
>+      player_object_->GetInterface(player_object_.Get(), SL_IID_BUFFERQUEUE,
>+                                   &simple_buffer_queue_),
>+      false);
>+
>+  // Register callback method for the Android Simple Buffer Queue interface.
>+  // This method will be called when the native audio layer needs audio data.
>+  RETURN_ON_ERROR((*simple_buffer_queue_)
>+                      ->RegisterCallback(simple_buffer_queue_,
>+                                         SimpleBufferQueueCallback, this),
>+                  false);
>+
>+  // Get the SLVolumeItf interface on the audio player.
>+  RETURN_ON_ERROR(player_object_->GetInterface(player_object_.Get(),
>+                                               SL_IID_VOLUME, &volume_),
>+                  false);
>+
>+  // TODO(henrika): might not be required to set volume to max here since it
>+  // seems to be default on most devices. Might be required for unit tests.
>+  // RETURN_ON_ERROR((*volume_)->SetVolumeLevel(volume_, 0), false);
>+
>+  return true;
>+}
>+
>+void OpenSLESPlayer::DestroyAudioPlayer() {
>+  ALOGD("DestroyAudioPlayer");
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  if (!player_object_.Get())
>+    return;
>+  (*simple_buffer_queue_)
>+      ->RegisterCallback(simple_buffer_queue_, nullptr, nullptr);
>+  player_object_.Reset();
>+  player_ = nullptr;
>+  simple_buffer_queue_ = nullptr;
>+  volume_ = nullptr;
>+}
>+
>+// static
>+void OpenSLESPlayer::SimpleBufferQueueCallback(
>+    SLAndroidSimpleBufferQueueItf caller,
>+    void* context) {
>+  OpenSLESPlayer* stream = reinterpret_cast<OpenSLESPlayer*>(context);
>+  stream->FillBufferQueue();
>+}
>+
>+void OpenSLESPlayer::FillBufferQueue() {
>+  RTC_DCHECK(thread_checker_opensles_.CalledOnValidThread());
>+  SLuint32 state = GetPlayState();
>+  if (state != SL_PLAYSTATE_PLAYING) {
>+    ALOGW("Buffer callback in non-playing state!");
>+    return;
>+  }
>+  EnqueuePlayoutData(false);
>+}
>+
>+void OpenSLESPlayer::EnqueuePlayoutData(bool silence) {
>+  // Check delta time between two successive callbacks and provide a warning
>+  // if it becomes very large.
>+  // TODO(henrika): using 150ms as upper limit but this value is rather random.
>+  const uint32_t current_time = rtc::Time();
>+  const uint32_t diff = current_time - last_play_time_;
>+  if (diff > 150) {
>+    ALOGW("Bad OpenSL ES playout timing, dT=%u [ms]", diff);
>+  }
>+  last_play_time_ = current_time;
>+  SLint8* audio_ptr8 =
>+      reinterpret_cast<SLint8*>(audio_buffers_[buffer_index_].get());
>+  if (silence) {
>+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+    // Avoid acquiring real audio data from WebRTC and fill the buffer with
>+    // zeros instead. Used to prime the buffer with silence and to avoid asking
>+    // for audio data from two different threads.
>+    memset(audio_ptr8, 0, audio_parameters_.GetBytesPerBuffer());
>+  } else {
>+    RTC_DCHECK(thread_checker_opensles_.CalledOnValidThread());
>+    // Read audio data from the WebRTC source using the FineAudioBuffer object
>+    // to adjust for differences in buffer size between WebRTC (10ms) and native
>+    // OpenSL ES. Use hardcoded delay estimate since OpenSL ES does not support
>+    // delay estimation.
>+    fine_audio_buffer_->GetPlayoutData(
>+        rtc::ArrayView<int16_t>(audio_buffers_[buffer_index_].get(),
>+                                audio_parameters_.frames_per_buffer() *
>+                                    audio_parameters_.channels()),
>+        25);
>+  }
>+  // Enqueue the decoded audio buffer for playback.
>+  SLresult err = (*simple_buffer_queue_)
>+                     ->Enqueue(simple_buffer_queue_, audio_ptr8,
>+                               audio_parameters_.GetBytesPerBuffer());
>+  if (SL_RESULT_SUCCESS != err) {
>+    ALOGE("Enqueue failed: %d", err);
>+  }
>+  buffer_index_ = (buffer_index_ + 1) % kNumOfOpenSLESBuffers;
>+}
>+
>+SLuint32 OpenSLESPlayer::GetPlayState() const {
>+  RTC_DCHECK(player_);
>+  SLuint32 state;
>+  SLresult err = (*player_)->GetPlayState(player_, &state);
>+  if (SL_RESULT_SUCCESS != err) {
>+    ALOGE("GetPlayState failed: %d", err);
>+  }
>+  return state;
>+}
>+
>+}  // namespace jni
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/opensles_player.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/opensles_player.h
>new file mode 100644
>index 00000000000..d8befe50d07
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/opensles_player.h
>@@ -0,0 +1,195 @@
>+/*
>+ *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef SDK_ANDROID_SRC_JNI_AUDIO_DEVICE_OPENSLES_PLAYER_H_
>+#define SDK_ANDROID_SRC_JNI_AUDIO_DEVICE_OPENSLES_PLAYER_H_
>+
>+#include <SLES/OpenSLES.h>
>+#include <SLES/OpenSLES_Android.h>
>+#include <SLES/OpenSLES_AndroidConfiguration.h>
>+
>+#include <memory>
>+#include "absl/types/optional.h"
>+#include "modules/audio_device/audio_device_buffer.h"
>+#include "modules/audio_device/fine_audio_buffer.h"
>+#include "modules/audio_device/include/audio_device_defines.h"
>+#include "rtc_base/thread_checker.h"
>+#include "sdk/android/src/jni/audio_device/audio_common.h"
>+#include "sdk/android/src/jni/audio_device/audio_device_module.h"
>+#include "sdk/android/src/jni/audio_device/opensles_common.h"
>+
>+namespace webrtc {
>+
>+class FineAudioBuffer;
>+
>+namespace jni {
>+
>+// Implements 16-bit mono PCM audio output support for Android using the
>+// C based OpenSL ES API. No calls from C/C++ to Java using JNI is done.
>+//
>+// An instance can be created on any thread, but must then be used on one and
>+// the same thread. All public methods must also be called on the same thread. A
>+// thread checker will RTC_DCHECK if any method is called on an invalid thread.
>+// Decoded audio buffers are requested on a dedicated internal thread managed by
>+// the OpenSL ES layer.
>+//
>+// The existing design forces the user to call InitPlayout() after Stoplayout()
>+// to be able to call StartPlayout() again. This is inline with how the Java-
>+// based implementation works.
>+//
>+// OpenSL ES is a native C API which have no Dalvik-related overhead such as
>+// garbage collection pauses and it supports reduced audio output latency.
>+// If the device doesn't claim this feature but supports API level 9 (Android
>+// platform version 2.3) or later, then we can still use the OpenSL ES APIs but
>+// the output latency may be higher.
>+class OpenSLESPlayer : public AudioOutput {
>+ public:
>+  // Beginning with API level 17 (Android 4.2), a buffer count of 2 or more is
>+  // required for lower latency. Beginning with API level 18 (Android 4.3), a
>+  // buffer count of 1 is sufficient for lower latency. In addition, the buffer
>+  // size and sample rate must be compatible with the device's native output
>+  // configuration provided via the audio manager at construction.
>+  // TODO(henrika): perhaps set this value dynamically based on OS version.
>+  static const int kNumOfOpenSLESBuffers = 2;
>+
>+  OpenSLESPlayer(const AudioParameters& audio_parameters,
>+                 std::unique_ptr<OpenSLEngineManager> engine_manager);
>+  ~OpenSLESPlayer() override;
>+
>+  int Init() override;
>+  int Terminate() override;
>+
>+  int InitPlayout() override;
>+  bool PlayoutIsInitialized() const override;
>+
>+  int StartPlayout() override;
>+  int StopPlayout() override;
>+  bool Playing() const override;
>+
>+  bool SpeakerVolumeIsAvailable() override;
>+  int SetSpeakerVolume(uint32_t volume) override;
>+  absl::optional<uint32_t> SpeakerVolume() const override;
>+  absl::optional<uint32_t> MaxSpeakerVolume() const override;
>+  absl::optional<uint32_t> MinSpeakerVolume() const override;
>+
>+  void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override;
>+
>+ private:
>+  // These callback methods are called when data is required for playout.
>+  // They are both called from an internal "OpenSL ES thread" which is not
>+  // attached to the Dalvik VM.
>+  static void SimpleBufferQueueCallback(SLAndroidSimpleBufferQueueItf caller,
>+                                        void* context);
>+  void FillBufferQueue();
>+  // Reads audio data in PCM format using the AudioDeviceBuffer.
>+  // Can be called both on the main thread (during Start()) and from the
>+  // internal audio thread while output streaming is active.
>+  // If the |silence| flag is set, the audio is filled with zeros instead of
>+  // asking the WebRTC layer for real audio data. This procedure is also known
>+  // as audio priming.
>+  void EnqueuePlayoutData(bool silence);
>+
>+  // Allocate memory for audio buffers which will be used to render audio
>+  // via the SLAndroidSimpleBufferQueueItf interface.
>+  void AllocateDataBuffers();
>+
>+  // Obtaines the SL Engine Interface from the existing global Engine object.
>+  // The interface exposes creation methods of all the OpenSL ES object types.
>+  // This method defines the |engine_| member variable.
>+  bool ObtainEngineInterface();
>+
>+  // Creates/destroys the output mix object.
>+  bool CreateMix();
>+  void DestroyMix();
>+
>+  // Creates/destroys the audio player and the simple-buffer object.
>+  // Also creates the volume object.
>+  bool CreateAudioPlayer();
>+  void DestroyAudioPlayer();
>+
>+  SLuint32 GetPlayState() const;
>+
>+  // Ensures that methods are called from the same thread as this object is
>+  // created on.
>+  rtc::ThreadChecker thread_checker_;
>+
>+  // Stores thread ID in first call to SimpleBufferQueueCallback() from internal
>+  // non-application thread which is not attached to the Dalvik JVM.
>+  // Detached during construction of this object.
>+  rtc::ThreadChecker thread_checker_opensles_;
>+
>+  const AudioParameters audio_parameters_;
>+
>+  // Raw pointer handle provided to us in AttachAudioBuffer(). Owned by the
>+  // AudioDeviceModuleImpl class and called by AudioDeviceModule::Create().
>+  AudioDeviceBuffer* audio_device_buffer_;
>+
>+  bool initialized_;
>+  bool playing_;
>+
>+  // PCM-type format definition.
>+  // TODO(henrika): add support for SLAndroidDataFormat_PCM_EX (android-21) if
>+  // 32-bit float representation is needed.
>+  SLDataFormat_PCM pcm_format_;
>+
>+  // Queue of audio buffers to be used by the player object for rendering
>+  // audio.
>+  std::unique_ptr<SLint16[]> audio_buffers_[kNumOfOpenSLESBuffers];
>+
>+  // FineAudioBuffer takes an AudioDeviceBuffer which delivers audio data
>+  // in chunks of 10ms. It then allows for this data to be pulled in
>+  // a finer or coarser granularity. I.e. interacting with this class instead
>+  // of directly with the AudioDeviceBuffer one can ask for any number of
>+  // audio data samples.
>+  // Example: native buffer size can be 192 audio frames at 48kHz sample rate.
>+  // WebRTC will provide 480 audio frames per 10ms but OpenSL ES asks for 192
>+  // in each callback (one every 4th ms). This class can then ask for 192 and
>+  // the FineAudioBuffer will ask WebRTC for new data approximately only every
>+  // second callback and also cache non-utilized audio.
>+  std::unique_ptr<FineAudioBuffer> fine_audio_buffer_;
>+
>+  // Keeps track of active audio buffer 'n' in the audio_buffers_[n] queue.
>+  // Example (kNumOfOpenSLESBuffers = 2): counts 0, 1, 0, 1, ...
>+  int buffer_index_;
>+
>+  std::unique_ptr<OpenSLEngineManager> engine_manager_;
>+  // This interface exposes creation methods for all the OpenSL ES object types.
>+  // It is the OpenSL ES API entry point.
>+  SLEngineItf engine_;
>+
>+  // Output mix object to be used by the player object.
>+  ScopedSLObjectItf output_mix_;
>+
>+  // The audio player media object plays out audio to the speakers. It also
>+  // supports volume control.
>+  ScopedSLObjectItf player_object_;
>+
>+  // This interface is supported on the audio player and it controls the state
>+  // of the audio player.
>+  SLPlayItf player_;
>+
>+  // The Android Simple Buffer Queue interface is supported on the audio player
>+  // and it provides methods to send audio data from the source to the audio
>+  // player for rendering.
>+  SLAndroidSimpleBufferQueueItf simple_buffer_queue_;
>+
>+  // This interface exposes controls for manipulating the objectâs audio volume
>+  // properties. This interface is supported on the Audio Player object.
>+  SLVolumeItf volume_;
>+
>+  // Last time the OpenSL ES layer asked for audio data to play out.
>+  uint32_t last_play_time_;
>+};
>+
>+}  // namespace jni
>+
>+}  // namespace webrtc
>+
>+#endif  // SDK_ANDROID_SRC_JNI_AUDIO_DEVICE_OPENSLES_PLAYER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/opensles_recorder.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/opensles_recorder.cc
>new file mode 100644
>index 00000000000..32d9e5c9cfc
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/opensles_recorder.cc
>@@ -0,0 +1,446 @@
>+/*
>+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "sdk/android/src/jni/audio_device/opensles_recorder.h"
>+
>+#include <android/log.h>
>+
>+#include "absl/memory/memory.h"
>+#include "api/array_view.h"
>+#include "modules/audio_device/fine_audio_buffer.h"
>+#include "rtc_base/arraysize.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/format_macros.h"
>+#include "rtc_base/platform_thread.h"
>+#include "rtc_base/timeutils.h"
>+#include "sdk/android/src/jni/audio_device/audio_common.h"
>+
>+#define TAG "OpenSLESRecorder"
>+#define ALOGV(...) __android_log_print(ANDROID_LOG_VERBOSE, TAG, __VA_ARGS__)
>+#define ALOGD(...) __android_log_print(ANDROID_LOG_DEBUG, TAG, __VA_ARGS__)
>+#define ALOGE(...) __android_log_print(ANDROID_LOG_ERROR, TAG, __VA_ARGS__)
>+#define ALOGW(...) __android_log_print(ANDROID_LOG_WARN, TAG, __VA_ARGS__)
>+#define ALOGI(...) __android_log_print(ANDROID_LOG_INFO, TAG, __VA_ARGS__)
>+
>+#define LOG_ON_ERROR(op)                                    \
>+  [](SLresult err) {                                        \
>+    if (err != SL_RESULT_SUCCESS) {                         \
>+      ALOGE("%s:%d %s failed: %s", __FILE__, __LINE__, #op, \
>+            GetSLErrorString(err));                         \
>+      return true;                                          \
>+    }                                                       \
>+    return false;                                           \
>+  }(op)
>+
>+namespace webrtc {
>+
>+namespace jni {
>+
>+OpenSLESRecorder::OpenSLESRecorder(const AudioParameters& audio_parameters,
>+                                   OpenSLEngineManager* engine_manager)
>+    : audio_parameters_(audio_parameters),
>+      audio_device_buffer_(nullptr),
>+      initialized_(false),
>+      recording_(false),
>+      engine_manager_(engine_manager),
>+      engine_(nullptr),
>+      recorder_(nullptr),
>+      simple_buffer_queue_(nullptr),
>+      buffer_index_(0),
>+      last_rec_time_(0) {
>+  ALOGD("ctor[tid=%d]", rtc::CurrentThreadId());
>+  // Detach from this thread since we want to use the checker to verify calls
>+  // from the internal  audio thread.
>+  thread_checker_opensles_.DetachFromThread();
>+  // Use native audio output parameters provided by the audio manager and
>+  // define the PCM format structure.
>+  pcm_format_ = CreatePCMConfiguration(audio_parameters_.channels(),
>+                                       audio_parameters_.sample_rate(),
>+                                       audio_parameters_.bits_per_sample());
>+}
>+
>+OpenSLESRecorder::~OpenSLESRecorder() {
>+  ALOGD("dtor[tid=%d]", rtc::CurrentThreadId());
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  Terminate();
>+  DestroyAudioRecorder();
>+  engine_ = nullptr;
>+  RTC_DCHECK(!engine_);
>+  RTC_DCHECK(!recorder_);
>+  RTC_DCHECK(!simple_buffer_queue_);
>+}
>+
>+int OpenSLESRecorder::Init() {
>+  ALOGD("Init[tid=%d]", rtc::CurrentThreadId());
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  if (audio_parameters_.channels() == 2) {
>+    ALOGD("Stereo mode is enabled");
>+  }
>+  return 0;
>+}
>+
>+int OpenSLESRecorder::Terminate() {
>+  ALOGD("Terminate[tid=%d]", rtc::CurrentThreadId());
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  StopRecording();
>+  return 0;
>+}
>+
>+int OpenSLESRecorder::InitRecording() {
>+  ALOGD("InitRecording[tid=%d]", rtc::CurrentThreadId());
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  RTC_DCHECK(!initialized_);
>+  RTC_DCHECK(!recording_);
>+  if (!ObtainEngineInterface()) {
>+    ALOGE("Failed to obtain SL Engine interface");
>+    return -1;
>+  }
>+  CreateAudioRecorder();
>+  initialized_ = true;
>+  buffer_index_ = 0;
>+  return 0;
>+}
>+
>+bool OpenSLESRecorder::RecordingIsInitialized() const {
>+  return initialized_;
>+}
>+
>+int OpenSLESRecorder::StartRecording() {
>+  ALOGD("StartRecording[tid=%d]", rtc::CurrentThreadId());
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  RTC_DCHECK(initialized_);
>+  RTC_DCHECK(!recording_);
>+  if (fine_audio_buffer_) {
>+    fine_audio_buffer_->ResetRecord();
>+  }
>+  // Add buffers to the queue before changing state to SL_RECORDSTATE_RECORDING
>+  // to ensure that recording starts as soon as the state is modified. On some
>+  // devices, SLAndroidSimpleBufferQueue::Clear() used in Stop() does not flush
>+  // the buffers as intended and we therefore check the number of buffers
>+  // already queued first. Enqueue() can return SL_RESULT_BUFFER_INSUFFICIENT
>+  // otherwise.
>+  int num_buffers_in_queue = GetBufferCount();
>+  for (int i = 0; i < kNumOfOpenSLESBuffers - num_buffers_in_queue; ++i) {
>+    if (!EnqueueAudioBuffer()) {
>+      recording_ = false;
>+      return -1;
>+    }
>+  }
>+  num_buffers_in_queue = GetBufferCount();
>+  RTC_DCHECK_EQ(num_buffers_in_queue, kNumOfOpenSLESBuffers);
>+  LogBufferState();
>+  // Start audio recording by changing the state to SL_RECORDSTATE_RECORDING.
>+  // Given that buffers are already enqueued, recording should start at once.
>+  // The macro returns -1 if recording fails to start.
>+  last_rec_time_ = rtc::Time();
>+  if (LOG_ON_ERROR(
>+          (*recorder_)->SetRecordState(recorder_, SL_RECORDSTATE_RECORDING))) {
>+    return -1;
>+  }
>+  recording_ = (GetRecordState() == SL_RECORDSTATE_RECORDING);
>+  RTC_DCHECK(recording_);
>+  return 0;
>+}
>+
>+int OpenSLESRecorder::StopRecording() {
>+  ALOGD("StopRecording[tid=%d]", rtc::CurrentThreadId());
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  if (!initialized_ || !recording_) {
>+    return 0;
>+  }
>+  // Stop recording by setting the record state to SL_RECORDSTATE_STOPPED.
>+  if (LOG_ON_ERROR(
>+          (*recorder_)->SetRecordState(recorder_, SL_RECORDSTATE_STOPPED))) {
>+    return -1;
>+  }
>+  // Clear the buffer queue to get rid of old data when resuming recording.
>+  if (LOG_ON_ERROR((*simple_buffer_queue_)->Clear(simple_buffer_queue_))) {
>+    return -1;
>+  }
>+  thread_checker_opensles_.DetachFromThread();
>+  initialized_ = false;
>+  recording_ = false;
>+  return 0;
>+}
>+
>+bool OpenSLESRecorder::Recording() const {
>+  return recording_;
>+}
>+
>+void OpenSLESRecorder::AttachAudioBuffer(AudioDeviceBuffer* audio_buffer) {
>+  ALOGD("AttachAudioBuffer");
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  RTC_CHECK(audio_buffer);
>+  audio_device_buffer_ = audio_buffer;
>+  // Ensure that the audio device buffer is informed about the native sample
>+  // rate used on the recording side.
>+  const int sample_rate_hz = audio_parameters_.sample_rate();
>+  ALOGD("SetRecordingSampleRate(%d)", sample_rate_hz);
>+  audio_device_buffer_->SetRecordingSampleRate(sample_rate_hz);
>+  // Ensure that the audio device buffer is informed about the number of
>+  // channels preferred by the OS on the recording side.
>+  const size_t channels = audio_parameters_.channels();
>+  ALOGD("SetRecordingChannels(%" PRIuS ")", channels);
>+  audio_device_buffer_->SetRecordingChannels(channels);
>+  // Allocated memory for internal data buffers given existing audio parameters.
>+  AllocateDataBuffers();
>+}
>+
>+bool OpenSLESRecorder::IsAcousticEchoCancelerSupported() const {
>+  return false;
>+}
>+
>+bool OpenSLESRecorder::IsNoiseSuppressorSupported() const {
>+  return false;
>+}
>+
>+int OpenSLESRecorder::EnableBuiltInAEC(bool enable) {
>+  ALOGD("EnableBuiltInAEC(%d)", enable);
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  ALOGE("Not implemented");
>+  return 0;
>+}
>+
>+int OpenSLESRecorder::EnableBuiltInNS(bool enable) {
>+  ALOGD("EnableBuiltInNS(%d)", enable);
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  ALOGE("Not implemented");
>+  return 0;
>+}
>+
>+bool OpenSLESRecorder::ObtainEngineInterface() {
>+  ALOGD("ObtainEngineInterface");
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  if (engine_)
>+    return true;
>+  // Get access to (or create if not already existing) the global OpenSL Engine
>+  // object.
>+  SLObjectItf engine_object = engine_manager_->GetOpenSLEngine();
>+  if (engine_object == nullptr) {
>+    ALOGE("Failed to access the global OpenSL engine");
>+    return false;
>+  }
>+  // Get the SL Engine Interface which is implicit.
>+  if (LOG_ON_ERROR(
>+          (*engine_object)
>+              ->GetInterface(engine_object, SL_IID_ENGINE, &engine_))) {
>+    return false;
>+  }
>+  return true;
>+}
>+
>+bool OpenSLESRecorder::CreateAudioRecorder() {
>+  ALOGD("CreateAudioRecorder");
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  if (recorder_object_.Get())
>+    return true;
>+  RTC_DCHECK(!recorder_);
>+  RTC_DCHECK(!simple_buffer_queue_);
>+
>+  // Audio source configuration.
>+  SLDataLocator_IODevice mic_locator = {SL_DATALOCATOR_IODEVICE,
>+                                        SL_IODEVICE_AUDIOINPUT,
>+                                        SL_DEFAULTDEVICEID_AUDIOINPUT, NULL};
>+  SLDataSource audio_source = {&mic_locator, NULL};
>+
>+  // Audio sink configuration.
>+  SLDataLocator_AndroidSimpleBufferQueue buffer_queue = {
>+      SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE,
>+      static_cast<SLuint32>(kNumOfOpenSLESBuffers)};
>+  SLDataSink audio_sink = {&buffer_queue, &pcm_format_};
>+
>+  // Create the audio recorder object (requires the RECORD_AUDIO permission).
>+  // Do not realize the recorder yet. Set the configuration first.
>+  const SLInterfaceID interface_id[] = {SL_IID_ANDROIDSIMPLEBUFFERQUEUE,
>+                                        SL_IID_ANDROIDCONFIGURATION};
>+  const SLboolean interface_required[] = {SL_BOOLEAN_TRUE, SL_BOOLEAN_TRUE};
>+  if (LOG_ON_ERROR((*engine_)->CreateAudioRecorder(
>+          engine_, recorder_object_.Receive(), &audio_source, &audio_sink,
>+          arraysize(interface_id), interface_id, interface_required))) {
>+    return false;
>+  }
>+
>+  // Configure the audio recorder (before it is realized).
>+  SLAndroidConfigurationItf recorder_config;
>+  if (LOG_ON_ERROR((recorder_object_->GetInterface(recorder_object_.Get(),
>+                                                   SL_IID_ANDROIDCONFIGURATION,
>+                                                   &recorder_config)))) {
>+    return false;
>+  }
>+
>+  // Uses the default microphone tuned for audio communication.
>+  // Note that, SL_ANDROID_RECORDING_PRESET_VOICE_RECOGNITION leads to a fast
>+  // track but also excludes usage of required effects like AEC, AGC and NS.
>+  // SL_ANDROID_RECORDING_PRESET_VOICE_COMMUNICATION
>+  SLint32 stream_type = SL_ANDROID_RECORDING_PRESET_VOICE_COMMUNICATION;
>+  if (LOG_ON_ERROR(((*recorder_config)
>+                        ->SetConfiguration(recorder_config,
>+                                           SL_ANDROID_KEY_RECORDING_PRESET,
>+                                           &stream_type, sizeof(SLint32))))) {
>+    return false;
>+  }
>+
>+  // The audio recorder can now be realized (in synchronous mode).
>+  if (LOG_ON_ERROR((recorder_object_->Realize(recorder_object_.Get(),
>+                                              SL_BOOLEAN_FALSE)))) {
>+    return false;
>+  }
>+
>+  // Get the implicit recorder interface (SL_IID_RECORD).
>+  if (LOG_ON_ERROR((recorder_object_->GetInterface(
>+          recorder_object_.Get(), SL_IID_RECORD, &recorder_)))) {
>+    return false;
>+  }
>+
>+  // Get the simple buffer queue interface (SL_IID_ANDROIDSIMPLEBUFFERQUEUE).
>+  // It was explicitly requested.
>+  if (LOG_ON_ERROR((recorder_object_->GetInterface(
>+          recorder_object_.Get(), SL_IID_ANDROIDSIMPLEBUFFERQUEUE,
>+          &simple_buffer_queue_)))) {
>+    return false;
>+  }
>+
>+  // Register the input callback for the simple buffer queue.
>+  // This callback will be called when receiving new data from the device.
>+  if (LOG_ON_ERROR(((*simple_buffer_queue_)
>+                        ->RegisterCallback(simple_buffer_queue_,
>+                                           SimpleBufferQueueCallback, this)))) {
>+    return false;
>+  }
>+  return true;
>+}
>+
>+void OpenSLESRecorder::DestroyAudioRecorder() {
>+  ALOGD("DestroyAudioRecorder");
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  if (!recorder_object_.Get())
>+    return;
>+  (*simple_buffer_queue_)
>+      ->RegisterCallback(simple_buffer_queue_, nullptr, nullptr);
>+  recorder_object_.Reset();
>+  recorder_ = nullptr;
>+  simple_buffer_queue_ = nullptr;
>+}
>+
>+void OpenSLESRecorder::SimpleBufferQueueCallback(
>+    SLAndroidSimpleBufferQueueItf buffer_queue,
>+    void* context) {
>+  OpenSLESRecorder* stream = static_cast<OpenSLESRecorder*>(context);
>+  stream->ReadBufferQueue();
>+}
>+
>+void OpenSLESRecorder::AllocateDataBuffers() {
>+  ALOGD("AllocateDataBuffers");
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  RTC_DCHECK(!simple_buffer_queue_);
>+  RTC_CHECK(audio_device_buffer_);
>+  // Create a modified audio buffer class which allows us to deliver any number
>+  // of samples (and not only multiple of 10ms) to match the native audio unit
>+  // buffer size.
>+  ALOGD("frames per native buffer: %" PRIuS,
>+        audio_parameters_.frames_per_buffer());
>+  ALOGD("frames per 10ms buffer: %" PRIuS,
>+        audio_parameters_.frames_per_10ms_buffer());
>+  ALOGD("bytes per native buffer: %" PRIuS,
>+        audio_parameters_.GetBytesPerBuffer());
>+  ALOGD("native sample rate: %d", audio_parameters_.sample_rate());
>+  RTC_DCHECK(audio_device_buffer_);
>+  fine_audio_buffer_ = absl::make_unique<FineAudioBuffer>(audio_device_buffer_);
>+  // Allocate queue of audio buffers that stores recorded audio samples.
>+  const int buffer_size_samples =
>+      audio_parameters_.frames_per_buffer() * audio_parameters_.channels();
>+  audio_buffers_.reset(new std::unique_ptr<SLint16[]>[kNumOfOpenSLESBuffers]);
>+  for (int i = 0; i < kNumOfOpenSLESBuffers; ++i) {
>+    audio_buffers_[i].reset(new SLint16[buffer_size_samples]);
>+  }
>+}
>+
>+void OpenSLESRecorder::ReadBufferQueue() {
>+  RTC_DCHECK(thread_checker_opensles_.CalledOnValidThread());
>+  SLuint32 state = GetRecordState();
>+  if (state != SL_RECORDSTATE_RECORDING) {
>+    ALOGW("Buffer callback in non-recording state!");
>+    return;
>+  }
>+  // Check delta time between two successive callbacks and provide a warning
>+  // if it becomes very large.
>+  // TODO(henrika): using 150ms as upper limit but this value is rather random.
>+  const uint32_t current_time = rtc::Time();
>+  const uint32_t diff = current_time - last_rec_time_;
>+  if (diff > 150) {
>+    ALOGW("Bad OpenSL ES record timing, dT=%u [ms]", diff);
>+  }
>+  last_rec_time_ = current_time;
>+  // Send recorded audio data to the WebRTC sink.
>+  // TODO(henrika): fix delay estimates. It is OK to use fixed values for now
>+  // since there is no support to turn off built-in EC in combination with
>+  // OpenSL ES anyhow. Hence, as is, the WebRTC based AEC (which would use
>+  // these estimates) will never be active.
>+  fine_audio_buffer_->DeliverRecordedData(
>+      rtc::ArrayView<const int16_t>(
>+          audio_buffers_[buffer_index_].get(),
>+          audio_parameters_.frames_per_buffer() * audio_parameters_.channels()),
>+      25);
>+  // Enqueue the utilized audio buffer and use if for recording again.
>+  EnqueueAudioBuffer();
>+}
>+
>+bool OpenSLESRecorder::EnqueueAudioBuffer() {
>+  SLresult err =
>+      (*simple_buffer_queue_)
>+          ->Enqueue(
>+              simple_buffer_queue_,
>+              reinterpret_cast<SLint8*>(audio_buffers_[buffer_index_].get()),
>+              audio_parameters_.GetBytesPerBuffer());
>+  if (SL_RESULT_SUCCESS != err) {
>+    ALOGE("Enqueue failed: %s", GetSLErrorString(err));
>+    return false;
>+  }
>+  buffer_index_ = (buffer_index_ + 1) % kNumOfOpenSLESBuffers;
>+  return true;
>+}
>+
>+SLuint32 OpenSLESRecorder::GetRecordState() const {
>+  RTC_DCHECK(recorder_);
>+  SLuint32 state;
>+  SLresult err = (*recorder_)->GetRecordState(recorder_, &state);
>+  if (SL_RESULT_SUCCESS != err) {
>+    ALOGE("GetRecordState failed: %s", GetSLErrorString(err));
>+  }
>+  return state;
>+}
>+
>+SLAndroidSimpleBufferQueueState OpenSLESRecorder::GetBufferQueueState() const {
>+  RTC_DCHECK(simple_buffer_queue_);
>+  // state.count: Number of buffers currently in the queue.
>+  // state.index: Index of the currently filling buffer. This is a linear index
>+  // that keeps a cumulative count of the number of buffers recorded.
>+  SLAndroidSimpleBufferQueueState state;
>+  SLresult err =
>+      (*simple_buffer_queue_)->GetState(simple_buffer_queue_, &state);
>+  if (SL_RESULT_SUCCESS != err) {
>+    ALOGE("GetState failed: %s", GetSLErrorString(err));
>+  }
>+  return state;
>+}
>+
>+void OpenSLESRecorder::LogBufferState() const {
>+  SLAndroidSimpleBufferQueueState state = GetBufferQueueState();
>+  ALOGD("state.count:%d state.index:%d", state.count, state.index);
>+}
>+
>+SLuint32 OpenSLESRecorder::GetBufferCount() {
>+  SLAndroidSimpleBufferQueueState state = GetBufferQueueState();
>+  return state.count;
>+}
>+
>+}  // namespace jni
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/opensles_recorder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/opensles_recorder.h
>new file mode 100644
>index 00000000000..6408be0f563
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/audio_device/opensles_recorder.h
>@@ -0,0 +1,192 @@
>+/*
>+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef SDK_ANDROID_SRC_JNI_AUDIO_DEVICE_OPENSLES_RECORDER_H_
>+#define SDK_ANDROID_SRC_JNI_AUDIO_DEVICE_OPENSLES_RECORDER_H_
>+
>+#include <SLES/OpenSLES.h>
>+#include <SLES/OpenSLES_Android.h>
>+#include <SLES/OpenSLES_AndroidConfiguration.h>
>+
>+#include <memory>
>+
>+#include "modules/audio_device/audio_device_buffer.h"
>+#include "modules/audio_device/fine_audio_buffer.h"
>+#include "modules/audio_device/include/audio_device_defines.h"
>+#include "rtc_base/thread_checker.h"
>+#include "sdk/android/src/jni/audio_device/audio_common.h"
>+#include "sdk/android/src/jni/audio_device/audio_device_module.h"
>+#include "sdk/android/src/jni/audio_device/opensles_common.h"
>+
>+namespace webrtc {
>+
>+class FineAudioBuffer;
>+
>+namespace jni {
>+
>+// Implements 16-bit mono PCM audio input support for Android using the
>+// C based OpenSL ES API. No calls from C/C++ to Java using JNI is done.
>+//
>+// An instance can be created on any thread, but must then be used on one and
>+// the same thread. All public methods must also be called on the same thread. A
>+// thread checker will RTC_DCHECK if any method is called on an invalid thread.
>+// Recorded audio buffers are provided on a dedicated internal thread managed by
>+// the OpenSL ES layer.
>+//
>+// The existing design forces the user to call InitRecording() after
>+// StopRecording() to be able to call StartRecording() again. This is inline
>+// with how the Java-based implementation works.
>+//
>+// As of API level 21, lower latency audio input is supported on select devices.
>+// To take advantage of this feature, first confirm that lower latency output is
>+// available. The capability for lower latency output is a prerequisite for the
>+// lower latency input feature. Then, create an AudioRecorder with the same
>+// sample rate and buffer size as would be used for output. OpenSL ES interfaces
>+// for input effects preclude the lower latency path.
>+// See https://developer.android.com/ndk/guides/audio/opensl-prog-notes.html
>+// for more details.
>+class OpenSLESRecorder : public AudioInput {
>+ public:
>+  // Beginning with API level 17 (Android 4.2), a buffer count of 2 or more is
>+  // required for lower latency. Beginning with API level 18 (Android 4.3), a
>+  // buffer count of 1 is sufficient for lower latency. In addition, the buffer
>+  // size and sample rate must be compatible with the device's native input
>+  // configuration provided via the audio manager at construction.
>+  // TODO(henrika): perhaps set this value dynamically based on OS version.
>+  static const int kNumOfOpenSLESBuffers = 2;
>+
>+  OpenSLESRecorder(const AudioParameters& audio_parameters,
>+                   OpenSLEngineManager* engine_manager);
>+  ~OpenSLESRecorder() override;
>+
>+  int Init() override;
>+  int Terminate() override;
>+
>+  int InitRecording() override;
>+  bool RecordingIsInitialized() const override;
>+
>+  int StartRecording() override;
>+  int StopRecording() override;
>+  bool Recording() const override;
>+
>+  void AttachAudioBuffer(AudioDeviceBuffer* audio_buffer) override;
>+
>+  // TODO(henrika): add support using OpenSL ES APIs when available.
>+  bool IsAcousticEchoCancelerSupported() const override;
>+  bool IsNoiseSuppressorSupported() const override;
>+  int EnableBuiltInAEC(bool enable) override;
>+  int EnableBuiltInNS(bool enable) override;
>+
>+ private:
>+  // Obtaines the SL Engine Interface from the existing global Engine object.
>+  // The interface exposes creation methods of all the OpenSL ES object types.
>+  // This method defines the |engine_| member variable.
>+  bool ObtainEngineInterface();
>+
>+  // Creates/destroys the audio recorder and the simple-buffer queue object.
>+  bool CreateAudioRecorder();
>+  void DestroyAudioRecorder();
>+
>+  // Allocate memory for audio buffers which will be used to capture audio
>+  // via the SLAndroidSimpleBufferQueueItf interface.
>+  void AllocateDataBuffers();
>+
>+  // These callback methods are called when data has been written to the input
>+  // buffer queue. They are both called from an internal "OpenSL ES thread"
>+  // which is not attached to the Dalvik VM.
>+  static void SimpleBufferQueueCallback(SLAndroidSimpleBufferQueueItf caller,
>+                                        void* context);
>+  void ReadBufferQueue();
>+
>+  // Wraps calls to SLAndroidSimpleBufferQueueState::Enqueue() and it can be
>+  // called both on the main thread (but before recording has started) and from
>+  // the internal audio thread while input streaming is active. It uses
>+  // |simple_buffer_queue_| but no lock is needed since the initial calls from
>+  // the main thread and the native callback thread are mutually exclusive.
>+  bool EnqueueAudioBuffer();
>+
>+  // Returns the current recorder state.
>+  SLuint32 GetRecordState() const;
>+
>+  // Returns the current buffer queue state.
>+  SLAndroidSimpleBufferQueueState GetBufferQueueState() const;
>+
>+  // Number of buffers currently in the queue.
>+  SLuint32 GetBufferCount();
>+
>+  // Prints a log message of the current queue state. Can be used for debugging
>+  // purposes.
>+  void LogBufferState() const;
>+
>+  // Ensures that methods are called from the same thread as this object is
>+  // created on.
>+  rtc::ThreadChecker thread_checker_;
>+
>+  // Stores thread ID in first call to SimpleBufferQueueCallback() from internal
>+  // non-application thread which is not attached to the Dalvik JVM.
>+  // Detached during construction of this object.
>+  rtc::ThreadChecker thread_checker_opensles_;
>+
>+  const AudioParameters audio_parameters_;
>+
>+  // Raw pointer handle provided to us in AttachAudioBuffer(). Owned by the
>+  // AudioDeviceModuleImpl class and called by AudioDeviceModule::Create().
>+  AudioDeviceBuffer* audio_device_buffer_;
>+
>+  // PCM-type format definition.
>+  // TODO(henrika): add support for SLAndroidDataFormat_PCM_EX (android-21) if
>+  // 32-bit float representation is needed.
>+  SLDataFormat_PCM pcm_format_;
>+
>+  bool initialized_;
>+  bool recording_;
>+
>+  OpenSLEngineManager* const engine_manager_;
>+  // This interface exposes creation methods for all the OpenSL ES object types.
>+  // It is the OpenSL ES API entry point.
>+  SLEngineItf engine_;
>+
>+  // The audio recorder media object records audio to the destination specified
>+  // by the data sink capturing it from the input specified by the data source.
>+  ScopedSLObjectItf recorder_object_;
>+
>+  // This interface is supported on the audio recorder object and it controls
>+  // the state of the audio recorder.
>+  SLRecordItf recorder_;
>+
>+  // The Android Simple Buffer Queue interface is supported on the audio
>+  // recorder. For recording, an app should enqueue empty buffers. When a
>+  // registered callback sends notification that the system has finished writing
>+  // data to the buffer, the app can read the buffer.
>+  SLAndroidSimpleBufferQueueItf simple_buffer_queue_;
>+
>+  // Consumes audio of native buffer size and feeds the WebRTC layer with 10ms
>+  // chunks of audio.
>+  std::unique_ptr<FineAudioBuffer> fine_audio_buffer_;
>+
>+  // Queue of audio buffers to be used by the recorder object for capturing
>+  // audio. They will be used in a Round-robin way and the size of each buffer
>+  // is given by AudioParameters::frames_per_buffer(), i.e., it corresponds to
>+  // the native OpenSL ES buffer size.
>+  std::unique_ptr<std::unique_ptr<SLint16[]>[]> audio_buffers_;
>+
>+  // Keeps track of active audio buffer 'n' in the audio_buffers_[n] queue.
>+  // Example (kNumOfOpenSLESBuffers = 2): counts 0, 1, 0, 1, ...
>+  int buffer_index_;
>+
>+  // Last time the OpenSL ES layer delivered recorded audio data.
>+  uint32_t last_rec_time_;
>+};
>+
>+}  // namespace jni
>+
>+}  // namespace webrtc
>+
>+#endif  // SDK_ANDROID_SRC_JNI_AUDIO_DEVICE_OPENSLES_RECORDER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/class_loader.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/class_loader.h
>index 52bb1ba5231..4b255a4660c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/class_loader.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/class_loader.h
>@@ -8,32 +8,16 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-// Android's FindClass() is tricky because the app-specific ClassLoader is not
>-// consulted when there is no app-specific frame on the stack (i.e. when called
>-// from a thread created from native C++ code). These helper functions provide a
>-// workaround for this.
>-// http://developer.android.com/training/articles/perf-jni.html#faq_FindClass
>+// Deprecated: use sdk/android/native_api/jni/class_loader.h instead.
> 
> #ifndef SDK_ANDROID_SRC_JNI_CLASS_LOADER_H_
> #define SDK_ANDROID_SRC_JNI_CLASS_LOADER_H_
> 
>-#include <jni.h>
>+#include "sdk/android/native_api/jni/class_loader.h"
> 
> namespace webrtc {
> namespace jni {
>-
>-// This method should be called from JNI_OnLoad and before any calls to
>-// FindClass.
>-void InitClassLoader(JNIEnv* env);
>-
>-// This function is identical to JNIEnv::FindClass except that it works from any
>-// thread. This function loads and returns a local reference to the class with
>-// the given name. The name argument is a fully-qualified class name. For
>-// example, the fully-qualified class name for the java.lang.String class is:
>-// "java/lang/String". This function will be used from the JNI generated code
>-// and should rarely be used manually.
>-jclass GetClass(JNIEnv* env, const char* name);
>-
>+using ::webrtc::InitClassLoader;
> }  // namespace jni
> }  // namespace webrtc
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/classreferenceholder.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/classreferenceholder.cc
>deleted file mode 100644
>index b1f8fdaaf66..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/classreferenceholder.cc
>+++ /dev/null
>@@ -1,147 +0,0 @@
>-/*
>- *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-#include "sdk/android/src/jni/classreferenceholder.h"
>-
>-#include "sdk/android/src/jni/class_loader.h"
>-#include "sdk/android/src/jni/jni_helpers.h"
>-
>-namespace webrtc {
>-namespace jni {
>-
>-// ClassReferenceHolder holds global reference to Java classes in app/webrtc.
>-class ClassReferenceHolder {
>- public:
>-  explicit ClassReferenceHolder(JNIEnv* jni);
>-  ~ClassReferenceHolder();
>-
>-  void FreeReferences(JNIEnv* jni);
>-  jclass GetClass(const std::string& name);
>-
>- private:
>-  void LoadClass(JNIEnv* jni, const std::string& name);
>-
>-  std::map<std::string, jclass> classes_;
>-};
>-
>-// Allocated in LoadGlobalClassReferenceHolder(),
>-// freed in FreeGlobalClassReferenceHolder().
>-static ClassReferenceHolder* g_class_reference_holder = nullptr;
>-
>-void LoadGlobalClassReferenceHolder() {
>-  JNIEnv* env = GetEnv();
>-  RTC_CHECK(g_class_reference_holder == nullptr);
>-  g_class_reference_holder = new ClassReferenceHolder(env);
>-  // TODO(magjed): This is a weird place to call the other class loader from,
>-  // but the only place that will keep backwards compatibility.
>-  InitClassLoader(env);
>-}
>-
>-void FreeGlobalClassReferenceHolder() {
>-  g_class_reference_holder->FreeReferences(AttachCurrentThreadIfNeeded());
>-  delete g_class_reference_holder;
>-  g_class_reference_holder = nullptr;
>-}
>-
>-ClassReferenceHolder::ClassReferenceHolder(JNIEnv* jni) {
>-  LoadClass(jni, "android/graphics/SurfaceTexture");
>-  LoadClass(jni, "java/lang/String");
>-  LoadClass(jni, "java/nio/ByteBuffer");
>-  LoadClass(jni, "java/util/ArrayList");
>-  LoadClass(jni, "java/util/LinkedHashMap");
>-  LoadClass(jni, "org/webrtc/Camera1Enumerator");
>-  LoadClass(jni, "org/webrtc/Camera2Enumerator");
>-  LoadClass(jni, "org/webrtc/CameraEnumerationAndroid");
>-  LoadClass(jni, "org/webrtc/EglBase");
>-  LoadClass(jni, "org/webrtc/EglBase$Context");
>-  LoadClass(jni, "org/webrtc/EglBase14$Context");
>-  LoadClass(jni, "org/webrtc/EncodedImage");
>-  LoadClass(jni, "org/webrtc/EncodedImage$FrameType");
>-  LoadClass(jni, "org/webrtc/MediaCodecVideoDecoder");
>-  LoadClass(jni, "org/webrtc/MediaCodecVideoDecoder$DecodedOutputBuffer");
>-  LoadClass(jni, "org/webrtc/MediaCodecVideoDecoder$DecodedTextureBuffer");
>-  LoadClass(jni, "org/webrtc/MediaCodecVideoDecoder$VideoCodecType");
>-  LoadClass(jni, "org/webrtc/MediaCodecVideoEncoder");
>-  LoadClass(jni, "org/webrtc/MediaCodecVideoEncoder$OutputBufferInfo");
>-  LoadClass(jni, "org/webrtc/MediaCodecVideoEncoder$VideoCodecType");
>-  LoadClass(jni, "org/webrtc/MediaSource$State");
>-  LoadClass(jni, "org/webrtc/NetworkMonitor");
>-  LoadClass(jni, "org/webrtc/NetworkMonitorAutoDetect$ConnectionType");
>-  LoadClass(jni, "org/webrtc/NetworkMonitorAutoDetect$IPAddress");
>-  LoadClass(jni, "org/webrtc/NetworkMonitorAutoDetect$NetworkInformation");
>-  LoadClass(jni, "org/webrtc/PeerConnection$BundlePolicy");
>-  LoadClass(jni, "org/webrtc/PeerConnection$CandidateNetworkPolicy");
>-  LoadClass(jni, "org/webrtc/PeerConnection$ContinualGatheringPolicy");
>-  LoadClass(jni, "org/webrtc/PeerConnection$IceConnectionState");
>-  LoadClass(jni, "org/webrtc/PeerConnection$IceGatheringState");
>-  LoadClass(jni, "org/webrtc/PeerConnection$IceTransportsType");
>-  LoadClass(jni, "org/webrtc/PeerConnection$KeyType");
>-  LoadClass(jni, "org/webrtc/PeerConnection$RtcpMuxPolicy");
>-  LoadClass(jni, "org/webrtc/PeerConnection$SignalingState");
>-  LoadClass(jni, "org/webrtc/PeerConnection$TcpCandidatePolicy");
>-  LoadClass(jni, "org/webrtc/PeerConnection$TlsCertPolicy");
>-  LoadClass(jni, "org/webrtc/PeerConnectionFactory");
>-  LoadClass(jni, "org/webrtc/RTCStats");
>-  LoadClass(jni, "org/webrtc/RTCStatsReport");
>-  LoadClass(jni, "org/webrtc/RtpReceiver");
>-  LoadClass(jni, "org/webrtc/RtpSender");
>-  LoadClass(jni, "org/webrtc/SessionDescription");
>-  LoadClass(jni, "org/webrtc/SessionDescription$Type");
>-  LoadClass(jni, "org/webrtc/StatsReport");
>-  LoadClass(jni, "org/webrtc/StatsReport$Value");
>-  LoadClass(jni, "org/webrtc/SurfaceTextureHelper");
>-  LoadClass(jni, "org/webrtc/VideoCapturer");
>-  LoadClass(jni, "org/webrtc/VideoCodecInfo");
>-  LoadClass(jni, "org/webrtc/VideoCodecStatus");
>-  LoadClass(jni, "org/webrtc/VideoFrame");
>-  LoadClass(jni, "org/webrtc/VideoFrame$Buffer");
>-  LoadClass(jni, "org/webrtc/VideoFrame$I420Buffer");
>-  LoadClass(jni, "org/webrtc/VideoFrame$TextureBuffer");
>-  LoadClass(jni, "org/webrtc/VideoRenderer$I420Frame");
>-  LoadClass(jni, "org/webrtc/VideoSink");
>-  LoadClass(jni, "org/webrtc/WrappedNativeI420Buffer");
>-}
>-
>-ClassReferenceHolder::~ClassReferenceHolder() {
>-  RTC_CHECK(classes_.empty()) << "Must call FreeReferences() before dtor!";
>-}
>-
>-void ClassReferenceHolder::FreeReferences(JNIEnv* jni) {
>-  for (std::map<std::string, jclass>::const_iterator it = classes_.begin();
>-      it != classes_.end(); ++it) {
>-    jni->DeleteGlobalRef(it->second);
>-  }
>-  classes_.clear();
>-}
>-
>-jclass ClassReferenceHolder::GetClass(const std::string& name) {
>-  std::map<std::string, jclass>::iterator it = classes_.find(name);
>-  RTC_CHECK(it != classes_.end()) << "Unexpected GetClass() call for: " << name;
>-  return it->second;
>-}
>-
>-void ClassReferenceHolder::LoadClass(JNIEnv* jni, const std::string& name) {
>-  jclass localRef = jni->FindClass(name.c_str());
>-  CHECK_EXCEPTION(jni) << "error during FindClass: " << name;
>-  RTC_CHECK(localRef) << name;
>-  jclass globalRef = reinterpret_cast<jclass>(jni->NewGlobalRef(localRef));
>-  CHECK_EXCEPTION(jni) << "error during NewGlobalRef: " << name;
>-  RTC_CHECK(globalRef) << name;
>-  bool inserted = classes_.insert(std::make_pair(name, globalRef)).second;
>-  RTC_CHECK(inserted) << "Duplicate class name: " << name;
>-}
>-
>-// Returns a global reference guaranteed to be valid for the lifetime of the
>-// process.
>-jclass FindClass(JNIEnv* jni, const char* name) {
>-  return g_class_reference_holder->GetClass(name);
>-}
>-
>-}  // namespace jni
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/classreferenceholder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/classreferenceholder.h
>index 5145c108e66..51759fc95ab 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/classreferenceholder.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/classreferenceholder.h
>@@ -8,35 +8,24 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-// Android's FindClass() is trickier than usual because the app-specific
>-// ClassLoader is not consulted when there is no app-specific frame on the
>-// stack.  Consequently, we only look up all classes once in app/webrtc.
>-// http://developer.android.com/training/articles/perf-jni.html#faq_FindClass
>-
> #ifndef SDK_ANDROID_SRC_JNI_CLASSREFERENCEHOLDER_H_
> #define SDK_ANDROID_SRC_JNI_CLASSREFERENCEHOLDER_H_
> 
>-// TODO(magjed): Remove this whole file and replace with either generated JNI
>-// code or class_loader.h.
>-
>-#include <jni.h>
>-#include <map>
>-#include <string>
>+// TODO(magjed): Update external clients to call webrtc::jni::InitClassLoader
>+// immediately instead.
>+#include "sdk/android/native_api/jni/class_loader.h"
>+#include "sdk/android/src/jni/jni_helpers.h"
> 
> namespace webrtc {
> namespace jni {
> 
>-// LoadGlobalClassReferenceHolder must be called in JNI_OnLoad.
>-void LoadGlobalClassReferenceHolder();
>-// FreeGlobalClassReferenceHolder must be called in JNI_UnLoad.
>-void FreeGlobalClassReferenceHolder();
>-
>-// Deprecated. Most cases of finding classes should be done with generated JNI
>-// code, and the few remaining cases should use the function from
>-// class_loader.h.
>-// Returns a global reference guaranteed to be valid for the lifetime of the
>-// process.
>-jclass FindClass(JNIEnv* jni, const char* name);
>+// Deprecated. Call webrtc::jni::InitClassLoader() immediately instead..
>+inline void LoadGlobalClassReferenceHolder() {
>+  webrtc::InitClassLoader(GetEnv());
>+}
>+
>+// Deprecated. Do not call at all.
>+inline void FreeGlobalClassReferenceHolder() {}
> 
> }  // namespace jni
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/encodedimage.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/encodedimage.cc
>index 81c5caa761d..c0ea28e2dc3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/encodedimage.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/encodedimage.cc
>@@ -13,27 +13,35 @@
> #include "common_video/include/video_frame.h"
> #include "rtc_base/timeutils.h"
> #include "sdk/android/generated_video_jni/jni/EncodedImage_jni.h"
>+#include "sdk/android/native_api/jni/java_types.h"
> #include "sdk/android/src/jni/jni_helpers.h"
> 
> namespace webrtc {
> namespace jni {
> 
>-jobject NativeToJavaFrameType(JNIEnv* env, FrameType frame_type) {
>+ScopedJavaLocalRef<jobject> NativeToJavaFrameType(JNIEnv* env,
>+                                                  FrameType frame_type) {
>   return Java_FrameType_fromNativeIndex(env, frame_type);
> }
> 
>-jobject NativeToJavaEncodedImage(JNIEnv* jni, const EncodedImage& image) {
>-  jobject buffer = jni->NewDirectByteBuffer(image._buffer, image._length);
>-  jobject frame_type = NativeToJavaFrameType(jni, image._frameType);
>-  jobject qp =
>-      (image.qp_ == -1) ? nullptr : NativeToJavaInteger(jni, image.qp_);
>+ScopedJavaLocalRef<jobject> NativeToJavaEncodedImage(
>+    JNIEnv* jni,
>+    const EncodedImage& image) {
>+  ScopedJavaLocalRef<jobject> buffer =
>+      NewDirectByteBuffer(jni, image._buffer, image._length);
>+  ScopedJavaLocalRef<jobject> frame_type =
>+      NativeToJavaFrameType(jni, image._frameType);
>+  ScopedJavaLocalRef<jobject> qp;
>+  if (image.qp_ != -1)
>+    qp = NativeToJavaInteger(jni, image.qp_);
>   return Java_EncodedImage_Constructor(
>-      jni, buffer, image._encodedWidth, image._encodedHeight,
>+      jni, buffer, static_cast<int>(image._encodedWidth),
>+      static_cast<int>(image._encodedHeight),
>       image.capture_time_ms_ * rtc::kNumNanosecsPerMillisec, frame_type,
>       static_cast<jint>(image.rotation_), image._completeFrame, qp);
> }
> 
>-jobjectArray NativeToJavaFrameTypeArray(
>+ScopedJavaLocalRef<jobjectArray> NativeToJavaFrameTypeArray(
>     JNIEnv* env,
>     const std::vector<FrameType>& frame_types) {
>   return NativeToJavaObjectArray(
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/encodedimage.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/encodedimage.h
>index 563b0d00956..66c226b1c9b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/encodedimage.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/encodedimage.h
>@@ -12,18 +12,21 @@
> #define SDK_ANDROID_SRC_JNI_ENCODEDIMAGE_H_
> 
> #include <jni.h>
>-
> #include "common_types.h"  // NOLINT(build/include)
> 
>+#include "sdk/android/native_api/jni/scoped_java_ref.h"
>+
> namespace webrtc {
> 
> class EncodedImage;
> 
> namespace jni {
> 
>-jobject NativeToJavaFrameType(JNIEnv* env, FrameType frame_type);
>-jobject NativeToJavaEncodedImage(JNIEnv* jni, const EncodedImage& image);
>-jobjectArray NativeToJavaFrameTypeArray(
>+ScopedJavaLocalRef<jobject> NativeToJavaFrameType(JNIEnv* env,
>+                                                  FrameType frame_type);
>+ScopedJavaLocalRef<jobject> NativeToJavaEncodedImage(JNIEnv* jni,
>+                                                     const EncodedImage& image);
>+ScopedJavaLocalRef<jobjectArray> NativeToJavaFrameTypeArray(
>     JNIEnv* env,
>     const std::vector<FrameType>& frame_types);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/h264utils.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/h264utils.cc
>new file mode 100644
>index 00000000000..d40266a55d5
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/h264utils.cc
>@@ -0,0 +1,29 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "sdk/android/src/jni/videocodecinfo.h"
>+
>+#include "common_video/h264/profile_level_id.h"
>+#include "sdk/android/generated_video_jni/jni/H264Utils_jni.h"
>+
>+namespace webrtc {
>+namespace jni {
>+
>+static jboolean JNI_H264Utils_IsSameH264Profile(
>+    JNIEnv* env,
>+    const JavaParamRef<jclass>&,
>+    const JavaParamRef<jobject>& params1,
>+    const JavaParamRef<jobject>& params2) {
>+  return H264::IsSameH264Profile(JavaToNativeStringMap(env, params1),
>+                                 JavaToNativeStringMap(env, params2));
>+}
>+
>+}  // namespace jni
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/hardwarevideoencoderfactory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/hardwarevideoencoderfactory.cc
>deleted file mode 100644
>index baeb426cdf1..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/hardwarevideoencoderfactory.cc
>+++ /dev/null
>@@ -1,32 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include <jni.h>
>-
>-#include "media/base/codec.h"
>-#include "media/base/h264_profile_level_id.h"
>-#include "media/base/mediaconstants.h"
>-#include "sdk/android/src/jni/jni_helpers.h"
>-
>-namespace webrtc {
>-namespace jni {
>-
>-JNI_FUNCTION_DECLARATION(jboolean,
>-                         HardwareVideoEncoderFactory_isSameH264Profile,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jobject params1,
>-                         jobject params2) {
>-  return H264::IsSameH264Profile(JavaToStdMapStrings(jni, params1),
>-                                 JavaToStdMapStrings(jni, params2));
>-}
>-
>-}  // namespace jni
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/javai420buffer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/javai420buffer.cc
>new file mode 100644
>index 00000000000..a98462fa7a1
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/javai420buffer.cc
>@@ -0,0 +1,64 @@
>+/*
>+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "sdk/android/generated_video_jni/jni/JavaI420Buffer_jni.h"
>+#include "third_party/libyuv/include/libyuv/scale.h"
>+
>+namespace webrtc {
>+namespace jni {
>+
>+static void JNI_JavaI420Buffer_CropAndScaleI420(
>+    JNIEnv* jni,
>+    const JavaParamRef<jclass>&,
>+    const JavaParamRef<jobject>& j_src_y,
>+    jint src_stride_y,
>+    const JavaParamRef<jobject>& j_src_u,
>+    jint src_stride_u,
>+    const JavaParamRef<jobject>& j_src_v,
>+    jint src_stride_v,
>+    jint crop_x,
>+    jint crop_y,
>+    jint crop_width,
>+    jint crop_height,
>+    const JavaParamRef<jobject>& j_dst_y,
>+    jint dst_stride_y,
>+    const JavaParamRef<jobject>& j_dst_u,
>+    jint dst_stride_u,
>+    const JavaParamRef<jobject>& j_dst_v,
>+    jint dst_stride_v,
>+    jint scale_width,
>+    jint scale_height) {
>+  uint8_t const* src_y =
>+      static_cast<uint8_t*>(jni->GetDirectBufferAddress(j_src_y.obj()));
>+  uint8_t const* src_u =
>+      static_cast<uint8_t*>(jni->GetDirectBufferAddress(j_src_u.obj()));
>+  uint8_t const* src_v =
>+      static_cast<uint8_t*>(jni->GetDirectBufferAddress(j_src_v.obj()));
>+  uint8_t* dst_y =
>+      static_cast<uint8_t*>(jni->GetDirectBufferAddress(j_dst_y.obj()));
>+  uint8_t* dst_u =
>+      static_cast<uint8_t*>(jni->GetDirectBufferAddress(j_dst_u.obj()));
>+  uint8_t* dst_v =
>+      static_cast<uint8_t*>(jni->GetDirectBufferAddress(j_dst_v.obj()));
>+
>+  // Perform cropping using pointer arithmetic.
>+  src_y += crop_x + crop_y * src_stride_y;
>+  src_u += crop_x / 2 + crop_y / 2 * src_stride_u;
>+  src_v += crop_x / 2 + crop_y / 2 * src_stride_v;
>+
>+  bool ret = libyuv::I420Scale(
>+      src_y, src_stride_y, src_u, src_stride_u, src_v, src_stride_v, crop_width,
>+      crop_height, dst_y, dst_stride_y, dst_u, dst_stride_u, dst_v,
>+      dst_stride_v, scale_width, scale_height, libyuv::kFilterBox);
>+  RTC_DCHECK_EQ(ret, 0) << "I420Scale failed";
>+}
>+
>+}  // namespace jni
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/jni_common.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/jni_common.cc
>index 8435c7820d4..2d4bbeded3e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/jni_common.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/jni_common.cc
>@@ -9,45 +9,39 @@
>  */
> 
> #include "rtc_base/refcount.h"
>+#include "sdk/android/generated_base_jni/jni/JniCommon_jni.h"
> #include "sdk/android/src/jni/jni_helpers.h"
> 
> namespace webrtc {
> namespace jni {
> 
>-JNI_FUNCTION_DECLARATION(void,
>-                         JniCommon_nativeAddRef,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jlong j_native_ref_counted_pointer) {
>+static void JNI_JniCommon_AddRef(JNIEnv* jni,
>+                                 const JavaParamRef<jclass>&,
>+                                 jlong j_native_ref_counted_pointer) {
>   reinterpret_cast<rtc::RefCountInterface*>(j_native_ref_counted_pointer)
>       ->AddRef();
> }
> 
>-JNI_FUNCTION_DECLARATION(void,
>-                         JniCommon_nativeReleaseRef,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jlong j_native_ref_counted_pointer) {
>+static void JNI_JniCommon_ReleaseRef(JNIEnv* jni,
>+                                     const JavaParamRef<jclass>&,
>+                                     jlong j_native_ref_counted_pointer) {
>   reinterpret_cast<rtc::RefCountInterface*>(j_native_ref_counted_pointer)
>       ->Release();
> }
> 
>-JNI_FUNCTION_DECLARATION(jobject,
>-                         JniCommon_allocateNativeByteBuffer,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jint size) {
>+static ScopedJavaLocalRef<jobject> JNI_JniCommon_AllocateByteBuffer(
>+    JNIEnv* jni,
>+    const JavaParamRef<jclass>&,
>+    jint size) {
>   void* new_data = ::operator new(size);
>-  jobject byte_buffer = jni->NewDirectByteBuffer(new_data, size);
>-  return byte_buffer;
>+  return NewDirectByteBuffer(jni, new_data, size);
> }
> 
>-JNI_FUNCTION_DECLARATION(void,
>-                         JniCommon_freeNativeByteBuffer,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jobject byte_buffer) {
>-  void* data = jni->GetDirectBufferAddress(byte_buffer);
>+static void JNI_JniCommon_FreeByteBuffer(
>+    JNIEnv* jni,
>+    const JavaParamRef<jclass>&,
>+    const JavaParamRef<jobject>& byte_buffer) {
>+  void* data = jni->GetDirectBufferAddress(byte_buffer.obj());
>   ::operator delete(data);
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/jni_generator_helper.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/jni_generator_helper.cc
>index c870e1c51c5..24a62125439 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/jni_generator_helper.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/jni_generator_helper.cc
>@@ -11,40 +11,32 @@
> #include "sdk/android/src/jni/jni_generator_helper.h"
> 
> #include "rtc_base/atomicops.h"
>-#include "sdk/android/src/jni/class_loader.h"
>+#include "sdk/android/native_api/jni/class_loader.h"
> 
>-namespace base {
>-namespace android {
>+namespace webrtc {
> 
> // If |atomic_class_id| set, it'll return immediately. Otherwise, it will look
> // up the class and store it. If there's a race, we take care to only store one
> // global reference (and the duplicated effort will happen only once).
> jclass LazyGetClass(JNIEnv* env,
>                     const char* class_name,
>-                    base::subtle::AtomicWord* atomic_class_id) {
>-  static_assert(sizeof(base::subtle::AtomicWord) >= sizeof(jclass),
>-                "AtomicWord can't be smaller than jclass");
>-  base::subtle::AtomicWord value =
>-      rtc::AtomicOps::AcquireLoadPtr(atomic_class_id);
>+                    std::atomic<jclass>* atomic_class_id) {
>+  const jclass value = std::atomic_load(atomic_class_id);
>   if (value)
>-    return reinterpret_cast<jclass>(value);
>-  jclass clazz = static_cast<jclass>(
>-      env->NewGlobalRef(webrtc::jni::GetClass(env, class_name)));
>-  RTC_CHECK(clazz) << class_name;
>-  base::subtle::AtomicWord null_aw = nullptr;
>-  base::subtle::AtomicWord cas_result = rtc::AtomicOps::CompareAndSwapPtr(
>-      atomic_class_id, null_aw,
>-      reinterpret_cast<base::subtle::AtomicWord>(clazz));
>-  if (cas_result == null_aw) {
>+    return value;
>+  webrtc::ScopedJavaGlobalRef<jclass> clazz(webrtc::GetClass(env, class_name));
>+  RTC_CHECK(!clazz.is_null()) << class_name;
>+  jclass cas_result = nullptr;
>+  if (std::atomic_compare_exchange_strong(atomic_class_id, &cas_result,
>+                                          clazz.obj())) {
>     // We sucessfully stored |clazz| in |atomic_class_id|, so we are
>     // intentionally leaking the global ref since it's now stored there.
>-    return clazz;
>+    return clazz.Release();
>   } else {
>     // Some other thread came before us and stored a global pointer in
>     // |atomic_class_id|. Relase our global ref and return the ref from the
>     // other thread.
>-    env->DeleteGlobalRef(clazz);
>-    return reinterpret_cast<jclass>(cas_result);
>+    return cas_result;
>   }
> }
> 
>@@ -56,21 +48,18 @@ jmethodID MethodID::LazyGet(JNIEnv* env,
>                             jclass clazz,
>                             const char* method_name,
>                             const char* jni_signature,
>-                            base::subtle::AtomicWord* atomic_method_id) {
>-  static_assert(sizeof(base::subtle::AtomicWord) >= sizeof(jmethodID),
>-                "AtomicWord can't be smaller than jMethodID");
>-  base::subtle::AtomicWord value =
>-      rtc::AtomicOps::AcquireLoadPtr(atomic_method_id);
>+                            std::atomic<jmethodID>* atomic_method_id) {
>+  const jmethodID value = std::atomic_load(atomic_method_id);
>   if (value)
>-    return reinterpret_cast<jmethodID>(value);
>-  jmethodID id =
>-      (type == MethodID::TYPE_STATIC)
>-          ? webrtc::jni::GetStaticMethodID(env, clazz, method_name,
>-                                           jni_signature)
>-          : webrtc::jni::GetMethodID(env, clazz, method_name, jni_signature);
>-  rtc::AtomicOps::CompareAndSwapPtr(
>-      atomic_method_id, base::subtle::AtomicWord(nullptr),
>-      reinterpret_cast<base::subtle::AtomicWord>(id));
>+    return value;
>+  auto get_method_ptr = type == MethodID::TYPE_STATIC
>+                            ? &JNIEnv::GetStaticMethodID
>+                            : &JNIEnv::GetMethodID;
>+  jmethodID id = (env->*get_method_ptr)(clazz, method_name, jni_signature);
>+  CHECK_EXCEPTION(env) << "error during GetMethodID: " << method_name << ", "
>+                       << jni_signature;
>+  RTC_CHECK(id) << method_name << ", " << jni_signature;
>+  std::atomic_store(atomic_method_id, id);
>   return id;
> }
> 
>@@ -80,14 +69,13 @@ template jmethodID MethodID::LazyGet<MethodID::TYPE_STATIC>(
>     jclass clazz,
>     const char* method_name,
>     const char* jni_signature,
>-    base::subtle::AtomicWord* atomic_method_id);
>+    std::atomic<jmethodID>* atomic_method_id);
> 
> template jmethodID MethodID::LazyGet<MethodID::TYPE_INSTANCE>(
>     JNIEnv* env,
>     jclass clazz,
>     const char* method_name,
>     const char* jni_signature,
>-    base::subtle::AtomicWord* atomic_method_id);
>+    std::atomic<jmethodID>* atomic_method_id);
> 
>-}  // namespace android
>-}  // namespace base
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/jni_generator_helper.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/jni_generator_helper.h
>index 7dec4f7a823..53e1d84127d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/jni_generator_helper.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/jni_generator_helper.h
>@@ -9,14 +9,17 @@
>  */
> // Do not include this file directly. It's intended to be used only by the JNI
> // generation script. We are exporting types in strange namespaces in order to
>-// be compatible with the generated code targeted for Chromium. We are bypassing
>-// the wrapping done by Chromium's JniIntWrapper, JavaRef, and
>-// ScopedJavaLocalRef, and use raw jobjects instead.
>+// be compatible with the generated code targeted for Chromium.
> 
> #ifndef SDK_ANDROID_SRC_JNI_JNI_GENERATOR_HELPER_H_
> #define SDK_ANDROID_SRC_JNI_JNI_GENERATOR_HELPER_H_
> 
>-#include "sdk/android/src/jni/jni_helpers.h"
>+#include <jni.h>
>+#include <atomic>
>+
>+#include "rtc_base/checks.h"
>+#include "sdk/android/native_api/jni/jni_int_wrapper.h"
>+#include "sdk/android/native_api/jni/scoped_java_ref.h"
> 
> #define CHECK_CLAZZ(env, jcaller, clazz, ...) RTC_DCHECK(clazz);
> #define CHECK_NATIVE_PTR(env, jcaller, native_ptr, method_name, ...) \
>@@ -26,55 +29,17 @@
> #define JNI_REGISTRATION_EXPORT __attribute__((visibility("default")))
> #define JNI_GENERATOR_EXPORT extern "C" JNIEXPORT JNICALL
> 
>+#define CHECK_EXCEPTION(jni)        \
>+  RTC_CHECK(!jni->ExceptionCheck()) \
>+      << (jni->ExceptionDescribe(), jni->ExceptionClear(), "")
>+
> namespace jni_generator {
> inline void CheckException(JNIEnv* env) {
>   CHECK_EXCEPTION(env);
> }
> }  // namespace jni_generator
> 
>-namespace {  // NOLINT(build/namespaces)
>-// Bypass JniIntWrapper.
>-// TODO(magjed): Start using Chromium's JniIntWrapper.
>-typedef jint JniIntWrapper;
>-inline jint as_jint(JniIntWrapper wrapper) {
>-  return wrapper;
>-}
>-}  // namespace
>-
>-namespace base {
>-
>-namespace subtle {
>-// This needs to be a type that is big enough to store a jobject/jclass.
>-typedef void* AtomicWord;
>-}  // namespace subtle
>-
>-namespace android {
>-
>-// Implement JavaRef and ScopedJavaLocalRef as a shallow wrapper on top of a
>-// jobject/jclass, with no scoped destruction.
>-// TODO(magjed): Start using Chromium's scoped Java refs.
>-template <typename T>
>-class JavaRef {
>- public:
>-  JavaRef() {}
>-  JavaRef(JNIEnv* env, T obj) : obj_(obj) {}
>-  T obj() const { return obj_; }
>-
>-  // Implicit on purpose.
>-  JavaRef(const T& obj) : obj_(obj) {}  // NOLINT(runtime/explicit)
>-  operator T() const { return obj_; }
>-
>- private:
>-  T obj_;
>-};
>-
>-// TODO(magjed): This looks weird, but it is safe. We don't use DeleteLocalRef
>-// in WebRTC, we use ScopedLocalRefFrame instead. We should probably switch to
>-// using DeleteLocalRef though.
>-template <typename T>
>-using ScopedJavaLocalRef = JavaRef<T>;
>-template <typename T>
>-using JavaParamRef = JavaRef<T>;
>+namespace webrtc {
> 
> // This function will initialize |atomic_class_id| to contain a global ref to
> // the given class, and will return that ref on subsequent calls. The caller is
>@@ -83,7 +48,7 @@ using JavaParamRef = JavaRef<T>;
> // |atomic_method_id|.
> jclass LazyGetClass(JNIEnv* env,
>                     const char* class_name,
>-                    base::subtle::AtomicWord* atomic_class_id);
>+                    std::atomic<jclass>* atomic_class_id);
> 
> // This class is a wrapper for JNIEnv Get(Static)MethodID.
> class MethodID {
>@@ -103,9 +68,21 @@ class MethodID {
>                            jclass clazz,
>                            const char* method_name,
>                            const char* jni_signature,
>-                           base::subtle::AtomicWord* atomic_method_id);
>+                           std::atomic<jmethodID>* atomic_method_id);
> };
> 
>+}  // namespace webrtc
>+
>+// Re-export relevant classes into the namespaces the script expects.
>+namespace base {
>+namespace android {
>+
>+using webrtc::JavaParamRef;
>+using webrtc::JavaRef;
>+using webrtc::ScopedJavaLocalRef;
>+using webrtc::LazyGetClass;
>+using webrtc::MethodID;
>+
> }  // namespace android
> }  // namespace base
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/jni_helpers.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/jni_helpers.cc
>index 694ebead2c9..53399abab18 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/jni_helpers.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/jni_helpers.cc
>@@ -9,365 +9,20 @@
>  */
> #include "sdk/android/src/jni/jni_helpers.h"
> 
>-#include <asm/unistd.h>
>-#include <sys/prctl.h>
>-#include <sys/syscall.h>
>-#include <unistd.h>
> #include <vector>
> 
>-#include "sdk/android/generated_external_classes_jni/jni/Boolean_jni.h"
>-#include "sdk/android/generated_external_classes_jni/jni/Double_jni.h"
>-#include "sdk/android/generated_external_classes_jni/jni/Integer_jni.h"
>-#include "sdk/android/generated_external_classes_jni/jni/Long_jni.h"
>-#include "sdk/android/src/jni/class_loader.h"
>-#include "sdk/android/src/jni/classreferenceholder.h"
>+#include "sdk/android/native_api/jni/java_types.h"
> 
> namespace webrtc {
> namespace jni {
> 
>-static JavaVM* g_jvm = nullptr;
>-
>-static pthread_once_t g_jni_ptr_once = PTHREAD_ONCE_INIT;
>-
>-// Key for per-thread JNIEnv* data.  Non-NULL in threads attached to |g_jvm| by
>-// AttachCurrentThreadIfNeeded(), NULL in unattached threads and threads that
>-// were attached by the JVM because of a Java->native call.
>-static pthread_key_t g_jni_ptr;
>-
>-JavaVM *GetJVM() {
>-  RTC_CHECK(g_jvm) << "JNI_OnLoad failed to run?";
>-  return g_jvm;
>-}
>-
>-// Return a |JNIEnv*| usable on this thread or NULL if this thread is detached.
>-JNIEnv* GetEnv() {
>-  void* env = nullptr;
>-  jint status = g_jvm->GetEnv(&env, JNI_VERSION_1_6);
>-  RTC_CHECK(((env != nullptr) && (status == JNI_OK)) ||
>-            ((env == nullptr) && (status == JNI_EDETACHED)))
>-      << "Unexpected GetEnv return: " << status << ":" << env;
>-  return reinterpret_cast<JNIEnv*>(env);
>-}
>-
>-static void ThreadDestructor(void* prev_jni_ptr) {
>-  // This function only runs on threads where |g_jni_ptr| is non-NULL, meaning
>-  // we were responsible for originally attaching the thread, so are responsible
>-  // for detaching it now.  However, because some JVM implementations (notably
>-  // Oracle's http://goo.gl/eHApYT) also use the pthread_key_create mechanism,
>-  // the JVMs accounting info for this thread may already be wiped out by the
>-  // time this is called. Thus it may appear we are already detached even though
>-  // it was our responsibility to detach!  Oh well.
>-  if (!GetEnv())
>-    return;
>-
>-  RTC_CHECK(GetEnv() == prev_jni_ptr)
>-      << "Detaching from another thread: " << prev_jni_ptr << ":" << GetEnv();
>-  jint status = g_jvm->DetachCurrentThread();
>-  RTC_CHECK(status == JNI_OK) << "Failed to detach thread: " << status;
>-  RTC_CHECK(!GetEnv()) << "Detaching was a successful no-op???";
>-}
>-
>-static void CreateJNIPtrKey() {
>-  RTC_CHECK(!pthread_key_create(&g_jni_ptr, &ThreadDestructor))
>-      << "pthread_key_create";
>-}
>-
>-jint InitGlobalJniVariables(JavaVM *jvm) {
>-  RTC_CHECK(!g_jvm) << "InitGlobalJniVariables!";
>-  g_jvm = jvm;
>-  RTC_CHECK(g_jvm) << "InitGlobalJniVariables handed NULL?";
>-
>-  RTC_CHECK(!pthread_once(&g_jni_ptr_once, &CreateJNIPtrKey)) << "pthread_once";
>-
>-  JNIEnv* jni = nullptr;
>-  if (jvm->GetEnv(reinterpret_cast<void**>(&jni), JNI_VERSION_1_6) != JNI_OK)
>-    return -1;
>-
>-  return JNI_VERSION_1_6;
>-}
>-
>-// Return thread ID as a string.
>-static std::string GetThreadId() {
>-  char buf[21];  // Big enough to hold a kuint64max plus terminating NULL.
>-  RTC_CHECK_LT(snprintf(buf, sizeof(buf), "%ld",
>-                        static_cast<long>(syscall(__NR_gettid))),
>-               sizeof(buf))
>-      << "Thread id is bigger than uint64??";
>-  return std::string(buf);
>-}
>-
>-// Return the current thread's name.
>-static std::string GetThreadName() {
>-  char name[17] = {0};
>-  if (prctl(PR_GET_NAME, name) != 0)
>-    return std::string("<noname>");
>-  return std::string(name);
>-}
>-
>-// Return a |JNIEnv*| usable on this thread.  Attaches to |g_jvm| if necessary.
>-JNIEnv* AttachCurrentThreadIfNeeded() {
>-  JNIEnv* jni = GetEnv();
>-  if (jni)
>-    return jni;
>-  RTC_CHECK(!pthread_getspecific(g_jni_ptr))
>-      << "TLS has a JNIEnv* but not attached?";
>-
>-  std::string name(GetThreadName() + " - " + GetThreadId());
>-  JavaVMAttachArgs args;
>-  args.version = JNI_VERSION_1_6;
>-  args.name = &name[0];
>-  args.group = nullptr;
>-  // Deal with difference in signatures between Oracle's jni.h and Android's.
>-#ifdef _JAVASOFT_JNI_H_  // Oracle's jni.h violates the JNI spec!
>-  void* env = nullptr;
>-#else
>-  JNIEnv* env = nullptr;
>-#endif
>-  RTC_CHECK(!g_jvm->AttachCurrentThread(&env, &args))
>-      << "Failed to attach thread";
>-  RTC_CHECK(env) << "AttachCurrentThread handed back NULL!";
>-  jni = reinterpret_cast<JNIEnv*>(env);
>-  RTC_CHECK(!pthread_setspecific(g_jni_ptr, jni)) << "pthread_setspecific";
>-  return jni;
>-}
>-
>-// Return a |jlong| that will correctly convert back to |ptr|.  This is needed
>-// because the alternative (of silently passing a 32-bit pointer to a vararg
>-// function expecting a 64-bit param) picks up garbage in the high 32 bits.
>-jlong jlongFromPointer(void* ptr) {
>-  static_assert(sizeof(intptr_t) <= sizeof(jlong),
>-                "Time to rethink the use of jlongs");
>-  // Going through intptr_t to be obvious about the definedness of the
>-  // conversion from pointer to integral type.  intptr_t to jlong is a standard
>-  // widening by the static_assert above.
>-  jlong ret = reinterpret_cast<intptr_t>(ptr);
>-  RTC_DCHECK(reinterpret_cast<void*>(ret) == ptr);
>-  return ret;
>-}
>-
>-// JNIEnv-helper methods that RTC_CHECK success: no Java exception thrown and
>-// found object/class/method/field is non-null.
>-jmethodID GetMethodID(
>-    JNIEnv* jni, jclass c, const std::string& name, const char* signature) {
>-  jmethodID m = jni->GetMethodID(c, name.c_str(), signature);
>-  CHECK_EXCEPTION(jni) << "error during GetMethodID: " << name << ", "
>-                       << signature;
>-  RTC_CHECK(m) << name << ", " << signature;
>-  return m;
>-}
>-
>-jmethodID GetStaticMethodID(
>-    JNIEnv* jni, jclass c, const char* name, const char* signature) {
>-  jmethodID m = jni->GetStaticMethodID(c, name, signature);
>-  CHECK_EXCEPTION(jni) << "error during GetStaticMethodID: " << name << ", "
>-                       << signature;
>-  RTC_CHECK(m) << name << ", " << signature;
>-  return m;
>-}
>-
>-jfieldID GetFieldID(
>-    JNIEnv* jni, jclass c, const char* name, const char* signature) {
>-  jfieldID f = jni->GetFieldID(c, name, signature);
>-  CHECK_EXCEPTION(jni) << "error during GetFieldID";
>-  RTC_CHECK(f) << name << ", " << signature;
>-  return f;
>-}
>-
>-jfieldID GetStaticFieldID(JNIEnv* jni,
>-                          jclass c,
>-                          const char* name,
>-                          const char* signature) {
>-  jfieldID f = jni->GetStaticFieldID(c, name, signature);
>-  CHECK_EXCEPTION(jni) << "error during GetStaticFieldID";
>-  RTC_CHECK(f) << name << ", " << signature;
>-  return f;
>-}
>-
>-jclass GetObjectClass(JNIEnv* jni, jobject object) {
>-  jclass c = jni->GetObjectClass(object);
>-  CHECK_EXCEPTION(jni) << "error during GetObjectClass";
>-  RTC_CHECK(c) << "GetObjectClass returned NULL";
>-  return c;
>-}
>-
>-jobject GetObjectField(JNIEnv* jni, jobject object, jfieldID id) {
>-  jobject o = jni->GetObjectField(object, id);
>-  CHECK_EXCEPTION(jni) << "error during GetObjectField";
>-  RTC_CHECK(!IsNull(jni, o)) << "GetObjectField returned NULL";
>-  return o;
>-}
>-
>-jobject GetStaticObjectField(JNIEnv* jni, jclass c, jfieldID id) {
>-  jobject o = jni->GetStaticObjectField(c, id);
>-  CHECK_EXCEPTION(jni) << "error during GetStaticObjectField";
>-  RTC_CHECK(!IsNull(jni, o)) << "GetStaticObjectField returned NULL";
>-  return o;
>-}
>-
>-jobject GetNullableObjectField(JNIEnv* jni, jobject object, jfieldID id) {
>-  jobject o = jni->GetObjectField(object, id);
>-  CHECK_EXCEPTION(jni) << "error during GetObjectField";
>-  return o;
>-}
>-
>-jstring GetStringField(JNIEnv* jni, jobject object, jfieldID id) {
>-  return static_cast<jstring>(GetObjectField(jni, object, id));
>-}
>-
>-jlong GetLongField(JNIEnv* jni, jobject object, jfieldID id) {
>-  jlong l = jni->GetLongField(object, id);
>-  CHECK_EXCEPTION(jni) << "error during GetLongField";
>-  return l;
>-}
>-
>-jint GetIntField(JNIEnv* jni, jobject object, jfieldID id) {
>-  jint i = jni->GetIntField(object, id);
>-  CHECK_EXCEPTION(jni) << "error during GetIntField";
>-  return i;
>-}
>-
>-bool GetBooleanField(JNIEnv* jni, jobject object, jfieldID id) {
>-  jboolean b = jni->GetBooleanField(object, id);
>-  CHECK_EXCEPTION(jni) << "error during GetBooleanField";
>-  return b;
>-}
>-
>-bool IsNull(JNIEnv* jni, jobject obj) {
>-  return jni->IsSameObject(obj, nullptr);
>-}
>-
>-// Given a jstring, reinterprets it to a new native string.
>-std::string JavaToStdString(JNIEnv* jni, const jstring& j_string) {
>-  // Invoke String.getBytes(String charsetName) method to convert |j_string|
>-  // to a byte array.
>-  const jclass string_class = GetObjectClass(jni, j_string);
>-  const jmethodID get_bytes =
>-      GetMethodID(jni, string_class, "getBytes", "(Ljava/lang/String;)[B");
>-  const jstring charset_name = jni->NewStringUTF("ISO-8859-1");
>-  CHECK_EXCEPTION(jni) << "error during NewStringUTF";
>-  const jbyteArray j_byte_array =
>-      (jbyteArray)jni->CallObjectMethod(j_string, get_bytes, charset_name);
>-  CHECK_EXCEPTION(jni) << "error during CallObjectMethod";
>-
>-  const size_t len = jni->GetArrayLength(j_byte_array);
>-  CHECK_EXCEPTION(jni) << "error during GetArrayLength";
>-  std::vector<char> buf(len);
>-  jni->GetByteArrayRegion(j_byte_array, 0, len,
>-                          reinterpret_cast<jbyte*>(&buf[0]));
>-  CHECK_EXCEPTION(jni) << "error during GetByteArrayRegion";
>-
>-  return std::string(buf.begin(), buf.end());
>-}
>-
>-// Given a list of jstrings, reinterprets it to a new vector of native strings.
>-std::vector<std::string> JavaToStdVectorStrings(JNIEnv* jni, jobject list) {
>-  std::vector<std::string> converted_list;
>-  if (list != nullptr) {
>-    for (jobject str : Iterable(jni, list)) {
>-      converted_list.push_back(
>-          JavaToStdString(jni, reinterpret_cast<jstring>(str)));
>-    }
>-  }
>-  return converted_list;
>-}
>-
>-rtc::Optional<int32_t> JavaToNativeOptionalInt(JNIEnv* jni, jobject integer) {
>-  if (IsNull(jni, integer))
>-    return rtc::nullopt;
>-  return JNI_Integer::Java_Integer_intValue(jni, integer);
>-}
>-
>-jobject NativeToJavaBoolean(JNIEnv* env, bool b) {
>-  return JNI_Boolean::Java_Boolean_ConstructorJLB_Z(env, b);
>-}
>-
>-jobject NativeToJavaInteger(JNIEnv* jni, int32_t i) {
>-  return JNI_Integer::Java_Integer_ConstructorJLI_I(jni, i);
>-}
>-
>-jobject NativeToJavaLong(JNIEnv* env, int64_t u) {
>-  return JNI_Long::Java_Long_ConstructorJLLO_J(env, u);
>-}
>-
>-jobject NativeToJavaDouble(JNIEnv* env, double d) {
>-  return JNI_Double::Java_Double_ConstructorJLD_D(env, d);
>-}
>-
>-jstring NativeToJavaString(JNIEnv* jni, const std::string& native) {
>-  jstring jstr = jni->NewStringUTF(native.c_str());
>-  CHECK_EXCEPTION(jni) << "error during NewStringUTF";
>-  return jstr;
>-}
>-
>-jobject NativeToJavaInteger(JNIEnv* jni,
>-                            const rtc::Optional<int32_t>& optional_int) {
>-  return optional_int ? NativeToJavaInteger(jni, *optional_int) : nullptr;
>-}
>-
>-// Return the (singleton) Java Enum object corresponding to |index|;
>-jobject JavaEnumFromIndex(JNIEnv* jni, jclass state_class,
>-                          const std::string& state_class_name, int index) {
>-  jmethodID state_values_id = GetStaticMethodID(
>-      jni, state_class, "values", ("()[L" + state_class_name  + ";").c_str());
>-  jobjectArray state_values = static_cast<jobjectArray>(
>-      jni->CallStaticObjectMethod(state_class, state_values_id));
>-  CHECK_EXCEPTION(jni) << "error during CallStaticObjectMethod";
>-  jobject ret = jni->GetObjectArrayElement(state_values, index);
>-  CHECK_EXCEPTION(jni) << "error during GetObjectArrayElement";
>-  return ret;
>-}
>-
>-jobject JavaEnumFromIndexAndClassName(JNIEnv* jni,
>-                                      const std::string& state_class_fragment,
>-                                      int index) {
>-  const std::string state_class = "org/webrtc/" + state_class_fragment;
>-  return JavaEnumFromIndex(jni, FindClass(jni, state_class.c_str()),
>-                           state_class, index);
>-}
>-
>-std::string GetJavaEnumName(JNIEnv* jni, jobject j_enum) {
>-  jclass enum_class = GetClass(jni, "java/lang/Enum");
>-  jmethodID nameMethod =
>-      GetMethodID(jni, enum_class, "name", "()Ljava/lang/String;");
>-  jstring name =
>-      reinterpret_cast<jstring>(jni->CallObjectMethod(j_enum, nameMethod));
>-  CHECK_EXCEPTION(jni);
>-  return JavaToStdString(jni, name);
>-}
>-
>-std::map<std::string, std::string> JavaToStdMapStrings(JNIEnv* jni,
>-                                                       jobject j_map) {
>-  jclass map_class = jni->FindClass("java/util/Map");
>-  jclass set_class = jni->FindClass("java/util/Set");
>-  jclass iterator_class = jni->FindClass("java/util/Iterator");
>-  jclass entry_class = jni->FindClass("java/util/Map$Entry");
>-  jmethodID entry_set_method =
>-      jni->GetMethodID(map_class, "entrySet", "()Ljava/util/Set;");
>-  jmethodID iterator_method =
>-      jni->GetMethodID(set_class, "iterator", "()Ljava/util/Iterator;");
>-  jmethodID has_next_method =
>-      jni->GetMethodID(iterator_class, "hasNext", "()Z");
>-  jmethodID next_method =
>-      jni->GetMethodID(iterator_class, "next", "()Ljava/lang/Object;");
>-  jmethodID get_key_method =
>-      jni->GetMethodID(entry_class, "getKey", "()Ljava/lang/Object;");
>-  jmethodID get_value_method =
>-      jni->GetMethodID(entry_class, "getValue", "()Ljava/lang/Object;");
>-
>-  jobject j_entry_set = jni->CallObjectMethod(j_map, entry_set_method);
>-  jobject j_iterator = jni->CallObjectMethod(j_entry_set, iterator_method);
>-
>-  std::map<std::string, std::string> result;
>-  while (jni->CallBooleanMethod(j_iterator, has_next_method)) {
>-    jobject j_entry = jni->CallObjectMethod(j_iterator, next_method);
>-    jstring j_key =
>-        static_cast<jstring>(jni->CallObjectMethod(j_entry, get_key_method));
>-    jstring j_value =
>-        static_cast<jstring>(jni->CallObjectMethod(j_entry, get_value_method));
>-    result[JavaToStdString(jni, j_key)] = JavaToStdString(jni, j_value);
>-  }
>-
>-  return result;
>+ScopedJavaLocalRef<jobject> NewDirectByteBuffer(JNIEnv* env,
>+                                                void* address,
>+                                                jlong capacity) {
>+  ScopedJavaLocalRef<jobject> buffer(
>+      env, env->NewDirectByteBuffer(address, capacity));
>+  CHECK_EXCEPTION(env) << "error NewDirectByteBuffer";
>+  return buffer;
> }
> 
> jobject NewGlobalRef(JNIEnv* jni, jobject o) {
>@@ -392,114 +47,5 @@ ScopedLocalRefFrame::~ScopedLocalRefFrame() {
>   jni_->PopLocalFrame(nullptr);
> }
> 
>-// Creates an iterator representing the end of any collection.
>-Iterable::Iterator::Iterator() : iterator_(nullptr) {}
>-
>-// Creates an iterator pointing to the beginning of the specified collection.
>-Iterable::Iterator::Iterator(JNIEnv* jni, jobject iterable) : jni_(jni) {
>-  jclass j_class = GetObjectClass(jni, iterable);
>-  jmethodID iterator_id =
>-      GetMethodID(jni, j_class, "iterator", "()Ljava/util/Iterator;");
>-  iterator_ = jni->CallObjectMethod(iterable, iterator_id);
>-  CHECK_EXCEPTION(jni) << "error during CallObjectMethod";
>-  RTC_CHECK(iterator_ != nullptr);
>-
>-  jclass iterator_class = GetObjectClass(jni, iterator_);
>-  has_next_id_ = GetMethodID(jni, iterator_class, "hasNext", "()Z");
>-  next_id_ = GetMethodID(jni, iterator_class, "next", "()Ljava/lang/Object;");
>-  remove_id_ = GetMethodID(jni, iterator_class, "remove", "()V");
>-
>-  // Start at the first element in the collection.
>-  ++(*this);
>-}
>-
>-// Move constructor - necessary to be able to return iterator types from
>-// functions.
>-Iterable::Iterator::Iterator(Iterator&& other)
>-    : jni_(std::move(other.jni_)),
>-      iterator_(std::move(other.iterator_)),
>-      value_(std::move(other.value_)),
>-      has_next_id_(std::move(other.has_next_id_)),
>-      next_id_(std::move(other.next_id_)),
>-      thread_checker_(std::move(other.thread_checker_)){};
>-
>-// Advances the iterator one step.
>-Iterable::Iterator& Iterable::Iterator::operator++() {
>-  RTC_CHECK(thread_checker_.CalledOnValidThread());
>-  if (AtEnd()) {
>-    // Can't move past the end.
>-    return *this;
>-  }
>-  bool has_next = jni_->CallBooleanMethod(iterator_, has_next_id_);
>-  CHECK_EXCEPTION(jni_) << "error during CallBooleanMethod";
>-  if (!has_next) {
>-    iterator_ = nullptr;
>-    value_ = nullptr;
>-    return *this;
>-  }
>-
>-  value_ = jni_->CallObjectMethod(iterator_, next_id_);
>-  CHECK_EXCEPTION(jni_) << "error during CallObjectMethod";
>-  return *this;
>-}
>-
>-void Iterable::Iterator::Remove() {
>-  jni_->CallVoidMethod(iterator_, remove_id_);
>-  CHECK_EXCEPTION(jni_) << "error during CallVoidMethod";
>-}
>-
>-// Provides a way to compare the iterator with itself and with the end iterator.
>-// Note: all other comparison results are undefined, just like for C++ input
>-// iterators.
>-bool Iterable::Iterator::operator==(const Iterable::Iterator& other) {
>-  // Two different active iterators should never be compared.
>-  RTC_DCHECK(this == &other || AtEnd() || other.AtEnd());
>-  return AtEnd() == other.AtEnd();
>-}
>-
>-jobject Iterable::Iterator::operator*() {
>-  RTC_CHECK(!AtEnd());
>-  return value_;
>-}
>-
>-bool Iterable::Iterator::AtEnd() const {
>-  RTC_CHECK(thread_checker_.CalledOnValidThread());
>-  return jni_ == nullptr || IsNull(jni_, iterator_);
>-}
>-
>-jobjectArray NativeToJavaIntegerArray(JNIEnv* env,
>-                                      const std::vector<int32_t>& container) {
>-  jobject (*convert_function)(JNIEnv*, int32_t) = &NativeToJavaInteger;
>-  return NativeToJavaObjectArray(env, container, java_lang_Integer_clazz(env),
>-                                 convert_function);
>-}
>-
>-jobjectArray NativeToJavaBooleanArray(JNIEnv* env,
>-                                      const std::vector<bool>& container) {
>-  return NativeToJavaObjectArray(env, container, java_lang_Boolean_clazz(env),
>-                                 &NativeToJavaBoolean);
>-}
>-
>-jobjectArray NativeToJavaDoubleArray(JNIEnv* env,
>-                                     const std::vector<double>& container) {
>-  return NativeToJavaObjectArray(env, container, java_lang_Double_clazz(env),
>-                                 &NativeToJavaDouble);
>-}
>-
>-jobjectArray NativeToJavaLongArray(JNIEnv* env,
>-                                   const std::vector<int64_t>& container) {
>-  return NativeToJavaObjectArray(env, container, java_lang_Long_clazz(env),
>-                                 &NativeToJavaLong);
>-}
>-
>-jobjectArray NativeToJavaStringArray(
>-    JNIEnv* env,
>-    const std::vector<std::string>& container) {
>-  // TODO(magjed): Remove this class when we can generate it from String.class
>-  // directly (the script currently chokes on that class).
>-  return NativeToJavaObjectArray(
>-      env, container, FindClass(env, "java/lang/String"), &NativeToJavaString);
>-}
>-
> }  // namespace jni
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/jni_helpers.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/jni_helpers.h
>index bfef3a88026..38b0d7edf42 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/jni_helpers.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/jni_helpers.h
>@@ -15,29 +15,11 @@
> #define SDK_ANDROID_SRC_JNI_JNI_HELPERS_H_
> 
> #include <jni.h>
>-#include <map>
> #include <string>
>-#include <vector>
> 
>-#include "api/optional.h"
>-#include "rtc_base/checks.h"
>-#include "rtc_base/constructormagic.h"
>-#include "rtc_base/refcount.h"
>-#include "rtc_base/thread_checker.h"
>-
>-// Abort the process if |jni| has a Java exception pending.
>-// This macros uses the comma operator to execute ExceptionDescribe
>-// and ExceptionClear ignoring their return values and sending ""
>-// to the error stream.
>-#define CHECK_EXCEPTION(jni)        \
>-  RTC_CHECK(!jni->ExceptionCheck()) \
>-      << (jni->ExceptionDescribe(), jni->ExceptionClear(), "")
>-
>-// Helper that calls ptr->Release() and aborts the process with a useful
>-// message if that didn't actually delete *ptr because of extra refcounts.
>-#define CHECK_RELEASE(ptr)                                                   \
>-  RTC_CHECK((ptr)->Release() == rtc::RefCountReleaseStatus::kDroppedLastRef) \
>-      << "Unexpected refcount."
>+#include "sdk/android/native_api/jni/java_types.h"
>+#include "sdk/android/native_api/jni/scoped_java_ref.h"
>+#include "sdk/android/src/jni/jvm.h"
> 
> // Convenience macro defining JNI-accessible methods in the org.webrtc package.
> // Eliminates unnecessary boilerplate and line-wraps, reducing visual clutter.
>@@ -47,91 +29,17 @@
> namespace webrtc {
> namespace jni {
> 
>-jint InitGlobalJniVariables(JavaVM *jvm);
>-
>-// Return a |JNIEnv*| usable on this thread or NULL if this thread is detached.
>-JNIEnv* GetEnv();
>-
>-JavaVM *GetJVM();
>-
>-// Return a |JNIEnv*| usable on this thread.  Attaches to |g_jvm| if necessary.
>-JNIEnv* AttachCurrentThreadIfNeeded();
>-
>-// Return a |jlong| that will correctly convert back to |ptr|.  This is needed
>-// because the alternative (of silently passing a 32-bit pointer to a vararg
>-// function expecting a 64-bit param) picks up garbage in the high 32 bits.
>-jlong jlongFromPointer(void* ptr);
>-
>-// JNIEnv-helper methods that RTC_CHECK success: no Java exception thrown and
>-// found object/class/method/field is non-null.
>-jmethodID GetMethodID(
>-    JNIEnv* jni, jclass c, const std::string& name, const char* signature);
>-
>-jmethodID GetStaticMethodID(
>-    JNIEnv* jni, jclass c, const char* name, const char* signature);
>-
>-jfieldID GetFieldID(JNIEnv* jni, jclass c, const char* name,
>-                    const char* signature);
>-
>-jfieldID GetStaticFieldID(JNIEnv* jni,
>-                          jclass c,
>-                          const char* name,
>-                          const char* signature);
>-
>-jclass GetObjectClass(JNIEnv* jni, jobject object);
>-
>-// Throws an exception if the object field is null.
>-jobject GetObjectField(JNIEnv* jni, jobject object, jfieldID id);
>-
>-jobject GetStaticObjectField(JNIEnv* jni, jclass c, jfieldID id);
>-
>-jobject GetNullableObjectField(JNIEnv* jni, jobject object, jfieldID id);
>-
>-jstring GetStringField(JNIEnv* jni, jobject object, jfieldID id);
>-
>-jlong GetLongField(JNIEnv* jni, jobject object, jfieldID id);
>-
>-jint GetIntField(JNIEnv* jni, jobject object, jfieldID id);
>-
>-bool GetBooleanField(JNIEnv* jni, jobject object, jfieldID id);
>-
>-// Returns true if |obj| == null in Java.
>-bool IsNull(JNIEnv* jni, jobject obj);
>-
>-// Given a (UTF-16) jstring return a new UTF-8 native string.
>-std::string JavaToStdString(JNIEnv* jni, const jstring& j_string);
>-
>-// Given a List of (UTF-16) jstrings
>-// return a new vector of UTF-8 native strings.
>-std::vector<std::string> JavaToStdVectorStrings(JNIEnv* jni, jobject list);
>-
>-rtc::Optional<int32_t> JavaToNativeOptionalInt(JNIEnv* jni, jobject integer);
>-
>-jobject NativeToJavaBoolean(JNIEnv* env, bool b);
>-jobject NativeToJavaInteger(JNIEnv* jni, int32_t i);
>-jobject NativeToJavaLong(JNIEnv* env, int64_t u);
>-jobject NativeToJavaDouble(JNIEnv* env, double d);
>-// Given a UTF-8 encoded |native| string return a new (UTF-16) jstring.
>-jstring NativeToJavaString(JNIEnv* jni, const std::string& native);
>-jobject NativeToJavaInteger(JNIEnv* jni,
>-                            const rtc::Optional<int32_t>& optional_int);
>-
>-// Return the (singleton) Java Enum object corresponding to |index|;
>-jobject JavaEnumFromIndex(JNIEnv* jni, jclass state_class,
>-                          const std::string& state_class_name, int index);
>-
>-// Return the (singleton) Java Enum object corresponding to |index|;
>-// |state_class_fragment| is something like "MediaSource$State".
>-jobject JavaEnumFromIndexAndClassName(JNIEnv* jni,
>-                                      const std::string& state_class_fragment,
>-                                      int index);
>+// TODO(sakal): Remove once clients have migrated.
>+using ::webrtc::JavaToStdMapStrings;
> 
>-// Parses Map<String, String> to std::map<std::string, std::string>.
>-std::map<std::string, std::string> JavaToStdMapStrings(JNIEnv* jni,
>-                                                       jobject j_map);
>+// Deprecated, use NativeToJavaPointer.
>+inline long jlongFromPointer(void* ptr) {
>+  return NativeToJavaPointer(ptr);
>+}
> 
>-// Returns the name of a Java enum.
>-std::string GetJavaEnumName(JNIEnv* jni, jobject j_enum);
>+ScopedJavaLocalRef<jobject> NewDirectByteBuffer(JNIEnv* env,
>+                                                void* address,
>+                                                jlong capacity);
> 
> jobject NewGlobalRef(JNIEnv* jni, jobject o);
> 
>@@ -149,124 +57,13 @@ class ScopedLocalRefFrame {
>   JNIEnv* jni_;
> };
> 
>-// Scoped holder for global Java refs.
>-template<class T>  // T is jclass, jobject, jintArray, etc.
>-class ScopedGlobalRef {
>- public:
>-  ScopedGlobalRef(JNIEnv* jni, T obj)
>-      : obj_(static_cast<T>(jni->NewGlobalRef(obj))) {}
>-  ~ScopedGlobalRef() {
>-    DeleteGlobalRef(AttachCurrentThreadIfNeeded(), obj_);
>-  }
>-  T operator*() const {
>-    return obj_;
>-  }
>- private:
>-  T obj_;
>-};
>-
>-// Provides a convenient way to iterate over a Java Iterable using the
>-// C++ range-for loop.
>-// E.g. for (jobject value : Iterable(jni, j_iterable)) { ... }
>-// Note: Since Java iterators cannot be duplicated, the iterator class is not
>-// copyable to prevent creating multiple C++ iterators that refer to the same
>-// Java iterator.
>-class Iterable {
>- public:
>-  Iterable(JNIEnv* jni, jobject iterable) : jni_(jni), iterable_(iterable) {}
>-
>-  class Iterator {
>-   public:
>-    // Creates an iterator representing the end of any collection.
>-    Iterator();
>-    // Creates an iterator pointing to the beginning of the specified
>-    // collection.
>-    Iterator(JNIEnv* jni, jobject iterable);
>-
>-    // Move constructor - necessary to be able to return iterator types from
>-    // functions.
>-    Iterator(Iterator&& other);
>-
>-    // Move assignment should not be used.
>-    Iterator& operator=(Iterator&&) = delete;
>-
>-    // Advances the iterator one step.
>-    Iterator& operator++();
>-
>-    // Removes the element the iterator is pointing to. Must still advance the
>-    // iterator afterwards.
>-    void Remove();
>-
>-    // Provides a way to compare the iterator with itself and with the end
>-    // iterator.
>-    // Note: all other comparison results are undefined, just like for C++ input
>-    // iterators.
>-    bool operator==(const Iterator& other);
>-    bool operator!=(const Iterator& other) { return !(*this == other); }
>-    jobject operator*();
>-
>-   private:
>-    bool AtEnd() const;
>-
>-    JNIEnv* jni_ = nullptr;
>-    jobject iterator_ = nullptr;
>-    jobject value_ = nullptr;
>-    jmethodID has_next_id_ = nullptr;
>-    jmethodID next_id_ = nullptr;
>-    jmethodID remove_id_ = nullptr;
>-    rtc::ThreadChecker thread_checker_;
>-
>-    RTC_DISALLOW_COPY_AND_ASSIGN(Iterator);
>-  };
>-
>-  Iterable::Iterator begin() { return Iterable::Iterator(jni_, iterable_); }
>-  Iterable::Iterator end() { return Iterable::Iterator(); }
>-
>- private:
>-  JNIEnv* jni_;
>-  jobject iterable_;
>-
>-  RTC_DISALLOW_COPY_AND_ASSIGN(Iterable);
>-};
>-
>-// Helper function for converting std::vector<T> into a Java array.
>-template <typename T, typename Convert>
>-jobjectArray NativeToJavaObjectArray(JNIEnv* env,
>-                                     const std::vector<T>& container,
>-                                     jclass clazz,
>-                                     Convert convert) {
>-  jobjectArray j_container =
>-      env->NewObjectArray(container.size(), clazz, nullptr);
>-  int i = 0;
>-  for (const T& element : container) {
>-    jobject j_element = convert(env, element);
>-    env->SetObjectArrayElement(j_container, i, j_element);
>-    // Delete local ref immediately since we might create a lot of local
>-    // references in this loop.
>-    env->DeleteLocalRef(j_element);
>-    ++i;
>-  }
>-  return j_container;
>-}
>-
>-jobjectArray NativeToJavaIntegerArray(JNIEnv* env,
>-                                      const std::vector<int32_t>& container);
>-jobjectArray NativeToJavaBooleanArray(JNIEnv* env,
>-                                      const std::vector<bool>& container);
>-jobjectArray NativeToJavaLongArray(JNIEnv* env,
>-                                   const std::vector<int64_t>& container);
>-jobjectArray NativeToJavaDoubleArray(JNIEnv* env,
>-                                     const std::vector<double>& container);
>-jobjectArray NativeToJavaStringArray(JNIEnv* env,
>-                                     const std::vector<std::string>& container);
>-
> }  // namespace jni
> }  // namespace webrtc
> 
> // TODO(magjed): Remove once external clients are updated.
> namespace webrtc_jni {
> 
>-using webrtc::jni::AttachCurrentThreadIfNeeded;
>+using webrtc::AttachCurrentThreadIfNeeded;
> using webrtc::jni::InitGlobalJniVariables;
> 
> }  // namespace webrtc_jni
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/jni_onload.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/jni_onload.cc
>index 81cc4889c28..83c61d6a08b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/jni_onload.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/jni_onload.cc
>@@ -19,7 +19,7 @@
> namespace webrtc {
> namespace jni {
> 
>-extern "C" jint JNIEXPORT JNICALL JNI_OnLoad(JavaVM *jvm, void *reserved) {
>+extern "C" jint JNIEXPORT JNICALL JNI_OnLoad(JavaVM* jvm, void* reserved) {
>   jint ret = InitGlobalJniVariables(jvm);
>   RTC_DCHECK_GE(ret, 0);
>   if (ret < 0)
>@@ -31,7 +31,7 @@ extern "C" jint JNIEXPORT JNICALL JNI_OnLoad(JavaVM *jvm, void *reserved) {
>   return ret;
> }
> 
>-extern "C" void JNIEXPORT JNICALL JNI_OnUnLoad(JavaVM *jvm, void *reserved) {
>+extern "C" void JNIEXPORT JNICALL JNI_OnUnLoad(JavaVM* jvm, void* reserved) {
>   FreeGlobalClassReferenceHolder();
>   RTC_CHECK(rtc::CleanupSSL()) << "Failed to CleanupSSL()";
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/jvm.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/jvm.cc
>new file mode 100644
>index 00000000000..eaa4f672124
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/jvm.cc
>@@ -0,0 +1,133 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "sdk/android/src/jni/jvm.h"
>+
>+#include <asm/unistd.h>
>+#include <pthread.h>
>+#include <sys/prctl.h>
>+#include <sys/syscall.h>
>+#include <unistd.h>
>+
>+#include <string>
>+
>+#include "rtc_base/checks.h"
>+
>+namespace webrtc {
>+namespace jni {
>+
>+static JavaVM* g_jvm = nullptr;
>+
>+static pthread_once_t g_jni_ptr_once = PTHREAD_ONCE_INIT;
>+
>+// Key for per-thread JNIEnv* data.  Non-NULL in threads attached to |g_jvm| by
>+// AttachCurrentThreadIfNeeded(), NULL in unattached threads and threads that
>+// were attached by the JVM because of a Java->native call.
>+static pthread_key_t g_jni_ptr;
>+
>+JavaVM* GetJVM() {
>+  RTC_CHECK(g_jvm) << "JNI_OnLoad failed to run?";
>+  return g_jvm;
>+}
>+
>+// Return a |JNIEnv*| usable on this thread or NULL if this thread is detached.
>+JNIEnv* GetEnv() {
>+  void* env = nullptr;
>+  jint status = g_jvm->GetEnv(&env, JNI_VERSION_1_6);
>+  RTC_CHECK(((env != nullptr) && (status == JNI_OK)) ||
>+            ((env == nullptr) && (status == JNI_EDETACHED)))
>+      << "Unexpected GetEnv return: " << status << ":" << env;
>+  return reinterpret_cast<JNIEnv*>(env);
>+}
>+
>+static void ThreadDestructor(void* prev_jni_ptr) {
>+  // This function only runs on threads where |g_jni_ptr| is non-NULL, meaning
>+  // we were responsible for originally attaching the thread, so are responsible
>+  // for detaching it now.  However, because some JVM implementations (notably
>+  // Oracle's http://goo.gl/eHApYT) also use the pthread_key_create mechanism,
>+  // the JVMs accounting info for this thread may already be wiped out by the
>+  // time this is called. Thus it may appear we are already detached even though
>+  // it was our responsibility to detach!  Oh well.
>+  if (!GetEnv())
>+    return;
>+
>+  RTC_CHECK(GetEnv() == prev_jni_ptr)
>+      << "Detaching from another thread: " << prev_jni_ptr << ":" << GetEnv();
>+  jint status = g_jvm->DetachCurrentThread();
>+  RTC_CHECK(status == JNI_OK) << "Failed to detach thread: " << status;
>+  RTC_CHECK(!GetEnv()) << "Detaching was a successful no-op???";
>+}
>+
>+static void CreateJNIPtrKey() {
>+  RTC_CHECK(!pthread_key_create(&g_jni_ptr, &ThreadDestructor))
>+      << "pthread_key_create";
>+}
>+
>+jint InitGlobalJniVariables(JavaVM* jvm) {
>+  RTC_CHECK(!g_jvm) << "InitGlobalJniVariables!";
>+  g_jvm = jvm;
>+  RTC_CHECK(g_jvm) << "InitGlobalJniVariables handed NULL?";
>+
>+  RTC_CHECK(!pthread_once(&g_jni_ptr_once, &CreateJNIPtrKey)) << "pthread_once";
>+
>+  JNIEnv* jni = nullptr;
>+  if (jvm->GetEnv(reinterpret_cast<void**>(&jni), JNI_VERSION_1_6) != JNI_OK)
>+    return -1;
>+
>+  return JNI_VERSION_1_6;
>+}
>+
>+// Return thread ID as a string.
>+static std::string GetThreadId() {
>+  char buf[21];  // Big enough to hold a kuint64max plus terminating NULL.
>+  RTC_CHECK_LT(snprintf(buf, sizeof(buf), "%ld",
>+                        static_cast<long>(syscall(__NR_gettid))),
>+               sizeof(buf))
>+      << "Thread id is bigger than uint64??";
>+  return std::string(buf);
>+}
>+
>+// Return the current thread's name.
>+static std::string GetThreadName() {
>+  char name[17] = {0};
>+  if (prctl(PR_GET_NAME, name) != 0)
>+    return std::string("<noname>");
>+  return std::string(name);
>+}
>+
>+// Return a |JNIEnv*| usable on this thread.  Attaches to |g_jvm| if necessary.
>+JNIEnv* AttachCurrentThreadIfNeeded() {
>+  JNIEnv* jni = GetEnv();
>+  if (jni)
>+    return jni;
>+  RTC_CHECK(!pthread_getspecific(g_jni_ptr))
>+      << "TLS has a JNIEnv* but not attached?";
>+
>+  std::string name(GetThreadName() + " - " + GetThreadId());
>+  JavaVMAttachArgs args;
>+  args.version = JNI_VERSION_1_6;
>+  args.name = &name[0];
>+  args.group = nullptr;
>+// Deal with difference in signatures between Oracle's jni.h and Android's.
>+#ifdef _JAVASOFT_JNI_H_  // Oracle's jni.h violates the JNI spec!
>+  void* env = nullptr;
>+#else
>+  JNIEnv* env = nullptr;
>+#endif
>+  RTC_CHECK(!g_jvm->AttachCurrentThread(&env, &args))
>+      << "Failed to attach thread";
>+  RTC_CHECK(env) << "AttachCurrentThread handed back NULL!";
>+  jni = reinterpret_cast<JNIEnv*>(env);
>+  RTC_CHECK(!pthread_setspecific(g_jni_ptr, jni)) << "pthread_setspecific";
>+  return jni;
>+}
>+
>+}  // namespace jni
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/jvm.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/jvm.h
>new file mode 100644
>index 00000000000..069a19a445b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/jvm.h
>@@ -0,0 +1,32 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef SDK_ANDROID_SRC_JNI_JVM_H_
>+#define SDK_ANDROID_SRC_JNI_JVM_H_
>+
>+#include <jni.h>
>+
>+namespace webrtc {
>+namespace jni {
>+
>+jint InitGlobalJniVariables(JavaVM* jvm);
>+
>+// Return a |JNIEnv*| usable on this thread or NULL if this thread is detached.
>+JNIEnv* GetEnv();
>+
>+JavaVM* GetJVM();
>+
>+// Return a |JNIEnv*| usable on this thread.  Attaches to |g_jvm| if necessary.
>+JNIEnv* AttachCurrentThreadIfNeeded();
>+
>+}  // namespace jni
>+}  // namespace webrtc
>+
>+#endif  // SDK_ANDROID_SRC_JNI_JVM_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/logging/logsink.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/logging/logsink.cc
>new file mode 100644
>index 00000000000..cfa3c60abb2
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/logging/logsink.cc
>@@ -0,0 +1,35 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#include "sdk/android/src/jni/logging/logsink.h"
>+
>+#include "sdk/android/generated_logging_jni/jni/JNILogging_jni.h"
>+
>+namespace webrtc {
>+namespace jni {
>+
>+JNILogSink::JNILogSink(JNIEnv* env, const JavaRef<jobject>& j_logging)
>+    : j_logging_(env, j_logging) {}
>+JNILogSink::~JNILogSink() = default;
>+
>+void JNILogSink::OnLogMessage(const std::string& msg,
>+                              rtc::LoggingSeverity severity,
>+                              const char* tag) {
>+  JNIEnv* env = AttachCurrentThreadIfNeeded();
>+  Java_JNILogging_logToInjectable(env, j_logging_, NativeToJavaString(env, msg),
>+                                  NativeToJavaInteger(env, severity),
>+                                  NativeToJavaString(env, tag));
>+}
>+
>+void JNILogSink::OnLogMessage(const std::string& msg) {
>+  RTC_NOTREACHED();
>+}
>+
>+}  // namespace jni
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/logging/logsink.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/logging/logsink.h
>new file mode 100644
>index 00000000000..bac51a06da9
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/logging/logsink.h
>@@ -0,0 +1,39 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#ifndef SDK_ANDROID_SRC_JNI_LOGGING_LOGSINK_H_
>+#define SDK_ANDROID_SRC_JNI_LOGGING_LOGSINK_H_
>+
>+#include <string>
>+
>+#include "rtc_base/logging.h"
>+#include "sdk/android/native_api/jni/java_types.h"
>+#include "sdk/android/src/jni/jni_helpers.h"
>+
>+namespace webrtc {
>+namespace jni {
>+
>+class JNILogSink : public rtc::LogSink {
>+ public:
>+  JNILogSink(JNIEnv* env, const JavaRef<jobject>& j_logging);
>+  ~JNILogSink() override;
>+
>+  void OnLogMessage(const std::string& msg,
>+                    rtc::LoggingSeverity severity,
>+                    const char* tag) override;
>+  void OnLogMessage(const std::string& msg) override;
>+
>+ private:
>+  const ScopedJavaGlobalRef<jobject> j_logging_;
>+};
>+
>+}  // namespace jni
>+}  // namespace webrtc
>+
>+#endif  // SDK_ANDROID_SRC_JNI_LOGGING_LOGSINK_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/nativecapturerobserver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/nativecapturerobserver.cc
>new file mode 100644
>index 00000000000..3dd97ce56ee
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/nativecapturerobserver.cc
>@@ -0,0 +1,76 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "sdk/android/src/jni/nativecapturerobserver.h"
>+
>+#include "api/videosourceproxy.h"
>+#include "rtc_base/logging.h"
>+#include "sdk/android/generated_video_jni/jni/NativeCapturerObserver_jni.h"
>+#include "sdk/android/native_api/jni/java_types.h"
>+#include "sdk/android/src/jni/androidvideotracksource.h"
>+
>+namespace webrtc {
>+namespace jni {
>+
>+namespace {
>+VideoRotation jintToVideoRotation(jint rotation) {
>+  RTC_DCHECK(rotation == 0 || rotation == 90 || rotation == 180 ||
>+             rotation == 270);
>+  return static_cast<VideoRotation>(rotation);
>+}
>+}  // namespace
>+
>+ScopedJavaLocalRef<jobject> CreateJavaNativeCapturerObserver(
>+    JNIEnv* env,
>+    rtc::scoped_refptr<AndroidVideoTrackSource> native_source) {
>+  return Java_NativeCapturerObserver_Constructor(
>+      env, NativeToJavaPointer(native_source.release()));
>+}
>+
>+static void JNI_NativeCapturerObserver_OnFrameCaptured(
>+    JNIEnv* jni,
>+    const JavaParamRef<jclass>&,
>+    jlong j_source,
>+    jint j_width,
>+    jint j_height,
>+    jint j_rotation,
>+    jlong j_timestamp_ns,
>+    const JavaParamRef<jobject>& j_video_frame_buffer) {
>+  AndroidVideoTrackSource* source =
>+      reinterpret_cast<AndroidVideoTrackSource*>(j_source);
>+  source->OnFrameCaptured(jni, j_width, j_height, j_timestamp_ns,
>+                          jintToVideoRotation(j_rotation),
>+                          j_video_frame_buffer);
>+}
>+
>+static void JNI_NativeCapturerObserver_CapturerStarted(
>+    JNIEnv* jni,
>+    const JavaParamRef<jclass>&,
>+    jlong j_source,
>+    jboolean j_success) {
>+  RTC_LOG(LS_INFO) << "NativeCapturerObserver_nativeCapturerStarted";
>+  AndroidVideoTrackSource* source =
>+      reinterpret_cast<AndroidVideoTrackSource*>(j_source);
>+  source->SetState(j_success ? AndroidVideoTrackSource::SourceState::kLive
>+                             : AndroidVideoTrackSource::SourceState::kEnded);
>+}
>+
>+static void JNI_NativeCapturerObserver_CapturerStopped(
>+    JNIEnv* jni,
>+    const JavaParamRef<jclass>&,
>+    jlong j_source) {
>+  RTC_LOG(LS_INFO) << "NativeCapturerObserver_nativeCapturerStopped";
>+  AndroidVideoTrackSource* source =
>+      reinterpret_cast<AndroidVideoTrackSource*>(j_source);
>+  source->SetState(AndroidVideoTrackSource::SourceState::kEnded);
>+}
>+
>+}  // namespace jni
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/nativecapturerobserver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/nativecapturerobserver.h
>new file mode 100644
>index 00000000000..57ddc075e72
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/nativecapturerobserver.h
>@@ -0,0 +1,29 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef SDK_ANDROID_SRC_JNI_NATIVECAPTUREROBSERVER_H_
>+#define SDK_ANDROID_SRC_JNI_NATIVECAPTUREROBSERVER_H_
>+
>+#include <jni.h>
>+
>+#include "sdk/android/native_api/jni/scoped_java_ref.h"
>+#include "sdk/android/src/jni/androidvideotracksource.h"
>+
>+namespace webrtc {
>+namespace jni {
>+
>+ScopedJavaLocalRef<jobject> CreateJavaNativeCapturerObserver(
>+    JNIEnv* env,
>+    rtc::scoped_refptr<AndroidVideoTrackSource> native_source);
>+
>+}  // namespace jni
>+}  // namespace webrtc
>+
>+#endif  // SDK_ANDROID_SRC_JNI_NATIVECAPTUREROBSERVER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/nv12buffer_jni.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/nv12buffer.cc
>similarity index 54%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/nv12buffer_jni.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/nv12buffer.cc
>index bc11b9390c4..23b1791ef98 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/nv12buffer_jni.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/nv12buffer.cc
>@@ -15,30 +15,30 @@
> #include "third_party/libyuv/include/libyuv/scale.h"
> 
> #include "rtc_base/checks.h"
>+#include "sdk/android/generated_video_jni/jni/NV12Buffer_jni.h"
> 
> namespace webrtc {
> namespace jni {
> 
>-extern "C" JNIEXPORT void JNICALL
>-Java_org_webrtc_NV12Buffer_nativeCropAndScale(JNIEnv* jni,
>-                                              jclass,
>-                                              jint crop_x,
>-                                              jint crop_y,
>-                                              jint crop_width,
>-                                              jint crop_height,
>-                                              jint scale_width,
>-                                              jint scale_height,
>-                                              jobject j_src,
>-                                              jint src_width,
>-                                              jint src_height,
>-                                              jint src_stride,
>-                                              jint src_slice_height,
>-                                              jobject j_dst_y,
>-                                              jint dst_stride_y,
>-                                              jobject j_dst_u,
>-                                              jint dst_stride_u,
>-                                              jobject j_dst_v,
>-                                              jint dst_stride_v) {
>+static void JNI_NV12Buffer_CropAndScale(JNIEnv* jni,
>+                                        const JavaParamRef<jclass>&,
>+                                        jint crop_x,
>+                                        jint crop_y,
>+                                        jint crop_width,
>+                                        jint crop_height,
>+                                        jint scale_width,
>+                                        jint scale_height,
>+                                        const JavaParamRef<jobject>& j_src,
>+                                        jint src_width,
>+                                        jint src_height,
>+                                        jint src_stride,
>+                                        jint src_slice_height,
>+                                        const JavaParamRef<jobject>& j_dst_y,
>+                                        jint dst_stride_y,
>+                                        const JavaParamRef<jobject>& j_dst_u,
>+                                        jint dst_stride_u,
>+                                        const JavaParamRef<jobject>& j_dst_v,
>+                                        jint dst_stride_v) {
>   const int src_stride_y = src_stride;
>   const int src_stride_uv = src_stride;
>   const int crop_chroma_x = crop_x / 2;
>@@ -50,16 +50,19 @@ Java_org_webrtc_NV12Buffer_nativeCropAndScale(JNIEnv* jni,
>   const int tmp_size = crop_chroma_height * (tmp_stride_u + tmp_stride_v);
> 
>   uint8_t const* src_y =
>-      static_cast<uint8_t const*>(jni->GetDirectBufferAddress(j_src));
>+      static_cast<uint8_t const*>(jni->GetDirectBufferAddress(j_src.obj()));
>   uint8_t const* src_uv = src_y + src_slice_height * src_stride_y;
> 
>-  uint8_t* dst_y = static_cast<uint8_t*>(jni->GetDirectBufferAddress(j_dst_y));
>-  uint8_t* dst_u = static_cast<uint8_t*>(jni->GetDirectBufferAddress(j_dst_u));
>-  uint8_t* dst_v = static_cast<uint8_t*>(jni->GetDirectBufferAddress(j_dst_v));
>+  uint8_t* dst_y =
>+      static_cast<uint8_t*>(jni->GetDirectBufferAddress(j_dst_y.obj()));
>+  uint8_t* dst_u =
>+      static_cast<uint8_t*>(jni->GetDirectBufferAddress(j_dst_u.obj()));
>+  uint8_t* dst_v =
>+      static_cast<uint8_t*>(jni->GetDirectBufferAddress(j_dst_v.obj()));
> 
>   // Crop using pointer arithmetic.
>   src_y += crop_x + crop_y * src_stride_y;
>-  src_uv += crop_chroma_x + crop_chroma_y * src_stride_uv;
>+  src_uv += 2 * crop_chroma_x + crop_chroma_y * src_stride_uv;
> 
>   std::vector<uint8_t> tmp_buffer(tmp_size);
>   uint8_t* tmp_u = tmp_buffer.data();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/nv21buffer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/nv21buffer.cc
>new file mode 100644
>index 00000000000..d619aac710c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/nv21buffer.cc
>@@ -0,0 +1,73 @@
>+/*
>+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <jni.h>
>+#include <vector>
>+
>+#include "third_party/libyuv/include/libyuv/convert.h"
>+#include "third_party/libyuv/include/libyuv/scale.h"
>+
>+#include "common_video/libyuv/include/webrtc_libyuv.h"
>+#include "rtc_base/checks.h"
>+#include "sdk/android/generated_video_jni/jni/NV21Buffer_jni.h"
>+
>+namespace webrtc {
>+namespace jni {
>+
>+static void JNI_NV21Buffer_CropAndScale(JNIEnv* jni,
>+                                        const JavaParamRef<jclass>&,
>+                                        jint crop_x,
>+                                        jint crop_y,
>+                                        jint crop_width,
>+                                        jint crop_height,
>+                                        jint scale_width,
>+                                        jint scale_height,
>+                                        const JavaParamRef<jbyteArray>& j_src,
>+                                        jint src_width,
>+                                        jint src_height,
>+                                        const JavaParamRef<jobject>& j_dst_y,
>+                                        jint dst_stride_y,
>+                                        const JavaParamRef<jobject>& j_dst_u,
>+                                        jint dst_stride_u,
>+                                        const JavaParamRef<jobject>& j_dst_v,
>+                                        jint dst_stride_v) {
>+  const int src_stride_y = src_width;
>+  const int src_stride_uv = src_width;
>+  const int crop_chroma_x = crop_x / 2;
>+  const int crop_chroma_y = crop_y / 2;
>+
>+  jboolean was_copy;
>+  jbyte* src_bytes = jni->GetByteArrayElements(j_src.obj(), &was_copy);
>+  RTC_DCHECK(!was_copy);
>+  uint8_t const* src_y = reinterpret_cast<uint8_t const*>(src_bytes);
>+  uint8_t const* src_uv = src_y + src_height * src_stride_y;
>+
>+  uint8_t* dst_y =
>+      static_cast<uint8_t*>(jni->GetDirectBufferAddress(j_dst_y.obj()));
>+  uint8_t* dst_u =
>+      static_cast<uint8_t*>(jni->GetDirectBufferAddress(j_dst_u.obj()));
>+  uint8_t* dst_v =
>+      static_cast<uint8_t*>(jni->GetDirectBufferAddress(j_dst_v.obj()));
>+
>+  // Crop using pointer arithmetic.
>+  src_y += crop_x + crop_y * src_stride_y;
>+  src_uv += 2 * crop_chroma_x + crop_chroma_y * src_stride_uv;
>+
>+  NV12ToI420Scaler scaler;
>+  // U- and V-planes are swapped because this is NV21 not NV12.
>+  scaler.NV12ToI420Scale(src_y, src_stride_y, src_uv, src_stride_uv, crop_width,
>+                         crop_height, dst_y, dst_stride_y, dst_v, dst_stride_v,
>+                         dst_u, dst_stride_u, scale_width, scale_height);
>+
>+  jni->ReleaseByteArrayElements(j_src.obj(), src_bytes, JNI_ABORT);
>+}
>+
>+}  // namespace jni
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/nv21buffer_jni.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/nv21buffer_jni.cc
>deleted file mode 100644
>index 2e107d7cbcb..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/nv21buffer_jni.cc
>+++ /dev/null
>@@ -1,75 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include <jni.h>
>-#include <vector>
>-
>-#include "third_party/libyuv/include/libyuv/convert.h"
>-#include "third_party/libyuv/include/libyuv/scale.h"
>-
>-#include "common_video/libyuv/include/webrtc_libyuv.h"
>-#include "rtc_base/checks.h"
>-
>-namespace webrtc {
>-namespace jni {
>-
>-extern "C" JNIEXPORT void JNICALL
>-Java_org_webrtc_NV21Buffer_nativeCropAndScale(JNIEnv* jni,
>-                                              jclass,
>-                                              jint crop_x,
>-                                              jint crop_y,
>-                                              jint crop_width,
>-                                              jint crop_height,
>-                                              jint scale_width,
>-                                              jint scale_height,
>-                                              jbyteArray j_src,
>-                                              jint src_width,
>-                                              jint src_height,
>-                                              jobject j_dst_y,
>-                                              jint dst_stride_y,
>-                                              jobject j_dst_u,
>-                                              jint dst_stride_u,
>-                                              jobject j_dst_v,
>-                                              jint dst_stride_v) {
>-  const int src_stride_y = src_width;
>-  const int src_stride_uv = src_width;
>-  const int crop_chroma_x = crop_x / 2;
>-  const int crop_chroma_y = crop_y / 2;
>-  const int crop_chroma_width = (crop_width + 1) / 2;
>-  const int crop_chroma_height = (crop_height + 1) / 2;
>-  const int tmp_stride_u = crop_chroma_width;
>-  const int tmp_stride_v = crop_chroma_width;
>-  const int tmp_size = crop_chroma_height * (tmp_stride_u + tmp_stride_v);
>-
>-  jboolean was_copy;
>-  jbyte* src_bytes = jni->GetByteArrayElements(j_src, &was_copy);
>-  RTC_DCHECK(!was_copy);
>-  uint8_t const* src_y = reinterpret_cast<uint8_t const*>(src_bytes);
>-  uint8_t const* src_uv = src_y + src_height * src_stride_y;
>-
>-  uint8_t* dst_y = static_cast<uint8_t*>(jni->GetDirectBufferAddress(j_dst_y));
>-  uint8_t* dst_u = static_cast<uint8_t*>(jni->GetDirectBufferAddress(j_dst_u));
>-  uint8_t* dst_v = static_cast<uint8_t*>(jni->GetDirectBufferAddress(j_dst_v));
>-
>-  // Crop using pointer arithmetic.
>-  src_y += crop_x + crop_y * src_stride_y;
>-  src_uv += crop_chroma_x + crop_chroma_y * src_stride_uv;
>-
>-  NV12ToI420Scaler scaler;
>-  // U- and V-planes are swapped because this is NV21 not NV12.
>-  scaler.NV12ToI420Scale(src_y, src_stride_y, src_uv, src_stride_uv, crop_width,
>-                         crop_height, dst_y, dst_stride_y, dst_v, dst_stride_v,
>-                         dst_u, dst_stride_u, scale_width, scale_height);
>-
>-  jni->ReleaseByteArrayElements(j_src, src_bytes, JNI_ABORT);
>-}
>-
>-}  // namespace jni
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/OWNERS
>deleted file mode 100644
>index a32f041ac89..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/OWNERS
>+++ /dev/null
>@@ -1 +0,0 @@
>-deadbeef@webrtc.org
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/session.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/androidnetworkmonitor.h
>similarity index 63%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/session.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/androidnetworkmonitor.h
>index 1a23f8363f7..3eee11b3a5d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/session.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/androidnetworkmonitor.h
>@@ -1,5 +1,5 @@
> /*
>- *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
>+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
>  *
>  *  Use of this source code is governed by a BSD-style license
>  *  that can be found in the LICENSE file in the root of the source
>@@ -8,5 +8,5 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-// TODO(deadbeef): Remove this file when Chrome build files no longer reference
>-// it.
>+// TODO(sakal): Remove this file once clients have update to the native API.
>+#include "sdk/android/src/jni/androidnetworkmonitor.h"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/audio_jni.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/audio.cc
>similarity index 91%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/audio_jni.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/audio.cc
>index 7dde0258c63..376c42178af 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/audio_jni.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/audio.cc
>@@ -8,7 +8,7 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "sdk/android/src/jni/pc/audio_jni.h"
>+#include "sdk/android/src/jni/pc/audio.h"
> 
> #include "api/audio_codecs/builtin_audio_decoder_factory.h"
> #include "api/audio_codecs/builtin_audio_encoder_factory.h"
>@@ -26,7 +26,7 @@ rtc::scoped_refptr<AudioEncoderFactory> CreateAudioEncoderFactory() {
> }
> 
> rtc::scoped_refptr<AudioProcessing> CreateAudioProcessing() {
>-  return AudioProcessing::Create();
>+  return AudioProcessingBuilder().Create();
> }
> 
> }  // namespace jni
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/audio_jni.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/audio.h
>similarity index 77%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/audio_jni.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/audio.h
>index d6998b8d355..49cf0699e10 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/audio_jni.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/audio.h
>@@ -8,13 +8,13 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#ifndef SDK_ANDROID_SRC_JNI_PC_AUDIO_JNI_H_
>-#define SDK_ANDROID_SRC_JNI_PC_AUDIO_JNI_H_
>+#ifndef SDK_ANDROID_SRC_JNI_PC_AUDIO_H_
>+#define SDK_ANDROID_SRC_JNI_PC_AUDIO_H_
> 
> // Adding 'nogncheck' to disable the gn include headers check.
> // We don't want this target depend on audio related targets
>-#include "api/audio_codecs/audio_decoder_factory.h"  // nogncheck
>-#include "api/audio_codecs/audio_encoder_factory.h"  // nogncheck
>+#include "api/audio_codecs/audio_decoder_factory.h"             // nogncheck
>+#include "api/audio_codecs/audio_encoder_factory.h"             // nogncheck
> #include "modules/audio_processing/include/audio_processing.h"  // nogncheck
> #include "rtc_base/scoped_ref_ptr.h"
> 
>@@ -30,4 +30,4 @@ rtc::scoped_refptr<AudioProcessing> CreateAudioProcessing();
> }  // namespace jni
> }  // namespace webrtc
> 
>-#endif  // SDK_ANDROID_SRC_JNI_PC_AUDIO_JNI_H_
>+#endif  // SDK_ANDROID_SRC_JNI_PC_AUDIO_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/audiotrack_jni.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/audiotrack.cc
>similarity index 70%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/audiotrack_jni.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/audiotrack.cc
>index 9662a6872cf..90df1213505 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/audiotrack_jni.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/audiotrack.cc
>@@ -9,17 +9,15 @@
>  */
> 
> #include "api/mediastreaminterface.h"
>-#include "sdk/android/src/jni/jni_helpers.h"
>+#include "sdk/android/generated_peerconnection_jni/jni/AudioTrack_jni.h"
> 
> namespace webrtc {
> namespace jni {
> 
>-JNI_FUNCTION_DECLARATION(void,
>-                         AudioTrack_nativeSetVolume,
>-                         JNIEnv*,
>-                         jclass,
>-                         jlong j_p,
>-                         jdouble volume) {
>+static void JNI_AudioTrack_SetVolume(JNIEnv*,
>+                                     const JavaParamRef<jclass>&,
>+                                     jlong j_p,
>+                                     jdouble volume) {
>   rtc::scoped_refptr<AudioSourceInterface> source(
>       reinterpret_cast<AudioTrackInterface*>(j_p)->GetSource());
>   source->SetVolume(volume);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/callsessionfilerotatinglogsink_jni.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/callsessionfilerotatinglogsink.cc
>similarity index 65%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/callsessionfilerotatinglogsink_jni.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/callsessionfilerotatinglogsink.cc
>index af142ae31be..da8d9b3c1e2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/callsessionfilerotatinglogsink_jni.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/callsessionfilerotatinglogsink.cc
>@@ -9,18 +9,19 @@
>  */
> 
> #include "rtc_base/logsinks.h"
>+#include "sdk/android/generated_peerconnection_jni/jni/CallSessionFileRotatingLogSink_jni.h"
>+#include "sdk/android/native_api/jni/java_types.h"
> #include "sdk/android/src/jni/jni_helpers.h"
> 
> namespace webrtc {
> namespace jni {
> 
>-JNI_FUNCTION_DECLARATION(jlong,
>-                         CallSessionFileRotatingLogSink_nativeAddSink,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jstring j_dirPath,
>-                         jint j_maxFileSize,
>-                         jint j_severity) {
>+static jlong JNI_CallSessionFileRotatingLogSink_AddSink(
>+    JNIEnv* jni,
>+    const JavaParamRef<jclass>&,
>+    const JavaParamRef<jstring>& j_dirPath,
>+    jint j_maxFileSize,
>+    jint j_severity) {
>   std::string dir_path = JavaToStdString(jni, j_dirPath);
>   rtc::CallSessionFileRotatingLogSink* sink =
>       new rtc::CallSessionFileRotatingLogSink(dir_path, j_maxFileSize);
>@@ -33,46 +34,46 @@ JNI_FUNCTION_DECLARATION(jlong,
>   }
>   rtc::LogMessage::AddLogToStream(
>       sink, static_cast<rtc::LoggingSeverity>(j_severity));
>-  return (jlong)sink;
>+  return jlongFromPointer(sink);
> }
> 
>-JNI_FUNCTION_DECLARATION(void,
>-                         CallSessionFileRotatingLogSink_nativeDeleteSink,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jlong j_sink) {
>+static void JNI_CallSessionFileRotatingLogSink_DeleteSink(
>+    JNIEnv* jni,
>+    const JavaParamRef<jclass>&,
>+    jlong j_sink) {
>   rtc::CallSessionFileRotatingLogSink* sink =
>       reinterpret_cast<rtc::CallSessionFileRotatingLogSink*>(j_sink);
>   rtc::LogMessage::RemoveLogToStream(sink);
>   delete sink;
> }
> 
>-JNI_FUNCTION_DECLARATION(jbyteArray,
>-                         CallSessionFileRotatingLogSink_nativeGetLogData,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jstring j_dirPath) {
>+static ScopedJavaLocalRef<jbyteArray>
>+JNI_CallSessionFileRotatingLogSink_GetLogData(
>+    JNIEnv* jni,
>+    const JavaParamRef<jclass>&,
>+    const JavaParamRef<jstring>& j_dirPath) {
>   std::string dir_path = JavaToStdString(jni, j_dirPath);
>   std::unique_ptr<rtc::CallSessionFileRotatingStream> stream(
>       new rtc::CallSessionFileRotatingStream(dir_path));
>   if (!stream->Open()) {
>     RTC_LOG_V(rtc::LoggingSeverity::LS_WARNING)
>         << "Failed to open CallSessionFileRotatingStream for path " << dir_path;
>-    return jni->NewByteArray(0);
>+    return ScopedJavaLocalRef<jbyteArray>(jni, jni->NewByteArray(0));
>   }
>   size_t log_size = 0;
>   if (!stream->GetSize(&log_size) || log_size == 0) {
>     RTC_LOG_V(rtc::LoggingSeverity::LS_WARNING)
>         << "CallSessionFileRotatingStream returns 0 size for path " << dir_path;
>-    return jni->NewByteArray(0);
>+    return ScopedJavaLocalRef<jbyteArray>(jni, jni->NewByteArray(0));
>   }
> 
>   size_t read = 0;
>   std::unique_ptr<jbyte> buffer(static_cast<jbyte*>(malloc(log_size)));
>   stream->ReadAll(buffer.get(), log_size, &read, nullptr);
> 
>-  jbyteArray result = jni->NewByteArray(read);
>-  jni->SetByteArrayRegion(result, 0, read, buffer.get());
>+  ScopedJavaLocalRef<jbyteArray> result =
>+      ScopedJavaLocalRef<jbyteArray>(jni, jni->NewByteArray(read));
>+  jni->SetByteArrayRegion(result.obj(), 0, read, buffer.get());
> 
>   return result;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/datachannel.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/datachannel.cc
>index c4be4318277..d07e00e39a5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/datachannel.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/datachannel.cc
>@@ -12,10 +12,11 @@
> 
> #include <limits>
> 
>+#include "absl/memory/memory.h"
> #include "api/datachannelinterface.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/ptr_util.h"
> #include "sdk/android/generated_peerconnection_jni/jni/DataChannel_jni.h"
>+#include "sdk/android/native_api/jni/java_types.h"
> #include "sdk/android/src/jni/jni_helpers.h"
> #include "sdk/android/src/jni/pc/datachannel.h"
> 
>@@ -27,48 +28,52 @@ namespace {
> // and dispatching the callback from C++ back to Java.
> class DataChannelObserverJni : public DataChannelObserver {
>  public:
>-  DataChannelObserverJni(JNIEnv* jni, jobject j_observer);
>-  virtual ~DataChannelObserverJni() {}
>+  DataChannelObserverJni(JNIEnv* jni, const JavaRef<jobject>& j_observer);
>+  ~DataChannelObserverJni() override {}
> 
>   void OnBufferedAmountChange(uint64_t previous_amount) override;
>   void OnStateChange() override;
>   void OnMessage(const DataBuffer& buffer) override;
> 
>  private:
>-  const ScopedGlobalRef<jobject> j_observer_global_;
>+  const ScopedJavaGlobalRef<jobject> j_observer_global_;
> };
> 
>-DataChannelObserverJni::DataChannelObserverJni(JNIEnv* jni, jobject j_observer)
>+DataChannelObserverJni::DataChannelObserverJni(
>+    JNIEnv* jni,
>+    const JavaRef<jobject>& j_observer)
>     : j_observer_global_(jni, j_observer) {}
> 
> void DataChannelObserverJni::OnBufferedAmountChange(uint64_t previous_amount) {
>   JNIEnv* env = AttachCurrentThreadIfNeeded();
>-  Java_Observer_onBufferedAmountChange(env, *j_observer_global_,
>+  Java_Observer_onBufferedAmountChange(env, j_observer_global_,
>                                        previous_amount);
> }
> 
> void DataChannelObserverJni::OnStateChange() {
>   JNIEnv* env = AttachCurrentThreadIfNeeded();
>-  Java_Observer_onStateChange(env, *j_observer_global_);
>+  Java_Observer_onStateChange(env, j_observer_global_);
> }
> 
> void DataChannelObserverJni::OnMessage(const DataBuffer& buffer) {
>   JNIEnv* env = AttachCurrentThreadIfNeeded();
>-  ScopedLocalRefFrame local_ref_frame(env);
>-  jobject byte_buffer = env->NewDirectByteBuffer(
>-      const_cast<char*>(buffer.data.data<char>()), buffer.data.size());
>-  jobject j_buffer = Java_Buffer_Constructor(env, byte_buffer, buffer.binary);
>-  Java_Observer_onMessage(env, *j_observer_global_, j_buffer);
>+  ScopedJavaLocalRef<jobject> byte_buffer = NewDirectByteBuffer(
>+      env, const_cast<char*>(buffer.data.data<char>()), buffer.data.size());
>+  ScopedJavaLocalRef<jobject> j_buffer =
>+      Java_Buffer_Constructor(env, byte_buffer, buffer.binary);
>+  Java_Observer_onMessage(env, j_observer_global_, j_buffer);
> }
> 
>-DataChannelInterface* ExtractNativeDC(JNIEnv* jni, jobject j_dc) {
>-  jlong j_d = Java_DataChannel_getNativeDataChannel(jni, j_dc);
>-  return reinterpret_cast<DataChannelInterface*>(j_d);
>+DataChannelInterface* ExtractNativeDC(JNIEnv* jni,
>+                                      const JavaParamRef<jobject>& j_dc) {
>+  return reinterpret_cast<DataChannelInterface*>(
>+      Java_DataChannel_getNativeDataChannel(jni, j_dc));
> }
> 
> }  // namespace
> 
>-DataChannelInit JavaToNativeDataChannelInit(JNIEnv* env, jobject j_init) {
>+DataChannelInit JavaToNativeDataChannelInit(JNIEnv* env,
>+                                            const JavaRef<jobject>& j_init) {
>   DataChannelInit init;
>   init.ordered = Java_Init_getOrdered(env, j_init);
>   init.maxRetransmitTime = Java_Init_getMaxRetransmitTimeMs(env, j_init);
>@@ -79,79 +84,72 @@ DataChannelInit JavaToNativeDataChannelInit(JNIEnv* env, jobject j_init) {
>   return init;
> }
> 
>-jobject WrapNativeDataChannel(
>+ScopedJavaLocalRef<jobject> WrapNativeDataChannel(
>     JNIEnv* env,
>     rtc::scoped_refptr<DataChannelInterface> channel) {
>+  if (!channel)
>+    return nullptr;
>   // Channel is now owned by Java object, and will be freed from there.
>-  return channel ? Java_DataChannel_Constructor(
>-                       env, jlongFromPointer(channel.release()))
>-                 : nullptr;
>+  return Java_DataChannel_Constructor(env, jlongFromPointer(channel.release()));
> }
> 
>-JNI_FUNCTION_DECLARATION(jlong,
>-                         DataChannel_registerObserverNative,
>-                         JNIEnv* jni,
>-                         jobject j_dc,
>-                         jobject j_observer) {
>-  auto observer = rtc::MakeUnique<DataChannelObserverJni>(jni, j_observer);
>+static jlong JNI_DataChannel_RegisterObserver(
>+    JNIEnv* jni,
>+    const JavaParamRef<jobject>& j_dc,
>+    const JavaParamRef<jobject>& j_observer) {
>+  auto observer = absl::make_unique<DataChannelObserverJni>(jni, j_observer);
>   ExtractNativeDC(jni, j_dc)->RegisterObserver(observer.get());
>   return jlongFromPointer(observer.release());
> }
> 
>-JNI_FUNCTION_DECLARATION(void,
>-                         DataChannel_unregisterObserverNative,
>-                         JNIEnv* jni,
>-                         jobject j_dc,
>-                         jlong native_observer) {
>+static void JNI_DataChannel_UnregisterObserver(
>+    JNIEnv* jni,
>+    const JavaParamRef<jobject>& j_dc,
>+    jlong native_observer) {
>   ExtractNativeDC(jni, j_dc)->UnregisterObserver();
>   delete reinterpret_cast<DataChannelObserverJni*>(native_observer);
> }
> 
>-JNI_FUNCTION_DECLARATION(jstring,
>-                         DataChannel_label,
>-                         JNIEnv* jni,
>-                         jobject j_dc) {
>+static ScopedJavaLocalRef<jstring> JNI_DataChannel_Label(
>+    JNIEnv* jni,
>+    const JavaParamRef<jobject>& j_dc) {
>   return NativeToJavaString(jni, ExtractNativeDC(jni, j_dc)->label());
> }
> 
>-JNI_FUNCTION_DECLARATION(jint, DataChannel_id, JNIEnv* jni, jobject j_dc) {
>+static jint JNI_DataChannel_Id(JNIEnv* jni, const JavaParamRef<jobject>& j_dc) {
>   int id = ExtractNativeDC(jni, j_dc)->id();
>   RTC_CHECK_LE(id, std::numeric_limits<int32_t>::max())
>       << "id overflowed jint!";
>   return static_cast<jint>(id);
> }
> 
>-JNI_FUNCTION_DECLARATION(jobject,
>-                         DataChannel_state,
>-                         JNIEnv* jni,
>-                         jobject j_dc) {
>+static ScopedJavaLocalRef<jobject> JNI_DataChannel_State(
>+    JNIEnv* jni,
>+    const JavaParamRef<jobject>& j_dc) {
>   return Java_State_fromNativeIndex(jni, ExtractNativeDC(jni, j_dc)->state());
> }
> 
>-JNI_FUNCTION_DECLARATION(jlong,
>-                         DataChannel_bufferedAmount,
>-                         JNIEnv* jni,
>-                         jobject j_dc) {
>+static jlong JNI_DataChannel_BufferedAmount(JNIEnv* jni,
>+                                            const JavaParamRef<jobject>& j_dc) {
>   uint64_t buffered_amount = ExtractNativeDC(jni, j_dc)->buffered_amount();
>   RTC_CHECK_LE(buffered_amount, std::numeric_limits<int64_t>::max())
>       << "buffered_amount overflowed jlong!";
>   return static_cast<jlong>(buffered_amount);
> }
> 
>-JNI_FUNCTION_DECLARATION(void, DataChannel_close, JNIEnv* jni, jobject j_dc) {
>+static void JNI_DataChannel_Close(JNIEnv* jni,
>+                                  const JavaParamRef<jobject>& j_dc) {
>   ExtractNativeDC(jni, j_dc)->Close();
> }
> 
>-JNI_FUNCTION_DECLARATION(jboolean,
>-                         DataChannel_sendNative,
>-                         JNIEnv* jni,
>-                         jobject j_dc,
>-                         jbyteArray data,
>-                         jboolean binary) {
>-  jbyte* bytes = jni->GetByteArrayElements(data, nullptr);
>+static jboolean JNI_DataChannel_Send(JNIEnv* jni,
>+                                     const JavaParamRef<jobject>& j_dc,
>+                                     const JavaParamRef<jbyteArray>& data,
>+                                     jboolean binary) {
>+  jbyte* bytes = jni->GetByteArrayElements(data.obj(), nullptr);
>   bool ret = ExtractNativeDC(jni, j_dc)->Send(DataBuffer(
>-      rtc::CopyOnWriteBuffer(bytes, jni->GetArrayLength(data)), binary));
>-  jni->ReleaseByteArrayElements(data, bytes, JNI_ABORT);
>+      rtc::CopyOnWriteBuffer(bytes, jni->GetArrayLength(data.obj())), binary));
>+  jni->ReleaseByteArrayElements(data.obj(), bytes, JNI_ABORT);
>   return ret;
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/datachannel.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/datachannel.h
>index 557520af441..c6786f64e7b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/datachannel.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/datachannel.h
>@@ -14,10 +14,12 @@
> namespace webrtc {
> namespace jni {
> 
>-DataChannelInit JavaToNativeDataChannelInit(JNIEnv* env, jobject j_init);
>+DataChannelInit JavaToNativeDataChannelInit(JNIEnv* env,
>+                                            const JavaRef<jobject>& j_init);
> 
>-jobject WrapNativeDataChannel(JNIEnv* env,
>-                              rtc::scoped_refptr<DataChannelInterface> channel);
>+ScopedJavaLocalRef<jobject> WrapNativeDataChannel(
>+    JNIEnv* env,
>+    rtc::scoped_refptr<DataChannelInterface> channel);
> 
> }  // namespace jni
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/defaultaudioprocessingfactory_jni.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/defaultaudioprocessingfactory_jni.cc
>deleted file mode 100644
>index a0d2c03dda0..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/defaultaudioprocessingfactory_jni.cc
>+++ /dev/null
>@@ -1,36 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include <memory>
>-
>-#include "modules/audio_processing/include/audio_processing.h"
>-#include "rtc_base/scoped_ref_ptr.h"
>-#include "sdk/android/src/jni/jni_helpers.h"
>-
>-namespace webrtc {
>-namespace jni {
>-
>-JNI_FUNCTION_DECLARATION(
>-    jlong,
>-    DefaultAudioProcessingFactory_nativeCreateAudioProcessing,
>-    JNIEnv*,
>-    jclass,
>-    jlong native_post_processor) {
>-  std::unique_ptr<PostProcessing> post_processor(
>-      reinterpret_cast<PostProcessing*>(native_post_processor));
>-  rtc::scoped_refptr<AudioProcessing> audio_processing =
>-      AudioProcessing::Create(webrtc::Config(), std::move(post_processor),
>-                              nullptr /* echo_control_factory */,
>-                              nullptr /* beamformer */);
>-  return jlongFromPointer(audio_processing.release());
>-}
>-
>-}  // namespace jni
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/dtmfsender.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/dtmfsender.cc
>new file mode 100644
>index 00000000000..fbddc6b0bfc
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/dtmfsender.cc
>@@ -0,0 +1,60 @@
>+/*
>+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/dtmfsenderinterface.h"
>+#include "sdk/android/generated_peerconnection_jni/jni/DtmfSender_jni.h"
>+#include "sdk/android/native_api/jni/java_types.h"
>+#include "sdk/android/src/jni/jni_helpers.h"
>+
>+namespace webrtc {
>+namespace jni {
>+
>+static jboolean JNI_DtmfSender_CanInsertDtmf(JNIEnv* jni,
>+                                             const JavaParamRef<jclass>&,
>+                                             jlong j_dtmf_sender_pointer) {
>+  return reinterpret_cast<DtmfSenderInterface*>(j_dtmf_sender_pointer)
>+      ->CanInsertDtmf();
>+}
>+
>+static jboolean JNI_DtmfSender_InsertDtmf(JNIEnv* jni,
>+                                          const JavaParamRef<jclass>&,
>+                                          jlong j_dtmf_sender_pointer,
>+                                          const JavaParamRef<jstring>& tones,
>+                                          jint duration,
>+                                          jint inter_tone_gap) {
>+  return reinterpret_cast<DtmfSenderInterface*>(j_dtmf_sender_pointer)
>+      ->InsertDtmf(JavaToStdString(jni, tones), duration, inter_tone_gap);
>+}
>+
>+static ScopedJavaLocalRef<jstring> JNI_DtmfSender_Tones(
>+    JNIEnv* jni,
>+    const JavaParamRef<jclass>&,
>+    jlong j_dtmf_sender_pointer) {
>+  return NativeToJavaString(
>+      jni,
>+      reinterpret_cast<DtmfSenderInterface*>(j_dtmf_sender_pointer)->tones());
>+}
>+
>+static jint JNI_DtmfSender_Duration(JNIEnv* jni,
>+                                    const JavaParamRef<jclass>&,
>+                                    jlong j_dtmf_sender_pointer) {
>+  return reinterpret_cast<DtmfSenderInterface*>(j_dtmf_sender_pointer)
>+      ->duration();
>+}
>+
>+static jint JNI_DtmfSender_InterToneGap(JNIEnv* jni,
>+                                        const JavaParamRef<jclass>&,
>+                                        jlong j_dtmf_sender_pointer) {
>+  return reinterpret_cast<DtmfSenderInterface*>(j_dtmf_sender_pointer)
>+      ->inter_tone_gap();
>+}
>+
>+}  // namespace jni
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/dtmfsender_jni.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/dtmfsender_jni.cc
>deleted file mode 100644
>index c8d5493d7e8..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/dtmfsender_jni.cc
>+++ /dev/null
>@@ -1,67 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "api/dtmfsenderinterface.h"
>-#include "sdk/android/src/jni/jni_helpers.h"
>-
>-namespace webrtc {
>-namespace jni {
>-
>-JNI_FUNCTION_DECLARATION(jboolean,
>-                         DtmfSender_nativeCanInsertDtmf,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jlong j_dtmf_sender_pointer) {
>-  return reinterpret_cast<DtmfSenderInterface*>(j_dtmf_sender_pointer)
>-      ->CanInsertDtmf();
>-}
>-
>-JNI_FUNCTION_DECLARATION(jboolean,
>-                         DtmfSender_nativeInsertDtmf,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jlong j_dtmf_sender_pointer,
>-                         jstring tones,
>-                         jint duration,
>-                         jint inter_tone_gap) {
>-  return reinterpret_cast<DtmfSenderInterface*>(j_dtmf_sender_pointer)
>-      ->InsertDtmf(JavaToStdString(jni, tones), duration, inter_tone_gap);
>-}
>-
>-JNI_FUNCTION_DECLARATION(jstring,
>-                         DtmfSender_nativeTones,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jlong j_dtmf_sender_pointer) {
>-  return NativeToJavaString(
>-      jni,
>-      reinterpret_cast<DtmfSenderInterface*>(j_dtmf_sender_pointer)->tones());
>-}
>-
>-JNI_FUNCTION_DECLARATION(jint,
>-                         DtmfSender_nativeDuration,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jlong j_dtmf_sender_pointer) {
>-  return reinterpret_cast<DtmfSenderInterface*>(j_dtmf_sender_pointer)
>-      ->duration();
>-}
>-
>-JNI_FUNCTION_DECLARATION(jint,
>-                         DtmfSender_nativeInterToneGap,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jlong j_dtmf_sender_pointer) {
>-  return reinterpret_cast<DtmfSenderInterface*>(j_dtmf_sender_pointer)
>-      ->inter_tone_gap();
>-}
>-
>-}  // namespace jni
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/icecandidate.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/icecandidate.cc
>new file mode 100644
>index 00000000000..b9d66a4679d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/icecandidate.cc
>@@ -0,0 +1,238 @@
>+/*
>+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "sdk/android/src/jni/pc/icecandidate.h"
>+
>+#include <string>
>+
>+#include "pc/webrtcsdp.h"
>+#include "sdk/android/generated_peerconnection_jni/jni/IceCandidate_jni.h"
>+#include "sdk/android/native_api/jni/java_types.h"
>+#include "sdk/android/src/jni/pc/mediastreamtrack.h"
>+
>+namespace webrtc {
>+namespace jni {
>+
>+namespace {
>+
>+ScopedJavaLocalRef<jobject> CreateJavaIceCandidate(
>+    JNIEnv* env,
>+    const std::string& sdp_mid,
>+    int sdp_mline_index,
>+    const std::string& sdp,
>+    const std::string server_url) {
>+  return Java_IceCandidate_Constructor(
>+      env, NativeToJavaString(env, sdp_mid), sdp_mline_index,
>+      NativeToJavaString(env, sdp), NativeToJavaString(env, server_url));
>+}
>+
>+}  // namespace
>+
>+cricket::Candidate JavaToNativeCandidate(JNIEnv* jni,
>+                                         const JavaRef<jobject>& j_candidate) {
>+  std::string sdp_mid =
>+      JavaToStdString(jni, Java_IceCandidate_getSdpMid(jni, j_candidate));
>+  std::string sdp =
>+      JavaToStdString(jni, Java_IceCandidate_getSdp(jni, j_candidate));
>+  cricket::Candidate candidate;
>+  if (!SdpDeserializeCandidate(sdp_mid, sdp, &candidate, NULL)) {
>+    RTC_LOG(LS_ERROR) << "SdpDescrializeCandidate failed with sdp " << sdp;
>+  }
>+  return candidate;
>+}
>+
>+ScopedJavaLocalRef<jobject> NativeToJavaCandidate(
>+    JNIEnv* env,
>+    const cricket::Candidate& candidate) {
>+  std::string sdp = SdpSerializeCandidate(candidate);
>+  RTC_CHECK(!sdp.empty()) << "got an empty ICE candidate";
>+  // sdp_mline_index is not used, pass an invalid value -1.
>+  return CreateJavaIceCandidate(env, candidate.transport_name(),
>+                                -1 /* sdp_mline_index */, sdp,
>+                                "" /* server_url */);
>+}
>+
>+ScopedJavaLocalRef<jobject> NativeToJavaIceCandidate(
>+    JNIEnv* env,
>+    const IceCandidateInterface& candidate) {
>+  std::string sdp;
>+  RTC_CHECK(candidate.ToString(&sdp)) << "got so far: " << sdp;
>+  return CreateJavaIceCandidate(env, candidate.sdp_mid(),
>+                                candidate.sdp_mline_index(), sdp,
>+                                candidate.candidate().url());
>+}
>+
>+ScopedJavaLocalRef<jobjectArray> NativeToJavaCandidateArray(
>+    JNIEnv* jni,
>+    const std::vector<cricket::Candidate>& candidates) {
>+  return NativeToJavaObjectArray(jni, candidates,
>+                                 org_webrtc_IceCandidate_clazz(jni),
>+                                 &NativeToJavaCandidate);
>+}
>+
>+PeerConnectionInterface::IceTransportsType JavaToNativeIceTransportsType(
>+    JNIEnv* jni,
>+    const JavaRef<jobject>& j_ice_transports_type) {
>+  std::string enum_name = GetJavaEnumName(jni, j_ice_transports_type);
>+
>+  if (enum_name == "ALL")
>+    return PeerConnectionInterface::kAll;
>+
>+  if (enum_name == "RELAY")
>+    return PeerConnectionInterface::kRelay;
>+
>+  if (enum_name == "NOHOST")
>+    return PeerConnectionInterface::kNoHost;
>+
>+  if (enum_name == "NONE")
>+    return PeerConnectionInterface::kNone;
>+
>+  RTC_CHECK(false) << "Unexpected IceTransportsType enum_name " << enum_name;
>+  return PeerConnectionInterface::kAll;
>+}
>+
>+PeerConnectionInterface::BundlePolicy JavaToNativeBundlePolicy(
>+    JNIEnv* jni,
>+    const JavaRef<jobject>& j_bundle_policy) {
>+  std::string enum_name = GetJavaEnumName(jni, j_bundle_policy);
>+
>+  if (enum_name == "BALANCED")
>+    return PeerConnectionInterface::kBundlePolicyBalanced;
>+
>+  if (enum_name == "MAXBUNDLE")
>+    return PeerConnectionInterface::kBundlePolicyMaxBundle;
>+
>+  if (enum_name == "MAXCOMPAT")
>+    return PeerConnectionInterface::kBundlePolicyMaxCompat;
>+
>+  RTC_CHECK(false) << "Unexpected BundlePolicy enum_name " << enum_name;
>+  return PeerConnectionInterface::kBundlePolicyBalanced;
>+}
>+
>+PeerConnectionInterface::RtcpMuxPolicy JavaToNativeRtcpMuxPolicy(
>+    JNIEnv* jni,
>+    const JavaRef<jobject>& j_rtcp_mux_policy) {
>+  std::string enum_name = GetJavaEnumName(jni, j_rtcp_mux_policy);
>+
>+  if (enum_name == "NEGOTIATE")
>+    return PeerConnectionInterface::kRtcpMuxPolicyNegotiate;
>+
>+  if (enum_name == "REQUIRE")
>+    return PeerConnectionInterface::kRtcpMuxPolicyRequire;
>+
>+  RTC_CHECK(false) << "Unexpected RtcpMuxPolicy enum_name " << enum_name;
>+  return PeerConnectionInterface::kRtcpMuxPolicyNegotiate;
>+}
>+
>+PeerConnectionInterface::TcpCandidatePolicy JavaToNativeTcpCandidatePolicy(
>+    JNIEnv* jni,
>+    const JavaRef<jobject>& j_tcp_candidate_policy) {
>+  std::string enum_name = GetJavaEnumName(jni, j_tcp_candidate_policy);
>+
>+  if (enum_name == "ENABLED")
>+    return PeerConnectionInterface::kTcpCandidatePolicyEnabled;
>+
>+  if (enum_name == "DISABLED")
>+    return PeerConnectionInterface::kTcpCandidatePolicyDisabled;
>+
>+  RTC_CHECK(false) << "Unexpected TcpCandidatePolicy enum_name " << enum_name;
>+  return PeerConnectionInterface::kTcpCandidatePolicyEnabled;
>+}
>+
>+PeerConnectionInterface::CandidateNetworkPolicy
>+JavaToNativeCandidateNetworkPolicy(
>+    JNIEnv* jni,
>+    const JavaRef<jobject>& j_candidate_network_policy) {
>+  std::string enum_name = GetJavaEnumName(jni, j_candidate_network_policy);
>+
>+  if (enum_name == "ALL")
>+    return PeerConnectionInterface::kCandidateNetworkPolicyAll;
>+
>+  if (enum_name == "LOW_COST")
>+    return PeerConnectionInterface::kCandidateNetworkPolicyLowCost;
>+
>+  RTC_CHECK(false) << "Unexpected CandidateNetworkPolicy enum_name "
>+                   << enum_name;
>+  return PeerConnectionInterface::kCandidateNetworkPolicyAll;
>+}
>+
>+rtc::KeyType JavaToNativeKeyType(JNIEnv* jni,
>+                                 const JavaRef<jobject>& j_key_type) {
>+  std::string enum_name = GetJavaEnumName(jni, j_key_type);
>+
>+  if (enum_name == "RSA")
>+    return rtc::KT_RSA;
>+  if (enum_name == "ECDSA")
>+    return rtc::KT_ECDSA;
>+
>+  RTC_CHECK(false) << "Unexpected KeyType enum_name " << enum_name;
>+  return rtc::KT_ECDSA;
>+}
>+
>+PeerConnectionInterface::ContinualGatheringPolicy
>+JavaToNativeContinualGatheringPolicy(
>+    JNIEnv* jni,
>+    const JavaRef<jobject>& j_gathering_policy) {
>+  std::string enum_name = GetJavaEnumName(jni, j_gathering_policy);
>+  if (enum_name == "GATHER_ONCE")
>+    return PeerConnectionInterface::GATHER_ONCE;
>+
>+  if (enum_name == "GATHER_CONTINUALLY")
>+    return PeerConnectionInterface::GATHER_CONTINUALLY;
>+
>+  RTC_CHECK(false) << "Unexpected ContinualGatheringPolicy enum name "
>+                   << enum_name;
>+  return PeerConnectionInterface::GATHER_ONCE;
>+}
>+
>+PeerConnectionInterface::TlsCertPolicy JavaToNativeTlsCertPolicy(
>+    JNIEnv* jni,
>+    const JavaRef<jobject>& j_ice_server_tls_cert_policy) {
>+  std::string enum_name = GetJavaEnumName(jni, j_ice_server_tls_cert_policy);
>+
>+  if (enum_name == "TLS_CERT_POLICY_SECURE")
>+    return PeerConnectionInterface::kTlsCertPolicySecure;
>+
>+  if (enum_name == "TLS_CERT_POLICY_INSECURE_NO_CHECK")
>+    return PeerConnectionInterface::kTlsCertPolicyInsecureNoCheck;
>+
>+  RTC_CHECK(false) << "Unexpected TlsCertPolicy enum_name " << enum_name;
>+  return PeerConnectionInterface::kTlsCertPolicySecure;
>+}
>+
>+absl::optional<rtc::AdapterType> JavaToNativeNetworkPreference(
>+    JNIEnv* jni,
>+    const JavaRef<jobject>& j_network_preference) {
>+  std::string enum_name = GetJavaEnumName(jni, j_network_preference);
>+
>+  if (enum_name == "UNKNOWN")
>+    return absl::nullopt;
>+
>+  if (enum_name == "ETHERNET")
>+    return rtc::ADAPTER_TYPE_ETHERNET;
>+
>+  if (enum_name == "WIFI")
>+    return rtc::ADAPTER_TYPE_WIFI;
>+
>+  if (enum_name == "CELLULAR")
>+    return rtc::ADAPTER_TYPE_CELLULAR;
>+
>+  if (enum_name == "VPN")
>+    return rtc::ADAPTER_TYPE_VPN;
>+
>+  if (enum_name == "LOOPBACK")
>+    return rtc::ADAPTER_TYPE_LOOPBACK;
>+
>+  RTC_CHECK(false) << "Unexpected NetworkPreference enum_name " << enum_name;
>+  return absl::nullopt;
>+}
>+
>+}  // namespace jni
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/icecandidate.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/icecandidate.h
>new file mode 100644
>index 00000000000..662b649ba68
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/icecandidate.h
>@@ -0,0 +1,85 @@
>+/*
>+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef SDK_ANDROID_SRC_JNI_PC_ICECANDIDATE_H_
>+#define SDK_ANDROID_SRC_JNI_PC_ICECANDIDATE_H_
>+
>+#include <vector>
>+
>+#include "api/datachannelinterface.h"
>+#include "api/jsep.h"
>+#include "api/jsepicecandidate.h"
>+#include "api/peerconnectioninterface.h"
>+#include "api/rtpparameters.h"
>+#include "rtc_base/sslidentity.h"
>+#include "sdk/android/src/jni/jni_helpers.h"
>+
>+namespace webrtc {
>+namespace jni {
>+
>+cricket::Candidate JavaToNativeCandidate(JNIEnv* jni,
>+                                         const JavaRef<jobject>& j_candidate);
>+
>+ScopedJavaLocalRef<jobject> NativeToJavaCandidate(
>+    JNIEnv* env,
>+    const cricket::Candidate& candidate);
>+
>+ScopedJavaLocalRef<jobject> NativeToJavaIceCandidate(
>+    JNIEnv* env,
>+    const IceCandidateInterface& candidate);
>+
>+ScopedJavaLocalRef<jobjectArray> NativeToJavaCandidateArray(
>+    JNIEnv* jni,
>+    const std::vector<cricket::Candidate>& candidates);
>+
>+/*****************************************************
>+ * Below are all things that go into RTCConfiguration.
>+ *****************************************************/
>+PeerConnectionInterface::IceTransportsType JavaToNativeIceTransportsType(
>+    JNIEnv* jni,
>+    const JavaRef<jobject>& j_ice_transports_type);
>+
>+PeerConnectionInterface::BundlePolicy JavaToNativeBundlePolicy(
>+    JNIEnv* jni,
>+    const JavaRef<jobject>& j_bundle_policy);
>+
>+PeerConnectionInterface::RtcpMuxPolicy JavaToNativeRtcpMuxPolicy(
>+    JNIEnv* jni,
>+    const JavaRef<jobject>& j_rtcp_mux_policy);
>+
>+PeerConnectionInterface::TcpCandidatePolicy JavaToNativeTcpCandidatePolicy(
>+    JNIEnv* jni,
>+    const JavaRef<jobject>& j_tcp_candidate_policy);
>+
>+PeerConnectionInterface::CandidateNetworkPolicy
>+JavaToNativeCandidateNetworkPolicy(
>+    JNIEnv* jni,
>+    const JavaRef<jobject>& j_candidate_network_policy);
>+
>+rtc::KeyType JavaToNativeKeyType(JNIEnv* jni,
>+                                 const JavaRef<jobject>& j_key_type);
>+
>+PeerConnectionInterface::ContinualGatheringPolicy
>+JavaToNativeContinualGatheringPolicy(
>+    JNIEnv* jni,
>+    const JavaRef<jobject>& j_gathering_policy);
>+
>+PeerConnectionInterface::TlsCertPolicy JavaToNativeTlsCertPolicy(
>+    JNIEnv* jni,
>+    const JavaRef<jobject>& j_ice_server_tls_cert_policy);
>+
>+absl::optional<rtc::AdapterType> JavaToNativeNetworkPreference(
>+    JNIEnv* jni,
>+    const JavaRef<jobject>& j_network_preference);
>+
>+}  // namespace jni
>+}  // namespace webrtc
>+
>+#endif  // SDK_ANDROID_SRC_JNI_PC_ICECANDIDATE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/java_native_conversion.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/java_native_conversion.cc
>deleted file mode 100644
>index 3f0e2e1e1bf..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/java_native_conversion.cc
>+++ /dev/null
>@@ -1,626 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "sdk/android/src/jni/pc/java_native_conversion.h"
>-
>-#include <string>
>-
>-#include "pc/webrtcsdp.h"
>-#include "sdk/android/generated_peerconnection_jni/jni/IceCandidate_jni.h"
>-#include "sdk/android/generated_peerconnection_jni/jni/MediaStreamTrack_jni.h"
>-#include "sdk/android/src/jni/classreferenceholder.h"
>-
>-namespace webrtc {
>-namespace jni {
>-
>-namespace {
>-
>-jobject CreateJavaIceCandidate(JNIEnv* env,
>-                               const std::string& sdp_mid,
>-                               int sdp_mline_index,
>-                               const std::string& sdp,
>-                               const std::string server_url) {
>-  return Java_IceCandidate_Constructor(
>-      env, NativeToJavaString(env, sdp_mid), sdp_mline_index,
>-      NativeToJavaString(env, sdp), NativeToJavaString(env, server_url));
>-}
>-
>-}  // namespace
>-
>-jobject NativeToJavaMediaType(JNIEnv* jni, cricket::MediaType media_type) {
>-  return Java_MediaType_fromNativeIndex(jni, media_type);
>-}
>-
>-cricket::MediaType JavaToNativeMediaType(JNIEnv* jni, jobject j_media_type) {
>-  return static_cast<cricket::MediaType>(
>-      Java_MediaType_getNative(jni, j_media_type));
>-}
>-
>-cricket::Candidate JavaToNativeCandidate(JNIEnv* jni, jobject j_candidate) {
>-  std::string sdp_mid =
>-      JavaToStdString(jni, Java_IceCandidate_getSdpMid(jni, j_candidate));
>-  std::string sdp =
>-      JavaToStdString(jni, Java_IceCandidate_getSdp(jni, j_candidate));
>-  cricket::Candidate candidate;
>-  if (!SdpDeserializeCandidate(sdp_mid, sdp, &candidate, NULL)) {
>-    RTC_LOG(LS_ERROR) << "SdpDescrializeCandidate failed with sdp " << sdp;
>-  }
>-  return candidate;
>-}
>-
>-jobject NativeToJavaCandidate(JNIEnv* env,
>-                              const cricket::Candidate& candidate) {
>-  std::string sdp = SdpSerializeCandidate(candidate);
>-  RTC_CHECK(!sdp.empty()) << "got an empty ICE candidate";
>-  // sdp_mline_index is not used, pass an invalid value -1.
>-  return CreateJavaIceCandidate(env, candidate.transport_name(),
>-                                -1 /* sdp_mline_index */, sdp,
>-                                "" /* server_url */);
>-}
>-
>-jobject NativeToJavaIceCandidate(JNIEnv* env,
>-                                 const IceCandidateInterface& candidate) {
>-  std::string sdp;
>-  RTC_CHECK(candidate.ToString(&sdp)) << "got so far: " << sdp;
>-  return CreateJavaIceCandidate(env, candidate.sdp_mid(),
>-                                candidate.sdp_mline_index(), sdp,
>-                                candidate.candidate().url());
>-}
>-
>-jobjectArray NativeToJavaCandidateArray(
>-    JNIEnv* jni,
>-    const std::vector<cricket::Candidate>& candidates) {
>-  return NativeToJavaObjectArray(jni, candidates,
>-                                 org_webrtc_IceCandidate_clazz(jni),
>-                                 &NativeToJavaCandidate);
>-}
>-
>-SessionDescriptionInterface* JavaToNativeSessionDescription(JNIEnv* jni,
>-                                                            jobject j_sdp) {
>-  jfieldID j_type_id = GetFieldID(jni, GetObjectClass(jni, j_sdp), "type",
>-                                  "Lorg/webrtc/SessionDescription$Type;");
>-  jobject j_type = GetObjectField(jni, j_sdp, j_type_id);
>-  jmethodID j_canonical_form_id =
>-      GetMethodID(jni, GetObjectClass(jni, j_type), "canonicalForm",
>-                  "()Ljava/lang/String;");
>-  jstring j_type_string =
>-      (jstring)jni->CallObjectMethod(j_type, j_canonical_form_id);
>-  CHECK_EXCEPTION(jni) << "error during CallObjectMethod";
>-  std::string std_type = JavaToStdString(jni, j_type_string);
>-
>-  jfieldID j_description_id = GetFieldID(jni, GetObjectClass(jni, j_sdp),
>-                                         "description", "Ljava/lang/String;");
>-  jstring j_description = (jstring)GetObjectField(jni, j_sdp, j_description_id);
>-  std::string std_description = JavaToStdString(jni, j_description);
>-
>-  return CreateSessionDescription(std_type, std_description, NULL);
>-}
>-
>-jobject NativeToJavaSessionDescription(
>-    JNIEnv* jni,
>-    const SessionDescriptionInterface* desc) {
>-  std::string sdp;
>-  RTC_CHECK(desc->ToString(&sdp)) << "got so far: " << sdp;
>-  jstring j_description = NativeToJavaString(jni, sdp);
>-
>-  jclass j_type_class = FindClass(jni, "org/webrtc/SessionDescription$Type");
>-  jmethodID j_type_from_canonical = GetStaticMethodID(
>-      jni, j_type_class, "fromCanonicalForm",
>-      "(Ljava/lang/String;)Lorg/webrtc/SessionDescription$Type;");
>-  jstring j_type_string = NativeToJavaString(jni, desc->type());
>-  jobject j_type = jni->CallStaticObjectMethod(
>-      j_type_class, j_type_from_canonical, j_type_string);
>-  CHECK_EXCEPTION(jni) << "error during CallObjectMethod";
>-
>-  jclass j_sdp_class = FindClass(jni, "org/webrtc/SessionDescription");
>-  jmethodID j_sdp_ctor =
>-      GetMethodID(jni, j_sdp_class, "<init>",
>-                  "(Lorg/webrtc/SessionDescription$Type;Ljava/lang/String;)V");
>-  jobject j_sdp =
>-      jni->NewObject(j_sdp_class, j_sdp_ctor, j_type, j_description);
>-  CHECK_EXCEPTION(jni) << "error during NewObject";
>-  return j_sdp;
>-}
>-
>-PeerConnectionFactoryInterface::Options
>-JavaToNativePeerConnectionFactoryOptions(JNIEnv* jni, jobject options) {
>-  jclass options_class = jni->GetObjectClass(options);
>-  jfieldID network_ignore_mask_field =
>-      jni->GetFieldID(options_class, "networkIgnoreMask", "I");
>-  int network_ignore_mask =
>-      jni->GetIntField(options, network_ignore_mask_field);
>-
>-  jfieldID disable_encryption_field =
>-      jni->GetFieldID(options_class, "disableEncryption", "Z");
>-  bool disable_encryption =
>-      jni->GetBooleanField(options, disable_encryption_field);
>-
>-  jfieldID disable_network_monitor_field =
>-      jni->GetFieldID(options_class, "disableNetworkMonitor", "Z");
>-  bool disable_network_monitor =
>-      jni->GetBooleanField(options, disable_network_monitor_field);
>-
>-  PeerConnectionFactoryInterface::Options native_options;
>-
>-  // This doesn't necessarily match the c++ version of this struct; feel free
>-  // to add more parameters as necessary.
>-  native_options.network_ignore_mask = network_ignore_mask;
>-  native_options.disable_encryption = disable_encryption;
>-  native_options.disable_network_monitor = disable_network_monitor;
>-  return native_options;
>-}
>-
>-PeerConnectionInterface::IceTransportsType JavaToNativeIceTransportsType(
>-    JNIEnv* jni,
>-    jobject j_ice_transports_type) {
>-  std::string enum_name = GetJavaEnumName(jni, j_ice_transports_type);
>-
>-  if (enum_name == "ALL")
>-    return PeerConnectionInterface::kAll;
>-
>-  if (enum_name == "RELAY")
>-    return PeerConnectionInterface::kRelay;
>-
>-  if (enum_name == "NOHOST")
>-    return PeerConnectionInterface::kNoHost;
>-
>-  if (enum_name == "NONE")
>-    return PeerConnectionInterface::kNone;
>-
>-  RTC_CHECK(false) << "Unexpected IceTransportsType enum_name " << enum_name;
>-  return PeerConnectionInterface::kAll;
>-}
>-
>-PeerConnectionInterface::BundlePolicy JavaToNativeBundlePolicy(
>-    JNIEnv* jni,
>-    jobject j_bundle_policy) {
>-  std::string enum_name = GetJavaEnumName(jni, j_bundle_policy);
>-
>-  if (enum_name == "BALANCED")
>-    return PeerConnectionInterface::kBundlePolicyBalanced;
>-
>-  if (enum_name == "MAXBUNDLE")
>-    return PeerConnectionInterface::kBundlePolicyMaxBundle;
>-
>-  if (enum_name == "MAXCOMPAT")
>-    return PeerConnectionInterface::kBundlePolicyMaxCompat;
>-
>-  RTC_CHECK(false) << "Unexpected BundlePolicy enum_name " << enum_name;
>-  return PeerConnectionInterface::kBundlePolicyBalanced;
>-}
>-
>-PeerConnectionInterface::RtcpMuxPolicy JavaToNativeRtcpMuxPolicy(
>-    JNIEnv* jni,
>-    jobject j_rtcp_mux_policy) {
>-  std::string enum_name = GetJavaEnumName(jni, j_rtcp_mux_policy);
>-
>-  if (enum_name == "NEGOTIATE")
>-    return PeerConnectionInterface::kRtcpMuxPolicyNegotiate;
>-
>-  if (enum_name == "REQUIRE")
>-    return PeerConnectionInterface::kRtcpMuxPolicyRequire;
>-
>-  RTC_CHECK(false) << "Unexpected RtcpMuxPolicy enum_name " << enum_name;
>-  return PeerConnectionInterface::kRtcpMuxPolicyNegotiate;
>-}
>-
>-PeerConnectionInterface::TcpCandidatePolicy JavaToNativeTcpCandidatePolicy(
>-    JNIEnv* jni,
>-    jobject j_tcp_candidate_policy) {
>-  std::string enum_name = GetJavaEnumName(jni, j_tcp_candidate_policy);
>-
>-  if (enum_name == "ENABLED")
>-    return PeerConnectionInterface::kTcpCandidatePolicyEnabled;
>-
>-  if (enum_name == "DISABLED")
>-    return PeerConnectionInterface::kTcpCandidatePolicyDisabled;
>-
>-  RTC_CHECK(false) << "Unexpected TcpCandidatePolicy enum_name " << enum_name;
>-  return PeerConnectionInterface::kTcpCandidatePolicyEnabled;
>-}
>-
>-PeerConnectionInterface::CandidateNetworkPolicy
>-JavaToNativeCandidateNetworkPolicy(JNIEnv* jni,
>-                                   jobject j_candidate_network_policy) {
>-  std::string enum_name = GetJavaEnumName(jni, j_candidate_network_policy);
>-
>-  if (enum_name == "ALL")
>-    return PeerConnectionInterface::kCandidateNetworkPolicyAll;
>-
>-  if (enum_name == "LOW_COST")
>-    return PeerConnectionInterface::kCandidateNetworkPolicyLowCost;
>-
>-  RTC_CHECK(false) << "Unexpected CandidateNetworkPolicy enum_name "
>-                   << enum_name;
>-  return PeerConnectionInterface::kCandidateNetworkPolicyAll;
>-}
>-
>-rtc::KeyType JavaToNativeKeyType(JNIEnv* jni, jobject j_key_type) {
>-  std::string enum_name = GetJavaEnumName(jni, j_key_type);
>-
>-  if (enum_name == "RSA")
>-    return rtc::KT_RSA;
>-  if (enum_name == "ECDSA")
>-    return rtc::KT_ECDSA;
>-
>-  RTC_CHECK(false) << "Unexpected KeyType enum_name " << enum_name;
>-  return rtc::KT_ECDSA;
>-}
>-
>-PeerConnectionInterface::ContinualGatheringPolicy
>-JavaToNativeContinualGatheringPolicy(JNIEnv* jni, jobject j_gathering_policy) {
>-  std::string enum_name = GetJavaEnumName(jni, j_gathering_policy);
>-  if (enum_name == "GATHER_ONCE")
>-    return PeerConnectionInterface::GATHER_ONCE;
>-
>-  if (enum_name == "GATHER_CONTINUALLY")
>-    return PeerConnectionInterface::GATHER_CONTINUALLY;
>-
>-  RTC_CHECK(false) << "Unexpected ContinualGatheringPolicy enum name "
>-                   << enum_name;
>-  return PeerConnectionInterface::GATHER_ONCE;
>-}
>-
>-PeerConnectionInterface::TlsCertPolicy JavaToNativeTlsCertPolicy(
>-    JNIEnv* jni,
>-    jobject j_ice_server_tls_cert_policy) {
>-  std::string enum_name = GetJavaEnumName(jni, j_ice_server_tls_cert_policy);
>-
>-  if (enum_name == "TLS_CERT_POLICY_SECURE")
>-    return PeerConnectionInterface::kTlsCertPolicySecure;
>-
>-  if (enum_name == "TLS_CERT_POLICY_INSECURE_NO_CHECK")
>-    return PeerConnectionInterface::kTlsCertPolicyInsecureNoCheck;
>-
>-  RTC_CHECK(false) << "Unexpected TlsCertPolicy enum_name " << enum_name;
>-  return PeerConnectionInterface::kTlsCertPolicySecure;
>-}
>-
>-void JavaToNativeIceServers(JNIEnv* jni,
>-                            jobject j_ice_servers,
>-                            PeerConnectionInterface::IceServers* ice_servers) {
>-  for (jobject j_ice_server : Iterable(jni, j_ice_servers)) {
>-    jclass j_ice_server_class = GetObjectClass(jni, j_ice_server);
>-    jfieldID j_ice_server_urls_id =
>-        GetFieldID(jni, j_ice_server_class, "urls", "Ljava/util/List;");
>-    jfieldID j_ice_server_username_id =
>-        GetFieldID(jni, j_ice_server_class, "username", "Ljava/lang/String;");
>-    jfieldID j_ice_server_password_id =
>-        GetFieldID(jni, j_ice_server_class, "password", "Ljava/lang/String;");
>-    jfieldID j_ice_server_tls_cert_policy_id =
>-        GetFieldID(jni, j_ice_server_class, "tlsCertPolicy",
>-                   "Lorg/webrtc/PeerConnection$TlsCertPolicy;");
>-    jobject j_ice_server_tls_cert_policy =
>-        GetObjectField(jni, j_ice_server, j_ice_server_tls_cert_policy_id);
>-    jfieldID j_ice_server_hostname_id =
>-        GetFieldID(jni, j_ice_server_class, "hostname", "Ljava/lang/String;");
>-    jfieldID j_ice_server_tls_alpn_protocols_id = GetFieldID(
>-        jni, j_ice_server_class, "tlsAlpnProtocols", "Ljava/util/List;");
>-    jfieldID j_ice_server_tls_elliptic_curves_id = GetFieldID(
>-        jni, j_ice_server_class, "tlsEllipticCurves", "Ljava/util/List;");
>-    jobject urls = GetObjectField(jni, j_ice_server, j_ice_server_urls_id);
>-    jstring username = reinterpret_cast<jstring>(
>-        GetObjectField(jni, j_ice_server, j_ice_server_username_id));
>-    jstring password = reinterpret_cast<jstring>(
>-        GetObjectField(jni, j_ice_server, j_ice_server_password_id));
>-    PeerConnectionInterface::TlsCertPolicy tls_cert_policy =
>-        JavaToNativeTlsCertPolicy(jni, j_ice_server_tls_cert_policy);
>-    jstring hostname = reinterpret_cast<jstring>(
>-        GetObjectField(jni, j_ice_server, j_ice_server_hostname_id));
>-    jobject tls_alpn_protocols = GetNullableObjectField(
>-        jni, j_ice_server, j_ice_server_tls_alpn_protocols_id);
>-    jobject tls_elliptic_curves = GetNullableObjectField(
>-        jni, j_ice_server, j_ice_server_tls_elliptic_curves_id);
>-    PeerConnectionInterface::IceServer server;
>-    server.urls = JavaToStdVectorStrings(jni, urls);
>-    server.username = JavaToStdString(jni, username);
>-    server.password = JavaToStdString(jni, password);
>-    server.tls_cert_policy = tls_cert_policy;
>-    server.hostname = JavaToStdString(jni, hostname);
>-    server.tls_alpn_protocols = JavaToStdVectorStrings(jni, tls_alpn_protocols);
>-    server.tls_elliptic_curves =
>-        JavaToStdVectorStrings(jni, tls_elliptic_curves);
>-    ice_servers->push_back(server);
>-  }
>-}
>-
>-void JavaToNativeRTCConfiguration(
>-    JNIEnv* jni,
>-    jobject j_rtc_config,
>-    PeerConnectionInterface::RTCConfiguration* rtc_config) {
>-  jclass j_rtc_config_class = GetObjectClass(jni, j_rtc_config);
>-
>-  jfieldID j_ice_transports_type_id =
>-      GetFieldID(jni, j_rtc_config_class, "iceTransportsType",
>-                 "Lorg/webrtc/PeerConnection$IceTransportsType;");
>-  jobject j_ice_transports_type =
>-      GetObjectField(jni, j_rtc_config, j_ice_transports_type_id);
>-
>-  jfieldID j_bundle_policy_id =
>-      GetFieldID(jni, j_rtc_config_class, "bundlePolicy",
>-                 "Lorg/webrtc/PeerConnection$BundlePolicy;");
>-  jobject j_bundle_policy =
>-      GetObjectField(jni, j_rtc_config, j_bundle_policy_id);
>-
>-  jfieldID j_rtcp_mux_policy_id =
>-      GetFieldID(jni, j_rtc_config_class, "rtcpMuxPolicy",
>-                 "Lorg/webrtc/PeerConnection$RtcpMuxPolicy;");
>-  jobject j_rtcp_mux_policy =
>-      GetObjectField(jni, j_rtc_config, j_rtcp_mux_policy_id);
>-
>-  jfieldID j_tcp_candidate_policy_id =
>-      GetFieldID(jni, j_rtc_config_class, "tcpCandidatePolicy",
>-                 "Lorg/webrtc/PeerConnection$TcpCandidatePolicy;");
>-  jobject j_tcp_candidate_policy =
>-      GetObjectField(jni, j_rtc_config, j_tcp_candidate_policy_id);
>-
>-  jfieldID j_candidate_network_policy_id =
>-      GetFieldID(jni, j_rtc_config_class, "candidateNetworkPolicy",
>-                 "Lorg/webrtc/PeerConnection$CandidateNetworkPolicy;");
>-  jobject j_candidate_network_policy =
>-      GetObjectField(jni, j_rtc_config, j_candidate_network_policy_id);
>-
>-  jfieldID j_ice_servers_id =
>-      GetFieldID(jni, j_rtc_config_class, "iceServers", "Ljava/util/List;");
>-  jobject j_ice_servers = GetObjectField(jni, j_rtc_config, j_ice_servers_id);
>-
>-  jfieldID j_audio_jitter_buffer_max_packets_id =
>-      GetFieldID(jni, j_rtc_config_class, "audioJitterBufferMaxPackets", "I");
>-  jfieldID j_audio_jitter_buffer_fast_accelerate_id = GetFieldID(
>-      jni, j_rtc_config_class, "audioJitterBufferFastAccelerate", "Z");
>-
>-  jfieldID j_ice_connection_receiving_timeout_id =
>-      GetFieldID(jni, j_rtc_config_class, "iceConnectionReceivingTimeout", "I");
>-
>-  jfieldID j_ice_backup_candidate_pair_ping_interval_id = GetFieldID(
>-      jni, j_rtc_config_class, "iceBackupCandidatePairPingInterval", "I");
>-
>-  jfieldID j_continual_gathering_policy_id =
>-      GetFieldID(jni, j_rtc_config_class, "continualGatheringPolicy",
>-                 "Lorg/webrtc/PeerConnection$ContinualGatheringPolicy;");
>-  jobject j_continual_gathering_policy =
>-      GetObjectField(jni, j_rtc_config, j_continual_gathering_policy_id);
>-
>-  jfieldID j_ice_candidate_pool_size_id =
>-      GetFieldID(jni, j_rtc_config_class, "iceCandidatePoolSize", "I");
>-  jfieldID j_presume_writable_when_fully_relayed_id = GetFieldID(
>-      jni, j_rtc_config_class, "presumeWritableWhenFullyRelayed", "Z");
>-
>-  jfieldID j_prune_turn_ports_id =
>-      GetFieldID(jni, j_rtc_config_class, "pruneTurnPorts", "Z");
>-
>-  jfieldID j_ice_check_min_interval_id = GetFieldID(
>-      jni, j_rtc_config_class, "iceCheckMinInterval", "Ljava/lang/Integer;");
>-
>-  jfieldID j_disable_ipv6_on_wifi_id =
>-      GetFieldID(jni, j_rtc_config_class, "disableIPv6OnWifi", "Z");
>-
>-  jfieldID j_max_ipv6_networks_id =
>-      GetFieldID(jni, j_rtc_config_class, "maxIPv6Networks", "I");
>-
>-  jfieldID j_ice_regather_interval_range_id =
>-      GetFieldID(jni, j_rtc_config_class, "iceRegatherIntervalRange",
>-                 "Lorg/webrtc/PeerConnection$IntervalRange;");
>-  jclass j_interval_range_class =
>-      jni->FindClass("org/webrtc/PeerConnection$IntervalRange");
>-  jmethodID get_min_id =
>-      GetMethodID(jni, j_interval_range_class, "getMin", "()I");
>-  jmethodID get_max_id =
>-      GetMethodID(jni, j_interval_range_class, "getMax", "()I");
>-
>-  jfieldID j_turn_customizer_type_id = GetFieldID(
>-      jni, j_rtc_config_class, "turnCustomizer", "Lorg/webrtc/TurnCustomizer;");
>-  jobject j_turn_customizer =
>-      GetNullableObjectField(jni, j_rtc_config, j_turn_customizer_type_id);
>-
>-  jclass j_turn_customizer_class = jni->FindClass("org/webrtc/TurnCustomizer");
>-  jfieldID j_native_turn_customizer_id =
>-      GetFieldID(jni, j_turn_customizer_class, "nativeTurnCustomizer", "J");
>-
>-  rtc_config->type = JavaToNativeIceTransportsType(jni, j_ice_transports_type);
>-  rtc_config->bundle_policy = JavaToNativeBundlePolicy(jni, j_bundle_policy);
>-  rtc_config->rtcp_mux_policy =
>-      JavaToNativeRtcpMuxPolicy(jni, j_rtcp_mux_policy);
>-  rtc_config->tcp_candidate_policy =
>-      JavaToNativeTcpCandidatePolicy(jni, j_tcp_candidate_policy);
>-  rtc_config->candidate_network_policy =
>-      JavaToNativeCandidateNetworkPolicy(jni, j_candidate_network_policy);
>-  JavaToNativeIceServers(jni, j_ice_servers, &rtc_config->servers);
>-  rtc_config->audio_jitter_buffer_max_packets =
>-      GetIntField(jni, j_rtc_config, j_audio_jitter_buffer_max_packets_id);
>-  rtc_config->audio_jitter_buffer_fast_accelerate = GetBooleanField(
>-      jni, j_rtc_config, j_audio_jitter_buffer_fast_accelerate_id);
>-  rtc_config->ice_connection_receiving_timeout =
>-      GetIntField(jni, j_rtc_config, j_ice_connection_receiving_timeout_id);
>-  rtc_config->ice_backup_candidate_pair_ping_interval = GetIntField(
>-      jni, j_rtc_config, j_ice_backup_candidate_pair_ping_interval_id);
>-  rtc_config->continual_gathering_policy =
>-      JavaToNativeContinualGatheringPolicy(jni, j_continual_gathering_policy);
>-  rtc_config->ice_candidate_pool_size =
>-      GetIntField(jni, j_rtc_config, j_ice_candidate_pool_size_id);
>-  rtc_config->prune_turn_ports =
>-      GetBooleanField(jni, j_rtc_config, j_prune_turn_ports_id);
>-  rtc_config->presume_writable_when_fully_relayed = GetBooleanField(
>-      jni, j_rtc_config, j_presume_writable_when_fully_relayed_id);
>-  jobject j_ice_check_min_interval =
>-      GetNullableObjectField(jni, j_rtc_config, j_ice_check_min_interval_id);
>-  rtc_config->ice_check_min_interval =
>-      JavaToNativeOptionalInt(jni, j_ice_check_min_interval);
>-  rtc_config->disable_ipv6_on_wifi =
>-      GetBooleanField(jni, j_rtc_config, j_disable_ipv6_on_wifi_id);
>-  rtc_config->max_ipv6_networks =
>-      GetIntField(jni, j_rtc_config, j_max_ipv6_networks_id);
>-  jobject j_ice_regather_interval_range = GetNullableObjectField(
>-      jni, j_rtc_config, j_ice_regather_interval_range_id);
>-  if (!IsNull(jni, j_ice_regather_interval_range)) {
>-    int min = jni->CallIntMethod(j_ice_regather_interval_range, get_min_id);
>-    int max = jni->CallIntMethod(j_ice_regather_interval_range, get_max_id);
>-    rtc_config->ice_regather_interval_range.emplace(min, max);
>-  }
>-
>-  if (!IsNull(jni, j_turn_customizer)) {
>-    rtc_config->turn_customizer = reinterpret_cast<webrtc::TurnCustomizer*>(
>-        GetLongField(jni, j_turn_customizer, j_native_turn_customizer_id));
>-  }
>-}
>-
>-void JavaToNativeRtpParameters(JNIEnv* jni,
>-                               jobject j_parameters,
>-                               RtpParameters* parameters) {
>-  RTC_CHECK(parameters != nullptr);
>-  jclass parameters_class = jni->FindClass("org/webrtc/RtpParameters");
>-  jfieldID encodings_id =
>-      GetFieldID(jni, parameters_class, "encodings", "Ljava/util/List;");
>-  jfieldID codecs_id =
>-      GetFieldID(jni, parameters_class, "codecs", "Ljava/util/List;");
>-
>-  // Convert encodings.
>-  jobject j_encodings = GetObjectField(jni, j_parameters, encodings_id);
>-  jclass j_encoding_parameters_class =
>-      jni->FindClass("org/webrtc/RtpParameters$Encoding");
>-  jfieldID active_id =
>-      GetFieldID(jni, j_encoding_parameters_class, "active", "Z");
>-  jfieldID bitrate_id = GetFieldID(jni, j_encoding_parameters_class,
>-                                   "maxBitrateBps", "Ljava/lang/Integer;");
>-  jfieldID ssrc_id =
>-      GetFieldID(jni, j_encoding_parameters_class, "ssrc", "Ljava/lang/Long;");
>-  jclass j_long_class = jni->FindClass("java/lang/Long");
>-  jmethodID long_value_id = GetMethodID(jni, j_long_class, "longValue", "()J");
>-
>-  for (jobject j_encoding_parameters : Iterable(jni, j_encodings)) {
>-    RtpEncodingParameters encoding;
>-    encoding.active = GetBooleanField(jni, j_encoding_parameters, active_id);
>-    jobject j_bitrate =
>-        GetNullableObjectField(jni, j_encoding_parameters, bitrate_id);
>-    encoding.max_bitrate_bps = JavaToNativeOptionalInt(jni, j_bitrate);
>-    jobject j_ssrc =
>-        GetNullableObjectField(jni, j_encoding_parameters, ssrc_id);
>-    if (!IsNull(jni, j_ssrc)) {
>-      jlong ssrc_value = jni->CallLongMethod(j_ssrc, long_value_id);
>-      CHECK_EXCEPTION(jni) << "error during CallLongMethod";
>-      encoding.ssrc = rtc::Optional<uint32_t>(ssrc_value);
>-    }
>-    parameters->encodings.push_back(encoding);
>-  }
>-
>-  // Convert codecs.
>-  jobject j_codecs = GetObjectField(jni, j_parameters, codecs_id);
>-  jclass codec_class = jni->FindClass("org/webrtc/RtpParameters$Codec");
>-  jfieldID payload_type_id = GetFieldID(jni, codec_class, "payloadType", "I");
>-  jfieldID name_id = GetFieldID(jni, codec_class, "name", "Ljava/lang/String;");
>-  jfieldID kind_id = GetFieldID(jni, codec_class, "kind",
>-                                "Lorg/webrtc/MediaStreamTrack$MediaType;");
>-  jfieldID clock_rate_id =
>-      GetFieldID(jni, codec_class, "clockRate", "Ljava/lang/Integer;");
>-  jfieldID num_channels_id =
>-      GetFieldID(jni, codec_class, "numChannels", "Ljava/lang/Integer;");
>-
>-  for (jobject j_codec : Iterable(jni, j_codecs)) {
>-    RtpCodecParameters codec;
>-    codec.payload_type = GetIntField(jni, j_codec, payload_type_id);
>-    codec.name = JavaToStdString(jni, GetStringField(jni, j_codec, name_id));
>-    codec.kind =
>-        JavaToNativeMediaType(jni, GetObjectField(jni, j_codec, kind_id));
>-    jobject j_clock_rate = GetNullableObjectField(jni, j_codec, clock_rate_id);
>-    codec.clock_rate = JavaToNativeOptionalInt(jni, j_clock_rate);
>-    jobject j_num_channels =
>-        GetNullableObjectField(jni, j_codec, num_channels_id);
>-    codec.num_channels = JavaToNativeOptionalInt(jni, j_num_channels);
>-    parameters->codecs.push_back(codec);
>-  }
>-}
>-
>-jobject NativeToJavaRtpParameters(JNIEnv* jni,
>-                                  const RtpParameters& parameters) {
>-  jclass parameters_class = jni->FindClass("org/webrtc/RtpParameters");
>-  jmethodID parameters_ctor =
>-      GetMethodID(jni, parameters_class, "<init>", "()V");
>-  jobject j_parameters = jni->NewObject(parameters_class, parameters_ctor);
>-  CHECK_EXCEPTION(jni) << "error during NewObject";
>-
>-  // Add encodings.
>-  jclass encoding_class = jni->FindClass("org/webrtc/RtpParameters$Encoding");
>-  jmethodID encoding_ctor = GetMethodID(jni, encoding_class, "<init>", "()V");
>-  jfieldID encodings_id =
>-      GetFieldID(jni, parameters_class, "encodings", "Ljava/util/List;");
>-  jobject j_encodings = GetObjectField(jni, j_parameters, encodings_id);
>-  jmethodID encodings_add = GetMethodID(jni, GetObjectClass(jni, j_encodings),
>-                                        "add", "(Ljava/lang/Object;)Z");
>-  jfieldID active_id = GetFieldID(jni, encoding_class, "active", "Z");
>-  jfieldID bitrate_id =
>-      GetFieldID(jni, encoding_class, "maxBitrateBps", "Ljava/lang/Integer;");
>-  jfieldID ssrc_id =
>-      GetFieldID(jni, encoding_class, "ssrc", "Ljava/lang/Long;");
>-
>-  jclass long_class = jni->FindClass("java/lang/Long");
>-  jmethodID long_ctor = GetMethodID(jni, long_class, "<init>", "(J)V");
>-
>-  for (const RtpEncodingParameters& encoding : parameters.encodings) {
>-    jobject j_encoding_parameters =
>-        jni->NewObject(encoding_class, encoding_ctor);
>-    CHECK_EXCEPTION(jni) << "error during NewObject";
>-    jni->SetBooleanField(j_encoding_parameters, active_id, encoding.active);
>-    CHECK_EXCEPTION(jni) << "error during SetBooleanField";
>-    jni->SetObjectField(j_encoding_parameters, bitrate_id,
>-                        NativeToJavaInteger(jni, encoding.max_bitrate_bps));
>-    if (encoding.ssrc) {
>-      jobject j_ssrc_value = jni->NewObject(long_class, long_ctor,
>-                                            static_cast<jlong>(*encoding.ssrc));
>-      CHECK_EXCEPTION(jni) << "error during NewObject";
>-      jni->SetObjectField(j_encoding_parameters, ssrc_id, j_ssrc_value);
>-      CHECK_EXCEPTION(jni) << "error during SetObjectField";
>-    }
>-    jboolean added = jni->CallBooleanMethod(j_encodings, encodings_add,
>-                                            j_encoding_parameters);
>-    CHECK_EXCEPTION(jni) << "error during CallBooleanMethod";
>-    RTC_CHECK(added);
>-  }
>-
>-  // Add codecs.
>-  jclass codec_class = jni->FindClass("org/webrtc/RtpParameters$Codec");
>-  jmethodID codec_ctor = GetMethodID(jni, codec_class, "<init>", "()V");
>-  jfieldID codecs_id =
>-      GetFieldID(jni, parameters_class, "codecs", "Ljava/util/List;");
>-  jobject j_codecs = GetObjectField(jni, j_parameters, codecs_id);
>-  jmethodID codecs_add = GetMethodID(jni, GetObjectClass(jni, j_codecs), "add",
>-                                     "(Ljava/lang/Object;)Z");
>-  jfieldID payload_type_id = GetFieldID(jni, codec_class, "payloadType", "I");
>-  jfieldID name_id = GetFieldID(jni, codec_class, "name", "Ljava/lang/String;");
>-  jfieldID kind_id = GetFieldID(jni, codec_class, "kind",
>-                                "Lorg/webrtc/MediaStreamTrack$MediaType;");
>-  jfieldID clock_rate_id =
>-      GetFieldID(jni, codec_class, "clockRate", "Ljava/lang/Integer;");
>-  jfieldID num_channels_id =
>-      GetFieldID(jni, codec_class, "numChannels", "Ljava/lang/Integer;");
>-
>-  for (const RtpCodecParameters& codec : parameters.codecs) {
>-    jobject j_codec = jni->NewObject(codec_class, codec_ctor);
>-    CHECK_EXCEPTION(jni) << "error during NewObject";
>-    jni->SetIntField(j_codec, payload_type_id, codec.payload_type);
>-    CHECK_EXCEPTION(jni) << "error during SetIntField";
>-    jni->SetObjectField(j_codec, name_id, NativeToJavaString(jni, codec.name));
>-    CHECK_EXCEPTION(jni) << "error during SetObjectField";
>-    jni->SetObjectField(j_codec, kind_id,
>-                        NativeToJavaMediaType(jni, codec.kind));
>-    CHECK_EXCEPTION(jni) << "error during SetObjectField";
>-    jni->SetObjectField(j_codec, clock_rate_id,
>-                        NativeToJavaInteger(jni, codec.clock_rate));
>-    jni->SetObjectField(j_codec, num_channels_id,
>-                        NativeToJavaInteger(jni, codec.num_channels));
>-    jboolean added = jni->CallBooleanMethod(j_codecs, codecs_add, j_codec);
>-    CHECK_EXCEPTION(jni) << "error during CallBooleanMethod";
>-    RTC_CHECK(added);
>-  }
>-
>-  return j_parameters;
>-}
>-
>-}  // namespace jni
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/java_native_conversion.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/java_native_conversion.h
>deleted file mode 100644
>index faa4084aab6..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/java_native_conversion.h
>+++ /dev/null
>@@ -1,110 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef SDK_ANDROID_SRC_JNI_PC_JAVA_NATIVE_CONVERSION_H_
>-#define SDK_ANDROID_SRC_JNI_PC_JAVA_NATIVE_CONVERSION_H_
>-
>-#include <vector>
>-
>-#include "api/datachannelinterface.h"
>-#include "api/jsep.h"
>-#include "api/jsepicecandidate.h"
>-#include "api/mediastreaminterface.h"
>-#include "api/mediatypes.h"
>-#include "api/peerconnectioninterface.h"
>-#include "api/rtpparameters.h"
>-#include "rtc_base/sslidentity.h"
>-#include "sdk/android/src/jni/jni_helpers.h"
>-
>-// This file contains helper methods for converting between simple C++ and Java
>-// PeerConnection-related structures. Similar to some methods in jni_helpers.h,
>-// but specifically for structures tied to the PeerConnection API.
>-
>-namespace webrtc {
>-namespace jni {
>-
>-cricket::MediaType JavaToNativeMediaType(JNIEnv* jni, jobject j_media_type);
>-
>-jobject NativeToJavaMediaType(JNIEnv* jni, cricket::MediaType media_type);
>-
>-cricket::Candidate JavaToNativeCandidate(JNIEnv* jni, jobject j_candidate);
>-
>-jobject NativeToJavaCandidate(JNIEnv* env, const cricket::Candidate& candidate);
>-
>-jobject NativeToJavaIceCandidate(JNIEnv* env,
>-                                 const IceCandidateInterface& candidate);
>-
>-jobjectArray NativeToJavaCandidateArray(
>-    JNIEnv* jni,
>-    const std::vector<cricket::Candidate>& candidates);
>-
>-SessionDescriptionInterface* JavaToNativeSessionDescription(JNIEnv* jni,
>-                                                            jobject j_sdp);
>-
>-jobject NativeToJavaSessionDescription(JNIEnv* jni,
>-                                       const SessionDescriptionInterface* desc);
>-
>-PeerConnectionFactoryInterface::Options
>-JavaToNativePeerConnectionFactoryOptions(JNIEnv* jni, jobject options);
>-
>-/*****************************************************
>- * Below are all things that go into RTCConfiguration.
>- *****************************************************/
>-PeerConnectionInterface::IceTransportsType JavaToNativeIceTransportsType(
>-    JNIEnv* jni,
>-    jobject j_ice_transports_type);
>-
>-PeerConnectionInterface::BundlePolicy JavaToNativeBundlePolicy(
>-    JNIEnv* jni,
>-    jobject j_bundle_policy);
>-
>-PeerConnectionInterface::RtcpMuxPolicy JavaToNativeRtcpMuxPolicy(
>-    JNIEnv* jni,
>-    jobject j_rtcp_mux_policy);
>-
>-PeerConnectionInterface::TcpCandidatePolicy JavaToNativeTcpCandidatePolicy(
>-    JNIEnv* jni,
>-    jobject j_tcp_candidate_policy);
>-
>-PeerConnectionInterface::CandidateNetworkPolicy
>-JavaToNativeCandidateNetworkPolicy(JNIEnv* jni,
>-                                   jobject j_candidate_network_policy);
>-
>-rtc::KeyType JavaToNativeKeyType(JNIEnv* jni, jobject j_key_type);
>-
>-PeerConnectionInterface::ContinualGatheringPolicy
>-JavaToNativeContinualGatheringPolicy(JNIEnv* jni, jobject j_gathering_policy);
>-
>-PeerConnectionInterface::TlsCertPolicy JavaToNativeTlsCertPolicy(
>-    JNIEnv* jni,
>-    jobject j_ice_server_tls_cert_policy);
>-
>-void JavaToNativeIceServers(JNIEnv* jni,
>-                            jobject j_ice_servers,
>-                            PeerConnectionInterface::IceServers* ice_servers);
>-
>-void JavaToNativeRTCConfiguration(
>-    JNIEnv* jni,
>-    jobject j_rtc_config,
>-    PeerConnectionInterface::RTCConfiguration* rtc_config);
>-
>-/*********************************************************
>- * RtpParameters, used for RtpSender and RtpReceiver APIs.
>- *********************************************************/
>-void JavaToNativeRtpParameters(JNIEnv* jni,
>-                               jobject j_parameters,
>-                               RtpParameters* parameters);
>-
>-jobject NativeToJavaRtpParameters(JNIEnv* jni, const RtpParameters& parameters);
>-
>-}  // namespace jni
>-}  // namespace webrtc
>-
>-#endif  // SDK_ANDROID_SRC_JNI_PC_JAVA_NATIVE_CONVERSION_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/logging_jni.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/logging.cc
>similarity index 88%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/logging_jni.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/logging.cc
>index 75256439f39..4fb8215b1e0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/logging_jni.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/logging.cc
>@@ -11,6 +11,7 @@
> #include <memory>
> 
> #include "rtc_base/logging.h"
>+#include "sdk/android/native_api/jni/java_types.h"
> #include "sdk/android/src/jni/jni_helpers.h"
> 
> namespace webrtc {
>@@ -48,9 +49,10 @@ JNI_FUNCTION_DECLARATION(void,
>                          jint j_severity,
>                          jstring j_tag,
>                          jstring j_message) {
>-  std::string message = JavaToStdString(jni, j_message);
>-  std::string tag = JavaToStdString(jni, j_tag);
>-  RTC_LOG_TAG(static_cast<rtc::LoggingSeverity>(j_severity), tag) << message;
>+  std::string message = JavaToStdString(jni, JavaParamRef<jstring>(j_message));
>+  std::string tag = JavaToStdString(jni, JavaParamRef<jstring>(j_tag));
>+  RTC_LOG_TAG(static_cast<rtc::LoggingSeverity>(j_severity), tag.c_str())
>+      << message;
> }
> 
> }  // namespace jni
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/media_jni.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/media.cc
>similarity index 70%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/media_jni.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/media.cc
>index 84507753001..dba85fbffa1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/media_jni.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/media.cc
>@@ -7,11 +7,13 @@
>  *  in the file PATENTS.  All contributing project authors may
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
>-#include "sdk/android/src/jni/pc/media_jni.h"
>+#include "sdk/android/src/jni/pc/media.h"
> 
>+#include <utility>
>+
>+#include "api/call/callfactoryinterface.h"
> #include "api/video_codecs/video_decoder_factory.h"
> #include "api/video_codecs/video_encoder_factory.h"
>-#include "call/callfactoryinterface.h"
> #include "logging/rtc_event_log/rtc_event_log_factory_interface.h"
> #include "media/engine/webrtcmediaengine.h"
> #include "modules/audio_device/include/audio_device.h"
>@@ -28,19 +30,6 @@ RtcEventLogFactoryInterface* CreateRtcEventLogFactory() {
>   return webrtc::CreateRtcEventLogFactory().release();
> }
> 
>-cricket::MediaEngineInterface* CreateMediaEngine(
>-    AudioDeviceModule* adm,
>-    const rtc::scoped_refptr<AudioEncoderFactory>& audio_encoder_factory,
>-    const rtc::scoped_refptr<AudioDecoderFactory>& audio_decoder_factory,
>-    cricket::WebRtcVideoEncoderFactory* video_encoder_factory,
>-    cricket::WebRtcVideoDecoderFactory* video_decoder_factory,
>-    rtc::scoped_refptr<AudioMixer> audio_mixer,
>-    rtc::scoped_refptr<AudioProcessing> audio_processor) {
>-  return cricket::WebRtcMediaEngineFactory::Create(
>-      adm, audio_encoder_factory, audio_decoder_factory, video_encoder_factory,
>-      video_decoder_factory, audio_mixer, audio_processor);
>-}
>-
> cricket::MediaEngineInterface* CreateMediaEngine(
>     rtc::scoped_refptr<AudioDeviceModule> adm,
>     rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/media_jni.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/media.h
>similarity index 68%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/media_jni.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/media.h
>index 6ed0e7e1d55..8a0bb521743 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/media_jni.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/media.h
>@@ -8,8 +8,10 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#ifndef SDK_ANDROID_SRC_JNI_PC_MEDIA_JNI_H_
>-#define SDK_ANDROID_SRC_JNI_PC_MEDIA_JNI_H_
>+#ifndef SDK_ANDROID_SRC_JNI_PC_MEDIA_H_
>+#define SDK_ANDROID_SRC_JNI_PC_MEDIA_H_
>+
>+#include <memory>
> 
> #include "rtc_base/scoped_ref_ptr.h"
> 
>@@ -27,8 +29,6 @@ class VideoDecoderFactory;
> 
> namespace cricket {
> class MediaEngineInterface;
>-class WebRtcVideoEncoderFactory;
>-class WebRtcVideoDecoderFactory;
> }  // namespace cricket
> 
> namespace webrtc {
>@@ -37,15 +37,6 @@ namespace jni {
> CallFactoryInterface* CreateCallFactory();
> RtcEventLogFactoryInterface* CreateRtcEventLogFactory();
> 
>-cricket::MediaEngineInterface* CreateMediaEngine(
>-    AudioDeviceModule* adm,
>-    const rtc::scoped_refptr<AudioEncoderFactory>& audio_encoder_factory,
>-    const rtc::scoped_refptr<AudioDecoderFactory>& audio_decoder_factory,
>-    cricket::WebRtcVideoEncoderFactory* video_encoder_factory,
>-    cricket::WebRtcVideoDecoderFactory* video_decoder_factory,
>-    rtc::scoped_refptr<AudioMixer> audio_mixer,
>-    rtc::scoped_refptr<AudioProcessing> audio_processor);
>-
> cricket::MediaEngineInterface* CreateMediaEngine(
>     rtc::scoped_refptr<AudioDeviceModule> adm,
>     rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
>@@ -58,4 +49,4 @@ cricket::MediaEngineInterface* CreateMediaEngine(
> }  // namespace jni
> }  // namespace webrtc
> 
>-#endif  // SDK_ANDROID_SRC_JNI_PC_MEDIA_JNI_H_
>+#endif  // SDK_ANDROID_SRC_JNI_PC_MEDIA_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/mediaconstraints_jni.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/mediaconstraints.cc
>similarity index 73%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/mediaconstraints_jni.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/mediaconstraints.cc
>index b7d05a2428d..e7a4bb50376 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/mediaconstraints_jni.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/mediaconstraints.cc
>@@ -8,10 +8,11 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "sdk/android/src/jni/pc/mediaconstraints_jni.h"
>+#include "sdk/android/src/jni/pc/mediaconstraints.h"
> 
>-#include "rtc_base/ptr_util.h"
>+#include "absl/memory/memory.h"
> #include "sdk/android/generated_peerconnection_jni/jni/MediaConstraints_jni.h"
>+#include "sdk/android/native_api/jni/java_types.h"
> #include "sdk/android/src/jni/jni_helpers.h"
> 
> namespace webrtc {
>@@ -22,13 +23,12 @@ namespace {
> // Helper for translating a List<Pair<String, String>> to a Constraints.
> MediaConstraintsInterface::Constraints PopulateConstraintsFromJavaPairList(
>     JNIEnv* env,
>-    jobject j_list) {
>+    const JavaRef<jobject>& j_list) {
>   MediaConstraintsInterface::Constraints constraints;
>-  for (jobject entry : Iterable(env, j_list)) {
>-    jstring j_key = Java_KeyValuePair_getKey(env, entry);
>-    jstring j_value = Java_KeyValuePair_getValue(env, entry);
>-    constraints.emplace_back(JavaToStdString(env, j_key),
>-                             JavaToStdString(env, j_value));
>+  for (const JavaRef<jobject>& entry : Iterable(env, j_list)) {
>+    constraints.emplace_back(
>+        JavaToStdString(env, Java_KeyValuePair_getKey(env, entry)),
>+        JavaToStdString(env, Java_KeyValuePair_getValue(env, entry)));
>   }
>   return constraints;
> }
>@@ -37,14 +37,14 @@ MediaConstraintsInterface::Constraints PopulateConstraintsFromJavaPairList(
> // the constructor returns the Java object is no longer needed.
> class MediaConstraintsJni : public MediaConstraintsInterface {
>  public:
>-  MediaConstraintsJni(JNIEnv* env, jobject j_constraints)
>+  MediaConstraintsJni(JNIEnv* env, const JavaRef<jobject>& j_constraints)
>       : mandatory_(PopulateConstraintsFromJavaPairList(
>             env,
>             Java_MediaConstraints_getMandatory(env, j_constraints))),
>         optional_(PopulateConstraintsFromJavaPairList(
>             env,
>             Java_MediaConstraints_getOptional(env, j_constraints))) {}
>-  virtual ~MediaConstraintsJni() = default;
>+  ~MediaConstraintsJni() override = default;
> 
>   // MediaConstraintsInterface.
>   const Constraints& GetMandatory() const override { return mandatory_; }
>@@ -59,8 +59,8 @@ class MediaConstraintsJni : public MediaConstraintsInterface {
> 
> std::unique_ptr<MediaConstraintsInterface> JavaToNativeMediaConstraints(
>     JNIEnv* env,
>-    jobject j_constraints) {
>-  return rtc::MakeUnique<MediaConstraintsJni>(env, j_constraints);
>+    const JavaRef<jobject>& j_constraints) {
>+  return absl::make_unique<MediaConstraintsJni>(env, j_constraints);
> }
> 
> }  // namespace jni
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/mediaconstraints_jni.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/mediaconstraints.h
>similarity index 70%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/mediaconstraints_jni.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/mediaconstraints.h
>index ce7d90dc899..2fbf34e74a0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/mediaconstraints_jni.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/mediaconstraints.h
>@@ -8,21 +8,23 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#ifndef SDK_ANDROID_SRC_JNI_PC_MEDIACONSTRAINTS_JNI_H_
>-#define SDK_ANDROID_SRC_JNI_PC_MEDIACONSTRAINTS_JNI_H_
>+#ifndef SDK_ANDROID_SRC_JNI_PC_MEDIACONSTRAINTS_H_
>+#define SDK_ANDROID_SRC_JNI_PC_MEDIACONSTRAINTS_H_
> 
> #include <jni.h>
>+#include <memory>
> 
> #include "api/mediaconstraintsinterface.h"
>+#include "sdk/android/native_api/jni/scoped_java_ref.h"
> 
> namespace webrtc {
> namespace jni {
> 
> std::unique_ptr<MediaConstraintsInterface> JavaToNativeMediaConstraints(
>     JNIEnv* env,
>-    jobject j_constraints);
>+    const JavaRef<jobject>& j_constraints);
> 
> }  // namespace jni
> }  // namespace webrtc
> 
>-#endif  // SDK_ANDROID_SRC_JNI_PC_MEDIACONSTRAINTS_JNI_H_
>+#endif  // SDK_ANDROID_SRC_JNI_PC_MEDIACONSTRAINTS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/mediasource.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/mediasource.cc
>new file mode 100644
>index 00000000000..dfd2ff39cb6
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/mediasource.cc
>@@ -0,0 +1,24 @@
>+/*
>+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/mediastreaminterface.h"
>+#include "sdk/android/generated_peerconnection_jni/jni/MediaSource_jni.h"
>+
>+namespace webrtc {
>+namespace jni {
>+
>+static ScopedJavaLocalRef<jobject>
>+JNI_MediaSource_GetState(JNIEnv* jni, const JavaParamRef<jclass>&, jlong j_p) {
>+  return Java_State_fromNativeIndex(
>+      jni, reinterpret_cast<MediaSourceInterface*>(j_p)->state());
>+}
>+
>+}  // namespace jni
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/mediastream.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/mediastream.cc
>new file mode 100644
>index 00000000000..d70996bf0b2
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/mediastream.cc
>@@ -0,0 +1,144 @@
>+/*
>+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "sdk/android/src/jni/pc/mediastream.h"
>+
>+#include "absl/memory/memory.h"
>+#include "sdk/android/generated_peerconnection_jni/jni/MediaStream_jni.h"
>+#include "sdk/android/native_api/jni/java_types.h"
>+#include "sdk/android/src/jni/jni_helpers.h"
>+
>+namespace webrtc {
>+namespace jni {
>+
>+JavaMediaStream::JavaMediaStream(
>+    JNIEnv* env,
>+    rtc::scoped_refptr<MediaStreamInterface> media_stream)
>+    : j_media_stream_(
>+          env,
>+          Java_MediaStream_Constructor(env,
>+                                       jlongFromPointer(media_stream.get()))),
>+      observer_(absl::make_unique<MediaStreamObserver>(media_stream)) {
>+  for (rtc::scoped_refptr<AudioTrackInterface> track :
>+       media_stream->GetAudioTracks()) {
>+    Java_MediaStream_addNativeAudioTrack(env, j_media_stream_,
>+                                         jlongFromPointer(track.release()));
>+  }
>+  for (rtc::scoped_refptr<VideoTrackInterface> track :
>+       media_stream->GetVideoTracks()) {
>+    Java_MediaStream_addNativeVideoTrack(env, j_media_stream_,
>+                                         jlongFromPointer(track.release()));
>+  }
>+
>+  // Create an observer to update the Java stream when the native stream's set
>+  // of tracks changes.
>+  observer_->SignalAudioTrackRemoved.connect(
>+      this, &JavaMediaStream::OnAudioTrackRemovedFromStream);
>+  observer_->SignalVideoTrackRemoved.connect(
>+      this, &JavaMediaStream::OnVideoTrackRemovedFromStream);
>+  observer_->SignalAudioTrackAdded.connect(
>+      this, &JavaMediaStream::OnAudioTrackAddedToStream);
>+  observer_->SignalVideoTrackAdded.connect(
>+      this, &JavaMediaStream::OnVideoTrackAddedToStream);
>+
>+  // |j_media_stream| holds one reference. Corresponding Release() is in
>+  // MediaStream_free, triggered by MediaStream.dispose().
>+  media_stream.release();
>+}
>+
>+JavaMediaStream::~JavaMediaStream() {
>+  JNIEnv* env = AttachCurrentThreadIfNeeded();
>+  // Remove the observer first, so it doesn't react to events during deletion.
>+  observer_ = nullptr;
>+  Java_MediaStream_dispose(env, j_media_stream_);
>+}
>+
>+void JavaMediaStream::OnAudioTrackAddedToStream(AudioTrackInterface* track,
>+                                                MediaStreamInterface* stream) {
>+  JNIEnv* env = AttachCurrentThreadIfNeeded();
>+  ScopedLocalRefFrame local_ref_frame(env);
>+  track->AddRef();
>+  Java_MediaStream_addNativeAudioTrack(env, j_media_stream_,
>+                                       jlongFromPointer(track));
>+}
>+
>+void JavaMediaStream::OnVideoTrackAddedToStream(VideoTrackInterface* track,
>+                                                MediaStreamInterface* stream) {
>+  JNIEnv* env = AttachCurrentThreadIfNeeded();
>+  ScopedLocalRefFrame local_ref_frame(env);
>+  track->AddRef();
>+  Java_MediaStream_addNativeVideoTrack(env, j_media_stream_,
>+                                       jlongFromPointer(track));
>+}
>+
>+void JavaMediaStream::OnAudioTrackRemovedFromStream(
>+    AudioTrackInterface* track,
>+    MediaStreamInterface* stream) {
>+  JNIEnv* env = AttachCurrentThreadIfNeeded();
>+  ScopedLocalRefFrame local_ref_frame(env);
>+  Java_MediaStream_removeAudioTrack(env, j_media_stream_,
>+                                    jlongFromPointer(track));
>+}
>+
>+void JavaMediaStream::OnVideoTrackRemovedFromStream(
>+    VideoTrackInterface* track,
>+    MediaStreamInterface* stream) {
>+  JNIEnv* env = AttachCurrentThreadIfNeeded();
>+  ScopedLocalRefFrame local_ref_frame(env);
>+  Java_MediaStream_removeVideoTrack(env, j_media_stream_,
>+                                    jlongFromPointer(track));
>+}
>+
>+jclass GetMediaStreamClass(JNIEnv* env) {
>+  return org_webrtc_MediaStream_clazz(env);
>+}
>+
>+static jboolean JNI_MediaStream_AddAudioTrackToNativeStream(
>+    JNIEnv* jni,
>+    const JavaParamRef<jclass>&,
>+    jlong pointer,
>+    jlong j_audio_track_pointer) {
>+  return reinterpret_cast<MediaStreamInterface*>(pointer)->AddTrack(
>+      reinterpret_cast<AudioTrackInterface*>(j_audio_track_pointer));
>+}
>+
>+static jboolean JNI_MediaStream_AddVideoTrackToNativeStream(
>+    JNIEnv* jni,
>+    const JavaParamRef<jclass>&,
>+    jlong pointer,
>+    jlong j_video_track_pointer) {
>+  return reinterpret_cast<MediaStreamInterface*>(pointer)->AddTrack(
>+      reinterpret_cast<VideoTrackInterface*>(j_video_track_pointer));
>+}
>+
>+static jboolean JNI_MediaStream_RemoveAudioTrack(JNIEnv* jni,
>+                                                 const JavaParamRef<jclass>&,
>+                                                 jlong pointer,
>+                                                 jlong j_audio_track_pointer) {
>+  return reinterpret_cast<MediaStreamInterface*>(pointer)->RemoveTrack(
>+      reinterpret_cast<AudioTrackInterface*>(j_audio_track_pointer));
>+}
>+
>+static jboolean JNI_MediaStream_RemoveVideoTrack(JNIEnv* jni,
>+                                                 const JavaParamRef<jclass>&,
>+                                                 jlong pointer,
>+                                                 jlong j_video_track_pointer) {
>+  return reinterpret_cast<MediaStreamInterface*>(pointer)->RemoveTrack(
>+      reinterpret_cast<VideoTrackInterface*>(j_video_track_pointer));
>+}
>+
>+static ScopedJavaLocalRef<jstring>
>+JNI_MediaStream_GetId(JNIEnv* jni, const JavaParamRef<jclass>&, jlong j_p) {
>+  return NativeToJavaString(jni,
>+                            reinterpret_cast<MediaStreamInterface*>(j_p)->id());
>+}
>+
>+}  // namespace jni
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/mediastream.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/mediastream.h
>new file mode 100644
>index 00000000000..479ab11e0c3
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/mediastream.h
>@@ -0,0 +1,54 @@
>+/*
>+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef SDK_ANDROID_SRC_JNI_PC_MEDIASTREAM_H_
>+#define SDK_ANDROID_SRC_JNI_PC_MEDIASTREAM_H_
>+
>+#include <jni.h>
>+#include <memory>
>+
>+#include "api/mediastreaminterface.h"
>+#include "pc/mediastreamobserver.h"
>+#include "sdk/android/src/jni/jni_helpers.h"
>+
>+namespace webrtc {
>+namespace jni {
>+
>+class JavaMediaStream : public sigslot::has_slots<> {
>+ public:
>+  explicit JavaMediaStream(
>+      JNIEnv* env,
>+      rtc::scoped_refptr<MediaStreamInterface> media_stream);
>+  ~JavaMediaStream() override;
>+
>+  const ScopedJavaGlobalRef<jobject>& j_media_stream() {
>+    return j_media_stream_;
>+  }
>+
>+ private:
>+  void OnAudioTrackAddedToStream(AudioTrackInterface* track,
>+                                 MediaStreamInterface* stream);
>+  void OnVideoTrackAddedToStream(VideoTrackInterface* track,
>+                                 MediaStreamInterface* stream);
>+  void OnAudioTrackRemovedFromStream(AudioTrackInterface* track,
>+                                     MediaStreamInterface* stream);
>+  void OnVideoTrackRemovedFromStream(VideoTrackInterface* track,
>+                                     MediaStreamInterface* stream);
>+
>+  ScopedJavaGlobalRef<jobject> j_media_stream_;
>+  std::unique_ptr<MediaStreamObserver> observer_;
>+};
>+
>+jclass GetMediaStreamClass(JNIEnv* env);
>+
>+}  // namespace jni
>+}  // namespace webrtc
>+
>+#endif  // SDK_ANDROID_SRC_JNI_PC_MEDIASTREAM_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/mediastream_jni.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/mediastream_jni.cc
>deleted file mode 100644
>index 92f4c9d0c42..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/mediastream_jni.cc
>+++ /dev/null
>@@ -1,71 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "api/mediastreaminterface.h"
>-#include "sdk/android/src/jni/jni_helpers.h"
>-
>-namespace webrtc {
>-namespace jni {
>-
>-JNI_FUNCTION_DECLARATION(jboolean,
>-                         MediaStream_addAudioTrackToNativeStream,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jlong pointer,
>-                         jlong j_audio_track_pointer) {
>-  return reinterpret_cast<MediaStreamInterface*>(pointer)->AddTrack(
>-      reinterpret_cast<AudioTrackInterface*>(j_audio_track_pointer));
>-}
>-
>-JNI_FUNCTION_DECLARATION(jboolean,
>-                         MediaStream_addVideoTrackToNativeStream,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jlong pointer,
>-                         jlong j_video_track_pointer) {
>-  return reinterpret_cast<MediaStreamInterface*>(pointer)->AddTrack(
>-      reinterpret_cast<VideoTrackInterface*>(j_video_track_pointer));
>-}
>-
>-JNI_FUNCTION_DECLARATION(jboolean,
>-                         MediaStream_removeNativeAudioTrack,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jlong pointer,
>-                         jlong j_audio_track_pointer) {
>-  return reinterpret_cast<MediaStreamInterface*>(pointer)->RemoveTrack(
>-      reinterpret_cast<AudioTrackInterface*>(j_audio_track_pointer));
>-}
>-
>-JNI_FUNCTION_DECLARATION(jboolean,
>-                         MediaStream_removeNativeVideoTrack,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jlong pointer,
>-                         jlong j_video_track_pointer) {
>-  return reinterpret_cast<MediaStreamInterface*>(pointer)->RemoveTrack(
>-      reinterpret_cast<VideoTrackInterface*>(j_video_track_pointer));
>-}
>-
>-JNI_FUNCTION_DECLARATION(jstring,
>-                         MediaStream_getNativeLabel,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jlong j_p) {
>-  return NativeToJavaString(
>-      jni, reinterpret_cast<MediaStreamInterface*>(j_p)->label());
>-}
>-
>-JNI_FUNCTION_DECLARATION(void, MediaStream_free, JNIEnv*, jclass, jlong j_p) {
>-  reinterpret_cast<MediaStreamInterface*>(j_p)->Release();
>-}
>-
>-}  // namespace jni
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/mediastreamtrack.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/mediastreamtrack.cc
>new file mode 100644
>index 00000000000..070b9d68521
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/mediastreamtrack.cc
>@@ -0,0 +1,72 @@
>+/*
>+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "sdk/android/src/jni/pc/mediastreamtrack.h"
>+
>+#include "api/mediastreaminterface.h"
>+#include "sdk/android/generated_peerconnection_jni/jni/MediaStreamTrack_jni.h"
>+#include "sdk/android/native_api/jni/java_types.h"
>+#include "sdk/android/src/jni/jni_helpers.h"
>+
>+namespace webrtc {
>+namespace jni {
>+
>+ScopedJavaLocalRef<jobject> NativeToJavaMediaType(
>+    JNIEnv* jni,
>+    cricket::MediaType media_type) {
>+  return Java_MediaType_fromNativeIndex(jni, media_type);
>+}
>+
>+cricket::MediaType JavaToNativeMediaType(JNIEnv* jni,
>+                                         const JavaRef<jobject>& j_media_type) {
>+  return static_cast<cricket::MediaType>(
>+      Java_MediaType_getNative(jni, j_media_type));
>+}
>+
>+static ScopedJavaLocalRef<jstring> JNI_MediaStreamTrack_GetId(
>+    JNIEnv* jni,
>+    const JavaParamRef<jclass>&,
>+    jlong j_p) {
>+  return NativeToJavaString(
>+      jni, reinterpret_cast<MediaStreamTrackInterface*>(j_p)->id());
>+}
>+
>+static ScopedJavaLocalRef<jstring> JNI_MediaStreamTrack_GetKind(
>+    JNIEnv* jni,
>+    const JavaParamRef<jclass>&,
>+    jlong j_p) {
>+  return NativeToJavaString(
>+      jni, reinterpret_cast<MediaStreamTrackInterface*>(j_p)->kind());
>+}
>+
>+static jboolean JNI_MediaStreamTrack_GetEnabled(JNIEnv* jni,
>+                                                const JavaParamRef<jclass>&,
>+                                                jlong j_p) {
>+  return reinterpret_cast<MediaStreamTrackInterface*>(j_p)->enabled();
>+}
>+
>+static ScopedJavaLocalRef<jobject> JNI_MediaStreamTrack_GetState(
>+    JNIEnv* jni,
>+    const JavaParamRef<jclass>&,
>+    jlong j_p) {
>+  return Java_State_fromNativeIndex(
>+      jni, reinterpret_cast<MediaStreamTrackInterface*>(j_p)->state());
>+}
>+
>+static jboolean JNI_MediaStreamTrack_SetEnabled(JNIEnv* jni,
>+                                                const JavaParamRef<jclass>&,
>+                                                jlong j_p,
>+                                                jboolean enabled) {
>+  return reinterpret_cast<MediaStreamTrackInterface*>(j_p)->set_enabled(
>+      enabled);
>+}
>+
>+}  // namespace jni
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/mediastreamtrack.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/mediastreamtrack.h
>new file mode 100644
>index 00000000000..8af425c9a41
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/mediastreamtrack.h
>@@ -0,0 +1,31 @@
>+/*
>+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef SDK_ANDROID_SRC_JNI_PC_MEDIASTREAMTRACK_H_
>+#define SDK_ANDROID_SRC_JNI_PC_MEDIASTREAMTRACK_H_
>+
>+#include <jni.h>
>+
>+#include "api/mediatypes.h"
>+#include "sdk/android/native_api/jni/scoped_java_ref.h"
>+
>+namespace webrtc {
>+namespace jni {
>+
>+ScopedJavaLocalRef<jobject> NativeToJavaMediaType(
>+    JNIEnv* jni,
>+    cricket::MediaType media_type);
>+cricket::MediaType JavaToNativeMediaType(JNIEnv* jni,
>+                                         const JavaRef<jobject>& j_media_type);
>+
>+}  // namespace jni
>+}  // namespace webrtc
>+
>+#endif  // SDK_ANDROID_SRC_JNI_PC_MEDIASTREAMTRACK_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/mediastreamtrack_jni.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/mediastreamtrack_jni.cc
>deleted file mode 100644
>index 52c3794a3c4..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/mediastreamtrack_jni.cc
>+++ /dev/null
>@@ -1,64 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "api/mediastreaminterface.h"
>-#include "sdk/android/src/jni/jni_helpers.h"
>-
>-namespace webrtc {
>-namespace jni {
>-
>-JNI_FUNCTION_DECLARATION(jstring,
>-                         MediaStreamTrack_getNativeId,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jlong j_p) {
>-  return NativeToJavaString(
>-      jni, reinterpret_cast<MediaStreamTrackInterface*>(j_p)->id());
>-}
>-
>-JNI_FUNCTION_DECLARATION(jstring,
>-                         MediaStreamTrack_getNativeKind,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jlong j_p) {
>-  return NativeToJavaString(
>-      jni, reinterpret_cast<MediaStreamTrackInterface*>(j_p)->kind());
>-}
>-
>-JNI_FUNCTION_DECLARATION(jboolean,
>-                         MediaStreamTrack_getNativeEnabled,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jlong j_p) {
>-  return reinterpret_cast<MediaStreamTrackInterface*>(j_p)->enabled();
>-}
>-
>-JNI_FUNCTION_DECLARATION(jobject,
>-                         MediaStreamTrack_getNativeState,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jlong j_p) {
>-  return JavaEnumFromIndexAndClassName(
>-      jni, "MediaStreamTrack$State",
>-      reinterpret_cast<MediaStreamTrackInterface*>(j_p)->state());
>-}
>-
>-JNI_FUNCTION_DECLARATION(jboolean,
>-                         MediaStreamTrack_setNativeEnabled,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jlong j_p,
>-                         jboolean enabled) {
>-  return reinterpret_cast<MediaStreamTrackInterface*>(j_p)->set_enabled(
>-      enabled);
>-}
>-
>-}  // namespace jni
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/null_audio_jni.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/null_audio.cc
>similarity index 94%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/null_audio_jni.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/null_audio.cc
>index 0482a680847..8d3e4960189 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/null_audio_jni.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/null_audio.cc
>@@ -8,7 +8,7 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "sdk/android/src/jni/pc/audio_jni.h"
>+#include "sdk/android/src/jni/pc/audio.h"
> 
> namespace webrtc {
> namespace jni {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/null_media_jni.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/null_media.cc
>similarity index 68%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/null_media_jni.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/null_media.cc
>index 25e5124771a..c59c476d51b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/null_media_jni.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/null_media.cc
>@@ -8,7 +8,7 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "sdk/android/src/jni/pc/media_jni.h"
>+#include "sdk/android/src/jni/pc/media.h"
> 
> namespace webrtc {
> namespace jni {
>@@ -21,17 +21,6 @@ RtcEventLogFactoryInterface* CreateRtcEventLogFactory() {
>   return nullptr;
> }
> 
>-cricket::MediaEngineInterface* CreateMediaEngine(
>-    AudioDeviceModule* adm,
>-    const rtc::scoped_refptr<AudioEncoderFactory>& audio_encoder_factory,
>-    const rtc::scoped_refptr<AudioDecoderFactory>& audio_decoder_factory,
>-    cricket::WebRtcVideoEncoderFactory* video_encoder_factory,
>-    cricket::WebRtcVideoDecoderFactory* video_decoder_factory,
>-    rtc::scoped_refptr<AudioMixer> audio_mixer,
>-    rtc::scoped_refptr<AudioProcessing> audio_processor) {
>-  return nullptr;
>-}
>-
> cricket::MediaEngineInterface* CreateMediaEngine(
>     rtc::scoped_refptr<AudioDeviceModule> adm,
>     rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/null_video.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/null_video.cc
>new file mode 100644
>index 00000000000..0e0b80d4612
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/null_video.cc
>@@ -0,0 +1,39 @@
>+/*
>+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "sdk/android/src/jni/pc/video.h"
>+
>+#include "api/video_codecs/video_decoder_factory.h"
>+#include "api/video_codecs/video_encoder_factory.h"
>+
>+namespace webrtc {
>+namespace jni {
>+
>+VideoEncoderFactory* CreateVideoEncoderFactory(
>+    JNIEnv* jni,
>+    const JavaRef<jobject>& j_encoder_factory) {
>+  return nullptr;
>+}
>+
>+VideoDecoderFactory* CreateVideoDecoderFactory(
>+    JNIEnv* jni,
>+    const JavaRef<jobject>& j_decoder_factory) {
>+  return nullptr;
>+}
>+
>+void* CreateVideoSource(JNIEnv* env,
>+                        rtc::Thread* signaling_thread,
>+                        rtc::Thread* worker_thread,
>+                        jboolean is_screencast) {
>+  return nullptr;
>+}
>+
>+}  // namespace jni
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/null_video_jni.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/null_video_jni.cc
>deleted file mode 100644
>index 88ed665b147..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/null_video_jni.cc
>+++ /dev/null
>@@ -1,50 +0,0 @@
>-/*
>- *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "sdk/android/src/jni/pc/video_jni.h"
>-
>-namespace webrtc {
>-namespace jni {
>-
>-VideoEncoderFactory* CreateVideoEncoderFactory(JNIEnv* jni,
>-                                               jobject j_encoder_factory) {
>-  return nullptr;
>-}
>-
>-VideoDecoderFactory* CreateVideoDecoderFactory(JNIEnv* jni,
>-                                               jobject j_decoder_factory) {
>-  return nullptr;
>-}
>-
>-cricket::WebRtcVideoEncoderFactory* CreateLegacyVideoEncoderFactory() {
>-  return nullptr;
>-}
>-
>-cricket::WebRtcVideoDecoderFactory* CreateLegacyVideoDecoderFactory() {
>-  return nullptr;
>-}
>-
>-VideoEncoderFactory* WrapLegacyVideoEncoderFactory(
>-    cricket::WebRtcVideoEncoderFactory* legacy_encoder_factory) {
>-  return nullptr;
>-}
>-
>-VideoDecoderFactory* WrapLegacyVideoDecoderFactory(
>-    cricket::WebRtcVideoDecoderFactory* legacy_decoder_factory) {
>-  return nullptr;
>-}
>-
>-jobject GetJavaSurfaceTextureHelper(
>-    const rtc::scoped_refptr<SurfaceTextureHelper>& surface_texture_helper) {
>-  return nullptr;
>-}
>-
>-}  // namespace jni
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/ownedfactoryandthreads.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/ownedfactoryandthreads.cc
>index abbc0367085..9987f252275 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/ownedfactoryandthreads.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/ownedfactoryandthreads.cc
>@@ -10,9 +10,8 @@
> 
> #include "sdk/android/src/jni/pc/ownedfactoryandthreads.h"
> 
>-#include "rtc_base/logging.h"
>-#include "sdk/android/src/jni/classreferenceholder.h"
> #include "sdk/android/src/jni/jni_helpers.h"
>+#include "sdk/android/src/jni/pc/peerconnectionfactory.h"
> 
> namespace webrtc {
> namespace jni {
>@@ -21,45 +20,33 @@ PeerConnectionFactoryInterface* factoryFromJava(jlong j_p) {
>   return reinterpret_cast<OwnedFactoryAndThreads*>(j_p)->factory();
> }
> 
>+OwnedFactoryAndThreads::OwnedFactoryAndThreads(
>+    std::unique_ptr<Thread> network_thread,
>+    std::unique_ptr<Thread> worker_thread,
>+    std::unique_ptr<Thread> signaling_thread,
>+    rtc::NetworkMonitorFactory* network_monitor_factory,
>+    PeerConnectionFactoryInterface* factory)
>+    : network_thread_(std::move(network_thread)),
>+      worker_thread_(std::move(worker_thread)),
>+      signaling_thread_(std::move(signaling_thread)),
>+      network_monitor_factory_(network_monitor_factory),
>+      factory_(factory) {}
>+
> OwnedFactoryAndThreads::~OwnedFactoryAndThreads() {
>-  CHECK_RELEASE(factory_);
>+  factory_->Release();
>   if (network_monitor_factory_ != nullptr) {
>     rtc::NetworkMonitorFactory::ReleaseFactory(network_monitor_factory_);
>   }
> }
> 
>-void OwnedFactoryAndThreads::JavaCallbackOnFactoryThreads() {
>-  JNIEnv* jni = AttachCurrentThreadIfNeeded();
>-  ScopedLocalRefFrame local_ref_frame(jni);
>-  jclass j_factory_class = FindClass(jni, "org/webrtc/PeerConnectionFactory");
>-  jmethodID m = nullptr;
>-  if (network_thread_->IsCurrent()) {
>-    RTC_LOG(LS_INFO) << "Network thread JavaCallback";
>-    m = GetStaticMethodID(jni, j_factory_class, "onNetworkThreadReady", "()V");
>-  }
>-  if (worker_thread_->IsCurrent()) {
>-    RTC_LOG(LS_INFO) << "Worker thread JavaCallback";
>-    m = GetStaticMethodID(jni, j_factory_class, "onWorkerThreadReady", "()V");
>-  }
>-  if (signaling_thread_->IsCurrent()) {
>-    RTC_LOG(LS_INFO) << "Signaling thread JavaCallback";
>-    m = GetStaticMethodID(jni, j_factory_class, "onSignalingThreadReady",
>-                          "()V");
>-  }
>-  if (m != nullptr) {
>-    jni->CallStaticVoidMethod(j_factory_class, m);
>-    CHECK_EXCEPTION(jni) << "error during JavaCallback::CallStaticVoidMethod";
>-  }
>-}
>-
> void OwnedFactoryAndThreads::InvokeJavaCallbacksOnFactoryThreads() {
>   RTC_LOG(LS_INFO) << "InvokeJavaCallbacksOnFactoryThreads.";
>   network_thread_->Invoke<void>(RTC_FROM_HERE,
>-                                [this] { JavaCallbackOnFactoryThreads(); });
>+                                &PeerConnectionFactoryNetworkThreadReady);
>   worker_thread_->Invoke<void>(RTC_FROM_HERE,
>-                               [this] { JavaCallbackOnFactoryThreads(); });
>+                               &PeerConnectionFactoryWorkerThreadReady);
>   signaling_thread_->Invoke<void>(RTC_FROM_HERE,
>-                                  [this] { JavaCallbackOnFactoryThreads(); });
>+                                  &PeerConnectionFactorySignalingThreadReady);
> }
> 
> }  // namespace jni
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/ownedfactoryandthreads.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/ownedfactoryandthreads.h
>index e4d1cf6b18a..2ea8570282d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/ownedfactoryandthreads.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/ownedfactoryandthreads.h
>@@ -18,8 +18,6 @@
> #include "api/peerconnectioninterface.h"
> #include "rtc_base/thread.h"
> 
>-using cricket::WebRtcVideoDecoderFactory;
>-using cricket::WebRtcVideoEncoderFactory;
> using rtc::Thread;
> 
> namespace webrtc {
>@@ -38,29 +36,15 @@ class OwnedFactoryAndThreads {
>   OwnedFactoryAndThreads(std::unique_ptr<Thread> network_thread,
>                          std::unique_ptr<Thread> worker_thread,
>                          std::unique_ptr<Thread> signaling_thread,
>-                         WebRtcVideoEncoderFactory* legacy_encoder_factory,
>-                         WebRtcVideoDecoderFactory* legacy_decoder_factory,
>                          rtc::NetworkMonitorFactory* network_monitor_factory,
>-                         PeerConnectionFactoryInterface* factory)
>-      : network_thread_(std::move(network_thread)),
>-        worker_thread_(std::move(worker_thread)),
>-        signaling_thread_(std::move(signaling_thread)),
>-        legacy_encoder_factory_(legacy_encoder_factory),
>-        legacy_decoder_factory_(legacy_decoder_factory),
>-        network_monitor_factory_(network_monitor_factory),
>-        factory_(factory) {}
>+                         PeerConnectionFactoryInterface* factory);
> 
>   ~OwnedFactoryAndThreads();
> 
>   PeerConnectionFactoryInterface* factory() { return factory_; }
>+  Thread* network_thread() { return network_thread_.get(); }
>   Thread* signaling_thread() { return signaling_thread_.get(); }
>   Thread* worker_thread() { return worker_thread_.get(); }
>-  WebRtcVideoEncoderFactory* legacy_encoder_factory() {
>-    return legacy_encoder_factory_;
>-  }
>-  WebRtcVideoDecoderFactory* legacy_decoder_factory() {
>-    return legacy_decoder_factory_;
>-  }
>   rtc::NetworkMonitorFactory* network_monitor_factory() {
>     return network_monitor_factory_;
>   }
>@@ -68,13 +52,9 @@ class OwnedFactoryAndThreads {
>   void InvokeJavaCallbacksOnFactoryThreads();
> 
>  private:
>-  void JavaCallbackOnFactoryThreads();
>-
>   const std::unique_ptr<Thread> network_thread_;
>   const std::unique_ptr<Thread> worker_thread_;
>   const std::unique_ptr<Thread> signaling_thread_;
>-  WebRtcVideoEncoderFactory* legacy_encoder_factory_;
>-  WebRtcVideoDecoderFactory* legacy_decoder_factory_;
>   rtc::NetworkMonitorFactory* network_monitor_factory_;
>   PeerConnectionFactoryInterface* factory_;  // Const after ctor except dtor.
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/peerconnection.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/peerconnection.cc
>new file mode 100644
>index 00000000000..31a0689eedc
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/peerconnection.cc
>@@ -0,0 +1,734 @@
>+/*
>+ *  Copyright 2013 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+// Lifecycle notes: objects are owned where they will be called; in other words
>+// FooObservers are owned by C++-land, and user-callable objects (e.g.
>+// PeerConnection and VideoTrack) are owned by Java-land.
>+// When this file (or other files in this directory) allocates C++
>+// RefCountInterfaces it AddRef()s an artificial ref simulating the jlong held
>+// in Java-land, and then Release()s the ref in the respective free call.
>+// Sometimes this AddRef is implicit in the construction of a scoped_refptr<>
>+// which is then .release()d. Any persistent (non-local) references from C++ to
>+// Java must be global or weak (in which case they must be checked before use)!
>+//
>+// Exception notes: pretty much all JNI calls can throw Java exceptions, so each
>+// call through a JNIEnv* pointer needs to be followed by an ExceptionCheck()
>+// call. In this file this is done in CHECK_EXCEPTION, making for much easier
>+// debugging in case of failure (the alternative is to wait for control to
>+// return to the Java frame that called code in this file, at which point it's
>+// impossible to tell which JNI call broke).
>+
>+#include "sdk/android/src/jni/pc/peerconnection.h"
>+
>+#include <limits>
>+#include <memory>
>+#include <string>
>+#include <utility>
>+
>+#include "absl/memory/memory.h"
>+#include "api/mediaconstraintsinterface.h"
>+#include "api/peerconnectioninterface.h"
>+#include "api/rtpreceiverinterface.h"
>+#include "api/rtpsenderinterface.h"
>+#include "api/rtptransceiverinterface.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/logging.h"
>+#include "sdk/android/generated_peerconnection_jni/jni/PeerConnection_jni.h"
>+#include "sdk/android/native_api/jni/java_types.h"
>+#include "sdk/android/src/jni/jni_helpers.h"
>+#include "sdk/android/src/jni/pc/datachannel.h"
>+#include "sdk/android/src/jni/pc/icecandidate.h"
>+#include "sdk/android/src/jni/pc/mediaconstraints.h"
>+#include "sdk/android/src/jni/pc/mediastreamtrack.h"
>+#include "sdk/android/src/jni/pc/rtcstatscollectorcallbackwrapper.h"
>+#include "sdk/android/src/jni/pc/rtpsender.h"
>+#include "sdk/android/src/jni/pc/sdpobserver.h"
>+#include "sdk/android/src/jni/pc/sessiondescription.h"
>+#include "sdk/android/src/jni/pc/statsobserver.h"
>+#include "sdk/android/src/jni/pc/turncustomizer.h"
>+
>+namespace webrtc {
>+namespace jni {
>+
>+namespace {
>+
>+PeerConnectionInterface* ExtractNativePC(JNIEnv* jni,
>+                                         const JavaRef<jobject>& j_pc) {
>+  return reinterpret_cast<OwnedPeerConnection*>(
>+             Java_PeerConnection_getNativeOwnedPeerConnection(jni, j_pc))
>+      ->pc();
>+}
>+
>+PeerConnectionInterface::IceServers JavaToNativeIceServers(
>+    JNIEnv* jni,
>+    const JavaRef<jobject>& j_ice_servers) {
>+  PeerConnectionInterface::IceServers ice_servers;
>+  for (const JavaRef<jobject>& j_ice_server : Iterable(jni, j_ice_servers)) {
>+    ScopedJavaLocalRef<jobject> j_ice_server_tls_cert_policy =
>+        Java_IceServer_getTlsCertPolicy(jni, j_ice_server);
>+    ScopedJavaLocalRef<jobject> urls =
>+        Java_IceServer_getUrls(jni, j_ice_server);
>+    ScopedJavaLocalRef<jstring> username =
>+        Java_IceServer_getUsername(jni, j_ice_server);
>+    ScopedJavaLocalRef<jstring> password =
>+        Java_IceServer_getPassword(jni, j_ice_server);
>+    PeerConnectionInterface::TlsCertPolicy tls_cert_policy =
>+        JavaToNativeTlsCertPolicy(jni, j_ice_server_tls_cert_policy);
>+    ScopedJavaLocalRef<jstring> hostname =
>+        Java_IceServer_getHostname(jni, j_ice_server);
>+    ScopedJavaLocalRef<jobject> tls_alpn_protocols =
>+        Java_IceServer_getTlsAlpnProtocols(jni, j_ice_server);
>+    ScopedJavaLocalRef<jobject> tls_elliptic_curves =
>+        Java_IceServer_getTlsEllipticCurves(jni, j_ice_server);
>+    PeerConnectionInterface::IceServer server;
>+    server.urls = JavaListToNativeVector<std::string, jstring>(
>+        jni, urls, &JavaToNativeString);
>+    server.username = JavaToNativeString(jni, username);
>+    server.password = JavaToNativeString(jni, password);
>+    server.tls_cert_policy = tls_cert_policy;
>+    server.hostname = JavaToNativeString(jni, hostname);
>+    server.tls_alpn_protocols = JavaListToNativeVector<std::string, jstring>(
>+        jni, tls_alpn_protocols, &JavaToNativeString);
>+    server.tls_elliptic_curves = JavaListToNativeVector<std::string, jstring>(
>+        jni, tls_elliptic_curves, &JavaToNativeString);
>+    ice_servers.push_back(server);
>+  }
>+  return ice_servers;
>+}
>+
>+SdpSemantics JavaToNativeSdpSemantics(JNIEnv* jni,
>+                                      const JavaRef<jobject>& j_sdp_semantics) {
>+  std::string enum_name = GetJavaEnumName(jni, j_sdp_semantics);
>+
>+  if (enum_name == "PLAN_B")
>+    return SdpSemantics::kPlanB;
>+
>+  if (enum_name == "UNIFIED_PLAN")
>+    return SdpSemantics::kUnifiedPlan;
>+
>+  RTC_NOTREACHED();
>+  return SdpSemantics::kPlanB;
>+}
>+
>+}  // namespace
>+
>+void JavaToNativeRTCConfiguration(
>+    JNIEnv* jni,
>+    const JavaRef<jobject>& j_rtc_config,
>+    PeerConnectionInterface::RTCConfiguration* rtc_config) {
>+  ScopedJavaLocalRef<jobject> j_ice_transports_type =
>+      Java_RTCConfiguration_getIceTransportsType(jni, j_rtc_config);
>+  ScopedJavaLocalRef<jobject> j_bundle_policy =
>+      Java_RTCConfiguration_getBundlePolicy(jni, j_rtc_config);
>+  ScopedJavaLocalRef<jobject> j_rtcp_mux_policy =
>+      Java_RTCConfiguration_getRtcpMuxPolicy(jni, j_rtc_config);
>+  ScopedJavaLocalRef<jobject> j_tcp_candidate_policy =
>+      Java_RTCConfiguration_getTcpCandidatePolicy(jni, j_rtc_config);
>+  ScopedJavaLocalRef<jobject> j_candidate_network_policy =
>+      Java_RTCConfiguration_getCandidateNetworkPolicy(jni, j_rtc_config);
>+  ScopedJavaLocalRef<jobject> j_ice_servers =
>+      Java_RTCConfiguration_getIceServers(jni, j_rtc_config);
>+  ScopedJavaLocalRef<jobject> j_continual_gathering_policy =
>+      Java_RTCConfiguration_getContinualGatheringPolicy(jni, j_rtc_config);
>+  ScopedJavaLocalRef<jobject> j_turn_customizer =
>+      Java_RTCConfiguration_getTurnCustomizer(jni, j_rtc_config);
>+  ScopedJavaLocalRef<jobject> j_network_preference =
>+      Java_RTCConfiguration_getNetworkPreference(jni, j_rtc_config);
>+  ScopedJavaLocalRef<jobject> j_sdp_semantics =
>+      Java_RTCConfiguration_getSdpSemantics(jni, j_rtc_config);
>+
>+  rtc_config->type = JavaToNativeIceTransportsType(jni, j_ice_transports_type);
>+  rtc_config->bundle_policy = JavaToNativeBundlePolicy(jni, j_bundle_policy);
>+  rtc_config->rtcp_mux_policy =
>+      JavaToNativeRtcpMuxPolicy(jni, j_rtcp_mux_policy);
>+  rtc_config->tcp_candidate_policy =
>+      JavaToNativeTcpCandidatePolicy(jni, j_tcp_candidate_policy);
>+  rtc_config->candidate_network_policy =
>+      JavaToNativeCandidateNetworkPolicy(jni, j_candidate_network_policy);
>+  rtc_config->servers = JavaToNativeIceServers(jni, j_ice_servers);
>+  rtc_config->audio_jitter_buffer_max_packets =
>+      Java_RTCConfiguration_getAudioJitterBufferMaxPackets(jni, j_rtc_config);
>+  rtc_config->audio_jitter_buffer_fast_accelerate =
>+      Java_RTCConfiguration_getAudioJitterBufferFastAccelerate(jni,
>+                                                               j_rtc_config);
>+  rtc_config->ice_connection_receiving_timeout =
>+      Java_RTCConfiguration_getIceConnectionReceivingTimeout(jni, j_rtc_config);
>+  rtc_config->ice_backup_candidate_pair_ping_interval =
>+      Java_RTCConfiguration_getIceBackupCandidatePairPingInterval(jni,
>+                                                                  j_rtc_config);
>+  rtc_config->continual_gathering_policy =
>+      JavaToNativeContinualGatheringPolicy(jni, j_continual_gathering_policy);
>+  rtc_config->ice_candidate_pool_size =
>+      Java_RTCConfiguration_getIceCandidatePoolSize(jni, j_rtc_config);
>+  rtc_config->prune_turn_ports =
>+      Java_RTCConfiguration_getPruneTurnPorts(jni, j_rtc_config);
>+  rtc_config->presume_writable_when_fully_relayed =
>+      Java_RTCConfiguration_getPresumeWritableWhenFullyRelayed(jni,
>+                                                               j_rtc_config);
>+  ScopedJavaLocalRef<jobject> j_ice_check_interval_strong_connectivity =
>+      Java_RTCConfiguration_getIceCheckIntervalStrongConnectivity(jni,
>+                                                                  j_rtc_config);
>+  rtc_config->ice_check_interval_strong_connectivity =
>+      JavaToNativeOptionalInt(jni, j_ice_check_interval_strong_connectivity);
>+  ScopedJavaLocalRef<jobject> j_ice_check_interval_weak_connectivity =
>+      Java_RTCConfiguration_getIceCheckIntervalWeakConnectivity(jni,
>+                                                                j_rtc_config);
>+  rtc_config->ice_check_interval_weak_connectivity =
>+      JavaToNativeOptionalInt(jni, j_ice_check_interval_weak_connectivity);
>+  ScopedJavaLocalRef<jobject> j_ice_check_min_interval =
>+      Java_RTCConfiguration_getIceCheckMinInterval(jni, j_rtc_config);
>+  rtc_config->ice_check_min_interval =
>+      JavaToNativeOptionalInt(jni, j_ice_check_min_interval);
>+  ScopedJavaLocalRef<jobject> j_ice_unwritable_timeout =
>+      Java_RTCConfiguration_getIceUnwritableTimeout(jni, j_rtc_config);
>+  rtc_config->ice_unwritable_timeout =
>+      JavaToNativeOptionalInt(jni, j_ice_unwritable_timeout);
>+  ScopedJavaLocalRef<jobject> j_ice_unwritable_min_checks =
>+      Java_RTCConfiguration_getIceUnwritableMinChecks(jni, j_rtc_config);
>+  rtc_config->ice_unwritable_min_checks =
>+      JavaToNativeOptionalInt(jni, j_ice_unwritable_min_checks);
>+  ScopedJavaLocalRef<jobject> j_stun_candidate_keepalive_interval =
>+      Java_RTCConfiguration_getStunCandidateKeepaliveInterval(jni,
>+                                                              j_rtc_config);
>+  rtc_config->stun_candidate_keepalive_interval =
>+      JavaToNativeOptionalInt(jni, j_stun_candidate_keepalive_interval);
>+  rtc_config->disable_ipv6_on_wifi =
>+      Java_RTCConfiguration_getDisableIPv6OnWifi(jni, j_rtc_config);
>+  rtc_config->max_ipv6_networks =
>+      Java_RTCConfiguration_getMaxIPv6Networks(jni, j_rtc_config);
>+  ScopedJavaLocalRef<jobject> j_ice_regather_interval_range =
>+      Java_RTCConfiguration_getIceRegatherIntervalRange(jni, j_rtc_config);
>+  if (!IsNull(jni, j_ice_regather_interval_range)) {
>+    int min = Java_IntervalRange_getMin(jni, j_ice_regather_interval_range);
>+    int max = Java_IntervalRange_getMax(jni, j_ice_regather_interval_range);
>+    rtc_config->ice_regather_interval_range.emplace(min, max);
>+  }
>+
>+  rtc_config->turn_customizer = GetNativeTurnCustomizer(jni, j_turn_customizer);
>+
>+  rtc_config->disable_ipv6 =
>+      Java_RTCConfiguration_getDisableIpv6(jni, j_rtc_config);
>+  rtc_config->media_config.enable_dscp =
>+      Java_RTCConfiguration_getEnableDscp(jni, j_rtc_config);
>+  rtc_config->media_config.video.enable_cpu_adaptation =
>+      Java_RTCConfiguration_getEnableCpuOveruseDetection(jni, j_rtc_config);
>+  rtc_config->enable_rtp_data_channel =
>+      Java_RTCConfiguration_getEnableRtpDataChannel(jni, j_rtc_config);
>+  rtc_config->media_config.video.suspend_below_min_bitrate =
>+      Java_RTCConfiguration_getSuspendBelowMinBitrate(jni, j_rtc_config);
>+  rtc_config->screencast_min_bitrate = JavaToNativeOptionalInt(
>+      jni, Java_RTCConfiguration_getScreencastMinBitrate(jni, j_rtc_config));
>+  rtc_config->combined_audio_video_bwe = JavaToNativeOptionalBool(
>+      jni, Java_RTCConfiguration_getCombinedAudioVideoBwe(jni, j_rtc_config));
>+  rtc_config->enable_dtls_srtp = JavaToNativeOptionalBool(
>+      jni, Java_RTCConfiguration_getEnableDtlsSrtp(jni, j_rtc_config));
>+  rtc_config->network_preference =
>+      JavaToNativeNetworkPreference(jni, j_network_preference);
>+  rtc_config->sdp_semantics = JavaToNativeSdpSemantics(jni, j_sdp_semantics);
>+  rtc_config->active_reset_srtp_params =
>+      Java_RTCConfiguration_getActiveResetSrtpParams(jni, j_rtc_config);
>+}
>+
>+rtc::KeyType GetRtcConfigKeyType(JNIEnv* env,
>+                                 const JavaRef<jobject>& j_rtc_config) {
>+  return JavaToNativeKeyType(
>+      env, Java_RTCConfiguration_getKeyType(env, j_rtc_config));
>+}
>+
>+PeerConnectionObserverJni::PeerConnectionObserverJni(
>+    JNIEnv* jni,
>+    const JavaRef<jobject>& j_observer)
>+    : j_observer_global_(jni, j_observer) {}
>+
>+PeerConnectionObserverJni::~PeerConnectionObserverJni() = default;
>+
>+void PeerConnectionObserverJni::OnIceCandidate(
>+    const IceCandidateInterface* candidate) {
>+  JNIEnv* env = AttachCurrentThreadIfNeeded();
>+  Java_Observer_onIceCandidate(env, j_observer_global_,
>+                               NativeToJavaIceCandidate(env, *candidate));
>+}
>+
>+void PeerConnectionObserverJni::OnIceCandidatesRemoved(
>+    const std::vector<cricket::Candidate>& candidates) {
>+  JNIEnv* env = AttachCurrentThreadIfNeeded();
>+  Java_Observer_onIceCandidatesRemoved(
>+      env, j_observer_global_, NativeToJavaCandidateArray(env, candidates));
>+}
>+
>+void PeerConnectionObserverJni::OnSignalingChange(
>+    PeerConnectionInterface::SignalingState new_state) {
>+  JNIEnv* env = AttachCurrentThreadIfNeeded();
>+  Java_Observer_onSignalingChange(
>+      env, j_observer_global_,
>+      Java_SignalingState_fromNativeIndex(env, new_state));
>+}
>+
>+void PeerConnectionObserverJni::OnIceConnectionChange(
>+    PeerConnectionInterface::IceConnectionState new_state) {
>+  JNIEnv* env = AttachCurrentThreadIfNeeded();
>+  Java_Observer_onIceConnectionChange(
>+      env, j_observer_global_,
>+      Java_IceConnectionState_fromNativeIndex(env, new_state));
>+}
>+
>+void PeerConnectionObserverJni::OnIceConnectionReceivingChange(bool receiving) {
>+  JNIEnv* env = AttachCurrentThreadIfNeeded();
>+  Java_Observer_onIceConnectionReceivingChange(env, j_observer_global_,
>+                                               receiving);
>+}
>+
>+void PeerConnectionObserverJni::OnIceGatheringChange(
>+    PeerConnectionInterface::IceGatheringState new_state) {
>+  JNIEnv* env = AttachCurrentThreadIfNeeded();
>+  Java_Observer_onIceGatheringChange(
>+      env, j_observer_global_,
>+      Java_IceGatheringState_fromNativeIndex(env, new_state));
>+}
>+
>+void PeerConnectionObserverJni::OnAddStream(
>+    rtc::scoped_refptr<MediaStreamInterface> stream) {
>+  JNIEnv* env = AttachCurrentThreadIfNeeded();
>+  Java_Observer_onAddStream(
>+      env, j_observer_global_,
>+      GetOrCreateJavaStream(env, stream).j_media_stream());
>+}
>+
>+void PeerConnectionObserverJni::OnRemoveStream(
>+    rtc::scoped_refptr<MediaStreamInterface> stream) {
>+  JNIEnv* env = AttachCurrentThreadIfNeeded();
>+  NativeToJavaStreamsMap::iterator it = remote_streams_.find(stream);
>+  RTC_CHECK(it != remote_streams_.end()) << "unexpected stream: " << stream;
>+  Java_Observer_onRemoveStream(env, j_observer_global_,
>+                               it->second.j_media_stream());
>+  remote_streams_.erase(it);
>+}
>+
>+void PeerConnectionObserverJni::OnDataChannel(
>+    rtc::scoped_refptr<DataChannelInterface> channel) {
>+  JNIEnv* env = AttachCurrentThreadIfNeeded();
>+  Java_Observer_onDataChannel(env, j_observer_global_,
>+                              WrapNativeDataChannel(env, channel));
>+}
>+
>+void PeerConnectionObserverJni::OnRenegotiationNeeded() {
>+  JNIEnv* env = AttachCurrentThreadIfNeeded();
>+  Java_Observer_onRenegotiationNeeded(env, j_observer_global_);
>+}
>+
>+void PeerConnectionObserverJni::OnAddTrack(
>+    rtc::scoped_refptr<RtpReceiverInterface> receiver,
>+    const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams) {
>+  JNIEnv* env = AttachCurrentThreadIfNeeded();
>+  ScopedJavaLocalRef<jobject> j_rtp_receiver =
>+      NativeToJavaRtpReceiver(env, receiver);
>+  rtp_receivers_.emplace_back(env, j_rtp_receiver);
>+
>+  Java_Observer_onAddTrack(env, j_observer_global_, j_rtp_receiver,
>+                           NativeToJavaMediaStreamArray(env, streams));
>+}
>+
>+void PeerConnectionObserverJni::OnTrack(
>+    rtc::scoped_refptr<RtpTransceiverInterface> transceiver) {
>+  JNIEnv* env = AttachCurrentThreadIfNeeded();
>+  ScopedJavaLocalRef<jobject> j_rtp_transceiver =
>+      NativeToJavaRtpTransceiver(env, transceiver);
>+  rtp_transceivers_.emplace_back(env, j_rtp_transceiver);
>+
>+  Java_Observer_onTrack(env, j_observer_global_, j_rtp_transceiver);
>+}
>+
>+// If the NativeToJavaStreamsMap contains the stream, return it.
>+// Otherwise, create a new Java MediaStream.
>+JavaMediaStream& PeerConnectionObserverJni::GetOrCreateJavaStream(
>+    JNIEnv* env,
>+    const rtc::scoped_refptr<MediaStreamInterface>& stream) {
>+  NativeToJavaStreamsMap::iterator it = remote_streams_.find(stream);
>+  if (it == remote_streams_.end()) {
>+    it = remote_streams_
>+             .emplace(std::piecewise_construct,
>+                      std::forward_as_tuple(stream.get()),
>+                      std::forward_as_tuple(env, stream))
>+             .first;
>+  }
>+  return it->second;
>+}
>+
>+ScopedJavaLocalRef<jobjectArray>
>+PeerConnectionObserverJni::NativeToJavaMediaStreamArray(
>+    JNIEnv* jni,
>+    const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams) {
>+  return NativeToJavaObjectArray(
>+      jni, streams, GetMediaStreamClass(jni),
>+      [this](JNIEnv* env, rtc::scoped_refptr<MediaStreamInterface> stream)
>+          -> const ScopedJavaGlobalRef<jobject>& {
>+        return GetOrCreateJavaStream(env, stream).j_media_stream();
>+      });
>+}
>+
>+OwnedPeerConnection::OwnedPeerConnection(
>+    rtc::scoped_refptr<PeerConnectionInterface> peer_connection,
>+    std::unique_ptr<PeerConnectionObserver> observer)
>+    : OwnedPeerConnection(peer_connection,
>+                          std::move(observer),
>+                          nullptr /* constraints */) {}
>+
>+OwnedPeerConnection::OwnedPeerConnection(
>+    rtc::scoped_refptr<PeerConnectionInterface> peer_connection,
>+    std::unique_ptr<PeerConnectionObserver> observer,
>+    std::unique_ptr<MediaConstraintsInterface> constraints)
>+    : peer_connection_(peer_connection),
>+      observer_(std::move(observer)),
>+      constraints_(std::move(constraints)) {}
>+
>+OwnedPeerConnection::~OwnedPeerConnection() {
>+  // Ensure that PeerConnection is destroyed before the observer.
>+  peer_connection_ = nullptr;
>+}
>+
>+static jlong JNI_PeerConnection_CreatePeerConnectionObserver(
>+    JNIEnv* jni,
>+    const JavaParamRef<jclass>&,
>+    const JavaParamRef<jobject>& j_observer) {
>+  return jlongFromPointer(new PeerConnectionObserverJni(jni, j_observer));
>+}
>+
>+static void JNI_PeerConnection_FreeOwnedPeerConnection(
>+    JNIEnv*,
>+    const JavaParamRef<jclass>&,
>+    jlong j_p) {
>+  delete reinterpret_cast<OwnedPeerConnection*>(j_p);
>+}
>+
>+static jlong JNI_PeerConnection_GetNativePeerConnection(
>+    JNIEnv* jni,
>+    const JavaParamRef<jobject>& j_pc) {
>+  return jlongFromPointer(ExtractNativePC(jni, j_pc));
>+}
>+
>+static ScopedJavaLocalRef<jobject> JNI_PeerConnection_GetLocalDescription(
>+    JNIEnv* jni,
>+    const JavaParamRef<jobject>& j_pc) {
>+  const SessionDescriptionInterface* sdp =
>+      ExtractNativePC(jni, j_pc)->local_description();
>+  return sdp ? NativeToJavaSessionDescription(jni, sdp) : nullptr;
>+}
>+
>+static ScopedJavaLocalRef<jobject> JNI_PeerConnection_GetRemoteDescription(
>+    JNIEnv* jni,
>+    const JavaParamRef<jobject>& j_pc) {
>+  const SessionDescriptionInterface* sdp =
>+      ExtractNativePC(jni, j_pc)->remote_description();
>+  return sdp ? NativeToJavaSessionDescription(jni, sdp) : nullptr;
>+}
>+
>+static ScopedJavaLocalRef<jobject> JNI_PeerConnection_CreateDataChannel(
>+    JNIEnv* jni,
>+    const JavaParamRef<jobject>& j_pc,
>+    const JavaParamRef<jstring>& j_label,
>+    const JavaParamRef<jobject>& j_init) {
>+  DataChannelInit init = JavaToNativeDataChannelInit(jni, j_init);
>+  rtc::scoped_refptr<DataChannelInterface> channel(
>+      ExtractNativePC(jni, j_pc)->CreateDataChannel(
>+          JavaToNativeString(jni, j_label), &init));
>+  return WrapNativeDataChannel(jni, channel);
>+}
>+
>+static void JNI_PeerConnection_CreateOffer(
>+    JNIEnv* jni,
>+    const JavaParamRef<jobject>& j_pc,
>+    const JavaParamRef<jobject>& j_observer,
>+    const JavaParamRef<jobject>& j_constraints) {
>+  std::unique_ptr<MediaConstraintsInterface> constraints =
>+      JavaToNativeMediaConstraints(jni, j_constraints);
>+  rtc::scoped_refptr<CreateSdpObserverJni> observer(
>+      new rtc::RefCountedObject<CreateSdpObserverJni>(jni, j_observer,
>+                                                      std::move(constraints)));
>+  ExtractNativePC(jni, j_pc)->CreateOffer(observer, observer->constraints());
>+}
>+
>+static void JNI_PeerConnection_CreateAnswer(
>+    JNIEnv* jni,
>+    const JavaParamRef<jobject>& j_pc,
>+    const JavaParamRef<jobject>& j_observer,
>+    const JavaParamRef<jobject>& j_constraints) {
>+  std::unique_ptr<MediaConstraintsInterface> constraints =
>+      JavaToNativeMediaConstraints(jni, j_constraints);
>+  rtc::scoped_refptr<CreateSdpObserverJni> observer(
>+      new rtc::RefCountedObject<CreateSdpObserverJni>(jni, j_observer,
>+                                                      std::move(constraints)));
>+  ExtractNativePC(jni, j_pc)->CreateAnswer(observer, observer->constraints());
>+}
>+
>+static void JNI_PeerConnection_SetLocalDescription(
>+    JNIEnv* jni,
>+    const JavaParamRef<jobject>& j_pc,
>+    const JavaParamRef<jobject>& j_observer,
>+    const JavaParamRef<jobject>& j_sdp) {
>+  rtc::scoped_refptr<SetSdpObserverJni> observer(
>+      new rtc::RefCountedObject<SetSdpObserverJni>(jni, j_observer, nullptr));
>+  ExtractNativePC(jni, j_pc)->SetLocalDescription(
>+      observer, JavaToNativeSessionDescription(jni, j_sdp).release());
>+}
>+
>+static void JNI_PeerConnection_SetRemoteDescription(
>+    JNIEnv* jni,
>+    const JavaParamRef<jobject>& j_pc,
>+    const JavaParamRef<jobject>& j_observer,
>+    const JavaParamRef<jobject>& j_sdp) {
>+  rtc::scoped_refptr<SetSdpObserverJni> observer(
>+      new rtc::RefCountedObject<SetSdpObserverJni>(jni, j_observer, nullptr));
>+  ExtractNativePC(jni, j_pc)->SetRemoteDescription(
>+      observer, JavaToNativeSessionDescription(jni, j_sdp).release());
>+}
>+
>+static void JNI_PeerConnection_SetAudioPlayout(
>+    JNIEnv* jni,
>+    const JavaParamRef<jobject>& j_pc,
>+    jboolean playout) {
>+  ExtractNativePC(jni, j_pc)->SetAudioPlayout(playout);
>+}
>+
>+static void JNI_PeerConnection_SetAudioRecording(
>+    JNIEnv* jni,
>+    const JavaParamRef<jobject>& j_pc,
>+    jboolean recording) {
>+  ExtractNativePC(jni, j_pc)->SetAudioRecording(recording);
>+}
>+
>+static jboolean JNI_PeerConnection_SetConfiguration(
>+    JNIEnv* jni,
>+    const JavaParamRef<jobject>& j_pc,
>+    const JavaParamRef<jobject>& j_rtc_config) {
>+  // Need to merge constraints into RTCConfiguration again, which are stored
>+  // in the OwnedPeerConnection object.
>+  OwnedPeerConnection* owned_pc = reinterpret_cast<OwnedPeerConnection*>(
>+      Java_PeerConnection_getNativeOwnedPeerConnection(jni, j_pc));
>+  PeerConnectionInterface::RTCConfiguration rtc_config(
>+      PeerConnectionInterface::RTCConfigurationType::kAggressive);
>+  JavaToNativeRTCConfiguration(jni, j_rtc_config, &rtc_config);
>+  if (owned_pc->constraints()) {
>+    CopyConstraintsIntoRtcConfiguration(owned_pc->constraints(), &rtc_config);
>+  }
>+  return owned_pc->pc()->SetConfiguration(rtc_config);
>+}
>+
>+static jboolean JNI_PeerConnection_AddIceCandidate(
>+    JNIEnv* jni,
>+    const JavaParamRef<jobject>& j_pc,
>+    const JavaParamRef<jstring>& j_sdp_mid,
>+    jint j_sdp_mline_index,
>+    const JavaParamRef<jstring>& j_candidate_sdp) {
>+  std::string sdp_mid = JavaToNativeString(jni, j_sdp_mid);
>+  std::string sdp = JavaToNativeString(jni, j_candidate_sdp);
>+  std::unique_ptr<IceCandidateInterface> candidate(
>+      CreateIceCandidate(sdp_mid, j_sdp_mline_index, sdp, nullptr));
>+  return ExtractNativePC(jni, j_pc)->AddIceCandidate(candidate.get());
>+}
>+
>+static jboolean JNI_PeerConnection_RemoveIceCandidates(
>+    JNIEnv* jni,
>+    const JavaParamRef<jobject>& j_pc,
>+    const JavaParamRef<jobjectArray>& j_candidates) {
>+  std::vector<cricket::Candidate> candidates =
>+      JavaToNativeVector<cricket::Candidate>(jni, j_candidates,
>+                                             &JavaToNativeCandidate);
>+  return ExtractNativePC(jni, j_pc)->RemoveIceCandidates(candidates);
>+}
>+
>+static jboolean JNI_PeerConnection_AddLocalStream(
>+    JNIEnv* jni,
>+    const JavaParamRef<jobject>& j_pc,
>+    jlong native_stream) {
>+  return ExtractNativePC(jni, j_pc)->AddStream(
>+      reinterpret_cast<MediaStreamInterface*>(native_stream));
>+}
>+
>+static void JNI_PeerConnection_RemoveLocalStream(
>+    JNIEnv* jni,
>+    const JavaParamRef<jobject>& j_pc,
>+    jlong native_stream) {
>+  ExtractNativePC(jni, j_pc)->RemoveStream(
>+      reinterpret_cast<MediaStreamInterface*>(native_stream));
>+}
>+
>+static ScopedJavaLocalRef<jobject> JNI_PeerConnection_CreateSender(
>+    JNIEnv* jni,
>+    const JavaParamRef<jobject>& j_pc,
>+    const JavaParamRef<jstring>& j_kind,
>+    const JavaParamRef<jstring>& j_stream_id) {
>+  std::string kind = JavaToNativeString(jni, j_kind);
>+  std::string stream_id = JavaToNativeString(jni, j_stream_id);
>+  rtc::scoped_refptr<RtpSenderInterface> sender =
>+      ExtractNativePC(jni, j_pc)->CreateSender(kind, stream_id);
>+  return NativeToJavaRtpSender(jni, sender);
>+}
>+
>+static ScopedJavaLocalRef<jobject> JNI_PeerConnection_GetSenders(
>+    JNIEnv* jni,
>+    const JavaParamRef<jobject>& j_pc) {
>+  return NativeToJavaList(jni, ExtractNativePC(jni, j_pc)->GetSenders(),
>+                          &NativeToJavaRtpSender);
>+}
>+
>+static ScopedJavaLocalRef<jobject> JNI_PeerConnection_GetReceivers(
>+    JNIEnv* jni,
>+    const JavaParamRef<jobject>& j_pc) {
>+  return NativeToJavaList(jni, ExtractNativePC(jni, j_pc)->GetReceivers(),
>+                          &NativeToJavaRtpReceiver);
>+}
>+
>+static ScopedJavaLocalRef<jobject> JNI_PeerConnection_GetTransceivers(
>+    JNIEnv* jni,
>+    const JavaParamRef<jobject>& j_pc) {
>+  return NativeToJavaList(jni, ExtractNativePC(jni, j_pc)->GetTransceivers(),
>+                          &NativeToJavaRtpTransceiver);
>+}
>+
>+static ScopedJavaLocalRef<jobject> JNI_PeerConnection_AddTrack(
>+    JNIEnv* jni,
>+    const JavaParamRef<jobject>& j_pc,
>+    const jlong native_track,
>+    const JavaParamRef<jobject>& j_stream_labels) {
>+  RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>> result =
>+      ExtractNativePC(jni, j_pc)->AddTrack(
>+          reinterpret_cast<MediaStreamTrackInterface*>(native_track),
>+          JavaListToNativeVector<std::string, jstring>(jni, j_stream_labels,
>+                                                       &JavaToNativeString));
>+  if (!result.ok()) {
>+    RTC_LOG(LS_ERROR) << "Failed to add track: " << result.error().message();
>+    return nullptr;
>+  } else {
>+    return NativeToJavaRtpSender(jni, result.MoveValue());
>+  }
>+}
>+
>+static jboolean JNI_PeerConnection_RemoveTrack(
>+    JNIEnv* jni,
>+    const JavaParamRef<jobject>& j_pc,
>+    jlong native_sender) {
>+  return ExtractNativePC(jni, j_pc)->RemoveTrack(
>+      reinterpret_cast<RtpSenderInterface*>(native_sender));
>+}
>+
>+static ScopedJavaLocalRef<jobject> JNI_PeerConnection_AddTransceiverWithTrack(
>+    JNIEnv* jni,
>+    const JavaParamRef<jobject>& j_pc,
>+    jlong native_track,
>+    const JavaParamRef<jobject>& j_init) {
>+  RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> result =
>+      ExtractNativePC(jni, j_pc)->AddTransceiver(
>+          reinterpret_cast<MediaStreamTrackInterface*>(native_track),
>+          JavaToNativeRtpTransceiverInit(jni, j_init));
>+  if (!result.ok()) {
>+    RTC_LOG(LS_ERROR) << "Failed to add transceiver: "
>+                      << result.error().message();
>+    return nullptr;
>+  } else {
>+    return NativeToJavaRtpTransceiver(jni, result.MoveValue());
>+  }
>+}
>+
>+static ScopedJavaLocalRef<jobject> JNI_PeerConnection_AddTransceiverOfType(
>+    JNIEnv* jni,
>+    const JavaParamRef<jobject>& j_pc,
>+    const JavaParamRef<jobject>& j_media_type,
>+    const JavaParamRef<jobject>& j_init) {
>+  RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> result =
>+      ExtractNativePC(jni, j_pc)->AddTransceiver(
>+          JavaToNativeMediaType(jni, j_media_type),
>+          JavaToNativeRtpTransceiverInit(jni, j_init));
>+  if (!result.ok()) {
>+    RTC_LOG(LS_ERROR) << "Failed to add transceiver: "
>+                      << result.error().message();
>+    return nullptr;
>+  } else {
>+    return NativeToJavaRtpTransceiver(jni, result.MoveValue());
>+  }
>+}
>+
>+static jboolean JNI_PeerConnection_OldGetStats(
>+    JNIEnv* jni,
>+    const JavaParamRef<jobject>& j_pc,
>+    const JavaParamRef<jobject>& j_observer,
>+    jlong native_track) {
>+  rtc::scoped_refptr<StatsObserverJni> observer(
>+      new rtc::RefCountedObject<StatsObserverJni>(jni, j_observer));
>+  return ExtractNativePC(jni, j_pc)->GetStats(
>+      observer, reinterpret_cast<MediaStreamTrackInterface*>(native_track),
>+      PeerConnectionInterface::kStatsOutputLevelStandard);
>+}
>+
>+static void JNI_PeerConnection_NewGetStats(
>+    JNIEnv* jni,
>+    const JavaParamRef<jobject>& j_pc,
>+    const JavaParamRef<jobject>& j_callback) {
>+  rtc::scoped_refptr<RTCStatsCollectorCallbackWrapper> callback(
>+      new rtc::RefCountedObject<RTCStatsCollectorCallbackWrapper>(jni,
>+                                                                  j_callback));
>+  ExtractNativePC(jni, j_pc)->GetStats(callback);
>+}
>+
>+static jboolean JNI_PeerConnection_SetBitrate(
>+    JNIEnv* jni,
>+    const JavaParamRef<jobject>& j_pc,
>+    const JavaParamRef<jobject>& j_min,
>+    const JavaParamRef<jobject>& j_current,
>+    const JavaParamRef<jobject>& j_max) {
>+  PeerConnectionInterface::BitrateParameters params;
>+  params.min_bitrate_bps = JavaToNativeOptionalInt(jni, j_min);
>+  params.current_bitrate_bps = JavaToNativeOptionalInt(jni, j_current);
>+  params.max_bitrate_bps = JavaToNativeOptionalInt(jni, j_max);
>+  return ExtractNativePC(jni, j_pc)->SetBitrate(params).ok();
>+}
>+
>+static jboolean JNI_PeerConnection_StartRtcEventLog(
>+    JNIEnv* jni,
>+    const JavaParamRef<jobject>& j_pc,
>+    int file_descriptor,
>+    int max_size_bytes) {
>+  return ExtractNativePC(jni, j_pc)->StartRtcEventLog(file_descriptor,
>+                                                      max_size_bytes);
>+}
>+
>+static void JNI_PeerConnection_StopRtcEventLog(
>+    JNIEnv* jni,
>+    const JavaParamRef<jobject>& j_pc) {
>+  ExtractNativePC(jni, j_pc)->StopRtcEventLog();
>+}
>+
>+static ScopedJavaLocalRef<jobject> JNI_PeerConnection_SignalingState(
>+    JNIEnv* env,
>+    const JavaParamRef<jobject>& j_pc) {
>+  return Java_SignalingState_fromNativeIndex(
>+      env, ExtractNativePC(env, j_pc)->signaling_state());
>+}
>+
>+static ScopedJavaLocalRef<jobject> JNI_PeerConnection_IceConnectionState(
>+    JNIEnv* env,
>+    const JavaParamRef<jobject>& j_pc) {
>+  return Java_IceConnectionState_fromNativeIndex(
>+      env, ExtractNativePC(env, j_pc)->ice_connection_state());
>+}
>+
>+static ScopedJavaLocalRef<jobject> JNI_PeerConnection_IceGatheringState(
>+    JNIEnv* env,
>+    const JavaParamRef<jobject>& j_pc) {
>+  return Java_IceGatheringState_fromNativeIndex(
>+      env, ExtractNativePC(env, j_pc)->ice_gathering_state());
>+}
>+
>+static void JNI_PeerConnection_Close(JNIEnv* jni,
>+                                     const JavaParamRef<jobject>& j_pc) {
>+  ExtractNativePC(jni, j_pc)->Close();
>+}
>+
>+}  // namespace jni
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/peerconnection.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/peerconnection.h
>new file mode 100644
>index 00000000000..f20166727ea
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/peerconnection.h
>@@ -0,0 +1,126 @@
>+/*
>+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef SDK_ANDROID_SRC_JNI_PC_PEERCONNECTION_H_
>+#define SDK_ANDROID_SRC_JNI_PC_PEERCONNECTION_H_
>+
>+#include <pc/mediastreamobserver.h>
>+#include <map>
>+#include <memory>
>+#include <vector>
>+
>+#include "api/peerconnectioninterface.h"
>+#include "sdk/android/src/jni/jni_helpers.h"
>+#include "sdk/android/src/jni/pc/mediaconstraints.h"
>+#include "sdk/android/src/jni/pc/mediastream.h"
>+#include "sdk/android/src/jni/pc/rtpreceiver.h"
>+#include "sdk/android/src/jni/pc/rtptransceiver.h"
>+
>+namespace webrtc {
>+namespace jni {
>+
>+void JavaToNativeRTCConfiguration(
>+    JNIEnv* jni,
>+    const JavaRef<jobject>& j_rtc_config,
>+    PeerConnectionInterface::RTCConfiguration* rtc_config);
>+
>+rtc::KeyType GetRtcConfigKeyType(JNIEnv* env,
>+                                 const JavaRef<jobject>& j_rtc_config);
>+
>+// Adapter between the C++ PeerConnectionObserver interface and the Java
>+// PeerConnection.Observer interface.  Wraps an instance of the Java interface
>+// and dispatches C++ callbacks to Java.
>+class PeerConnectionObserverJni : public PeerConnectionObserver {
>+ public:
>+  PeerConnectionObserverJni(JNIEnv* jni, const JavaRef<jobject>& j_observer);
>+  ~PeerConnectionObserverJni() override;
>+
>+  // Implementation of PeerConnectionObserver interface, which propagates
>+  // the callbacks to the Java observer.
>+  void OnIceCandidate(const IceCandidateInterface* candidate) override;
>+  void OnIceCandidatesRemoved(
>+      const std::vector<cricket::Candidate>& candidates) override;
>+  void OnSignalingChange(
>+      PeerConnectionInterface::SignalingState new_state) override;
>+  void OnIceConnectionChange(
>+      PeerConnectionInterface::IceConnectionState new_state) override;
>+  void OnIceConnectionReceivingChange(bool receiving) override;
>+  void OnIceGatheringChange(
>+      PeerConnectionInterface::IceGatheringState new_state) override;
>+  void OnAddStream(rtc::scoped_refptr<MediaStreamInterface> stream) override;
>+  void OnRemoveStream(rtc::scoped_refptr<MediaStreamInterface> stream) override;
>+  void OnDataChannel(rtc::scoped_refptr<DataChannelInterface> channel) override;
>+  void OnRenegotiationNeeded() override;
>+  void OnAddTrack(rtc::scoped_refptr<RtpReceiverInterface> receiver,
>+                  const std::vector<rtc::scoped_refptr<MediaStreamInterface>>&
>+                      streams) override;
>+  void OnTrack(
>+      rtc::scoped_refptr<RtpTransceiverInterface> transceiver) override;
>+
>+ private:
>+  typedef std::map<MediaStreamInterface*, JavaMediaStream>
>+      NativeToJavaStreamsMap;
>+  typedef std::map<MediaStreamTrackInterface*, RtpReceiverInterface*>
>+      NativeMediaStreamTrackToNativeRtpReceiver;
>+
>+  // If the NativeToJavaStreamsMap contains the stream, return it.
>+  // Otherwise, create a new Java MediaStream. Returns a global jobject.
>+  JavaMediaStream& GetOrCreateJavaStream(
>+      JNIEnv* env,
>+      const rtc::scoped_refptr<MediaStreamInterface>& stream);
>+
>+  // Converts array of streams, creating or re-using Java streams as necessary.
>+  ScopedJavaLocalRef<jobjectArray> NativeToJavaMediaStreamArray(
>+      JNIEnv* jni,
>+      const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams);
>+
>+  const ScopedJavaGlobalRef<jobject> j_observer_global_;
>+
>+  // C++ -> Java remote streams.
>+  NativeToJavaStreamsMap remote_streams_;
>+  std::vector<JavaRtpReceiverGlobalOwner> rtp_receivers_;
>+  // Holds a reference to the Java transceivers given to the AddTrack
>+  // callback, so that the shared ownership by the Java object will be
>+  // properly disposed.
>+  std::vector<JavaRtpTransceiverGlobalOwner> rtp_transceivers_;
>+};
>+
>+// PeerConnection doesn't take ownership of the observer. In Java API, we don't
>+// want the client to have to manually dispose the observer. To solve this, this
>+// wrapper class is used for object ownership.
>+//
>+// Also stores reference to the deprecated PeerConnection constraints for now.
>+class OwnedPeerConnection {
>+ public:
>+  OwnedPeerConnection(
>+      rtc::scoped_refptr<PeerConnectionInterface> peer_connection,
>+      std::unique_ptr<PeerConnectionObserver> observer);
>+  // Deprecated. PC constraints are deprecated.
>+  OwnedPeerConnection(
>+      rtc::scoped_refptr<PeerConnectionInterface> peer_connection,
>+      std::unique_ptr<PeerConnectionObserver> observer,
>+      std::unique_ptr<MediaConstraintsInterface> constraints);
>+  ~OwnedPeerConnection();
>+
>+  PeerConnectionInterface* pc() const { return peer_connection_.get(); }
>+  const MediaConstraintsInterface* constraints() const {
>+    return constraints_.get();
>+  }
>+
>+ private:
>+  rtc::scoped_refptr<PeerConnectionInterface> peer_connection_;
>+  std::unique_ptr<PeerConnectionObserver> observer_;
>+  std::unique_ptr<MediaConstraintsInterface> constraints_;
>+};
>+
>+}  // namespace jni
>+}  // namespace webrtc
>+
>+#endif  // SDK_ANDROID_SRC_JNI_PC_PEERCONNECTION_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/peerconnection_jni.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/peerconnection_jni.cc
>deleted file mode 100644
>index c3f2d12d42f..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/peerconnection_jni.cc
>+++ /dev/null
>@@ -1,415 +0,0 @@
>-/*
>- *  Copyright 2013 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-// Lifecycle notes: objects are owned where they will be called; in other words
>-// FooObservers are owned by C++-land, and user-callable objects (e.g.
>-// PeerConnection and VideoTrack) are owned by Java-land.
>-// When this file (or other files in this directory) allocates C++
>-// RefCountInterfaces it AddRef()s an artificial ref simulating the jlong held
>-// in Java-land, and then Release()s the ref in the respective free call.
>-// Sometimes this AddRef is implicit in the construction of a scoped_refptr<>
>-// which is then .release()d. Any persistent (non-local) references from C++ to
>-// Java must be global or weak (in which case they must be checked before use)!
>-//
>-// Exception notes: pretty much all JNI calls can throw Java exceptions, so each
>-// call through a JNIEnv* pointer needs to be followed by an ExceptionCheck()
>-// call. In this file this is done in CHECK_EXCEPTION, making for much easier
>-// debugging in case of failure (the alternative is to wait for control to
>-// return to the Java frame that called code in this file, at which point it's
>-// impossible to tell which JNI call broke).
>-
>-#include <limits>
>-#include <memory>
>-#include <utility>
>-
>-#include "api/mediaconstraintsinterface.h"
>-#include "api/peerconnectioninterface.h"
>-#include "api/rtpreceiverinterface.h"
>-#include "api/rtpsenderinterface.h"
>-#include "rtc_base/checks.h"
>-#include "rtc_base/logging.h"
>-#include "sdk/android/src/jni/classreferenceholder.h"
>-#include "sdk/android/src/jni/jni_helpers.h"
>-#include "sdk/android/src/jni/pc/datachannel.h"
>-#include "sdk/android/src/jni/pc/java_native_conversion.h"
>-#include "sdk/android/src/jni/pc/mediaconstraints_jni.h"
>-#include "sdk/android/src/jni/pc/peerconnectionobserver_jni.h"
>-#include "sdk/android/src/jni/pc/rtcstatscollectorcallbackwrapper.h"
>-#include "sdk/android/src/jni/pc/sdpobserver_jni.h"
>-#include "sdk/android/src/jni/pc/statsobserver_jni.h"
>-
>-namespace webrtc {
>-namespace jni {
>-
>-static rtc::scoped_refptr<PeerConnectionInterface> ExtractNativePC(
>-    JNIEnv* jni,
>-    jobject j_pc) {
>-  jfieldID native_pc_id =
>-      GetFieldID(jni, GetObjectClass(jni, j_pc), "nativePeerConnection", "J");
>-  jlong j_p = GetLongField(jni, j_pc, native_pc_id);
>-  return rtc::scoped_refptr<PeerConnectionInterface>(
>-      reinterpret_cast<PeerConnectionInterface*>(j_p));
>-}
>-
>-JNI_FUNCTION_DECLARATION(void,
>-                         PeerConnection_freeObserver,
>-                         JNIEnv*,
>-                         jclass,
>-                         jlong j_p) {
>-  PeerConnectionObserverJni* p =
>-      reinterpret_cast<PeerConnectionObserverJni*>(j_p);
>-  delete p;
>-}
>-
>-JNI_FUNCTION_DECLARATION(jobject,
>-                         PeerConnection_getLocalDescription,
>-                         JNIEnv* jni,
>-                         jobject j_pc) {
>-  const SessionDescriptionInterface* sdp =
>-      ExtractNativePC(jni, j_pc)->local_description();
>-  return sdp ? NativeToJavaSessionDescription(jni, sdp) : NULL;
>-}
>-
>-JNI_FUNCTION_DECLARATION(jobject,
>-                         PeerConnection_getRemoteDescription,
>-                         JNIEnv* jni,
>-                         jobject j_pc) {
>-  const SessionDescriptionInterface* sdp =
>-      ExtractNativePC(jni, j_pc)->remote_description();
>-  return sdp ? NativeToJavaSessionDescription(jni, sdp) : NULL;
>-}
>-
>-JNI_FUNCTION_DECLARATION(jobject,
>-                         PeerConnection_createDataChannel,
>-                         JNIEnv* jni,
>-                         jobject j_pc,
>-                         jstring j_label,
>-                         jobject j_init) {
>-  DataChannelInit init = JavaToNativeDataChannelInit(jni, j_init);
>-  rtc::scoped_refptr<DataChannelInterface> channel(
>-      ExtractNativePC(jni, j_pc)->CreateDataChannel(
>-          JavaToStdString(jni, j_label), &init));
>-  return WrapNativeDataChannel(jni, channel);
>-}
>-
>-JNI_FUNCTION_DECLARATION(void,
>-                         PeerConnection_createOffer,
>-                         JNIEnv* jni,
>-                         jobject j_pc,
>-                         jobject j_observer,
>-                         jobject j_constraints) {
>-  std::unique_ptr<MediaConstraintsInterface> constraints =
>-      JavaToNativeMediaConstraints(jni, j_constraints);
>-  rtc::scoped_refptr<CreateSdpObserverJni> observer(
>-      new rtc::RefCountedObject<CreateSdpObserverJni>(jni, j_observer,
>-                                                      std::move(constraints)));
>-  ExtractNativePC(jni, j_pc)->CreateOffer(observer, observer->constraints());
>-}
>-
>-JNI_FUNCTION_DECLARATION(void,
>-                         PeerConnection_createAnswer,
>-                         JNIEnv* jni,
>-                         jobject j_pc,
>-                         jobject j_observer,
>-                         jobject j_constraints) {
>-  std::unique_ptr<MediaConstraintsInterface> constraints =
>-      JavaToNativeMediaConstraints(jni, j_constraints);
>-  rtc::scoped_refptr<CreateSdpObserverJni> observer(
>-      new rtc::RefCountedObject<CreateSdpObserverJni>(jni, j_observer,
>-                                                      std::move(constraints)));
>-  ExtractNativePC(jni, j_pc)->CreateAnswer(observer, observer->constraints());
>-}
>-
>-JNI_FUNCTION_DECLARATION(void,
>-                         PeerConnection_setLocalDescription,
>-                         JNIEnv* jni,
>-                         jobject j_pc,
>-                         jobject j_observer,
>-                         jobject j_sdp) {
>-  rtc::scoped_refptr<SetSdpObserverJni> observer(
>-      new rtc::RefCountedObject<SetSdpObserverJni>(jni, j_observer, nullptr));
>-  ExtractNativePC(jni, j_pc)->SetLocalDescription(
>-      observer, JavaToNativeSessionDescription(jni, j_sdp));
>-}
>-
>-JNI_FUNCTION_DECLARATION(void,
>-                         PeerConnection_setRemoteDescription,
>-                         JNIEnv* jni,
>-                         jobject j_pc,
>-                         jobject j_observer,
>-                         jobject j_sdp) {
>-  rtc::scoped_refptr<SetSdpObserverJni> observer(
>-      new rtc::RefCountedObject<SetSdpObserverJni>(jni, j_observer, nullptr));
>-  ExtractNativePC(jni, j_pc)->SetRemoteDescription(
>-      observer, JavaToNativeSessionDescription(jni, j_sdp));
>-}
>-
>-JNI_FUNCTION_DECLARATION(void,
>-                         PeerConnection_setAudioPlayout,
>-                         JNIEnv* jni,
>-                         jobject j_pc,
>-                         jboolean playout) {
>-  ExtractNativePC(jni, j_pc)->SetAudioPlayout(playout);
>-}
>-
>-JNI_FUNCTION_DECLARATION(void,
>-                         PeerConnection_setAudioRecording,
>-                         JNIEnv* jni,
>-                         jobject j_pc,
>-                         jboolean recording) {
>-  ExtractNativePC(jni, j_pc)->SetAudioRecording(recording);
>-}
>-
>-JNI_FUNCTION_DECLARATION(jboolean,
>-                         PeerConnection_nativeSetConfiguration,
>-                         JNIEnv* jni,
>-                         jobject j_pc,
>-                         jobject j_rtc_config,
>-                         jlong native_observer) {
>-  // Need to merge constraints into RTCConfiguration again, which are stored
>-  // in the observer object.
>-  PeerConnectionObserverJni* observer =
>-      reinterpret_cast<PeerConnectionObserverJni*>(native_observer);
>-  PeerConnectionInterface::RTCConfiguration rtc_config(
>-      PeerConnectionInterface::RTCConfigurationType::kAggressive);
>-  JavaToNativeRTCConfiguration(jni, j_rtc_config, &rtc_config);
>-  CopyConstraintsIntoRtcConfiguration(observer->constraints(), &rtc_config);
>-  return ExtractNativePC(jni, j_pc)->SetConfiguration(rtc_config);
>-}
>-
>-JNI_FUNCTION_DECLARATION(jboolean,
>-                         PeerConnection_nativeAddIceCandidate,
>-                         JNIEnv* jni,
>-                         jobject j_pc,
>-                         jstring j_sdp_mid,
>-                         jint j_sdp_mline_index,
>-                         jstring j_candidate_sdp) {
>-  std::string sdp_mid = JavaToStdString(jni, j_sdp_mid);
>-  std::string sdp = JavaToStdString(jni, j_candidate_sdp);
>-  std::unique_ptr<IceCandidateInterface> candidate(
>-      CreateIceCandidate(sdp_mid, j_sdp_mline_index, sdp, nullptr));
>-  return ExtractNativePC(jni, j_pc)->AddIceCandidate(candidate.get());
>-}
>-
>-JNI_FUNCTION_DECLARATION(jboolean,
>-                         PeerConnection_nativeRemoveIceCandidates,
>-                         JNIEnv* jni,
>-                         jobject j_pc,
>-                         jobjectArray j_candidates) {
>-  std::vector<cricket::Candidate> candidates;
>-  size_t num_candidates = jni->GetArrayLength(j_candidates);
>-  for (size_t i = 0; i < num_candidates; ++i) {
>-    jobject j_candidate = jni->GetObjectArrayElement(j_candidates, i);
>-    candidates.push_back(JavaToNativeCandidate(jni, j_candidate));
>-  }
>-  return ExtractNativePC(jni, j_pc)->RemoveIceCandidates(candidates);
>-}
>-
>-JNI_FUNCTION_DECLARATION(jboolean,
>-                         PeerConnection_nativeAddLocalStream,
>-                         JNIEnv* jni,
>-                         jobject j_pc,
>-                         jlong native_stream) {
>-  return ExtractNativePC(jni, j_pc)->AddStream(
>-      reinterpret_cast<MediaStreamInterface*>(native_stream));
>-}
>-
>-JNI_FUNCTION_DECLARATION(void,
>-                         PeerConnection_nativeRemoveLocalStream,
>-                         JNIEnv* jni,
>-                         jobject j_pc,
>-                         jlong native_stream) {
>-  ExtractNativePC(jni, j_pc)->RemoveStream(
>-      reinterpret_cast<MediaStreamInterface*>(native_stream));
>-}
>-
>-JNI_FUNCTION_DECLARATION(jobject,
>-                         PeerConnection_nativeCreateSender,
>-                         JNIEnv* jni,
>-                         jobject j_pc,
>-                         jstring j_kind,
>-                         jstring j_stream_id) {
>-  jclass j_rtp_sender_class = FindClass(jni, "org/webrtc/RtpSender");
>-  jmethodID j_rtp_sender_ctor =
>-      GetMethodID(jni, j_rtp_sender_class, "<init>", "(J)V");
>-
>-  std::string kind = JavaToStdString(jni, j_kind);
>-  std::string stream_id = JavaToStdString(jni, j_stream_id);
>-  rtc::scoped_refptr<RtpSenderInterface> sender =
>-      ExtractNativePC(jni, j_pc)->CreateSender(kind, stream_id);
>-  if (!sender.get()) {
>-    return nullptr;
>-  }
>-  jlong nativeSenderPtr = jlongFromPointer(sender.get());
>-  jobject j_sender =
>-      jni->NewObject(j_rtp_sender_class, j_rtp_sender_ctor, nativeSenderPtr);
>-  CHECK_EXCEPTION(jni) << "error during NewObject";
>-  // Sender is now owned by the Java object, and will be freed from
>-  // RtpSender.dispose(), called by PeerConnection.dispose() or getSenders().
>-  sender->AddRef();
>-  return j_sender;
>-}
>-
>-JNI_FUNCTION_DECLARATION(jobject,
>-                         PeerConnection_nativeGetSenders,
>-                         JNIEnv* jni,
>-                         jobject j_pc) {
>-  jclass j_array_list_class = FindClass(jni, "java/util/ArrayList");
>-  jmethodID j_array_list_ctor =
>-      GetMethodID(jni, j_array_list_class, "<init>", "()V");
>-  jmethodID j_array_list_add =
>-      GetMethodID(jni, j_array_list_class, "add", "(Ljava/lang/Object;)Z");
>-  jobject j_senders = jni->NewObject(j_array_list_class, j_array_list_ctor);
>-  CHECK_EXCEPTION(jni) << "error during NewObject";
>-
>-  jclass j_rtp_sender_class = FindClass(jni, "org/webrtc/RtpSender");
>-  jmethodID j_rtp_sender_ctor =
>-      GetMethodID(jni, j_rtp_sender_class, "<init>", "(J)V");
>-
>-  auto senders = ExtractNativePC(jni, j_pc)->GetSenders();
>-  for (const auto& sender : senders) {
>-    jlong nativeSenderPtr = jlongFromPointer(sender.get());
>-    jobject j_sender =
>-        jni->NewObject(j_rtp_sender_class, j_rtp_sender_ctor, nativeSenderPtr);
>-    CHECK_EXCEPTION(jni) << "error during NewObject";
>-    // Sender is now owned by the Java object, and will be freed from
>-    // RtpSender.dispose(), called by PeerConnection.dispose() or getSenders().
>-    sender->AddRef();
>-    jni->CallBooleanMethod(j_senders, j_array_list_add, j_sender);
>-    CHECK_EXCEPTION(jni) << "error during CallBooleanMethod";
>-  }
>-  return j_senders;
>-}
>-
>-JNI_FUNCTION_DECLARATION(jobject,
>-                         PeerConnection_nativeGetReceivers,
>-                         JNIEnv* jni,
>-                         jobject j_pc) {
>-  jclass j_array_list_class = FindClass(jni, "java/util/ArrayList");
>-  jmethodID j_array_list_ctor =
>-      GetMethodID(jni, j_array_list_class, "<init>", "()V");
>-  jmethodID j_array_list_add =
>-      GetMethodID(jni, j_array_list_class, "add", "(Ljava/lang/Object;)Z");
>-  jobject j_receivers = jni->NewObject(j_array_list_class, j_array_list_ctor);
>-  CHECK_EXCEPTION(jni) << "error during NewObject";
>-
>-  jclass j_rtp_receiver_class = FindClass(jni, "org/webrtc/RtpReceiver");
>-  jmethodID j_rtp_receiver_ctor =
>-      GetMethodID(jni, j_rtp_receiver_class, "<init>", "(J)V");
>-
>-  auto receivers = ExtractNativePC(jni, j_pc)->GetReceivers();
>-  for (const auto& receiver : receivers) {
>-    jlong nativeReceiverPtr = jlongFromPointer(receiver.get());
>-    jobject j_receiver = jni->NewObject(j_rtp_receiver_class,
>-                                        j_rtp_receiver_ctor, nativeReceiverPtr);
>-    CHECK_EXCEPTION(jni) << "error during NewObject";
>-    // Receiver is now owned by Java object, and will be freed from there.
>-    receiver->AddRef();
>-    jni->CallBooleanMethod(j_receivers, j_array_list_add, j_receiver);
>-    CHECK_EXCEPTION(jni) << "error during CallBooleanMethod";
>-  }
>-  return j_receivers;
>-}
>-
>-JNI_FUNCTION_DECLARATION(bool,
>-                         PeerConnection_nativeOldGetStats,
>-                         JNIEnv* jni,
>-                         jobject j_pc,
>-                         jobject j_observer,
>-                         jlong native_track) {
>-  rtc::scoped_refptr<StatsObserverJni> observer(
>-      new rtc::RefCountedObject<StatsObserverJni>(jni, j_observer));
>-  return ExtractNativePC(jni, j_pc)->GetStats(
>-      observer, reinterpret_cast<MediaStreamTrackInterface*>(native_track),
>-      PeerConnectionInterface::kStatsOutputLevelStandard);
>-}
>-
>-JNI_FUNCTION_DECLARATION(void,
>-                         PeerConnection_nativeNewGetStats,
>-                         JNIEnv* jni,
>-                         jobject j_pc,
>-                         jobject j_callback) {
>-  rtc::scoped_refptr<RTCStatsCollectorCallbackWrapper> callback(
>-      new rtc::RefCountedObject<RTCStatsCollectorCallbackWrapper>(jni,
>-                                                                  j_callback));
>-  ExtractNativePC(jni, j_pc)->GetStats(callback);
>-}
>-
>-JNI_FUNCTION_DECLARATION(jboolean,
>-                         PeerConnection_setBitrate,
>-                         JNIEnv* jni,
>-                         jobject j_pc,
>-                         jobject j_min,
>-                         jobject j_current,
>-                         jobject j_max) {
>-  PeerConnectionInterface::BitrateParameters params;
>-  params.min_bitrate_bps = JavaToNativeOptionalInt(jni, j_min);
>-  params.current_bitrate_bps = JavaToNativeOptionalInt(jni, j_current);
>-  params.max_bitrate_bps = JavaToNativeOptionalInt(jni, j_max);
>-  return ExtractNativePC(jni, j_pc)->SetBitrate(params).ok();
>-}
>-
>-JNI_FUNCTION_DECLARATION(bool,
>-                         PeerConnection_nativeStartRtcEventLog,
>-                         JNIEnv* jni,
>-                         jobject j_pc,
>-                         int file_descriptor,
>-                         int max_size_bytes) {
>-  return ExtractNativePC(jni, j_pc)->StartRtcEventLog(file_descriptor,
>-                                                      max_size_bytes);
>-}
>-
>-JNI_FUNCTION_DECLARATION(void,
>-                         PeerConnection_nativeStopRtcEventLog,
>-                         JNIEnv* jni,
>-                         jobject j_pc) {
>-  ExtractNativePC(jni, j_pc)->StopRtcEventLog();
>-}
>-
>-JNI_FUNCTION_DECLARATION(jobject,
>-                         PeerConnection_signalingState,
>-                         JNIEnv* jni,
>-                         jobject j_pc) {
>-  PeerConnectionInterface::SignalingState state =
>-      ExtractNativePC(jni, j_pc)->signaling_state();
>-  return JavaEnumFromIndexAndClassName(jni, "PeerConnection$SignalingState",
>-                                       state);
>-}
>-
>-JNI_FUNCTION_DECLARATION(jobject,
>-                         PeerConnection_iceConnectionState,
>-                         JNIEnv* jni,
>-                         jobject j_pc) {
>-  PeerConnectionInterface::IceConnectionState state =
>-      ExtractNativePC(jni, j_pc)->ice_connection_state();
>-  return JavaEnumFromIndexAndClassName(jni, "PeerConnection$IceConnectionState",
>-                                       state);
>-}
>-
>-JNI_FUNCTION_DECLARATION(jobject,
>-                         PeerConnection_iceGatheringState,
>-                         JNIEnv* jni,
>-                         jobject j_pc) {
>-  PeerConnectionInterface::IceGatheringState state =
>-      ExtractNativePC(jni, j_pc)->ice_gathering_state();
>-  return JavaEnumFromIndexAndClassName(jni, "PeerConnection$IceGatheringState",
>-                                       state);
>-}
>-
>-JNI_FUNCTION_DECLARATION(void,
>-                         PeerConnection_close,
>-                         JNIEnv* jni,
>-                         jobject j_pc) {
>-  ExtractNativePC(jni, j_pc)->Close();
>-  return;
>-}
>-
>-}  // namespace jni
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/peerconnectionfactory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/peerconnectionfactory.cc
>new file mode 100644
>index 00000000000..ba1be384532
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/peerconnectionfactory.cc
>@@ -0,0 +1,482 @@
>+/*
>+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "sdk/android/src/jni/pc/peerconnectionfactory.h"
>+
>+#include <memory>
>+#include <utility>
>+
>+#include "api/video_codecs/video_decoder_factory.h"
>+#include "api/video_codecs/video_encoder_factory.h"
>+#include "media/base/mediaengine.h"
>+#include "modules/audio_device/include/audio_device.h"
>+#include "modules/utility/include/jvm_android.h"
>+// We don't depend on the audio processing module implementation.
>+// The user may pass in a nullptr.
>+#include "modules/audio_processing/include/audio_processing.h"  // nogncheck
>+#include "rtc_base/event_tracer.h"
>+#include "rtc_base/stringutils.h"
>+#include "rtc_base/thread.h"
>+#include "sdk/android/generated_peerconnection_jni/jni/PeerConnectionFactory_jni.h"
>+#include "sdk/android/native_api/jni/java_types.h"
>+#include "sdk/android/src/jni/jni_helpers.h"
>+#include "sdk/android/src/jni/logging/logsink.h"
>+#include "sdk/android/src/jni/pc/androidnetworkmonitor.h"
>+#include "sdk/android/src/jni/pc/audio.h"
>+#include "sdk/android/src/jni/pc/icecandidate.h"
>+#include "sdk/android/src/jni/pc/media.h"
>+#include "sdk/android/src/jni/pc/ownedfactoryandthreads.h"
>+#include "sdk/android/src/jni/pc/peerconnection.h"
>+#include "sdk/android/src/jni/pc/sslcertificateverifierwrapper.h"
>+#include "sdk/android/src/jni/pc/video.h"
>+#include "system_wrappers/include/field_trial.h"
>+// Adding 'nogncheck' to disable the gn include headers check.
>+// We don't want to depend on 'system_wrappers:field_trial_default' because
>+// clients should be able to provide their own implementation.
>+#include "system_wrappers/include/field_trial_default.h"  // nogncheck
>+
>+namespace webrtc {
>+namespace jni {
>+
>+namespace {
>+PeerConnectionFactoryInterface::Options
>+JavaToNativePeerConnectionFactoryOptions(JNIEnv* jni,
>+                                         const JavaRef<jobject>& options) {
>+  int network_ignore_mask = Java_Options_getNetworkIgnoreMask(jni, options);
>+  bool disable_encryption = Java_Options_getDisableEncryption(jni, options);
>+  bool disable_network_monitor =
>+      Java_Options_getDisableNetworkMonitor(jni, options);
>+  bool enable_aes128_sha1_32_crypto_cipher =
>+      Java_Options_getEnableAes128Sha1_32CryptoCipher(jni, options);
>+  bool enable_gcm_crypto_suites =
>+      Java_Options_getEnableGcmCryptoSuites(jni, options);
>+
>+  PeerConnectionFactoryInterface::Options native_options;
>+
>+  // This doesn't necessarily match the c++ version of this struct; feel free
>+  // to add more parameters as necessary.
>+  native_options.network_ignore_mask = network_ignore_mask;
>+  native_options.disable_encryption = disable_encryption;
>+  native_options.disable_network_monitor = disable_network_monitor;
>+
>+  native_options.crypto_options.enable_aes128_sha1_32_crypto_cipher =
>+      enable_aes128_sha1_32_crypto_cipher;
>+  native_options.crypto_options.enable_gcm_crypto_suites =
>+      enable_gcm_crypto_suites;
>+  return native_options;
>+}
>+}  // namespace
>+
>+// Note: Some of the video-specific PeerConnectionFactory methods are
>+// implemented in "video.cc". This is done so that if an application
>+// doesn't need video support, it can just link with "null_video.cc"
>+// instead of "video.cc", which doesn't bring in the video-specific
>+// dependencies.
>+
>+// Field trials initialization string
>+static std::unique_ptr<std::string> field_trials_init_string;
>+
>+// Set in PeerConnectionFactory_initializeAndroidGlobals().
>+static bool factory_static_initialized = false;
>+
>+// Set in PeerConnectionFactory_InjectLoggable().
>+static std::unique_ptr<JNILogSink> jni_log_sink;
>+
>+void PeerConnectionFactoryNetworkThreadReady() {
>+  RTC_LOG(LS_INFO) << "Network thread JavaCallback";
>+  JNIEnv* env = AttachCurrentThreadIfNeeded();
>+  Java_PeerConnectionFactory_onNetworkThreadReady(env);
>+}
>+
>+void PeerConnectionFactoryWorkerThreadReady() {
>+  RTC_LOG(LS_INFO) << "Worker thread JavaCallback";
>+  JNIEnv* env = AttachCurrentThreadIfNeeded();
>+  Java_PeerConnectionFactory_onWorkerThreadReady(env);
>+}
>+
>+void PeerConnectionFactorySignalingThreadReady() {
>+  RTC_LOG(LS_INFO) << "Signaling thread JavaCallback";
>+  JNIEnv* env = AttachCurrentThreadIfNeeded();
>+  Java_PeerConnectionFactory_onSignalingThreadReady(env);
>+}
>+
>+jobject NativeToJavaPeerConnectionFactory(
>+    JNIEnv* jni,
>+    rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> pcf,
>+    std::unique_ptr<rtc::Thread> network_thread,
>+    std::unique_ptr<rtc::Thread> worker_thread,
>+    std::unique_ptr<rtc::Thread> signaling_thread,
>+    rtc::NetworkMonitorFactory* network_monitor_factory) {
>+  jni::OwnedFactoryAndThreads* owned_factory = new jni::OwnedFactoryAndThreads(
>+      std::move(network_thread), std::move(worker_thread),
>+      std::move(signaling_thread), network_monitor_factory, pcf.release());
>+  owned_factory->InvokeJavaCallbacksOnFactoryThreads();
>+
>+  return Java_PeerConnectionFactory_Constructor(
>+             jni, NativeToJavaPointer(owned_factory))
>+      .Release();
>+}
>+
>+static void JNI_PeerConnectionFactory_InitializeAndroidGlobals(
>+    JNIEnv* jni,
>+    const JavaParamRef<jclass>&) {
>+  if (!factory_static_initialized) {
>+    JVM::Initialize(GetJVM());
>+    factory_static_initialized = true;
>+  }
>+}
>+
>+static void JNI_PeerConnectionFactory_InitializeFieldTrials(
>+    JNIEnv* jni,
>+    const JavaParamRef<jclass>&,
>+    const JavaParamRef<jstring>& j_trials_init_string) {
>+  if (j_trials_init_string.is_null()) {
>+    field_trials_init_string = nullptr;
>+    field_trial::InitFieldTrialsFromString(nullptr);
>+    return;
>+  }
>+  field_trials_init_string = absl::make_unique<std::string>(
>+      JavaToNativeString(jni, j_trials_init_string));
>+  RTC_LOG(LS_INFO) << "initializeFieldTrials: " << *field_trials_init_string;
>+  field_trial::InitFieldTrialsFromString(field_trials_init_string->c_str());
>+}
>+
>+static void JNI_PeerConnectionFactory_InitializeInternalTracer(
>+    JNIEnv* jni,
>+    const JavaParamRef<jclass>&) {
>+  rtc::tracing::SetupInternalTracer();
>+}
>+
>+static ScopedJavaLocalRef<jstring>
>+JNI_PeerConnectionFactory_FindFieldTrialsFullName(
>+    JNIEnv* jni,
>+    const JavaParamRef<jclass>&,
>+    const JavaParamRef<jstring>& j_name) {
>+  return NativeToJavaString(
>+      jni, field_trial::FindFullName(JavaToStdString(jni, j_name)));
>+}
>+
>+static jboolean JNI_PeerConnectionFactory_StartInternalTracingCapture(
>+    JNIEnv* jni,
>+    const JavaParamRef<jclass>&,
>+    const JavaParamRef<jstring>& j_event_tracing_filename) {
>+  if (j_event_tracing_filename.is_null())
>+    return false;
>+
>+  const char* init_string =
>+      jni->GetStringUTFChars(j_event_tracing_filename.obj(), NULL);
>+  RTC_LOG(LS_INFO) << "Starting internal tracing to: " << init_string;
>+  bool ret = rtc::tracing::StartInternalCapture(init_string);
>+  jni->ReleaseStringUTFChars(j_event_tracing_filename.obj(), init_string);
>+  return ret;
>+}
>+
>+static void JNI_PeerConnectionFactory_StopInternalTracingCapture(
>+    JNIEnv* jni,
>+    const JavaParamRef<jclass>&) {
>+  rtc::tracing::StopInternalCapture();
>+}
>+
>+static void JNI_PeerConnectionFactory_ShutdownInternalTracer(
>+    JNIEnv* jni,
>+    const JavaParamRef<jclass>&) {
>+  rtc::tracing::ShutdownInternalTracer();
>+}
>+
>+jlong CreatePeerConnectionFactoryForJava(
>+    JNIEnv* jni,
>+    const JavaParamRef<jobject>& jcontext,
>+    const JavaParamRef<jobject>& joptions,
>+    rtc::scoped_refptr<AudioDeviceModule> audio_device_module,
>+    const JavaParamRef<jobject>& jencoder_factory,
>+    const JavaParamRef<jobject>& jdecoder_factory,
>+    rtc::scoped_refptr<AudioProcessing> audio_processor,
>+    std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory) {
>+  // talk/ assumes pretty widely that the current Thread is ThreadManager'd, but
>+  // ThreadManager only WrapCurrentThread()s the thread where it is first
>+  // created.  Since the semantics around when auto-wrapping happens in
>+  // webrtc/rtc_base/ are convoluted, we simply wrap here to avoid having to
>+  // think about ramifications of auto-wrapping there.
>+  rtc::ThreadManager::Instance()->WrapCurrentThread();
>+
>+  std::unique_ptr<rtc::Thread> network_thread =
>+      rtc::Thread::CreateWithSocketServer();
>+  network_thread->SetName("network_thread", nullptr);
>+  RTC_CHECK(network_thread->Start()) << "Failed to start thread";
>+
>+  std::unique_ptr<rtc::Thread> worker_thread = rtc::Thread::Create();
>+  worker_thread->SetName("worker_thread", nullptr);
>+  RTC_CHECK(worker_thread->Start()) << "Failed to start thread";
>+
>+  std::unique_ptr<rtc::Thread> signaling_thread = rtc::Thread::Create();
>+  signaling_thread->SetName("signaling_thread", NULL);
>+  RTC_CHECK(signaling_thread->Start()) << "Failed to start thread";
>+
>+  rtc::NetworkMonitorFactory* network_monitor_factory = nullptr;
>+  auto audio_encoder_factory = CreateAudioEncoderFactory();
>+  auto audio_decoder_factory = CreateAudioDecoderFactory();
>+
>+  PeerConnectionFactoryInterface::Options options;
>+  bool has_options = !joptions.is_null();
>+  if (has_options) {
>+    options = JavaToNativePeerConnectionFactoryOptions(jni, joptions);
>+  }
>+
>+  // Do not create network_monitor_factory only if the options are
>+  // provided and disable_network_monitor therein is set to true.
>+  if (!(has_options && options.disable_network_monitor)) {
>+    network_monitor_factory = new AndroidNetworkMonitorFactory();
>+    rtc::NetworkMonitorFactory::SetFactory(network_monitor_factory);
>+  }
>+
>+  rtc::scoped_refptr<AudioMixer> audio_mixer = nullptr;
>+  std::unique_ptr<CallFactoryInterface> call_factory(CreateCallFactory());
>+  std::unique_ptr<RtcEventLogFactoryInterface> rtc_event_log_factory(
>+      CreateRtcEventLogFactory());
>+
>+  std::unique_ptr<cricket::MediaEngineInterface> media_engine(CreateMediaEngine(
>+      audio_device_module, audio_encoder_factory, audio_decoder_factory,
>+      std::unique_ptr<VideoEncoderFactory>(
>+          CreateVideoEncoderFactory(jni, jencoder_factory)),
>+      std::unique_ptr<VideoDecoderFactory>(
>+          CreateVideoDecoderFactory(jni, jdecoder_factory)),
>+      audio_mixer, audio_processor));
>+
>+  rtc::scoped_refptr<PeerConnectionFactoryInterface> factory(
>+      CreateModularPeerConnectionFactory(
>+          network_thread.get(), worker_thread.get(), signaling_thread.get(),
>+          std::move(media_engine), std::move(call_factory),
>+          std::move(rtc_event_log_factory), std::move(fec_controller_factory)));
>+  RTC_CHECK(factory) << "Failed to create the peer connection factory; "
>+                     << "WebRTC/libjingle init likely failed on this device";
>+  // TODO(honghaiz): Maybe put the options as the argument of
>+  // CreatePeerConnectionFactory.
>+  if (has_options) {
>+    factory->SetOptions(options);
>+  }
>+  OwnedFactoryAndThreads* owned_factory = new OwnedFactoryAndThreads(
>+      std::move(network_thread), std::move(worker_thread),
>+      std::move(signaling_thread), network_monitor_factory, factory.release());
>+  owned_factory->InvokeJavaCallbacksOnFactoryThreads();
>+  return jlongFromPointer(owned_factory);
>+}
>+
>+static jlong JNI_PeerConnectionFactory_CreatePeerConnectionFactory(
>+    JNIEnv* jni,
>+    const JavaParamRef<jclass>&,
>+    const JavaParamRef<jobject>& jcontext,
>+    const JavaParamRef<jobject>& joptions,
>+    jlong native_audio_device_module,
>+    const JavaParamRef<jobject>& jencoder_factory,
>+    const JavaParamRef<jobject>& jdecoder_factory,
>+    jlong native_audio_processor,
>+    jlong native_fec_controller_factory) {
>+  rtc::scoped_refptr<AudioProcessing> audio_processor =
>+      reinterpret_cast<AudioProcessing*>(native_audio_processor);
>+  std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory(
>+      reinterpret_cast<FecControllerFactoryInterface*>(
>+          native_fec_controller_factory));
>+  return CreatePeerConnectionFactoryForJava(
>+      jni, jcontext, joptions,
>+      reinterpret_cast<AudioDeviceModule*>(native_audio_device_module),
>+      jencoder_factory, jdecoder_factory,
>+      audio_processor ? audio_processor : CreateAudioProcessing(),
>+      std::move(fec_controller_factory));
>+}
>+
>+static void JNI_PeerConnectionFactory_FreeFactory(JNIEnv*,
>+                                                  const JavaParamRef<jclass>&,
>+                                                  jlong j_p) {
>+  delete reinterpret_cast<OwnedFactoryAndThreads*>(j_p);
>+  field_trial::InitFieldTrialsFromString(nullptr);
>+  field_trials_init_string = nullptr;
>+}
>+
>+static void JNI_PeerConnectionFactory_InvokeThreadsCallbacks(
>+    JNIEnv*,
>+    const JavaParamRef<jclass>&,
>+    jlong j_p) {
>+  OwnedFactoryAndThreads* factory =
>+      reinterpret_cast<OwnedFactoryAndThreads*>(j_p);
>+  factory->InvokeJavaCallbacksOnFactoryThreads();
>+}
>+
>+static jlong JNI_PeerConnectionFactory_CreateLocalMediaStream(
>+    JNIEnv* jni,
>+    const JavaParamRef<jclass>&,
>+    jlong native_factory,
>+    const JavaParamRef<jstring>& label) {
>+  rtc::scoped_refptr<PeerConnectionFactoryInterface> factory(
>+      factoryFromJava(native_factory));
>+  rtc::scoped_refptr<MediaStreamInterface> stream(
>+      factory->CreateLocalMediaStream(JavaToStdString(jni, label)));
>+  return jlongFromPointer(stream.release());
>+}
>+
>+static jlong JNI_PeerConnectionFactory_CreateAudioSource(
>+    JNIEnv* jni,
>+    const JavaParamRef<jclass>&,
>+    jlong native_factory,
>+    const JavaParamRef<jobject>& j_constraints) {
>+  std::unique_ptr<MediaConstraintsInterface> constraints =
>+      JavaToNativeMediaConstraints(jni, j_constraints);
>+  rtc::scoped_refptr<PeerConnectionFactoryInterface> factory(
>+      factoryFromJava(native_factory));
>+  cricket::AudioOptions options;
>+  CopyConstraintsIntoAudioOptions(constraints.get(), &options);
>+  rtc::scoped_refptr<AudioSourceInterface> source(
>+      factory->CreateAudioSource(options));
>+  return jlongFromPointer(source.release());
>+}
>+
>+jlong JNI_PeerConnectionFactory_CreateAudioTrack(
>+    JNIEnv* jni,
>+    const JavaParamRef<jclass>&,
>+    jlong native_factory,
>+    const JavaParamRef<jstring>& id,
>+    jlong native_source) {
>+  rtc::scoped_refptr<PeerConnectionFactoryInterface> factory(
>+      factoryFromJava(native_factory));
>+  rtc::scoped_refptr<AudioTrackInterface> track(factory->CreateAudioTrack(
>+      JavaToStdString(jni, id),
>+      reinterpret_cast<AudioSourceInterface*>(native_source)));
>+  return jlongFromPointer(track.release());
>+}
>+
>+static jboolean JNI_PeerConnectionFactory_StartAecDump(
>+    JNIEnv* jni,
>+    const JavaParamRef<jclass>&,
>+    jlong native_factory,
>+    jint file,
>+    jint filesize_limit_bytes) {
>+  rtc::scoped_refptr<PeerConnectionFactoryInterface> factory(
>+      factoryFromJava(native_factory));
>+  return factory->StartAecDump(file, filesize_limit_bytes);
>+}
>+
>+static void JNI_PeerConnectionFactory_StopAecDump(JNIEnv* jni,
>+                                                  const JavaParamRef<jclass>&,
>+                                                  jlong native_factory) {
>+  rtc::scoped_refptr<PeerConnectionFactoryInterface> factory(
>+      factoryFromJava(native_factory));
>+  factory->StopAecDump();
>+}
>+
>+static jlong JNI_PeerConnectionFactory_CreatePeerConnection(
>+    JNIEnv* jni,
>+    const JavaParamRef<jclass>&,
>+    jlong factory,
>+    const JavaParamRef<jobject>& j_rtc_config,
>+    const JavaParamRef<jobject>& j_constraints,
>+    jlong observer_p,
>+    const JavaParamRef<jobject>& j_sslCertificateVerifier) {
>+  rtc::scoped_refptr<PeerConnectionFactoryInterface> f(
>+      reinterpret_cast<PeerConnectionFactoryInterface*>(
>+          factoryFromJava(factory)));
>+  std::unique_ptr<PeerConnectionObserver> observer(
>+      reinterpret_cast<PeerConnectionObserver*>(observer_p));
>+
>+  PeerConnectionInterface::RTCConfiguration rtc_config(
>+      PeerConnectionInterface::RTCConfigurationType::kAggressive);
>+  JavaToNativeRTCConfiguration(jni, j_rtc_config, &rtc_config);
>+
>+  // Generate non-default certificate.
>+  rtc::KeyType key_type = GetRtcConfigKeyType(jni, j_rtc_config);
>+  if (key_type != rtc::KT_DEFAULT) {
>+    rtc::scoped_refptr<rtc::RTCCertificate> certificate =
>+        rtc::RTCCertificateGenerator::GenerateCertificate(
>+            rtc::KeyParams(key_type), absl::nullopt);
>+    if (!certificate) {
>+      RTC_LOG(LS_ERROR) << "Failed to generate certificate. KeyType: "
>+                        << key_type;
>+      return 0;
>+    }
>+    rtc_config.certificates.push_back(certificate);
>+  }
>+
>+  std::unique_ptr<MediaConstraintsInterface> constraints;
>+  if (!j_constraints.is_null()) {
>+    constraints = JavaToNativeMediaConstraints(jni, j_constraints);
>+    CopyConstraintsIntoRtcConfiguration(constraints.get(), &rtc_config);
>+  }
>+
>+  PeerConnectionDependencies peer_connection_dependencies(observer.get());
>+  if (!j_sslCertificateVerifier.is_null()) {
>+    peer_connection_dependencies.tls_cert_verifier =
>+        absl::make_unique<SSLCertificateVerifierWrapper>(
>+            jni, j_sslCertificateVerifier);
>+  }
>+
>+  rtc::scoped_refptr<PeerConnectionInterface> pc(f->CreatePeerConnection(
>+      rtc_config, std::move(peer_connection_dependencies)));
>+
>+  return jlongFromPointer(
>+      new OwnedPeerConnection(pc, std::move(observer), std::move(constraints)));
>+}
>+
>+static jlong JNI_PeerConnectionFactory_CreateVideoSource(
>+    JNIEnv* jni,
>+    const JavaParamRef<jclass>&,
>+    jlong native_factory,
>+    jboolean is_screencast) {
>+  OwnedFactoryAndThreads* factory =
>+      reinterpret_cast<OwnedFactoryAndThreads*>(native_factory);
>+  return jlongFromPointer(CreateVideoSource(jni, factory->signaling_thread(),
>+                                            factory->worker_thread(),
>+                                            is_screencast));
>+}
>+
>+static jlong JNI_PeerConnectionFactory_CreateVideoTrack(
>+    JNIEnv* jni,
>+    const JavaParamRef<jclass>&,
>+    jlong native_factory,
>+    const JavaParamRef<jstring>& id,
>+    jlong native_source) {
>+  rtc::scoped_refptr<PeerConnectionFactoryInterface> factory(
>+      factoryFromJava(native_factory));
>+  rtc::scoped_refptr<VideoTrackInterface> track(factory->CreateVideoTrack(
>+      JavaToStdString(jni, id),
>+      reinterpret_cast<VideoTrackSourceInterface*>(native_source)));
>+  return jlongFromPointer(track.release());
>+}
>+
>+static jlong JNI_PeerConnectionFactory_GetNativePeerConnectionFactory(
>+    JNIEnv* jni,
>+    const JavaParamRef<jclass>&,
>+    jlong native_factory) {
>+  return jlongFromPointer(factoryFromJava(native_factory));
>+}
>+
>+static void JNI_PeerConnectionFactory_InjectLoggable(
>+    JNIEnv* jni,
>+    const JavaParamRef<jclass>&,
>+    const JavaParamRef<jobject>& j_logging,
>+    jint nativeSeverity) {
>+  // If there is already a LogSink, remove it from LogMessage.
>+  if (jni_log_sink) {
>+    rtc::LogMessage::RemoveLogToStream(jni_log_sink.get());
>+  }
>+  jni_log_sink = absl::make_unique<JNILogSink>(jni, j_logging);
>+  rtc::LogMessage::AddLogToStream(
>+      jni_log_sink.get(), static_cast<rtc::LoggingSeverity>(nativeSeverity));
>+  rtc::LogMessage::LogToDebug(rtc::LS_NONE);
>+}
>+
>+static void JNI_PeerConnectionFactory_DeleteLoggable(
>+    JNIEnv* jni,
>+    const JavaParamRef<jclass>&) {
>+  if (jni_log_sink) {
>+    rtc::LogMessage::RemoveLogToStream(jni_log_sink.get());
>+    jni_log_sink.reset();
>+  }
>+}
>+
>+}  // namespace jni
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/peerconnectionfactory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/peerconnectionfactory.h
>new file mode 100644
>index 00000000000..6f09776df93
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/peerconnectionfactory.h
>@@ -0,0 +1,37 @@
>+/*
>+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef SDK_ANDROID_SRC_JNI_PC_PEERCONNECTIONFACTORY_H_
>+#define SDK_ANDROID_SRC_JNI_PC_PEERCONNECTIONFACTORY_H_
>+
>+#include <jni.h>
>+#include "api/peerconnectioninterface.h"
>+#include "rtc_base/thread.h"
>+
>+namespace webrtc {
>+namespace jni {
>+
>+void PeerConnectionFactoryNetworkThreadReady();
>+void PeerConnectionFactoryWorkerThreadReady();
>+void PeerConnectionFactorySignalingThreadReady();
>+
>+// Creates java PeerConnectionFactory with specified |pcf|.
>+jobject NativeToJavaPeerConnectionFactory(
>+    JNIEnv* jni,
>+    rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> pcf,
>+    std::unique_ptr<rtc::Thread> network_thread,
>+    std::unique_ptr<rtc::Thread> worker_thread,
>+    std::unique_ptr<rtc::Thread> signaling_thread,
>+    rtc::NetworkMonitorFactory* network_monitor_factory = nullptr);
>+
>+}  // namespace jni
>+}  // namespace webrtc
>+
>+#endif  // SDK_ANDROID_SRC_JNI_PC_PEERCONNECTIONFACTORY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/peerconnectionfactory_jni.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/peerconnectionfactory_jni.cc
>deleted file mode 100644
>index 5882243e2e1..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/peerconnectionfactory_jni.cc
>+++ /dev/null
>@@ -1,440 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include <memory>
>-#include <utility>
>-
>-#include "api/peerconnectioninterface.h"
>-#include "api/video_codecs/video_decoder_factory.h"
>-#include "api/video_codecs/video_encoder_factory.h"
>-#include "media/base/mediaengine.h"
>-#include "modules/audio_device/include/audio_device.h"
>-#include "modules/utility/include/jvm_android.h"
>-// We don't depend on the audio processing module implementation.
>-// The user may pass in a nullptr.
>-#include "modules/audio_processing/include/audio_processing.h"  // nogncheck
>-#include "rtc_base/event_tracer.h"
>-#include "rtc_base/stringutils.h"
>-#include "rtc_base/thread.h"
>-#include "sdk/android/src/jni/jni_helpers.h"
>-#include "sdk/android/src/jni/pc/androidnetworkmonitor_jni.h"
>-#include "sdk/android/src/jni/pc/audio_jni.h"
>-#include "sdk/android/src/jni/pc/java_native_conversion.h"
>-#include "sdk/android/src/jni/pc/media_jni.h"
>-#include "sdk/android/src/jni/pc/ownedfactoryandthreads.h"
>-#include "sdk/android/src/jni/pc/peerconnectionobserver_jni.h"
>-#include "sdk/android/src/jni/pc/video_jni.h"
>-#include "system_wrappers/include/field_trial.h"
>-// Adding 'nogncheck' to disable the gn include headers check.
>-// We don't want to depend on 'system_wrappers:field_trial_default' because
>-// clients should be able to provide their own implementation.
>-#include "system_wrappers/include/field_trial_default.h"  // nogncheck
>-
>-namespace webrtc {
>-namespace jni {
>-
>-// Note: Some of the video-specific PeerConnectionFactory methods are
>-// implemented in "video_jni.cc". This is done so that if an application
>-// doesn't need video support, it can just link with "null_video_jni.cc"
>-// instead of "video_jni.cc", which doesn't bring in the video-specific
>-// dependencies.
>-
>-// Field trials initialization string
>-static char* field_trials_init_string = nullptr;
>-
>-// Set in PeerConnectionFactory_initializeAndroidGlobals().
>-static bool factory_static_initialized = false;
>-static bool video_hw_acceleration_enabled = true;
>-
>-JNI_FUNCTION_DECLARATION(jlong,
>-                         PeerConnectionFactory_nativeCreateObserver,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jobject j_observer) {
>-  return (jlong) new PeerConnectionObserverJni(jni, j_observer);
>-}
>-
>-JNI_FUNCTION_DECLARATION(void,
>-                         PeerConnectionFactory_nativeInitializeAndroidGlobals,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jobject context,
>-                         jboolean video_hw_acceleration) {
>-  video_hw_acceleration_enabled = video_hw_acceleration;
>-  if (!factory_static_initialized) {
>-    JVM::Initialize(GetJVM());
>-    factory_static_initialized = true;
>-  }
>-}
>-
>-JNI_FUNCTION_DECLARATION(void,
>-                         PeerConnectionFactory_initializeFieldTrials,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jstring j_trials_init_string) {
>-  field_trials_init_string = NULL;
>-  if (j_trials_init_string != NULL) {
>-    const char* init_string =
>-        jni->GetStringUTFChars(j_trials_init_string, NULL);
>-    int init_string_length = jni->GetStringUTFLength(j_trials_init_string);
>-    field_trials_init_string = new char[init_string_length + 1];
>-    rtc::strcpyn(field_trials_init_string, init_string_length + 1, init_string);
>-    jni->ReleaseStringUTFChars(j_trials_init_string, init_string);
>-    RTC_LOG(LS_INFO) << "initializeFieldTrials: " << field_trials_init_string;
>-  }
>-  field_trial::InitFieldTrialsFromString(field_trials_init_string);
>-}
>-
>-JNI_FUNCTION_DECLARATION(void,
>-                         PeerConnectionFactory_nativeInitializeInternalTracer,
>-                         JNIEnv* jni,
>-                         jclass) {
>-  rtc::tracing::SetupInternalTracer();
>-}
>-
>-JNI_FUNCTION_DECLARATION(jstring,
>-                         PeerConnectionFactory_nativeFieldTrialsFindFullName,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jstring j_name) {
>-  return NativeToJavaString(
>-      jni, field_trial::FindFullName(JavaToStdString(jni, j_name)));
>-}
>-
>-JNI_FUNCTION_DECLARATION(jboolean,
>-                         PeerConnectionFactory_startInternalTracingCapture,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jstring j_event_tracing_filename) {
>-  if (!j_event_tracing_filename)
>-    return false;
>-
>-  const char* init_string =
>-      jni->GetStringUTFChars(j_event_tracing_filename, NULL);
>-  RTC_LOG(LS_INFO) << "Starting internal tracing to: " << init_string;
>-  bool ret = rtc::tracing::StartInternalCapture(init_string);
>-  jni->ReleaseStringUTFChars(j_event_tracing_filename, init_string);
>-  return ret;
>-}
>-
>-JNI_FUNCTION_DECLARATION(void,
>-                         PeerConnectionFactory_stopInternalTracingCapture,
>-                         JNIEnv* jni,
>-                         jclass) {
>-  rtc::tracing::StopInternalCapture();
>-}
>-
>-JNI_FUNCTION_DECLARATION(void,
>-                         PeerConnectionFactory_nativeShutdownInternalTracer,
>-                         JNIEnv* jni,
>-                         jclass) {
>-  rtc::tracing::ShutdownInternalTracer();
>-}
>-
>-jlong CreatePeerConnectionFactoryForJava(
>-    JNIEnv* jni,
>-    jobject joptions,
>-    jobject jencoder_factory,
>-    jobject jdecoder_factory,
>-    rtc::scoped_refptr<AudioProcessing> audio_processor) {
>-  // talk/ assumes pretty widely that the current Thread is ThreadManager'd, but
>-  // ThreadManager only WrapCurrentThread()s the thread where it is first
>-  // created.  Since the semantics around when auto-wrapping happens in
>-  // webrtc/rtc_base/ are convoluted, we simply wrap here to avoid having to
>-  // think about ramifications of auto-wrapping there.
>-  rtc::ThreadManager::Instance()->WrapCurrentThread();
>-
>-  std::unique_ptr<rtc::Thread> network_thread =
>-      rtc::Thread::CreateWithSocketServer();
>-  network_thread->SetName("network_thread", nullptr);
>-  RTC_CHECK(network_thread->Start()) << "Failed to start thread";
>-
>-  std::unique_ptr<rtc::Thread> worker_thread = rtc::Thread::Create();
>-  worker_thread->SetName("worker_thread", nullptr);
>-  RTC_CHECK(worker_thread->Start()) << "Failed to start thread";
>-
>-  std::unique_ptr<rtc::Thread> signaling_thread = rtc::Thread::Create();
>-  signaling_thread->SetName("signaling_thread", NULL);
>-  RTC_CHECK(signaling_thread->Start()) << "Failed to start thread";
>-
>-  rtc::NetworkMonitorFactory* network_monitor_factory = nullptr;
>-  auto audio_encoder_factory = CreateAudioEncoderFactory();
>-  auto audio_decoder_factory = CreateAudioDecoderFactory();
>-
>-  PeerConnectionFactoryInterface::Options options;
>-  bool has_options = joptions != NULL;
>-  if (has_options) {
>-    options = JavaToNativePeerConnectionFactoryOptions(jni, joptions);
>-  }
>-
>-  // Do not create network_monitor_factory only if the options are
>-  // provided and disable_network_monitor therein is set to true.
>-  if (!(has_options && options.disable_network_monitor)) {
>-    network_monitor_factory = new AndroidNetworkMonitorFactory();
>-    rtc::NetworkMonitorFactory::SetFactory(network_monitor_factory);
>-  }
>-
>-  AudioDeviceModule* adm = nullptr;
>-  rtc::scoped_refptr<AudioMixer> audio_mixer = nullptr;
>-  std::unique_ptr<CallFactoryInterface> call_factory(CreateCallFactory());
>-  std::unique_ptr<RtcEventLogFactoryInterface> rtc_event_log_factory(
>-      CreateRtcEventLogFactory());
>-
>-  cricket::WebRtcVideoEncoderFactory* legacy_video_encoder_factory = nullptr;
>-  cricket::WebRtcVideoDecoderFactory* legacy_video_decoder_factory = nullptr;
>-  std::unique_ptr<cricket::MediaEngineInterface> media_engine;
>-  if (jencoder_factory == nullptr && jdecoder_factory == nullptr) {
>-    // This uses the legacy API, which automatically uses the internal SW
>-    // codecs in WebRTC.
>-    if (video_hw_acceleration_enabled) {
>-      legacy_video_encoder_factory = CreateLegacyVideoEncoderFactory();
>-      legacy_video_decoder_factory = CreateLegacyVideoDecoderFactory();
>-    }
>-    media_engine.reset(CreateMediaEngine(
>-        adm, audio_encoder_factory, audio_decoder_factory,
>-        legacy_video_encoder_factory, legacy_video_decoder_factory, audio_mixer,
>-        audio_processor));
>-  } else {
>-    // This uses the new API, does not automatically include software codecs.
>-    std::unique_ptr<VideoEncoderFactory> video_encoder_factory = nullptr;
>-    if (jencoder_factory == nullptr) {
>-      legacy_video_encoder_factory = CreateLegacyVideoEncoderFactory();
>-      video_encoder_factory = std::unique_ptr<VideoEncoderFactory>(
>-          WrapLegacyVideoEncoderFactory(legacy_video_encoder_factory));
>-    } else {
>-      video_encoder_factory = std::unique_ptr<VideoEncoderFactory>(
>-          CreateVideoEncoderFactory(jni, jencoder_factory));
>-    }
>-
>-    std::unique_ptr<VideoDecoderFactory> video_decoder_factory = nullptr;
>-    if (jdecoder_factory == nullptr) {
>-      legacy_video_decoder_factory = CreateLegacyVideoDecoderFactory();
>-      video_decoder_factory = std::unique_ptr<VideoDecoderFactory>(
>-          WrapLegacyVideoDecoderFactory(legacy_video_decoder_factory));
>-    } else {
>-      video_decoder_factory = std::unique_ptr<VideoDecoderFactory>(
>-          CreateVideoDecoderFactory(jni, jdecoder_factory));
>-    }
>-
>-    rtc::scoped_refptr<AudioDeviceModule> adm_scoped = nullptr;
>-    media_engine.reset(CreateMediaEngine(
>-        adm_scoped, audio_encoder_factory, audio_decoder_factory,
>-        std::move(video_encoder_factory), std::move(video_decoder_factory),
>-        audio_mixer, audio_processor));
>-  }
>-
>-  rtc::scoped_refptr<PeerConnectionFactoryInterface> factory(
>-      CreateModularPeerConnectionFactory(
>-          network_thread.get(), worker_thread.get(), signaling_thread.get(),
>-          std::move(media_engine), std::move(call_factory),
>-          std::move(rtc_event_log_factory)));
>-  RTC_CHECK(factory) << "Failed to create the peer connection factory; "
>-                     << "WebRTC/libjingle init likely failed on this device";
>-  // TODO(honghaiz): Maybe put the options as the argument of
>-  // CreatePeerConnectionFactory.
>-  if (has_options) {
>-    factory->SetOptions(options);
>-  }
>-  OwnedFactoryAndThreads* owned_factory = new OwnedFactoryAndThreads(
>-      std::move(network_thread), std::move(worker_thread),
>-      std::move(signaling_thread), legacy_video_encoder_factory,
>-      legacy_video_decoder_factory, network_monitor_factory, factory.release());
>-  owned_factory->InvokeJavaCallbacksOnFactoryThreads();
>-  return jlongFromPointer(owned_factory);
>-}
>-
>-JNI_FUNCTION_DECLARATION(
>-    jlong,
>-    PeerConnectionFactory_nativeCreatePeerConnectionFactory,
>-    JNIEnv* jni,
>-    jclass,
>-    jobject joptions,
>-    jobject jencoder_factory,
>-    jobject jdecoder_factory) {
>-  return CreatePeerConnectionFactoryForJava(jni, joptions, jencoder_factory,
>-                                            jdecoder_factory,
>-                                            CreateAudioProcessing());
>-}
>-
>-JNI_FUNCTION_DECLARATION(
>-    jlong,
>-    PeerConnectionFactory_nativeCreatePeerConnectionFactoryWithAudioProcessing,
>-    JNIEnv* jni,
>-    jclass,
>-    jobject joptions,
>-    jobject jencoder_factory,
>-    jobject jdecoder_factory,
>-    jlong native_audio_processor) {
>-  rtc::scoped_refptr<AudioProcessing> audio_processor =
>-      reinterpret_cast<AudioProcessing*>(native_audio_processor);
>-  RTC_DCHECK(audio_processor);
>-  return CreatePeerConnectionFactoryForJava(jni, joptions, jencoder_factory,
>-                                            jdecoder_factory, audio_processor);
>-}
>-
>-JNI_FUNCTION_DECLARATION(void,
>-                         PeerConnectionFactory_nativeFreeFactory,
>-                         JNIEnv*,
>-                         jclass,
>-                         jlong j_p) {
>-  delete reinterpret_cast<OwnedFactoryAndThreads*>(j_p);
>-  if (field_trials_init_string) {
>-    field_trial::InitFieldTrialsFromString(NULL);
>-    delete field_trials_init_string;
>-    field_trials_init_string = NULL;
>-  }
>-}
>-
>-JNI_FUNCTION_DECLARATION(void,
>-                         PeerConnectionFactory_nativeThreadsCallbacks,
>-                         JNIEnv*,
>-                         jclass,
>-                         jlong j_p) {
>-  OwnedFactoryAndThreads* factory =
>-      reinterpret_cast<OwnedFactoryAndThreads*>(j_p);
>-  factory->InvokeJavaCallbacksOnFactoryThreads();
>-}
>-
>-JNI_FUNCTION_DECLARATION(jlong,
>-                         PeerConnectionFactory_nativeCreateLocalMediaStream,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jlong native_factory,
>-                         jstring label) {
>-  rtc::scoped_refptr<PeerConnectionFactoryInterface> factory(
>-      factoryFromJava(native_factory));
>-  rtc::scoped_refptr<MediaStreamInterface> stream(
>-      factory->CreateLocalMediaStream(JavaToStdString(jni, label)));
>-  return (jlong)stream.release();
>-}
>-
>-JNI_FUNCTION_DECLARATION(jlong,
>-                         PeerConnectionFactory_nativeCreateAudioSource,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jlong native_factory,
>-                         jobject j_constraints) {
>-  std::unique_ptr<MediaConstraintsInterface> constraints =
>-      JavaToNativeMediaConstraints(jni, j_constraints);
>-  rtc::scoped_refptr<PeerConnectionFactoryInterface> factory(
>-      factoryFromJava(native_factory));
>-  cricket::AudioOptions options;
>-  CopyConstraintsIntoAudioOptions(constraints.get(), &options);
>-  rtc::scoped_refptr<AudioSourceInterface> source(
>-      factory->CreateAudioSource(options));
>-  return (jlong)source.release();
>-}
>-
>-JNI_FUNCTION_DECLARATION(jlong,
>-                         PeerConnectionFactory_nativeCreateAudioTrack,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jlong native_factory,
>-                         jstring id,
>-                         jlong native_source) {
>-  rtc::scoped_refptr<PeerConnectionFactoryInterface> factory(
>-      factoryFromJava(native_factory));
>-  rtc::scoped_refptr<AudioTrackInterface> track(factory->CreateAudioTrack(
>-      JavaToStdString(jni, id),
>-      reinterpret_cast<AudioSourceInterface*>(native_source)));
>-  return (jlong)track.release();
>-}
>-
>-JNI_FUNCTION_DECLARATION(jboolean,
>-                         PeerConnectionFactory_nativeStartAecDump,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jlong native_factory,
>-                         jint file,
>-                         jint filesize_limit_bytes) {
>-  rtc::scoped_refptr<PeerConnectionFactoryInterface> factory(
>-      factoryFromJava(native_factory));
>-  return factory->StartAecDump(file, filesize_limit_bytes);
>-}
>-
>-JNI_FUNCTION_DECLARATION(void,
>-                         PeerConnectionFactory_nativeStopAecDump,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jlong native_factory) {
>-  rtc::scoped_refptr<PeerConnectionFactoryInterface> factory(
>-      factoryFromJava(native_factory));
>-  factory->StopAecDump();
>-}
>-
>-JNI_FUNCTION_DECLARATION(void,
>-                         PeerConnectionFactory_nativeSetOptions,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jlong native_factory,
>-                         jobject options) {
>-  rtc::scoped_refptr<PeerConnectionFactoryInterface> factory(
>-      factoryFromJava(native_factory));
>-  PeerConnectionFactoryInterface::Options options_to_set =
>-      JavaToNativePeerConnectionFactoryOptions(jni, options);
>-  factory->SetOptions(options_to_set);
>-
>-  if (options_to_set.disable_network_monitor) {
>-    OwnedFactoryAndThreads* owner =
>-        reinterpret_cast<OwnedFactoryAndThreads*>(native_factory);
>-    if (owner->network_monitor_factory()) {
>-      rtc::NetworkMonitorFactory::ReleaseFactory(
>-          owner->network_monitor_factory());
>-      owner->clear_network_monitor_factory();
>-    }
>-  }
>-}
>-
>-JNI_FUNCTION_DECLARATION(jlong,
>-                         PeerConnectionFactory_nativeCreatePeerConnection,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jlong factory,
>-                         jobject j_rtc_config,
>-                         jobject j_constraints,
>-                         jlong observer_p) {
>-  rtc::scoped_refptr<PeerConnectionFactoryInterface> f(
>-      reinterpret_cast<PeerConnectionFactoryInterface*>(
>-          factoryFromJava(factory)));
>-
>-  PeerConnectionInterface::RTCConfiguration rtc_config(
>-      PeerConnectionInterface::RTCConfigurationType::kAggressive);
>-  JavaToNativeRTCConfiguration(jni, j_rtc_config, &rtc_config);
>-
>-  jclass j_rtc_config_class = GetObjectClass(jni, j_rtc_config);
>-  jfieldID j_key_type_id = GetFieldID(jni, j_rtc_config_class, "keyType",
>-                                      "Lorg/webrtc/PeerConnection$KeyType;");
>-  jobject j_key_type = GetObjectField(jni, j_rtc_config, j_key_type_id);
>-
>-  // Generate non-default certificate.
>-  rtc::KeyType key_type = JavaToNativeKeyType(jni, j_key_type);
>-  if (key_type != rtc::KT_DEFAULT) {
>-    rtc::scoped_refptr<rtc::RTCCertificate> certificate =
>-        rtc::RTCCertificateGenerator::GenerateCertificate(
>-            rtc::KeyParams(key_type), rtc::Optional<uint64_t>());
>-    if (!certificate) {
>-      RTC_LOG(LS_ERROR) << "Failed to generate certificate. KeyType: "
>-                        << key_type;
>-      return 0;
>-    }
>-    rtc_config.certificates.push_back(certificate);
>-  }
>-
>-  PeerConnectionObserverJni* observer =
>-      reinterpret_cast<PeerConnectionObserverJni*>(observer_p);
>-  observer->SetConstraints(JavaToNativeMediaConstraints(jni, j_constraints));
>-  CopyConstraintsIntoRtcConfiguration(observer->constraints(), &rtc_config);
>-  rtc::scoped_refptr<PeerConnectionInterface> pc(
>-      f->CreatePeerConnection(rtc_config, nullptr, nullptr, observer));
>-  return (jlong)pc.release();
>-}
>-
>-}  // namespace jni
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/peerconnectionobserver_jni.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/peerconnectionobserver_jni.cc
>deleted file mode 100644
>index 4ad107eb91c..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/peerconnectionobserver_jni.cc
>+++ /dev/null
>@@ -1,320 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "sdk/android/src/jni/pc/peerconnectionobserver_jni.h"
>-
>-#include <pc/mediastreamobserver.h>
>-#include <string>
>-
>-#include "rtc_base/ptr_util.h"
>-#include "sdk/android/generated_peerconnection_jni/jni/MediaStream_jni.h"
>-#include "sdk/android/src/jni/classreferenceholder.h"
>-#include "sdk/android/src/jni/jni_helpers.h"
>-#include "sdk/android/src/jni/pc/datachannel.h"
>-#include "sdk/android/src/jni/pc/java_native_conversion.h"
>-
>-namespace webrtc {
>-namespace jni {
>-
>-// Convenience, used since callbacks occur on the signaling thread, which may
>-// be a non-Java thread.
>-static JNIEnv* jni() {
>-  return AttachCurrentThreadIfNeeded();
>-}
>-
>-PeerConnectionObserverJni::PeerConnectionObserverJni(JNIEnv* jni,
>-                                                     jobject j_observer)
>-    : j_observer_global_(jni, j_observer),
>-      j_observer_class_(jni, GetObjectClass(jni, *j_observer_global_)),
>-      j_rtp_receiver_class_(jni, FindClass(jni, "org/webrtc/RtpReceiver")),
>-      j_rtp_receiver_ctor_(
>-          GetMethodID(jni, *j_rtp_receiver_class_, "<init>", "(J)V")) {}
>-
>-PeerConnectionObserverJni::~PeerConnectionObserverJni() {
>-  ScopedLocalRefFrame local_ref_frame(jni());
>-  while (!remote_streams_.empty())
>-    DisposeRemoteStream(remote_streams_.begin());
>-  while (!rtp_receivers_.empty())
>-    DisposeRtpReceiver(rtp_receivers_.begin());
>-}
>-
>-void PeerConnectionObserverJni::OnIceCandidate(
>-    const IceCandidateInterface* candidate) {
>-  JNIEnv* env = AttachCurrentThreadIfNeeded();
>-  ScopedLocalRefFrame local_ref_frame(env);
>-  jobject j_candidate = NativeToJavaIceCandidate(env, *candidate);
>-
>-  jmethodID m = GetMethodID(env, *j_observer_class_, "onIceCandidate",
>-                            "(Lorg/webrtc/IceCandidate;)V");
>-  env->CallVoidMethod(*j_observer_global_, m, j_candidate);
>-  CHECK_EXCEPTION(env) << "error during CallVoidMethod";
>-}
>-
>-void PeerConnectionObserverJni::OnIceCandidatesRemoved(
>-    const std::vector<cricket::Candidate>& candidates) {
>-  ScopedLocalRefFrame local_ref_frame(jni());
>-  jobjectArray candidates_array = NativeToJavaCandidateArray(jni(), candidates);
>-  jmethodID m = GetMethodID(jni(), *j_observer_class_, "onIceCandidatesRemoved",
>-                            "([Lorg/webrtc/IceCandidate;)V");
>-  jni()->CallVoidMethod(*j_observer_global_, m, candidates_array);
>-  CHECK_EXCEPTION(jni()) << "Error during CallVoidMethod";
>-}
>-
>-void PeerConnectionObserverJni::OnSignalingChange(
>-    PeerConnectionInterface::SignalingState new_state) {
>-  ScopedLocalRefFrame local_ref_frame(jni());
>-  jmethodID m = GetMethodID(jni(), *j_observer_class_, "onSignalingChange",
>-                            "(Lorg/webrtc/PeerConnection$SignalingState;)V");
>-  jobject new_state_enum = JavaEnumFromIndexAndClassName(
>-      jni(), "PeerConnection$SignalingState", new_state);
>-  jni()->CallVoidMethod(*j_observer_global_, m, new_state_enum);
>-  CHECK_EXCEPTION(jni()) << "error during CallVoidMethod";
>-}
>-
>-void PeerConnectionObserverJni::OnIceConnectionChange(
>-    PeerConnectionInterface::IceConnectionState new_state) {
>-  ScopedLocalRefFrame local_ref_frame(jni());
>-  jmethodID m =
>-      GetMethodID(jni(), *j_observer_class_, "onIceConnectionChange",
>-                  "(Lorg/webrtc/PeerConnection$IceConnectionState;)V");
>-  jobject new_state_enum = JavaEnumFromIndexAndClassName(
>-      jni(), "PeerConnection$IceConnectionState", new_state);
>-  jni()->CallVoidMethod(*j_observer_global_, m, new_state_enum);
>-  CHECK_EXCEPTION(jni()) << "error during CallVoidMethod";
>-}
>-
>-void PeerConnectionObserverJni::OnIceConnectionReceivingChange(bool receiving) {
>-  ScopedLocalRefFrame local_ref_frame(jni());
>-  jmethodID m = GetMethodID(jni(), *j_observer_class_,
>-                            "onIceConnectionReceivingChange", "(Z)V");
>-  jni()->CallVoidMethod(*j_observer_global_, m, receiving);
>-  CHECK_EXCEPTION(jni()) << "error during CallVoidMethod";
>-}
>-
>-void PeerConnectionObserverJni::OnIceGatheringChange(
>-    PeerConnectionInterface::IceGatheringState new_state) {
>-  ScopedLocalRefFrame local_ref_frame(jni());
>-  jmethodID m = GetMethodID(jni(), *j_observer_class_, "onIceGatheringChange",
>-                            "(Lorg/webrtc/PeerConnection$IceGatheringState;)V");
>-  jobject new_state_enum = JavaEnumFromIndexAndClassName(
>-      jni(), "PeerConnection$IceGatheringState", new_state);
>-  jni()->CallVoidMethod(*j_observer_global_, m, new_state_enum);
>-  CHECK_EXCEPTION(jni()) << "error during CallVoidMethod";
>-}
>-
>-void PeerConnectionObserverJni::OnAddStream(
>-    rtc::scoped_refptr<MediaStreamInterface> stream) {
>-  ScopedLocalRefFrame local_ref_frame(jni());
>-  // The stream could be added into the remote_streams_ map when calling
>-  // OnAddTrack.
>-  jobject j_stream = GetOrCreateJavaStream(stream);
>-
>-  for (const auto& track : stream->GetAudioTracks()) {
>-    AddNativeAudioTrackToJavaStream(track, j_stream);
>-  }
>-  for (const auto& track : stream->GetVideoTracks()) {
>-    AddNativeVideoTrackToJavaStream(track, j_stream);
>-  }
>-
>-  jmethodID m = GetMethodID(jni(), *j_observer_class_, "onAddStream",
>-                            "(Lorg/webrtc/MediaStream;)V");
>-  jni()->CallVoidMethod(*j_observer_global_, m, j_stream);
>-  CHECK_EXCEPTION(jni()) << "error during CallVoidMethod";
>-
>-  // Create an observer to update the Java stream when the native stream's set
>-  // of tracks changes.
>-  auto observer = rtc::MakeUnique<MediaStreamObserver>(stream);
>-  observer->SignalAudioTrackRemoved.connect(
>-      this, &PeerConnectionObserverJni::OnAudioTrackRemovedFromStream);
>-  observer->SignalVideoTrackRemoved.connect(
>-      this, &PeerConnectionObserverJni::OnVideoTrackRemovedFromStream);
>-  observer->SignalAudioTrackAdded.connect(
>-      this, &PeerConnectionObserverJni::OnAudioTrackAddedToStream);
>-  observer->SignalVideoTrackAdded.connect(
>-      this, &PeerConnectionObserverJni::OnVideoTrackAddedToStream);
>-  stream_observers_.push_back(std::move(observer));
>-}
>-
>-void PeerConnectionObserverJni::AddNativeAudioTrackToJavaStream(
>-    rtc::scoped_refptr<AudioTrackInterface> track,
>-    jobject j_stream) {
>-  JNIEnv* env = AttachCurrentThreadIfNeeded();
>-  Java_MediaStream_addNativeAudioTrack(env, j_stream,
>-                                       jlongFromPointer(track.release()));
>-}
>-
>-void PeerConnectionObserverJni::AddNativeVideoTrackToJavaStream(
>-    rtc::scoped_refptr<VideoTrackInterface> track,
>-    jobject j_stream) {
>-  JNIEnv* env = AttachCurrentThreadIfNeeded();
>-  Java_MediaStream_addNativeVideoTrack(env, j_stream,
>-                                       jlongFromPointer(track.release()));
>-}
>-
>-void PeerConnectionObserverJni::OnAudioTrackAddedToStream(
>-    AudioTrackInterface* track,
>-    MediaStreamInterface* stream) {
>-  ScopedLocalRefFrame local_ref_frame(jni());
>-  jobject j_stream = GetOrCreateJavaStream(stream);
>-  AddNativeAudioTrackToJavaStream(track, j_stream);
>-}
>-
>-void PeerConnectionObserverJni::OnVideoTrackAddedToStream(
>-    VideoTrackInterface* track,
>-    MediaStreamInterface* stream) {
>-  ScopedLocalRefFrame local_ref_frame(jni());
>-  jobject j_stream = GetOrCreateJavaStream(stream);
>-  AddNativeVideoTrackToJavaStream(track, j_stream);
>-}
>-
>-void PeerConnectionObserverJni::OnAudioTrackRemovedFromStream(
>-    AudioTrackInterface* track,
>-    MediaStreamInterface* stream) {
>-  JNIEnv* env = AttachCurrentThreadIfNeeded();
>-  ScopedLocalRefFrame local_ref_frame(env);
>-  jobject j_stream = GetOrCreateJavaStream(stream);
>-  Java_MediaStream_removeAudioTrack(env, j_stream, jlongFromPointer(track));
>-}
>-
>-void PeerConnectionObserverJni::OnVideoTrackRemovedFromStream(
>-    VideoTrackInterface* track,
>-    MediaStreamInterface* stream) {
>-  JNIEnv* env = AttachCurrentThreadIfNeeded();
>-  ScopedLocalRefFrame local_ref_frame(env);
>-  jobject j_stream = GetOrCreateJavaStream(stream);
>-  Java_MediaStream_removeVideoTrack(env, j_stream, jlongFromPointer(track));
>-}
>-
>-void PeerConnectionObserverJni::OnRemoveStream(
>-    rtc::scoped_refptr<MediaStreamInterface> stream) {
>-  ScopedLocalRefFrame local_ref_frame(jni());
>-  NativeToJavaStreamsMap::iterator it = remote_streams_.find(stream);
>-  RTC_CHECK(it != remote_streams_.end())
>-      << "unexpected stream: " << std::hex << stream;
>-  jobject j_stream = it->second;
>-  jmethodID m = GetMethodID(jni(), *j_observer_class_, "onRemoveStream",
>-                            "(Lorg/webrtc/MediaStream;)V");
>-  jni()->CallVoidMethod(*j_observer_global_, m, j_stream);
>-  CHECK_EXCEPTION(jni()) << "error during CallVoidMethod";
>-
>-  // Release the refptr reference so that DisposeRemoteStream can assert
>-  // it removes the final reference.
>-  stream = nullptr;
>-  DisposeRemoteStream(it);
>-}
>-
>-void PeerConnectionObserverJni::OnDataChannel(
>-    rtc::scoped_refptr<DataChannelInterface> channel) {
>-  JNIEnv* env = AttachCurrentThreadIfNeeded();
>-  ScopedLocalRefFrame local_ref_frame(env);
>-  jobject j_channel = WrapNativeDataChannel(env, channel);
>-  jmethodID m = GetMethodID(env, *j_observer_class_, "onDataChannel",
>-                            "(Lorg/webrtc/DataChannel;)V");
>-  env->CallVoidMethod(*j_observer_global_, m, j_channel);
>-  CHECK_EXCEPTION(env) << "error during CallVoidMethod";
>-}
>-
>-void PeerConnectionObserverJni::OnRenegotiationNeeded() {
>-  ScopedLocalRefFrame local_ref_frame(jni());
>-  jmethodID m =
>-      GetMethodID(jni(), *j_observer_class_, "onRenegotiationNeeded", "()V");
>-  jni()->CallVoidMethod(*j_observer_global_, m);
>-  CHECK_EXCEPTION(jni()) << "error during CallVoidMethod";
>-}
>-
>-void PeerConnectionObserverJni::OnAddTrack(
>-    rtc::scoped_refptr<RtpReceiverInterface> receiver,
>-    const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams) {
>-  ScopedLocalRefFrame local_ref_frame(jni());
>-  jobject j_rtp_receiver =
>-      jni()->NewObject(*j_rtp_receiver_class_, j_rtp_receiver_ctor_,
>-                       jlongFromPointer(receiver.get()));
>-  CHECK_EXCEPTION(jni()) << "error during NewObject";
>-  receiver->AddRef();
>-  rtp_receivers_[receiver] = NewGlobalRef(jni(), j_rtp_receiver);
>-
>-  jobjectArray j_stream_array = NativeToJavaMediaStreamArray(jni(), streams);
>-  jmethodID m =
>-      GetMethodID(jni(), *j_observer_class_, "onAddTrack",
>-                  "(Lorg/webrtc/RtpReceiver;[Lorg/webrtc/MediaStream;)V");
>-  jni()->CallVoidMethod(*j_observer_global_, m, j_rtp_receiver, j_stream_array);
>-  CHECK_EXCEPTION(jni()) << "Error during CallVoidMethod";
>-}
>-
>-void PeerConnectionObserverJni::SetConstraints(
>-    std::unique_ptr<MediaConstraintsInterface> constraints) {
>-  RTC_CHECK(!constraints_.get()) << "constraints already set!";
>-  constraints_ = std::move(constraints);
>-}
>-
>-void PeerConnectionObserverJni::DisposeRemoteStream(
>-    const NativeToJavaStreamsMap::iterator& it) {
>-  MediaStreamInterface* stream = it->first;
>-  jobject j_stream = it->second;
>-
>-  // Remove the observer first, so it doesn't react to events during deletion.
>-  stream_observers_.erase(
>-      std::remove_if(
>-          stream_observers_.begin(), stream_observers_.end(),
>-          [stream](const std::unique_ptr<MediaStreamObserver>& observer) {
>-            return observer->stream() == stream;
>-          }),
>-      stream_observers_.end());
>-
>-  remote_streams_.erase(it);
>-  JNIEnv* env = AttachCurrentThreadIfNeeded();
>-  Java_MediaStream_dispose(env, j_stream);
>-  DeleteGlobalRef(env, j_stream);
>-}
>-
>-void PeerConnectionObserverJni::DisposeRtpReceiver(
>-    const NativeToJavaRtpReceiverMap::iterator& it) {
>-  jobject j_rtp_receiver = it->second;
>-  rtp_receivers_.erase(it);
>-  jni()->CallVoidMethod(
>-      j_rtp_receiver,
>-      GetMethodID(jni(), *j_rtp_receiver_class_, "dispose", "()V"));
>-  CHECK_EXCEPTION(jni()) << "error during RtpReceiver.dispose()";
>-  DeleteGlobalRef(jni(), j_rtp_receiver);
>-}
>-
>-// If the NativeToJavaStreamsMap contains the stream, return it.
>-// Otherwise, create a new Java MediaStream.
>-jobject PeerConnectionObserverJni::GetOrCreateJavaStream(
>-    const rtc::scoped_refptr<MediaStreamInterface>& stream) {
>-  NativeToJavaStreamsMap::iterator it = remote_streams_.find(stream);
>-  if (it != remote_streams_.end()) {
>-    return it->second;
>-  }
>-  // Java MediaStream holds one reference. Corresponding Release() is in
>-  // MediaStream_free, triggered by MediaStream.dispose().
>-  stream->AddRef();
>-  jobject j_stream =
>-      Java_MediaStream_Constructor(jni(), jlongFromPointer(stream.get()));
>-
>-  remote_streams_[stream] = NewGlobalRef(jni(), j_stream);
>-  return j_stream;
>-}
>-
>-jobjectArray PeerConnectionObserverJni::NativeToJavaMediaStreamArray(
>-    JNIEnv* jni,
>-    const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams) {
>-  jobjectArray java_streams = jni->NewObjectArray(
>-      streams.size(), org_webrtc_MediaStream_clazz(jni), nullptr);
>-  CHECK_EXCEPTION(jni) << "error during NewObjectArray";
>-  for (size_t i = 0; i < streams.size(); ++i) {
>-    jobject j_stream = GetOrCreateJavaStream(streams[i]);
>-    jni->SetObjectArrayElement(java_streams, i, j_stream);
>-  }
>-  return java_streams;
>-}
>-
>-}  // namespace jni
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/peerconnectionobserver_jni.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/peerconnectionobserver_jni.h
>deleted file mode 100644
>index 6f95a385679..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/peerconnectionobserver_jni.h
>+++ /dev/null
>@@ -1,118 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef SDK_ANDROID_SRC_JNI_PC_PEERCONNECTIONOBSERVER_JNI_H_
>-#define SDK_ANDROID_SRC_JNI_PC_PEERCONNECTIONOBSERVER_JNI_H_
>-
>-#include <pc/mediastreamobserver.h>
>-#include <map>
>-#include <memory>
>-#include <vector>
>-
>-#include "api/peerconnectioninterface.h"
>-#include "sdk/android/src/jni/jni_helpers.h"
>-#include "sdk/android/src/jni/pc/mediaconstraints_jni.h"
>-
>-namespace webrtc {
>-namespace jni {
>-
>-// Adapter between the C++ PeerConnectionObserver interface and the Java
>-// PeerConnection.Observer interface.  Wraps an instance of the Java interface
>-// and dispatches C++ callbacks to Java.
>-class PeerConnectionObserverJni : public PeerConnectionObserver,
>-                                  public sigslot::has_slots<> {
>- public:
>-  PeerConnectionObserverJni(JNIEnv* jni, jobject j_observer);
>-  virtual ~PeerConnectionObserverJni();
>-
>-  // Implementation of PeerConnectionObserver interface, which propagates
>-  // the callbacks to the Java observer.
>-  void OnIceCandidate(const IceCandidateInterface* candidate) override;
>-  void OnIceCandidatesRemoved(
>-      const std::vector<cricket::Candidate>& candidates) override;
>-  void OnSignalingChange(
>-      PeerConnectionInterface::SignalingState new_state) override;
>-  void OnIceConnectionChange(
>-      PeerConnectionInterface::IceConnectionState new_state) override;
>-  void OnIceConnectionReceivingChange(bool receiving) override;
>-  void OnIceGatheringChange(
>-      PeerConnectionInterface::IceGatheringState new_state) override;
>-  void OnAddStream(rtc::scoped_refptr<MediaStreamInterface> stream) override;
>-  void OnRemoveStream(rtc::scoped_refptr<MediaStreamInterface> stream) override;
>-  void OnDataChannel(rtc::scoped_refptr<DataChannelInterface> channel) override;
>-  void OnRenegotiationNeeded() override;
>-  void OnAddTrack(rtc::scoped_refptr<RtpReceiverInterface> receiver,
>-                  const std::vector<rtc::scoped_refptr<MediaStreamInterface>>&
>-                      streams) override;
>-
>-  void SetConstraints(std::unique_ptr<MediaConstraintsInterface> constraints);
>-  const MediaConstraintsInterface* constraints() { return constraints_.get(); }
>-
>- private:
>-  typedef std::map<MediaStreamInterface*, jobject> NativeToJavaStreamsMap;
>-  typedef std::map<RtpReceiverInterface*, jobject> NativeToJavaRtpReceiverMap;
>-  typedef std::map<MediaStreamTrackInterface*, jobject>
>-      NativeToJavaMediaTrackMap;
>-  typedef std::map<MediaStreamTrackInterface*, RtpReceiverInterface*>
>-      NativeMediaStreamTrackToNativeRtpReceiver;
>-
>-  void DisposeRemoteStream(const NativeToJavaStreamsMap::iterator& it);
>-  void DisposeRtpReceiver(const NativeToJavaRtpReceiverMap::iterator& it);
>-
>-  // If the NativeToJavaStreamsMap contains the stream, return it.
>-  // Otherwise, create a new Java MediaStream.
>-  jobject GetOrCreateJavaStream(
>-      const rtc::scoped_refptr<MediaStreamInterface>& stream);
>-
>-  // Converts array of streams, creating or re-using Java streams as necessary.
>-  jobjectArray NativeToJavaMediaStreamArray(
>-      JNIEnv* jni,
>-      const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams);
>-
>-  // The three methods below must be called from within a local ref
>-  // frame (e.g., using ScopedLocalRefFrame), otherwise they will
>-  // leak references.
>-  //
>-  // Create a Java track object to wrap |track|, and add it to |j_stream|.
>-  void AddNativeAudioTrackToJavaStream(
>-      rtc::scoped_refptr<AudioTrackInterface> track,
>-      jobject j_stream);
>-  void AddNativeVideoTrackToJavaStream(
>-      rtc::scoped_refptr<VideoTrackInterface> track,
>-      jobject j_stream);
>-
>-  // Callbacks invoked when a native stream changes, and the Java stream needs
>-  // to be updated; MediaStreamObserver is used to make this simpler.
>-  void OnAudioTrackAddedToStream(AudioTrackInterface* track,
>-                                 MediaStreamInterface* stream);
>-  void OnVideoTrackAddedToStream(VideoTrackInterface* track,
>-                                 MediaStreamInterface* stream);
>-  void OnAudioTrackRemovedFromStream(AudioTrackInterface* track,
>-                                     MediaStreamInterface* stream);
>-  void OnVideoTrackRemovedFromStream(VideoTrackInterface* track,
>-                                     MediaStreamInterface* stream);
>-
>-  const ScopedGlobalRef<jobject> j_observer_global_;
>-  const ScopedGlobalRef<jclass> j_observer_class_;
>-  const ScopedGlobalRef<jclass> j_rtp_receiver_class_;
>-  const jmethodID j_rtp_receiver_ctor_;
>-
>-  // C++ -> Java remote streams. The stored jobects are global refs and must be
>-  // manually deleted upon removal. Use DisposeRemoteStream().
>-  NativeToJavaStreamsMap remote_streams_;
>-  NativeToJavaRtpReceiverMap rtp_receivers_;
>-  std::unique_ptr<MediaConstraintsInterface> constraints_;
>-  std::vector<std::unique_ptr<webrtc::MediaStreamObserver>> stream_observers_;
>-};
>-
>-}  // namespace jni
>-}  // namespace webrtc
>-
>-#endif  // SDK_ANDROID_SRC_JNI_PC_PEERCONNECTIONOBSERVER_JNI_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/rtcstatscollectorcallbackwrapper.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/rtcstatscollectorcallbackwrapper.cc
>index f4d0b81a347..966d2efb75c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/rtcstatscollectorcallbackwrapper.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/rtcstatscollectorcallbackwrapper.cc
>@@ -18,26 +18,28 @@
> #include "sdk/android/generated_peerconnection_jni/jni/RTCStatsCollectorCallback_jni.h"
> #include "sdk/android/generated_peerconnection_jni/jni/RTCStatsReport_jni.h"
> #include "sdk/android/generated_peerconnection_jni/jni/RTCStats_jni.h"
>-#include "sdk/android/src/jni/classreferenceholder.h"
>+#include "sdk/android/native_api/jni/java_types.h"
> 
> namespace webrtc {
> namespace jni {
> 
> namespace {
> 
>-jobject NativeToJavaBigInteger(JNIEnv* env, uint64_t u) {
>+ScopedJavaLocalRef<jobject> NativeToJavaBigInteger(JNIEnv* env, uint64_t u) {
>   return JNI_BigInteger::Java_BigInteger_ConstructorJMBI_JLS(
>       env, NativeToJavaString(env, rtc::ToString(u)));
> }
> 
>-jobjectArray NativeToJavaBigIntegerArray(
>+ScopedJavaLocalRef<jobjectArray> NativeToJavaBigIntegerArray(
>     JNIEnv* env,
>     const std::vector<uint64_t>& container) {
>   return NativeToJavaObjectArray(
>       env, container, java_math_BigInteger_clazz(env), &NativeToJavaBigInteger);
> }
> 
>-jobject MemberToJava(JNIEnv* env, const RTCStatsMemberInterface& member) {
>+ScopedJavaLocalRef<jobject> MemberToJava(
>+    JNIEnv* env,
>+    const RTCStatsMemberInterface& member) {
>   switch (member.type()) {
>     case RTCStatsMemberInterface::kBool:
>       return NativeToJavaBoolean(env, *member.cast_to<RTCStatsMember<bool>>());
>@@ -97,71 +99,45 @@ jobject MemberToJava(JNIEnv* env, const RTCStatsMemberInterface& member) {
>   return nullptr;
> }
> 
>+ScopedJavaLocalRef<jobject> NativeToJavaRtcStats(JNIEnv* env,
>+                                                 const RTCStats& stats) {
>+  JavaMapBuilder builder(env);
>+  for (auto* const member : stats.Members()) {
>+    if (!member->is_defined())
>+      continue;
>+    builder.put(NativeToJavaString(env, member->name()),
>+                MemberToJava(env, *member));
>+  }
>+  return Java_RTCStats_create(
>+      env, stats.timestamp_us(), NativeToJavaString(env, stats.type()),
>+      NativeToJavaString(env, stats.id()), builder.GetJavaMap());
>+}
>+
>+ScopedJavaLocalRef<jobject> NativeToJavaRtcStatsReport(
>+    JNIEnv* env,
>+    const rtc::scoped_refptr<const RTCStatsReport>& report) {
>+  ScopedJavaLocalRef<jobject> j_stats_map =
>+      NativeToJavaMap(env, *report, [](JNIEnv* env, const RTCStats& stats) {
>+        return std::make_pair(NativeToJavaString(env, stats.id()),
>+                              NativeToJavaRtcStats(env, stats));
>+      });
>+  return Java_RTCStatsReport_create(env, report->timestamp_us(), j_stats_map);
>+}
>+
> }  // namespace
> 
> RTCStatsCollectorCallbackWrapper::RTCStatsCollectorCallbackWrapper(
>     JNIEnv* jni,
>-    jobject j_callback)
>-    : j_callback_global_(jni, j_callback),
>-      j_linked_hash_map_class_(FindClass(jni, "java/util/LinkedHashMap")),
>-      j_linked_hash_map_ctor_(
>-          GetMethodID(jni, j_linked_hash_map_class_, "<init>", "()V")),
>-      j_linked_hash_map_put_(GetMethodID(
>-          jni,
>-          j_linked_hash_map_class_,
>-          "put",
>-          "(Ljava/lang/Object;Ljava/lang/Object;)Ljava/lang/Object;")) {}
>+    const JavaRef<jobject>& j_callback)
>+    : j_callback_global_(jni, j_callback) {}
>+
>+RTCStatsCollectorCallbackWrapper::~RTCStatsCollectorCallbackWrapper() = default;
> 
> void RTCStatsCollectorCallbackWrapper::OnStatsDelivered(
>     const rtc::scoped_refptr<const RTCStatsReport>& report) {
>   JNIEnv* jni = AttachCurrentThreadIfNeeded();
>-  ScopedLocalRefFrame local_ref_frame(jni);
>-  jobject j_report = ReportToJava(jni, report);
>-  Java_RTCStatsCollectorCallback_onStatsDelivered(jni, *j_callback_global_,
>-                                                  j_report);
>-}
>-
>-jobject RTCStatsCollectorCallbackWrapper::ReportToJava(
>-    JNIEnv* jni,
>-    const rtc::scoped_refptr<const RTCStatsReport>& report) {
>-  jobject j_stats_map =
>-      jni->NewObject(j_linked_hash_map_class_, j_linked_hash_map_ctor_);
>-  CHECK_EXCEPTION(jni) << "error during NewObject";
>-  for (const RTCStats& stats : *report) {
>-    // Create a local reference frame for each RTCStats, since there is a
>-    // maximum number of references that can be created in one frame.
>-    ScopedLocalRefFrame local_ref_frame(jni);
>-    jstring j_id = NativeToJavaString(jni, stats.id());
>-    jobject j_stats = StatsToJava(jni, stats);
>-    jni->CallObjectMethod(j_stats_map, j_linked_hash_map_put_, j_id, j_stats);
>-    CHECK_EXCEPTION(jni) << "error during CallObjectMethod";
>-  }
>-  jobject j_report =
>-      Java_RTCStatsReport_create(jni, report->timestamp_us(), j_stats_map);
>-  return j_report;
>-}
>-
>-jobject RTCStatsCollectorCallbackWrapper::StatsToJava(JNIEnv* jni,
>-                                                      const RTCStats& stats) {
>-  jstring j_type = NativeToJavaString(jni, stats.type());
>-  jstring j_id = NativeToJavaString(jni, stats.id());
>-  jobject j_members =
>-      jni->NewObject(j_linked_hash_map_class_, j_linked_hash_map_ctor_);
>-  for (const RTCStatsMemberInterface* member : stats.Members()) {
>-    if (!member->is_defined()) {
>-      continue;
>-    }
>-    // Create a local reference frame for each member as well.
>-    ScopedLocalRefFrame local_ref_frame(jni);
>-    jstring j_name = NativeToJavaString(jni, member->name());
>-    jobject j_member = MemberToJava(jni, *member);
>-    jni->CallObjectMethod(j_members, j_linked_hash_map_put_, j_name, j_member);
>-    CHECK_EXCEPTION(jni) << "error during CallObjectMethod";
>-  }
>-  jobject j_stats =
>-      Java_RTCStats_create(jni, stats.timestamp_us(), j_type, j_id, j_members);
>-  CHECK_EXCEPTION(jni) << "error during NewObject";
>-  return j_stats;
>+  Java_RTCStatsCollectorCallback_onStatsDelivered(
>+      jni, j_callback_global_, NativeToJavaRtcStatsReport(jni, report));
> }
> 
> }  // namespace jni
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/rtcstatscollectorcallbackwrapper.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/rtcstatscollectorcallbackwrapper.h
>index d9cc6923d1b..a2794ff2f36 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/rtcstatscollectorcallbackwrapper.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/rtcstatscollectorcallbackwrapper.h
>@@ -24,21 +24,15 @@ namespace jni {
> // Java.
> class RTCStatsCollectorCallbackWrapper : public RTCStatsCollectorCallback {
>  public:
>-  RTCStatsCollectorCallbackWrapper(JNIEnv* jni, jobject j_callback);
>+  RTCStatsCollectorCallbackWrapper(JNIEnv* jni,
>+                                   const JavaRef<jobject>& j_callback);
>+  ~RTCStatsCollectorCallbackWrapper() override;
> 
>   void OnStatsDelivered(
>       const rtc::scoped_refptr<const RTCStatsReport>& report) override;
> 
>  private:
>-  // Helper functions for converting C++ RTCStatsReport to Java equivalent.
>-  jobject ReportToJava(JNIEnv* jni,
>-                       const rtc::scoped_refptr<const RTCStatsReport>& report);
>-  jobject StatsToJava(JNIEnv* jni, const RTCStats& stats);
>-
>-  const ScopedGlobalRef<jobject> j_callback_global_;
>-  const jclass j_linked_hash_map_class_;
>-  const jmethodID j_linked_hash_map_ctor_;
>-  const jmethodID j_linked_hash_map_put_;
>+  const ScopedJavaGlobalRef<jobject> j_callback_global_;
> };
> 
> }  // namespace jni
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/rtpparameters.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/rtpparameters.cc
>new file mode 100644
>index 00000000000..d547d36f052
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/rtpparameters.cc
>@@ -0,0 +1,150 @@
>+/*
>+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "sdk/android/src/jni/pc/rtpparameters.h"
>+
>+#include "sdk/android/generated_peerconnection_jni/jni/RtpParameters_jni.h"
>+#include "sdk/android/native_api/jni/java_types.h"
>+#include "sdk/android/src/jni/jni_helpers.h"
>+#include "sdk/android/src/jni/pc/mediastreamtrack.h"
>+
>+namespace webrtc {
>+namespace jni {
>+
>+namespace {
>+
>+ScopedJavaLocalRef<jobject> NativeToJavaRtpEncodingParameter(
>+    JNIEnv* env,
>+    const RtpEncodingParameters& encoding) {
>+  return Java_Encoding_Constructor(
>+      env, encoding.active, NativeToJavaInteger(env, encoding.max_bitrate_bps),
>+      NativeToJavaInteger(env, encoding.min_bitrate_bps),
>+      encoding.ssrc ? NativeToJavaLong(env, *encoding.ssrc) : nullptr);
>+}
>+
>+ScopedJavaLocalRef<jobject> NativeToJavaRtpCodecParameter(
>+    JNIEnv* env,
>+    const RtpCodecParameters& codec) {
>+  return Java_Codec_Constructor(env, codec.payload_type,
>+                                NativeToJavaString(env, codec.name),
>+                                NativeToJavaMediaType(env, codec.kind),
>+                                NativeToJavaInteger(env, codec.clock_rate),
>+                                NativeToJavaInteger(env, codec.num_channels),
>+                                NativeToJavaStringMap(env, codec.parameters));
>+}
>+
>+ScopedJavaLocalRef<jobject> NativeToJavaRtpRtcpParameters(
>+    JNIEnv* env,
>+    const RtcpParameters& rtcp) {
>+  return Java_Rtcp_Constructor(env, NativeToJavaString(env, rtcp.cname),
>+                               rtcp.reduced_size);
>+}
>+
>+ScopedJavaLocalRef<jobject> NativeToJavaRtpHeaderExtensionParameter(
>+    JNIEnv* env,
>+    const RtpExtension& extension) {
>+  return Java_HeaderExtension_Constructor(
>+      env, NativeToJavaString(env, extension.uri), extension.id,
>+      extension.encrypt);
>+}
>+
>+}  // namespace
>+
>+RtpEncodingParameters JavaToNativeRtpEncodingParameters(
>+    JNIEnv* jni,
>+    const JavaRef<jobject>& j_encoding_parameters) {
>+  RtpEncodingParameters encoding;
>+  encoding.active = Java_Encoding_getActive(jni, j_encoding_parameters);
>+  ScopedJavaLocalRef<jobject> j_max_bitrate =
>+      Java_Encoding_getMaxBitrateBps(jni, j_encoding_parameters);
>+  encoding.max_bitrate_bps = JavaToNativeOptionalInt(jni, j_max_bitrate);
>+  ScopedJavaLocalRef<jobject> j_min_bitrate =
>+      Java_Encoding_getMinBitrateBps(jni, j_encoding_parameters);
>+  encoding.min_bitrate_bps = JavaToNativeOptionalInt(jni, j_min_bitrate);
>+  ScopedJavaLocalRef<jobject> j_ssrc =
>+      Java_Encoding_getSsrc(jni, j_encoding_parameters);
>+  if (!IsNull(jni, j_ssrc))
>+    encoding.ssrc = JavaToNativeLong(jni, j_ssrc);
>+  return encoding;
>+}
>+
>+RtpParameters JavaToNativeRtpParameters(JNIEnv* jni,
>+                                        const JavaRef<jobject>& j_parameters) {
>+  RtpParameters parameters;
>+
>+  ScopedJavaLocalRef<jstring> j_transaction_id =
>+      Java_RtpParameters_getTransactionId(jni, j_parameters);
>+  parameters.transaction_id = JavaToNativeString(jni, j_transaction_id);
>+
>+  ScopedJavaLocalRef<jobject> j_rtcp =
>+      Java_RtpParameters_getRtcp(jni, j_parameters);
>+  ScopedJavaLocalRef<jstring> j_rtcp_cname = Java_Rtcp_getCname(jni, j_rtcp);
>+  jboolean j_rtcp_reduced_size = Java_Rtcp_getReducedSize(jni, j_rtcp);
>+  parameters.rtcp.cname = JavaToNativeString(jni, j_rtcp_cname);
>+  parameters.rtcp.reduced_size = j_rtcp_reduced_size;
>+
>+  ScopedJavaLocalRef<jobject> j_header_extensions =
>+      Java_RtpParameters_getHeaderExtensions(jni, j_parameters);
>+  for (const JavaRef<jobject>& j_header_extension :
>+       Iterable(jni, j_header_extensions)) {
>+    RtpExtension header_extension;
>+    header_extension.uri = JavaToStdString(
>+        jni, Java_HeaderExtension_getUri(jni, j_header_extension));
>+    header_extension.id = Java_HeaderExtension_getId(jni, j_header_extension);
>+    header_extension.encrypt =
>+        Java_HeaderExtension_getEncrypted(jni, j_header_extension);
>+    parameters.header_extensions.push_back(header_extension);
>+  }
>+
>+  // Convert encodings.
>+  ScopedJavaLocalRef<jobject> j_encodings =
>+      Java_RtpParameters_getEncodings(jni, j_parameters);
>+  for (const JavaRef<jobject>& j_encoding_parameters :
>+       Iterable(jni, j_encodings)) {
>+    RtpEncodingParameters encoding =
>+        JavaToNativeRtpEncodingParameters(jni, j_encoding_parameters);
>+    parameters.encodings.push_back(encoding);
>+  }
>+
>+  // Convert codecs.
>+  ScopedJavaLocalRef<jobject> j_codecs =
>+      Java_RtpParameters_getCodecs(jni, j_parameters);
>+  for (const JavaRef<jobject>& j_codec : Iterable(jni, j_codecs)) {
>+    RtpCodecParameters codec;
>+    codec.payload_type = Java_Codec_getPayloadType(jni, j_codec);
>+    codec.name = JavaToStdString(jni, Java_Codec_getName(jni, j_codec));
>+    codec.kind = JavaToNativeMediaType(jni, Java_Codec_getKind(jni, j_codec));
>+    codec.clock_rate =
>+        JavaToNativeOptionalInt(jni, Java_Codec_getClockRate(jni, j_codec));
>+    codec.num_channels =
>+        JavaToNativeOptionalInt(jni, Java_Codec_getNumChannels(jni, j_codec));
>+    auto parameters_map =
>+        JavaToNativeStringMap(jni, Java_Codec_getParameters(jni, j_codec));
>+    codec.parameters.insert(parameters_map.begin(), parameters_map.end());
>+    parameters.codecs.push_back(codec);
>+  }
>+  return parameters;
>+}
>+
>+ScopedJavaLocalRef<jobject> NativeToJavaRtpParameters(
>+    JNIEnv* env,
>+    const RtpParameters& parameters) {
>+  return Java_RtpParameters_Constructor(
>+      env, NativeToJavaString(env, parameters.transaction_id),
>+      NativeToJavaRtpRtcpParameters(env, parameters.rtcp),
>+      NativeToJavaList(env, parameters.header_extensions,
>+                       &NativeToJavaRtpHeaderExtensionParameter),
>+      NativeToJavaList(env, parameters.encodings,
>+                       &NativeToJavaRtpEncodingParameter),
>+      NativeToJavaList(env, parameters.codecs, &NativeToJavaRtpCodecParameter));
>+}
>+
>+}  // namespace jni
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/rtpparameters.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/rtpparameters.h
>new file mode 100644
>index 00000000000..ae79793b1af
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/rtpparameters.h
>@@ -0,0 +1,35 @@
>+/*
>+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef SDK_ANDROID_SRC_JNI_PC_RTPPARAMETERS_H_
>+#define SDK_ANDROID_SRC_JNI_PC_RTPPARAMETERS_H_
>+
>+#include <jni.h>
>+
>+#include "api/rtpparameters.h"
>+#include "sdk/android/native_api/jni/scoped_java_ref.h"
>+
>+namespace webrtc {
>+namespace jni {
>+
>+RtpEncodingParameters JavaToNativeRtpEncodingParameters(
>+    JNIEnv* env,
>+    const JavaRef<jobject>& j_encoding_parameters);
>+
>+RtpParameters JavaToNativeRtpParameters(JNIEnv* jni,
>+                                        const JavaRef<jobject>& j_parameters);
>+ScopedJavaLocalRef<jobject> NativeToJavaRtpParameters(
>+    JNIEnv* jni,
>+    const RtpParameters& parameters);
>+
>+}  // namespace jni
>+}  // namespace webrtc
>+
>+#endif  // SDK_ANDROID_SRC_JNI_PC_RTPPARAMETERS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/rtpreceiver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/rtpreceiver.cc
>new file mode 100644
>index 00000000000..049df49e7d2
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/rtpreceiver.cc
>@@ -0,0 +1,133 @@
>+/*
>+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "sdk/android/src/jni/pc/rtpreceiver.h"
>+
>+#include "sdk/android/generated_peerconnection_jni/jni/RtpReceiver_jni.h"
>+#include "sdk/android/native_api/jni/java_types.h"
>+#include "sdk/android/src/jni/jni_helpers.h"
>+#include "sdk/android/src/jni/pc/mediastreamtrack.h"
>+#include "sdk/android/src/jni/pc/rtpparameters.h"
>+
>+namespace webrtc {
>+namespace jni {
>+
>+namespace {
>+
>+// Adapter between the C++ RtpReceiverObserverInterface and the Java
>+// RtpReceiver.Observer interface. Wraps an instance of the Java interface and
>+// dispatches C++ callbacks to Java.
>+class RtpReceiverObserverJni : public RtpReceiverObserverInterface {
>+ public:
>+  RtpReceiverObserverJni(JNIEnv* env, const JavaRef<jobject>& j_observer)
>+      : j_observer_global_(env, j_observer) {}
>+
>+  ~RtpReceiverObserverJni() override = default;
>+
>+  void OnFirstPacketReceived(cricket::MediaType media_type) override {
>+    JNIEnv* const env = AttachCurrentThreadIfNeeded();
>+    Java_Observer_onFirstPacketReceived(env, j_observer_global_,
>+                                        NativeToJavaMediaType(env, media_type));
>+  }
>+
>+ private:
>+  const ScopedJavaGlobalRef<jobject> j_observer_global_;
>+};
>+
>+}  // namespace
>+
>+ScopedJavaLocalRef<jobject> NativeToJavaRtpReceiver(
>+    JNIEnv* env,
>+    rtc::scoped_refptr<RtpReceiverInterface> receiver) {
>+  // Receiver is now owned by Java object, and will be freed from there.
>+  return Java_RtpReceiver_Constructor(env,
>+                                      jlongFromPointer(receiver.release()));
>+}
>+
>+JavaRtpReceiverGlobalOwner::JavaRtpReceiverGlobalOwner(
>+    JNIEnv* env,
>+    const JavaRef<jobject>& j_receiver)
>+    : j_receiver_(env, j_receiver) {}
>+
>+JavaRtpReceiverGlobalOwner::JavaRtpReceiverGlobalOwner(
>+    JavaRtpReceiverGlobalOwner&& other) = default;
>+
>+JavaRtpReceiverGlobalOwner::~JavaRtpReceiverGlobalOwner() {
>+  if (j_receiver_.obj())
>+    Java_RtpReceiver_dispose(AttachCurrentThreadIfNeeded(), j_receiver_);
>+}
>+
>+static jlong JNI_RtpReceiver_GetTrack(JNIEnv* jni,
>+                                      const JavaParamRef<jclass>&,
>+                                      jlong j_rtp_receiver_pointer) {
>+  // MediaStreamTrack will have shared ownership by the MediaStreamTrack Java
>+  // object.
>+  return jlongFromPointer(
>+      reinterpret_cast<RtpReceiverInterface*>(j_rtp_receiver_pointer)
>+          ->track()
>+          .release());
>+}
>+
>+static jboolean JNI_RtpReceiver_SetParameters(
>+    JNIEnv* jni,
>+    const JavaParamRef<jclass>&,
>+    jlong j_rtp_receiver_pointer,
>+    const JavaParamRef<jobject>& j_parameters) {
>+  RtpParameters parameters = JavaToNativeRtpParameters(jni, j_parameters);
>+  return reinterpret_cast<RtpReceiverInterface*>(j_rtp_receiver_pointer)
>+      ->SetParameters(parameters);
>+}
>+
>+static ScopedJavaLocalRef<jobject> JNI_RtpReceiver_GetParameters(
>+    JNIEnv* jni,
>+    const JavaParamRef<jclass>&,
>+    jlong j_rtp_receiver_pointer) {
>+  RtpParameters parameters =
>+      reinterpret_cast<RtpReceiverInterface*>(j_rtp_receiver_pointer)
>+          ->GetParameters();
>+  return NativeToJavaRtpParameters(jni, parameters);
>+}
>+
>+static ScopedJavaLocalRef<jstring> JNI_RtpReceiver_GetId(
>+    JNIEnv* jni,
>+    const JavaParamRef<jclass>&,
>+    jlong j_rtp_receiver_pointer) {
>+  return NativeToJavaString(
>+      jni,
>+      reinterpret_cast<RtpReceiverInterface*>(j_rtp_receiver_pointer)->id());
>+}
>+
>+static jlong JNI_RtpReceiver_SetObserver(
>+    JNIEnv* jni,
>+    const JavaParamRef<jclass>&,
>+    jlong j_rtp_receiver_pointer,
>+    const JavaParamRef<jobject>& j_observer) {
>+  RtpReceiverObserverJni* rtpReceiverObserver =
>+      new RtpReceiverObserverJni(jni, j_observer);
>+  reinterpret_cast<RtpReceiverInterface*>(j_rtp_receiver_pointer)
>+      ->SetObserver(rtpReceiverObserver);
>+  return jlongFromPointer(rtpReceiverObserver);
>+}
>+
>+static void JNI_RtpReceiver_UnsetObserver(JNIEnv* jni,
>+                                          const JavaParamRef<jclass>&,
>+                                          jlong j_rtp_receiver_pointer,
>+                                          jlong j_observer_pointer) {
>+  reinterpret_cast<RtpReceiverInterface*>(j_rtp_receiver_pointer)
>+      ->SetObserver(nullptr);
>+  RtpReceiverObserverJni* observer =
>+      reinterpret_cast<RtpReceiverObserverJni*>(j_observer_pointer);
>+  if (observer) {
>+    delete observer;
>+  }
>+}
>+
>+}  // namespace jni
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/rtpreceiver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/rtpreceiver.h
>new file mode 100644
>index 00000000000..c2e9c31aedf
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/rtpreceiver.h
>@@ -0,0 +1,41 @@
>+/*
>+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef SDK_ANDROID_SRC_JNI_PC_RTPRECEIVER_H_
>+#define SDK_ANDROID_SRC_JNI_PC_RTPRECEIVER_H_
>+
>+#include <jni.h>
>+
>+#include "api/rtpreceiverinterface.h"
>+#include "sdk/android/native_api/jni/scoped_java_ref.h"
>+
>+namespace webrtc {
>+namespace jni {
>+
>+ScopedJavaLocalRef<jobject> NativeToJavaRtpReceiver(
>+    JNIEnv* env,
>+    rtc::scoped_refptr<RtpReceiverInterface> receiver);
>+
>+// Takes ownership of the passed |j_receiver| and stores it as a global
>+// reference. Will call dispose() in the dtor.
>+class JavaRtpReceiverGlobalOwner {
>+ public:
>+  JavaRtpReceiverGlobalOwner(JNIEnv* env, const JavaRef<jobject>& j_receiver);
>+  JavaRtpReceiverGlobalOwner(JavaRtpReceiverGlobalOwner&& other);
>+  ~JavaRtpReceiverGlobalOwner();
>+
>+ private:
>+  ScopedJavaGlobalRef<jobject> j_receiver_;
>+};
>+
>+}  // namespace jni
>+}  // namespace webrtc
>+
>+#endif  // SDK_ANDROID_SRC_JNI_PC_RTPRECEIVER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/rtpreceiver_jni.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/rtpreceiver_jni.cc
>deleted file mode 100644
>index eb60cc0c0fa..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/rtpreceiver_jni.cc
>+++ /dev/null
>@@ -1,96 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "api/rtpreceiverinterface.h"
>-#include "sdk/android/src/jni/jni_helpers.h"
>-#include "sdk/android/src/jni/pc/java_native_conversion.h"
>-#include "sdk/android/src/jni/pc/rtpreceiverobserver_jni.h"
>-
>-namespace webrtc {
>-namespace jni {
>-
>-JNI_FUNCTION_DECLARATION(jlong,
>-                         RtpReceiver_nativeGetTrack,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jlong j_rtp_receiver_pointer,
>-                         jlong j_track_pointer) {
>-  return jlongFromPointer(
>-      reinterpret_cast<RtpReceiverInterface*>(j_rtp_receiver_pointer)
>-          ->track()
>-          .release());
>-}
>-
>-JNI_FUNCTION_DECLARATION(jboolean,
>-                         RtpReceiver_nativeSetParameters,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jlong j_rtp_receiver_pointer,
>-                         jobject j_parameters) {
>-  if (IsNull(jni, j_parameters)) {
>-    return false;
>-  }
>-  RtpParameters parameters;
>-  JavaToNativeRtpParameters(jni, j_parameters, &parameters);
>-  return reinterpret_cast<RtpReceiverInterface*>(j_rtp_receiver_pointer)
>-      ->SetParameters(parameters);
>-}
>-
>-JNI_FUNCTION_DECLARATION(jobject,
>-                         RtpReceiver_nativeGetParameters,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jlong j_rtp_receiver_pointer) {
>-  RtpParameters parameters =
>-      reinterpret_cast<RtpReceiverInterface*>(j_rtp_receiver_pointer)
>-          ->GetParameters();
>-  return NativeToJavaRtpParameters(jni, parameters);
>-}
>-
>-JNI_FUNCTION_DECLARATION(jstring,
>-                         RtpReceiver_nativeId,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jlong j_rtp_receiver_pointer) {
>-  return NativeToJavaString(
>-      jni,
>-      reinterpret_cast<RtpReceiverInterface*>(j_rtp_receiver_pointer)->id());
>-}
>-
>-JNI_FUNCTION_DECLARATION(jlong,
>-                         RtpReceiver_nativeSetObserver,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jlong j_rtp_receiver_pointer,
>-                         jobject j_observer) {
>-  RtpReceiverObserverJni* rtpReceiverObserver =
>-      new RtpReceiverObserverJni(jni, j_observer);
>-  reinterpret_cast<RtpReceiverInterface*>(j_rtp_receiver_pointer)
>-      ->SetObserver(rtpReceiverObserver);
>-  return jlongFromPointer(rtpReceiverObserver);
>-}
>-
>-JNI_FUNCTION_DECLARATION(void,
>-                         RtpReceiver_nativeUnsetObserver,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jlong j_rtp_receiver_pointer,
>-                         jlong j_observer_pointer) {
>-  reinterpret_cast<RtpReceiverInterface*>(j_rtp_receiver_pointer)
>-      ->SetObserver(nullptr);
>-  RtpReceiverObserverJni* observer =
>-      reinterpret_cast<RtpReceiverObserverJni*>(j_observer_pointer);
>-  if (observer) {
>-    delete observer;
>-  }
>-}
>-
>-}  // namespace jni
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/rtpreceiverobserver_jni.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/rtpreceiverobserver_jni.cc
>deleted file mode 100644
>index 8e900631da7..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/rtpreceiverobserver_jni.cc
>+++ /dev/null
>@@ -1,34 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "sdk/android/src/jni/pc/rtpreceiverobserver_jni.h"
>-
>-#include "sdk/android/src/jni/pc/java_native_conversion.h"
>-
>-namespace webrtc {
>-namespace jni {
>-
>-void RtpReceiverObserverJni::OnFirstPacketReceived(
>-    cricket::MediaType media_type) {
>-  JNIEnv* const jni = AttachCurrentThreadIfNeeded();
>-
>-  jmethodID j_on_first_packet_received_mid = GetMethodID(
>-      jni, GetObjectClass(jni, *j_observer_global_), "onFirstPacketReceived",
>-      "(Lorg/webrtc/MediaStreamTrack$MediaType;)V");
>-  // Get the Java version of media type.
>-  jobject JavaMediaType = NativeToJavaMediaType(jni, media_type);
>-  // Trigger the callback function.
>-  jni->CallVoidMethod(*j_observer_global_, j_on_first_packet_received_mid,
>-                      JavaMediaType);
>-  CHECK_EXCEPTION(jni) << "error during CallVoidMethod";
>-}
>-
>-}  // namespace jni
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/rtpreceiverobserver_jni.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/rtpreceiverobserver_jni.h
>deleted file mode 100644
>index c6455338e8e..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/rtpreceiverobserver_jni.h
>+++ /dev/null
>@@ -1,39 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef SDK_ANDROID_SRC_JNI_PC_RTPRECEIVEROBSERVER_JNI_H_
>-#define SDK_ANDROID_SRC_JNI_PC_RTPRECEIVEROBSERVER_JNI_H_
>-
>-#include "api/rtpreceiverinterface.h"
>-#include "sdk/android/src/jni/jni_helpers.h"
>-
>-namespace webrtc {
>-namespace jni {
>-
>-// Adapter between the C++ RtpReceiverObserverInterface and the Java
>-// RtpReceiver.Observer interface. Wraps an instance of the Java interface and
>-// dispatches C++ callbacks to Java.
>-class RtpReceiverObserverJni : public RtpReceiverObserverInterface {
>- public:
>-  RtpReceiverObserverJni(JNIEnv* jni, jobject j_observer)
>-      : j_observer_global_(jni, j_observer) {}
>-
>-  ~RtpReceiverObserverJni() override {}
>-
>-  void OnFirstPacketReceived(cricket::MediaType media_type) override;
>-
>- private:
>-  const ScopedGlobalRef<jobject> j_observer_global_;
>-};
>-
>-}  // namespace jni
>-}  // namespace webrtc
>-
>-#endif  // SDK_ANDROID_SRC_JNI_PC_RTPRECEIVEROBSERVER_JNI_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/rtpsender.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/rtpsender.cc
>new file mode 100644
>index 00000000000..772dbcd613a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/rtpsender.cc
>@@ -0,0 +1,91 @@
>+/*
>+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "sdk/android/src/jni/pc/rtpsender.h"
>+
>+#include "sdk/android/generated_peerconnection_jni/jni/RtpSender_jni.h"
>+#include "sdk/android/native_api/jni/java_types.h"
>+#include "sdk/android/src/jni/jni_helpers.h"
>+#include "sdk/android/src/jni/pc/rtpparameters.h"
>+
>+namespace webrtc {
>+namespace jni {
>+
>+ScopedJavaLocalRef<jobject> NativeToJavaRtpSender(
>+    JNIEnv* env,
>+    rtc::scoped_refptr<RtpSenderInterface> sender) {
>+  if (!sender)
>+    return nullptr;
>+  // Sender is now owned by the Java object, and will be freed from
>+  // RtpSender.dispose(), called by PeerConnection.dispose() or getSenders().
>+  return Java_RtpSender_Constructor(env, jlongFromPointer(sender.release()));
>+}
>+
>+static jboolean JNI_RtpSender_SetTrack(JNIEnv* jni,
>+                                       const JavaParamRef<jclass>&,
>+                                       jlong j_rtp_sender_pointer,
>+                                       jlong j_track_pointer) {
>+  return reinterpret_cast<RtpSenderInterface*>(j_rtp_sender_pointer)
>+      ->SetTrack(reinterpret_cast<MediaStreamTrackInterface*>(j_track_pointer));
>+}
>+
>+jlong JNI_RtpSender_GetTrack(JNIEnv* jni,
>+                             const JavaParamRef<jclass>&,
>+                             jlong j_rtp_sender_pointer) {
>+  // MediaStreamTrack will have shared ownership by the MediaStreamTrack Java
>+  // object.
>+  return jlongFromPointer(
>+      reinterpret_cast<RtpSenderInterface*>(j_rtp_sender_pointer)
>+          ->track()
>+          .release());
>+}
>+
>+jlong JNI_RtpSender_GetDtmfSender(JNIEnv* jni,
>+                                  const JavaParamRef<jclass>&,
>+                                  jlong j_rtp_sender_pointer) {
>+  return jlongFromPointer(
>+      reinterpret_cast<RtpSenderInterface*>(j_rtp_sender_pointer)
>+          ->GetDtmfSender()
>+          .release());
>+}
>+
>+jboolean JNI_RtpSender_SetParameters(
>+    JNIEnv* jni,
>+    const JavaParamRef<jclass>&,
>+    jlong j_rtp_sender_pointer,
>+    const JavaParamRef<jobject>& j_parameters) {
>+  if (IsNull(jni, j_parameters)) {
>+    return false;
>+  }
>+  RtpParameters parameters = JavaToNativeRtpParameters(jni, j_parameters);
>+  return reinterpret_cast<RtpSenderInterface*>(j_rtp_sender_pointer)
>+      ->SetParameters(parameters)
>+      .ok();
>+}
>+
>+ScopedJavaLocalRef<jobject> JNI_RtpSender_GetParameters(
>+    JNIEnv* jni,
>+    const JavaParamRef<jclass>&,
>+    jlong j_rtp_sender_pointer) {
>+  RtpParameters parameters =
>+      reinterpret_cast<RtpSenderInterface*>(j_rtp_sender_pointer)
>+          ->GetParameters();
>+  return NativeToJavaRtpParameters(jni, parameters);
>+}
>+
>+ScopedJavaLocalRef<jstring> JNI_RtpSender_GetId(JNIEnv* jni,
>+                                                const JavaParamRef<jclass>&,
>+                                                jlong j_rtp_sender_pointer) {
>+  return NativeToJavaString(
>+      jni, reinterpret_cast<RtpSenderInterface*>(j_rtp_sender_pointer)->id());
>+}
>+
>+}  // namespace jni
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/rtpsender.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/rtpsender.h
>new file mode 100644
>index 00000000000..5d30ed52e40
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/rtpsender.h
>@@ -0,0 +1,29 @@
>+/*
>+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef SDK_ANDROID_SRC_JNI_PC_RTPSENDER_H_
>+#define SDK_ANDROID_SRC_JNI_PC_RTPSENDER_H_
>+
>+#include <jni.h>
>+
>+#include "api/rtpsenderinterface.h"
>+#include "sdk/android/native_api/jni/scoped_java_ref.h"
>+
>+namespace webrtc {
>+namespace jni {
>+
>+ScopedJavaLocalRef<jobject> NativeToJavaRtpSender(
>+    JNIEnv* env,
>+    rtc::scoped_refptr<RtpSenderInterface> sender);
>+
>+}  // namespace jni
>+}  // namespace webrtc
>+
>+#endif  // SDK_ANDROID_SRC_JNI_PC_RTPSENDER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/rtpsender_jni.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/rtpsender_jni.cc
>deleted file mode 100644
>index a30bc6c87e7..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/rtpsender_jni.cc
>+++ /dev/null
>@@ -1,86 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "api/rtpsenderinterface.h"
>-#include "sdk/android/src/jni/jni_helpers.h"
>-#include "sdk/android/src/jni/pc/java_native_conversion.h"
>-
>-namespace webrtc {
>-namespace jni {
>-
>-JNI_FUNCTION_DECLARATION(jboolean,
>-                         RtpSender_nativeSetTrack,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jlong j_rtp_sender_pointer,
>-                         jlong j_track_pointer) {
>-  return reinterpret_cast<RtpSenderInterface*>(j_rtp_sender_pointer)
>-      ->SetTrack(reinterpret_cast<MediaStreamTrackInterface*>(j_track_pointer));
>-}
>-
>-JNI_FUNCTION_DECLARATION(jlong,
>-                         RtpSender_nativeGetTrack,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jlong j_rtp_sender_pointer) {
>-  return jlongFromPointer(
>-      reinterpret_cast<RtpSenderInterface*>(j_rtp_sender_pointer)
>-          ->track()
>-          .release());
>-}
>-
>-JNI_FUNCTION_DECLARATION(jlong,
>-                         RtpSender_nativeGetDtmfSender,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jlong j_rtp_sender_pointer) {
>-  return jlongFromPointer(
>-      reinterpret_cast<RtpSenderInterface*>(j_rtp_sender_pointer)
>-          ->GetDtmfSender()
>-          .release());
>-}
>-
>-JNI_FUNCTION_DECLARATION(jboolean,
>-                         RtpSender_nativeSetParameters,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jlong j_rtp_sender_pointer,
>-                         jobject j_parameters) {
>-  if (IsNull(jni, j_parameters)) {
>-    return false;
>-  }
>-  RtpParameters parameters;
>-  JavaToNativeRtpParameters(jni, j_parameters, &parameters);
>-  return reinterpret_cast<RtpSenderInterface*>(j_rtp_sender_pointer)
>-      ->SetParameters(parameters);
>-}
>-
>-JNI_FUNCTION_DECLARATION(jobject,
>-                         RtpSender_nativeGetParameters,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jlong j_rtp_sender_pointer) {
>-  RtpParameters parameters =
>-      reinterpret_cast<RtpSenderInterface*>(j_rtp_sender_pointer)
>-          ->GetParameters();
>-  return NativeToJavaRtpParameters(jni, parameters);
>-}
>-
>-JNI_FUNCTION_DECLARATION(jstring,
>-                         RtpSender_nativeId,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jlong j_rtp_sender_pointer) {
>-  return NativeToJavaString(
>-      jni, reinterpret_cast<RtpSenderInterface*>(j_rtp_sender_pointer)->id());
>-}
>-
>-}  // namespace jni
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/rtptransceiver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/rtptransceiver.cc
>new file mode 100644
>index 00000000000..fa20d8031fb
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/rtptransceiver.cc
>@@ -0,0 +1,165 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "sdk/android/src/jni/pc/rtptransceiver.h"
>+
>+#include <string>
>+
>+#include "sdk/android/generated_peerconnection_jni/jni/RtpTransceiver_jni.h"
>+#include "sdk/android/native_api/jni/java_types.h"
>+#include "sdk/android/src/jni/jni_helpers.h"
>+#include "sdk/android/src/jni/pc/mediastreamtrack.h"
>+#include "sdk/android/src/jni/pc/rtpparameters.h"
>+#include "sdk/android/src/jni/pc/rtpreceiver.h"
>+#include "sdk/android/src/jni/pc/rtpsender.h"
>+
>+namespace webrtc {
>+namespace jni {
>+
>+namespace {
>+
>+ScopedJavaLocalRef<jobject> NativeToJavaRtpTransceiverDirection(
>+    JNIEnv* jni,
>+    RtpTransceiverDirection rtp_transceiver_direction) {
>+  return Java_RtpTransceiverDirection_fromNativeIndex(
>+      jni, static_cast<int>(rtp_transceiver_direction));
>+}
>+
>+}  // namespace
>+
>+RtpTransceiverInit JavaToNativeRtpTransceiverInit(
>+    JNIEnv* jni,
>+    const JavaRef<jobject>& j_init) {
>+  RtpTransceiverInit init;
>+
>+  // Convert the direction.
>+  init.direction = static_cast<RtpTransceiverDirection>(
>+      Java_RtpTransceiverInit_getDirectionNativeIndex(jni, j_init));
>+
>+  // Convert the stream ids.
>+  ScopedJavaLocalRef<jobject> j_stream_ids =
>+      Java_RtpTransceiverInit_getStreamIds(jni, j_init);
>+  init.stream_ids = JavaListToNativeVector<std::string, jstring>(
>+      jni, j_stream_ids, &JavaToNativeString);
>+  return init;
>+}
>+
>+ScopedJavaLocalRef<jobject> NativeToJavaRtpTransceiver(
>+    JNIEnv* env,
>+    rtc::scoped_refptr<RtpTransceiverInterface> transceiver) {
>+  if (!transceiver) {
>+    return nullptr;
>+  }
>+  // Transceiver will now have shared ownership by the Java object.
>+  return Java_RtpTransceiver_Constructor(
>+      env, jlongFromPointer(transceiver.release()));
>+}
>+
>+JavaRtpTransceiverGlobalOwner::JavaRtpTransceiverGlobalOwner(
>+    JNIEnv* env,
>+    const JavaRef<jobject>& j_transceiver)
>+    : j_transceiver_(env, j_transceiver){};
>+
>+JavaRtpTransceiverGlobalOwner::JavaRtpTransceiverGlobalOwner(
>+    JavaRtpTransceiverGlobalOwner&& other) = default;
>+
>+JavaRtpTransceiverGlobalOwner::~JavaRtpTransceiverGlobalOwner() {
>+  if (j_transceiver_.obj()) {
>+    Java_RtpTransceiver_dispose(AttachCurrentThreadIfNeeded(), j_transceiver_);
>+  }
>+}
>+
>+ScopedJavaLocalRef<jobject> JNI_RtpTransceiver_GetMediaType(
>+    JNIEnv* jni,
>+    const base::android::JavaParamRef<jclass>&,
>+    jlong j_rtp_transceiver_pointer) {
>+  return NativeToJavaMediaType(
>+      jni, reinterpret_cast<RtpTransceiverInterface*>(j_rtp_transceiver_pointer)
>+               ->media_type());
>+}
>+
>+ScopedJavaLocalRef<jstring> JNI_RtpTransceiver_GetMid(
>+    JNIEnv* jni,
>+    const base::android::JavaParamRef<jclass>&,
>+    jlong j_rtp_transceiver_pointer) {
>+  absl::optional<std::string> mid =
>+      reinterpret_cast<RtpTransceiverInterface*>(j_rtp_transceiver_pointer)
>+          ->mid();
>+  return NativeToJavaString(jni, mid);
>+}
>+
>+ScopedJavaLocalRef<jobject> JNI_RtpTransceiver_GetSender(
>+    JNIEnv* jni,
>+    const base::android::JavaParamRef<jclass>&,
>+    jlong j_rtp_transceiver_pointer) {
>+  return NativeToJavaRtpSender(
>+      jni, reinterpret_cast<RtpTransceiverInterface*>(j_rtp_transceiver_pointer)
>+               ->sender());
>+}
>+
>+ScopedJavaLocalRef<jobject> JNI_RtpTransceiver_GetReceiver(
>+    JNIEnv* jni,
>+    const base::android::JavaParamRef<jclass>&,
>+    jlong j_rtp_transceiver_pointer) {
>+  return NativeToJavaRtpReceiver(
>+      jni, reinterpret_cast<RtpTransceiverInterface*>(j_rtp_transceiver_pointer)
>+               ->receiver());
>+}
>+
>+jboolean JNI_RtpTransceiver_Stopped(JNIEnv* jni,
>+                                    const base::android::JavaParamRef<jclass>&,
>+                                    jlong j_rtp_transceiver_pointer) {
>+  return reinterpret_cast<RtpTransceiverInterface*>(j_rtp_transceiver_pointer)
>+      ->stopped();
>+}
>+
>+ScopedJavaLocalRef<jobject> JNI_RtpTransceiver_Direction(
>+    JNIEnv* jni,
>+    const base::android::JavaParamRef<jclass>&,
>+    jlong j_rtp_transceiver_pointer) {
>+  return NativeToJavaRtpTransceiverDirection(
>+      jni, reinterpret_cast<RtpTransceiverInterface*>(j_rtp_transceiver_pointer)
>+               ->direction());
>+}
>+
>+ScopedJavaLocalRef<jobject> JNI_RtpTransceiver_CurrentDirection(
>+    JNIEnv* jni,
>+    const base::android::JavaParamRef<jclass>&,
>+    jlong j_rtp_transceiver_pointer) {
>+  absl::optional<RtpTransceiverDirection> direction =
>+      reinterpret_cast<RtpTransceiverInterface*>(j_rtp_transceiver_pointer)
>+          ->current_direction();
>+  return direction ? NativeToJavaRtpTransceiverDirection(jni, *direction)
>+                   : nullptr;
>+}
>+
>+void JNI_RtpTransceiver_Stop(JNIEnv* jni,
>+                             const base::android::JavaParamRef<jclass>&,
>+                             jlong j_rtp_transceiver_pointer) {
>+  reinterpret_cast<RtpTransceiverInterface*>(j_rtp_transceiver_pointer)->Stop();
>+}
>+
>+void JNI_RtpTransceiver_SetDirection(
>+    JNIEnv* jni,
>+    const base::android::JavaParamRef<jclass>&,
>+    jlong j_rtp_transceiver_pointer,
>+    const base::android::JavaParamRef<jobject>& j_rtp_transceiver_direction) {
>+  if (IsNull(jni, j_rtp_transceiver_direction)) {
>+    return;
>+  }
>+  RtpTransceiverDirection direction = static_cast<RtpTransceiverDirection>(
>+      Java_RtpTransceiverDirection_getNativeIndex(jni,
>+                                                  j_rtp_transceiver_direction));
>+  reinterpret_cast<RtpTransceiverInterface*>(j_rtp_transceiver_pointer)
>+      ->SetDirection(direction);
>+}
>+
>+}  // namespace jni
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/rtptransceiver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/rtptransceiver.h
>new file mode 100644
>index 00000000000..61f3e0cafc2
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/rtptransceiver.h
>@@ -0,0 +1,46 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef SDK_ANDROID_SRC_JNI_PC_RTPTRANSCEIVER_H_
>+#define SDK_ANDROID_SRC_JNI_PC_RTPTRANSCEIVER_H_
>+
>+#include <jni.h>
>+
>+#include "api/rtptransceiverinterface.h"
>+#include "sdk/android/native_api/jni/scoped_java_ref.h"
>+
>+namespace webrtc {
>+namespace jni {
>+
>+RtpTransceiverInit JavaToNativeRtpTransceiverInit(
>+    JNIEnv* jni,
>+    const JavaRef<jobject>& j_init);
>+
>+ScopedJavaLocalRef<jobject> NativeToJavaRtpTransceiver(
>+    JNIEnv* env,
>+    rtc::scoped_refptr<RtpTransceiverInterface> transceiver);
>+
>+// This takes ownership of the of the |j_transceiver| and stores it as a global
>+// reference. This calls the Java Transceiver's dispose() method with the dtor.
>+class JavaRtpTransceiverGlobalOwner {
>+ public:
>+  JavaRtpTransceiverGlobalOwner(JNIEnv* env,
>+                                const JavaRef<jobject>& j_transceiver);
>+  JavaRtpTransceiverGlobalOwner(JavaRtpTransceiverGlobalOwner&& other);
>+  ~JavaRtpTransceiverGlobalOwner();
>+
>+ private:
>+  ScopedJavaGlobalRef<jobject> j_transceiver_;
>+};
>+
>+}  // namespace jni
>+}  // namespace webrtc
>+
>+#endif  // SDK_ANDROID_SRC_JNI_PC_RTPTRANSCEIVER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/sdpobserver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/sdpobserver.cc
>new file mode 100644
>index 00000000000..87989ab0470
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/sdpobserver.cc
>@@ -0,0 +1,68 @@
>+/*
>+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "sdk/android/src/jni/pc/sdpobserver.h"
>+
>+#include <utility>
>+
>+#include "api/mediaconstraintsinterface.h"
>+#include "sdk/android/generated_peerconnection_jni/jni/SdpObserver_jni.h"
>+#include "sdk/android/native_api/jni/java_types.h"
>+#include "sdk/android/src/jni/jni_helpers.h"
>+
>+namespace webrtc {
>+namespace jni {
>+
>+CreateSdpObserverJni::CreateSdpObserverJni(
>+    JNIEnv* env,
>+    const JavaRef<jobject>& j_observer,
>+    std::unique_ptr<MediaConstraintsInterface> constraints)
>+    : j_observer_global_(env, j_observer),
>+      constraints_(std::move(constraints)) {}
>+
>+CreateSdpObserverJni::~CreateSdpObserverJni() = default;
>+
>+void CreateSdpObserverJni::OnSuccess(SessionDescriptionInterface* desc) {
>+  JNIEnv* env = AttachCurrentThreadIfNeeded();
>+  Java_SdpObserver_onCreateSuccess(env, j_observer_global_,
>+                                   NativeToJavaSessionDescription(env, desc));
>+  // OnSuccess transfers ownership of the description (there's a TODO to make
>+  // it use unique_ptr...).
>+  delete desc;
>+}
>+
>+void CreateSdpObserverJni::OnFailure(webrtc::RTCError error) {
>+  JNIEnv* env = AttachCurrentThreadIfNeeded();
>+  Java_SdpObserver_onCreateFailure(env, j_observer_global_,
>+                                   NativeToJavaString(env, error.message()));
>+}
>+
>+SetSdpObserverJni::SetSdpObserverJni(
>+    JNIEnv* env,
>+    const JavaRef<jobject>& j_observer,
>+    std::unique_ptr<MediaConstraintsInterface> constraints)
>+    : j_observer_global_(env, j_observer),
>+      constraints_(std::move(constraints)) {}
>+
>+SetSdpObserverJni::~SetSdpObserverJni() = default;
>+
>+void SetSdpObserverJni::OnSuccess() {
>+  JNIEnv* env = AttachCurrentThreadIfNeeded();
>+  Java_SdpObserver_onSetSuccess(env, j_observer_global_);
>+}
>+
>+void SetSdpObserverJni::OnFailure(webrtc::RTCError error) {
>+  JNIEnv* env = AttachCurrentThreadIfNeeded();
>+  Java_SdpObserver_onSetFailure(env, j_observer_global_,
>+                                NativeToJavaString(env, error.message()));
>+}
>+
>+}  // namespace jni
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/sdpobserver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/sdpobserver.h
>new file mode 100644
>index 00000000000..6673fececf9
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/sdpobserver.h
>@@ -0,0 +1,61 @@
>+/*
>+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef SDK_ANDROID_SRC_JNI_PC_SDPOBSERVER_H_
>+#define SDK_ANDROID_SRC_JNI_PC_SDPOBSERVER_H_
>+
>+#include <memory>
>+#include <string>
>+
>+#include "api/peerconnectioninterface.h"
>+#include "sdk/android/src/jni/jni_helpers.h"
>+#include "sdk/android/src/jni/pc/sessiondescription.h"
>+
>+namespace webrtc {
>+namespace jni {
>+
>+class CreateSdpObserverJni : public CreateSessionDescriptionObserver {
>+ public:
>+  CreateSdpObserverJni(JNIEnv* env,
>+                       const JavaRef<jobject>& j_observer,
>+                       std::unique_ptr<MediaConstraintsInterface> constraints);
>+  ~CreateSdpObserverJni() override;
>+
>+  MediaConstraintsInterface* constraints() { return constraints_.get(); }
>+
>+  void OnSuccess(SessionDescriptionInterface* desc) override;
>+  void OnFailure(RTCError error) override;
>+
>+ private:
>+  const ScopedJavaGlobalRef<jobject> j_observer_global_;
>+  std::unique_ptr<MediaConstraintsInterface> constraints_;
>+};
>+
>+class SetSdpObserverJni : public SetSessionDescriptionObserver {
>+ public:
>+  SetSdpObserverJni(JNIEnv* env,
>+                    const JavaRef<jobject>& j_observer,
>+                    std::unique_ptr<MediaConstraintsInterface> constraints);
>+  ~SetSdpObserverJni() override;
>+
>+  MediaConstraintsInterface* constraints() { return constraints_.get(); }
>+
>+  void OnSuccess() override;
>+  void OnFailure(RTCError error) override;
>+
>+ private:
>+  const ScopedJavaGlobalRef<jobject> j_observer_global_;
>+  std::unique_ptr<MediaConstraintsInterface> constraints_;
>+};
>+
>+}  // namespace jni
>+}  // namespace webrtc
>+
>+#endif  // SDK_ANDROID_SRC_JNI_PC_SDPOBSERVER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/sdpobserver_jni.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/sdpobserver_jni.h
>deleted file mode 100644
>index 5ed3b59acf0..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/sdpobserver_jni.h
>+++ /dev/null
>@@ -1,110 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef SDK_ANDROID_SRC_JNI_PC_SDPOBSERVER_JNI_H_
>-#define SDK_ANDROID_SRC_JNI_PC_SDPOBSERVER_JNI_H_
>-
>-#include <memory>
>-#include <string>
>-
>-#include "api/peerconnectioninterface.h"
>-#include "sdk/android/src/jni/jni_helpers.h"
>-
>-namespace webrtc {
>-namespace jni {
>-
>-// Adapter for a Java StatsObserver presenting a C++
>-// CreateSessionDescriptionObserver or SetSessionDescriptionObserver and
>-// dispatching the callback from C++ back to Java.
>-template <class T>  // T is one of {Create,Set}SessionDescriptionObserver.
>-class SdpObserverJni : public T {
>- public:
>-  SdpObserverJni(JNIEnv* jni,
>-                 jobject j_observer,
>-                 std::unique_ptr<MediaConstraintsInterface> constraints)
>-      : constraints_(std::move(constraints)),
>-        j_observer_global_(jni, j_observer),
>-        j_observer_class_(jni, GetObjectClass(jni, j_observer)) {}
>-
>-  virtual ~SdpObserverJni() {}
>-
>-  // Can't mark override because of templating.
>-  virtual void OnSuccess() {
>-    ScopedLocalRefFrame local_ref_frame(jni());
>-    jmethodID m = GetMethodID(jni(), *j_observer_class_, "onSetSuccess", "()V");
>-    jni()->CallVoidMethod(*j_observer_global_, m);
>-    CHECK_EXCEPTION(jni()) << "error during CallVoidMethod";
>-  }
>-
>-  // Can't mark override because of templating.
>-  virtual void OnSuccess(SessionDescriptionInterface* desc) {
>-    ScopedLocalRefFrame local_ref_frame(jni());
>-    jmethodID m = GetMethodID(jni(), *j_observer_class_, "onCreateSuccess",
>-                              "(Lorg/webrtc/SessionDescription;)V");
>-    jobject j_sdp = NativeToJavaSessionDescription(jni(), desc);
>-    jni()->CallVoidMethod(*j_observer_global_, m, j_sdp);
>-    CHECK_EXCEPTION(jni()) << "error during CallVoidMethod";
>-    // OnSuccess transfers ownership of the description (there's a TODO to make
>-    // it use unique_ptr...).
>-    delete desc;
>-  }
>-
>-  MediaConstraintsInterface* constraints() { return constraints_.get(); }
>-
>- protected:
>-  // Common implementation for failure of Set & Create types, distinguished by
>-  // |op| being "Set" or "Create".
>-  void DoOnFailure(const std::string& op, const std::string& error) {
>-    jmethodID m = GetMethodID(jni(), *j_observer_class_, "on" + op + "Failure",
>-                              "(Ljava/lang/String;)V");
>-    jstring j_error_string = NativeToJavaString(jni(), error);
>-    jni()->CallVoidMethod(*j_observer_global_, m, j_error_string);
>-    CHECK_EXCEPTION(jni()) << "error during CallVoidMethod";
>-  }
>-
>-  JNIEnv* jni() { return AttachCurrentThreadIfNeeded(); }
>-
>- private:
>-  std::unique_ptr<MediaConstraintsInterface> constraints_;
>-  const ScopedGlobalRef<jobject> j_observer_global_;
>-  const ScopedGlobalRef<jclass> j_observer_class_;
>-};
>-
>-class CreateSdpObserverJni
>-    : public SdpObserverJni<CreateSessionDescriptionObserver> {
>- public:
>-  CreateSdpObserverJni(JNIEnv* jni,
>-                       jobject j_observer,
>-                       std::unique_ptr<MediaConstraintsInterface> constraints)
>-      : SdpObserverJni(jni, j_observer, std::move(constraints)) {}
>-
>-  void OnFailure(const std::string& error) override {
>-    ScopedLocalRefFrame local_ref_frame(jni());
>-    SdpObserverJni::DoOnFailure(std::string("Create"), error);
>-  }
>-};
>-
>-class SetSdpObserverJni : public SdpObserverJni<SetSessionDescriptionObserver> {
>- public:
>-  SetSdpObserverJni(JNIEnv* jni,
>-                    jobject j_observer,
>-                    std::unique_ptr<MediaConstraintsInterface> constraints)
>-      : SdpObserverJni(jni, j_observer, std::move(constraints)) {}
>-
>-  void OnFailure(const std::string& error) override {
>-    ScopedLocalRefFrame local_ref_frame(jni());
>-    SdpObserverJni::DoOnFailure(std::string("Set"), error);
>-  }
>-};
>-
>-}  // namespace jni
>-}  // namespace webrtc
>-
>-#endif  // SDK_ANDROID_SRC_JNI_PC_SDPOBSERVER_JNI_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/sessiondescription.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/sessiondescription.cc
>new file mode 100644
>index 00000000000..bd3806d5626
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/sessiondescription.cc
>@@ -0,0 +1,50 @@
>+/*
>+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "sdk/android/src/jni/pc/sessiondescription.h"
>+
>+#include <string>
>+
>+#include "rtc_base/logging.h"
>+#include "sdk/android/generated_peerconnection_jni/jni/SessionDescription_jni.h"
>+#include "sdk/android/native_api/jni/java_types.h"
>+#include "sdk/android/src/jni/jni_helpers.h"
>+
>+namespace webrtc {
>+namespace jni {
>+
>+std::unique_ptr<SessionDescriptionInterface> JavaToNativeSessionDescription(
>+    JNIEnv* jni,
>+    const JavaRef<jobject>& j_sdp) {
>+  std::string std_type = JavaToStdString(
>+      jni, Java_SessionDescription_getTypeInCanonicalForm(jni, j_sdp));
>+  std::string std_description =
>+      JavaToStdString(jni, Java_SessionDescription_getDescription(jni, j_sdp));
>+  absl::optional<SdpType> sdp_type_maybe = SdpTypeFromString(std_type);
>+  if (!sdp_type_maybe) {
>+    RTC_LOG(LS_ERROR) << "Unexpected SDP type: " << std_type;
>+    return nullptr;
>+  }
>+  return CreateSessionDescription(*sdp_type_maybe, std_description);
>+}
>+
>+ScopedJavaLocalRef<jobject> NativeToJavaSessionDescription(
>+    JNIEnv* jni,
>+    const SessionDescriptionInterface* desc) {
>+  std::string sdp;
>+  RTC_CHECK(desc->ToString(&sdp)) << "got so far: " << sdp;
>+  return Java_SessionDescription_Constructor(
>+      jni,
>+      Java_Type_fromCanonicalForm(jni, NativeToJavaString(jni, desc->type())),
>+      NativeToJavaString(jni, sdp));
>+}
>+
>+}  // namespace jni
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/sessiondescription.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/sessiondescription.h
>new file mode 100644
>index 00000000000..a7ef6671582
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/sessiondescription.h
>@@ -0,0 +1,34 @@
>+/*
>+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef SDK_ANDROID_SRC_JNI_PC_SESSIONDESCRIPTION_H_
>+#define SDK_ANDROID_SRC_JNI_PC_SESSIONDESCRIPTION_H_
>+
>+#include <jni.h>
>+#include <memory>
>+
>+#include "api/jsep.h"
>+#include "sdk/android/native_api/jni/scoped_java_ref.h"
>+
>+namespace webrtc {
>+namespace jni {
>+
>+std::unique_ptr<SessionDescriptionInterface> JavaToNativeSessionDescription(
>+    JNIEnv* jni,
>+    const JavaRef<jobject>& j_sdp);
>+
>+ScopedJavaLocalRef<jobject> NativeToJavaSessionDescription(
>+    JNIEnv* jni,
>+    const SessionDescriptionInterface* desc);
>+
>+}  // namespace jni
>+}  // namespace webrtc
>+
>+#endif  // SDK_ANDROID_SRC_JNI_PC_SESSIONDESCRIPTION_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/sslcertificateverifierwrapper.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/sslcertificateverifierwrapper.cc
>new file mode 100644
>index 00000000000..2ff9a462e37
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/sslcertificateverifierwrapper.cc
>@@ -0,0 +1,44 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "sdk/android/src/jni/pc/sslcertificateverifierwrapper.h"
>+#include "sdk/android/generated_peerconnection_jni/jni/SSLCertificateVerifier_jni.h"
>+#include "sdk/android/native_api/jni/class_loader.h"
>+#include "sdk/android/native_api/jni/java_types.h"
>+
>+namespace webrtc {
>+namespace jni {
>+
>+SSLCertificateVerifierWrapper::SSLCertificateVerifierWrapper(
>+    JNIEnv* jni,
>+    const JavaRef<jobject>& ssl_certificate_verifier)
>+    : ssl_certificate_verifier_(jni, ssl_certificate_verifier) {}
>+
>+SSLCertificateVerifierWrapper::~SSLCertificateVerifierWrapper() = default;
>+
>+bool SSLCertificateVerifierWrapper::Verify(
>+    const rtc::SSLCertificate& certificate) {
>+  JNIEnv* jni = AttachCurrentThreadIfNeeded();
>+
>+  // Serialize the der encoding of the cert into a jbyteArray
>+  rtc::Buffer cert_der_buffer;
>+  certificate.ToDER(&cert_der_buffer);
>+  ScopedJavaLocalRef<jbyteArray> jni_buffer(
>+      jni, jni->NewByteArray(cert_der_buffer.size()));
>+  jni->SetByteArrayRegion(
>+      jni_buffer.obj(), 0, cert_der_buffer.size(),
>+      reinterpret_cast<const jbyte*>(cert_der_buffer.data()));
>+
>+  return Java_SSLCertificateVerifier_verify(jni, ssl_certificate_verifier_,
>+                                            jni_buffer);
>+}
>+
>+}  // namespace jni
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/sslcertificateverifierwrapper.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/sslcertificateverifierwrapper.h
>new file mode 100644
>index 00000000000..719fa44494b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/sslcertificateverifierwrapper.h
>@@ -0,0 +1,41 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef SDK_ANDROID_SRC_JNI_PC_SSLCERTIFICATEVERIFIERWRAPPER_H_
>+#define SDK_ANDROID_SRC_JNI_PC_SSLCERTIFICATEVERIFIERWRAPPER_H_
>+
>+#include <jni.h>
>+#include <vector>
>+
>+#include "rtc_base/sslcertificate.h"
>+#include "sdk/android/src/jni/jni_helpers.h"
>+
>+namespace webrtc {
>+namespace jni {
>+
>+// Wrapper for Java SSLCertifiacteVerifier class. Delegates method calls through
>+// JNI and wraps the encoder inside SSLCertificateVerifierWrapper.
>+class SSLCertificateVerifierWrapper : public rtc::SSLCertificateVerifier {
>+ public:
>+  SSLCertificateVerifierWrapper(
>+      JNIEnv* jni,
>+      const JavaRef<jobject>& ssl_certificate_verifier);
>+  ~SSLCertificateVerifierWrapper() override;
>+
>+  bool Verify(const rtc::SSLCertificate& certificate) override;
>+
>+ private:
>+  const ScopedJavaGlobalRef<jobject> ssl_certificate_verifier_;
>+};
>+
>+}  // namespace jni
>+}  // namespace webrtc
>+
>+#endif  // SDK_ANDROID_SRC_JNI_PC_SSLCERTIFICATEVERIFIERWRAPPER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/statsobserver_jni.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/statsobserver.cc
>similarity index 62%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/statsobserver_jni.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/statsobserver.cc
>index f91a491bc1b..f30a34c34d6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/statsobserver_jni.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/statsobserver.cc
>@@ -8,10 +8,13 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "sdk/android/src/jni/pc/statsobserver_jni.h"
>+#include "sdk/android/src/jni/pc/statsobserver.h"
>+
>+#include <vector>
> 
> #include "sdk/android/generated_peerconnection_jni/jni/StatsObserver_jni.h"
> #include "sdk/android/generated_peerconnection_jni/jni/StatsReport_jni.h"
>+#include "sdk/android/native_api/jni/java_types.h"
> #include "sdk/android/src/jni/jni_helpers.h"
> 
> namespace webrtc {
>@@ -19,17 +22,17 @@ namespace jni {
> 
> namespace {
> 
>-jobject NativeToJavaStatsReportValue(
>+ScopedJavaLocalRef<jobject> NativeToJavaStatsReportValue(
>     JNIEnv* env,
>     const rtc::scoped_refptr<StatsReport::Value>& value_ptr) {
>   // Should we use the '.name' enum value here instead of converting the
>   // name to a string?
>-  jstring j_name = NativeToJavaString(env, value_ptr->display_name());
>-  jstring j_value = NativeToJavaString(env, value_ptr->ToString());
>-  return Java_Value_Constructor(env, j_name, j_value);
>+  return Java_Value_Constructor(
>+      env, NativeToJavaString(env, value_ptr->display_name()),
>+      NativeToJavaString(env, value_ptr->ToString()));
> }
> 
>-jobjectArray NativeToJavaStatsReportValueArray(
>+ScopedJavaLocalRef<jobjectArray> NativeToJavaStatsReportValueArray(
>     JNIEnv* env,
>     const StatsReport::Values& value_map) {
>   // Ignore the keys and make an array out of the values.
>@@ -41,29 +44,30 @@ jobjectArray NativeToJavaStatsReportValueArray(
>                                  &NativeToJavaStatsReportValue);
> }
> 
>-jobject NativeToJavaStatsReport(JNIEnv* env, const StatsReport& report) {
>-  jstring j_id = NativeToJavaString(env, report.id()->ToString());
>-  jstring j_type = NativeToJavaString(env, report.TypeToString());
>-  jobjectArray j_values =
>-      NativeToJavaStatsReportValueArray(env, report.values());
>-  return Java_StatsReport_Constructor(env, j_id, j_type, report.timestamp(),
>-                                      j_values);
>+ScopedJavaLocalRef<jobject> NativeToJavaStatsReport(JNIEnv* env,
>+                                                    const StatsReport& report) {
>+  return Java_StatsReport_Constructor(
>+      env, NativeToJavaString(env, report.id()->ToString()),
>+      NativeToJavaString(env, report.TypeToString()), report.timestamp(),
>+      NativeToJavaStatsReportValueArray(env, report.values()));
> }
> 
> }  // namespace
> 
>-StatsObserverJni::StatsObserverJni(JNIEnv* jni, jobject j_observer)
>+StatsObserverJni::StatsObserverJni(JNIEnv* jni,
>+                                   const JavaRef<jobject>& j_observer)
>     : j_observer_global_(jni, j_observer) {}
> 
>+StatsObserverJni::~StatsObserverJni() = default;
>+
> void StatsObserverJni::OnComplete(const StatsReports& reports) {
>   JNIEnv* env = AttachCurrentThreadIfNeeded();
>-  ScopedLocalRefFrame local_ref_frame(env);
>-  jobjectArray j_reports =
>+  ScopedJavaLocalRef<jobjectArray> j_reports =
>       NativeToJavaObjectArray(env, reports, org_webrtc_StatsReport_clazz(env),
>                               [](JNIEnv* env, const StatsReport* report) {
>                                 return NativeToJavaStatsReport(env, *report);
>                               });
>-  Java_StatsObserver_onComplete(env, *j_observer_global_, j_reports);
>+  Java_StatsObserver_onComplete(env, j_observer_global_, j_reports);
> }
> 
> }  // namespace jni
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/statsobserver_jni.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/statsobserver.h
>similarity index 73%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/statsobserver_jni.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/statsobserver.h
>index ec5d4fdd701..d21fb98abe0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/statsobserver_jni.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/statsobserver.h
>@@ -8,8 +8,8 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#ifndef SDK_ANDROID_SRC_JNI_PC_STATSOBSERVER_JNI_H_
>-#define SDK_ANDROID_SRC_JNI_PC_STATSOBSERVER_JNI_H_
>+#ifndef SDK_ANDROID_SRC_JNI_PC_STATSOBSERVER_H_
>+#define SDK_ANDROID_SRC_JNI_PC_STATSOBSERVER_H_
> 
> #include "api/peerconnectioninterface.h"
> #include "sdk/android/src/jni/jni_helpers.h"
>@@ -21,15 +21,16 @@ namespace jni {
> // dispatching the callback from C++ back to Java.
> class StatsObserverJni : public StatsObserver {
>  public:
>-  StatsObserverJni(JNIEnv* jni, jobject j_observer);
>+  StatsObserverJni(JNIEnv* jni, const JavaRef<jobject>& j_observer);
>+  ~StatsObserverJni() override;
> 
>   void OnComplete(const StatsReports& reports) override;
> 
>  private:
>-  const ScopedGlobalRef<jobject> j_observer_global_;
>+  const ScopedJavaGlobalRef<jobject> j_observer_global_;
> };
> 
> }  // namespace jni
> }  // namespace webrtc
> 
>-#endif  // SDK_ANDROID_SRC_JNI_PC_STATSOBSERVER_JNI_H_
>+#endif  // SDK_ANDROID_SRC_JNI_PC_STATSOBSERVER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/turncustomizer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/turncustomizer.cc
>new file mode 100644
>index 00000000000..3857a540ff6
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/turncustomizer.cc
>@@ -0,0 +1,36 @@
>+/*
>+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/turncustomizer.h"
>+#include "sdk/android/generated_peerconnection_jni/jni/TurnCustomizer_jni.h"
>+#include "sdk/android/native_api/jni/java_types.h"
>+#include "sdk/android/src/jni/jni_helpers.h"
>+
>+namespace webrtc {
>+namespace jni {
>+
>+TurnCustomizer* GetNativeTurnCustomizer(
>+    JNIEnv* env,
>+    const JavaRef<jobject>& j_turn_customizer) {
>+  if (IsNull(env, j_turn_customizer))
>+    return nullptr;
>+  return reinterpret_cast<webrtc::TurnCustomizer*>(
>+      Java_TurnCustomizer_getNativeTurnCustomizer(env, j_turn_customizer));
>+}
>+
>+static void JNI_TurnCustomizer_FreeTurnCustomizer(
>+    JNIEnv* jni,
>+    const JavaParamRef<jclass>&,
>+    jlong j_turn_customizer_pointer) {
>+  delete reinterpret_cast<TurnCustomizer*>(j_turn_customizer_pointer);
>+}
>+
>+}  // namespace jni
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/turncustomizer_jni.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/turncustomizer.h
>similarity index 60%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/turncustomizer_jni.cc
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/turncustomizer.h
>index f1c33db4742..ea6997310f8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/turncustomizer_jni.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/turncustomizer.h
>@@ -8,19 +8,20 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>+#ifndef SDK_ANDROID_SRC_JNI_PC_TURNCUSTOMIZER_H_
>+#define SDK_ANDROID_SRC_JNI_PC_TURNCUSTOMIZER_H_
>+
> #include "api/turncustomizer.h"
>-#include "sdk/android/src/jni/jni_helpers.h"
>+#include "sdk/android/native_api/jni/scoped_java_ref.h"
> 
> namespace webrtc {
> namespace jni {
> 
>-JNI_FUNCTION_DECLARATION(void,
>-                         TurnCustomizer_nativeFreeTurnCustomizer,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jlong j_turn_customizer_pointer) {
>-  delete reinterpret_cast<TurnCustomizer*>(j_turn_customizer_pointer);
>-}
>+TurnCustomizer* GetNativeTurnCustomizer(
>+    JNIEnv* env,
>+    const JavaRef<jobject>& j_turn_customizer);
> 
> }  // namespace jni
> }  // namespace webrtc
>+
>+#endif  // SDK_ANDROID_SRC_JNI_PC_TURNCUSTOMIZER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/video.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/video.cc
>new file mode 100644
>index 00000000000..cb1307a367c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/video.cc
>@@ -0,0 +1,52 @@
>+/*
>+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "sdk/android/src/jni/pc/video.h"
>+
>+#include <jni.h>
>+#include <memory>
>+
>+#include "api/video_codecs/video_decoder_factory.h"
>+#include "api/video_codecs/video_encoder_factory.h"
>+#include "api/videosourceproxy.h"
>+#include "media/engine/convert_legacy_video_factory.h"
>+#include "rtc_base/logging.h"
>+#include "sdk/android/src/jni/androidvideotracksource.h"
>+#include "sdk/android/src/jni/videodecoderfactorywrapper.h"
>+#include "sdk/android/src/jni/videoencoderfactorywrapper.h"
>+
>+namespace webrtc {
>+namespace jni {
>+
>+VideoEncoderFactory* CreateVideoEncoderFactory(
>+    JNIEnv* jni,
>+    const JavaRef<jobject>& j_encoder_factory) {
>+  return new VideoEncoderFactoryWrapper(jni, j_encoder_factory);
>+}
>+
>+VideoDecoderFactory* CreateVideoDecoderFactory(
>+    JNIEnv* jni,
>+    const JavaRef<jobject>& j_decoder_factory) {
>+  return new VideoDecoderFactoryWrapper(jni, j_decoder_factory);
>+}
>+
>+void* CreateVideoSource(JNIEnv* env,
>+                        rtc::Thread* signaling_thread,
>+                        rtc::Thread* worker_thread,
>+                        jboolean is_screencast) {
>+  rtc::scoped_refptr<AndroidVideoTrackSource> source(
>+      new rtc::RefCountedObject<AndroidVideoTrackSource>(signaling_thread, env,
>+                                                         is_screencast));
>+  return VideoTrackSourceProxy::Create(signaling_thread, worker_thread, source)
>+      .release();
>+}
>+
>+}  // namespace jni
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/video.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/video.h
>new file mode 100644
>index 00000000000..183770b8206
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/video.h
>@@ -0,0 +1,44 @@
>+/*
>+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef SDK_ANDROID_SRC_JNI_PC_VIDEO_H_
>+#define SDK_ANDROID_SRC_JNI_PC_VIDEO_H_
>+
>+#include <jni.h>
>+
>+#include "rtc_base/scoped_ref_ptr.h"
>+#include "rtc_base/thread.h"
>+#include "sdk/android/native_api/jni/scoped_java_ref.h"
>+
>+namespace webrtc {
>+class VideoEncoderFactory;
>+class VideoDecoderFactory;
>+}  // namespace webrtc
>+
>+namespace webrtc {
>+namespace jni {
>+
>+VideoEncoderFactory* CreateVideoEncoderFactory(
>+    JNIEnv* jni,
>+    const JavaRef<jobject>& j_encoder_factory);
>+
>+VideoDecoderFactory* CreateVideoDecoderFactory(
>+    JNIEnv* jni,
>+    const JavaRef<jobject>& j_decoder_factory);
>+
>+void* CreateVideoSource(JNIEnv* env,
>+                        rtc::Thread* signaling_thread,
>+                        rtc::Thread* worker_thread,
>+                        jboolean is_screencast);
>+
>+}  // namespace jni
>+}  // namespace webrtc
>+
>+#endif  // SDK_ANDROID_SRC_JNI_PC_VIDEO_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/video_jni.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/video_jni.cc
>deleted file mode 100644
>index f7f60ff6597..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/video_jni.cc
>+++ /dev/null
>@@ -1,146 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include <jni.h>
>-
>-#include "api/video_codecs/video_decoder_factory.h"
>-#include "api/video_codecs/video_encoder_factory.h"
>-#include "api/videosourceproxy.h"
>-#include "media/engine/convert_legacy_video_factory.h"
>-#include "media/engine/webrtcvideodecoderfactory.h"
>-#include "media/engine/webrtcvideoencoderfactory.h"
>-#include "rtc_base/logging.h"
>-#include "sdk/android/src/jni/androidmediadecoder_jni.h"
>-#include "sdk/android/src/jni/androidmediaencoder_jni.h"
>-#include "sdk/android/src/jni/androidvideotracksource.h"
>-#include "sdk/android/src/jni/classreferenceholder.h"
>-#include "sdk/android/src/jni/pc/ownedfactoryandthreads.h"
>-#include "sdk/android/src/jni/surfacetexturehelper_jni.h"
>-#include "sdk/android/src/jni/videodecoderfactorywrapper.h"
>-#include "sdk/android/src/jni/videoencoderfactorywrapper.h"
>-
>-namespace webrtc {
>-namespace jni {
>-
>-VideoEncoderFactory* CreateVideoEncoderFactory(JNIEnv* jni,
>-                                               jobject j_encoder_factory) {
>-  return new VideoEncoderFactoryWrapper(jni, j_encoder_factory);
>-}
>-
>-VideoDecoderFactory* CreateVideoDecoderFactory(JNIEnv* jni,
>-                                               jobject j_decoder_factory) {
>-  return new VideoDecoderFactoryWrapper(jni, j_decoder_factory);
>-}
>-
>-cricket::WebRtcVideoEncoderFactory* CreateLegacyVideoEncoderFactory() {
>-  return new MediaCodecVideoEncoderFactory();
>-}
>-
>-cricket::WebRtcVideoDecoderFactory* CreateLegacyVideoDecoderFactory() {
>-  return new MediaCodecVideoDecoderFactory();
>-}
>-
>-VideoEncoderFactory* WrapLegacyVideoEncoderFactory(
>-    cricket::WebRtcVideoEncoderFactory* legacy_encoder_factory) {
>-  return ConvertVideoEncoderFactory(
>-             std::unique_ptr<cricket::WebRtcVideoEncoderFactory>(
>-                 legacy_encoder_factory))
>-      .release();
>-}
>-
>-VideoDecoderFactory* WrapLegacyVideoDecoderFactory(
>-    cricket::WebRtcVideoDecoderFactory* legacy_decoder_factory) {
>-  return ConvertVideoDecoderFactory(
>-             std::unique_ptr<cricket::WebRtcVideoDecoderFactory>(
>-                 legacy_decoder_factory))
>-      .release();
>-}
>-
>-jobject GetJavaSurfaceTextureHelper(
>-    const rtc::scoped_refptr<SurfaceTextureHelper>& surface_texture_helper) {
>-  return surface_texture_helper
>-             ? surface_texture_helper->GetJavaSurfaceTextureHelper()
>-             : nullptr;
>-}
>-
>-JNI_FUNCTION_DECLARATION(jlong,
>-                         PeerConnectionFactory_nativeCreateVideoSource,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jlong native_factory,
>-                         jobject j_surface_texture_helper,
>-                         jboolean is_screencast) {
>-  OwnedFactoryAndThreads* factory =
>-      reinterpret_cast<OwnedFactoryAndThreads*>(native_factory);
>-
>-  rtc::scoped_refptr<AndroidVideoTrackSource> source(
>-      new rtc::RefCountedObject<AndroidVideoTrackSource>(
>-          factory->signaling_thread(), jni, j_surface_texture_helper,
>-          is_screencast));
>-  rtc::scoped_refptr<VideoTrackSourceProxy> proxy_source =
>-      VideoTrackSourceProxy::Create(factory->signaling_thread(),
>-                                    factory->worker_thread(), source);
>-
>-  return (jlong)proxy_source.release();
>-}
>-
>-JNI_FUNCTION_DECLARATION(jlong,
>-                         PeerConnectionFactory_nativeCreateVideoTrack,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jlong native_factory,
>-                         jstring id,
>-                         jlong native_source) {
>-  rtc::scoped_refptr<PeerConnectionFactoryInterface> factory(
>-      factoryFromJava(native_factory));
>-  rtc::scoped_refptr<VideoTrackInterface> track(factory->CreateVideoTrack(
>-      JavaToStdString(jni, id),
>-      reinterpret_cast<VideoTrackSourceInterface*>(native_source)));
>-  return (jlong)track.release();
>-}
>-
>-JNI_FUNCTION_DECLARATION(
>-    void,
>-    PeerConnectionFactory_nativeSetVideoHwAccelerationOptions,
>-    JNIEnv* jni,
>-    jclass,
>-    jlong native_factory,
>-    jobject local_egl_context,
>-    jobject remote_egl_context) {
>-  OwnedFactoryAndThreads* owned_factory =
>-      reinterpret_cast<OwnedFactoryAndThreads*>(native_factory);
>-
>-  jclass j_eglbase14_context_class =
>-      FindClass(jni, "org/webrtc/EglBase14$Context");
>-
>-  if (owned_factory->legacy_encoder_factory()) {
>-    MediaCodecVideoEncoderFactory* encoder_factory =
>-        static_cast<MediaCodecVideoEncoderFactory*>(
>-            owned_factory->legacy_encoder_factory());
>-    if (encoder_factory &&
>-        jni->IsInstanceOf(local_egl_context, j_eglbase14_context_class)) {
>-      RTC_LOG(LS_INFO) << "Set EGL context for HW encoding.";
>-      encoder_factory->SetEGLContext(jni, local_egl_context);
>-    }
>-  }
>-
>-  if (owned_factory->legacy_decoder_factory()) {
>-    MediaCodecVideoDecoderFactory* decoder_factory =
>-        static_cast<MediaCodecVideoDecoderFactory*>(
>-            owned_factory->legacy_decoder_factory());
>-    if (decoder_factory) {
>-      RTC_LOG(LS_INFO) << "Set EGL context for HW decoding.";
>-      decoder_factory->SetEGLContext(jni, remote_egl_context);
>-    }
>-  }
>-}
>-
>-}  // namespace jni
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/video_jni.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/video_jni.h
>deleted file mode 100644
>index a73c093d11c..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/pc/video_jni.h
>+++ /dev/null
>@@ -1,53 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef SDK_ANDROID_SRC_JNI_PC_VIDEO_JNI_H_
>-#define SDK_ANDROID_SRC_JNI_PC_VIDEO_JNI_H_
>-
>-#include <jni.h>
>-
>-#include "rtc_base/scoped_ref_ptr.h"
>-
>-namespace cricket {
>-class WebRtcVideoEncoderFactory;
>-class WebRtcVideoDecoderFactory;
>-}  // namespace cricket
>-
>-namespace webrtc {
>-class VideoEncoderFactory;
>-class VideoDecoderFactory;
>-}  // namespace webrtc
>-
>-namespace webrtc {
>-namespace jni {
>-
>-class SurfaceTextureHelper;
>-
>-VideoEncoderFactory* CreateVideoEncoderFactory(JNIEnv* jni,
>-                                               jobject j_encoder_factory);
>-
>-VideoDecoderFactory* CreateVideoDecoderFactory(JNIEnv* jni,
>-                                               jobject j_decoder_factory);
>-
>-cricket::WebRtcVideoEncoderFactory* CreateLegacyVideoEncoderFactory();
>-cricket::WebRtcVideoDecoderFactory* CreateLegacyVideoDecoderFactory();
>-
>-VideoEncoderFactory* WrapLegacyVideoEncoderFactory(
>-    cricket::WebRtcVideoEncoderFactory* legacy_encoder_factory);
>-VideoDecoderFactory* WrapLegacyVideoDecoderFactory(
>-    cricket::WebRtcVideoDecoderFactory* legacy_decoder_factory);
>-
>-jobject GetJavaSurfaceTextureHelper(
>-    const rtc::scoped_refptr<SurfaceTextureHelper>& surface_texture_helper);
>-
>-}  // namespace jni
>-}  // namespace webrtc
>-
>-#endif  // SDK_ANDROID_SRC_JNI_PC_VIDEO_JNI_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/surfacetexturehelper_jni.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/surfacetexturehelper_jni.cc
>deleted file mode 100644
>index 0b7cf4ebaf1..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/surfacetexturehelper_jni.cc
>+++ /dev/null
>@@ -1,64 +0,0 @@
>-/*
>- *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-
>-#include "sdk/android/src/jni/surfacetexturehelper_jni.h"
>-
>-#include "rtc_base/bind.h"
>-#include "rtc_base/logging.h"
>-#include "sdk/android/generated_video_jni/jni/SurfaceTextureHelper_jni.h"
>-
>-namespace webrtc {
>-namespace jni {
>-
>-rtc::scoped_refptr<SurfaceTextureHelper> SurfaceTextureHelper::create(
>-    JNIEnv* jni,
>-    const char* thread_name,
>-    jobject j_egl_context) {
>-  jobject j_surface_texture_helper = Java_SurfaceTextureHelper_create(
>-      jni, jni->NewStringUTF(thread_name), j_egl_context);
>-  CHECK_EXCEPTION(jni)
>-      << "error during initialization of Java SurfaceTextureHelper";
>-  if (IsNull(jni, j_surface_texture_helper))
>-    return nullptr;
>-  return new rtc::RefCountedObject<SurfaceTextureHelper>(
>-      jni, j_surface_texture_helper);
>-}
>-
>-SurfaceTextureHelper::SurfaceTextureHelper(JNIEnv* jni,
>-                                           jobject j_surface_texture_helper)
>-    : j_surface_texture_helper_(jni, j_surface_texture_helper) {}
>-
>-SurfaceTextureHelper::~SurfaceTextureHelper() {
>-  RTC_LOG(LS_INFO) << "SurfaceTextureHelper dtor";
>-  JNIEnv* jni = AttachCurrentThreadIfNeeded();
>-  Java_SurfaceTextureHelper_dispose(jni, *j_surface_texture_helper_);
>-}
>-
>-jobject SurfaceTextureHelper::GetJavaSurfaceTextureHelper() const {
>-  return *j_surface_texture_helper_;
>-}
>-
>-void SurfaceTextureHelper::ReturnTextureFrame() const {
>-  JNIEnv* jni = AttachCurrentThreadIfNeeded();
>-  Java_SurfaceTextureHelper_returnTextureFrame(jni, *j_surface_texture_helper_);
>-}
>-
>-rtc::scoped_refptr<VideoFrameBuffer> SurfaceTextureHelper::CreateTextureFrame(
>-    int width,
>-    int height,
>-    const NativeHandleImpl& native_handle) {
>-  return new rtc::RefCountedObject<AndroidTextureBuffer>(
>-      width, height, native_handle, *j_surface_texture_helper_,
>-      rtc::Bind(&SurfaceTextureHelper::ReturnTextureFrame, this));
>-}
>-
>-}  // namespace jni
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/surfacetexturehelper_jni.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/surfacetexturehelper_jni.h
>deleted file mode 100644
>index b227b52cc02..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/surfacetexturehelper_jni.h
>+++ /dev/null
>@@ -1,69 +0,0 @@
>-/*
>- *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef SDK_ANDROID_SRC_JNI_SURFACETEXTUREHELPER_JNI_H_
>-#define SDK_ANDROID_SRC_JNI_SURFACETEXTUREHELPER_JNI_H_
>-
>-#include <jni.h>
>-
>-#include "api/video/video_frame_buffer.h"
>-#include "rtc_base/refcount.h"
>-#include "rtc_base/scoped_ref_ptr.h"
>-#include "sdk/android/src/jni/jni_helpers.h"
>-#include "sdk/android/src/jni/videoframe.h"
>-
>-namespace webrtc {
>-namespace jni {
>-
>-// Helper class to create and synchronize access to an Android SurfaceTexture.
>-// It is used for creating VideoFrameBuffers from a SurfaceTexture when
>-// the SurfaceTexture has been updated.
>-// When the VideoFrameBuffer is released, this class returns the buffer to the
>-// java SurfaceTextureHelper so it can be updated safely. The VideoFrameBuffer
>-// can be released on an arbitrary thread.
>-// SurfaceTextureHelper is reference counted to make sure that it is not
>-// destroyed while a VideoFrameBuffer is in use.
>-// This class is the C++ counterpart of the java class SurfaceTextureHelper.
>-// It owns the corresponding java object, and calls the java dispose
>-// method when destroyed.
>-// Usage:
>-// 1. Create an instance of this class.
>-// 2. Get the Java SurfaceTextureHelper with GetJavaSurfaceTextureHelper().
>-// 3. Register a listener to the Java SurfaceListener and start producing
>-// new buffers.
>-// 4. Call CreateTextureFrame to wrap the Java texture in a VideoFrameBuffer.
>-class SurfaceTextureHelper : public rtc::RefCountInterface {
>- public:
>-  // Might return null if creating the Java SurfaceTextureHelper fails.
>-  static rtc::scoped_refptr<SurfaceTextureHelper> create(
>-      JNIEnv* jni, const char* thread_name, jobject j_egl_context);
>-
>-  jobject GetJavaSurfaceTextureHelper() const;
>-
>-  rtc::scoped_refptr<VideoFrameBuffer> CreateTextureFrame(
>-      int width,
>-      int height,
>-      const NativeHandleImpl& native_handle);
>-
>-  // May be called on arbitrary thread.
>-  void ReturnTextureFrame() const;
>-
>- protected:
>-  ~SurfaceTextureHelper();
>-  SurfaceTextureHelper(JNIEnv* jni, jobject j_surface_texture_helper);
>-
>- private:
>-  const ScopedGlobalRef<jobject> j_surface_texture_helper_;
>-};
>-
>-}  // namespace jni
>-}  // namespace webrtc
>-
>-#endif  // SDK_ANDROID_SRC_JNI_SURFACETEXTUREHELPER_JNI_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/video_renderer_jni.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/video_renderer_jni.cc
>deleted file mode 100644
>index ba439b9de1d..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/video_renderer_jni.cc
>+++ /dev/null
>@@ -1,198 +0,0 @@
>-/*
>- *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include <jni.h>
>-
>-#include "api/video/video_frame.h"
>-#include "media/base/videosinkinterface.h"
>-#include "sdk/android/src/jni/classreferenceholder.h"
>-#include "sdk/android/src/jni/jni_helpers.h"
>-#include "sdk/android/src/jni/videoframe.h"
>-
>-namespace webrtc {
>-namespace jni {
>-
>-// Wrapper dispatching rtc::VideoSinkInterface to a Java VideoRenderer
>-// instance.
>-class JavaVideoRendererWrapper : public rtc::VideoSinkInterface<VideoFrame> {
>- public:
>-  JavaVideoRendererWrapper(JNIEnv* jni, jobject j_callbacks)
>-      : j_callbacks_(jni, j_callbacks),
>-        j_render_frame_id_(
>-            GetMethodID(jni,
>-                        GetObjectClass(jni, j_callbacks),
>-                        "renderFrame",
>-                        "(Lorg/webrtc/VideoRenderer$I420Frame;)V")),
>-        j_frame_class_(jni,
>-                       FindClass(jni, "org/webrtc/VideoRenderer$I420Frame")),
>-        j_i420_frame_ctor_id_(GetMethodID(jni,
>-                                          *j_frame_class_,
>-                                          "<init>",
>-                                          "(III[I[Ljava/nio/ByteBuffer;J)V")),
>-        j_texture_frame_ctor_id_(
>-            GetMethodID(jni, *j_frame_class_, "<init>", "(IIII[FJ)V")),
>-        j_byte_buffer_class_(jni, FindClass(jni, "java/nio/ByteBuffer")) {
>-    CHECK_EXCEPTION(jni);
>-  }
>-
>-  virtual ~JavaVideoRendererWrapper() {}
>-
>-  void OnFrame(const VideoFrame& video_frame) override {
>-    ScopedLocalRefFrame local_ref_frame(jni());
>-
>-    jobject j_frame;
>-    if (video_frame.video_frame_buffer()->type() ==
>-        VideoFrameBuffer::Type::kNative) {
>-      AndroidVideoFrameBuffer* android_buffer =
>-          static_cast<AndroidVideoFrameBuffer*>(
>-              video_frame.video_frame_buffer().get());
>-      switch (android_buffer->android_type()) {
>-        case AndroidVideoFrameBuffer::AndroidType::kTextureBuffer:
>-          j_frame = ToJavaTextureFrame(&video_frame);
>-          break;
>-        case AndroidVideoFrameBuffer::AndroidType::kJavaBuffer:
>-          j_frame = static_cast<AndroidVideoBuffer*>(android_buffer)
>-                        ->ToJavaI420Frame(jni(), video_frame.rotation());
>-          break;
>-        default:
>-          RTC_NOTREACHED();
>-      }
>-    } else {
>-      j_frame = ToJavaI420Frame(&video_frame);
>-    }
>-    // |j_callbacks_| is responsible for releasing |j_frame| with
>-    // VideoRenderer.renderFrameDone().
>-    jni()->CallVoidMethod(*j_callbacks_, j_render_frame_id_, j_frame);
>-    CHECK_EXCEPTION(jni());
>-  }
>-
>- private:
>-  // Make a shallow copy of |frame| to be used with Java. The callee has
>-  // ownership of the frame, and the frame should be released with
>-  // VideoRenderer.releaseNativeFrame().
>-  static jlong javaShallowCopy(const VideoFrame* frame) {
>-    return jlongFromPointer(new VideoFrame(*frame));
>-  }
>-
>-  // Return a VideoRenderer.I420Frame referring to the data in |frame|.
>-  jobject ToJavaI420Frame(const VideoFrame* frame) {
>-    jintArray strides = jni()->NewIntArray(3);
>-    jint* strides_array = jni()->GetIntArrayElements(strides, NULL);
>-    rtc::scoped_refptr<I420BufferInterface> i420_buffer =
>-        frame->video_frame_buffer()->ToI420();
>-    strides_array[0] = i420_buffer->StrideY();
>-    strides_array[1] = i420_buffer->StrideU();
>-    strides_array[2] = i420_buffer->StrideV();
>-    jni()->ReleaseIntArrayElements(strides, strides_array, 0);
>-    jobjectArray planes = jni()->NewObjectArray(3, *j_byte_buffer_class_, NULL);
>-    jobject y_buffer = jni()->NewDirectByteBuffer(
>-        const_cast<uint8_t*>(i420_buffer->DataY()),
>-        i420_buffer->StrideY() * i420_buffer->height());
>-    size_t chroma_height = i420_buffer->ChromaHeight();
>-    jobject u_buffer =
>-        jni()->NewDirectByteBuffer(const_cast<uint8_t*>(i420_buffer->DataU()),
>-                                   i420_buffer->StrideU() * chroma_height);
>-    jobject v_buffer =
>-        jni()->NewDirectByteBuffer(const_cast<uint8_t*>(i420_buffer->DataV()),
>-                                   i420_buffer->StrideV() * chroma_height);
>-
>-    jni()->SetObjectArrayElement(planes, 0, y_buffer);
>-    jni()->SetObjectArrayElement(planes, 1, u_buffer);
>-    jni()->SetObjectArrayElement(planes, 2, v_buffer);
>-    return jni()->NewObject(*j_frame_class_, j_i420_frame_ctor_id_,
>-                            frame->width(), frame->height(),
>-                            static_cast<int>(frame->rotation()), strides,
>-                            planes, javaShallowCopy(frame));
>-  }
>-
>-  // Return a VideoRenderer.I420Frame referring texture object in |frame|.
>-  jobject ToJavaTextureFrame(const VideoFrame* frame) {
>-    NativeHandleImpl handle =
>-        static_cast<AndroidTextureBuffer*>(frame->video_frame_buffer().get())
>-            ->native_handle_impl();
>-    jfloatArray sampling_matrix = handle.sampling_matrix.ToJava(jni());
>-
>-    return jni()->NewObject(
>-        *j_frame_class_, j_texture_frame_ctor_id_, frame->width(),
>-        frame->height(), static_cast<int>(frame->rotation()),
>-        handle.oes_texture_id, sampling_matrix, javaShallowCopy(frame));
>-  }
>-
>-  JNIEnv* jni() { return AttachCurrentThreadIfNeeded(); }
>-
>-  ScopedGlobalRef<jobject> j_callbacks_;
>-  jmethodID j_render_frame_id_;
>-  ScopedGlobalRef<jclass> j_frame_class_;
>-  jmethodID j_i420_frame_ctor_id_;
>-  jmethodID j_texture_frame_ctor_id_;
>-  ScopedGlobalRef<jclass> j_byte_buffer_class_;
>-};
>-
>-JNI_FUNCTION_DECLARATION(void,
>-                         VideoRenderer_freeWrappedVideoRenderer,
>-                         JNIEnv*,
>-                         jclass,
>-                         jlong j_p) {
>-  delete reinterpret_cast<JavaVideoRendererWrapper*>(j_p);
>-}
>-
>-JNI_FUNCTION_DECLARATION(void,
>-                         VideoRenderer_releaseNativeFrame,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jlong j_frame_ptr) {
>-  delete reinterpret_cast<const VideoFrame*>(j_frame_ptr);
>-}
>-
>-JNI_FUNCTION_DECLARATION(jlong,
>-                         VideoRenderer_nativeWrapVideoRenderer,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jobject j_callbacks) {
>-  std::unique_ptr<JavaVideoRendererWrapper> renderer(
>-      new JavaVideoRendererWrapper(jni, j_callbacks));
>-  return (jlong)renderer.release();
>-}
>-
>-JNI_FUNCTION_DECLARATION(void,
>-                         VideoRenderer_nativeCopyPlane,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jobject j_src_buffer,
>-                         jint width,
>-                         jint height,
>-                         jint src_stride,
>-                         jobject j_dst_buffer,
>-                         jint dst_stride) {
>-  size_t src_size = jni->GetDirectBufferCapacity(j_src_buffer);
>-  size_t dst_size = jni->GetDirectBufferCapacity(j_dst_buffer);
>-  RTC_CHECK(src_stride >= width) << "Wrong source stride " << src_stride;
>-  RTC_CHECK(dst_stride >= width) << "Wrong destination stride " << dst_stride;
>-  RTC_CHECK(src_size >= src_stride * height)
>-      << "Insufficient source buffer capacity " << src_size;
>-  RTC_CHECK(dst_size >= dst_stride * height)
>-      << "Insufficient destination buffer capacity " << dst_size;
>-  uint8_t* src =
>-      reinterpret_cast<uint8_t*>(jni->GetDirectBufferAddress(j_src_buffer));
>-  uint8_t* dst =
>-      reinterpret_cast<uint8_t*>(jni->GetDirectBufferAddress(j_dst_buffer));
>-  if (src_stride == dst_stride) {
>-    memcpy(dst, src, src_stride * height);
>-  } else {
>-    for (int i = 0; i < height; i++) {
>-      memcpy(dst, src, width);
>-      src += src_stride;
>-      dst += dst_stride;
>-    }
>-  }
>-}
>-
>-}  // namespace jni
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videocodecinfo.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videocodecinfo.cc
>index 2412649199c..20a641d1392 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videocodecinfo.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videocodecinfo.cc
>@@ -10,47 +10,27 @@
> 
> #include "sdk/android/src/jni/videocodecinfo.h"
> 
>-#include "sdk/android/src/jni/classreferenceholder.h"
>+#include "sdk/android/generated_video_jni/jni/VideoCodecInfo_jni.h"
>+#include "sdk/android/native_api/jni/java_types.h"
> #include "sdk/android/src/jni/jni_helpers.h"
> 
> namespace webrtc {
> namespace jni {
> 
>-SdpVideoFormat VideoCodecInfoToSdpVideoFormat(JNIEnv* jni, jobject j_info) {
>-  jclass video_codec_info_class = FindClass(jni, "org/webrtc/VideoCodecInfo");
>-  jfieldID name_field =
>-      GetFieldID(jni, video_codec_info_class, "name", "Ljava/lang/String;");
>-  jfieldID params_field =
>-      GetFieldID(jni, video_codec_info_class, "params", "Ljava/util/Map;");
>-
>-  jobject j_params = jni->GetObjectField(j_info, params_field);
>-  jstring j_name =
>-      static_cast<jstring>(jni->GetObjectField(j_info, name_field));
>-
>-  return SdpVideoFormat(JavaToStdString(jni, j_name),
>-                        JavaToStdMapStrings(jni, j_params));
>+SdpVideoFormat VideoCodecInfoToSdpVideoFormat(JNIEnv* jni,
>+                                              const JavaRef<jobject>& j_info) {
>+  return SdpVideoFormat(
>+      JavaToNativeString(jni, Java_VideoCodecInfo_getName(jni, j_info)),
>+      JavaToNativeStringMap(jni, Java_VideoCodecInfo_getParams(jni, j_info)));
> }
> 
>-jobject SdpVideoFormatToVideoCodecInfo(JNIEnv* jni,
>-                                       const SdpVideoFormat& format) {
>-  jclass hash_map_class = jni->FindClass("java/util/HashMap");
>-  jmethodID hash_map_constructor =
>-      jni->GetMethodID(hash_map_class, "<init>", "()V");
>-  jmethodID put_method = jni->GetMethodID(
>-      hash_map_class, "put",
>-      "(Ljava/lang/Object;Ljava/lang/Object;)Ljava/lang/Object;");
>-  jclass video_codec_info_class = FindClass(jni, "org/webrtc/VideoCodecInfo");
>-  jmethodID video_codec_info_constructor = jni->GetMethodID(
>-      video_codec_info_class, "<init>", "(Ljava/lang/String;Ljava/util/Map;)V");
>-
>-  jobject j_params = jni->NewObject(hash_map_class, hash_map_constructor);
>-  for (auto const& param : format.parameters) {
>-    jni->CallObjectMethod(j_params, put_method,
>-                          NativeToJavaString(jni, param.first),
>-                          NativeToJavaString(jni, param.second));
>-  }
>-  return jni->NewObject(video_codec_info_class, video_codec_info_constructor,
>-                        NativeToJavaString(jni, format.name), j_params);
>+ScopedJavaLocalRef<jobject> SdpVideoFormatToVideoCodecInfo(
>+    JNIEnv* jni,
>+    const SdpVideoFormat& format) {
>+  ScopedJavaLocalRef<jobject> j_params =
>+      NativeToJavaStringMap(jni, format.parameters);
>+  return Java_VideoCodecInfo_Constructor(
>+      jni, NativeToJavaString(jni, format.name), j_params);
> }
> 
> }  // namespace jni
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videocodecinfo.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videocodecinfo.h
>index ea72273a55a..2829c02b07a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videocodecinfo.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videocodecinfo.h
>@@ -14,13 +14,16 @@
> #include <jni.h>
> 
> #include "api/video_codecs/sdp_video_format.h"
>+#include "sdk/android/src/jni/jni_helpers.h"
> 
> namespace webrtc {
> namespace jni {
> 
>-SdpVideoFormat VideoCodecInfoToSdpVideoFormat(JNIEnv* jni, jobject info);
>-jobject SdpVideoFormatToVideoCodecInfo(JNIEnv* jni,
>-                                       const SdpVideoFormat& format);
>+SdpVideoFormat VideoCodecInfoToSdpVideoFormat(JNIEnv* jni,
>+                                              const JavaRef<jobject>& info);
>+ScopedJavaLocalRef<jobject> SdpVideoFormatToVideoCodecInfo(
>+    JNIEnv* jni,
>+    const SdpVideoFormat& format);
> 
> }  // namespace jni
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videocodecstatus.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videocodecstatus.cc
>index 6641f30ac40..b4a0a525f2a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videocodecstatus.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videocodecstatus.cc
>@@ -15,8 +15,9 @@
> namespace webrtc {
> namespace jni {
> 
>-int32_t JavaToNativeVideoCodecStatus(JNIEnv* env,
>-                                     jobject j_video_codec_status) {
>+int32_t JavaToNativeVideoCodecStatus(
>+    JNIEnv* env,
>+    const JavaRef<jobject>& j_video_codec_status) {
>   return Java_VideoCodecStatus_getNumber(env, j_video_codec_status);
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videocodecstatus.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videocodecstatus.h
>index a2ad39ed737..2804654e414 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videocodecstatus.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videocodecstatus.h
>@@ -14,9 +14,13 @@
> #include <jni.h>
> #include <stdint.h>
> 
>+#include "sdk/android/native_api/jni/scoped_java_ref.h"
>+
> namespace webrtc {
> namespace jni {
>-int32_t JavaToNativeVideoCodecStatus(JNIEnv* env, jobject j_video_codec_status);
>+int32_t JavaToNativeVideoCodecStatus(
>+    JNIEnv* env,
>+    const JavaRef<jobject>& j_video_codec_status);
> }  // namespace jni
> }  // namespace webrtc
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videodecoderfactorywrapper.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videodecoderfactorywrapper.cc
>index e687289ad39..94b44676e1f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videodecoderfactorywrapper.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videodecoderfactorywrapper.cc
>@@ -14,34 +14,38 @@
> #include "api/video_codecs/video_decoder.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "rtc_base/logging.h"
>-#include "sdk/android/src/jni/wrappednativecodec.h"
>+#include "sdk/android/generated_video_jni/jni/VideoDecoderFactory_jni.h"
>+#include "sdk/android/native_api/jni/java_types.h"
>+#include "sdk/android/src/jni/videocodecinfo.h"
>+#include "sdk/android/src/jni/videodecoderwrapper.h"
> 
> namespace webrtc {
> namespace jni {
> 
>-VideoDecoderFactoryWrapper::VideoDecoderFactoryWrapper(JNIEnv* jni,
>-                                                       jobject decoder_factory)
>-    : decoder_factory_(jni, decoder_factory) {
>-  jclass decoder_factory_class = jni->GetObjectClass(*decoder_factory_);
>-  create_decoder_method_ =
>-      jni->GetMethodID(decoder_factory_class, "createDecoder",
>-                       "(Ljava/lang/String;)Lorg/webrtc/VideoDecoder;");
>-}
>+VideoDecoderFactoryWrapper::VideoDecoderFactoryWrapper(
>+    JNIEnv* jni,
>+    const JavaRef<jobject>& decoder_factory)
>+    : decoder_factory_(jni, decoder_factory) {}
>+VideoDecoderFactoryWrapper::~VideoDecoderFactoryWrapper() = default;
> 
> std::unique_ptr<VideoDecoder> VideoDecoderFactoryWrapper::CreateVideoDecoder(
>     const SdpVideoFormat& format) {
>   JNIEnv* jni = AttachCurrentThreadIfNeeded();
>-  ScopedLocalRefFrame local_ref_frame(jni);
>-  jstring name = NativeToJavaString(jni, format.name);
>-  jobject decoder =
>-      jni->CallObjectMethod(*decoder_factory_, create_decoder_method_, name);
>-  return decoder != nullptr ? JavaToNativeVideoDecoder(jni, decoder) : nullptr;
>+  ScopedJavaLocalRef<jobject> j_codec_info =
>+      SdpVideoFormatToVideoCodecInfo(jni, format);
>+  ScopedJavaLocalRef<jobject> decoder = Java_VideoDecoderFactory_createDecoder(
>+      jni, decoder_factory_, j_codec_info);
>+  if (!decoder.obj())
>+    return nullptr;
>+  return JavaToNativeVideoDecoder(jni, decoder);
> }
> 
> std::vector<SdpVideoFormat> VideoDecoderFactoryWrapper::GetSupportedFormats()
>     const {
>-  // TODO(andersc): VideoDecoderFactory.java does not have this method.
>-  return std::vector<SdpVideoFormat>();
>+  JNIEnv* env = AttachCurrentThreadIfNeeded();
>+  return JavaToNativeVector<SdpVideoFormat>(
>+      env, Java_VideoDecoderFactory_getSupportedCodecs(env, decoder_factory_),
>+      &VideoCodecInfoToSdpVideoFormat);
> }
> 
> }  // namespace jni
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videodecoderfactorywrapper.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videodecoderfactorywrapper.h
>index 3ba5ab69a00..a1716b10ba6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videodecoderfactorywrapper.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videodecoderfactorywrapper.h
>@@ -23,15 +23,16 @@ namespace jni {
> // JNI and wraps the decoder inside VideoDecoderWrapper.
> class VideoDecoderFactoryWrapper : public VideoDecoderFactory {
>  public:
>-  VideoDecoderFactoryWrapper(JNIEnv* jni, jobject decoder_factory);
>+  VideoDecoderFactoryWrapper(JNIEnv* jni,
>+                             const JavaRef<jobject>& decoder_factory);
>+  ~VideoDecoderFactoryWrapper() override;
> 
>   std::vector<SdpVideoFormat> GetSupportedFormats() const override;
>   std::unique_ptr<VideoDecoder> CreateVideoDecoder(
>       const SdpVideoFormat& format) override;
> 
>  private:
>-  const ScopedGlobalRef<jobject> decoder_factory_;
>-  jmethodID create_decoder_method_;
>+  const ScopedJavaGlobalRef<jobject> decoder_factory_;
> };
> 
> }  // namespace jni
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videodecoderfallback.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videodecoderfallback.cc
>index 92ad6ad9a5f..abb042ff661 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videodecoderfallback.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videodecoderfallback.cc
>@@ -10,27 +10,28 @@
> 
> #include <jni.h>
> 
>-#include "media/engine/videodecodersoftwarefallbackwrapper.h"
>+#include "api/video_codecs/video_decoder_software_fallback_wrapper.h"
>+#include "sdk/android/generated_video_jni/jni/VideoDecoderFallback_jni.h"
> #include "sdk/android/src/jni/jni_helpers.h"
>-#include "sdk/android/src/jni/wrappednativecodec.h"
>+#include "sdk/android/src/jni/videodecoderwrapper.h"
> 
> namespace webrtc {
> namespace jni {
> 
>-JNI_FUNCTION_DECLARATION(jlong,
>-                         VideoDecoderFallback_createNativeDecoder,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jobject j_fallback_decoder,
>-                         jobject j_primary_decoder) {
>+static jlong JNI_VideoDecoderFallback_CreateDecoder(
>+    JNIEnv* jni,
>+    const JavaParamRef<jclass>&,
>+    const JavaParamRef<jobject>& j_fallback_decoder,
>+    const JavaParamRef<jobject>& j_primary_decoder) {
>   std::unique_ptr<VideoDecoder> fallback_decoder =
>       JavaToNativeVideoDecoder(jni, j_fallback_decoder);
>   std::unique_ptr<VideoDecoder> primary_decoder =
>       JavaToNativeVideoDecoder(jni, j_primary_decoder);
> 
>-  VideoDecoderSoftwareFallbackWrapper* nativeWrapper =
>-      new VideoDecoderSoftwareFallbackWrapper(std::move(fallback_decoder),
>-                                              std::move(primary_decoder));
>+  VideoDecoder* nativeWrapper =
>+      CreateVideoDecoderSoftwareFallbackWrapper(std::move(fallback_decoder),
>+                                                std::move(primary_decoder))
>+          .release();
> 
>   return jlongFromPointer(nativeWrapper);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videodecoderwrapper.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videodecoderwrapper.cc
>index 30377db0a0d..7bd2b5ae460 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videodecoderwrapper.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videodecoderwrapper.cc
>@@ -17,7 +17,7 @@
> #include "rtc_base/logging.h"
> #include "sdk/android/generated_video_jni/jni/VideoDecoderWrapper_jni.h"
> #include "sdk/android/generated_video_jni/jni/VideoDecoder_jni.h"
>-#include "sdk/android/src/jni/classreferenceholder.h"
>+#include "sdk/android/native_api/jni/java_types.h"
> #include "sdk/android/src/jni/encodedimage.h"
> #include "sdk/android/src/jni/videocodecstatus.h"
> #include "sdk/android/src/jni/videoframe.h"
>@@ -30,42 +30,49 @@ namespace {
> const int64_t kNumRtpTicksPerMillisec = 90000 / rtc::kNumMillisecsPerSec;
> 
> template <typename Dst, typename Src>
>-inline rtc::Optional<Dst> cast_optional(const rtc::Optional<Src>& value) {
>-  return value ? rtc::Optional<Dst>(rtc::dchecked_cast<Dst, Src>(*value))
>-               : rtc::nullopt;
>+inline absl::optional<Dst> cast_optional(const absl::optional<Src>& value) {
>+  return value ? absl::optional<Dst>(rtc::dchecked_cast<Dst, Src>(*value))
>+               : absl::nullopt;
> }
> }  // namespace
> 
>-VideoDecoderWrapper::VideoDecoderWrapper(JNIEnv* jni, jobject decoder)
>-    : decoder_(jni, decoder) {
>-  initialized_ = false;
>-  // QP parsing starts enabled and we disable it if the decoder provides frames.
>-  qp_parsing_enabled_ = true;
>-
>-  implementation_name_ = JavaToStdString(
>-      jni, Java_VideoDecoder_getImplementationName(jni, decoder));
>+VideoDecoderWrapper::VideoDecoderWrapper(JNIEnv* jni,
>+                                         const JavaRef<jobject>& decoder)
>+    : decoder_(jni, decoder),
>+      implementation_name_(JavaToStdString(
>+          jni,
>+          Java_VideoDecoder_getImplementationName(jni, decoder))),
>+      initialized_(false),
>+      qp_parsing_enabled_(true)  // QP parsing starts enabled and we disable it
>+                                 // if the decoder provides frames.
>+
>+{
>+  decoder_thread_checker_.DetachFromThread();
> }
> 
>+VideoDecoderWrapper::~VideoDecoderWrapper() = default;
>+
> int32_t VideoDecoderWrapper::InitDecode(const VideoCodec* codec_settings,
>                                         int32_t number_of_cores) {
>+  RTC_DCHECK_RUN_ON(&decoder_thread_checker_);
>   JNIEnv* jni = AttachCurrentThreadIfNeeded();
>-  ScopedLocalRefFrame local_ref_frame(jni);
>-
>   codec_settings_ = *codec_settings;
>   number_of_cores_ = number_of_cores;
>   return InitDecodeInternal(jni);
> }
> 
> int32_t VideoDecoderWrapper::InitDecodeInternal(JNIEnv* jni) {
>-  jobject settings = Java_Settings_Constructor(
>+  ScopedJavaLocalRef<jobject> settings = Java_Settings_Constructor(
>       jni, number_of_cores_, codec_settings_.width, codec_settings_.height);
> 
>-  jobject callback = Java_VideoDecoderWrapper_createDecoderCallback(
>-      jni, jlongFromPointer(this));
>+  ScopedJavaLocalRef<jobject> callback =
>+      Java_VideoDecoderWrapper_createDecoderCallback(jni,
>+                                                     jlongFromPointer(this));
> 
>-  jobject ret =
>-      Java_VideoDecoder_initDecode(jni, *decoder_, settings, callback);
>-  if (JavaToNativeVideoCodecStatus(jni, ret) == WEBRTC_VIDEO_CODEC_OK) {
>+  int32_t status = JavaToNativeVideoCodecStatus(
>+      jni, Java_VideoDecoder_initDecode(jni, decoder_, settings, callback));
>+  RTC_LOG(LS_INFO) << "initDecode: " << status;
>+  if (status == WEBRTC_VIDEO_CODEC_OK) {
>     initialized_ = true;
>   }
> 
>@@ -73,23 +80,20 @@ int32_t VideoDecoderWrapper::InitDecodeInternal(JNIEnv* jni) {
>   // providing QP values.
>   qp_parsing_enabled_ = true;
> 
>-  return HandleReturnCode(jni, ret);
>+  return status;
> }
> 
> int32_t VideoDecoderWrapper::Decode(
>     const EncodedImage& image_param,
>     bool missing_frames,
>-    const RTPFragmentationHeader* fragmentation,
>     const CodecSpecificInfo* codec_specific_info,
>     int64_t render_time_ms) {
>+  RTC_DCHECK_RUN_ON(&decoder_thread_checker_);
>   if (!initialized_) {
>     // Most likely initializing the codec failed.
>     return WEBRTC_VIDEO_CODEC_FALLBACK_SOFTWARE;
>   }
> 
>-  JNIEnv* jni = AttachCurrentThreadIfNeeded();
>-  ScopedLocalRefFrame local_ref_frame(jni);
>-
>   // Make a mutable copy so we can modify the timestamp.
>   EncodedImage input_image(image_param);
>   // We use RTP timestamp for capture time because capture_time_ms_ is always 0.
>@@ -102,67 +106,87 @@ int32_t VideoDecoderWrapper::Decode(
>   frame_extra_info.timestamp_rtp = input_image._timeStamp;
>   frame_extra_info.timestamp_ntp = input_image.ntp_time_ms_;
>   frame_extra_info.qp =
>-      qp_parsing_enabled_ ? ParseQP(input_image) : rtc::nullopt;
>-  frame_extra_infos_.push_back(frame_extra_info);
>+      qp_parsing_enabled_ ? ParseQP(input_image) : absl::nullopt;
>+  {
>+    rtc::CritScope cs(&frame_extra_infos_lock_);
>+    frame_extra_infos_.push_back(frame_extra_info);
>+  }
> 
>-  jobject jinput_image = NativeToJavaEncodedImage(jni, input_image);
>-  jobject ret = Java_VideoDecoder_decode(jni, *decoder_, jinput_image, nullptr);
>-  return HandleReturnCode(jni, ret);
>+  JNIEnv* env = AttachCurrentThreadIfNeeded();
>+  ScopedJavaLocalRef<jobject> jinput_image =
>+      NativeToJavaEncodedImage(env, input_image);
>+  ScopedJavaLocalRef<jobject> decode_info;
>+  ScopedJavaLocalRef<jobject> ret =
>+      Java_VideoDecoder_decode(env, decoder_, jinput_image, decode_info);
>+  return HandleReturnCode(env, ret, "decode");
> }
> 
> int32_t VideoDecoderWrapper::RegisterDecodeCompleteCallback(
>     DecodedImageCallback* callback) {
>+  RTC_DCHECK_RUNS_SERIALIZED(&callback_race_checker_);
>   callback_ = callback;
>   return WEBRTC_VIDEO_CODEC_OK;
> }
> 
> int32_t VideoDecoderWrapper::Release() {
>   JNIEnv* jni = AttachCurrentThreadIfNeeded();
>-  ScopedLocalRefFrame local_ref_frame(jni);
>-  jobject ret = Java_VideoDecoder_release(jni, *decoder_);
>-  frame_extra_infos_.clear();
>+  int32_t status = JavaToNativeVideoCodecStatus(
>+      jni, Java_VideoDecoder_release(jni, decoder_));
>+  RTC_LOG(LS_INFO) << "release: " << status;
>+  {
>+    rtc::CritScope cs(&frame_extra_infos_lock_);
>+    frame_extra_infos_.clear();
>+  }
>   initialized_ = false;
>-  return HandleReturnCode(jni, ret);
>+  // It is allowed to reinitialize the codec on a different thread.
>+  decoder_thread_checker_.DetachFromThread();
>+  return status;
> }
> 
> bool VideoDecoderWrapper::PrefersLateDecoding() const {
>   JNIEnv* jni = AttachCurrentThreadIfNeeded();
>-  return Java_VideoDecoder_getPrefersLateDecoding(jni, *decoder_);
>+  return Java_VideoDecoder_getPrefersLateDecoding(jni, decoder_);
> }
> 
> const char* VideoDecoderWrapper::ImplementationName() const {
>   return implementation_name_.c_str();
> }
> 
>-void VideoDecoderWrapper::OnDecodedFrame(JNIEnv* env,
>-                                         jobject j_caller,
>-                                         jobject j_frame,
>-                                         jobject j_decode_time_ms,
>-                                         jobject j_qp) {
>-  const uint64_t timestamp_ns = GetJavaVideoFrameTimestampNs(env, j_frame);
>+void VideoDecoderWrapper::OnDecodedFrame(
>+    JNIEnv* env,
>+    const JavaRef<jobject>& j_caller,
>+    const JavaRef<jobject>& j_frame,
>+    const JavaRef<jobject>& j_decode_time_ms,
>+    const JavaRef<jobject>& j_qp) {
>+  RTC_DCHECK_RUNS_SERIALIZED(&callback_race_checker_);
>+  const int64_t timestamp_ns = GetJavaVideoFrameTimestampNs(env, j_frame);
> 
>   FrameExtraInfo frame_extra_info;
>-  do {
>-    if (frame_extra_infos_.empty()) {
>-      RTC_LOG(LS_WARNING) << "Java decoder produced an unexpected frame: "
>-                          << timestamp_ns;
>-      return;
>-    }
>+  {
>+    rtc::CritScope cs(&frame_extra_infos_lock_);
>+
>+    do {
>+      if (frame_extra_infos_.empty()) {
>+        RTC_LOG(LS_WARNING)
>+            << "Java decoder produced an unexpected frame: " << timestamp_ns;
>+        return;
>+      }
> 
>-    frame_extra_info = frame_extra_infos_.front();
>-    frame_extra_infos_.pop_front();
>-    // If the decoder might drop frames so iterate through the queue until we
>-    // find a matching timestamp.
>-  } while (frame_extra_info.timestamp_ns != timestamp_ns);
>+      frame_extra_info = frame_extra_infos_.front();
>+      frame_extra_infos_.pop_front();
>+      // If the decoder might drop frames so iterate through the queue until we
>+      // find a matching timestamp.
>+    } while (frame_extra_info.timestamp_ns != timestamp_ns);
>+  }
> 
>   VideoFrame frame =
>       JavaToNativeFrame(env, j_frame, frame_extra_info.timestamp_rtp);
>   frame.set_ntp_time_ms(frame_extra_info.timestamp_ntp);
> 
>-  rtc::Optional<int32_t> decoding_time_ms =
>+  absl::optional<int32_t> decoding_time_ms =
>       JavaToNativeOptionalInt(env, j_decode_time_ms);
> 
>-  rtc::Optional<uint8_t> decoder_qp =
>+  absl::optional<uint8_t> decoder_qp =
>       cast_optional<uint8_t, int32_t>(JavaToNativeOptionalInt(env, j_qp));
>   // If the decoder provides QP values itself, no need to parse the bitstream.
>   // Enable QP parsing if decoder does not provide QP values itself.
>@@ -171,28 +195,44 @@ void VideoDecoderWrapper::OnDecodedFrame(JNIEnv* env,
>                      decoder_qp ? decoder_qp : frame_extra_info.qp);
> }
> 
>-int32_t VideoDecoderWrapper::HandleReturnCode(JNIEnv* jni, jobject code) {
>-  int32_t value = JavaToNativeVideoCodecStatus(jni, code);
>-  if (value < 0) {  // Any errors are represented by negative values.
>-    // Reset the codec.
>-    if (Release() == WEBRTC_VIDEO_CODEC_OK) {
>-      InitDecodeInternal(jni);
>-    }
>+VideoDecoderWrapper::FrameExtraInfo::FrameExtraInfo() = default;
>+VideoDecoderWrapper::FrameExtraInfo::FrameExtraInfo(const FrameExtraInfo&) =
>+    default;
>+VideoDecoderWrapper::FrameExtraInfo::~FrameExtraInfo() = default;
> 
>-    RTC_LOG(LS_WARNING) << "Falling back to software decoder.";
>-    return WEBRTC_VIDEO_CODEC_FALLBACK_SOFTWARE;
>-  } else {
>+int32_t VideoDecoderWrapper::HandleReturnCode(JNIEnv* jni,
>+                                              const JavaRef<jobject>& j_value,
>+                                              const char* method_name) {
>+  int32_t value = JavaToNativeVideoCodecStatus(jni, j_value);
>+  if (value >= 0) {  // OK or NO_OUTPUT
>     return value;
>   }
>+
>+  RTC_LOG(LS_WARNING) << method_name << ": " << value;
>+  if (value == WEBRTC_VIDEO_CODEC_FALLBACK_SOFTWARE ||
>+      value == WEBRTC_VIDEO_CODEC_UNINITIALIZED) {  // Critical error.
>+    RTC_LOG(LS_WARNING) << "Java decoder requested software fallback.";
>+    return WEBRTC_VIDEO_CODEC_FALLBACK_SOFTWARE;
>+  }
>+
>+  // Try resetting the codec.
>+  if (Release() == WEBRTC_VIDEO_CODEC_OK &&
>+      InitDecodeInternal(jni) == WEBRTC_VIDEO_CODEC_OK) {
>+    RTC_LOG(LS_WARNING) << "Reset Java decoder.";
>+    return WEBRTC_VIDEO_CODEC_ERROR;
>+  }
>+
>+  RTC_LOG(LS_WARNING) << "Unable to reset Java decoder.";
>+  return WEBRTC_VIDEO_CODEC_FALLBACK_SOFTWARE;
> }
> 
>-rtc::Optional<uint8_t> VideoDecoderWrapper::ParseQP(
>+absl::optional<uint8_t> VideoDecoderWrapper::ParseQP(
>     const EncodedImage& input_image) {
>   if (input_image.qp_ != -1) {
>     return input_image.qp_;
>   }
> 
>-  rtc::Optional<uint8_t> qp;
>+  absl::optional<uint8_t> qp;
>   switch (codec_settings_.codecType) {
>     case kVideoCodecVP8: {
>       int qp_int;
>@@ -223,5 +263,19 @@ rtc::Optional<uint8_t> VideoDecoderWrapper::ParseQP(
>   return qp;
> }
> 
>+std::unique_ptr<VideoDecoder> JavaToNativeVideoDecoder(
>+    JNIEnv* jni,
>+    const JavaRef<jobject>& j_decoder) {
>+  const jlong native_decoder =
>+      Java_VideoDecoder_createNativeVideoDecoder(jni, j_decoder);
>+  VideoDecoder* decoder;
>+  if (native_decoder == 0) {
>+    decoder = new VideoDecoderWrapper(jni, j_decoder);
>+  } else {
>+    decoder = reinterpret_cast<VideoDecoder*>(native_decoder);
>+  }
>+  return std::unique_ptr<VideoDecoder>(decoder);
>+}
>+
> }  // namespace jni
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videodecoderwrapper.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videodecoderwrapper.h
>index 14ced839e70..eb2958a9087 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videodecoderwrapper.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videodecoderwrapper.h
>@@ -12,10 +12,13 @@
> #define SDK_ANDROID_SRC_JNI_VIDEODECODERWRAPPER_H_
> 
> #include <jni.h>
>+#include <atomic>
> #include <deque>
> 
> #include "api/video_codecs/video_decoder.h"
> #include "common_video/h264/h264_bitstream_parser.h"
>+#include "rtc_base/race_checker.h"
>+#include "rtc_base/thread_checker.h"
> #include "sdk/android/src/jni/jni_helpers.h"
> 
> namespace webrtc {
>@@ -24,21 +27,24 @@ namespace jni {
> // Wraps a Java decoder and delegates all calls to it.
> class VideoDecoderWrapper : public VideoDecoder {
>  public:
>-  VideoDecoderWrapper(JNIEnv* jni, jobject decoder);
>+  VideoDecoderWrapper(JNIEnv* jni, const JavaRef<jobject>& decoder);
>+  ~VideoDecoderWrapper() override;
> 
>   int32_t InitDecode(const VideoCodec* codec_settings,
>                      int32_t number_of_cores) override;
> 
>   int32_t Decode(const EncodedImage& input_image,
>                  bool missing_frames,
>-                 const RTPFragmentationHeader* fragmentation,
>                  const CodecSpecificInfo* codec_specific_info,
>                  int64_t render_time_ms) override;
> 
>   int32_t RegisterDecodeCompleteCallback(
>       DecodedImageCallback* callback) override;
> 
>-  int32_t Release() override;
>+  // TODO(sakal): This is not always called on the correct thread. It is called
>+  // from VCMGenericDecoder destructor which is on a different thread but is
>+  // still safe and synchronous.
>+  int32_t Release() override RTC_NO_THREAD_SAFETY_ANALYSIS;
> 
>   // Returns true if the decoder prefer to decode frames late.
>   // That is, it can not decode infinite number of frames before the decoded
>@@ -49,10 +55,10 @@ class VideoDecoderWrapper : public VideoDecoder {
> 
>   // Wraps the frame to a AndroidVideoBuffer and passes it to the callback.
>   void OnDecodedFrame(JNIEnv* env,
>-                      jobject j_caller,
>-                      jobject j_frame,
>-                      jobject j_decode_time_ms,
>-                      jobject j_qp);
>+                      const JavaRef<jobject>& j_caller,
>+                      const JavaRef<jobject>& j_frame,
>+                      const JavaRef<jobject>& j_decode_time_ms,
>+                      const JavaRef<jobject>& j_qp);
> 
>  private:
>   struct FrameExtraInfo {
>@@ -60,31 +66,57 @@ class VideoDecoderWrapper : public VideoDecoder {
> 
>     uint32_t timestamp_rtp;
>     int64_t timestamp_ntp;
>-    rtc::Optional<uint8_t> qp;
>+    absl::optional<uint8_t> qp;
>+
>+    FrameExtraInfo();
>+    FrameExtraInfo(const FrameExtraInfo&);
>+    ~FrameExtraInfo();
>   };
> 
>-  int32_t InitDecodeInternal(JNIEnv* jni);
>+  int32_t InitDecodeInternal(JNIEnv* jni) RTC_RUN_ON(decoder_thread_checker_);
> 
>   // Takes Java VideoCodecStatus, handles it and returns WEBRTC_VIDEO_CODEC_*
>   // status code.
>-  int32_t HandleReturnCode(JNIEnv* jni, jobject code);
>+  int32_t HandleReturnCode(JNIEnv* jni,
>+                           const JavaRef<jobject>& j_value,
>+                           const char* method_name)
>+      RTC_RUN_ON(decoder_thread_checker_);
>+
>+  absl::optional<uint8_t> ParseQP(const EncodedImage& input_image)
>+      RTC_RUN_ON(decoder_thread_checker_);
>+
>+  const ScopedJavaGlobalRef<jobject> decoder_;
>+  const std::string implementation_name_;
> 
>-  rtc::Optional<uint8_t> ParseQP(const EncodedImage& input_image);
>+  rtc::ThreadChecker decoder_thread_checker_;
>+  // Callbacks must be executed sequentially on an arbitrary thread. We do not
>+  // own this thread so a thread checker cannot be used.
>+  rtc::RaceChecker callback_race_checker_;
> 
>-  VideoCodec codec_settings_;
>-  int32_t number_of_cores_;
>+  // Initialized on InitDecode and immutable after that.
>+  VideoCodec codec_settings_ RTC_GUARDED_BY(decoder_thread_checker_);
>+  int32_t number_of_cores_ RTC_GUARDED_BY(decoder_thread_checker_);
> 
>-  bool initialized_;
>-  std::deque<FrameExtraInfo> frame_extra_infos_;
>-  bool qp_parsing_enabled_;
>-  H264BitstreamParser h264_bitstream_parser_;
>-  std::string implementation_name_;
>+  bool initialized_ RTC_GUARDED_BY(decoder_thread_checker_);
>+  H264BitstreamParser h264_bitstream_parser_
>+      RTC_GUARDED_BY(decoder_thread_checker_);
> 
>-  DecodedImageCallback* callback_;
>+  DecodedImageCallback* callback_ RTC_GUARDED_BY(callback_race_checker_);
> 
>-  const ScopedGlobalRef<jobject> decoder_;
>+  // Accessed both on the decoder thread and the callback thread.
>+  std::atomic<bool> qp_parsing_enabled_;
>+  rtc::CriticalSection frame_extra_infos_lock_;
>+  std::deque<FrameExtraInfo> frame_extra_infos_
>+      RTC_GUARDED_BY(frame_extra_infos_lock_);
> };
> 
>+/* If the j_decoder is a wrapped native decoder, unwrap it. If it is not,
>+ * wrap it in a VideoDecoderWrapper.
>+ */
>+std::unique_ptr<VideoDecoder> JavaToNativeVideoDecoder(
>+    JNIEnv* jni,
>+    const JavaRef<jobject>& j_decoder);
>+
> }  // namespace jni
> }  // namespace webrtc
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videoencoderfactorywrapper.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videoencoderfactorywrapper.cc
>index 255031320e6..44026662dbc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videoencoderfactorywrapper.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videoencoderfactorywrapper.cc
>@@ -13,68 +13,57 @@
> #include "api/video_codecs/video_encoder.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "rtc_base/logging.h"
>-#include "sdk/android/src/jni/class_loader.h"
>+#include "sdk/android/generated_video_jni/jni/VideoEncoderFactory_jni.h"
>+#include "sdk/android/native_api/jni/class_loader.h"
>+#include "sdk/android/native_api/jni/java_types.h"
> #include "sdk/android/src/jni/videocodecinfo.h"
>-#include "sdk/android/src/jni/wrappednativecodec.h"
>+#include "sdk/android/src/jni/videoencoderwrapper.h"
> 
> namespace webrtc {
> namespace jni {
> 
>-VideoEncoderFactoryWrapper::VideoEncoderFactoryWrapper(JNIEnv* jni,
>-                                                       jobject encoder_factory)
>+VideoEncoderFactoryWrapper::VideoEncoderFactoryWrapper(
>+    JNIEnv* jni,
>+    const JavaRef<jobject>& encoder_factory)
>     : encoder_factory_(jni, encoder_factory) {
>-  jclass encoder_factory_class = jni->GetObjectClass(*encoder_factory_);
>-  create_encoder_method_ = jni->GetMethodID(
>-      encoder_factory_class, "createEncoder",
>-      "(Lorg/webrtc/VideoCodecInfo;)Lorg/webrtc/VideoEncoder;");
>-  get_supported_codecs_method_ =
>-      jni->GetMethodID(encoder_factory_class, "getSupportedCodecs",
>-                       "()[Lorg/webrtc/VideoCodecInfo;");
>-
>-  supported_formats_ = GetSupportedFormats(jni);
>+  const ScopedJavaLocalRef<jobjectArray> j_supported_codecs =
>+      Java_VideoEncoderFactory_getSupportedCodecs(jni, encoder_factory);
>+  supported_formats_ = JavaToNativeVector<SdpVideoFormat>(
>+      jni, j_supported_codecs, &VideoCodecInfoToSdpVideoFormat);
> }
>+VideoEncoderFactoryWrapper::~VideoEncoderFactoryWrapper() = default;
> 
> std::unique_ptr<VideoEncoder> VideoEncoderFactoryWrapper::CreateVideoEncoder(
>     const SdpVideoFormat& format) {
>   JNIEnv* jni = AttachCurrentThreadIfNeeded();
>-  ScopedLocalRefFrame local_ref_frame(jni);
>-  jobject j_codec_info = SdpVideoFormatToVideoCodecInfo(jni, format);
>-  jobject encoder = jni->CallObjectMethod(*encoder_factory_,
>-                                          create_encoder_method_, j_codec_info);
>-  return encoder != nullptr ? JavaToNativeVideoEncoder(jni, encoder) : nullptr;
>+  ScopedJavaLocalRef<jobject> j_codec_info =
>+      SdpVideoFormatToVideoCodecInfo(jni, format);
>+  ScopedJavaLocalRef<jobject> encoder = Java_VideoEncoderFactory_createEncoder(
>+      jni, encoder_factory_, j_codec_info);
>+  if (!encoder.obj())
>+    return nullptr;
>+  return JavaToNativeVideoEncoder(jni, encoder);
>+}
>+
>+std::vector<SdpVideoFormat> VideoEncoderFactoryWrapper::GetSupportedFormats()
>+    const {
>+  return supported_formats_;
> }
> 
> VideoEncoderFactory::CodecInfo VideoEncoderFactoryWrapper::QueryVideoEncoder(
>     const SdpVideoFormat& format) const {
>   JNIEnv* jni = AttachCurrentThreadIfNeeded();
>-  ScopedLocalRefFrame local_ref_frame(jni);
>-
>-  jobject j_codec_info = SdpVideoFormatToVideoCodecInfo(jni, format);
>-  jobject encoder = jni->CallObjectMethod(*encoder_factory_,
>-                                          create_encoder_method_, j_codec_info);
>+  ScopedJavaLocalRef<jobject> j_codec_info =
>+      SdpVideoFormatToVideoCodecInfo(jni, format);
>+  ScopedJavaLocalRef<jobject> encoder = Java_VideoEncoderFactory_createEncoder(
>+      jni, encoder_factory_, j_codec_info);
> 
>   CodecInfo codec_info;
>   // Check if this is a wrapped native software encoder implementation.
>-  codec_info.is_hardware_accelerated = !IsWrappedSoftwareEncoder(jni, encoder);
>+  codec_info.is_hardware_accelerated = IsHardwareVideoEncoder(jni, encoder);
>   codec_info.has_internal_source = false;
>   return codec_info;
> }
> 
>-std::vector<SdpVideoFormat> VideoEncoderFactoryWrapper::GetSupportedFormats(
>-    JNIEnv* jni) const {
>-  const jobjectArray j_supported_codecs = static_cast<jobjectArray>(
>-      jni->CallObjectMethod(*encoder_factory_, get_supported_codecs_method_));
>-  const jsize supported_codecs_count = jni->GetArrayLength(j_supported_codecs);
>-
>-  std::vector<SdpVideoFormat> supported_formats;
>-  for (jsize i = 0; i < supported_codecs_count; i++) {
>-    jobject j_supported_codec =
>-        jni->GetObjectArrayElement(j_supported_codecs, i);
>-    supported_formats.push_back(
>-        VideoCodecInfoToSdpVideoFormat(jni, j_supported_codec));
>-  }
>-  return supported_formats;
>-}
>-
> }  // namespace jni
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videoencoderfactorywrapper.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videoencoderfactorywrapper.h
>index 26174be2922..2c2dadcaecb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videoencoderfactorywrapper.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videoencoderfactorywrapper.h
>@@ -25,26 +25,20 @@ namespace jni {
> // JNI and wraps the encoder inside VideoEncoderWrapper.
> class VideoEncoderFactoryWrapper : public VideoEncoderFactory {
>  public:
>-  VideoEncoderFactoryWrapper(JNIEnv* jni, jobject encoder_factory);
>+  VideoEncoderFactoryWrapper(JNIEnv* jni,
>+                             const JavaRef<jobject>& encoder_factory);
>+  ~VideoEncoderFactoryWrapper() override;
> 
>   std::unique_ptr<VideoEncoder> CreateVideoEncoder(
>       const SdpVideoFormat& format) override;
> 
>   // Returns a list of supported codecs in order of preference.
>-  std::vector<SdpVideoFormat> GetSupportedFormats() const override {
>-    return supported_formats_;
>-  }
>+  std::vector<SdpVideoFormat> GetSupportedFormats() const override;
> 
>   CodecInfo QueryVideoEncoder(const SdpVideoFormat& format) const override;
> 
>  private:
>-  std::vector<SdpVideoFormat> GetSupportedFormats(JNIEnv* jni) const;
>-
>-  const ScopedGlobalRef<jobject> encoder_factory_;
>-
>-  jmethodID create_encoder_method_;
>-  jmethodID get_supported_codecs_method_;
>-
>+  const ScopedJavaGlobalRef<jobject> encoder_factory_;
>   std::vector<SdpVideoFormat> supported_formats_;
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videoencoderfallback.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videoencoderfallback.cc
>index b151f15fed8..88ea3df18c2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videoencoderfallback.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videoencoderfallback.cc
>@@ -10,27 +10,28 @@
> 
> #include <jni.h>
> 
>-#include "media/engine/videoencodersoftwarefallbackwrapper.h"
>+#include "api/video_codecs/video_encoder_software_fallback_wrapper.h"
>+#include "sdk/android/generated_video_jni/jni/VideoEncoderFallback_jni.h"
> #include "sdk/android/src/jni/jni_helpers.h"
>-#include "sdk/android/src/jni/wrappednativecodec.h"
>+#include "sdk/android/src/jni/videoencoderwrapper.h"
> 
> namespace webrtc {
> namespace jni {
> 
>-JNI_FUNCTION_DECLARATION(jlong,
>-                         VideoEncoderFallback_createNativeEncoder,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jobject j_fallback_encoder,
>-                         jobject j_primary_encoder) {
>+static jlong JNI_VideoEncoderFallback_CreateEncoder(
>+    JNIEnv* jni,
>+    const JavaParamRef<jclass>&,
>+    const JavaParamRef<jobject>& j_fallback_encoder,
>+    const JavaParamRef<jobject>& j_primary_encoder) {
>   std::unique_ptr<VideoEncoder> fallback_encoder =
>       JavaToNativeVideoEncoder(jni, j_fallback_encoder);
>   std::unique_ptr<VideoEncoder> primary_encoder =
>       JavaToNativeVideoEncoder(jni, j_primary_encoder);
> 
>-  VideoEncoderSoftwareFallbackWrapper* nativeWrapper =
>-      new VideoEncoderSoftwareFallbackWrapper(std::move(fallback_encoder),
>-                                              std::move(primary_encoder));
>+  VideoEncoder* nativeWrapper =
>+      CreateVideoEncoderSoftwareFallbackWrapper(std::move(fallback_encoder),
>+                                                std::move(primary_encoder))
>+          .release();
> 
>   return jlongFromPointer(nativeWrapper);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videoencoderwrapper.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videoencoderwrapper.cc
>index 13b0e37c16c..9500a1f7a9a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videoencoderwrapper.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videoencoderwrapper.cc
>@@ -19,39 +19,30 @@
> #include "modules/video_coding/utility/vp8_header_parser.h"
> #include "modules/video_coding/utility/vp9_uncompressed_header_parser.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/random.h"
>-#include "rtc_base/timeutils.h"
> #include "sdk/android/generated_video_jni/jni/VideoEncoderWrapper_jni.h"
> #include "sdk/android/generated_video_jni/jni/VideoEncoder_jni.h"
>-#include "sdk/android/src/jni/class_loader.h"
>+#include "sdk/android/native_api/jni/class_loader.h"
>+#include "sdk/android/native_api/jni/java_types.h"
> #include "sdk/android/src/jni/encodedimage.h"
> #include "sdk/android/src/jni/videocodecstatus.h"
> 
> namespace webrtc {
> namespace jni {
> 
>-static const int kMaxJavaEncoderResets = 3;
>-
>-VideoEncoderWrapper::VideoEncoderWrapper(JNIEnv* jni, jobject j_encoder)
>-    : encoder_(jni, j_encoder),
>-      frame_type_class_(jni,
>-                        GetClass(jni, "org/webrtc/EncodedImage$FrameType")),
>-      int_array_class_(jni, jni->FindClass("[I")) {
>+VideoEncoderWrapper::VideoEncoderWrapper(JNIEnv* jni,
>+                                         const JavaRef<jobject>& j_encoder)
>+    : encoder_(jni, j_encoder), int_array_class_(GetClass(jni, "[I")) {
>   implementation_name_ = GetImplementationName(jni);
> 
>   initialized_ = false;
>   num_resets_ = 0;
>-
>-  Random random(rtc::TimeMicros());
>-  picture_id_ = random.Rand<uint16_t>() & 0x7FFF;
>-  tl0_pic_idx_ = random.Rand<uint8_t>();
> }
>+VideoEncoderWrapper::~VideoEncoderWrapper() = default;
> 
> int32_t VideoEncoderWrapper::InitEncode(const VideoCodec* codec_settings,
>                                         int32_t number_of_cores,
>                                         size_t max_payload_size) {
>   JNIEnv* jni = AttachCurrentThreadIfNeeded();
>-  ScopedLocalRefFrame local_ref_frame(jni);
> 
>   number_of_cores_ = number_of_cores;
>   codec_settings_ = *codec_settings;
>@@ -69,27 +60,30 @@ int32_t VideoEncoderWrapper::InitEncodeInternal(JNIEnv* jni) {
>       break;
>     case kVideoCodecVP9:
>       automatic_resize_on = codec_settings_.VP9()->automaticResizeOn;
>+      gof_.SetGofInfoVP9(TemporalStructureMode::kTemporalStructureMode1);
>+      gof_idx_ = 0;
>       break;
>     default:
>       automatic_resize_on = true;
>   }
> 
>-  jobject settings = Java_Settings_Constructor(
>+  ScopedJavaLocalRef<jobject> settings = Java_Settings_Constructor(
>       jni, number_of_cores_, codec_settings_.width, codec_settings_.height,
>-      codec_settings_.startBitrate, codec_settings_.maxFramerate,
>-      automatic_resize_on);
>+      static_cast<int>(codec_settings_.startBitrate),
>+      static_cast<int>(codec_settings_.maxFramerate), automatic_resize_on);
> 
>-  jobject callback = Java_VideoEncoderWrapper_createEncoderCallback(
>-      jni, jlongFromPointer(this));
>+  ScopedJavaLocalRef<jobject> callback =
>+      Java_VideoEncoderWrapper_createEncoderCallback(jni,
>+                                                     jlongFromPointer(this));
> 
>-  jobject ret =
>-      Java_VideoEncoder_initEncode(jni, *encoder_, settings, callback);
>+  int32_t status = JavaToNativeVideoCodecStatus(
>+      jni, Java_VideoEncoder_initEncode(jni, encoder_, settings, callback));
>+  RTC_LOG(LS_INFO) << "initEncode: " << status;
> 
>-  if (JavaToNativeVideoCodecStatus(jni, ret) == WEBRTC_VIDEO_CODEC_OK) {
>+  if (status == WEBRTC_VIDEO_CODEC_OK) {
>     initialized_ = true;
>   }
>-
>-  return HandleReturnCode(jni, ret);
>+  return status;
> }
> 
> int32_t VideoEncoderWrapper::RegisterEncodeCompleteCallback(
>@@ -100,12 +94,15 @@ int32_t VideoEncoderWrapper::RegisterEncodeCompleteCallback(
> 
> int32_t VideoEncoderWrapper::Release() {
>   JNIEnv* jni = AttachCurrentThreadIfNeeded();
>-  ScopedLocalRefFrame local_ref_frame(jni);
>-  jobject ret = Java_VideoEncoder_release(jni, *encoder_);
>+
>+  int32_t status = JavaToNativeVideoCodecStatus(
>+      jni, Java_VideoEncoder_release(jni, encoder_));
>+  RTC_LOG(LS_INFO) << "release: " << status;
>   frame_extra_infos_.clear();
>   initialized_ = false;
>   encoder_queue_ = nullptr;
>-  return HandleReturnCode(jni, ret);
>+
>+  return status;
> }
> 
> int32_t VideoEncoderWrapper::Encode(
>@@ -118,61 +115,97 @@ int32_t VideoEncoderWrapper::Encode(
>   }
> 
>   JNIEnv* jni = AttachCurrentThreadIfNeeded();
>-  ScopedLocalRefFrame local_ref_frame(jni);
> 
>   // Construct encode info.
>-  jobjectArray j_frame_types = NativeToJavaFrameTypeArray(jni, *frame_types);
>-  jobject encode_info = Java_EncodeInfo_Constructor(jni, j_frame_types);
>+  ScopedJavaLocalRef<jobjectArray> j_frame_types =
>+      NativeToJavaFrameTypeArray(jni, *frame_types);
>+  ScopedJavaLocalRef<jobject> encode_info =
>+      Java_EncodeInfo_Constructor(jni, j_frame_types);
> 
>   FrameExtraInfo info;
>   info.capture_time_ns = frame.timestamp_us() * rtc::kNumNanosecsPerMicrosec;
>   info.timestamp_rtp = frame.timestamp();
>   frame_extra_infos_.push_back(info);
> 
>-  jobject ret = Java_VideoEncoder_encode(
>-      jni, *encoder_, NativeToJavaFrame(jni, frame), encode_info);
>-  return HandleReturnCode(jni, ret);
>+  ScopedJavaLocalRef<jobject> j_frame = NativeToJavaVideoFrame(jni, frame);
>+  ScopedJavaLocalRef<jobject> ret =
>+      Java_VideoEncoder_encode(jni, encoder_, j_frame, encode_info);
>+  ReleaseJavaVideoFrame(jni, j_frame);
>+  return HandleReturnCode(jni, ret, "encode");
> }
> 
> int32_t VideoEncoderWrapper::SetChannelParameters(uint32_t packet_loss,
>                                                   int64_t rtt) {
>   JNIEnv* jni = AttachCurrentThreadIfNeeded();
>-  ScopedLocalRefFrame local_ref_frame(jni);
>-  jobject ret = Java_VideoEncoder_setChannelParameters(
>-      jni, *encoder_, (jshort)packet_loss, (jlong)rtt);
>-  return HandleReturnCode(jni, ret);
>+  ScopedJavaLocalRef<jobject> ret = Java_VideoEncoder_setChannelParameters(
>+      jni, encoder_, (jshort)packet_loss, (jlong)rtt);
>+  return HandleReturnCode(jni, ret, "setChannelParameters");
> }
> 
> int32_t VideoEncoderWrapper::SetRateAllocation(
>-    const BitrateAllocation& allocation,
>+    const VideoBitrateAllocation& allocation,
>     uint32_t framerate) {
>   JNIEnv* jni = AttachCurrentThreadIfNeeded();
>-  ScopedLocalRefFrame local_ref_frame(jni);
> 
>-  jobject j_bitrate_allocation = ToJavaBitrateAllocation(jni, allocation);
>-  jobject ret = Java_VideoEncoder_setRateAllocation(
>-      jni, *encoder_, j_bitrate_allocation, (jint)framerate);
>-  return HandleReturnCode(jni, ret);
>+  ScopedJavaLocalRef<jobject> j_bitrate_allocation =
>+      ToJavaBitrateAllocation(jni, allocation);
>+  ScopedJavaLocalRef<jobject> ret = Java_VideoEncoder_setRateAllocation(
>+      jni, encoder_, j_bitrate_allocation, (jint)framerate);
>+  return HandleReturnCode(jni, ret, "setRateAllocation");
> }
> 
> VideoEncoderWrapper::ScalingSettings VideoEncoderWrapper::GetScalingSettings()
>     const {
>   JNIEnv* jni = AttachCurrentThreadIfNeeded();
>-  ScopedLocalRefFrame local_ref_frame(jni);
>-  jobject j_scaling_settings =
>-      Java_VideoEncoder_getScalingSettings(jni, *encoder_);
>+  ScopedJavaLocalRef<jobject> j_scaling_settings =
>+      Java_VideoEncoder_getScalingSettings(jni, encoder_);
>   bool isOn =
>       Java_VideoEncoderWrapper_getScalingSettingsOn(jni, j_scaling_settings);
> 
>-  rtc::Optional<int> low = JavaToNativeOptionalInt(
>+  if (!isOn)
>+    return ScalingSettings::kOff;
>+
>+  absl::optional<int> low = JavaToNativeOptionalInt(
>       jni,
>       Java_VideoEncoderWrapper_getScalingSettingsLow(jni, j_scaling_settings));
>-  rtc::Optional<int> high = JavaToNativeOptionalInt(
>+  absl::optional<int> high = JavaToNativeOptionalInt(
>       jni,
>       Java_VideoEncoderWrapper_getScalingSettingsHigh(jni, j_scaling_settings));
> 
>-  return (low && high) ? ScalingSettings(isOn, *low, *high)
>-                       : ScalingSettings(isOn);
>+  if (low && high)
>+    return ScalingSettings(*low, *high);
>+
>+  switch (codec_settings_.codecType) {
>+    case kVideoCodecVP8: {
>+      // Same as in vp8_impl.cc.
>+      static const int kLowVp8QpThreshold = 29;
>+      static const int kHighVp8QpThreshold = 95;
>+      return ScalingSettings(low.value_or(kLowVp8QpThreshold),
>+                             high.value_or(kHighVp8QpThreshold));
>+    }
>+    case kVideoCodecVP9: {
>+      // QP is obtained from VP9-bitstream, so the QP corresponds to the
>+      // bitstream range of [0, 255] and not the user-level range of [0,63].
>+      static const int kLowVp9QpThreshold = 96;
>+      static const int kHighVp9QpThreshold = 185;
>+
>+      return VideoEncoder::ScalingSettings(kLowVp9QpThreshold,
>+                                           kHighVp9QpThreshold);
>+    }
>+    case kVideoCodecH264: {
>+      // Same as in h264_encoder_impl.cc.
>+      static const int kLowH264QpThreshold = 24;
>+      static const int kHighH264QpThreshold = 37;
>+      return ScalingSettings(low.value_or(kLowH264QpThreshold),
>+                             high.value_or(kHighH264QpThreshold));
>+    }
>+    default:
>+      return ScalingSettings::kOff;
>+  }
>+}
>+
>+bool VideoEncoderWrapper::SupportsNativeHandle() const {
>+  return true;
> }
> 
> const char* VideoEncoderWrapper::ImplementationName() const {
>@@ -180,79 +213,114 @@ const char* VideoEncoderWrapper::ImplementationName() const {
> }
> 
> void VideoEncoderWrapper::OnEncodedFrame(JNIEnv* jni,
>-                                         jobject j_caller,
>-                                         jobject j_buffer,
>+                                         const JavaRef<jobject>& j_caller,
>+                                         const JavaRef<jobject>& j_buffer,
>                                          jint encoded_width,
>                                          jint encoded_height,
>                                          jlong capture_time_ns,
>                                          jint frame_type,
>                                          jint rotation,
>                                          jboolean complete_frame,
>-                                         jobject j_qp) {
>+                                         const JavaRef<jobject>& j_qp) {
>   const uint8_t* buffer =
>-      static_cast<uint8_t*>(jni->GetDirectBufferAddress(j_buffer));
>-  const size_t buffer_size = jni->GetDirectBufferCapacity(j_buffer);
>+      static_cast<uint8_t*>(jni->GetDirectBufferAddress(j_buffer.obj()));
>+  const size_t buffer_size = jni->GetDirectBufferCapacity(j_buffer.obj());
> 
>   std::vector<uint8_t> buffer_copy(buffer_size);
>   memcpy(buffer_copy.data(), buffer, buffer_size);
>   const int qp = JavaToNativeOptionalInt(jni, j_qp).value_or(-1);
> 
>+  struct Lambda {
>+    VideoEncoderWrapper* video_encoder_wrapper;
>+    std::vector<uint8_t> task_buffer;
>+    int qp;
>+    jint encoded_width;
>+    jint encoded_height;
>+    jlong capture_time_ns;
>+    jint frame_type;
>+    jint rotation;
>+    jboolean complete_frame;
>+    std::deque<FrameExtraInfo>* frame_extra_infos;
>+    EncodedImageCallback* callback;
>+
>+    void operator()() const {
>+      // Encoded frames are delivered in the order received, but some of them
>+      // may be dropped, so remove records of frames older than the current one.
>+      //
>+      // NOTE: if the current frame is associated with Encoder A, in the time
>+      // since this frame was received, Encoder A could have been Release()'ed,
>+      // Encoder B InitEncode()'ed (due to reuse of Encoder A), and frames
>+      // received by Encoder B. Thus there may be frame_extra_infos entries that
>+      // don't belong to us, and we need to be careful not to remove them.
>+      // Removing only those entries older than the current frame provides this
>+      // guarantee.
>+      while (!frame_extra_infos->empty() &&
>+             frame_extra_infos->front().capture_time_ns < capture_time_ns) {
>+        frame_extra_infos->pop_front();
>+      }
>+      if (frame_extra_infos->empty() ||
>+          frame_extra_infos->front().capture_time_ns != capture_time_ns) {
>+        RTC_LOG(LS_WARNING)
>+            << "Java encoder produced an unexpected frame with timestamp: "
>+            << capture_time_ns;
>+        return;
>+      }
>+      FrameExtraInfo frame_extra_info = std::move(frame_extra_infos->front());
>+      frame_extra_infos->pop_front();
>+
>+      RTPFragmentationHeader header =
>+          video_encoder_wrapper->ParseFragmentationHeader(task_buffer);
>+      EncodedImage frame(const_cast<uint8_t*>(task_buffer.data()),
>+                         task_buffer.size(), task_buffer.size());
>+      frame._encodedWidth = encoded_width;
>+      frame._encodedHeight = encoded_height;
>+      frame._timeStamp = frame_extra_info.timestamp_rtp;
>+      frame.capture_time_ms_ = capture_time_ns / rtc::kNumNanosecsPerMillisec;
>+      frame._frameType = (FrameType)frame_type;
>+      frame.rotation_ = (VideoRotation)rotation;
>+      frame._completeFrame = complete_frame;
>+      if (qp == -1) {
>+        frame.qp_ = video_encoder_wrapper->ParseQp(task_buffer);
>+      } else {
>+        frame.qp_ = qp;
>+      }
>+
>+      CodecSpecificInfo info(
>+          video_encoder_wrapper->ParseCodecSpecificInfo(frame));
>+      callback->OnEncodedImage(frame, &info, &header);
>+    }
>+  };
>+
>   encoder_queue_->PostTask(
>-      [
>-        this, task_buffer = std::move(buffer_copy), qp, encoded_width,
>-        encoded_height, capture_time_ns, frame_type, rotation, complete_frame
>-      ]() {
>-        FrameExtraInfo frame_extra_info;
>-        do {
>-          if (frame_extra_infos_.empty()) {
>-            RTC_LOG(LS_WARNING)
>-                << "Java encoder produced an unexpected frame with timestamp: "
>-                << capture_time_ns;
>-            return;
>-          }
>-
>-          frame_extra_info = frame_extra_infos_.front();
>-          frame_extra_infos_.pop_front();
>-          // The encoder might drop frames so iterate through the queue until
>-          // we find a matching timestamp.
>-        } while (frame_extra_info.capture_time_ns != capture_time_ns);
>-
>-        RTPFragmentationHeader header = ParseFragmentationHeader(task_buffer);
>-        EncodedImage frame(const_cast<uint8_t*>(task_buffer.data()),
>-                           task_buffer.size(), task_buffer.size());
>-        frame._encodedWidth = encoded_width;
>-        frame._encodedHeight = encoded_height;
>-        frame._timeStamp = frame_extra_info.timestamp_rtp;
>-        frame.capture_time_ms_ = capture_time_ns / rtc::kNumNanosecsPerMillisec;
>-        frame._frameType = (FrameType)frame_type;
>-        frame.rotation_ = (VideoRotation)rotation;
>-        frame._completeFrame = complete_frame;
>-        if (qp == -1) {
>-          frame.qp_ = ParseQp(task_buffer);
>-        } else {
>-          frame.qp_ = qp;
>-        }
>-
>-        CodecSpecificInfo info(ParseCodecSpecificInfo(frame));
>-        callback_->OnEncodedImage(frame, &info, &header);
>-      });
>+      Lambda{this, std::move(buffer_copy), qp, encoded_width, encoded_height,
>+             capture_time_ns, frame_type, rotation, complete_frame,
>+             &frame_extra_infos_, callback_});
> }
> 
>-int32_t VideoEncoderWrapper::HandleReturnCode(JNIEnv* jni, jobject code) {
>-  int32_t value = JavaToNativeVideoCodecStatus(jni, code);
>-  if (value < 0) {  // Any errors are represented by negative values.
>-    // Try resetting the codec.
>-    if (++num_resets_ <= kMaxJavaEncoderResets &&
>-        Release() == WEBRTC_VIDEO_CODEC_OK) {
>-      RTC_LOG(LS_WARNING) << "Reset Java encoder: " << num_resets_;
>-      return InitEncodeInternal(jni);
>-    }
>+int32_t VideoEncoderWrapper::HandleReturnCode(JNIEnv* jni,
>+                                              const JavaRef<jobject>& j_value,
>+                                              const char* method_name) {
>+  int32_t value = JavaToNativeVideoCodecStatus(jni, j_value);
>+  if (value >= 0) {  // OK or NO_OUTPUT
>+    return value;
>+  }
> 
>-    RTC_LOG(LS_WARNING) << "Falling back to software decoder.";
>+  RTC_LOG(LS_WARNING) << method_name << ": " << value;
>+  if (value == WEBRTC_VIDEO_CODEC_FALLBACK_SOFTWARE ||
>+      value == WEBRTC_VIDEO_CODEC_UNINITIALIZED) {  // Critical error.
>+    RTC_LOG(LS_WARNING) << "Java encoder requested software fallback.";
>     return WEBRTC_VIDEO_CODEC_FALLBACK_SOFTWARE;
>-  } else {
>-    return value;
>   }
>+
>+  // Try resetting the codec.
>+  if (Release() == WEBRTC_VIDEO_CODEC_OK &&
>+      InitEncodeInternal(jni) == WEBRTC_VIDEO_CODEC_OK) {
>+    RTC_LOG(LS_WARNING) << "Reset Java encoder.";
>+    return WEBRTC_VIDEO_CODEC_ERROR;
>+  }
>+
>+  RTC_LOG(LS_WARNING) << "Unable to reset Java encoder.";
>+  return WEBRTC_VIDEO_CODEC_FALLBACK_SOFTWARE;
> }
> 
> RTPFragmentationHeader VideoEncoderWrapper::ParseFragmentationHeader(
>@@ -319,23 +387,19 @@ CodecSpecificInfo VideoEncoderWrapper::ParseCodecSpecificInfo(
> 
>   switch (codec_settings_.codecType) {
>     case kVideoCodecVP8:
>-      info.codecSpecific.VP8.pictureId = picture_id_;
>       info.codecSpecific.VP8.nonReference = false;
>       info.codecSpecific.VP8.simulcastIdx = 0;
>       info.codecSpecific.VP8.temporalIdx = kNoTemporalIdx;
>       info.codecSpecific.VP8.layerSync = false;
>-      info.codecSpecific.VP8.tl0PicIdx = kNoTl0PicIdx;
>       info.codecSpecific.VP8.keyIdx = kNoKeyIdx;
>       break;
>     case kVideoCodecVP9:
>       if (key_frame) {
>         gof_idx_ = 0;
>       }
>-      info.codecSpecific.VP9.picture_id = picture_id_;
>       info.codecSpecific.VP9.inter_pic_predicted = key_frame ? false : true;
>       info.codecSpecific.VP9.flexible_mode = false;
>       info.codecSpecific.VP9.ss_data_available = key_frame ? true : false;
>-      info.codecSpecific.VP9.tl0_pic_idx = tl0_pic_idx_++;
>       info.codecSpecific.VP9.temporal_idx = kNoTemporalIdx;
>       info.codecSpecific.VP9.spatial_idx = kNoSpatialIdx;
>       info.codecSpecific.VP9.temporal_up_switch = true;
>@@ -343,6 +407,8 @@ CodecSpecificInfo VideoEncoderWrapper::ParseCodecSpecificInfo(
>       info.codecSpecific.VP9.gof_idx =
>           static_cast<uint8_t>(gof_idx_++ % gof_.num_frames_in_gof);
>       info.codecSpecific.VP9.num_spatial_layers = 1;
>+      info.codecSpecific.VP9.first_frame_in_picture = true;
>+      info.codecSpecific.VP9.end_of_picture = true;
>       info.codecSpecific.VP9.spatial_layer_resolution_present = false;
>       if (info.codecSpecific.VP9.ss_data_available) {
>         info.codecSpecific.VP9.spatial_layer_resolution_present = true;
>@@ -355,36 +421,54 @@ CodecSpecificInfo VideoEncoderWrapper::ParseCodecSpecificInfo(
>       break;
>   }
> 
>-  picture_id_ = (picture_id_ + 1) & 0x7FFF;
>-
>   return info;
> }
> 
>-jobject VideoEncoderWrapper::ToJavaBitrateAllocation(
>+ScopedJavaLocalRef<jobject> VideoEncoderWrapper::ToJavaBitrateAllocation(
>     JNIEnv* jni,
>-    const BitrateAllocation& allocation) {
>-  jobjectArray j_allocation_array = jni->NewObjectArray(
>-      kMaxSpatialLayers, *int_array_class_, nullptr /* initial */);
>+    const VideoBitrateAllocation& allocation) {
>+  ScopedJavaLocalRef<jobjectArray> j_allocation_array(
>+      jni, jni->NewObjectArray(kMaxSpatialLayers, int_array_class_.obj(),
>+                               nullptr /* initial */));
>   for (int spatial_i = 0; spatial_i < kMaxSpatialLayers; ++spatial_i) {
>-    jintArray j_array_spatial_layer = jni->NewIntArray(kMaxTemporalStreams);
>-    jint* array_spatial_layer =
>-        jni->GetIntArrayElements(j_array_spatial_layer, nullptr /* isCopy */);
>+    ScopedJavaLocalRef<jintArray> j_array_spatial_layer(
>+        jni, jni->NewIntArray(kMaxTemporalStreams));
>+    jint* array_spatial_layer = jni->GetIntArrayElements(
>+        j_array_spatial_layer.obj(), nullptr /* isCopy */);
>     for (int temporal_i = 0; temporal_i < kMaxTemporalStreams; ++temporal_i) {
>       array_spatial_layer[temporal_i] =
>           allocation.GetBitrate(spatial_i, temporal_i);
>     }
>-    jni->ReleaseIntArrayElements(j_array_spatial_layer, array_spatial_layer,
>-                                 JNI_COMMIT);
>+    jni->ReleaseIntArrayElements(j_array_spatial_layer.obj(),
>+                                 array_spatial_layer, JNI_COMMIT);
> 
>-    jni->SetObjectArrayElement(j_allocation_array, spatial_i,
>-                               j_array_spatial_layer);
>+    jni->SetObjectArrayElement(j_allocation_array.obj(), spatial_i,
>+                               j_array_spatial_layer.obj());
>   }
>   return Java_BitrateAllocation_Constructor(jni, j_allocation_array);
> }
> 
> std::string VideoEncoderWrapper::GetImplementationName(JNIEnv* jni) const {
>-  jstring jname = Java_VideoEncoder_getImplementationName(jni, *encoder_);
>-  return JavaToStdString(jni, jname);
>+  return JavaToStdString(
>+      jni, Java_VideoEncoder_getImplementationName(jni, encoder_));
>+}
>+
>+std::unique_ptr<VideoEncoder> JavaToNativeVideoEncoder(
>+    JNIEnv* jni,
>+    const JavaRef<jobject>& j_encoder) {
>+  const jlong native_encoder =
>+      Java_VideoEncoder_createNativeVideoEncoder(jni, j_encoder);
>+  VideoEncoder* encoder;
>+  if (native_encoder == 0) {
>+    encoder = new VideoEncoderWrapper(jni, j_encoder);
>+  } else {
>+    encoder = reinterpret_cast<VideoEncoder*>(native_encoder);
>+  }
>+  return std::unique_ptr<VideoEncoder>(encoder);
>+}
>+
>+bool IsHardwareVideoEncoder(JNIEnv* jni, const JavaRef<jobject>& j_encoder) {
>+  return Java_VideoEncoder_isHardwareEncoder(jni, j_encoder);
> }
> 
> }  // namespace jni
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videoencoderwrapper.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videoencoderwrapper.h
>index 629f28a6ec1..c30560503cb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videoencoderwrapper.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videoencoderwrapper.h
>@@ -29,7 +29,8 @@ namespace jni {
> // Wraps a Java encoder and delegates all calls to it.
> class VideoEncoderWrapper : public VideoEncoder {
>  public:
>-  VideoEncoderWrapper(JNIEnv* jni, jobject j_encoder);
>+  VideoEncoderWrapper(JNIEnv* jni, const JavaRef<jobject>& j_encoder);
>+  ~VideoEncoderWrapper() override;
> 
>   int32_t InitEncode(const VideoCodec* codec_settings,
>                      int32_t number_of_cores,
>@@ -46,30 +47,30 @@ class VideoEncoderWrapper : public VideoEncoder {
> 
>   int32_t SetChannelParameters(uint32_t packet_loss, int64_t rtt) override;
> 
>-  int32_t SetRateAllocation(const BitrateAllocation& allocation,
>+  int32_t SetRateAllocation(const VideoBitrateAllocation& allocation,
>                             uint32_t framerate) override;
> 
>   ScalingSettings GetScalingSettings() const override;
> 
>-  bool SupportsNativeHandle() const override { return true; }
>+  bool SupportsNativeHandle() const override;
> 
>   // Should only be called by JNI.
>   void OnEncodedFrame(JNIEnv* jni,
>-                      jobject j_caller,
>-                      jobject j_buffer,
>+                      const JavaRef<jobject>& j_caller,
>+                      const JavaRef<jobject>& j_buffer,
>                       jint encoded_width,
>                       jint encoded_height,
>                       jlong capture_time_ms,
>                       jint frame_type,
>                       jint rotation,
>                       jboolean complete_frame,
>-                      jobject j_qp);
>+                      const JavaRef<jobject>& j_qp);
> 
>   const char* ImplementationName() const override;
> 
>  private:
>   struct FrameExtraInfo {
>-    uint64_t capture_time_ns;  // Used as an identifier of the frame.
>+    int64_t capture_time_ns;  // Used as an identifier of the frame.
> 
>     uint32_t timestamp_rtp;
>   };
>@@ -78,19 +79,21 @@ class VideoEncoderWrapper : public VideoEncoder {
> 
>   // Takes Java VideoCodecStatus, handles it and returns WEBRTC_VIDEO_CODEC_*
>   // status code.
>-  int32_t HandleReturnCode(JNIEnv* jni, jobject code);
>+  int32_t HandleReturnCode(JNIEnv* jni,
>+                           const JavaRef<jobject>& j_value,
>+                           const char* method_name);
> 
>   RTPFragmentationHeader ParseFragmentationHeader(
>       const std::vector<uint8_t>& buffer);
>   int ParseQp(const std::vector<uint8_t>& buffer);
>   CodecSpecificInfo ParseCodecSpecificInfo(const EncodedImage& frame);
>-  jobject ToJavaBitrateAllocation(JNIEnv* jni,
>-                                  const BitrateAllocation& allocation);
>+  ScopedJavaLocalRef<jobject> ToJavaBitrateAllocation(
>+      JNIEnv* jni,
>+      const VideoBitrateAllocation& allocation);
>   std::string GetImplementationName(JNIEnv* jni) const;
> 
>-  const ScopedGlobalRef<jobject> encoder_;
>-  const ScopedGlobalRef<jclass> frame_type_class_;
>-  const ScopedGlobalRef<jclass> int_array_class_;
>+  const ScopedJavaGlobalRef<jobject> encoder_;
>+  const ScopedJavaGlobalRef<jclass> int_array_class_;
> 
>   std::string implementation_name_;
> 
>@@ -103,16 +106,21 @@ class VideoEncoderWrapper : public VideoEncoder {
>   VideoCodec codec_settings_;
>   H264BitstreamParser h264_bitstream_parser_;
> 
>-  // RTP state.
>-  uint16_t picture_id_;
>-  uint8_t tl0_pic_idx_;
>-
>   // VP9 variables to populate codec specific structure.
>   GofInfoVP9 gof_;  // Contains each frame's temporal information for
>                     // non-flexible VP9 mode.
>   size_t gof_idx_;
> };
> 
>+/* If the j_encoder is a wrapped native encoder, unwrap it. If it is not,
>+ * wrap it in a VideoEncoderWrapper.
>+ */
>+std::unique_ptr<VideoEncoder> JavaToNativeVideoEncoder(
>+    JNIEnv* jni,
>+    const JavaRef<jobject>& j_encoder);
>+
>+bool IsHardwareVideoEncoder(JNIEnv* jni, const JavaRef<jobject>& j_encoder);
>+
> }  // namespace jni
> }  // namespace webrtc
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videofilerenderer_jni.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videofilerenderer_jni.cc
>deleted file mode 100644
>index 09623f39622..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videofilerenderer_jni.cc
>+++ /dev/null
>@@ -1,69 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include <jni.h>
>-
>-#include "third_party/libyuv/include/libyuv/scale.h"
>-#include "rtc_base/checks.h"
>-#include "rtc_base/logging.h"
>-
>-namespace webrtc {
>-namespace jni {
>-
>-extern "C" JNIEXPORT void JNICALL
>-Java_org_webrtc_VideoFileRenderer_nativeI420Scale(JNIEnv* jni,
>-                                                  jclass,
>-                                                  jobject j_src_buffer_y,
>-                                                  jint j_src_stride_y,
>-                                                  jobject j_src_buffer_u,
>-                                                  jint j_src_stride_u,
>-                                                  jobject j_src_buffer_v,
>-                                                  jint j_src_stride_v,
>-                                                  jint width,
>-                                                  jint height,
>-                                                  jbyteArray j_dst_buffer,
>-                                                  jint dstWidth,
>-                                                  jint dstHeight) {
>-  size_t src_size_y = jni->GetDirectBufferCapacity(j_src_buffer_y);
>-  size_t src_size_u = jni->GetDirectBufferCapacity(j_src_buffer_u);
>-  size_t src_size_v = jni->GetDirectBufferCapacity(j_src_buffer_v);
>-  size_t dst_size = jni->GetDirectBufferCapacity(j_dst_buffer);
>-  int dst_stride = dstWidth;
>-  RTC_CHECK_GE(src_size_y, j_src_stride_y * height);
>-  RTC_CHECK_GE(src_size_u, j_src_stride_u * height / 4);
>-  RTC_CHECK_GE(src_size_v, j_src_stride_v * height / 4);
>-  RTC_CHECK_GE(dst_size, dst_stride * dstHeight * 3 / 2);
>-  uint8_t* src_y =
>-      reinterpret_cast<uint8_t*>(jni->GetDirectBufferAddress(j_src_buffer_y));
>-  uint8_t* src_u =
>-      reinterpret_cast<uint8_t*>(jni->GetDirectBufferAddress(j_src_buffer_u));
>-  uint8_t* src_v =
>-      reinterpret_cast<uint8_t*>(jni->GetDirectBufferAddress(j_src_buffer_v));
>-  uint8_t* dst =
>-      reinterpret_cast<uint8_t*>(jni->GetDirectBufferAddress(j_dst_buffer));
>-
>-  uint8_t* dst_y = dst;
>-  size_t dst_stride_y = dst_stride;
>-  uint8_t* dst_u = dst + dst_stride * dstHeight;
>-  size_t dst_stride_u = dst_stride / 2;
>-  uint8_t* dst_v = dst + dst_stride * dstHeight * 5 / 4;
>-  size_t dst_stride_v = dst_stride / 2;
>-
>-  int ret = libyuv::I420Scale(
>-      src_y, j_src_stride_y, src_u, j_src_stride_u, src_v, j_src_stride_v,
>-      width, height, dst_y, dst_stride_y, dst_u, dst_stride_u, dst_v,
>-      dst_stride_v, dstWidth, dstHeight, libyuv::kFilterBilinear);
>-  if (ret) {
>-    RTC_LOG(LS_ERROR) << "Error scaling I420 frame: " << ret;
>-  }
>-}
>-
>-}  // namespace jni
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videoframe.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videoframe.cc
>index 5dbc5fbb8f8..dd13cb3522c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videoframe.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videoframe.cc
>@@ -13,7 +13,6 @@
> #include <memory>
> 
> #include "common_video/include/video_frame_buffer.h"
>-#include "libyuv/scale.h"
> #include "rtc_base/bind.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/keep_ref_until_done.h"
>@@ -21,10 +20,8 @@
> #include "rtc_base/scoped_ref_ptr.h"
> #include "rtc_base/timeutils.h"
> #include "sdk/android/generated_video_jni/jni/VideoFrame_jni.h"
>-#include "sdk/android/src/jni/classreferenceholder.h"
> #include "sdk/android/src/jni/jni_helpers.h"
> #include "sdk/android/src/jni/wrapped_native_i420_buffer.h"
>-#include "system_wrappers/include/aligned_malloc.h"
> 
> namespace webrtc {
> namespace jni {
>@@ -36,8 +33,11 @@ class AndroidVideoI420Buffer : public I420BufferInterface {
>   // Adopts and takes ownership of the Java VideoFrame.Buffer. I.e. retain()
>   // will not be called, but release() will be called when the returned
>   // AndroidVideoBuffer is destroyed.
>-  static rtc::scoped_refptr<AndroidVideoI420Buffer>
>-  Adopt(JNIEnv* jni, int width, int height, jobject j_video_frame_buffer);
>+  static rtc::scoped_refptr<AndroidVideoI420Buffer> Adopt(
>+      JNIEnv* jni,
>+      int width,
>+      int height,
>+      const JavaRef<jobject>& j_video_frame_buffer);
> 
>  protected:
>   // Should not be called directly. Adopts the buffer. Use Adopt() instead for
>@@ -45,8 +45,8 @@ class AndroidVideoI420Buffer : public I420BufferInterface {
>   AndroidVideoI420Buffer(JNIEnv* jni,
>                          int width,
>                          int height,
>-                         jobject j_video_frame_buffer);
>-  ~AndroidVideoI420Buffer();
>+                         const JavaRef<jobject>& j_video_frame_buffer);
>+  ~AndroidVideoI420Buffer() override;
> 
>  private:
>   const uint8_t* DataY() const override { return data_y_; }
>@@ -63,7 +63,7 @@ class AndroidVideoI420Buffer : public I420BufferInterface {
>   const int width_;
>   const int height_;
>   // Holds a VideoFrame.I420Buffer.
>-  const ScopedGlobalRef<jobject> j_video_frame_buffer_;
>+  const ScopedJavaGlobalRef<jobject> j_video_frame_buffer_;
> 
>   const uint8_t* data_y_;
>   const uint8_t* data_u_;
>@@ -77,25 +77,32 @@ rtc::scoped_refptr<AndroidVideoI420Buffer> AndroidVideoI420Buffer::Adopt(
>     JNIEnv* jni,
>     int width,
>     int height,
>-    jobject j_video_frame_buffer) {
>+    const JavaRef<jobject>& j_video_frame_buffer) {
>   return new rtc::RefCountedObject<AndroidVideoI420Buffer>(
>       jni, width, height, j_video_frame_buffer);
> }
> 
>-AndroidVideoI420Buffer::AndroidVideoI420Buffer(JNIEnv* jni,
>-                                               int width,
>-                                               int height,
>-                                               jobject j_video_frame_buffer)
>+AndroidVideoI420Buffer::AndroidVideoI420Buffer(
>+    JNIEnv* jni,
>+    int width,
>+    int height,
>+    const JavaRef<jobject>& j_video_frame_buffer)
>     : width_(width),
>       height_(height),
>       j_video_frame_buffer_(jni, j_video_frame_buffer) {
>-  jobject j_data_y = Java_I420Buffer_getDataY(jni, j_video_frame_buffer);
>-  jobject j_data_u = Java_I420Buffer_getDataU(jni, j_video_frame_buffer);
>-  jobject j_data_v = Java_I420Buffer_getDataV(jni, j_video_frame_buffer);
>-
>-  data_y_ = static_cast<const uint8_t*>(jni->GetDirectBufferAddress(j_data_y));
>-  data_u_ = static_cast<const uint8_t*>(jni->GetDirectBufferAddress(j_data_u));
>-  data_v_ = static_cast<const uint8_t*>(jni->GetDirectBufferAddress(j_data_v));
>+  ScopedJavaLocalRef<jobject> j_data_y =
>+      Java_I420Buffer_getDataY(jni, j_video_frame_buffer);
>+  ScopedJavaLocalRef<jobject> j_data_u =
>+      Java_I420Buffer_getDataU(jni, j_video_frame_buffer);
>+  ScopedJavaLocalRef<jobject> j_data_v =
>+      Java_I420Buffer_getDataV(jni, j_video_frame_buffer);
>+
>+  data_y_ =
>+      static_cast<const uint8_t*>(jni->GetDirectBufferAddress(j_data_y.obj()));
>+  data_u_ =
>+      static_cast<const uint8_t*>(jni->GetDirectBufferAddress(j_data_u.obj()));
>+  data_v_ =
>+      static_cast<const uint8_t*>(jni->GetDirectBufferAddress(j_data_v.obj()));
> 
>   stride_y_ = Java_I420Buffer_getStrideY(jni, j_video_frame_buffer);
>   stride_u_ = Java_I420Buffer_getStrideU(jni, j_video_frame_buffer);
>@@ -104,215 +111,45 @@ AndroidVideoI420Buffer::AndroidVideoI420Buffer(JNIEnv* jni,
> 
> AndroidVideoI420Buffer::~AndroidVideoI420Buffer() {
>   JNIEnv* jni = AttachCurrentThreadIfNeeded();
>-  Java_Buffer_release(jni, *j_video_frame_buffer_);
>+  Java_Buffer_release(jni, j_video_frame_buffer_);
> }
> 
> }  // namespace
> 
>-int64_t GetJavaVideoFrameTimestampNs(JNIEnv* jni, jobject j_video_frame) {
>+int64_t GetJavaVideoFrameTimestampNs(JNIEnv* jni,
>+                                     const JavaRef<jobject>& j_video_frame) {
>   return Java_VideoFrame_getTimestampNs(jni, j_video_frame);
> }
> 
>-Matrix::Matrix(JNIEnv* jni, jfloatArray a) {
>-  RTC_CHECK_EQ(16, jni->GetArrayLength(a));
>-  jfloat* ptr = jni->GetFloatArrayElements(a, nullptr);
>-  for (int i = 0; i < 16; ++i) {
>-    elem_[i] = ptr[i];
>-  }
>-  jni->ReleaseFloatArrayElements(a, ptr, 0);
>-}
>-
>-jfloatArray Matrix::ToJava(JNIEnv* jni) const {
>-  jfloatArray matrix = jni->NewFloatArray(16);
>-  jni->SetFloatArrayRegion(matrix, 0, 16, elem_);
>-  return matrix;
>-}
>-
>-void Matrix::Rotate(VideoRotation rotation) {
>-  // Texture coordinates are in the range 0 to 1. The transformation of the last
>-  // row in each rotation matrix is needed for proper translation, e.g, to
>-  // mirror x, we don't replace x by -x, but by 1-x.
>-  switch (rotation) {
>-    case kVideoRotation_0:
>-      break;
>-    case kVideoRotation_90: {
>-      const float ROTATE_90[16] =
>-          { elem_[4], elem_[5], elem_[6], elem_[7],
>-            -elem_[0], -elem_[1], -elem_[2], -elem_[3],
>-            elem_[8], elem_[9], elem_[10], elem_[11],
>-            elem_[0] + elem_[12], elem_[1] + elem_[13],
>-            elem_[2] + elem_[14], elem_[3] + elem_[15]};
>-      memcpy(elem_, ROTATE_90, sizeof(elem_));
>-    } break;
>-    case kVideoRotation_180: {
>-      const float ROTATE_180[16] =
>-          { -elem_[0], -elem_[1], -elem_[2], -elem_[3],
>-            -elem_[4], -elem_[5], -elem_[6], -elem_[7],
>-            elem_[8], elem_[9], elem_[10], elem_[11],
>-            elem_[0] + elem_[4] + elem_[12], elem_[1] + elem_[5] + elem_[13],
>-            elem_[2] + elem_[6] + elem_[14], elem_[3] + elem_[11]+ elem_[15]};
>-        memcpy(elem_, ROTATE_180, sizeof(elem_));
>-    } break;
>-    case kVideoRotation_270: {
>-      const float ROTATE_270[16] =
>-          { -elem_[4], -elem_[5], -elem_[6], -elem_[7],
>-            elem_[0], elem_[1], elem_[2], elem_[3],
>-            elem_[8], elem_[9], elem_[10], elem_[11],
>-            elem_[4] + elem_[12], elem_[5] + elem_[13],
>-            elem_[6] + elem_[14], elem_[7] + elem_[15]};
>-      memcpy(elem_, ROTATE_270, sizeof(elem_));
>-    } break;
>-  }
>-}
>-
>-// Calculates result = a * b, in column-major order.
>-void Matrix::Multiply(const float a[16], const float b[16], float result[16]) {
>-  for (int i = 0; i < 4; ++i) {
>-    for (int j = 0; j < 4; ++j) {
>-      float sum = 0;
>-      for (int k = 0; k < 4; ++k) {
>-        sum += a[k * 4 + j] * b[i * 4 + k];
>-      }
>-      result[i * 4 + j] = sum;
>-    }
>-  }
>-}
>-
>-// Center crop by keeping xFraction of the width and yFraction of the height,
>-// so e.g. cropping from 640x480 to 640x360 would use
>-// xFraction=1, yFraction=360/480.
>-void Matrix::Crop(float xFraction,
>-                  float yFraction,
>-                  float xOffset,
>-                  float yOffset) {
>-  const float crop_matrix[16] =
>-      {xFraction, 0, 0, 0,
>-       0, yFraction, 0, 0,
>-       0, 0, 1, 0,
>-       xOffset, yOffset, 0, 1};
>-  const Matrix old = *this;
>-  Multiply(crop_matrix, old.elem_, this->elem_);
>-}
>-
>-// Aligning pointer to 64 bytes for improved performance, e.g. use SIMD.
>-static const int kBufferAlignment = 64;
>-
>-NativeHandleImpl::NativeHandleImpl(int id, const Matrix& matrix)
>-    : oes_texture_id(id), sampling_matrix(matrix) {}
>-
>-NativeHandleImpl::NativeHandleImpl(JNIEnv* jni,
>-                                   jint j_oes_texture_id,
>-                                   jfloatArray j_transform_matrix)
>-    : oes_texture_id(j_oes_texture_id),
>-      sampling_matrix(jni, j_transform_matrix) {}
>-
>-AndroidTextureBuffer::AndroidTextureBuffer(
>-    int width,
>-    int height,
>-    const NativeHandleImpl& native_handle,
>-    jobject surface_texture_helper,
>-    const rtc::Callback0<void>& no_longer_used)
>-    : width_(width),
>-      height_(height),
>-      native_handle_(native_handle),
>-      surface_texture_helper_(surface_texture_helper),
>-      no_longer_used_cb_(no_longer_used) {}
>-
>-AndroidTextureBuffer::~AndroidTextureBuffer() {
>-  no_longer_used_cb_();
>-}
>-
>-VideoFrameBuffer::Type AndroidTextureBuffer::type() const {
>-  return Type::kNative;
>-}
>-
>-NativeHandleImpl AndroidTextureBuffer::native_handle_impl() const {
>-  return native_handle_;
>-}
>-
>-int AndroidTextureBuffer::width() const {
>-  return width_;
>-}
>-
>-int AndroidTextureBuffer::height() const {
>-  return height_;
>-}
>-
>-rtc::scoped_refptr<I420BufferInterface> AndroidTextureBuffer::ToI420() {
>-  int uv_width = (width() + 7) / 8;
>-  int stride = 8 * uv_width;
>-  int uv_height = (height() + 1) / 2;
>-  size_t size = stride * (height() + uv_height);
>-  // The data is owned by the frame, and the normal case is that the
>-  // data is deleted by the frame's destructor callback.
>-  //
>-  // TODO(nisse): Use an I420BufferPool. We then need to extend that
>-  // class, and I420Buffer, to support our memory layout.
>-  // TODO(nisse): Depending on
>-  // system_wrappers/include/aligned_malloc.h violate current DEPS
>-  // rules. We get away for now only because it is indirectly included
>-  // by i420_buffer.h
>-  std::unique_ptr<uint8_t, AlignedFreeDeleter> yuv_data(
>-      static_cast<uint8_t*>(AlignedMalloc(size, kBufferAlignment)));
>-  // See YuvConverter.java for the required layout.
>-  uint8_t* y_data = yuv_data.get();
>-  uint8_t* u_data = y_data + height() * stride;
>-  uint8_t* v_data = u_data + stride / 2;
>-
>-  rtc::scoped_refptr<I420BufferInterface> copy = webrtc::WrapI420Buffer(
>-      width(), height(), y_data, stride, u_data, stride, v_data, stride,
>-      rtc::Bind(&AlignedFree, yuv_data.release()));
>-
>-  JNIEnv* jni = AttachCurrentThreadIfNeeded();
>-  ScopedLocalRefFrame local_ref_frame(jni);
>-
>-  // TODO(sakal): This call to a deperecated method will be removed when
>-  // AndroidTextureBuffer is removed.
>-  jmethodID transform_mid = GetMethodID(
>-      jni,
>-      GetObjectClass(jni, surface_texture_helper_),
>-      "textureToYUV",
>-      "(Ljava/nio/ByteBuffer;IIII[F)V");
>-
>-  jobject byte_buffer = jni->NewDirectByteBuffer(y_data, size);
>-
>-  jfloatArray sampling_matrix = native_handle_.sampling_matrix.ToJava(jni);
>-  jni->CallVoidMethod(surface_texture_helper_,
>-                      transform_mid,
>-                      byte_buffer, width(), height(), stride,
>-                      native_handle_.oes_texture_id, sampling_matrix);
>-  CHECK_EXCEPTION(jni) << "textureToYUV throwed an exception";
>-
>-  return copy;
>-}
>-
> rtc::scoped_refptr<AndroidVideoBuffer> AndroidVideoBuffer::Adopt(
>     JNIEnv* jni,
>-    jobject j_video_frame_buffer) {
>+    const JavaRef<jobject>& j_video_frame_buffer) {
>   return new rtc::RefCountedObject<AndroidVideoBuffer>(jni,
>                                                        j_video_frame_buffer);
> }
> 
> rtc::scoped_refptr<AndroidVideoBuffer> AndroidVideoBuffer::Create(
>     JNIEnv* jni,
>-    jobject j_video_frame_buffer) {
>+    const JavaRef<jobject>& j_video_frame_buffer) {
>   Java_Buffer_retain(jni, j_video_frame_buffer);
>   return Adopt(jni, j_video_frame_buffer);
> }
> 
>-AndroidVideoBuffer::AndroidVideoBuffer(JNIEnv* jni,
>-                                       jobject j_video_frame_buffer)
>+AndroidVideoBuffer::AndroidVideoBuffer(
>+    JNIEnv* jni,
>+    const JavaRef<jobject>& j_video_frame_buffer)
>     : width_(Java_Buffer_getWidth(jni, j_video_frame_buffer)),
>       height_(Java_Buffer_getHeight(jni, j_video_frame_buffer)),
>       j_video_frame_buffer_(jni, j_video_frame_buffer) {}
> 
> AndroidVideoBuffer::~AndroidVideoBuffer() {
>   JNIEnv* jni = AttachCurrentThreadIfNeeded();
>-  Java_Buffer_release(jni, *j_video_frame_buffer_);
>+  Java_Buffer_release(jni, j_video_frame_buffer_);
> }
> 
>-jobject AndroidVideoBuffer::video_frame_buffer() const {
>-  return *j_video_frame_buffer_;
>+const ScopedJavaGlobalRef<jobject>& AndroidVideoBuffer::video_frame_buffer()
>+    const {
>+  return j_video_frame_buffer_;
> }
> 
> rtc::scoped_refptr<AndroidVideoBuffer> AndroidVideoBuffer::CropAndScale(
>@@ -323,9 +160,9 @@ rtc::scoped_refptr<AndroidVideoBuffer> AndroidVideoBuffer::CropAndScale(
>     int crop_height,
>     int scale_width,
>     int scale_height) {
>-  return Adopt(jni, Java_Buffer_cropAndScale(
>-                        jni, *j_video_frame_buffer_, crop_x, crop_y, crop_width,
>-                        crop_height, scale_width, scale_height));
>+  return Adopt(jni, Java_Buffer_cropAndScale(jni, j_video_frame_buffer_, crop_x,
>+                                             crop_y, crop_width, crop_height,
>+                                             scale_width, scale_height));
> }
> 
> VideoFrameBuffer::Type AndroidVideoBuffer::type() const {
>@@ -342,33 +179,19 @@ int AndroidVideoBuffer::height() const {
> 
> rtc::scoped_refptr<I420BufferInterface> AndroidVideoBuffer::ToI420() {
>   JNIEnv* jni = AttachCurrentThreadIfNeeded();
>-  ScopedLocalRefFrame local_ref_frame(jni);
>-  jobject j_i420_buffer = Java_Buffer_toI420(jni, *j_video_frame_buffer_);
>+  ScopedJavaLocalRef<jobject> j_i420_buffer =
>+      Java_Buffer_toI420(jni, j_video_frame_buffer_);
> 
>   // We don't need to retain the buffer because toI420 returns a new object that
>   // we are assumed to take the ownership of.
>   return AndroidVideoI420Buffer::Adopt(jni, width_, height_, j_i420_buffer);
> }
> 
>-jobject AndroidVideoBuffer::ToJavaI420Frame(JNIEnv* jni, int rotation) {
>-  jclass j_byte_buffer_class = jni->FindClass("java/nio/ByteBuffer");
>-  jclass j_i420_frame_class =
>-      FindClass(jni, "org/webrtc/VideoRenderer$I420Frame");
>-  jmethodID j_i420_frame_ctor_id = GetMethodID(
>-      jni, j_i420_frame_class, "<init>", "(ILorg/webrtc/VideoFrame$Buffer;J)V");
>-  // Java code just uses the native frame to hold a reference to the buffer so
>-  // this is okay.
>-  VideoFrame* native_frame =
>-      new VideoFrame(this, 0 /* timestamp */, 0 /* render_time_ms */,
>-                     VideoRotation::kVideoRotation_0 /* rotation */);
>-  return jni->NewObject(j_i420_frame_class, j_i420_frame_ctor_id, rotation,
>-                        *j_video_frame_buffer_, jlongFromPointer(native_frame));
>-}
>-
> VideoFrame JavaToNativeFrame(JNIEnv* jni,
>-                             jobject j_video_frame,
>+                             const JavaRef<jobject>& j_video_frame,
>                              uint32_t timestamp_rtp) {
>-  jobject j_video_frame_buffer = Java_VideoFrame_getBuffer(jni, j_video_frame);
>+  ScopedJavaLocalRef<jobject> j_video_frame_buffer =
>+      Java_VideoFrame_getBuffer(jni, j_video_frame);
>   int rotation = Java_VideoFrame_getRotation(jni, j_video_frame);
>   int64_t timestamp_ns = Java_VideoFrame_getTimestampNs(jni, j_video_frame);
>   rtc::scoped_refptr<AndroidVideoBuffer> buffer =
>@@ -378,77 +201,31 @@ VideoFrame JavaToNativeFrame(JNIEnv* jni,
>                     static_cast<VideoRotation>(rotation));
> }
> 
>-static bool IsJavaVideoBuffer(rtc::scoped_refptr<VideoFrameBuffer> buffer) {
>-  if (buffer->type() != VideoFrameBuffer::Type::kNative) {
>-    return false;
>-  }
>-  AndroidVideoFrameBuffer* android_buffer =
>-      static_cast<AndroidVideoFrameBuffer*>(buffer.get());
>-  return android_buffer->android_type() ==
>-         AndroidVideoFrameBuffer::AndroidType::kJavaBuffer;
>-}
>-
>-jobject NativeToJavaFrame(JNIEnv* jni, const VideoFrame& frame) {
>+ScopedJavaLocalRef<jobject> NativeToJavaVideoFrame(JNIEnv* jni,
>+                                                   const VideoFrame& frame) {
>   rtc::scoped_refptr<VideoFrameBuffer> buffer = frame.video_frame_buffer();
>-  jobject j_buffer;
>-  if (IsJavaVideoBuffer(buffer)) {
>-    RTC_DCHECK(buffer->type() == VideoFrameBuffer::Type::kNative);
>-    AndroidVideoFrameBuffer* android_buffer =
>-        static_cast<AndroidVideoFrameBuffer*>(buffer.get());
>-    RTC_DCHECK(android_buffer->android_type() ==
>-               AndroidVideoFrameBuffer::AndroidType::kJavaBuffer);
>-    AndroidVideoBuffer* android_video_buffer =
>-        static_cast<AndroidVideoBuffer*>(android_buffer);
>-    j_buffer = android_video_buffer->video_frame_buffer();
>+
>+  if (buffer->type() == VideoFrameBuffer::Type::kNative) {
>+    AndroidVideoBuffer* android_buffer =
>+        static_cast<AndroidVideoBuffer*>(buffer.get());
>+    ScopedJavaLocalRef<jobject> j_video_frame_buffer(
>+        jni, android_buffer->video_frame_buffer());
>+    Java_Buffer_retain(jni, j_video_frame_buffer);
>+    return Java_VideoFrame_Constructor(
>+        jni, j_video_frame_buffer, static_cast<jint>(frame.rotation()),
>+        static_cast<jlong>(frame.timestamp_us() *
>+                           rtc::kNumNanosecsPerMicrosec));
>   } else {
>-    j_buffer = WrapI420Buffer(jni, buffer->ToI420());
>+    return Java_VideoFrame_Constructor(
>+        jni, WrapI420Buffer(jni, buffer->ToI420()),
>+        static_cast<jint>(frame.rotation()),
>+        static_cast<jlong>(frame.timestamp_us() *
>+                           rtc::kNumNanosecsPerMicrosec));
>   }
>-  return Java_VideoFrame_Constructor(
>-      jni, j_buffer, static_cast<jint>(frame.rotation()),
>-      static_cast<jlong>(frame.timestamp_us() * rtc::kNumNanosecsPerMicrosec));
> }
> 
>-extern "C" JNIEXPORT void JNICALL
>-Java_org_webrtc_VideoFrame_cropAndScaleI420Native(JNIEnv* jni,
>-                                                  jclass,
>-                                                  jobject j_src_y,
>-                                                  jint src_stride_y,
>-                                                  jobject j_src_u,
>-                                                  jint src_stride_u,
>-                                                  jobject j_src_v,
>-                                                  jint src_stride_v,
>-                                                  jint crop_x,
>-                                                  jint crop_y,
>-                                                  jint crop_width,
>-                                                  jint crop_height,
>-                                                  jobject j_dst_y,
>-                                                  jint dst_stride_y,
>-                                                  jobject j_dst_u,
>-                                                  jint dst_stride_u,
>-                                                  jobject j_dst_v,
>-                                                  jint dst_stride_v,
>-                                                  jint scale_width,
>-                                                  jint scale_height) {
>-  uint8_t const* src_y =
>-      static_cast<uint8_t*>(jni->GetDirectBufferAddress(j_src_y));
>-  uint8_t const* src_u =
>-      static_cast<uint8_t*>(jni->GetDirectBufferAddress(j_src_u));
>-  uint8_t const* src_v =
>-      static_cast<uint8_t*>(jni->GetDirectBufferAddress(j_src_v));
>-  uint8_t* dst_y = static_cast<uint8_t*>(jni->GetDirectBufferAddress(j_dst_y));
>-  uint8_t* dst_u = static_cast<uint8_t*>(jni->GetDirectBufferAddress(j_dst_u));
>-  uint8_t* dst_v = static_cast<uint8_t*>(jni->GetDirectBufferAddress(j_dst_v));
>-
>-  // Perform cropping using pointer arithmetic.
>-  src_y += crop_x + crop_y * src_stride_y;
>-  src_u += crop_x / 2 + crop_y / 2 * src_stride_u;
>-  src_v += crop_x / 2 + crop_y / 2 * src_stride_v;
>-
>-  bool ret = libyuv::I420Scale(
>-      src_y, src_stride_y, src_u, src_stride_u, src_v, src_stride_v, crop_width,
>-      crop_height, dst_y, dst_stride_y, dst_u, dst_stride_u, dst_v,
>-      dst_stride_v, scale_width, scale_height, libyuv::kFilterBox);
>-  RTC_DCHECK_EQ(ret, 0) << "I420Scale failed";
>+void ReleaseJavaVideoFrame(JNIEnv* jni, const JavaRef<jobject>& j_video_frame) {
>+  Java_VideoFrame_release(jni, j_video_frame);
> }
> 
> }  // namespace jni
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videoframe.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videoframe.h
>index 26206b8631a..107e03e17bf 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videoframe.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videoframe.h
>@@ -22,103 +22,23 @@
> namespace webrtc {
> namespace jni {
> 
>-// Open gl texture matrix, in column-major order. Operations are
>-// in-place.
>-class Matrix {
>- public:
>-  Matrix(JNIEnv* jni, jfloatArray a);
>-
>-  static Matrix fromAndroidGraphicsMatrix(JNIEnv* jni, jobject j_matrix);
>-
>-  jfloatArray ToJava(JNIEnv* jni) const;
>-
>-  // Crop arguments are relative to original size.
>-  void Crop(float cropped_width,
>-            float cropped_height,
>-            float crop_x,
>-            float crop_y);
>-
>-  void Rotate(VideoRotation rotation);
>-
>- private:
>-  Matrix() {}
>-
>-  static void Multiply(const float a[16], const float b[16], float result[16]);
>-  float elem_[16];
>-};
>-
>-// Wrapper for texture object.
>-struct NativeHandleImpl {
>-  NativeHandleImpl(JNIEnv* jni,
>-                   jint j_oes_texture_id,
>-                   jfloatArray j_transform_matrix);
>-
>-  NativeHandleImpl(int id, const Matrix& matrix);
>-
>-  const int oes_texture_id;
>-  Matrix sampling_matrix;
>-};
>-
>-// Base class to differentiate between the old texture frames and the new
>-// Java-based frames.
>-// TODO(sakal): Remove this and AndroidTextureBuffer once they are no longer
>-// needed.
>-class AndroidVideoFrameBuffer : public VideoFrameBuffer {
>- public:
>-  enum class AndroidType { kTextureBuffer, kJavaBuffer };
>-
>-  virtual AndroidType android_type() = 0;
>-};
>-
>-class AndroidTextureBuffer : public AndroidVideoFrameBuffer {
>- public:
>-  AndroidTextureBuffer(int width,
>-                       int height,
>-                       const NativeHandleImpl& native_handle,
>-                       jobject surface_texture_helper,
>-                       const rtc::Callback0<void>& no_longer_used);
>-  ~AndroidTextureBuffer();
>-
>-  NativeHandleImpl native_handle_impl() const;
>-
>- private:
>-  Type type() const override;
>-  int width() const override;
>-  int height() const override;
>-
>-  rtc::scoped_refptr<I420BufferInterface> ToI420() override;
>-
>-  AndroidType android_type() override { return AndroidType::kTextureBuffer; }
>-
>-  const int width_;
>-  const int height_;
>-  NativeHandleImpl native_handle_;
>-  // Raw object pointer, relying on the caller, i.e.,
>-  // AndroidVideoCapturerJni or the C++ SurfaceTextureHelper, to keep
>-  // a global reference. TODO(nisse): Make this a reference to the C++
>-  // SurfaceTextureHelper instead, but that requires some refactoring
>-  // of AndroidVideoCapturerJni.
>-  jobject surface_texture_helper_;
>-  rtc::Callback0<void> no_longer_used_cb_;
>-};
>-
>-class AndroidVideoBuffer : public AndroidVideoFrameBuffer {
>+class AndroidVideoBuffer : public VideoFrameBuffer {
>  public:
>   // Creates a native VideoFrameBuffer from a Java VideoFrame.Buffer.
>   static rtc::scoped_refptr<AndroidVideoBuffer> Create(
>       JNIEnv* jni,
>-      jobject j_video_frame_buffer);
>+      const JavaRef<jobject>& j_video_frame_buffer);
> 
>   // Similar to the Create() above, but adopts and takes ownership of the Java
>   // VideoFrame.Buffer. I.e. retain() will not be called, but release() will be
>   // called when the returned AndroidVideoBuffer is destroyed.
>   static rtc::scoped_refptr<AndroidVideoBuffer> Adopt(
>       JNIEnv* jni,
>-      jobject j_video_frame_buffer);
>+      const JavaRef<jobject>& j_video_frame_buffer);
> 
>   ~AndroidVideoBuffer() override;
> 
>-  jobject video_frame_buffer() const;
>+  const ScopedJavaGlobalRef<jobject>& video_frame_buffer() const;
> 
>   // Crops a region defined by |crop_x|, |crop_y|, |crop_width| and
>   // |crop_height|. Scales it to size |scale_width| x |scale_height|.
>@@ -130,13 +50,10 @@ class AndroidVideoBuffer : public AndroidVideoFrameBuffer {
>                                                       int scale_width,
>                                                       int scale_height);
> 
>-  // Returns an instance of VideoRenderer.I420Frame (deprecated)
>-  jobject ToJavaI420Frame(JNIEnv* jni, int rotation);
>-
>  protected:
>   // Should not be called directly. Adopts the Java VideoFrame.Buffer. Use
>   // Create() or Adopt() instead for clarity.
>-  AndroidVideoBuffer(JNIEnv* jni, jobject j_video_frame_buffer);
>+  AndroidVideoBuffer(JNIEnv* jni, const JavaRef<jobject>& j_video_frame_buffer);
> 
>  private:
>   Type type() const override;
>@@ -145,21 +62,24 @@ class AndroidVideoBuffer : public AndroidVideoFrameBuffer {
> 
>   rtc::scoped_refptr<I420BufferInterface> ToI420() override;
> 
>-  AndroidType android_type() override { return AndroidType::kJavaBuffer; }
>-
>   const int width_;
>   const int height_;
>   // Holds a VideoFrame.Buffer.
>-  const ScopedGlobalRef<jobject> j_video_frame_buffer_;
>+  const ScopedJavaGlobalRef<jobject> j_video_frame_buffer_;
> };
> 
> VideoFrame JavaToNativeFrame(JNIEnv* jni,
>-                             jobject j_video_frame,
>+                             const JavaRef<jobject>& j_video_frame,
>                              uint32_t timestamp_rtp);
> 
>-jobject NativeToJavaFrame(JNIEnv* jni, const VideoFrame& frame);
>+// NOTE: Returns a new video frame that has to be released by calling
>+// ReleaseJavaVideoFrame.
>+ScopedJavaLocalRef<jobject> NativeToJavaVideoFrame(JNIEnv* jni,
>+                                                   const VideoFrame& frame);
>+void ReleaseJavaVideoFrame(JNIEnv* jni, const JavaRef<jobject>& j_video_frame);
> 
>-int64_t GetJavaVideoFrameTimestampNs(JNIEnv* jni, jobject j_video_frame);
>+int64_t GetJavaVideoFrameTimestampNs(JNIEnv* jni,
>+                                     const JavaRef<jobject>& j_video_frame);
> 
> }  // namespace jni
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videosink.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videosink.cc
>new file mode 100644
>index 00000000000..bbdda31431f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videosink.cc
>@@ -0,0 +1,32 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "sdk/android/src/jni/videosink.h"
>+
>+#include "sdk/android/generated_video_jni/jni/VideoSink_jni.h"
>+#include "sdk/android/src/jni/videoframe.h"
>+
>+namespace webrtc {
>+namespace jni {
>+
>+VideoSinkWrapper::VideoSinkWrapper(JNIEnv* jni, const JavaRef<jobject>& j_sink)
>+    : j_sink_(jni, j_sink) {}
>+
>+VideoSinkWrapper::~VideoSinkWrapper() {}
>+
>+void VideoSinkWrapper::OnFrame(const VideoFrame& frame) {
>+  JNIEnv* jni = AttachCurrentThreadIfNeeded();
>+  ScopedJavaLocalRef<jobject> j_frame = NativeToJavaVideoFrame(jni, frame);
>+  Java_VideoSink_onFrame(jni, j_sink_, j_frame);
>+  ReleaseJavaVideoFrame(jni, j_frame);
>+}
>+
>+}  // namespace jni
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videosink.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videosink.h
>new file mode 100644
>index 00000000000..7efe44405f2
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videosink.h
>@@ -0,0 +1,36 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef SDK_ANDROID_SRC_JNI_VIDEOSINK_H_
>+#define SDK_ANDROID_SRC_JNI_VIDEOSINK_H_
>+
>+#include <jni.h>
>+
>+#include "api/mediastreaminterface.h"
>+#include "sdk/android/src/jni/jni_helpers.h"
>+
>+namespace webrtc {
>+namespace jni {
>+
>+class VideoSinkWrapper : public rtc::VideoSinkInterface<VideoFrame> {
>+ public:
>+  VideoSinkWrapper(JNIEnv* jni, const JavaRef<jobject>& j_sink);
>+  ~VideoSinkWrapper() override;
>+
>+ private:
>+  void OnFrame(const VideoFrame& frame) override;
>+
>+  const ScopedJavaGlobalRef<jobject> j_sink_;
>+};
>+
>+}  // namespace jni
>+}  // namespace webrtc
>+
>+#endif  // SDK_ANDROID_SRC_JNI_VIDEOSINK_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videosource.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videosource.cc
>new file mode 100644
>index 00000000000..09b538ebc26
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videosource.cc
>@@ -0,0 +1,46 @@
>+/*
>+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/videosourceproxy.h"
>+#include "rtc_base/logging.h"
>+#include "sdk/android/generated_video_jni/jni/VideoSource_jni.h"
>+#include "sdk/android/native_api/jni/java_types.h"
>+#include "sdk/android/src/jni/androidvideotracksource.h"
>+
>+namespace webrtc {
>+namespace jni {
>+
>+namespace {
>+AndroidVideoTrackSource* AndroidVideoTrackSourceFromJavaProxy(jlong j_proxy) {
>+  auto* proxy_source = reinterpret_cast<VideoTrackSourceProxy*>(j_proxy);
>+  return reinterpret_cast<AndroidVideoTrackSource*>(proxy_source->internal());
>+}
>+}  // namespace
>+
>+static jlong JNI_VideoSource_GetInternalSource(JNIEnv* jni,
>+                                               const JavaParamRef<jclass>&,
>+                                               jlong j_source) {
>+  return NativeToJavaPointer(AndroidVideoTrackSourceFromJavaProxy(j_source));
>+}
>+
>+static void JNI_VideoSource_AdaptOutputFormat(JNIEnv* jni,
>+                                              const JavaParamRef<jclass>&,
>+                                              jlong j_source,
>+                                              jint j_width,
>+                                              jint j_height,
>+                                              jint j_fps) {
>+  RTC_LOG(LS_INFO) << "VideoSource_nativeAdaptOutputFormat";
>+  AndroidVideoTrackSource* source =
>+      AndroidVideoTrackSourceFromJavaProxy(j_source);
>+  source->OnOutputFormatRequest(j_width, j_height, j_fps);
>+}
>+
>+}  // namespace jni
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videotrack.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videotrack.cc
>new file mode 100644
>index 00000000000..0595b753cfb
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videotrack.cc
>@@ -0,0 +1,53 @@
>+/*
>+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <jni.h>
>+
>+#include "api/mediastreaminterface.h"
>+#include "sdk/android/generated_video_jni/jni/VideoTrack_jni.h"
>+#include "sdk/android/src/jni/jni_helpers.h"
>+#include "sdk/android/src/jni/videosink.h"
>+
>+namespace webrtc {
>+namespace jni {
>+
>+static void JNI_VideoTrack_AddSink(JNIEnv* jni,
>+                                   const JavaParamRef<jclass>&,
>+                                   jlong j_native_track,
>+                                   jlong j_native_sink) {
>+  reinterpret_cast<VideoTrackInterface*>(j_native_track)
>+      ->AddOrUpdateSink(
>+          reinterpret_cast<rtc::VideoSinkInterface<VideoFrame>*>(j_native_sink),
>+          rtc::VideoSinkWants());
>+}
>+
>+static void JNI_VideoTrack_RemoveSink(JNIEnv* jni,
>+                                      const JavaParamRef<jclass>&,
>+                                      jlong j_native_track,
>+                                      jlong j_native_sink) {
>+  reinterpret_cast<VideoTrackInterface*>(j_native_track)
>+      ->RemoveSink(reinterpret_cast<rtc::VideoSinkInterface<VideoFrame>*>(
>+          j_native_sink));
>+}
>+
>+static jlong JNI_VideoTrack_WrapSink(JNIEnv* jni,
>+                                     const JavaParamRef<jclass>&,
>+                                     const JavaParamRef<jobject>& sink) {
>+  return jlongFromPointer(new VideoSinkWrapper(jni, sink));
>+}
>+
>+static void JNI_VideoTrack_FreeSink(JNIEnv* jni,
>+                                    const JavaParamRef<jclass>&,
>+                                    jlong j_native_sink) {
>+  delete reinterpret_cast<rtc::VideoSinkInterface<VideoFrame>*>(j_native_sink);
>+}
>+
>+}  // namespace jni
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videotrack_jni.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videotrack_jni.cc
>deleted file mode 100644
>index 7a9dcd0c469..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/videotrack_jni.cc
>+++ /dev/null
>@@ -1,81 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include <jni.h>
>-
>-#include "api/mediastreaminterface.h"
>-#include "rtc_base/logging.h"
>-#include "sdk/android/generated_video_jni/jni/VideoSink_jni.h"
>-#include "sdk/android/src/jni/classreferenceholder.h"
>-#include "sdk/android/src/jni/jni_helpers.h"
>-#include "sdk/android/src/jni/videoframe.h"
>-
>-namespace webrtc {
>-namespace jni {
>-
>-namespace {
>-
>-class VideoSinkWrapper : public rtc::VideoSinkInterface<VideoFrame> {
>- public:
>-  VideoSinkWrapper(JNIEnv* jni, jobject j_sink);
>-  ~VideoSinkWrapper() override {}
>-
>- private:
>-  void OnFrame(const VideoFrame& frame) override;
>-
>-  const ScopedGlobalRef<jobject> j_sink_;
>-};
>-
>-VideoSinkWrapper::VideoSinkWrapper(JNIEnv* jni, jobject j_sink)
>-    : j_sink_(jni, j_sink) {}
>-
>-void VideoSinkWrapper::OnFrame(const VideoFrame& frame) {
>-  JNIEnv* jni = AttachCurrentThreadIfNeeded();
>-  ScopedLocalRefFrame local_ref_frame(jni);
>-  Java_VideoSink_onFrame(jni, *j_sink_, NativeToJavaFrame(jni, frame));
>-}
>-
>-}  // namespace
>-
>-extern "C" JNIEXPORT void JNICALL
>-Java_org_webrtc_VideoTrack_nativeAddSink(JNIEnv* jni,
>-                                         jclass,
>-                                         jlong j_native_track,
>-                                         jlong j_native_sink) {
>-  reinterpret_cast<VideoTrackInterface*>(j_native_track)
>-      ->AddOrUpdateSink(
>-          reinterpret_cast<rtc::VideoSinkInterface<VideoFrame>*>(j_native_sink),
>-          rtc::VideoSinkWants());
>-}
>-
>-extern "C" JNIEXPORT void JNICALL
>-Java_org_webrtc_VideoTrack_nativeRemoveSink(JNIEnv* jni,
>-                                            jclass,
>-                                            jlong j_native_track,
>-                                            jlong j_native_sink) {
>-  reinterpret_cast<VideoTrackInterface*>(j_native_track)
>-      ->RemoveSink(reinterpret_cast<rtc::VideoSinkInterface<VideoFrame>*>(
>-          j_native_sink));
>-}
>-
>-extern "C" JNIEXPORT jlong JNICALL
>-Java_org_webrtc_VideoTrack_nativeWrapSink(JNIEnv* jni, jclass, jobject sink) {
>-  return jlongFromPointer(new VideoSinkWrapper(jni, sink));
>-}
>-
>-extern "C" JNIEXPORT void JNICALL
>-Java_org_webrtc_VideoTrack_nativeFreeSink(JNIEnv* jni,
>-                                          jclass,
>-                                          jlong j_native_sink) {
>-  delete reinterpret_cast<rtc::VideoSinkInterface<VideoFrame>*>(j_native_sink);
>-}
>-
>-}  // namespace jni
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/vp8codec.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/vp8codec.cc
>index 000652ce804..295a03a6107 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/vp8codec.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/vp8codec.cc
>@@ -11,22 +11,20 @@
> #include <jni.h>
> 
> #include "modules/video_coding/codecs/vp8/include/vp8.h"
>+#include "sdk/android/generated_vp8_jni/jni/VP8Decoder_jni.h"
>+#include "sdk/android/generated_vp8_jni/jni/VP8Encoder_jni.h"
> #include "sdk/android/src/jni/jni_helpers.h"
> 
> namespace webrtc {
> namespace jni {
> 
>-JNI_FUNCTION_DECLARATION(jlong,
>-                         VP8Encoder_createNativeEncoder,
>-                         JNIEnv* jni,
>-                         jobject) {
>+static jlong JNI_VP8Encoder_CreateEncoder(JNIEnv* jni,
>+                                          const JavaParamRef<jclass>&) {
>   return jlongFromPointer(VP8Encoder::Create().release());
> }
> 
>-JNI_FUNCTION_DECLARATION(jlong,
>-                         VP8Decoder_createNativeDecoder,
>-                         JNIEnv* jni,
>-                         jobject) {
>+static jlong JNI_VP8Decoder_CreateDecoder(JNIEnv* jni,
>+                                          const JavaParamRef<jclass>&) {
>   return jlongFromPointer(VP8Decoder::Create().release());
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/vp9codec.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/vp9codec.cc
>index 8ec51ae336e..ca712e6d865 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/vp9codec.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/vp9codec.cc
>@@ -11,37 +11,31 @@
> #include <jni.h>
> 
> #include "modules/video_coding/codecs/vp9/include/vp9.h"
>+#include "sdk/android/generated_vp9_jni/jni/VP9Decoder_jni.h"
>+#include "sdk/android/generated_vp9_jni/jni/VP9Encoder_jni.h"
> #include "sdk/android/src/jni/jni_helpers.h"
> 
> namespace webrtc {
> namespace jni {
> 
>-JNI_FUNCTION_DECLARATION(jlong,
>-                         VP9Encoder_createNativeEncoder,
>-                         JNIEnv* jni,
>-                         jobject) {
>+static jlong JNI_VP9Encoder_CreateEncoder(JNIEnv* jni,
>+                                          const JavaParamRef<jclass>& w) {
>   return jlongFromPointer(VP9Encoder::Create().release());
> }
> 
>-JNI_FUNCTION_DECLARATION(jboolean,
>-                         VP9Encoder_isSupported,
>-                         JNIEnv* jni,
>-                         jclass) {
>-  return VP9Encoder::IsSupported();
>+static jboolean JNI_VP9Encoder_IsSupported(JNIEnv* jni,
>+                                           const JavaParamRef<jclass>&) {
>+  return !SupportedVP9Codecs().empty();
> }
> 
>-JNI_FUNCTION_DECLARATION(jlong,
>-                         VP9Decoder_createNativeDecoder,
>-                         JNIEnv* jni,
>-                         jobject) {
>+static jlong JNI_VP9Decoder_CreateDecoder(JNIEnv* jni,
>+                                          const JavaParamRef<jclass>& w) {
>   return jlongFromPointer(VP9Decoder::Create().release());
> }
> 
>-JNI_FUNCTION_DECLARATION(jboolean,
>-                         VP9Decoder_isSupported,
>-                         JNIEnv* jni,
>-                         jclass) {
>-  return VP9Decoder::IsSupported();
>+static jboolean JNI_VP9Decoder_IsSupported(JNIEnv* jni,
>+                                           const JavaParamRef<jclass>&) {
>+  return !SupportedVP9Codecs().empty();
> }
> 
> }  // namespace jni
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/wrapped_native_i420_buffer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/wrapped_native_i420_buffer.cc
>index 2a09bafa4be..eab28405531 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/wrapped_native_i420_buffer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/wrapped_native_i420_buffer.cc
>@@ -10,41 +10,30 @@
> 
> #include "sdk/android/src/jni/wrapped_native_i420_buffer.h"
> 
>-#include "sdk/android/src/jni/classreferenceholder.h"
>+#include "sdk/android/generated_video_jni/jni/WrappedNativeI420Buffer_jni.h"
> #include "sdk/android/src/jni/jni_helpers.h"
> 
> namespace webrtc {
> namespace jni {
> 
> // TODO(magjed): Write a test for this function.
>-jobject WrapI420Buffer(
>+ScopedJavaLocalRef<jobject> WrapI420Buffer(
>     JNIEnv* jni,
>     const rtc::scoped_refptr<I420BufferInterface>& i420_buffer) {
>-  jclass j_wrapped_native_i420_buffer_class =
>-      FindClass(jni, "org/webrtc/WrappedNativeI420Buffer");
>-  jmethodID j_wrapped_native_i420_buffer_ctor_id =
>-      GetMethodID(jni, j_wrapped_native_i420_buffer_class, "<init>",
>-                  "(IILjava/nio/ByteBuffer;ILjava/nio/ByteBuffer;ILjava/nio/"
>-                  "ByteBuffer;IJ)V");
>-
>-  jobject y_buffer =
>-      jni->NewDirectByteBuffer(const_cast<uint8_t*>(i420_buffer->DataY()),
>-                               i420_buffer->StrideY() * i420_buffer->height());
>-  jobject u_buffer = jni->NewDirectByteBuffer(
>-      const_cast<uint8_t*>(i420_buffer->DataU()),
>-      i420_buffer->StrideU() * i420_buffer->ChromaHeight());
>-  jobject v_buffer = jni->NewDirectByteBuffer(
>-      const_cast<uint8_t*>(i420_buffer->DataV()),
>-      i420_buffer->StrideV() * i420_buffer->ChromaHeight());
>-
>-  jobject j_wrapped_native_i420_buffer = jni->NewObject(
>-      j_wrapped_native_i420_buffer_class, j_wrapped_native_i420_buffer_ctor_id,
>-      i420_buffer->width(), i420_buffer->height(), y_buffer,
>+  ScopedJavaLocalRef<jobject> y_buffer =
>+      NewDirectByteBuffer(jni, const_cast<uint8_t*>(i420_buffer->DataY()),
>+                          i420_buffer->StrideY() * i420_buffer->height());
>+  ScopedJavaLocalRef<jobject> u_buffer =
>+      NewDirectByteBuffer(jni, const_cast<uint8_t*>(i420_buffer->DataU()),
>+                          i420_buffer->StrideU() * i420_buffer->ChromaHeight());
>+  ScopedJavaLocalRef<jobject> v_buffer =
>+      NewDirectByteBuffer(jni, const_cast<uint8_t*>(i420_buffer->DataV()),
>+                          i420_buffer->StrideV() * i420_buffer->ChromaHeight());
>+
>+  return Java_WrappedNativeI420Buffer_Constructor(
>+      jni, i420_buffer->width(), i420_buffer->height(), y_buffer,
>       i420_buffer->StrideY(), u_buffer, i420_buffer->StrideU(), v_buffer,
>       i420_buffer->StrideV(), jlongFromPointer(i420_buffer.get()));
>-  CHECK_EXCEPTION(jni);
>-
>-  return j_wrapped_native_i420_buffer;
> }
> 
> }  // namespace jni
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/wrapped_native_i420_buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/wrapped_native_i420_buffer.h
>index 5de98d81cc2..70ad062cc6f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/wrapped_native_i420_buffer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/wrapped_native_i420_buffer.h
>@@ -14,13 +14,14 @@
> #include <jni.h>
> 
> #include "api/video/video_frame_buffer.h"
>+#include "sdk/android/native_api/jni/scoped_java_ref.h"
> 
> namespace webrtc {
> namespace jni {
> 
> // This function wraps the C++ I420 buffer and returns a Java
> // VideoFrame.I420Buffer as a jobject.
>-jobject WrapI420Buffer(
>+ScopedJavaLocalRef<jobject> WrapI420Buffer(
>     JNIEnv* jni,
>     const rtc::scoped_refptr<I420BufferInterface>& i420_buffer);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/wrappednativecodec.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/wrappednativecodec.cc
>deleted file mode 100644
>index 27a8977a683..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/wrappednativecodec.cc
>+++ /dev/null
>@@ -1,60 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "sdk/android/src/jni/wrappednativecodec.h"
>-
>-#include "sdk/android/generated_video_jni/jni/WrappedNativeVideoDecoder_jni.h"
>-#include "sdk/android/generated_video_jni/jni/WrappedNativeVideoEncoder_jni.h"
>-#include "sdk/android/src/jni/class_loader.h"
>-#include "sdk/android/src/jni/jni_helpers.h"
>-#include "sdk/android/src/jni/videodecoderwrapper.h"
>-#include "sdk/android/src/jni/videoencoderwrapper.h"
>-
>-namespace webrtc {
>-namespace jni {
>-
>-std::unique_ptr<VideoDecoder> JavaToNativeVideoDecoder(JNIEnv* jni,
>-                                                       jobject j_decoder) {
>-  jclass wrapped_native_decoder_class =
>-      GetClass(jni, "org/webrtc/WrappedNativeVideoDecoder");
>-
>-  VideoDecoder* decoder;
>-  if (jni->IsInstanceOf(j_decoder, wrapped_native_decoder_class)) {
>-    jlong native_decoder =
>-        Java_WrappedNativeVideoDecoder_createNativeDecoder(jni, j_decoder);
>-    decoder = reinterpret_cast<VideoDecoder*>(native_decoder);
>-  } else {
>-    decoder = new VideoDecoderWrapper(jni, j_decoder);
>-  }
>-
>-  return std::unique_ptr<VideoDecoder>(decoder);
>-}
>-
>-std::unique_ptr<VideoEncoder> JavaToNativeVideoEncoder(JNIEnv* jni,
>-                                                       jobject j_encoder) {
>-  VideoEncoder* encoder;
>-  if (Java_WrappedNativeVideoEncoder_isInstanceOf(jni, j_encoder)) {
>-    jlong native_encoder =
>-        Java_WrappedNativeVideoEncoder_createNativeEncoder(jni, j_encoder);
>-    encoder = reinterpret_cast<VideoEncoder*>(native_encoder);
>-  } else {
>-    encoder = new VideoEncoderWrapper(jni, j_encoder);
>-  }
>-
>-  return std::unique_ptr<VideoEncoder>(encoder);
>-}
>-
>-bool IsWrappedSoftwareEncoder(JNIEnv* jni, jobject j_encoder) {
>-  return Java_WrappedNativeVideoEncoder_isWrappedSoftwareEncoder(jni,
>-                                                                 j_encoder);
>-}
>-
>-}  // namespace jni
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/wrappednativecodec.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/wrappednativecodec.h
>deleted file mode 100644
>index 04942960289..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/wrappednativecodec.h
>+++ /dev/null
>@@ -1,40 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef SDK_ANDROID_SRC_JNI_WRAPPEDNATIVECODEC_H_
>-#define SDK_ANDROID_SRC_JNI_WRAPPEDNATIVECODEC_H_
>-
>-#include <jni.h>
>-#include <memory>
>-
>-#include "api/video_codecs/video_decoder.h"
>-#include "api/video_codecs/video_encoder.h"
>-
>-namespace webrtc {
>-namespace jni {
>-
>-/* If the j_decoder is a wrapped native decoder, unwrap it. If it is not,
>- * wrap it in a VideoDecoderWrapper.
>- */
>-std::unique_ptr<VideoDecoder> JavaToNativeVideoDecoder(JNIEnv* jni,
>-                                                       jobject j_decoder);
>-
>-/* If the j_encoder is a wrapped native encoder, unwrap it. If it is not,
>- * wrap it in a VideoEncoderWrapper.
>- */
>-std::unique_ptr<VideoEncoder> JavaToNativeVideoEncoder(JNIEnv* jni,
>-                                                       jobject j_encoder);
>-
>-bool IsWrappedSoftwareEncoder(JNIEnv* jni, jobject j_encoder);
>-
>-}  // namespace jni
>-}  // namespace webrtc
>-
>-#endif  // SDK_ANDROID_SRC_JNI_WRAPPEDNATIVECODEC_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/yuvhelper.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/yuvhelper.cc
>index bdf5375c4fd..1acd1428661 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/yuvhelper.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/jni/yuvhelper.cc
>@@ -10,75 +10,129 @@
> 
> #include <jni.h>
> 
>+#include "sdk/android/generated_video_jni/jni/YuvHelper_jni.h"
> #include "sdk/android/src/jni/jni_helpers.h"
> #include "third_party/libyuv/include/libyuv/convert.h"
>+#include "third_party/libyuv/include/libyuv/planar_functions.h"
> 
> namespace webrtc {
> namespace jni {
> 
>-JNI_FUNCTION_DECLARATION(void,
>-                         YuvHelper_I420Copy,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jobject j_src_y,
>-                         jint src_stride_y,
>-                         jobject j_src_u,
>-                         jint src_stride_u,
>-                         jobject j_src_v,
>-                         jint src_stride_v,
>-                         jobject j_dst_y,
>-                         jint dst_stride_y,
>-                         jobject j_dst_u,
>-                         jint dst_stride_u,
>-                         jobject j_dst_v,
>-                         jint dst_stride_v,
>-                         jint width,
>-                         jint height) {
>+void JNI_YuvHelper_CopyPlane(JNIEnv* jni,
>+                             const JavaParamRef<jclass>&,
>+                             const JavaParamRef<jobject>& j_src,
>+                             jint src_stride,
>+                             const JavaParamRef<jobject>& j_dst,
>+                             jint dst_stride,
>+                             jint width,
>+                             jint height) {
>+  const uint8_t* src =
>+      static_cast<const uint8_t*>(jni->GetDirectBufferAddress(j_src.obj()));
>+  uint8_t* dst =
>+      static_cast<uint8_t*>(jni->GetDirectBufferAddress(j_dst.obj()));
>+
>+  libyuv::CopyPlane(src, src_stride, dst, dst_stride, width, height);
>+}
>+
>+void JNI_YuvHelper_I420Copy(JNIEnv* jni,
>+                            const JavaParamRef<jclass>&,
>+                            const JavaParamRef<jobject>& j_src_y,
>+                            jint src_stride_y,
>+                            const JavaParamRef<jobject>& j_src_u,
>+                            jint src_stride_u,
>+                            const JavaParamRef<jobject>& j_src_v,
>+                            jint src_stride_v,
>+                            const JavaParamRef<jobject>& j_dst_y,
>+                            jint dst_stride_y,
>+                            const JavaParamRef<jobject>& j_dst_u,
>+                            jint dst_stride_u,
>+                            const JavaParamRef<jobject>& j_dst_v,
>+                            jint dst_stride_v,
>+                            jint width,
>+                            jint height) {
>   const uint8_t* src_y =
>-      static_cast<const uint8_t*>(jni->GetDirectBufferAddress(j_src_y));
>+      static_cast<const uint8_t*>(jni->GetDirectBufferAddress(j_src_y.obj()));
>   const uint8_t* src_u =
>-      static_cast<const uint8_t*>(jni->GetDirectBufferAddress(j_src_u));
>+      static_cast<const uint8_t*>(jni->GetDirectBufferAddress(j_src_u.obj()));
>   const uint8_t* src_v =
>-      static_cast<const uint8_t*>(jni->GetDirectBufferAddress(j_src_v));
>-  uint8_t* dst_y = static_cast<uint8_t*>(jni->GetDirectBufferAddress(j_dst_y));
>-  uint8_t* dst_u = static_cast<uint8_t*>(jni->GetDirectBufferAddress(j_dst_u));
>-  uint8_t* dst_v = static_cast<uint8_t*>(jni->GetDirectBufferAddress(j_dst_v));
>+      static_cast<const uint8_t*>(jni->GetDirectBufferAddress(j_src_v.obj()));
>+  uint8_t* dst_y =
>+      static_cast<uint8_t*>(jni->GetDirectBufferAddress(j_dst_y.obj()));
>+  uint8_t* dst_u =
>+      static_cast<uint8_t*>(jni->GetDirectBufferAddress(j_dst_u.obj()));
>+  uint8_t* dst_v =
>+      static_cast<uint8_t*>(jni->GetDirectBufferAddress(j_dst_v.obj()));
> 
>   libyuv::I420Copy(src_y, src_stride_y, src_u, src_stride_u, src_v,
>                    src_stride_v, dst_y, dst_stride_y, dst_u, dst_stride_u,
>                    dst_v, dst_stride_v, width, height);
> }
> 
>-JNI_FUNCTION_DECLARATION(void,
>-                         YuvHelper_I420ToNV12,
>-                         JNIEnv* jni,
>-                         jclass,
>-                         jobject j_src_y,
>-                         jint src_stride_y,
>-                         jobject j_src_u,
>-                         jint src_stride_u,
>-                         jobject j_src_v,
>-                         jint src_stride_v,
>-                         jobject j_dst_y,
>-                         jint dst_stride_y,
>-                         jobject j_dst_uv,
>-                         jint dst_stride_uv,
>-                         jint width,
>-                         jint height) {
>+static void JNI_YuvHelper_I420ToNV12(JNIEnv* jni,
>+                                     const JavaParamRef<jclass>&,
>+                                     const JavaParamRef<jobject>& j_src_y,
>+                                     jint src_stride_y,
>+                                     const JavaParamRef<jobject>& j_src_u,
>+                                     jint src_stride_u,
>+                                     const JavaParamRef<jobject>& j_src_v,
>+                                     jint src_stride_v,
>+                                     const JavaParamRef<jobject>& j_dst_y,
>+                                     jint dst_stride_y,
>+                                     const JavaParamRef<jobject>& j_dst_uv,
>+                                     jint dst_stride_uv,
>+                                     jint width,
>+                                     jint height) {
>   const uint8_t* src_y =
>-      static_cast<const uint8_t*>(jni->GetDirectBufferAddress(j_src_y));
>+      static_cast<const uint8_t*>(jni->GetDirectBufferAddress(j_src_y.obj()));
>   const uint8_t* src_u =
>-      static_cast<const uint8_t*>(jni->GetDirectBufferAddress(j_src_u));
>+      static_cast<const uint8_t*>(jni->GetDirectBufferAddress(j_src_u.obj()));
>   const uint8_t* src_v =
>-      static_cast<const uint8_t*>(jni->GetDirectBufferAddress(j_src_v));
>-  uint8_t* dst_y = static_cast<uint8_t*>(jni->GetDirectBufferAddress(j_dst_y));
>+      static_cast<const uint8_t*>(jni->GetDirectBufferAddress(j_src_v.obj()));
>+  uint8_t* dst_y =
>+      static_cast<uint8_t*>(jni->GetDirectBufferAddress(j_dst_y.obj()));
>   uint8_t* dst_uv =
>-      static_cast<uint8_t*>(jni->GetDirectBufferAddress(j_dst_uv));
>+      static_cast<uint8_t*>(jni->GetDirectBufferAddress(j_dst_uv.obj()));
> 
>   libyuv::I420ToNV12(src_y, src_stride_y, src_u, src_stride_u, src_v,
>                      src_stride_v, dst_y, dst_stride_y, dst_uv, dst_stride_uv,
>                      width, height);
> }
> 
>+void JNI_YuvHelper_I420Rotate(JNIEnv* jni,
>+                              const JavaParamRef<jclass>&,
>+                              const JavaParamRef<jobject>& j_src_y,
>+                              jint src_stride_y,
>+                              const JavaParamRef<jobject>& j_src_u,
>+                              jint src_stride_u,
>+                              const JavaParamRef<jobject>& j_src_v,
>+                              jint src_stride_v,
>+                              const JavaParamRef<jobject>& j_dst_y,
>+                              jint dst_stride_y,
>+                              const JavaParamRef<jobject>& j_dst_u,
>+                              jint dst_stride_u,
>+                              const JavaParamRef<jobject>& j_dst_v,
>+                              jint dst_stride_v,
>+                              jint src_width,
>+                              jint src_height,
>+                              jint rotation_mode) {
>+  const uint8_t* src_y =
>+      static_cast<const uint8_t*>(jni->GetDirectBufferAddress(j_src_y.obj()));
>+  const uint8_t* src_u =
>+      static_cast<const uint8_t*>(jni->GetDirectBufferAddress(j_src_u.obj()));
>+  const uint8_t* src_v =
>+      static_cast<const uint8_t*>(jni->GetDirectBufferAddress(j_src_v.obj()));
>+  uint8_t* dst_y =
>+      static_cast<uint8_t*>(jni->GetDirectBufferAddress(j_dst_y.obj()));
>+  uint8_t* dst_u =
>+      static_cast<uint8_t*>(jni->GetDirectBufferAddress(j_dst_u.obj()));
>+  uint8_t* dst_v =
>+      static_cast<uint8_t*>(jni->GetDirectBufferAddress(j_dst_v.obj()));
>+
>+  libyuv::I420Rotate(src_y, src_stride_y, src_u, src_stride_u, src_v,
>+                     src_stride_v, dst_y, dst_stride_y, dst_u, dst_stride_u,
>+                     dst_v, dst_stride_v, src_width, src_height,
>+                     static_cast<libyuv::RotationMode>(rotation_mode));
>+}
>+
> }  // namespace jni
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/tests/src/org/webrtc/CodecTestHelper.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/tests/src/org/webrtc/CodecTestHelper.java
>new file mode 100644
>index 00000000000..8067408c5ce
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/tests/src/org/webrtc/CodecTestHelper.java
>@@ -0,0 +1,64 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc;
>+
>+import static com.google.common.truth.Truth.assertThat;
>+import static com.google.common.truth.Truth.assertWithMessage;
>+
>+import java.nio.ByteBuffer;
>+import java.util.Random;
>+import org.webrtc.JavaI420Buffer;
>+import org.webrtc.VideoFrame;
>+
>+/**
>+ * Helper methods for {@link HardwareVideoEncoderTest} and {@link HardwareVideoDecoderTest}.
>+ */
>+class CodecTestHelper {
>+  static void assertEqualContents(byte[] expected, ByteBuffer actual, int offset, int size) {
>+    assertThat(size).isEqualTo(expected.length);
>+    assertThat(actual.capacity()).isAtLeast(offset + size);
>+    for (int i = 0; i < expected.length; i++) {
>+      assertWithMessage("At index: " + i).that(actual.get(offset + i)).isEqualTo(expected[i]);
>+    }
>+  }
>+
>+  static byte[] generateRandomData(int length) {
>+    Random random = new Random();
>+    byte[] data = new byte[length];
>+    random.nextBytes(data);
>+    return data;
>+  }
>+
>+  static VideoFrame.I420Buffer wrapI420(int width, int height, byte[] data) {
>+    final int posY = 0;
>+    final int posU = width * height;
>+    final int posV = posU + width * height / 4;
>+    final int endV = posV + width * height / 4;
>+
>+    ByteBuffer buffer = ByteBuffer.allocateDirect(data.length);
>+    buffer.put(data);
>+
>+    buffer.limit(posU);
>+    buffer.position(posY);
>+    ByteBuffer dataY = buffer.slice();
>+
>+    buffer.limit(posV);
>+    buffer.position(posU);
>+    ByteBuffer dataU = buffer.slice();
>+
>+    buffer.limit(endV);
>+    buffer.position(posV);
>+    ByteBuffer dataV = buffer.slice();
>+
>+    return JavaI420Buffer.wrap(width, height, dataY, width, dataU, width / 2, dataV, width / 2,
>+        /* releaseCallback= */ null);
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/tests/src/org/webrtc/FakeMediaCodecWrapper.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/tests/src/org/webrtc/FakeMediaCodecWrapper.java
>new file mode 100644
>index 00000000000..c07cfc62426
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/tests/src/org/webrtc/FakeMediaCodecWrapper.java
>@@ -0,0 +1,314 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc;
>+
>+import android.graphics.SurfaceTexture;
>+import android.media.MediaCodec;
>+import android.media.MediaCodecInfo.CodecCapabilities;
>+import android.media.MediaCrypto;
>+import android.media.MediaFormat;
>+import android.os.Bundle;
>+import android.view.Surface;
>+import java.nio.ByteBuffer;
>+import java.util.ArrayList;
>+import java.util.List;
>+import javax.annotation.Nullable;
>+
>+/**
>+ * Fake MediaCodec that implements the basic state machine.
>+ *
>+ * @note This class is only intended for single-threaded tests and is not thread-safe.
>+ */
>+public class FakeMediaCodecWrapper implements MediaCodecWrapper {
>+  private static final int NUM_INPUT_BUFFERS = 10;
>+  private static final int NUM_OUTPUT_BUFFERS = 10;
>+  private static final int MAX_ENCODED_DATA_SIZE_BYTES = 1_000;
>+
>+  /**
>+   * MediaCodec state as defined by:
>+   * https://developer.android.com/reference/android/media/MediaCodec.html
>+   */
>+  public enum State {
>+    STOPPED_CONFIGURED(Primary.STOPPED),
>+    STOPPED_UNINITIALIZED(Primary.STOPPED),
>+    STOPPED_ERROR(Primary.STOPPED),
>+    EXECUTING_FLUSHED(Primary.EXECUTING),
>+    EXECUTING_RUNNING(Primary.EXECUTING),
>+    EXECUTING_END_OF_STREAM(Primary.EXECUTING),
>+    RELEASED(Primary.RELEASED);
>+
>+    public enum Primary { STOPPED, EXECUTING, RELEASED }
>+
>+    private final Primary primary;
>+
>+    State(Primary primary) {
>+      this.primary = primary;
>+    }
>+
>+    public Primary getPrimary() {
>+      return primary;
>+    }
>+  }
>+
>+  /** Represents an output buffer that will be returned by dequeueOutputBuffer. */
>+  public static class QueuedOutputBufferInfo {
>+    private int index;
>+    private int offset;
>+    private int size;
>+    private long presentationTimeUs;
>+    private int flags;
>+
>+    private QueuedOutputBufferInfo(
>+        int index, int offset, int size, long presentationTimeUs, int flags) {
>+      this.index = index;
>+      this.offset = offset;
>+      this.size = size;
>+      this.presentationTimeUs = presentationTimeUs;
>+      this.flags = flags;
>+    }
>+
>+    public static QueuedOutputBufferInfo create(
>+        int index, int offset, int size, long presentationTimeUs, int flags) {
>+      return new QueuedOutputBufferInfo(index, offset, size, presentationTimeUs, flags);
>+    }
>+
>+    public int getIndex() {
>+      return index;
>+    }
>+
>+    public int getOffset() {
>+      return offset;
>+    }
>+
>+    public int getSize() {
>+      return size;
>+    }
>+
>+    public long getPresentationTimeUs() {
>+      return presentationTimeUs;
>+    }
>+
>+    public int getFlags() {
>+      return flags;
>+    }
>+  }
>+
>+  private State state = State.STOPPED_UNINITIALIZED;
>+  private @Nullable MediaFormat configuredFormat;
>+  private int configuredFlags;
>+  private final MediaFormat outputFormat;
>+  private final ByteBuffer[] inputBuffers = new ByteBuffer[NUM_INPUT_BUFFERS];
>+  private final ByteBuffer[] outputBuffers = new ByteBuffer[NUM_OUTPUT_BUFFERS];
>+  private final boolean[] inputBufferReserved = new boolean[NUM_INPUT_BUFFERS];
>+  private final boolean[] outputBufferReserved = new boolean[NUM_OUTPUT_BUFFERS];
>+  private final List<QueuedOutputBufferInfo> queuedOutputBuffers = new ArrayList<>();
>+
>+  public FakeMediaCodecWrapper(MediaFormat outputFormat) {
>+    this.outputFormat = outputFormat;
>+  }
>+
>+  /** Returns the current simulated state of MediaCodec. */
>+  public State getState() {
>+    return state;
>+  }
>+
>+  /** Gets the last configured media format passed to configure. */
>+  public @Nullable MediaFormat getConfiguredFormat() {
>+    return configuredFormat;
>+  }
>+
>+  /** Returns the last flags passed to configure. */
>+  public int getConfiguredFlags() {
>+    return configuredFlags;
>+  }
>+
>+  /**
>+   * Adds a texture buffer that will be returned by dequeueOutputBuffer. Returns index of the
>+   * buffer.
>+   */
>+  public int addOutputTexture(long presentationTimestampUs, int flags) {
>+    int index = getFreeOutputBuffer();
>+    queuedOutputBuffers.add(QueuedOutputBufferInfo.create(
>+        index, /* offset= */ 0, /* size= */ 0, presentationTimestampUs, flags));
>+    return index;
>+  }
>+
>+  /**
>+   * Adds a byte buffer buffer that will be returned by dequeueOutputBuffer. Returns index of the
>+   * buffer.
>+   */
>+  public int addOutputData(byte[] data, long presentationTimestampUs, int flags) {
>+    int index = getFreeOutputBuffer();
>+    ByteBuffer outputBuffer = outputBuffers[index];
>+
>+    outputBuffer.clear();
>+    outputBuffer.put(data);
>+    outputBuffer.rewind();
>+
>+    queuedOutputBuffers.add(QueuedOutputBufferInfo.create(
>+        index, /* offset= */ 0, data.length, presentationTimestampUs, flags));
>+    return index;
>+  }
>+
>+  /**
>+   * Returns the first output buffer that is not reserved and reserves it. It will be stay reserved
>+   * until released with releaseOutputBuffer.
>+   */
>+  private int getFreeOutputBuffer() {
>+    for (int i = 0; i < NUM_OUTPUT_BUFFERS; i++) {
>+      if (!outputBufferReserved[i]) {
>+        outputBufferReserved[i] = true;
>+        return i;
>+      }
>+    }
>+    throw new RuntimeException("All output buffers reserved!");
>+  }
>+
>+  @Override
>+  public void configure(MediaFormat format, Surface surface, MediaCrypto crypto, int flags) {
>+    if (state != State.STOPPED_UNINITIALIZED) {
>+      throw new IllegalStateException("Expected state STOPPED_UNINITIALIZED but was " + state);
>+    }
>+    state = State.STOPPED_CONFIGURED;
>+    configuredFormat = format;
>+    configuredFlags = flags;
>+
>+    final int width = configuredFormat.getInteger(MediaFormat.KEY_WIDTH);
>+    final int height = configuredFormat.getInteger(MediaFormat.KEY_HEIGHT);
>+    final int yuvSize = width * height * 3 / 2;
>+    final int inputBufferSize;
>+    final int outputBufferSize;
>+
>+    if ((flags & MediaCodec.CONFIGURE_FLAG_ENCODE) != 0) {
>+      final int colorFormat = configuredFormat.getInteger(MediaFormat.KEY_COLOR_FORMAT);
>+
>+      inputBufferSize = colorFormat == CodecCapabilities.COLOR_FormatSurface ? 0 : yuvSize;
>+      outputBufferSize = MAX_ENCODED_DATA_SIZE_BYTES;
>+    } else {
>+      inputBufferSize = MAX_ENCODED_DATA_SIZE_BYTES;
>+      outputBufferSize = surface != null ? 0 : yuvSize;
>+    }
>+
>+    for (int i = 0; i < inputBuffers.length; i++) {
>+      inputBuffers[i] = ByteBuffer.allocateDirect(inputBufferSize);
>+    }
>+    for (int i = 0; i < outputBuffers.length; i++) {
>+      outputBuffers[i] = ByteBuffer.allocateDirect(outputBufferSize);
>+    }
>+  }
>+
>+  @Override
>+  public void start() {
>+    if (state != State.STOPPED_CONFIGURED) {
>+      throw new IllegalStateException("Expected state STOPPED_CONFIGURED but was " + state);
>+    }
>+    state = State.EXECUTING_RUNNING;
>+  }
>+
>+  @Override
>+  public void flush() {
>+    if (state.getPrimary() != State.Primary.EXECUTING) {
>+      throw new IllegalStateException("Expected state EXECUTING but was " + state);
>+    }
>+    state = State.EXECUTING_FLUSHED;
>+  }
>+
>+  @Override
>+  public void stop() {
>+    if (state.getPrimary() != State.Primary.EXECUTING) {
>+      throw new IllegalStateException("Expected state EXECUTING but was " + state);
>+    }
>+    state = State.STOPPED_UNINITIALIZED;
>+  }
>+
>+  @Override
>+  public void release() {
>+    state = State.RELEASED;
>+  }
>+
>+  @Override
>+  public int dequeueInputBuffer(long timeoutUs) {
>+    if (state != State.EXECUTING_FLUSHED && state != State.EXECUTING_RUNNING) {
>+      throw new IllegalStateException(
>+          "Expected state EXECUTING_FLUSHED or EXECUTING_RUNNING but was " + state);
>+    }
>+    state = State.EXECUTING_RUNNING;
>+
>+    for (int i = 0; i < NUM_INPUT_BUFFERS; i++) {
>+      if (!inputBufferReserved[i]) {
>+        inputBufferReserved[i] = true;
>+        return i;
>+      }
>+    }
>+    return MediaCodec.INFO_TRY_AGAIN_LATER;
>+  }
>+
>+  @Override
>+  public void queueInputBuffer(
>+      int index, int offset, int size, long presentationTimeUs, int flags) {
>+    if (state.getPrimary() != State.Primary.EXECUTING) {
>+      throw new IllegalStateException("Expected state EXECUTING but was " + state);
>+    }
>+    if (flags != 0) {
>+      throw new UnsupportedOperationException(
>+          "Flags are not implemented in FakeMediaCodecWrapper.");
>+    }
>+  }
>+
>+  @Override
>+  public int dequeueOutputBuffer(MediaCodec.BufferInfo info, long timeoutUs) {
>+    if (state.getPrimary() != State.Primary.EXECUTING) {
>+      throw new IllegalStateException("Expected state EXECUTING but was " + state);
>+    }
>+
>+    if (queuedOutputBuffers.isEmpty()) {
>+      return MediaCodec.INFO_TRY_AGAIN_LATER;
>+    }
>+    QueuedOutputBufferInfo outputBufferInfo = queuedOutputBuffers.remove(/* index= */ 0);
>+    info.set(outputBufferInfo.getOffset(), outputBufferInfo.getSize(),
>+        outputBufferInfo.getPresentationTimeUs(), outputBufferInfo.getFlags());
>+    return outputBufferInfo.getIndex();
>+  }
>+
>+  @Override
>+  public void releaseOutputBuffer(int index, boolean render) {
>+    if (state.getPrimary() != State.Primary.EXECUTING) {
>+      throw new IllegalStateException("Expected state EXECUTING but was " + state);
>+    }
>+    if (!outputBufferReserved[index]) {
>+      throw new RuntimeException("Released output buffer was not in use.");
>+    }
>+    outputBufferReserved[index] = false;
>+  }
>+
>+  @Override
>+  public ByteBuffer[] getInputBuffers() {
>+    return inputBuffers;
>+  }
>+
>+  @Override
>+  public ByteBuffer[] getOutputBuffers() {
>+    return outputBuffers;
>+  }
>+
>+  @Override
>+  public MediaFormat getOutputFormat() {
>+    return outputFormat;
>+  }
>+
>+  @Override
>+  public Surface createInputSurface() {
>+    return new Surface(new SurfaceTexture(/* texName= */ 0));
>+  }
>+
>+  @Override
>+  public void setParameters(Bundle params) {}
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/tests/src/org/webrtc/GlGenericDrawerTest.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/tests/src/org/webrtc/GlGenericDrawerTest.java
>new file mode 100644
>index 00000000000..06b53672a0a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/tests/src/org/webrtc/GlGenericDrawerTest.java
>@@ -0,0 +1,158 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc;
>+
>+import static org.junit.Assert.assertEquals;
>+import static org.mockito.Mockito.mock;
>+import static org.mockito.Mockito.times;
>+import static org.mockito.Mockito.verify;
>+import static org.mockito.Mockito.verifyNoMoreInteractions;
>+
>+import org.chromium.testing.local.LocalRobolectricTestRunner;
>+import org.junit.After;
>+import org.junit.Before;
>+import org.junit.Test;
>+import org.junit.runner.RunWith;
>+import org.robolectric.annotation.Config;
>+import org.webrtc.GlShader;
>+
>+@RunWith(LocalRobolectricTestRunner.class)
>+@Config(manifest = Config.NONE)
>+public class GlGenericDrawerTest {
>+  // Simplest possible valid generic fragment shader.
>+  private static final String FRAGMENT_SHADER = "void main() {\n"
>+      + "  gl_FragColor = sample(tc);\n"
>+      + "}\n";
>+  private static final int TEXTURE_ID = 3;
>+  private static final float[] TEX_MATRIX =
>+      new float[] {1, 2, 3, 4, -1, -2, -3, -4, 0, 0, 1, 0, 0, 0, 0, 1};
>+  private static final int FRAME_WIDTH = 640;
>+  private static final int FRAME_HEIGHT = 480;
>+  private static final int VIEWPORT_X = 3;
>+  private static final int VIEWPORT_Y = 5;
>+  private static final int VIEWPORT_WIDTH = 500;
>+  private static final int VIEWPORT_HEIGHT = 500;
>+
>+  // Replace OpenGLES GlShader dependency with a mock.
>+  private class GlGenericDrawerForTest extends GlGenericDrawer {
>+    public GlGenericDrawerForTest(String genericFragmentSource, ShaderCallbacks shaderCallbacks) {
>+      super(genericFragmentSource, shaderCallbacks);
>+    }
>+
>+    @Override
>+    GlShader createShader(ShaderType shaderType) {
>+      return mockedShader;
>+    }
>+  }
>+
>+  private GlShader mockedShader;
>+  private GlGenericDrawer glGenericDrawer;
>+  private GlGenericDrawer.ShaderCallbacks mockedCallbacks;
>+
>+  @Before
>+  public void setUp() {
>+    mockedShader = mock(GlShader.class);
>+    mockedCallbacks = mock(GlGenericDrawer.ShaderCallbacks.class);
>+    glGenericDrawer = new GlGenericDrawerForTest(FRAGMENT_SHADER, mockedCallbacks);
>+  }
>+
>+  @After
>+  public void tearDown() {
>+    verifyNoMoreInteractions(mockedCallbacks);
>+  }
>+
>+  @Test
>+  public void testOesFragmentShader() {
>+    final String expectedOesFragmentShader = "#extension GL_OES_EGL_image_external : require\n"
>+        + "precision mediump float;\n"
>+        + "varying vec2 tc;\n"
>+        + "uniform samplerExternalOES tex;\n"
>+        + "void main() {\n"
>+        + "  gl_FragColor = texture2D(tex, tc);\n"
>+        + "}\n";
>+    final String oesFragmentShader =
>+        GlGenericDrawer.createFragmentShaderString(FRAGMENT_SHADER, GlGenericDrawer.ShaderType.OES);
>+    assertEquals(expectedOesFragmentShader, oesFragmentShader);
>+  }
>+
>+  @Test
>+  public void testRgbFragmentShader() {
>+    final String expectedRgbFragmentShader = "precision mediump float;\n"
>+        + "varying vec2 tc;\n"
>+        + "uniform sampler2D tex;\n"
>+        + "void main() {\n"
>+        + "  gl_FragColor = texture2D(tex, tc);\n"
>+        + "}\n";
>+    final String rgbFragmentShader =
>+        GlGenericDrawer.createFragmentShaderString(FRAGMENT_SHADER, GlGenericDrawer.ShaderType.RGB);
>+    assertEquals(expectedRgbFragmentShader, rgbFragmentShader);
>+  }
>+
>+  @Test
>+  public void testYuvFragmentShader() {
>+    final String expectedYuvFragmentShader = "precision mediump float;\n"
>+        + "varying vec2 tc;\n"
>+        + "uniform sampler2D y_tex;\n"
>+        + "uniform sampler2D u_tex;\n"
>+        + "uniform sampler2D v_tex;\n"
>+        + "vec4 sample(vec2 p) {\n"
>+        + "  float y = texture2D(y_tex, p).r;\n"
>+        + "  float u = texture2D(u_tex, p).r - 0.5;\n"
>+        + "  float v = texture2D(v_tex, p).r - 0.5;\n"
>+        + "  return vec4(y + 1.403 * v, y - 0.344 * u - 0.714 * v, y + 1.77 * u, 1);\n"
>+        + "}\n"
>+        + "void main() {\n"
>+        + "  gl_FragColor = sample(tc);\n"
>+        + "}\n";
>+    final String yuvFragmentShader =
>+        GlGenericDrawer.createFragmentShaderString(FRAGMENT_SHADER, GlGenericDrawer.ShaderType.YUV);
>+    assertEquals(expectedYuvFragmentShader, yuvFragmentShader);
>+  }
>+
>+  @Test
>+  public void testShaderCallbacksOneRgbFrame() {
>+    glGenericDrawer.drawRgb(TEXTURE_ID, TEX_MATRIX, FRAME_WIDTH, FRAME_HEIGHT, VIEWPORT_X,
>+        VIEWPORT_Y, VIEWPORT_WIDTH, VIEWPORT_HEIGHT);
>+
>+    verify(mockedCallbacks).onNewShader(mockedShader);
>+    verify(mockedCallbacks)
>+        .onPrepareShader(
>+            mockedShader, TEX_MATRIX, FRAME_WIDTH, FRAME_HEIGHT, VIEWPORT_WIDTH, VIEWPORT_HEIGHT);
>+  }
>+
>+  @Test
>+  public void testShaderCallbacksTwoRgbFrames() {
>+    glGenericDrawer.drawRgb(TEXTURE_ID, TEX_MATRIX, FRAME_WIDTH, FRAME_HEIGHT, VIEWPORT_X,
>+        VIEWPORT_Y, VIEWPORT_WIDTH, VIEWPORT_HEIGHT);
>+    glGenericDrawer.drawRgb(TEXTURE_ID, TEX_MATRIX, FRAME_WIDTH, FRAME_HEIGHT, VIEWPORT_X,
>+        VIEWPORT_Y, VIEWPORT_WIDTH, VIEWPORT_HEIGHT);
>+
>+    // Expect only one shader to be created, but two frames to be drawn.
>+    verify(mockedCallbacks, times(1)).onNewShader(mockedShader);
>+    verify(mockedCallbacks, times(2))
>+        .onPrepareShader(
>+            mockedShader, TEX_MATRIX, FRAME_WIDTH, FRAME_HEIGHT, VIEWPORT_WIDTH, VIEWPORT_HEIGHT);
>+  }
>+
>+  @Test
>+  public void testShaderCallbacksChangingShaderType() {
>+    glGenericDrawer.drawRgb(TEXTURE_ID, TEX_MATRIX, FRAME_WIDTH, FRAME_HEIGHT, VIEWPORT_X,
>+        VIEWPORT_Y, VIEWPORT_WIDTH, VIEWPORT_HEIGHT);
>+    glGenericDrawer.drawOes(TEXTURE_ID, TEX_MATRIX, FRAME_WIDTH, FRAME_HEIGHT, VIEWPORT_X,
>+        VIEWPORT_Y, VIEWPORT_WIDTH, VIEWPORT_HEIGHT);
>+
>+    // Expect two shaders to be created, and two frames to be drawn.
>+    verify(mockedCallbacks, times(2)).onNewShader(mockedShader);
>+    verify(mockedCallbacks, times(2))
>+        .onPrepareShader(
>+            mockedShader, TEX_MATRIX, FRAME_WIDTH, FRAME_HEIGHT, VIEWPORT_WIDTH, VIEWPORT_HEIGHT);
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/tests/src/org/webrtc/HardwareVideoDecoderTest.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/tests/src/org/webrtc/HardwareVideoDecoderTest.java
>new file mode 100644
>index 00000000000..f47d06902aa
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/tests/src/org/webrtc/HardwareVideoDecoderTest.java
>@@ -0,0 +1,422 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc;
>+
>+import static com.google.common.truth.Truth.assertThat;
>+import static org.mockito.Matchers.any;
>+import static org.mockito.Matchers.anyInt;
>+import static org.mockito.Matchers.anyLong;
>+import static org.mockito.Matchers.eq;
>+import static org.mockito.Mockito.doThrow;
>+import static org.mockito.Mockito.inOrder;
>+import static org.mockito.Mockito.mock;
>+import static org.mockito.Mockito.spy;
>+import static org.mockito.Mockito.verify;
>+import static org.robolectric.Shadows.shadowOf;
>+
>+import android.graphics.Matrix;
>+import android.media.MediaCodec.BufferInfo;
>+import android.media.MediaCodecInfo.CodecCapabilities;
>+import android.media.MediaFormat;
>+import android.os.Handler;
>+import java.nio.ByteBuffer;
>+import java.util.concurrent.Callable;
>+import java.util.concurrent.TimeUnit;
>+import org.chromium.testing.local.LocalRobolectricTestRunner;
>+import org.junit.Before;
>+import org.junit.Ignore;
>+import org.junit.Test;
>+import org.junit.runner.RunWith;
>+import org.mockito.ArgumentCaptor;
>+import org.mockito.InOrder;
>+import org.mockito.Mock;
>+import org.mockito.MockitoAnnotations;
>+import org.robolectric.annotation.Config;
>+import org.robolectric.shadows.ShadowSystemClock;
>+import org.webrtc.EglBase;
>+import org.webrtc.EncodedImage;
>+import org.webrtc.EncodedImage.FrameType;
>+import org.webrtc.FakeMediaCodecWrapper.State;
>+import org.webrtc.SurfaceTextureHelper;
>+import org.webrtc.TextureBufferImpl;
>+import org.webrtc.VideoCodecStatus;
>+import org.webrtc.VideoDecoder;
>+import org.webrtc.VideoDecoder.DecodeInfo;
>+import org.webrtc.VideoFrame;
>+import org.webrtc.VideoFrame.I420Buffer;
>+import org.webrtc.VideoFrame.TextureBuffer.Type;
>+import org.webrtc.VideoSink;
>+import org.webrtc.YuvConverter;
>+
>+@RunWith(LocalRobolectricTestRunner.class)
>+@Config(manifest = Config.NONE)
>+public class HardwareVideoDecoderTest {
>+  private static final VideoDecoder.Settings TEST_DECODER_SETTINGS =
>+      new VideoDecoder.Settings(/* numberOfCores= */ 1, /* width= */ 640, /* height= */ 480);
>+  private static final int COLOR_FORMAT = CodecCapabilities.COLOR_FormatYUV420Planar;
>+  private static final long POLL_DELAY_MS = 10;
>+  private static final long DELIVER_DECODED_IMAGE_DELAY_MS = 10;
>+
>+  private static final byte[] ENCODED_TEST_DATA = new byte[] {1, 2, 3, 4, 5, 6, 7, 8, 9, 10};
>+
>+  private class TestDecoder extends HardwareVideoDecoder {
>+    private final Object deliverDecodedFrameLock = new Object();
>+    private boolean deliverDecodedFrameDone = true;
>+
>+    public TestDecoder(MediaCodecWrapperFactory mediaCodecFactory, String codecName,
>+        VideoCodecType codecType, int colorFormat, EglBase.Context sharedContext) {
>+      super(mediaCodecFactory, codecName, codecType, colorFormat, sharedContext);
>+    }
>+
>+    public void waitDeliverDecodedFrame() throws InterruptedException {
>+      synchronized (deliverDecodedFrameLock) {
>+        deliverDecodedFrameDone = false;
>+        deliverDecodedFrameLock.notifyAll();
>+        while (!deliverDecodedFrameDone) {
>+          deliverDecodedFrameLock.wait();
>+        }
>+      }
>+    }
>+
>+    @SuppressWarnings("WaitNotInLoop") // This method is called inside a loop.
>+    @Override
>+    protected void deliverDecodedFrame() {
>+      synchronized (deliverDecodedFrameLock) {
>+        if (deliverDecodedFrameDone) {
>+          try {
>+            deliverDecodedFrameLock.wait(DELIVER_DECODED_IMAGE_DELAY_MS);
>+          } catch (InterruptedException e) {
>+            Thread.currentThread().interrupt();
>+            return;
>+          }
>+        }
>+        if (deliverDecodedFrameDone) {
>+          return;
>+        }
>+        super.deliverDecodedFrame();
>+        deliverDecodedFrameDone = true;
>+        deliverDecodedFrameLock.notifyAll();
>+      }
>+    }
>+
>+    @Override
>+    protected SurfaceTextureHelper createSurfaceTextureHelper() {
>+      return mockSurfaceTextureHelper;
>+    }
>+
>+    @Override
>+    protected void releaseSurface() {}
>+
>+    @Override
>+    protected VideoFrame.I420Buffer allocateI420Buffer(int width, int height) {
>+      int chromaHeight = (height + 1) / 2;
>+      int strideUV = (width + 1) / 2;
>+      int yPos = 0;
>+      int uPos = yPos + width * height;
>+      int vPos = uPos + strideUV * chromaHeight;
>+
>+      ByteBuffer buffer = ByteBuffer.allocateDirect(width * height + 2 * strideUV * chromaHeight);
>+
>+      buffer.position(yPos);
>+      buffer.limit(uPos);
>+      ByteBuffer dataY = buffer.slice();
>+
>+      buffer.position(uPos);
>+      buffer.limit(vPos);
>+      ByteBuffer dataU = buffer.slice();
>+
>+      buffer.position(vPos);
>+      buffer.limit(vPos + strideUV * chromaHeight);
>+      ByteBuffer dataV = buffer.slice();
>+
>+      return JavaI420Buffer.wrap(width, height, dataY, width, dataU, strideUV, dataV, strideUV,
>+          /* releaseCallback= */ null);
>+    }
>+
>+    @Override
>+    protected void copyPlane(
>+        ByteBuffer src, int srcStride, ByteBuffer dst, int dstStride, int width, int height) {
>+      for (int y = 0; y < height; y++) {
>+        for (int x = 0; x < width; x++) {
>+          dst.put(y * dstStride + x, src.get(y * srcStride + x));
>+        }
>+      }
>+    }
>+  }
>+
>+  private class TestDecoderBuilder {
>+    private VideoCodecType codecType = VideoCodecType.VP8;
>+    private boolean useSurface = true;
>+
>+    public TestDecoderBuilder setCodecType(VideoCodecType codecType) {
>+      this.codecType = codecType;
>+      return this;
>+    }
>+
>+    public TestDecoderBuilder setUseSurface(boolean useSurface) {
>+      this.useSurface = useSurface;
>+      return this;
>+    }
>+
>+    public TestDecoder build() {
>+      return new TestDecoder((String name)
>+                                 -> fakeMediaCodecWrapper,
>+          /* codecName= */ "org.webrtc.testdecoder", codecType, COLOR_FORMAT,
>+          useSurface ? mockEglBaseContext : null);
>+    }
>+  }
>+
>+  private EncodedImage createTestEncodedImage() {
>+    return EncodedImage.builder()
>+        .setBuffer(ByteBuffer.wrap(ENCODED_TEST_DATA))
>+        .setFrameType(FrameType.VideoFrameKey)
>+        .setCompleteFrame(true)
>+        .createEncodedImage();
>+  }
>+
>+  @Mock private EglBase.Context mockEglBaseContext;
>+  @Mock private SurfaceTextureHelper mockSurfaceTextureHelper;
>+  @Mock private VideoDecoder.Callback mockDecoderCallback;
>+  private FakeMediaCodecWrapper fakeMediaCodecWrapper;
>+
>+  @Before
>+  public void setUp() {
>+    MockitoAnnotations.initMocks(this);
>+    MediaFormat outputFormat = new MediaFormat();
>+    // TODO(sakal): Add more details to output format as needed.
>+    fakeMediaCodecWrapper = spy(new FakeMediaCodecWrapper(outputFormat));
>+  }
>+
>+  @Test
>+  public void testInit() {
>+    // Set-up.
>+    HardwareVideoDecoder decoder =
>+        new TestDecoderBuilder().setCodecType(VideoCodecType.VP8).build();
>+
>+    // Test.
>+    assertThat(decoder.initDecode(TEST_DECODER_SETTINGS, mockDecoderCallback))
>+        .isEqualTo(VideoCodecStatus.OK);
>+
>+    // Verify.
>+    assertThat(fakeMediaCodecWrapper.getState()).isEqualTo(State.EXECUTING_RUNNING);
>+
>+    MediaFormat mediaFormat = fakeMediaCodecWrapper.getConfiguredFormat();
>+    assertThat(mediaFormat).isNotNull();
>+    assertThat(mediaFormat.getInteger(MediaFormat.KEY_WIDTH))
>+        .isEqualTo(TEST_DECODER_SETTINGS.width);
>+    assertThat(mediaFormat.getInteger(MediaFormat.KEY_HEIGHT))
>+        .isEqualTo(TEST_DECODER_SETTINGS.height);
>+    assertThat(mediaFormat.getString(MediaFormat.KEY_MIME))
>+        .isEqualTo(VideoCodecType.VP8.mimeType());
>+  }
>+
>+  @Test
>+  public void testRelease() {
>+    // Set-up.
>+    HardwareVideoDecoder decoder = new TestDecoderBuilder().build();
>+    decoder.initDecode(TEST_DECODER_SETTINGS, mockDecoderCallback);
>+
>+    // Test.
>+    assertThat(decoder.release()).isEqualTo(VideoCodecStatus.OK);
>+
>+    // Verify.
>+    assertThat(fakeMediaCodecWrapper.getState()).isEqualTo(State.RELEASED);
>+  }
>+
>+  @Test
>+  public void testReleaseMultipleTimes() {
>+    // Set-up.
>+    HardwareVideoDecoder decoder = new TestDecoderBuilder().build();
>+    decoder.initDecode(TEST_DECODER_SETTINGS, mockDecoderCallback);
>+
>+    // Test.
>+    assertThat(decoder.release()).isEqualTo(VideoCodecStatus.OK);
>+    assertThat(decoder.release()).isEqualTo(VideoCodecStatus.OK);
>+
>+    // Verify.
>+    assertThat(fakeMediaCodecWrapper.getState()).isEqualTo(State.RELEASED);
>+  }
>+
>+  @Test
>+  public void testDecodeQueuesData() {
>+    // Set-up.
>+    HardwareVideoDecoder decoder = new TestDecoderBuilder().build();
>+    decoder.initDecode(TEST_DECODER_SETTINGS, mockDecoderCallback);
>+
>+    // Test.
>+    assertThat(decoder.decode(createTestEncodedImage(),
>+                   new DecodeInfo(/* isMissingFrames= */ false, /* renderTimeMs= */ 0)))
>+        .isEqualTo(VideoCodecStatus.OK);
>+
>+    // Verify.
>+    ArgumentCaptor<Integer> indexCaptor = ArgumentCaptor.forClass(Integer.class);
>+    ArgumentCaptor<Integer> offsetCaptor = ArgumentCaptor.forClass(Integer.class);
>+    ArgumentCaptor<Integer> sizeCaptor = ArgumentCaptor.forClass(Integer.class);
>+    verify(fakeMediaCodecWrapper)
>+        .queueInputBuffer(indexCaptor.capture(), offsetCaptor.capture(), sizeCaptor.capture(),
>+            /* presentationTimeUs= */ anyLong(),
>+            /* flags= */ eq(0));
>+
>+    ByteBuffer inputBuffer = fakeMediaCodecWrapper.getInputBuffers()[indexCaptor.getValue()];
>+    CodecTestHelper.assertEqualContents(
>+        ENCODED_TEST_DATA, inputBuffer, offsetCaptor.getValue(), sizeCaptor.getValue());
>+  }
>+
>+  @Test
>+  public void testDeliversOutputByteBuffers() throws InterruptedException {
>+    final byte[] testOutputData = CodecTestHelper.generateRandomData(
>+        TEST_DECODER_SETTINGS.width * TEST_DECODER_SETTINGS.height * 3 / 2);
>+    final I420Buffer expectedDeliveredBuffer = CodecTestHelper.wrapI420(
>+        TEST_DECODER_SETTINGS.width, TEST_DECODER_SETTINGS.height, testOutputData);
>+
>+    // Set-up.
>+    TestDecoder decoder = new TestDecoderBuilder().setUseSurface(/* useSurface = */ false).build();
>+    decoder.initDecode(TEST_DECODER_SETTINGS, mockDecoderCallback);
>+    decoder.decode(createTestEncodedImage(),
>+        new DecodeInfo(/* isMissingFrames= */ false, /* renderTimeMs= */ 0));
>+    fakeMediaCodecWrapper.addOutputData(
>+        testOutputData, /* presentationTimestampUs= */ 0, /* flags= */ 0);
>+
>+    // Test.
>+    decoder.waitDeliverDecodedFrame();
>+
>+    // Verify.
>+    ArgumentCaptor<VideoFrame> videoFrameCaptor = ArgumentCaptor.forClass(VideoFrame.class);
>+    verify(mockDecoderCallback)
>+        .onDecodedFrame(videoFrameCaptor.capture(),
>+            /* decodeTimeMs= */ any(Integer.class),
>+            /* qp= */ any());
>+
>+    VideoFrame videoFrame = videoFrameCaptor.getValue();
>+    assertThat(videoFrame).isNotNull();
>+    assertThat(videoFrame.getRotatedWidth()).isEqualTo(TEST_DECODER_SETTINGS.width);
>+    assertThat(videoFrame.getRotatedHeight()).isEqualTo(TEST_DECODER_SETTINGS.height);
>+    assertThat(videoFrame.getRotation()).isEqualTo(0);
>+    I420Buffer deliveredBuffer = videoFrame.getBuffer().toI420();
>+    assertThat(deliveredBuffer.getDataY()).isEqualTo(expectedDeliveredBuffer.getDataY());
>+    assertThat(deliveredBuffer.getDataU()).isEqualTo(expectedDeliveredBuffer.getDataU());
>+    assertThat(deliveredBuffer.getDataV()).isEqualTo(expectedDeliveredBuffer.getDataV());
>+  }
>+
>+  @Test
>+  public void testRendersOutputTexture() throws InterruptedException {
>+    // Set-up.
>+    TestDecoder decoder = new TestDecoderBuilder().build();
>+    decoder.initDecode(TEST_DECODER_SETTINGS, mockDecoderCallback);
>+    decoder.decode(createTestEncodedImage(),
>+        new DecodeInfo(/* isMissingFrames= */ false, /* renderTimeMs= */ 0));
>+    int bufferIndex =
>+        fakeMediaCodecWrapper.addOutputTexture(/* presentationTimestampUs= */ 0, /* flags= */ 0);
>+
>+    // Test.
>+    decoder.waitDeliverDecodedFrame();
>+
>+    // Verify.
>+    verify(fakeMediaCodecWrapper).releaseOutputBuffer(bufferIndex, /* render= */ true);
>+  }
>+
>+  @Test
>+  public void testSurfaceTextureStall_FramesDropped() throws InterruptedException {
>+    final int numFrames = 10;
>+    // Maximum number of frame the decoder can keep queued on the output side.
>+    final int maxQueuedBuffers = 3;
>+
>+    // Set-up.
>+    TestDecoder decoder = new TestDecoderBuilder().build();
>+    decoder.initDecode(TEST_DECODER_SETTINGS, mockDecoderCallback);
>+
>+    // Test.
>+    int[] bufferIndices = new int[numFrames];
>+    for (int i = 0; i < 10; i++) {
>+      decoder.decode(createTestEncodedImage(),
>+          new DecodeInfo(/* isMissingFrames= */ false, /* renderTimeMs= */ 0));
>+      bufferIndices[i] =
>+          fakeMediaCodecWrapper.addOutputTexture(/* presentationTimestampUs= */ 0, /* flags= */ 0);
>+      decoder.waitDeliverDecodedFrame();
>+    }
>+
>+    // Verify.
>+    InOrder releaseOrder = inOrder(fakeMediaCodecWrapper);
>+    releaseOrder.verify(fakeMediaCodecWrapper)
>+        .releaseOutputBuffer(bufferIndices[0], /* render= */ true);
>+    for (int i = 1; i < numFrames - maxQueuedBuffers; i++) {
>+      releaseOrder.verify(fakeMediaCodecWrapper)
>+          .releaseOutputBuffer(bufferIndices[i], /* render= */ false);
>+    }
>+  }
>+
>+  @Test
>+  public void testDeliversRenderedBuffers() throws InterruptedException {
>+    // Set-up.
>+    TestDecoder decoder = new TestDecoderBuilder().build();
>+    decoder.initDecode(TEST_DECODER_SETTINGS, mockDecoderCallback);
>+    decoder.decode(createTestEncodedImage(),
>+        new DecodeInfo(/* isMissingFrames= */ false, /* renderTimeMs= */ 0));
>+    fakeMediaCodecWrapper.addOutputTexture(/* presentationTimestampUs= */ 0, /* flags= */ 0);
>+
>+    // Render the output buffer.
>+    decoder.waitDeliverDecodedFrame();
>+
>+    ArgumentCaptor<VideoSink> videoSinkCaptor = ArgumentCaptor.forClass(VideoSink.class);
>+    verify(mockSurfaceTextureHelper).startListening(videoSinkCaptor.capture());
>+
>+    // Test.
>+    Runnable releaseCallback = mock(Runnable.class);
>+    VideoFrame.TextureBuffer outputTextureBuffer =
>+        new TextureBufferImpl(TEST_DECODER_SETTINGS.width, TEST_DECODER_SETTINGS.height, Type.OES,
>+            /* id= */ 0,
>+            /* transformMatrix= */ new Matrix(),
>+            /* toI420Handler= */ new Handler(), new YuvConverter(), releaseCallback);
>+    VideoFrame outputVideoFrame =
>+        new VideoFrame(outputTextureBuffer, /* rotation= */ 0, /* timestampNs= */ 0);
>+    videoSinkCaptor.getValue().onFrame(outputVideoFrame);
>+    outputVideoFrame.release();
>+
>+    // Verify.
>+    ArgumentCaptor<VideoFrame> videoFrameCaptor = ArgumentCaptor.forClass(VideoFrame.class);
>+    verify(mockDecoderCallback)
>+        .onDecodedFrame(videoFrameCaptor.capture(),
>+            /* decodeTimeMs= */ any(Integer.class),
>+            /* qp= */ any());
>+
>+    VideoFrame videoFrame = videoFrameCaptor.getValue();
>+    assertThat(videoFrame).isNotNull();
>+    assertThat(videoFrame.getBuffer()).isEqualTo(outputTextureBuffer);
>+
>+    verify(releaseCallback).run();
>+  }
>+
>+  @Test
>+  public void testConfigureExceptionTriggerSWFallback() {
>+    // Set-up.
>+    doThrow(new IllegalStateException("Fake error"))
>+        .when(fakeMediaCodecWrapper)
>+        .configure(any(), any(), any(), anyInt());
>+
>+    HardwareVideoDecoder decoder = new TestDecoderBuilder().build();
>+
>+    // Test.
>+    assertThat(decoder.initDecode(TEST_DECODER_SETTINGS, mockDecoderCallback))
>+        .isEqualTo(VideoCodecStatus.FALLBACK_SOFTWARE);
>+  }
>+
>+  @Test
>+  public void testStartExceptionTriggerSWFallback() {
>+    // Set-up.
>+    doThrow(new IllegalStateException("Fake error")).when(fakeMediaCodecWrapper).start();
>+
>+    HardwareVideoDecoder decoder = new TestDecoderBuilder().build();
>+
>+    // Test.
>+    assertThat(decoder.initDecode(TEST_DECODER_SETTINGS, mockDecoderCallback))
>+        .isEqualTo(VideoCodecStatus.FALLBACK_SOFTWARE);
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/tests/src/org/webrtc/HardwareVideoEncoderTest.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/tests/src/org/webrtc/HardwareVideoEncoderTest.java
>new file mode 100644
>index 00000000000..bf5fcea2bbd
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/tests/src/org/webrtc/HardwareVideoEncoderTest.java
>@@ -0,0 +1,271 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc;
>+
>+import static com.google.common.truth.Truth.assertThat;
>+import static org.mockito.Matchers.any;
>+import static org.mockito.Matchers.anyInt;
>+import static org.mockito.Matchers.anyLong;
>+import static org.mockito.Mockito.spy;
>+import static org.mockito.Mockito.verify;
>+import static org.robolectric.Shadows.shadowOf;
>+
>+import android.media.MediaCodec;
>+import android.media.MediaCodecInfo;
>+import android.media.MediaFormat;
>+import java.nio.ByteBuffer;
>+import java.util.HashMap;
>+import java.util.Map;
>+import java.util.concurrent.Callable;
>+import java.util.concurrent.TimeUnit;
>+import org.chromium.testing.local.LocalRobolectricTestRunner;
>+import org.junit.Before;
>+import org.junit.Test;
>+import org.junit.runner.RunWith;
>+import org.mockito.ArgumentCaptor;
>+import org.mockito.Mock;
>+import org.mockito.MockitoAnnotations;
>+import org.robolectric.annotation.Config;
>+import org.robolectric.shadows.ShadowSystemClock;
>+import org.webrtc.EglBase.Context;
>+import org.webrtc.EncodedImage;
>+import org.webrtc.EncodedImage.FrameType;
>+import org.webrtc.FakeMediaCodecWrapper.State;
>+import org.webrtc.VideoCodecStatus;
>+import org.webrtc.VideoEncoder;
>+import org.webrtc.VideoEncoder.CodecSpecificInfo;
>+import org.webrtc.VideoEncoder.EncodeInfo;
>+import org.webrtc.VideoEncoder.Settings;
>+import org.webrtc.VideoFrame;
>+import org.webrtc.VideoFrame.Buffer;
>+import org.webrtc.VideoFrame.I420Buffer;
>+
>+@RunWith(LocalRobolectricTestRunner.class)
>+@Config(manifest = Config.NONE)
>+public class HardwareVideoEncoderTest {
>+  private static final VideoEncoder.Settings TEST_ENCODER_SETTINGS = new Settings(
>+      /* numberOfCores= */ 1,
>+      /* width= */ 640,
>+      /* height= */ 480,
>+      /* startBitrate= */ 10000,
>+      /* maxFramerate= */ 30,
>+      /* automaticResizeOn= */ true);
>+  private static final long POLL_DELAY_MS = 10;
>+  private static final long DELIVER_ENCODED_IMAGE_DELAY_MS = 10;
>+
>+  private static class TestEncoder extends HardwareVideoEncoder {
>+    private final Object deliverEncodedImageLock = new Object();
>+    private boolean deliverEncodedImageDone = true;
>+
>+    TestEncoder(MediaCodecWrapperFactory mediaCodecWrapperFactory, String codecName,
>+        VideoCodecType codecType, Integer surfaceColorFormat, Integer yuvColorFormat,
>+        Map<String, String> params, int keyFrameIntervalSec, int forceKeyFrameIntervalMs,
>+        BitrateAdjuster bitrateAdjuster, EglBase14.Context sharedContext) {
>+      super(mediaCodecWrapperFactory, codecName, codecType, surfaceColorFormat, yuvColorFormat,
>+          params, keyFrameIntervalSec, forceKeyFrameIntervalMs, bitrateAdjuster, sharedContext);
>+    }
>+
>+    public void waitDeliverEncodedImage() throws InterruptedException {
>+      synchronized (deliverEncodedImageLock) {
>+        deliverEncodedImageDone = false;
>+        deliverEncodedImageLock.notifyAll();
>+        while (!deliverEncodedImageDone) {
>+          deliverEncodedImageLock.wait();
>+        }
>+      }
>+    }
>+
>+    @SuppressWarnings("WaitNotInLoop") // This method is called inside a loop.
>+    @Override
>+    protected void deliverEncodedImage() {
>+      synchronized (deliverEncodedImageLock) {
>+        if (deliverEncodedImageDone) {
>+          try {
>+            deliverEncodedImageLock.wait(DELIVER_ENCODED_IMAGE_DELAY_MS);
>+          } catch (InterruptedException e) {
>+            Thread.currentThread().interrupt();
>+            return;
>+          }
>+        }
>+        if (deliverEncodedImageDone) {
>+          return;
>+        }
>+        super.deliverEncodedImage();
>+        deliverEncodedImageDone = true;
>+        deliverEncodedImageLock.notifyAll();
>+      }
>+    }
>+
>+    @Override
>+    protected void fillInputBuffer(ByteBuffer buffer, Buffer videoFrameBuffer) {
>+      I420Buffer i420Buffer = videoFrameBuffer.toI420();
>+      buffer.put(i420Buffer.getDataY());
>+      buffer.put(i420Buffer.getDataU());
>+      buffer.put(i420Buffer.getDataV());
>+      buffer.flip();
>+      i420Buffer.release();
>+    }
>+  }
>+
>+  private class TestEncoderBuilder {
>+    private VideoCodecType codecType = VideoCodecType.VP8;
>+
>+    public TestEncoderBuilder setCodecType(VideoCodecType codecType) {
>+      this.codecType = codecType;
>+      return this;
>+    }
>+
>+    public TestEncoder build() {
>+      return new TestEncoder((String name)
>+                                 -> fakeMediaCodecWrapper,
>+          "org.webrtc.testencoder", codecType,
>+          /* surfaceColorFormat= */ null,
>+          /* yuvColorFormat= */ MediaCodecInfo.CodecCapabilities.COLOR_FormatYUV420Planar,
>+          /* params= */ new HashMap<>(),
>+          /* keyFrameIntervalSec= */ 0,
>+          /* forceKeyFrameIntervalMs= */ 0,
>+          /* bitrateAdjuster= */ new BaseBitrateAdjuster(),
>+          /* sharedContext= */ null);
>+    }
>+  }
>+
>+  @Mock VideoEncoder.Callback mockEncoderCallback;
>+  private FakeMediaCodecWrapper fakeMediaCodecWrapper;
>+
>+  @Before
>+  public void setUp() {
>+    MockitoAnnotations.initMocks(this);
>+    MediaFormat outputFormat = new MediaFormat();
>+    // TODO(sakal): Add more details to output format as needed.
>+    fakeMediaCodecWrapper = spy(new FakeMediaCodecWrapper(outputFormat));
>+  }
>+
>+  @Test
>+  public void testInit() {
>+    // Set-up.
>+    HardwareVideoEncoder encoder =
>+        new TestEncoderBuilder().setCodecType(VideoCodecType.VP8).build();
>+
>+    // Test.
>+    assertThat(encoder.initEncode(TEST_ENCODER_SETTINGS, mockEncoderCallback))
>+        .isEqualTo(VideoCodecStatus.OK);
>+
>+    // Verify.
>+    assertThat(fakeMediaCodecWrapper.getState()).isEqualTo(State.EXECUTING_RUNNING);
>+
>+    MediaFormat mediaFormat = fakeMediaCodecWrapper.getConfiguredFormat();
>+    assertThat(mediaFormat).isNotNull();
>+    assertThat(mediaFormat.getInteger(MediaFormat.KEY_WIDTH))
>+        .isEqualTo(TEST_ENCODER_SETTINGS.width);
>+    assertThat(mediaFormat.getInteger(MediaFormat.KEY_HEIGHT))
>+        .isEqualTo(TEST_ENCODER_SETTINGS.height);
>+    assertThat(mediaFormat.getString(MediaFormat.KEY_MIME))
>+        .isEqualTo(VideoCodecType.VP8.mimeType());
>+
>+    assertThat(fakeMediaCodecWrapper.getConfiguredFlags())
>+        .isEqualTo(MediaCodec.CONFIGURE_FLAG_ENCODE);
>+  }
>+
>+  @Test
>+  public void testEncodeByteBuffer() {
>+    // Set-up.
>+    HardwareVideoEncoder encoder = new TestEncoderBuilder().build();
>+    encoder.initEncode(TEST_ENCODER_SETTINGS, mockEncoderCallback);
>+
>+    // Test.
>+    byte[] i420 = CodecTestHelper.generateRandomData(
>+        TEST_ENCODER_SETTINGS.width * TEST_ENCODER_SETTINGS.height * 3 / 2);
>+    final VideoFrame.I420Buffer testBuffer =
>+        CodecTestHelper.wrapI420(TEST_ENCODER_SETTINGS.width, TEST_ENCODER_SETTINGS.height, i420);
>+    final VideoFrame testFrame =
>+        new VideoFrame(testBuffer, /* rotation= */ 0, /* timestampNs= */ 0);
>+    assertThat(encoder.encode(testFrame, new EncodeInfo(new FrameType[] {FrameType.VideoFrameKey})))
>+        .isEqualTo(VideoCodecStatus.OK);
>+
>+    // Verify.
>+    ArgumentCaptor<Integer> indexCaptor = ArgumentCaptor.forClass(Integer.class);
>+    ArgumentCaptor<Integer> offsetCaptor = ArgumentCaptor.forClass(Integer.class);
>+    ArgumentCaptor<Integer> sizeCaptor = ArgumentCaptor.forClass(Integer.class);
>+    verify(fakeMediaCodecWrapper)
>+        .queueInputBuffer(indexCaptor.capture(), offsetCaptor.capture(), sizeCaptor.capture(),
>+            anyLong(), anyInt());
>+    ByteBuffer buffer = fakeMediaCodecWrapper.getInputBuffers()[indexCaptor.getValue()];
>+    CodecTestHelper.assertEqualContents(
>+        i420, buffer, offsetCaptor.getValue(), sizeCaptor.getValue());
>+  }
>+
>+  @Test
>+  public void testDeliversOutputData() throws InterruptedException {
>+    final int outputDataLength = 100;
>+
>+    // Set-up.
>+    TestEncoder encoder = new TestEncoderBuilder().build();
>+    encoder.initEncode(TEST_ENCODER_SETTINGS, mockEncoderCallback);
>+    byte[] i420 = CodecTestHelper.generateRandomData(
>+        TEST_ENCODER_SETTINGS.width * TEST_ENCODER_SETTINGS.height * 3 / 2);
>+    final VideoFrame.I420Buffer testBuffer =
>+        CodecTestHelper.wrapI420(TEST_ENCODER_SETTINGS.width, TEST_ENCODER_SETTINGS.height, i420);
>+    final VideoFrame testFrame =
>+        new VideoFrame(testBuffer, /* rotation= */ 0, /* timestampNs= */ 42);
>+    encoder.encode(testFrame, new EncodeInfo(new FrameType[] {FrameType.VideoFrameKey}));
>+
>+    // Test.
>+    byte[] outputData = CodecTestHelper.generateRandomData(outputDataLength);
>+    fakeMediaCodecWrapper.addOutputData(outputData,
>+        /* presentationTimestampUs= */ 0,
>+        /* flags= */ MediaCodec.BUFFER_FLAG_SYNC_FRAME);
>+
>+    encoder.waitDeliverEncodedImage();
>+
>+    // Verify.
>+    ArgumentCaptor<EncodedImage> videoFrameCaptor = ArgumentCaptor.forClass(EncodedImage.class);
>+    verify(mockEncoderCallback)
>+        .onEncodedFrame(videoFrameCaptor.capture(), any(CodecSpecificInfo.class));
>+
>+    EncodedImage videoFrame = videoFrameCaptor.getValue();
>+    assertThat(videoFrame).isNotNull();
>+    assertThat(videoFrame.encodedWidth).isEqualTo(TEST_ENCODER_SETTINGS.width);
>+    assertThat(videoFrame.encodedHeight).isEqualTo(TEST_ENCODER_SETTINGS.height);
>+    assertThat(videoFrame.rotation).isEqualTo(0);
>+    assertThat(videoFrame.captureTimeNs).isEqualTo(42);
>+    assertThat(videoFrame.completeFrame).isTrue();
>+    assertThat(videoFrame.frameType).isEqualTo(FrameType.VideoFrameKey);
>+    CodecTestHelper.assertEqualContents(
>+        outputData, videoFrame.buffer, /* offset= */ 0, videoFrame.buffer.capacity());
>+  }
>+
>+  @Test
>+  public void testRelease() {
>+    // Set-up.
>+    HardwareVideoEncoder encoder = new TestEncoderBuilder().build();
>+    encoder.initEncode(TEST_ENCODER_SETTINGS, mockEncoderCallback);
>+
>+    // Test.
>+    assertThat(encoder.release()).isEqualTo(VideoCodecStatus.OK);
>+
>+    // Verify.
>+    assertThat(fakeMediaCodecWrapper.getState()).isEqualTo(State.RELEASED);
>+  }
>+
>+  @Test
>+  public void testReleaseMultipleTimes() {
>+    // Set-up.
>+    HardwareVideoEncoder encoder = new TestEncoderBuilder().build();
>+    encoder.initEncode(TEST_ENCODER_SETTINGS, mockEncoderCallback);
>+
>+    // Test.
>+    assertThat(encoder.release()).isEqualTo(VideoCodecStatus.OK);
>+    assertThat(encoder.release()).isEqualTo(VideoCodecStatus.OK);
>+
>+    // Verify.
>+    assertThat(fakeMediaCodecWrapper.getState()).isEqualTo(State.RELEASED);
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/tests/src/org/webrtc/ScalingSettingsTest.java b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/tests/src/org/webrtc/ScalingSettingsTest.java
>new file mode 100644
>index 00000000000..a092a7c4853
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/tests/src/org/webrtc/ScalingSettingsTest.java
>@@ -0,0 +1,29 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+package org.webrtc;
>+
>+import static org.junit.Assert.assertEquals;
>+
>+import org.chromium.testing.local.LocalRobolectricTestRunner;
>+import org.junit.Test;
>+import org.junit.runner.RunWith;
>+import org.robolectric.annotation.Config;
>+import org.webrtc.VideoEncoder.ScalingSettings;
>+
>+@RunWith(LocalRobolectricTestRunner.class)
>+@Config(manifest = Config.NONE)
>+public class ScalingSettingsTest {
>+  @Test
>+  public void testToString() {
>+    assertEquals("[ 1, 2 ]", new ScalingSettings(1, 2).toString());
>+    assertEquals("OFF", ScalingSettings.OFF.toString());
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Audio/RTCAudioSession+Private.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Audio/RTCAudioSession+Private.h
>index 8be8aadb0de..f163b5a1331 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Audio/RTCAudioSession+Private.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Audio/RTCAudioSession+Private.h
>@@ -77,14 +77,12 @@ NS_ASSUME_NONNULL_BEGIN
> - (NSError *)configurationErrorWithDescription:(NSString *)description;
> 
> // Properties and methods for tests.
>-@property(nonatomic, readonly)
>-    std::vector<__weak id<RTCAudioSessionDelegate> > delegates;
>+@property(nonatomic, readonly) std::vector<__weak id<RTCAudioSessionDelegate> > delegates;
> 
> - (void)notifyDidBeginInterruption;
>-- (void)notifyDidEndInterruptionWithShouldResumeSession:
>-    (BOOL)shouldResumeSession;
>+- (void)notifyDidEndInterruptionWithShouldResumeSession:(BOOL)shouldResumeSession;
> - (void)notifyDidChangeRouteWithReason:(AVAudioSessionRouteChangeReason)reason
>-    previousRoute:(AVAudioSessionRouteDescription *)previousRoute;
>+                         previousRoute:(AVAudioSessionRouteDescription *)previousRoute;
> - (void)notifyMediaServicesWereLost;
> - (void)notifyMediaServicesWereReset;
> - (void)notifyDidChangeCanPlayOrRecord:(BOOL)canPlayOrRecord;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Audio/RTCAudioSession.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Audio/RTCAudioSession.mm
>index 67a70da8002..307b97959e4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Audio/RTCAudioSession.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Audio/RTCAudioSession.mm
>@@ -335,6 +335,7 @@ - (BOOL)setActive:(BOOL)active
>   if (!active && activationCount == 0) {
>     RTCLogWarning(@"Attempting to deactivate without prior activation.");
>   }
>+  [self notifyWillSetActive:active];
>   BOOL success = YES;
>   BOOL isActive = self.isActive;
>   // Keep a local error so we can log it.
>@@ -365,9 +366,11 @@ - (BOOL)setActive:(BOOL)active
>     if (active) {
>       [self incrementActivationCount];
>     }
>+    [self notifyDidSetActive:active];
>   } else {
>     RTCLogError(@"Failed to setActive:%d. Error: %@",
>                 active, error.localizedDescription);
>+    [self notifyFailedToSetActive:active error:error];
>   }
>   // Decrement activation count on deactivation whether or not it succeeded.
>   if (!active) {
>@@ -565,9 +568,11 @@ - (void)handleSilenceSecondaryAudioHintNotification:(NSNotification *)notificati
> }
> 
> - (void)handleApplicationDidBecomeActive:(NSNotification *)notification {
>+  BOOL isInterrupted = self.isInterrupted;
>   RTCLog(@"Application became active after an interruption. Treating as interruption "
>-         " end. isInterrupted changed from %d to 0.", self.isInterrupted);
>-  if (self.isInterrupted) {
>+          "end. isInterrupted changed from %d to 0.",
>+         isInterrupted);
>+  if (isInterrupted) {
>     self.isInterrupted = NO;
>     [self updateAudioSessionAfterEvent];
>   }
>@@ -806,14 +811,26 @@ - (void)audioSessionDidActivate:(AVAudioSession *)session {
>   if (_session != session) {
>     RTCLogError(@"audioSessionDidActivate called on different AVAudioSession");
>   }
>+  RTCLog(@"Audio session was externally activated.");
>   [self incrementActivationCount];
>   self.isActive = YES;
>+  // When a CallKit call begins, it's possible that we receive an interruption
>+  // begin without a corresponding end. Since we know that we have an activated
>+  // audio session at this point, just clear any saved interruption flag since
>+  // the app may never be foregrounded during the duration of the call.
>+  if (self.isInterrupted) {
>+    RTCLog(@"Clearing interrupted state due to external activation.");
>+    self.isInterrupted = NO;
>+  }
>+  // Treat external audio session activation as an end interruption event.
>+  [self notifyDidEndInterruptionWithShouldResumeSession:YES];
> }
> 
> - (void)audioSessionDidDeactivate:(AVAudioSession *)session {
>   if (_session != session) {
>     RTCLogError(@"audioSessionDidDeactivate called on different AVAudioSession");
>   }
>+  RTCLog(@"Audio session was externally deactivated.");
>   self.isActive = NO;
>   [self decrementActivationCount];
> }
>@@ -931,4 +948,31 @@ - (void)notifyDidDetectPlayoutGlitch:(int64_t)totalNumberOfGlitches {
>   }
> }
> 
>+- (void)notifyWillSetActive:(BOOL)active {
>+  for (id delegate : self.delegates) {
>+    SEL sel = @selector(audioSession:willSetActive:);
>+    if ([delegate respondsToSelector:sel]) {
>+      [delegate audioSession:self willSetActive:active];
>+    }
>+  }
>+}
>+
>+- (void)notifyDidSetActive:(BOOL)active {
>+  for (id delegate : self.delegates) {
>+    SEL sel = @selector(audioSession:didSetActive:);
>+    if ([delegate respondsToSelector:sel]) {
>+      [delegate audioSession:self didSetActive:active];
>+    }
>+  }
>+}
>+
>+- (void)notifyFailedToSetActive:(BOOL)active error:(NSError *)error {
>+  for (id delegate : self.delegates) {
>+    SEL sel = @selector(audioSession:failedToSetActive:error:);
>+    if ([delegate respondsToSelector:sel]) {
>+      [delegate audioSession:self failedToSetActive:active error:error];
>+    }
>+  }
>+}
>+
> @end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Audio/RTCNativeAudioSessionDelegateAdapter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Audio/RTCNativeAudioSessionDelegateAdapter.h
>new file mode 100644
>index 00000000000..7fb21842604
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Audio/RTCNativeAudioSessionDelegateAdapter.h
>@@ -0,0 +1,33 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#import "sdk/objc/Framework/Headers/WebRTC/RTCAudioSession.h"
>+
>+NS_ASSUME_NONNULL_BEGIN
>+
>+namespace webrtc {
>+class AudioSessionObserver;
>+}
>+
>+/** Adapter that forwards RTCAudioSessionDelegate calls to the appropriate
>+ *  methods on the AudioSessionObserver.
>+ */
>+@interface RTCNativeAudioSessionDelegateAdapter : NSObject <RTCAudioSessionDelegate>
>+
>+- (instancetype)init NS_UNAVAILABLE;
>+
>+/** |observer| is a raw pointer and should be kept alive
>+ *  for this object's lifetime.
>+ */
>+- (instancetype)initWithObserver:(webrtc::AudioSessionObserver *)observer NS_DESIGNATED_INITIALIZER;
>+
>+@end
>+
>+NS_ASSUME_NONNULL_END
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Audio/RTCNativeAudioSessionDelegateAdapter.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Audio/RTCNativeAudioSessionDelegateAdapter.mm
>new file mode 100644
>index 00000000000..a443e51e38f
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Audio/RTCNativeAudioSessionDelegateAdapter.mm
>@@ -0,0 +1,89 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#import "RTCNativeAudioSessionDelegateAdapter.h"
>+
>+#include "sdk/objc/Framework/Native/src/audio/audio_session_observer.h"
>+
>+#import "WebRTC/RTCLogging.h"
>+
>+@implementation RTCNativeAudioSessionDelegateAdapter {
>+  webrtc::AudioSessionObserver *_observer;
>+}
>+
>+- (instancetype)initWithObserver:(webrtc::AudioSessionObserver *)observer {
>+  RTC_DCHECK(observer);
>+  if (self = [super init]) {
>+    _observer = observer;
>+  }
>+  return self;
>+}
>+
>+#pragma mark - RTCAudioSessionDelegate
>+
>+- (void)audioSessionDidBeginInterruption:(RTCAudioSession *)session {
>+  _observer->OnInterruptionBegin();
>+}
>+
>+- (void)audioSessionDidEndInterruption:(RTCAudioSession *)session
>+                   shouldResumeSession:(BOOL)shouldResumeSession {
>+  _observer->OnInterruptionEnd();
>+}
>+
>+- (void)audioSessionDidChangeRoute:(RTCAudioSession *)session
>+           reason:(AVAudioSessionRouteChangeReason)reason
>+    previousRoute:(AVAudioSessionRouteDescription *)previousRoute {
>+  switch (reason) {
>+    case AVAudioSessionRouteChangeReasonUnknown:
>+    case AVAudioSessionRouteChangeReasonNewDeviceAvailable:
>+    case AVAudioSessionRouteChangeReasonOldDeviceUnavailable:
>+    case AVAudioSessionRouteChangeReasonCategoryChange:
>+      // It turns out that we see a category change (at least in iOS 9.2)
>+      // when making a switch from a BT device to e.g. Speaker using the
>+      // iOS Control Center and that we therefore must check if the sample
>+      // rate has changed. And if so is the case, restart the audio unit.
>+    case AVAudioSessionRouteChangeReasonOverride:
>+    case AVAudioSessionRouteChangeReasonWakeFromSleep:
>+    case AVAudioSessionRouteChangeReasonNoSuitableRouteForCategory:
>+      _observer->OnValidRouteChange();
>+      break;
>+    case AVAudioSessionRouteChangeReasonRouteConfigurationChange:
>+      // The set of input and output ports has not changed, but their
>+      // configuration has, e.g., a portâs selected data source has
>+      // changed. Ignore this type of route change since we are focusing
>+      // on detecting headset changes.
>+      RTCLog(@"Ignoring RouteConfigurationChange");
>+      break;
>+  }
>+}
>+
>+- (void)audioSessionMediaServerTerminated:(RTCAudioSession *)session {
>+}
>+
>+- (void)audioSessionMediaServerReset:(RTCAudioSession *)session {
>+}
>+
>+- (void)audioSession:(RTCAudioSession *)session
>+    didChangeCanPlayOrRecord:(BOOL)canPlayOrRecord {
>+  _observer->OnCanPlayOrRecordChange(canPlayOrRecord);
>+}
>+
>+- (void)audioSessionDidStartPlayOrRecord:(RTCAudioSession *)session {
>+}
>+
>+- (void)audioSessionDidStopPlayOrRecord:(RTCAudioSession *)session {
>+}
>+
>+- (void)audioSession:(RTCAudioSession *)audioSession
>+    didChangeOutputVolume:(float)outputVolume {
>+  _observer->OnChangedOutputVolume();
>+}
>+
>+@end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Common/NSString+StdString.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Common/NSString+StdString.h
>index 8bf6cc94be0..7f51a9f3c67 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Common/NSString+StdString.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Common/NSString+StdString.h
>@@ -19,7 +19,7 @@ NS_ASSUME_NONNULL_BEGIN
> @property(nonatomic, readonly) std::string stdString;
> 
> + (std::string)stdStringForString:(NSString *)nsString;
>-+ (NSString *)stringForStdString:(const std::string&)stdString;
>++ (NSString *)stringForStdString:(const std::string &)stdString;
> 
> @end
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Common/RTCCallbackLogger.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Common/RTCCallbackLogger.mm
>new file mode 100644
>index 00000000000..4802c12b742
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Common/RTCCallbackLogger.mm
>@@ -0,0 +1,92 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#import "WebRTC/RTCCallbackLogger.h"
>+
>+#include <memory>
>+
>+#include "rtc_base/checks.h"
>+#include "rtc_base/logging.h"
>+#include "rtc_base/logsinks.h"
>+
>+class CallbackLogSink : public rtc::LogSink {
>+ public:
>+  CallbackLogSink(void (^callbackHandler)(NSString *message)) {
>+    callback_handler_ = callbackHandler;
>+  }
>+
>+  ~CallbackLogSink() override { callback_handler_ = nil; }
>+
>+  void OnLogMessage(const std::string &message) override {
>+    if (callback_handler_) {
>+      callback_handler_([NSString stringWithUTF8String:message.c_str()]);
>+    }
>+  }
>+
>+ private:
>+  void (^callback_handler_)(NSString *message);
>+};
>+
>+@implementation RTCCallbackLogger {
>+  BOOL _hasStarted;
>+  std::unique_ptr<CallbackLogSink> _logSink;
>+}
>+
>+@synthesize severity = _severity;
>+
>+- (instancetype)init {
>+  self = [super init];
>+  if (self != nil) {
>+    _severity = RTCLoggingSeverityInfo;
>+  }
>+  return self;
>+}
>+
>+- (void)dealloc {
>+  [self stop];
>+}
>+
>+- (void)start:(nullable void (^)(NSString *))callback {
>+  if (_hasStarted) {
>+    return;
>+  }
>+
>+  _logSink.reset(new CallbackLogSink(callback));
>+
>+  rtc::LogMessage::AddLogToStream(_logSink.get(), [self rtcSeverity]);
>+  _hasStarted = YES;
>+}
>+
>+- (void)stop {
>+  if (!_hasStarted) {
>+    return;
>+  }
>+  RTC_DCHECK(_logSink);
>+  rtc::LogMessage::RemoveLogToStream(_logSink.get());
>+  _hasStarted = NO;
>+  _logSink.reset();
>+}
>+
>+#pragma mark - Private
>+
>+- (rtc::LoggingSeverity)rtcSeverity {
>+  switch (_severity) {
>+    case RTCLoggingSeverityVerbose:
>+      return rtc::LS_VERBOSE;
>+    case RTCLoggingSeverityInfo:
>+      return rtc::LS_INFO;
>+    case RTCLoggingSeverityWarning:
>+      return rtc::LS_WARNING;
>+    case RTCLoggingSeverityError:
>+      return rtc::LS_ERROR;
>+  }
>+}
>+
>+@end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Common/RTCFieldTrials.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Common/RTCFieldTrials.mm
>index c95bcb4a408..9cbd640f0c5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Common/RTCFieldTrials.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Common/RTCFieldTrials.mm
>@@ -20,6 +20,9 @@
> #include "system_wrappers/include/field_trial_default.h"  // nogncheck
> 
> NSString * const kRTCFieldTrialAudioSendSideBweKey = @"WebRTC-Audio-SendSideBwe";
>+NSString * const kRTCFieldTrialAudioSendSideBweForVideoKey = @"WebRTC-Audio-SendSideBwe-For-Video";
>+NSString * const kRTCFieldTrialAudioForceNoTWCCKey = @"WebRTC-Audio-ForceNoTWCC";
>+NSString * const kRTCFieldTrialAudioForceABWENoTWCCKey = @"WebRTC-Audio-ABWENoTWCC";
> NSString * const kRTCFieldTrialSendSideBweWithOverheadKey = @"WebRTC-SendSideBwe-WithOverhead";
> NSString * const kRTCFieldTrialFlexFec03AdvertisedKey = @"WebRTC-FlexFEC-03-Advertised";
> NSString * const kRTCFieldTrialFlexFec03Key = @"WebRTC-FlexFEC-03";
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Common/RTCUIApplicationStatusObserver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Common/RTCUIApplicationStatusObserver.h
>index 8ee12b62499..c7172288033 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Common/RTCUIApplicationStatusObserver.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Common/RTCUIApplicationStatusObserver.h
>@@ -13,6 +13,7 @@
> #import <Foundation/Foundation.h>
> 
> __attribute__((objc_runtime_name("WK_RTCUIApplicationStatusObserver")))
>+NS_EXTENSION_UNAVAILABLE_IOS("Application status not available in app extensions.")
> @interface RTCUIApplicationStatusObserver : NSObject
> 
> + (instancetype)sharedInstance;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Common/UIDevice+RTCDevice.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Common/UIDevice+RTCDevice.mm
>index b2cfb2094b4..997666d5035 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Common/UIDevice+RTCDevice.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Common/UIDevice+RTCDevice.mm
>@@ -21,6 +21,7 @@ + (RTCDeviceType)deviceType {
>     @"iPhone1,2": @(RTCDeviceTypeIPhone3G),
>     @"iPhone2,1": @(RTCDeviceTypeIPhone3GS),
>     @"iPhone3,1": @(RTCDeviceTypeIPhone4),
>+    @"iPhone3,2": @(RTCDeviceTypeIPhone4),
>     @"iPhone3,3": @(RTCDeviceTypeIPhone4Verizon),
>     @"iPhone4,1": @(RTCDeviceTypeIPhone4S),
>     @"iPhone5,1": @(RTCDeviceTypeIPhone5GSM),
>@@ -33,6 +34,7 @@ + (RTCDeviceType)deviceType {
>     @"iPhone7,2": @(RTCDeviceTypeIPhone6),
>     @"iPhone8,1": @(RTCDeviceTypeIPhone6S),
>     @"iPhone8,2": @(RTCDeviceTypeIPhone6SPlus),
>+    @"iPhone8,4": @(RTCDeviceTypeIPhoneSE),
>     @"iPhone9,1": @(RTCDeviceTypeIPhone7),
>     @"iPhone9,2": @(RTCDeviceTypeIPhone7Plus),
>     @"iPhone9,3": @(RTCDeviceTypeIPhone7),
>@@ -48,6 +50,7 @@ + (RTCDeviceType)deviceType {
>     @"iPod3,1": @(RTCDeviceTypeIPodTouch3G),
>     @"iPod4,1": @(RTCDeviceTypeIPodTouch4G),
>     @"iPod5,1": @(RTCDeviceTypeIPodTouch5G),
>+    @"iPod7,1": @(RTCDeviceTypeIPodTouch6G),
>     @"iPad1,1": @(RTCDeviceTypeIPad),
>     @"iPad2,1": @(RTCDeviceTypeIPad2Wifi),
>     @"iPad2,2": @(RTCDeviceTypeIPad2GSM),
>@@ -64,8 +67,29 @@ + (RTCDeviceType)deviceType {
>     @"iPad3,6": @(RTCDeviceTypeIPad4GSM_CDMA),
>     @"iPad4,1": @(RTCDeviceTypeIPadAirWifi),
>     @"iPad4,2": @(RTCDeviceTypeIPadAirCellular),
>+    @"iPad4,3": @(RTCDeviceTypeIPadAirWifiCellular),
>     @"iPad4,4": @(RTCDeviceTypeIPadMini2GWifi),
>     @"iPad4,5": @(RTCDeviceTypeIPadMini2GCellular),
>+    @"iPad4,6": @(RTCDeviceTypeIPadMini2GWifiCellular),
>+    @"iPad4,7": @(RTCDeviceTypeIPadMini3),
>+    @"iPad4,8": @(RTCDeviceTypeIPadMini3),
>+    @"iPad4,9": @(RTCDeviceTypeIPadMini3),
>+    @"iPad5,1": @(RTCDeviceTypeIPadMini4),
>+    @"iPad5,2": @(RTCDeviceTypeIPadMini4),
>+    @"iPad5,3": @(RTCDeviceTypeIPadAir2),
>+    @"iPad5,4": @(RTCDeviceTypeIPadAir2),
>+    @"iPad6,3": @(RTCDeviceTypeIPadPro9Inch),
>+    @"iPad6,4": @(RTCDeviceTypeIPadPro9Inch),
>+    @"iPad6,7": @(RTCDeviceTypeIPadPro12Inch),
>+    @"iPad6,8": @(RTCDeviceTypeIPadPro12Inch),
>+    @"iPad6,11": @(RTCDeviceTypeIPad5),
>+    @"iPad6,12": @(RTCDeviceTypeIPad5),
>+    @"iPad7,1": @(RTCDeviceTypeIPadPro12Inch2),
>+    @"iPad7,2": @(RTCDeviceTypeIPadPro12Inch2),
>+    @"iPad7,3": @(RTCDeviceTypeIPadPro10Inch),
>+    @"iPad7,4": @(RTCDeviceTypeIPadPro10Inch),
>+    @"iPad7,5": @(RTCDeviceTypeIPad6),
>+    @"iPad7,6": @(RTCDeviceTypeIPad6),
>     @"i386": @(RTCDeviceTypeSimulatori386),
>     @"x86_64": @(RTCDeviceTypeSimulatorx86_64),
>   };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Common/helpers.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Common/helpers.h
>index ca56d518015..7fd9b34a991 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Common/helpers.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Common/helpers.h
>@@ -18,7 +18,7 @@ namespace ios {
> 
> bool CheckAndLogError(BOOL success, NSError* error);
> 
>-NSString *NSStringFromStdString(const std::string& stdString);
>+NSString* NSStringFromStdString(const std::string& stdString);
> std::string StdStringFromNSString(NSString* nsString);
> 
> // Return thread ID as a string.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/session.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Common/noop.mm
>similarity index 57%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/session.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Common/noop.mm
>index a98a5efe13a..16a8e6d5c10 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/session.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Common/noop.mm
>@@ -1,5 +1,5 @@
> /*
>- *  Copyright 2004 The WebRTC Project Authors. All rights reserved.
>+ *  Copyright 2015 The WebRTC Project Authors. All rights reserved.
>  *
>  *  Use of this source code is governed by a BSD-style license
>  *  that can be found in the LICENSE file in the root of the source
>@@ -8,6 +8,6 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-// TODO(deadbeef): Remove this file when Chrome build files no longer reference
>-// it.
>-#error "DONT INCLUDE THIS"
>+// This file is only needed to make ninja happy on some platforms.
>+// On some platforms it is not possible to link an rtc_static_library
>+// without any source file listed in the GN target.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLI420Renderer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLI420Renderer.h
>index 315b1f5a8da..e5987fe22a2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLI420Renderer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLI420Renderer.h
>@@ -13,6 +13,5 @@
> #import "RTCMTLRenderer.h"
> 
> NS_AVAILABLE(10_11, 9_0)
>-__attribute__((objc_runtime_name("WK_RTCMTLI420Renderer")))
> @interface RTCMTLI420Renderer : RTCMTLRenderer
> @end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLI420Renderer.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLI420Renderer.mm
>index 027db05b391..f7a75d8056c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLI420Renderer.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLI420Renderer.mm
>@@ -19,10 +19,10 @@
> 
> #import "RTCMTLRenderer+Private.h"
> 
>-#define MTL_STRINGIFY(s) @ #s
>-
> static NSString *const shaderSource = MTL_STRINGIFY(
>-    using namespace metal; typedef struct {
>+    using namespace metal;
>+
>+    typedef struct {
>       packed_float2 position;
>       packed_float2 texcoord;
>     } Vertex;
>@@ -89,8 +89,25 @@ - (NSString *)shaderSource {
>   return shaderSource;
> }
> 
>+- (void)getWidth:(nonnull int *)width
>+          height:(nonnull int *)height
>+       cropWidth:(nonnull int *)cropWidth
>+      cropHeight:(nonnull int *)cropHeight
>+           cropX:(nonnull int *)cropX
>+           cropY:(nonnull int *)cropY
>+         ofFrame:(nonnull RTCVideoFrame *)frame {
>+  *width = frame.width;
>+  *height = frame.height;
>+  *cropWidth = frame.width;
>+  *cropHeight = frame.height;
>+  *cropX = 0;
>+  *cropY = 0;
>+}
>+
> - (BOOL)setupTexturesForFrame:(nonnull RTCVideoFrame *)frame {
>-  [super setupTexturesForFrame:frame];
>+  if (![super setupTexturesForFrame:frame]) {
>+    return NO;
>+  }
> 
>   id<MTLDevice> device = [self currentMetalDevice];
>   if (!device) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLNSVideoView.m b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLNSVideoView.m
>index ff23bc6e6e1..869858e6b4c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLNSVideoView.m
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLNSVideoView.m
>@@ -27,6 +27,7 @@ @implementation RTCMTLNSVideoView {
>   id<RTCMTLRenderer> _renderer;
> }
> 
>+@synthesize delegate = _delegate;
> @synthesize renderer = _renderer;
> @synthesize metalView = _metalView;
> @synthesize videoFrame = _videoFrame;
>@@ -105,6 +106,9 @@ - (void)mtkView:(MTKView *)view drawableSizeWillChange:(CGSize)size {
> 
> - (void)setSize:(CGSize)size {
>   _metalView.drawableSize = size;
>+  dispatch_async(dispatch_get_main_queue(), ^{
>+    [self.delegate videoView:self didChangeVideoSize:size];
>+  });
>   [_metalView draw];
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLNV12Renderer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLNV12Renderer.h
>index 8bbcb204d25..866b7ea17ee 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLNV12Renderer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLNV12Renderer.h
>@@ -13,7 +13,6 @@
> #import "RTCMTLRenderer.h"
> 
> NS_AVAILABLE(10_11, 9_0)
>-__attribute__((objc_runtime_name("WK_RTCMTLNV12Renderer")))
> @interface RTCMTLNV12Renderer : RTCMTLRenderer
> 
> @end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLNV12Renderer.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLNV12Renderer.mm
>index 079b5fb76c5..a968014c35b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLNV12Renderer.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLNV12Renderer.mm
>@@ -18,11 +18,12 @@
> #import "WebRTC/RTCVideoFrameBuffer.h"
> 
> #import "RTCMTLRenderer+Private.h"
>-
>-#define MTL_STRINGIFY(s) @ #s
>+#include "rtc_base/checks.h"
> 
> static NSString *const shaderSource = MTL_STRINGIFY(
>-    using namespace metal; typedef struct {
>+    using namespace metal;
>+
>+    typedef struct {
>       packed_float2 position;
>       packed_float2 texcoord;
>     } Vertex;
>@@ -66,26 +67,47 @@ @implementation RTCMTLNV12Renderer {
> 
> - (BOOL)addRenderingDestination:(__kindof MTKView *)view {
>   if ([super addRenderingDestination:view]) {
>-    [self initializeTextureCache];
>-    return YES;
>+    return [self initializeTextureCache];
>   }
>   return NO;
> }
> 
>-- (void)initializeTextureCache {
>+- (BOOL)initializeTextureCache {
>   CVReturn status = CVMetalTextureCacheCreate(kCFAllocatorDefault, nil, [self currentMetalDevice],
>                                               nil, &_textureCache);
>   if (status != kCVReturnSuccess) {
>     RTCLogError(@"Metal: Failed to initialize metal texture cache. Return status is %d", status);
>+    return NO;
>   }
>+
>+  return YES;
> }
> 
> - (NSString *)shaderSource {
>   return shaderSource;
> }
> 
>+- (void)getWidth:(nonnull int *)width
>+          height:(nonnull int *)height
>+       cropWidth:(nonnull int *)cropWidth
>+      cropHeight:(nonnull int *)cropHeight
>+           cropX:(nonnull int *)cropX
>+           cropY:(nonnull int *)cropY
>+         ofFrame:(nonnull RTCVideoFrame *)frame {
>+  RTCCVPixelBuffer *pixelBuffer = (RTCCVPixelBuffer *)frame.buffer;
>+  *width = CVPixelBufferGetWidth(pixelBuffer.pixelBuffer);
>+  *height = CVPixelBufferGetHeight(pixelBuffer.pixelBuffer);
>+  *cropWidth = pixelBuffer.cropWidth;
>+  *cropHeight = pixelBuffer.cropHeight;
>+  *cropX = pixelBuffer.cropX;
>+  *cropY = pixelBuffer.cropY;
>+}
>+
> - (BOOL)setupTexturesForFrame:(nonnull RTCVideoFrame *)frame {
>-  [super setupTexturesForFrame:frame];
>+  RTC_DCHECK([frame.buffer isKindOfClass:[RTCCVPixelBuffer class]]);
>+  if (![super setupTexturesForFrame:frame]) {
>+    return NO;
>+  }
>   CVPixelBufferRef pixelBuffer = ((RTCCVPixelBuffer *)frame.buffer).pixelBuffer;
> 
>   id<MTLTexture> lumaTexture = nil;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLRGBRenderer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLRGBRenderer.h
>new file mode 100644
>index 00000000000..9db422cd220
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLRGBRenderer.h
>@@ -0,0 +1,22 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#import <Foundation/Foundation.h>
>+
>+#import "RTCMTLRenderer.h"
>+
>+/** @abstract RGB/BGR renderer.
>+ *  @discussion This renderer handles both kCVPixelFormatType_32BGRA and
>+ * kCVPixelFormatType_32ARGB.
>+ */
>+NS_AVAILABLE(10_11, 9_0)
>+@interface RTCMTLRGBRenderer : RTCMTLRenderer
>+
>+@end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLRGBRenderer.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLRGBRenderer.mm
>new file mode 100644
>index 00000000000..e5a370421aa
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLRGBRenderer.mm
>@@ -0,0 +1,163 @@
>+/*
>+ *  Copyright 2018 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#import "RTCMTLRGBRenderer.h"
>+
>+#import <Metal/Metal.h>
>+#import <MetalKit/MetalKit.h>
>+
>+#import "WebRTC/RTCLogging.h"
>+#import "WebRTC/RTCVideoFrame.h"
>+#import "WebRTC/RTCVideoFrameBuffer.h"
>+
>+#import "RTCMTLRenderer+Private.h"
>+#include "rtc_base/checks.h"
>+
>+static NSString *const shaderSource = MTL_STRINGIFY(
>+    using namespace metal;
>+
>+    typedef struct {
>+      packed_float2 position;
>+      packed_float2 texcoord;
>+    } Vertex;
>+
>+    typedef struct {
>+      float4 position[[position]];
>+      float2 texcoord;
>+    } VertexIO;
>+
>+    vertex VertexIO vertexPassthrough(device Vertex * verticies[[buffer(0)]],
>+                                      uint vid[[vertex_id]]) {
>+      VertexIO out;
>+      device Vertex &v = verticies[vid];
>+      out.position = float4(float2(v.position), 0.0, 1.0);
>+      out.texcoord = v.texcoord;
>+      return out;
>+    }
>+
>+    fragment half4 fragmentColorConversion(
>+        VertexIO in[[stage_in]], texture2d<half, access::sample> texture[[texture(0)]],
>+                                           constant bool &isARGB[[buffer(0)]]) {
>+      constexpr sampler s(address::clamp_to_edge, filter::linear);
>+
>+      half4 out = texture.sample(s, in.texcoord);
>+      if (isARGB) {
>+        out = half4(out.g, out.b, out.a, out.r);
>+      }
>+
>+      return out;
>+    });
>+
>+@implementation RTCMTLRGBRenderer {
>+  // Textures.
>+  CVMetalTextureCacheRef _textureCache;
>+  id<MTLTexture> _texture;
>+
>+  // Uniforms.
>+  id<MTLBuffer> _uniformsBuffer;
>+}
>+
>+- (BOOL)addRenderingDestination:(__kindof MTKView *)view {
>+  if ([super addRenderingDestination:view]) {
>+    return [self initializeTextureCache];
>+  }
>+  return NO;
>+}
>+
>+- (BOOL)initializeTextureCache {
>+  CVReturn status = CVMetalTextureCacheCreate(kCFAllocatorDefault, nil, [self currentMetalDevice],
>+                                              nil, &_textureCache);
>+  if (status != kCVReturnSuccess) {
>+    RTCLogError(@"Metal: Failed to initialize metal texture cache. Return status is %d", status);
>+    return NO;
>+  }
>+
>+  return YES;
>+}
>+
>+- (NSString *)shaderSource {
>+  return shaderSource;
>+}
>+
>+- (void)getWidth:(nonnull int *)width
>+          height:(nonnull int *)height
>+       cropWidth:(nonnull int *)cropWidth
>+      cropHeight:(nonnull int *)cropHeight
>+           cropX:(nonnull int *)cropX
>+           cropY:(nonnull int *)cropY
>+         ofFrame:(nonnull RTCVideoFrame *)frame {
>+  RTCCVPixelBuffer *pixelBuffer = (RTCCVPixelBuffer *)frame.buffer;
>+  *width = CVPixelBufferGetWidth(pixelBuffer.pixelBuffer);
>+  *height = CVPixelBufferGetHeight(pixelBuffer.pixelBuffer);
>+  *cropWidth = pixelBuffer.cropWidth;
>+  *cropHeight = pixelBuffer.cropHeight;
>+  *cropX = pixelBuffer.cropX;
>+  *cropY = pixelBuffer.cropY;
>+}
>+
>+- (BOOL)setupTexturesForFrame:(nonnull RTCVideoFrame *)frame {
>+  RTC_DCHECK([frame.buffer isKindOfClass:[RTCCVPixelBuffer class]]);
>+  if (![super setupTexturesForFrame:frame]) {
>+    return NO;
>+  }
>+  CVPixelBufferRef pixelBuffer = ((RTCCVPixelBuffer *)frame.buffer).pixelBuffer;
>+
>+  id<MTLTexture> gpuTexture = nil;
>+  CVMetalTextureRef textureOut = nullptr;
>+  bool isARGB;
>+
>+  int width = CVPixelBufferGetWidth(pixelBuffer);
>+  int height = CVPixelBufferGetHeight(pixelBuffer);
>+  OSType pixelFormat = CVPixelBufferGetPixelFormatType(pixelBuffer);
>+
>+  MTLPixelFormat mtlPixelFormat;
>+  if (pixelFormat == kCVPixelFormatType_32BGRA) {
>+    mtlPixelFormat = MTLPixelFormatBGRA8Unorm;
>+    isARGB = false;
>+  } else if (pixelFormat == kCVPixelFormatType_32ARGB) {
>+    mtlPixelFormat = MTLPixelFormatRGBA8Unorm;
>+    isARGB = true;
>+  } else {
>+    RTC_NOTREACHED();
>+    return NO;
>+  }
>+
>+  CVReturn result = CVMetalTextureCacheCreateTextureFromImage(
>+                kCFAllocatorDefault, _textureCache, pixelBuffer, nil, mtlPixelFormat,
>+                width, height, 0, &textureOut);
>+  if (result == kCVReturnSuccess) {
>+    gpuTexture = CVMetalTextureGetTexture(textureOut);
>+  }
>+  CVBufferRelease(textureOut);
>+
>+  if (gpuTexture != nil) {
>+    _texture = gpuTexture;
>+    _uniformsBuffer =
>+        [[self currentMetalDevice] newBufferWithBytes:&isARGB
>+                                               length:sizeof(isARGB)
>+                                              options:MTLResourceCPUCacheModeDefaultCache];
>+    return YES;
>+  }
>+
>+  return NO;
>+}
>+
>+- (void)uploadTexturesToRenderEncoder:(id<MTLRenderCommandEncoder>)renderEncoder {
>+  [renderEncoder setFragmentTexture:_texture atIndex:0];
>+  [renderEncoder setFragmentBuffer:_uniformsBuffer offset:0 atIndex:0];
>+}
>+
>+- (void)dealloc {
>+  if (_textureCache) {
>+    CFRelease(_textureCache);
>+  }
>+}
>+
>+@end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLRenderer+Private.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLRenderer+Private.h
>index dfeb87b5fe5..e50ae77dde5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLRenderer+Private.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLRenderer+Private.h
>@@ -11,11 +11,20 @@
> #import <Metal/Metal.h>
> #import "RTCMTLRenderer.h"
> 
>+#define MTL_STRINGIFY(s) @ #s
>+
> NS_ASSUME_NONNULL_BEGIN
> @interface RTCMTLRenderer (Private)
> - (nullable id<MTLDevice>)currentMetalDevice;
> - (NSString *)shaderSource;
> - (BOOL)setupTexturesForFrame:(nonnull RTCVideoFrame *)frame;
> - (void)uploadTexturesToRenderEncoder:(id<MTLRenderCommandEncoder>)renderEncoder;
>+- (void)getWidth:(nonnull int *)width
>+          height:(nonnull int *)height
>+       cropWidth:(nonnull int *)cropWidth
>+      cropHeight:(nonnull int *)cropHeight
>+           cropX:(nonnull int *)cropX
>+           cropY:(nonnull int *)cropY
>+         ofFrame:(nonnull RTCVideoFrame *)frame;
> @end
> NS_ASSUME_NONNULL_END
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLRenderer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLRenderer.h
>index 110e2124656..f4a293b5c0c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLRenderer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLRenderer.h
>@@ -21,8 +21,7 @@ NS_ASSUME_NONNULL_BEGIN
> /**
>  * Protocol defining ability to render RTCVideoFrame in Metal enabled views.
>  */
>-__attribute__((objc_runtime_name("WK_RTCMTLRenderer")))
>-@protocol RTCMTLRenderer<NSObject>
>+@protocol RTCMTLRenderer <NSObject>
> 
> /**
>  * Method to be implemented to perform actual rendering of the provided frame.
>@@ -47,11 +46,16 @@ __attribute__((objc_runtime_name("WK_RTCMTLRenderer")))
> @end
> 
> /**
>- * Implementation of RTCMTLRenderer protocol for rendering native nv12 video frames.
>+ * Implementation of RTCMTLRenderer protocol.
>  */
> NS_AVAILABLE(10_11, 9_0)
>-__attribute__((objc_runtime_name("WK_RTCMTLRenderer")))
>-@interface RTCMTLRenderer : NSObject<RTCMTLRenderer>
>+@interface RTCMTLRenderer : NSObject <RTCMTLRenderer>
>+
>+/** @abstract   A wrapped RTCVideoRotation, or nil.
>+    @discussion When not nil, the rotation of the actual frame is ignored when rendering.
>+ */
>+@property(atomic, nullable) NSValue *rotationOverride;
>+
> @end
> 
> NS_ASSUME_NONNULL_END
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLRenderer.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLRenderer.mm
>index 7fcba49a312..5363aaeda21 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLRenderer.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLRenderer.mm
>@@ -15,6 +15,7 @@
> 
> #import "WebRTC/RTCLogging.h"
> #import "WebRTC/RTCVideoFrame.h"
>+#import "WebRTC/RTCVideoFrameBuffer.h"
> 
> #include "api/video/video_rotation.h"
> #include "rtc_base/checks.h"
>@@ -28,31 +29,57 @@ static NSString *const commandBufferLabel = @"RTCCommandBuffer";
> static NSString *const renderEncoderLabel = @"RTCEncoder";
> static NSString *const renderEncoderDebugGroup = @"RTCDrawFrame";
> 
>-static const float cubeVertexData[64] = {
>-    -1.0, -1.0, 0.0, 1.0, 1.0, -1.0, 1.0, 1.0, -1.0, 1.0, 0.0, 0.0, 1.0, 1.0, 1.0, 0.0,
>-
>-    // rotation = 90, offset = 16.
>-    -1.0, -1.0, 1.0, 1.0, 1.0, -1.0, 1.0, 0.0, -1.0, 1.0, 0.0, 1.0, 1.0, 1.0, 0.0, 0.0,
>-
>-    // rotation = 180, offset = 32.
>-    -1.0, -1.0, 1.0, 0.0, 1.0, -1.0, 0.0, 0.0, -1.0, 1.0, 1.0, 1.0, 1.0, 1.0, 0.0, 1.0,
>-
>-    // rotation = 270, offset = 48.
>-    -1.0, -1.0, 0.0, 0.0, 1.0, -1.0, 0.0, 1.0, -1.0, 1.0, 1.0, 0.0, 1.0, 1.0, 1.0, 1.0,
>-};
>-
>-static inline int offsetForRotation(RTCVideoRotation rotation) {
>+// Computes the texture coordinates given rotation and cropping.
>+static inline void getCubeVertexData(int cropX,
>+                                     int cropY,
>+                                     int cropWidth,
>+                                     int cropHeight,
>+                                     size_t frameWidth,
>+                                     size_t frameHeight,
>+                                     RTCVideoRotation rotation,
>+                                     float *buffer) {
>+  // The computed values are the adjusted texture coordinates, in [0..1].
>+  // For the left and top, 0.0 means no cropping and e.g. 0.2 means we're skipping 20% of the
>+  // left/top edge.
>+  // For the right and bottom, 1.0 means no cropping and e.g. 0.8 means we're skipping 20% of the
>+  // right/bottom edge (i.e. render up to 80% of the width/height).
>+  float cropLeft = cropX / (float)frameWidth;
>+  float cropRight = (cropX + cropWidth) / (float)frameWidth;
>+  float cropTop = cropY / (float)frameHeight;
>+  float cropBottom = (cropY + cropHeight) / (float)frameHeight;
>+
>+  // These arrays map the view coordinates to texture coordinates, taking cropping and rotation
>+  // into account. The first two columns are view coordinates, the last two are texture coordinates.
>   switch (rotation) {
>-    case RTCVideoRotation_0:
>-      return 0;
>-    case RTCVideoRotation_90:
>-      return 16;
>-    case RTCVideoRotation_180:
>-      return 32;
>-    case RTCVideoRotation_270:
>-      return 48;
>+    case RTCVideoRotation_0: {
>+      float values[16] = {-1.0, -1.0, cropLeft, cropBottom,
>+                           1.0, -1.0, cropRight, cropBottom,
>+                          -1.0,  1.0, cropLeft, cropTop,
>+                           1.0,  1.0, cropRight, cropTop};
>+      memcpy(buffer, &values, sizeof(values));
>+    } break;
>+    case RTCVideoRotation_90: {
>+      float values[16] = {-1.0, -1.0, cropRight, cropBottom,
>+                           1.0, -1.0, cropRight, cropTop,
>+                          -1.0,  1.0, cropLeft, cropBottom,
>+                           1.0,  1.0, cropLeft, cropTop};
>+      memcpy(buffer, &values, sizeof(values));
>+    } break;
>+    case RTCVideoRotation_180: {
>+      float values[16] = {-1.0, -1.0, cropRight, cropTop,
>+                           1.0, -1.0, cropLeft, cropTop,
>+                          -1.0,  1.0, cropRight, cropBottom,
>+                           1.0,  1.0, cropLeft, cropBottom};
>+      memcpy(buffer, &values, sizeof(values));
>+    } break;
>+    case RTCVideoRotation_270: {
>+      float values[16] = {-1.0, -1.0, cropLeft, cropTop,
>+                           1.0, -1.0, cropLeft, cropBottom,
>+                          -1.0, 1.0, cropRight, cropTop,
>+                           1.0, 1.0, cropRight, cropBottom};
>+      memcpy(buffer, &values, sizeof(values));
>+    } break;
>   }
>-  return 0;
> }
> 
> // The max number of command buffers in flight (submitted to GPU).
>@@ -75,14 +102,20 @@ @implementation RTCMTLRenderer {
>   // Buffers.
>   id<MTLBuffer> _vertexBuffer;
> 
>-  // RTC Frame parameters.
>-  int _offset;
>+  // Values affecting the vertex buffer. Stored for comparison to avoid unnecessary recreation.
>+  int _oldFrameWidth;
>+  int _oldFrameHeight;
>+  int _oldCropWidth;
>+  int _oldCropHeight;
>+  int _oldCropX;
>+  int _oldCropY;
>+  RTCVideoRotation _oldRotation;
> }
> 
>+@synthesize rotationOverride = _rotationOverride;
>+
> - (instancetype)init {
>   if (self = [super init]) {
>-    // _offset of 0 is equal to rotation of 0.
>-    _offset = 0;
>     _inflight_semaphore = dispatch_semaphore_create(kMaxInflightBuffers);
>   }
> 
>@@ -98,9 +131,17 @@ - (BOOL)addRenderingDestination:(__kindof MTKView *)view {
> - (BOOL)setupWithView:(__kindof MTKView *)view {
>   BOOL success = NO;
>   if ([self setupMetal]) {
>-    [self setupView:view];
>+    _view = view;
>+    view.device = _device;
>+    view.preferredFramesPerSecond = 30;
>+    view.autoResizeDrawable = NO;
>+
>     [self loadAssets];
>-    [self setupBuffers];
>+
>+    float vertexBufferArray[16] = {0};
>+    _vertexBuffer = [_device newBufferWithBytes:vertexBufferArray
>+                                         length:sizeof(vertexBufferArray)
>+                                        options:MTLResourceCPUCacheModeWriteCombined];
>     success = YES;
>   }
>   return success;
>@@ -120,8 +161,63 @@ - (void)uploadTexturesToRenderEncoder:(id<MTLRenderCommandEncoder>)renderEncoder
>   RTC_NOTREACHED() << "Virtual method not implemented in subclass.";
> }
> 
>+- (void)getWidth:(int *)width
>+          height:(int *)height
>+       cropWidth:(int *)cropWidth
>+      cropHeight:(int *)cropHeight
>+           cropX:(int *)cropX
>+           cropY:(int *)cropY
>+         ofFrame:(nonnull RTCVideoFrame *)frame {
>+  RTC_NOTREACHED() << "Virtual method not implemented in subclass.";
>+}
>+
> - (BOOL)setupTexturesForFrame:(nonnull RTCVideoFrame *)frame {
>-  _offset = offsetForRotation(frame.rotation);
>+  // Apply rotation override if set.
>+  RTCVideoRotation rotation;
>+  NSValue *rotationOverride = self.rotationOverride;
>+  if (rotationOverride) {
>+#if defined(__IPHONE_11_0) && (__IPHONE_OS_VERSION_MAX_ALLOWED >= __IPHONE_11_0)
>+    if (@available(iOS 11, *)) {
>+      [rotationOverride getValue:&rotation size:sizeof(rotation)];
>+    } else
>+#endif
>+    {
>+      [rotationOverride getValue:&rotation];
>+    }
>+  } else {
>+    rotation = frame.rotation;
>+  }
>+
>+  int frameWidth, frameHeight, cropWidth, cropHeight, cropX, cropY;
>+  [self getWidth:&frameWidth
>+          height:&frameHeight
>+       cropWidth:&cropWidth
>+      cropHeight:&cropHeight
>+           cropX:&cropX
>+           cropY:&cropY
>+         ofFrame:frame];
>+
>+  // Recompute the texture cropping and recreate vertexBuffer if necessary.
>+  if (cropX != _oldCropX || cropY != _oldCropY || cropWidth != _oldCropWidth ||
>+      cropHeight != _oldCropHeight || rotation != _oldRotation || frameWidth != _oldFrameWidth ||
>+      frameHeight != _oldFrameHeight) {
>+    getCubeVertexData(cropX,
>+                      cropY,
>+                      cropWidth,
>+                      cropHeight,
>+                      frameWidth,
>+                      frameHeight,
>+                      rotation,
>+                      (float *)_vertexBuffer.contents);
>+    _oldCropX = cropX;
>+    _oldCropY = cropY;
>+    _oldCropWidth = cropWidth;
>+    _oldCropHeight = cropHeight;
>+    _oldRotation = rotation;
>+    _oldFrameWidth = frameWidth;
>+    _oldFrameHeight = frameHeight;
>+  }
>+
>   return YES;
> }
> 
>@@ -139,9 +235,10 @@ - (BOOL)setupMetal {
> 
>   // Load metal library from source.
>   NSError *libraryError = nil;
>+  NSString *shaderSource = [self shaderSource];
> 
>   id<MTLLibrary> sourceLibrary =
>-      [_device newLibraryWithSource:[self shaderSource] options:NULL error:&libraryError];
>+      [_device newLibraryWithSource:shaderSource options:NULL error:&libraryError];
> 
>   if (libraryError) {
>     RTCLogError(@"Metal: Library with source failed\n%@", libraryError);
>@@ -157,16 +254,6 @@ - (BOOL)setupMetal {
>   return YES;
> }
> 
>-- (void)setupView:(__kindof MTKView *)view {
>-  view.device = _device;
>-
>-  view.preferredFramesPerSecond = 30;
>-  view.autoResizeDrawable = NO;
>-
>-  // We need to keep reference to the view as it's needed down the rendering pipeline.
>-  _view = view;
>-}
>-
> - (void)loadAssets {
>   id<MTLFunction> vertexFunction = [_defaultLibrary newFunctionWithName:vertexFunctionName];
>   id<MTLFunction> fragmentFunction = [_defaultLibrary newFunctionWithName:fragmentFunctionName];
>@@ -185,12 +272,6 @@ - (void)loadAssets {
>   }
> }
> 
>-- (void)setupBuffers {
>-  _vertexBuffer = [_device newBufferWithBytes:cubeVertexData
>-                                       length:sizeof(cubeVertexData)
>-                                      options:MTLResourceOptionCPUCacheModeDefault];
>-}
>-
> - (void)render {
>   // Wait until the inflight (curently sent to GPU) command buffer
>   // has completed the GPU work.
>@@ -214,7 +295,7 @@ - (void)render {
>     // Set context state.
>     [renderEncoder pushDebugGroup:renderEncoderDebugGroup];
>     [renderEncoder setRenderPipelineState:_pipelineState];
>-    [renderEncoder setVertexBuffer:_vertexBuffer offset:_offset * sizeof(float) atIndex:0];
>+    [renderEncoder setVertexBuffer:_vertexBuffer offset:0 atIndex:0];
>     [self uploadTexturesToRenderEncoder:renderEncoder];
> 
>     [renderEncoder drawPrimitives:MTLPrimitiveTypeTriangleStrip
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLVideoView.m b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLVideoView.m
>index 48c3f33a0c0..e0fd5241771 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLVideoView.m
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLVideoView.m
>@@ -19,26 +19,36 @@
> 
> #import "RTCMTLI420Renderer.h"
> #import "RTCMTLNV12Renderer.h"
>+#import "RTCMTLRGBRenderer.h"
> 
> // To avoid unreconized symbol linker errors, we're taking advantage of the objc runtime.
> // Linking errors occur when compiling for architectures that don't support Metal.
> #define MTKViewClass NSClassFromString(@"MTKView")
> #define RTCMTLNV12RendererClass NSClassFromString(@"RTCMTLNV12Renderer")
> #define RTCMTLI420RendererClass NSClassFromString(@"RTCMTLI420Renderer")
>+#define RTCMTLRGBRendererClass NSClassFromString(@"RTCMTLRGBRenderer")
> 
> @interface RTCMTLVideoView () <MTKViewDelegate>
>-@property(nonatomic, strong) RTCMTLI420Renderer *rendererI420;
>-@property(nonatomic, strong) RTCMTLNV12Renderer *rendererNV12;
>-@property(nonatomic, strong) MTKView *metalView;
>-@property(atomic, strong) RTCVideoFrame *videoFrame;
>+@property(nonatomic) RTCMTLI420Renderer *rendererI420;
>+@property(nonatomic) RTCMTLNV12Renderer *rendererNV12;
>+@property(nonatomic) RTCMTLRGBRenderer *rendererRGB;
>+@property(nonatomic) MTKView *metalView;
>+@property(atomic) RTCVideoFrame *videoFrame;
>+@property(nonatomic) CGSize videoFrameSize;
>+@property(nonatomic) int64_t lastFrameTimeNs;
> @end
> 
> @implementation RTCMTLVideoView
> 
>+@synthesize delegate = _delegate;
> @synthesize rendererI420 = _rendererI420;
> @synthesize rendererNV12 = _rendererNV12;
>+@synthesize rendererRGB = _rendererRGB;
> @synthesize metalView = _metalView;
> @synthesize videoFrame = _videoFrame;
>+@synthesize videoFrameSize = _videoFrameSize;
>+@synthesize lastFrameTimeNs = _lastFrameTimeNs;
>+@synthesize rotationOverride = _rotationOverride;
> 
> - (instancetype)initWithFrame:(CGRect)frameRect {
>   self = [super initWithFrame:frameRect];
>@@ -56,19 +66,34 @@ - (instancetype)initWithCoder:(NSCoder *)aCoder {
>   return self;
> }
> 
>+- (BOOL)isEnabled {
>+  return !self.metalView.paused;
>+}
>+
>+- (void)setEnabled:(BOOL)enabled {
>+  self.metalView.paused = !enabled;
>+}
>+
>+- (UIViewContentMode)videoContentMode {
>+  return self.metalView.contentMode;
>+}
>+
>+- (void)setVideoContentMode:(UIViewContentMode)mode {
>+  self.metalView.contentMode = mode;
>+}
>+
> #pragma mark - Private
> 
> + (BOOL)isMetalAvailable {
> #if defined(RTC_SUPPORTS_METAL)
>-  return YES;
>+  return MTLCreateSystemDefaultDevice() != nil;
> #else
>   return NO;
> #endif
> }
> 
> + (MTKView *)createMetalView:(CGRect)frame {
>-  MTKView *view = [[MTKViewClass alloc] initWithFrame:frame];
>-  return view;
>+  return [[MTKViewClass alloc] initWithFrame:frame];
> }
> 
> + (RTCMTLNV12Renderer *)createNV12Renderer {
>@@ -79,70 +104,149 @@ + (RTCMTLI420Renderer *)createI420Renderer {
>   return [[RTCMTLI420RendererClass alloc] init];
> }
> 
>++ (RTCMTLRGBRenderer *)createRGBRenderer {
>+  return [[RTCMTLRGBRenderer alloc] init];
>+}
>+
> - (void)configure {
>   NSAssert([RTCMTLVideoView isMetalAvailable], @"Metal not availiable on this device");
> 
>-  _metalView = [RTCMTLVideoView createMetalView:self.bounds];
>-  [self configureMetalView];
>+  self.metalView = [RTCMTLVideoView createMetalView:self.bounds];
>+  self.metalView.delegate = self;
>+  self.metalView.contentMode = UIViewContentModeScaleAspectFill;
>+  [self addSubview:self.metalView];
>+  self.videoFrameSize = CGSizeZero;
> }
> 
>-- (void)configureMetalView {
>-  if (_metalView) {
>-    _metalView.delegate = self;
>-    [self addSubview:_metalView];
>-    _metalView.contentMode = UIViewContentModeScaleAspectFit;
>-  }
>-}
>-
>-#pragma mark - Private
>-
> - (void)layoutSubviews {
>   [super layoutSubviews];
>-  _metalView.frame = self.bounds;
>+
>+  CGRect bounds = self.bounds;
>+  self.metalView.frame = bounds;
>+  if (!CGSizeEqualToSize(self.videoFrameSize, CGSizeZero)) {
>+    self.metalView.drawableSize = [self drawableSize];
>+  } else {
>+    self.metalView.drawableSize = bounds.size;
>+  }
> }
> 
> #pragma mark - MTKViewDelegate methods
> 
> - (void)drawInMTKView:(nonnull MTKView *)view {
>   NSAssert(view == self.metalView, @"Receiving draw callbacks from foreign instance.");
>-  if (!self.videoFrame) {
>+  RTCVideoFrame *videoFrame = self.videoFrame;
>+  // Skip rendering if we've already rendered this frame.
>+  if (!videoFrame || videoFrame.timeStampNs == self.lastFrameTimeNs) {
>     return;
>   }
> 
>-  id<RTCMTLRenderer> renderer = nil;
>-  if ([self.videoFrame.buffer isKindOfClass:[RTCCVPixelBuffer class]]) {
>-    if (!self.rendererNV12) {
>-      self.rendererNV12 = [RTCMTLVideoView createNV12Renderer];
>-      if (![self.rendererNV12 addRenderingDestination:self.metalView]) {
>-        self.rendererNV12 = nil;
>-        RTCLogError(@"Failed to create NV12 renderer");
>+  RTCMTLRenderer *renderer;
>+  if ([videoFrame.buffer isKindOfClass:[RTCCVPixelBuffer class]]) {
>+    RTCCVPixelBuffer *buffer = (RTCCVPixelBuffer*)videoFrame.buffer;
>+    const OSType pixelFormat = CVPixelBufferGetPixelFormatType(buffer.pixelBuffer);
>+    if (pixelFormat == kCVPixelFormatType_32BGRA || pixelFormat == kCVPixelFormatType_32ARGB) {
>+      if (!self.rendererRGB) {
>+        self.rendererRGB = [RTCMTLVideoView createRGBRenderer];
>+        if (![self.rendererRGB addRenderingDestination:self.metalView]) {
>+          self.rendererRGB = nil;
>+          RTCLogError(@"Failed to create RGB renderer");
>+          return;
>+        }
>       }
>+      renderer = self.rendererRGB;
>+    } else {
>+      if (!self.rendererNV12) {
>+        self.rendererNV12 = [RTCMTLVideoView createNV12Renderer];
>+        if (![self.rendererNV12 addRenderingDestination:self.metalView]) {
>+          self.rendererNV12 = nil;
>+          RTCLogError(@"Failed to create NV12 renderer");
>+          return;
>+        }
>+      }
>+      renderer = self.rendererNV12;
>     }
>-    renderer = self.rendererNV12;
>   } else {
>     if (!self.rendererI420) {
>       self.rendererI420 = [RTCMTLVideoView createI420Renderer];
>       if (![self.rendererI420 addRenderingDestination:self.metalView]) {
>         self.rendererI420 = nil;
>         RTCLogError(@"Failed to create I420 renderer");
>+        return;
>       }
>     }
>     renderer = self.rendererI420;
>   }
> 
>-  [renderer drawFrame:self.videoFrame];
>+  renderer.rotationOverride = self.rotationOverride;
>+
>+  [renderer drawFrame:videoFrame];
>+  self.lastFrameTimeNs = videoFrame.timeStampNs;
> }
> 
> - (void)mtkView:(MTKView *)view drawableSizeWillChange:(CGSize)size {
> }
> 
>+#pragma mark -
>+
>+- (void)setRotationOverride:(NSValue *)rotationOverride {
>+  _rotationOverride = rotationOverride;
>+
>+  self.metalView.drawableSize = [self drawableSize];
>+  [self setNeedsLayout];
>+}
>+
>+- (RTCVideoRotation)frameRotation {
>+  if (self.rotationOverride) {
>+    RTCVideoRotation rotation;
>+    if (@available(iOS 11, *)) {
>+      [self.rotationOverride getValue:&rotation size:sizeof(rotation)];
>+    } else {
>+      [self.rotationOverride getValue:&rotation];
>+    }
>+    return rotation;
>+  }
>+
>+  return self.videoFrame.rotation;
>+}
>+
>+- (CGSize)drawableSize {
>+  // Flip width/height if the rotations are not the same.
>+  CGSize videoFrameSize = self.videoFrameSize;
>+  RTCVideoRotation frameRotation = [self frameRotation];
>+
>+  BOOL useLandscape =
>+      (frameRotation == RTCVideoRotation_0) || (frameRotation == RTCVideoRotation_180);
>+  BOOL sizeIsLandscape = (self.videoFrame.rotation == RTCVideoRotation_0) ||
>+      (self.videoFrame.rotation == RTCVideoRotation_180);
>+
>+  if (useLandscape == sizeIsLandscape) {
>+    return videoFrameSize;
>+  } else {
>+    return CGSizeMake(videoFrameSize.height, videoFrameSize.width);
>+  }
>+}
>+
> #pragma mark - RTCVideoRenderer
> 
> - (void)setSize:(CGSize)size {
>-  self.metalView.drawableSize = size;
>+  __weak RTCMTLVideoView *weakSelf = self;
>+  dispatch_async(dispatch_get_main_queue(), ^{
>+    RTCMTLVideoView *strongSelf = weakSelf;
>+
>+    strongSelf.videoFrameSize = size;
>+    CGSize drawableSize = [strongSelf drawableSize];
>+
>+    strongSelf.metalView.drawableSize = drawableSize;
>+    [strongSelf setNeedsLayout];
>+    [strongSelf.delegate videoView:self didChangeVideoSize:size];
>+  });
> }
> 
> - (void)renderFrame:(nullable RTCVideoFrame *)frame {
>+  if (!self.isEnabled) {
>+    return;
>+  }
>+
>   if (frame == nil) {
>     RTCLogInfo(@"Incoming frame is nil. Exiting render callback.");
>     return;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/OWNERS
>deleted file mode 100644
>index a32f041ac89..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/OWNERS
>+++ /dev/null
>@@ -1 +0,0 @@
>-deadbeef@webrtc.org
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCAVFoundationVideoSource+Private.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCAVFoundationVideoSource+Private.h
>deleted file mode 100644
>index 7a4de08240b..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCAVFoundationVideoSource+Private.h
>+++ /dev/null
>@@ -1,27 +0,0 @@
>-/*
>- *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#import "WebRTC/RTCAVFoundationVideoSource.h"
>-
>-#include "avfoundationvideocapturer.h"
>-
>-NS_ASSUME_NONNULL_BEGIN
>-
>-@interface RTCAVFoundationVideoSource ()
>-
>-@property(nonatomic, readonly) webrtc::AVFoundationVideoCapturer *capturer;
>-
>-/** Initialize an RTCAVFoundationVideoSource with constraints. */
>-- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
>-                    constraints:(nullable RTCMediaConstraints *)constraints;
>-
>-@end
>-
>-NS_ASSUME_NONNULL_END
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCAVFoundationVideoSource.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCAVFoundationVideoSource.mm
>deleted file mode 100644
>index b004191f8b3..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCAVFoundationVideoSource.mm
>+++ /dev/null
>@@ -1,62 +0,0 @@
>-/*
>- *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#import "RTCAVFoundationVideoSource+Private.h"
>-
>-#import "RTCMediaConstraints+Private.h"
>-#import "RTCPeerConnectionFactory+Private.h"
>-#import "RTCVideoSource+Private.h"
>-
>-@implementation RTCAVFoundationVideoSource {
>-  webrtc::AVFoundationVideoCapturer *_capturer;
>-}
>-
>-- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
>-                    constraints:(RTCMediaConstraints *)constraints {
>-  NSParameterAssert(factory);
>-  // We pass ownership of the capturer to the source, but since we own
>-  // the source, it should be ok to keep a raw pointer to the
>-  // capturer.
>-  _capturer = new webrtc::AVFoundationVideoCapturer();
>-  rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> source =
>-      factory.nativeFactory->CreateVideoSource(
>-          std::unique_ptr<cricket::VideoCapturer>(_capturer),
>-          constraints.nativeConstraints.get());
>-
>-  return [super initWithNativeVideoSource:source];
>-}
>-
>-- (void)adaptOutputFormatToWidth:(int)width
>-                          height:(int)height
>-                             fps:(int)fps {
>-  self.capturer->AdaptOutputFormat(width, height, fps);
>-}
>-
>-- (BOOL)canUseBackCamera {
>-  return self.capturer->CanUseBackCamera();
>-}
>-
>-- (BOOL)useBackCamera {
>-  return self.capturer->GetUseBackCamera();
>-}
>-
>-- (void)setUseBackCamera:(BOOL)useBackCamera {
>-  self.capturer->SetUseBackCamera(useBackCamera);
>-}
>-
>-- (AVCaptureSession *)captureSession {
>-  return self.capturer->GetCaptureSession();
>-}
>-
>-- (webrtc::AVFoundationVideoCapturer *)capturer {
>-  return _capturer;
>-}
>-
>-@end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCAudioSource+Private.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCAudioSource+Private.h
>index 43aff02691f..63fff688926 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCAudioSource+Private.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCAudioSource+Private.h
>@@ -21,12 +21,12 @@
> @property(nonatomic, readonly) rtc::scoped_refptr<webrtc::AudioSourceInterface> nativeAudioSource;
> 
> /** Initialize an RTCAudioSource from a native AudioSourceInterface. */
>-- (instancetype)initWithNativeAudioSource:
>-    (rtc::scoped_refptr<webrtc::AudioSourceInterface>)nativeAudioSource
>+- (instancetype)initWithFactory:(RTCPeerConnectionFactory*)factory
>+              nativeAudioSource:(rtc::scoped_refptr<webrtc::AudioSourceInterface>)nativeAudioSource
>     NS_DESIGNATED_INITIALIZER;
> 
>-- (instancetype)initWithNativeMediaSource:
>-    (rtc::scoped_refptr<webrtc::MediaSourceInterface>)nativeMediaSource
>-                                     type:(RTCMediaSourceType)type NS_UNAVAILABLE;
>+- (instancetype)initWithFactory:(RTCPeerConnectionFactory*)factory
>+              nativeMediaSource:(rtc::scoped_refptr<webrtc::MediaSourceInterface>)nativeMediaSource
>+                           type:(RTCMediaSourceType)type NS_UNAVAILABLE;
> 
> @end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCAudioSource.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCAudioSource.mm
>index 310171d76b5..a6822f67024 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCAudioSource.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCAudioSource.mm
>@@ -18,19 +18,23 @@ @implementation RTCAudioSource {
> @synthesize volume = _volume;
> @synthesize nativeAudioSource = _nativeAudioSource;
> 
>-- (instancetype)initWithNativeAudioSource:
>-    (rtc::scoped_refptr<webrtc::AudioSourceInterface>)nativeAudioSource {
>+- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
>+              nativeAudioSource:
>+                  (rtc::scoped_refptr<webrtc::AudioSourceInterface>)nativeAudioSource {
>+  RTC_DCHECK(factory);
>   RTC_DCHECK(nativeAudioSource);
>-  if (self = [super initWithNativeMediaSource:nativeAudioSource
>-                                         type:RTCMediaSourceTypeAudio]) {
>+
>+  if (self = [super initWithFactory:factory
>+                  nativeMediaSource:nativeAudioSource
>+                               type:RTCMediaSourceTypeAudio]) {
>     _nativeAudioSource = nativeAudioSource;
>   }
>   return self;
> }
> 
>-- (instancetype)initWithNativeMediaSource:
>-    (rtc::scoped_refptr<webrtc::MediaSourceInterface>)nativeMediaSource
>-                                     type:(RTCMediaSourceType)type {
>+- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
>+              nativeMediaSource:(rtc::scoped_refptr<webrtc::MediaSourceInterface>)nativeMediaSource
>+                           type:(RTCMediaSourceType)type {
>   RTC_NOTREACHED();
>   return nil;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCAudioTrack+Private.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCAudioTrack+Private.h
>index 176d03faaa5..7a2f2c82cdc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCAudioTrack+Private.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCAudioTrack+Private.h
>@@ -18,8 +18,7 @@ NS_ASSUME_NONNULL_BEGIN
> @interface RTCAudioTrack ()
> 
> /** AudioTrackInterface created or passed in at construction. */
>-@property(nonatomic, readonly)
>-    rtc::scoped_refptr<webrtc::AudioTrackInterface> nativeAudioTrack;
>+@property(nonatomic, readonly) rtc::scoped_refptr<webrtc::AudioTrackInterface> nativeAudioTrack;
> 
> /** Initialize an RTCAudioTrack with an id. */
> - (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCAudioTrack.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCAudioTrack.mm
>index e26088f59d2..73de401eeb3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCAudioTrack.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCAudioTrack.mm
>@@ -31,18 +31,19 @@ - (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
>   std::string nativeId = [NSString stdStringForString:trackId];
>   rtc::scoped_refptr<webrtc::AudioTrackInterface> track =
>       factory.nativeFactory->CreateAudioTrack(nativeId, source.nativeAudioSource);
>-  if (self = [self initWithNativeTrack:track type:RTCMediaStreamTrackTypeAudio]) {
>+  if (self = [self initWithFactory:factory nativeTrack:track type:RTCMediaStreamTrackTypeAudio]) {
>     _source = source;
>   }
>   return self;
> }
> 
>-- (instancetype)initWithNativeTrack:
>-    (rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack
>-                               type:(RTCMediaStreamTrackType)type {
>+- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
>+                    nativeTrack:(rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack
>+                           type:(RTCMediaStreamTrackType)type {
>+  NSParameterAssert(factory);
>   NSParameterAssert(nativeTrack);
>   NSParameterAssert(type == RTCMediaStreamTrackTypeAudio);
>-  return [super initWithNativeTrack:nativeTrack type:type];
>+  return [super initWithFactory:factory nativeTrack:nativeTrack type:type];
> }
> 
> 
>@@ -51,7 +52,8 @@ - (RTCAudioSource *)source {
>     rtc::scoped_refptr<webrtc::AudioSourceInterface> source =
>         self.nativeAudioTrack->GetSource();
>     if (source) {
>-      _source = [[RTCAudioSource alloc] initWithNativeAudioSource:source.get()];
>+      _source =
>+          [[RTCAudioSource alloc] initWithFactory:self.factory nativeAudioSource:source.get()];
>     }
>   }
>   return _source;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCCameraVideoCapturer.m b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCCameraVideoCapturer.m
>index 89060efa58e..8f93937c750 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCCameraVideoCapturer.m
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCCameraVideoCapturer.m
>@@ -46,13 +46,23 @@ @implementation RTCCameraVideoCapturer {
> @synthesize frameQueue = _frameQueue;
> @synthesize captureSession = _captureSession;
> 
>+- (instancetype)init {
>+  return [self initWithDelegate:nil captureSession:[[AVCaptureSession alloc] init]];
>+}
>+
> - (instancetype)initWithDelegate:(__weak id<RTCVideoCapturerDelegate>)delegate {
>+  return [self initWithDelegate:delegate captureSession:[[AVCaptureSession alloc] init]];
>+}
>+
>+// This initializer is used for testing.
>+- (instancetype)initWithDelegate:(__weak id<RTCVideoCapturerDelegate>)delegate
>+                  captureSession:(AVCaptureSession *)captureSession {
>   if (self = [super initWithDelegate:delegate]) {
>     // Create the capture session and all relevant inputs and outputs. We need
>     // to do this in init because the application may want the capture session
>     // before we start the capturer for e.g. AVCapturePreviewLayer. All objects
>     // created here are retained until dealloc and never recreated.
>-    if (![self setupCaptureSession]) {
>+    if (![self setupCaptureSession:captureSession]) {
>       return nil;
>     }
>     NSNotificationCenter *center = [NSNotificationCenter defaultCenter];
>@@ -100,7 +110,16 @@ - (void)dealloc {
> }
> 
> + (NSArray<AVCaptureDevice *> *)captureDevices {
>+#if defined(WEBRTC_IOS) && defined(__IPHONE_10_0) && \
>+    __IPHONE_OS_VERSION_MIN_REQUIRED >= __IPHONE_10_0
>+  AVCaptureDeviceDiscoverySession *session = [AVCaptureDeviceDiscoverySession
>+      discoverySessionWithDeviceTypes:@[ AVCaptureDeviceTypeBuiltInWideAngleCamera ]
>+                            mediaType:AVMediaTypeVideo
>+                             position:AVCaptureDevicePositionUnspecified];
>+  return session.devices;
>+#else
>   return [AVCaptureDevice devicesWithMediaType:AVMediaTypeVideo];
>+#endif
> }
> 
> + (NSArray<AVCaptureDeviceFormat *> *)supportedFormatsForDevice:(AVCaptureDevice *)device {
>@@ -146,6 +165,7 @@ - (void)startCaptureWithDevice:(AVCaptureDevice *)device
>                         if (completionHandler) {
>                           completionHandler(error);
>                         }
>+                        _willBeRunning = NO;
>                         return;
>                       }
>                       [self reconfigureCaptureSessionInput];
>@@ -161,7 +181,7 @@ - (void)startCaptureWithDevice:(AVCaptureDevice *)device
>                     }];
> }
> 
>-- (void)stopCaptureWithCompletionHandler:(nullable void (^)())completionHandler {
>+- (void)stopCaptureWithCompletionHandler:(nullable void (^)(void))completionHandler {
>   _willBeRunning = NO;
>   [RTCDispatcher
>       dispatchAsyncOnType:RTCDispatcherTypeCaptureSession
>@@ -372,16 +392,16 @@ - (void)handleApplicationDidBecomeActive:(NSNotification *)notification {
> - (dispatch_queue_t)frameQueue {
>   if (!_frameQueue) {
>     _frameQueue =
>-        dispatch_queue_create("org.webrtc.avfoundationvideocapturer.video", DISPATCH_QUEUE_SERIAL);
>+        dispatch_queue_create("org.webrtc.cameravideocapturer.video", DISPATCH_QUEUE_SERIAL);
>     dispatch_set_target_queue(_frameQueue,
>                               dispatch_get_global_queue(DISPATCH_QUEUE_PRIORITY_HIGH, 0));
>   }
>   return _frameQueue;
> }
> 
>-- (BOOL)setupCaptureSession {
>+- (BOOL)setupCaptureSession:(AVCaptureSession *)captureSession {
>   NSAssert(_captureSession == nil, @"Setup capture session called twice.");
>-  _captureSession = [[AVCaptureSession alloc] init];
>+  _captureSession = captureSession;
> #if defined(WEBRTC_IOS)
>   _captureSession.sessionPreset = AVCaptureSessionPresetInputPriority;
>   _captureSession.usesApplicationAudioSession = NO;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCCertificate.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCCertificate.mm
>new file mode 100644
>index 00000000000..34fa837bdca
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCCertificate.mm
>@@ -0,0 +1,70 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#import "WebRTC/RTCCertificate.h"
>+#import "WebRTC/RTCLogging.h"
>+
>+#include "rtc_base/logging.h"
>+#include "rtc_base/rtccertificategenerator.h"
>+#include "rtc_base/sslidentity.h"
>+
>+@implementation RTCCertificate
>+
>+@synthesize private_key = _private_key;
>+@synthesize certificate = _certificate;
>+
>+- (id)copyWithZone:(NSZone *)zone {
>+  id copy = [[[self class] alloc] initWithPrivateKey:[self.private_key copyWithZone:zone]
>+                                         certificate:[self.certificate copyWithZone:zone]];
>+  return copy;
>+}
>+
>+- (instancetype)initWithPrivateKey:(NSString *)private_key certificate:(NSString *)certificate {
>+  if (self = [super init]) {
>+    _private_key = [private_key copy];
>+    _certificate = [certificate copy];
>+  }
>+  return self;
>+}
>+
>++ (nullable RTCCertificate *)generateCertificateWithParams:(NSDictionary *)params {
>+  rtc::KeyType keyType = rtc::KT_ECDSA;
>+  NSString *keyTypeString = [params valueForKey:@"name"];
>+  if (keyTypeString && [keyTypeString isEqualToString:@"RSASSA-PKCS1-v1_5"]) {
>+    keyType = rtc::KT_RSA;
>+  }
>+
>+  NSNumber *expires = [params valueForKey:@"expires"];
>+  rtc::scoped_refptr<rtc::RTCCertificate> cc_certificate = nullptr;
>+  if (expires != nil) {
>+    uint64_t expirationTimestamp = [expires unsignedLongLongValue];
>+    cc_certificate = rtc::RTCCertificateGenerator::GenerateCertificate(rtc::KeyParams(keyType),
>+                                                                       expirationTimestamp);
>+  } else {
>+    cc_certificate =
>+        rtc::RTCCertificateGenerator::GenerateCertificate(rtc::KeyParams(keyType), absl::nullopt);
>+  }
>+  if (!cc_certificate) {
>+    RTCLogError(@"Failed to generate certificate.");
>+    return nullptr;
>+  }
>+  // grab PEMs and create an NS RTCCerticicate
>+  rtc::RTCCertificatePEM pem = cc_certificate->ToPEM();
>+  std::string pem_private_key = pem.private_key();
>+  std::string pem_certificate = pem.certificate();
>+  RTC_LOG(LS_INFO) << "CERT PEM ";
>+  RTC_LOG(LS_INFO) << pem_certificate;
>+
>+  RTCCertificate *cert = [[RTCCertificate alloc] initWithPrivateKey:@(pem_private_key.c_str())
>+                                                        certificate:@(pem_certificate.c_str())];
>+  return cert;
>+}
>+
>+@end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCConfiguration+Native.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCConfiguration+Native.h
>index f539d1ced79..64653e45f6b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCConfiguration+Native.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCConfiguration+Native.h
>@@ -18,9 +18,10 @@ NS_ASSUME_NONNULL_BEGIN
> 
> /** Optional TurnCustomizer.
>  *  With this class one can modify outgoing TURN messages.
>- *  The object passed in must remain valid until PeerConnection::Close() is called.
>+ *  The object passed in must remain valid until PeerConnection::Close() is
>+ * called.
>  */
>-@property(nonatomic, nullable) webrtc::TurnCustomizer *turnCustomizer;
>+@property(nonatomic, nullable) webrtc::TurnCustomizer* turnCustomizer;
> 
> @end
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCConfiguration+Private.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCConfiguration+Private.h
>index d9a991650b7..ad57336b704 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCConfiguration+Private.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCConfiguration+Private.h
>@@ -16,48 +16,54 @@ NS_ASSUME_NONNULL_BEGIN
> 
> @interface RTCConfiguration ()
> 
>-+ (webrtc::PeerConnectionInterface::IceTransportsType)
>-    nativeTransportsTypeForTransportPolicy:(RTCIceTransportPolicy)policy;
>++ (webrtc::PeerConnectionInterface::IceTransportsType)nativeTransportsTypeForTransportPolicy:
>+        (RTCIceTransportPolicy)policy;
> 
> + (RTCIceTransportPolicy)transportPolicyForTransportsType:
>-    (webrtc::PeerConnectionInterface::IceTransportsType)nativeType;
>+        (webrtc::PeerConnectionInterface::IceTransportsType)nativeType;
> 
> + (NSString *)stringForTransportPolicy:(RTCIceTransportPolicy)policy;
> 
> + (webrtc::PeerConnectionInterface::BundlePolicy)nativeBundlePolicyForPolicy:
>-    (RTCBundlePolicy)policy;
>+        (RTCBundlePolicy)policy;
> 
> + (RTCBundlePolicy)bundlePolicyForNativePolicy:
>-    (webrtc::PeerConnectionInterface::BundlePolicy)nativePolicy;
>+        (webrtc::PeerConnectionInterface::BundlePolicy)nativePolicy;
> 
> + (NSString *)stringForBundlePolicy:(RTCBundlePolicy)policy;
> 
> + (webrtc::PeerConnectionInterface::RtcpMuxPolicy)nativeRtcpMuxPolicyForPolicy:
>-    (RTCRtcpMuxPolicy)policy;
>+        (RTCRtcpMuxPolicy)policy;
> 
> + (RTCRtcpMuxPolicy)rtcpMuxPolicyForNativePolicy:
>-    (webrtc::PeerConnectionInterface::RtcpMuxPolicy)nativePolicy;
>+        (webrtc::PeerConnectionInterface::RtcpMuxPolicy)nativePolicy;
> 
> + (NSString *)stringForRtcpMuxPolicy:(RTCRtcpMuxPolicy)policy;
> 
>-+ (webrtc::PeerConnectionInterface::TcpCandidatePolicy)
>-    nativeTcpCandidatePolicyForPolicy:(RTCTcpCandidatePolicy)policy;
>++ (webrtc::PeerConnectionInterface::TcpCandidatePolicy)nativeTcpCandidatePolicyForPolicy:
>+        (RTCTcpCandidatePolicy)policy;
> 
> + (RTCTcpCandidatePolicy)tcpCandidatePolicyForNativePolicy:
>-    (webrtc::PeerConnectionInterface::TcpCandidatePolicy)nativePolicy;
>+        (webrtc::PeerConnectionInterface::TcpCandidatePolicy)nativePolicy;
> 
> + (NSString *)stringForTcpCandidatePolicy:(RTCTcpCandidatePolicy)policy;
> 
>-+ (webrtc::PeerConnectionInterface::CandidateNetworkPolicy)
>-    nativeCandidateNetworkPolicyForPolicy:(RTCCandidateNetworkPolicy)policy;
>++ (webrtc::PeerConnectionInterface::CandidateNetworkPolicy)nativeCandidateNetworkPolicyForPolicy:
>+        (RTCCandidateNetworkPolicy)policy;
> 
> + (RTCCandidateNetworkPolicy)candidateNetworkPolicyForNativePolicy:
>-    (webrtc::PeerConnectionInterface::CandidateNetworkPolicy)nativePolicy;
>+        (webrtc::PeerConnectionInterface::CandidateNetworkPolicy)nativePolicy;
> 
> + (NSString *)stringForCandidateNetworkPolicy:(RTCCandidateNetworkPolicy)policy;
> 
> + (rtc::KeyType)nativeEncryptionKeyTypeForKeyType:(RTCEncryptionKeyType)keyType;
> 
>++ (webrtc::SdpSemantics)nativeSdpSemanticsForSdpSemantics:(RTCSdpSemantics)sdpSemantics;
>+
>++ (RTCSdpSemantics)sdpSemanticsForNativeSdpSemantics:(webrtc::SdpSemantics)sdpSemantics;
>+
>++ (NSString *)stringForSdpSemantics:(RTCSdpSemantics)sdpSemantics;
>+
> /**
>  * RTCConfiguration struct representation of this RTCConfiguration. This is
>  * needed to pass to the underlying C++ APIs.
>@@ -65,7 +71,7 @@ NS_ASSUME_NONNULL_BEGIN
> - (nullable webrtc::PeerConnectionInterface::RTCConfiguration *)createNativeConfiguration;
> 
> - (instancetype)initWithNativeConfiguration:
>-    (const webrtc::PeerConnectionInterface::RTCConfiguration &)config NS_DESIGNATED_INITIALIZER;
>+        (const webrtc::PeerConnectionInterface::RTCConfiguration &)config NS_DESIGNATED_INITIALIZER;
> 
> @end
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCConfiguration.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCConfiguration.mm
>index faf34d63d0f..17483774fc0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCConfiguration.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCConfiguration.mm
>@@ -23,6 +23,7 @@
> @implementation RTCConfiguration
> 
> @synthesize iceServers = _iceServers;
>+@synthesize certificate = _certificate;
> @synthesize iceTransportPolicy = _iceTransportPolicy;
> @synthesize bundlePolicy = _bundlePolicy;
> @synthesize rtcpMuxPolicy = _rtcpMuxPolicy;
>@@ -30,6 +31,7 @@ @synthesize tcpCandidatePolicy = _tcpCandidatePolicy;
> @synthesize candidateNetworkPolicy = _candidateNetworkPolicy;
> @synthesize continualGatheringPolicy = _continualGatheringPolicy;
> @synthesize maxIPv6Networks = _maxIPv6Networks;
>+@synthesize disableLinkLocalNetworks = _disableLinkLocalNetworks;
> @synthesize audioJitterBufferMaxPackets = _audioJitterBufferMaxPackets;
> @synthesize audioJitterBufferFastAccelerate = _audioJitterBufferFastAccelerate;
> @synthesize iceConnectionReceivingTimeout = _iceConnectionReceivingTimeout;
>@@ -42,7 +44,9 @@ @synthesize shouldPresumeWritableWhenFullyRelayed =
>     _shouldPresumeWritableWhenFullyRelayed;
> @synthesize iceCheckMinInterval = _iceCheckMinInterval;
> @synthesize iceRegatherIntervalRange = _iceRegatherIntervalRange;
>+@synthesize sdpSemantics = _sdpSemantics;
> @synthesize turnCustomizer = _turnCustomizer;
>+@synthesize activeResetSrtpParams = _activeResetSrtpParams;
> 
> - (instancetype)init {
>   // Copy defaults.
>@@ -60,6 +64,14 @@ - (instancetype)initWithNativeConfiguration:
>       [iceServers addObject:iceServer];
>     }
>     _iceServers = iceServers;
>+    if (!config.certificates.empty()) {
>+      rtc::scoped_refptr<rtc::RTCCertificate> native_cert;
>+      native_cert = config.certificates[0];
>+      rtc::RTCCertificatePEM native_pem = native_cert->ToPEM();
>+      _certificate =
>+          [[RTCCertificate alloc] initWithPrivateKey:@(native_pem.private_key().c_str())
>+                                         certificate:@(native_pem.certificate().c_str())];
>+    }
>     _iceTransportPolicy =
>         [[self class] transportPolicyForTransportsType:config.type];
>     _bundlePolicy =
>@@ -75,6 +87,7 @@ - (instancetype)initWithNativeConfiguration:
>     _continualGatheringPolicy =
>         [[self class] continualGatheringPolicyForNativePolicy:nativePolicy];
>     _maxIPv6Networks = config.max_ipv6_networks;
>+    _disableLinkLocalNetworks = config.disable_link_local_networks;
>     _audioJitterBufferMaxPackets = config.audio_jitter_buffer_max_packets;
>     _audioJitterBufferFastAccelerate = config.audio_jitter_buffer_fast_accelerate;
>     _iceConnectionReceivingTimeout = config.ice_connection_receiving_timeout;
>@@ -94,33 +107,40 @@ - (instancetype)initWithNativeConfiguration:
>       _iceRegatherIntervalRange =
>           [[RTCIntervalRange alloc] initWithNativeIntervalRange:nativeIntervalRange];
>     }
>+    _sdpSemantics = [[self class] sdpSemanticsForNativeSdpSemantics:config.sdp_semantics];
>     _turnCustomizer = config.turn_customizer;
>+    _activeResetSrtpParams = config.active_reset_srtp_params;
>   }
>   return self;
> }
> 
> - (NSString *)description {
>-  return
>-      [NSString stringWithFormat:
>-                    @"RTCConfiguration: "
>-                    @"{\n%@\n%@\n%@\n%@\n%@\n%@\n%@\n%d\n%d\n%d\n%d\n%d\n%d\n%d\n%@\n%@\n%d\n}\n",
>-                    _iceServers,
>-                    [[self class] stringForTransportPolicy:_iceTransportPolicy],
>-                    [[self class] stringForBundlePolicy:_bundlePolicy],
>-                    [[self class] stringForRtcpMuxPolicy:_rtcpMuxPolicy],
>-                    [[self class] stringForTcpCandidatePolicy:_tcpCandidatePolicy],
>-                    [[self class] stringForCandidateNetworkPolicy:_candidateNetworkPolicy],
>-                    [[self class] stringForContinualGatheringPolicy:_continualGatheringPolicy],
>-                    _audioJitterBufferMaxPackets,
>-                    _audioJitterBufferFastAccelerate,
>-                    _iceConnectionReceivingTimeout,
>-                    _iceBackupCandidatePairPingInterval,
>-                    _iceCandidatePoolSize,
>-                    _shouldPruneTurnPorts,
>-                    _shouldPresumeWritableWhenFullyRelayed,
>-                    _iceCheckMinInterval,
>-                    _iceRegatherIntervalRange,
>-                    _maxIPv6Networks];
>+  static NSString *formatString =
>+      @"RTCConfiguration: "
>+      @"{\n%@\n%@\n%@\n%@\n%@\n%@\n%@\n%@\n%d\n%d\n%d\n%d\n%d\n%d\n%d\n%@\n%@\n%d\n%d\n%d\n}\n";
>+
>+  return [NSString
>+      stringWithFormat:formatString,
>+                       _iceServers,
>+                       [[self class] stringForTransportPolicy:_iceTransportPolicy],
>+                       [[self class] stringForBundlePolicy:_bundlePolicy],
>+                       [[self class] stringForRtcpMuxPolicy:_rtcpMuxPolicy],
>+                       [[self class] stringForTcpCandidatePolicy:_tcpCandidatePolicy],
>+                       [[self class] stringForCandidateNetworkPolicy:_candidateNetworkPolicy],
>+                       [[self class] stringForContinualGatheringPolicy:_continualGatheringPolicy],
>+                       [[self class] stringForSdpSemantics:_sdpSemantics],
>+                       _audioJitterBufferMaxPackets,
>+                       _audioJitterBufferFastAccelerate,
>+                       _iceConnectionReceivingTimeout,
>+                       _iceBackupCandidatePairPingInterval,
>+                       _iceCandidatePoolSize,
>+                       _shouldPruneTurnPorts,
>+                       _shouldPresumeWritableWhenFullyRelayed,
>+                       _iceCheckMinInterval,
>+                       _iceRegatherIntervalRange,
>+                       _disableLinkLocalNetworks,
>+                       _maxIPv6Networks,
>+                       _activeResetSrtpParams];
> }
> 
> #pragma mark - Private
>@@ -147,6 +167,7 @@ - (webrtc::PeerConnectionInterface::RTCConfiguration *)
>   nativeConfig->continual_gathering_policy = [[self class]
>       nativeContinualGatheringPolicyForPolicy:_continualGatheringPolicy];
>   nativeConfig->max_ipv6_networks = _maxIPv6Networks;
>+  nativeConfig->disable_link_local_networks = _disableLinkLocalNetworks;
>   nativeConfig->audio_jitter_buffer_max_packets = _audioJitterBufferMaxPackets;
>   nativeConfig->audio_jitter_buffer_fast_accelerate =
>       _audioJitterBufferFastAccelerate  ? true : false;
>@@ -156,34 +177,51 @@ - (webrtc::PeerConnectionInterface::RTCConfiguration *)
>       _iceBackupCandidatePairPingInterval;
>   rtc::KeyType keyType =
>       [[self class] nativeEncryptionKeyTypeForKeyType:_keyType];
>-  // Generate non-default certificate.
>-  if (keyType != rtc::KT_DEFAULT) {
>-    rtc::scoped_refptr<rtc::RTCCertificate> certificate =
>-        rtc::RTCCertificateGenerator::GenerateCertificate(
>-            rtc::KeyParams(keyType), rtc::Optional<uint64_t>());
>+  if (_certificate != nullptr) {
>+    // if offered a pemcert use it...
>+    RTC_LOG(LS_INFO) << "Have configured cert - using it.";
>+    std::string pem_private_key = [[_certificate private_key] UTF8String];
>+    std::string pem_certificate = [[_certificate certificate] UTF8String];
>+    rtc::RTCCertificatePEM pem = rtc::RTCCertificatePEM(pem_private_key, pem_certificate);
>+    rtc::scoped_refptr<rtc::RTCCertificate> certificate = rtc::RTCCertificate::FromPEM(pem);
>+    RTC_LOG(LS_INFO) << "Created cert from PEM strings.";
>     if (!certificate) {
>-      RTCLogError(@"Failed to generate certificate.");
>+      RTC_LOG(LS_ERROR) << "Failed to generate certificate from PEM.";
>       return nullptr;
>     }
>     nativeConfig->certificates.push_back(certificate);
>+  } else {
>+    RTC_LOG(LS_INFO) << "Don't have configured cert.";
>+    // Generate non-default certificate.
>+    if (keyType != rtc::KT_DEFAULT) {
>+      rtc::scoped_refptr<rtc::RTCCertificate> certificate =
>+          rtc::RTCCertificateGenerator::GenerateCertificate(rtc::KeyParams(keyType),
>+                                                            absl::optional<uint64_t>());
>+      if (!certificate) {
>+        RTCLogError(@"Failed to generate certificate.");
>+        return nullptr;
>+      }
>+      nativeConfig->certificates.push_back(certificate);
>+    }
>   }
>   nativeConfig->ice_candidate_pool_size = _iceCandidatePoolSize;
>   nativeConfig->prune_turn_ports = _shouldPruneTurnPorts ? true : false;
>   nativeConfig->presume_writable_when_fully_relayed =
>       _shouldPresumeWritableWhenFullyRelayed ? true : false;
>   if (_iceCheckMinInterval != nil) {
>-    nativeConfig->ice_check_min_interval =
>-        rtc::Optional<int>(_iceCheckMinInterval.intValue);
>+    nativeConfig->ice_check_min_interval = absl::optional<int>(_iceCheckMinInterval.intValue);
>   }
>   if (_iceRegatherIntervalRange != nil) {
>     std::unique_ptr<rtc::IntervalRange> nativeIntervalRange(
>         _iceRegatherIntervalRange.nativeIntervalRange);
>     nativeConfig->ice_regather_interval_range =
>-        rtc::Optional<rtc::IntervalRange>(*nativeIntervalRange);
>+        absl::optional<rtc::IntervalRange>(*nativeIntervalRange);
>   }
>+  nativeConfig->sdp_semantics = [[self class] nativeSdpSemanticsForSdpSemantics:_sdpSemantics];
>   if (_turnCustomizer) {
>     nativeConfig->turn_customizer = _turnCustomizer;
>   }
>+  nativeConfig->active_reset_srtp_params = _activeResetSrtpParams ? true : false;
>   return nativeConfig.release();
> }
> 
>@@ -392,4 +430,31 @@ + (rtc::KeyType)nativeEncryptionKeyTypeForKeyType:
>   }
> }
> 
>++ (webrtc::SdpSemantics)nativeSdpSemanticsForSdpSemantics:(RTCSdpSemantics)sdpSemantics {
>+  switch (sdpSemantics) {
>+    case RTCSdpSemanticsPlanB:
>+      return webrtc::SdpSemantics::kPlanB;
>+    case RTCSdpSemanticsUnifiedPlan:
>+      return webrtc::SdpSemantics::kUnifiedPlan;
>+  }
>+}
>+
>++ (RTCSdpSemantics)sdpSemanticsForNativeSdpSemantics:(webrtc::SdpSemantics)sdpSemantics {
>+  switch (sdpSemantics) {
>+    case webrtc::SdpSemantics::kPlanB:
>+      return RTCSdpSemanticsPlanB;
>+    case webrtc::SdpSemantics::kUnifiedPlan:
>+      return RTCSdpSemanticsUnifiedPlan;
>+  }
>+}
>+
>++ (NSString *)stringForSdpSemantics:(RTCSdpSemantics)sdpSemantics {
>+  switch (sdpSemantics) {
>+    case RTCSdpSemanticsPlanB:
>+      return @"PLAN_B";
>+    case RTCSdpSemanticsUnifiedPlan:
>+      return @"UNIFIED_PLAN";
>+  }
>+}
>+
> @end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCDataChannel+Private.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCDataChannel+Private.h
>index ca806cafb9f..9849fd95c9b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCDataChannel+Private.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCDataChannel+Private.h
>@@ -15,6 +15,8 @@
> 
> NS_ASSUME_NONNULL_BEGIN
> 
>+@class RTCPeerConnectionFactory;
>+
> @interface RTCDataBuffer ()
> 
> /**
>@@ -24,23 +26,22 @@ NS_ASSUME_NONNULL_BEGIN
> @property(nonatomic, readonly) const webrtc::DataBuffer *nativeDataBuffer;
> 
> /** Initialize an RTCDataBuffer from a native DataBuffer. */
>-- (instancetype)initWithNativeBuffer:(const webrtc::DataBuffer&)nativeBuffer;
>+- (instancetype)initWithNativeBuffer:(const webrtc::DataBuffer &)nativeBuffer;
> 
> @end
> 
>-
> @interface RTCDataChannel ()
> 
> /** Initialize an RTCDataChannel from a native DataChannelInterface. */
>-- (instancetype)initWithNativeDataChannel:
>-    (rtc::scoped_refptr<webrtc::DataChannelInterface>)nativeDataChannel
>+- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
>+              nativeDataChannel:(rtc::scoped_refptr<webrtc::DataChannelInterface>)nativeDataChannel
>     NS_DESIGNATED_INITIALIZER;
> 
>-+ (webrtc::DataChannelInterface::DataState)
>-    nativeDataChannelStateForState:(RTCDataChannelState)state;
>++ (webrtc::DataChannelInterface::DataState)nativeDataChannelStateForState:
>+        (RTCDataChannelState)state;
> 
> + (RTCDataChannelState)dataChannelStateForNativeState:
>-    (webrtc::DataChannelInterface::DataState)nativeState;
>+        (webrtc::DataChannelInterface::DataState)nativeState;
> 
> + (NSString *)stringForState:(RTCDataChannelState)state;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCDataChannel.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCDataChannel.mm
>index 706e43e2ed5..06ca453bd02 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCDataChannel.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCDataChannel.mm
>@@ -85,6 +85,7 @@ @end
> 
> 
> @implementation RTCDataChannel {
>+  RTCPeerConnectionFactory *_factory;
>   rtc::scoped_refptr<webrtc::DataChannelInterface> _nativeDataChannel;
>   std::unique_ptr<webrtc::DataChannelDelegateAdapter> _observer;
>   BOOL _isObserverRegistered;
>@@ -165,10 +166,12 @@ - (NSString *)description {
> 
> #pragma mark - Private
> 
>-- (instancetype)initWithNativeDataChannel:
>-    (rtc::scoped_refptr<webrtc::DataChannelInterface>)nativeDataChannel {
>+- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
>+              nativeDataChannel:
>+                  (rtc::scoped_refptr<webrtc::DataChannelInterface>)nativeDataChannel {
>   NSParameterAssert(nativeDataChannel);
>   if (self = [super init]) {
>+    _factory = factory;
>     _nativeDataChannel = nativeDataChannel;
>     _observer.reset(new webrtc::DataChannelDelegateAdapter(self));
>     _nativeDataChannel->RegisterObserver(_observer.get());
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCDefaultVideoDecoderFactory.m b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCDefaultVideoDecoderFactory.m
>index ba27ad2b29b..5f292c9c382 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCDefaultVideoDecoderFactory.m
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCDefaultVideoDecoderFactory.m
>@@ -37,7 +37,7 @@ @implementation RTCDefaultVideoDecoderFactory
>     [[RTCVideoCodecInfo alloc] initWithName:kRTCVideoCodecH264Name],
>     [[RTCVideoCodecInfo alloc] initWithName:kRTCVideoCodecVp8Name],
> #if !defined(RTC_DISABLE_VP9)
>-    [[RTCVideoCodecInfo alloc] initWithName:kRTCVideoCodecVp9Name]
>+    [[RTCVideoCodecInfo alloc] initWithName:kRTCVideoCodecVp9Name],
> #endif
>   ];
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCDefaultVideoEncoderFactory.m b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCDefaultVideoEncoderFactory.m
>index 154fac746b6..a4b8d710f43 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCDefaultVideoEncoderFactory.m
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCDefaultVideoEncoderFactory.m
>@@ -23,7 +23,7 @@ @implementation RTCDefaultVideoEncoderFactory
> 
> + (NSArray<RTCVideoCodecInfo *> *)supportedCodecs {
>   NSDictionary<NSString *, NSString *> *constrainedHighParams = @{
>-    @"profile-level-id" : kRTCLevel31ConstrainedHigh,
>+    @"profile-level-id" : kRTCMaxSupportedH264ProfileLevelConstrainedHigh,
>     @"level-asymmetry-allowed" : @"1",
>     @"packetization-mode" : @"1",
>   };
>@@ -32,7 +32,7 @@ @implementation RTCDefaultVideoEncoderFactory
>                                    parameters:constrainedHighParams];
> 
>   NSDictionary<NSString *, NSString *> *constrainedBaselineParams = @{
>-    @"profile-level-id" : kRTCLevel31ConstrainedBaseline,
>+    @"profile-level-id" : kRTCMaxSupportedH264ProfileLevelConstrainedBaseline,
>     @"level-asymmetry-allowed" : @"1",
>     @"packetization-mode" : @"1",
>   };
>@@ -46,13 +46,14 @@ @implementation RTCDefaultVideoEncoderFactory
>   RTCVideoCodecInfo *vp9Info = [[RTCVideoCodecInfo alloc] initWithName:kRTCVideoCodecVp9Name];
> #endif
> 
>-  return @[ constrainedHighInfo,
>-            constrainedBaselineInfo,
>-            vp8Info,
>+  return @[
>+    constrainedHighInfo,
>+    constrainedBaselineInfo,
>+    vp8Info,
> #if !defined(RTC_DISABLE_VP9)
>-            vp9Info
>+    vp9Info,
> #endif
>-            ];
>+  ];
> }
> 
> - (id<RTCVideoEncoder>)createEncoder:(RTCVideoCodecInfo *)info {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCDtmfSender+Private.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCDtmfSender+Private.h
>index 04f10e8b399..197aeb83a42 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCDtmfSender+Private.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCDtmfSender+Private.h
>@@ -14,7 +14,6 @@
> 
> NS_ASSUME_NONNULL_BEGIN
> 
>-__attribute__((objc_runtime_name("WK_RTCDtmfSender")))
> @interface RTCDtmfSender : NSObject <RTCDtmfSender>
> 
> @property(nonatomic, readonly) rtc::scoped_refptr<webrtc::DtmfSenderInterface> nativeDtmfSender;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCH264ProfileLevelId.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCH264ProfileLevelId.mm
>new file mode 100644
>index 00000000000..04a5689417c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCH264ProfileLevelId.mm
>@@ -0,0 +1,58 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ *
>+ */
>+
>+#import "WebRTC/RTCVideoCodecH264.h"
>+
>+#include "media/base/h264_profile_level_id.h"
>+
>+@interface RTCH264ProfileLevelId ()
>+
>+@property(nonatomic, assign) RTCH264Profile profile;
>+@property(nonatomic, assign) RTCH264Level level;
>+@property(nonatomic, strong) NSString *hexString;
>+
>+@end
>+
>+@implementation RTCH264ProfileLevelId
>+
>+@synthesize profile = _profile;
>+@synthesize level = _level;
>+@synthesize hexString = _hexString;
>+
>+- (instancetype)initWithHexString:(NSString *)hexString {
>+  if (self = [super init]) {
>+    self.hexString = hexString;
>+
>+    absl::optional<webrtc::H264::ProfileLevelId> profile_level_id =
>+        webrtc::H264::ParseProfileLevelId([hexString cStringUsingEncoding:NSUTF8StringEncoding]);
>+    if (profile_level_id.has_value()) {
>+      self.profile = static_cast<RTCH264Profile>(profile_level_id->profile);
>+      self.level = static_cast<RTCH264Level>(profile_level_id->level);
>+    }
>+  }
>+  return self;
>+}
>+
>+- (instancetype)initWithProfile:(RTCH264Profile)profile level:(RTCH264Level)level {
>+  if (self = [super init]) {
>+    self.profile = profile;
>+    self.level = level;
>+
>+    absl::optional<std::string> hex_string =
>+        webrtc::H264::ProfileLevelIdToString(webrtc::H264::ProfileLevelId(
>+            static_cast<webrtc::H264::Profile>(profile), static_cast<webrtc::H264::Level>(level)));
>+    self.hexString =
>+        [NSString stringWithCString:hex_string.value_or("").c_str() encoding:NSUTF8StringEncoding];
>+  }
>+  return self;
>+}
>+
>+@end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCIceCandidate+Private.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCIceCandidate+Private.h
>index 6c47a398539..47e934cbb3d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCIceCandidate+Private.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCIceCandidate+Private.h
>@@ -22,15 +22,13 @@ NS_ASSUME_NONNULL_BEGIN
>  * The native IceCandidateInterface representation of this RTCIceCandidate
>  * object. This is needed to pass to the underlying C++ APIs.
>  */
>-@property(nonatomic, readonly)
>-    std::unique_ptr<webrtc::IceCandidateInterface> nativeCandidate;
>+@property(nonatomic, readonly) std::unique_ptr<webrtc::IceCandidateInterface> nativeCandidate;
> 
> /**
>  * Initialize an RTCIceCandidate from a native IceCandidateInterface. No
>  * ownership is taken of the native candidate.
>  */
>-- (instancetype)initWithNativeCandidate:
>-    (const webrtc::IceCandidateInterface *)candidate;
>+- (instancetype)initWithNativeCandidate:(const webrtc::IceCandidateInterface *)candidate;
> 
> @end
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCIceServer+Private.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCIceServer+Private.h
>index 19a845b3fec..0f6b7b7b833 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCIceServer+Private.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCIceServer+Private.h
>@@ -20,12 +20,10 @@ NS_ASSUME_NONNULL_BEGIN
>  * IceServer struct representation of this RTCIceServer object's data.
>  * This is needed to pass to the underlying C++ APIs.
>  */
>-@property(nonatomic, readonly)
>-    webrtc::PeerConnectionInterface::IceServer nativeServer;
>+@property(nonatomic, readonly) webrtc::PeerConnectionInterface::IceServer nativeServer;
> 
> /** Initialize an RTCIceServer from a native IceServer. */
>-- (instancetype)initWithNativeServer:
>-    (webrtc::PeerConnectionInterface::IceServer)nativeServer;
>+- (instancetype)initWithNativeServer:(webrtc::PeerConnectionInterface::IceServer)nativeServer;
> 
> @end
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCIntervalRange+Private.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCIntervalRange+Private.h
>index a80c99a1aeb..690daf340e9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCIntervalRange+Private.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCIntervalRange+Private.h
>@@ -16,12 +16,10 @@ NS_ASSUME_NONNULL_BEGIN
> 
> @interface RTCIntervalRange ()
> 
>-@property(nonatomic, readonly)
>-    std::unique_ptr<rtc::IntervalRange> nativeIntervalRange;
>+@property(nonatomic, readonly) std::unique_ptr<rtc::IntervalRange> nativeIntervalRange;
> 
> - (instancetype)initWithNativeIntervalRange:(const rtc::IntervalRange &)config;
> 
> @end
> 
> NS_ASSUME_NONNULL_END
>-
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCMediaConstraints+Private.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCMediaConstraints+Private.h
>index 1955c8c2651..1c8eb288957 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCMediaConstraints+Private.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCMediaConstraints+Private.h
>@@ -18,13 +18,12 @@ namespace webrtc {
> 
> class MediaConstraints : public MediaConstraintsInterface {
>  public:
>-  virtual ~MediaConstraints();
>+  ~MediaConstraints() override;
>   MediaConstraints();
>-  MediaConstraints(
>-      const MediaConstraintsInterface::Constraints& mandatory,
>-      const MediaConstraintsInterface::Constraints& optional);
>-  virtual const Constraints& GetMandatory() const;
>-  virtual const Constraints& GetOptional() const;
>+  MediaConstraints(const MediaConstraintsInterface::Constraints& mandatory,
>+                   const MediaConstraintsInterface::Constraints& optional);
>+  const Constraints& GetMandatory() const override;
>+  const Constraints& GetOptional() const override;
> 
>  private:
>   MediaConstraintsInterface::Constraints mandatory_;
>@@ -33,7 +32,6 @@ class MediaConstraints : public MediaConstraintsInterface {
> 
> }  // namespace webrtc
> 
>-
> NS_ASSUME_NONNULL_BEGIN
> 
> @interface RTCMediaConstraints ()
>@@ -45,9 +43,8 @@ NS_ASSUME_NONNULL_BEGIN
> - (std::unique_ptr<webrtc::MediaConstraints>)nativeConstraints;
> 
> /** Return a native Constraints object representing these constraints */
>-+ (webrtc::MediaConstraintsInterface::Constraints)
>-    nativeConstraintsForConstraints:
>-        (NSDictionary<NSString *, NSString *> *)constraints;
>++ (webrtc::MediaConstraintsInterface::Constraints)nativeConstraintsForConstraints:
>+        (NSDictionary<NSString*, NSString*>*)constraints;
> 
> @end
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCMediaConstraints.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCMediaConstraints.mm
>index c686b5643c3..34e5899ad43 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCMediaConstraints.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCMediaConstraints.mm
>@@ -30,8 +30,6 @@ NSString * const kRTCMediaConstraintsMinFrameRate =
>     @(webrtc::MediaConstraintsInterface::kMinFrameRate);
> NSString * const kRTCMediaConstraintsMaxFrameRate =
>     @(webrtc::MediaConstraintsInterface::kMaxFrameRate);
>-NSString * const kRTCMediaConstraintsLevelControl =
>-    @(webrtc::MediaConstraintsInterface::kLevelControl);
> NSString * const kRTCMediaConstraintsAudioNetworkAdaptorConfig =
>     @(webrtc::MediaConstraintsInterface::kAudioNetworkAdaptorConfig);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCMediaSource+Private.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCMediaSource+Private.h
>index 1656d16d75e..9883faf1c15 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCMediaSource+Private.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCMediaSource+Private.h
>@@ -14,6 +14,8 @@
> 
> NS_ASSUME_NONNULL_BEGIN
> 
>+@class RTCPeerConnectionFactory;
>+
> typedef NS_ENUM(NSInteger, RTCMediaSourceType) {
>   RTCMediaSourceTypeAudio,
>   RTCMediaSourceTypeVideo,
>@@ -21,19 +23,15 @@ typedef NS_ENUM(NSInteger, RTCMediaSourceType) {
> 
> @interface RTCMediaSource ()
> 
>-@property(nonatomic, readonly)
>-    rtc::scoped_refptr<webrtc::MediaSourceInterface> nativeMediaSource;
>+@property(nonatomic, readonly) rtc::scoped_refptr<webrtc::MediaSourceInterface> nativeMediaSource;
> 
>-- (instancetype)initWithNativeMediaSource:
>-   (rtc::scoped_refptr<webrtc::MediaSourceInterface>)nativeMediaSource
>-                                     type:(RTCMediaSourceType)type
>-    NS_DESIGNATED_INITIALIZER;
>+- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
>+              nativeMediaSource:(rtc::scoped_refptr<webrtc::MediaSourceInterface>)nativeMediaSource
>+                           type:(RTCMediaSourceType)type NS_DESIGNATED_INITIALIZER;
> 
>-+ (webrtc::MediaSourceInterface::SourceState)nativeSourceStateForState:
>-    (RTCSourceState)state;
>++ (webrtc::MediaSourceInterface::SourceState)nativeSourceStateForState:(RTCSourceState)state;
> 
>-+ (RTCSourceState)sourceStateForNativeState:
>-    (webrtc::MediaSourceInterface::SourceState)nativeState;
>++ (RTCSourceState)sourceStateForNativeState:(webrtc::MediaSourceInterface::SourceState)nativeState;
> 
> + (NSString *)stringForState:(RTCSourceState)state;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCMediaSource.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCMediaSource.mm
>index 0f8cadc07d9..6ec41c3b507 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCMediaSource.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCMediaSource.mm
>@@ -13,16 +13,19 @@
> #include "rtc_base/checks.h"
> 
> @implementation RTCMediaSource {
>+  RTCPeerConnectionFactory *_factory;
>   RTCMediaSourceType _type;
> }
> 
> @synthesize nativeMediaSource = _nativeMediaSource;
> 
>-- (instancetype)initWithNativeMediaSource:
>-   (rtc::scoped_refptr<webrtc::MediaSourceInterface>)nativeMediaSource
>-                                     type:(RTCMediaSourceType)type {
>+- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
>+              nativeMediaSource:(rtc::scoped_refptr<webrtc::MediaSourceInterface>)nativeMediaSource
>+                           type:(RTCMediaSourceType)type {
>+  RTC_DCHECK(factory);
>   RTC_DCHECK(nativeMediaSource);
>   if (self = [super init]) {
>+    _factory = factory;
>     _nativeMediaSource = nativeMediaSource;
>     _type = type;
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCMediaStream+Private.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCMediaStream+Private.h
>index 1d0847c12ca..39860664578 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCMediaStream+Private.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCMediaStream+Private.h
>@@ -20,16 +20,14 @@ NS_ASSUME_NONNULL_BEGIN
>  * MediaStreamInterface representation of this RTCMediaStream object. This is
>  * needed to pass to the underlying C++ APIs.
>  */
>-@property(nonatomic, readonly)
>-    rtc::scoped_refptr<webrtc::MediaStreamInterface> nativeMediaStream;
>+@property(nonatomic, readonly) rtc::scoped_refptr<webrtc::MediaStreamInterface> nativeMediaStream;
> 
> /** Initialize an RTCMediaStream with an id. */
>-- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
>-                       streamId:(NSString *)streamId;
>+- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory streamId:(NSString *)streamId;
> 
> /** Initialize an RTCMediaStream from a native MediaStreamInterface. */
>-- (instancetype)initWithNativeMediaStream:
>-    (rtc::scoped_refptr<webrtc::MediaStreamInterface>)nativeMediaStream;
>+- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
>+              nativeMediaStream:(rtc::scoped_refptr<webrtc::MediaStreamInterface>)nativeMediaStream;
> 
> @end
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCMediaStream.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCMediaStream.mm
>index 0737dee8a93..c8bcfd91a41 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCMediaStream.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCMediaStream.mm
>@@ -19,6 +19,7 @@
> #import "RTCVideoTrack+Private.h"
> 
> @implementation RTCMediaStream {
>+  RTCPeerConnectionFactory *_factory;
>   NSMutableArray *_audioTracks;
>   NSMutableArray *_videoTracks;
>   rtc::scoped_refptr<webrtc::MediaStreamInterface> _nativeMediaStream;
>@@ -31,7 +32,7 @@ - (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
>   std::string nativeId = [NSString stdStringForString:streamId];
>   rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
>       factory.nativeFactory->CreateLocalMediaStream(nativeId);
>-  return [self initWithNativeMediaStream:stream];
>+  return [self initWithFactory:factory nativeMediaStream:stream];
> }
> 
> - (NSArray<RTCAudioTrack *> *)audioTracks {
>@@ -43,7 +44,7 @@ - (NSArray<RTCVideoTrack *> *)videoTracks {
> }
> 
> - (NSString *)streamId {
>-  return [NSString stringForStdString:_nativeMediaStream->label()];
>+  return [NSString stringForStdString:_nativeMediaStream->id()];
> }
> 
> - (void)addAudioTrack:(RTCAudioTrack *)audioTrack {
>@@ -91,10 +92,13 @@ - (rtc::scoped_refptr<webrtc::MediaStreamInterface>)nativeMediaStream {
>   return _nativeMediaStream;
> }
> 
>-- (instancetype)initWithNativeMediaStream:
>-    (rtc::scoped_refptr<webrtc::MediaStreamInterface>)nativeMediaStream {
>+- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
>+              nativeMediaStream:
>+                  (rtc::scoped_refptr<webrtc::MediaStreamInterface>)nativeMediaStream {
>   NSParameterAssert(nativeMediaStream);
>   if (self = [super init]) {
>+    _factory = factory;
>+
>     webrtc::AudioTrackVector audioTracks = nativeMediaStream->GetAudioTracks();
>     webrtc::VideoTrackVector videoTracks = nativeMediaStream->GetVideoTracks();
> 
>@@ -105,14 +109,14 @@ - (instancetype)initWithNativeMediaStream:
>     for (auto &track : audioTracks) {
>       RTCMediaStreamTrackType type = RTCMediaStreamTrackTypeAudio;
>       RTCAudioTrack *audioTrack =
>-          [[RTCAudioTrack alloc] initWithNativeTrack:track type:type];
>+          [[RTCAudioTrack alloc] initWithFactory:_factory nativeTrack:track type:type];
>       [_audioTracks addObject:audioTrack];
>     }
> 
>     for (auto &track : videoTracks) {
>       RTCMediaStreamTrackType type = RTCMediaStreamTrackTypeVideo;
>       RTCVideoTrack *videoTrack =
>-          [[RTCVideoTrack alloc] initWithNativeTrack:track type:type];
>+          [[RTCVideoTrack alloc] initWithFactory:_factory nativeTrack:track type:type];
>       [_videoTracks addObject:videoTrack];
>     }
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCMediaStreamTrack+Private.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCMediaStreamTrack+Private.h
>index cb93f116f0d..6effeaac195 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCMediaStreamTrack+Private.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCMediaStreamTrack+Private.h
>@@ -19,36 +19,42 @@ typedef NS_ENUM(NSInteger, RTCMediaStreamTrackType) {
> 
> NS_ASSUME_NONNULL_BEGIN
> 
>+@class RTCPeerConnectionFactory;
>+
> @interface RTCMediaStreamTrack ()
> 
>+@property(nonatomic, readonly) RTCPeerConnectionFactory *factory;
>+
> /**
>  * The native MediaStreamTrackInterface passed in or created during
>  * construction.
>  */
>-@property(nonatomic, readonly)
>-    rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> nativeTrack;
>+@property(nonatomic, readonly) rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> nativeTrack;
> 
> /**
>  * Initialize an RTCMediaStreamTrack from a native MediaStreamTrackInterface.
>  */
>-- (instancetype)initWithNativeTrack:
>-    (rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack
>-                               type:(RTCMediaStreamTrackType)type
>-    NS_DESIGNATED_INITIALIZER;
>+- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
>+                    nativeTrack:(rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack
>+                           type:(RTCMediaStreamTrackType)type NS_DESIGNATED_INITIALIZER;
> 
>-- (instancetype)initWithNativeTrack:
>-    (rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack;
>+- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
>+                    nativeTrack:(rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack;
> 
> - (BOOL)isEqualToTrack:(RTCMediaStreamTrack *)track;
> 
> + (webrtc::MediaStreamTrackInterface::TrackState)nativeTrackStateForState:
>-    (RTCMediaStreamTrackState)state;
>+        (RTCMediaStreamTrackState)state;
> 
> + (RTCMediaStreamTrackState)trackStateForNativeState:
>-    (webrtc::MediaStreamTrackInterface::TrackState)nativeState;
>+        (webrtc::MediaStreamTrackInterface::TrackState)nativeState;
> 
> + (NSString *)stringForState:(RTCMediaStreamTrackState)state;
> 
>++ (RTCMediaStreamTrack *)mediaTrackForNativeTrack:
>+                             (rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack
>+                                          factory:(RTCPeerConnectionFactory *)factory;
>+
> @end
> 
> NS_ASSUME_NONNULL_END
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCMediaStreamTrack.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCMediaStreamTrack.mm
>index 208550f6c86..07bb0094c1a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCMediaStreamTrack.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCMediaStreamTrack.mm
>@@ -8,7 +8,9 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>+#import "RTCAudioTrack+Private.h"
> #import "RTCMediaStreamTrack+Private.h"
>+#import "RTCVideoTrack+Private.h"
> 
> #import "NSString+StdString.h"
> 
>@@ -18,6 +20,7 @@ NSString * const kRTCMediaStreamTrackKindVideo =
>     @(webrtc::MediaStreamTrackInterface::kVideoKind);
> 
> @implementation RTCMediaStreamTrack {
>+  RTCPeerConnectionFactory *_factory;
>   rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> _nativeTrack;
>   RTCMediaStreamTrackType _type;
> }
>@@ -71,29 +74,31 @@ - (rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack {
>   return _nativeTrack;
> }
> 
>-- (instancetype)initWithNativeTrack:
>-    (rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack
>-                               type:(RTCMediaStreamTrackType)type {
>+@synthesize factory = _factory;
>+
>+- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
>+                    nativeTrack:(rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack
>+                           type:(RTCMediaStreamTrackType)type {
>   NSParameterAssert(nativeTrack);
>+  NSParameterAssert(factory);
>   if (self = [super init]) {
>+    _factory = factory;
>     _nativeTrack = nativeTrack;
>     _type = type;
>   }
>   return self;
> }
> 
>-- (instancetype)initWithNativeTrack:
>-    (rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack {
>+- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
>+                    nativeTrack:(rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack {
>   NSParameterAssert(nativeTrack);
>   if (nativeTrack->kind() ==
>       std::string(webrtc::MediaStreamTrackInterface::kAudioKind)) {
>-    return [self initWithNativeTrack:nativeTrack
>-                                type:RTCMediaStreamTrackTypeAudio];
>+    return [self initWithFactory:factory nativeTrack:nativeTrack type:RTCMediaStreamTrackTypeAudio];
>   }
>   if (nativeTrack->kind() ==
>       std::string(webrtc::MediaStreamTrackInterface::kVideoKind)) {
>-    return [self initWithNativeTrack:nativeTrack
>-                                type:RTCMediaStreamTrackTypeVideo];
>+    return [self initWithFactory:factory nativeTrack:nativeTrack type:RTCMediaStreamTrackTypeVideo];
>   }
>   return nil;
> }
>@@ -134,4 +139,22 @@ + (NSString *)stringForState:(RTCMediaStreamTrackState)state {
>   }
> }
> 
>++ (RTCMediaStreamTrack *)mediaTrackForNativeTrack:
>+                             (rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeTrack
>+                                          factory:(RTCPeerConnectionFactory *)factory {
>+  NSParameterAssert(nativeTrack);
>+  NSParameterAssert(factory);
>+  if (nativeTrack->kind() == webrtc::MediaStreamTrackInterface::kAudioKind) {
>+    return [[RTCAudioTrack alloc] initWithFactory:factory
>+                                      nativeTrack:nativeTrack
>+                                             type:RTCMediaStreamTrackTypeAudio];
>+  } else if (nativeTrack->kind() == webrtc::MediaStreamTrackInterface::kVideoKind) {
>+    return [[RTCVideoTrack alloc] initWithFactory:factory
>+                                      nativeTrack:nativeTrack
>+                                             type:RTCMediaStreamTrackTypeVideo];
>+  } else {
>+    return [[RTCMediaStreamTrack alloc] initWithFactory:factory nativeTrack:nativeTrack];
>+  }
>+}
>+
> @end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCMetrics.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCMetrics.mm
>index 279a2bec311..2e80216dea6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCMetrics.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCMetrics.mm
>@@ -12,11 +12,11 @@
> 
> #import "RTCMetricsSampleInfo+Private.h"
> 
>-void RTCEnableMetrics() {
>+void RTCEnableMetrics(void) {
>   webrtc::metrics::Enable();
> }
> 
>-NSArray<RTCMetricsSampleInfo *> *RTCGetAndResetMetrics() {
>+NSArray<RTCMetricsSampleInfo *> *RTCGetAndResetMetrics(void) {
>   std::map<std::string, std::unique_ptr<webrtc::metrics::SampleInfo>>
>       histograms;
>   webrtc::metrics::GetAndReset(&histograms);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCMetricsSampleInfo+Private.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCMetricsSampleInfo+Private.h
>index 48451df46a1..ceaca8f53fe 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCMetricsSampleInfo+Private.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCMetricsSampleInfo+Private.h
>@@ -20,8 +20,7 @@ NS_ASSUME_NONNULL_BEGIN
> @interface RTCMetricsSampleInfo ()
> 
> /** Initialize an RTCMetricsSampleInfo object from native SampleInfo. */
>-- (instancetype)initWithNativeSampleInfo:
>-    (const webrtc::metrics::SampleInfo &)info;
>+- (instancetype)initWithNativeSampleInfo:(const webrtc::metrics::SampleInfo &)info;
> 
> @end
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnection+DataChannel.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnection+DataChannel.mm
>index d17d9ac2f46..c6f2b0bc9fd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnection+DataChannel.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnection+DataChannel.mm
>@@ -27,7 +27,7 @@ - (nullable RTCDataChannel *)dataChannelForLabel:(NSString *)label
>   if (!dataChannel) {
>     return nil;
>   }
>-  return [[RTCDataChannel alloc] initWithNativeDataChannel:dataChannel];
>+  return [[RTCDataChannel alloc] initWithFactory:self.factory nativeDataChannel:dataChannel];
> }
> 
> @end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnection+Private.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnection+Private.h
>index e1017f5e8f2..1d02b636913 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnection+Private.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnection+Private.h
>@@ -21,86 +21,83 @@ namespace webrtc {
>  * id<RTCPeerConnectionDelegate> and call methods on that interface.
>  */
> class PeerConnectionDelegateAdapter : public PeerConnectionObserver {
>-
>  public:
>   PeerConnectionDelegateAdapter(RTCPeerConnection *peerConnection);
>-  virtual ~PeerConnectionDelegateAdapter();
>+  ~PeerConnectionDelegateAdapter() override;
> 
>-  void OnSignalingChange(
>-      PeerConnectionInterface::SignalingState new_state) override;
>+  void OnSignalingChange(PeerConnectionInterface::SignalingState new_state) override;
> 
>   void OnAddStream(rtc::scoped_refptr<MediaStreamInterface> stream) override;
> 
>   void OnRemoveStream(rtc::scoped_refptr<MediaStreamInterface> stream) override;
> 
>-  void OnDataChannel(
>-      rtc::scoped_refptr<DataChannelInterface> data_channel) override;
>+  void OnTrack(rtc::scoped_refptr<RtpTransceiverInterface> transceiver) override;
>+
>+  void OnDataChannel(rtc::scoped_refptr<DataChannelInterface> data_channel) override;
> 
>   void OnRenegotiationNeeded() override;
> 
>-  void OnIceConnectionChange(
>-      PeerConnectionInterface::IceConnectionState new_state) override;
>+  void OnIceConnectionChange(PeerConnectionInterface::IceConnectionState new_state) override;
> 
>-  void OnIceGatheringChange(
>-      PeerConnectionInterface::IceGatheringState new_state) override;
>+  void OnIceGatheringChange(PeerConnectionInterface::IceGatheringState new_state) override;
> 
>   void OnIceCandidate(const IceCandidateInterface *candidate) override;
> 
>-  void OnIceCandidatesRemoved(
>-      const std::vector<cricket::Candidate>& candidates) override;
>+  void OnIceCandidatesRemoved(const std::vector<cricket::Candidate> &candidates) override;
>+
>+  void OnAddTrack(rtc::scoped_refptr<RtpReceiverInterface> receiver,
>+                  const std::vector<rtc::scoped_refptr<MediaStreamInterface>> &streams) override;
> 
>  private:
>   __weak RTCPeerConnection *peer_connection_;
> };
> 
>-} // namespace webrtc
>-
>+}  // namespace webrtc
> 
> @interface RTCPeerConnection ()
> 
>+/** The factory used to create this RTCPeerConnection */
>+@property(nonatomic, readonly) RTCPeerConnectionFactory *factory;
>+
> /** The native PeerConnectionInterface created during construction. */
>-@property(nonatomic, readonly)
>-    rtc::scoped_refptr<webrtc::PeerConnectionInterface> nativePeerConnection;
>+@property(nonatomic, readonly) rtc::scoped_refptr<webrtc::PeerConnectionInterface>
>+    nativePeerConnection;
> 
> /** Initialize an RTCPeerConnection with a configuration, constraints, and
>  *  delegate.
>  */
>-- (instancetype)initWithFactory:
>-    (RTCPeerConnectionFactory *)factory
>-                  configuration:
>-    (RTCConfiguration *)configuration
>-                    constraints:
>-    (RTCMediaConstraints *)constraints
>-                       delegate:
>-    (nullable id<RTCPeerConnectionDelegate>)delegate
>+- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
>+                  configuration:(RTCConfiguration *)configuration
>+                    constraints:(RTCMediaConstraints *)constraints
>+                       delegate:(nullable id<RTCPeerConnectionDelegate>)delegate
>     NS_DESIGNATED_INITIALIZER;
> 
> + (webrtc::PeerConnectionInterface::SignalingState)nativeSignalingStateForState:
>-    (RTCSignalingState)state;
>+        (RTCSignalingState)state;
> 
> + (RTCSignalingState)signalingStateForNativeState:
>-    (webrtc::PeerConnectionInterface::SignalingState)nativeState;
>+        (webrtc::PeerConnectionInterface::SignalingState)nativeState;
> 
> + (NSString *)stringForSignalingState:(RTCSignalingState)state;
> 
>-+ (webrtc::PeerConnectionInterface::IceConnectionState)
>-    nativeIceConnectionStateForState:(RTCIceConnectionState)state;
>++ (webrtc::PeerConnectionInterface::IceConnectionState)nativeIceConnectionStateForState:
>+        (RTCIceConnectionState)state;
> 
> + (RTCIceConnectionState)iceConnectionStateForNativeState:
>-    (webrtc::PeerConnectionInterface::IceConnectionState)nativeState;
>+        (webrtc::PeerConnectionInterface::IceConnectionState)nativeState;
> 
> + (NSString *)stringForIceConnectionState:(RTCIceConnectionState)state;
> 
>-+ (webrtc::PeerConnectionInterface::IceGatheringState)
>-    nativeIceGatheringStateForState:(RTCIceGatheringState)state;
>++ (webrtc::PeerConnectionInterface::IceGatheringState)nativeIceGatheringStateForState:
>+        (RTCIceGatheringState)state;
> 
> + (RTCIceGatheringState)iceGatheringStateForNativeState:
>-    (webrtc::PeerConnectionInterface::IceGatheringState)nativeState;
>+        (webrtc::PeerConnectionInterface::IceGatheringState)nativeState;
> 
> + (NSString *)stringForIceGatheringState:(RTCIceGatheringState)state;
> 
>-+ (webrtc::PeerConnectionInterface::StatsOutputLevel)
>-    nativeStatsOutputLevelForLevel:(RTCStatsOutputLevel)level;
>++ (webrtc::PeerConnectionInterface::StatsOutputLevel)nativeStatsOutputLevelForLevel:
>+        (RTCStatsOutputLevel)level;
> 
> @end
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnection+Stats.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnection+Stats.mm
>index fb16a31b3b1..e39972261b0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnection+Stats.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnection+Stats.mm
>@@ -25,9 +25,7 @@ class StatsObserverAdapter : public StatsObserver {
>     completion_handler_ = completionHandler;
>   }
> 
>-  ~StatsObserverAdapter() {
>-    completion_handler_ = nil;
>-  }
>+  ~StatsObserverAdapter() override { completion_handler_ = nil; }
> 
>   void OnComplete(const StatsReports& reports) override {
>     RTC_DCHECK(completion_handler_);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnection.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnection.mm
>index 6376ea04297..103b3c095dc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnection.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnection.mm
>@@ -17,10 +17,12 @@
> #import "RTCLegacyStatsReport+Private.h"
> #import "RTCMediaConstraints+Private.h"
> #import "RTCMediaStream+Private.h"
>+#import "RTCMediaStreamTrack+Private.h"
> #import "RTCPeerConnection+Native.h"
> #import "RTCPeerConnectionFactory+Private.h"
> #import "RTCRtpReceiver+Private.h"
> #import "RTCRtpSender+Private.h"
>+#import "RTCRtpTransceiver+Private.h"
> #import "RTCSessionDescription+Private.h"
> #import "WebRTC/RTCLogging.h"
> 
>@@ -44,9 +46,7 @@ class CreateSessionDescriptionObserverAdapter
>     completion_handler_ = completionHandler;
>   }
> 
>-  ~CreateSessionDescriptionObserverAdapter() {
>-    completion_handler_ = nil;
>-  }
>+  ~CreateSessionDescriptionObserverAdapter() override { completion_handler_ = nil; }
> 
>   void OnSuccess(SessionDescriptionInterface *desc) override {
>     RTC_DCHECK(completion_handler_);
>@@ -59,9 +59,10 @@ class CreateSessionDescriptionObserverAdapter
>     completion_handler_ = nil;
>   }
> 
>-  void OnFailure(const std::string& error) override {
>+  void OnFailure(RTCError error) override {
>     RTC_DCHECK(completion_handler_);
>-    NSString* str = [NSString stringForStdString:error];
>+    // TODO(hta): Add handling of error.type()
>+    NSString *str = [NSString stringForStdString:error.message()];
>     NSError* err =
>         [NSError errorWithDomain:kRTCPeerConnectionErrorDomain
>                             code:kRTCPeerConnnectionSessionDescriptionError
>@@ -83,9 +84,7 @@ class SetSessionDescriptionObserverAdapter :
>     completion_handler_ = completionHandler;
>   }
> 
>-  ~SetSessionDescriptionObserverAdapter() {
>-    completion_handler_ = nil;
>-  }
>+  ~SetSessionDescriptionObserverAdapter() override { completion_handler_ = nil; }
> 
>   void OnSuccess() override {
>     RTC_DCHECK(completion_handler_);
>@@ -93,9 +92,10 @@ class SetSessionDescriptionObserverAdapter :
>     completion_handler_ = nil;
>   }
> 
>-  void OnFailure(const std::string& error) override {
>+  void OnFailure(RTCError error) override {
>     RTC_DCHECK(completion_handler_);
>-    NSString* str = [NSString stringForStdString:error];
>+    // TODO(hta): Add handling of error.type()
>+    NSString *str = [NSString stringForStdString:error.message()];
>     NSError* err =
>         [NSError errorWithDomain:kRTCPeerConnectionErrorDomain
>                             code:kRTCPeerConnnectionSessionDescriptionError
>@@ -128,27 +128,41 @@ void PeerConnectionDelegateAdapter::OnSignalingChange(
> 
> void PeerConnectionDelegateAdapter::OnAddStream(
>     rtc::scoped_refptr<MediaStreamInterface> stream) {
>-  RTCMediaStream *mediaStream =
>-      [[RTCMediaStream alloc] initWithNativeMediaStream:stream];
>   RTCPeerConnection *peer_connection = peer_connection_;
>+  RTCMediaStream *mediaStream =
>+      [[RTCMediaStream alloc] initWithFactory:peer_connection.factory nativeMediaStream:stream];
>   [peer_connection.delegate peerConnection:peer_connection
>                               didAddStream:mediaStream];
> }
> 
> void PeerConnectionDelegateAdapter::OnRemoveStream(
>     rtc::scoped_refptr<MediaStreamInterface> stream) {
>-  RTCMediaStream *mediaStream =
>-      [[RTCMediaStream alloc] initWithNativeMediaStream:stream];
>   RTCPeerConnection *peer_connection = peer_connection_;
>+  RTCMediaStream *mediaStream =
>+      [[RTCMediaStream alloc] initWithFactory:peer_connection.factory nativeMediaStream:stream];
>+
>   [peer_connection.delegate peerConnection:peer_connection
>                            didRemoveStream:mediaStream];
> }
> 
>+void PeerConnectionDelegateAdapter::OnTrack(
>+    rtc::scoped_refptr<RtpTransceiverInterface> nativeTransceiver) {
>+  RTCPeerConnection *peer_connection = peer_connection_;
>+  RTCRtpTransceiver *transceiver =
>+      [[RTCRtpTransceiver alloc] initWithFactory:peer_connection.factory
>+                            nativeRtpTransceiver:nativeTransceiver];
>+  if ([peer_connection.delegate
>+          respondsToSelector:@selector(peerConnection:didStartReceivingOnTransceiver:)]) {
>+    [peer_connection.delegate peerConnection:peer_connection
>+              didStartReceivingOnTransceiver:transceiver];
>+  }
>+}
>+
> void PeerConnectionDelegateAdapter::OnDataChannel(
>     rtc::scoped_refptr<DataChannelInterface> data_channel) {
>-  RTCDataChannel *dataChannel =
>-      [[RTCDataChannel alloc] initWithNativeDataChannel:data_channel];
>   RTCPeerConnection *peer_connection = peer_connection_;
>+  RTCDataChannel *dataChannel = [[RTCDataChannel alloc] initWithFactory:peer_connection.factory
>+                                                      nativeDataChannel:data_channel];
>   [peer_connection.delegate peerConnection:peer_connection
>                         didOpenDataChannel:dataChannel];
> }
>@@ -201,10 +215,32 @@ void PeerConnectionDelegateAdapter::OnIceCandidatesRemoved(
>                     didRemoveIceCandidates:ice_candidates];
> }
> 
>+void PeerConnectionDelegateAdapter::OnAddTrack(
>+    rtc::scoped_refptr<RtpReceiverInterface> receiver,
>+    const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams) {
>+  RTCPeerConnection *peer_connection = peer_connection_;
>+  if ([peer_connection.delegate
>+          respondsToSelector:@selector(peerConnection:didAddReceiver:streams:)]) {
>+    NSMutableArray *mediaStreams = [NSMutableArray arrayWithCapacity:streams.size()];
>+    for (const auto& nativeStream : streams) {
>+      RTCMediaStream *mediaStream = [[RTCMediaStream alloc] initWithFactory:peer_connection.factory
>+                                                          nativeMediaStream:nativeStream];
>+      [mediaStreams addObject:mediaStream];
>+    }
>+    RTCRtpReceiver *rtpReceiver =
>+        [[RTCRtpReceiver alloc] initWithFactory:peer_connection.factory nativeRtpReceiver:receiver];
>+
>+    [peer_connection.delegate peerConnection:peer_connection
>+                              didAddReceiver:rtpReceiver
>+                                     streams:mediaStreams];
>+  }
>+}
>+
> }  // namespace webrtc
> 
> 
> @implementation RTCPeerConnection {
>+  RTCPeerConnectionFactory *_factory;
>   NSMutableArray<RTCMediaStream *> *_localStreams;
>   std::unique_ptr<webrtc::PeerConnectionDelegateAdapter> _observer;
>   rtc::scoped_refptr<webrtc::PeerConnectionInterface> _peerConnection;
>@@ -213,6 +249,7 @@ @implementation RTCPeerConnection {
> }
> 
> @synthesize delegate = _delegate;
>+@synthesize factory = _factory;
> 
> - (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
>                   configuration:(RTCConfiguration *)configuration
>@@ -237,6 +274,7 @@ - (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
>     if (!_peerConnection) {
>       return nil;
>     }
>+    _factory = factory;
>     _localStreams = [[NSMutableArray alloc] init];
>     _delegate = delegate;
>   }
>@@ -334,6 +372,65 @@ - (void)removeStream:(RTCMediaStream *)stream {
>   [_localStreams removeObject:stream];
> }
> 
>+- (RTCRtpSender *)addTrack:(RTCMediaStreamTrack *)track streamIds:(NSArray<NSString *> *)streamIds {
>+  std::vector<std::string> nativeStreamIds;
>+  for (NSString *streamId in streamIds) {
>+    nativeStreamIds.push_back([streamId UTF8String]);
>+  }
>+  webrtc::RTCErrorOr<rtc::scoped_refptr<webrtc::RtpSenderInterface>> nativeSenderOrError =
>+      _peerConnection->AddTrack(track.nativeTrack, nativeStreamIds);
>+  if (!nativeSenderOrError.ok()) {
>+    RTCLogError(@"Failed to add track %@: %s", track, nativeSenderOrError.error().message());
>+    return nil;
>+  }
>+  return [[RTCRtpSender alloc] initWithFactory:self.factory
>+                               nativeRtpSender:nativeSenderOrError.MoveValue()];
>+}
>+
>+- (BOOL)removeTrack:(RTCRtpSender *)sender {
>+  bool result = _peerConnection->RemoveTrack(sender.nativeRtpSender);
>+  if (!result) {
>+    RTCLogError(@"Failed to remote track %@", sender);
>+  }
>+  return result;
>+}
>+
>+- (RTCRtpTransceiver *)addTransceiverWithTrack:(RTCMediaStreamTrack *)track {
>+  return [self addTransceiverWithTrack:track init:[[RTCRtpTransceiverInit alloc] init]];
>+}
>+
>+- (RTCRtpTransceiver *)addTransceiverWithTrack:(RTCMediaStreamTrack *)track
>+                                          init:(RTCRtpTransceiverInit *)init {
>+  webrtc::RTCErrorOr<rtc::scoped_refptr<webrtc::RtpTransceiverInterface>> nativeTransceiverOrError =
>+      _peerConnection->AddTransceiver(track.nativeTrack, init.nativeInit);
>+  if (!nativeTransceiverOrError.ok()) {
>+    RTCLogError(
>+        @"Failed to add transceiver %@: %s", track, nativeTransceiverOrError.error().message());
>+    return nil;
>+  }
>+  return [[RTCRtpTransceiver alloc] initWithFactory:self.factory
>+                               nativeRtpTransceiver:nativeTransceiverOrError.MoveValue()];
>+}
>+
>+- (RTCRtpTransceiver *)addTransceiverOfType:(RTCRtpMediaType)mediaType {
>+  return [self addTransceiverOfType:mediaType init:[[RTCRtpTransceiverInit alloc] init]];
>+}
>+
>+- (RTCRtpTransceiver *)addTransceiverOfType:(RTCRtpMediaType)mediaType
>+                                       init:(RTCRtpTransceiverInit *)init {
>+  webrtc::RTCErrorOr<rtc::scoped_refptr<webrtc::RtpTransceiverInterface>> nativeTransceiverOrError =
>+      _peerConnection->AddTransceiver([RTCRtpReceiver nativeMediaTypeForMediaType:mediaType],
>+                                      init.nativeInit);
>+  if (!nativeTransceiverOrError.ok()) {
>+    RTCLogError(@"Failed to add transceiver %@: %s",
>+                [RTCRtpReceiver stringForMediaType:mediaType],
>+                nativeTransceiverOrError.error().message());
>+    return nil;
>+  }
>+  return [[RTCRtpTransceiver alloc] initWithFactory:self.factory
>+                               nativeRtpTransceiver:nativeTransceiverOrError.MoveValue()];
>+}
>+
> - (void)offerForConstraints:(RTCMediaConstraints *)constraints
>           completionHandler:
>     (void (^)(RTCSessionDescription *sessionDescription,
>@@ -375,13 +472,13 @@ - (BOOL)setBweMinBitrateBps:(nullable NSNumber *)minBitrateBps
>               maxBitrateBps:(nullable NSNumber *)maxBitrateBps {
>   webrtc::PeerConnectionInterface::BitrateParameters params;
>   if (minBitrateBps != nil) {
>-    params.min_bitrate_bps = rtc::Optional<int>(minBitrateBps.intValue);
>+    params.min_bitrate_bps = absl::optional<int>(minBitrateBps.intValue);
>   }
>   if (currentBitrateBps != nil) {
>-    params.current_bitrate_bps = rtc::Optional<int>(currentBitrateBps.intValue);
>+    params.current_bitrate_bps = absl::optional<int>(currentBitrateBps.intValue);
>   }
>   if (maxBitrateBps != nil) {
>-    params.max_bitrate_bps = rtc::Optional<int>(maxBitrateBps.intValue);
>+    params.max_bitrate_bps = absl::optional<int>(maxBitrateBps.intValue);
>   }
>   return _peerConnection->SetBitrate(params).ok();
> }
>@@ -423,8 +520,8 @@ - (RTCRtpSender *)senderWithKind:(NSString *)kind
>   rtc::scoped_refptr<webrtc::RtpSenderInterface> nativeSender(
>       _peerConnection->CreateSender(nativeKind, nativeStreamId));
>   return nativeSender ?
>-      [[RTCRtpSender alloc] initWithNativeRtpSender:nativeSender]
>-      : nil;
>+      [[RTCRtpSender alloc] initWithFactory:self.factory nativeRtpSender:nativeSender] :
>+      nil;
> }
> 
> - (NSArray<RTCRtpSender *> *)senders {
>@@ -433,7 +530,7 @@ - (NSArray<RTCRtpSender *> *)senders {
>   NSMutableArray *senders = [[NSMutableArray alloc] init];
>   for (const auto &nativeSender : nativeSenders) {
>     RTCRtpSender *sender =
>-        [[RTCRtpSender alloc] initWithNativeRtpSender:nativeSender];
>+        [[RTCRtpSender alloc] initWithFactory:self.factory nativeRtpSender:nativeSender];
>     [senders addObject:sender];
>   }
>   return senders;
>@@ -445,12 +542,24 @@ - (NSArray<RTCRtpReceiver *> *)receivers {
>   NSMutableArray *receivers = [[NSMutableArray alloc] init];
>   for (const auto &nativeReceiver : nativeReceivers) {
>     RTCRtpReceiver *receiver =
>-        [[RTCRtpReceiver alloc] initWithNativeRtpReceiver:nativeReceiver];
>+        [[RTCRtpReceiver alloc] initWithFactory:self.factory nativeRtpReceiver:nativeReceiver];
>     [receivers addObject:receiver];
>   }
>   return receivers;
> }
> 
>+- (NSArray<RTCRtpTransceiver *> *)transceivers {
>+  std::vector<rtc::scoped_refptr<webrtc::RtpTransceiverInterface>> nativeTransceivers(
>+      _peerConnection->GetTransceivers());
>+  NSMutableArray *transceivers = [[NSMutableArray alloc] init];
>+  for (auto nativeTransceiver : nativeTransceivers) {
>+    RTCRtpTransceiver *transceiver = [[RTCRtpTransceiver alloc] initWithFactory:self.factory
>+                                                           nativeRtpTransceiver:nativeTransceiver];
>+    [transceivers addObject:transceiver];
>+  }
>+  return transceivers;
>+}
>+
> #pragma mark - Private
> 
> + (webrtc::PeerConnectionInterface::SignalingState)nativeSignalingStateForState:
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactory+Native.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactory+Native.h
>index 32e23903b01..3d460fe4d80 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactory+Native.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactory+Native.h
>@@ -23,13 +23,6 @@ class AudioProcessing;
> 
> }  // namespace webrtc
> 
>-namespace cricket {
>-
>-class WebRtcVideoEncoderFactory;
>-class WebRtcVideoDecoderFactory;
>-
>-}  // namespace cricket
>-
> NS_ASSUME_NONNULL_BEGIN
> 
> /**
>@@ -56,18 +49,6 @@ NS_ASSUME_NONNULL_BEGIN
>                             audioProcessingModule:
>                                 (rtc::scoped_refptr<webrtc::AudioProcessing>)audioProcessingModule;
> 
>-/* Initialize object with legacy injectable native audio/video encoder/decoder factories
>-   TODO(andersc): Remove this when backwards compatiblity is no longer needed.
>- */
>-- (instancetype)
>-    initWithNativeAudioEncoderFactory:
>-        (rtc::scoped_refptr<webrtc::AudioEncoderFactory>)audioEncoderFactory
>-            nativeAudioDecoderFactory:
>-                (rtc::scoped_refptr<webrtc::AudioDecoderFactory>)audioDecoderFactory
>-      legacyNativeVideoEncoderFactory:(cricket::WebRtcVideoEncoderFactory*)videoEncoderFactory
>-      legacyNativeVideoDecoderFactory:(cricket::WebRtcVideoDecoderFactory*)videoDecoderFactory
>-                    audioDeviceModule:(nullable webrtc::AudioDeviceModule *)audioDeviceModule;
>-
> @end
> 
> NS_ASSUME_NONNULL_END
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactory+Private.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactory+Private.h
>index 1e5d1795f44..50944eecbb2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactory+Private.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactory+Private.h
>@@ -23,7 +23,8 @@ NS_ASSUME_NONNULL_BEGIN
>  * C++ APIs.
>  */
> @property(nonatomic, readonly)
>-    rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> nativeFactory;
>+    rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface>
>+        nativeFactory;
> 
> @end
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactory.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactory.mm
>index 31d56954783..4c801f05d6d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactory.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactory.mm
>@@ -13,7 +13,6 @@
> #import "RTCPeerConnectionFactoryOptions+Private.h"
> 
> #import "NSString+StdString.h"
>-#import "RTCAVFoundationVideoSource+Private.h"
> #import "RTCAudioSource+Private.h"
> #import "RTCAudioTrack+Private.h"
> #import "RTCMediaConstraints+Private.h"
>@@ -24,24 +23,31 @@
> #import "WebRTC/RTCLogging.h"
> #import "WebRTC/RTCVideoCodecFactory.h"
> #ifndef HAVE_NO_MEDIA
>-#include "VideoToolbox/objc_video_decoder_factory.h"
>-#include "VideoToolbox/objc_video_encoder_factory.h"
> #import "WebRTC/RTCVideoCodecH264.h"
> // The no-media version PeerConnectionFactory doesn't depend on these files, but the gn check tool
> // is not smart enough to take the #ifdef into account.
> #include "api/audio_codecs/builtin_audio_decoder_factory.h"     // nogncheck
> #include "api/audio_codecs/builtin_audio_encoder_factory.h"     // nogncheck
>+#include "media/engine/convert_legacy_video_factory.h"          // nogncheck
> #include "modules/audio_device/include/audio_device.h"          // nogncheck
> #include "modules/audio_processing/include/audio_processing.h"  // nogncheck
>+
>+#include "sdk/objc/Framework/Native/api/video_decoder_factory.h"
>+#include "sdk/objc/Framework/Native/api/video_encoder_factory.h"
>+#include "sdk/objc/Framework/Native/src/objc_video_decoder_factory.h"
>+#include "sdk/objc/Framework/Native/src/objc_video_encoder_factory.h"
>+#endif
>+
>+#if defined(WEBRTC_IOS)
>+#import "sdk/objc/Framework/Native/api/audio_device_module.h"
> #endif
> 
>-#include "Video/objcvideotracksource.h"
>-#include "api/videosourceproxy.h"
> // Adding the nogncheck to disable the including header check.
> // The no-media version PeerConnectionFactory doesn't depend on media related
> // C++ target.
> // TODO(zhihuang): Remove nogncheck once MediaEngineInterface is moved to C++
> // API layer.
>+#include "absl/memory/memory.h"
> #include "media/engine/webrtcmediaengine.h"  // nogncheck
> 
> @implementation RTCPeerConnectionFactory {
>@@ -53,18 +59,26 @@ @implementation RTCPeerConnectionFactory {
> 
> @synthesize nativeFactory = _nativeFactory;
> 
>+- (rtc::scoped_refptr<webrtc::AudioDeviceModule>)audioDeviceModule {
>+#if defined(WEBRTC_IOS)
>+  return webrtc::CreateAudioDeviceModule();
>+#else
>+  return nullptr;
>+#endif
>+}
>+
> - (instancetype)init {
> #ifdef HAVE_NO_MEDIA
>   return [self initWithNoMedia];
> #else
>   return [self initWithNativeAudioEncoderFactory:webrtc::CreateBuiltinAudioEncoderFactory()
>                        nativeAudioDecoderFactory:webrtc::CreateBuiltinAudioDecoderFactory()
>-                 legacyNativeVideoEncoderFactory:new webrtc::ObjCVideoEncoderFactory(
>+                       nativeVideoEncoderFactory:webrtc::ObjCToNativeVideoEncoderFactory(
>                                                      [[RTCVideoEncoderFactoryH264 alloc] init])
>-                 legacyNativeVideoDecoderFactory:new webrtc::ObjCVideoDecoderFactory(
>+                       nativeVideoDecoderFactory:webrtc::ObjCToNativeVideoDecoderFactory(
>                                                      [[RTCVideoDecoderFactoryH264 alloc] init])
>-                               audioDeviceModule:nullptr];
>-
>+                               audioDeviceModule:[self audioDeviceModule]
>+                           audioProcessingModule:nullptr];
> #endif
> }
> 
>@@ -76,16 +90,16 @@ - (instancetype)initWithEncoderFactory:(nullable id<RTCVideoEncoderFactory>)enco
>   std::unique_ptr<webrtc::VideoEncoderFactory> native_encoder_factory;
>   std::unique_ptr<webrtc::VideoDecoderFactory> native_decoder_factory;
>   if (encoderFactory) {
>-    native_encoder_factory.reset(new webrtc::ObjCVideoEncoderFactory(encoderFactory));
>+    native_encoder_factory = webrtc::ObjCToNativeVideoEncoderFactory(encoderFactory);
>   }
>   if (decoderFactory) {
>-    native_decoder_factory.reset(new webrtc::ObjCVideoDecoderFactory(decoderFactory));
>+    native_decoder_factory = webrtc::ObjCToNativeVideoDecoderFactory(decoderFactory);
>   }
>   return [self initWithNativeAudioEncoderFactory:webrtc::CreateBuiltinAudioEncoderFactory()
>                        nativeAudioDecoderFactory:webrtc::CreateBuiltinAudioDecoderFactory()
>                        nativeVideoEncoderFactory:std::move(native_encoder_factory)
>                        nativeVideoDecoderFactory:std::move(native_decoder_factory)
>-                               audioDeviceModule:nullptr
>+                               audioDeviceModule:[self audioDeviceModule]
>                            audioProcessingModule:nullptr];
> #endif
> }
>@@ -93,14 +107,17 @@ - (instancetype)initWithEncoderFactory:(nullable id<RTCVideoEncoderFactory>)enco
> - (instancetype)initNative {
>   if (self = [super init]) {
>     _networkThread = rtc::Thread::CreateWithSocketServer();
>+    _networkThread->SetName("network_thread", _networkThread.get());
>     BOOL result = _networkThread->Start();
>     NSAssert(result, @"Failed to start network thread.");
> 
>     _workerThread = rtc::Thread::Create();
>+    _workerThread->SetName("worker_thread", _workerThread.get());
>     result = _workerThread->Start();
>     NSAssert(result, @"Failed to start worker thread.");
> 
>     _signalingThread = rtc::Thread::Create();
>+    _signalingThread->SetName("signaling_thread", _signalingThread.get());
>     result = _signalingThread->Start();
>     NSAssert(result, @"Failed to start signaling thread.");
>   }
>@@ -153,40 +170,17 @@ - (instancetype)initWithNativeAudioEncoderFactory:
> #endif
> }
> 
>-- (instancetype)
>-    initWithNativeAudioEncoderFactory:
>-        (rtc::scoped_refptr<webrtc::AudioEncoderFactory>)audioEncoderFactory
>-            nativeAudioDecoderFactory:
>-                (rtc::scoped_refptr<webrtc::AudioDecoderFactory>)audioDecoderFactory
>-      legacyNativeVideoEncoderFactory:(cricket::WebRtcVideoEncoderFactory *)videoEncoderFactory
>-      legacyNativeVideoDecoderFactory:(cricket::WebRtcVideoDecoderFactory *)videoDecoderFactory
>-                    audioDeviceModule:(nullable webrtc::AudioDeviceModule *)audioDeviceModule {
>-#ifdef HAVE_NO_MEDIA
>-  return [self initWithNoMedia];
>-#else
>-  if (self = [self initNative]) {
>-    _nativeFactory = webrtc::CreatePeerConnectionFactory(_networkThread.get(),
>-                                                         _workerThread.get(),
>-                                                         _signalingThread.get(),
>-                                                         audioDeviceModule,
>-                                                         audioEncoderFactory,
>-                                                         audioDecoderFactory,
>-                                                         videoEncoderFactory,
>-                                                         videoDecoderFactory);
>-    NSAssert(_nativeFactory, @"Failed to initialize PeerConnectionFactory!");
>-  }
>-  return self;
>-#endif
>-}
>-
> - (RTCAudioSource *)audioSourceWithConstraints:(nullable RTCMediaConstraints *)constraints {
>   std::unique_ptr<webrtc::MediaConstraints> nativeConstraints;
>   if (constraints) {
>     nativeConstraints = constraints.nativeConstraints;
>   }
>+  cricket::AudioOptions options;
>+  CopyConstraintsIntoAudioOptions(nativeConstraints.get(), &options);
>+
>   rtc::scoped_refptr<webrtc::AudioSourceInterface> source =
>-      _nativeFactory->CreateAudioSource(nativeConstraints.get());
>-  return [[RTCAudioSource alloc] initWithNativeAudioSource:source];
>+      _nativeFactory->CreateAudioSource(options);
>+  return [[RTCAudioSource alloc] initWithFactory:self nativeAudioSource:source];
> }
> 
> - (RTCAudioTrack *)audioTrackWithTrackId:(NSString *)trackId {
>@@ -201,22 +195,10 @@ - (RTCAudioTrack *)audioTrackWithSource:(RTCAudioSource *)source
>                                         trackId:trackId];
> }
> 
>-- (RTCAVFoundationVideoSource *)avFoundationVideoSourceWithConstraints:
>-    (nullable RTCMediaConstraints *)constraints {
>-#ifdef HAVE_NO_MEDIA
>-  return nil;
>-#else
>-  return [[RTCAVFoundationVideoSource alloc] initWithFactory:self constraints:constraints];
>-#endif
>-}
>-
> - (RTCVideoSource *)videoSource {
>-  rtc::scoped_refptr<webrtc::ObjcVideoTrackSource> objcVideoTrackSource(
>-      new rtc::RefCountedObject<webrtc::ObjcVideoTrackSource>());
>-  return [[RTCVideoSource alloc]
>-      initWithNativeVideoSource:webrtc::VideoTrackSourceProxy::Create(_signalingThread.get(),
>-                                                                      _workerThread.get(),
>-                                                                      objcVideoTrackSource)];
>+  return [[RTCVideoSource alloc] initWithFactory:self
>+                                 signalingThread:_signalingThread.get()
>+                                    workerThread:_workerThread.get()];
> }
> 
> - (RTCVideoTrack *)videoTrackWithSource:(RTCVideoSource *)source
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactoryBuilder+DefaultComponents.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactoryBuilder+DefaultComponents.h
>new file mode 100644
>index 00000000000..070a0e74a56
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactoryBuilder+DefaultComponents.h
>@@ -0,0 +1,21 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#import "RTCPeerConnectionFactoryBuilder.h"
>+
>+NS_ASSUME_NONNULL_BEGIN
>+
>+@interface RTCPeerConnectionFactoryBuilder (DefaultComponents)
>+
>++ (RTCPeerConnectionFactoryBuilder *)defaultBuilder;
>+
>+@end
>+
>+NS_ASSUME_NONNULL_END
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactoryBuilder+DefaultComponents.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactoryBuilder+DefaultComponents.mm
>new file mode 100644
>index 00000000000..9fb50de101c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactoryBuilder+DefaultComponents.mm
>@@ -0,0 +1,48 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#import "RTCPeerConnectionFactory+Native.h"
>+#import "RTCPeerConnectionFactoryBuilder+DefaultComponents.h"
>+
>+#import "WebRTC/RTCVideoCodecH264.h"
>+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
>+#include "api/audio_codecs/builtin_audio_encoder_factory.h"
>+#include "sdk/objc/Framework/Native/api/video_decoder_factory.h"
>+#include "sdk/objc/Framework/Native/api/video_encoder_factory.h"
>+
>+#if defined(WEBRTC_IOS)
>+#import "sdk/objc/Framework/Native/api/audio_device_module.h"
>+#endif
>+
>+@implementation RTCPeerConnectionFactoryBuilder (DefaultComponents)
>+
>++ (RTCPeerConnectionFactoryBuilder *)defaultBuilder {
>+  RTCPeerConnectionFactoryBuilder *builder = [[RTCPeerConnectionFactoryBuilder alloc] init];
>+  auto audioEncoderFactory = webrtc::CreateBuiltinAudioEncoderFactory();
>+  [builder setAudioEncoderFactory:audioEncoderFactory];
>+
>+  auto audioDecoderFactory = webrtc::CreateBuiltinAudioDecoderFactory();
>+  [builder setAudioDecoderFactory:audioDecoderFactory];
>+
>+  auto videoEncoderFactory =
>+      webrtc::ObjCToNativeVideoEncoderFactory([[RTCVideoEncoderFactoryH264 alloc] init]);
>+  [builder setVideoEncoderFactory:std::move(videoEncoderFactory)];
>+
>+  auto videoDecoderFactory =
>+      webrtc::ObjCToNativeVideoDecoderFactory([[RTCVideoDecoderFactoryH264 alloc] init]);
>+  [builder setVideoDecoderFactory:std::move(videoDecoderFactory)];
>+
>+#if defined(WEBRTC_IOS)
>+  [builder setAudioDeviceModule:webrtc::CreateAudioDeviceModule()];
>+#endif
>+  return builder;
>+}
>+
>+@end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactoryBuilder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactoryBuilder.h
>new file mode 100644
>index 00000000000..27bdcd9f56b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactoryBuilder.h
>@@ -0,0 +1,48 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#import "WebRTC/RTCPeerConnectionFactory.h"
>+
>+#include "rtc_base/scoped_ref_ptr.h"
>+
>+namespace webrtc {
>+
>+class AudioDeviceModule;
>+class AudioEncoderFactory;
>+class AudioDecoderFactory;
>+class VideoEncoderFactory;
>+class VideoDecoderFactory;
>+class AudioProcessing;
>+
>+}  // namespace webrtc
>+
>+NS_ASSUME_NONNULL_BEGIN
>+
>+@interface RTCPeerConnectionFactoryBuilder : NSObject
>+
>++ (RTCPeerConnectionFactoryBuilder *)builder;
>+
>+- (RTCPeerConnectionFactory *)createPeerConnectionFactory;
>+
>+- (void)setVideoEncoderFactory:(std::unique_ptr<webrtc::VideoEncoderFactory>)videoEncoderFactory;
>+
>+- (void)setVideoDecoderFactory:(std::unique_ptr<webrtc::VideoDecoderFactory>)videoDecoderFactory;
>+
>+- (void)setAudioEncoderFactory:(rtc::scoped_refptr<webrtc::AudioEncoderFactory>)audioEncoderFactory;
>+
>+- (void)setAudioDecoderFactory:(rtc::scoped_refptr<webrtc::AudioDecoderFactory>)audioDecoderFactory;
>+
>+- (void)setAudioDeviceModule:(rtc::scoped_refptr<webrtc::AudioDeviceModule>)audioDeviceModule;
>+
>+- (void)setAudioProcessingModule:(rtc::scoped_refptr<webrtc::AudioProcessing>)audioProcessingModule;
>+
>+@end
>+
>+NS_ASSUME_NONNULL_END
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactoryBuilder.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactoryBuilder.mm
>new file mode 100644
>index 00000000000..a26a639e28e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactoryBuilder.mm
>@@ -0,0 +1,71 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#import "RTCPeerConnectionFactoryBuilder.h"
>+#import "RTCPeerConnectionFactory+Native.h"
>+
>+#include "api/audio_codecs/audio_decoder_factory.h"
>+#include "api/audio_codecs/audio_encoder_factory.h"
>+#include "api/video_codecs/video_decoder_factory.h"
>+#include "api/video_codecs/video_encoder_factory.h"
>+#include "modules/audio_device/include/audio_device.h"
>+#include "modules/audio_processing/include/audio_processing.h"
>+
>+@implementation RTCPeerConnectionFactoryBuilder {
>+  std::unique_ptr<webrtc::VideoEncoderFactory> _videoEncoderFactory;
>+  std::unique_ptr<webrtc::VideoDecoderFactory> _videoDecoderFactory;
>+  rtc::scoped_refptr<webrtc::AudioEncoderFactory> _audioEncoderFactory;
>+  rtc::scoped_refptr<webrtc::AudioDecoderFactory> _audioDecoderFactory;
>+  rtc::scoped_refptr<webrtc::AudioDeviceModule> _audioDeviceModule;
>+  rtc::scoped_refptr<webrtc::AudioProcessing> _audioProcessingModule;
>+}
>+
>++ (RTCPeerConnectionFactoryBuilder *)builder {
>+  return [[RTCPeerConnectionFactoryBuilder alloc] init];
>+}
>+
>+- (RTCPeerConnectionFactory *)createPeerConnectionFactory {
>+  RTCPeerConnectionFactory *factory = [RTCPeerConnectionFactory alloc];
>+  return [factory initWithNativeAudioEncoderFactory:_audioEncoderFactory
>+                          nativeAudioDecoderFactory:_audioDecoderFactory
>+                          nativeVideoEncoderFactory:std::move(_videoEncoderFactory)
>+                          nativeVideoDecoderFactory:std::move(_videoDecoderFactory)
>+                                  audioDeviceModule:_audioDeviceModule
>+                              audioProcessingModule:_audioProcessingModule];
>+}
>+
>+- (void)setVideoEncoderFactory:(std::unique_ptr<webrtc::VideoEncoderFactory>)videoEncoderFactory {
>+  _videoEncoderFactory = std::move(videoEncoderFactory);
>+}
>+
>+- (void)setVideoDecoderFactory:(std::unique_ptr<webrtc::VideoDecoderFactory>)videoDecoderFactory {
>+  _videoDecoderFactory = std::move(videoDecoderFactory);
>+}
>+
>+- (void)setAudioEncoderFactory:
>+        (rtc::scoped_refptr<webrtc::AudioEncoderFactory>)audioEncoderFactory {
>+  _audioEncoderFactory = audioEncoderFactory;
>+}
>+
>+- (void)setAudioDecoderFactory:
>+        (rtc::scoped_refptr<webrtc::AudioDecoderFactory>)audioDecoderFactory {
>+  _audioDecoderFactory = audioDecoderFactory;
>+}
>+
>+- (void)setAudioDeviceModule:(rtc::scoped_refptr<webrtc::AudioDeviceModule>)audioDeviceModule {
>+  _audioDeviceModule = audioDeviceModule;
>+}
>+
>+- (void)setAudioProcessingModule:
>+        (rtc::scoped_refptr<webrtc::AudioProcessing>)audioProcessingModule {
>+  _audioProcessingModule = audioProcessingModule;
>+}
>+
>+@end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactoryOptions.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactoryOptions.mm
>index f0cc6a6c81f..103a130390c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactoryOptions.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactoryOptions.mm
>@@ -34,6 +34,8 @@ @synthesize ignoreVPNNetworkAdapter = _ignoreVPNNetworkAdapter;
> @synthesize ignoreCellularNetworkAdapter = _ignoreCellularNetworkAdapter;
> @synthesize ignoreWiFiNetworkAdapter = _ignoreWiFiNetworkAdapter;
> @synthesize ignoreEthernetNetworkAdapter = _ignoreEthernetNetworkAdapter;
>+@synthesize enableAes128Sha1_32CryptoCipher = _enableAes128Sha1_32CryptoCipher;
>+@synthesize enableGcmCryptoSuites = _enableGcmCryptoSuites;
> 
> - (instancetype)init {
>   return [super init];
>@@ -50,6 +52,9 @@ - (webrtc::PeerConnectionFactoryInterface::Options)nativeOptions {
>   setNetworkBit(&options, rtc::ADAPTER_TYPE_WIFI, self.ignoreWiFiNetworkAdapter);
>   setNetworkBit(&options, rtc::ADAPTER_TYPE_ETHERNET, self.ignoreEthernetNetworkAdapter);
> 
>+  options.crypto_options.enable_aes128_sha1_32_crypto_cipher = self.enableAes128Sha1_32CryptoCipher;
>+  options.crypto_options.enable_gcm_crypto_suites = self.enableGcmCryptoSuites;
>+
>   return options;
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtcpParameters+Private.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtcpParameters+Private.h
>new file mode 100644
>index 00000000000..4157ffe7463
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtcpParameters+Private.h
>@@ -0,0 +1,27 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#import "WebRTC/RTCRtcpParameters.h"
>+
>+#include "api/rtpparameters.h"
>+
>+NS_ASSUME_NONNULL_BEGIN
>+
>+@interface RTCRtcpParameters ()
>+
>+/** Returns the equivalent native RtcpParameters structure. */
>+@property(nonatomic, readonly) webrtc::RtcpParameters nativeParameters;
>+
>+/** Initialize the object with a native RtcpParameters structure. */
>+- (instancetype)initWithNativeParameters:(const webrtc::RtcpParameters &)nativeParameters;
>+
>+@end
>+
>+NS_ASSUME_NONNULL_END
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtcpParameters.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtcpParameters.mm
>new file mode 100644
>index 00000000000..1c8a31bf207
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtcpParameters.mm
>@@ -0,0 +1,39 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#import "RTCRtcpParameters+Private.h"
>+
>+#import "NSString+StdString.h"
>+
>+@implementation RTCRtcpParameters
>+
>+@synthesize cname = _cname;
>+@synthesize isReducedSize = _isReducedSize;
>+
>+- (instancetype)init {
>+  return [super init];
>+}
>+
>+- (instancetype)initWithNativeParameters:(const webrtc::RtcpParameters &)nativeParameters {
>+  if (self = [self init]) {
>+    _cname = [NSString stringForStdString:nativeParameters.cname];
>+    _isReducedSize = nativeParameters.reduced_size;
>+  }
>+  return self;
>+}
>+
>+- (webrtc::RtcpParameters)nativeParameters {
>+  webrtc::RtcpParameters parameters;
>+  parameters.cname = [NSString stdStringForString:_cname];
>+  parameters.reduced_size = _isReducedSize;
>+  return parameters;
>+}
>+
>+@end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtpCodecParameters+Private.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtpCodecParameters+Private.h
>index f33a8cf1bed..c6ca462b1f3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtpCodecParameters+Private.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtpCodecParameters+Private.h
>@@ -20,8 +20,7 @@ NS_ASSUME_NONNULL_BEGIN
> @property(nonatomic, readonly) webrtc::RtpCodecParameters nativeParameters;
> 
> /** Initialize the object with a native RtpCodecParameters structure. */
>-- (instancetype)initWithNativeParameters:
>-    (const webrtc::RtpCodecParameters &)nativeParameters;
>+- (instancetype)initWithNativeParameters:(const webrtc::RtpCodecParameters &)nativeParameters;
> 
> @end
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtpCodecParameters.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtpCodecParameters.mm
>index 1a3bcf4dac7..7951cee45d5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtpCodecParameters.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtpCodecParameters.mm
>@@ -41,6 +41,7 @@ @synthesize name = _name;
> @synthesize kind = _kind;
> @synthesize clockRate = _clockRate;
> @synthesize numChannels = _numChannels;
>+@synthesize parameters = _parameters;
> 
> - (instancetype)init {
>   return [super init];
>@@ -68,6 +69,12 @@ - (instancetype)initWithNativeParameters:
>     if (nativeParameters.num_channels) {
>       _numChannels = [NSNumber numberWithInt:*nativeParameters.num_channels];
>     }
>+    NSMutableDictionary *parameters = [NSMutableDictionary dictionary];
>+    for (const auto &parameter : nativeParameters.parameters) {
>+      [parameters setObject:[NSString stringForStdString:parameter.second]
>+                     forKey:[NSString stringForStdString:parameter.first]];
>+    }
>+    _parameters = parameters;
>   }
>   return self;
> }
>@@ -86,10 +93,15 @@ - (webrtc::RtpCodecParameters)nativeParameters {
>     RTC_NOTREACHED();
>   }
>   if (_clockRate != nil) {
>-    parameters.clock_rate = rtc::Optional<int>(_clockRate.intValue);
>+    parameters.clock_rate = absl::optional<int>(_clockRate.intValue);
>   }
>   if (_numChannels != nil) {
>-    parameters.num_channels = rtc::Optional<int>(_numChannels.intValue);
>+    parameters.num_channels = absl::optional<int>(_numChannels.intValue);
>+  }
>+  for (NSString *paramKey in _parameters.allKeys) {
>+    std::string key = [NSString stdStringForString:paramKey];
>+    std::string value = [NSString stdStringForString:_parameters[paramKey]];
>+    parameters.parameters[key] = value;
>   }
>   return parameters;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtpEncodingParameters+Private.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtpEncodingParameters+Private.h
>index 673a485689a..c64501f062b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtpEncodingParameters+Private.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtpEncodingParameters+Private.h
>@@ -20,8 +20,7 @@ NS_ASSUME_NONNULL_BEGIN
> @property(nonatomic, readonly) webrtc::RtpEncodingParameters nativeParameters;
> 
> /** Initialize the object with a native RtpEncodingParameters structure. */
>-- (instancetype)initWithNativeParameters:
>-    (const webrtc::RtpEncodingParameters &)nativeParameters;
>+- (instancetype)initWithNativeParameters:(const webrtc::RtpEncodingParameters &)nativeParameters;
> 
> @end
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtpEncodingParameters.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtpEncodingParameters.mm
>index 8e72d18fca6..299e318bdcf 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtpEncodingParameters.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtpEncodingParameters.mm
>@@ -14,6 +14,7 @@ @implementation RTCRtpEncodingParameters
> 
> @synthesize isActive = _isActive;
> @synthesize maxBitrateBps = _maxBitrateBps;
>+@synthesize minBitrateBps = _minBitrateBps;
> @synthesize ssrc = _ssrc;
> 
> - (instancetype)init {
>@@ -28,6 +29,10 @@ - (instancetype)initWithNativeParameters:
>       _maxBitrateBps =
>           [NSNumber numberWithInt:*nativeParameters.max_bitrate_bps];
>     }
>+    if (nativeParameters.min_bitrate_bps) {
>+      _minBitrateBps =
>+          [NSNumber numberWithInt:*nativeParameters.min_bitrate_bps];
>+    }
>     if (nativeParameters.ssrc) {
>       _ssrc = [NSNumber numberWithUnsignedLong:*nativeParameters.ssrc];
>     }
>@@ -39,10 +44,13 @@ - (webrtc::RtpEncodingParameters)nativeParameters {
>   webrtc::RtpEncodingParameters parameters;
>   parameters.active = _isActive;
>   if (_maxBitrateBps != nil) {
>-    parameters.max_bitrate_bps = rtc::Optional<int>(_maxBitrateBps.intValue);
>+    parameters.max_bitrate_bps = absl::optional<int>(_maxBitrateBps.intValue);
>+  }
>+  if (_minBitrateBps != nil) {
>+    parameters.min_bitrate_bps = absl::optional<int>(_minBitrateBps.intValue);
>   }
>   if (_ssrc != nil) {
>-    parameters.ssrc = rtc::Optional<uint32_t>(_ssrc.unsignedLongValue);
>+    parameters.ssrc = absl::optional<uint32_t>(_ssrc.unsignedLongValue);
>   }
>   return parameters;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtpHeaderExtension+Private.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtpHeaderExtension+Private.h
>new file mode 100644
>index 00000000000..0bc236b5942
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtpHeaderExtension+Private.h
>@@ -0,0 +1,27 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#import "WebRTC/RTCRtpHeaderExtension.h"
>+
>+#include "api/rtpparameters.h"
>+
>+NS_ASSUME_NONNULL_BEGIN
>+
>+@interface RTCRtpHeaderExtension ()
>+
>+/** Returns the equivalent native RtpExtension structure. */
>+@property(nonatomic, readonly) webrtc::RtpExtension nativeParameters;
>+
>+/** Initialize the object with a native RtpExtension structure. */
>+- (instancetype)initWithNativeParameters:(const webrtc::RtpExtension &)nativeParameters;
>+
>+@end
>+
>+NS_ASSUME_NONNULL_END
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtpHeaderExtension.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtpHeaderExtension.mm
>new file mode 100644
>index 00000000000..3f448d73fb0
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtpHeaderExtension.mm
>@@ -0,0 +1,42 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#import "RTCRtpHeaderExtension+Private.h"
>+
>+#import "NSString+StdString.h"
>+
>+@implementation RTCRtpHeaderExtension
>+
>+@synthesize uri = _uri;
>+@synthesize id = _id;
>+@synthesize encrypted = _encrypted;
>+
>+- (instancetype)init {
>+  return [super init];
>+}
>+
>+- (instancetype)initWithNativeParameters:(const webrtc::RtpExtension &)nativeParameters {
>+  if (self = [self init]) {
>+    _uri = [NSString stringForStdString:nativeParameters.uri];
>+    _id = nativeParameters.id;
>+    _encrypted = nativeParameters.encrypt;
>+  }
>+  return self;
>+}
>+
>+- (webrtc::RtpExtension)nativeParameters {
>+  webrtc::RtpExtension extension;
>+  extension.uri = [NSString stdStringForString:_uri];
>+  extension.id = _id;
>+  extension.encrypt = _encrypted;
>+  return extension;
>+}
>+
>+@end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtpParameters+Private.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtpParameters+Private.h
>index 4a0daaf88e0..41d1b7c6015 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtpParameters+Private.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtpParameters+Private.h
>@@ -20,8 +20,7 @@ NS_ASSUME_NONNULL_BEGIN
> @property(nonatomic, readonly) webrtc::RtpParameters nativeParameters;
> 
> /** Initialize the object with a native RtpParameters structure. */
>-- (instancetype)initWithNativeParameters:
>-    (const webrtc::RtpParameters &)nativeParameters;
>+- (instancetype)initWithNativeParameters:(const webrtc::RtpParameters &)nativeParameters;
> 
> @end
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtpParameters.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtpParameters.mm
>index 5e791066df1..97800ca6395 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtpParameters.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtpParameters.mm
>@@ -10,11 +10,17 @@
> 
> #import "RTCRtpParameters+Private.h"
> 
>+#import "NSString+StdString.h"
>+#import "RTCRtcpParameters+Private.h"
> #import "RTCRtpCodecParameters+Private.h"
> #import "RTCRtpEncodingParameters+Private.h"
>+#import "RTCRtpHeaderExtension+Private.h"
> 
> @implementation RTCRtpParameters
> 
>+@synthesize transactionId = _transactionId;
>+@synthesize rtcp = _rtcp;
>+@synthesize headerExtensions = _headerExtensions;
> @synthesize encodings = _encodings;
> @synthesize codecs = _codecs;
> 
>@@ -25,6 +31,16 @@ - (instancetype)init {
> - (instancetype)initWithNativeParameters:
>     (const webrtc::RtpParameters &)nativeParameters {
>   if (self = [self init]) {
>+    _transactionId = [NSString stringForStdString:nativeParameters.transaction_id];
>+    _rtcp = [[RTCRtcpParameters alloc] initWithNativeParameters:nativeParameters.rtcp];
>+
>+    NSMutableArray *headerExtensions = [[NSMutableArray alloc] init];
>+    for (const auto &headerExtension : nativeParameters.header_extensions) {
>+      [headerExtensions
>+          addObject:[[RTCRtpHeaderExtension alloc] initWithNativeParameters:headerExtension]];
>+    }
>+    _headerExtensions = headerExtensions;
>+
>     NSMutableArray *encodings = [[NSMutableArray alloc] init];
>     for (const auto &encoding : nativeParameters.encodings) {
>       [encodings addObject:[[RTCRtpEncodingParameters alloc]
>@@ -43,7 +59,12 @@ - (instancetype)initWithNativeParameters:
> }
> 
> - (webrtc::RtpParameters)nativeParameters {
>-    webrtc::RtpParameters parameters;
>+  webrtc::RtpParameters parameters;
>+  parameters.transaction_id = [NSString stdStringForString:_transactionId];
>+  parameters.rtcp = [_rtcp nativeParameters];
>+  for (RTCRtpHeaderExtension *headerExtension in _headerExtensions) {
>+    parameters.header_extensions.push_back(headerExtension.nativeParameters);
>+  }
>   for (RTCRtpEncodingParameters *encoding in _encodings) {
>     parameters.encodings.push_back(encoding.nativeParameters);
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtpReceiver+Private.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtpReceiver+Private.h
>index 98738114f3b..6ec49bede45 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtpReceiver+Private.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtpReceiver+Private.h
>@@ -14,6 +14,8 @@
> 
> NS_ASSUME_NONNULL_BEGIN
> 
>+@class RTCPeerConnectionFactory;
>+
> namespace webrtc {
> 
> class RtpReceiverDelegateAdapter : public RtpReceiverObserverInterface {
>@@ -30,16 +32,19 @@ class RtpReceiverDelegateAdapter : public RtpReceiverObserverInterface {
> 
> @interface RTCRtpReceiver ()
> 
>-@property(nonatomic, readonly)
>-    rtc::scoped_refptr<webrtc::RtpReceiverInterface> nativeRtpReceiver;
>+@property(nonatomic, readonly) rtc::scoped_refptr<webrtc::RtpReceiverInterface> nativeRtpReceiver;
> 
> /** Initialize an RTCRtpReceiver with a native RtpReceiverInterface. */
>-- (instancetype)initWithNativeRtpReceiver:
>-    (rtc::scoped_refptr<webrtc::RtpReceiverInterface>)nativeRtpReceiver
>+- (instancetype)initWithFactory:(RTCPeerConnectionFactory*)factory
>+              nativeRtpReceiver:(rtc::scoped_refptr<webrtc::RtpReceiverInterface>)nativeRtpReceiver
>     NS_DESIGNATED_INITIALIZER;
> 
> + (RTCRtpMediaType)mediaTypeForNativeMediaType:(cricket::MediaType)nativeMediaType;
> 
>++ (cricket::MediaType)nativeMediaTypeForMediaType:(RTCRtpMediaType)mediaType;
>+
>++ (NSString*)stringForMediaType:(RTCRtpMediaType)mediaType;
>+
> @end
> 
> NS_ASSUME_NONNULL_END
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtpReceiver.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtpReceiver.mm
>index cb2771a29cd..895c451a044 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtpReceiver.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtpReceiver.mm
>@@ -36,6 +36,7 @@ void RtpReceiverDelegateAdapter::OnFirstPacketReceived(
> }  // namespace webrtc
> 
> @implementation RTCRtpReceiver {
>+  RTCPeerConnectionFactory *_factory;
>   rtc::scoped_refptr<webrtc::RtpReceiverInterface> _nativeRtpReceiver;
>   std::unique_ptr<webrtc::RtpReceiverDelegateAdapter> _observer;
> }
>@@ -62,7 +63,7 @@ - (nullable RTCMediaStreamTrack *)track {
>   rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> nativeTrack(
>     _nativeRtpReceiver->track());
>   if (nativeTrack) {
>-    return [[RTCMediaStreamTrack alloc] initWithNativeTrack:nativeTrack];
>+    return [RTCMediaStreamTrack mediaTrackForNativeTrack:nativeTrack factory:_factory];
>   }
>   return nil;
> }
>@@ -72,6 +73,12 @@ - (NSString *)description {
>       self.receiverId];
> }
> 
>+- (void)dealloc {
>+  if (_nativeRtpReceiver) {
>+    _nativeRtpReceiver->SetObserver(nullptr);
>+  }
>+}
>+
> - (BOOL)isEqual:(id)object {
>   if (self == object) {
>     return YES;
>@@ -96,9 +103,11 @@ - (rtc::scoped_refptr<webrtc::RtpReceiverInterface>)nativeRtpReceiver {
>   return _nativeRtpReceiver;
> }
> 
>-- (instancetype)initWithNativeRtpReceiver:
>-    (rtc::scoped_refptr<webrtc::RtpReceiverInterface>)nativeRtpReceiver {
>+- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
>+              nativeRtpReceiver:
>+                  (rtc::scoped_refptr<webrtc::RtpReceiverInterface>)nativeRtpReceiver {
>   if (self = [super init]) {
>+    _factory = factory;
>     _nativeRtpReceiver = nativeRtpReceiver;
>     RTCLogInfo(
>         @"RTCRtpReceiver(%p): created receiver: %@", self, self.description);
>@@ -120,4 +129,26 @@ + (RTCRtpMediaType)mediaTypeForNativeMediaType:
>   }
> }
> 
>++ (cricket::MediaType)nativeMediaTypeForMediaType:(RTCRtpMediaType)mediaType {
>+  switch (mediaType) {
>+    case RTCRtpMediaTypeAudio:
>+      return cricket::MEDIA_TYPE_AUDIO;
>+    case RTCRtpMediaTypeVideo:
>+      return cricket::MEDIA_TYPE_VIDEO;
>+    case RTCRtpMediaTypeData:
>+      return cricket::MEDIA_TYPE_DATA;
>+  }
>+}
>+
>++ (NSString *)stringForMediaType:(RTCRtpMediaType)mediaType {
>+  switch (mediaType) {
>+    case RTCRtpMediaTypeAudio:
>+      return @"AUDIO";
>+    case RTCRtpMediaTypeVideo:
>+      return @"VIDEO";
>+    case RTCRtpMediaTypeData:
>+      return @"DATA";
>+  }
>+}
>+
> @end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtpSender+Private.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtpSender+Private.h
>index e70be8349e3..5b671ae7c7a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtpSender+Private.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtpSender+Private.h
>@@ -14,14 +14,15 @@
> 
> NS_ASSUME_NONNULL_BEGIN
> 
>+@class RTCPeerConnectionFactory;
>+
> @interface RTCRtpSender ()
> 
>-@property(nonatomic, readonly)
>-    rtc::scoped_refptr<webrtc::RtpSenderInterface> nativeRtpSender;
>+@property(nonatomic, readonly) rtc::scoped_refptr<webrtc::RtpSenderInterface> nativeRtpSender;
> 
> /** Initialize an RTCRtpSender with a native RtpSenderInterface. */
>-- (instancetype)initWithNativeRtpSender:
>-    (rtc::scoped_refptr<webrtc::RtpSenderInterface>)nativeRtpSender
>+- (instancetype)initWithFactory:(RTCPeerConnectionFactory*)factory
>+                nativeRtpSender:(rtc::scoped_refptr<webrtc::RtpSenderInterface>)nativeRtpSender
>     NS_DESIGNATED_INITIALIZER;
> 
> @end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtpSender.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtpSender.mm
>index de3d69c958a..6a46edfef3c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtpSender.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtpSender.mm
>@@ -19,6 +19,7 @@
> #include "api/mediastreaminterface.h"
> 
> @implementation RTCRtpSender {
>+  RTCPeerConnectionFactory *_factory;
>   rtc::scoped_refptr<webrtc::RtpSenderInterface> _nativeRtpSender;
> }
> 
>@@ -34,7 +35,7 @@ - (RTCRtpParameters *)parameters {
> }
> 
> - (void)setParameters:(RTCRtpParameters *)parameters {
>-  if (!_nativeRtpSender->SetParameters(parameters.nativeParameters)) {
>+  if (!_nativeRtpSender->SetParameters(parameters.nativeParameters).ok()) {
>     RTCLogError(@"RTCRtpSender(%p): Failed to set parameters: %@", self,
>         parameters);
>   }
>@@ -44,7 +45,7 @@ - (RTCMediaStreamTrack *)track {
>   rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> nativeTrack(
>     _nativeRtpSender->track());
>   if (nativeTrack) {
>-    return [[RTCMediaStreamTrack alloc] initWithNativeTrack:nativeTrack];
>+    return [RTCMediaStreamTrack mediaTrackForNativeTrack:nativeTrack factory:_factory];
>   }
>   return nil;
> }
>@@ -84,10 +85,12 @@ - (rtc::scoped_refptr<webrtc::RtpSenderInterface>)nativeRtpSender {
>   return _nativeRtpSender;
> }
> 
>-- (instancetype)initWithNativeRtpSender:
>-    (rtc::scoped_refptr<webrtc::RtpSenderInterface>)nativeRtpSender {
>+- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
>+                nativeRtpSender:(rtc::scoped_refptr<webrtc::RtpSenderInterface>)nativeRtpSender {
>+  NSParameterAssert(factory);
>   NSParameterAssert(nativeRtpSender);
>   if (self = [super init]) {
>+    _factory = factory;
>     _nativeRtpSender = nativeRtpSender;
>     rtc::scoped_refptr<webrtc::DtmfSenderInterface> nativeDtmfSender(
>         _nativeRtpSender->GetDtmfSender());
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtpTransceiver+Private.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtpTransceiver+Private.h
>new file mode 100644
>index 00000000000..ee361ea8bf7
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtpTransceiver+Private.h
>@@ -0,0 +1,44 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#import "WebRTC/RTCRtpTransceiver.h"
>+
>+#include "api/rtptransceiverinterface.h"
>+
>+NS_ASSUME_NONNULL_BEGIN
>+
>+@class RTCPeerConnectionFactory;
>+
>+@interface RTCRtpTransceiverInit ()
>+
>+@property(nonatomic, readonly) webrtc::RtpTransceiverInit nativeInit;
>+
>+@end
>+
>+@interface RTCRtpTransceiver ()
>+
>+@property(nonatomic, readonly) rtc::scoped_refptr<webrtc::RtpTransceiverInterface>
>+    nativeRtpTransceiver;
>+
>+/** Initialize an RTCRtpTransceiver with a native RtpTransceiverInterface. */
>+- (instancetype)initWithFactory:(RTCPeerConnectionFactory*)factory
>+           nativeRtpTransceiver:
>+               (rtc::scoped_refptr<webrtc::RtpTransceiverInterface>)nativeRtpTransceiver
>+    NS_DESIGNATED_INITIALIZER;
>+
>++ (webrtc::RtpTransceiverDirection)nativeRtpTransceiverDirectionFromDirection:
>+        (RTCRtpTransceiverDirection)direction;
>+
>++ (RTCRtpTransceiverDirection)rtpTransceiverDirectionFromNativeDirection:
>+        (webrtc::RtpTransceiverDirection)nativeDirection;
>+
>+@end
>+
>+NS_ASSUME_NONNULL_END
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtpTransceiver.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtpTransceiver.mm
>new file mode 100644
>index 00000000000..cf39d5beff1
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCRtpTransceiver.mm
>@@ -0,0 +1,169 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#import "RTCRtpTransceiver+Private.h"
>+
>+#import "NSString+StdString.h"
>+#import "RTCRtpEncodingParameters+Private.h"
>+#import "RTCRtpParameters+Private.h"
>+#import "RTCRtpReceiver+Private.h"
>+#import "RTCRtpSender+Private.h"
>+#import "WebRTC/RTCLogging.h"
>+
>+@implementation RTCRtpTransceiverInit
>+
>+@synthesize direction = _direction;
>+@synthesize streamIds = _streamIds;
>+@synthesize sendEncodings = _sendEncodings;
>+
>+- (instancetype)init {
>+  if (self = [super init]) {
>+    _direction = RTCRtpTransceiverDirectionSendRecv;
>+  }
>+  return self;
>+}
>+
>+- (webrtc::RtpTransceiverInit)nativeInit {
>+  webrtc::RtpTransceiverInit init;
>+  init.direction = [RTCRtpTransceiver nativeRtpTransceiverDirectionFromDirection:_direction];
>+  for (NSString *streamId in _streamIds) {
>+    init.stream_ids.push_back([streamId UTF8String]);
>+  }
>+  for (RTCRtpEncodingParameters *sendEncoding in _sendEncodings) {
>+    init.send_encodings.push_back(sendEncoding.nativeParameters);
>+  }
>+  return init;
>+}
>+
>+@end
>+
>+@implementation RTCRtpTransceiver {
>+  RTCPeerConnectionFactory *_factory;
>+  rtc::scoped_refptr<webrtc::RtpTransceiverInterface> _nativeRtpTransceiver;
>+}
>+
>+- (RTCRtpMediaType)mediaType {
>+  return [RTCRtpReceiver mediaTypeForNativeMediaType:_nativeRtpTransceiver->media_type()];
>+}
>+
>+- (NSString *)mid {
>+  if (_nativeRtpTransceiver->mid()) {
>+    return [NSString stringForStdString:*_nativeRtpTransceiver->mid()];
>+  } else {
>+    return nil;
>+  }
>+}
>+
>+@synthesize sender = _sender;
>+@synthesize receiver = _receiver;
>+
>+- (BOOL)isStopped {
>+  return _nativeRtpTransceiver->stopped();
>+}
>+
>+- (RTCRtpTransceiverDirection)direction {
>+  return [RTCRtpTransceiver
>+      rtpTransceiverDirectionFromNativeDirection:_nativeRtpTransceiver->direction()];
>+}
>+
>+- (void)setDirection:(RTCRtpTransceiverDirection)direction {
>+  _nativeRtpTransceiver->SetDirection(
>+      [RTCRtpTransceiver nativeRtpTransceiverDirectionFromDirection:direction]);
>+}
>+
>+- (BOOL)currentDirection:(RTCRtpTransceiverDirection *)currentDirectionOut {
>+  if (_nativeRtpTransceiver->current_direction()) {
>+    *currentDirectionOut = [RTCRtpTransceiver
>+        rtpTransceiverDirectionFromNativeDirection:*_nativeRtpTransceiver->current_direction()];
>+    return YES;
>+  } else {
>+    return NO;
>+  }
>+}
>+
>+- (void)stop {
>+  _nativeRtpTransceiver->Stop();
>+}
>+
>+- (NSString *)description {
>+  return [NSString
>+      stringWithFormat:@"RTCRtpTransceiver {\n  sender: %@\n  receiver: %@\n}", _sender, _receiver];
>+}
>+
>+- (BOOL)isEqual:(id)object {
>+  if (self == object) {
>+    return YES;
>+  }
>+  if (object == nil) {
>+    return NO;
>+  }
>+  if (![object isMemberOfClass:[self class]]) {
>+    return NO;
>+  }
>+  RTCRtpTransceiver *transceiver = (RTCRtpTransceiver *)object;
>+  return _nativeRtpTransceiver == transceiver.nativeRtpTransceiver;
>+}
>+
>+- (NSUInteger)hash {
>+  return (NSUInteger)_nativeRtpTransceiver.get();
>+}
>+
>+#pragma mark - Private
>+
>+- (rtc::scoped_refptr<webrtc::RtpTransceiverInterface>)nativeRtpTransceiver {
>+  return _nativeRtpTransceiver;
>+}
>+
>+- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
>+           nativeRtpTransceiver:
>+               (rtc::scoped_refptr<webrtc::RtpTransceiverInterface>)nativeRtpTransceiver {
>+  NSParameterAssert(factory);
>+  NSParameterAssert(nativeRtpTransceiver);
>+  if (self = [super init]) {
>+    _factory = factory;
>+    _nativeRtpTransceiver = nativeRtpTransceiver;
>+    _sender = [[RTCRtpSender alloc] initWithFactory:_factory
>+                                    nativeRtpSender:nativeRtpTransceiver->sender()];
>+    _receiver = [[RTCRtpReceiver alloc] initWithFactory:_factory
>+                                      nativeRtpReceiver:nativeRtpTransceiver->receiver()];
>+    RTCLogInfo(@"RTCRtpTransceiver(%p): created transceiver: %@", self, self.description);
>+  }
>+  return self;
>+}
>+
>++ (webrtc::RtpTransceiverDirection)nativeRtpTransceiverDirectionFromDirection:
>+        (RTCRtpTransceiverDirection)direction {
>+  switch (direction) {
>+    case RTCRtpTransceiverDirectionSendRecv:
>+      return webrtc::RtpTransceiverDirection::kSendRecv;
>+    case RTCRtpTransceiverDirectionSendOnly:
>+      return webrtc::RtpTransceiverDirection::kSendOnly;
>+    case RTCRtpTransceiverDirectionRecvOnly:
>+      return webrtc::RtpTransceiverDirection::kRecvOnly;
>+    case RTCRtpTransceiverDirectionInactive:
>+      return webrtc::RtpTransceiverDirection::kInactive;
>+  }
>+}
>+
>++ (RTCRtpTransceiverDirection)rtpTransceiverDirectionFromNativeDirection:
>+        (webrtc::RtpTransceiverDirection)nativeDirection {
>+  switch (nativeDirection) {
>+    case webrtc::RtpTransceiverDirection::kSendRecv:
>+      return RTCRtpTransceiverDirectionSendRecv;
>+    case webrtc::RtpTransceiverDirection::kSendOnly:
>+      return RTCRtpTransceiverDirectionSendOnly;
>+    case webrtc::RtpTransceiverDirection::kRecvOnly:
>+      return RTCRtpTransceiverDirectionRecvOnly;
>+    case webrtc::RtpTransceiverDirection::kInactive:
>+      return RTCRtpTransceiverDirectionInactive;
>+  }
>+}
>+
>+@end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCSSLAdapter.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCSSLAdapter.mm
>index a64ff49c56c..dfe54b5e20e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCSSLAdapter.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCSSLAdapter.mm
>@@ -13,13 +13,13 @@
> #include "rtc_base/checks.h"
> #include "rtc_base/ssladapter.h"
> 
>-BOOL RTCInitializeSSL() {
>+BOOL RTCInitializeSSL(void) {
>   BOOL initialized = rtc::InitializeSSL();
>   RTC_DCHECK(initialized);
>   return initialized;
> }
> 
>-BOOL RTCCleanupSSL() {
>+BOOL RTCCleanupSSL(void) {
>   BOOL cleanedUp = rtc::CleanupSSL();
>   RTC_DCHECK(cleanedUp);
>   return cleanedUp;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCSessionDescription+Private.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCSessionDescription+Private.h
>index 4c58fa726a0..08a6f55c58a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCSessionDescription+Private.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCSessionDescription+Private.h
>@@ -29,7 +29,7 @@ NS_ASSUME_NONNULL_BEGIN
>  * description.
>  */
> - (instancetype)initWithNativeDescription:
>-    (const webrtc::SessionDescriptionInterface *)nativeDescription;
>+        (const webrtc::SessionDescriptionInterface *)nativeDescription;
> 
> + (std::string)stdStringForType:(RTCSdpType)type;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCTracing.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCTracing.mm
>index f29c161d96e..7bcd6496e3a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCTracing.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCTracing.mm
>@@ -12,7 +12,7 @@
> 
> #include "rtc_base/event_tracer.h"
> 
>-void RTCSetupInternalTracer() {
>+void RTCSetupInternalTracer(void) {
>   rtc::tracing::SetupInternalTracer();
> }
> 
>@@ -20,10 +20,10 @@ BOOL RTCStartInternalCapture(NSString *filePath) {
>   return rtc::tracing::StartInternalCapture(filePath.UTF8String);
> }
> 
>-void RTCStopInternalCapture() {
>+void RTCStopInternalCapture(void) {
>   rtc::tracing::StopInternalCapture();
> }
> 
>-void RTCShutdownInternalTracer() {
>+void RTCShutdownInternalTracer(void) {
>   rtc::tracing::ShutdownInternalTracer();
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoCapturer.m b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoCapturer.m
>index 9d4d99dd7ce..3639fa75cac 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoCapturer.m
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoCapturer.m
>@@ -15,7 +15,6 @@ @implementation RTCVideoCapturer
> @synthesize delegate = _delegate;
> 
> - (instancetype)initWithDelegate:(id<RTCVideoCapturerDelegate>)delegate {
>-  NSAssert(delegate != nil, @"delegate cannot be nil");
>   if (self = [super init]) {
>     _delegate = delegate;
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoCodec+Private.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoCodec+Private.h
>index 97ee5138ad6..4a2cf5228bb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoCodec+Private.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoCodec+Private.h
>@@ -53,10 +53,6 @@ NS_ASSUME_NONNULL_BEGIN
> - (instancetype)initWithNativeSdpVideoFormat:(webrtc::SdpVideoFormat)format;
> - (webrtc::SdpVideoFormat)nativeSdpVideoFormat;
> 
>-/* TODO(andersc): These are deprecated, remove when no longer in use. */
>-- (instancetype)initWithNativeVideoCodec:(cricket::VideoCodec)videoCodec;
>-- (cricket::VideoCodec)nativeVideoCodec;
>-
> @end
> 
> NS_ASSUME_NONNULL_END
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoCodec.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoCodec.mm
>index c784ad0512b..d7521262d4d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoCodec.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoCodec.mm
>@@ -12,15 +12,70 @@
> 
> #import "NSString+StdString.h"
> #import "RTCVideoCodec+Private.h"
>+#if defined(WEBRTC_IOS)
>+#import "UIDevice+H264Profile.h"
>+#endif
> #import "WebRTC/RTCVideoCodecFactory.h"
> 
> #include "media/base/mediaconstants.h"
> 
>-NSString *const kRTCVideoCodecVp8Name = @"VP8";
>-NSString *const kRTCVideoCodecVp9Name = @"VP9";
>-NSString *const kRTCVideoCodecH264Name = @"H264";
>+namespace {
>+
>+NSString *MaxSupportedProfileLevelConstrainedHigh();
>+NSString *MaxSupportedProfileLevelConstrainedBaseline();
>+
>+}  // namespace
>+
>+NSString *const kRTCVideoCodecVp8Name = @(cricket::kVp8CodecName);
>+NSString *const kRTCVideoCodecVp9Name = @(cricket::kVp9CodecName);
>+NSString *const kRTCVideoCodecH264Name = @(cricket::kH264CodecName);
> NSString *const kRTCLevel31ConstrainedHigh = @"640c1f";
> NSString *const kRTCLevel31ConstrainedBaseline = @"42e01f";
>+NSString *const kRTCMaxSupportedH264ProfileLevelConstrainedHigh =
>+    MaxSupportedProfileLevelConstrainedHigh();
>+NSString *const kRTCMaxSupportedH264ProfileLevelConstrainedBaseline =
>+    MaxSupportedProfileLevelConstrainedBaseline();
>+
>+namespace {
>+
>+#if defined(WEBRTC_IOS)
>+
>+using namespace webrtc::H264;
>+
>+NSString *MaxSupportedLevelForProfile(Profile profile) {
>+  const absl::optional<ProfileLevelId> profileLevelId = [UIDevice maxSupportedH264Profile];
>+  if (profileLevelId && profileLevelId->profile >= profile) {
>+    const absl::optional<std::string> profileString =
>+        ProfileLevelIdToString(ProfileLevelId(profile, profileLevelId->level));
>+    if (profileString) {
>+      return [NSString stringForStdString:*profileString];
>+    }
>+  }
>+  return nil;
>+}
>+#endif
>+
>+NSString *MaxSupportedProfileLevelConstrainedBaseline() {
>+#if defined(WEBRTC_IOS)
>+  NSString *profile = MaxSupportedLevelForProfile(webrtc::H264::kProfileConstrainedBaseline);
>+  if (profile != nil) {
>+    return profile;
>+  }
>+#endif
>+  return kRTCLevel31ConstrainedBaseline;
>+}
>+
>+NSString *MaxSupportedProfileLevelConstrainedHigh() {
>+#if defined(WEBRTC_IOS)
>+  NSString *profile = MaxSupportedLevelForProfile(webrtc::H264::kProfileConstrainedHigh);
>+  if (profile != nil) {
>+    return profile;
>+  }
>+#endif
>+  return kRTCLevel31ConstrainedHigh;
>+}
>+
>+}  // namespace
> 
> @implementation RTCVideoCodecInfo
> 
>@@ -50,11 +105,6 @@ - (instancetype)initWithNativeSdpVideoFormat:(webrtc::SdpVideoFormat)format {
>   return [self initWithName:[NSString stringForStdString:format.name] parameters:params];
> }
> 
>-- (instancetype)initWithNativeVideoCodec:(cricket::VideoCodec)videoCodec {
>-  return [self
>-      initWithNativeSdpVideoFormat:webrtc::SdpVideoFormat(videoCodec.name, videoCodec.params)];
>-}
>-
> - (BOOL)isEqualToCodecInfo:(RTCVideoCodecInfo *)info {
>   if (!info ||
>       ![self.name isEqualToString:info.name] ||
>@@ -87,16 +137,6 @@ - (webrtc::SdpVideoFormat)nativeSdpVideoFormat {
>   return webrtc::SdpVideoFormat([NSString stdStringForString:_name], parameters);
> }
> 
>-- (cricket::VideoCodec)nativeVideoCodec {
>-  cricket::VideoCodec codec([NSString stdStringForString:_name]);
>-  for (NSString *paramKey in _parameters.allKeys) {
>-    codec.SetParam([NSString stdStringForString:paramKey],
>-                   [NSString stdStringForString:_parameters[paramKey]]);
>-  }
>-
>-  return codec;
>-}
>-
> #pragma mark - NSCoding
> 
> - (instancetype)initWithCoder:(NSCoder *)decoder {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoCodecH264.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoCodecH264.mm
>index f862453b5f8..51583d09eb6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoCodecH264.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoCodecH264.mm
>@@ -16,14 +16,6 @@
> #import "WebRTC/RTCVideoCodec.h"
> 
> #include "rtc_base/timeutils.h"
>-#include "sdk/objc/Framework/Classes/Video/objc_frame_buffer.h"
>-#include "system_wrappers/include/field_trial.h"
>-
>-const char kHighProfileExperiment[] = "WebRTC-H264HighProfile";
>-
>-bool IsHighProfileEnabled() {
>-  return webrtc::field_trial::IsEnabled(kHighProfileExperiment);
>-}
> 
> // H264 specific settings.
> @implementation RTCCodecSpecificInfoH264
>@@ -49,19 +41,17 @@ - (NSArray<RTCVideoCodecInfo *> *)supportedCodecs {
>   NSMutableArray<RTCVideoCodecInfo *> *codecs = [NSMutableArray array];
>   NSString *codecName = kRTCVideoCodecH264Name;
> 
>-  if (IsHighProfileEnabled()) {
>-    NSDictionary<NSString *, NSString *> *constrainedHighParams = @{
>-      @"profile-level-id" : kRTCLevel31ConstrainedHigh,
>-      @"level-asymmetry-allowed" : @"1",
>-      @"packetization-mode" : @"1",
>-    };
>-    RTCVideoCodecInfo *constrainedHighInfo =
>-        [[RTCVideoCodecInfo alloc] initWithName:codecName parameters:constrainedHighParams];
>-    [codecs addObject:constrainedHighInfo];
>-  }
>+  NSDictionary<NSString *, NSString *> *constrainedHighParams = @{
>+    @"profile-level-id" : kRTCMaxSupportedH264ProfileLevelConstrainedHigh,
>+    @"level-asymmetry-allowed" : @"1",
>+    @"packetization-mode" : @"1",
>+  };
>+  RTCVideoCodecInfo *constrainedHighInfo =
>+      [[RTCVideoCodecInfo alloc] initWithName:codecName parameters:constrainedHighParams];
>+  [codecs addObject:constrainedHighInfo];
> 
>   NSDictionary<NSString *, NSString *> *constrainedBaselineParams = @{
>-    @"profile-level-id" : kRTCLevel31ConstrainedBaseline,
>+    @"profile-level-id" : kRTCMaxSupportedH264ProfileLevelConstrainedBaseline,
>     @"level-asymmetry-allowed" : @"1",
>     @"packetization-mode" : @"1",
>   };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoFrame.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoFrame.mm
>index 19dd2452cd0..9a5d3065d69 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoFrame.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoFrame.mm
>@@ -8,20 +8,9 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#import "RTCVideoFrame+Private.h"
>-
> #import "WebRTC/RTCVideoFrame.h"
>-#import "WebRTC/RTCVideoFrameBuffer.h"
>-
>-#include "api/video/video_frame.h"
>-#include "rtc_base/timeutils.h"
> 
>-id<RTCVideoFrameBuffer> nativeToRtcFrameBuffer(
>-    const rtc::scoped_refptr<webrtc::VideoFrameBuffer> &buffer) {
>-  return buffer->type() == webrtc::VideoFrameBuffer::Type::kNative ?
>-      static_cast<webrtc::ObjCFrameBuffer *>(buffer.get())->wrapped_frame_buffer() :
>-      [[RTCI420Buffer alloc] initWithFrameBuffer:buffer->ToI420()];
>-}
>+#import "WebRTC/RTCVideoFrameBuffer.h"
> 
> @implementation RTCVideoFrame {
>   RTCVideoRotation _rotation;
>@@ -92,24 +81,4 @@ - (instancetype)initWithBuffer:(id<RTCVideoFrameBuffer>)buffer
>   return self;
> }
> 
>-- (instancetype)initWithNativeVideoFrame:(const webrtc::VideoFrame &)frame {
>-  if (self = [self initWithBuffer:nativeToRtcFrameBuffer(frame.video_frame_buffer())
>-                         rotation:RTCVideoRotation(frame.rotation())
>-                      timeStampNs:frame.timestamp_us() * rtc::kNumNanosecsPerMicrosec]) {
>-    self.timeStamp = frame.timestamp();
>-  }
>-
>-  return self;
>-}
>-
>-- (webrtc::VideoFrame)nativeVideoFrame {
>-  rtc::scoped_refptr<webrtc::VideoFrameBuffer> frameBuffer =
>-      new rtc::RefCountedObject<webrtc::ObjCFrameBuffer>(self.buffer);
>-  webrtc::VideoFrame videoFrame(frameBuffer,
>-                                (webrtc::VideoRotation)self.rotation,
>-                                self.timeStampNs / rtc::kNumNanosecsPerMicrosec);
>-  videoFrame.set_timestamp(self.timeStamp);
>-  return videoFrame;
>-}
>-
> @end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoRendererAdapter+Private.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoRendererAdapter+Private.h
>index bdec24d5b7f..80a2140bfa3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoRendererAdapter+Private.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoRendererAdapter+Private.h
>@@ -30,8 +30,7 @@ NS_ASSUME_NONNULL_BEGIN
>  * to this interface will be adapted and passed to the RTCVideoRenderer supplied
>  * during construction. This pointer is unsafe and owned by this class.
>  */
>-@property(nonatomic, readonly)
>-    rtc::VideoSinkInterface<webrtc::VideoFrame> *nativeVideoRenderer;
>+@property(nonatomic, readonly) rtc::VideoSinkInterface<webrtc::VideoFrame> *nativeVideoRenderer;
> 
> /** Initialize an RTCVideoRendererAdapter with an RTCVideoRenderer. */
> - (instancetype)initWithNativeRenderer:(id<RTCVideoRenderer>)videoRenderer
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoRendererAdapter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoRendererAdapter.h
>index 5b4570310d4..b0b6f044884 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoRendererAdapter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoRendererAdapter.h
>@@ -18,7 +18,6 @@ NS_ASSUME_NONNULL_BEGIN
>  * adapter adapts calls made to that interface to the RTCVideoRenderer supplied
>  * during construction.
>  */
>-__attribute__((objc_runtime_name("WK_RTCVideoRendererAdapter")))
> @interface RTCVideoRendererAdapter : NSObject
> 
> - (instancetype)init NS_UNAVAILABLE;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoRendererAdapter.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoRendererAdapter.mm
>index 79cc1a71dd7..f5c4f011f51 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoRendererAdapter.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoRendererAdapter.mm
>@@ -9,14 +9,14 @@
>  */
> 
> #import "RTCI420Buffer+Private.h"
>-#import "RTCVideoFrame+Private.h"
> #import "RTCVideoRendererAdapter+Private.h"
> #import "WebRTC/RTCVideoFrame.h"
> #import "WebRTC/RTCVideoFrameBuffer.h"
>-#import "objc_frame_buffer.h"
> 
> #include <memory>
> 
>+#include "sdk/objc/Framework/Native/api/video_frame.h"
>+
> namespace webrtc {
> 
> class VideoRendererAdapter
>@@ -28,7 +28,7 @@ class VideoRendererAdapter
>   }
> 
>   void OnFrame(const webrtc::VideoFrame& nativeVideoFrame) override {
>-    RTCVideoFrame* videoFrame = [[RTCVideoFrame alloc] initWithNativeVideoFrame:nativeVideoFrame];
>+    RTCVideoFrame* videoFrame = NativeToObjCVideoFrame(nativeVideoFrame);
> 
>     CGSize current_size = (videoFrame.rotation % 180 == 0)
>                               ? CGSizeMake(videoFrame.width, videoFrame.height)
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoSource+Private.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoSource+Private.h
>index 1f34e047c2b..5eea2f9896f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoSource+Private.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoSource+Private.h
>@@ -22,18 +22,22 @@ NS_ASSUME_NONNULL_BEGIN
>  * The VideoTrackSourceInterface object passed to this RTCVideoSource during
>  * construction.
>  */
>-@property(nonatomic, readonly)
>-    rtc::scoped_refptr<webrtc::VideoTrackSourceInterface>
>-        nativeVideoSource;
>+@property(nonatomic, readonly) rtc::scoped_refptr<webrtc::VideoTrackSourceInterface>
>+    nativeVideoSource;
> 
> /** Initialize an RTCVideoSource from a native VideoTrackSourceInterface. */
>-- (instancetype)initWithNativeVideoSource:
>-    (rtc::scoped_refptr<webrtc::VideoTrackSourceInterface>)nativeVideoSource
>+- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
>+              nativeVideoSource:
>+                  (rtc::scoped_refptr<webrtc::VideoTrackSourceInterface>)nativeVideoSource
>     NS_DESIGNATED_INITIALIZER;
> 
>-- (instancetype)initWithNativeMediaSource:
>-    (rtc::scoped_refptr<webrtc::MediaSourceInterface>)nativeMediaSource
>-                                     type:(RTCMediaSourceType)type NS_UNAVAILABLE;
>+- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
>+              nativeMediaSource:(rtc::scoped_refptr<webrtc::MediaSourceInterface>)nativeMediaSource
>+                           type:(RTCMediaSourceType)type NS_UNAVAILABLE;
>+
>+- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
>+                signalingThread:(rtc::Thread *)signalingThread
>+                   workerThread:(rtc::Thread *)workerThread;
> 
> @end
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoSource.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoSource.mm
>index fccf92e2e28..63b80144cf6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoSource.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoSource.mm
>@@ -12,50 +12,64 @@
> 
> #include "api/videosourceproxy.h"
> #include "rtc_base/checks.h"
>-#include "sdk/objc/Framework/Classes/Video/objcvideotracksource.h"
>+#include "sdk/objc/Framework/Native/src/objc_video_track_source.h"
> 
>-static webrtc::ObjcVideoTrackSource *getObjcVideoSource(
>+static webrtc::ObjCVideoTrackSource *getObjCVideoSource(
>     const rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> nativeSource) {
>   webrtc::VideoTrackSourceProxy *proxy_source =
>       static_cast<webrtc::VideoTrackSourceProxy *>(nativeSource.get());
>-  return static_cast<webrtc::ObjcVideoTrackSource *>(proxy_source->internal());
>+  return static_cast<webrtc::ObjCVideoTrackSource *>(proxy_source->internal());
> }
> 
>-// TODO(magjed): Refactor this class and target ObjcVideoTrackSource only once
>+// TODO(magjed): Refactor this class and target ObjCVideoTrackSource only once
> // RTCAVFoundationVideoSource is gone. See http://crbug/webrtc/7177 for more
> // info.
> @implementation RTCVideoSource {
>   rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> _nativeVideoSource;
> }
> 
>-- (instancetype)initWithNativeVideoSource:
>-    (rtc::scoped_refptr<webrtc::VideoTrackSourceInterface>)nativeVideoSource {
>+- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
>+              nativeVideoSource:
>+                  (rtc::scoped_refptr<webrtc::VideoTrackSourceInterface>)nativeVideoSource {
>+  RTC_DCHECK(factory);
>   RTC_DCHECK(nativeVideoSource);
>-  if (self = [super initWithNativeMediaSource:nativeVideoSource
>-                                         type:RTCMediaSourceTypeVideo]) {
>+  if (self = [super initWithFactory:factory
>+                  nativeMediaSource:nativeVideoSource
>+                               type:RTCMediaSourceTypeVideo]) {
>     _nativeVideoSource = nativeVideoSource;
>   }
>   return self;
> }
> 
>-- (instancetype)initWithNativeMediaSource:
>-    (rtc::scoped_refptr<webrtc::MediaSourceInterface>)nativeMediaSource
>-                                     type:(RTCMediaSourceType)type {
>+- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
>+              nativeMediaSource:(rtc::scoped_refptr<webrtc::MediaSourceInterface>)nativeMediaSource
>+                           type:(RTCMediaSourceType)type {
>   RTC_NOTREACHED();
>   return nil;
> }
> 
>+- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
>+                signalingThread:(rtc::Thread *)signalingThread
>+                   workerThread:(rtc::Thread *)workerThread {
>+  rtc::scoped_refptr<webrtc::ObjCVideoTrackSource> objCVideoTrackSource(
>+      new rtc::RefCountedObject<webrtc::ObjCVideoTrackSource>());
>+
>+  return [self initWithFactory:factory
>+             nativeVideoSource:webrtc::VideoTrackSourceProxy::Create(
>+                                   signalingThread, workerThread, objCVideoTrackSource)];
>+}
>+
> - (NSString *)description {
>   NSString *stateString = [[self class] stringForState:self.state];
>   return [NSString stringWithFormat:@"RTCVideoSource( %p ): %@", self, stateString];
> }
> 
> - (void)capturer:(RTCVideoCapturer *)capturer didCaptureVideoFrame:(RTCVideoFrame *)frame {
>-  getObjcVideoSource(_nativeVideoSource)->OnCapturedFrame(frame);
>+  getObjCVideoSource(_nativeVideoSource)->OnCapturedFrame(frame);
> }
> 
> - (void)adaptOutputFormatToWidth:(int)width height:(int)height fps:(int)fps {
>-  getObjcVideoSource(_nativeVideoSource)->OnOutputFormatRequest(width, height, fps);
>+  getObjCVideoSource(_nativeVideoSource)->OnOutputFormatRequest(width, height, fps);
> }
> 
> #pragma mark - Private
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoTrack+Private.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoTrack+Private.h
>index 2cb6a13f259..14639be9fb5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoTrack+Private.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoTrack+Private.h
>@@ -17,8 +17,7 @@ NS_ASSUME_NONNULL_BEGIN
> @interface RTCVideoTrack ()
> 
> /** VideoTrackInterface created or passed in at construction. */
>-@property(nonatomic, readonly)
>-    rtc::scoped_refptr<webrtc::VideoTrackInterface> nativeVideoTrack;
>+@property(nonatomic, readonly) rtc::scoped_refptr<webrtc::VideoTrackInterface> nativeVideoTrack;
> 
> /** Initialize an RTCVideoTrack with its source and an id. */
> - (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoTrack.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoTrack.mm
>index 2f031107397..c9eb35c8757 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoTrack.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoTrack.mm
>@@ -32,18 +32,20 @@ - (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
>   rtc::scoped_refptr<webrtc::VideoTrackInterface> track =
>       factory.nativeFactory->CreateVideoTrack(nativeId,
>                                               source.nativeVideoSource);
>-  if (self = [self initWithNativeTrack:track type:RTCMediaStreamTrackTypeVideo]) {
>+  if (self = [self initWithFactory:factory nativeTrack:track type:RTCMediaStreamTrackTypeVideo]) {
>     _source = source;
>   }
>   return self;
> }
> 
>-- (instancetype)initWithNativeTrack:
>-    (rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeMediaTrack
>-                               type:(RTCMediaStreamTrackType)type {
>+- (instancetype)initWithFactory:(RTCPeerConnectionFactory *)factory
>+                    nativeTrack:
>+                        (rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>)nativeMediaTrack
>+                           type:(RTCMediaStreamTrackType)type {
>+  NSParameterAssert(factory);
>   NSParameterAssert(nativeMediaTrack);
>   NSParameterAssert(type == RTCMediaStreamTrackTypeVideo);
>-  if (self = [super initWithNativeTrack:nativeMediaTrack type:type]) {
>+  if (self = [super initWithFactory:factory nativeTrack:nativeMediaTrack type:type]) {
>     _adapters = [NSMutableArray array];
>   }
>   return self;
>@@ -60,7 +62,8 @@ - (RTCVideoSource *)source {
>     rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> source =
>         self.nativeVideoTrack->GetSource();
>     if (source) {
>-      _source = [[RTCVideoSource alloc] initWithNativeVideoSource:source.get()];
>+      _source =
>+          [[RTCVideoSource alloc] initWithFactory:self.factory nativeVideoSource:source.get()];
>     }
>   }
>   return _source;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCWrappedNativeVideoDecoder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCWrappedNativeVideoDecoder.h
>index cb29b692640..74cfaeb861f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCWrappedNativeVideoDecoder.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCWrappedNativeVideoDecoder.h
>@@ -14,7 +14,6 @@
> #include "api/video_codecs/video_decoder.h"
> #include "media/base/codec.h"
> 
>-__attribute__((objc_runtime_name("WK_RTCWrappedNativeVideoDecoder")))
> @interface RTCWrappedNativeVideoDecoder : NSObject <RTCVideoDecoder>
> 
> - (instancetype)initWithNativeDecoder:(std::unique_ptr<webrtc::VideoDecoder>)decoder;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCWrappedNativeVideoDecoder.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCWrappedNativeVideoDecoder.mm
>index 7529ebf0286..2d4d45e3785 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCWrappedNativeVideoDecoder.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCWrappedNativeVideoDecoder.mm
>@@ -35,6 +35,11 @@ - (void)setCallback:(RTCVideoDecoderCallback)callback {
>   RTC_NOTREACHED();
> }
> 
>+- (NSInteger)startDecodeWithNumberOfCores:(int)numberOfCores {
>+  RTC_NOTREACHED();
>+  return 0;
>+}
>+
> - (NSInteger)startDecodeWithSettings:(RTCVideoEncoderSettings *)settings
>                        numberOfCores:(int)numberOfCores {
>   RTC_NOTREACHED();
>@@ -47,10 +52,9 @@ - (NSInteger)releaseDecoder {
> }
> 
> - (NSInteger)decode:(RTCEncodedImage *)encodedImage
>-          missingFrames:(BOOL)missingFrames
>-    fragmentationHeader:(RTCRtpFragmentationHeader *)fragmentationHeader
>-      codecSpecificInfo:(nullable id<RTCCodecSpecificInfo>)info
>-           renderTimeMs:(int64_t)renderTimeMs {
>+        missingFrames:(BOOL)missingFrames
>+    codecSpecificInfo:(nullable id<RTCCodecSpecificInfo>)info
>+         renderTimeMs:(int64_t)renderTimeMs {
>   RTC_NOTREACHED();
>   return 0;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCWrappedNativeVideoEncoder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCWrappedNativeVideoEncoder.h
>index 29eff125a54..5b95d33eba9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCWrappedNativeVideoEncoder.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCWrappedNativeVideoEncoder.h
>@@ -15,7 +15,6 @@
> #include "api/video_codecs/video_encoder.h"
> #include "media/base/codec.h"
> 
>-__attribute__((objc_runtime_name("WK_RTCWrappedNativeVideoEncoder")))
> @interface RTCWrappedNativeVideoEncoder : NSObject <RTCVideoEncoder>
> 
> - (instancetype)initWithNativeEncoder:(std::unique_ptr<webrtc::VideoEncoder>)encoder;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/UI/RTCEAGLVideoView.m b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/UI/RTCEAGLVideoView.m
>index 8f4da302fab..a26741479fa 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/UI/RTCEAGLVideoView.m
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/UI/RTCEAGLVideoView.m
>@@ -23,7 +23,6 @@
> // refreshes, which should be 30fps. We wrap the display link in order to avoid
> // a retain cycle since CADisplayLink takes a strong reference onto its target.
> // The timer is paused by default.
>-__attribute__((objc_runtime_name("WK_RTCDisplayLinkTimer")))
> @interface RTCDisplayLinkTimer : NSObject
> 
> @property(nonatomic) BOOL isPaused;
>@@ -46,8 +45,11 @@ - (instancetype)initWithTimerHandler:(void (^)(void))timerHandler {
>         [CADisplayLink displayLinkWithTarget:self
>                                     selector:@selector(displayLinkDidFire:)];
>     _displayLink.paused = YES;
>-    // Set to half of screen refresh, which should be 30fps.
>+#if __IPHONE_OS_VERSION_MIN_REQUIRED >= __IPHONE_10_0
>+    _displayLink.preferredFramesPerSecond = 30;
>+#else
>     [_displayLink setFrameInterval:2];
>+#endif
>     [_displayLink addToRunLoop:[NSRunLoop currentRunLoop]
>                        forMode:NSRunLoopCommonModes];
>   }
>@@ -104,7 +106,9 @@ @implementation RTCEAGLVideoView {
>   id<RTCVideoViewShading> _shader;
>   RTCNV12TextureCache *_nv12TextureCache;
>   RTCI420TextureCache *_i420TextureCache;
>-  RTCVideoFrame *_lastDrawnFrame;
>+  // As timestamps should be unique between frames, will store last
>+  // drawn frame timestamp instead of the whole frame to reduce memory usage.
>+  int64_t _lastDrawnFrameTimeStampNs;
> }
> 
> @synthesize delegate = _delegate;
>@@ -230,7 +234,7 @@ - (void)glkView:(GLKView *)view drawInRect:(CGRect)rect {
>   // The renderer will draw the frame to the framebuffer corresponding to the
>   // one used by |view|.
>   RTCVideoFrame *frame = self.videoFrame;
>-  if (!frame || frame == _lastDrawnFrame) {
>+  if (!frame || frame.timeStampNs == _lastDrawnFrameTimeStampNs) {
>     return;
>   }
>   [self ensureGLContext];
>@@ -247,6 +251,8 @@ - (void)glkView:(GLKView *)view drawInRect:(CGRect)rect {
>                                       yPlane:_nv12TextureCache.yTexture
>                                      uvPlane:_nv12TextureCache.uvTexture];
>       [_nv12TextureCache releaseTextures];
>+
>+      _lastDrawnFrameTimeStampNs = self.videoFrame.timeStampNs;
>     }
>   } else {
>     if (!_i420TextureCache) {
>@@ -259,6 +265,8 @@ - (void)glkView:(GLKView *)view drawInRect:(CGRect)rect {
>                                     yPlane:_i420TextureCache.yTexture
>                                     uPlane:_i420TextureCache.uTexture
>                                     vPlane:_i420TextureCache.vTexture];
>+
>+    _lastDrawnFrameTimeStampNs = self.videoFrame.timeStampNs;
>   }
> }
> 
>@@ -282,7 +290,7 @@ - (void)renderFrame:(RTCVideoFrame *)frame {
> - (void)displayLinkTimerDidFire {
>   // Don't render unless video frame have changed or the view content
>   // has explicitly been marked dirty.
>-  if (!_isDirty && _lastDrawnFrame == self.videoFrame) {
>+  if (!_isDirty && _lastDrawnFrameTimeStampNs == self.videoFrame.timeStampNs) {
>     return;
>   }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCAVFoundationVideoCapturerInternal.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCAVFoundationVideoCapturerInternal.h
>deleted file mode 100644
>index 217437aff2d..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCAVFoundationVideoCapturerInternal.h
>+++ /dev/null
>@@ -1,50 +0,0 @@
>-/*
>- *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#import <AVFoundation/AVFoundation.h>
>-#import <Foundation/Foundation.h>
>-
>-#ifdef __cplusplus
>-#include "avfoundationvideocapturer.h"
>-#endif
>-
>-NS_ASSUME_NONNULL_BEGIN
>-// This class is an implementation detail of AVFoundationVideoCapturer and handles
>-// the ObjC integration with the AVFoundation APIs.
>-// It is meant to be owned by an instance of AVFoundationVideoCapturer.
>-// The reason for this is because other webrtc objects own cricket::VideoCapturer, which is not
>-// ref counted. To prevent bad behavior we do not expose this class directly.
>-__attribute__((objc_runtime_name("WK_RTCAVFoundationVideoCapturerInternal")))
>-@interface RTCAVFoundationVideoCapturerInternal
>-    : NSObject <AVCaptureVideoDataOutputSampleBufferDelegate>
>-
>-@property(nonatomic, readonly) AVCaptureSession *captureSession;
>-@property(nonatomic, readonly) dispatch_queue_t frameQueue;
>-@property(nonatomic, readonly) BOOL canUseBackCamera;
>-@property(nonatomic, assign) BOOL useBackCamera;  // Defaults to NO.
>-@property(atomic, assign) BOOL isRunning;  // Whether the capture session is running.
>-@property(atomic, assign) BOOL hasStarted;  // Whether we have an unmatched start.
>-
>-// We keep a pointer back to AVFoundationVideoCapturer to make callbacks on it
>-// when we receive frames. This is safe because this object should be owned by
>-// it.
>-- (instancetype)initWithCapturer:(webrtc::AVFoundationVideoCapturer *)capturer;
>-- (AVCaptureDevice *)getActiveCaptureDevice;
>-
>-- (nullable AVCaptureDevice *)frontCaptureDevice;
>-- (nullable AVCaptureDevice *)backCaptureDevice;
>-
>-// Starts and stops the capture session asynchronously. We cannot do this
>-// synchronously without blocking a WebRTC thread.
>-- (void)start;
>-- (void)stop;
>-
>-@end
>-NS_ASSUME_NONNULL_END
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCAVFoundationVideoCapturerInternal.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCAVFoundationVideoCapturerInternal.mm
>deleted file mode 100644
>index b3722b05510..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCAVFoundationVideoCapturerInternal.mm
>+++ /dev/null
>@@ -1,515 +0,0 @@
>-/*
>- *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#import "RTCAVFoundationVideoCapturerInternal.h"
>-
>-#import <Foundation/Foundation.h>
>-#if TARGET_OS_IPHONE
>-#import <UIKit/UIKit.h>
>-#import "WebRTC/UIDevice+RTCDevice.h"
>-#endif
>-
>-#import "AVCaptureSession+DevicePosition.h"
>-#import "RTCDispatcher+Private.h"
>-#import "WebRTC/RTCLogging.h"
>-
>-#include "avfoundationformatmapper.h"
>-
>-@implementation RTCAVFoundationVideoCapturerInternal {
>-  // Keep pointers to inputs for convenience.
>-  AVCaptureDeviceInput *_frontCameraInput;
>-  AVCaptureDeviceInput *_backCameraInput;
>-  AVCaptureVideoDataOutput *_videoDataOutput;
>-  // The cricket::VideoCapturer that owns this class. Should never be NULL.
>-  webrtc::AVFoundationVideoCapturer *_capturer;
>-  BOOL _hasRetriedOnFatalError;
>-  BOOL _isRunning;
>-  BOOL _hasStarted;
>-  rtc::CriticalSection _crit;
>-#if TARGET_OS_IPHONE
>-  UIDeviceOrientation _orientation;
>-#endif
>-}
>-
>-@synthesize captureSession = _captureSession;
>-@synthesize frameQueue = _frameQueue;
>-@synthesize useBackCamera = _useBackCamera;
>-
>-@synthesize isRunning = _isRunning;
>-@synthesize hasStarted = _hasStarted;
>-
>-// This is called from the thread that creates the video source, which is likely
>-// the main thread.
>-- (instancetype)initWithCapturer:(webrtc::AVFoundationVideoCapturer *)capturer {
>-  RTC_DCHECK(capturer);
>-  if (self = [super init]) {
>-    _capturer = capturer;
>-    // Create the capture session and all relevant inputs and outputs. We need
>-    // to do this in init because the application may want the capture session
>-    // before we start the capturer for e.g. AVCapturePreviewLayer. All objects
>-    // created here are retained until dealloc and never recreated.
>-    if (![self setupCaptureSession]) {
>-      return nil;
>-    }
>-    NSNotificationCenter *center = [NSNotificationCenter defaultCenter];
>-#if TARGET_OS_IPHONE
>-    _orientation = UIDeviceOrientationPortrait;
>-    [center addObserver:self
>-               selector:@selector(deviceOrientationDidChange:)
>-                   name:UIDeviceOrientationDidChangeNotification
>-                 object:nil];
>-    [center addObserver:self
>-               selector:@selector(handleCaptureSessionInterruption:)
>-                   name:AVCaptureSessionWasInterruptedNotification
>-                 object:_captureSession];
>-    [center addObserver:self
>-               selector:@selector(handleCaptureSessionInterruptionEnded:)
>-                   name:AVCaptureSessionInterruptionEndedNotification
>-                 object:_captureSession];
>-    [center addObserver:self
>-               selector:@selector(handleApplicationDidBecomeActive:)
>-                   name:UIApplicationDidBecomeActiveNotification
>-                 object:[UIApplication sharedApplication]];
>-#endif
>-    [center addObserver:self
>-               selector:@selector(handleCaptureSessionRuntimeError:)
>-                   name:AVCaptureSessionRuntimeErrorNotification
>-                 object:_captureSession];
>-    [center addObserver:self
>-               selector:@selector(handleCaptureSessionDidStartRunning:)
>-                   name:AVCaptureSessionDidStartRunningNotification
>-                 object:_captureSession];
>-    [center addObserver:self
>-               selector:@selector(handleCaptureSessionDidStopRunning:)
>-                   name:AVCaptureSessionDidStopRunningNotification
>-                 object:_captureSession];
>-  }
>-  return self;
>-}
>-
>-- (void)dealloc {
>-  RTC_DCHECK(!self.hasStarted);
>-  [[NSNotificationCenter defaultCenter] removeObserver:self];
>-  _capturer = nullptr;
>-}
>-
>-- (AVCaptureSession *)captureSession {
>-  return _captureSession;
>-}
>-
>-- (AVCaptureDevice *)getActiveCaptureDevice {
>-  return self.useBackCamera ? _backCameraInput.device : _frontCameraInput.device;
>-}
>-
>-- (nullable AVCaptureDevice *)frontCaptureDevice {
>-  return _frontCameraInput.device;
>-}
>-
>-- (nullable AVCaptureDevice *)backCaptureDevice {
>-  return _backCameraInput.device;
>-}
>-
>-- (dispatch_queue_t)frameQueue {
>-  if (!_frameQueue) {
>-    _frameQueue =
>-        dispatch_queue_create("org.webrtc.avfoundationvideocapturer.video", DISPATCH_QUEUE_SERIAL);
>-    dispatch_set_target_queue(_frameQueue,
>-                              dispatch_get_global_queue(DISPATCH_QUEUE_PRIORITY_HIGH, 0));
>-  }
>-  return _frameQueue;
>-}
>-
>-// Called from any thread (likely main thread).
>-- (BOOL)canUseBackCamera {
>-  return _backCameraInput != nil;
>-}
>-
>-// Called from any thread (likely main thread).
>-- (BOOL)useBackCamera {
>-  @synchronized(self) {
>-    return _useBackCamera;
>-  }
>-}
>-
>-// Called from any thread (likely main thread).
>-- (void)setUseBackCamera:(BOOL)useBackCamera {
>-  if (!self.canUseBackCamera) {
>-    if (useBackCamera) {
>-      RTCLogWarning(@"No rear-facing camera exists or it cannot be used;"
>-                     "not switching.");
>-    }
>-    return;
>-  }
>-  @synchronized(self) {
>-    if (_useBackCamera == useBackCamera) {
>-      return;
>-    }
>-    _useBackCamera = useBackCamera;
>-    [self updateSessionInputForUseBackCamera:useBackCamera];
>-  }
>-}
>-
>-// Called from WebRTC thread.
>-- (void)start {
>-  if (self.hasStarted) {
>-    return;
>-  }
>-  self.hasStarted = YES;
>-  [RTCDispatcher
>-      dispatchAsyncOnType:RTCDispatcherTypeCaptureSession
>-                    block:^{
>-                        [self updateOrientation];
>-#if TARGET_OS_IPHONE
>-                        [[UIDevice currentDevice] beginGeneratingDeviceOrientationNotifications];
>-#endif
>-                        AVCaptureSession *captureSession = self.captureSession;
>-                        [captureSession startRunning];
>-                    }];
>-}
>-
>-// Called from same thread as start.
>-- (void)stop {
>-  if (!self.hasStarted) {
>-    return;
>-  }
>-  self.hasStarted = NO;
>-  // Due to this async block, it's possible that the ObjC object outlives the
>-  // C++ one. In order to not invoke functions on the C++ object, we set
>-  // hasStarted immediately instead of dispatching it async.
>-  [RTCDispatcher
>-      dispatchAsyncOnType:RTCDispatcherTypeCaptureSession
>-                    block:^{
>-                        [_videoDataOutput setSampleBufferDelegate:nil queue:nullptr];
>-                        [_captureSession stopRunning];
>-#if TARGET_OS_IPHONE
>-                        [[UIDevice currentDevice] endGeneratingDeviceOrientationNotifications];
>-#endif
>-                    }];
>-}
>-
>-#pragma mark iOS notifications
>-
>-#if TARGET_OS_IPHONE
>-- (void)deviceOrientationDidChange:(NSNotification *)notification {
>-  [RTCDispatcher dispatchAsyncOnType:RTCDispatcherTypeCaptureSession
>-                               block:^{
>-                                   [self updateOrientation];
>-                               }];
>-}
>-#endif
>-
>-#pragma mark AVCaptureVideoDataOutputSampleBufferDelegate
>-
>-- (void)captureOutput:(AVCaptureOutput *)captureOutput
>-    didOutputSampleBuffer:(CMSampleBufferRef)sampleBuffer
>-           fromConnection:(AVCaptureConnection *)connection {
>-  NSParameterAssert(captureOutput == _videoDataOutput);
>-  if (!self.hasStarted) {
>-    return;
>-  }
>-
>-#if TARGET_OS_IPHONE
>-  // Default to portrait orientation on iPhone.
>-  webrtc::VideoRotation rotation = webrtc::kVideoRotation_90;
>-  BOOL usingFrontCamera = NO;
>-  // Check the image's EXIF for the camera the image came from as the image could have been
>-  // delayed as we set alwaysDiscardsLateVideoFrames to NO.
>-  AVCaptureDevicePosition cameraPosition =
>-      [AVCaptureSession devicePositionForSampleBuffer:sampleBuffer];
>-  if (cameraPosition != AVCaptureDevicePositionUnspecified) {
>-    usingFrontCamera = AVCaptureDevicePositionFront == cameraPosition;
>-  } else {
>-    AVCaptureDeviceInput *deviceInput =
>-        (AVCaptureDeviceInput *)((AVCaptureInputPort *)connection.inputPorts.firstObject).input;
>-    usingFrontCamera = AVCaptureDevicePositionFront == deviceInput.device.position;
>-  }
>-  switch (_orientation) {
>-    case UIDeviceOrientationPortrait:
>-      rotation = webrtc::kVideoRotation_90;
>-      break;
>-    case UIDeviceOrientationPortraitUpsideDown:
>-      rotation = webrtc::kVideoRotation_270;
>-      break;
>-    case UIDeviceOrientationLandscapeLeft:
>-      rotation = usingFrontCamera ? webrtc::kVideoRotation_180 : webrtc::kVideoRotation_0;
>-      break;
>-    case UIDeviceOrientationLandscapeRight:
>-      rotation = usingFrontCamera ? webrtc::kVideoRotation_0 : webrtc::kVideoRotation_180;
>-      break;
>-    case UIDeviceOrientationFaceUp:
>-    case UIDeviceOrientationFaceDown:
>-    case UIDeviceOrientationUnknown:
>-      // Ignore.
>-      break;
>-  }
>-#else
>-  // No rotation on Mac.
>-  webrtc::VideoRotation rotation = webrtc::kVideoRotation_0;
>-#endif
>-
>-  _capturer->CaptureSampleBuffer(sampleBuffer, rotation);
>-}
>-
>-- (void)captureOutput:(AVCaptureOutput *)captureOutput
>-    didDropSampleBuffer:(CMSampleBufferRef)sampleBuffer
>-         fromConnection:(AVCaptureConnection *)connection {
>-  RTCLogError(@"Dropped sample buffer.");
>-}
>-
>-#pragma mark - AVCaptureSession notifications
>-
>-- (void)handleCaptureSessionInterruption:(NSNotification *)notification {
>-  NSString *reasonString = nil;
>-#if TARGET_OS_IPHONE
>-  NSNumber *reason = notification.userInfo[AVCaptureSessionInterruptionReasonKey];
>-  if (reason) {
>-    switch (reason.intValue) {
>-      case AVCaptureSessionInterruptionReasonVideoDeviceNotAvailableInBackground:
>-        reasonString = @"VideoDeviceNotAvailableInBackground";
>-        break;
>-      case AVCaptureSessionInterruptionReasonAudioDeviceInUseByAnotherClient:
>-        reasonString = @"AudioDeviceInUseByAnotherClient";
>-        break;
>-      case AVCaptureSessionInterruptionReasonVideoDeviceInUseByAnotherClient:
>-        reasonString = @"VideoDeviceInUseByAnotherClient";
>-        break;
>-      case AVCaptureSessionInterruptionReasonVideoDeviceNotAvailableWithMultipleForegroundApps:
>-        reasonString = @"VideoDeviceNotAvailableWithMultipleForegroundApps";
>-        break;
>-    }
>-  }
>-#endif
>-  RTCLog(@"Capture session interrupted: %@", reasonString);
>-  // TODO(tkchin): Handle this case.
>-}
>-
>-- (void)handleCaptureSessionInterruptionEnded:(NSNotification *)notification {
>-  RTCLog(@"Capture session interruption ended.");
>-  // TODO(tkchin): Handle this case.
>-}
>-
>-- (void)handleCaptureSessionRuntimeError:(NSNotification *)notification {
>-  NSError *error = [notification.userInfo objectForKey:AVCaptureSessionErrorKey];
>-  RTCLogError(@"Capture session runtime error: %@", error);
>-
>-  [RTCDispatcher dispatchAsyncOnType:RTCDispatcherTypeCaptureSession
>-                               block:^{
>-#if TARGET_OS_IPHONE
>-                                   if (error.code == AVErrorMediaServicesWereReset) {
>-                                     [self handleNonFatalError];
>-                                   } else {
>-                                     [self handleFatalError];
>-                                   }
>-#else
>-                                   [self handleFatalError];
>-#endif
>-                               }];
>-}
>-
>-- (void)handleCaptureSessionDidStartRunning:(NSNotification *)notification {
>-  RTCLog(@"Capture session started.");
>-
>-  self.isRunning = YES;
>-  [RTCDispatcher dispatchAsyncOnType:RTCDispatcherTypeCaptureSession
>-                               block:^{
>-                                   // If we successfully restarted after an unknown error,
>-                                   // allow future retries on fatal errors.
>-                                   _hasRetriedOnFatalError = NO;
>-                               }];
>-}
>-
>-- (void)handleCaptureSessionDidStopRunning:(NSNotification *)notification {
>-  RTCLog(@"Capture session stopped.");
>-  self.isRunning = NO;
>-}
>-
>-- (void)handleFatalError {
>-  [RTCDispatcher
>-      dispatchAsyncOnType:RTCDispatcherTypeCaptureSession
>-                    block:^{
>-                        if (!_hasRetriedOnFatalError) {
>-                          RTCLogWarning(@"Attempting to recover from fatal capture error.");
>-                          [self handleNonFatalError];
>-                          _hasRetriedOnFatalError = YES;
>-                        } else {
>-                          RTCLogError(@"Previous fatal error recovery failed.");
>-                        }
>-                    }];
>-}
>-
>-- (void)handleNonFatalError {
>-  [RTCDispatcher dispatchAsyncOnType:RTCDispatcherTypeCaptureSession
>-                               block:^{
>-                                   if (self.hasStarted) {
>-                                     RTCLog(@"Restarting capture session after error.");
>-                                     [self.captureSession startRunning];
>-                                   }
>-                               }];
>-}
>-
>-#if TARGET_OS_IPHONE
>-
>-#pragma mark - UIApplication notifications
>-
>-- (void)handleApplicationDidBecomeActive:(NSNotification *)notification {
>-  [RTCDispatcher dispatchAsyncOnType:RTCDispatcherTypeCaptureSession
>-                               block:^{
>-                                   if (self.hasStarted && !self.captureSession.isRunning) {
>-                                     RTCLog(@"Restarting capture session on active.");
>-                                     [self.captureSession startRunning];
>-                                   }
>-                               }];
>-}
>-
>-#endif  // TARGET_OS_IPHONE
>-
>-#pragma mark - Private
>-
>-- (BOOL)setupCaptureSession {
>-  AVCaptureSession *captureSession = [[AVCaptureSession alloc] init];
>-#if defined(WEBRTC_IOS)
>-  captureSession.usesApplicationAudioSession = NO;
>-#endif
>-  // Add the output.
>-  AVCaptureVideoDataOutput *videoDataOutput = [self videoDataOutput];
>-  if (![captureSession canAddOutput:videoDataOutput]) {
>-    RTCLogError(@"Video data output unsupported.");
>-    return NO;
>-  }
>-  [captureSession addOutput:videoDataOutput];
>-
>-  // Get the front and back cameras. If there isn't a front camera
>-  // give up.
>-  AVCaptureDeviceInput *frontCameraInput = [self frontCameraInput];
>-  AVCaptureDeviceInput *backCameraInput = [self backCameraInput];
>-  if (!frontCameraInput) {
>-    RTCLogError(@"No front camera for capture session.");
>-    return NO;
>-  }
>-
>-  // Add the inputs.
>-  if (![captureSession canAddInput:frontCameraInput] ||
>-      (backCameraInput && ![captureSession canAddInput:backCameraInput])) {
>-    RTCLogError(@"Session does not support capture inputs.");
>-    return NO;
>-  }
>-  AVCaptureDeviceInput *input = self.useBackCamera ? backCameraInput : frontCameraInput;
>-  [captureSession addInput:input];
>-
>-  _captureSession = captureSession;
>-  return YES;
>-}
>-
>-- (AVCaptureVideoDataOutput *)videoDataOutput {
>-  if (!_videoDataOutput) {
>-    // Make the capturer output NV12. Ideally we want I420 but that's not
>-    // currently supported on iPhone / iPad.
>-    AVCaptureVideoDataOutput *videoDataOutput = [[AVCaptureVideoDataOutput alloc] init];
>-    videoDataOutput.videoSettings = @{
>-      (NSString *)
>-      // TODO(denicija): Remove this color conversion and use the original capture format directly.
>-      kCVPixelBufferPixelFormatTypeKey : @(kCVPixelFormatType_420YpCbCr8BiPlanarFullRange)
>-    };
>-    videoDataOutput.alwaysDiscardsLateVideoFrames = NO;
>-    [videoDataOutput setSampleBufferDelegate:self queue:self.frameQueue];
>-    _videoDataOutput = videoDataOutput;
>-  }
>-  return _videoDataOutput;
>-}
>-
>-- (AVCaptureDevice *)videoCaptureDeviceForPosition:(AVCaptureDevicePosition)position {
>-  for (AVCaptureDevice *captureDevice in [AVCaptureDevice devicesWithMediaType:AVMediaTypeVideo]) {
>-    if (captureDevice.position == position) {
>-      return captureDevice;
>-    }
>-  }
>-  return nil;
>-}
>-
>-- (AVCaptureDeviceInput *)frontCameraInput {
>-  if (!_frontCameraInput) {
>-#if TARGET_OS_IPHONE
>-    AVCaptureDevice *frontCameraDevice =
>-        [self videoCaptureDeviceForPosition:AVCaptureDevicePositionFront];
>-#else
>-    AVCaptureDevice *frontCameraDevice =
>-        [AVCaptureDevice defaultDeviceWithMediaType:AVMediaTypeVideo];
>-#endif
>-    if (!frontCameraDevice) {
>-      RTCLogWarning(@"Failed to find front capture device.");
>-      return nil;
>-    }
>-    NSError *error = nil;
>-    AVCaptureDeviceInput *frontCameraInput =
>-        [AVCaptureDeviceInput deviceInputWithDevice:frontCameraDevice error:&error];
>-    if (!frontCameraInput) {
>-      RTCLogError(@"Failed to create front camera input: %@", error.localizedDescription);
>-      return nil;
>-    }
>-    _frontCameraInput = frontCameraInput;
>-  }
>-  return _frontCameraInput;
>-}
>-
>-- (AVCaptureDeviceInput *)backCameraInput {
>-  if (!_backCameraInput) {
>-    AVCaptureDevice *backCameraDevice =
>-        [self videoCaptureDeviceForPosition:AVCaptureDevicePositionBack];
>-    if (!backCameraDevice) {
>-      RTCLogWarning(@"Failed to find front capture device.");
>-      return nil;
>-    }
>-    NSError *error = nil;
>-    AVCaptureDeviceInput *backCameraInput =
>-        [AVCaptureDeviceInput deviceInputWithDevice:backCameraDevice error:&error];
>-    if (!backCameraInput) {
>-      RTCLogError(@"Failed to create front camera input: %@", error.localizedDescription);
>-      return nil;
>-    }
>-    _backCameraInput = backCameraInput;
>-  }
>-  return _backCameraInput;
>-}
>-
>-// Called from capture session queue.
>-- (void)updateOrientation {
>-#if TARGET_OS_IPHONE
>-  _orientation = [UIDevice currentDevice].orientation;
>-#endif
>-}
>-
>-// Update the current session input to match what's stored in _useBackCamera.
>-- (void)updateSessionInputForUseBackCamera:(BOOL)useBackCamera {
>-  [RTCDispatcher
>-      dispatchAsyncOnType:RTCDispatcherTypeCaptureSession
>-                    block:^{
>-                      [_captureSession beginConfiguration];
>-                      AVCaptureDeviceInput *oldInput = _backCameraInput;
>-                      AVCaptureDeviceInput *newInput = _frontCameraInput;
>-                      if (useBackCamera) {
>-                        oldInput = _frontCameraInput;
>-                        newInput = _backCameraInput;
>-                      }
>-                      if (oldInput) {
>-                        // Ok to remove this even if it's not attached. Will be no-op.
>-                        [_captureSession removeInput:oldInput];
>-                      }
>-                      if (newInput) {
>-                        [_captureSession addInput:newInput];
>-                      }
>-                      [self updateOrientation];
>-                      AVCaptureDevice *newDevice = newInput.device;
>-                      const cricket::VideoFormat *format = _capturer->GetCaptureFormat();
>-                      webrtc::SetFormatForCaptureDevice(newDevice, _captureSession, *format);
>-                      [_captureSession commitConfiguration];
>-                    }];
>-}
>-
>-@end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCCVPixelBuffer.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCCVPixelBuffer.mm
>index 203896758e8..9e5ac7320bd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCCVPixelBuffer.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCCVPixelBuffer.mm
>@@ -13,9 +13,13 @@
> #include "common_video/libyuv/include/webrtc_libyuv.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>-
> #include "third_party/libyuv/include/libyuv.h"
> 
>+#if !defined(NDEBUG) && defined(WEBRTC_IOS)
>+#import <UIKit/UIKit.h>
>+#import <VideoToolbox/VideoToolbox.h>
>+#endif
>+
> @implementation RTCCVPixelBuffer {
>   int _width;
>   int _height;
>@@ -28,6 +32,8 @@ @implementation RTCCVPixelBuffer {
> @synthesize pixelBuffer = _pixelBuffer;
> @synthesize cropX = _cropX;
> @synthesize cropY = _cropY;
>+@synthesize cropWidth = _cropWidth;
>+@synthesize cropHeight = _cropHeight;
> 
> + (NSSet<NSNumber*>*)supportedPixelFormats {
>   return [NSSet setWithObjects:@(kCVPixelFormatType_420YpCbCr8BiPlanarFullRange),
>@@ -112,16 +118,29 @@ - (int)bufferSizeForCroppingAndScalingToWidth:(int)width height:(int)height {
>   return 0;
> }
> 
>-- (BOOL)cropAndScaleTo:(CVPixelBufferRef)outputPixelBuffer withTempBuffer:(uint8_t*)tmpBuffer {
>+- (BOOL)cropAndScaleTo:(CVPixelBufferRef)outputPixelBuffer
>+        withTempBuffer:(nullable uint8_t*)tmpBuffer {
>   const OSType srcPixelFormat = CVPixelBufferGetPixelFormatType(_pixelBuffer);
>+  const OSType dstPixelFormat = CVPixelBufferGetPixelFormatType(outputPixelBuffer);
>+
>   switch (srcPixelFormat) {
>     case kCVPixelFormatType_420YpCbCr8BiPlanarFullRange:
>     case kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange: {
>-      [self cropAndScaleNV12To:outputPixelBuffer withTempBuffer:tmpBuffer];
>+      size_t dstWidth = CVPixelBufferGetWidth(outputPixelBuffer);
>+      size_t dstHeight = CVPixelBufferGetHeight(outputPixelBuffer);
>+      if (dstWidth > 0 && dstHeight > 0) {
>+        RTC_DCHECK(dstPixelFormat == kCVPixelFormatType_420YpCbCr8BiPlanarFullRange ||
>+                   dstPixelFormat == kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange);
>+        if ([self requiresScalingToWidth:dstWidth height:dstHeight]) {
>+          RTC_DCHECK(tmpBuffer);
>+        }
>+        [self cropAndScaleNV12To:outputPixelBuffer withTempBuffer:tmpBuffer];
>+      }
>       break;
>     }
>     case kCVPixelFormatType_32BGRA:
>     case kCVPixelFormatType_32ARGB: {
>+      RTC_DCHECK(srcPixelFormat == dstPixelFormat);
>       [self cropAndScaleARGBTo:outputPixelBuffer];
>       break;
>     }
>@@ -143,10 +162,10 @@ - (id<RTCI420Buffer>)toI420 {
>     case kCVPixelFormatType_420YpCbCr8BiPlanarFullRange:
>     case kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange: {
>       const uint8_t* srcY =
>-          static_cast<const uint8_t*>(CVPixelBufferGetBaseAddressOfPlane(_pixelBuffer, 0));
>+          static_cast<uint8_t*>(CVPixelBufferGetBaseAddressOfPlane(_pixelBuffer, 0));
>       const int srcYStride = CVPixelBufferGetBytesPerRowOfPlane(_pixelBuffer, 0);
>       const uint8_t* srcUV =
>-          static_cast<const uint8_t*>(CVPixelBufferGetBaseAddressOfPlane(_pixelBuffer, 1));
>+          static_cast<uint8_t*>(CVPixelBufferGetBaseAddressOfPlane(_pixelBuffer, 1));
>       const int srcUVStride = CVPixelBufferGetBytesPerRowOfPlane(_pixelBuffer, 1);
> 
>       // Crop just by modifying pointers.
>@@ -173,32 +192,52 @@ - (id<RTCI420Buffer>)toI420 {
>     }
>     case kCVPixelFormatType_32BGRA:
>     case kCVPixelFormatType_32ARGB: {
>-      const uint8_t* src =
>-          static_cast<const uint8_t*>(CVPixelBufferGetBaseAddressOfPlane(_pixelBuffer, 0));
>+      CVPixelBufferRef scaledPixelBuffer = NULL;
>+      CVPixelBufferRef sourcePixelBuffer = NULL;
>+      if ([self requiresCropping] ||
>+          [self requiresScalingToWidth:i420Buffer.width height:i420Buffer.height]) {
>+        CVPixelBufferCreate(
>+            NULL, i420Buffer.width, i420Buffer.height, pixelFormat, NULL, &scaledPixelBuffer);
>+        [self cropAndScaleTo:scaledPixelBuffer withTempBuffer:NULL];
>+
>+        CVPixelBufferLockBaseAddress(scaledPixelBuffer, kCVPixelBufferLock_ReadOnly);
>+        sourcePixelBuffer = scaledPixelBuffer;
>+      } else {
>+        sourcePixelBuffer = _pixelBuffer;
>+      }
>+      const uint8_t* src = static_cast<uint8_t*>(CVPixelBufferGetBaseAddress(sourcePixelBuffer));
>+      const size_t bytesPerRow = CVPixelBufferGetBytesPerRow(sourcePixelBuffer);
> 
>-      uint32 libyuvPixelFormat = 0;
>       if (pixelFormat == kCVPixelFormatType_32BGRA) {
>-        libyuvPixelFormat = libyuv::FOURCC_ARGB;
>+        // Corresponds to libyuv::FOURCC_ARGB
>+        libyuv::ARGBToI420(src,
>+                           bytesPerRow,
>+                           i420Buffer.mutableDataY,
>+                           i420Buffer.strideY,
>+                           i420Buffer.mutableDataU,
>+                           i420Buffer.strideU,
>+                           i420Buffer.mutableDataV,
>+                           i420Buffer.strideV,
>+                           i420Buffer.width,
>+                           i420Buffer.height);
>       } else if (pixelFormat == kCVPixelFormatType_32ARGB) {
>-        libyuvPixelFormat = libyuv::FOURCC_ABGR;
>+        // Corresponds to libyuv::FOURCC_BGRA
>+        libyuv::BGRAToI420(src,
>+                           bytesPerRow,
>+                           i420Buffer.mutableDataY,
>+                           i420Buffer.strideY,
>+                           i420Buffer.mutableDataU,
>+                           i420Buffer.strideU,
>+                           i420Buffer.mutableDataV,
>+                           i420Buffer.strideV,
>+                           i420Buffer.width,
>+                           i420Buffer.height);
>       }
> 
>-      libyuv::ConvertToI420(src,
>-                            0,
>-                            i420Buffer.mutableDataY,
>-                            i420Buffer.strideY,
>-                            i420Buffer.mutableDataU,
>-                            i420Buffer.strideU,
>-                            i420Buffer.mutableDataV,
>-                            i420Buffer.strideV,
>-                            _cropX,
>-                            _cropY,
>-                            _cropWidth,
>-                            _cropHeight,
>-                            i420Buffer.width,
>-                            i420Buffer.height,
>-                            libyuv::kRotate0,
>-                            libyuvPixelFormat);
>+      if (scaledPixelBuffer) {
>+        CVPixelBufferUnlockBaseAddress(scaledPixelBuffer, kCVPixelBufferLock_ReadOnly);
>+        CVBufferRelease(scaledPixelBuffer);
>+      }
>       break;
>     }
>     default: { RTC_NOTREACHED() << "Unsupported pixel format."; }
>@@ -209,6 +248,20 @@ - (id<RTCI420Buffer>)toI420 {
>   return i420Buffer;
> }
> 
>+#pragma mark - Debugging
>+
>+#if !defined(NDEBUG) && defined(WEBRTC_IOS)
>+- (id)debugQuickLookObject {
>+  CGImageRef cgImage;
>+  VTCreateCGImageFromCVPixelBuffer(_pixelBuffer, NULL, &cgImage);
>+  UIImage *image = [UIImage imageWithCGImage:cgImage scale:1.0 orientation:UIImageOrientationUp];
>+  CGImageRelease(cgImage);
>+  return image;
>+}
>+#endif
>+
>+#pragma mark - Private
>+
> - (void)cropAndScaleNV12To:(CVPixelBufferRef)outputPixelBuffer withTempBuffer:(uint8_t*)tmpBuffer {
>   // Prepare output pointers.
>   CVReturn cvRet = CVPixelBufferLockBaseAddress(outputPixelBuffer, 0);
>@@ -226,11 +279,9 @@ - (void)cropAndScaleNV12To:(CVPixelBufferRef)outputPixelBuffer withTempBuffer:(u
> 
>   // Prepare source pointers.
>   CVPixelBufferLockBaseAddress(_pixelBuffer, kCVPixelBufferLock_ReadOnly);
>-  const uint8_t* srcY =
>-      static_cast<const uint8_t*>(CVPixelBufferGetBaseAddressOfPlane(_pixelBuffer, 0));
>+  const uint8_t* srcY = static_cast<uint8_t*>(CVPixelBufferGetBaseAddressOfPlane(_pixelBuffer, 0));
>   const int srcYStride = CVPixelBufferGetBytesPerRowOfPlane(_pixelBuffer, 0);
>-  const uint8_t* srcUV =
>-      static_cast<const uint8_t*>(CVPixelBufferGetBaseAddressOfPlane(_pixelBuffer, 1));
>+  const uint8_t* srcUV = static_cast<uint8_t*>(CVPixelBufferGetBaseAddressOfPlane(_pixelBuffer, 1));
>   const int srcUVStride = CVPixelBufferGetBytesPerRowOfPlane(_pixelBuffer, 1);
> 
>   // Crop just by modifying pointers.
>@@ -264,17 +315,20 @@ - (void)cropAndScaleARGBTo:(CVPixelBufferRef)outputPixelBuffer {
>   const int dstWidth = CVPixelBufferGetWidth(outputPixelBuffer);
>   const int dstHeight = CVPixelBufferGetHeight(outputPixelBuffer);
> 
>-  uint8_t* dst =
>-      reinterpret_cast<uint8_t*>(CVPixelBufferGetBaseAddressOfPlane(outputPixelBuffer, 0));
>-  const int dstStride = CVPixelBufferGetBytesPerRowOfPlane(outputPixelBuffer, 0);
>+  uint8_t* dst = reinterpret_cast<uint8_t*>(CVPixelBufferGetBaseAddress(outputPixelBuffer));
>+  const int dstStride = CVPixelBufferGetBytesPerRow(outputPixelBuffer);
> 
>   // Prepare source pointers.
>   CVPixelBufferLockBaseAddress(_pixelBuffer, kCVPixelBufferLock_ReadOnly);
>-  const uint8_t* src = static_cast<const uint8_t*>(CVPixelBufferGetBaseAddress(_pixelBuffer));
>+  const uint8_t* src = static_cast<uint8_t*>(CVPixelBufferGetBaseAddress(_pixelBuffer));
>   const int srcStride = CVPixelBufferGetBytesPerRow(_pixelBuffer);
> 
>-  // Crop just by modifying pointers.
>-  src += srcStride * _cropY + _cropX;
>+  // Crop just by modifying pointers. Need to ensure that src pointer points to a byte corresponding
>+  // to the start of a new pixel (byte with B for BGRA) so that libyuv scales correctly.
>+  const int bytesPerPixel = 4;
>+  src += srcStride * _cropY + (_cropX * bytesPerPixel);
>+
>+  // kCVPixelFormatType_32BGRA corresponds to libyuv::FOURCC_ARGB
>   libyuv::ARGBScale(src,
>                     srcStride,
>                     _cropWidth,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCDefaultShader.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCDefaultShader.h
>index cc2ff4e46be..ed5125e2e8e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCDefaultShader.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCDefaultShader.h
>@@ -16,7 +16,6 @@ NS_ASSUME_NONNULL_BEGIN
>  *  RTCEAGLVideoView if no external shader is specified. This shader will render
>  *  the video in a rectangle without any color or geometric transformations.
>  */
>-__attribute__((objc_runtime_name("WK_RTCDefaultShader")))
> @interface RTCDefaultShader : NSObject<RTCVideoViewShading>
> 
> @end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCDefaultShader.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCDefaultShader.mm
>index 3caf1442628..c5fbde12c82 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCDefaultShader.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCDefaultShader.mm
>@@ -20,7 +20,7 @@
> #import "RTCShader.h"
> #import "WebRTC/RTCLogging.h"
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> 
> static const int kYTextureUnit = 0;
> static const int kUTextureUnit = 1;
>@@ -73,7 +73,7 @@ @implementation RTCDefaultShader {
>   GLuint _vertexBuffer;
>   GLuint _vertexArray;
>   // Store current rotation and only upload new vertex data when rotation changes.
>-  rtc::Optional<RTCVideoRotation> _currentRotation;
>+  absl::optional<RTCVideoRotation> _currentRotation;
> 
>   GLuint _i420Program;
>   GLuint _nv12Program;
>@@ -144,7 +144,7 @@ - (BOOL)prepareVertexBufferWithRotation:(RTCVideoRotation)rotation {
> #endif
>   glBindBuffer(GL_ARRAY_BUFFER, _vertexBuffer);
>   if (!_currentRotation || rotation != *_currentRotation) {
>-    _currentRotation = rtc::Optional<RTCVideoRotation>(rotation);
>+    _currentRotation = absl::optional<RTCVideoRotation>(rotation);
>     RTCSetVertexData(*_currentRotation);
>   }
>   return YES;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCI420Buffer.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCI420Buffer.mm
>index a428a8d6cbb..ae4092689ff 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCI420Buffer.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCI420Buffer.mm
>@@ -8,10 +8,15 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#import "WebRTC/RTCVideoFrameBuffer.h"
>+#import "RTCI420Buffer+Private.h"
> 
> #include "api/video/i420_buffer.h"
> 
>+#if !defined(NDEBUG) && defined(WEBRTC_IOS)
>+#import <UIKit/UIKit.h>
>+#include "third_party/libyuv/include/libyuv.h"
>+#endif
>+
> @implementation RTCI420Buffer {
>  @protected
>   rtc::scoped_refptr<webrtc::I420BufferInterface> _i420Buffer;
>@@ -25,6 +30,18 @@ - (instancetype)initWithWidth:(int)width height:(int)height {
>   return self;
> }
> 
>+- (instancetype)initWithWidth:(int)width
>+                       height:(int)height
>+                        dataY:(const uint8_t *)dataY
>+                        dataU:(const uint8_t *)dataU
>+                        dataV:(const uint8_t *)dataV {
>+  if (self = [super init]) {
>+    _i420Buffer = webrtc::I420Buffer::Copy(
>+        width, height, dataY, width, dataU, (width + 1) / 2, dataV, (width + 1) / 2);
>+  }
>+  return self;
>+}
>+
> - (instancetype)initWithWidth:(int)width
>                        height:(int)height
>                       strideY:(int)strideY
>@@ -89,8 +106,42 @@ - (id<RTCI420Buffer>)toI420 {
>   return self;
> }
> 
>+#pragma mark - Private
>+
>+- (rtc::scoped_refptr<webrtc::I420BufferInterface>)nativeI420Buffer {
>+  return _i420Buffer;
>+}
>+
>+#pragma mark - Debugging
>+
>+#if !defined(NDEBUG) && defined(WEBRTC_IOS)
>+- (id)debugQuickLookObject {
>+  UIGraphicsBeginImageContext(CGSizeMake(_i420Buffer->width(), _i420Buffer->height()));
>+  CGContextRef c = UIGraphicsGetCurrentContext();
>+  uint8_t *ctxData = (uint8_t *)CGBitmapContextGetData(c);
>+
>+  libyuv::I420ToARGB(_i420Buffer->DataY(),
>+                     _i420Buffer->StrideY(),
>+                     _i420Buffer->DataU(),
>+                     _i420Buffer->StrideU(),
>+                     _i420Buffer->DataV(),
>+                     _i420Buffer->StrideV(),
>+                     ctxData,
>+                     CGBitmapContextGetBytesPerRow(c),
>+                     CGBitmapContextGetWidth(c),
>+                     CGBitmapContextGetHeight(c));
>+
>+  UIImage *image = UIGraphicsGetImageFromCurrentImageContext();
>+  UIGraphicsEndImageContext();
>+
>+  return image;
>+}
>+#endif
>+
> @end
> 
>+#pragma mark -
>+
> @implementation RTCMutableI420Buffer
> 
> - (uint8_t *)mutableDataY {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCI420TextureCache.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCI420TextureCache.h
>index 6726a8842b7..19f1bc99934 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCI420TextureCache.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCI420TextureCache.h
>@@ -11,7 +11,6 @@
> #import "RTCOpenGLDefines.h"
> #import "WebRTC/RTCVideoFrame.h"
> 
>-__attribute__((objc_runtime_name("WK_RTCI420TextureCache")))
> @interface RTCI420TextureCache : NSObject
> 
> @property(nonatomic, readonly) GLuint yTexture;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCNV12TextureCache.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCNV12TextureCache.h
>index d59b73c9a93..9cba8232714 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCNV12TextureCache.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCNV12TextureCache.h
>@@ -14,7 +14,6 @@
> 
> NS_ASSUME_NONNULL_BEGIN
> 
>-__attribute__((objc_runtime_name("WK_RTCNV12TextureCache")))
> @interface RTCNV12TextureCache : NSObject
> 
> @property(nonatomic, readonly) GLuint yTexture;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCShader.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCShader.h
>index 4d92d3dd102..67afabb1d8a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCShader.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCShader.h
>@@ -12,9 +12,10 @@
> 
> RTC_EXTERN const char kRTCVertexShaderSource[];
> 
>-RTC_EXTERN GLuint RTCCreateShader(GLenum type, const GLchar *source);
>+RTC_EXTERN GLuint RTCCreateShader(GLenum type, const GLchar* source);
> RTC_EXTERN GLuint RTCCreateProgram(GLuint vertexShader, GLuint fragmentShader);
>-RTC_EXTERN GLuint RTCCreateProgramFromFragmentSource(const char fragmentShaderSource[]);
>+RTC_EXTERN GLuint
>+RTCCreateProgramFromFragmentSource(const char fragmentShaderSource[]);
> RTC_EXTERN BOOL RTCCreateVertexBuffer(GLuint* vertexBuffer,
>                                       GLuint* vertexArray);
> RTC_EXTERN void RTCSetVertexData(RTCVideoRotation rotation);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/UIDevice+H264Profile.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/UIDevice+H264Profile.h
>new file mode 100644
>index 00000000000..bb6f6ce520b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/UIDevice+H264Profile.h
>@@ -0,0 +1,19 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#import <UIKit/UIKit.h>
>+
>+#include "media/base/h264_profile_level_id.h"
>+
>+@interface UIDevice (H264Profile)
>+
>++ (absl::optional<webrtc::H264::ProfileLevelId>)maxSupportedH264Profile;
>+
>+@end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/UIDevice+H264Profile.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/UIDevice+H264Profile.mm
>new file mode 100644
>index 00000000000..196e34e4efd
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/UIDevice+H264Profile.mm
>@@ -0,0 +1,108 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#import "UIDevice+H264Profile.h"
>+#import "WebRTC/UIDevice+RTCDevice.h"
>+
>+#include <algorithm>
>+
>+namespace {
>+
>+using namespace webrtc::H264;
>+
>+struct SupportedH264Profile {
>+  const RTCDeviceType deviceType;
>+  const ProfileLevelId profile;
>+};
>+
>+constexpr SupportedH264Profile kH264MaxSupportedProfiles[] = {
>+    // iPhones with at least iOS 9
>+    {RTCDeviceTypeIPhoneX, {kProfileHigh, kLevel5_2}},       // https://support.apple.com/kb/SP770
>+    {RTCDeviceTypeIPhone8, {kProfileHigh, kLevel5_2}},       // https://support.apple.com/kb/SP767
>+    {RTCDeviceTypeIPhone8Plus, {kProfileHigh, kLevel5_2}},   // https://support.apple.com/kb/SP768
>+    {RTCDeviceTypeIPhone7, {kProfileHigh, kLevel5_1}},       // https://support.apple.com/kb/SP743
>+    {RTCDeviceTypeIPhone7Plus, {kProfileHigh, kLevel5_1}},   // https://support.apple.com/kb/SP744
>+    {RTCDeviceTypeIPhoneSE, {kProfileHigh, kLevel4_2}},      // https://support.apple.com/kb/SP738
>+    {RTCDeviceTypeIPhone6S, {kProfileHigh, kLevel4_2}},      // https://support.apple.com/kb/SP726
>+    {RTCDeviceTypeIPhone6SPlus, {kProfileHigh, kLevel4_2}},  // https://support.apple.com/kb/SP727
>+    {RTCDeviceTypeIPhone6, {kProfileHigh, kLevel4_2}},       // https://support.apple.com/kb/SP705
>+    {RTCDeviceTypeIPhone6Plus, {kProfileHigh, kLevel4_2}},   // https://support.apple.com/kb/SP706
>+    {RTCDeviceTypeIPhone5SGSM, {kProfileHigh, kLevel4_2}},   // https://support.apple.com/kb/SP685
>+    {RTCDeviceTypeIPhone5SGSM_CDMA,
>+     {kProfileHigh, kLevel4_2}},                           // https://support.apple.com/kb/SP685
>+    {RTCDeviceTypeIPhone5GSM, {kProfileHigh, kLevel4_1}},  // https://support.apple.com/kb/SP655
>+    {RTCDeviceTypeIPhone5GSM_CDMA,
>+     {kProfileHigh, kLevel4_1}},                            // https://support.apple.com/kb/SP655
>+    {RTCDeviceTypeIPhone5CGSM, {kProfileHigh, kLevel4_1}},  // https://support.apple.com/kb/SP684
>+    {RTCDeviceTypeIPhone5CGSM_CDMA,
>+     {kProfileHigh, kLevel4_1}},                         // https://support.apple.com/kb/SP684
>+    {RTCDeviceTypeIPhone4S, {kProfileHigh, kLevel4_1}},  // https://support.apple.com/kb/SP643
>+
>+    // iPods with at least iOS 9
>+    {RTCDeviceTypeIPodTouch6G, {kProfileMain, kLevel4_1}},  // https://support.apple.com/kb/SP720
>+    {RTCDeviceTypeIPodTouch5G, {kProfileMain, kLevel3_1}},  // https://support.apple.com/kb/SP657
>+
>+    // iPads with at least iOS 9
>+    {RTCDeviceTypeIPad2Wifi, {kProfileHigh, kLevel4_1}},     // https://support.apple.com/kb/SP622
>+    {RTCDeviceTypeIPad2GSM, {kProfileHigh, kLevel4_1}},      // https://support.apple.com/kb/SP622
>+    {RTCDeviceTypeIPad2CDMA, {kProfileHigh, kLevel4_1}},     // https://support.apple.com/kb/SP622
>+    {RTCDeviceTypeIPad2Wifi2, {kProfileHigh, kLevel4_1}},    // https://support.apple.com/kb/SP622
>+    {RTCDeviceTypeIPadMiniWifi, {kProfileHigh, kLevel4_1}},  // https://support.apple.com/kb/SP661
>+    {RTCDeviceTypeIPadMiniGSM, {kProfileHigh, kLevel4_1}},   // https://support.apple.com/kb/SP661
>+    {RTCDeviceTypeIPadMiniGSM_CDMA,
>+     {kProfileHigh, kLevel4_1}},                              // https://support.apple.com/kb/SP661
>+    {RTCDeviceTypeIPad3Wifi, {kProfileHigh, kLevel4_1}},      // https://support.apple.com/kb/SP647
>+    {RTCDeviceTypeIPad3GSM_CDMA, {kProfileHigh, kLevel4_1}},  // https://support.apple.com/kb/SP647
>+    {RTCDeviceTypeIPad3GSM, {kProfileHigh, kLevel4_1}},       // https://support.apple.com/kb/SP647
>+    {RTCDeviceTypeIPad4Wifi, {kProfileHigh, kLevel4_1}},      // https://support.apple.com/kb/SP662
>+    {RTCDeviceTypeIPad4GSM, {kProfileHigh, kLevel4_1}},       // https://support.apple.com/kb/SP662
>+    {RTCDeviceTypeIPad4GSM_CDMA, {kProfileHigh, kLevel4_1}},  // https://support.apple.com/kb/SP662
>+    {RTCDeviceTypeIPad5, {kProfileHigh, kLevel4_2}},          // https://support.apple.com/kb/SP751
>+    {RTCDeviceTypeIPad6, {kProfileHigh, kLevel4_2}},          // https://support.apple.com/kb/SP774
>+    {RTCDeviceTypeIPadAirWifi, {kProfileHigh, kLevel4_2}},    // https://support.apple.com/kb/SP692
>+    {RTCDeviceTypeIPadAirCellular,
>+     {kProfileHigh, kLevel4_2}},  // https://support.apple.com/kb/SP692
>+    {RTCDeviceTypeIPadAirWifiCellular,
>+     {kProfileHigh, kLevel4_2}},                               // https://support.apple.com/kb/SP692
>+    {RTCDeviceTypeIPadAir2, {kProfileHigh, kLevel4_2}},        // https://support.apple.com/kb/SP708
>+    {RTCDeviceTypeIPadMini2GWifi, {kProfileHigh, kLevel4_2}},  // https://support.apple.com/kb/SP693
>+    {RTCDeviceTypeIPadMini2GCellular,
>+     {kProfileHigh, kLevel4_2}},  // https://support.apple.com/kb/SP693
>+    {RTCDeviceTypeIPadMini2GWifiCellular,
>+     {kProfileHigh, kLevel4_2}},                               // https://support.apple.com/kb/SP693
>+    {RTCDeviceTypeIPadMini3, {kProfileHigh, kLevel4_2}},       // https://support.apple.com/kb/SP709
>+    {RTCDeviceTypeIPadMini4, {kProfileHigh, kLevel4_2}},       // https://support.apple.com/kb/SP725
>+    {RTCDeviceTypeIPadPro9Inch, {kProfileHigh, kLevel4_2}},    // https://support.apple.com/kb/SP739
>+    {RTCDeviceTypeIPadPro12Inch, {kProfileHigh, kLevel4_2}},   // https://support.apple.com/kb/sp723
>+    {RTCDeviceTypeIPadPro12Inch2, {kProfileHigh, kLevel4_2}},  // https://support.apple.com/kb/SP761
>+    {RTCDeviceTypeIPadPro10Inch, {kProfileHigh, kLevel4_2}},   // https://support.apple.com/kb/SP762
>+};
>+
>+absl::optional<ProfileLevelId> FindMaxSupportedProfileForDevice(RTCDeviceType deviceType) {
>+  const auto* result = std::find_if(std::begin(kH264MaxSupportedProfiles),
>+                                    std::end(kH264MaxSupportedProfiles),
>+                                    [deviceType](const SupportedH264Profile& supportedProfile) {
>+                                      return supportedProfile.deviceType == deviceType;
>+                                    });
>+  if (result != std::end(kH264MaxSupportedProfiles)) {
>+    return result->profile;
>+  }
>+  return absl::nullopt;
>+}
>+
>+}  // namespace
>+
>+@implementation UIDevice (H264Profile)
>+
>++ (absl::optional<webrtc::H264::ProfileLevelId>)maxSupportedH264Profile {
>+  return FindMaxSupportedProfileForDevice([self deviceType]);
>+}
>+
>+@end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/avfoundationformatmapper.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/avfoundationformatmapper.h
>deleted file mode 100644
>index 23ed7011269..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/avfoundationformatmapper.h
>+++ /dev/null
>@@ -1,29 +0,0 @@
>-/*
>- *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include <set>
>-
>-#import <AVFoundation/AVFoundation.h>
>-#import <Foundation/Foundation.h>
>-
>-#include "media/base/videocapturer.h"
>-
>-namespace webrtc {
>-// Mapping from AVCaptureDeviceFormat to cricket::VideoFormat for given input
>-// device.
>-std::set<cricket::VideoFormat> GetSupportedVideoFormatsForDevice(
>-    AVCaptureDevice* device);
>-
>-// Sets device format for the provided capture device. Returns YES/NO depending
>-// on success.
>-bool SetFormatForCaptureDevice(AVCaptureDevice* device,
>-                               AVCaptureSession* session,
>-                               const cricket::VideoFormat& format);
>-}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/avfoundationformatmapper.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/avfoundationformatmapper.mm
>deleted file mode 100644
>index 19afa337683..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/avfoundationformatmapper.mm
>+++ /dev/null
>@@ -1,135 +0,0 @@
>-/*
>- *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "avfoundationformatmapper.h"
>-
>-#import "WebRTC/RTCLogging.h"
>-
>-// TODO(denicija): add support for higher frame rates.
>-// See http://crbug/webrtc/6355 for more info.
>-static const int kFramesPerSecond = 30;
>-
>-static inline BOOL IsMediaSubTypeSupported(FourCharCode mediaSubType) {
>-  return (mediaSubType == kCVPixelFormatType_420YpCbCr8PlanarFullRange ||
>-          mediaSubType == kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange);
>-}
>-
>-static inline BOOL IsFrameRateWithinRange(int fps, AVFrameRateRange* range) {
>-  return range.minFrameRate <= fps && range.maxFrameRate >= fps;
>-}
>-
>-// Returns filtered array of device formats based on predefined constraints our
>-// stack imposes.
>-static NSArray<AVCaptureDeviceFormat*>* GetEligibleDeviceFormats(
>-    const AVCaptureDevice* device,
>-    int supportedFps) {
>-  NSMutableArray<AVCaptureDeviceFormat*>* eligibleDeviceFormats =
>-      [NSMutableArray array];
>-
>-  for (AVCaptureDeviceFormat* format in device.formats) {
>-    // Filter out subTypes that we currently don't support in the stack
>-    FourCharCode mediaSubType =
>-        CMFormatDescriptionGetMediaSubType(format.formatDescription);
>-    if (!IsMediaSubTypeSupported(mediaSubType)) {
>-      continue;
>-    }
>-
>-    // Filter out frame rate ranges that we currently don't support in the stack
>-    for (AVFrameRateRange* frameRateRange in format.videoSupportedFrameRateRanges) {
>-      if (IsFrameRateWithinRange(supportedFps, frameRateRange)) {
>-        [eligibleDeviceFormats addObject:format];
>-        break;
>-      }
>-    }
>-  }
>-
>-  return [eligibleDeviceFormats copy];
>-}
>-
>-// Mapping from cricket::VideoFormat to AVCaptureDeviceFormat.
>-static AVCaptureDeviceFormat* GetDeviceFormatForVideoFormat(
>-    const AVCaptureDevice* device,
>-    const cricket::VideoFormat& videoFormat) {
>-  AVCaptureDeviceFormat* desiredDeviceFormat = nil;
>-  NSArray<AVCaptureDeviceFormat*>* eligibleFormats =
>-      GetEligibleDeviceFormats(device, videoFormat.framerate());
>-
>-  for (AVCaptureDeviceFormat* deviceFormat in eligibleFormats) {
>-    CMVideoDimensions dimension =
>-        CMVideoFormatDescriptionGetDimensions(deviceFormat.formatDescription);
>-    FourCharCode mediaSubType =
>-        CMFormatDescriptionGetMediaSubType(deviceFormat.formatDescription);
>-
>-    if (videoFormat.width == dimension.width &&
>-        videoFormat.height == dimension.height) {
>-      if (mediaSubType == kCVPixelFormatType_420YpCbCr8BiPlanarFullRange) {
>-        // This is the preferred format so no need to wait for better option.
>-        return deviceFormat;
>-      } else {
>-        // This is good candidate, but let's wait for something better.
>-        desiredDeviceFormat = deviceFormat;
>-      }
>-    }
>-  }
>-
>-  return desiredDeviceFormat;
>-}
>-namespace webrtc {
>-std::set<cricket::VideoFormat> GetSupportedVideoFormatsForDevice(
>-    AVCaptureDevice* device) {
>-  std::set<cricket::VideoFormat> supportedFormats;
>-
>-  NSArray<AVCaptureDeviceFormat*>* eligibleFormats =
>-      GetEligibleDeviceFormats(device, kFramesPerSecond);
>-
>-  for (AVCaptureDeviceFormat* deviceFormat in eligibleFormats) {
>-    CMVideoDimensions dimension =
>-        CMVideoFormatDescriptionGetDimensions(deviceFormat.formatDescription);
>-    cricket::VideoFormat format = cricket::VideoFormat(
>-        dimension.width, dimension.height,
>-        cricket::VideoFormat::FpsToInterval(kFramesPerSecond),
>-        cricket::FOURCC_NV12);
>-    supportedFormats.insert(format);
>-  }
>-
>-  return supportedFormats;
>-}
>-
>-bool SetFormatForCaptureDevice(AVCaptureDevice* device,
>-                               AVCaptureSession* session,
>-                               const cricket::VideoFormat& format) {
>-  AVCaptureDeviceFormat* deviceFormat =
>-      GetDeviceFormatForVideoFormat(device, format);
>-  const int fps = cricket::VideoFormat::IntervalToFps(format.interval);
>-
>-  NSError* error = nil;
>-  bool success = true;
>-  [session beginConfiguration];
>-  if ([device lockForConfiguration:&error]) {
>-    @try {
>-      device.activeFormat = deviceFormat;
>-      device.activeVideoMinFrameDuration = CMTimeMake(1, fps);
>-    } @catch (NSException* exception) {
>-      RTCLogError(@"Failed to set active format!\n User info:%@",
>-                  exception.userInfo);
>-      success = false;
>-    }
>-
>-    [device unlockForConfiguration];
>-  } else {
>-    RTCLogError(@"Failed to lock device %@. Error: %@", device, error.userInfo);
>-    success = false;
>-  }
>-  [session commitConfiguration];
>-
>-  return success;
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/avfoundationvideocapturer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/avfoundationvideocapturer.h
>deleted file mode 100644
>index 459a047ec1b..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/avfoundationvideocapturer.h
>+++ /dev/null
>@@ -1,72 +0,0 @@
>-/*
>- *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef SDK_OBJC_FRAMEWORK_CLASSES_VIDEO_AVFOUNDATIONVIDEOCAPTURER_H_
>-#define SDK_OBJC_FRAMEWORK_CLASSES_VIDEO_AVFOUNDATIONVIDEOCAPTURER_H_
>-
>-#import <AVFoundation/AVFoundation.h>
>-
>-#include "api/video/video_frame.h"
>-#include "common_video/include/i420_buffer_pool.h"
>-#include "media/base/videocapturer.h"
>-
>-@class RTCAVFoundationVideoCapturerInternal;
>-
>-namespace rtc {
>-class Thread;
>-}  // namespace rtc
>-
>-namespace webrtc {
>-
>-class AVFoundationVideoCapturer : public cricket::VideoCapturer {
>- public:
>-  AVFoundationVideoCapturer();
>-  ~AVFoundationVideoCapturer();
>-
>-  cricket::CaptureState Start(const cricket::VideoFormat& format) override;
>-  void Stop() override;
>-  bool IsRunning() override;
>-  bool IsScreencast() const override {
>-    return false;
>-  }
>-  bool GetPreferredFourccs(std::vector<uint32_t> *fourccs) override {
>-    fourccs->push_back(cricket::FOURCC_NV12);
>-    return true;
>-  }
>-
>-  // Returns the active capture session. Calls to the capture session should
>-  // occur on the RTCDispatcherTypeCaptureSession queue in RTCDispatcher.
>-  AVCaptureSession* GetCaptureSession();
>-
>-  // Returns whether the rear-facing camera can be used.
>-  // e.g. It can't be used because it doesn't exist.
>-  bool CanUseBackCamera() const;
>-
>-  // Switches the camera being used (either front or back).
>-  void SetUseBackCamera(bool useBackCamera);
>-  bool GetUseBackCamera() const;
>-
>-  // Converts the sample buffer into a cricket::CapturedFrame and signals the
>-  // frame for capture.
>-  void CaptureSampleBuffer(CMSampleBufferRef sample_buffer,
>-                           webrtc::VideoRotation rotation);
>-
>-  // Called to adjust the size of output frames to supplied |width| and
>-  // |height|. Also drops frames to make the output match |fps|.
>-  void AdaptOutputFormat(int width, int height, int fps);
>-
>- private:
>-  RTCAVFoundationVideoCapturerInternal *_capturer;
>-  webrtc::I420BufferPool _buffer_pool;
>-};  // AVFoundationVideoCapturer
>-
>-}  // namespace webrtc
>-
>-#endif  // SDK_OBJC_FRAMEWORK_CLASSES_VIDEO_AVFOUNDATIONVIDEOCAPTURER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/avfoundationvideocapturer.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/avfoundationvideocapturer.mm
>deleted file mode 100644
>index 77cb88b5ef6..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/avfoundationvideocapturer.mm
>+++ /dev/null
>@@ -1,179 +0,0 @@
>-/*
>- *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "avfoundationvideocapturer.h"
>-
>-#import <AVFoundation/AVFoundation.h>
>-
>-#import "RTCAVFoundationVideoCapturerInternal.h"
>-#import "RTCDispatcher+Private.h"
>-#import "WebRTC/RTCLogging.h"
>-#import "WebRTC/RTCVideoFrameBuffer.h"
>-
>-#include "avfoundationformatmapper.h"
>-
>-#include "api/video/video_rotation.h"
>-#include "rtc_base/bind.h"
>-#include "rtc_base/checks.h"
>-#include "rtc_base/logging.h"
>-#include "rtc_base/thread.h"
>-#include "sdk/objc/Framework/Classes/Video/objc_frame_buffer.h"
>-
>-namespace webrtc {
>-
>-enum AVFoundationVideoCapturerMessageType : uint32_t {
>-  kMessageTypeFrame,
>-};
>-
>-AVFoundationVideoCapturer::AVFoundationVideoCapturer() : _capturer(nil) {
>-  _capturer =
>-      [[RTCAVFoundationVideoCapturerInternal alloc] initWithCapturer:this];
>-
>-  std::set<cricket::VideoFormat> front_camera_video_formats =
>-      GetSupportedVideoFormatsForDevice([_capturer frontCaptureDevice]);
>-  std::set<cricket::VideoFormat> back_camera_video_formats =
>-      GetSupportedVideoFormatsForDevice([_capturer backCaptureDevice]);
>-  std::vector<cricket::VideoFormat> intersection_video_formats;
>-  if (back_camera_video_formats.empty()) {
>-    intersection_video_formats.assign(front_camera_video_formats.begin(),
>-                                      front_camera_video_formats.end());
>-
>-  } else if (front_camera_video_formats.empty()) {
>-    intersection_video_formats.assign(back_camera_video_formats.begin(),
>-                                      back_camera_video_formats.end());
>-  } else {
>-    std::set_intersection(
>-        front_camera_video_formats.begin(), front_camera_video_formats.end(),
>-        back_camera_video_formats.begin(), back_camera_video_formats.end(),
>-        std::back_inserter(intersection_video_formats));
>-  }
>-  SetSupportedFormats(intersection_video_formats);
>-}
>-
>-AVFoundationVideoCapturer::~AVFoundationVideoCapturer() {
>-  _capturer = nil;
>-}
>-
>-cricket::CaptureState AVFoundationVideoCapturer::Start(
>-    const cricket::VideoFormat& format) {
>-  if (!_capturer) {
>-    RTC_LOG(LS_ERROR) << "Failed to create AVFoundation capturer.";
>-    return cricket::CaptureState::CS_FAILED;
>-  }
>-  if (_capturer.isRunning) {
>-    RTC_LOG(LS_ERROR) << "The capturer is already running.";
>-    return cricket::CaptureState::CS_FAILED;
>-  }
>-
>-  AVCaptureDevice* device = [_capturer getActiveCaptureDevice];
>-  AVCaptureSession* session = _capturer.captureSession;
>-
>-  if (!SetFormatForCaptureDevice(device, session, format)) {
>-    return cricket::CaptureState::CS_FAILED;
>-  }
>-
>-  SetCaptureFormat(&format);
>-  // This isn't super accurate because it takes a while for the AVCaptureSession
>-  // to spin up, and this call returns async.
>-  // TODO(tkchin): make this better.
>-  [_capturer start];
>-  SetCaptureState(cricket::CaptureState::CS_RUNNING);
>-
>-  return cricket::CaptureState::CS_STARTING;
>-}
>-
>-void AVFoundationVideoCapturer::Stop() {
>-  [_capturer stop];
>-  SetCaptureFormat(NULL);
>-}
>-
>-bool AVFoundationVideoCapturer::IsRunning() {
>-  return _capturer.isRunning;
>-}
>-
>-AVCaptureSession* AVFoundationVideoCapturer::GetCaptureSession() {
>-  return _capturer.captureSession;
>-}
>-
>-bool AVFoundationVideoCapturer::CanUseBackCamera() const {
>-  return _capturer.canUseBackCamera;
>-}
>-
>-void AVFoundationVideoCapturer::SetUseBackCamera(bool useBackCamera) {
>-  _capturer.useBackCamera = useBackCamera;
>-}
>-
>-bool AVFoundationVideoCapturer::GetUseBackCamera() const {
>-  return _capturer.useBackCamera;
>-}
>-
>-void AVFoundationVideoCapturer::AdaptOutputFormat(int width, int height, int fps) {
>-  cricket::VideoFormat format(width, height, cricket::VideoFormat::FpsToInterval(fps), 0);
>-  video_adapter()->OnOutputFormatRequest(format);
>-}
>-
>-void AVFoundationVideoCapturer::CaptureSampleBuffer(
>-    CMSampleBufferRef sample_buffer, VideoRotation rotation) {
>-  if (CMSampleBufferGetNumSamples(sample_buffer) != 1 ||
>-      !CMSampleBufferIsValid(sample_buffer) ||
>-      !CMSampleBufferDataIsReady(sample_buffer)) {
>-    return;
>-  }
>-
>-  CVImageBufferRef image_buffer = CMSampleBufferGetImageBuffer(sample_buffer);
>-  if (image_buffer == NULL) {
>-    return;
>-  }
>-
>-  int captured_width = CVPixelBufferGetWidth(image_buffer);
>-  int captured_height = CVPixelBufferGetHeight(image_buffer);
>-
>-  int adapted_width;
>-  int adapted_height;
>-  int crop_width;
>-  int crop_height;
>-  int crop_x;
>-  int crop_y;
>-  int64_t translated_camera_time_us;
>-
>-  if (!AdaptFrame(captured_width, captured_height,
>-                  rtc::TimeNanos() / rtc::kNumNanosecsPerMicrosec,
>-                  rtc::TimeMicros(), &adapted_width, &adapted_height,
>-                  &crop_width, &crop_height, &crop_x, &crop_y,
>-                  &translated_camera_time_us)) {
>-    return;
>-  }
>-
>-  RTCCVPixelBuffer* rtcPixelBuffer = [[RTCCVPixelBuffer alloc] initWithPixelBuffer:image_buffer
>-                                                                      adaptedWidth:adapted_width
>-                                                                     adaptedHeight:adapted_height
>-                                                                         cropWidth:crop_width
>-                                                                        cropHeight:crop_height
>-                                                                             cropX:crop_x
>-                                                                             cropY:crop_y];
>-  rtc::scoped_refptr<VideoFrameBuffer> buffer =
>-      new rtc::RefCountedObject<ObjCFrameBuffer>(rtcPixelBuffer);
>-
>-  // Applying rotation is only supported for legacy reasons and performance is
>-  // not critical here.
>-  if (apply_rotation() && rotation != kVideoRotation_0) {
>-    buffer = I420Buffer::Rotate(*buffer->ToI420(), rotation);
>-    if (rotation == kVideoRotation_90 || rotation == kVideoRotation_270) {
>-      std::swap(captured_width, captured_height);
>-    }
>-
>-    rotation = kVideoRotation_0;
>-  }
>-
>-  OnFrame(webrtc::VideoFrame(buffer, rotation, translated_camera_time_us),
>-          captured_width, captured_height);
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/corevideo_frame_buffer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/corevideo_frame_buffer.cc
>deleted file mode 100644
>index 3ddaa71e722..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/corevideo_frame_buffer.cc
>+++ /dev/null
>@@ -1,178 +0,0 @@
>-/*
>- *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "sdk/objc/Framework/Classes/Video/corevideo_frame_buffer.h"
>-
>-#include "api/video/i420_buffer.h"
>-#include "common_video/libyuv/include/webrtc_libyuv.h"
>-#include "rtc_base/checks.h"
>-#include "rtc_base/logging.h"
>-#include "third_party/libyuv/include/libyuv/convert.h"
>-
>-namespace webrtc {
>-
>-CoreVideoFrameBuffer::CoreVideoFrameBuffer(CVPixelBufferRef pixel_buffer,
>-                                           int adapted_width,
>-                                           int adapted_height,
>-                                           int crop_width,
>-                                           int crop_height,
>-                                           int crop_x,
>-                                           int crop_y)
>-    : pixel_buffer_(pixel_buffer),
>-      width_(adapted_width),
>-      height_(adapted_height),
>-      buffer_width_(CVPixelBufferGetWidth(pixel_buffer)),
>-      buffer_height_(CVPixelBufferGetHeight(pixel_buffer)),
>-      crop_width_(crop_width),
>-      crop_height_(crop_height),
>-      // Can only crop at even pixels.
>-      crop_x_(crop_x & ~1),
>-      crop_y_(crop_y & ~1) {
>-  CVBufferRetain(pixel_buffer_);
>-}
>-
>-CoreVideoFrameBuffer::CoreVideoFrameBuffer(CVPixelBufferRef pixel_buffer)
>-    : pixel_buffer_(pixel_buffer),
>-      width_(CVPixelBufferGetWidth(pixel_buffer)),
>-      height_(CVPixelBufferGetHeight(pixel_buffer)),
>-      buffer_width_(width_),
>-      buffer_height_(height_),
>-      crop_width_(width_),
>-      crop_height_(height_),
>-      crop_x_(0),
>-      crop_y_(0) {
>-  CVBufferRetain(pixel_buffer_);
>-}
>-
>-CoreVideoFrameBuffer::~CoreVideoFrameBuffer() {
>-  CVBufferRelease(pixel_buffer_);
>-}
>-
>-VideoFrameBuffer::Type CoreVideoFrameBuffer::type() const {
>-  return Type::kNative;
>-}
>-
>-int CoreVideoFrameBuffer::width() const {
>-  return width_;
>-}
>-
>-int CoreVideoFrameBuffer::height() const {
>-  return height_;
>-}
>-
>-rtc::scoped_refptr<I420BufferInterface> CoreVideoFrameBuffer::ToI420() {
>-  const OSType pixel_format = CVPixelBufferGetPixelFormatType(pixel_buffer_);
>-  RTC_DCHECK(pixel_format == kCVPixelFormatType_420YpCbCr8BiPlanarFullRange ||
>-             pixel_format == kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange);
>-
>-  CVPixelBufferLockBaseAddress(pixel_buffer_, kCVPixelBufferLock_ReadOnly);
>-  const uint8_t* src_y = static_cast<const uint8_t*>(
>-      CVPixelBufferGetBaseAddressOfPlane(pixel_buffer_, 0));
>-  const int src_y_stride = CVPixelBufferGetBytesPerRowOfPlane(pixel_buffer_, 0);
>-  const uint8_t* src_uv = static_cast<const uint8_t*>(
>-      CVPixelBufferGetBaseAddressOfPlane(pixel_buffer_, 1));
>-  const int src_uv_stride =
>-      CVPixelBufferGetBytesPerRowOfPlane(pixel_buffer_, 1);
>-
>-  // Crop just by modifying pointers.
>-  src_y += src_y_stride * crop_y_ + crop_x_;
>-  src_uv += src_uv_stride * (crop_y_ / 2) + crop_x_;
>-
>-  // TODO(magjed): Use a frame buffer pool.
>-  NV12ToI420Scaler nv12_to_i420_scaler;
>-  rtc::scoped_refptr<I420Buffer> buffer =
>-      new rtc::RefCountedObject<I420Buffer>(width_, height_);
>-  nv12_to_i420_scaler.NV12ToI420Scale(
>-      src_y, src_y_stride,
>-      src_uv, src_uv_stride,
>-      crop_width_, crop_height_,
>-      buffer->MutableDataY(), buffer->StrideY(),
>-      buffer->MutableDataU(), buffer->StrideU(),
>-      buffer->MutableDataV(), buffer->StrideV(),
>-      buffer->width(), buffer->height());
>-
>-  CVPixelBufferUnlockBaseAddress(pixel_buffer_, kCVPixelBufferLock_ReadOnly);
>-
>-  return buffer;
>-}
>-
>-bool CoreVideoFrameBuffer::RequiresCropping() const {
>-  return crop_width_ != buffer_width_ || crop_height_ != buffer_height_;
>-}
>-
>-bool CoreVideoFrameBuffer::CropAndScaleTo(
>-    std::vector<uint8_t>* tmp_buffer,
>-    CVPixelBufferRef output_pixel_buffer) const {
>-  // Prepare output pointers.
>-  RTC_DCHECK_EQ(CVPixelBufferGetPixelFormatType(output_pixel_buffer),
>-                kCVPixelFormatType_420YpCbCr8BiPlanarFullRange);
>-  CVReturn cv_ret = CVPixelBufferLockBaseAddress(output_pixel_buffer, 0);
>-  if (cv_ret != kCVReturnSuccess) {
>-    RTC_LOG(LS_ERROR) << "Failed to lock base address: " << cv_ret;
>-    return false;
>-  }
>-  const int dst_width = CVPixelBufferGetWidth(output_pixel_buffer);
>-  const int dst_height = CVPixelBufferGetHeight(output_pixel_buffer);
>-  uint8_t* dst_y = reinterpret_cast<uint8_t*>(
>-      CVPixelBufferGetBaseAddressOfPlane(output_pixel_buffer, 0));
>-  const int dst_y_stride =
>-      CVPixelBufferGetBytesPerRowOfPlane(output_pixel_buffer, 0);
>-  uint8_t* dst_uv = reinterpret_cast<uint8_t*>(
>-      CVPixelBufferGetBaseAddressOfPlane(output_pixel_buffer, 1));
>-  const int dst_uv_stride =
>-      CVPixelBufferGetBytesPerRowOfPlane(output_pixel_buffer, 1);
>-
>-  // Prepare source pointers.
>-  const OSType src_pixel_format =
>-      CVPixelBufferGetPixelFormatType(pixel_buffer_);
>-  RTC_DCHECK(
>-      src_pixel_format == kCVPixelFormatType_420YpCbCr8BiPlanarFullRange ||
>-      src_pixel_format == kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange);
>-  CVPixelBufferLockBaseAddress(pixel_buffer_, kCVPixelBufferLock_ReadOnly);
>-  const uint8_t* src_y = static_cast<const uint8_t*>(
>-      CVPixelBufferGetBaseAddressOfPlane(pixel_buffer_, 0));
>-  const int src_y_stride = CVPixelBufferGetBytesPerRowOfPlane(pixel_buffer_, 0);
>-  const uint8_t* src_uv = static_cast<const uint8_t*>(
>-      CVPixelBufferGetBaseAddressOfPlane(pixel_buffer_, 1));
>-  const int src_uv_stride =
>-      CVPixelBufferGetBytesPerRowOfPlane(pixel_buffer_, 1);
>-
>-  // Crop just by modifying pointers.
>-  src_y += src_y_stride * crop_y_ + crop_x_;
>-  src_uv += src_uv_stride * (crop_y_ / 2) + crop_x_;
>-
>-  if (crop_width_ == dst_width && crop_height_ == dst_height) {
>-    tmp_buffer->clear();
>-    tmp_buffer->shrink_to_fit();
>-  } else {
>-    const int src_chroma_width = (crop_width_ + 1) / 2;
>-    const int src_chroma_height = (crop_height_ + 1) / 2;
>-    const int dst_chroma_width = (dst_width + 1) / 2;
>-    const int dst_chroma_height = (dst_height + 1) / 2;
>-    tmp_buffer->resize(src_chroma_width * src_chroma_height * 2 +
>-                       dst_chroma_width * dst_chroma_height * 2);
>-    tmp_buffer->shrink_to_fit();
>-  }
>-
>-  NV12Scale(tmp_buffer->data(),
>-            src_y, src_y_stride,
>-            src_uv, src_uv_stride,
>-            crop_width_, crop_height_,
>-            dst_y, dst_y_stride,
>-            dst_uv, dst_uv_stride,
>-            dst_width, dst_height);
>-
>-  CVPixelBufferUnlockBaseAddress(pixel_buffer_, kCVPixelBufferLock_ReadOnly);
>-  CVPixelBufferUnlockBaseAddress(output_pixel_buffer, 0);
>-
>-  return true;
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/corevideo_frame_buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/corevideo_frame_buffer.h
>deleted file mode 100644
>index f0b094b9842..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/corevideo_frame_buffer.h
>+++ /dev/null
>@@ -1,66 +0,0 @@
>-/*
>- *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef SDK_OBJC_FRAMEWORK_CLASSES_VIDEO_COREVIDEO_FRAME_BUFFER_H_
>-#define SDK_OBJC_FRAMEWORK_CLASSES_VIDEO_COREVIDEO_FRAME_BUFFER_H_
>-
>-#include <CoreVideo/CoreVideo.h>
>-
>-#include <vector>
>-
>-#include "common_video/include/video_frame_buffer.h"
>-
>-namespace webrtc {
>-
>-class CoreVideoFrameBuffer : public VideoFrameBuffer {
>- public:
>-  explicit CoreVideoFrameBuffer(CVPixelBufferRef pixel_buffer);
>-  CoreVideoFrameBuffer(CVPixelBufferRef pixel_buffer,
>-                       int adapted_width,
>-                       int adapted_height,
>-                       int crop_width,
>-                       int crop_height,
>-                       int crop_x,
>-                       int crop_y);
>-  ~CoreVideoFrameBuffer() override;
>-
>-  CVPixelBufferRef pixel_buffer() { return pixel_buffer_; }
>-
>-  // Returns true if the internal pixel buffer needs to be cropped.
>-  bool RequiresCropping() const;
>-  // Crop and scales the internal pixel buffer to the output pixel buffer. The
>-  // tmp buffer is used for intermediary splitting the UV channels. This
>-  // function returns true if successful.
>-  bool CropAndScaleTo(std::vector<uint8_t>* tmp_buffer,
>-                      CVPixelBufferRef output_pixel_buffer) const;
>-
>- private:
>-  Type type() const override;
>-  int width() const override;
>-  int height() const override;
>-  rtc::scoped_refptr<I420BufferInterface> ToI420() override;
>-
>-  CVPixelBufferRef pixel_buffer_;
>-  // buffer_width/height is the actual pixel buffer resolution. The
>-  // width_/height_ is the resolution we will scale to in ToI420(). Cropping
>-  // happens before scaling, so: buffer_width >= crop_width >= width().
>-  const int width_;
>-  const int height_;
>-  const int buffer_width_;
>-  const int buffer_height_;
>-  const int crop_width_;
>-  const int crop_height_;
>-  const int crop_x_;
>-  const int crop_y_;
>-};
>-
>-}  // namespace webrtc
>-
>-#endif  // SDK_OBJC_FRAMEWORK_CLASSES_VIDEO_COREVIDEO_FRAME_BUFFER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoDecoderH264.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoDecoderH264.mm
>index 1e102cf62a7..2871b8a5ef1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoDecoderH264.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoDecoderH264.mm
>@@ -78,7 +78,7 @@ @implementation RTCVideoDecoderH264 {
> 
> - (instancetype)init {
>   if (self = [super init]) {
>-#if defined(WEBRTC_IOS)
>+#if defined(WEBRTC_IOS) && !defined(RTC_APPRTCMOBILE_BROADCAST_EXTENSION)
>     [RTCUIApplicationStatusObserver prepareForUse];
>     _error = noErr;
> #endif
>@@ -92,16 +92,19 @@ - (void)dealloc {
>   [self setVideoFormat:nullptr];
> }
> 
>+- (NSInteger)startDecodeWithNumberOfCores:(int)numberOfCores {
>+  return WEBRTC_VIDEO_CODEC_OK;
>+}
>+
> - (NSInteger)startDecodeWithSettings:(RTCVideoEncoderSettings *)settings
>                        numberOfCores:(int)numberOfCores {
>   return WEBRTC_VIDEO_CODEC_OK;
> }
> 
> - (NSInteger)decode:(RTCEncodedImage *)inputImage
>-          missingFrames:(BOOL)missingFrames
>-    fragmentationHeader:(RTCRtpFragmentationHeader *)fragmentationHeader
>-      codecSpecificInfo:(nullable id<RTCCodecSpecificInfo>)info
>-           renderTimeMs:(int64_t)renderTimeMs {
>+        missingFrames:(BOOL)missingFrames
>+    codecSpecificInfo:(nullable id<RTCCodecSpecificInfo>)info
>+         renderTimeMs:(int64_t)renderTimeMs {
>   RTC_DCHECK(inputImage.buffer);
> 
>   if (_error != noErr) {
>@@ -110,7 +113,7 @@ - (NSInteger)decode:(RTCEncodedImage *)inputImage
>     return WEBRTC_VIDEO_CODEC_ERROR;
>   }
> 
>-#if defined(WEBRTC_IOS)
>+#if defined(WEBRTC_IOS) && !defined(RTC_APPRTCMOBILE_BROADCAST_EXTENSION)
>   if (![[RTCUIApplicationStatusObserver sharedInstance] isApplicationActive]) {
>     // Ignore all decode requests when app isn't active. In this state, the
>     // hardware decoder has been invalidated by the OS.
>@@ -120,21 +123,17 @@ - (NSInteger)decode:(RTCEncodedImage *)inputImage
>     return WEBRTC_VIDEO_CODEC_NO_OUTPUT;
>   }
> #endif
>-  if (webrtc::H264AnnexBBufferHasVideoFormatDescription((uint8_t *)inputImage.buffer.bytes,
>-                                                        inputImage.buffer.length)) {
>-    rtc::ScopedCFTypeRef<CMVideoFormatDescriptionRef> inputFormat =
>-        rtc::ScopedCF(webrtc::CreateVideoFormatDescription((uint8_t *)inputImage.buffer.bytes,
>-                                                           inputImage.buffer.length));
>-    if (inputFormat) {
>-      // Check if the video format has changed, and reinitialize decoder if
>-      // needed.
>-      if (!CMFormatDescriptionEqual(inputFormat.get(), _videoFormat)) {
>-        [self setVideoFormat:inputFormat.get()];
>-
>-        int resetDecompressionSessionError = [self resetDecompressionSession];
>-        if (resetDecompressionSessionError != WEBRTC_VIDEO_CODEC_OK) {
>-          return resetDecompressionSessionError;
>-        }
>+  rtc::ScopedCFTypeRef<CMVideoFormatDescriptionRef> inputFormat =
>+      rtc::ScopedCF(webrtc::CreateVideoFormatDescription((uint8_t *)inputImage.buffer.bytes,
>+                                                         inputImage.buffer.length));
>+  if (inputFormat) {
>+    // Check if the video format has changed, and reinitialize decoder if
>+    // needed.
>+    if (!CMFormatDescriptionEqual(inputFormat.get(), _videoFormat)) {
>+      [self setVideoFormat:inputFormat.get()];
>+      int resetDecompressionSessionError = [self resetDecompressionSession];
>+      if (resetDecompressionSessionError != WEBRTC_VIDEO_CODEC_OK) {
>+        return resetDecompressionSessionError;
>       }
>     }
>   }
>@@ -236,12 +235,13 @@ - (int)resetDecompressionSession {
>     pixelFormat = nullptr;
>   }
>   VTDecompressionOutputCallbackRecord record = {
>-      decompressionOutputCallback, nullptr,
>+      decompressionOutputCallback, (__bridge void *)self,
>   };
>   OSStatus status = VTDecompressionSessionCreate(
>       nullptr, _videoFormat, nullptr, attributes, &record, &_decompressionSession);
>   CFRelease(attributes);
>   if (status != noErr) {
>+    RTC_LOG(LS_ERROR) << "Failed to create decompression session: " << status;
>     [self destroyDecompressionSession];
>     return WEBRTC_VIDEO_CODEC_ERROR;
>   }
>@@ -260,7 +260,9 @@ - (void)configureDecompressionSession {
> - (void)destroyDecompressionSession {
>   if (_decompressionSession) {
> #if defined(WEBRTC_IOS)
>-    VTDecompressionSessionWaitForAsynchronousFrames(_decompressionSession);
>+    if ([UIDevice isIOS11OrLater]) {
>+      VTDecompressionSessionWaitForAsynchronousFrames(_decompressionSession);
>+    }
> #endif
>     VTDecompressionSessionInvalidate(_decompressionSession);
>     CFRelease(_decompressionSession);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoEncoderH264.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoEncoderH264.mm
>index 16a841d9458..d47818f763f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoEncoderH264.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoEncoderH264.mm
>@@ -210,7 +210,7 @@ void compressionOutputCallback(void *encoder,
> // returned. The user must initialize the encoder with a resolution and
> // framerate conforming to the selected H264 level regardless.
> CFStringRef ExtractProfile(webrtc::SdpVideoFormat videoFormat) {
>-  const rtc::Optional<webrtc::H264::ProfileLevelId> profile_level_id =
>+  const absl::optional<webrtc::H264::ProfileLevelId> profile_level_id =
>       webrtc::H264::ParseSdpProfileLevelId(videoFormat.parameters);
>   RTC_DCHECK(profile_level_id);
>   switch (profile_level_id->profile) {
>@@ -339,8 +339,7 @@ @implementation RTCVideoEncoderH264 {
> - (instancetype)initWithCodecInfo:(RTCVideoCodecInfo *)codecInfo {
>   if (self = [super init]) {
>     _codecInfo = codecInfo;
>-    _bitrateAdjuster.reset(new webrtc::BitrateAdjuster(
>-        webrtc::Clock::GetRealTimeClock(), .5, .95));
>+    _bitrateAdjuster.reset(new webrtc::BitrateAdjuster(.5, .95));
>     _packetizationMode = RTCH264PacketizationModeNonInterleaved;
>     _profile = ExtractProfile([codecInfo nativeSdpVideoFormat]);
>     RTC_LOG(LS_INFO) << "Using profile " << CFStringToString(_profile);
>@@ -859,7 +858,7 @@ - (void)frameWasEncoded:(OSStatus)status
>   frame.rotation = rotation;
>   frame.contentType = (_mode == RTCVideoCodecModeScreensharing) ? RTCVideoContentTypeScreenshare :
>                                                                   RTCVideoContentTypeUnspecified;
>-  frame.flags = webrtc::TimingFrameFlags::kInvalid;
>+  frame.flags = webrtc::VideoSendTiming::kInvalid;
> 
>   int qp;
>   _h264BitstreamParser.ParseBitstream(buffer->data(), buffer->size());
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/helpers.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/helpers.cc
>index d22a6f9087a..ac957f1b497 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/helpers.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/helpers.cc
>@@ -35,56 +35,56 @@ std::string CFStringToString(const CFStringRef cf_string) {
> }
> 
> // Convenience function for setting a VT property.
>-void SetVTSessionProperty(CompressionSessionRef session,
>+void SetVTSessionProperty(VTSessionRef session,
>                           CFStringRef key,
>                           int32_t value) {
>   CFNumberRef cfNum =
>       CFNumberCreate(kCFAllocatorDefault, kCFNumberSInt32Type, &value);
>-  OSStatus status = CompressionSessionSetProperty(session, key, cfNum);
>+  OSStatus status = VTSessionSetProperty(session, key, cfNum);
>   CFRelease(cfNum);
>   if (status != noErr) {
>     std::string key_string = CFStringToString(key);
>-    RTC_LOG(LS_ERROR) << "CompressionSessionSetProperty failed to set: " << key_string
>+    RTC_LOG(LS_ERROR) << "VTSessionSetProperty failed to set: " << key_string
>                       << " to " << value << ": " << status;
>   }
> }
> 
> // Convenience function for setting a VT property.
>-void SetVTSessionProperty(CompressionSessionRef session,
>+void SetVTSessionProperty(VTSessionRef session,
>                           CFStringRef key,
>                           uint32_t value) {
>   int64_t value_64 = value;
>   CFNumberRef cfNum =
>       CFNumberCreate(kCFAllocatorDefault, kCFNumberSInt64Type, &value_64);
>-  OSStatus status = CompressionSessionSetProperty(session, key, cfNum);
>+  OSStatus status = VTSessionSetProperty(session, key, cfNum);
>   CFRelease(cfNum);
>   if (status != noErr) {
>     std::string key_string = CFStringToString(key);
>-    RTC_LOG(LS_ERROR) << "CompressionSessionSetProperty failed to set: " << key_string
>+    RTC_LOG(LS_ERROR) << "VTSessionSetProperty failed to set: " << key_string
>                       << " to " << value << ": " << status;
>   }
> }
> 
> // Convenience function for setting a VT property.
>-void SetVTSessionProperty(CompressionSessionRef session, CFStringRef key, bool value) {
>+void SetVTSessionProperty(VTSessionRef session, CFStringRef key, bool value) {
>   CFBooleanRef cf_bool = (value) ? kCFBooleanTrue : kCFBooleanFalse;
>-  OSStatus status = CompressionSessionSetProperty(session, key, cf_bool);
>+  OSStatus status = VTSessionSetProperty(session, key, cf_bool);
>   if (status != noErr) {
>     std::string key_string = CFStringToString(key);
>-    RTC_LOG(LS_ERROR) << "CompressionSessionSetProperty failed to set: " << key_string
>+    RTC_LOG(LS_ERROR) << "VTSessionSetProperty failed to set: " << key_string
>                       << " to " << value << ": " << status;
>   }
> }
> 
> // Convenience function for setting a VT property.
>-void SetVTSessionProperty(CompressionSessionRef session,
>+void SetVTSessionProperty(VTSessionRef session,
>                           CFStringRef key,
>                           CFStringRef value) {
>-  OSStatus status = CompressionSessionSetProperty(session, key, value);
>+  OSStatus status = VTSessionSetProperty(session, key, value);
>   if (status != noErr) {
>     std::string key_string = CFStringToString(key);
>     std::string val_string = CFStringToString(value);
>-    RTC_LOG(LS_ERROR) << "CompressionSessionSetProperty failed to set: " << key_string
>+    RTC_LOG(LS_ERROR) << "VTSessionSetProperty failed to set: " << key_string
>                       << " to " << val_string << ": " << status;
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/helpers.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/helpers.h
>index 0c669e8bce6..0683ea79e56 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/helpers.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/helpers.h
>@@ -16,8 +16,6 @@
> #include <VideoToolbox/VideoToolbox.h>
> #include <string>
> 
>-#include "sdk/WebKit/EncoderUtilities.h"
>-
> // Convenience function for creating a dictionary.
> inline CFDictionaryRef CreateCFTypeDictionary(CFTypeRef* keys,
>                                               CFTypeRef* values,
>@@ -31,18 +29,18 @@ inline CFDictionaryRef CreateCFTypeDictionary(CFTypeRef* keys,
> std::string CFStringToString(const CFStringRef cf_string);
> 
> // Convenience function for setting a VT property.
>-void SetVTSessionProperty(CompressionSessionRef session, CFStringRef key, int32_t value);
>+void SetVTSessionProperty(VTSessionRef session, CFStringRef key, int32_t value);
> 
> // Convenience function for setting a VT property.
>-void SetVTSessionProperty(CompressionSessionRef session,
>+void SetVTSessionProperty(VTSessionRef session,
>                           CFStringRef key,
>                           uint32_t value);
> 
> // Convenience function for setting a VT property.
>-void SetVTSessionProperty(CompressionSessionRef session, CFStringRef key, bool value);
>+void SetVTSessionProperty(VTSessionRef session, CFStringRef key, bool value);
> 
> // Convenience function for setting a VT property.
>-void SetVTSessionProperty(CompressionSessionRef session,
>+void SetVTSessionProperty(VTSessionRef session,
>                           CFStringRef key,
>                           CFStringRef value);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/nalu_rewriter.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/nalu_rewriter.cc
>index f6ee10688e5..479ef3c6bba 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/nalu_rewriter.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/nalu_rewriter.cc
>@@ -33,7 +33,7 @@ bool H264CMSampleBufferToAnnexBBuffer(
>     CMSampleBufferRef avcc_sample_buffer,
>     bool is_keyframe,
>     rtc::Buffer* annexb_buffer,
>-    std::unique_ptr<RTPFragmentationHeader> *out_header) {
>+    std::unique_ptr<RTPFragmentationHeader>* out_header) {
>   RTC_DCHECK(avcc_sample_buffer);
>   RTC_DCHECK(out_header);
>   out_header->reset(nullptr);
>@@ -167,11 +167,10 @@ bool H264AnnexBBufferToCMSampleBuffer(const uint8_t* annexb_buffer,
>   *out_sample_buffer = nullptr;
> 
>   AnnexBBufferReader reader(annexb_buffer, annexb_buffer_size);
>-  if (H264AnnexBBufferHasVideoFormatDescription(annexb_buffer,
>-                                                annexb_buffer_size)) {
>-    // Advance past the SPS and PPS.
>-    const uint8_t* data = nullptr;
>-    size_t data_len = 0;
>+  if (reader.SeekToNextNaluOfType(kSps)) {
>+    // Buffer contains an SPS NALU - skip it and the following PPS
>+    const uint8_t* data;
>+    size_t data_len;
>     if (!reader.ReadNalu(&data, &data_len)) {
>       RTC_LOG(LS_ERROR) << "Failed to read SPS";
>       return false;
>@@ -180,6 +179,9 @@ bool H264AnnexBBufferToCMSampleBuffer(const uint8_t* annexb_buffer,
>       RTC_LOG(LS_ERROR) << "Failed to read PPS";
>       return false;
>     }
>+  } else {
>+    // No SPS NALU - start reading from the first NALU in the buffer
>+    reader.SeekToStart();
>   }
> 
>   // Allocate memory as a block buffer.
>@@ -246,47 +248,15 @@ bool H264AnnexBBufferToCMSampleBuffer(const uint8_t* annexb_buffer,
>   return true;
> }
> 
>-bool H264AnnexBBufferHasVideoFormatDescription(const uint8_t* annexb_buffer,
>-                                               size_t annexb_buffer_size) {
>-  RTC_DCHECK(annexb_buffer);
>-  RTC_DCHECK_GT(annexb_buffer_size, 4);
>-
>-  // The buffer we receive via RTP has 00 00 00 01 start code artifically
>-  // embedded by the RTP depacketizer. Extract NALU information.
>-  // TODO(tkchin): handle potential case where sps and pps are delivered
>-  // separately.
>-  NaluType first_nalu_type = ParseNaluType(annexb_buffer[4]);
>-  bool is_first_nalu_type_sps = first_nalu_type == kSps;
>-  if (is_first_nalu_type_sps)
>-    return true;
>-  bool is_first_nalu_type_aud = first_nalu_type == kAud;
>-  // Start code + access unit delimiter + start code = 4 + 2 + 4 = 10.
>-  if (!is_first_nalu_type_aud || annexb_buffer_size <= 10u)
>-    return false;
>-  NaluType second_nalu_type = ParseNaluType(annexb_buffer[10]);
>-  bool is_second_nalu_type_sps = second_nalu_type == kSps;
>-  return is_second_nalu_type_sps;
>-}
>-
> CMVideoFormatDescriptionRef CreateVideoFormatDescription(
>     const uint8_t* annexb_buffer,
>     size_t annexb_buffer_size) {
>-  if (!H264AnnexBBufferHasVideoFormatDescription(annexb_buffer,
>-                                                 annexb_buffer_size)) {
>-    return nullptr;
>-  }
>-  AnnexBBufferReader reader(annexb_buffer, annexb_buffer_size);
>-  CMVideoFormatDescriptionRef description = nullptr;
>-  OSStatus status = noErr;
>-  // Parse the SPS and PPS into a CMVideoFormatDescription.
>   const uint8_t* param_set_ptrs[2] = {};
>   size_t param_set_sizes[2] = {};
>-  // Skip AUD.
>-  if (ParseNaluType(annexb_buffer[4]) == kAud) {
>-    if (!reader.ReadNalu(&param_set_ptrs[0], &param_set_sizes[0])) {
>-      RTC_LOG(LS_ERROR) << "Failed to read AUD";
>-      return nullptr;
>-    }
>+  AnnexBBufferReader reader(annexb_buffer, annexb_buffer_size);
>+  // Skip everyting before the SPS, then read the SPS and PPS
>+  if (!reader.SeekToNextNaluOfType(kSps)) {
>+    return nullptr;
>   }
>   if (!reader.ReadNalu(&param_set_ptrs[0], &param_set_sizes[0])) {
>     RTC_LOG(LS_ERROR) << "Failed to read SPS";
>@@ -296,9 +266,11 @@ CMVideoFormatDescriptionRef CreateVideoFormatDescription(
>     RTC_LOG(LS_ERROR) << "Failed to read PPS";
>     return nullptr;
>   }
>-  status = CMVideoFormatDescriptionCreateFromH264ParameterSets(
>-      kCFAllocatorDefault, 2, param_set_ptrs, param_set_sizes, 4,
>-      &description);
>+
>+  // Parse the SPS and PPS into a CMVideoFormatDescription.
>+  CMVideoFormatDescriptionRef description = nullptr;
>+  OSStatus status = CMVideoFormatDescriptionCreateFromH264ParameterSets(
>+      kCFAllocatorDefault, 2, param_set_ptrs, param_set_sizes, 4, &description);
>   if (status != noErr) {
>     RTC_LOG(LS_ERROR) << "Failed to create video format description.";
>     return nullptr;
>@@ -314,6 +286,8 @@ AnnexBBufferReader::AnnexBBufferReader(const uint8_t* annexb_buffer,
>   offset_ = offsets_.begin();
> }
> 
>+AnnexBBufferReader::~AnnexBBufferReader() = default;
>+
> bool AnnexBBufferReader::ReadNalu(const uint8_t** out_nalu,
>                                   size_t* out_length) {
>   RTC_DCHECK(out_nalu);
>@@ -337,6 +311,19 @@ size_t AnnexBBufferReader::BytesRemaining() const {
>   return length_ - offset_->start_offset;
> }
> 
>+void AnnexBBufferReader::SeekToStart() {
>+  offset_ = offsets_.begin();
>+}
>+
>+bool AnnexBBufferReader::SeekToNextNaluOfType(NaluType type) {
>+  for (; offset_ != offsets_.end(); ++offset_) {
>+    if (offset_->payload_size < 1)
>+      continue;
>+    if (ParseNaluType(*(start_ + offset_->payload_start_offset)) == type)
>+      return true;
>+  }
>+  return false;
>+}
> AvccBufferWriter::AvccBufferWriter(uint8_t* const avcc_buffer, size_t length)
>     : start_(avcc_buffer), offset_(0), length_(length) {
>   RTC_DCHECK(avcc_buffer);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/nalu_rewriter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/nalu_rewriter.h
>index fed83eb8831..cd5a51079ab 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/nalu_rewriter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/nalu_rewriter.h
>@@ -33,7 +33,7 @@ bool H264CMSampleBufferToAnnexBBuffer(
>     CMSampleBufferRef avcc_sample_buffer,
>     bool is_keyframe,
>     rtc::Buffer* annexb_buffer,
>-    std::unique_ptr<RTPFragmentationHeader> *out_header);
>+    std::unique_ptr<RTPFragmentationHeader>* out_header);
> 
> // Converts a buffer received from RTP into a sample buffer suitable for the
> // VideoToolbox decoder. The RTP buffer is in annex b format whereas the sample
>@@ -46,11 +46,6 @@ bool H264AnnexBBufferToCMSampleBuffer(const uint8_t* annexb_buffer,
>                                       CMVideoFormatDescriptionRef video_format,
>                                       CMSampleBufferRef* out_sample_buffer);
> 
>-// Returns true if the type of the first NALU in the supplied Annex B buffer is
>-// the SPS type.
>-bool H264AnnexBBufferHasVideoFormatDescription(const uint8_t* annexb_buffer,
>-                                               size_t annexb_buffer_size);
>-
> // Returns a video format description created from the sps/pps information in
> // the Annex B buffer. If there is no such information, nullptr is returned.
> // The caller is responsible for releasing the description.
>@@ -62,7 +57,7 @@ CMVideoFormatDescriptionRef CreateVideoFormatDescription(
> class AnnexBBufferReader final {
>  public:
>   AnnexBBufferReader(const uint8_t* annexb_buffer, size_t length);
>-  ~AnnexBBufferReader() {}
>+  ~AnnexBBufferReader();
>   AnnexBBufferReader(const AnnexBBufferReader& other) = delete;
>   void operator=(const AnnexBBufferReader& other) = delete;
> 
>@@ -74,6 +69,15 @@ class AnnexBBufferReader final {
>   // If the buffer has no remaining NALUs this will return zero.
>   size_t BytesRemaining() const;
> 
>+  // Reset the reader to start reading from the first NALU
>+  void SeekToStart();
>+
>+  // Seek to the next position that holds a NALU of the desired type,
>+  // or the end if no such NALU is found.
>+  // Return true if a NALU of the desired type is found, false if we
>+  // reached the end instead
>+  bool SeekToNextNaluOfType(H264::NaluType type);
>+
>  private:
>   // Returns the the next offset that contains NALU data.
>   size_t FindNextNaluHeader(const uint8_t* start,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/nalu_rewriter_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/nalu_rewriter_unittest.cc
>index f5d5406e7b1..d907ef84874 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/nalu_rewriter_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/nalu_rewriter_unittest.cc
>@@ -11,42 +11,25 @@
> 
> #include <memory>
> 
>+#include "common_video/h264/h264_common.h"
> #include "rtc_base/arraysize.h"
> #include "sdk/objc/Framework/Classes/VideoToolbox/nalu_rewriter.h"
> #include "test/gtest.h"
> 
> namespace webrtc {
> 
>+using H264::kSps;
>+
> static const uint8_t NALU_TEST_DATA_0[] = {0xAA, 0xBB, 0xCC};
> static const uint8_t NALU_TEST_DATA_1[] = {0xDE, 0xAD, 0xBE, 0xEF};
> 
>-TEST(H264VideoToolboxNaluTest, TestHasVideoFormatDescription) {
>-  const uint8_t sps_buffer[] = {0x00, 0x00, 0x00, 0x01, 0x27};
>-  EXPECT_TRUE(H264AnnexBBufferHasVideoFormatDescription(sps_buffer,
>-                                                        arraysize(sps_buffer)));
>-  const uint8_t aud_sps_buffer[] = {0x00, 0x00, 0x00, 0x01, 0x29, 0x10,
>-                                    0x00, 0x00, 0x00, 0x01, 0x27, 0xFF};
>-  EXPECT_TRUE(H264AnnexBBufferHasVideoFormatDescription(
>-      aud_sps_buffer, arraysize(aud_sps_buffer)));
>-  const uint8_t other_buffer[] = {0x00, 0x00, 0x00, 0x01, 0x28};
>-  EXPECT_FALSE(H264AnnexBBufferHasVideoFormatDescription(
>-      other_buffer, arraysize(other_buffer)));
>-  const uint8_t aud_other_buffer[] = {0x00, 0x00, 0x00, 0x01, 0x29,
>-                                      0x00, 0x00, 0x00, 0x01, 0x28};
>-  EXPECT_FALSE(H264AnnexBBufferHasVideoFormatDescription(
>-      aud_other_buffer, arraysize(aud_other_buffer)));
>-}
>-
> TEST(H264VideoToolboxNaluTest, TestCreateVideoFormatDescription) {
>   const uint8_t sps_pps_buffer[] = {
>-    // SPS nalu.
>-    0x00, 0x00, 0x00, 0x01,
>-    0x27, 0x42, 0x00, 0x1E, 0xAB, 0x40, 0xF0, 0x28, 0xD3, 0x70, 0x20, 0x20,
>-    0x20, 0x20,
>-    // PPS nalu.
>-    0x00, 0x00, 0x00, 0x01,
>-    0x28, 0xCE, 0x3C, 0x30
>-  };
>+      // SPS nalu.
>+      0x00, 0x00, 0x00, 0x01, 0x27, 0x42, 0x00, 0x1E, 0xAB, 0x40, 0xF0, 0x28,
>+      0xD3, 0x70, 0x20, 0x20, 0x20, 0x20,
>+      // PPS nalu.
>+      0x00, 0x00, 0x00, 0x01, 0x28, 0xCE, 0x3C, 0x30};
>   CMVideoFormatDescriptionRef description =
>       CreateVideoFormatDescription(sps_pps_buffer, arraysize(sps_pps_buffer));
>   EXPECT_TRUE(description);
>@@ -54,9 +37,27 @@ TEST(H264VideoToolboxNaluTest, TestCreateVideoFormatDescription) {
>     CFRelease(description);
>     description = nullptr;
>   }
>+
>+  const uint8_t sps_pps_not_at_start_buffer[] = {
>+      // Add some non-SPS/PPS NALUs at the beginning
>+      0x00, 0x00, 0x00, 0x01, 0x00, 0x00, 0x01, 0xFF, 0x00, 0x00, 0x00, 0x01,
>+      0xAB, 0x33, 0x21,
>+      // SPS nalu.
>+      0x00, 0x00, 0x01, 0x27, 0x42, 0x00, 0x1E, 0xAB, 0x40, 0xF0, 0x28, 0xD3,
>+      0x70, 0x20, 0x20, 0x20, 0x20,
>+      // PPS nalu.
>+      0x00, 0x00, 0x01, 0x28, 0xCE, 0x3C, 0x30};
>+  description = CreateVideoFormatDescription(
>+      sps_pps_not_at_start_buffer, arraysize(sps_pps_not_at_start_buffer));
>+  EXPECT_TRUE(description);
>+  if (description) {
>+    CFRelease(description);
>+    description = nullptr;
>+  }
>+
>   const uint8_t other_buffer[] = {0x00, 0x00, 0x00, 0x01, 0x28};
>-  EXPECT_FALSE(CreateVideoFormatDescription(other_buffer,
>-                                            arraysize(other_buffer)));
>+  EXPECT_FALSE(
>+      CreateVideoFormatDescription(other_buffer, arraysize(other_buffer)));
> }
> 
> TEST(AnnexBBufferReaderTest, TestReadEmptyInput) {
>@@ -140,6 +141,34 @@ TEST(AnnexBBufferReaderTest, TestReadMultipleNalus) {
>   EXPECT_EQ(0u, nalu_length);
> }
> 
>+TEST(AnnexBBufferReaderTest, TestFindNextNaluOfType) {
>+  const uint8_t notSps = 0x1F;
>+  const uint8_t annex_b_test_data[] = {
>+      0x00, 0x00,   0x00, 0x01,   kSps, 0x00,   0x00, 0x01, notSps,
>+      0x00, 0x00,   0x01, notSps, 0xDD, 0x00,   0x00, 0x01, notSps,
>+      0xEE, 0xFF,   0x00, 0x00,   0x00, 0xFF,   0x00, 0x00, 0x00,
>+      0x01, 0x00,   0x00, 0x00,   0x01, kSps,   0xBB, 0x00, 0x00,
>+      0x01, notSps, 0x00, 0x00,   0x01, notSps, 0xDD, 0x00, 0x00,
>+      0x01, notSps, 0xEE, 0xFF,   0x00, 0x00,   0x00, 0x01};
>+
>+  AnnexBBufferReader reader(annex_b_test_data, arraysize(annex_b_test_data));
>+  const uint8_t* nalu = nullptr;
>+  size_t nalu_length = 0;
>+  EXPECT_EQ(arraysize(annex_b_test_data), reader.BytesRemaining());
>+  EXPECT_TRUE(reader.FindNextNaluOfType(kSps));
>+  EXPECT_TRUE(reader.ReadNalu(&nalu, &nalu_length));
>+  EXPECT_EQ(annex_b_test_data + 4, nalu);
>+  EXPECT_EQ(1u, nalu_length);
>+
>+  EXPECT_TRUE(reader.FindNextNaluOfType(kSps));
>+  EXPECT_TRUE(reader.ReadNalu(&nalu, &nalu_length));
>+  EXPECT_EQ(annex_b_test_data + 32, nalu);
>+  EXPECT_EQ(2u, nalu_length);
>+
>+  EXPECT_FALSE(reader.FindNextNaluOfType(kSps));
>+  EXPECT_FALSE(reader.ReadNalu(&nalu, &nalu_length));
>+}
>+
> TEST(AvccBufferWriterTest, TestEmptyOutputBuffer) {
>   const uint8_t expected_buffer[] = {0x00};
>   const size_t buffer_size = 1;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/objc_video_decoder_factory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/objc_video_decoder_factory.h
>deleted file mode 100644
>index 13467e3898e..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/objc_video_decoder_factory.h
>+++ /dev/null
>@@ -1,51 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef SDK_OBJC_FRAMEWORK_CLASSES_PEERCONNECTION_OBJC_VIDEO_DECODER_FACTORY_H_
>-#define SDK_OBJC_FRAMEWORK_CLASSES_PEERCONNECTION_OBJC_VIDEO_DECODER_FACTORY_H_
>-
>-#include "api/video_codecs/video_decoder_factory.h"
>-#include "media/base/codec.h"
>-#include "media/engine/webrtcvideodecoderfactory.h"
>-
>-__attribute__((objc_runtime_name("WK_RTCVideoDecoderFactory")))
>-@protocol RTCVideoDecoderFactory;
>-
>-namespace webrtc {
>-
>-// TODO(andersc): Remove the inheritance from cricket::WebRtcVideoDecoderFactory
>-// when the legacy path in [RTCPeerConnectionFactory init] is no longer needed.
>-class ObjCVideoDecoderFactory : public VideoDecoderFactory,
>-                                public cricket::WebRtcVideoDecoderFactory {
>- public:
>-  explicit ObjCVideoDecoderFactory(id<RTCVideoDecoderFactory>);
>-  ~ObjCVideoDecoderFactory();
>-
>-  id<RTCVideoDecoderFactory> wrapped_decoder_factory() const;
>-
>-  std::vector<SdpVideoFormat> GetSupportedFormats() const override;
>-  std::unique_ptr<VideoDecoder> CreateVideoDecoder(
>-      const SdpVideoFormat& format) override;
>-
>-  // Needed for WebRtcVideoDecoderFactory interface.
>-  webrtc::VideoDecoder* CreateVideoDecoderWithParams(
>-      const cricket::VideoCodec& codec,
>-      cricket::VideoDecoderParams params) override;
>-  webrtc::VideoDecoder* CreateVideoDecoder(
>-      webrtc::VideoCodecType type) override;
>-  void DestroyVideoDecoder(webrtc::VideoDecoder* decoder) override;
>-
>- private:
>-  id<RTCVideoDecoderFactory> decoder_factory_;
>-};
>-
>-}  // namespace webrtc
>-
>-#endif  // SDK_OBJC_FRAMEWORK_CLASSES_PEERCONNECTION_OBJC_VIDEO_DECODER_FACTORY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/objc_video_encoder_factory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/objc_video_encoder_factory.h
>deleted file mode 100644
>index 727922bfe4e..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/objc_video_encoder_factory.h
>+++ /dev/null
>@@ -1,54 +0,0 @@
>-/*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef SDK_OBJC_FRAMEWORK_CLASSES_PEERCONNECTION_OBJC_VIDEO_ENCODER_FACTORY_H_
>-#define SDK_OBJC_FRAMEWORK_CLASSES_PEERCONNECTION_OBJC_VIDEO_ENCODER_FACTORY_H_
>-
>-#import <Foundation/Foundation.h>
>-
>-#include "api/video_codecs/video_encoder_factory.h"
>-#include "media/engine/webrtcvideoencoderfactory.h"
>-
>-__attribute__((objc_runtime_name("WK_RTCVideoEncoderFactory")))
>-@protocol RTCVideoEncoderFactory;
>-
>-namespace webrtc {
>-
>-// TODO(andersc): Remove the inheritance from cricket::WebRtcVideoEncoderFactory
>-// when the legacy path in [RTCPeerConnectionFactory init] is no longer needed.
>-class ObjCVideoEncoderFactory : public VideoEncoderFactory,
>-                                public cricket::WebRtcVideoEncoderFactory {
>- public:
>-  explicit ObjCVideoEncoderFactory(id<RTCVideoEncoderFactory>);
>-  ~ObjCVideoEncoderFactory();
>-
>-  id<RTCVideoEncoderFactory> wrapped_encoder_factory() const;
>-
>-  std::vector<SdpVideoFormat> GetSupportedFormats() const override;
>-  std::unique_ptr<VideoEncoder> CreateVideoEncoder(
>-      const SdpVideoFormat& format) override;
>-  CodecInfo QueryVideoEncoder(const SdpVideoFormat& format) const override;
>-
>-  // Needed for WebRtcVideoEncoderFactory interface.
>-  webrtc::VideoEncoder* CreateVideoEncoder(
>-      const cricket::VideoCodec& codec) override;
>-  const std::vector<cricket::VideoCodec>& supported_codecs() const override;
>-  void DestroyVideoEncoder(webrtc::VideoEncoder* encoder) override;
>-
>- private:
>-  id<RTCVideoEncoderFactory> encoder_factory_;
>-
>-  // Needed for WebRtcVideoEncoderFactory interface.
>-  mutable std::vector<cricket::VideoCodec> supported_codecs_;
>-};
>-
>-}  // namespace webrtc
>-
>-#endif  // SDK_OBJC_FRAMEWORK_CLASSES_PEERCONNECTION_OBJC_VIDEO_ENCODER_FACTORY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/OWNERS
>deleted file mode 100644
>index 4b431af19a2..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/OWNERS
>+++ /dev/null
>@@ -1,27 +0,0 @@
>-# Headers for classes closely related to the PeerConnection interface.
>-per-file RTCAudioSource.h=deadbeef@webrtc.org
>-per-file RTCAudioTrack.h=deadbeef@webrtc.org
>-per-file RTCConfiguration.h=deadbeef@webrtc.org
>-per-file RTCDataChannelConfiguration.h=deadbeef@webrtc.org
>-per-file RTCDataChannel.h=deadbeef@webrtc.org
>-per-file RTCIceCandidate.h=deadbeef@webrtc.org
>-per-file RTCIceServer.h=deadbeef@webrtc.org
>-per-file RTCIntervalRange.h=deadbeef@webrtc.org
>-per-file RTCLegacyStatsReport.h=deadbeef@webrtc.org
>-per-file RTCMediaConstraints.h=deadbeef@webrtc.org
>-per-file RTCMediaSource.h=deadbeef@webrtc.org
>-per-file RTCMediaStream.h=deadbeef@webrtc.org
>-per-file RTCMediaStreamTrack.h=deadbeef@webrtc.org
>-per-file RTCPeerConnectionFactory.h=deadbeef@webrtc.org
>-per-file RTCPeerConnection.h=deadbeef@webrtc.org
>-per-file RTCRtpCodecParameters.h=deadbeef@webrtc.org
>-per-file RTCRtpEncodingParameters.h=deadbeef@webrtc.org
>-per-file RTCRtpParameters.h=deadbeef@webrtc.org
>-per-file RTCRtpReceiver.h=deadbeef@webrtc.org
>-per-file RTCRtpSender.h=deadbeef@webrtc.org
>-per-file RTCSessionDescription.h=deadbeef@webrtc.org
>-per-file RTCSSLAdapter.h=deadbeef@webrtc.org
>-per-file RTCTracing.h=deadbeef@webrtc.org
>-per-file RTCVideoCapturer.h=deadbeef@webrtc.org
>-per-file RTCVideoSource.h=deadbeef@webrtc.org
>-per-file RTCVideoTrack.h=deadbeef@webrtc.org
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAVFoundationVideoSource.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAVFoundationVideoSource.h
>deleted file mode 100644
>index de73b477099..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAVFoundationVideoSource.h
>+++ /dev/null
>@@ -1,56 +0,0 @@
>-/*
>- *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#import <WebRTC/RTCMacros.h>
>-#import <WebRTC/RTCVideoSource.h>
>-
>-@class AVCaptureSession;
>-@class RTCMediaConstraints;
>-@class RTCPeerConnectionFactory;
>-
>-NS_ASSUME_NONNULL_BEGIN
>-
>-/**
>- * DEPRECATED Use RTCCameraVideoCapturer instead.
>- *
>- * RTCAVFoundationVideoSource is a video source that uses
>- * webrtc::AVFoundationVideoCapturer. We do not currently provide a wrapper for
>- * that capturer because cricket::VideoCapturer is not ref counted and we cannot
>- * guarantee its lifetime. Instead, we expose its properties through the ref
>- * counted video source interface.
>- */
>-RTC_EXPORT
>-__attribute__((objc_runtime_name("WK_RTCAVFoundationVideoSource")))
>-@interface RTCAVFoundationVideoSource : RTCVideoSource
>-
>-- (instancetype)init NS_UNAVAILABLE;
>-
>-/**
>- * Calling this function will cause frames to be scaled down to the
>- * requested resolution. Also, frames will be cropped to match the
>- * requested aspect ratio, and frames will be dropped to match the
>- * requested fps. The requested aspect ratio is orientation agnostic and
>- * will be adjusted to maintain the input orientation, so it doesn't
>- * matter if e.g. 1280x720 or 720x1280 is requested.
>- */
>-- (void)adaptOutputFormatToWidth:(int)width height:(int)height fps:(int)fps;
>-
>-/** Returns whether rear-facing camera is available for use. */
>-@property(nonatomic, readonly) BOOL canUseBackCamera;
>-
>-/** Switches the camera being used (either front or back). */
>-@property(nonatomic, assign) BOOL useBackCamera;
>-
>-/** Returns the active capture session. */
>-@property(nonatomic, readonly) AVCaptureSession *captureSession;
>-
>-@end
>-
>-NS_ASSUME_NONNULL_END
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSession.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSession.h
>index e44aa39ae55..8e55852844b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSession.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSession.h
>@@ -15,7 +15,7 @@
> 
> NS_ASSUME_NONNULL_BEGIN
> 
>-extern NSString * const kRTCAudioSessionErrorDomain;
>+extern NSString *const kRTCAudioSessionErrorDomain;
> /** Method that requires lock was called without lock. */
> extern NSInteger const kRTCAudioSessionErrorLockRequired;
> /** Unknown configuration error occurred. */
>@@ -47,8 +47,8 @@ __attribute__((objc_runtime_name("WK_RTCAudioSessionDelegate")))
>  *  route.
>  */
> - (void)audioSessionDidChangeRoute:(RTCAudioSession *)session
>-           reason:(AVAudioSessionRouteChangeReason)reason
>-    previousRoute:(AVAudioSessionRouteDescription *)previousRoute;
>+                            reason:(AVAudioSessionRouteChangeReason)reason
>+                     previousRoute:(AVAudioSessionRouteDescription *)previousRoute;
> 
> /** Called on a system notification thread when AVAudioSession media server
>  *  terminates.
>@@ -62,8 +62,7 @@ __attribute__((objc_runtime_name("WK_RTCAudioSessionDelegate")))
> 
> // TODO(tkchin): Maybe handle SilenceSecondaryAudioHintNotification.
> 
>-- (void)audioSession:(RTCAudioSession *)session
>-    didChangeCanPlayOrRecord:(BOOL)canPlayOrRecord;
>+- (void)audioSession:(RTCAudioSession *)session didChangeCanPlayOrRecord:(BOOL)canPlayOrRecord;
> 
> /** Called on a WebRTC thread when the audio device is notified to begin
>  *  playback or recording.
>@@ -76,8 +75,7 @@ __attribute__((objc_runtime_name("WK_RTCAudioSessionDelegate")))
> - (void)audioSessionDidStopPlayOrRecord:(RTCAudioSession *)session;
> 
> /** Called when the AVAudioSession output volume value changes. */
>-- (void)audioSession:(RTCAudioSession *)audioSession
>-    didChangeOutputVolume:(float)outputVolume;
>+- (void)audioSession:(RTCAudioSession *)audioSession didChangeOutputVolume:(float)outputVolume;
> 
> /** Called when the audio device detects a playout glitch. The argument is the
>  *  number of glitches detected so far in the current audio playout session.
>@@ -85,6 +83,20 @@ __attribute__((objc_runtime_name("WK_RTCAudioSessionDelegate")))
> - (void)audioSession:(RTCAudioSession *)audioSession
>     didDetectPlayoutGlitch:(int64_t)totalNumberOfGlitches;
> 
>+/** Called when the audio session is about to change the active state.
>+ */
>+- (void)audioSession:(RTCAudioSession *)audioSession willSetActive:(BOOL)active;
>+
>+/** Called after the audio session sucessfully changed the active state.
>+ */
>+- (void)audioSession:(RTCAudioSession *)audioSession didSetActive:(BOOL)active;
>+
>+/** Called after the audio session failed to change the active state.
>+ */
>+- (void)audioSession:(RTCAudioSession *)audioSession
>+    failedToSetActive:(BOOL)active
>+                error:(NSError *)error;
>+
> @end
> 
> /** This is a protocol used to inform RTCAudioSession when the audio session
>@@ -159,14 +171,10 @@ __attribute__((objc_runtime_name("WK_RTCAudioSession")))
> @property(readonly) float inputGain;
> @property(readonly) BOOL inputGainSettable;
> @property(readonly) BOOL inputAvailable;
>-@property(readonly, nullable)
>-    NSArray<AVAudioSessionDataSourceDescription *> * inputDataSources;
>-@property(readonly, nullable)
>-  AVAudioSessionDataSourceDescription *inputDataSource;
>-@property(readonly, nullable)
>-    NSArray<AVAudioSessionDataSourceDescription *> * outputDataSources;
>-@property(readonly, nullable)
>-    AVAudioSessionDataSourceDescription *outputDataSource;
>+@property(readonly, nullable) NSArray<AVAudioSessionDataSourceDescription *> *inputDataSources;
>+@property(readonly, nullable) AVAudioSessionDataSourceDescription *inputDataSource;
>+@property(readonly, nullable) NSArray<AVAudioSessionDataSourceDescription *> *outputDataSources;
>+@property(readonly, nullable) AVAudioSessionDataSourceDescription *outputDataSource;
> @property(readonly) double sampleRate;
> @property(readonly) double preferredSampleRate;
> @property(readonly) NSInteger inputNumberOfChannels;
>@@ -200,8 +208,7 @@ __attribute__((objc_runtime_name("WK_RTCAudioSession")))
>  *  AVAudioSessionSetActiveOptionNotifyOthersOnDeactivation option is passed to
>  *  AVAudioSession.
>  */
>-- (BOOL)setActive:(BOOL)active
>-            error:(NSError **)outError;
>+- (BOOL)setActive:(BOOL)active error:(NSError **)outError;
> 
> // The following methods are proxies for the associated methods on
> // AVAudioSession. |lockForConfiguration| must be called before using them
>@@ -213,22 +220,18 @@ __attribute__((objc_runtime_name("WK_RTCAudioSession")))
> - (BOOL)setMode:(NSString *)mode error:(NSError **)outError;
> - (BOOL)setInputGain:(float)gain error:(NSError **)outError;
> - (BOOL)setPreferredSampleRate:(double)sampleRate error:(NSError **)outError;
>-- (BOOL)setPreferredIOBufferDuration:(NSTimeInterval)duration
>-                               error:(NSError **)outError;
>-- (BOOL)setPreferredInputNumberOfChannels:(NSInteger)count
>-                                    error:(NSError **)outError;
>-- (BOOL)setPreferredOutputNumberOfChannels:(NSInteger)count
>-                                     error:(NSError **)outError;
>-- (BOOL)overrideOutputAudioPort:(AVAudioSessionPortOverride)portOverride
>-                          error:(NSError **)outError;
>-- (BOOL)setPreferredInput:(AVAudioSessionPortDescription *)inPort
>-                    error:(NSError **)outError;
>+- (BOOL)setPreferredIOBufferDuration:(NSTimeInterval)duration error:(NSError **)outError;
>+- (BOOL)setPreferredInputNumberOfChannels:(NSInteger)count error:(NSError **)outError;
>+- (BOOL)setPreferredOutputNumberOfChannels:(NSInteger)count error:(NSError **)outError;
>+- (BOOL)overrideOutputAudioPort:(AVAudioSessionPortOverride)portOverride error:(NSError **)outError;
>+- (BOOL)setPreferredInput:(AVAudioSessionPortDescription *)inPort error:(NSError **)outError;
> - (BOOL)setInputDataSource:(AVAudioSessionDataSourceDescription *)dataSource
>                      error:(NSError **)outError;
> - (BOOL)setOutputDataSource:(AVAudioSessionDataSourceDescription *)dataSource
>                       error:(NSError **)outError;
> @end
> 
>+__attribute__((objc_runtime_name("WK_RTCAudioSession")))
> @interface RTCAudioSession (Configuration)
> 
> /** Applies the configuration to the current session. Attempts to set all
>@@ -236,8 +239,7 @@ __attribute__((objc_runtime_name("WK_RTCAudioSession")))
>  *  returned.
>  *  |lockForConfiguration| must be called first.
>  */
>-- (BOOL)setConfiguration:(RTCAudioSessionConfiguration *)configuration
>-                   error:(NSError **)outError;
>+- (BOOL)setConfiguration:(RTCAudioSessionConfiguration *)configuration error:(NSError **)outError;
> 
> /** Convenience method that calls both setConfiguration and setActive.
>  *  |lockForConfiguration| must be called first.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioTrack.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioTrack.h
>index bab1b1a0020..71d797dbd75 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioTrack.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioTrack.h
>@@ -16,7 +16,7 @@ NS_ASSUME_NONNULL_BEGIN
> @class RTCAudioSource;
> 
> RTC_EXPORT
>-__attribute__((objc_runtime_name("WK_RTCAudioTrack")))
>+__attribute__((objc_runtime_name("WK_RTCMediaStreamTrack")))
> @interface RTCAudioTrack : RTCMediaStreamTrack
> 
> - (instancetype)init NS_UNAVAILABLE;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCCallbackLogger.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCCallbackLogger.h
>new file mode 100644
>index 00000000000..cdfe17f6071
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCCallbackLogger.h
>@@ -0,0 +1,35 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#import <Foundation/Foundation.h>
>+
>+#import <WebRTC/RTCLogging.h>
>+#import <WebRTC/RTCMacros.h>
>+
>+NS_ASSUME_NONNULL_BEGIN
>+
>+// This class intercepts WebRTC logs and forwards them to a registered block.
>+// This class is not threadsafe.
>+RTC_EXPORT
>+@interface RTCCallbackLogger : NSObject
>+
>+// The severity level to capture. The default is kRTCLoggingSeverityInfo.
>+@property(nonatomic, assign) RTCLoggingSeverity severity;
>+
>+// The callback will be called on the same thread that does the logging, so
>+// if the logging callback can be slow it may be a good idea to implement
>+// dispatching to some other queue.
>+- (void)start:(nullable void (^)(NSString*))callback;
>+
>+- (void)stop;
>+
>+@end
>+
>+NS_ASSUME_NONNULL_END
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCCameraPreviewView.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCCameraPreviewView.h
>index 9f9a59fc87d..56ef9281729 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCCameraPreviewView.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCCameraPreviewView.h
>@@ -14,7 +14,6 @@
> #import <WebRTC/RTCMacros.h>
> 
> @class AVCaptureSession;
>-@class RTCAVFoundationVideoSource;
> 
> /** RTCCameraPreviewView is a view that renders local video from an
>  *  AVCaptureSession.
>@@ -27,6 +26,6 @@ __attribute__((objc_runtime_name("WK_RTCCameraPreviewView")))
>  *  is assigned to AVCaptureVideoPreviewLayer async in the same
>  *  queue that the AVCaptureSession is started/stopped.
>  */
>-@property(nonatomic, strong) AVCaptureSession *captureSession;
>+@property(nonatomic, strong) AVCaptureSession* captureSession;
> 
> @end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCCameraVideoCapturer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCCameraVideoCapturer.h
>index b2e012778f3..61868c5469c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCCameraVideoCapturer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCCameraVideoCapturer.h
>@@ -8,8 +8,8 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#import <Foundation/Foundation.h>
> #import <AVFoundation/AVFoundation.h>
>+#import <Foundation/Foundation.h>
> 
> #import <WebRTC/RTCMacros.h>
> #import <WebRTC/RTCVideoCapturer.h>
>@@ -19,7 +19,7 @@ NS_ASSUME_NONNULL_BEGIN
> RTC_EXPORT
> // Camera capture that implements RTCVideoCapturer. Delivers frames to a RTCVideoCapturerDelegate
> // (usually RTCVideoSource).
>-__attribute__((objc_runtime_name("WK_RTCCameraVideoCapturer")))
>+NS_EXTENSION_UNAVAILABLE_IOS("Camera not available in app extensions.")
> @interface RTCCameraVideoCapturer : RTCVideoCapturer
> 
> // Capture session that is used for capturing. Valid from initialization to dealloc.
>@@ -42,7 +42,7 @@ __attribute__((objc_runtime_name("WK_RTCCameraVideoCapturer")))
>                            fps:(NSInteger)fps
>              completionHandler:(nullable void (^)(NSError *))completionHandler;
> // Stops the capture session asynchronously and notifies callback on completion.
>-- (void)stopCaptureWithCompletionHandler:(nullable void (^)())completionHandler;
>+- (void)stopCaptureWithCompletionHandler:(nullable void (^)(void))completionHandler;
> 
> // Starts the capture session asynchronously.
> - (void)startCaptureWithDevice:(AVCaptureDevice *)device
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCCertificate.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCCertificate.h
>new file mode 100644
>index 00000000000..6b4ea4323a0
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCCertificate.h
>@@ -0,0 +1,44 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#import <Foundation/Foundation.h>
>+
>+#import <WebRTC/RTCMacros.h>
>+
>+NS_ASSUME_NONNULL_BEGIN
>+
>+RTC_EXPORT
>+@interface RTCCertificate : NSObject <NSCopying>
>+
>+/** Private key in PEM. */
>+@property(nonatomic, readonly, copy) NSString *private_key;
>+
>+/** Public key in an x509 cert encoded in PEM. */
>+@property(nonatomic, readonly, copy) NSString *certificate;
>+
>+/**
>+ * Initialize an RTCCertificate with PEM strings for private_key and certificate.
>+ */
>+- (instancetype)initWithPrivateKey:(NSString *)private_key
>+                       certificate:(NSString *)certificate NS_DESIGNATED_INITIALIZER;
>+
>+- (instancetype)init NS_UNAVAILABLE;
>+
>+/** Generate a new certificate for 're' use.
>+ *
>+ *  Optional dictionary of parameters. Defaults to KeyType ECDSA if none are
>+ *  provided.
>+ *  - name: "ECDSA" or "RSASSA-PKCS1-v1_5"
>+ */
>++ (nullable RTCCertificate *)generateCertificateWithParams:(NSDictionary *)params;
>+
>+@end
>+
>+NS_ASSUME_NONNULL_END
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCConfiguration.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCConfiguration.h
>index bf8e50c0504..458b20c7ac9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCConfiguration.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCConfiguration.h
>@@ -10,6 +10,7 @@
> 
> #import <Foundation/Foundation.h>
> 
>+#import <WebRTC/RTCCertificate.h>
> #import <WebRTC/RTCMacros.h>
> 
> @class RTCIceServer;
>@@ -34,10 +35,7 @@ typedef NS_ENUM(NSInteger, RTCBundlePolicy) {
> };
> 
> /** Represents the rtcp mux policy. */
>-typedef NS_ENUM(NSInteger, RTCRtcpMuxPolicy) {
>-  RTCRtcpMuxPolicyNegotiate,
>-  RTCRtcpMuxPolicyRequire
>-};
>+typedef NS_ENUM(NSInteger, RTCRtcpMuxPolicy) { RTCRtcpMuxPolicyNegotiate, RTCRtcpMuxPolicyRequire };
> 
> /** Represents the tcp candidate policy. */
> typedef NS_ENUM(NSInteger, RTCTcpCandidatePolicy) {
>@@ -63,8 +61,13 @@ typedef NS_ENUM(NSInteger, RTCEncryptionKeyType) {
>   RTCEncryptionKeyTypeECDSA,
> };
> 
>-NS_ASSUME_NONNULL_BEGIN
>+/** Represents the chosen SDP semantics for the RTCPeerConnection. */
>+typedef NS_ENUM(NSInteger, RTCSdpSemantics) {
>+  RTCSdpSemanticsPlanB,
>+  RTCSdpSemanticsUnifiedPlan,
>+};
> 
>+NS_ASSUME_NONNULL_BEGIN
> RTC_EXPORT
> __attribute__((objc_runtime_name("WK_RTCConfiguration")))
> @interface RTCConfiguration : NSObject
>@@ -72,6 +75,9 @@ __attribute__((objc_runtime_name("WK_RTCConfiguration")))
> /** An array of Ice Servers available to be used by ICE. */
> @property(nonatomic, copy) NSArray<RTCIceServer *> *iceServers;
> 
>+/** An RTCCertificate for 're' use. */
>+@property(nonatomic, nullable) RTCCertificate *certificate;
>+
> /** Which candidates the ICE agent is allowed to use. The W3C calls it
>  * |iceTransportPolicy|, while in C++ it is called |type|. */
> @property(nonatomic, assign) RTCIceTransportPolicy iceTransportPolicy;
>@@ -83,8 +89,7 @@ __attribute__((objc_runtime_name("WK_RTCConfiguration")))
> @property(nonatomic, assign) RTCRtcpMuxPolicy rtcpMuxPolicy;
> @property(nonatomic, assign) RTCTcpCandidatePolicy tcpCandidatePolicy;
> @property(nonatomic, assign) RTCCandidateNetworkPolicy candidateNetworkPolicy;
>-@property(nonatomic, assign)
>-    RTCContinualGatheringPolicy continualGatheringPolicy;
>+@property(nonatomic, assign) RTCContinualGatheringPolicy continualGatheringPolicy;
> 
> /** By default, the PeerConnection will use a limited number of IPv6 network
>  *  interfaces, in order to avoid too many ICE candidate pairs being created
>@@ -94,6 +99,12 @@ __attribute__((objc_runtime_name("WK_RTCConfiguration")))
>  */
> @property(nonatomic, assign) int maxIPv6Networks;
> 
>+/** Exclude link-local network interfaces
>+ *  from considertaion for gathering ICE candidates.
>+ *  Defaults to NO.
>+ */
>+@property(nonatomic, assign) BOOL disableLinkLocalNetworks;
>+
> @property(nonatomic, assign) int audioJitterBufferMaxPackets;
> @property(nonatomic, assign) BOOL audioJitterBufferFastAccelerate;
> @property(nonatomic, assign) int iceConnectionReceivingTimeout;
>@@ -127,6 +138,36 @@ __attribute__((objc_runtime_name("WK_RTCConfiguration")))
>  */
> @property(nonatomic, strong, nullable) RTCIntervalRange *iceRegatherIntervalRange;
> 
>+/** Configure the SDP semantics used by this PeerConnection. Note that the
>+ *  WebRTC 1.0 specification requires UnifiedPlan semantics. The
>+ *  RTCRtpTransceiver API is only available with UnifiedPlan semantics.
>+ *
>+ *  PlanB will cause RTCPeerConnection to create offers and answers with at
>+ *  most one audio and one video m= section with multiple RTCRtpSenders and
>+ *  RTCRtpReceivers specified as multiple a=ssrc lines within the section. This
>+ *  will also cause RTCPeerConnection to ignore all but the first m= section of
>+ *  the same media type.
>+ *
>+ *  UnifiedPlan will cause RTCPeerConnection to create offers and answers with
>+ *  multiple m= sections where each m= section maps to one RTCRtpSender and one
>+ *  RTCRtpReceiver (an RTCRtpTransceiver), either both audio or both video. This
>+ *  will also cause RTCPeerConnection to ignore all but the first a=ssrc lines
>+ *  that form a Plan B stream.
>+ *
>+ *  For users who wish to send multiple audio/video streams and need to stay
>+ *  interoperable with legacy WebRTC implementations or use legacy APIs,
>+ *  specify PlanB.
>+ *
>+ *  For all other users, specify UnifiedPlan.
>+ */
>+@property(nonatomic, assign) RTCSdpSemantics sdpSemantics;
>+
>+/** Actively reset the SRTP parameters when the DTLS transports underneath are
>+ *  changed after offer/answer negotiation. This is only intended to be a
>+ *  workaround for crbug.com/835958
>+ */
>+@property(nonatomic, assign) BOOL activeResetSrtpParams;
>+
> - (instancetype)init;
> 
> @end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDataChannel.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDataChannel.h
>index 1785513753a..b525a84fc82 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDataChannel.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDataChannel.h
>@@ -35,10 +35,8 @@ __attribute__((objc_runtime_name("WK_RTCDataBuffer")))
> 
> @end
> 
>-
> @class RTCDataChannel;
> RTC_EXPORT
>-__attribute__((objc_runtime_name("WK_RTCDataChannelDelegate")))
> @protocol RTCDataChannelDelegate <NSObject>
> 
> /** The data channel state changed. */
>@@ -50,12 +48,10 @@ __attribute__((objc_runtime_name("WK_RTCDataChannelDelegate")))
> 
> @optional
> /** The data channel's |bufferedAmount| changed. */
>-- (void)dataChannel:(RTCDataChannel *)dataChannel
>-    didChangeBufferedAmount:(uint64_t)amount;
>+- (void)dataChannel:(RTCDataChannel *)dataChannel didChangeBufferedAmount:(uint64_t)amount;
> 
> @end
> 
>-
> /** Represents the state of the data channel. */
> typedef NS_ENUM(NSInteger, RTCDataChannelState) {
>   RTCDataChannelStateConnecting,
>@@ -81,8 +77,7 @@ __attribute__((objc_runtime_name("WK_RTCDataChannel")))
> @property(nonatomic, readonly) BOOL isOrdered;
> 
> /** Deprecated. Use maxPacketLifeTime. */
>-@property(nonatomic, readonly) NSUInteger maxRetransmitTime
>-    DEPRECATED_ATTRIBUTE;
>+@property(nonatomic, readonly) NSUInteger maxRetransmitTime DEPRECATED_ATTRIBUTE;
> 
> /**
>  * The length of the time window (in milliseconds) during which transmissions
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDataChannelConfiguration.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDataChannelConfiguration.h
>index 0b8913ce549..567e8838c65 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDataChannelConfiguration.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDataChannelConfiguration.h
>@@ -46,7 +46,7 @@ __attribute__((objc_runtime_name("WK_RTCDataChannelConfiguration")))
> @property(nonatomic, assign) int channelId;
> 
> /** Set by the application and opaque to the WebRTC implementation. */
>-@property(nonatomic) NSString *protocol;
>+@property(nonatomic) NSString* protocol;
> 
> @end
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDispatcher.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDispatcher.h
>index 2166277bc17..5e149fce0b1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDispatcher.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDispatcher.h
>@@ -35,8 +35,7 @@ __attribute__((objc_runtime_name("WK_RTCDispatcher")))
>  *  @param dispatchType The queue type to dispatch on.
>  *  @param block The block to dispatch asynchronously.
>  */
>-+ (void)dispatchAsyncOnType:(RTCDispatcherQueueType)dispatchType
>-                      block:(dispatch_block_t)block;
>++ (void)dispatchAsyncOnType:(RTCDispatcherQueueType)dispatchType block:(dispatch_block_t)block;
> 
> /** Returns YES if run on queue for the dispatchType otherwise NO.
>  *  Useful for asserting that a method is run on a correct queue.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDtmfSender.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDtmfSender.h
>index b4b6c6cf7d6..c69af026b7c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDtmfSender.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDtmfSender.h
>@@ -15,7 +15,6 @@
> NS_ASSUME_NONNULL_BEGIN
> 
> RTC_EXPORT
>-__attribute__((objc_runtime_name("WK_RTCDtmfSender")))
> @protocol RTCDtmfSender <NSObject>
> 
> /**
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCEAGLVideoView.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCEAGLVideoView.h
>index d4d3b54fee7..8fdd3c3c2f5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCEAGLVideoView.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCEAGLVideoView.h
>@@ -18,12 +18,9 @@
> NS_ASSUME_NONNULL_BEGIN
> 
> @class RTCEAGLVideoView;
>-RTC_EXPORT
>-__attribute__((objc_runtime_name("WK_RTCEAGLVideoViewDelegate")))
>-@protocol RTCEAGLVideoViewDelegate
>-
>-- (void)videoView:(RTCEAGLVideoView *)videoView didChangeVideoSize:(CGSize)size;
> 
>+RTC_EXPORT
>+@protocol RTCEAGLVideoViewDelegate <RTCVideoViewDelegate>
> @end
> 
> /**
>@@ -31,10 +28,10 @@ __attribute__((objc_runtime_name("WK_RTCEAGLVideoViewDelegate")))
>  * bounds using OpenGLES 2.0 or OpenGLES 3.0.
>  */
> RTC_EXPORT
>-__attribute__((objc_runtime_name("WK_RTCEAGLVideoView")))
>+NS_EXTENSION_UNAVAILABLE_IOS("Rendering not available in app extensions.")
> @interface RTCEAGLVideoView : UIView <RTCVideoRenderer>
> 
>-@property(nonatomic, weak) id<RTCEAGLVideoViewDelegate> delegate;
>+@property(nonatomic, weak) id<RTCVideoViewDelegate> delegate;
> 
> - (instancetype)initWithFrame:(CGRect)frame
>                        shader:(id<RTCVideoViewShading>)shader NS_DESIGNATED_INITIALIZER;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCFieldTrials.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCFieldTrials.h
>index 78d69610d8c..3ce94ba9379 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCFieldTrials.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCFieldTrials.h
>@@ -14,6 +14,9 @@
> 
> /** The only valid value for the following if set is kRTCFieldTrialEnabledValue. */
> RTC_EXTERN NSString * const kRTCFieldTrialAudioSendSideBweKey;
>+RTC_EXTERN NSString * const kRTCFieldTrialAudioSendSideBweForVideoKey;
>+RTC_EXTERN NSString * const kRTCFieldTrialAudioForceNoTWCCKey;
>+RTC_EXTERN NSString * const kRTCFieldTrialAudioForceABWENoTWCCKey;
> RTC_EXTERN NSString * const kRTCFieldTrialSendSideBweWithOverheadKey;
> RTC_EXTERN NSString * const kRTCFieldTrialFlexFec03AdvertisedKey;
> RTC_EXTERN NSString * const kRTCFieldTrialFlexFec03Key;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCFileLogger.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCFileLogger.h
>index 33df4efcf0f..a4e05a70b85 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCFileLogger.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCFileLogger.h
>@@ -53,13 +53,11 @@ __attribute__((objc_runtime_name("WK_RTCFileLogger")))
> - (instancetype)init;
> 
> // Create file logger with default rotation type.
>-- (instancetype)initWithDirPath:(NSString *)dirPath
>-                    maxFileSize:(NSUInteger)maxFileSize;
>+- (instancetype)initWithDirPath:(NSString *)dirPath maxFileSize:(NSUInteger)maxFileSize;
> 
> - (instancetype)initWithDirPath:(NSString *)dirPath
>                     maxFileSize:(NSUInteger)maxFileSize
>-                   rotationType:(RTCFileLoggerRotationType)rotationType
>-    NS_DESIGNATED_INITIALIZER;
>+                   rotationType:(RTCFileLoggerRotationType)rotationType NS_DESIGNATED_INITIALIZER;
> 
> // Starts writing WebRTC logs to disk if not already started. Overwrites any
> // existing file(s).
>@@ -75,4 +73,3 @@ __attribute__((objc_runtime_name("WK_RTCFileLogger")))
> @end
> 
> NS_ASSUME_NONNULL_END
>-
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCFileVideoCapturer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCFileVideoCapturer.h
>index 70f482e35c7..7b898e494cf 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCFileVideoCapturer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCFileVideoCapturer.h
>@@ -26,7 +26,6 @@ typedef void (^RTCFileVideoCapturerErrorBlock)(NSError *error);
> RTC_EXPORT
> 
> NS_CLASS_AVAILABLE_IOS(10)
>-__attribute__((objc_runtime_name("WK_RTCFileVideoCapturer")))
> @interface RTCFileVideoCapturer : RTCVideoCapturer
> 
> /**
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCIceCandidate.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCIceCandidate.h
>index 902925203ea..59083dca07c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCIceCandidate.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCIceCandidate.h
>@@ -43,8 +43,7 @@ __attribute__((objc_runtime_name("WK_RTCIceCandidate")))
>  */
> - (instancetype)initWithSdp:(NSString *)sdp
>               sdpMLineIndex:(int)sdpMLineIndex
>-                     sdpMid:(nullable NSString *)sdpMid
>-    NS_DESIGNATED_INITIALIZER;
>+                     sdpMid:(nullable NSString *)sdpMid NS_DESIGNATED_INITIALIZER;
> 
> @end
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCIntervalRange.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCIntervalRange.h
>index 11eb4dc6233..1bb27b629c7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCIntervalRange.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCIntervalRange.h
>@@ -19,11 +19,8 @@ __attribute__((objc_runtime_name("WK_RTCIntervalRange")))
> @property(nonatomic, readonly) NSInteger max;
> 
> - (instancetype)init;
>-- (instancetype)initWithMin:(NSInteger)min
>-                        max:(NSInteger)max
>-    NS_DESIGNATED_INITIALIZER;
>+- (instancetype)initWithMin:(NSInteger)min max:(NSInteger)max NS_DESIGNATED_INITIALIZER;
> 
> @end
> 
> NS_ASSUME_NONNULL_END
>-
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCLogging.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCLogging.h
>index a65dcad82ff..b758785115a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCLogging.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCLogging.h
>@@ -33,12 +33,9 @@ RTC_EXTERN NSString* RTCFileName(const char* filePath);
> 
> // Some convenience macros.
> 
>-#define RTCLogString(format, ...)                    \
>-  [NSString stringWithFormat:@"(%@:%d %s): " format, \
>-      RTCFileName(__FILE__),                         \
>-      __LINE__,                                      \
>-      __FUNCTION__,                                  \
>-      ##__VA_ARGS__]
>+#define RTCLogString(format, ...)                                           \
>+  [NSString stringWithFormat:@"(%@:%d %s): " format, RTCFileName(__FILE__), \
>+                             __LINE__, __FUNCTION__, ##__VA_ARGS__]
> 
> #define RTCLogFormat(severity, format, ...)                     \
>   do {                                                          \
>@@ -46,17 +43,17 @@ RTC_EXTERN NSString* RTCFileName(const char* filePath);
>     RTCLogEx(severity, log_string);                             \
>   } while (false)
> 
>-#define RTCLogVerbose(format, ...)                                \
>-  RTCLogFormat(RTCLoggingSeverityVerbose, format, ##__VA_ARGS__) \
>+#define RTCLogVerbose(format, ...) \
>+  RTCLogFormat(RTCLoggingSeverityVerbose, format, ##__VA_ARGS__)
> 
>-#define RTCLogInfo(format, ...)                                   \
>-  RTCLogFormat(RTCLoggingSeverityInfo, format, ##__VA_ARGS__)    \
>+#define RTCLogInfo(format, ...) \
>+  RTCLogFormat(RTCLoggingSeverityInfo, format, ##__VA_ARGS__)
> 
>-#define RTCLogWarning(format, ...)                                \
>-  RTCLogFormat(RTCLoggingSeverityWarning, format, ##__VA_ARGS__) \
>+#define RTCLogWarning(format, ...) \
>+  RTCLogFormat(RTCLoggingSeverityWarning, format, ##__VA_ARGS__)
> 
>-#define RTCLogError(format, ...)                                  \
>-  RTCLogFormat(RTCLoggingSeverityError, format, ##__VA_ARGS__)   \
>+#define RTCLogError(format, ...) \
>+  RTCLogFormat(RTCLoggingSeverityError, format, ##__VA_ARGS__)
> 
> #if !defined(NDEBUG)
> #define RTCLogDebug(format, ...) RTCLogInfo(format, ##__VA_ARGS__)
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMTLNSVideoView.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMTLNSVideoView.h
>index b18fc777402..d28d554d9ad 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMTLNSVideoView.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMTLNSVideoView.h
>@@ -12,7 +12,9 @@
> #import "WebRTC/RTCVideoRenderer.h"
> 
> NS_AVAILABLE_MAC(10.11)
>-__attribute__((objc_runtime_name("WK_RTCMTLNSVideoView")))
>-@interface RTCMTLNSVideoView : NSView<RTCVideoRenderer>
>+@interface RTCMTLNSVideoView : NSView <RTCVideoRenderer>
>+
>+@property(nonatomic, weak) id<RTCVideoViewDelegate> delegate;
>+
> + (BOOL)isMetalAvailable;
> @end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMTLVideoView.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMTLVideoView.h
>index b62d0889b06..06154950357 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMTLVideoView.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMTLVideoView.h
>@@ -10,6 +10,7 @@
> 
> #import <Foundation/Foundation.h>
> 
>+#import "WebRTC/RTCVideoFrame.h"
> #import "WebRTC/RTCVideoRenderer.h"
> 
> // Check if metal is supported in WebRTC.
>@@ -31,8 +32,20 @@ NS_ASSUME_NONNULL_BEGIN
> NS_CLASS_AVAILABLE_IOS(9)
> 
> RTC_EXPORT
>-__attribute__((objc_runtime_name("WK_RTCMTLVideoView")))
>-@interface RTCMTLVideoView : UIView <RTCVideoRenderer>
>+@interface RTCMTLVideoView : UIView<RTCVideoRenderer>
>+
>+@property(nonatomic, weak) id<RTCVideoViewDelegate> delegate;
>+
>+@property(nonatomic) UIViewContentMode videoContentMode;
>+
>+/** @abstract Enables/disables rendering.
>+ */
>+@property(nonatomic, getter=isEnabled) BOOL enabled;
>+
>+/** @abstract Wrapped RTCVideoRotation, or nil.
>+ */
>+@property(nonatomic, nullable) NSValue* rotationOverride;
> 
> @end
>+
> NS_ASSUME_NONNULL_END
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaConstraints.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaConstraints.h
>index e6321940c44..8b358e9c59d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaConstraints.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaConstraints.h
>@@ -15,29 +15,28 @@
> NS_ASSUME_NONNULL_BEGIN
> 
> /** Constraint keys for media sources. */
>-RTC_EXTERN NSString * const kRTCMediaConstraintsMinAspectRatio;
>-RTC_EXTERN NSString * const kRTCMediaConstraintsMaxAspectRatio;
>-RTC_EXTERN NSString * const kRTCMediaConstraintsMaxWidth;
>-RTC_EXTERN NSString * const kRTCMediaConstraintsMinWidth;
>-RTC_EXTERN NSString * const kRTCMediaConstraintsMaxHeight;
>-RTC_EXTERN NSString * const kRTCMediaConstraintsMinHeight;
>-RTC_EXTERN NSString * const kRTCMediaConstraintsMaxFrameRate;
>-RTC_EXTERN NSString * const kRTCMediaConstraintsMinFrameRate;
>-RTC_EXTERN NSString * const kRTCMediaConstraintsLevelControl;
>+RTC_EXTERN NSString *const kRTCMediaConstraintsMinAspectRatio;
>+RTC_EXTERN NSString *const kRTCMediaConstraintsMaxAspectRatio;
>+RTC_EXTERN NSString *const kRTCMediaConstraintsMaxWidth;
>+RTC_EXTERN NSString *const kRTCMediaConstraintsMinWidth;
>+RTC_EXTERN NSString *const kRTCMediaConstraintsMaxHeight;
>+RTC_EXTERN NSString *const kRTCMediaConstraintsMinHeight;
>+RTC_EXTERN NSString *const kRTCMediaConstraintsMaxFrameRate;
>+RTC_EXTERN NSString *const kRTCMediaConstraintsMinFrameRate;
> /** The value for this key should be a base64 encoded string containing
>  *  the data from the serialized configuration proto.
>  */
>-RTC_EXTERN NSString * const kRTCMediaConstraintsAudioNetworkAdaptorConfig;
>+RTC_EXTERN NSString *const kRTCMediaConstraintsAudioNetworkAdaptorConfig;
> 
> /** Constraint keys for generating offers and answers. */
>-RTC_EXTERN NSString * const kRTCMediaConstraintsIceRestart;
>-RTC_EXTERN NSString * const kRTCMediaConstraintsOfferToReceiveAudio;
>-RTC_EXTERN NSString * const kRTCMediaConstraintsOfferToReceiveVideo;
>-RTC_EXTERN NSString * const kRTCMediaConstraintsVoiceActivityDetection;
>+RTC_EXTERN NSString *const kRTCMediaConstraintsIceRestart;
>+RTC_EXTERN NSString *const kRTCMediaConstraintsOfferToReceiveAudio;
>+RTC_EXTERN NSString *const kRTCMediaConstraintsOfferToReceiveVideo;
>+RTC_EXTERN NSString *const kRTCMediaConstraintsVoiceActivityDetection;
> 
> /** Constraint values for Boolean parameters. */
>-RTC_EXTERN NSString * const kRTCMediaConstraintsValueTrue;
>-RTC_EXTERN NSString * const kRTCMediaConstraintsValueFalse;
>+RTC_EXTERN NSString *const kRTCMediaConstraintsValueTrue;
>+RTC_EXTERN NSString *const kRTCMediaConstraintsValueFalse;
> 
> RTC_EXPORT
> __attribute__((objc_runtime_name("WK_RTCMediaConstraints")))
>@@ -46,10 +45,9 @@ __attribute__((objc_runtime_name("WK_RTCMediaConstraints")))
> - (instancetype)init NS_UNAVAILABLE;
> 
> /** Initialize with mandatory and/or optional constraints. */
>-- (instancetype)initWithMandatoryConstraints:
>-    (nullable NSDictionary<NSString *, NSString *> *)mandatory
>-                         optionalConstraints:
>-    (nullable NSDictionary<NSString *, NSString *> *)optional
>+- (instancetype)
>+    initWithMandatoryConstraints:(nullable NSDictionary<NSString *, NSString *> *)mandatory
>+             optionalConstraints:(nullable NSDictionary<NSString *, NSString *> *)optional
>     NS_DESIGNATED_INITIALIZER;
> 
> @end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaStreamTrack.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaStreamTrack.h
>index a1161c2ac2c..914d8f2e047 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaStreamTrack.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaStreamTrack.h
>@@ -22,8 +22,8 @@ typedef NS_ENUM(NSInteger, RTCMediaStreamTrackState) {
> 
> NS_ASSUME_NONNULL_BEGIN
> 
>-RTC_EXTERN NSString * const kRTCMediaStreamTrackKindAudio;
>-RTC_EXTERN NSString * const kRTCMediaStreamTrackKindVideo;
>+RTC_EXTERN NSString *const kRTCMediaStreamTrackKindAudio;
>+RTC_EXTERN NSString *const kRTCMediaStreamTrackKindVideo;
> 
> RTC_EXPORT
> __attribute__((objc_runtime_name("WK_RTCMediaStreamTrack")))
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMetrics.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMetrics.h
>index fa2d48fa075..cc898882196 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMetrics.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMetrics.h
>@@ -17,7 +17,7 @@
>  * Enables gathering of metrics (which can be fetched with
>  * RTCGetAndResetMetrics). Must be called before any other call into WebRTC.
>  */
>-RTC_EXTERN void RTCEnableMetrics();
>+RTC_EXTERN void RTCEnableMetrics(void);
> 
> /** Gets and clears native histograms. */
>-RTC_EXTERN NSArray<RTCMetricsSampleInfo *> *RTCGetAndResetMetrics();
>+RTC_EXTERN NSArray<RTCMetricsSampleInfo*>* RTCGetAndResetMetrics(void);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCNSGLVideoView.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCNSGLVideoView.h
>index e29ec9f3c9d..77e301f189a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCNSGLVideoView.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCNSGLVideoView.h
>@@ -20,17 +20,13 @@
> NS_ASSUME_NONNULL_BEGIN
> 
> @class RTCNSGLVideoView;
>-__attribute__((objc_runtime_name("WK_RTCNSGLVideoViewDelegate")))
>-@protocol RTCNSGLVideoViewDelegate
>-
>-- (void)videoView:(RTCNSGLVideoView *)videoView didChangeVideoSize:(CGSize)size;
> 
>+@protocol RTCNSGLVideoViewDelegate <RTCVideoViewDelegate>
> @end
> 
>-__attribute__((objc_runtime_name("WK_RTCNSGLVideoView")))
> @interface RTCNSGLVideoView : NSOpenGLView <RTCVideoRenderer>
> 
>-@property(nonatomic, weak) id<RTCNSGLVideoViewDelegate> delegate;
>+@property(nonatomic, weak) id<RTCVideoViewDelegate> delegate;
> 
> - (instancetype)initWithFrame:(NSRect)frameRect
>                   pixelFormat:(NSOpenGLPixelFormat *)format
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnection.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnection.h
>index 041232e341b..0e930c524e1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnection.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnection.h
>@@ -22,12 +22,16 @@
> @class RTCPeerConnectionFactory;
> @class RTCRtpReceiver;
> @class RTCRtpSender;
>+@class RTCRtpTransceiver;
>+@class RTCRtpTransceiverInit;
> @class RTCSessionDescription;
> @class RTCLegacyStatsReport;
> 
>+typedef NS_ENUM(NSInteger, RTCRtpMediaType);
>+
> NS_ASSUME_NONNULL_BEGIN
> 
>-extern NSString * const kRTCPeerConnectionErrorDomain;
>+extern NSString *const kRTCPeerConnectionErrorDomain;
> extern int const kRTCSessionDescriptionErrorCode;
> 
> /** Represents the signaling state of the peer connection. */
>@@ -69,7 +73,6 @@ typedef NS_ENUM(NSInteger, RTCStatsOutputLevel) {
> @class RTCPeerConnection;
> 
> RTC_EXPORT
>-__attribute__((objc_runtime_name("WK_RTCPeerConnectionDelegate")))
> @protocol RTCPeerConnectionDelegate <NSObject>
> 
> /** Called when the SignalingState changed. */
>@@ -77,12 +80,12 @@ __attribute__((objc_runtime_name("WK_RTCPeerConnectionDelegate")))
>     didChangeSignalingState:(RTCSignalingState)stateChanged;
> 
> /** Called when media is received on a new stream from remote peer. */
>-- (void)peerConnection:(RTCPeerConnection *)peerConnection
>-          didAddStream:(RTCMediaStream *)stream;
>+- (void)peerConnection:(RTCPeerConnection *)peerConnection didAddStream:(RTCMediaStream *)stream;
> 
>-/** Called when a remote peer closes a stream. */
>-- (void)peerConnection:(RTCPeerConnection *)peerConnection
>-       didRemoveStream:(RTCMediaStream *)stream;
>+/** Called when a remote peer closes a stream.
>+ *  This is not called when RTCSdpSemanticsUnifiedPlan is specified.
>+ */
>+- (void)peerConnection:(RTCPeerConnection *)peerConnection didRemoveStream:(RTCMediaStream *)stream;
> 
> /** Called when negotiation is needed, for example ICE has restarted. */
> - (void)peerConnectionShouldNegotiate:(RTCPeerConnection *)peerConnection;
>@@ -107,6 +110,20 @@ __attribute__((objc_runtime_name("WK_RTCPeerConnectionDelegate")))
> - (void)peerConnection:(RTCPeerConnection *)peerConnection
>     didOpenDataChannel:(RTCDataChannel *)dataChannel;
> 
>+/** Called when signaling indicates a transceiver will be receiving media from
>+ *  the remote endpoint.
>+ *  This is only called with RTCSdpSemanticsUnifiedPlan specified.
>+ */
>+@optional
>+- (void)peerConnection:(RTCPeerConnection *)peerConnection
>+    didStartReceivingOnTransceiver:(RTCRtpTransceiver *)transceiver;
>+
>+/** Called when a receiver and its track are created. */
>+@optional
>+- (void)peerConnection:(RTCPeerConnection *)peerConnection
>+        didAddReceiver:(RTCRtpReceiver *)rtpReceiver
>+               streams:(NSArray<RTCMediaStream *> *)mediaStreams;
>+
> @end
> 
> RTC_EXPORT
>@@ -117,11 +134,12 @@ __attribute__((objc_runtime_name("WK_RTCPeerConnection")))
>  *  streams being added or removed.
>  */
> @property(nonatomic, weak, nullable) id<RTCPeerConnectionDelegate> delegate;
>+/** This property is not available with RTCSdpSemanticsUnifiedPlan. Please use
>+ *  |senders| instead.
>+ */
> @property(nonatomic, readonly) NSArray<RTCMediaStream *> *localStreams;
>-@property(nonatomic, readonly, nullable)
>-    RTCSessionDescription *localDescription;
>-@property(nonatomic, readonly, nullable)
>-    RTCSessionDescription *remoteDescription;
>+@property(nonatomic, readonly, nullable) RTCSessionDescription *localDescription;
>+@property(nonatomic, readonly, nullable) RTCSessionDescription *remoteDescription;
> @property(nonatomic, readonly) RTCSignalingState signalingState;
> @property(nonatomic, readonly) RTCIceConnectionState iceConnectionState;
> @property(nonatomic, readonly) RTCIceGatheringState iceGatheringState;
>@@ -139,6 +157,14 @@ __attribute__((objc_runtime_name("WK_RTCPeerConnection")))
>  */
> @property(nonatomic, readonly) NSArray<RTCRtpReceiver *> *receivers;
> 
>+/** Gets all RTCRtpTransceivers associated with this peer connection.
>+ *  Note: reading this property returns different instances of
>+ *  RTCRtpTransceiver. Use isEqual: instead of == to compare RTCRtpTransceiver
>+ *  instances.
>+ *  This is only available with RTCSdpSemanticsUnifiedPlan specified.
>+ */
>+@property(nonatomic, readonly) NSArray<RTCRtpTransceiver *> *transceivers;
>+
> - (instancetype)init NS_UNAVAILABLE;
> 
> /** Sets the PeerConnection's global configuration to |configuration|.
>@@ -158,33 +184,86 @@ __attribute__((objc_runtime_name("WK_RTCPeerConnection")))
> /** Remove a group of remote candidates from the ICE Agent. */
> - (void)removeIceCandidates:(NSArray<RTCIceCandidate *> *)candidates;
> 
>-/** Add a new media stream to be sent on this peer connection. */
>+/** Add a new media stream to be sent on this peer connection.
>+ *  This method is not supported with RTCSdpSemanticsUnifiedPlan. Please use
>+ *  addTrack instead.
>+ */
> - (void)addStream:(RTCMediaStream *)stream;
> 
>-/** Remove the given media stream from this peer connection. */
>+/** Remove the given media stream from this peer connection.
>+ *  This method is not supported with RTCSdpSemanticsUnifiedPlan. Please use
>+ *  removeTrack instead.
>+ */
> - (void)removeStream:(RTCMediaStream *)stream;
> 
>+/** Add a new media stream track to be sent on this peer connection, and return
>+ *  the newly created RTCRtpSender. The RTCRtpSender will be associated with
>+ *  the streams specified in the |streamIds| list.
>+ *
>+ *  Errors: If an error occurs, returns nil. An error can occur if:
>+ *  - A sender already exists for the track.
>+ *  - The peer connection is closed.
>+ */
>+- (RTCRtpSender *)addTrack:(RTCMediaStreamTrack *)track streamIds:(NSArray<NSString *> *)streamIds;
>+
>+/** With PlanB semantics, removes an RTCRtpSender from this peer connection.
>+ *
>+ *  With UnifiedPlan semantics, sets sender's track to null and removes the
>+ *  send component from the associated RTCRtpTransceiver's direction.
>+ *
>+ *  Returns YES on success.
>+ */
>+- (BOOL)removeTrack:(RTCRtpSender *)sender;
>+
>+/** addTransceiver creates a new RTCRtpTransceiver and adds it to the set of
>+ *  transceivers. Adding a transceiver will cause future calls to CreateOffer
>+ *  to add a media description for the corresponding transceiver.
>+ *
>+ *  The initial value of |mid| in the returned transceiver is nil. Setting a
>+ *  new session description may change it to a non-nil value.
>+ *
>+ *  https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnection-addtransceiver
>+ *
>+ *  Optionally, an RtpTransceiverInit structure can be specified to configure
>+ *  the transceiver from construction. If not specified, the transceiver will
>+ *  default to having a direction of kSendRecv and not be part of any streams.
>+ *
>+ *  These methods are only available when Unified Plan is enabled (see
>+ *  RTCConfiguration).
>+ */
>+
>+/** Adds a transceiver with a sender set to transmit the given track. The kind
>+ *  of the transceiver (and sender/receiver) will be derived from the kind of
>+ *  the track.
>+ */
>+- (RTCRtpTransceiver *)addTransceiverWithTrack:(RTCMediaStreamTrack *)track;
>+- (RTCRtpTransceiver *)addTransceiverWithTrack:(RTCMediaStreamTrack *)track
>+                                          init:(RTCRtpTransceiverInit *)init;
>+
>+/** Adds a transceiver with the given kind. Can either be RTCRtpMediaTypeAudio
>+ *  or RTCRtpMediaTypeVideo.
>+ */
>+- (RTCRtpTransceiver *)addTransceiverOfType:(RTCRtpMediaType)mediaType;
>+- (RTCRtpTransceiver *)addTransceiverOfType:(RTCRtpMediaType)mediaType
>+                                       init:(RTCRtpTransceiverInit *)init;
>+
> /** Generate an SDP offer. */
> - (void)offerForConstraints:(RTCMediaConstraints *)constraints
>-          completionHandler:(nullable void (^)
>-    (RTCSessionDescription * _Nullable sdp,
>-     NSError * _Nullable error))completionHandler;
>+          completionHandler:(nullable void (^)(RTCSessionDescription *_Nullable sdp,
>+                                               NSError *_Nullable error))completionHandler;
> 
> /** Generate an SDP answer. */
> - (void)answerForConstraints:(RTCMediaConstraints *)constraints
>-           completionHandler:(nullable void (^)
>-    (RTCSessionDescription * _Nullable sdp,
>-     NSError * _Nullable error))completionHandler;
>+           completionHandler:(nullable void (^)(RTCSessionDescription *_Nullable sdp,
>+                                                NSError *_Nullable error))completionHandler;
> 
> /** Apply the supplied RTCSessionDescription as the local description. */
> - (void)setLocalDescription:(RTCSessionDescription *)sdp
>-          completionHandler:
>-    (nullable void (^)(NSError * _Nullable error))completionHandler;
>+          completionHandler:(nullable void (^)(NSError *_Nullable error))completionHandler;
> 
> /** Apply the supplied RTCSessionDescription as the remote description. */
> - (void)setRemoteDescription:(RTCSessionDescription *)sdp
>-           completionHandler:
>-    (nullable void (^)(NSError * _Nullable error))completionHandler;
>+           completionHandler:(nullable void (^)(NSError *_Nullable error))completionHandler;
> 
> /** Limits the bandwidth allocated for all RTP streams sent by this
>  *  PeerConnection. Nil parameters will be unchanged. Setting
>@@ -196,21 +275,24 @@ __attribute__((objc_runtime_name("WK_RTCPeerConnection")))
>               maxBitrateBps:(nullable NSNumber *)maxBitrateBps;
> 
> /** Start or stop recording an Rtc EventLog. */
>-- (BOOL)startRtcEventLogWithFilePath:(NSString *)filePath
>-                      maxSizeInBytes:(int64_t)maxSizeInBytes;
>+- (BOOL)startRtcEventLogWithFilePath:(NSString *)filePath maxSizeInBytes:(int64_t)maxSizeInBytes;
> - (void)stopRtcEventLog;
> 
> @end
> 
>+__attribute__((objc_runtime_name("WK_RTCPeerConnection")))
> @interface RTCPeerConnection (Media)
> 
> /** Create an RTCRtpSender with the specified kind and media stream ID.
>  *  See RTCMediaStreamTrack.h for available kinds.
>+ *  This method is not supported with RTCSdpSemanticsUnifiedPlan. Please use
>+ *  addTransceiver instead.
>  */
> - (RTCRtpSender *)senderWithKind:(NSString *)kind streamId:(NSString *)streamId;
> 
> @end
> 
>+__attribute__((objc_runtime_name("WK_RTCPeerConnection")))
> @interface RTCPeerConnection (DataChannel)
> 
> /** Create a new data channel with the given label and configuration. */
>@@ -219,16 +301,15 @@ __attribute__((objc_runtime_name("WK_RTCPeerConnection")))
> 
> @end
> 
>+__attribute__((objc_runtime_name("WK_RTCPeerConnection")))
> @interface RTCPeerConnection (Stats)
> 
> /** Gather stats for the given RTCMediaStreamTrack. If |mediaStreamTrack| is nil
>  *  statistics are gathered for all tracks.
>  */
>-- (void)statsForTrack:
>-    (nullable RTCMediaStreamTrack *)mediaStreamTrack
>+- (void)statsForTrack:(nullable RTCMediaStreamTrack *)mediaStreamTrack
>      statsOutputLevel:(RTCStatsOutputLevel)statsOutputLevel
>-    completionHandler:
>-    (nullable void (^)(NSArray<RTCLegacyStatsReport *> *stats))completionHandler;
>+    completionHandler:(nullable void (^)(NSArray<RTCLegacyStatsReport *> *stats))completionHandler;
> 
> @end
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnectionFactory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnectionFactory.h
>index 8acaf43ce26..3fc70e65488 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnectionFactory.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnectionFactory.h
>@@ -14,7 +14,6 @@
> 
> NS_ASSUME_NONNULL_BEGIN
> 
>-@class RTCAVFoundationVideoSource;
> @class RTCAudioSource;
> @class RTCAudioTrack;
> @class RTCConfiguration;
>@@ -24,11 +23,8 @@ NS_ASSUME_NONNULL_BEGIN
> @class RTCVideoSource;
> @class RTCVideoTrack;
> @class RTCPeerConnectionFactoryOptions;
>-__attribute__((objc_runtime_name("WK_RTCPeerConnectionDelegate")))
> @protocol RTCPeerConnectionDelegate;
>-__attribute__((objc_runtime_name("WK_RTCVideoDecoderFactory")))
> @protocol RTCVideoDecoderFactory;
>-__attribute__((objc_runtime_name("WK_RTCVideoEncoderFactory")))
> @protocol RTCVideoEncoderFactory;
> 
> RTC_EXPORT
>@@ -51,12 +47,7 @@ __attribute__((objc_runtime_name("WK_RTCPeerConnectionFactory")))
> - (RTCAudioTrack *)audioTrackWithTrackId:(NSString *)trackId;
> 
> /** Initialize an RTCAudioTrack with a source and an id. */
>-- (RTCAudioTrack *)audioTrackWithSource:(RTCAudioSource *)source
>-                                trackId:(NSString *)trackId;
>-
>-/** Initialize an RTCAVFoundationVideoSource with constraints. */
>-- (RTCAVFoundationVideoSource *)avFoundationVideoSourceWithConstraints:
>-    (nullable RTCMediaConstraints *)constraints;
>+- (RTCAudioTrack *)audioTrackWithSource:(RTCAudioSource *)source trackId:(NSString *)trackId;
> 
> /** Initialize a generic RTCVideoSource. The RTCVideoSource should be passed to a RTCVideoCapturer
>  *  implementation, e.g. RTCCameraVideoCapturer, in order to produce frames.
>@@ -64,8 +55,7 @@ __attribute__((objc_runtime_name("WK_RTCPeerConnectionFactory")))
> - (RTCVideoSource *)videoSource;
> 
> /** Initialize an RTCVideoTrack with a source and an id. */
>-- (RTCVideoTrack *)videoTrackWithSource:(RTCVideoSource *)source
>-                                trackId:(NSString *)trackId;
>+- (RTCVideoTrack *)videoTrackWithSource:(RTCVideoSource *)source trackId:(NSString *)trackId;
> 
> /** Initialize an RTCMediaStream with an id. */
> - (RTCMediaStream *)mediaStreamWithStreamId:(NSString *)streamId;
>@@ -73,19 +63,16 @@ __attribute__((objc_runtime_name("WK_RTCPeerConnectionFactory")))
> /** Initialize an RTCPeerConnection with a configuration, constraints, and
>  *  delegate.
>  */
>-- (RTCPeerConnection *)peerConnectionWithConfiguration:
>-    (RTCConfiguration *)configuration
>-                                           constraints:
>-    (RTCMediaConstraints *)constraints
>+- (RTCPeerConnection *)peerConnectionWithConfiguration:(RTCConfiguration *)configuration
>+                                           constraints:(RTCMediaConstraints *)constraints
>                                               delegate:
>-    (nullable id<RTCPeerConnectionDelegate>)delegate;
>+                                                  (nullable id<RTCPeerConnectionDelegate>)delegate;
> 
> /** Set the options to be used for subsequently created RTCPeerConnections */
> - (void)setOptions:(nonnull RTCPeerConnectionFactoryOptions *)options;
> 
> /** Start an AecDump recording. This API call will likely change in the future. */
>-- (BOOL)startAecDumpWithFilePath:(NSString *)filePath
>-                  maxSizeInBytes:(int64_t)maxSizeInBytes;
>+- (BOOL)startAecDumpWithFilePath:(NSString *)filePath maxSizeInBytes:(int64_t)maxSizeInBytes;
> 
> /* Stop an active AecDump recording */
> - (void)stopAecDump;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnectionFactoryOptions.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnectionFactoryOptions.h
>index 9a91ae218ae..725c834440f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnectionFactoryOptions.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnectionFactoryOptions.h
>@@ -15,7 +15,6 @@
> NS_ASSUME_NONNULL_BEGIN
> 
> RTC_EXPORT
>-__attribute__((objc_runtime_name("WK_RTCPeerConnectionFactoryOptions")))
> @interface RTCPeerConnectionFactoryOptions : NSObject
> 
> @property(nonatomic, assign) BOOL disableEncryption;
>@@ -32,6 +31,10 @@ __attribute__((objc_runtime_name("WK_RTCPeerConnectionFactoryOptions")))
> 
> @property(nonatomic, assign) BOOL ignoreEthernetNetworkAdapter;
> 
>+@property(nonatomic, assign) BOOL enableAes128Sha1_32CryptoCipher;
>+
>+@property(nonatomic, assign) BOOL enableGcmCryptoSuites;
>+
> - (instancetype)init NS_DESIGNATED_INITIALIZER;
> 
> @end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtcpParameters.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtcpParameters.h
>new file mode 100644
>index 00000000000..54b254c3735
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtcpParameters.h
>@@ -0,0 +1,30 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#import <Foundation/Foundation.h>
>+
>+#import <WebRTC/RTCMacros.h>
>+
>+NS_ASSUME_NONNULL_BEGIN
>+
>+RTC_EXPORT
>+@interface RTCRtcpParameters : NSObject
>+
>+/** The Canonical Name used by RTCP. */
>+@property(nonatomic, readonly, copy) NSString *cname;
>+
>+/** Whether reduced size RTCP is configured or compound RTCP. */
>+@property(nonatomic, assign) BOOL isReducedSize;
>+
>+- (instancetype)init NS_DESIGNATED_INITIALIZER;
>+
>+@end
>+
>+NS_ASSUME_NONNULL_END
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpCodecParameters.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpCodecParameters.h
>index 2725e53d31e..341e4289884 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpCodecParameters.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpCodecParameters.h
>@@ -14,22 +14,22 @@
> 
> NS_ASSUME_NONNULL_BEGIN
> 
>-RTC_EXTERN const NSString * const kRTCRtxCodecName;
>-RTC_EXTERN const NSString * const kRTCRedCodecName;
>-RTC_EXTERN const NSString * const kRTCUlpfecCodecName;
>-RTC_EXTERN const NSString * const kRTCFlexfecCodecName;
>-RTC_EXTERN const NSString * const kRTCOpusCodecName;
>-RTC_EXTERN const NSString * const kRTCIsacCodecName;
>-RTC_EXTERN const NSString * const kRTCL16CodecName;
>-RTC_EXTERN const NSString * const kRTCG722CodecName;
>-RTC_EXTERN const NSString * const kRTCIlbcCodecName;
>-RTC_EXTERN const NSString * const kRTCPcmuCodecName;
>-RTC_EXTERN const NSString * const kRTCPcmaCodecName;
>-RTC_EXTERN const NSString * const kRTCDtmfCodecName;
>-RTC_EXTERN const NSString * const kRTCComfortNoiseCodecName;
>-RTC_EXTERN const NSString * const kRTCVp8CodecName;
>-RTC_EXTERN const NSString * const kRTCVp9CodecName;
>-RTC_EXTERN const NSString * const kRTCH264CodecName;
>+RTC_EXTERN const NSString *const kRTCRtxCodecName;
>+RTC_EXTERN const NSString *const kRTCRedCodecName;
>+RTC_EXTERN const NSString *const kRTCUlpfecCodecName;
>+RTC_EXTERN const NSString *const kRTCFlexfecCodecName;
>+RTC_EXTERN const NSString *const kRTCOpusCodecName;
>+RTC_EXTERN const NSString *const kRTCIsacCodecName;
>+RTC_EXTERN const NSString *const kRTCL16CodecName;
>+RTC_EXTERN const NSString *const kRTCG722CodecName;
>+RTC_EXTERN const NSString *const kRTCIlbcCodecName;
>+RTC_EXTERN const NSString *const kRTCPcmuCodecName;
>+RTC_EXTERN const NSString *const kRTCPcmaCodecName;
>+RTC_EXTERN const NSString *const kRTCDtmfCodecName;
>+RTC_EXTERN const NSString *const kRTCComfortNoiseCodecName;
>+RTC_EXTERN const NSString *const kRTCVp8CodecName;
>+RTC_EXTERN const NSString *const kRTCVp9CodecName;
>+RTC_EXTERN const NSString *const kRTCH264CodecName;
> 
> /** Defined in http://w3c.github.io/webrtc-pc/#idl-def-RTCRtpCodecParameters */
> RTC_EXPORT
>@@ -64,6 +64,9 @@ __attribute__((objc_runtime_name("WK_RTCRtpCodecParameters")))
>  **/
> @property(nonatomic, readonly, nullable) NSNumber *numChannels;
> 
>+/** The "format specific parameters" field from the "a=fmtp" line in the SDP */
>+@property(nonatomic, readonly, nonnull) NSDictionary *parameters;
>+
> - (instancetype)init NS_DESIGNATED_INITIALIZER;
> 
> @end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpEncodingParameters.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpEncodingParameters.h
>index 4d2e1e67152..1230e05b004 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpEncodingParameters.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpEncodingParameters.h
>@@ -26,6 +26,13 @@ __attribute__((objc_runtime_name("WK_RTCRtpEncodingParameters")))
>  */
> @property(nonatomic, copy, nullable) NSNumber *maxBitrateBps;
> 
>+/** The minimum bitrate to use for the encoding, or nil if there is no
>+ *  limit.
>+ *
>+ *  Not implemented.
>+ */
>+@property(nonatomic, copy, nullable) NSNumber *minBitrateBps;
>+
> /** The SSRC being used by this encoding. */
> @property(nonatomic, readonly, nullable) NSNumber *ssrc;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpHeaderExtension.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpHeaderExtension.h
>new file mode 100644
>index 00000000000..b89eb063e29
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpHeaderExtension.h
>@@ -0,0 +1,33 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#import <Foundation/Foundation.h>
>+
>+#import <WebRTC/RTCMacros.h>
>+
>+NS_ASSUME_NONNULL_BEGIN
>+
>+RTC_EXPORT
>+@interface RTCRtpHeaderExtension : NSObject
>+
>+/** The URI of the RTP header extension, as defined in RFC5285. */
>+@property(nonatomic, readonly, copy) NSString *uri;
>+
>+/** The value put in the RTP packet to identify the header extension. */
>+@property(nonatomic, readonly) int id;
>+
>+/** Whether the header extension is encrypted or not. */
>+@property(nonatomic, readonly, getter=isEncrypted) BOOL encrypted;
>+
>+- (instancetype)init NS_DESIGNATED_INITIALIZER;
>+
>+@end
>+
>+NS_ASSUME_NONNULL_END
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpParameters.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpParameters.h
>index 1972749d9a1..ea74644f7e0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpParameters.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpParameters.h
>@@ -11,8 +11,10 @@
> #import <Foundation/Foundation.h>
> 
> #import <WebRTC/RTCMacros.h>
>+#import <WebRTC/RTCRtcpParameters.h>
> #import <WebRTC/RTCRtpCodecParameters.h>
> #import <WebRTC/RTCRtpEncodingParameters.h>
>+#import <WebRTC/RTCRtpHeaderExtension.h>
> 
> NS_ASSUME_NONNULL_BEGIN
> 
>@@ -20,6 +22,15 @@ RTC_EXPORT
> __attribute__((objc_runtime_name("WK_RTCRtpParameters")))
> @interface RTCRtpParameters : NSObject
> 
>+/** A unique identifier for the last set of parameters applied. */
>+@property(nonatomic, copy) NSString *transactionId;
>+
>+/** Parameters used for RTCP. */
>+@property(nonatomic, readonly, copy) RTCRtcpParameters *rtcp;
>+
>+/** An array containing parameters for RTP header extensions. */
>+@property(nonatomic, readonly, copy) NSArray<RTCRtpHeaderExtension *> *headerExtensions;
>+
> /** The currently active encodings in the order of preference. */
> @property(nonatomic, copy) NSArray<RTCRtpEncodingParameters *> *encodings;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpReceiver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpReceiver.h
>index 324271876b7..a9b697220a2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpReceiver.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpReceiver.h
>@@ -26,7 +26,6 @@ typedef NS_ENUM(NSInteger, RTCRtpMediaType) {
> @class RTCRtpReceiver;
> 
> RTC_EXPORT
>-__attribute__((objc_runtime_name("WK_RTCRtpReceiverDelegate")))
> @protocol RTCRtpReceiverDelegate <NSObject>
> 
> /** Called when the first RTP packet is received.
>@@ -47,7 +46,6 @@ __attribute__((objc_runtime_name("WK_RTCRtpReceiverDelegate")))
> @end
> 
> RTC_EXPORT
>-__attribute__((objc_runtime_name("WK_RTCRtpReceiver")))
> @protocol RTCRtpReceiver <NSObject>
> 
> /** A unique identifier for this receiver. */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpSender.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpSender.h
>index bf2fbc6ac51..01ed2f749dd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpSender.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpSender.h
>@@ -18,7 +18,6 @@
> NS_ASSUME_NONNULL_BEGIN
> 
> RTC_EXPORT
>-__attribute__((objc_runtime_name("WK_RTCRtpSender")))
> @protocol RTCRtpSender <NSObject>
> 
> /** A unique identifier for this sender. */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpTransceiver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpTransceiver.h
>new file mode 100644
>index 00000000000..27fccc46c2a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpTransceiver.h
>@@ -0,0 +1,129 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#import <Foundation/Foundation.h>
>+
>+#import <WebRTC/RTCMacros.h>
>+#import <WebRTC/RTCRtpReceiver.h>
>+#import <WebRTC/RTCRtpSender.h>
>+
>+NS_ASSUME_NONNULL_BEGIN
>+
>+/** https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiverdirection */
>+typedef NS_ENUM(NSInteger, RTCRtpTransceiverDirection) {
>+  RTCRtpTransceiverDirectionSendRecv,
>+  RTCRtpTransceiverDirectionSendOnly,
>+  RTCRtpTransceiverDirectionRecvOnly,
>+  RTCRtpTransceiverDirectionInactive,
>+};
>+
>+/** Structure for initializing an RTCRtpTransceiver in a call to
>+ *  RTCPeerConnection.addTransceiver.
>+ *  https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiverinit
>+ */
>+RTC_EXPORT
>+@interface RTCRtpTransceiverInit : NSObject
>+
>+/** Direction of the RTCRtpTransceiver. See RTCRtpTransceiver.direction. */
>+@property(nonatomic) RTCRtpTransceiverDirection direction;
>+
>+/** The added RTCRtpTransceiver will be added to these streams. */
>+@property(nonatomic) NSArray<NSString *> *streamIds;
>+
>+/** TODO(bugs.webrtc.org/7600): Not implemented. */
>+@property(nonatomic) NSArray<RTCRtpEncodingParameters *> *sendEncodings;
>+
>+@end
>+
>+@class RTCRtpTransceiver;
>+
>+/** The RTCRtpTransceiver maps to the RTCRtpTransceiver defined by the WebRTC
>+ *  specification. A transceiver represents a combination of an RTCRtpSender
>+ *  and an RTCRtpReceiver that share a common mid. As defined in JSEP, an
>+ *  RTCRtpTransceiver is said to be associated with a media description if its
>+ *  mid property is non-nil; otherwise, it is said to be disassociated.
>+ *  JSEP: https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24
>+ *
>+ *  Note that RTCRtpTransceivers are only supported when using
>+ *  RTCPeerConnection with Unified Plan SDP.
>+ *
>+ *  WebRTC specification for RTCRtpTransceiver, the JavaScript analog:
>+ *  https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver
>+ */
>+RTC_EXPORT
>+@protocol RTCRtpTransceiver <NSObject>
>+
>+/** Media type of the transceiver. The sender and receiver will also have this
>+ *  type.
>+ */
>+@property(nonatomic, readonly) RTCRtpMediaType mediaType;
>+
>+/** The mid attribute is the mid negotiated and present in the local and
>+ *  remote descriptions. Before negotiation is complete, the mid value may be
>+ *  nil. After rollbacks, the value may change from a non-nil value to nil.
>+ *  https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-mid
>+ */
>+@property(nonatomic, readonly) NSString *mid;
>+
>+/** The sender attribute exposes the RTCRtpSender corresponding to the RTP
>+ *  media that may be sent with the transceiver's mid. The sender is always
>+ *  present, regardless of the direction of media.
>+ *  https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-sender
>+ */
>+@property(nonatomic, readonly) RTCRtpSender *sender;
>+
>+/** The receiver attribute exposes the RTCRtpReceiver corresponding to the RTP
>+ *  media that may be received with the transceiver's mid. The receiver is
>+ *  always present, regardless of the direction of media.
>+ *  https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-receiver
>+ */
>+@property(nonatomic, readonly) RTCRtpReceiver *receiver;
>+
>+/** The isStopped attribute indicates that the sender of this transceiver will
>+ *  no longer send, and that the receiver will no longer receive. It is true if
>+ *  either stop has been called or if setting the local or remote description
>+ *  has caused the RTCRtpTransceiver to be stopped.
>+ *  https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-stopped
>+ */
>+@property(nonatomic, readonly) BOOL isStopped;
>+
>+/** The direction attribute indicates the preferred direction of this
>+ *  transceiver, which will be used in calls to createOffer and createAnswer.
>+ *  An update of directionality does not take effect immediately. Instead,
>+ *  future calls to createOffer and createAnswer mark the corresponding media
>+ *  descriptions as sendrecv, sendonly, recvonly, or inactive.
>+ *  https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-direction
>+ */
>+@property(nonatomic) RTCRtpTransceiverDirection direction;
>+
>+/** The currentDirection attribute indicates the current direction negotiated
>+ *  for this transceiver. If this transceiver has never been represented in an
>+ *  offer/answer exchange, or if the transceiver is stopped, the value is not
>+ *  present and this method returns NO.
>+ *  https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-currentdirection
>+ */
>+- (BOOL)currentDirection:(RTCRtpTransceiverDirection *)currentDirectionOut;
>+
>+/** The stop method irreversibly stops the RTCRtpTransceiver. The sender of
>+ *  this transceiver will no longer send, the receiver will no longer receive.
>+ *  https://w3c.github.io/webrtc-pc/#dom-rtcrtptransceiver-stop
>+ */
>+- (void)stop;
>+
>+@end
>+
>+RTC_EXPORT
>+@interface RTCRtpTransceiver : NSObject <RTCRtpTransceiver>
>+
>+- (instancetype)init NS_UNAVAILABLE;
>+
>+@end
>+
>+NS_ASSUME_NONNULL_END
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCSSLAdapter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCSSLAdapter.h
>index c862d297928..a0da827c357 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCSSLAdapter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCSSLAdapter.h
>@@ -16,5 +16,5 @@
>  * Initialize and clean up the SSL library. Failure is fatal. These call the
>  * corresponding functions in webrtc/rtc_base/ssladapter.h.
>  */
>-RTC_EXTERN BOOL RTCInitializeSSL();
>-RTC_EXTERN BOOL RTCCleanupSSL();
>+RTC_EXTERN BOOL RTCInitializeSSL(void);
>+RTC_EXTERN BOOL RTCCleanupSSL(void);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCSessionDescription.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCSessionDescription.h
>index ba05e22066f..5ac62d67d3b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCSessionDescription.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCSessionDescription.h
>@@ -37,8 +37,7 @@ __attribute__((objc_runtime_name("WK_RTCSessionDescription")))
> - (instancetype)init NS_UNAVAILABLE;
> 
> /** Initialize a session description with a type and SDP string. */
>-- (instancetype)initWithType:(RTCSdpType)type sdp:(NSString *)sdp
>-    NS_DESIGNATED_INITIALIZER;
>+- (instancetype)initWithType:(RTCSdpType)type sdp:(NSString *)sdp NS_DESIGNATED_INITIALIZER;
> 
> + (NSString *)stringForType:(RTCSdpType)type;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCTracing.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCTracing.h
>index 136479118b5..899a57001ee 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCTracing.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCTracing.h
>@@ -12,10 +12,10 @@
> 
> #import <WebRTC/RTCMacros.h>
> 
>-RTC_EXTERN void RTCSetupInternalTracer();
>+RTC_EXTERN void RTCSetupInternalTracer(void);
> /** Starts capture to specified file. Must be a valid writable path.
>  *  Returns YES if capture starts.
>  */
>-RTC_EXTERN BOOL RTCStartInternalCapture(NSString *filePath);
>-RTC_EXTERN void RTCStopInternalCapture();
>-RTC_EXTERN void RTCShutdownInternalTracer();
>+RTC_EXTERN BOOL RTCStartInternalCapture(NSString* filePath);
>+RTC_EXTERN void RTCStopInternalCapture(void);
>+RTC_EXTERN void RTCShutdownInternalTracer(void);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCapturer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCapturer.h
>index 72215872bb9..60151a1ac15 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCapturer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCapturer.h
>@@ -15,7 +15,6 @@ NS_ASSUME_NONNULL_BEGIN
> @class RTCVideoCapturer;
> 
> RTC_EXPORT
>-__attribute__((objc_runtime_name("WK_RTCVideoCapturerDelegate")))
> @protocol RTCVideoCapturerDelegate <NSObject>
> - (void)capturer:(RTCVideoCapturer *)capturer didCaptureVideoFrame:(RTCVideoFrame *)frame;
> @end
>@@ -24,7 +23,7 @@ RTC_EXPORT
> __attribute__((objc_runtime_name("WK_RTCVideoCapturer")))
> @interface RTCVideoCapturer : NSObject
> 
>-@property(nonatomic, readonly, weak) id<RTCVideoCapturerDelegate> delegate;
>+@property(nonatomic, weak) id<RTCVideoCapturerDelegate> delegate;
> 
> - (instancetype)initWithDelegate:(id<RTCVideoCapturerDelegate>)delegate;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodec.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodec.h
>index 345c22e53c2..6fccf44fb93 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodec.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodec.h
>@@ -20,6 +20,8 @@ RTC_EXPORT extern NSString *const kRTCVideoCodecVp9Name;
> RTC_EXPORT extern NSString *const kRTCVideoCodecH264Name;
> RTC_EXPORT extern NSString *const kRTCLevel31ConstrainedHigh;
> RTC_EXPORT extern NSString *const kRTCLevel31ConstrainedBaseline;
>+RTC_EXPORT extern NSString *const kRTCMaxSupportedH264ProfileLevelConstrainedHigh;
>+RTC_EXPORT extern NSString *const kRTCMaxSupportedH264ProfileLevelConstrainedBaseline;
> 
> /** Represents an encoded frame's type. */
> typedef NS_ENUM(NSUInteger, RTCFrameType) {
>@@ -73,7 +75,6 @@ __attribute__((objc_runtime_name("WK_RTCRtpFragmentationHeader")))
>  *  Corresponds to webrtc::CodecSpecificInfo.
>  */
> RTC_EXPORT
>-__attribute__((objc_runtime_name("WK_RTCCodecSpecificInfo")))
> @protocol RTCCodecSpecificInfo <NSObject>
> 
> @end
>@@ -91,7 +92,7 @@ typedef NS_ENUM(NSUInteger, RTCVideoCodecMode) {
>   RTCVideoCodecModeScreensharing,
> };
> 
>-/** Holds information to identify a codec. Corresponds to cricket::VideoCodec. */
>+/** Holds information to identify a codec. Corresponds to webrtc::SdpVideoFormat. */
> RTC_EXPORT
> __attribute__((objc_runtime_name("WK_RTCVideoCodecInfo")))
> @interface RTCVideoCodecInfo : NSObject <NSCoding>
>@@ -147,7 +148,6 @@ __attribute__((objc_runtime_name("WK_RTCVideoEncoderQpThresholds")))
> 
> /** Protocol for encoder implementations. */
> RTC_EXPORT
>-__attribute__((objc_runtime_name("WK_RTCVideoEncoder")))
> @protocol RTCVideoEncoder <NSObject>
> 
> - (void)setCallback:(RTCVideoEncoderCallback)callback;
>@@ -169,20 +169,23 @@ __attribute__((objc_runtime_name("WK_RTCVideoEncoder")))
> 
> /** Protocol for decoder implementations. */
> RTC_EXPORT
>-__attribute__((objc_runtime_name("WK_RTCVideoDecoder")))
> @protocol RTCVideoDecoder <NSObject>
> 
> - (void)setCallback:(RTCVideoDecoderCallback)callback;
> - (NSInteger)startDecodeWithSettings:(RTCVideoEncoderSettings *)settings
>-                       numberOfCores:(int)numberOfCores;
>+                       numberOfCores:(int)numberOfCores
>+    DEPRECATED_MSG_ATTRIBUTE("use startDecodeWithNumberOfCores: instead");
> - (NSInteger)releaseDecoder;
> - (NSInteger)decode:(RTCEncodedImage *)encodedImage
>-          missingFrames:(BOOL)missingFrames
>-    fragmentationHeader:(RTCRtpFragmentationHeader *)fragmentationHeader
>-      codecSpecificInfo:(nullable id<RTCCodecSpecificInfo>)info
>-           renderTimeMs:(int64_t)renderTimeMs;
>+        missingFrames:(BOOL)missingFrames
>+    codecSpecificInfo:(nullable id<RTCCodecSpecificInfo>)info
>+         renderTimeMs:(int64_t)renderTimeMs;
> - (NSString *)implementationName;
> 
>+// TODO(andersc): Make non-optional when `startDecodeWithSettings:numberOfCores:` is removed.
>+@optional
>+- (NSInteger)startDecodeWithNumberOfCores:(int)numberOfCores;
>+
> @end
> 
> NS_ASSUME_NONNULL_END
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodecFactory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodecFactory.h
>index 9fb1cb71d99..e9053025de7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodecFactory.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodecFactory.h
>@@ -17,7 +17,6 @@ NS_ASSUME_NONNULL_BEGIN
> 
> /** RTCVideoEncoderFactory is an Objective-C version of webrtc::VideoEncoderFactory. */
> RTC_EXPORT
>-__attribute__((objc_runtime_name("WK_RTCVideoEncoderFactory")))
> @protocol RTCVideoEncoderFactory <NSObject>
> 
> - (nullable id<RTCVideoEncoder>)createEncoder:(RTCVideoCodecInfo *)info;
>@@ -27,7 +26,6 @@ __attribute__((objc_runtime_name("WK_RTCVideoEncoderFactory")))
> 
> /** RTCVideoDecoderFactory is an Objective-C version of webrtc::VideoDecoderFactory. */
> RTC_EXPORT
>-__attribute__((objc_runtime_name("WK_RTCVideoDecoderFactory")))
> @protocol RTCVideoDecoderFactory <NSObject>
> 
> - (nullable id<RTCVideoDecoder>)createDecoder:(RTCVideoCodecInfo *)info;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodecH264.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodecH264.h
>index c8d7fc007f1..84bf9b75028 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodecH264.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodecH264.h
>@@ -27,10 +27,52 @@ __attribute__((objc_runtime_name("WK_RTCCodecSpecificInfoH264")))
> 
> @end
> 
>+/** H264 Profiles and levels. */
>+typedef NS_ENUM(NSUInteger, RTCH264Profile) {
>+  RTCH264ProfileConstrainedBaseline,
>+  RTCH264ProfileBaseline,
>+  RTCH264ProfileMain,
>+  RTCH264ProfileConstrainedHigh,
>+  RTCH264ProfileHigh,
>+};
>+
>+typedef NS_ENUM(NSUInteger, RTCH264Level) {
>+  RTCH264Level1_b = 0,
>+  RTCH264Level1 = 10,
>+  RTCH264Level1_1 = 11,
>+  RTCH264Level1_2 = 12,
>+  RTCH264Level1_3 = 13,
>+  RTCH264Level2 = 20,
>+  RTCH264Level2_1 = 21,
>+  RTCH264Level2_2 = 22,
>+  RTCH264Level3 = 30,
>+  RTCH264Level3_1 = 31,
>+  RTCH264Level3_2 = 32,
>+  RTCH264Level4 = 40,
>+  RTCH264Level4_1 = 41,
>+  RTCH264Level4_2 = 42,
>+  RTCH264Level5 = 50,
>+  RTCH264Level5_1 = 51,
>+  RTCH264Level5_2 = 52
>+};
>+
>+RTC_EXPORT
>+__attribute__((objc_runtime_name("WK_RTCH264ProfileLevelId")))
>+@interface RTCH264ProfileLevelId : NSObject
>+
>+@property(nonatomic, readonly) RTCH264Profile profile;
>+@property(nonatomic, readonly) RTCH264Level level;
>+@property(nonatomic, readonly) NSString *hexString;
>+
>+- (instancetype)initWithHexString:(NSString *)hexString;
>+- (instancetype)initWithProfile:(RTCH264Profile)profile level:(RTCH264Level)level;
>+
>+@end
>+
> /** Encoder. */
> RTC_EXPORT
> __attribute__((objc_runtime_name("WK_RTCVideoEncoderH264")))
>-@interface RTCVideoEncoderH264 : NSObject<RTCVideoEncoder>
>+@interface RTCVideoEncoderH264 : NSObject <RTCVideoEncoder>
> 
> - (instancetype)initWithCodecInfo:(RTCVideoCodecInfo *)codecInfo;
> 
>@@ -39,17 +81,17 @@ __attribute__((objc_runtime_name("WK_RTCVideoEncoderH264")))
> /** Decoder. */
> RTC_EXPORT
> __attribute__((objc_runtime_name("WK_RTCVideoDecoderH264")))
>-@interface RTCVideoDecoderH264 : NSObject<RTCVideoDecoder>
>+@interface RTCVideoDecoderH264 : NSObject <RTCVideoDecoder>
> @end
> 
> /** Encoder factory. */
> RTC_EXPORT
> __attribute__((objc_runtime_name("WK_RTCVideoEncoderFactoryH264")))
>-@interface RTCVideoEncoderFactoryH264 : NSObject<RTCVideoEncoderFactory>
>+@interface RTCVideoEncoderFactoryH264 : NSObject <RTCVideoEncoderFactory>
> @end
> 
> /** Decoder factory. */
> RTC_EXPORT
> __attribute__((objc_runtime_name("WK_RTCVideoDecoderFactoryH264")))
>-@interface RTCVideoDecoderFactoryH264 : NSObject<RTCVideoDecoderFactory>
>+@interface RTCVideoDecoderFactoryH264 : NSObject <RTCVideoDecoderFactory>
> @end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoFrame.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoFrame.h
>index 5b8fba602df..809363674b4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoFrame.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoFrame.h
>@@ -22,7 +22,6 @@ typedef NS_ENUM(NSInteger, RTCVideoRotation) {
>   RTCVideoRotation_270 = 270,
> };
> 
>-__attribute__((objc_runtime_name("WK_RTCVideoFrameBuffer")))
> @protocol RTCVideoFrameBuffer;
> 
> // RTCVideoFrame is an ObjectiveC version of webrtc::VideoFrame.
>@@ -46,7 +45,7 @@ __attribute__((objc_runtime_name("WK_RTCVideoFrame")))
> @property(nonatomic, readonly) id<RTCVideoFrameBuffer> buffer;
> 
> - (instancetype)init NS_UNAVAILABLE;
>-- (instancetype)new NS_UNAVAILABLE;
>+- (instancetype) new NS_UNAVAILABLE;
> 
> /** Initialize an RTCVideoFrame from a pixel buffer, rotation, and timestamp.
>  *  Deprecated - initialize with a RTCCVPixelBuffer instead
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoFrameBuffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoFrameBuffer.h
>index 8b825d8050c..3f342536e38 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoFrameBuffer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoFrameBuffer.h
>@@ -13,12 +13,10 @@
> 
> NS_ASSUME_NONNULL_BEGIN
> 
>-__attribute__((objc_runtime_name("WK_RTCI420Buffer")))
> @protocol RTCI420Buffer;
> 
> // RTCVideoFrameBuffer is an ObjectiveC version of webrtc::VideoFrameBuffer.
> RTC_EXPORT
>-__attribute__((objc_runtime_name("WK_RTCVideoFrameBuffer")))
> @protocol RTCVideoFrameBuffer <NSObject>
> 
> @property(nonatomic, readonly) int width;
>@@ -29,7 +27,6 @@ __attribute__((objc_runtime_name("WK_RTCVideoFrameBuffer")))
> @end
> 
> /** Protocol for RTCVideoFrameBuffers containing YUV planar data. */
>-__attribute__((objc_runtime_name("WK_RTCYUVPlanarBuffer")))
> @protocol RTCYUVPlanarBuffer <RTCVideoFrameBuffer>
> 
> @property(nonatomic, readonly) int chromaWidth;
>@@ -41,6 +38,11 @@ __attribute__((objc_runtime_name("WK_RTCYUVPlanarBuffer")))
> @property(nonatomic, readonly) int strideU;
> @property(nonatomic, readonly) int strideV;
> 
>+- (instancetype)initWithWidth:(int)width
>+                       height:(int)height
>+                        dataY:(const uint8_t *)dataY
>+                        dataU:(const uint8_t *)dataU
>+                        dataV:(const uint8_t *)dataV;
> - (instancetype)initWithWidth:(int)width height:(int)height;
> - (instancetype)initWithWidth:(int)width
>                        height:(int)height
>@@ -51,7 +53,6 @@ __attribute__((objc_runtime_name("WK_RTCYUVPlanarBuffer")))
> @end
> 
> /** Extension of the YUV planar data buffer with mutable data access */
>-__attribute__((objc_runtime_name("WK_RTCMutableYUVPlanarBuffer")))
> @protocol RTCMutableYUVPlanarBuffer <RTCYUVPlanarBuffer>
> 
> @property(nonatomic, readonly) uint8_t *mutableDataY;
>@@ -61,12 +62,10 @@ __attribute__((objc_runtime_name("WK_RTCMutableYUVPlanarBuffer")))
> @end
> 
> /** Protocol for RTCYUVPlanarBuffers containing I420 data */
>-__attribute__((objc_runtime_name("WK_RTCI420Buffer")))
> @protocol RTCI420Buffer <RTCYUVPlanarBuffer>
> @end
> 
> /** Extension of the I420 buffer with mutable data access */
>-__attribute__((objc_runtime_name("WK_RTCMutableI420Buffer")))
> @protocol RTCMutableI420Buffer <RTCI420Buffer, RTCMutableYUVPlanarBuffer>
> @end
> 
>@@ -78,6 +77,8 @@ __attribute__((objc_runtime_name("WK_RTCCVPixelBuffer")))
> @property(nonatomic, readonly) CVPixelBufferRef pixelBuffer;
> @property(nonatomic, readonly) int cropX;
> @property(nonatomic, readonly) int cropY;
>+@property(nonatomic, readonly) int cropWidth;
>+@property(nonatomic, readonly) int cropHeight;
> 
> + (NSSet<NSNumber *> *)supportedPixelFormats;
> 
>@@ -93,10 +94,13 @@ __attribute__((objc_runtime_name("WK_RTCCVPixelBuffer")))
> - (BOOL)requiresCropping;
> - (BOOL)requiresScalingToWidth:(int)width height:(int)height;
> - (int)bufferSizeForCroppingAndScalingToWidth:(int)width height:(int)height;
>+
> /** The minimum size of the |tmpBuffer| must be the number of bytes returned from the
>  * bufferSizeForCroppingAndScalingToWidth:height: method.
>+ * If that size is 0, the |tmpBuffer| may be nil.
>  */
>-- (BOOL)cropAndScaleTo:(CVPixelBufferRef)outputPixelBuffer withTempBuffer:(uint8_t *)tmpBuffer;
>+- (BOOL)cropAndScaleTo:(CVPixelBufferRef)outputPixelBuffer
>+        withTempBuffer:(nullable uint8_t *)tmpBuffer;
> 
> @end
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoRenderer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoRenderer.h
>index c821a1e3008..7d91f2f14d0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoRenderer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoRenderer.h
>@@ -20,7 +20,6 @@ NS_ASSUME_NONNULL_BEGIN
> @class RTCVideoFrame;
> 
> RTC_EXPORT
>-__attribute__((objc_runtime_name("WK_RTCVideoRenderer")))
> @protocol RTCVideoRenderer <NSObject>
> 
> /** The size of the frame. */
>@@ -31,4 +30,11 @@ __attribute__((objc_runtime_name("WK_RTCVideoRenderer")))
> 
> @end
> 
>+RTC_EXPORT
>+@protocol RTCVideoViewDelegate
>+
>+- (void)videoView:(id<RTCVideoRenderer>)videoView didChangeVideoSize:(CGSize)size;
>+
>+@end
>+
> NS_ASSUME_NONNULL_END
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoTrack.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoTrack.h
>index ecbdbb93c68..e381c3d1e3e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoTrack.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoTrack.h
>@@ -14,7 +14,6 @@
> 
> NS_ASSUME_NONNULL_BEGIN
> 
>-__attribute__((objc_runtime_name("WK_RTCVideoRenderer")))
> @protocol RTCVideoRenderer;
> @class RTCPeerConnectionFactory;
> @class RTCVideoSource;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoViewShading.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoViewShading.h
>index 5e33ff235ca..6f3ece94a1f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoViewShading.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoViewShading.h
>@@ -19,7 +19,6 @@ NS_ASSUME_NONNULL_BEGIN
>  * rendering for the RTCEAGLVideoView/RTCNSGLVideoView.
>  */
> RTC_EXPORT
>-__attribute__((objc_runtime_name("WK_RTCVideoViewShading")))
> @protocol RTCVideoViewShading <NSObject>
> 
> /** Callback for I420 frames. Each plane is given as a texture. */
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/UIDevice+RTCDevice.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/UIDevice+RTCDevice.h
>index 79d2db8c4aa..eb39ea9cc4d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/UIDevice+RTCDevice.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/UIDevice+RTCDevice.h
>@@ -30,6 +30,7 @@ typedef NS_ENUM(NSInteger, RTCDeviceType) {
>   RTCDeviceTypeIPhone6SPlus,
>   RTCDeviceTypeIPhone7,
>   RTCDeviceTypeIPhone7Plus,
>+  RTCDeviceTypeIPhoneSE,
>   RTCDeviceTypeIPhone8,
>   RTCDeviceTypeIPhone8Plus,
>   RTCDeviceTypeIPhoneX,
>@@ -38,6 +39,7 @@ typedef NS_ENUM(NSInteger, RTCDeviceType) {
>   RTCDeviceTypeIPodTouch3G,
>   RTCDeviceTypeIPodTouch4G,
>   RTCDeviceTypeIPodTouch5G,
>+  RTCDeviceTypeIPodTouch6G,
>   RTCDeviceTypeIPad,
>   RTCDeviceTypeIPad2Wifi,
>   RTCDeviceTypeIPad2GSM,
>@@ -52,10 +54,21 @@ typedef NS_ENUM(NSInteger, RTCDeviceType) {
>   RTCDeviceTypeIPad4Wifi,
>   RTCDeviceTypeIPad4GSM,
>   RTCDeviceTypeIPad4GSM_CDMA,
>+  RTCDeviceTypeIPad5,
>+  RTCDeviceTypeIPad6,
>   RTCDeviceTypeIPadAirWifi,
>   RTCDeviceTypeIPadAirCellular,
>+  RTCDeviceTypeIPadAirWifiCellular,
>+  RTCDeviceTypeIPadAir2,
>   RTCDeviceTypeIPadMini2GWifi,
>   RTCDeviceTypeIPadMini2GCellular,
>+  RTCDeviceTypeIPadMini2GWifiCellular,
>+  RTCDeviceTypeIPadMini3,
>+  RTCDeviceTypeIPadMini4,
>+  RTCDeviceTypeIPadPro9Inch,
>+  RTCDeviceTypeIPadPro12Inch,
>+  RTCDeviceTypeIPadPro12Inch2,
>+  RTCDeviceTypeIPadPro10Inch,
>   RTCDeviceTypeSimulatori386,
>   RTCDeviceTypeSimulatorx86_64,
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/api/audio_device_module.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/api/audio_device_module.h
>new file mode 100644
>index 00000000000..aaeb39b18c2
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/api/audio_device_module.h
>@@ -0,0 +1,24 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef SDK_OBJC_FRAMEWORK_NATIVE_API_AUDIO_DEVICE_MODULE_H_
>+#define SDK_OBJC_FRAMEWORK_NATIVE_API_AUDIO_DEVICE_MODULE_H_
>+
>+#include <memory>
>+
>+#include "modules/audio_device/include/audio_device.h"
>+
>+namespace webrtc {
>+
>+rtc::scoped_refptr<AudioDeviceModule> CreateAudioDeviceModule();
>+
>+}  // namespace webrtc
>+
>+#endif  // SDK_OBJC_FRAMEWORK_NATIVE_API_AUDIO_DEVICE_MODULE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/api/audio_device_module.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/api/audio_device_module.mm
>new file mode 100644
>index 00000000000..2afa6dfc210
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/api/audio_device_module.mm
>@@ -0,0 +1,30 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "audio_device_module.h"
>+
>+#include "rtc_base/logging.h"
>+#include "rtc_base/refcountedobject.h"
>+
>+#include "sdk/objc/Framework/Native/src/audio/audio_device_module_ios.h"
>+
>+namespace webrtc {
>+
>+rtc::scoped_refptr<AudioDeviceModule> CreateAudioDeviceModule() {
>+  RTC_LOG(INFO) << __FUNCTION__;
>+#if defined(WEBRTC_IOS)
>+  return new rtc::RefCountedObject<ios_adm::AudioDeviceModuleIOS>();
>+#else
>+  RTC_LOG(LERROR)
>+      << "current platform is not supported => this module will self destruct!";
>+  return nullptr;
>+#endif
>+}
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/api/video_capturer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/api/video_capturer.h
>new file mode 100644
>index 00000000000..f601f1840d8
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/api/video_capturer.h
>@@ -0,0 +1,28 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef SDK_OBJC_FRAMEWORK_NATIVE_API_VIDEO_CAPTURER_H_
>+#define SDK_OBJC_FRAMEWORK_NATIVE_API_VIDEO_CAPTURER_H_
>+
>+#import "WebRTC/RTCVideoCapturer.h"
>+
>+#include "api/mediastreaminterface.h"
>+#include "rtc_base/scoped_ref_ptr.h"
>+
>+namespace webrtc {
>+
>+rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> ObjCToNativeVideoCapturer(
>+    RTCVideoCapturer* objc_video_capturer,
>+    rtc::Thread* signaling_thread,
>+    rtc::Thread* worker_thread);
>+
>+}  // namespace webrtc
>+
>+#endif  // SDK_OBJC_FRAMEWORK_NATIVE_API_VIDEO_CAPTURER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/api/video_capturer.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/api/video_capturer.mm
>new file mode 100644
>index 00000000000..4868c2e21bf
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/api/video_capturer.mm
>@@ -0,0 +1,35 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "sdk/objc/Framework/Native/api/video_capturer.h"
>+
>+#include "absl/memory/memory.h"
>+#include "api/videosourceproxy.h"
>+#include "sdk/objc/Framework/Native/src/objc_video_track_source.h"
>+
>+namespace webrtc {
>+
>+rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> ObjCToNativeVideoCapturer(
>+    RTCVideoCapturer *objc_video_capturer,
>+    rtc::Thread *signaling_thread,
>+    rtc::Thread *worker_thread) {
>+  RTCObjCVideoSourceAdapter *adapter = [[RTCObjCVideoSourceAdapter alloc] init];
>+  rtc::scoped_refptr<webrtc::ObjCVideoTrackSource> objc_video_track_source(
>+      new rtc::RefCountedObject<webrtc::ObjCVideoTrackSource>(adapter));
>+  rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> video_source =
>+      webrtc::VideoTrackSourceProxy::Create(
>+          signaling_thread, worker_thread, objc_video_track_source);
>+
>+  objc_video_capturer.delegate = adapter;
>+
>+  return video_source;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/api/video_decoder_factory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/api/video_decoder_factory.h
>new file mode 100644
>index 00000000000..9fda6a95272
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/api/video_decoder_factory.h
>@@ -0,0 +1,27 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef SDK_OBJC_FRAMEWORK_NATIVE_API_VIDEO_DECODER_FACTORY_H_
>+#define SDK_OBJC_FRAMEWORK_NATIVE_API_VIDEO_DECODER_FACTORY_H_
>+
>+#include <memory>
>+
>+#import "WebRTC/RTCVideoCodecFactory.h"
>+
>+#include "api/video_codecs/video_decoder_factory.h"
>+
>+namespace webrtc {
>+
>+std::unique_ptr<VideoDecoderFactory> ObjCToNativeVideoDecoderFactory(
>+    id<RTCVideoDecoderFactory> objc_video_decoder_factory);
>+
>+}  // namespace webrtc
>+
>+#endif  // SDK_OBJC_FRAMEWORK_NATIVE_API_VIDEO_DECODER_FACTORY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/api/video_decoder_factory.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/api/video_decoder_factory.mm
>new file mode 100644
>index 00000000000..32d286a78a3
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/api/video_decoder_factory.mm
>@@ -0,0 +1,23 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "sdk/objc/Framework/Native/api/video_decoder_factory.h"
>+
>+#include "absl/memory/memory.h"
>+#include "sdk/objc/Framework/Native/src/objc_video_decoder_factory.h"
>+
>+namespace webrtc {
>+
>+std::unique_ptr<VideoDecoderFactory> ObjCToNativeVideoDecoderFactory(
>+    id<RTCVideoDecoderFactory> objc_video_decoder_factory) {
>+  return absl::make_unique<ObjCVideoDecoderFactory>(objc_video_decoder_factory);
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/api/video_encoder_factory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/api/video_encoder_factory.h
>new file mode 100644
>index 00000000000..865a707b008
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/api/video_encoder_factory.h
>@@ -0,0 +1,27 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef SDK_OBJC_FRAMEWORK_NATIVE_API_VIDEO_ENCODER_FACTORY_H_
>+#define SDK_OBJC_FRAMEWORK_NATIVE_API_VIDEO_ENCODER_FACTORY_H_
>+
>+#include <memory>
>+
>+#import "WebRTC/RTCVideoCodecFactory.h"
>+
>+#include "api/video_codecs/video_encoder_factory.h"
>+
>+namespace webrtc {
>+
>+std::unique_ptr<VideoEncoderFactory> ObjCToNativeVideoEncoderFactory(
>+    id<RTCVideoEncoderFactory> objc_video_encoder_factory);
>+
>+}  // namespace webrtc
>+
>+#endif  // SDK_OBJC_FRAMEWORK_NATIVE_API_VIDEO_ENCODER_FACTORY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/api/video_encoder_factory.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/api/video_encoder_factory.mm
>new file mode 100644
>index 00000000000..f94c41a5624
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/api/video_encoder_factory.mm
>@@ -0,0 +1,23 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "sdk/objc/Framework/Native/api/video_encoder_factory.h"
>+
>+#include "absl/memory/memory.h"
>+#include "sdk/objc/Framework/Native/src/objc_video_encoder_factory.h"
>+
>+namespace webrtc {
>+
>+std::unique_ptr<VideoEncoderFactory> ObjCToNativeVideoEncoderFactory(
>+    id<RTCVideoEncoderFactory> objc_video_encoder_factory) {
>+  return absl::make_unique<ObjCVideoEncoderFactory>(objc_video_encoder_factory);
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoFrame+Private.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/api/video_frame.h
>similarity index 51%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoFrame+Private.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/api/video_frame.h
>index 43d3fbf68e3..c64d6fc28a4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoFrame+Private.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/api/video_frame.h
>@@ -1,5 +1,5 @@
> /*
>- *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>  *
>  *  Use of this source code is governed by a BSD-style license
>  *  that can be found in the LICENSE file in the root of the source
>@@ -8,20 +8,17 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#import "RTCI420Buffer+Private.h"
>+#ifndef SDK_OBJC_FRAMEWORK_NATIVE_API_VIDEO_FRAME_H_
>+#define SDK_OBJC_FRAMEWORK_NATIVE_API_VIDEO_FRAME_H_
>+
> #import "WebRTC/RTCVideoFrame.h"
>-#import "WebRTC/RTCVideoFrameBuffer.h"
> 
> #include "api/video/video_frame.h"
>-#include "sdk/objc/Framework/Classes/Video/objc_frame_buffer.h"
>-
>-NS_ASSUME_NONNULL_BEGIN
> 
>-@interface RTCVideoFrame ()
>+namespace webrtc {
> 
>-- (instancetype)initWithNativeVideoFrame:(const webrtc::VideoFrame &)frame;
>-- (webrtc::VideoFrame)nativeVideoFrame;
>+RTCVideoFrame* NativeToObjCVideoFrame(const VideoFrame& frame);
> 
>-@end
>+}  // namespace webrtc
> 
>-NS_ASSUME_NONNULL_END
>+#endif  // SDK_OBJC_FRAMEWORK_NATIVE_API_VIDEO_FRAME_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/api/video_frame.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/api/video_frame.mm
>new file mode 100644
>index 00000000000..7fe9124beef
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/api/video_frame.mm
>@@ -0,0 +1,21 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "sdk/objc/Framework/Native/api/video_frame.h"
>+
>+#include "sdk/objc/Framework/Native/src/objc_video_frame.h"
>+
>+namespace webrtc {
>+
>+RTCVideoFrame* NativeToObjCVideoFrame(const VideoFrame& frame) {
>+  return ToObjCVideoFrame(frame);
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/api/video_frame_buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/api/video_frame_buffer.h
>new file mode 100644
>index 00000000000..1009c76bb7e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/api/video_frame_buffer.h
>@@ -0,0 +1,29 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef SDK_OBJC_FRAMEWORK_NATIVE_API_VIDEO_FRAME_BUFFER_H_
>+#define SDK_OBJC_FRAMEWORK_NATIVE_API_VIDEO_FRAME_BUFFER_H_
>+
>+#import "WebRTC/RTCVideoFrameBuffer.h"
>+
>+#include "common_video/include/video_frame_buffer.h"
>+#include "rtc_base/scoped_ref_ptr.h"
>+
>+namespace webrtc {
>+
>+rtc::scoped_refptr<VideoFrameBuffer> ObjCToNativeVideoFrameBuffer(
>+    id<RTCVideoFrameBuffer> objc_video_frame_buffer);
>+
>+id<RTCVideoFrameBuffer> NativeToObjCVideoFrameBuffer(
>+    const rtc::scoped_refptr<VideoFrameBuffer>& buffer);
>+
>+}  // namespace webrtc
>+
>+#endif  // SDK_OBJC_FRAMEWORK_NATIVE_API_VIDEO_FRAME_BUFFER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/api/video_frame_buffer.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/api/video_frame_buffer.mm
>new file mode 100644
>index 00000000000..149c22a01e7
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/api/video_frame_buffer.mm
>@@ -0,0 +1,27 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "sdk/objc/Framework/Native/api/video_frame_buffer.h"
>+
>+#include "sdk/objc/Framework/Native/src/objc_frame_buffer.h"
>+
>+namespace webrtc {
>+
>+rtc::scoped_refptr<VideoFrameBuffer> ObjCToNativeVideoFrameBuffer(
>+    id<RTCVideoFrameBuffer> objc_video_frame_buffer) {
>+  return new rtc::RefCountedObject<ObjCFrameBuffer>(objc_video_frame_buffer);
>+}
>+
>+id<RTCVideoFrameBuffer> NativeToObjCVideoFrameBuffer(
>+    const rtc::scoped_refptr<VideoFrameBuffer> &buffer) {
>+  return ToObjCVideoFrameBuffer(buffer);
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/api/video_renderer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/api/video_renderer.h
>new file mode 100644
>index 00000000000..ffba02b3be6
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/api/video_renderer.h
>@@ -0,0 +1,28 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef SDK_OBJC_FRAMEWORK_NATIVE_API_VIDEO_RENDERER_H_
>+#define SDK_OBJC_FRAMEWORK_NATIVE_API_VIDEO_RENDERER_H_
>+
>+#import "WebRTC/RTCVideoRenderer.h"
>+
>+#include <memory>
>+
>+#include "api/video/video_frame.h"
>+#include "api/video/video_sink_interface.h"
>+
>+namespace webrtc {
>+
>+std::unique_ptr<rtc::VideoSinkInterface<VideoFrame>> ObjCToNativeVideoRenderer(
>+    id<RTCVideoRenderer> objc_video_renderer);
>+
>+}  // namespace webrtc
>+
>+#endif  // SDK_OBJC_FRAMEWORK_NATIVE_API_VIDEO_RENDERER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/api/video_renderer.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/api/video_renderer.mm
>new file mode 100644
>index 00000000000..963e474835d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/api/video_renderer.mm
>@@ -0,0 +1,23 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "sdk/objc/Framework/Native/api/video_renderer.h"
>+
>+#include "absl/memory/memory.h"
>+#include "sdk/objc/Framework/Native/src/objc_video_renderer.h"
>+
>+namespace webrtc {
>+
>+std::unique_ptr<rtc::VideoSinkInterface<VideoFrame>> ObjCToNativeVideoRenderer(
>+    id<RTCVideoRenderer> objc_video_renderer) {
>+  return absl::make_unique<ObjCVideoRenderer>(objc_video_renderer);
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/audio/audio_device_ios.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/audio/audio_device_ios.h
>new file mode 100644
>index 00000000000..23e54b1409c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/audio/audio_device_ios.h
>@@ -0,0 +1,296 @@
>+/*
>+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_DEVICE_IOS_AUDIO_DEVICE_IOS_H_
>+#define MODULES_AUDIO_DEVICE_IOS_AUDIO_DEVICE_IOS_H_
>+
>+#include <memory>
>+
>+#include "audio_session_observer.h"
>+#include "modules/audio_device/audio_device_generic.h"
>+#include "rtc_base/buffer.h"
>+#include "rtc_base/thread.h"
>+#include "rtc_base/thread_annotations.h"
>+#include "rtc_base/thread_checker.h"
>+#include "sdk/objc/Framework/Headers/WebRTC/RTCMacros.h"
>+#include "voice_processing_audio_unit.h"
>+
>+RTC_FWD_DECL_OBJC_CLASS(RTCNativeAudioSessionDelegateAdapter);
>+
>+namespace webrtc {
>+
>+class FineAudioBuffer;
>+
>+namespace ios_adm {
>+
>+// Implements full duplex 16-bit mono PCM audio support for iOS using a
>+// Voice-Processing (VP) I/O audio unit in Core Audio. The VP I/O audio unit
>+// supports audio echo cancellation. It also adds automatic gain control,
>+// adjustment of voice-processing quality and muting.
>+//
>+// An instance must be created and destroyed on one and the same thread.
>+// All supported public methods must also be called on the same thread.
>+// A thread checker will RTC_DCHECK if any supported method is called on an
>+// invalid thread.
>+//
>+// Recorded audio will be delivered on a real-time internal I/O thread in the
>+// audio unit. The audio unit will also ask for audio data to play out on this
>+// same thread.
>+class AudioDeviceIOS : public AudioDeviceGeneric,
>+                       public AudioSessionObserver,
>+                       public VoiceProcessingAudioUnitObserver,
>+                       public rtc::MessageHandler {
>+ public:
>+  AudioDeviceIOS();
>+  ~AudioDeviceIOS() override;
>+
>+  void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override;
>+
>+  InitStatus Init() override;
>+  int32_t Terminate() override;
>+  bool Initialized() const override;
>+
>+  int32_t InitPlayout() override;
>+  bool PlayoutIsInitialized() const override;
>+
>+  int32_t InitRecording() override;
>+  bool RecordingIsInitialized() const override;
>+
>+  int32_t StartPlayout() override;
>+  int32_t StopPlayout() override;
>+  bool Playing() const override;
>+
>+  int32_t StartRecording() override;
>+  int32_t StopRecording() override;
>+  bool Recording() const override;
>+
>+  // These methods returns hard-coded delay values and not dynamic delay
>+  // estimates. The reason is that iOS supports a built-in AEC and the WebRTC
>+  // AEC will always be disabled in the Libjingle layer to avoid running two
>+  // AEC implementations at the same time. And, it saves resources to avoid
>+  // updating these delay values continuously.
>+  // TODO(henrika): it would be possible to mark these two methods as not
>+  // implemented since they are only called for A/V-sync purposes today and
>+  // A/V-sync is not supported on iOS. However, we avoid adding error messages
>+  // the log by using these dummy implementations instead.
>+  int32_t PlayoutDelay(uint16_t& delayMS) const override;
>+
>+  // Native audio parameters stored during construction.
>+  // These methods are unique for the iOS implementation.
>+  int GetPlayoutAudioParameters(AudioParameters* params) const override;
>+  int GetRecordAudioParameters(AudioParameters* params) const override;
>+
>+  // These methods are currently not fully implemented on iOS:
>+
>+  // See audio_device_not_implemented.cc for trivial implementations.
>+  int32_t ActiveAudioLayer(
>+      AudioDeviceModule::AudioLayer& audioLayer) const override;
>+  int32_t PlayoutIsAvailable(bool& available) override;
>+  int32_t RecordingIsAvailable(bool& available) override;
>+  int16_t PlayoutDevices() override;
>+  int16_t RecordingDevices() override;
>+  int32_t PlayoutDeviceName(uint16_t index,
>+                            char name[kAdmMaxDeviceNameSize],
>+                            char guid[kAdmMaxGuidSize]) override;
>+  int32_t RecordingDeviceName(uint16_t index,
>+                              char name[kAdmMaxDeviceNameSize],
>+                              char guid[kAdmMaxGuidSize]) override;
>+  int32_t SetPlayoutDevice(uint16_t index) override;
>+  int32_t SetPlayoutDevice(
>+      AudioDeviceModule::WindowsDeviceType device) override;
>+  int32_t SetRecordingDevice(uint16_t index) override;
>+  int32_t SetRecordingDevice(
>+      AudioDeviceModule::WindowsDeviceType device) override;
>+  int32_t InitSpeaker() override;
>+  bool SpeakerIsInitialized() const override;
>+  int32_t InitMicrophone() override;
>+  bool MicrophoneIsInitialized() const override;
>+  int32_t SpeakerVolumeIsAvailable(bool& available) override;
>+  int32_t SetSpeakerVolume(uint32_t volume) override;
>+  int32_t SpeakerVolume(uint32_t& volume) const override;
>+  int32_t MaxSpeakerVolume(uint32_t& maxVolume) const override;
>+  int32_t MinSpeakerVolume(uint32_t& minVolume) const override;
>+  int32_t MicrophoneVolumeIsAvailable(bool& available) override;
>+  int32_t SetMicrophoneVolume(uint32_t volume) override;
>+  int32_t MicrophoneVolume(uint32_t& volume) const override;
>+  int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const override;
>+  int32_t MinMicrophoneVolume(uint32_t& minVolume) const override;
>+  int32_t MicrophoneMuteIsAvailable(bool& available) override;
>+  int32_t SetMicrophoneMute(bool enable) override;
>+  int32_t MicrophoneMute(bool& enabled) const override;
>+  int32_t SpeakerMuteIsAvailable(bool& available) override;
>+  int32_t SetSpeakerMute(bool enable) override;
>+  int32_t SpeakerMute(bool& enabled) const override;
>+  int32_t StereoPlayoutIsAvailable(bool& available) override;
>+  int32_t SetStereoPlayout(bool enable) override;
>+  int32_t StereoPlayout(bool& enabled) const override;
>+  int32_t StereoRecordingIsAvailable(bool& available) override;
>+  int32_t SetStereoRecording(bool enable) override;
>+  int32_t StereoRecording(bool& enabled) const override;
>+
>+  // AudioSessionObserver methods. May be called from any thread.
>+  void OnInterruptionBegin() override;
>+  void OnInterruptionEnd() override;
>+  void OnValidRouteChange() override;
>+  void OnCanPlayOrRecordChange(bool can_play_or_record) override;
>+  void OnChangedOutputVolume() override;
>+
>+  // VoiceProcessingAudioUnitObserver methods.
>+  OSStatus OnDeliverRecordedData(AudioUnitRenderActionFlags* flags,
>+                                 const AudioTimeStamp* time_stamp,
>+                                 UInt32 bus_number,
>+                                 UInt32 num_frames,
>+                                 AudioBufferList* io_data) override;
>+  OSStatus OnGetPlayoutData(AudioUnitRenderActionFlags* flags,
>+                            const AudioTimeStamp* time_stamp,
>+                            UInt32 bus_number,
>+                            UInt32 num_frames,
>+                            AudioBufferList* io_data) override;
>+
>+  // Handles messages from posts.
>+  void OnMessage(rtc::Message* msg) override;
>+
>+  bool IsInterrupted();
>+
>+ private:
>+  // Called by the relevant AudioSessionObserver methods on |thread_|.
>+  void HandleInterruptionBegin();
>+  void HandleInterruptionEnd();
>+  void HandleValidRouteChange();
>+  void HandleCanPlayOrRecordChange(bool can_play_or_record);
>+  void HandleSampleRateChange(float sample_rate);
>+  void HandlePlayoutGlitchDetected();
>+  void HandleOutputVolumeChange();
>+
>+  // Uses current |playout_parameters_| and |record_parameters_| to inform the
>+  // audio device buffer (ADB) about our internal audio parameters.
>+  void UpdateAudioDeviceBuffer();
>+
>+  // Since the preferred audio parameters are only hints to the OS, the actual
>+  // values may be different once the AVAudioSession has been activated.
>+  // This method asks for the current hardware parameters and takes actions
>+  // if they should differ from what we have asked for initially. It also
>+  // defines |playout_parameters_| and |record_parameters_|.
>+  void SetupAudioBuffersForActiveAudioSession();
>+
>+  // Creates the audio unit.
>+  bool CreateAudioUnit();
>+
>+  // Updates the audio unit state based on current state.
>+  void UpdateAudioUnit(bool can_play_or_record);
>+
>+  // Configures the audio session for WebRTC.
>+  bool ConfigureAudioSession();
>+  // Unconfigures the audio session.
>+  void UnconfigureAudioSession();
>+
>+  // Activates our audio session, creates and initializes the voice-processing
>+  // audio unit and verifies that we got the preferred native audio parameters.
>+  bool InitPlayOrRecord();
>+
>+  // Closes and deletes the voice-processing I/O unit.
>+  void ShutdownPlayOrRecord();
>+
>+  // Resets thread-checkers before a call is restarted.
>+  void PrepareForNewStart();
>+
>+  // Ensures that methods are called from the same thread as this object is
>+  // created on.
>+  rtc::ThreadChecker thread_checker_;
>+
>+  // Native I/O audio thread checker.
>+  rtc::ThreadChecker io_thread_checker_;
>+
>+  // Thread that this object is created on.
>+  rtc::Thread* thread_;
>+
>+  // Raw pointer handle provided to us in AttachAudioBuffer(). Owned by the
>+  // AudioDeviceModuleImpl class and called by AudioDeviceModule::Create().
>+  // The AudioDeviceBuffer is a member of the AudioDeviceModuleImpl instance
>+  // and therefore outlives this object.
>+  AudioDeviceBuffer* audio_device_buffer_;
>+
>+  // Contains audio parameters (sample rate, #channels, buffer size etc.) for
>+  // the playout and recording sides. These structure is set in two steps:
>+  // first, native sample rate and #channels are defined in Init(). Next, the
>+  // audio session is activated and we verify that the preferred parameters
>+  // were granted by the OS. At this stage it is also possible to add a third
>+  // component to the parameters; the native I/O buffer duration.
>+  // A RTC_CHECK will be hit if we for some reason fail to open an audio session
>+  // using the specified parameters.
>+  AudioParameters playout_parameters_;
>+  AudioParameters record_parameters_;
>+
>+  // The AudioUnit used to play and record audio.
>+  std::unique_ptr<VoiceProcessingAudioUnit> audio_unit_;
>+
>+  // FineAudioBuffer takes an AudioDeviceBuffer which delivers audio data
>+  // in chunks of 10ms. It then allows for this data to be pulled in
>+  // a finer or coarser granularity. I.e. interacting with this class instead
>+  // of directly with the AudioDeviceBuffer one can ask for any number of
>+  // audio data samples. Is also supports a similar scheme for the recording
>+  // side.
>+  // Example: native buffer size can be 128 audio frames at 16kHz sample rate.
>+  // WebRTC will provide 480 audio frames per 10ms but iOS asks for 128
>+  // in each callback (one every 8ms). This class can then ask for 128 and the
>+  // FineAudioBuffer will ask WebRTC for new data only when needed and also
>+  // cache non-utilized audio between callbacks. On the recording side, iOS
>+  // can provide audio data frames of size 128 and these are accumulated until
>+  // enough data to supply one 10ms call exists. This 10ms chunk is then sent
>+  // to WebRTC and the remaining part is stored.
>+  std::unique_ptr<FineAudioBuffer> fine_audio_buffer_;
>+
>+  // Temporary storage for recorded data. AudioUnitRender() renders into this
>+  // array as soon as a frame of the desired buffer size has been recorded.
>+  // On real iOS devices, the size will be fixed and set once. For iOS
>+  // simulators, the size can vary from callback to callback and the size
>+  // will be changed dynamically to account for this behavior.
>+  rtc::BufferT<int16_t> record_audio_buffer_;
>+
>+  // Set to 1 when recording is active and 0 otherwise.
>+  volatile int recording_;
>+
>+  // Set to 1 when playout is active and 0 otherwise.
>+  volatile int playing_;
>+
>+  // Set to true after successful call to Init(), false otherwise.
>+  bool initialized_ RTC_GUARDED_BY(thread_checker_);
>+
>+  // Set to true after successful call to InitRecording() or InitPlayout(),
>+  // false otherwise.
>+  bool audio_is_initialized_;
>+
>+  // Set to true if audio session is interrupted, false otherwise.
>+  bool is_interrupted_;
>+
>+  // Audio interruption observer instance.
>+  RTCNativeAudioSessionDelegateAdapter* audio_session_observer_
>+      RTC_GUARDED_BY(thread_checker_);
>+
>+  // Set to true if we've activated the audio session.
>+  bool has_configured_session_ RTC_GUARDED_BY(thread_checker_);
>+
>+  // Counts number of detected audio glitches on the playout side.
>+  int64_t num_detected_playout_glitches_ RTC_GUARDED_BY(thread_checker_);
>+  int64_t last_playout_time_ RTC_GUARDED_BY(io_thread_checker_);
>+
>+  // Counts number of playout callbacks per call.
>+  // The value isupdated on the native I/O thread and later read on the
>+  // creating thread (see thread_checker_) but at this stage no audio is
>+  // active. Hence, it is a "thread safe" design and no lock is needed.
>+  int64_t num_playout_callbacks_;
>+
>+  // Contains the time for when the last output volume change was detected.
>+  int64_t last_output_volume_change_time_ RTC_GUARDED_BY(thread_checker_);
>+};
>+}  // namespace ios_adm
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_DEVICE_IOS_AUDIO_DEVICE_IOS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/audio/audio_device_ios.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/audio/audio_device_ios.mm
>new file mode 100644
>index 00000000000..bb0a352f7da
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/audio/audio_device_ios.mm
>@@ -0,0 +1,1120 @@
>+/*
>+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#import <AVFoundation/AVFoundation.h>
>+#import <Foundation/Foundation.h>
>+
>+#include "audio_device_ios.h"
>+
>+#include <cmath>
>+
>+#include "api/array_view.h"
>+#include "modules/audio_device/fine_audio_buffer.h"
>+#include "rtc_base/atomicops.h"
>+#include "rtc_base/bind.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/criticalsection.h"
>+#include "rtc_base/logging.h"
>+#include "rtc_base/thread.h"
>+#include "rtc_base/thread_annotations.h"
>+#include "rtc_base/timeutils.h"
>+#include "sdk/objc/Framework/Classes/Common/helpers.h"
>+#include "system_wrappers/include/metrics.h"
>+
>+#import "WebRTC/RTCLogging.h"
>+#import "sdk/objc/Framework/Classes/Audio/RTCAudioSession+Private.h"
>+#import "sdk/objc/Framework/Classes/Audio/RTCNativeAudioSessionDelegateAdapter.h"
>+#import "sdk/objc/Framework/Headers/WebRTC/RTCAudioSession.h"
>+#import "sdk/objc/Framework/Headers/WebRTC/RTCAudioSessionConfiguration.h"
>+
>+namespace webrtc {
>+namespace ios_adm {
>+
>+#define LOGI() RTC_LOG(LS_INFO) << "AudioDeviceIOS::"
>+
>+#define LOG_AND_RETURN_IF_ERROR(error, message)    \
>+  do {                                             \
>+    OSStatus err = error;                          \
>+    if (err) {                                     \
>+      RTC_LOG(LS_ERROR) << message << ": " << err; \
>+      return false;                                \
>+    }                                              \
>+  } while (0)
>+
>+#define LOG_IF_ERROR(error, message)               \
>+  do {                                             \
>+    OSStatus err = error;                          \
>+    if (err) {                                     \
>+      RTC_LOG(LS_ERROR) << message << ": " << err; \
>+    }                                              \
>+  } while (0)
>+
>+// Hardcoded delay estimates based on real measurements.
>+// TODO(henrika): these value is not used in combination with built-in AEC.
>+// Can most likely be removed.
>+const UInt16 kFixedPlayoutDelayEstimate = 30;
>+const UInt16 kFixedRecordDelayEstimate = 30;
>+
>+enum AudioDeviceMessageType : uint32_t {
>+  kMessageTypeInterruptionBegin,
>+  kMessageTypeInterruptionEnd,
>+  kMessageTypeValidRouteChange,
>+  kMessageTypeCanPlayOrRecordChange,
>+  kMessageTypePlayoutGlitchDetected,
>+  kMessageOutputVolumeChange,
>+};
>+
>+using ios::CheckAndLogError;
>+
>+#if !defined(NDEBUG)
>+// Returns true when the code runs on a device simulator.
>+static bool DeviceIsSimulator() {
>+  return ios::GetDeviceName() == "x86_64";
>+}
>+
>+// Helper method that logs essential device information strings.
>+static void LogDeviceInfo() {
>+  RTC_LOG(LS_INFO) << "LogDeviceInfo";
>+  @autoreleasepool {
>+    RTC_LOG(LS_INFO) << " system name: " << ios::GetSystemName();
>+    RTC_LOG(LS_INFO) << " system version: " << ios::GetSystemVersionAsString();
>+    RTC_LOG(LS_INFO) << " device type: " << ios::GetDeviceType();
>+    RTC_LOG(LS_INFO) << " device name: " << ios::GetDeviceName();
>+    RTC_LOG(LS_INFO) << " process name: " << ios::GetProcessName();
>+    RTC_LOG(LS_INFO) << " process ID: " << ios::GetProcessID();
>+    RTC_LOG(LS_INFO) << " OS version: " << ios::GetOSVersionString();
>+    RTC_LOG(LS_INFO) << " processing cores: " << ios::GetProcessorCount();
>+    RTC_LOG(LS_INFO) << " low power mode: " << ios::GetLowPowerModeEnabled();
>+#if TARGET_IPHONE_SIMULATOR
>+    RTC_LOG(LS_INFO) << " TARGET_IPHONE_SIMULATOR is defined";
>+#endif
>+    RTC_LOG(LS_INFO) << " DeviceIsSimulator: " << DeviceIsSimulator();
>+  }
>+}
>+#endif  // !defined(NDEBUG)
>+
>+AudioDeviceIOS::AudioDeviceIOS()
>+    : audio_device_buffer_(nullptr),
>+      audio_unit_(nullptr),
>+      recording_(0),
>+      playing_(0),
>+      initialized_(false),
>+      audio_is_initialized_(false),
>+      is_interrupted_(false),
>+      has_configured_session_(false),
>+      num_detected_playout_glitches_(0),
>+      last_playout_time_(0),
>+      num_playout_callbacks_(0),
>+      last_output_volume_change_time_(0) {
>+  LOGI() << "ctor" << ios::GetCurrentThreadDescription();
>+  io_thread_checker_.DetachFromThread();
>+  thread_checker_.DetachFromThread();
>+  thread_ = rtc::Thread::Current();
>+
>+  audio_session_observer_ = [[RTCNativeAudioSessionDelegateAdapter alloc] initWithObserver:this];
>+}
>+
>+AudioDeviceIOS::~AudioDeviceIOS() {
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  LOGI() << "~dtor" << ios::GetCurrentThreadDescription();
>+  Terminate();
>+  audio_session_observer_ = nil;
>+}
>+
>+void AudioDeviceIOS::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
>+  LOGI() << "AttachAudioBuffer";
>+  RTC_DCHECK(audioBuffer);
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  audio_device_buffer_ = audioBuffer;
>+}
>+
>+AudioDeviceGeneric::InitStatus AudioDeviceIOS::Init() {
>+  LOGI() << "Init";
>+  io_thread_checker_.DetachFromThread();
>+  thread_checker_.DetachFromThread();
>+
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  if (initialized_) {
>+    return InitStatus::OK;
>+  }
>+#if !defined(NDEBUG)
>+  LogDeviceInfo();
>+#endif
>+  // Store the preferred sample rate and preferred number of channels already
>+  // here. They have not been set and confirmed yet since configureForWebRTC
>+  // is not called until audio is about to start. However, it makes sense to
>+  // store the parameters now and then verify at a later stage.
>+  RTCAudioSessionConfiguration* config = [RTCAudioSessionConfiguration webRTCConfiguration];
>+  playout_parameters_.reset(config.sampleRate, config.outputNumberOfChannels);
>+  record_parameters_.reset(config.sampleRate, config.inputNumberOfChannels);
>+  // Ensure that the audio device buffer (ADB) knows about the internal audio
>+  // parameters. Note that, even if we are unable to get a mono audio session,
>+  // we will always tell the I/O audio unit to do a channel format conversion
>+  // to guarantee mono on the "input side" of the audio unit.
>+  UpdateAudioDeviceBuffer();
>+  initialized_ = true;
>+  return InitStatus::OK;
>+}
>+
>+int32_t AudioDeviceIOS::Terminate() {
>+  LOGI() << "Terminate";
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  if (!initialized_) {
>+    return 0;
>+  }
>+  StopPlayout();
>+  StopRecording();
>+  initialized_ = false;
>+  return 0;
>+}
>+
>+bool AudioDeviceIOS::Initialized() const {
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  return initialized_;
>+}
>+
>+int32_t AudioDeviceIOS::InitPlayout() {
>+  LOGI() << "InitPlayout";
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  RTC_DCHECK(initialized_);
>+  RTC_DCHECK(!audio_is_initialized_);
>+  RTC_DCHECK(!playing_);
>+  if (!audio_is_initialized_) {
>+    if (!InitPlayOrRecord()) {
>+      RTC_LOG_F(LS_ERROR) << "InitPlayOrRecord failed for InitPlayout!";
>+      return -1;
>+    }
>+  }
>+  audio_is_initialized_ = true;
>+  return 0;
>+}
>+
>+bool AudioDeviceIOS::PlayoutIsInitialized() const {
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  return audio_is_initialized_;
>+}
>+
>+bool AudioDeviceIOS::RecordingIsInitialized() const {
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  return audio_is_initialized_;
>+}
>+
>+int32_t AudioDeviceIOS::InitRecording() {
>+  LOGI() << "InitRecording";
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  RTC_DCHECK(initialized_);
>+  RTC_DCHECK(!audio_is_initialized_);
>+  RTC_DCHECK(!recording_);
>+  if (!audio_is_initialized_) {
>+    if (!InitPlayOrRecord()) {
>+      RTC_LOG_F(LS_ERROR) << "InitPlayOrRecord failed for InitRecording!";
>+      return -1;
>+    }
>+  }
>+  audio_is_initialized_ = true;
>+  return 0;
>+}
>+
>+int32_t AudioDeviceIOS::StartPlayout() {
>+  LOGI() << "StartPlayout";
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  RTC_DCHECK(audio_is_initialized_);
>+  RTC_DCHECK(!playing_);
>+  RTC_DCHECK(audio_unit_);
>+  if (fine_audio_buffer_) {
>+    fine_audio_buffer_->ResetPlayout();
>+  }
>+  if (!recording_ && audio_unit_->GetState() == VoiceProcessingAudioUnit::kInitialized) {
>+    if (!audio_unit_->Start()) {
>+      RTCLogError(@"StartPlayout failed to start audio unit.");
>+      return -1;
>+    }
>+    RTC_LOG(LS_INFO) << "Voice-Processing I/O audio unit is now started";
>+  }
>+  rtc::AtomicOps::ReleaseStore(&playing_, 1);
>+  num_playout_callbacks_ = 0;
>+  num_detected_playout_glitches_ = 0;
>+  return 0;
>+}
>+
>+int32_t AudioDeviceIOS::StopPlayout() {
>+  LOGI() << "StopPlayout";
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  if (!audio_is_initialized_ || !playing_) {
>+    return 0;
>+  }
>+  if (!recording_) {
>+    ShutdownPlayOrRecord();
>+    audio_is_initialized_ = false;
>+  }
>+  rtc::AtomicOps::ReleaseStore(&playing_, 0);
>+
>+  // Derive average number of calls to OnGetPlayoutData() between detected
>+  // audio glitches and add the result to a histogram.
>+  int average_number_of_playout_callbacks_between_glitches = 100000;
>+  RTC_DCHECK_GE(num_playout_callbacks_, num_detected_playout_glitches_);
>+  if (num_detected_playout_glitches_ > 0) {
>+    average_number_of_playout_callbacks_between_glitches =
>+        num_playout_callbacks_ / num_detected_playout_glitches_;
>+  }
>+  RTC_HISTOGRAM_COUNTS_100000("WebRTC.Audio.AveragePlayoutCallbacksBetweenGlitches",
>+                              average_number_of_playout_callbacks_between_glitches);
>+  RTCLog(@"Average number of playout callbacks between glitches: %d",
>+         average_number_of_playout_callbacks_between_glitches);
>+  return 0;
>+}
>+
>+bool AudioDeviceIOS::Playing() const {
>+  return playing_;
>+}
>+
>+int32_t AudioDeviceIOS::StartRecording() {
>+  LOGI() << "StartRecording";
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  RTC_DCHECK(audio_is_initialized_);
>+  RTC_DCHECK(!recording_);
>+  RTC_DCHECK(audio_unit_);
>+  if (fine_audio_buffer_) {
>+    fine_audio_buffer_->ResetRecord();
>+  }
>+  if (!playing_ && audio_unit_->GetState() == VoiceProcessingAudioUnit::kInitialized) {
>+    if (!audio_unit_->Start()) {
>+      RTCLogError(@"StartRecording failed to start audio unit.");
>+      return -1;
>+    }
>+    RTC_LOG(LS_INFO) << "Voice-Processing I/O audio unit is now started";
>+  }
>+  rtc::AtomicOps::ReleaseStore(&recording_, 1);
>+  return 0;
>+}
>+
>+int32_t AudioDeviceIOS::StopRecording() {
>+  LOGI() << "StopRecording";
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  if (!audio_is_initialized_ || !recording_) {
>+    return 0;
>+  }
>+  if (!playing_) {
>+    ShutdownPlayOrRecord();
>+    audio_is_initialized_ = false;
>+  }
>+  rtc::AtomicOps::ReleaseStore(&recording_, 0);
>+  return 0;
>+}
>+
>+bool AudioDeviceIOS::Recording() const {
>+  return recording_;
>+}
>+
>+int32_t AudioDeviceIOS::PlayoutDelay(uint16_t& delayMS) const {
>+  delayMS = kFixedPlayoutDelayEstimate;
>+  return 0;
>+}
>+
>+int AudioDeviceIOS::GetPlayoutAudioParameters(AudioParameters* params) const {
>+  LOGI() << "GetPlayoutAudioParameters";
>+  RTC_DCHECK(playout_parameters_.is_valid());
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  *params = playout_parameters_;
>+  return 0;
>+}
>+
>+int AudioDeviceIOS::GetRecordAudioParameters(AudioParameters* params) const {
>+  LOGI() << "GetRecordAudioParameters";
>+  RTC_DCHECK(record_parameters_.is_valid());
>+  RTC_DCHECK(thread_checker_.CalledOnValidThread());
>+  *params = record_parameters_;
>+  return 0;
>+}
>+
>+void AudioDeviceIOS::OnInterruptionBegin() {
>+  RTC_DCHECK(thread_);
>+  LOGI() << "OnInterruptionBegin";
>+  thread_->Post(RTC_FROM_HERE, this, kMessageTypeInterruptionBegin);
>+}
>+
>+void AudioDeviceIOS::OnInterruptionEnd() {
>+  RTC_DCHECK(thread_);
>+  LOGI() << "OnInterruptionEnd";
>+  thread_->Post(RTC_FROM_HERE, this, kMessageTypeInterruptionEnd);
>+}
>+
>+void AudioDeviceIOS::OnValidRouteChange() {
>+  RTC_DCHECK(thread_);
>+  thread_->Post(RTC_FROM_HERE, this, kMessageTypeValidRouteChange);
>+}
>+
>+void AudioDeviceIOS::OnCanPlayOrRecordChange(bool can_play_or_record) {
>+  RTC_DCHECK(thread_);
>+  thread_->Post(RTC_FROM_HERE,
>+                this,
>+                kMessageTypeCanPlayOrRecordChange,
>+                new rtc::TypedMessageData<bool>(can_play_or_record));
>+}
>+
>+void AudioDeviceIOS::OnChangedOutputVolume() {
>+  RTC_DCHECK(thread_);
>+  thread_->Post(RTC_FROM_HERE, this, kMessageOutputVolumeChange);
>+}
>+
>+OSStatus AudioDeviceIOS::OnDeliverRecordedData(AudioUnitRenderActionFlags* flags,
>+                                               const AudioTimeStamp* time_stamp,
>+                                               UInt32 bus_number,
>+                                               UInt32 num_frames,
>+                                               AudioBufferList* /* io_data */) {
>+  RTC_DCHECK_RUN_ON(&io_thread_checker_);
>+  OSStatus result = noErr;
>+  // Simply return if recording is not enabled.
>+  if (!rtc::AtomicOps::AcquireLoad(&recording_)) return result;
>+
>+  // Set the size of our own audio buffer and clear it first to avoid copying
>+  // in combination with potential reallocations.
>+  // On real iOS devices, the size will only be set once (at first callback).
>+  record_audio_buffer_.Clear();
>+  record_audio_buffer_.SetSize(num_frames);
>+
>+  // Allocate AudioBuffers to be used as storage for the received audio.
>+  // The AudioBufferList structure works as a placeholder for the
>+  // AudioBuffer structure, which holds a pointer to the actual data buffer
>+  // in |record_audio_buffer_|. Recorded audio will be rendered into this memory
>+  // at each input callback when calling AudioUnitRender().
>+  AudioBufferList audio_buffer_list;
>+  audio_buffer_list.mNumberBuffers = 1;
>+  AudioBuffer* audio_buffer = &audio_buffer_list.mBuffers[0];
>+  audio_buffer->mNumberChannels = record_parameters_.channels();
>+  audio_buffer->mDataByteSize =
>+      record_audio_buffer_.size() * VoiceProcessingAudioUnit::kBytesPerSample;
>+  audio_buffer->mData = reinterpret_cast<int8_t*>(record_audio_buffer_.data());
>+
>+  // Obtain the recorded audio samples by initiating a rendering cycle.
>+  // Since it happens on the input bus, the |io_data| parameter is a reference
>+  // to the preallocated audio buffer list that the audio unit renders into.
>+  // We can make the audio unit provide a buffer instead in io_data, but we
>+  // currently just use our own.
>+  // TODO(henrika): should error handling be improved?
>+  result = audio_unit_->Render(flags, time_stamp, bus_number, num_frames, &audio_buffer_list);
>+  if (result != noErr) {
>+    RTCLogError(@"Failed to render audio.");
>+    return result;
>+  }
>+
>+  // Get a pointer to the recorded audio and send it to the WebRTC ADB.
>+  // Use the FineAudioBuffer instance to convert between native buffer size
>+  // and the 10ms buffer size used by WebRTC.
>+  fine_audio_buffer_->DeliverRecordedData(record_audio_buffer_, kFixedRecordDelayEstimate);
>+  return noErr;
>+}
>+
>+OSStatus AudioDeviceIOS::OnGetPlayoutData(AudioUnitRenderActionFlags* flags,
>+                                          const AudioTimeStamp* time_stamp,
>+                                          UInt32 bus_number,
>+                                          UInt32 num_frames,
>+                                          AudioBufferList* io_data) {
>+  RTC_DCHECK_RUN_ON(&io_thread_checker_);
>+  // Verify 16-bit, noninterleaved mono PCM signal format.
>+  RTC_DCHECK_EQ(1, io_data->mNumberBuffers);
>+  AudioBuffer* audio_buffer = &io_data->mBuffers[0];
>+  RTC_DCHECK_EQ(1, audio_buffer->mNumberChannels);
>+
>+  // Produce silence and give audio unit a hint about it if playout is not
>+  // activated.
>+  if (!rtc::AtomicOps::AcquireLoad(&playing_)) {
>+    const size_t size_in_bytes = audio_buffer->mDataByteSize;
>+    RTC_CHECK_EQ(size_in_bytes / VoiceProcessingAudioUnit::kBytesPerSample, num_frames);
>+    *flags |= kAudioUnitRenderAction_OutputIsSilence;
>+    memset(static_cast<int8_t*>(audio_buffer->mData), 0, size_in_bytes);
>+    return noErr;
>+  }
>+
>+  // Measure time since last call to OnGetPlayoutData() and see if it is larger
>+  // than a well defined threshold which depends on the current IO buffer size.
>+  // If so, we have an indication of a glitch in the output audio since the
>+  // core audio layer will most likely run dry in this state.
>+  ++num_playout_callbacks_;
>+  const int64_t now_time = rtc::TimeMillis();
>+  if (time_stamp->mSampleTime != num_frames) {
>+    const int64_t delta_time = now_time - last_playout_time_;
>+    const int glitch_threshold = 1.6 * playout_parameters_.GetBufferSizeInMilliseconds();
>+    if (delta_time > glitch_threshold) {
>+      RTCLogWarning(@"Possible playout audio glitch detected.\n"
>+                     "  Time since last OnGetPlayoutData was %lld ms.\n",
>+                    delta_time);
>+      // Exclude extreme delta values since they do most likely not correspond
>+      // to a real glitch. Instead, the most probable cause is that a headset
>+      // has been plugged in or out. There are more direct ways to detect
>+      // audio device changes (see HandleValidRouteChange()) but experiments
>+      // show that using it leads to more complex implementations.
>+      // TODO(henrika): more tests might be needed to come up with an even
>+      // better upper limit.
>+      if (glitch_threshold < 120 && delta_time > 120) {
>+        RTCLog(@"Glitch warning is ignored. Probably caused by device switch.");
>+      } else {
>+        thread_->Post(RTC_FROM_HERE, this, kMessageTypePlayoutGlitchDetected);
>+      }
>+    }
>+  }
>+  last_playout_time_ = now_time;
>+
>+  // Read decoded 16-bit PCM samples from WebRTC (using a size that matches
>+  // the native I/O audio unit) and copy the result to the audio buffer in the
>+  // |io_data| destination.
>+  fine_audio_buffer_->GetPlayoutData(
>+      rtc::ArrayView<int16_t>(static_cast<int16_t*>(audio_buffer->mData), num_frames),
>+      kFixedPlayoutDelayEstimate);
>+  return noErr;
>+}
>+
>+void AudioDeviceIOS::OnMessage(rtc::Message* msg) {
>+  switch (msg->message_id) {
>+    case kMessageTypeInterruptionBegin:
>+      HandleInterruptionBegin();
>+      break;
>+    case kMessageTypeInterruptionEnd:
>+      HandleInterruptionEnd();
>+      break;
>+    case kMessageTypeValidRouteChange:
>+      HandleValidRouteChange();
>+      break;
>+    case kMessageTypeCanPlayOrRecordChange: {
>+      rtc::TypedMessageData<bool>* data = static_cast<rtc::TypedMessageData<bool>*>(msg->pdata);
>+      HandleCanPlayOrRecordChange(data->data());
>+      delete data;
>+      break;
>+    }
>+    case kMessageTypePlayoutGlitchDetected:
>+      HandlePlayoutGlitchDetected();
>+      break;
>+    case kMessageOutputVolumeChange:
>+      HandleOutputVolumeChange();
>+      break;
>+  }
>+}
>+
>+void AudioDeviceIOS::HandleInterruptionBegin() {
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  RTCLog(@"Interruption begin. IsInterrupted changed from %d to 1.", is_interrupted_);
>+  if (audio_unit_ && audio_unit_->GetState() == VoiceProcessingAudioUnit::kStarted) {
>+    RTCLog(@"Stopping the audio unit due to interruption begin.");
>+    if (!audio_unit_->Stop()) {
>+      RTCLogError(@"Failed to stop the audio unit for interruption begin.");
>+    } else {
>+      PrepareForNewStart();
>+    }
>+  }
>+  is_interrupted_ = true;
>+}
>+
>+void AudioDeviceIOS::HandleInterruptionEnd() {
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  RTCLog(@"Interruption ended. IsInterrupted changed from %d to 0. "
>+          "Updating audio unit state.",
>+         is_interrupted_);
>+  is_interrupted_ = false;
>+  UpdateAudioUnit([RTCAudioSession sharedInstance].canPlayOrRecord);
>+}
>+
>+void AudioDeviceIOS::HandleValidRouteChange() {
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  RTCAudioSession* session = [RTCAudioSession sharedInstance];
>+  RTCLog(@"%@", session);
>+  HandleSampleRateChange(session.sampleRate);
>+}
>+
>+void AudioDeviceIOS::HandleCanPlayOrRecordChange(bool can_play_or_record) {
>+  RTCLog(@"Handling CanPlayOrRecord change to: %d", can_play_or_record);
>+  UpdateAudioUnit(can_play_or_record);
>+}
>+
>+void AudioDeviceIOS::HandleSampleRateChange(float sample_rate) {
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  RTCLog(@"Handling sample rate change to %f.", sample_rate);
>+
>+  // Don't do anything if we're interrupted.
>+  if (is_interrupted_) {
>+    RTCLog(@"Ignoring sample rate change to %f due to interruption.", sample_rate);
>+    return;
>+  }
>+
>+  // If we don't have an audio unit yet, or the audio unit is uninitialized,
>+  // there is no work to do.
>+  if (!audio_unit_ || audio_unit_->GetState() < VoiceProcessingAudioUnit::kInitialized) {
>+    return;
>+  }
>+
>+  // The audio unit is already initialized or started.
>+  // Check to see if the sample rate or buffer size has changed.
>+  RTCAudioSession* session = [RTCAudioSession sharedInstance];
>+  const double session_sample_rate = session.sampleRate;
>+  const NSTimeInterval session_buffer_duration = session.IOBufferDuration;
>+  const size_t session_frames_per_buffer =
>+      static_cast<size_t>(session_sample_rate * session_buffer_duration + .5);
>+  const double current_sample_rate = playout_parameters_.sample_rate();
>+  const size_t current_frames_per_buffer = playout_parameters_.frames_per_buffer();
>+  RTCLog(@"Handling playout sample rate change to: %f\n"
>+          "  Session sample rate: %f frames_per_buffer: %lu\n"
>+          "  ADM sample rate: %f frames_per_buffer: %lu",
>+         sample_rate,
>+         session_sample_rate,
>+         (unsigned long)session_frames_per_buffer,
>+         current_sample_rate,
>+         (unsigned long)current_frames_per_buffer);
>+
>+  // Sample rate and buffer size are the same, no work to do.
>+  if (std::abs(current_sample_rate - session_sample_rate) <= DBL_EPSILON &&
>+      current_frames_per_buffer == session_frames_per_buffer) {
>+    RTCLog(@"Ignoring sample rate change since audio parameters are intact.");
>+    return;
>+  }
>+
>+  // Extra sanity check to ensure that the new sample rate is valid.
>+  if (session_sample_rate <= 0.0) {
>+    RTCLogError(@"Sample rate is invalid: %f", session_sample_rate);
>+    return;
>+  }
>+
>+  // We need to adjust our format and buffer sizes.
>+  // The stream format is about to be changed and it requires that we first
>+  // stop and uninitialize the audio unit to deallocate its resources.
>+  RTCLog(@"Stopping and uninitializing audio unit to adjust buffers.");
>+  bool restart_audio_unit = false;
>+  if (audio_unit_->GetState() == VoiceProcessingAudioUnit::kStarted) {
>+    audio_unit_->Stop();
>+    restart_audio_unit = true;
>+    PrepareForNewStart();
>+  }
>+  if (audio_unit_->GetState() == VoiceProcessingAudioUnit::kInitialized) {
>+    audio_unit_->Uninitialize();
>+  }
>+
>+  // Allocate new buffers given the new stream format.
>+  SetupAudioBuffersForActiveAudioSession();
>+
>+  // Initialize the audio unit again with the new sample rate.
>+  RTC_DCHECK_EQ(playout_parameters_.sample_rate(), session_sample_rate);
>+  if (!audio_unit_->Initialize(session_sample_rate)) {
>+    RTCLogError(@"Failed to initialize the audio unit with sample rate: %f", session_sample_rate);
>+    return;
>+  }
>+
>+  // Restart the audio unit if it was already running.
>+  if (restart_audio_unit && !audio_unit_->Start()) {
>+    RTCLogError(@"Failed to start audio unit with sample rate: %f", session_sample_rate);
>+    return;
>+  }
>+  RTCLog(@"Successfully handled sample rate change.");
>+}
>+
>+void AudioDeviceIOS::HandlePlayoutGlitchDetected() {
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  // Don't update metrics if we're interrupted since a "glitch" is expected
>+  // in this state.
>+  if (is_interrupted_) {
>+    RTCLog(@"Ignoring audio glitch due to interruption.");
>+    return;
>+  }
>+  // Avoid doing glitch detection for two seconds after a volume change
>+  // has been detected to reduce the risk of false alarm.
>+  if (last_output_volume_change_time_ > 0 &&
>+      rtc::TimeSince(last_output_volume_change_time_) < 2000) {
>+    RTCLog(@"Ignoring audio glitch due to recent output volume change.");
>+    return;
>+  }
>+  num_detected_playout_glitches_++;
>+  RTCLog(@"Number of detected playout glitches: %lld", num_detected_playout_glitches_);
>+
>+  int64_t glitch_count = num_detected_playout_glitches_;
>+  dispatch_async(dispatch_get_main_queue(), ^{
>+    RTCAudioSession* session = [RTCAudioSession sharedInstance];
>+    [session notifyDidDetectPlayoutGlitch:glitch_count];
>+  });
>+}
>+
>+void AudioDeviceIOS::HandleOutputVolumeChange() {
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  RTCLog(@"Output volume change detected.");
>+  // Store time of this detection so it can be used to defer detection of
>+  // glitches too close in time to this event.
>+  last_output_volume_change_time_ = rtc::TimeMillis();
>+}
>+
>+void AudioDeviceIOS::UpdateAudioDeviceBuffer() {
>+  LOGI() << "UpdateAudioDevicebuffer";
>+  // AttachAudioBuffer() is called at construction by the main class but check
>+  // just in case.
>+  RTC_DCHECK(audio_device_buffer_) << "AttachAudioBuffer must be called first";
>+  RTC_DCHECK_GT(playout_parameters_.sample_rate(), 0);
>+  RTC_DCHECK_GT(record_parameters_.sample_rate(), 0);
>+  RTC_DCHECK_EQ(playout_parameters_.channels(), 1);
>+  RTC_DCHECK_EQ(record_parameters_.channels(), 1);
>+  // Inform the audio device buffer (ADB) about the new audio format.
>+  audio_device_buffer_->SetPlayoutSampleRate(playout_parameters_.sample_rate());
>+  audio_device_buffer_->SetPlayoutChannels(playout_parameters_.channels());
>+  audio_device_buffer_->SetRecordingSampleRate(record_parameters_.sample_rate());
>+  audio_device_buffer_->SetRecordingChannels(record_parameters_.channels());
>+}
>+
>+void AudioDeviceIOS::SetupAudioBuffersForActiveAudioSession() {
>+  LOGI() << "SetupAudioBuffersForActiveAudioSession";
>+  // Verify the current values once the audio session has been activated.
>+  RTCAudioSession* session = [RTCAudioSession sharedInstance];
>+  double sample_rate = session.sampleRate;
>+  NSTimeInterval io_buffer_duration = session.IOBufferDuration;
>+  RTCLog(@"%@", session);
>+
>+  // Log a warning message for the case when we are unable to set the preferred
>+  // hardware sample rate but continue and use the non-ideal sample rate after
>+  // reinitializing the audio parameters. Most BT headsets only support 8kHz or
>+  // 16kHz.
>+  RTCAudioSessionConfiguration* webRTCConfig = [RTCAudioSessionConfiguration webRTCConfiguration];
>+  if (sample_rate != webRTCConfig.sampleRate) {
>+    RTC_LOG(LS_WARNING) << "Unable to set the preferred sample rate";
>+  }
>+
>+  // Crash reports indicates that it can happen in rare cases that the reported
>+  // sample rate is less than or equal to zero. If that happens and if a valid
>+  // sample rate has already been set during initialization, the best guess we
>+  // can do is to reuse the current sample rate.
>+  if (sample_rate <= DBL_EPSILON && playout_parameters_.sample_rate() > 0) {
>+    RTCLogError(@"Reported rate is invalid: %f. "
>+                 "Using %d as sample rate instead.",
>+                sample_rate, playout_parameters_.sample_rate());
>+    sample_rate = playout_parameters_.sample_rate();
>+  }
>+
>+  // At this stage, we also know the exact IO buffer duration and can add
>+  // that info to the existing audio parameters where it is converted into
>+  // number of audio frames.
>+  // Example: IO buffer size = 0.008 seconds <=> 128 audio frames at 16kHz.
>+  // Hence, 128 is the size we expect to see in upcoming render callbacks.
>+  playout_parameters_.reset(sample_rate, playout_parameters_.channels(), io_buffer_duration);
>+  RTC_DCHECK(playout_parameters_.is_complete());
>+  record_parameters_.reset(sample_rate, record_parameters_.channels(), io_buffer_duration);
>+  RTC_DCHECK(record_parameters_.is_complete());
>+  RTC_LOG(LS_INFO) << " frames per I/O buffer: " << playout_parameters_.frames_per_buffer();
>+  RTC_LOG(LS_INFO) << " bytes per I/O buffer: " << playout_parameters_.GetBytesPerBuffer();
>+  RTC_DCHECK_EQ(playout_parameters_.GetBytesPerBuffer(), record_parameters_.GetBytesPerBuffer());
>+
>+  // Update the ADB parameters since the sample rate might have changed.
>+  UpdateAudioDeviceBuffer();
>+
>+  // Create a modified audio buffer class which allows us to ask for,
>+  // or deliver, any number of samples (and not only multiple of 10ms) to match
>+  // the native audio unit buffer size.
>+  RTC_DCHECK(audio_device_buffer_);
>+  fine_audio_buffer_.reset(new FineAudioBuffer(audio_device_buffer_));
>+}
>+
>+bool AudioDeviceIOS::CreateAudioUnit() {
>+  RTC_DCHECK(!audio_unit_);
>+
>+  audio_unit_.reset(new VoiceProcessingAudioUnit(this));
>+  if (!audio_unit_->Init()) {
>+    audio_unit_.reset();
>+    return false;
>+  }
>+
>+  return true;
>+}
>+
>+void AudioDeviceIOS::UpdateAudioUnit(bool can_play_or_record) {
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  RTCLog(@"Updating audio unit state. CanPlayOrRecord=%d IsInterrupted=%d",
>+         can_play_or_record,
>+         is_interrupted_);
>+
>+  if (is_interrupted_) {
>+    RTCLog(@"Ignoring audio unit update due to interruption.");
>+    return;
>+  }
>+
>+  // If we're not initialized we don't need to do anything. Audio unit will
>+  // be initialized on initialization.
>+  if (!audio_is_initialized_) return;
>+
>+  // If we're initialized, we must have an audio unit.
>+  RTC_DCHECK(audio_unit_);
>+
>+  bool should_initialize_audio_unit = false;
>+  bool should_uninitialize_audio_unit = false;
>+  bool should_start_audio_unit = false;
>+  bool should_stop_audio_unit = false;
>+
>+  switch (audio_unit_->GetState()) {
>+    case VoiceProcessingAudioUnit::kInitRequired:
>+      RTCLog(@"VPAU state: InitRequired");
>+      RTC_NOTREACHED();
>+      break;
>+    case VoiceProcessingAudioUnit::kUninitialized:
>+      RTCLog(@"VPAU state: Uninitialized");
>+      should_initialize_audio_unit = can_play_or_record;
>+      should_start_audio_unit = should_initialize_audio_unit && (playing_ || recording_);
>+      break;
>+    case VoiceProcessingAudioUnit::kInitialized:
>+      RTCLog(@"VPAU state: Initialized");
>+      should_start_audio_unit = can_play_or_record && (playing_ || recording_);
>+      should_uninitialize_audio_unit = !can_play_or_record;
>+      break;
>+    case VoiceProcessingAudioUnit::kStarted:
>+      RTCLog(@"VPAU state: Started");
>+      RTC_DCHECK(playing_ || recording_);
>+      should_stop_audio_unit = !can_play_or_record;
>+      should_uninitialize_audio_unit = should_stop_audio_unit;
>+      break;
>+  }
>+
>+  if (should_initialize_audio_unit) {
>+    RTCLog(@"Initializing audio unit for UpdateAudioUnit");
>+    ConfigureAudioSession();
>+    SetupAudioBuffersForActiveAudioSession();
>+    if (!audio_unit_->Initialize(playout_parameters_.sample_rate())) {
>+      RTCLogError(@"Failed to initialize audio unit.");
>+      return;
>+    }
>+  }
>+
>+  if (should_start_audio_unit) {
>+    RTCLog(@"Starting audio unit for UpdateAudioUnit");
>+    // Log session settings before trying to start audio streaming.
>+    RTCAudioSession* session = [RTCAudioSession sharedInstance];
>+    RTCLog(@"%@", session);
>+    if (!audio_unit_->Start()) {
>+      RTCLogError(@"Failed to start audio unit.");
>+      return;
>+    }
>+  }
>+
>+  if (should_stop_audio_unit) {
>+    RTCLog(@"Stopping audio unit for UpdateAudioUnit");
>+    if (!audio_unit_->Stop()) {
>+      RTCLogError(@"Failed to stop audio unit.");
>+      return;
>+    }
>+  }
>+
>+  if (should_uninitialize_audio_unit) {
>+    RTCLog(@"Uninitializing audio unit for UpdateAudioUnit");
>+    audio_unit_->Uninitialize();
>+    UnconfigureAudioSession();
>+  }
>+}
>+
>+bool AudioDeviceIOS::ConfigureAudioSession() {
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  RTCLog(@"Configuring audio session.");
>+  if (has_configured_session_) {
>+    RTCLogWarning(@"Audio session already configured.");
>+    return false;
>+  }
>+  RTCAudioSession* session = [RTCAudioSession sharedInstance];
>+  [session lockForConfiguration];
>+  bool success = [session configureWebRTCSession:nil];
>+  [session unlockForConfiguration];
>+  if (success) {
>+    has_configured_session_ = true;
>+    RTCLog(@"Configured audio session.");
>+  } else {
>+    RTCLog(@"Failed to configure audio session.");
>+  }
>+  return success;
>+}
>+
>+void AudioDeviceIOS::UnconfigureAudioSession() {
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  RTCLog(@"Unconfiguring audio session.");
>+  if (!has_configured_session_) {
>+    RTCLogWarning(@"Audio session already unconfigured.");
>+    return;
>+  }
>+  RTCAudioSession* session = [RTCAudioSession sharedInstance];
>+  [session lockForConfiguration];
>+  [session unconfigureWebRTCSession:nil];
>+  [session endWebRTCSession:nil];
>+  [session unlockForConfiguration];
>+  has_configured_session_ = false;
>+  RTCLog(@"Unconfigured audio session.");
>+}
>+
>+bool AudioDeviceIOS::InitPlayOrRecord() {
>+  LOGI() << "InitPlayOrRecord";
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+
>+  // There should be no audio unit at this point.
>+  if (!CreateAudioUnit()) {
>+    return false;
>+  }
>+
>+  RTCAudioSession* session = [RTCAudioSession sharedInstance];
>+  // Subscribe to audio session events.
>+  [session pushDelegate:audio_session_observer_];
>+  is_interrupted_ = session.isInterrupted ? true : false;
>+
>+  // Lock the session to make configuration changes.
>+  [session lockForConfiguration];
>+  NSError* error = nil;
>+  if (![session beginWebRTCSession:&error]) {
>+    [session unlockForConfiguration];
>+    RTCLogError(@"Failed to begin WebRTC session: %@", error.localizedDescription);
>+    audio_unit_.reset();
>+    return false;
>+  }
>+
>+  // If we are ready to play or record, and if the audio session can be
>+  // configured, then initialize the audio unit.
>+  if (session.canPlayOrRecord) {
>+    if (!ConfigureAudioSession()) {
>+      // One possible reason for failure is if an attempt was made to use the
>+      // audio session during or after a Media Services failure.
>+      // See AVAudioSessionErrorCodeMediaServicesFailed for details.
>+      [session unlockForConfiguration];
>+      audio_unit_.reset();
>+      return false;
>+    }
>+    SetupAudioBuffersForActiveAudioSession();
>+    audio_unit_->Initialize(playout_parameters_.sample_rate());
>+  }
>+
>+  // Release the lock.
>+  [session unlockForConfiguration];
>+  return true;
>+}
>+
>+void AudioDeviceIOS::ShutdownPlayOrRecord() {
>+  LOGI() << "ShutdownPlayOrRecord";
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+
>+  // Stop the audio unit to prevent any additional audio callbacks.
>+  audio_unit_->Stop();
>+
>+  // Close and delete the voice-processing I/O unit.
>+  audio_unit_.reset();
>+
>+  // Detach thread checker for the AURemoteIO::IOThread to ensure that the
>+  // next session uses a fresh thread id.
>+  io_thread_checker_.DetachFromThread();
>+
>+  // Remove audio session notification observers.
>+  RTCAudioSession* session = [RTCAudioSession sharedInstance];
>+  [session removeDelegate:audio_session_observer_];
>+
>+  // All I/O should be stopped or paused prior to deactivating the audio
>+  // session, hence we deactivate as last action.
>+  UnconfigureAudioSession();
>+}
>+
>+void AudioDeviceIOS::PrepareForNewStart() {
>+  LOGI() << "PrepareForNewStart";
>+  // The audio unit has been stopped and preparations are needed for an upcoming
>+  // restart. It will result in audio callbacks from a new native I/O thread
>+  // which means that we must detach thread checkers here to be prepared for an
>+  // upcoming new audio stream.
>+  io_thread_checker_.DetachFromThread();
>+  // The audio device buffer must also be informed about the interrupted
>+  // state so it can detach its thread checkers as well.
>+  if (audio_device_buffer_) {
>+    audio_device_buffer_->NativeAudioPlayoutInterrupted();
>+    audio_device_buffer_->NativeAudioRecordingInterrupted();
>+  }
>+}
>+
>+bool AudioDeviceIOS::IsInterrupted() {
>+  return is_interrupted_;
>+}
>+
>+#pragma mark - Not Implemented
>+
>+int32_t AudioDeviceIOS::ActiveAudioLayer(AudioDeviceModule::AudioLayer& audioLayer) const {
>+  audioLayer = AudioDeviceModule::kPlatformDefaultAudio;
>+  return 0;
>+}
>+
>+int16_t AudioDeviceIOS::PlayoutDevices() {
>+  // TODO(henrika): improve.
>+  RTC_LOG_F(LS_WARNING) << "Not implemented";
>+  return (int16_t)1;
>+}
>+
>+int16_t AudioDeviceIOS::RecordingDevices() {
>+  // TODO(henrika): improve.
>+  RTC_LOG_F(LS_WARNING) << "Not implemented";
>+  return (int16_t)1;
>+}
>+
>+int32_t AudioDeviceIOS::InitSpeaker() {
>+  return 0;
>+}
>+
>+bool AudioDeviceIOS::SpeakerIsInitialized() const {
>+  return true;
>+}
>+
>+int32_t AudioDeviceIOS::SpeakerVolumeIsAvailable(bool& available) {
>+  available = false;
>+  return 0;
>+}
>+
>+int32_t AudioDeviceIOS::SetSpeakerVolume(uint32_t volume) {
>+  RTC_NOTREACHED() << "Not implemented";
>+  return -1;
>+}
>+
>+int32_t AudioDeviceIOS::SpeakerVolume(uint32_t& volume) const {
>+  RTC_NOTREACHED() << "Not implemented";
>+  return -1;
>+}
>+
>+int32_t AudioDeviceIOS::MaxSpeakerVolume(uint32_t& maxVolume) const {
>+  RTC_NOTREACHED() << "Not implemented";
>+  return -1;
>+}
>+
>+int32_t AudioDeviceIOS::MinSpeakerVolume(uint32_t& minVolume) const {
>+  RTC_NOTREACHED() << "Not implemented";
>+  return -1;
>+}
>+
>+int32_t AudioDeviceIOS::SpeakerMuteIsAvailable(bool& available) {
>+  available = false;
>+  return 0;
>+}
>+
>+int32_t AudioDeviceIOS::SetSpeakerMute(bool enable) {
>+  RTC_NOTREACHED() << "Not implemented";
>+  return -1;
>+}
>+
>+int32_t AudioDeviceIOS::SpeakerMute(bool& enabled) const {
>+  RTC_NOTREACHED() << "Not implemented";
>+  return -1;
>+}
>+
>+int32_t AudioDeviceIOS::SetPlayoutDevice(uint16_t index) {
>+  RTC_LOG_F(LS_WARNING) << "Not implemented";
>+  return 0;
>+}
>+
>+int32_t AudioDeviceIOS::SetPlayoutDevice(AudioDeviceModule::WindowsDeviceType) {
>+  RTC_NOTREACHED() << "Not implemented";
>+  return -1;
>+}
>+
>+int32_t AudioDeviceIOS::InitMicrophone() {
>+  return 0;
>+}
>+
>+bool AudioDeviceIOS::MicrophoneIsInitialized() const {
>+  return true;
>+}
>+
>+int32_t AudioDeviceIOS::MicrophoneMuteIsAvailable(bool& available) {
>+  available = false;
>+  return 0;
>+}
>+
>+int32_t AudioDeviceIOS::SetMicrophoneMute(bool enable) {
>+  RTC_NOTREACHED() << "Not implemented";
>+  return -1;
>+}
>+
>+int32_t AudioDeviceIOS::MicrophoneMute(bool& enabled) const {
>+  RTC_NOTREACHED() << "Not implemented";
>+  return -1;
>+}
>+
>+int32_t AudioDeviceIOS::StereoRecordingIsAvailable(bool& available) {
>+  available = false;
>+  return 0;
>+}
>+
>+int32_t AudioDeviceIOS::SetStereoRecording(bool enable) {
>+  RTC_LOG_F(LS_WARNING) << "Not implemented";
>+  return -1;
>+}
>+
>+int32_t AudioDeviceIOS::StereoRecording(bool& enabled) const {
>+  enabled = false;
>+  return 0;
>+}
>+
>+int32_t AudioDeviceIOS::StereoPlayoutIsAvailable(bool& available) {
>+  available = false;
>+  return 0;
>+}
>+
>+int32_t AudioDeviceIOS::SetStereoPlayout(bool enable) {
>+  RTC_LOG_F(LS_WARNING) << "Not implemented";
>+  return -1;
>+}
>+
>+int32_t AudioDeviceIOS::StereoPlayout(bool& enabled) const {
>+  enabled = false;
>+  return 0;
>+}
>+
>+int32_t AudioDeviceIOS::MicrophoneVolumeIsAvailable(bool& available) {
>+  available = false;
>+  return 0;
>+}
>+
>+int32_t AudioDeviceIOS::SetMicrophoneVolume(uint32_t volume) {
>+  RTC_NOTREACHED() << "Not implemented";
>+  return -1;
>+}
>+
>+int32_t AudioDeviceIOS::MicrophoneVolume(uint32_t& volume) const {
>+  RTC_NOTREACHED() << "Not implemented";
>+  return -1;
>+}
>+
>+int32_t AudioDeviceIOS::MaxMicrophoneVolume(uint32_t& maxVolume) const {
>+  RTC_NOTREACHED() << "Not implemented";
>+  return -1;
>+}
>+
>+int32_t AudioDeviceIOS::MinMicrophoneVolume(uint32_t& minVolume) const {
>+  RTC_NOTREACHED() << "Not implemented";
>+  return -1;
>+}
>+
>+int32_t AudioDeviceIOS::PlayoutDeviceName(uint16_t index,
>+                                          char name[kAdmMaxDeviceNameSize],
>+                                          char guid[kAdmMaxGuidSize]) {
>+  RTC_NOTREACHED() << "Not implemented";
>+  return -1;
>+}
>+
>+int32_t AudioDeviceIOS::RecordingDeviceName(uint16_t index,
>+                                            char name[kAdmMaxDeviceNameSize],
>+                                            char guid[kAdmMaxGuidSize]) {
>+  RTC_NOTREACHED() << "Not implemented";
>+  return -1;
>+}
>+
>+int32_t AudioDeviceIOS::SetRecordingDevice(uint16_t index) {
>+  RTC_LOG_F(LS_WARNING) << "Not implemented";
>+  return 0;
>+}
>+
>+int32_t AudioDeviceIOS::SetRecordingDevice(AudioDeviceModule::WindowsDeviceType) {
>+  RTC_NOTREACHED() << "Not implemented";
>+  return -1;
>+}
>+
>+int32_t AudioDeviceIOS::PlayoutIsAvailable(bool& available) {
>+  available = true;
>+  return 0;
>+}
>+
>+int32_t AudioDeviceIOS::RecordingIsAvailable(bool& available) {
>+  available = true;
>+  return 0;
>+}
>+
>+}  // namespace ios_adm
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/audio/audio_device_module_ios.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/audio/audio_device_module_ios.h
>new file mode 100644
>index 00000000000..8e1ace44fd6
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/audio/audio_device_module_ios.h
>@@ -0,0 +1,139 @@
>+/*
>+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef SDK_IOS_NATIVE_API_AUDIO_DEVICE_MODULE_AUDIO_DEVICE_IOS_H_
>+#define SDK_IOS_NATIVE_API_AUDIO_DEVICE_MODULE_AUDIO_DEVICE_IOS_H_
>+
>+#include <memory>
>+
>+#include "audio_device_ios.h"
>+
>+#include "modules/audio_device/audio_device_buffer.h"
>+#include "modules/audio_device/include/audio_device.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/criticalsection.h"
>+
>+namespace webrtc {
>+
>+class AudioDeviceGeneric;
>+
>+namespace ios_adm {
>+
>+class AudioDeviceModuleIOS : public AudioDeviceModule {
>+ public:
>+  int32_t AttachAudioBuffer();
>+
>+  AudioDeviceModuleIOS();
>+  ~AudioDeviceModuleIOS() override;
>+
>+  // Retrieve the currently utilized audio layer
>+  int32_t ActiveAudioLayer(AudioLayer* audioLayer) const override;
>+
>+  // Full-duplex transportation of PCM audio
>+  int32_t RegisterAudioCallback(AudioTransport* audioCallback) override;
>+
>+  // Main initializaton and termination
>+  int32_t Init() override;
>+  int32_t Terminate() override;
>+  bool Initialized() const override;
>+
>+  // Device enumeration
>+  int16_t PlayoutDevices() override;
>+  int16_t RecordingDevices() override;
>+  int32_t PlayoutDeviceName(uint16_t index,
>+                            char name[kAdmMaxDeviceNameSize],
>+                            char guid[kAdmMaxGuidSize]) override;
>+  int32_t RecordingDeviceName(uint16_t index,
>+                              char name[kAdmMaxDeviceNameSize],
>+                              char guid[kAdmMaxGuidSize]) override;
>+
>+  // Device selection
>+  int32_t SetPlayoutDevice(uint16_t index) override;
>+  int32_t SetPlayoutDevice(WindowsDeviceType device) override;
>+  int32_t SetRecordingDevice(uint16_t index) override;
>+  int32_t SetRecordingDevice(WindowsDeviceType device) override;
>+
>+  // Audio transport initialization
>+  int32_t PlayoutIsAvailable(bool* available) override;
>+  int32_t InitPlayout() override;
>+  bool PlayoutIsInitialized() const override;
>+  int32_t RecordingIsAvailable(bool* available) override;
>+  int32_t InitRecording() override;
>+  bool RecordingIsInitialized() const override;
>+
>+  // Audio transport control
>+  int32_t StartPlayout() override;
>+  int32_t StopPlayout() override;
>+  bool Playing() const override;
>+  int32_t StartRecording() override;
>+  int32_t StopRecording() override;
>+  bool Recording() const override;
>+
>+  // Audio mixer initialization
>+  int32_t InitSpeaker() override;
>+  bool SpeakerIsInitialized() const override;
>+  int32_t InitMicrophone() override;
>+  bool MicrophoneIsInitialized() const override;
>+
>+  // Speaker volume controls
>+  int32_t SpeakerVolumeIsAvailable(bool* available) override;
>+  int32_t SetSpeakerVolume(uint32_t volume) override;
>+  int32_t SpeakerVolume(uint32_t* volume) const override;
>+  int32_t MaxSpeakerVolume(uint32_t* maxVolume) const override;
>+  int32_t MinSpeakerVolume(uint32_t* minVolume) const override;
>+
>+  // Microphone volume controls
>+  int32_t MicrophoneVolumeIsAvailable(bool* available) override;
>+  int32_t SetMicrophoneVolume(uint32_t volume) override;
>+  int32_t MicrophoneVolume(uint32_t* volume) const override;
>+  int32_t MaxMicrophoneVolume(uint32_t* maxVolume) const override;
>+  int32_t MinMicrophoneVolume(uint32_t* minVolume) const override;
>+
>+  // Speaker mute control
>+  int32_t SpeakerMuteIsAvailable(bool* available) override;
>+  int32_t SetSpeakerMute(bool enable) override;
>+  int32_t SpeakerMute(bool* enabled) const override;
>+
>+  // Microphone mute control
>+  int32_t MicrophoneMuteIsAvailable(bool* available) override;
>+  int32_t SetMicrophoneMute(bool enable) override;
>+  int32_t MicrophoneMute(bool* enabled) const override;
>+
>+  // Stereo support
>+  int32_t StereoPlayoutIsAvailable(bool* available) const override;
>+  int32_t SetStereoPlayout(bool enable) override;
>+  int32_t StereoPlayout(bool* enabled) const override;
>+  int32_t StereoRecordingIsAvailable(bool* available) const override;
>+  int32_t SetStereoRecording(bool enable) override;
>+  int32_t StereoRecording(bool* enabled) const override;
>+
>+  // Delay information and control
>+  int32_t PlayoutDelay(uint16_t* delayMS) const override;
>+
>+  bool BuiltInAECIsAvailable() const override;
>+  int32_t EnableBuiltInAEC(bool enable) override;
>+  bool BuiltInAGCIsAvailable() const override;
>+  int32_t EnableBuiltInAGC(bool enable) override;
>+  bool BuiltInNSIsAvailable() const override;
>+  int32_t EnableBuiltInNS(bool enable) override;
>+
>+#if defined(WEBRTC_IOS)
>+  int GetPlayoutAudioParameters(AudioParameters* params) const override;
>+  int GetRecordAudioParameters(AudioParameters* params) const override;
>+#endif  // WEBRTC_IOS
>+ private:
>+  bool initialized_ = false;
>+  std::unique_ptr<AudioDeviceIOS> audio_device_;
>+  std::unique_ptr<AudioDeviceBuffer> audio_device_buffer_;
>+};
>+}  // namespace ios_adm
>+}  // namespace webrtc
>+
>+#endif  // SDK_IOS_NATIVE_API_AUDIO_DEVICE_MODULE_AUDIO_DEVICE_IOS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/audio/audio_device_module_ios.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/audio/audio_device_module_ios.mm
>new file mode 100644
>index 00000000000..3e918a25d49
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/audio/audio_device_module_ios.mm
>@@ -0,0 +1,673 @@
>+/*
>+ *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "audio_device_module_ios.h"
>+
>+#include "modules/audio_device/audio_device_config.h"
>+#include "modules/audio_device/audio_device_generic.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/logging.h"
>+#include "rtc_base/refcount.h"
>+#include "rtc_base/refcountedobject.h"
>+#include "system_wrappers/include/metrics.h"
>+
>+#if defined(WEBRTC_IOS)
>+#include "audio_device_ios.h"
>+#endif
>+
>+#define CHECKinitialized_() \
>+  {                         \
>+    if (!initialized_) {    \
>+      return -1;            \
>+    };                      \
>+  }
>+
>+#define CHECKinitialized__BOOL() \
>+  {                              \
>+    if (!initialized_) {         \
>+      return false;              \
>+    };                           \
>+  }
>+
>+
>+namespace webrtc {
>+namespace ios_adm {
>+
>+  AudioDeviceModuleIOS::AudioDeviceModuleIOS() {
>+    RTC_LOG(INFO) << "current platform is IOS";
>+    RTC_LOG(INFO) << "iPhone Audio APIs will be utilized.";
>+  }
>+
>+  int32_t AudioDeviceModuleIOS::AttachAudioBuffer() {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    audio_device_->AttachAudioBuffer(audio_device_buffer_.get());
>+    return 0;
>+  }
>+
>+  AudioDeviceModuleIOS::~AudioDeviceModuleIOS() {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+  }
>+
>+  int32_t AudioDeviceModuleIOS::ActiveAudioLayer(AudioLayer* audioLayer) const {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    AudioLayer activeAudio;
>+    if (audio_device_->ActiveAudioLayer(activeAudio) == -1) {
>+      return -1;
>+    }
>+    *audioLayer = activeAudio;
>+    return 0;
>+  }
>+
>+  int32_t AudioDeviceModuleIOS::Init() {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    if (initialized_)
>+      return 0;
>+
>+    audio_device_buffer_.reset(new webrtc::AudioDeviceBuffer());
>+    audio_device_.reset(new ios_adm::AudioDeviceIOS());
>+    RTC_CHECK(audio_device_);
>+
>+    this->AttachAudioBuffer();
>+
>+    AudioDeviceGeneric::InitStatus status = audio_device_->Init();
>+    RTC_HISTOGRAM_ENUMERATION(
>+        "WebRTC.Audio.InitializationResult", static_cast<int>(status),
>+        static_cast<int>(AudioDeviceGeneric::InitStatus::NUM_STATUSES));
>+    if (status != AudioDeviceGeneric::InitStatus::OK) {
>+      RTC_LOG(LS_ERROR) << "Audio device initialization failed.";
>+      return -1;
>+    }
>+    initialized_ = true;
>+    return 0;
>+  }
>+
>+  int32_t AudioDeviceModuleIOS::Terminate() {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    if (!initialized_)
>+      return 0;
>+    if (audio_device_->Terminate() == -1) {
>+      return -1;
>+    }
>+    initialized_ = false;
>+    return 0;
>+  }
>+
>+  bool AudioDeviceModuleIOS::Initialized() const {
>+    RTC_LOG(INFO) << __FUNCTION__ << ": " << initialized_;
>+    return initialized_;
>+  }
>+
>+  int32_t AudioDeviceModuleIOS::InitSpeaker() {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    CHECKinitialized_();
>+    return audio_device_->InitSpeaker();
>+  }
>+
>+  int32_t AudioDeviceModuleIOS::InitMicrophone() {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    CHECKinitialized_();
>+    return audio_device_->InitMicrophone();
>+  }
>+
>+  int32_t AudioDeviceModuleIOS::SpeakerVolumeIsAvailable(bool* available) {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    CHECKinitialized_();
>+    bool isAvailable = false;
>+    if (audio_device_->SpeakerVolumeIsAvailable(isAvailable) == -1) {
>+      return -1;
>+    }
>+    *available = isAvailable;
>+    RTC_LOG(INFO) << "output: " << isAvailable;
>+    return 0;
>+  }
>+
>+  int32_t AudioDeviceModuleIOS::SetSpeakerVolume(uint32_t volume) {
>+    RTC_LOG(INFO) << __FUNCTION__ << "(" << volume << ")";
>+    CHECKinitialized_();
>+    return audio_device_->SetSpeakerVolume(volume);
>+  }
>+
>+  int32_t AudioDeviceModuleIOS::SpeakerVolume(uint32_t* volume) const {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    CHECKinitialized_();
>+    uint32_t level = 0;
>+    if (audio_device_->SpeakerVolume(level) == -1) {
>+      return -1;
>+    }
>+    *volume = level;
>+    RTC_LOG(INFO) << "output: " << *volume;
>+    return 0;
>+  }
>+
>+  bool AudioDeviceModuleIOS::SpeakerIsInitialized() const {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    CHECKinitialized__BOOL();
>+    bool isInitialized = audio_device_->SpeakerIsInitialized();
>+    RTC_LOG(INFO) << "output: " << isInitialized;
>+    return isInitialized;
>+  }
>+
>+  bool AudioDeviceModuleIOS::MicrophoneIsInitialized() const {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    CHECKinitialized__BOOL();
>+    bool isInitialized = audio_device_->MicrophoneIsInitialized();
>+    RTC_LOG(INFO) << "output: " << isInitialized;
>+    return isInitialized;
>+  }
>+
>+  int32_t AudioDeviceModuleIOS::MaxSpeakerVolume(uint32_t* maxVolume) const {
>+    CHECKinitialized_();
>+    uint32_t maxVol = 0;
>+    if (audio_device_->MaxSpeakerVolume(maxVol) == -1) {
>+      return -1;
>+    }
>+    *maxVolume = maxVol;
>+    return 0;
>+  }
>+
>+  int32_t AudioDeviceModuleIOS::MinSpeakerVolume(uint32_t* minVolume) const {
>+    CHECKinitialized_();
>+    uint32_t minVol = 0;
>+    if (audio_device_->MinSpeakerVolume(minVol) == -1) {
>+      return -1;
>+    }
>+    *minVolume = minVol;
>+    return 0;
>+  }
>+
>+  int32_t AudioDeviceModuleIOS::SpeakerMuteIsAvailable(bool* available) {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    CHECKinitialized_();
>+    bool isAvailable = false;
>+    if (audio_device_->SpeakerMuteIsAvailable(isAvailable) == -1) {
>+      return -1;
>+    }
>+    *available = isAvailable;
>+    RTC_LOG(INFO) << "output: " << isAvailable;
>+    return 0;
>+  }
>+
>+  int32_t AudioDeviceModuleIOS::SetSpeakerMute(bool enable) {
>+    RTC_LOG(INFO) << __FUNCTION__ << "(" << enable << ")";
>+    CHECKinitialized_();
>+    return audio_device_->SetSpeakerMute(enable);
>+  }
>+
>+  int32_t AudioDeviceModuleIOS::SpeakerMute(bool* enabled) const {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    CHECKinitialized_();
>+    bool muted = false;
>+    if (audio_device_->SpeakerMute(muted) == -1) {
>+      return -1;
>+    }
>+    *enabled = muted;
>+    RTC_LOG(INFO) << "output: " << muted;
>+    return 0;
>+  }
>+
>+  int32_t AudioDeviceModuleIOS::MicrophoneMuteIsAvailable(bool* available) {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    CHECKinitialized_();
>+    bool isAvailable = false;
>+    if (audio_device_->MicrophoneMuteIsAvailable(isAvailable) == -1) {
>+      return -1;
>+    }
>+    *available = isAvailable;
>+    RTC_LOG(INFO) << "output: " << isAvailable;
>+    return 0;
>+  }
>+
>+  int32_t AudioDeviceModuleIOS::SetMicrophoneMute(bool enable) {
>+    RTC_LOG(INFO) << __FUNCTION__ << "(" << enable << ")";
>+    CHECKinitialized_();
>+    return (audio_device_->SetMicrophoneMute(enable));
>+  }
>+
>+  int32_t AudioDeviceModuleIOS::MicrophoneMute(bool* enabled) const {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    CHECKinitialized_();
>+    bool muted = false;
>+    if (audio_device_->MicrophoneMute(muted) == -1) {
>+      return -1;
>+    }
>+    *enabled = muted;
>+    RTC_LOG(INFO) << "output: " << muted;
>+    return 0;
>+  }
>+
>+  int32_t AudioDeviceModuleIOS::MicrophoneVolumeIsAvailable(bool* available) {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    CHECKinitialized_();
>+    bool isAvailable = false;
>+    if (audio_device_->MicrophoneVolumeIsAvailable(isAvailable) == -1) {
>+      return -1;
>+    }
>+    *available = isAvailable;
>+    RTC_LOG(INFO) << "output: " << isAvailable;
>+    return 0;
>+  }
>+
>+  int32_t AudioDeviceModuleIOS::SetMicrophoneVolume(uint32_t volume) {
>+    RTC_LOG(INFO) << __FUNCTION__ << "(" << volume << ")";
>+    CHECKinitialized_();
>+    return (audio_device_->SetMicrophoneVolume(volume));
>+  }
>+
>+  int32_t AudioDeviceModuleIOS::MicrophoneVolume(uint32_t* volume) const {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    CHECKinitialized_();
>+    uint32_t level = 0;
>+    if (audio_device_->MicrophoneVolume(level) == -1) {
>+      return -1;
>+    }
>+    *volume = level;
>+    RTC_LOG(INFO) << "output: " << *volume;
>+    return 0;
>+  }
>+
>+  int32_t AudioDeviceModuleIOS::StereoRecordingIsAvailable(
>+      bool* available) const {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    CHECKinitialized_();
>+    bool isAvailable = false;
>+    if (audio_device_->StereoRecordingIsAvailable(isAvailable) == -1) {
>+      return -1;
>+    }
>+    *available = isAvailable;
>+    RTC_LOG(INFO) << "output: " << isAvailable;
>+    return 0;
>+  }
>+
>+  int32_t AudioDeviceModuleIOS::SetStereoRecording(bool enable) {
>+    RTC_LOG(INFO) << __FUNCTION__ << "(" << enable << ")";
>+    CHECKinitialized_();
>+    if (audio_device_->RecordingIsInitialized()) {
>+      RTC_LOG(WARNING) << "recording in stereo is not supported";
>+      return -1;
>+    }
>+    if (audio_device_->SetStereoRecording(enable) == -1) {
>+      RTC_LOG(WARNING) << "failed to change stereo recording";
>+      return -1;
>+    }
>+    int8_t nChannels(1);
>+    if (enable) {
>+      nChannels = 2;
>+    }
>+    audio_device_buffer_.get()->SetRecordingChannels(nChannels);
>+    return 0;
>+  }
>+
>+  int32_t AudioDeviceModuleIOS::StereoRecording(bool* enabled) const {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    CHECKinitialized_();
>+    bool stereo = false;
>+    if (audio_device_->StereoRecording(stereo) == -1) {
>+      return -1;
>+    }
>+    *enabled = stereo;
>+    RTC_LOG(INFO) << "output: " << stereo;
>+    return 0;
>+  }
>+
>+  int32_t AudioDeviceModuleIOS::StereoPlayoutIsAvailable(bool* available) const {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    CHECKinitialized_();
>+    bool isAvailable = false;
>+    if (audio_device_->StereoPlayoutIsAvailable(isAvailable) == -1) {
>+      return -1;
>+    }
>+    *available = isAvailable;
>+    RTC_LOG(INFO) << "output: " << isAvailable;
>+    return 0;
>+  }
>+
>+  int32_t AudioDeviceModuleIOS::SetStereoPlayout(bool enable) {
>+    RTC_LOG(INFO) << __FUNCTION__ << "(" << enable << ")";
>+    CHECKinitialized_();
>+    if (audio_device_->PlayoutIsInitialized()) {
>+      RTC_LOG(LERROR)
>+      << "unable to set stereo mode while playing side is initialized";
>+      return -1;
>+    }
>+    if (audio_device_->SetStereoPlayout(enable)) {
>+      RTC_LOG(WARNING) << "stereo playout is not supported";
>+      return -1;
>+    }
>+    int8_t nChannels(1);
>+    if (enable) {
>+      nChannels = 2;
>+    }
>+    audio_device_buffer_.get()->SetPlayoutChannels(nChannels);
>+    return 0;
>+  }
>+
>+  int32_t AudioDeviceModuleIOS::StereoPlayout(bool* enabled) const {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    CHECKinitialized_();
>+    bool stereo = false;
>+    if (audio_device_->StereoPlayout(stereo) == -1) {
>+      return -1;
>+    }
>+    *enabled = stereo;
>+    RTC_LOG(INFO) << "output: " << stereo;
>+    return 0;
>+  }
>+
>+  int32_t AudioDeviceModuleIOS::PlayoutIsAvailable(bool* available) {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    CHECKinitialized_();
>+    bool isAvailable = false;
>+    if (audio_device_->PlayoutIsAvailable(isAvailable) == -1) {
>+      return -1;
>+    }
>+    *available = isAvailable;
>+    RTC_LOG(INFO) << "output: " << isAvailable;
>+    return 0;
>+  }
>+
>+  int32_t AudioDeviceModuleIOS::RecordingIsAvailable(bool* available) {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    CHECKinitialized_();
>+    bool isAvailable = false;
>+    if (audio_device_->RecordingIsAvailable(isAvailable) == -1) {
>+      return -1;
>+    }
>+    *available = isAvailable;
>+    RTC_LOG(INFO) << "output: " << isAvailable;
>+    return 0;
>+  }
>+
>+  int32_t AudioDeviceModuleIOS::MaxMicrophoneVolume(uint32_t* maxVolume) const {
>+    CHECKinitialized_();
>+    uint32_t maxVol(0);
>+    if (audio_device_->MaxMicrophoneVolume(maxVol) == -1) {
>+      return -1;
>+    }
>+    *maxVolume = maxVol;
>+    return 0;
>+  }
>+
>+  int32_t AudioDeviceModuleIOS::MinMicrophoneVolume(uint32_t* minVolume) const {
>+    CHECKinitialized_();
>+    uint32_t minVol(0);
>+    if (audio_device_->MinMicrophoneVolume(minVol) == -1) {
>+      return -1;
>+    }
>+    *minVolume = minVol;
>+    return 0;
>+  }
>+
>+  int16_t AudioDeviceModuleIOS::PlayoutDevices() {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    CHECKinitialized_();
>+    uint16_t nPlayoutDevices = audio_device_->PlayoutDevices();
>+    RTC_LOG(INFO) << "output: " << nPlayoutDevices;
>+    return (int16_t)(nPlayoutDevices);
>+  }
>+
>+  int32_t AudioDeviceModuleIOS::SetPlayoutDevice(uint16_t index) {
>+    RTC_LOG(INFO) << __FUNCTION__ << "(" << index << ")";
>+    CHECKinitialized_();
>+    return audio_device_->SetPlayoutDevice(index);
>+  }
>+
>+  int32_t AudioDeviceModuleIOS::SetPlayoutDevice(WindowsDeviceType device) {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    CHECKinitialized_();
>+    return audio_device_->SetPlayoutDevice(device);
>+  }
>+
>+  int32_t AudioDeviceModuleIOS::PlayoutDeviceName(
>+      uint16_t index,
>+      char name[kAdmMaxDeviceNameSize],
>+      char guid[kAdmMaxGuidSize]) {
>+    RTC_LOG(INFO) << __FUNCTION__ << "(" << index << ", ...)";
>+    CHECKinitialized_();
>+    if (name == NULL) {
>+      return -1;
>+    }
>+    if (audio_device_->PlayoutDeviceName(index, name, guid) == -1) {
>+      return -1;
>+    }
>+    if (name != NULL) {
>+      RTC_LOG(INFO) << "output: name = " << name;
>+    }
>+    if (guid != NULL) {
>+      RTC_LOG(INFO) << "output: guid = " << guid;
>+    }
>+    return 0;
>+  }
>+
>+  int32_t AudioDeviceModuleIOS::RecordingDeviceName(
>+      uint16_t index,
>+      char name[kAdmMaxDeviceNameSize],
>+      char guid[kAdmMaxGuidSize]) {
>+    RTC_LOG(INFO) << __FUNCTION__ << "(" << index << ", ...)";
>+    CHECKinitialized_();
>+    if (name == NULL) {
>+      return -1;
>+    }
>+    if (audio_device_->RecordingDeviceName(index, name, guid) == -1) {
>+      return -1;
>+    }
>+    if (name != NULL) {
>+      RTC_LOG(INFO) << "output: name = " << name;
>+    }
>+    if (guid != NULL) {
>+      RTC_LOG(INFO) << "output: guid = " << guid;
>+    }
>+    return 0;
>+  }
>+
>+  int16_t AudioDeviceModuleIOS::RecordingDevices() {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    CHECKinitialized_();
>+    uint16_t nRecordingDevices = audio_device_->RecordingDevices();
>+    RTC_LOG(INFO) << "output: " << nRecordingDevices;
>+    return (int16_t)nRecordingDevices;
>+  }
>+
>+  int32_t AudioDeviceModuleIOS::SetRecordingDevice(uint16_t index) {
>+    RTC_LOG(INFO) << __FUNCTION__ << "(" << index << ")";
>+    CHECKinitialized_();
>+    return audio_device_->SetRecordingDevice(index);
>+  }
>+
>+  int32_t AudioDeviceModuleIOS::SetRecordingDevice(WindowsDeviceType device) {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    CHECKinitialized_();
>+    return audio_device_->SetRecordingDevice(device);
>+  }
>+
>+  int32_t AudioDeviceModuleIOS::InitPlayout() {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    CHECKinitialized_();
>+    if (PlayoutIsInitialized()) {
>+      return 0;
>+    }
>+    int32_t result = audio_device_->InitPlayout();
>+    RTC_LOG(INFO) << "output: " << result;
>+    RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.InitPlayoutSuccess",
>+                          static_cast<int>(result == 0));
>+    return result;
>+  }
>+
>+  int32_t AudioDeviceModuleIOS::InitRecording() {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    CHECKinitialized_();
>+    if (RecordingIsInitialized()) {
>+      return 0;
>+    }
>+    int32_t result = audio_device_->InitRecording();
>+    RTC_LOG(INFO) << "output: " << result;
>+    RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.InitRecordingSuccess",
>+                          static_cast<int>(result == 0));
>+    return result;
>+  }
>+
>+  bool AudioDeviceModuleIOS::PlayoutIsInitialized() const {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    CHECKinitialized__BOOL();
>+    return audio_device_->PlayoutIsInitialized();
>+  }
>+
>+  bool AudioDeviceModuleIOS::RecordingIsInitialized() const {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    CHECKinitialized__BOOL();
>+    return audio_device_->RecordingIsInitialized();
>+  }
>+
>+  int32_t AudioDeviceModuleIOS::StartPlayout() {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    CHECKinitialized_();
>+    if (Playing()) {
>+      return 0;
>+    }
>+    audio_device_buffer_.get()->StartPlayout();
>+    int32_t result = audio_device_->StartPlayout();
>+    RTC_LOG(INFO) << "output: " << result;
>+    RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StartPlayoutSuccess",
>+                          static_cast<int>(result == 0));
>+    return result;
>+  }
>+
>+  int32_t AudioDeviceModuleIOS::StopPlayout() {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    CHECKinitialized_();
>+    int32_t result = audio_device_->StopPlayout();
>+    audio_device_buffer_.get()->StopPlayout();
>+    RTC_LOG(INFO) << "output: " << result;
>+    RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StopPlayoutSuccess",
>+                          static_cast<int>(result == 0));
>+    return result;
>+  }
>+
>+  bool AudioDeviceModuleIOS::Playing() const {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    CHECKinitialized__BOOL();
>+    return audio_device_->Playing();
>+  }
>+
>+  int32_t AudioDeviceModuleIOS::StartRecording() {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    CHECKinitialized_();
>+    if (Recording()) {
>+      return 0;
>+    }
>+    audio_device_buffer_.get()->StartRecording();
>+    int32_t result = audio_device_->StartRecording();
>+    RTC_LOG(INFO) << "output: " << result;
>+    RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StartRecordingSuccess",
>+                          static_cast<int>(result == 0));
>+    return result;
>+  }
>+
>+  int32_t AudioDeviceModuleIOS::StopRecording() {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    CHECKinitialized_();
>+    int32_t result = audio_device_->StopRecording();
>+    audio_device_buffer_.get()->StopRecording();
>+    RTC_LOG(INFO) << "output: " << result;
>+    RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.StopRecordingSuccess",
>+                          static_cast<int>(result == 0));
>+    return result;
>+  }
>+
>+  bool AudioDeviceModuleIOS::Recording() const {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    CHECKinitialized__BOOL();
>+    return audio_device_->Recording();
>+  }
>+
>+  int32_t AudioDeviceModuleIOS::RegisterAudioCallback(
>+      AudioTransport* audioCallback) {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    return audio_device_buffer_.get()->RegisterAudioCallback(audioCallback);
>+  }
>+
>+  int32_t AudioDeviceModuleIOS::PlayoutDelay(uint16_t* delayMS) const {
>+    CHECKinitialized_();
>+    uint16_t delay = 0;
>+    if (audio_device_->PlayoutDelay(delay) == -1) {
>+      RTC_LOG(LERROR) << "failed to retrieve the playout delay";
>+      return -1;
>+    }
>+    *delayMS = delay;
>+    return 0;
>+  }
>+
>+  bool AudioDeviceModuleIOS::BuiltInAECIsAvailable() const {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    CHECKinitialized__BOOL();
>+    bool isAvailable = audio_device_->BuiltInAECIsAvailable();
>+    RTC_LOG(INFO) << "output: " << isAvailable;
>+    return isAvailable;
>+  }
>+
>+  int32_t AudioDeviceModuleIOS::EnableBuiltInAEC(bool enable) {
>+    RTC_LOG(INFO) << __FUNCTION__ << "(" << enable << ")";
>+    CHECKinitialized_();
>+    int32_t ok = audio_device_->EnableBuiltInAEC(enable);
>+    RTC_LOG(INFO) << "output: " << ok;
>+    return ok;
>+  }
>+
>+  bool AudioDeviceModuleIOS::BuiltInAGCIsAvailable() const {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    CHECKinitialized__BOOL();
>+    bool isAvailable = audio_device_->BuiltInAGCIsAvailable();
>+    RTC_LOG(INFO) << "output: " << isAvailable;
>+    return isAvailable;
>+  }
>+
>+  int32_t AudioDeviceModuleIOS::EnableBuiltInAGC(bool enable) {
>+    RTC_LOG(INFO) << __FUNCTION__ << "(" << enable << ")";
>+    CHECKinitialized_();
>+    int32_t ok = audio_device_->EnableBuiltInAGC(enable);
>+    RTC_LOG(INFO) << "output: " << ok;
>+    return ok;
>+  }
>+
>+  bool AudioDeviceModuleIOS::BuiltInNSIsAvailable() const {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    CHECKinitialized__BOOL();
>+    bool isAvailable = audio_device_->BuiltInNSIsAvailable();
>+    RTC_LOG(INFO) << "output: " << isAvailable;
>+    return isAvailable;
>+  }
>+
>+  int32_t AudioDeviceModuleIOS::EnableBuiltInNS(bool enable) {
>+    RTC_LOG(INFO) << __FUNCTION__ << "(" << enable << ")";
>+    CHECKinitialized_();
>+    int32_t ok = audio_device_->EnableBuiltInNS(enable);
>+    RTC_LOG(INFO) << "output: " << ok;
>+    return ok;
>+  }
>+
>+#if defined(WEBRTC_IOS)
>+  int AudioDeviceModuleIOS::GetPlayoutAudioParameters(
>+      AudioParameters* params) const {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    int r = audio_device_->GetPlayoutAudioParameters(params);
>+    RTC_LOG(INFO) << "output: " << r;
>+    return r;
>+  }
>+
>+  int AudioDeviceModuleIOS::GetRecordAudioParameters(
>+      AudioParameters* params) const {
>+    RTC_LOG(INFO) << __FUNCTION__;
>+    int r = audio_device_->GetRecordAudioParameters(params);
>+    RTC_LOG(INFO) << "output: " << r;
>+    return r;
>+  }
>+#endif  // WEBRTC_IOS
>+
>+}
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/audio/audio_session_observer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/audio/audio_session_observer.h
>new file mode 100644
>index 00000000000..13a979bd4eb
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/audio/audio_session_observer.h
>@@ -0,0 +1,42 @@
>+/*
>+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_DEVICE_IOS_AUDIO_SESSION_OBSERVER_H_
>+#define MODULES_AUDIO_DEVICE_IOS_AUDIO_SESSION_OBSERVER_H_
>+
>+#include "rtc_base/asyncinvoker.h"
>+#include "rtc_base/thread.h"
>+
>+namespace webrtc {
>+
>+// Observer interface for listening to AVAudioSession events.
>+class AudioSessionObserver {
>+ public:
>+  // Called when audio session interruption begins.
>+  virtual void OnInterruptionBegin() = 0;
>+
>+  // Called when audio session interruption ends.
>+  virtual void OnInterruptionEnd() = 0;
>+
>+  // Called when audio route changes.
>+  virtual void OnValidRouteChange() = 0;
>+
>+  // Called when the ability to play or record changes.
>+  virtual void OnCanPlayOrRecordChange(bool can_play_or_record) = 0;
>+
>+  virtual void OnChangedOutputVolume() = 0;
>+
>+ protected:
>+  virtual ~AudioSessionObserver() {}
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  //  MODULES_AUDIO_DEVICE_IOS_AUDIO_SESSION_OBSERVER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/audio/voice_processing_audio_unit.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/audio/voice_processing_audio_unit.h
>new file mode 100644
>index 00000000000..ae5e1a90d0a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/audio/voice_processing_audio_unit.h
>@@ -0,0 +1,139 @@
>+/*
>+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef MODULES_AUDIO_DEVICE_IOS_VOICE_PROCESSING_AUDIO_UNIT_H_
>+#define MODULES_AUDIO_DEVICE_IOS_VOICE_PROCESSING_AUDIO_UNIT_H_
>+
>+#include <AudioUnit/AudioUnit.h>
>+
>+namespace webrtc {
>+namespace ios_adm {
>+
>+class VoiceProcessingAudioUnitObserver {
>+ public:
>+  // Callback function called on a real-time priority I/O thread from the audio
>+  // unit. This method is used to signal that recorded audio is available.
>+  virtual OSStatus OnDeliverRecordedData(AudioUnitRenderActionFlags* flags,
>+                                         const AudioTimeStamp* time_stamp,
>+                                         UInt32 bus_number,
>+                                         UInt32 num_frames,
>+                                         AudioBufferList* io_data) = 0;
>+
>+  // Callback function called on a real-time priority I/O thread from the audio
>+  // unit. This method is used to provide audio samples to the audio unit.
>+  virtual OSStatus OnGetPlayoutData(AudioUnitRenderActionFlags* io_action_flags,
>+                                    const AudioTimeStamp* time_stamp,
>+                                    UInt32 bus_number,
>+                                    UInt32 num_frames,
>+                                    AudioBufferList* io_data) = 0;
>+
>+ protected:
>+  ~VoiceProcessingAudioUnitObserver() {}
>+};
>+
>+// Convenience class to abstract away the management of a Voice Processing
>+// I/O Audio Unit. The Voice Processing I/O unit has the same characteristics
>+// as the Remote I/O unit (supports full duplex low-latency audio input and
>+// output) and adds AEC for for two-way duplex communication. It also adds AGC,
>+// adjustment of voice-processing quality, and muting. Hence, ideal for
>+// VoIP applications.
>+class VoiceProcessingAudioUnit {
>+ public:
>+  explicit VoiceProcessingAudioUnit(VoiceProcessingAudioUnitObserver* observer);
>+  ~VoiceProcessingAudioUnit();
>+
>+  // TODO(tkchin): enum for state and state checking.
>+  enum State : int32_t {
>+    // Init() should be called.
>+    kInitRequired,
>+    // Audio unit created but not initialized.
>+    kUninitialized,
>+    // Initialized but not started. Equivalent to stopped.
>+    kInitialized,
>+    // Initialized and started.
>+    kStarted,
>+  };
>+
>+  // Number of bytes per audio sample for 16-bit signed integer representation.
>+  static const UInt32 kBytesPerSample;
>+
>+  // Initializes this class by creating the underlying audio unit instance.
>+  // Creates a Voice-Processing I/O unit and configures it for full-duplex
>+  // audio. The selected stream format is selected to avoid internal resampling
>+  // and to match the 10ms callback rate for WebRTC as well as possible.
>+  // Does not intialize the audio unit.
>+  bool Init();
>+
>+  VoiceProcessingAudioUnit::State GetState() const;
>+
>+  // Initializes the underlying audio unit with the given sample rate.
>+  bool Initialize(Float64 sample_rate);
>+
>+  // Starts the underlying audio unit.
>+  bool Start();
>+
>+  // Stops the underlying audio unit.
>+  bool Stop();
>+
>+  // Uninitializes the underlying audio unit.
>+  bool Uninitialize();
>+
>+  // Calls render on the underlying audio unit.
>+  OSStatus Render(AudioUnitRenderActionFlags* flags,
>+                  const AudioTimeStamp* time_stamp,
>+                  UInt32 output_bus_number,
>+                  UInt32 num_frames,
>+                  AudioBufferList* io_data);
>+
>+ private:
>+  // The C API used to set callbacks requires static functions. When these are
>+  // called, they will invoke the relevant instance method by casting
>+  // in_ref_con to VoiceProcessingAudioUnit*.
>+  static OSStatus OnGetPlayoutData(void* in_ref_con,
>+                                   AudioUnitRenderActionFlags* flags,
>+                                   const AudioTimeStamp* time_stamp,
>+                                   UInt32 bus_number,
>+                                   UInt32 num_frames,
>+                                   AudioBufferList* io_data);
>+  static OSStatus OnDeliverRecordedData(void* in_ref_con,
>+                                        AudioUnitRenderActionFlags* flags,
>+                                        const AudioTimeStamp* time_stamp,
>+                                        UInt32 bus_number,
>+                                        UInt32 num_frames,
>+                                        AudioBufferList* io_data);
>+
>+  // Notifies observer that samples are needed for playback.
>+  OSStatus NotifyGetPlayoutData(AudioUnitRenderActionFlags* flags,
>+                                const AudioTimeStamp* time_stamp,
>+                                UInt32 bus_number,
>+                                UInt32 num_frames,
>+                                AudioBufferList* io_data);
>+  // Notifies observer that recorded samples are available for render.
>+  OSStatus NotifyDeliverRecordedData(AudioUnitRenderActionFlags* flags,
>+                                     const AudioTimeStamp* time_stamp,
>+                                     UInt32 bus_number,
>+                                     UInt32 num_frames,
>+                                     AudioBufferList* io_data);
>+
>+  // Returns the predetermined format with a specific sample rate. See
>+  // implementation file for details on format.
>+  AudioStreamBasicDescription GetFormat(Float64 sample_rate) const;
>+
>+  // Deletes the underlying audio unit.
>+  void DisposeAudioUnit();
>+
>+  VoiceProcessingAudioUnitObserver* observer_;
>+  AudioUnit vpio_unit_;
>+  VoiceProcessingAudioUnit::State state_;
>+};
>+}  // namespace ios_adm
>+}  // namespace webrtc
>+
>+#endif  // MODULES_AUDIO_DEVICE_IOS_VOICE_PROCESSING_AUDIO_UNIT_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/audio/voice_processing_audio_unit.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/audio/voice_processing_audio_unit.mm
>new file mode 100644
>index 00000000000..eddb1397c18
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/audio/voice_processing_audio_unit.mm
>@@ -0,0 +1,470 @@
>+/*
>+ *  Copyright 2016 The WebRTC Project Authors. All rights reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#import "voice_processing_audio_unit.h"
>+
>+#include "rtc_base/checks.h"
>+#include "rtc_base/system/fallthrough.h"
>+#include "system_wrappers/include/metrics.h"
>+
>+#import "WebRTC/RTCLogging.h"
>+#import "sdk/objc/Framework/Headers/WebRTC/RTCAudioSessionConfiguration.h"
>+
>+#if !defined(NDEBUG)
>+static void LogStreamDescription(AudioStreamBasicDescription description) {
>+  char formatIdString[5];
>+  UInt32 formatId = CFSwapInt32HostToBig(description.mFormatID);
>+  bcopy(&formatId, formatIdString, 4);
>+  formatIdString[4] = '\0';
>+  RTCLog(@"AudioStreamBasicDescription: {\n"
>+          "  mSampleRate: %.2f\n"
>+          "  formatIDString: %s\n"
>+          "  mFormatFlags: 0x%X\n"
>+          "  mBytesPerPacket: %u\n"
>+          "  mFramesPerPacket: %u\n"
>+          "  mBytesPerFrame: %u\n"
>+          "  mChannelsPerFrame: %u\n"
>+          "  mBitsPerChannel: %u\n"
>+          "  mReserved: %u\n}",
>+         description.mSampleRate, formatIdString,
>+         static_cast<unsigned int>(description.mFormatFlags),
>+         static_cast<unsigned int>(description.mBytesPerPacket),
>+         static_cast<unsigned int>(description.mFramesPerPacket),
>+         static_cast<unsigned int>(description.mBytesPerFrame),
>+         static_cast<unsigned int>(description.mChannelsPerFrame),
>+         static_cast<unsigned int>(description.mBitsPerChannel),
>+         static_cast<unsigned int>(description.mReserved));
>+}
>+#endif
>+
>+namespace webrtc {
>+namespace ios_adm {
>+
>+// Calls to AudioUnitInitialize() can fail if called back-to-back on different
>+// ADM instances. A fall-back solution is to allow multiple sequential calls
>+// with as small delay between each. This factor sets the max number of allowed
>+// initialization attempts.
>+static const int kMaxNumberOfAudioUnitInitializeAttempts = 5;
>+// A VP I/O unit's bus 1 connects to input hardware (microphone).
>+static const AudioUnitElement kInputBus = 1;
>+// A VP I/O unit's bus 0 connects to output hardware (speaker).
>+static const AudioUnitElement kOutputBus = 0;
>+
>+// Returns the automatic gain control (AGC) state on the processed microphone
>+// signal. Should be on by default for Voice Processing audio units.
>+static OSStatus GetAGCState(AudioUnit audio_unit, UInt32* enabled) {
>+  RTC_DCHECK(audio_unit);
>+  UInt32 size = sizeof(*enabled);
>+  OSStatus result = AudioUnitGetProperty(audio_unit,
>+                                         kAUVoiceIOProperty_VoiceProcessingEnableAGC,
>+                                         kAudioUnitScope_Global,
>+                                         kInputBus,
>+                                         enabled,
>+                                         &size);
>+  RTCLog(@"VPIO unit AGC: %u", static_cast<unsigned int>(*enabled));
>+  return result;
>+}
>+
>+VoiceProcessingAudioUnit::VoiceProcessingAudioUnit(
>+    VoiceProcessingAudioUnitObserver* observer)
>+    : observer_(observer), vpio_unit_(nullptr), state_(kInitRequired) {
>+  RTC_DCHECK(observer);
>+}
>+
>+VoiceProcessingAudioUnit::~VoiceProcessingAudioUnit() {
>+  DisposeAudioUnit();
>+}
>+
>+const UInt32 VoiceProcessingAudioUnit::kBytesPerSample = 2;
>+
>+bool VoiceProcessingAudioUnit::Init() {
>+  RTC_DCHECK_EQ(state_, kInitRequired);
>+
>+  // Create an audio component description to identify the Voice Processing
>+  // I/O audio unit.
>+  AudioComponentDescription vpio_unit_description;
>+  vpio_unit_description.componentType = kAudioUnitType_Output;
>+  vpio_unit_description.componentSubType = kAudioUnitSubType_VoiceProcessingIO;
>+  vpio_unit_description.componentManufacturer = kAudioUnitManufacturer_Apple;
>+  vpio_unit_description.componentFlags = 0;
>+  vpio_unit_description.componentFlagsMask = 0;
>+
>+  // Obtain an audio unit instance given the description.
>+  AudioComponent found_vpio_unit_ref =
>+      AudioComponentFindNext(nullptr, &vpio_unit_description);
>+
>+  // Create a Voice Processing IO audio unit.
>+  OSStatus result = noErr;
>+  result = AudioComponentInstanceNew(found_vpio_unit_ref, &vpio_unit_);
>+  if (result != noErr) {
>+    vpio_unit_ = nullptr;
>+    RTCLogError(@"AudioComponentInstanceNew failed. Error=%ld.", (long)result);
>+    return false;
>+  }
>+
>+  // Enable input on the input scope of the input element.
>+  UInt32 enable_input = 1;
>+  result = AudioUnitSetProperty(vpio_unit_, kAudioOutputUnitProperty_EnableIO,
>+                                kAudioUnitScope_Input, kInputBus, &enable_input,
>+                                sizeof(enable_input));
>+  if (result != noErr) {
>+    DisposeAudioUnit();
>+    RTCLogError(@"Failed to enable input on input scope of input element. "
>+                 "Error=%ld.",
>+                (long)result);
>+    return false;
>+  }
>+
>+  // Enable output on the output scope of the output element.
>+  UInt32 enable_output = 1;
>+  result = AudioUnitSetProperty(vpio_unit_, kAudioOutputUnitProperty_EnableIO,
>+                                kAudioUnitScope_Output, kOutputBus,
>+                                &enable_output, sizeof(enable_output));
>+  if (result != noErr) {
>+    DisposeAudioUnit();
>+    RTCLogError(@"Failed to enable output on output scope of output element. "
>+                 "Error=%ld.",
>+                (long)result);
>+    return false;
>+  }
>+
>+  // Specify the callback function that provides audio samples to the audio
>+  // unit.
>+  AURenderCallbackStruct render_callback;
>+  render_callback.inputProc = OnGetPlayoutData;
>+  render_callback.inputProcRefCon = this;
>+  result = AudioUnitSetProperty(
>+      vpio_unit_, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input,
>+      kOutputBus, &render_callback, sizeof(render_callback));
>+  if (result != noErr) {
>+    DisposeAudioUnit();
>+    RTCLogError(@"Failed to specify the render callback on the output bus. "
>+                 "Error=%ld.",
>+                (long)result);
>+    return false;
>+  }
>+
>+  // Disable AU buffer allocation for the recorder, we allocate our own.
>+  // TODO(henrika): not sure that it actually saves resource to make this call.
>+  UInt32 flag = 0;
>+  result = AudioUnitSetProperty(
>+      vpio_unit_, kAudioUnitProperty_ShouldAllocateBuffer,
>+      kAudioUnitScope_Output, kInputBus, &flag, sizeof(flag));
>+  if (result != noErr) {
>+    DisposeAudioUnit();
>+    RTCLogError(@"Failed to disable buffer allocation on the input bus. "
>+                 "Error=%ld.",
>+                (long)result);
>+    return false;
>+  }
>+
>+  // Specify the callback to be called by the I/O thread to us when input audio
>+  // is available. The recorded samples can then be obtained by calling the
>+  // AudioUnitRender() method.
>+  AURenderCallbackStruct input_callback;
>+  input_callback.inputProc = OnDeliverRecordedData;
>+  input_callback.inputProcRefCon = this;
>+  result = AudioUnitSetProperty(vpio_unit_,
>+                                kAudioOutputUnitProperty_SetInputCallback,
>+                                kAudioUnitScope_Global, kInputBus,
>+                                &input_callback, sizeof(input_callback));
>+  if (result != noErr) {
>+    DisposeAudioUnit();
>+    RTCLogError(@"Failed to specify the input callback on the input bus. "
>+                 "Error=%ld.",
>+                (long)result);
>+    return false;
>+  }
>+
>+  state_ = kUninitialized;
>+  return true;
>+}
>+
>+VoiceProcessingAudioUnit::State VoiceProcessingAudioUnit::GetState() const {
>+  return state_;
>+}
>+
>+bool VoiceProcessingAudioUnit::Initialize(Float64 sample_rate) {
>+  RTC_DCHECK_GE(state_, kUninitialized);
>+  RTCLog(@"Initializing audio unit with sample rate: %f", sample_rate);
>+
>+  OSStatus result = noErr;
>+  AudioStreamBasicDescription format = GetFormat(sample_rate);
>+  UInt32 size = sizeof(format);
>+#if !defined(NDEBUG)
>+  LogStreamDescription(format);
>+#endif
>+
>+  // Set the format on the output scope of the input element/bus.
>+  result =
>+      AudioUnitSetProperty(vpio_unit_, kAudioUnitProperty_StreamFormat,
>+                           kAudioUnitScope_Output, kInputBus, &format, size);
>+  if (result != noErr) {
>+    RTCLogError(@"Failed to set format on output scope of input bus. "
>+                 "Error=%ld.",
>+                (long)result);
>+    return false;
>+  }
>+
>+  // Set the format on the input scope of the output element/bus.
>+  result =
>+      AudioUnitSetProperty(vpio_unit_, kAudioUnitProperty_StreamFormat,
>+                           kAudioUnitScope_Input, kOutputBus, &format, size);
>+  if (result != noErr) {
>+    RTCLogError(@"Failed to set format on input scope of output bus. "
>+                 "Error=%ld.",
>+                (long)result);
>+    return false;
>+  }
>+
>+  // Initialize the Voice Processing I/O unit instance.
>+  // Calls to AudioUnitInitialize() can fail if called back-to-back on
>+  // different ADM instances. The error message in this case is -66635 which is
>+  // undocumented. Tests have shown that calling AudioUnitInitialize a second
>+  // time, after a short sleep, avoids this issue.
>+  // See webrtc:5166 for details.
>+  int failed_initalize_attempts = 0;
>+  result = AudioUnitInitialize(vpio_unit_);
>+  while (result != noErr) {
>+    RTCLogError(@"Failed to initialize the Voice Processing I/O unit. "
>+                 "Error=%ld.",
>+                (long)result);
>+    ++failed_initalize_attempts;
>+    if (failed_initalize_attempts == kMaxNumberOfAudioUnitInitializeAttempts) {
>+      // Max number of initialization attempts exceeded, hence abort.
>+      RTCLogError(@"Too many initialization attempts.");
>+      return false;
>+    }
>+    RTCLog(@"Pause 100ms and try audio unit initialization again...");
>+    [NSThread sleepForTimeInterval:0.1f];
>+    result = AudioUnitInitialize(vpio_unit_);
>+  }
>+  if (result == noErr) {
>+    RTCLog(@"Voice Processing I/O unit is now initialized.");
>+  }
>+
>+  // AGC should be enabled by default for Voice Processing I/O units but it is
>+  // checked below and enabled explicitly if needed. This scheme is used
>+  // to be absolutely sure that the AGC is enabled since we have seen cases
>+  // where only zeros are recorded and a disabled AGC could be one of the
>+  // reasons why it happens.
>+  int agc_was_enabled_by_default = 0;
>+  UInt32 agc_is_enabled = 0;
>+  result = GetAGCState(vpio_unit_, &agc_is_enabled);
>+  if (result != noErr) {
>+    RTCLogError(@"Failed to get AGC state (1st attempt). "
>+                 "Error=%ld.",
>+                (long)result);
>+    // Example of error code: kAudioUnitErr_NoConnection (-10876).
>+    // All error codes related to audio units are negative and are therefore
>+    // converted into a postive value to match the UMA APIs.
>+    RTC_HISTOGRAM_COUNTS_SPARSE_100000(
>+        "WebRTC.Audio.GetAGCStateErrorCode1", (-1) * result);
>+  } else if (agc_is_enabled) {
>+    // Remember that the AGC was enabled by default. Will be used in UMA.
>+    agc_was_enabled_by_default = 1;
>+  } else {
>+    // AGC was initially disabled => try to enable it explicitly.
>+    UInt32 enable_agc = 1;
>+    result =
>+        AudioUnitSetProperty(vpio_unit_,
>+                             kAUVoiceIOProperty_VoiceProcessingEnableAGC,
>+                             kAudioUnitScope_Global, kInputBus, &enable_agc,
>+                             sizeof(enable_agc));
>+    if (result != noErr) {
>+      RTCLogError(@"Failed to enable the built-in AGC. "
>+                   "Error=%ld.",
>+                  (long)result);
>+      RTC_HISTOGRAM_COUNTS_SPARSE_100000(
>+          "WebRTC.Audio.SetAGCStateErrorCode", (-1) * result);
>+    }
>+    result = GetAGCState(vpio_unit_, &agc_is_enabled);
>+    if (result != noErr) {
>+      RTCLogError(@"Failed to get AGC state (2nd attempt). "
>+                   "Error=%ld.",
>+                  (long)result);
>+      RTC_HISTOGRAM_COUNTS_SPARSE_100000(
>+          "WebRTC.Audio.GetAGCStateErrorCode2", (-1) * result);
>+    }
>+  }
>+
>+  // Track if the built-in AGC was enabled by default (as it should) or not.
>+  RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.BuiltInAGCWasEnabledByDefault",
>+                        agc_was_enabled_by_default);
>+  RTCLog(@"WebRTC.Audio.BuiltInAGCWasEnabledByDefault: %d",
>+         agc_was_enabled_by_default);
>+  // As a final step, add an UMA histogram for tracking the AGC state.
>+  // At this stage, the AGC should be enabled, and if it is not, more work is
>+  // needed to find out the root cause.
>+  RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.BuiltInAGCIsEnabled", agc_is_enabled);
>+  RTCLog(@"WebRTC.Audio.BuiltInAGCIsEnabled: %u",
>+         static_cast<unsigned int>(agc_is_enabled));
>+
>+  state_ = kInitialized;
>+  return true;
>+}
>+
>+bool VoiceProcessingAudioUnit::Start() {
>+  RTC_DCHECK_GE(state_, kUninitialized);
>+  RTCLog(@"Starting audio unit.");
>+
>+  OSStatus result = AudioOutputUnitStart(vpio_unit_);
>+  if (result != noErr) {
>+    RTCLogError(@"Failed to start audio unit. Error=%ld", (long)result);
>+    return false;
>+  } else {
>+    RTCLog(@"Started audio unit");
>+  }
>+  state_ = kStarted;
>+  return true;
>+}
>+
>+bool VoiceProcessingAudioUnit::Stop() {
>+  RTC_DCHECK_GE(state_, kUninitialized);
>+  RTCLog(@"Stopping audio unit.");
>+
>+  OSStatus result = AudioOutputUnitStop(vpio_unit_);
>+  if (result != noErr) {
>+    RTCLogError(@"Failed to stop audio unit. Error=%ld", (long)result);
>+    return false;
>+  } else {
>+    RTCLog(@"Stopped audio unit");
>+  }
>+
>+  state_ = kInitialized;
>+  return true;
>+}
>+
>+bool VoiceProcessingAudioUnit::Uninitialize() {
>+  RTC_DCHECK_GE(state_, kUninitialized);
>+  RTCLog(@"Unintializing audio unit.");
>+
>+  OSStatus result = AudioUnitUninitialize(vpio_unit_);
>+  if (result != noErr) {
>+    RTCLogError(@"Failed to uninitialize audio unit. Error=%ld", (long)result);
>+    return false;
>+  } else {
>+    RTCLog(@"Uninitialized audio unit.");
>+  }
>+
>+  state_ = kUninitialized;
>+  return true;
>+}
>+
>+OSStatus VoiceProcessingAudioUnit::Render(AudioUnitRenderActionFlags* flags,
>+                                          const AudioTimeStamp* time_stamp,
>+                                          UInt32 output_bus_number,
>+                                          UInt32 num_frames,
>+                                          AudioBufferList* io_data) {
>+  RTC_DCHECK(vpio_unit_) << "Init() not called.";
>+
>+  OSStatus result = AudioUnitRender(vpio_unit_, flags, time_stamp,
>+                                    output_bus_number, num_frames, io_data);
>+  if (result != noErr) {
>+    RTCLogError(@"Failed to render audio unit. Error=%ld", (long)result);
>+  }
>+  return result;
>+}
>+
>+OSStatus VoiceProcessingAudioUnit::OnGetPlayoutData(
>+    void* in_ref_con,
>+    AudioUnitRenderActionFlags* flags,
>+    const AudioTimeStamp* time_stamp,
>+    UInt32 bus_number,
>+    UInt32 num_frames,
>+    AudioBufferList* io_data) {
>+  VoiceProcessingAudioUnit* audio_unit =
>+      static_cast<VoiceProcessingAudioUnit*>(in_ref_con);
>+  return audio_unit->NotifyGetPlayoutData(flags, time_stamp, bus_number,
>+                                          num_frames, io_data);
>+}
>+
>+OSStatus VoiceProcessingAudioUnit::OnDeliverRecordedData(
>+    void* in_ref_con,
>+    AudioUnitRenderActionFlags* flags,
>+    const AudioTimeStamp* time_stamp,
>+    UInt32 bus_number,
>+    UInt32 num_frames,
>+    AudioBufferList* io_data) {
>+  VoiceProcessingAudioUnit* audio_unit =
>+      static_cast<VoiceProcessingAudioUnit*>(in_ref_con);
>+  return audio_unit->NotifyDeliverRecordedData(flags, time_stamp, bus_number,
>+                                               num_frames, io_data);
>+}
>+
>+OSStatus VoiceProcessingAudioUnit::NotifyGetPlayoutData(
>+    AudioUnitRenderActionFlags* flags,
>+    const AudioTimeStamp* time_stamp,
>+    UInt32 bus_number,
>+    UInt32 num_frames,
>+    AudioBufferList* io_data) {
>+  return observer_->OnGetPlayoutData(flags, time_stamp, bus_number, num_frames,
>+                                     io_data);
>+}
>+
>+OSStatus VoiceProcessingAudioUnit::NotifyDeliverRecordedData(
>+    AudioUnitRenderActionFlags* flags,
>+    const AudioTimeStamp* time_stamp,
>+    UInt32 bus_number,
>+    UInt32 num_frames,
>+    AudioBufferList* io_data) {
>+  return observer_->OnDeliverRecordedData(flags, time_stamp, bus_number,
>+                                          num_frames, io_data);
>+}
>+
>+AudioStreamBasicDescription VoiceProcessingAudioUnit::GetFormat(
>+    Float64 sample_rate) const {
>+  // Set the application formats for input and output:
>+  // - use same format in both directions
>+  // - avoid resampling in the I/O unit by using the hardware sample rate
>+  // - linear PCM => noncompressed audio data format with one frame per packet
>+  // - no need to specify interleaving since only mono is supported
>+  AudioStreamBasicDescription format;
>+  RTC_DCHECK_EQ(1, kRTCAudioSessionPreferredNumberOfChannels);
>+  format.mSampleRate = sample_rate;
>+  format.mFormatID = kAudioFormatLinearPCM;
>+  format.mFormatFlags =
>+      kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsPacked;
>+  format.mBytesPerPacket = kBytesPerSample;
>+  format.mFramesPerPacket = 1;  // uncompressed.
>+  format.mBytesPerFrame = kBytesPerSample;
>+  format.mChannelsPerFrame = kRTCAudioSessionPreferredNumberOfChannels;
>+  format.mBitsPerChannel = 8 * kBytesPerSample;
>+  return format;
>+}
>+
>+void VoiceProcessingAudioUnit::DisposeAudioUnit() {
>+  if (vpio_unit_) {
>+    switch (state_) {
>+      case kStarted:
>+        Stop();
>+        // Fall through.
>+        RTC_FALLTHROUGH();
>+      case kInitialized:
>+        Uninitialize();
>+        break;
>+      case kUninitialized:
>+        RTC_FALLTHROUGH();
>+      case kInitRequired:
>+        break;
>+    }
>+
>+    RTCLog(@"Disposing audio unit.");
>+    OSStatus result = AudioComponentInstanceDispose(vpio_unit_);
>+    if (result != noErr) {
>+      RTCLogError(@"AudioComponentInstanceDispose failed. Error=%ld.",
>+                  (long)result);
>+    }
>+    vpio_unit_ = nullptr;
>+  }
>+}
>+
>+}  // namespace ios_adm
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/objc_frame_buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/objc_frame_buffer.h
>similarity index 78%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/objc_frame_buffer.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/objc_frame_buffer.h
>index 163028eb29d..dc9784a17db 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/objc_frame_buffer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/objc_frame_buffer.h
>@@ -8,14 +8,13 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#ifndef SDK_OBJC_FRAMEWORK_CLASSES_VIDEO_OBJC_FRAME_BUFFER_H_
>-#define SDK_OBJC_FRAMEWORK_CLASSES_VIDEO_OBJC_FRAME_BUFFER_H_
>+#ifndef SDK_OBJC_FRAMEWORK_NATIVE_SRC_OBJC_FRAME_BUFFER_H_
>+#define SDK_OBJC_FRAMEWORK_NATIVE_SRC_OBJC_FRAME_BUFFER_H_
> 
> #import <CoreVideo/CoreVideo.h>
> 
> #include "common_video/include/video_frame_buffer.h"
> 
>-__attribute__((objc_runtime_name("WK_RTCVideoFrameBuffer")))
> @protocol RTCVideoFrameBuffer;
> 
> namespace webrtc {
>@@ -40,6 +39,9 @@ class ObjCFrameBuffer : public VideoFrameBuffer {
>   int height_;
> };
> 
>+id<RTCVideoFrameBuffer> ToObjCVideoFrameBuffer(
>+    const rtc::scoped_refptr<VideoFrameBuffer>& buffer);
>+
> }  // namespace webrtc
> 
>-#endif  // SDK_OBJC_FRAMEWORK_CLASSES_VIDEO_OBJC_FRAME_BUFFER_H_
>+#endif  // SDK_OBJC_FRAMEWORK_NATIVE_SRC_OBJC_FRAME_BUFFER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/objc_frame_buffer.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/objc_frame_buffer.mm
>similarity index 82%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/objc_frame_buffer.mm
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/objc_frame_buffer.mm
>index 3658562eb29..b9434140463 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/objc_frame_buffer.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/objc_frame_buffer.mm
>@@ -8,9 +8,10 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "sdk/objc/Framework/Classes/Video/objc_frame_buffer.h"
>+#include "sdk/objc/Framework/Native/src/objc_frame_buffer.h"
> 
> #import "WebRTC/RTCVideoFrameBuffer.h"
>+#import "sdk/objc/Framework/Classes/Video/RTCI420Buffer+Private.h"
> 
> namespace webrtc {
> 
>@@ -75,4 +76,12 @@ id<RTCVideoFrameBuffer> ObjCFrameBuffer::wrapped_frame_buffer() const {
>   return frame_buffer_;
> }
> 
>+id<RTCVideoFrameBuffer> ToObjCVideoFrameBuffer(const rtc::scoped_refptr<VideoFrameBuffer>& buffer) {
>+  if (buffer->type() == VideoFrameBuffer::Type::kNative) {
>+    return static_cast<ObjCFrameBuffer*>(buffer.get())->wrapped_frame_buffer();
>+  } else {
>+    return [[RTCI420Buffer alloc] initWithFrameBuffer:buffer->ToI420()];
>+  }
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/objc_video_decoder_factory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/objc_video_decoder_factory.h
>new file mode 100644
>index 00000000000..3e2980896e1
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/objc_video_decoder_factory.h
>@@ -0,0 +1,38 @@
>+/*
>+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef SDK_OBJC_FRAMEWORK_NATIVE_SRC_OBJC_VIDEO_DECODER_FACTORY_H_
>+#define SDK_OBJC_FRAMEWORK_NATIVE_SRC_OBJC_VIDEO_DECODER_FACTORY_H_
>+
>+#include "api/video_codecs/video_decoder_factory.h"
>+#include "media/base/codec.h"
>+
>+@protocol RTCVideoDecoderFactory;
>+
>+namespace webrtc {
>+
>+class ObjCVideoDecoderFactory : public VideoDecoderFactory {
>+ public:
>+  explicit ObjCVideoDecoderFactory(id<RTCVideoDecoderFactory>);
>+  ~ObjCVideoDecoderFactory() override;
>+
>+  id<RTCVideoDecoderFactory> wrapped_decoder_factory() const;
>+
>+  std::vector<SdpVideoFormat> GetSupportedFormats() const override;
>+  std::unique_ptr<VideoDecoder> CreateVideoDecoder(
>+      const SdpVideoFormat& format) override;
>+
>+ private:
>+  id<RTCVideoDecoderFactory> decoder_factory_;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // SDK_OBJC_FRAMEWORK_NATIVE_SRC_OBJC_VIDEO_DECODER_FACTORY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/objc_video_decoder_factory.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/objc_video_decoder_factory.mm
>similarity index 78%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/objc_video_decoder_factory.mm
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/objc_video_decoder_factory.mm
>index ab615a39687..13956cfe190 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/objc_video_decoder_factory.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/objc_video_decoder_factory.mm
>@@ -8,16 +8,16 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "sdk/objc/Framework/Classes/VideoToolbox/objc_video_decoder_factory.h"
>+#include "sdk/objc/Framework/Native/src/objc_video_decoder_factory.h"
> 
>-#import "NSString+StdString.h"
>-#import "RTCVideoCodec+Private.h"
>-#import "RTCWrappedNativeVideoDecoder.h"
> #import "WebRTC/RTCVideoCodec.h"
> #import "WebRTC/RTCVideoCodecFactory.h"
> #import "WebRTC/RTCVideoCodecH264.h"
> #import "WebRTC/RTCVideoFrame.h"
> #import "WebRTC/RTCVideoFrameBuffer.h"
>+#import "sdk/objc/Framework/Classes/Common/NSString+StdString.h"
>+#import "sdk/objc/Framework/Classes/PeerConnection/RTCVideoCodec+Private.h"
>+#import "sdk/objc/Framework/Classes/PeerConnection/RTCWrappedNativeVideoDecoder.h"
> 
> #include "api/video_codecs/sdp_video_format.h"
> #include "api/video_codecs/video_decoder.h"
>@@ -26,7 +26,7 @@
> #include "modules/video_coding/include/video_error_codes.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/timeutils.h"
>-#include "sdk/objc/Framework/Classes/Video/objc_frame_buffer.h"
>+#include "sdk/objc/Framework/Native/src/objc_frame_buffer.h"
> 
> namespace webrtc {
> 
>@@ -37,20 +37,23 @@ class ObjCVideoDecoder : public VideoDecoder {
>       : decoder_(decoder), implementation_name_([decoder implementationName].stdString) {}
> 
>   int32_t InitDecode(const VideoCodec *codec_settings, int32_t number_of_cores) {
>-    RTCVideoEncoderSettings *settings =
>-        [[RTCVideoEncoderSettings alloc] initWithNativeVideoCodec:codec_settings];
>-    return [decoder_ startDecodeWithSettings:settings numberOfCores:number_of_cores];
>+    if ([decoder_ respondsToSelector:@selector(startDecodeWithNumberOfCores:)]) {
>+      return [decoder_ startDecodeWithNumberOfCores:number_of_cores];
>+    } else {
>+#pragma clang diagnostic push
>+#pragma clang diagnostic ignored "-Wdeprecated-declarations"
>+      RTCVideoEncoderSettings *settings = [[RTCVideoEncoderSettings alloc] init];
>+      return [decoder_ startDecodeWithSettings:settings numberOfCores:number_of_cores];
>+#pragma clang diagnostic pop
>+    }
>   }
> 
>   int32_t Decode(const EncodedImage &input_image,
>                  bool missing_frames,
>-                 const RTPFragmentationHeader *fragmentation,
>                  const CodecSpecificInfo *codec_specific_info = NULL,
>                  int64_t render_time_ms = -1) {
>     RTCEncodedImage *encodedImage =
>         [[RTCEncodedImage alloc] initWithNativeEncodedImage:input_image];
>-    RTCRtpFragmentationHeader *header =
>-        [[RTCRtpFragmentationHeader alloc] initWithNativeFragmentationHeader:fragmentation];
> 
>     // webrtc::CodecSpecificInfo only handles a hard coded list of codecs
>     id<RTCCodecSpecificInfo> rtcCodecSpecificInfo = nil;
>@@ -65,7 +68,6 @@ class ObjCVideoDecoder : public VideoDecoder {
> 
>     return [decoder_ decode:encodedImage
>               missingFrames:missing_frames
>-        fragmentationHeader:header
>           codecSpecificInfo:rtcCodecSpecificInfo
>                renderTimeMs:render_time_ms];
>   }
>@@ -133,22 +135,4 @@ std::vector<SdpVideoFormat> ObjCVideoDecoderFactory::GetSupportedFormats() const
>   return supported_formats;
> }
> 
>-// WebRtcVideoDecoderFactory
>-
>-VideoDecoder *ObjCVideoDecoderFactory::CreateVideoDecoderWithParams(
>-    const cricket::VideoCodec &codec, cricket::VideoDecoderParams params) {
>-  return CreateVideoDecoder(SdpVideoFormat(codec.name, codec.params)).release();
>-}
>-
>-VideoDecoder *ObjCVideoDecoderFactory::CreateVideoDecoder(VideoCodecType type) {
>-  // This is implemented to avoid hiding an overloaded virtual function
>-  RTC_NOTREACHED();
>-  return nullptr;
>-}
>-
>-void ObjCVideoDecoderFactory::DestroyVideoDecoder(VideoDecoder *decoder) {
>-  delete decoder;
>-  decoder = nullptr;
>-}
>-
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/objc_video_encoder_factory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/objc_video_encoder_factory.h
>new file mode 100644
>index 00000000000..43a3bf431ef
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/objc_video_encoder_factory.h
>@@ -0,0 +1,40 @@
>+/*
>+ *  Copyright 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef SDK_OBJC_FRAMEWORK_NATIVE_SRC_OBJC_VIDEO_ENCODER_FACTORY_H_
>+#define SDK_OBJC_FRAMEWORK_NATIVE_SRC_OBJC_VIDEO_ENCODER_FACTORY_H_
>+
>+#import <Foundation/Foundation.h>
>+
>+#include "api/video_codecs/video_encoder_factory.h"
>+
>+@protocol RTCVideoEncoderFactory;
>+
>+namespace webrtc {
>+
>+class ObjCVideoEncoderFactory : public VideoEncoderFactory {
>+ public:
>+  explicit ObjCVideoEncoderFactory(id<RTCVideoEncoderFactory>);
>+  ~ObjCVideoEncoderFactory() override;
>+
>+  id<RTCVideoEncoderFactory> wrapped_encoder_factory() const;
>+
>+  std::vector<SdpVideoFormat> GetSupportedFormats() const override;
>+  std::unique_ptr<VideoEncoder> CreateVideoEncoder(
>+      const SdpVideoFormat& format) override;
>+  CodecInfo QueryVideoEncoder(const SdpVideoFormat& format) const override;
>+
>+ private:
>+  id<RTCVideoEncoderFactory> encoder_factory_;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // SDK_OBJC_FRAMEWORK_NATIVE_SRC_OBJC_VIDEO_ENCODER_FACTORY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/objc_video_encoder_factory.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/objc_video_encoder_factory.mm
>similarity index 71%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/objc_video_encoder_factory.mm
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/objc_video_encoder_factory.mm
>index 13804867302..3794da67b9c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/objc_video_encoder_factory.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/objc_video_encoder_factory.mm
>@@ -8,20 +8,17 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "sdk/objc/Framework/Classes/VideoToolbox/objc_video_encoder_factory.h"
>+#include "sdk/objc/Framework/Native/src/objc_video_encoder_factory.h"
> 
> #include <string>
> 
>-#import "NSString+StdString.h"
>-#import "RTCI420Buffer+Private.h"
>-#import "RTCVideoCodec+Private.h"
>-#import "RTCVideoFrame+Private.h"
>-#import "RTCWrappedNativeVideoEncoder.h"
> #import "WebRTC/RTCVideoCodec.h"
> #import "WebRTC/RTCVideoCodecFactory.h"
> #import "WebRTC/RTCVideoCodecH264.h"
>-#import "WebRTC/RTCVideoFrame.h"
>-#import "WebRTC/RTCVideoFrameBuffer.h"
>+#import "sdk/objc/Framework/Classes/Common/NSString+StdString.h"
>+#import "sdk/objc/Framework/Classes/PeerConnection/RTCVideoCodec+Private.h"
>+#import "sdk/objc/Framework/Classes/PeerConnection/RTCWrappedNativeVideoEncoder.h"
>+#import "sdk/objc/Framework/Classes/Video/RTCI420Buffer+Private.h"
> 
> #include "api/video/video_frame.h"
> #include "api/video_codecs/sdp_video_format.h"
>@@ -31,7 +28,7 @@
> #include "modules/video_coding/include/video_error_codes.h"
> #include "rtc_base/logging.h"
> #include "sdk/objc/Framework/Classes/Common/helpers.h"
>-#include "sdk/objc/Framework/Classes/Video/objc_frame_buffer.h"
>+#include "sdk/objc/Framework/Native/src/objc_video_frame.h"
> 
> namespace webrtc {
> 
>@@ -77,24 +74,13 @@ class ObjCVideoEncoder : public VideoEncoder {
>   int32_t Encode(const VideoFrame &frame,
>                  const CodecSpecificInfo *codec_specific_info,
>                  const std::vector<FrameType> *frame_types) {
>-    // CodecSpecificInfo only handles a hard coded list of codecs
>-    id<RTCCodecSpecificInfo> rtcCodecSpecificInfo = nil;
>-    if (codec_specific_info) {
>-      if (strcmp(codec_specific_info->codec_name, cricket::kH264CodecName) == 0) {
>-        RTCCodecSpecificInfoH264 *h264Info = [[RTCCodecSpecificInfoH264 alloc] init];
>-        h264Info.packetizationMode =
>-            (RTCH264PacketizationMode)codec_specific_info->codecSpecific.H264.packetization_mode;
>-        rtcCodecSpecificInfo = h264Info;
>-      }
>-    }
>-
>     NSMutableArray<NSNumber *> *rtcFrameTypes = [NSMutableArray array];
>     for (size_t i = 0; i < frame_types->size(); ++i) {
>       [rtcFrameTypes addObject:@(RTCFrameType(frame_types->at(i)))];
>     }
> 
>-    return [encoder_ encode:[[RTCVideoFrame alloc] initWithNativeVideoFrame:frame]
>-          codecSpecificInfo:rtcCodecSpecificInfo
>+    return [encoder_ encode:ToObjCVideoFrame(frame)
>+          codecSpecificInfo:nil
>                  frameTypes:rtcFrameTypes];
>   }
> 
>@@ -107,10 +93,9 @@ class ObjCVideoEncoder : public VideoEncoder {
>   bool SupportsNativeHandle() const { return true; }
> 
>   VideoEncoder::ScalingSettings GetScalingSettings() const {
>-    RTCVideoEncoderQpThresholds* qp_thresholds = [encoder_ scalingSettings];
>-    return qp_thresholds ?
>-        ScalingSettings(true /* enabled */, qp_thresholds.low, qp_thresholds.high) :
>-        ScalingSettings(false /* enabled */);
>+    RTCVideoEncoderQpThresholds *qp_thresholds = [encoder_ scalingSettings];
>+    return qp_thresholds ? ScalingSettings(qp_thresholds.low, qp_thresholds.high) :
>+                           ScalingSettings::kOff;
>   }
> 
>   const char *ImplementationName() const { return implementation_name_.c_str(); }
>@@ -164,28 +149,4 @@ std::unique_ptr<VideoEncoder> ObjCVideoEncoderFactory::CreateVideoEncoder(
>   }
> }
> 
>-// WebRtcVideoEncoderFactory
>-
>-VideoEncoder *ObjCVideoEncoderFactory::CreateVideoEncoder(const cricket::VideoCodec &codec) {
>-  RTCVideoCodecInfo *info = [[RTCVideoCodecInfo alloc]
>-      initWithNativeSdpVideoFormat:SdpVideoFormat(codec.name, codec.params)];
>-  id<RTCVideoEncoder> encoder = [encoder_factory_ createEncoder:info];
>-  return new ObjCVideoEncoder(encoder);
>-}
>-
>-const std::vector<cricket::VideoCodec> &ObjCVideoEncoderFactory::supported_codecs() const {
>-  supported_codecs_.clear();
>-  for (RTCVideoCodecInfo *supportedCodec in encoder_factory_.supportedCodecs) {
>-    SdpVideoFormat format = [supportedCodec nativeSdpVideoFormat];
>-    supported_codecs_.push_back(cricket::VideoCodec(format));
>-  }
>-
>-  return supported_codecs_;
>-}
>-
>-void ObjCVideoEncoderFactory::DestroyVideoEncoder(VideoEncoder *encoder) {
>-  delete encoder;
>-  encoder = nullptr;
>-}
>-
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/objc_video_frame.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/objc_video_frame.h
>new file mode 100644
>index 00000000000..e3ef3df8af5
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/objc_video_frame.h
>@@ -0,0 +1,24 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef SDK_OBJC_FRAMEWORK_NATIVE_SRC_OBJC_VIDEO_FRAME_H_
>+#define SDK_OBJC_FRAMEWORK_NATIVE_SRC_OBJC_VIDEO_FRAME_H_
>+
>+#import "WebRTC/RTCVideoFrame.h"
>+
>+#include "api/video/video_frame.h"
>+
>+namespace webrtc {
>+
>+RTCVideoFrame* ToObjCVideoFrame(const VideoFrame& frame);
>+
>+}  // namespace webrtc
>+
>+#endif  // SDK_OBJC_FRAMEWORK_NATIVE_SRC_OBJC_VIDEO_FRAME_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/objc_video_frame.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/objc_video_frame.mm
>new file mode 100644
>index 00000000000..f0eee193dcf
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/objc_video_frame.mm
>@@ -0,0 +1,28 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "sdk/objc/Framework/Native/src/objc_video_frame.h"
>+
>+#include "rtc_base/timeutils.h"
>+#include "sdk/objc/Framework/Native/src/objc_frame_buffer.h"
>+
>+namespace webrtc {
>+
>+RTCVideoFrame *ToObjCVideoFrame(const VideoFrame &frame) {
>+  RTCVideoFrame *videoFrame =
>+      [[RTCVideoFrame alloc] initWithBuffer:ToObjCVideoFrameBuffer(frame.video_frame_buffer())
>+                                   rotation:RTCVideoRotation(frame.rotation())
>+                                timeStampNs:frame.timestamp_us() * rtc::kNumNanosecsPerMicrosec];
>+  videoFrame.timeStamp = frame.timestamp();
>+
>+  return videoFrame;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/objc_video_renderer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/objc_video_renderer.h
>new file mode 100644
>index 00000000000..9b7c9b00fd4
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/objc_video_renderer.h
>@@ -0,0 +1,36 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef SDK_OBJC_FRAMEWORK_NATIVE_SRC_OBJC_VIDEO_RENDERER_H_
>+#define SDK_OBJC_FRAMEWORK_NATIVE_SRC_OBJC_VIDEO_RENDERER_H_
>+
>+#import <CoreGraphics/CoreGraphics.h>
>+#import <Foundation/Foundation.h>
>+
>+#include "api/video/video_frame.h"
>+#include "api/video/video_sink_interface.h"
>+
>+@protocol RTCVideoRenderer;
>+
>+namespace webrtc {
>+
>+class ObjCVideoRenderer : public rtc::VideoSinkInterface<VideoFrame> {
>+ public:
>+  ObjCVideoRenderer(id<RTCVideoRenderer> renderer);
>+  void OnFrame(const VideoFrame& nativeVideoFrame) override;
>+
>+ private:
>+  id<RTCVideoRenderer> renderer_;
>+  CGSize size_;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // SDK_OBJC_FRAMEWORK_NATIVE_SRC_OBJC_VIDEO_RENDERER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/objc_video_renderer.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/objc_video_renderer.mm
>new file mode 100644
>index 00000000000..aa8f0230689
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/objc_video_renderer.mm
>@@ -0,0 +1,37 @@
>+/*
>+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "sdk/objc/Framework/Native/src/objc_video_renderer.h"
>+
>+#import "WebRTC/RTCVideoFrame.h"
>+#import "WebRTC/RTCVideoRenderer.h"
>+
>+#include "sdk/objc/Framework/Native/src/objc_video_frame.h"
>+
>+namespace webrtc {
>+
>+ObjCVideoRenderer::ObjCVideoRenderer(id<RTCVideoRenderer> renderer)
>+    : renderer_(renderer), size_(CGSizeZero) {}
>+
>+void ObjCVideoRenderer::OnFrame(const VideoFrame& nativeVideoFrame) {
>+  RTCVideoFrame* videoFrame = ToObjCVideoFrame(nativeVideoFrame);
>+
>+  CGSize current_size = (videoFrame.rotation % 180 == 0) ?
>+      CGSizeMake(videoFrame.width, videoFrame.height) :
>+      CGSizeMake(videoFrame.height, videoFrame.width);
>+
>+  if (!CGSizeEqualToSize(size_, current_size)) {
>+    size_ = current_size;
>+    [renderer_ setSize:size_];
>+  }
>+  [renderer_ renderFrame:videoFrame];
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/objcvideotracksource.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/objc_video_track_source.h
>similarity index 74%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/objcvideotracksource.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/objc_video_track_source.h
>index 656fd661189..e26b2815710 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/objcvideotracksource.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/objc_video_track_source.h
>@@ -11,38 +11,47 @@
> #ifndef SDK_OBJC_FRAMEWORK_CLASSES_VIDEO_OBJCVIDEOTRACKSOURCE_H_
> #define SDK_OBJC_FRAMEWORK_CLASSES_VIDEO_OBJCVIDEOTRACKSOURCE_H_
> 
>+#import "WebRTC/RTCVideoCapturer.h"
>+
> #include "WebRTC/RTCMacros.h"
> #include "media/base/adaptedvideotracksource.h"
> #include "rtc_base/timestampaligner.h"
> 
> RTC_FWD_DECL_OBJC_CLASS(RTCVideoFrame);
> 
>+@interface RTCObjCVideoSourceAdapter : NSObject<RTCVideoCapturerDelegate>
>+@end
>+
> namespace webrtc {
> 
>-class ObjcVideoTrackSource : public rtc::AdaptedVideoTrackSource {
>+class ObjCVideoTrackSource : public rtc::AdaptedVideoTrackSource {
>  public:
>-  ObjcVideoTrackSource();
>+  ObjCVideoTrackSource();
>+  explicit ObjCVideoTrackSource(RTCObjCVideoSourceAdapter* adapter);
> 
>   // This class can not be used for implementing screen casting. Hopefully, this
>   // function will be removed before we add that to iOS/Mac.
>-  bool is_screencast() const override { return false; }
>+  bool is_screencast() const override;
> 
>   // Indicates that the encoder should denoise video before encoding it.
>   // If it is not set, the default configuration is used which is different
>   // depending on video codec.
>-  rtc::Optional<bool> needs_denoising() const override { return false; }
>+  absl::optional<bool> needs_denoising() const override;
> 
>-  SourceState state() const override { return SourceState::kLive; }
>+  SourceState state() const override;
> 
>-  bool remote() const override { return false; }
>+  bool remote() const override;
> 
>-  // Called by RTCVideoSource.
>   void OnCapturedFrame(RTCVideoFrame* frame);
>+
>+  // Called by RTCVideoSource.
>   void OnOutputFormatRequest(int width, int height, int fps);
> 
>  private:
>   rtc::VideoBroadcaster broadcaster_;
>   rtc::TimestampAligner timestamp_aligner_;
>+
>+  RTCObjCVideoSourceAdapter* adapter_;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/objcvideotracksource.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/objc_video_track_source.mm
>similarity index 60%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/objcvideotracksource.mm
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/objc_video_track_source.mm
>index eca8852c0fb..abb03781bf5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/objcvideotracksource.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Native/src/objc_video_track_source.mm
>@@ -8,24 +8,59 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "sdk/objc/Framework/Classes/Video/objcvideotracksource.h"
>+#include "sdk/objc/Framework/Native/src/objc_video_track_source.h"
> 
> #import "WebRTC/RTCVideoFrame.h"
> #import "WebRTC/RTCVideoFrameBuffer.h"
> 
> #include "api/video/i420_buffer.h"
>-#include "sdk/objc/Framework/Classes/Video/objc_frame_buffer.h"
>+#include "sdk/objc/Framework/Native/src/objc_frame_buffer.h"
>+
>+@interface RTCObjCVideoSourceAdapter ()
>+@property(nonatomic) webrtc::ObjCVideoTrackSource *objCVideoTrackSource;
>+@end
>+
>+@implementation RTCObjCVideoSourceAdapter
>+
>+@synthesize objCVideoTrackSource = _objCVideoTrackSource;
>+
>+- (void)capturer:(RTCVideoCapturer *)capturer didCaptureVideoFrame:(RTCVideoFrame *)frame {
>+  _objCVideoTrackSource->OnCapturedFrame(frame);
>+}
>+
>+@end
> 
> namespace webrtc {
> 
>-ObjcVideoTrackSource::ObjcVideoTrackSource() {}
>+ObjCVideoTrackSource::ObjCVideoTrackSource() :
>+  AdaptedVideoTrackSource(/* required resolution alignment */ 4) {}
>+
>+ObjCVideoTrackSource::ObjCVideoTrackSource(RTCObjCVideoSourceAdapter *adapter) : adapter_(adapter) {
>+  adapter_.objCVideoTrackSource = this;
>+}
>+
>+bool ObjCVideoTrackSource::is_screencast() const {
>+  return false;
>+}
>+
>+absl::optional<bool> ObjCVideoTrackSource::needs_denoising() const {
>+  return false;
>+}
>+
>+MediaSourceInterface::SourceState ObjCVideoTrackSource::state() const {
>+  return SourceState::kLive;
>+}
>+
>+bool ObjCVideoTrackSource::remote() const {
>+  return false;
>+}
> 
>-void ObjcVideoTrackSource::OnOutputFormatRequest(int width, int height, int fps) {
>+void ObjCVideoTrackSource::OnOutputFormatRequest(int width, int height, int fps) {
>   cricket::VideoFormat format(width, height, cricket::VideoFormat::FpsToInterval(fps), 0);
>   video_adapter()->OnOutputFormatRequest(format);
> }
> 
>-void ObjcVideoTrackSource::OnCapturedFrame(RTCVideoFrame* frame) {
>+void ObjCVideoTrackSource::OnCapturedFrame(RTCVideoFrame *frame) {
>   const int64_t timestamp_us = frame.timeStampNs / rtc::kNumNanosecsPerMicrosec;
>   const int64_t translated_timestamp_us =
>       timestamp_aligner_.TranslateTimestamp(timestamp_us, rtc::TimeMicros());
>@@ -36,8 +71,15 @@ void ObjcVideoTrackSource::OnCapturedFrame(RTCVideoFrame* frame) {
>   int crop_height;
>   int crop_x;
>   int crop_y;
>-  if (!AdaptFrame(frame.width, frame.height, timestamp_us, &adapted_width, &adapted_height,
>-                  &crop_width, &crop_height, &crop_x, &crop_y)) {
>+  if (!AdaptFrame(frame.width,
>+                  frame.height,
>+                  timestamp_us,
>+                  &adapted_width,
>+                  &adapted_height,
>+                  &crop_width,
>+                  &crop_height,
>+                  &crop_x,
>+                  &crop_y)) {
>     return;
>   }
> 
>@@ -67,13 +109,13 @@ void ObjcVideoTrackSource::OnCapturedFrame(RTCVideoFrame* frame) {
> 
>   // Applying rotation is only supported for legacy reasons and performance is
>   // not critical here.
>-  webrtc::VideoRotation rotation = static_cast<webrtc::VideoRotation>(frame.rotation);
>+  VideoRotation rotation = static_cast<VideoRotation>(frame.rotation);
>   if (apply_rotation() && rotation != kVideoRotation_0) {
>     buffer = I420Buffer::Rotate(*buffer->ToI420(), rotation);
>     rotation = kVideoRotation_0;
>   }
> 
>-  OnFrame(webrtc::VideoFrame(buffer, rotation, translated_timestamp_us));
>+  OnFrame(VideoFrame(buffer, rotation, translated_timestamp_us));
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/ObjCVideoTrackSource_xctest.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/ObjCVideoTrackSource_xctest.mm
>new file mode 100644
>index 00000000000..a32ccce314b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/ObjCVideoTrackSource_xctest.mm
>@@ -0,0 +1,393 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#import <Foundation/Foundation.h>
>+#import <XCTest/XCTest.h>
>+
>+#include "sdk/objc/Framework/Native/src/objc_video_track_source.h"
>+
>+#import "Video/RTCI420Buffer+Private.h"
>+#import "WebRTC/RTCVideoFrame.h"
>+#import "WebRTC/RTCVideoFrameBuffer.h"
>+#include "common_video/libyuv/include/webrtc_libyuv.h"
>+#include "media/base/fakevideorenderer.h"
>+#include "rtc_base/refcountedobject.h"
>+#include "rtc_base/scoped_ref_ptr.h"
>+#include "sdk/objc/Framework/Native/api/video_frame.h"
>+#import "sdk/objc/Framework/UnitTests/frame_buffer_helpers.h"
>+
>+typedef void (^VideoSinkCallback)(RTCVideoFrame *);
>+
>+namespace {
>+
>+class ObjCCallbackVideoSink : public rtc::VideoSinkInterface<webrtc::VideoFrame> {
>+ public:
>+  ObjCCallbackVideoSink(VideoSinkCallback callback) : callback_(callback) {}
>+
>+  void OnFrame(const webrtc::VideoFrame &frame) override {
>+    callback_(NativeToObjCVideoFrame(frame));
>+  }
>+
>+ private:
>+  VideoSinkCallback callback_;
>+};
>+
>+}  // namespace
>+
>+@interface ObjCVideoTrackSourceTests : XCTestCase
>+@end
>+
>+@implementation ObjCVideoTrackSourceTests {
>+  rtc::scoped_refptr<webrtc::ObjCVideoTrackSource> _video_source;
>+}
>+
>+- (void)setUp {
>+  _video_source = new rtc::RefCountedObject<webrtc::ObjCVideoTrackSource>();
>+}
>+
>+- (void)tearDown {
>+  _video_source = NULL;
>+}
>+
>+- (void)testOnCapturedFrameAdaptsFrame {
>+  CVPixelBufferRef pixelBufferRef = NULL;
>+  CVPixelBufferCreate(
>+      NULL, 720, 1280, kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange, NULL, &pixelBufferRef);
>+
>+  RTCCVPixelBuffer *buffer = [[RTCCVPixelBuffer alloc] initWithPixelBuffer:pixelBufferRef];
>+
>+  RTCVideoFrame *frame =
>+      [[RTCVideoFrame alloc] initWithBuffer:buffer rotation:RTCVideoRotation_0 timeStampNs:0];
>+
>+  cricket::FakeVideoRenderer *video_renderer = new cricket::FakeVideoRenderer();
>+  const rtc::VideoSinkWants video_sink_wants;
>+  rtc::VideoSourceInterface<webrtc::VideoFrame> *video_source_interface = _video_source;
>+  video_source_interface->AddOrUpdateSink(video_renderer, video_sink_wants);
>+
>+  _video_source->OnOutputFormatRequest(640, 360, 30);
>+  _video_source->OnCapturedFrame(frame);
>+
>+  XCTAssertEqual(video_renderer->num_rendered_frames(), 1);
>+  XCTAssertEqual(video_renderer->width(), 360);
>+  XCTAssertEqual(video_renderer->height(), 640);
>+
>+  CVBufferRelease(pixelBufferRef);
>+}
>+
>+- (void)testOnCapturedFrameAdaptsFrameWithAlignment {
>+  // Requesting to adapt 1280x720 to 912x514 gives 639x360 without alignment. The 639 causes issues
>+  // with some hardware encoders (e.g. HEVC) so in this test we verify that the alignment is set and
>+  // respected.
>+
>+  CVPixelBufferRef pixelBufferRef = NULL;
>+  CVPixelBufferCreate(
>+      NULL, 720, 1280, kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange, NULL, &pixelBufferRef);
>+
>+  RTCCVPixelBuffer *buffer = [[RTCCVPixelBuffer alloc] initWithPixelBuffer:pixelBufferRef];
>+
>+  RTCVideoFrame *frame =
>+      [[RTCVideoFrame alloc] initWithBuffer:buffer rotation:RTCVideoRotation_0 timeStampNs:0];
>+
>+  cricket::FakeVideoRenderer *video_renderer = new cricket::FakeVideoRenderer();
>+  const rtc::VideoSinkWants video_sink_wants;
>+  rtc::VideoSourceInterface<webrtc::VideoFrame> *video_source_interface = _video_source;
>+  video_source_interface->AddOrUpdateSink(video_renderer, video_sink_wants);
>+
>+  _video_source->OnOutputFormatRequest(912, 514, 30);
>+  _video_source->OnCapturedFrame(frame);
>+
>+  XCTAssertEqual(video_renderer->num_rendered_frames(), 1);
>+  XCTAssertEqual(video_renderer->width(), 360);
>+  XCTAssertEqual(video_renderer->height(), 640);
>+
>+  CVBufferRelease(pixelBufferRef);
>+}
>+
>+- (void)testOnCapturedFrameWithoutAdaptation {
>+  CVPixelBufferRef pixelBufferRef = NULL;
>+  CVPixelBufferCreate(
>+      NULL, 360, 640, kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange, NULL, &pixelBufferRef);
>+
>+  RTCCVPixelBuffer *buffer = [[RTCCVPixelBuffer alloc] initWithPixelBuffer:pixelBufferRef];
>+  RTCVideoFrame *frame =
>+      [[RTCVideoFrame alloc] initWithBuffer:buffer rotation:RTCVideoRotation_0 timeStampNs:0];
>+
>+  XCTestExpectation *callbackExpectation = [self expectationWithDescription:@"videoSinkCallback"];
>+  ObjCCallbackVideoSink callback_video_sink(^void(RTCVideoFrame *outputFrame) {
>+    XCTAssertEqual(frame.width, outputFrame.width);
>+    XCTAssertEqual(frame.height, outputFrame.height);
>+
>+    RTCCVPixelBuffer *outputBuffer = outputFrame.buffer;
>+    XCTAssertEqual(buffer.cropX, outputBuffer.cropX);
>+    XCTAssertEqual(buffer.cropY, outputBuffer.cropY);
>+    XCTAssertEqual(buffer.pixelBuffer, outputBuffer.pixelBuffer);
>+
>+    [callbackExpectation fulfill];
>+  });
>+
>+  const rtc::VideoSinkWants video_sink_wants;
>+  rtc::VideoSourceInterface<webrtc::VideoFrame> *video_source_interface = _video_source;
>+  video_source_interface->AddOrUpdateSink(&callback_video_sink, video_sink_wants);
>+
>+  _video_source->OnOutputFormatRequest(640, 360, 30);
>+  _video_source->OnCapturedFrame(frame);
>+
>+  [self waitForExpectations:@[ callbackExpectation ] timeout:10.0];
>+  CVBufferRelease(pixelBufferRef);
>+}
>+
>+- (void)testOnCapturedFrameCVPixelBufferNeedsAdaptation {
>+  CVPixelBufferRef pixelBufferRef = NULL;
>+  CVPixelBufferCreate(
>+      NULL, 720, 1280, kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange, NULL, &pixelBufferRef);
>+
>+  RTCCVPixelBuffer *buffer = [[RTCCVPixelBuffer alloc] initWithPixelBuffer:pixelBufferRef];
>+  RTCVideoFrame *frame =
>+      [[RTCVideoFrame alloc] initWithBuffer:buffer rotation:RTCVideoRotation_0 timeStampNs:0];
>+
>+  XCTestExpectation *callbackExpectation = [self expectationWithDescription:@"videoSinkCallback"];
>+  ObjCCallbackVideoSink callback_video_sink(^void(RTCVideoFrame *outputFrame) {
>+    XCTAssertEqual(outputFrame.width, 360);
>+    XCTAssertEqual(outputFrame.height, 640);
>+
>+    RTCCVPixelBuffer *outputBuffer = outputFrame.buffer;
>+    XCTAssertEqual(outputBuffer.cropX, 0);
>+    XCTAssertEqual(outputBuffer.cropY, 0);
>+    XCTAssertEqual(buffer.pixelBuffer, outputBuffer.pixelBuffer);
>+
>+    [callbackExpectation fulfill];
>+  });
>+
>+  const rtc::VideoSinkWants video_sink_wants;
>+  rtc::VideoSourceInterface<webrtc::VideoFrame> *video_source_interface = _video_source;
>+  video_source_interface->AddOrUpdateSink(&callback_video_sink, video_sink_wants);
>+
>+  _video_source->OnOutputFormatRequest(640, 360, 30);
>+  _video_source->OnCapturedFrame(frame);
>+
>+  [self waitForExpectations:@[ callbackExpectation ] timeout:10.0];
>+  CVBufferRelease(pixelBufferRef);
>+}
>+
>+- (void)testOnCapturedFrameCVPixelBufferNeedsCropping {
>+  CVPixelBufferRef pixelBufferRef = NULL;
>+  CVPixelBufferCreate(
>+      NULL, 380, 640, kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange, NULL, &pixelBufferRef);
>+
>+  RTCCVPixelBuffer *buffer = [[RTCCVPixelBuffer alloc] initWithPixelBuffer:pixelBufferRef];
>+  RTCVideoFrame *frame =
>+      [[RTCVideoFrame alloc] initWithBuffer:buffer rotation:RTCVideoRotation_0 timeStampNs:0];
>+
>+  XCTestExpectation *callbackExpectation = [self expectationWithDescription:@"videoSinkCallback"];
>+  ObjCCallbackVideoSink callback_video_sink(^void(RTCVideoFrame *outputFrame) {
>+    XCTAssertEqual(outputFrame.width, 360);
>+    XCTAssertEqual(outputFrame.height, 640);
>+
>+    RTCCVPixelBuffer *outputBuffer = outputFrame.buffer;
>+    XCTAssertEqual(outputBuffer.cropX, 10);
>+    XCTAssertEqual(outputBuffer.cropY, 0);
>+    XCTAssertEqual(buffer.pixelBuffer, outputBuffer.pixelBuffer);
>+
>+    [callbackExpectation fulfill];
>+  });
>+
>+  const rtc::VideoSinkWants video_sink_wants;
>+  rtc::VideoSourceInterface<webrtc::VideoFrame> *video_source_interface = _video_source;
>+  video_source_interface->AddOrUpdateSink(&callback_video_sink, video_sink_wants);
>+
>+  _video_source->OnOutputFormatRequest(640, 360, 30);
>+  _video_source->OnCapturedFrame(frame);
>+
>+  [self waitForExpectations:@[ callbackExpectation ] timeout:10.0];
>+  CVBufferRelease(pixelBufferRef);
>+}
>+
>+- (void)testOnCapturedFramePreAdaptedCVPixelBufferNeedsAdaptation {
>+  CVPixelBufferRef pixelBufferRef = NULL;
>+  CVPixelBufferCreate(
>+      NULL, 720, 1280, kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange, NULL, &pixelBufferRef);
>+
>+  RTCCVPixelBuffer *buffer = [[RTCCVPixelBuffer alloc] initWithPixelBuffer:pixelBufferRef
>+                                                              adaptedWidth:700
>+                                                             adaptedHeight:700
>+                                                                 cropWidth:720
>+                                                                cropHeight:1280
>+                                                                     cropX:0
>+                                                                     cropY:0];
>+  RTCVideoFrame *frame =
>+      [[RTCVideoFrame alloc] initWithBuffer:buffer rotation:RTCVideoRotation_0 timeStampNs:0];
>+
>+  XCTestExpectation *callbackExpectation = [self expectationWithDescription:@"videoSinkCallback"];
>+  ObjCCallbackVideoSink callback_video_sink(^void(RTCVideoFrame *outputFrame) {
>+    XCTAssertEqual(outputFrame.width, 300);
>+    XCTAssertEqual(outputFrame.height, 516);
>+
>+    RTCCVPixelBuffer *outputBuffer = outputFrame.buffer;
>+    XCTAssertEqual(outputBuffer.cropX, 150);
>+    XCTAssertEqual(outputBuffer.cropY, 6);
>+    XCTAssertEqual(outputBuffer.cropWidth, 400);
>+    XCTAssertEqual(outputBuffer.cropHeight, 688);
>+    XCTAssertEqual(buffer.pixelBuffer, outputBuffer.pixelBuffer);
>+
>+    [callbackExpectation fulfill];
>+  });
>+
>+  const rtc::VideoSinkWants video_sink_wants;
>+  rtc::VideoSourceInterface<webrtc::VideoFrame> *video_source_interface = _video_source;
>+  video_source_interface->AddOrUpdateSink(&callback_video_sink, video_sink_wants);
>+
>+  _video_source->OnOutputFormatRequest(640, 360, 30);
>+  _video_source->OnCapturedFrame(frame);
>+
>+  [self waitForExpectations:@[ callbackExpectation ] timeout:10.0];
>+  CVBufferRelease(pixelBufferRef);
>+}
>+
>+- (void)testOnCapturedFramePreCroppedCVPixelBufferNeedsCropping {
>+  CVPixelBufferRef pixelBufferRef = NULL;
>+  CVPixelBufferCreate(
>+      NULL, 380, 640, kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange, NULL, &pixelBufferRef);
>+
>+  RTCCVPixelBuffer *buffer = [[RTCCVPixelBuffer alloc] initWithPixelBuffer:pixelBufferRef
>+                                                              adaptedWidth:370
>+                                                             adaptedHeight:640
>+                                                                 cropWidth:370
>+                                                                cropHeight:640
>+                                                                     cropX:10
>+                                                                     cropY:0];
>+  RTCVideoFrame *frame =
>+      [[RTCVideoFrame alloc] initWithBuffer:buffer rotation:RTCVideoRotation_0 timeStampNs:0];
>+
>+  XCTestExpectation *callbackExpectation = [self expectationWithDescription:@"videoSinkCallback"];
>+  ObjCCallbackVideoSink callback_video_sink(^void(RTCVideoFrame *outputFrame) {
>+    XCTAssertEqual(outputFrame.width, 360);
>+    XCTAssertEqual(outputFrame.height, 640);
>+
>+    RTCCVPixelBuffer *outputBuffer = outputFrame.buffer;
>+    XCTAssertEqual(outputBuffer.cropX, 14);
>+    XCTAssertEqual(outputBuffer.cropY, 0);
>+    XCTAssertEqual(outputBuffer.cropWidth, 360);
>+    XCTAssertEqual(outputBuffer.cropHeight, 640);
>+    XCTAssertEqual(buffer.pixelBuffer, outputBuffer.pixelBuffer);
>+
>+    [callbackExpectation fulfill];
>+  });
>+
>+  const rtc::VideoSinkWants video_sink_wants;
>+  rtc::VideoSourceInterface<webrtc::VideoFrame> *video_source_interface = _video_source;
>+  video_source_interface->AddOrUpdateSink(&callback_video_sink, video_sink_wants);
>+
>+  _video_source->OnOutputFormatRequest(640, 360, 30);
>+  _video_source->OnCapturedFrame(frame);
>+
>+  [self waitForExpectations:@[ callbackExpectation ] timeout:10.0];
>+  CVBufferRelease(pixelBufferRef);
>+}
>+
>+- (void)testOnCapturedFrameSmallerPreCroppedCVPixelBufferNeedsCropping {
>+  CVPixelBufferRef pixelBufferRef = NULL;
>+  CVPixelBufferCreate(
>+      NULL, 380, 640, kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange, NULL, &pixelBufferRef);
>+
>+  RTCCVPixelBuffer *buffer = [[RTCCVPixelBuffer alloc] initWithPixelBuffer:pixelBufferRef
>+                                                              adaptedWidth:300
>+                                                             adaptedHeight:640
>+                                                                 cropWidth:300
>+                                                                cropHeight:640
>+                                                                     cropX:40
>+                                                                     cropY:0];
>+  RTCVideoFrame *frame =
>+      [[RTCVideoFrame alloc] initWithBuffer:buffer rotation:RTCVideoRotation_0 timeStampNs:0];
>+
>+  XCTestExpectation *callbackExpectation = [self expectationWithDescription:@"videoSinkCallback"];
>+  ObjCCallbackVideoSink callback_video_sink(^void(RTCVideoFrame *outputFrame) {
>+    XCTAssertEqual(outputFrame.width, 300);
>+    XCTAssertEqual(outputFrame.height, 536);
>+
>+    RTCCVPixelBuffer *outputBuffer = outputFrame.buffer;
>+    XCTAssertEqual(outputBuffer.cropX, 40);
>+    XCTAssertEqual(outputBuffer.cropY, 52);
>+    XCTAssertEqual(outputBuffer.cropWidth, 300);
>+    XCTAssertEqual(outputBuffer.cropHeight, 536);
>+    XCTAssertEqual(buffer.pixelBuffer, outputBuffer.pixelBuffer);
>+
>+    [callbackExpectation fulfill];
>+  });
>+
>+  const rtc::VideoSinkWants video_sink_wants;
>+  rtc::VideoSourceInterface<webrtc::VideoFrame> *video_source_interface = _video_source;
>+  video_source_interface->AddOrUpdateSink(&callback_video_sink, video_sink_wants);
>+
>+  _video_source->OnOutputFormatRequest(640, 360, 30);
>+  _video_source->OnCapturedFrame(frame);
>+
>+  [self waitForExpectations:@[ callbackExpectation ] timeout:10.0];
>+  CVBufferRelease(pixelBufferRef);
>+}
>+
>+- (void)testOnCapturedFrameI420BufferNeedsAdaptation {
>+  rtc::scoped_refptr<webrtc::I420Buffer> i420Buffer = CreateI420Gradient(720, 1280);
>+  RTCI420Buffer *buffer = [[RTCI420Buffer alloc] initWithFrameBuffer:i420Buffer];
>+  RTCVideoFrame *frame =
>+      [[RTCVideoFrame alloc] initWithBuffer:buffer rotation:RTCVideoRotation_0 timeStampNs:0];
>+
>+  XCTestExpectation *callbackExpectation = [self expectationWithDescription:@"videoSinkCallback"];
>+  ObjCCallbackVideoSink callback_video_sink(^void(RTCVideoFrame *outputFrame) {
>+    XCTAssertEqual(outputFrame.width, 360);
>+    XCTAssertEqual(outputFrame.height, 640);
>+
>+    RTCI420Buffer *outputBuffer = (RTCI420Buffer *)outputFrame.buffer;
>+
>+    double psnr = I420PSNR(*[buffer nativeI420Buffer], *[outputBuffer nativeI420Buffer]);
>+    XCTAssertEqual(psnr, webrtc::kPerfectPSNR);
>+
>+    [callbackExpectation fulfill];
>+  });
>+
>+  const rtc::VideoSinkWants video_sink_wants;
>+  rtc::VideoSourceInterface<webrtc::VideoFrame> *video_source_interface = _video_source;
>+  video_source_interface->AddOrUpdateSink(&callback_video_sink, video_sink_wants);
>+
>+  _video_source->OnOutputFormatRequest(640, 360, 30);
>+  _video_source->OnCapturedFrame(frame);
>+
>+  [self waitForExpectations:@[ callbackExpectation ] timeout:10.0];
>+}
>+
>+- (void)testOnCapturedFrameI420BufferNeedsCropping {
>+  rtc::scoped_refptr<webrtc::I420Buffer> i420Buffer = CreateI420Gradient(380, 640);
>+  RTCI420Buffer *buffer = [[RTCI420Buffer alloc] initWithFrameBuffer:i420Buffer];
>+  RTCVideoFrame *frame =
>+      [[RTCVideoFrame alloc] initWithBuffer:buffer rotation:RTCVideoRotation_0 timeStampNs:0];
>+
>+  XCTestExpectation *callbackExpectation = [self expectationWithDescription:@"videoSinkCallback"];
>+  ObjCCallbackVideoSink callback_video_sink(^void(RTCVideoFrame *outputFrame) {
>+    XCTAssertEqual(outputFrame.width, 360);
>+    XCTAssertEqual(outputFrame.height, 640);
>+
>+    RTCI420Buffer *outputBuffer = (RTCI420Buffer *)outputFrame.buffer;
>+
>+    double psnr = I420PSNR(*[buffer nativeI420Buffer], *[outputBuffer nativeI420Buffer]);
>+    XCTAssertGreaterThanOrEqual(psnr, 40);
>+
>+    [callbackExpectation fulfill];
>+  });
>+
>+  const rtc::VideoSinkWants video_sink_wants;
>+  rtc::VideoSourceInterface<webrtc::VideoFrame> *video_source_interface = _video_source;
>+  video_source_interface->AddOrUpdateSink(&callback_video_sink, video_sink_wants);
>+
>+  _video_source->OnOutputFormatRequest(640, 360, 30);
>+  _video_source->OnCapturedFrame(frame);
>+
>+  [self waitForExpectations:@[ callbackExpectation ] timeout:10.0];
>+}
>+
>+@end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/RTCAudioDeviceModule_xctest.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/RTCAudioDeviceModule_xctest.mm
>new file mode 100644
>index 00000000000..739ec6af8cb
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/RTCAudioDeviceModule_xctest.mm
>@@ -0,0 +1,590 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#import <XCTest/XCTest.h>
>+
>+#if defined(WEBRTC_IOS)
>+#import "sdk/objc/Framework/Native/api/audio_device_module.h"
>+#endif
>+
>+#include "system_wrappers/include/event_wrapper.h"
>+
>+#include "rtc_base/scoped_ref_ptr.h"
>+
>+typedef int32_t(^NeedMorePlayDataBlock)(const size_t nSamples,
>+                                        const size_t nBytesPerSample,
>+                                        const size_t nChannels,
>+                                        const uint32_t samplesPerSec,
>+                                        void* audioSamples,
>+                                        size_t& nSamplesOut,
>+                                        int64_t* elapsed_time_ms,
>+                                        int64_t* ntp_time_ms);
>+
>+typedef int32_t(^RecordedDataIsAvailableBlock)(const void* audioSamples,
>+                                               const size_t nSamples,
>+                                               const size_t nBytesPerSample,
>+                                               const size_t nChannels,
>+                                               const uint32_t samplesPerSec,
>+                                               const uint32_t totalDelayMS,
>+                                               const int32_t clockDrift,
>+                                               const uint32_t currentMicLevel,
>+                                               const bool keyPressed,
>+                                               uint32_t& newMicLevel);
>+
>+
>+// This class implements the AudioTransport API and forwards all methods to the appropriate blocks.
>+class MockAudioTransport : public webrtc::AudioTransport {
>+public:
>+  MockAudioTransport() {}
>+  ~MockAudioTransport() override {}
>+
>+  void expectNeedMorePlayData(NeedMorePlayDataBlock block) {
>+    needMorePlayDataBlock = block;
>+  }
>+
>+  void expectRecordedDataIsAvailable(RecordedDataIsAvailableBlock block) {
>+    recordedDataIsAvailableBlock = block;
>+  }
>+
>+  int32_t NeedMorePlayData(const size_t nSamples,
>+                           const size_t nBytesPerSample,
>+                           const size_t nChannels,
>+                           const uint32_t samplesPerSec,
>+                           void* audioSamples,
>+                           size_t& nSamplesOut,
>+                           int64_t* elapsed_time_ms,
>+                           int64_t* ntp_time_ms) override {
>+    return needMorePlayDataBlock(nSamples,
>+                                 nBytesPerSample,
>+                                 nChannels,
>+                                 samplesPerSec,
>+                                 audioSamples,
>+                                 nSamplesOut,
>+                                 elapsed_time_ms,
>+                                 ntp_time_ms);
>+  }
>+
>+  int32_t RecordedDataIsAvailable(const void* audioSamples,
>+                                  const size_t nSamples,
>+                                  const size_t nBytesPerSample,
>+                                  const size_t nChannels,
>+                                  const uint32_t samplesPerSec,
>+                                  const uint32_t totalDelayMS,
>+                                  const int32_t clockDrift,
>+                                  const uint32_t currentMicLevel,
>+                                  const bool keyPressed,
>+                                  uint32_t& newMicLevel) override {
>+    return recordedDataIsAvailableBlock(audioSamples,
>+                                        nSamples,
>+                                        nBytesPerSample,
>+                                        nChannels,
>+                                        samplesPerSec,
>+                                        totalDelayMS,
>+                                        clockDrift,
>+                                        currentMicLevel,
>+                                        keyPressed,
>+                                        newMicLevel);
>+  }
>+
>+  void PullRenderData(int bits_per_sample,
>+                      int sample_rate,
>+                      size_t number_of_channels,
>+                      size_t number_of_frames,
>+                      void* audio_data,
>+                      int64_t* elapsed_time_ms,
>+                      int64_t* ntp_time_ms) override {}
>+
>+ private:
>+  NeedMorePlayDataBlock needMorePlayDataBlock;
>+  RecordedDataIsAvailableBlock recordedDataIsAvailableBlock;
>+};
>+
>+// Number of callbacks (input or output) the tests waits for before we set
>+// an event indicating that the test was OK.
>+static const NSUInteger kNumCallbacks = 10;
>+// Max amount of time we wait for an event to be set while counting callbacks.
>+static const NSTimeInterval kTestTimeOutInSec = 20.0;
>+// Number of bits per PCM audio sample.
>+static const NSUInteger kBitsPerSample = 16;
>+// Number of bytes per PCM audio sample.
>+static const NSUInteger kBytesPerSample = kBitsPerSample / 8;
>+// Average number of audio callbacks per second assuming 10ms packet size.
>+static const NSUInteger kNumCallbacksPerSecond = 100;
>+// Play out a test file during this time (unit is in seconds).
>+static const NSUInteger kFilePlayTimeInSec = 15;
>+// Run the full-duplex test during this time (unit is in seconds).
>+// Note that first |kNumIgnoreFirstCallbacks| are ignored.
>+static const NSUInteger kFullDuplexTimeInSec = 10;
>+// Wait for the callback sequence to stabilize by ignoring this amount of the
>+// initial callbacks (avoids initial FIFO access).
>+// Only used in the RunPlayoutAndRecordingInFullDuplex test.
>+static const NSUInteger kNumIgnoreFirstCallbacks = 50;
>+
>+@interface RTCAudioDeviceModuleTests : XCTestCase {
>+
>+  rtc::scoped_refptr<webrtc::AudioDeviceModule> audioDeviceModule;
>+  webrtc::AudioParameters playoutParameters;
>+  webrtc::AudioParameters recordParameters;
>+  MockAudioTransport mock;
>+}
>+
>+@end
>+
>+@implementation RTCAudioDeviceModuleTests
>+
>+- (void)setUp {
>+  [super setUp];
>+  audioDeviceModule = webrtc::CreateAudioDeviceModule();
>+  XCTAssertEqual(0, audioDeviceModule->Init());
>+  XCTAssertEqual(0, audioDeviceModule->GetPlayoutAudioParameters(&playoutParameters));
>+  XCTAssertEqual(0, audioDeviceModule->GetRecordAudioParameters(&recordParameters));
>+}
>+
>+- (void)tearDown {
>+  XCTAssertEqual(0, audioDeviceModule->Terminate());
>+  audioDeviceModule = nullptr;
>+  [super tearDown];
>+}
>+
>+- (void)startPlayout {
>+  XCTAssertFalse(audioDeviceModule->Playing());
>+  XCTAssertEqual(0, audioDeviceModule->InitPlayout());
>+  XCTAssertTrue(audioDeviceModule->PlayoutIsInitialized());
>+  XCTAssertEqual(0, audioDeviceModule->StartPlayout());
>+  XCTAssertTrue(audioDeviceModule->Playing());
>+}
>+
>+- (void)stopPlayout {
>+  XCTAssertEqual(0, audioDeviceModule->StopPlayout());
>+  XCTAssertFalse(audioDeviceModule->Playing());
>+}
>+
>+- (void)startRecording{
>+  XCTAssertFalse(audioDeviceModule->Recording());
>+  XCTAssertEqual(0, audioDeviceModule->InitRecording());
>+  XCTAssertTrue(audioDeviceModule->RecordingIsInitialized());
>+  XCTAssertEqual(0, audioDeviceModule->StartRecording());
>+  XCTAssertTrue(audioDeviceModule->Recording());
>+}
>+
>+- (void)stopRecording{
>+  XCTAssertEqual(0, audioDeviceModule->StopRecording());
>+  XCTAssertFalse(audioDeviceModule->Recording());
>+}
>+
>+- (NSURL*)fileURLForSampleRate:(int)sampleRate {
>+  XCTAssertTrue(sampleRate == 48000 || sampleRate == 44100 || sampleRate == 16000);
>+  NSString *filename = [NSString stringWithFormat:@"audio_short%d", sampleRate / 1000];
>+  NSURL *url = [[NSBundle mainBundle] URLForResource:filename withExtension:@"pcm"];
>+  XCTAssertNotNil(url);
>+
>+  return url;
>+}
>+
>+#pragma mark - Tests
>+
>+- (void)testConstructDestruct {
>+  // Using the test fixture to create and destruct the audio device module.
>+}
>+
>+- (void)testInitTerminate {
>+  // Initialization is part of the test fixture.
>+  XCTAssertTrue(audioDeviceModule->Initialized());
>+  XCTAssertEqual(0, audioDeviceModule->Terminate());
>+  XCTAssertFalse(audioDeviceModule->Initialized());
>+}
>+
>+// Tests that playout can be initiated, started and stopped. No audio callback
>+// is registered in this test.
>+// Failing when running on real iOS devices: bugs.webrtc.org/6889.
>+- (void)DISABLED_testStartStopPlayout {
>+  [self startPlayout];
>+  [self stopPlayout];
>+  [self startPlayout];
>+  [self stopPlayout];
>+}
>+
>+// Tests that recording can be initiated, started and stopped. No audio callback
>+// is registered in this test.
>+// Can sometimes fail when running on real devices: bugs.webrtc.org/7888.
>+- (void)DISABLED_testStartStopRecording {
>+  [self startRecording];
>+  [self stopRecording];
>+  [self startRecording];
>+  [self stopRecording];
>+}
>+// Verify that calling StopPlayout() will leave us in an uninitialized state
>+// which will require a new call to InitPlayout(). This test does not call
>+// StartPlayout() while being uninitialized since doing so will hit a
>+// RTC_DCHECK.
>+- (void)testStopPlayoutRequiresInitToRestart {
>+  XCTAssertEqual(0, audioDeviceModule->InitPlayout());
>+  XCTAssertEqual(0, audioDeviceModule->StartPlayout());
>+  XCTAssertEqual(0, audioDeviceModule->StopPlayout());
>+  XCTAssertFalse(audioDeviceModule->PlayoutIsInitialized());
>+}
>+
>+// Verify that we can create two ADMs and start playing on the second ADM.
>+// Only the first active instance shall activate an audio session and the
>+// last active instance shall deactivate the audio session. The test does not
>+// explicitly verify correct audio session calls but instead focuses on
>+// ensuring that audio starts for both ADMs.
>+// Failing when running on real iOS devices: bugs.webrtc.org/6889.
>+- (void)DISABLED_testStartPlayoutOnTwoInstances {
>+  // Create and initialize a second/extra ADM instance. The default ADM is
>+  // created by the test harness.
>+  rtc::scoped_refptr<webrtc::AudioDeviceModule> secondAudioDeviceModule =
>+      webrtc::CreateAudioDeviceModule();
>+  XCTAssertNotEqual(secondAudioDeviceModule.get(), nullptr);
>+  XCTAssertEqual(0, secondAudioDeviceModule->Init());
>+
>+  // Start playout for the default ADM but don't wait here. Instead use the
>+  // upcoming second stream for that. We set the same expectation on number
>+  // of callbacks as for the second stream.
>+  mock.expectNeedMorePlayData(^int32_t(const size_t nSamples,
>+                                       const size_t nBytesPerSample,
>+                                       const size_t nChannels,
>+                                       const uint32_t samplesPerSec,
>+                                       void *audioSamples,
>+                                       size_t &nSamplesOut,
>+                                       int64_t *elapsed_time_ms,
>+                                       int64_t *ntp_time_ms) {
>+    nSamplesOut = nSamples;
>+    XCTAssertEqual(nSamples, playoutParameters.frames_per_10ms_buffer());
>+    XCTAssertEqual(nBytesPerSample, kBytesPerSample);
>+    XCTAssertEqual(nChannels, playoutParameters.channels());
>+    XCTAssertEqual((int) samplesPerSec, playoutParameters.sample_rate());
>+    XCTAssertNotEqual((void*)NULL, audioSamples);
>+
>+    return 0;
>+  });
>+
>+  XCTAssertEqual(0, audioDeviceModule->RegisterAudioCallback(&mock));
>+  [self startPlayout];
>+
>+  // Initialize playout for the second ADM. If all is OK, the second ADM shall
>+  // reuse the audio session activated when the first ADM started playing.
>+  // This call will also ensure that we avoid a problem related to initializing
>+  // two different audio unit instances back to back (see webrtc:5166 for
>+  // details).
>+  XCTAssertEqual(0, secondAudioDeviceModule->InitPlayout());
>+  XCTAssertTrue(secondAudioDeviceModule->PlayoutIsInitialized());
>+
>+  // Start playout for the second ADM and verify that it starts as intended.
>+  // Passing this test ensures that initialization of the second audio unit
>+  // has been done successfully and that there is no conflict with the already
>+  // playing first ADM.
>+  XCTestExpectation *playoutExpectation = [self expectationWithDescription:@"NeedMorePlayoutData"];
>+  playoutExpectation.expectedFulfillmentCount = kNumCallbacks;
>+
>+  MockAudioTransport mock2;
>+  mock2.expectNeedMorePlayData(^int32_t(const size_t nSamples,
>+                                        const size_t nBytesPerSample,
>+                                        const size_t nChannels,
>+                                        const uint32_t samplesPerSec,
>+                                        void *audioSamples,
>+                                        size_t &nSamplesOut,
>+                                        int64_t *elapsed_time_ms,
>+                                        int64_t *ntp_time_ms) {
>+    nSamplesOut = nSamples;
>+    XCTAssertEqual(nSamples, playoutParameters.frames_per_10ms_buffer());
>+    XCTAssertEqual(nBytesPerSample, kBytesPerSample);
>+    XCTAssertEqual(nChannels, playoutParameters.channels());
>+    XCTAssertEqual((int) samplesPerSec, playoutParameters.sample_rate());
>+    XCTAssertNotEqual((void*)NULL, audioSamples);
>+    [playoutExpectation fulfill];
>+
>+    return 0;
>+  });
>+
>+  XCTAssertEqual(0, secondAudioDeviceModule->RegisterAudioCallback(&mock2));
>+  XCTAssertEqual(0, secondAudioDeviceModule->StartPlayout());
>+  XCTAssertTrue(secondAudioDeviceModule->Playing());
>+  [self waitForExpectationsWithTimeout:kTestTimeOutInSec handler:nil];
>+  XCTAssertEqual(0, secondAudioDeviceModule->StopPlayout());
>+  XCTAssertFalse(secondAudioDeviceModule->Playing());
>+  XCTAssertFalse(secondAudioDeviceModule->PlayoutIsInitialized());
>+
>+  XCTAssertEqual(0, secondAudioDeviceModule->Terminate());
>+}
>+
>+// Start playout and verify that the native audio layer starts asking for real
>+// audio samples to play out using the NeedMorePlayData callback.
>+- (void)testStartPlayoutVerifyCallbacks {
>+
>+  XCTestExpectation *playoutExpectation = [self expectationWithDescription:@"NeedMorePlayoutData"];
>+  playoutExpectation.expectedFulfillmentCount = kNumCallbacks;
>+
>+  mock.expectNeedMorePlayData(^int32_t(const size_t nSamples,
>+                                       const size_t nBytesPerSample,
>+                                       const size_t nChannels,
>+                                       const uint32_t samplesPerSec,
>+                                       void *audioSamples,
>+                                       size_t &nSamplesOut,
>+                                       int64_t *elapsed_time_ms,
>+                                       int64_t *ntp_time_ms) {
>+    nSamplesOut = nSamples;
>+    XCTAssertEqual(nSamples, playoutParameters.frames_per_10ms_buffer());
>+    XCTAssertEqual(nBytesPerSample, kBytesPerSample);
>+    XCTAssertEqual(nChannels, playoutParameters.channels());
>+    XCTAssertEqual((int) samplesPerSec, playoutParameters.sample_rate());
>+    XCTAssertNotEqual((void*)NULL, audioSamples);
>+    [playoutExpectation fulfill];
>+
>+    return 0;
>+  });
>+
>+  XCTAssertEqual(0, audioDeviceModule->RegisterAudioCallback(&mock));
>+
>+  [self startPlayout];
>+  [self waitForExpectationsWithTimeout:kTestTimeOutInSec handler:nil];
>+  [self stopPlayout];
>+}
>+
>+// Start recording and verify that the native audio layer starts feeding real
>+// audio samples via the RecordedDataIsAvailable callback.
>+- (void)testStartRecordingVerifyCallbacks {
>+  XCTestExpectation *recordExpectation =
>+  [self expectationWithDescription:@"RecordedDataIsAvailable"];
>+  recordExpectation.expectedFulfillmentCount = kNumCallbacks;
>+
>+  mock.expectRecordedDataIsAvailable(^(const void* audioSamples,
>+                                       const size_t nSamples,
>+                                       const size_t nBytesPerSample,
>+                                       const size_t nChannels,
>+                                       const uint32_t samplesPerSec,
>+                                       const uint32_t totalDelayMS,
>+                                       const int32_t clockDrift,
>+                                       const uint32_t currentMicLevel,
>+                                       const bool keyPressed,
>+                                       uint32_t& newMicLevel) {
>+    XCTAssertNotEqual((void*)NULL, audioSamples);
>+    XCTAssertEqual(nSamples, recordParameters.frames_per_10ms_buffer());
>+    XCTAssertEqual(nBytesPerSample, kBytesPerSample);
>+    XCTAssertEqual(nChannels, recordParameters.channels());
>+    XCTAssertEqual((int) samplesPerSec, recordParameters.sample_rate());
>+    XCTAssertEqual(0, clockDrift);
>+    XCTAssertEqual(0u, currentMicLevel);
>+    XCTAssertFalse(keyPressed);
>+    [recordExpectation fulfill];
>+
>+    return 0;
>+  });
>+
>+  XCTAssertEqual(0, audioDeviceModule->RegisterAudioCallback(&mock));
>+  [self startRecording];
>+  [self waitForExpectationsWithTimeout:kTestTimeOutInSec handler:nil];
>+  [self stopRecording];
>+}
>+
>+// Start playout and recording (full-duplex audio) and verify that audio is
>+// active in both directions.
>+- (void)testStartPlayoutAndRecordingVerifyCallbacks {
>+  XCTestExpectation *playoutExpectation = [self expectationWithDescription:@"NeedMorePlayoutData"];
>+  __block NSUInteger callbackCount = 0;
>+
>+  XCTestExpectation *recordExpectation =
>+  [self expectationWithDescription:@"RecordedDataIsAvailable"];
>+  recordExpectation.expectedFulfillmentCount = kNumCallbacks;
>+
>+  mock.expectNeedMorePlayData(^int32_t(const size_t nSamples,
>+                                       const size_t nBytesPerSample,
>+                                       const size_t nChannels,
>+                                       const uint32_t samplesPerSec,
>+                                       void *audioSamples,
>+                                       size_t &nSamplesOut,
>+                                       int64_t *elapsed_time_ms,
>+                                       int64_t *ntp_time_ms) {
>+    nSamplesOut = nSamples;
>+    XCTAssertEqual(nSamples, playoutParameters.frames_per_10ms_buffer());
>+    XCTAssertEqual(nBytesPerSample, kBytesPerSample);
>+    XCTAssertEqual(nChannels, playoutParameters.channels());
>+    XCTAssertEqual((int) samplesPerSec, playoutParameters.sample_rate());
>+    XCTAssertNotEqual((void*)NULL, audioSamples);
>+    if (callbackCount++ >= kNumCallbacks) {
>+      [playoutExpectation fulfill];
>+    }
>+
>+    return 0;
>+  });
>+
>+  mock.expectRecordedDataIsAvailable(^(const void* audioSamples,
>+                                       const size_t nSamples,
>+                                       const size_t nBytesPerSample,
>+                                       const size_t nChannels,
>+                                       const uint32_t samplesPerSec,
>+                                       const uint32_t totalDelayMS,
>+                                       const int32_t clockDrift,
>+                                       const uint32_t currentMicLevel,
>+                                       const bool keyPressed,
>+                                       uint32_t& newMicLevel) {
>+    XCTAssertNotEqual((void*)NULL, audioSamples);
>+    XCTAssertEqual(nSamples, recordParameters.frames_per_10ms_buffer());
>+    XCTAssertEqual(nBytesPerSample, kBytesPerSample);
>+    XCTAssertEqual(nChannels, recordParameters.channels());
>+    XCTAssertEqual((int) samplesPerSec, recordParameters.sample_rate());
>+    XCTAssertEqual(0, clockDrift);
>+    XCTAssertEqual(0u, currentMicLevel);
>+    XCTAssertFalse(keyPressed);
>+    [recordExpectation fulfill];
>+
>+    return 0;
>+  });
>+
>+  XCTAssertEqual(0, audioDeviceModule->RegisterAudioCallback(&mock));
>+  [self startPlayout];
>+  [self startRecording];
>+  [self waitForExpectationsWithTimeout:kTestTimeOutInSec handler:nil];
>+  [self stopRecording];
>+  [self stopPlayout];
>+}
>+
>+// Start playout and read audio from an external PCM file when the audio layer
>+// asks for data to play out. Real audio is played out in this test but it does
>+// not contain any explicit verification that the audio quality is perfect.
>+- (void)testRunPlayoutWithFileAsSource {
>+  XCTAssertEqual(1u, playoutParameters.channels());
>+
>+  // Using XCTestExpectation to count callbacks is very slow.
>+  XCTestExpectation *playoutExpectation = [self expectationWithDescription:@"NeedMorePlayoutData"];
>+  const int expectedCallbackCount = kFilePlayTimeInSec * kNumCallbacksPerSecond;
>+  __block int callbackCount = 0;
>+
>+  NSURL *fileURL = [self fileURLForSampleRate:playoutParameters.sample_rate()];
>+  NSInputStream *inputStream = [[NSInputStream alloc] initWithURL:fileURL];
>+
>+  mock.expectNeedMorePlayData(^int32_t(const size_t nSamples,
>+                                       const size_t nBytesPerSample,
>+                                       const size_t nChannels,
>+                                       const uint32_t samplesPerSec,
>+                                       void *audioSamples,
>+                                       size_t &nSamplesOut,
>+                                       int64_t *elapsed_time_ms,
>+                                       int64_t *ntp_time_ms) {
>+    [inputStream read:(uint8_t *)audioSamples maxLength:nSamples*nBytesPerSample*nChannels];
>+    nSamplesOut = nSamples;
>+    if (callbackCount++ == expectedCallbackCount) {
>+      [playoutExpectation fulfill];
>+    }
>+
>+    return 0;
>+  });
>+
>+  XCTAssertEqual(0, audioDeviceModule->RegisterAudioCallback(&mock));
>+  [self startPlayout];
>+  NSTimeInterval waitTimeout = kFilePlayTimeInSec * 2.0;
>+  [self waitForExpectationsWithTimeout:waitTimeout handler:nil];
>+  [self stopPlayout];
>+}
>+
>+- (void)testDevices {
>+  // Device enumeration is not supported. Verify fixed values only.
>+  XCTAssertEqual(1, audioDeviceModule->PlayoutDevices());
>+  XCTAssertEqual(1, audioDeviceModule->RecordingDevices());
>+}
>+
>+// Start playout and recording and store recorded data in an intermediate FIFO
>+// buffer from which the playout side then reads its samples in the same order
>+// as they were stored. Under ideal circumstances, a callback sequence would
>+// look like: ...+-+-+-+-+-+-+-..., where '+' means 'packet recorded' and '-'
>+// means 'packet played'. Under such conditions, the FIFO would only contain
>+// one packet on average. However, under more realistic conditions, the size
>+// of the FIFO will vary more due to an unbalance between the two sides.
>+// This test tries to verify that the device maintains a balanced callback-
>+// sequence by running in loopback for ten seconds while measuring the size
>+// (max and average) of the FIFO. The size of the FIFO is increased by the
>+// recording side and decreased by the playout side.
>+// TODO(henrika): tune the final test parameters after running tests on several
>+// different devices.
>+- (void)testRunPlayoutAndRecordingInFullDuplex {
>+  XCTAssertEqual(recordParameters.channels(), playoutParameters.channels());
>+  XCTAssertEqual(recordParameters.sample_rate(), playoutParameters.sample_rate());
>+
>+  XCTestExpectation *playoutExpectation = [self expectationWithDescription:@"NeedMorePlayoutData"];
>+  __block NSUInteger playoutCallbacks = 0;
>+  NSUInteger expectedPlayoutCallbacks = kFullDuplexTimeInSec * kNumCallbacksPerSecond;
>+
>+  // FIFO queue and measurements
>+  NSMutableArray *fifoBuffer = [NSMutableArray arrayWithCapacity:20];
>+  __block NSUInteger fifoMaxSize = 0;
>+  __block NSUInteger fifoTotalWrittenElements = 0;
>+  __block NSUInteger fifoWriteCount = 0;
>+
>+  mock.expectRecordedDataIsAvailable(^(const void* audioSamples,
>+                                       const size_t nSamples,
>+                                       const size_t nBytesPerSample,
>+                                       const size_t nChannels,
>+                                       const uint32_t samplesPerSec,
>+                                       const uint32_t totalDelayMS,
>+                                       const int32_t clockDrift,
>+                                       const uint32_t currentMicLevel,
>+                                       const bool keyPressed,
>+                                       uint32_t& newMicLevel) {
>+    if (fifoWriteCount++ < kNumIgnoreFirstCallbacks) {
>+      return 0;
>+    }
>+
>+    NSData *data = [NSData dataWithBytes:audioSamples length:nSamples*nBytesPerSample*nChannels];
>+    @synchronized(fifoBuffer) {
>+      [fifoBuffer addObject:data];
>+      fifoMaxSize = MAX(fifoMaxSize, fifoBuffer.count);
>+      fifoTotalWrittenElements += fifoBuffer.count;
>+    }
>+
>+    return 0;
>+  });
>+
>+  mock.expectNeedMorePlayData(^int32_t(const size_t nSamples,
>+                                       const size_t nBytesPerSample,
>+                                       const size_t nChannels,
>+                                       const uint32_t samplesPerSec,
>+                                       void *audioSamples,
>+                                       size_t &nSamplesOut,
>+                                       int64_t *elapsed_time_ms,
>+                                       int64_t *ntp_time_ms) {
>+    nSamplesOut = nSamples;
>+    NSData *data;
>+    @synchronized(fifoBuffer) {
>+      data = fifoBuffer.firstObject;
>+      if (data) {
>+        [fifoBuffer removeObjectAtIndex:0];
>+      }
>+    }
>+
>+    if (data) {
>+      memcpy(audioSamples, (char*) data.bytes, data.length);
>+    } else {
>+      memset(audioSamples, 0, nSamples*nBytesPerSample*nChannels);
>+    }
>+
>+    if (playoutCallbacks++ == expectedPlayoutCallbacks) {
>+      [playoutExpectation fulfill];
>+    }
>+    return 0;
>+  });
>+
>+  XCTAssertEqual(0, audioDeviceModule->RegisterAudioCallback(&mock));
>+  [self startRecording];
>+  [self startPlayout];
>+  NSTimeInterval waitTimeout = kFullDuplexTimeInSec * 2.0;
>+  [self waitForExpectationsWithTimeout:waitTimeout handler:nil];
>+
>+  size_t fifoAverageSize =
>+      (fifoTotalWrittenElements == 0)
>+        ? 0.0
>+        : 0.5 + (double)fifoTotalWrittenElements / (fifoWriteCount - kNumIgnoreFirstCallbacks);
>+
>+  [self stopPlayout];
>+  [self stopRecording];
>+  XCTAssertLessThan(fifoAverageSize, 10u);
>+  XCTAssertLessThan(fifoMaxSize, 20u);
>+}
>+
>+@end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/RTCAudioDevice_xctest.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/RTCAudioDevice_xctest.mm
>new file mode 100644
>index 00000000000..fa99d9038de
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/RTCAudioDevice_xctest.mm
>@@ -0,0 +1,112 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#import <XCTest/XCTest.h>
>+#import "sdk/objc/Framework/Native/src/audio/audio_device_ios.h"
>+#import "sdk/objc/Framework/Native/api/audio_device_module.h"
>+#import "sdk/objc/Framework/Classes/Audio/RTCAudioSession+Private.h"
>+
>+@interface RTCAudioDeviceTests: XCTestCase {
>+  rtc::scoped_refptr<webrtc::AudioDeviceModule> _audioDeviceModule;
>+  std::unique_ptr<webrtc::ios_adm::AudioDeviceIOS> _audio_device;
>+}
>+
>+@property(nonatomic) RTCAudioSession *audioSession;
>+
>+@end
>+
>+@implementation RTCAudioDeviceTests
>+
>+@synthesize audioSession = _audioSession;
>+
>+- (void)setUp {
>+  [super setUp];
>+
>+  _audioDeviceModule = webrtc::CreateAudioDeviceModule();
>+  _audio_device.reset(new webrtc::ios_adm::AudioDeviceIOS());
>+  self.audioSession = [RTCAudioSession sharedInstance];
>+
>+  NSError *error = nil;
>+  [self.audioSession lockForConfiguration];
>+  [self.audioSession setCategory:AVAudioSessionCategoryPlayAndRecord
>+                     withOptions:0
>+                           error:&error];
>+  XCTAssertNil(error);
>+
>+  [self.audioSession setMode:AVAudioSessionModeVoiceChat error:&error];
>+  XCTAssertNil(error);
>+
>+  [self.audioSession setActive:YES error:&error];
>+  XCTAssertNil(error);
>+
>+  [self.audioSession unlockForConfiguration];
>+}
>+
>+- (void)tearDown {
>+  _audio_device->Terminate();
>+  _audio_device.reset(nullptr);
>+  _audioDeviceModule = nullptr;
>+  [self.audioSession notifyDidEndInterruptionWithShouldResumeSession:NO];
>+
>+  [super tearDown];
>+}
>+
>+// Verifies that the AudioDeviceIOS is_interrupted_ flag is reset correctly
>+// after an iOS AVAudioSessionInterruptionTypeEnded notification event.
>+// AudioDeviceIOS listens to RTCAudioSession interrupted notifications by:
>+// - In AudioDeviceIOS.InitPlayOrRecord registers its audio_session_observer_
>+//   callback with RTCAudioSession's delegate list.
>+// - When RTCAudioSession receives an iOS audio interrupted notification, it
>+//   passes the notification to callbacks in its delegate list which sets
>+//   AudioDeviceIOS's is_interrupted_ flag to true.
>+// - When AudioDeviceIOS.ShutdownPlayOrRecord is called, its
>+//   audio_session_observer_ callback is removed from RTCAudioSessions's
>+//   delegate list.
>+//   So if RTCAudioSession receives an iOS end audio interruption notification,
>+//   AudioDeviceIOS is not notified as its callback is not in RTCAudioSession's
>+//   delegate list. This causes AudioDeviceIOS's is_interrupted_ flag to be in
>+//   the wrong (true) state and the audio session will ignore audio changes.
>+// As RTCAudioSession keeps its own interrupted state, the fix is to initialize
>+// AudioDeviceIOS's is_interrupted_ flag to RTCAudioSession's isInterrupted
>+// flag in AudioDeviceIOS.InitPlayOrRecord.
>+- (void)testInterruptedAudioSession {
>+  XCTAssertTrue(self.audioSession.isActive);
>+  XCTAssertTrue([self.audioSession.category isEqual:AVAudioSessionCategoryPlayAndRecord] ||
>+                [self.audioSession.category isEqual:AVAudioSessionCategoryPlayback]);
>+  XCTAssertEqual(AVAudioSessionModeVoiceChat, self.audioSession.mode);
>+
>+  std::unique_ptr<webrtc::AudioDeviceBuffer> audio_buffer;
>+  audio_buffer.reset(new webrtc::AudioDeviceBuffer());
>+  _audio_device->AttachAudioBuffer(audio_buffer.get());
>+  XCTAssertEqual(webrtc::AudioDeviceGeneric::InitStatus::OK, _audio_device->Init());
>+  XCTAssertEqual(0, _audio_device->InitPlayout());
>+  XCTAssertEqual(0, _audio_device->StartPlayout());
>+
>+  // Force interruption.
>+  [self.audioSession notifyDidBeginInterruption];
>+
>+  // Wait for notification to propagate.
>+  rtc::MessageQueueManager::ProcessAllMessageQueues();
>+  XCTAssertTrue(_audio_device->IsInterrupted());
>+
>+  // Force it for testing.
>+  _audio_device->StopPlayout();
>+
>+  [self.audioSession notifyDidEndInterruptionWithShouldResumeSession:YES];
>+  // Wait for notification to propagate.
>+  rtc::MessageQueueManager::ProcessAllMessageQueues();
>+  XCTAssertTrue(_audio_device->IsInterrupted());
>+
>+  _audio_device->Init();
>+  _audio_device->InitPlayout();
>+  XCTAssertFalse(_audio_device->IsInterrupted());
>+}
>+
>+@end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/RTCAudioSessionTest.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/RTCAudioSessionTest.mm
>index 951b5ea2600..ee52242ecfa 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/RTCAudioSessionTest.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/RTCAudioSessionTest.mm
>@@ -299,7 +299,7 @@ namespace webrtc {
> 
> class AudioSessionTest : public ::testing::Test {
>  protected:
>-  void TearDown() {
>+  void TearDown() override {
>     RTCAudioSession *session = [RTCAudioSession sharedInstance];
>     for (id<RTCAudioSessionDelegate> delegate : session.delegates) {
>       [session removeDelegate:delegate];
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/RTCCVPixelBuffer_xctest.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/RTCCVPixelBuffer_xctest.mm
>new file mode 100644
>index 00000000000..834bb222b99
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/RTCCVPixelBuffer_xctest.mm
>@@ -0,0 +1,296 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#import <Foundation/Foundation.h>
>+#import <XCTest/XCTest.h>
>+
>+#import "Video/RTCI420Buffer+Private.h"
>+#import "WebRTC/RTCVideoFrame.h"
>+#import "WebRTC/RTCVideoFrameBuffer.h"
>+#include "common_video/libyuv/include/webrtc_libyuv.h"
>+#import "sdk/objc/Framework/UnitTests/frame_buffer_helpers.h"
>+#include "third_party/libyuv/include/libyuv.h"
>+
>+@interface RTCCVPixelBufferTests : XCTestCase
>+@end
>+
>+@implementation RTCCVPixelBufferTests {
>+}
>+
>+- (void)testRequiresCroppingNoCrop {
>+  CVPixelBufferRef pixelBufferRef = NULL;
>+  CVPixelBufferCreate(
>+      NULL, 720, 1280, kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange, NULL, &pixelBufferRef);
>+  RTCCVPixelBuffer *buffer = [[RTCCVPixelBuffer alloc] initWithPixelBuffer:pixelBufferRef];
>+
>+  XCTAssertFalse([buffer requiresCropping]);
>+
>+  CVBufferRelease(pixelBufferRef);
>+}
>+
>+- (void)testRequiresCroppingWithCrop {
>+  CVPixelBufferRef pixelBufferRef = NULL;
>+  CVPixelBufferCreate(
>+      NULL, 720, 1280, kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange, NULL, &pixelBufferRef);
>+  RTCCVPixelBuffer *croppedBuffer = [[RTCCVPixelBuffer alloc] initWithPixelBuffer:pixelBufferRef
>+                                                                     adaptedWidth:720
>+                                                                    adaptedHeight:1280
>+                                                                        cropWidth:360
>+                                                                       cropHeight:640
>+                                                                            cropX:100
>+                                                                            cropY:100];
>+
>+  XCTAssertTrue([croppedBuffer requiresCropping]);
>+
>+  CVBufferRelease(pixelBufferRef);
>+}
>+
>+- (void)testRequiresScalingNoScale {
>+  CVPixelBufferRef pixelBufferRef = NULL;
>+  CVPixelBufferCreate(
>+      NULL, 720, 1280, kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange, NULL, &pixelBufferRef);
>+
>+  RTCCVPixelBuffer *buffer = [[RTCCVPixelBuffer alloc] initWithPixelBuffer:pixelBufferRef];
>+  XCTAssertFalse([buffer requiresScalingToWidth:720 height:1280]);
>+
>+  CVBufferRelease(pixelBufferRef);
>+}
>+
>+- (void)testRequiresScalingWithScale {
>+  CVPixelBufferRef pixelBufferRef = NULL;
>+  CVPixelBufferCreate(
>+      NULL, 720, 1280, kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange, NULL, &pixelBufferRef);
>+
>+  RTCCVPixelBuffer *buffer = [[RTCCVPixelBuffer alloc] initWithPixelBuffer:pixelBufferRef];
>+  XCTAssertTrue([buffer requiresScalingToWidth:360 height:640]);
>+
>+  CVBufferRelease(pixelBufferRef);
>+}
>+
>+- (void)testRequiresScalingWithScaleAndMatchingCrop {
>+  CVPixelBufferRef pixelBufferRef = NULL;
>+  CVPixelBufferCreate(
>+      NULL, 720, 1280, kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange, NULL, &pixelBufferRef);
>+
>+  RTCCVPixelBuffer *buffer = [[RTCCVPixelBuffer alloc] initWithPixelBuffer:pixelBufferRef
>+                                                              adaptedWidth:720
>+                                                             adaptedHeight:1280
>+                                                                 cropWidth:360
>+                                                                cropHeight:640
>+                                                                     cropX:100
>+                                                                     cropY:100];
>+  XCTAssertFalse([buffer requiresScalingToWidth:360 height:640]);
>+
>+  CVBufferRelease(pixelBufferRef);
>+}
>+
>+- (void)testBufferSize_NV12 {
>+  CVPixelBufferRef pixelBufferRef = NULL;
>+  CVPixelBufferCreate(
>+      NULL, 720, 1280, kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange, NULL, &pixelBufferRef);
>+
>+  RTCCVPixelBuffer *buffer = [[RTCCVPixelBuffer alloc] initWithPixelBuffer:pixelBufferRef];
>+  XCTAssertEqual([buffer bufferSizeForCroppingAndScalingToWidth:360 height:640], 576000);
>+
>+  CVBufferRelease(pixelBufferRef);
>+}
>+
>+- (void)testBufferSize_RGB {
>+  CVPixelBufferRef pixelBufferRef = NULL;
>+  CVPixelBufferCreate(NULL, 720, 1280, kCVPixelFormatType_32BGRA, NULL, &pixelBufferRef);
>+
>+  RTCCVPixelBuffer *buffer = [[RTCCVPixelBuffer alloc] initWithPixelBuffer:pixelBufferRef];
>+  XCTAssertEqual([buffer bufferSizeForCroppingAndScalingToWidth:360 height:640], 0);
>+
>+  CVBufferRelease(pixelBufferRef);
>+}
>+
>+- (void)testCropAndScale_NV12 {
>+  [self cropAndScaleTestWithNV12];
>+}
>+
>+- (void)testCropAndScaleNoOp_NV12 {
>+  [self cropAndScaleTestWithNV12InputFormat:kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange
>+                               outputFormat:kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange
>+                                 outputSize:CGSizeMake(720, 1280)];
>+}
>+
>+- (void)testCropAndScale_NV12FullToVideo {
>+  [self cropAndScaleTestWithNV12InputFormat:kCVPixelFormatType_420YpCbCr8BiPlanarFullRange
>+                               outputFormat:kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange];
>+}
>+
>+- (void)testCropAndScaleZeroSizeFrame_NV12 {
>+  [self cropAndScaleTestWithNV12InputFormat:kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange
>+                               outputFormat:kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange
>+                                 outputSize:CGSizeMake(0, 0)];
>+}
>+
>+- (void)testCropAndScaleToSmallFormat_NV12 {
>+  [self cropAndScaleTestWithNV12InputFormat:kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange
>+                               outputFormat:kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange
>+                                 outputSize:CGSizeMake(148, 320)];
>+}
>+
>+- (void)testCropAndScaleToOddFormat_NV12 {
>+  [self cropAndScaleTestWithNV12InputFormat:kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange
>+                               outputFormat:kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange
>+                                 outputSize:CGSizeMake(361, 640)];
>+}
>+
>+- (void)testCropAndScale_32BGRA {
>+  [self cropAndScaleTestWithRGBPixelFormat:kCVPixelFormatType_32BGRA];
>+}
>+
>+- (void)testCropAndScale_32ARGB {
>+  [self cropAndScaleTestWithRGBPixelFormat:kCVPixelFormatType_32ARGB];
>+}
>+
>+- (void)testCropAndScaleWithSmallCropInfo_32ARGB {
>+  [self cropAndScaleTestWithRGBPixelFormat:kCVPixelFormatType_32ARGB cropX:2 cropY:3];
>+}
>+
>+- (void)testCropAndScaleWithLargeCropInfo_32ARGB {
>+  [self cropAndScaleTestWithRGBPixelFormat:kCVPixelFormatType_32ARGB cropX:200 cropY:300];
>+}
>+
>+- (void)testToI420_NV12 {
>+  [self toI420WithPixelFormat:kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange];
>+}
>+
>+- (void)testToI420_32BGRA {
>+  [self toI420WithPixelFormat:kCVPixelFormatType_32BGRA];
>+}
>+
>+- (void)testToI420_32ARGB {
>+  [self toI420WithPixelFormat:kCVPixelFormatType_32ARGB];
>+}
>+
>+#pragma mark - Shared test code
>+
>+- (void)cropAndScaleTestWithNV12 {
>+  [self cropAndScaleTestWithNV12InputFormat:kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange
>+                               outputFormat:kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange];
>+}
>+
>+- (void)cropAndScaleTestWithNV12InputFormat:(OSType)inputFormat outputFormat:(OSType)outputFormat {
>+  [self cropAndScaleTestWithNV12InputFormat:(OSType)inputFormat
>+                               outputFormat:(OSType)outputFormat
>+                                 outputSize:CGSizeMake(360, 640)];
>+}
>+
>+- (void)cropAndScaleTestWithNV12InputFormat:(OSType)inputFormat
>+                               outputFormat:(OSType)outputFormat
>+                                 outputSize:(CGSize)outputSize {
>+  CVPixelBufferRef pixelBufferRef = NULL;
>+  CVPixelBufferCreate(NULL, 720, 1280, inputFormat, NULL, &pixelBufferRef);
>+
>+  rtc::scoped_refptr<webrtc::I420Buffer> i420Buffer = CreateI420Gradient(720, 1280);
>+  CopyI420BufferToCVPixelBuffer(i420Buffer, pixelBufferRef);
>+
>+  RTCCVPixelBuffer *buffer = [[RTCCVPixelBuffer alloc] initWithPixelBuffer:pixelBufferRef];
>+  XCTAssertEqual(buffer.width, 720);
>+  XCTAssertEqual(buffer.height, 1280);
>+
>+  CVPixelBufferRef outputPixelBufferRef = NULL;
>+  CVPixelBufferCreate(
>+      NULL, outputSize.width, outputSize.height, outputFormat, NULL, &outputPixelBufferRef);
>+
>+  std::vector<uint8_t> frameScaleBuffer;
>+  if ([buffer requiresScalingToWidth:outputSize.width height:outputSize.height]) {
>+    int size =
>+        [buffer bufferSizeForCroppingAndScalingToWidth:outputSize.width height:outputSize.height];
>+    frameScaleBuffer.resize(size);
>+  } else {
>+    frameScaleBuffer.clear();
>+  }
>+  frameScaleBuffer.shrink_to_fit();
>+
>+  [buffer cropAndScaleTo:outputPixelBufferRef withTempBuffer:frameScaleBuffer.data()];
>+
>+  RTCCVPixelBuffer *scaledBuffer =
>+      [[RTCCVPixelBuffer alloc] initWithPixelBuffer:outputPixelBufferRef];
>+  XCTAssertEqual(scaledBuffer.width, outputSize.width);
>+  XCTAssertEqual(scaledBuffer.height, outputSize.height);
>+
>+  if (outputSize.width > 0 && outputSize.height > 0) {
>+    RTCI420Buffer *originalBufferI420 = [buffer toI420];
>+    RTCI420Buffer *scaledBufferI420 = [scaledBuffer toI420];
>+    double psnr =
>+        I420PSNR(*[originalBufferI420 nativeI420Buffer], *[scaledBufferI420 nativeI420Buffer]);
>+    XCTAssertEqual(psnr, webrtc::kPerfectPSNR);
>+  }
>+
>+  CVBufferRelease(pixelBufferRef);
>+}
>+
>+- (void)cropAndScaleTestWithRGBPixelFormat:(OSType)pixelFormat {
>+  [self cropAndScaleTestWithRGBPixelFormat:pixelFormat cropX:0 cropY:0];
>+}
>+
>+- (void)cropAndScaleTestWithRGBPixelFormat:(OSType)pixelFormat cropX:(int)cropX cropY:(int)cropY {
>+  CVPixelBufferRef pixelBufferRef = NULL;
>+  CVPixelBufferCreate(NULL, 720, 1280, pixelFormat, NULL, &pixelBufferRef);
>+
>+  DrawGradientInRGBPixelBuffer(pixelBufferRef);
>+
>+  RTCCVPixelBuffer *buffer =
>+      [[RTCCVPixelBuffer alloc] initWithPixelBuffer:pixelBufferRef
>+                                       adaptedWidth:CVPixelBufferGetWidth(pixelBufferRef)
>+                                      adaptedHeight:CVPixelBufferGetHeight(pixelBufferRef)
>+                                          cropWidth:CVPixelBufferGetWidth(pixelBufferRef) - cropX
>+                                         cropHeight:CVPixelBufferGetHeight(pixelBufferRef) - cropY
>+                                              cropX:cropX
>+                                              cropY:cropY];
>+
>+  XCTAssertEqual(buffer.width, 720);
>+  XCTAssertEqual(buffer.height, 1280);
>+
>+  CVPixelBufferRef outputPixelBufferRef = NULL;
>+  CVPixelBufferCreate(NULL, 360, 640, pixelFormat, NULL, &outputPixelBufferRef);
>+  [buffer cropAndScaleTo:outputPixelBufferRef withTempBuffer:NULL];
>+
>+  RTCCVPixelBuffer *scaledBuffer =
>+      [[RTCCVPixelBuffer alloc] initWithPixelBuffer:outputPixelBufferRef];
>+  XCTAssertEqual(scaledBuffer.width, 360);
>+  XCTAssertEqual(scaledBuffer.height, 640);
>+
>+  RTCI420Buffer *originalBufferI420 = [buffer toI420];
>+  RTCI420Buffer *scaledBufferI420 = [scaledBuffer toI420];
>+  double psnr =
>+      I420PSNR(*[originalBufferI420 nativeI420Buffer], *[scaledBufferI420 nativeI420Buffer]);
>+  XCTAssertEqual(psnr, webrtc::kPerfectPSNR);
>+
>+  CVBufferRelease(pixelBufferRef);
>+}
>+
>+- (void)toI420WithPixelFormat:(OSType)pixelFormat {
>+  rtc::scoped_refptr<webrtc::I420Buffer> i420Buffer = CreateI420Gradient(360, 640);
>+
>+  CVPixelBufferRef pixelBufferRef = NULL;
>+  CVPixelBufferCreate(NULL, 360, 640, pixelFormat, NULL, &pixelBufferRef);
>+
>+  CopyI420BufferToCVPixelBuffer(i420Buffer, pixelBufferRef);
>+
>+  RTCCVPixelBuffer *buffer = [[RTCCVPixelBuffer alloc] initWithPixelBuffer:pixelBufferRef];
>+  RTCI420Buffer *fromCVPixelBuffer = [buffer toI420];
>+
>+  double psnr = I420PSNR(*i420Buffer, *[fromCVPixelBuffer nativeI420Buffer]);
>+  double target = webrtc::kPerfectPSNR;
>+  if (pixelFormat != kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange) {
>+    // libyuv's I420ToRGB functions seem to lose some quality.
>+    target = 19.0;
>+  }
>+  XCTAssertGreaterThanOrEqual(psnr, target);
>+
>+  CVBufferRelease(pixelBufferRef);
>+}
>+
>+@end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/RTCCallbackLogger_xctest.m b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/RTCCallbackLogger_xctest.m
>new file mode 100644
>index 00000000000..4cc9f157452
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/RTCCallbackLogger_xctest.m
>@@ -0,0 +1,107 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#import "WebRTC/RTCCallbackLogger.h"
>+
>+#import <XCTest/XCTest.h>
>+
>+@interface RTCCallbackLoggerTests : XCTestCase
>+
>+@property(nonatomic, strong) RTCCallbackLogger *logger;
>+
>+@end
>+
>+@implementation RTCCallbackLoggerTests
>+
>+@synthesize logger;
>+
>+- (void)setUp {
>+  self.logger = [[RTCCallbackLogger alloc] init];
>+}
>+
>+- (void)tearDown {
>+  self.logger = nil;
>+}
>+
>+- (void)testDefaultSeverityLevel {
>+  XCTAssertEqual(self.logger.severity, RTCLoggingSeverityInfo);
>+}
>+
>+- (void)testCallbackGetsCalledForAppropriateLevel {
>+  self.logger.severity = RTCLoggingSeverityWarning;
>+
>+  XCTestExpectation *callbackExpectation = [self expectationWithDescription:@"callbackWarning"];
>+
>+  [self.logger start:^(NSString *message) {
>+    XCTAssertTrue([message hasSuffix:@"Horrible error\n"]);
>+    [callbackExpectation fulfill];
>+  }];
>+
>+  RTCLogError("Horrible error");
>+
>+  [self waitForExpectations:@[ callbackExpectation ] timeout:10.0];
>+}
>+
>+- (void)testCallbackDoesNotGetCalledForOtherLevels {
>+  self.logger.severity = RTCLoggingSeverityError;
>+
>+  XCTestExpectation *callbackExpectation = [self expectationWithDescription:@"callbackError"];
>+
>+  [self.logger start:^(NSString *message) {
>+    XCTAssertTrue([message hasSuffix:@"Horrible error\n"]);
>+    [callbackExpectation fulfill];
>+  }];
>+
>+  RTCLogInfo("Just some info");
>+  RTCLogWarning("Warning warning");
>+  RTCLogError("Horrible error");
>+
>+  [self waitForExpectations:@[ callbackExpectation ] timeout:10.0];
>+}
>+
>+- (void)testStartingWithNilCallbackDoesNotCrash {
>+  [self.logger start:nil];
>+
>+  RTCLogError("Horrible error");
>+}
>+
>+- (void)testStopCallbackLogger {
>+  XCTestExpectation *callbackExpectation = [self expectationWithDescription:@"stopped"];
>+
>+  [self.logger start:^(NSString *message) {
>+    [callbackExpectation fulfill];
>+  }];
>+
>+  [self.logger stop];
>+
>+  RTCLogInfo("Just some info");
>+
>+  XCTWaiter *waiter = [[XCTWaiter alloc] init];
>+  XCTWaiterResult result = [waiter waitForExpectations:@[ callbackExpectation ] timeout:1.0];
>+  XCTAssertEqual(result, XCTWaiterResultTimedOut);
>+}
>+
>+- (void)testDestroyingCallbackLogger {
>+  XCTestExpectation *callbackExpectation = [self expectationWithDescription:@"destroyed"];
>+
>+  [self.logger start:^(NSString *message) {
>+    [callbackExpectation fulfill];
>+  }];
>+
>+  self.logger = nil;
>+
>+  RTCLogInfo("Just some info");
>+
>+  XCTWaiter *waiter = [[XCTWaiter alloc] init];
>+  XCTWaiterResult result = [waiter waitForExpectations:@[ callbackExpectation ] timeout:1.0];
>+  XCTAssertEqual(result, XCTWaiterResultTimedOut);
>+}
>+
>+@end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/RTCCameraVideoCapturerTests.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/RTCCameraVideoCapturerTests.mm
>index f543069c5ff..0af17c4414a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/RTCCameraVideoCapturerTests.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/RTCCameraVideoCapturerTests.mm
>@@ -60,20 +60,24 @@ CMSampleBufferRef createTestSampleBufferRef() {
> }
> #endif
> @interface RTCCameraVideoCapturer (Tests)<AVCaptureVideoDataOutputSampleBufferDelegate>
>+- (instancetype)initWithDelegate:(__weak id<RTCVideoCapturerDelegate>)delegate
>+                  captureSession:(AVCaptureSession *)captureSession;
> @end
> 
> @interface RTCCameraVideoCapturerTests : NSObject
> @property(nonatomic, strong) id delegateMock;
> @property(nonatomic, strong) id deviceMock;
> @property(nonatomic, strong) id captureConnectionMock;
>+@property(nonatomic, strong) id captureSessionMock;
> @property(nonatomic, strong) RTCCameraVideoCapturer *capturer;
> @end
> 
> @implementation RTCCameraVideoCapturerTests
> @synthesize delegateMock = _delegateMock;
>+@synthesize deviceMock = _deviceMock;
> @synthesize captureConnectionMock = _captureConnectionMock;
>+@synthesize captureSessionMock = _captureSessionMock;
> @synthesize capturer = _capturer;
>-@synthesize deviceMock = _deviceMock;
> 
> - (void)setup {
>   self.delegateMock = OCMProtocolMock(@protocol(RTCVideoCapturerDelegate));
>@@ -82,6 +86,21 @@ - (void)setup {
>   self.deviceMock = [self createDeviceMock];
> }
> 
>+- (void)setupWithMockedCaptureSession {
>+  self.captureSessionMock = OCMStrictClassMock([AVCaptureSession class]);
>+  OCMStub([self.captureSessionMock setSessionPreset:[OCMArg any]]);
>+  OCMStub([self.captureSessionMock setUsesApplicationAudioSession:NO]);
>+  OCMStub([self.captureSessionMock canAddOutput:[OCMArg any]]).andReturn(YES);
>+  OCMStub([self.captureSessionMock addOutput:[OCMArg any]]);
>+  OCMStub([self.captureSessionMock beginConfiguration]);
>+  OCMStub([self.captureSessionMock commitConfiguration]);
>+  self.delegateMock = OCMProtocolMock(@protocol(RTCVideoCapturerDelegate));
>+  self.captureConnectionMock = OCMClassMock([AVCaptureConnection class]);
>+  self.capturer = [[RTCCameraVideoCapturer alloc] initWithDelegate:self.delegateMock
>+                                                    captureSession:self.captureSessionMock];
>+  self.deviceMock = [self createDeviceMock];
>+}
>+
> - (void)tearDown {
>   [self.delegateMock stopMocking];
>   [self.deviceMock stopMocking];
>@@ -144,9 +163,11 @@ - (void)testSupportedFormatsForDevice {
>   NSArray *supportedFormats = [RTCCameraVideoCapturer supportedFormatsForDevice:self.deviceMock];
> 
>   // then
>-  EXPECT_EQ(supportedFormats.count, 2u);
>+  EXPECT_EQ(supportedFormats.count, 3u);
>   EXPECT_TRUE([supportedFormats containsObject:validFormat1]);
>   EXPECT_TRUE([supportedFormats containsObject:validFormat2]);
>+  EXPECT_TRUE([supportedFormats containsObject:invalidFormat]);
>+
>   // cleanup
>   [validFormat1 stopMocking];
>   [validFormat2 stopMocking];
>@@ -156,15 +177,6 @@ - (void)testSupportedFormatsForDevice {
>   invalidFormat = nil;
> }
> 
>-- (void)testCaptureDevices {
>-  OCMStub([self.deviceMock devicesWithMediaType:AVMediaTypeVideo]).andReturn(@[ [NSObject new] ]);
>-  OCMStub([self.deviceMock devicesWithMediaType:AVMediaTypeAudio]).andReturn(@[ [NSObject new] ]);
>-
>-  NSArray *captureDevices = [RTCCameraVideoCapturer captureDevices];
>-
>-  EXPECT_EQ(captureDevices.count, 1u);
>-}
>-
> - (void)testDelegateCallbackNotCalledWhenInvalidBuffer {
>   // given
>   CMSampleBufferRef sampleBuffer = nullptr;
>@@ -182,13 +194,7 @@ - (void)testDelegateCallbackNotCalledWhenInvalidBuffer {
> 
> - (void)testDelegateCallbackWithValidBufferAndOrientationUpdate {
> #if TARGET_OS_IPHONE
>-  // given
>-  UIDevice *currentDeviceMock = OCMClassMock([UIDevice class]);
>-  // UpsideDown -> RTCVideoRotation_270.
>-  OCMStub(currentDeviceMock.orientation).andReturn(UIDeviceOrientationPortraitUpsideDown);
>-  id classMock = OCMClassMock([UIDevice class]);
>-  OCMStub([classMock currentDevice]).andReturn(currentDeviceMock);
>-
>+  [UIDevice.currentDevice setValue:@(UIDeviceOrientationPortraitUpsideDown) forKey:@"orientation"];
>   CMSampleBufferRef sampleBuffer = createTestSampleBufferRef();
> 
>   // then
>@@ -210,11 +216,6 @@ - (void)testDelegateCallbackWithValidBufferAndOrientationUpdate {
>                 fromConnection:self.captureConnectionMock];
> 
>   [self.delegateMock verify];
>-
>-  [(id)currentDeviceMock stopMocking];
>-  currentDeviceMock = nil;
>-  [classMock stopMocking];
>-  classMock = nil;
>   CFRelease(sampleBuffer);
> #endif
> }
>@@ -234,11 +235,7 @@ - (void)testRotationCamera:(AVCaptureDevicePosition)camera
>   OCMStub(inputPortMock.device).andReturn(captureDeviceMock);
>   OCMStub(captureDeviceMock.position).andReturn(camera);
> 
>-  // UpsideDown -> RTCVideoRotation_0.
>-  UIDevice *currentDeviceMock = OCMClassMock([UIDevice class]);
>-  OCMStub(currentDeviceMock.orientation).andReturn(deviceOrientation);
>-  id classMock = OCMClassMock([UIDevice class]);
>-  OCMStub([classMock currentDevice]).andReturn(currentDeviceMock);
>+  [UIDevice.currentDevice setValue:@(deviceOrientation) forKey:@"orientation"];
> 
>   CMSampleBufferRef sampleBuffer = createTestSampleBufferRef();
> 
>@@ -272,10 +269,6 @@ - (void)testRotationCamera:(AVCaptureDevicePosition)camera
> 
>   [self.delegateMock verify];
> 
>-  [(id)currentDeviceMock stopMocking];
>-  currentDeviceMock = nil;
>-  [classMock stopMocking];
>-  classMock = nil;
>   CFRelease(sampleBuffer);
> #endif
> }
>@@ -300,11 +293,7 @@ - (void)testRotationFrame {
>   OCMStub(inputPortMock.device).andReturn(captureDeviceMock);
>   OCMStub(captureDeviceMock.position).andReturn(AVCaptureDevicePositionFront);
> 
>-  // UpsideDown -> RTCVideoRotation_0.
>-  UIDevice *currentDeviceMock = OCMClassMock([UIDevice class]);
>-  OCMStub(currentDeviceMock.orientation).andReturn(UIDeviceOrientationLandscapeLeft);
>-  id classMock = OCMClassMock([UIDevice class]);
>-  OCMStub([classMock currentDevice]).andReturn(currentDeviceMock);
>+  [UIDevice.currentDevice setValue:@(UIDeviceOrientationLandscapeLeft) forKey:@"orientation"];
> 
>   CMSampleBufferRef sampleBuffer = createTestSampleBufferRef();
> 
>@@ -329,11 +318,6 @@ - (void)testRotationFrame {
>                 fromConnection:self.captureConnectionMock];
> 
>   [self.delegateMock verify];
>-
>-  [(id)currentDeviceMock stopMocking];
>-  currentDeviceMock = nil;
>-  [classMock stopMocking];
>-  classMock = nil;
>   CFRelease(sampleBuffer);
> #endif
> }
>@@ -349,59 +333,155 @@ - (void)testImageExif {
> #endif
> }
> 
>-@end
>+- (void)testStartingAndStoppingCapture {
>+  id expectedDeviceInputMock = OCMClassMock([AVCaptureDeviceInput class]);
>+  id captureDeviceInputMock = OCMClassMock([AVCaptureDeviceInput class]);
>+  OCMStub([captureDeviceInputMock deviceInputWithDevice:self.deviceMock error:[OCMArg setTo:nil]])
>+      .andReturn(expectedDeviceInputMock);
>+
>+  OCMStub([self.deviceMock lockForConfiguration:[OCMArg setTo:nil]]).andReturn(YES);
>+  OCMStub([self.deviceMock unlockForConfiguration]);
>+  OCMStub([_captureSessionMock canAddInput:expectedDeviceInputMock]).andReturn(YES);
>+  OCMStub([_captureSessionMock inputs]).andReturn(@[ expectedDeviceInputMock ]);
>+  OCMStub([_captureSessionMock removeInput:expectedDeviceInputMock]);
>+
>+  // Set expectation that the capture session should be started with correct device.
>+  OCMExpect([_captureSessionMock addInput:expectedDeviceInputMock]);
>+  OCMExpect([_captureSessionMock startRunning]);
>+  OCMExpect([_captureSessionMock stopRunning]);
>+
>+  id format = OCMClassMock([AVCaptureDeviceFormat class]);
>+  [self.capturer startCaptureWithDevice:self.deviceMock format:format fps:30];
>+  [self.capturer stopCapture];
>+
>+  // Start capture code is dispatched async.
>+  OCMVerifyAllWithDelay(_captureSessionMock, 15);
>+}
> 
>-// TODO(kthelgason): Reenable these tests on simulator.
>-// See bugs.webrtc.org/7813
>-#if TARGET_IPHONE_SIMULATOR
>-#define MAYBE_TEST(f, name) TEST(f, DISABLED_##name)
>-#else
>-#define MAYBE_TEST TEST
>-#endif
>+- (void)testStartCaptureFailingToLockForConfiguration {
>+  // The captureSessionMock is a strict mock, so this test will crash if the startCapture
>+  // method does not return when failing to lock for configuration.
>+  OCMExpect([self.deviceMock lockForConfiguration:[OCMArg setTo:nil]]).andReturn(NO);
>+
>+  id format = OCMClassMock([AVCaptureDeviceFormat class]);
>+  [self.capturer startCaptureWithDevice:self.deviceMock format:format fps:30];
>+
>+  // Start capture code is dispatched async.
>+  OCMVerifyAllWithDelay(self.deviceMock, 15);
>+}
>+
>+- (void)testStartingAndStoppingCaptureWithCallbacks {
>+  id expectedDeviceInputMock = OCMClassMock([AVCaptureDeviceInput class]);
>+  id captureDeviceInputMock = OCMClassMock([AVCaptureDeviceInput class]);
>+  OCMStub([captureDeviceInputMock deviceInputWithDevice:self.deviceMock error:[OCMArg setTo:nil]])
>+      .andReturn(expectedDeviceInputMock);
>+
>+  OCMStub([self.deviceMock lockForConfiguration:[OCMArg setTo:nil]]).andReturn(YES);
>+  OCMStub([self.deviceMock unlockForConfiguration]);
>+  OCMStub([_captureSessionMock canAddInput:expectedDeviceInputMock]).andReturn(YES);
>+  OCMStub([_captureSessionMock inputs]).andReturn(@[ expectedDeviceInputMock ]);
>+  OCMStub([_captureSessionMock removeInput:expectedDeviceInputMock]);
>+
>+  // Set expectation that the capture session should be started with correct device.
>+  OCMExpect([_captureSessionMock addInput:expectedDeviceInputMock]);
>+  OCMExpect([_captureSessionMock startRunning]);
>+  OCMExpect([_captureSessionMock stopRunning]);
>+
>+  dispatch_semaphore_t completedStopSemaphore = dispatch_semaphore_create(0);
>+
>+  __block BOOL completedStart = NO;
>+  id format = OCMClassMock([AVCaptureDeviceFormat class]);
>+  [self.capturer startCaptureWithDevice:self.deviceMock
>+                                 format:format
>+                                    fps:30
>+                      completionHandler:^(NSError *error) {
>+                        EXPECT_EQ(error, nil);
>+                        completedStart = YES;
>+                      }];
>+
>+  __block BOOL completedStop = NO;
>+  [self.capturer stopCaptureWithCompletionHandler:^{
>+    completedStop = YES;
>+    dispatch_semaphore_signal(completedStopSemaphore);
>+  }];
>+
>+  dispatch_semaphore_wait(completedStopSemaphore,
>+                          dispatch_time(DISPATCH_TIME_NOW, 15.0 * NSEC_PER_SEC));
>+  OCMVerifyAllWithDelay(_captureSessionMock, 15);
>+  EXPECT_TRUE(completedStart);
>+  EXPECT_TRUE(completedStop);
>+}
>+
>+- (void)testStartCaptureFailingToLockForConfigurationWithCallback {
>+  id expectedDeviceInputMock = OCMClassMock([AVCaptureDeviceInput class]);
>+  id captureDeviceInputMock = OCMClassMock([AVCaptureDeviceInput class]);
>+  OCMStub([captureDeviceInputMock deviceInputWithDevice:self.deviceMock error:[OCMArg setTo:nil]])
>+      .andReturn(expectedDeviceInputMock);
>+
>+  id errorMock = OCMClassMock([NSError class]);
>+
>+  OCMStub([self.deviceMock lockForConfiguration:[OCMArg setTo:errorMock]]).andReturn(NO);
>+  OCMStub([_captureSessionMock canAddInput:expectedDeviceInputMock]).andReturn(YES);
>+  OCMStub([self.deviceMock unlockForConfiguration]);
>+
>+  OCMExpect([_captureSessionMock addInput:expectedDeviceInputMock]);
> 
>-MAYBE_TEST(RTCCameraVideoCapturerTests, SetupSession) {
>+  dispatch_semaphore_t completedStartSemaphore = dispatch_semaphore_create(0);
>+  __block NSError *callbackError = nil;
>+
>+  id format = OCMClassMock([AVCaptureDeviceFormat class]);
>+  [self.capturer startCaptureWithDevice:self.deviceMock
>+                                 format:format
>+                                    fps:30
>+                      completionHandler:^(NSError *error) {
>+                        callbackError = error;
>+                        dispatch_semaphore_signal(completedStartSemaphore);
>+                      }];
>+
>+  long ret = dispatch_semaphore_wait(completedStartSemaphore,
>+                                     dispatch_time(DISPATCH_TIME_NOW, 15.0 * NSEC_PER_SEC));
>+  EXPECT_EQ(ret, 0);
>+  EXPECT_EQ(callbackError, errorMock);
>+}
>+
>+@end
>+
>+TEST(RTCCameraVideoCapturerTests, SetupSession) {
>   RTCCameraVideoCapturerTests *test = [[RTCCameraVideoCapturerTests alloc] init];
>   [test setup];
>   [test testSetupSession];
>   [test tearDown];
> }
> 
>-MAYBE_TEST(RTCCameraVideoCapturerTests, SetupSessionOutput) {
>+TEST(RTCCameraVideoCapturerTests, SetupSessionOutput) {
>   RTCCameraVideoCapturerTests *test = [[RTCCameraVideoCapturerTests alloc] init];
>   [test setup];
>   [test testSetupSessionOutput];
>   [test tearDown];
> }
> 
>-MAYBE_TEST(RTCCameraVideoCapturerTests, SupportedFormatsForDevice) {
>+TEST(RTCCameraVideoCapturerTests, SupportedFormatsForDevice) {
>   RTCCameraVideoCapturerTests *test = [[RTCCameraVideoCapturerTests alloc] init];
>   [test setup];
>   [test testSupportedFormatsForDevice];
>   [test tearDown];
> }
> 
>-MAYBE_TEST(RTCCameraVideoCapturerTests, CaptureDevices) {
>-  RTCCameraVideoCapturerTests *test = [[RTCCameraVideoCapturerTests alloc] init];
>-  [test setup];
>-  [test testCaptureDevices];
>-  [test tearDown];
>-}
>-
>-MAYBE_TEST(RTCCameraVideoCapturerTests, DelegateCallbackNotCalledWhenInvalidBuffer) {
>+TEST(RTCCameraVideoCapturerTests, DelegateCallbackNotCalledWhenInvalidBuffer) {
>   RTCCameraVideoCapturerTests *test = [[RTCCameraVideoCapturerTests alloc] init];
>   [test setup];
>   [test testDelegateCallbackNotCalledWhenInvalidBuffer];
>   [test tearDown];
> }
> 
>-MAYBE_TEST(RTCCameraVideoCapturerTests, DelegateCallbackWithValidBufferAndOrientationUpdate) {
>+TEST(RTCCameraVideoCapturerTests, DelegateCallbackWithValidBufferAndOrientationUpdate) {
>   RTCCameraVideoCapturerTests *test = [[RTCCameraVideoCapturerTests alloc] init];
>   [test setup];
>   [test testDelegateCallbackWithValidBufferAndOrientationUpdate];
>   [test tearDown];
> }
> 
>-MAYBE_TEST(RTCCameraVideoCapturerTests, RotationCameraBackLandscapeLeft) {
>+TEST(RTCCameraVideoCapturerTests, RotationCameraBackLandscapeLeft) {
>   RTCCameraVideoCapturerTests *test = [[RTCCameraVideoCapturerTests alloc] init];
>   [test setup];
>   [test testRotationCamera:AVCaptureDevicePositionBack
>@@ -409,7 +489,7 @@ MAYBE_TEST(RTCCameraVideoCapturerTests, RotationCameraBackLandscapeLeft) {
>   [test tearDown];
> }
> 
>-MAYBE_TEST(RTCCameraVideoCapturerTests, RotationCameraFrontLandscapeLeft) {
>+TEST(RTCCameraVideoCapturerTests, RotationCameraFrontLandscapeLeft) {
>   RTCCameraVideoCapturerTests *test = [[RTCCameraVideoCapturerTests alloc] init];
>   [test setup];
>   [test testRotationCamera:AVCaptureDevicePositionFront
>@@ -417,7 +497,7 @@ MAYBE_TEST(RTCCameraVideoCapturerTests, RotationCameraFrontLandscapeLeft) {
>   [test tearDown];
> }
> 
>-MAYBE_TEST(RTCCameraVideoCapturerTests, RotationCameraBackLandscapeRight) {
>+TEST(RTCCameraVideoCapturerTests, RotationCameraBackLandscapeRight) {
>   RTCCameraVideoCapturerTests *test = [[RTCCameraVideoCapturerTests alloc] init];
>   [test setup];
>   [test testRotationCamera:AVCaptureDevicePositionBack
>@@ -425,7 +505,7 @@ MAYBE_TEST(RTCCameraVideoCapturerTests, RotationCameraBackLandscapeRight) {
>   [test tearDown];
> }
> 
>-MAYBE_TEST(RTCCameraVideoCapturerTests, RotationCameraFrontLandscapeRight) {
>+TEST(RTCCameraVideoCapturerTests, RotationCameraFrontLandscapeRight) {
>   RTCCameraVideoCapturerTests *test = [[RTCCameraVideoCapturerTests alloc] init];
>   [test setup];
>   [test testRotationCamera:AVCaptureDevicePositionFront
>@@ -433,16 +513,44 @@ MAYBE_TEST(RTCCameraVideoCapturerTests, RotationCameraFrontLandscapeRight) {
>   [test tearDown];
> }
> 
>-MAYBE_TEST(RTCCameraVideoCapturerTests, RotationCameraFrame) {
>+TEST(RTCCameraVideoCapturerTests, RotationCameraFrame) {
>   RTCCameraVideoCapturerTests *test = [[RTCCameraVideoCapturerTests alloc] init];
>   [test setup];
>   [test testRotationFrame];
>   [test tearDown];
> }
> 
>-MAYBE_TEST(RTCCameraVideoCapturerTests, ImageExif) {
>+TEST(RTCCameraVideoCapturerTests, ImageExif) {
>   RTCCameraVideoCapturerTests *test = [[RTCCameraVideoCapturerTests alloc] init];
>   [test setup];
>   [test testImageExif];
>   [test tearDown];
> }
>+
>+TEST(RTCCameraVideoCapturerTests, StartAndStopCapture) {
>+  RTCCameraVideoCapturerTests *test = [[RTCCameraVideoCapturerTests alloc] init];
>+  [test setupWithMockedCaptureSession];
>+  [test testStartingAndStoppingCapture];
>+  [test tearDown];
>+}
>+
>+TEST(RTCCameraVideoCapturerTests, StartCaptureFailingToLockForConfiguration) {
>+  RTCCameraVideoCapturerTests *test = [[RTCCameraVideoCapturerTests alloc] init];
>+  [test setupWithMockedCaptureSession];
>+  [test testStartCaptureFailingToLockForConfiguration];
>+  [test tearDown];
>+}
>+
>+TEST(RTCCameraVideoCapturerTests, StartAndStopCaptureWithCallbacks) {
>+  RTCCameraVideoCapturerTests *test = [[RTCCameraVideoCapturerTests alloc] init];
>+  [test setupWithMockedCaptureSession];
>+  [test testStartingAndStoppingCaptureWithCallbacks];
>+  [test tearDown];
>+}
>+
>+TEST(RTCCameraVideoCapturerTests, StartCaptureFailingToLockForConfigurationWithCallback) {
>+  RTCCameraVideoCapturerTests *test = [[RTCCameraVideoCapturerTests alloc] init];
>+  [test setupWithMockedCaptureSession];
>+  [test testStartCaptureFailingToLockForConfigurationWithCallback];
>+  [test tearDown];
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/RTCCertificateTest.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/RTCCertificateTest.mm
>new file mode 100644
>index 00000000000..64269f7b8c8
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/RTCCertificateTest.mm
>@@ -0,0 +1,77 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#import <Foundation/Foundation.h>
>+
>+#include <vector>
>+
>+#include "rtc_base/gunit.h"
>+
>+#import "NSString+StdString.h"
>+#import "RTCConfiguration+Private.h"
>+#import "WebRTC/RTCConfiguration.h"
>+#import "WebRTC/RTCIceServer.h"
>+#import "WebRTC/RTCMediaConstraints.h"
>+#import "WebRTC/RTCPeerConnection.h"
>+#import "WebRTC/RTCPeerConnectionFactory.h"
>+
>+@interface RTCCertificateTest : NSObject
>+- (void)testCertificateIsUsedInConfig;
>+@end
>+
>+@implementation RTCCertificateTest
>+
>+- (void)testCertificateIsUsedInConfig {
>+  RTCConfiguration *originalConfig = [[RTCConfiguration alloc] init];
>+
>+  NSArray *urlStrings = @[ @"stun:stun1.example.net" ];
>+  RTCIceServer *server = [[RTCIceServer alloc] initWithURLStrings:urlStrings];
>+  originalConfig.iceServers = @[ server ];
>+
>+  // Generate a new certificate.
>+  RTCCertificate *originalCertificate = [RTCCertificate generateCertificateWithParams:@{
>+    @"expires" : @100000,
>+    @"name" : @"RSASSA-PKCS1-v1_5"
>+  }];
>+
>+  // Store certificate in configuration.
>+  originalConfig.certificate = originalCertificate;
>+
>+  RTCMediaConstraints *contraints =
>+      [[RTCMediaConstraints alloc] initWithMandatoryConstraints:@{} optionalConstraints:nil];
>+  RTCPeerConnectionFactory *factory = [[RTCPeerConnectionFactory alloc] init];
>+
>+  // Create PeerConnection with this certificate.
>+  RTCPeerConnection *peerConnection =
>+      [factory peerConnectionWithConfiguration:originalConfig constraints:contraints delegate:nil];
>+
>+  // Retrieve certificate from the configuration.
>+  RTCConfiguration *retrievedConfig = peerConnection.configuration;
>+
>+  // Extract PEM strings from original certificate.
>+  std::string originalPrivateKeyField = [[originalCertificate private_key] UTF8String];
>+  std::string originalCertificateField = [[originalCertificate certificate] UTF8String];
>+
>+  // Extract PEM strings from certificate retrieved from configuration.
>+  RTCCertificate *retrievedCertificate = retrievedConfig.certificate;
>+  std::string retrievedPrivateKeyField = [[retrievedCertificate private_key] UTF8String];
>+  std::string retrievedCertificateField = [[retrievedCertificate certificate] UTF8String];
>+
>+  // Check that the original certificate and retrieved certificate match.
>+  EXPECT_EQ(originalPrivateKeyField, retrievedPrivateKeyField);
>+  EXPECT_EQ(retrievedCertificateField, retrievedCertificateField);
>+}
>+
>+@end
>+
>+TEST(CertificateTest, CertificateIsUsedInConfig) {
>+  RTCCertificateTest *test = [[RTCCertificateTest alloc] init];
>+  [test testCertificateIsUsedInConfig];
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/RTCH264ProfileLevelId_xctest.m b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/RTCH264ProfileLevelId_xctest.m
>new file mode 100644
>index 00000000000..4f0c7c7a270
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/RTCH264ProfileLevelId_xctest.m
>@@ -0,0 +1,46 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#import "WebRTC/RTCVideoCodecH264.h"
>+
>+#import <XCTest/XCTest.h>
>+
>+@interface RTCH264ProfileLevelIdTests : XCTestCase
>+
>+@end
>+
>+static NSString *level31ConstrainedHigh = @"640c1f";
>+static NSString *level31ConstrainedBaseline = @"42e01f";
>+
>+@implementation RTCH264ProfileLevelIdTests
>+
>+- (void)testInitWithString {
>+  RTCH264ProfileLevelId *profileLevelId =
>+      [[RTCH264ProfileLevelId alloc] initWithHexString:level31ConstrainedHigh];
>+  XCTAssertEqual(profileLevelId.profile, RTCH264ProfileConstrainedHigh);
>+  XCTAssertEqual(profileLevelId.level, RTCH264Level3_1);
>+
>+  profileLevelId = [[RTCH264ProfileLevelId alloc] initWithHexString:level31ConstrainedBaseline];
>+  XCTAssertEqual(profileLevelId.profile, RTCH264ProfileConstrainedBaseline);
>+  XCTAssertEqual(profileLevelId.level, RTCH264Level3_1);
>+}
>+
>+- (void)testInitWithProfileAndLevel {
>+  RTCH264ProfileLevelId *profileLevelId =
>+      [[RTCH264ProfileLevelId alloc] initWithProfile:RTCH264ProfileConstrainedHigh
>+                                               level:RTCH264Level3_1];
>+  XCTAssertEqualObjects(profileLevelId.hexString, level31ConstrainedHigh);
>+
>+  profileLevelId = [[RTCH264ProfileLevelId alloc] initWithProfile:RTCH264ProfileConstrainedBaseline
>+                                                            level:RTCH264Level3_1];
>+  XCTAssertEqualObjects(profileLevelId.hexString, level31ConstrainedBaseline);
>+}
>+
>+@end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/RTCMTLVideoView_xctest.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/RTCMTLVideoView_xctest.m
>similarity index 53%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/RTCMTLVideoView_xctest.mm
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/RTCMTLVideoView_xctest.m
>index 602c699dba8..8f157f1d99e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/RTCMTLVideoView_xctest.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/RTCMTLVideoView_xctest.m
>@@ -11,6 +11,7 @@
> #import <XCTest/XCTest.h>
> 
> #import <Foundation/Foundation.h>
>+#import <MetalKit/MetalKit.h>
> #import <OCMock/OCMock.h>
> 
> #include <WebRTC/RTCMTLVideoView.h>
>@@ -62,24 +63,20 @@ - (id)frameMockWithCVPixelBuffer:(BOOL)hasCVPixelBuffer {
>   id frameMock = OCMClassMock([RTCVideoFrame class]);
>   if (hasCVPixelBuffer) {
>     CVPixelBufferRef pixelBufferRef;
>-    CVPixelBufferCreate(kCFAllocatorDefault,
>-                        200,
>-                        200,
>-                        kCVPixelFormatType_420YpCbCr8Planar,
>-                        nullptr,
>-                        &pixelBufferRef);
>+    CVPixelBufferCreate(
>+        kCFAllocatorDefault, 200, 200, kCVPixelFormatType_420YpCbCr8Planar, nil, &pixelBufferRef);
>     OCMStub([frameMock buffer])
>         .andReturn([[RTCCVPixelBuffer alloc] initWithPixelBuffer:pixelBufferRef]);
>   } else {
>     OCMStub([frameMock buffer]).andReturn([[RTCI420Buffer alloc] initWithWidth:200 height:200]);
>   }
>+  OCMStub([frameMock timeStampNs]).andReturn(arc4random_uniform(INT_MAX));
>   return frameMock;
> }
> 
>-- (id)rendererMockWithSuccessfulSetup:(BOOL)sucess {
>-  id rendererMock = OCMProtocolMock(@protocol(RTCMTLRenderer));
>-  OCMStub([rendererMock addRenderingDestination:[OCMArg any]]).andReturn(sucess);
>-
>+- (id)rendererMockWithSuccessfulSetup:(BOOL)success {
>+  id rendererMock = OCMClassMock([RTCMTLRenderer class]);
>+  OCMStub([rendererMock addRenderingDestination:[OCMArg any]]).andReturn(success);
>   return rendererMock;
> }
> 
>@@ -89,6 +86,7 @@ - (void)startMockingNilView {
> }
> 
> #pragma mark - Test cases
>+
> - (void)testInitAssertsIfMetalUnavailabe {
>   // given
>   OCMStub([self.classMock isMetalAvailable]).andReturn(NO);
>@@ -167,4 +165,110 @@ - (void)testRTCVideoRenderFrameCallbackNV12 {
>   [self.classMock verify];
> }
> 
>+- (void)testRTCVideoRenderWorksAfterReconstruction {
>+  OCMStub([self.classMock isMetalAvailable]).andReturn(YES);
>+  self.rendererNV12Mock = [self rendererMockWithSuccessfulSetup:YES];
>+  self.frameMock = [self frameMockWithCVPixelBuffer:YES];
>+
>+  OCMExpect([self.rendererNV12Mock drawFrame:self.frameMock]);
>+  OCMExpect([self.classMock createNV12Renderer]).andReturn(self.rendererNV12Mock);
>+  [[self.classMock reject] createI420Renderer];
>+
>+  RTCMTLVideoView *realView = [[RTCMTLVideoView alloc] init];
>+
>+  [realView renderFrame:self.frameMock];
>+  [realView drawInMTKView:nil];
>+  [self.rendererNV12Mock verify];
>+  [self.classMock verify];
>+
>+  // Recreate view.
>+  realView = [[RTCMTLVideoView alloc] init];
>+  OCMExpect([self.rendererNV12Mock drawFrame:self.frameMock]);
>+  // View hould reinit renderer.
>+  OCMExpect([self.classMock createNV12Renderer]).andReturn(self.rendererNV12Mock);
>+
>+  [realView renderFrame:self.frameMock];
>+  [realView drawInMTKView:nil];
>+  [self.rendererNV12Mock verify];
>+  [self.classMock verify];
>+}
>+
>+- (void)testDontRedrawOldFrame {
>+  OCMStub([self.classMock isMetalAvailable]).andReturn(YES);
>+  self.rendererNV12Mock = [self rendererMockWithSuccessfulSetup:YES];
>+  self.frameMock = [self frameMockWithCVPixelBuffer:YES];
>+
>+  OCMExpect([self.rendererNV12Mock drawFrame:self.frameMock]);
>+  OCMExpect([self.classMock createNV12Renderer]).andReturn(self.rendererNV12Mock);
>+  [[self.classMock reject] createI420Renderer];
>+
>+  RTCMTLVideoView *realView = [[RTCMTLVideoView alloc] init];
>+  [realView renderFrame:self.frameMock];
>+  [realView drawInMTKView:nil];
>+
>+  [self.rendererNV12Mock verify];
>+  [self.classMock verify];
>+
>+  [[self.rendererNV12Mock reject] drawFrame:[OCMArg any]];
>+
>+  [realView renderFrame:self.frameMock];
>+  [realView drawInMTKView:nil];
>+
>+  [self.rendererNV12Mock verify];
>+}
>+
>+- (void)testDoDrawNewFrame {
>+  OCMStub([self.classMock isMetalAvailable]).andReturn(YES);
>+  self.rendererNV12Mock = [self rendererMockWithSuccessfulSetup:YES];
>+  self.frameMock = [self frameMockWithCVPixelBuffer:YES];
>+
>+  OCMExpect([self.rendererNV12Mock drawFrame:self.frameMock]);
>+  OCMExpect([self.classMock createNV12Renderer]).andReturn(self.rendererNV12Mock);
>+  [[self.classMock reject] createI420Renderer];
>+
>+  RTCMTLVideoView *realView = [[RTCMTLVideoView alloc] init];
>+  [realView renderFrame:self.frameMock];
>+  [realView drawInMTKView:nil];
>+
>+  [self.rendererNV12Mock verify];
>+  [self.classMock verify];
>+
>+  // Get new frame.
>+  self.frameMock = [self frameMockWithCVPixelBuffer:YES];
>+  OCMExpect([self.rendererNV12Mock drawFrame:self.frameMock]);
>+
>+  [realView renderFrame:self.frameMock];
>+  [realView drawInMTKView:nil];
>+
>+  [self.rendererNV12Mock verify];
>+}
>+
>+- (void)testReportsSizeChangesToDelegate {
>+  OCMStub([self.classMock isMetalAvailable]).andReturn(YES);
>+
>+  id delegateMock = OCMProtocolMock(@protocol(RTCVideoViewDelegate));
>+  CGSize size = CGSizeMake(640, 480);
>+  OCMExpect([delegateMock videoView:[OCMArg any] didChangeVideoSize:size]);
>+
>+  RTCMTLVideoView *realView = [[RTCMTLVideoView alloc] init];
>+  realView.delegate = delegateMock;
>+  [realView setSize:size];
>+
>+  // Delegate method is invoked with a dispatch_async.
>+  OCMVerifyAllWithDelay(delegateMock, 1);
>+}
>+
>+- (void)testSetContentMode {
>+  OCMStub([self.classMock isMetalAvailable]).andReturn(YES);
>+  id metalKitView = OCMClassMock([MTKView class]);
>+  [[[[self.classMock stub] ignoringNonObjectArgs] andReturn:metalKitView]
>+      createMetalView:CGRectZero];
>+  OCMExpect([metalKitView setContentMode:UIViewContentModeScaleAspectFill]);
>+
>+  RTCMTLVideoView *realView = [[RTCMTLVideoView alloc] init];
>+  [realView setVideoContentMode:UIViewContentModeScaleAspectFill];
>+
>+  OCMVerify(metalKitView);
>+}
>+
> @end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/RTCPeerConnectionFactoryBuilderTest.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/RTCPeerConnectionFactoryBuilderTest.mm
>new file mode 100644
>index 00000000000..b09cc12d2d4
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/RTCPeerConnectionFactoryBuilderTest.mm
>@@ -0,0 +1,93 @@
>+/*
>+ *  Copyright 2015 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#import <Foundation/Foundation.h>
>+#ifdef __cplusplus
>+extern "C" {
>+#endif
>+#import <OCMock/OCMock.h>
>+#ifdef __cplusplus
>+}
>+#endif
>+#import "sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactory+Native.h"
>+#import "sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactoryBuilder+DefaultComponents.h"
>+#import "sdk/objc/Framework/Classes/PeerConnection/RTCPeerConnectionFactoryBuilder.h"
>+
>+#include "api/audio_codecs/builtin_audio_decoder_factory.h"
>+#include "api/audio_codecs/builtin_audio_encoder_factory.h"
>+#include "api/video_codecs/video_decoder_factory.h"
>+#include "api/video_codecs/video_encoder_factory.h"
>+#include "modules/audio_device/include/audio_device.h"
>+#include "modules/audio_processing/include/audio_processing.h"
>+
>+#include "rtc_base/gunit.h"
>+#include "rtc_base/system/unused.h"
>+
>+@interface RTCPeerConnectionFactoryBuilderTest : NSObject
>+- (void)testBuilder;
>+- (void)testDefaultComponentsBuilder;
>+@end
>+
>+@implementation RTCPeerConnectionFactoryBuilderTest
>+
>+- (void)testBuilder {
>+  id factoryMock = OCMStrictClassMock([RTCPeerConnectionFactory class]);
>+  OCMExpect([factoryMock alloc]).andReturn(factoryMock);
>+#ifdef HAVE_NO_MEDIA
>+  RTC_UNUSED([[[factoryMock expect] andReturn:factoryMock] initWithNoMedia]);
>+#else
>+  RTC_UNUSED([[[[factoryMock expect] andReturn:factoryMock] ignoringNonObjectArgs]
>+      initWithNativeAudioEncoderFactory:nullptr
>+              nativeAudioDecoderFactory:nullptr
>+              nativeVideoEncoderFactory:nullptr
>+              nativeVideoDecoderFactory:nullptr
>+                      audioDeviceModule:nullptr
>+                  audioProcessingModule:nullptr]);
>+#endif
>+  RTCPeerConnectionFactoryBuilder* builder = [[RTCPeerConnectionFactoryBuilder alloc] init];
>+  RTCPeerConnectionFactory* peerConnectionFactory = [builder createPeerConnectionFactory];
>+  EXPECT_TRUE(peerConnectionFactory != nil);
>+  OCMVerifyAll(factoryMock);
>+}
>+
>+- (void)testDefaultComponentsBuilder {
>+  id factoryMock = OCMStrictClassMock([RTCPeerConnectionFactory class]);
>+  OCMExpect([factoryMock alloc]).andReturn(factoryMock);
>+#ifdef HAVE_NO_MEDIA
>+  RTC_UNUSED([[[factoryMock expect] andReturn:factoryMock] initWithNoMedia]);
>+#else
>+  RTC_UNUSED([[[[factoryMock expect] andReturn:factoryMock] ignoringNonObjectArgs]
>+      initWithNativeAudioEncoderFactory:nullptr
>+              nativeAudioDecoderFactory:nullptr
>+              nativeVideoEncoderFactory:nullptr
>+              nativeVideoDecoderFactory:nullptr
>+                      audioDeviceModule:nullptr
>+                  audioProcessingModule:nullptr]);
>+#endif
>+  RTCPeerConnectionFactoryBuilder* builder = [RTCPeerConnectionFactoryBuilder defaultBuilder];
>+  RTCPeerConnectionFactory* peerConnectionFactory = [builder createPeerConnectionFactory];
>+  EXPECT_TRUE(peerConnectionFactory != nil);
>+  OCMVerifyAll(factoryMock);
>+}
>+@end
>+
>+TEST(RTCPeerConnectionFactoryBuilderTest, BuilderTest) {
>+  @autoreleasepool {
>+    RTCPeerConnectionFactoryBuilderTest* test = [[RTCPeerConnectionFactoryBuilderTest alloc] init];
>+    [test testBuilder];
>+  }
>+}
>+
>+TEST(RTCPeerConnectionFactoryBuilderTest, DefaultComponentsBuilderTest) {
>+  @autoreleasepool {
>+    RTCPeerConnectionFactoryBuilderTest* test = [[RTCPeerConnectionFactoryBuilderTest alloc] init];
>+    [test testDefaultComponentsBuilder];
>+  }
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/RTCPeerConnectionFactory_xctest.m b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/RTCPeerConnectionFactory_xctest.m
>new file mode 100644
>index 00000000000..5608c69dbc1
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/RTCPeerConnectionFactory_xctest.m
>@@ -0,0 +1,317 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#import <WebRTC/RTCAudioSource.h>
>+#import <WebRTC/RTCConfiguration.h>
>+#import <WebRTC/RTCDataChannel.h>
>+#import <WebRTC/RTCDataChannelConfiguration.h>
>+#import <WebRTC/RTCMediaConstraints.h>
>+#import <WebRTC/RTCMediaStreamTrack.h>
>+#import <WebRTC/RTCPeerConnection.h>
>+#import <WebRTC/RTCPeerConnectionFactory.h>
>+#import <WebRTC/RTCRtpReceiver.h>
>+#import <WebRTC/RTCRtpSender.h>
>+#import <WebRTC/RTCRtpTransceiver.h>
>+#import <WebRTC/RTCVideoSource.h>
>+
>+#import <XCTest/XCTest.h>
>+
>+@interface RTCPeerConnectionFactoryTests : XCTestCase
>+@end
>+
>+@implementation RTCPeerConnectionFactoryTests
>+
>+- (void)testPeerConnectionLifetime {
>+  @autoreleasepool {
>+    RTCConfiguration *config = [[RTCConfiguration alloc] init];
>+
>+    RTCMediaConstraints *constraints =
>+        [[RTCMediaConstraints alloc] initWithMandatoryConstraints:@{} optionalConstraints:nil];
>+
>+    RTCPeerConnectionFactory *factory;
>+    RTCPeerConnection *peerConnection;
>+
>+    @autoreleasepool {
>+      factory = [[RTCPeerConnectionFactory alloc] init];
>+      peerConnection =
>+          [factory peerConnectionWithConfiguration:config constraints:constraints delegate:nil];
>+      [peerConnection close];
>+      factory = nil;
>+    }
>+    peerConnection = nil;
>+  }
>+
>+  XCTAssertTrue(true, @"Expect test does not crash");
>+}
>+
>+- (void)testMediaStreamLifetime {
>+  @autoreleasepool {
>+    RTCPeerConnectionFactory *factory;
>+    RTCMediaStream *mediaStream;
>+
>+    @autoreleasepool {
>+      factory = [[RTCPeerConnectionFactory alloc] init];
>+      mediaStream = [factory mediaStreamWithStreamId:@"mediaStream"];
>+      factory = nil;
>+    }
>+    mediaStream = nil;
>+  }
>+
>+  XCTAssertTrue(true, "Expect test does not crash");
>+}
>+
>+- (void)testDataChannelLifetime {
>+  @autoreleasepool {
>+    RTCConfiguration *config = [[RTCConfiguration alloc] init];
>+    RTCMediaConstraints *constraints =
>+        [[RTCMediaConstraints alloc] initWithMandatoryConstraints:@{} optionalConstraints:nil];
>+    RTCDataChannelConfiguration *dataChannelConfig = [[RTCDataChannelConfiguration alloc] init];
>+
>+    RTCPeerConnectionFactory *factory;
>+    RTCPeerConnection *peerConnection;
>+    RTCDataChannel *dataChannel;
>+
>+    @autoreleasepool {
>+      factory = [[RTCPeerConnectionFactory alloc] init];
>+      peerConnection =
>+          [factory peerConnectionWithConfiguration:config constraints:constraints delegate:nil];
>+      dataChannel =
>+          [peerConnection dataChannelForLabel:@"test_channel" configuration:dataChannelConfig];
>+      XCTAssertNotNil(dataChannel);
>+      [peerConnection close];
>+      peerConnection = nil;
>+      factory = nil;
>+    }
>+    dataChannel = nil;
>+  }
>+
>+  XCTAssertTrue(true, "Expect test does not crash");
>+}
>+
>+- (void)testRTCRtpTransceiverLifetime {
>+  @autoreleasepool {
>+    RTCConfiguration *config = [[RTCConfiguration alloc] init];
>+    config.sdpSemantics = RTCSdpSemanticsUnifiedPlan;
>+    RTCMediaConstraints *contraints =
>+        [[RTCMediaConstraints alloc] initWithMandatoryConstraints:@{} optionalConstraints:nil];
>+    RTCRtpTransceiverInit *init = [[RTCRtpTransceiverInit alloc] init];
>+
>+    RTCPeerConnectionFactory *factory;
>+    RTCPeerConnection *peerConnection;
>+    RTCRtpTransceiver *tranceiver;
>+
>+    @autoreleasepool {
>+      factory = [[RTCPeerConnectionFactory alloc] init];
>+      peerConnection =
>+          [factory peerConnectionWithConfiguration:config constraints:contraints delegate:nil];
>+      tranceiver = [peerConnection addTransceiverOfType:RTCRtpMediaTypeAudio init:init];
>+      XCTAssertNotNil(tranceiver);
>+      [peerConnection close];
>+      peerConnection = nil;
>+      factory = nil;
>+    }
>+    tranceiver = nil;
>+  }
>+
>+  XCTAssertTrue(true, "Expect test does not crash");
>+}
>+
>+- (void)testRTCRtpSenderLifetime {
>+  @autoreleasepool {
>+    RTCConfiguration *config = [[RTCConfiguration alloc] init];
>+    RTCMediaConstraints *constraints =
>+        [[RTCMediaConstraints alloc] initWithMandatoryConstraints:@{} optionalConstraints:nil];
>+
>+    RTCPeerConnectionFactory *factory;
>+    RTCPeerConnection *peerConnection;
>+    RTCRtpSender *sender;
>+
>+    @autoreleasepool {
>+      factory = [[RTCPeerConnectionFactory alloc] init];
>+      peerConnection =
>+          [factory peerConnectionWithConfiguration:config constraints:constraints delegate:nil];
>+      sender = [peerConnection senderWithKind:kRTCMediaStreamTrackKindVideo streamId:@"stream"];
>+      XCTAssertNotNil(sender);
>+      [peerConnection close];
>+      peerConnection = nil;
>+      factory = nil;
>+    }
>+    sender = nil;
>+  }
>+
>+  XCTAssertTrue(true, "Expect test does not crash");
>+}
>+
>+- (void)testRTCRtpReceiverLifetime {
>+  @autoreleasepool {
>+    RTCConfiguration *config = [[RTCConfiguration alloc] init];
>+    RTCMediaConstraints *constraints =
>+        [[RTCMediaConstraints alloc] initWithMandatoryConstraints:@{} optionalConstraints:nil];
>+
>+    RTCPeerConnectionFactory *factory;
>+    RTCPeerConnection *pc1;
>+    RTCPeerConnection *pc2;
>+
>+    NSArray<RTCRtpReceiver *> *receivers1;
>+    NSArray<RTCRtpReceiver *> *receivers2;
>+
>+    @autoreleasepool {
>+      factory = [[RTCPeerConnectionFactory alloc] init];
>+      pc1 = [factory peerConnectionWithConfiguration:config constraints:constraints delegate:nil];
>+      [pc1 senderWithKind:kRTCMediaStreamTrackKindAudio streamId:@"stream"];
>+
>+      pc2 = [factory peerConnectionWithConfiguration:config constraints:constraints delegate:nil];
>+      [pc2 senderWithKind:kRTCMediaStreamTrackKindAudio streamId:@"stream"];
>+
>+      NSTimeInterval negotiationTimeout = 15;
>+      XCTAssertTrue([self negotiatePeerConnection:pc1
>+                               withPeerConnection:pc2
>+                               negotiationTimeout:negotiationTimeout]);
>+
>+      XCTAssertEqual(pc1.signalingState, RTCSignalingStateStable);
>+      XCTAssertEqual(pc2.signalingState, RTCSignalingStateStable);
>+
>+      receivers1 = pc1.receivers;
>+      receivers2 = pc2.receivers;
>+      XCTAssertTrue(receivers1.count > 0);
>+      XCTAssertTrue(receivers2.count > 0);
>+      [pc1 close];
>+      [pc2 close];
>+      pc1 = nil;
>+      pc2 = nil;
>+      factory = nil;
>+    }
>+    receivers1 = nil;
>+    receivers2 = nil;
>+  }
>+
>+  XCTAssertTrue(true, "Expect test does not crash");
>+}
>+
>+- (void)testAudioSourceLifetime {
>+  @autoreleasepool {
>+    RTCPeerConnectionFactory *factory;
>+    RTCAudioSource *audioSource;
>+
>+    @autoreleasepool {
>+      factory = [[RTCPeerConnectionFactory alloc] init];
>+      audioSource = [factory audioSourceWithConstraints:nil];
>+      XCTAssertNotNil(audioSource);
>+      factory = nil;
>+    }
>+    audioSource = nil;
>+  }
>+
>+  XCTAssertTrue(true, "Expect test does not crash");
>+}
>+
>+- (void)testVideoSourceLifetime {
>+  @autoreleasepool {
>+    RTCPeerConnectionFactory *factory;
>+    RTCVideoSource *videoSource;
>+
>+    @autoreleasepool {
>+      factory = [[RTCPeerConnectionFactory alloc] init];
>+      videoSource = [factory videoSource];
>+      XCTAssertNotNil(videoSource);
>+      factory = nil;
>+    }
>+    videoSource = nil;
>+  }
>+
>+  XCTAssertTrue(true, "Expect test does not crash");
>+}
>+
>+- (void)testAudioTrackLifetime {
>+  @autoreleasepool {
>+    RTCPeerConnectionFactory *factory;
>+    RTCAudioTrack *audioTrack;
>+
>+    @autoreleasepool {
>+      factory = [[RTCPeerConnectionFactory alloc] init];
>+      audioTrack = [factory audioTrackWithTrackId:@"audioTrack"];
>+      XCTAssertNotNil(audioTrack);
>+      factory = nil;
>+    }
>+    audioTrack = nil;
>+  }
>+
>+  XCTAssertTrue(true, "Expect test does not crash");
>+}
>+
>+- (void)testVideoTrackLifetime {
>+  @autoreleasepool {
>+    RTCPeerConnectionFactory *factory;
>+    RTCVideoTrack *videoTrack;
>+
>+    @autoreleasepool {
>+      factory = [[RTCPeerConnectionFactory alloc] init];
>+      videoTrack = [factory videoTrackWithSource:[factory videoSource] trackId:@"videoTrack"];
>+      XCTAssertNotNil(videoTrack);
>+      factory = nil;
>+    }
>+    videoTrack = nil;
>+  }
>+
>+  XCTAssertTrue(true, "Expect test does not crash");
>+}
>+
>+- (bool)negotiatePeerConnection:(RTCPeerConnection *)pc1
>+             withPeerConnection:(RTCPeerConnection *)pc2
>+             negotiationTimeout:(NSTimeInterval)timeout {
>+  __weak RTCPeerConnection *weakPC1 = pc1;
>+  __weak RTCPeerConnection *weakPC2 = pc2;
>+  RTCMediaConstraints *sdpConstraints =
>+      [[RTCMediaConstraints alloc] initWithMandatoryConstraints:@{
>+        kRTCMediaConstraintsOfferToReceiveAudio : kRTCMediaConstraintsValueTrue
>+      }
>+                                            optionalConstraints:nil];
>+
>+  dispatch_semaphore_t negotiatedSem = dispatch_semaphore_create(0);
>+  [weakPC1 offerForConstraints:sdpConstraints
>+             completionHandler:^(RTCSessionDescription *offer, NSError *error) {
>+               XCTAssertNil(error);
>+               XCTAssertNotNil(offer);
>+               [weakPC1
>+                   setLocalDescription:offer
>+                     completionHandler:^(NSError *error) {
>+                       XCTAssertNil(error);
>+                       [weakPC2
>+                           setRemoteDescription:offer
>+                              completionHandler:^(NSError *error) {
>+                                XCTAssertNil(error);
>+                                [weakPC2
>+                                    answerForConstraints:sdpConstraints
>+                                       completionHandler:^(RTCSessionDescription *answer,
>+                                                           NSError *error) {
>+                                         XCTAssertNil(error);
>+                                         XCTAssertNotNil(answer);
>+                                         [weakPC2
>+                                             setLocalDescription:answer
>+                                               completionHandler:^(NSError *error) {
>+                                                 XCTAssertNil(error);
>+                                                 [weakPC1
>+                                                     setRemoteDescription:answer
>+                                                        completionHandler:^(NSError *error) {
>+                                                          XCTAssertNil(error);
>+                                                          dispatch_semaphore_signal(negotiatedSem);
>+                                                        }];
>+                                               }];
>+                                       }];
>+                              }];
>+                     }];
>+             }];
>+
>+  return 0 ==
>+      dispatch_semaphore_wait(negotiatedSem,
>+                              dispatch_time(DISPATCH_TIME_NOW, (int64_t)(timeout * NSEC_PER_SEC)));
>+}
>+
>+@end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/RTCPeerConnectionTest.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/RTCPeerConnectionTest.mm
>index 65bedde704a..63617cbe5a9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/RTCPeerConnectionTest.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/RTCPeerConnectionTest.mm
>@@ -40,8 +40,8 @@ - (void)testConfigurationGetter {
>   config.tcpCandidatePolicy = RTCTcpCandidatePolicyDisabled;
>   config.candidateNetworkPolicy = RTCCandidateNetworkPolicyLowCost;
>   const int maxPackets = 60;
>-  const int timeout = 1;
>-  const int interval = 2;
>+  const int timeout = 1500;
>+  const int interval = 2000;
>   config.audioJitterBufferMaxPackets = maxPackets;
>   config.audioJitterBufferFastAccelerate = YES;
>   config.iceConnectionReceivingTimeout = timeout;
>@@ -49,6 +49,7 @@ - (void)testConfigurationGetter {
>   config.continualGatheringPolicy =
>       RTCContinualGatheringPolicyGatherContinually;
>   config.shouldPruneTurnPorts = YES;
>+  config.activeResetSrtpParams = YES;
> 
>   RTCMediaConstraints *contraints = [[RTCMediaConstraints alloc] initWithMandatoryConstraints:@{}
>       optionalConstraints:nil];
>@@ -87,6 +88,7 @@ - (void)testConfigurationGetter {
>             newConfig.iceBackupCandidatePairPingInterval);
>   EXPECT_EQ(config.continualGatheringPolicy, newConfig.continualGatheringPolicy);
>   EXPECT_EQ(config.shouldPruneTurnPorts, newConfig.shouldPruneTurnPorts);
>+  EXPECT_EQ(config.activeResetSrtpParams, newConfig.activeResetSrtpParams);
> }
> 
> @end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/audio_short16.pcm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/audio_short16.pcm
>new file mode 100644
>index 00000000000..15a0f1811c7
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/audio_short16.pcm differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/audio_short44.pcm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/audio_short44.pcm
>new file mode 100644
>index 00000000000..011cdce959b
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/audio_short44.pcm differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/audio_short48.pcm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/audio_short48.pcm
>new file mode 100644
>index 00000000000..06fd8261cd3
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/audio_short48.pcm differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/frame_buffer_helpers.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/frame_buffer_helpers.h
>new file mode 100644
>index 00000000000..76c0d15c7eb
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/frame_buffer_helpers.h
>@@ -0,0 +1,22 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#import <AVFoundation/AVFoundation.h>
>+
>+#include "api/video/i420_buffer.h"
>+
>+void DrawGradientInRGBPixelBuffer(CVPixelBufferRef pixelBuffer);
>+
>+rtc::scoped_refptr<webrtc::I420Buffer> CreateI420Gradient(int width,
>+                                                          int height);
>+
>+void CopyI420BufferToCVPixelBuffer(
>+    rtc::scoped_refptr<webrtc::I420Buffer> i420Buffer,
>+    CVPixelBufferRef pixelBuffer);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/frame_buffer_helpers.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/frame_buffer_helpers.mm
>new file mode 100644
>index 00000000000..c6a7550facb
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/frame_buffer_helpers.mm
>@@ -0,0 +1,126 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "sdk/objc/Framework/UnitTests/frame_buffer_helpers.h"
>+
>+#include "third_party/libyuv/include/libyuv.h"
>+
>+void DrawGradientInRGBPixelBuffer(CVPixelBufferRef pixelBuffer) {
>+  CVPixelBufferLockBaseAddress(pixelBuffer, kCVPixelBufferLock_ReadOnly);
>+  void* baseAddr = CVPixelBufferGetBaseAddress(pixelBuffer);
>+  size_t width = CVPixelBufferGetWidth(pixelBuffer);
>+  size_t height = CVPixelBufferGetHeight(pixelBuffer);
>+  CGColorSpaceRef colorSpace = CGColorSpaceCreateDeviceRGB();
>+  int byteOrder = CVPixelBufferGetPixelFormatType(pixelBuffer) == kCVPixelFormatType_32ARGB ?
>+      kCGBitmapByteOrder32Little :
>+      0;
>+  CGContextRef cgContext = CGBitmapContextCreate(baseAddr,
>+                                                 width,
>+                                                 height,
>+                                                 8,
>+                                                 CVPixelBufferGetBytesPerRow(pixelBuffer),
>+                                                 colorSpace,
>+                                                 byteOrder | kCGImageAlphaNoneSkipLast);
>+
>+  // Create a gradient
>+  CGFloat colors[] = {
>+      1.0, 1.0, 1.0, 1.0, 1.0, 0.0, 0.0, 1.0, 0.0, 1.0, 0.0, 1.0, 0.0, 0.0, 1.0, 1.0,
>+  };
>+  CGGradientRef gradient = CGGradientCreateWithColorComponents(colorSpace, colors, NULL, 4);
>+
>+  CGContextDrawLinearGradient(
>+      cgContext, gradient, CGPointMake(0, 0), CGPointMake(width, height), 0);
>+  CGGradientRelease(gradient);
>+
>+  CGImageRef cgImage = CGBitmapContextCreateImage(cgContext);
>+  CGContextRelease(cgContext);
>+  CGImageRelease(cgImage);
>+  CGColorSpaceRelease(colorSpace);
>+
>+  CVPixelBufferUnlockBaseAddress(pixelBuffer, kCVPixelBufferLock_ReadOnly);
>+}
>+
>+rtc::scoped_refptr<webrtc::I420Buffer> CreateI420Gradient(int width, int height) {
>+  rtc::scoped_refptr<webrtc::I420Buffer> buffer(webrtc::I420Buffer::Create(width, height));
>+  // Initialize with gradient, Y = 128(x/w + y/h), U = 256 x/w, V = 256 y/h
>+  for (int x = 0; x < width; x++) {
>+    for (int y = 0; y < height; y++) {
>+      buffer->MutableDataY()[x + y * width] = 128 * (x * height + y * width) / (width * height);
>+    }
>+  }
>+  int chroma_width = buffer->ChromaWidth();
>+  int chroma_height = buffer->ChromaHeight();
>+  for (int x = 0; x < chroma_width; x++) {
>+    for (int y = 0; y < chroma_height; y++) {
>+      buffer->MutableDataU()[x + y * chroma_width] = 255 * x / (chroma_width - 1);
>+      buffer->MutableDataV()[x + y * chroma_width] = 255 * y / (chroma_height - 1);
>+    }
>+  }
>+  return buffer;
>+}
>+
>+void CopyI420BufferToCVPixelBuffer(rtc::scoped_refptr<webrtc::I420Buffer> i420Buffer,
>+                                   CVPixelBufferRef pixelBuffer) {
>+  CVPixelBufferLockBaseAddress(pixelBuffer, 0);
>+
>+  const OSType pixelFormat = CVPixelBufferGetPixelFormatType(pixelBuffer);
>+  if (pixelFormat == kCVPixelFormatType_420YpCbCr8BiPlanarVideoRange ||
>+      pixelFormat == kCVPixelFormatType_420YpCbCr8BiPlanarFullRange) {
>+    // NV12
>+    uint8_t* dstY = static_cast<uint8_t*>(CVPixelBufferGetBaseAddressOfPlane(pixelBuffer, 0));
>+    const int dstYStride = CVPixelBufferGetBytesPerRowOfPlane(pixelBuffer, 0);
>+    uint8_t* dstUV = static_cast<uint8_t*>(CVPixelBufferGetBaseAddressOfPlane(pixelBuffer, 1));
>+    const int dstUVStride = CVPixelBufferGetBytesPerRowOfPlane(pixelBuffer, 1);
>+
>+    libyuv::I420ToNV12(i420Buffer->DataY(),
>+                       i420Buffer->StrideY(),
>+                       i420Buffer->DataU(),
>+                       i420Buffer->StrideU(),
>+                       i420Buffer->DataV(),
>+                       i420Buffer->StrideV(),
>+                       dstY,
>+                       dstYStride,
>+                       dstUV,
>+                       dstUVStride,
>+                       i420Buffer->width(),
>+                       i420Buffer->height());
>+  } else {
>+    uint8_t* dst = static_cast<uint8_t*>(CVPixelBufferGetBaseAddress(pixelBuffer));
>+    const int bytesPerRow = CVPixelBufferGetBytesPerRow(pixelBuffer);
>+
>+    if (pixelFormat == kCVPixelFormatType_32BGRA) {
>+      // Corresponds to libyuv::FOURCC_ARGB
>+      libyuv::I420ToARGB(i420Buffer->DataY(),
>+                         i420Buffer->StrideY(),
>+                         i420Buffer->DataU(),
>+                         i420Buffer->StrideU(),
>+                         i420Buffer->DataV(),
>+                         i420Buffer->StrideV(),
>+                         dst,
>+                         bytesPerRow,
>+                         i420Buffer->width(),
>+                         i420Buffer->height());
>+    } else if (pixelFormat == kCVPixelFormatType_32ARGB) {
>+      // Corresponds to libyuv::FOURCC_BGRA
>+      libyuv::I420ToBGRA(i420Buffer->DataY(),
>+                         i420Buffer->StrideY(),
>+                         i420Buffer->DataU(),
>+                         i420Buffer->StrideU(),
>+                         i420Buffer->DataV(),
>+                         i420Buffer->StrideV(),
>+                         dst,
>+                         bytesPerRow,
>+                         i420Buffer->width(),
>+                         i420Buffer->height());
>+    }
>+  }
>+
>+  CVPixelBufferUnlockBaseAddress(pixelBuffer, 0);
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/objc_video_decoder_factory_tests.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/objc_video_decoder_factory_tests.mm
>index d98b12a085c..a8ea43cdf04 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/objc_video_decoder_factory_tests.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/objc_video_decoder_factory_tests.mm
>@@ -11,7 +11,7 @@
> #import <Foundation/Foundation.h>
> #import <OCMock/OCMock.h>
> 
>-#include "sdk/objc/Framework/Classes/VideoToolbox/objc_video_decoder_factory.h"
>+#include "sdk/objc/Framework/Native/src/objc_video_decoder_factory.h"
> 
> #import "WebRTC/RTCVideoCodec.h"
> #import "WebRTC/RTCVideoCodecFactory.h"
>@@ -23,11 +23,9 @@
> 
> id<RTCVideoDecoderFactory> CreateDecoderFactoryReturning(int return_code) {
>   id decoderMock = OCMProtocolMock(@protocol(RTCVideoDecoder));
>-  OCMStub([decoderMock startDecodeWithSettings:[OCMArg any] numberOfCores:1])
>-      .andReturn(return_code);
>+  OCMStub([decoderMock startDecodeWithNumberOfCores:1]).andReturn(return_code);
>   OCMStub([decoderMock decode:[OCMArg any]
>                     missingFrames:NO
>-              fragmentationHeader:[OCMArg any]
>                 codecSpecificInfo:[OCMArg any]
>                      renderTimeMs:0])
>       .andReturn(return_code);
>@@ -48,58 +46,55 @@ id<RTCVideoDecoderFactory> CreateErrorDecoderFactory() {
>   return CreateDecoderFactoryReturning(WEBRTC_VIDEO_CODEC_ERROR);
> }
> 
>-webrtc::VideoDecoder *GetObjCDecoder(id<RTCVideoDecoderFactory> factory) {
>+std::unique_ptr<webrtc::VideoDecoder> GetObjCDecoder(id<RTCVideoDecoderFactory> factory) {
>   webrtc::ObjCVideoDecoderFactory decoder_factory(factory);
>-  return decoder_factory.CreateVideoDecoderWithParams(cricket::VideoCodec(cricket::kH264CodecName),
>-                                                      {});
>+  return decoder_factory.CreateVideoDecoder(webrtc::SdpVideoFormat(cricket::kH264CodecName));
> }
> 
> #pragma mark -
> 
> TEST(ObjCVideoDecoderFactoryTest, InitDecodeReturnsOKOnSuccess) {
>-  webrtc::VideoDecoder *decoder = GetObjCDecoder(CreateOKDecoderFactory());
>+  std::unique_ptr<webrtc::VideoDecoder> decoder = GetObjCDecoder(CreateOKDecoderFactory());
> 
>-  auto settings = new webrtc::VideoCodec();
>+  auto* settings = new webrtc::VideoCodec();
>   EXPECT_EQ(decoder->InitDecode(settings, 1), WEBRTC_VIDEO_CODEC_OK);
> }
> 
> TEST(ObjCVideoDecoderFactoryTest, InitDecodeReturnsErrorOnFail) {
>-  webrtc::VideoDecoder *decoder = GetObjCDecoder(CreateErrorDecoderFactory());
>+  std::unique_ptr<webrtc::VideoDecoder> decoder = GetObjCDecoder(CreateErrorDecoderFactory());
> 
>-  auto settings = new webrtc::VideoCodec();
>+  auto* settings = new webrtc::VideoCodec();
>   EXPECT_EQ(decoder->InitDecode(settings, 1), WEBRTC_VIDEO_CODEC_ERROR);
> }
> 
> TEST(ObjCVideoDecoderFactoryTest, DecodeReturnsOKOnSuccess) {
>-  webrtc::VideoDecoder *decoder = GetObjCDecoder(CreateOKDecoderFactory());
>+  std::unique_ptr<webrtc::VideoDecoder> decoder = GetObjCDecoder(CreateOKDecoderFactory());
> 
>   webrtc::EncodedImage encoded_image;
>-  webrtc::RTPFragmentationHeader header;
>   webrtc::CodecSpecificInfo info;
>   info.codecType = webrtc::kVideoCodecH264;
> 
>-  EXPECT_EQ(decoder->Decode(encoded_image, false, &header, &info, 0), WEBRTC_VIDEO_CODEC_OK);
>+  EXPECT_EQ(decoder->Decode(encoded_image, false, &info, 0), WEBRTC_VIDEO_CODEC_OK);
> }
> 
> TEST(ObjCVideoDecoderFactoryTest, DecodeReturnsErrorOnFail) {
>-  webrtc::VideoDecoder *decoder = GetObjCDecoder(CreateErrorDecoderFactory());
>+  std::unique_ptr<webrtc::VideoDecoder> decoder = GetObjCDecoder(CreateErrorDecoderFactory());
> 
>   webrtc::EncodedImage encoded_image;
>-  webrtc::RTPFragmentationHeader header;
>   webrtc::CodecSpecificInfo info;
>   info.codecType = webrtc::kVideoCodecH264;
> 
>-  EXPECT_EQ(decoder->Decode(encoded_image, false, &header, &info, 0), WEBRTC_VIDEO_CODEC_ERROR);
>+  EXPECT_EQ(decoder->Decode(encoded_image, false, &info, 0), WEBRTC_VIDEO_CODEC_ERROR);
> }
> 
> TEST(ObjCVideoDecoderFactoryTest, ReleaseDecodeReturnsOKOnSuccess) {
>-  webrtc::VideoDecoder *decoder = GetObjCDecoder(CreateOKDecoderFactory());
>+  std::unique_ptr<webrtc::VideoDecoder> decoder = GetObjCDecoder(CreateOKDecoderFactory());
> 
>   EXPECT_EQ(decoder->Release(), WEBRTC_VIDEO_CODEC_OK);
> }
> 
> TEST(ObjCVideoDecoderFactoryTest, ReleaseDecodeReturnsErrorOnFail) {
>-  webrtc::VideoDecoder *decoder = GetObjCDecoder(CreateErrorDecoderFactory());
>+  std::unique_ptr<webrtc::VideoDecoder> decoder = GetObjCDecoder(CreateErrorDecoderFactory());
> 
>   EXPECT_EQ(decoder->Release(), WEBRTC_VIDEO_CODEC_ERROR);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/objc_video_encoder_factory_tests.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/objc_video_encoder_factory_tests.mm
>index 1a3588cd544..9377751d054 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/objc_video_encoder_factory_tests.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/UnitTests/objc_video_encoder_factory_tests.mm
>@@ -11,16 +11,17 @@
> #import <Foundation/Foundation.h>
> #import <OCMock/OCMock.h>
> 
>-#include "sdk/objc/Framework/Classes/VideoToolbox/objc_video_encoder_factory.h"
>+#include "sdk/objc/Framework/Native/src/objc_video_encoder_factory.h"
> 
> #import "WebRTC/RTCVideoCodec.h"
> #import "WebRTC/RTCVideoCodecFactory.h"
> #import "WebRTC/RTCVideoFrameBuffer.h"
>+#include "api/video_codecs/sdp_video_format.h"
> #include "modules/include/module_common_types.h"
> #include "modules/video_coding/include/video_codec_interface.h"
> #include "modules/video_coding/include/video_error_codes.h"
> #include "rtc_base/gunit.h"
>-#include "sdk/objc/Framework/Classes/Video/objc_frame_buffer.h"
>+#include "sdk/objc/Framework/Native/src/objc_frame_buffer.h"
> 
> id<RTCVideoEncoderFactory> CreateEncoderFactoryReturning(int return_code) {
>   id encoderMock = OCMProtocolMock(@protocol(RTCVideoEncoder));
>@@ -46,30 +47,30 @@ id<RTCVideoEncoderFactory> CreateErrorEncoderFactory() {
>   return CreateEncoderFactoryReturning(WEBRTC_VIDEO_CODEC_ERROR);
> }
> 
>-webrtc::VideoEncoder *GetObjCEncoder(id<RTCVideoEncoderFactory> factory) {
>+std::unique_ptr<webrtc::VideoEncoder> GetObjCEncoder(id<RTCVideoEncoderFactory> factory) {
>   webrtc::ObjCVideoEncoderFactory encoder_factory(factory);
>-  cricket::VideoCodec codec("H264");
>-  return encoder_factory.CreateVideoEncoder(codec);
>+  webrtc::SdpVideoFormat format("H264");
>+  return encoder_factory.CreateVideoEncoder(format);
> }
> 
> #pragma mark -
> 
> TEST(ObjCVideoEncoderFactoryTest, InitEncodeReturnsOKOnSuccess) {
>-  webrtc::VideoEncoder *encoder = GetObjCEncoder(CreateOKEncoderFactory());
>+  std::unique_ptr<webrtc::VideoEncoder> encoder = GetObjCEncoder(CreateOKEncoderFactory());
> 
>-  auto settings = new webrtc::VideoCodec();
>+  auto* settings = new webrtc::VideoCodec();
>   EXPECT_EQ(encoder->InitEncode(settings, 1, 0), WEBRTC_VIDEO_CODEC_OK);
> }
> 
> TEST(ObjCVideoEncoderFactoryTest, InitEncodeReturnsErrorOnFail) {
>-  webrtc::VideoEncoder *encoder = GetObjCEncoder(CreateErrorEncoderFactory());
>+  std::unique_ptr<webrtc::VideoEncoder> encoder = GetObjCEncoder(CreateErrorEncoderFactory());
> 
>-  auto settings = new webrtc::VideoCodec();
>+  auto* settings = new webrtc::VideoCodec();
>   EXPECT_EQ(encoder->InitEncode(settings, 1, 0), WEBRTC_VIDEO_CODEC_ERROR);
> }
> 
> TEST(ObjCVideoEncoderFactoryTest, EncodeReturnsOKOnSuccess) {
>-  webrtc::VideoEncoder *encoder = GetObjCEncoder(CreateOKEncoderFactory());
>+  std::unique_ptr<webrtc::VideoEncoder> encoder = GetObjCEncoder(CreateOKEncoderFactory());
> 
>   CVPixelBufferRef pixel_buffer;
>   CVPixelBufferCreate(kCFAllocatorDefault, 640, 480, kCVPixelFormatType_32ARGB, nil, &pixel_buffer);
>@@ -86,7 +87,7 @@ TEST(ObjCVideoEncoderFactoryTest, EncodeReturnsOKOnSuccess) {
> }
> 
> TEST(ObjCVideoEncoderFactoryTest, EncodeReturnsErrorOnFail) {
>-  webrtc::VideoEncoder *encoder = GetObjCEncoder(CreateErrorEncoderFactory());
>+  std::unique_ptr<webrtc::VideoEncoder> encoder = GetObjCEncoder(CreateErrorEncoderFactory());
> 
>   CVPixelBufferRef pixel_buffer;
>   CVPixelBufferCreate(kCFAllocatorDefault, 640, 480, kCVPixelFormatType_32ARGB, nil, &pixel_buffer);
>@@ -103,31 +104,31 @@ TEST(ObjCVideoEncoderFactoryTest, EncodeReturnsErrorOnFail) {
> }
> 
> TEST(ObjCVideoEncoderFactoryTest, ReleaseEncodeReturnsOKOnSuccess) {
>-  webrtc::VideoEncoder *encoder = GetObjCEncoder(CreateOKEncoderFactory());
>+  std::unique_ptr<webrtc::VideoEncoder> encoder = GetObjCEncoder(CreateOKEncoderFactory());
> 
>   EXPECT_EQ(encoder->Release(), WEBRTC_VIDEO_CODEC_OK);
> }
> 
> TEST(ObjCVideoEncoderFactoryTest, ReleaseEncodeReturnsErrorOnFail) {
>-  webrtc::VideoEncoder *encoder = GetObjCEncoder(CreateErrorEncoderFactory());
>+  std::unique_ptr<webrtc::VideoEncoder> encoder = GetObjCEncoder(CreateErrorEncoderFactory());
> 
>   EXPECT_EQ(encoder->Release(), WEBRTC_VIDEO_CODEC_ERROR);
> }
> 
> TEST(ObjCVideoEncoderFactoryTest, SetChannelParametersAlwaysReturnsOK) {
>-  webrtc::VideoEncoder *encoder = GetObjCEncoder(CreateErrorEncoderFactory());
>+  std::unique_ptr<webrtc::VideoEncoder> encoder = GetObjCEncoder(CreateErrorEncoderFactory());
> 
>   EXPECT_EQ(encoder->SetChannelParameters(1, 1), WEBRTC_VIDEO_CODEC_OK);
> }
> 
> TEST(ObjCVideoEncoderFactoryTest, SetRatesReturnsOKOnSuccess) {
>-  webrtc::VideoEncoder *encoder = GetObjCEncoder(CreateOKEncoderFactory());
>+  std::unique_ptr<webrtc::VideoEncoder> encoder = GetObjCEncoder(CreateOKEncoderFactory());
> 
>   EXPECT_EQ(encoder->SetRates(0, 0), WEBRTC_VIDEO_CODEC_OK);
> }
> 
> TEST(ObjCVideoEncoderFactoryTest, SetRatesReturnsErrorOnFail) {
>-  webrtc::VideoEncoder *encoder = GetObjCEncoder(CreateErrorEncoderFactory());
>+  std::unique_ptr<webrtc::VideoEncoder> encoder = GetObjCEncoder(CreateErrorEncoderFactory());
> 
>   EXPECT_EQ(encoder->SetRates(0, 0), WEBRTC_VIDEO_CODEC_ERROR);
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/OWNERS
>index 17414d13779..4695e82a8de 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/OWNERS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/OWNERS
>@@ -1,5 +1,10 @@
>-denicija@webrtc.org
>+# Normal code changes.
> kthelgason@webrtc.org
>+andersc@webrtc.org
>+peterhanspers@webrtc.org
>+denicija@webrtc.org
>+
>+# Rubberstamps of e.g. reverts and critical bug fixes.
> magjed@webrtc.org
> tkchin@webrtc.org
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/WebRTC.podspec b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/WebRTC.podspec
>index 37ab166b0c5..a29b315fcf4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/WebRTC.podspec
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/WebRTC.podspec
>@@ -11,7 +11,7 @@ Pod::Spec.new do |s|
>   s.license      = { :type => "BSD", :file => "LICENSE" }
>   s.author    = "Google Inc."
> 
>-  s.platform     = :ios, "8.0"
>+  s.platform     = :ios, "9.0"
> 
>   s.public_header_files = "WebRTC.framework/Headers/**/*.h"
>   s.vendored_frameworks = "WebRTC.framework"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/stats/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/stats/BUILD.gn
>index 92231cef02a..d62dbd0f1bd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/stats/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/stats/BUILD.gn
>@@ -51,11 +51,6 @@ if (rtc_include_tests) {
>       "rtcstatsreport_unittest.cc",
>     ]
> 
>-    if (!build_with_chromium && is_clang) {
>-      # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>-      suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-    }
>-
>     deps = [
>       ":rtc_stats",
>       ":rtc_stats_test_utils",
>@@ -66,7 +61,6 @@ if (rtc_include_tests) {
>       "../rtc_base:rtc_base_tests_utils",
>       "../rtc_base:rtc_json",
>       "../system_wrappers:metrics_default",
>-      "//testing/gmock",
>     ]
> 
>     if (is_android) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/stats/rtcstats.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/stats/rtcstats.cc
>index e4460bd90ce..a27466079f9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/stats/rtcstats.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/stats/rtcstats.cc
>@@ -21,14 +21,14 @@ namespace {
> 
> // Produces "[a,b,c]". Works for non-vector |RTCStatsMemberInterface::Type|
> // types.
>-template<typename T>
>+template <typename T>
> std::string VectorToString(const std::vector<T>& vector) {
>   if (vector.empty())
>     return "[]";
>   std::ostringstream oss;
>-  oss << "[" << rtc::ToString<T>(vector[0]);
>+  oss << "[" << rtc::ToString(vector[0]);
>   for (size_t i = 1; i < vector.size(); ++i) {
>-    oss << "," << rtc::ToString<T>(vector[i]);
>+    oss << "," << rtc::ToString(vector[i]);
>   }
>   oss << "]";
>   return oss.str();
>@@ -36,14 +36,14 @@ std::string VectorToString(const std::vector<T>& vector) {
> 
> // Produces "[\"a\",\"b\",\"c\"]". Works for vectors of both const char* and
> // std::string element types.
>-template<typename T>
>+template <typename T>
> std::string VectorOfStringsToString(const std::vector<T>& strings) {
>   if (strings.empty())
>     return "[]";
>   std::ostringstream oss;
>-  oss << "[\"" << rtc::ToString<T>(strings[0]) << '\"';
>+  oss << "[\"" << rtc::ToString(strings[0]) << '\"';
>   for (size_t i = 1; i < strings.size(); ++i) {
>-    oss << ",\"" << rtc::ToString<T>(strings[i]) << '\"';
>+    oss << ",\"" << rtc::ToString(strings[i]) << '\"';
>   }
>   oss << "]";
>   return oss.str();
>@@ -118,8 +118,7 @@ std::vector<const RTCStatsMemberInterface*> RTCStats::Members() const {
> }
> 
> std::vector<const RTCStatsMemberInterface*>
>-RTCStats::MembersOfThisObjectAndAncestors(
>-    size_t additional_capacity) const {
>+RTCStats::MembersOfThisObjectAndAncestors(size_t additional_capacity) const {
>   std::vector<const RTCStatsMemberInterface*> members;
>   members.reserve(additional_capacity);
>   return members;
>@@ -184,12 +183,7 @@ WEBRTC_DEFINE_RTCSTATSMEMBER(double,
>                              false,
>                              rtc::ToString(value_),
>                              ToStringAsDouble(value_));
>-WEBRTC_DEFINE_RTCSTATSMEMBER(std::string,
>-                             kString,
>-                             false,
>-                             true,
>-                             value_,
>-                             value_);
>+WEBRTC_DEFINE_RTCSTATSMEMBER(std::string, kString, false, true, value_, value_);
> WEBRTC_DEFINE_RTCSTATSMEMBER(std::vector<bool>,
>                              kSequenceBool,
>                              true,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/stats/rtcstats_objects.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/stats/rtcstats_objects.cc
>index b6bc4c0f9e4..b711a277588 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/stats/rtcstats_objects.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/stats/rtcstats_objects.cc
>@@ -55,31 +55,25 @@ WEBRTC_RTCSTATS_IMPL(RTCCertificateStats, RTCStats, "certificate",
>     &issuer_certificate_id);
> // clang-format on
> 
>-RTCCertificateStats::RTCCertificateStats(
>-    const std::string& id, int64_t timestamp_us)
>-    : RTCCertificateStats(std::string(id), timestamp_us) {
>-}
>+RTCCertificateStats::RTCCertificateStats(const std::string& id,
>+                                         int64_t timestamp_us)
>+    : RTCCertificateStats(std::string(id), timestamp_us) {}
> 
>-RTCCertificateStats::RTCCertificateStats(
>-    std::string&& id, int64_t timestamp_us)
>+RTCCertificateStats::RTCCertificateStats(std::string&& id, int64_t timestamp_us)
>     : RTCStats(std::move(id), timestamp_us),
>       fingerprint("fingerprint"),
>       fingerprint_algorithm("fingerprintAlgorithm"),
>       base64_certificate("base64Certificate"),
>-      issuer_certificate_id("issuerCertificateId") {
>-}
>+      issuer_certificate_id("issuerCertificateId") {}
> 
>-RTCCertificateStats::RTCCertificateStats(
>-    const RTCCertificateStats& other)
>+RTCCertificateStats::RTCCertificateStats(const RTCCertificateStats& other)
>     : RTCStats(other.id(), other.timestamp_us()),
>       fingerprint(other.fingerprint),
>       fingerprint_algorithm(other.fingerprint_algorithm),
>       base64_certificate(other.base64_certificate),
>-      issuer_certificate_id(other.issuer_certificate_id) {
>-}
>+      issuer_certificate_id(other.issuer_certificate_id) {}
> 
>-RTCCertificateStats::~RTCCertificateStats() {
>-}
>+RTCCertificateStats::~RTCCertificateStats() {}
> 
> // clang-format off
> WEBRTC_RTCSTATS_IMPL(RTCCodecStats, RTCStats, "codec",
>@@ -91,35 +85,28 @@ WEBRTC_RTCSTATS_IMPL(RTCCodecStats, RTCStats, "codec",
>     &implementation);
> // clang-format on
> 
>-RTCCodecStats::RTCCodecStats(
>-    const std::string& id, int64_t timestamp_us)
>-    : RTCCodecStats(std::string(id), timestamp_us) {
>-}
>+RTCCodecStats::RTCCodecStats(const std::string& id, int64_t timestamp_us)
>+    : RTCCodecStats(std::string(id), timestamp_us) {}
> 
>-RTCCodecStats::RTCCodecStats(
>-    std::string&& id, int64_t timestamp_us)
>+RTCCodecStats::RTCCodecStats(std::string&& id, int64_t timestamp_us)
>     : RTCStats(std::move(id), timestamp_us),
>       payload_type("payloadType"),
>       mime_type("mimeType"),
>       clock_rate("clockRate"),
>       channels("channels"),
>       sdp_fmtp_line("sdpFmtpLine"),
>-      implementation("implementation") {
>-}
>+      implementation("implementation") {}
> 
>-RTCCodecStats::RTCCodecStats(
>-    const RTCCodecStats& other)
>+RTCCodecStats::RTCCodecStats(const RTCCodecStats& other)
>     : RTCStats(other.id(), other.timestamp_us()),
>       payload_type(other.payload_type),
>       mime_type(other.mime_type),
>       clock_rate(other.clock_rate),
>       channels(other.channels),
>       sdp_fmtp_line(other.sdp_fmtp_line),
>-      implementation(other.implementation) {
>-}
>+      implementation(other.implementation) {}
> 
>-RTCCodecStats::~RTCCodecStats() {
>-}
>+RTCCodecStats::~RTCCodecStats() {}
> 
> // clang-format off
> WEBRTC_RTCSTATS_IMPL(RTCDataChannelStats, RTCStats, "data-channel",
>@@ -133,13 +120,11 @@ WEBRTC_RTCSTATS_IMPL(RTCDataChannelStats, RTCStats, "data-channel",
>     &bytes_received);
> // clang-format on
> 
>-RTCDataChannelStats::RTCDataChannelStats(
>-    const std::string& id, int64_t timestamp_us)
>-    : RTCDataChannelStats(std::string(id), timestamp_us) {
>-}
>+RTCDataChannelStats::RTCDataChannelStats(const std::string& id,
>+                                         int64_t timestamp_us)
>+    : RTCDataChannelStats(std::string(id), timestamp_us) {}
> 
>-RTCDataChannelStats::RTCDataChannelStats(
>-    std::string&& id, int64_t timestamp_us)
>+RTCDataChannelStats::RTCDataChannelStats(std::string&& id, int64_t timestamp_us)
>     : RTCStats(std::move(id), timestamp_us),
>       label("label"),
>       protocol("protocol"),
>@@ -148,11 +133,9 @@ RTCDataChannelStats::RTCDataChannelStats(
>       messages_sent("messagesSent"),
>       bytes_sent("bytesSent"),
>       messages_received("messagesReceived"),
>-      bytes_received("bytesReceived") {
>-}
>+      bytes_received("bytesReceived") {}
> 
>-RTCDataChannelStats::RTCDataChannelStats(
>-    const RTCDataChannelStats& other)
>+RTCDataChannelStats::RTCDataChannelStats(const RTCDataChannelStats& other)
>     : RTCStats(other.id(), other.timestamp_us()),
>       label(other.label),
>       protocol(other.protocol),
>@@ -161,11 +144,9 @@ RTCDataChannelStats::RTCDataChannelStats(
>       messages_sent(other.messages_sent),
>       bytes_sent(other.bytes_sent),
>       messages_received(other.messages_received),
>-      bytes_received(other.bytes_received) {
>-}
>+      bytes_received(other.bytes_received) {}
> 
>-RTCDataChannelStats::~RTCDataChannelStats() {
>-}
>+RTCDataChannelStats::~RTCDataChannelStats() {}
> 
> // clang-format off
> WEBRTC_RTCSTATS_IMPL(RTCIceCandidatePairStats, RTCStats, "candidate-pair",
>@@ -195,13 +176,12 @@ WEBRTC_RTCSTATS_IMPL(RTCIceCandidatePairStats, RTCStats, "candidate-pair",
>     &consent_responses_sent);
> // clang-format on
> 
>-RTCIceCandidatePairStats::RTCIceCandidatePairStats(
>-    const std::string& id, int64_t timestamp_us)
>-    : RTCIceCandidatePairStats(std::string(id), timestamp_us) {
>-}
>+RTCIceCandidatePairStats::RTCIceCandidatePairStats(const std::string& id,
>+                                                   int64_t timestamp_us)
>+    : RTCIceCandidatePairStats(std::string(id), timestamp_us) {}
> 
>-RTCIceCandidatePairStats::RTCIceCandidatePairStats(
>-    std::string&& id, int64_t timestamp_us)
>+RTCIceCandidatePairStats::RTCIceCandidatePairStats(std::string&& id,
>+                                                   int64_t timestamp_us)
>     : RTCStats(std::move(id), timestamp_us),
>       transport_id("transportId"),
>       local_candidate_id("localCandidateId"),
>@@ -226,8 +206,7 @@ RTCIceCandidatePairStats::RTCIceCandidatePairStats(
>       consent_requests_received("consentRequestsReceived"),
>       consent_requests_sent("consentRequestsSent"),
>       consent_responses_received("consentResponsesReceived"),
>-      consent_responses_sent("consentResponsesSent") {
>-}
>+      consent_responses_sent("consentResponsesSent") {}
> 
> RTCIceCandidatePairStats::RTCIceCandidatePairStats(
>     const RTCIceCandidatePairStats& other)
>@@ -255,14 +234,12 @@ RTCIceCandidatePairStats::RTCIceCandidatePairStats(
>       consent_requests_received(other.consent_requests_received),
>       consent_requests_sent(other.consent_requests_sent),
>       consent_responses_received(other.consent_responses_received),
>-      consent_responses_sent(other.consent_responses_sent) {
>-}
>+      consent_responses_sent(other.consent_responses_sent) {}
> 
>-RTCIceCandidatePairStats::~RTCIceCandidatePairStats() {
>-}
>+RTCIceCandidatePairStats::~RTCIceCandidatePairStats() {}
> 
> // clang-format off
>-WEBRTC_RTCSTATS_IMPL(RTCIceCandidateStats, RTCStats, "ice-candidate",
>+WEBRTC_RTCSTATS_IMPL(RTCIceCandidateStats, RTCStats, "abstract-ice-candidate",
>     &transport_id,
>     &is_remote,
>     &network_type,
>@@ -275,10 +252,10 @@ WEBRTC_RTCSTATS_IMPL(RTCIceCandidateStats, RTCStats, "ice-candidate",
>     &deleted);
> // clang-format on
> 
>-RTCIceCandidateStats::RTCIceCandidateStats(
>-    const std::string& id, int64_t timestamp_us, bool is_remote)
>-    : RTCIceCandidateStats(std::string(id), timestamp_us, is_remote) {
>-}
>+RTCIceCandidateStats::RTCIceCandidateStats(const std::string& id,
>+                                           int64_t timestamp_us,
>+                                           bool is_remote)
>+    : RTCIceCandidateStats(std::string(id), timestamp_us, is_remote) {}
> 
> RTCIceCandidateStats::RTCIceCandidateStats(std::string&& id,
>                                            int64_t timestamp_us,
>@@ -308,19 +285,20 @@ RTCIceCandidateStats::RTCIceCandidateStats(const RTCIceCandidateStats& other)
>       url(other.url),
>       deleted(other.deleted) {}
> 
>-RTCIceCandidateStats::~RTCIceCandidateStats() {
>-}
>+RTCIceCandidateStats::~RTCIceCandidateStats() {}
> 
> const char RTCLocalIceCandidateStats::kType[] = "local-candidate";
> 
>-RTCLocalIceCandidateStats::RTCLocalIceCandidateStats(
>-    const std::string& id, int64_t timestamp_us)
>-    : RTCIceCandidateStats(id, timestamp_us, false) {
>-}
>+RTCLocalIceCandidateStats::RTCLocalIceCandidateStats(const std::string& id,
>+                                                     int64_t timestamp_us)
>+    : RTCIceCandidateStats(id, timestamp_us, false) {}
>+
>+RTCLocalIceCandidateStats::RTCLocalIceCandidateStats(std::string&& id,
>+                                                     int64_t timestamp_us)
>+    : RTCIceCandidateStats(std::move(id), timestamp_us, false) {}
> 
>-RTCLocalIceCandidateStats::RTCLocalIceCandidateStats(
>-    std::string&& id, int64_t timestamp_us)
>-    : RTCIceCandidateStats(std::move(id), timestamp_us, false) {
>+std::unique_ptr<RTCStats> RTCLocalIceCandidateStats::copy() const {
>+  return std::unique_ptr<RTCStats>(new RTCLocalIceCandidateStats(*this));
> }
> 
> const char* RTCLocalIceCandidateStats::type() const {
>@@ -329,14 +307,16 @@ const char* RTCLocalIceCandidateStats::type() const {
> 
> const char RTCRemoteIceCandidateStats::kType[] = "remote-candidate";
> 
>-RTCRemoteIceCandidateStats::RTCRemoteIceCandidateStats(
>-    const std::string& id, int64_t timestamp_us)
>-    : RTCIceCandidateStats(id, timestamp_us, true) {
>-}
>+RTCRemoteIceCandidateStats::RTCRemoteIceCandidateStats(const std::string& id,
>+                                                       int64_t timestamp_us)
>+    : RTCIceCandidateStats(id, timestamp_us, true) {}
>+
>+RTCRemoteIceCandidateStats::RTCRemoteIceCandidateStats(std::string&& id,
>+                                                       int64_t timestamp_us)
>+    : RTCIceCandidateStats(std::move(id), timestamp_us, true) {}
> 
>-RTCRemoteIceCandidateStats::RTCRemoteIceCandidateStats(
>-    std::string&& id, int64_t timestamp_us)
>-    : RTCIceCandidateStats(std::move(id), timestamp_us, true) {
>+std::unique_ptr<RTCStats> RTCRemoteIceCandidateStats::copy() const {
>+  return std::unique_ptr<RTCStats>(new RTCRemoteIceCandidateStats(*this));
> }
> 
> const char* RTCRemoteIceCandidateStats::type() const {
>@@ -349,27 +329,21 @@ WEBRTC_RTCSTATS_IMPL(RTCMediaStreamStats, RTCStats, "stream",
>     &track_ids);
> // clang-format on
> 
>-RTCMediaStreamStats::RTCMediaStreamStats(
>-    const std::string& id, int64_t timestamp_us)
>-    : RTCMediaStreamStats(std::string(id), timestamp_us) {
>-}
>+RTCMediaStreamStats::RTCMediaStreamStats(const std::string& id,
>+                                         int64_t timestamp_us)
>+    : RTCMediaStreamStats(std::string(id), timestamp_us) {}
> 
>-RTCMediaStreamStats::RTCMediaStreamStats(
>-    std::string&& id, int64_t timestamp_us)
>+RTCMediaStreamStats::RTCMediaStreamStats(std::string&& id, int64_t timestamp_us)
>     : RTCStats(std::move(id), timestamp_us),
>       stream_identifier("streamIdentifier"),
>-      track_ids("trackIds") {
>-}
>+      track_ids("trackIds") {}
> 
>-RTCMediaStreamStats::RTCMediaStreamStats(
>-    const RTCMediaStreamStats& other)
>+RTCMediaStreamStats::RTCMediaStreamStats(const RTCMediaStreamStats& other)
>     : RTCStats(other.id(), other.timestamp_us()),
>       stream_identifier(other.stream_identifier),
>-      track_ids(other.track_ids) {
>-}
>+      track_ids(other.track_ids) {}
> 
>-RTCMediaStreamStats::~RTCMediaStreamStats() {
>-}
>+RTCMediaStreamStats::~RTCMediaStreamStats() {}
> 
> // clang-format off
> WEBRTC_RTCSTATS_IMPL(RTCMediaStreamTrackStats, RTCStats, "track",
>@@ -383,6 +357,7 @@ WEBRTC_RTCSTATS_IMPL(RTCMediaStreamTrackStats, RTCStats, "track",
>                      &frame_height,
>                      &frames_per_second,
>                      &frames_sent,
>+                     &huge_frames_sent,
>                      &frames_received,
>                      &frames_decoded,
>                      &frames_dropped,
>@@ -399,10 +374,10 @@ WEBRTC_RTCSTATS_IMPL(RTCMediaStreamTrackStats, RTCStats, "track",
>                      &concealment_events);
> // clang-format on
> 
>-RTCMediaStreamTrackStats::RTCMediaStreamTrackStats(
>-    const std::string& id, int64_t timestamp_us, const char* kind)
>-    : RTCMediaStreamTrackStats(std::string(id), timestamp_us, kind) {
>-}
>+RTCMediaStreamTrackStats::RTCMediaStreamTrackStats(const std::string& id,
>+                                                   int64_t timestamp_us,
>+                                                   const char* kind)
>+    : RTCMediaStreamTrackStats(std::string(id), timestamp_us, kind) {}
> 
> RTCMediaStreamTrackStats::RTCMediaStreamTrackStats(std::string&& id,
>                                                    int64_t timestamp_us,
>@@ -418,6 +393,7 @@ RTCMediaStreamTrackStats::RTCMediaStreamTrackStats(std::string&& id,
>       frame_height("frameHeight"),
>       frames_per_second("framesPerSecond"),
>       frames_sent("framesSent"),
>+      huge_frames_sent("hugeFramesSent"),
>       frames_received("framesReceived"),
>       frames_decoded("framesDecoded"),
>       frames_dropped("framesDropped"),
>@@ -449,6 +425,7 @@ RTCMediaStreamTrackStats::RTCMediaStreamTrackStats(
>       frame_height(other.frame_height),
>       frames_per_second(other.frames_per_second),
>       frames_sent(other.frames_sent),
>+      huge_frames_sent(other.huge_frames_sent),
>       frames_received(other.frames_received),
>       frames_decoded(other.frames_decoded),
>       frames_dropped(other.frames_dropped),
>@@ -464,8 +441,7 @@ RTCMediaStreamTrackStats::RTCMediaStreamTrackStats(
>       concealed_samples(other.concealed_samples),
>       concealment_events(other.concealment_events) {}
> 
>-RTCMediaStreamTrackStats::~RTCMediaStreamTrackStats() {
>-}
>+RTCMediaStreamTrackStats::~RTCMediaStreamTrackStats() {}
> 
> // clang-format off
> WEBRTC_RTCSTATS_IMPL(RTCPeerConnectionStats, RTCStats, "peer-connection",
>@@ -473,27 +449,23 @@ WEBRTC_RTCSTATS_IMPL(RTCPeerConnectionStats, RTCStats, "peer-connection",
>     &data_channels_closed);
> // clang-format on
> 
>-RTCPeerConnectionStats::RTCPeerConnectionStats(
>-    const std::string& id, int64_t timestamp_us)
>-    : RTCPeerConnectionStats(std::string(id), timestamp_us) {
>-}
>+RTCPeerConnectionStats::RTCPeerConnectionStats(const std::string& id,
>+                                               int64_t timestamp_us)
>+    : RTCPeerConnectionStats(std::string(id), timestamp_us) {}
> 
>-RTCPeerConnectionStats::RTCPeerConnectionStats(
>-    std::string&& id, int64_t timestamp_us)
>+RTCPeerConnectionStats::RTCPeerConnectionStats(std::string&& id,
>+                                               int64_t timestamp_us)
>     : RTCStats(std::move(id), timestamp_us),
>       data_channels_opened("dataChannelsOpened"),
>-      data_channels_closed("dataChannelsClosed") {
>-}
>+      data_channels_closed("dataChannelsClosed") {}
> 
> RTCPeerConnectionStats::RTCPeerConnectionStats(
>     const RTCPeerConnectionStats& other)
>     : RTCStats(other.id(), other.timestamp_us()),
>       data_channels_opened(other.data_channels_opened),
>-      data_channels_closed(other.data_channels_closed) {
>-}
>+      data_channels_closed(other.data_channels_closed) {}
> 
>-RTCPeerConnectionStats::~RTCPeerConnectionStats() {
>-}
>+RTCPeerConnectionStats::~RTCPeerConnectionStats() {}
> 
> // clang-format off
> WEBRTC_RTCSTATS_IMPL(RTCRTPStreamStats, RTCStats, "rtp",
>@@ -511,13 +483,11 @@ WEBRTC_RTCSTATS_IMPL(RTCRTPStreamStats, RTCStats, "rtp",
>     &qp_sum);
> // clang-format on
> 
>-RTCRTPStreamStats::RTCRTPStreamStats(
>-    const std::string& id, int64_t timestamp_us)
>-    : RTCRTPStreamStats(std::string(id), timestamp_us) {
>-}
>+RTCRTPStreamStats::RTCRTPStreamStats(const std::string& id,
>+                                     int64_t timestamp_us)
>+    : RTCRTPStreamStats(std::string(id), timestamp_us) {}
> 
>-RTCRTPStreamStats::RTCRTPStreamStats(
>-    std::string&& id, int64_t timestamp_us)
>+RTCRTPStreamStats::RTCRTPStreamStats(std::string&& id, int64_t timestamp_us)
>     : RTCStats(std::move(id), timestamp_us),
>       ssrc("ssrc"),
>       associate_stats_id("associateStatsId"),
>@@ -530,11 +500,9 @@ RTCRTPStreamStats::RTCRTPStreamStats(
>       pli_count("pliCount"),
>       nack_count("nackCount"),
>       sli_count("sliCount"),
>-      qp_sum("qpSum") {
>-}
>+      qp_sum("qpSum") {}
> 
>-RTCRTPStreamStats::RTCRTPStreamStats(
>-    const RTCRTPStreamStats& other)
>+RTCRTPStreamStats::RTCRTPStreamStats(const RTCRTPStreamStats& other)
>     : RTCStats(other.id(), other.timestamp_us()),
>       ssrc(other.ssrc),
>       associate_stats_id(other.associate_stats_id),
>@@ -547,11 +515,9 @@ RTCRTPStreamStats::RTCRTPStreamStats(
>       pli_count(other.pli_count),
>       nack_count(other.nack_count),
>       sli_count(other.sli_count),
>-      qp_sum(other.qp_sum) {
>-}
>+      qp_sum(other.qp_sum) {}
> 
>-RTCRTPStreamStats::~RTCRTPStreamStats() {
>-}
>+RTCRTPStreamStats::~RTCRTPStreamStats() {}
> 
> // clang-format off
> WEBRTC_RTCSTATS_IMPL(
>@@ -575,13 +541,12 @@ WEBRTC_RTCSTATS_IMPL(
>     &frames_decoded);
> // clang-format on
> 
>-RTCInboundRTPStreamStats::RTCInboundRTPStreamStats(
>-    const std::string& id, int64_t timestamp_us)
>-    : RTCInboundRTPStreamStats(std::string(id), timestamp_us) {
>-}
>+RTCInboundRTPStreamStats::RTCInboundRTPStreamStats(const std::string& id,
>+                                                   int64_t timestamp_us)
>+    : RTCInboundRTPStreamStats(std::string(id), timestamp_us) {}
> 
>-RTCInboundRTPStreamStats::RTCInboundRTPStreamStats(
>-    std::string&& id, int64_t timestamp_us)
>+RTCInboundRTPStreamStats::RTCInboundRTPStreamStats(std::string&& id,
>+                                                   int64_t timestamp_us)
>     : RTCRTPStreamStats(std::move(id), timestamp_us),
>       packets_received("packetsReceived"),
>       bytes_received("bytesReceived"),
>@@ -599,8 +564,7 @@ RTCInboundRTPStreamStats::RTCInboundRTPStreamStats(
>       burst_discard_rate("burstDiscardRate"),
>       gap_loss_rate("gapLossRate"),
>       gap_discard_rate("gapDiscardRate"),
>-      frames_decoded("framesDecoded") {
>-}
>+      frames_decoded("framesDecoded") {}
> 
> RTCInboundRTPStreamStats::RTCInboundRTPStreamStats(
>     const RTCInboundRTPStreamStats& other)
>@@ -621,11 +585,9 @@ RTCInboundRTPStreamStats::RTCInboundRTPStreamStats(
>       burst_discard_rate(other.burst_discard_rate),
>       gap_loss_rate(other.gap_loss_rate),
>       gap_discard_rate(other.gap_discard_rate),
>-      frames_decoded(other.frames_decoded) {
>-}
>+      frames_decoded(other.frames_decoded) {}
> 
>-RTCInboundRTPStreamStats::~RTCInboundRTPStreamStats() {
>-}
>+RTCInboundRTPStreamStats::~RTCInboundRTPStreamStats() {}
> 
> // clang-format off
> WEBRTC_RTCSTATS_IMPL(
>@@ -636,19 +598,17 @@ WEBRTC_RTCSTATS_IMPL(
>     &frames_encoded);
> // clang-format on
> 
>-RTCOutboundRTPStreamStats::RTCOutboundRTPStreamStats(
>-    const std::string& id, int64_t timestamp_us)
>-    : RTCOutboundRTPStreamStats(std::string(id), timestamp_us) {
>-}
>+RTCOutboundRTPStreamStats::RTCOutboundRTPStreamStats(const std::string& id,
>+                                                     int64_t timestamp_us)
>+    : RTCOutboundRTPStreamStats(std::string(id), timestamp_us) {}
> 
>-RTCOutboundRTPStreamStats::RTCOutboundRTPStreamStats(
>-    std::string&& id, int64_t timestamp_us)
>+RTCOutboundRTPStreamStats::RTCOutboundRTPStreamStats(std::string&& id,
>+                                                     int64_t timestamp_us)
>     : RTCRTPStreamStats(std::move(id), timestamp_us),
>       packets_sent("packetsSent"),
>       bytes_sent("bytesSent"),
>       target_bitrate("targetBitrate"),
>-      frames_encoded("framesEncoded") {
>-}
>+      frames_encoded("framesEncoded") {}
> 
> RTCOutboundRTPStreamStats::RTCOutboundRTPStreamStats(
>     const RTCOutboundRTPStreamStats& other)
>@@ -656,11 +616,9 @@ RTCOutboundRTPStreamStats::RTCOutboundRTPStreamStats(
>       packets_sent(other.packets_sent),
>       bytes_sent(other.bytes_sent),
>       target_bitrate(other.target_bitrate),
>-      frames_encoded(other.frames_encoded) {
>-}
>+      frames_encoded(other.frames_encoded) {}
> 
>-RTCOutboundRTPStreamStats::~RTCOutboundRTPStreamStats() {
>-}
>+RTCOutboundRTPStreamStats::~RTCOutboundRTPStreamStats() {}
> 
> // clang-format off
> WEBRTC_RTCSTATS_IMPL(RTCTransportStats, RTCStats, "transport",
>@@ -673,13 +631,11 @@ WEBRTC_RTCSTATS_IMPL(RTCTransportStats, RTCStats, "transport",
>     &remote_certificate_id);
> // clang-format on
> 
>-RTCTransportStats::RTCTransportStats(
>-    const std::string& id, int64_t timestamp_us)
>-    : RTCTransportStats(std::string(id), timestamp_us) {
>-}
>+RTCTransportStats::RTCTransportStats(const std::string& id,
>+                                     int64_t timestamp_us)
>+    : RTCTransportStats(std::string(id), timestamp_us) {}
> 
>-RTCTransportStats::RTCTransportStats(
>-    std::string&& id, int64_t timestamp_us)
>+RTCTransportStats::RTCTransportStats(std::string&& id, int64_t timestamp_us)
>     : RTCStats(std::move(id), timestamp_us),
>       bytes_sent("bytesSent"),
>       bytes_received("bytesReceived"),
>@@ -687,11 +643,9 @@ RTCTransportStats::RTCTransportStats(
>       dtls_state("dtlsState"),
>       selected_candidate_pair_id("selectedCandidatePairId"),
>       local_certificate_id("localCertificateId"),
>-      remote_certificate_id("remoteCertificateId") {
>-}
>+      remote_certificate_id("remoteCertificateId") {}
> 
>-RTCTransportStats::RTCTransportStats(
>-    const RTCTransportStats& other)
>+RTCTransportStats::RTCTransportStats(const RTCTransportStats& other)
>     : RTCStats(other.id(), other.timestamp_us()),
>       bytes_sent(other.bytes_sent),
>       bytes_received(other.bytes_received),
>@@ -699,10 +653,8 @@ RTCTransportStats::RTCTransportStats(
>       dtls_state(other.dtls_state),
>       selected_candidate_pair_id(other.selected_candidate_pair_id),
>       local_certificate_id(other.local_certificate_id),
>-      remote_certificate_id(other.remote_certificate_id) {
>-}
>+      remote_certificate_id(other.remote_certificate_id) {}
> 
>-RTCTransportStats::~RTCTransportStats() {
>-}
>+RTCTransportStats::~RTCTransportStats() {}
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/stats/rtcstats_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/stats/rtcstats_unittest.cc
>index 7148c9fc2f4..de3e18c4e05 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/stats/rtcstats_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/stats/rtcstats_unittest.cc
>@@ -42,28 +42,27 @@ class RTCChildStats : public RTCStats {
>   WEBRTC_RTCSTATS_DECL();
> 
>   RTCChildStats(const std::string& id, int64_t timestamp_us)
>-      : RTCStats(id, timestamp_us),
>-        child_int("childInt") {}
>+      : RTCStats(id, timestamp_us), child_int("childInt") {}
> 
>   RTCStatsMember<int32_t> child_int;
> };
> 
>-WEBRTC_RTCSTATS_IMPL(RTCChildStats, RTCStats, "child-stats",
>-    &child_int);
>+WEBRTC_RTCSTATS_IMPL(RTCChildStats, RTCStats, "child-stats", &child_int);
> 
> class RTCGrandChildStats : public RTCChildStats {
>  public:
>   WEBRTC_RTCSTATS_DECL();
> 
>   RTCGrandChildStats(const std::string& id, int64_t timestamp_us)
>-      : RTCChildStats(id, timestamp_us),
>-        grandchild_int("grandchildInt") {}
>+      : RTCChildStats(id, timestamp_us), grandchild_int("grandchildInt") {}
> 
>   RTCStatsMember<int32_t> grandchild_int;
> };
> 
>-WEBRTC_RTCSTATS_IMPL(RTCGrandChildStats, RTCChildStats, "grandchild-stats",
>-      &grandchild_int);
>+WEBRTC_RTCSTATS_IMPL(RTCGrandChildStats,
>+                     RTCChildStats,
>+                     "grandchild-stats",
>+                     &grandchild_int);
> 
> TEST(RTCStatsTest, RTCStatsAndMembers) {
>   RTCTestStats stats("testId", 42);
>@@ -123,7 +122,7 @@ TEST(RTCStatsTest, RTCStatsAndMembers) {
>   EXPECT_EQ(*stats.m_sequence_double, sequence_double);
>   EXPECT_EQ(*stats.m_sequence_string, sequence_string);
> 
>-  int32_t numbers[] = { 4, 8, 15, 16, 23, 42 };
>+  int32_t numbers[] = {4, 8, 15, 16, 23, 42};
>   std::vector<int32_t> numbers_sequence(&numbers[0], &numbers[6]);
>   stats.m_sequence_int32->clear();
>   stats.m_sequence_int32->insert(stats.m_sequence_int32->end(),
>@@ -156,11 +155,11 @@ TEST(RTCStatsTest, EqualityOperator) {
>   EXPECT_NE(stats_with_all_values.m_int32, stats_with_all_values.m_uint32);
> 
>   RTCTestStats one_member_different[] = {
>-    stats_with_all_values, stats_with_all_values, stats_with_all_values,
>-    stats_with_all_values, stats_with_all_values, stats_with_all_values,
>-    stats_with_all_values, stats_with_all_values, stats_with_all_values,
>-    stats_with_all_values, stats_with_all_values, stats_with_all_values,
>-    stats_with_all_values, stats_with_all_values,
>+      stats_with_all_values, stats_with_all_values, stats_with_all_values,
>+      stats_with_all_values, stats_with_all_values, stats_with_all_values,
>+      stats_with_all_values, stats_with_all_values, stats_with_all_values,
>+      stats_with_all_values, stats_with_all_values, stats_with_all_values,
>+      stats_with_all_values, stats_with_all_values,
>   };
>   for (size_t i = 0; i < 14; ++i) {
>     EXPECT_EQ(stats_with_all_values, one_member_different[i]);
>@@ -331,6 +330,13 @@ TEST(RTCStatsTest, RTCStatsPrintsValidJson) {
>   std::cout << stats.ToJson() << std::endl;
> }
> 
>+TEST(RTCStatsTest, IsStandardized) {
>+  RTCStatsMember<int32_t> standardized("standardized");
>+  RTCNonStandardStatsMember<int32_t> unstandardized("unstandardized");
>+  EXPECT_TRUE(standardized.is_standardized());
>+  EXPECT_FALSE(unstandardized.is_standardized());
>+}
>+
> // Death tests.
> // Disabled on Android because death tests misbehave on Android, see
> // base/test/gtest_util.h.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/stats/rtcstatsreport.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/stats/rtcstatsreport.cc
>index 90025f7120b..33f6b6a121c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/stats/rtcstatsreport.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/stats/rtcstatsreport.cc
>@@ -17,17 +17,12 @@ namespace webrtc {
> RTCStatsReport::ConstIterator::ConstIterator(
>     const rtc::scoped_refptr<const RTCStatsReport>& report,
>     StatsMap::const_iterator it)
>-    : report_(report),
>-      it_(it) {
>-}
>+    : report_(report), it_(it) {}
> 
> RTCStatsReport::ConstIterator::ConstIterator(const ConstIterator&& other)
>-    : report_(std::move(other.report_)),
>-      it_(std::move(other.it_)) {
>-}
>+    : report_(std::move(other.report_)), it_(std::move(other.it_)) {}
> 
>-RTCStatsReport::ConstIterator::~ConstIterator() {
>-}
>+RTCStatsReport::ConstIterator::~ConstIterator() {}
> 
> RTCStatsReport::ConstIterator& RTCStatsReport::ConstIterator::operator++() {
>   ++it_;
>@@ -63,18 +58,25 @@ rtc::scoped_refptr<RTCStatsReport> RTCStatsReport::Create(
> }
> 
> RTCStatsReport::RTCStatsReport(int64_t timestamp_us)
>-    : timestamp_us_(timestamp_us) {
>-}
>+    : timestamp_us_(timestamp_us) {}
> 
>-RTCStatsReport::~RTCStatsReport() {
>+RTCStatsReport::~RTCStatsReport() {}
>+
>+rtc::scoped_refptr<RTCStatsReport> RTCStatsReport::Copy() const {
>+  rtc::scoped_refptr<RTCStatsReport> copy = Create(timestamp_us_);
>+  for (auto it = stats_.begin(); it != stats_.end(); ++it) {
>+    copy->AddStats(it->second->copy());
>+  }
>+  return copy;
> }
> 
> void RTCStatsReport::AddStats(std::unique_ptr<const RTCStats> stats) {
>-  auto result = stats_.insert(std::make_pair(std::string(stats->id()),
>-                              std::move(stats)));
>-  RTC_DCHECK(result.second) <<
>-      "A stats object with ID " << result.first->second->id() << " is already "
>-      "present in this stats report.";
>+  auto result =
>+      stats_.insert(std::make_pair(std::string(stats->id()), std::move(stats)));
>+  RTC_DCHECK(result.second)
>+      << "A stats object with ID " << result.first->second->id()
>+      << " is already "
>+         "present in this stats report.";
> }
> 
> const RTCStats* RTCStatsReport::Get(const std::string& id) const {
>@@ -84,6 +86,15 @@ const RTCStats* RTCStatsReport::Get(const std::string& id) const {
>   return nullptr;
> }
> 
>+std::unique_ptr<const RTCStats> RTCStatsReport::Take(const std::string& id) {
>+  StatsMap::iterator it = stats_.find(id);
>+  if (it == stats_.end())
>+    return nullptr;
>+  std::unique_ptr<const RTCStats> stats = std::move(it->second);
>+  stats_.erase(it);
>+  return stats;
>+}
>+
> void RTCStatsReport::TakeMembersFrom(
>     rtc::scoped_refptr<RTCStatsReport> victim) {
>   for (StatsMap::iterator it = victim->stats_.begin();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/stats/rtcstatsreport_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/stats/rtcstatsreport_unittest.cc
>index 5483ee3083f..4df99e175b5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/stats/rtcstatsreport_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/stats/rtcstatsreport_unittest.cc
>@@ -21,42 +21,36 @@ class RTCTestStats1 : public RTCStats {
>   WEBRTC_RTCSTATS_DECL();
> 
>   RTCTestStats1(const std::string& id, int64_t timestamp_us)
>-      : RTCStats(id, timestamp_us),
>-        integer("integer") {}
>+      : RTCStats(id, timestamp_us), integer("integer") {}
> 
>   RTCStatsMember<int32_t> integer;
> };
> 
>-WEBRTC_RTCSTATS_IMPL(RTCTestStats1, RTCStats, "test-stats-1",
>-    &integer);
>+WEBRTC_RTCSTATS_IMPL(RTCTestStats1, RTCStats, "test-stats-1", &integer);
> 
> class RTCTestStats2 : public RTCStats {
>  public:
>   WEBRTC_RTCSTATS_DECL();
> 
>   RTCTestStats2(const std::string& id, int64_t timestamp_us)
>-      : RTCStats(id, timestamp_us),
>-        number("number") {}
>+      : RTCStats(id, timestamp_us), number("number") {}
> 
>   RTCStatsMember<double> number;
> };
> 
>-WEBRTC_RTCSTATS_IMPL(RTCTestStats2, RTCStats, "test-stats-2",
>-    &number);
>+WEBRTC_RTCSTATS_IMPL(RTCTestStats2, RTCStats, "test-stats-2", &number);
> 
> class RTCTestStats3 : public RTCStats {
>  public:
>   WEBRTC_RTCSTATS_DECL();
> 
>   RTCTestStats3(const std::string& id, int64_t timestamp_us)
>-      : RTCStats(id, timestamp_us),
>-        string("string") {}
>+      : RTCStats(id, timestamp_us), string("string") {}
> 
>   RTCStatsMember<std::string> string;
> };
> 
>-WEBRTC_RTCSTATS_IMPL(RTCTestStats3, RTCStats, "test-stats-3",
>-    &string);
>+WEBRTC_RTCSTATS_IMPL(RTCTestStats3, RTCStats, "test-stats-3", &string);
> 
> TEST(RTCStatsReport, AddAndGetStats) {
>   rtc::scoped_refptr<RTCStatsReport> report = RTCStatsReport::Create(1337);
>@@ -110,6 +104,19 @@ TEST(RTCStatsReport, StatsOrder) {
>   EXPECT_EQ(i, static_cast<int64_t>(7));
> }
> 
>+TEST(RTCStatsReport, Take) {
>+  rtc::scoped_refptr<RTCStatsReport> report = RTCStatsReport::Create(0);
>+  report->AddStats(std::unique_ptr<RTCStats>(new RTCTestStats1("A", 1)));
>+  report->AddStats(std::unique_ptr<RTCStats>(new RTCTestStats1("B", 2)));
>+  EXPECT_TRUE(report->Get("A"));
>+  EXPECT_EQ(report->size(), 2u);
>+  auto a = report->Take("A");
>+  EXPECT_TRUE(a);
>+  EXPECT_EQ(report->size(), 1u);
>+  EXPECT_FALSE(report->Get("A"));
>+  EXPECT_FALSE(report->Take("A"));
>+}
>+
> TEST(RTCStatsReport, TakeMembersFrom) {
>   rtc::scoped_refptr<RTCStatsReport> a = RTCStatsReport::Create(1337);
>   EXPECT_EQ(a->timestamp_us(), 1337u);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/stats/test/rtcteststats.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/stats/test/rtcteststats.cc
>index 5b6dffa5b81..162f329802d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/stats/test/rtcteststats.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/stats/test/rtcteststats.cc
>@@ -12,21 +12,23 @@
> 
> namespace webrtc {
> 
>-WEBRTC_RTCSTATS_IMPL(RTCTestStats, RTCStats, "test-stats",
>-    &m_bool,
>-    &m_int32,
>-    &m_uint32,
>-    &m_int64,
>-    &m_uint64,
>-    &m_double,
>-    &m_string,
>-    &m_sequence_bool,
>-    &m_sequence_int32,
>-    &m_sequence_uint32,
>-    &m_sequence_int64,
>-    &m_sequence_uint64,
>-    &m_sequence_double,
>-    &m_sequence_string);
>+WEBRTC_RTCSTATS_IMPL(RTCTestStats,
>+                     RTCStats,
>+                     "test-stats",
>+                     &m_bool,
>+                     &m_int32,
>+                     &m_uint32,
>+                     &m_int64,
>+                     &m_uint64,
>+                     &m_double,
>+                     &m_string,
>+                     &m_sequence_bool,
>+                     &m_sequence_int32,
>+                     &m_sequence_uint32,
>+                     &m_sequence_int64,
>+                     &m_sequence_uint64,
>+                     &m_sequence_double,
>+                     &m_sequence_string);
> 
> RTCTestStats::RTCTestStats(const std::string& id, int64_t timestamp_us)
>     : RTCStats(id, timestamp_us),
>@@ -43,8 +45,7 @@ RTCTestStats::RTCTestStats(const std::string& id, int64_t timestamp_us)
>       m_sequence_int64("mSequenceInt64"),
>       m_sequence_uint64("mSequenceUint64"),
>       m_sequence_double("mSequenceDouble"),
>-      m_sequence_string("mSequenceString") {
>-}
>+      m_sequence_string("mSequenceString") {}
> 
> RTCTestStats::RTCTestStats(const RTCTestStats& other)
>     : RTCStats(other.id(), other.timestamp_us()),
>@@ -61,10 +62,8 @@ RTCTestStats::RTCTestStats(const RTCTestStats& other)
>       m_sequence_int64(other.m_sequence_int64),
>       m_sequence_uint64(other.m_sequence_uint64),
>       m_sequence_double(other.m_sequence_double),
>-      m_sequence_string(other.m_sequence_string) {
>-}
>+      m_sequence_string(other.m_sequence_string) {}
> 
>-RTCTestStats::~RTCTestStats() {
>-}
>+RTCTestStats::~RTCTestStats() {}
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/style-guide.md b/Source/ThirdParty/libwebrtc/Source/webrtc/style-guide.md
>index d8fbaae6d49..50f45fc6e88 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/style-guide.md
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/style-guide.md
>@@ -20,6 +20,40 @@ both.
> [chr-style]: https://chromium.googlesource.com/chromium/src/+/HEAD/styleguide/c++/c++.md
> [goog-style]: https://google.github.io/styleguide/cppguide.html
> 
>+### C++ version
>+
>+WebRTC is written in C++11, but with some restrictions:
>+
>+* We only allow the subset of C++11 (language and library) in the
>+  âallowedâ section of [this Chromium page][chromium-cpp11].
>+* We only allow the subset of C++11 that is also valid C++14;
>+  otherwise, users would not be able to compile WebRTC in C++14 mode.
>+
>+[chromium-cpp11]: https://chromium-cpp.appspot.com/
>+
>+### <a name="h-cc-pairs"></a>`.h` and `.cc` files come in pairs
>+
>+`.h` and `.cc` files should come in pairs, with the same name (except
>+for the file type suffix), in the same directory, in the same build
>+target.
>+
>+* If a declaration in `path/to/foo.h` has a definition in some `.cc`
>+  file, it should be in `path/to/foo.cc`.
>+* If a definition in `path/to/foo.cc` file has a declaration in some
>+  `.h` file, it should be in `path/to/foo.h`.
>+* Omit the `.cc` file if it would have been empty, but still list the
>+  `.h` file in a build target.
>+* Omit the `.h` file if it would have been empty. (This can happen
>+  with unit test `.cc` files, and with `.cc` files that define
>+  `main`.)
>+
>+This makes the source code easier to navigate and organize, and
>+precludes some questionable build system practices such as having
>+build targets that donât pull in definitions for everything they
>+declare.
>+
>+[Examples and exceptions](style-guide/h-cc-pairs.md).
>+
> ### ArrayView
> 
> When passing an array of values to a function, use `rtc::ArrayView`
>@@ -34,7 +68,7 @@ instead of                          | use
> `const T* ptr, size_t num_elements` | `ArrayView<const T>`
> `T* ptr, size_t num_elements`       | `ArrayView<T>`
> 
>-See [the source](webrtc/api/array_view.h) for more detailed docs.
>+See [the source](api/array_view.h) for more detailed docs.
> 
> ### sigslot
> 
>@@ -75,6 +109,26 @@ implements the signal. It may:
>   sigslot::signal<int>& SignalFoo() { return bar_.SignalFoo(); }
>   ```
> 
>+### std::bind
>+
>+Donât use `std::bind`âthere are pitfalls, and lambdas are almost as
>+succinct and already familiar to modern C++ programmers.
>+
>+### std::function
>+
>+`std::function` is allowed, but remember that itâs not the right tool
>+for every occasion. Prefer to use interfaces when that makes sense,
>+and consider `rtc::FunctionView` for cases where the callee will not
>+save the function object.
>+
>+### Forward declarations
>+
>+WebRTC follows the [Google][goog-forward-declarations] C++ style guide
>+with respect to forward declarations. In summary: avoid using forward
>+declarations where possible; just `#include` the headers you need.
>+
>+[goog-forward-declarations]: https://google.github.io/styleguide/cppguide.html#Forward_Declarations
>+
> ## **C**
> 
> Thereâs a substantial chunk of legacy C code in WebRTC, and a lot of
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/style-guide/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/style-guide/OWNERS
>new file mode 100644
>index 00000000000..22ccb83d440
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/style-guide/OWNERS
>@@ -0,0 +1,2 @@
>+danilchap@webrtc.org
>+kwiberg@webrtc.org
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/style-guide/h-cc-pairs.md b/Source/ThirdParty/libwebrtc/Source/webrtc/style-guide/h-cc-pairs.md
>new file mode 100644
>index 00000000000..1a24e49d09b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/style-guide/h-cc-pairs.md
>@@ -0,0 +1,29 @@
>+# `.h` and `.cc` files come in pairs
>+
>+This is an overflow page for [this](../style-guide.md#h-cc-pairs)
>+style rule.
>+
>+## Example violations
>+
>+Example violations, which should be avoided in new code:
>+
>+* Declarations in `path/to/include/foo.h`, definitions in
>+  `path/to/source/foo.cc`. **Fix:** The `.h` and `.cc` files should be
>+  in the same directory.
>+* Declarations in `foo.h`, definitions in both `foo_bar.cc` and
>+  `foo_baz.cc`. **Fix:** The `.h` and `.cc` files should come in
>+  pairs, so either split `foo.h` into `foo_bar.h` and `foo_baz.h`, or
>+  merge `foo_bar.cc` and `foo_baz.cc` into `foo.cc`.
>+
>+## Exception for platform-specific code
>+
>+If the functions in a header file need different implementations for
>+different platforms, we allow the following arrangement:
>+
>+* Declarations in `foo.h`.
>+* A complete set of matching definitions in `foo_win.cc`, another
>+  complete set of matching definitions in `foo_mac.cc`, and so on.
>+* As per the main rule, these files should all be in the same
>+  directory and in the same build target. The build target should use
>+  platform conditionals to ensure that exactly one of the `.cc` files
>+  are included.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/BUILD.gn
>index 2e8d71f63de..106c63b0556 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/BUILD.gn
>@@ -15,35 +15,20 @@ import("../webrtc.gni")
> rtc_static_library("system_wrappers") {
>   visibility = [ "*" ]
>   sources = [
>-    "include/aligned_array.h",
>-    "include/aligned_malloc.h",
>     "include/clock.h",
>     "include/cpu_info.h",
>     "include/event_wrapper.h",
>-    "include/file_wrapper.h",
>     "include/ntp_time.h",
>     "include/rtp_to_ntp_estimator.h",
>-    "include/rw_lock_wrapper.h",
>     "include/sleep.h",
>-    "include/timestamp_extrapolator.h",
>-    "source/aligned_malloc.cc",
>     "source/clock.cc",
>     "source/cpu_features.cc",
>     "source/cpu_info.cc",
>     "source/event.cc",
>-    "source/event_timer_posix.cc",
>-    "source/event_timer_posix.h",
>     "source/event_timer_win.cc",
>     "source/event_timer_win.h",
>-    "source/file_impl.cc",
>     "source/rtp_to_ntp_estimator.cc",
>-    "source/rw_lock.cc",
>-    "source/rw_lock_posix.cc",
>-    "source/rw_lock_posix.h",
>-    "source/rw_lock_win.cc",
>-    "source/rw_lock_win.h",
>     "source/sleep.cc",
>-    "source/timestamp_extrapolator.cc",
>   ]
> 
>   defines = []
>@@ -54,12 +39,20 @@ rtc_static_library("system_wrappers") {
>     ":metrics_api",
>     ":runtime_enabled_features_api",
>     "..:webrtc_common",
>-    "../:typedefs",
>-    "../api:optional",
>     "../modules:module_api_public",
>     "../rtc_base:checks",
>+    "../rtc_base/synchronization:rw_lock_wrapper",
>+    "../rtc_base/system:arch",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> 
>+  if (is_posix || is_fuchsia) {
>+    sources += [
>+      "source/event_timer_posix.cc",
>+      "source/event_timer_posix.h",
>+    ]
>+  }
>+
>   if (is_android) {
>     defines += [ "WEBRTC_THREAD_RR" ]
> 
>@@ -90,24 +83,14 @@ rtc_static_library("system_wrappers") {
>     defines += [ "WEBRTC_THREAD_RR" ]
>   }
> 
>-  # TODO(jschuh): Bug 1348: fix this warning.
>-  configs += [ "//build/config/compiler:no_size_t_to_int_warning" ]
>-
>   if (is_win) {
>     libs += [ "winmm.lib" ]
> 
>-    cflags = [ "/wd4334" ]  # Ignore warning on shift operator promotion.
>-
>     # Windows needs ../rtc_base:rtc_base due to include of
>     # webrtc/rtc_base/win32.h in source/clock.cc.
>     deps += [ "../rtc_base:rtc_base" ]
>   }
> 
>-  if (is_win && is_clang) {
>-    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>-    suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-  }
>-
>   deps += [
>     "../rtc_base:rtc_base_approved",
>     "../rtc_base:rtc_numerics",
>@@ -120,13 +103,6 @@ rtc_source_set("cpu_features_api") {
>   ]
>   deps = [
>     "..:webrtc_common",
>-    "../:typedefs",
>-  ]
>-}
>-
>-rtc_source_set("asm_defines") {
>-  sources = [
>-    "include/asm_defines.h",
>   ]
> }
> 
>@@ -137,6 +113,7 @@ rtc_source_set("field_trial_api") {
> }
> 
> rtc_source_set("runtime_enabled_features_api") {
>+  visibility = [ "*" ]
>   sources = [
>     "include/runtime_enabled_features.h",
>   ]
>@@ -166,6 +143,7 @@ rtc_source_set("field_trial_default") {
> }
> 
> rtc_source_set("runtime_enabled_features_default") {
>+  visibility = [ "*" ]
>   sources = [
>     "source/runtime_enabled_features_default.cc",
>   ]
>@@ -215,6 +193,7 @@ if (is_linux) {
>     ]
>     deps = [
>       ":cpu_features_api",
>+      "../rtc_base/system:arch",
>     ]
>   }
> }
>@@ -223,20 +202,15 @@ if (rtc_include_tests) {
>   rtc_test("system_wrappers_unittests") {
>     testonly = true
>     sources = [
>-      "source/aligned_array_unittest.cc",
>-      "source/aligned_malloc_unittest.cc",
>       "source/clock_unittest.cc",
>-      "source/event_timer_posix_unittest.cc",
>       "source/metrics_default_unittest.cc",
>       "source/metrics_unittest.cc",
>       "source/ntp_time_unittest.cc",
>       "source/rtp_to_ntp_estimator_unittest.cc",
>     ]
>-    configs += [ "//build/config/compiler:no_size_t_to_int_warning" ]
> 
>-    if (!build_with_chromium && is_clang) {
>-      # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>-      suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>+    if (is_posix || is_fuchsia) {
>+      sources += [ "source/event_timer_posix_unittest.cc" ]
>     }
> 
>     deps = [
>@@ -244,7 +218,6 @@ if (rtc_include_tests) {
>       ":metrics_default",
>       ":system_wrappers",
>       "..:webrtc_common",
>-      "../:typedefs",
>       "../rtc_base:rtc_base_approved",
>       "../test:test_main",
>       "//testing/gtest",
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/OWNERS
>index ca991001573..13b08a36ef8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/OWNERS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/OWNERS
>@@ -1,8 +1,8 @@
>-perkj@webrtc.org
> henrika@webrtc.org
>-henrikg@webrtc.org
> mflodman@webrtc.org
> niklas.enbom@webrtc.org
>+nisse@webrtc.org
>+perkj@webrtc.org
> 
> # These are for the common case of adding or renaming files. If you're doing
> # structural changes, please get a review from a reviewer in this file.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/clock.h b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/clock.h
>index aec5ca5be90..4b6eab8edce 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/clock.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/clock.h
>@@ -13,9 +13,8 @@
> 
> #include <memory>
> 
>+#include "rtc_base/synchronization/rw_lock_wrapper.h"
> #include "system_wrappers/include/ntp_time.h"
>-#include "system_wrappers/include/rw_lock_wrapper.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/cpu_features_wrapper.h b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/cpu_features_wrapper.h
>index 07ee912cb2b..739161afcaf 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/cpu_features_wrapper.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/cpu_features_wrapper.h
>@@ -11,12 +11,12 @@
> #ifndef SYSTEM_WRAPPERS_INCLUDE_CPU_FEATURES_WRAPPER_H_
> #define SYSTEM_WRAPPERS_INCLUDE_CPU_FEATURES_WRAPPER_H_
> 
>+#include <stdint.h>
>+
> #if defined(__cplusplus) || defined(c_plusplus)
> extern "C" {
> #endif
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>-
> // List of features in x86.
> typedef enum { kSSE2, kSSE3 } CPUFeature;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/cpu_info.h b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/cpu_info.h
>index dbd5d606aa3..ab546c72141 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/cpu_info.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/cpu_info.h
>@@ -11,7 +11,7 @@
> #ifndef SYSTEM_WRAPPERS_INCLUDE_CPU_INFO_H_
> #define SYSTEM_WRAPPERS_INCLUDE_CPU_INFO_H_
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include <stdint.h>
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/metrics.h b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/metrics.h
>index 13f2483b322..99b8194f1aa 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/metrics.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/metrics.h
>@@ -127,10 +127,13 @@
> 
> // Histogram for enumerators (evenly spaced buckets).
> // |boundary| should be above the max enumerator sample.
>+//
>+// TODO(qingsi): Refactor the default implementation given by RtcHistogram,
>+// which is already sparse, and remove the boundary argument from the macro.
> #define RTC_HISTOGRAM_ENUMERATION_SPARSE(name, sample, boundary) \
>-  RTC_HISTOGRAM_COMMON_BLOCK_SLOW(                               \
>+  RTC_HISTOGRAM_COMMON_BLOCK(                                    \
>       name, sample,                                              \
>-      webrtc::metrics::HistogramFactoryGetEnumeration(name, boundary))
>+      webrtc::metrics::SparseHistogramFactoryGetEnumeration(name, boundary))
> 
> // Histogram for percentage (evenly spaced buckets).
> #define RTC_HISTOGRAM_PERCENTAGE(name, sample) \
>@@ -263,6 +266,11 @@ Histogram* HistogramFactoryGetCountsLinear(const std::string& name,
> Histogram* HistogramFactoryGetEnumeration(const std::string& name,
>                                           int boundary);
> 
>+// Get sparse histogram for enumerators.
>+// |boundary| should be above the max enumerator sample.
>+Histogram* SparseHistogramFactoryGetEnumeration(const std::string& name,
>+                                                int boundary);
>+
> // Function for adding a |sample| to a histogram.
> void HistogramAdd(Histogram* histogram_pointer, int sample);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/rtp_to_ntp_estimator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/rtp_to_ntp_estimator.h
>index 7c0757c5463..d7009d8b1f4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/rtp_to_ntp_estimator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/include/rtp_to_ntp_estimator.h
>@@ -13,11 +13,10 @@
> 
> #include <list>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> #include "modules/include/module_common_types_public.h"
> #include "rtc_base/numerics/moving_median_filter.h"
> #include "system_wrappers/include/ntp_time.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> // Class for converting an RTP timestamp to the NTP domain in milliseconds.
>@@ -72,7 +71,7 @@ class RtpToNtpEstimator {
>   bool Estimate(int64_t rtp_timestamp, int64_t* rtp_timestamp_ms) const;
> 
>   // Returns estimated rtp to ntp linear transform parameters.
>-  const rtc::Optional<Parameters> params() const;
>+  const absl::optional<Parameters> params() const;
> 
>   static const int kMaxInvalidSamples = 3;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/clock.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/clock.cc
>index 21d963f5d7c..c9940fbe99e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/clock.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/clock.cc
>@@ -15,7 +15,7 @@
> // Windows needs to be included before mmsystem.h
> #include "rtc_base/win32.h"
> 
>-#include <MMSystem.h>
>+#include <mmsystem.h>
> 
> #elif defined(WEBRTC_POSIX)
> 
>@@ -25,8 +25,8 @@
> #endif  // defined(WEBRTC_POSIX)
> 
> #include "rtc_base/criticalsection.h"
>+#include "rtc_base/synchronization/rw_lock_wrapper.h"
> #include "rtc_base/timeutils.h"
>-#include "system_wrappers/include/rw_lock_wrapper.h"
> 
> namespace webrtc {
> 
>@@ -90,7 +90,7 @@ class WindowsRealTimeClock : public RealTimeClock {
>         num_timer_wraps_(0),
>         ref_point_(GetSystemReferencePoint()) {}
> 
>-  virtual ~WindowsRealTimeClock() {}
>+  ~WindowsRealTimeClock() override {}
> 
>  protected:
>   struct ReferencePoint {
>@@ -221,10 +221,9 @@ Clock* Clock::GetRealTimeClock() {
>   }
>   return g_shared_clock;
> #elif defined(WEBRTC_POSIX)
>-  static std::aligned_storage<sizeof(UnixRealTimeClock), std::alignment_of<UnixRealTimeClock>::value>::type clock_storage;
>-  static UnixRealTimeClock* clock = new (&clock_storage) UnixRealTimeClock;
>-  return clock;
>-#else  // defined(WEBRTC_POSIX)
>+  static UnixRealTimeClock clock;
>+  return &clock;
>+#else   // defined(WEBRTC_POSIX)
>   return nullptr;
> #endif  // !defined(WEBRTC_WIN) || defined(WEBRTC_POSIX)
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/cpu_features.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/cpu_features.cc
>index 7417f53d596..d433a77f70e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/cpu_features.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/cpu_features.cc
>@@ -10,13 +10,13 @@
> 
> // Parts of this file derived from Chromium's base/cpu.cc.
> 
>+#include "rtc_base/system/arch.h"
> #include "system_wrappers/include/cpu_features_wrapper.h"
> 
> #if defined(WEBRTC_ARCH_X86_FAMILY) && defined(_MSC_VER)
> #include <intrin.h>
> #endif
> 
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> // No CPU feature is available => straight C path.
> int GetCPUInfoNoASM(CPUFeature feature) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/cpu_features_linux.c b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/cpu_features_linux.c
>index 9c5645068a5..004de5a6a93 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/cpu_features_linux.c
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/cpu_features_linux.c
>@@ -22,6 +22,7 @@
> #include <errno.h>
> #include <link.h>
> #endif
>+#include "rtc_base/system/arch.h"
> #include "system_wrappers/include/cpu_features_wrapper.h"
> 
> #if defined(WEBRTC_ARCH_ARM_FAMILY)
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/cpu_info.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/cpu_info.cc
>index 0bfe01582f0..c17d59d7f7b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/cpu_info.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/cpu_info.cc
>@@ -12,10 +12,6 @@
> 
> #if defined(WEBRTC_WIN)
> #include <windows.h>
>-#include <winsock2.h>
>-#ifndef EXCLUDE_D3D9
>-#include <d3d9.h>
>-#endif
> #elif defined(WEBRTC_LINUX)
> #include <unistd.h>
> #endif
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/event_timer_posix.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/event_timer_posix.cc
>index 2a6a580c642..e79aa9900af 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/event_timer_posix.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/event_timer_posix.cc
>@@ -10,6 +10,10 @@
> 
> #include "system_wrappers/source/event_timer_posix.h"
> 
>+#if defined(WEBRTC_ANDROID)
>+#include <android/api-level.h>
>+#endif
>+
> #include <errno.h>
> #include <pthread.h>
> #include <signal.h>
>@@ -20,6 +24,17 @@
> 
> #include "rtc_base/checks.h"
> 
>+#if defined(HAVE_PTHREAD_COND_TIMEDWAIT_MONOTONIC)
>+// Chromium build is always defining this macro if __ANDROID_API__ < 20.
>+#undef HAVE_PTHREAD_COND_TIMEDWAIT_MONOTONIC
>+#endif
>+
>+#if defined(WEBRTC_ANDROID) && defined(__ANDROID_API__)
>+#define HAVE_PTHREAD_COND_TIMEDWAIT_MONOTONIC (__ANDROID_API__ < 21)
>+#else
>+#define HAVE_PTHREAD_COND_TIMEDWAIT_MONOTONIC 0
>+#endif
>+
> namespace webrtc {
> 
> // static
>@@ -48,8 +63,7 @@ EventTimerPosix::EventTimerPosix()
> // all supported Android platforms support pthread_condattr_setclock.
> // TODO(sprang): Add support for monotonic clock on Apple platforms.
> #if !(defined(WEBRTC_MAC) || defined(WEBRTC_IOS)) && \
>-    !(defined(WEBRTC_ANDROID) &&                     \
>-      defined(HAVE_PTHREAD_COND_TIMEDWAIT_MONOTONIC))
>+    !(defined(WEBRTC_ANDROID) && HAVE_PTHREAD_COND_TIMEDWAIT_MONOTONIC)
>   pthread_condattr_setclock(&cond_attr, CLOCK_MONOTONIC);
> #endif
>   pthread_cond_init(&cond_, &cond_attr);
>@@ -96,7 +110,7 @@ EventTypeWrapper EventTimerPosix::Wait(unsigned long timeout_ms) {
>         end_at.tv_nsec -= kNanosecondsPerSecond;
>       }
>       while (ret_val == 0 && !event_set_) {
>-#if defined(WEBRTC_ANDROID) && defined(HAVE_PTHREAD_COND_TIMEDWAIT_MONOTONIC)
>+#if defined(WEBRTC_ANDROID) && HAVE_PTHREAD_COND_TIMEDWAIT_MONOTONIC
>         ret_val = pthread_cond_timedwait_monotonic_np(&cond_, &mutex_, &end_at);
> #else
>         ret_val = pthread_cond_timedwait(&cond_, &mutex_, &end_at);
>@@ -129,7 +143,7 @@ EventTypeWrapper EventTimerPosix::Wait(timespec* end_at, bool reset_event) {
>   }
> 
>   while (ret_val == 0 && !event_set_) {
>-#if defined(WEBRTC_ANDROID) && defined(HAVE_PTHREAD_COND_TIMEDWAIT_MONOTONIC)
>+#if defined(WEBRTC_ANDROID) && HAVE_PTHREAD_COND_TIMEDWAIT_MONOTONIC
>     ret_val = pthread_cond_timedwait_monotonic_np(&cond_, &mutex_, end_at);
> #else
>     ret_val = pthread_cond_timedwait(&cond_, &mutex_, end_at);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/event_timer_posix_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/event_timer_posix_unittest.cc
>index e0c5cbc4778..88fd90aef89 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/event_timer_posix_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/event_timer_posix_unittest.cc
>@@ -35,7 +35,7 @@ class EventTimerPosixTest : public testing::Test, public EventTimerPosix {
>         main_event_(false, true),
>         process_thread_id_(0),
>         process_thread_(nullptr) {}
>-  virtual ~EventTimerPosixTest() {}
>+  ~EventTimerPosixTest() override {}
> 
>   rtc::PlatformThread* CreateThread() override {
>     EXPECT_TRUE(process_thread_ == nullptr);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/event_timer_win.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/event_timer_win.cc
>index b6a93feb822..c2ad9f372a2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/event_timer_win.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/event_timer_win.cc
>@@ -10,7 +10,7 @@
> 
> #include "system_wrappers/source/event_timer_win.h"
> 
>-#include "Mmsystem.h"
>+#include "mmsystem.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/event_timer_win.h b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/event_timer_win.h
>index 5631a3fbc4a..c99f8c66d66 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/event_timer_win.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/event_timer_win.h
>@@ -11,24 +11,24 @@
> #ifndef SYSTEM_WRAPPERS_SOURCE_EVENT_WIN_H_
> #define SYSTEM_WRAPPERS_SOURCE_EVENT_WIN_H_
> 
>+#include <stdint.h>
>+
> #include <windows.h>
> 
> #include "system_wrappers/include/event_wrapper.h"
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>-
> namespace webrtc {
> 
> class EventTimerWin : public EventTimerWrapper {
>  public:
>   EventTimerWin();
>-  virtual ~EventTimerWin();
>+  ~EventTimerWin() override;
> 
>-  virtual EventTypeWrapper Wait(unsigned long max_time);
>-  virtual bool Set();
>+  EventTypeWrapper Wait(unsigned long max_time) override;
>+  bool Set() override;
> 
>-  virtual bool StartTimer(bool periodic, unsigned long time);
>-  virtual bool StopTimer();
>+  bool StartTimer(bool periodic, unsigned long time) override;
>+  bool StopTimer() override;
> 
>  private:
>   HANDLE event_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/metrics_default.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/metrics_default.cc
>index fbb29563012..7b62c8103f1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/metrics_default.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/metrics_default.cc
>@@ -247,6 +247,12 @@ Histogram* HistogramFactoryGetEnumeration(const std::string& name,
>   return map->GetEnumerationHistogram(name, boundary);
> }
> 
>+// Our default implementation reuses the non-sparse histogram.
>+Histogram* SparseHistogramFactoryGetEnumeration(const std::string& name,
>+                                                int boundary) {
>+  return HistogramFactoryGetEnumeration(name, boundary);
>+}
>+
> // Fast path. Adds |sample| to cached |histogram_pointer|.
> void HistogramAdd(Histogram* histogram_pointer, int sample) {
>   RtcHistogram* ptr = reinterpret_cast<RtcHistogram*>(histogram_pointer);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/metrics_default_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/metrics_default_unittest.cc
>index fa253a9bfed..7491a2f513b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/metrics_default_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/metrics_default_unittest.cc
>@@ -54,7 +54,7 @@ class MetricsDefaultTest : public ::testing::Test {
>   MetricsDefaultTest() {}
> 
>  protected:
>-  virtual void SetUp() { metrics::Reset(); }
>+  void SetUp() override { metrics::Reset(); }
> };
> 
> TEST_F(MetricsDefaultTest, Reset) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/metrics_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/metrics_unittest.cc
>index 53d43cd2b18..4218ad751d3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/metrics_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/metrics_unittest.cc
>@@ -29,7 +29,7 @@ class MetricsTest : public ::testing::Test {
>   MetricsTest() {}
> 
>  protected:
>-  virtual void SetUp() { metrics::Reset(); }
>+  void SetUp() override { metrics::Reset(); }
> };
> 
> TEST_F(MetricsTest, InitiallyNoSamples) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/rtp_to_ntp_estimator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/rtp_to_ntp_estimator.cc
>index 21f64eca25a..730c4f6691f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/rtp_to_ntp_estimator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/rtp_to_ntp_estimator.cc
>@@ -21,7 +21,6 @@ const size_t kNumRtcpReportsToUse = 2;
> // Number of parameters samples used to smooth.
> const size_t kNumSamplesToSmooth = 20;
> 
>-
> // Calculates the RTP timestamp frequency from two pairs of NTP/RTP timestamps.
> bool CalculateFrequency(int64_t ntp_ms1,
>                         uint32_t rtp_timestamp1,
>@@ -194,9 +193,9 @@ bool RtpToNtpEstimator::Estimate(int64_t rtp_timestamp,
>   return true;
> }
> 
>-const rtc::Optional<RtpToNtpEstimator::Parameters> RtpToNtpEstimator::params()
>+const absl::optional<RtpToNtpEstimator::Parameters> RtpToNtpEstimator::params()
>     const {
>-  rtc::Optional<Parameters> res;
>+  absl::optional<Parameters> res;
>   if (params_calculated_) {
>     res.emplace(smoothing_filter_.GetFilteredValue());
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/runtime_enabled_features_default.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/runtime_enabled_features_default.cc
>index 9827e7347aa..1d040a91b0e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/runtime_enabled_features_default.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/system_wrappers/source/runtime_enabled_features_default.cc
>@@ -12,13 +12,8 @@
> 
> #include "rtc_base/flags.h"
> 
>-#include <functional>
>-#include <utility>
>-
> namespace flags {
>-DEFINE_bool(enable_dual_stream_mode,
>-            false,
>-            "Enables dual video stream mode.");
>+DEFINE_bool(enable_dual_stream_mode, false, "Enables dual video stream mode.");
> }
> 
> namespace webrtc {
>@@ -26,7 +21,7 @@ namespace runtime_enabled_features {
> 
> bool IsFeatureEnabled(std::string feature_name) {
>   if (feature_name == kDualStreamModeFeatureName)
>-    return flags::FLAG_enable_dual_stream_mode();
>+    return flags::FLAG_enable_dual_stream_mode;
>   return false;
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/test/BUILD.gn
>index 9a50e1cc79e..9b4ba418f71 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/BUILD.gn
>@@ -11,9 +11,6 @@ import("//build/config/ui.gni")
> if (is_android) {
>   import("//build/config/android/rules.gni")
> }
>-if (rtc_use_memcheck) {
>-  import("//tools_webrtc/valgrind/valgrind-webrtc.gni")
>-}
> 
> group("test") {
>   testonly = true
>@@ -36,6 +33,7 @@ group("test") {
> }
> 
> rtc_source_set("video_test_common") {
>+  visibility = [ "*" ]
>   testonly = true
>   sources = [
>     "fake_texture_frame.cc",
>@@ -60,16 +58,20 @@ rtc_source_set("video_test_common") {
> 
>   deps = [
>     "..:webrtc_common",
>-    "../api:optional",
>-    "../api:video_frame_api_i420",
>+    "../api:libjingle_peerconnection_api",
>+    "../api/video:video_frame",
>+    "../api/video:video_frame_i010",
>+    "../api/video:video_frame_i420",
>     "../api/video_codecs:video_codecs_api",
>     "../call:video_stream_api",
>     "../common_video",
>     "../media:rtc_media_base",
>     "../modules/video_capture:video_capture_module",
>-    "../rtc_base:rtc_base_approved",
>+    "../rtc_base:checks",
>+    "../rtc_base:rtc_base",
>     "../rtc_base:rtc_task_queue",
>     "../system_wrappers",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
> 
>@@ -93,12 +95,16 @@ rtc_source_set("rtp_test_utils") {
>     "..:webrtc_common",
>     "../api:array_view",
>     "../modules/rtp_rtcp",
>+    "../modules/rtp_rtcp:rtp_rtcp_format",
>+    "../rtc_base:checks",
>     "../rtc_base:rtc_base_approved",
>+    "../rtc_base/system:arch",
>     "//testing/gtest",
>   ]
> }
> 
> rtc_source_set("field_trial") {
>+  visibility = [ "*" ]
>   testonly = true
>   sources = [
>     "field_trial.cc",
>@@ -107,11 +113,24 @@ rtc_source_set("field_trial") {
> 
>   deps = [
>     "..:webrtc_common",
>-    "../system_wrappers",
>+    "../system_wrappers:field_trial_api",
>     "../system_wrappers:field_trial_default",
>   ]
> }
> 
>+rtc_source_set("perf_test") {
>+  visibility = [ "*" ]
>+  sources = [
>+    "testsupport/perf_test.cc",
>+    "testsupport/perf_test.h",
>+  ]
>+  deps = [
>+    "..:webrtc_common",
>+    "../api:array_view",
>+    "../rtc_base:rtc_base_approved",
>+  ]
>+}
>+
> if (is_ios) {
>   rtc_source_set("test_support_objc") {
>     testonly = true
>@@ -121,37 +140,45 @@ if (is_ios) {
>       "ios/test_support.mm",
>     ]
>     deps = [
>+      ":perf_test",
>       "../sdk:common_objc",
>     ]
>   }
> }
> 
>+config("suppress_warning_4373") {
>+  if (is_win) {
>+    cflags = [
>+      # MSVC has a bug which generates this warning when using mocks; see the
>+      # section on warning 4373 in he googlemock FAQ. This warning isn't the
>+      # least relevant for us, anyway.
>+      "/wd4373",
>+    ]
>+  }
>+}
>+
> rtc_source_set("test_support") {
>+  visibility = [ "*" ]
>   testonly = true
> 
>+  all_dependent_configs = [
>+    ":suppress_warning_4373",
>+    "//third_party/googletest:gmock_config",
>+    "//third_party/googletest:gtest_config",
>+  ]
>+
>   sources = [
>     "gmock.h",
>     "gtest.h",
>-    "testsupport/packet_reader.cc",
>-    "testsupport/packet_reader.h",
>-    "testsupport/perf_test.cc",
>-    "testsupport/perf_test.h",
>-    "testsupport/unittest_utils.h",
>   ]
> 
>-  public_deps = [
>-    ":fileutils",
>-  ]
>+  public_deps = []
>   if (is_ios) {
>     public_deps += [ ":test_support_objc" ]
>   }
> 
>   deps = [
>-    "..:webrtc_common",
>-    "../common_video",
>-    "../rtc_base:gtest_prod",
>     "../rtc_base:rtc_base_approved",
>-    "../system_wrappers",
>     "//testing/gmock",
>     "//testing/gtest",
>   ]
>@@ -161,21 +188,14 @@ rtc_source_set("test_support") {
>     suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>   }
> 
>-  if (is_android) {
>-    deps += [ "//base:base" ]
>-  }
>-
>   if (is_ios && !build_with_chromium) {
>     deps += [ "../sdk:common_objc" ]
>   }
>-
>-  if (rtc_use_memcheck) {
>-    data = valgrind_webrtc_dependencies
>-  }
> }
> 
> if (rtc_include_tests) {
>   rtc_source_set("test_main") {
>+    visibility = [ "*" ]
>     testonly = true
>     sources = [
>       "test_main.cc",
>@@ -186,9 +206,12 @@ if (rtc_include_tests) {
>     ]
>     deps = [
>       ":field_trial",
>+      ":fileutils",
>+      ":perf_test",
>       "../rtc_base:rtc_base_approved",
>+      "../system_wrappers:field_trial_default",
>       "../system_wrappers:metrics_default",
>-      "//testing/gmock",
>+      "../system_wrappers:runtime_enabled_features_default",
>       "//testing/gtest",
>     ]
>   }
>@@ -199,8 +222,6 @@ if (rtc_include_tests) {
>     sources = [
>       "testsupport/frame_reader.h",
>       "testsupport/frame_writer.h",
>-      "testsupport/metrics/video_metrics.cc",
>-      "testsupport/metrics/video_metrics.h",
>       "testsupport/mock/mock_frame_reader.h",
>       "testsupport/y4m_frame_writer.cc",
>       "testsupport/yuv_frame_reader.cc",
>@@ -208,16 +229,18 @@ if (rtc_include_tests) {
>     ]
> 
>     deps = [
>+      ":fileutils",
>       ":test_support",
>       ":video_test_common",
>       "..:webrtc_common",
>-      "../api:video_frame_api",
>-      "../api:video_frame_api_i420",
>+      "../api/video:video_frame",
>+      "../api/video:video_frame_i420",
>       "../common_video",
>+      "../rtc_base:checks",
>       "../rtc_base:rtc_base_approved",
>       "../system_wrappers",
>-      "//testing/gmock",
>       "//testing/gtest",
>+      "//third_party/libyuv",
>     ]
> 
>     if (!is_ios) {
>@@ -227,10 +250,6 @@ if (rtc_include_tests) {
>       sources += [ "testsupport/jpeg_frame_writer_ios.cc" ]
>     }
> 
>-    public_deps = [
>-      ":fileutils",
>-    ]
>-
>     if (!build_with_chromium && is_clang) {
>       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>@@ -243,10 +262,6 @@ if (rtc_include_tests) {
>     if (is_ios) {
>       deps += [ "../sdk:common_objc" ]
>     }
>-
>-    if (rtc_use_memcheck) {
>-      data = valgrind_webrtc_dependencies
>-    }
>   }
> 
>   rtc_source_set("test_support_test_artifacts") {
>@@ -283,26 +298,29 @@ if (rtc_include_tests) {
> 
>   rtc_test("test_support_unittests") {
>     deps = [
>-      ":fake_audio_device",
>+      ":direct_transport",
>+      ":fileutils",
>+      ":perf_test",
>       ":rtp_test_utils",
>-      "../api:video_frame_api",
>-      "../api:video_frame_api_i420",
>-      "../call:call_interfaces",
>-      "../common_audio",
>-      "../modules/rtp_rtcp",
>+      ":test_main",
>+      ":test_support",
>+      ":test_support_test_artifacts",
>+      ":video_test_common",
>+      ":video_test_support",
>+      "../api/video:video_frame_i420",
>+      "../modules/rtp_rtcp:rtp_rtcp",
>       "../rtc_base:rtc_base_approved",
>-      "../system_wrappers",
>+      "../test:single_threaded_task_queue",
>+      "//testing/gtest",
>+      "//third_party/abseil-cpp/absl/memory",
>     ]
>     sources = [
>-      "fake_audio_device_unittest.cc",
>-      "fake_network_pipe_unittest.cc",
>+      "direct_transport_unittest.cc",
>       "frame_generator_unittest.cc",
>       "rtp_file_reader_unittest.cc",
>       "rtp_file_writer_unittest.cc",
>       "single_threaded_task_queue_unittest.cc",
>       "testsupport/always_passing_unittest.cc",
>-      "testsupport/metrics/video_metrics_unittest.cc",
>-      "testsupport/packet_reader_unittest.cc",
>       "testsupport/perf_test_unittest.cc",
>       "testsupport/test_artifacts_unittest.cc",
>       "testsupport/y4m_frame_writer_unittest.cc",
>@@ -310,14 +328,6 @@ if (rtc_include_tests) {
>       "testsupport/yuv_frame_writer_unittest.cc",
>     ]
> 
>-    # TODO(jschuh): Bug 1348: fix this warning.
>-    configs += [ "//build/config/compiler:no_size_t_to_int_warning" ]
>-
>-    if (is_win) {
>-      # virtual override w/different const/volatile signature.
>-      cflags = [ "/wd4373" ]
>-    }
>-
>     if (!build_with_chromium && is_clang) {
>       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>@@ -332,20 +342,11 @@ if (rtc_include_tests) {
>     if (is_ios) {
>       deps += [ ":test_support_unittests_bundle_data" ]
>     }
>-
>-    deps += [
>-      ":direct_transport",
>-      ":fileutils_unittests",
>-      ":test_common",
>-      ":test_main",
>-      ":test_support_test_artifacts",
>-      ":video_test_common",
>-      ":video_test_support",
>-      "../modules/video_capture",
>-      "../rtc_base:rtc_base_approved",
>-      "//testing/gmock",
>-      "//testing/gtest",
>-    ]
>+    if (!is_android && !build_with_chromium) {
>+      # This is needed in order to avoid:
>+      # undefined symbol: webrtc::videocapturemodule::VideoCaptureImpl::Create
>+      deps += [ "../modules/video_capture:video_capture_internal_impl" ]
>+    }
>   }
> }
> 
>@@ -357,6 +358,7 @@ if (is_ios) {
>     ]
>     deps = [
>       "..:webrtc_common",
>+      "../rtc_base:checks",
>       "../rtc_base:rtc_base_approved",
>       "../sdk:common_objc",
>     ]
>@@ -365,20 +367,17 @@ if (is_ios) {
> 
> rtc_source_set("fileutils") {
>   testonly = true
>-  visibility = [
>-    ":fileutils_unittests",
>-    ":test_support_test_artifacts",
>-    ":test_support",
>-    ":video_test_support",
>-  ]
>+  visibility = [ "*" ]
>   sources = [
>     "testsupport/fileutils.cc",
>     "testsupport/fileutils.h",
>   ]
>   deps = [
>     "..:webrtc_common",
>-    "../api:optional",
>+    "../rtc_base:checks",
>     "../rtc_base:rtc_base_approved",
>+    "../rtc_base/system:arch",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
>   if (is_ios) {
>     deps += [ ":fileutils_objc" ]
>@@ -440,20 +439,19 @@ rtc_source_set("fileutils_unittests") {
>   deps = [
>     ":fileutils",
>     ":test_support",
>-    "../api:optional",
>+    "../rtc_base:checks",
>     "../rtc_base:rtc_base_approved",
>-    "//testing/gmock",
>     "//testing/gtest",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
> 
> rtc_source_set("direct_transport") {
>+  visibility = [ "*" ]
>   testonly = true
>   sources = [
>     "direct_transport.cc",
>     "direct_transport.h",
>-    "fake_network_pipe.cc",
>-    "fake_network_pipe.h",
>   ]
>   if (!build_with_chromium && is_clang) {
>     # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>@@ -462,14 +460,16 @@ rtc_source_set("direct_transport") {
>   deps = [
>     "..:webrtc_common",
>     "../api:transport_api",
>-    "../call",
>+    "../call:call_interfaces",
>     "../modules/rtp_rtcp",
>     "../rtc_base:rtc_base_approved",
>     "../rtc_base:sequenced_task_checker",
>     "../system_wrappers",
>+    "//third_party/abseil-cpp/absl/memory",
>   ]
>   public_deps = [
>     ":single_threaded_task_queue",
>+    "../call:fake_network",
>   ]
> }
> 
>@@ -480,27 +480,9 @@ rtc_source_set("single_threaded_task_queue") {
>     "single_threaded_task_queue.h",
>   ]
>   deps = [
>+    "../rtc_base:checks",
>     "../rtc_base:rtc_base_approved",
>-  ]
>-}
>-
>-rtc_source_set("fake_audio_device") {
>-  testonly = true
>-  sources = [
>-    "fake_audio_device.cc",
>-    "fake_audio_device.h",
>-  ]
>-  if (!build_with_chromium && is_clang) {
>-    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>-    suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-  }
>-  deps = [
>-    "..:webrtc_common",
>-    "../api:array_view",
>-    "../common_audio:common_audio",
>-    "../modules/audio_device:audio_device",
>-    "../rtc_base:rtc_base_approved",
>-    "../system_wrappers:system_wrappers",
>+    "//third_party/abseil-cpp/absl/memory",
>   ]
> }
> 
>@@ -515,6 +497,7 @@ rtc_source_set("test_common") {
>     "constants.h",
>     "drifting_clock.cc",
>     "drifting_clock.h",
>+    "encoder_proxy_factory.h",
>     "encoder_settings.cc",
>     "encoder_settings.h",
>     "fake_decoder.cc",
>@@ -522,11 +505,12 @@ rtc_source_set("test_common") {
>     "fake_encoder.cc",
>     "fake_encoder.h",
>     "fake_videorenderer.h",
>+    "function_video_decoder_factory.h",
>+    "function_video_encoder_factory.h",
>     "layer_filtering_transport.cc",
>     "layer_filtering_transport.h",
>+    "mock_transport.cc",
>     "mock_transport.h",
>-    "mock_voe_channel_proxy.h",
>-    "mock_voice_engine.h",
>     "null_transport.cc",
>     "null_transport.h",
>     "rtp_rtcp_observer.h",
>@@ -548,59 +532,60 @@ rtc_source_set("test_common") {
> 
>   deps = [
>     ":direct_transport",
>-    ":fake_audio_device",
>+    ":fileutils",
>     ":rtp_test_utils",
>     ":test_support",
>     ":video_test_common",
>     "..:webrtc_common",
>+    "../api:libjingle_peerconnection_api",
>     "../api:transport_api",
>-    "../api:video_frame_api",
>-    "../api:video_frame_api_i420",
>     "../api/audio_codecs:builtin_audio_decoder_factory",
>     "../api/audio_codecs:builtin_audio_encoder_factory",
>+    "../api/video:video_frame",
>+    "../api/video:video_frame_i420",
>     "../api/video_codecs:video_codecs_api",
>     "../audio",
>     "../call",
>+    "../call:call_interfaces",
>     "../call:rtp_sender",
>     "../call:video_stream_api",
>     "../common_video",
>     "../logging:rtc_event_log_api",
>+    "../logging:rtc_event_log_impl_base",
>+    "../media:rtc_internal_video_codecs",
>+    "../media:rtc_media_base",
>+    "../modules/audio_device",
>+    "../modules/audio_device:audio_device_impl",
>     "../modules/audio_device:mock_audio_device",
>     "../modules/audio_mixer:audio_mixer_impl",
>     "../modules/audio_processing",
>+    "../modules/congestion_controller/bbr",
>     "../modules/rtp_rtcp",
>     "../modules/rtp_rtcp:mock_rtp_rtcp",
>+    "../modules/rtp_rtcp:rtp_rtcp_format",
>+    "../modules/video_coding:video_codec_interface",
>     "../modules/video_coding:video_coding_utility",
>     "../modules/video_coding:webrtc_h264",
>+    "../modules/video_coding:webrtc_multiplex",
>     "../modules/video_coding:webrtc_vp8",
>     "../modules/video_coding:webrtc_vp9",
>+    "../rtc_base:checks",
>     "../rtc_base:rtc_base_approved",
>     "../rtc_base:rtc_task_queue",
>     "../rtc_base:sequenced_task_checker",
>+    "../rtc_base/experiments:congestion_controller_experiment",
>     "../system_wrappers",
>+    "../system_wrappers:field_trial_api",
>+    "../system_wrappers:runtime_enabled_features_api",
>     "../video",
>-    "../voice_engine",
>-    "//testing/gmock",
>     "//testing/gtest",
>+    "//third_party/abseil-cpp/absl/memory",
>   ]
>   if (!is_android && !build_with_chromium) {
>     deps += [ "../modules/video_capture:video_capture_internal_impl" ]
>   }
> }
> 
>-config("test_renderer_exported_config") {
>-  if (is_win && is_clang) {
>-    # GN orders flags on a target before flags from configs. The default config
>-    # adds -Wall, and this flag have to be after -Wall -- so they need to
>-    # come from a config and cannot be on the target directly.
>-    cflags = [
>-      "-Wno-bool-conversion",
>-      "-Wno-comment",
>-      "-Wno-delete-non-virtual-dtor",
>-    ]
>-  }
>-}
>-
> rtc_source_set("test_renderer") {
>   public_deps = [
>     ":test_renderer_generic",
>@@ -651,22 +636,24 @@ rtc_source_set("test_renderer_generic") {
>   deps = [
>     ":test_support",
>     "..:webrtc_common",
>+    "../api/video:video_frame",
>     "../common_video",
>-    "../modules/media_file",
>+    "../media:rtc_media_base",
>+    "../rtc_base:checks",
>     "../rtc_base:rtc_base_approved",
>     "//testing/gtest",
>   ]
>-  if (!(is_linux && use_x11) && !is_mac && !is_win) {
>+  if (!(is_linux && rtc_use_x11) && !is_mac && !is_win) {
>     sources += [ "null_platform_renderer.cc" ]
>   }
>-  if ((is_linux && use_x11) || is_mac) {
>+  if ((is_linux && rtc_use_x11) || is_mac) {
>     sources += [
>       "gl/gl_renderer.cc",
>       "gl/gl_renderer.h",
>     ]
>   }
> 
>-  if (is_linux && use_x11) {
>+  if (is_linux && rtc_use_x11) {
>     sources += [
>       "linux/glx_renderer.cc",
>       "linux/glx_renderer.h",
>@@ -685,8 +672,6 @@ rtc_source_set("test_renderer_generic") {
>     ]
>   }
> 
>-  public_configs = [ ":test_renderer_exported_config" ]
>-
>   if (!build_with_chromium && is_clang) {
>     # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>     suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>@@ -710,7 +695,6 @@ rtc_source_set("audio_codec_mocks") {
>     "../api/audio_codecs:audio_codecs_api",
>     "../api/audio_codecs:builtin_audio_decoder_factory",
>     "../rtc_base:rtc_base_approved",
>-    "//testing/gmock",
>   ]
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/DEPS b/Source/ThirdParty/libwebrtc/Source/webrtc/test/DEPS
>index 39f107b9b2d..cd8b2e2bb5a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/DEPS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/DEPS
>@@ -6,17 +6,18 @@ include_rules = [
>   "+common_video",
>   "+logging/rtc_event_log",
>   "+media/base",
>+  "+media/engine",
>   "+modules/audio_coding",
>   "+modules/audio_device",
>   "+modules/audio_mixer",
>   "+modules/audio_processing",
>-  "+modules/media_file",
>+  "+modules/congestion_controller/bbr",
>   "+modules/rtp_rtcp",
>   "+modules/video_capture",
>   "+modules/video_coding",
>   "+sdk",
>   "+system_wrappers",
>-  "+voice_engine",
>+  "+third_party/libyuv",
> ]
> 
> specific_include_rules = {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/test/OWNERS
>index c58c863e161..e33219134c0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/OWNERS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/OWNERS
>@@ -1,2 +1,3 @@
> phoglund@webrtc.org
> stefan@webrtc.org
>+srte@webrtc.org
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/android/AndroidManifest.xml b/Source/ThirdParty/libwebrtc/Source/webrtc/test/android/AndroidManifest.xml
>index b5157a75dfc..f8eb2e500c7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/android/AndroidManifest.xml
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/android/AndroidManifest.xml
>@@ -26,7 +26,7 @@ be found in the AUTHORS file in the root of the source tree.
>     <uses-permission android:name="android.permission.WRITE_EXTERNAL_STORAGE"/>
> 
>     <application android:label="NativeTests"
>-            android:name="org.chromium.base.BaseChromiumApplication">
>+            android:name="org.chromium.native_test.NativeTestApplication">
>         <uses-library android:name="android.test.runner"/>
>         <activity android:name=".RTCNativeUnitTestActivity"
>                 android:label="NativeTest"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/call_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/call_test.cc
>index 3210e6460db..4df054b5cc6 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/call_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/call_test.cc
>@@ -12,16 +12,18 @@
> 
> #include <algorithm>
> 
>+#include "absl/memory/memory.h"
> #include "api/audio_codecs/builtin_audio_decoder_factory.h"
> #include "api/audio_codecs/builtin_audio_encoder_factory.h"
>+#include "api/video_codecs/video_encoder_config.h"
> #include "call/rtp_transport_controller_send.h"
>-#include "call/video_config.h"
> #include "modules/audio_mixer/audio_mixer_impl.h"
>+#include "modules/congestion_controller/bbr/bbr_factory.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/event.h"
>-#include "rtc_base/ptr_util.h"
>+#include "rtc_base/experiments/congestion_controller_experiment.h"
>+#include "test/fake_encoder.h"
> #include "test/testsupport/fileutils.h"
>-#include "voice_engine/include/voe_base.h"
> 
> namespace webrtc {
> namespace test {
>@@ -32,27 +34,29 @@ const int kVideoRotationRtpExtensionId = 4;
> 
> CallTest::CallTest()
>     : clock_(Clock::GetRealTimeClock()),
>-      event_log_(RtcEventLog::CreateNull()),
>+      send_event_log_(RtcEventLog::CreateNull()),
>+      recv_event_log_(RtcEventLog::CreateNull()),
>       sender_call_transport_controller_(nullptr),
>-      video_send_config_(nullptr),
>-      video_send_stream_(nullptr),
>       audio_send_config_(nullptr),
>       audio_send_stream_(nullptr),
>-      fake_encoder_(clock_),
>+      bbr_network_controller_factory_(new BbrNetworkControllerFactory()),
>+      fake_encoder_factory_([this]() {
>+        auto encoder = absl::make_unique<test::FakeEncoder>(clock_);
>+        encoder->SetMaxBitrate(fake_encoder_max_bitrate_);
>+        return encoder;
>+      }),
>       num_video_streams_(1),
>       num_audio_streams_(0),
>       num_flexfec_streams_(0),
>-      decoder_factory_(CreateBuiltinAudioDecoderFactory()),
>-      encoder_factory_(CreateBuiltinAudioEncoderFactory()),
>-      task_queue_("CallTestTaskQueue"),
>-      fake_send_audio_device_(nullptr),
>-      fake_recv_audio_device_(nullptr) {}
>+      audio_decoder_factory_(CreateBuiltinAudioDecoderFactory()),
>+      audio_encoder_factory_(CreateBuiltinAudioEncoderFactory()),
>+      task_queue_("CallTestTaskQueue") {}
> 
> CallTest::~CallTest() {
>   task_queue_.SendTask([this]() {
>-    fake_send_audio_device_.reset();
>-    fake_recv_audio_device_.reset();
>-    frame_generator_capturer_.reset();
>+    fake_send_audio_device_ = nullptr;
>+    fake_recv_audio_device_ = nullptr;
>+    video_capturers_.clear();
>   });
> }
> 
>@@ -62,18 +66,20 @@ void CallTest::RunBaseTest(BaseTest* test) {
>     num_audio_streams_ = test->GetNumAudioStreams();
>     num_flexfec_streams_ = test->GetNumFlexfecStreams();
>     RTC_DCHECK(num_video_streams_ > 0 || num_audio_streams_ > 0);
>-    Call::Config send_config(test->GetSenderCallConfig());
>+    Call::Config send_config(send_event_log_.get());
>+    test->ModifySenderCallConfig(&send_config);
>     if (num_audio_streams_ > 0) {
>       CreateFakeAudioDevices(test->CreateCapturer(), test->CreateRenderer());
>       test->OnFakeAudioDevicesCreated(fake_send_audio_device_.get(),
>                                       fake_recv_audio_device_.get());
>-      apm_send_ = AudioProcessing::Create();
>-      apm_recv_ = AudioProcessing::Create();
>-      CreateVoiceEngines();
>+      apm_send_ = AudioProcessingBuilder().Create();
>+      apm_recv_ = AudioProcessingBuilder().Create();
>+      EXPECT_EQ(0, fake_send_audio_device_->Init());
>+      EXPECT_EQ(0, fake_recv_audio_device_->Init());
>       AudioState::Config audio_state_config;
>-      audio_state_config.voice_engine = voe_send_.voice_engine;
>       audio_state_config.audio_mixer = AudioMixerImpl::Create();
>       audio_state_config.audio_processing = apm_send_;
>+      audio_state_config.audio_device_module = fake_send_audio_device_;
>       send_config.audio_state = AudioState::Create(audio_state_config);
>       fake_send_audio_device_->RegisterAudioCallback(
>           send_config.audio_state->audio_transport());
>@@ -84,15 +90,17 @@ void CallTest::RunBaseTest(BaseTest* test) {
>           sender_call_transport_controller_);
>     }
>     if (test->ShouldCreateReceivers()) {
>-      Call::Config recv_config(test->GetReceiverCallConfig());
>+      Call::Config recv_config(recv_event_log_.get());
>+      test->ModifyReceiverCallConfig(&recv_config);
>       if (num_audio_streams_ > 0) {
>         AudioState::Config audio_state_config;
>-        audio_state_config.voice_engine = voe_recv_.voice_engine;
>         audio_state_config.audio_mixer = AudioMixerImpl::Create();
>         audio_state_config.audio_processing = apm_recv_;
>+        audio_state_config.audio_device_module = fake_recv_audio_device_;
>         recv_config.audio_state = AudioState::Create(audio_state_config);
>         fake_recv_audio_device_->RegisterAudioCallback(
>-            recv_config.audio_state->audio_transport());      }
>+            recv_config.audio_state->audio_transport());
>+      }
>       CreateReceiverCall(recv_config);
>     }
>     test->OnCallsCreated(sender_call_.get(), receiver_call_.get());
>@@ -119,8 +127,8 @@ void CallTest::RunBaseTest(BaseTest* test) {
>       CreateMatchingReceiveConfigs(receive_transport_.get());
>     }
>     if (num_video_streams_ > 0) {
>-      test->ModifyVideoConfigs(&video_send_config_, &video_receive_configs_,
>-                               &video_encoder_config_);
>+      test->ModifyVideoConfigs(GetVideoSendConfig(), &video_receive_configs_,
>+                               GetVideoEncoderConfig());
>     }
>     if (num_audio_streams_ > 0) {
>       test->ModifyAudioConfigs(&audio_send_config_, &audio_receive_configs_);
>@@ -135,7 +143,7 @@ void CallTest::RunBaseTest(BaseTest* test) {
>     }
>     if (num_video_streams_ > 0) {
>       CreateVideoStreams();
>-      test->OnVideoStreamsCreated(video_send_stream_, video_receive_streams_);
>+      test->OnVideoStreamsCreated(GetVideoSendStream(), video_receive_streams_);
>     }
>     if (num_audio_streams_ > 0) {
>       CreateAudioStreams();
>@@ -148,7 +156,7 @@ void CallTest::RunBaseTest(BaseTest* test) {
>       int frame_rate = kDefaultFramerate;
>       test->ModifyVideoCaptureStartResolution(&width, &height, &frame_rate);
>       CreateFrameGeneratorCapturer(frame_rate, width, height);
>-      test->OnFrameGeneratorCapturerCreated(frame_generator_capturer_.get());
>+      test->OnFrameGeneratorCapturerCreated(frame_generator_capturer_);
>     }
> 
>     Start();
>@@ -163,24 +171,41 @@ void CallTest::RunBaseTest(BaseTest* test) {
>     send_transport_.reset();
>     receive_transport_.reset();
>     DestroyCalls();
>-    if (num_audio_streams_ > 0)
>-      DestroyVoiceEngines();
>   });
> }
> 
>+void CallTest::CreateCalls() {
>+  CreateCalls(Call::Config(send_event_log_.get()),
>+              Call::Config(recv_event_log_.get()));
>+}
>+
> void CallTest::CreateCalls(const Call::Config& sender_config,
>                            const Call::Config& receiver_config) {
>   CreateSenderCall(sender_config);
>   CreateReceiverCall(receiver_config);
> }
> 
>-void CallTest::CreateSenderCall(const Call::Config& config) {
>-  sender_call_transport_controller_ = new RtpTransportControllerSend(
>-      Clock::GetRealTimeClock(), config.event_log);
>+void CallTest::CreateSenderCall() {
>+  CreateSenderCall(Call::Config(send_event_log_.get()));
>+}
> 
>-  sender_call_.reset(
>-      Call::Create(config, std::unique_ptr<RtpTransportControllerSend>(
>-                               sender_call_transport_controller_)));
>+void CallTest::CreateSenderCall(const Call::Config& config) {
>+  NetworkControllerFactoryInterface* injected_factory =
>+      config.network_controller_factory;
>+  if (!injected_factory) {
>+    if (CongestionControllerExperiment::BbrControllerEnabled()) {
>+      RTC_LOG(LS_INFO) << "Using BBR network controller factory";
>+      injected_factory = bbr_network_controller_factory_.get();
>+    } else {
>+      RTC_LOG(LS_INFO) << "Using default network controller factory";
>+    }
>+  }
>+  std::unique_ptr<RtpTransportControllerSend> controller_send =
>+      absl::make_unique<RtpTransportControllerSend>(
>+          Clock::GetRealTimeClock(), config.event_log, injected_factory,
>+          config.bitrate_config);
>+  sender_call_transport_controller_ = controller_send.get();
>+  sender_call_.reset(Call::Create(config, std::move(controller_send)));
> }
> 
> void CallTest::CreateReceiverCall(const Call::Config& config) {
>@@ -192,102 +217,218 @@ void CallTest::DestroyCalls() {
>   receiver_call_.reset();
> }
> 
>-void CallTest::CreateSendConfig(size_t num_video_streams,
>-                                size_t num_audio_streams,
>-                                size_t num_flexfec_streams,
>-                                Transport* send_transport) {
>-  RTC_DCHECK(num_video_streams <= kNumSsrcs);
>-  RTC_DCHECK_LE(num_audio_streams, 1);
>-  RTC_DCHECK_LE(num_flexfec_streams, 1);
>-  RTC_DCHECK(num_audio_streams == 0 || voe_send_.channel_id >= 0);
>-  if (num_video_streams > 0) {
>-    video_send_config_ = VideoSendStream::Config(send_transport);
>-    video_send_config_.encoder_settings.encoder = &fake_encoder_;
>-    video_send_config_.encoder_settings.payload_name = "FAKE";
>-    video_send_config_.encoder_settings.payload_type =
>-        kFakeVideoSendPayloadType;
>-    video_send_config_.rtp.extensions.push_back(
>-        RtpExtension(RtpExtension::kTransportSequenceNumberUri,
>-                     kTransportSequenceNumberExtensionId));
>-    video_send_config_.rtp.extensions.push_back(RtpExtension(
>-        RtpExtension::kVideoContentTypeUri, kVideoContentTypeExtensionId));
>-    FillEncoderConfiguration(num_video_streams, &video_encoder_config_);
>-
>-    for (size_t i = 0; i < num_video_streams; ++i)
>-      video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[i]);
>-    video_send_config_.rtp.extensions.push_back(RtpExtension(
>-        RtpExtension::kVideoRotationUri, kVideoRotationRtpExtensionId));
>+void CallTest::CreateVideoSendConfig(VideoSendStream::Config* video_config,
>+                                     size_t num_video_streams,
>+                                     size_t num_used_ssrcs,
>+                                     Transport* send_transport) {
>+  RTC_DCHECK_LE(num_video_streams + num_used_ssrcs, kNumSsrcs);
>+  *video_config = VideoSendStream::Config(send_transport);
>+  video_config->encoder_settings.encoder_factory = &fake_encoder_factory_;
>+  video_config->rtp.payload_name = "FAKE";
>+  video_config->rtp.payload_type = kFakeVideoSendPayloadType;
>+  video_config->rtp.extensions.push_back(
>+      RtpExtension(RtpExtension::kTransportSequenceNumberUri,
>+                   kTransportSequenceNumberExtensionId));
>+  video_config->rtp.extensions.push_back(RtpExtension(
>+      RtpExtension::kVideoContentTypeUri, kVideoContentTypeExtensionId));
>+  if (video_encoder_configs_.empty()) {
>+    video_encoder_configs_.emplace_back();
>+    FillEncoderConfiguration(kVideoCodecGeneric, num_video_streams,
>+                             &video_encoder_configs_.back());
>   }
> 
>+  for (size_t i = 0; i < num_video_streams; ++i)
>+    video_config->rtp.ssrcs.push_back(kVideoSendSsrcs[num_used_ssrcs + i]);
>+  video_config->rtp.extensions.push_back(RtpExtension(
>+      RtpExtension::kVideoRotationUri, kVideoRotationRtpExtensionId));
>+}
>+
>+void CallTest::CreateAudioAndFecSendConfigs(size_t num_audio_streams,
>+                                            size_t num_flexfec_streams,
>+                                            Transport* send_transport) {
>+  RTC_DCHECK_LE(num_audio_streams, 1);
>+  RTC_DCHECK_LE(num_flexfec_streams, 1);
>   if (num_audio_streams > 0) {
>-    audio_send_config_ = AudioSendStream::Config(send_transport);
>-    audio_send_config_.voe_channel_id = voe_send_.channel_id;
>-    audio_send_config_.rtp.ssrc = kAudioSendSsrc;
>-    audio_send_config_.send_codec_spec =
>-        rtc::Optional<AudioSendStream::Config::SendCodecSpec>(
>-            {kAudioSendPayloadType, {"opus", 48000, 2, {{"stereo", "1"}}}});
>-    audio_send_config_.encoder_factory = encoder_factory_;
>+    AudioSendStream::Config audio_send_config(send_transport);
>+    audio_send_config.rtp.ssrc = kAudioSendSsrc;
>+    audio_send_config.send_codec_spec = AudioSendStream::Config::SendCodecSpec(
>+        kAudioSendPayloadType, {"opus", 48000, 2, {{"stereo", "1"}}});
>+    audio_send_config.encoder_factory = audio_encoder_factory_;
>+    SetAudioConfig(audio_send_config);
>   }
> 
>   // TODO(brandtr): Update this when we support multistream protection.
>   if (num_flexfec_streams > 0) {
>-    video_send_config_.rtp.flexfec.payload_type = kFlexfecPayloadType;
>-    video_send_config_.rtp.flexfec.ssrc = kFlexfecSendSsrc;
>-    video_send_config_.rtp.flexfec.protected_media_ssrcs = {kVideoSendSsrcs[0]};
>+    SetSendFecConfig({kVideoSendSsrcs[0]});
>   }
> }
> 
>-void CallTest::CreateMatchingReceiveConfigs(Transport* rtcp_send_transport) {
>-  video_receive_configs_.clear();
>-  allocated_decoders_.clear();
>-  if (num_video_streams_ > 0) {
>-    RTC_DCHECK(!video_send_config_.rtp.ssrcs.empty());
>-    VideoReceiveStream::Config video_config(rtcp_send_transport);
>-    video_config.rtp.remb = false;
>-    video_config.rtp.transport_cc = true;
>-    video_config.rtp.local_ssrc = kReceiverLocalVideoSsrc;
>-    for (const RtpExtension& extension : video_send_config_.rtp.extensions)
>-      video_config.rtp.extensions.push_back(extension);
>-    video_config.renderer = &fake_renderer_;
>-    for (size_t i = 0; i < video_send_config_.rtp.ssrcs.size(); ++i) {
>-      VideoReceiveStream::Decoder decoder =
>-          test::CreateMatchingDecoder(video_send_config_.encoder_settings);
>-      allocated_decoders_.push_back(
>-          std::unique_ptr<VideoDecoder>(decoder.decoder));
>-      video_config.decoders.clear();
>-      video_config.decoders.push_back(decoder);
>-      video_config.rtp.remote_ssrc = video_send_config_.rtp.ssrcs[i];
>-      video_receive_configs_.push_back(video_config.Copy());
>+void CallTest::SetAudioConfig(const AudioSendStream::Config& config) {
>+  audio_send_config_ = config;
>+}
>+
>+void CallTest::SetSendFecConfig(std::vector<uint32_t> video_send_ssrcs) {
>+  GetVideoSendConfig()->rtp.flexfec.payload_type = kFlexfecPayloadType;
>+  GetVideoSendConfig()->rtp.flexfec.ssrc = kFlexfecSendSsrc;
>+  GetVideoSendConfig()->rtp.flexfec.protected_media_ssrcs = video_send_ssrcs;
>+}
>+
>+void CallTest::SetSendUlpFecConfig(VideoSendStream::Config* send_config) {
>+  send_config->rtp.ulpfec.red_payload_type = kRedPayloadType;
>+  send_config->rtp.ulpfec.ulpfec_payload_type = kUlpfecPayloadType;
>+  send_config->rtp.ulpfec.red_rtx_payload_type = kRtxRedPayloadType;
>+}
>+
>+void CallTest::SetReceiveUlpFecConfig(
>+    VideoReceiveStream::Config* receive_config) {
>+  receive_config->rtp.red_payload_type = kRedPayloadType;
>+  receive_config->rtp.ulpfec_payload_type = kUlpfecPayloadType;
>+  receive_config->rtp.rtx_associated_payload_types[kRtxRedPayloadType] =
>+      kRedPayloadType;
>+}
>+
>+void CallTest::CreateSendConfig(size_t num_video_streams,
>+                                size_t num_audio_streams,
>+                                size_t num_flexfec_streams,
>+                                Transport* send_transport) {
>+  if (num_video_streams > 0) {
>+    video_send_configs_.clear();
>+    video_send_configs_.emplace_back(nullptr);
>+    CreateVideoSendConfig(&video_send_configs_.back(), num_video_streams, 0,
>+                          send_transport);
>+  }
>+  CreateAudioAndFecSendConfigs(num_audio_streams, num_flexfec_streams,
>+                               send_transport);
>+}
>+
>+void CallTest::CreateMatchingVideoReceiveConfigs(
>+    const VideoSendStream::Config& video_send_config,
>+    Transport* rtcp_send_transport) {
>+  CreateMatchingVideoReceiveConfigs(video_send_config, rtcp_send_transport,
>+                                    true, absl::nullopt, false, 0);
>+}
>+
>+void CallTest::CreateMatchingVideoReceiveConfigs(
>+    const VideoSendStream::Config& video_send_config,
>+    Transport* rtcp_send_transport,
>+    bool send_side_bwe,
>+    absl::optional<size_t> decode_sub_stream,
>+    bool receiver_reference_time_report,
>+    int rtp_history_ms) {
>+  AddMatchingVideoReceiveConfigs(
>+      &video_receive_configs_, video_send_config, rtcp_send_transport,
>+      send_side_bwe, decode_sub_stream, receiver_reference_time_report,
>+      rtp_history_ms);
>+}
>+
>+void CallTest::AddMatchingVideoReceiveConfigs(
>+    std::vector<VideoReceiveStream::Config>* receive_configs,
>+    const VideoSendStream::Config& video_send_config,
>+    Transport* rtcp_send_transport,
>+    bool send_side_bwe,
>+    absl::optional<size_t> decode_sub_stream,
>+    bool receiver_reference_time_report,
>+    int rtp_history_ms) {
>+  RTC_DCHECK(!video_send_config.rtp.ssrcs.empty());
>+  VideoReceiveStream::Config default_config(rtcp_send_transport);
>+  default_config.rtp.remb = !send_side_bwe;
>+  default_config.rtp.transport_cc = send_side_bwe;
>+  default_config.rtp.local_ssrc = kReceiverLocalVideoSsrc;
>+  for (const RtpExtension& extension : video_send_config.rtp.extensions)
>+    default_config.rtp.extensions.push_back(extension);
>+  default_config.rtp.nack.rtp_history_ms = rtp_history_ms;
>+  // Enable RTT calculation so NTP time estimator will work.
>+  default_config.rtp.rtcp_xr.receiver_reference_time_report =
>+      receiver_reference_time_report;
>+  default_config.renderer = &fake_renderer_;
>+
>+  for (size_t i = 0; i < video_send_config.rtp.ssrcs.size(); ++i) {
>+    VideoReceiveStream::Config video_recv_config(default_config.Copy());
>+    video_recv_config.decoders.clear();
>+    if (!video_send_config.rtp.rtx.ssrcs.empty()) {
>+      video_recv_config.rtp.rtx_ssrc = video_send_config.rtp.rtx.ssrcs[i];
>+      video_recv_config.rtp.rtx_associated_payload_types[kSendRtxPayloadType] =
>+          video_send_config.rtp.payload_type;
>     }
>-    video_receive_configs_[0].rtp.protected_by_flexfec =
>-        (num_flexfec_streams_ == 1);
>+    video_recv_config.rtp.remote_ssrc = video_send_config.rtp.ssrcs[i];
>+    VideoReceiveStream::Decoder decoder;
>+
>+    // Force fake decoders on non-selected simulcast streams.
>+    if (!decode_sub_stream || i == *decode_sub_stream) {
>+      decoder = test::CreateMatchingDecoder(video_send_config);
>+    } else {
>+      decoder.decoder = new test::FakeDecoder();
>+      decoder.payload_type = video_send_config.rtp.payload_type;
>+      decoder.payload_name = video_send_config.rtp.payload_name;
>+    }
>+    allocated_decoders_.emplace_back(decoder.decoder);
>+    video_recv_config.decoders.push_back(decoder);
>+    receive_configs->emplace_back(std::move(video_recv_config));
>   }
>+}
> 
>+void CallTest::CreateMatchingAudioAndFecConfigs(
>+    Transport* rtcp_send_transport) {
>   RTC_DCHECK_GE(1, num_audio_streams_);
>   if (num_audio_streams_ == 1) {
>-    RTC_DCHECK_LE(0, voe_send_.channel_id);
>-    AudioReceiveStream::Config audio_config;
>-    audio_config.rtp.local_ssrc = kReceiverLocalAudioSsrc;
>-    audio_config.rtcp_send_transport = rtcp_send_transport;
>-    audio_config.voe_channel_id = voe_recv_.channel_id;
>-    audio_config.rtp.remote_ssrc = audio_send_config_.rtp.ssrc;
>-    audio_config.decoder_factory = decoder_factory_;
>-    audio_config.decoder_map = {{kAudioSendPayloadType, {"opus", 48000, 2}}};
>-    audio_receive_configs_.push_back(audio_config);
>+    CreateMatchingAudioConfigs(rtcp_send_transport, "");
>   }
> 
>   // TODO(brandtr): Update this when we support multistream protection.
>   RTC_DCHECK(num_flexfec_streams_ <= 1);
>   if (num_flexfec_streams_ == 1) {
>-    FlexfecReceiveStream::Config config(rtcp_send_transport);
>-    config.payload_type = kFlexfecPayloadType;
>-    config.remote_ssrc = kFlexfecSendSsrc;
>-    config.protected_media_ssrcs = {kVideoSendSsrcs[0]};
>-    config.local_ssrc = kReceiverLocalVideoSsrc;
>-    for (const RtpExtension& extension : video_send_config_.rtp.extensions)
>-      config.rtp_header_extensions.push_back(extension);
>-    flexfec_receive_configs_.push_back(config);
>+    CreateMatchingFecConfig(rtcp_send_transport, *GetVideoSendConfig());
>+    for (const RtpExtension& extension : GetVideoSendConfig()->rtp.extensions)
>+      GetFlexFecConfig()->rtp_header_extensions.push_back(extension);
>+  }
>+}
>+
>+void CallTest::CreateMatchingAudioConfigs(Transport* transport,
>+                                          std::string sync_group) {
>+  audio_receive_configs_.push_back(CreateMatchingAudioConfig(
>+      audio_send_config_, audio_decoder_factory_, transport, sync_group));
>+}
>+
>+AudioReceiveStream::Config CallTest::CreateMatchingAudioConfig(
>+    const AudioSendStream::Config& send_config,
>+    rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory,
>+    Transport* transport,
>+    std::string sync_group) {
>+  AudioReceiveStream::Config audio_config;
>+  audio_config.rtp.local_ssrc = kReceiverLocalAudioSsrc;
>+  audio_config.rtcp_send_transport = transport;
>+  audio_config.rtp.remote_ssrc = send_config.rtp.ssrc;
>+  audio_config.rtp.transport_cc =
>+      send_config.send_codec_spec
>+          ? send_config.send_codec_spec->transport_cc_enabled
>+          : false;
>+  audio_config.rtp.extensions = send_config.rtp.extensions;
>+  audio_config.decoder_factory = audio_decoder_factory;
>+  audio_config.decoder_map = {{kAudioSendPayloadType, {"opus", 48000, 2}}};
>+  audio_config.sync_group = sync_group;
>+  return audio_config;
>+}
>+
>+void CallTest::CreateMatchingFecConfig(
>+    Transport* transport,
>+    const VideoSendStream::Config& send_config) {
>+  FlexfecReceiveStream::Config config(transport);
>+  config.payload_type = send_config.rtp.flexfec.payload_type;
>+  config.remote_ssrc = send_config.rtp.flexfec.ssrc;
>+  config.protected_media_ssrcs = send_config.rtp.flexfec.protected_media_ssrcs;
>+  config.local_ssrc = kReceiverLocalVideoSsrc;
>+  if (!video_receive_configs_.empty())
>+    video_receive_configs_[0].rtp.protected_by_flexfec = true;
>+  flexfec_receive_configs_.push_back(config);
>+}
>+
>+void CallTest::CreateMatchingReceiveConfigs(Transport* rtcp_send_transport) {
>+  video_receive_configs_.clear();
>+  allocated_decoders_.clear();
>+  for (VideoSendStream::Config& video_send_config : video_send_configs_) {
>+    CreateMatchingVideoReceiveConfigs(video_send_config, rtcp_send_transport);
>   }
>+  CreateMatchingAudioAndFecConfigs(rtcp_send_transport);
> }
> 
> void CallTest::CreateFrameGeneratorCapturerWithDrift(Clock* clock,
>@@ -295,40 +436,41 @@ void CallTest::CreateFrameGeneratorCapturerWithDrift(Clock* clock,
>                                                      int framerate,
>                                                      int width,
>                                                      int height) {
>-  frame_generator_capturer_.reset(test::FrameGeneratorCapturer::Create(
>-      width, height, framerate * speed, clock));
>-  video_send_stream_->SetSource(
>-      frame_generator_capturer_.get(),
>-      VideoSendStream::DegradationPreference::kMaintainFramerate);
>+  video_sources_.clear();
>+  video_capturers_.clear();
>+  frame_generator_capturer_ = test::FrameGeneratorCapturer::Create(
>+      width, height, absl::nullopt, absl::nullopt, framerate * speed, clock);
>+  video_capturers_.emplace_back(
>+      std::unique_ptr<FrameGeneratorCapturer>(frame_generator_capturer_));
>+  video_sources_.push_back(video_capturers_.back().get());
>+  ConnectVideoSourcesToStreams();
> }
> 
> void CallTest::CreateFrameGeneratorCapturer(int framerate,
>                                             int width,
>                                             int height) {
>-  frame_generator_capturer_.reset(
>-      test::FrameGeneratorCapturer::Create(width, height, framerate, clock_));
>-  video_send_stream_->SetSource(
>-      frame_generator_capturer_.get(),
>-      VideoSendStream::DegradationPreference::kMaintainFramerate);
>+  video_sources_.clear();
>+  video_capturers_.clear();
>+  frame_generator_capturer_ = test::FrameGeneratorCapturer::Create(
>+      width, height, absl::nullopt, absl::nullopt, framerate, clock_);
>+  video_capturers_.emplace_back(
>+      std::unique_ptr<FrameGeneratorCapturer>(frame_generator_capturer_));
>+  video_sources_.push_back(video_capturers_.back().get());
>+  ConnectVideoSourcesToStreams();
> }
> 
> void CallTest::CreateFakeAudioDevices(
>-    std::unique_ptr<FakeAudioDevice::Capturer> capturer,
>-    std::unique_ptr<FakeAudioDevice::Renderer> renderer) {
>-  fake_send_audio_device_.reset(new FakeAudioDevice(
>-      std::move(capturer), nullptr, 1.f));
>-  fake_recv_audio_device_.reset(new FakeAudioDevice(
>-      nullptr, std::move(renderer), 1.f));
>+    std::unique_ptr<TestAudioDeviceModule::Capturer> capturer,
>+    std::unique_ptr<TestAudioDeviceModule::Renderer> renderer) {
>+  fake_send_audio_device_ = TestAudioDeviceModule::CreateTestAudioDeviceModule(
>+      std::move(capturer), nullptr, 1.f);
>+  fake_recv_audio_device_ = TestAudioDeviceModule::CreateTestAudioDeviceModule(
>+      nullptr, std::move(renderer), 1.f);
> }
> 
> void CallTest::CreateVideoStreams() {
>-  RTC_DCHECK(video_send_stream_ == nullptr);
>   RTC_DCHECK(video_receive_streams_.empty());
>-  RTC_DCHECK(audio_send_stream_ == nullptr);
>-  RTC_DCHECK(audio_receive_streams_.empty());
>-
>-  video_send_stream_ = sender_call_->CreateVideoSendStream(
>-      video_send_config_.Copy(), video_encoder_config_.Copy());
>+  CreateVideoSendStreams();
>   for (size_t i = 0; i < video_receive_configs_.size(); ++i) {
>     video_receive_streams_.push_back(receiver_call_->CreateVideoReceiveStream(
>         video_receive_configs_[i].Copy()));
>@@ -337,7 +479,51 @@ void CallTest::CreateVideoStreams() {
>   AssociateFlexfecStreamsWithVideoStreams();
> }
> 
>+void CallTest::CreateVideoSendStreams() {
>+  RTC_DCHECK(video_send_streams_.empty());
>+
>+  // We currently only support testing external fec controllers with a single
>+  // VideoSendStream.
>+  if (fec_controller_factory_.get()) {
>+    RTC_DCHECK_LE(video_send_configs_.size(), 1);
>+  }
>+
>+  // TODO(http://crbug/818127):
>+  // Remove this workaround when ALR is not screenshare-specific.
>+  std::list<size_t> streams_creation_order;
>+  for (size_t i = 0; i < video_send_configs_.size(); ++i) {
>+    // If dual streams are created, add the screenshare stream last.
>+    if (video_encoder_configs_[i].content_type ==
>+        VideoEncoderConfig::ContentType::kScreen) {
>+      streams_creation_order.push_back(i);
>+    } else {
>+      streams_creation_order.push_front(i);
>+    }
>+  }
>+
>+  video_send_streams_.resize(video_send_configs_.size(), nullptr);
>+
>+  for (size_t i : streams_creation_order) {
>+    if (fec_controller_factory_.get()) {
>+      video_send_streams_[i] = sender_call_->CreateVideoSendStream(
>+          video_send_configs_[i].Copy(), video_encoder_configs_[i].Copy(),
>+          fec_controller_factory_->CreateFecController());
>+    } else {
>+      video_send_streams_[i] = sender_call_->CreateVideoSendStream(
>+          video_send_configs_[i].Copy(), video_encoder_configs_[i].Copy());
>+    }
>+  }
>+}
>+
>+void CallTest::CreateVideoSendStream(const VideoEncoderConfig& encoder_config) {
>+  RTC_DCHECK(video_send_streams_.empty());
>+  video_send_streams_.push_back(sender_call_->CreateVideoSendStream(
>+      GetVideoSendConfig()->Copy(), encoder_config.Copy()));
>+}
>+
> void CallTest::CreateAudioStreams() {
>+  RTC_DCHECK(audio_send_stream_ == nullptr);
>+  RTC_DCHECK(audio_receive_streams_.empty());
>   audio_send_stream_ = sender_call_->CreateAudioSendStream(audio_send_config_);
>   for (size_t i = 0; i < audio_receive_configs_.size(); ++i) {
>     audio_receive_streams_.push_back(
>@@ -355,6 +541,12 @@ void CallTest::CreateFlexfecStreams() {
>   AssociateFlexfecStreamsWithVideoStreams();
> }
> 
>+void CallTest::ConnectVideoSourcesToStreams() {
>+  for (size_t i = 0; i < video_sources_.size(); ++i)
>+    video_send_streams_[i]->SetSource(video_sources_[i],
>+                                      degradation_preference_);
>+}
>+
> void CallTest::AssociateFlexfecStreamsWithVideoStreams() {
>   // All FlexFEC streams protect all of the video streams.
>   for (FlexfecReceiveStream* flexfec_recv_stream : flexfec_receive_streams_) {
>@@ -373,31 +565,47 @@ void CallTest::DissociateFlexfecStreamsFromVideoStreams() {
> }
> 
> void CallTest::Start() {
>-  if (video_send_stream_)
>-    video_send_stream_->Start();
>-  for (VideoReceiveStream* video_recv_stream : video_receive_streams_)
>-    video_recv_stream->Start();
>+  StartVideoStreams();
>   if (audio_send_stream_) {
>     audio_send_stream_->Start();
>   }
>   for (AudioReceiveStream* audio_recv_stream : audio_receive_streams_)
>     audio_recv_stream->Start();
>-  if (frame_generator_capturer_.get() != NULL)
>-    frame_generator_capturer_->Start();
>+  StartVideoCapture();
>+}
>+
>+void CallTest::StartVideoStreams() {
>+  for (VideoSendStream* video_send_stream : video_send_streams_)
>+    video_send_stream->Start();
>+  for (VideoReceiveStream* video_recv_stream : video_receive_streams_)
>+    video_recv_stream->Start();
>+}
>+
>+void CallTest::StartVideoCapture() {
>+  for (auto& capturer : video_capturers_)
>+    capturer->Start();
> }
> 
> void CallTest::Stop() {
>-  if (frame_generator_capturer_.get() != NULL)
>-    frame_generator_capturer_->Stop();
>+  StopVideoCapture();
>   for (AudioReceiveStream* audio_recv_stream : audio_receive_streams_)
>     audio_recv_stream->Stop();
>   if (audio_send_stream_) {
>     audio_send_stream_->Stop();
>   }
>+  StopVideoStreams();
>+}
>+
>+void CallTest::StopVideoCapture() {
>+  for (auto& capturer : video_capturers_)
>+    capturer->Stop();
>+}
>+
>+void CallTest::StopVideoStreams() {
>+  for (VideoSendStream* video_send_stream : video_send_streams_)
>+    video_send_stream->Stop();
>   for (VideoReceiveStream* video_recv_stream : video_receive_streams_)
>     video_recv_stream->Stop();
>-  if (video_send_stream_)
>-    video_send_stream_->Stop();
> }
> 
> void CallTest::DestroyStreams() {
>@@ -409,9 +617,7 @@ void CallTest::DestroyStreams() {
>   for (AudioReceiveStream* audio_recv_stream : audio_receive_streams_)
>     receiver_call_->DestroyAudioReceiveStream(audio_recv_stream);
> 
>-  if (video_send_stream_)
>-    sender_call_->DestroyVideoSendStream(video_send_stream_);
>-  video_send_stream_ = nullptr;
>+  DestroyVideoSendStreams();
> 
>   for (VideoReceiveStream* video_recv_stream : video_receive_streams_)
>     receiver_call_->DestroyVideoReceiveStream(video_recv_stream);
>@@ -423,67 +629,55 @@ void CallTest::DestroyStreams() {
>   allocated_decoders_.clear();
> }
> 
>+void CallTest::DestroyVideoSendStreams() {
>+  for (VideoSendStream* video_send_stream : video_send_streams_)
>+    sender_call_->DestroyVideoSendStream(video_send_stream);
>+  video_send_streams_.clear();
>+}
>+
> void CallTest::SetFakeVideoCaptureRotation(VideoRotation rotation) {
>   frame_generator_capturer_->SetFakeRotation(rotation);
> }
> 
>-void CallTest::CreateVoiceEngines() {
>-  voe_send_.voice_engine = VoiceEngine::Create();
>-  voe_send_.base = VoEBase::GetInterface(voe_send_.voice_engine);
>-  EXPECT_EQ(0, fake_send_audio_device_->Init());
>-  EXPECT_EQ(0, voe_send_.base->Init(fake_send_audio_device_.get(),
>-                                    apm_send_.get(), decoder_factory_));
>-  VoEBase::ChannelConfig config;
>-  config.enable_voice_pacing = true;
>-  voe_send_.channel_id = voe_send_.base->CreateChannel(config);
>-  EXPECT_GE(voe_send_.channel_id, 0);
>+void CallTest::SetVideoDegradation(DegradationPreference preference) {
>+  GetVideoSendStream()->SetSource(frame_generator_capturer_, preference);
>+}
>+
>+VideoSendStream::Config* CallTest::GetVideoSendConfig() {
>+  return &video_send_configs_[0];
>+}
> 
>-  voe_recv_.voice_engine = VoiceEngine::Create();
>-  voe_recv_.base = VoEBase::GetInterface(voe_recv_.voice_engine);
>-  EXPECT_EQ(0, fake_recv_audio_device_->Init());
>-  EXPECT_EQ(0, voe_recv_.base->Init(fake_recv_audio_device_.get(),
>-                                    apm_recv_.get(), decoder_factory_));
>-  voe_recv_.channel_id = voe_recv_.base->CreateChannel();
>-  EXPECT_GE(voe_recv_.channel_id, 0);
>+void CallTest::SetVideoSendConfig(const VideoSendStream::Config& config) {
>+  video_send_configs_.clear();
>+  video_send_configs_.push_back(config.Copy());
> }
> 
>-void CallTest::DestroyVoiceEngines() {
>-  voe_recv_.base->DeleteChannel(voe_recv_.channel_id);
>-  voe_recv_.channel_id = -1;
>-  voe_recv_.base->Release();
>-  voe_recv_.base = nullptr;
>+VideoEncoderConfig* CallTest::GetVideoEncoderConfig() {
>+  return &video_encoder_configs_[0];
>+}
> 
>-  voe_send_.base->DeleteChannel(voe_send_.channel_id);
>-  voe_send_.channel_id = -1;
>-  voe_send_.base->Release();
>-  voe_send_.base = nullptr;
>+void CallTest::SetVideoEncoderConfig(const VideoEncoderConfig& config) {
>+  video_encoder_configs_.clear();
>+  video_encoder_configs_.push_back(config.Copy());
>+}
> 
>-  VoiceEngine::Delete(voe_send_.voice_engine);
>-  voe_send_.voice_engine = nullptr;
>-  VoiceEngine::Delete(voe_recv_.voice_engine);
>-  voe_recv_.voice_engine = nullptr;
>+VideoSendStream* CallTest::GetVideoSendStream() {
>+  return video_send_streams_[0];
>+}
>+FlexfecReceiveStream::Config* CallTest::GetFlexFecConfig() {
>+  return &flexfec_receive_configs_[0];
> }
> 
>+constexpr size_t CallTest::kNumSsrcs;
> const int CallTest::kDefaultWidth;
> const int CallTest::kDefaultHeight;
> const int CallTest::kDefaultFramerate;
> const int CallTest::kDefaultTimeoutMs = 30 * 1000;
> const int CallTest::kLongTimeoutMs = 120 * 1000;
>-const uint8_t CallTest::kVideoSendPayloadType = 100;
>-const uint8_t CallTest::kFakeVideoSendPayloadType = 125;
>-const uint8_t CallTest::kSendRtxPayloadType = 98;
>-const uint8_t CallTest::kRedPayloadType = 118;
>-const uint8_t CallTest::kRtxRedPayloadType = 99;
>-const uint8_t CallTest::kUlpfecPayloadType = 119;
>-const uint8_t CallTest::kFlexfecPayloadType = 120;
>-const uint8_t CallTest::kAudioSendPayloadType = 103;
>-const uint8_t CallTest::kPayloadTypeH264 = 122;
>-const uint8_t CallTest::kPayloadTypeVP8 = 123;
>-const uint8_t CallTest::kPayloadTypeVP9 = 124;
>-const uint32_t CallTest::kSendRtxSsrcs[kNumSsrcs] = {0xBADCAFD, 0xBADCAFE,
>-                                                     0xBADCAFF};
>-const uint32_t CallTest::kVideoSendSsrcs[kNumSsrcs] = {0xC0FFED, 0xC0FFEE,
>-                                                       0xC0FFEF};
>+const uint32_t CallTest::kSendRtxSsrcs[kNumSsrcs] = {
>+    0xBADCAFD, 0xBADCAFE, 0xBADCAFF, 0xBADCB00, 0xBADCB01, 0xBADCB02};
>+const uint32_t CallTest::kVideoSendSsrcs[kNumSsrcs] = {
>+    0xC0FFED, 0xC0FFEE, 0xC0FFEF, 0xC0FFF0, 0xC0FFF1, 0xC0FFF2};
> const uint32_t CallTest::kAudioSendSsrc = 0xDEADBEEF;
> const uint32_t CallTest::kFlexfecSendSsrc = 0xBADBEEF;
> const uint32_t CallTest::kReceiverLocalVideoSsrc = 0x123456;
>@@ -504,39 +698,32 @@ const std::map<uint8_t, MediaType> CallTest::payload_type_map_ = {
>     {CallTest::kAudioSendPayloadType, MediaType::AUDIO},
>     {CallTest::kDefaultKeepalivePayloadType, MediaType::ANY}};
> 
>-BaseTest::BaseTest() : event_log_(RtcEventLog::CreateNull()) {}
>+BaseTest::BaseTest() {}
> 
>-BaseTest::BaseTest(unsigned int timeout_ms)
>-    : RtpRtcpObserver(timeout_ms), event_log_(RtcEventLog::CreateNull()) {}
>+BaseTest::BaseTest(int timeout_ms) : RtpRtcpObserver(timeout_ms) {}
> 
>-BaseTest::~BaseTest() {
>-}
>+BaseTest::~BaseTest() {}
> 
>-std::unique_ptr<FakeAudioDevice::Capturer> BaseTest::CreateCapturer() {
>-  return FakeAudioDevice::CreatePulsedNoiseCapturer(256, 48000);
>+std::unique_ptr<TestAudioDeviceModule::Capturer> BaseTest::CreateCapturer() {
>+  return TestAudioDeviceModule::CreatePulsedNoiseCapturer(256, 48000);
> }
> 
>-std::unique_ptr<FakeAudioDevice::Renderer> BaseTest::CreateRenderer() {
>-  return FakeAudioDevice::CreateDiscardRenderer(48000);
>+std::unique_ptr<TestAudioDeviceModule::Renderer> BaseTest::CreateRenderer() {
>+  return TestAudioDeviceModule::CreateDiscardRenderer(48000);
> }
> 
>-void BaseTest::OnFakeAudioDevicesCreated(FakeAudioDevice* send_audio_device,
>-                                         FakeAudioDevice* recv_audio_device) {
>-}
>+void BaseTest::OnFakeAudioDevicesCreated(
>+    TestAudioDeviceModule* send_audio_device,
>+    TestAudioDeviceModule* recv_audio_device) {}
> 
>-Call::Config BaseTest::GetSenderCallConfig() {
>-  return Call::Config(event_log_.get());
>-}
>+void BaseTest::ModifySenderCallConfig(Call::Config* config) {}
> 
>-Call::Config BaseTest::GetReceiverCallConfig() {
>-  return Call::Config(event_log_.get());
>-}
>+void BaseTest::ModifyReceiverCallConfig(Call::Config* config) {}
> 
> void BaseTest::OnRtpTransportControllerSendCreated(
>     RtpTransportControllerSend* controller) {}
> 
>-void BaseTest::OnCallsCreated(Call* sender_call, Call* receiver_call) {
>-}
>+void BaseTest::OnCallsCreated(Call* sender_call, Call* receiver_call) {}
> 
> test::PacketTransport* BaseTest::CreateSendTransport(
>     SingleThreadedTaskQueueForTesting* task_queue,
>@@ -593,14 +780,11 @@ void BaseTest::OnFlexfecStreamsCreated(
>     const std::vector<FlexfecReceiveStream*>& receive_streams) {}
> 
> void BaseTest::OnFrameGeneratorCapturerCreated(
>-    FrameGeneratorCapturer* frame_generator_capturer) {
>-}
>+    FrameGeneratorCapturer* frame_generator_capturer) {}
> 
>-void BaseTest::OnStreamsStopped() {
>-}
>+void BaseTest::OnStreamsStopped() {}
> 
>-SendTest::SendTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
>-}
>+SendTest::SendTest(int timeout_ms) : BaseTest(timeout_ms) {}
> 
> bool SendTest::ShouldCreateReceivers() const {
>   return false;
>@@ -608,8 +792,7 @@ bool SendTest::ShouldCreateReceivers() const {
> 
> EndToEndTest::EndToEndTest() {}
> 
>-EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
>-}
>+EndToEndTest::EndToEndTest(int timeout_ms) : BaseTest(timeout_ms) {}
> 
> bool EndToEndTest::ShouldCreateReceivers() const {
>   return true;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/call_test.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/call_test.h
>index cf15aacf114..54be39ac988 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/call_test.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/call_test.h
>@@ -16,19 +16,16 @@
> #include "call/call.h"
> #include "call/rtp_transport_controller_send.h"
> #include "logging/rtc_event_log/rtc_event_log.h"
>+#include "modules/audio_device/include/test_audio_device.h"
> #include "test/encoder_settings.h"
>-#include "test/fake_audio_device.h"
> #include "test/fake_decoder.h"
>-#include "test/fake_encoder.h"
> #include "test/fake_videorenderer.h"
> #include "test/frame_generator_capturer.h"
>+#include "test/function_video_encoder_factory.h"
> #include "test/rtp_rtcp_observer.h"
> #include "test/single_threaded_task_queue.h"
> 
> namespace webrtc {
>-
>-class VoEBase;
>-
> namespace test {
> 
> class BaseTest;
>@@ -38,23 +35,26 @@ class CallTest : public ::testing::Test {
>   CallTest();
>   virtual ~CallTest();
> 
>-  static const size_t kNumSsrcs = 3;
>+  static constexpr size_t kNumSsrcs = 6;
>+  static const int kNumSimulcastStreams = 3;
>   static const int kDefaultWidth = 320;
>   static const int kDefaultHeight = 180;
>   static const int kDefaultFramerate = 30;
>   static const int kDefaultTimeoutMs;
>   static const int kLongTimeoutMs;
>-  static const uint8_t kVideoSendPayloadType;
>-  static const uint8_t kSendRtxPayloadType;
>-  static const uint8_t kFakeVideoSendPayloadType;
>-  static const uint8_t kRedPayloadType;
>-  static const uint8_t kRtxRedPayloadType;
>-  static const uint8_t kUlpfecPayloadType;
>-  static const uint8_t kFlexfecPayloadType;
>-  static const uint8_t kAudioSendPayloadType;
>-  static const uint8_t kPayloadTypeH264;
>-  static const uint8_t kPayloadTypeVP8;
>-  static const uint8_t kPayloadTypeVP9;
>+  enum classPayloadTypes : uint8_t {
>+    kSendRtxPayloadType = 98,
>+    kRtxRedPayloadType = 99,
>+    kVideoSendPayloadType = 100,
>+    kAudioSendPayloadType = 103,
>+    kRedPayloadType = 118,
>+    kUlpfecPayloadType = 119,
>+    kFlexfecPayloadType = 120,
>+    kPayloadTypeH264 = 122,
>+    kPayloadTypeVP8 = 123,
>+    kPayloadTypeVP9 = 124,
>+    kFakeVideoSendPayloadType = 125,
>+  };
>   static const uint32_t kSendRtxSsrcs[kNumSsrcs];
>   static const uint32_t kVideoSendSsrcs[kNumSsrcs];
>   static const uint32_t kAudioSendSsrc;
>@@ -66,22 +66,64 @@ class CallTest : public ::testing::Test {
>   static const std::map<uint8_t, MediaType> payload_type_map_;
> 
>  protected:
>-  // RunBaseTest overwrites the audio_state and the voice_engine of the send and
>-  // receive Call configs to simplify test code and avoid having old VoiceEngine
>-  // APIs in the tests.
>+  // RunBaseTest overwrites the audio_state of the send and receive Call configs
>+  // to simplify test code.
>   void RunBaseTest(BaseTest* test);
> 
>+  void CreateCalls();
>   void CreateCalls(const Call::Config& sender_config,
>                    const Call::Config& receiver_config);
>+  void CreateSenderCall();
>   void CreateSenderCall(const Call::Config& config);
>   void CreateReceiverCall(const Call::Config& config);
>   void DestroyCalls();
> 
>+  void CreateVideoSendConfig(VideoSendStream::Config* video_config,
>+                             size_t num_video_streams,
>+                             size_t num_used_ssrcs,
>+                             Transport* send_transport);
>+  void CreateAudioAndFecSendConfigs(size_t num_audio_streams,
>+                                    size_t num_flexfec_streams,
>+                                    Transport* send_transport);
>+  void SetAudioConfig(const AudioSendStream::Config& config);
>+
>+  void SetSendFecConfig(std::vector<uint32_t> video_send_ssrcs);
>+  void SetSendUlpFecConfig(VideoSendStream::Config* send_config);
>+  void SetReceiveUlpFecConfig(VideoReceiveStream::Config* receive_config);
>   void CreateSendConfig(size_t num_video_streams,
>                         size_t num_audio_streams,
>                         size_t num_flexfec_streams,
>                         Transport* send_transport);
> 
>+  void CreateMatchingVideoReceiveConfigs(
>+      const VideoSendStream::Config& video_send_config,
>+      Transport* rtcp_send_transport);
>+  void CreateMatchingVideoReceiveConfigs(
>+      const VideoSendStream::Config& video_send_config,
>+      Transport* rtcp_send_transport,
>+      bool send_side_bwe,
>+      absl::optional<size_t> decode_sub_stream,
>+      bool receiver_reference_time_report,
>+      int rtp_history_ms);
>+  void AddMatchingVideoReceiveConfigs(
>+      std::vector<VideoReceiveStream::Config>* receive_configs,
>+      const VideoSendStream::Config& video_send_config,
>+      Transport* rtcp_send_transport,
>+      bool send_side_bwe,
>+      absl::optional<size_t> decode_sub_stream,
>+      bool receiver_reference_time_report,
>+      int rtp_history_ms);
>+
>+  void CreateMatchingAudioAndFecConfigs(Transport* rtcp_send_transport);
>+  void CreateMatchingAudioConfigs(Transport* transport, std::string sync_group);
>+  static AudioReceiveStream::Config CreateMatchingAudioConfig(
>+      const AudioSendStream::Config& send_config,
>+      rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory,
>+      Transport* transport,
>+      std::string sync_group);
>+  void CreateMatchingFecConfig(
>+      Transport* transport,
>+      const VideoSendStream::Config& video_send_config);
>   void CreateMatchingReceiveConfigs(Transport* rtcp_send_transport);
> 
>   void CreateFrameGeneratorCapturerWithDrift(Clock* drift_clock,
>@@ -91,30 +133,49 @@ class CallTest : public ::testing::Test {
>                                              int height);
>   void CreateFrameGeneratorCapturer(int framerate, int width, int height);
>   void CreateFakeAudioDevices(
>-      std::unique_ptr<FakeAudioDevice::Capturer> capturer,
>-      std::unique_ptr<FakeAudioDevice::Renderer> renderer);
>+      std::unique_ptr<TestAudioDeviceModule::Capturer> capturer,
>+      std::unique_ptr<TestAudioDeviceModule::Renderer> renderer);
> 
>   void CreateVideoStreams();
>+  void CreateVideoSendStreams();
>+  void CreateVideoSendStream(const VideoEncoderConfig& encoder_config);
>   void CreateAudioStreams();
>   void CreateFlexfecStreams();
> 
>+  void ConnectVideoSourcesToStreams();
>+
>   void AssociateFlexfecStreamsWithVideoStreams();
>   void DissociateFlexfecStreamsFromVideoStreams();
> 
>   void Start();
>+  void StartVideoStreams();
>+  void StartVideoCapture();
>   void Stop();
>+  void StopVideoCapture();
>+  void StopVideoStreams();
>   void DestroyStreams();
>+  void DestroyVideoSendStreams();
>   void SetFakeVideoCaptureRotation(VideoRotation rotation);
> 
>+  void SetVideoDegradation(DegradationPreference preference);
>+
>+  VideoSendStream::Config* GetVideoSendConfig();
>+  void SetVideoSendConfig(const VideoSendStream::Config& config);
>+  VideoEncoderConfig* GetVideoEncoderConfig();
>+  void SetVideoEncoderConfig(const VideoEncoderConfig& config);
>+  VideoSendStream* GetVideoSendStream();
>+  FlexfecReceiveStream::Config* GetFlexFecConfig();
>+
>   Clock* const clock_;
> 
>-  std::unique_ptr<webrtc::RtcEventLog> event_log_;
>+  std::unique_ptr<webrtc::RtcEventLog> send_event_log_;
>+  std::unique_ptr<webrtc::RtcEventLog> recv_event_log_;
>   std::unique_ptr<Call> sender_call_;
>   RtpTransportControllerSend* sender_call_transport_controller_;
>   std::unique_ptr<PacketTransport> send_transport_;
>-  VideoSendStream::Config video_send_config_;
>-  VideoEncoderConfig video_encoder_config_;
>-  VideoSendStream* video_send_stream_;
>+  std::vector<VideoSendStream::Config> video_send_configs_;
>+  std::vector<VideoEncoderConfig> video_encoder_configs_;
>+  std::vector<VideoSendStream*> video_send_streams_;
>   AudioSendStream::Config audio_send_config_;
>   AudioSendStream* audio_send_stream_;
> 
>@@ -127,50 +188,40 @@ class CallTest : public ::testing::Test {
>   std::vector<FlexfecReceiveStream::Config> flexfec_receive_configs_;
>   std::vector<FlexfecReceiveStream*> flexfec_receive_streams_;
> 
>-  std::unique_ptr<test::FrameGeneratorCapturer> frame_generator_capturer_;
>-  test::FakeEncoder fake_encoder_;
>+  test::FrameGeneratorCapturer* frame_generator_capturer_;
>+  std::vector<rtc::VideoSourceInterface<VideoFrame>*> video_sources_;
>+  std::vector<std::unique_ptr<VideoCapturer>> video_capturers_;
>+  DegradationPreference degradation_preference_ =
>+      DegradationPreference::MAINTAIN_FRAMERATE;
>+
>+  std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory_;
>+  std::unique_ptr<NetworkControllerFactoryInterface>
>+      bbr_network_controller_factory_;
>+
>+  test::FunctionVideoEncoderFactory fake_encoder_factory_;
>+  int fake_encoder_max_bitrate_ = -1;
>   std::vector<std::unique_ptr<VideoDecoder>> allocated_decoders_;
>+  // Number of simulcast substreams.
>   size_t num_video_streams_;
>   size_t num_audio_streams_;
>   size_t num_flexfec_streams_;
>-  rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
>-  rtc::scoped_refptr<AudioEncoderFactory> encoder_factory_;
>+  rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory_;
>+  rtc::scoped_refptr<AudioEncoderFactory> audio_encoder_factory_;
>   test::FakeVideoRenderer fake_renderer_;
> 
>   SingleThreadedTaskQueueForTesting task_queue_;
> 
>  private:
>-  // TODO(holmer): Remove once VoiceEngine is fully refactored to the new API.
>-  // These methods are used to set up legacy voice engines and channels which is
>-  // necessary while voice engine is being refactored to the new stream API.
>-  struct VoiceEngineState {
>-    VoiceEngineState()
>-        : voice_engine(nullptr),
>-          base(nullptr),
>-          channel_id(-1) {}
>-
>-    VoiceEngine* voice_engine;
>-    VoEBase* base;
>-    int channel_id;
>-  };
>-
>-  void CreateVoiceEngines();
>-  void DestroyVoiceEngines();
>-
>-  VoiceEngineState voe_send_;
>-  VoiceEngineState voe_recv_;
>   rtc::scoped_refptr<AudioProcessing> apm_send_;
>   rtc::scoped_refptr<AudioProcessing> apm_recv_;
>-
>-  // The audio devices must outlive the voice engines.
>-  std::unique_ptr<test::FakeAudioDevice> fake_send_audio_device_;
>-  std::unique_ptr<test::FakeAudioDevice> fake_recv_audio_device_;
>+  rtc::scoped_refptr<TestAudioDeviceModule> fake_send_audio_device_;
>+  rtc::scoped_refptr<TestAudioDeviceModule> fake_recv_audio_device_;
> };
> 
> class BaseTest : public RtpRtcpObserver {
>  public:
>   BaseTest();
>-  explicit BaseTest(unsigned int timeout_ms);
>+  explicit BaseTest(int timeout_ms);
>   virtual ~BaseTest();
> 
>   virtual void PerformTest() = 0;
>@@ -180,13 +231,15 @@ class BaseTest : public RtpRtcpObserver {
>   virtual size_t GetNumAudioStreams() const;
>   virtual size_t GetNumFlexfecStreams() const;
> 
>-  virtual std::unique_ptr<FakeAudioDevice::Capturer> CreateCapturer();
>-  virtual std::unique_ptr<FakeAudioDevice::Renderer> CreateRenderer();
>-  virtual void OnFakeAudioDevicesCreated(FakeAudioDevice* send_audio_device,
>-                                         FakeAudioDevice* recv_audio_device);
>+  virtual std::unique_ptr<TestAudioDeviceModule::Capturer> CreateCapturer();
>+  virtual std::unique_ptr<TestAudioDeviceModule::Renderer> CreateRenderer();
>+  virtual void OnFakeAudioDevicesCreated(
>+      TestAudioDeviceModule* send_audio_device,
>+      TestAudioDeviceModule* recv_audio_device);
>+
>+  virtual void ModifySenderCallConfig(Call::Config* config);
>+  virtual void ModifyReceiverCallConfig(Call::Config* config);
> 
>-  virtual Call::Config GetSenderCallConfig();
>-  virtual Call::Config GetReceiverCallConfig();
>   virtual void OnRtpTransportControllerSendCreated(
>       RtpTransportControllerSend* controller);
>   virtual void OnCallsCreated(Call* sender_call, Call* receiver_call);
>@@ -224,13 +277,11 @@ class BaseTest : public RtpRtcpObserver {
>       FrameGeneratorCapturer* frame_generator_capturer);
> 
>   virtual void OnStreamsStopped();
>-
>-  std::unique_ptr<webrtc::RtcEventLog> event_log_;
> };
> 
> class SendTest : public BaseTest {
>  public:
>-  explicit SendTest(unsigned int timeout_ms);
>+  explicit SendTest(int timeout_ms);
> 
>   bool ShouldCreateReceivers() const override;
> };
>@@ -238,7 +289,7 @@ class SendTest : public BaseTest {
> class EndToEndTest : public BaseTest {
>  public:
>   EndToEndTest();
>-  explicit EndToEndTest(unsigned int timeout_ms);
>+  explicit EndToEndTest(int timeout_ms);
> 
>   bool ShouldCreateReceivers() const override;
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/configurable_frame_size_encoder.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/configurable_frame_size_encoder.cc
>index cbfb35bf339..4fb8df473b5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/configurable_frame_size_encoder.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/configurable_frame_size_encoder.cc
>@@ -42,8 +42,8 @@ int32_t ConfigurableFrameSizeEncoder::Encode(
>     const VideoFrame& inputImage,
>     const CodecSpecificInfo* codecSpecificInfo,
>     const std::vector<FrameType>* frame_types) {
>-  EncodedImage encodedImage(
>-      buffer_.get(), current_frame_size_, max_frame_size_);
>+  EncodedImage encodedImage(buffer_.get(), current_frame_size_,
>+                            max_frame_size_);
>   encodedImage._completeFrame = true;
>   encodedImage._encodedHeight = inputImage.height();
>   encodedImage._encodedWidth = inputImage.width();
>@@ -74,15 +74,11 @@ int32_t ConfigurableFrameSizeEncoder::SetChannelParameters(uint32_t packet_loss,
> }
> 
> int32_t ConfigurableFrameSizeEncoder::SetRateAllocation(
>-    const BitrateAllocation& allocation,
>+    const VideoBitrateAllocation& allocation,
>     uint32_t framerate) {
>   return WEBRTC_VIDEO_CODEC_OK;
> }
> 
>-int32_t ConfigurableFrameSizeEncoder::SetPeriodicKeyFrames(bool enable) {
>-  return WEBRTC_VIDEO_CODEC_OK;
>-}
>-
> int32_t ConfigurableFrameSizeEncoder::SetFrameSize(size_t size) {
>   RTC_DCHECK_LE(size, max_frame_size_);
>   current_frame_size_ = size;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/configurable_frame_size_encoder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/configurable_frame_size_encoder.h
>index 8d3cbc65afe..b8c3b830d4b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/configurable_frame_size_encoder.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/configurable_frame_size_encoder.h
>@@ -39,11 +39,9 @@ class ConfigurableFrameSizeEncoder : public VideoEncoder {
> 
>   int32_t SetChannelParameters(uint32_t packet_loss, int64_t rtt) override;
> 
>-  int32_t SetRateAllocation(const BitrateAllocation& allocation,
>+  int32_t SetRateAllocation(const VideoBitrateAllocation& allocation,
>                             uint32_t framerate) override;
> 
>-  int32_t SetPeriodicKeyFrames(bool enable) override;
>-
>   int32_t SetFrameSize(size_t size);
> 
>  private:
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/direct_transport.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/direct_transport.cc
>index dddbe589092..93221a6055b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/direct_transport.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/direct_transport.cc
>@@ -9,14 +9,32 @@
>  */
> #include "test/direct_transport.h"
> 
>+#include "absl/memory/memory.h"
> #include "call/call.h"
>-#include "rtc_base/ptr_util.h"
>+#include "modules/rtp_rtcp/include/rtp_header_parser.h"
> #include "system_wrappers/include/clock.h"
> #include "test/single_threaded_task_queue.h"
> 
> namespace webrtc {
> namespace test {
> 
>+Demuxer::Demuxer(const std::map<uint8_t, MediaType>& payload_type_map)
>+    : payload_type_map_(payload_type_map) {}
>+
>+MediaType Demuxer::GetMediaType(const uint8_t* packet_data,
>+                                const size_t packet_length) const {
>+  if (!RtpHeaderParser::IsRtcp(packet_data, packet_length)) {
>+    RTC_CHECK_GE(packet_length, 2);
>+    const uint8_t payload_type = packet_data[1] & 0x7f;
>+    std::map<uint8_t, MediaType>::const_iterator it =
>+        payload_type_map_.find(payload_type);
>+    RTC_CHECK(it != payload_type_map_.end())
>+        << "payload type " << static_cast<int>(payload_type) << " unknown.";
>+    return it->second;
>+  }
>+  return MediaType::ANY;
>+}
>+
> DirectTransport::DirectTransport(
>     SingleThreadedTaskQueueForTesting* task_queue,
>     Call* send_call,
>@@ -24,39 +42,30 @@ DirectTransport::DirectTransport(
>     : DirectTransport(task_queue,
>                       FakeNetworkPipe::Config(),
>                       send_call,
>-                      payload_type_map) {
>-}
>+                      payload_type_map) {}
> 
> DirectTransport::DirectTransport(
>     SingleThreadedTaskQueueForTesting* task_queue,
>     const FakeNetworkPipe::Config& config,
>     Call* send_call,
>     const std::map<uint8_t, MediaType>& payload_type_map)
>-    : DirectTransport(
>-          task_queue,
>-          config,
>-          send_call,
>-          std::unique_ptr<Demuxer>(new DemuxerImpl(payload_type_map))) {
>-}
>-
>-DirectTransport::DirectTransport(SingleThreadedTaskQueueForTesting* task_queue,
>-                                 const FakeNetworkPipe::Config& config,
>-                                 Call* send_call,
>-                                 std::unique_ptr<Demuxer> demuxer)
>     : send_call_(send_call),
>       clock_(Clock::GetRealTimeClock()),
>       task_queue_(task_queue),
>-      fake_network_(rtc::MakeUnique<FakeNetworkPipe>(clock_, config,
>-                                                      std::move(demuxer))) {
>+      demuxer_(payload_type_map),
>+      fake_network_(absl::make_unique<FakeNetworkPipe>(clock_, config)) {
>   Start();
> }
> 
>-DirectTransport::DirectTransport(SingleThreadedTaskQueueForTesting* task_queue,
>-                                 std::unique_ptr<FakeNetworkPipe> pipe,
>-                                 Call* send_call)
>+DirectTransport::DirectTransport(
>+    SingleThreadedTaskQueueForTesting* task_queue,
>+    std::unique_ptr<FakeNetworkPipe> pipe,
>+    Call* send_call,
>+    const std::map<uint8_t, MediaType>& payload_type_map)
>     : send_call_(send_call),
>       clock_(Clock::GetRealTimeClock()),
>       task_queue_(task_queue),
>+      demuxer_(payload_type_map),
>       fake_network_(std::move(pipe)) {
>   Start();
> }
>@@ -68,6 +77,10 @@ DirectTransport::~DirectTransport() {
>   task_queue_->CancelTask(next_scheduled_task_);
> }
> 
>+void DirectTransport::SetClockOffset(int64_t offset_ms) {
>+  fake_network_->SetClockOffset(offset_ms);
>+}
>+
> void DirectTransport::SetConfig(const FakeNetworkPipe::Config& config) {
>   fake_network_->SetConfig(config);
> }
>@@ -90,15 +103,22 @@ bool DirectTransport::SendRtp(const uint8_t* data,
>                                 clock_->TimeInMilliseconds());
>     send_call_->OnSentPacket(sent_packet);
>   }
>-  fake_network_->SendPacket(data, length);
>+  SendPacket(data, length);
>   return true;
> }
> 
> bool DirectTransport::SendRtcp(const uint8_t* data, size_t length) {
>-  fake_network_->SendPacket(data, length);
>+  SendPacket(data, length);
>   return true;
> }
> 
>+void DirectTransport::SendPacket(const uint8_t* data, size_t length) {
>+  MediaType media_type = demuxer_.GetMediaType(data, length);
>+  int64_t send_time = clock_->TimeInMicroseconds();
>+  fake_network_->DeliverPacket(media_type, rtc::CopyOnWriteBuffer(data, length),
>+                               send_time);
>+}
>+
> int DirectTransport::GetAverageDelayMs() {
>   return fake_network_->AverageDelay();
> }
>@@ -118,9 +138,8 @@ void DirectTransport::SendPackets() {
>   fake_network_->Process();
> 
>   int64_t delay_ms = fake_network_->TimeUntilNextProcess();
>-  next_scheduled_task_ = task_queue_->PostDelayedTask([this]() {
>-    SendPackets();
>-  }, delay_ms);
>+  next_scheduled_task_ =
>+      task_queue_->PostDelayedTask([this]() { SendPackets(); }, delay_ms);
> }
> }  // namespace test
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/direct_transport.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/direct_transport.h
>index 85cd892ec2f..2d473073e91 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/direct_transport.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/direct_transport.h
>@@ -16,9 +16,9 @@
> 
> #include "api/call/transport.h"
> #include "call/call.h"
>+#include "call/fake_network_pipe.h"
> #include "rtc_base/sequenced_task_checker.h"
> #include "rtc_base/thread_annotations.h"
>-#include "test/fake_network_pipe.h"
> #include "test/single_threaded_task_queue.h"
> 
> namespace webrtc {
>@@ -27,6 +27,15 @@ class Clock;
> class PacketReceiver;
> 
> namespace test {
>+class Demuxer {
>+ public:
>+  explicit Demuxer(const std::map<uint8_t, MediaType>& payload_type_map);
>+  ~Demuxer() = default;
>+  MediaType GetMediaType(const uint8_t* packet_data,
>+                         const size_t packet_length) const;
>+  const std::map<uint8_t, MediaType> payload_type_map_;
>+  RTC_DISALLOW_COPY_AND_ASSIGN(Demuxer);
>+};
> 
> // Objects of this class are expected to be allocated and destroyed  on the
> // same task-queue - the one that's passed in via the constructor.
>@@ -42,15 +51,14 @@ class DirectTransport : public Transport {
>                   const std::map<uint8_t, MediaType>& payload_type_map);
> 
>   DirectTransport(SingleThreadedTaskQueueForTesting* task_queue,
>-                  const FakeNetworkPipe::Config& config,
>+                  std::unique_ptr<FakeNetworkPipe> pipe,
>                   Call* send_call,
>-                  std::unique_ptr<Demuxer> demuxer);
>-
>-  DirectTransport(SingleThreadedTaskQueueForTesting* task_queue,
>-                  std::unique_ptr<FakeNetworkPipe> pipe, Call* send_call);
>+                  const std::map<uint8_t, MediaType>& payload_type_map);
> 
>   ~DirectTransport() override;
> 
>+  void SetClockOffset(int64_t offset_ms);
>+
>   void SetConfig(const FakeNetworkPipe::Config& config);
> 
>   RTC_DEPRECATED void StopSending();
>@@ -67,6 +75,7 @@ class DirectTransport : public Transport {
> 
>  private:
>   void SendPackets();
>+  void SendPacket(const uint8_t* data, size_t length);
>   void Start();
> 
>   Call* const send_call_;
>@@ -76,7 +85,8 @@ class DirectTransport : public Transport {
>   SingleThreadedTaskQueueForTesting::TaskId next_scheduled_task_
>       RTC_GUARDED_BY(&sequence_checker_);
> 
>-  std::unique_ptr<FakeNetworkPipe> fake_network_;
>+  const Demuxer demuxer_;
>+  const std::unique_ptr<FakeNetworkPipe> fake_network_;
> 
>   rtc::SequencedTaskChecker sequence_checker_;
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/direct_transport_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/direct_transport_unittest.cc
>new file mode 100644
>index 00000000000..d35c4a4c033
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/direct_transport_unittest.cc
>@@ -0,0 +1,30 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#include "test/direct_transport.h"
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+namespace test {
>+TEST(DemuxerTest, Demuxing) {
>+  constexpr uint8_t kVideoPayloadType = 100;
>+  constexpr uint8_t kAudioPayloadType = 101;
>+  constexpr size_t kPacketSize = 10;
>+  Demuxer demuxer({{kVideoPayloadType, MediaType::VIDEO},
>+                   {kAudioPayloadType, MediaType::AUDIO}});
>+
>+  uint8_t data[kPacketSize];
>+  memset(data, 0, kPacketSize);
>+  data[1] = kVideoPayloadType;
>+  EXPECT_EQ(demuxer.GetMediaType(data, kPacketSize), MediaType::VIDEO);
>+  data[1] = kAudioPayloadType;
>+  EXPECT_EQ(demuxer.GetMediaType(data, kPacketSize), MediaType::AUDIO);
>+}
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/encoder_proxy_factory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/encoder_proxy_factory.h
>new file mode 100644
>index 00000000000..8672a464923
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/encoder_proxy_factory.h
>@@ -0,0 +1,103 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef TEST_ENCODER_PROXY_FACTORY_H_
>+#define TEST_ENCODER_PROXY_FACTORY_H_
>+
>+#include <memory>
>+#include <vector>
>+
>+#include "absl/memory/memory.h"
>+#include "api/video_codecs/video_encoder.h"
>+#include "api/video_codecs/video_encoder_factory.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+// An encoder factory with a single underlying VideoEncoder object,
>+// intended for test purposes. Each call to CreateVideoEncoder returns
>+// a proxy for the same encoder, typically an instance of FakeEncoder.
>+class EncoderProxyFactory final : public VideoEncoderFactory {
>+ public:
>+  explicit EncoderProxyFactory(VideoEncoder* encoder) : encoder_(encoder) {
>+    codec_info_.is_hardware_accelerated = false;
>+    codec_info_.has_internal_source = false;
>+  }
>+
>+  // Unused by tests.
>+  std::vector<SdpVideoFormat> GetSupportedFormats() const override {
>+    RTC_NOTREACHED();
>+    return {};
>+  }
>+
>+  CodecInfo QueryVideoEncoder(const SdpVideoFormat& format) const override {
>+    return codec_info_;
>+  }
>+
>+  std::unique_ptr<VideoEncoder> CreateVideoEncoder(
>+      const SdpVideoFormat& format) override {
>+    return absl::make_unique<EncoderProxy>(encoder_);
>+  }
>+
>+  void SetIsHardwareAccelerated(bool is_hardware_accelerated) {
>+    codec_info_.is_hardware_accelerated = is_hardware_accelerated;
>+  }
>+  void SetHasInternalSource(bool has_internal_source) {
>+    codec_info_.has_internal_source = has_internal_source;
>+  }
>+
>+ private:
>+  // Wrapper class, since CreateVideoEncoder needs to surrender
>+  // ownership to the object it returns.
>+  class EncoderProxy final : public VideoEncoder {
>+   public:
>+    explicit EncoderProxy(VideoEncoder* encoder) : encoder_(encoder) {}
>+
>+   private:
>+    int32_t Encode(const VideoFrame& input_image,
>+                   const CodecSpecificInfo* codec_specific_info,
>+                   const std::vector<FrameType>* frame_types) override {
>+      return encoder_->Encode(input_image, codec_specific_info, frame_types);
>+    }
>+    int32_t InitEncode(const VideoCodec* config,
>+                       int32_t number_of_cores,
>+                       size_t max_payload_size) override {
>+      return encoder_->InitEncode(config, number_of_cores, max_payload_size);
>+    }
>+    VideoEncoder::ScalingSettings GetScalingSettings() const override {
>+      return encoder_->GetScalingSettings();
>+    }
>+    int32_t RegisterEncodeCompleteCallback(
>+        EncodedImageCallback* callback) override {
>+      return encoder_->RegisterEncodeCompleteCallback(callback);
>+    }
>+    int32_t Release() override { return encoder_->Release(); }
>+    int32_t SetChannelParameters(uint32_t packet_loss, int64_t rtt) override {
>+      return encoder_->SetChannelParameters(packet_loss, rtt);
>+    }
>+    int32_t SetRateAllocation(const VideoBitrateAllocation& rate_allocation,
>+                              uint32_t framerate) override {
>+      return encoder_->SetRateAllocation(rate_allocation, framerate);
>+    }
>+    const char* ImplementationName() const override {
>+      return encoder_->ImplementationName();
>+    }
>+
>+    VideoEncoder* const encoder_;
>+  };
>+
>+  VideoEncoder* const encoder_;
>+  CodecInfo codec_info_;
>+};
>+
>+}  // namespace test
>+}  // namespace webrtc
>+
>+#endif  // TEST_ENCODER_PROXY_FACTORY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/encoder_settings.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/encoder_settings.cc
>index bd700cf6d28..8e21598c2d7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/encoder_settings.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/encoder_settings.cc
>@@ -12,7 +12,9 @@
> #include <algorithm>
> #include <string>
> 
>+#include "media/engine/internaldecoderfactory.h"
> #include "modules/video_coding/codecs/h264/include/h264.h"
>+#include "modules/video_coding/codecs/multiplex/include/multiplex_decoder_adapter.h"
> #include "modules/video_coding/codecs/vp8/include/vp8.h"
> #include "modules/video_coding/codecs/vp9/include/vp9.h"
> #include "rtc_base/refcountedobject.h"
>@@ -46,15 +48,50 @@ std::vector<VideoStream> CreateVideoStreams(
>     stream_settings[i].max_framerate = 30;
>     stream_settings[i].min_bitrate_bps =
>         DefaultVideoStreamFactory::kDefaultMinBitratePerStream[i];
>-    stream_settings[i].target_bitrate_bps = stream_settings[i].max_bitrate_bps =
>-        std::min(bitrate_left_bps,
>-                 DefaultVideoStreamFactory::kMaxBitratePerStream[i]);
>+
>+    int target_bitrate_bps = -1;
>+    int max_bitrate_bps = -1;
>+    // Use configured values instead of default values if values has been
>+    // configured.
>+    if (i < encoder_config.simulcast_layers.size()) {
>+      const VideoStream& stream = encoder_config.simulcast_layers[i];
>+
>+      max_bitrate_bps =
>+          stream.max_bitrate_bps > 0
>+              ? stream.max_bitrate_bps
>+              : DefaultVideoStreamFactory::kMaxBitratePerStream[i];
>+      max_bitrate_bps = std::min(bitrate_left_bps, max_bitrate_bps);
>+
>+      target_bitrate_bps =
>+          stream.target_bitrate_bps > 0
>+              ? stream.target_bitrate_bps
>+              : DefaultVideoStreamFactory::kMaxBitratePerStream[i];
>+      target_bitrate_bps = std::min(max_bitrate_bps, target_bitrate_bps);
>+    } else {
>+      max_bitrate_bps = std::min(
>+          bitrate_left_bps, DefaultVideoStreamFactory::kMaxBitratePerStream[i]);
>+      target_bitrate_bps = max_bitrate_bps;
>+    }
>+
>+    RTC_DCHECK_NE(target_bitrate_bps, -1);
>+    RTC_DCHECK_NE(max_bitrate_bps, -1);
>+    stream_settings[i].target_bitrate_bps = target_bitrate_bps;
>+    stream_settings[i].max_bitrate_bps = max_bitrate_bps;
>     stream_settings[i].max_qp = 56;
>+
>+    if (i < encoder_config.simulcast_layers.size()) {
>+      // Higher level controls are setting the active configuration for the
>+      // VideoStream.
>+      stream_settings[i].active = encoder_config.simulcast_layers[i].active;
>+    } else {
>+      stream_settings[i].active = true;
>+    }
>     bitrate_left_bps -= stream_settings[i].target_bitrate_bps;
>   }
> 
>   stream_settings[encoder_config.number_of_streams - 1].max_bitrate_bps +=
>       bitrate_left_bps;
>+  stream_settings[0].bitrate_priority = encoder_config.bitrate_priority;
> 
>   return stream_settings;
> }
>@@ -68,14 +105,17 @@ std::vector<VideoStream> DefaultVideoStreamFactory::CreateEncoderStreams(
>   return CreateVideoStreams(width, height, encoder_config);
> }
> 
>-void FillEncoderConfiguration(size_t num_streams,
>+void FillEncoderConfiguration(VideoCodecType codec_type,
>+                              size_t num_streams,
>                               VideoEncoderConfig* configuration) {
>   RTC_DCHECK_LE(num_streams, DefaultVideoStreamFactory::kMaxNumberOfStreams);
> 
>+  configuration->codec_type = codec_type;
>   configuration->number_of_streams = num_streams;
>   configuration->video_stream_factory =
>       new rtc::RefCountedObject<DefaultVideoStreamFactory>();
>   configuration->max_bitrate_bps = 0;
>+  configuration->simulcast_layers = std::vector<VideoStream>(num_streams);
>   for (size_t i = 0; i < num_streams; ++i) {
>     configuration->max_bitrate_bps +=
>         DefaultVideoStreamFactory::kMaxBitratePerStream[i];
>@@ -83,20 +123,31 @@ void FillEncoderConfiguration(size_t num_streams,
> }
> 
> VideoReceiveStream::Decoder CreateMatchingDecoder(
>-    const VideoSendStream::Config::EncoderSettings& encoder_settings) {
>+    int payload_type,
>+    const std::string& payload_name) {
>   VideoReceiveStream::Decoder decoder;
>-  decoder.payload_type = encoder_settings.payload_type;
>-  decoder.payload_name = encoder_settings.payload_name;
>-  if (encoder_settings.payload_name == "H264") {
>+  decoder.payload_type = payload_type;
>+  decoder.payload_name = payload_name;
>+  if (payload_name == "H264") {
>     decoder.decoder = H264Decoder::Create().release();
>-  } else if (encoder_settings.payload_name == "VP8") {
>+  } else if (payload_name == "VP8") {
>     decoder.decoder = VP8Decoder::Create().release();
>-  } else if (encoder_settings.payload_name == "VP9") {
>+  } else if (payload_name == "VP9") {
>     decoder.decoder = VP9Decoder::Create().release();
>+  } else if (payload_name == "multiplex") {
>+    decoder.decoder = new MultiplexDecoderAdapter(
>+        new InternalDecoderFactory(), SdpVideoFormat(cricket::kVp9CodecName));
>   } else {
>     decoder.decoder = new FakeDecoder();
>   }
>   return decoder;
> }
>+
>+VideoReceiveStream::Decoder CreateMatchingDecoder(
>+    const VideoSendStream::Config& config) {
>+  return CreateMatchingDecoder(config.rtp.payload_type,
>+                               config.rtp.payload_name);
>+}
>+
> }  // namespace test
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/encoder_settings.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/encoder_settings.h
>index 25cfa298975..c05ac938162 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/encoder_settings.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/encoder_settings.h
>@@ -45,11 +45,16 @@ std::vector<VideoStream> CreateVideoStreams(
>     int height,
>     const webrtc::VideoEncoderConfig& encoder_config);
> 
>-void FillEncoderConfiguration(size_t num_streams,
>+void FillEncoderConfiguration(VideoCodecType codec_type,
>+                              size_t num_streams,
>                               VideoEncoderConfig* configuration);
> 
> VideoReceiveStream::Decoder CreateMatchingDecoder(
>-    const VideoSendStream::Config::EncoderSettings& encoder_settings);
>+    int payload_type,
>+    const std::string& payload_name);
>+
>+VideoReceiveStream::Decoder CreateMatchingDecoder(
>+    const VideoSendStream::Config& config);
> }  // namespace test
> }  // namespace webrtc
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_audio_device.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_audio_device.cc
>deleted file mode 100644
>index eef8bfd0159..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_audio_device.cc
>+++ /dev/null
>@@ -1,380 +0,0 @@
>-/*
>- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "test/fake_audio_device.h"
>-
>-#include <algorithm>
>-#include <utility>
>-
>-#include "common_audio/wav_file.h"
>-#include "rtc_base/checks.h"
>-#include "rtc_base/random.h"
>-#include "system_wrappers/include/event_wrapper.h"
>-
>-namespace webrtc {
>-
>-namespace {
>-
>-constexpr int kFrameLengthMs = 10;
>-constexpr int kFramesPerSecond = 1000 / kFrameLengthMs;
>-
>-// Assuming 10ms audio packets..
>-class PulsedNoiseCapturer final : public test::FakeAudioDevice::Capturer {
>- public:
>-  PulsedNoiseCapturer(int16_t max_amplitude, int sampling_frequency_in_hz)
>-      : sampling_frequency_in_hz_(sampling_frequency_in_hz),
>-        fill_with_zero_(false),
>-        random_generator_(1),
>-        max_amplitude_(max_amplitude) {
>-    RTC_DCHECK_GT(max_amplitude, 0);
>-  }
>-
>-  int SamplingFrequency() const override {
>-    return sampling_frequency_in_hz_;
>-  }
>-
>-  bool Capture(rtc::BufferT<int16_t>* buffer) override {
>-    fill_with_zero_ = !fill_with_zero_;
>-    buffer->SetData(
>-        test::FakeAudioDevice::SamplesPerFrame(sampling_frequency_in_hz_),
>-        [&](rtc::ArrayView<int16_t> data) {
>-      if (fill_with_zero_) {
>-        std::fill(data.begin(), data.end(), 0);
>-      } else {
>-        std::generate(data.begin(), data.end(), [&]() {
>-          return random_generator_.Rand(-max_amplitude_, max_amplitude_);
>-        });
>-      }
>-      return data.size();
>-    });
>-    return true;
>-  }
>-
>- private:
>-  int sampling_frequency_in_hz_;
>-  bool fill_with_zero_;
>-  Random random_generator_;
>-  const int16_t max_amplitude_;
>-};
>-
>-class WavFileReader final : public test::FakeAudioDevice::Capturer {
>- public:
>-  WavFileReader(std::string filename, int sampling_frequency_in_hz)
>-      : sampling_frequency_in_hz_(sampling_frequency_in_hz),
>-        wav_reader_(filename) {
>-    RTC_CHECK_EQ(wav_reader_.sample_rate(), sampling_frequency_in_hz);
>-    RTC_CHECK_EQ(wav_reader_.num_channels(), 1);
>-  }
>-
>-  int SamplingFrequency() const override {
>-    return sampling_frequency_in_hz_;
>-  }
>-
>-  bool Capture(rtc::BufferT<int16_t>* buffer) override {
>-    buffer->SetData(
>-        test::FakeAudioDevice::SamplesPerFrame(sampling_frequency_in_hz_),
>-        [&](rtc::ArrayView<int16_t> data) {
>-      return wav_reader_.ReadSamples(data.size(), data.data());
>-    });
>-    return buffer->size() > 0;
>-  }
>-
>- private:
>-  int sampling_frequency_in_hz_;
>-  WavReader wav_reader_;
>-};
>-
>-class WavFileWriter final : public test::FakeAudioDevice::Renderer {
>- public:
>-  WavFileWriter(std::string filename, int sampling_frequency_in_hz)
>-      : sampling_frequency_in_hz_(sampling_frequency_in_hz),
>-        wav_writer_(filename, sampling_frequency_in_hz, 1) {}
>-
>-  int SamplingFrequency() const override {
>-    return sampling_frequency_in_hz_;
>-  }
>-
>-  bool Render(rtc::ArrayView<const int16_t> data) override {
>-    wav_writer_.WriteSamples(data.data(), data.size());
>-    return true;
>-  }
>-
>- private:
>-  int sampling_frequency_in_hz_;
>-  WavWriter wav_writer_;
>-};
>-
>-class BoundedWavFileWriter : public test::FakeAudioDevice::Renderer {
>- public:
>-  BoundedWavFileWriter(std::string filename, int sampling_frequency_in_hz)
>-      : sampling_frequency_in_hz_(sampling_frequency_in_hz),
>-        wav_writer_(filename, sampling_frequency_in_hz, 1),
>-        silent_audio_(test::FakeAudioDevice::SamplesPerFrame(
>-            sampling_frequency_in_hz), 0),
>-        started_writing_(false),
>-        trailing_zeros_(0) {}
>-
>-  int SamplingFrequency() const override {
>-    return sampling_frequency_in_hz_;
>-  }
>-
>-  bool Render(rtc::ArrayView<const int16_t> data) override {
>-    const int16_t kAmplitudeThreshold = 5;
>-
>-    const int16_t* begin = data.begin();
>-    const int16_t* end = data.end();
>-    if (!started_writing_) {
>-      // Cut off silence at the beginning.
>-      while (begin < end) {
>-        if (std::abs(*begin) > kAmplitudeThreshold) {
>-          started_writing_ = true;
>-          break;
>-        }
>-        ++begin;
>-      }
>-    }
>-    if (started_writing_) {
>-      // Cut off silence at the end.
>-      while (begin < end) {
>-        if (*(end - 1) != 0) {
>-          break;
>-        }
>-        --end;
>-      }
>-      if (begin < end) {
>-        // If it turns out that the silence was not final, need to write all the
>-        // skipped zeros and continue writing audio.
>-        while (trailing_zeros_ > 0) {
>-          const size_t zeros_to_write = std::min(trailing_zeros_,
>-                                                 silent_audio_.size());
>-          wav_writer_.WriteSamples(silent_audio_.data(), zeros_to_write);
>-          trailing_zeros_ -= zeros_to_write;
>-        }
>-        wav_writer_.WriteSamples(begin, end - begin);
>-      }
>-      // Save the number of zeros we skipped in case this needs to be restored.
>-      trailing_zeros_ += data.end() - end;
>-    }
>-    return true;
>-  }
>-
>- private:
>-  int sampling_frequency_in_hz_;
>-  WavWriter wav_writer_;
>-  std::vector<int16_t> silent_audio_;
>-  bool started_writing_;
>-  size_t trailing_zeros_;
>-};
>-
>-
>-class DiscardRenderer final : public test::FakeAudioDevice::Renderer {
>- public:
>-  explicit DiscardRenderer(int sampling_frequency_in_hz)
>-      : sampling_frequency_in_hz_(sampling_frequency_in_hz) {}
>-
>-  int SamplingFrequency() const override {
>-    return sampling_frequency_in_hz_;
>-  }
>-
>-  bool Render(rtc::ArrayView<const int16_t> data) override {
>-    return true;
>-  }
>-
>- private:
>-  int sampling_frequency_in_hz_;
>-};
>-
>-}  // namespace
>-namespace test {
>-
>-size_t FakeAudioDevice::SamplesPerFrame(int sampling_frequency_in_hz) {
>-  return rtc::CheckedDivExact(sampling_frequency_in_hz, kFramesPerSecond);
>-}
>-
>-std::unique_ptr<FakeAudioDevice::Capturer>
>-    FakeAudioDevice::CreatePulsedNoiseCapturer(
>-        int16_t max_amplitude, int sampling_frequency_in_hz) {
>-  return std::unique_ptr<FakeAudioDevice::Capturer>(
>-      new PulsedNoiseCapturer(max_amplitude, sampling_frequency_in_hz));
>-}
>-
>-std::unique_ptr<FakeAudioDevice::Capturer> FakeAudioDevice::CreateWavFileReader(
>-    std::string filename, int sampling_frequency_in_hz) {
>-  return std::unique_ptr<FakeAudioDevice::Capturer>(
>-      new WavFileReader(filename, sampling_frequency_in_hz));
>-}
>-
>-std::unique_ptr<FakeAudioDevice::Capturer> FakeAudioDevice::CreateWavFileReader(
>-    std::string filename) {
>-  int sampling_frequency_in_hz = WavReader(filename).sample_rate();
>-  return std::unique_ptr<FakeAudioDevice::Capturer>(
>-      new WavFileReader(filename, sampling_frequency_in_hz));
>-}
>-
>-std::unique_ptr<FakeAudioDevice::Renderer> FakeAudioDevice::CreateWavFileWriter(
>-    std::string filename, int sampling_frequency_in_hz) {
>-  return std::unique_ptr<FakeAudioDevice::Renderer>(
>-      new WavFileWriter(filename, sampling_frequency_in_hz));
>-}
>-
>-std::unique_ptr<FakeAudioDevice::Renderer>
>-    FakeAudioDevice::CreateBoundedWavFileWriter(
>-        std::string filename, int sampling_frequency_in_hz) {
>-  return std::unique_ptr<FakeAudioDevice::Renderer>(
>-      new BoundedWavFileWriter(filename, sampling_frequency_in_hz));
>-}
>-
>-std::unique_ptr<FakeAudioDevice::Renderer>
>-    FakeAudioDevice::CreateDiscardRenderer(int sampling_frequency_in_hz) {
>-  return std::unique_ptr<FakeAudioDevice::Renderer>(
>-      new DiscardRenderer(sampling_frequency_in_hz));
>-}
>-
>-
>-FakeAudioDevice::FakeAudioDevice(std::unique_ptr<Capturer> capturer,
>-                                 std::unique_ptr<Renderer> renderer,
>-                                 float speed)
>-    : capturer_(std::move(capturer)),
>-      renderer_(std::move(renderer)),
>-      speed_(speed),
>-      audio_callback_(nullptr),
>-      rendering_(false),
>-      capturing_(false),
>-      done_rendering_(true, true),
>-      done_capturing_(true, true),
>-      tick_(EventTimerWrapper::Create()),
>-      thread_(FakeAudioDevice::Run, this, "FakeAudioDevice") {
>-  auto good_sample_rate = [](int sr) {
>-    return sr == 8000 || sr == 16000 || sr == 32000
>-        || sr == 44100 || sr == 48000;
>-  };
>-
>-  if (renderer_) {
>-    const int sample_rate = renderer_->SamplingFrequency();
>-    playout_buffer_.resize(SamplesPerFrame(sample_rate), 0);
>-    RTC_CHECK(good_sample_rate(sample_rate));
>-  }
>-  if (capturer_) {
>-    RTC_CHECK(good_sample_rate(capturer_->SamplingFrequency()));
>-  }
>-}
>-
>-FakeAudioDevice::~FakeAudioDevice() {
>-  StopPlayout();
>-  StopRecording();
>-  thread_.Stop();
>-}
>-
>-int32_t FakeAudioDevice::StartPlayout() {
>-  rtc::CritScope cs(&lock_);
>-  RTC_CHECK(renderer_);
>-  rendering_ = true;
>-  done_rendering_.Reset();
>-  return 0;
>-}
>-
>-int32_t FakeAudioDevice::StopPlayout() {
>-  rtc::CritScope cs(&lock_);
>-  rendering_ = false;
>-  done_rendering_.Set();
>-  return 0;
>-}
>-
>-int32_t FakeAudioDevice::StartRecording() {
>-  rtc::CritScope cs(&lock_);
>-  RTC_CHECK(capturer_);
>-  capturing_ = true;
>-  done_capturing_.Reset();
>-  return 0;
>-}
>-
>-int32_t FakeAudioDevice::StopRecording() {
>-  rtc::CritScope cs(&lock_);
>-  capturing_ = false;
>-  done_capturing_.Set();
>-  return 0;
>-}
>-
>-int32_t FakeAudioDevice::Init() {
>-  RTC_CHECK(tick_->StartTimer(true, kFrameLengthMs / speed_));
>-  thread_.Start();
>-  thread_.SetPriority(rtc::kHighPriority);
>-  return 0;
>-}
>-
>-int32_t FakeAudioDevice::RegisterAudioCallback(AudioTransport* callback) {
>-  rtc::CritScope cs(&lock_);
>-  RTC_DCHECK(callback || audio_callback_);
>-  audio_callback_ = callback;
>-  return 0;
>-}
>-
>-bool FakeAudioDevice::Playing() const {
>-  rtc::CritScope cs(&lock_);
>-  return rendering_;
>-}
>-
>-bool FakeAudioDevice::Recording() const {
>-  rtc::CritScope cs(&lock_);
>-  return capturing_;
>-}
>-
>-bool FakeAudioDevice::WaitForPlayoutEnd(int timeout_ms) {
>-  return done_rendering_.Wait(timeout_ms);
>-}
>-
>-bool FakeAudioDevice::WaitForRecordingEnd(int timeout_ms) {
>-  return done_capturing_.Wait(timeout_ms);
>-}
>-
>-bool FakeAudioDevice::Run(void* obj) {
>-  static_cast<FakeAudioDevice*>(obj)->ProcessAudio();
>-  return true;
>-}
>-
>-void FakeAudioDevice::ProcessAudio() {
>-  {
>-    rtc::CritScope cs(&lock_);
>-    if (capturing_) {
>-      // Capture 10ms of audio. 2 bytes per sample.
>-      const bool keep_capturing = capturer_->Capture(&recording_buffer_);
>-      uint32_t new_mic_level;
>-      if (recording_buffer_.size() > 0) {
>-        audio_callback_->RecordedDataIsAvailable(
>-            recording_buffer_.data(), recording_buffer_.size(), 2, 1,
>-            capturer_->SamplingFrequency(), 0, 0, 0, false, new_mic_level);
>-      }
>-      if (!keep_capturing) {
>-        capturing_ = false;
>-        done_capturing_.Set();
>-      }
>-    }
>-    if (rendering_) {
>-      size_t samples_out;
>-      int64_t elapsed_time_ms;
>-      int64_t ntp_time_ms;
>-      const int sampling_frequency = renderer_->SamplingFrequency();
>-      audio_callback_->NeedMorePlayData(
>-          SamplesPerFrame(sampling_frequency), 2, 1, sampling_frequency,
>-          playout_buffer_.data(), samples_out, &elapsed_time_ms, &ntp_time_ms);
>-      const bool keep_rendering = renderer_->Render(
>-          rtc::ArrayView<const int16_t>(playout_buffer_.data(), samples_out));
>-      if (!keep_rendering) {
>-        rendering_ = false;
>-        done_rendering_.Set();
>-      }
>-    }
>-  }
>-  tick_->Wait(WEBRTC_EVENT_INFINITE);
>-}
>-
>-
>-}  // namespace test
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_audio_device.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_audio_device.h
>deleted file mode 100644
>index f71fd016d9a..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_audio_device.h
>+++ /dev/null
>@@ -1,144 +0,0 @@
>-/*
>- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-#ifndef TEST_FAKE_AUDIO_DEVICE_H_
>-#define TEST_FAKE_AUDIO_DEVICE_H_
>-
>-#include <memory>
>-#include <string>
>-#include <vector>
>-
>-#include "api/array_view.h"
>-#include "modules/audio_device/include/fake_audio_device.h"
>-#include "rtc_base/buffer.h"
>-#include "rtc_base/criticalsection.h"
>-#include "rtc_base/event.h"
>-#include "rtc_base/platform_thread.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>-
>-namespace webrtc {
>-
>-class EventTimerWrapper;
>-
>-namespace test {
>-
>-// FakeAudioDevice implements an AudioDevice module that can act both as a
>-// capturer and a renderer. It will use 10ms audio frames.
>-class FakeAudioDevice : public FakeAudioDeviceModule {
>- public:
>-  // Returns the number of samples that Capturers and Renderers with this
>-  // sampling frequency will work with every time Capture or Render is called.
>-  static size_t SamplesPerFrame(int sampling_frequency_in_hz);
>-
>-  class Capturer {
>-   public:
>-    virtual ~Capturer() {}
>-    // Returns the sampling frequency in Hz of the audio data that this
>-    // capturer produces.
>-    virtual int SamplingFrequency() const = 0;
>-    // Replaces the contents of |buffer| with 10ms of captured audio data
>-    // (see FakeAudioDevice::SamplesPerFrame). Returns true if the capturer can
>-    // keep producing data, or false when the capture finishes.
>-    virtual bool Capture(rtc::BufferT<int16_t>* buffer) = 0;
>-  };
>-
>-  class Renderer {
>-   public:
>-    virtual ~Renderer() {}
>-    // Returns the sampling frequency in Hz of the audio data that this
>-    // renderer receives.
>-    virtual int SamplingFrequency() const = 0;
>-    // Renders the passed audio data and returns true if the renderer wants
>-    // to keep receiving data, or false otherwise.
>-    virtual bool Render(rtc::ArrayView<const int16_t> data) = 0;
>-  };
>-
>-  // Creates a new FakeAudioDevice. When capturing or playing, 10 ms audio
>-  // frames will be processed every 10ms / |speed|.
>-  // |capturer| is an object that produces audio data. Can be nullptr if this
>-  // device is never used for recording.
>-  // |renderer| is an object that receives audio data that would have been
>-  // played out. Can be nullptr if this device is never used for playing.
>-  // Use one of the Create... functions to get these instances.
>-  FakeAudioDevice(std::unique_ptr<Capturer> capturer,
>-                  std::unique_ptr<Renderer> renderer,
>-                  float speed = 1);
>-  ~FakeAudioDevice() override;
>-
>-  // Returns a Capturer instance that generates a signal where every second
>-  // frame is zero and every second frame is evenly distributed random noise
>-  // with max amplitude |max_amplitude|.
>-  static std::unique_ptr<Capturer> CreatePulsedNoiseCapturer(
>-      int16_t max_amplitude, int sampling_frequency_in_hz);
>-
>-  // Returns a Capturer instance that gets its data from a file.
>-  static std::unique_ptr<Capturer> CreateWavFileReader(
>-      std::string filename, int sampling_frequency_in_hz);
>-
>-  // Returns a Capturer instance that gets its data from a file.
>-  // Automatically detects sample rate.
>-  static std::unique_ptr<Capturer> CreateWavFileReader(std::string filename);
>-
>-  // Returns a Renderer instance that writes its data to a file.
>-  static std::unique_ptr<Renderer> CreateWavFileWriter(
>-      std::string filename, int sampling_frequency_in_hz);
>-
>-  // Returns a Renderer instance that writes its data to a WAV file, cutting
>-  // off silence at the beginning (not necessarily perfect silence, see
>-  // kAmplitudeThreshold) and at the end (only actual 0 samples in this case).
>-  static std::unique_ptr<Renderer> CreateBoundedWavFileWriter(
>-      std::string filename, int sampling_frequency_in_hz);
>-
>-  // Returns a Renderer instance that does nothing with the audio data.
>-  static std::unique_ptr<Renderer> CreateDiscardRenderer(
>-      int sampling_frequency_in_hz);
>-
>-  int32_t Init() override;
>-  int32_t RegisterAudioCallback(AudioTransport* callback) override;
>-
>-  int32_t StartPlayout() override;
>-  int32_t StopPlayout() override;
>-  int32_t StartRecording() override;
>-  int32_t StopRecording() override;
>-
>-  bool Playing() const override;
>-  bool Recording() const override;
>-
>-  // Blocks until the Renderer refuses to receive data.
>-  // Returns false if |timeout_ms| passes before that happens.
>-  bool WaitForPlayoutEnd(int timeout_ms = rtc::Event::kForever);
>-  // Blocks until the Recorder stops producing data.
>-  // Returns false if |timeout_ms| passes before that happens.
>-  bool WaitForRecordingEnd(int timeout_ms = rtc::Event::kForever);
>-
>- private:
>-  static bool Run(void* obj);
>-  void ProcessAudio();
>-
>-  const std::unique_ptr<Capturer> capturer_ RTC_GUARDED_BY(lock_);
>-  const std::unique_ptr<Renderer> renderer_ RTC_GUARDED_BY(lock_);
>-  const float speed_;
>-
>-  rtc::CriticalSection lock_;
>-  AudioTransport* audio_callback_ RTC_GUARDED_BY(lock_);
>-  bool rendering_ RTC_GUARDED_BY(lock_);
>-  bool capturing_ RTC_GUARDED_BY(lock_);
>-  rtc::Event done_rendering_;
>-  rtc::Event done_capturing_;
>-
>-  std::vector<int16_t> playout_buffer_ RTC_GUARDED_BY(lock_);
>-  rtc::BufferT<int16_t> recording_buffer_ RTC_GUARDED_BY(lock_);
>-
>-  std::unique_ptr<EventTimerWrapper> tick_;
>-  rtc::PlatformThread thread_;
>-};
>-}  // namespace test
>-}  // namespace webrtc
>-
>-#endif  // TEST_FAKE_AUDIO_DEVICE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_audio_device_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_audio_device_unittest.cc
>deleted file mode 100644
>index 02ece3cd85b..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_audio_device_unittest.cc
>+++ /dev/null
>@@ -1,131 +0,0 @@
>-/*
>- *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include <algorithm>
>-#include <array>
>-
>-#include "common_audio/wav_file.h"
>-#include "common_audio/wav_header.h"
>-#include "test/fake_audio_device.h"
>-#include "test/gtest.h"
>-#include "test/testsupport/fileutils.h"
>-
>-namespace webrtc {
>-namespace test {
>-
>-namespace {
>-void RunTest(const std::vector<int16_t>& input_samples,
>-             const std::vector<int16_t>& expected_samples,
>-             size_t samples_per_frame) {
>-  const ::testing::TestInfo* const test_info =
>-      ::testing::UnitTest::GetInstance()->current_test_info();
>-
>-  const std::string output_filename = test::OutputPath() +
>-      "BoundedWavFileWriterTest_" + test_info->name() + ".wav";
>-
>-  static const size_t kSamplesPerFrame = 8;
>-  static const int kSampleRate = kSamplesPerFrame * 100;
>-  EXPECT_EQ(FakeAudioDevice::SamplesPerFrame(kSampleRate), kSamplesPerFrame);
>-
>-  {
>-    std::unique_ptr<FakeAudioDevice::Renderer> writer =
>-        FakeAudioDevice::CreateBoundedWavFileWriter(output_filename, 800);
>-
>-    for (size_t i = 0; i < input_samples.size(); i += kSamplesPerFrame) {
>-      EXPECT_TRUE(writer->Render(rtc::ArrayView<const int16_t>(
>-          &input_samples[i],
>-          std::min(kSamplesPerFrame, input_samples.size() - i))));
>-    }
>-  }
>-
>-  {
>-    WavReader reader(output_filename);
>-    std::vector<int16_t> read_samples(expected_samples.size());
>-    EXPECT_EQ(expected_samples.size(),
>-              reader.ReadSamples(read_samples.size(), read_samples.data()));
>-    EXPECT_EQ(expected_samples, read_samples);
>-
>-    EXPECT_EQ(0u, reader.ReadSamples(read_samples.size(), read_samples.data()));
>-  }
>-
>-  remove(output_filename.c_str());
>-}
>-}  // namespace
>-
>-TEST(BoundedWavFileWriterTest, NoSilence) {
>-  static const std::vector<int16_t> kInputSamples = {
>-      75, 1234, 243, -1231, -22222, 0, 3, 88,
>-      1222, -1213, -13222, -7, -3525, 5787, -25247, 8
>-  };
>-  static const std::vector<int16_t> kExpectedSamples = kInputSamples;
>-  RunTest(kInputSamples, kExpectedSamples, 8);
>-}
>-
>-TEST(BoundedWavFileWriterTest, SomeStartSilence) {
>-  static const std::vector<int16_t> kInputSamples = {
>-      0, 0, 0, 0, 3, 0, 0, 0,
>-      0, 3, -13222, -7, -3525, 5787, -25247, 8
>-  };
>-  static const std::vector<int16_t> kExpectedSamples(kInputSamples.begin() + 10,
>-                                                     kInputSamples.end());
>-  RunTest(kInputSamples, kExpectedSamples, 8);
>-}
>-
>-TEST(BoundedWavFileWriterTest, NegativeStartSilence) {
>-  static const std::vector<int16_t> kInputSamples = {
>-      0, -4, -6, 0, 3, 0, 0, 0,
>-      0, 3, -13222, -7, -3525, 5787, -25247, 8
>-  };
>-  static const std::vector<int16_t> kExpectedSamples(kInputSamples.begin() + 2,
>-                                                     kInputSamples.end());
>-  RunTest(kInputSamples, kExpectedSamples, 8);
>-}
>-
>-TEST(BoundedWavFileWriterTest, SomeEndSilence) {
>-  static const std::vector<int16_t> kInputSamples = {
>-      75, 1234, 243, -1231, -22222, 0, 1, 0,
>-      0, 0, 0, 0, 0, 0, 0, 0
>-  };
>-  static const std::vector<int16_t> kExpectedSamples(kInputSamples.begin(),
>-                                                     kInputSamples.end() - 9);
>-  RunTest(kInputSamples, kExpectedSamples, 8);
>-}
>-
>-TEST(BoundedWavFileWriterTest, DoubleEndSilence) {
>-  static const std::vector<int16_t> kInputSamples = {
>-      75, 1234, 243, -1231, -22222, 0, 0, 0,
>-      0, -1213, -13222, -7, -3525, 5787, 0, 0
>-  };
>-  static const std::vector<int16_t> kExpectedSamples(kInputSamples.begin(),
>-                                                     kInputSamples.end() - 2);
>-  RunTest(kInputSamples, kExpectedSamples, 8);
>-}
>-
>-TEST(BoundedWavFileWriterTest, DoubleSilence) {
>-  static const std::vector<int16_t> kInputSamples = {
>-      0, -1213, -13222, -7, -3525, 5787, 0, 0
>-  };
>-  static const std::vector<int16_t> kExpectedSamples(kInputSamples.begin() + 1,
>-                                                     kInputSamples.end() - 2);
>-  RunTest(kInputSamples, kExpectedSamples, 8);
>-}
>-
>-TEST(BoundedWavFileWriterTest, EndSilenceCutoff) {
>-  static const std::vector<int16_t> kInputSamples = {
>-      75, 1234, 243, -1231, -22222, 0, 1, 0,
>-      0, 0, 0
>-  };
>-  static const std::vector<int16_t> kExpectedSamples(kInputSamples.begin(),
>-                                                     kInputSamples.end() - 4);
>-  RunTest(kInputSamples, kExpectedSamples, 8);
>-}
>-
>-}  // namespace test
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_decoder.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_decoder.cc
>index fe8aef71677..eaf12d07ce2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_decoder.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_decoder.cc
>@@ -12,25 +12,32 @@
> 
> #include "api/video/i420_buffer.h"
> #include "rtc_base/timeutils.h"
>+#include "test/call_test.h"
> #include "test/gtest.h"
> 
> namespace webrtc {
> namespace test {
> 
>-FakeDecoder::FakeDecoder() : callback_(NULL) {}
>+FakeDecoder::FakeDecoder()
>+    : callback_(NULL),
>+      width_(CallTest::kDefaultWidth),
>+      height_(CallTest::kDefaultHeight) {}
> 
> int32_t FakeDecoder::InitDecode(const VideoCodec* config,
>                                 int32_t number_of_cores) {
>-  config_ = *config;
>   return WEBRTC_VIDEO_CODEC_OK;
> }
> 
> int32_t FakeDecoder::Decode(const EncodedImage& input,
>                             bool missing_frames,
>-                            const RTPFragmentationHeader* fragmentation,
>                             const CodecSpecificInfo* codec_specific_info,
>                             int64_t render_time_ms) {
>-  VideoFrame frame(I420Buffer::Create(config_.width, config_.height),
>+  if (input._encodedWidth > 0 && input._encodedHeight > 0) {
>+    width_ = input._encodedWidth;
>+    height_ = input._encodedHeight;
>+  }
>+
>+  VideoFrame frame(I420Buffer::Create(width_, height_),
>                    webrtc::kVideoRotation_0,
>                    render_time_ms * rtc::kNumMicrosecsPerMillisec);
>   frame.set_timestamp(input._timeStamp);
>@@ -58,7 +65,6 @@ const char* FakeDecoder::ImplementationName() const {
> 
> int32_t FakeH264Decoder::Decode(const EncodedImage& input,
>                                 bool missing_frames,
>-                                const RTPFragmentationHeader* fragmentation,
>                                 const CodecSpecificInfo* codec_specific_info,
>                                 int64_t render_time_ms) {
>   uint8_t value = 0;
>@@ -75,10 +81,7 @@ int32_t FakeH264Decoder::Decode(const EncodedImage& input,
>     }
>     ++value;
>   }
>-  return FakeDecoder::Decode(input,
>-                             missing_frames,
>-                             fragmentation,
>-                             codec_specific_info,
>+  return FakeDecoder::Decode(input, missing_frames, codec_specific_info,
>                              render_time_ms);
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_decoder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_decoder.h
>index 61dacf065f3..a90c861f1ac 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_decoder.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_decoder.h
>@@ -29,7 +29,6 @@ class FakeDecoder : public VideoDecoder {
> 
>   int32_t Decode(const EncodedImage& input,
>                  bool missing_frames,
>-                 const RTPFragmentationHeader* fragmentation,
>                  const CodecSpecificInfo* codec_specific_info,
>                  int64_t render_time_ms) override;
> 
>@@ -43,8 +42,9 @@ class FakeDecoder : public VideoDecoder {
>   static const char* kImplementationName;
> 
>  private:
>-  VideoCodec config_;
>   DecodedImageCallback* callback_;
>+  int width_;
>+  int height_;
> };
> 
> class FakeH264Decoder : public FakeDecoder {
>@@ -53,23 +53,10 @@ class FakeH264Decoder : public FakeDecoder {
> 
>   int32_t Decode(const EncodedImage& input,
>                  bool missing_frames,
>-                 const RTPFragmentationHeader* fragmentation,
>                  const CodecSpecificInfo* codec_specific_info,
>                  int64_t render_time_ms) override;
> };
> 
>-class FakeNullDecoder : public FakeDecoder {
>- public:
>-  virtual ~FakeNullDecoder() {}
>-
>-  int32_t Decode(const EncodedImage& input,
>-                 bool missing_frames,
>-                 const RTPFragmentationHeader* fragmentation,
>-                 const CodecSpecificInfo* codec_specific_info,
>-                 int64_t render_time_ms) override {
>-    return 0;
>-  }
>-};
> }  // namespace test
> }  // namespace webrtc
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_encoder.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_encoder.cc
>index a7dbbd9048e..12a77e3f9e4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_encoder.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_encoder.cc
>@@ -116,8 +116,8 @@ int32_t FakeEncoder::Encode(const VideoFrame& input_image,
>         (simulcast_streams[i].minBitrate * 1000) / framerate);
>     size_t max_stream_bits = static_cast<size_t>(
>         (simulcast_streams[i].maxBitrate * 1000) / framerate);
>-    size_t stream_bits = (bits_available > max_stream_bits) ? max_stream_bits :
>-        bits_available;
>+    size_t stream_bits =
>+        (bits_available > max_stream_bits) ? max_stream_bits : bits_available;
>     size_t stream_bytes = (stream_bits + 7) / 8;
>     if (keyframe) {
>       // The first frame is a key frame and should be larger.
>@@ -150,7 +150,7 @@ int32_t FakeEncoder::Encode(const VideoFrame& input_image,
>     encoded._encodedWidth = simulcast_streams[i].width;
>     encoded._encodedHeight = simulcast_streams[i].height;
>     encoded.rotation_ = input_image.rotation();
>-    encoded.content_type_ = (mode == kScreensharing)
>+    encoded.content_type_ = (mode == VideoCodecMode::kScreensharing)
>                                 ? VideoContentType::SCREENSHARE
>                                 : VideoContentType::UNSPECIFIED;
>     specifics.codec_name = ImplementationName();
>@@ -172,14 +172,17 @@ int32_t FakeEncoder::RegisterEncodeCompleteCallback(
>   return 0;
> }
> 
>-int32_t FakeEncoder::Release() { return 0; }
>+int32_t FakeEncoder::Release() {
>+  return 0;
>+}
> 
> int32_t FakeEncoder::SetChannelParameters(uint32_t packet_loss, int64_t rtt) {
>   return 0;
> }
> 
>-int32_t FakeEncoder::SetRateAllocation(const BitrateAllocation& rate_allocation,
>-                                       uint32_t framerate) {
>+int32_t FakeEncoder::SetRateAllocation(
>+    const VideoBitrateAllocation& rate_allocation,
>+    uint32_t framerate) {
>   rtc::CritScope cs(&crit_sect_);
>   target_bitrate_ = rate_allocation;
>   configured_input_framerate_ = framerate;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_encoder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_encoder.h
>index 06c0e0392de..071019abf39 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_encoder.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_encoder.h
>@@ -11,8 +11,8 @@
> #ifndef TEST_FAKE_ENCODER_H_
> #define TEST_FAKE_ENCODER_H_
> 
>-#include <vector>
> #include <memory>
>+#include <vector>
> 
> #include "api/video_codecs/video_encoder.h"
> #include "common_types.h"  // NOLINT(build/include)
>@@ -42,7 +42,7 @@ class FakeEncoder : public VideoEncoder {
>       EncodedImageCallback* callback) override;
>   int32_t Release() override;
>   int32_t SetChannelParameters(uint32_t packet_loss, int64_t rtt) override;
>-  int32_t SetRateAllocation(const BitrateAllocation& rate_allocation,
>+  int32_t SetRateAllocation(const VideoBitrateAllocation& rate_allocation,
>                             uint32_t framerate) override;
>   const char* ImplementationName() const override;
>   int GetConfiguredInputFramerate() const;
>@@ -53,7 +53,7 @@ class FakeEncoder : public VideoEncoder {
>   Clock* const clock_;
>   VideoCodec config_ RTC_GUARDED_BY(crit_sect_);
>   EncodedImageCallback* callback_ RTC_GUARDED_BY(crit_sect_);
>-  BitrateAllocation target_bitrate_ RTC_GUARDED_BY(crit_sect_);
>+  VideoBitrateAllocation target_bitrate_ RTC_GUARDED_BY(crit_sect_);
>   int configured_input_framerate_ RTC_GUARDED_BY(crit_sect_);
>   int max_target_bitrate_kbps_ RTC_GUARDED_BY(crit_sect_);
>   bool pending_keyframe_ RTC_GUARDED_BY(crit_sect_);
>@@ -95,7 +95,7 @@ class DelayedEncoder : public test::FakeEncoder {
>                  const std::vector<FrameType>* frame_types) override;
> 
>  private:
>-  int delay_ms_ RTC_ACCESS_ON(sequence_checker_);
>+  int delay_ms_ RTC_GUARDED_BY(sequence_checker_);
>   rtc::SequencedTaskChecker sequence_checker_;
> };
> 
>@@ -125,9 +125,9 @@ class MultithreadedFakeH264Encoder : public test::FakeH264Encoder {
>  protected:
>   class EncodeTask;
> 
>-  int current_queue_ RTC_ACCESS_ON(sequence_checker_);
>-  std::unique_ptr<rtc::TaskQueue> queue1_ RTC_ACCESS_ON(sequence_checker_);
>-  std::unique_ptr<rtc::TaskQueue> queue2_ RTC_ACCESS_ON(sequence_checker_);
>+  int current_queue_ RTC_GUARDED_BY(sequence_checker_);
>+  std::unique_ptr<rtc::TaskQueue> queue1_ RTC_GUARDED_BY(sequence_checker_);
>+  std::unique_ptr<rtc::TaskQueue> queue2_ RTC_GUARDED_BY(sequence_checker_);
>   rtc::SequencedTaskChecker sequence_checker_;
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_network_pipe.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_network_pipe.cc
>deleted file mode 100644
>index 6678f1c3744..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_network_pipe.cc
>+++ /dev/null
>@@ -1,255 +0,0 @@
>-/*
>- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "test/fake_network_pipe.h"
>-
>-#include <assert.h>
>-#include <math.h>
>-#include <string.h>
>-
>-#include <algorithm>
>-#include <cmath>
>-
>-#include "call/call.h"
>-#include "modules/rtp_rtcp/include/rtp_header_parser.h"
>-#include "rtc_base/logging.h"
>-#include "system_wrappers/include/clock.h"
>-
>-namespace webrtc {
>-
>-namespace {
>-constexpr int64_t kDefaultProcessIntervalMs = 5;
>-}
>-
>-DemuxerImpl::DemuxerImpl(const std::map<uint8_t, MediaType>& payload_type_map)
>-    : packet_receiver_(nullptr), payload_type_map_(payload_type_map) {}
>-
>-void DemuxerImpl::SetReceiver(PacketReceiver* receiver) {
>-  packet_receiver_ = receiver;
>-}
>-
>-void DemuxerImpl::DeliverPacket(const NetworkPacket* packet,
>-                                const PacketTime& packet_time) {
>-  // No packet receiver means that this demuxer will terminate the flow of
>-  // packets.
>-  if (!packet_receiver_)
>-    return;
>-  const uint8_t* const packet_data = packet->data();
>-  const size_t packet_length = packet->data_length();
>-  MediaType media_type = MediaType::ANY;
>-  if (!RtpHeaderParser::IsRtcp(packet_data, packet_length)) {
>-    RTC_CHECK_GE(packet_length, 2);
>-    const uint8_t payload_type = packet_data[1] & 0x7f;
>-    std::map<uint8_t, MediaType>::const_iterator it =
>-        payload_type_map_.find(payload_type);
>-    RTC_CHECK(it != payload_type_map_.end())
>-        << "payload type " << static_cast<int>(payload_type) << " unknown.";
>-    media_type = it->second;
>-  }
>-  packet_receiver_->DeliverPacket(media_type, packet_data, packet_length,
>-                                  packet_time);
>-}
>-
>-FakeNetworkPipe::FakeNetworkPipe(Clock* clock,
>-                                 const FakeNetworkPipe::Config& config,
>-                                 std::unique_ptr<Demuxer> demuxer)
>-    : FakeNetworkPipe(clock, config, std::move(demuxer), 1) {}
>-
>-FakeNetworkPipe::FakeNetworkPipe(Clock* clock,
>-                                 const FakeNetworkPipe::Config& config,
>-                                 std::unique_ptr<Demuxer> demuxer,
>-                                 uint64_t seed)
>-    : clock_(clock),
>-      demuxer_(std::move(demuxer)),
>-      random_(seed),
>-      config_(),
>-      dropped_packets_(0),
>-      sent_packets_(0),
>-      total_packet_delay_(0),
>-      bursting_(false),
>-      next_process_time_(clock_->TimeInMilliseconds()),
>-      last_log_time_(clock_->TimeInMilliseconds()) {
>-  SetConfig(config);
>-}
>-
>-FakeNetworkPipe::~FakeNetworkPipe() {
>-  while (!capacity_link_.empty()) {
>-    delete capacity_link_.front();
>-    capacity_link_.pop();
>-  }
>-  while (!delay_link_.empty()) {
>-    delete *delay_link_.begin();
>-    delay_link_.erase(delay_link_.begin());
>-  }
>-}
>-
>-void FakeNetworkPipe::SetReceiver(PacketReceiver* receiver) {
>-  RTC_CHECK(demuxer_);
>-  demuxer_->SetReceiver(receiver);
>-}
>-
>-void FakeNetworkPipe::SetConfig(const FakeNetworkPipe::Config& config) {
>-  rtc::CritScope crit(&lock_);
>-  config_ = config;  // Shallow copy of the struct.
>-  double prob_loss = config.loss_percent / 100.0;
>-  if (config_.avg_burst_loss_length == -1) {
>-    // Uniform loss
>-    prob_loss_bursting_ = prob_loss;
>-    prob_start_bursting_ = prob_loss;
>-  } else {
>-    // Lose packets according to a gilbert-elliot model.
>-    int avg_burst_loss_length = config.avg_burst_loss_length;
>-    int min_avg_burst_loss_length = std::ceil(prob_loss / (1 - prob_loss));
>-
>-    RTC_CHECK_GT(avg_burst_loss_length, min_avg_burst_loss_length)
>-        << "For a total packet loss of " << config.loss_percent << "%% then"
>-        << " avg_burst_loss_length must be " << min_avg_burst_loss_length + 1
>-        << " or higher.";
>-
>-    prob_loss_bursting_ = (1.0 - 1.0 / avg_burst_loss_length);
>-    prob_start_bursting_ = prob_loss / (1 - prob_loss) / avg_burst_loss_length;
>-  }
>-}
>-
>-void FakeNetworkPipe::SendPacket(const uint8_t* data, size_t data_length) {
>-  RTC_CHECK(demuxer_);
>-  rtc::CritScope crit(&lock_);
>-  if (config_.queue_length_packets > 0 &&
>-      capacity_link_.size() >= config_.queue_length_packets) {
>-    // Too many packet on the link, drop this one.
>-    ++dropped_packets_;
>-    return;
>-  }
>-
>-  int64_t time_now = clock_->TimeInMilliseconds();
>-
>-  // Delay introduced by the link capacity.
>-  int64_t capacity_delay_ms = 0;
>-  if (config_.link_capacity_kbps > 0) {
>-    const int bytes_per_millisecond = config_.link_capacity_kbps / 8;
>-    // To round to the closest millisecond we add half a milliseconds worth of
>-    // bytes to the delay calculation.
>-    capacity_delay_ms = (data_length + capacity_delay_error_bytes_ +
>-                         bytes_per_millisecond / 2) /
>-                        bytes_per_millisecond;
>-    capacity_delay_error_bytes_ +=
>-        data_length - capacity_delay_ms * bytes_per_millisecond;
>-  }
>-  int64_t network_start_time = time_now;
>-
>-  // Check if there already are packets on the link and change network start
>-  // time forward if there is.
>-  if (!capacity_link_.empty() &&
>-      network_start_time < capacity_link_.back()->arrival_time())
>-    network_start_time = capacity_link_.back()->arrival_time();
>-
>-  int64_t arrival_time = network_start_time + capacity_delay_ms;
>-  NetworkPacket* packet = new NetworkPacket(data, data_length, time_now,
>-                                            arrival_time);
>-  capacity_link_.push(packet);
>-}
>-
>-float FakeNetworkPipe::PercentageLoss() {
>-  rtc::CritScope crit(&lock_);
>-  if (sent_packets_ == 0)
>-    return 0;
>-
>-  return static_cast<float>(dropped_packets_) /
>-      (sent_packets_ + dropped_packets_);
>-}
>-
>-int FakeNetworkPipe::AverageDelay() {
>-  rtc::CritScope crit(&lock_);
>-  if (sent_packets_ == 0)
>-    return 0;
>-
>-  return static_cast<int>(total_packet_delay_ /
>-                          static_cast<int64_t>(sent_packets_));
>-}
>-
>-void FakeNetworkPipe::Process() {
>-  int64_t time_now = clock_->TimeInMilliseconds();
>-  std::queue<NetworkPacket*> packets_to_deliver;
>-  {
>-    rtc::CritScope crit(&lock_);
>-    if (time_now - last_log_time_ > 5000) {
>-      int64_t queueing_delay_ms = 0;
>-      if (!capacity_link_.empty()) {
>-        queueing_delay_ms = time_now - capacity_link_.front()->send_time();
>-      }
>-      RTC_LOG(LS_INFO) << "Network queue: " << queueing_delay_ms << " ms.";
>-      last_log_time_ = time_now;
>-    }
>-    // Check the capacity link first.
>-    while (!capacity_link_.empty() &&
>-           time_now >= capacity_link_.front()->arrival_time()) {
>-      // Time to get this packet.
>-      NetworkPacket* packet = capacity_link_.front();
>-      capacity_link_.pop();
>-
>-      // Drop packets at an average rate of |config_.loss_percent| with
>-      // and average loss burst length of |config_.avg_burst_loss_length|.
>-      if ((bursting_ && random_.Rand<double>() < prob_loss_bursting_) ||
>-          (!bursting_ && random_.Rand<double>() < prob_start_bursting_)) {
>-        bursting_ = true;
>-        delete packet;
>-        continue;
>-      } else {
>-        bursting_ = false;
>-      }
>-
>-      int arrival_time_jitter = random_.Gaussian(
>-          config_.queue_delay_ms, config_.delay_standard_deviation_ms);
>-
>-      // If reordering is not allowed then adjust arrival_time_jitter
>-      // to make sure all packets are sent in order.
>-      if (!config_.allow_reordering && !delay_link_.empty() &&
>-          packet->arrival_time() + arrival_time_jitter <
>-              (*delay_link_.rbegin())->arrival_time()) {
>-        arrival_time_jitter =
>-            (*delay_link_.rbegin())->arrival_time() - packet->arrival_time();
>-      }
>-      packet->IncrementArrivalTime(arrival_time_jitter);
>-      delay_link_.insert(packet);
>-    }
>-
>-    // Check the extra delay queue.
>-    while (!delay_link_.empty() &&
>-           time_now >= (*delay_link_.begin())->arrival_time()) {
>-      // Deliver this packet.
>-      NetworkPacket* packet = *delay_link_.begin();
>-      packets_to_deliver.push(packet);
>-      delay_link_.erase(delay_link_.begin());
>-      // |time_now| might be later than when the packet should have arrived, due
>-      // to NetworkProcess being called too late. For stats, use the time it
>-      // should have been on the link.
>-      total_packet_delay_ += packet->arrival_time() - packet->send_time();
>-    }
>-    sent_packets_ += packets_to_deliver.size();
>-  }
>-  while (!packets_to_deliver.empty()) {
>-    NetworkPacket* packet = packets_to_deliver.front();
>-    packets_to_deliver.pop();
>-    demuxer_->DeliverPacket(packet, PacketTime());
>-    delete packet;
>-  }
>-
>-  next_process_time_ = !delay_link_.empty()
>-                           ? (*delay_link_.begin())->arrival_time()
>-                           : time_now + kDefaultProcessIntervalMs;
>-}
>-
>-int64_t FakeNetworkPipe::TimeUntilNextProcess() const {
>-  rtc::CritScope crit(&lock_);
>-  return std::max<int64_t>(next_process_time_ - clock_->TimeInMilliseconds(),
>-                           0);
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_network_pipe.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_network_pipe.h
>deleted file mode 100644
>index 1cfaa382f49..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_network_pipe.h
>+++ /dev/null
>@@ -1,185 +0,0 @@
>-/*
>- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef TEST_FAKE_NETWORK_PIPE_H_
>-#define TEST_FAKE_NETWORK_PIPE_H_
>-
>-#include <string.h>
>-#include <map>
>-#include <memory>
>-#include <queue>
>-#include <set>
>-
>-#include "common_types.h"  // NOLINT(build/include)
>-#include "rtc_base/constructormagic.h"
>-#include "rtc_base/criticalsection.h"
>-#include "rtc_base/random.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>-
>-namespace webrtc {
>-
>-class Clock;
>-class PacketReceiver;
>-enum class MediaType;
>-
>-class NetworkPacket {
>- public:
>-  NetworkPacket(const uint8_t* data,
>-                size_t length,
>-                int64_t send_time,
>-                int64_t arrival_time)
>-      : data_(new uint8_t[length]),
>-        data_length_(length),
>-        send_time_(send_time),
>-        arrival_time_(arrival_time) {
>-    memcpy(data_.get(), data, length);
>-  }
>-
>-  uint8_t* data() const { return data_.get(); }
>-  size_t data_length() const { return data_length_; }
>-  int64_t send_time() const { return send_time_; }
>-  int64_t arrival_time() const { return arrival_time_; }
>-  void IncrementArrivalTime(int64_t extra_delay) {
>-    arrival_time_ += extra_delay;
>-  }
>-
>- private:
>-  // The packet data.
>-  std::unique_ptr<uint8_t[]> data_;
>-  // Length of data_.
>-  size_t data_length_;
>-  // The time the packet was sent out on the network.
>-  const int64_t send_time_;
>-  // The time the packet should arrive at the receiver.
>-  int64_t arrival_time_;
>-};
>-
>-class Demuxer {
>- public:
>-  virtual ~Demuxer() = default;
>-  virtual void SetReceiver(PacketReceiver* receiver) = 0;
>-  virtual void DeliverPacket(const NetworkPacket* packet,
>-                             const PacketTime& packet_time) = 0;
>-};
>-
>-class DemuxerImpl final : public Demuxer {
>- public:
>-  explicit DemuxerImpl(const std::map<uint8_t, MediaType>& payload_type_map);
>-
>-  void SetReceiver(PacketReceiver* receiver) override;
>-  void DeliverPacket(const NetworkPacket* packet,
>-                     const PacketTime& packet_time) override;
>-
>- private:
>-  PacketReceiver* packet_receiver_;
>-  const std::map<uint8_t, MediaType> payload_type_map_;
>-  RTC_DISALLOW_COPY_AND_ASSIGN(DemuxerImpl);
>-};
>-
>-// Class faking a network link. This is a simple and naive solution just faking
>-// capacity and adding an extra transport delay in addition to the capacity
>-// introduced delay.
>-
>-class FakeNetworkPipe {
>- public:
>-  struct Config {
>-    Config() {}
>-    // Queue length in number of packets.
>-    size_t queue_length_packets = 0;
>-    // Delay in addition to capacity induced delay.
>-    int queue_delay_ms = 0;
>-    // Standard deviation of the extra delay.
>-    int delay_standard_deviation_ms = 0;
>-    // Link capacity in kbps.
>-    int link_capacity_kbps = 0;
>-    // Random packet loss.
>-    int loss_percent = 0;
>-    // If packets are allowed to be reordered.
>-    bool allow_reordering = false;
>-    // The average length of a burst of lost packets.
>-    int avg_burst_loss_length = -1;
>-  };
>-
>-  FakeNetworkPipe(Clock* clock,
>-                  const FakeNetworkPipe::Config& config,
>-                  std::unique_ptr<Demuxer> demuxer);
>-  FakeNetworkPipe(Clock* clock,
>-                  const FakeNetworkPipe::Config& config,
>-                  std::unique_ptr<Demuxer> demuxer,
>-                  uint64_t seed);
>-  virtual ~FakeNetworkPipe();
>-
>-  // Sets a new configuration. This won't affect packets already in the pipe.
>-  void SetConfig(const FakeNetworkPipe::Config& config);
>-
>-  // Sends a new packet to the link.
>-  virtual void SendPacket(const uint8_t* packet, size_t packet_length);
>-
>-  // Must not be called in parallel with SendPacket or Process.
>-  void SetReceiver(PacketReceiver* receiver);
>-
>-  // Processes the network queues and trigger PacketReceiver::IncomingPacket for
>-  // packets ready to be delivered.
>-  virtual void Process();
>-  int64_t TimeUntilNextProcess() const;
>-
>-  // Get statistics.
>-  float PercentageLoss();
>-  int AverageDelay();
>-  size_t dropped_packets() { return dropped_packets_; }
>-  size_t sent_packets() { return sent_packets_; }
>-
>- protected:
>-  Clock* const clock_;
>-  rtc::CriticalSection lock_;
>-  const std::unique_ptr<Demuxer> demuxer_;
>-  std::queue<NetworkPacket*> capacity_link_;
>-  Random random_;
>-
>-  // Since we need to access both the packet with the earliest and latest
>-  // arrival time we need to use a multiset to keep all packets sorted,
>-  // hence, we cannot use a priority queue.
>-  struct PacketArrivalTimeComparator {
>-    bool operator()(const NetworkPacket* p1, const NetworkPacket* p2) {
>-      return p1->arrival_time() < p2->arrival_time();
>-    }
>-  };
>-  std::multiset<NetworkPacket*, PacketArrivalTimeComparator> delay_link_;
>-
>-  // Link configuration.
>-  Config config_;
>-
>-  // Statistics.
>-  size_t dropped_packets_;
>-  size_t sent_packets_;
>-  int64_t total_packet_delay_;
>-
>-  // Are we currently dropping a burst of packets?
>-  bool bursting_;
>-
>-  // The probability to drop the packet if we are currently dropping a
>-  // burst of packet
>-  double prob_loss_bursting_;
>-
>-  // The probability to drop a burst of packets.
>-  double prob_start_bursting_;
>-
>-  int64_t next_process_time_;
>-
>-  int64_t last_log_time_;
>-
>-  int64_t capacity_delay_error_bytes_ = 0;
>-
>-  RTC_DISALLOW_COPY_AND_ASSIGN(FakeNetworkPipe);
>-};
>-
>-}  // namespace webrtc
>-
>-#endif  // TEST_FAKE_NETWORK_PIPE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_texture_frame.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_texture_frame.cc
>index 4c4ea94fd29..c6f186e23bc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_texture_frame.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_texture_frame.cc
>@@ -21,5 +21,24 @@ VideoFrame FakeNativeBuffer::CreateFrame(int width,
>   return VideoFrame(new rtc::RefCountedObject<FakeNativeBuffer>(width, height),
>                     timestamp, render_time_ms, rotation);
> }
>+
>+VideoFrameBuffer::Type FakeNativeBuffer::type() const {
>+  return Type::kNative;
>+}
>+
>+int FakeNativeBuffer::width() const {
>+  return width_;
>+}
>+
>+int FakeNativeBuffer::height() const {
>+  return height_;
>+}
>+
>+rtc::scoped_refptr<I420BufferInterface> FakeNativeBuffer::ToI420() {
>+  rtc::scoped_refptr<I420Buffer> buffer = I420Buffer::Create(width_, height_);
>+  I420Buffer::SetBlack(buffer);
>+  return buffer;
>+}
>+
> }  // namespace test
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_texture_frame.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_texture_frame.h
>index 17261ff8aea..1b25112e011 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_texture_frame.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_texture_frame.h
>@@ -28,16 +28,12 @@ class FakeNativeBuffer : public VideoFrameBuffer {
> 
>   FakeNativeBuffer(int width, int height) : width_(width), height_(height) {}
> 
>-  Type type() const override { return Type::kNative; }
>-  int width() const override { return width_; }
>-  int height() const override { return height_; }
>+  Type type() const override;
>+  int width() const override;
>+  int height() const override;
> 
>  private:
>-  rtc::scoped_refptr<I420BufferInterface> ToI420() override {
>-    rtc::scoped_refptr<I420Buffer> buffer = I420Buffer::Create(width_, height_);
>-    I420Buffer::SetBlack(buffer);
>-    return buffer;
>-  }
>+  rtc::scoped_refptr<I420BufferInterface> ToI420() override;
> 
>   const int width_;
>   const int height_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_videorenderer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_videorenderer.h
>index bc9c1db710a..a2c953ff77c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_videorenderer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fake_videorenderer.h
>@@ -12,7 +12,7 @@
> #define TEST_FAKE_VIDEORENDERER_H_
> 
> #include "api/video/video_frame.h"
>-#include "media/base/videosinkinterface.h"
>+#include "api/video/video_sink_interface.h"
> 
> namespace webrtc {
> namespace test {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/field_trial.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/field_trial.cc
>index 00e5b69041b..c4d8b3c3379 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/field_trial.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/field_trial.cc
>@@ -26,9 +26,8 @@ bool field_trials_initiated_ = false;
> }  // namespace
> 
> namespace test {
>-// Note: this code is copied from src/base/metrics/field_trial.cc since the aim
>-// is to mimic chromium --force-fieldtrials.
>-void InitFieldTrialsFromString(const std::string& trials_string) {
>+
>+void ValidateFieldTrialsStringOrDie(const std::string& trials_string) {
>   static const char kPersistentStringSeparator = '/';
> 
>   // Catch an error if this is called more than once.
>@@ -44,8 +43,8 @@ void InitFieldTrialsFromString(const std::string& trials_string) {
>     size_t name_end = trials_string.find(kPersistentStringSeparator, next_item);
>     if (name_end == trials_string.npos || next_item == name_end)
>       break;
>-    size_t group_name_end = trials_string.find(kPersistentStringSeparator,
>-                                               name_end + 1);
>+    size_t group_name_end =
>+        trials_string.find(kPersistentStringSeparator, name_end + 1);
>     if (group_name_end == trials_string.npos || name_end + 1 == group_name_end)
>       break;
>     std::string name(trials_string, next_item, name_end - next_item);
>@@ -63,7 +62,7 @@ void InitFieldTrialsFromString(const std::string& trials_string) {
> 
>     // Successfully parsed all field trials from the string.
>     if (next_item == trials_string.length()) {
>-      webrtc::field_trial::InitFieldTrialsFromString(trials_string.c_str());
>+      // webrtc::field_trial::InitFieldTrialsFromString(trials_string.c_str());
>       return;
>     }
>   }
>@@ -75,11 +74,12 @@ void InitFieldTrialsFromString(const std::string& trials_string) {
> }
> 
> ScopedFieldTrials::ScopedFieldTrials(const std::string& config)
>-  : previous_field_trials_(webrtc::field_trial::GetFieldTrialString()) {
>+    : previous_field_trials_(webrtc::field_trial::GetFieldTrialString()) {
>   assert(field_trials_initiated_);
>   field_trials_initiated_ = false;
>   current_field_trials_ = config;
>-  InitFieldTrialsFromString(current_field_trials_);
>+  ValidateFieldTrialsStringOrDie(current_field_trials_);
>+  webrtc::field_trial::InitFieldTrialsFromString(current_field_trials_.c_str());
> }
> 
> ScopedFieldTrials::~ScopedFieldTrials() {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/field_trial.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/field_trial.h
>index 99ace4e8648..2d8f37561a2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/field_trial.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/field_trial.h
>@@ -11,8 +11,8 @@
> #ifndef TEST_FIELD_TRIAL_H_
> #define TEST_FIELD_TRIAL_H_
> 
>-#include <string>
> #include <map>
>+#include <string>
> 
> namespace webrtc {
> namespace test {
>@@ -30,7 +30,7 @@ namespace test {
> //
> // Note: This method crashes with an error message if an invalid config is
> // passed to it. That can be used to find out if a binary is parsing the flags.
>-void InitFieldTrialsFromString(const std::string& config);
>+void ValidateFieldTrialsStringOrDie(const std::string& config);
> 
> // This class is used to override field-trial configs within specific tests.
> // After this class goes out of scope previous field trials will be restored.
>@@ -38,6 +38,7 @@ class ScopedFieldTrials {
>  public:
>   explicit ScopedFieldTrials(const std::string& config);
>   ~ScopedFieldTrials();
>+
>  private:
>   std::string current_field_trials_;
>   const char* previous_field_trials_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/frame_generator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/frame_generator.cc
>index e5d99503015..45d5f2f76ed 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/frame_generator.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/frame_generator.cc
>@@ -15,6 +15,7 @@
> 
> #include <memory>
> 
>+#include "api/video/i010_buffer.h"
> #include "api/video/i420_buffer.h"
> #include "common_video/include/video_frame_buffer.h"
> #include "common_video/libyuv/include/webrtc_libyuv.h"
>@@ -28,12 +29,17 @@ namespace webrtc {
> namespace test {
> namespace {
> 
>+// Helper method for keeping a reference to passed pointers.
>+void KeepBufferRefs(rtc::scoped_refptr<webrtc::VideoFrameBuffer>,
>+                    rtc::scoped_refptr<webrtc::VideoFrameBuffer>) {}
>+
> // SquareGenerator is a FrameGenerator that draws a given amount of randomly
> // sized and colored squares. Between each new generated frame, the squares
> // are moved slightly towards the lower right corner.
> class SquareGenerator : public FrameGenerator {
>  public:
>-  SquareGenerator(int width, int height, int num_squares) {
>+  SquareGenerator(int width, int height, OutputType type, int num_squares)
>+      : type_(type) {
>     ChangeResolution(width, height);
>     for (int i = 0; i < num_squares; ++i) {
>       squares_.emplace_back(new Square(width, height, i + 1));
>@@ -48,21 +54,50 @@ class SquareGenerator : public FrameGenerator {
>     RTC_CHECK(height_ > 0);
>   }
> 
>-  VideoFrame* NextFrame() override {
>-    rtc::CritScope lock(&crit_);
>-
>-    rtc::scoped_refptr<I420Buffer> buffer(I420Buffer::Create(width_, height_));
>-
>-    memset(buffer->MutableDataY(), 127, height_ * buffer->StrideY());
>+  rtc::scoped_refptr<I420Buffer> CreateI420Buffer(int width, int height) {
>+    rtc::scoped_refptr<I420Buffer> buffer(I420Buffer::Create(width, height));
>+    memset(buffer->MutableDataY(), 127, height * buffer->StrideY());
>     memset(buffer->MutableDataU(), 127,
>            buffer->ChromaHeight() * buffer->StrideU());
>     memset(buffer->MutableDataV(), 127,
>            buffer->ChromaHeight() * buffer->StrideV());
>+    return buffer;
>+  }
>+
>+  VideoFrame* NextFrame() override {
>+    rtc::CritScope lock(&crit_);
>+
>+    rtc::scoped_refptr<VideoFrameBuffer> buffer = nullptr;
>+    switch (type_) {
>+      case OutputType::I420:
>+      case OutputType::I010: {
>+        buffer = CreateI420Buffer(width_, height_);
>+        break;
>+      }
>+      case OutputType::I420A: {
>+        rtc::scoped_refptr<I420Buffer> yuv_buffer =
>+            CreateI420Buffer(width_, height_);
>+        rtc::scoped_refptr<I420Buffer> axx_buffer =
>+            CreateI420Buffer(width_, height_);
>+        buffer = WrapI420ABuffer(
>+            yuv_buffer->width(), yuv_buffer->height(), yuv_buffer->DataY(),
>+            yuv_buffer->StrideY(), yuv_buffer->DataU(), yuv_buffer->StrideU(),
>+            yuv_buffer->DataV(), yuv_buffer->StrideV(), axx_buffer->DataY(),
>+            axx_buffer->StrideY(),
>+            rtc::Bind(&KeepBufferRefs, yuv_buffer, axx_buffer));
>+        break;
>+      }
>+    }
> 
>     for (const auto& square : squares_)
>       square->Draw(buffer);
> 
>-    frame_.reset(new VideoFrame(buffer, 0, 0, webrtc::kVideoRotation_0));
>+    if (type_ == OutputType::I010) {
>+      buffer = I010Buffer::Copy(*buffer->ToI420());
>+    }
>+
>+    frame_.reset(
>+        new VideoFrame(buffer, webrtc::kVideoRotation_0, 0 /* timestamp_us */));
>     return frame_.get();
>   }
> 
>@@ -76,25 +111,41 @@ class SquareGenerator : public FrameGenerator {
>           length_(random_generator_.Rand(1, width > 4 ? width / 4 : 1)),
>           yuv_y_(random_generator_.Rand(0, 255)),
>           yuv_u_(random_generator_.Rand(0, 255)),
>-          yuv_v_(random_generator_.Rand(0, 255)) {}
>+          yuv_v_(random_generator_.Rand(0, 255)),
>+          yuv_a_(random_generator_.Rand(0, 255)) {}
> 
>-    void Draw(const rtc::scoped_refptr<I420Buffer>& buffer) {
>+    void Draw(const rtc::scoped_refptr<VideoFrameBuffer>& frame_buffer) {
>+      RTC_DCHECK(frame_buffer->type() == VideoFrameBuffer::Type::kI420 ||
>+                 frame_buffer->type() == VideoFrameBuffer::Type::kI420A);
>+      rtc::scoped_refptr<I420BufferInterface> buffer = frame_buffer->ToI420();
>       x_ = (x_ + random_generator_.Rand(0, 4)) % (buffer->width() - length_);
>       y_ = (y_ + random_generator_.Rand(0, 4)) % (buffer->height() - length_);
>-        for (int y = y_; y < y_ + length_; ++y) {
>-          uint8_t* pos_y =
>-              (buffer->MutableDataY() + x_ + y * buffer->StrideY());
>-          memset(pos_y, yuv_y_, length_);
>-        }
>-
>-        for (int y = y_; y < y_ + length_; y = y + 2) {
>-          uint8_t* pos_u =
>-              (buffer->MutableDataU() + x_ / 2 + y / 2 * buffer->StrideU());
>-          memset(pos_u, yuv_u_, length_ / 2);
>-          uint8_t* pos_v =
>-              (buffer->MutableDataV() + x_ / 2 + y / 2 * buffer->StrideV());
>-          memset(pos_v, yuv_v_, length_ / 2);
>-        }
>+      for (int y = y_; y < y_ + length_; ++y) {
>+        uint8_t* pos_y = (const_cast<uint8_t*>(buffer->DataY()) + x_ +
>+                          y * buffer->StrideY());
>+        memset(pos_y, yuv_y_, length_);
>+      }
>+
>+      for (int y = y_; y < y_ + length_; y = y + 2) {
>+        uint8_t* pos_u = (const_cast<uint8_t*>(buffer->DataU()) + x_ / 2 +
>+                          y / 2 * buffer->StrideU());
>+        memset(pos_u, yuv_u_, length_ / 2);
>+        uint8_t* pos_v = (const_cast<uint8_t*>(buffer->DataV()) + x_ / 2 +
>+                          y / 2 * buffer->StrideV());
>+        memset(pos_v, yuv_v_, length_ / 2);
>+      }
>+
>+      if (frame_buffer->type() == VideoFrameBuffer::Type::kI420)
>+        return;
>+
>+      // Optionally draw on alpha plane if given.
>+      const webrtc::I420ABufferInterface* yuva_buffer =
>+          frame_buffer->GetI420A();
>+      for (int y = y_; y < y_ + length_; ++y) {
>+        uint8_t* pos_y = (const_cast<uint8_t*>(yuva_buffer->DataA()) + x_ +
>+                          y * yuva_buffer->StrideA());
>+        memset(pos_y, yuv_a_, length_);
>+      }
>     }
> 
>    private:
>@@ -105,9 +156,11 @@ class SquareGenerator : public FrameGenerator {
>     const uint8_t yuv_y_;
>     const uint8_t yuv_u_;
>     const uint8_t yuv_v_;
>+    const uint8_t yuv_a_;
>   };
> 
>   rtc::CriticalSection crit_;
>+  const OutputType type_;
>   int width_ RTC_GUARDED_BY(&crit_);
>   int height_ RTC_GUARDED_BY(&crit_);
>   std::vector<std::unique_ptr<Square>> squares_ RTC_GUARDED_BY(&crit_);
>@@ -146,8 +199,8 @@ class YuvFileGenerator : public FrameGenerator {
>     if (++current_display_count_ >= frame_display_count_)
>       current_display_count_ = 0;
> 
>-    temp_frame_.reset(
>-        new VideoFrame(last_read_buffer_, 0, 0, webrtc::kVideoRotation_0));
>+    temp_frame_.reset(new VideoFrame(
>+        last_read_buffer_, webrtc::kVideoRotation_0, 0 /* timestamp_us */));
>     return temp_frame_.get();
>   }
> 
>@@ -203,8 +256,8 @@ class SlideGenerator : public FrameGenerator {
>     if (++current_display_count_ >= frame_display_count_)
>       current_display_count_ = 0;
> 
>-    frame_.reset(
>-        new VideoFrame(buffer_, 0, 0, webrtc::kVideoRotation_0));
>+    frame_.reset(new VideoFrame(buffer_, webrtc::kVideoRotation_0,
>+                                0 /* timestamp_us */));
>     return frame_.get();
>   }
> 
>@@ -336,13 +389,13 @@ class ScrollingImageFrameGenerator : public FrameGenerator {
>     int offset_v = (i420_buffer->StrideV() * (pixels_scrolled_y / 2)) +
>                    (pixels_scrolled_x / 2);
> 
>-    current_frame_ = rtc::Optional<webrtc::VideoFrame>(webrtc::VideoFrame(
>+    current_frame_ = webrtc::VideoFrame(
>         new rtc::RefCountedObject<webrtc::WrappedI420Buffer>(
>             target_width_, target_height_, &i420_buffer->DataY()[offset_y],
>             i420_buffer->StrideY(), &i420_buffer->DataU()[offset_u],
>             i420_buffer->StrideU(), &i420_buffer->DataV()[offset_v],
>             i420_buffer->StrideV(), KeepRefUntilDone(i420_buffer)),
>-        kVideoRotation_0, 0));
>+        kVideoRotation_0, 0);
>   }
> 
>   Clock* const clock_;
>@@ -355,7 +408,7 @@ class ScrollingImageFrameGenerator : public FrameGenerator {
> 
>   size_t current_frame_num_;
>   VideoFrame* current_source_frame_;
>-  rtc::Optional<VideoFrame> current_frame_;
>+  absl::optional<VideoFrame> current_frame_;
>   YuvFileGenerator file_generator_;
> };
> 
>@@ -396,15 +449,12 @@ bool FrameForwarder::has_sinks() const {
> 
> std::unique_ptr<FrameGenerator> FrameGenerator::CreateSquareGenerator(
>     int width,
>-    int height) {
>-  return std::unique_ptr<FrameGenerator>(
>-      new SquareGenerator(width, height, 10));
>-}
>-
>-std::unique_ptr<FrameGenerator>
>-FrameGenerator::CreateSquareGenerator(int width, int height, int num_squares) {
>+    int height,
>+    absl::optional<OutputType> type,
>+    absl::optional<int> num_squares) {
>   return std::unique_ptr<FrameGenerator>(
>-      new SquareGenerator(width, height, num_squares));
>+      new SquareGenerator(width, height, type.value_or(OutputType::I420),
>+                          num_squares.value_or(10)));
> }
> 
> std::unique_ptr<FrameGenerator> FrameGenerator::CreateSlideGenerator(
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/frame_generator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/frame_generator.h
>index d37166f2fa6..a92c3b4b5aa 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/frame_generator.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/frame_generator.h
>@@ -15,9 +15,8 @@
> #include <vector>
> 
> #include "api/video/video_frame.h"
>-#include "media/base/videosourceinterface.h"
>+#include "api/video/video_source_interface.h"
> #include "rtc_base/criticalsection.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> class Clock;
>@@ -58,13 +57,17 @@ class FrameGenerator {
>     RTC_NOTREACHED();
>   }
> 
>+  enum class OutputType { I420, I420A, I010 };
>+
>   // Creates a frame generator that produces frames with small squares that
>   // move randomly towards the lower right corner.
>-  static std::unique_ptr<FrameGenerator> CreateSquareGenerator(int width,
>-                                                               int height);
>-  static std::unique_ptr<FrameGenerator> CreateSquareGenerator(int width,
>-                                                               int height,
>-                                                               int num_squares);
>+  // |type| has the default value OutputType::I420. |num_squares| has the
>+  // default value 10.
>+  static std::unique_ptr<FrameGenerator> CreateSquareGenerator(
>+      int width,
>+      int height,
>+      absl::optional<OutputType> type,
>+      absl::optional<int> num_squares);
> 
>   // Creates a frame generator that repeatedly plays a set of yuv files.
>   // The frame_repeat_count determines how many times each frame is shown,
>@@ -95,8 +98,8 @@ class FrameGenerator {
> 
>   // Creates a frame generator that produces randomly generated slides.
>   // frame_repeat_count determines how many times each slide is shown.
>-  static std::unique_ptr<FrameGenerator> CreateSlideGenerator(
>-      int width, int height, int frame_repeat_count);
>+  static std::unique_ptr<FrameGenerator>
>+  CreateSlideGenerator(int width, int height, int frame_repeat_count);
> };
> }  // namespace test
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/frame_generator_capturer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/frame_generator_capturer.cc
>index a8983b7ba20..184fdd76ef0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/frame_generator_capturer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/frame_generator_capturer.cc
>@@ -13,14 +13,13 @@
> #include <utility>
> #include <vector>
> 
>+#include "call/video_send_stream.h"
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/logging.h"
> #include "rtc_base/platform_thread.h"
> #include "rtc_base/task_queue.h"
> #include "rtc_base/timeutils.h"
> #include "system_wrappers/include/clock.h"
>-#include "test/frame_generator.h"
>-#include "call/video_send_stream.h"
> 
> namespace webrtc {
> namespace test {
>@@ -85,25 +84,16 @@ class FrameGeneratorCapturer::InsertFrameTask : public rtc::QueuedTask {
>   int64_t intended_run_time_ms_;
> };
> 
>-FrameGeneratorCapturer* FrameGeneratorCapturer::Create(int width,
>-                                                       int height,
>-                                                       int target_fps,
>-                                                       Clock* clock) {
>-  std::unique_ptr<FrameGeneratorCapturer> capturer(new FrameGeneratorCapturer(
>-      clock, FrameGenerator::CreateSquareGenerator(width, height), target_fps));
>-  if (!capturer->Init())
>-    return nullptr;
>-
>-  return capturer.release();
>-}
>-
>-FrameGeneratorCapturer* FrameGeneratorCapturer::Create(int width,
>-                                                       int height,
>-                                                       int num_squares,
>-                                                       int target_fps,
>-                                                       Clock* clock) {
>+FrameGeneratorCapturer* FrameGeneratorCapturer::Create(
>+    int width,
>+    int height,
>+    absl::optional<FrameGenerator::OutputType> type,
>+    absl::optional<int> num_squares,
>+    int target_fps,
>+    Clock* clock) {
>   std::unique_ptr<FrameGeneratorCapturer> capturer(new FrameGeneratorCapturer(
>-      clock, FrameGenerator::CreateSquareGenerator(width, height, num_squares),
>+      clock,
>+      FrameGenerator::CreateSquareGenerator(width, height, type, num_squares),
>       target_fps));
>   if (!capturer->Init())
>     return nullptr;
>@@ -135,8 +125,8 @@ FrameGeneratorCapturer* FrameGeneratorCapturer::CreateSlideGenerator(
>     int target_fps,
>     Clock* clock) {
>   std::unique_ptr<FrameGeneratorCapturer> capturer(new FrameGeneratorCapturer(
>-      clock, FrameGenerator::CreateSlideGenerator(width, height,
>-                                                  frame_repeat_count),
>+      clock,
>+      FrameGenerator::CreateSlideGenerator(width, height, frame_repeat_count),
>       target_fps));
>   if (!capturer->Init())
>     return nullptr;
>@@ -155,8 +145,7 @@ FrameGeneratorCapturer::FrameGeneratorCapturer(
>       frame_generator_(std::move(frame_generator)),
>       target_fps_(target_fps),
>       first_frame_capture_time_(-1),
>-      task_queue_("FrameGenCapQ",
>-                  rtc::TaskQueue::Priority::HIGH) {
>+      task_queue_("FrameGenCapQ", rtc::TaskQueue::Priority::HIGH) {
>   RTC_DCHECK(frame_generator_);
>   RTC_DCHECK_GT(target_fps, 0);
> }
>@@ -197,7 +186,7 @@ void FrameGeneratorCapturer::InsertFrame() {
>     }
> 
>     if (sink_) {
>-      rtc::Optional<VideoFrame> out_frame = AdaptFrame(*frame);
>+      absl::optional<VideoFrame> out_frame = AdaptFrame(*frame);
>       if (out_frame)
>         sink_->OnFrame(*out_frame);
>     }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/frame_generator_capturer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/frame_generator_capturer.h
>index 3a3fe82431c..63219900d8f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/frame_generator_capturer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/frame_generator_capturer.h
>@@ -16,13 +16,11 @@
> #include "api/video/video_frame.h"
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/task_queue.h"
>+#include "test/frame_generator.h"
> #include "test/video_capturer.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>-class EventTimerWrapper;
>-
> namespace test {
> 
> class FrameGenerator;
>@@ -40,16 +38,15 @@ class FrameGeneratorCapturer : public VideoCapturer {
>     virtual ~SinkWantsObserver() {}
>   };
> 
>-  static FrameGeneratorCapturer* Create(int width,
>-                                        int height,
>-                                        int target_fps,
>-                                        Clock* clock);
>-
>-  static FrameGeneratorCapturer* Create(int width,
>-                                        int height,
>-                                        int num_squares,
>-                                        int target_fps,
>-                                        Clock* clock);
>+  // |type| has the default value OutputType::I420. |num_squares| has the
>+  // default value 10.
>+  static FrameGeneratorCapturer* Create(
>+      int width,
>+      int height,
>+      absl::optional<FrameGenerator::OutputType> type,
>+      absl::optional<int> num_squares,
>+      int target_fps,
>+      Clock* clock);
> 
>   static FrameGeneratorCapturer* CreateFromYuvFile(const std::string& file_name,
>                                                    size_t width,
>@@ -100,7 +97,7 @@ class FrameGeneratorCapturer : public VideoCapturer {
>   std::unique_ptr<FrameGenerator> frame_generator_;
> 
>   int target_fps_ RTC_GUARDED_BY(&lock_);
>-  rtc::Optional<int> wanted_fps_ RTC_GUARDED_BY(&lock_);
>+  absl::optional<int> wanted_fps_ RTC_GUARDED_BY(&lock_);
>   VideoRotation fake_rotation_ = kVideoRotation_0;
> 
>   int64_t first_frame_capture_time_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/frame_generator_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/frame_generator_unittest.cc
>index ba4627dc3b8..14e2f291c54 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/frame_generator_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/frame_generator_unittest.cc
>@@ -169,8 +169,8 @@ TEST_F(FrameGeneratorTest, SlideGenerator) {
>   const int kGenCount = 9;
>   const int kRepeatCount = 3;
>   std::unique_ptr<FrameGenerator> generator(
>-      FrameGenerator::CreateSlideGenerator(
>-          kFrameWidth, kFrameHeight, kRepeatCount));
>+      FrameGenerator::CreateSlideGenerator(kFrameWidth, kFrameHeight,
>+                                           kRepeatCount));
>   uint64_t hashes[kGenCount];
>   for (int i = 0; i < kGenCount; ++i) {
>     hashes[i] = Hash(generator->NextFrame());
>@@ -178,9 +178,9 @@ TEST_F(FrameGeneratorTest, SlideGenerator) {
>   // Check that the buffer changes only every |kRepeatCount| frames.
>   for (int i = 1; i < kGenCount; ++i) {
>     if (i % kRepeatCount == 0) {
>-      EXPECT_NE(hashes[i-1], hashes[i]);
>+      EXPECT_NE(hashes[i - 1], hashes[i]);
>     } else {
>-      EXPECT_EQ(hashes[i-1], hashes[i]);
>+      EXPECT_EQ(hashes[i - 1], hashes[i]);
>     }
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/frame_utils.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/frame_utils.cc
>index 1d5fe7c8f48..833f57162c2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/frame_utils.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/frame_utils.cc
>@@ -59,18 +59,17 @@ bool FrameBufsEqual(const rtc::scoped_refptr<webrtc::VideoFrameBuffer>& f1,
> 
>   rtc::scoped_refptr<webrtc::I420BufferInterface> f1_i420 = f1->ToI420();
>   rtc::scoped_refptr<webrtc::I420BufferInterface> f2_i420 = f2->ToI420();
>-  return EqualPlane(f1_i420->DataY(), f2_i420->DataY(),
>-                    f1_i420->StrideY(), f2_i420->StrideY(),
>-                    f1_i420->width(), f1_i420->height()) &&
>-         EqualPlane(f1_i420->DataU(), f2_i420->DataU(),
>-                    f1_i420->StrideU(), f2_i420->StrideU(),
>-                    f1_i420->ChromaWidth(), f1_i420->ChromaHeight()) &&
>-         EqualPlane(f1_i420->DataV(), f2_i420->DataV(),
>-                    f1_i420->StrideV(), f2_i420->StrideV(),
>-                    f1_i420->ChromaWidth(), f1_i420->ChromaHeight());
>+  return EqualPlane(f1_i420->DataY(), f2_i420->DataY(), f1_i420->StrideY(),
>+                    f2_i420->StrideY(), f1_i420->width(), f1_i420->height()) &&
>+         EqualPlane(f1_i420->DataU(), f2_i420->DataU(), f1_i420->StrideU(),
>+                    f2_i420->StrideU(), f1_i420->ChromaWidth(),
>+                    f1_i420->ChromaHeight()) &&
>+         EqualPlane(f1_i420->DataV(), f2_i420->DataV(), f1_i420->StrideV(),
>+                    f2_i420->StrideV(), f1_i420->ChromaWidth(),
>+                    f1_i420->ChromaHeight());
> }
> 
>-rtc::scoped_refptr<I420Buffer> ReadI420Buffer(int width, int height, FILE *f) {
>+rtc::scoped_refptr<I420Buffer> ReadI420Buffer(int width, int height, FILE* f) {
>   int half_width = (width + 1) / 2;
>   rtc::scoped_refptr<I420Buffer> buffer(
>       // Explicit stride, no padding between rows.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/frame_utils.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/frame_utils.h
>index 80466e1c291..c8293111995 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/frame_utils.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/frame_utils.h
>@@ -40,7 +40,7 @@ bool FramesEqual(const webrtc::VideoFrame& f1, const webrtc::VideoFrame& f2);
> bool FrameBufsEqual(const rtc::scoped_refptr<webrtc::VideoFrameBuffer>& f1,
>                     const rtc::scoped_refptr<webrtc::VideoFrameBuffer>& f2);
> 
>-rtc::scoped_refptr<I420Buffer> ReadI420Buffer(int width, int height, FILE *);
>+rtc::scoped_refptr<I420Buffer> ReadI420Buffer(int width, int height, FILE*);
> 
> }  // namespace test
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/function_video_decoder_factory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/function_video_decoder_factory.h
>new file mode 100644
>index 00000000000..00b01680a11
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/function_video_decoder_factory.h
>@@ -0,0 +1,51 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef TEST_FUNCTION_VIDEO_DECODER_FACTORY_H_
>+#define TEST_FUNCTION_VIDEO_DECODER_FACTORY_H_
>+
>+#include <functional>
>+#include <memory>
>+#include <utility>
>+#include <vector>
>+
>+#include "api/video_codecs/sdp_video_format.h"
>+#include "api/video_codecs/video_decoder_factory.h"
>+#include "rtc_base/checks.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+// A decoder factory producing decoders by calling a supplied create function.
>+class FunctionVideoDecoderFactory final : public VideoDecoderFactory {
>+ public:
>+  explicit FunctionVideoDecoderFactory(
>+      std::function<std::unique_ptr<VideoDecoder>()> create)
>+      : create_(std::move(create)) {}
>+
>+  // Unused by tests.
>+  std::vector<SdpVideoFormat> GetSupportedFormats() const override {
>+    RTC_NOTREACHED();
>+    return {};
>+  }
>+
>+  std::unique_ptr<VideoDecoder> CreateVideoDecoder(
>+      const SdpVideoFormat& /* format */) override {
>+    return create_();
>+  }
>+
>+ private:
>+  const std::function<std::unique_ptr<VideoDecoder>()> create_;
>+};
>+
>+}  // namespace test
>+}  // namespace webrtc
>+
>+#endif  // TEST_FUNCTION_VIDEO_DECODER_FACTORY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/function_video_encoder_factory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/function_video_encoder_factory.h
>new file mode 100644
>index 00000000000..567397f6c0c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/function_video_encoder_factory.h
>@@ -0,0 +1,65 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef TEST_FUNCTION_VIDEO_ENCODER_FACTORY_H_
>+#define TEST_FUNCTION_VIDEO_ENCODER_FACTORY_H_
>+
>+#include <functional>
>+#include <memory>
>+#include <utility>
>+#include <vector>
>+
>+#include "api/video_codecs/sdp_video_format.h"
>+#include "api/video_codecs/video_encoder_factory.h"
>+#include "rtc_base/checks.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+// An encoder factory producing encoders by calling a supplied create
>+// function.
>+class FunctionVideoEncoderFactory final : public VideoEncoderFactory {
>+ public:
>+  explicit FunctionVideoEncoderFactory(
>+      std::function<std::unique_ptr<VideoEncoder>()> create)
>+      : create_([create](const SdpVideoFormat&) { return create(); }) {}
>+  explicit FunctionVideoEncoderFactory(
>+      std::function<std::unique_ptr<VideoEncoder>(const SdpVideoFormat&)>
>+          create)
>+      : create_(std::move(create)) {}
>+
>+  // Unused by tests.
>+  std::vector<SdpVideoFormat> GetSupportedFormats() const override {
>+    RTC_NOTREACHED();
>+    return {};
>+  }
>+
>+  CodecInfo QueryVideoEncoder(
>+      const SdpVideoFormat& /* format */) const override {
>+    CodecInfo codec_info;
>+    codec_info.is_hardware_accelerated = false;
>+    codec_info.has_internal_source = false;
>+    return codec_info;
>+  }
>+
>+  std::unique_ptr<VideoEncoder> CreateVideoEncoder(
>+      const SdpVideoFormat& format) override {
>+    return create_(format);
>+  }
>+
>+ private:
>+  const std::function<std::unique_ptr<VideoEncoder>(const SdpVideoFormat&)>
>+      create_;
>+};
>+
>+}  // namespace test
>+}  // namespace webrtc
>+
>+#endif  // TEST_FUNCTION_VIDEO_ENCODER_FACTORY_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/BUILD.gn
>index 8a1cb97df28..f361d592618 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/BUILD.gn
>@@ -18,14 +18,28 @@ rtc_static_library("webrtc_fuzzer_main") {
>     "../../rtc_base:rtc_base_approved",
>     "../../system_wrappers:field_trial_default",
>     "../../system_wrappers:metrics_default",
>+    "../../system_wrappers:runtime_enabled_features_default",
>     "//testing/libfuzzer:libfuzzer_main",
>   ]
> }
> 
>+rtc_static_library("fuzz_data_helper") {
>+  sources = [
>+    "fuzz_data_helper.cc",
>+    "fuzz_data_helper.h",
>+  ]
>+  deps = [
>+    "../../api:array_view",
>+    "../../modules/rtp_rtcp:rtp_rtcp_format",
>+  ]
>+  visibility = [ ":*" ]  # Only targets in this file can depend on this.
>+}
>+
> template("webrtc_fuzzer_test") {
>   fuzzer_test(target_name) {
>     forward_variables_from(invoker, "*")
>     deps += [
>+      ":fuzz_data_helper",
>       ":webrtc_fuzzer_main",
> 
>       # Link unconditionally with webrtc's TaskQueue, regardless of
>@@ -96,12 +110,25 @@ webrtc_fuzzer_test("h264_bitstream_parser_fuzzer") {
>   ]
> }
> 
>+webrtc_fuzzer_test("forward_error_correction_fuzzer") {
>+  sources = [
>+    "forward_error_correction_fuzzer.cc",
>+  ]
>+  deps = [
>+    "../../modules/rtp_rtcp",
>+    "../../modules/rtp_rtcp:rtp_rtcp_format",
>+    "../../rtc_base:rtc_base_approved",
>+  ]
>+  libfuzzer_options = [ "max_len=5000" ]
>+}
>+
> webrtc_fuzzer_test("flexfec_header_reader_fuzzer") {
>   sources = [
>     "flexfec_header_reader_fuzzer.cc",
>   ]
>   deps = [
>     "../../modules/rtp_rtcp",
>+    "../../modules/rtp_rtcp:rtp_rtcp_format",
>     "../../rtc_base:rtc_base_approved",
>   ]
> }
>@@ -112,6 +139,7 @@ webrtc_fuzzer_test("flexfec_sender_fuzzer") {
>   ]
>   deps = [
>     "../../modules/rtp_rtcp",
>+    "../../modules/rtp_rtcp:rtp_rtcp_format",
>     "../../system_wrappers",
>   ]
>   libfuzzer_options = [ "max_len=200" ]
>@@ -124,6 +152,7 @@ webrtc_fuzzer_test("ulpfec_header_reader_fuzzer") {
>   deps = [
>     "../../modules/rtp_rtcp",
>     "../../modules/rtp_rtcp:fec_test_helper",
>+    "../../modules/rtp_rtcp:rtp_rtcp_format",
>     "../../rtc_base:rtc_base_approved",
>   ]
> }
>@@ -133,8 +162,11 @@ webrtc_fuzzer_test("ulpfec_generator_fuzzer") {
>     "ulpfec_generator_fuzzer.cc",
>   ]
>   deps = [
>+    "../../modules:module_api_public",
>     "../../modules/rtp_rtcp",
>     "../../modules/rtp_rtcp:fec_test_helper",
>+    "../../modules/rtp_rtcp:rtp_rtcp_format",
>+    "../../rtc_base:checks",
>     "../../rtc_base:rtc_base_approved",
>   ]
> }
>@@ -145,6 +177,7 @@ webrtc_fuzzer_test("ulpfec_receiver_fuzzer") {
>   ]
>   deps = [
>     "../../modules/rtp_rtcp",
>+    "../../modules/rtp_rtcp:rtp_rtcp_format",
>     "../../rtc_base:rtc_base_approved",
>   ]
>   libfuzzer_options = [ "max_len=2000" ]
>@@ -156,6 +189,7 @@ webrtc_fuzzer_test("flexfec_receiver_fuzzer") {
>   ]
>   deps = [
>     "../../modules/rtp_rtcp",
>+    "../../modules/rtp_rtcp:rtp_rtcp_format",
>     "../../rtc_base:rtc_base_approved",
>   ]
>   libfuzzer_options = [ "max_len=2000" ]
>@@ -169,7 +203,7 @@ webrtc_fuzzer_test("packet_buffer_fuzzer") {
>     "../../modules/video_coding/",
>     "../../system_wrappers",
>   ]
>-  libfuzzer_options = [ "max_len=2000" ]
>+  libfuzzer_options = [ "max_len=200000" ]
> }
> 
> webrtc_fuzzer_test("rtcp_receiver_fuzzer") {
>@@ -178,6 +212,7 @@ webrtc_fuzzer_test("rtcp_receiver_fuzzer") {
>   ]
>   deps = [
>     "../../modules/rtp_rtcp",
>+    "../../rtc_base:checks",
>     "../../rtc_base:rtc_base_approved",
>     "../../system_wrappers:system_wrappers",
>   ]
>@@ -189,7 +224,7 @@ webrtc_fuzzer_test("rtp_packet_fuzzer") {
>     "rtp_packet_fuzzer.cc",
>   ]
>   deps = [
>-    "../../modules/rtp_rtcp",
>+    "../../modules/rtp_rtcp:rtp_rtcp_format",
>   ]
>   seed_corpus = "corpora/rtp-corpus"
> }
>@@ -200,6 +235,7 @@ webrtc_fuzzer_test("rtp_header_fuzzer") {
>   ]
>   deps = [
>     "../../modules/rtp_rtcp",
>+    "../../modules/rtp_rtcp:rtp_rtcp_format",
>   ]
> }
> 
>@@ -209,11 +245,11 @@ webrtc_fuzzer_test("congestion_controller_feedback_fuzzer") {
>   ]
>   deps = [
>     "../../logging:rtc_event_log_api",
>-    "../../logging:rtc_event_log_impl",
>+    "../../logging:rtc_event_log_impl_base",
>     "../../modules/congestion_controller",
>     "../../modules/pacing",
>     "../../modules/remote_bitrate_estimator:remote_bitrate_estimator",
>-    "../../modules/rtp_rtcp",
>+    "../../modules/rtp_rtcp:rtp_rtcp_format",
>   ]
> }
> 
>@@ -224,10 +260,11 @@ rtc_static_library("audio_decoder_fuzzer") {
>   ]
>   deps = [
>     "../..:webrtc_common",
>-    "../../api:optional",
>     "../../api/audio_codecs:audio_codecs_api",
>-    "../../modules/rtp_rtcp",
>+    "../../modules/rtp_rtcp:rtp_rtcp_format",
>+    "../../rtc_base:checks",
>     "../../rtc_base:rtc_base_approved",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
> 
>@@ -239,6 +276,8 @@ webrtc_fuzzer_test("audio_decoder_ilbc_fuzzer") {
>     ":audio_decoder_fuzzer",
>     "../../modules/audio_coding:ilbc",
>   ]
>+
>+  libfuzzer_options = [ "max_len=10000" ]
> }
> 
> webrtc_fuzzer_test("audio_decoder_isac_fuzzer") {
>@@ -249,6 +288,8 @@ webrtc_fuzzer_test("audio_decoder_isac_fuzzer") {
>     ":audio_decoder_fuzzer",
>     "../../modules/audio_coding:isac",
>   ]
>+
>+  libfuzzer_options = [ "max_len=20000" ]
> }
> 
> webrtc_fuzzer_test("audio_decoder_isac_incoming_packet_fuzzer") {
>@@ -259,6 +300,8 @@ webrtc_fuzzer_test("audio_decoder_isac_incoming_packet_fuzzer") {
>     ":audio_decoder_fuzzer",
>     "../../modules/audio_coding:isac",
>   ]
>+
>+  libfuzzer_options = [ "max_len=20000" ]
> }
> 
> webrtc_fuzzer_test("audio_decoder_isacfix_fuzzer") {
>@@ -269,6 +312,8 @@ webrtc_fuzzer_test("audio_decoder_isacfix_fuzzer") {
>     ":audio_decoder_fuzzer",
>     "../../modules/audio_coding:isac_fix",
>   ]
>+
>+  libfuzzer_options = [ "max_len=20000" ]
> }
> 
> webrtc_fuzzer_test("audio_decoder_opus_fuzzer") {
>@@ -291,12 +336,25 @@ webrtc_fuzzer_test("audio_decoder_opus_redundant_fuzzer") {
>   ]
> }
> 
>+webrtc_fuzzer_test("audio_encoder_opus_fuzzer") {
>+  sources = [
>+    "audio_encoder_opus_fuzzer.cc",
>+  ]
>+  deps = [
>+    "../../api:array_view",
>+    "../../api/audio_codecs/opus:audio_encoder_opus",
>+    "../../rtc_base:checks",
>+    "../../rtc_base:rtc_base_approved",
>+  ]
>+}
>+
> webrtc_fuzzer_test("turn_unwrap_fuzzer") {
>   sources = [
>     "turn_unwrap_fuzzer.cc",
>   ]
>   deps = [
>     "../../media:media",
>+    "../../media:rtc_media_base",
>   ]
> }
> 
>@@ -310,10 +368,12 @@ webrtc_fuzzer_test("neteq_rtp_fuzzer") {
>     "../../modules/audio_coding:neteq_test_tools",
>     "../../modules/audio_coding:neteq_tools_minimal",
>     "../../modules/audio_coding:pcm16b",
>-    "../../modules/rtp_rtcp",
>+    "../../modules/rtp_rtcp:rtp_rtcp_format",
>     "../../rtc_base:rtc_base_approved",
>     "../../rtc_base:rtc_base_tests_utils",
>   ]
>+
>+  libfuzzer_options = [ "max_len=100000" ]
> }
> 
> webrtc_fuzzer_test("neteq_signal_fuzzer") {
>@@ -326,10 +386,11 @@ webrtc_fuzzer_test("neteq_signal_fuzzer") {
>     "../../modules/audio_coding:neteq_test_tools",
>     "../../modules/audio_coding:neteq_tools_minimal",
>     "../../modules/audio_coding:pcm16b",
>-    "../../modules/rtp_rtcp",
>     "../../rtc_base:rtc_base_approved",
>     "../../rtc_base:rtc_base_tests_utils",
>   ]
>+
>+  libfuzzer_options = [ "max_len=100000" ]
> }
> 
> webrtc_fuzzer_test("residual_echo_detector_fuzzer") {
>@@ -338,6 +399,7 @@ webrtc_fuzzer_test("residual_echo_detector_fuzzer") {
>   ]
>   deps = [
>     "../../modules/audio_processing:audio_processing",
>+    "../../rtc_base:checks",
>     "../../rtc_base:rtc_base_approved",
>   ]
> }
>@@ -347,9 +409,11 @@ webrtc_fuzzer_test("sdp_parser_fuzzer") {
>     "sdp_parser_fuzzer.cc",
>   ]
>   deps = [
>+    "../../api:libjingle_peerconnection_api",
>     "../../pc:libjingle_peerconnection",
>   ]
>   seed_corpus = "corpora/sdp-corpus"
>+  libfuzzer_options = [ "max_len=16384" ]
> }
> 
> webrtc_fuzzer_test("stun_parser_fuzzer") {
>@@ -390,22 +454,94 @@ webrtc_fuzzer_test("transport_feedback_packet_loss_tracker_fuzzer") {
>   ]
>   deps = [
>     "../../api:array_view",
>-    "../../modules/rtp_rtcp",
>+    "../../audio",
>+    "../../modules/rtp_rtcp:rtp_rtcp_format",
>+    "../../rtc_base:rtc_base_approved",
>+  ]
>+}
>+rtc_static_library("audio_processing_fuzzer_helper") {
>+  sources = [
>+    "audio_processing_fuzzer_helper.cc",
>+    "audio_processing_fuzzer_helper.h",
>+  ]
>+  deps = [
>+    ":fuzz_data_helper",
>+    "../../api/audio:audio_frame_api",
>+    "../../modules/audio_processing",
>+    "../../rtc_base:checks",
>     "../../rtc_base:rtc_base_approved",
>-    "../../voice_engine",
>+    "//third_party/abseil-cpp/absl/types:optional",
>   ]
> }
> 
> webrtc_fuzzer_test("audio_processing_fuzzer") {
>   sources = [
>-    "audio_processing_fuzzer.cc",
>-    "audio_processing_fuzzer.h",
>-    "audio_processing_fuzzer_configs.cc",
>+    "audio_processing_configs_fuzzer.cc",
>+  ]
>+  deps = [
>+    ":audio_processing_fuzzer_helper",
>+    "../../api/audio:aec3_factory",
>+    "../../modules/audio_processing",
>+    "../../modules/audio_processing/aec3",
>+    "../../modules/audio_processing/aec_dump:mock_aec_dump",
>+    "../../rtc_base:ptr_util",
>+    "../../rtc_base:rtc_base_approved",
>+    "../../rtc_base:safe_minmax",
>+    "../../system_wrappers:field_trial_default",
>+    "//third_party/abseil-cpp/absl/memory",
>+  ]
>+  seed_corpus = "corpora/audio_processing-corpus"
>+}
>+
>+webrtc_fuzzer_test("agc_fuzzer") {
>+  sources = [
>+    "agc_fuzzer.cc",
>   ]
>   deps = [
>-    "../../api:optional",
>-    "../../modules:module_api",
>+    ":fuzz_data_helper",
>     "../../modules/audio_processing",
>     "../../rtc_base:rtc_base_approved",
>+    "../../rtc_base:safe_minmax",
>+    "//third_party/abseil-cpp/absl/memory",
>+  ]
>+
>+  seed_corpus = "corpora/agc-corpus"
>+  libfuzzer_options = [ "max_len=200000" ]
>+}
>+
>+webrtc_fuzzer_test("comfort_noise_decoder_fuzzer") {
>+  sources = [
>+    "comfort_noise_decoder_fuzzer.cc",
>+  ]
>+  deps = [
>+    "../../api:array_view",
>+    "../../modules/audio_coding:cng",
>+    "../../rtc_base:rtc_base_approved",
>+  ]
>+
>+  libfuzzer_options = [ "max_len=5000" ]
>+}
>+
>+webrtc_fuzzer_test("rtp_frame_reference_finder_fuzzer") {
>+  sources = [
>+    "rtp_frame_reference_finder_fuzzer.cc",
>+  ]
>+  deps = [
>+    "../../modules/video_coding/",
>+    "../../rtc_base:ptr_util",
>+    "../../system_wrappers",
>+    "//third_party/abseil-cpp/absl/memory",
>+  ]
>+  libfuzzer_options = [ "max_len=20000" ]
>+}
>+
>+webrtc_fuzzer_test("frame_buffer2_fuzzer") {
>+  sources = [
>+    "frame_buffer2_fuzzer.cc",
>+  ]
>+  deps = [
>+    "../../modules/video_coding/",
>+    "../../system_wrappers:system_wrappers",
>   ]
>+  libfuzzer_options = [ "max_len=10000" ]
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/DEPS b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/DEPS
>index cdbb5660e99..222cd0202ea 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/DEPS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/DEPS
>@@ -1,3 +1,3 @@
> include_rules = [
>-  "+webrtc",
>+  "+audio",
> ]
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/OWNERS
>index 6e7454380ea..1ba807eff3b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/OWNERS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/OWNERS
>@@ -1 +1,3 @@
> henrik.lundin@webrtc.org
>+aleloi@webrtc.org
>+saza@webrtc.org
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/agc_fuzzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/agc_fuzzer.cc
>new file mode 100644
>index 00000000000..5bd921e74da
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/agc_fuzzer.cc
>@@ -0,0 +1,118 @@
>+/*
>+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "absl/memory/memory.h"
>+#include "modules/audio_processing/audio_buffer.h"
>+#include "modules/audio_processing/gain_control_impl.h"
>+#include "modules/audio_processing/include/audio_processing.h"
>+#include "rtc_base/numerics/safe_minmax.h"
>+#include "rtc_base/thread_annotations.h"
>+#include "test/fuzzers/fuzz_data_helper.h"
>+
>+namespace webrtc {
>+namespace {
>+
>+void FillAudioBuffer(test::FuzzDataHelper* fuzz_data, AudioBuffer* buffer) {
>+  float* const* channels = buffer->channels_f();
>+  for (size_t i = 0; i < buffer->num_channels(); ++i) {
>+    for (size_t j = 0; j < buffer->num_frames(); ++j) {
>+      channels[i][j] =
>+          static_cast<float>(fuzz_data->ReadOrDefaultValue<int16_t>(0));
>+    }
>+  }
>+}
>+
>+// This function calls the GainControl functions that are overriden as private
>+// in GainControlInterface.
>+void FuzzGainControllerConfig(test::FuzzDataHelper* fuzz_data,
>+                              GainControl* gc) {
>+  GainControl::Mode modes[] = {GainControl::Mode::kAdaptiveAnalog,
>+                               GainControl::Mode::kAdaptiveDigital,
>+                               GainControl::Mode::kFixedDigital};
>+  GainControl::Mode mode = fuzz_data->SelectOneOf(modes);
>+  const bool enable_limiter = fuzz_data->ReadOrDefaultValue(true);
>+  // The values are capped to comply with the API of webrtc::GainControl.
>+  const int analog_level_min =
>+      rtc::SafeClamp<int>(fuzz_data->ReadOrDefaultValue<uint16_t>(0), 0, 65534);
>+  const int analog_level_max =
>+      rtc::SafeClamp<int>(fuzz_data->ReadOrDefaultValue<uint16_t>(65535),
>+                          analog_level_min + 1, 65535);
>+  const int stream_analog_level =
>+      rtc::SafeClamp<int>(fuzz_data->ReadOrDefaultValue<uint16_t>(30000),
>+                          analog_level_min, analog_level_max);
>+  const int gain =
>+      rtc::SafeClamp<int>(fuzz_data->ReadOrDefaultValue<int8_t>(30), -1, 100);
>+  const int target_level_dbfs =
>+      rtc::SafeClamp<int>(fuzz_data->ReadOrDefaultValue<int8_t>(15), -1, 35);
>+
>+  gc->set_mode(mode);
>+  gc->enable_limiter(enable_limiter);
>+  if (mode == GainControl::Mode::kAdaptiveAnalog) {
>+    gc->set_analog_level_limits(analog_level_min, analog_level_max);
>+    gc->set_stream_analog_level(stream_analog_level);
>+  }
>+  gc->set_compression_gain_db(gain);
>+  gc->set_target_level_dbfs(target_level_dbfs);
>+  gc->Enable(true);
>+
>+  static_cast<void>(gc->is_enabled());
>+  static_cast<void>(gc->mode());
>+  static_cast<void>(gc->analog_level_minimum());
>+  static_cast<void>(gc->analog_level_maximum());
>+  static_cast<void>(gc->stream_analog_level());
>+  static_cast<void>(gc->compression_gain_db());
>+  static_cast<void>(gc->stream_is_saturated());
>+  static_cast<void>(gc->target_level_dbfs());
>+  static_cast<void>(gc->is_limiter_enabled());
>+}
>+
>+void FuzzGainController(test::FuzzDataHelper* fuzz_data, GainControlImpl* gci) {
>+  using Rate = ::webrtc::AudioProcessing::NativeRate;
>+  const Rate rate_kinds[] = {Rate::kSampleRate8kHz, Rate::kSampleRate16kHz,
>+                             Rate::kSampleRate32kHz, Rate::kSampleRate48kHz};
>+
>+  const auto sample_rate_hz =
>+      static_cast<size_t>(fuzz_data->SelectOneOf(rate_kinds));
>+  const size_t samples_per_frame = sample_rate_hz / 100;
>+  const bool num_channels = fuzz_data->ReadOrDefaultValue(true) ? 2 : 1;
>+
>+  gci->Initialize(num_channels, sample_rate_hz);
>+  FuzzGainControllerConfig(fuzz_data, gci);
>+
>+  // The audio buffer is used for both capture and render.
>+  AudioBuffer audio(samples_per_frame, num_channels, samples_per_frame,
>+                    num_channels, samples_per_frame);
>+
>+  std::vector<int16_t> packed_render_audio(samples_per_frame);
>+
>+  while (fuzz_data->CanReadBytes(1)) {
>+    FillAudioBuffer(fuzz_data, &audio);
>+
>+    const bool stream_has_echo = fuzz_data->ReadOrDefaultValue(true);
>+    gci->AnalyzeCaptureAudio(&audio);
>+    gci->ProcessCaptureAudio(&audio, stream_has_echo);
>+
>+    FillAudioBuffer(fuzz_data, &audio);
>+
>+    gci->PackRenderAudioBuffer(&audio, &packed_render_audio);
>+    gci->ProcessRenderAudio(packed_render_audio);
>+  }
>+}
>+
>+}  // namespace
>+
>+void FuzzOneInput(const uint8_t* data, size_t size) {
>+  test::FuzzDataHelper fuzz_data(rtc::ArrayView<const uint8_t>(data, size));
>+  rtc::CriticalSection crit_capture;
>+  rtc::CriticalSection crit_render;
>+  auto gci = absl::make_unique<GainControlImpl>(&crit_render, &crit_capture);
>+  FuzzGainController(&fuzz_data, gci.get());
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/audio_decoder_fuzzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/audio_decoder_fuzzer.cc
>index f6ac4cb0bbb..2b4003b5e13 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/audio_decoder_fuzzer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/audio_decoder_fuzzer.cc
>@@ -12,8 +12,8 @@
> 
> #include <limits>
> 
>+#include "absl/types/optional.h"
> #include "api/audio_codecs/audio_decoder.h"
>-#include "api/optional.h"
> #include "modules/rtp_rtcp/source/byte_io.h"
> #include "rtc_base/checks.h"
> 
>@@ -51,8 +51,13 @@ void FuzzAudioDecoder(DecoderFunctionType decode_type,
>   const uint8_t* data_ptr = data;
>   size_t remaining_size = size;
>   size_t packet_len;
>-  while (ParseInt<size_t, 2>(&data_ptr, &remaining_size, &packet_len) &&
>-         packet_len <= remaining_size) {
>+  constexpr size_t kMaxNumFuzzedPackets = 200;
>+  for (size_t num_packets = 0; num_packets < kMaxNumFuzzedPackets;
>+       ++num_packets) {
>+    if (!(ParseInt<size_t, 2>(&data_ptr, &remaining_size, &packet_len) &&
>+          packet_len <= remaining_size)) {
>+      break;
>+    }
>     AudioDecoder::SpeechType speech_type;
>     switch (decode_type) {
>       case DecoderFunctionType::kNormalDecode:
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/audio_decoder_fuzzer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/audio_decoder_fuzzer.h
>index 504ecf6c02d..b935004220f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/audio_decoder_fuzzer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/audio_decoder_fuzzer.h
>@@ -12,8 +12,7 @@
> #define TEST_FUZZERS_AUDIO_DECODER_FUZZER_H_
> 
> #include <stddef.h>
>-
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include <stdint.h>
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/audio_encoder_opus_fuzzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/audio_encoder_opus_fuzzer.cc
>new file mode 100644
>index 00000000000..50c285616b9
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/audio_encoder_opus_fuzzer.cc
>@@ -0,0 +1,64 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "api/array_view.h"
>+#include "api/audio_codecs/opus/audio_encoder_opus.h"
>+#include "rtc_base/buffer.h"
>+#include "rtc_base/checks.h"
>+#include "test/fuzzers/fuzz_data_helper.h"
>+
>+namespace webrtc {
>+namespace {
>+
>+// This function reads bytes from |data_view|, interprets them
>+// as RTP timestamp and input samples, and sends them for encoding. The process
>+// continues until no more data is available.
>+void FuzzAudioEncoder(rtc::ArrayView<const uint8_t> data_view,
>+                      AudioEncoder* encoder) {
>+  test::FuzzDataHelper data(data_view);
>+  const size_t block_size_samples =
>+      encoder->SampleRateHz() / 100 * encoder->NumChannels();
>+  const size_t block_size_bytes = block_size_samples * sizeof(int16_t);
>+  if (data_view.size() / block_size_bytes > 1000) {
>+    // If the size of the fuzzer data is more than 1000 input blocks (i.e., more
>+    // than 10 seconds), then don't fuzz at all for the fear of timing out.
>+    return;
>+  }
>+
>+  rtc::BufferT<int16_t> input_aligned(block_size_samples);
>+  rtc::Buffer encoded;
>+
>+  // Each round in the loop below will need one block of samples + a 32-bit
>+  // timestamp from the fuzzer input.
>+  const size_t bytes_to_read = block_size_bytes + sizeof(uint32_t);
>+  while (data.CanReadBytes(bytes_to_read)) {
>+    const uint32_t timestamp = data.Read<uint32_t>();
>+    auto byte_array = data.ReadByteArray(block_size_bytes);
>+    // Align the data by copying to another array.
>+    RTC_DCHECK_EQ(input_aligned.size() * sizeof(int16_t),
>+                  byte_array.size() * sizeof(uint8_t));
>+    memcpy(input_aligned.data(), byte_array.data(), byte_array.size());
>+    auto info = encoder->Encode(timestamp, input_aligned, &encoded);
>+  }
>+}
>+
>+}  // namespace
>+
>+void FuzzOneInput(const uint8_t* data, size_t size) {
>+  AudioEncoderOpus::Config config;
>+  config.frame_size_ms = 20;
>+  RTC_CHECK(config.IsOk());
>+  constexpr int kPayloadType = 100;
>+  std::unique_ptr<AudioEncoder> enc =
>+      AudioEncoderOpus::MakeAudioEncoder(config, kPayloadType);
>+  FuzzAudioEncoder(rtc::ArrayView<const uint8_t>(data, size), enc.get());
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/audio_processing_configs_fuzzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/audio_processing_configs_fuzzer.cc
>new file mode 100644
>index 00000000000..bcbfabbfef9
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/audio_processing_configs_fuzzer.cc
>@@ -0,0 +1,168 @@
>+/*
>+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <bitset>
>+#include <string>
>+
>+#include "absl/memory/memory.h"
>+#include "api/audio/echo_canceller3_factory.h"
>+#include "modules/audio_processing/aec_dump/mock_aec_dump.h"
>+#include "modules/audio_processing/include/audio_processing.h"
>+#include "rtc_base/arraysize.h"
>+#include "rtc_base/numerics/safe_minmax.h"
>+#include "system_wrappers/include/field_trial_default.h"
>+#include "test/fuzzers/audio_processing_fuzzer_helper.h"
>+#include "test/fuzzers/fuzz_data_helper.h"
>+
>+namespace webrtc {
>+namespace {
>+
>+const std::string kFieldTrialNames[] = {
>+    "WebRTC-Aec3TransparentModeKillSwitch",
>+    "WebRTC-Aec3StationaryRenderImprovementsKillSwitch",
>+    "WebRTC-Aec3EnforceDelayAfterRealignmentKillSwitch",
>+    "WebRTC-Aec3UseShortDelayEstimatorWindow",
>+    "WebRTC-Aec3ReverbBasedOnRenderKillSwitch",
>+    "WebRTC-Aec3ReverbModellingKillSwitch",
>+    "WebRTC-Aec3FilterAnalyzerPreprocessorKillSwitch",
>+    "WebRTC-Aec3TransparencyImprovementsKillSwitch",
>+    "WebRTC-Aec3SoftTransparentModeKillSwitch",
>+    "WebRTC-Aec3OverrideEchoPathGainKillSwitch",
>+    "WebRTC-Aec3ZeroExternalDelayHeadroomKillSwitch",
>+    "WebRTC-Aec3DownSamplingFactor8KillSwitch",
>+    "WebRTC-Aec3EnforceSkewHysteresis1",
>+    "WebRTC-Aec3EnforceSkewHysteresis2",
>+    "WebRTC-Aec3NewSuppressionKillSwitch",
>+    "WebRTC-Aec3LinearModeWithDivergedFilterKillSwitch",
>+    "WebRTC-Aec3MisadjustmentEstimatorKillSwitch",
>+    "WebRTC-Aec3RapidAgcGainRecoveryKillSwitch",
>+    "WebRTC-Aec3SlowFilterAdaptationKillSwitch",
>+    "WebRTC-Aec3SmoothUpdatesTailFreqRespKillSwitch",
>+    "WebRTC-Aec3SuppressorNearendAveragingKillSwitch",
>+    "WebRTC-Aec3AgcGainChangeResponseKillSwitch",
>+    "WebRTC-Aec3ShadowFilterJumpstartKillSwitch",
>+    "WebRTC-Aec3EarlyLinearFilterUsageKillSwitch",
>+    "WebRTC-Aec3ShortInitialStateKillSwitch",
>+    "WebRTC-Aec3StandardNonlinearReverbModelKillSwitch",
>+    "WebRTC-Aec3EnableAdaptiveEchoReverbEstimation"};
>+
>+std::unique_ptr<AudioProcessing> CreateApm(test::FuzzDataHelper* fuzz_data,
>+                                           std::string* field_trial_string) {
>+  // Parse boolean values for optionally enabling different
>+  // configurable public components of APM.
>+  bool exp_agc = fuzz_data->ReadOrDefaultValue(true);
>+  bool exp_ns = fuzz_data->ReadOrDefaultValue(true);
>+  static_cast<void>(fuzz_data->ReadOrDefaultValue(true));
>+  bool ef = fuzz_data->ReadOrDefaultValue(true);
>+  bool raf = fuzz_data->ReadOrDefaultValue(true);
>+  static_cast<void>(fuzz_data->ReadOrDefaultValue(true));
>+  bool ie = fuzz_data->ReadOrDefaultValue(true);
>+  bool red = fuzz_data->ReadOrDefaultValue(true);
>+  bool hpf = fuzz_data->ReadOrDefaultValue(true);
>+  bool aec3 = fuzz_data->ReadOrDefaultValue(true);
>+
>+  bool use_aec = fuzz_data->ReadOrDefaultValue(true);
>+  bool use_aecm = fuzz_data->ReadOrDefaultValue(true);
>+  bool use_agc = fuzz_data->ReadOrDefaultValue(true);
>+  bool use_ns = fuzz_data->ReadOrDefaultValue(true);
>+  bool use_le = fuzz_data->ReadOrDefaultValue(true);
>+  bool use_vad = fuzz_data->ReadOrDefaultValue(true);
>+  bool use_agc_limiter = fuzz_data->ReadOrDefaultValue(true);
>+  bool use_agc2_limiter = fuzz_data->ReadOrDefaultValue(true);
>+
>+  // Read an int8 value, but don't let it be too large or small.
>+  const float gain_controller2_gain_db =
>+      rtc::SafeClamp<int>(fuzz_data->ReadOrDefaultValue<int8_t>(0), -50, 50);
>+
>+  constexpr size_t kNumFieldTrials = arraysize(kFieldTrialNames);
>+  // Verify that the read data type has enough bits to fuzz the field trials.
>+  using FieldTrialBitmaskType = uint32_t;
>+  RTC_DCHECK_LE(kNumFieldTrials, sizeof(FieldTrialBitmaskType) * 8);
>+  std::bitset<kNumFieldTrials> field_trial_bitmask(
>+      fuzz_data->ReadOrDefaultValue<FieldTrialBitmaskType>(0));
>+  for (size_t i = 0; i < kNumFieldTrials; ++i) {
>+    if (field_trial_bitmask[i]) {
>+      *field_trial_string += kFieldTrialNames[i] + "/Enabled/";
>+    }
>+  }
>+  field_trial::InitFieldTrialsFromString(field_trial_string->c_str());
>+
>+  // Ignore a few bytes. Bytes from this segment will be used for
>+  // future config flag changes. We assume 40 bytes is enough for
>+  // configuring the APM.
>+  constexpr size_t kSizeOfConfigSegment = 40;
>+  RTC_DCHECK(kSizeOfConfigSegment >= fuzz_data->BytesRead());
>+  static_cast<void>(
>+      fuzz_data->ReadByteArray(kSizeOfConfigSegment - fuzz_data->BytesRead()));
>+
>+  // Filter out incompatible settings that lead to CHECK failures.
>+  if ((use_aecm && use_aec) ||      // These settings cause CHECK failure.
>+      (use_aecm && aec3 && use_ns)  // These settings trigger webrtc:9489.
>+      ) {
>+    return nullptr;
>+  }
>+
>+  // Components can be enabled through webrtc::Config and
>+  // webrtc::AudioProcessingConfig.
>+  Config config;
>+
>+  std::unique_ptr<EchoControlFactory> echo_control_factory;
>+  if (aec3) {
>+    echo_control_factory.reset(new EchoCanceller3Factory());
>+  }
>+
>+  config.Set<ExperimentalAgc>(new ExperimentalAgc(exp_agc));
>+  config.Set<ExperimentalNs>(new ExperimentalNs(exp_ns));
>+  config.Set<ExtendedFilter>(new ExtendedFilter(ef));
>+  config.Set<RefinedAdaptiveFilter>(new RefinedAdaptiveFilter(raf));
>+  config.Set<DelayAgnostic>(new DelayAgnostic(true));
>+  config.Set<Intelligibility>(new Intelligibility(ie));
>+
>+  std::unique_ptr<AudioProcessing> apm(
>+      AudioProcessingBuilder()
>+          .SetEchoControlFactory(std::move(echo_control_factory))
>+          .Create(config));
>+
>+  apm->AttachAecDump(
>+      absl::make_unique<testing::NiceMock<webrtc::test::MockAecDump>>());
>+
>+  webrtc::AudioProcessing::Config apm_config;
>+  apm_config.residual_echo_detector.enabled = red;
>+  apm_config.high_pass_filter.enabled = hpf;
>+  apm_config.gain_controller2.enabled = use_agc2_limiter;
>+
>+  apm_config.gain_controller2.fixed_gain_db = gain_controller2_gain_db;
>+
>+  apm->ApplyConfig(apm_config);
>+
>+  apm->echo_cancellation()->Enable(use_aec);
>+  apm->echo_control_mobile()->Enable(use_aecm);
>+  apm->gain_control()->Enable(use_agc);
>+  apm->noise_suppression()->Enable(use_ns);
>+  apm->level_estimator()->Enable(use_le);
>+  apm->voice_detection()->Enable(use_vad);
>+  apm->gain_control()->enable_limiter(use_agc_limiter);
>+
>+  return apm;
>+}
>+}  // namespace
>+
>+void FuzzOneInput(const uint8_t* data, size_t size) {
>+  test::FuzzDataHelper fuzz_data(rtc::ArrayView<const uint8_t>(data, size));
>+  // This string must be in scope during execution, according to documentation
>+  // for field_trial_default.h. Hence it's created here and not in CreateApm.
>+  std::string field_trial_string = "";
>+  auto apm = CreateApm(&fuzz_data, &field_trial_string);
>+
>+  if (apm) {
>+    FuzzAudioProcessing(&fuzz_data, std::move(apm));
>+  }
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/audio_processing_fuzzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/audio_processing_fuzzer.cc
>deleted file mode 100644
>index a44d446096b..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/audio_processing_fuzzer.cc
>+++ /dev/null
>@@ -1,156 +0,0 @@
>-/*
>- *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "test/fuzzers/audio_processing_fuzzer.h"
>-
>-#include <algorithm>
>-#include <array>
>-#include <cmath>
>-
>-#include "modules/audio_processing/include/audio_processing.h"
>-#include "modules/include/module_common_types.h"
>-#include "rtc_base/checks.h"
>-
>-namespace webrtc {
>-namespace {
>-size_t ByteToNativeRate(uint8_t data) {
>-  using Rate = AudioProcessing::NativeRate;
>-  switch (data % 4) {
>-    case 0:
>-      return static_cast<size_t>(Rate::kSampleRate8kHz);
>-    case 1:
>-      return static_cast<size_t>(Rate::kSampleRate16kHz);
>-    case 2:
>-      return static_cast<size_t>(Rate::kSampleRate32kHz);
>-    default:
>-      return static_cast<size_t>(Rate::kSampleRate48kHz);
>-  }
>-}
>-
>-template <class T>
>-bool ParseSequence(size_t size,
>-                   const uint8_t** data,
>-                   size_t* remaining_size,
>-                   T* result_data) {
>-  const size_t data_size_bytes = sizeof(T) * size;
>-  if (data_size_bytes > *remaining_size) {
>-    return false;
>-  }
>-
>-  std::copy(*data, *data + data_size_bytes,
>-            reinterpret_cast<uint8_t*>(result_data));
>-
>-  *data += data_size_bytes;
>-  *remaining_size -= data_size_bytes;
>-  return true;
>-}
>-
>-void FuzzAudioProcessing(const uint8_t* data,
>-                         size_t size,
>-                         bool is_float,
>-                         AudioProcessing* apm) {
>-  AudioFrame fixed_frame;
>-  std::array<float, 480> float_frame{};
>-  float* const first_channel = &float_frame[0];
>-
>-  while (size > 0) {
>-    // Decide input/output rate for this iteration.
>-    const auto input_rate_byte = ParseByte(&data, &size);
>-    const auto output_rate_byte = ParseByte(&data, &size);
>-    if (!input_rate_byte || !output_rate_byte) {
>-      return;
>-    }
>-    const auto input_rate_hz = ByteToNativeRate(*input_rate_byte);
>-    const auto output_rate_hz = ByteToNativeRate(*output_rate_byte);
>-
>-    const size_t samples_per_input_channel =
>-        rtc::CheckedDivExact(input_rate_hz, 100ul);
>-    fixed_frame.samples_per_channel_ = samples_per_input_channel;
>-    fixed_frame.sample_rate_hz_ = input_rate_hz;
>-
>-    // Two channels breaks AEC3.
>-    fixed_frame.num_channels_ = 1;
>-
>-    // Fill the arrays with audio samples from the data.
>-    if (is_float) {
>-      if (!ParseSequence(samples_per_input_channel, &data, &size,
>-                         &float_frame[0])) {
>-        return;
>-      }
>-    } else if (!ParseSequence(samples_per_input_channel, &data, &size,
>-                              fixed_frame.mutable_data())) {
>-      return;
>-    }
>-
>-    // Filter obviously wrong values like inf/nan and values that will
>-    // lead to inf/nan in calculations. 1e6 leads to DCHECKS failing.
>-    for (auto& x : float_frame) {
>-      if (!std::isnormal(x) || std::abs(x) > 1e5) {
>-        x = 0;
>-      }
>-    }
>-
>-    // Make the APM call depending on capture/render mode and float /
>-    // fix interface.
>-    const auto is_capture = ParseBool(&data, &size);
>-    if (!is_capture) {
>-      return;
>-    }
>-    if (*is_capture) {
>-      auto apm_return_code =
>-          is_float ? (apm->ProcessStream(
>-                         &first_channel, StreamConfig(input_rate_hz, 1),
>-                         StreamConfig(output_rate_hz, 1), &first_channel))
>-                   : (apm->ProcessStream(&fixed_frame));
>-      RTC_DCHECK_NE(apm_return_code, AudioProcessing::kBadDataLengthError);
>-    } else {
>-      auto apm_return_code =
>-          is_float ? (apm->ProcessReverseStream(
>-                         &first_channel, StreamConfig(input_rate_hz, 1),
>-                         StreamConfig(output_rate_hz, 1), &first_channel))
>-                   : (apm->ProcessReverseStream(&fixed_frame));
>-      RTC_DCHECK_NE(apm_return_code, AudioProcessing::kBadDataLengthError);
>-    }
>-  }
>-}
>-
>-}  // namespace
>-
>-rtc::Optional<bool> ParseBool(const uint8_t** data, size_t* remaining_size) {
>-  if (1 > *remaining_size) {
>-    return rtc::nullopt;
>-  }
>-  auto res = (**data) % 2;
>-  *data += 1;
>-  *remaining_size -= 1;
>-  return res;
>-}
>-
>-rtc::Optional<uint8_t> ParseByte(const uint8_t** data, size_t* remaining_size) {
>-  if (1 > *remaining_size) {
>-    return rtc::nullopt;
>-  }
>-  auto res = **data;
>-  *data += 1;
>-  *remaining_size -= 1;
>-  return res;
>-}
>-
>-void FuzzAudioProcessing(const uint8_t* data,
>-                         size_t size,
>-                         std::unique_ptr<AudioProcessing> apm) {
>-  const auto is_float = ParseBool(&data, &size);
>-  if (!is_float) {
>-    return;
>-  }
>-
>-  FuzzAudioProcessing(data, size, *is_float, apm.get());
>-}
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/audio_processing_fuzzer_configs.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/audio_processing_fuzzer_configs.cc
>deleted file mode 100644
>index 15e0beeba9a..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/audio_processing_fuzzer_configs.cc
>+++ /dev/null
>@@ -1,99 +0,0 @@
>-/*
>- *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "modules/audio_processing/include/audio_processing.h"
>-#include "test/fuzzers/audio_processing_fuzzer.h"
>-
>-#include "api/optional.h"
>-
>-namespace webrtc {
>-
>-std::unique_ptr<AudioProcessing> CreateAPM(const uint8_t** data,
>-                                           size_t* remaining_size) {
>-  // Parse boolean values for optionally enabling different
>-  // configurable public components of APM.
>-  auto exp_agc = ParseBool(data, remaining_size);
>-  auto exp_ns = ParseBool(data, remaining_size);
>-  auto bf = ParseBool(data, remaining_size);
>-  auto ef = ParseBool(data, remaining_size);
>-  auto raf = ParseBool(data, remaining_size);
>-  auto da = ParseBool(data, remaining_size);
>-  auto ie = ParseBool(data, remaining_size);
>-  auto red = ParseBool(data, remaining_size);
>-  auto lc = ParseBool(data, remaining_size);
>-  auto hpf = ParseBool(data, remaining_size);
>-  auto aec3 = ParseBool(data, remaining_size);
>-
>-  auto use_aec = ParseBool(data, remaining_size);
>-  auto use_aecm = ParseBool(data, remaining_size);
>-  auto use_agc = ParseBool(data, remaining_size);
>-  auto use_ns = ParseBool(data, remaining_size);
>-  auto use_le = ParseBool(data, remaining_size);
>-  auto use_vad = ParseBool(data, remaining_size);
>-  auto use_agc_limiter = ParseBool(data, remaining_size);
>-
>-  if (!(exp_agc && exp_ns && bf && ef && raf && da && ie && red && lc && hpf &&
>-        aec3 && use_aec && use_aecm && use_agc && use_ns && use_le && use_vad &&
>-        use_agc_limiter)) {
>-    return nullptr;
>-  }
>-
>-  // Filter out incompatible settings that lead to CHECK failures.
>-  if (*use_aecm && *use_aec) {
>-    return nullptr;
>-  }
>-
>-  // Components can be enabled through webrtc::Config and
>-  // webrtc::AudioProcessingConfig.
>-  Config config;
>-
>-  std::unique_ptr<EchoControlFactory> echo_control_factory;
>-  if (*aec3) {
>-    echo_control_factory.reset(new EchoCanceller3Factory());
>-  }
>-
>-  config.Set<ExperimentalAgc>(new ExperimentalAgc(*exp_agc));
>-  config.Set<ExperimentalNs>(new ExperimentalNs(*exp_ns));
>-  if (*bf) {
>-    config.Set<Beamforming>(new Beamforming());
>-  }
>-  config.Set<ExtendedFilter>(new ExtendedFilter(*ef));
>-  config.Set<RefinedAdaptiveFilter>(new RefinedAdaptiveFilter(*raf));
>-  config.Set<DelayAgnostic>(new DelayAgnostic(*da));
>-  config.Set<Intelligibility>(new Intelligibility(*ie));
>-
>-  std::unique_ptr<AudioProcessing> apm(AudioProcessing::Create(
>-      config, nullptr, std::move(echo_control_factory), nullptr));
>-
>-  webrtc::AudioProcessing::Config apm_config;
>-  apm_config.residual_echo_detector.enabled = *red;
>-  apm_config.level_controller.enabled = *lc;
>-  apm_config.high_pass_filter.enabled = *hpf;
>-
>-  apm->ApplyConfig(apm_config);
>-
>-  apm->echo_cancellation()->Enable(*use_aec);
>-  apm->echo_control_mobile()->Enable(*use_aecm);
>-  apm->gain_control()->Enable(*use_agc);
>-  apm->noise_suppression()->Enable(*use_ns);
>-  apm->level_estimator()->Enable(*use_le);
>-  apm->voice_detection()->Enable(*use_vad);
>-  apm->gain_control()->enable_limiter(*use_agc_limiter);
>-
>-  return apm;
>-}
>-
>-void FuzzOneInput(const uint8_t* data, size_t size) {
>-  auto apm = CreateAPM(&data, &size);
>-  if (apm) {
>-    FuzzAudioProcessing(data, size, std::move(apm));
>-  }
>-}
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/audio_processing_fuzzer_helper.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/audio_processing_fuzzer_helper.cc
>new file mode 100644
>index 00000000000..9989111c4d1
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/audio_processing_fuzzer_helper.cc
>@@ -0,0 +1,142 @@
>+/*
>+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "test/fuzzers/audio_processing_fuzzer_helper.h"
>+
>+#include <algorithm>
>+#include <array>
>+#include <cmath>
>+#include <limits>
>+
>+#include "api/audio/audio_frame.h"
>+#include "modules/audio_processing/include/audio_processing.h"
>+#include "rtc_base/checks.h"
>+
>+namespace webrtc {
>+namespace {
>+bool ValidForApm(float x) {
>+  return std::isfinite(x) && -1.0f <= x && x <= 1.0f;
>+}
>+
>+void GenerateFloatFrame(test::FuzzDataHelper* fuzz_data,
>+                        size_t input_rate,
>+                        size_t num_channels,
>+                        float* const* float_frames) {
>+  const size_t samples_per_input_channel =
>+      rtc::CheckedDivExact(input_rate, 100ul);
>+  RTC_DCHECK_LE(samples_per_input_channel, 480);
>+  for (size_t i = 0; i < num_channels; ++i) {
>+    std::fill(float_frames[i], float_frames[i] + samples_per_input_channel, 0);
>+    const size_t read_bytes = sizeof(float) * samples_per_input_channel;
>+    if (fuzz_data->CanReadBytes(read_bytes)) {
>+      rtc::ArrayView<const uint8_t> byte_array =
>+          fuzz_data->ReadByteArray(read_bytes);
>+      memmove(float_frames[i], byte_array.begin(), read_bytes);
>+    }
>+
>+    // Sanitize input.
>+    for (size_t j = 0; j < samples_per_input_channel; ++j) {
>+      if (!ValidForApm(float_frames[i][j])) {
>+        float_frames[i][j] = 0.f;
>+      }
>+    }
>+  }
>+}
>+
>+void GenerateFixedFrame(test::FuzzDataHelper* fuzz_data,
>+                        size_t input_rate,
>+                        size_t num_channels,
>+                        AudioFrame* fixed_frame) {
>+  const size_t samples_per_input_channel =
>+      rtc::CheckedDivExact(input_rate, 100ul);
>+  fixed_frame->samples_per_channel_ = samples_per_input_channel;
>+  fixed_frame->sample_rate_hz_ = input_rate;
>+  fixed_frame->num_channels_ = num_channels;
>+
>+  RTC_DCHECK_LE(samples_per_input_channel * num_channels,
>+                AudioFrame::kMaxDataSizeSamples);
>+  for (size_t i = 0; i < samples_per_input_channel * num_channels; ++i) {
>+    fixed_frame->mutable_data()[i] = fuzz_data->ReadOrDefaultValue<int16_t>(0);
>+  }
>+}
>+}  // namespace
>+
>+void FuzzAudioProcessing(test::FuzzDataHelper* fuzz_data,
>+                         std::unique_ptr<AudioProcessing> apm) {
>+  AudioFrame fixed_frame;
>+  std::array<float, 480> float_frame1;
>+  std::array<float, 480> float_frame2;
>+  std::array<float* const, 2> float_frame_ptrs = {
>+      &float_frame1[0], &float_frame2[0],
>+  };
>+  float* const* ptr_to_float_frames = &float_frame_ptrs[0];
>+
>+  using Rate = AudioProcessing::NativeRate;
>+  const Rate rate_kinds[] = {Rate::kSampleRate8kHz, Rate::kSampleRate16kHz,
>+                             Rate::kSampleRate32kHz, Rate::kSampleRate48kHz};
>+
>+  // We may run out of fuzz data in the middle of a loop iteration. In
>+  // that case, default values will be used for the rest of that
>+  // iteration.
>+  while (fuzz_data->CanReadBytes(1)) {
>+    const bool is_float = fuzz_data->ReadOrDefaultValue(true);
>+    // Decide input/output rate for this iteration.
>+    const auto input_rate =
>+        static_cast<size_t>(fuzz_data->SelectOneOf(rate_kinds));
>+    const auto output_rate =
>+        static_cast<size_t>(fuzz_data->SelectOneOf(rate_kinds));
>+
>+    const int num_channels = fuzz_data->ReadOrDefaultValue(true) ? 2 : 1;
>+    const uint8_t stream_delay = fuzz_data->ReadOrDefaultValue<uint8_t>(0);
>+
>+    // API call needed for AEC-2 and AEC-m to run.
>+    apm->set_stream_delay_ms(stream_delay);
>+
>+    const bool key_pressed = fuzz_data->ReadOrDefaultValue(true);
>+    apm->set_stream_key_pressed(key_pressed);
>+
>+    // Make the APM call depending on capture/render mode and float /
>+    // fix interface.
>+    const bool is_capture = fuzz_data->ReadOrDefaultValue(true);
>+
>+    // Fill the arrays with audio samples from the data.
>+    int apm_return_code = AudioProcessing::Error::kNoError;
>+    if (is_float) {
>+      GenerateFloatFrame(fuzz_data, input_rate, num_channels,
>+                         ptr_to_float_frames);
>+      if (is_capture) {
>+        apm_return_code = apm->ProcessStream(
>+            ptr_to_float_frames, StreamConfig(input_rate, num_channels),
>+            StreamConfig(output_rate, num_channels), ptr_to_float_frames);
>+      } else {
>+        apm_return_code = apm->ProcessReverseStream(
>+            ptr_to_float_frames, StreamConfig(input_rate, 1),
>+            StreamConfig(output_rate, 1), ptr_to_float_frames);
>+      }
>+    } else {
>+      GenerateFixedFrame(fuzz_data, input_rate, num_channels, &fixed_frame);
>+
>+      if (is_capture) {
>+        apm_return_code = apm->ProcessStream(&fixed_frame);
>+      } else {
>+        apm_return_code = apm->ProcessReverseStream(&fixed_frame);
>+      }
>+    }
>+
>+    // Make calls to stats gathering functions to cover these
>+    // codeways.
>+    static_cast<void>(apm->GetStatistics());
>+    static_cast<void>(apm->GetStatistics(true));
>+    static_cast<void>(apm->UpdateHistogramsOnCallEnd());
>+
>+    RTC_DCHECK_NE(apm_return_code, AudioProcessing::kBadDataLengthError);
>+  }
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/audio_processing_fuzzer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/audio_processing_fuzzer_helper.h
>similarity index 60%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/audio_processing_fuzzer.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/audio_processing_fuzzer_helper.h
>index 337d9b215dd..697ed8db613 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/audio_processing_fuzzer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/audio_processing_fuzzer_helper.h
>@@ -8,20 +8,18 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#ifndef TEST_FUZZERS_AUDIO_PROCESSING_FUZZER_H_
>-#define TEST_FUZZERS_AUDIO_PROCESSING_FUZZER_H_
>+#ifndef TEST_FUZZERS_AUDIO_PROCESSING_FUZZER_HELPER_H_
>+#define TEST_FUZZERS_AUDIO_PROCESSING_FUZZER_HELPER_H_
> 
> #include <memory>
> 
> #include "modules/audio_processing/include/audio_processing.h"
>+#include "test/fuzzers/fuzz_data_helper.h"
> namespace webrtc {
> 
>-rtc::Optional<bool> ParseBool(const uint8_t** data, size_t* remaining_size);
>-rtc::Optional<uint8_t> ParseByte(const uint8_t** data, size_t* remaining_size);
>-
>-void FuzzAudioProcessing(const uint8_t* data,
>-                         size_t size,
>+void FuzzAudioProcessing(test::FuzzDataHelper* fuzz_data,
>                          std::unique_ptr<AudioProcessing> apm);
>+
> }  // namespace webrtc
> 
>-#endif  // TEST_FUZZERS_AUDIO_PROCESSING_FUZZER_H_
>+#endif  // TEST_FUZZERS_AUDIO_PROCESSING_FUZZER_HELPER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/comfort_noise_decoder_fuzzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/comfort_noise_decoder_fuzzer.cc
>new file mode 100644
>index 00000000000..50166d7df77
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/comfort_noise_decoder_fuzzer.cc
>@@ -0,0 +1,56 @@
>+/*
>+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <algorithm>
>+
>+#include "api/array_view.h"
>+#include "modules/audio_coding/codecs/cng/webrtc_cng.h"
>+#include "rtc_base/buffer.h"
>+#include "test/fuzzers/fuzz_data_helper.h"
>+
>+namespace webrtc {
>+namespace test {
>+namespace {
>+
>+void FuzzOneInputTest(rtc::ArrayView<const uint8_t> data) {
>+  FuzzDataHelper fuzz_data(data);
>+  ComfortNoiseDecoder cng_decoder;
>+
>+  while (1) {
>+    if (!fuzz_data.CanReadBytes(1))
>+      break;
>+    const uint8_t sid_frame_len = fuzz_data.Read<uint8_t>();
>+    auto sid_frame = fuzz_data.ReadByteArray(sid_frame_len);
>+    if (sid_frame.empty())
>+      break;
>+    cng_decoder.UpdateSid(sid_frame);
>+    if (!fuzz_data.CanReadBytes(3))
>+      break;
>+    constexpr bool kTrueOrFalse[] = {true, false};
>+    const bool new_period = fuzz_data.SelectOneOf(kTrueOrFalse);
>+    constexpr size_t kOutputSizes[] = {80, 160, 320, 480};
>+    const size_t output_size = fuzz_data.SelectOneOf(kOutputSizes);
>+    const size_t num_generate_calls =
>+        std::min(fuzz_data.Read<uint8_t>(), static_cast<uint8_t>(17));
>+    rtc::BufferT<int16_t> output(output_size);
>+    for (size_t i = 0; i < num_generate_calls; ++i) {
>+      cng_decoder.Generate(output, new_period);
>+    }
>+  }
>+}
>+
>+}  // namespace
>+}  // namespace test
>+
>+void FuzzOneInput(const uint8_t* data, size_t size) {
>+  test::FuzzOneInputTest(rtc::ArrayView<const uint8_t>(data, size));
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/corpora/agc-corpus/agc-1 b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/corpora/agc-corpus/agc-1
>new file mode 100644
>index 00000000000..cda107c5341
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/corpora/agc-corpus/agc-1 differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/corpora/agc-corpus/agc-2 b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/corpora/agc-corpus/agc-2
>new file mode 100644
>index 00000000000..bf1a98dccd7
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/corpora/agc-corpus/agc-2 differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/corpora/agc-corpus/agc-3 b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/corpora/agc-corpus/agc-3
>new file mode 100644
>index 00000000000..f28283502a8
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/corpora/agc-corpus/agc-3 differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/corpora/agc-corpus/agc-4 b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/corpora/agc-corpus/agc-4
>new file mode 100644
>index 00000000000..7b5d1cd7908
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/corpora/agc-corpus/agc-4 differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/corpora/audio_processing-corpus/audio-processing-0 b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/corpora/audio_processing-corpus/audio-processing-0
>new file mode 100644
>index 00000000000..bc8471e92b9
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/corpora/audio_processing-corpus/audio-processing-0 differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/corpora/audio_processing-corpus/audio-processing-1 b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/corpora/audio_processing-corpus/audio-processing-1
>new file mode 100644
>index 00000000000..3fe154394a1
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/corpora/audio_processing-corpus/audio-processing-1 differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/corpora/audio_processing-corpus/audio-processing-2 b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/corpora/audio_processing-corpus/audio-processing-2
>new file mode 100644
>index 00000000000..cbd68b6fd53
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/corpora/audio_processing-corpus/audio-processing-2 differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/corpora/audio_processing-corpus/audio-processing-3 b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/corpora/audio_processing-corpus/audio-processing-3
>new file mode 100644
>index 00000000000..50f85fbcae5
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/corpora/audio_processing-corpus/audio-processing-3 differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/corpora/rtp-corpus/rtp-5 b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/corpora/rtp-corpus/rtp-5
>new file mode 100644
>index 00000000000..7dd66320072
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/corpora/rtp-corpus/rtp-5 differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/corpora/rtp-corpus/rtp-6 b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/corpora/rtp-corpus/rtp-6
>new file mode 100644
>index 00000000000..3a1145cd0f7
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/corpora/rtp-corpus/rtp-6 differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/corpora/rtp-corpus/rtp-7 b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/corpora/rtp-corpus/rtp-7
>new file mode 100644
>index 00000000000..6c4a9feb524
>Binary files /dev/null and b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/corpora/rtp-corpus/rtp-7 differ
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/flexfec_receiver_fuzzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/flexfec_receiver_fuzzer.cc
>index 14208bd7ec4..d96a328fa2c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/flexfec_receiver_fuzzer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/flexfec_receiver_fuzzer.cc
>@@ -14,7 +14,6 @@
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> #include "modules/rtp_rtcp/source/byte_io.h"
> #include "modules/rtp_rtcp/source/rtp_packet_received.h"
>-#include "rtc_base/basictypes.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/flexfec_sender_fuzzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/flexfec_sender_fuzzer.cc
>index 64bc3ed2471..8e79f95cf3f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/flexfec_sender_fuzzer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/flexfec_sender_fuzzer.cc
>@@ -12,8 +12,8 @@
> 
> #include "modules/rtp_rtcp/include/flexfec_sender.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>-#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
> #include "modules/rtp_rtcp/source/byte_io.h"
>+#include "modules/rtp_rtcp/source/rtp_packet_to_send.h"
> #include "system_wrappers/include/clock.h"
> 
> namespace webrtc {
>@@ -23,6 +23,7 @@ namespace {
> constexpr int kFlexfecPayloadType = 123;
> constexpr uint32_t kMediaSsrc = 1234;
> constexpr uint32_t kFlexfecSsrc = 5678;
>+const char kNoMid[] = "";
> const std::vector<RtpExtension> kNoRtpHeaderExtensions;
> const std::vector<RtpExtensionSize> kNoRtpHeaderExtensionSizes;
> 
>@@ -35,7 +36,7 @@ void FuzzOneInput(const uint8_t* data, size_t size) {
>   }
> 
>   SimulatedClock clock(1 + data[i++]);
>-  FlexfecSender sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc,
>+  FlexfecSender sender(kFlexfecPayloadType, kFlexfecSsrc, kMediaSsrc, kNoMid,
>                        kNoRtpHeaderExtensions, kNoRtpHeaderExtensionSizes,
>                        nullptr /* rtp_state */, &clock);
>   FecProtectionParams params = {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/forward_error_correction_fuzzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/forward_error_correction_fuzzer.cc
>new file mode 100644
>index 00000000000..9d5b872434a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/forward_error_correction_fuzzer.cc
>@@ -0,0 +1,114 @@
>+/*
>+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <memory>
>+
>+#include "modules/rtp_rtcp/source/byte_io.h"
>+#include "modules/rtp_rtcp/source/forward_error_correction.h"
>+#include "rtc_base/bytebuffer.h"
>+#include "rtc_base/scoped_ref_ptr.h"
>+
>+namespace webrtc {
>+
>+namespace {
>+constexpr uint32_t kMediaSsrc = 100200300;
>+constexpr uint32_t kFecSsrc = 111222333;
>+
>+constexpr size_t kPacketSize = 50;
>+constexpr size_t kMaxPacketsInBuffer = 48;
>+}  // namespace
>+
>+void FuzzOneInput(const uint8_t* data, size_t size) {
>+  // Object under test.
>+  std::unique_ptr<ForwardErrorCorrection> fec =
>+      ForwardErrorCorrection::CreateFlexfec(kFecSsrc, kMediaSsrc);
>+
>+  // Entropy from fuzzer.
>+  rtc::ByteBufferReader fuzz_buffer(reinterpret_cast<const char*>(data), size);
>+
>+  // Initial stream state.
>+  uint16_t media_seqnum;
>+  if (!fuzz_buffer.ReadUInt16(&media_seqnum))
>+    return;
>+  const uint16_t original_media_seqnum = media_seqnum;
>+  uint16_t fec_seqnum;
>+  if (!fuzz_buffer.ReadUInt16(&fec_seqnum))
>+    return;
>+
>+  // Existing packets in the packet buffer.
>+  ForwardErrorCorrection::RecoveredPacketList recovered_packets;
>+  uint8_t num_existing_recovered_packets;
>+  if (!fuzz_buffer.ReadUInt8(&num_existing_recovered_packets))
>+    return;
>+  for (size_t i = 0; i < num_existing_recovered_packets % kMaxPacketsInBuffer;
>+       ++i) {
>+    ForwardErrorCorrection::RecoveredPacket* recovered_packet =
>+        new ForwardErrorCorrection::RecoveredPacket();
>+    recovered_packet->pkt = rtc::scoped_refptr<ForwardErrorCorrection::Packet>(
>+        new ForwardErrorCorrection::Packet());
>+    recovered_packet->pkt->length = kPacketSize;
>+    recovered_packet->ssrc = kMediaSsrc;
>+    recovered_packet->seq_num = media_seqnum++;
>+    recovered_packets.emplace_back(recovered_packet);
>+  }
>+
>+  // New packets received from the network.
>+  ForwardErrorCorrection::ReceivedPacket received_packet;
>+  received_packet.pkt = rtc::scoped_refptr<ForwardErrorCorrection::Packet>(
>+      new ForwardErrorCorrection::Packet());
>+  received_packet.pkt->length = kPacketSize;
>+  uint8_t* packet_buffer = received_packet.pkt->data;
>+  uint8_t reordering;
>+  uint16_t seq_num_diff;
>+  uint8_t packet_type;
>+  uint8_t packet_loss;
>+  while (true) {
>+    if (!fuzz_buffer.ReadBytes(reinterpret_cast<char*>(packet_buffer),
>+                               kPacketSize)) {
>+      return;
>+    }
>+    if (!fuzz_buffer.ReadUInt8(&reordering))
>+      return;
>+    if (!fuzz_buffer.ReadUInt16(&seq_num_diff))
>+      return;
>+    if (!fuzz_buffer.ReadUInt8(&packet_type))
>+      return;
>+    if (!fuzz_buffer.ReadUInt8(&packet_loss))
>+      return;
>+
>+    if (reordering % 10 != 0)
>+      seq_num_diff = 0;
>+
>+    if (packet_type % 2 == 0) {
>+      received_packet.is_fec = true;
>+      received_packet.ssrc = kFecSsrc;
>+      received_packet.seq_num = seq_num_diff + fec_seqnum++;
>+
>+      // Overwrite parts of the FlexFEC header for fuzzing efficiency.
>+      packet_buffer[0] = 0;                                       // R, F bits.
>+      ByteWriter<uint8_t>::WriteBigEndian(&packet_buffer[8], 1);  // SSRCCount.
>+      ByteWriter<uint32_t>::WriteBigEndian(&packet_buffer[12],
>+                                           kMediaSsrc);  // SSRC_i.
>+      ByteWriter<uint16_t>::WriteBigEndian(
>+          &packet_buffer[16], original_media_seqnum);  // SN base_i.
>+    } else {
>+      received_packet.is_fec = false;
>+      received_packet.ssrc = kMediaSsrc;
>+      received_packet.seq_num = seq_num_diff + media_seqnum++;
>+    }
>+
>+    if (packet_loss % 10 == 0)
>+      continue;
>+
>+    fec->DecodeFec(received_packet, &recovered_packets);
>+  }
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/frame_buffer2_fuzzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/frame_buffer2_fuzzer.cc
>new file mode 100644
>index 00000000000..57f0c74a7ed
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/frame_buffer2_fuzzer.cc
>@@ -0,0 +1,99 @@
>+/*
>+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/video_coding/frame_buffer2.h"
>+
>+#include "modules/video_coding/jitter_estimator.h"
>+#include "modules/video_coding/timing.h"
>+#include "system_wrappers/include/clock.h"
>+
>+namespace webrtc {
>+
>+namespace {
>+
>+// When DataReader runs out of data provided in the constructor it will
>+// just set/return 0 instead.
>+struct DataReader {
>+  DataReader(const uint8_t* data, size_t size) : data_(data), size_(size) {}
>+
>+  void CopyTo(void* destination, size_t dest_size) {
>+    memset(destination, 0, dest_size);
>+
>+    size_t bytes_to_copy = std::min(size_ - offset_, dest_size);
>+    memcpy(destination, data_ + offset_, bytes_to_copy);
>+    offset_ += bytes_to_copy;
>+  }
>+
>+  template <typename T>
>+  T GetNum() {
>+    T res;
>+    if (offset_ + sizeof(res) < size_) {
>+      memcpy(&res, data_ + offset_, sizeof(res));
>+      offset_ += sizeof(res);
>+      return res;
>+    }
>+
>+    offset_ = size_;
>+    return T(0);
>+  }
>+
>+  bool MoreToRead() { return offset_ < size_; }
>+
>+  const uint8_t* const data_;
>+  size_t size_;
>+  size_t offset_ = 0;
>+};
>+
>+class FuzzyFrameObject : public video_coding::EncodedFrame {
>+ public:
>+  FuzzyFrameObject() {}
>+  ~FuzzyFrameObject() {}
>+
>+  bool GetBitstream(uint8_t* destination) const override { return false; }
>+  uint32_t Timestamp() const override { return timestamp; }
>+  int64_t ReceivedTime() const override { return 0; }
>+  int64_t RenderTime() const override { return _renderTimeMs; }
>+};
>+}  // namespace
>+
>+void FuzzOneInput(const uint8_t* data, size_t size) {
>+  DataReader reader(data, size);
>+  Clock* clock = Clock::GetRealTimeClock();
>+  VCMJitterEstimator jitter_estimator(clock, 0, 0);
>+  VCMTiming timing(clock);
>+  video_coding::FrameBuffer frame_buffer(clock, &jitter_estimator, &timing,
>+                                         nullptr);
>+
>+  while (reader.MoreToRead()) {
>+    if (reader.GetNum<uint8_t>() & 1) {
>+      std::unique_ptr<FuzzyFrameObject> frame(new FuzzyFrameObject());
>+      frame->id.picture_id = reader.GetNum<int64_t>();
>+      frame->id.spatial_layer = reader.GetNum<uint8_t>();
>+      frame->timestamp = reader.GetNum<uint32_t>();
>+      frame->num_references = reader.GetNum<uint8_t>() %
>+                              video_coding::EncodedFrame::kMaxFrameReferences;
>+
>+      for (size_t r = 0; r < frame->num_references; ++r)
>+        frame->references[r] = reader.GetNum<int64_t>();
>+
>+      frame_buffer.InsertFrame(std::move(frame));
>+    } else {
>+      // Since we are not trying to trigger race conditions it does not make
>+      // sense to have a wait time > 0.
>+      const int kWaitTimeMs = 0;
>+
>+      std::unique_ptr<video_coding::EncodedFrame> frame(new FuzzyFrameObject());
>+      bool keyframe_required = reader.GetNum<uint8_t>() % 2;
>+      frame_buffer.NextFrame(kWaitTimeMs, &frame, keyframe_required);
>+    }
>+  }
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/stereo/include/stereo_globals.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/fuzz_data_helper.cc
>similarity index 59%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/stereo/include/stereo_globals.h
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/fuzz_data_helper.cc
>index 9f9ad0edb2a..866f7bc4b86 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/stereo/include/stereo_globals.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/fuzz_data_helper.cc
>@@ -8,17 +8,13 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#ifndef MODULES_VIDEO_CODING_CODECS_STEREO_INCLUDE_STEREO_GLOBALS_H_
>-#define MODULES_VIDEO_CODING_CODECS_STEREO_INCLUDE_STEREO_GLOBALS_H_
>+#include "test/fuzzers/fuzz_data_helper.h"
> 
> namespace webrtc {
>+namespace test {
> 
>-struct StereoIndices {
>-  uint8_t frame_index;
>-  uint8_t frame_count;
>-  uint16_t picture_index;
>-};
>+FuzzDataHelper::FuzzDataHelper(rtc::ArrayView<const uint8_t> data)
>+    : data_(data) {}
> 
>+}  // namespace test
> }  // namespace webrtc
>-
>-#endif  // MODULES_VIDEO_CODING_CODECS_STEREO_INCLUDE_STEREO_GLOBALS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/fuzz_data_helper.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/fuzz_data_helper.h
>new file mode 100644
>index 00000000000..4606de1b26d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/fuzz_data_helper.h
>@@ -0,0 +1,105 @@
>+/*
>+ *  Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef TEST_FUZZERS_FUZZ_DATA_HELPER_H_
>+#define TEST_FUZZERS_FUZZ_DATA_HELPER_H_
>+
>+#include <limits>
>+
>+#include "api/array_view.h"
>+#include "modules/rtp_rtcp/source/byte_io.h"
>+
>+namespace webrtc {
>+namespace test {
>+
>+// Helper class to take care of the fuzzer input, read from it, and keep track
>+// of when the end of the data has been reached.
>+class FuzzDataHelper {
>+ public:
>+  explicit FuzzDataHelper(rtc::ArrayView<const uint8_t> data);
>+
>+  // Returns true if n bytes can be read.
>+  bool CanReadBytes(size_t n) const { return data_ix_ + n <= data_.size(); }
>+
>+  // Reads and returns data of type T.
>+  template <typename T>
>+  T Read() {
>+    RTC_CHECK(CanReadBytes(sizeof(T)));
>+    T x = ByteReader<T>::ReadLittleEndian(&data_[data_ix_]);
>+    data_ix_ += sizeof(T);
>+    return x;
>+  }
>+
>+  // Reads and returns data of type T. Returns default_value if not enough
>+  // fuzzer input remains to read a T.
>+  template <typename T>
>+  T ReadOrDefaultValue(T default_value) {
>+    if (!CanReadBytes(sizeof(T))) {
>+      return default_value;
>+    }
>+    return Read<T>();
>+  }
>+
>+  // Like ReadOrDefaultValue, but replaces the value 0 with default_value.
>+  template <typename T>
>+  T ReadOrDefaultValueNotZero(T default_value) {
>+    static_assert(std::is_integral<T>::value, "");
>+    T x = ReadOrDefaultValue(default_value);
>+    return x == 0 ? default_value : x;
>+  }
>+
>+  // Returns one of the elements from the provided input array. The selection
>+  // is based on the fuzzer input data. If not enough fuzzer data is available,
>+  // the method will return the first element in the input array. The reason for
>+  // not flagging this as an error is to allow the method to be called from
>+  // class constructors, and in constructors we typically do not handle
>+  // errors. The code will work anyway, and the fuzzer will likely see that
>+  // providing more data will actually make this method return something else.
>+  template <typename T, size_t N>
>+  T SelectOneOf(const T (&select_from)[N]) {
>+    static_assert(N <= std::numeric_limits<uint8_t>::max(), "");
>+    // Read an index between 0 and select_from.size() - 1 from the fuzzer data.
>+    uint8_t index = ReadOrDefaultValue<uint8_t>(0) % N;
>+    return select_from[index];
>+  }
>+
>+  rtc::ArrayView<const uint8_t> ReadByteArray(size_t bytes) {
>+    if (!CanReadBytes(bytes)) {
>+      return rtc::ArrayView<const uint8_t>(nullptr, 0);
>+    }
>+    const size_t index_to_return = data_ix_;
>+    data_ix_ += bytes;
>+    return data_.subview(index_to_return, bytes);
>+  }
>+
>+  // If sizeof(T) > BytesLeft then the remaining bytes will be used and the rest
>+  // of the object will be zero initialized.
>+  template <typename T>
>+  void CopyTo(T* object) {
>+    memset(object, 0, sizeof(T));
>+
>+    size_t bytes_to_copy = std::min(BytesLeft(), sizeof(T));
>+    memcpy(object, data_.data() + data_ix_, bytes_to_copy);
>+    data_ix_ += bytes_to_copy;
>+  }
>+
>+  size_t BytesRead() const { return data_ix_; }
>+
>+  size_t BytesLeft() const { return data_.size() - data_ix_; };
>+
>+ private:
>+  rtc::ArrayView<const uint8_t> data_;
>+  size_t data_ix_ = 0;
>+};
>+
>+}  // namespace test
>+}  // namespace webrtc
>+
>+#endif  // TEST_FUZZERS_FUZZ_DATA_HELPER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/neteq_rtp_fuzzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/neteq_rtp_fuzzer.cc
>index 0f26367b686..8aa6be59dd0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/neteq_rtp_fuzzer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/neteq_rtp_fuzzer.cc
>@@ -65,11 +65,11 @@ class FuzzRtpInput : public NetEqInput {
>     FuzzHeader();
>   }
> 
>-  rtc::Optional<int64_t> NextPacketTime() const override {
>+  absl::optional<int64_t> NextPacketTime() const override {
>     return packet_->time_ms;
>   }
> 
>-  rtc::Optional<int64_t> NextOutputEventTime() const override {
>+  absl::optional<int64_t> NextOutputEventTime() const override {
>     return input_->NextOutputEventTime();
>   }
> 
>@@ -85,7 +85,7 @@ class FuzzRtpInput : public NetEqInput {
> 
>   bool ended() const override { return ended_; }
> 
>-  rtc::Optional<RTPHeader> NextHeader() const override {
>+  absl::optional<RTPHeader> NextHeader() const override {
>     RTC_DCHECK(packet_);
>     return packet_->header;
>   }
>@@ -129,29 +129,13 @@ void FuzzOneInputTest(const uint8_t* data, size_t size) {
>   std::unique_ptr<AudioChecksum> output(new AudioChecksum);
>   NetEqTest::Callbacks callbacks;
>   NetEq::Config config;
>-  NetEqTest::DecoderMap codecs;
>-  codecs[0] = std::make_pair(NetEqDecoder::kDecoderPCMu, "pcmu");
>-  codecs[8] = std::make_pair(NetEqDecoder::kDecoderPCMa, "pcma");
>-  codecs[103] = std::make_pair(NetEqDecoder::kDecoderISAC, "isac");
>-  codecs[104] = std::make_pair(NetEqDecoder::kDecoderISACswb, "isac-swb");
>-  codecs[111] = std::make_pair(NetEqDecoder::kDecoderOpus, "opus");
>-  codecs[93] = std::make_pair(NetEqDecoder::kDecoderPCM16B, "pcm16-nb");
>-  codecs[94] = std::make_pair(NetEqDecoder::kDecoderPCM16Bwb, "pcm16-wb");
>-  codecs[96] =
>-      std::make_pair(NetEqDecoder::kDecoderPCM16Bswb48kHz, "pcm16-swb48");
>-  codecs[9] = std::make_pair(NetEqDecoder::kDecoderG722, "g722");
>-  codecs[106] = std::make_pair(NetEqDecoder::kDecoderAVT, "avt");
>-  codecs[114] = std::make_pair(NetEqDecoder::kDecoderAVT16kHz, "avt-16");
>-  codecs[115] = std::make_pair(NetEqDecoder::kDecoderAVT32kHz, "avt-32");
>-  codecs[116] = std::make_pair(NetEqDecoder::kDecoderAVT48kHz, "avt-48");
>-  codecs[117] = std::make_pair(NetEqDecoder::kDecoderRED, "red");
>-  codecs[13] = std::make_pair(NetEqDecoder::kDecoderCNGnb, "cng-nb");
>-  codecs[98] = std::make_pair(NetEqDecoder::kDecoderCNGwb, "cng-wb");
>-  codecs[99] = std::make_pair(NetEqDecoder::kDecoderCNGswb32kHz, "cng-swb32");
>-  codecs[100] = std::make_pair(NetEqDecoder::kDecoderCNGswb48kHz, "cng-swb48");
>-  // This is the payload type that will be used for encoding.
>-  codecs[kPayloadType] =
>-      std::make_pair(NetEqDecoder::kDecoderPCM16Bswb32kHz, "pcm16-swb32");
>+  auto codecs = NetEqTest::StandardDecoderMap();
>+  // kPayloadType is the payload type that will be used for encoding. Verify
>+  // that it is included in the standard decoder map, and that it points to the
>+  // expected decoder type.
>+  RTC_CHECK_EQ(codecs.count(kPayloadType), 1);
>+  RTC_CHECK(codecs[kPayloadType].first == NetEqDecoder::kDecoderPCM16Bswb32kHz);
>+
>   NetEqTest::ExtDecoderMap ext_codecs;
> 
>   NetEqTest test(config, codecs, ext_codecs, std::move(input),
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/neteq_signal_fuzzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/neteq_signal_fuzzer.cc
>index f0b79113aeb..16e776072bc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/neteq_signal_fuzzer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/neteq_signal_fuzzer.cc
>@@ -18,75 +18,19 @@
> #include "modules/audio_coding/neteq/tools/audio_checksum.h"
> #include "modules/audio_coding/neteq/tools/encode_neteq_input.h"
> #include "modules/audio_coding/neteq/tools/neteq_test.h"
>-#include "modules/rtp_rtcp/source/byte_io.h"
> #include "rtc_base/numerics/safe_conversions.h"
> #include "rtc_base/random.h"
>+#include "test/fuzzers/fuzz_data_helper.h"
> 
> namespace webrtc {
> namespace test {
> namespace {
>-// Helper class to take care of the fuzzer input, read from it, and keep track
>-// of when the end of the data has been reached.
>-class FuzzData {
>- public:
>-  explicit FuzzData(rtc::ArrayView<const uint8_t> data) : data_(data) {}
>-
>-  // Returns true if n bytes can be read.
>-  bool CanReadBytes(size_t n) const { return data_ix_ + n <= data_.size(); }
>-
>-  // Reads and returns data of type T.
>-  template <typename T>
>-  T Read() {
>-    RTC_CHECK(CanReadBytes(sizeof(T)));
>-    T x = ByteReader<T>::ReadLittleEndian(&data_[data_ix_]);
>-    data_ix_ += sizeof(T);
>-    return x;
>-  }
>-
>-  // Reads and returns data of type T. Returns default_value if not enough
>-  // fuzzer input remains to read a T.
>-  template <typename T>
>-  T ReadOrDefaultValue(T default_value) {
>-    if (!CanReadBytes(sizeof(T))) {
>-      return default_value;
>-    }
>-    return Read<T>();
>-  }
>-
>-  // Like ReadOrDefaultValue, but replaces the value 0 with default_value.
>-  template <typename T>
>-  T ReadOrDefaultValueNotZero(T default_value) {
>-    static_assert(std::is_integral<T>::value, "");
>-    T x = ReadOrDefaultValue(default_value);
>-    return x == 0 ? default_value : x;
>-  }
>-
>-  // Returns one of the elements from the provided input array. The selection
>-  // is based on the fuzzer input data. If not enough fuzzer data is available,
>-  // the method will return the first element in the input array. The reason for
>-  // not flaggin this as an error is that the method is called from the
>-  // FuzzSignalInput constructor, and in constructors we typically do not handle
>-  // errors. The code will work anyway, and the fuzzer will likely see that
>-  // providing more data will actually make this method return something else.
>-  template <typename T>
>-  T SelectOneOf(rtc::ArrayView<const T> select_from) {
>-    RTC_CHECK_LE(select_from.size(), std::numeric_limits<uint8_t>::max());
>-    // Read an index between 0 and select_from.size() - 1 from the fuzzer data.
>-    uint8_t index = ReadOrDefaultValue<uint8_t>(0) % select_from.size();
>-    return select_from[index];
>-  }
>-
>- private:
>-  rtc::ArrayView<const uint8_t> data_;
>-  size_t data_ix_ = 0;
>-};
>-
> // Generate a mixture of sine wave and gaussian noise.
> class SineAndNoiseGenerator : public EncodeNetEqInput::Generator {
>  public:
>   // The noise generator is seeded with a value from the fuzzer data, but 0 is
>   // avoided (since it is not allowed by the Random class).
>-  SineAndNoiseGenerator(int sample_rate_hz, FuzzData* fuzz_data)
>+  SineAndNoiseGenerator(int sample_rate_hz, FuzzDataHelper* fuzz_data)
>       : sample_rate_hz_(sample_rate_hz),
>         fuzz_data_(*fuzz_data),
>         noise_generator_(fuzz_data_.ReadOrDefaultValueNotZero<uint64_t>(1)) {}
>@@ -117,13 +61,13 @@ class SineAndNoiseGenerator : public EncodeNetEqInput::Generator {
>   const double kPi = std::acos(-1);
>   std::vector<int16_t> samples_;
>   double phase_ = 0.0;
>-  FuzzData& fuzz_data_;
>+  FuzzDataHelper& fuzz_data_;
>   Random noise_generator_;
> };
> 
> class FuzzSignalInput : public NetEqInput {
>  public:
>-  explicit FuzzSignalInput(FuzzData* fuzz_data,
>+  explicit FuzzSignalInput(FuzzDataHelper* fuzz_data,
>                            int sample_rate,
>                            uint8_t payload_type)
>       : fuzz_data_(*fuzz_data) {
>@@ -141,15 +85,14 @@ class FuzzSignalInput : public NetEqInput {
>     // call to NetEq::GetAudio. 10 ms is nominal, 9 and 11 ms will both lead to
>     // clock drift (in different directions).
>     constexpr int output_event_periods[] = {9, 10, 11};
>-    output_event_period_ms_ =
>-        fuzz_data_.SelectOneOf(rtc::ArrayView<const int>(output_event_periods));
>+    output_event_period_ms_ = fuzz_data_.SelectOneOf(output_event_periods);
>   }
> 
>-  rtc::Optional<int64_t> NextPacketTime() const override {
>+  absl::optional<int64_t> NextPacketTime() const override {
>     return packet_->time_ms;
>   }
> 
>-  rtc::Optional<int64_t> NextOutputEventTime() const override {
>+  absl::optional<int64_t> NextOutputEventTime() const override {
>     return next_output_event_ms_;
>   }
> 
>@@ -181,14 +124,14 @@ class FuzzSignalInput : public NetEqInput {
> 
>   bool ended() const override { return ended_; }
> 
>-  rtc::Optional<RTPHeader> NextHeader() const override {
>+  absl::optional<RTPHeader> NextHeader() const override {
>     RTC_DCHECK(packet_);
>     return packet_->header;
>   }
> 
>  private:
>   bool ended_ = false;
>-  FuzzData& fuzz_data_;
>+  FuzzDataHelper& fuzz_data_;
>   std::unique_ptr<EncodeNetEqInput> input_;
>   std::unique_ptr<PacketData> packet_;
>   int64_t next_output_event_ms_ = 0;
>@@ -199,13 +142,13 @@ class FuzzSignalInput : public NetEqInput {
> void FuzzOneInputTest(const uint8_t* data, size_t size) {
>   if (size < 1)
>     return;
>-  FuzzData fuzz_data(rtc::ArrayView<const uint8_t>(data, size));
>+
>+  FuzzDataHelper fuzz_data(rtc::ArrayView<const uint8_t>(data, size));
> 
>   // Allowed sample rates and payload types used in the test.
>   std::pair<int, uint8_t> rate_types[] = {
>       {8000, 93}, {16000, 94}, {32000, 95}, {48000, 96}};
>-  const auto rate_type = fuzz_data.SelectOneOf(
>-      rtc::ArrayView<const std::pair<int, uint8_t>>(rate_types));
>+  const auto rate_type = fuzz_data.SelectOneOf(rate_types);
>   const int sample_rate = rate_type.first;
>   const uint8_t payload_type = rate_type.second;
> 
>@@ -221,31 +164,22 @@ void FuzzOneInputTest(const uint8_t* data, size_t size) {
>   NetEq::Config config;
>   config.enable_post_decode_vad = true;
>   config.enable_fast_accelerate = true;
>-  NetEqTest::DecoderMap codecs;
>-  codecs[0] = std::make_pair(NetEqDecoder::kDecoderPCMu, "pcmu");
>-  codecs[8] = std::make_pair(NetEqDecoder::kDecoderPCMa, "pcma");
>-  codecs[103] = std::make_pair(NetEqDecoder::kDecoderISAC, "isac");
>-  codecs[104] = std::make_pair(NetEqDecoder::kDecoderISACswb, "isac-swb");
>-  codecs[111] = std::make_pair(NetEqDecoder::kDecoderOpus, "opus");
>-  codecs[9] = std::make_pair(NetEqDecoder::kDecoderG722, "g722");
>-  codecs[106] = std::make_pair(NetEqDecoder::kDecoderAVT, "avt");
>-  codecs[114] = std::make_pair(NetEqDecoder::kDecoderAVT16kHz, "avt-16");
>-  codecs[115] = std::make_pair(NetEqDecoder::kDecoderAVT32kHz, "avt-32");
>-  codecs[116] = std::make_pair(NetEqDecoder::kDecoderAVT48kHz, "avt-48");
>-  codecs[117] = std::make_pair(NetEqDecoder::kDecoderRED, "red");
>-  codecs[13] = std::make_pair(NetEqDecoder::kDecoderCNGnb, "cng-nb");
>-  codecs[98] = std::make_pair(NetEqDecoder::kDecoderCNGwb, "cng-wb");
>-  codecs[99] = std::make_pair(NetEqDecoder::kDecoderCNGswb32kHz, "cng-swb32");
>-  codecs[100] = std::make_pair(NetEqDecoder::kDecoderCNGswb48kHz, "cng-swb48");
>-  // One of these payload types will be used for encoding.
>-  codecs[rate_types[0].second] =
>-      std::make_pair(NetEqDecoder::kDecoderPCM16B, "pcm16-nb");
>-  codecs[rate_types[1].second] =
>-      std::make_pair(NetEqDecoder::kDecoderPCM16Bwb, "pcm16-wb");
>-  codecs[rate_types[2].second] =
>-      std::make_pair(NetEqDecoder::kDecoderPCM16Bswb32kHz, "pcm16-swb32");
>-  codecs[rate_types[3].second] =
>-      std::make_pair(NetEqDecoder::kDecoderPCM16Bswb48kHz, "pcm16-swb48");
>+  auto codecs = NetEqTest::StandardDecoderMap();
>+  // rate_types contains the payload types that will be used for encoding.
>+  // Verify that they all are included in the standard decoder map, and that
>+  // they point to the expected decoder types.
>+  RTC_CHECK_EQ(codecs.count(rate_types[0].second), 1);
>+  RTC_CHECK(codecs[rate_types[0].second].first == NetEqDecoder::kDecoderPCM16B);
>+  RTC_CHECK_EQ(codecs.count(rate_types[1].second), 1);
>+  RTC_CHECK(codecs[rate_types[1].second].first ==
>+            NetEqDecoder::kDecoderPCM16Bwb);
>+  RTC_CHECK_EQ(codecs.count(rate_types[2].second), 1);
>+  RTC_CHECK(codecs[rate_types[2].second].first ==
>+            NetEqDecoder::kDecoderPCM16Bswb32kHz);
>+  RTC_CHECK_EQ(codecs.count(rate_types[3].second), 1);
>+  RTC_CHECK(codecs[rate_types[3].second].first ==
>+            NetEqDecoder::kDecoderPCM16Bswb48kHz);
>+
>   NetEqTest::ExtDecoderMap ext_codecs;
> 
>   NetEqTest test(config, codecs, ext_codecs, std::move(input),
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/packet_buffer_fuzzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/packet_buffer_fuzzer.cc
>index df6baf48e5d..b500f92b403 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/packet_buffer_fuzzer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/packet_buffer_fuzzer.cc
>@@ -10,9 +10,9 @@
> 
> #include "modules/video_coding/packet_buffer.h"
> #include "system_wrappers/include/clock.h"
>+#include "test/fuzzers/fuzz_data_helper.h"
> 
> namespace webrtc {
>-
> namespace {
> class NullCallback : public video_coding::OnReceivedFrameCallback {
>   void OnReceivedFrame(std::unique_ptr<video_coding::RtpFrameObject> frame) {}
>@@ -20,26 +20,32 @@ class NullCallback : public video_coding::OnReceivedFrameCallback {
> }  // namespace
> 
> void FuzzOneInput(const uint8_t* data, size_t size) {
>-  // Two bytes for the sequence number,
>-  // one byte for |is_first_packet_in_frame| and |markerBit|.
>-  constexpr size_t kMinDataNeeded = 3;
>-  if (size < kMinDataNeeded) {
>-    return;
>-  }
>-
>   VCMPacket packet;
>   NullCallback callback;
>   SimulatedClock clock(0);
>   rtc::scoped_refptr<video_coding::PacketBuffer> packet_buffer(
>       video_coding::PacketBuffer::Create(&clock, 8, 1024, &callback));
>+  test::FuzzDataHelper helper(rtc::ArrayView<const uint8_t>(data, size));
>+
>+  while (helper.BytesLeft()) {
>+    // The RTPVideoHeader is a complex type, so overwriting it with random data
>+    // will put it in an invalid state. Therefore we save/restore it.
>+    uint8_t video_header_backup[sizeof(packet.video_header)];
>+    memcpy(&video_header_backup, &packet.video_header,
>+           sizeof(packet.video_header));
>+
>+    helper.CopyTo(&packet);
>+
>+    memcpy(&packet.video_header, &video_header_backup,
>+           sizeof(packet.video_header));
>+
>+    // The packet buffer owns the payload of the packet.
>+    uint8_t payload_size;
>+    helper.CopyTo(&payload_size);
>+    packet.sizeBytes = payload_size;
>+    packet.dataPtr = new uint8_t[payload_size];
> 
>-  size_t i = kMinDataNeeded;
>-  while (i < size) {
>-    memcpy(&packet.seqNum, &data[i - kMinDataNeeded], 2);
>-    packet.is_first_packet_in_frame = data[i] & 1;
>-    packet.markerBit = data[i] & 2;
>     packet_buffer->InsertPacket(&packet);
>-    i += kMinDataNeeded;
>   }
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/pseudotcp_parser_fuzzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/pseudotcp_parser_fuzzer.cc
>index 90123c3206c..bab09df715c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/pseudotcp_parser_fuzzer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/pseudotcp_parser_fuzzer.cc
>@@ -1,12 +1,12 @@
> /*
>-*  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
>-*
>-*  Use of this source code is governed by a BSD-style license
>-*  that can be found in the LICENSE file in the root of the source
>-*  tree. An additional intellectual property rights grant can be found
>-*  in the file PATENTS.  All contributing project authors may
>-*  be found in the AUTHORS file in the root of the source tree.
>-*/
>+ *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
> 
> #include <stddef.h>
> #include <stdint.h>
>@@ -30,9 +30,8 @@ class FakeIPseudoTcpNotify : public cricket::IPseudoTcpNotify {
> };
> 
> struct Environment {
>-  explicit Environment(cricket::IPseudoTcpNotify* notifier):
>-      ptcp(notifier, 0) {
>-  }
>+  explicit Environment(cricket::IPseudoTcpNotify* notifier)
>+      : ptcp(notifier, 0) {}
> 
>   // We need the thread to avoid some uninteresting crashes, since the
>   // production code expects there to be a thread object available.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/residual_echo_detector_fuzzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/residual_echo_detector_fuzzer.cc
>index c3c81237c8d..b0405ee9e19 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/residual_echo_detector_fuzzer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/residual_echo_detector_fuzzer.cc
>@@ -17,6 +17,7 @@
> 
> #include "modules/audio_processing/residual_echo_detector.h"
> #include "rtc_base/checks.h"
>+#include "rtc_base/refcountedobject.h"
> 
> namespace webrtc {
> 
>@@ -42,10 +43,11 @@ void FuzzOneInput(const uint8_t* data, size_t size) {
>   read_idx += 2;
>   std::bitset<16> call_order(call_order_int);
> 
>-  ResidualEchoDetector echo_detector;
>+  rtc::scoped_refptr<ResidualEchoDetector> echo_detector =
>+      new rtc::RefCountedObject<ResidualEchoDetector>();
>   std::vector<float> input(1);
>   // Call AnalyzeCaptureAudio once to prevent the flushing of the buffer.
>-  echo_detector.AnalyzeCaptureAudio(input);
>+  echo_detector->AnalyzeCaptureAudio(input);
>   for (size_t i = 0; i < 2 * kNrOfUpdates; ++i) {
>     // Convert 4 input bytes to a float.
>     RTC_DCHECK_LE(read_idx + sizeof(float), size);
>@@ -56,9 +58,9 @@ void FuzzOneInput(const uint8_t* data, size_t size) {
>       continue;
>     }
>     if (call_order[i]) {
>-      echo_detector.AnalyzeRenderAudio(input);
>+      echo_detector->AnalyzeRenderAudio(input);
>     } else {
>-      echo_detector.AnalyzeCaptureAudio(input);
>+      echo_detector->AnalyzeCaptureAudio(input);
>     }
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/rtcp_receiver_fuzzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/rtcp_receiver_fuzzer.cc
>index 8782f0c0a84..d87e002f679 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/rtcp_receiver_fuzzer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/rtcp_receiver_fuzzer.cc
>@@ -18,11 +18,11 @@ class NullModuleRtpRtcp : public RTCPReceiver::ModuleRtpRtcp {
>  public:
>   void SetTmmbn(std::vector<rtcp::TmmbItem>) override {}
>   void OnRequestSendReport() override {}
>-  void OnReceivedNack(const std::vector<uint16_t>&) override {};
>-  void OnReceivedRtcpReportBlocks(const ReportBlockList&) override {};
>+  void OnReceivedNack(const std::vector<uint16_t>&) override{};
>+  void OnReceivedRtcpReportBlocks(const ReportBlockList&) override{};
> };
> 
>-}
>+}  // namespace
> 
> void FuzzOneInput(const uint8_t* data, size_t size) {
>   NullModuleRtpRtcp rtp_rtcp_module;
>@@ -34,4 +34,3 @@ void FuzzOneInput(const uint8_t* data, size_t size) {
>   receiver.IncomingPacket(data, size);
> }
> }  // namespace webrtc
>-
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/rtp_frame_reference_finder_fuzzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/rtp_frame_reference_finder_fuzzer.cc
>new file mode 100644
>index 00000000000..fd98892a048
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/rtp_frame_reference_finder_fuzzer.cc
>@@ -0,0 +1,116 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/video_coding/rtp_frame_reference_finder.h"
>+
>+#include "absl/memory/memory.h"
>+#include "modules/video_coding/frame_object.h"
>+#include "modules/video_coding/packet_buffer.h"
>+#include "system_wrappers/include/clock.h"
>+
>+namespace webrtc {
>+
>+namespace {
>+struct DataReader {
>+  DataReader(const uint8_t* data, size_t size) : data_(data), size_(size) {}
>+
>+  void CopyTo(void* destination, size_t size) {
>+    uint8_t* dest = reinterpret_cast<uint8_t*>(destination);
>+    size_t num_bytes = std::min(size_ - offset_, size);
>+    memcpy(dest, data_ + offset_, num_bytes);
>+    offset_ += num_bytes;
>+
>+    size -= num_bytes;
>+    if (size > 0)
>+      memset(dest + num_bytes, 0, size);
>+  }
>+
>+  template <typename T>
>+  T GetNum() {
>+    T res;
>+    if (offset_ + sizeof(res) < size_) {
>+      memcpy(&res, data_ + offset_, sizeof(res));
>+      offset_ += sizeof(res);
>+      return res;
>+    }
>+
>+    offset_ = size_;
>+    return T(0);
>+  }
>+
>+  bool MoreToRead() { return offset_ < size_; }
>+
>+  const uint8_t* data_;
>+  size_t size_;
>+  size_t offset_ = 0;
>+};
>+
>+class NullCallback : public video_coding::OnCompleteFrameCallback {
>+  void OnCompleteFrame(
>+      std::unique_ptr<video_coding::EncodedFrame> frame) override {}
>+};
>+
>+class FuzzyPacketBuffer : public video_coding::PacketBuffer {
>+ public:
>+  explicit FuzzyPacketBuffer(DataReader* reader)
>+      : PacketBuffer(nullptr, 2, 4, nullptr), reader(reader) {
>+    switch (reader->GetNum<uint8_t>() % 3) {
>+      case 0:
>+        codec = kVideoCodecVP8;
>+        break;
>+      case 1:
>+        codec = kVideoCodecVP9;
>+        break;
>+      case 2:
>+        codec = kVideoCodecH264;
>+        break;
>+    }
>+  }
>+
>+  VCMPacket* GetPacket(uint16_t seq_num) override {
>+    auto packet_it = packets.find(seq_num);
>+    if (packet_it != packets.end())
>+      return &packet_it->second;
>+
>+    VCMPacket* packet = &packets[seq_num];
>+    packet->codec = codec;
>+    packet->markerBit = true;
>+    reader->CopyTo(packet, sizeof(packet));
>+    return packet;
>+  }
>+
>+  bool GetBitstream(const video_coding::RtpFrameObject& frame,
>+                    uint8_t* destination) override {
>+    return true;
>+  }
>+
>+  void ReturnFrame(video_coding::RtpFrameObject* frame) override {}
>+
>+ private:
>+  std::map<uint16_t, VCMPacket> packets;
>+  VideoCodecType codec;
>+  DataReader* const reader;
>+};
>+}  // namespace
>+
>+void FuzzOneInput(const uint8_t* data, size_t size) {
>+  DataReader reader(data, size);
>+  rtc::scoped_refptr<FuzzyPacketBuffer> pb(new FuzzyPacketBuffer(&reader));
>+  NullCallback cb;
>+  video_coding::RtpFrameReferenceFinder reference_finder(&cb);
>+
>+  while (reader.MoreToRead()) {
>+    auto frame = absl::make_unique<video_coding::RtpFrameObject>(
>+        pb, reader.GetNum<uint16_t>(), reader.GetNum<uint16_t>(), 0, 0, 0);
>+    reference_finder.ManageFrame(std::move(frame));
>+  }
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/rtp_packet_fuzzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/rtp_packet_fuzzer.cc
>index c89cb46dee0..0e35c8ee41a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/rtp_packet_fuzzer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/rtp_packet_fuzzer.cc
>@@ -9,6 +9,7 @@
>  */
> 
> #include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
>+#include "modules/rtp_rtcp/source/rtp_generic_frame_descriptor_extension.h"
> #include "modules/rtp_rtcp/source/rtp_header_extensions.h"
> #include "modules/rtp_rtcp/source/rtp_packet_received.h"
> 
>@@ -109,6 +110,11 @@ void FuzzOneInput(const uint8_t* data, size_t size) {
>         packet.GetExtension<RtpMid>(&mid);
>         break;
>       }
>+      case kRtpExtensionGenericFrameDescriptor: {
>+        RtpGenericFrameDescriptor descriptor;
>+        packet.GetExtension<RtpGenericFrameDescriptorExtension>(&descriptor);
>+        break;
>+      }
>     }
>   }
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/transport_feedback_packet_loss_tracker_fuzzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/transport_feedback_packet_loss_tracker_fuzzer.cc
>index 31bc1639e2e..2168d219f9d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/transport_feedback_packet_loss_tracker_fuzzer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/transport_feedback_packet_loss_tracker_fuzzer.cc
>@@ -11,10 +11,10 @@
> #include <algorithm>
> 
> #include "api/array_view.h"
>+#include "audio/transport_feedback_packet_loss_tracker.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> #include "modules/rtp_rtcp/source/byte_io.h"
> #include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
>-#include "voice_engine/transport_feedback_packet_loss_tracker.h"
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/ulpfec_generator_fuzzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/ulpfec_generator_fuzzer.cc
>index 03728f6bacb..ce9d8fdbc8d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/ulpfec_generator_fuzzer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/ulpfec_generator_fuzzer.cc
>@@ -10,6 +10,7 @@
> 
> #include <memory>
> 
>+#include "modules/include/module_common_types_public.h"
> #include "modules/rtp_rtcp/source/byte_io.h"
> #include "modules/rtp_rtcp/source/fec_test_helper.h"
> #include "modules/rtp_rtcp/source/ulpfec_generator.h"
>@@ -31,7 +32,7 @@ void FuzzOneInput(const uint8_t* data, size_t size) {
>       data[i++] % 128, static_cast<int>(data[i++] % 10), kFecMaskBursty};
>   generator.SetFecParameters(params);
>   uint16_t seq_num = data[i++];
>-
>+  uint16_t prev_seq_num = 0;
>   while (i + 3 < size) {
>     size_t rtp_header_length = data[i++] % 10 + 12;
>     size_t payload_size = data[i++] % 10;
>@@ -40,19 +41,26 @@ void FuzzOneInput(const uint8_t* data, size_t size) {
>     std::unique_ptr<uint8_t[]> packet(
>         new uint8_t[payload_size + rtp_header_length]);
>     memcpy(packet.get(), &data[i], payload_size + rtp_header_length);
>+
>+    // Make sure sequence numbers are increasing.
>     ByteWriter<uint16_t>::WriteBigEndian(&packet[2], seq_num++);
>     i += payload_size + rtp_header_length;
>-    // Make sure sequence numbers are increasing.
>     const bool protect = data[i++] % 2 == 1;
>-    if (protect) {
>+
>+    // Check the sequence numbers are monotonic. In rare case the packets number
>+    // may loop around and in the same FEC-protected group the packet sequence
>+    // number became out of order.
>+    if (protect && IsNewerSequenceNumber(seq_num, prev_seq_num) &&
>+        seq_num < prev_seq_num + kUlpfecMaxMediaPackets) {
>       generator.AddRtpPacketAndGenerateFec(packet.get(), payload_size,
>                                            rtp_header_length);
>+      prev_seq_num = seq_num;
>     }
>     const size_t num_fec_packets = generator.NumAvailableFecPackets();
>     if (num_fec_packets > 0) {
>       std::vector<std::unique_ptr<RedPacket>> fec_packets =
>-          generator.GetUlpfecPacketsAsRed(kRedPayloadType, kFecPayloadType, 100,
>-                                          rtp_header_length);
>+          generator.GetUlpfecPacketsAsRed(kRedPayloadType, kFecPayloadType,
>+                                          100);
>       RTC_CHECK_EQ(num_fec_packets, fec_packets.size());
>     }
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/ulpfec_receiver_fuzzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/ulpfec_receiver_fuzzer.cc
>index c850fec996a..6b74f3ac95d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/ulpfec_receiver_fuzzer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/ulpfec_receiver_fuzzer.cc
>@@ -10,11 +10,10 @@
> 
> #include <algorithm>
> 
>-#include "modules/rtp_rtcp/include/ulpfec_receiver.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>+#include "modules/rtp_rtcp/include/ulpfec_receiver.h"
> #include "modules/rtp_rtcp/source/byte_io.h"
> #include "modules/rtp_rtcp/source/rtp_packet_received.h"
>-#include "rtc_base/basictypes.h"
> 
> namespace webrtc {
> 
>@@ -68,8 +67,8 @@ void FuzzOneInput(const uint8_t* data, size_t size) {
>     RTPHeader parsed_header;
>     if (parsed_packet.Parse(packet.get(), packet_length)) {
>       parsed_packet.GetHeader(&parsed_header);
>-      receiver->AddReceivedRedPacket(parsed_header, packet.get(),
>-                                     packet_length, 0);
>+      receiver->AddReceivedRedPacket(parsed_header, packet.get(), packet_length,
>+                                     0);
>     }
>   }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/webrtc_fuzzer_main.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/webrtc_fuzzer_main.cc
>index ca4bc15c710..a52dd231beb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/webrtc_fuzzer_main.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/fuzzers/webrtc_fuzzer_main.cc
>@@ -20,21 +20,21 @@ void InitializeWebRtcFuzzDefaults() {
>   if (g_initialized)
>     return;
> 
>-  // Remove default logging to prevent huge slowdowns.
>-  // TODO(pbos): Disable in Chromium: http://crbug.com/561667
>+// Remove default logging to prevent huge slowdowns.
>+// TODO(pbos): Disable in Chromium: http://crbug.com/561667
> #if !defined(WEBRTC_CHROMIUM_BUILD)
>   rtc::LogMessage::LogToDebug(rtc::LS_NONE);
> #endif  // !defined(WEBRTC_CHROMIUM_BUILD)
> 
>   g_initialized = true;
> }
>-}
>+}  // namespace
> 
> namespace webrtc {
> extern void FuzzOneInput(const uint8_t* data, size_t size);
> }  // namespace webrtc
> 
>-extern "C" int LLVMFuzzerTestOneInput(const unsigned char *data, size_t size) {
>+extern "C" int LLVMFuzzerTestOneInput(const unsigned char* data, size_t size) {
>   InitializeWebRtcFuzzDefaults();
>   webrtc::FuzzOneInput(data, size);
>   return 0;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/gl/gl_renderer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/gl/gl_renderer.cc
>index ef7ab9c1262..10162ee056d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/gl/gl_renderer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/gl/gl_renderer.cc
>@@ -108,5 +108,5 @@ void GlRenderer::OnFrame(const webrtc::VideoFrame& frame) {
>   glBindTexture(GL_TEXTURE_2D, 0);
>   glFlush();
> }
>-}  // test
>-}  // webrtc
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/gl/gl_renderer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/gl/gl_renderer.h
>index 382739d307d..531e3f56ce2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/gl/gl_renderer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/gl/gl_renderer.h
>@@ -18,8 +18,6 @@
> #endif
> 
> #include "test/video_renderer.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>-
> 
> namespace webrtc {
> namespace test {
>@@ -44,7 +42,7 @@ class GlRenderer : public VideoRenderer {
> 
>   void ResizeVideo(size_t width, size_t height);
> };
>-}  // test
>-}  // webrtc
>+}  // namespace test
>+}  // namespace webrtc
> 
> #endif  // TEST_GL_GL_RENDERER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/gtest.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/gtest.h
>index 928ec496f08..fa4396420e4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/gtest.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/gtest.h
>@@ -14,6 +14,7 @@
> #include "rtc_base/ignore_wundef.h"
> 
> RTC_PUSH_IGNORING_WUNDEF()
>+#include "testing/gtest/include/gtest/gtest-spi.h"
> #include "testing/gtest/include/gtest/gtest.h"
> RTC_POP_IGNORING_WUNDEF()
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/ios/test_support.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/ios/test_support.h
>index 7bc2993b9e2..2c5b600ce80 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/ios/test_support.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/ios/test_support.h
>@@ -17,7 +17,10 @@ namespace test {
> // This is necessary as iOS doesn't like processes without a gui
> // running for longer than a few seconds.
> void RunTestsFromIOSApp();
>-void InitTestSuite(int (*test_suite)(void), int argc, char* argv[]);
>+void InitTestSuite(int (*test_suite)(void),
>+                   int argc,
>+                   char* argv[],
>+                   bool save_chartjson_result);
> 
> }  // namespace test
> }  // namespace rtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/ios/test_support.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/test/ios/test_support.mm
>index 603bc8c19c7..5b7f2f93505 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/ios/test_support.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/ios/test_support.mm
>@@ -11,8 +11,10 @@
> #import <UIKit/UIKit.h>
> 
> #include "test/ios/test_support.h"
>+#include "test/testsupport/perf_test.h"
> 
> #import "sdk/objc/Framework/Classes/Common/RTCUIApplicationStatusObserver.h"
>+#include "sdk/objc/Framework/Classes/Common/helpers.h"
> 
> // Springboard will kill any iOS app that fails to check in after launch within
> // a given time. Starting a UIApplication before invoking TestSuite::Run
>@@ -31,6 +33,7 @@
> static int (*g_test_suite)(void) = NULL;
> static int g_argc;
> static char **g_argv;
>+static bool g_save_chartjson_result;
> 
> @interface UIApplication (Testing)
> - (void)_terminateWithStatus:(int)status;
>@@ -75,6 +78,20 @@ - (BOOL)application:(UIApplication *)application
> - (void)runTests {
>   int exitStatus = g_test_suite();
> 
>+  if (g_save_chartjson_result) {
>+    // Stores data into a json file under the app's document directory.
>+    NSString* fileName = @"perf_result.json";
>+    NSArray<NSString*>* outputDirectories = NSSearchPathForDirectoriesInDomains(
>+        NSDocumentDirectory, NSUserDomainMask, YES);
>+    if ([outputDirectories count] != 0) {
>+      NSString* outputPath =
>+          [outputDirectories[0] stringByAppendingPathComponent:fileName];
>+
>+      webrtc::test::WritePerfResults(
>+          webrtc::ios::StdStringFromNSString(outputPath));
>+    }
>+  }
>+
>   // If a test app is too fast, it will exit before Instruments has has a
>   // a chance to initialize and no test results will be seen.
>   // TODO(crbug.com/137010): Figure out how much time is actually needed, and
>@@ -93,10 +110,14 @@ @end
> namespace rtc {
> namespace test {
> 
>-void InitTestSuite(int (*test_suite)(void), int argc, char *argv[]) {
>+// Note: This is not thread safe, and must be called from the same thread as
>+// runTests above.
>+void InitTestSuite(int (*test_suite)(void), int argc, char *argv[],
>+                   bool save_chartjson_result) {
>   g_test_suite = test_suite;
>   g_argc = argc;
>   g_argv = argv;
>+  g_save_chartjson_result = save_chartjson_result;
> }
> 
> void RunTestsFromIOSApp() {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/layer_filtering_transport.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/layer_filtering_transport.cc
>index a8a337da650..1f36619bddc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/layer_filtering_transport.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/layer_filtering_transport.cc
>@@ -20,6 +20,46 @@
> namespace webrtc {
> namespace test {
> 
>+LayerFilteringTransport::LayerFilteringTransport(
>+    SingleThreadedTaskQueueForTesting* task_queue,
>+    const FakeNetworkPipe::Config& config,
>+    Call* send_call,
>+    uint8_t vp8_video_payload_type,
>+    uint8_t vp9_video_payload_type,
>+    int selected_tl,
>+    int selected_sl,
>+    const std::map<uint8_t, MediaType>& payload_type_map,
>+    uint32_t ssrc_to_filter_min,
>+    uint32_t ssrc_to_filter_max)
>+    : DirectTransport(task_queue, config, send_call, payload_type_map),
>+      vp8_video_payload_type_(vp8_video_payload_type),
>+      vp9_video_payload_type_(vp9_video_payload_type),
>+      selected_tl_(selected_tl),
>+      selected_sl_(selected_sl),
>+      discarded_last_packet_(false),
>+      ssrc_to_filter_min_(ssrc_to_filter_min),
>+      ssrc_to_filter_max_(ssrc_to_filter_max) {}
>+
>+LayerFilteringTransport::LayerFilteringTransport(
>+    SingleThreadedTaskQueueForTesting* task_queue,
>+    std::unique_ptr<FakeNetworkPipe> pipe,
>+    Call* send_call,
>+    uint8_t vp8_video_payload_type,
>+    uint8_t vp9_video_payload_type,
>+    int selected_tl,
>+    int selected_sl,
>+    const std::map<uint8_t, MediaType>& payload_type_map,
>+    uint32_t ssrc_to_filter_min,
>+    uint32_t ssrc_to_filter_max)
>+    : DirectTransport(task_queue, std::move(pipe), send_call, payload_type_map),
>+      vp8_video_payload_type_(vp8_video_payload_type),
>+      vp9_video_payload_type_(vp9_video_payload_type),
>+      selected_tl_(selected_tl),
>+      selected_sl_(selected_sl),
>+      discarded_last_packet_(false),
>+      ssrc_to_filter_min_(ssrc_to_filter_min),
>+      ssrc_to_filter_max_(ssrc_to_filter_max) {}
>+
> LayerFilteringTransport::LayerFilteringTransport(
>     SingleThreadedTaskQueueForTesting* task_queue,
>     const FakeNetworkPipe::Config& config,
>@@ -34,7 +74,9 @@ LayerFilteringTransport::LayerFilteringTransport(
>       vp9_video_payload_type_(vp9_video_payload_type),
>       selected_tl_(selected_tl),
>       selected_sl_(selected_sl),
>-      discarded_last_packet_(false) {}
>+      discarded_last_packet_(false),
>+      ssrc_to_filter_min_(0),
>+      ssrc_to_filter_max_(0xFFFFFFFF) {}
> 
> LayerFilteringTransport::LayerFilteringTransport(
>     SingleThreadedTaskQueueForTesting* task_queue,
>@@ -43,13 +85,16 @@ LayerFilteringTransport::LayerFilteringTransport(
>     uint8_t vp8_video_payload_type,
>     uint8_t vp9_video_payload_type,
>     int selected_tl,
>-    int selected_sl)
>-    : DirectTransport(task_queue, std::move(pipe), send_call),
>+    int selected_sl,
>+    const std::map<uint8_t, MediaType>& payload_type_map)
>+    : DirectTransport(task_queue, std::move(pipe), send_call, payload_type_map),
>       vp8_video_payload_type_(vp8_video_payload_type),
>       vp9_video_payload_type_(vp9_video_payload_type),
>       selected_tl_(selected_tl),
>       selected_sl_(selected_sl),
>-      discarded_last_packet_(false) {}
>+      discarded_last_packet_(false),
>+      ssrc_to_filter_min_(0),
>+      ssrc_to_filter_max_(0xFFFFFFFF) {}
> 
> bool LayerFilteringTransport::DiscardedLastPacket() const {
>   return discarded_last_packet_;
>@@ -68,6 +113,11 @@ bool LayerFilteringTransport::SendRtp(const uint8_t* packet,
>   RTPHeader header;
>   parser.Parse(&header);
> 
>+  if (header.ssrc < ssrc_to_filter_min_ || header.ssrc > ssrc_to_filter_max_) {
>+    // Nothing to change, forward the packet immediately.
>+    return test::DirectTransport::SendRtp(packet, length, options);
>+  }
>+
>   RTC_DCHECK_LE(length, IP_PACKET_SIZE);
>   uint8_t temp_buffer[IP_PACKET_SIZE];
>   memcpy(temp_buffer, packet, length);
>@@ -82,29 +132,73 @@ bool LayerFilteringTransport::SendRtp(const uint8_t* packet,
> 
>     const bool is_vp8 = header.payloadType == vp8_video_payload_type_;
>     std::unique_ptr<RtpDepacketizer> depacketizer(
>-        RtpDepacketizer::Create(is_vp8 ? kRtpVideoVp8 : kRtpVideoVp9));
>+        RtpDepacketizer::Create(is_vp8 ? kVideoCodecVP8 : kVideoCodecVP9));
>     RtpDepacketizer::ParsedPayload parsed_payload;
>     if (depacketizer->Parse(&parsed_payload, payload, payload_data_length)) {
>-      const int temporal_idx = static_cast<int>(
>-          is_vp8 ? parsed_payload.type.Video.codecHeader.VP8.temporalIdx
>-                 : parsed_payload.type.Video.codecHeader.VP9.temporal_idx);
>-      const int spatial_idx = static_cast<int>(
>-          is_vp8 ? kNoSpatialIdx
>-                 : parsed_payload.type.Video.codecHeader.VP9.spatial_idx);
>-      if (selected_sl_ >= 0 && spatial_idx == selected_sl_ &&
>-          parsed_payload.type.Video.codecHeader.VP9.end_of_frame) {
>+      int temporal_idx;
>+      int spatial_idx;
>+      bool non_ref_for_inter_layer_pred;
>+      bool end_of_frame;
>+
>+      if (is_vp8) {
>+        temporal_idx = parsed_payload.video_header().vp8().temporalIdx;
>+        spatial_idx = kNoSpatialIdx;
>+        num_active_spatial_layers_ = 1;
>+        non_ref_for_inter_layer_pred = false;
>+        end_of_frame = true;
>+      } else {
>+        const auto& vp9_header = absl::get<RTPVideoHeaderVP9>(
>+            parsed_payload.video_header().video_type_header);
>+        temporal_idx = vp9_header.temporal_idx;
>+        spatial_idx = vp9_header.spatial_idx;
>+        non_ref_for_inter_layer_pred = vp9_header.non_ref_for_inter_layer_pred;
>+        end_of_frame = vp9_header.end_of_frame;
>+
>+        // The number of spatial layers is sent in ssData, which is included
>+        // only in the first packet of the first spatial layer of a key frame.
>+        if (!vp9_header.inter_pic_predicted &&
>+            vp9_header.beginning_of_frame == 1 && spatial_idx == 0) {
>+          num_active_spatial_layers_ = vp9_header.num_spatial_layers;
>+        }
>+      }
>+
>+      if (spatial_idx == kNoSpatialIdx)
>+        num_active_spatial_layers_ = 1;
>+
>+      RTC_CHECK_GT(num_active_spatial_layers_, 0);
>+
>+      if (selected_sl_ >= 0 &&
>+          spatial_idx ==
>+              std::min(num_active_spatial_layers_ - 1, selected_sl_) &&
>+          end_of_frame) {
>         // This layer is now the last in the superframe.
>         set_marker_bit = true;
>-      } else if ((selected_tl_ >= 0 && temporal_idx != kNoTemporalIdx &&
>-                  temporal_idx > selected_tl_) ||
>-                 (selected_sl_ >= 0 && spatial_idx != kNoSpatialIdx &&
>-                  spatial_idx > selected_sl_)) {
>-        // Truncate packet to a padding packet.
>-        length = header.headerLength + 1;
>-        temp_buffer[0] |= (1 << 5);  // P = 1.
>-        temp_buffer[1] &= 0x7F;      // M = 0.
>-        discarded_last_packet_ = true;
>-        temp_buffer[header.headerLength] = 1;  // One byte of padding.
>+      } else {
>+        const bool higher_temporal_layer =
>+            (selected_tl_ >= 0 && temporal_idx != kNoTemporalIdx &&
>+             temporal_idx > selected_tl_);
>+
>+        const bool higher_spatial_layer =
>+            (selected_sl_ >= 0 && spatial_idx != kNoSpatialIdx &&
>+             spatial_idx > selected_sl_);
>+
>+        // Filter out non-reference lower spatial layers since they are not
>+        // needed for decoding of target spatial layer.
>+        const bool lower_non_ref_spatial_layer =
>+            (selected_sl_ >= 0 && spatial_idx != kNoSpatialIdx &&
>+             spatial_idx <
>+                 std::min(num_active_spatial_layers_ - 1, selected_sl_) &&
>+             non_ref_for_inter_layer_pred);
>+
>+        if (higher_temporal_layer || higher_spatial_layer ||
>+            lower_non_ref_spatial_layer) {
>+          // Truncate packet to a padding packet.
>+          length = header.headerLength + 1;
>+          temp_buffer[0] |= (1 << 5);  // P = 1.
>+          temp_buffer[1] &= 0x7F;      // M = 0.
>+          discarded_last_packet_ = true;
>+          temp_buffer[header.headerLength] = 1;  // One byte of padding.
>+        }
>       }
>     } else {
>       RTC_NOTREACHED() << "Parse error";
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/layer_filtering_transport.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/layer_filtering_transport.h
>index fe3b3d6d707..bd5f2e8ab9e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/layer_filtering_transport.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/layer_filtering_transport.h
>@@ -10,19 +10,29 @@
> #ifndef TEST_LAYER_FILTERING_TRANSPORT_H_
> #define TEST_LAYER_FILTERING_TRANSPORT_H_
> 
>+#include <map>
>+
> #include "call/call.h"
>+#include "call/fake_network_pipe.h"
> #include "test/direct_transport.h"
>-#include "test/fake_network_pipe.h"
> #include "test/single_threaded_task_queue.h"
> 
>-#include <map>
>-
> namespace webrtc {
> 
> namespace test {
> 
> class LayerFilteringTransport : public test::DirectTransport {
>  public:
>+  LayerFilteringTransport(SingleThreadedTaskQueueForTesting* task_queue,
>+                          const FakeNetworkPipe::Config& config,
>+                          Call* send_call,
>+                          uint8_t vp8_video_payload_type,
>+                          uint8_t vp9_video_payload_type,
>+                          int selected_tl,
>+                          int selected_sl,
>+                          const std::map<uint8_t, MediaType>& payload_type_map,
>+                          uint32_t ssrc_to_filter_min,
>+                          uint32_t ssrc_to_filter_max);
>   LayerFilteringTransport(SingleThreadedTaskQueueForTesting* task_queue,
>                           const FakeNetworkPipe::Config& config,
>                           Call* send_call,
>@@ -37,7 +47,18 @@ class LayerFilteringTransport : public test::DirectTransport {
>                           uint8_t vp8_video_payload_type,
>                           uint8_t vp9_video_payload_type,
>                           int selected_tl,
>-                          int selected_sl);
>+                          int selected_sl,
>+                          const std::map<uint8_t, MediaType>& payload_type_map,
>+                          uint32_t ssrc_to_filter_min,
>+                          uint32_t ssrc_to_filter_max);
>+  LayerFilteringTransport(SingleThreadedTaskQueueForTesting* task_queue,
>+                          std::unique_ptr<FakeNetworkPipe> pipe,
>+                          Call* send_call,
>+                          uint8_t vp8_video_payload_type,
>+                          uint8_t vp9_video_payload_type,
>+                          int selected_tl,
>+                          int selected_sl,
>+                          const std::map<uint8_t, MediaType>& payload_type_map);
>   bool DiscardedLastPacket() const;
>   bool SendRtp(const uint8_t* data,
>                size_t length,
>@@ -52,6 +73,9 @@ class LayerFilteringTransport : public test::DirectTransport {
>   const int selected_tl_;
>   const int selected_sl_;
>   bool discarded_last_packet_;
>+  int num_active_spatial_layers_;
>+  const uint32_t ssrc_to_filter_min_;
>+  const uint32_t ssrc_to_filter_max_;
> };
> 
> }  // namespace test
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/linux/glx_renderer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/linux/glx_renderer.cc
>index 4dbf1c97d9e..aaff84bc8f5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/linux/glx_renderer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/linux/glx_renderer.cc
>@@ -22,15 +22,14 @@ namespace webrtc {
> namespace test {
> 
> GlxRenderer::GlxRenderer(size_t width, size_t height)
>-    : width_(width),
>-      height_(height),
>-      display_(NULL),
>-      context_(NULL) {
>+    : width_(width), height_(height), display_(NULL), context_(NULL) {
>   assert(width > 0);
>   assert(height > 0);
> }
> 
>-GlxRenderer::~GlxRenderer() { Destroy(); }
>+GlxRenderer::~GlxRenderer() {
>+  Destroy();
>+}
> 
> bool GlxRenderer::Init(const char* window_title) {
>   if ((display_ = XOpenDisplay(NULL)) == NULL) {
>@@ -41,9 +40,10 @@ bool GlxRenderer::Init(const char* window_title) {
>   int screen = DefaultScreen(display_);
> 
>   XVisualInfo* vi;
>-  int attr_list[] = { GLX_DOUBLEBUFFER, GLX_RGBA, GLX_RED_SIZE, 4,
>-                      GLX_GREEN_SIZE, 4, GLX_BLUE_SIZE, 4, GLX_DEPTH_SIZE, 16,
>-                      None, };
>+  int attr_list[] = {
>+      GLX_DOUBLEBUFFER, GLX_RGBA, GLX_RED_SIZE,   4,  GLX_GREEN_SIZE, 4,
>+      GLX_BLUE_SIZE,    4,        GLX_DEPTH_SIZE, 16, None,
>+  };
> 
>   if ((vi = glXChooseVisual(display_, screen, attr_list)) == NULL) {
>     Destroy();
>@@ -106,7 +106,8 @@ void GlxRenderer::Destroy() {
>   }
> }
> 
>-GlxRenderer* GlxRenderer::Create(const char* window_title, size_t width,
>+GlxRenderer* GlxRenderer::Create(const char* window_title,
>+                                 size_t width,
>                                  size_t height) {
>   GlxRenderer* glx_renderer = new GlxRenderer(width, height);
>   if (!glx_renderer->Init(window_title)) {
>@@ -174,5 +175,5 @@ void GlxRenderer::OnFrame(const webrtc::VideoFrame& frame) {
>     abort();
>   }
> }
>-}  // test
>-}  // webrtc
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/linux/glx_renderer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/linux/glx_renderer.h
>index 63ef6242629..0c486df97f8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/linux/glx_renderer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/linux/glx_renderer.h
>@@ -15,14 +15,14 @@
> #include <X11/Xlib.h>
> 
> #include "test/gl/gl_renderer.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> namespace test {
> 
> class GlxRenderer : public GlRenderer {
>  public:
>-  static GlxRenderer* Create(const char* window_title, size_t width,
>+  static GlxRenderer* Create(const char* window_title,
>+                             size_t width,
>                              size_t height);
>   virtual ~GlxRenderer();
> 
>@@ -41,7 +41,7 @@ class GlxRenderer : public GlRenderer {
>   Window window_;
>   GLXContext context_;
> };
>-}  // test
>-}  // webrtc
>+}  // namespace test
>+}  // namespace webrtc
> 
> #endif  // TEST_LINUX_GLX_RENDERER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/linux/video_renderer_linux.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/linux/video_renderer_linux.cc
>index 942848942fa..da0b792a07b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/linux/video_renderer_linux.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/linux/video_renderer_linux.cc
>@@ -23,5 +23,5 @@ VideoRenderer* VideoRenderer::CreatePlatformRenderer(const char* window_title,
>   }
>   return NULL;
> }
>-}  // test
>-}  // webrtc
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/mac/video_renderer_mac.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/mac/video_renderer_mac.h
>index 19349c037ac..e6499f07ffb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/mac/video_renderer_mac.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/mac/video_renderer_mac.h
>@@ -34,7 +34,7 @@ class MacRenderer : public GlRenderer {
> 
>   RTC_DISALLOW_COPY_AND_ASSIGN(MacRenderer);
> };
>-}  // test
>-}  // webrtc
>+}  // namespace test
>+}  // namespace webrtc
> 
> #endif  // TEST_MAC_VIDEO_RENDERER_MAC_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/mock_audio_decoder_factory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/mock_audio_decoder_factory.h
>index bc504d871e2..247f9fa8d44 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/mock_audio_decoder_factory.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/mock_audio_decoder_factory.h
>@@ -26,13 +26,16 @@ class MockAudioDecoderFactory : public AudioDecoderFactory {
>  public:
>   MOCK_METHOD0(GetSupportedDecoders, std::vector<AudioCodecSpec>());
>   MOCK_METHOD1(IsSupportedDecoder, bool(const SdpAudioFormat&));
>-  std::unique_ptr<AudioDecoder> MakeAudioDecoder(const SdpAudioFormat& format) {
>+  std::unique_ptr<AudioDecoder> MakeAudioDecoder(
>+      const SdpAudioFormat& format,
>+      absl::optional<AudioCodecPairId> codec_pair_id) {
>     std::unique_ptr<AudioDecoder> return_value;
>-    MakeAudioDecoderMock(format, &return_value);
>+    MakeAudioDecoderMock(format, codec_pair_id, &return_value);
>     return return_value;
>   }
>-  MOCK_METHOD2(MakeAudioDecoderMock,
>+  MOCK_METHOD3(MakeAudioDecoderMock,
>                void(const SdpAudioFormat& format,
>+                    absl::optional<AudioCodecPairId> codec_pair_id,
>                     std::unique_ptr<AudioDecoder>* return_value));
> 
>   // Creates a MockAudioDecoderFactory with no formats and that may not be
>@@ -51,7 +54,7 @@ class MockAudioDecoderFactory : public AudioDecoderFactory {
>     EXPECT_CALL(*factory.get(), GetSupportedDecoders()).Times(AnyNumber());
>     ON_CALL(*factory, IsSupportedDecoder(_)).WillByDefault(Return(false));
>     EXPECT_CALL(*factory, IsSupportedDecoder(_)).Times(AnyNumber());
>-    EXPECT_CALL(*factory.get(), MakeAudioDecoderMock(_, _)).Times(0);
>+    EXPECT_CALL(*factory.get(), MakeAudioDecoderMock(_, _, _)).Times(0);
>     return factory;
>   }
> 
>@@ -72,9 +75,10 @@ class MockAudioDecoderFactory : public AudioDecoderFactory {
>     EXPECT_CALL(*factory.get(), GetSupportedDecoders()).Times(AnyNumber());
>     ON_CALL(*factory, IsSupportedDecoder(_)).WillByDefault(Return(false));
>     EXPECT_CALL(*factory, IsSupportedDecoder(_)).Times(AnyNumber());
>-    ON_CALL(*factory.get(), MakeAudioDecoderMock(_, _))
>-        .WillByDefault(SetArgPointee<1>(nullptr));
>-    EXPECT_CALL(*factory.get(), MakeAudioDecoderMock(_, _)).Times(AnyNumber());
>+    ON_CALL(*factory.get(), MakeAudioDecoderMock(_, _, _))
>+        .WillByDefault(SetArgPointee<2>(nullptr));
>+    EXPECT_CALL(*factory.get(), MakeAudioDecoderMock(_, _, _))
>+        .Times(AnyNumber());
>     return factory;
>   }
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/mock_audio_encoder.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/mock_audio_encoder.cc
>index cd89bdd4e8d..36615111a5a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/mock_audio_encoder.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/mock_audio_encoder.cc
>@@ -13,9 +13,7 @@
> namespace webrtc {
> 
> MockAudioEncoder::MockAudioEncoder() = default;
>-MockAudioEncoder::~MockAudioEncoder() {
>-  Die();
>-}
>+MockAudioEncoder::~MockAudioEncoder() = default;
> 
> MockAudioEncoder::FakeEncoding::FakeEncoding(
>     const AudioEncoder::EncodedInfo& info)
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/mock_audio_encoder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/mock_audio_encoder.h
>index 8196cf18a76..60425e0613d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/mock_audio_encoder.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/mock_audio_encoder.h
>@@ -27,7 +27,6 @@ class MockAudioEncoder : public AudioEncoder {
>   // http://crbug.com/428099.
>   MockAudioEncoder();
>   ~MockAudioEncoder();
>-  MOCK_METHOD0(Die, void());
>   MOCK_METHOD1(Mark, void(std::string desc));
>   MOCK_CONST_METHOD0(SampleRateHz, int());
>   MOCK_CONST_METHOD0(NumChannels, size_t());
>@@ -44,7 +43,7 @@ class MockAudioEncoder : public AudioEncoder {
>   MOCK_METHOD1(SetMaxPayloadSize, void(int max_payload_size_bytes));
>   MOCK_METHOD2(OnReceivedUplinkBandwidth,
>                void(int target_audio_bitrate_bps,
>-                    rtc::Optional<int64_t> probing_interval_ms));
>+                    absl::optional<int64_t> probing_interval_ms));
>   MOCK_METHOD1(OnReceivedUplinkPacketLossFraction,
>                void(float uplink_packet_loss_fraction));
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/mock_audio_encoder_factory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/mock_audio_encoder_factory.h
>index 9b63f7d18ff..340602c35fe 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/mock_audio_encoder_factory.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/mock_audio_encoder_factory.h
>@@ -15,26 +15,30 @@
> #include <vector>
> 
> #include "api/audio_codecs/audio_encoder_factory.h"
>+#include "rtc_base/refcountedobject.h"
> #include "rtc_base/scoped_ref_ptr.h"
> #include "test/gmock.h"
> 
> namespace webrtc {
> 
>-class MockAudioEncoderFactory : public AudioEncoderFactory {
>+class MockAudioEncoderFactory : public testing::NiceMock<AudioEncoderFactory> {
>  public:
>   MOCK_METHOD0(GetSupportedEncoders, std::vector<AudioCodecSpec>());
>   MOCK_METHOD1(QueryAudioEncoder,
>-               rtc::Optional<AudioCodecInfo>(const SdpAudioFormat& format));
>+               absl::optional<AudioCodecInfo>(const SdpAudioFormat& format));
> 
>-  std::unique_ptr<AudioEncoder> MakeAudioEncoder(int payload_type,
>-                                                 const SdpAudioFormat& format) {
>+  std::unique_ptr<AudioEncoder> MakeAudioEncoder(
>+      int payload_type,
>+      const SdpAudioFormat& format,
>+      absl::optional<AudioCodecPairId> codec_pair_id) {
>     std::unique_ptr<AudioEncoder> return_value;
>-    MakeAudioEncoderMock(payload_type, format, &return_value);
>+    MakeAudioEncoderMock(payload_type, format, codec_pair_id, &return_value);
>     return return_value;
>   }
>-  MOCK_METHOD3(MakeAudioEncoderMock,
>+  MOCK_METHOD4(MakeAudioEncoderMock,
>                void(int payload_type,
>                     const SdpAudioFormat& format,
>+                    absl::optional<AudioCodecPairId> codec_pair_id,
>                     std::unique_ptr<AudioEncoder>* return_value));
> 
>   // Creates a MockAudioEncoderFactory with no formats and that may not be
>@@ -51,11 +55,11 @@ class MockAudioEncoderFactory : public AudioEncoderFactory {
>     ON_CALL(*factory.get(), GetSupportedEncoders())
>         .WillByDefault(Return(std::vector<webrtc::AudioCodecSpec>()));
>     ON_CALL(*factory.get(), QueryAudioEncoder(_))
>-        .WillByDefault(Return(rtc::Optional<AudioCodecInfo>()));
>+        .WillByDefault(Return(absl::nullopt));
> 
>     EXPECT_CALL(*factory.get(), GetSupportedEncoders()).Times(AnyNumber());
>     EXPECT_CALL(*factory.get(), QueryAudioEncoder(_)).Times(AnyNumber());
>-    EXPECT_CALL(*factory.get(), MakeAudioEncoderMock(_, _, _)).Times(0);
>+    EXPECT_CALL(*factory.get(), MakeAudioEncoderMock(_, _, _, _)).Times(0);
>     return factory;
>   }
> 
>@@ -74,13 +78,13 @@ class MockAudioEncoderFactory : public AudioEncoderFactory {
>     ON_CALL(*factory.get(), GetSupportedEncoders())
>         .WillByDefault(Return(std::vector<webrtc::AudioCodecSpec>()));
>     ON_CALL(*factory.get(), QueryAudioEncoder(_))
>-        .WillByDefault(Return(rtc::Optional<AudioCodecInfo>()));
>-    ON_CALL(*factory.get(), MakeAudioEncoderMock(_, _, _))
>-        .WillByDefault(SetArgPointee<2>(nullptr));
>+        .WillByDefault(Return(absl::nullopt));
>+    ON_CALL(*factory.get(), MakeAudioEncoderMock(_, _, _, _))
>+        .WillByDefault(SetArgPointee<3>(nullptr));
> 
>     EXPECT_CALL(*factory.get(), GetSupportedEncoders()).Times(AnyNumber());
>     EXPECT_CALL(*factory.get(), QueryAudioEncoder(_)).Times(AnyNumber());
>-    EXPECT_CALL(*factory.get(), MakeAudioEncoderMock(_, _, _))
>+    EXPECT_CALL(*factory.get(), MakeAudioEncoderMock(_, _, _, _))
>         .Times(AnyNumber());
>     return factory;
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/mock_transport.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/mock_transport.cc
>new file mode 100644
>index 00000000000..3878b3bdbee
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/mock_transport.cc
>@@ -0,0 +1,18 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "test/mock_transport.h"
>+
>+namespace webrtc {
>+
>+MockTransport::MockTransport() = default;
>+MockTransport::~MockTransport() = default;
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/mock_transport.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/mock_transport.h
>index 9837593f334..7eaf3c94c32 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/mock_transport.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/mock_transport.h
>@@ -18,6 +18,9 @@ namespace webrtc {
> 
> class MockTransport : public Transport {
>  public:
>+  MockTransport();
>+  ~MockTransport();
>+
>   MOCK_METHOD3(SendRtp,
>                bool(const uint8_t* data,
>                     size_t len,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/mock_voice_engine.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/mock_voice_engine.h
>deleted file mode 100644
>index 57c6b58c675..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/mock_voice_engine.h
>+++ /dev/null
>@@ -1,124 +0,0 @@
>-/*
>- *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef AUDIO_MOCK_VOICE_ENGINE_H_
>-#define AUDIO_MOCK_VOICE_ENGINE_H_
>-
>-#include <memory>
>-
>-#include "modules/audio_device/include/mock_audio_transport.h"
>-#include "modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
>-#include "test/gmock.h"
>-#include "test/mock_voe_channel_proxy.h"
>-#include "voice_engine/voice_engine_impl.h"
>-
>-namespace webrtc {
>-namespace voe {
>-class TransmitMixer;
>-}  // namespace voe
>-
>-namespace test {
>-
>-// NOTE: This class inherits from VoiceEngineImpl so that its clients will be
>-// able to get the various interfaces as usual, via T::GetInterface().
>-class MockVoiceEngine : public VoiceEngineImpl {
>- public:
>-  // TODO(nisse): Valid overrides commented out, because the gmock
>-  // methods don't use any override declarations, and we want to avoid
>-  // warnings from -Winconsistent-missing-override. See
>-  // http://crbug.com/428099.
>-  MockVoiceEngine(
>-      rtc::scoped_refptr<AudioDecoderFactory> decoder_factory = nullptr)
>-      : decoder_factory_(decoder_factory) {
>-    // Increase ref count so this object isn't automatically deleted whenever
>-    // interfaces are Release():d.
>-    ++_ref_count;
>-    // We add this default behavior to make the mock easier to use in tests. It
>-    // will create a NiceMock of a voe::ChannelProxy.
>-    // TODO(ossu): As long as AudioReceiveStream is implemented as a wrapper
>-    // around Channel, we need to make sure ChannelProxy returns the same
>-    // decoder factory as the one passed in when creating an AudioReceiveStream.
>-    ON_CALL(*this, ChannelProxyFactory(testing::_))
>-        .WillByDefault(testing::Invoke([this](int channel_id) {
>-          auto* proxy =
>-              new testing::NiceMock<webrtc::test::MockVoEChannelProxy>();
>-          EXPECT_CALL(*proxy, GetAudioDecoderFactory())
>-              .WillRepeatedly(testing::ReturnRef(decoder_factory_));
>-          EXPECT_CALL(*proxy, SetReceiveCodecs(testing::_))
>-              .WillRepeatedly(testing::Invoke(
>-                  [](const std::map<int, SdpAudioFormat>& codecs) {
>-                    EXPECT_THAT(codecs, testing::IsEmpty());
>-                  }));
>-          EXPECT_CALL(*proxy, GetRtpRtcp(testing::_, testing::_))
>-              .WillRepeatedly(
>-                  testing::SetArgPointee<0>(GetMockRtpRtcp(channel_id)));
>-          return proxy;
>-        }));
>-
>-    ON_CALL(*this, audio_transport())
>-        .WillByDefault(testing::Return(&mock_audio_transport_));
>-  }
>-  virtual ~MockVoiceEngine() /* override */ {
>-    // Decrease ref count before base class d-tor is called; otherwise it will
>-    // trigger an assertion.
>-    --_ref_count;
>-  }
>-
>-  // These need to be the same each call to channel_id and must not leak.
>-  MockRtpRtcp* GetMockRtpRtcp(int channel_id) {
>-    if (mock_rtp_rtcps_.find(channel_id) == mock_rtp_rtcps_.end()) {
>-      mock_rtp_rtcps_[channel_id].reset(new ::testing::NiceMock<MockRtpRtcp>);
>-    }
>-    return mock_rtp_rtcps_[channel_id].get();
>-  }
>-
>-  // Allows injecting a ChannelProxy factory.
>-  MOCK_METHOD1(ChannelProxyFactory, voe::ChannelProxy*(int channel_id));
>-
>-  // VoiceEngineImpl
>-  virtual std::unique_ptr<voe::ChannelProxy> GetChannelProxy(
>-      int channel_id) /* override */ {
>-    return std::unique_ptr<voe::ChannelProxy>(ChannelProxyFactory(channel_id));
>-  }
>-
>-  // VoEBase
>-  MOCK_METHOD3(
>-      Init,
>-      int(AudioDeviceModule* external_adm,
>-          AudioProcessing* external_apm,
>-          const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory));
>-  MOCK_METHOD0(transmit_mixer, voe::TransmitMixer*());
>-  MOCK_METHOD0(Terminate, void());
>-  MOCK_METHOD0(CreateChannel, int());
>-  MOCK_METHOD1(CreateChannel, int(const ChannelConfig& config));
>-  MOCK_METHOD1(DeleteChannel, int(int channel));
>-  MOCK_METHOD1(StartPlayout, int(int channel));
>-  MOCK_METHOD1(StopPlayout, int(int channel));
>-  MOCK_METHOD1(StartSend, int(int channel));
>-  MOCK_METHOD1(StopSend, int(int channel));
>-  MOCK_METHOD0(audio_transport, AudioTransport*());
>-
>- private:
>-  // TODO(ossu): I'm not particularly happy about keeping the decoder factory
>-  // here, but due to how gmock is implemented, I cannot just keep it in the
>-  // functor implementing the default version of ChannelProxyFactory, above.
>-  // GMock creates an unfortunate copy of the functor, which would cause us to
>-  // return a dangling reference. Fortunately, this should go away once
>-  // voe::Channel does.
>-  rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
>-
>-  std::map<int, std::unique_ptr<MockRtpRtcp>> mock_rtp_rtcps_;
>-
>-  MockAudioTransport mock_audio_transport_;
>-};
>-}  // namespace test
>-}  // namespace webrtc
>-
>-#endif  // AUDIO_MOCK_VOICE_ENGINE_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/null_platform_renderer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/null_platform_renderer.cc
>index 48e4cc0d8bb..7ea604ead8e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/null_platform_renderer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/null_platform_renderer.cc
>@@ -18,5 +18,5 @@ VideoRenderer* VideoRenderer::CreatePlatformRenderer(const char* window_title,
>                                                      size_t height) {
>   return NULL;
> }
>-}  // test
>-}  // webrtc
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/rtp_file_reader.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/rtp_file_reader.cc
>index 8d7207d4e8a..c9eb98681cd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/rtp_file_reader.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/rtp_file_reader.cc
>@@ -21,11 +21,12 @@
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/format_macros.h"
> #include "rtc_base/logging.h"
>+#include "rtc_base/system/arch.h"
> 
> namespace webrtc {
> namespace test {
> 
>-static const size_t kFirstLineLength = 40;
>+static const size_t kFirstLineLength = 80;
> static uint16_t kPacketHeaderSize = 8;
> 
> #define TRY(expr)                           \
>@@ -90,7 +91,7 @@ class InterleavedRtpFileReader : public RtpFileReaderImpl {
>     uint32_t len = 0;
>     TRY(ReadUint32(&len, file_));
>     if (packet->length < len) {
>-      RTC_FATAL() << "Packet is too large to fit: " << len << " bytes vs "
>+      FATAL() << "Packet is too large to fit: " << len << " bytes vs "
>               << packet->length
>               << " bytes allocated. Consider increasing the buffer "
>                  "size";
>@@ -178,7 +179,7 @@ class RtpDumpReader : public RtpFileReaderImpl {
>     // Use 'len' here because a 'plen' of 0 specifies rtcp.
>     len -= kPacketHeaderSize;
>     if (packet->length < len) {
>-      RTC_FATAL() << "Packet is too large to fit: " << len << " bytes vs "
>+      FATAL() << "Packet is too large to fit: " << len << " bytes vs "
>               << packet->length
>               << " bytes allocated. Consider increasing the buffer "
>                  "size";
>@@ -241,17 +242,17 @@ const uint32_t kPcapBOMNoSwapOrder = 0xa1b2c3d4UL;
> class PcapReader : public RtpFileReaderImpl {
>  public:
>   PcapReader()
>-    : file_(NULL),
>-      swap_pcap_byte_order_(false),
>+      : file_(NULL),
>+        swap_pcap_byte_order_(false),
> #ifdef WEBRTC_ARCH_BIG_ENDIAN
>-      swap_network_byte_order_(false),
>+        swap_network_byte_order_(false),
> #else
>-      swap_network_byte_order_(true),
>+        swap_network_byte_order_(true),
> #endif
>-      read_buffer_(),
>-      packets_by_ssrc_(),
>-      packets_(),
>-      next_packet_it_() {
>+        read_buffer_(),
>+        packets_by_ssrc_(),
>+        packets_(),
>+        next_packet_it_() {
>   }
> 
>   virtual ~PcapReader() {
>@@ -304,7 +305,7 @@ class PcapReader : public RtpFileReaderImpl {
>     printf("Total RTP/RTCP packets: %" PRIuS "\n", packets_.size());
> 
>     for (SsrcMapIterator mit = packets_by_ssrc_.begin();
>-        mit != packets_by_ssrc_.end(); ++mit) {
>+         mit != packets_by_ssrc_.end(); ++mit) {
>       uint32_t ssrc = mit->first;
>       const std::vector<uint32_t>& packet_indices = mit->second;
>       uint8_t pt = packets_[packet_indices[0]].rtp_header.payloadType;
>@@ -364,14 +365,14 @@ class PcapReader : public RtpFileReaderImpl {
>  private:
>   // A marker of an RTP packet within the file.
>   struct RtpPacketMarker {
>-    uint32_t packet_number;   // One-based index (like in WireShark)
>+    uint32_t packet_number;  // One-based index (like in WireShark)
>     uint32_t time_offset_ms;
>     uint32_t source_ip;
>     uint32_t dest_ip;
>     uint16_t source_port;
>     uint16_t dest_port;
>     RTPHeader rtp_header;
>-    int32_t pos_in_file;      // Byte offset of payload from start of file.
>+    int32_t pos_in_file;  // Byte offset of payload from start of file.
>     uint32_t payload_length;
>   };
> 
>@@ -505,8 +506,8 @@ class PcapReader : public RtpFileReaderImpl {
> 
>   uint32_t CalcTimeDelta(uint32_t ts_sec, uint32_t ts_usec, uint32_t start_ms) {
>     // Round to nearest ms.
>-    uint64_t t2_ms = ((static_cast<uint64_t>(ts_sec) * 1000000) + ts_usec +
>-        500) / 1000;
>+    uint64_t t2_ms =
>+        ((static_cast<uint64_t>(ts_sec) * 1000000) + ts_usec + 500) / 1000;
>     uint64_t t1_ms = static_cast<uint64_t>(start_ms);
>     if (t2_ms < t1_ms) {
>       return 0;
>@@ -577,7 +578,7 @@ class PcapReader : public RtpFileReaderImpl {
>     if ((!expect_network_order && swap_pcap_byte_order_) ||
>         (expect_network_order && swap_network_byte_order_)) {
>       tmp = ((tmp >> 24) & 0x000000ff) | (tmp << 24) |
>-          ((tmp >> 8) & 0x0000ff00) | ((tmp << 8) & 0x00ff0000);
>+            ((tmp >> 8) & 0x0000ff00) | ((tmp << 8) & 0x00ff0000);
>     }
>     *out = tmp;
>     return kResultSuccess;
>@@ -611,7 +612,7 @@ class PcapReader : public RtpFileReaderImpl {
>     if ((!expect_network_order && swap_pcap_byte_order_) ||
>         (expect_network_order && swap_network_byte_order_)) {
>       tmp = ((tmp >> 24) & 0x000000ff) | (tmp << 24) |
>-          ((tmp >> 8) & 0x0000ff00) | ((tmp << 8) & 0x00ff0000);
>+            ((tmp >> 8) & 0x0000ff00) | ((tmp << 8) & 0x00ff0000);
>     }
>     *out = tmp;
>     return kResultSuccess;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/rtp_file_reader.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/rtp_file_reader.h
>index c044d3de267..4913522ba70 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/rtp_file_reader.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/rtp_file_reader.h
>@@ -36,8 +36,7 @@ class RtpFileReader {
>   enum FileFormat { kPcap, kRtpDump, kLengthPacketInterleaved };
> 
>   virtual ~RtpFileReader() {}
>-  static RtpFileReader* Create(FileFormat format,
>-                               const std::string& filename);
>+  static RtpFileReader* Create(FileFormat format, const std::string& filename);
>   static RtpFileReader* Create(FileFormat format,
>                                const std::string& filename,
>                                const std::set<uint32_t>& ssrc_filter);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/rtp_rtcp_observer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/rtp_rtcp_observer.h
>index 5163d2248ee..d946b5f675e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/rtp_rtcp_observer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/rtp_rtcp_observer.h
>@@ -14,6 +14,7 @@
> #include <memory>
> #include <vector>
> 
>+#include "call/video_send_stream.h"
> #include "modules/rtp_rtcp/include/rtp_header_parser.h"
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/event.h"
>@@ -21,8 +22,6 @@
> #include "test/constants.h"
> #include "test/direct_transport.h"
> #include "test/gtest.h"
>-#include "typedefs.h"  // NOLINT(build/include)
>-#include "call/video_send_stream.h"
> 
> namespace {
> const int kShortTimeoutMs = 500;
>@@ -106,10 +105,15 @@ class PacketTransport : public test::DirectTransport {
>         transport_type_(transport_type) {}
> 
>   PacketTransport(SingleThreadedTaskQueueForTesting* task_queue,
>-                  Call* send_call, RtpRtcpObserver* observer,
>+                  Call* send_call,
>+                  RtpRtcpObserver* observer,
>                   TransportType transport_type,
>+                  const std::map<uint8_t, MediaType>& payload_type_map,
>                   std::unique_ptr<FakeNetworkPipe> nw_pipe)
>-      : test::DirectTransport(task_queue, std::move(nw_pipe), send_call),
>+      : test::DirectTransport(task_queue,
>+                              std::move(nw_pipe),
>+                              send_call,
>+                              payload_type_map),
>         observer_(observer),
>         transport_type_(transport_type) {}
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/run_loop.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/run_loop.cc
>index 0d00ecd5e90..dc2ec18eea7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/run_loop.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/run_loop.cc
>@@ -16,7 +16,8 @@ namespace test {
> 
> void PressEnterToContinue() {
>   puts(">> Press ENTER to continue...");
>-  while (getc(stdin) != '\n' && !feof(stdin));
>+  while (getc(stdin) != '\n' && !feof(stdin))
>+    ;
> }
> }  // namespace test
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/run_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/run_test.cc
>index ba0bf34d233..9f6126cf27a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/run_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/run_test.cc
>@@ -15,7 +15,7 @@
> namespace webrtc {
> namespace test {
> 
>-void RunTest(void(*test)()) {
>+void RunTest(void (*test)()) {
>   (*test)();
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/run_test.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/run_test.h
>index ed908570fd4..bd05584365b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/run_test.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/run_test.h
>@@ -14,7 +14,7 @@ namespace webrtc {
> namespace test {
> 
> // Running a test function on a separate thread, if required by the OS.
>-void RunTest(void(*test)());
>+void RunTest(void (*test)());
> 
> }  // namespace test
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/single_threaded_task_queue.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/single_threaded_task_queue.cc
>index 8a8cbea6143..ad7a50327b9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/single_threaded_task_queue.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/single_threaded_task_queue.cc
>@@ -12,9 +12,9 @@
> 
> #include <utility>
> 
>+#include "absl/memory/memory.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/numerics/safe_conversions.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/timeutils.h"
> 
> namespace webrtc {
>@@ -70,7 +70,8 @@ SingleThreadedTaskQueueForTesting::PostDelayedTask(Task task,
>       break;
>     }
>   }
>-  tasks_.insert(it, rtc::MakeUnique<QueuedTask>(id, earliest_exec_time, task));
>+  tasks_.insert(it,
>+                absl::make_unique<QueuedTask>(id, earliest_exec_time, task));
> 
>   // This class is optimized for simplicty, not for performance. This will wake
>   // the thread up even if the next task in the queue is only scheduled for
>@@ -110,7 +111,7 @@ void SingleThreadedTaskQueueForTesting::RunLoop() {
>     std::unique_ptr<QueuedTask> queued_task;
> 
>     // An empty queue would lead to sleeping until the queue becoems non-empty.
>-    // A queue where the earliest task is shceduled for later than now, will
>+    // A queue where the earliest task is scheduled for later than now, will
>     // lead to sleeping until the time of the next scheduled task (or until
>     // more tasks are scheduled).
>     int wait_time = rtc::Event::kForever;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/single_threaded_task_queue_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/single_threaded_task_queue_unittest.cc
>index 5bd40d704a2..084f0d7e8cb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/single_threaded_task_queue_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/single_threaded_task_queue_unittest.cc
>@@ -14,8 +14,8 @@
> #include <memory>
> #include <vector>
> 
>+#include "absl/memory/memory.h"
> #include "rtc_base/event.h"
>-#include "rtc_base/ptr_util.h"
> #include "test/gtest.h"
> 
> namespace webrtc {
>@@ -60,7 +60,7 @@ TEST(SingleThreadedTaskQueueForTestingTest,
> 
>   std::vector<std::unique_ptr<rtc::Event>> done_events;
>   for (size_t i = 0; i < kCount; i++) {
>-    done_events.emplace_back(rtc::MakeUnique<rtc::Event>(false, false));
>+    done_events.emplace_back(absl::make_unique<rtc::Event>(false, false));
>   }
> 
>   // To avoid the tasks which comprise the actual test from running before they
>@@ -68,9 +68,8 @@ TEST(SingleThreadedTaskQueueForTestingTest,
>   // queue at any given time, post one waiting task that would block the
>   // task-queue, and unblock only after all tasks have been posted.
>   rtc::Event rendezvous(true, false);
>-  task_queue.PostTask([&rendezvous]() {
>-    ASSERT_TRUE(rendezvous.Wait(kMaxWaitTimeMs));
>-  });
>+  task_queue.PostTask(
>+      [&rendezvous]() { ASSERT_TRUE(rendezvous.Wait(kMaxWaitTimeMs)); });
> 
>   // Post the tasks which comprise the test.
>   for (size_t i = 0; i < kCount; i++) {
>@@ -126,9 +125,8 @@ TEST(SingleThreadedTaskQueueForTestingTest, TasksExecutedInSequence) {
>   // Prevent the chain from being set in motion before we've had time to
>   // schedule it all, lest the queue only contain one task at a time.
>   rtc::Event rendezvous(true, false);
>-  task_queue.PostTask([&rendezvous]() {
>-    ASSERT_TRUE(rendezvous.Wait(kMaxWaitTimeMs));
>-  });
>+  task_queue.PostTask(
>+      [&rendezvous]() { ASSERT_TRUE(rendezvous.Wait(kMaxWaitTimeMs)); });
> 
>   for (size_t i = 0; i < 3; i++) {
>     task_queue.PostTask([&accumulator, i]() {  // |i| passed by value.
>@@ -139,9 +137,7 @@ TEST(SingleThreadedTaskQueueForTestingTest, TasksExecutedInSequence) {
> 
>   // The test will wait for the task-queue to finish.
>   rtc::Event done(true, false);
>-  task_queue.PostTask([&done]() {
>-    done.Set();
>-  });
>+  task_queue.PostTask([&done]() { done.Set(); });
> 
>   rendezvous.Set();  // Set the chain in motion.
> 
>@@ -159,10 +155,12 @@ TEST(SingleThreadedTaskQueueForTestingTest, ExecutesPostedDelayedTask) {
>   constexpr int64_t delay_ms = 20;
>   static_assert(delay_ms < kMaxWaitTimeMs / 2, "Delay too long for tests.");
> 
>-  task_queue.PostDelayedTask([&executed, &done]() {
>-    executed.store(true);
>-    done.Set();
>-  }, delay_ms);
>+  task_queue.PostDelayedTask(
>+      [&executed, &done]() {
>+        executed.store(true);
>+        done.Set();
>+      },
>+      delay_ms);
>   ASSERT_TRUE(done.Wait(kMaxWaitTimeMs));
> 
>   EXPECT_TRUE(executed.load());
>@@ -176,9 +174,7 @@ TEST(SingleThreadedTaskQueueForTestingTest, DoesNotExecuteDelayedTaskTooSoon) {
>   constexpr int64_t delay_ms = 2000;
>   static_assert(delay_ms < kMaxWaitTimeMs / 2, "Delay too long for tests.");
> 
>-  task_queue.PostDelayedTask([&executed]() {
>-    executed.store(true);
>-  }, delay_ms);
>+  task_queue.PostDelayedTask([&executed]() { executed.store(true); }, delay_ms);
> 
>   // Wait less than is enough, make sure the task was not yet executed.
>   rtc::Event not_done(true, false);
>@@ -262,16 +258,11 @@ TEST(SingleThreadedTaskQueueForTestingTest, ExternalThreadCancelsTask) {
>   // Prevent the to-be-cancelled task from being executed before we've had
>   // time to cancel it.
>   rtc::Event rendezvous(true, false);
>-  task_queue.PostTask([&rendezvous]() {
>-    ASSERT_TRUE(rendezvous.Wait(kMaxWaitTimeMs));
>-  });
>+  task_queue.PostTask(
>+      [&rendezvous]() { ASSERT_TRUE(rendezvous.Wait(kMaxWaitTimeMs)); });
> 
>-  TaskId cancelled_task_id = task_queue.PostTask([]() {
>-    EXPECT_TRUE(false);
>-  });
>-  task_queue.PostTask([&done]() {
>-    done.Set();
>-  });
>+  TaskId cancelled_task_id = task_queue.PostTask([]() { EXPECT_TRUE(false); });
>+  task_queue.PostTask([&done]() { done.Set(); });
> 
>   task_queue.CancelTask(cancelled_task_id);
> 
>@@ -292,9 +283,8 @@ TEST(SingleThreadedTaskQueueForTestingTest, InternalThreadCancelsTask) {
> 
>   // Prevent the chain from being set-off before we've set everything up.
>   rtc::Event rendezvous(true, false);
>-  task_queue.PostTask([&rendezvous]() {
>-    ASSERT_TRUE(rendezvous.Wait(kMaxWaitTimeMs));
>-  });
>+  task_queue.PostTask(
>+      [&rendezvous]() { ASSERT_TRUE(rendezvous.Wait(kMaxWaitTimeMs)); });
> 
>   // This is the canceller-task. It takes cancelled_task_id by reference,
>   // because the ID will only become known after the cancelled task is
>@@ -306,15 +296,11 @@ TEST(SingleThreadedTaskQueueForTestingTest, InternalThreadCancelsTask) {
>   task_queue.PostTask(canceller_task);
> 
>   // This task will be cancelled by the task before it.
>-  auto cancelled_task = []() {
>-    EXPECT_TRUE(false);
>-  };
>+  auto cancelled_task = []() { EXPECT_TRUE(false); };
>   cancelled_task_id = task_queue.PostTask(cancelled_task);
> 
>   // When this task runs, it will allow the test to be finished.
>-  auto completion_marker_task = [&done]() {
>-    done.Set();
>-  };
>+  auto completion_marker_task = [&done]() { done.Set(); };
>   task_queue.PostTask(completion_marker_task);
> 
>   rendezvous.Set();  // Set the chain in motion.
>@@ -341,7 +327,7 @@ TEST(SingleThreadedTaskQueueForTestingTest, SendTask) {
> TEST(SingleThreadedTaskQueueForTestingTest,
>      DestructTaskQueueWhileTasksPending) {
>   auto task_queue =
>-      rtc::MakeUnique<SingleThreadedTaskQueueForTesting>("task_queue");
>+      absl::make_unique<SingleThreadedTaskQueueForTesting>("task_queue");
> 
>   std::atomic<size_t> counter(0);
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/statistics.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/statistics.cc
>index c43dde96fa9..192366aef8e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/statistics.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/statistics.cc
>@@ -11,23 +11,40 @@
> 
> #include <math.h>
> 
>+#include <algorithm>
>+
> namespace webrtc {
> namespace test {
> 
>-Statistics::Statistics() : sum_(0.0), sum_squared_(0.0), count_(0) {}
>+Statistics::Statistics()
>+    : sum_(0.0),
>+      sum_squared_(0.0),
>+      max_(std::numeric_limits<double>::min()),
>+      min_(std::numeric_limits<double>::max()),
>+      count_(0) {}
> 
> void Statistics::AddSample(double sample) {
>   sum_ += sample;
>   sum_squared_ += sample * sample;
>+  max_ = std::max(max_, sample);
>+  min_ = std::min(min_, sample);
>   ++count_;
> }
> 
>+double Statistics::Max() const {
>+  return max_;
>+}
>+
> double Statistics::Mean() const {
>   if (count_ == 0)
>     return 0.0;
>   return sum_ / count_;
> }
> 
>+double Statistics::Min() const {
>+  return min_;
>+}
>+
> double Statistics::Variance() const {
>   if (count_ == 0)
>     return 0.0;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/statistics.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/statistics.h
>index d52d92dc110..2fa706f6032 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/statistics.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/statistics.h
>@@ -10,7 +10,7 @@
> #ifndef TEST_STATISTICS_H_
> #define TEST_STATISTICS_H_
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>+#include <stdint.h>
> 
> namespace webrtc {
> namespace test {
>@@ -21,13 +21,17 @@ class Statistics {
> 
>   void AddSample(double sample);
> 
>+  double Max() const;
>   double Mean() const;
>+  double Min() const;
>   double Variance() const;
>   double StandardDeviation() const;
> 
>  private:
>   double sum_;
>   double sum_squared_;
>+  double max_;
>+  double min_;
>   uint64_t count_;
> };
> }  // namespace test
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/test_main.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/test_main.cc
>index 8bd2ed2244e..277d420c675 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/test_main.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/test_main.cc
>@@ -10,24 +10,52 @@
> 
> #include "rtc_base/flags.h"
> #include "rtc_base/logging.h"
>+#include "system_wrappers/include/field_trial_default.h"
> #include "system_wrappers/include/metrics_default.h"
> #include "test/field_trial.h"
> #include "test/gmock.h"
> #include "test/gtest.h"
> #include "test/testsupport/fileutils.h"
>+#include "test/testsupport/perf_test.h"
> 
> #if defined(WEBRTC_IOS)
> #include "test/ios/test_support.h"
> 
>-DEFINE_string(NSTreatUnknownArgumentsAsOpen, "",
>-    "Intentionally ignored flag intended for iOS simulator.");
>-DEFINE_string(ApplePersistenceIgnoreState, "",
>-    "Intentionally ignored flag intended for iOS simulator.");
>+DEFINE_string(NSTreatUnknownArgumentsAsOpen,
>+              "",
>+              "Intentionally ignored flag intended for iOS simulator.");
>+DEFINE_string(ApplePersistenceIgnoreState,
>+              "",
>+              "Intentionally ignored flag intended for iOS simulator.");
>+DEFINE_bool(
>+    save_chartjson_result,
>+    false,
>+    "Store the perf results in Documents/perf_result.json in the format "
>+    "described by "
>+    "https://github.com/catapult-project/catapult/blob/master/dashboard/docs/"
>+    "data-format.md.");
>+
>+#else
>+
>+DEFINE_string(isolated_script_test_output,
>+              "",
>+              "Intentionally ignored flag intended for Chromium.");
>+
>+DEFINE_string(
>+    isolated_script_test_perf_output,
>+    "",
>+    "Path where the perf results should be stored in the JSON format described "
>+    "by "
>+    "https://github.com/catapult-project/catapult/blob/master/dashboard/docs/"
>+    "data-format.md.");
>+
> #endif
> 
> DEFINE_bool(logs, false, "print logs to stderr");
> 
>-DEFINE_string(force_fieldtrials, "",
>+DEFINE_string(
>+    force_fieldtrials,
>+    "",
>     "Field trials control experimental feature code which can be forced. "
>     "E.g. running with --force_fieldtrials=WebRTC-FooFeature/Enable/"
>     " will assign the group Enable to field trial WebRTC-FooFeature.");
>@@ -51,15 +79,30 @@ int main(int argc, char* argv[]) {
>   }
> 
>   webrtc::test::SetExecutablePath(argv[0]);
>-  std::string fieldtrials = FLAG_force_fieldtrials;
>-  webrtc::test::InitFieldTrialsFromString(fieldtrials);
>+  webrtc::test::ValidateFieldTrialsStringOrDie(FLAG_force_fieldtrials);
>+  // InitFieldTrialsFromString stores the char*, so the char array must outlive
>+  // the application.
>+  webrtc::field_trial::InitFieldTrialsFromString(FLAG_force_fieldtrials);
>   webrtc::metrics::Enable();
> 
>   rtc::LogMessage::SetLogToStderr(FLAG_logs);
>+
> #if defined(WEBRTC_IOS)
>-  rtc::test::InitTestSuite(RUN_ALL_TESTS, argc, argv);
>+
>+  rtc::test::InitTestSuite(RUN_ALL_TESTS, argc, argv,
>+                           FLAG_save_chartjson_result);
>   rtc::test::RunTestsFromIOSApp();
>-#endif
> 
>-  return RUN_ALL_TESTS();
>+#else
>+
>+  int exit_code = RUN_ALL_TESTS();
>+
>+  std::string chartjson_result_file = FLAG_isolated_script_test_perf_output;
>+  if (!chartjson_result_file.empty()) {
>+    webrtc::test::WritePerfResults(chartjson_result_file);
>+  }
>+
>+  return exit_code;
>+
>+#endif
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/fileutils.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/fileutils.cc
>index 2aff75a4d91..370ee06b29d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/fileutils.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/fileutils.cc
>@@ -31,8 +31,8 @@
> #endif
> 
> #include <sys/stat.h>  // To check for directory existence.
>-#ifndef S_ISDIR  // Not defined in stat.h on Windows.
>-#define S_ISDIR(mode) (((mode) & S_IFMT) == S_IFDIR)
>+#ifndef S_ISDIR        // Not defined in stat.h on Windows.
>+#define S_ISDIR(mode) (((mode)&S_IFMT) == S_IFDIR)
> #endif
> 
> #include <stdio.h>
>@@ -43,7 +43,7 @@
> #include <utility>
> 
> #include "rtc_base/checks.h"
>-#include "typedefs.h"  // NOLINT(build/include)  // For architecture defines
>+#include "rtc_base/stringutils.h"
> 
> namespace webrtc {
> namespace test {
>@@ -91,9 +91,9 @@ void SetExecutablePath(const std::string& path) {
>   if (path.find(working_dir) != std::string::npos) {
>     temp_path = path.substr(working_dir.length() + 1);
>   }
>-  // On Windows, when tests are run under memory tools like DrMemory and TSan,
>-  // slashes occur in the path as directory separators. Make sure we replace
>-  // such cases with backslashes in order for the paths to be correct.
>+// On Windows, when tests are run under memory tools like DrMemory and TSan,
>+// slashes occur in the path as directory separators. Make sure we replace
>+// such cases with backslashes in order for the paths to be correct.
> #ifdef WIN32
>   std::replace(temp_path.begin(), temp_path.end(), '/', '\\');
> #endif
>@@ -111,8 +111,8 @@ bool FileExists(const std::string& file_name) {
> 
> bool DirExists(const std::string& directory_name) {
>   struct stat directory_info = {0};
>-  return stat(directory_name.c_str(), &directory_info) == 0 && S_ISDIR(
>-      directory_info.st_mode);
>+  return stat(directory_name.c_str(), &directory_info) == 0 &&
>+         S_ISDIR(directory_info.st_mode);
> }
> 
> #ifdef WEBRTC_ANDROID
>@@ -189,11 +189,11 @@ std::string WorkingDir() {
> 
> // Generate a temporary filename in a safe way.
> // Largely copied from talk/base/{unixfilesystem,win32filesystem}.cc.
>-std::string TempFilename(const std::string &dir, const std::string &prefix) {
>+std::string TempFilename(const std::string& dir, const std::string& prefix) {
> #ifdef WIN32
>   wchar_t filename[MAX_PATH];
>-  if (::GetTempFileName(rtc::ToUtf16(dir).c_str(),
>-                        rtc::ToUtf16(prefix).c_str(), 0, filename) != 0)
>+  if (::GetTempFileName(rtc::ToUtf16(dir).c_str(), rtc::ToUtf16(prefix).c_str(),
>+                        0, filename) != 0)
>     return rtc::ToUtf8(filename);
>   assert(false);
>   return "";
>@@ -201,8 +201,7 @@ std::string TempFilename(const std::string &dir, const std::string &prefix) {
>   int len = dir.size() + prefix.size() + 2 + 6;
>   std::unique_ptr<char[]> tempname(new char[len]);
> 
>-  snprintf(tempname.get(), len, "%s/%sXXXXXX", dir.c_str(),
>-           prefix.c_str());
>+  snprintf(tempname.get(), len, "%s/%sXXXXXX", dir.c_str(), prefix.c_str());
>   int fd = ::mkstemp(tempname.get());
>   if (fd == -1) {
>     assert(false);
>@@ -215,9 +214,16 @@ std::string TempFilename(const std::string &dir, const std::string &prefix) {
> #endif
> }
> 
>-rtc::Optional<std::vector<std::string>> ReadDirectory(std::string path) {
>+std::string GenerateTempFilename(const std::string& dir,
>+                                 const std::string& prefix) {
>+  std::string filename = TempFilename(dir, prefix);
>+  RemoveFile(filename);
>+  return filename;
>+}
>+
>+absl::optional<std::vector<std::string>> ReadDirectory(std::string path) {
>   if (path.length() == 0)
>-    return rtc::Optional<std::vector<std::string>>();
>+    return absl::optional<std::vector<std::string>>();
> 
> #if defined(WEBRTC_WIN)
>   // Append separator character if needed.
>@@ -228,7 +234,7 @@ rtc::Optional<std::vector<std::string>> ReadDirectory(std::string path) {
>   WIN32_FIND_DATA data;
>   HANDLE handle = ::FindFirstFile(rtc::ToUtf16(path + '*').c_str(), &data);
>   if (handle == INVALID_HANDLE_VALUE)
>-    return rtc::Optional<std::vector<std::string>>();
>+    return absl::optional<std::vector<std::string>>();
> 
>   // Populate output.
>   std::vector<std::string> found_entries;
>@@ -249,7 +255,7 @@ rtc::Optional<std::vector<std::string>> ReadDirectory(std::string path) {
>   // Init.
>   DIR* dir = ::opendir(path.c_str());
>   if (dir == nullptr)
>-    return rtc::Optional<std::vector<std::string>>();
>+    return absl::optional<std::vector<std::string>>();
> 
>   // Populate output.
>   std::vector<std::string> found_entries;
>@@ -263,7 +269,7 @@ rtc::Optional<std::vector<std::string>> ReadDirectory(std::string path) {
>   closedir(dir);
> #endif
> 
>-  return rtc::Optional<std::vector<std::string>>(std::move(found_entries));
>+  return absl::optional<std::vector<std::string>>(std::move(found_entries));
> }
> 
> bool CreateDir(const std::string& directory_name) {
>@@ -271,7 +277,8 @@ bool CreateDir(const std::string& directory_name) {
>   // Check if the path exists already:
>   if (stat(directory_name.c_str(), &path_info) == 0) {
>     if (!S_ISDIR(path_info.st_mode)) {
>-      fprintf(stderr, "Path %s exists but is not a directory! Remove this "
>+      fprintf(stderr,
>+              "Path %s exists but is not a directory! Remove this "
>               "file and re-run to create the directory.\n",
>               directory_name.c_str());
>       return false;
>@@ -280,7 +287,7 @@ bool CreateDir(const std::string& directory_name) {
> #ifdef WIN32
>     return _mkdir(directory_name.c_str()) == 0;
> #else
>-    return mkdir(directory_name.c_str(),  S_IRWXU | S_IRWXG | S_IRWXO) == 0;
>+    return mkdir(directory_name.c_str(), S_IRWXU | S_IRWXG | S_IRWXO) == 0;
> #endif
>   }
>   return true;
>@@ -288,9 +295,9 @@ bool CreateDir(const std::string& directory_name) {
> 
> bool RemoveDir(const std::string& directory_name) {
> #ifdef WIN32
>-    return RemoveDirectoryA(directory_name.c_str()) != FALSE;
>+  return RemoveDirectoryA(directory_name.c_str()) != FALSE;
> #else
>-    return rmdir(directory_name.c_str()) == 0;
>+  return rmdir(directory_name.c_str()) == 0;
> #endif
> }
> 
>@@ -318,35 +325,23 @@ std::string ResourcePath(const std::string& name,
>   platform = "android";
> #endif  // WEBRTC_ANDROID
> 
>-#ifdef WEBRTC_ARCH_64_BITS
>-  std::string architecture = "64";
>-#else
>-  std::string architecture = "32";
>-#endif  // WEBRTC_ARCH_64_BITS
>-
>-  std::string resources_path = ProjectRootPath() + kResourcesDirName +
>-      kPathDelimiter;
>-  std::string resource_file = resources_path + name + "_" + platform + "_" +
>-      architecture + "." + extension;
>-  if (FileExists(resource_file)) {
>-    return resource_file;
>-  }
>-  // Try without architecture.
>-  resource_file = resources_path + name + "_" + platform + "." + extension;
>+  std::string resources_path =
>+      ProjectRootPath() + kResourcesDirName + kPathDelimiter;
>+  std::string resource_file =
>+      resources_path + name + "_" + platform + "." + extension;
>   if (FileExists(resource_file)) {
>     return resource_file;
>   }
>-  // Try without platform.
>-  resource_file = resources_path + name + "_" + architecture + "." + extension;
>-  if (FileExists(resource_file)) {
>-    return resource_file;
>-  }
>-
>-  // Fall back on name without architecture or platform.
>+  // Fall back on name without platform.
>   return resources_path + name + "." + extension;
> #endif  // defined (WEBRTC_IOS)
> }
> 
>+std::string JoinFilename(const std::string& dir, const std::string& name) {
>+  RTC_CHECK(!dir.empty()) << "Special cases not implemented.";
>+  return dir + kPathDelimiter + name;
>+}
>+
> size_t GetFileSize(const std::string& filename) {
>   FILE* f = fopen(filename.c_str(), "rb");
>   size_t size = 0;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/fileutils.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/fileutils.h
>index f0560cf979c..8cef0e36b5a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/fileutils.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/fileutils.h
>@@ -16,7 +16,7 @@
> #include <string>
> #include <vector>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> 
> namespace webrtc {
> namespace test {
>@@ -40,7 +40,13 @@ std::string OutputPath();
> 
> // Generates an empty file with a unique name in the specified directory and
> // returns the file name and path.
>-std::string TempFilename(const std::string &dir, const std::string &prefix);
>+// TODO(titovartem) rename to TempFile and next method to TempFilename
>+std::string TempFilename(const std::string& dir, const std::string& prefix);
>+
>+// Generates a unique file name that can be used for file creation. Doesn't
>+// create any files.
>+std::string GenerateTempFilename(const std::string& dir,
>+                                 const std::string& prefix);
> 
> // Returns a path to a resource file for the currently executing platform.
> // Adapts to what filenames are currently present in the
>@@ -61,8 +67,10 @@ std::string TempFilename(const std::string &dir, const std::string &prefix);
> //           If a directory path is prepended to the filename, a subdirectory
> //           hierarchy reflecting that path is assumed to be present.
> //    extension - File extension, without the dot, i.e. "bmp" or "yuv".
>-std::string ResourcePath(const std::string& name,
>-                         const std::string& extension);
>+std::string ResourcePath(const std::string& name, const std::string& extension);
>+
>+// Joins directory name and file name, separated by the path delimiter.
>+std::string JoinFilename(const std::string& dir, const std::string& name);
> 
> // Gets the current working directory for the executing program.
> // Returns "./" if for some reason it is not possible to find the working
>@@ -73,7 +81,7 @@ std::string WorkingDir();
> // of strings with one element for each found file or directory. Each element is
> // a path created by prepending |dir| to the file/directory name. "." and ".."
> // are never added in the returned vector.
>-rtc::Optional<std::vector<std::string>> ReadDirectory(std::string path);
>+absl::optional<std::vector<std::string>> ReadDirectory(std::string path);
> 
> // Creates a directory if it not already exists.
> // Returns true if successful. Will print an error message to stderr and return
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/fileutils_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/fileutils_unittest.cc
>index 79988c0dd34..dc1f1bc7f7f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/fileutils_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/fileutils_unittest.cc
>@@ -17,9 +17,8 @@
> #include <list>
> #include <string>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> #include "rtc_base/checks.h"
>-#include "rtc_base/pathutils.h"
> #include "test/gtest.h"
> 
> #ifdef WIN32
>@@ -42,7 +41,7 @@ namespace {
> void CleanDir(const std::string& dir, size_t* num_deleted_entries) {
>   RTC_DCHECK(num_deleted_entries);
>   *num_deleted_entries = 0;
>-  rtc::Optional<std::vector<std::string>> dir_content = ReadDirectory(dir);
>+  absl::optional<std::vector<std::string>> dir_content = ReadDirectory(dir);
>   EXPECT_TRUE(dir_content);
>   for (const auto& entry : *dir_content) {
>     if (DirExists(entry)) {
>@@ -70,19 +69,17 @@ void WriteStringInFile(const std::string& what, const std::string& file_path) {
> // gtest framework).
> class FileUtilsTest : public testing::Test {
>  protected:
>-  FileUtilsTest() {
>-  }
>+  FileUtilsTest() {}
>   ~FileUtilsTest() override {}
>   // Runs before the first test
>   static void SetUpTestCase() {
>     original_working_dir_ = webrtc::test::WorkingDir();
>   }
>-  void SetUp() override {
>-    ASSERT_EQ(chdir(original_working_dir_.c_str()), 0);
>-  }
>+  void SetUp() override { ASSERT_EQ(chdir(original_working_dir_.c_str()), 0); }
>   void TearDown() override {
>     ASSERT_EQ(chdir(original_working_dir_.c_str()), 0);
>   }
>+
>  private:
>   static std::string original_working_dir_;
> };
>@@ -127,6 +124,19 @@ TEST_F(FileUtilsTest, TempFilename) {
>   remove(temp_filename.c_str());
> }
> 
>+TEST_F(FileUtilsTest, GenerateTempFilename) {
>+  std::string temp_filename = webrtc::test::GenerateTempFilename(
>+      webrtc::test::OutputPath(), "TempFilenameTest");
>+  ASSERT_FALSE(webrtc::test::FileExists(temp_filename))
>+      << "File exists: " << temp_filename;
>+  FILE* file = fopen(temp_filename.c_str(), "wb");
>+  ASSERT_TRUE(file != NULL) << "Failed to open file: " << temp_filename;
>+  ASSERT_GT(fprintf(file, "%s", "Dummy data"), 0)
>+      << "Failed to write to file: " << temp_filename;
>+  fclose(file);
>+  remove(temp_filename.c_str());
>+}
>+
> // Only tests that the code executes
> #if defined(WEBRTC_IOS)
> #define MAYBE_CreateDir DISABLED_CreateDir
>@@ -174,8 +184,8 @@ TEST_F(FileUtilsTest, GetFileSizeExistingFile) {
>       webrtc::test::OutputPath(), "fileutils_unittest");
>   FILE* file = fopen(temp_filename.c_str(), "wb");
>   ASSERT_TRUE(file != NULL) << "Failed to open file: " << temp_filename;
>-  ASSERT_GT(fprintf(file, "%s",  "Dummy data"), 0) <<
>-      "Failed to write to file: " << temp_filename;
>+  ASSERT_GT(fprintf(file, "%s", "Dummy data"), 0)
>+      << "Failed to write to file: " << temp_filename;
>   fclose(file);
>   ASSERT_GT(webrtc::test::GetFileSize(std::string(temp_filename.c_str())), 0u);
>   remove(temp_filename.c_str());
>@@ -226,7 +236,7 @@ TEST_F(FileUtilsTest, WriteReadDeleteFilesAndDirs) {
>   EXPECT_TRUE(DirExists(temp_subdir));
> 
>   // Checks.
>-  rtc::Optional<std::vector<std::string>> dir_content =
>+  absl::optional<std::vector<std::string>> dir_content =
>       ReadDirectory(temp_directory);
>   EXPECT_TRUE(dir_content);
>   EXPECT_EQ(2u, dir_content->size());
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/frame_reader.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/frame_reader.h
>index 2e19af344e6..de8962e1266 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/frame_reader.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/frame_reader.h
>@@ -16,7 +16,6 @@
> #include <string>
> 
> #include "rtc_base/scoped_ref_ptr.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> class I420Buffer;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/frame_writer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/frame_writer.h
>index 7556050a214..b91e57c963d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/frame_writer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/frame_writer.h
>@@ -16,7 +16,6 @@
> #include <string>
> 
> #include "api/video/video_frame.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> namespace test {
>@@ -86,7 +85,7 @@ class Y4mFrameWriterImpl : public YuvFrameWriterImpl {
> // LibJpeg is not available on iOS. This class will do nothing on iOS.
> class JpegFrameWriter {
>  public:
>-  JpegFrameWriter(const std::string &output_filename);
>+  JpegFrameWriter(const std::string& output_filename);
>   // Quality can be from 0 (worst) to 100 (best). Best quality is still lossy.
>   // WriteFrame can be called only once. Subsequent calls will fail.
>   bool WriteFrame(const VideoFrame& input_frame, int quality);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/iosfileutils.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/iosfileutils.mm
>index a9156db6564..5e9f71c8b77 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/iosfileutils.mm
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/iosfileutils.mm
>@@ -15,7 +15,6 @@
> 
> #include "rtc_base/checks.h"
> #include "sdk/objc/Framework/Classes/Common/helpers.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> namespace test {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/jpeg_frame_writer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/jpeg_frame_writer.cc
>index 4888f65aeba..537e85cfa5e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/jpeg_frame_writer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/jpeg_frame_writer.cc
>@@ -10,7 +10,6 @@
> 
> #include <stdio.h>
> 
>-
> #include "common_types.h"  // NOLINT(build/include)
> #include "common_video/libyuv/include/webrtc_libyuv.h"
> #include "rtc_base/checks.h"
>@@ -29,7 +28,7 @@ extern "C" {
> namespace webrtc {
> namespace test {
> 
>-JpegFrameWriter::JpegFrameWriter(const std::string &output_filename)
>+JpegFrameWriter::JpegFrameWriter(const std::string& output_filename)
>     : frame_written_(false),
>       output_filename_(output_filename),
>       output_file_(nullptr) {}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/jpeg_frame_writer_ios.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/jpeg_frame_writer_ios.cc
>index 567e75e2543..e72fea102f1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/jpeg_frame_writer_ios.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/jpeg_frame_writer_ios.cc
>@@ -12,7 +12,6 @@
> #include "rtc_base/logging.h"
> #include "test/testsupport/frame_writer.h"
> 
>-
> namespace webrtc {
> namespace test {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/metrics/video_metrics.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/metrics/video_metrics.cc
>deleted file mode 100644
>index ea5d3f2b5f2..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/metrics/video_metrics.cc
>+++ /dev/null
>@@ -1,193 +0,0 @@
>-/*
>- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "test/testsupport/metrics/video_metrics.h"
>-
>-#include <assert.h>
>-#include <stdio.h>
>-
>-#include <algorithm>  // min_element, max_element
>-#include <memory>
>-
>-#include "api/video/i420_buffer.h"
>-#include "api/video/video_frame.h"
>-#include "common_video/libyuv/include/webrtc_libyuv.h"
>-#include "test/frame_utils.h"
>-#include "libyuv/convert.h"
>-
>-namespace webrtc {
>-namespace test {
>-
>-// Copy here so our callers won't need to include libyuv for this constant.
>-double kMetricsPerfectPSNR = kPerfectPSNR;
>-
>-// Used for calculating min and max values.
>-static bool LessForFrameResultValue(const FrameResult& s1,
>-                                    const FrameResult& s2) {
>-  return s1.value < s2.value;
>-}
>-
>-enum VideoMetricsType { kPSNR, kSSIM, kBoth };
>-
>-// Calculates metrics for a frame and adds statistics to the result for it.
>-void CalculateFrame(VideoMetricsType video_metrics_type,
>-                    const I420BufferInterface& ref,
>-                    const I420BufferInterface& test,
>-                    int frame_number,
>-                    QualityMetricsResult* result) {
>-  FrameResult frame_result = {0, 0};
>-  frame_result.frame_number = frame_number;
>-  switch (video_metrics_type) {
>-    case kPSNR:
>-      frame_result.value = I420PSNR(ref, test);
>-      break;
>-    case kSSIM:
>-      frame_result.value = I420SSIM(ref, test);
>-      break;
>-    default:
>-      assert(false);
>-  }
>-  result->frames.push_back(frame_result);
>-}
>-
>-// Calculates average, min and max values for the supplied struct, if non-NULL.
>-void CalculateStats(QualityMetricsResult* result) {
>-  if (result == NULL || result->frames.size() == 0) {
>-    return;
>-  }
>-  // Calculate average.
>-  std::vector<FrameResult>::iterator iter;
>-  double metrics_values_sum = 0.0;
>-  for (iter = result->frames.begin(); iter != result->frames.end(); ++iter) {
>-    metrics_values_sum += iter->value;
>-  }
>-  result->average = metrics_values_sum / result->frames.size();
>-
>-  // Calculate min/max statistics.
>-  iter = std::min_element(result->frames.begin(), result->frames.end(),
>-                     LessForFrameResultValue);
>-  result->min = iter->value;
>-  result->min_frame_number = iter->frame_number;
>-  iter = std::max_element(result->frames.begin(), result->frames.end(),
>-                     LessForFrameResultValue);
>-  result->max = iter->value;
>-  result->max_frame_number = iter->frame_number;
>-}
>-
>-// Single method that handles all combinations of video metrics calculation, to
>-// minimize code duplication. Either psnr_result or ssim_result may be NULL,
>-// depending on which VideoMetricsType is targeted.
>-int CalculateMetrics(VideoMetricsType video_metrics_type,
>-                     const char* ref_filename,
>-                     const char* test_filename,
>-                     int width,
>-                     int height,
>-                     QualityMetricsResult* psnr_result,
>-                     QualityMetricsResult* ssim_result) {
>-  assert(ref_filename != NULL);
>-  assert(test_filename != NULL);
>-  assert(width > 0);
>-  assert(height > 0);
>-
>-  FILE* ref_fp = fopen(ref_filename, "rb");
>-  if (ref_fp == NULL) {
>-    // Cannot open reference file.
>-    fprintf(stderr, "Cannot open file %s\n", ref_filename);
>-    return -1;
>-  }
>-  FILE* test_fp = fopen(test_filename, "rb");
>-  if (test_fp == NULL) {
>-    // Cannot open test file.
>-    fprintf(stderr, "Cannot open file %s\n", test_filename);
>-    fclose(ref_fp);
>-    return -2;
>-  }
>-  int frame_number = 0;
>-
>-  // Read reference and test frames.
>-  for (;;) {
>-    rtc::scoped_refptr<I420Buffer> ref_i420_buffer(
>-        test::ReadI420Buffer(width, height, ref_fp));
>-    if (!ref_i420_buffer)
>-      break;
>-
>-    rtc::scoped_refptr<I420Buffer> test_i420_buffer(
>-        test::ReadI420Buffer(width, height, test_fp));
>-
>-    if (!test_i420_buffer)
>-      break;
>-
>-    switch (video_metrics_type) {
>-      case kPSNR:
>-        CalculateFrame(kPSNR, *ref_i420_buffer, *test_i420_buffer, frame_number,
>-                       psnr_result);
>-        break;
>-      case kSSIM:
>-        CalculateFrame(kSSIM, *ref_i420_buffer, *test_i420_buffer, frame_number,
>-                       ssim_result);
>-        break;
>-      case kBoth:
>-        CalculateFrame(kPSNR, *ref_i420_buffer, *test_i420_buffer, frame_number,
>-                       psnr_result);
>-        CalculateFrame(kSSIM, *ref_i420_buffer, *test_i420_buffer, frame_number,
>-                       ssim_result);
>-        break;
>-    }
>-    frame_number++;
>-  }
>-  int return_code = 0;
>-  if (frame_number == 0) {
>-    fprintf(stderr, "Tried to measure video metrics from empty files "
>-            "(reference file: %s  test file: %s)\n", ref_filename,
>-            test_filename);
>-    return_code = -3;
>-  } else {
>-    CalculateStats(psnr_result);
>-    CalculateStats(ssim_result);
>-  }
>-  fclose(ref_fp);
>-  fclose(test_fp);
>-  return return_code;
>-}
>-
>-int I420MetricsFromFiles(const char* ref_filename,
>-                         const char* test_filename,
>-                         int width,
>-                         int height,
>-                         QualityMetricsResult* psnr_result,
>-                         QualityMetricsResult* ssim_result) {
>-  assert(psnr_result != NULL);
>-  assert(ssim_result != NULL);
>-  return CalculateMetrics(kBoth, ref_filename, test_filename, width, height,
>-                          psnr_result, ssim_result);
>-}
>-
>-int I420PSNRFromFiles(const char* ref_filename,
>-                      const char* test_filename,
>-                      int width,
>-                      int height,
>-                      QualityMetricsResult* result) {
>-  assert(result != NULL);
>-  return CalculateMetrics(kPSNR, ref_filename, test_filename, width, height,
>-                          result, NULL);
>-}
>-
>-int I420SSIMFromFiles(const char* ref_filename,
>-                      const char* test_filename,
>-                      int width,
>-                      int height,
>-                      QualityMetricsResult* result) {
>-  assert(result != NULL);
>-  return CalculateMetrics(kSSIM, ref_filename, test_filename, width, height,
>-                          NULL, result);
>-}
>-
>-}  // namespace test
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/metrics/video_metrics.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/metrics/video_metrics.h
>deleted file mode 100644
>index f72ea9b7bb4..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/metrics/video_metrics.h
>+++ /dev/null
>@@ -1,121 +0,0 @@
>-/*
>- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef TESTSUPPORT_METRICS_VIDEO_METRICS_H_
>-#define TESTSUPPORT_METRICS_VIDEO_METRICS_H_
>-
>-#include <limits>
>-#include <vector>
>-
>-namespace webrtc {
>-namespace test {
>-
>-// The highest PSNR value our algorithms will return.
>-extern double kMetricsPerfectPSNR;
>-
>-// Contains video quality metrics result for a single frame.
>-struct FrameResult {
>-  int frame_number;
>-  double value;
>-};
>-
>-// Result from a PSNR/SSIM calculation operation.
>-// The frames in this data structure are 0-indexed.
>-struct QualityMetricsResult {
>-  QualityMetricsResult() :
>-    average(0.0),
>-    min(std::numeric_limits<double>::max()),
>-    max(std::numeric_limits<double>::min()),
>-    min_frame_number(-1),
>-    max_frame_number(-1)
>-  {};
>-  double average;
>-  double min;
>-  double max;
>-  int min_frame_number;
>-  int max_frame_number;
>-  std::vector<FrameResult> frames;
>-};
>-
>-// Calculates PSNR and SSIM values for the reference and test video files
>-// (must be in I420 format). All calculated values are filled into the
>-// QualityMetricsResult structs.
>-//
>-// PSNR values have the unit decibel (dB) where a high value means the test file
>-// is similar to the reference file. The higher value, the more similar. The
>-// maximum PSNR value is kMetricsInfinitePSNR. For more info about PSNR, see
>-// http://en.wikipedia.org/wiki/PSNR.
>-//
>-// SSIM values range between -1.0 and 1.0, where 1.0 means the files are
>-// identical. For more info about SSIM, see http://en.wikipedia.org/wiki/SSIM
>-// This function only compares video frames up to the point when the shortest
>-// video ends.
>-// Return value:
>-//  0 if successful, negative on errors:
>-// -1 if the source file cannot be opened
>-// -2 if the test file cannot be opened
>-// -3 if any of the files are empty
>-// -4 if any arguments are invalid.
>-int I420MetricsFromFiles(const char* ref_filename,
>-                         const char* test_filename,
>-                         int width,
>-                         int height,
>-                         QualityMetricsResult* psnr_result,
>-                         QualityMetricsResult* ssim_result);
>-
>-// Calculates PSNR values for the reference and test video files (must be in
>-// I420 format). All calculated values are filled into the QualityMetricsResult
>-// struct.
>-//
>-// PSNR values have the unit decibel (dB) where a high value means the test file
>-// is similar to the reference file. The higher value, the more similar. The
>-// maximum PSNR value is kMetricsInfinitePSNR. For more info about PSNR, see
>-// http://en.wikipedia.org/wiki/PSNR.
>-//
>-// This function only compares video frames up to the point when the shortest
>-// video ends.
>-//
>-// Return value:
>-//  0 if successful, negative on errors:
>-// -1 if the source file cannot be opened
>-// -2 if the test file cannot be opened
>-// -3 if any of the files are empty
>-// -4 if any arguments are invalid.
>-int I420PSNRFromFiles(const char* ref_filename,
>-                      const char* test_filename,
>-                      int width,
>-                      int height,
>-                      QualityMetricsResult* result);
>-
>-// Calculates SSIM values for the reference and test video files (must be in
>-// I420 format). All calculated values are filled into the QualityMetricsResult
>-// struct.
>-// SSIM values range between -1.0 and 1.0, where 1.0 means the files are
>-// identical.
>-// This function only compares video frames up to the point when the shortest
>-// video ends.
>-// For more info about SSIM, see http://en.wikipedia.org/wiki/SSIM
>-//
>-// Return value:
>-//  0 if successful, negative on errors:
>-// -1 if the source file cannot be opened
>-// -2 if the test file cannot be opened
>-// -3 if any of the files are empty
>-// -4 if any arguments are invalid.
>-int I420SSIMFromFiles(const char* ref_filename,
>-                      const char* test_filename,
>-                      int width,
>-                      int height,
>-                      QualityMetricsResult* result);
>-
>-}  // namespace test
>-}  // namespace webrtc
>-
>-#endif // TESTSUPPORT_METRICS_VIDEO_METRICS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/metrics/video_metrics_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/metrics/video_metrics_unittest.cc
>deleted file mode 100644
>index 5b00458761f..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/metrics/video_metrics_unittest.cc
>+++ /dev/null
>@@ -1,147 +0,0 @@
>-/*
>- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "test/testsupport/metrics/video_metrics.h"
>-
>-#include "test/gtest.h"
>-#include "test/testsupport/fileutils.h"
>-
>-namespace webrtc {
>-
>-static const int kWidth = 352;
>-static const int kHeight = 288;
>-
>-static const int kMissingReferenceFileReturnCode = -1;
>-static const int kMissingTestFileReturnCode = -2;
>-static const int kEmptyFileReturnCode = -3;
>-static const double kPsnrPerfectResult =  48.0;
>-static const double kSsimPerfectResult = 1.0;
>-
>-class VideoMetricsTest: public testing::Test {
>- protected:
>-  VideoMetricsTest() {
>-    video_file_ = webrtc::test::ResourcePath("foreman_cif_short", "yuv");
>-  }
>-  virtual ~VideoMetricsTest() {}
>-  void SetUp() {
>-    non_existing_file_ = webrtc::test::OutputPath() +
>-        "video_metrics_unittest_non_existing";
>-    remove(non_existing_file_.c_str());  // To be sure it doesn't exist.
>-
>-    // Create an empty file:
>-    empty_file_ = webrtc::test::TempFilename(
>-        webrtc::test::OutputPath(), "video_metrics_unittest_empty_file");
>-    FILE* dummy = fopen(empty_file_.c_str(), "wb");
>-    fclose(dummy);
>-  }
>-  void TearDown() {
>-    remove(empty_file_.c_str());
>-  }
>-  webrtc::test::QualityMetricsResult psnr_result_;
>-  webrtc::test::QualityMetricsResult ssim_result_;
>-  std::string non_existing_file_;
>-  std::string empty_file_;
>-  std::string video_file_;
>-};
>-
>-// Tests that it is possible to run with the same reference as test file
>-TEST_F(VideoMetricsTest, ReturnsPerfectResultForIdenticalFilesPSNR) {
>-  EXPECT_EQ(0, I420PSNRFromFiles(video_file_.c_str(), video_file_.c_str(),
>-                                 kWidth, kHeight, &psnr_result_));
>-  EXPECT_EQ(kPsnrPerfectResult, psnr_result_.average);
>-}
>-
>-TEST_F(VideoMetricsTest, ReturnsPerfectResultForIdenticalFilesSSIM) {
>-  EXPECT_EQ(0, I420SSIMFromFiles(video_file_.c_str(), video_file_.c_str(),
>-                                 kWidth, kHeight, &ssim_result_));
>-  EXPECT_EQ(kSsimPerfectResult, ssim_result_.average);
>-}
>-
>-TEST_F(VideoMetricsTest, ReturnsPerfectResultForIdenticalFilesBothMetrics) {
>-  EXPECT_EQ(0, I420MetricsFromFiles(video_file_.c_str(), video_file_.c_str(),
>-                                    kWidth, kHeight, &psnr_result_,
>-                                    &ssim_result_));
>-  EXPECT_EQ(kPsnrPerfectResult, psnr_result_.average);
>-  EXPECT_EQ(kSsimPerfectResult, ssim_result_.average);
>-}
>-
>-// Tests that the right return code is given when the reference file is missing.
>-TEST_F(VideoMetricsTest, MissingReferenceFilePSNR) {
>-  EXPECT_EQ(kMissingReferenceFileReturnCode,
>-            I420PSNRFromFiles(non_existing_file_.c_str(),
>-                              video_file_.c_str(), kWidth, kHeight,
>-                              &ssim_result_));
>-}
>-
>-TEST_F(VideoMetricsTest, MissingReferenceFileSSIM) {
>-  EXPECT_EQ(kMissingReferenceFileReturnCode,
>-            I420SSIMFromFiles(non_existing_file_.c_str(),
>-                              video_file_.c_str(), kWidth, kHeight,
>-                              &ssim_result_));
>-}
>-
>-TEST_F(VideoMetricsTest, MissingReferenceFileBothMetrics) {
>-  EXPECT_EQ(kMissingReferenceFileReturnCode,
>-            I420MetricsFromFiles(non_existing_file_.c_str(),
>-                                 video_file_.c_str(), kWidth, kHeight,
>-                                 &psnr_result_, &ssim_result_));
>-}
>-
>-// Tests that the right return code is given when the test file is missing.
>-TEST_F(VideoMetricsTest, MissingTestFilePSNR) {
>-  EXPECT_EQ(kMissingTestFileReturnCode,
>-            I420PSNRFromFiles(video_file_.c_str(), non_existing_file_.c_str(),
>-                              kWidth, kHeight, &ssim_result_));
>-}
>-
>-TEST_F(VideoMetricsTest, MissingTestFileSSIM) {
>-  EXPECT_EQ(kMissingTestFileReturnCode,
>-            I420SSIMFromFiles(video_file_.c_str(), non_existing_file_.c_str(),
>-                              kWidth, kHeight, &ssim_result_));
>-}
>-
>-TEST_F(VideoMetricsTest, MissingTestFileBothMetrics) {
>-  EXPECT_EQ(kMissingTestFileReturnCode,
>-            I420MetricsFromFiles(video_file_.c_str(),
>-                                 non_existing_file_.c_str(), kWidth, kHeight,
>-                                 &psnr_result_, &ssim_result_));
>-}
>-
>-// Tests that the method can be executed with empty files.
>-TEST_F(VideoMetricsTest, EmptyFilesPSNR) {
>-  EXPECT_EQ(kEmptyFileReturnCode,
>-            I420PSNRFromFiles(empty_file_.c_str(), video_file_.c_str(),
>-                              kWidth, kHeight, &ssim_result_));
>-  EXPECT_EQ(kEmptyFileReturnCode,
>-            I420PSNRFromFiles(video_file_.c_str(), empty_file_.c_str(),
>-                              kWidth, kHeight, &ssim_result_));
>-}
>-
>-TEST_F(VideoMetricsTest, EmptyFilesSSIM) {
>-  EXPECT_EQ(kEmptyFileReturnCode,
>-            I420SSIMFromFiles(empty_file_.c_str(), video_file_.c_str(),
>-                              kWidth, kHeight, &ssim_result_));
>-  EXPECT_EQ(kEmptyFileReturnCode,
>-            I420SSIMFromFiles(video_file_.c_str(), empty_file_.c_str(),
>-                              kWidth, kHeight, &ssim_result_));
>-}
>-
>-TEST_F(VideoMetricsTest, EmptyFilesBothMetrics) {
>-  EXPECT_EQ(kEmptyFileReturnCode,
>-            I420MetricsFromFiles(empty_file_.c_str(), video_file_.c_str(),
>-                                 kWidth, kHeight,
>-                                 &psnr_result_, &ssim_result_));
>-  EXPECT_EQ(kEmptyFileReturnCode,
>-              I420MetricsFromFiles(video_file_.c_str(), empty_file_.c_str(),
>-                                   kWidth, kHeight,
>-                                   &psnr_result_, &ssim_result_));
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/packet_reader.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/packet_reader.cc
>deleted file mode 100644
>index ba52f94e578..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/packet_reader.cc
>+++ /dev/null
>@@ -1,49 +0,0 @@
>-/*
>- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "test/testsupport/packet_reader.h"
>-
>-#include <assert.h>
>-#include <stdio.h>
>-#include <algorithm>
>-
>-namespace webrtc {
>-namespace test {
>-
>-PacketReader::PacketReader()
>-    : initialized_(false) {}
>-
>-PacketReader::~PacketReader() {}
>-
>-void PacketReader::InitializeReading(uint8_t* data,
>-                                     size_t data_length_in_bytes,
>-                                     size_t packet_size_in_bytes) {
>-  assert(data);
>-  assert(packet_size_in_bytes > 0);
>-  data_ = data;
>-  data_length_ = data_length_in_bytes;
>-  packet_size_ = packet_size_in_bytes;
>-  currentIndex_ = 0;
>-  initialized_ = true;
>-}
>-
>-int PacketReader::NextPacket(uint8_t** packet_pointer) {
>-  if (!initialized_) {
>-    fprintf(stderr, "Attempting to use uninitialized PacketReader!\n");
>-    return -1;
>-  }
>-  *packet_pointer = data_ + currentIndex_;
>-  size_t old_index = currentIndex_;
>-  currentIndex_ = std::min(currentIndex_ + packet_size_, data_length_);
>-  return static_cast<int>(currentIndex_ - old_index);
>-}
>-
>-}  // namespace test
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/packet_reader.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/packet_reader.h
>deleted file mode 100644
>index dbdb3da9e9b..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/packet_reader.h
>+++ /dev/null
>@@ -1,54 +0,0 @@
>-/*
>- *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef TEST_TESTSUPPORT_PACKET_READER_H_
>-#define TEST_TESTSUPPORT_PACKET_READER_H_
>-
>-#include <cstddef>
>-#include "typedefs.h"  // NOLINT(build/include)
>-
>-namespace webrtc {
>-namespace test {
>-
>-// Reads chunks of data to simulate network packets from a byte array.
>-class PacketReader {
>- public:
>-  PacketReader();
>-  virtual ~PacketReader();
>-
>-  // Inizializes a new reading operation. Must be done before invoking the
>-  // NextPacket method.
>-  // * data_length_in_bytes is the length of the data byte array.
>-  //   0 length will result in no packets are read.
>-  // * packet_size_in_bytes is the number of bytes to read in each NextPacket
>-  //   method call. Must be > 0
>-  virtual void InitializeReading(uint8_t* data, size_t data_length_in_bytes,
>-                                 size_t packet_size_in_bytes);
>-
>-  // Moves the supplied pointer to the beginning of the next packet.
>-  // Returns:
>-  // *  The size of the packet ready to read (lower than the packet size for
>-  //    the last packet)
>-  // *  0 if there are no more packets to read
>-  // * -1 if InitializeReading has not been called (also prints to stderr).
>-  virtual int NextPacket(uint8_t** packet_pointer);
>-
>- private:
>-  uint8_t* data_;
>-  size_t data_length_;
>-  size_t packet_size_;
>-  size_t currentIndex_;
>-  bool initialized_;
>-};
>-
>-}  // namespace test
>-}  // namespace webrtc
>-
>-#endif  // TEST_TESTSUPPORT_PACKET_READER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/packet_reader_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/packet_reader_unittest.cc
>deleted file mode 100644
>index 3255151d050..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/packet_reader_unittest.cc
>+++ /dev/null
>@@ -1,125 +0,0 @@
>-/*
>- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "test/testsupport/packet_reader.h"
>-
>-#include "test/gtest.h"
>-#include "test/testsupport/unittest_utils.h"
>-
>-namespace webrtc {
>-namespace test {
>-
>-class PacketReaderTest: public PacketRelatedTest {
>- protected:
>-  PacketReaderTest() {}
>-  virtual ~PacketReaderTest() {}
>-  void SetUp() {
>-    reader_ = new PacketReader();
>-  }
>-  void TearDown() {
>-    delete reader_;
>-  }
>-  void VerifyPacketData(size_t expected_length,
>-                        int actual_length,
>-                        uint8_t* original_data_pointer,
>-                        uint8_t* new_data_pointer) {
>-    EXPECT_EQ(static_cast<int>(expected_length), actual_length);
>-    EXPECT_EQ(*original_data_pointer, *new_data_pointer);
>-    EXPECT_EQ(0, memcmp(original_data_pointer, new_data_pointer,
>-                        actual_length));
>-  }
>-  PacketReader* reader_;
>-};
>-
>-// Test lack of initialization
>-TEST_F(PacketReaderTest, Uninitialized) {
>-  uint8_t* data_pointer = NULL;
>-  EXPECT_EQ(-1, reader_->NextPacket(&data_pointer));
>-  EXPECT_EQ(NULL, data_pointer);
>-}
>-
>-TEST_F(PacketReaderTest, InitializeZeroLengthArgument) {
>-  reader_->InitializeReading(packet_data_, 0, kPacketSizeInBytes);
>-  ASSERT_EQ(0, reader_->NextPacket(&packet_data_pointer_));
>-}
>-
>-// Test with something smaller than one packet
>-TEST_F(PacketReaderTest, NormalSmallData) {
>-  const int kDataLengthInBytes = 1499;
>-  uint8_t data[kDataLengthInBytes];
>-  uint8_t* data_pointer = data;
>-  memset(data, 1, kDataLengthInBytes);
>-
>-  reader_->InitializeReading(data, kDataLengthInBytes, kPacketSizeInBytes);
>-  int length_to_read = reader_->NextPacket(&data_pointer);
>-  VerifyPacketData(kDataLengthInBytes, length_to_read, data, data_pointer);
>-  EXPECT_EQ(0, data_pointer - data);  // pointer hasn't moved
>-
>-  // Reading another one shall result in 0 bytes:
>-  length_to_read = reader_->NextPacket(&data_pointer);
>-  EXPECT_EQ(0, length_to_read);
>-  EXPECT_EQ(kDataLengthInBytes, data_pointer - data);
>-}
>-
>-// Test with data length that exactly matches one packet
>-TEST_F(PacketReaderTest, NormalOnePacketData) {
>-  uint8_t data[kPacketSizeInBytes];
>-  uint8_t* data_pointer = data;
>-  memset(data, 1, kPacketSizeInBytes);
>-
>-  reader_->InitializeReading(data, kPacketSizeInBytes, kPacketSizeInBytes);
>-  int length_to_read = reader_->NextPacket(&data_pointer);
>-  VerifyPacketData(kPacketSizeInBytes, length_to_read, data, data_pointer);
>-  EXPECT_EQ(0, data_pointer - data);  // pointer hasn't moved
>-
>-  // Reading another one shall result in 0 bytes:
>-  length_to_read = reader_->NextPacket(&data_pointer);
>-  EXPECT_EQ(0, length_to_read);
>-  EXPECT_EQ(kPacketSizeInBytes, static_cast<size_t>(data_pointer - data));
>-}
>-
>-// Test with data length that will result in 3 packets
>-TEST_F(PacketReaderTest, NormalLargeData) {
>-  reader_->InitializeReading(packet_data_, kPacketDataLength,
>-                             kPacketSizeInBytes);
>-
>-  int length_to_read = reader_->NextPacket(&packet_data_pointer_);
>-  VerifyPacketData(kPacketSizeInBytes, length_to_read,
>-                   packet1_, packet_data_pointer_);
>-
>-  length_to_read = reader_->NextPacket(&packet_data_pointer_);
>-  VerifyPacketData(kPacketSizeInBytes, length_to_read,
>-                   packet2_, packet_data_pointer_);
>-
>-  length_to_read = reader_->NextPacket(&packet_data_pointer_);
>-  VerifyPacketData(1u, length_to_read,
>-                   packet3_, packet_data_pointer_);
>-
>-  // Reading another one shall result in 0 bytes:
>-  length_to_read = reader_->NextPacket(&packet_data_pointer_);
>-  EXPECT_EQ(0, length_to_read);
>-  EXPECT_EQ(kPacketDataLength,
>-            static_cast<size_t>(packet_data_pointer_ - packet_data_));
>-}
>-
>-// Test with empty data.
>-TEST_F(PacketReaderTest, EmptyData) {
>-  const int kDataLengthInBytes = 0;
>-  // But don't really try to allocate a zero-length array...
>-  uint8_t data[kPacketSizeInBytes];
>-  uint8_t* data_pointer = data;
>-  reader_->InitializeReading(data, kDataLengthInBytes, kPacketSizeInBytes);
>-  EXPECT_EQ(kDataLengthInBytes, reader_->NextPacket(&data_pointer));
>-  // Do it again to make sure nothing changes
>-  EXPECT_EQ(kDataLengthInBytes, reader_->NextPacket(&data_pointer));
>-}
>-
>-}  // namespace test
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/perf_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/perf_test.cc
>index b7578d101e0..c59c07149c7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/perf_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/perf_test.cc
>@@ -8,32 +8,145 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-// A stripped-down version of Chromium's chrome/test/perf/perf_test.cc.
>-// ResultsToString(), PrintResult(size_t value) and AppendResult(size_t value)
>-// have been modified. The remainder are identical to the Chromium version.
>-
> #include "test/testsupport/perf_test.h"
>+#include "rtc_base/criticalsection.h"
> 
>-#include <sstream>
> #include <stdio.h>
>+#include <fstream>
>+#include <map>
>+#include <sstream>
> #include <vector>
> 
> namespace {
> 
>-void PrintResultsImpl(const std::string& graph_name,
>+template <typename Container>
>+void OutputListToStream(std::ostream* ostream, const Container& values) {
>+  const char* sep = "";
>+  for (const auto& v : values) {
>+    (*ostream) << sep << v;
>+    sep = ",";
>+  }
>+}
>+
>+class PerfResultsLogger {
>+ public:
>+  PerfResultsLogger() : crit_(), output_(stdout), graphs_() {}
>+  void ClearResults() {
>+    rtc::CritScope lock(&crit_);
>+    graphs_.clear();
>+  }
>+  void SetOutput(FILE* output) {
>+    rtc::CritScope lock(&crit_);
>+    output_ = output;
>+  }
>+  void LogResult(const std::string& graph_name,
>+                 const std::string& trace_name,
>+                 const double value,
>+                 const std::string& units,
>+                 const bool important) {
>+    std::ostringstream value_stream;
>+    value_stream.precision(8);
>+    value_stream << value;
>+    LogResultsImpl(graph_name, trace_name, value_stream.str(), units,
>+                   important);
>+
>+    std::ostringstream json_stream;
>+    json_stream << '"' << trace_name << R"(":{)";
>+    json_stream << R"("type":"scalar",)";
>+    json_stream << R"("value":)" << value << ',';
>+    json_stream << R"("units":")" << units << R"("})";
>+    rtc::CritScope lock(&crit_);
>+    graphs_[graph_name].push_back(json_stream.str());
>+  }
>+  void LogResultMeanAndError(const std::string& graph_name,
>+                             const std::string& trace_name,
>+                             const double mean,
>+                             const double error,
>+                             const std::string& units,
>+                             const bool important) {
>+    std::ostringstream value_stream;
>+    value_stream.precision(8);
>+    value_stream << '{' << mean << ',' << error << '}';
>+    LogResultsImpl(graph_name, trace_name, value_stream.str(), units,
>+                   important);
>+
>+    std::ostringstream json_stream;
>+    json_stream << '"' << trace_name << R"(":{)";
>+    json_stream << R"("type":"list_of_scalar_values",)";
>+    json_stream << R"("values":[)" << mean << "],";
>+    json_stream << R"("std":)" << error << ',';
>+    json_stream << R"("units":")" << units << R"("})";
>+    rtc::CritScope lock(&crit_);
>+    graphs_[graph_name].push_back(json_stream.str());
>+  }
>+  void LogResultList(const std::string& graph_name,
>+                     const std::string& trace_name,
>+                     const rtc::ArrayView<const double> values,
>+                     const std::string& units,
>+                     const bool important) {
>+    std::ostringstream value_stream;
>+    value_stream.precision(8);
>+    value_stream << '[';
>+    OutputListToStream(&value_stream, values);
>+    value_stream << ']';
>+    LogResultsImpl(graph_name, trace_name, value_stream.str(), units,
>+                   important);
>+
>+    std::ostringstream json_stream;
>+    json_stream << '"' << trace_name << R"(":{)";
>+    json_stream << R"("type":"list_of_scalar_values",)";
>+    json_stream << R"("values":)" << value_stream.str() << ',';
>+    json_stream << R"("units":")" << units << R"("})";
>+    rtc::CritScope lock(&crit_);
>+    graphs_[graph_name].push_back(json_stream.str());
>+  }
>+  std::string ToJSON() const;
>+
>+ private:
>+  void LogResultsImpl(const std::string& graph_name,
>                       const std::string& trace,
>                       const std::string& values,
>                       const std::string& units,
>                       bool important) {
>-  // <*>RESULT <graph_name>: <trace_name>= <value> <units>
>-  // <*>RESULT <graph_name>: <trace_name>= {<mean>, <std deviation>} <units>
>-  // <*>RESULT <graph_name>: <trace_name>= [<value>,value,value,...,] <units>
>+    // <*>RESULT <graph_name>: <trace_name>= <value> <units>
>+    // <*>RESULT <graph_name>: <trace_name>= {<mean>, <std deviation>} <units>
>+    // <*>RESULT <graph_name>: <trace_name>= [<value>,value,value,...,] <units>
>+    rtc::CritScope lock(&crit_);
>+
>+    if (important) {
>+      fprintf(output_, "*");
>+    }
>+    fprintf(output_, "RESULT %s: %s= %s %s\n", graph_name.c_str(),
>+            trace.c_str(), values.c_str(), units.c_str());
>+  }
>+
>+  rtc::CriticalSection crit_;
>+  FILE* output_ RTC_GUARDED_BY(&crit_);
>+  std::map<std::string, std::vector<std::string>> graphs_
>+      RTC_GUARDED_BY(&crit_);
>+};
> 
>-  if (important) {
>-    printf("*");
>+std::string PerfResultsLogger::ToJSON() const {
>+  std::ostringstream json_stream;
>+  json_stream << R"({"format_version":"1.0",)";
>+  json_stream << R"("charts":{)";
>+  rtc::CritScope lock(&crit_);
>+  for (auto graphs_it = graphs_.begin(); graphs_it != graphs_.end();
>+       ++graphs_it) {
>+    if (graphs_it != graphs_.begin())
>+      json_stream << ',';
>+    json_stream << '"' << graphs_it->first << "\":";
>+    json_stream << '{';
>+    OutputListToStream(&json_stream, graphs_it->second);
>+    json_stream << '}';
>   }
>-  printf("RESULT %s: %s= %s %s\n", graph_name.c_str(), trace.c_str(),
>-         values.c_str(), units.c_str());
>+  json_stream << "}}";
>+  return json_stream.str();
>+}
>+
>+PerfResultsLogger& GetPerfResultsLogger() {
>+  static PerfResultsLogger* const logger_ = new PerfResultsLogger();
>+  return *logger_;
> }
> 
> }  // namespace
>@@ -41,16 +154,33 @@ void PrintResultsImpl(const std::string& graph_name,
> namespace webrtc {
> namespace test {
> 
>+void ClearPerfResults() {
>+  GetPerfResultsLogger().ClearResults();
>+}
>+
>+void SetPerfResultsOutput(FILE* output) {
>+  GetPerfResultsLogger().SetOutput(output);
>+}
>+
>+std::string GetPerfResultsJSON() {
>+  return GetPerfResultsLogger().ToJSON();
>+}
>+
>+void WritePerfResults(const std::string& output_path) {
>+  std::string json_results = GetPerfResultsJSON();
>+  std::fstream json_file(output_path, std::fstream::out);
>+  json_file << json_results;
>+  json_file.close();
>+}
>+
> void PrintResult(const std::string& measurement,
>                  const std::string& modifier,
>                  const std::string& trace,
>                  const double value,
>                  const std::string& units,
>                  bool important) {
>-  std::ostringstream value_stream;
>-  value_stream << value;
>-  PrintResultsImpl(measurement + modifier, trace, value_stream.str(), units,
>-                   important);
>+  GetPerfResultsLogger().LogResult(measurement + modifier, trace, value, units,
>+                                   important);
> }
> 
> void PrintResultMeanAndError(const std::string& measurement,
>@@ -60,32 +190,18 @@ void PrintResultMeanAndError(const std::string& measurement,
>                              const double error,
>                              const std::string& units,
>                              bool important) {
>-  std::ostringstream value_stream;
>-  value_stream << '{' << mean << ',' << error << '}';
>-  PrintResultsImpl(measurement + modifier, trace, value_stream.str(), units,
>-                   important);
>+  GetPerfResultsLogger().LogResultMeanAndError(measurement + modifier, trace,
>+                                               mean, error, units, important);
> }
> 
> void PrintResultList(const std::string& measurement,
>                      const std::string& modifier,
>                      const std::string& trace,
>-                     const std::vector<double>& values,
>+                     const rtc::ArrayView<const double> values,
>                      const std::string& units,
>                      bool important) {
>-  std::ostringstream value_stream;
>-  value_stream << '[';
>-  if (!values.empty()) {
>-    auto it = values.begin();
>-    while (true) {
>-      value_stream << *it;
>-      if (++it == values.end())
>-        break;
>-      value_stream << ',';
>-    }
>-  }
>-  value_stream << ']';
>-  PrintResultsImpl(measurement + modifier, trace, value_stream.str(), units,
>-                   important);
>+  GetPerfResultsLogger().LogResultList(measurement + modifier, trace, values,
>+                                       units, important);
> }
> 
> }  // namespace test
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/perf_test.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/perf_test.h
>index 31b7b1345cd..4f718bb5e70 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/perf_test.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/perf_test.h
>@@ -8,16 +8,13 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-// A stripped-down version of Chromium's chrome/test/perf/perf_test.h.
>-// Several functions have been removed; the prototypes of the remainder have
>-// not been changed.
>-
> #ifndef TEST_TESTSUPPORT_PERF_TEST_H_
> #define TEST_TESTSUPPORT_PERF_TEST_H_
> 
>+#include "api/array_view.h"
>+
> #include <sstream>
> #include <string>
>-#include <vector>
> 
> namespace webrtc {
> namespace test {
>@@ -53,7 +50,6 @@ void PrintResultMeanAndError(const std::string& measurement,
>                              const std::string& units,
>                              bool important);
> 
>-
> // Like PrintResult(), but prints an entire list of results. The |values|
> // will generally be a list of comma-separated numbers. A typical
> // post-processing step might produce plots of their mean and standard
>@@ -61,10 +57,25 @@ void PrintResultMeanAndError(const std::string& measurement,
> void PrintResultList(const std::string& measurement,
>                      const std::string& modifier,
>                      const std::string& trace,
>-                     const std::vector<double>& values,
>+                     rtc::ArrayView<const double> values,
>                      const std::string& units,
>                      bool important);
> 
>+// Returns all perf results to date in a JSON string formatted as described in
>+// https://github.com/catapult-project/catapult/blob/master/dashboard/docs/data-format.md
>+std::string GetPerfResultsJSON();
>+
>+// Writes the JSON representation of the perf results returned by
>+// GetPerfResultsJSON() to the file in output_path.
>+void WritePerfResults(const std::string& output_path);
>+
>+// By default, perf results are printed to stdout. Set the FILE* to where they
>+// should be printing instead.
>+void SetPerfResultsOutput(FILE* output);
>+
>+// You shouldn't use this function. It's only used to test the functions above.
>+void ClearPerfResults();
>+
> }  // namespace test
> }  // namespace webrtc
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/perf_test_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/perf_test_unittest.cc
>index 466e2b3e693..ac2b8b97722 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/perf_test_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/perf_test_unittest.cc
>@@ -14,25 +14,94 @@
> 
> #include "test/gtest.h"
> 
>+namespace {
>+
>+const char* kJsonExpected = R"({
>+  "format_version":"1.0",
>+  "charts":{
>+    "foobar":{
>+      "baz_v":{
>+        "type":"scalar",
>+        "value":7,
>+        "units":"widgets"
>+      },
>+      "baz_me":{
>+        "type":"list_of_scalar_values",
>+        "values":[1],
>+        "std":2,
>+        "units":"lemurs"
>+      },
>+      "baz_vl":{
>+        "type":"list_of_scalar_values",
>+        "values":[1,2,3],
>+        "units":"units"
>+      }
>+    },
>+    "measurementmodifier":{
>+      "trace":{
>+        "type":"scalar",
>+        "value":42,
>+        "units":"units"
>+      }
>+    }
>+  }
>+})";
>+
>+std::string RemoveSpaces(std::string s) {
>+  s.erase(std::remove(s.begin(), s.end(), ' '), s.end());
>+  s.erase(std::remove(s.begin(), s.end(), '\n'), s.end());
>+  return s;
>+}
>+
>+}  // namespace
>+
> namespace webrtc {
> namespace test {
> 
>-TEST(PerfTest, AppendResult) {
>+class PerfTest : public ::testing::Test {
>+ protected:
>+  void TearDown() override { ClearPerfResults(); }
>+};
>+
>+#if defined(WEBRTC_IOS)
>+#define MAYBE_TestPrintResult DISABLED_TestPrintResult
>+#else
>+#define MAYBE_TestPrintResult TestPrintResult
>+#endif
>+TEST_F(PerfTest, MAYBE_TestPrintResult) {
>   testing::internal::CaptureStdout();
>-  std::string expected = "RESULT measurementmodifier: trace= 42 units\n";
>+  std::string expected;
>+
>+  expected += "RESULT measurementmodifier: trace= 42 units\n";
>   PrintResult("measurement", "modifier", "trace", 42, "units", false);
> 
>-  expected += "*RESULT foobar: baz= 7 widgets\n";
>-  PrintResult("foo", "bar", "baz", 7, "widgets", true);
>+  expected += "*RESULT foobar: baz_v= 1423730 widgets\n";
>+  PrintResult("foo", "bar", "baz_v", 1423730, "widgets", true);
> 
>-  expected += "RESULT foobar: baz= {1,2} lemurs\n";
>-  PrintResultMeanAndError("foo", "bar", "baz", 1, 2, "lemurs", false);
>+  expected += "RESULT foobar: baz_me= {1,2} lemurs\n";
>+  PrintResultMeanAndError("foo", "bar", "baz_me", 1, 2, "lemurs", false);
> 
>-  expected += "RESULT foobar: baz= [1,2,3] units\n";
>-  PrintResultList("foo", "bar", "baz", {1, 2, 3}, "units", false);
>+  const double kListOfScalars[] = {1, 2, 3};
>+  expected += "RESULT foobar: baz_vl= [1,2,3] units\n";
>+  PrintResultList("foo", "bar", "baz_vl", kListOfScalars, "units", false);
> 
>-  std::string output = testing::internal::GetCapturedStdout();
>-  EXPECT_EQ(expected, output);
>+  EXPECT_EQ(expected, testing::internal::GetCapturedStdout());
>+}
>+
>+TEST_F(PerfTest, TestGetPerfResultsJSON) {
>+  PrintResult("measurement", "modifier", "trace", 42, "units", false);
>+  PrintResult("foo", "bar", "baz_v", 7, "widgets", true);
>+  PrintResultMeanAndError("foo", "bar", "baz_me", 1, 2, "lemurs", false);
>+  const double kListOfScalars[] = {1, 2, 3};
>+  PrintResultList("foo", "bar", "baz_vl", kListOfScalars, "units", false);
>+
>+  EXPECT_EQ(RemoveSpaces(kJsonExpected), GetPerfResultsJSON());
>+}
>+
>+TEST_F(PerfTest, TestClearPerfResults) {
>+  PrintResult("measurement", "modifier", "trace", 42, "units", false);
>+  ClearPerfResults();
>+  EXPECT_EQ(R"({"format_version":"1.0","charts":{}})", GetPerfResultsJSON());
> }
> 
> }  // namespace test
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/test_artifacts.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/test_artifacts.cc
>index 3d97a944724..0f7e0cd4ff8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/test_artifacts.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/test_artifacts.cc
>@@ -15,7 +15,6 @@
> #include "rtc_base/file.h"
> #include "rtc_base/flags.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/pathutils.h"
> #include "test/testsupport/fileutils.h"
> 
> namespace {
>@@ -23,7 +22,7 @@ const std::string& DefaultArtifactPath() {
>   static const std::string path = webrtc::test::OutputPath();
>   return path;
> }
>-}
>+}  // namespace
> 
> DEFINE_string(test_artifacts_dir,
>               DefaultArtifactPath().c_str(),
>@@ -55,7 +54,7 @@ bool WriteToTestArtifactsDir(const char* filename,
>   }
> 
>   rtc::File output =
>-      rtc::File::Create(rtc::Pathname(FLAG_test_artifacts_dir, filename));
>+      rtc::File::Create(JoinFilename(FLAG_test_artifacts_dir, filename));
> 
>   return output.IsOpen() && output.Write(buffer, length) == length;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/test_artifacts_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/test_artifacts_unittest.cc
>index 251c5cde88c..c423cd90ead 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/test_artifacts_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/test_artifacts_unittest.cc
>@@ -19,6 +19,7 @@
> #include "rtc_base/pathutils.h"
> #include "rtc_base/platform_file.h"
> #include "test/gtest.h"
>+#include "test/testsupport/fileutils.h"
> 
> DECLARE_string(test_artifacts_dir);
> 
>@@ -42,7 +43,7 @@ TEST(IsolatedOutputTest, ShouldBeAbleToWriteContent) {
>   const char* filename = "a-file";
>   const char* content = "some-contents";
>   if (WriteToTestArtifactsDir(filename, content)) {
>-    rtc::Pathname out_file(FLAG_test_artifacts_dir, filename);
>+    std::string out_file = JoinFilename(FLAG_test_artifacts_dir, filename);
>     rtc::File input = rtc::File::Open(out_file);
>     EXPECT_TRUE(input.IsOpen());
>     EXPECT_TRUE(input.Seek(0));
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/unittest_utils.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/unittest_utils.h
>deleted file mode 100644
>index 5fee2ca454e..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/testsupport/unittest_utils.h
>+++ /dev/null
>@@ -1,56 +0,0 @@
>-/*
>- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef TEST_TESTSUPPORT_UNITTEST_UTILS_H_
>-#define TEST_TESTSUPPORT_UNITTEST_UTILS_H_
>-
>-namespace webrtc {
>-namespace test {
>-
>-const size_t kPacketSizeInBytes = 1500;
>-const size_t kPacketDataLength = kPacketSizeInBytes * 2 + 1;
>-const int kPacketDataNumberOfPackets = 3;
>-
>-// A base test fixture for packet related tests. Contains
>-// two full prepared packets with 1s, 2s in their data and a third packet with
>-// a single 3 in it (size=1).
>-// A packet data structure is also available, that contains these three packets
>-// in order.
>-class PacketRelatedTest: public testing::Test {
>- protected:
>-  // Tree packet byte arrays with data used for verification:
>-  uint8_t packet1_[kPacketSizeInBytes];
>-  uint8_t packet2_[kPacketSizeInBytes];
>-  uint8_t packet3_[1];
>-  // Construct a data structure containing these packets
>-  uint8_t packet_data_[kPacketDataLength];
>-  uint8_t* packet_data_pointer_;
>-
>-  PacketRelatedTest() {
>-    packet_data_pointer_ = packet_data_;
>-
>-    memset(packet1_, 1, kPacketSizeInBytes);
>-    memset(packet2_, 2, kPacketSizeInBytes);
>-    memset(packet3_, 3, 1);
>-    // Fill the packet_data:
>-    memcpy(packet_data_pointer_, packet1_, kPacketSizeInBytes);
>-    memcpy(packet_data_pointer_ + kPacketSizeInBytes, packet2_,
>-           kPacketSizeInBytes);
>-    memcpy(packet_data_pointer_ + kPacketSizeInBytes * 2, packet3_, 1);
>-  }
>-  virtual ~PacketRelatedTest() {}
>-  void SetUp() {}
>-  void TearDown() {}
>-};
>-
>-}  // namespace test
>-}  // namespace webrtc
>-
>-#endif  // TEST_TESTSUPPORT_UNITTEST_UTILS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/vcm_capturer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/vcm_capturer.cc
>index 82b5cbc4551..3e4e0f63c0c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/vcm_capturer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/vcm_capturer.cc
>@@ -10,9 +10,9 @@
> 
> #include "test/vcm_capturer.h"
> 
>+#include "call/video_send_stream.h"
> #include "modules/video_capture/video_capture_factory.h"
> #include "rtc_base/logging.h"
>-#include "call/video_send_stream.h"
> namespace webrtc {
> namespace test {
> 
>@@ -68,7 +68,6 @@ VcmCapturer* VcmCapturer::Create(size_t width,
>   return vcm_capturer.release();
> }
> 
>-
> void VcmCapturer::Start() {
>   rtc::CritScope lock(&crit_);
>   started_ = true;
>@@ -103,16 +102,18 @@ void VcmCapturer::Destroy() {
>   vcm_ = nullptr;
> }
> 
>-VcmCapturer::~VcmCapturer() { Destroy(); }
>+VcmCapturer::~VcmCapturer() {
>+  Destroy();
>+}
> 
> void VcmCapturer::OnFrame(const VideoFrame& frame) {
>   rtc::CritScope lock(&crit_);
>   if (started_ && sink_) {
>-    rtc::Optional<VideoFrame> out_frame = AdaptFrame(frame);
>+    absl::optional<VideoFrame> out_frame = AdaptFrame(frame);
>     if (out_frame)
>       sink_->OnFrame(*out_frame);
>   }
> }
> 
>-}  // test
>-}  // webrtc
>+}  // namespace test
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/vcm_capturer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/vcm_capturer.h
>index 9d158c3fcde..524cfe40382 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/vcm_capturer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/vcm_capturer.h
>@@ -22,9 +22,8 @@
> namespace webrtc {
> namespace test {
> 
>-class VcmCapturer
>-    : public VideoCapturer,
>-      public rtc::VideoSinkInterface<VideoFrame> {
>+class VcmCapturer : public VideoCapturer,
>+                    public rtc::VideoSinkInterface<VideoFrame> {
>  public:
>   static VcmCapturer* Create(size_t width,
>                              size_t height,
>@@ -55,7 +54,7 @@ class VcmCapturer
>   VideoCaptureCapability capability_;
> };
> 
>-}  // test
>-}  // webrtc
>+}  // namespace test
>+}  // namespace webrtc
> 
> #endif  // TEST_VCM_CAPTURER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/video_capturer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/video_capturer.cc
>index b95936feb92..4faf4491cbb 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/video_capturer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/video_capturer.cc
>@@ -10,7 +10,6 @@
> 
> #include "test/video_capturer.h"
> 
>-#include "rtc_base/basictypes.h"
> #include "rtc_base/constructormagic.h"
> 
> namespace webrtc {
>@@ -18,7 +17,7 @@ namespace test {
> VideoCapturer::VideoCapturer() : video_adapter_(new cricket::VideoAdapter()) {}
> VideoCapturer::~VideoCapturer() {}
> 
>-rtc::Optional<VideoFrame> VideoCapturer::AdaptFrame(const VideoFrame& frame) {
>+absl::optional<VideoFrame> VideoCapturer::AdaptFrame(const VideoFrame& frame) {
>   int cropped_width = 0;
>   int cropped_height = 0;
>   int out_width = 0;
>@@ -28,10 +27,10 @@ rtc::Optional<VideoFrame> VideoCapturer::AdaptFrame(const VideoFrame& frame) {
>           frame.width(), frame.height(), frame.timestamp_us() * 1000,
>           &cropped_width, &cropped_height, &out_width, &out_height)) {
>     // Drop frame in order to respect frame rate constraint.
>-    return rtc::Optional<VideoFrame>();
>+    return absl::nullopt;
>   }
> 
>-  rtc::Optional<VideoFrame> out_frame;
>+  absl::optional<VideoFrame> out_frame;
>   if (out_height != frame.height() || out_width != frame.width()) {
>     // Video adapter has requested a down-scale. Allocate a new buffer and
>     // return scaled version.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/video_capturer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/video_capturer.h
>index eba5d611116..d117b9682b1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/video_capturer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/video_capturer.h
>@@ -14,11 +14,11 @@
> 
> #include <memory>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> #include "api/video/i420_buffer.h"
> #include "api/video/video_frame.h"
>+#include "api/video/video_source_interface.h"
> #include "media/base/videoadapter.h"
>-#include "media/base/videosourceinterface.h"
> #include "rtc_base/criticalsection.h"
> 
> namespace cricket {
>@@ -41,7 +41,7 @@ class VideoCapturer : public rtc::VideoSourceInterface<VideoFrame> {
>                        const rtc::VideoSinkWants& wants) override;
> 
>  protected:
>-  rtc::Optional<VideoFrame> AdaptFrame(const VideoFrame& frame);
>+  absl::optional<VideoFrame> AdaptFrame(const VideoFrame& frame);
>   rtc::VideoSinkWants GetSinkWants();
> 
>  private:
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/video_codec_settings.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/video_codec_settings.h
>index 54cf717a77c..dadc35c69d8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/video_codec_settings.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/video_codec_settings.h
>@@ -26,82 +26,50 @@ const uint16_t kTestOutlierFrameSizePercent = 250;
> 
> static void CodecSettings(VideoCodecType codec_type, VideoCodec* settings) {
>   memset(settings, 0, sizeof(VideoCodec));
>+
>+  settings->plType = kTestPayloadType;
>+
>+  settings->width = kTestWidth;
>+  settings->height = kTestHeight;
>+
>+  settings->startBitrate = kTestStartBitrateKbps;
>+  settings->maxBitrate = 0;
>+  settings->minBitrate = kTestMinBitrateKbps;
>+  settings->targetBitrate = 0;
>+
>+  settings->maxFramerate = kTestFrameRate;
>+
>+  settings->active = true;
>+
>+  settings->qpMax = 56;  // See webrtcvideoengine.h.
>+  settings->numberOfSimulcastStreams = 0;
>+
>+  settings->timing_frame_thresholds = {
>+      kTestTimingFramesDelayMs, kTestOutlierFrameSizePercent,
>+  };
>+
>   switch (codec_type) {
>     case kVideoCodecVP8:
>-      strncpy(settings->plName, "VP8", 4);
>       settings->codecType = kVideoCodecVP8;
>-      // 96 to 127 dynamic payload types for video codecs.
>-      settings->plType = kTestPayloadType;
>-      settings->startBitrate = kTestStartBitrateKbps;
>-      settings->minBitrate = kTestMinBitrateKbps;
>-      settings->maxBitrate = 0;
>-      settings->maxFramerate = kTestFrameRate;
>-      settings->width = kTestWidth;
>-      settings->height = kTestHeight;
>-      settings->numberOfSimulcastStreams = 0;
>-      settings->qpMax = 56;
>-      settings->timing_frame_thresholds = {
>-          kTestTimingFramesDelayMs, kTestOutlierFrameSizePercent,
>-      };
>       *(settings->VP8()) = VideoEncoder::GetDefaultVp8Settings();
>       return;
>     case kVideoCodecVP9:
>-      strncpy(settings->plName, "VP9", 4);
>       settings->codecType = kVideoCodecVP9;
>-      // 96 to 127 dynamic payload types for video codecs.
>-      settings->plType = kTestPayloadType;
>-      settings->startBitrate = 100;
>-      settings->minBitrate = kTestMinBitrateKbps;
>-      settings->maxBitrate = 0;
>-      settings->maxFramerate = kTestFrameRate;
>-      settings->width = kTestWidth;
>-      settings->height = kTestHeight;
>-      settings->numberOfSimulcastStreams = 0;
>-      settings->qpMax = 56;
>-      settings->timing_frame_thresholds = {
>-          kTestTimingFramesDelayMs, kTestOutlierFrameSizePercent,
>-      };
>       *(settings->VP9()) = VideoEncoder::GetDefaultVp9Settings();
>       return;
>     case kVideoCodecH264:
>-      strncpy(settings->plName, "H264", 5);
>       settings->codecType = kVideoCodecH264;
>-      // 96 to 127 dynamic payload types for video codecs.
>-      settings->plType = kTestPayloadType;
>-      settings->startBitrate = kTestStartBitrateKbps;
>-      settings->minBitrate = kTestMinBitrateKbps;
>-      settings->maxBitrate = 0;
>-      settings->maxFramerate = kTestFrameRate;
>-      settings->width = kTestWidth;
>-      settings->height = kTestHeight;
>-      settings->numberOfSimulcastStreams = 0;
>-      settings->qpMax = 56;
>-      settings->timing_frame_thresholds = {
>-          kTestTimingFramesDelayMs, kTestOutlierFrameSizePercent,
>-      };
>+      // TODO(brandtr): Set |qpMax| here, when the OpenH264 wrapper supports it.
>       *(settings->H264()) = VideoEncoder::GetDefaultH264Settings();
>       return;
>     case kVideoCodecI420:
>-      strncpy(settings->plName, "I420", 5);
>       settings->codecType = kVideoCodecI420;
>-      // 96 to 127 dynamic payload types for video codecs.
>-      settings->plType = kTestPayloadType;
>       // Bitrate needed for this size and framerate.
>       settings->startBitrate =
>           3 * kTestWidth * kTestHeight * 8 * kTestFrameRate / 1000 / 2;
>       settings->maxBitrate = settings->startBitrate;
>-      settings->maxFramerate = kTestFrameRate;
>-      settings->width = kTestWidth;
>-      settings->height = kTestHeight;
>-      settings->minBitrate = kTestMinBitrateKbps;
>-      settings->numberOfSimulcastStreams = 0;
>       return;
>-    case kVideoCodecRED:
>-    case kVideoCodecULPFEC:
>-    case kVideoCodecFlexfec:
>-    case kVideoCodecStereo:
>-    case kVideoCodecGeneric:
>-    case kVideoCodecUnknown:
>+    default:
>       RTC_NOTREACHED();
>       return;
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/video_renderer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/video_renderer.cc
>index 68a20f3752f..75ab72a8839 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/video_renderer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/video_renderer.cc
>@@ -11,8 +11,6 @@
> 
> // TODO(pbos): Android renderer
> 
>-#include "typedefs.h"  // NOLINT(build/include)
>-
> namespace webrtc {
> namespace test {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/video_renderer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/video_renderer.h
>index 6a00e8803b3..9e580f6f598 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/video_renderer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/video_renderer.h
>@@ -12,7 +12,7 @@
> 
> #include <stddef.h>
> 
>-#include "media/base/videosinkinterface.h"
>+#include "api/video/video_sink_interface.h"
> 
> namespace webrtc {
> class VideoFrame;
>@@ -22,7 +22,8 @@ class VideoRenderer : public rtc::VideoSinkInterface<VideoFrame> {
>  public:
>   // Creates a platform-specific renderer if possible, or a null implementation
>   // if failing.
>-  static VideoRenderer* Create(const char* window_title, size_t width,
>+  static VideoRenderer* Create(const char* window_title,
>+                               size_t width,
>                                size_t height);
>   // Returns a renderer rendering to a platform specific window if possible,
>   // NULL if none can be created.
>@@ -30,8 +31,10 @@ class VideoRenderer : public rtc::VideoSinkInterface<VideoFrame> {
>   // platform renderer could not be created. This occurs, for instance, when
>   // running without an X environment on Linux.
>   static VideoRenderer* CreatePlatformRenderer(const char* window_title,
>-                                               size_t width, size_t height);
>+                                               size_t width,
>+                                               size_t height);
>   virtual ~VideoRenderer() {}
>+
>  protected:
>   VideoRenderer() {}
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/win/d3d_renderer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/win/d3d_renderer.cc
>index 1f791a91ef4..18c6cd78a97 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/win/d3d_renderer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/win/d3d_renderer.cc
>@@ -15,7 +15,7 @@
> namespace webrtc {
> namespace test {
> 
>-#define D3DFVF_CUSTOMVERTEX (D3DFVF_XYZ|D3DFVF_TEX1)
>+#define D3DFVF_CUSTOMVERTEX (D3DFVF_XYZ | D3DFVF_TEX1)
> 
> struct D3dCustomVertex {
>   float x, y, z;
>@@ -42,9 +42,13 @@ D3dRenderer::D3dRenderer(size_t width, size_t height)
>   RTC_DCHECK_GT(height, 0);
> }
> 
>-D3dRenderer::~D3dRenderer() { Destroy(); }
>+D3dRenderer::~D3dRenderer() {
>+  Destroy();
>+}
> 
>-LRESULT WINAPI D3dRenderer::WindowProc(HWND hwnd, UINT msg, WPARAM wparam,
>+LRESULT WINAPI D3dRenderer::WindowProc(HWND hwnd,
>+                                       UINT msg,
>+                                       WPARAM wparam,
>                                        LPARAM lparam) {
>   if (msg == WM_DESTROY || (msg == WM_CHAR && wparam == VK_RETURN)) {
>     PostQuitMessage(0);
>@@ -68,17 +72,9 @@ void D3dRenderer::Destroy() {
> }
> 
> bool D3dRenderer::Init(const char* window_title) {
>-  hwnd_ = CreateWindowA(kD3DClassName,
>-                        window_title,
>-                        WS_OVERLAPPEDWINDOW,
>-                        0,
>-                        0,
>-                        static_cast<int>(width_),
>-                        static_cast<int>(height_),
>-                        NULL,
>-                        NULL,
>-                        NULL,
>-                        NULL);
>+  hwnd_ = CreateWindowA(kD3DClassName, window_title, WS_OVERLAPPEDWINDOW, 0, 0,
>+                        static_cast<int>(width_), static_cast<int>(height_),
>+                        NULL, NULL, NULL, NULL);
> 
>   if (hwnd_ == NULL) {
>     Destroy();
>@@ -97,11 +93,8 @@ bool D3dRenderer::Init(const char* window_title) {
>   d3d_params.SwapEffect = D3DSWAPEFFECT_COPY;
> 
>   IDirect3DDevice9* d3d_device;
>-  if (d3d_->CreateDevice(D3DADAPTER_DEFAULT,
>-                         D3DDEVTYPE_HAL,
>-                         hwnd_,
>-                         D3DCREATE_SOFTWARE_VERTEXPROCESSING,
>-                         &d3d_params,
>+  if (d3d_->CreateDevice(D3DADAPTER_DEFAULT, D3DDEVTYPE_HAL, hwnd_,
>+                         D3DCREATE_SOFTWARE_VERTEXPROCESSING, &d3d_params,
>                          &d3d_device) != D3D_OK) {
>     Destroy();
>     return false;
>@@ -112,11 +105,8 @@ bool D3dRenderer::Init(const char* window_title) {
>   IDirect3DVertexBuffer9* vertex_buffer;
>   const int kRectVertices = 4;
>   if (d3d_device_->CreateVertexBuffer(kRectVertices * sizeof(D3dCustomVertex),
>-                                      0,
>-                                      D3DFVF_CUSTOMVERTEX,
>-                                      D3DPOOL_MANAGED,
>-                                      &vertex_buffer,
>-                                      NULL) != D3D_OK) {
>+                                      0, D3DFVF_CUSTOMVERTEX, D3DPOOL_MANAGED,
>+                                      &vertex_buffer, NULL) != D3D_OK) {
>     Destroy();
>     return false;
>   }
>@@ -148,13 +138,13 @@ D3dRenderer* D3dRenderer::Create(const char* window_title,
> 
>     wc_atom = RegisterClassA(&wc);
>     if (wc_atom == 0)
>-      return false;
>+      return nullptr;
>   }
> 
>   D3dRenderer* d3d_renderer = new D3dRenderer(width, height);
>   if (!d3d_renderer->Init(window_title)) {
>     delete d3d_renderer;
>-    return NULL;
>+    return nullptr;
>   }
> 
>   return d3d_renderer;
>@@ -166,22 +156,17 @@ void D3dRenderer::Resize(size_t width, size_t height) {
>   IDirect3DTexture9* texture;
> 
>   d3d_device_->CreateTexture(static_cast<UINT>(width_),
>-                             static_cast<UINT>(height_),
>-                             1,
>-                             0,
>-                             D3DFMT_A8R8G8B8,
>-                             D3DPOOL_MANAGED,
>-                             &texture,
>-                             NULL);
>+                             static_cast<UINT>(height_), 1, 0, D3DFMT_A8R8G8B8,
>+                             D3DPOOL_MANAGED, &texture, NULL);
>   texture_ = texture;
>   texture->Release();
> 
>   // Vertices for the video frame to be rendered to.
>   static const D3dCustomVertex rect[] = {
>-    {-1.0f, -1.0f, 0.0f, 0.0f, 1.0f},
>-    {-1.0f, 1.0f, 0.0f, 0.0f, 0.0f},
>-    {1.0f, -1.0f, 0.0f, 1.0f, 1.0f},
>-    {1.0f, 1.0f, 0.0f, 1.0f, 0.0f},
>+      {-1.0f, -1.0f, 0.0f, 0.0f, 1.0f},
>+      {-1.0f, 1.0f, 0.0f, 0.0f, 0.0f},
>+      {1.0f, -1.0f, 0.0f, 1.0f, 1.0f},
>+      {1.0f, 1.0f, 0.0f, 1.0f, 0.0f},
>   };
> 
>   void* buf_data;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/win/d3d_renderer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/test/win/d3d_renderer.h
>index 68a5a802f2e..a43be447f69 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/win/d3d_renderer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/win/d3d_renderer.h
>@@ -12,18 +12,18 @@
> 
> #include <Windows.h>
> #include <d3d9.h>
>-#pragma comment(lib, "d3d9.lib")       // located in DirectX SDK
>+#pragma comment(lib, "d3d9.lib")  // located in DirectX SDK
> 
> #include "rtc_base/scoped_ref_ptr.h"
> #include "test/video_renderer.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> namespace test {
> 
> class D3dRenderer : public VideoRenderer {
>  public:
>-  static D3dRenderer* Create(const char* window_title, size_t width,
>+  static D3dRenderer* Create(const char* window_title,
>+                             size_t width,
>                              size_t height);
>   virtual ~D3dRenderer();
> 
>@@ -32,7 +32,9 @@ class D3dRenderer : public VideoRenderer {
>  private:
>   D3dRenderer(size_t width, size_t height);
> 
>-  static LRESULT WINAPI WindowProc(HWND hwnd, UINT msg, WPARAM wparam,
>+  static LRESULT WINAPI WindowProc(HWND hwnd,
>+                                   UINT msg,
>+                                   WPARAM wparam,
>                                    LPARAM lparam);
>   bool Init(const char* window_title);
>   void Resize(size_t width, size_t height);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/win/run_loop_win.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/win/run_loop_win.cc
>index 494adc47d60..28dc9334e08 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/win/run_loop_win.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/win/run_loop_win.cc
>@@ -11,9 +11,9 @@
> 
> #include <assert.h>
> 
>+#include <Windows.h>
> #include <conio.h>
> #include <stdio.h>
>-#include <Windows.h>
> 
> namespace webrtc {
> namespace test {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/OWNERS
>deleted file mode 100644
>index e69de29bb2d..00000000000
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/author_line_count.sh b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/author_line_count.sh
>deleted file mode 100755
>index e69de29bb2d..00000000000
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/barcode_tools/DEPS b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/barcode_tools/DEPS
>deleted file mode 100644
>index e69de29bb2d..00000000000
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/barcode_tools/README b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/barcode_tools/README
>deleted file mode 100644
>index e69de29bb2d..00000000000
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/barcode_tools/barcode_decoder.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/barcode_tools/barcode_decoder.py
>deleted file mode 100755
>index e69de29bb2d..00000000000
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/barcode_tools/barcode_encoder.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/barcode_tools/barcode_encoder.py
>deleted file mode 100755
>index e69de29bb2d..00000000000
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/barcode_tools/build_zxing.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/barcode_tools/build_zxing.py
>deleted file mode 100755
>index e69de29bb2d..00000000000
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/barcode_tools/helper_functions.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/barcode_tools/helper_functions.py
>deleted file mode 100644
>index e69de29bb2d..00000000000
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/barcode_tools/yuv_cropper.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/barcode_tools/yuv_cropper.py
>deleted file mode 100755
>index e69de29bb2d..00000000000
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/class_usage.sh b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/class_usage.sh
>deleted file mode 100755
>index e69de29bb2d..00000000000
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/compare_videos.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/compare_videos.py
>deleted file mode 100755
>index e69de29bb2d..00000000000
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/event_log_visualizer/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/event_log_visualizer/OWNERS
>deleted file mode 100644
>index e69de29bb2d..00000000000
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/event_log_visualizer/chart.proto b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/event_log_visualizer/chart.proto
>deleted file mode 100644
>index e69de29bb2d..00000000000
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/header_usage.sh b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/header_usage.sh
>deleted file mode 100755
>index e69de29bb2d..00000000000
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/loopback_test/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/loopback_test/OWNERS
>deleted file mode 100644
>index e69de29bb2d..00000000000
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/loopback_test/README b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/loopback_test/README
>deleted file mode 100644
>index e69de29bb2d..00000000000
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/loopback_test/adapter.js b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/loopback_test/adapter.js
>deleted file mode 100644
>index e69de29bb2d..00000000000
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/loopback_test/loopback_test.html b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/loopback_test/loopback_test.html
>deleted file mode 100644
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>deleted file mode 100644
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>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/loopback_test/record-test.sh b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/loopback_test/record-test.sh
>deleted file mode 100755
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>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/loopback_test/run-server.sh b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/loopback_test/run-server.sh
>deleted file mode 100755
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>deleted file mode 100644
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>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/network_tester/androidapp/AndroidManifest.xml b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/network_tester/androidapp/AndroidManifest.xml
>deleted file mode 100755
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>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/network_tester/androidapp/res/layout/activity_main.xml b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/network_tester/androidapp/res/layout/activity_main.xml
>deleted file mode 100755
>index e69de29bb2d..00000000000
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/network_tester/androidapp/res/values-v17/styles.xml b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/network_tester/androidapp/res/values-v17/styles.xml
>deleted file mode 100755
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>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/network_tester/androidapp/res/values-w820dp/dimens.xml b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/network_tester/androidapp/res/values-w820dp/dimens.xml
>deleted file mode 100755
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>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/network_tester/androidapp/res/values/colors.xml b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/network_tester/androidapp/res/values/colors.xml
>deleted file mode 100755
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>deleted file mode 100755
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>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/network_tester/androidapp/res/values/strings.xml b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/network_tester/androidapp/res/values/strings.xml
>deleted file mode 100755
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>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/network_tester/androidapp/src/com/google/media/networktester/MainActivity.java b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/network_tester/androidapp/src/com/google/media/networktester/MainActivity.java
>deleted file mode 100644
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>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/network_tester/androidapp/src/com/google/media/networktester/NetworkTester.java b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/network_tester/androidapp/src/com/google/media/networktester/NetworkTester.java
>deleted file mode 100644
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>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/network_tester/create_network_tester_config.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/network_tester/create_network_tester_config.py
>deleted file mode 100644
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>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/network_tester/jni.cpp b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/network_tester/jni.cpp
>deleted file mode 100644
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>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/network_tester/network_tester_config.proto b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/network_tester/network_tester_config.proto
>deleted file mode 100644
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>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/network_tester/network_tester_packet.proto b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/network_tester/network_tester_packet.proto
>deleted file mode 100644
>index e69de29bb2d..00000000000
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/network_tester/parse_packet_log.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/network_tester/parse_packet_log.py
>deleted file mode 100755
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>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/py_event_log_analyzer/README b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/py_event_log_analyzer/README
>deleted file mode 100644
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>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/py_event_log_analyzer/misc.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/py_event_log_analyzer/misc.py
>deleted file mode 100644
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>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/py_event_log_analyzer/misc_test.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/py_event_log_analyzer/misc_test.py
>deleted file mode 100755
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>deleted file mode 100644
>index e69de29bb2d..00000000000
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/py_event_log_analyzer/rtp_analyzer.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/py_event_log_analyzer/rtp_analyzer.py
>deleted file mode 100644
>index e69de29bb2d..00000000000
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/py_event_log_analyzer/rtp_analyzer.sh b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/py_event_log_analyzer/rtp_analyzer.sh
>deleted file mode 100755
>index e69de29bb2d..00000000000
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/py_event_log_analyzer/rtp_analyzer_test.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/py_event_log_analyzer/rtp_analyzer_test.py
>deleted file mode 100755
>index e69de29bb2d..00000000000
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/rtcbot/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/rtcbot/OWNERS
>deleted file mode 100644
>index e69de29bb2d..00000000000
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/rtcbot/README b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/rtcbot/README
>deleted file mode 100644
>index e69de29bb2d..00000000000
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/rtcbot/bot/api.js b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/rtcbot/bot/api.js
>deleted file mode 100644
>index e69de29bb2d..00000000000
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/rtcbot/bot/browser/bot.js b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/rtcbot/bot/browser/bot.js
>deleted file mode 100644
>index e69de29bb2d..00000000000
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/rtcbot/bot/browser/index.html b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/rtcbot/bot/browser/index.html
>deleted file mode 100644
>index e69de29bb2d..00000000000
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/rtcbot/botmanager.js b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/rtcbot/botmanager.js
>deleted file mode 100644
>index e69de29bb2d..00000000000
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/rtcbot/main.js b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/rtcbot/main.js
>deleted file mode 100644
>index e69de29bb2d..00000000000
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/rtcbot/rtcBotReportVisualizer/index.html b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/rtcbot/rtcBotReportVisualizer/index.html
>deleted file mode 100644
>index e69de29bb2d..00000000000
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/rtcbot/rtcBotReportVisualizer/main.js b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/rtcbot/rtcBotReportVisualizer/main.js
>deleted file mode 100644
>index e69de29bb2d..00000000000
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/rtcbot/test.js b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/rtcbot/test.js
>deleted file mode 100644
>index e69de29bb2d..00000000000
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/rtcbot/test/oneWayVideoStreamingWithDownloadingFile.js b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/rtcbot/test/oneWayVideoStreamingWithDownloadingFile.js
>deleted file mode 100644
>index e69de29bb2d..00000000000
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/rtcbot/test/ping_pong.js b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/rtcbot/test/ping_pong.js
>deleted file mode 100644
>index e69de29bb2d..00000000000
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/rtcbot/test/simple_offer_answer.js b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/rtcbot/test/simple_offer_answer.js
>deleted file mode 100644
>index e69de29bb2d..00000000000
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/rtcbot/test/three_bots_video_conference.js b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/rtcbot/test/three_bots_video_conference.js
>deleted file mode 100644
>index e69de29bb2d..00000000000
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/rtcbot/test/two_way_video_streaming.js b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/rtcbot/test/two_way_video_streaming.js
>deleted file mode 100644
>index e69de29bb2d..00000000000
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/rtcbot/test/webrtc_video_streaming.js b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/rtcbot/test/webrtc_video_streaming.js
>deleted file mode 100644
>index e69de29bb2d..00000000000
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/testing/README.md b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/testing/README.md
>deleted file mode 100644
>index e69de29bb2d..00000000000
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/testing/build_apprtc.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/testing/build_apprtc.py
>deleted file mode 100755
>index e69de29bb2d..00000000000
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/testing/download_apprtc.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/testing/download_apprtc.py
>deleted file mode 100755
>index e69de29bb2d..00000000000
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/testing/golang/linux/go.tar.gz.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/testing/golang/linux/go.tar.gz.sha1
>deleted file mode 100644
>index e69de29bb2d..00000000000
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/testing/golang/mac/go.tar.gz.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/testing/golang/mac/go.tar.gz.sha1
>deleted file mode 100644
>index e69de29bb2d..00000000000
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/testing/golang/win/go.zip.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/testing/golang/win/go.zip.sha1
>deleted file mode 100644
>index e69de29bb2d..00000000000
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/testing/prebuilt_apprtc.zip.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/testing/prebuilt_apprtc.zip.sha1
>deleted file mode 100644
>index e69de29bb2d..00000000000
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/testing/setup_apprtc.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/testing/setup_apprtc.py
>deleted file mode 100755
>index e69de29bb2d..00000000000
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/testing/utils.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/testing/utils.py
>deleted file mode 100755
>index e69de29bb2d..00000000000
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/video_analysis.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/video_analysis.py
>deleted file mode 100755
>index e69de29bb2d..00000000000
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/video_analysis_test.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools/video_analysis_test.py
>deleted file mode 100755
>index e69de29bb2d..00000000000
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/OWNERS
>index 7f84d3bd69c..ae45ba7186a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/OWNERS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/OWNERS
>@@ -1,2 +1,3 @@
>-phoglund@webrtc.org
>+mbonadei@webrtc.org
>+oprypin@webrtc.org
> phoglund@webrtc.org
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/PRESUBMIT.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/PRESUBMIT.py
>index 2f7b835ba96..80e20a348a1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/PRESUBMIT.py
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/PRESUBMIT.py
>@@ -6,6 +6,7 @@
> # in the file PATENTS.  All contributing project authors may
> # be found in the AUTHORS file in the root of the source tree.
> 
>+
> def _LicenseHeader(input_api):
>   """Returns the license header regexp."""
>   # Accept any year number from 2003 to the current year
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/android/AndroidManifest.xml b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/android/AndroidManifest.xml
>deleted file mode 100644
>index bb6d354fef2..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/android/AndroidManifest.xml
>+++ /dev/null
>@@ -1,14 +0,0 @@
>-<?xml version="1.0" encoding="utf-8"?>
>-
>-<manifest xmlns:android="http://schemas.android.com/apk/res/android"
>-    package="dummy.package">
>-
>-      <uses-sdk android:minSdkVersion="16" android:targetSdkVersion="23" />
>-
>-</manifest>
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/android/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/android/OWNERS
>new file mode 100644
>index 00000000000..3c4e54174e6
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/android/OWNERS
>@@ -0,0 +1 @@
>+sakal@webrtc.org
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ios/AppRTCMobile-Prefix.pch b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/android/__init__.py
>similarity index 100%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/ios/AppRTCMobile-Prefix.pch
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/android/__init__.py
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/android/adb_shell.sh b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/android/adb_shell.sh
>old mode 100644
>new mode 100755
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/android/build_aar.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/android/build_aar.py
>old mode 100644
>new mode 100755
>index 0ddca2fd5ec..81e545d11c7
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/android/build_aar.py
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/android/build_aar.py
>@@ -66,7 +66,22 @@ def _ParseArgs():
>   parser.add_argument('--verbose', action='store_true', default=False,
>       help='Debug logging.')
>   parser.add_argument('--extra-gn-args', default=[], nargs='*',
>-      help='Additional GN args to be used during Ninja generation.')
>+      help="""Additional GN arguments to be used during Ninja generation.
>+              These are passed to gn inside `--args` switch and
>+              applied after any other arguments and will
>+              override any values defined by the script.
>+              Example of building debug aar file:
>+              build_aar.py --extra-gn-args='is_debug=true'""")
>+  parser.add_argument('--extra-ninja-switches', default=[], nargs='*',
>+      help="""Additional Ninja switches to be used during compilation.
>+              These are applied after any other Ninja switches.
>+              Example of enabling verbose Ninja output:
>+              build_aar.py --extra-ninja-switches='-v'""")
>+  parser.add_argument('--extra-gn-switches', default=[], nargs='*',
>+      help="""Additional GN switches to be used during compilation.
>+              These are applied after any other GN switches.
>+              Example of enabling verbose GN output:
>+              build_aar.py --extra-gn-switches='-v'""")
>   return parser.parse_args()
> 
> 
>@@ -88,9 +103,9 @@ def _RunNinja(output_directory, args):
> 
> def _EncodeForGN(value):
>   """Encodes value as a GN literal."""
>-  if type(value) is str:
>+  if isinstance(value, str):
>     return '"' + value + '"'
>-  elif type(value) is bool:
>+  elif isinstance(value, bool):
>     return repr(value).lower()
>   else:
>     return repr(value)
>@@ -127,7 +142,8 @@ def _GetArmVersion(arch):
>     raise Exception('Unknown arch: ' + arch)
> 
> 
>-def Build(build_dir, arch, use_goma, extra_gn_args):
>+def Build(build_dir, arch, use_goma, extra_gn_args, extra_gn_switches,
>+          extra_ninja_switches):
>   """Generates target architecture using GN and builds it using ninja."""
>   logging.info('Building: %s', arch)
>   output_directory = _GetOutputDirectory(build_dir, arch)
>@@ -145,11 +161,14 @@ def Build(build_dir, arch, use_goma, extra_gn_args):
>   gn_args_str = '--args=' + ' '.join([
>       k + '=' + _EncodeForGN(v) for k, v in gn_args.items()] + extra_gn_args)
> 
>-  _RunGN(['gen', output_directory, gn_args_str])
>+  gn_args_list = ['gen', output_directory, gn_args_str]
>+  gn_args_list.extend(extra_gn_switches)
>+  _RunGN(gn_args_list)
> 
>   ninja_args = TARGETS[:]
>   if use_goma:
>     ninja_args.extend(['-j', '200'])
>+  ninja_args.extend(extra_ninja_switches)
>   _RunNinja(output_directory, ninja_args)
> 
> 
>@@ -179,12 +198,16 @@ def GenerateLicenses(output_dir, build_dir, archs):
> 
> 
> def BuildAar(archs, output_file, use_goma=False, extra_gn_args=None,
>-             ext_build_dir=None):
>+             ext_build_dir=None, extra_gn_switches=None,
>+             extra_ninja_switches=None):
>   extra_gn_args = extra_gn_args or []
>+  extra_gn_switches = extra_gn_switches or []
>+  extra_ninja_switches = extra_ninja_switches or []
>   build_dir = ext_build_dir if ext_build_dir else tempfile.mkdtemp()
> 
>   for arch in archs:
>-    Build(build_dir, arch, use_goma, extra_gn_args)
>+    Build(build_dir, arch, use_goma, extra_gn_args, extra_gn_switches,
>+          extra_ninja_switches)
> 
>   with zipfile.ZipFile(output_file, 'w') as aar_file:
>     # Architecture doesn't matter here, arbitrarily using the first one.
>@@ -204,7 +227,7 @@ def main():
>   logging.basicConfig(level=logging.DEBUG if args.verbose else logging.INFO)
> 
>   BuildAar(args.arch, args.output, args.use_goma, args.extra_gn_args,
>-           args.build_dir)
>+           args.build_dir, args.extra_gn_switches, args.extra_ninja_switches)
> 
> 
> if __name__ == '__main__':
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/android/profiling/perf_setup.sh b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/android/profiling/perf_setup.sh
>old mode 100644
>new mode 100755
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/android/profiling/utilities.sh b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/android/profiling/utilities.sh
>old mode 100644
>new mode 100755
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/android/release_aar.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/android/release_aar.py
>index c2ae06516a6..bb22b113238 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/android/release_aar.py
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/android/release_aar.py
>@@ -52,6 +52,8 @@ UPLOAD_TRIES = 3
> UPLOAD_RETRY_BASE_SLEEP_SECONDS = 2.0
> GRADLEW_BIN = os.path.join(CHECKOUT_ROOT,
>                            'examples/androidtests/third_party/gradle/gradlew')
>+ADB_BIN = os.path.join(CHECKOUT_ROOT,
>+                       'third_party/android_tools/sdk/platform-tools/adb')
> AAR_PROJECT_DIR = os.path.join(CHECKOUT_ROOT, 'examples/aarproject')
> AAR_PROJECT_GRADLE = os.path.join(AAR_PROJECT_DIR, 'build.gradle')
> AAR_PROJECT_APP_GRADLE = os.path.join(AAR_PROJECT_DIR, 'app', 'build.gradle')
>@@ -67,6 +69,9 @@ def _ParseArgs():
>       help='Skips running the tests.')
>   parser.add_argument('--publish', action='store_true', default=False,
>       help='Automatically publishes the library if the tests pass.')
>+  parser.add_argument('--build-dir', default=None,
>+      help='Temporary directory to store the build files. If not specified, '
>+           'a new directory will be created.')
>   parser.add_argument('--verbose', action='store_true', default=False,
>       help='Debug logging.')
>   return parser.parse_args()
>@@ -168,6 +173,12 @@ def _TestAAR(tmp_dir, username, password, version):
>     with open(AAR_PROJECT_APP_GRADLE, 'w') as gradle_app_file:
>       gradle_app_file.write(gradle_app)
> 
>+    # Uninstall any existing version of AppRTCMobile.
>+    logging.info('Uninstalling previous AppRTCMobile versions. It is okay for '
>+                 'these commands to fail if AppRTCMobile is not installed.')
>+    subprocess.call([ADB_BIN, 'uninstall', 'org.appspot.apprtc'])
>+    subprocess.call([ADB_BIN, 'uninstall', 'org.appspot.apprtc.test'])
>+
>     # Run tests.
>     try:
>       # First clean the project.
>@@ -219,7 +230,7 @@ def _DeleteUnpublishedVersion(user, password, version):
>     raise Exception('Failed to delete version. Response: %s' % response)
> 
> 
>-def ReleaseAar(use_goma, skip_tests, publish):
>+def ReleaseAar(use_goma, skip_tests, publish, build_dir):
>   version = '1.0.' + _GetCommitPos()
>   commit = _GetCommitHash()
>   logging.info('Releasing AAR version %s with hash %s', version, commit)
>@@ -230,18 +241,21 @@ def ReleaseAar(use_goma, skip_tests, publish):
>     raise Exception('Environment variables BINTRAY_USER and BINTRAY_API_KEY '
>                     'must be defined.')
> 
>-  tmp_dir = tempfile.mkdtemp()
>+  # If build directory is not specified, create a temporary directory.
>+  use_tmp_dir = not build_dir
>+  if use_tmp_dir:
>+    build_dir = tempfile.mkdtemp()
> 
>   try:
>     base_name = ARTIFACT_ID + '-' + version
>-    aar_file = os.path.join(tmp_dir, base_name + '.aar')
>-    third_party_licenses_file = os.path.join(tmp_dir, 'LICENSE.md')
>-    pom_file = os.path.join(tmp_dir, base_name + '.pom')
>+    aar_file = os.path.join(build_dir, base_name + '.aar')
>+    third_party_licenses_file = os.path.join(build_dir, 'LICENSE.md')
>+    pom_file = os.path.join(build_dir, base_name + '.pom')
> 
>-    logging.info('Building at %s', tmp_dir)
>+    logging.info('Building at %s', build_dir)
>     BuildAar(ARCHS, aar_file,
>              use_goma=use_goma,
>-             ext_build_dir=os.path.join(tmp_dir, 'aar-build'))
>+             ext_build_dir=os.path.join(build_dir, 'aar-build'))
>     _GeneratePom(pom_file, version, commit)
> 
>     _UploadFile(user, api_key, aar_file, version, base_name + '.aar')
>@@ -249,7 +263,7 @@ def ReleaseAar(use_goma, skip_tests, publish):
>                 'THIRD_PARTY_LICENSES.md')
>     _UploadFile(user, api_key, pom_file, version, base_name + '.pom')
> 
>-    tests_pass = skip_tests or _TestAAR(tmp_dir, user, api_key, version)
>+    tests_pass = skip_tests or _TestAAR(build_dir, user, api_key, version)
>     if not tests_pass:
>       logging.info('Discarding library.')
>       _PublishAAR(user, api_key, version, {'discard': True})
>@@ -263,13 +277,14 @@ def ReleaseAar(use_goma, skip_tests, publish):
>       logging.info('Note: The library has not not been published automatically.'
>                    ' Please do so manually if desired.')
>   finally:
>-    shutil.rmtree(tmp_dir, True)
>+    if use_tmp_dir:
>+      shutil.rmtree(build_dir, True)
> 
> 
> def main():
>   args = _ParseArgs()
>   logging.basicConfig(level=logging.DEBUG if args.verbose else logging.INFO)
>-  ReleaseAar(args.use_goma, args.skip_tests, args.publish)
>+  ReleaseAar(args.use_goma, args.skip_tests, args.publish, args.build_dir)
> 
> 
> if __name__ == '__main__':
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/android/suppressions.xml b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/android/suppressions.xml
>index ff304e650cb..67e5b416530 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/android/suppressions.xml
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/android/suppressions.xml
>@@ -33,4 +33,19 @@
>   <issue id="all">
>     <ignore path="**/org/chromium/**/*.java" />
>   </issue>
>+
>+  <issue id="NewApi">
>+    
>+    <ignore regexp="Call requires API level 19.*`java.util.Objects#requireNonNull`"/>
>+    
>+    <ignore regexp="Class requires API level 19.*java.lang.AutoCloseable"/>
>+    <ignore regexp="Call requires API level 19.*java.lang.AutoCloseable#close"/>
>+    
>+    <ignore regexp="Try-with-resources requires API level 19"/>
>+    <ignore regexp="Call requires API level 19.*`java.lang.Throwable#addSuppressed`"/>
>+    
>+    <ignore regexp="Default method requires API level 24"/>
>+    <ignore regexp="Static interface  method requires API level 24"/>
>+  </issue>
>+
> </lint>
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/android/templates/pom.jinja b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/android/templates/pom.jinja
>index 90ca51ca9c2..0236772dbf7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/android/templates/pom.jinja
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/android/templates/pom.jinja
>@@ -15,4 +15,12 @@
>     <project.build.commitid>{{ commit }}</project.build.commitid>
>   </properties>
> 
>+  <dependencies>
>+    <dependency>
>+      <groupId>com.google.code.findbugs</groupId>
>+      <artifactId>jsr305</artifactId>
>+      <version>3.+</version>
>+    </dependency>
>+  </dependencies>
>+
> </project>
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/autoroller/roll_deps.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/autoroller/roll_deps.py
>old mode 100644
>new mode 100755
>index aaf49fc4da9..5ddbcd156af
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/autoroller/roll_deps.py
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/autoroller/roll_deps.py
>@@ -19,16 +19,17 @@ import subprocess
> import sys
> import urllib2
> 
>-
> # Skip these dependencies (list without solution name prefix).
> DONT_AUTOROLL_THESE = [
>-  'src/third_party/winsdk_samples',
>   'src/examples/androidtests/third_party/gradle',
>+  'src/third_party/ffmpeg',
> ]
> 
> # Run these CQ trybots in addition to the default ones in infra/config/cq.cfg.
> EXTRA_TRYBOTS = (
>-    'master.internal.tryserver.corp.webrtc:linux_internal'
>+  # TODO(bugs.webrtc.org/9615): Temporarily switched to compile_lite, switch
>+  # back to linux_internal when it has been fixed.
>+  'master.internal.tryserver.corp.webrtc:linux_internal_compile_lite'
> )
> 
> WEBRTC_URL = 'https://webrtc.googlesource.com/src'
>@@ -48,6 +49,7 @@ CHECKOUT_ROOT_DIR = os.path.realpath(os.path.join(CHECKOUT_SRC_DIR, os.pardir))
> 
> sys.path.append(os.path.join(CHECKOUT_SRC_DIR, 'build'))
> import find_depot_tools
>+
> find_depot_tools.add_depot_tools_to_path()
> 
> CLANG_UPDATE_SCRIPT_URL_PATH = 'tools/clang/scripts/update.py'
>@@ -55,8 +57,16 @@ CLANG_UPDATE_SCRIPT_LOCAL_PATH = os.path.join(CHECKOUT_SRC_DIR, 'tools',
>                                               'clang', 'scripts', 'update.py')
> 
> DepsEntry = collections.namedtuple('DepsEntry', 'path url revision')
>-ChangedDep = collections.namedtuple('ChangedDep',
>-                                    'path url current_rev new_rev')
>+ChangedDep = collections.namedtuple(
>+    'ChangedDep', 'path url current_rev new_rev')
>+CipdDepsEntry = collections.namedtuple('CipdDepsEntry', 'path packages')
>+ChangedCipdPackage = collections.namedtuple(
>+    'ChangedCipdPackage', 'path package current_version new_version')
>+
>+ChromiumRevisionUpdate = collections.namedtuple('ChromiumRevisionUpdate',
>+                                                ('current_chromium_rev '
>+                                                 'new_chromium_rev '))
>+
> 
> class RollError(Exception):
>   pass
>@@ -72,7 +82,7 @@ def ParseDepsDict(deps_content):
>     'Var': VarLookup(local_scope),
>     'deps_os': {},
>   }
>-  exec(deps_content, global_scope, local_scope)
>+  exec (deps_content, global_scope, local_scope)
>   return local_scope
> 
> 
>@@ -91,14 +101,14 @@ def ParseCommitPosition(commit_message):
>   for line in reversed(commit_message.splitlines()):
>     m = COMMIT_POSITION_RE.match(line.strip())
>     if m:
>-      return m.group(1)
>+      return int(m.group(1))
>   logging.error('Failed to parse commit position id from:\n%s\n',
>                 commit_message)
>   sys.exit(-1)
> 
> 
> def _RunCommand(command, working_dir=None, ignore_exit_code=False,
>-                extra_env=None):
>+    extra_env=None, input_data=None):
>   """Runs a command and returns the output from that command.
> 
>   If the command fails (exit code != 0), the function will exit the process.
>@@ -110,15 +120,15 @@ def _RunCommand(command, working_dir=None, ignore_exit_code=False,
>   logging.debug('CMD: %s CWD: %s', ' '.join(command), working_dir)
>   env = os.environ.copy()
>   if extra_env:
>-    assert all(type(value) == str for value in extra_env.values())
>+    assert all(isinstance(value, str) for value in extra_env.values())
>     logging.debug('extra env: %s', extra_env)
>     env.update(extra_env)
>-  p = subprocess.Popen(command, stdout=subprocess.PIPE,
>+  p = subprocess.Popen(command,
>+                       stdin=subprocess.PIPE,
>+                       stdout=subprocess.PIPE,
>                        stderr=subprocess.PIPE, env=env,
>                        cwd=working_dir, universal_newlines=True)
>-  std_output = p.stdout.read()
>-  err_output = p.stderr.read()
>-  p.wait()
>+  std_output, err_output = p.communicate(input_data)
>   p.stdout.close()
>   p.stderr.close()
>   if not ignore_exit_code and p.returncode != 0:
>@@ -207,12 +217,17 @@ def GetMatchingDepsEntries(depsentry_dict, dir_path):
> def BuildDepsentryDict(deps_dict):
>   """Builds a dict of paths to DepsEntry objects from a raw parsed deps dict."""
>   result = {}
>+
>   def AddDepsEntries(deps_subdict):
>-    for path, deps_url in deps_subdict.iteritems():
>-      if isinstance(deps_url, dict):
>-        deps_url = deps_url['url']
>-      if not result.has_key(path):
>-        url, revision = deps_url.split('@') if deps_url else (None, None)
>+    for path, dep in deps_subdict.iteritems():
>+      if path in result:
>+        continue
>+      if not isinstance(dep, dict):
>+        dep = {'url': dep}
>+      if dep.get('dep_type') == 'cipd':
>+        result[path] = CipdDepsEntry(path, dep['packages'])
>+      else:
>+        url, revision = dep['url'].split('@')
>         result[path] = DepsEntry(path, url, revision)
> 
>   AddDepsEntries(deps_dict['deps'])
>@@ -221,6 +236,20 @@ def BuildDepsentryDict(deps_dict):
>   return result
> 
> 
>+def _FindChangedCipdPackages(path, old_pkgs, new_pkgs):
>+  assert ({p['package'] for p in old_pkgs} ==
>+          {p['package'] for p in new_pkgs})
>+  for old_pkg in old_pkgs:
>+    for new_pkg in new_pkgs:
>+      old_version = old_pkg['version']
>+      new_version = new_pkg['version']
>+      if (old_pkg['package'] == new_pkg['package'] and
>+          old_version != new_version):
>+        logging.debug('Roll dependency %s to %s', path, new_version)
>+        yield ChangedCipdPackage(path, old_pkg['package'],
>+                                 old_version, new_version)
>+
>+
> def CalculateChangedDeps(webrtc_deps, new_cr_deps):
>   """
>   Calculate changed deps entries based on entries defined in the WebRTC DEPS
>@@ -244,16 +273,27 @@ def CalculateChangedDeps(webrtc_deps, new_cr_deps):
>       continue
>     cr_deps_entry = new_cr_entries.get(path)
>     if cr_deps_entry:
>+      assert type(cr_deps_entry) is type(webrtc_deps_entry)
>+
>+      if isinstance(cr_deps_entry, CipdDepsEntry):
>+        result.extend(_FindChangedCipdPackages(path, webrtc_deps_entry.packages,
>+                                               cr_deps_entry.packages))
>+        continue
>+
>       # Use the revision from Chromium's DEPS file.
>       new_rev = cr_deps_entry.revision
>       assert webrtc_deps_entry.url == cr_deps_entry.url, (
>-        'WebRTC DEPS entry %s has a different URL (%s) than Chromium (%s).' %
>-        (path, webrtc_deps_entry.url, cr_deps_entry.url))
>+          'WebRTC DEPS entry %s has a different URL (%s) than Chromium (%s).' %
>+          (path, webrtc_deps_entry.url, cr_deps_entry.url))
>     else:
>-      # Use the HEAD of the deps repo.
>-      stdout, _ = _RunCommand(['git', 'ls-remote', webrtc_deps_entry.url,
>-                               'HEAD'])
>-      new_rev = stdout.strip().split('\t')[0]
>+      if isinstance(webrtc_deps_entry, DepsEntry):
>+        # Use the HEAD of the deps repo.
>+        stdout, _ = _RunCommand(['git', 'ls-remote', webrtc_deps_entry.url,
>+                                'HEAD'])
>+        new_rev = stdout.strip().split('\t')[0]
>+      else:
>+        raise RollError('WebRTC DEPS entry %s is missing from Chromium. Remove '
>+                        'it or add it to DONT_AUTOROLL_THESE.' % path)
> 
>     # Check if an update is necessary.
>     if webrtc_deps_entry.revision != new_rev:
>@@ -276,31 +316,35 @@ def CalculateChangedClang(new_cr_rev):
>   current_rev = GetClangRev(current_lines)
> 
>   new_clang_update_py = ReadRemoteCrFile(CLANG_UPDATE_SCRIPT_URL_PATH,
>-                                             new_cr_rev).splitlines()
>+                                         new_cr_rev).splitlines()
>   new_rev = GetClangRev(new_clang_update_py)
>   return ChangedDep(CLANG_UPDATE_SCRIPT_LOCAL_PATH, None, current_rev, new_rev)
> 
> 
>-def GenerateCommitMessage(current_cr_rev, new_cr_rev, current_commit_pos,
>-                          new_commit_pos, changed_deps_list, clang_change):
>-  current_cr_rev = current_cr_rev[0:10]
>-  new_cr_rev = new_cr_rev[0:10]
>+def GenerateCommitMessage(rev_update, current_commit_pos,
>+    new_commit_pos, changed_deps_list, clang_change):
>+  current_cr_rev = rev_update.current_chromium_rev[0:10]
>+  new_cr_rev = rev_update.new_chromium_rev[0:10]
>   rev_interval = '%s..%s' % (current_cr_rev, new_cr_rev)
>   git_number_interval = '%s:%s' % (current_commit_pos, new_commit_pos)
> 
>   commit_msg = ['Roll chromium_revision %s (%s)\n' % (rev_interval,
>-                                                    git_number_interval)]
>-  commit_msg.append('Change log: %s' % (CHROMIUM_LOG_TEMPLATE % rev_interval))
>-  commit_msg.append('Full diff: %s\n' % (CHROMIUM_COMMIT_TEMPLATE %
>-                                         rev_interval))
>+                                                      git_number_interval),
>+                'Change log: %s' % (CHROMIUM_LOG_TEMPLATE % rev_interval),
>+                'Full diff: %s\n' % (CHROMIUM_COMMIT_TEMPLATE %
>+                                     rev_interval)]
>   tbr_authors = ''
>   if changed_deps_list:
>     commit_msg.append('Changed dependencies:')
> 
>     for c in changed_deps_list:
>-      commit_msg.append('* %s: %s/+log/%s..%s' % (c.path, c.url,
>-                                                  c.current_rev[0:10],
>-                                                  c.new_rev[0:10]))
>+      if isinstance(c, ChangedCipdPackage):
>+        commit_msg.append('* %s: %s..%s' % (c.path, c.current_version,
>+                                            c.new_version))
>+      else:
>+        commit_msg.append('* %s: %s/+log/%s..%s' % (c.path, c.url,
>+                                                    c.current_rev[0:10],
>+                                                    c.new_rev[0:10]))
>       if 'libvpx' in c.path:
>         tbr_authors += 'marpan@webrtc.org, '
> 
>@@ -329,14 +373,14 @@ def GenerateCommitMessage(current_cr_rev, new_cr_rev, current_commit_pos,
>   return '\n'.join(commit_msg)
> 
> 
>-def UpdateDepsFile(deps_filename, old_cr_revision, new_cr_revision,
>-                   changed_deps):
>+def UpdateDepsFile(deps_filename, rev_update, changed_deps):
>   """Update the DEPS file with the new revision."""
> 
>   # Update the chromium_revision variable.
>   with open(deps_filename, 'rb') as deps_file:
>     deps_content = deps_file.read()
>-  deps_content = deps_content.replace(old_cr_revision, new_cr_revision)
>+  deps_content = deps_content.replace(rev_update.current_chromium_rev,
>+                                      rev_update.new_chromium_rev)
>   with open(deps_filename, 'wb') as deps_file:
>     deps_file.write(deps_content)
> 
>@@ -351,11 +395,13 @@ def UpdateDepsFile(deps_filename, old_cr_revision, new_cr_revision,
>           'platforms in the target_os list, i.e.\n'
>           'target_os = ["android", "unix", "mac", "ios", "win"];\n'
>           'Then run "gclient sync" again.' % local_dep_dir)
>-    _, stderr = _RunCommand(
>-      ['roll-dep-svn', '--no-verify-revision', dep.path, dep.new_rev],
>-      working_dir=CHECKOUT_SRC_DIR, ignore_exit_code=True)
>-    if stderr:
>-      logging.warning('roll-dep-svn: %s', stderr)
>+    if isinstance(dep, ChangedCipdPackage):
>+      package = dep.package.format()  # Eliminate double curly brackets
>+      update = '%s:%s@%s' % (dep.path, package, dep.new_version)
>+    else:
>+      update = '%s@%s' % (dep.path, dep.new_rev)
>+    _RunCommand(['gclient', 'setdep', '--revision', update],
>+                working_dir=CHECKOUT_SRC_DIR)
> 
> 
> def _IsTreeClean():
>@@ -403,14 +449,42 @@ def _LocalCommit(commit_msg, dry_run):
>     _RunCommand(['git', 'commit', '-m', commit_msg])
> 
> 
>-def _UploadCL(dry_run, skip_cq=False):
>-  logging.info('Uploading CL...')
>-  if not dry_run:
>-    cmd = ['git', 'cl', 'upload', '-f', '--gerrit']
>-    if not skip_cq:
>-      logging.info('Sending the CL to the CQ...')
>-      cmd.extend(['--use-commit-queue', '--send-mail'])
>-    _RunCommand(cmd, extra_env={'EDITOR': 'true', 'SKIP_GCE_AUTH_FOR_GIT': '1'})
>+def ChooseCQMode(skip_cq, cq_over, current_commit_pos, new_commit_pos):
>+  if skip_cq:
>+    return 0
>+  if (new_commit_pos - current_commit_pos) < cq_over:
>+    return 1
>+  return 2
>+
>+
>+def _UploadCL(commit_queue_mode):
>+  """Upload the committed changes as a changelist to Gerrit.
>+
>+  commit_queue_mode:
>+    - 2: Submit to commit queue.
>+    - 1: Run trybots but do not submit to CQ.
>+    - 0: Skip CQ, upload only.
>+  """
>+  cmd = ['git', 'cl', 'upload', '-f', '--gerrit']
>+  if commit_queue_mode >= 2:
>+    logging.info('Sending the CL to the CQ...')
>+    cmd.extend(['--use-commit-queue', '--send-mail'])
>+  elif commit_queue_mode >= 1:
>+    logging.info('Starting CQ dry run...')
>+    cmd.extend(['--cq-dry-run'])
>+  _RunCommand(cmd, extra_env={'EDITOR': 'true', 'SKIP_GCE_AUTH_FOR_GIT': '1'})
>+
>+
>+def GetRollRevisionRanges(opts, webrtc_deps):
>+  current_cr_rev = webrtc_deps['vars']['chromium_revision']
>+  new_cr_rev = opts.revision
>+  if not new_cr_rev:
>+    stdout, _ = _RunCommand(['git', 'ls-remote', CHROMIUM_SRC_URL, 'HEAD'])
>+    head_rev = stdout.strip().split('\t')[0]
>+    logging.info('No revision specified. Using HEAD: %s', head_rev)
>+    new_cr_rev = head_rev
>+
>+  return ChromiumRevisionUpdate(current_cr_rev, new_cr_rev)
> 
> 
> def main():
>@@ -432,8 +506,12 @@ def main():
>                  default=False,
>                  help=('Ignore if the current branch is not master or if there '
>                        'are uncommitted changes (default: %(default)s).'))
>-  p.add_argument('--skip-cq', action='store_true', default=False,
>-                 help='Skip sending the CL to the CQ (default: %(default)s)')
>+  grp = p.add_mutually_exclusive_group()
>+  grp.add_argument('--skip-cq', action='store_true', default=False,
>+                   help='Skip sending the CL to the CQ (default: %(default)s)')
>+  grp.add_argument('--cq-over', type=int, default=1,
>+                   help=('Commit queue dry run if the revision difference '
>+                         'is below this number (default: %(default)s)'))
>   p.add_argument('-v', '--verbose', action='store_true', default=False,
>                  help='Be extra verbose in printing of log messages.')
>   opts = p.parse_args()
>@@ -453,35 +531,34 @@ def main():
>   if not opts.ignore_unclean_workdir:
>     _EnsureUpdatedMasterBranch(opts.dry_run)
> 
>-  new_cr_rev = opts.revision
>-  if not new_cr_rev:
>-    stdout, _ = _RunCommand(['git', 'ls-remote', CHROMIUM_SRC_URL, 'HEAD'])
>-    head_rev = stdout.strip().split('\t')[0]
>-    logging.info('No revision specified. Using HEAD: %s', head_rev)
>-    new_cr_rev = head_rev
>-
>   deps_filename = os.path.join(CHECKOUT_SRC_DIR, 'DEPS')
>   webrtc_deps = ParseLocalDepsFile(deps_filename)
>-  current_cr_rev = webrtc_deps['vars']['chromium_revision']
>+  rev_update = GetRollRevisionRanges(opts, webrtc_deps)
> 
>-  current_commit_pos = ParseCommitPosition(ReadRemoteCrCommit(current_cr_rev))
>-  new_commit_pos = ParseCommitPosition(ReadRemoteCrCommit(new_cr_rev))
>+  current_commit_pos = ParseCommitPosition(
>+      ReadRemoteCrCommit(rev_update.current_chromium_rev))
>+  new_commit_pos = ParseCommitPosition(
>+      ReadRemoteCrCommit(rev_update.new_chromium_rev))
> 
>-  new_cr_deps = ParseRemoteCrDepsFile(new_cr_rev)
>+  new_cr_deps = ParseRemoteCrDepsFile(rev_update.new_chromium_rev)
>   changed_deps = CalculateChangedDeps(webrtc_deps, new_cr_deps)
>-  clang_change = CalculateChangedClang(new_cr_rev)
>-  commit_msg = GenerateCommitMessage(current_cr_rev, new_cr_rev,
>+  clang_change = CalculateChangedClang(rev_update.new_chromium_rev)
>+  commit_msg = GenerateCommitMessage(rev_update,
>                                      current_commit_pos, new_commit_pos,
>                                      changed_deps, clang_change)
>   logging.debug('Commit message:\n%s', commit_msg)
> 
>   _CreateRollBranch(opts.dry_run)
>-  UpdateDepsFile(deps_filename, current_cr_rev, new_cr_rev, changed_deps)
>+  UpdateDepsFile(deps_filename, rev_update, changed_deps)
>   if _IsTreeClean():
>     logging.info("No DEPS changes detected, skipping CL creation.")
>   else:
>     _LocalCommit(commit_msg, opts.dry_run)
>-    _UploadCL(opts.dry_run, opts.skip_cq)
>+    commit_queue_mode = ChooseCQMode(opts.skip_cq, opts.cq_over,
>+                                     current_commit_pos, new_commit_pos)
>+    logging.info('Uploading CL...')
>+    if not opts.dry_run:
>+      _UploadCL(commit_queue_mode)
>   return 0
> 
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/autoroller/unittests/roll_deps_test.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/autoroller/unittests/roll_deps_test.py
>old mode 100644
>new mode 100755
>index 5c48a86c161..2c19198aaef
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/autoroller/unittests/roll_deps_test.py
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/autoroller/unittests/roll_deps_test.py
>@@ -19,8 +19,9 @@ SCRIPT_DIR = os.path.dirname(os.path.abspath(__file__))
> PARENT_DIR = os.path.join(SCRIPT_DIR, os.pardir)
> sys.path.append(PARENT_DIR)
> import roll_deps
>-from roll_deps import CalculateChangedDeps, GetMatchingDepsEntries, \
>-  ParseDepsDict, ParseLocalDepsFile, UpdateDepsFile
>+from roll_deps import CalculateChangedDeps, ChooseCQMode, \
>+  GetMatchingDepsEntries, ParseDepsDict, ParseLocalDepsFile, UpdateDepsFile, \
>+  ChromiumRevisionUpdate
> 
> 
> TEST_DATA_VARS = {
>@@ -67,10 +68,10 @@ class FakeCmd(object):
> class TestRollChromiumRevision(unittest.TestCase):
>   def setUp(self):
>     self._output_dir = tempfile.mkdtemp()
>-    for test_file in glob.glob(os.path.join(SCRIPT_DIR, 'testdata', '*')):
>+    test_data_dir = os.path.join(SCRIPT_DIR, 'testdata', 'roll_deps')
>+    for test_file in glob.glob(os.path.join(test_data_dir, '*')):
>       shutil.copy(test_file, self._output_dir)
>     self._webrtc_depsfile = os.path.join(self._output_dir, 'DEPS')
>-    self._old_cr_depsfile = os.path.join(self._output_dir, 'DEPS.chromium.old')
>     self._new_cr_depsfile = os.path.join(self._output_dir, 'DEPS.chromium.new')
> 
>     self.fake = FakeCmd()
>@@ -91,7 +92,9 @@ class TestRollChromiumRevision(unittest.TestCase):
>     new_rev = 'aaaaabbbbbcccccdddddeeeeefffff0000011111'
> 
>     current_rev = TEST_DATA_VARS['chromium_revision']
>-    UpdateDepsFile(self._webrtc_depsfile, current_rev, new_rev, [])
>+    UpdateDepsFile(self._webrtc_depsfile,
>+                   ChromiumRevisionUpdate(current_rev, new_rev),
>+                   [])
>     with open(self._webrtc_depsfile) as deps_file:
>       deps_contents = deps_file.read()
>       self.assertTrue(new_rev in deps_contents,
>@@ -107,7 +110,7 @@ class TestRollChromiumRevision(unittest.TestCase):
>       self.assertEquals(vars_dict[variable_name], TEST_DATA_VARS[variable_name])
>     AssertVar('chromium_git')
>     AssertVar('chromium_revision')
>-    self.assertEquals(len(local_scope['deps']), 2)
>+    self.assertEquals(len(local_scope['deps']), 3)
>     self.assertEquals(len(local_scope['deps_os']), 1)
> 
>   def testGetMatchingDepsEntriesReturnsPathInSimpleCase(self):
>@@ -130,7 +133,7 @@ class TestRollChromiumRevision(unittest.TestCase):
>     webrtc_deps = ParseLocalDepsFile(self._webrtc_depsfile)
>     new_cr_deps = ParseLocalDepsFile(self._new_cr_depsfile)
>     changed_deps = CalculateChangedDeps(webrtc_deps, new_cr_deps)
>-    self.assertEquals(len(changed_deps), 2)
>+    self.assertEquals(len(changed_deps), 3)
>     self.assertEquals(changed_deps[0].path, 'src/build')
>     self.assertEquals(changed_deps[0].current_rev, BUILD_OLD_REV)
>     self.assertEquals(changed_deps[0].new_rev, BUILD_NEW_REV)
>@@ -139,6 +142,22 @@ class TestRollChromiumRevision(unittest.TestCase):
>     self.assertEquals(changed_deps[1].current_rev, BUILDTOOLS_OLD_REV)
>     self.assertEquals(changed_deps[1].new_rev, BUILDTOOLS_NEW_REV)
> 
>+    self.assertEquals(changed_deps[2].path, 'src/third_party/xstream')
>+    self.assertEquals(changed_deps[2].package, 'chromium/third_party/xstream')
>+    self.assertEquals(changed_deps[2].current_version, 'version:1.4.8-cr0')
>+    self.assertEquals(changed_deps[2].new_version, 'version:1.10.0-cr0')
>+
>+
>+class TestChooseCQMode(unittest.TestCase):
>+  def testSkip(self):
>+    self.assertEquals(ChooseCQMode(True, 99, 500000, 500100), 0)
>+
>+  def testDryRun(self):
>+    self.assertEquals(ChooseCQMode(False, 101, 500000, 500100), 1)
>+
>+  def testSubmit(self):
>+    self.assertEquals(ChooseCQMode(False, 100, 500000, 500100), 2)
>+
> 
> def _SetupGitLsRemoteCall(cmd_fake, url, revision):
>   cmd = ['git', 'ls-remote', url, revision]
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/autoroller/unittests/testdata/DEPS.chromium.new b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/autoroller/unittests/testdata/DEPS.chromium.new
>deleted file mode 100644
>index d53083ceec5..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/autoroller/unittests/testdata/DEPS.chromium.new
>+++ /dev/null
>@@ -1,13 +0,0 @@
>-# DEPS file for unit tests.
>-
>-vars = {
>-  'chromium_git': 'https://chromium.googlesource.com',
>-
>-  # This is updated compared to the DEPS.chromium.old file.
>-  'buildtools_revision': '55ad626b08ef971fd82a62b7abb325359542952b',
>-}
>-
>-deps = {
>-  'src/buildtools':
>-    Var('chromium_git') + '/chromium/buildtools.git' + '@' +  Var('buildtools_revision'),
>-}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/autoroller/unittests/testdata/DEPS.chromium.old b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/autoroller/unittests/testdata/DEPS.chromium.old
>deleted file mode 100644
>index dd6ddaec813..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/autoroller/unittests/testdata/DEPS.chromium.old
>+++ /dev/null
>@@ -1,13 +0,0 @@
>-# DEPS file for unit tests.
>-
>-vars = {
>-  'chromium_git': 'https://chromium.googlesource.com',
>-
>-  # This is and older revision than DEPS.chromium.new file.
>-  'buildtools_revision': '64e38f0cebdde27aa0cfb405f330063582f9ac76',
>-}
>-
>-deps = {
>-  'src/buildtools':
>-    Var('chromium_git') + '/chromium/buildtools.git' + '@' +  Var('buildtools_revision'),
>-}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/autoroller/unittests/testdata/DEPS b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/autoroller/unittests/testdata/roll_deps/DEPS
>similarity index 55%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/autoroller/unittests/testdata/DEPS
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/autoroller/unittests/testdata/roll_deps/DEPS
>index ff30f0b481a..fcd65a5c9f4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/autoroller/unittests/testdata/DEPS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/autoroller/unittests/testdata/roll_deps/DEPS
>@@ -13,12 +13,25 @@ deps = {
>   # Entry that's also a DEPS entry in the Chromium DEPS file.
>   'src/buildtools':
>     Var('chromium_git') + '/chromium/buildtools.git' + '@' + '64e38f0cebdde27aa0cfb405f330063582f9ac76',
>+
>+  # Entry that's also a CIPD entry in the Chromium DEPS file.
>+  'src/third_party/xstream': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/xstream',
>+              'version': 'version:1.4.8-cr0',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
> }
> 
> deps_os = {
>   # Entry only present in WebRTC, not Chromium.
>-  'win': {
>-    'src/third_party/winsdk_samples':
>-      Var('chromium_git') + '/external/webrtc/deps/third_party/winsdk_samples_v71@e71b549167a665d7424d6f1dadfbff4b4aad1589',
>+  'android': {
>+    'src/examples/androidtests/third_party/gradle':
>+      Var('chromium_git') + '/external/github.com/gradle/gradle.git' + '@' +
>+        '89af43c4d0506f69980f00dde78c97b2f81437f8',
>   },
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/autoroller/unittests/testdata/roll_deps/DEPS.chromium.new b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/autoroller/unittests/testdata/roll_deps/DEPS.chromium.new
>new file mode 100644
>index 00000000000..41526c7b3f8
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/autoroller/unittests/testdata/roll_deps/DEPS.chromium.new
>@@ -0,0 +1,25 @@
>+# DEPS file for unit tests.
>+
>+vars = {
>+  'chromium_git': 'https://chromium.googlesource.com',
>+
>+  # This is updated compared to the DEPS file.
>+  'buildtools_revision': '55ad626b08ef971fd82a62b7abb325359542952b',
>+}
>+
>+deps = {
>+  'src/buildtools':
>+    Var('chromium_git') + '/chromium/buildtools.git' + '@' +  Var('buildtools_revision'),
>+
>+  'src/third_party/xstream': {
>+      'packages': [
>+          {
>+              'package': 'chromium/third_party/xstream',
>+              # This is updated compared to the DEPS file.
>+              'version': 'version:1.10.0-cr0',
>+          },
>+      ],
>+      'condition': 'checkout_android',
>+      'dep_type': 'cipd',
>+  },
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/coverage/generate_coverage_command.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/coverage/generate_coverage_command.py
>new file mode 100644
>index 00000000000..c6b38d067ff
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/coverage/generate_coverage_command.py
>@@ -0,0 +1,55 @@
>+#!/usr/bin/env python
>+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+#
>+# Use of this source code is governed by a BSD-style license
>+# that can be found in the LICENSE file in the root of the source
>+# tree. An additional intellectual property rights grant can be found
>+# in the file PATENTS.  All contributing project authors may
>+# be found in the AUTHORS file in the root of the source tree.
>+
>+"""Generates a command-line for coverage.py. Useful for manual coverage runs.
>+
>+Before running the generated command line, do this:
>+
>+gn gen out/coverage --args='use_clang_coverage=true is_component_build=false'
>+"""
>+
>+import sys
>+
>+TESTS = [
>+  'video_capture_tests',
>+  'webrtc_nonparallel_tests',
>+  'video_engine_tests',
>+  'tools_unittests',
>+  'test_support_unittests',
>+  'system_wrappers_unittests',
>+  'rtc_unittests',
>+  'rtc_stats_unittests',
>+  'rtc_pc_unittests',
>+  'rtc_media_unittests',
>+  'peerconnection_unittests',
>+  'ortc_unittests',
>+  'modules_unittests',
>+  'modules_tests',
>+  'low_bandwidth_audio_test',
>+  'common_video_unittests',
>+  'common_audio_unittests',
>+  'audio_decoder_unittests'
>+]
>+
>+def main():
>+  cmd = ([sys.executable, 'tools/code_coverage/coverage.py'] + TESTS +
>+         ['-b out/coverage', '-o out/report'] +
>+         ['-i=\'.*/out/.*|.*/third_party/.*\''] +
>+         ['-c \'out/coverage/%s\'' % t for t in TESTS])
>+
>+  def WithXvfb(binary):
>+    return '-c \'%s testing/xvfb.py %s\'' % (sys.executable, binary)
>+  modules_unittests = 'out/coverage/modules_unittests'
>+  cmd[cmd.index('-c \'%s\'' % modules_unittests)] = WithXvfb(modules_unittests)
>+
>+  print ' '.join(cmd)
>+  return 0
>+
>+if __name__ == '__main__':
>+  sys.exit(main())
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/cpu/cpu_mon.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/cpu/cpu_mon.py
>index 9a13723ed2d..128a5c0bf6f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/cpu/cpu_mon.py
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/cpu/cpu_mon.py
>@@ -61,7 +61,7 @@ def main():
>   snapshots = []
>   while True:
>     snapshot = GrabCpuSamples(sample_count)
>-    if snapshot == None:
>+    if snapshot is None:
>       break
>     snapshots.append(snapshot)
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/download_tools.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/download_tools.py
>old mode 100644
>new mode 100755
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ensure_webcam_is_running.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ensure_webcam_is_running.py
>new file mode 100755
>index 00000000000..10153a14513
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ensure_webcam_is_running.py
>@@ -0,0 +1,124 @@
>+#!/usr/bin/env python
>+# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
>+#
>+# Use of this source code is governed by a BSD-style license
>+# that can be found in the LICENSE file in the root of the source
>+# tree. An additional intellectual property rights grant can be found
>+# in the file PATENTS.  All contributing project authors may
>+# be found in the AUTHORS file in the root of the source tree.
>+
>+"""Checks if a virtual webcam is running and starts it if not.
>+
>+Returns a non-zero return code if the webcam could not be started.
>+
>+Prerequisites:
>+* The Python interpreter must have the psutil package installed.
>+* Windows: a scheduled task named 'ManyCam' must exist and be configured to
>+  launch ManyCam preconfigured to auto-play the test clip.
>+* Mac: ManyCam must be installed in the default location and be preconfigured
>+  to auto-play the test clip.
>+* Linux: The v4l2loopback kernel module must be compiled and loaded to the
>+  kernel already and the v4l2_file_player application must be compiled and put
>+  in the location specified below.
>+
>+NOTICE: When running this script as a buildbot step, make sure to set
>+usePTY=False for the build step when adding it, or the subprocess will die as
>+soon the step has executed.
>+
>+If any command line arguments are passed to the script, it is executed as a
>+command in a subprocess.
>+"""
>+
>+import os
>+# psutil is not installed on non-Linux machines by default.
>+import psutil  # pylint: disable=F0401
>+import subprocess
>+import sys
>+import time
>+
>+
>+WEBCAM_WIN = ('schtasks', '/run', '/tn', 'ManyCam')
>+WEBCAM_MAC = ('open', '/Applications/ManyCam/ManyCam.app')
>+E = os.path.expandvars
>+WEBCAM_LINUX = (
>+    E('$HOME/fake-webcam-driver/linux/v4l2_file_player/v4l2_file_player'),
>+    E('$HOME/webrtc_video_quality/reference_video.yuv'),
>+    '640', '480', '/dev/video0',
>+)
>+
>+
>+def IsWebCamRunning():
>+  if sys.platform == 'win32':
>+    process_name = 'ManyCam.exe'
>+  elif sys.platform.startswith('darwin'):
>+    process_name = 'ManyCam'
>+  elif sys.platform.startswith('linux'):
>+    process_name = 'v4l2_file_player'
>+  else:
>+    raise Exception('Unsupported platform: %s' % sys.platform)
>+  for p in psutil.process_iter():
>+    try:
>+      if process_name == p.name:
>+        print 'Found a running virtual webcam (%s with PID %s)' % (p.name,
>+                                                                   p.pid)
>+        return True
>+    except psutil.AccessDenied:
>+      pass  # This is normal if we query sys processes, etc.
>+  return False
>+
>+
>+def StartWebCam():
>+  try:
>+    if sys.platform == 'win32':
>+      subprocess.check_call(WEBCAM_WIN)
>+      print 'Successfully launched virtual webcam.'
>+    elif sys.platform.startswith('darwin'):
>+      subprocess.check_call(WEBCAM_MAC)
>+      print 'Successfully launched virtual webcam.'
>+    elif sys.platform.startswith('linux'):
>+
>+      # Must redirect stdout/stderr/stdin to avoid having the subprocess
>+      # being killed when the parent shell dies (happens on the bots).
>+      process = subprocess.Popen(WEBCAM_LINUX, stdout=subprocess.PIPE,
>+                                 stderr=subprocess.PIPE,
>+                                 stdin=subprocess.PIPE)
>+      # If the v4l2loopback module is not loaded or incorrectly configured,
>+      # the process will still launch but will die immediately.
>+      # Wait for a second and then check for aliveness to catch such errors.
>+      time.sleep(1)
>+      if process.poll() is None:
>+        print 'Successfully launched virtual webcam with PID %s' % process.pid
>+      else:
>+        print 'Failed to launch virtual webcam.'
>+        return False
>+
>+  except Exception as e:
>+    print 'Failed to launch virtual webcam: %s' % e
>+    return False
>+
>+  return True
>+
>+
>+def _ForcePythonInterpreter(cmd):
>+  """Returns the fixed command line to call the right python executable."""
>+  out = cmd[:]
>+  if out[0] == 'python':
>+    out[0] = sys.executable
>+  elif out[0].endswith('.py'):
>+    out.insert(0, sys.executable)
>+  return out
>+
>+
>+def Main(argv):
>+  if not IsWebCamRunning():
>+    if not StartWebCam():
>+      return 1
>+
>+  if argv:
>+    return subprocess.call(_ForcePythonInterpreter(argv))
>+  else:
>+    return 0
>+
>+
>+if __name__ == '__main__':
>+  sys.exit(Main(sys.argv[1:]))
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/get_landmines.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/get_landmines.py
>old mode 100644
>new mode 100755
>index a452835458d..91980844f41
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/get_landmines.py
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/get_landmines.py
>@@ -21,7 +21,7 @@ sys.path.insert(0, os.path.join(CHECKOUT_ROOT, 'build'))
> import landmine_utils
> 
> 
>-platform = landmine_utils.platform  # pylint: disable=invalid-name
>+host_os = landmine_utils.host_os  # pylint: disable=invalid-name
> 
> 
> def print_landmines():  # pylint: disable=invalid-name
>@@ -35,14 +35,7 @@ def print_landmines():  # pylint: disable=invalid-name
>   # landmine.
>   # See the Chromium version in src/build/get_landmines.py for usage examples.
>   print 'Clobber to remove out/{Debug,Release}/args.gn (webrtc:5070)'
>-  if platform() == 'android':
>-    print ('Clobber to remove artifacts on Android causing lint errors after '
>-           'rolling in https://codereview.webrtc.org/2293863002')
>-    print ('Clobber to remove old AppRTCDemo artifacts after renaming to '
>-           'AppRTCMobile in https://codereview.webrtc.org/2373443005')
>-    print ('Clobber to fix Android x86/x64 builds after '
>-           'https://codereview.webrtc.org/1414343008/')
>-  if platform() == 'win':
>+  if host_os() == 'win':
>     print 'Clobber to resolve some issues with corrupt .pdb files on bots.'
>     print 'Clobber due to corrupt .pdb files (after #14623)'
>     print 'Clobber due to Win 64-bit Debug linking error (crbug.com/668961)'
>@@ -50,9 +43,7 @@ def print_landmines():  # pylint: disable=invalid-name
>            'https://codereview.webrtc.org/2786603002')
>     print ('Clobber due to Win Debug linking errors in '
>            'https://codereview.webrtc.org/2832063003/')
>-  if platform() == 'mac':
>-    # platform == 'ios' doesn't work since it assumes GYP_DEFINES is set, which
>-    # is no longer the case.
>+  if host_os() == 'mac':
>     print 'Clobber due to iOS compile errors (crbug.com/694721)'
>     print 'Clobber to unblock https://codereview.webrtc.org/2709573003'
>     print ('Clobber to fix https://codereview.webrtc.org/2709573003 after '
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/gtest-parallel-wrapper.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/gtest-parallel-wrapper.py
>old mode 100644
>new mode 100755
>index 42e523343ed..875e0885efe
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/gtest-parallel-wrapper.py
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/gtest-parallel-wrapper.py
>@@ -18,12 +18,9 @@ In particular, this translates the GTEST_SHARD_INDEX and GTEST_TOTAL_SHARDS
> environment variables to the --shard_index and --shard_count flags, and renames
> the --isolated-script-test-output flag to --dump_json_test_results.
> 
>-All flags before '--' will be passed as arguments to gtest-parallel, and
>-(almost) all flags after '--' will be passed as arguments to the test
>-executable.
>-The exception is that --isolated-script-test-output and
>---isolated-script-test-chartson-output are expected to be after '--', so they
>-are processed and removed from there.
>+Flags before '--' will be attempted to be understood as arguments to
>+gtest-parallel. If gtest-parallel doesn't recognize the flag or the flag is
>+after '--', the flag will be passed on to the test executable.
> 
> If the --store-test-artifacts flag is set, an --output_dir must be also
> specified.
>@@ -38,25 +35,27 @@ For example:
>   gtest-parallel-wrapper.py some_test \
>       --some_flag=some_value \
>       --another_flag \
>-      --output_dir=SOME_OUTPUT_DIR
>-      -- \
>+      --output_dir=SOME_OUTPUT_DIR \
>       --store-test-artifacts
>       --isolated-script-test-output=SOME_DIR \
>       --isolated-script-test-perf-output=SOME_OTHER_DIR \
>+      -- \
>       --foo=bar \
>       --baz
> 
> Will be converted into:
> 
>-  python gtest-parallel some_test \
>-      --shard_count 1 \
>+  python gtest-parallel \
>       --shard_index 0 \
>-      --some_flag=some_value \
>-      --another_flag \
>+      --shard_count 1 \
>       --output_dir=SOME_OUTPUT_DIR \
>       --dump_json_test_results=SOME_DIR \
>+      some_test \
>       -- \
>       --test_artifacts_dir=SOME_OUTPUT_DIR/test_artifacts \
>+      --some_flag=some_value \
>+      --another_flag \
>+      --isolated-script-test-perf-output=SOME_OTHER_DIR \
>       --foo=bar \
>       --baz
> 
>@@ -83,68 +82,80 @@ def _CatFiles(file_list, output_file):
>       os.remove(filename)
> 
> 
>-def _GetOutputDir(gtest_parallel_args):
>-  parser = argparse.ArgumentParser()
>-  parser.add_argument('--output_dir', type=str, default=None)
>-  options, _ = parser.parse_known_args(gtest_parallel_args)
>-  return options.output_dir
>-
>-
>-def _ParseArgs():
>-  if '--' in sys.argv:
>-    argv_index = sys.argv.index('--')
>-  else:
>-    argv_index = len(sys.argv)
>+class ReconstructibleArgumentGroup(object):
>+  """An argument group that can be converted back into a command line.
>+
>+  This acts like ArgumentParser.add_argument_group, but names of arguments added
>+  to it are also kept in a list, so that parsed options from
>+  ArgumentParser.parse_args can be reconstructed back into a command line (list
>+  of args) based on the list of wanted keys."""
>+  def __init__(self, parser, *args, **kwargs):
>+    self._group = parser.add_argument_group(*args, **kwargs)
>+    self._keys = []
>+
>+  def AddArgument(self, *args, **kwargs):
>+    arg = self._group.add_argument(*args, **kwargs)
>+    self._keys.append(arg.dest)
>+
>+  def RemakeCommandLine(self, options):
>+    result = []
>+    for key in self._keys:
>+      value = getattr(options, key)
>+      if value is True:
>+        result.append('--%s' % key)
>+      elif value is not None:
>+        result.append('--%s=%s' % (key, value))
>+    return result
>+
>+
>+def ParseArgs(argv=None):
>+  parser = argparse.ArgumentParser(argv)
>+
>+  gtest_group = ReconstructibleArgumentGroup(parser,
>+                                             'Arguments to gtest-parallel')
>+  # These options will be passed unchanged to gtest-parallel.
>+  gtest_group.AddArgument('-d', '--output_dir')
>+  gtest_group.AddArgument('-r', '--repeat')
>+  gtest_group.AddArgument('--retry_failed')
>+  gtest_group.AddArgument('-w', '--workers')
>+  gtest_group.AddArgument('--gtest_color')
>+  gtest_group.AddArgument('--gtest_filter')
>+  gtest_group.AddArgument('--gtest_also_run_disabled_tests',
>+                          action='store_true', default=None)
>+  gtest_group.AddArgument('--timeout')
> 
>-  gtest_parallel_args = sys.argv[1:argv_index]
>-  executable_args = sys.argv[argv_index + 1:]
>-
>-  parser = argparse.ArgumentParser()
>-  parser.add_argument('--isolated-script-test-output', type=str, default=None)
>+  # --isolated-script-test-output is used to upload results to the flakiness
>+  # dashboard. This translation is made because gtest-parallel expects the flag
>+  # to be called --dump_json_test_results instead.
>+  gtest_group.AddArgument('--isolated-script-test-output',
>+                          dest='dump_json_test_results')
> 
>   # Needed when the test wants to store test artifacts, because it doesn't know
>   # what will be the swarming output dir.
>-  parser.add_argument('--store-test-artifacts', action='store_true',
>-                      default=False)
>-
>-  # We don't need to implement this flag, and possibly can't, since it's
>-  # intended for results of Telemetry tests. See
>-  # https://chromium.googlesource.com/external/github.com/catapult-project/catapult/+/HEAD/dashboard/docs/data-format.md
>-  parser.add_argument('--isolated-script-test-perf-output', type=str,
>-                      default=None)
>+  parser.add_argument('--store-test-artifacts', action='store_true')
> 
>   # No-sandbox is a Chromium-specific flag, ignore it.
>   # TODO(oprypin): Remove (bugs.webrtc.org/8115)
>-  parser.add_argument('--no-sandbox', action='store_true', default=False)
>+  parser.add_argument('--no-sandbox', action='store_true',
>+                      help=argparse.SUPPRESS)
> 
>-  # We have to do this, since --isolated-script-test-output is passed as an
>-  # argument to the executable by the swarming scripts, and we want to pass it
>-  # to gtest-parallel instead.
>-  options, executable_args = parser.parse_known_args(executable_args)
>+  parser.add_argument('executable')
>+  parser.add_argument('executable_args', nargs='*')
> 
>-  # --isolated-script-test-output is used to upload results to the flakiness
>-  # dashboard. This translation is made because gtest-parallel expects the flag
>-  # to be called --dump_json_test_results instead.
>-  if options.isolated_script_test_output:
>-    gtest_parallel_args += [
>-        '--dump_json_test_results',
>-        options.isolated_script_test_output,
>-    ]
>+  options, unrecognized_args = parser.parse_known_args(argv)
> 
>-  output_dir = _GetOutputDir(gtest_parallel_args)
>-  test_artifacts_dir = None
>+  executable_args = options.executable_args + unrecognized_args
> 
>   if options.store_test_artifacts:
>-    assert output_dir, (
>+    assert options.output_dir, (
>         '--output_dir must be specified for storing test artifacts.')
>-    test_artifacts_dir = os.path.join(output_dir, 'test_artifacts')
>-    if not os.path.isdir(test_artifacts_dir):
>-      os.makedirs(test_artifacts_dir)
>+    test_artifacts_dir = os.path.join(options.output_dir, 'test_artifacts')
>+
>+    executable_args.insert(0, '--test_artifacts_dir=%s' % test_artifacts_dir)
>+  else:
>+    test_artifacts_dir = None
> 
>-    executable_args += [
>-        '--test_artifacts_dir',
>-        test_artifacts_dir,
>-    ]
>+  gtest_parallel_args = gtest_group.RemakeCommandLine(options)
> 
>   # GTEST_SHARD_INDEX and GTEST_TOTAL_SHARDS must be removed from the
>   # environment. Otherwise it will be picked up by the binary, causing a bug
>@@ -153,14 +164,15 @@ def _ParseArgs():
>   gtest_shard_index = test_env.pop('GTEST_SHARD_INDEX', '0')
>   gtest_total_shards = test_env.pop('GTEST_TOTAL_SHARDS', '1')
> 
>-  gtest_parallel_args += [
>-      '--shard_count',
>-      gtest_total_shards,
>-      '--shard_index',
>-      gtest_shard_index,
>-  ] + ['--'] + executable_args
>+  gtest_parallel_args.insert(0, '--shard_index=%s' % gtest_shard_index)
>+  gtest_parallel_args.insert(1, '--shard_count=%s' % gtest_total_shards)
>+
>+  gtest_parallel_args.append(options.executable)
>+  if executable_args:
>+    gtest_parallel_args += ['--'] + executable_args
> 
>-  return Args(gtest_parallel_args, test_env, output_dir, test_artifacts_dir)
>+  return Args(gtest_parallel_args, test_env, options.output_dir,
>+              test_artifacts_dir)
> 
> 
> def main():
>@@ -168,32 +180,36 @@ def main():
>   gtest_parallel_path = os.path.join(
>       webrtc_root, 'third_party', 'gtest-parallel', 'gtest-parallel')
> 
>-  args = _ParseArgs()
>+  gtest_parallel_args, test_env, output_dir, test_artifacts_dir = ParseArgs()
> 
>   command = [
>       sys.executable,
>       gtest_parallel_path,
>-  ] + args.gtest_parallel_args
>+  ] + gtest_parallel_args
>+
>+  if output_dir and not os.path.isdir(output_dir):
>+    os.makedirs(output_dir)
>+  if test_artifacts_dir and not os.path.isdir(test_artifacts_dir):
>+    os.makedirs(test_artifacts_dir)
> 
>   print 'gtest-parallel-wrapper: Executing command %s' % ' '.join(command)
>   sys.stdout.flush()
> 
>-  exit_code = subprocess.call(command, env=args.test_env, cwd=os.getcwd())
>+  exit_code = subprocess.call(command, env=test_env, cwd=os.getcwd())
> 
>-  if args.output_dir:
>+  if output_dir:
>     for test_status in 'passed', 'failed', 'interrupted':
>-      logs_dir = os.path.join(args.output_dir, 'gtest-parallel-logs',
>-                              test_status)
>+      logs_dir = os.path.join(output_dir, 'gtest-parallel-logs', test_status)
>       if not os.path.isdir(logs_dir):
>         continue
>       logs = [os.path.join(logs_dir, log) for log in os.listdir(logs_dir)]
>-      log_file = os.path.join(args.output_dir, '%s-tests.log' % test_status)
>+      log_file = os.path.join(output_dir, '%s-tests.log' % test_status)
>       _CatFiles(logs, log_file)
>       os.rmdir(logs_dir)
> 
>-  if args.test_artifacts_dir:
>-    shutil.make_archive(args.test_artifacts_dir, 'zip', args.test_artifacts_dir)
>-    shutil.rmtree(args.test_artifacts_dir)
>+  if test_artifacts_dir:
>+    shutil.make_archive(test_artifacts_dir, 'zip', test_artifacts_dir)
>+    shutil.rmtree(test_artifacts_dir)
> 
>   return exit_code
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/gtest_parallel_wrapper_test.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/gtest_parallel_wrapper_test.py
>new file mode 100755
>index 00000000000..e731d922437
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/gtest_parallel_wrapper_test.py
>@@ -0,0 +1,129 @@
>+#!/usr/bin/env python
>+
>+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+#
>+# Use of this source code is governed by a BSD-style license
>+# that can be found in the LICENSE file in the root of the source
>+# tree. An additional intellectual property rights grant can be found
>+# in the file PATENTS.  All contributing project authors may
>+# be found in the AUTHORS file in the root of the source tree.
>+
>+from contextlib import contextmanager
>+
>+import os
>+import tempfile
>+import unittest
>+
>+script = __import__('gtest-parallel-wrapper')  # pylint: disable=invalid-name
>+
>+
>+@contextmanager
>+def TemporaryDirectory():
>+  tmp_dir = tempfile.mkdtemp()
>+  yield tmp_dir
>+  os.rmdir(tmp_dir)
>+
>+
>+class GtestParallelWrapperTest(unittest.TestCase):
>+  @classmethod
>+  def _Expected(cls, gtest_parallel_args):
>+    return ['--shard_index=0', '--shard_count=1'] + gtest_parallel_args
>+
>+  def testOverwrite(self):
>+    result = script.ParseArgs(['--timeout=123', 'exec', '--timeout', '124'])
>+    expected = self._Expected(['--timeout=124', 'exec'])
>+    self.assertEqual(result.gtest_parallel_args, expected)
>+
>+  def testMixing(self):
>+    result = script.ParseArgs(
>+        ['--timeout=123', '--param1', 'exec', '--param2', '--timeout', '124'])
>+    expected = self._Expected(
>+        ['--timeout=124', 'exec', '--', '--param1', '--param2'])
>+    self.assertEqual(result.gtest_parallel_args, expected)
>+
>+  def testMixingPositional(self):
>+    result = script.ParseArgs(['--timeout=123', 'exec', '--foo1', 'bar1',
>+                               '--timeout', '124', '--foo2', 'bar2'])
>+    expected = self._Expected(['--timeout=124', 'exec', '--', '--foo1', 'bar1',
>+                               '--foo2', 'bar2'])
>+    self.assertEqual(result.gtest_parallel_args, expected)
>+
>+  def testDoubleDash1(self):
>+    result = script.ParseArgs(
>+        ['--timeout', '123', 'exec', '--', '--timeout', '124'])
>+    expected = self._Expected(
>+        ['--timeout=123', 'exec', '--', '--timeout', '124'])
>+    self.assertEqual(result.gtest_parallel_args, expected)
>+
>+  def testDoubleDash2(self):
>+    result = script.ParseArgs(['--timeout=123', '--', 'exec', '--timeout=124'])
>+    expected = self._Expected(['--timeout=123', 'exec', '--', '--timeout=124'])
>+    self.assertEqual(result.gtest_parallel_args, expected)
>+
>+  def testArtifacts(self):
>+    with TemporaryDirectory() as tmp_dir:
>+      output_dir = os.path.join(tmp_dir, 'foo')
>+      result = script.ParseArgs(['exec', '--store-test-artifacts',
>+                                 '--output_dir', output_dir])
>+      exp_artifacts_dir = os.path.join(output_dir, 'test_artifacts')
>+      exp = self._Expected(['--output_dir=' + output_dir, 'exec', '--',
>+                            '--test_artifacts_dir=' + exp_artifacts_dir])
>+      self.assertEqual(result.gtest_parallel_args, exp)
>+      self.assertEqual(result.output_dir, output_dir)
>+      self.assertRegexpMatches(result.test_artifacts_dir,
>+                               output_dir + '.test_artifacts')
>+
>+  def testNoDirsSpecified(self):
>+    result = script.ParseArgs(['exec'])
>+    self.assertEqual(result.output_dir, None)
>+    self.assertEqual(result.test_artifacts_dir, None)
>+
>+  def testOutputDirSpecified(self):
>+    result = script.ParseArgs(['exec', '--output_dir', '/tmp/foo'])
>+    self.assertEqual(result.output_dir, '/tmp/foo')
>+    self.assertEqual(result.test_artifacts_dir, None)
>+
>+  def testJsonTestResults(self):
>+    result = script.ParseArgs(['--isolated-script-test-output', '/tmp/foo',
>+                               'exec'])
>+    expected = self._Expected(['--dump_json_test_results=/tmp/foo', 'exec'])
>+    self.assertEqual(result.gtest_parallel_args, expected)
>+
>+  def testShortArg(self):
>+    result = script.ParseArgs(['-d', '/tmp/foo', 'exec'])
>+    expected = self._Expected(['--output_dir=/tmp/foo', 'exec'])
>+    self.assertEqual(result.gtest_parallel_args, expected)
>+    self.assertEqual(result.output_dir, '/tmp/foo')
>+
>+  def testBoolArg(self):
>+    result = script.ParseArgs(['--gtest_also_run_disabled_tests', 'exec'])
>+    expected = self._Expected(['--gtest_also_run_disabled_tests', 'exec'])
>+    self.assertEqual(result.gtest_parallel_args, expected)
>+
>+  def testNoArgs(self):
>+    result = script.ParseArgs(['exec'])
>+    expected = self._Expected(['exec'])
>+    self.assertEqual(result.gtest_parallel_args, expected)
>+
>+  def testDocExample(self):
>+    with TemporaryDirectory() as tmp_dir:
>+      output_dir = os.path.join(tmp_dir, 'foo')
>+      result = script.ParseArgs([
>+          'some_test', '--some_flag=some_value', '--another_flag',
>+          '--output_dir=' + output_dir, '--store-test-artifacts',
>+          '--isolated-script-test-output=SOME_DIR',
>+          '--isolated-script-test-perf-output=SOME_OTHER_DIR',
>+          '--foo=bar', '--baz'])
>+      expected_artifacts_dir = os.path.join(output_dir, 'test_artifacts')
>+      expected = self._Expected([
>+          '--output_dir=' + output_dir, '--dump_json_test_results=SOME_DIR',
>+          'some_test', '--',
>+          '--test_artifacts_dir=' + expected_artifacts_dir,
>+          '--some_flag=some_value', '--another_flag',
>+          '--isolated-script-test-perf-output=SOME_OTHER_DIR',
>+          '--foo=bar', '--baz'])
>+      self.assertEqual(result.gtest_parallel_args, expected)
>+
>+
>+if __name__ == '__main__':
>+  unittest.main()
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/SDK/PodTest/PodTest/AppDelegate.m b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/SDK/PodTest/PodTest/AppDelegate.m
>deleted file mode 100644
>index 9eaa1682673..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/SDK/PodTest/PodTest/AppDelegate.m
>+++ /dev/null
>@@ -1,26 +0,0 @@
>-/*
>- *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#import "AppDelegate.h"
>-#import <WebRTC/WebRTC.h>
>-
>-@implementation AppDelegate
>-
>-- (BOOL)application:(UIApplication *)application
>-    didFinishLaunchingWithOptions:(NSDictionary *)launchOptions {
>-  RTCInitializeSSL();
>-  return YES;
>-}
>-
>-- (void)applicationWillTerminate:(UIApplication *)application {
>-  RTCCleanupSSL();
>-}
>-
>-@end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/SDK/PodTest/PodTest/Assets.xcassets/AppIcon.appiconset/Contents.json b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/SDK/PodTest/PodTest/Assets.xcassets/AppIcon.appiconset/Contents.json
>deleted file mode 100644
>index 36d2c80d889..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/SDK/PodTest/PodTest/Assets.xcassets/AppIcon.appiconset/Contents.json
>+++ /dev/null
>@@ -1,68 +0,0 @@
>-{
>-  "images" : [
>-    {
>-      "idiom" : "iphone",
>-      "size" : "29x29",
>-      "scale" : "2x"
>-    },
>-    {
>-      "idiom" : "iphone",
>-      "size" : "29x29",
>-      "scale" : "3x"
>-    },
>-    {
>-      "idiom" : "iphone",
>-      "size" : "40x40",
>-      "scale" : "2x"
>-    },
>-    {
>-      "idiom" : "iphone",
>-      "size" : "40x40",
>-      "scale" : "3x"
>-    },
>-    {
>-      "idiom" : "iphone",
>-      "size" : "60x60",
>-      "scale" : "2x"
>-    },
>-    {
>-      "idiom" : "iphone",
>-      "size" : "60x60",
>-      "scale" : "3x"
>-    },
>-    {
>-      "idiom" : "ipad",
>-      "size" : "29x29",
>-      "scale" : "1x"
>-    },
>-    {
>-      "idiom" : "ipad",
>-      "size" : "29x29",
>-      "scale" : "2x"
>-    },
>-    {
>-      "idiom" : "ipad",
>-      "size" : "40x40",
>-      "scale" : "1x"
>-    },
>-    {
>-      "idiom" : "ipad",
>-      "size" : "40x40",
>-      "scale" : "2x"
>-    },
>-    {
>-      "idiom" : "ipad",
>-      "size" : "76x76",
>-      "scale" : "1x"
>-    },
>-    {
>-      "idiom" : "ipad",
>-      "size" : "76x76",
>-      "scale" : "2x"
>-    }
>-  ],
>-  "info" : {
>-    "version" : 1,
>-    "author" : "xcode"
>-  }
>-}
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/SDK/PodTest/PodTest/Base.lproj/LaunchScreen.storyboard b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/SDK/PodTest/PodTest/Base.lproj/LaunchScreen.storyboard
>deleted file mode 100644
>index 2e721e1833f..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/SDK/PodTest/PodTest/Base.lproj/LaunchScreen.storyboard
>+++ /dev/null
>@@ -1,27 +0,0 @@
>-<?xml version="1.0" encoding="UTF-8" standalone="no"?>
>-<document type="com.apple.InterfaceBuilder3.CocoaTouch.Storyboard.XIB" version="3.0" toolsVersion="8150" systemVersion="15A204g" targetRuntime="iOS.CocoaTouch" propertyAccessControl="none" useAutolayout="YES" launchScreen="YES" useTraitCollections="YES" initialViewController="01J-lp-oVM">
>-    <dependencies>
>-        <plugIn identifier="com.apple.InterfaceBuilder.IBCocoaTouchPlugin" version="8122"/>
>-    </dependencies>
>-    <scenes>
>-        
>-        <scene sceneID="EHf-IW-A2E">
>-            <objects>
>-                <viewController id="01J-lp-oVM" sceneMemberID="viewController">
>-                    <layoutGuides>
>-                        <viewControllerLayoutGuide type="top" id="Llm-lL-Icb"/>
>-                        <viewControllerLayoutGuide type="bottom" id="xb3-aO-Qok"/>
>-                    </layoutGuides>
>-                    <view key="view" contentMode="scaleToFill" id="Ze5-6b-2t3">
>-                        <rect key="frame" x="0.0" y="0.0" width="600" height="600"/>
>-                        <autoresizingMask key="autoresizingMask" widthSizable="YES" heightSizable="YES"/>
>-                        <animations/>
>-                        <color key="backgroundColor" white="1" alpha="1" colorSpace="custom" customColorSpace="calibratedWhite"/>
>-                    </view>
>-                </viewController>
>-                <placeholder placeholderIdentifier="IBFirstResponder" id="iYj-Kq-Ea1" userLabel="First Responder" sceneMemberID="firstResponder"/>
>-            </objects>
>-            <point key="canvasLocation" x="53" y="375"/>
>-        </scene>
>-    </scenes>
>-</document>
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/SDK/PodTest/PodTest/Base.lproj/Main.storyboard b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/SDK/PodTest/PodTest/Base.lproj/Main.storyboard
>deleted file mode 100644
>index f56d2f3bb56..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/SDK/PodTest/PodTest/Base.lproj/Main.storyboard
>+++ /dev/null
>@@ -1,25 +0,0 @@
>-<?xml version="1.0" encoding="UTF-8" standalone="no"?>
>-<document type="com.apple.InterfaceBuilder3.CocoaTouch.Storyboard.XIB" version="3.0" toolsVersion="6211" systemVersion="14A298i" targetRuntime="iOS.CocoaTouch" propertyAccessControl="none" useAutolayout="YES" useTraitCollections="YES" initialViewController="BYZ-38-t0r">
>-    <dependencies>
>-        <plugIn identifier="com.apple.InterfaceBuilder.IBCocoaTouchPlugin" version="6204"/>
>-    </dependencies>
>-    <scenes>
>-        
>-        <scene sceneID="tne-QT-ifu">
>-            <objects>
>-                <viewController id="BYZ-38-t0r" customClass="ViewController" customModuleProvider="" sceneMemberID="viewController">
>-                    <layoutGuides>
>-                        <viewControllerLayoutGuide type="top" id="y3c-jy-aDJ"/>
>-                        <viewControllerLayoutGuide type="bottom" id="wfy-db-euE"/>
>-                    </layoutGuides>
>-                    <view key="view" contentMode="scaleToFill" id="8bC-Xf-vdC">
>-                        <rect key="frame" x="0.0" y="0.0" width="600" height="600"/>
>-                        <autoresizingMask key="autoresizingMask" widthSizable="YES" heightSizable="YES"/>
>-                        <color key="backgroundColor" white="1" alpha="1" colorSpace="custom" customColorSpace="calibratedWhite"/>
>-                    </view>
>-                </viewController>
>-                <placeholder placeholderIdentifier="IBFirstResponder" id="dkx-z0-nzr" sceneMemberID="firstResponder"/>
>-            </objects>
>-        </scene>
>-    </scenes>
>-</document>
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/SDK/PodTest/PodTest/Info.plist b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/SDK/PodTest/PodTest/Info.plist
>deleted file mode 100644
>index 40c6215d906..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/SDK/PodTest/PodTest/Info.plist
>+++ /dev/null
>@@ -1,47 +0,0 @@
>-<?xml version="1.0" encoding="UTF-8"?>
>-<!DOCTYPE plist PUBLIC "-//Apple//DTD PLIST 1.0//EN" "http://www.apple.com/DTDs/PropertyList-1.0.dtd">
>-<plist version="1.0">
>-<dict>
>-	<key>CFBundleDevelopmentRegion</key>
>-	<string>en</string>
>-	<key>CFBundleExecutable</key>
>-	<string>$(EXECUTABLE_NAME)</string>
>-	<key>CFBundleIdentifier</key>
>-	<string>$(PRODUCT_BUNDLE_IDENTIFIER)</string>
>-	<key>CFBundleInfoDictionaryVersion</key>
>-	<string>6.0</string>
>-	<key>CFBundleName</key>
>-	<string>$(PRODUCT_NAME)</string>
>-	<key>CFBundlePackageType</key>
>-	<string>APPL</string>
>-	<key>CFBundleShortVersionString</key>
>-	<string>1.0</string>
>-	<key>CFBundleSignature</key>
>-	<string>????</string>
>-	<key>CFBundleVersion</key>
>-	<string>1</string>
>-	<key>LSRequiresIPhoneOS</key>
>-	<true/>
>-	<key>UILaunchStoryboardName</key>
>-	<string>LaunchScreen</string>
>-	<key>UIMainStoryboardFile</key>
>-	<string>Main</string>
>-	<key>UIRequiredDeviceCapabilities</key>
>-	<array>
>-		<string>armv7</string>
>-	</array>
>-	<key>UISupportedInterfaceOrientations</key>
>-	<array>
>-		<string>UIInterfaceOrientationPortrait</string>
>-		<string>UIInterfaceOrientationLandscapeLeft</string>
>-		<string>UIInterfaceOrientationLandscapeRight</string>
>-	</array>
>-	<key>UISupportedInterfaceOrientations~ipad</key>
>-	<array>
>-		<string>UIInterfaceOrientationPortrait</string>
>-		<string>UIInterfaceOrientationPortraitUpsideDown</string>
>-		<string>UIInterfaceOrientationLandscapeLeft</string>
>-		<string>UIInterfaceOrientationLandscapeRight</string>
>-	</array>
>-</dict>
>-</plist>
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/SDK/PodTest/PodTest/ViewController.m b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/SDK/PodTest/PodTest/ViewController.m
>deleted file mode 100644
>index 39a8c76ff27..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/SDK/PodTest/PodTest/ViewController.m
>+++ /dev/null
>@@ -1,27 +0,0 @@
>-/*
>- *  Copyright 2016 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#import "ViewController.h"
>-#import <WebRTC/WebRTC.h>
>-
>-@interface ViewController ()
>-@property (nonatomic, strong) RTCPeerConnectionFactory *factory;
>-@property (nonatomic, strong) RTCAVFoundationVideoSource *videoSource;
>-@end
>-
>-@implementation ViewController
>-
>-- (void)viewDidLoad {
>-  [super viewDidLoad];
>-  self.factory = [[RTCPeerConnectionFactory alloc] init];
>-  self.videoSource = [self.factory avFoundationVideoSourceWithConstraints:nil];
>-}
>-
>-@end
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/SDK/PodTest/Podfile b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/SDK/PodTest/Podfile
>deleted file mode 100644
>index fec22d917fa..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/SDK/PodTest/Podfile
>+++ /dev/null
>@@ -1,9 +0,0 @@
>-# Uncomment this line to define a global platform for your project
>-# platform :ios, '8.0'
>-# Uncomment this line if you're using Swift
>-# use_frameworks!
>-
>-target 'PodTest' do
>-pod "WebRTC", :path => "../../../../../out_ios_libs"
>-end
>-
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/build_ios_libs.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/build_ios_libs.py
>old mode 100644
>new mode 100755
>index b48304e0322..46508cf15f3
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/build_ios_libs.py
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/build_ios_libs.py
>@@ -30,7 +30,6 @@ sys.path.append(os.path.join(SRC_DIR, 'build'))
> import find_depot_tools
> 
> SDK_OUTPUT_DIR = os.path.join(SRC_DIR, 'out_ios_libs')
>-SDK_LIB_NAME = 'librtc_sdk_objc.a'
> SDK_FRAMEWORK_NAME = 'WebRTC.framework'
> 
> DEFAULT_ARCHS = ENABLED_ARCHS = ['arm64', 'arm', 'x64', 'x86']
>@@ -43,10 +42,6 @@ from generate_licenses import LicenseBuilder
> 
> def _ParseArgs():
>   parser = argparse.ArgumentParser(description=__doc__)
>-  parser.add_argument('-b', '--build_type', default='framework',
>-      choices=['framework', 'static_only'],
>-      help='The build type. Can be "framework" or "static_only". '
>-           'Defaults to "framework".')
>   parser.add_argument('--build_config', default='release',
>       choices=['debug', 'release'],
>       help='The build config. Can be "debug" or "release". '
>@@ -98,7 +93,7 @@ def _CleanTemporary(output_dir, architectures):
> 
> def BuildWebRTC(output_dir, target_arch, flavor, gn_target_name,
>                 ios_deployment_target, libvpx_build_vp9, use_bitcode,
>-                use_goma, extra_gn_args, static_only):
>+                use_goma, extra_gn_args):
>   output_dir = os.path.join(output_dir, target_arch + '_libs')
>   gn_args = ['target_os="ios"', 'ios_enable_code_signing=false',
>              'use_xcode_clang=true', 'is_component_build=false']
>@@ -145,15 +140,6 @@ def BuildWebRTC(output_dir, target_arch, flavor, gn_target_name,
>     cmd.extend(['-j', '200'])
>   _RunCommand(cmd)
> 
>-  # Strip debug symbols to reduce size.
>-  if static_only:
>-    gn_target_path = os.path.join(output_dir, 'obj', 'sdk',
>-                                  'lib%s.a' % gn_target_name)
>-    cmd = ['strip', '-S', gn_target_path, '-o',
>-           os.path.join(output_dir, 'lib%s.a' % gn_target_name)]
>-    _RunCommand(cmd)
>-
>-
> def main():
>   args = _ParseArgs()
> 
>@@ -170,76 +156,56 @@ def main():
>     _CleanTemporary(args.output_dir, architectures)
>     return 0
> 
>-  # Ignoring x86 except for static libraries for now because of a GN build issue
>-  # where the generated dynamic framework has the wrong architectures.
>-  if 'x86' in architectures and args.build_type != 'static_only':
>-    architectures.remove('x86')
>-
>-  # Generate static or dynamic.
>-  if args.build_type == 'static_only':
>-    gn_target_name = 'rtc_sdk_objc'
>-  elif args.build_type == 'framework':
>-    gn_target_name = 'framework_objc'
>-    if not args.bitcode:
>-      gn_args.append('enable_dsyms=true')
>-    gn_args.append('enable_stripping=true')
>-  else:
>-    raise ValueError('Build type "%s" is not supported.' % args.build_type)
>+  gn_target_name = 'framework_objc'
>+  if not args.bitcode:
>+    gn_args.append('enable_dsyms=true')
>+  gn_args.append('enable_stripping=true')
> 
> 
>   # Build all architectures.
>   for arch in architectures:
>     BuildWebRTC(args.output_dir, arch, args.build_config, gn_target_name,
>                 IOS_DEPLOYMENT_TARGET, LIBVPX_BUILD_VP9, args.bitcode,
>-                args.use_goma, gn_args, args.build_type == 'static_only')
>+                args.use_goma, gn_args)
> 
>   # Create FAT archive.
>-  if args.build_type == 'static_only':
>-    lib_paths = [os.path.join(args.output_dir, arch + '_libs', SDK_LIB_NAME)
>-                 for arch in architectures]
>-    out_lib_path = os.path.join(args.output_dir, SDK_LIB_NAME)
>-    # Combine the slices.
>-    cmd = ['lipo'] + lib_paths + ['-create', '-output', out_lib_path]
>-    _RunCommand(cmd)
>-
>-  elif args.build_type == 'framework':
>-    lib_paths = [os.path.join(args.output_dir, arch + '_libs')
>-                 for arch in architectures]
>+  lib_paths = [os.path.join(args.output_dir, arch + '_libs')
>+                for arch in architectures]
>+
>+  # Combine the slices.
>+  dylib_path = os.path.join(SDK_FRAMEWORK_NAME, 'WebRTC')
>+  # Dylibs will be combined, all other files are the same across archs.
>+  # Use distutils instead of shutil to support merging folders.
>+  distutils.dir_util.copy_tree(
>+      os.path.join(lib_paths[0], SDK_FRAMEWORK_NAME),
>+      os.path.join(args.output_dir, SDK_FRAMEWORK_NAME))
>+  logging.info('Merging framework slices.')
>+  dylib_paths = [os.path.join(path, dylib_path) for path in lib_paths]
>+  out_dylib_path = os.path.join(args.output_dir, dylib_path)
>+  try:
>+    os.remove(out_dylib_path)
>+  except OSError:
>+    pass
>+  cmd = ['lipo'] + dylib_paths + ['-create', '-output', out_dylib_path]
>+  _RunCommand(cmd)
> 
>-    # Combine the slices.
>-    dylib_path = os.path.join(SDK_FRAMEWORK_NAME, 'WebRTC')
>-    # Dylibs will be combined, all other files are the same across archs.
>-    # Use distutils instead of shutil to support merging folders.
>-    distutils.dir_util.copy_tree(
>-        os.path.join(lib_paths[0], SDK_FRAMEWORK_NAME),
>-        os.path.join(args.output_dir, SDK_FRAMEWORK_NAME))
>-    logging.info('Merging framework slices.')
>-    dylib_paths = [os.path.join(path, dylib_path) for path in lib_paths]
>-    out_dylib_path = os.path.join(args.output_dir, dylib_path)
>+  # Merge the dSYM slices.
>+  lib_dsym_dir_path = os.path.join(lib_paths[0], 'WebRTC.dSYM')
>+  if os.path.isdir(lib_dsym_dir_path):
>+    distutils.dir_util.copy_tree(lib_dsym_dir_path,
>+                                  os.path.join(args.output_dir, 'WebRTC.dSYM'))
>+    logging.info('Merging dSYM slices.')
>+    dsym_path = os.path.join('WebRTC.dSYM', 'Contents', 'Resources', 'DWARF',
>+                              'WebRTC')
>+    lib_dsym_paths = [os.path.join(path, dsym_path) for path in lib_paths]
>+    out_dsym_path = os.path.join(args.output_dir, dsym_path)
>     try:
>-      os.remove(out_dylib_path)
>+      os.remove(out_dsym_path)
>     except OSError:
>       pass
>-    cmd = ['lipo'] + dylib_paths + ['-create', '-output', out_dylib_path]
>+    cmd = ['lipo'] + lib_dsym_paths + ['-create', '-output', out_dsym_path]
>     _RunCommand(cmd)
> 
>-    # Merge the dSYM slices.
>-    lib_dsym_dir_path = os.path.join(lib_paths[0], 'WebRTC.dSYM')
>-    if os.path.isdir(lib_dsym_dir_path):
>-      distutils.dir_util.copy_tree(lib_dsym_dir_path,
>-                                   os.path.join(args.output_dir, 'WebRTC.dSYM'))
>-      logging.info('Merging dSYM slices.')
>-      dsym_path = os.path.join('WebRTC.dSYM', 'Contents', 'Resources', 'DWARF',
>-                               'WebRTC')
>-      lib_dsym_paths = [os.path.join(path, dsym_path) for path in lib_paths]
>-      out_dsym_path = os.path.join(args.output_dir, dsym_path)
>-      try:
>-        os.remove(out_dsym_path)
>-      except OSError:
>-        pass
>-      cmd = ['lipo'] + lib_dsym_paths + ['-create', '-output', out_dsym_path]
>-      _RunCommand(cmd)
>-
>     # Generate the license file.
>     ninja_dirs = [os.path.join(args.output_dir, arch + '_libs')
>                   for arch in architectures]
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/build_ios_libs.sh b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/build_ios_libs.sh
>old mode 100644
>new mode 100755
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc.branches/iOS (beta).json b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc.branches/iOS (beta).json
>index ad87ca184ed..5ab132ede96 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc.branches/iOS (beta).json	
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc.branches/iOS (beta).json	
>@@ -2,7 +2,7 @@
>   "comments": [
>     "Tests for 64-bit iOS simulators."
>   ],
>-  "xcode version": "8.0",
>+  "xcode build version": "9C40b",
>   "gn_args": [
>     "goma_dir=\"$(goma_dir)\"",
>     "is_component_build=false",
>@@ -11,14 +11,24 @@
>     "target_os=\"ios\"",
>     "use_goma=true"
>   ],
>-  "additional_compile_targets": [ "all" ],
>+  "additional_compile_targets": [],
>   "configuration": "Debug",
>-  "sdk": "iphonesimulator10.0",
>+  "sdk": "iphonesimulator10.3",
>   "tests": [
>     {
>       "include": "common_tests.json",
>       "device type": "iPhone 6s",
>-      "os": "10.0"
>+      "os": "10.3",
>+      "dimensions": [
>+        {
>+          "os": "Mac-10.12",
>+          "pool": "Chrome"
>+        },
>+        {
>+          "os": "Mac-10.13.3",
>+          "pool": "Chrome"
>+        }
>+      ]
>     }
>   ]
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc.branches/iOS (stable).json b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc.branches/iOS (stable).json
>index ad87ca184ed..5ab132ede96 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc.branches/iOS (stable).json	
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc.branches/iOS (stable).json	
>@@ -2,7 +2,7 @@
>   "comments": [
>     "Tests for 64-bit iOS simulators."
>   ],
>-  "xcode version": "8.0",
>+  "xcode build version": "9C40b",
>   "gn_args": [
>     "goma_dir=\"$(goma_dir)\"",
>     "is_component_build=false",
>@@ -11,14 +11,24 @@
>     "target_os=\"ios\"",
>     "use_goma=true"
>   ],
>-  "additional_compile_targets": [ "all" ],
>+  "additional_compile_targets": [],
>   "configuration": "Debug",
>-  "sdk": "iphonesimulator10.0",
>+  "sdk": "iphonesimulator10.3",
>   "tests": [
>     {
>       "include": "common_tests.json",
>       "device type": "iPhone 6s",
>-      "os": "10.0"
>+      "os": "10.3",
>+      "dimensions": [
>+        {
>+          "os": "Mac-10.12",
>+          "pool": "Chrome"
>+        },
>+        {
>+          "os": "Mac-10.13.3",
>+          "pool": "Chrome"
>+        }
>+      ]
>     }
>   ]
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS32_Debug.json b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS32_Debug.json
>index 4f86af67a0d..4ae5e994159 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS32_Debug.json
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS32_Debug.json
>@@ -2,7 +2,7 @@
>   "comments": [
>     "Builder for 32-bit devices."
>   ],
>-  "xcode version": "8.0",
>+  "xcode build version": "9C40b",
>   "gn_args": [
>     "goma_dir=\"$(goma_dir)\"",
>     "ios_enable_code_signing=false",
>@@ -14,7 +14,7 @@
>   ],
>   "additional_compile_targets": [ "all" ],
>   "configuration": "Debug",
>-  "sdk": "iphoneos10.0",
>+  "sdk": "iphoneos10.3",
>   "tests": [
>   ]
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS32_Release.json b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS32_Release.json
>index 1b5ba85d975..ed915458bb0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS32_Release.json
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS32_Release.json
>@@ -2,7 +2,7 @@
>   "comments": [
>     "Builder for 32-bit devices."
>   ],
>-  "xcode version": "8.0",
>+  "xcode build version": "9C40b",
>   "gn_args": [
>     "goma_dir=\"$(goma_dir)\"",
>     "dcheck_always_on=true",
>@@ -15,7 +15,7 @@
>   ],
>   "additional_compile_targets": [ "all" ],
>   "configuration": "Release",
>-  "sdk": "iphoneos10.0",
>+  "sdk": "iphoneos10.3",
>   "tests": [
>   ]
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS32_Sim_Debug_(iOS_9.0).json b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS32_Sim_Debug_(iOS_9.0).json
>index b3c0028f592..54e3c17fec5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS32_Sim_Debug_(iOS_9.0).json
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS32_Sim_Debug_(iOS_9.0).json
>@@ -2,7 +2,7 @@
>   "comments": [
>     "Tests for 32-bit iOS simulators."
>   ],
>-  "xcode version": "8.0",
>+  "xcode build version": "9C40b",
>   "gn_args": [
>     "goma_dir=\"$(goma_dir)\"",
>     "is_component_build=false",
>@@ -13,12 +13,22 @@
>   ],
>   "additional_compile_targets": [ "all" ],
>   "configuration": "Debug",
>-  "sdk": "iphonesimulator9.0",
>+  "sdk": "iphonesimulator9.3",
>   "tests": [
>     {
>       "include": "common_tests.json",
>       "device type": "iPhone 5",
>-      "os": "9.0"
>+      "os": "9.3",
>+      "dimensions": [
>+        {
>+          "os": "Mac-10.12",
>+          "pool": "Chrome"
>+        },
>+        {
>+          "os": "Mac-10.13",
>+          "pool": "Chrome"
>+        }
>+      ]
>     }
>   ]
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS64_Debug.json b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS64_Debug.json
>index 5d7b77bc3d6..2af5805c33d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS64_Debug.json
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS64_Debug.json
>@@ -2,7 +2,7 @@
>   "comments": [
>     "Builder for 64-bit devices."
>   ],
>-  "xcode version": "8.0",
>+  "xcode build version": "9C40b",
>   "gn_args": [
>     "goma_dir=\"$(goma_dir)\"",
>     "ios_enable_code_signing=false",
>@@ -14,7 +14,7 @@
>   ],
>   "additional_compile_targets": [ "all" ],
>   "configuration": "Debug",
>-  "sdk": "iphoneos10.0",
>+  "sdk": "iphoneos10.3",
>   "tests": [
>   ]
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS64_Release.json b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS64_Release.json
>index 71242ae740f..a59ef83fd66 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS64_Release.json
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS64_Release.json
>@@ -2,7 +2,7 @@
>   "comments": [
>     "Builder for 64-bit devices."
>   ],
>-  "xcode version": "8.0",
>+  "xcode build version": "9C40b",
>   "gn_args": [
>     "goma_dir=\"$(goma_dir)\"",
>     "dcheck_always_on=true",
>@@ -15,7 +15,7 @@
>   ],
>   "additional_compile_targets": [ "all" ],
>   "configuration": "Release",
>-  "sdk": "iphoneos10.0",
>+  "sdk": "iphoneos10.3",
>   "tests": [
>   ]
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS64_Sim_Debug_(iOS_10.0).json b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS64_Sim_Debug_(iOS_10.0).json
>index ad87ca184ed..f5c9d34fb6d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS64_Sim_Debug_(iOS_10.0).json
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS64_Sim_Debug_(iOS_10.0).json
>@@ -2,7 +2,7 @@
>   "comments": [
>     "Tests for 64-bit iOS simulators."
>   ],
>-  "xcode version": "8.0",
>+  "xcode build version": "9C40b",
>   "gn_args": [
>     "goma_dir=\"$(goma_dir)\"",
>     "is_component_build=false",
>@@ -13,12 +13,23 @@
>   ],
>   "additional_compile_targets": [ "all" ],
>   "configuration": "Debug",
>-  "sdk": "iphonesimulator10.0",
>+  "sdk": "iphonesimulator10.3",
>   "tests": [
>     {
>       "include": "common_tests.json",
>       "device type": "iPhone 6s",
>-      "os": "10.0"
>+      "os": "10.3",
>+      "xcode build version": "8E3004b",
>+      "dimensions": [
>+        {
>+          "os": "Mac-10.12",
>+          "pool": "Chrome"
>+        },
>+        {
>+          "os": "Mac-10.13.3",
>+          "pool": "Chrome"
>+        }
>+      ]
>     }
>   ]
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS64_Sim_Debug_(iOS_11.0).json b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS64_Sim_Debug_(iOS_11.0).json
>index 0f71b12c132..c25e3d1b14b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS64_Sim_Debug_(iOS_11.0).json
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS64_Sim_Debug_(iOS_11.0).json
>@@ -2,7 +2,7 @@
>   "comments": [
>     "Tests for 64-bit iOS simulators."
>   ],
>-  "xcode version": "8.0",
>+  "xcode build version": "9C40b",
>   "gn_args": [
>     "goma_dir=\"$(goma_dir)\"",
>     "is_component_build=false",
>@@ -18,7 +18,17 @@
>     {
>       "include": "common_tests.json",
>       "device type": "iPhone 7",
>-      "os": "11.0"
>+      "os": "11.0",
>+      "dimensions": [
>+        {
>+          "os": "Mac-10.12",
>+          "pool": "Chrome"
>+        },
>+        {
>+          "os": "Mac-10.13",
>+          "pool": "Chrome"
>+        }
>+      ]
>     }
>   ]
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS64_Sim_Debug_(iOS_9.0).json b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS64_Sim_Debug_(iOS_9.0).json
>index d13fa95d36f..929897747ab 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS64_Sim_Debug_(iOS_9.0).json
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/client.webrtc/iOS64_Sim_Debug_(iOS_9.0).json
>@@ -2,7 +2,7 @@
>   "comments": [
>     "Tests for 64-bit iOS simulators."
>   ],
>-  "xcode version": "8.0",
>+  "xcode build version": "9C40b",
>   "gn_args": [
>     "goma_dir=\"$(goma_dir)\"",
>     "is_component_build=false",
>@@ -13,12 +13,22 @@
>   ],
>   "additional_compile_targets": [ "all" ],
>   "configuration": "Debug",
>-  "sdk": "iphonesimulator9.0",
>+  "sdk": "iphonesimulator9.3",
>   "tests": [
>     {
>       "include": "common_tests.json",
>       "device type": "iPhone 6s",
>-      "os": "9.0"
>+      "os": "9.3",
>+      "dimensions": [
>+        {
>+          "os": "Mac-10.12",
>+          "pool": "Chrome"
>+        },
>+        {
>+          "os": "Mac-10.13",
>+          "pool": "Chrome"
>+        }
>+      ]
>     }
>   ]
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/generate_umbrella_header.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/generate_umbrella_header.py
>new file mode 100644
>index 00000000000..d8a87bf72a5
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/generate_umbrella_header.py
>@@ -0,0 +1,40 @@
>+# Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+#
>+# Use of this source code is governed by a BSD-style license
>+# that can be found in the LICENSE file in the root of the source
>+# tree. An additional intellectual property rights grant can be found
>+# in the file PATENTS.  All contributing project authors may
>+# be found in the AUTHORS file in the root of the source tree.
>+
>+import argparse
>+import datetime
>+import sys
>+
>+
>+def GenerateUmbrellaHeader():
>+  parser = argparse.ArgumentParser(description='Generate umbrella header')
>+  parser.add_argument("-o", "--out", type=str, help="Output file.")
>+  parser.add_argument("-s", "--sources", default=[], type=str, nargs='+',
>+                      help="Headers to include.")
>+
>+  args = parser.parse_args()
>+
>+  with open(args.out, "w") as outfile:
>+    outfile.write("""/*
>+     *  Copyright %d The WebRTC project authors. All Rights Reserved.
>+     *
>+     *  Use of this source code is governed by a BSD-style license
>+     *  that can be found in the LICENSE file in the root of the source
>+     *  tree. An additional intellectual property rights grant can be found
>+     *  in the file PATENTS.  All contributing project authors may
>+     *  be found in the AUTHORS file in the root of the source tree.
>+     */\n\n""" % datetime.datetime.now().year)
>+
>+    for s in args.sources:
>+      outfile.write("#import <{}>\n".format(s))
>+
>+  return 0
>+
>+
>+if __name__ == '__main__':
>+  sys.exit(GenerateUmbrellaHeader())
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/internal.client.webrtc/iOS64_Debug.json b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/internal.client.webrtc/iOS64_Debug.json
>index 51e5e0a00fb..60d916d476d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/internal.client.webrtc/iOS64_Debug.json
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/internal.client.webrtc/iOS64_Debug.json
>@@ -2,7 +2,7 @@
>   "comments": [
>     "Tests for 64-bit iOS devices."
>   ],
>-  "xcode version": "9.0",
>+  "xcode build version": "9C40b",
>   "gn_args": [
>     "goma_dir=\"$(goma_dir)\"",
>     "is_component_build=false",
>@@ -15,14 +15,20 @@
>     "MAC_TOOLCHAIN_REVISION": "8E2002-1"
>   },
>   "use_analyze": false,
>-  "additional_compile_targets": [ "all" ],
>+  "additional_compile_targets": [],
>   "configuration": "Debug",
>   "bucket": "chromium-webrtc",
>   "tests": [
>     {
>       "include": "real_device_tests.json",
>       "device type": "iPhone 6s",
>-      "os": "10.3.3"
>+      "os": "10.3.3",
>+      "dimensions": [
>+        {
>+          "os": "Mac",
>+          "pool": "Chrome"
>+        }
>+      ]
>     }
>   ]
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/internal.client.webrtc/iOS64_Perf.json b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/internal.client.webrtc/iOS64_Perf.json
>index fd2a7867e53..58383fae4a0 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/internal.client.webrtc/iOS64_Perf.json
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/internal.client.webrtc/iOS64_Perf.json
>@@ -2,7 +2,7 @@
>   "comments": [
>     "Tests for 64-bit iOS devices."
>   ],
>-  "xcode version": "9.0",
>+  "xcode build version": "9C40b",
>   "gn_args": [
>     "goma_dir=\"$(goma_dir)\"",
>     "is_component_build=false",
>@@ -15,14 +15,20 @@
>     "MAC_TOOLCHAIN_REVISION": "8E2002-1"
>   },
>   "use_analyze": false,
>-  "additional_compile_targets": [ "all" ],
>+  "additional_compile_targets": [],
>   "configuration": "Release",
>   "bucket": "chromium-webrtc",
>   "tests": [
>     {
>       "include": "perf_tests.json",
>       "device type": "iPhone 7",
>-      "os": "11.0.3"
>+      "os": "11.0.3",
>+      "dimensions": [
>+        {
>+          "os": "Mac",
>+          "pool": "Chrome"
>+        }
>+      ]
>     }
>   ]
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/internal.client.webrtc/iOS64_Release.json b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/internal.client.webrtc/iOS64_Release.json
>index 3348039dad6..a894dba9a39 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/internal.client.webrtc/iOS64_Release.json
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/internal.client.webrtc/iOS64_Release.json
>@@ -2,7 +2,7 @@
>   "comments": [
>     "Tests for 64-bit iOS devices."
>   ],
>-  "xcode version": "9.0",
>+  "xcode build version": "9C40b",
>   "gn_args": [
>     "dcheck_always_on=true",
>     "goma_dir=\"$(goma_dir)\"",
>@@ -16,14 +16,20 @@
>     "MAC_TOOLCHAIN_REVISION": "8E2002-1"
>   },
>   "use_analyze": false,
>-  "additional_compile_targets": [ "all" ],
>+  "additional_compile_targets": [],
>   "configuration": "Release",
>   "bucket": "chromium-webrtc",
>   "tests": [
>     {
>       "include": "real_device_tests.json",
>-      "device type": "iPhone 7",
>-      "os": "11.0"
>+      "device type": "iPhone 6s",
>+      "os": "10.3.3",
>+      "dimensions": [
>+        {
>+          "os": "Mac",
>+          "pool": "Chrome"
>+        }
>+      ]
>     }
>   ],
>   "expiration_time": 3600
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/internal.tryserver.webrtc/ios_arm64_dbg.json b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/internal.tryserver.webrtc/ios_arm64_dbg.json
>index 51e5e0a00fb..60d916d476d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/internal.tryserver.webrtc/ios_arm64_dbg.json
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/internal.tryserver.webrtc/ios_arm64_dbg.json
>@@ -2,7 +2,7 @@
>   "comments": [
>     "Tests for 64-bit iOS devices."
>   ],
>-  "xcode version": "9.0",
>+  "xcode build version": "9C40b",
>   "gn_args": [
>     "goma_dir=\"$(goma_dir)\"",
>     "is_component_build=false",
>@@ -15,14 +15,20 @@
>     "MAC_TOOLCHAIN_REVISION": "8E2002-1"
>   },
>   "use_analyze": false,
>-  "additional_compile_targets": [ "all" ],
>+  "additional_compile_targets": [],
>   "configuration": "Debug",
>   "bucket": "chromium-webrtc",
>   "tests": [
>     {
>       "include": "real_device_tests.json",
>       "device type": "iPhone 6s",
>-      "os": "10.3.3"
>+      "os": "10.3.3",
>+      "dimensions": [
>+        {
>+          "os": "Mac",
>+          "pool": "Chrome"
>+        }
>+      ]
>     }
>   ]
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/internal.tryserver.webrtc/ios_arm64.json b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/internal.tryserver.webrtc/ios_arm64_perf.json
>similarity index 52%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/internal.tryserver.webrtc/ios_arm64.json
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/internal.tryserver.webrtc/ios_arm64_perf.json
>index 51e5e0a00fb..566876949e8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/internal.tryserver.webrtc/ios_arm64.json
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/internal.tryserver.webrtc/ios_arm64_perf.json
>@@ -2,11 +2,11 @@
>   "comments": [
>     "Tests for 64-bit iOS devices."
>   ],
>-  "xcode version": "9.0",
>+  "xcode build version": "9C40b",
>   "gn_args": [
>     "goma_dir=\"$(goma_dir)\"",
>     "is_component_build=false",
>-    "is_debug=true",
>+    "is_debug=false",
>     "target_cpu=\"arm64\"",
>     "target_os=\"ios\"",
>     "use_goma=true"
>@@ -15,14 +15,20 @@
>     "MAC_TOOLCHAIN_REVISION": "8E2002-1"
>   },
>   "use_analyze": false,
>-  "additional_compile_targets": [ "all" ],
>-  "configuration": "Debug",
>+  "additional_compile_targets": [],
>+  "configuration": "Release",
>   "bucket": "chromium-webrtc",
>   "tests": [
>     {
>-      "include": "real_device_tests.json",
>-      "device type": "iPhone 6s",
>-      "os": "10.3.3"
>+      "include": "perf_trybot_tests.json",
>+      "device type": "iPhone 7",
>+      "os": "11.0.3",
>+      "dimensions": [
>+        {
>+          "os": "Mac",
>+          "pool": "Chrome"
>+        }
>+      ]
>     }
>   ]
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/internal.tryserver.webrtc/ios_arm64_rel.json b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/internal.tryserver.webrtc/ios_arm64_rel.json
>index 3348039dad6..a894dba9a39 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/internal.tryserver.webrtc/ios_arm64_rel.json
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/internal.tryserver.webrtc/ios_arm64_rel.json
>@@ -2,7 +2,7 @@
>   "comments": [
>     "Tests for 64-bit iOS devices."
>   ],
>-  "xcode version": "9.0",
>+  "xcode build version": "9C40b",
>   "gn_args": [
>     "dcheck_always_on=true",
>     "goma_dir=\"$(goma_dir)\"",
>@@ -16,14 +16,20 @@
>     "MAC_TOOLCHAIN_REVISION": "8E2002-1"
>   },
>   "use_analyze": false,
>-  "additional_compile_targets": [ "all" ],
>+  "additional_compile_targets": [],
>   "configuration": "Release",
>   "bucket": "chromium-webrtc",
>   "tests": [
>     {
>       "include": "real_device_tests.json",
>-      "device type": "iPhone 7",
>-      "os": "11.0"
>+      "device type": "iPhone 6s",
>+      "os": "10.3.3",
>+      "dimensions": [
>+        {
>+          "os": "Mac",
>+          "pool": "Chrome"
>+        }
>+      ]
>     }
>   ],
>   "expiration_time": 3600
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/merge_ios_libs.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/merge_ios_libs.py
>old mode 100644
>new mode 100755
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tests/common_tests.json b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tests/common_tests.json
>index c0ad8952883..de23231bd6d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tests/common_tests.json
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tests/common_tests.json
>@@ -53,9 +53,6 @@
>     {
>       "app": "video_engine_tests"
>     },
>-    {
>-      "app": "voice_engine_unittests"
>-    },
>     {
>       "app": "webrtc_nonparallel_tests"
>     }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tests/perf_tests.json b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tests/perf_tests.json
>index bc6f0b9277d..a249b98445a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tests/perf_tests.json
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tests/perf_tests.json
>@@ -2,7 +2,10 @@
>   "tests": [
>     {
>       "app": "webrtc_perf_tests",
>-      "bot_id": "build15-a7",
>+      "test args": [
>+        "--save_chartjson_result"
>+      ],
>+      "bot id": "build15-a7",
>       "pool": "WebRTC"
>     }
>   ]
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tests/perf_trybot_tests.json b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tests/perf_trybot_tests.json
>new file mode 100644
>index 00000000000..02a792ad2a2
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tests/perf_trybot_tests.json
>@@ -0,0 +1,12 @@
>+{
>+  "tests": [
>+    {
>+      "app": "webrtc_perf_tests",
>+      "test args": [
>+        "--save_chartjson_result"
>+      ],
>+      "bot id": "build16-a7",
>+      "pool": "WebRTC"
>+    }
>+  ]
>+}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tests/real_device_tests.json b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tests/real_device_tests.json
>index 11341f8506f..8417470ddb8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tests/real_device_tests.json
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tests/real_device_tests.json
>@@ -35,9 +35,6 @@
>     },
>     {
>       "app": "video_engine_tests"
>-    },
>-    {
>-      "app": "voice_engine_unittests"
>     }
>   ]
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios32_sim_ios9_dbg.json b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios32_sim_ios9_dbg.json
>index 6c4d6980359..8eeece96ff4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios32_sim_ios9_dbg.json
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios32_sim_ios9_dbg.json
>@@ -2,7 +2,7 @@
>   "comments": [
>     "Tests for 32-bit iOS simulators."
>   ],
>-  "xcode version": "8.0",
>+  "xcode build version": "9C40b",
>   "gn_args": [
>     "goma_dir=\"$(goma_dir)\"",
>     "is_component_build=false",
>@@ -14,12 +14,22 @@
>   "use_analyze": false,
>   "additional_compile_targets": [ "all" ],
>   "configuration": "Debug",
>-  "sdk": "iphonesimulator9.0",
>+  "sdk": "iphonesimulator9.3",
>   "tests": [
>     {
>       "include": "common_tests.json",
>       "device type": "iPhone 5",
>-      "os": "9.0"
>+      "os": "9.3",
>+      "dimensions": [
>+        {
>+          "os": "Mac-10.12",
>+          "pool": "Chrome"
>+        },
>+        {
>+          "os": "Mac-10.13",
>+          "pool": "Chrome"
>+        }
>+      ]
>     }
>   ]
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios64_sim_ios10_dbg.json b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios64_sim_ios10_dbg.json
>index 25573e89a84..5e98758ae0b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios64_sim_ios10_dbg.json
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios64_sim_ios10_dbg.json
>@@ -2,7 +2,7 @@
>   "comments": [
>     "Tests for 64-bit iOS simulators."
>   ],
>-  "xcode version": "8.0",
>+  "xcode build version": "9C40b",
>   "gn_args": [
>     "goma_dir=\"$(goma_dir)\"",
>     "is_component_build=false",
>@@ -14,12 +14,23 @@
>   "use_analyze": false,
>   "additional_compile_targets": [ "all" ],
>   "configuration": "Debug",
>-  "sdk": "iphonesimulator10.0",
>+  "sdk": "iphonesimulator10.3",
>   "tests": [
>     {
>       "include": "common_tests.json",
>       "device type": "iPhone 6s",
>-      "os": "10.0"
>+      "os": "10.3",
>+      "xcode build version": "8E3004b",
>+      "dimensions": [
>+        {
>+          "os": "Mac-10.12",
>+          "pool": "Chrome"
>+        },
>+        {
>+          "os": "Mac-10.13.3",
>+          "pool": "Chrome"
>+        }
>+      ]
>     }
>   ]
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios64_sim_ios11_dbg.json b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios64_sim_ios11_dbg.json
>index eb5f26ada01..c145ddaf400 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios64_sim_ios11_dbg.json
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios64_sim_ios11_dbg.json
>@@ -2,7 +2,7 @@
>   "comments": [
>     "Tests for 64-bit iOS simulators."
>   ],
>-  "xcode version": "8.0",
>+  "xcode build version": "9C40b",
>   "gn_args": [
>     "goma_dir=\"$(goma_dir)\"",
>     "is_component_build=false",
>@@ -19,7 +19,17 @@
>     {
>       "include": "common_tests.json",
>       "device type": "iPhone 7",
>-      "os": "11.0"
>+      "os": "11.0",
>+      "dimensions": [
>+        {
>+          "os": "Mac-10.12",
>+          "pool": "Chrome"
>+        },
>+        {
>+          "os": "Mac-10.13",
>+          "pool": "Chrome"
>+        }
>+      ]
>     }
>   ]
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios64_sim_ios9_dbg.json b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios64_sim_ios9_dbg.json
>index 0cabd240231..2c0d62be4c7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios64_sim_ios9_dbg.json
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios64_sim_ios9_dbg.json
>@@ -2,7 +2,7 @@
>   "comments": [
>     "Tests for 64-bit iOS simulators."
>   ],
>-  "xcode version": "8.0",
>+  "xcode build version": "9C40b",
>   "gn_args": [
>     "goma_dir=\"$(goma_dir)\"",
>     "is_component_build=false",
>@@ -14,12 +14,22 @@
>   "use_analyze": false,
>   "additional_compile_targets": [ "all" ],
>   "configuration": "Debug",
>-  "sdk": "iphonesimulator9.0",
>+  "sdk": "iphonesimulator9.3",
>   "tests": [
>     {
>       "include": "common_tests.json",
>       "device type": "iPhone 6s",
>-      "os": "9.0"
>+      "os": "9.3",
>+      "dimensions": [
>+        {
>+          "os": "Mac-10.12",
>+          "pool": "Chrome"
>+        },
>+        {
>+          "os": "Mac-10.13",
>+          "pool": "Chrome"
>+        }
>+      ]
>     }
>   ]
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios_arm64_dbg.json b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios_arm64_dbg.json
>index 07e2e6e30cf..a718b7acad8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios_arm64_dbg.json
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios_arm64_dbg.json
>@@ -2,7 +2,7 @@
>   "comments": [
>     "Builder for 64-bit devices."
>   ],
>-  "xcode version": "8.0",
>+  "xcode build version": "9C40b",
>   "gn_args": [
>     "goma_dir=\"$(goma_dir)\"",
>     "ios_enable_code_signing=false",
>@@ -15,7 +15,7 @@
>   "use_analyze": false,
>   "additional_compile_targets": [ "all" ],
>   "configuration": "Debug",
>-  "sdk": "iphoneos10.0",
>+  "sdk": "iphoneos10.3",
>   "tests": [
>   ]
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios_arm64_rel.json b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios_arm64_rel.json
>index 1f346dd6e1b..7d2f923b574 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios_arm64_rel.json
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios_arm64_rel.json
>@@ -2,7 +2,7 @@
>   "comments": [
>     "Builder for 64-bit devices."
>   ],
>-  "xcode version": "8.0",
>+  "xcode build version": "9C40b",
>   "gn_args": [
>     "goma_dir=\"$(goma_dir)\"",
>     "dcheck_always_on=true",
>@@ -16,7 +16,7 @@
>   "use_analyze": false,
>   "additional_compile_targets": [ "all" ],
>   "configuration": "Release",
>-  "sdk": "iphoneos10.0",
>+  "sdk": "iphoneos10.3",
>   "tests": [
>   ]
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios_dbg.json b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios_dbg.json
>index ea039ca4206..4c5aca5ef7e 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios_dbg.json
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios_dbg.json
>@@ -2,7 +2,7 @@
>   "comments": [
>     "Builder for 32-bit devices."
>   ],
>-  "xcode version": "8.0",
>+  "xcode build version": "9C40b",
>   "gn_args": [
>     "goma_dir=\"$(goma_dir)\"",
>     "ios_enable_code_signing=false",
>@@ -15,7 +15,7 @@
>   "use_analyze": false,
>   "additional_compile_targets": [ "all" ],
>   "configuration": "Debug",
>-  "sdk": "iphoneos10.0",
>+  "sdk": "iphoneos10.3",
>   "tests": [
>   ]
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios_rel.json b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios_rel.json
>index 154a3ed92ab..81c16db60a4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios_rel.json
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/ios/tryserver.webrtc/ios_rel.json
>@@ -2,7 +2,7 @@
>   "comments": [
>     "Builder for 32-bit devices."
>   ],
>-  "xcode version": "8.0",
>+  "xcode build version": "9C40b",
>   "gn_args": [
>     "goma_dir=\"$(goma_dir)\"",
>     "dcheck_always_on=true",
>@@ -16,7 +16,7 @@
>   "use_analyze": false,
>   "additional_compile_targets": [ "all" ],
>   "configuration": "Release",
>-  "sdk": "iphoneos10.0",
>+  "sdk": "iphoneos10.3",
>   "tests": [
>   ]
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/tests/ARDSDPUtils_xctest.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/libs/__init__.py
>similarity index 100%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/examples/objc/AppRTCMobile/tests/ARDSDPUtils_xctest.mm
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/libs/__init__.py
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/libs/generate_licenses.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/libs/generate_licenses.py
>old mode 100644
>new mode 100755
>index 5cb8011978b..7f2ec46aab2
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/libs/generate_licenses.py
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/libs/generate_licenses.py
>@@ -22,14 +22,18 @@ import subprocess
> 
> 
> LIB_TO_LICENSES_DICT = {
>+    'abseil-cpp': ['third_party/abseil-cpp/LICENSE'],
>     'android_tools': ['third_party/android_tools/LICENSE'],
>     'auto': ['third_party/auto/src/LICENSE.txt'],
>+    'bazel': ['third_party/bazel/LICENSE'],
>     'boringssl': ['third_party/boringssl/src/LICENSE'],
>     'errorprone': ['third_party/errorprone/LICENSE'],
>     'expat': ['third_party/expat/files/COPYING'],
>+    'fiat': ['third_party/boringssl/src/third_party/fiat/LICENSE'],
>     'guava': ['third_party/guava/LICENSE'],
>     'ijar': ['third_party/ijar/LICENSE'],
>     'jsoncpp': ['third_party/jsoncpp/LICENSE'],
>+    'jsr-305': ['third_party/jsr-305/src/ri/LICENSE'],
>     'libc++': ['buildtools/third_party/libc++/trunk/LICENSE.TXT'],
>     'libc++abi': ['buildtools/third_party/libc++abi/trunk/LICENSE.TXT'],
>     'libevent': ['base/third_party/libevent/LICENSE'],
>@@ -37,15 +41,24 @@ LIB_TO_LICENSES_DICT = {
>     'libsrtp': ['third_party/libsrtp/LICENSE'],
>     'libvpx': ['third_party/libvpx/source/libvpx/LICENSE'],
>     'libyuv': ['third_party/libyuv/LICENSE'],
>-    'openmax_dl': ['third_party/openmax_dl/LICENSE'],
>     'opus': ['third_party/opus/src/COPYING'],
>     'protobuf': ['third_party/protobuf/LICENSE'],
>+    'rnnoise': ['third_party/rnnoise/COPYING'],
>     'usrsctp': ['third_party/usrsctp/LICENSE'],
>-    'webrtc': ['LICENSE', 'LICENSE_THIRD_PARTY'],
>-    'zlib': ['LICENSE', 'third_party/zlib/LICENSE'],
>+    'webrtc': ['LICENSE'],
>+    'zlib': ['third_party/zlib/LICENSE'],
>+    'base64': ['rtc_base/third_party/base64/LICENSE'],
>+    'sigslot': ['rtc_base/third_party/sigslot/LICENSE'],
>+    'portaudio': ['modules/third_party/portaudio/LICENSE'],
>+    'fft': ['modules/third_party/fft/LICENSE'],
>+    'g711': ['modules/third_party/g711/LICENSE'],
>+    'g722': ['modules/third_party/g722/LICENSE'],
>+    'fft4g': ['common_audio/third_party/fft4g/LICENSE'],
>+    'spl_sqrt_floor': ['common_audio/third_party/spl_sqrt_floor/LICENSE'],
> 
>     # Compile time dependencies, no license needed:
>     'yasm': [],
>+    'ow2_asm': [],
> }
> 
> SCRIPT_DIR = os.path.dirname(os.path.realpath(sys.argv[0]))
>@@ -53,7 +66,7 @@ CHECKOUT_ROOT = os.path.abspath(os.path.join(SCRIPT_DIR, os.pardir, os.pardir))
> sys.path.append(os.path.join(CHECKOUT_ROOT, 'build'))
> import find_depot_tools
> 
>-THIRD_PARTY_LIB_REGEX = r'^.*/third_party/([\w+]+).*$'
>+THIRD_PARTY_LIB_REGEX = r'^.*/third_party/([\w\-+]+).*$'
> 
> class LicenseBuilder(object):
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/libs/generate_licenses_test.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/libs/generate_licenses_test.py
>old mode 100644
>new mode 100755
>index 9ac541adf95..c28a226af24
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/libs/generate_licenses_test.py
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/libs/generate_licenses_test.py
>@@ -16,8 +16,8 @@ SRC = os.path.abspath(os.path.join(
>                       os.path.dirname((__file__)), os.pardir, os.pardir))
> sys.path.append(os.path.join(SRC, 'third_party', 'pymock'))
> 
>-import mock
> import unittest
>+import mock
> 
> from generate_licenses import LicenseBuilder
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/mb/gn_isolate_map.pyl b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/mb/gn_isolate_map.pyl
>index 7240efa7ffc..5832dd1dd4b 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/mb/gn_isolate_map.pyl
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/mb/gn_isolate_map.pyl
>@@ -19,12 +19,12 @@
>     "label": "//:All",
>     "type": "additional_compile_target",
>   },
>-  "AppRTCMobileTest": {
>-    "label": "//examples:AppRTCMobileTest",
>+  "AppRTCMobile_test_apk": {
>+    "label": "//examples:AppRTCMobile_test_apk",
>     "type": "additional_compile_target",
>   },
>-  "AppRTCMobileTestStubbedVideoIO": {
>-    "label": "//examples:AppRTCMobileTestStubbedVideoIO",
>+  "AppRTCMobile_stubbed_video_io_test_apk": {
>+    "label": "//examples:AppRTCMobile_stubbed_video_io_test_apk",
>     "type": "additional_compile_target",
>   },
>   "android_junit_tests": {
>@@ -55,8 +55,8 @@
>       "isac_speech_and_misc_wb.pcm",
>     ]
>   },
>-  "libjingle_peerconnection_android_unittest": {
>-    "label": "//sdk/android:libjingle_peerconnection_android_unittest",
>+  "android_instrumentation_test_apk": {
>+    "label": "//sdk/android:android_instrumentation_test_apk",
>     "type": "additional_compile_target",
>   },
>   "low_bandwidth_audio_test": {
>@@ -67,7 +67,7 @@
>     ],
>   },
>   "low_bandwidth_audio_perf_test": {
>-    "label": "//audio:low_bandwidth_audio_test",
>+    "label": "//audio:low_bandwidth_audio_perf_test",
>     "type": "script",
>     "script": "//audio/test/low_bandwidth_audio_test.py",
>     "args": [
>@@ -118,13 +118,20 @@
>     "label": "//rtc_tools:tools_unittests",
>     "type": "console_test_launcher",
>   },
>+  "video_capture_tests": {
>+    "label": "//modules/video_capture:video_capture_tests",
>+    "type": "console_test_launcher",
>+    "use_webcam": True,
>+  },
>   "video_engine_tests": {
>     "label": "//:video_engine_tests",
>     "type": "console_test_launcher",
>   },
>-  "voice_engine_unittests": {
>-    "label": "//voice_engine:voice_engine_unittests",
>-    "type": "console_test_launcher",
>+  "video_quality_loopback_test": {
>+    "label": "//examples:AppRTCMobile_stubbed_video_io_test_apk",
>+    "type": "script",
>+    "script": "//examples/androidtests/video_quality_loopback_test.py",
>+    "args": ["."],
>   },
>   "webrtc_nonparallel_tests": {
>     "label": "//:webrtc_nonparallel_tests",
>@@ -133,6 +140,6 @@
>   "webrtc_perf_tests": {
>     "label": "//:webrtc_perf_tests",
>     "timeout": 3600,  # 1 hour
>-    "type": "non_parallel_console_test_launcher",
>+    "type": "raw",
>   },
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/mb/mb b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/mb/mb
>old mode 100644
>new mode 100755
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/mb/mb.bat b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/mb/mb.bat
>old mode 100644
>new mode 100755
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/mb/mb.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/mb/mb.py
>old mode 100644
>new mode 100755
>index ca021d356e0..7f196bb7338
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/mb/mb.py
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/mb/mb.py
>@@ -7,9 +7,9 @@
> # in the file PATENTS.  All contributing project authors may
> # be found in the AUTHORS file in the root of the source tree.
> 
>-"""MB - the Meta-Build wrapper around GYP and GN
>+"""MB - the Meta-Build wrapper around GN.
> 
>-MB is a wrapper script for GYP and GN that can be used to generate build files
>+MB is a wrapper script for GN that can be used to generate build files
> for sets of canned configurations and analyze them.
> """
> 
>@@ -36,7 +36,6 @@ SCRIPT_DIR = os.path.dirname(os.path.realpath(__file__))
> SRC_DIR = os.path.dirname(os.path.dirname(SCRIPT_DIR))
> sys.path = [os.path.join(SRC_DIR, 'build')] + sys.path
> 
>-import find_depot_tools
> import gn_helpers
> 
> 
>@@ -97,16 +96,10 @@ class MetaBuildWrapper(object):
>                              '(default is %(default)s)')
>       subp.add_argument('-g', '--goma-dir',
>                         help='path to goma directory')
>-      subp.add_argument('--gyp-script', metavar='PATH',
>-                        default=self.PathJoin('build', 'gyp_chromium'),
>-                        help='path to gyp script relative to project root '
>-                             '(default is %(default)s)')
>       subp.add_argument('--android-version-code',
>-                        help='Sets GN arg android_default_version_code and '
>-                             'GYP_DEFINE app_manifest_version_code')
>+                        help='Sets GN arg android_default_version_code')
>       subp.add_argument('--android-version-name',
>-                        help='Sets GN arg android_default_version_name and '
>-                             'GYP_DEFINE app_manifest_version_name')
>+                        help='Sets GN arg android_default_version_name')
>       subp.add_argument('-n', '--dryrun', action='store_true',
>                         help='Do a dry run (i.e., do nothing, just print '
>                              'the commands that will run)')
>@@ -187,7 +180,6 @@ class MetaBuildWrapper(object):
>         '    --test-launcher-retry-limit=0'
>         '\n'
>     )
>-
>     AddCommonOptions(subp)
>     subp.add_argument('-j', '--jobs', dest='jobs', type=int,
>                       help='Number of jobs to pass to ninja')
>@@ -199,6 +191,11 @@ class MetaBuildWrapper(object):
>                             ' This can be either a regular path or a '
>                             'GN-style source-relative path like '
>                             '//out/Default.'))
>+    subp.add_argument('-s', '--swarmed', action='store_true',
>+                      help='Run under swarming')
>+    subp.add_argument('-d', '--dimension', default=[], action='append', nargs=2,
>+                      dest='dimensions', metavar='FOO bar',
>+                      help='dimension to filter on')
>     subp.add_argument('target', nargs=1,
>                       help='ninja target to build and run')
>     subp.add_argument('extra_args', nargs='*',
>@@ -214,26 +211,6 @@ class MetaBuildWrapper(object):
>                       help='path to config file (default is %(default)s)')
>     subp.set_defaults(func=self.CmdValidate)
> 
>-    subp = subps.add_parser('audit',
>-                            help='Audit the config file to track progress')
>-    subp.add_argument('-f', '--config-file', metavar='PATH',
>-                      default=self.default_config,
>-                      help='path to config file (default is %(default)s)')
>-    subp.add_argument('-i', '--internal', action='store_true',
>-                      help='check internal masters also')
>-    subp.add_argument('-m', '--master', action='append',
>-                      help='master to audit (default is all non-internal '
>-                           'masters in file)')
>-    subp.add_argument('-u', '--url-template', action='store',
>-                      default='https://build.chromium.org/p/'
>-                              '{master}/json/builders',
>-                      help='URL scheme for JSON APIs to buildbot '
>-                           '(default: %(default)s) ')
>-    subp.add_argument('-c', '--check-compile', action='store_true',
>-                      help='check whether tbd and master-only bots actually'
>-                           ' do compiles')
>-    subp.set_defaults(func=self.CmdAudit)
>-
>     subp = subps.add_parser('help',
>                             help='Get help on a subcommand.')
>     subp.add_argument(nargs='?', action='store', dest='subcommand',
>@@ -261,11 +238,7 @@ class MetaBuildWrapper(object):
> 
>   def CmdAnalyze(self):
>     vals = self.Lookup()
>-    self.ClobberIfNeeded(vals)
>-    if vals['type'] == 'gn':
>-      return self.RunGNAnalyze(vals)
>-    else:
>-      return self.RunGYPAnalyze(vals)
>+    return self.RunGNAnalyze(vals)
> 
>   def CmdExport(self):
>     self.ReadConfigFile()
>@@ -297,11 +270,7 @@ class MetaBuildWrapper(object):
> 
>   def CmdGen(self):
>     vals = self.Lookup()
>-    self.ClobberIfNeeded(vals)
>-    if vals['type'] == 'gn':
>-      return self.RunGNGen(vals)
>-    else:
>-      return self.RunGYPGen(vals)
>+    return self.RunGNGen(vals)
> 
>   def CmdHelp(self):
>     if self.args.subcommand:
>@@ -313,21 +282,14 @@ class MetaBuildWrapper(object):
>     vals = self.GetConfig()
>     if not vals:
>       return 1
>-
>-    if vals['type'] == 'gn':
>-      return self.RunGNIsolate(vals)
>-    else:
>-      return self.Build('%s_run' % self.args.target[0])
>+    return self.RunGNIsolate(vals)
> 
>   def CmdLookup(self):
>     vals = self.Lookup()
>-    if vals['type'] == 'gn':
>-      cmd = self.GNCmd('gen', '_path_')
>-      gn_args = self.GNArgs(vals)
>-      self.Print('\nWriting """\\\n%s""" to _path_/args.gn.\n' % gn_args)
>-      env = None
>-    else:
>-      cmd, env = self.GYPCmd('_path_', vals)
>+    cmd = self.GNCmd('gen', '_path_')
>+    gn_args = self.GNArgs(vals)
>+    self.Print('\nWriting """\\\n%s""" to _path_/args.gn.\n' % gn_args)
>+    env = None
> 
>     self.PrintCmd(cmd, env)
>     return 0
>@@ -340,31 +302,67 @@ class MetaBuildWrapper(object):
>     build_dir = self.args.path[0]
>     target = self.args.target[0]
> 
>-    if vals['type'] == 'gn':
>-      if self.args.build:
>-        ret = self.Build(target)
>-        if ret:
>-          return ret
>-      ret = self.RunGNIsolate(vals)
>+    if self.args.build:
>+      ret = self.Build(target)
>       if ret:
>         return ret
>+    ret = self.RunGNIsolate(vals)
>+    if ret:
>+      return ret
>+
>+    if self.args.swarmed:
>+      return self._RunUnderSwarming(build_dir, target)
>     else:
>-      ret = self.Build('%s_run' % target)
>-      if ret:
>-        return ret
>+      return self._RunLocallyIsolated(build_dir, target)
>+
>+  def _RunUnderSwarming(self, build_dir, target):
>+    # TODO(dpranke): Look up the information for the target in
>+    # the //testing/buildbot.json file, if possible, so that we
>+    # can determine the isolate target, command line, and additional
>+    # swarming parameters, if possible.
>+    #
>+    # TODO(dpranke): Also, add support for sharding and merging results.
>+    dimensions = []
>+    for k, v in self.args.dimensions:
>+      dimensions += ['-d', k, v]
> 
>     cmd = [
>         self.executable,
>         self.PathJoin('tools', 'swarming_client', 'isolate.py'),
>-        'run',
>+        'archive',
>         '-s',
>         self.ToSrcRelPath('%s/%s.isolated' % (build_dir, target)),
>-    ]
>-    if self.args.extra_args:
>-        cmd += ['--'] + self.args.extra_args
>+        '-I', 'isolateserver.appspot.com',
>+      ]
>+    ret, out, _ = self.Run(cmd, force_verbose=False)
>+    if ret:
>+      return ret
> 
>-    ret, _, _ = self.Run(cmd, force_verbose=False, buffer_output=False)
>+    isolated_hash = out.splitlines()[0].split()[0]
>+    cmd = [
>+        self.executable,
>+        self.PathJoin('tools', 'swarming_client', 'swarming.py'),
>+          'run',
>+          '-s', isolated_hash,
>+          '-I', 'isolateserver.appspot.com',
>+          '-S', 'chromium-swarm.appspot.com',
>+      ] + dimensions
>+    if self.args.extra_args:
>+      cmd += ['--'] + self.args.extra_args
>+    ret, _, _ = self.Run(cmd, force_verbose=True, buffer_output=False)
>+    return ret
> 
>+  def _RunLocallyIsolated(self, build_dir, target):
>+    cmd = [
>+        self.executable,
>+        self.PathJoin('tools', 'swarming_client', 'isolate.py'),
>+        'run',
>+        '-s',
>+        self.ToSrcRelPath('%s/%s.isolated' % (build_dir, target)),
>+      ]
>+    if self.args.extra_args:
>+      cmd += ['--'] + self.args.extra_args
>+    ret, _, _ = self.Run(cmd, force_verbose=True, buffer_output=False)
>     return ret
> 
>   def CmdValidate(self, print_ok=True):
>@@ -428,154 +426,26 @@ class MetaBuildWrapper(object):
>       self.Print('mb config file %s looks ok.' % self.args.config_file)
>     return 0
> 
>-  def CmdAudit(self):
>-    """Track the progress of the GYP->GN migration on the bots."""
>-
>-    # First, make sure the config file is okay, but don't print anything
>-    # if it is (it will throw an error if it isn't).
>-    self.CmdValidate(print_ok=False)
>-
>-    stats = OrderedDict()
>-    STAT_MASTER_ONLY = 'Master only'
>-    STAT_CONFIG_ONLY = 'Config only'
>-    STAT_TBD = 'Still TBD'
>-    STAT_GYP = 'Still GYP'
>-    STAT_DONE = 'Done (on GN)'
>-    stats[STAT_MASTER_ONLY] = 0
>-    stats[STAT_CONFIG_ONLY] = 0
>-    stats[STAT_TBD] = 0
>-    stats[STAT_GYP] = 0
>-    stats[STAT_DONE] = 0
>-
>-    def PrintBuilders(heading, builders, notes):
>-      stats.setdefault(heading, 0)
>-      stats[heading] += len(builders)
>-      if builders:
>-        self.Print('  %s:' % heading)
>-        for builder in sorted(builders):
>-          self.Print('    %s%s' % (builder, notes[builder]))
>-
>-    self.ReadConfigFile()
>-
>-    masters = self.args.master or self.masters
>-    for master in sorted(masters):
>-      url = self.args.url_template.replace('{master}', master)
>-
>-      self.Print('Auditing %s' % master)
>-
>-      MASTERS_TO_SKIP = (
>-        'client.skia',
>-        'client.v8.fyi',
>-        'tryserver.v8',
>-      )
>-      if master in MASTERS_TO_SKIP:
>-        # Skip these bots because converting them is the responsibility of
>-        # those teams and out of scope for the Chromium migration to GN.
>-        self.Print('  Skipped (out of scope)')
>-        self.Print('')
>-        continue
>-
>-      INTERNAL_MASTERS = ('official.desktop', 'official.desktop.continuous',
>-                          'internal.client.kitchensync')
>-      if master in INTERNAL_MASTERS and not self.args.internal:
>-        # Skip these because the servers aren't accessible by default ...
>-        self.Print('  Skipped (internal)')
>-        self.Print('')
>-        continue
>-
>-      try:
>-        # Fetch the /builders contents from the buildbot master. The
>-        # keys of the dict are the builder names themselves.
>-        json_contents = self.Fetch(url)
>-        d = json.loads(json_contents)
>-      except Exception as e:
>-        self.Print(str(e))
>-        return 1
>-
>-      config_builders = set(self.masters[master])
>-      master_builders = set(d.keys())
>-      both = master_builders & config_builders
>-      master_only = master_builders - config_builders
>-      config_only = config_builders - master_builders
>-      tbd = set()
>-      gyp = set()
>-      done = set()
>-      notes = {builder: '' for builder in config_builders | master_builders}
>-
>-      for builder in both:
>-        config = self.masters[master][builder]
>-        if config == 'tbd':
>-          tbd.add(builder)
>-        elif isinstance(config, dict):
>-          vals = self.FlattenConfig(config.values()[0])
>-          if vals['type'] == 'gyp':
>-            gyp.add(builder)
>-          else:
>-            done.add(builder)
>-        elif config.startswith('//'):
>-          done.add(builder)
>-        else:
>-          vals = self.FlattenConfig(config)
>-          if vals['type'] == 'gyp':
>-            gyp.add(builder)
>-          else:
>-            done.add(builder)
>-
>-      if self.args.check_compile and (tbd or master_only):
>-        either = tbd | master_only
>-        for builder in either:
>-          notes[builder] = ' (' + self.CheckCompile(master, builder) +')'
>-
>-      if master_only or config_only or tbd or gyp:
>-        PrintBuilders(STAT_MASTER_ONLY, master_only, notes)
>-        PrintBuilders(STAT_CONFIG_ONLY, config_only, notes)
>-        PrintBuilders(STAT_TBD, tbd, notes)
>-        PrintBuilders(STAT_GYP, gyp, notes)
>-      else:
>-        self.Print('  All GN!')
>-
>-      stats[STAT_DONE] += len(done)
>-
>-      self.Print('')
>-
>-    fmt = '{:<27} {:>4}'
>-    self.Print(fmt.format('Totals', str(sum(int(v) for v in stats.values()))))
>-    self.Print(fmt.format('-' * 27, '----'))
>-    for stat, count in stats.items():
>-      self.Print(fmt.format(stat, str(count)))
>-
>-    return 0
>-
>   def GetConfig(self):
>     build_dir = self.args.path[0]
> 
>     vals = self.DefaultVals()
>     if self.args.builder or self.args.master or self.args.config:
>       vals = self.Lookup()
>-      if vals['type'] == 'gn':
>-        # Re-run gn gen in order to ensure the config is consistent with the
>-        # build dir.
>-        self.RunGNGen(vals)
>+      # Re-run gn gen in order to ensure the config is consistent with the
>+      # build dir.
>+      self.RunGNGen(vals)
>       return vals
> 
>-    mb_type_path = self.PathJoin(self.ToAbsPath(build_dir), 'mb_type')
>-    if not self.Exists(mb_type_path):
>-      toolchain_path = self.PathJoin(self.ToAbsPath(build_dir),
>-                                     'toolchain.ninja')
>-      if not self.Exists(toolchain_path):
>-        self.Print('Must either specify a path to an existing GN build dir '
>-                   'or pass in a -m/-b pair or a -c flag to specify the '
>-                   'configuration')
>-        return {}
>-      else:
>-        mb_type = 'gn'
>-    else:
>-      mb_type = self.ReadFile(mb_type_path).strip()
>-
>-    if mb_type == 'gn':
>-      vals['gn_args'] = self.GNArgsFromDir(build_dir)
>-    vals['type'] = mb_type
>+    toolchain_path = self.PathJoin(self.ToAbsPath(build_dir),
>+                                   'toolchain.ninja')
>+    if not self.Exists(toolchain_path):
>+      self.Print('Must either specify a path to an existing GN build dir '
>+                 'or pass in a -m/-b pair or a -c flag to specify the '
>+                 'configuration')
>+      return {}
> 
>+    vals['gn_args'] = self.GNArgsFromDir(build_dir)
>     return vals
> 
>   def GNArgsFromDir(self, build_dir):
>@@ -607,37 +477,21 @@ class MetaBuildWrapper(object):
>           raise MBErr('Config "%s" not found in %s' %
>                       (config, self.args.config_file))
>         vals = self.FlattenConfig(config)
>-
>-    # Do some basic sanity checking on the config so that we
>-    # don't have to do this in every caller.
>-    if 'type' not in vals:
>-        vals['type'] = 'gn'
>-    assert vals['type'] in ('gn', 'gyp'), (
>-        'Unknown meta-build type "%s"' % vals['gn_args'])
>-
>     return vals
> 
>   def ReadIOSBotConfig(self):
>     if not self.args.master or not self.args.builder:
>       return {}
>-    path = self.PathJoin(self.src_dir, 'tools_webrtc', 'ios',
>-                         self.args.master,
>+    path = self.PathJoin(self.src_dir, 'tools_webrtc', 'ios', self.args.master,
>                          self.args.builder.replace(' ', '_') + '.json')
>     if not self.Exists(path):
>       return {}
> 
>     contents = json.loads(self.ReadFile(path))
>-    gyp_vals = contents.get('GYP_DEFINES', {})
>-    if isinstance(gyp_vals, dict):
>-      gyp_defines = ' '.join('%s=%s' % (k, v) for k, v in gyp_vals.items())
>-    else:
>-      gyp_defines = ' '.join(gyp_vals)
>     gn_args = ' '.join(contents.get('gn_args', []))
> 
>     vals = self.DefaultVals()
>     vals['gn_args'] = gn_args
>-    vals['gyp_defines'] = gyp_defines
>-    vals['type'] = contents.get('mb_type', 'gn')
>     return vals
> 
>   def ReadConfigFile(self):
>@@ -655,14 +509,14 @@ class MetaBuildWrapper(object):
>     self.mixins = contents['mixins']
> 
>   def ReadIsolateMap(self):
>-    if not self.Exists(self.args.isolate_map_file):
>-      raise MBErr('isolate map file not found at %s' %
>-                  self.args.isolate_map_file)
>+    isolate_map = self.args.isolate_map_file
>+    if not self.Exists(isolate_map):
>+      raise MBErr('isolate map file not found at %s' % isolate_map)
>     try:
>-      return ast.literal_eval(self.ReadFile(self.args.isolate_map_file))
>+      return ast.literal_eval(self.ReadFile(isolate_map))
>     except SyntaxError as e:
>-      raise MBErr('Failed to parse isolate map file "%s": %s' %
>-                  (self.args.isolate_map_file, e))
>+      raise MBErr(
>+          'Failed to parse isolate map file "%s": %s' % (isolate_map, e))
> 
>   def ConfigFromArgs(self):
>     if self.args.config:
>@@ -713,9 +567,6 @@ class MetaBuildWrapper(object):
>       'args_file': '',
>       'cros_passthrough': False,
>       'gn_args': '',
>-      'gyp_defines': '',
>-      'gyp_crosscompile': False,
>-      'type': 'gn',
>     }
> 
>   def FlattenMixins(self, mixins, vals, visited):
>@@ -734,50 +585,11 @@ class MetaBuildWrapper(object):
>           vals['gn_args'] += ' ' + mixin_vals['gn_args']
>         else:
>           vals['gn_args'] = mixin_vals['gn_args']
>-      if 'gyp_crosscompile' in mixin_vals:
>-        vals['gyp_crosscompile'] = mixin_vals['gyp_crosscompile']
>-      if 'gyp_defines' in mixin_vals:
>-        if vals['gyp_defines']:
>-          vals['gyp_defines'] += ' ' + mixin_vals['gyp_defines']
>-        else:
>-          vals['gyp_defines'] = mixin_vals['gyp_defines']
>-      if 'type' in mixin_vals:
>-        vals['type'] = mixin_vals['type']
> 
>       if 'mixins' in mixin_vals:
>         self.FlattenMixins(mixin_vals['mixins'], vals, visited)
>     return vals
> 
>-  def ClobberIfNeeded(self, vals):
>-    path = self.args.path[0]
>-    build_dir = self.ToAbsPath(path)
>-    mb_type_path = self.PathJoin(build_dir, 'mb_type')
>-    needs_clobber = False
>-    new_mb_type = vals['type']
>-    if self.Exists(build_dir):
>-      if self.Exists(mb_type_path):
>-        old_mb_type = self.ReadFile(mb_type_path)
>-        if old_mb_type != new_mb_type:
>-          self.Print("Build type mismatch: was %s, will be %s, clobbering %s" %
>-                     (old_mb_type, new_mb_type, path))
>-          needs_clobber = True
>-      else:
>-        # There is no 'mb_type' file in the build directory, so this probably
>-        # means that the prior build(s) were not done through mb, and we
>-        # have no idea if this was a GYP build or a GN build. Clobber it
>-        # to be safe.
>-        self.Print("%s/mb_type missing, clobbering to be safe" % path)
>-        needs_clobber = True
>-
>-    if self.args.dryrun:
>-      return
>-
>-    if needs_clobber:
>-      self.RemoveDirectory(build_dir)
>-
>-    self.MaybeMakeDirectory(build_dir)
>-    self.WriteFile(mb_type_path, new_mb_type)
>-
>   def RunGNGen(self, vals):
>     build_dir = self.args.path[0]
> 
>@@ -956,7 +768,7 @@ class MetaBuildWrapper(object):
>     return err, labels
> 
>   def GNCmd(self, subcommand, path, *args):
>-    if self.platform == 'linux2':
>+    if self.platform.startswith('linux'):
>       subdir, exe = 'linux64', 'gn'
>     elif self.platform == 'darwin':
>       subdir, exe = 'mac', 'gn'
>@@ -999,67 +811,40 @@ class MetaBuildWrapper(object):
>       gn_args = ('import("%s")\n' % vals['args_file']) + gn_args
>     return gn_args
> 
>-  def RunGYPGen(self, vals):
>-    path = self.args.path[0]
>-
>-    output_dir = self.ParseGYPConfigPath(path)
>-    cmd, env = self.GYPCmd(output_dir, vals)
>-    ret, _, _ = self.Run(cmd, env=env)
>-    return ret
>-
>-  def RunGYPAnalyze(self, vals):
>-    output_dir = self.ParseGYPConfigPath(self.args.path[0])
>-    if self.args.verbose:
>-      inp = self.ReadInputJSON(['files', 'test_targets',
>-                                'additional_compile_targets'])
>-      self.Print()
>-      self.Print('analyze input:')
>-      self.PrintJSON(inp)
>-      self.Print()
>-
>-    cmd, env = self.GYPCmd(output_dir, vals)
>-    cmd.extend(['-f', 'analyzer',
>-                '-G', 'config_path=%s' % self.args.input_path[0],
>-                '-G', 'analyzer_output_path=%s' % self.args.output_path[0]])
>-    ret, _, _ = self.Run(cmd, env=env)
>-    if not ret and self.args.verbose:
>-      outp = json.loads(self.ReadFile(self.args.output_path[0]))
>-      self.Print()
>-      self.Print('analyze output:')
>-      self.PrintJSON(outp)
>-      self.Print()
>-
>-    return ret
>-
>   def GetIsolateCommand(self, target, vals):
>     isolate_map = self.ReadIsolateMap()
>     test_type = isolate_map[target]['type']
> 
>-    android = 'target_os="android"' in vals['gn_args']
>-    is_linux = self.platform == 'linux2' and not android
>+    is_android = 'target_os="android"' in vals['gn_args']
>+    is_linux = self.platform.startswith('linux') and not is_android
> 
>     if test_type == 'nontest':
>       self.WriteFailureAndRaise('We should not be isolating %s.' % target,
>                                 output_path=None)
>     if test_type not in ('console_test_launcher', 'windowed_test_launcher',
>-                         'non_parallel_console_test_launcher',
>+                         'non_parallel_console_test_launcher', 'raw',
>                          'additional_compile_target', 'junit_test', 'script'):
>       self.WriteFailureAndRaise('No command line for %s found (test type %s).'
>                                 % (target, test_type), output_path=None)
> 
>     cmdline = []
>-    extra_files = []
>+    extra_files = [
>+      '../../.vpython',
>+      '../../testing/test_env.py',
>+    ]
> 
>-    if android:
>+    if test_type == 'script':
>+      cmdline = ['../../' + self.ToSrcRelPath(isolate_map[target]['script'])]
>+    elif is_android:
>       cmdline = ['../../build/android/test_wrapper/logdog_wrapper.py',
>                  '--target', target,
>                  '--logdog-bin-cmd', '../../bin/logdog_butler',
>                  '--logcat-output-file', '${ISOLATED_OUTDIR}/logcats',
>                  '--store-tombstones']
>-    elif test_type == "script":
>-      cmdline = ['../../' + self.ToSrcRelPath(isolate_map[target]['script'])]
>     else:
>-      extra_files = ['../../testing/test_env.py']
>+      if isolate_map[target].get('use_webcam', False):
>+        cmdline.append('../../tools_webrtc/ensure_webcam_is_running.py')
>+        extra_files.append('../../tools_webrtc/ensure_webcam_is_running.py')
> 
>       # This needs to mirror the settings in //build/config/ui.gni:
>       # use_x11 = is_linux && !use_ozone.
>@@ -1067,25 +852,12 @@ class MetaBuildWrapper(object):
> 
>       xvfb = use_x11 and test_type == 'windowed_test_launcher'
>       if xvfb:
>-        extra_files += [
>-            '../../testing/xvfb.py',
>-        ]
>-
>-      # Memcheck is only supported for linux. Ignore in other platforms.
>-      memcheck = is_linux and 'rtc_use_memcheck=true' in vals['gn_args']
>-      memcheck_cmdline = [
>-          'bash',
>-          '../../tools_webrtc/valgrind/webrtc_tests.sh',
>-          '--tool',
>-          'memcheck',
>-          '--target',
>-          'Release',
>-          '--build-dir',
>-          '..',
>-          '--test',
>-      ]
>+        cmdline.append('../../testing/xvfb.py')
>+        extra_files.append('../../testing/xvfb.py')
>+      else:
>+        cmdline.append('../../testing/test_env.py')
> 
>-      if not memcheck:
>+      if test_type != 'raw':
>         extra_files += [
>             '../../third_party/gtest-parallel/gtest-parallel',
>             '../../third_party/gtest-parallel/gtest_parallel.py',
>@@ -1094,7 +866,7 @@ class MetaBuildWrapper(object):
>         sep = '\\' if self.platform == 'win32' else '/'
>         output_dir = '${ISOLATED_OUTDIR}' + sep + 'test_logs'
>         timeout = isolate_map[target].get('timeout', 900)
>-        gtest_parallel_wrapper = [
>+        cmdline += [
>             '../../tools_webrtc/gtest-parallel-wrapper.py',
>             '--output_dir=%s' % output_dir,
>             '--gtest_color=no',
>@@ -1104,27 +876,23 @@ class MetaBuildWrapper(object):
>             '--timeout=%s' % timeout,
>             '--retry_failed=3',
>         ]
>-
>-      asan = 'is_asan=true' in vals['gn_args']
>-      lsan = 'is_lsan=true' in vals['gn_args']
>-      msan = 'is_msan=true' in vals['gn_args']
>-      tsan = 'is_tsan=true' in vals['gn_args']
>+        if test_type == 'non_parallel_console_test_launcher':
>+          # Still use the gtest-parallel-wrapper.py script since we need it to
>+          # run tests on swarming, but don't execute tests in parallel.
>+          cmdline.append('--workers=1')
> 
>       executable_prefix = '.\\' if self.platform == 'win32' else './'
>       executable_suffix = '.exe' if self.platform == 'win32' else ''
>       executable = executable_prefix + target + executable_suffix
> 
>-      cmdline = (['../../testing/xvfb.py'] if xvfb else
>-                 ['../../testing/test_env.py'])
>-      cmdline += memcheck_cmdline if memcheck else gtest_parallel_wrapper
>       cmdline.append(executable)
>-      if test_type == 'non_parallel_console_test_launcher' and not memcheck:
>-        # Still use the gtest-parallel-wrapper.py script since we need it to
>-        # run tests on swarming, but don't execute tests in parallel.
>-        cmdline.append('--workers=1')
>+
>+      asan = 'is_asan=true' in vals['gn_args']
>+      lsan = 'is_lsan=true' in vals['gn_args']
>+      msan = 'is_msan=true' in vals['gn_args']
>+      tsan = 'is_tsan=true' in vals['gn_args']
> 
>       cmdline.extend([
>-          '--',
>           '--asan=%d' % asan,
>           '--lsan=%d' % lsan,
>           '--msan=%d' % msan,
>@@ -1146,86 +914,6 @@ class MetaBuildWrapper(object):
>       return path[2:].replace('/', self.sep)
>     return self.RelPath(path, self.src_dir)
> 
>-  def ParseGYPConfigPath(self, path):
>-    rpath = self.ToSrcRelPath(path)
>-    output_dir, _, _ = rpath.rpartition(self.sep)
>-    return output_dir
>-
>-  def GYPCmd(self, output_dir, vals):
>-    if vals['cros_passthrough']:
>-      if not 'GYP_DEFINES' in os.environ:
>-        raise MBErr('MB is expecting GYP_DEFINES to be in the environment')
>-      gyp_defines = os.environ['GYP_DEFINES']
>-      if not 'chromeos=1' in gyp_defines:
>-        raise MBErr('GYP_DEFINES is missing chromeos=1: (GYP_DEFINES=%s)' %
>-                    gyp_defines)
>-    else:
>-      gyp_defines = vals['gyp_defines']
>-
>-    goma_dir = self.args.goma_dir
>-
>-    # GYP uses shlex.split() to split the gyp defines into separate arguments,
>-    # so we can support backslashes and and spaces in arguments by quoting
>-    # them, even on Windows, where this normally wouldn't work.
>-    if goma_dir and ('\\' in goma_dir or ' ' in goma_dir):
>-      goma_dir = "'%s'" % goma_dir
>-
>-    if goma_dir:
>-      gyp_defines += ' gomadir=%s' % goma_dir
>-
>-    android_version_code = self.args.android_version_code
>-    if android_version_code:
>-      gyp_defines += ' app_manifest_version_code=%s' % android_version_code
>-
>-    android_version_name = self.args.android_version_name
>-    if android_version_name:
>-      gyp_defines += ' app_manifest_version_name=%s' % android_version_name
>-
>-    cmd = [
>-        self.executable,
>-        self.args.gyp_script,
>-        '-G',
>-        'output_dir=' + output_dir,
>-    ]
>-
>-    # Ensure that we have an environment that only contains
>-    # the exact values of the GYP variables we need.
>-    env = os.environ.copy()
>-
>-    # This is a terrible hack to work around the fact that
>-    # //tools/clang/scripts/update.py is invoked by GYP and GN but
>-    # currently relies on an environment variable to figure out
>-    # what revision to embed in the command line #defines.
>-    # For GN, we've made this work via a gn arg that will cause update.py
>-    # to get an additional command line arg, but getting that to work
>-    # via GYP_DEFINES has proven difficult, so we rewrite the GYP_DEFINES
>-    # to get rid of the arg and add the old var in, instead.
>-    # See crbug.com/582737 for more on this. This can hopefully all
>-    # go away with GYP.
>-    m = re.search('llvm_force_head_revision=1\s*', gyp_defines)
>-    if m:
>-      env['LLVM_FORCE_HEAD_REVISION'] = '1'
>-      gyp_defines = gyp_defines.replace(m.group(0), '')
>-
>-    # This is another terrible hack to work around the fact that
>-    # GYP sets the link concurrency to use via the GYP_LINK_CONCURRENCY
>-    # environment variable, and not via a proper GYP_DEFINE. See
>-    # crbug.com/611491 for more on this.
>-    m = re.search('gyp_link_concurrency=(\d+)(\s*)', gyp_defines)
>-    if m:
>-      env['GYP_LINK_CONCURRENCY'] = m.group(1)
>-      gyp_defines = gyp_defines.replace(m.group(0), '')
>-
>-    env['GYP_GENERATORS'] = 'ninja'
>-    if 'GYP_CHROMIUM_NO_ACTION' in env:
>-      del env['GYP_CHROMIUM_NO_ACTION']
>-    if 'GYP_CROSSCOMPILE' in env:
>-      del env['GYP_CROSSCOMPILE']
>-    env['GYP_DEFINES'] = gyp_defines
>-    if vals['gyp_crosscompile']:
>-      env['GYP_CROSSCOMPILE'] = '1'
>-    return cmd, env
>-
>   def RunGNAnalyze(self, vals):
>     # Analyze runs before 'gn gen' now, so we need to run gn gen
>     # in order to ensure that we have a build directory.
>@@ -1392,9 +1080,6 @@ class MetaBuildWrapper(object):
>       if env and var in env:
>         self.Print('%s%s=%s' % (env_prefix, var, env_quoter(env[var])))
> 
>-    print_env('GYP_CROSSCOMPILE')
>-    print_env('GYP_DEFINES')
>-    print_env('GYP_LINK_CONCURRENCY')
>     print_env('LLVM_FORCE_HEAD_REVISION')
> 
>     if cmd[0] == self.executable:
>@@ -1406,11 +1091,7 @@ class MetaBuildWrapper(object):
> 
>   def Build(self, target):
>     build_dir = self.ToSrcRelPath(self.args.path[0])
>-    ninja_cmd = [
>-      os.path.join(find_depot_tools.DEPOT_TOOLS_PATH, 'ninja'),
>-      '-C',
>-      build_dir,
>-    ]
>+    ninja_cmd = ['ninja', '-C', build_dir]
>     if self.args.jobs:
>       ninja_cmd.extend(['-j', '%d' % self.args.jobs])
>     ninja_cmd.append(target)
>@@ -1535,7 +1216,6 @@ def QuoteForSet(arg):
> 
> def QuoteForCmd(arg):
>   # First, escape the arg so that CommandLineToArgvW will parse it properly.
>-  # From //tools/gyp/pylib/gyp/msvs_emulation.py:23.
>   if arg == '' or ' ' in arg or '"' in arg:
>     quote_re = re.compile(r'(\\*)"')
>     arg = '"%s"' % (quote_re.sub(lambda mo: 2 * mo.group(1) + '\\"', arg))
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/mb/mb_config.pyl b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/mb/mb_config.pyl
>index 4ee4d5cf159..dca973b071c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/mb/mb_config.pyl
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/mb/mb_config.pyl
>@@ -32,49 +32,78 @@
>       'Mac64 Debug': 'debug_bot_x64',
>       'Mac64 Release': 'release_bot_x64',
>       'Mac Asan': 'mac_asan_clang_release_bot_x64',
>-      'Mac64 Release [large tests]': 'release_bot_x64',
> 
>       # Linux
>-      'Linux32 Debug': 'debug_bot_x86',
>+      'Linux32 Debug': 'no_h264_debug_bot_x86',
>       'Linux32 Release': 'release_bot_x86',
>       'Linux32 Debug (ARM)': 'debug_bot_arm',
>       'Linux32 Release (ARM)': 'release_bot_arm',
>       'Linux64 Debug': 'debug_bot_x64',
>       'Linux64 Release': 'release_bot_x64',
>+      'Linux64 Builder': 'pure_release_bot_x64',
>       'Linux64 Debug (ARM)': 'debug_bot_arm64',
>       'Linux64 Release (ARM)': 'release_bot_arm64',
>       'Linux64 Release (GCC)': 'gcc_release_bot_x64',
>       'Linux Asan': 'asan_lsan_clang_release_bot_x64',
>-      'Linux Memcheck': 'memcheck_release_bot_x64',
>       'Linux MSan': 'msan_clang_release_bot_x64',
>       'Linux Tsan v2': 'tsan_clang_release_bot_x64',
>       'Linux UBSan': 'ubsan_clang_release_bot_x64',
>       'Linux UBSan vptr': 'ubsan_vptr_clang_release_bot_x64',
>-      'Linux64 Release [large tests]': 'release_bot_x64',
>       'Linux64 Release (Libfuzzer)': 'libfuzzer_asan_release_bot_x64',
>+      # "More configs" bots will build all the following configs in sequence.
>+      # This is using MB's "phases" feature.
>+      'Linux (more configs)': {
>+        'intelligibility_enhancer_no_include_tests':
>+          'intelligibility_enhancer_no_include_tests_x64',
>+        'bwe_test_logging':
>+          'bwe_test_logging_x64',
>+        'dummy_audio_file_devices_no_protobuf':
>+          'dummy_audio_file_devices_no_protobuf_x64',
>+        'rtti_no_sctp':
>+          'rtti_no_sctp_x64',
>+      },
> 
>       # Android
>       'Android32 (M Nexus5X)': 'android_release_bot_arm',
>       'Android32 (M Nexus5X)(dbg)': 'android_debug_static_bot_arm',
>+      'Android32 Builder arm': 'android_pure_release_bot_arm',
>       'Android64 (M Nexus5X)': 'android_release_bot_arm64',
>       'Android64 (M Nexus5X)(dbg)': 'android_debug_static_bot_arm64',
>+      'Android64 Builder arm64': 'android_pure_release_bot_arm64',
>       'Android32 Builder x86': 'android_release_bot_x86',
>       'Android32 Builder x86 (dbg)': 'android_debug_static_bot_x86',
>       'Android64 Builder x64 (dbg)': 'android_debug_static_bot_x64',
>-      'Android32 Builder MIPS (dbg)': 'android_debug_static_bot_mipsel',
>+      'Android32 (more configs)': {
>+        'intelligibility_enhancer_no_include_tests':
>+          'intelligibility_enhancer_no_include_tests_android_arm',
>+        'bwe_test_logging':
>+          'bwe_test_logging_android_arm',
>+        'dummy_audio_file_devices_no_protobuf':
>+          'dummy_audio_file_devices_no_protobuf_android_arm',
>+        'rtti_no_sctp':
>+          'rtti_no_sctp_android_arm',
>+      },
> 
>       # Windows
>-      'Win32 Debug': 'debug_bot_x86',
>-      'Win32 Release': 'release_bot_x86',
>-      'Win64 Debug': 'debug_bot_x64',
>-      'Win64 Release': 'release_bot_x64',
>+      'Win32 Debug': 'win_msvc_debug_bot_x86',
>+      'Win32 Release': 'win_msvc_release_bot_x86',
>+      'Win64 Debug': 'win_msvc_debug_bot_x64',
>+      'Win64 Release': 'win_msvc_release_bot_x64',
>       'Win32 Debug (Clang)': 'win_clang_debug_bot_x86',
>       'Win32 Release (Clang)': 'win_clang_release_bot_x86',
>       'Win64 Debug (Clang)': 'win_clang_debug_bot_x64',
>       'Win64 Release (Clang)': 'win_clang_release_bot_x64',
>       'Win32 ASan': 'win_asan_clang_release_bot_x86',
>-      'Win32 Release (MSVC)': 'win_msvc_release_bot_x86',
>-      'Win32 Release [large tests]': 'release_bot_x86',
>+      'Win (more configs)': {
>+        'intelligibility_enhancer_no_include_tests':
>+          'intelligibility_enhancer_no_include_tests_x86',
>+        'bwe_test_logging':
>+          'bwe_test_logging_x86',
>+        'dummy_audio_file_devices_no_protobuf':
>+          'dummy_audio_file_devices_no_protobuf_x86',
>+        'rtti_no_sctp':
>+          'rtti_no_sctp_x86',
>+      },
>     },
>     'client.webrtc.branches': {
>       # iOS
>@@ -152,68 +181,92 @@
> 
>       # Mac
>       'mac_compile_dbg': 'debug_bot_x64',
>-      'mac_compile_rel': 'release_bot_x64',
>+      'mac_compile_rel': 'pure_release_bot_x64',
>       'mac_dbg': 'debug_bot_x64',
>       'mac_rel': 'release_bot_x64',
>       'mac_asan': 'mac_asan_clang_release_bot_x64',
>-      'mac_baremetal': 'release_bot_x64',
>       'mac_experimental': 'release_bot_x64',
> 
>       # Linux
>       'linux_compile_dbg': 'debug_bot_x64',
>-      'linux_compile_rel': 'release_bot_x64',
>+      'linux_compile_rel': 'pure_release_bot_x64',
>       'linux_dbg': 'debug_bot_x64',
>       'linux_rel': 'release_bot_x64',
>       'linux_arm64_dbg': 'debug_bot_arm64',
>       'linux_arm64_rel': 'release_bot_arm64',
>-      'linux32_dbg': 'debug_bot_x86',
>+      'linux32_dbg': 'no_h264_debug_bot_x86',
>       'linux32_rel': 'release_bot_x86',
>       'linux32_arm_dbg': 'debug_bot_arm',
>       'linux32_arm_rel': 'release_bot_arm',
>       'linux_asan': 'asan_lsan_clang_release_bot_x64',
>       'linux_gcc_rel': 'gcc_release_bot_x64',
>-      'linux_memcheck': 'memcheck_release_bot_x64',
>       'linux_msan': 'msan_clang_release_bot_x64',
>       'linux_tsan2': 'tsan_clang_release_bot_x64',
>       'linux_ubsan': 'ubsan_clang_release_bot_x64',
>       'linux_ubsan_vptr': 'ubsan_vptr_clang_release_bot_x64',
>-      'linux_baremetal': 'release_bot_x64',
>       'linux_experimental': 'release_bot_x64',
>       'linux_libfuzzer_rel': 'libfuzzer_asan_release_bot_x64',
>+      'linux_more_configs': {
>+        'intelligibility_enhancer_no_include_tests':
>+          'intelligibility_enhancer_no_include_tests_x64',
>+        'bwe_test_logging':
>+          'bwe_test_logging_x64',
>+        'dummy_audio_file_devices_no_protobuf':
>+          'dummy_audio_file_devices_no_protobuf_x64',
>+        'rtti_no_sctp':
>+          'rtti_no_sctp_x64',
>+      },
> 
>       # Android
>       'android_compile_dbg': 'android_debug_static_bot_arm',
>-      'android_compile_rel': 'android_release_bot_arm',
>+      'android_compile_rel': 'android_pure_release_bot_arm',
>       'android_compile_arm64_dbg': 'android_debug_static_bot_arm64',
>-      'android_compile_arm64_rel': 'android_release_bot_arm64',
>+      'android_compile_arm64_rel': 'android_pure_release_bot_arm64',
>       'android_compile_x86_dbg': 'android_debug_static_bot_x86',
>       'android_compile_x86_rel': 'android_release_bot_x86',
>       'android_compile_x64_dbg': 'android_debug_static_bot_x64',
>-      'android_compile_mips_dbg': 'android_debug_static_bot_mipsel',
>       'android_dbg': 'android_debug_static_bot_arm',
>       'android_rel': 'android_release_bot_arm',
>       'android_arm64_rel': 'android_release_bot_arm64',
>       'android_experimental': 'android_release_bot_arm',
>+      'android_more_configs': {
>+        'intelligibility_enhancer_no_include_tests':
>+          'intelligibility_enhancer_no_include_tests_android_arm',
>+        'bwe_test_logging':
>+          'bwe_test_logging_android_arm',
>+        'dummy_audio_file_devices_no_protobuf':
>+          'dummy_audio_file_devices_no_protobuf_android_arm',
>+        'rtti_no_sctp':
>+          'rtti_no_sctp_android_arm',
>+      },
> 
>       # Windows
>       'win_compile_dbg': 'debug_bot_x86',
>-      'win_compile_rel': 'release_bot_x86',
>+      'win_compile_rel': 'pure_release_bot_x86',
>       'win_compile_x64_dbg': 'debug_bot_x64',
>-      'win_compile_x64_rel': 'release_bot_x64',
>-      'win_dbg': 'debug_bot_x86',
>-      'win_rel': 'release_bot_x86',
>-      'win_x64_dbg': 'debug_bot_x64',
>-      'win_x64_rel': 'release_bot_x64',
>+      'win_compile_x64_rel': 'pure_release_bot_x64',
>+      'win_dbg': 'win_msvc_debug_bot_x86',
>+      'win_rel': 'win_msvc_release_bot_x86',
>+      'win_x64_dbg': 'win_msvc_debug_bot_x64',
>+      'win_x64_rel': 'win_msvc_release_bot_x64',
>       'win_clang_dbg': 'win_clang_debug_bot_x86',
>       'win_clang_rel': 'win_clang_release_bot_x86',
>       'win_x64_clang_dbg': 'win_clang_debug_bot_x64',
>       'win_x64_clang_rel': 'win_clang_release_bot_x64',
>       'win_asan': 'win_asan_clang_release_bot_x86',
>-      'win_msvc_rel': 'win_msvc_release_bot_x86',
>-      'win_baremetal': 'release_bot_x86',
>       'win_experimental': 'release_bot_x86',
>       'win_x64_win8': 'debug_bot_x64',
>       'win_x64_win10': 'debug_bot_x64',
>+      'win_more_configs': {
>+        'intelligibility_enhancer_no_include_tests':
>+          'intelligibility_enhancer_no_include_tests_x86',
>+        'bwe_test_logging':
>+          'bwe_test_logging_x86',
>+        'dummy_audio_file_devices_no_protobuf':
>+          'dummy_audio_file_devices_no_protobuf_x86',
>+        'rtti_no_sctp':
>+          'rtti_no_sctp_x86',
>+      },
>     }
>   },
> 
>@@ -231,43 +284,52 @@
>       'gn', 'openh264', 'debug', 'arm'
>     ],
>     'release_bot_arm': [
>-      'gn', 'openh264_release_bot', 'arm'
>+      'gn', 'openh264', 'release_bot', 'arm'
>     ],
>     'debug_bot_arm64': [
>-      'gn', 'openh264_debug_bot', 'arm64'
>+      'gn', 'openh264', 'debug_bot', 'arm64'
>     ],
>     'release_bot_arm64': [
>-      'gn', 'openh264_release_bot', 'arm64'
>+      'gn', 'openh264', 'release_bot', 'arm64'
>     ],
>     'asan_lsan_clang_release_bot_x64': [
>-      'gn', 'asan', 'lsan', 'clang', 'openh264_release_bot', 'x64'
>+      'gn', 'asan', 'lsan', 'clang', 'openh264', 'release_bot', 'x64'
>     ],
>     'msan_clang_release_bot_x64': [
>-      'gn', 'msan', 'clang', 'openh264_release_bot', 'x64'
>+      'gn', 'msan', 'clang', 'openh264', 'release_bot', 'x64'
>     ],
>     'tsan_clang_release_bot_x64': [
>-      'gn', 'tsan', 'clang', 'openh264_release_bot', 'x64'
>+      'gn', 'tsan', 'clang', 'openh264', 'release_bot', 'x64'
>     ],
>     'ubsan_clang_release_bot_x64': [
>-      'gn', 'ubsan', 'clang', 'openh264_release_bot', 'x64'
>+      'gn', 'ubsan', 'clang', 'openh264', 'release_bot', 'x64'
>     ],
>     'ubsan_vptr_clang_release_bot_x64': [
>-      'gn', 'ubsan_vptr', 'clang', 'openh264_release_bot', 'x64'
>+      'gn', 'ubsan_vptr', 'clang', 'openh264', 'release_bot', 'x64'
>     ],
>     'debug_bot_x86': [
>-      'gn', 'openh264_debug_bot', 'x86'
>+      'gn', 'openh264', 'debug_bot', 'x86'
>+    ],
>+    'no_h264_debug_bot_x86': [
>+      'gn', 'debug_bot', 'x86'
>     ],
>     'release_bot_x86': [
>-      'gn', 'openh264_release_bot', 'x86'
>+      'gn', 'openh264', 'release_bot', 'x86'
>     ],
>     'debug_bot_x64': [
>-      'gn', 'openh264_debug_bot', 'x64'
>+      'gn', 'openh264', 'debug_bot', 'x64'
>     ],
>     'codesearch_gen_linux_bot': [
>-      'gn', 'openh264_debug_bot', 'minimal_symbols'
>+      'gn', 'openh264', 'debug_bot', 'minimal_symbols'
>     ],
>     'release_bot_x64': [
>-      'gn', 'openh264_release_bot', 'x64'
>+      'gn', 'openh264', 'release_bot', 'x64'
>+    ],
>+    'pure_release_bot_x86': [
>+      'gn', 'openh264', 'pure_release_bot', 'x86'
>+    ],
>+    'pure_release_bot_x64': [
>+      'gn', 'openh264', 'pure_release_bot', 'x64'
>     ],
>     'perf_release_bot_x86': [
>       'gn', 'openh264', 'release', 'goma', 'x86'
>@@ -276,13 +338,9 @@
>       'gn', 'openh264', 'release', 'goma', 'x64'
>     ],
>     'libfuzzer_asan_release_bot_x64': [
>-      'libfuzzer', 'asan', 'optimize_for_fuzzing', 'openh264_release_bot',
>+      'libfuzzer', 'asan', 'optimize_for_fuzzing', 'openh264', 'release_bot',
>       'x64', 'no_rtc_tests'
>     ],
>-    'memcheck_release_bot_x64': [
>-      'memcheck', 'gn', 'openh264_release_bot', 'x64', 'use_default_libcxx',
>-      'no_lld'
>-    ],
> 
>     # iOS
>     # The 'ios' config is just used for auditing. iOS bots
>@@ -293,28 +351,37 @@
> 
>     # Windows
>     'win_clang_debug_bot_x86': [
>-      'gn', 'clang', 'openh264_debug_bot', 'x86',
>+      'gn', 'clang', 'openh264', 'debug_bot', 'x86',
>     ],
>     'win_clang_release_bot_x86': [
>-      'gn', 'clang', 'openh264_release_bot', 'x86',
>+      'gn', 'clang', 'openh264', 'release_bot', 'x86',
>     ],
>     'win_clang_debug_bot_x64': [
>-      'gn', 'clang', 'openh264_debug_bot', 'x64',
>+      'gn', 'clang', 'openh264', 'debug_bot', 'x64',
>     ],
>     'win_clang_release_bot_x64': [
>-      'gn', 'clang', 'openh264_release_bot', 'x64',
>+      'gn', 'clang', 'openh264', 'release_bot', 'x64',
>+    ],
>+    'win_msvc_debug_bot_x86': [
>+      'gn', 'no_clang', 'openh264', 'debug_bot', 'x86', 'no_lld',
>     ],
>     'win_msvc_release_bot_x86': [
>-      'gn', 'no_clang', 'openh264_release_bot', 'x86',
>+      'gn', 'no_clang', 'openh264', 'release_bot', 'x86', 'no_lld',
>+    ],
>+    'win_msvc_debug_bot_x64': [
>+      'gn', 'no_clang', 'openh264', 'debug_bot', 'x64', 'no_lld'
>+    ],
>+    'win_msvc_release_bot_x64': [
>+      'gn', 'no_clang', 'openh264', 'release_bot', 'x64', 'no_lld',
>     ],
>     'win_asan_clang_release_bot_x86': [
>-      'gn', 'asan', 'clang',  'full_symbols', 'openh264_release_bot', 'x86',
>+      'gn', 'asan', 'clang',  'full_symbols', 'openh264', 'release_bot', 'x86',
>       'win_fastlink',
>     ],
> 
>     # Mac
>     'mac_asan_clang_release_bot_x64': [
>-      'gn', 'asan', 'clang', 'openh264_release_bot', 'x64',
>+      'gn', 'asan', 'clang', 'openh264', 'release_bot', 'x64',
>     ],
> 
>     # Android
>@@ -324,21 +391,24 @@
>     'android_release_bot_arm': [
>       'android', 'gn', 'release_bot', 'arm'
>     ],
>+    'android_pure_release_bot_arm': [
>+      'android', 'gn', 'pure_release_bot', 'arm'
>+    ],
>     'android_debug_static_bot_x86': [
>       'android', 'gn', 'debug_static_bot', 'x86'
>     ],
>     'android_release_bot_x86': [
>       'android', 'gn', 'release_bot', 'x86'
>     ],
>-    'android_debug_static_bot_mipsel': [
>-      'android', 'gn', 'debug_static_bot', 'mipsel'
>-    ],
>     'android_debug_static_bot_arm64': [
>       'android', 'gn', 'debug_static_bot', 'arm64'
>     ],
>     'android_release_bot_arm64': [
>       'android', 'gn', 'release_bot', 'arm64'
>     ],
>+    'android_pure_release_bot_arm64': [
>+      'android', 'gn', 'pure_release_bot', 'arm64'
>+    ],
>     'android_debug_static_bot_x64': [
>       'android', 'gn', 'debug_static_bot', 'x64'
>     ],
>@@ -352,6 +422,48 @@
>       'android', 'gn', 'release', 'goma', 'arm64'
>     ],
> 
>+    # More configs
>+    'intelligibility_enhancer_no_include_tests_x64': [
>+      'debug_bot', 'x64', 'intelligibility_enhancer', 'no_include_tests'
>+    ],
>+    'bwe_test_logging_x64': [
>+      'debug_bot', 'x64', 'bwe_test_logging'
>+    ],
>+    'dummy_audio_file_devices_no_protobuf_x64': [
>+      'debug_bot', 'x64', 'dummy_audio_file_devices', 'no_protobuf'
>+    ],
>+    'rtti_no_sctp_x64': [
>+      'debug_bot', 'x64', 'rtti', 'no_sctp'
>+    ],
>+
>+    'intelligibility_enhancer_no_include_tests_x86': [
>+      'debug_bot', 'x86', 'intelligibility_enhancer', 'no_include_tests'
>+    ],
>+    'bwe_test_logging_x86': [
>+      'debug_bot', 'x86', 'bwe_test_logging'
>+    ],
>+    'dummy_audio_file_devices_no_protobuf_x86': [
>+      'debug_bot', 'x86', 'dummy_audio_file_devices', 'no_protobuf'
>+    ],
>+    'rtti_no_sctp_x86': [
>+      'debug_bot', 'x86', 'rtti', 'no_sctp'
>+    ],
>+
>+    'intelligibility_enhancer_no_include_tests_android_arm': [
>+      'android', 'debug_static_bot', 'arm',
>+      'intelligibility_enhancer', 'no_include_tests'
>+    ],
>+    'bwe_test_logging_android_arm': [
>+      'android', 'debug_static_bot', 'arm', 'bwe_test_logging'
>+    ],
>+    'dummy_audio_file_devices_no_protobuf_android_arm': [
>+      'android', 'debug_static_bot', 'arm',
>+      'dummy_audio_file_devices', 'no_protobuf'
>+    ],
>+    'rtti_no_sctp_android_arm': [
>+      'android', 'debug_static_bot', 'arm', 'rtti', 'no_sctp'
>+    ],
>+
>     # This is used for tracking purposes; any bot that uses this config
>     # should never actually run MB.
>     'none': [
>@@ -378,6 +490,7 @@
>       'gn_args': 'is_asan=true',
>     },
> 
>+    # is_clang=true by default, this is only to guard from upstream changes.
>     'clang': {
>       'gn_args': 'is_clang=true',
>     },
>@@ -429,18 +542,10 @@
>       'gn_args': 'is_lsan=true',
>     },
> 
>-    'memcheck': {
>-      'gn_args': 'rtc_use_memcheck=true',
>-    },
>-
>     'minimal_symbols': {
>       'gn_args': 'symbol_level=1 strip_absolute_paths_from_debug_symbols=true',
>     },
> 
>-    'mipsel': {
>-      'gn_args': 'target_cpu="mipsel"',
>-    },
>-
>     'msan': {
>       'gn_args': ('is_msan=true msan_track_origins=2 '
>                   'use_prebuilt_instrumented_libraries=true'),
>@@ -462,24 +567,20 @@
>       'gn_args': 'ffmpeg_branding="Chrome" rtc_use_h264=true',
>     },
> 
>-    'openh264_debug_bot': {
>-      'mixins': ['openh264', 'debug_bot'],
>-    },
>-
>-    'openh264_release_bot': {
>-      'mixins': ['openh264', 'release_bot'],
>-    },
>-
>     'optimize_for_fuzzing': {
>       'gn_args': 'optimize_for_fuzzing=true',
>     },
> 
>+    'pure_release_bot': {
>+      'mixins': ['release', 'goma'],
>+    },
>+
>     'release': {
>       'gn_args': 'is_debug=false',
>     },
> 
>     'release_bot': {
>-      'mixins': ['release', 'dcheck_always_on', 'goma'],
>+      'mixins': ['pure_release_bot', 'dcheck_always_on'],
>     },
> 
>     'tsan': {
>@@ -506,8 +607,32 @@
>       'gn_args': 'target_cpu="x86"',
>     },
> 
>-    'use_default_libcxx': {
>-      'gn_args': 'use_custom_libcxx=false',
>-    }
>+    'intelligibility_enhancer': {
>+      'gn_args': 'rtc_enable_intelligibility_enhancer=true',
>+    },
>+
>+    'no_include_tests': {
>+      'gn_args': 'rtc_include_tests=false',
>+    },
>+
>+    'bwe_test_logging': {
>+      'gn_args': 'rtc_enable_bwe_test_logging=true',
>+    },
>+
>+    'dummy_audio_file_devices': {
>+      'gn_args': 'rtc_use_dummy_audio_file_devices=true',
>+    },
>+
>+    'no_protobuf': {
>+      'gn_args': 'rtc_enable_protobuf=false',
>+    },
>+
>+    'rtti': {
>+      'gn_args': 'use_rtti=true',
>+    },
>+
>+    'no_sctp': {
>+      'gn_args': 'rtc_enable_sctp=false',
>+    },
>   },
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/mb/mb_unittest.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/mb/mb_unittest.py
>old mode 100644
>new mode 100755
>index dd475317ba4..09093c06bae
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/mb/mb_unittest.py
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/mb/mb_unittest.py
>@@ -69,8 +69,6 @@ class FakeMBW(mb.MetaBuildWrapper):
>     self.files[path] = contents
> 
>   def Call(self, cmd, env=None, buffer_output=True):
>-    if env:
>-      self.cross_compile = env.get('GYP_CROSSCOMPILE')
>     self.calls.append(cmd)
>     if self.cmds:
>       return self.cmds.pop(0)
>@@ -116,49 +114,32 @@ TEST_CONFIG = """\
>   'masters': {
>     'chromium': {},
>     'fake_master': {
>-      'fake_builder': 'gyp_rel_bot',
>-      'fake_gn_builder': 'gn_rel_bot',
>-      'fake_gyp_crosscompile_builder': 'gyp_crosscompile',
>-      'fake_gn_debug_builder': 'gn_debug_goma',
>-      'fake_gyp_builder': 'gyp_debug',
>-      'fake_gn_args_bot': '//build/args/bots/fake_master/fake_gn_args_bot.gn',
>-      'fake_memcheck_bot': 'gn_memcheck_bot',
>-      'fake_multi_phase': { 'phase_1': 'gn_phase_1', 'phase_2': 'gn_phase_2'},
>-      'fake_android_bot': 'gn_android_bot',
>+      'fake_builder': 'rel_bot',
>+      'fake_debug_builder': 'debug_goma',
>+      'fake_args_bot': '//build/args/bots/fake_master/fake_args_bot.gn',
>+      'fake_multi_phase': { 'phase_1': 'phase_1', 'phase_2': 'phase_2'},
>+      'fake_android_bot': 'android_bot',
>     },
>   },
>   'configs': {
>-    'gyp_rel_bot': ['gyp', 'rel', 'goma'],
>-    'gn_debug_goma': ['gn', 'debug', 'goma'],
>-    'gyp_debug': ['gyp', 'debug', 'fake_feature1'],
>-    'gn_rel_bot': ['gn', 'rel', 'goma'],
>-    'gyp_crosscompile': ['gyp', 'crosscompile'],
>-    'gn_phase_1': ['gn', 'phase_1'],
>-    'gn_phase_2': ['gn', 'phase_2'],
>-    'gn_memcheck_bot': ['gn', 'memcheck'],
>-    'gn_android_bot': ['gn', 'android'],
>+    'rel_bot': ['rel', 'goma', 'fake_feature1'],
>+    'debug_goma': ['debug', 'goma'],
>+    'phase_1': ['phase_1'],
>+    'phase_2': ['phase_2'],
>+    'android_bot': ['android'],
>   },
>   'mixins': {
>-    'crosscompile': {
>-      'gyp_crosscompile': True,
>-    },
>     'fake_feature1': {
>       'gn_args': 'enable_doom_melon=true',
>-      'gyp_defines': 'doom_melon=1',
>     },
>-    'gyp': {'type': 'gyp'},
>-    'gn': {'type': 'gn'},
>     'goma': {
>       'gn_args': 'use_goma=true',
>-      'gyp_defines': 'goma=1',
>     },
>     'phase_1': {
>       'gn_args': 'phase=1',
>-      'gyp_args': 'phase=1',
>     },
>     'phase_2': {
>       'gn_args': 'phase=2',
>-      'gyp_args': 'phase=2',
>     },
>     'rel': {
>       'gn_args': 'is_debug=false',
>@@ -166,9 +147,6 @@ TEST_CONFIG = """\
>     'debug': {
>       'gn_args': 'is_debug=true',
>     },
>-    'memcheck': {
>-      'gn_args': 'rtc_use_memcheck=true',
>-    },
>     'android': {
>       'gn_args': 'target_os="android"',
>     }
>@@ -176,29 +154,6 @@ TEST_CONFIG = """\
> }
> """
> 
>-GYP_HACKS_CONFIG = """\
>-{
>-  'masters': {
>-    'chromium': {},
>-    'fake_master': {
>-      'fake_builder': 'fake_config',
>-    },
>-  },
>-  'configs': {
>-    'fake_config': ['fake_mixin'],
>-  },
>-  'mixins': {
>-    'fake_mixin': {
>-      'type': 'gyp',
>-      'gn_args': '',
>-      'gyp_defines':
>-         ('foo=bar llvm_force_head_revision=1 '
>-          'gyp_link_concurrency=1 baz=1'),
>-    },
>-  },
>-}
>-"""
>-
> 
> class UnitTest(unittest.TestCase):
>   def fake_mbw(self, files=None, win32=False):
>@@ -214,7 +169,7 @@ class UnitTest(unittest.TestCase):
>         },
>       }''')
>     mbw.files.setdefault(
>-        mbw.ToAbsPath('//build/args/bots/fake_master/fake_gn_args_bot.gn'),
>+        mbw.ToAbsPath('//build/args/bots/fake_master/fake_args_bot.gn'),
>         'is_debug = false\n')
>     if files:
>       for path, contents in files.items():
>@@ -234,38 +189,7 @@ class UnitTest(unittest.TestCase):
>       self.assertEqual(mbw.err, err)
>     return mbw
> 
>-  def test_clobber(self):
>-    files = {
>-      '/fake_src/out/Debug': None,
>-      '/fake_src/out/Debug/mb_type': None,
>-    }
>-    mbw = self.fake_mbw(files)
>-
>-    # The first time we run this, the build dir doesn't exist, so no clobber.
>-    self.check(['gen', '-c', 'gn_debug_goma', '//out/Debug'], mbw=mbw, ret=0)
>-    self.assertEqual(mbw.rmdirs, [])
>-    self.assertEqual(mbw.files['/fake_src/out/Debug/mb_type'], 'gn')
>-
>-    # The second time we run this, the build dir exists and matches, so no
>-    # clobber.
>-    self.check(['gen', '-c', 'gn_debug_goma', '//out/Debug'], mbw=mbw, ret=0)
>-    self.assertEqual(mbw.rmdirs, [])
>-    self.assertEqual(mbw.files['/fake_src/out/Debug/mb_type'], 'gn')
>-
>-    # Now we switch build types; this should result in a clobber.
>-    self.check(['gen', '-c', 'gyp_debug', '//out/Debug'], mbw=mbw, ret=0)
>-    self.assertEqual(mbw.rmdirs, ['/fake_src/out/Debug'])
>-    self.assertEqual(mbw.files['/fake_src/out/Debug/mb_type'], 'gyp')
>-
>-    # Now we delete mb_type; this checks the case where the build dir
>-    # exists but wasn't populated by mb; this should also result in a clobber.
>-    del mbw.files['/fake_src/out/Debug/mb_type']
>-    self.check(['gen', '-c', 'gyp_debug', '//out/Debug'], mbw=mbw, ret=0)
>-    self.assertEqual(mbw.rmdirs,
>-                     ['/fake_src/out/Debug', '/fake_src/out/Debug'])
>-    self.assertEqual(mbw.files['/fake_src/out/Debug/mb_type'], 'gyp')
>-
>-  def test_gn_analyze(self):
>+  def test_analyze(self):
>     files = {'/tmp/in.json': '''{\
>                "files": ["foo/foo_unittest.cc"],
>                "test_targets": ["foo_unittests"],
>@@ -280,7 +204,7 @@ class UnitTest(unittest.TestCase):
>     mbw = self.fake_mbw(files)
>     mbw.Call = lambda cmd, env=None, buffer_output=True: (0, '', '')
> 
>-    self.check(['analyze', '-c', 'gn_debug_goma', '//out/Default',
>+    self.check(['analyze', '-c', 'debug_goma', '//out/Default',
>                 '/tmp/in.json', '/tmp/out.json'], mbw=mbw, ret=0)
>     out = json.loads(mbw.files['/tmp/out.json'])
>     self.assertEqual(out, {
>@@ -289,9 +213,9 @@ class UnitTest(unittest.TestCase):
>       'test_targets': ['foo_unittests']
>     })
> 
>-  def test_gn_gen(self):
>+  def test_gen(self):
>     mbw = self.fake_mbw()
>-    self.check(['gen', '-c', 'gn_debug_goma', '//out/Default', '-g', '/goma'],
>+    self.check(['gen', '-c', 'debug_goma', '//out/Default', '-g', '/goma'],
>                mbw=mbw, ret=0)
>     self.assertMultiLineEqual(mbw.files['/fake_src/out/Default/args.gn'],
>                               ('goma_dir = "/goma"\n'
>@@ -304,7 +228,7 @@ class UnitTest(unittest.TestCase):
>                   mbw.out)
> 
>     mbw = self.fake_mbw(win32=True)
>-    self.check(['gen', '-c', 'gn_debug_goma', '-g', 'c:\\goma', '//out/Debug'],
>+    self.check(['gen', '-c', 'debug_goma', '-g', 'c:\\goma', '//out/Debug'],
>                mbw=mbw, ret=0)
>     self.assertMultiLineEqual(mbw.files['c:\\fake_src\\out\\Debug\\args.gn'],
>                               ('goma_dir = "c:\\\\goma"\n'
>@@ -314,45 +238,44 @@ class UnitTest(unittest.TestCase):
>                   '--check\n', mbw.out)
> 
>     mbw = self.fake_mbw()
>-    self.check(['gen', '-m', 'fake_master', '-b', 'fake_gn_args_bot',
>+    self.check(['gen', '-m', 'fake_master', '-b', 'fake_args_bot',
>                 '//out/Debug'],
>                mbw=mbw, ret=0)
>     self.assertEqual(
>         mbw.files['/fake_src/out/Debug/args.gn'],
>-        'import("//build/args/bots/fake_master/fake_gn_args_bot.gn")\n')
>+        'import("//build/args/bots/fake_master/fake_args_bot.gn")\n')
> 
> 
>-  def test_gn_gen_fails(self):
>+  def test_gen_fails(self):
>     mbw = self.fake_mbw()
>     mbw.Call = lambda cmd, env=None, buffer_output=True: (1, '', '')
>-    self.check(['gen', '-c', 'gn_debug_goma', '//out/Default'], mbw=mbw, ret=1)
>+    self.check(['gen', '-c', 'debug_goma', '//out/Default'], mbw=mbw, ret=1)
> 
>-  def test_gn_gen_swarming(self):
>+  def test_gen_swarming(self):
>     files = {
>-      '/tmp/swarming_targets': 'cc_perftests\n',
>+      '/tmp/swarming_targets': 'base_unittests\n',
>       '/fake_src/testing/buildbot/gn_isolate_map.pyl': (
>-          "{'cc_perftests': {"
>-          "  'label': '//cc:cc_perftests',"
>-          "  'type': 'console_test_launcher',"
>+          "{'base_unittests': {"
>+          "  'label': '//base:base_unittests',"
>+          "  'type': 'raw',"
>+          "  'args': [],"
>           "}}\n"
>       ),
>-      'c:\\fake_src\out\Default\cc_perftests.exe.runtime_deps': (
>-          "cc_perftests\n"
>+      '/fake_src/out/Default/base_unittests.runtime_deps': (
>+          "base_unittests\n"
>       ),
>     }
>-    mbw = self.fake_mbw(files=files, win32=True)
>+    mbw = self.fake_mbw(files)
>     self.check(['gen',
>-                '-c', 'gn_debug_goma',
>+                '-c', 'debug_goma',
>                 '--swarming-targets-file', '/tmp/swarming_targets',
>-                '--isolate-map-file',
>-                '/fake_src/testing/buildbot/gn_isolate_map.pyl',
>                 '//out/Default'], mbw=mbw, ret=0)
>-    self.assertIn('c:\\fake_src\\out\\Default\\cc_perftests.isolate',
>+    self.assertIn('/fake_src/out/Default/base_unittests.isolate',
>                   mbw.files)
>-    self.assertIn('c:\\fake_src\\out\\Default\\cc_perftests.isolated.gen.json',
>+    self.assertIn('/fake_src/out/Default/base_unittests.isolated.gen.json',
>                   mbw.files)
> 
>-  def test_gn_gen_swarming_android(self):
>+  def test_gen_swarming_android(self):
>     test_files = {
>       '/tmp/swarming_targets': 'base_unittests\n',
>       '/fake_src/testing/buildbot/gn_isolate_map.pyl': (
>@@ -365,7 +288,7 @@ class UnitTest(unittest.TestCase):
>           "base_unittests\n"
>       ),
>     }
>-    mbw = self.check(['gen', '-c', 'gn_android_bot', '//out/Default',
>+    mbw = self.check(['gen', '-c', 'android_bot', '//out/Default',
>                       '--swarming-targets-file', '/tmp/swarming_targets',
>                       '--isolate-map-file',
>                       '/fake_src/testing/buildbot/gn_isolate_map.pyl'],
>@@ -376,7 +299,8 @@ class UnitTest(unittest.TestCase):
>     files = isolate_file_contents['variables']['files']
>     command = isolate_file_contents['variables']['command']
> 
>-    self.assertEqual(files, ['base_unittests'])
>+    self.assertEqual(files, ['../../.vpython', '../../testing/test_env.py',
>+                             'base_unittests'])
>     self.assertEqual(command, [
>         '../../build/android/test_wrapper/logdog_wrapper.py',
>         '--target', 'base_unittests',
>@@ -385,7 +309,7 @@ class UnitTest(unittest.TestCase):
>         '--store-tombstones',
>     ])
> 
>-  def test_gn_gen_swarming_android_junit_test(self):
>+  def test_gen_swarming_android_junit_test(self):
>     test_files = {
>       '/tmp/swarming_targets': 'base_unittests\n',
>       '/fake_src/testing/buildbot/gn_isolate_map.pyl': (
>@@ -398,7 +322,7 @@ class UnitTest(unittest.TestCase):
>           "base_unittests\n"
>       ),
>     }
>-    mbw = self.check(['gen', '-c', 'gn_android_bot', '//out/Default',
>+    mbw = self.check(['gen', '-c', 'android_bot', '//out/Default',
>                       '--swarming-targets-file', '/tmp/swarming_targets',
>                       '--isolate-map-file',
>                       '/fake_src/testing/buildbot/gn_isolate_map.pyl'],
>@@ -409,7 +333,8 @@ class UnitTest(unittest.TestCase):
>     files = isolate_file_contents['variables']['files']
>     command = isolate_file_contents['variables']['command']
> 
>-    self.assertEqual(files, ['base_unittests'])
>+    self.assertEqual(files, ['../../.vpython', '../../testing/test_env.py',
>+                             'base_unittests'])
>     self.assertEqual(command, [
>         '../../build/android/test_wrapper/logdog_wrapper.py',
>         '--target', 'base_unittests',
>@@ -418,7 +343,7 @@ class UnitTest(unittest.TestCase):
>         '--store-tombstones',
>     ])
> 
>-  def test_gn_gen_timeout(self):
>+  def test_gen_timeout(self):
>     test_files = {
>       '/tmp/swarming_targets': 'base_unittests\n',
>       '/fake_src/testing/buildbot/gn_isolate_map.pyl': (
>@@ -432,7 +357,7 @@ class UnitTest(unittest.TestCase):
>           "base_unittests\n"
>       ),
>     }
>-    mbw = self.check(['gen', '-c', 'gn_debug_goma', '//out/Default',
>+    mbw = self.check(['gen', '-c', 'debug_goma', '//out/Default',
>                       '--swarming-targets-file', '/tmp/swarming_targets',
>                       '--isolate-map-file',
>                       '/fake_src/testing/buildbot/gn_isolate_map.pyl'],
>@@ -444,6 +369,7 @@ class UnitTest(unittest.TestCase):
>     command = isolate_file_contents['variables']['command']
> 
>     self.assertEqual(files, [
>+        '../../.vpython',
>         '../../testing/test_env.py',
>         '../../third_party/gtest-parallel/gtest-parallel',
>         '../../third_party/gtest-parallel/gtest_parallel.py',
>@@ -457,16 +383,15 @@ class UnitTest(unittest.TestCase):
>         '--gtest_color=no',
>         '--timeout=500',
>         '--retry_failed=3',
>-        './base_unittests',
>         '--workers=1',
>-        '--',
>+        './base_unittests',
>         '--asan=0',
>         '--lsan=0',
>         '--msan=0',
>         '--tsan=0',
>     ])
> 
>-  def test_gn_gen_script(self):
>+  def test_gen_script(self):
>     test_files = {
>       '/tmp/swarming_targets': 'base_unittests_script\n',
>       '/fake_src/testing/buildbot/gn_isolate_map.pyl': (
>@@ -481,7 +406,7 @@ class UnitTest(unittest.TestCase):
>           "base_unittests_script.py\n"
>       ),
>     }
>-    mbw = self.check(['gen', '-c', 'gn_debug_goma', '//out/Default',
>+    mbw = self.check(['gen', '-c', 'debug_goma', '//out/Default',
>                       '--swarming-targets-file', '/tmp/swarming_targets',
>                       '--isolate-map-file',
>                       '/fake_src/testing/buildbot/gn_isolate_map.pyl'],
>@@ -494,14 +419,52 @@ class UnitTest(unittest.TestCase):
>     command = isolate_file_contents['variables']['command']
> 
>     self.assertEqual(files, [
>-        'base_unittests',
>-        'base_unittests_script.py',
>+        '../../.vpython', '../../testing/test_env.py',
>+        'base_unittests', 'base_unittests_script.py',
>     ])
>     self.assertEqual(command, [
>         '../../base/base_unittests_script.py',
>     ])
> 
>-  def test_gn_gen_non_parallel_console_test_launcher(self):
>+  def test_gen_raw(self):
>+    test_files = {
>+      '/tmp/swarming_targets': 'base_unittests\n',
>+      '/fake_src/testing/buildbot/gn_isolate_map.pyl': (
>+          "{'base_unittests': {"
>+          "  'label': '//base:base_unittests',"
>+          "  'type': 'raw',"
>+          "}}\n"
>+      ),
>+      '/fake_src/out/Default/base_unittests.runtime_deps': (
>+          "base_unittests\n"
>+      ),
>+    }
>+    mbw = self.check(['gen', '-c', 'debug_goma', '//out/Default',
>+                      '--swarming-targets-file', '/tmp/swarming_targets',
>+                      '--isolate-map-file',
>+                      '/fake_src/testing/buildbot/gn_isolate_map.pyl'],
>+                     files=test_files, ret=0)
>+
>+    isolate_file = mbw.files['/fake_src/out/Default/base_unittests.isolate']
>+    isolate_file_contents = ast.literal_eval(isolate_file)
>+    files = isolate_file_contents['variables']['files']
>+    command = isolate_file_contents['variables']['command']
>+
>+    self.assertEqual(files, [
>+        '../../.vpython',
>+        '../../testing/test_env.py',
>+        'base_unittests',
>+    ])
>+    self.assertEqual(command, [
>+        '../../testing/test_env.py',
>+        './base_unittests',
>+        '--asan=0',
>+        '--lsan=0',
>+        '--msan=0',
>+        '--tsan=0',
>+    ])
>+
>+  def test_gen_non_parallel_console_test_launcher(self):
>     test_files = {
>       '/tmp/swarming_targets': 'base_unittests\n',
>       '/fake_src/testing/buildbot/gn_isolate_map.pyl': (
>@@ -514,7 +477,7 @@ class UnitTest(unittest.TestCase):
>           "base_unittests\n"
>       ),
>     }
>-    mbw = self.check(['gen', '-c', 'gn_debug_goma', '//out/Default',
>+    mbw = self.check(['gen', '-c', 'debug_goma', '//out/Default',
>                       '--swarming-targets-file', '/tmp/swarming_targets',
>                       '--isolate-map-file',
>                       '/fake_src/testing/buildbot/gn_isolate_map.pyl'],
>@@ -526,6 +489,7 @@ class UnitTest(unittest.TestCase):
>     command = isolate_file_contents['variables']['command']
> 
>     self.assertEqual(files, [
>+        '../../.vpython',
>         '../../testing/test_env.py',
>         '../../third_party/gtest-parallel/gtest-parallel',
>         '../../third_party/gtest-parallel/gtest_parallel.py',
>@@ -539,16 +503,15 @@ class UnitTest(unittest.TestCase):
>         '--gtest_color=no',
>         '--timeout=900',
>         '--retry_failed=3',
>-        './base_unittests',
>         '--workers=1',
>-        '--',
>+        './base_unittests',
>         '--asan=0',
>         '--lsan=0',
>         '--msan=0',
>         '--tsan=0',
>     ])
> 
>-  def test_gn_isolate_windowed_test_launcher_linux(self):
>+  def test_isolate_windowed_test_launcher_linux(self):
>     test_files = {
>       '/tmp/swarming_targets': 'base_unittests\n',
>       '/fake_src/testing/buildbot/gn_isolate_map.pyl': (
>@@ -562,7 +525,7 @@ class UnitTest(unittest.TestCase):
>           "some_resource_file\n"
>       ),
>     }
>-    mbw = self.check(['gen', '-c', 'gn_debug_goma', '//out/Default',
>+    mbw = self.check(['gen', '-c', 'debug_goma', '//out/Default',
>                       '--swarming-targets-file', '/tmp/swarming_targets',
>                       '--isolate-map-file',
>                       '/fake_src/testing/buildbot/gn_isolate_map.pyl'],
>@@ -574,6 +537,7 @@ class UnitTest(unittest.TestCase):
>     command = isolate_file_contents['variables']['command']
> 
>     self.assertEqual(files, [
>+        '../../.vpython',
>         '../../testing/test_env.py',
>         '../../testing/xvfb.py',
>         '../../third_party/gtest-parallel/gtest-parallel',
>@@ -590,14 +554,13 @@ class UnitTest(unittest.TestCase):
>         '--timeout=900',
>         '--retry_failed=3',
>         './base_unittests',
>-        '--',
>         '--asan=0',
>         '--lsan=0',
>         '--msan=0',
>         '--tsan=0',
>     ])
> 
>-  def test_gn_gen_windowed_test_launcher_win(self):
>+  def test_gen_windowed_test_launcher_win(self):
>     files = {
>       '/tmp/swarming_targets': 'unittests\n',
>       '/fake_src/testing/buildbot/gn_isolate_map.pyl': (
>@@ -613,7 +576,7 @@ class UnitTest(unittest.TestCase):
>     }
>     mbw = self.fake_mbw(files=files, win32=True)
>     self.check(['gen',
>-                '-c', 'gn_debug_goma',
>+                '-c', 'debug_goma',
>                 '--swarming-targets-file', '/tmp/swarming_targets',
>                 '--isolate-map-file',
>                 '/fake_src/testing/buildbot/gn_isolate_map.pyl',
>@@ -625,6 +588,7 @@ class UnitTest(unittest.TestCase):
>     command = isolate_file_contents['variables']['command']
> 
>     self.assertEqual(files, [
>+        '../../.vpython',
>         '../../testing/test_env.py',
>         '../../third_party/gtest-parallel/gtest-parallel',
>         '../../third_party/gtest-parallel/gtest_parallel.py',
>@@ -640,14 +604,13 @@ class UnitTest(unittest.TestCase):
>         '--timeout=900',
>         '--retry_failed=3',
>         r'.\unittests.exe',
>-        '--',
>         '--asan=0',
>         '--lsan=0',
>         '--msan=0',
>         '--tsan=0',
>     ])
> 
>-  def test_gn_gen_console_test_launcher(self):
>+  def test_gen_console_test_launcher(self):
>     test_files = {
>       '/tmp/swarming_targets': 'base_unittests\n',
>       '/fake_src/testing/buildbot/gn_isolate_map.pyl': (
>@@ -660,7 +623,7 @@ class UnitTest(unittest.TestCase):
>           "base_unittests\n"
>       ),
>     }
>-    mbw = self.check(['gen', '-c', 'gn_debug_goma', '//out/Default',
>+    mbw = self.check(['gen', '-c', 'debug_goma', '//out/Default',
>                       '--swarming-targets-file', '/tmp/swarming_targets',
>                       '--isolate-map-file',
>                       '/fake_src/testing/buildbot/gn_isolate_map.pyl'],
>@@ -672,6 +635,7 @@ class UnitTest(unittest.TestCase):
>     command = isolate_file_contents['variables']['command']
> 
>     self.assertEqual(files, [
>+        '../../.vpython',
>         '../../testing/test_env.py',
>         '../../third_party/gtest-parallel/gtest-parallel',
>         '../../third_party/gtest-parallel/gtest_parallel.py',
>@@ -686,65 +650,64 @@ class UnitTest(unittest.TestCase):
>         '--timeout=900',
>         '--retry_failed=3',
>         './base_unittests',
>-        '--',
>         '--asan=0',
>         '--lsan=0',
>         '--msan=0',
>         '--tsan=0',
>     ])
> 
>-  def test_gn_isolate_console_test_launcher_memcheck(self):
>+  def test_isolate_test_launcher_with_webcam(self):
>     test_files = {
>       '/tmp/swarming_targets': 'base_unittests\n',
>       '/fake_src/testing/buildbot/gn_isolate_map.pyl': (
>           "{'base_unittests': {"
>           "  'label': '//base:base_unittests',"
>           "  'type': 'console_test_launcher',"
>+          "  'use_webcam': True,"
>           "}}\n"
>       ),
>-      '/fake_src/out/Release/base_unittests.runtime_deps': (
>+      '/fake_src/out/Default/base_unittests.runtime_deps': (
>           "base_unittests\n"
>-          "lots_of_memcheck_dependencies\n"
>-          "../../tools_webrtc/valgrind/webrtc_tests.sh\n"
>+          "some_resource_file\n"
>       ),
>     }
>-    mbw = self.check(['gen', '-c', 'gn_memcheck_bot', '//out/Release',
>+    mbw = self.check(['gen', '-c', 'debug_goma', '//out/Default',
>                       '--swarming-targets-file', '/tmp/swarming_targets',
>                       '--isolate-map-file',
>                       '/fake_src/testing/buildbot/gn_isolate_map.pyl'],
>                      files=test_files, ret=0)
> 
>-    isolate_file = mbw.files['/fake_src/out/Release/base_unittests.isolate']
>+    isolate_file = mbw.files['/fake_src/out/Default/base_unittests.isolate']
>     isolate_file_contents = ast.literal_eval(isolate_file)
>     files = isolate_file_contents['variables']['files']
>     command = isolate_file_contents['variables']['command']
> 
>     self.assertEqual(files, [
>+        '../../.vpython',
>         '../../testing/test_env.py',
>-        '../../tools_webrtc/valgrind/webrtc_tests.sh',
>+        '../../third_party/gtest-parallel/gtest-parallel',
>+        '../../third_party/gtest-parallel/gtest_parallel.py',
>+        '../../tools_webrtc/ensure_webcam_is_running.py',
>+        '../../tools_webrtc/gtest-parallel-wrapper.py',
>         'base_unittests',
>-        'lots_of_memcheck_dependencies',
>+        'some_resource_file',
>     ])
>     self.assertEqual(command, [
>+        '../../tools_webrtc/ensure_webcam_is_running.py',
>         '../../testing/test_env.py',
>-        'bash',
>-        '../../tools_webrtc/valgrind/webrtc_tests.sh',
>-        '--tool',
>-        'memcheck',
>-        '--target',
>-        'Release',
>-        '--build-dir',
>-        '..',
>-        '--test',
>+        '../../tools_webrtc/gtest-parallel-wrapper.py',
>+        '--output_dir=${ISOLATED_OUTDIR}/test_logs',
>+        '--gtest_color=no',
>+        '--timeout=900',
>+        '--retry_failed=3',
>         './base_unittests',
>-        '--',
>         '--asan=0',
>         '--lsan=0',
>         '--msan=0',
>         '--tsan=0',
>     ])
> 
>-  def test_gn_isolate(self):
>+  def test_isolate(self):
>     files = {
>       '/fake_src/out/Default/toolchain.ninja': "",
>       '/fake_src/testing/buildbot/gn_isolate_map.pyl': (
>@@ -757,7 +720,7 @@ class UnitTest(unittest.TestCase):
>           "base_unittests\n"
>       ),
>     }
>-    self.check(['isolate', '-c', 'gn_debug_goma', '//out/Default',
>+    self.check(['isolate', '-c', 'debug_goma', '//out/Default',
>                 'base_unittests'], files=files, ret=0)
> 
>     # test running isolate on an existing build_dir
>@@ -768,7 +731,7 @@ class UnitTest(unittest.TestCase):
>     self.check(['isolate', '//out/Default', 'base_unittests'],
>                files=files, ret=0)
> 
>-  def test_gn_run(self):
>+  def test_run(self):
>     files = {
>       '/fake_src/testing/buildbot/gn_isolate_map.pyl': (
>           "{'base_unittests': {"
>@@ -780,55 +743,50 @@ class UnitTest(unittest.TestCase):
>           "base_unittests\n"
>       ),
>     }
>-    self.check(['run', '-c', 'gn_debug_goma', '//out/Default',
>+    self.check(['run', '-c', 'debug_goma', '//out/Default',
>                 'base_unittests'], files=files, ret=0)
> 
>-  def test_gn_lookup(self):
>-    self.check(['lookup', '-c', 'gn_debug_goma'], ret=0)
>+  def test_run_swarmed(self):
>+    files = {
>+      '/fake_src/testing/buildbot/gn_isolate_map.pyl': (
>+          "{'base_unittests': {"
>+          "  'label': '//base:base_unittests',"
>+          "  'type': 'raw',"
>+          "  'args': [],"
>+          "}}\n"
>+      ),
>+      '/fake_src/out/Default/base_unittests.runtime_deps': (
>+          "base_unittests\n"
>+      ),
>+    }
>+
>+    def run_stub(cmd, **_kwargs):
>+      if 'isolate.py' in cmd[1]:
>+        return 0, 'fake_hash base_unittests', ''
>+      else:
>+        return 0, '', ''
>+
>+    mbw = self.fake_mbw(files=files)
>+    mbw.Run = run_stub
>+    self.check(['run', '-s', '-c', 'debug_goma', '//out/Default',
>+                'base_unittests'], mbw=mbw, ret=0)
>+    self.check(['run', '-s', '-c', 'debug_goma', '-d', 'os', 'Win7',
>+                '//out/Default', 'base_unittests'], mbw=mbw, ret=0)
>+
>+  def test_lookup(self):
>+    self.check(['lookup', '-c', 'debug_goma'], ret=0)
> 
>-  def test_gn_lookup_goma_dir_expansion(self):
>-    self.check(['lookup', '-c', 'gn_rel_bot', '-g', '/foo'], ret=0,
>+  def test_lookup_goma_dir_expansion(self):
>+    self.check(['lookup', '-c', 'rel_bot', '-g', '/foo'], ret=0,
>                out=('\n'
>                     'Writing """\\\n'
>+                    'enable_doom_melon = true\n'
>                     'goma_dir = "/foo"\n'
>                     'is_debug = false\n'
>                     'use_goma = true\n'
>                     '""" to _path_/args.gn.\n\n'
>                     '/fake_src/buildtools/linux64/gn gen _path_\n'))
> 
>-  def test_gyp_analyze(self):
>-    mbw = self.check(['analyze', '-c', 'gyp_rel_bot', '//out/Release',
>-                      '/tmp/in.json', '/tmp/out.json'], ret=0)
>-    self.assertIn('analyzer', mbw.calls[0])
>-
>-  def test_gyp_crosscompile(self):
>-    mbw = self.fake_mbw()
>-    self.check(['gen', '-c', 'gyp_crosscompile', '//out/Release'],
>-               mbw=mbw, ret=0)
>-    self.assertTrue(mbw.cross_compile)
>-
>-  def test_gyp_gen(self):
>-    self.check(['gen', '-c', 'gyp_rel_bot', '-g', '/goma', '//out/Release'],
>-               ret=0,
>-               out=("GYP_DEFINES='goma=1 gomadir=/goma'\n"
>-                    "python build/gyp_chromium -G output_dir=out\n"))
>-
>-    mbw = self.fake_mbw(win32=True)
>-    self.check(['gen', '-c', 'gyp_rel_bot', '-g', 'c:\\goma', '//out/Release'],
>-               mbw=mbw, ret=0,
>-               out=("set GYP_DEFINES=goma=1 gomadir='c:\\goma'\n"
>-                    "python build\\gyp_chromium -G output_dir=out\n"))
>-
>-  def test_gyp_gen_fails(self):
>-    mbw = self.fake_mbw()
>-    mbw.Call = lambda cmd, env=None, buffer_output=True: (1, '', '')
>-    self.check(['gen', '-c', 'gyp_rel_bot', '//out/Release'], mbw=mbw, ret=1)
>-
>-  def test_gyp_lookup_goma_dir_expansion(self):
>-    self.check(['lookup', '-c', 'gyp_rel_bot', '-g', '/foo'], ret=0,
>-               out=("GYP_DEFINES='goma=1 gomadir=/foo'\n"
>-                    "python build/gyp_chromium -G output_dir=_path_\n"))
>-
>   def test_help(self):
>     orig_stdout = sys.stdout
>     try:
>@@ -846,7 +804,7 @@ class UnitTest(unittest.TestCase):
>     self.assertIn('Must specify a build --phase', mbw.out)
> 
>     # Check that passing a --phase to a single-phase builder fails.
>-    mbw = self.check(['lookup', '-m', 'fake_master', '-b', 'fake_gn_builder',
>+    mbw = self.check(['lookup', '-m', 'fake_master', '-b', 'fake_builder',
>                       '--phase', 'phase_1'], ret=1)
>     self.assertIn('Must not specify a build --phase', mbw.out)
> 
>@@ -868,16 +826,6 @@ class UnitTest(unittest.TestCase):
>     mbw = self.fake_mbw()
>     self.check(['validate'], mbw=mbw, ret=0)
> 
>-  def test_gyp_env_hacks(self):
>-    mbw = self.fake_mbw()
>-    mbw.files[mbw.default_config] = GYP_HACKS_CONFIG
>-    self.check(['lookup', '-c', 'fake_config'], mbw=mbw,
>-               ret=0,
>-               out=("GYP_DEFINES='foo=bar baz=1'\n"
>-                    "GYP_LINK_CONCURRENCY=1\n"
>-                    "LLVM_FORCE_HEAD_REVISION=1\n"
>-                    "python build/gyp_chromium -G output_dir=_path_\n"))
>-
> 
> if __name__ == '__main__':
>   unittest.main()
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/network_emulator/emulate.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/network_emulator/emulate.py
>old mode 100644
>new mode 100755
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/presubmit_checks_lib/check_orphan_headers.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/presubmit_checks_lib/check_orphan_headers.py
>index 31be7196ba7..deb445cd8ca 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/presubmit_checks_lib/check_orphan_headers.py
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/presubmit_checks_lib/check_orphan_headers.py
>@@ -114,7 +114,6 @@ def GetHeadersInBuildGnFileSources(file_content, target_abs_path):
>         source_file = source_file_match.group('source_file')
>         if source_file.endswith('.h'):
>           source_file_tokens = string.split(source_file, '/')
>-          # pylint: disable=star-args
>           headers_in_sources.add(os.path.join(target_abs_path,
>                                               *source_file_tokens))
>   return headers_in_sources
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/presubmit_checks_lib/check_orphan_headers_test.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/presubmit_checks_lib/check_orphan_headers_test.py
>old mode 100644
>new mode 100755
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/presubmit_checks_lib/check_package_boundaries_test.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/presubmit_checks_lib/check_package_boundaries_test.py
>old mode 100644
>new mode 100755
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/presubmit_checks_lib/gn_check_test.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/presubmit_checks_lib/gn_check_test.py
>old mode 100644
>new mode 100755
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/presubmit_checks_lib/testdata/all_build_files/subpackage1/subsubpackage1/BUILD.gn
>similarity index 100%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/tools/BUILD.gn
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/presubmit_checks_lib/testdata/all_build_files/subpackage1/subsubpackage1/BUILD.gn
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/network_tester/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/presubmit_checks_lib/testdata/all_build_files/subpackage2/subsubpackage2/BUILD.gn
>similarity index 100%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/tools/network_tester/BUILD.gn
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/presubmit_checks_lib/testdata/all_build_files/subpackage2/subsubpackage2/BUILD.gn
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/presubmit_checks_lib/testdata/circular_dependency/.gn.rej b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/presubmit_checks_lib/testdata/circular_dependency/.gn.rej
>deleted file mode 100644
>index 8e7613aaac1..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/presubmit_checks_lib/testdata/circular_dependency/.gn.rej
>+++ /dev/null
>@@ -1,3 +0,0 @@
>-diff a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/presubmit_checks_lib/testdata/circular_dependency/.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/presubmit_checks_lib/testdata/circular_dependency/.gn	(rejected hunks)
>-@@ -0,0 +1 @@
>-+buildconfig = "//BUILDCONFIG.gn"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/I420BufferImpl.java b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/presubmit_checks_lib/testdata/common_prefix/call/BUILD.gn
>similarity index 100%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/I420BufferImpl.java
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/presubmit_checks_lib/testdata/common_prefix/call/BUILD.gn
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/YuvConverter.java b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/presubmit_checks_lib/testdata/multiple_errors_multiple_targets/subpackage1/BUILD.gn
>similarity index 100%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/sdk/android/src/java/org/webrtc/YuvConverter.java
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/presubmit_checks_lib/testdata/multiple_errors_multiple_targets/subpackage1/BUILD.gn
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools/DEPS b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/presubmit_checks_lib/testdata/multiple_errors_multiple_targets/subpackage2/BUILD.gn
>similarity index 100%
>rename from Source/ThirdParty/libwebrtc/Source/webrtc/tools/DEPS
>rename to Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/presubmit_checks_lib/testdata/multiple_errors_multiple_targets/subpackage2/BUILD.gn
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/sanitizers/lsan_suppressions_webrtc.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/sanitizers/lsan_suppressions_webrtc.cc
>index 49dee1b5624..a1953868878 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/sanitizers/lsan_suppressions_webrtc.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/sanitizers/lsan_suppressions_webrtc.cc
>@@ -21,90 +21,85 @@
> // for the instructions on writing suppressions.
> char kLSanDefaultSuppressions[] =
> 
>-// ============ Leaks in third-party code shared with Chromium =============
>-// These entries are copied from build/sanitizers/lsan_suppressions.cc in
>-// Chromium. Please don't add new entries here unless they're present in there.
>-
>-// False positives in libfontconfig. http://crbug.com/39050
>-"leak:libfontconfig\n"
>-
>-// Leaks in Nvidia's libGL.
>-"leak:libGL.so\n"
>-
>-// XRandR has several one time leaks.
>-"leak:libxrandr\n"
>-
>-// xrandr leak. http://crbug.com/119677
>-"leak:XRRFindDisplay\n"
>-
>-// ========== Leaks in third-party code not shared with Chromium ===========
>-
>-// None known so far.
>-
>-// ================ Leaks in WebRTC code ================
>-// PLEASE DO NOT ADD SUPPRESSIONS FOR NEW LEAKS.
>-// Instead, commits that introduce memory leaks should be reverted. Suppressing
>-// the leak is acceptable in some cases when reverting is impossible, i.e. when
>-// enabling leak detection for the first time for a test target with
>-// pre-existing leaks.
>-
>-// rtc_unittest
>-// https://code.google.com/p/webrtc/issues/detail?id=3827 for details.
>-"leak:rtc::unstarted_task_test_DoNotDeleteTask2_Test::TestBody\n"
>-"leak:rtc::HttpServer::HandleConnection\n"
>-"leak:rtc::HttpServer::Connection::onHttpHeaderComplete\n"
>-"leak:rtc::HttpResponseData::set_success\n"
>-"leak:rtc::HttpData::changeHeader\n"
>-// https://code.google.com/p/webrtc/issues/detail?id=4149 for details.
>-"leak:StartDNSLookup\n"
>-// https://code.google.com/p/webrtc/issues/detail?id=2527
>-"leak:HangoutPubSubClientTest::HangoutPubSubClientTest\n"
>-"leak:MucRoomConfigTaskTest::SetUp\n"
>-"leak:MucRoomDiscoveryTaskTest::SetUp\n"
>-"leak:MucRoomLookupTaskTest::SetUp\n"
>-"leak:MucRoomUniqueHangoutIdTaskTest::SetUp\n"
>-"leak:PingTaskTest::SetUp\n"
>-"leak:PingXmppClient::SendStanza\n"
>-"leak:PubSubClientTest::PubSubClientTest\n"
>-"leak:PubSubTasksTest::PubSubTasksTest\n"
>-
>-// rtc_media_unittests
>-"leak:cricket::FakeNetworkInterface::SetOption\n"
>-"leak:CodecTest_TestCodecOperators_Test::TestBody\n"
>-"leak:VideoEngineTest*::ConstrainNewCodecBody\n"
>-"leak:VideoMediaChannelTest*::AddRemoveRecvStreams\n"
>-"leak:WebRtcVideoCapturerTest_TestCapture_Test::TestBody\n"
>-"leak:WebRtcVideoEngineTestFake_MultipleSendStreamsWithOneCapturer_Test::TestBody\n"
>-"leak:WebRtcVideoEngineTestFake_SetBandwidthInConference_Test::TestBody\n"
>-"leak:WebRtcVideoEngineTestFake_SetSendCodecsRejectBadFormat_Test::TestBody\n"
>-
>-// peerconnection_unittests
>-// https://code.google.com/p/webrtc/issues/detail?id=2528
>-"leak:cricket::FakeVideoMediaChannel::~FakeVideoMediaChannel\n"
>-"leak:cricket::MediaSessionDescriptionFactory::CreateAnswer\n"
>-"leak:cricket::MediaSessionDescriptionFactory::CreateOffer\n"
>-"leak:DtmfSenderTest_InsertEmptyTonesToCancelPreviousTask_Test::TestBody\n"
>-"leak:sigslot::_signal_base2*::~_signal_base2\n"
>-"leak:testing::internal::CmpHelperEQ\n"
>-"leak:webrtc::AudioDeviceLinuxALSA::InitMicrophone\n"
>-"leak:webrtc::AudioDeviceLinuxALSA::InitSpeaker\n"
>-"leak:webrtc::CreateIceCandidate\n"
>-"leak:webrtc::FakeConstraints::AddOptional\n"
>-"leak:webrtc::WebRtcIdentityRequestObserver::OnSuccess\n"
>-"leak:webrtc::WebRtcSessionDescriptionFactory::InternalCreateAnswer\n"
>-"leak:webrtc::WebRtcSessionDescriptionFactory::InternalCreateOffer\n"
>-"leak:PeerConnectionInterfaceTest_SsrcInOfferAnswer_Test::TestBody\n"
>-"leak:PeerConnectionInterfaceTest_CloseAndTestMethods_Test::TestBody\n"
>-"leak:WebRtcSdpTest::TestDeserializeRtcpFb\n"
>-"leak:WebRtcSdpTest::TestSerialize\n"
>-"leak:WebRtcSdpTest_SerializeSessionDescriptionWithDataChannelAndBandwidth_Test::TestBody\n"
>-"leak:WebRtcSdpTest_SerializeSessionDescriptionWithBandwidth_Test::TestBody\n"
>-"leak:WebRtcSessionTest::SetLocalDescriptionExpectError\n"
>-"leak:WebRtcSessionTest_TestAVOfferWithAudioOnlyAnswer_Test::TestBody\n"
>-
>-// PLEASE READ ABOVE BEFORE ADDING NEW SUPPRESSIONS.
>-
>-// End of suppressions.
>-;  // Please keep this semicolon.
>+    // ============ Leaks in third-party code shared with Chromium =============
>+    // These entries are copied from build/sanitizers/lsan_suppressions.cc in
>+    // Chromium. Please don't add new entries here unless they're present in
>+    // there.
>+
>+    // False positives in libfontconfig. http://crbug.com/39050
>+    "leak:libfontconfig\n"
>+
>+    // Leaks in Nvidia's libGL.
>+    "leak:libGL.so\n"
>+
>+    // XRandR has several one time leaks.
>+    "leak:libxrandr\n"
>+
>+    // xrandr leak. http://crbug.com/119677
>+    "leak:XRRFindDisplay\n"
>+
>+    // ========== Leaks in third-party code not shared with Chromium ===========
>+
>+    // None known so far.
>+
>+    // ================ Leaks in WebRTC code ================
>+    // PLEASE DO NOT ADD SUPPRESSIONS FOR NEW LEAKS.
>+    // Instead, commits that introduce memory leaks should be reverted.
>+    // Suppressing the leak is acceptable in some cases when reverting is
>+    // impossible, i.e. when enabling leak detection for the first time for a
>+    // test target with pre-existing leaks.
>+
>+    // rtc_unittest
>+    // https://code.google.com/p/webrtc/issues/detail?id=3827 for details.
>+    "leak:rtc::unstarted_task_test_DoNotDeleteTask2_Test::TestBody\n"
>+    "leak:rtc::HttpServer::HandleConnection\n"
>+    "leak:rtc::HttpServer::Connection::onHttpHeaderComplete\n"
>+    "leak:rtc::HttpResponseData::set_success\n"
>+    "leak:rtc::HttpData::changeHeader\n"
>+    // https://code.google.com/p/webrtc/issues/detail?id=4149 for details.
>+    "leak:StartDNSLookup\n"
>+
>+    // rtc_media_unittests
>+    "leak:cricket::FakeNetworkInterface::SetOption\n"
>+    "leak:CodecTest_TestCodecOperators_Test::TestBody\n"
>+    "leak:VideoEngineTest*::ConstrainNewCodecBody\n"
>+    "leak:VideoMediaChannelTest*::AddRemoveRecvStreams\n"
>+    "leak:WebRtcVideoCapturerTest_TestCapture_Test::TestBody\n"
>+    "leak:WebRtcVideoEngineTestFake_MultipleSendStreamsWithOneCapturer_Test::"
>+    "TestBody\n"
>+    "leak:WebRtcVideoEngineTestFake_SetBandwidthInConference_Test::TestBody\n"
>+    "leak:WebRtcVideoEngineTestFake_SetSendCodecsRejectBadFormat_Test::"
>+    "TestBody\n"
>+
>+    // peerconnection_unittests
>+    // https://code.google.com/p/webrtc/issues/detail?id=2528
>+    "leak:cricket::FakeVideoMediaChannel::~FakeVideoMediaChannel\n"
>+    "leak:cricket::MediaSessionDescriptionFactory::CreateAnswer\n"
>+    "leak:cricket::MediaSessionDescriptionFactory::CreateOffer\n"
>+    "leak:DtmfSenderTest_InsertEmptyTonesToCancelPreviousTask_Test::TestBody\n"
>+    "leak:sigslot::_signal_base2*::~_signal_base2\n"
>+    "leak:testing::internal::CmpHelperEQ\n"
>+    "leak:webrtc::AudioDeviceLinuxALSA::InitMicrophone\n"
>+    "leak:webrtc::AudioDeviceLinuxALSA::InitSpeaker\n"
>+    "leak:webrtc::CreateIceCandidate\n"
>+    "leak:webrtc::FakeConstraints::AddOptional\n"
>+    "leak:webrtc::WebRtcIdentityRequestObserver::OnSuccess\n"
>+    "leak:webrtc::WebRtcSessionDescriptionFactory::InternalCreateAnswer\n"
>+    "leak:webrtc::WebRtcSessionDescriptionFactory::InternalCreateOffer\n"
>+    "leak:PeerConnectionInterfaceTest_SsrcInOfferAnswer_Test::TestBody\n"
>+    "leak:PeerConnectionInterfaceTest_CloseAndTestMethods_Test::TestBody\n"
>+    "leak:WebRtcSdpTest::TestDeserializeRtcpFb\n"
>+    "leak:WebRtcSdpTest::TestSerialize\n"
>+    "leak:WebRtcSdpTest_SerializeSessionDescriptionWithDataChannelAndBandwidth_"
>+    "Test::TestBody\n"
>+    "leak:WebRtcSdpTest_SerializeSessionDescriptionWithBandwidth_Test::"
>+    "TestBody\n"
>+    "leak:WebRtcSessionTest::SetLocalDescriptionExpectError\n"
>+    "leak:WebRtcSessionTest_TestAVOfferWithAudioOnlyAnswer_Test::TestBody\n"
>+
>+    // PLEASE READ ABOVE BEFORE ADDING NEW SUPPRESSIONS.
>+
>+    // End of suppressions.
>+    ;  // Please keep this semicolon.
> 
> #endif  // LEAK_SANITIZER
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc
>index ca4977b197d..7ff65b573b1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc
>@@ -21,63 +21,51 @@
> // for the instructions on writing suppressions.
> char kTSanDefaultSuppressions[] =
> 
>-// WebRTC specific suppressions.
>-
>-// Split up suppressions covered previously by thread.cc and messagequeue.cc.
>-"race:vp8cx_remove_encoder_threads\n"
>-"race:third_party/libvpx/source/libvpx/vp9/common/vp9_scan.h\n"
>-
>-// Race in pulse initialization.
>-// https://code.google.com/p/webrtc/issues/detail?id=5152
>-"race:webrtc::AudioDeviceLinuxPulse::Init\n"
>-
>-// rtc_unittests
>-// https://code.google.com/p/webrtc/issues/detail?id=3911 for details.
>-"race:rtc::MessageQueueManager::Clear\n"
>-"race:rtc::Thread::Clear\n"
>-// https://code.google.com/p/webrtc/issues/detail?id=2080
>-"race:rtc_base/logging.cc\n"
>-"race:rtc_base/sharedexclusivelock_unittest.cc\n"
>-"race:rtc_base/signalthread_unittest.cc\n"
>-// https://code.google.com/p/webrtc/issues/detail?id=4456
>-"deadlock:rtc::MessageQueueManager::Clear\n"
>-"deadlock:rtc::MessageQueueManager::ClearInternal\n"
>-
>-// rtc_pc_unittests
>-// https://code.google.com/p/webrtc/issues/detail?id=2079
>-"race:rtc_base/testclient.cc\n"
>-"race:rtc_base/virtualsocketserver.cc\n"
>-"race:talk/p2p/base/stunserver_unittest.cc\n"
>-
>-// third_party/usrsctp
>-// TODO(jiayl): https://code.google.com/p/webrtc/issues/detail?id=3492
>-"race:user_sctp_timer_iterate\n"
>-
>-// https://code.google.com/p/webrtc/issues/detail?id=5151
>-"race:sctp_close\n"
>-
>-// Potential deadlocks detected after roll in r6516.
>-// https://code.google.com/p/webrtc/issues/detail?id=3509
>-"deadlock:webrtc::test::UdpSocketManagerPosixImpl::RemoveSocket\n"
>-
>-// TODO(pbos): Trace events are racy due to lack of proper POD atomics.
>-// https://code.google.com/p/webrtc/issues/detail?id=2497
>-"race:*trace_event_unique_catstatic*\n"
>-
>-// https://code.google.com/p/webrtc/issues/detail?id=4719
>-"race:webrtc::voe::TransmitMixer::PrepareDemux\n"
>-"race:webrtc::voe::TransmitMixer::EnableStereoChannelSwapping\n"
>-
>-// Race between InitCpuFlags and TestCpuFlag in libyuv.
>-// https://code.google.com/p/libyuv/issues/detail?id=508
>-"race:InitCpuFlags\n"
>-
>-// Test-only race due to PeerConnection::session() being virtual for testing.
>-// The stats collector may call session() before or after the destructor begins
>-// executing, which modifies the vtable.
>-"race:*RTCStatsIntegrationTest_GetsStatsWhileDestroyingPeerConnections_Test::TestBody\n"
>-
>-// End of suppressions.
>-;  // Please keep this semicolon.
>+    // WebRTC specific suppressions.
>+
>+    // Split up suppressions covered previously by thread.cc and
>+    // messagequeue.cc.
>+    "race:vp8cx_remove_encoder_threads\n"
>+    "race:third_party/libvpx/source/libvpx/vp9/common/vp9_scan.h\n"
>+
>+    // rtc_unittests
>+    // https://code.google.com/p/webrtc/issues/detail?id=2080
>+    "race:rtc_base/logging.cc\n"
>+    "race:rtc_base/sharedexclusivelock_unittest.cc\n"
>+    "race:rtc_base/signalthread_unittest.cc\n"
>+
>+    // rtc_pc_unittests
>+    // https://code.google.com/p/webrtc/issues/detail?id=2079
>+    "race:rtc_base/testclient.cc\n"
>+    "race:rtc_base/virtualsocketserver.cc\n"
>+    "race:talk/p2p/base/stunserver_unittest.cc\n"
>+
>+    // third_party/usrsctp
>+    // TODO(jiayl): https://code.google.com/p/webrtc/issues/detail?id=3492
>+    "race:user_sctp_timer_iterate\n"
>+
>+    // https://code.google.com/p/webrtc/issues/detail?id=5151
>+    "race:sctp_close\n"
>+
>+    // Potential deadlocks detected after roll in r6516.
>+    // https://code.google.com/p/webrtc/issues/detail?id=3509
>+    "deadlock:webrtc::test::UdpSocketManagerPosixImpl::RemoveSocket\n"
>+
>+    // TODO(pbos): Trace events are racy due to lack of proper POD atomics.
>+    // https://code.google.com/p/webrtc/issues/detail?id=2497
>+    "race:*trace_event_unique_catstatic*\n"
>+
>+    // Race between InitCpuFlags and TestCpuFlag in libyuv.
>+    // https://code.google.com/p/libyuv/issues/detail?id=508
>+    "race:InitCpuFlags\n"
>+
>+    // Test-only race due to PeerConnection::session() being virtual for
>+    // testing. The stats collector may call session() before or after the
>+    // destructor begins executing, which modifies the vtable.
>+    "race:*RTCStatsIntegrationTest_GetsStatsWhileDestroyingPeerConnections_"
>+    "Test::TestBody\n"
>+
>+    // End of suppressions.
>+    ;  // Please keep this semicolon.
> 
> #endif  // THREAD_SANITIZER
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/chrome_tests.bat b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/chrome_tests.bat
>deleted file mode 100644
>index 9d4c8ca8d34..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/chrome_tests.bat
>+++ /dev/null
>@@ -1,53 +0,0 @@
>-@echo off
>-:: Copyright (c) 2011 The Chromium Authors. All rights reserved.
>-:: Use of this source code is governed by a BSD-style license that can be
>-:: found in the LICENSE file.
>-
>-setlocal
>-
>-set THISDIR=%~dp0
>-set TOOL_NAME="unknown"
>-
>-:: Get the tool name and put it into TOOL_NAME {{{1
>-:: NB: SHIFT command doesn't modify %*
>-:PARSE_ARGS_LOOP
>-  if %1 == () GOTO:TOOLNAME_NOT_FOUND
>-  if %1 == --tool GOTO:TOOLNAME_FOUND
>-  SHIFT
>-  goto :PARSE_ARGS_LOOP
>-
>-:TOOLNAME_NOT_FOUND
>-echo "Please specify a tool (e.g. drmemory) by using --tool flag"
>-exit /B 1
>-
>-:TOOLNAME_FOUND
>-SHIFT
>-set TOOL_NAME=%1
>-:: }}}
>-if "%TOOL_NAME%" == "drmemory"          GOTO :SETUP_DRMEMORY
>-if "%TOOL_NAME%" == "drmemory_light"    GOTO :SETUP_DRMEMORY
>-if "%TOOL_NAME%" == "drmemory_full"     GOTO :SETUP_DRMEMORY
>-if "%TOOL_NAME%" == "drmemory_pattern"  GOTO :SETUP_DRMEMORY
>-echo "Unknown tool: `%TOOL_NAME%`! Only drmemory is supported right now"
>-exit /B 1
>-
>-:SETUP_DRMEMORY
>-:: Set up DRMEMORY_COMMAND to invoke Dr. Memory {{{1
>-set DRMEMORY_PATH=%THISDIR%..\..\third_party\drmemory
>-set DRMEMORY_SFX=%DRMEMORY_PATH%\drmemory-windows-sfx.exe
>-if EXIST %DRMEMORY_SFX% GOTO DRMEMORY_BINARY_OK
>-echo "Can't find Dr. Memory executables."
>-echo "See http://www.chromium.org/developers/how-tos/using-valgrind/dr-memory"
>-echo "for the instructions on how to get them."
>-exit /B 1
>-
>-:DRMEMORY_BINARY_OK
>-%DRMEMORY_SFX% -o%DRMEMORY_PATH%\unpacked -y
>-set DRMEMORY_COMMAND=%DRMEMORY_PATH%\unpacked\bin\drmemory.exe
>-:: }}}
>-goto :RUN_TESTS
>-
>-:RUN_TESTS
>-set PYTHONPATH=%THISDIR%../python/google
>-set RUNNING_ON_VALGRIND=yes
>-python %THISDIR%/chrome_tests.py %*
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/chrome_tests.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/chrome_tests.py
>deleted file mode 100644
>index fe899bce147..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/chrome_tests.py
>+++ /dev/null
>@@ -1,869 +0,0 @@
>-#!/usr/bin/env python
>-# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>-#
>-# Use of this source code is governed by a BSD-style license
>-# that can be found in the LICENSE file in the root of the source
>-# tree. An additional intellectual property rights grant can be found
>-# in the file PATENTS.  All contributing project authors may
>-# be found in the AUTHORS file in the root of the source tree.
>-
>-''' Runs various chrome tests through valgrind_test.py.'''
>-
>-import glob
>-import logging
>-import multiprocessing
>-import optparse
>-import os
>-import stat
>-import subprocess
>-import sys
>-
>-import logging_utils
>-import path_utils
>-
>-import common
>-import valgrind_test
>-
>-class TestNotFound(Exception): pass
>-
>-class MultipleGTestFiltersSpecified(Exception): pass
>-
>-class BuildDirNotFound(Exception): pass
>-
>-class BuildDirAmbiguous(Exception): pass
>-
>-class ExecutableNotFound(Exception): pass
>-
>-class BadBinary(Exception): pass
>-
>-class ChromeTests:
>-  SLOW_TOOLS = ["memcheck", "drmemory"]
>-  LAYOUT_TESTS_DEFAULT_CHUNK_SIZE = 300
>-
>-  def __init__(self, options, args, test):
>-    if ':' in test:
>-      (self._test, self._gtest_filter) = test.split(':', 1)
>-    else:
>-      self._test = test
>-      self._gtest_filter = options.gtest_filter
>-
>-    if self._test not in self._test_list:
>-      raise TestNotFound("Unknown test: %s" % test)
>-
>-    if options.gtest_filter and options.gtest_filter != self._gtest_filter:
>-      raise MultipleGTestFiltersSpecified("Can not specify both --gtest_filter "
>-                                          "and --test %s" % test)
>-
>-    self._options = options
>-    self._args = args
>-
>-    script_dir = path_utils.ScriptDir()
>-    # Compute the top of the tree (the "source dir") from the script dir (where
>-    # this script lives).  We assume that the script dir is in tools/valgrind/
>-    # relative to the top of the tree.
>-    self._source_dir = os.path.dirname(os.path.dirname(script_dir))
>-    # since this path is used for string matching, make sure it's always
>-    # an absolute Unix-style path
>-    self._source_dir = os.path.abspath(self._source_dir).replace('\\', '/')
>-    valgrind_test_script = os.path.join(script_dir, "valgrind_test.py")
>-    self._command_preamble = ["--source-dir=%s" % (self._source_dir)]
>-
>-    if not self._options.build_dir:
>-      dirs = [
>-        os.path.join(self._source_dir, "xcodebuild", "Debug"),
>-        os.path.join(self._source_dir, "out", "Debug"),
>-        os.path.join(self._source_dir, "build", "Debug"),
>-      ]
>-      build_dir = [d for d in dirs if os.path.isdir(d)]
>-      if len(build_dir) > 1:
>-        raise BuildDirAmbiguous("Found more than one suitable build dir:\n"
>-                                "%s\nPlease specify just one "
>-                                "using --build-dir" % ", ".join(build_dir))
>-      elif build_dir:
>-        self._options.build_dir = build_dir[0]
>-      else:
>-        self._options.build_dir = None
>-
>-    if self._options.build_dir:
>-      build_dir = os.path.abspath(self._options.build_dir)
>-      self._command_preamble += ["--build-dir=%s" % (self._options.build_dir)]
>-
>-  def _EnsureBuildDirFound(self):
>-    if not self._options.build_dir:
>-      raise BuildDirNotFound("Oops, couldn't find a build dir, please "
>-                             "specify it manually using --build-dir")
>-
>-  def _DefaultCommand(self, tool, exe=None, valgrind_test_args=None):
>-    '''Generates the default command array that most tests will use.'''
>-    if exe and common.IsWindows():
>-      exe += '.exe'
>-
>-    cmd = list(self._command_preamble)
>-
>-    # Find all suppressions matching the following pattern:
>-    # tools/valgrind/TOOL/suppressions[_PLATFORM].txt
>-    # and list them with --suppressions= prefix.
>-    script_dir = path_utils.ScriptDir()
>-    tool_name = tool.ToolName();
>-    suppression_file = os.path.join(script_dir, tool_name, "suppressions.txt")
>-    if os.path.exists(suppression_file):
>-      cmd.append("--suppressions=%s" % suppression_file)
>-    # Platform-specific suppression
>-    for platform in common.PlatformNames():
>-      platform_suppression_file = \
>-          os.path.join(script_dir, tool_name, 'suppressions_%s.txt' % platform)
>-      if os.path.exists(platform_suppression_file):
>-        cmd.append("--suppressions=%s" % platform_suppression_file)
>-
>-    if tool_name == "drmemory":
>-      if self._options.drmemory_ops:
>-        # prepending " " to avoid Dr. Memory's option confusing optparse
>-        cmd += ["--drmemory_ops", " " + self._options.drmemory_ops]
>-
>-    if self._options.valgrind_tool_flags:
>-      cmd += self._options.valgrind_tool_flags.split(" ")
>-    if self._options.keep_logs:
>-      cmd += ["--keep_logs"]
>-    if valgrind_test_args != None:
>-      for arg in valgrind_test_args:
>-        cmd.append(arg)
>-    if exe:
>-      self._EnsureBuildDirFound()
>-      exe_path = os.path.join(self._options.build_dir, exe)
>-      if not os.path.exists(exe_path):
>-        raise ExecutableNotFound("Couldn't find '%s'" % exe_path)
>-
>-      # Make sure we don't try to test ASan-built binaries
>-      # with other dynamic instrumentation-based tools.
>-      # TODO(timurrrr): also check TSan and MSan?
>-      # `nm` might not be available, so use try-except.
>-      try:
>-        # Do not perform this check on OS X, as 'nm' on 10.6 can't handle
>-        # binaries built with Clang 3.5+.
>-        if not common.IsMac():
>-          nm_output = subprocess.check_output(["nm", exe_path])
>-          if nm_output.find("__asan_init") != -1:
>-            raise BadBinary("You're trying to run an executable instrumented "
>-                            "with AddressSanitizer under %s. Please provide "
>-                            "an uninstrumented executable." % tool_name)
>-      except OSError:
>-        pass
>-
>-      cmd.append(exe_path)
>-      # Valgrind runs tests slowly, so slow tests hurt more; show elapased time
>-      # so we can find the slowpokes.
>-      cmd.append("--gtest_print_time")
>-      # Built-in test launcher for gtest-based executables runs tests using
>-      # multiple process by default. Force the single-process mode back.
>-      cmd.append("--single-process-tests")
>-    if self._options.gtest_repeat:
>-      cmd.append("--gtest_repeat=%s" % self._options.gtest_repeat)
>-    if self._options.gtest_shuffle:
>-      cmd.append("--gtest_shuffle")
>-    if self._options.gtest_break_on_failure:
>-      cmd.append("--gtest_break_on_failure")
>-    if self._options.test_launcher_bot_mode:
>-      cmd.append("--test-launcher-bot-mode")
>-    if self._options.test_launcher_total_shards is not None:
>-      cmd.append("--test-launcher-total-shards=%d"
>-                 % self._options.test_launcher_total_shards)
>-    if self._options.test_launcher_shard_index is not None:
>-      cmd.append("--test-launcher-shard-index=%d"
>-                 % self._options.test_launcher_shard_index)
>-    return cmd
>-
>-  def Run(self):
>-    ''' Runs the test specified by command-line argument --test '''
>-    logging.info("running test %s" % (self._test))
>-    return self._test_list[self._test](self)
>-
>-  def _AppendGtestFilter(self, tool, name, cmd):
>-    '''Append an appropriate --gtest_filter flag to the googletest binary
>-       invocation.
>-       If the user passed their own filter mentioning only one test, just use
>-       it. Otherwise, filter out tests listed in the appropriate gtest_exclude
>-       files.
>-    '''
>-    if (self._gtest_filter and
>-        ":" not in self._gtest_filter and
>-        "?" not in self._gtest_filter and
>-        "*" not in self._gtest_filter):
>-      cmd.append("--gtest_filter=%s" % self._gtest_filter)
>-      return
>-
>-    filters = []
>-    gtest_files_dir = os.path.join(path_utils.ScriptDir(), "gtest_exclude")
>-
>-    gtest_filter_files = [
>-        os.path.join(gtest_files_dir, name + ".gtest-%s.txt" % tool.ToolName())]
>-    # Use ".gtest.txt" files only for slow tools, as they now contain
>-    # Valgrind- and Dr.Memory-specific filters.
>-    # TODO(glider): rename the files to ".gtest_slow.txt"
>-    if tool.ToolName() in ChromeTests.SLOW_TOOLS:
>-      gtest_filter_files += [os.path.join(gtest_files_dir, name + ".gtest.txt")]
>-    for platform_suffix in common.PlatformNames():
>-      gtest_filter_files += [
>-        os.path.join(gtest_files_dir, name + ".gtest_%s.txt" % platform_suffix),
>-        os.path.join(gtest_files_dir, name + ".gtest-%s_%s.txt" % \
>-            (tool.ToolName(), platform_suffix))]
>-    logging.info("Reading gtest exclude filter files:")
>-    for filename in gtest_filter_files:
>-      # strip the leading absolute path (may be very long on the bot)
>-      # and the following / or \.
>-      readable_filename = filename.replace("\\", "/")  # '\' on Windows
>-      readable_filename = readable_filename.replace(self._source_dir, "")[1:]
>-      if not os.path.exists(filename):
>-        logging.info("  \"%s\" - not found" % readable_filename)
>-        continue
>-      logging.info("  \"%s\" - OK" % readable_filename)
>-      f = open(filename, 'r')
>-      for line in f.readlines():
>-        if line.startswith("#") or line.startswith("//") or line.isspace():
>-          continue
>-        line = line.rstrip()
>-        test_prefixes = ["FLAKY", "FAILS"]
>-        for p in test_prefixes:
>-          # Strip prefixes from the test names.
>-          line = line.replace(".%s_" % p, ".")
>-        # Exclude the original test name.
>-        filters.append(line)
>-        if line[-2:] != ".*":
>-          # List all possible prefixes if line doesn't end with ".*".
>-          for p in test_prefixes:
>-            filters.append(line.replace(".", ".%s_" % p))
>-    # Get rid of duplicates.
>-    filters = set(filters)
>-    gtest_filter = self._gtest_filter
>-    if len(filters):
>-      if gtest_filter:
>-        gtest_filter += ":"
>-        if gtest_filter.find("-") < 0:
>-          gtest_filter += "-"
>-      else:
>-        gtest_filter = "-"
>-      gtest_filter += ":".join(filters)
>-    if gtest_filter:
>-      cmd.append("--gtest_filter=%s" % gtest_filter)
>-
>-  @staticmethod
>-  def ShowTests():
>-    test_to_names = {}
>-    for name, test_function in ChromeTests._test_list.iteritems():
>-      test_to_names.setdefault(test_function, []).append(name)
>-
>-    name_to_aliases = {}
>-    for names in test_to_names.itervalues():
>-      names.sort(key=lambda name: len(name))
>-      name_to_aliases[names[0]] = names[1:]
>-
>-    print
>-    print "Available tests:"
>-    print "----------------"
>-    for name, aliases in sorted(name_to_aliases.iteritems()):
>-      if aliases:
>-        print "   {} (aka {})".format(name, ', '.join(aliases))
>-      else:
>-        print "   {}".format(name)
>-
>-  def SetupLdPath(self, requires_build_dir):
>-    if requires_build_dir:
>-      self._EnsureBuildDirFound()
>-    elif not self._options.build_dir:
>-      return
>-
>-    # Append build_dir to LD_LIBRARY_PATH so external libraries can be loaded.
>-    if (os.getenv("LD_LIBRARY_PATH")):
>-      os.putenv("LD_LIBRARY_PATH", "%s:%s" % (os.getenv("LD_LIBRARY_PATH"),
>-                                              self._options.build_dir))
>-    else:
>-      os.putenv("LD_LIBRARY_PATH", self._options.build_dir)
>-
>-  def SimpleTest(self, module, name, valgrind_test_args=None, cmd_args=None):
>-    tool = valgrind_test.CreateTool(self._options.valgrind_tool)
>-    cmd = self._DefaultCommand(tool, name, valgrind_test_args)
>-    self._AppendGtestFilter(tool, name, cmd)
>-    cmd.extend(['--test-tiny-timeout=1000'])
>-    if cmd_args:
>-      cmd.extend(cmd_args)
>-
>-    self.SetupLdPath(True)
>-    return tool.Run(cmd, module)
>-
>-  def RunCmdLine(self):
>-    tool = valgrind_test.CreateTool(self._options.valgrind_tool)
>-    cmd = self._DefaultCommand(tool, None, self._args)
>-    self.SetupLdPath(False)
>-    return tool.Run(cmd, None)
>-
>-  def TestAccessibility(self):
>-    return self.SimpleTest("accessibility", "accessibility_unittests")
>-
>-  def TestAddressInput(self):
>-    return self.SimpleTest("addressinput", "libaddressinput_unittests")
>-
>-  def TestAngle(self):
>-    return self.SimpleTest("angle", "angle_unittests")
>-
>-  def TestAppList(self):
>-    return self.SimpleTest("app_list", "app_list_unittests")
>-
>-  def TestAsh(self):
>-    return self.SimpleTest("ash", "ash_unittests")
>-
>-  def TestAura(self):
>-    return self.SimpleTest("aura", "aura_unittests")
>-
>-  def TestBase(self):
>-    return self.SimpleTest("base", "base_unittests")
>-
>-  def TestBlinkHeap(self):
>-    return self.SimpleTest("blink_heap", "blink_heap_unittests")
>-
>-  def TestBlinkPlatform(self):
>-    return self.SimpleTest("blink_platform", "blink_platform_unittests")
>-
>-  def TestCacheInvalidation(self):
>-    return self.SimpleTest("cacheinvalidation", "cacheinvalidation_unittests")
>-
>-  def TestCast(self):
>-    return self.SimpleTest("chrome", "cast_unittests")
>-
>-  def TestCC(self):
>-    return self.SimpleTest("cc", "cc_unittests",
>-                           cmd_args=[
>-                               "--cc-layer-tree-test-long-timeout"])
>-
>-  def TestChromeApp(self):
>-    return self.SimpleTest("chrome_app", "chrome_app_unittests")
>-
>-  def TestChromeElf(self):
>-    return self.SimpleTest("chrome_elf", "chrome_elf_unittests")
>-
>-  def TestChromeDriver(self):
>-    return self.SimpleTest("chromedriver", "chromedriver_unittests")
>-
>-  def TestChromeOS(self):
>-    return self.SimpleTest("chromeos", "chromeos_unittests")
>-
>-  def TestComponents(self):
>-    return self.SimpleTest("components", "components_unittests")
>-
>-  def TestCompositor(self):
>-    return self.SimpleTest("compositor", "compositor_unittests")
>-
>-  def TestContent(self):
>-    return self.SimpleTest("content", "content_unittests")
>-
>-  def TestCourgette(self):
>-    return self.SimpleTest("courgette", "courgette_unittests")
>-
>-  def TestCrypto(self):
>-    return self.SimpleTest("crypto", "crypto_unittests")
>-
>-  def TestDevice(self):
>-    return self.SimpleTest("device", "device_unittests")
>-
>-  def TestDisplay(self):
>-    return self.SimpleTest("display", "display_unittests")
>-
>-  def TestEvents(self):
>-    return self.SimpleTest("events", "events_unittests")
>-
>-  def TestExtensions(self):
>-    return self.SimpleTest("extensions", "extensions_unittests")
>-
>-  def TestFFmpegRegressions(self):
>-    return self.SimpleTest("chrome", "ffmpeg_regression_tests")
>-
>-  def TestGCM(self):
>-    return self.SimpleTest("gcm", "gcm_unit_tests")
>-
>-  def TestGfx(self):
>-    return self.SimpleTest("gfx", "gfx_unittests")
>-
>-  def TestGin(self):
>-    return self.SimpleTest("gin", "gin_unittests")
>-
>-  def TestGoogleApis(self):
>-    return self.SimpleTest("google_apis", "google_apis_unittests")
>-
>-  def TestGPU(self):
>-    return self.SimpleTest("gpu", "gpu_unittests")
>-
>-  def TestIpc(self):
>-    return self.SimpleTest("ipc", "ipc_tests",
>-                           valgrind_test_args=["--trace_children"])
>-
>-  def TestInstallerUtil(self):
>-    return self.SimpleTest("installer_util", "installer_util_unittests")
>-
>-  def TestInstallStatic(self):
>-    return self.SimpleTest("install_static", "install_static_unittests")
>-
>-  def TestJingle(self):
>-    return self.SimpleTest("chrome", "jingle_unittests")
>-
>-  def TestKeyboard(self):
>-    return self.SimpleTest("keyboard", "keyboard_unittests")
>-
>-  def TestLatency(self):
>-    return self.SimpleTest("latency", "latency_unittests")
>-
>-  def TestMedia(self):
>-    return self.SimpleTest("chrome", "media_unittests")
>-
>-  def TestMessageCenter(self):
>-    return self.SimpleTest("message_center", "message_center_unittests")
>-
>-  def TestMidi(self):
>-    return self.SimpleTest("chrome", "midi_unittests")
>-
>-  def TestMojoCommon(self):
>-    return self.SimpleTest("mojo_common", "mojo_common_unittests")
>-
>-  def TestMojoPublicBindings(self):
>-    return self.SimpleTest("mojo_public_bindings",
>-                           "mojo_public_bindings_unittests")
>-
>-  def TestMojoPublicSystem(self):
>-    return self.SimpleTest("mojo_public_system",
>-                           "mojo_public_system_unittests")
>-
>-  def TestMojoPublicSysPerf(self):
>-    return self.SimpleTest("mojo_public_sysperf",
>-                           "mojo_public_system_perftests")
>-
>-  def TestMojoSystem(self):
>-    return self.SimpleTest("mojo_system", "mojo_system_unittests")
>-
>-  def TestNet(self):
>-    return self.SimpleTest("net", "net_unittests")
>-
>-  def TestNetPerf(self):
>-    return self.SimpleTest("net", "net_perftests")
>-
>-  def TestPhoneNumber(self):
>-    return self.SimpleTest("phonenumber", "libphonenumber_unittests")
>-
>-  def TestPPAPI(self):
>-    return self.SimpleTest("chrome", "ppapi_unittests")
>-
>-  def TestPrinting(self):
>-    return self.SimpleTest("chrome", "printing_unittests")
>-
>-  def TestRemoting(self):
>-    return self.SimpleTest("chrome", "remoting_unittests",
>-                           cmd_args=[
>-                               "--ui-test-action-timeout=60000",
>-                               "--ui-test-action-max-timeout=150000"])
>-
>-  def TestSkia(self):
>-    return self.SimpleTest("skia", "skia_unittests")
>-
>-  def TestSql(self):
>-    return self.SimpleTest("chrome", "sql_unittests")
>-
>-  def TestStorage(self):
>-    return self.SimpleTest("storage", "storage_unittests")
>-
>-  def TestLinuxSandbox(self):
>-    return self.SimpleTest("sandbox", "sandbox_linux_unittests")
>-
>-  def TestUnit(self):
>-    # http://crbug.com/51716
>-    # Disabling all unit tests
>-    # Problems reappeared after r119922
>-    if common.IsMac() and (self._options.valgrind_tool == "memcheck"):
>-      logging.warning("unit_tests are disabled for memcheck on MacOS.")
>-      return 0;
>-    return self.SimpleTest("chrome", "unit_tests")
>-
>-  def TestUIBaseUnit(self):
>-    return self.SimpleTest("chrome", "ui_base_unittests")
>-
>-  def TestUIChromeOS(self):
>-    return self.SimpleTest("chrome", "ui_chromeos_unittests")
>-
>-  def TestURL(self):
>-    return self.SimpleTest("chrome", "url_unittests")
>-
>-  def TestViews(self):
>-    return self.SimpleTest("views", "views_unittests")
>-
>-
>-  # Valgrind timeouts are in seconds.
>-  UI_VALGRIND_ARGS = ["--timeout=14400", "--trace_children", "--indirect"]
>-  # UI test timeouts are in milliseconds.
>-  UI_TEST_ARGS = ["--ui-test-action-timeout=60000",
>-                  "--ui-test-action-max-timeout=150000",
>-                  "--no-sandbox"]
>-
>-  # TODO(thestig) fine-tune these values.
>-  # Valgrind timeouts are in seconds.
>-  BROWSER_VALGRIND_ARGS = ["--timeout=50000", "--trace_children", "--indirect"]
>-  # Browser test timeouts are in milliseconds.
>-  BROWSER_TEST_ARGS = ["--ui-test-action-timeout=400000",
>-                       "--ui-test-action-max-timeout=800000",
>-                       "--no-sandbox"]
>-
>-  def TestBrowser(self):
>-    return self.SimpleTest("chrome", "browser_tests",
>-                           valgrind_test_args=self.BROWSER_VALGRIND_ARGS,
>-                           cmd_args=self.BROWSER_TEST_ARGS)
>-
>-  def TestContentBrowser(self):
>-    return self.SimpleTest("content", "content_browsertests",
>-                           valgrind_test_args=self.BROWSER_VALGRIND_ARGS,
>-                           cmd_args=self.BROWSER_TEST_ARGS)
>-
>-  def TestInteractiveUI(self):
>-    return self.SimpleTest("chrome", "interactive_ui_tests",
>-                           valgrind_test_args=self.UI_VALGRIND_ARGS,
>-                           cmd_args=self.UI_TEST_ARGS)
>-
>-  def TestSyncIntegration(self):
>-    return self.SimpleTest("chrome", "sync_integration_tests",
>-                           valgrind_test_args=self.UI_VALGRIND_ARGS,
>-                           cmd_args=(["--ui-test-action-max-timeout=450000"]))
>-
>-  def TestLayoutChunk(self, chunk_num, chunk_size):
>-    # Run tests [chunk_num*chunk_size .. (chunk_num+1)*chunk_size) from the
>-    # list of tests.  Wrap around to beginning of list at end.
>-    # If chunk_size is zero, run all tests in the list once.
>-    # If a text file is given as argument, it is used as the list of tests.
>-    assert((chunk_size == 0) != (len(self._args) == 0))
>-    # Build the ginormous commandline in 'cmd'.
>-    # It's going to be roughly
>-    #  python valgrind_test.py ...
>-    # but we'll use the --indirect flag to valgrind_test.py
>-    # to avoid valgrinding python.
>-    # Start by building the valgrind_test.py commandline.
>-    tool = valgrind_test.CreateTool(self._options.valgrind_tool)
>-    cmd = self._DefaultCommand(tool)
>-    cmd.append("--trace_children")
>-    cmd.append("--indirect_webkit_layout")
>-    cmd.append("--ignore_exit_code")
>-    # Now build script_cmd, the run-webkits-tests commandline.
>-    # Store each chunk in its own directory so that we can find the data later
>-    chunk_dir = os.path.join("layout", "chunk_%05d" % chunk_num)
>-    out_dir = os.path.join(path_utils.ScriptDir(), "latest")
>-    out_dir = os.path.join(out_dir, chunk_dir)
>-    if os.path.exists(out_dir):
>-      old_files = glob.glob(os.path.join(out_dir, "*.txt"))
>-      for f in old_files:
>-        os.remove(f)
>-    else:
>-      os.makedirs(out_dir)
>-    script = os.path.join(self._source_dir, "third_party", "WebKit", "Tools",
>-                          "Scripts", "run-webkit-tests")
>-    # http://crbug.com/260627: After the switch to content_shell from DRT, each
>-    # test now brings up 3 processes.  Under Valgrind, they become memory bound
>-    # and can eventually OOM if we don't reduce the total count.
>-    # It'd be nice if content_shell automatically throttled the startup of new
>-    # tests if we're low on memory.
>-    jobs = max(1, int(multiprocessing.cpu_count() * 0.3))
>-    script_cmd = ["python", script, "-v",
>-                  # run a separate DumpRenderTree for each test
>-                  "--batch-size=1",
>-                  "--fully-parallel",
>-                  "--child-processes=%d" % jobs,
>-                  "--time-out-ms=800000",
>-                  "--no-retry-failures",  # retrying takes too much time
>-                  # http://crbug.com/176908: Don't launch a browser when done.
>-                  "--no-show-results",
>-                  "--nocheck-sys-deps",
>-                  "--additional-driver-flag=--no-sandbox"]
>-    # Pass build mode to run-webkit-tests.  We aren't passed it directly,
>-    # so parse it out of build_dir.  run-webkit-tests can only handle
>-    # the two values "Release" and "Debug".
>-    # TODO(Hercules): unify how all our scripts pass around build mode
>-    # (--mode / --target / --build-dir / --debug)
>-    if self._options.build_dir:
>-      build_root, mode = os.path.split(self._options.build_dir)
>-      script_cmd.extend(["--build-directory", build_root, "--target", mode])
>-    if (chunk_size > 0):
>-      script_cmd.append("--run-chunk=%d:%d" % (chunk_num, chunk_size))
>-    if len(self._args):
>-      # if the arg is a txt file, then treat it as a list of tests
>-      if os.path.isfile(self._args[0]) and self._args[0][-4:] == ".txt":
>-        script_cmd.append("--test-list=%s" % self._args[0])
>-      else:
>-        script_cmd.extend(self._args)
>-    self._AppendGtestFilter(tool, "layout", script_cmd)
>-    # Now run script_cmd with the wrapper in cmd
>-    cmd.extend(["--"])
>-    cmd.extend(script_cmd)
>-
>-    # Layout tests often times fail quickly, but the buildbot remains green.
>-    # Detect this situation when running with the default chunk size.
>-    if chunk_size == self.LAYOUT_TESTS_DEFAULT_CHUNK_SIZE:
>-      min_runtime_in_seconds=120
>-    else:
>-      min_runtime_in_seconds=0
>-    ret = tool.Run(cmd, "layout", min_runtime_in_seconds=min_runtime_in_seconds)
>-    return ret
>-
>-
>-  def TestLayout(self):
>-    # A "chunk file" is maintained in the local directory so that each test
>-    # runs a slice of the layout tests of size chunk_size that increments with
>-    # each run.  Since tests can be added and removed from the layout tests at
>-    # any time, this is not going to give exact coverage, but it will allow us
>-    # to continuously run small slices of the layout tests under valgrind rather
>-    # than having to run all of them in one shot.
>-    chunk_size = self._options.num_tests
>-    if chunk_size == 0 or len(self._args):
>-      return self.TestLayoutChunk(0, 0)
>-    chunk_num = 0
>-    chunk_file = os.path.join("valgrind_layout_chunk.txt")
>-    logging.info("Reading state from " + chunk_file)
>-    try:
>-      f = open(chunk_file)
>-      if f:
>-        chunk_str = f.read()
>-        if len(chunk_str):
>-          chunk_num = int(chunk_str)
>-        # This should be enough so that we have a couple of complete runs
>-        # of test data stored in the archive (although note that when we loop
>-        # that we almost guaranteed won't be at the end of the test list)
>-        if chunk_num > 10000:
>-          chunk_num = 0
>-        f.close()
>-    except IOError, (errno, strerror):
>-      logging.error("error reading from file %s (%d, %s)" % (chunk_file,
>-                    errno, strerror))
>-    # Save the new chunk size before running the tests. Otherwise if a
>-    # particular chunk hangs the bot, the chunk number will never get
>-    # incremented and the bot will be wedged.
>-    logging.info("Saving state to " + chunk_file)
>-    try:
>-      f = open(chunk_file, "w")
>-      chunk_num += 1
>-      f.write("%d" % chunk_num)
>-      f.close()
>-    except IOError, (errno, strerror):
>-      logging.error("error writing to file %s (%d, %s)" % (chunk_file, errno,
>-                    strerror))
>-    # Since we're running small chunks of the layout tests, it's important to
>-    # mark the ones that have errors in them.  These won't be visible in the
>-    # summary list for long, but will be useful for someone reviewing this bot.
>-    return self.TestLayoutChunk(chunk_num, chunk_size)
>-
>-  # The known list of tests.
>-  # Recognise the original abbreviations as well as full executable names.
>-  _test_list = {
>-    "cmdline" : RunCmdLine,
>-    "addressinput": TestAddressInput,
>-    "libaddressinput_unittests": TestAddressInput,
>-    "accessibility": TestAccessibility,
>-    "angle": TestAngle,          "angle_unittests": TestAngle,
>-    "app_list": TestAppList,     "app_list_unittests": TestAppList,
>-    "ash": TestAsh,              "ash_unittests": TestAsh,
>-    "aura": TestAura,            "aura_unittests": TestAura,
>-    "base": TestBase,            "base_unittests": TestBase,
>-    "blink_heap": TestBlinkHeap,
>-    "blink_platform": TestBlinkPlatform,
>-    "browser": TestBrowser,      "browser_tests": TestBrowser,
>-    "cacheinvalidation": TestCacheInvalidation,
>-    "cacheinvalidation_unittests": TestCacheInvalidation,
>-    "cast": TestCast,            "cast_unittests": TestCast,
>-    "cc": TestCC,                "cc_unittests": TestCC,
>-    "chrome_app": TestChromeApp,
>-    "chrome_elf": TestChromeElf,
>-    "chromedriver": TestChromeDriver,
>-    "chromeos": TestChromeOS,    "chromeos_unittests": TestChromeOS,
>-    "components": TestComponents,"components_unittests": TestComponents,
>-    "compositor": TestCompositor,"compositor_unittests": TestCompositor,
>-    "content": TestContent,      "content_unittests": TestContent,
>-    "content_browsertests": TestContentBrowser,
>-    "courgette": TestCourgette,  "courgette_unittests": TestCourgette,
>-    "crypto": TestCrypto,        "crypto_unittests": TestCrypto,
>-    "device": TestDevice,        "device_unittests": TestDevice,
>-    "display": TestDisplay,      "display_unittests": TestDisplay,
>-    "events": TestEvents,        "events_unittests": TestEvents,
>-    "extensions": TestExtensions, "extensions_unittests": TestExtensions,
>-    "ffmpeg_regression_tests": TestFFmpegRegressions,
>-    "gcm": TestGCM,              "gcm_unit_tests": TestGCM,
>-    "gin": TestGin,              "gin_unittests": TestGin,
>-    "gfx": TestGfx,              "gfx_unittests": TestGfx,
>-    "google_apis": TestGoogleApis,
>-    "gpu": TestGPU,              "gpu_unittests": TestGPU,
>-    "ipc": TestIpc,              "ipc_tests": TestIpc,
>-    "installer_util": TestInstallerUtil,
>-    "installer_util_unittests": TestInstallerUtil,
>-    "install_static_unittests": TestInstallStatic,
>-    "interactive_ui": TestInteractiveUI,
>-    "jingle": TestJingle,        "jingle_unittests": TestJingle,
>-    "keyboard": TestKeyboard,    "keyboard_unittests": TestKeyboard,
>-    "latency": TestLatency,      "latency_unittests": TestLatency,
>-    "layout": TestLayout,        "layout_tests": TestLayout,
>-    "media": TestMedia,          "media_unittests": TestMedia,
>-    "message_center": TestMessageCenter,
>-    "message_center_unittests" : TestMessageCenter,
>-    "midi": TestMidi,             "midi_unittests": TestMidi,
>-    "mojo_common": TestMojoCommon,
>-    "mojo_common_unittests": TestMojoCommon,
>-    "mojo_system": TestMojoSystem,
>-    "mojo_system_unittests": TestMojoSystem,
>-    "mojo_public_system": TestMojoPublicSystem,
>-    "mojo_public_system_unittests": TestMojoPublicSystem,
>-    "mojo_public_bindings": TestMojoPublicBindings,
>-    "mojo_public_bindings_unittests": TestMojoPublicBindings,
>-    "mojo_public_sysperf": TestMojoPublicSysPerf,
>-    "net": TestNet,              "net_unittests": TestNet,
>-    "net_perf": TestNetPerf,     "net_perftests": TestNetPerf,
>-    "phonenumber": TestPhoneNumber,
>-    "libphonenumber_unittests": TestPhoneNumber,
>-    "ppapi": TestPPAPI,          "ppapi_unittests": TestPPAPI,
>-    "printing": TestPrinting,    "printing_unittests": TestPrinting,
>-    "remoting": TestRemoting,    "remoting_unittests": TestRemoting,
>-    "sandbox": TestLinuxSandbox, "sandbox_linux_unittests": TestLinuxSandbox,
>-    "skia": TestSkia,            "skia_unittests": TestSkia,
>-    "sql": TestSql,              "sql_unittests": TestSql,
>-    "storage": TestStorage,      "storage_unittests": TestStorage,
>-    "sync_integration_tests": TestSyncIntegration,
>-    "sync_integration": TestSyncIntegration,
>-    "ui_base_unit": TestUIBaseUnit,       "ui_base_unittests": TestUIBaseUnit,
>-    "ui_chromeos": TestUIChromeOS, "ui_chromeos_unittests": TestUIChromeOS,
>-    "unit": TestUnit,            "unit_tests": TestUnit,
>-    "url": TestURL,              "url_unittests": TestURL,
>-    "views": TestViews,          "views_unittests": TestViews,
>-    "webkit": TestLayout,
>-  }
>-
>-
>-def _main():
>-  parser = optparse.OptionParser("usage: %prog -b <dir> -t <test> "
>-                                 "[-t <test> ...]")
>-
>-  parser.add_option("--help-tests", dest="help_tests", action="store_true",
>-                    default=False, help="List all available tests")
>-  parser.add_option("-b", "--build-dir",
>-                    help="the location of the compiler output")
>-  parser.add_option("--target", help="Debug or Release")
>-  parser.add_option("-t", "--test", action="append", default=[],
>-                    help="which test to run, supports test:gtest_filter format "
>-                         "as well.")
>-  parser.add_option("--baseline", action="store_true", default=False,
>-                    help="generate baseline data instead of validating")
>-  parser.add_option("-f", "--force", action="store_true", default=False,
>-                    help="run a broken test anyway")
>-  parser.add_option("--gtest_filter",
>-                    help="additional arguments to --gtest_filter")
>-  parser.add_option("--gtest_repeat", help="argument for --gtest_repeat")
>-  parser.add_option("--gtest_shuffle", action="store_true", default=False,
>-                    help="Randomize tests' orders on every iteration.")
>-  parser.add_option("--gtest_break_on_failure", action="store_true",
>-                    default=False,
>-                    help="Drop in to debugger on assertion failure. Also "
>-                         "useful for forcing tests to exit with a stack dump "
>-                         "on the first assertion failure when running with "
>-                         "--gtest_repeat=-1")
>-  parser.add_option("-v", "--verbose", action="store_true", default=False,
>-                    help="verbose output - enable debug log messages")
>-  parser.add_option("--tool", dest="valgrind_tool", default="memcheck",
>-                    help="specify a valgrind tool to run the tests under")
>-  parser.add_option("--tool_flags", dest="valgrind_tool_flags", default="",
>-                    help="specify custom flags for the selected valgrind tool")
>-  parser.add_option("--keep_logs", action="store_true", default=False,
>-                    help="store memory tool logs in the <tool>.logs directory "
>-                         "instead of /tmp.\nThis can be useful for tool "
>-                         "developers/maintainers.\nPlease note that the <tool>"
>-                         ".logs directory will be clobbered on tool startup.")
>-  parser.add_option("-n", "--num_tests", type="int",
>-                    default=ChromeTests.LAYOUT_TESTS_DEFAULT_CHUNK_SIZE,
>-                    help="for layout tests: # of subtests per run.  0 for all.")
>-  parser.add_option("--test-launcher-bot-mode", action="store_true",
>-                    help="run the tests with --test-launcher-bot-mode")
>-  parser.add_option("--test-launcher-total-shards", type=int,
>-                    help="run the tests with --test-launcher-total-shards")
>-  parser.add_option("--test-launcher-shard-index", type=int,
>-                    help="run the tests with --test-launcher-shard-index")
>-  parser.add_option("--drmemory_ops",
>-                    help="extra options passed to Dr. Memory")
>-
>-  options, args = parser.parse_args()
>-
>-  # Bake target into build_dir.
>-  if options.target and options.build_dir:
>-    assert (options.target !=
>-            os.path.basename(os.path.dirname(options.build_dir)))
>-    options.build_dir = os.path.join(os.path.abspath(options.build_dir),
>-                                     options.target)
>-
>-  if options.verbose:
>-    logging_utils.config_root(logging.DEBUG)
>-  else:
>-    logging_utils.config_root()
>-
>-  if options.help_tests:
>-    ChromeTests.ShowTests()
>-    return 0
>-
>-  if not options.test:
>-    parser.error("--test not specified")
>-
>-  if len(options.test) != 1 and options.gtest_filter:
>-    parser.error("--gtest_filter and multiple tests don't make sense together")
>-
>-  BROKEN_TESTS = {
>-    'drmemory_light': [
>-      'addressinput',
>-      'aura',
>-      'base_unittests',
>-      'cc',
>-      'components', # x64 only?
>-      'content',
>-      'gfx',
>-      'mojo_public_bindings',
>-    ],
>-    'drmemory_full': [
>-      'addressinput',
>-      'aura',
>-      'base_unittests',
>-      'blink_heap',
>-      'blink_platform',
>-      'browser_tests',
>-      'cast',
>-      'cc',
>-      'chromedriver',
>-      'compositor',
>-      'content',
>-      'content_browsertests',
>-      'device',
>-      'events',
>-      'extensions',
>-      'gfx',
>-      'google_apis',
>-      'gpu',
>-      'ipc_tests',
>-      'jingle',
>-      'keyboard',
>-      'media',
>-      'midi',
>-      'mojo_common',
>-      'mojo_public_bindings',
>-      'mojo_public_sysperf',
>-      'mojo_public_system',
>-      'mojo_system',
>-      'net',
>-      'remoting',
>-      'unit',
>-      'url',
>-    ],
>-  }
>-
>-  for t in options.test:
>-    if t in BROKEN_TESTS[options.valgrind_tool] and not options.force:
>-      logging.info("Skipping broken %s test %s -- see crbug.com/633693" %
>-                   (options.valgrind_tool, t))
>-      return 0
>-
>-    tests = ChromeTests(options, args, t)
>-    ret = tests.Run()
>-    if ret: return ret
>-  return 0
>-
>-
>-if __name__ == "__main__":
>-  sys.exit(_main())
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/chrome_tests.sh b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/chrome_tests.sh
>deleted file mode 100644
>index dc17684fecc..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/chrome_tests.sh
>+++ /dev/null
>@@ -1,94 +0,0 @@
>-#!/bin/bash
>-
>-# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>-#
>-# Use of this source code is governed by a BSD-style license
>-# that can be found in the LICENSE file in the root of the source
>-# tree. An additional intellectual property rights grant can be found
>-# in the file PATENTS.  All contributing project authors may
>-# be found in the AUTHORS file in the root of the source tree.
>-
>-# Set up some paths and re-direct the arguments to chrome_tests.py
>-
>-export THISDIR=`dirname $0`
>-ARGV_COPY="$@"
>-
>-# We need to set CHROME_VALGRIND iff using Memcheck:
>-#   tools/valgrind/chrome_tests.sh --tool memcheck
>-# or
>-#   tools/valgrind/chrome_tests.sh --tool=memcheck
>-tool="memcheck"  # Default to memcheck.
>-while (( "$#" ))
>-do
>-  if [[ "$1" == "--tool" ]]
>-  then
>-    tool="$2"
>-    shift
>-  elif [[ "$1" =~ --tool=(.*) ]]
>-  then
>-    tool="${BASH_REMATCH[1]}"
>-  fi
>-  shift
>-done
>-
>-NEEDS_VALGRIND=0
>-NEEDS_DRMEMORY=0
>-
>-case "$tool" in
>-  "memcheck")
>-    NEEDS_VALGRIND=1
>-    ;;
>-  "drmemory" | "drmemory_light" | "drmemory_full" | "drmemory_pattern")
>-    NEEDS_DRMEMORY=1
>-    ;;
>-esac
>-
>-if [ "$NEEDS_VALGRIND" == "1" ]
>-then
>-  export CHROME_VALGRIND=`sh $THISDIR/locate_valgrind.sh`
>-  if [ "$CHROME_VALGRIND" = "" ]
>-  then
>-    # locate_valgrind.sh failed
>-    exit 1
>-  fi
>-  echo "Using valgrind binaries from ${CHROME_VALGRIND}"
>-
>-  PATH="${CHROME_VALGRIND}/bin:$PATH"
>-  # We need to set these variables to override default lib paths hard-coded into
>-  # Valgrind binary.
>-  export VALGRIND_LIB="$CHROME_VALGRIND/lib/valgrind"
>-  export VALGRIND_LIB_INNER="$CHROME_VALGRIND/lib/valgrind"
>-
>-  # Clean up some /tmp directories that might be stale due to interrupted
>-  # chrome_tests.py execution.
>-  # FYI:
>-  #   -mtime +1  <- only print files modified more than 24h ago,
>-  #   -print0/-0 are needed to handle possible newlines in the filenames.
>-  echo "Cleanup /tmp from Valgrind stuff"
>-  find /tmp -maxdepth 1 \(\
>-        -name "vgdb-pipe-*" -or -name "vg_logs_*" -or -name "valgrind.*" \
>-      \) -mtime +1 -print0 | xargs -0 rm -rf
>-fi
>-
>-if [ "$NEEDS_DRMEMORY" == "1" ]
>-then
>-  if [ -z "$DRMEMORY_COMMAND" ]
>-  then
>-    DRMEMORY_PATH="$THISDIR/../../third_party/drmemory"
>-    DRMEMORY_SFX="$DRMEMORY_PATH/drmemory-windows-sfx.exe"
>-    if [ ! -f "$DRMEMORY_SFX" ]
>-    then
>-      echo "Can't find Dr. Memory executables."
>-      echo "See http://www.chromium.org/developers/how-tos/using-valgrind/dr-memory"
>-      echo "for the instructions on how to get them."
>-      exit 1
>-    fi
>-
>-    chmod +x "$DRMEMORY_SFX"  # Cygwin won't run it without +x.
>-    "$DRMEMORY_SFX" -o"$DRMEMORY_PATH/unpacked" -y
>-    export DRMEMORY_COMMAND="$DRMEMORY_PATH/unpacked/bin/drmemory.exe"
>-  fi
>-fi
>-
>-PYTHONPATH=$THISDIR/../python/google python \
>-           "$THISDIR/chrome_tests.py" $ARGV_COPY
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/common.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/common.py
>deleted file mode 100644
>index e9ee51e48e5..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/common.py
>+++ /dev/null
>@@ -1,256 +0,0 @@
>-# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>-#
>-# Use of this source code is governed by a BSD-style license
>-# that can be found in the LICENSE file in the root of the source
>-# tree. An additional intellectual property rights grant can be found
>-# in the file PATENTS.  All contributing project authors may
>-# be found in the AUTHORS file in the root of the source tree.
>-
>-import logging
>-import platform
>-import os
>-import signal
>-import subprocess
>-import sys
>-import time
>-
>-
>-class NotImplementedError(Exception):
>-  pass
>-
>-
>-class TimeoutError(Exception):
>-  pass
>-
>-
>-def RunSubprocessInBackground(proc):
>-  """Runs a subprocess in the background. Returns a handle to the process."""
>-  logging.info("running %s in the background" % " ".join(proc))
>-  return subprocess.Popen(proc)
>-
>-
>-def RunSubprocess(proc, timeout=0):
>-  """ Runs a subprocess, until it finishes or |timeout| is exceeded and the
>-  process is killed with taskkill.  A |timeout| <= 0  means no timeout.
>-
>-  Args:
>-    proc: list of process components (exe + args)
>-    timeout: how long to wait before killing, <= 0 means wait forever
>-  """
>-
>-  logging.info("running %s, timeout %d sec" % (" ".join(proc), timeout))
>-  sys.stdout.flush()
>-  sys.stderr.flush()
>-
>-  # Manually read and print out stdout and stderr.
>-  # By default, the subprocess is supposed to inherit these from its parent,
>-  # however when run under buildbot, it seems unable to read data from a
>-  # grandchild process, so we have to read the child and print the data as if
>-  # it came from us for buildbot to read it.  We're not sure why this is
>-  # necessary.
>-  # TODO(erikkay): should we buffer stderr and stdout separately?
>-  p = subprocess.Popen(proc, universal_newlines=True,
>-                       bufsize=0,  # unbuffered
>-                       stdout=subprocess.PIPE, stderr=subprocess.STDOUT)
>-
>-  logging.info("started subprocess")
>-
>-  did_timeout = False
>-  if timeout > 0:
>-    wait_until = time.time() + timeout
>-  while p.poll() is None and not did_timeout:
>-    # Have to use readline rather than readlines() or "for line in p.stdout:",
>-    # otherwise we get buffered even with bufsize=0.
>-    line = p.stdout.readline()
>-    while line and not did_timeout:
>-      sys.stdout.write(line)
>-      sys.stdout.flush()
>-      line = p.stdout.readline()
>-      if timeout > 0:
>-        did_timeout = time.time() > wait_until
>-
>-  if did_timeout:
>-    logging.info("process timed out")
>-  else:
>-    logging.info("process ended, did not time out")
>-
>-  if did_timeout:
>-    if IsWindows():
>-      subprocess.call(["taskkill", "/T", "/F", "/PID", str(p.pid)])
>-    else:
>-      # Does this kill all children, too?
>-      os.kill(p.pid, signal.SIGINT)
>-    logging.error("KILLED %d" % p.pid)
>-    # Give the process a chance to actually die before continuing
>-    # so that cleanup can happen safely.
>-    time.sleep(1.0)
>-    logging.error("TIMEOUT waiting for %s" % proc[0])
>-    raise TimeoutError(proc[0])
>-  else:
>-    for line in p.stdout:
>-      sys.stdout.write(line)
>-    if not IsMac():   # stdout flush fails on Mac
>-      logging.info("flushing stdout")
>-      sys.stdout.flush()
>-
>-  logging.info("collecting result code")
>-  result = p.poll()
>-  if result:
>-    logging.error("%s exited with non-zero result code %d" % (proc[0], result))
>-  return result
>-
>-
>-def IsLinux():
>-  return sys.platform.startswith('linux')
>-
>-
>-def IsMac():
>-  return sys.platform.startswith('darwin')
>-
>-
>-def IsWindows():
>-  return sys.platform == 'cygwin' or sys.platform.startswith('win')
>-
>-
>-def WindowsVersionName():
>-  """Returns the name of the Windows version if it is known, or None.
>-
>-  Possible return values are: xp, vista, 7, 8, or None
>-  """
>-  if sys.platform == 'cygwin':
>-    # Windows version number is hiding in system name.  Looks like:
>-    # CYGWIN_NT-6.1-WOW64
>-    try:
>-      version_str = platform.uname()[0].split('-')[1]
>-    except:
>-      return None
>-  elif sys.platform.startswith('win'):
>-    # Normal Windows version string.  Mine: 6.1.7601
>-    version_str = platform.version()
>-  else:
>-    return None
>-
>-  parts = version_str.split('.')
>-  try:
>-    major = int(parts[0])
>-    minor = int(parts[1])
>-  except:
>-    return None  # Can't parse, unknown version.
>-
>-  if major == 5:
>-    return 'xp'
>-  elif major == 6 and minor == 0:
>-    return 'vista'
>-  elif major == 6 and minor == 1:
>-    return '7'
>-  elif major == 6 and minor == 2:
>-    return '8'  # Future proof.  ;)
>-  return None
>-
>-
>-def PlatformNames():
>-  """Return an array of string to be used in paths for the platform
>-  (e.g. suppressions, gtest filters, ignore files etc.)
>-  The first element of the array describes the 'main' platform
>-  """
>-  if IsLinux():
>-    return ['linux']
>-  if IsMac():
>-    return ['mac']
>-  if IsWindows():
>-    names = ['win32']
>-    version_name = WindowsVersionName()
>-    if version_name is not None:
>-      names.append('win-%s' % version_name)
>-    return names
>-  raise NotImplementedError('Unknown platform "%s".' % sys.platform)
>-
>-
>-def PutEnvAndLog(env_name, env_value):
>-  os.putenv(env_name, env_value)
>-  logging.info('export %s=%s', env_name, env_value)
>-
>-def BoringCallers(mangled, use_re_wildcards):
>-  """Return a list of 'boring' function names (optinally mangled)
>-  with */? wildcards (optionally .*/.).
>-  Boring = we drop off the bottom of stack traces below such functions.
>-  """
>-
>-  need_mangling = [
>-    # Don't show our testing framework:
>-    ("testing::Test::Run",     "_ZN7testing4Test3RunEv"),
>-    ("testing::TestInfo::Run", "_ZN7testing8TestInfo3RunEv"),
>-    ("testing::internal::Handle*ExceptionsInMethodIfSupported*",
>-     "_ZN7testing8internal3?Handle*ExceptionsInMethodIfSupported*"),
>-
>-    # Depend on scheduling:
>-    ("MessageLoop::Run",     "_ZN11MessageLoop3RunEv"),
>-    ("MessageLoop::RunTask", "_ZN11MessageLoop7RunTask*"),
>-    ("RunnableMethod*",      "_ZN14RunnableMethod*"),
>-    ("DispatchToMethod*",    "_Z*16DispatchToMethod*"),
>-    ("base::internal::Invoker*::DoInvoke*",
>-     "_ZN4base8internal8Invoker*DoInvoke*"),  # Invoker{1,2,3}
>-    ("base::internal::RunnableAdapter*::Run*",
>-     "_ZN4base8internal15RunnableAdapter*Run*"),
>-  ]
>-
>-  ret = []
>-  for pair in need_mangling:
>-    ret.append(pair[1 if mangled else 0])
>-
>-  ret += [
>-    # Also don't show the internals of libc/pthread.
>-    "start_thread",
>-    "main",
>-    "BaseThreadInitThunk",
>-  ]
>-
>-  if use_re_wildcards:
>-    for i in range(0, len(ret)):
>-      ret[i] = ret[i].replace('*', '.*').replace('?', '.')
>-
>-  return ret
>-
>-def NormalizeWindowsPath(path):
>-  """If we're using Cygwin Python, turn the path into a Windows path.
>-
>-  Don't turn forward slashes into backslashes for easier copy-pasting and
>-  escaping.
>-
>-  TODO(rnk): If we ever want to cut out the subprocess invocation, we can use
>-  _winreg to get the root Cygwin directory from the registry key:
>-  HKEY_LOCAL_MACHINE\SOFTWARE\Cygwin\setup\rootdir.
>-  """
>-  if sys.platform.startswith("cygwin"):
>-    p = subprocess.Popen(["cygpath", "-m", path],
>-                         stdout=subprocess.PIPE,
>-                         stderr=subprocess.PIPE)
>-    (out, err) = p.communicate()
>-    if err:
>-      logging.warning("WARNING: cygpath error: %s", err)
>-    return out.strip()
>-  else:
>-    return path
>-
>-############################
>-# Common output format code
>-
>-def PrintUsedSuppressionsList(suppcounts):
>-  """ Prints out the list of used suppressions in a format common to all the
>-      memory tools. If the list is empty, prints nothing and returns False,
>-      otherwise True.
>-
>-      suppcounts: a dictionary of used suppression counts,
>-                  Key -> name, Value -> count.
>-  """
>-  if not suppcounts:
>-    return False
>-
>-  print "-----------------------------------------------------"
>-  print "Suppressions used:"
>-  print "  count name"
>-  for (name, count) in sorted(suppcounts.items(), key=lambda (k,v): (v,k)):
>-    print "%7d %s" % (count, name)
>-  print "-----------------------------------------------------"
>-  sys.stdout.flush()
>-  return True
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/gdb_helper.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/gdb_helper.py
>deleted file mode 100644
>index d127f760080..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/gdb_helper.py
>+++ /dev/null
>@@ -1,91 +0,0 @@
>-# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>-#
>-# Use of this source code is governed by a BSD-style license
>-# that can be found in the LICENSE file in the root of the source
>-# tree. An additional intellectual property rights grant can be found
>-# in the file PATENTS.  All contributing project authors may
>-# be found in the AUTHORS file in the root of the source tree.
>-
>-''' A bunch of helper functions for querying gdb.'''
>-
>-import logging
>-import os
>-import re
>-import tempfile
>-
>-GDB_LINE_RE = re.compile(r'Line ([0-9]*) of "([^"]*)".*')
>-
>-def _GdbOutputToFileLine(output_line):
>-  ''' Parse the gdb output line, return a pair (file, line num) '''
>-  match =  GDB_LINE_RE.match(output_line)
>-  if match:
>-    return match.groups()[1], match.groups()[0]
>-  else:
>-    return None
>-
>-def ResolveAddressesWithinABinary(binary_name, load_address, address_list):
>-  ''' For each address, return a pair (file, line num) '''
>-  commands = tempfile.NamedTemporaryFile()
>-  commands.write('add-symbol-file "%s" %s\n' % (binary_name, load_address))
>-  for addr in address_list:
>-    commands.write('info line *%s\n' % addr)
>-  commands.write('quit\n')
>-  commands.flush()
>-  gdb_commandline = 'gdb -batch -x %s 2>/dev/null' % commands.name
>-  gdb_pipe = os.popen(gdb_commandline)
>-  result = gdb_pipe.readlines()
>-
>-  address_count = 0
>-  ret = {}
>-  for line in result:
>-    if line.startswith('Line'):
>-      ret[address_list[address_count]] = _GdbOutputToFileLine(line)
>-      address_count += 1
>-    if line.startswith('No line'):
>-      ret[address_list[address_count]] = (None, None)
>-      address_count += 1
>-  gdb_pipe.close()
>-  commands.close()
>-  return ret
>-
>-class AddressTable(object):
>-  ''' Object to do batched line number lookup. '''
>-  def __init__(self):
>-    self._load_addresses = {}
>-    self._binaries = {}
>-    self._all_resolved = False
>-
>-  def AddBinaryAt(self, binary, load_address):
>-    ''' Register a new shared library or executable. '''
>-    self._load_addresses[binary] = load_address
>-
>-  def Add(self, binary, address):
>-    ''' Register a lookup request. '''
>-    if binary == '':
>-      logging.warn('adding address %s in empty binary?' % address)
>-    if binary in self._binaries:
>-      self._binaries[binary].append(address)
>-    else:
>-      self._binaries[binary] = [address]
>-    self._all_resolved = False
>-
>-  def ResolveAll(self):
>-    ''' Carry out all lookup requests. '''
>-    self._translation = {}
>-    for binary in self._binaries.keys():
>-      if binary != '' and binary in self._load_addresses:
>-        load_address = self._load_addresses[binary]
>-        addr = ResolveAddressesWithinABinary(
>-            binary, load_address, self._binaries[binary])
>-        self._translation[binary] = addr
>-    self._all_resolved = True
>-
>-  def GetFileLine(self, binary, addr):
>-    ''' Get the (filename, linenum) result of a previously-registered lookup
>-    request.
>-    '''
>-    if self._all_resolved:
>-      if binary in self._translation:
>-        if addr in self._translation[binary]:
>-          return self._translation[binary][addr]
>-    return (None, None)
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/gtest_exclude/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/gtest_exclude/OWNERS
>deleted file mode 100644
>index 241e560df69..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/gtest_exclude/OWNERS
>+++ /dev/null
>@@ -1,2 +0,0 @@
>-*
>-
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/gtest_exclude/modules_tests.gtest-memcheck.txt b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/gtest_exclude/modules_tests.gtest-memcheck.txt
>deleted file mode 100644
>index efa4be59515..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/gtest_exclude/modules_tests.gtest-memcheck.txt
>+++ /dev/null
>@@ -1,9 +0,0 @@
>-# Tests that are too slow.
>-AudioCodingModuleTest.TestAllCodecs*
>-AudioCodingModuleTest.TestEncodeDecode*
>-AudioCodingModuleTest.TestFEC*
>-AudioCodingModuleTest.TestIsac*
>-AudioCodingModuleTest.TwoWayCommunication*
>-AudioCodingModuleTest.TestStereo*
>-AudioCodingModuleTest.TestVADDTX*
>-FecTest.FecTest
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/gtest_exclude/modules_unittests.gtest-memcheck.txt b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/gtest_exclude/modules_unittests.gtest-memcheck.txt
>deleted file mode 100644
>index 8588487ae46..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/gtest_exclude/modules_unittests.gtest-memcheck.txt
>+++ /dev/null
>@@ -1,2 +0,0 @@
>-# Tests that are too slow.
>-CommonFormats/AudioProcessingTest*
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/gtest_exclude/ortc_unittests.gtest-memcheck.txt b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/gtest_exclude/ortc_unittests.gtest-memcheck.txt
>deleted file mode 100644
>index 50e032b61c3..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/gtest_exclude/ortc_unittests.gtest-memcheck.txt
>+++ /dev/null
>@@ -1,3 +0,0 @@
>-# Tests that are failing when run under memcheck.
>-# https://code.google.com/p/webrtc/issues/detail?id=4387
>-OrtcFactoryIntegrationTest.*
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/gtest_exclude/peerconnection_unittests.gtest-memcheck.txt b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/gtest_exclude/peerconnection_unittests.gtest-memcheck.txt
>deleted file mode 100644
>index 65a1752090b..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/gtest_exclude/peerconnection_unittests.gtest-memcheck.txt
>+++ /dev/null
>@@ -1,7 +0,0 @@
>-# Tests that are failing when run under memcheck.
>-# https://code.google.com/p/webrtc/issues/detail?id=4387
>-DtmfSenderTest.*
>-PeerConnectionEndToEndTest.*
>-PeerConnectionIntegrationTest.*
>-PeerConnectionInterfaceTest.*
>-RTCStatsIntegrationTest.*
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/gtest_exclude/rtc_media_unittests.gtest-memcheck.txt b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/gtest_exclude/rtc_media_unittests.gtest-memcheck.txt
>deleted file mode 100644
>index 2ee09de77c4..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/gtest_exclude/rtc_media_unittests.gtest-memcheck.txt
>+++ /dev/null
>@@ -1,11 +0,0 @@
>-#TODO(wu): https://code.google.com/p/webrtc/issues/detail?id=2380
>-WebRtcVideoMediaChannelTest.TwoStreamsSendAndUnsignalledRecv
>-
>-#TODO(jiayl): https://code.google.com/p/webrtc/issues/detail?id=3492
>-SctpDataMediaChannelTest.*
>-
>-# TODO(pbos): https://code.google.com/p/webrtc/issues/detail?id=4963
>-WebRtcVideoChannel2BaseTest.*
>-
>-#TODO(honghaiz): https://bugs.chromium.org/p/webrtc/issues/detail?id=7421
>-P2PTransportChannelMultihomedTest.TestContinualGatheringOnNewInterface
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/gtest_exclude/rtc_media_unittests.gtest-memcheck_mac.txt b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/gtest_exclude/rtc_media_unittests.gtest-memcheck_mac.txt
>deleted file mode 100644
>index 05ce6180fd9..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/gtest_exclude/rtc_media_unittests.gtest-memcheck_mac.txt
>+++ /dev/null
>@@ -1 +0,0 @@
>-WebRtcVideoMediaChannelTest.AddRemoveCapturerMultipleSources
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/gtest_exclude/rtc_pc_unittests.gtest-memcheck.txt b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/gtest_exclude/rtc_pc_unittests.gtest-memcheck.txt
>deleted file mode 100644
>index 32978e8e6f4..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/gtest_exclude/rtc_pc_unittests.gtest-memcheck.txt
>+++ /dev/null
>@@ -1,8 +0,0 @@
>-# TODO(wu): Reenable P2PTransportChannelTest after issue 2158 is resolved.
>-P2PTransportChannelTest.*
>-P2PTransportChannelSameNatTest.TestConesBehindSameCone
>-PortTest.TestSendStunMessageAsIce
>-PseudoTcpTest.TestSendBothUseLargeWindowScale
>-# Issue 3447
>-P2PTransportChannelMultihomedTest.TestFailover
>-P2PTransportChannelMultihomedTest.TestDrain
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/gtest_exclude/rtc_unittests.gtest-memcheck.txt b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/gtest_exclude/rtc_unittests.gtest-memcheck.txt
>deleted file mode 100644
>index 4cace192ef2..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/gtest_exclude/rtc_unittests.gtest-memcheck.txt
>+++ /dev/null
>@@ -1,15 +0,0 @@
>-ThreadTest.Release
>-# TODO(wu): Reenable P2PTransportChannelTest after issue 2158 is resolved.
>-P2PTransportChannelTest.*
>-P2PTransportChannelSameNatTest.TestConesBehindSameCone
>-PortTest.TestSendStunMessageAsIce
>-PseudoTcpTest.TestSendBothUseLargeWindowScale
>-# Issue 3447
>-P2PTransportChannelMultihomedTest.TestFailover
>-P2PTransportChannelMultihomedTest.TestDrain
>-# Issue 4743
>-PortAllocatorTest.TestDisableAllPorts
>-PortAllocatorTest.TestGetAllPortsNoUdpSockets
>-PortAllocatorTest.TestGetAllPortsNoUdpSocketsNoTcpListen
>-# Issue 5981
>-TurnPortTest.*
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/gtest_exclude/video_engine_tests.gtest-memcheck.txt b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/gtest_exclude/video_engine_tests.gtest-memcheck.txt
>deleted file mode 100644
>index 87ec10dcbd9..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/gtest_exclude/video_engine_tests.gtest-memcheck.txt
>+++ /dev/null
>@@ -1,18 +0,0 @@
>-# Fails when run under memcheck
>-EndToEndTest.CanSwitchToUseAllSsrcs
>-EndToEndTest.SendsAndReceivesVP9
>-TestWithNewVideoJitterBuffer/EndToEndTest.SendsAndReceivesH264/*
>-VideoSendStreamTest.VP9FlexMode
>-EndToEndTest.InitialProbing
>-EndToEndTest.TriggerMidCallProbing
>-# Times out due to using a real VP8 encoder.
>-*EndToEndTest.VerifyHistogramStatsWithRed*
>-
>-# https://bugs.chromium.org/p/webrtc/issues/detail?id=5134
>-*EndToEndTest.AssignsTransportSequenceNumbers*
>-
>-# https://bugs.chromium.org/p/webrtc/issues/detail?id=6886
>-VideoSendStreamTest.RemoveOverheadFromBandwidth
>-
>-# https://bugs.chromium.org/p/webrtc/issues/detail?id=7145
>-*EndToEndTest.VerifyNackStats*
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/gtest_exclude/webrtc_nonparallel_tests.gtest-memcheck.txt b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/gtest_exclude/webrtc_nonparallel_tests.gtest-memcheck.txt
>deleted file mode 100644
>index 0f866b780b5..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/gtest_exclude/webrtc_nonparallel_tests.gtest-memcheck.txt
>+++ /dev/null
>@@ -1,8 +0,0 @@
>-# Fails when run under memcheck
>-# https://code.google.com/p/webrtc/issues/detail?id=4958
>-PhysicalSocketTest.TestUdpReadyToSendIPv4
>-PhysicalSocketTest.TestUdpIPv4
>-PhysicalSocketTest.TestSocketRecvTimestampIPv4
>-# Relies on actual CPU timing, so can be flaky on memcheck.
>-CpuTimeTest.TwoThreads
>-CpuTimeTest.Sleeping
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/locate_valgrind.sh b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/locate_valgrind.sh
>deleted file mode 100644
>index d9594f48483..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/locate_valgrind.sh
>+++ /dev/null
>@@ -1,73 +0,0 @@
>-#!/bin/bash
>-
>-# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>-#
>-# Use of this source code is governed by a BSD-style license
>-# that can be found in the LICENSE file in the root of the source
>-# tree. An additional intellectual property rights grant can be found
>-# in the file PATENTS.  All contributing project authors may
>-# be found in the AUTHORS file in the root of the source tree.
>-
>-# Prints a path to Valgrind binaries to be used for Chromium.
>-# Select the valgrind from third_party/valgrind by default,
>-# but allow users to override this default without editing scripts and
>-# without specifying a commandline option
>-
>-export THISDIR=`dirname $0`
>-
>-# User may use their own valgrind by giving its path with CHROME_VALGRIND env.
>-if [ "$CHROME_VALGRIND" = "" ]
>-then
>-  # Guess which binaries we should use by uname
>-  case "$(uname -a)" in
>-  *Linux*x86_64*)
>-    PLATFORM="linux_x64"
>-    ;;
>-  *Linux*86*)
>-    PLATFORM="linux_x86"
>-    ;;
>-  *Darwin*9.[678].[01]*i386*)
>-    # Didn't test other kernels.
>-    PLATFORM="mac"
>-    ;;
>-  *Darwin*10.[0-9].[0-9]*i386*)
>-    PLATFORM="mac_10.6"
>-    ;;
>-  *Darwin*10.[0-9].[0-9]*x86_64*)
>-    PLATFORM="mac_10.6"
>-    ;;
>-  *Darwin*11.[0-9].[0-9]*x86_64*)
>-    PLATFORM="mac_10.7"
>-    ;;
>-  *)
>-    (echo "Sorry, your platform is not supported:" &&
>-     uname -a
>-     echo
>-     echo "If you're on Mac OS X, please see http://crbug.com/441425") >&2
>-    exit 42
>-  esac
>-
>-  # The binaries should be in third_party/valgrind
>-  # (checked out from deps/third_party/valgrind/binaries).
>-  CHROME_VALGRIND="$THISDIR/../../third_party/valgrind/$PLATFORM"
>-
>-  # TODO(timurrrr): readlink -f is not present on Mac...
>-  if [ "$PLATFORM" != "mac" ] && \
>-    [ "$PLATFORM" != "mac_10.6" ] && \
>-    [ "$PLATFORM" != "mac_10.7" ]
>-  then
>-    # Get rid of all "../" dirs
>-    CHROME_VALGRIND=$(readlink -f $CHROME_VALGRIND)
>-  fi
>-fi
>-
>-if ! test -x $CHROME_VALGRIND/bin/valgrind
>-then
>-  echo "Oops, could not find Valgrind binaries in your checkout." >&2
>-  echo "Please see" >&2
>-  echo "  http://dev.chromium.org/developers/how-tos/using-valgrind/get-valgrind" >&2
>-  echo "for the instructions on how to download pre-built binaries." >&2
>-  exit 1
>-fi
>-
>-echo $CHROME_VALGRIND
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/memcheck/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/memcheck/OWNERS
>deleted file mode 100644
>index 72e8ffc0db8..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/memcheck/OWNERS
>+++ /dev/null
>@@ -1 +0,0 @@
>-*
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/memcheck/PRESUBMIT.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/memcheck/PRESUBMIT.py
>deleted file mode 100644
>index 927302d7df5..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/memcheck/PRESUBMIT.py
>+++ /dev/null
>@@ -1,100 +0,0 @@
>-#!/usr/bin/env python
>-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>-#
>-# Use of this source code is governed by a BSD-style license
>-# that can be found in the LICENSE file in the root of the source
>-# tree. An additional intellectual property rights grant can be found
>-# in the file PATENTS.  All contributing project authors may
>-# be found in the AUTHORS file in the root of the source tree.
>-
>-"""
>-Copied from Chrome's src/tools/valgrind/memcheck/PRESUBMIT.py
>-
>-See http://dev.chromium.org/developers/how-tos/depottools/presubmit-scripts
>-for more details on the presubmit API built into gcl.
>-"""
>-
>-import os
>-import re
>-import sys
>-
>-def CheckChange(input_api, output_api):
>-  """Checks the memcheck suppressions files for bad data."""
>-
>-  # Add the path to the Chrome valgrind dir to the import path:
>-  tools_vg_path = os.path.join(input_api.PresubmitLocalPath(), '..', '..', '..',
>-                               'tools', 'valgrind')
>-  sys.path.append(tools_vg_path)
>-  import suppressions
>-
>-  sup_regex = re.compile('suppressions.*\.txt$')
>-  suppressions = {}
>-  errors = []
>-  check_for_memcheck = False
>-  # skip_next_line has 3 possible values:
>-  # - False: don't skip the next line.
>-  # - 'skip_suppression_name': the next line is a suppression name, skip.
>-  # - 'skip_param': the next line is a system call parameter error, skip.
>-  skip_next_line = False
>-  for f in filter(lambda x: sup_regex.search(x.LocalPath()),
>-                  input_api.AffectedFiles()):
>-    for line, line_num in zip(f.NewContents(),
>-                              xrange(1, len(f.NewContents()) + 1)):
>-      line = line.lstrip()
>-      if line.startswith('#') or not line:
>-        continue
>-
>-      if skip_next_line:
>-        if skip_next_line == 'skip_suppression_name':
>-          if 'insert_a_suppression_name_here' in line:
>-            errors.append('"insert_a_suppression_name_here" is not a valid '
>-                          'suppression name')
>-          if suppressions.has_key(line):
>-            if f.LocalPath() == suppressions[line][1]:
>-              errors.append('suppression with name "%s" at %s line %s '
>-                            'has already been defined at line %s' %
>-                            (line, f.LocalPath(), line_num,
>-                             suppressions[line][1]))
>-            else:
>-              errors.append('suppression with name "%s" at %s line %s '
>-                            'has already been defined at %s line %s' %
>-                            (line, f.LocalPath(), line_num,
>-                             suppressions[line][0], suppressions[line][1]))
>-          else:
>-            suppressions[line] = (f, line_num)
>-            check_for_memcheck = True;
>-        skip_next_line = False
>-        continue
>-      if check_for_memcheck:
>-        if not line.startswith('Memcheck:'):
>-          errors.append('"%s" should be "Memcheck:..." in %s line %s' %
>-                        (line, f.LocalPath(), line_num))
>-        check_for_memcheck = False;
>-      if line == '{':
>-        skip_next_line = 'skip_suppression_name'
>-        continue
>-      if line == "Memcheck:Param":
>-        skip_next_line = 'skip_param'
>-        continue
>-
>-      if (line.startswith('fun:') or line.startswith('obj:') or
>-          line.startswith('Memcheck:') or line == '}' or
>-          line == '...'):
>-        continue
>-      errors.append('"%s" is probably wrong: %s line %s' % (line, f.LocalPath(),
>-                                                            line_num))
>-  if errors:
>-    return [output_api.PresubmitError('\n'.join(errors))]
>-  return []
>-
>-
>-def CheckChangeOnUpload(input_api, output_api):
>-  return CheckChange(input_api, output_api)
>-
>-
>-def CheckChangeOnCommit(input_api, output_api):
>-  return CheckChange(input_api, output_api)
>-
>-
>-def GetPreferredTrySlaves():
>-  return ['linux_memcheck']
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/memcheck/suppressions.txt b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/memcheck/suppressions.txt
>deleted file mode 100644
>index 75b1ce62caf..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/memcheck/suppressions.txt
>+++ /dev/null
>@@ -1,460 +0,0 @@
>-# This file is used in addition to the one already maintained in Chrome.
>-# It acts as a place holder for future additions for WebRTC.
>-# It must exist for the Python wrapper script to work properly.
>-
>-# There are three kinds of suppressions in this file.
>-# 1. third_party libraries
>-# 2. webrtc stuff
>-# 3. libjingle stuff (talk folder)
>-#-----------------------------------------------------------------------
>-
>-# third_party libraries
>-
>-{
>-   libc_free_hitting_gce_vms
>-   Memcheck:Free
>-   fun:free
>-   fun:__libc_freeres
>-   fun:_vgnU_freeres
>-   ...
>-}
>-{
>-   Ubuntu_Precise_Itoa_Optimized_Code
>-   Memcheck:Uninitialized
>-   fun:_itoa_word
>-   fun:vfprintf
>-   fun:vsnprintf
>-   fun:snprintf
>-   ...
>-}
>-# See: https://bugs.chromium.org/p/webrtc/issues/detail?id=7973
>-{
>-   libc_getaddrinfo_uninitialized
>-   Memcheck:Param
>-   sendmsg(mmsg[0].msg_hdr)
>-   fun:sendmmsg
>-   fun:__libc_res_nsend
>-   fun:__libc_res_nquery
>-   fun:__libc_res_nsearch
>-   fun:_nss_dns_gethostbyname4_r
>-   fun:gaih_inet
>-   fun:getaddrinfo
>-   ...
>-}
>-
>-#-----------------------------------------------------------------------
>-# webrtc stuff
>-
>-{
>-   bug_6784
>-   Memcheck:Leak
>-   fun:_Znw*
>-   fun:_ZN3rtc10HttpServer16HandleConnectionEPNS_15StreamInterfaceE
>-   fun:_ZN3rtc12_GLOBAL__N_122CreateClientConnectionERNS_10HttpServerERNS0_17HttpServerMonitorEb
>-   fun:_ZN3rtc47HttpServer_SignalsCloseAfterForcedCloseAll_Test8TestBodyEv
>-}
>-{
>-   bug_6773
>-   Memcheck:Uninitialized
>-   fun:vfprintf
>-   fun:vsnprintf
>-   fun:snprintf
>-   ...
>-}
>-{
>-   bug_3446
>-   Memcheck:Uninitialized
>-   fun:vp8cx_pick_filter_level_fast
>-   fun:vp8_loopfilter_frame
>-   fun:encode_frame_to_data_rate
>-   fun:vp8_get_compressed_data
>-   fun:vp8e_encode
>-   fun:vpx_codec_encode
>-   ...
>-   fun:_ZN6webrtc4test18VideoProcessorImpl12ProcessFrameEi
>-   fun:_ZN6webrtc29VideoProcessorIntegrationTest22ProcessFramesAndVerifyENS_14QualityMetricsENS_11RateProfileENS_15CodecConfigParsEPNS_18RateControlMetricsE
>-   fun:_ZN6webrtc70VideoProcessorIntegrationTest_ProcessNoLossSpatialResizeFrameDrop_Test8TestBodyEv
>-}
>-{
>-   bug_1976_1
>-   Memcheck:Unaddressable
>-   fun:pthread_mutex_unlock
>-   fun:_ZN9rtc15CriticalSection5LeaveEv
>-   fun:_ZN9rtc9CritScopeD1Ev
>-   ...
>-   fun:_ZN9rtc6Thread15ProcessMessagesEi
>-   fun:_ZN9rtc6Thread3RunEv
>-   fun:_ZN9rtc6Thread6PreRunEPv
>-}
>-{
>-   bug_2100_3
>-   Memcheck:Uninitialized
>-   fun:tls1_enc
>-   fun:ssl3_get_record
>-   fun:ssl3_read_bytes
>-   fun:ssl3_read_internal
>-   fun:ssl3_read
>-   fun:SSL_read
>-   fun:_ZN9rtc20OpenSSLStreamAdapter4ReadEPvmPmPi
>-   ...
>-}
>-{
>-   bug_2100_4
>-   Memcheck:Uninitialized
>-   fun:_ZN7testing8internal11CmpHelperEQIjhEENS_15AssertionResultEPKcS4_RKT_RKT0_
>-   fun:_ZN7testing8internal8EqHelperILb0EE7CompareIjhEENS_15AssertionResultEPKcS6_RKT_RKT0_
>-   fun:_ZN24SSLStreamAdapterTestDTLS8ReadDataEPN9rtc15StreamInterfaceE
>-   ...
>-}
>-{
>-   bug_2100_5
>-   Memcheck:Uninitialized
>-   fun:dtls1_process_record
>-   fun:dtls1_get_record
>-   fun:dtls1_read_bytes
>-   fun:ssl3_read_internal
>-   fun:ssl3_read
>-   fun:SSL_read
>-   fun:_ZN9rtc20OpenSSLStreamAdapter4ReadEPvmPmPi
>-   ...
>-}
>-{
>-   BIO_new_mem_buf_1
>-   Memcheck:Leak
>-   fun:malloc
>-   fun:default_malloc_ex
>-   fun:CRYPTO_malloc
>-   fun:BUF_MEM_new
>-   fun:mem_new
>-   fun:BIO_set
>-   fun:BIO_new
>-   fun:BIO_new_mem_buf
>-   fun:_ZN9rtc18OpenSSLCertificate13FromPEMStringERKSs
>-   ...
>-}
>-{
>-   BIO_new_mem_buf_2
>-   Memcheck:Leak
>-   fun:malloc
>-   fun:default_malloc_ex
>-   fun:CRYPTO_malloc
>-   fun:BUF_MEM_new
>-   fun:mem_new
>-   fun:BIO_set
>-   fun:BIO_new
>-   fun:BIO_new_mem_buf
>-   fun:_ZN9rtc15OpenSSLIdentity14FromPEMStringsERKSsS2_
>-}
>-{
>-   SignalsCloseAfterForcedCloseAll
>-   Memcheck:Leak
>-   fun:_Znw*
>-   fun:_ZN3rtc10HttpServer10Connection12BeginProcessEPNS_15StreamInterfaceE
>-   ...
>-}
>-{
>-   SignalsCloseAfterForcedCloseAll2
>-   Memcheck:Leak
>-   fun:_Znw*
>-   fun:_ZN9__gnu_cxx13new_allocatorISt13_Rb_tree_nodeISt4pairIKSsSsEEE8allocateEmPKv
>-   fun:_ZNSt8_Rb_treeISsSt4pairIKSsSsESt10_Select1stIS2_EN3rtc5ilessESaIS2_EE11_M_get_nodeEv
>-   fun:_ZNSt8_Rb_treeISsSt4pairIKSsSsESt10_Select1stIS2_EN3rtc5ilessESaIS2_EE14_M_create_nodeIJS2_EEEPSt13_Rb_tree_nodeIS2_EDpOT_
>-   fun:_ZNSt8_Rb_treeISsSt4pairIKSsSsESt10_Select1stIS2_EN3rtc5ilessESaIS2_EE10_M_insert_IS2_EESt17_Rb_tree_iteratorIS2_EPKSt18_Rb_tree_node_baseSE_OT_
>-   fun:_ZNSt8_Rb_treeISsSt4pairIKSsSsESt10_Select1stIS2_EN3rtc5ilessESaIS2_EE15_M_insert_equalIS2_EESt17_Rb_tree_iteratorIS2_EOT_
>-   fun:_ZNSt8multimapISsSsN3rtc5ilessESaISt4pairIKSsSsEEE6insertIS4_vEESt17_Rb_tree_iteratorIS4_EOT_
>-   fun:_ZN3rtc8HttpData12changeHeaderERKSsS2_NS0_13HeaderCombineE
>-   fun:_ZN3rtc8HttpData9setHeaderERKSsS2_b
>-   fun:_ZN3rtc8HttpData9setHeaderENS_10HttpHeaderERKSsb
>-   ...
>-}
>-{
>-   DoNotDeleteTask2
>-   Memcheck:Leak
>-   fun:_Znw*
>-   ...
>-   fun:_ZN3rtc41unstarted_task_test_DoNotDeleteTask2_Test8TestBodyEv
>-}
>-{
>-   bug_329_1
>-   Memcheck:Unaddressable
>-   fun:I422ToARGBRow_SSSE3
>-   fun:I420ToARGB
>-   fun:ConvertFromI420
>-   fun:_ZN6webrtc15ConvertFromI420EPKhiNS_9VideoTypeEiiiPh
>-   fun:_ZN6webrtc15VideoX11Channel12DeliverFrameEPhij
>-   fun:_ZN6webrtc15VideoX11Channel11RenderFrameEjRNS_10VideoFrameE
>-   fun:_ZN6webrtc19IncomingVideoStream26IncomingVideoStreamProcessEv
>-   fun:_ZN6webrtc19IncomingVideoStream28IncomingVideoStreamThreadFunEPv
>-   fun:_ZN6webrtc11ThreadPosix3RunEv
>-   fun:StartThread
>-}
>-
>-{
>-   bug_329_3
>-   Memcheck:Unaddressable
>-   fun:I422ToARGBRow_SSSE3
>-   fun:I420ToARGB
>-   fun:ConvertFromI420
>-   fun:_ZN6webrtc15ConvertFromI420EPKhiNS_9VideoTypeEiiiPh
>-   fun:_ZN6webrtc15VideoX11Channel12DeliverFrameEPhij
>-   fun:_ZN6webrtc15VideoX11Channel11RenderFrameEjRNS_10VideoFrameE
>-   fun:_ZN6webrtc19IncomingVideoStream26IncomingVideoStreamProcessEv
>-   fun:_ZN6webrtc19IncomingVideoStream28IncomingVideoStreamThreadFunEPv
>-   fun:_ZN6webrtc11ThreadPosix3RunEv
>-   fun:StartThread
>-}
>-
>-{
>-   bug_329_7
>-   Memcheck:Unaddressable
>-   fun:I422ToARGBRow_SSSE3
>-   fun:I420ToARGB
>-   fun:ConvertFromI420
>-   fun:_ZN6webrtc15ConvertFromI420ERKNS_10VideoFrameEiNS_9VideoTypeEiPh
>-   fun:_ZN6webrtc15VideoX11Channel12DeliverFrameERKNS_10VideoFrameE
>-   fun:_ZN6webrtc15VideoX11Channel11RenderFrameEjRNS_10VideoFrameE
>-   fun:_ZN6webrtc19IncomingVideoStream26IncomingVideoStreamProcessEv
>-   fun:_ZN6webrtc19IncomingVideoStream28IncomingVideoStreamThreadFunEPv
>-   fun:_ZN6webrtc11ThreadPosix3RunEv
>-   fun:StartThread
>-}
>-
>-{
>-   bug_329_8
>-   Memcheck:Unaddressable
>-   fun:I422ToARGBRow_SSSE3
>-   fun:I420ToARGB
>-   fun:ConvertFromI420
>-   fun:_ZN6webrtc15ConvertFromI420ERKNS_10VideoFrameEiNS_9VideoTypeEiPh
>-   fun:_ZN6webrtc15VideoX11Channel12DeliverFrameERKNS_10VideoFrameE
>-   fun:_ZN6webrtc15VideoX11Channel11RenderFrameEjRNS_10VideoFrameE
>-   fun:_ZN6webrtc19IncomingVideoStream26IncomingVideoStreamProcessEv
>-   fun:_ZN6webrtc19IncomingVideoStream28IncomingVideoStreamThreadFunEPv
>-   fun:_ZN6webrtc11ThreadPosix3RunEv
>-   fun:StartThread
>-}
>-
>-{
>-   bug_898_1
>-   Memcheck:Uninitialized
>-   ...
>-   fun:_ZN16FakeMediaProcess7ProcessEiN6webrtc15ProcessingTypesEPsiib
>-   ...
>-   fun:_ZN6webrtc21AudioDeviceLinuxPulse17PlayThreadProcessEv
>-   fun:_ZN6webrtc21AudioDeviceLinuxPulse14PlayThreadFuncEPv
>-   fun:_ZN6webrtc11ThreadPosix3RunEv
>-   fun:StartThread
>-}
>-
>-{
>-   bug_898_2
>-   Memcheck:Uninitialized
>-   ...
>-   fun:_ZN16FakeMediaProcess7ProcessEiN6webrtc15ProcessingTypesEPsiib
>-   ...
>-   fun:_ZN6webrtc21AudioDeviceLinuxPulse16RecThreadProcessEv
>-   fun:_ZN6webrtc21AudioDeviceLinuxPulse13RecThreadFuncEPv
>-   fun:_ZN6webrtc11ThreadPosix3RunEv
>-   fun:StartThread
>-}
>-
>-{
>-   bug_891
>-   Memcheck:Unaddressable
>-   fun:XShmPutImage
>-   fun:_ZN6webrtc15VideoX11Channel12DeliverFrameEPhij
>-   fun:_ZN6webrtc15VideoX11Channel11RenderFrameEjRNS_10VideoFrameE
>-   fun:_ZN6webrtc19IncomingVideoStream26IncomingVideoStreamProcessEv
>-   fun:_ZN6webrtc19IncomingVideoStream28IncomingVideoStreamThreadFunEPv
>-   fun:_ZN6webrtc11ThreadPosix3RunEv
>-   fun:StartThread
>-}
>-
>-{
>-   ignore_common_benevolent_trace_library_errors
>-   Memcheck:Uninitialized
>-   ...
>-   fun:vsnprintf
>-   fun:_ZN6webrtc5Trace3AddENS_10TraceLevelENS_11TraceModuleEiPKcz
>-   ...
>-}
>-
>-{
>-   bug_270_1
>-   Memcheck:Uninitialized
>-   ...
>-   fun:WebRtcNetEQ_RecOutInternal
>-   fun:WebRtcNetEQ_RecOut
>-   ...
>-}
>-
>-{
>-   bug_270_2
>-   Memcheck:Uninitialized
>-   fun:WebRtcNetEQ_GetSpeechOutputType
>-   ...
>-}
>-
>-
>-#-----------------------------------------------------------------------
>-# libjingle stuff (talk folder)
>-{
>-   bug_2056_1
>-   Memcheck:Leak
>-   fun:malloc
>-   fun:_dl_close_worker
>-   fun:_dl_close
>-   fun:_dl_catch_error
>-   fun:_dlerror_run
>-   fun:dlclose
>-   ...
>-   fun:snd_device_name_hint
>-   fun:_ZNK6webrtc20AudioDeviceLinuxALSA14GetDevicesInfoEibiPci
>-   ...
>-}
>-{
>-   bug_2100_1
>-   Memcheck:Uninitialized
>-   fun:_ZN7testing8internal11CmpHelperEQIiiEENS_15AssertionResultEPKcS4_RKT_RKT0_
>-   fun:_ZN7testing8internal8EqHelperILb1EE7CompareIiiEENS_15AssertionResultEPKcS6_RKT_RKT0_PNS0_8EnableIfIXntsrNS0_10is_pointerISA_EE5valueEE4typeE
>-   ...
>-}
>-{
>-   bug_2100_2
>-   Memcheck:Uninitialized
>-   fun:bcmp
>-   fun:_ZN23SSLStreamAdapterTestTLS12TestTransferEi
>-   ...
>-}
>-# For tests that create XMPPClient which can only be deleted by task runner.
>-{
>-   HangoutPubSubClientTest
>-   Memcheck:Leak
>-   fun:_Znw*
>-   fun:_ZN*HangoutPubSubClientTest*Ev
>-}
>-{
>-   MucRoomConfigTaskTest
>-   Memcheck:Leak
>-   fun:_Znw*
>-   fun:_ZN21MucRoomConfigTaskTest5SetUpEv
>-}
>-{
>-   MucRoomDiscoveryTaskTest
>-   Memcheck:Leak
>-   fun:_Znw*
>-   fun:_ZN24MucRoomDiscoveryTaskTest5SetUpEv
>-}
>-{
>-   MucRoomLookupTaskTest
>-   Memcheck:Leak
>-   fun:_Znw*
>-   fun:_ZN21MucRoomLookupTaskTest5SetUpEv
>-}
>-{
>-   MucRoomUniqueHangoutIdTaskTest
>-   Memcheck:Leak
>-   fun:_Znw*
>-   fun:_ZN30MucRoomUniqueHangoutIdTaskTest5SetUpEv
>-}
>-{
>-   PingTaskTest
>-   Memcheck:Leak
>-   fun:_Znw*
>-   fun:_ZN12PingTaskTest5SetUpEv
>-}
>-{
>-   PubSubClientTest
>-   Memcheck:Leak
>-   fun:_Znw*
>-   fun:_ZN*PubSubClientTest*Ev
>-}
>-{
>-   PubSubTasksTest
>-   Memcheck:Leak
>-   fun:_Znw*
>-   fun:_ZN*PubSubTasksTest*Ev
>-}
>-# failures which can't be reproduced locally
>-{
>-   bug_2153_1
>-   Memcheck:Leak
>-   fun:calloc
>-   ...
>-   fun:_ZN6webrtc26AudioMixerManagerLinuxALSA11OpenSpeakerEPc
>-   fun:_ZN6webrtc20AudioDeviceLinuxALSA11InitSpeakerEv
>-   ...
>-}
>-{
>-   bug_2153_3
>-   Memcheck:Leak
>-   fun:calloc
>-   obj:*
>-   fun:_ZN6webrtc26AudioMixerManagerLinuxALSA14OpenMicrophoneEPc
>-   fun:_ZN6webrtc20AudioDeviceLinuxALSA14InitMicrophoneEv
>-   ...
>-}
>-{
>-   bug_2263_1
>-   Memcheck:Uninitialized
>-   fun:_ZN14DtlsTestClient12VerifyPacketEPKcmPj
>-   ...
>-}
>-{
>-   bug_3478
>-   Memcheck:Leak
>-   fun:_Znw*
>-   fun:_ZNK3rtc18FakeSSLCertificate12GetReferenceEv
>-   fun:_ZN3rtc18FakeSSLCertificate7DupCertES0_
>-   fun:_ZSt9transformIN9__gnu_cxx17__normal_iteratorIPKN3rtc18FakeSSLCertificateESt6vectorIS3_SaIS3_EEEENS1_IPPNS2_14SSLCertificateES6_ISB_SaISB_EEEEPFPS3_S3_EET0_T_SK_SJ_T1_
>-   fun:_ZNK3rtc18FakeSSLCertificate8GetChainEPPNS_12SSLCertChainE
>-   fun:_ZN6webrtc14StatsCollector21AddCertificateReportsEPKN3rtc14SSLCertificateE
>-   fun:_ZN6webrtc14StatsCollector18ExtractSessionInfoEv
>-   fun:_ZN6webrtc14StatsCollector11UpdateStatsENS_23PeerConnectionInterface16StatsOutputLevelE
>-   fun:_ZN12_GLOBAL__N_118StatsCollectorTest22TestCertificateReportsERKN3rtc18FakeSSLCertificateERKSt6vectorISsSaISsEES4_S9_
>-   fun:_ZN12_GLOBAL__N_156StatsCollectorTest_ChainedCertificateReportsCreated_Test8TestBodyEv
>-}
>-{
>-   bug_5983
>-   Memcheck:Uninitialized
>-   fun:_ZN6webrtc11DataChannel4InitERKNS_23InternalDataChannelInitE
>-   ...
>-}
>-{
>-   bug_5984
>-   Memcheck:Uninitialized
>-   fun:_ZN6webrtc15GainControlImpl19ProcessCaptureAudioEPNS_11AudioBufferEb
>-   ...
>-}
>-{
>-   bug_5987
>-   Memcheck:Leak
>-   fun:_Znw*
>-   fun:_ZN6webrtc51AudioEncoderCopyRedDeathTest_NullSpeechEncoder_Test8TestBodyEv
>-}
>-{
>-   bug_5988
>-   Memcheck:Leak
>-   fun:_Znw*
>-   fun:_ZNSt8_Rb_treeISsSt4pairIKSsSsESt10_Select1stIS2_EN3rtc5ilessESaIS2_EE15_M_insert_equalIS2_EESt17_Rb_tree_iteratorIS2_EOT_
>-   fun:_ZN3rtc8HttpData12changeHeaderERKSsS2_NS0_13HeaderCombineE
>-   ...
>-   fun:_ZN3rtc47HttpServer_SignalsCloseAfterForcedCloseAll_Test8TestBodyEv
>-}
>-{
>-   bug_5989
>-   Memcheck:Unaddressable
>-   ...
>-   fun:_ZNK3rtc15CriticalSection5EnterEv
>-   ...
>-}
>-{
>-   bug_687087
>-   Memcheck:Uninitialized
>-   fun:vfprintf
>-   fun:vsnprintf
>-   fun:snprintf
>-   ...
>-   fun:_ZNK6webrtc11MockRtpRtcp11FlexfecSsrcEv
>-   ...
>-}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/memcheck/suppressions_win32.txt b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/memcheck/suppressions_win32.txt
>deleted file mode 100644
>index ec20df46d8b..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/memcheck/suppressions_win32.txt
>+++ /dev/null
>@@ -1,11 +0,0 @@
>-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>-#
>-# Use of this source code is governed by a BSD-style license
>-# that can be found in the LICENSE file in the root of the source
>-# tree. An additional intellectual property rights grant can be found
>-# in the file PATENTS.  All contributing project authors may
>-# be found in the AUTHORS file in the root of the source tree.
>-
>-# This file is used in addition to the one already maintained in Chrome.
>-# It acts as a place holder for future additions for WebRTC.
>-# It must exist for the Python wrapper script to work properly.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/memcheck_analyze.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/memcheck_analyze.py
>deleted file mode 100644
>index 80e85eb4ab1..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/memcheck_analyze.py
>+++ /dev/null
>@@ -1,644 +0,0 @@
>-#!/usr/bin/env python
>-# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>-#
>-# Use of this source code is governed by a BSD-style license
>-# that can be found in the LICENSE file in the root of the source
>-# tree. An additional intellectual property rights grant can be found
>-# in the file PATENTS.  All contributing project authors may
>-# be found in the AUTHORS file in the root of the source tree.
>-
>-# memcheck_analyze.py
>-
>-''' Given a valgrind XML file, parses errors and uniques them.'''
>-
>-import gdb_helper
>-
>-from collections import defaultdict
>-import hashlib
>-import logging
>-import optparse
>-import os
>-import re
>-import subprocess
>-import sys
>-import time
>-from xml.dom.minidom import parse
>-from xml.parsers.expat import ExpatError
>-
>-import common
>-
>-# Global symbol table (yuck)
>-TheAddressTable = None
>-
>-# These are regexps that define functions (using C++ mangled names)
>-# we don't want to see in stack traces while pretty printing
>-# or generating suppressions.
>-# Just stop printing the stack/suppression frames when the current one
>-# matches any of these.
>-_BORING_CALLERS = common.BoringCallers(mangled=True, use_re_wildcards=True)
>-
>-def getTextOf(top_node, name):
>-  ''' Returns all text in all DOM nodes with a certain |name| that are children
>-  of |top_node|.
>-  '''
>-
>-  text = ""
>-  for nodes_named in top_node.getElementsByTagName(name):
>-    text += "".join([node.data for node in nodes_named.childNodes
>-                     if node.nodeType == node.TEXT_NODE])
>-  return text
>-
>-def getCDATAOf(top_node, name):
>-  ''' Returns all CDATA in all DOM nodes with a certain |name| that are children
>-  of |top_node|.
>-  '''
>-
>-  text = ""
>-  for nodes_named in top_node.getElementsByTagName(name):
>-    text += "".join([node.data for node in nodes_named.childNodes
>-                     if node.nodeType == node.CDATA_SECTION_NODE])
>-  if (text == ""):
>-    return None
>-  return text
>-
>-def shortenFilePath(source_dir, directory):
>-  '''Returns a string with the string prefix |source_dir| removed from
>-  |directory|.'''
>-  prefixes_to_cut = ["build/src/", "valgrind/coregrind/", "out/Release/../../"]
>-
>-  if source_dir:
>-    prefixes_to_cut.append(source_dir)
>-
>-  for p in prefixes_to_cut:
>-    index = directory.rfind(p)
>-    if index != -1:
>-      directory = directory[index + len(p):]
>-
>-  return directory
>-
>-# Constants that give real names to the abbreviations in valgrind XML output.
>-INSTRUCTION_POINTER = "ip"
>-OBJECT_FILE = "obj"
>-FUNCTION_NAME = "fn"
>-SRC_FILE_DIR = "dir"
>-SRC_FILE_NAME = "file"
>-SRC_LINE = "line"
>-
>-def gatherFrames(node, source_dir):
>-  frames = []
>-  for frame in node.getElementsByTagName("frame"):
>-    frame_dict = {
>-      INSTRUCTION_POINTER : getTextOf(frame, INSTRUCTION_POINTER),
>-      OBJECT_FILE         : getTextOf(frame, OBJECT_FILE),
>-      FUNCTION_NAME       : getTextOf(frame, FUNCTION_NAME),
>-      SRC_FILE_DIR        : shortenFilePath(
>-          source_dir, getTextOf(frame, SRC_FILE_DIR)),
>-      SRC_FILE_NAME       : getTextOf(frame, SRC_FILE_NAME),
>-      SRC_LINE            : getTextOf(frame, SRC_LINE)
>-    }
>-
>-    # Ignore this frame and all the following if it's a "boring" function.
>-    enough_frames = False
>-    for regexp in _BORING_CALLERS:
>-      if re.match("^%s$" % regexp, frame_dict[FUNCTION_NAME]):
>-        enough_frames = True
>-        break
>-    if enough_frames:
>-      break
>-
>-    frames += [frame_dict]
>-
>-    global TheAddressTable
>-    if TheAddressTable != None and frame_dict[SRC_LINE] == "":
>-      # Try using gdb
>-      TheAddressTable.Add(frame_dict[OBJECT_FILE],
>-                          frame_dict[INSTRUCTION_POINTER])
>-  return frames
>-
>-class ValgrindError:
>-  ''' Takes a <DOM Element: error> node and reads all the data from it. A
>-  ValgrindError is immutable and is hashed on its pretty printed output.
>-  '''
>-
>-  def __init__(self, source_dir, error_node, commandline, testcase):
>-    ''' Copies all the relevant information out of the DOM and into object
>-    properties.
>-
>-    Args:
>-      error_node: The <error></error> DOM node we're extracting from.
>-      source_dir: Prefix that should be stripped from the <dir> node.
>-      commandline: The command that was run under valgrind
>-      testcase: The test case name, if known.
>-    '''
>-
>-    # Valgrind errors contain one <what><stack> pair, plus an optional
>-    # <auxwhat><stack> pair, plus an optional <origin><what><stack></origin>,
>-    # plus (since 3.5.0) a <suppression></suppression> pair.
>-    # (Origin is nicely enclosed; too bad the other two aren't.)
>-    # The most common way to see all three in one report is
>-    # a syscall with a parameter that points to uninitialized memory, e.g.
>-    # Format:
>-    # <error>
>-    #   <unique>0x6d</unique>
>-    #   <tid>1</tid>
>-    #   <kind>SyscallParam</kind>
>-    #   <what>Syscall param write(buf) points to uninitialised byte(s)</what>
>-    #   <stack>
>-    #     <frame>
>-    #     ...
>-    #     </frame>
>-    #   </stack>
>-    #   <auxwhat>Address 0x5c9af4f is 7 bytes inside a block of ...</auxwhat>
>-    #   <stack>
>-    #     <frame>
>-    #     ...
>-    #     </frame>
>-    #   </stack>
>-    #   <origin>
>-    #   <what>Uninitialised value was created by a heap allocation</what>
>-    #   <stack>
>-    #     <frame>
>-    #     ...
>-    #     </frame>
>-    #   </stack>
>-    #   </origin>
>-    #   <suppression>
>-    #     <sname>insert_a_suppression_name_here</sname>
>-    #     <skind>Memcheck:Param</skind>
>-    #     <skaux>write(buf)</skaux>
>-    #     <sframe> <fun>__write_nocancel</fun> </sframe>
>-    #     ...
>-    #     <sframe> <fun>main</fun> </sframe>
>-    #     <rawtext>
>-    # <![CDATA[
>-    # {
>-    #    <insert_a_suppression_name_here>
>-    #    Memcheck:Param
>-    #    write(buf)
>-    #    fun:__write_nocancel
>-    #    ...
>-    #    fun:main
>-    # }
>-    # ]]>
>-    #     </rawtext>
>-    #   </suppression>
>-    # </error>
>-    #
>-    # Each frame looks like this:
>-    #  <frame>
>-    #    <ip>0x83751BC</ip>
>-    #    <obj>/data/dkegel/chrome-build/src/out/Release/base_unittests</obj>
>-    #    <fn>_ZN7testing8internal12TestInfoImpl7RunTestEPNS_8TestInfoE</fn>
>-    #    <dir>/data/dkegel/chrome-build/src/testing/gtest/src</dir>
>-    #    <file>gtest-internal-inl.h</file>
>-    #    <line>655</line>
>-    #  </frame>
>-    # although the dir, file, and line elements are missing if there is
>-    # no debug info.
>-
>-    self._kind = getTextOf(error_node, "kind")
>-    self._backtraces = []
>-    self._suppression = None
>-    self._commandline = commandline
>-    self._testcase = testcase
>-    self._additional = []
>-
>-    # Iterate through the nodes, parsing <what|auxwhat><stack> pairs.
>-    description = None
>-    for node in error_node.childNodes:
>-      if node.localName == "what" or node.localName == "auxwhat":
>-        description = "".join([n.data for n in node.childNodes
>-                              if n.nodeType == n.TEXT_NODE])
>-      elif node.localName == "xwhat":
>-        description = getTextOf(node, "text")
>-      elif node.localName == "stack":
>-        assert description
>-        self._backtraces.append([description, gatherFrames(node, source_dir)])
>-        description = None
>-      elif node.localName == "origin":
>-        description = getTextOf(node, "what")
>-        stack = node.getElementsByTagName("stack")[0]
>-        frames = gatherFrames(stack, source_dir)
>-        self._backtraces.append([description, frames])
>-        description = None
>-        stack = None
>-        frames = None
>-      elif description and node.localName != None:
>-        # The lastest description has no stack, e.g. "Address 0x28 is unknown"
>-        self._additional.append(description)
>-        description = None
>-
>-      if node.localName == "suppression":
>-        self._suppression = getCDATAOf(node, "rawtext");
>-
>-  def __str__(self):
>-    ''' Pretty print the type and backtrace(s) of this specific error,
>-        including suppression (which is just a mangled backtrace).'''
>-    output = ""
>-    output += "\n" # Make sure the ### is at the beginning of line.
>-    output += "### BEGIN MEMORY TOOL REPORT (error hash=#%016X#)\n" % \
>-        self.ErrorHash()
>-    if (self._commandline):
>-      output += self._commandline + "\n"
>-
>-    output += self._kind + "\n"
>-    for backtrace in self._backtraces:
>-      output += backtrace[0] + "\n"
>-      filter = subprocess.Popen("c++filt -n", stdin=subprocess.PIPE,
>-                                stdout=subprocess.PIPE,
>-                                stderr=subprocess.STDOUT,
>-                                shell=True,
>-                                close_fds=True)
>-      buf = ""
>-      for frame in backtrace[1]:
>-        buf +=  (frame[FUNCTION_NAME] or frame[INSTRUCTION_POINTER]) + "\n"
>-      (stdoutbuf, stderrbuf) = filter.communicate(buf.encode('latin-1'))
>-      demangled_names = stdoutbuf.split("\n")
>-
>-      i = 0
>-      for frame in backtrace[1]:
>-        output += ("  " + demangled_names[i])
>-        i = i + 1
>-
>-        global TheAddressTable
>-        if TheAddressTable != None and frame[SRC_FILE_DIR] == "":
>-           # Try using gdb
>-           foo = TheAddressTable.GetFileLine(frame[OBJECT_FILE],
>-                                             frame[INSTRUCTION_POINTER])
>-           if foo[0] != None:
>-             output += (" (" + foo[0] + ":" + foo[1] + ")")
>-        elif frame[SRC_FILE_DIR] != "":
>-          output += (" (" + frame[SRC_FILE_DIR] + "/" + frame[SRC_FILE_NAME] +
>-                     ":" + frame[SRC_LINE] + ")")
>-        else:
>-          output += " (" + frame[OBJECT_FILE] + ")"
>-        output += "\n"
>-
>-    for additional in self._additional:
>-      output += additional + "\n"
>-
>-    assert self._suppression != None, "Your Valgrind doesn't generate " \
>-                                      "suppressions - is it too old?"
>-
>-    if self._testcase:
>-      output += "The report came from the `%s` test.\n" % self._testcase
>-    output += "Suppression (error hash=#%016X#):\n" % self.ErrorHash()
>-    output += ("  For more info on using suppressions see "
>-               "http://dev.chromium.org/developers/tree-sheriffs/sheriff-details-chromium/memory-sheriff#TOC-Suppressing-memory-reports")
>-
>-    # Widen suppression slightly to make portable between mac and linux
>-    # TODO(timurrrr): Oops, these transformations should happen
>-    # BEFORE calculating the hash!
>-    supp = self._suppression;
>-    supp = supp.replace("fun:_Znwj", "fun:_Znw*")
>-    supp = supp.replace("fun:_Znwm", "fun:_Znw*")
>-    supp = supp.replace("fun:_Znaj", "fun:_Zna*")
>-    supp = supp.replace("fun:_Znam", "fun:_Zna*")
>-
>-    # Make suppressions even less platform-dependent.
>-    for sz in [1, 2, 4, 8]:
>-      supp = supp.replace("Memcheck:Addr%d" % sz, "Memcheck:Unaddressable")
>-      supp = supp.replace("Memcheck:Value%d" % sz, "Memcheck:Uninitialized")
>-    supp = supp.replace("Memcheck:Cond", "Memcheck:Uninitialized")
>-
>-    # Split into lines so we can enforce length limits
>-    supplines = supp.split("\n")
>-    supp = None  # to avoid re-use
>-
>-    # Truncate at line 26 (VG_MAX_SUPP_CALLERS plus 2 for name and type)
>-    # or at the first 'boring' caller.
>-    # (https://bugs.kde.org/show_bug.cgi?id=199468 proposes raising
>-    # VG_MAX_SUPP_CALLERS, but we're probably fine with it as is.)
>-    newlen = min(26, len(supplines));
>-
>-    # Drop boring frames and all the following.
>-    enough_frames = False
>-    for frameno in range(newlen):
>-      for boring_caller in _BORING_CALLERS:
>-        if re.match("^ +fun:%s$" % boring_caller, supplines[frameno]):
>-          newlen = frameno
>-          enough_frames = True
>-          break
>-      if enough_frames:
>-        break
>-    if (len(supplines) > newlen):
>-      supplines = supplines[0:newlen]
>-      supplines.append("}")
>-
>-    for frame in range(len(supplines)):
>-      # Replace the always-changing anonymous namespace prefix with "*".
>-      m = re.match("( +fun:)_ZN.*_GLOBAL__N_.*\.cc_" +
>-                   "[0-9a-fA-F]{8}_[0-9a-fA-F]{8}(.*)",
>-                   supplines[frame])
>-      if m:
>-        supplines[frame] = "*".join(m.groups())
>-
>-    output += "\n".join(supplines) + "\n"
>-    output += "### END MEMORY TOOL REPORT (error hash=#%016X#)\n" % \
>-        self.ErrorHash()
>-
>-    return output
>-
>-  def UniqueString(self):
>-    ''' String to use for object identity. Don't print this, use str(obj)
>-    instead.'''
>-    rep = self._kind + " "
>-    for backtrace in self._backtraces:
>-      for frame in backtrace[1]:
>-        rep += frame[FUNCTION_NAME]
>-
>-        if frame[SRC_FILE_DIR] != "":
>-          rep += frame[SRC_FILE_DIR] + "/" + frame[SRC_FILE_NAME]
>-        else:
>-          rep += frame[OBJECT_FILE]
>-
>-    return rep
>-
>-  # This is a device-independent hash identifying the suppression.
>-  # By printing out this hash we can find duplicate reports between tests and
>-  # different shards running on multiple buildbots
>-  def ErrorHash(self):
>-    return int(hashlib.md5(self.UniqueString()).hexdigest()[:16], 16)
>-
>-  def __hash__(self):
>-    return hash(self.UniqueString())
>-  def __eq__(self, rhs):
>-    return self.UniqueString() == rhs
>-
>-def log_is_finished(f, force_finish):
>-  f.seek(0)
>-  prev_line = ""
>-  while True:
>-    line = f.readline()
>-    if line == "":
>-      if not force_finish:
>-        return False
>-      # Okay, the log is not finished but we can make it up to be parseable:
>-      if prev_line.strip() in ["</error>", "</errorcounts>", "</status>"]:
>-        f.write("</valgrindoutput>\n")
>-        return True
>-      return False
>-    if '</valgrindoutput>' in line:
>-      # Valgrind often has garbage after </valgrindoutput> upon crash.
>-      f.truncate()
>-      return True
>-    prev_line = line
>-
>-class MemcheckAnalyzer:
>-  ''' Given a set of Valgrind XML files, parse all the errors out of them,
>-  unique them and output the results.'''
>-
>-  SANITY_TEST_SUPPRESSIONS = {
>-      "Memcheck sanity test 01 (memory leak).": 1,
>-      "Memcheck sanity test 02 (malloc/read left).": 1,
>-      "Memcheck sanity test 03 (malloc/read right).": 1,
>-      "Memcheck sanity test 04 (malloc/write left).": 1,
>-      "Memcheck sanity test 05 (malloc/write right).": 1,
>-      "Memcheck sanity test 06 (new/read left).": 1,
>-      "Memcheck sanity test 07 (new/read right).": 1,
>-      "Memcheck sanity test 08 (new/write left).": 1,
>-      "Memcheck sanity test 09 (new/write right).": 1,
>-      "Memcheck sanity test 10 (write after free).": 1,
>-      "Memcheck sanity test 11 (write after delete).": 1,
>-      "Memcheck sanity test 12 (array deleted without []).": 1,
>-      "Memcheck sanity test 13 (single element deleted with []).": 1,
>-      "Memcheck sanity test 14 (malloc/read uninit).": 1,
>-      "Memcheck sanity test 15 (new/read uninit).": 1,
>-  }
>-
>-  # Max time to wait for memcheck logs to complete.
>-  LOG_COMPLETION_TIMEOUT = 180.0
>-
>-  def __init__(self, source_dir, show_all_leaks=False, use_gdb=False):
>-    '''Create a parser for Memcheck logs.
>-
>-    Args:
>-      source_dir: Path to top of source tree for this build
>-      show_all_leaks: Whether to show even less important leaks
>-      use_gdb: Whether to use gdb to resolve source filenames and line numbers
>-               in the report stacktraces
>-    '''
>-    self._source_dir = source_dir
>-    self._show_all_leaks = show_all_leaks
>-    self._use_gdb = use_gdb
>-
>-    # Contains the set of unique errors
>-    self._errors = set()
>-
>-    # Contains the time when the we started analyzing the first log file.
>-    # This variable is used to skip incomplete logs after some timeout.
>-    self._analyze_start_time = None
>-
>-
>-  def Report(self, files, testcase, check_sanity=False):
>-    '''Reads in a set of files and prints Memcheck report.
>-
>-    Args:
>-      files: A list of filenames.
>-      check_sanity: if true, search for SANITY_TEST_SUPPRESSIONS
>-    '''
>-    # Beyond the detailed errors parsed by ValgrindError above,
>-    # the xml file contain records describing suppressions that were used:
>-    # <suppcounts>
>-    #  <pair>
>-    #    <count>28</count>
>-    #    <name>pango_font_leak_todo</name>
>-    #  </pair>
>-    #  <pair>
>-    #    <count>378</count>
>-    #    <name>bug_13243</name>
>-    #  </pair>
>-    # </suppcounts
>-    # Collect these and print them at the end.
>-    #
>-    # With our patch for https://bugs.kde.org/show_bug.cgi?id=205000 in,
>-    # the file also includes records of the form
>-    # <load_obj><obj>/usr/lib/libgcc_s.1.dylib</obj><ip>0x27000</ip></load_obj>
>-    # giving the filename and load address of each binary that was mapped
>-    # into the process.
>-
>-    global TheAddressTable
>-    if self._use_gdb:
>-      TheAddressTable = gdb_helper.AddressTable()
>-    else:
>-      TheAddressTable = None
>-    cur_report_errors = set()
>-    suppcounts = defaultdict(int)
>-    badfiles = set()
>-
>-    if self._analyze_start_time == None:
>-      self._analyze_start_time = time.time()
>-    start_time = self._analyze_start_time
>-
>-    parse_failed = False
>-    for file in files:
>-      # Wait up to three minutes for valgrind to finish writing all files,
>-      # but after that, just skip incomplete files and warn.
>-      f = open(file, "r+")
>-      pid = re.match(".*\.([0-9]+)$", file)
>-      if pid:
>-        pid = pid.groups()[0]
>-      found = False
>-      running = True
>-      firstrun = True
>-      skip = False
>-      origsize = os.path.getsize(file)
>-      while (running and not found and not skip and
>-             (firstrun or
>-              ((time.time() - start_time) < self.LOG_COMPLETION_TIMEOUT))):
>-        firstrun = False
>-        f.seek(0)
>-        if pid:
>-          # Make sure the process is still running so we don't wait for
>-          # 3 minutes if it was killed. See http://crbug.com/17453
>-          ps_out = subprocess.Popen("ps p %s" % pid, shell=True,
>-                                    stdout=subprocess.PIPE).stdout
>-          if len(ps_out.readlines()) < 2:
>-            running = False
>-        else:
>-          skip = True
>-          running = False
>-        found = log_is_finished(f, False)
>-        if not running and not found:
>-          logging.warn("Valgrind process PID = %s is not running but its "
>-                       "XML log has not been finished correctly.\n"
>-                       "Make it up by adding some closing tags manually." % pid)
>-          found = log_is_finished(f, not running)
>-        if running and not found:
>-          time.sleep(1)
>-      f.close()
>-      if not found:
>-        badfiles.add(file)
>-      else:
>-        newsize = os.path.getsize(file)
>-        if origsize > newsize+1:
>-          logging.warn(str(origsize - newsize) +
>-                       " bytes of junk were after </valgrindoutput> in %s!" %
>-                       file)
>-        try:
>-          parsed_file = parse(file);
>-        except ExpatError, e:
>-          parse_failed = True
>-          logging.warn("could not parse %s: %s" % (file, e))
>-          lineno = e.lineno - 1
>-          context_lines = 5
>-          context_start = max(0, lineno - context_lines)
>-          context_end = lineno + context_lines + 1
>-          context_file = open(file, "r")
>-          for i in range(0, context_start):
>-            context_file.readline()
>-          for i in range(context_start, context_end):
>-            context_data = context_file.readline().rstrip()
>-            if i != lineno:
>-              logging.warn("  %s" % context_data)
>-            else:
>-              logging.warn("> %s" % context_data)
>-          context_file.close()
>-          continue
>-        if TheAddressTable != None:
>-          load_objs = parsed_file.getElementsByTagName("load_obj")
>-          for load_obj in load_objs:
>-            obj = getTextOf(load_obj, "obj")
>-            ip = getTextOf(load_obj, "ip")
>-            TheAddressTable.AddBinaryAt(obj, ip)
>-
>-        commandline = None
>-        preamble = parsed_file.getElementsByTagName("preamble")[0];
>-        for node in preamble.getElementsByTagName("line"):
>-          if node.localName == "line":
>-            for x in node.childNodes:
>-              if x.nodeType == node.TEXT_NODE and "Command" in x.data:
>-                commandline = x.data
>-                break
>-
>-        raw_errors = parsed_file.getElementsByTagName("error")
>-        for raw_error in raw_errors:
>-          # Ignore "possible" leaks for now by default.
>-          if (self._show_all_leaks or
>-              getTextOf(raw_error, "kind") != "Leak_PossiblyLost"):
>-            error = ValgrindError(self._source_dir,
>-                                  raw_error, commandline, testcase)
>-            if error not in cur_report_errors:
>-              # We haven't seen such errors doing this report yet...
>-              if error in self._errors:
>-                # ... but we saw it in earlier reports, e.g. previous UI test
>-                cur_report_errors.add("This error was already printed in "
>-                                      "some other test, see 'hash=#%016X#'" % \
>-                                      error.ErrorHash())
>-              else:
>-                # ... and we haven't seen it in other tests as well
>-                self._errors.add(error)
>-                cur_report_errors.add(error)
>-
>-        suppcountlist = parsed_file.getElementsByTagName("suppcounts")
>-        if len(suppcountlist) > 0:
>-          suppcountlist = suppcountlist[0]
>-          for node in suppcountlist.getElementsByTagName("pair"):
>-            count = getTextOf(node, "count");
>-            name = getTextOf(node, "name");
>-            suppcounts[name] += int(count)
>-
>-    if len(badfiles) > 0:
>-      logging.warn("valgrind didn't finish writing %d files?!" % len(badfiles))
>-      for file in badfiles:
>-        logging.warn("Last 20 lines of %s :" % file)
>-        os.system("tail -n 20 '%s' 1>&2" % file)
>-
>-    if parse_failed:
>-      logging.error("FAIL! Couldn't parse Valgrind output file")
>-      return -2
>-
>-    common.PrintUsedSuppressionsList(suppcounts)
>-
>-    retcode = 0
>-    if cur_report_errors:
>-      logging.error("FAIL! There were %s errors: " % len(cur_report_errors))
>-
>-      if TheAddressTable != None:
>-        TheAddressTable.ResolveAll()
>-
>-      for error in cur_report_errors:
>-        logging.error(error)
>-
>-      retcode = -1
>-
>-    # Report tool's insanity even if there were errors.
>-    if check_sanity:
>-      remaining_sanity_supp = MemcheckAnalyzer.SANITY_TEST_SUPPRESSIONS
>-      for (name, count) in suppcounts.iteritems():
>-        # Workaround for http://crbug.com/334074
>-        if (name in remaining_sanity_supp and
>-            remaining_sanity_supp[name] <= count):
>-          del remaining_sanity_supp[name]
>-      if remaining_sanity_supp:
>-        logging.error("FAIL! Sanity check failed!")
>-        logging.info("The following test errors were not handled: ")
>-        for (name, count) in remaining_sanity_supp.iteritems():
>-          logging.info("  * %dx %s" % (count, name))
>-        retcode = -3
>-
>-    if retcode != 0:
>-      return retcode
>-
>-    logging.info("PASS! No errors found!")
>-    return 0
>-
>-
>-def _main():
>-  '''For testing only. The MemcheckAnalyzer class should be imported instead.'''
>-  parser = optparse.OptionParser("usage: %prog [options] <files to analyze>")
>-  parser.add_option("", "--source-dir",
>-                    help="path to top of source tree for this build"
>-                    "(used to normalize source paths in baseline)")
>-
>-  (options, args) = parser.parse_args()
>-  if len(args) == 0:
>-    parser.error("no filename specified")
>-  filenames = args
>-
>-  analyzer = MemcheckAnalyzer(options.source_dir, use_gdb=True)
>-  return analyzer.Report(filenames, None)
>-
>-
>-if __name__ == "__main__":
>-  sys.exit(_main())
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/python-google.gni b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/python-google.gni
>deleted file mode 100644
>index 654029bf25b..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/python-google.gni
>+++ /dev/null
>@@ -1,27 +0,0 @@
>-# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
>-#
>-# Use of this source code is governed by a BSD-style license
>-# that can be found in the LICENSE file in the root of the source
>-# tree. An additional intellectual property rights grant can be found
>-# in the file PATENTS.  All contributing project authors may
>-# be found in the AUTHORS file in the root of the source tree.
>-
>-python_google = [
>-  "../tools/python/google/__init__.py",
>-  "../tools/python/google/gethash_timer.py",
>-  "../tools/python/google/httpd_config/httpd.conf",
>-  "../tools/python/google/httpd_config/httpd2.conf",
>-  "../tools/python/google/httpd_config/httpd2.pem",
>-  "../tools/python/google/httpd_config/httpd2_linux.conf",
>-  "../tools/python/google/httpd_config/httpd2_mac.conf",
>-  "../tools/python/google/httpd_config/mime.types",
>-  "../tools/python/google/httpd_config/root_ca_cert.pem",
>-  "../tools/python/google/httpd_utils.py",
>-  "../tools/python/google/logging_utils.py",
>-  "../tools/python/google/path_utils.py",
>-  "../tools/python/google/platform_utils.py",
>-  "../tools/python/google/platform_utils_linux.py",
>-  "../tools/python/google/platform_utils_mac.py",
>-  "../tools/python/google/platform_utils_win.py",
>-  "../tools/python/google/process_utils.py",
>-]
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/valgrind-binaries.gni b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/valgrind-binaries.gni
>deleted file mode 100644
>index df11b32635f..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/valgrind-binaries.gni
>+++ /dev/null
>@@ -1,425 +0,0 @@
>-# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
>-#
>-# Use of this source code is governed by a BSD-style license
>-# that can be found in the LICENSE file in the root of the source
>-# tree. An additional intellectual property rights grant can be found
>-# in the file PATENTS.  All contributing project authors may
>-# be found in the AUTHORS file in the root of the source tree.
>-
>-valgrind_binaries = []
>-
>-if (is_mac) {
>-  valgrind_binaries += [
>-    "../third_party/valgrind/mac_10.6/bin/ms_print",
>-    "../third_party/valgrind/mac_10.6/bin/no_op_client_for_valgrind",
>-    "../third_party/valgrind/mac_10.6/bin/valgrind",
>-    "../third_party/valgrind/mac_10.6/bin/valgrind-listener",
>-    "../third_party/valgrind/mac_10.6/bin/vgdb",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/32bit-core-valgrind-s1.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/32bit-core-valgrind-s2.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/32bit-core.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/32bit-linux-valgrind-s1.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/32bit-linux-valgrind-s2.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/32bit-linux.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/32bit-sse-valgrind-s1.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/32bit-sse-valgrind-s2.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/32bit-sse.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/64bit-avx-valgrind-s1.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/64bit-avx-valgrind-s2.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/64bit-avx.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/64bit-core-valgrind-s1.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/64bit-core-valgrind-s2.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/64bit-core.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/64bit-linux-valgrind-s1.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/64bit-linux-valgrind-s2.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/64bit-linux.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/64bit-sse-valgrind-s1.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/64bit-sse-valgrind-s2.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/64bit-sse.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/amd64-avx-coresse-valgrind.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/amd64-avx-coresse.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/amd64-avx-linux-valgrind.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/amd64-avx-linux.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/amd64-coresse-valgrind.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/amd64-linux-valgrind.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/arm-core-valgrind-s1.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/arm-core-valgrind-s2.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/arm-core.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/arm-vfpv3-valgrind-s1.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/arm-vfpv3-valgrind-s2.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/arm-vfpv3.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/arm-with-vfpv3-valgrind.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/arm-with-vfpv3.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/default.supp",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/i386-coresse-valgrind.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/i386-linux-valgrind.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/libmpiwrap-amd64-darwin.so",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/libmpiwrap-amd64-darwin.so.dSYM/Contents/Info.plist",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/libmpiwrap-amd64-darwin.so.dSYM/Contents/Resources/DWARF/libmpiwrap-amd64-darwin.so",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/libmpiwrap-x86-darwin.so",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/libmpiwrap-x86-darwin.so.dSYM/Contents/Info.plist",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/libmpiwrap-x86-darwin.so.dSYM/Contents/Resources/DWARF/libmpiwrap-x86-darwin.so",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/memcheck-amd64-darwin",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/memcheck-amd64-darwin.dSYM/Contents/Info.plist",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/memcheck-amd64-darwin.dSYM/Contents/Resources/DWARF/memcheck-amd64-darwin",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/memcheck-x86-darwin",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/memcheck-x86-darwin.dSYM/Contents/Info.plist",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/memcheck-x86-darwin.dSYM/Contents/Resources/DWARF/memcheck-x86-darwin",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/mips-cp0-valgrind-s1.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/mips-cp0-valgrind-s2.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/mips-cp0.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/mips-cpu-valgrind-s1.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/mips-cpu-valgrind-s2.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/mips-cpu.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/mips-fpu-valgrind-s1.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/mips-fpu-valgrind-s2.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/mips-fpu.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/mips-linux-valgrind.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/mips-linux.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/power-altivec-valgrind-s1.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/power-altivec-valgrind-s2.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/power-altivec.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/power-core-valgrind-s1.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/power-core-valgrind-s2.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/power-core.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/power-fpu-valgrind-s1.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/power-fpu-valgrind-s2.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/power-fpu.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/power-linux-valgrind-s1.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/power-linux-valgrind-s2.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/power-linux.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/power64-core-valgrind-s1.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/power64-core-valgrind-s2.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/power64-core.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/power64-linux-valgrind-s1.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/power64-linux-valgrind-s2.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/power64-linux.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/powerpc-altivec32l-valgrind.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/powerpc-altivec32l.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/powerpc-altivec64l-valgrind.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/powerpc-altivec64l.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/s390-acr-valgrind-s1.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/s390-acr-valgrind-s2.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/s390-acr.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/s390-fpr-valgrind-s1.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/s390-fpr-valgrind-s2.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/s390-fpr.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/s390x-core64-valgrind-s1.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/s390x-core64-valgrind-s2.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/s390x-core64.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/s390x-generic-valgrind.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/s390x-generic.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/s390x-linux64-valgrind-s1.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/s390x-linux64-valgrind-s2.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/s390x-linux64.xml",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/vgpreload_core-amd64-darwin.so",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/vgpreload_core-amd64-darwin.so.dSYM/Contents/Info.plist",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/vgpreload_core-amd64-darwin.so.dSYM/Contents/Resources/DWARF/vgpreload_core-amd64-darwin.so",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/vgpreload_core-x86-darwin.so",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/vgpreload_core-x86-darwin.so.dSYM/Contents/Info.plist",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/vgpreload_core-x86-darwin.so.dSYM/Contents/Resources/DWARF/vgpreload_core-x86-darwin.so",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/vgpreload_memcheck-amd64-darwin.so",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/vgpreload_memcheck-amd64-darwin.so.dSYM/Contents/Info.plist",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/vgpreload_memcheck-amd64-darwin.so.dSYM/Contents/Resources/DWARF/vgpreload_memcheck-amd64-darwin.so",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/vgpreload_memcheck-x86-darwin.so",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/vgpreload_memcheck-x86-darwin.so.dSYM/Contents/Info.plist",
>-    "../third_party/valgrind/mac_10.6/lib/valgrind/vgpreload_memcheck-x86-darwin.so.dSYM/Contents/Resources/DWARF/vgpreload_memcheck-x86-darwin.so",
>-    "../third_party/valgrind/mac_10.7/bin/valgrind",
>-    "../third_party/valgrind/mac_10.7/bin/valgrind-listener",
>-    "../third_party/valgrind/mac_10.7/bin/vgdb",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/32bit-core-valgrind-s1.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/32bit-core-valgrind-s2.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/32bit-core.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/32bit-linux-valgrind-s1.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/32bit-linux-valgrind-s2.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/32bit-linux.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/32bit-sse-valgrind-s1.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/32bit-sse-valgrind-s2.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/32bit-sse.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/64bit-avx-valgrind-s1.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/64bit-avx-valgrind-s2.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/64bit-avx.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/64bit-core-valgrind-s1.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/64bit-core-valgrind-s2.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/64bit-core.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/64bit-linux-valgrind-s1.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/64bit-linux-valgrind-s2.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/64bit-linux.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/64bit-sse-valgrind-s1.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/64bit-sse-valgrind-s2.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/64bit-sse.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/amd64-avx-coresse-valgrind.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/amd64-avx-coresse.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/amd64-avx-linux-valgrind.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/amd64-avx-linux.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/amd64-coresse-valgrind.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/amd64-linux-valgrind.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/arm-core-valgrind-s1.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/arm-core-valgrind-s2.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/arm-core.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/arm-vfpv3-valgrind-s1.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/arm-vfpv3-valgrind-s2.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/arm-vfpv3.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/arm-with-vfpv3-valgrind.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/arm-with-vfpv3.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/default.supp",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/i386-coresse-valgrind.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/i386-linux-valgrind.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/memcheck-amd64-darwin",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/memcheck-amd64-darwin.dSYM/Contents/Info.plist",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/memcheck-amd64-darwin.dSYM/Contents/Resources/DWARF/memcheck-amd64-darwin",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/memcheck-x86-darwin",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/memcheck-x86-darwin.dSYM/Contents/Info.plist",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/memcheck-x86-darwin.dSYM/Contents/Resources/DWARF/memcheck-x86-darwin",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/mips-cp0-valgrind-s1.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/mips-cp0-valgrind-s2.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/mips-cp0.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/mips-cpu-valgrind-s1.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/mips-cpu-valgrind-s2.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/mips-cpu.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/mips-fpu-valgrind-s1.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/mips-fpu-valgrind-s2.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/mips-fpu.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/mips-linux-valgrind.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/mips-linux.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/power-altivec-valgrind-s1.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/power-altivec-valgrind-s2.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/power-altivec.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/power-core-valgrind-s1.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/power-core-valgrind-s2.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/power-core.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/power-fpu-valgrind-s1.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/power-fpu-valgrind-s2.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/power-fpu.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/power-linux-valgrind-s1.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/power-linux-valgrind-s2.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/power-linux.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/power64-core-valgrind-s1.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/power64-core-valgrind-s2.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/power64-core.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/power64-linux-valgrind-s1.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/power64-linux-valgrind-s2.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/power64-linux.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/powerpc-altivec32l-valgrind.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/powerpc-altivec32l.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/powerpc-altivec64l-valgrind.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/powerpc-altivec64l.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/s390-acr-valgrind-s1.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/s390-acr-valgrind-s2.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/s390-acr.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/s390-fpr-valgrind-s1.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/s390-fpr-valgrind-s2.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/s390-fpr.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/s390x-core64-valgrind-s1.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/s390x-core64-valgrind-s2.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/s390x-core64.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/s390x-generic-valgrind.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/s390x-generic.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/s390x-linux64-valgrind-s1.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/s390x-linux64-valgrind-s2.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/s390x-linux64.xml",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/vgpreload_core-amd64-darwin.so.dSYM/Contents/Info.plist",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/vgpreload_core-x86-darwin.so.dSYM/Contents/Info.plist",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/vgpreload_memcheck-amd64-darwin.so.dSYM/Contents/Info.plist",
>-    "../third_party/valgrind/mac_10.7/lib/valgrind/vgpreload_memcheck-x86-darwin.so.dSYM/Contents/Info.plist",
>-  ]
>-} else if (is_linux) {
>-  if (target_cpu == "x64") {
>-    valgrind_binaries += [
>-      "../third_party/valgrind/linux_x64/bin/ms_print",
>-      "../third_party/valgrind/linux_x64/bin/no_op_client_for_valgrind",
>-      "../third_party/valgrind/linux_x64/bin/valgrind",
>-      "../third_party/valgrind/linux_x64/bin/valgrind-listener",
>-      "../third_party/valgrind/linux_x64/bin/valgrind-tsan.sh",
>-      "../third_party/valgrind/linux_x64/bin/vgdb",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/32bit-core-valgrind-s1.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/32bit-core-valgrind-s2.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/32bit-core.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/32bit-linux-valgrind-s1.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/32bit-linux-valgrind-s2.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/32bit-linux.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/32bit-sse-valgrind-s1.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/32bit-sse-valgrind-s2.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/32bit-sse.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/64bit-avx-valgrind-s1.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/64bit-avx-valgrind-s2.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/64bit-avx.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/64bit-core-valgrind-s1.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/64bit-core-valgrind-s2.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/64bit-core.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/64bit-linux-valgrind-s1.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/64bit-linux-valgrind-s2.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/64bit-linux.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/64bit-sse-valgrind-s1.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/64bit-sse-valgrind-s2.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/64bit-sse.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/amd64-avx-coresse-valgrind.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/amd64-avx-coresse.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/amd64-avx-linux-valgrind.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/amd64-avx-linux.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/amd64-coresse-valgrind.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/amd64-linux-valgrind.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/arm-core-valgrind-s1.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/arm-core-valgrind-s2.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/arm-core.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/arm-vfpv3-valgrind-s1.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/arm-vfpv3-valgrind-s2.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/arm-vfpv3.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/arm-with-vfpv3-valgrind.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/arm-with-vfpv3.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/default.supp",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/i386-coresse-valgrind.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/i386-linux-valgrind.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/memcheck-amd64-linux",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/memcheck-x86-linux",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/mips-cp0-valgrind-s1.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/mips-cp0-valgrind-s2.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/mips-cp0.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/mips-cpu-valgrind-s1.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/mips-cpu-valgrind-s2.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/mips-cpu.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/mips-fpu-valgrind-s1.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/mips-fpu-valgrind-s2.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/mips-fpu.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/mips-linux-valgrind.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/mips-linux.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/power-altivec-valgrind-s1.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/power-altivec-valgrind-s2.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/power-altivec.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/power-core-valgrind-s1.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/power-core-valgrind-s2.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/power-core.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/power-fpu-valgrind-s1.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/power-fpu-valgrind-s2.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/power-fpu.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/power-linux-valgrind-s1.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/power-linux-valgrind-s2.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/power-linux.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/power64-core-valgrind-s1.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/power64-core-valgrind-s2.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/power64-core.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/power64-linux-valgrind-s1.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/power64-linux-valgrind-s2.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/power64-linux.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/powerpc-altivec32l-valgrind.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/powerpc-altivec32l.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/powerpc-altivec64l-valgrind.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/powerpc-altivec64l.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/s390-acr-valgrind-s1.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/s390-acr-valgrind-s2.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/s390-acr.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/s390-fpr-valgrind-s1.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/s390-fpr-valgrind-s2.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/s390-fpr.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/s390x-core64-valgrind-s1.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/s390x-core64-valgrind-s2.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/s390x-core64.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/s390x-generic-valgrind.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/s390x-generic.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/s390x-linux64-valgrind-s1.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/s390x-linux64-valgrind-s2.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/s390x-linux64.xml",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/vgpreload_core-amd64-linux.so",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/vgpreload_core-x86-linux.so",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/vgpreload_memcheck-amd64-linux.so",
>-      "../third_party/valgrind/linux_x64/lib/valgrind/vgpreload_memcheck-x86-linux.so",
>-    ]
>-  } else if (target_cpu == "x86") {
>-    valgrind_binaries += [
>-      "../third_party/valgrind/linux_x86/bin/ms_print",
>-      "../third_party/valgrind/linux_x86/bin/no_op_client_for_valgrind",
>-      "../third_party/valgrind/linux_x86/bin/valgrind",
>-      "../third_party/valgrind/linux_x86/bin/valgrind-listener",
>-      "../third_party/valgrind/linux_x86/bin/valgrind-tsan.sh",
>-      "../third_party/valgrind/linux_x86/bin/vgdb",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/32bit-core-valgrind-s1.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/32bit-core-valgrind-s2.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/32bit-core.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/32bit-linux-valgrind-s1.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/32bit-linux-valgrind-s2.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/32bit-linux.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/32bit-sse-valgrind-s1.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/32bit-sse-valgrind-s2.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/32bit-sse.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/64bit-avx-valgrind-s1.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/64bit-avx-valgrind-s2.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/64bit-avx.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/64bit-core-valgrind-s1.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/64bit-core-valgrind-s2.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/64bit-core.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/64bit-linux-valgrind-s1.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/64bit-linux-valgrind-s2.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/64bit-linux.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/64bit-sse-valgrind-s1.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/64bit-sse-valgrind-s2.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/64bit-sse.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/amd64-avx-coresse-valgrind.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/amd64-avx-coresse.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/amd64-avx-linux-valgrind.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/amd64-avx-linux.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/amd64-coresse-valgrind.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/amd64-linux-valgrind.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/arm-core-valgrind-s1.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/arm-core-valgrind-s2.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/arm-core.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/arm-vfpv3-valgrind-s1.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/arm-vfpv3-valgrind-s2.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/arm-vfpv3.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/arm-with-vfpv3-valgrind.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/arm-with-vfpv3.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/default.supp",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/i386-coresse-valgrind.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/i386-linux-valgrind.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/memcheck-x86-linux",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/mips-cp0-valgrind-s1.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/mips-cp0-valgrind-s2.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/mips-cp0.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/mips-cpu-valgrind-s1.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/mips-cpu-valgrind-s2.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/mips-cpu.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/mips-fpu-valgrind-s1.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/mips-fpu-valgrind-s2.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/mips-fpu.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/mips-linux-valgrind.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/mips-linux.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/power-altivec-valgrind-s1.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/power-altivec-valgrind-s2.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/power-altivec.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/power-core-valgrind-s1.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/power-core-valgrind-s2.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/power-core.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/power-fpu-valgrind-s1.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/power-fpu-valgrind-s2.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/power-fpu.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/power-linux-valgrind-s1.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/power-linux-valgrind-s2.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/power-linux.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/power64-core-valgrind-s1.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/power64-core-valgrind-s2.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/power64-core.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/power64-linux-valgrind-s1.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/power64-linux-valgrind-s2.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/power64-linux.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/powerpc-altivec32l-valgrind.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/powerpc-altivec32l.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/powerpc-altivec64l-valgrind.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/powerpc-altivec64l.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/s390-acr-valgrind-s1.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/s390-acr-valgrind-s2.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/s390-acr.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/s390-fpr-valgrind-s1.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/s390-fpr-valgrind-s2.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/s390-fpr.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/s390x-core64-valgrind-s1.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/s390x-core64-valgrind-s2.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/s390x-core64.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/s390x-generic-valgrind.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/s390x-generic.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/s390x-linux64-valgrind-s1.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/s390x-linux64-valgrind-s2.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/s390x-linux64.xml",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/vgpreload_core-x86-linux.so",
>-      "../third_party/valgrind/linux_x86/lib/valgrind/vgpreload_memcheck-x86-linux.so",
>-    ]
>-  }
>-}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/valgrind-webrtc.gni b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/valgrind-webrtc.gni
>deleted file mode 100644
>index b0f45d35dd8..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/valgrind-webrtc.gni
>+++ /dev/null
>@@ -1,43 +0,0 @@
>-# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
>-#
>-# Use of this source code is governed by a BSD-style license
>-# that can be found in the LICENSE file in the root of the source
>-# tree. An additional intellectual property rights grant can be found
>-# in the file PATENTS.  All contributing project authors may
>-# be found in the AUTHORS file in the root of the source tree.
>-
>-# TODO(ehmaldonado): Replace by the new paths once the upstream changes land.
>-# See https://codereview.chromium.org/2526883002/ for valgrind-binaries.gni and
>-# https://codereview.chromium.org/2517953007/ for python-google.gni and
>-# valgrind.gni
>-import("python-google.gni")
>-import("valgrind-binaries.gni")
>-import("valgrind.gni")
>-
>-valgrind_webrtc_dependencies = [
>-                                 "../tools_webrtc/valgrind/gtest_exclude/modules_tests.gtest-memcheck.txt",
>-                                 "../tools_webrtc/valgrind/gtest_exclude/modules_unittests.gtest-memcheck.txt",
>-                                 "../tools_webrtc/valgrind/gtest_exclude/ortc_unittests.gtest-memcheck.txt",
>-                                 "../tools_webrtc/valgrind/gtest_exclude/peerconnection_unittests.gtest-memcheck.txt",
>-                                 "../tools_webrtc/valgrind/gtest_exclude/rtc_media_unittests.gtest-memcheck.txt",
>-                                 "../tools_webrtc/valgrind/gtest_exclude/rtc_media_unittests.gtest-memcheck_mac.txt",
>-                                 "../tools_webrtc/valgrind/gtest_exclude/rtc_pc_unittests.gtest-memcheck.txt",
>-                                 "../tools_webrtc/valgrind/gtest_exclude/rtc_unittests.gtest-memcheck.txt",
>-                                 "../tools_webrtc/valgrind/gtest_exclude/video_engine_tests.gtest-memcheck.txt",
>-                                 "../tools_webrtc/valgrind/gtest_exclude/webrtc_nonparallel_tests.gtest-memcheck.txt",
>-                                 "../tools_webrtc/valgrind/memcheck/suppressions.txt",
>-                                 "../tools_webrtc/valgrind/memcheck/suppressions_mac.txt",
>-                                 "../tools_webrtc/valgrind/memcheck/suppressions_win32.txt",
>-                                 "../tools_webrtc/valgrind/chrome_tests.bat",
>-                                 "../tools_webrtc/valgrind/chrome_tests.py",
>-                                 "../tools_webrtc/valgrind/chrome_tests.sh",
>-                                 "../tools_webrtc/valgrind/common.py",
>-                                 "../tools_webrtc/valgrind/gdb_helper.py",
>-                                 "../tools_webrtc/valgrind/locate_valgrind.sh",
>-                                 "../tools_webrtc/valgrind/memcheck_analyze.py",
>-                                 "../tools_webrtc/valgrind/valgrind.sh",
>-                                 "../tools_webrtc/valgrind/valgrind_test.py",
>-                                 "../tools_webrtc/valgrind/webrtc_tests.py",
>-                                 "../tools_webrtc/valgrind/webrtc_tests.sh",
>-                               ] + valgrind_dependencies + valgrind_binaries +
>-                               python_google
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/valgrind.gni b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/valgrind.gni
>deleted file mode 100644
>index f5f01f838a4..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/valgrind.gni
>+++ /dev/null
>@@ -1,56 +0,0 @@
>-# Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
>-#
>-# Use of this source code is governed by a BSD-style license
>-# that can be found in the LICENSE file in the root of the source
>-# tree. An additional intellectual property rights grant can be found
>-# in the file PATENTS.  All contributing project authors may
>-# be found in the AUTHORS file in the root of the source tree.
>-
>-import("//build_overrides/build.gni")
>-
>-valgrind_dependencies = [
>-  "../tools/valgrind/asan/asan_symbolize.py",
>-  "../tools/valgrind/asan/third_party/__init__.py",
>-  "../tools/valgrind/asan/third_party/asan_symbolize.py",
>-  "../tools/valgrind/fixed_suppressions.sh",
>-  "../tools/valgrind/memcheck/suppressions.txt",
>-  "../tools/valgrind/memcheck/suppressions_linux.txt",
>-  "../tools/valgrind/regrind.sh",
>-  "../tools/valgrind/scan-build.py",
>-  "../tools/valgrind/suppressions.py",
>-  "../tools/valgrind/test_suppressions.py",
>-  "../tools/valgrind/unused_suppressions.py",
>-  "../tools/valgrind/waterfall.sh",
>-]
>-
>-if (build_with_chromium) {
>-  valgrind_dependencies += [
>-    "../tools/valgrind/gtest_exclude/ash_unittests.gtest-memcheck.txt",
>-    "../tools/valgrind/gtest_exclude/aura_unittests.gtest.txt",
>-    "../tools/valgrind/gtest_exclude/base_unittests.gtest.txt",
>-    "../tools/valgrind/gtest_exclude/base_unittests.gtest_win-8.txt",
>-    "../tools/valgrind/gtest_exclude/base_unittests.gtest_win32.txt",
>-    "../tools/valgrind/gtest_exclude/blink_platform_unittests.gtest_win32.txt",
>-    "../tools/valgrind/gtest_exclude/browser_tests.gtest-memcheck.txt",
>-    "../tools/valgrind/gtest_exclude/chromeos_unittests.gtest.txt",
>-    "../tools/valgrind/gtest_exclude/components_unittests.gtest.txt",
>-    "../tools/valgrind/gtest_exclude/content_unittests.gtest.txt",
>-    "../tools/valgrind/gtest_exclude/extensions_unittests.gtest-memcheck.txt",
>-    "../tools/valgrind/gtest_exclude/interactive_ui_tests.gtest.txt",
>-    "../tools/valgrind/gtest_exclude/ipc_tests.gtest.txt",
>-    "../tools/valgrind/gtest_exclude/media_unittests.gtest.txt",
>-    "../tools/valgrind/gtest_exclude/message_center_unittests.gtest.txt",
>-    "../tools/valgrind/gtest_exclude/net_unittests.gtest-memcheck.txt",
>-    "../tools/valgrind/gtest_exclude/net_unittests.gtest.txt",
>-    "../tools/valgrind/gtest_exclude/net_unittests.gtest_linux.txt",
>-    "../tools/valgrind/gtest_exclude/remoting_unittests.gtest_win-8.txt",
>-    "../tools/valgrind/gtest_exclude/sandbox_linux_unittests.gtest.txt",
>-    "../tools/valgrind/gtest_exclude/suppressions.txt",
>-    "../tools/valgrind/gtest_exclude/sync_unit_tests.gtest-asan.txt",
>-    "../tools/valgrind/gtest_exclude/ui_base_unittests.gtest-memcheck.txt",
>-    "../tools/valgrind/gtest_exclude/ui_unittests.gtest-memcheck_linux.txt",
>-    "../tools/valgrind/gtest_exclude/unit_tests.gtest-memcheck.txt",
>-    "../tools/valgrind/gtest_exclude/unit_tests.gtest.txt",
>-    "../tools/valgrind/gtest_exclude/unit_tests.gtest_linux.txt",
>-  ]
>-}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/valgrind.sh b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/valgrind.sh
>deleted file mode 100644
>index a0976e5e0a2..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/valgrind.sh
>+++ /dev/null
>@@ -1,110 +0,0 @@
>-#!/bin/bash
>-
>-# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>-#
>-# Use of this source code is governed by a BSD-style license
>-# that can be found in the LICENSE file in the root of the source
>-# tree. An additional intellectual property rights grant can be found
>-# in the file PATENTS.  All contributing project authors may
>-# be found in the AUTHORS file in the root of the source tree.
>-
>-# This is a small script for manually launching valgrind, along with passing
>-# it the suppression file, and some helpful arguments (automatically attaching
>-# the debugger on failures, etc).  Run it from your repo root, something like:
>-#  $ sh ./tools/valgrind/valgrind.sh ./out/Debug/chrome
>-#
>-# This is mostly intended for running the chrome browser interactively.
>-# To run unit tests, you probably want to run chrome_tests.sh instead.
>-# That's the script used by the valgrind buildbot.
>-
>-export THISDIR=`dirname $0`
>-
>-setup_memcheck() {
>-  RUN_COMMAND="valgrind"
>-
>-  # Prompt to attach gdb when there was an error detected.
>-  DEFAULT_TOOL_FLAGS=("--db-command=gdb -nw %f %p" "--db-attach=yes" \
>-                      # Keep the registers in gdb in sync with the code.
>-                      "--vex-iropt-register-updates=allregs-at-mem-access" \
>-                      # Overwrite newly allocated or freed objects
>-                      # with 0x41 to catch inproper use.
>-                      "--malloc-fill=41" "--free-fill=41" \
>-                      # Increase the size of stacks being tracked.
>-                      "--num-callers=30")
>-}
>-
>-setup_unknown() {
>-  echo "Unknown tool \"$TOOL_NAME\" specified, the result is not guaranteed"
>-  DEFAULT_TOOL_FLAGS=()
>-}
>-
>-set -e
>-
>-if [ $# -eq 0 ]; then
>-  echo "usage: <command to run> <arguments ...>"
>-  exit 1
>-fi
>-
>-TOOL_NAME="memcheck"
>-declare -a DEFAULT_TOOL_FLAGS[0]
>-
>-# Select a tool different from memcheck with --tool=TOOL as a first argument
>-TMP_STR=`echo $1 | sed 's/^\-\-tool=//'`
>-if [ "$TMP_STR" != "$1" ]; then
>-  TOOL_NAME="$TMP_STR"
>-  shift
>-fi
>-
>-if echo "$@" | grep "\-\-tool" ; then
>-  echo "--tool=TOOL must be the first argument" >&2
>-  exit 1
>-fi
>-
>-case $TOOL_NAME in
>-  memcheck*)  setup_memcheck "$1";;
>-  *)          setup_unknown;;
>-esac
>-
>-
>-SUPPRESSIONS="$THISDIR/$TOOL_NAME/suppressions.txt"
>-
>-CHROME_VALGRIND=`sh $THISDIR/locate_valgrind.sh`
>-if [ "$CHROME_VALGRIND" = "" ]
>-then
>-  # locate_valgrind.sh failed
>-  exit 1
>-fi
>-echo "Using valgrind binaries from ${CHROME_VALGRIND}"
>-
>-set -x
>-PATH="${CHROME_VALGRIND}/bin:$PATH"
>-# We need to set these variables to override default lib paths hard-coded into
>-# Valgrind binary.
>-export VALGRIND_LIB="$CHROME_VALGRIND/lib/valgrind"
>-export VALGRIND_LIB_INNER="$CHROME_VALGRIND/lib/valgrind"
>-
>-# G_SLICE=always-malloc: make glib use system malloc
>-# NSS_DISABLE_UNLOAD=1: make nss skip dlclosing dynamically loaded modules,
>-# which would result in "obj:*" in backtraces.
>-# NSS_DISABLE_ARENA_FREE_LIST=1: make nss use system malloc
>-# G_DEBUG=fatal_warnings: make  GTK abort on any critical or warning assertions.
>-# If it crashes on you in the Options menu, you hit bug 19751,
>-# comment out the G_DEBUG=fatal_warnings line.
>-#
>-# GTEST_DEATH_TEST_USE_FORK=1: make gtest death tests valgrind-friendly
>-#
>-# When everyone has the latest valgrind, we might want to add
>-#  --show-possibly-lost=no
>-# to ignore possible but not definite leaks.
>-
>-G_SLICE=always-malloc \
>-NSS_DISABLE_UNLOAD=1 \
>-NSS_DISABLE_ARENA_FREE_LIST=1 \
>-G_DEBUG=fatal_warnings \
>-GTEST_DEATH_TEST_USE_FORK=1 \
>-$RUN_COMMAND \
>-  --trace-children=yes \
>-  --leak-check=yes \
>-  --suppressions="$SUPPRESSIONS" \
>-  "${DEFAULT_TOOL_FLAGS[@]}" \
>-  "$@"
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/valgrind_test.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/valgrind_test.py
>deleted file mode 100644
>index 0fd3d97f111..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/valgrind_test.py
>+++ /dev/null
>@@ -1,517 +0,0 @@
>-#!/usr/bin/env python
>-# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
>-#
>-# Use of this source code is governed by a BSD-style license
>-# that can be found in the LICENSE file in the root of the source
>-# tree. An additional intellectual property rights grant can be found
>-# in the file PATENTS.  All contributing project authors may
>-# be found in the AUTHORS file in the root of the source tree.
>-
>-"""Runs an exe through Valgrind and puts the intermediate files in a
>-directory.
>-"""
>-
>-import datetime
>-import glob
>-import logging
>-import optparse
>-import os
>-import re
>-import shutil
>-import stat
>-import subprocess
>-import sys
>-import tempfile
>-
>-import common
>-
>-import memcheck_analyze
>-
>-class BaseTool(object):
>-  """Abstract class for running dynamic error detection tools.
>-
>-  Always subclass this and implement ToolCommand with framework- and
>-  tool-specific stuff.
>-  """
>-
>-  def __init__(self):
>-    temp_parent_dir = None
>-    self.log_parent_dir = ""
>-    if common.IsWindows():
>-      # gpu process on Windows Vista+ runs at Low Integrity and can only
>-      # write to certain directories (http://crbug.com/119131)
>-      #
>-      # TODO(bruening): if scripts die in middle and don't clean up temp
>-      # dir, we'll accumulate files in profile dir.  should remove
>-      # really old files automatically.
>-      profile = os.getenv("USERPROFILE")
>-      if profile:
>-        self.log_parent_dir = profile + "\\AppData\\LocalLow\\"
>-        if os.path.exists(self.log_parent_dir):
>-          self.log_parent_dir = common.NormalizeWindowsPath(self.log_parent_dir)
>-          temp_parent_dir = self.log_parent_dir
>-    # Generated every time (even when overridden)
>-    self.temp_dir = tempfile.mkdtemp(prefix="vg_logs_", dir=temp_parent_dir)
>-    self.log_dir = self.temp_dir # overridable by --keep_logs
>-    self.option_parser_hooks = []
>-    # TODO(glider): we may not need some of the env vars on some of the
>-    # platforms.
>-    self._env = {
>-      "G_SLICE" : "always-malloc",
>-      "NSS_DISABLE_UNLOAD" : "1",
>-      "NSS_DISABLE_ARENA_FREE_LIST" : "1",
>-      "GTEST_DEATH_TEST_USE_FORK": "1",
>-    }
>-
>-  def ToolName(self):
>-    raise NotImplementedError, "This method should be implemented " \
>-                               "in the tool-specific subclass"
>-
>-  def Analyze(self, check_sanity=False):
>-    raise NotImplementedError, "This method should be implemented " \
>-                               "in the tool-specific subclass"
>-
>-  def RegisterOptionParserHook(self, hook):
>-    # Frameworks and tools can add their own flags to the parser.
>-    self.option_parser_hooks.append(hook)
>-
>-  def CreateOptionParser(self):
>-    # Defines Chromium-specific flags.
>-    self._parser = optparse.OptionParser("usage: %prog [options] <program to "
>-                                         "test>")
>-    self._parser.disable_interspersed_args()
>-    self._parser.add_option("-t", "--timeout",
>-                      dest="timeout", metavar="TIMEOUT", default=10000,
>-                      help="timeout in seconds for the run (default 10000)")
>-    self._parser.add_option("", "--build-dir",
>-                            help="the location of the compiler output")
>-    self._parser.add_option("", "--source-dir",
>-                            help="path to top of source tree for this build"
>-                                 "(used to normalize source paths in baseline)")
>-    self._parser.add_option("", "--gtest_filter", default="",
>-                            help="which test case to run")
>-    self._parser.add_option("", "--gtest_repeat",
>-                            help="how many times to run each test")
>-    self._parser.add_option("", "--gtest_print_time", action="store_true",
>-                            default=False,
>-                            help="show how long each test takes")
>-    self._parser.add_option("", "--ignore_exit_code", action="store_true",
>-                            default=False,
>-                            help="ignore exit code of the test "
>-                                 "(e.g. test failures)")
>-    self._parser.add_option("", "--keep_logs", action="store_true",
>-                            default=False,
>-                            help="store memory tool logs in the <tool>.logs "
>-                                 "directory instead of /tmp.\nThis can be "
>-                                 "useful for tool developers/maintainers.\n"
>-                                 "Please note that the <tool>.logs directory "
>-                                 "will be clobbered on tool startup.")
>-
>-    # To add framework- or tool-specific flags, please add a hook using
>-    # RegisterOptionParserHook in the corresponding subclass.
>-    # See ValgrindTool for an example.
>-    for hook in self.option_parser_hooks:
>-      hook(self, self._parser)
>-
>-  def ParseArgv(self, args):
>-    self.CreateOptionParser()
>-
>-    # self._tool_flags will store those tool flags which we don't parse
>-    # manually in this script.
>-    self._tool_flags = []
>-    known_args = []
>-
>-    """ We assume that the first argument not starting with "-" is a program
>-    name and all the following flags should be passed to the program.
>-    TODO(timurrrr): customize optparse instead
>-    """
>-    while len(args) > 0 and args[0][:1] == "-":
>-      arg = args[0]
>-      if (arg == "--"):
>-        break
>-      if self._parser.has_option(arg.split("=")[0]):
>-        known_args += [arg]
>-      else:
>-        self._tool_flags += [arg]
>-      args = args[1:]
>-
>-    if len(args) > 0:
>-      known_args += args
>-
>-    self._options, self._args = self._parser.parse_args(known_args)
>-
>-    self._timeout = int(self._options.timeout)
>-    self._source_dir = self._options.source_dir
>-    if self._options.keep_logs:
>-      # log_parent_dir has trailing slash if non-empty
>-      self.log_dir = self.log_parent_dir + "%s.logs" % self.ToolName()
>-      if os.path.exists(self.log_dir):
>-        shutil.rmtree(self.log_dir)
>-      os.mkdir(self.log_dir)
>-      logging.info("Logs are in " + self.log_dir)
>-
>-    self._ignore_exit_code = self._options.ignore_exit_code
>-    if self._options.gtest_filter != "":
>-      self._args.append("--gtest_filter=%s" % self._options.gtest_filter)
>-    if self._options.gtest_repeat:
>-      self._args.append("--gtest_repeat=%s" % self._options.gtest_repeat)
>-    if self._options.gtest_print_time:
>-      self._args.append("--gtest_print_time")
>-
>-    return True
>-
>-  def Setup(self, args):
>-    return self.ParseArgv(args)
>-
>-  def ToolCommand(self):
>-    raise NotImplementedError, "This method should be implemented " \
>-                               "in the tool-specific subclass"
>-
>-  def Cleanup(self):
>-    # You may override it in the tool-specific subclass
>-    pass
>-
>-  def Execute(self):
>-    """ Execute the app to be tested after successful instrumentation.
>-    Full execution command-line provided by subclassers via proc."""
>-    logging.info("starting execution...")
>-    proc = self.ToolCommand()
>-    for var in self._env:
>-      common.PutEnvAndLog(var, self._env[var])
>-    return common.RunSubprocess(proc, self._timeout)
>-
>-  def RunTestsAndAnalyze(self, check_sanity):
>-    exec_retcode = self.Execute()
>-    analyze_retcode = self.Analyze(check_sanity)
>-
>-    if analyze_retcode:
>-      logging.error("Analyze failed.")
>-      logging.info("Search the log for '[ERROR]' to see the error reports.")
>-      return analyze_retcode
>-
>-    if exec_retcode:
>-      if self._ignore_exit_code:
>-        logging.info("Test execution failed, but the exit code is ignored.")
>-      else:
>-        logging.error("Test execution failed.")
>-        return exec_retcode
>-    else:
>-      logging.info("Test execution completed successfully.")
>-
>-    if not analyze_retcode:
>-      logging.info("Analysis completed successfully.")
>-
>-    return 0
>-
>-  def Main(self, args, check_sanity, min_runtime_in_seconds):
>-    """Call this to run through the whole process: Setup, Execute, Analyze"""
>-    start_time = datetime.datetime.now()
>-    retcode = -1
>-    if self.Setup(args):
>-      retcode = self.RunTestsAndAnalyze(check_sanity)
>-      shutil.rmtree(self.temp_dir, ignore_errors=True)
>-      self.Cleanup()
>-    else:
>-      logging.error("Setup failed")
>-    end_time = datetime.datetime.now()
>-    runtime_in_seconds = (end_time - start_time).seconds
>-    hours = runtime_in_seconds / 3600
>-    seconds = runtime_in_seconds % 3600
>-    minutes = seconds / 60
>-    seconds = seconds % 60
>-    logging.info("elapsed time: %02d:%02d:%02d" % (hours, minutes, seconds))
>-    if (min_runtime_in_seconds > 0 and
>-        runtime_in_seconds < min_runtime_in_seconds):
>-      logging.error("Layout tests finished too quickly. "
>-                    "It should have taken at least %d seconds. "
>-                    "Something went wrong?" % min_runtime_in_seconds)
>-      retcode = -1
>-    return retcode
>-
>-  def Run(self, args, module, min_runtime_in_seconds=0):
>-    MODULES_TO_SANITY_CHECK = ["base"]
>-
>-    check_sanity = module in MODULES_TO_SANITY_CHECK
>-    return self.Main(args, check_sanity, min_runtime_in_seconds)
>-
>-
>-class ValgrindTool(BaseTool):
>-  """Abstract class for running Valgrind tools.
>-
>-  Always subclass this and implement ToolSpecificFlags() and
>-  ExtendOptionParser() for tool-specific stuff.
>-  """
>-  def __init__(self):
>-    super(ValgrindTool, self).__init__()
>-    self.RegisterOptionParserHook(ValgrindTool.ExtendOptionParser)
>-
>-  def UseXML(self):
>-    # Override if tool prefers nonxml output
>-    return True
>-
>-  def ExtendOptionParser(self, parser):
>-    parser.add_option("", "--suppressions", default=[],
>-                            action="append",
>-                            help="path to a valgrind suppression file")
>-    parser.add_option("", "--indirect", action="store_true",
>-                            default=False,
>-                            help="set BROWSER_WRAPPER rather than "
>-                                 "running valgrind directly")
>-    parser.add_option("", "--indirect_webkit_layout", action="store_true",
>-                            default=False,
>-                            help="set --wrapper rather than running Dr. Memory "
>-                                 "directly.")
>-    parser.add_option("", "--trace_children", action="store_true",
>-                            default=False,
>-                            help="also trace child processes")
>-    parser.add_option("", "--num-callers",
>-                            dest="num_callers", default=30,
>-                            help="number of callers to show in stack traces")
>-    parser.add_option("", "--generate_dsym", action="store_true",
>-                          default=False,
>-                          help="Generate .dSYM file on Mac if needed. Slow!")
>-
>-  def Setup(self, args):
>-    if not BaseTool.Setup(self, args):
>-      return False
>-    return True
>-
>-  def ToolCommand(self):
>-    """Get the valgrind command to run."""
>-    # Note that self._args begins with the exe to be run.
>-    tool_name = self.ToolName()
>-
>-    # Construct the valgrind command.
>-    if 'CHROME_VALGRIND' in os.environ:
>-      path = os.path.join(os.environ['CHROME_VALGRIND'], "bin", "valgrind")
>-    else:
>-      path = "valgrind"
>-    proc = [path, "--tool=%s" % tool_name]
>-
>-    proc += ["--num-callers=%i" % int(self._options.num_callers)]
>-
>-    if self._options.trace_children:
>-      proc += ["--trace-children=yes"]
>-      proc += ["--trace-children-skip='*dbus-daemon*'"]
>-      proc += ["--trace-children-skip='*dbus-launch*'"]
>-      proc += ["--trace-children-skip='*perl*'"]
>-      proc += ["--trace-children-skip='*python*'"]
>-      # This is really Python, but for some reason Valgrind follows it.
>-      proc += ["--trace-children-skip='*lsb_release*'"]
>-
>-    proc += self.ToolSpecificFlags()
>-    proc += self._tool_flags
>-
>-    suppression_count = 0
>-    for suppression_file in self._options.suppressions:
>-      if os.path.exists(suppression_file):
>-        suppression_count += 1
>-        proc += ["--suppressions=%s" % suppression_file]
>-
>-    if not suppression_count:
>-      logging.warning("WARNING: NOT USING SUPPRESSIONS!")
>-
>-    logfilename = self.log_dir + ("/%s." % tool_name) + "%p"
>-    if self.UseXML():
>-      proc += ["--xml=yes", "--xml-file=" + logfilename]
>-    else:
>-      proc += ["--log-file=" + logfilename]
>-
>-    # The Valgrind command is constructed.
>-
>-    # Handle --indirect_webkit_layout separately.
>-    if self._options.indirect_webkit_layout:
>-      # Need to create the wrapper before modifying |proc|.
>-      wrapper = self.CreateBrowserWrapper(proc, webkit=True)
>-      proc = self._args
>-      proc.append("--wrapper")
>-      proc.append(wrapper)
>-      return proc
>-
>-    if self._options.indirect:
>-      wrapper = self.CreateBrowserWrapper(proc)
>-      os.environ["BROWSER_WRAPPER"] = wrapper
>-      logging.info('export BROWSER_WRAPPER=' + wrapper)
>-      proc = []
>-    proc += self._args
>-    return proc
>-
>-  def ToolSpecificFlags(self):
>-    raise NotImplementedError, "This method should be implemented " \
>-                               "in the tool-specific subclass"
>-
>-  def CreateBrowserWrapper(self, proc, webkit=False):
>-    """The program being run invokes Python or something else that can't stand
>-    to be valgrinded, and also invokes the Chrome browser. In this case, use a
>-    magic wrapper to only valgrind the Chrome browser. Build the wrapper here.
>-    Returns the path to the wrapper. It's up to the caller to use the wrapper
>-    appropriately.
>-    """
>-    command = " ".join(proc)
>-    # Add the PID of the browser wrapper to the logfile names so we can
>-    # separate log files for different UI tests at the analyze stage.
>-    command = command.replace("%p", "$$.%p")
>-
>-    (fd, indirect_fname) = tempfile.mkstemp(dir=self.log_dir,
>-                                            prefix="browser_wrapper.",
>-                                            text=True)
>-    f = os.fdopen(fd, "w")
>-    f.write('#!/bin/bash\n'
>-            'echo "Started Valgrind wrapper for this test, PID=$$" >&2\n')
>-
>-    f.write('DIR=`dirname $0`\n'
>-            'TESTNAME_FILE=$DIR/testcase.$$.name\n\n')
>-
>-    if webkit:
>-      # Webkit layout_tests pass the URL as the first line of stdin.
>-      f.write('tee $TESTNAME_FILE | %s "$@"\n' % command)
>-    else:
>-      # Try to get the test case name by looking at the program arguments.
>-      # i.e. Chromium ui_tests used --test-name arg.
>-      # TODO(timurrrr): This doesn't handle "--test-name Test.Name"
>-      # TODO(timurrrr): ui_tests are dead. Where do we use the non-webkit
>-      # wrapper now? browser_tests? What do they do?
>-      f.write('for arg in $@\ndo\n'
>-              '  if [[ "$arg" =~ --test-name=(.*) ]]\n  then\n'
>-              '    echo ${BASH_REMATCH[1]} >$TESTNAME_FILE\n'
>-              '  fi\n'
>-              'done\n\n'
>-              '%s "$@"\n' % command)
>-
>-    f.close()
>-    os.chmod(indirect_fname, stat.S_IRUSR|stat.S_IXUSR)
>-    return indirect_fname
>-
>-  def CreateAnalyzer(self):
>-    raise NotImplementedError, "This method should be implemented " \
>-                               "in the tool-specific subclass"
>-
>-  def GetAnalyzeResults(self, check_sanity=False):
>-    # Glob all the files in the log directory
>-    filenames = glob.glob(self.log_dir + "/" + self.ToolName() + ".*")
>-
>-    # If we have browser wrapper, the logfiles are named as
>-    # "toolname.wrapper_PID.valgrind_PID".
>-    # Let's extract the list of wrapper_PIDs and name it ppids
>-    ppids = set([int(f.split(".")[-2]) \
>-                for f in filenames if re.search("\.[0-9]+\.[0-9]+$", f)])
>-
>-    analyzer = self.CreateAnalyzer()
>-    if len(ppids) == 0:
>-      # Fast path - no browser wrapper was set.
>-      return analyzer.Report(filenames, None, check_sanity)
>-
>-    ret = 0
>-    for ppid in ppids:
>-      testcase_name = None
>-      try:
>-        f = open(self.log_dir + ("/testcase.%d.name" % ppid))
>-        testcase_name = f.read().strip()
>-        f.close()
>-        wk_layout_prefix="third_party/WebKit/LayoutTests/"
>-        wk_prefix_at = testcase_name.rfind(wk_layout_prefix)
>-        if wk_prefix_at != -1:
>-          testcase_name = testcase_name[wk_prefix_at + len(wk_layout_prefix):]
>-      except IOError:
>-        pass
>-      print "====================================================="
>-      print " Below is the report for valgrind wrapper PID=%d." % ppid
>-      if testcase_name:
>-        print " It was used while running the `%s` test." % testcase_name
>-      else:
>-        print " You can find the corresponding test"
>-        print " by searching the above log for 'PID=%d'" % ppid
>-      sys.stdout.flush()
>-
>-      ppid_filenames = [f for f in filenames \
>-                        if re.search("\.%d\.[0-9]+$" % ppid, f)]
>-      # check_sanity won't work with browser wrappers
>-      assert check_sanity == False
>-      ret |= analyzer.Report(ppid_filenames, testcase_name)
>-      print "====================================================="
>-      sys.stdout.flush()
>-
>-    if ret != 0:
>-      print ""
>-      print "The Valgrind reports are grouped by test names."
>-      print "Each test has its PID printed in the log when the test was run"
>-      print "and at the beginning of its Valgrind report."
>-      print "Hint: you can search for the reports by Ctrl+F -> `=#`"
>-      sys.stdout.flush()
>-
>-    return ret
>-
>-
>-# TODO(timurrrr): Split into a separate file.
>-class Memcheck(ValgrindTool):
>-  """Memcheck
>-  Dynamic memory error detector for Linux & Mac
>-
>-  http://valgrind.org/info/tools.html#memcheck
>-  """
>-
>-  def __init__(self):
>-    super(Memcheck, self).__init__()
>-    self.RegisterOptionParserHook(Memcheck.ExtendOptionParser)
>-
>-  def ToolName(self):
>-    return "memcheck"
>-
>-  def ExtendOptionParser(self, parser):
>-    parser.add_option("--leak-check", "--leak_check", type="string",
>-                      default="yes",  # --leak-check=yes is equivalent of =full
>-                      help="perform leak checking at the end of the run")
>-    parser.add_option("", "--show_all_leaks", action="store_true",
>-                      default=False,
>-                      help="also show less blatant leaks")
>-    parser.add_option("", "--track_origins", action="store_true",
>-                      default=False,
>-                      help="Show whence uninitialized bytes came. 30% slower.")
>-
>-  def ToolSpecificFlags(self):
>-    ret = ["--gen-suppressions=all", "--demangle=no"]
>-    ret += ["--leak-check=%s" % self._options.leak_check]
>-
>-    if self._options.show_all_leaks:
>-      ret += ["--show-reachable=yes"]
>-    else:
>-      ret += ["--show-possibly-lost=no"]
>-
>-    if self._options.track_origins:
>-      ret += ["--track-origins=yes"]
>-
>-    # TODO(glider): this is a temporary workaround for http://crbug.com/51716
>-    # Let's see whether it helps.
>-    if common.IsMac():
>-      ret += ["--smc-check=all"]
>-
>-    return ret
>-
>-  def CreateAnalyzer(self):
>-    use_gdb = common.IsMac()
>-    return memcheck_analyze.MemcheckAnalyzer(self._source_dir,
>-                                            self._options.show_all_leaks,
>-                                            use_gdb=use_gdb)
>-
>-  def Analyze(self, check_sanity=False):
>-    ret = self.GetAnalyzeResults(check_sanity)
>-
>-    if ret != 0:
>-      logging.info("Please see http://dev.chromium.org/developers/how-tos/"
>-                   "using-valgrind for the info on Memcheck/Valgrind")
>-    return ret
>-
>-
>-class ToolFactory:
>-  def Create(self, tool_name):
>-    if tool_name == "memcheck":
>-      return Memcheck()
>-    try:
>-      platform_name = common.PlatformNames()[0]
>-    except common.NotImplementedError:
>-      platform_name = sys.platform + "(Unknown)"
>-    raise RuntimeError, "Unknown tool (tool=%s, platform=%s)" % (tool_name,
>-                                                                 platform_name)
>-
>-def CreateTool(tool):
>-  return ToolFactory().Create(tool)
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/webrtc_tests.py b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/webrtc_tests.py
>deleted file mode 100644
>index 7feb830f098..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/webrtc_tests.py
>+++ /dev/null
>@@ -1,161 +0,0 @@
>-#!/usr/bin/env python
>-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>-#
>-# Use of this source code is governed by a BSD-style license
>-# that can be found in the LICENSE file in the root of the source
>-# tree. An additional intellectual property rights grant can be found
>-# in the file PATENTS.  All contributing project authors may
>-# be found in the AUTHORS file in the root of the source tree.
>-
>-"""Runs various WebRTC tests through valgrind_test.py.
>-
>-This script inherits the chrome_tests.py in Chrome, but allows running any test
>-instead of only the hard-coded ones. It uses the -t cmdline flag to do this, and
>-only supports specifying a single test for each run.
>-
>-Suppression files:
>-The Chrome valgrind directory we use as a DEPS dependency contains the following
>-suppression files:
>-  valgrind/memcheck/suppressions.txt
>-  valgrind/memcheck/suppressions_mac.txt
>-Since they're referenced from the chrome_tests.py script, we have similar files
>-below the directory of this script. When executing, this script will setup both
>-Chrome's suppression files and our own, so we can easily maintain WebRTC
>-specific suppressions in our own files.
>-"""
>-
>-import argparse
>-import logging
>-import optparse
>-import os
>-import sys
>-
>-import logging_utils
>-import path_utils
>-
>-import chrome_tests
>-
>-
>-class WebRTCTest(chrome_tests.ChromeTests):
>-  """Class that handles setup of suppressions for WebRTC.
>-
>-  Everything else is inherited from chrome_tests.ChromeTests.
>-  """
>-
>-  def __init__(self, test_name, options, args, test_in_chrome_tests):
>-    """Create a WebRTC test.
>-    Args:
>-      test_name: Short name for the test executable (no path).
>-      options: options to pass to ChromeTests.
>-      args: args to pass to ChromeTests.
>-      test_in_chrome_tests: The name of the test configuration in ChromeTests.
>-    """
>-    self._test_name = test_name
>-    chrome_tests.ChromeTests.__init__(self, options, args, test_in_chrome_tests)
>-
>-  def _DefaultCommand(self, tool, exe=None, valgrind_test_args=None):
>-    """Override command-building method so we can add more suppressions."""
>-    cmd = chrome_tests.ChromeTests._DefaultCommand(self, tool, exe,
>-                                                   valgrind_test_args)
>-
>-    # Add gtest filters, if found.
>-    chrome_tests.ChromeTests._AppendGtestFilter(self, tool, self._test_name,
>-                                                cmd)
>-
>-    # When ChromeTests._DefaultCommand has executed, it has setup suppression
>-    # files based on what's found in the memcheck/ subdirectory of
>-    # this script's location. If Mac or Windows is executing, additional
>-    # platform specific files have also been added.
>-    # Since only the ones located below this directory are added, we must also
>-    # add the ones maintained by Chrome, located in ../../tools/valgrind.
>-
>-    # The idea is to look for --suppression arguments in the cmd list and add a
>-    # modified copy of each suppression file, for the corresponding file in
>-    # ../../tools/valgrind.
>-    script_dir = path_utils.ScriptDir()
>-    checkout_src = os.path.abspath(os.path.join(script_dir, os.pardir,
>-                                                os.pardir))
>-    new_dir = os.path.join(checkout_src, 'tools', 'valgrind')
>-    add_suppressions = []
>-    for token in cmd:
>-      if '--suppressions' in token:
>-        add_suppressions.append(token.replace(script_dir, new_dir))
>-    return add_suppressions + cmd
>-
>-
>-def main(_):
>-  parser = optparse.OptionParser(
>-      'usage: %prog -b <dir> -t <test> -- <test args>')
>-  parser.disable_interspersed_args()
>-  parser.add_option('-b', '--build-dir',
>-                    help=('Location of the compiler output. Can only be used '
>-                          'when the test argument does not contain this path.'))
>-  parser.add_option("--target", help="Debug or Release")
>-  parser.add_option('-t', '--test', help='Test to run.')
>-  parser.add_option('', '--baseline', action='store_true', default=False,
>-                    help='Generate baseline data instead of validating')
>-  parser.add_option('', '--gtest_filter',
>-                    help='Additional arguments to --gtest_filter')
>-  parser.add_option('', '--gtest_repeat',
>-                    help='Argument for --gtest_repeat')
>-  parser.add_option("--gtest_shuffle", action="store_true", default=False,
>-                    help="Randomize tests' orders on every iteration.")
>-  parser.add_option("--gtest_break_on_failure", action="store_true",
>-                    default=False,
>-                    help="Drop in to debugger on assertion failure. Also "
>-                         "useful for forcing tests to exit with a stack dump "
>-                         "on the first assertion failure when running with "
>-                         "--gtest_repeat=-1")
>-  parser.add_option('-v', '--verbose', action='store_true', default=False,
>-                    help='Verbose output - enable debug log messages')
>-  parser.add_option('', '--tool', dest='valgrind_tool', default='memcheck',
>-                    help='Specify a valgrind tool to run the tests under')
>-  parser.add_option('', '--tool_flags', dest='valgrind_tool_flags', default='',
>-                    help='Specify custom flags for the selected valgrind tool')
>-  parser.add_option('', '--keep_logs', action='store_true', default=False,
>-                    help=('Store memory tool logs in the <tool>.logs directory '
>-                          'instead of /tmp.\nThis can be useful for tool '
>-                          'developers/maintainers.\nPlease note that the <tool>'
>-                          '.logs directory will be clobbered on tool startup.'))
>-  parser.add_option("--test-launcher-bot-mode", action="store_true",
>-                    help="run the tests with --test-launcher-bot-mode")
>-  parser.add_option("--test-launcher-total-shards", type=int,
>-                    help="run the tests with --test-launcher-total-shards")
>-  parser.add_option("--test-launcher-shard-index", type=int,
>-                    help="run the tests with --test-launcher-shard-index")
>-  options, args = parser.parse_args()
>-
>-  ignore_parser = argparse.ArgumentParser()
>-  # Ignore Chromium-specific flags
>-  # TODO(oprypin): Remove (bugs.webrtc.org/8115)
>-  ignore_parser.add_argument('--isolated-script-test-output',
>-                             type=str, default=None)
>-  ignore_parser.add_argument('--isolated-script-test-perf-output',
>-                             type=str, default=None)
>-  _, args = ignore_parser.parse_known_args(args)
>-
>-  if options.verbose:
>-    logging_utils.config_root(logging.DEBUG)
>-  else:
>-    logging_utils.config_root()
>-
>-  if not options.test:
>-    parser.error('--test not specified')
>-
>-  # Support build dir both with and without the target.
>-  if (options.target and options.build_dir and
>-      not options.build_dir.endswith(options.target)):
>-    options.build_dir = os.path.join(options.build_dir, options.target)
>-
>-  # If --build_dir is provided, prepend it to the test executable if needed.
>-  test_executable = options.test
>-  if options.build_dir and not test_executable.startswith(options.build_dir):
>-    test_executable = os.path.join(options.build_dir, test_executable)
>-  args = [test_executable] + args
>-
>-  test = WebRTCTest(options.test, options, args, 'cmdline')
>-  return test.Run()
>-
>-if __name__ == '__main__':
>-  return_code = main(sys.argv)
>-  sys.exit(return_code)
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/webrtc_tests.sh b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/webrtc_tests.sh
>deleted file mode 100644
>index e8cd285474e..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/valgrind/webrtc_tests.sh
>+++ /dev/null
>@@ -1,102 +0,0 @@
>-#!/bin/bash
>-
>-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
>-#
>-# Use of this source code is governed by a BSD-style license
>-# that can be found in the LICENSE file in the root of the source
>-# tree. An additional intellectual property rights grant can be found
>-# in the file PATENTS.  All contributing project authors may
>-# be found in the AUTHORS file in the root of the source tree.
>-
>-# Set up some paths and re-direct the arguments to webrtc_tests.py
>-
>-# This script is a copy of the chrome_tests.sh wrapper script with the following
>-# changes:
>-# - The locate_valgrind.sh of Chromium's Valgrind scripts dir is used to locate
>-#   the Valgrind framework install. If it fails a fallback path is used instead
>-#   (../../chromium/src/third_party/valgrind/linux_x64) and a warning message
>-#   is showed by |show_locate_valgrind_failed_warning|.
>-# - webrtc_tests.py is invoked instead of chrome_tests.py.
>-# - Chromium's Valgrind scripts directory is added to the PYTHONPATH to make it
>-#   possible to execute the Python scripts properly.
>-
>-export THISDIR=`dirname $0`
>-ARGV_COPY="$@"
>-
>-# We need to set CHROME_VALGRIND iff using Memcheck:
>-#   tools_webrtc/valgrind/webrtc_tests.sh --tool memcheck
>-# or
>-#   tools_webrtc/valgrind/webrtc_tests.sh --tool=memcheck
>-tool="memcheck"  # Default to memcheck.
>-while (( "$#" ))
>-do
>-  if [[ "$1" == "--tool" ]]
>-  then
>-    tool="$2"
>-    shift
>-  elif [[ "$1" =~ --tool=(.*) ]]
>-  then
>-    tool="${BASH_REMATCH[1]}"
>-  fi
>-  shift
>-done
>-
>-NEEDS_VALGRIND=0
>-
>-case "$tool" in
>-  "memcheck")
>-    NEEDS_VALGRIND=1
>-    ;;
>-esac
>-
>-# For WebRTC, we'll use the locate_valgrind.sh script in Chromium's Valgrind
>-# scripts dir to locate the Valgrind framework install
>-CHROME_VALGRIND_SCRIPTS=$THISDIR/../../tools/valgrind
>-
>-if [ "$NEEDS_VALGRIND" == "1" ]
>-then
>-  CHROME_VALGRIND=`sh $THISDIR/locate_valgrind.sh`
>-  if [ "$CHROME_VALGRIND" = "" ]
>-  then
>-    CHROME_VALGRIND=../../src/third_party/valgrind/linux_x64
>-    echo
>-    echo "-------------------- WARNING ------------------------"
>-    echo "locate_valgrind.sh failed."
>-    echo "Using $CHROME_VALGRIND as a fallback location."
>-    echo "This might be because:"
>-    echo "1) This is a swarming bot"
>-    echo "2) You haven't set up the valgrind binaries correctly."
>-    echo "In this case, please make sure you have followed the instructions at"
>-    echo "http://www.chromium.org/developers/how-tos/using-valgrind/get-valgrind"
>-    echo "Notice: In the .gclient file, you need to add this for the 'src'"
>-    echo "solution since our directory structure is different from Chromium's:"
>-    echo "\"custom_deps\": {"
>-    echo "  \"src/third_party/valgrind\":"
>-    echo "      \"https://chromium.googlesource.com/chromium/deps/valgrind/binaries\","
>-    echo "},"
>-    echo "-----------------------------------------------------"
>-    echo
>-  fi
>-  echo "Using valgrind binaries from ${CHROME_VALGRIND}"
>-
>-  PATH="${CHROME_VALGRIND}/bin:$PATH"
>-  # We need to set these variables to override default lib paths hard-coded into
>-  # Valgrind binary.
>-  export VALGRIND_LIB="$CHROME_VALGRIND/lib/valgrind"
>-  export VALGRIND_LIB_INNER="$CHROME_VALGRIND/lib/valgrind"
>-
>-  # Clean up some /tmp directories that might be stale due to interrupted
>-  # chrome_tests.py execution.
>-  # FYI:
>-  #   -mtime +1  <- only print files modified more than 24h ago,
>-  #   -print0/-0 are needed to handle possible newlines in the filenames.
>-  echo "Cleanup /tmp from Valgrind stuff"
>-  find /tmp -maxdepth 1 \(\
>-        -name "vgdb-pipe-*" -or -name "vg_logs_*" -or -name "valgrind.*" \
>-      \) -mtime +1 -print0 | xargs -0 rm -rf
>-fi
>-
>-# Add Chrome's Valgrind scripts dir to the PYTHON_PATH since it contains
>-# the scripts that are needed for this script to run
>-PYTHONPATH=$THISDIR/../../tools/python/google:$CHROME_VALGRIND_SCRIPTS python \
>-           "$THISDIR/webrtc_tests.py" $ARGV_COPY
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/video_quality_toolchain/linux/frame_analyzer.sha1 b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/video_quality_toolchain/linux/frame_analyzer.sha1
>new file mode 100644
>index 00000000000..9f7d4e9f746
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/video_quality_toolchain/linux/frame_analyzer.sha1
>@@ -0,0 +1 @@
>+fe3e1e5253a1a49277e3ec9c0699ccf059cd8601
>\ No newline at end of file
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/whitespace.txt b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/whitespace.txt
>index 633f23318e2..685a8768f31 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/whitespace.txt
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/tools_webrtc/whitespace.txt
>@@ -9,4 +9,7 @@ Holidays are coming, beware! Bought any gifts yet?
> Lemur was here :)
> Kjellander as well :-o
> 
>-Foo Bar Baz
>+Foo Bar Baz Bur
>+
>+Alios ego vidi ventos; alias prospexi animo procellas
>+  - Cicero
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/typedefs.h b/Source/ThirdParty/libwebrtc/Source/webrtc/typedefs.h
>deleted file mode 100644
>index 629cded1afd..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/typedefs.h
>+++ /dev/null
>@@ -1,125 +0,0 @@
>-/*
>- *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-// This file contains platform-specific typedefs and defines.
>-// Much of it is derived from Chromium's build/build_config.h.
>-
>-#ifndef TYPEDEFS_H_
>-#define TYPEDEFS_H_
>-
>-// Processor architecture detection.  For more info on what's defined, see:
>-//   http://msdn.microsoft.com/en-us/library/b0084kay.aspx
>-//   http://www.agner.org/optimize/calling_conventions.pdf
>-//   or with gcc, run: "echo | gcc -E -dM -"
>-#if defined(_M_X64) || defined(__x86_64__)
>-#define WEBRTC_ARCH_X86_FAMILY
>-#define WEBRTC_ARCH_X86_64
>-#define WEBRTC_ARCH_64_BITS
>-#define WEBRTC_ARCH_LITTLE_ENDIAN
>-#elif defined(__aarch64__)
>-#define WEBRTC_ARCH_ARM_FAMILY
>-#define WEBRTC_ARCH_64_BITS
>-#define WEBRTC_ARCH_LITTLE_ENDIAN
>-#elif defined(_M_IX86) || defined(__i386__)
>-#define WEBRTC_ARCH_X86_FAMILY
>-#define WEBRTC_ARCH_X86
>-#define WEBRTC_ARCH_32_BITS
>-#define WEBRTC_ARCH_LITTLE_ENDIAN
>-#elif defined(__ARMEL__)
>-#define WEBRTC_ARCH_ARM_FAMILY
>-#define WEBRTC_ARCH_32_BITS
>-#define WEBRTC_ARCH_LITTLE_ENDIAN
>-#elif defined(__MIPSEL__)
>-#define WEBRTC_ARCH_MIPS_FAMILY
>-#if defined(__LP64__)
>-#define WEBRTC_ARCH_64_BITS
>-#else
>-#define WEBRTC_ARCH_32_BITS
>-#endif
>-#define WEBRTC_ARCH_LITTLE_ENDIAN
>-#elif defined(__pnacl__)
>-#define WEBRTC_ARCH_32_BITS
>-#define WEBRTC_ARCH_LITTLE_ENDIAN
>-#else
>-#error Please add support for your architecture in typedefs.h
>-#endif
>-
>-#if !(defined(WEBRTC_ARCH_LITTLE_ENDIAN) ^ defined(WEBRTC_ARCH_BIG_ENDIAN))
>-#error Define either WEBRTC_ARCH_LITTLE_ENDIAN or WEBRTC_ARCH_BIG_ENDIAN
>-#endif
>-
>-// TODO(zhongwei.yao): WEBRTC_CPU_DETECTION is only used in one place; we should
>-// probably just remove it.
>-#if (defined(WEBRTC_ARCH_X86_FAMILY) && !defined(__SSE2__))
>-#define WEBRTC_CPU_DETECTION
>-#endif
>-
>-#include <stdint.h>
>-
>-// Annotate a function indicating the caller must examine the return value.
>-// Use like:
>-//   int foo() RTC_WARN_UNUSED_RESULT;
>-// To explicitly ignore a result, cast to void.
>-// TODO(kwiberg): Remove when we can use [[nodiscard]] from C++17.
>-#if defined(__clang__)
>-#define RTC_WARN_UNUSED_RESULT __attribute__((__warn_unused_result__))
>-#elif defined(__GNUC__)
>-// gcc has a __warn_unused_result__ attribute, but you can't quiet it by
>-// casting to void, so we don't use it.
>-#define RTC_WARN_UNUSED_RESULT
>-#else
>-#define RTC_WARN_UNUSED_RESULT
>-#endif
>-
>-// Put after a variable that might not be used, to prevent compiler warnings:
>-//   int result ATTRIBUTE_UNUSED = DoSomething();
>-//   assert(result == 17);
>-// Deprecated since it only works with GCC & clang. See RTC_UNUSED below.
>-// TODO(terelius): Remove.
>-#ifndef ATTRIBUTE_UNUSED
>-#if defined(__GNUC__) || defined(__clang__)
>-#define ATTRIBUTE_UNUSED __attribute__ ((__unused__))
>-#else
>-#define ATTRIBUTE_UNUSED
>-#endif
>-#endif
>-
>-// Macro to be used for switch-case fallthrough (required for enabling
>-// -Wimplicit-fallthrough warning on Clang).
>-#ifndef FALLTHROUGH
>-#if defined(__clang__)
>-#define FALLTHROUGH() [[clang::fallthrough]]
>-#else
>-#define FALLTHROUGH() do { } while (0)
>-#endif
>-#endif
>-
>-#ifndef NO_RETURN
>-// Annotate a function that will not return control flow to the caller.
>-#if defined(_MSC_VER)
>-#define NO_RETURN __declspec(noreturn)
>-#elif defined(__GNUC__)
>-#define NO_RETURN __attribute__ ((__noreturn__))
>-#else
>-#define NO_RETURN
>-#endif
>-#endif
>-
>-// Prevent the compiler from warning about an unused variable. For example:
>-//   int result = DoSomething();
>-//   assert(result == 17);
>-//   RTC_UNUSED(result);
>-// Note: In most cases it is better to remove the unused variable rather than
>-// suppressing the compiler warning.
>-#ifndef RTC_UNUSED
>-#define RTC_UNUSED(x) static_cast<void>(x)
>-#endif  // RTC_UNUSED
>-
>-#endif  // TYPEDEFS_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/video/BUILD.gn
>index cc6755689a9..5f9455c5001 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/BUILD.gn
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/BUILD.gn
>@@ -14,10 +14,6 @@ rtc_static_library("video") {
>     "call_stats.h",
>     "encoder_rtcp_feedback.cc",
>     "encoder_rtcp_feedback.h",
>-    "overuse_frame_detector.cc",
>-    "overuse_frame_detector.h",
>-    "payload_router.cc",
>-    "payload_router.h",
>     "quality_threshold.cc",
>     "quality_threshold.h",
>     "receive_statistics_proxy.cc",
>@@ -38,14 +34,16 @@ rtc_static_library("video") {
>     "stream_synchronization.h",
>     "transport_adapter.cc",
>     "transport_adapter.h",
>+    "video_quality_observer.cc",
>+    "video_quality_observer.h",
>     "video_receive_stream.cc",
>     "video_receive_stream.h",
>     "video_send_stream.cc",
>     "video_send_stream.h",
>+    "video_send_stream_impl.cc",
>+    "video_send_stream_impl.h",
>     "video_stream_decoder.cc",
>     "video_stream_decoder.h",
>-    "video_stream_encoder.cc",
>-    "video_stream_encoder.h",
>   ]
> 
>   if (!build_with_chromium && is_clang) {
>@@ -55,23 +53,36 @@ rtc_static_library("video") {
> 
>   deps = [
>     "..:webrtc_common",
>-    "../:typedefs",
>+    "../api:fec_controller_api",
>     "../api:libjingle_peerconnection_api",
>-    "../api:optional",
>     "../api:transport_api",
>-    "../api:video_frame_api",
>-    "../api:video_frame_api_i420",
>+    "../api/video:video_bitrate_allocator",
>+    "../api/video:video_frame",
>+    "../api/video:video_frame_i420",
>+    "../api/video:video_stream_encoder",
>+    "../api/video:video_stream_encoder_create",
>     "../api/video_codecs:video_codecs_api",
>     "../call:bitrate_allocator",
>     "../call:call_interfaces",
>     "../call:rtp_interfaces",
>+    "../call:rtp_sender",
>     "../call:video_stream_api",
>     "../modules/rtp_rtcp:rtp_rtcp_format",
>+    "../modules/rtp_rtcp:rtp_video_header",
>+    "../modules/video_coding:codec_globals_headers",
>+    "../modules/video_coding:nack_module",
>+    "../modules/video_coding:packet",
>     "../modules/video_coding:video_codec_interface",
>     "../rtc_base:checks",
>+    "../rtc_base:rate_limiter",
>+    "../rtc_base:stringutils",
>     "../rtc_base/experiments:alr_experiment",
>+    "../rtc_base/experiments:quality_scaling_experiment",
>+    "../rtc_base/system:fallthrough",
>     "../system_wrappers:field_trial_api",
>     "../system_wrappers:metrics_api",
>+    "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/types:optional",
> 
>     # For RtxReceiveStream.
>     "../call:rtp_receiver",
>@@ -79,22 +90,20 @@ rtc_static_library("video") {
>     "../logging:rtc_event_log_api",
>     "../modules:module_api",
>     "../modules/bitrate_controller",
>-    "../modules/congestion_controller",
>     "../modules/pacing",
>     "../modules/remote_bitrate_estimator",
>     "../modules/rtp_rtcp",
>     "../modules/utility",
>     "../modules/video_coding",
>     "../modules/video_coding:video_coding_utility",
>-    "../modules/video_coding:webrtc_vp8_helpers",
>     "../modules/video_processing",
>     "../rtc_base:rtc_base_approved",
>     "../rtc_base:rtc_numerics",
>     "../rtc_base:rtc_task_queue",
>     "../rtc_base:sequenced_task_checker",
>     "../rtc_base:weak_ptr",
>+    "../rtc_base/time:timestamp_extrapolator",
>     "../system_wrappers",
>-    "../voice_engine",
>   ]
> 
>   if (!build_with_mozilla) {
>@@ -102,26 +111,116 @@ rtc_static_library("video") {
>   }
> }
> 
>+rtc_source_set("video_stream_decoder_impl") {
>+  visibility = [ "*" ]
>+
>+  sources = [
>+    "video_stream_decoder_impl.cc",
>+    "video_stream_decoder_impl.h",
>+  ]
>+
>+  deps = [
>+    "../api/video:encoded_frame",
>+    "../api/video:video_frame",
>+    "../api/video:video_stream_decoder",
>+    "../api/video_codecs:video_codecs_api",
>+    "../modules/video_coding:video_coding",
>+    "../rtc_base:rtc_base_approved",
>+    "../rtc_base:rtc_task_queue_api",
>+    "../system_wrappers:system_wrappers",
>+    "//third_party/abseil-cpp/absl/memory",
>+    "//third_party/abseil-cpp/absl/types:optional",
>+  ]
>+}
>+
>+rtc_source_set("video_stream_encoder_impl") {
>+  visibility = [ "*" ]
>+
>+  # visibility = [ "../api/video:video_stream_encoder_create" ]
>+  # In modules/video_coding, there's a dependency video_coding --> webrtc_vp8
>+  allow_poison = [ "software_video_codecs" ]  # TODO(bugs.webrtc.org/7925): Remove.
>+  sources = [
>+    "overuse_frame_detector.cc",
>+    "overuse_frame_detector.h",
>+    "video_stream_encoder.cc",
>+    "video_stream_encoder.h",
>+  ]
>+
>+  if (!build_with_chromium && is_clang) {
>+    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>+    suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>+  }
>+
>+  deps = [
>+    "../api/video:video_bitrate_allocator",
>+    "../api/video:video_frame",
>+    "../api/video:video_frame_i420",
>+    "../api/video:video_stream_encoder",
>+    "../api/video_codecs:video_codecs_api",
>+    "../common_video:common_video",
>+    "../modules/video_coding",
>+    "../modules/video_coding:video_coding_utility",
>+    "../rtc_base:checks",
>+    "../rtc_base:criticalsection",
>+    "../rtc_base:logging",
>+    "../rtc_base:macromagic",
>+    "../rtc_base:rtc_base_approved",
>+    "../rtc_base:rtc_event",
>+    "../rtc_base:rtc_numerics",
>+    "../rtc_base:rtc_task_queue",
>+    "../rtc_base:sequenced_task_checker",
>+    "../rtc_base:timeutils",
>+    "../rtc_base/experiments:quality_scaling_experiment",
>+    "../rtc_base/system:fallthrough",
>+    "../system_wrappers:field_trial_api",
>+    "//third_party/abseil-cpp/absl/types:optional",
>+  ]
>+}
>+
> if (rtc_include_tests) {
>+  rtc_source_set("video_mocks") {
>+    testonly = true
>+    sources = [
>+      "test/mock_video_stream_encoder.h",
>+    ]
>+    deps = [
>+      "../api/video:video_stream_encoder",
>+      "../test:test_support",
>+    ]
>+  }
>   rtc_source_set("video_quality_test") {
>     testonly = true
>-    visibility = [ ":*" ]  # Only targets in this file can depend on this.
>+
>+    # Only targets in this file and api/ can depend on this.
>+    visibility = [
>+      ":*",
>+      "../api:create_video_quality_test_fixture_api",
>+    ]
>     sources = [
>+      "video_analyzer.cc",
>+      "video_analyzer.h",
>       "video_quality_test.cc",
>       "video_quality_test.h",
>     ]
>     deps = [
>+      "../api:fec_controller_api",
>+      "../api:video_quality_test_fixture_api",
>+      "../call:fake_network",
>       "../logging:rtc_event_log_api",
>+      "../logging:rtc_event_log_impl_output",
>       "../media:rtc_audio_video",
>       "../media:rtc_internal_video_codecs",
>       "../modules/audio_mixer:audio_mixer_impl",
>       "../modules/rtp_rtcp",
>+      "../modules/video_coding:video_coding",
>       "../modules/video_coding:webrtc_h264",
>+      "../modules/video_coding:webrtc_multiplex",
>       "../modules/video_coding:webrtc_vp8",
>       "../modules/video_coding:webrtc_vp9",
>       "../rtc_base:rtc_base_approved",
>       "../rtc_base:rtc_base_tests_utils",
>       "../system_wrappers",
>+      "../test:fileutils",
>       "../test:perf_test",
>       "../test:rtp_test_utils",
>       "../test:test_common",
>@@ -130,7 +229,6 @@ if (rtc_include_tests) {
>       "../test:test_support_test_artifacts",
>       "../test:video_test_common",
>       "../test:video_test_support",
>-      "../voice_engine",
>     ]
>     if (!build_with_chromium && is_clang) {
>       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>@@ -146,8 +244,12 @@ if (rtc_include_tests) {
>     ]
>     deps = [
>       ":video_quality_test",
>+      "../media:rtc_vp9_profile",
>       "../modules/pacing:pacing",
>+      "../modules/video_coding:webrtc_vp9",
>+      "../rtc_base:rtc_base_approved",
>       "../rtc_base/experiments:alr_experiment",
>+      "../system_wrappers:field_trial_default",
>       "../test:field_trial",
>       "../test:test_common",
>       "../test:test_support",
>@@ -169,7 +271,9 @@ if (rtc_include_tests) {
>     ]
>     deps = [
>       ":video_quality_test",
>+      "../rtc_base:logging",
>       "../rtc_base:rtc_base_approved",
>+      "../system_wrappers:field_trial_default",
>       "../system_wrappers:metrics_default",
>       "../system_wrappers:runtime_enabled_features_default",
>       "../test:field_trial",
>@@ -178,7 +282,6 @@ if (rtc_include_tests) {
>       "../test:test_common",
>       "../test:test_renderer",
>       "../test:test_support",
>-      "//testing/gmock",
>       "//testing/gtest",
>     ]
>     if (!build_with_chromium && is_clang) {
>@@ -195,7 +298,9 @@ if (rtc_include_tests) {
> 
>     deps = [
>       ":video_quality_test",
>+      "../rtc_base:logging",
>       "../rtc_base:rtc_base_approved",
>+      "../system_wrappers:field_trial_default",
>       "../system_wrappers:metrics_default",
>       "../system_wrappers:runtime_enabled_features_default",
>       "../test:field_trial",
>@@ -219,7 +324,9 @@ if (rtc_include_tests) {
>     ]
>     deps = [
>       ":video_quality_test",
>+      "../rtc_base:logging",
>       "../rtc_base:rtc_base_approved",
>+      "../system_wrappers:field_trial_default",
>       "../system_wrappers:metrics_default",
>       "../system_wrappers:runtime_enabled_features_default",
>       "../test:field_trial",
>@@ -228,7 +335,6 @@ if (rtc_include_tests) {
>       "../test:test_common",
>       "../test:test_renderer",
>       "../test:test_support",
>-      "//testing/gmock",
>       "//testing/gtest",
>     ]
>     if (!build_with_chromium && is_clang) {
>@@ -244,7 +350,6 @@ if (rtc_include_tests) {
>     ]
>     deps = [
>       "..:webrtc_common",
>-      "../:typedefs",
>       "../api/video_codecs:video_codecs_api",
>       "../call:call_interfaces",
>       "../common_video",
>@@ -279,10 +384,29 @@ if (rtc_include_tests) {
>     sources = [
>       "call_stats_unittest.cc",
>       "encoder_rtcp_feedback_unittest.cc",
>-      "end_to_end_tests.cc",
>+      "end_to_end_tests/bandwidth_tests.cc",
>+      "end_to_end_tests/call_operation_tests.cc",
>+      "end_to_end_tests/codec_tests.cc",
>+      "end_to_end_tests/config_tests.cc",
>+      "end_to_end_tests/extended_reports_tests.cc",
>+      "end_to_end_tests/fec_tests.cc",
>+      "end_to_end_tests/histogram_tests.cc",
>+      "end_to_end_tests/log_tests.cc",
>+      "end_to_end_tests/multi_codec_receive_tests.cc",
>+      "end_to_end_tests/multi_stream_tester.cc",
>+      "end_to_end_tests/multi_stream_tester.h",
>+      "end_to_end_tests/multi_stream_tests.cc",
>+      "end_to_end_tests/network_state_tests.cc",
>+      "end_to_end_tests/probing_tests.cc",
>+      "end_to_end_tests/receive_time_tests.cc",
>+      "end_to_end_tests/retransmission_tests.cc",
>+      "end_to_end_tests/rtp_rtcp_tests.cc",
>+      "end_to_end_tests/ssrc_tests.cc",
>+      "end_to_end_tests/stats_tests.cc",
>+      "end_to_end_tests/transport_feedback_tests.cc",
>       "overuse_frame_detector_unittest.cc",
>-      "payload_router_unittest.cc",
>       "picture_id_tests.cc",
>+      "quality_scaling_tests.cc",
>       "quality_threshold_unittest.cc",
>       "receive_statistics_proxy_unittest.cc",
>       "report_block_stats_unittest.cc",
>@@ -292,16 +416,19 @@ if (rtc_include_tests) {
>       "stats_counter_unittest.cc",
>       "stream_synchronization_unittest.cc",
>       "video_receive_stream_unittest.cc",
>+      "video_send_stream_impl_unittest.cc",
>       "video_send_stream_tests.cc",
>       "video_stream_encoder_unittest.cc",
>     ]
>     deps = [
>       ":video",
>-      "../api:optional",
>-      "../api:video_frame_api",
>-      "../api:video_frame_api_i420",
>+      ":video_mocks",
>+      ":video_stream_encoder_impl",
>+      "../api/video:video_frame",
>+      "../api/video:video_frame_i420",
>       "../api/video_codecs:video_codecs_api",
>       "../call:call_interfaces",
>+      "../call:mock_bitrate_allocator",
>       "../call:mock_rtp_interfaces",
>       "../call:rtp_receiver",
>       "../call:rtp_sender",
>@@ -320,17 +447,21 @@ if (rtc_include_tests) {
>       "../modules/rtp_rtcp:rtp_rtcp_format",
>       "../modules/utility",
>       "../modules/video_coding",
>+      "../modules/video_coding:codec_globals_headers",
>+      "../modules/video_coding:packet",
>       "../modules/video_coding:video_codec_interface",
>       "../modules/video_coding:video_coding_utility",
>       "../modules/video_coding:webrtc_h264",
>+      "../modules/video_coding:webrtc_multiplex",
>       "../modules/video_coding:webrtc_vp8",
>-      "../modules/video_coding:webrtc_vp8_helpers",
>       "../modules/video_coding:webrtc_vp9",
>       "../rtc_base:checks",
>       "../rtc_base:rate_limiter",
>       "../rtc_base:rtc_base_approved",
>       "../rtc_base:rtc_base_tests_utils",
>       "../rtc_base:rtc_numerics",
>+      "../rtc_base:rtc_task_queue",
>+      "../rtc_base:rtc_task_queue_for_test",
>       "../rtc_base/experiments:alr_experiment",
>       "../system_wrappers",
>       "../system_wrappers:field_trial_default",
>@@ -338,13 +469,15 @@ if (rtc_include_tests) {
>       "../system_wrappers:metrics_default",
>       "../test:direct_transport",
>       "../test:field_trial",
>+      "../test:fileutils",
>       "../test:perf_test",
>       "../test:rtp_test_utils",
>       "../test:test_common",
>       "../test:test_support",
>       "../test:video_test_common",
>-      "//testing/gmock",
>       "//testing/gtest",
>+      "//third_party/abseil-cpp/absl/memory",
>+      "//third_party/abseil-cpp/absl/types:optional",
>     ]
>     if (!build_with_chromium && is_clang) {
>       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/DEPS b/Source/ThirdParty/libwebrtc/Source/webrtc/video/DEPS
>index b94455edd00..288ecfdfe1f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/DEPS
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/DEPS
>@@ -15,5 +15,4 @@ include_rules = [
>   "+modules/video_capture",
>   "+modules/video_processing",
>   "+system_wrappers",
>-  "+voice_engine",
> ]
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/call_stats.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/call_stats.cc
>index c9f019929f6..8948e68e5a5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/call_stats.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/call_stats.cc
>@@ -12,177 +12,209 @@
> 
> #include <algorithm>
> 
>-#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>+#include "modules/utility/include/process_thread.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/constructormagic.h"
>+#include "rtc_base/location.h"
>+#include "rtc_base/logging.h"
>+#include "rtc_base/task_queue.h"
> #include "system_wrappers/include/metrics.h"
> 
> namespace webrtc {
> namespace {
>-// Time interval for updating the observers.
>-const int64_t kUpdateIntervalMs = 1000;
>-// Weight factor to apply to the average rtt.
>-const float kWeightFactor = 0.3f;
> 
> void RemoveOldReports(int64_t now, std::list<CallStats::RttTime>* reports) {
>-  // A rtt report is considered valid for this long.
>-  const int64_t kRttTimeoutMs = 1500;
>-  while (!reports->empty() &&
>-         (now - reports->front().time) > kRttTimeoutMs) {
>-    reports->pop_front();
>-  }
>+  static constexpr const int64_t kRttTimeoutMs = 1500;
>+  reports->remove_if(
>+      [&now](CallStats::RttTime& r) { return now - r.time > kRttTimeoutMs; });
> }
> 
>-int64_t GetMaxRttMs(std::list<CallStats::RttTime>* reports) {
>-  if (reports->empty())
>-    return -1;
>-  int64_t max_rtt_ms = 0;
>-  for (const CallStats::RttTime& rtt_time : *reports)
>+int64_t GetMaxRttMs(const std::list<CallStats::RttTime>& reports) {
>+  int64_t max_rtt_ms = -1;
>+  for (const CallStats::RttTime& rtt_time : reports)
>     max_rtt_ms = std::max(rtt_time.rtt, max_rtt_ms);
>   return max_rtt_ms;
> }
> 
>-int64_t GetAvgRttMs(std::list<CallStats::RttTime>* reports) {
>-  if (reports->empty()) {
>-    return -1;
>-  }
>+int64_t GetAvgRttMs(const std::list<CallStats::RttTime>& reports) {
>+  RTC_DCHECK(!reports.empty());
>   int64_t sum = 0;
>-  for (std::list<CallStats::RttTime>::const_iterator it = reports->begin();
>-       it != reports->end(); ++it) {
>+  for (std::list<CallStats::RttTime>::const_iterator it = reports.begin();
>+       it != reports.end(); ++it) {
>     sum += it->rtt;
>   }
>-  return sum / reports->size();
>+  return sum / reports.size();
> }
> 
>-void UpdateAvgRttMs(std::list<CallStats::RttTime>* reports, int64_t* avg_rtt) {
>+int64_t GetNewAvgRttMs(const std::list<CallStats::RttTime>& reports,
>+                       int64_t prev_avg_rtt) {
>+  if (reports.empty())
>+    return -1;  // Reset (invalid average).
>+
>   int64_t cur_rtt_ms = GetAvgRttMs(reports);
>-  if (cur_rtt_ms == -1) {
>-    // Reset.
>-    *avg_rtt = -1;
>-    return;
>-  }
>-  if (*avg_rtt == -1) {
>-    // Initialize.
>-    *avg_rtt = cur_rtt_ms;
>-    return;
>-  }
>-  *avg_rtt = *avg_rtt * (1.0f - kWeightFactor) + cur_rtt_ms * kWeightFactor;
>+  if (prev_avg_rtt == -1)
>+    return cur_rtt_ms;  // New initial average value.
>+
>+  // Weight factor to apply to the average rtt.
>+  // We weigh the old average at 70% against the new average (30%).
>+  constexpr const float kWeightFactor = 0.3f;
>+  return prev_avg_rtt * (1.0f - kWeightFactor) + cur_rtt_ms * kWeightFactor;
> }
>-}  // namespace
> 
>-class RtcpObserver : public RtcpRttStats {
>+// This class is used to de-register a Module from a ProcessThread to satisfy
>+// threading requirements of the Module (CallStats).
>+// The guarantee offered by TemporaryDeregistration is that while its in scope,
>+// no calls to |TimeUntilNextProcess| or |Process()| will occur and therefore
>+// synchronization with those methods, is not necessary.
>+class TemporaryDeregistration {
>  public:
>-  explicit RtcpObserver(CallStats* owner) : owner_(owner) {}
>-  virtual ~RtcpObserver() {}
>-
>-  virtual void OnRttUpdate(int64_t rtt) {
>-    owner_->OnRttUpdate(rtt);
>+  TemporaryDeregistration(Module* module,
>+                          ProcessThread* process_thread,
>+                          bool thread_running)
>+      : module_(module),
>+        process_thread_(process_thread),
>+        deregistered_(thread_running) {
>+    if (thread_running)
>+      process_thread_->DeRegisterModule(module_);
>   }
>-
>-  // Returns the average RTT.
>-  virtual int64_t LastProcessedRtt() const {
>-    return owner_->avg_rtt_ms();
>+  ~TemporaryDeregistration() {
>+    if (deregistered_)
>+      process_thread_->RegisterModule(module_, RTC_FROM_HERE);
>   }
> 
>  private:
>-  CallStats* owner_;
>-
>-  RTC_DISALLOW_COPY_AND_ASSIGN(RtcpObserver);
>+  Module* const module_;
>+  ProcessThread* const process_thread_;
>+  const bool deregistered_;
> };
> 
>-CallStats::CallStats(Clock* clock)
>+}  // namespace
>+
>+CallStats::CallStats(Clock* clock, ProcessThread* process_thread)
>     : clock_(clock),
>-      rtcp_rtt_stats_(new RtcpObserver(this)),
>       last_process_time_(clock_->TimeInMilliseconds()),
>       max_rtt_ms_(-1),
>       avg_rtt_ms_(-1),
>       sum_avg_rtt_ms_(0),
>       num_avg_rtt_(0),
>-      time_of_first_rtt_ms_(-1) {}
>+      time_of_first_rtt_ms_(-1),
>+      process_thread_(process_thread),
>+      process_thread_running_(false) {
>+  RTC_DCHECK(process_thread_);
>+  process_thread_checker_.DetachFromThread();
>+}
> 
> CallStats::~CallStats() {
>+  RTC_DCHECK_RUN_ON(&construction_thread_checker_);
>+  RTC_DCHECK(!process_thread_running_);
>   RTC_DCHECK(observers_.empty());
>+
>   UpdateHistograms();
> }
> 
> int64_t CallStats::TimeUntilNextProcess() {
>+  RTC_DCHECK_RUN_ON(&process_thread_checker_);
>   return last_process_time_ + kUpdateIntervalMs - clock_->TimeInMilliseconds();
> }
> 
> void CallStats::Process() {
>-  rtc::CritScope cs(&crit_);
>+  RTC_DCHECK_RUN_ON(&process_thread_checker_);
>   int64_t now = clock_->TimeInMilliseconds();
>-  if (now < last_process_time_ + kUpdateIntervalMs)
>-    return;
>-
>   last_process_time_ = now;
> 
>+  int64_t avg_rtt_ms = avg_rtt_ms_;
>   RemoveOldReports(now, &reports_);
>-  max_rtt_ms_ = GetMaxRttMs(&reports_);
>-  UpdateAvgRttMs(&reports_, &avg_rtt_ms_);
>+  max_rtt_ms_ = GetMaxRttMs(reports_);
>+  avg_rtt_ms = GetNewAvgRttMs(reports_, avg_rtt_ms);
>+  {
>+    rtc::CritScope lock(&avg_rtt_ms_lock_);
>+    avg_rtt_ms_ = avg_rtt_ms;
>+  }
> 
>   // If there is a valid rtt, update all observers with the max rtt.
>   if (max_rtt_ms_ >= 0) {
>-    RTC_DCHECK_GE(avg_rtt_ms_, 0);
>-    for (std::list<CallStatsObserver*>::iterator it = observers_.begin();
>-         it != observers_.end(); ++it) {
>-      (*it)->OnRttUpdate(avg_rtt_ms_, max_rtt_ms_);
>-    }
>+    RTC_DCHECK_GE(avg_rtt_ms, 0);
>+    for (CallStatsObserver* observer : observers_)
>+      observer->OnRttUpdate(avg_rtt_ms, max_rtt_ms_);
>     // Sum for Histogram of average RTT reported over the entire call.
>-    sum_avg_rtt_ms_ += avg_rtt_ms_;
>+    sum_avg_rtt_ms_ += avg_rtt_ms;
>     ++num_avg_rtt_;
>   }
> }
> 
>-int64_t CallStats::avg_rtt_ms() const {
>-  rtc::CritScope cs(&crit_);
>-  return avg_rtt_ms_;
>-}
>+void CallStats::ProcessThreadAttached(ProcessThread* process_thread) {
>+  RTC_DCHECK_RUN_ON(&construction_thread_checker_);
>+  RTC_DCHECK(!process_thread || process_thread_ == process_thread);
>+  process_thread_running_ = process_thread != nullptr;
> 
>-RtcpRttStats* CallStats::rtcp_rtt_stats() const {
>-  return rtcp_rtt_stats_.get();
>+  // Whether we just got attached or detached, we clear the
>+  // |process_thread_checker_| so that it can be used to protect variables
>+  // in either the process thread when it starts again, or UpdateHistograms()
>+  // (mutually exclusive).
>+  process_thread_checker_.DetachFromThread();
> }
> 
> void CallStats::RegisterStatsObserver(CallStatsObserver* observer) {
>-  rtc::CritScope cs(&crit_);
>-  for (std::list<CallStatsObserver*>::iterator it = observers_.begin();
>-       it != observers_.end(); ++it) {
>-    if (*it == observer)
>-      return;
>-  }
>-  observers_.push_back(observer);
>+  RTC_DCHECK_RUN_ON(&construction_thread_checker_);
>+  TemporaryDeregistration deregister(this, process_thread_,
>+                                     process_thread_running_);
>+
>+  auto it = std::find(observers_.begin(), observers_.end(), observer);
>+  if (it == observers_.end())
>+    observers_.push_back(observer);
> }
> 
> void CallStats::DeregisterStatsObserver(CallStatsObserver* observer) {
>-  rtc::CritScope cs(&crit_);
>-  for (std::list<CallStatsObserver*>::iterator it = observers_.begin();
>-       it != observers_.end(); ++it) {
>-    if (*it == observer) {
>-      observers_.erase(it);
>-      return;
>-    }
>-  }
>+  RTC_DCHECK_RUN_ON(&construction_thread_checker_);
>+  TemporaryDeregistration deregister(this, process_thread_,
>+                                     process_thread_running_);
>+  observers_.remove(observer);
>+}
>+
>+int64_t CallStats::LastProcessedRtt() const {
>+  rtc::CritScope cs(&avg_rtt_ms_lock_);
>+  return avg_rtt_ms_;
> }
> 
> void CallStats::OnRttUpdate(int64_t rtt) {
>-  rtc::CritScope cs(&crit_);
>   int64_t now_ms = clock_->TimeInMilliseconds();
>-  reports_.push_back(RttTime(rtt, now_ms));
>-  if (time_of_first_rtt_ms_ == -1)
>-    time_of_first_rtt_ms_ = now_ms;
>+  process_thread_->PostTask(rtc::NewClosure([rtt, now_ms, this]() {
>+    RTC_DCHECK_RUN_ON(&process_thread_checker_);
>+    reports_.push_back(RttTime(rtt, now_ms));
>+    if (time_of_first_rtt_ms_ == -1)
>+      time_of_first_rtt_ms_ = now_ms;
>+
>+    process_thread_->WakeUp(this);
>+  }));
> }
> 
> void CallStats::UpdateHistograms() {
>-  rtc::CritScope cs(&crit_);
>-  if (time_of_first_rtt_ms_ == -1 || num_avg_rtt_ < 1)
>-    return;
>-
>-  int64_t elapsed_sec =
>-      (clock_->TimeInMilliseconds() - time_of_first_rtt_ms_) / 1000;
>-  if (elapsed_sec >= metrics::kMinRunTimeInSeconds) {
>-    int64_t avg_rtt_ms = (sum_avg_rtt_ms_ + num_avg_rtt_ / 2) / num_avg_rtt_;
>-    RTC_HISTOGRAM_COUNTS_10000(
>-        "WebRTC.Video.AverageRoundTripTimeInMilliseconds", avg_rtt_ms);
>+  RTC_DCHECK_RUN_ON(&construction_thread_checker_);
>+  RTC_DCHECK(!process_thread_running_);
>+
>+  // The extra scope is because we have two 'dcheck run on' thread checkers.
>+  // This is a special case since it's safe to access variables on the current
>+  // thread that normally are only touched on the process thread.
>+  // Since we're not attached to the process thread and/or the process thread
>+  // isn't running, it's OK to touch these variables here.
>+  {
>+    // This method is called on the ctor thread (usually from the dtor, unless
>+    // a test calls it). It's a requirement that the function be called when
>+    // the process thread is not running (a condition that's met at destruction
>+    // time), and thanks to that, we don't need a lock to synchronize against
>+    // it.
>+    RTC_DCHECK_RUN_ON(&process_thread_checker_);
>+
>+    if (time_of_first_rtt_ms_ == -1 || num_avg_rtt_ < 1)
>+      return;
>+
>+    int64_t elapsed_sec =
>+        (clock_->TimeInMilliseconds() - time_of_first_rtt_ms_) / 1000;
>+    if (elapsed_sec >= metrics::kMinRunTimeInSeconds) {
>+      int64_t avg_rtt_ms = (sum_avg_rtt_ms_ + num_avg_rtt_ / 2) / num_avg_rtt_;
>+      RTC_HISTOGRAM_COUNTS_10000(
>+          "WebRTC.Video.AverageRoundTripTimeInMilliseconds", avg_rtt_ms);
>+    }
>   }
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/call_stats.h b/Source/ThirdParty/libwebrtc/Source/webrtc/video/call_stats.h
>index af5c45c996e..930c3efffd9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/call_stats.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/call_stats.h
>@@ -15,71 +15,105 @@
> #include <memory>
> 
> #include "modules/include/module.h"
>+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/criticalsection.h"
>+#include "rtc_base/thread_checker.h"
> #include "system_wrappers/include/clock.h"
> 
> namespace webrtc {
> 
> class CallStatsObserver;
>-class RtcpRttStats;
> 
> // CallStats keeps track of statistics for a call.
>-class CallStats : public Module {
>+class CallStats : public Module, public RtcpRttStats {
>  public:
>-  friend class RtcpObserver;
>+  // Time interval for updating the observers.
>+  static constexpr int64_t kUpdateIntervalMs = 1000;
> 
>-  explicit CallStats(Clock* clock);
>-  ~CallStats();
>-
>-  // Implements Module, to use the process thread.
>-  int64_t TimeUntilNextProcess() override;
>-  void Process() override;
>-
>-  // Returns a RtcpRttStats to register at a statistics provider. The object
>-  // has the same lifetime as the CallStats instance.
>-  RtcpRttStats* rtcp_rtt_stats() const;
>+  CallStats(Clock* clock, ProcessThread* process_thread);
>+  ~CallStats() override;
> 
>   // Registers/deregisters a new observer to receive statistics updates.
>+  // Must be called from the construction thread.
>   void RegisterStatsObserver(CallStatsObserver* observer);
>   void DeregisterStatsObserver(CallStatsObserver* observer);
> 
>+  // Expose |LastProcessedRtt()| from RtcpRttStats to the public interface, as
>+  // it is the part of the API that is needed by direct users of CallStats.
>+  // TODO(tommi): Threading or lifetime guarantees are not explicit in how
>+  // CallStats is used as RtcpRttStats or how pointers are cached in a
>+  // few different places (distributed via Call). It would be good to clarify
>+  // from what thread/TQ calls to OnRttUpdate and LastProcessedRtt need to be
>+  // allowed.
>+  int64_t LastProcessedRtt() const override;
>+
>+  // Exposed for tests to test histogram support.
>+  void UpdateHistogramsForTest() { UpdateHistograms(); }
>+
>   // Helper struct keeping track of the time a rtt value is reported.
>   struct RttTime {
>-    RttTime(int64_t new_rtt, int64_t rtt_time)
>-        : rtt(new_rtt), time(rtt_time) {}
>+    RttTime(int64_t new_rtt, int64_t rtt_time) : rtt(new_rtt), time(rtt_time) {}
>     const int64_t rtt;
>     const int64_t time;
>   };
> 
>- protected:
>-  void OnRttUpdate(int64_t rtt);
>+ private:
>+  // RtcpRttStats implementation.
>+  void OnRttUpdate(int64_t rtt) override;
>+
>+  // Implements Module, to use the process thread.
>+  int64_t TimeUntilNextProcess() override;
>+  void Process() override;
> 
>-  int64_t avg_rtt_ms() const;
>+  // TODO(tommi): Use this to know when we're attached to the process thread?
>+  // Alternatively, inject that pointer via the ctor since the call_stats
>+  // test code, isn't using a processthread atm.
>+  void ProcessThreadAttached(ProcessThread* process_thread) override;
> 
>- private:
>+  // This method must only be called when the process thread is not
>+  // running, and from the construction thread.
>   void UpdateHistograms();
> 
>   Clock* const clock_;
>-  // Protecting all members.
>-  rtc::CriticalSection crit_;
>-  // Observer receiving statistics updates.
>-  std::unique_ptr<RtcpRttStats> rtcp_rtt_stats_;
>+
>   // The last time 'Process' resulted in statistic update.
>-  int64_t last_process_time_;
>+  int64_t last_process_time_ RTC_GUARDED_BY(process_thread_checker_);
>   // The last RTT in the statistics update (zero if there is no valid estimate).
>-  int64_t max_rtt_ms_;
>+  int64_t max_rtt_ms_ RTC_GUARDED_BY(process_thread_checker_);
>+
>+  // Accessed from random threads (seemingly). Consider atomic.
>+  // |avg_rtt_ms_| is allowed to be read on the process thread without a lock.
>+  // |avg_rtt_ms_lock_| must be held elsewhere for reading.
>+  // |avg_rtt_ms_lock_| must be held on the process thread for writing.
>   int64_t avg_rtt_ms_;
>-  int64_t sum_avg_rtt_ms_ RTC_GUARDED_BY(crit_);
>-  int64_t num_avg_rtt_ RTC_GUARDED_BY(crit_);
>-  int64_t time_of_first_rtt_ms_ RTC_GUARDED_BY(crit_);
>+
>+  // Protects |avg_rtt_ms_|.
>+  rtc::CriticalSection avg_rtt_ms_lock_;
>+
>+  // |sum_avg_rtt_ms_|, |num_avg_rtt_| and |time_of_first_rtt_ms_| are only used
>+  // on the ProcessThread when running. When the Process Thread is not running,
>+  // (and only then) they can be used in UpdateHistograms(), usually called from
>+  // the dtor.
>+  int64_t sum_avg_rtt_ms_ RTC_GUARDED_BY(process_thread_checker_);
>+  int64_t num_avg_rtt_ RTC_GUARDED_BY(process_thread_checker_);
>+  int64_t time_of_first_rtt_ms_ RTC_GUARDED_BY(process_thread_checker_);
> 
>   // All Rtt reports within valid time interval, oldest first.
>-  std::list<RttTime> reports_;
>+  std::list<RttTime> reports_ RTC_GUARDED_BY(process_thread_checker_);
> 
>   // Observers getting stats reports.
>+  // When attached to ProcessThread, this is read-only. In order to allow
>+  // modification, we detach from the process thread while the observer
>+  // list is updated, to avoid races. This allows us to not require a lock
>+  // for the observers_ list, which makes the most common case lock free.
>   std::list<CallStatsObserver*> observers_;
> 
>+  rtc::ThreadChecker construction_thread_checker_;
>+  rtc::ThreadChecker process_thread_checker_;
>+  ProcessThread* const process_thread_;
>+  bool process_thread_running_ RTC_GUARDED_BY(construction_thread_checker_);
>+
>   RTC_DISALLOW_COPY_AND_ASSIGN(CallStats);
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/call_stats_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/call_stats_unittest.cc
>index 989722d29fe..33076c518c2 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/call_stats_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/call_stats_unittest.cc
>@@ -8,17 +8,23 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>+#include "video/call_stats.h"
>+
> #include <memory>
> 
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>+#include "modules/utility/include/process_thread.h"
>+#include "rtc_base/event.h"
>+#include "rtc_base/location.h"
>+#include "rtc_base/task_queue.h"
> #include "system_wrappers/include/metrics.h"
> #include "system_wrappers/include/metrics_default.h"
> #include "test/gmock.h"
> #include "test/gtest.h"
>-#include "video/call_stats.h"
> 
> using ::testing::_;
> using ::testing::AnyNumber;
>+using ::testing::InvokeWithoutArgs;
> using ::testing::Return;
> 
> namespace webrtc {
>@@ -33,184 +39,277 @@ class MockStatsObserver : public CallStatsObserver {
> 
> class CallStatsTest : public ::testing::Test {
>  public:
>-  CallStatsTest() : fake_clock_(12345) {}
>+  CallStatsTest() {
>+    process_thread_->RegisterModule(&call_stats_, RTC_FROM_HERE);
>+    process_thread_->Start();
>+  }
>+  ~CallStatsTest() override {
>+    process_thread_->Stop();
>+    process_thread_->DeRegisterModule(&call_stats_);
>+  }
> 
>  protected:
>-  virtual void SetUp() { call_stats_.reset(new CallStats(&fake_clock_)); }
>-  SimulatedClock fake_clock_;
>-  std::unique_ptr<CallStats> call_stats_;
>+  std::unique_ptr<ProcessThread> process_thread_{
>+      ProcessThread::Create("CallStats")};
>+  SimulatedClock fake_clock_{12345};
>+  CallStats call_stats_{&fake_clock_, process_thread_.get()};
> };
> 
> TEST_F(CallStatsTest, AddAndTriggerCallback) {
>+  rtc::Event event(false, false);
>+
>+  static constexpr const int64_t kRtt = 25;
>+
>   MockStatsObserver stats_observer;
>-  RtcpRttStats* rtcp_rtt_stats = call_stats_->rtcp_rtt_stats();
>-  call_stats_->RegisterStatsObserver(&stats_observer);
>-  fake_clock_.AdvanceTimeMilliseconds(1000);
>+  EXPECT_CALL(stats_observer, OnRttUpdate(kRtt, kRtt))
>+      .Times(1)
>+      .WillOnce(InvokeWithoutArgs([&event] { event.Set(); }));
>+
>+  RtcpRttStats* rtcp_rtt_stats = &call_stats_;
>+  call_stats_.RegisterStatsObserver(&stats_observer);
>   EXPECT_EQ(-1, rtcp_rtt_stats->LastProcessedRtt());
> 
>-  const int64_t kRtt = 25;
>   rtcp_rtt_stats->OnRttUpdate(kRtt);
>-  EXPECT_CALL(stats_observer, OnRttUpdate(kRtt, kRtt)).Times(1);
>-  call_stats_->Process();
>-  EXPECT_EQ(kRtt, rtcp_rtt_stats->LastProcessedRtt());
> 
>-  const int64_t kRttTimeOutMs = 1500 + 10;
>-  fake_clock_.AdvanceTimeMilliseconds(kRttTimeOutMs);
>-  EXPECT_CALL(stats_observer, OnRttUpdate(_, _)).Times(0);
>-  call_stats_->Process();
>-  EXPECT_EQ(-1, rtcp_rtt_stats->LastProcessedRtt());
>+  EXPECT_TRUE(event.Wait(1000));
>+  EXPECT_EQ(kRtt, rtcp_rtt_stats->LastProcessedRtt());
> 
>-  call_stats_->DeregisterStatsObserver(&stats_observer);
>+  call_stats_.DeregisterStatsObserver(&stats_observer);
> }
> 
> TEST_F(CallStatsTest, ProcessTime) {
>+  rtc::Event event(false, false);
>+
>+  static constexpr const int64_t kRtt = 100;
>+  static constexpr const int64_t kRtt2 = 80;
>+
>+  RtcpRttStats* rtcp_rtt_stats = &call_stats_;
>+
>   MockStatsObserver stats_observer;
>-  call_stats_->RegisterStatsObserver(&stats_observer);
>-  RtcpRttStats* rtcp_rtt_stats = call_stats_->rtcp_rtt_stats();
>-  rtcp_rtt_stats->OnRttUpdate(100);
>-
>-  // Time isn't updated yet.
>-  EXPECT_CALL(stats_observer, OnRttUpdate(_, _)).Times(0);
>-  call_stats_->Process();
>-
>-  // Advance clock and verify we get an update.
>-  fake_clock_.AdvanceTimeMilliseconds(1000);
>-  EXPECT_CALL(stats_observer, OnRttUpdate(_, _)).Times(1);
>-  call_stats_->Process();
>-
>-  // Advance clock just too little to get an update.
>-  fake_clock_.AdvanceTimeMilliseconds(999);
>-  rtcp_rtt_stats->OnRttUpdate(100);
>-  EXPECT_CALL(stats_observer, OnRttUpdate(_, _)).Times(0);
>-  call_stats_->Process();
>-
>-  // Advance enough to trigger a new update.
>-  fake_clock_.AdvanceTimeMilliseconds(1);
>-  EXPECT_CALL(stats_observer, OnRttUpdate(_, _)).Times(1);
>-  call_stats_->Process();
>-
>-  call_stats_->DeregisterStatsObserver(&stats_observer);
>+
>+  EXPECT_CALL(stats_observer, OnRttUpdate(kRtt, kRtt))
>+      .Times(2)
>+      .WillOnce(InvokeWithoutArgs([this] {
>+        // Advance clock and verify we get an update.
>+        fake_clock_.AdvanceTimeMilliseconds(CallStats::kUpdateIntervalMs);
>+      }))
>+      .WillRepeatedly(InvokeWithoutArgs([this, rtcp_rtt_stats] {
>+        rtcp_rtt_stats->OnRttUpdate(kRtt2);
>+        // Advance clock just too little to get an update.
>+        fake_clock_.AdvanceTimeMilliseconds(CallStats::kUpdateIntervalMs - 1);
>+      }));
>+
>+  // In case you're reading this and wondering how this number is arrived at,
>+  // please see comments in the ChangeRtt test that go into some detail.
>+  static constexpr const int64_t kLastAvg = 94;
>+  EXPECT_CALL(stats_observer, OnRttUpdate(kLastAvg, kRtt2))
>+      .Times(1)
>+      .WillOnce(InvokeWithoutArgs([&event] { event.Set(); }));
>+
>+  call_stats_.RegisterStatsObserver(&stats_observer);
>+
>+  rtcp_rtt_stats->OnRttUpdate(kRtt);
>+  EXPECT_TRUE(event.Wait(1000));
>+
>+  call_stats_.DeregisterStatsObserver(&stats_observer);
> }
> 
> // Verify all observers get correct estimates and observers can be added and
> // removed.
> TEST_F(CallStatsTest, MultipleObservers) {
>   MockStatsObserver stats_observer_1;
>-  call_stats_->RegisterStatsObserver(&stats_observer_1);
>+  call_stats_.RegisterStatsObserver(&stats_observer_1);
>   // Add the second observer twice, there should still be only one report to the
>   // observer.
>   MockStatsObserver stats_observer_2;
>-  call_stats_->RegisterStatsObserver(&stats_observer_2);
>-  call_stats_->RegisterStatsObserver(&stats_observer_2);
>+  call_stats_.RegisterStatsObserver(&stats_observer_2);
>+  call_stats_.RegisterStatsObserver(&stats_observer_2);
> 
>-  RtcpRttStats* rtcp_rtt_stats = call_stats_->rtcp_rtt_stats();
>-  const int64_t kRtt = 100;
>-  rtcp_rtt_stats->OnRttUpdate(kRtt);
>+  RtcpRttStats* rtcp_rtt_stats = &call_stats_;
>+  static constexpr const int64_t kRtt = 100;
> 
>   // Verify both observers are updated.
>-  fake_clock_.AdvanceTimeMilliseconds(1000);
>-  EXPECT_CALL(stats_observer_1, OnRttUpdate(kRtt, kRtt)).Times(1);
>-  EXPECT_CALL(stats_observer_2, OnRttUpdate(kRtt, kRtt)).Times(1);
>-  call_stats_->Process();
>+  rtc::Event ev1(false, false);
>+  rtc::Event ev2(false, false);
>+  EXPECT_CALL(stats_observer_1, OnRttUpdate(kRtt, kRtt))
>+      .Times(AnyNumber())
>+      .WillOnce(InvokeWithoutArgs([&ev1] { ev1.Set(); }))
>+      .WillRepeatedly(Return());
>+  EXPECT_CALL(stats_observer_2, OnRttUpdate(kRtt, kRtt))
>+      .Times(AnyNumber())
>+      .WillOnce(InvokeWithoutArgs([&ev2] { ev2.Set(); }))
>+      .WillRepeatedly(Return());
>+  rtcp_rtt_stats->OnRttUpdate(kRtt);
>+  ASSERT_TRUE(ev1.Wait(100));
>+  ASSERT_TRUE(ev2.Wait(100));
> 
>   // Deregister the second observer and verify update is only sent to the first
>   // observer.
>-  call_stats_->DeregisterStatsObserver(&stats_observer_2);
>-  rtcp_rtt_stats->OnRttUpdate(kRtt);
>-  fake_clock_.AdvanceTimeMilliseconds(1000);
>-  EXPECT_CALL(stats_observer_1, OnRttUpdate(kRtt, kRtt)).Times(1);
>+  call_stats_.DeregisterStatsObserver(&stats_observer_2);
>+
>+  EXPECT_CALL(stats_observer_1, OnRttUpdate(kRtt, kRtt))
>+      .Times(AnyNumber())
>+      .WillOnce(InvokeWithoutArgs([&ev1] { ev1.Set(); }))
>+      .WillRepeatedly(Return());
>   EXPECT_CALL(stats_observer_2, OnRttUpdate(kRtt, kRtt)).Times(0);
>-  call_stats_->Process();
>+  rtcp_rtt_stats->OnRttUpdate(kRtt);
>+  ASSERT_TRUE(ev1.Wait(100));
> 
>   // Deregister the first observer.
>-  call_stats_->DeregisterStatsObserver(&stats_observer_1);
>-  rtcp_rtt_stats->OnRttUpdate(kRtt);
>-  fake_clock_.AdvanceTimeMilliseconds(1000);
>+  call_stats_.DeregisterStatsObserver(&stats_observer_1);
>+
>+  // Now make sure we don't get any callbacks.
>   EXPECT_CALL(stats_observer_1, OnRttUpdate(kRtt, kRtt)).Times(0);
>   EXPECT_CALL(stats_observer_2, OnRttUpdate(kRtt, kRtt)).Times(0);
>-  call_stats_->Process();
>+  rtcp_rtt_stats->OnRttUpdate(kRtt);
>+
>+  // Force a call to Process().
>+  process_thread_->WakeUp(&call_stats_);
>+
>+  // Flush the queue on the process thread to make sure we return after
>+  // Process() has been called.
>+  rtc::Event event(false, false);
>+  process_thread_->PostTask(rtc::NewClosure([&event]() { event.Set(); }));
>+  event.Wait(rtc::Event::kForever);
> }
> 
> // Verify increasing and decreasing rtt triggers callbacks with correct values.
> TEST_F(CallStatsTest, ChangeRtt) {
>+  // TODO(tommi): This test assumes things about how old reports are removed
>+  // inside of call_stats.cc. The threshold ms value is 1500ms, but it's not
>+  // clear here that how the clock is advanced, affects that algorithm and
>+  // subsequently the average reported rtt.
>+
>   MockStatsObserver stats_observer;
>-  call_stats_->RegisterStatsObserver(&stats_observer);
>-  RtcpRttStats* rtcp_rtt_stats = call_stats_->rtcp_rtt_stats();
>-
>-  // Advance clock to be ready for an update.
>-  fake_clock_.AdvanceTimeMilliseconds(1000);
>-
>-  // Set a first value and verify the callback is triggered.
>-  const int64_t kFirstRtt = 100;
>-  rtcp_rtt_stats->OnRttUpdate(kFirstRtt);
>-  EXPECT_CALL(stats_observer, OnRttUpdate(kFirstRtt, kFirstRtt)).Times(1);
>-  call_stats_->Process();
>-
>-  // Increase rtt and verify the new value is reported.
>-  fake_clock_.AdvanceTimeMilliseconds(1000);
>-  const int64_t kHighRtt = kFirstRtt + 20;
>-  const int64_t kAvgRtt1 = 103;
>-  rtcp_rtt_stats->OnRttUpdate(kHighRtt);
>-  EXPECT_CALL(stats_observer, OnRttUpdate(kAvgRtt1, kHighRtt)).Times(1);
>-  call_stats_->Process();
>+  call_stats_.RegisterStatsObserver(&stats_observer);
>+  RtcpRttStats* rtcp_rtt_stats = &call_stats_;
>+
>+  rtc::Event event(false, false);
>+
>+  static constexpr const int64_t kFirstRtt = 100;
>+  static constexpr const int64_t kLowRtt = kFirstRtt - 20;
>+  static constexpr const int64_t kHighRtt = kFirstRtt + 20;
>+
>+  EXPECT_CALL(stats_observer, OnRttUpdate(kFirstRtt, kFirstRtt))
>+      .Times(1)
>+      .WillOnce(InvokeWithoutArgs([&rtcp_rtt_stats, this] {
>+        fake_clock_.AdvanceTimeMilliseconds(1000);
>+        rtcp_rtt_stats->OnRttUpdate(kHighRtt);  // Reported at T1 (1000ms).
>+      }));
>+
>+  // TODO(tommi): This relies on the internal algorithms of call_stats.cc.
>+  // There's a weight factor there (0.3), that weighs the previous average to
>+  // the new one by 70%, so the number 103 in this case is arrived at like so:
>+  // (100) / 1 * 0.7 + (100+120)/2 * 0.3 = 103
>+  static constexpr const int64_t kAvgRtt1 = 103;
>+  EXPECT_CALL(stats_observer, OnRttUpdate(kAvgRtt1, kHighRtt))
>+      .Times(1)
>+      .WillOnce(InvokeWithoutArgs([&rtcp_rtt_stats, this] {
>+        // This interacts with an internal implementation detail in call_stats
>+        // that decays the oldest rtt value. See more below.
>+        fake_clock_.AdvanceTimeMilliseconds(1000);
>+        rtcp_rtt_stats->OnRttUpdate(kLowRtt);  // Reported at T2 (2000ms).
>+      }));
> 
>   // Increase time enough for a new update, but not too much to make the
>   // rtt invalid. Report a lower rtt and verify the old/high value still is sent
>   // in the callback.
>-  fake_clock_.AdvanceTimeMilliseconds(1000);
>-  const int64_t kLowRtt = kFirstRtt - 20;
>-  const int64_t kAvgRtt2 = 102;
>-  rtcp_rtt_stats->OnRttUpdate(kLowRtt);
>-  EXPECT_CALL(stats_observer, OnRttUpdate(kAvgRtt2, kHighRtt)).Times(1);
>-  call_stats_->Process();
>-
>-  // Advance time to make the high report invalid, the lower rtt should now be
>-  // in the callback.
>-  fake_clock_.AdvanceTimeMilliseconds(1000);
>-  const int64_t kAvgRtt3 = 95;
>-  EXPECT_CALL(stats_observer, OnRttUpdate(kAvgRtt3, kLowRtt)).Times(1);
>-  call_stats_->Process();
> 
>-  call_stats_->DeregisterStatsObserver(&stats_observer);
>+  // Here, enough time must have passed in order to remove exactly the first
>+  // report and nothing else (>1500ms has passed since the first rtt).
>+  // So, this value is arrived by doing:
>+  // (kAvgRtt1)/1 * 0.7 + (kHighRtt+kLowRtt)/2 * 0.3 = 102.1
>+  static constexpr const int64_t kAvgRtt2 = 102;
>+  EXPECT_CALL(stats_observer, OnRttUpdate(kAvgRtt2, kHighRtt))
>+      .Times(1)
>+      .WillOnce(InvokeWithoutArgs([this] {
>+        // Advance time to make the high report invalid, the lower rtt should
>+        // now be in the callback.
>+        fake_clock_.AdvanceTimeMilliseconds(1000);
>+      }));
>+
>+  static constexpr const int64_t kAvgRtt3 = 95;
>+  EXPECT_CALL(stats_observer, OnRttUpdate(kAvgRtt3, kLowRtt))
>+      .Times(1)
>+      .WillOnce(InvokeWithoutArgs([&event] { event.Set(); }));
>+
>+  // Trigger the first rtt value and set off the chain of callbacks.
>+  rtcp_rtt_stats->OnRttUpdate(kFirstRtt);  // Reported at T0 (0ms).
>+  EXPECT_TRUE(event.Wait(1000));
>+
>+  call_stats_.DeregisterStatsObserver(&stats_observer);
> }
> 
> TEST_F(CallStatsTest, LastProcessedRtt) {
>+  rtc::Event event(false, false);
>   MockStatsObserver stats_observer;
>-  call_stats_->RegisterStatsObserver(&stats_observer);
>-  RtcpRttStats* rtcp_rtt_stats = call_stats_->rtcp_rtt_stats();
>-  fake_clock_.AdvanceTimeMilliseconds(1000);
>+  call_stats_.RegisterStatsObserver(&stats_observer);
>+  RtcpRttStats* rtcp_rtt_stats = &call_stats_;
>+
>+  static constexpr const int64_t kRttLow = 10;
>+  static constexpr const int64_t kRttHigh = 30;
>+  // The following two average numbers dependend on average + weight
>+  // calculations in call_stats.cc.
>+  static constexpr const int64_t kAvgRtt1 = 13;
>+  static constexpr const int64_t kAvgRtt2 = 15;
>+
>+  EXPECT_CALL(stats_observer, OnRttUpdate(kRttLow, kRttLow))
>+      .Times(1)
>+      .WillOnce(InvokeWithoutArgs([rtcp_rtt_stats] {
>+        EXPECT_EQ(kRttLow, rtcp_rtt_stats->LastProcessedRtt());
>+        // Don't advance the clock to make sure that low and high rtt values
>+        // are associated with the same time stamp.
>+        rtcp_rtt_stats->OnRttUpdate(kRttHigh);
>+      }));
>+
>+  EXPECT_CALL(stats_observer, OnRttUpdate(kAvgRtt1, kRttHigh))
>+      .Times(1)
>+      .WillOnce(InvokeWithoutArgs([rtcp_rtt_stats, this] {
>+        EXPECT_EQ(kAvgRtt1, rtcp_rtt_stats->LastProcessedRtt());
>+        fake_clock_.AdvanceTimeMilliseconds(CallStats::kUpdateIntervalMs);
>+        rtcp_rtt_stats->OnRttUpdate(kRttLow);
>+        rtcp_rtt_stats->OnRttUpdate(kRttHigh);
>+      }));
>+
>+  EXPECT_CALL(stats_observer, OnRttUpdate(kAvgRtt2, kRttHigh))
>+      .Times(1)
>+      .WillOnce(InvokeWithoutArgs([rtcp_rtt_stats, &event] {
>+        EXPECT_EQ(kAvgRtt2, rtcp_rtt_stats->LastProcessedRtt());
>+        event.Set();
>+      }));
> 
>   // Set a first values and verify that LastProcessedRtt initially returns the
>   // average rtt.
>-  const int64_t kRttLow = 10;
>-  const int64_t kRttHigh = 30;
>-  const int64_t kAvgRtt = 20;
>-  rtcp_rtt_stats->OnRttUpdate(kRttLow);
>-  rtcp_rtt_stats->OnRttUpdate(kRttHigh);
>-  EXPECT_CALL(stats_observer, OnRttUpdate(kAvgRtt, kRttHigh)).Times(1);
>-  call_stats_->Process();
>-  EXPECT_EQ(kAvgRtt, rtcp_rtt_stats->LastProcessedRtt());
>-
>-  // Update values and verify LastProcessedRtt.
>-  fake_clock_.AdvanceTimeMilliseconds(1000);
>+  fake_clock_.AdvanceTimeMilliseconds(CallStats::kUpdateIntervalMs);
>   rtcp_rtt_stats->OnRttUpdate(kRttLow);
>-  rtcp_rtt_stats->OnRttUpdate(kRttHigh);
>-  EXPECT_CALL(stats_observer, OnRttUpdate(kAvgRtt, kRttHigh)).Times(1);
>-  call_stats_->Process();
>-  EXPECT_EQ(kAvgRtt, rtcp_rtt_stats->LastProcessedRtt());
>+  EXPECT_TRUE(event.Wait(1000));
>+  EXPECT_EQ(kAvgRtt2, rtcp_rtt_stats->LastProcessedRtt());
> 
>-  call_stats_->DeregisterStatsObserver(&stats_observer);
>+  call_stats_.DeregisterStatsObserver(&stats_observer);
> }
> 
> TEST_F(CallStatsTest, ProducesHistogramMetrics) {
>   metrics::Reset();
>-  const int64_t kRtt = 123;
>-  RtcpRttStats* rtcp_rtt_stats = call_stats_->rtcp_rtt_stats();
>+  rtc::Event event(false, false);
>+  static constexpr const int64_t kRtt = 123;
>+  RtcpRttStats* rtcp_rtt_stats = &call_stats_;
>+  MockStatsObserver stats_observer;
>+  call_stats_.RegisterStatsObserver(&stats_observer);
>+  EXPECT_CALL(stats_observer, OnRttUpdate(kRtt, kRtt))
>+      .Times(AnyNumber())
>+      .WillOnce(InvokeWithoutArgs([&event] { event.Set(); }))
>+      .WillRepeatedly(Return());
>+
>   rtcp_rtt_stats->OnRttUpdate(kRtt);
>-  fake_clock_.AdvanceTimeMilliseconds(metrics::kMinRunTimeInSeconds * 1000);
>+  fake_clock_.AdvanceTimeMilliseconds(metrics::kMinRunTimeInSeconds *
>+                                      CallStats::kUpdateIntervalMs);
>   rtcp_rtt_stats->OnRttUpdate(kRtt);
>-  call_stats_->Process();
>-  call_stats_.reset();
>+  EXPECT_TRUE(event.Wait(1000));
>+
>+  call_stats_.DeregisterStatsObserver(&stats_observer);
>+
>+  process_thread_->Stop();
>+  call_stats_.UpdateHistogramsForTest();
> 
>   EXPECT_EQ(1, metrics::NumSamples(
>                    "WebRTC.Video.AverageRoundTripTimeInMilliseconds"));
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/encoder_rtcp_feedback.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/encoder_rtcp_feedback.cc
>index 5a1194f9660..8b680b260f7 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/encoder_rtcp_feedback.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/encoder_rtcp_feedback.cc
>@@ -10,8 +10,8 @@
> 
> #include "video/encoder_rtcp_feedback.h"
> 
>+#include "api/video/video_stream_encoder_interface.h"
> #include "rtc_base/checks.h"
>-#include "video/video_stream_encoder.h"
> 
> static const int kMinKeyFrameRequestIntervalMs = 300;
> 
>@@ -19,7 +19,7 @@ namespace webrtc {
> 
> EncoderRtcpFeedback::EncoderRtcpFeedback(Clock* clock,
>                                          const std::vector<uint32_t>& ssrcs,
>-                                         VideoStreamEncoder* encoder)
>+                                         VideoStreamEncoderInterface* encoder)
>     : clock_(clock),
>       ssrcs_(ssrcs),
>       video_stream_encoder_(encoder),
>@@ -60,7 +60,8 @@ void EncoderRtcpFeedback::OnReceivedIntraFrameRequest(uint32_t ssrc) {
>     time_last_intra_request_ms_[index] = now_ms;
>   }
> 
>-  video_stream_encoder_->OnReceivedIntraFrameRequest(index);
>+  // Always produce key frame for all streams.
>+  video_stream_encoder_->SendKeyFrame();
> }
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/encoder_rtcp_feedback.h b/Source/ThirdParty/libwebrtc/Source/webrtc/video/encoder_rtcp_feedback.h
>index fd57558fc6e..f6fa4a25f54 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/encoder_rtcp_feedback.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/encoder_rtcp_feedback.h
>@@ -15,17 +15,16 @@
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> #include "rtc_base/criticalsection.h"
> #include "system_wrappers/include/clock.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>-class VideoStreamEncoder;
>+class VideoStreamEncoderInterface;
> 
> class EncoderRtcpFeedback : public RtcpIntraFrameObserver {
>  public:
>   EncoderRtcpFeedback(Clock* clock,
>-                       const std::vector<uint32_t>& ssrcs,
>-                       VideoStreamEncoder* encoder);
>+                      const std::vector<uint32_t>& ssrcs,
>+                      VideoStreamEncoderInterface* encoder);
>   void OnReceivedIntraFrameRequest(uint32_t ssrc) override;
> 
>  private:
>@@ -34,7 +33,7 @@ class EncoderRtcpFeedback : public RtcpIntraFrameObserver {
> 
>   Clock* const clock_;
>   const std::vector<uint32_t> ssrcs_;
>-  VideoStreamEncoder* const video_stream_encoder_;
>+  VideoStreamEncoderInterface* const video_stream_encoder_;
> 
>   rtc::CriticalSection crit_;
>   std::vector<int64_t> time_last_intra_request_ms_ RTC_GUARDED_BY(crit_);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/encoder_rtcp_feedback_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/encoder_rtcp_feedback_unittest.cc
>index 8acbec41f9b..1ce387b9917 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/encoder_rtcp_feedback_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/encoder_rtcp_feedback_unittest.cc
>@@ -14,35 +14,17 @@
> 
> #include "test/gmock.h"
> #include "test/gtest.h"
>-#include "video/send_statistics_proxy.h"
>-#include "video/video_stream_encoder.h"
>+#include "video/test/mock_video_stream_encoder.h"
> 
> using ::testing::NiceMock;
> 
> namespace webrtc {
> 
>-class MockVideoStreamEncoder : public VideoStreamEncoder {
>- public:
>-  explicit MockVideoStreamEncoder(SendStatisticsProxy* send_stats_proxy)
>-      : VideoStreamEncoder(1,
>-                           send_stats_proxy,
>-                           VideoSendStream::Config::EncoderSettings("fake", 0,
>-                                                                    nullptr),
>-                           nullptr,
>-                           std::unique_ptr<OveruseFrameDetector>()) {}
>-  ~MockVideoStreamEncoder() { Stop(); }
>-
>-  MOCK_METHOD1(OnReceivedIntraFrameRequest, void(size_t));
>-};
>-
> class VieKeyRequestTest : public ::testing::Test {
>  public:
>   VieKeyRequestTest()
>       : simulated_clock_(123456789),
>-        send_stats_proxy_(&simulated_clock_,
>-                          VideoSendStream::Config(nullptr),
>-                          VideoEncoderConfig::ContentType::kRealtimeVideo),
>-        encoder_(&send_stats_proxy_),
>+        encoder_(),
>         encoder_rtcp_feedback_(
>             &simulated_clock_,
>             std::vector<uint32_t>(1, VieKeyRequestTest::kSsrc),
>@@ -52,24 +34,23 @@ class VieKeyRequestTest : public ::testing::Test {
>   const uint32_t kSsrc = 1234;
> 
>   SimulatedClock simulated_clock_;
>-  SendStatisticsProxy send_stats_proxy_;
>-  MockVideoStreamEncoder encoder_;
>+  testing::StrictMock<MockVideoStreamEncoder> encoder_;
>   EncoderRtcpFeedback encoder_rtcp_feedback_;
> };
> 
> TEST_F(VieKeyRequestTest, CreateAndTriggerRequests) {
>-  EXPECT_CALL(encoder_, OnReceivedIntraFrameRequest(0)).Times(1);
>+  EXPECT_CALL(encoder_, SendKeyFrame()).Times(1);
>   encoder_rtcp_feedback_.OnReceivedIntraFrameRequest(kSsrc);
> }
> 
> TEST_F(VieKeyRequestTest, TooManyOnReceivedIntraFrameRequest) {
>-  EXPECT_CALL(encoder_, OnReceivedIntraFrameRequest(0)).Times(1);
>+  EXPECT_CALL(encoder_, SendKeyFrame()).Times(1);
>   encoder_rtcp_feedback_.OnReceivedIntraFrameRequest(kSsrc);
>   encoder_rtcp_feedback_.OnReceivedIntraFrameRequest(kSsrc);
>   simulated_clock_.AdvanceTimeMilliseconds(10);
>   encoder_rtcp_feedback_.OnReceivedIntraFrameRequest(kSsrc);
> 
>-  EXPECT_CALL(encoder_, OnReceivedIntraFrameRequest(0)).Times(1);
>+  EXPECT_CALL(encoder_, SendKeyFrame()).Times(1);
>   simulated_clock_.AdvanceTimeMilliseconds(300);
>   encoder_rtcp_feedback_.OnReceivedIntraFrameRequest(kSsrc);
>   encoder_rtcp_feedback_.OnReceivedIntraFrameRequest(kSsrc);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests.cc
>deleted file mode 100644
>index 6df63eef8b3..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests.cc
>+++ /dev/null
>@@ -1,5110 +0,0 @@
>-/*
>- *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-#include <algorithm>
>-#include <list>
>-#include <map>
>-#include <memory>
>-#include <sstream>
>-#include <string>
>-#include <vector>
>-
>-#include "api/optional.h"
>-#include "api/video_codecs/video_encoder.h"
>-#include "call/call.h"
>-#include "common_video/include/frame_callback.h"
>-#include "logging/rtc_event_log/rtc_event_log.h"
>-#include "media/base/fakevideorenderer.h"
>-#include "media/base/mediaconstants.h"
>-#include "media/engine/internalencoderfactory.h"
>-#include "media/engine/simulcast_encoder_adapter.h"
>-#include "media/engine/webrtcvideoencoderfactory.h"
>-#include "modules/include/module_common_types.h"
>-#include "modules/rtp_rtcp/include/rtp_rtcp.h"
>-#include "modules/rtp_rtcp/source/byte_io.h"
>-#include "modules/rtp_rtcp/source/rtcp_packet/nack.h"
>-#include "modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h"
>-#include "modules/rtp_rtcp/source/rtp_format.h"
>-#include "modules/rtp_rtcp/source/rtp_utility.h"
>-#include "modules/video_coding/codecs/h264/include/h264.h"
>-#include "modules/video_coding/codecs/vp8/include/vp8.h"
>-#include "modules/video_coding/codecs/vp9/include/vp9.h"
>-#include "modules/video_coding/include/video_coding_defines.h"
>-#include "rtc_base/checks.h"
>-#include "rtc_base/event.h"
>-#include "rtc_base/file.h"
>-#include "rtc_base/ptr_util.h"
>-#include "rtc_base/random.h"
>-#include "rtc_base/rate_limiter.h"
>-#include "system_wrappers/include/metrics.h"
>-#include "system_wrappers/include/metrics_default.h"
>-#include "system_wrappers/include/sleep.h"
>-#include "test/call_test.h"
>-#include "test/direct_transport.h"
>-#include "test/encoder_settings.h"
>-#include "test/fake_decoder.h"
>-#include "test/fake_encoder.h"
>-#include "test/field_trial.h"
>-#include "test/frame_generator.h"
>-#include "test/frame_generator_capturer.h"
>-#include "test/gmock.h"
>-#include "test/gtest.h"
>-#include "test/null_transport.h"
>-#include "test/rtcp_packet_parser.h"
>-#include "test/rtp_rtcp_observer.h"
>-#include "test/testsupport/fileutils.h"
>-#include "test/testsupport/perf_test.h"
>-#include "video/transport_adapter.h"
>-
>-namespace webrtc {
>-
>-namespace {
>-constexpr int kSilenceTimeoutMs = 2000;
>-}
>-
>-class EndToEndTest : public test::CallTest,
>-                     public testing::WithParamInterface<std::string> {
>- public:
>-  EndToEndTest() : field_trial_(GetParam()) {}
>-
>-  virtual ~EndToEndTest() {
>-    EXPECT_EQ(nullptr, video_send_stream_);
>-    EXPECT_TRUE(video_receive_streams_.empty());
>-  }
>-
>- protected:
>-  class UnusedTransport : public Transport {
>-   private:
>-    bool SendRtp(const uint8_t* packet,
>-                 size_t length,
>-                 const PacketOptions& options) override {
>-      ADD_FAILURE() << "Unexpected RTP sent.";
>-      return false;
>-    }
>-
>-    bool SendRtcp(const uint8_t* packet, size_t length) override {
>-      ADD_FAILURE() << "Unexpected RTCP sent.";
>-      return false;
>-    }
>-  };
>-
>-  class RequiredTransport : public Transport {
>-   public:
>-    RequiredTransport(bool rtp_required, bool rtcp_required)
>-        : need_rtp_(rtp_required), need_rtcp_(rtcp_required) {}
>-    ~RequiredTransport() {
>-      if (need_rtp_) {
>-        ADD_FAILURE() << "Expected RTP packet not sent.";
>-      }
>-      if (need_rtcp_) {
>-        ADD_FAILURE() << "Expected RTCP packet not sent.";
>-      }
>-    }
>-
>-   private:
>-    bool SendRtp(const uint8_t* packet,
>-                 size_t length,
>-                 const PacketOptions& options) override {
>-      rtc::CritScope lock(&crit_);
>-      need_rtp_ = false;
>-      return true;
>-    }
>-
>-    bool SendRtcp(const uint8_t* packet, size_t length) override {
>-      rtc::CritScope lock(&crit_);
>-      need_rtcp_ = false;
>-      return true;
>-    }
>-    bool need_rtp_;
>-    bool need_rtcp_;
>-    rtc::CriticalSection crit_;
>-  };
>-
>-  void DecodesRetransmittedFrame(bool enable_rtx, bool enable_red);
>-  void ReceivesPliAndRecovers(int rtp_history_ms);
>-  void RespectsRtcpMode(RtcpMode rtcp_mode);
>-  void TestSendsSetSsrcs(size_t num_ssrcs, bool send_single_ssrc_first);
>-  void TestRtpStatePreservation(bool use_rtx, bool provoke_rtcpsr_before_rtp);
>-  void VerifyHistogramStats(bool use_rtx, bool use_fec, bool screenshare);
>-  void VerifyNewVideoSendStreamsRespectNetworkState(
>-      MediaType network_to_bring_up,
>-      VideoEncoder* encoder,
>-      Transport* transport);
>-  void VerifyNewVideoReceiveStreamsRespectNetworkState(
>-      MediaType network_to_bring_up,
>-      Transport* transport);
>-
>-  test::ScopedFieldTrials field_trial_;
>-};
>-
>-TEST_P(EndToEndTest, ReceiverCanBeStartedTwice) {
>-  CreateCalls(Call::Config(event_log_.get()), Call::Config(event_log_.get()));
>-
>-  test::NullTransport transport;
>-  CreateSendConfig(1, 0, 0, &transport);
>-  CreateMatchingReceiveConfigs(&transport);
>-
>-  CreateVideoStreams();
>-
>-  video_receive_streams_[0]->Start();
>-  video_receive_streams_[0]->Start();
>-
>-  DestroyStreams();
>-}
>-
>-TEST_P(EndToEndTest, ReceiverCanBeStoppedTwice) {
>-  CreateCalls(Call::Config(event_log_.get()), Call::Config(event_log_.get()));
>-
>-  test::NullTransport transport;
>-  CreateSendConfig(1, 0, 0, &transport);
>-  CreateMatchingReceiveConfigs(&transport);
>-
>-  CreateVideoStreams();
>-
>-  video_receive_streams_[0]->Stop();
>-  video_receive_streams_[0]->Stop();
>-
>-  DestroyStreams();
>-}
>-
>-TEST_P(EndToEndTest, ReceiverCanBeStoppedAndRestarted) {
>-  CreateCalls(Call::Config(event_log_.get()), Call::Config(event_log_.get()));
>-
>-  test::NullTransport transport;
>-  CreateSendConfig(1, 0, 0, &transport);
>-  CreateMatchingReceiveConfigs(&transport);
>-
>-  CreateVideoStreams();
>-
>-  video_receive_streams_[0]->Stop();
>-  video_receive_streams_[0]->Start();
>-  video_receive_streams_[0]->Stop();
>-
>-  DestroyStreams();
>-}
>-
>-TEST_P(EndToEndTest, RendersSingleDelayedFrame) {
>-  static const int kWidth = 320;
>-  static const int kHeight = 240;
>-  // This constant is chosen to be higher than the timeout in the video_render
>-  // module. This makes sure that frames aren't dropped if there are no other
>-  // frames in the queue.
>-  static const int kRenderDelayMs = 1000;
>-
>-  class Renderer : public rtc::VideoSinkInterface<VideoFrame> {
>-   public:
>-    Renderer() : event_(false, false) {}
>-
>-    void OnFrame(const VideoFrame& video_frame) override {
>-      SleepMs(kRenderDelayMs);
>-      event_.Set();
>-    }
>-
>-    bool Wait() { return event_.Wait(kDefaultTimeoutMs); }
>-
>-    rtc::Event event_;
>-  } renderer;
>-
>-  test::FrameForwarder frame_forwarder;
>-  std::unique_ptr<test::DirectTransport> sender_transport;
>-  std::unique_ptr<test::DirectTransport> receiver_transport;
>-
>-  task_queue_.SendTask([this, &renderer, &frame_forwarder, &sender_transport,
>-                        &receiver_transport]() {
>-    CreateCalls(Call::Config(event_log_.get()), Call::Config(event_log_.get()));
>-
>-    sender_transport = rtc::MakeUnique<test::DirectTransport>(
>-        &task_queue_, sender_call_.get(), payload_type_map_);
>-    receiver_transport = rtc::MakeUnique<test::DirectTransport>(
>-        &task_queue_, receiver_call_.get(), payload_type_map_);
>-    sender_transport->SetReceiver(receiver_call_->Receiver());
>-    receiver_transport->SetReceiver(sender_call_->Receiver());
>-
>-    CreateSendConfig(1, 0, 0, sender_transport.get());
>-    CreateMatchingReceiveConfigs(receiver_transport.get());
>-
>-    video_receive_configs_[0].renderer = &renderer;
>-
>-    CreateVideoStreams();
>-    Start();
>-
>-    // Create frames that are smaller than the send width/height, this is done
>-    // to check that the callbacks are done after processing video.
>-    std::unique_ptr<test::FrameGenerator> frame_generator(
>-        test::FrameGenerator::CreateSquareGenerator(kWidth, kHeight));
>-    video_send_stream_->SetSource(
>-        &frame_forwarder,
>-        VideoSendStream::DegradationPreference::kMaintainFramerate);
>-
>-    frame_forwarder.IncomingCapturedFrame(*frame_generator->NextFrame());
>-  });
>-
>-  EXPECT_TRUE(renderer.Wait())
>-      << "Timed out while waiting for the frame to render.";
>-
>-  task_queue_.SendTask([this, &sender_transport, &receiver_transport]() {
>-    Stop();
>-    DestroyStreams();
>-    sender_transport.reset();
>-    receiver_transport.reset();
>-    DestroyCalls();
>-  });
>-}
>-
>-TEST_P(EndToEndTest, TransmitsFirstFrame) {
>-  class Renderer : public rtc::VideoSinkInterface<VideoFrame> {
>-   public:
>-    Renderer() : event_(false, false) {}
>-
>-    void OnFrame(const VideoFrame& video_frame) override { event_.Set(); }
>-
>-    bool Wait() { return event_.Wait(kDefaultTimeoutMs); }
>-
>-    rtc::Event event_;
>-  } renderer;
>-
>-  std::unique_ptr<test::FrameGenerator> frame_generator;
>-  test::FrameForwarder frame_forwarder;
>-
>-  std::unique_ptr<test::DirectTransport> sender_transport;
>-  std::unique_ptr<test::DirectTransport> receiver_transport;
>-
>-  task_queue_.SendTask([this, &renderer, &frame_generator, &frame_forwarder,
>-                        &sender_transport, &receiver_transport]() {
>-    CreateCalls(Call::Config(event_log_.get()), Call::Config(event_log_.get()));
>-
>-    sender_transport = rtc::MakeUnique<test::DirectTransport>(
>-        &task_queue_, sender_call_.get(), payload_type_map_);
>-    receiver_transport = rtc::MakeUnique<test::DirectTransport>(
>-        &task_queue_, receiver_call_.get(), payload_type_map_);
>-    sender_transport->SetReceiver(receiver_call_->Receiver());
>-    receiver_transport->SetReceiver(sender_call_->Receiver());
>-
>-    CreateSendConfig(1, 0, 0, sender_transport.get());
>-    CreateMatchingReceiveConfigs(receiver_transport.get());
>-    video_receive_configs_[0].renderer = &renderer;
>-
>-    CreateVideoStreams();
>-    Start();
>-
>-    frame_generator = test::FrameGenerator::CreateSquareGenerator(
>-        kDefaultWidth, kDefaultHeight);
>-    video_send_stream_->SetSource(
>-        &frame_forwarder,
>-        VideoSendStream::DegradationPreference::kMaintainFramerate);
>-    frame_forwarder.IncomingCapturedFrame(*frame_generator->NextFrame());
>-  });
>-
>-  EXPECT_TRUE(renderer.Wait())
>-      << "Timed out while waiting for the frame to render.";
>-
>-  task_queue_.SendTask([this, &sender_transport, &receiver_transport]() {
>-    Stop();
>-    DestroyStreams();
>-    sender_transport.reset();
>-    receiver_transport.reset();
>-    DestroyCalls();
>-  });
>-}
>-
>-class CodecObserver : public test::EndToEndTest,
>-                      public rtc::VideoSinkInterface<VideoFrame> {
>- public:
>-  CodecObserver(int no_frames_to_wait_for,
>-                VideoRotation rotation_to_test,
>-                const std::string& payload_name,
>-                std::unique_ptr<webrtc::VideoEncoder> encoder,
>-                std::unique_ptr<webrtc::VideoDecoder> decoder)
>-      : EndToEndTest(4 * webrtc::EndToEndTest::kDefaultTimeoutMs),
>-        // TODO(hta): This timeout (120 seconds) is excessive.
>-        // https://bugs.webrtc.org/6830
>-        no_frames_to_wait_for_(no_frames_to_wait_for),
>-        expected_rotation_(rotation_to_test),
>-        payload_name_(payload_name),
>-        encoder_(std::move(encoder)),
>-        decoder_(std::move(decoder)),
>-        frame_counter_(0) {}
>-
>-  void PerformTest() override {
>-    EXPECT_TRUE(Wait())
>-        << "Timed out while waiting for enough frames to be decoded.";
>-  }
>-
>-  void ModifyVideoConfigs(
>-      VideoSendStream::Config* send_config,
>-      std::vector<VideoReceiveStream::Config>* receive_configs,
>-      VideoEncoderConfig* encoder_config) override {
>-    send_config->encoder_settings.encoder = encoder_.get();
>-    send_config->encoder_settings.payload_name = payload_name_;
>-    send_config->encoder_settings.payload_type =
>-        test::CallTest::kVideoSendPayloadType;
>-
>-    (*receive_configs)[0].renderer = this;
>-    (*receive_configs)[0].decoders.resize(1);
>-    (*receive_configs)[0].decoders[0].payload_type =
>-        send_config->encoder_settings.payload_type;
>-    (*receive_configs)[0].decoders[0].payload_name =
>-        send_config->encoder_settings.payload_name;
>-    (*receive_configs)[0].decoders[0].decoder = decoder_.get();
>-  }
>-
>-  void OnFrame(const VideoFrame& video_frame) override {
>-    EXPECT_EQ(expected_rotation_, video_frame.rotation());
>-    if (++frame_counter_ == no_frames_to_wait_for_)
>-      observation_complete_.Set();
>-  }
>-
>-  void OnFrameGeneratorCapturerCreated(
>-      test::FrameGeneratorCapturer* frame_generator_capturer) override {
>-    frame_generator_capturer->SetFakeRotation(expected_rotation_);
>-  }
>-
>- private:
>-  int no_frames_to_wait_for_;
>-  VideoRotation expected_rotation_;
>-  std::string payload_name_;
>-  std::unique_ptr<webrtc::VideoEncoder> encoder_;
>-  std::unique_ptr<webrtc::VideoDecoder> decoder_;
>-  int frame_counter_;
>-};
>-
>-TEST_P(EndToEndTest, SendsAndReceivesVP8) {
>-  CodecObserver test(5, kVideoRotation_0, "VP8", VP8Encoder::Create(),
>-                     VP8Decoder::Create());
>-  RunBaseTest(&test);
>-}
>-
>-TEST_P(EndToEndTest, SendsAndReceivesVP8Rotation90) {
>-  CodecObserver test(5, kVideoRotation_90, "VP8", VP8Encoder::Create(),
>-                     VP8Decoder::Create());
>-  RunBaseTest(&test);
>-}
>-
>-#if !defined(RTC_DISABLE_VP9)
>-TEST_P(EndToEndTest, SendsAndReceivesVP9) {
>-  CodecObserver test(500, kVideoRotation_0, "VP9", VP9Encoder::Create(),
>-                     VP9Decoder::Create());
>-  RunBaseTest(&test);
>-}
>-
>-TEST_P(EndToEndTest, SendsAndReceivesVP9VideoRotation90) {
>-  CodecObserver test(5, kVideoRotation_90, "VP9", VP9Encoder::Create(),
>-                     VP9Decoder::Create());
>-  RunBaseTest(&test);
>-}
>-#endif  // !defined(RTC_DISABLE_VP9)
>-
>-#if defined(WEBRTC_USE_H264)
>-class EndToEndTestH264 : public EndToEndTest {};
>-
>-const auto h264_field_trial_combinations = ::testing::Values(
>-    "WebRTC-SpsPpsIdrIsH264Keyframe/Disabled/WebRTC-RoundRobinPacing/Disabled/",
>-    "WebRTC-SpsPpsIdrIsH264Keyframe/Enabled/WebRTC-RoundRobinPacing/Disabled/",
>-    "WebRTC-SpsPpsIdrIsH264Keyframe/Disabled/WebRTC-RoundRobinPacing/Enabled/",
>-    "WebRTC-SpsPpsIdrIsH264Keyframe/Enabled/WebRTC-RoundRobinPacing/Enabled/");
>-INSTANTIATE_TEST_CASE_P(SpsPpsIdrIsKeyframe,
>-                        EndToEndTestH264,
>-                        h264_field_trial_combinations);
>-
>-TEST_P(EndToEndTestH264, SendsAndReceivesH264) {
>-  CodecObserver test(500, kVideoRotation_0, "H264",
>-                     H264Encoder::Create(cricket::VideoCodec("H264")),
>-                     H264Decoder::Create());
>-  RunBaseTest(&test);
>-}
>-
>-TEST_P(EndToEndTestH264, SendsAndReceivesH264VideoRotation90) {
>-  CodecObserver test(5, kVideoRotation_90, "H264",
>-                     H264Encoder::Create(cricket::VideoCodec("H264")),
>-                     H264Decoder::Create());
>-  RunBaseTest(&test);
>-}
>-
>-TEST_P(EndToEndTestH264, SendsAndReceivesH264PacketizationMode0) {
>-  cricket::VideoCodec codec = cricket::VideoCodec("H264");
>-  codec.SetParam(cricket::kH264FmtpPacketizationMode, "0");
>-  CodecObserver test(500, kVideoRotation_0, "H264", H264Encoder::Create(codec),
>-                     H264Decoder::Create());
>-  RunBaseTest(&test);
>-}
>-
>-TEST_P(EndToEndTestH264, SendsAndReceivesH264PacketizationMode1) {
>-  cricket::VideoCodec codec = cricket::VideoCodec("H264");
>-  codec.SetParam(cricket::kH264FmtpPacketizationMode, "1");
>-  CodecObserver test(500, kVideoRotation_0, "H264", H264Encoder::Create(codec),
>-                     H264Decoder::Create());
>-  RunBaseTest(&test);
>-}
>-#endif  // defined(WEBRTC_USE_H264)
>-
>-TEST_P(EndToEndTest, ReceiverUsesLocalSsrc) {
>-  class SyncRtcpObserver : public test::EndToEndTest {
>-   public:
>-    SyncRtcpObserver() : EndToEndTest(kDefaultTimeoutMs) {}
>-
>-    Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
>-      test::RtcpPacketParser parser;
>-      EXPECT_TRUE(parser.Parse(packet, length));
>-      EXPECT_EQ(kReceiverLocalVideoSsrc, parser.sender_ssrc());
>-      observation_complete_.Set();
>-
>-      return SEND_PACKET;
>-    }
>-
>-    void PerformTest() override {
>-      EXPECT_TRUE(Wait())
>-          << "Timed out while waiting for a receiver RTCP packet to be sent.";
>-    }
>-  } test;
>-
>-  RunBaseTest(&test);
>-}
>-
>-TEST_P(EndToEndTest, ReceivesAndRetransmitsNack) {
>-  static const int kNumberOfNacksToObserve = 2;
>-  static const int kLossBurstSize = 2;
>-  static const int kPacketsBetweenLossBursts = 9;
>-  class NackObserver : public test::EndToEndTest {
>-   public:
>-    NackObserver()
>-        : EndToEndTest(kLongTimeoutMs),
>-          sent_rtp_packets_(0),
>-          packets_left_to_drop_(0),
>-          nacks_left_(kNumberOfNacksToObserve) {}
>-
>-   private:
>-    Action OnSendRtp(const uint8_t* packet, size_t length) override {
>-      rtc::CritScope lock(&crit_);
>-      RTPHeader header;
>-      EXPECT_TRUE(parser_->Parse(packet, length, &header));
>-
>-      // Never drop retransmitted packets.
>-      if (dropped_packets_.find(header.sequenceNumber) !=
>-          dropped_packets_.end()) {
>-        retransmitted_packets_.insert(header.sequenceNumber);
>-        return SEND_PACKET;
>-      }
>-
>-      if (nacks_left_ <= 0 &&
>-          retransmitted_packets_.size() == dropped_packets_.size()) {
>-        observation_complete_.Set();
>-      }
>-
>-      ++sent_rtp_packets_;
>-
>-      // Enough NACKs received, stop dropping packets.
>-      if (nacks_left_ <= 0)
>-        return SEND_PACKET;
>-
>-      // Check if it's time for a new loss burst.
>-      if (sent_rtp_packets_ % kPacketsBetweenLossBursts == 0)
>-        packets_left_to_drop_ = kLossBurstSize;
>-
>-      // Never drop padding packets as those won't be retransmitted.
>-      if (packets_left_to_drop_ > 0 && header.paddingLength == 0) {
>-        --packets_left_to_drop_;
>-        dropped_packets_.insert(header.sequenceNumber);
>-        return DROP_PACKET;
>-      }
>-
>-      return SEND_PACKET;
>-    }
>-
>-    Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
>-      rtc::CritScope lock(&crit_);
>-      test::RtcpPacketParser parser;
>-      EXPECT_TRUE(parser.Parse(packet, length));
>-      nacks_left_ -= parser.nack()->num_packets();
>-      return SEND_PACKET;
>-    }
>-
>-    void ModifyVideoConfigs(
>-        VideoSendStream::Config* send_config,
>-        std::vector<VideoReceiveStream::Config>* receive_configs,
>-        VideoEncoderConfig* encoder_config) override {
>-      send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
>-      (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
>-    }
>-
>-    void PerformTest() override {
>-      EXPECT_TRUE(Wait())
>-          << "Timed out waiting for packets to be NACKed, retransmitted and "
>-             "rendered.";
>-    }
>-
>-    rtc::CriticalSection crit_;
>-    std::set<uint16_t> dropped_packets_;
>-    std::set<uint16_t> retransmitted_packets_;
>-    uint64_t sent_rtp_packets_;
>-    int packets_left_to_drop_;
>-    int nacks_left_ RTC_GUARDED_BY(&crit_);
>-  } test;
>-
>-  RunBaseTest(&test);
>-}
>-
>-TEST_P(EndToEndTest, ReceivesNackAndRetransmitsAudio) {
>-  class NackObserver : public test::EndToEndTest {
>-   public:
>-    NackObserver()
>-        : EndToEndTest(kLongTimeoutMs),
>-          local_ssrc_(0),
>-          remote_ssrc_(0),
>-          receive_transport_(nullptr) {}
>-
>-   private:
>-    size_t GetNumVideoStreams() const override { return 0; }
>-    size_t GetNumAudioStreams() const override { return 1; }
>-
>-    test::PacketTransport* CreateReceiveTransport(
>-        test::SingleThreadedTaskQueueForTesting* task_queue) override {
>-      test::PacketTransport* receive_transport = new test::PacketTransport(
>-          task_queue, nullptr, this, test::PacketTransport::kReceiver,
>-          payload_type_map_, FakeNetworkPipe::Config());
>-      receive_transport_ = receive_transport;
>-      return receive_transport;
>-    }
>-
>-    Action OnSendRtp(const uint8_t* packet, size_t length) override {
>-      RTPHeader header;
>-      EXPECT_TRUE(parser_->Parse(packet, length, &header));
>-
>-      if (!sequence_number_to_retransmit_) {
>-        sequence_number_to_retransmit_ = header.sequenceNumber;
>-
>-        // Don't ask for retransmission straight away, may be deduped in pacer.
>-      } else if (header.sequenceNumber == *sequence_number_to_retransmit_) {
>-        observation_complete_.Set();
>-      } else {
>-        // Send a NACK as often as necessary until retransmission is received.
>-        rtcp::Nack nack;
>-        nack.SetSenderSsrc(local_ssrc_);
>-        nack.SetMediaSsrc(remote_ssrc_);
>-        uint16_t nack_list[] = {*sequence_number_to_retransmit_};
>-        nack.SetPacketIds(nack_list, 1);
>-        rtc::Buffer buffer = nack.Build();
>-
>-        EXPECT_TRUE(receive_transport_->SendRtcp(buffer.data(), buffer.size()));
>-      }
>-
>-      return SEND_PACKET;
>-    }
>-
>-    void ModifyAudioConfigs(
>-        AudioSendStream::Config* send_config,
>-        std::vector<AudioReceiveStream::Config>* receive_configs) override {
>-      send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
>-      (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
>-      local_ssrc_ = (*receive_configs)[0].rtp.local_ssrc;
>-      remote_ssrc_ = (*receive_configs)[0].rtp.remote_ssrc;
>-    }
>-
>-    void PerformTest() override {
>-      EXPECT_TRUE(Wait())
>-          << "Timed out waiting for packets to be NACKed, retransmitted and "
>-             "rendered.";
>-    }
>-
>-    uint32_t local_ssrc_;
>-    uint32_t remote_ssrc_;
>-    Transport* receive_transport_;
>-    rtc::Optional<uint16_t> sequence_number_to_retransmit_;
>-  } test;
>-
>-  RunBaseTest(&test);
>-}
>-
>-TEST_P(EndToEndTest, ReceivesUlpfec) {
>-  class UlpfecRenderObserver : public test::EndToEndTest,
>-                               public rtc::VideoSinkInterface<VideoFrame> {
>-   public:
>-    UlpfecRenderObserver()
>-        : EndToEndTest(kDefaultTimeoutMs),
>-          encoder_(VP8Encoder::Create()),
>-          random_(0xcafef00d1),
>-          num_packets_sent_(0) {}
>-
>-   private:
>-    Action OnSendRtp(const uint8_t* packet, size_t length) override {
>-      rtc::CritScope lock(&crit_);
>-      RTPHeader header;
>-      EXPECT_TRUE(parser_->Parse(packet, length, &header));
>-
>-      EXPECT_TRUE(header.payloadType == kVideoSendPayloadType ||
>-                  header.payloadType == kRedPayloadType)
>-          << "Unknown payload type received.";
>-      EXPECT_EQ(kVideoSendSsrcs[0], header.ssrc) << "Unknown SSRC received.";
>-
>-      // Parse RED header.
>-      int encapsulated_payload_type = -1;
>-      if (header.payloadType == kRedPayloadType) {
>-        encapsulated_payload_type =
>-            static_cast<int>(packet[header.headerLength]);
>-
>-        EXPECT_TRUE(encapsulated_payload_type == kVideoSendPayloadType ||
>-                    encapsulated_payload_type == kUlpfecPayloadType)
>-            << "Unknown encapsulated payload type received.";
>-      }
>-
>-      // To minimize test flakiness, always let ULPFEC packets through.
>-      if (encapsulated_payload_type == kUlpfecPayloadType) {
>-        return SEND_PACKET;
>-      }
>-
>-      // Simulate 5% video packet loss after rampup period. Record the
>-      // corresponding timestamps that were dropped.
>-      if (num_packets_sent_++ > 100 && random_.Rand(1, 100) <= 5) {
>-        if (encapsulated_payload_type == kVideoSendPayloadType) {
>-          dropped_sequence_numbers_.insert(header.sequenceNumber);
>-          dropped_timestamps_.insert(header.timestamp);
>-        }
>-        return DROP_PACKET;
>-      }
>-
>-      return SEND_PACKET;
>-    }
>-
>-    void OnFrame(const VideoFrame& video_frame) override {
>-      rtc::CritScope lock(&crit_);
>-      // Rendering frame with timestamp of packet that was dropped -> FEC
>-      // protection worked.
>-      auto it = dropped_timestamps_.find(video_frame.timestamp());
>-      if (it != dropped_timestamps_.end()) {
>-        observation_complete_.Set();
>-      }
>-    }
>-
>-    void ModifyVideoConfigs(
>-        VideoSendStream::Config* send_config,
>-        std::vector<VideoReceiveStream::Config>* receive_configs,
>-        VideoEncoderConfig* encoder_config) override {
>-      // Use VP8 instead of FAKE, since the latter does not have PictureID
>-      // in the packetization headers.
>-      send_config->encoder_settings.encoder = encoder_.get();
>-      send_config->encoder_settings.payload_name = "VP8";
>-      send_config->encoder_settings.payload_type = kVideoSendPayloadType;
>-      VideoReceiveStream::Decoder decoder =
>-          test::CreateMatchingDecoder(send_config->encoder_settings);
>-      decoder_.reset(decoder.decoder);
>-      (*receive_configs)[0].decoders.clear();
>-      (*receive_configs)[0].decoders.push_back(decoder);
>-
>-      // Enable ULPFEC over RED.
>-      send_config->rtp.ulpfec.red_payload_type = kRedPayloadType;
>-      send_config->rtp.ulpfec.ulpfec_payload_type = kUlpfecPayloadType;
>-      (*receive_configs)[0].rtp.red_payload_type = kRedPayloadType;
>-      (*receive_configs)[0].rtp.ulpfec_payload_type = kUlpfecPayloadType;
>-
>-      (*receive_configs)[0].renderer = this;
>-    }
>-
>-    void PerformTest() override {
>-      EXPECT_TRUE(Wait())
>-          << "Timed out waiting for dropped frames to be rendered.";
>-    }
>-
>-    rtc::CriticalSection crit_;
>-    std::unique_ptr<VideoEncoder> encoder_;
>-    std::unique_ptr<VideoDecoder> decoder_;
>-    std::set<uint32_t> dropped_sequence_numbers_ RTC_GUARDED_BY(crit_);
>-    // Several packets can have the same timestamp.
>-    std::multiset<uint32_t> dropped_timestamps_ RTC_GUARDED_BY(crit_);
>-    Random random_;
>-    int num_packets_sent_ RTC_GUARDED_BY(crit_);
>-  } test;
>-
>-  RunBaseTest(&test);
>-}
>-
>-class FlexfecRenderObserver : public test::EndToEndTest,
>-                              public rtc::VideoSinkInterface<VideoFrame> {
>- public:
>-  static constexpr uint32_t kVideoLocalSsrc = 123;
>-  static constexpr uint32_t kFlexfecLocalSsrc = 456;
>-
>-  explicit FlexfecRenderObserver(bool enable_nack, bool expect_flexfec_rtcp)
>-      : test::EndToEndTest(test::CallTest::kDefaultTimeoutMs),
>-        enable_nack_(enable_nack),
>-        expect_flexfec_rtcp_(expect_flexfec_rtcp),
>-        received_flexfec_rtcp_(false),
>-        random_(0xcafef00d1),
>-        num_packets_sent_(0) {}
>-
>-  size_t GetNumFlexfecStreams() const override { return 1; }
>-
>- private:
>-  Action OnSendRtp(const uint8_t* packet, size_t length) override {
>-    rtc::CritScope lock(&crit_);
>-    RTPHeader header;
>-    EXPECT_TRUE(parser_->Parse(packet, length, &header));
>-
>-    EXPECT_TRUE(header.payloadType ==
>-                    test::CallTest::kFakeVideoSendPayloadType ||
>-                header.payloadType == test::CallTest::kFlexfecPayloadType ||
>-                (enable_nack_ &&
>-                 header.payloadType == test::CallTest::kSendRtxPayloadType))
>-        << "Unknown payload type received.";
>-    EXPECT_TRUE(
>-        header.ssrc == test::CallTest::kVideoSendSsrcs[0] ||
>-        header.ssrc == test::CallTest::kFlexfecSendSsrc ||
>-        (enable_nack_ && header.ssrc == test::CallTest::kSendRtxSsrcs[0]))
>-        << "Unknown SSRC received.";
>-
>-    // To reduce test flakiness, always let FlexFEC packets through.
>-    if (header.payloadType == test::CallTest::kFlexfecPayloadType) {
>-      EXPECT_EQ(test::CallTest::kFlexfecSendSsrc, header.ssrc);
>-
>-      return SEND_PACKET;
>-    }
>-
>-    // To reduce test flakiness, always let RTX packets through.
>-    if (header.payloadType == test::CallTest::kSendRtxPayloadType) {
>-      EXPECT_EQ(test::CallTest::kSendRtxSsrcs[0], header.ssrc);
>-
>-      // Parse RTX header.
>-      uint16_t original_sequence_number =
>-          ByteReader<uint16_t>::ReadBigEndian(&packet[header.headerLength]);
>-
>-      // From the perspective of FEC, a retransmitted packet is no longer
>-      // dropped, so remove it from list of dropped packets.
>-      auto seq_num_it =
>-          dropped_sequence_numbers_.find(original_sequence_number);
>-      if (seq_num_it != dropped_sequence_numbers_.end()) {
>-        dropped_sequence_numbers_.erase(seq_num_it);
>-        auto ts_it = dropped_timestamps_.find(header.timestamp);
>-        EXPECT_NE(ts_it, dropped_timestamps_.end());
>-        dropped_timestamps_.erase(ts_it);
>-      }
>-
>-      return SEND_PACKET;
>-    }
>-
>-    // Simulate 5% video packet loss after rampup period. Record the
>-    // corresponding timestamps that were dropped.
>-    if (num_packets_sent_++ > 100 && random_.Rand(1, 100) <= 5) {
>-      EXPECT_EQ(test::CallTest::kFakeVideoSendPayloadType, header.payloadType);
>-      EXPECT_EQ(test::CallTest::kVideoSendSsrcs[0], header.ssrc);
>-
>-      dropped_sequence_numbers_.insert(header.sequenceNumber);
>-      dropped_timestamps_.insert(header.timestamp);
>-
>-      return DROP_PACKET;
>-    }
>-
>-    return SEND_PACKET;
>-  }
>-
>-  Action OnReceiveRtcp(const uint8_t* data, size_t length) override {
>-    test::RtcpPacketParser parser;
>-
>-    parser.Parse(data, length);
>-    if (parser.sender_ssrc() == kFlexfecLocalSsrc) {
>-      EXPECT_EQ(1, parser.receiver_report()->num_packets());
>-      const std::vector<rtcp::ReportBlock>& report_blocks =
>-          parser.receiver_report()->report_blocks();
>-      if (!report_blocks.empty()) {
>-        EXPECT_EQ(1U, report_blocks.size());
>-        EXPECT_EQ(test::CallTest::kFlexfecSendSsrc,
>-                  report_blocks[0].source_ssrc());
>-        rtc::CritScope lock(&crit_);
>-        received_flexfec_rtcp_ = true;
>-      }
>-    }
>-
>-    return SEND_PACKET;
>-  }
>-
>-  test::PacketTransport* CreateSendTransport(
>-      test::SingleThreadedTaskQueueForTesting* task_queue,
>-      Call* sender_call) override {
>-    // At low RTT (< kLowRttNackMs) -> NACK only, no FEC.
>-    const int kNetworkDelayMs = 100;
>-    FakeNetworkPipe::Config config;
>-    config.queue_delay_ms = kNetworkDelayMs;
>-    return new test::PacketTransport(task_queue, sender_call, this,
>-                                     test::PacketTransport::kSender,
>-                                     test::CallTest::payload_type_map_, config);
>-  }
>-
>-  void OnFrame(const VideoFrame& video_frame) override {
>-    EXPECT_EQ(kVideoRotation_90, video_frame.rotation());
>-
>-    rtc::CritScope lock(&crit_);
>-    // Rendering frame with timestamp of packet that was dropped -> FEC
>-    // protection worked.
>-    auto it = dropped_timestamps_.find(video_frame.timestamp());
>-    if (it != dropped_timestamps_.end()) {
>-      if (!expect_flexfec_rtcp_ || received_flexfec_rtcp_) {
>-        observation_complete_.Set();
>-      }
>-    }
>-  }
>-
>-  void ModifyVideoConfigs(
>-      VideoSendStream::Config* send_config,
>-      std::vector<VideoReceiveStream::Config>* receive_configs,
>-      VideoEncoderConfig* encoder_config) override {
>-    (*receive_configs)[0].rtp.local_ssrc = kVideoLocalSsrc;
>-    (*receive_configs)[0].renderer = this;
>-
>-    if (enable_nack_) {
>-      send_config->rtp.nack.rtp_history_ms = test::CallTest::kNackRtpHistoryMs;
>-      send_config->rtp.rtx.ssrcs.push_back(test::CallTest::kSendRtxSsrcs[0]);
>-      send_config->rtp.rtx.payload_type = test::CallTest::kSendRtxPayloadType;
>-
>-      (*receive_configs)[0].rtp.nack.rtp_history_ms =
>-          test::CallTest::kNackRtpHistoryMs;
>-      (*receive_configs)[0].rtp.rtx_ssrc = test::CallTest::kSendRtxSsrcs[0];
>-      (*receive_configs)[0]
>-          .rtp
>-          .rtx_associated_payload_types[test::CallTest::kSendRtxPayloadType] =
>-          test::CallTest::kVideoSendPayloadType;
>-    }
>-  }
>-
>-  void OnFrameGeneratorCapturerCreated(
>-      test::FrameGeneratorCapturer* frame_generator_capturer) override {
>-    frame_generator_capturer->SetFakeRotation(kVideoRotation_90);
>-  }
>-
>-  void ModifyFlexfecConfigs(
>-      std::vector<FlexfecReceiveStream::Config>* receive_configs) override {
>-    (*receive_configs)[0].local_ssrc = kFlexfecLocalSsrc;
>-  }
>-
>-  void PerformTest() override {
>-    EXPECT_TRUE(Wait())
>-        << "Timed out waiting for dropped frames to be rendered.";
>-  }
>-
>-  rtc::CriticalSection crit_;
>-  std::set<uint32_t> dropped_sequence_numbers_ RTC_GUARDED_BY(crit_);
>-  // Several packets can have the same timestamp.
>-  std::multiset<uint32_t> dropped_timestamps_ RTC_GUARDED_BY(crit_);
>-  const bool enable_nack_;
>-  const bool expect_flexfec_rtcp_;
>-  bool received_flexfec_rtcp_ RTC_GUARDED_BY(crit_);
>-  Random random_;
>-  int num_packets_sent_;
>-};
>-
>-TEST_P(EndToEndTest, RecoversWithFlexfec) {
>-  FlexfecRenderObserver test(false, false);
>-  RunBaseTest(&test);
>-}
>-
>-TEST_P(EndToEndTest, RecoversWithFlexfecAndNack) {
>-  FlexfecRenderObserver test(true, false);
>-  RunBaseTest(&test);
>-}
>-
>-TEST_P(EndToEndTest, RecoversWithFlexfecAndSendsCorrespondingRtcp) {
>-  FlexfecRenderObserver test(false, true);
>-  RunBaseTest(&test);
>-}
>-
>-TEST_P(EndToEndTest, ReceivedUlpfecPacketsNotNacked) {
>-  class UlpfecNackObserver : public test::EndToEndTest {
>-   public:
>-    UlpfecNackObserver()
>-        : EndToEndTest(kDefaultTimeoutMs),
>-          state_(kFirstPacket),
>-          ulpfec_sequence_number_(0),
>-          has_last_sequence_number_(false),
>-          last_sequence_number_(0),
>-          encoder_(VP8Encoder::Create()),
>-          decoder_(VP8Decoder::Create()) {}
>-
>-   private:
>-    Action OnSendRtp(const uint8_t* packet, size_t length) override {
>-      rtc::CritScope lock_(&crit_);
>-      RTPHeader header;
>-      EXPECT_TRUE(parser_->Parse(packet, length, &header));
>-
>-      int encapsulated_payload_type = -1;
>-      if (header.payloadType == kRedPayloadType) {
>-        encapsulated_payload_type =
>-            static_cast<int>(packet[header.headerLength]);
>-        if (encapsulated_payload_type != kFakeVideoSendPayloadType)
>-          EXPECT_EQ(kUlpfecPayloadType, encapsulated_payload_type);
>-      } else {
>-        EXPECT_EQ(kFakeVideoSendPayloadType, header.payloadType);
>-      }
>-
>-      if (has_last_sequence_number_ &&
>-          !IsNewerSequenceNumber(header.sequenceNumber,
>-                                 last_sequence_number_)) {
>-        // Drop retransmitted packets.
>-        return DROP_PACKET;
>-      }
>-      last_sequence_number_ = header.sequenceNumber;
>-      has_last_sequence_number_ = true;
>-
>-      bool ulpfec_packet = encapsulated_payload_type == kUlpfecPayloadType;
>-      switch (state_) {
>-        case kFirstPacket:
>-          state_ = kDropEveryOtherPacketUntilUlpfec;
>-          break;
>-        case kDropEveryOtherPacketUntilUlpfec:
>-          if (ulpfec_packet) {
>-            state_ = kDropAllMediaPacketsUntilUlpfec;
>-          } else if (header.sequenceNumber % 2 == 0) {
>-            return DROP_PACKET;
>-          }
>-          break;
>-        case kDropAllMediaPacketsUntilUlpfec:
>-          if (!ulpfec_packet)
>-            return DROP_PACKET;
>-          ulpfec_sequence_number_ = header.sequenceNumber;
>-          state_ = kDropOneMediaPacket;
>-          break;
>-        case kDropOneMediaPacket:
>-          if (ulpfec_packet)
>-            return DROP_PACKET;
>-          state_ = kPassOneMediaPacket;
>-          return DROP_PACKET;
>-          break;
>-        case kPassOneMediaPacket:
>-          if (ulpfec_packet)
>-            return DROP_PACKET;
>-          // Pass one media packet after dropped packet after last FEC,
>-          // otherwise receiver might never see a seq_no after
>-          // |ulpfec_sequence_number_|
>-          state_ = kVerifyUlpfecPacketNotInNackList;
>-          break;
>-        case kVerifyUlpfecPacketNotInNackList:
>-          // Continue to drop packets. Make sure no frame can be decoded.
>-          if (ulpfec_packet || header.sequenceNumber % 2 == 0)
>-            return DROP_PACKET;
>-          break;
>-      }
>-      return SEND_PACKET;
>-    }
>-
>-    Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
>-      rtc::CritScope lock_(&crit_);
>-      if (state_ == kVerifyUlpfecPacketNotInNackList) {
>-        test::RtcpPacketParser rtcp_parser;
>-        rtcp_parser.Parse(packet, length);
>-        const std::vector<uint16_t>& nacks = rtcp_parser.nack()->packet_ids();
>-        EXPECT_TRUE(std::find(nacks.begin(), nacks.end(),
>-                              ulpfec_sequence_number_) == nacks.end())
>-            << "Got nack for ULPFEC packet";
>-        if (!nacks.empty() &&
>-            IsNewerSequenceNumber(nacks.back(), ulpfec_sequence_number_)) {
>-          observation_complete_.Set();
>-        }
>-      }
>-      return SEND_PACKET;
>-    }
>-
>-    test::PacketTransport* CreateSendTransport(
>-        test::SingleThreadedTaskQueueForTesting* task_queue,
>-        Call* sender_call) override {
>-      // At low RTT (< kLowRttNackMs) -> NACK only, no FEC.
>-      // Configure some network delay.
>-      const int kNetworkDelayMs = 50;
>-      FakeNetworkPipe::Config config;
>-      config.queue_delay_ms = kNetworkDelayMs;
>-      return new test::PacketTransport(task_queue, sender_call, this,
>-                                       test::PacketTransport::kSender,
>-                                       payload_type_map_, config);
>-    }
>-
>-    // TODO(holmer): Investigate why we don't send FEC packets when the bitrate
>-    // is 10 kbps.
>-    Call::Config GetSenderCallConfig() override {
>-      Call::Config config(event_log_.get());
>-      const int kMinBitrateBps = 30000;
>-      config.bitrate_config.min_bitrate_bps = kMinBitrateBps;
>-      return config;
>-    }
>-
>-    void ModifyVideoConfigs(
>-        VideoSendStream::Config* send_config,
>-        std::vector<VideoReceiveStream::Config>* receive_configs,
>-        VideoEncoderConfig* encoder_config) override {
>-      // Configure hybrid NACK/FEC.
>-      send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
>-      send_config->rtp.ulpfec.red_payload_type = kRedPayloadType;
>-      send_config->rtp.ulpfec.ulpfec_payload_type = kUlpfecPayloadType;
>-      // Set codec to VP8, otherwise NACK/FEC hybrid will be disabled.
>-      send_config->encoder_settings.encoder = encoder_.get();
>-      send_config->encoder_settings.payload_name = "VP8";
>-      send_config->encoder_settings.payload_type = kFakeVideoSendPayloadType;
>-
>-      (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
>-      (*receive_configs)[0].rtp.red_payload_type = kRedPayloadType;
>-      (*receive_configs)[0].rtp.ulpfec_payload_type = kUlpfecPayloadType;
>-
>-      (*receive_configs)[0].decoders.resize(1);
>-      (*receive_configs)[0].decoders[0].payload_type =
>-          send_config->encoder_settings.payload_type;
>-      (*receive_configs)[0].decoders[0].payload_name =
>-          send_config->encoder_settings.payload_name;
>-      (*receive_configs)[0].decoders[0].decoder = decoder_.get();
>-    }
>-
>-    void PerformTest() override {
>-      EXPECT_TRUE(Wait())
>-          << "Timed out while waiting for FEC packets to be received.";
>-    }
>-
>-    enum {
>-      kFirstPacket,
>-      kDropEveryOtherPacketUntilUlpfec,
>-      kDropAllMediaPacketsUntilUlpfec,
>-      kDropOneMediaPacket,
>-      kPassOneMediaPacket,
>-      kVerifyUlpfecPacketNotInNackList,
>-    } state_;
>-
>-    rtc::CriticalSection crit_;
>-    uint16_t ulpfec_sequence_number_ RTC_GUARDED_BY(&crit_);
>-    bool has_last_sequence_number_;
>-    uint16_t last_sequence_number_;
>-    std::unique_ptr<webrtc::VideoEncoder> encoder_;
>-    std::unique_ptr<webrtc::VideoDecoder> decoder_;
>-  } test;
>-
>-  RunBaseTest(&test);
>-}
>-
>-// This test drops second RTP packet with a marker bit set, makes sure it's
>-// retransmitted and renders. Retransmission SSRCs are also checked.
>-void EndToEndTest::DecodesRetransmittedFrame(bool enable_rtx, bool enable_red) {
>-  static const int kDroppedFrameNumber = 10;
>-  class RetransmissionObserver : public test::EndToEndTest,
>-                                 public rtc::VideoSinkInterface<VideoFrame> {
>-   public:
>-    RetransmissionObserver(bool enable_rtx, bool enable_red)
>-        : EndToEndTest(kDefaultTimeoutMs),
>-          payload_type_(GetPayloadType(false, enable_red)),
>-          retransmission_ssrc_(enable_rtx ? kSendRtxSsrcs[0]
>-                                          : kVideoSendSsrcs[0]),
>-          retransmission_payload_type_(GetPayloadType(enable_rtx, enable_red)),
>-          encoder_(VP8Encoder::Create()),
>-          marker_bits_observed_(0),
>-          retransmitted_timestamp_(0) {}
>-
>-   private:
>-    Action OnSendRtp(const uint8_t* packet, size_t length) override {
>-      rtc::CritScope lock(&crit_);
>-      RTPHeader header;
>-      EXPECT_TRUE(parser_->Parse(packet, length, &header));
>-
>-      // Ignore padding-only packets over RTX.
>-      if (header.payloadType != payload_type_) {
>-        EXPECT_EQ(retransmission_ssrc_, header.ssrc);
>-        if (length == header.headerLength + header.paddingLength)
>-          return SEND_PACKET;
>-      }
>-
>-      if (header.timestamp == retransmitted_timestamp_) {
>-        EXPECT_EQ(retransmission_ssrc_, header.ssrc);
>-        EXPECT_EQ(retransmission_payload_type_, header.payloadType);
>-        return SEND_PACKET;
>-      }
>-
>-      // Found the final packet of the frame to inflict loss to, drop this and
>-      // expect a retransmission.
>-      if (header.payloadType == payload_type_ && header.markerBit &&
>-          ++marker_bits_observed_ == kDroppedFrameNumber) {
>-        // This should be the only dropped packet.
>-        EXPECT_EQ(0u, retransmitted_timestamp_);
>-        retransmitted_timestamp_ = header.timestamp;
>-        if (std::find(rendered_timestamps_.begin(), rendered_timestamps_.end(),
>-                      retransmitted_timestamp_) != rendered_timestamps_.end()) {
>-          // Frame was rendered before last packet was scheduled for sending.
>-          // This is extremly rare but possible scenario because prober able to
>-          // resend packet before it was send.
>-          // TODO(danilchap): Remove this corner case when prober would not be
>-          // able to sneak in between packet saved to history for resending and
>-          // pacer notified about existance of that packet for sending.
>-          // See https://bugs.chromium.org/p/webrtc/issues/detail?id=5540 for
>-          // details.
>-          observation_complete_.Set();
>-        }
>-        return DROP_PACKET;
>-      }
>-
>-      return SEND_PACKET;
>-    }
>-
>-    void OnFrame(const VideoFrame& frame) override {
>-      EXPECT_EQ(kVideoRotation_90, frame.rotation());
>-      {
>-        rtc::CritScope lock(&crit_);
>-        if (frame.timestamp() == retransmitted_timestamp_)
>-          observation_complete_.Set();
>-        rendered_timestamps_.push_back(frame.timestamp());
>-      }
>-      orig_renderer_->OnFrame(frame);
>-    }
>-
>-    void ModifyVideoConfigs(
>-        VideoSendStream::Config* send_config,
>-        std::vector<VideoReceiveStream::Config>* receive_configs,
>-        VideoEncoderConfig* encoder_config) override {
>-      send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
>-
>-      // Insert ourselves into the rendering pipeline.
>-      RTC_DCHECK(!orig_renderer_);
>-      orig_renderer_ = (*receive_configs)[0].renderer;
>-      RTC_DCHECK(orig_renderer_);
>-      (*receive_configs)[0].disable_prerenderer_smoothing = true;
>-      (*receive_configs)[0].renderer = this;
>-
>-      (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
>-
>-      if (payload_type_ == kRedPayloadType) {
>-        send_config->rtp.ulpfec.ulpfec_payload_type = kUlpfecPayloadType;
>-        send_config->rtp.ulpfec.red_payload_type = kRedPayloadType;
>-        if (retransmission_ssrc_ == kSendRtxSsrcs[0])
>-          send_config->rtp.ulpfec.red_rtx_payload_type = kRtxRedPayloadType;
>-        (*receive_configs)[0].rtp.ulpfec_payload_type =
>-            send_config->rtp.ulpfec.ulpfec_payload_type;
>-        (*receive_configs)[0].rtp.red_payload_type =
>-            send_config->rtp.ulpfec.red_payload_type;
>-      }
>-
>-      if (retransmission_ssrc_ == kSendRtxSsrcs[0]) {
>-        send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[0]);
>-        send_config->rtp.rtx.payload_type = kSendRtxPayloadType;
>-        (*receive_configs)[0].rtp.rtx_ssrc = kSendRtxSsrcs[0];
>-        (*receive_configs)[0]
>-            .rtp.rtx_associated_payload_types[(payload_type_ == kRedPayloadType)
>-                                                  ? kRtxRedPayloadType
>-                                                  : kSendRtxPayloadType] =
>-            payload_type_;
>-      }
>-      // Configure encoding and decoding with VP8, since generic packetization
>-      // doesn't support FEC with NACK.
>-      RTC_DCHECK_EQ(1, (*receive_configs)[0].decoders.size());
>-      send_config->encoder_settings.encoder = encoder_.get();
>-      send_config->encoder_settings.payload_name = "VP8";
>-      (*receive_configs)[0].decoders[0].payload_name = "VP8";
>-    }
>-
>-    void OnFrameGeneratorCapturerCreated(
>-        test::FrameGeneratorCapturer* frame_generator_capturer) override {
>-      frame_generator_capturer->SetFakeRotation(kVideoRotation_90);
>-    }
>-
>-    void PerformTest() override {
>-      EXPECT_TRUE(Wait())
>-          << "Timed out while waiting for retransmission to render.";
>-    }
>-
>-    int GetPayloadType(bool use_rtx, bool use_fec) {
>-      if (use_fec) {
>-        if (use_rtx)
>-          return kRtxRedPayloadType;
>-        return kRedPayloadType;
>-      }
>-      if (use_rtx)
>-        return kSendRtxPayloadType;
>-      return kFakeVideoSendPayloadType;
>-    }
>-
>-    rtc::CriticalSection crit_;
>-    rtc::VideoSinkInterface<VideoFrame>* orig_renderer_ = nullptr;
>-    const int payload_type_;
>-    const uint32_t retransmission_ssrc_;
>-    const int retransmission_payload_type_;
>-    std::unique_ptr<VideoEncoder> encoder_;
>-    const std::string payload_name_;
>-    int marker_bits_observed_;
>-    uint32_t retransmitted_timestamp_ RTC_GUARDED_BY(&crit_);
>-    std::vector<uint32_t> rendered_timestamps_ RTC_GUARDED_BY(&crit_);
>-  } test(enable_rtx, enable_red);
>-
>-  RunBaseTest(&test);
>-}
>-
>-TEST_P(EndToEndTest, DecodesRetransmittedFrame) {
>-  DecodesRetransmittedFrame(false, false);
>-}
>-
>-TEST_P(EndToEndTest, DecodesRetransmittedFrameOverRtx) {
>-  DecodesRetransmittedFrame(true, false);
>-}
>-
>-TEST_P(EndToEndTest, DecodesRetransmittedFrameByRed) {
>-  DecodesRetransmittedFrame(false, true);
>-}
>-
>-TEST_P(EndToEndTest, DecodesRetransmittedFrameByRedOverRtx) {
>-  DecodesRetransmittedFrame(true, true);
>-}
>-
>-void EndToEndTest::ReceivesPliAndRecovers(int rtp_history_ms) {
>-  static const int kPacketsToDrop = 1;
>-
>-  class PliObserver : public test::EndToEndTest,
>-                      public rtc::VideoSinkInterface<VideoFrame> {
>-   public:
>-    explicit PliObserver(int rtp_history_ms)
>-        : EndToEndTest(kLongTimeoutMs),
>-          rtp_history_ms_(rtp_history_ms),
>-          nack_enabled_(rtp_history_ms > 0),
>-          highest_dropped_timestamp_(0),
>-          frames_to_drop_(0),
>-          received_pli_(false) {}
>-
>-   private:
>-    Action OnSendRtp(const uint8_t* packet, size_t length) override {
>-      rtc::CritScope lock(&crit_);
>-      RTPHeader header;
>-      EXPECT_TRUE(parser_->Parse(packet, length, &header));
>-
>-      // Drop all retransmitted packets to force a PLI.
>-      if (header.timestamp <= highest_dropped_timestamp_)
>-        return DROP_PACKET;
>-
>-      if (frames_to_drop_ > 0) {
>-        highest_dropped_timestamp_ = header.timestamp;
>-        --frames_to_drop_;
>-        return DROP_PACKET;
>-      }
>-
>-      return SEND_PACKET;
>-    }
>-
>-    Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
>-      rtc::CritScope lock(&crit_);
>-      test::RtcpPacketParser parser;
>-      EXPECT_TRUE(parser.Parse(packet, length));
>-      if (!nack_enabled_)
>-        EXPECT_EQ(0, parser.nack()->num_packets());
>-      if (parser.pli()->num_packets() > 0)
>-        received_pli_ = true;
>-      return SEND_PACKET;
>-    }
>-
>-    void OnFrame(const VideoFrame& video_frame) override {
>-      rtc::CritScope lock(&crit_);
>-      if (received_pli_ &&
>-          video_frame.timestamp() > highest_dropped_timestamp_) {
>-        observation_complete_.Set();
>-      }
>-      if (!received_pli_)
>-        frames_to_drop_ = kPacketsToDrop;
>-    }
>-
>-    void ModifyVideoConfigs(
>-        VideoSendStream::Config* send_config,
>-        std::vector<VideoReceiveStream::Config>* receive_configs,
>-        VideoEncoderConfig* encoder_config) override {
>-      send_config->rtp.nack.rtp_history_ms = rtp_history_ms_;
>-      (*receive_configs)[0].rtp.nack.rtp_history_ms = rtp_history_ms_;
>-      (*receive_configs)[0].renderer = this;
>-    }
>-
>-    void PerformTest() override {
>-      EXPECT_TRUE(Wait()) << "Timed out waiting for PLI to be "
>-                             "received and a frame to be "
>-                             "rendered afterwards.";
>-    }
>-
>-    rtc::CriticalSection crit_;
>-    int rtp_history_ms_;
>-    bool nack_enabled_;
>-    uint32_t highest_dropped_timestamp_ RTC_GUARDED_BY(&crit_);
>-    int frames_to_drop_ RTC_GUARDED_BY(&crit_);
>-    bool received_pli_ RTC_GUARDED_BY(&crit_);
>-  } test(rtp_history_ms);
>-
>-  RunBaseTest(&test);
>-}
>-
>-TEST_P(EndToEndTest, ReceivesPliAndRecoversWithNack) {
>-  ReceivesPliAndRecovers(1000);
>-}
>-
>-TEST_P(EndToEndTest, ReceivesPliAndRecoversWithoutNack) {
>-  ReceivesPliAndRecovers(0);
>-}
>-
>-TEST_P(EndToEndTest, UnknownRtpPacketGivesUnknownSsrcReturnCode) {
>-  class PacketInputObserver : public PacketReceiver {
>-   public:
>-    explicit PacketInputObserver(PacketReceiver* receiver)
>-        : receiver_(receiver), delivered_packet_(false, false) {}
>-
>-    bool Wait() { return delivered_packet_.Wait(kDefaultTimeoutMs); }
>-
>-   private:
>-    DeliveryStatus DeliverPacket(MediaType media_type,
>-                                 rtc::CopyOnWriteBuffer packet,
>-                                 const PacketTime& packet_time) override {
>-      if (RtpHeaderParser::IsRtcp(packet.cdata(), packet.size())) {
>-        return receiver_->DeliverPacket(media_type, std::move(packet),
>-                                        packet_time);
>-      } else {
>-        DeliveryStatus delivery_status = receiver_->DeliverPacket(
>-            media_type, std::move(packet), packet_time);
>-        EXPECT_EQ(DELIVERY_UNKNOWN_SSRC, delivery_status);
>-        delivered_packet_.Set();
>-        return delivery_status;
>-      }
>-    }
>-
>-    PacketReceiver* receiver_;
>-    rtc::Event delivered_packet_;
>-  };
>-
>-  std::unique_ptr<test::DirectTransport> send_transport;
>-  std::unique_ptr<test::DirectTransport> receive_transport;
>-  std::unique_ptr<PacketInputObserver> input_observer;
>-
>-  task_queue_.SendTask([this, &send_transport, &receive_transport,
>-                        &input_observer]() {
>-    CreateCalls(Call::Config(event_log_.get()), Call::Config(event_log_.get()));
>-
>-    send_transport = rtc::MakeUnique<test::DirectTransport>(
>-        &task_queue_, sender_call_.get(), payload_type_map_);
>-    receive_transport = rtc::MakeUnique<test::DirectTransport>(
>-        &task_queue_, receiver_call_.get(), payload_type_map_);
>-    input_observer =
>-        rtc::MakeUnique<PacketInputObserver>(receiver_call_->Receiver());
>-    send_transport->SetReceiver(input_observer.get());
>-    receive_transport->SetReceiver(sender_call_->Receiver());
>-
>-    CreateSendConfig(1, 0, 0, send_transport.get());
>-    CreateMatchingReceiveConfigs(receive_transport.get());
>-
>-    CreateVideoStreams();
>-    CreateFrameGeneratorCapturer(kDefaultFramerate, kDefaultWidth,
>-                                 kDefaultHeight);
>-    Start();
>-
>-    receiver_call_->DestroyVideoReceiveStream(video_receive_streams_[0]);
>-    video_receive_streams_.clear();
>-  });
>-
>-  // Wait() waits for a received packet.
>-  EXPECT_TRUE(input_observer->Wait());
>-
>-  task_queue_.SendTask([this, &send_transport, &receive_transport]() {
>-    Stop();
>-    DestroyStreams();
>-    send_transport.reset();
>-    receive_transport.reset();
>-    DestroyCalls();
>-  });
>-}
>-
>-void EndToEndTest::RespectsRtcpMode(RtcpMode rtcp_mode) {
>-  static const int kNumCompoundRtcpPacketsToObserve = 10;
>-  class RtcpModeObserver : public test::EndToEndTest {
>-   public:
>-    explicit RtcpModeObserver(RtcpMode rtcp_mode)
>-        : EndToEndTest(kDefaultTimeoutMs),
>-          rtcp_mode_(rtcp_mode),
>-          sent_rtp_(0),
>-          sent_rtcp_(0) {}
>-
>-   private:
>-    Action OnSendRtp(const uint8_t* packet, size_t length) override {
>-      rtc::CritScope lock(&crit_);
>-      if (++sent_rtp_ % 3 == 0)
>-        return DROP_PACKET;
>-
>-      return SEND_PACKET;
>-    }
>-
>-    Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
>-      rtc::CritScope lock(&crit_);
>-      ++sent_rtcp_;
>-      test::RtcpPacketParser parser;
>-      EXPECT_TRUE(parser.Parse(packet, length));
>-
>-      EXPECT_EQ(0, parser.sender_report()->num_packets());
>-
>-      switch (rtcp_mode_) {
>-        case RtcpMode::kCompound:
>-          // TODO(holmer): We shouldn't send transport feedback alone if
>-          // compound RTCP is negotiated.
>-          if (parser.receiver_report()->num_packets() == 0 &&
>-              parser.transport_feedback()->num_packets() == 0) {
>-            ADD_FAILURE() << "Received RTCP packet without receiver report for "
>-                             "RtcpMode::kCompound.";
>-            observation_complete_.Set();
>-          }
>-
>-          if (sent_rtcp_ >= kNumCompoundRtcpPacketsToObserve)
>-            observation_complete_.Set();
>-
>-          break;
>-        case RtcpMode::kReducedSize:
>-          if (parser.receiver_report()->num_packets() == 0)
>-            observation_complete_.Set();
>-          break;
>-        case RtcpMode::kOff:
>-          RTC_NOTREACHED();
>-          break;
>-      }
>-
>-      return SEND_PACKET;
>-    }
>-
>-    void ModifyVideoConfigs(
>-        VideoSendStream::Config* send_config,
>-        std::vector<VideoReceiveStream::Config>* receive_configs,
>-        VideoEncoderConfig* encoder_config) override {
>-      send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
>-      (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
>-      (*receive_configs)[0].rtp.rtcp_mode = rtcp_mode_;
>-    }
>-
>-    void PerformTest() override {
>-      EXPECT_TRUE(Wait())
>-          << (rtcp_mode_ == RtcpMode::kCompound
>-                  ? "Timed out before observing enough compound packets."
>-                  : "Timed out before receiving a non-compound RTCP packet.");
>-    }
>-
>-    RtcpMode rtcp_mode_;
>-    rtc::CriticalSection crit_;
>-    // Must be protected since RTCP can be sent by both the process thread
>-    // and the pacer thread.
>-    int sent_rtp_ RTC_GUARDED_BY(&crit_);
>-    int sent_rtcp_ RTC_GUARDED_BY(&crit_);
>-  } test(rtcp_mode);
>-
>-  RunBaseTest(&test);
>-}
>-
>-TEST_P(EndToEndTest, UsesRtcpCompoundMode) {
>-  RespectsRtcpMode(RtcpMode::kCompound);
>-}
>-
>-TEST_P(EndToEndTest, UsesRtcpReducedSizeMode) {
>-  RespectsRtcpMode(RtcpMode::kReducedSize);
>-}
>-
>-// Test sets up a Call multiple senders with different resolutions and SSRCs.
>-// Another is set up to receive all three of these with different renderers.
>-class MultiStreamTest {
>- public:
>-  static constexpr size_t kNumStreams = 3;
>-  const uint8_t kVideoPayloadType = 124;
>-  const std::map<uint8_t, MediaType> payload_type_map_ = {
>-      {kVideoPayloadType, MediaType::VIDEO}};
>-
>-  struct CodecSettings {
>-    uint32_t ssrc;
>-    int width;
>-    int height;
>-  } codec_settings[kNumStreams];
>-
>-  explicit MultiStreamTest(test::SingleThreadedTaskQueueForTesting* task_queue)
>-      : task_queue_(task_queue) {
>-    // TODO(sprang): Cleanup when msvc supports explicit initializers for array.
>-    codec_settings[0] = {1, 640, 480};
>-    codec_settings[1] = {2, 320, 240};
>-    codec_settings[2] = {3, 240, 160};
>-  }
>-
>-  virtual ~MultiStreamTest() {}
>-
>-  void RunTest() {
>-    webrtc::RtcEventLogNullImpl event_log;
>-    Call::Config config(&event_log);
>-    std::unique_ptr<Call> sender_call;
>-    std::unique_ptr<Call> receiver_call;
>-    std::unique_ptr<test::DirectTransport> sender_transport;
>-    std::unique_ptr<test::DirectTransport> receiver_transport;
>-
>-    VideoSendStream* send_streams[kNumStreams];
>-    VideoReceiveStream* receive_streams[kNumStreams];
>-    test::FrameGeneratorCapturer* frame_generators[kNumStreams];
>-    std::vector<std::unique_ptr<VideoDecoder>> allocated_decoders;
>-    std::unique_ptr<VideoEncoder> encoders[kNumStreams];
>-
>-    task_queue_->SendTask([&]() {
>-      sender_call = rtc::WrapUnique(Call::Create(config));
>-      receiver_call = rtc::WrapUnique(Call::Create(config));
>-      sender_transport =
>-          rtc::WrapUnique(CreateSendTransport(task_queue_, sender_call.get()));
>-      receiver_transport = rtc::WrapUnique(
>-          CreateReceiveTransport(task_queue_, receiver_call.get()));
>-
>-      sender_transport->SetReceiver(receiver_call->Receiver());
>-      receiver_transport->SetReceiver(sender_call->Receiver());
>-
>-      for (size_t i = 0; i < kNumStreams; ++i)
>-        encoders[i] = VP8Encoder::Create();
>-
>-      for (size_t i = 0; i < kNumStreams; ++i) {
>-        uint32_t ssrc = codec_settings[i].ssrc;
>-        int width = codec_settings[i].width;
>-        int height = codec_settings[i].height;
>-
>-        VideoSendStream::Config send_config(sender_transport.get());
>-        send_config.rtp.ssrcs.push_back(ssrc);
>-        send_config.encoder_settings.encoder = encoders[i].get();
>-        send_config.encoder_settings.payload_name = "VP8";
>-        send_config.encoder_settings.payload_type = kVideoPayloadType;
>-        VideoEncoderConfig encoder_config;
>-        test::FillEncoderConfiguration(1, &encoder_config);
>-        encoder_config.max_bitrate_bps = 100000;
>-
>-        UpdateSendConfig(i, &send_config, &encoder_config,
>-                         &frame_generators[i]);
>-
>-        send_streams[i] = sender_call->CreateVideoSendStream(
>-            send_config.Copy(), encoder_config.Copy());
>-        send_streams[i]->Start();
>-
>-        VideoReceiveStream::Config receive_config(receiver_transport.get());
>-        receive_config.rtp.remote_ssrc = ssrc;
>-        receive_config.rtp.local_ssrc = test::CallTest::kReceiverLocalVideoSsrc;
>-        VideoReceiveStream::Decoder decoder =
>-            test::CreateMatchingDecoder(send_config.encoder_settings);
>-        allocated_decoders.push_back(
>-            std::unique_ptr<VideoDecoder>(decoder.decoder));
>-        receive_config.decoders.push_back(decoder);
>-
>-        UpdateReceiveConfig(i, &receive_config);
>-
>-        receive_streams[i] =
>-            receiver_call->CreateVideoReceiveStream(std::move(receive_config));
>-        receive_streams[i]->Start();
>-
>-        frame_generators[i] = test::FrameGeneratorCapturer::Create(
>-            width, height, 30, Clock::GetRealTimeClock());
>-        send_streams[i]->SetSource(
>-            frame_generators[i],
>-            VideoSendStream::DegradationPreference::kMaintainFramerate);
>-        frame_generators[i]->Start();
>-      }
>-    });
>-
>-    Wait();
>-
>-    task_queue_->SendTask([&]() {
>-      for (size_t i = 0; i < kNumStreams; ++i) {
>-        frame_generators[i]->Stop();
>-        sender_call->DestroyVideoSendStream(send_streams[i]);
>-        receiver_call->DestroyVideoReceiveStream(receive_streams[i]);
>-        delete frame_generators[i];
>-      }
>-
>-      sender_transport.reset();
>-      receiver_transport.reset();
>-
>-      sender_call.reset();
>-      receiver_call.reset();
>-    });
>-  }
>-
>- protected:
>-  virtual void Wait() = 0;
>-  // Note: frame_generator is a point-to-pointer, since the actual instance
>-  // hasn't been created at the time of this call. Only when packets/frames
>-  // start flowing should this be dereferenced.
>-  virtual void UpdateSendConfig(
>-      size_t stream_index,
>-      VideoSendStream::Config* send_config,
>-      VideoEncoderConfig* encoder_config,
>-      test::FrameGeneratorCapturer** frame_generator) {}
>-  virtual void UpdateReceiveConfig(size_t stream_index,
>-                                   VideoReceiveStream::Config* receive_config) {
>-  }
>-  virtual test::DirectTransport* CreateSendTransport(
>-      test::SingleThreadedTaskQueueForTesting* task_queue,
>-      Call* sender_call) {
>-    return new test::DirectTransport(task_queue, sender_call,
>-                                     payload_type_map_);
>-  }
>-  virtual test::DirectTransport* CreateReceiveTransport(
>-      test::SingleThreadedTaskQueueForTesting* task_queue,
>-      Call* receiver_call) {
>-    return new test::DirectTransport(task_queue, receiver_call,
>-                                     payload_type_map_);
>-  }
>-
>-  test::SingleThreadedTaskQueueForTesting* const task_queue_;
>-};
>-
>-// Each renderer verifies that it receives the expected resolution, and as soon
>-// as every renderer has received a frame, the test finishes.
>-TEST_P(EndToEndTest, SendsAndReceivesMultipleStreams) {
>-  class VideoOutputObserver : public rtc::VideoSinkInterface<VideoFrame> {
>-   public:
>-    VideoOutputObserver(const MultiStreamTest::CodecSettings& settings,
>-                        uint32_t ssrc,
>-                        test::FrameGeneratorCapturer** frame_generator)
>-        : settings_(settings),
>-          ssrc_(ssrc),
>-          frame_generator_(frame_generator),
>-          done_(false, false) {}
>-
>-    void OnFrame(const VideoFrame& video_frame) override {
>-      EXPECT_EQ(settings_.width, video_frame.width());
>-      EXPECT_EQ(settings_.height, video_frame.height());
>-      (*frame_generator_)->Stop();
>-      done_.Set();
>-    }
>-
>-    uint32_t Ssrc() { return ssrc_; }
>-
>-    bool Wait() { return done_.Wait(kDefaultTimeoutMs); }
>-
>-   private:
>-    const MultiStreamTest::CodecSettings& settings_;
>-    const uint32_t ssrc_;
>-    test::FrameGeneratorCapturer** const frame_generator_;
>-    rtc::Event done_;
>-  };
>-
>-  class Tester : public MultiStreamTest {
>-   public:
>-    explicit Tester(test::SingleThreadedTaskQueueForTesting* task_queue)
>-        : MultiStreamTest(task_queue) {}
>-    virtual ~Tester() {}
>-
>-   protected:
>-    void Wait() override {
>-      for (const auto& observer : observers_) {
>-        EXPECT_TRUE(observer->Wait()) << "Time out waiting for from on ssrc "
>-                                      << observer->Ssrc();
>-      }
>-    }
>-
>-    void UpdateSendConfig(
>-        size_t stream_index,
>-        VideoSendStream::Config* send_config,
>-        VideoEncoderConfig* encoder_config,
>-        test::FrameGeneratorCapturer** frame_generator) override {
>-      observers_[stream_index].reset(new VideoOutputObserver(
>-          codec_settings[stream_index], send_config->rtp.ssrcs.front(),
>-          frame_generator));
>-    }
>-
>-    void UpdateReceiveConfig(
>-        size_t stream_index,
>-        VideoReceiveStream::Config* receive_config) override {
>-      receive_config->renderer = observers_[stream_index].get();
>-    }
>-
>-   private:
>-    std::unique_ptr<VideoOutputObserver> observers_[kNumStreams];
>-  } tester(&task_queue_);
>-
>-  tester.RunTest();
>-}
>-
>-TEST_P(EndToEndTest, AssignsTransportSequenceNumbers) {
>-  static const int kExtensionId = 5;
>-
>-  class RtpExtensionHeaderObserver : public test::DirectTransport {
>-   public:
>-    RtpExtensionHeaderObserver(
>-        test::SingleThreadedTaskQueueForTesting* task_queue,
>-        Call* sender_call,
>-        const uint32_t& first_media_ssrc,
>-        const std::map<uint32_t, uint32_t>& ssrc_map,
>-        const std::map<uint8_t, MediaType>& payload_type_map)
>-        : DirectTransport(task_queue, sender_call, payload_type_map),
>-          done_(false, false),
>-          parser_(RtpHeaderParser::Create()),
>-          first_media_ssrc_(first_media_ssrc),
>-          rtx_to_media_ssrcs_(ssrc_map),
>-          padding_observed_(false),
>-          rtx_padding_observed_(false),
>-          retransmit_observed_(false),
>-          started_(false) {
>-      parser_->RegisterRtpHeaderExtension(kRtpExtensionTransportSequenceNumber,
>-                                          kExtensionId);
>-    }
>-    virtual ~RtpExtensionHeaderObserver() {}
>-
>-    bool SendRtp(const uint8_t* data,
>-                 size_t length,
>-                 const PacketOptions& options) override {
>-      {
>-        rtc::CritScope cs(&lock_);
>-
>-        if (IsDone())
>-          return false;
>-
>-        if (started_) {
>-          RTPHeader header;
>-          EXPECT_TRUE(parser_->Parse(data, length, &header));
>-          bool drop_packet = false;
>-
>-          EXPECT_TRUE(header.extension.hasTransportSequenceNumber);
>-          EXPECT_EQ(options.packet_id,
>-                    header.extension.transportSequenceNumber);
>-          if (!streams_observed_.empty()) {
>-            // Unwrap packet id and verify uniqueness.
>-            int64_t packet_id = unwrapper_.Unwrap(options.packet_id);
>-            EXPECT_TRUE(received_packed_ids_.insert(packet_id).second);
>-          }
>-
>-          // Drop (up to) every 17th packet, so we get retransmits.
>-          // Only drop media, and not on the first stream (otherwise it will be
>-          // hard to distinguish from padding, which is always sent on the first
>-          // stream).
>-          if (header.payloadType != kSendRtxPayloadType &&
>-              header.ssrc != first_media_ssrc_ &&
>-              header.extension.transportSequenceNumber % 17 == 0) {
>-            dropped_seq_[header.ssrc].insert(header.sequenceNumber);
>-            drop_packet = true;
>-          }
>-
>-          if (header.payloadType == kSendRtxPayloadType) {
>-            uint16_t original_sequence_number =
>-                ByteReader<uint16_t>::ReadBigEndian(&data[header.headerLength]);
>-            uint32_t original_ssrc =
>-                rtx_to_media_ssrcs_.find(header.ssrc)->second;
>-            std::set<uint16_t>* seq_no_map = &dropped_seq_[original_ssrc];
>-            auto it = seq_no_map->find(original_sequence_number);
>-            if (it != seq_no_map->end()) {
>-              retransmit_observed_ = true;
>-              seq_no_map->erase(it);
>-            } else {
>-              rtx_padding_observed_ = true;
>-            }
>-          } else {
>-            streams_observed_.insert(header.ssrc);
>-          }
>-
>-          if (IsDone())
>-            done_.Set();
>-
>-          if (drop_packet)
>-            return true;
>-        }
>-      }
>-
>-      return test::DirectTransport::SendRtp(data, length, options);
>-    }
>-
>-    bool IsDone() {
>-      bool observed_types_ok =
>-          streams_observed_.size() == MultiStreamTest::kNumStreams &&
>-          retransmit_observed_ && rtx_padding_observed_;
>-      if (!observed_types_ok)
>-        return false;
>-      // We should not have any gaps in the sequence number range.
>-      size_t seqno_range =
>-          *received_packed_ids_.rbegin() - *received_packed_ids_.begin() + 1;
>-      return seqno_range == received_packed_ids_.size();
>-    }
>-
>-    bool Wait() {
>-      {
>-        // Can't be sure until this point that rtx_to_media_ssrcs_ etc have
>-        // been initialized and are OK to read.
>-        rtc::CritScope cs(&lock_);
>-        started_ = true;
>-      }
>-      return done_.Wait(kDefaultTimeoutMs);
>-    }
>-
>-    rtc::CriticalSection lock_;
>-    rtc::Event done_;
>-    std::unique_ptr<RtpHeaderParser> parser_;
>-    SequenceNumberUnwrapper unwrapper_;
>-    std::set<int64_t> received_packed_ids_;
>-    std::set<uint32_t> streams_observed_;
>-    std::map<uint32_t, std::set<uint16_t>> dropped_seq_;
>-    const uint32_t& first_media_ssrc_;
>-    const std::map<uint32_t, uint32_t>& rtx_to_media_ssrcs_;
>-    bool padding_observed_;
>-    bool rtx_padding_observed_;
>-    bool retransmit_observed_;
>-    bool started_;
>-  };
>-
>-  class TransportSequenceNumberTester : public MultiStreamTest {
>-   public:
>-    explicit TransportSequenceNumberTester(
>-        test::SingleThreadedTaskQueueForTesting* task_queue)
>-        : MultiStreamTest(task_queue),
>-          first_media_ssrc_(0),
>-          observer_(nullptr) {}
>-    virtual ~TransportSequenceNumberTester() {}
>-
>-   protected:
>-    void Wait() override {
>-      RTC_DCHECK(observer_);
>-      EXPECT_TRUE(observer_->Wait());
>-    }
>-
>-    void UpdateSendConfig(
>-        size_t stream_index,
>-        VideoSendStream::Config* send_config,
>-        VideoEncoderConfig* encoder_config,
>-        test::FrameGeneratorCapturer** frame_generator) override {
>-      send_config->rtp.extensions.clear();
>-      send_config->rtp.extensions.push_back(RtpExtension(
>-          RtpExtension::kTransportSequenceNumberUri, kExtensionId));
>-
>-      // Force some padding to be sent. Note that since we do send media
>-      // packets we can not guarantee that a padding only packet is sent.
>-      // Instead, padding will most likely be send as an RTX packet.
>-      const int kPaddingBitrateBps = 50000;
>-      encoder_config->max_bitrate_bps = 200000;
>-      encoder_config->min_transmit_bitrate_bps =
>-          encoder_config->max_bitrate_bps + kPaddingBitrateBps;
>-
>-      // Configure RTX for redundant payload padding.
>-      send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
>-      send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[stream_index]);
>-      send_config->rtp.rtx.payload_type = kSendRtxPayloadType;
>-      rtx_to_media_ssrcs_[kSendRtxSsrcs[stream_index]] =
>-          send_config->rtp.ssrcs[0];
>-
>-      if (stream_index == 0)
>-        first_media_ssrc_ = send_config->rtp.ssrcs[0];
>-    }
>-
>-    void UpdateReceiveConfig(
>-        size_t stream_index,
>-        VideoReceiveStream::Config* receive_config) override {
>-      receive_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
>-      receive_config->rtp.extensions.clear();
>-      receive_config->rtp.extensions.push_back(RtpExtension(
>-          RtpExtension::kTransportSequenceNumberUri, kExtensionId));
>-      receive_config->renderer = &fake_renderer_;
>-    }
>-
>-    test::DirectTransport* CreateSendTransport(
>-        test::SingleThreadedTaskQueueForTesting* task_queue,
>-        Call* sender_call) override {
>-      std::map<uint8_t, MediaType> payload_type_map =
>-          MultiStreamTest::payload_type_map_;
>-      RTC_DCHECK(payload_type_map.find(kSendRtxPayloadType) ==
>-                 payload_type_map.end());
>-      payload_type_map[kSendRtxPayloadType] = MediaType::VIDEO;
>-      observer_ = new RtpExtensionHeaderObserver(
>-          task_queue, sender_call, first_media_ssrc_, rtx_to_media_ssrcs_,
>-          payload_type_map);
>-      return observer_;
>-    }
>-
>-   private:
>-    test::FakeVideoRenderer fake_renderer_;
>-    uint32_t first_media_ssrc_;
>-    std::map<uint32_t, uint32_t> rtx_to_media_ssrcs_;
>-    RtpExtensionHeaderObserver* observer_;
>-  } tester(&task_queue_);
>-
>-  tester.RunTest();
>-}
>-
>-class TransportFeedbackTester : public test::EndToEndTest {
>- public:
>-  TransportFeedbackTester(bool feedback_enabled,
>-                          size_t num_video_streams,
>-                          size_t num_audio_streams)
>-      : EndToEndTest(::webrtc::EndToEndTest::kDefaultTimeoutMs),
>-        feedback_enabled_(feedback_enabled),
>-        num_video_streams_(num_video_streams),
>-        num_audio_streams_(num_audio_streams),
>-        receiver_call_(nullptr) {
>-    // Only one stream of each supported for now.
>-    EXPECT_LE(num_video_streams, 1u);
>-    EXPECT_LE(num_audio_streams, 1u);
>-  }
>-
>- protected:
>-  Action OnSendRtcp(const uint8_t* data, size_t length) override {
>-    EXPECT_FALSE(HasTransportFeedback(data, length));
>-    return SEND_PACKET;
>-  }
>-
>-  Action OnReceiveRtcp(const uint8_t* data, size_t length) override {
>-    if (HasTransportFeedback(data, length))
>-      observation_complete_.Set();
>-    return SEND_PACKET;
>-  }
>-
>-  bool HasTransportFeedback(const uint8_t* data, size_t length) const {
>-    test::RtcpPacketParser parser;
>-    EXPECT_TRUE(parser.Parse(data, length));
>-    return parser.transport_feedback()->num_packets() > 0;
>-  }
>-
>-  void PerformTest() override {
>-    const int64_t kDisabledFeedbackTimeoutMs = 5000;
>-    EXPECT_EQ(feedback_enabled_,
>-              observation_complete_.Wait(feedback_enabled_
>-                                             ? test::CallTest::kDefaultTimeoutMs
>-                                             : kDisabledFeedbackTimeoutMs));
>-  }
>-
>-  void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
>-    receiver_call_ = receiver_call;
>-  }
>-
>-  size_t GetNumVideoStreams() const override { return num_video_streams_; }
>-  size_t GetNumAudioStreams() const override { return num_audio_streams_; }
>-
>-  void ModifyVideoConfigs(
>-      VideoSendStream::Config* send_config,
>-      std::vector<VideoReceiveStream::Config>* receive_configs,
>-      VideoEncoderConfig* encoder_config) override {
>-    (*receive_configs)[0].rtp.transport_cc = feedback_enabled_;
>-  }
>-
>-  void ModifyAudioConfigs(
>-      AudioSendStream::Config* send_config,
>-      std::vector<AudioReceiveStream::Config>* receive_configs) override {
>-    send_config->rtp.extensions.clear();
>-    send_config->rtp.extensions.push_back(
>-        RtpExtension(RtpExtension::kTransportSequenceNumberUri, kExtensionId));
>-    (*receive_configs)[0].rtp.extensions.clear();
>-    (*receive_configs)[0].rtp.extensions = send_config->rtp.extensions;
>-    (*receive_configs)[0].rtp.transport_cc = feedback_enabled_;
>-  }
>-
>- private:
>-  static const int kExtensionId = 5;
>-  const bool feedback_enabled_;
>-  const size_t num_video_streams_;
>-  const size_t num_audio_streams_;
>-  Call* receiver_call_;
>-};
>-
>-TEST_P(EndToEndTest, VideoReceivesTransportFeedback) {
>-  TransportFeedbackTester test(true, 1, 0);
>-  RunBaseTest(&test);
>-}
>-
>-TEST_P(EndToEndTest, VideoTransportFeedbackNotConfigured) {
>-  TransportFeedbackTester test(false, 1, 0);
>-  RunBaseTest(&test);
>-}
>-
>-TEST_P(EndToEndTest, AudioReceivesTransportFeedback) {
>-  TransportFeedbackTester test(true, 0, 1);
>-  RunBaseTest(&test);
>-}
>-
>-TEST_P(EndToEndTest, AudioTransportFeedbackNotConfigured) {
>-  TransportFeedbackTester test(false, 0, 1);
>-  RunBaseTest(&test);
>-}
>-
>-TEST_P(EndToEndTest, AudioVideoReceivesTransportFeedback) {
>-  TransportFeedbackTester test(true, 1, 1);
>-  RunBaseTest(&test);
>-}
>-
>-TEST_P(EndToEndTest, StopsSendingMediaWithoutFeedback) {
>-  test::ScopedFieldTrials override_field_trials(
>-      "WebRTC-CwndExperiment/Enabled-250/");
>-
>-  class TransportFeedbackTester : public test::EndToEndTest {
>-   public:
>-    TransportFeedbackTester(size_t num_video_streams, size_t num_audio_streams)
>-        : EndToEndTest(::webrtc::EndToEndTest::kDefaultTimeoutMs),
>-          num_video_streams_(num_video_streams),
>-          num_audio_streams_(num_audio_streams),
>-          media_sent_(0),
>-          padding_sent_(0) {
>-      // Only one stream of each supported for now.
>-      EXPECT_LE(num_video_streams, 1u);
>-      EXPECT_LE(num_audio_streams, 1u);
>-    }
>-
>-   protected:
>-    Action OnSendRtp(const uint8_t* packet, size_t length) override {
>-      RTPHeader header;
>-      EXPECT_TRUE(parser_->Parse(packet, length, &header));
>-      const bool only_padding =
>-          header.headerLength + header.paddingLength == length;
>-      rtc::CritScope lock(&crit_);
>-      if (only_padding) {
>-        ++padding_sent_;
>-      } else {
>-        ++media_sent_;
>-        EXPECT_LT(media_sent_, 40) << "Media sent without feedback.";
>-      }
>-
>-      return SEND_PACKET;
>-    }
>-
>-    Action OnReceiveRtcp(const uint8_t* data, size_t length) override {
>-      rtc::CritScope lock(&crit_);
>-      if (media_sent_ > 20 && HasTransportFeedback(data, length)) {
>-        return DROP_PACKET;
>-      }
>-      return SEND_PACKET;
>-    }
>-
>-    bool HasTransportFeedback(const uint8_t* data, size_t length) const {
>-      test::RtcpPacketParser parser;
>-      EXPECT_TRUE(parser.Parse(data, length));
>-      return parser.transport_feedback()->num_packets() > 0;
>-    }
>-
>-    Call::Config GetSenderCallConfig() override {
>-      Call::Config config = EndToEndTest::GetSenderCallConfig();
>-      config.bitrate_config.max_bitrate_bps = 300000;
>-      return config;
>-    }
>-
>-    void PerformTest() override {
>-      const int64_t kDisabledFeedbackTimeoutMs = 10000;
>-      observation_complete_.Wait(kDisabledFeedbackTimeoutMs);
>-      rtc::CritScope lock(&crit_);
>-      EXPECT_GT(padding_sent_, 0);
>-    }
>-
>-    size_t GetNumVideoStreams() const override { return num_video_streams_; }
>-    size_t GetNumAudioStreams() const override { return num_audio_streams_; }
>-
>-   private:
>-    const size_t num_video_streams_;
>-    const size_t num_audio_streams_;
>-    rtc::CriticalSection crit_;
>-    int media_sent_ RTC_GUARDED_BY(crit_);
>-    int padding_sent_ RTC_GUARDED_BY(crit_);
>-  } test(1, 0);
>-  RunBaseTest(&test);
>-}
>-
>-TEST_P(EndToEndTest, ObserversEncodedFrames) {
>-  class EncodedFrameTestObserver : public EncodedFrameObserver {
>-   public:
>-    EncodedFrameTestObserver()
>-        : length_(0), frame_type_(kEmptyFrame), called_(false, false) {}
>-    virtual ~EncodedFrameTestObserver() {}
>-
>-    virtual void EncodedFrameCallback(const EncodedFrame& encoded_frame) {
>-      frame_type_ = encoded_frame.frame_type_;
>-      length_ = encoded_frame.length_;
>-      buffer_.reset(new uint8_t[length_]);
>-      memcpy(buffer_.get(), encoded_frame.data_, length_);
>-      called_.Set();
>-    }
>-
>-    bool Wait() { return called_.Wait(kDefaultTimeoutMs); }
>-
>-    void ExpectEqualFrames(const EncodedFrameTestObserver& observer) {
>-      ASSERT_EQ(length_, observer.length_)
>-          << "Observed frames are of different lengths.";
>-      EXPECT_EQ(frame_type_, observer.frame_type_)
>-          << "Observed frames have different frame types.";
>-      EXPECT_EQ(0, memcmp(buffer_.get(), observer.buffer_.get(), length_))
>-          << "Observed encoded frames have different content.";
>-    }
>-
>-   private:
>-    std::unique_ptr<uint8_t[]> buffer_;
>-    size_t length_;
>-    FrameType frame_type_;
>-    rtc::Event called_;
>-  };
>-
>-  EncodedFrameTestObserver post_encode_observer;
>-  EncodedFrameTestObserver pre_decode_observer;
>-  test::FrameForwarder forwarder;
>-  std::unique_ptr<test::FrameGenerator> frame_generator;
>-
>-  std::unique_ptr<test::DirectTransport> sender_transport;
>-  std::unique_ptr<test::DirectTransport> receiver_transport;
>-
>-  task_queue_.SendTask([&]() {
>-    CreateCalls(Call::Config(event_log_.get()), Call::Config(event_log_.get()));
>-
>-    sender_transport = rtc::MakeUnique<test::DirectTransport>(
>-        &task_queue_, sender_call_.get(), payload_type_map_);
>-    receiver_transport = rtc::MakeUnique<test::DirectTransport>(
>-        &task_queue_, receiver_call_.get(), payload_type_map_);
>-    sender_transport->SetReceiver(receiver_call_->Receiver());
>-    receiver_transport->SetReceiver(sender_call_->Receiver());
>-
>-    CreateSendConfig(1, 0, 0, sender_transport.get());
>-    CreateMatchingReceiveConfigs(receiver_transport.get());
>-    video_send_config_.post_encode_callback = &post_encode_observer;
>-    video_receive_configs_[0].pre_decode_callback = &pre_decode_observer;
>-
>-    CreateVideoStreams();
>-    Start();
>-
>-    frame_generator = test::FrameGenerator::CreateSquareGenerator(
>-        kDefaultWidth, kDefaultHeight);
>-    video_send_stream_->SetSource(
>-        &forwarder, VideoSendStream::DegradationPreference::kMaintainFramerate);
>-    forwarder.IncomingCapturedFrame(*frame_generator->NextFrame());
>-  });
>-
>-  EXPECT_TRUE(post_encode_observer.Wait())
>-      << "Timed out while waiting for send-side encoded-frame callback.";
>-
>-  EXPECT_TRUE(pre_decode_observer.Wait())
>-      << "Timed out while waiting for pre-decode encoded-frame callback.";
>-
>-  post_encode_observer.ExpectEqualFrames(pre_decode_observer);
>-
>-  task_queue_.SendTask([this, &sender_transport, &receiver_transport]() {
>-    Stop();
>-    DestroyStreams();
>-    sender_transport.reset();
>-    receiver_transport.reset();
>-    DestroyCalls();
>-  });
>-}
>-
>-TEST_P(EndToEndTest, ReceiveStreamSendsRemb) {
>-  class RembObserver : public test::EndToEndTest {
>-   public:
>-    RembObserver() : EndToEndTest(kDefaultTimeoutMs) {}
>-
>-    void ModifyVideoConfigs(
>-        VideoSendStream::Config* send_config,
>-        std::vector<VideoReceiveStream::Config>* receive_configs,
>-        VideoEncoderConfig* encoder_config) override {
>-      send_config->rtp.extensions.clear();
>-      send_config->rtp.extensions.push_back(RtpExtension(
>-          RtpExtension::kAbsSendTimeUri, test::kAbsSendTimeExtensionId));
>-      (*receive_configs)[0].rtp.remb = true;
>-      (*receive_configs)[0].rtp.transport_cc = false;
>-    }
>-
>-    Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
>-      test::RtcpPacketParser parser;
>-      EXPECT_TRUE(parser.Parse(packet, length));
>-
>-      if (parser.remb()->num_packets() > 0) {
>-        EXPECT_EQ(kReceiverLocalVideoSsrc, parser.remb()->sender_ssrc());
>-        EXPECT_LT(0U, parser.remb()->bitrate_bps());
>-        EXPECT_EQ(1U, parser.remb()->ssrcs().size());
>-        EXPECT_EQ(kVideoSendSsrcs[0], parser.remb()->ssrcs()[0]);
>-        observation_complete_.Set();
>-      }
>-
>-      return SEND_PACKET;
>-    }
>-    void PerformTest() override {
>-      EXPECT_TRUE(Wait()) << "Timed out while waiting for a "
>-                             "receiver RTCP REMB packet to be "
>-                             "sent.";
>-    }
>-  } test;
>-
>-  RunBaseTest(&test);
>-}
>-
>-class BandwidthStatsTest : public test::EndToEndTest {
>- public:
>-  explicit BandwidthStatsTest(bool send_side_bwe)
>-      : EndToEndTest(test::CallTest::kDefaultTimeoutMs),
>-        sender_call_(nullptr),
>-        receiver_call_(nullptr),
>-        has_seen_pacer_delay_(false),
>-        send_side_bwe_(send_side_bwe) {}
>-
>-  void ModifyVideoConfigs(
>-      VideoSendStream::Config* send_config,
>-      std::vector<VideoReceiveStream::Config>* receive_configs,
>-      VideoEncoderConfig* encoder_config) override {
>-    if (!send_side_bwe_) {
>-      send_config->rtp.extensions.clear();
>-      send_config->rtp.extensions.push_back(RtpExtension(
>-          RtpExtension::kAbsSendTimeUri, test::kAbsSendTimeExtensionId));
>-      (*receive_configs)[0].rtp.remb = true;
>-      (*receive_configs)[0].rtp.transport_cc = false;
>-    }
>-  }
>-
>-  Action OnSendRtp(const uint8_t* packet, size_t length) override {
>-    Call::Stats sender_stats = sender_call_->GetStats();
>-    Call::Stats receiver_stats = receiver_call_->GetStats();
>-    if (!has_seen_pacer_delay_)
>-      has_seen_pacer_delay_ = sender_stats.pacer_delay_ms > 0;
>-    if (sender_stats.send_bandwidth_bps > 0 && has_seen_pacer_delay_) {
>-      if (send_side_bwe_ || receiver_stats.recv_bandwidth_bps > 0)
>-        observation_complete_.Set();
>-    }
>-    return SEND_PACKET;
>-  }
>-
>-  void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
>-    sender_call_ = sender_call;
>-    receiver_call_ = receiver_call;
>-  }
>-
>-  void PerformTest() override {
>-    EXPECT_TRUE(Wait()) << "Timed out while waiting for "
>-                           "non-zero bandwidth stats.";
>-  }
>-
>- private:
>-  Call* sender_call_;
>-  Call* receiver_call_;
>-  bool has_seen_pacer_delay_;
>-  const bool send_side_bwe_;
>-};
>-
>-TEST_P(EndToEndTest, VerifySendSideBweStats) {
>-  BandwidthStatsTest test(true);
>-  RunBaseTest(&test);
>-}
>-
>-TEST_P(EndToEndTest, VerifyRecvSideBweStats) {
>-  BandwidthStatsTest test(false);
>-  RunBaseTest(&test);
>-}
>-
>-// Verifies that it's possible to limit the send BWE by sending a REMB.
>-// This is verified by allowing the send BWE to ramp-up to >1000 kbps,
>-// then have the test generate a REMB of 500 kbps and verify that the send BWE
>-// is reduced to exactly 500 kbps. Then a REMB of 1000 kbps is generated and the
>-// test verifies that the send BWE ramps back up to exactly 1000 kbps.
>-TEST_P(EndToEndTest, RembWithSendSideBwe) {
>-  class BweObserver : public test::EndToEndTest {
>-   public:
>-    BweObserver()
>-        : EndToEndTest(kDefaultTimeoutMs),
>-          sender_call_(nullptr),
>-          clock_(Clock::GetRealTimeClock()),
>-          sender_ssrc_(0),
>-          remb_bitrate_bps_(1000000),
>-          receive_transport_(nullptr),
>-          stop_event_(false, false),
>-          poller_thread_(&BitrateStatsPollingThread,
>-                         this,
>-                         "BitrateStatsPollingThread"),
>-          state_(kWaitForFirstRampUp),
>-          retransmission_rate_limiter_(clock_, 1000) {}
>-
>-    ~BweObserver() {}
>-
>-    test::PacketTransport* CreateReceiveTransport(
>-        test::SingleThreadedTaskQueueForTesting* task_queue) override {
>-      receive_transport_ = new test::PacketTransport(
>-          task_queue, nullptr, this, test::PacketTransport::kReceiver,
>-          payload_type_map_, FakeNetworkPipe::Config());
>-      return receive_transport_;
>-    }
>-
>-    Call::Config GetSenderCallConfig() override {
>-      Call::Config config(event_log_.get());
>-      // Set a high start bitrate to reduce the test completion time.
>-      config.bitrate_config.start_bitrate_bps = remb_bitrate_bps_;
>-      return config;
>-    }
>-
>-    void ModifyVideoConfigs(
>-        VideoSendStream::Config* send_config,
>-        std::vector<VideoReceiveStream::Config>* receive_configs,
>-        VideoEncoderConfig* encoder_config) override {
>-      ASSERT_EQ(1u, send_config->rtp.ssrcs.size());
>-      sender_ssrc_ = send_config->rtp.ssrcs[0];
>-
>-      encoder_config->max_bitrate_bps = 2000000;
>-
>-      ASSERT_EQ(1u, receive_configs->size());
>-      RtpRtcp::Configuration config;
>-      config.receiver_only = true;
>-      config.clock = clock_;
>-      config.outgoing_transport = receive_transport_;
>-      config.retransmission_rate_limiter = &retransmission_rate_limiter_;
>-      rtp_rtcp_.reset(RtpRtcp::CreateRtpRtcp(config));
>-      rtp_rtcp_->SetRemoteSSRC((*receive_configs)[0].rtp.remote_ssrc);
>-      rtp_rtcp_->SetSSRC((*receive_configs)[0].rtp.local_ssrc);
>-      rtp_rtcp_->SetRTCPStatus(RtcpMode::kReducedSize);
>-    }
>-
>-    void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
>-      sender_call_ = sender_call;
>-    }
>-
>-    static void BitrateStatsPollingThread(void* obj) {
>-      static_cast<BweObserver*>(obj)->PollStats();
>-    }
>-
>-    void PollStats() {
>-      do {
>-        if (sender_call_) {
>-          Call::Stats stats = sender_call_->GetStats();
>-          switch (state_) {
>-            case kWaitForFirstRampUp:
>-              if (stats.send_bandwidth_bps >= remb_bitrate_bps_) {
>-                state_ = kWaitForRemb;
>-                remb_bitrate_bps_ /= 2;
>-                rtp_rtcp_->SetRemb(
>-                    remb_bitrate_bps_,
>-                    std::vector<uint32_t>(&sender_ssrc_, &sender_ssrc_ + 1));
>-                rtp_rtcp_->SendRTCP(kRtcpRr);
>-              }
>-              break;
>-
>-            case kWaitForRemb:
>-              if (stats.send_bandwidth_bps == remb_bitrate_bps_) {
>-                state_ = kWaitForSecondRampUp;
>-                remb_bitrate_bps_ *= 2;
>-                rtp_rtcp_->SetRemb(
>-                    remb_bitrate_bps_,
>-                    std::vector<uint32_t>(&sender_ssrc_, &sender_ssrc_ + 1));
>-                rtp_rtcp_->SendRTCP(kRtcpRr);
>-              }
>-              break;
>-
>-            case kWaitForSecondRampUp:
>-              if (stats.send_bandwidth_bps == remb_bitrate_bps_) {
>-                observation_complete_.Set();
>-              }
>-              break;
>-          }
>-        }
>-      } while (!stop_event_.Wait(1000));
>-    }
>-
>-    void PerformTest() override {
>-      poller_thread_.Start();
>-      EXPECT_TRUE(Wait())
>-          << "Timed out while waiting for bitrate to change according to REMB.";
>-      stop_event_.Set();
>-      poller_thread_.Stop();
>-    }
>-
>-   private:
>-    enum TestState { kWaitForFirstRampUp, kWaitForRemb, kWaitForSecondRampUp };
>-
>-    Call* sender_call_;
>-    Clock* const clock_;
>-    uint32_t sender_ssrc_;
>-    int remb_bitrate_bps_;
>-    std::unique_ptr<RtpRtcp> rtp_rtcp_;
>-    test::PacketTransport* receive_transport_;
>-    rtc::Event stop_event_;
>-    rtc::PlatformThread poller_thread_;
>-    TestState state_;
>-    RateLimiter retransmission_rate_limiter_;
>-  } test;
>-
>-  RunBaseTest(&test);
>-}
>-
>-TEST_P(EndToEndTest, StopSendingKeyframeRequestsForInactiveStream) {
>-  class KeyframeRequestObserver : public test::EndToEndTest {
>-   public:
>-    explicit KeyframeRequestObserver(
>-        test::SingleThreadedTaskQueueForTesting* task_queue)
>-        : clock_(Clock::GetRealTimeClock()), task_queue_(task_queue) {}
>-
>-    void OnVideoStreamsCreated(
>-        VideoSendStream* send_stream,
>-        const std::vector<VideoReceiveStream*>& receive_streams) override {
>-      RTC_DCHECK_EQ(1, receive_streams.size());
>-      send_stream_ = send_stream;
>-      receive_stream_ = receive_streams[0];
>-    }
>-
>-    void PerformTest() override {
>-      bool frame_decoded = false;
>-      int64_t start_time = clock_->TimeInMilliseconds();
>-      while (clock_->TimeInMilliseconds() - start_time <= 5000) {
>-        if (receive_stream_->GetStats().frames_decoded > 0) {
>-          frame_decoded = true;
>-          break;
>-        }
>-        SleepMs(100);
>-      }
>-      ASSERT_TRUE(frame_decoded);
>-      task_queue_->SendTask([this]() { send_stream_->Stop(); });
>-      SleepMs(10000);
>-      ASSERT_EQ(
>-          1U, receive_stream_->GetStats().rtcp_packet_type_counts.pli_packets);
>-    }
>-
>-   private:
>-    Clock* clock_;
>-    VideoSendStream* send_stream_;
>-    VideoReceiveStream* receive_stream_;
>-    test::SingleThreadedTaskQueueForTesting* const task_queue_;
>-  } test(&task_queue_);
>-
>-  RunBaseTest(&test);
>-}
>-
>-class ProbingTest : public test::EndToEndTest {
>- public:
>-  explicit ProbingTest(int start_bitrate_bps)
>-      : clock_(Clock::GetRealTimeClock()),
>-        start_bitrate_bps_(start_bitrate_bps),
>-        state_(0),
>-        sender_call_(nullptr) {}
>-
>-  ~ProbingTest() {}
>-
>-  Call::Config GetSenderCallConfig() override {
>-    Call::Config config(event_log_.get());
>-    config.bitrate_config.start_bitrate_bps = start_bitrate_bps_;
>-    return config;
>-  }
>-
>-  void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
>-    sender_call_ = sender_call;
>-  }
>-
>- protected:
>-  Clock* const clock_;
>-  const int start_bitrate_bps_;
>-  int state_;
>-  Call* sender_call_;
>-};
>-
>-// Flaky under MemorySanitizer: bugs.webrtc.org/7419
>-// Flaky on iOS bots: bugs.webrtc.org/7851
>-#if defined(MEMORY_SANITIZER)
>-TEST_P(EndToEndTest, DISABLED_InitialProbing) {
>-#elif defined(TARGET_IPHONE_SIMULATOR) && TARGET_IPHONE_SIMULATOR
>-TEST_P(EndToEndTest, DISABLED_InitialProbing) {
>-#else
>-TEST_P(EndToEndTest, InitialProbing) {
>-#endif
>-  class InitialProbingTest : public ProbingTest {
>-   public:
>-    explicit InitialProbingTest(bool* success)
>-        : ProbingTest(300000), success_(success) {
>-      *success_ = false;
>-    }
>-
>-    void PerformTest() override {
>-      int64_t start_time_ms = clock_->TimeInMilliseconds();
>-      do {
>-        if (clock_->TimeInMilliseconds() - start_time_ms > kTimeoutMs)
>-          break;
>-
>-        Call::Stats stats = sender_call_->GetStats();
>-        // Initial probing is done with a x3 and x6 multiplier of the start
>-        // bitrate, so a x4 multiplier is a high enough threshold.
>-        if (stats.send_bandwidth_bps > 4 * 300000) {
>-          *success_ = true;
>-          break;
>-        }
>-      } while (!observation_complete_.Wait(20));
>-    }
>-
>-   private:
>-    const int kTimeoutMs = 1000;
>-    bool* const success_;
>-  };
>-
>-  bool success = false;
>-  const int kMaxAttempts = 3;
>-  for (int i = 0; i < kMaxAttempts; ++i) {
>-    InitialProbingTest test(&success);
>-    RunBaseTest(&test);
>-    if (success)
>-      return;
>-  }
>-  EXPECT_TRUE(success) << "Failed to perform mid initial probing ("
>-                       << kMaxAttempts << " attempts).";
>-}
>-
>-// Fails on Linux MSan: bugs.webrtc.org/7428
>-#if defined(MEMORY_SANITIZER)
>-TEST_P(EndToEndTest, DISABLED_TriggerMidCallProbing) {
>-// Fails on iOS bots: bugs.webrtc.org/7851
>-#elif defined(TARGET_IPHONE_SIMULATOR) && TARGET_IPHONE_SIMULATOR
>-TEST_P(EndToEndTest, DISABLED_TriggerMidCallProbing) {
>-#else
>-TEST_P(EndToEndTest, TriggerMidCallProbing) {
>-#endif
>-
>-  class TriggerMidCallProbingTest : public ProbingTest {
>-   public:
>-    TriggerMidCallProbingTest(
>-        test::SingleThreadedTaskQueueForTesting* task_queue,
>-        bool* success)
>-        : ProbingTest(300000), success_(success), task_queue_(task_queue) {}
>-
>-    void PerformTest() override {
>-      *success_ = false;
>-      int64_t start_time_ms = clock_->TimeInMilliseconds();
>-      do {
>-        if (clock_->TimeInMilliseconds() - start_time_ms > kTimeoutMs)
>-          break;
>-
>-        Call::Stats stats = sender_call_->GetStats();
>-
>-        switch (state_) {
>-          case 0:
>-            if (stats.send_bandwidth_bps > 5 * 300000) {
>-              Call::Config::BitrateConfig bitrate_config;
>-              bitrate_config.max_bitrate_bps = 100000;
>-              task_queue_->SendTask([this, &bitrate_config]() {
>-                sender_call_->SetBitrateConfig(bitrate_config);
>-              });
>-              ++state_;
>-            }
>-            break;
>-          case 1:
>-            if (stats.send_bandwidth_bps < 110000) {
>-              Call::Config::BitrateConfig bitrate_config;
>-              bitrate_config.max_bitrate_bps = 2500000;
>-              task_queue_->SendTask([this, &bitrate_config]() {
>-                sender_call_->SetBitrateConfig(bitrate_config);
>-              });
>-              ++state_;
>-            }
>-            break;
>-          case 2:
>-            // During high cpu load the pacer will not be able to pace packets
>-            // at the correct speed, but if we go from 110 to 1250 kbps
>-            // in 5 seconds then it is due to probing.
>-            if (stats.send_bandwidth_bps > 1250000) {
>-              *success_ = true;
>-              observation_complete_.Set();
>-            }
>-            break;
>-        }
>-      } while (!observation_complete_.Wait(20));
>-    }
>-
>-   private:
>-    const int kTimeoutMs = 5000;
>-    bool* const success_;
>-    test::SingleThreadedTaskQueueForTesting* const task_queue_;
>-  };
>-
>-  bool success = false;
>-  const int kMaxAttempts = 3;
>-  for (int i = 0; i < kMaxAttempts; ++i) {
>-    TriggerMidCallProbingTest test(&task_queue_, &success);
>-    RunBaseTest(&test);
>-    if (success)
>-      return;
>-  }
>-  EXPECT_TRUE(success) << "Failed to perform mid call probing (" << kMaxAttempts
>-                       << " attempts).";
>-}
>-
>-TEST_P(EndToEndTest, VerifyNackStats) {
>-  static const int kPacketNumberToDrop = 200;
>-  class NackObserver : public test::EndToEndTest {
>-   public:
>-    NackObserver()
>-        : EndToEndTest(kLongTimeoutMs),
>-          sent_rtp_packets_(0),
>-          dropped_rtp_packet_(0),
>-          dropped_rtp_packet_requested_(false),
>-          send_stream_(nullptr),
>-          start_runtime_ms_(-1) {}
>-
>-   private:
>-    Action OnSendRtp(const uint8_t* packet, size_t length) override {
>-      rtc::CritScope lock(&crit_);
>-      if (++sent_rtp_packets_ == kPacketNumberToDrop) {
>-        std::unique_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
>-        RTPHeader header;
>-        EXPECT_TRUE(parser->Parse(packet, length, &header));
>-        dropped_rtp_packet_ = header.sequenceNumber;
>-        return DROP_PACKET;
>-      }
>-      VerifyStats();
>-      return SEND_PACKET;
>-    }
>-
>-    Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
>-      rtc::CritScope lock(&crit_);
>-      test::RtcpPacketParser rtcp_parser;
>-      rtcp_parser.Parse(packet, length);
>-      const std::vector<uint16_t>& nacks = rtcp_parser.nack()->packet_ids();
>-      if (!nacks.empty() && std::find(
>-          nacks.begin(), nacks.end(), dropped_rtp_packet_) != nacks.end()) {
>-        dropped_rtp_packet_requested_ = true;
>-      }
>-      return SEND_PACKET;
>-    }
>-
>-    void VerifyStats() RTC_EXCLUSIVE_LOCKS_REQUIRED(&crit_) {
>-      if (!dropped_rtp_packet_requested_)
>-        return;
>-      int send_stream_nack_packets = 0;
>-      int receive_stream_nack_packets = 0;
>-      VideoSendStream::Stats stats = send_stream_->GetStats();
>-      for (std::map<uint32_t, VideoSendStream::StreamStats>::const_iterator it =
>-           stats.substreams.begin(); it != stats.substreams.end(); ++it) {
>-        const VideoSendStream::StreamStats& stream_stats = it->second;
>-        send_stream_nack_packets +=
>-            stream_stats.rtcp_packet_type_counts.nack_packets;
>-      }
>-      for (size_t i = 0; i < receive_streams_.size(); ++i) {
>-        VideoReceiveStream::Stats stats = receive_streams_[i]->GetStats();
>-        receive_stream_nack_packets +=
>-            stats.rtcp_packet_type_counts.nack_packets;
>-      }
>-      if (send_stream_nack_packets >= 1 && receive_stream_nack_packets >= 1) {
>-        // NACK packet sent on receive stream and received on sent stream.
>-        if (MinMetricRunTimePassed())
>-          observation_complete_.Set();
>-      }
>-    }
>-
>-    bool MinMetricRunTimePassed() {
>-      int64_t now = Clock::GetRealTimeClock()->TimeInMilliseconds();
>-      if (start_runtime_ms_ == -1) {
>-        start_runtime_ms_ = now;
>-        return false;
>-      }
>-      int64_t elapsed_sec = (now - start_runtime_ms_) / 1000;
>-      return elapsed_sec > metrics::kMinRunTimeInSeconds;
>-    }
>-
>-    void ModifyVideoConfigs(
>-        VideoSendStream::Config* send_config,
>-        std::vector<VideoReceiveStream::Config>* receive_configs,
>-        VideoEncoderConfig* encoder_config) override {
>-      send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
>-      (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
>-      (*receive_configs)[0].renderer = &fake_renderer_;
>-    }
>-
>-    void OnVideoStreamsCreated(
>-        VideoSendStream* send_stream,
>-        const std::vector<VideoReceiveStream*>& receive_streams) override {
>-      send_stream_ = send_stream;
>-      receive_streams_ = receive_streams;
>-    }
>-
>-    void PerformTest() override {
>-      EXPECT_TRUE(Wait()) << "Timed out waiting for packet to be NACKed.";
>-    }
>-
>-    test::FakeVideoRenderer fake_renderer_;
>-    rtc::CriticalSection crit_;
>-    uint64_t sent_rtp_packets_;
>-    uint16_t dropped_rtp_packet_ RTC_GUARDED_BY(&crit_);
>-    bool dropped_rtp_packet_requested_ RTC_GUARDED_BY(&crit_);
>-    std::vector<VideoReceiveStream*> receive_streams_;
>-    VideoSendStream* send_stream_;
>-    int64_t start_runtime_ms_;
>-  } test;
>-
>-  metrics::Reset();
>-  RunBaseTest(&test);
>-
>-  EXPECT_EQ(
>-      1, metrics::NumSamples("WebRTC.Video.UniqueNackRequestsSentInPercent"));
>-  EXPECT_EQ(1, metrics::NumSamples(
>-                   "WebRTC.Video.UniqueNackRequestsReceivedInPercent"));
>-  EXPECT_GT(metrics::MinSample("WebRTC.Video.NackPacketsSentPerMinute"), 0);
>-}
>-
>-void EndToEndTest::VerifyHistogramStats(bool use_rtx,
>-                                        bool use_fec,
>-                                        bool screenshare) {
>-  class StatsObserver : public test::EndToEndTest,
>-                        public rtc::VideoSinkInterface<VideoFrame> {
>-   public:
>-    StatsObserver(bool use_rtx, bool use_fec, bool screenshare)
>-        : EndToEndTest(kLongTimeoutMs),
>-          use_rtx_(use_rtx),
>-          use_fec_(use_fec),
>-          screenshare_(screenshare),
>-          // This test uses NACK, so to send FEC we can't use a fake encoder.
>-          vp8_encoder_(use_fec ? VP8Encoder::Create() : nullptr),
>-          sender_call_(nullptr),
>-          receiver_call_(nullptr),
>-          start_runtime_ms_(-1),
>-          num_frames_received_(0) {}
>-
>-   private:
>-    void OnFrame(const VideoFrame& video_frame) override {
>-      // The RTT is needed to estimate |ntp_time_ms| which is used by
>-      // end-to-end delay stats. Therefore, start counting received frames once
>-      // |ntp_time_ms| is valid.
>-      if (video_frame.ntp_time_ms() > 0 &&
>-          Clock::GetRealTimeClock()->CurrentNtpInMilliseconds() >=
>-              video_frame.ntp_time_ms()) {
>-        rtc::CritScope lock(&crit_);
>-        ++num_frames_received_;
>-      }
>-    }
>-
>-    Action OnSendRtp(const uint8_t* packet, size_t length) override {
>-      if (MinMetricRunTimePassed() && MinNumberOfFramesReceived())
>-        observation_complete_.Set();
>-
>-      return SEND_PACKET;
>-    }
>-
>-    bool MinMetricRunTimePassed() {
>-      int64_t now = Clock::GetRealTimeClock()->TimeInMilliseconds();
>-      if (start_runtime_ms_ == -1) {
>-        start_runtime_ms_ = now;
>-        return false;
>-      }
>-      int64_t elapsed_sec = (now - start_runtime_ms_) / 1000;
>-      return elapsed_sec > metrics::kMinRunTimeInSeconds * 2;
>-    }
>-
>-    bool MinNumberOfFramesReceived() const {
>-      const int kMinRequiredHistogramSamples = 200;
>-      rtc::CritScope lock(&crit_);
>-      return num_frames_received_ > kMinRequiredHistogramSamples;
>-    }
>-
>-    void ModifyVideoConfigs(
>-        VideoSendStream::Config* send_config,
>-        std::vector<VideoReceiveStream::Config>* receive_configs,
>-        VideoEncoderConfig* encoder_config) override {
>-      // NACK
>-      send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
>-      (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
>-      (*receive_configs)[0].renderer = this;
>-      // FEC
>-      if (use_fec_) {
>-        send_config->rtp.ulpfec.ulpfec_payload_type = kUlpfecPayloadType;
>-        send_config->rtp.ulpfec.red_payload_type = kRedPayloadType;
>-        send_config->encoder_settings.encoder = vp8_encoder_.get();
>-        send_config->encoder_settings.payload_name = "VP8";
>-        (*receive_configs)[0].decoders[0].payload_name = "VP8";
>-        (*receive_configs)[0].rtp.red_payload_type = kRedPayloadType;
>-        (*receive_configs)[0].rtp.ulpfec_payload_type = kUlpfecPayloadType;
>-      }
>-      // RTX
>-      if (use_rtx_) {
>-        send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[0]);
>-        send_config->rtp.rtx.payload_type = kSendRtxPayloadType;
>-        (*receive_configs)[0].rtp.rtx_ssrc = kSendRtxSsrcs[0];
>-        (*receive_configs)[0]
>-            .rtp.rtx_associated_payload_types[kSendRtxPayloadType] =
>-            kFakeVideoSendPayloadType;
>-        if (use_fec_) {
>-          send_config->rtp.ulpfec.red_rtx_payload_type = kRtxRedPayloadType;
>-          (*receive_configs)[0]
>-              .rtp.rtx_associated_payload_types[kRtxRedPayloadType] =
>-              kSendRtxPayloadType;
>-        }
>-      }
>-      // RTT needed for RemoteNtpTimeEstimator for the receive stream.
>-      (*receive_configs)[0].rtp.rtcp_xr.receiver_reference_time_report = true;
>-      encoder_config->content_type =
>-          screenshare_ ? VideoEncoderConfig::ContentType::kScreen
>-                       : VideoEncoderConfig::ContentType::kRealtimeVideo;
>-    }
>-
>-    void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
>-      sender_call_ = sender_call;
>-      receiver_call_ = receiver_call;
>-    }
>-
>-    void PerformTest() override {
>-      EXPECT_TRUE(Wait()) << "Timed out waiting for packet to be NACKed.";
>-    }
>-
>-    rtc::CriticalSection crit_;
>-    const bool use_rtx_;
>-    const bool use_fec_;
>-    const bool screenshare_;
>-    const std::unique_ptr<VideoEncoder> vp8_encoder_;
>-    Call* sender_call_;
>-    Call* receiver_call_;
>-    int64_t start_runtime_ms_;
>-    int num_frames_received_ RTC_GUARDED_BY(&crit_);
>-  } test(use_rtx, use_fec, screenshare);
>-
>-  metrics::Reset();
>-  RunBaseTest(&test);
>-
>-  std::string video_prefix =
>-      screenshare ? "WebRTC.Video.Screenshare." : "WebRTC.Video.";
>-  // The content type extension is disabled in non screenshare test,
>-  // therefore no slicing on simulcast id should be present.
>-  std::string video_suffix = screenshare ? ".S0" : "";
>-  // Verify that stats have been updated once.
>-  EXPECT_EQ(2, metrics::NumSamples("WebRTC.Call.LifetimeInSeconds"));
>-  EXPECT_EQ(1, metrics::NumSamples(
>-                   "WebRTC.Call.TimeReceivingVideoRtpPacketsInSeconds"));
>-  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Call.VideoBitrateReceivedInKbps"));
>-  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Call.RtcpBitrateReceivedInBps"));
>-  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Call.BitrateReceivedInKbps"));
>-  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Call.EstimatedSendBitrateInKbps"));
>-  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Call.PacerBitrateInKbps"));
>-
>-  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.SendStreamLifetimeInSeconds"));
>-  EXPECT_EQ(1,
>-            metrics::NumSamples("WebRTC.Video.ReceiveStreamLifetimeInSeconds"));
>-
>-  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.NackPacketsSentPerMinute"));
>-  EXPECT_EQ(1,
>-            metrics::NumSamples(video_prefix + "NackPacketsReceivedPerMinute"));
>-  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.FirPacketsSentPerMinute"));
>-  EXPECT_EQ(1,
>-            metrics::NumSamples(video_prefix + "FirPacketsReceivedPerMinute"));
>-  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.PliPacketsSentPerMinute"));
>-  EXPECT_EQ(1,
>-            metrics::NumSamples(video_prefix + "PliPacketsReceivedPerMinute"));
>-
>-  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "KeyFramesSentInPermille"));
>-  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.KeyFramesReceivedInPermille"));
>-
>-  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "SentPacketsLostInPercent"));
>-  EXPECT_EQ(1,
>-            metrics::NumSamples("WebRTC.Video.ReceivedPacketsLostInPercent"));
>-
>-  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "InputWidthInPixels"));
>-  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "InputHeightInPixels"));
>-  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "SentWidthInPixels"));
>-  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "SentHeightInPixels"));
>-  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "ReceivedWidthInPixels"));
>-  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "ReceivedHeightInPixels"));
>-
>-  EXPECT_EQ(1, metrics::NumEvents(video_prefix + "InputWidthInPixels",
>-                                  kDefaultWidth));
>-  EXPECT_EQ(1, metrics::NumEvents(video_prefix + "InputHeightInPixels",
>-                                  kDefaultHeight));
>-  EXPECT_EQ(
>-      1, metrics::NumEvents(video_prefix + "SentWidthInPixels", kDefaultWidth));
>-  EXPECT_EQ(1, metrics::NumEvents(video_prefix + "SentHeightInPixels",
>-                                  kDefaultHeight));
>-  EXPECT_EQ(1, metrics::NumEvents(video_prefix + "ReceivedWidthInPixels",
>-                                  kDefaultWidth));
>-  EXPECT_EQ(1, metrics::NumEvents(video_prefix + "ReceivedHeightInPixels",
>-                                  kDefaultHeight));
>-
>-  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "InputFramesPerSecond"));
>-  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "SentFramesPerSecond"));
>-  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.DecodedFramesPerSecond"));
>-  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.RenderFramesPerSecond"));
>-
>-  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.JitterBufferDelayInMs"));
>-  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.TargetDelayInMs"));
>-  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.CurrentDelayInMs"));
>-  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.OnewayDelayInMs"));
>-
>-  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "EndToEndDelayInMs" +
>-                                   video_suffix));
>-  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "EndToEndDelayMaxInMs" +
>-                                   video_suffix));
>-  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "InterframeDelayInMs" +
>-                                   video_suffix));
>-  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "InterframeDelayMaxInMs" +
>-                                   video_suffix));
>-
>-  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.RenderSqrtPixelsPerSecond"));
>-
>-  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "EncodeTimeInMs"));
>-  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.DecodeTimeInMs"));
>-
>-  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "NumberOfPauseEvents"));
>-  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "PausedTimeInPercent"));
>-
>-  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "BitrateSentInKbps"));
>-  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.BitrateReceivedInKbps"));
>-  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "MediaBitrateSentInKbps"));
>-  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.MediaBitrateReceivedInKbps"));
>-  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "PaddingBitrateSentInKbps"));
>-  EXPECT_EQ(1,
>-            metrics::NumSamples("WebRTC.Video.PaddingBitrateReceivedInKbps"));
>-  EXPECT_EQ(
>-      1, metrics::NumSamples(video_prefix + "RetransmittedBitrateSentInKbps"));
>-  EXPECT_EQ(1, metrics::NumSamples(
>-                   "WebRTC.Video.RetransmittedBitrateReceivedInKbps"));
>-
>-  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.SendDelayInMs"));
>-  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "SendSideDelayInMs"));
>-  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "SendSideDelayMaxInMs"));
>-
>-  int num_rtx_samples = use_rtx ? 1 : 0;
>-  EXPECT_EQ(num_rtx_samples,
>-            metrics::NumSamples("WebRTC.Video.RtxBitrateSentInKbps"));
>-  EXPECT_EQ(num_rtx_samples,
>-            metrics::NumSamples("WebRTC.Video.RtxBitrateReceivedInKbps"));
>-
>-  int num_red_samples = use_fec ? 1 : 0;
>-  EXPECT_EQ(num_red_samples,
>-            metrics::NumSamples("WebRTC.Video.FecBitrateSentInKbps"));
>-  EXPECT_EQ(num_red_samples,
>-            metrics::NumSamples("WebRTC.Video.FecBitrateReceivedInKbps"));
>-  EXPECT_EQ(num_red_samples,
>-            metrics::NumSamples("WebRTC.Video.ReceivedFecPacketsInPercent"));
>-}
>-
>-#if defined(WEBRTC_WIN)
>-// Disabled due to flakiness on Windows (bugs.webrtc.org/7483).
>-#define MAYBE_ContentTypeSwitches DISABLED_ContentTypeSwitches
>-#else
>-#define MAYBE_ContentTypeSwitches ContentTypeSwitches
>-#endif
>-TEST_P(EndToEndTest, MAYBE_ContentTypeSwitches) {
>-  class StatsObserver : public test::BaseTest,
>-                        public rtc::VideoSinkInterface<VideoFrame> {
>-   public:
>-    StatsObserver() : BaseTest(kLongTimeoutMs), num_frames_received_(0) {}
>-
>-    bool ShouldCreateReceivers() const override { return true; }
>-
>-    void OnFrame(const VideoFrame& video_frame) override {
>-      // The RTT is needed to estimate |ntp_time_ms| which is used by
>-      // end-to-end delay stats. Therefore, start counting received frames once
>-      // |ntp_time_ms| is valid.
>-      if (video_frame.ntp_time_ms() > 0 &&
>-          Clock::GetRealTimeClock()->CurrentNtpInMilliseconds() >=
>-              video_frame.ntp_time_ms()) {
>-        rtc::CritScope lock(&crit_);
>-        ++num_frames_received_;
>-      }
>-    }
>-
>-    Action OnSendRtp(const uint8_t* packet, size_t length) override {
>-      if (MinNumberOfFramesReceived())
>-        observation_complete_.Set();
>-      return SEND_PACKET;
>-    }
>-
>-    bool MinNumberOfFramesReceived() const {
>-      // Have some room for frames with wrong content type during switch.
>-      const int kMinRequiredHistogramSamples = 200+50;
>-      rtc::CritScope lock(&crit_);
>-      return num_frames_received_ > kMinRequiredHistogramSamples;
>-    }
>-
>-    // May be called several times.
>-    void PerformTest() override {
>-      EXPECT_TRUE(Wait()) << "Timed out waiting for enough packets.";
>-      // Reset frame counter so next PerformTest() call will do something.
>-      {
>-        rtc::CritScope lock(&crit_);
>-        num_frames_received_ = 0;
>-      }
>-    }
>-
>-    rtc::CriticalSection crit_;
>-    int num_frames_received_ RTC_GUARDED_BY(&crit_);
>-  } test;
>-
>-  metrics::Reset();
>-
>-  Call::Config send_config(test.GetSenderCallConfig());
>-  Call::Config recv_config(test.GetReceiverCallConfig());
>-  VideoEncoderConfig encoder_config_with_screenshare;
>-
>-  task_queue_.SendTask([this, &test, &send_config,
>-                        &recv_config, &encoder_config_with_screenshare]() {
>-    CreateSenderCall(send_config);
>-    CreateReceiverCall(recv_config);
>-
>-    receive_transport_.reset(test.CreateReceiveTransport(&task_queue_));
>-    send_transport_.reset(
>-        test.CreateSendTransport(&task_queue_, sender_call_.get()));
>-    send_transport_->SetReceiver(receiver_call_->Receiver());
>-    receive_transport_->SetReceiver(sender_call_->Receiver());
>-
>-    receiver_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
>-    CreateSendConfig(1, 0, 0, send_transport_.get());
>-    CreateMatchingReceiveConfigs(receive_transport_.get());
>-
>-    // Modify send and receive configs.
>-    video_send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
>-    video_receive_configs_[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
>-    video_receive_configs_[0].renderer = &test;
>-    // RTT needed for RemoteNtpTimeEstimator for the receive stream.
>-    video_receive_configs_[0].rtp.rtcp_xr.receiver_reference_time_report = true;
>-    // Start with realtime video.
>-    video_encoder_config_.content_type =
>-        VideoEncoderConfig::ContentType::kRealtimeVideo;
>-    // Second encoder config for the second part of the test uses screenshare
>-    encoder_config_with_screenshare = video_encoder_config_.Copy();
>-    encoder_config_with_screenshare.content_type =
>-        VideoEncoderConfig::ContentType::kScreen;
>-
>-    CreateVideoStreams();
>-    CreateFrameGeneratorCapturer(kDefaultFramerate, kDefaultWidth,
>-                                 kDefaultHeight);
>-    Start();
>-  });
>-
>-  test.PerformTest();
>-
>-  // Replace old send stream.
>-  task_queue_.SendTask([this, &encoder_config_with_screenshare]() {
>-    sender_call_->DestroyVideoSendStream(video_send_stream_);
>-    video_send_stream_ = sender_call_->CreateVideoSendStream(
>-        video_send_config_.Copy(), encoder_config_with_screenshare.Copy());
>-    video_send_stream_->SetSource(
>-        frame_generator_capturer_.get(),
>-        VideoSendStream::DegradationPreference::kBalanced);
>-    video_send_stream_->Start();
>-  });
>-
>-  // Continue to run test but now with screenshare.
>-  test.PerformTest();
>-
>-  task_queue_.SendTask([this]() {
>-    Stop();
>-    DestroyStreams();
>-    send_transport_.reset();
>-    receive_transport_.reset();
>-    DestroyCalls();
>-  });
>-
>-  // Verify that stats have been updated for both screenshare and video.
>-  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.EndToEndDelayInMs"));
>-  EXPECT_EQ(1,
>-            metrics::NumSamples("WebRTC.Video.Screenshare.EndToEndDelayInMs"));
>-  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.EndToEndDelayMaxInMs"));
>-  EXPECT_EQ(
>-      1, metrics::NumSamples("WebRTC.Video.Screenshare.EndToEndDelayMaxInMs"));
>-  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.InterframeDelayInMs"));
>-  EXPECT_EQ(1,
>-            metrics::NumSamples(
>-                "WebRTC.Video.Screenshare.InterframeDelayInMs"));
>-  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.InterframeDelayMaxInMs"));
>-  EXPECT_EQ(1,
>-            metrics::NumSamples(
>-                "WebRTC.Video.Screenshare.InterframeDelayMaxInMs"));
>-}
>-
>-TEST_P(EndToEndTest, VerifyHistogramStatsWithRtx) {
>-  const bool kEnabledRtx = true;
>-  const bool kEnabledRed = false;
>-  const bool kScreenshare = false;
>-  VerifyHistogramStats(kEnabledRtx, kEnabledRed, kScreenshare);
>-}
>-
>-TEST_P(EndToEndTest, VerifyHistogramStatsWithRed) {
>-  const bool kEnabledRtx = false;
>-  const bool kEnabledRed = true;
>-  const bool kScreenshare = false;
>-  VerifyHistogramStats(kEnabledRtx, kEnabledRed, kScreenshare);
>-}
>-
>-TEST_P(EndToEndTest, VerifyHistogramStatsWithScreenshare) {
>-  const bool kEnabledRtx = false;
>-  const bool kEnabledRed = false;
>-  const bool kScreenshare = true;
>-  VerifyHistogramStats(kEnabledRtx, kEnabledRed, kScreenshare);
>-}
>-
>-void EndToEndTest::TestSendsSetSsrcs(size_t num_ssrcs,
>-                                     bool send_single_ssrc_first) {
>-  class SendsSetSsrcs : public test::EndToEndTest {
>-   public:
>-    SendsSetSsrcs(const uint32_t* ssrcs,
>-                  size_t num_ssrcs,
>-                  bool send_single_ssrc_first)
>-        : EndToEndTest(kDefaultTimeoutMs),
>-          num_ssrcs_(num_ssrcs),
>-          send_single_ssrc_first_(send_single_ssrc_first),
>-          ssrcs_to_observe_(num_ssrcs),
>-          expect_single_ssrc_(send_single_ssrc_first),
>-          send_stream_(nullptr) {
>-      for (size_t i = 0; i < num_ssrcs; ++i)
>-        valid_ssrcs_[ssrcs[i]] = true;
>-    }
>-
>-   private:
>-    Action OnSendRtp(const uint8_t* packet, size_t length) override {
>-      RTPHeader header;
>-      EXPECT_TRUE(parser_->Parse(packet, length, &header));
>-
>-      EXPECT_TRUE(valid_ssrcs_[header.ssrc])
>-          << "Received unknown SSRC: " << header.ssrc;
>-
>-      if (!valid_ssrcs_[header.ssrc])
>-        observation_complete_.Set();
>-
>-      if (!is_observed_[header.ssrc]) {
>-        is_observed_[header.ssrc] = true;
>-        --ssrcs_to_observe_;
>-        if (expect_single_ssrc_) {
>-          expect_single_ssrc_ = false;
>-          observation_complete_.Set();
>-        }
>-      }
>-
>-      if (ssrcs_to_observe_ == 0)
>-        observation_complete_.Set();
>-
>-      return SEND_PACKET;
>-    }
>-
>-    size_t GetNumVideoStreams() const override { return num_ssrcs_; }
>-
>-    // This test use other VideoStream settings than the the default settings
>-    // implemented in DefaultVideoStreamFactory. Therefore  this test implement
>-    // its own VideoEncoderConfig::VideoStreamFactoryInterface which is created
>-    // in ModifyVideoConfigs.
>-    class VideoStreamFactory
>-        : public VideoEncoderConfig::VideoStreamFactoryInterface {
>-     public:
>-      VideoStreamFactory() {}
>-
>-     private:
>-      std::vector<VideoStream> CreateEncoderStreams(
>-          int width,
>-          int height,
>-          const VideoEncoderConfig& encoder_config) override {
>-        std::vector<VideoStream> streams =
>-            test::CreateVideoStreams(width, height, encoder_config);
>-        // Set low simulcast bitrates to not have to wait for bandwidth ramp-up.
>-        for (size_t i = 0; i < encoder_config.number_of_streams; ++i) {
>-          streams[i].min_bitrate_bps = 10000;
>-          streams[i].target_bitrate_bps = 15000;
>-          streams[i].max_bitrate_bps = 20000;
>-        }
>-        return streams;
>-      }
>-    };
>-
>-    void ModifyVideoConfigs(
>-        VideoSendStream::Config* send_config,
>-        std::vector<VideoReceiveStream::Config>* receive_configs,
>-        VideoEncoderConfig* encoder_config) override {
>-      encoder_config->video_stream_factory =
>-          new rtc::RefCountedObject<VideoStreamFactory>();
>-      video_encoder_config_all_streams_ = encoder_config->Copy();
>-      if (send_single_ssrc_first_)
>-        encoder_config->number_of_streams = 1;
>-    }
>-
>-    void OnVideoStreamsCreated(
>-        VideoSendStream* send_stream,
>-        const std::vector<VideoReceiveStream*>& receive_streams) override {
>-      send_stream_ = send_stream;
>-    }
>-
>-    void PerformTest() override {
>-      EXPECT_TRUE(Wait()) << "Timed out while waiting for "
>-                          << (send_single_ssrc_first_ ? "first SSRC."
>-                                                      : "SSRCs.");
>-
>-      if (send_single_ssrc_first_) {
>-        // Set full simulcast and continue with the rest of the SSRCs.
>-        send_stream_->ReconfigureVideoEncoder(
>-            std::move(video_encoder_config_all_streams_));
>-        EXPECT_TRUE(Wait()) << "Timed out while waiting on additional SSRCs.";
>-      }
>-    }
>-
>-   private:
>-    std::map<uint32_t, bool> valid_ssrcs_;
>-    std::map<uint32_t, bool> is_observed_;
>-
>-    const size_t num_ssrcs_;
>-    const bool send_single_ssrc_first_;
>-
>-    size_t ssrcs_to_observe_;
>-    bool expect_single_ssrc_;
>-
>-    VideoSendStream* send_stream_;
>-    VideoEncoderConfig video_encoder_config_all_streams_;
>-  } test(kVideoSendSsrcs, num_ssrcs, send_single_ssrc_first);
>-
>-  RunBaseTest(&test);
>-}
>-
>-TEST_P(EndToEndTest, ReportsSetEncoderRates) {
>-  class EncoderRateStatsTest : public test::EndToEndTest,
>-                               public test::FakeEncoder {
>-   public:
>-    explicit EncoderRateStatsTest(
>-        test::SingleThreadedTaskQueueForTesting* task_queue)
>-        : EndToEndTest(kDefaultTimeoutMs),
>-          FakeEncoder(Clock::GetRealTimeClock()),
>-          task_queue_(task_queue),
>-          send_stream_(nullptr),
>-          bitrate_kbps_(0) {}
>-
>-    void OnVideoStreamsCreated(
>-        VideoSendStream* send_stream,
>-        const std::vector<VideoReceiveStream*>& receive_streams) override {
>-      send_stream_ = send_stream;
>-    }
>-
>-    void ModifyVideoConfigs(
>-        VideoSendStream::Config* send_config,
>-        std::vector<VideoReceiveStream::Config>* receive_configs,
>-        VideoEncoderConfig* encoder_config) override {
>-      send_config->encoder_settings.encoder = this;
>-      RTC_DCHECK_EQ(1, encoder_config->number_of_streams);
>-    }
>-
>-    int32_t SetRateAllocation(const BitrateAllocation& rate_allocation,
>-                              uint32_t framerate) override {
>-      // Make sure not to trigger on any default zero bitrates.
>-      if (rate_allocation.get_sum_bps() == 0)
>-        return 0;
>-      rtc::CritScope lock(&crit_);
>-      bitrate_kbps_ = rate_allocation.get_sum_kbps();
>-      observation_complete_.Set();
>-      return 0;
>-    }
>-
>-    void PerformTest() override {
>-      ASSERT_TRUE(Wait())
>-          << "Timed out while waiting for encoder SetRates() call.";
>-
>-      task_queue_->SendTask([this]() {
>-        WaitForEncoderTargetBitrateMatchStats();
>-        send_stream_->Stop();
>-        WaitForStatsReportZeroTargetBitrate();
>-        send_stream_->Start();
>-        WaitForEncoderTargetBitrateMatchStats();
>-      });
>-    }
>-
>-    void WaitForEncoderTargetBitrateMatchStats() {
>-      for (int i = 0; i < kDefaultTimeoutMs; ++i) {
>-        VideoSendStream::Stats stats = send_stream_->GetStats();
>-        {
>-          rtc::CritScope lock(&crit_);
>-          if ((stats.target_media_bitrate_bps + 500) / 1000 ==
>-              static_cast<int>(bitrate_kbps_)) {
>-            return;
>-          }
>-        }
>-        SleepMs(1);
>-      }
>-      FAIL()
>-          << "Timed out waiting for stats reporting the currently set bitrate.";
>-    }
>-
>-    void WaitForStatsReportZeroTargetBitrate() {
>-      for (int i = 0; i < kDefaultTimeoutMs; ++i) {
>-        if (send_stream_->GetStats().target_media_bitrate_bps == 0) {
>-          return;
>-        }
>-        SleepMs(1);
>-      }
>-      FAIL() << "Timed out waiting for stats reporting zero bitrate.";
>-    }
>-
>-   private:
>-    test::SingleThreadedTaskQueueForTesting* const task_queue_;
>-    rtc::CriticalSection crit_;
>-    VideoSendStream* send_stream_;
>-    uint32_t bitrate_kbps_ RTC_GUARDED_BY(crit_);
>-  } test(&task_queue_);
>-
>-  RunBaseTest(&test);
>-}
>-
>-TEST_P(EndToEndTest, GetStats) {
>-  static const int kStartBitrateBps = 3000000;
>-  static const int kExpectedRenderDelayMs = 20;
>-
>-  class ReceiveStreamRenderer : public rtc::VideoSinkInterface<VideoFrame> {
>-   public:
>-    ReceiveStreamRenderer() {}
>-
>-   private:
>-    void OnFrame(const VideoFrame& video_frame) override {}
>-  };
>-
>-  class StatsObserver : public test::EndToEndTest,
>-                        public rtc::VideoSinkInterface<VideoFrame> {
>-   public:
>-    StatsObserver()
>-        : EndToEndTest(kLongTimeoutMs),
>-          encoder_(Clock::GetRealTimeClock(), 10),
>-          send_stream_(nullptr),
>-          expected_send_ssrcs_(),
>-          check_stats_event_(false, false) {}
>-
>-   private:
>-    Action OnSendRtp(const uint8_t* packet, size_t length) override {
>-      // Drop every 25th packet => 4% loss.
>-      static const int kPacketLossFrac = 25;
>-      RTPHeader header;
>-      RtpUtility::RtpHeaderParser parser(packet, length);
>-      if (parser.Parse(&header) &&
>-          expected_send_ssrcs_.find(header.ssrc) !=
>-              expected_send_ssrcs_.end() &&
>-          header.sequenceNumber % kPacketLossFrac == 0) {
>-        return DROP_PACKET;
>-      }
>-      check_stats_event_.Set();
>-      return SEND_PACKET;
>-    }
>-
>-    Action OnSendRtcp(const uint8_t* packet, size_t length) override {
>-      check_stats_event_.Set();
>-      return SEND_PACKET;
>-    }
>-
>-    Action OnReceiveRtp(const uint8_t* packet, size_t length) override {
>-      check_stats_event_.Set();
>-      return SEND_PACKET;
>-    }
>-
>-    Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
>-      check_stats_event_.Set();
>-      return SEND_PACKET;
>-    }
>-
>-    void OnFrame(const VideoFrame& video_frame) override {
>-      // Ensure that we have at least 5ms send side delay.
>-      SleepMs(5);
>-    }
>-
>-    bool CheckReceiveStats() {
>-      for (size_t i = 0; i < receive_streams_.size(); ++i) {
>-        VideoReceiveStream::Stats stats = receive_streams_[i]->GetStats();
>-        EXPECT_EQ(expected_receive_ssrcs_[i], stats.ssrc);
>-
>-        // Make sure all fields have been populated.
>-        // TODO(pbos): Use CompoundKey if/when we ever know that all stats are
>-        // always filled for all receivers.
>-        receive_stats_filled_["IncomingRate"] |=
>-            stats.network_frame_rate != 0 || stats.total_bitrate_bps != 0;
>-
>-        send_stats_filled_["DecoderImplementationName"] |=
>-            stats.decoder_implementation_name ==
>-            test::FakeDecoder::kImplementationName;
>-        receive_stats_filled_["RenderDelayAsHighAsExpected"] |=
>-            stats.render_delay_ms >= kExpectedRenderDelayMs;
>-
>-        receive_stats_filled_["FrameCallback"] |= stats.decode_frame_rate != 0;
>-
>-        receive_stats_filled_["FrameRendered"] |= stats.render_frame_rate != 0;
>-
>-        receive_stats_filled_["StatisticsUpdated"] |=
>-            stats.rtcp_stats.packets_lost != 0 ||
>-            stats.rtcp_stats.extended_highest_sequence_number != 0 ||
>-            stats.rtcp_stats.fraction_lost != 0 || stats.rtcp_stats.jitter != 0;
>-
>-        receive_stats_filled_["DataCountersUpdated"] |=
>-            stats.rtp_stats.transmitted.payload_bytes != 0 ||
>-            stats.rtp_stats.fec.packets != 0 ||
>-            stats.rtp_stats.transmitted.header_bytes != 0 ||
>-            stats.rtp_stats.transmitted.packets != 0 ||
>-            stats.rtp_stats.transmitted.padding_bytes != 0 ||
>-            stats.rtp_stats.retransmitted.packets != 0;
>-
>-        receive_stats_filled_["CodecStats"] |=
>-            stats.target_delay_ms != 0 || stats.discarded_packets != 0;
>-
>-        receive_stats_filled_["FrameCounts"] |=
>-            stats.frame_counts.key_frames != 0 ||
>-            stats.frame_counts.delta_frames != 0;
>-
>-        receive_stats_filled_["CName"] |= !stats.c_name.empty();
>-
>-        receive_stats_filled_["RtcpPacketTypeCount"] |=
>-            stats.rtcp_packet_type_counts.fir_packets != 0 ||
>-            stats.rtcp_packet_type_counts.nack_packets != 0 ||
>-            stats.rtcp_packet_type_counts.pli_packets != 0 ||
>-            stats.rtcp_packet_type_counts.nack_requests != 0 ||
>-            stats.rtcp_packet_type_counts.unique_nack_requests != 0;
>-
>-        assert(stats.current_payload_type == -1 ||
>-               stats.current_payload_type == kFakeVideoSendPayloadType);
>-        receive_stats_filled_["IncomingPayloadType"] |=
>-            stats.current_payload_type == kFakeVideoSendPayloadType;
>-      }
>-
>-      return AllStatsFilled(receive_stats_filled_);
>-    }
>-
>-    bool CheckSendStats() {
>-      RTC_DCHECK(send_stream_);
>-      VideoSendStream::Stats stats = send_stream_->GetStats();
>-
>-      size_t expected_num_streams =
>-          kNumSimulcastStreams + expected_send_ssrcs_.size();
>-      send_stats_filled_["NumStreams"] |=
>-          stats.substreams.size() == expected_num_streams;
>-
>-      send_stats_filled_["CpuOveruseMetrics"] |=
>-          stats.avg_encode_time_ms != 0 && stats.encode_usage_percent != 0;
>-
>-      send_stats_filled_["EncoderImplementationName"] |=
>-          stats.encoder_implementation_name ==
>-          test::FakeEncoder::kImplementationName;
>-
>-      send_stats_filled_["EncoderPreferredBitrate"] |=
>-          stats.preferred_media_bitrate_bps > 0;
>-
>-      for (std::map<uint32_t, VideoSendStream::StreamStats>::const_iterator it =
>-               stats.substreams.begin();
>-           it != stats.substreams.end(); ++it) {
>-        if (expected_send_ssrcs_.find(it->first) == expected_send_ssrcs_.end())
>-          continue;  // Probably RTX.
>-
>-        send_stats_filled_[CompoundKey("CapturedFrameRate", it->first)] |=
>-            stats.input_frame_rate != 0;
>-
>-        const VideoSendStream::StreamStats& stream_stats = it->second;
>-
>-        send_stats_filled_[CompoundKey("StatisticsUpdated", it->first)] |=
>-            stream_stats.rtcp_stats.packets_lost != 0 ||
>-            stream_stats.rtcp_stats.extended_highest_sequence_number != 0 ||
>-            stream_stats.rtcp_stats.fraction_lost != 0;
>-
>-        send_stats_filled_[CompoundKey("DataCountersUpdated", it->first)] |=
>-            stream_stats.rtp_stats.fec.packets != 0 ||
>-            stream_stats.rtp_stats.transmitted.padding_bytes != 0 ||
>-            stream_stats.rtp_stats.retransmitted.packets != 0 ||
>-            stream_stats.rtp_stats.transmitted.packets != 0;
>-
>-        send_stats_filled_[CompoundKey("BitrateStatisticsObserver.Total",
>-                                       it->first)] |=
>-            stream_stats.total_bitrate_bps != 0;
>-
>-        send_stats_filled_[CompoundKey("BitrateStatisticsObserver.Retransmit",
>-                                       it->first)] |=
>-            stream_stats.retransmit_bitrate_bps != 0;
>-
>-        send_stats_filled_[CompoundKey("FrameCountObserver", it->first)] |=
>-            stream_stats.frame_counts.delta_frames != 0 ||
>-            stream_stats.frame_counts.key_frames != 0;
>-
>-        send_stats_filled_[CompoundKey("OutgoingRate", it->first)] |=
>-            stats.encode_frame_rate != 0;
>-
>-        send_stats_filled_[CompoundKey("Delay", it->first)] |=
>-            stream_stats.avg_delay_ms != 0 || stream_stats.max_delay_ms != 0;
>-
>-        // TODO(pbos): Use CompoundKey when the test makes sure that all SSRCs
>-        // report dropped packets.
>-        send_stats_filled_["RtcpPacketTypeCount"] |=
>-            stream_stats.rtcp_packet_type_counts.fir_packets != 0 ||
>-            stream_stats.rtcp_packet_type_counts.nack_packets != 0 ||
>-            stream_stats.rtcp_packet_type_counts.pli_packets != 0 ||
>-            stream_stats.rtcp_packet_type_counts.nack_requests != 0 ||
>-            stream_stats.rtcp_packet_type_counts.unique_nack_requests != 0;
>-      }
>-
>-      return AllStatsFilled(send_stats_filled_);
>-    }
>-
>-    std::string CompoundKey(const char* name, uint32_t ssrc) {
>-      std::ostringstream oss;
>-      oss << name << "_" << ssrc;
>-      return oss.str();
>-    }
>-
>-    bool AllStatsFilled(const std::map<std::string, bool>& stats_map) {
>-      for (const auto& stat : stats_map) {
>-        if (!stat.second)
>-          return false;
>-      }
>-      return true;
>-    }
>-
>-    test::PacketTransport* CreateSendTransport(
>-        test::SingleThreadedTaskQueueForTesting* task_queue,
>-        Call* sender_call) override {
>-      FakeNetworkPipe::Config network_config;
>-      network_config.loss_percent = 5;
>-      return new test::PacketTransport(task_queue, sender_call, this,
>-                                       test::PacketTransport::kSender,
>-                                       payload_type_map_, network_config);
>-    }
>-
>-    Call::Config GetSenderCallConfig() override {
>-      Call::Config config = EndToEndTest::GetSenderCallConfig();
>-      config.bitrate_config.start_bitrate_bps = kStartBitrateBps;
>-      return config;
>-    }
>-
>-    // This test use other VideoStream settings than the the default settings
>-    // implemented in DefaultVideoStreamFactory. Therefore  this test implement
>-    // its own VideoEncoderConfig::VideoStreamFactoryInterface which is created
>-    // in ModifyVideoConfigs.
>-    class VideoStreamFactory
>-        : public VideoEncoderConfig::VideoStreamFactoryInterface {
>-     public:
>-      VideoStreamFactory() {}
>-
>-     private:
>-      std::vector<VideoStream> CreateEncoderStreams(
>-          int width,
>-          int height,
>-          const VideoEncoderConfig& encoder_config) override {
>-        std::vector<VideoStream> streams =
>-            test::CreateVideoStreams(width, height, encoder_config);
>-        // Set low simulcast bitrates to not have to wait for bandwidth ramp-up.
>-        for (size_t i = 0; i < encoder_config.number_of_streams; ++i) {
>-          streams[i].min_bitrate_bps = 10000;
>-          streams[i].target_bitrate_bps = 15000;
>-          streams[i].max_bitrate_bps = 20000;
>-        }
>-        return streams;
>-      }
>-    };
>-
>-    void ModifyVideoConfigs(
>-        VideoSendStream::Config* send_config,
>-        std::vector<VideoReceiveStream::Config>* receive_configs,
>-        VideoEncoderConfig* encoder_config) override {
>-      encoder_config->video_stream_factory =
>-          new rtc::RefCountedObject<VideoStreamFactory>();
>-      send_config->pre_encode_callback = this;  // Used to inject delay.
>-      expected_cname_ = send_config->rtp.c_name = "SomeCName";
>-
>-      send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
>-      send_config->rtp.rtx.payload_type = kSendRtxPayloadType;
>-
>-      const std::vector<uint32_t>& ssrcs = send_config->rtp.ssrcs;
>-      for (size_t i = 0; i < ssrcs.size(); ++i) {
>-        expected_send_ssrcs_.insert(ssrcs[i]);
>-        expected_receive_ssrcs_.push_back(
>-            (*receive_configs)[i].rtp.remote_ssrc);
>-        (*receive_configs)[i].render_delay_ms = kExpectedRenderDelayMs;
>-        (*receive_configs)[i].renderer = &receive_stream_renderer_;
>-        (*receive_configs)[i].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
>-
>-        (*receive_configs)[i].rtp.rtx_ssrc = kSendRtxSsrcs[i];
>-        (*receive_configs)[i]
>-            .rtp.rtx_associated_payload_types[kSendRtxPayloadType] =
>-            kFakeVideoSendPayloadType;
>-      }
>-
>-      for (size_t i = 0; i < kNumSimulcastStreams; ++i)
>-        send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[i]);
>-
>-      // Use a delayed encoder to make sure we see CpuOveruseMetrics stats that
>-      // are non-zero.
>-      send_config->encoder_settings.encoder = &encoder_;
>-    }
>-
>-    size_t GetNumVideoStreams() const override { return kNumSimulcastStreams; }
>-
>-    void OnVideoStreamsCreated(
>-        VideoSendStream* send_stream,
>-        const std::vector<VideoReceiveStream*>& receive_streams) override {
>-      send_stream_ = send_stream;
>-      receive_streams_ = receive_streams;
>-    }
>-
>-    void PerformTest() override {
>-      Clock* clock = Clock::GetRealTimeClock();
>-      int64_t now = clock->TimeInMilliseconds();
>-      int64_t stop_time = now + test::CallTest::kLongTimeoutMs;
>-      bool receive_ok = false;
>-      bool send_ok = false;
>-
>-      while (now < stop_time) {
>-        if (!receive_ok)
>-          receive_ok = CheckReceiveStats();
>-        if (!send_ok)
>-          send_ok = CheckSendStats();
>-
>-        if (receive_ok && send_ok)
>-          return;
>-
>-        int64_t time_until_timout_ = stop_time - now;
>-        if (time_until_timout_ > 0)
>-          check_stats_event_.Wait(time_until_timout_);
>-        now = clock->TimeInMilliseconds();
>-      }
>-
>-      ADD_FAILURE() << "Timed out waiting for filled stats.";
>-      for (std::map<std::string, bool>::const_iterator it =
>-               receive_stats_filled_.begin();
>-           it != receive_stats_filled_.end(); ++it) {
>-        if (!it->second) {
>-          ADD_FAILURE() << "Missing receive stats: " << it->first;
>-        }
>-      }
>-
>-      for (std::map<std::string, bool>::const_iterator it =
>-               send_stats_filled_.begin();
>-           it != send_stats_filled_.end(); ++it) {
>-        if (!it->second) {
>-          ADD_FAILURE() << "Missing send stats: " << it->first;
>-        }
>-      }
>-    }
>-
>-    test::DelayedEncoder encoder_;
>-    std::vector<VideoReceiveStream*> receive_streams_;
>-    std::map<std::string, bool> receive_stats_filled_;
>-
>-    VideoSendStream* send_stream_;
>-    std::map<std::string, bool> send_stats_filled_;
>-
>-    std::vector<uint32_t> expected_receive_ssrcs_;
>-    std::set<uint32_t> expected_send_ssrcs_;
>-    std::string expected_cname_;
>-
>-    rtc::Event check_stats_event_;
>-    ReceiveStreamRenderer receive_stream_renderer_;
>-  } test;
>-
>-  RunBaseTest(&test);
>-}
>-
>-TEST_P(EndToEndTest, TimingFramesAreReported) {
>-  static const int kExtensionId = 5;
>-
>-  class StatsObserver : public test::EndToEndTest {
>-   public:
>-    StatsObserver() : EndToEndTest(kLongTimeoutMs) {}
>-
>-   private:
>-    void ModifyVideoConfigs(
>-        VideoSendStream::Config* send_config,
>-        std::vector<VideoReceiveStream::Config>* receive_configs,
>-        VideoEncoderConfig* encoder_config) override {
>-      send_config->rtp.extensions.clear();
>-      send_config->rtp.extensions.push_back(
>-          RtpExtension(RtpExtension::kVideoTimingUri, kExtensionId));
>-      for (size_t i = 0; i < receive_configs->size(); ++i) {
>-        (*receive_configs)[i].rtp.extensions.clear();
>-        (*receive_configs)[i].rtp.extensions.push_back(
>-            RtpExtension(RtpExtension::kVideoTimingUri, kExtensionId));
>-      }
>-    }
>-
>-    void OnVideoStreamsCreated(
>-        VideoSendStream* send_stream,
>-        const std::vector<VideoReceiveStream*>& receive_streams) override {
>-      receive_streams_ = receive_streams;
>-    }
>-
>-    void PerformTest() override {
>-      // No frames reported initially.
>-      for (size_t i = 0; i < receive_streams_.size(); ++i) {
>-        EXPECT_FALSE(receive_streams_[i]->GetStats().timing_frame_info);
>-      }
>-      // Wait for at least one timing frame to be sent with 100ms grace period.
>-      SleepMs(kDefaultTimingFramesDelayMs + 100);
>-      // Check that timing frames are reported for each stream.
>-      for (size_t i = 0; i < receive_streams_.size(); ++i) {
>-        EXPECT_TRUE(receive_streams_[i]->GetStats().timing_frame_info);
>-      }
>-    }
>-
>-    std::vector<VideoReceiveStream*> receive_streams_;
>-  } test;
>-
>-  RunBaseTest(&test);
>-}
>-
>-class RtcpXrObserver : public test::EndToEndTest {
>- public:
>-  RtcpXrObserver(bool enable_rrtr, bool enable_target_bitrate,
>-                 bool enable_zero_target_bitrate)
>-      : EndToEndTest(test::CallTest::kDefaultTimeoutMs),
>-        enable_rrtr_(enable_rrtr),
>-        enable_target_bitrate_(enable_target_bitrate),
>-        enable_zero_target_bitrate_(enable_zero_target_bitrate),
>-        sent_rtcp_sr_(0),
>-        sent_rtcp_rr_(0),
>-        sent_rtcp_rrtr_(0),
>-        sent_rtcp_target_bitrate_(false),
>-        sent_zero_rtcp_target_bitrate_(false),
>-        sent_rtcp_dlrr_(0) {}
>-
>- private:
>-  // Receive stream should send RR packets (and RRTR packets if enabled).
>-  Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
>-    rtc::CritScope lock(&crit_);
>-    test::RtcpPacketParser parser;
>-    EXPECT_TRUE(parser.Parse(packet, length));
>-
>-    sent_rtcp_rr_ += parser.receiver_report()->num_packets();
>-    EXPECT_EQ(0, parser.sender_report()->num_packets());
>-    EXPECT_GE(1, parser.xr()->num_packets());
>-    if (parser.xr()->num_packets() > 0) {
>-      if (parser.xr()->rrtr())
>-        ++sent_rtcp_rrtr_;
>-      EXPECT_FALSE(parser.xr()->dlrr());
>-    }
>-
>-    return SEND_PACKET;
>-  }
>-  // Send stream should send SR packets (and DLRR packets if enabled).
>-  Action OnSendRtcp(const uint8_t* packet, size_t length) override {
>-    rtc::CritScope lock(&crit_);
>-    test::RtcpPacketParser parser;
>-    EXPECT_TRUE(parser.Parse(packet, length));
>-
>-    sent_rtcp_sr_ += parser.sender_report()->num_packets();
>-    EXPECT_LE(parser.xr()->num_packets(), 1);
>-    if (parser.xr()->num_packets() > 0) {
>-      EXPECT_FALSE(parser.xr()->rrtr());
>-      if (parser.xr()->dlrr())
>-        ++sent_rtcp_dlrr_;
>-      if (parser.xr()->target_bitrate()) {
>-        sent_rtcp_target_bitrate_ = true;
>-        for (const rtcp::TargetBitrate::BitrateItem& item :
>-                 parser.xr()->target_bitrate()->GetTargetBitrates()) {
>-          if (item.target_bitrate_kbps == 0) {
>-            sent_zero_rtcp_target_bitrate_ = true;
>-            break;
>-          }
>-        }
>-      }
>-    }
>-
>-    if (sent_rtcp_sr_ > kNumRtcpReportPacketsToObserve &&
>-        sent_rtcp_rr_ > kNumRtcpReportPacketsToObserve &&
>-        (sent_rtcp_target_bitrate_ || !enable_target_bitrate_) &&
>-        (sent_zero_rtcp_target_bitrate_ || !enable_zero_target_bitrate_)) {
>-      if (enable_rrtr_) {
>-        EXPECT_GT(sent_rtcp_rrtr_, 0);
>-        EXPECT_GT(sent_rtcp_dlrr_, 0);
>-      } else {
>-        EXPECT_EQ(sent_rtcp_rrtr_, 0);
>-        EXPECT_EQ(sent_rtcp_dlrr_, 0);
>-      }
>-      EXPECT_EQ(enable_target_bitrate_, sent_rtcp_target_bitrate_);
>-      EXPECT_EQ(enable_zero_target_bitrate_, sent_zero_rtcp_target_bitrate_);
>-      observation_complete_.Set();
>-    }
>-    return SEND_PACKET;
>-  }
>-
>-  size_t GetNumVideoStreams() const override {
>-    // When sending a zero target bitrate, we use two spatial layers so that
>-    // we'll still have a layer with non-zero bitrate.
>-    return enable_zero_target_bitrate_ ? 2 : 1;
>-  }
>-
>-  // This test uses VideoStream settings different from the the default one
>-  // implemented in DefaultVideoStreamFactory, so it implements its own
>-  // VideoEncoderConfig::VideoStreamFactoryInterface which is created
>-  // in ModifyVideoConfigs.
>-  class ZeroTargetVideoStreamFactory
>-      : public VideoEncoderConfig::VideoStreamFactoryInterface {
>-   public:
>-    ZeroTargetVideoStreamFactory() {}
>-
>-   private:
>-    std::vector<VideoStream> CreateEncoderStreams(
>-        int width,
>-        int height,
>-        const VideoEncoderConfig& encoder_config) override {
>-      std::vector<VideoStream> streams =
>-          test::CreateVideoStreams(width, height, encoder_config);
>-      // Set one of the streams' target bitrates to zero to test that a
>-      // bitrate of 0 can be signalled.
>-      streams[encoder_config.number_of_streams-1].min_bitrate_bps = 0;
>-      streams[encoder_config.number_of_streams-1].target_bitrate_bps = 0;
>-      streams[encoder_config.number_of_streams-1].max_bitrate_bps = 0;
>-      return streams;
>-    }
>-  };
>-
>-  void ModifyVideoConfigs(
>-      VideoSendStream::Config* send_config,
>-      std::vector<VideoReceiveStream::Config>* receive_configs,
>-      VideoEncoderConfig* encoder_config) override {
>-    if (enable_zero_target_bitrate_) {
>-      encoder_config->video_stream_factory =
>-          new rtc::RefCountedObject<ZeroTargetVideoStreamFactory>();
>-
>-      // Configure VP8 to be able to use simulcast.
>-      send_config->encoder_settings.payload_name = "VP8";
>-      (*receive_configs)[0].decoders.resize(1);
>-      (*receive_configs)[0].decoders[0].payload_type =
>-          send_config->encoder_settings.payload_type;
>-      (*receive_configs)[0].decoders[0].payload_name =
>-          send_config->encoder_settings.payload_name;
>-    }
>-    if (enable_target_bitrate_) {
>-      // TargetBitrate only signaled for screensharing.
>-      encoder_config->content_type = VideoEncoderConfig::ContentType::kScreen;
>-    }
>-    (*receive_configs)[0].rtp.rtcp_mode = RtcpMode::kReducedSize;
>-    (*receive_configs)[0].rtp.rtcp_xr.receiver_reference_time_report =
>-        enable_rrtr_;
>-  }
>-
>-  void PerformTest() override {
>-    EXPECT_TRUE(Wait())
>-        << "Timed out while waiting for RTCP SR/RR packets to be sent.";
>-  }
>-
>-  static const int kNumRtcpReportPacketsToObserve = 5;
>-
>-  rtc::CriticalSection crit_;
>-  const bool enable_rrtr_;
>-  const bool enable_target_bitrate_;
>-  const bool enable_zero_target_bitrate_;
>-  int sent_rtcp_sr_;
>-  int sent_rtcp_rr_ RTC_GUARDED_BY(&crit_);
>-  int sent_rtcp_rrtr_ RTC_GUARDED_BY(&crit_);
>-  bool sent_rtcp_target_bitrate_ RTC_GUARDED_BY(&crit_);
>-  bool sent_zero_rtcp_target_bitrate_ RTC_GUARDED_BY(&crit_);
>-  int sent_rtcp_dlrr_;
>-};
>-
>-TEST_P(EndToEndTest, TestExtendedReportsWithRrtrWithoutTargetBitrate) {
>-  RtcpXrObserver test(/*enable_rrtr=*/true, /*enable_target_bitrate=*/false,
>-                      /*enable_zero_target_bitrate=*/false);
>-  RunBaseTest(&test);
>-}
>-
>-TEST_P(EndToEndTest, TestExtendedReportsWithoutRrtrWithoutTargetBitrate) {
>-  RtcpXrObserver test(/*enable_rrtr=*/false, /*enable_target_bitrate=*/false,
>-                      /*enable_zero_target_bitrate=*/false);
>-  RunBaseTest(&test);
>-}
>-
>-TEST_P(EndToEndTest, TestExtendedReportsWithRrtrWithTargetBitrate) {
>-  RtcpXrObserver test(/*enable_rrtr=*/true, /*enable_target_bitrate=*/true,
>-                      /*enable_zero_target_bitrate=*/false);
>-  RunBaseTest(&test);
>-}
>-
>-TEST_P(EndToEndTest, TestExtendedReportsWithoutRrtrWithTargetBitrate) {
>-  RtcpXrObserver test(/*enable_rrtr=*/false, /*enable_target_bitrate=*/true,
>-                      /*enable_zero_target_bitrate=*/false);
>-  RunBaseTest(&test);
>-}
>-
>-TEST_P(EndToEndTest, TestExtendedReportsCanSignalZeroTargetBitrate) {
>-  RtcpXrObserver test(/*enable_rrtr=*/false, /*enable_target_bitrate=*/true,
>-                      /*enable_zero_target_bitrate=*/true);
>-  RunBaseTest(&test);
>-}
>-
>-TEST_P(EndToEndTest, TestReceivedRtpPacketStats) {
>-  static const size_t kNumRtpPacketsToSend = 5;
>-  class ReceivedRtpStatsObserver : public test::EndToEndTest {
>-   public:
>-    ReceivedRtpStatsObserver()
>-        : EndToEndTest(kDefaultTimeoutMs),
>-          receive_stream_(nullptr),
>-          sent_rtp_(0) {}
>-
>-   private:
>-    void OnVideoStreamsCreated(
>-        VideoSendStream* send_stream,
>-        const std::vector<VideoReceiveStream*>& receive_streams) override {
>-      receive_stream_ = receive_streams[0];
>-    }
>-
>-    Action OnSendRtp(const uint8_t* packet, size_t length) override {
>-      if (sent_rtp_ >= kNumRtpPacketsToSend) {
>-        VideoReceiveStream::Stats stats = receive_stream_->GetStats();
>-        if (kNumRtpPacketsToSend == stats.rtp_stats.transmitted.packets) {
>-          observation_complete_.Set();
>-        }
>-        return DROP_PACKET;
>-      }
>-      ++sent_rtp_;
>-      return SEND_PACKET;
>-    }
>-
>-    void PerformTest() override {
>-      EXPECT_TRUE(Wait())
>-          << "Timed out while verifying number of received RTP packets.";
>-    }
>-
>-    VideoReceiveStream* receive_stream_;
>-    uint32_t sent_rtp_;
>-  } test;
>-
>-  RunBaseTest(&test);
>-}
>-
>-TEST_P(EndToEndTest, SendsSetSsrc) {
>-  TestSendsSetSsrcs(1, false);
>-}
>-
>-TEST_P(EndToEndTest, SendsSetSimulcastSsrcs) {
>-  TestSendsSetSsrcs(kNumSimulcastStreams, false);
>-}
>-
>-TEST_P(EndToEndTest, CanSwitchToUseAllSsrcs) {
>-  TestSendsSetSsrcs(kNumSimulcastStreams, true);
>-}
>-
>-TEST_P(EndToEndTest, DISABLED_RedundantPayloadsTransmittedOnAllSsrcs) {
>-  class ObserveRedundantPayloads: public test::EndToEndTest {
>-   public:
>-    ObserveRedundantPayloads()
>-        : EndToEndTest(kDefaultTimeoutMs),
>-          ssrcs_to_observe_(kNumSimulcastStreams) {
>-      for (size_t i = 0; i < kNumSimulcastStreams; ++i) {
>-        registered_rtx_ssrc_[kSendRtxSsrcs[i]] = true;
>-      }
>-        }
>-
>-   private:
>-    Action OnSendRtp(const uint8_t* packet, size_t length) override {
>-      RTPHeader header;
>-      EXPECT_TRUE(parser_->Parse(packet, length, &header));
>-
>-      if (!registered_rtx_ssrc_[header.ssrc])
>-        return SEND_PACKET;
>-
>-      EXPECT_LE(header.headerLength + header.paddingLength, length);
>-      const bool packet_is_redundant_payload =
>-          header.headerLength + header.paddingLength < length;
>-
>-      if (!packet_is_redundant_payload)
>-        return SEND_PACKET;
>-
>-      if (!observed_redundant_retransmission_[header.ssrc]) {
>-        observed_redundant_retransmission_[header.ssrc] = true;
>-        if (--ssrcs_to_observe_ == 0)
>-          observation_complete_.Set();
>-      }
>-
>-      return SEND_PACKET;
>-    }
>-
>-    size_t GetNumVideoStreams() const override { return kNumSimulcastStreams; }
>-
>-    // This test use other VideoStream settings than the the default settings
>-    // implemented in DefaultVideoStreamFactory. Therefore  this test implement
>-    // its own VideoEncoderConfig::VideoStreamFactoryInterface which is created
>-    // in ModifyVideoConfigs.
>-    class VideoStreamFactory
>-        : public VideoEncoderConfig::VideoStreamFactoryInterface {
>-     public:
>-      VideoStreamFactory() {}
>-
>-     private:
>-      std::vector<VideoStream> CreateEncoderStreams(
>-          int width,
>-          int height,
>-          const VideoEncoderConfig& encoder_config) override {
>-        std::vector<VideoStream> streams =
>-            test::CreateVideoStreams(width, height, encoder_config);
>-        // Set low simulcast bitrates to not have to wait for bandwidth ramp-up.
>-        for (size_t i = 0; i < encoder_config.number_of_streams; ++i) {
>-          streams[i].min_bitrate_bps = 10000;
>-          streams[i].target_bitrate_bps = 15000;
>-          streams[i].max_bitrate_bps = 20000;
>-        }
>-        return streams;
>-      }
>-    };
>-
>-    void ModifyVideoConfigs(
>-        VideoSendStream::Config* send_config,
>-        std::vector<VideoReceiveStream::Config>* receive_configs,
>-        VideoEncoderConfig* encoder_config) override {
>-      // Set low simulcast bitrates to not have to wait for bandwidth ramp-up.
>-      encoder_config->video_stream_factory =
>-          new rtc::RefCountedObject<VideoStreamFactory>();
>-      send_config->rtp.rtx.payload_type = kSendRtxPayloadType;
>-
>-      for (size_t i = 0; i < kNumSimulcastStreams; ++i)
>-        send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[i]);
>-
>-      // Significantly higher than max bitrates for all video streams -> forcing
>-      // padding to trigger redundant padding on all RTX SSRCs.
>-      encoder_config->min_transmit_bitrate_bps = 100000;
>-    }
>-
>-    void PerformTest() override {
>-      EXPECT_TRUE(Wait())
>-          << "Timed out while waiting for redundant payloads on all SSRCs.";
>-    }
>-
>-   private:
>-    size_t ssrcs_to_observe_;
>-    std::map<uint32_t, bool> observed_redundant_retransmission_;
>-    std::map<uint32_t, bool> registered_rtx_ssrc_;
>-  } test;
>-
>-  RunBaseTest(&test);
>-}
>-
>-void EndToEndTest::TestRtpStatePreservation(bool use_rtx,
>-                                            bool provoke_rtcpsr_before_rtp) {
>-  // This test uses other VideoStream settings than the the default settings
>-  // implemented in DefaultVideoStreamFactory. Therefore this test implements
>-  // its own VideoEncoderConfig::VideoStreamFactoryInterface which is created
>-  // in ModifyVideoConfigs.
>-  class VideoStreamFactory
>-      : public VideoEncoderConfig::VideoStreamFactoryInterface {
>-   public:
>-    VideoStreamFactory() {}
>-
>-   private:
>-    std::vector<VideoStream> CreateEncoderStreams(
>-        int width,
>-        int height,
>-        const VideoEncoderConfig& encoder_config) override {
>-      std::vector<VideoStream> streams =
>-          test::CreateVideoStreams(width, height, encoder_config);
>-
>-      if (encoder_config.number_of_streams > 1) {
>-        // Lower bitrates so that all streams send initially.
>-        RTC_DCHECK_EQ(3, encoder_config.number_of_streams);
>-        for (size_t i = 0; i < encoder_config.number_of_streams; ++i) {
>-          streams[i].min_bitrate_bps = 10000;
>-          streams[i].target_bitrate_bps = 15000;
>-          streams[i].max_bitrate_bps = 20000;
>-        }
>-      } else {
>-        // Use the same total bitrates when sending a single stream to avoid
>-        // lowering
>-        // the bitrate estimate and requiring a subsequent rampup.
>-        streams[0].min_bitrate_bps = 3 * 10000;
>-        streams[0].target_bitrate_bps = 3 * 15000;
>-        streams[0].max_bitrate_bps = 3 * 20000;
>-      }
>-      return streams;
>-    }
>-  };
>-
>-  class RtpSequenceObserver : public test::RtpRtcpObserver {
>-   public:
>-    explicit RtpSequenceObserver(bool use_rtx)
>-        : test::RtpRtcpObserver(kDefaultTimeoutMs),
>-          ssrcs_to_observe_(kNumSimulcastStreams) {
>-      for (size_t i = 0; i < kNumSimulcastStreams; ++i) {
>-        ssrc_is_rtx_[kVideoSendSsrcs[i]] = false;
>-        if (use_rtx)
>-          ssrc_is_rtx_[kSendRtxSsrcs[i]] = true;
>-      }
>-    }
>-
>-    void ResetExpectedSsrcs(size_t num_expected_ssrcs) {
>-      rtc::CritScope lock(&crit_);
>-      ssrc_observed_.clear();
>-      ssrcs_to_observe_ = num_expected_ssrcs;
>-    }
>-
>-   private:
>-    void ValidateTimestampGap(uint32_t ssrc,
>-                              uint32_t timestamp,
>-                              bool only_padding)
>-        RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_) {
>-      static const int32_t kMaxTimestampGap = kDefaultTimeoutMs * 90;
>-      auto timestamp_it = last_observed_timestamp_.find(ssrc);
>-      if (timestamp_it == last_observed_timestamp_.end()) {
>-        EXPECT_FALSE(only_padding);
>-        last_observed_timestamp_[ssrc] = timestamp;
>-      } else {
>-        // Verify timestamps are reasonably close.
>-        uint32_t latest_observed = timestamp_it->second;
>-        // Wraparound handling is unnecessary here as long as an int variable
>-        // is used to store the result.
>-        int32_t timestamp_gap = timestamp - latest_observed;
>-        EXPECT_LE(std::abs(timestamp_gap), kMaxTimestampGap)
>-            << "Gap in timestamps (" << latest_observed << " -> " << timestamp
>-            << ") too large for SSRC: " << ssrc << ".";
>-        timestamp_it->second = timestamp;
>-      }
>-    }
>-
>-    Action OnSendRtp(const uint8_t* packet, size_t length) override {
>-      RTPHeader header;
>-      EXPECT_TRUE(parser_->Parse(packet, length, &header));
>-      const uint32_t ssrc = header.ssrc;
>-      const int64_t sequence_number =
>-          seq_numbers_unwrapper_.Unwrap(header.sequenceNumber);
>-      const uint32_t timestamp = header.timestamp;
>-      const bool only_padding =
>-          header.headerLength + header.paddingLength == length;
>-
>-      EXPECT_TRUE(ssrc_is_rtx_.find(ssrc) != ssrc_is_rtx_.end())
>-          << "Received SSRC that wasn't configured: " << ssrc;
>-
>-      static const int64_t kMaxSequenceNumberGap = 100;
>-      std::list<int64_t>* seq_numbers = &last_observed_seq_numbers_[ssrc];
>-      if (seq_numbers->empty()) {
>-        seq_numbers->push_back(sequence_number);
>-      } else {
>-        // We shouldn't get replays of previous sequence numbers.
>-        for (int64_t observed : *seq_numbers) {
>-          EXPECT_NE(observed, sequence_number)
>-              << "Received sequence number " << sequence_number
>-              << " for SSRC " << ssrc << " 2nd time.";
>-        }
>-        // Verify sequence numbers are reasonably close.
>-        int64_t latest_observed = seq_numbers->back();
>-        int64_t sequence_number_gap = sequence_number - latest_observed;
>-        EXPECT_LE(std::abs(sequence_number_gap), kMaxSequenceNumberGap)
>-            << "Gap in sequence numbers (" << latest_observed << " -> "
>-            << sequence_number << ") too large for SSRC: " << ssrc << ".";
>-        seq_numbers->push_back(sequence_number);
>-        if (seq_numbers->size() >= kMaxSequenceNumberGap) {
>-          seq_numbers->pop_front();
>-        }
>-      }
>-
>-      if (!ssrc_is_rtx_[ssrc]) {
>-        rtc::CritScope lock(&crit_);
>-        ValidateTimestampGap(ssrc, timestamp, only_padding);
>-
>-        // Wait for media packets on all ssrcs.
>-        if (!ssrc_observed_[ssrc] && !only_padding) {
>-          ssrc_observed_[ssrc] = true;
>-          if (--ssrcs_to_observe_ == 0)
>-            observation_complete_.Set();
>-        }
>-      }
>-
>-      return SEND_PACKET;
>-    }
>-
>-    Action OnSendRtcp(const uint8_t* packet, size_t length) override {
>-      test::RtcpPacketParser rtcp_parser;
>-      rtcp_parser.Parse(packet, length);
>-      if (rtcp_parser.sender_report()->num_packets() > 0) {
>-        uint32_t ssrc = rtcp_parser.sender_report()->sender_ssrc();
>-        uint32_t rtcp_timestamp = rtcp_parser.sender_report()->rtp_timestamp();
>-
>-        rtc::CritScope lock(&crit_);
>-        ValidateTimestampGap(ssrc, rtcp_timestamp, false);
>-      }
>-      return SEND_PACKET;
>-    }
>-
>-    SequenceNumberUnwrapper seq_numbers_unwrapper_;
>-    std::map<uint32_t, std::list<int64_t>> last_observed_seq_numbers_;
>-    std::map<uint32_t, uint32_t> last_observed_timestamp_;
>-    std::map<uint32_t, bool> ssrc_is_rtx_;
>-
>-    rtc::CriticalSection crit_;
>-    size_t ssrcs_to_observe_ RTC_GUARDED_BY(crit_);
>-    std::map<uint32_t, bool> ssrc_observed_ RTC_GUARDED_BY(crit_);
>-  } observer(use_rtx);
>-
>-  std::unique_ptr<test::PacketTransport> send_transport;
>-  std::unique_ptr<test::PacketTransport> receive_transport;
>-
>-  Call::Config config(event_log_.get());
>-  VideoEncoderConfig one_stream;
>-
>-  task_queue_.SendTask([this, &observer, &send_transport, &receive_transport,
>-                        &config, &one_stream, use_rtx]() {
>-    CreateCalls(config, config);
>-
>-    send_transport = rtc::MakeUnique<test::PacketTransport>(
>-        &task_queue_, sender_call_.get(), &observer,
>-        test::PacketTransport::kSender, payload_type_map_,
>-        FakeNetworkPipe::Config());
>-    receive_transport = rtc::MakeUnique<test::PacketTransport>(
>-        &task_queue_, nullptr, &observer, test::PacketTransport::kReceiver,
>-        payload_type_map_, FakeNetworkPipe::Config());
>-    send_transport->SetReceiver(receiver_call_->Receiver());
>-    receive_transport->SetReceiver(sender_call_->Receiver());
>-
>-    CreateSendConfig(kNumSimulcastStreams, 0, 0, send_transport.get());
>-
>-    if (use_rtx) {
>-      for (size_t i = 0; i < kNumSimulcastStreams; ++i) {
>-        video_send_config_.rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[i]);
>-      }
>-      video_send_config_.rtp.rtx.payload_type = kSendRtxPayloadType;
>-    }
>-
>-    video_encoder_config_.video_stream_factory =
>-        new rtc::RefCountedObject<VideoStreamFactory>();
>-    // Use the same total bitrates when sending a single stream to avoid
>-    // lowering the bitrate estimate and requiring a subsequent rampup.
>-    one_stream = video_encoder_config_.Copy();
>-    // one_stream.streams.resize(1);
>-    one_stream.number_of_streams = 1;
>-    CreateMatchingReceiveConfigs(receive_transport.get());
>-
>-    CreateVideoStreams();
>-    CreateFrameGeneratorCapturer(30, 1280, 720);
>-
>-    Start();
>-  });
>-
>-  EXPECT_TRUE(observer.Wait())
>-      << "Timed out waiting for all SSRCs to send packets.";
>-
>-  // Test stream resetting more than once to make sure that the state doesn't
>-  // get set once (this could be due to using std::map::insert for instance).
>-  for (size_t i = 0; i < 3; ++i) {
>-    task_queue_.SendTask([&]() {
>-      frame_generator_capturer_->Stop();
>-      sender_call_->DestroyVideoSendStream(video_send_stream_);
>-
>-      // Re-create VideoSendStream with only one stream.
>-      video_send_stream_ = sender_call_->CreateVideoSendStream(
>-          video_send_config_.Copy(), one_stream.Copy());
>-      video_send_stream_->Start();
>-      if (provoke_rtcpsr_before_rtp) {
>-        // Rapid Resync Request forces sending RTCP Sender Report back.
>-        // Using this request speeds up this test because then there is no need
>-        // to wait for a second for periodic Sender Report.
>-        rtcp::RapidResyncRequest force_send_sr_back_request;
>-        rtc::Buffer packet = force_send_sr_back_request.Build();
>-        static_cast<webrtc::test::DirectTransport*>(receive_transport.get())
>-            ->SendRtcp(packet.data(), packet.size());
>-      }
>-      CreateFrameGeneratorCapturer(30, 1280, 720);
>-      frame_generator_capturer_->Start();
>-    });
>-
>-    observer.ResetExpectedSsrcs(1);
>-    EXPECT_TRUE(observer.Wait()) << "Timed out waiting for single RTP packet.";
>-
>-    // Reconfigure back to use all streams.
>-    task_queue_.SendTask([this]() {
>-      video_send_stream_->ReconfigureVideoEncoder(video_encoder_config_.Copy());
>-    });
>-    observer.ResetExpectedSsrcs(kNumSimulcastStreams);
>-    EXPECT_TRUE(observer.Wait())
>-        << "Timed out waiting for all SSRCs to send packets.";
>-
>-    // Reconfigure down to one stream.
>-    task_queue_.SendTask([this, &one_stream]() {
>-      video_send_stream_->ReconfigureVideoEncoder(one_stream.Copy());
>-    });
>-    observer.ResetExpectedSsrcs(1);
>-    EXPECT_TRUE(observer.Wait()) << "Timed out waiting for single RTP packet.";
>-
>-    // Reconfigure back to use all streams.
>-    task_queue_.SendTask([this]() {
>-      video_send_stream_->ReconfigureVideoEncoder(video_encoder_config_.Copy());
>-    });
>-    observer.ResetExpectedSsrcs(kNumSimulcastStreams);
>-    EXPECT_TRUE(observer.Wait())
>-        << "Timed out waiting for all SSRCs to send packets.";
>-  }
>-
>-  task_queue_.SendTask([this, &send_transport, &receive_transport]() {
>-    Stop();
>-    DestroyStreams();
>-    send_transport.reset();
>-    receive_transport.reset();
>-    DestroyCalls();
>-  });
>-}
>-
>-TEST_P(EndToEndTest, RestartingSendStreamPreservesRtpState) {
>-  TestRtpStatePreservation(false, false);
>-}
>-
>-TEST_P(EndToEndTest, RestartingSendStreamPreservesRtpStatesWithRtx) {
>-  TestRtpStatePreservation(true, false);
>-}
>-
>-TEST_P(EndToEndTest, RestartingSendStreamKeepsRtpAndRtcpTimestampsSynced) {
>-  TestRtpStatePreservation(true, true);
>-}
>-
>-// This test is flaky on linux_memcheck. Disable on all linux bots until
>-// flakyness has been fixed.
>-// https://bugs.chromium.org/p/webrtc/issues/detail?id=7737
>-#if defined(WEBRTC_LINUX)
>-TEST_P(EndToEndTest, DISABLED_TestFlexfecRtpStatePreservation) {
>-#else
>-TEST_P(EndToEndTest, TestFlexfecRtpStatePreservation) {
>-#endif
>-  class RtpSequenceObserver : public test::RtpRtcpObserver {
>-   public:
>-    RtpSequenceObserver()
>-        : test::RtpRtcpObserver(kDefaultTimeoutMs),
>-          num_flexfec_packets_sent_(0) {}
>-
>-    void ResetPacketCount() {
>-      rtc::CritScope lock(&crit_);
>-      num_flexfec_packets_sent_ = 0;
>-    }
>-
>-   private:
>-    Action OnSendRtp(const uint8_t* packet, size_t length) override {
>-      rtc::CritScope lock(&crit_);
>-
>-      RTPHeader header;
>-      EXPECT_TRUE(parser_->Parse(packet, length, &header));
>-      const uint16_t sequence_number = header.sequenceNumber;
>-      const uint32_t timestamp = header.timestamp;
>-      const uint32_t ssrc = header.ssrc;
>-
>-      if (ssrc == kVideoSendSsrcs[0] || ssrc == kSendRtxSsrcs[0]) {
>-        return SEND_PACKET;
>-      }
>-      EXPECT_EQ(kFlexfecSendSsrc, ssrc) << "Unknown SSRC sent.";
>-
>-      ++num_flexfec_packets_sent_;
>-
>-      // If this is the first packet, we have nothing to compare to.
>-      if (!last_observed_sequence_number_) {
>-        last_observed_sequence_number_.emplace(sequence_number);
>-        last_observed_timestamp_.emplace(timestamp);
>-
>-        return SEND_PACKET;
>-      }
>-
>-      // Verify continuity and monotonicity of RTP sequence numbers.
>-      EXPECT_EQ(static_cast<uint16_t>(*last_observed_sequence_number_ + 1),
>-                sequence_number);
>-      last_observed_sequence_number_.emplace(sequence_number);
>-
>-      // Timestamps should be non-decreasing...
>-      const bool timestamp_is_same_or_newer =
>-          timestamp == *last_observed_timestamp_ ||
>-          IsNewerTimestamp(timestamp, *last_observed_timestamp_);
>-      EXPECT_TRUE(timestamp_is_same_or_newer);
>-      // ...but reasonably close in time.
>-      const int k10SecondsInRtpTimestampBase = 10 * kVideoPayloadTypeFrequency;
>-      EXPECT_TRUE(IsNewerTimestamp(
>-          *last_observed_timestamp_ + k10SecondsInRtpTimestampBase, timestamp));
>-      last_observed_timestamp_.emplace(timestamp);
>-
>-      // Pass test when enough packets have been let through.
>-      if (num_flexfec_packets_sent_ >= 10) {
>-        observation_complete_.Set();
>-      }
>-
>-      return SEND_PACKET;
>-    }
>-
>-    rtc::Optional<uint16_t> last_observed_sequence_number_
>-        RTC_GUARDED_BY(crit_);
>-    rtc::Optional<uint32_t> last_observed_timestamp_ RTC_GUARDED_BY(crit_);
>-    size_t num_flexfec_packets_sent_ RTC_GUARDED_BY(crit_);
>-    rtc::CriticalSection crit_;
>-  } observer;
>-
>-  static constexpr int kFrameMaxWidth = 320;
>-  static constexpr int kFrameMaxHeight = 180;
>-  static constexpr int kFrameRate = 15;
>-
>-  Call::Config config(event_log_.get());
>-
>-  std::unique_ptr<test::PacketTransport> send_transport;
>-  std::unique_ptr<test::PacketTransport> receive_transport;
>-  std::unique_ptr<VideoEncoder> encoder;
>-
>-  task_queue_.SendTask([&]() {
>-    CreateCalls(config, config);
>-
>-    FakeNetworkPipe::Config lossy_delayed_link;
>-    lossy_delayed_link.loss_percent = 2;
>-    lossy_delayed_link.queue_delay_ms = 50;
>-
>-    send_transport = rtc::MakeUnique<test::PacketTransport>(
>-        &task_queue_, sender_call_.get(), &observer,
>-        test::PacketTransport::kSender, payload_type_map_, lossy_delayed_link);
>-    send_transport->SetReceiver(receiver_call_->Receiver());
>-
>-    FakeNetworkPipe::Config flawless_link;
>-    receive_transport = rtc::MakeUnique<test::PacketTransport>(
>-        &task_queue_, nullptr, &observer, test::PacketTransport::kReceiver,
>-        payload_type_map_, flawless_link);
>-    receive_transport->SetReceiver(sender_call_->Receiver());
>-
>-    // For reduced flakyness, we use a real VP8 encoder together with NACK
>-    // and RTX.
>-    const int kNumVideoStreams = 1;
>-    const int kNumFlexfecStreams = 1;
>-    CreateSendConfig(kNumVideoStreams, 0, kNumFlexfecStreams,
>-                     send_transport.get());
>-    encoder = VP8Encoder::Create();
>-    video_send_config_.encoder_settings.encoder = encoder.get();
>-    video_send_config_.encoder_settings.payload_name = "VP8";
>-    video_send_config_.encoder_settings.payload_type = kVideoSendPayloadType;
>-    video_send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
>-    video_send_config_.rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[0]);
>-    video_send_config_.rtp.rtx.payload_type = kSendRtxPayloadType;
>-
>-    CreateMatchingReceiveConfigs(receive_transport.get());
>-    video_receive_configs_[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
>-    video_receive_configs_[0].rtp.rtx_ssrc = kSendRtxSsrcs[0];
>-    video_receive_configs_[0]
>-        .rtp.rtx_associated_payload_types[kSendRtxPayloadType] =
>-        kVideoSendPayloadType;
>-
>-    // The matching FlexFEC receive config is not created by
>-    // CreateMatchingReceiveConfigs since this is not a test::BaseTest.
>-    // Set up the receive config manually instead.
>-    FlexfecReceiveStream::Config flexfec_receive_config(
>-        receive_transport.get());
>-    flexfec_receive_config.payload_type =
>-        video_send_config_.rtp.flexfec.payload_type;
>-    flexfec_receive_config.remote_ssrc = video_send_config_.rtp.flexfec.ssrc;
>-    flexfec_receive_config.protected_media_ssrcs =
>-        video_send_config_.rtp.flexfec.protected_media_ssrcs;
>-    flexfec_receive_config.local_ssrc = kReceiverLocalVideoSsrc;
>-    flexfec_receive_config.transport_cc = true;
>-    flexfec_receive_config.rtp_header_extensions.emplace_back(
>-        RtpExtension::kTransportSequenceNumberUri,
>-        test::kTransportSequenceNumberExtensionId);
>-    flexfec_receive_configs_.push_back(flexfec_receive_config);
>-
>-    CreateFlexfecStreams();
>-    CreateVideoStreams();
>-
>-    // RTCP might be disabled if the network is "down".
>-    sender_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
>-    receiver_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
>-
>-    CreateFrameGeneratorCapturer(kFrameRate, kFrameMaxWidth, kFrameMaxHeight);
>-
>-    Start();
>-  });
>-
>-  // Initial test.
>-  EXPECT_TRUE(observer.Wait()) << "Timed out waiting for packets.";
>-
>-  task_queue_.SendTask([this, &observer]() {
>-    // Ensure monotonicity when the VideoSendStream is restarted.
>-    Stop();
>-    observer.ResetPacketCount();
>-    Start();
>-  });
>-
>-  EXPECT_TRUE(observer.Wait()) << "Timed out waiting for packets.";
>-
>-  task_queue_.SendTask([this, &observer]() {
>-    // Ensure monotonicity when the VideoSendStream is recreated.
>-    frame_generator_capturer_->Stop();
>-    sender_call_->DestroyVideoSendStream(video_send_stream_);
>-    observer.ResetPacketCount();
>-    video_send_stream_ = sender_call_->CreateVideoSendStream(
>-        video_send_config_.Copy(), video_encoder_config_.Copy());
>-    video_send_stream_->Start();
>-    CreateFrameGeneratorCapturer(kFrameRate, kFrameMaxWidth, kFrameMaxHeight);
>-    frame_generator_capturer_->Start();
>-  });
>-
>-  EXPECT_TRUE(observer.Wait()) << "Timed out waiting for packets.";
>-
>-  // Cleanup.
>-  task_queue_.SendTask([this, &send_transport, &receive_transport]() {
>-    Stop();
>-    DestroyStreams();
>-    send_transport.reset();
>-    receive_transport.reset();
>-    DestroyCalls();
>-  });
>-}
>-
>-TEST_P(EndToEndTest, RespectsNetworkState) {
>-  // TODO(pbos): Remove accepted downtime packets etc. when signaling network
>-  // down blocks until no more packets will be sent.
>-
>-  // Pacer will send from its packet list and then send required padding before
>-  // checking paused_ again. This should be enough for one round of pacing,
>-  // otherwise increase.
>-  static const int kNumAcceptedDowntimeRtp = 5;
>-  // A single RTCP may be in the pipeline.
>-  static const int kNumAcceptedDowntimeRtcp = 1;
>-  class NetworkStateTest : public test::EndToEndTest, public test::FakeEncoder {
>-   public:
>-    explicit NetworkStateTest(
>-        test::SingleThreadedTaskQueueForTesting* task_queue)
>-        : EndToEndTest(kDefaultTimeoutMs),
>-          FakeEncoder(Clock::GetRealTimeClock()),
>-          task_queue_(task_queue),
>-          encoded_frames_(false, false),
>-          packet_event_(false, false),
>-          sender_call_(nullptr),
>-          receiver_call_(nullptr),
>-          sender_state_(kNetworkUp),
>-          sender_rtp_(0),
>-          sender_padding_(0),
>-          sender_rtcp_(0),
>-          receiver_rtcp_(0),
>-          down_frames_(0) {}
>-
>-    Action OnSendRtp(const uint8_t* packet, size_t length) override {
>-      rtc::CritScope lock(&test_crit_);
>-      RTPHeader header;
>-      EXPECT_TRUE(parser_->Parse(packet, length, &header));
>-      if (length == header.headerLength + header.paddingLength)
>-        ++sender_padding_;
>-      ++sender_rtp_;
>-      packet_event_.Set();
>-      return SEND_PACKET;
>-    }
>-
>-    Action OnSendRtcp(const uint8_t* packet, size_t length) override {
>-      rtc::CritScope lock(&test_crit_);
>-      ++sender_rtcp_;
>-      packet_event_.Set();
>-      return SEND_PACKET;
>-    }
>-
>-    Action OnReceiveRtp(const uint8_t* packet, size_t length) override {
>-      ADD_FAILURE() << "Unexpected receiver RTP, should not be sending.";
>-      return SEND_PACKET;
>-    }
>-
>-    Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
>-      rtc::CritScope lock(&test_crit_);
>-      ++receiver_rtcp_;
>-      packet_event_.Set();
>-      return SEND_PACKET;
>-    }
>-
>-    void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
>-      sender_call_ = sender_call;
>-      receiver_call_ = receiver_call;
>-    }
>-
>-    void ModifyVideoConfigs(
>-        VideoSendStream::Config* send_config,
>-        std::vector<VideoReceiveStream::Config>* receive_configs,
>-        VideoEncoderConfig* encoder_config) override {
>-      send_config->encoder_settings.encoder = this;
>-    }
>-
>-    void PerformTest() override {
>-      EXPECT_TRUE(encoded_frames_.Wait(kDefaultTimeoutMs))
>-          << "No frames received by the encoder.";
>-
>-      task_queue_->SendTask([this]() {
>-        // Wait for packets from both sender/receiver.
>-        WaitForPacketsOrSilence(false, false);
>-
>-        // Sender-side network down for audio; there should be no effect on
>-        // video
>-        sender_call_->SignalChannelNetworkState(MediaType::AUDIO, kNetworkDown);
>-        WaitForPacketsOrSilence(false, false);
>-
>-        // Receiver-side network down for audio; no change expected
>-        receiver_call_->SignalChannelNetworkState(MediaType::AUDIO,
>-                                                  kNetworkDown);
>-        WaitForPacketsOrSilence(false, false);
>-
>-        // Sender-side network down.
>-        sender_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkDown);
>-        {
>-          rtc::CritScope lock(&test_crit_);
>-          // After network goes down we shouldn't be encoding more frames.
>-          sender_state_ = kNetworkDown;
>-        }
>-        // Wait for receiver-packets and no sender packets.
>-        WaitForPacketsOrSilence(true, false);
>-
>-        // Receiver-side network down.
>-        receiver_call_->SignalChannelNetworkState(MediaType::VIDEO,
>-                                                  kNetworkDown);
>-        WaitForPacketsOrSilence(true, true);
>-
>-        // Network up for audio for both sides; video is still not expected to
>-        // start
>-        sender_call_->SignalChannelNetworkState(MediaType::AUDIO, kNetworkUp);
>-        receiver_call_->SignalChannelNetworkState(MediaType::AUDIO, kNetworkUp);
>-        WaitForPacketsOrSilence(true, true);
>-
>-        // Network back up again for both.
>-        {
>-          rtc::CritScope lock(&test_crit_);
>-          // It's OK to encode frames again, as we're about to bring up the
>-          // network.
>-          sender_state_ = kNetworkUp;
>-        }
>-        sender_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
>-        receiver_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
>-        WaitForPacketsOrSilence(false, false);
>-
>-        // TODO(skvlad): add tests to verify that the audio streams are stopped
>-        // when the network goes down for audio once the workaround in
>-        // paced_sender.cc is removed.
>-      });
>-    }
>-
>-    int32_t Encode(const VideoFrame& input_image,
>-                   const CodecSpecificInfo* codec_specific_info,
>-                   const std::vector<FrameType>* frame_types) override {
>-      {
>-        rtc::CritScope lock(&test_crit_);
>-        if (sender_state_ == kNetworkDown) {
>-          ++down_frames_;
>-          EXPECT_LE(down_frames_, 1)
>-              << "Encoding more than one frame while network is down.";
>-          if (down_frames_ > 1)
>-            encoded_frames_.Set();
>-        } else {
>-          encoded_frames_.Set();
>-        }
>-      }
>-      return test::FakeEncoder::Encode(
>-          input_image, codec_specific_info, frame_types);
>-    }
>-
>-   private:
>-    void WaitForPacketsOrSilence(bool sender_down, bool receiver_down) {
>-      int64_t initial_time_ms = clock_->TimeInMilliseconds();
>-      int initial_sender_rtp;
>-      int initial_sender_rtcp;
>-      int initial_receiver_rtcp;
>-      {
>-        rtc::CritScope lock(&test_crit_);
>-        initial_sender_rtp = sender_rtp_;
>-        initial_sender_rtcp = sender_rtcp_;
>-        initial_receiver_rtcp = receiver_rtcp_;
>-      }
>-      bool sender_done = false;
>-      bool receiver_done = false;
>-      while (!sender_done || !receiver_done) {
>-        packet_event_.Wait(kSilenceTimeoutMs);
>-        int64_t time_now_ms = clock_->TimeInMilliseconds();
>-        rtc::CritScope lock(&test_crit_);
>-        if (sender_down) {
>-          ASSERT_LE(sender_rtp_ - initial_sender_rtp - sender_padding_,
>-                    kNumAcceptedDowntimeRtp)
>-              << "RTP sent during sender-side downtime.";
>-          ASSERT_LE(sender_rtcp_ - initial_sender_rtcp,
>-                    kNumAcceptedDowntimeRtcp)
>-              << "RTCP sent during sender-side downtime.";
>-          if (time_now_ms - initial_time_ms >=
>-              static_cast<int64_t>(kSilenceTimeoutMs)) {
>-            sender_done = true;
>-          }
>-        } else {
>-          if (sender_rtp_ > initial_sender_rtp + kNumAcceptedDowntimeRtp)
>-            sender_done = true;
>-        }
>-        if (receiver_down) {
>-          ASSERT_LE(receiver_rtcp_ - initial_receiver_rtcp,
>-                    kNumAcceptedDowntimeRtcp)
>-              << "RTCP sent during receiver-side downtime.";
>-          if (time_now_ms - initial_time_ms >=
>-              static_cast<int64_t>(kSilenceTimeoutMs)) {
>-            receiver_done = true;
>-          }
>-        } else {
>-          if (receiver_rtcp_ > initial_receiver_rtcp + kNumAcceptedDowntimeRtcp)
>-            receiver_done = true;
>-        }
>-      }
>-    }
>-
>-    test::SingleThreadedTaskQueueForTesting* const task_queue_;
>-    rtc::CriticalSection test_crit_;
>-    rtc::Event encoded_frames_;
>-    rtc::Event packet_event_;
>-    Call* sender_call_;
>-    Call* receiver_call_;
>-    NetworkState sender_state_ RTC_GUARDED_BY(test_crit_);
>-    int sender_rtp_ RTC_GUARDED_BY(test_crit_);
>-    int sender_padding_ RTC_GUARDED_BY(test_crit_);
>-    int sender_rtcp_ RTC_GUARDED_BY(test_crit_);
>-    int receiver_rtcp_ RTC_GUARDED_BY(test_crit_);
>-    int down_frames_ RTC_GUARDED_BY(test_crit_);
>-  } test(&task_queue_);
>-
>-  RunBaseTest(&test);
>-}
>-
>-TEST_P(EndToEndTest, CallReportsRttForSender) {
>-  static const int kSendDelayMs = 30;
>-  static const int kReceiveDelayMs = 70;
>-
>-  std::unique_ptr<test::DirectTransport> sender_transport;
>-  std::unique_ptr<test::DirectTransport> receiver_transport;
>-
>-  task_queue_.SendTask([this, &sender_transport, &receiver_transport]() {
>-    FakeNetworkPipe::Config config;
>-    config.queue_delay_ms = kSendDelayMs;
>-    CreateCalls(Call::Config(event_log_.get()), Call::Config(event_log_.get()));
>-    sender_transport = rtc::MakeUnique<test::DirectTransport>(
>-        &task_queue_, config, sender_call_.get(), payload_type_map_);
>-    config.queue_delay_ms = kReceiveDelayMs;
>-    receiver_transport = rtc::MakeUnique<test::DirectTransport>(
>-        &task_queue_, config, receiver_call_.get(), payload_type_map_);
>-    sender_transport->SetReceiver(receiver_call_->Receiver());
>-    receiver_transport->SetReceiver(sender_call_->Receiver());
>-
>-    CreateSendConfig(1, 0, 0, sender_transport.get());
>-    CreateMatchingReceiveConfigs(receiver_transport.get());
>-
>-    CreateVideoStreams();
>-    CreateFrameGeneratorCapturer(kDefaultFramerate, kDefaultWidth,
>-                                 kDefaultHeight);
>-    Start();
>-  });
>-
>-  int64_t start_time_ms = clock_->TimeInMilliseconds();
>-  while (true) {
>-    Call::Stats stats = sender_call_->GetStats();
>-    ASSERT_GE(start_time_ms + kDefaultTimeoutMs,
>-              clock_->TimeInMilliseconds())
>-        << "No RTT stats before timeout!";
>-    if (stats.rtt_ms != -1) {
>-      // To avoid failures caused by rounding or minor ntp clock adjustments,
>-      // relax expectation by 1ms.
>-      constexpr int kAllowedErrorMs = 1;
>-      EXPECT_GE(stats.rtt_ms, kSendDelayMs + kReceiveDelayMs - kAllowedErrorMs);
>-      break;
>-    }
>-    SleepMs(10);
>-  }
>-
>-  task_queue_.SendTask([this, &sender_transport, &receiver_transport]() {
>-    Stop();
>-    DestroyStreams();
>-    sender_transport.reset();
>-    receiver_transport.reset();
>-    DestroyCalls();
>-  });
>-}
>-
>-void EndToEndTest::VerifyNewVideoSendStreamsRespectNetworkState(
>-    MediaType network_to_bring_up,
>-    VideoEncoder* encoder,
>-    Transport* transport) {
>-  task_queue_.SendTask([this, network_to_bring_up, encoder, transport]() {
>-    CreateSenderCall(Call::Config(event_log_.get()));
>-    sender_call_->SignalChannelNetworkState(network_to_bring_up, kNetworkUp);
>-
>-    CreateSendConfig(1, 0, 0, transport);
>-    video_send_config_.encoder_settings.encoder = encoder;
>-    CreateVideoStreams();
>-    CreateFrameGeneratorCapturer(kDefaultFramerate, kDefaultWidth,
>-                                 kDefaultHeight);
>-
>-    Start();
>-  });
>-
>-  SleepMs(kSilenceTimeoutMs);
>-
>-  task_queue_.SendTask([this]() {
>-    Stop();
>-    DestroyStreams();
>-    DestroyCalls();
>-  });
>-}
>-
>-void EndToEndTest::VerifyNewVideoReceiveStreamsRespectNetworkState(
>-    MediaType network_to_bring_up,
>-    Transport* transport) {
>-  std::unique_ptr<test::DirectTransport> sender_transport;
>-
>-  task_queue_.SendTask([this, &sender_transport, network_to_bring_up,
>-                        transport]() {
>-    Call::Config config(event_log_.get());
>-    CreateCalls(config, config);
>-    receiver_call_->SignalChannelNetworkState(network_to_bring_up, kNetworkUp);
>-    sender_transport = rtc::MakeUnique<test::DirectTransport>(
>-        &task_queue_, sender_call_.get(), payload_type_map_);
>-    sender_transport->SetReceiver(receiver_call_->Receiver());
>-    CreateSendConfig(1, 0, 0, sender_transport.get());
>-    CreateMatchingReceiveConfigs(transport);
>-    CreateVideoStreams();
>-    CreateFrameGeneratorCapturer(kDefaultFramerate, kDefaultWidth,
>-                                 kDefaultHeight);
>-    Start();
>-  });
>-
>-  SleepMs(kSilenceTimeoutMs);
>-
>-  task_queue_.SendTask([this, &sender_transport]() {
>-    Stop();
>-    DestroyStreams();
>-    sender_transport.reset();
>-    DestroyCalls();
>-  });
>-}
>-
>-TEST_P(EndToEndTest, NewVideoSendStreamsRespectVideoNetworkDown) {
>-  class UnusedEncoder : public test::FakeEncoder {
>-   public:
>-    UnusedEncoder() : FakeEncoder(Clock::GetRealTimeClock()) {}
>-
>-    int32_t InitEncode(const VideoCodec* config,
>-                       int32_t number_of_cores,
>-                       size_t max_payload_size) override {
>-      EXPECT_GT(config->startBitrate, 0u);
>-      return 0;
>-    }
>-    int32_t Encode(const VideoFrame& input_image,
>-                   const CodecSpecificInfo* codec_specific_info,
>-                   const std::vector<FrameType>* frame_types) override {
>-      ADD_FAILURE() << "Unexpected frame encode.";
>-      return test::FakeEncoder::Encode(input_image, codec_specific_info,
>-                                       frame_types);
>-    }
>-  };
>-
>-  UnusedEncoder unused_encoder;
>-  UnusedTransport unused_transport;
>-  VerifyNewVideoSendStreamsRespectNetworkState(
>-      MediaType::AUDIO, &unused_encoder, &unused_transport);
>-}
>-
>-TEST_P(EndToEndTest, NewVideoSendStreamsIgnoreAudioNetworkDown) {
>-  class RequiredEncoder : public test::FakeEncoder {
>-   public:
>-    RequiredEncoder()
>-        : FakeEncoder(Clock::GetRealTimeClock()), encoded_frame_(false) {}
>-    ~RequiredEncoder() {
>-      if (!encoded_frame_) {
>-        ADD_FAILURE() << "Didn't encode an expected frame";
>-      }
>-    }
>-    int32_t Encode(const VideoFrame& input_image,
>-                   const CodecSpecificInfo* codec_specific_info,
>-                   const std::vector<FrameType>* frame_types) override {
>-      encoded_frame_ = true;
>-      return test::FakeEncoder::Encode(input_image, codec_specific_info,
>-                                       frame_types);
>-    }
>-
>-   private:
>-    bool encoded_frame_;
>-  };
>-
>-  RequiredTransport required_transport(true /*rtp*/, false /*rtcp*/);
>-  RequiredEncoder required_encoder;
>-  VerifyNewVideoSendStreamsRespectNetworkState(
>-      MediaType::VIDEO, &required_encoder, &required_transport);
>-}
>-
>-TEST_P(EndToEndTest, NewVideoReceiveStreamsRespectVideoNetworkDown) {
>-  UnusedTransport transport;
>-  VerifyNewVideoReceiveStreamsRespectNetworkState(MediaType::AUDIO, &transport);
>-}
>-
>-TEST_P(EndToEndTest, NewVideoReceiveStreamsIgnoreAudioNetworkDown) {
>-  RequiredTransport transport(false /*rtp*/, true /*rtcp*/);
>-  VerifyNewVideoReceiveStreamsRespectNetworkState(MediaType::VIDEO, &transport);
>-}
>-
>-void VerifyEmptyNackConfig(const NackConfig& config) {
>-  EXPECT_EQ(0, config.rtp_history_ms)
>-      << "Enabling NACK requires rtcp-fb: nack negotiation.";
>-}
>-
>-void VerifyEmptyUlpfecConfig(const UlpfecConfig& config) {
>-  EXPECT_EQ(-1, config.ulpfec_payload_type)
>-      << "Enabling ULPFEC requires rtpmap: ulpfec negotiation.";
>-  EXPECT_EQ(-1, config.red_payload_type)
>-      << "Enabling ULPFEC requires rtpmap: red negotiation.";
>-  EXPECT_EQ(-1, config.red_rtx_payload_type)
>-      << "Enabling RTX in ULPFEC requires rtpmap: rtx negotiation.";
>-}
>-
>-void VerifyEmptyFlexfecConfig(
>-    const VideoSendStream::Config::Rtp::Flexfec& config) {
>-  EXPECT_EQ(-1, config.payload_type)
>-      << "Enabling FlexFEC requires rtpmap: flexfec negotiation.";
>-  EXPECT_EQ(0U, config.ssrc)
>-      << "Enabling FlexFEC requires ssrc-group: FEC-FR negotiation.";
>-  EXPECT_TRUE(config.protected_media_ssrcs.empty())
>-      << "Enabling FlexFEC requires ssrc-group: FEC-FR negotiation.";
>-}
>-
>-TEST_P(EndToEndTest, VerifyDefaultSendConfigParameters) {
>-  VideoSendStream::Config default_send_config(nullptr);
>-  EXPECT_EQ(0, default_send_config.rtp.nack.rtp_history_ms)
>-      << "Enabling NACK require rtcp-fb: nack negotiation.";
>-  EXPECT_TRUE(default_send_config.rtp.rtx.ssrcs.empty())
>-      << "Enabling RTX requires rtpmap: rtx negotiation.";
>-  EXPECT_TRUE(default_send_config.rtp.extensions.empty())
>-      << "Enabling RTP extensions require negotiation.";
>-
>-  VerifyEmptyNackConfig(default_send_config.rtp.nack);
>-  VerifyEmptyUlpfecConfig(default_send_config.rtp.ulpfec);
>-  VerifyEmptyFlexfecConfig(default_send_config.rtp.flexfec);
>-}
>-
>-TEST_P(EndToEndTest, VerifyDefaultVideoReceiveConfigParameters) {
>-  VideoReceiveStream::Config default_receive_config(nullptr);
>-  EXPECT_EQ(RtcpMode::kCompound, default_receive_config.rtp.rtcp_mode)
>-      << "Reduced-size RTCP require rtcp-rsize to be negotiated.";
>-  EXPECT_FALSE(default_receive_config.rtp.remb)
>-      << "REMB require rtcp-fb: goog-remb to be negotiated.";
>-  EXPECT_FALSE(
>-      default_receive_config.rtp.rtcp_xr.receiver_reference_time_report)
>-      << "RTCP XR settings require rtcp-xr to be negotiated.";
>-  EXPECT_EQ(0U, default_receive_config.rtp.rtx_ssrc)
>-      << "Enabling RTX requires ssrc-group: FID negotiation";
>-  EXPECT_TRUE(default_receive_config.rtp.rtx_associated_payload_types.empty())
>-      << "Enabling RTX requires rtpmap: rtx negotiation.";
>-  EXPECT_TRUE(default_receive_config.rtp.extensions.empty())
>-      << "Enabling RTP extensions require negotiation.";
>-
>-  VerifyEmptyNackConfig(default_receive_config.rtp.nack);
>-  EXPECT_EQ(-1, default_receive_config.rtp.ulpfec_payload_type)
>-      << "Enabling ULPFEC requires rtpmap: ulpfec negotiation.";
>-  EXPECT_EQ(-1, default_receive_config.rtp.red_payload_type)
>-      << "Enabling ULPFEC requires rtpmap: red negotiation.";
>-}
>-
>-TEST_P(EndToEndTest, VerifyDefaultFlexfecReceiveConfigParameters) {
>-  test::NullTransport rtcp_send_transport;
>-  FlexfecReceiveStream::Config default_receive_config(&rtcp_send_transport);
>-  EXPECT_EQ(-1, default_receive_config.payload_type)
>-      << "Enabling FlexFEC requires rtpmap: flexfec negotiation.";
>-  EXPECT_EQ(0U, default_receive_config.remote_ssrc)
>-      << "Enabling FlexFEC requires ssrc-group: FEC-FR negotiation.";
>-  EXPECT_TRUE(default_receive_config.protected_media_ssrcs.empty())
>-      << "Enabling FlexFEC requires ssrc-group: FEC-FR negotiation.";
>-}
>-
>-TEST_P(EndToEndTest, TransportSeqNumOnAudioAndVideo) {
>-  static constexpr int kExtensionId = 8;
>-  static constexpr size_t kMinPacketsToWaitFor = 50;
>-  class TransportSequenceNumberTest : public test::EndToEndTest {
>-   public:
>-    TransportSequenceNumberTest()
>-        : EndToEndTest(kDefaultTimeoutMs),
>-          video_observed_(false),
>-          audio_observed_(false) {
>-      parser_->RegisterRtpHeaderExtension(kRtpExtensionTransportSequenceNumber,
>-                                          kExtensionId);
>-    }
>-
>-    size_t GetNumVideoStreams() const override { return 1; }
>-    size_t GetNumAudioStreams() const override { return 1; }
>-
>-    void ModifyAudioConfigs(
>-        AudioSendStream::Config* send_config,
>-        std::vector<AudioReceiveStream::Config>* receive_configs) override {
>-      send_config->rtp.extensions.clear();
>-      send_config->rtp.extensions.push_back(RtpExtension(
>-          RtpExtension::kTransportSequenceNumberUri, kExtensionId));
>-      (*receive_configs)[0].rtp.extensions.clear();
>-      (*receive_configs)[0].rtp.extensions = send_config->rtp.extensions;
>-    }
>-
>-    Action OnSendRtp(const uint8_t* packet, size_t length) override {
>-      RTPHeader header;
>-      EXPECT_TRUE(parser_->Parse(packet, length, &header));
>-      EXPECT_TRUE(header.extension.hasTransportSequenceNumber);
>-      // Unwrap packet id and verify uniqueness.
>-      int64_t packet_id =
>-          unwrapper_.Unwrap(header.extension.transportSequenceNumber);
>-      EXPECT_TRUE(received_packet_ids_.insert(packet_id).second);
>-
>-      if (header.ssrc == kVideoSendSsrcs[0])
>-        video_observed_ = true;
>-      if (header.ssrc == kAudioSendSsrc)
>-        audio_observed_ = true;
>-      if (audio_observed_ && video_observed_ &&
>-          received_packet_ids_.size() >= kMinPacketsToWaitFor) {
>-        size_t packet_id_range =
>-            *received_packet_ids_.rbegin() - *received_packet_ids_.begin() + 1;
>-        EXPECT_EQ(received_packet_ids_.size(), packet_id_range);
>-        observation_complete_.Set();
>-      }
>-      return SEND_PACKET;
>-    }
>-
>-    void PerformTest() override {
>-      EXPECT_TRUE(Wait()) << "Timed out while waiting for audio and video "
>-                             "packets with transport sequence number.";
>-    }
>-
>-    void ExpectSuccessful() {
>-      EXPECT_TRUE(video_observed_);
>-      EXPECT_TRUE(audio_observed_);
>-      EXPECT_GE(received_packet_ids_.size(), kMinPacketsToWaitFor);
>-    }
>-
>-   private:
>-    bool video_observed_;
>-    bool audio_observed_;
>-    SequenceNumberUnwrapper unwrapper_;
>-    std::set<int64_t> received_packet_ids_;
>-  } test;
>-
>-  RunBaseTest(&test);
>-  // Double check conditions for successful test to produce better error
>-  // message when the test fail.
>-  test.ExpectSuccessful();
>-}
>-
>-class EndToEndLogTest : public EndToEndTest {
>-  void SetUp() { paths_.clear(); }
>-  void TearDown() {
>-    for (const auto& path : paths_) {
>-      rtc::RemoveFile(path);
>-    }
>-  }
>-
>- public:
>-  int AddFile() {
>-    paths_.push_back(test::TempFilename(test::OutputPath(), "test_file"));
>-    return static_cast<int>(paths_.size()) - 1;
>-  }
>-
>-  rtc::PlatformFile OpenFile(int idx) {
>-    return rtc::OpenPlatformFile(paths_[idx]);
>-  }
>-
>-  void LogSend(bool open) {
>-    if (open) {
>-      video_send_stream_->EnableEncodedFrameRecording(
>-          std::vector<rtc::PlatformFile>(1, OpenFile(AddFile())), 0);
>-    } else {
>-      video_send_stream_->DisableEncodedFrameRecording();
>-    }
>-  }
>-  void LogReceive(bool open) {
>-    if (open) {
>-      video_receive_streams_[0]->EnableEncodedFrameRecording(
>-          OpenFile(AddFile()), 0);
>-    } else {
>-      video_receive_streams_[0]->DisableEncodedFrameRecording();
>-    }
>-  }
>-
>-  std::vector<std::string> paths_;
>-};
>-
>-TEST_P(EndToEndLogTest, LogsEncodedFramesWhenRequested) {
>-  static const int kNumFramesToRecord = 10;
>-  class LogEncodingObserver : public test::EndToEndTest,
>-                              public EncodedFrameObserver {
>-   public:
>-    explicit LogEncodingObserver(EndToEndLogTest* fixture)
>-        : EndToEndTest(kDefaultTimeoutMs),
>-          fixture_(fixture),
>-          recorded_frames_(0) {}
>-
>-    void PerformTest() override {
>-      fixture_->LogSend(true);
>-      fixture_->LogReceive(true);
>-      ASSERT_TRUE(Wait()) << "Timed out while waiting for frame logging.";
>-    }
>-
>-    void ModifyVideoConfigs(
>-        VideoSendStream::Config* send_config,
>-        std::vector<VideoReceiveStream::Config>* receive_configs,
>-        VideoEncoderConfig* encoder_config) override {
>-      encoder_ = VP8Encoder::Create();
>-      decoder_ = VP8Decoder::Create();
>-
>-      send_config->post_encode_callback = this;
>-      send_config->encoder_settings.payload_name = "VP8";
>-      send_config->encoder_settings.encoder = encoder_.get();
>-
>-      (*receive_configs)[0].decoders.resize(1);
>-      (*receive_configs)[0].decoders[0].payload_type =
>-          send_config->encoder_settings.payload_type;
>-      (*receive_configs)[0].decoders[0].payload_name =
>-          send_config->encoder_settings.payload_name;
>-      (*receive_configs)[0].decoders[0].decoder = decoder_.get();
>-    }
>-
>-    void EncodedFrameCallback(const EncodedFrame& encoded_frame) override {
>-      rtc::CritScope lock(&crit_);
>-      if (recorded_frames_++ > kNumFramesToRecord) {
>-        fixture_->LogSend(false);
>-        fixture_->LogReceive(false);
>-        rtc::File send_file(fixture_->OpenFile(0));
>-        rtc::File receive_file(fixture_->OpenFile(1));
>-        uint8_t out[100];
>-        // If logging has worked correctly neither file should be empty, i.e.
>-        // we should be able to read something from them.
>-        EXPECT_LT(0u, send_file.Read(out, 100));
>-        EXPECT_LT(0u, receive_file.Read(out, 100));
>-        observation_complete_.Set();
>-      }
>-    }
>-
>-   private:
>-    EndToEndLogTest* const fixture_;
>-    std::unique_ptr<VideoEncoder> encoder_;
>-    std::unique_ptr<VideoDecoder> decoder_;
>-    rtc::CriticalSection crit_;
>-    int recorded_frames_ RTC_GUARDED_BY(crit_);
>-  } test(this);
>-
>-  RunBaseTest(&test);
>-}
>-
>-INSTANTIATE_TEST_CASE_P(RoundRobin,
>-                        EndToEndTest,
>-                        ::testing::Values("WebRTC-RoundRobinPacing/Disabled/",
>-                                          "WebRTC-RoundRobinPacing/Enabled/"));
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/bandwidth_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/bandwidth_tests.cc
>new file mode 100644
>index 00000000000..3b1df3c5903
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/bandwidth_tests.cc
>@@ -0,0 +1,355 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/rtp_rtcp/include/rtp_rtcp.h"
>+#include "rtc_base/rate_limiter.h"
>+#include "system_wrappers/include/sleep.h"
>+#include "test/call_test.h"
>+#include "test/encoder_proxy_factory.h"
>+#include "test/fake_encoder.h"
>+#include "test/field_trial.h"
>+#include "test/gtest.h"
>+#include "test/rtcp_packet_parser.h"
>+#include "test/rtp_rtcp_observer.h"
>+
>+namespace webrtc {
>+
>+class BandwidthEndToEndTest : public test::CallTest,
>+                              public testing::WithParamInterface<std::string> {
>+ public:
>+  BandwidthEndToEndTest() : field_trial_(GetParam()) {}
>+
>+ private:
>+  test::ScopedFieldTrials field_trial_;
>+};
>+
>+INSTANTIATE_TEST_CASE_P(RoundRobin,
>+                        BandwidthEndToEndTest,
>+                        ::testing::Values("WebRTC-RoundRobinPacing/Disabled/",
>+                                          "WebRTC-RoundRobinPacing/Enabled/"));
>+
>+TEST_P(BandwidthEndToEndTest, ReceiveStreamSendsRemb) {
>+  class RembObserver : public test::EndToEndTest {
>+   public:
>+    RembObserver() : EndToEndTest(kDefaultTimeoutMs) {}
>+
>+    void ModifyVideoConfigs(
>+        VideoSendStream::Config* send_config,
>+        std::vector<VideoReceiveStream::Config>* receive_configs,
>+        VideoEncoderConfig* encoder_config) override {
>+      send_config->rtp.extensions.clear();
>+      send_config->rtp.extensions.push_back(RtpExtension(
>+          RtpExtension::kAbsSendTimeUri, test::kAbsSendTimeExtensionId));
>+      (*receive_configs)[0].rtp.remb = true;
>+      (*receive_configs)[0].rtp.transport_cc = false;
>+    }
>+
>+    Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
>+      test::RtcpPacketParser parser;
>+      EXPECT_TRUE(parser.Parse(packet, length));
>+
>+      if (parser.remb()->num_packets() > 0) {
>+        EXPECT_EQ(kReceiverLocalVideoSsrc, parser.remb()->sender_ssrc());
>+        EXPECT_LT(0U, parser.remb()->bitrate_bps());
>+        EXPECT_EQ(1U, parser.remb()->ssrcs().size());
>+        EXPECT_EQ(kVideoSendSsrcs[0], parser.remb()->ssrcs()[0]);
>+        observation_complete_.Set();
>+      }
>+
>+      return SEND_PACKET;
>+    }
>+    void PerformTest() override {
>+      EXPECT_TRUE(Wait()) << "Timed out while waiting for a "
>+                             "receiver RTCP REMB packet to be "
>+                             "sent.";
>+    }
>+  } test;
>+
>+  RunBaseTest(&test);
>+}
>+
>+class BandwidthStatsTest : public test::EndToEndTest {
>+ public:
>+  explicit BandwidthStatsTest(bool send_side_bwe)
>+      : EndToEndTest(test::CallTest::kDefaultTimeoutMs),
>+        sender_call_(nullptr),
>+        receiver_call_(nullptr),
>+        has_seen_pacer_delay_(false),
>+        send_side_bwe_(send_side_bwe) {}
>+
>+  void ModifyVideoConfigs(
>+      VideoSendStream::Config* send_config,
>+      std::vector<VideoReceiveStream::Config>* receive_configs,
>+      VideoEncoderConfig* encoder_config) override {
>+    if (!send_side_bwe_) {
>+      send_config->rtp.extensions.clear();
>+      send_config->rtp.extensions.push_back(RtpExtension(
>+          RtpExtension::kAbsSendTimeUri, test::kAbsSendTimeExtensionId));
>+      (*receive_configs)[0].rtp.remb = true;
>+      (*receive_configs)[0].rtp.transport_cc = false;
>+    }
>+  }
>+
>+  Action OnSendRtp(const uint8_t* packet, size_t length) override {
>+    Call::Stats sender_stats = sender_call_->GetStats();
>+    Call::Stats receiver_stats = receiver_call_->GetStats();
>+    if (!has_seen_pacer_delay_)
>+      has_seen_pacer_delay_ = sender_stats.pacer_delay_ms > 0;
>+    if (sender_stats.send_bandwidth_bps > 0 && has_seen_pacer_delay_) {
>+      if (send_side_bwe_ || receiver_stats.recv_bandwidth_bps > 0)
>+        observation_complete_.Set();
>+    }
>+    return SEND_PACKET;
>+  }
>+
>+  void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
>+    sender_call_ = sender_call;
>+    receiver_call_ = receiver_call;
>+  }
>+
>+  void PerformTest() override {
>+    EXPECT_TRUE(Wait()) << "Timed out while waiting for "
>+                           "non-zero bandwidth stats.";
>+  }
>+
>+ private:
>+  Call* sender_call_;
>+  Call* receiver_call_;
>+  bool has_seen_pacer_delay_;
>+  const bool send_side_bwe_;
>+};
>+
>+TEST_P(BandwidthEndToEndTest, VerifySendSideBweStats) {
>+  BandwidthStatsTest test(true);
>+  RunBaseTest(&test);
>+}
>+
>+TEST_P(BandwidthEndToEndTest, VerifyRecvSideBweStats) {
>+  BandwidthStatsTest test(false);
>+  RunBaseTest(&test);
>+}
>+
>+// Verifies that it's possible to limit the send BWE by sending a REMB.
>+// This is verified by allowing the send BWE to ramp-up to >1000 kbps,
>+// then have the test generate a REMB of 500 kbps and verify that the send BWE
>+// is reduced to exactly 500 kbps. Then a REMB of 1000 kbps is generated and the
>+// test verifies that the send BWE ramps back up to exactly 1000 kbps.
>+TEST_P(BandwidthEndToEndTest, RembWithSendSideBwe) {
>+  class BweObserver : public test::EndToEndTest {
>+   public:
>+    BweObserver()
>+        : EndToEndTest(kDefaultTimeoutMs),
>+          sender_call_(nullptr),
>+          clock_(Clock::GetRealTimeClock()),
>+          sender_ssrc_(0),
>+          remb_bitrate_bps_(1000000),
>+          receive_transport_(nullptr),
>+          stop_event_(false, false),
>+          poller_thread_(&BitrateStatsPollingThread,
>+                         this,
>+                         "BitrateStatsPollingThread"),
>+          state_(kWaitForFirstRampUp),
>+          retransmission_rate_limiter_(clock_, 1000) {}
>+
>+    ~BweObserver() {}
>+
>+    test::PacketTransport* CreateReceiveTransport(
>+        test::SingleThreadedTaskQueueForTesting* task_queue) override {
>+      receive_transport_ = new test::PacketTransport(
>+          task_queue, nullptr, this, test::PacketTransport::kReceiver,
>+          payload_type_map_, FakeNetworkPipe::Config());
>+      return receive_transport_;
>+    }
>+
>+    void ModifySenderCallConfig(Call::Config* config) override {
>+      // Set a high start bitrate to reduce the test completion time.
>+      config->bitrate_config.start_bitrate_bps = remb_bitrate_bps_;
>+    }
>+
>+    void ModifyVideoConfigs(
>+        VideoSendStream::Config* send_config,
>+        std::vector<VideoReceiveStream::Config>* receive_configs,
>+        VideoEncoderConfig* encoder_config) override {
>+      ASSERT_EQ(1u, send_config->rtp.ssrcs.size());
>+      sender_ssrc_ = send_config->rtp.ssrcs[0];
>+
>+      encoder_config->max_bitrate_bps = 2000000;
>+
>+      ASSERT_EQ(1u, receive_configs->size());
>+      RtpRtcp::Configuration config;
>+      config.receiver_only = true;
>+      config.clock = clock_;
>+      config.outgoing_transport = receive_transport_;
>+      config.retransmission_rate_limiter = &retransmission_rate_limiter_;
>+      rtp_rtcp_.reset(RtpRtcp::CreateRtpRtcp(config));
>+      rtp_rtcp_->SetRemoteSSRC((*receive_configs)[0].rtp.remote_ssrc);
>+      rtp_rtcp_->SetSSRC((*receive_configs)[0].rtp.local_ssrc);
>+      rtp_rtcp_->SetRTCPStatus(RtcpMode::kReducedSize);
>+    }
>+
>+    void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
>+      sender_call_ = sender_call;
>+    }
>+
>+    static void BitrateStatsPollingThread(void* obj) {
>+      static_cast<BweObserver*>(obj)->PollStats();
>+    }
>+
>+    void PollStats() {
>+      do {
>+        if (sender_call_) {
>+          Call::Stats stats = sender_call_->GetStats();
>+          switch (state_) {
>+            case kWaitForFirstRampUp:
>+              if (stats.send_bandwidth_bps >= remb_bitrate_bps_) {
>+                state_ = kWaitForRemb;
>+                remb_bitrate_bps_ /= 2;
>+                rtp_rtcp_->SetRemb(
>+                    remb_bitrate_bps_,
>+                    std::vector<uint32_t>(&sender_ssrc_, &sender_ssrc_ + 1));
>+                rtp_rtcp_->SendRTCP(kRtcpRr);
>+              }
>+              break;
>+
>+            case kWaitForRemb:
>+              if (stats.send_bandwidth_bps == remb_bitrate_bps_) {
>+                state_ = kWaitForSecondRampUp;
>+                remb_bitrate_bps_ *= 2;
>+                rtp_rtcp_->SetRemb(
>+                    remb_bitrate_bps_,
>+                    std::vector<uint32_t>(&sender_ssrc_, &sender_ssrc_ + 1));
>+                rtp_rtcp_->SendRTCP(kRtcpRr);
>+              }
>+              break;
>+
>+            case kWaitForSecondRampUp:
>+              if (stats.send_bandwidth_bps == remb_bitrate_bps_) {
>+                observation_complete_.Set();
>+              }
>+              break;
>+          }
>+        }
>+      } while (!stop_event_.Wait(1000));
>+    }
>+
>+    void PerformTest() override {
>+      poller_thread_.Start();
>+      EXPECT_TRUE(Wait())
>+          << "Timed out while waiting for bitrate to change according to REMB.";
>+      stop_event_.Set();
>+      poller_thread_.Stop();
>+    }
>+
>+   private:
>+    enum TestState { kWaitForFirstRampUp, kWaitForRemb, kWaitForSecondRampUp };
>+
>+    Call* sender_call_;
>+    Clock* const clock_;
>+    uint32_t sender_ssrc_;
>+    int remb_bitrate_bps_;
>+    std::unique_ptr<RtpRtcp> rtp_rtcp_;
>+    test::PacketTransport* receive_transport_;
>+    rtc::Event stop_event_;
>+    rtc::PlatformThread poller_thread_;
>+    TestState state_;
>+    RateLimiter retransmission_rate_limiter_;
>+  } test;
>+
>+  RunBaseTest(&test);
>+}
>+
>+TEST_P(BandwidthEndToEndTest, ReportsSetEncoderRates) {
>+  class EncoderRateStatsTest : public test::EndToEndTest,
>+                               public test::FakeEncoder {
>+   public:
>+    explicit EncoderRateStatsTest(
>+        test::SingleThreadedTaskQueueForTesting* task_queue)
>+        : EndToEndTest(kDefaultTimeoutMs),
>+          FakeEncoder(Clock::GetRealTimeClock()),
>+          task_queue_(task_queue),
>+          send_stream_(nullptr),
>+          encoder_factory_(this),
>+          bitrate_kbps_(0) {}
>+
>+    void OnVideoStreamsCreated(
>+        VideoSendStream* send_stream,
>+        const std::vector<VideoReceiveStream*>& receive_streams) override {
>+      send_stream_ = send_stream;
>+    }
>+
>+    void ModifyVideoConfigs(
>+        VideoSendStream::Config* send_config,
>+        std::vector<VideoReceiveStream::Config>* receive_configs,
>+        VideoEncoderConfig* encoder_config) override {
>+      send_config->encoder_settings.encoder_factory = &encoder_factory_;
>+      RTC_DCHECK_EQ(1, encoder_config->number_of_streams);
>+    }
>+
>+    int32_t SetRateAllocation(const VideoBitrateAllocation& rate_allocation,
>+                              uint32_t framerate) override {
>+      // Make sure not to trigger on any default zero bitrates.
>+      if (rate_allocation.get_sum_bps() == 0)
>+        return 0;
>+      rtc::CritScope lock(&crit_);
>+      bitrate_kbps_ = rate_allocation.get_sum_kbps();
>+      observation_complete_.Set();
>+      return 0;
>+    }
>+
>+    void PerformTest() override {
>+      ASSERT_TRUE(Wait())
>+          << "Timed out while waiting for encoder SetRates() call.";
>+
>+      task_queue_->SendTask([this]() {
>+        WaitForEncoderTargetBitrateMatchStats();
>+        send_stream_->Stop();
>+        WaitForStatsReportZeroTargetBitrate();
>+        send_stream_->Start();
>+        WaitForEncoderTargetBitrateMatchStats();
>+      });
>+    }
>+
>+    void WaitForEncoderTargetBitrateMatchStats() {
>+      for (int i = 0; i < kDefaultTimeoutMs; ++i) {
>+        VideoSendStream::Stats stats = send_stream_->GetStats();
>+        {
>+          rtc::CritScope lock(&crit_);
>+          if ((stats.target_media_bitrate_bps + 500) / 1000 ==
>+              static_cast<int>(bitrate_kbps_)) {
>+            return;
>+          }
>+        }
>+        SleepMs(1);
>+      }
>+      FAIL()
>+          << "Timed out waiting for stats reporting the currently set bitrate.";
>+    }
>+
>+    void WaitForStatsReportZeroTargetBitrate() {
>+      for (int i = 0; i < kDefaultTimeoutMs; ++i) {
>+        if (send_stream_->GetStats().target_media_bitrate_bps == 0) {
>+          return;
>+        }
>+        SleepMs(1);
>+      }
>+      FAIL() << "Timed out waiting for stats reporting zero bitrate.";
>+    }
>+
>+   private:
>+    test::SingleThreadedTaskQueueForTesting* const task_queue_;
>+    rtc::CriticalSection crit_;
>+    VideoSendStream* send_stream_;
>+    test::EncoderProxyFactory encoder_factory_;
>+    uint32_t bitrate_kbps_ RTC_GUARDED_BY(crit_);
>+  } test(&task_queue_);
>+
>+  RunBaseTest(&test);
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/call_operation_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/call_operation_tests.cc
>new file mode 100644
>index 00000000000..1861058fbb5
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/call_operation_tests.cc
>@@ -0,0 +1,273 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "system_wrappers/include/sleep.h"
>+#include "test/call_test.h"
>+#include "test/field_trial.h"
>+#include "test/frame_generator.h"
>+#include "test/gtest.h"
>+#include "test/null_transport.h"
>+
>+namespace webrtc {
>+
>+class CallOperationEndToEndTest
>+    : public test::CallTest,
>+      public testing::WithParamInterface<std::string> {
>+ public:
>+  CallOperationEndToEndTest() : field_trial_(GetParam()) {}
>+
>+ private:
>+  test::ScopedFieldTrials field_trial_;
>+};
>+
>+INSTANTIATE_TEST_CASE_P(
>+    FieldTrials,
>+    CallOperationEndToEndTest,
>+    ::testing::Values("WebRTC-RoundRobinPacing/Disabled/",
>+                      "WebRTC-RoundRobinPacing/Enabled/",
>+                      "WebRTC-TaskQueueCongestionControl/Enabled/"));
>+
>+TEST_P(CallOperationEndToEndTest, ReceiverCanBeStartedTwice) {
>+  CreateCalls();
>+
>+  test::NullTransport transport;
>+  CreateSendConfig(1, 0, 0, &transport);
>+  CreateMatchingReceiveConfigs(&transport);
>+
>+  CreateVideoStreams();
>+
>+  video_receive_streams_[0]->Start();
>+  video_receive_streams_[0]->Start();
>+
>+  DestroyStreams();
>+}
>+
>+TEST_P(CallOperationEndToEndTest, ReceiverCanBeStoppedTwice) {
>+  CreateCalls();
>+
>+  test::NullTransport transport;
>+  CreateSendConfig(1, 0, 0, &transport);
>+  CreateMatchingReceiveConfigs(&transport);
>+
>+  CreateVideoStreams();
>+
>+  video_receive_streams_[0]->Stop();
>+  video_receive_streams_[0]->Stop();
>+
>+  DestroyStreams();
>+}
>+
>+TEST_P(CallOperationEndToEndTest, ReceiverCanBeStoppedAndRestarted) {
>+  CreateCalls();
>+
>+  test::NullTransport transport;
>+  CreateSendConfig(1, 0, 0, &transport);
>+  CreateMatchingReceiveConfigs(&transport);
>+
>+  CreateVideoStreams();
>+
>+  video_receive_streams_[0]->Stop();
>+  video_receive_streams_[0]->Start();
>+  video_receive_streams_[0]->Stop();
>+
>+  DestroyStreams();
>+}
>+
>+TEST_P(CallOperationEndToEndTest, RendersSingleDelayedFrame) {
>+  static const int kWidth = 320;
>+  static const int kHeight = 240;
>+  // This constant is chosen to be higher than the timeout in the video_render
>+  // module. This makes sure that frames aren't dropped if there are no other
>+  // frames in the queue.
>+  static const int kRenderDelayMs = 1000;
>+
>+  class Renderer : public rtc::VideoSinkInterface<VideoFrame> {
>+   public:
>+    Renderer() : event_(false, false) {}
>+
>+    void OnFrame(const VideoFrame& video_frame) override {
>+      SleepMs(kRenderDelayMs);
>+      event_.Set();
>+    }
>+
>+    bool Wait() { return event_.Wait(kDefaultTimeoutMs); }
>+
>+    rtc::Event event_;
>+  } renderer;
>+
>+  test::FrameForwarder frame_forwarder;
>+  std::unique_ptr<test::DirectTransport> sender_transport;
>+  std::unique_ptr<test::DirectTransport> receiver_transport;
>+
>+  task_queue_.SendTask([this, &renderer, &frame_forwarder, &sender_transport,
>+                        &receiver_transport]() {
>+    CreateCalls();
>+
>+    sender_transport = absl::make_unique<test::DirectTransport>(
>+        &task_queue_, sender_call_.get(), payload_type_map_);
>+    receiver_transport = absl::make_unique<test::DirectTransport>(
>+        &task_queue_, receiver_call_.get(), payload_type_map_);
>+    sender_transport->SetReceiver(receiver_call_->Receiver());
>+    receiver_transport->SetReceiver(sender_call_->Receiver());
>+
>+    CreateSendConfig(1, 0, 0, sender_transport.get());
>+    CreateMatchingReceiveConfigs(receiver_transport.get());
>+
>+    video_receive_configs_[0].renderer = &renderer;
>+
>+    CreateVideoStreams();
>+    Start();
>+
>+    // Create frames that are smaller than the send width/height, this is done
>+    // to check that the callbacks are done after processing video.
>+    std::unique_ptr<test::FrameGenerator> frame_generator(
>+        test::FrameGenerator::CreateSquareGenerator(
>+            kWidth, kHeight, absl::nullopt, absl::nullopt));
>+    GetVideoSendStream()->SetSource(&frame_forwarder,
>+                                    DegradationPreference::MAINTAIN_FRAMERATE);
>+
>+    frame_forwarder.IncomingCapturedFrame(*frame_generator->NextFrame());
>+  });
>+
>+  EXPECT_TRUE(renderer.Wait())
>+      << "Timed out while waiting for the frame to render.";
>+
>+  task_queue_.SendTask([this, &sender_transport, &receiver_transport]() {
>+    Stop();
>+    DestroyStreams();
>+    sender_transport.reset();
>+    receiver_transport.reset();
>+    DestroyCalls();
>+  });
>+}
>+
>+TEST_P(CallOperationEndToEndTest, TransmitsFirstFrame) {
>+  class Renderer : public rtc::VideoSinkInterface<VideoFrame> {
>+   public:
>+    Renderer() : event_(false, false) {}
>+
>+    void OnFrame(const VideoFrame& video_frame) override { event_.Set(); }
>+
>+    bool Wait() { return event_.Wait(kDefaultTimeoutMs); }
>+
>+    rtc::Event event_;
>+  } renderer;
>+
>+  std::unique_ptr<test::FrameGenerator> frame_generator;
>+  test::FrameForwarder frame_forwarder;
>+
>+  std::unique_ptr<test::DirectTransport> sender_transport;
>+  std::unique_ptr<test::DirectTransport> receiver_transport;
>+
>+  task_queue_.SendTask([this, &renderer, &frame_generator, &frame_forwarder,
>+                        &sender_transport, &receiver_transport]() {
>+    CreateCalls();
>+
>+    sender_transport = absl::make_unique<test::DirectTransport>(
>+        &task_queue_, sender_call_.get(), payload_type_map_);
>+    receiver_transport = absl::make_unique<test::DirectTransport>(
>+        &task_queue_, receiver_call_.get(), payload_type_map_);
>+    sender_transport->SetReceiver(receiver_call_->Receiver());
>+    receiver_transport->SetReceiver(sender_call_->Receiver());
>+
>+    CreateSendConfig(1, 0, 0, sender_transport.get());
>+    CreateMatchingReceiveConfigs(receiver_transport.get());
>+    video_receive_configs_[0].renderer = &renderer;
>+
>+    CreateVideoStreams();
>+    Start();
>+
>+    frame_generator = test::FrameGenerator::CreateSquareGenerator(
>+        kDefaultWidth, kDefaultHeight, absl::nullopt, absl::nullopt);
>+    GetVideoSendStream()->SetSource(&frame_forwarder,
>+                                    DegradationPreference::MAINTAIN_FRAMERATE);
>+    frame_forwarder.IncomingCapturedFrame(*frame_generator->NextFrame());
>+  });
>+
>+  EXPECT_TRUE(renderer.Wait())
>+      << "Timed out while waiting for the frame to render.";
>+
>+  task_queue_.SendTask([this, &sender_transport, &receiver_transport]() {
>+    Stop();
>+    DestroyStreams();
>+    sender_transport.reset();
>+    receiver_transport.reset();
>+    DestroyCalls();
>+  });
>+}
>+
>+TEST_P(CallOperationEndToEndTest, ObserversEncodedFrames) {
>+  class EncodedFrameTestObserver : public EncodedFrameObserver {
>+   public:
>+    EncodedFrameTestObserver()
>+        : length_(0), frame_type_(kEmptyFrame), called_(false, false) {}
>+    virtual ~EncodedFrameTestObserver() {}
>+
>+    virtual void EncodedFrameCallback(const EncodedFrame& encoded_frame) {
>+      frame_type_ = encoded_frame.frame_type_;
>+      length_ = encoded_frame.length_;
>+      buffer_.reset(new uint8_t[length_]);
>+      memcpy(buffer_.get(), encoded_frame.data_, length_);
>+      called_.Set();
>+    }
>+
>+    bool Wait() { return called_.Wait(kDefaultTimeoutMs); }
>+
>+   private:
>+    std::unique_ptr<uint8_t[]> buffer_;
>+    size_t length_;
>+    FrameType frame_type_;
>+    rtc::Event called_;
>+  };
>+
>+  EncodedFrameTestObserver post_encode_observer;
>+  test::FrameForwarder forwarder;
>+  std::unique_ptr<test::FrameGenerator> frame_generator;
>+
>+  std::unique_ptr<test::DirectTransport> sender_transport;
>+  std::unique_ptr<test::DirectTransport> receiver_transport;
>+
>+  task_queue_.SendTask([&]() {
>+    CreateCalls();
>+
>+    sender_transport = absl::make_unique<test::DirectTransport>(
>+        &task_queue_, sender_call_.get(), payload_type_map_);
>+    receiver_transport = absl::make_unique<test::DirectTransport>(
>+        &task_queue_, receiver_call_.get(), payload_type_map_);
>+    sender_transport->SetReceiver(receiver_call_->Receiver());
>+    receiver_transport->SetReceiver(sender_call_->Receiver());
>+
>+    CreateSendConfig(1, 0, 0, sender_transport.get());
>+    CreateMatchingReceiveConfigs(receiver_transport.get());
>+    GetVideoSendConfig()->post_encode_callback = &post_encode_observer;
>+
>+    CreateVideoStreams();
>+    Start();
>+
>+    frame_generator = test::FrameGenerator::CreateSquareGenerator(
>+        kDefaultWidth, kDefaultHeight, absl::nullopt, absl::nullopt);
>+    GetVideoSendStream()->SetSource(&forwarder,
>+                                    DegradationPreference::MAINTAIN_FRAMERATE);
>+    forwarder.IncomingCapturedFrame(*frame_generator->NextFrame());
>+  });
>+
>+  EXPECT_TRUE(post_encode_observer.Wait())
>+      << "Timed out while waiting for send-side encoded-frame callback.";
>+
>+  task_queue_.SendTask([this, &sender_transport, &receiver_transport]() {
>+    Stop();
>+    DestroyStreams();
>+    sender_transport.reset();
>+    receiver_transport.reset();
>+    DestroyCalls();
>+  });
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/codec_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/codec_tests.cc
>new file mode 100644
>index 00000000000..58af0a7c1d1
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/codec_tests.cc
>@@ -0,0 +1,216 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "media/engine/internaldecoderfactory.h"
>+#include "media/engine/internalencoderfactory.h"
>+#include "modules/video_coding/codecs/h264/include/h264.h"
>+#include "modules/video_coding/codecs/multiplex/include/multiplex_decoder_adapter.h"
>+#include "modules/video_coding/codecs/multiplex/include/multiplex_encoder_adapter.h"
>+#include "modules/video_coding/codecs/vp8/include/vp8.h"
>+#include "modules/video_coding/codecs/vp9/include/vp9.h"
>+#include "test/call_test.h"
>+#include "test/encoder_settings.h"
>+#include "test/field_trial.h"
>+#include "test/function_video_encoder_factory.h"
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+
>+class CodecEndToEndTest : public test::CallTest,
>+                          public testing::WithParamInterface<std::string> {
>+ public:
>+  CodecEndToEndTest() : field_trial_(GetParam()) {}
>+
>+ private:
>+  test::ScopedFieldTrials field_trial_;
>+};
>+
>+INSTANTIATE_TEST_CASE_P(RoundRobin,
>+                        CodecEndToEndTest,
>+                        ::testing::Values("WebRTC-RoundRobinPacing/Disabled/",
>+                                          "WebRTC-RoundRobinPacing/Enabled/"));
>+
>+class CodecObserver : public test::EndToEndTest,
>+                      public rtc::VideoSinkInterface<VideoFrame> {
>+ public:
>+  CodecObserver(int no_frames_to_wait_for,
>+                VideoRotation rotation_to_test,
>+                const std::string& payload_name,
>+                VideoEncoderFactory* encoder_factory,
>+                std::unique_ptr<webrtc::VideoDecoder> decoder)
>+      : EndToEndTest(4 * CodecEndToEndTest::kDefaultTimeoutMs),
>+        // TODO(hta): This timeout (120 seconds) is excessive.
>+        // https://bugs.webrtc.org/6830
>+        no_frames_to_wait_for_(no_frames_to_wait_for),
>+        expected_rotation_(rotation_to_test),
>+        payload_name_(payload_name),
>+        encoder_factory_(encoder_factory),
>+        decoder_(std::move(decoder)),
>+        frame_counter_(0) {}
>+
>+  void PerformTest() override {
>+    EXPECT_TRUE(Wait())
>+        << "Timed out while waiting for enough frames to be decoded.";
>+  }
>+
>+  void ModifyVideoConfigs(
>+      VideoSendStream::Config* send_config,
>+      std::vector<VideoReceiveStream::Config>* receive_configs,
>+      VideoEncoderConfig* encoder_config) override {
>+    encoder_config->codec_type = PayloadStringToCodecType(payload_name_);
>+    send_config->encoder_settings.encoder_factory = encoder_factory_;
>+    send_config->rtp.payload_name = payload_name_;
>+    send_config->rtp.payload_type = test::CallTest::kVideoSendPayloadType;
>+
>+    (*receive_configs)[0].renderer = this;
>+    (*receive_configs)[0].decoders.resize(1);
>+    (*receive_configs)[0].decoders[0].payload_type =
>+        send_config->rtp.payload_type;
>+    (*receive_configs)[0].decoders[0].payload_name =
>+        send_config->rtp.payload_name;
>+    (*receive_configs)[0].decoders[0].decoder = decoder_.get();
>+  }
>+
>+  void OnFrame(const VideoFrame& video_frame) override {
>+    EXPECT_EQ(expected_rotation_, video_frame.rotation());
>+    if (++frame_counter_ == no_frames_to_wait_for_)
>+      observation_complete_.Set();
>+  }
>+
>+  void OnFrameGeneratorCapturerCreated(
>+      test::FrameGeneratorCapturer* frame_generator_capturer) override {
>+    frame_generator_capturer->SetFakeRotation(expected_rotation_);
>+  }
>+
>+ private:
>+  int no_frames_to_wait_for_;
>+  VideoRotation expected_rotation_;
>+  std::string payload_name_;
>+  VideoEncoderFactory* encoder_factory_;
>+  std::unique_ptr<webrtc::VideoDecoder> decoder_;
>+  int frame_counter_;
>+};
>+
>+TEST_P(CodecEndToEndTest, SendsAndReceivesVP8) {
>+  test::FunctionVideoEncoderFactory encoder_factory(
>+      []() { return VP8Encoder::Create(); });
>+  CodecObserver test(5, kVideoRotation_0, "VP8", &encoder_factory,
>+                     VP8Decoder::Create());
>+  RunBaseTest(&test);
>+}
>+
>+TEST_P(CodecEndToEndTest, SendsAndReceivesVP8Rotation90) {
>+  test::FunctionVideoEncoderFactory encoder_factory(
>+      []() { return VP8Encoder::Create(); });
>+  CodecObserver test(5, kVideoRotation_90, "VP8", &encoder_factory,
>+                     VP8Decoder::Create());
>+  RunBaseTest(&test);
>+}
>+
>+#if !defined(RTC_DISABLE_VP9)
>+TEST_P(CodecEndToEndTest, SendsAndReceivesVP9) {
>+  test::FunctionVideoEncoderFactory encoder_factory(
>+      []() { return VP9Encoder::Create(); });
>+  CodecObserver test(500, kVideoRotation_0, "VP9", &encoder_factory,
>+                     VP9Decoder::Create());
>+  RunBaseTest(&test);
>+}
>+
>+TEST_P(CodecEndToEndTest, SendsAndReceivesVP9VideoRotation90) {
>+  test::FunctionVideoEncoderFactory encoder_factory(
>+      []() { return VP9Encoder::Create(); });
>+  CodecObserver test(5, kVideoRotation_90, "VP9", &encoder_factory,
>+                     VP9Decoder::Create());
>+  RunBaseTest(&test);
>+}
>+
>+// Mutiplex tests are using VP9 as the underlying implementation.
>+TEST_P(CodecEndToEndTest, SendsAndReceivesMultiplex) {
>+  InternalEncoderFactory internal_encoder_factory;
>+  InternalDecoderFactory decoder_factory;
>+  test::FunctionVideoEncoderFactory encoder_factory(
>+      [&internal_encoder_factory]() {
>+        return absl::make_unique<MultiplexEncoderAdapter>(
>+            &internal_encoder_factory, SdpVideoFormat(cricket::kVp9CodecName));
>+      });
>+  CodecObserver test(
>+      5, kVideoRotation_0, "multiplex", &encoder_factory,
>+      absl::make_unique<MultiplexDecoderAdapter>(
>+          &decoder_factory, SdpVideoFormat(cricket::kVp9CodecName)));
>+  RunBaseTest(&test);
>+}
>+
>+TEST_P(CodecEndToEndTest, SendsAndReceivesMultiplexVideoRotation90) {
>+  InternalEncoderFactory internal_encoder_factory;
>+  InternalDecoderFactory decoder_factory;
>+  test::FunctionVideoEncoderFactory encoder_factory(
>+      [&internal_encoder_factory]() {
>+        return absl::make_unique<MultiplexEncoderAdapter>(
>+            &internal_encoder_factory, SdpVideoFormat(cricket::kVp9CodecName));
>+      });
>+  CodecObserver test(
>+      5, kVideoRotation_90, "multiplex", &encoder_factory,
>+      absl::make_unique<MultiplexDecoderAdapter>(
>+          &decoder_factory, SdpVideoFormat(cricket::kVp9CodecName)));
>+  RunBaseTest(&test);
>+}
>+
>+#endif  // !defined(RTC_DISABLE_VP9)
>+
>+#if defined(WEBRTC_USE_H264)
>+class EndToEndTestH264 : public CodecEndToEndTest {};
>+
>+const auto h264_field_trial_combinations = ::testing::Values(
>+    "WebRTC-SpsPpsIdrIsH264Keyframe/Disabled/WebRTC-RoundRobinPacing/Disabled/",
>+    "WebRTC-SpsPpsIdrIsH264Keyframe/Enabled/WebRTC-RoundRobinPacing/Disabled/",
>+    "WebRTC-SpsPpsIdrIsH264Keyframe/Disabled/WebRTC-RoundRobinPacing/Enabled/",
>+    "WebRTC-SpsPpsIdrIsH264Keyframe/Enabled/WebRTC-RoundRobinPacing/Enabled/");
>+INSTANTIATE_TEST_CASE_P(SpsPpsIdrIsKeyframe,
>+                        EndToEndTestH264,
>+                        h264_field_trial_combinations);
>+
>+TEST_P(EndToEndTestH264, SendsAndReceivesH264) {
>+  test::FunctionVideoEncoderFactory encoder_factory(
>+      []() { return H264Encoder::Create(cricket::VideoCodec("H264")); });
>+  CodecObserver test(500, kVideoRotation_0, "H264", &encoder_factory,
>+                     H264Decoder::Create());
>+  RunBaseTest(&test);
>+}
>+
>+TEST_P(EndToEndTestH264, SendsAndReceivesH264VideoRotation90) {
>+  test::FunctionVideoEncoderFactory encoder_factory(
>+      []() { return H264Encoder::Create(cricket::VideoCodec("H264")); });
>+  CodecObserver test(5, kVideoRotation_90, "H264", &encoder_factory,
>+                     H264Decoder::Create());
>+  RunBaseTest(&test);
>+}
>+
>+TEST_P(EndToEndTestH264, SendsAndReceivesH264PacketizationMode0) {
>+  cricket::VideoCodec codec = cricket::VideoCodec("H264");
>+  codec.SetParam(cricket::kH264FmtpPacketizationMode, "0");
>+  test::FunctionVideoEncoderFactory encoder_factory(
>+      [codec]() { return H264Encoder::Create(codec); });
>+  CodecObserver test(500, kVideoRotation_0, "H264", &encoder_factory,
>+                     H264Decoder::Create());
>+  RunBaseTest(&test);
>+}
>+
>+TEST_P(EndToEndTestH264, SendsAndReceivesH264PacketizationMode1) {
>+  cricket::VideoCodec codec = cricket::VideoCodec("H264");
>+  codec.SetParam(cricket::kH264FmtpPacketizationMode, "1");
>+  test::FunctionVideoEncoderFactory encoder_factory(
>+      [codec]() { return H264Encoder::Create(codec); });
>+  CodecObserver test(500, kVideoRotation_0, "H264", &encoder_factory,
>+                     H264Decoder::Create());
>+  RunBaseTest(&test);
>+}
>+#endif  // defined(WEBRTC_USE_H264)
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/config_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/config_tests.cc
>new file mode 100644
>index 00000000000..d32b1118b7c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/config_tests.cc
>@@ -0,0 +1,92 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "test/call_test.h"
>+#include "test/gtest.h"
>+#include "test/null_transport.h"
>+
>+namespace webrtc {
>+
>+class ConfigEndToEndTest : public test::CallTest {};
>+
>+namespace {
>+void VerifyEmptyNackConfig(const NackConfig& config) {
>+  EXPECT_EQ(0, config.rtp_history_ms)
>+      << "Enabling NACK requires rtcp-fb: nack negotiation.";
>+}
>+
>+void VerifyEmptyUlpfecConfig(const UlpfecConfig& config) {
>+  EXPECT_EQ(-1, config.ulpfec_payload_type)
>+      << "Enabling ULPFEC requires rtpmap: ulpfec negotiation.";
>+  EXPECT_EQ(-1, config.red_payload_type)
>+      << "Enabling ULPFEC requires rtpmap: red negotiation.";
>+  EXPECT_EQ(-1, config.red_rtx_payload_type)
>+      << "Enabling RTX in ULPFEC requires rtpmap: rtx negotiation.";
>+}
>+
>+void VerifyEmptyFlexfecConfig(const RtpConfig::Flexfec& config) {
>+  EXPECT_EQ(-1, config.payload_type)
>+      << "Enabling FlexFEC requires rtpmap: flexfec negotiation.";
>+  EXPECT_EQ(0U, config.ssrc)
>+      << "Enabling FlexFEC requires ssrc-group: FEC-FR negotiation.";
>+  EXPECT_TRUE(config.protected_media_ssrcs.empty())
>+      << "Enabling FlexFEC requires ssrc-group: FEC-FR negotiation.";
>+}
>+}  // namespace
>+
>+TEST_F(ConfigEndToEndTest, VerifyDefaultSendConfigParameters) {
>+  VideoSendStream::Config default_send_config(nullptr);
>+  EXPECT_EQ(0, default_send_config.rtp.nack.rtp_history_ms)
>+      << "Enabling NACK require rtcp-fb: nack negotiation.";
>+  EXPECT_TRUE(default_send_config.rtp.rtx.ssrcs.empty())
>+      << "Enabling RTX requires rtpmap: rtx negotiation.";
>+  EXPECT_TRUE(default_send_config.rtp.extensions.empty())
>+      << "Enabling RTP extensions require negotiation.";
>+
>+  VerifyEmptyNackConfig(default_send_config.rtp.nack);
>+  VerifyEmptyUlpfecConfig(default_send_config.rtp.ulpfec);
>+  VerifyEmptyFlexfecConfig(default_send_config.rtp.flexfec);
>+}
>+
>+TEST_F(ConfigEndToEndTest, VerifyDefaultVideoReceiveConfigParameters) {
>+  VideoReceiveStream::Config default_receive_config(nullptr);
>+  EXPECT_EQ(RtcpMode::kCompound, default_receive_config.rtp.rtcp_mode)
>+      << "Reduced-size RTCP require rtcp-rsize to be negotiated.";
>+  EXPECT_FALSE(default_receive_config.rtp.remb)
>+      << "REMB require rtcp-fb: goog-remb to be negotiated.";
>+  EXPECT_FALSE(
>+      default_receive_config.rtp.rtcp_xr.receiver_reference_time_report)
>+      << "RTCP XR settings require rtcp-xr to be negotiated.";
>+  EXPECT_EQ(0U, default_receive_config.rtp.rtx_ssrc)
>+      << "Enabling RTX requires ssrc-group: FID negotiation";
>+  EXPECT_TRUE(default_receive_config.rtp.rtx_associated_payload_types.empty())
>+      << "Enabling RTX requires rtpmap: rtx negotiation.";
>+  EXPECT_TRUE(default_receive_config.rtp.extensions.empty())
>+      << "Enabling RTP extensions require negotiation.";
>+
>+  VerifyEmptyNackConfig(default_receive_config.rtp.nack);
>+  EXPECT_EQ(-1, default_receive_config.rtp.ulpfec_payload_type)
>+      << "Enabling ULPFEC requires rtpmap: ulpfec negotiation.";
>+  EXPECT_EQ(-1, default_receive_config.rtp.red_payload_type)
>+      << "Enabling ULPFEC requires rtpmap: red negotiation.";
>+}
>+
>+TEST_F(ConfigEndToEndTest, VerifyDefaultFlexfecReceiveConfigParameters) {
>+  test::NullTransport rtcp_send_transport;
>+  FlexfecReceiveStream::Config default_receive_config(&rtcp_send_transport);
>+  EXPECT_EQ(-1, default_receive_config.payload_type)
>+      << "Enabling FlexFEC requires rtpmap: flexfec negotiation.";
>+  EXPECT_EQ(0U, default_receive_config.remote_ssrc)
>+      << "Enabling FlexFEC requires ssrc-group: FEC-FR negotiation.";
>+  EXPECT_TRUE(default_receive_config.protected_media_ssrcs.empty())
>+      << "Enabling FlexFEC requires ssrc-group: FEC-FR negotiation.";
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/extended_reports_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/extended_reports_tests.cc
>new file mode 100644
>index 00000000000..322a9c301ff
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/extended_reports_tests.cc
>@@ -0,0 +1,205 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "test/call_test.h"
>+#include "test/gtest.h"
>+#include "test/rtcp_packet_parser.h"
>+
>+namespace webrtc {
>+
>+class ExtendedReportsEndToEndTest : public test::CallTest {};
>+
>+class RtcpXrObserver : public test::EndToEndTest {
>+ public:
>+  RtcpXrObserver(bool enable_rrtr,
>+                 bool enable_target_bitrate,
>+                 bool enable_zero_target_bitrate)
>+      : EndToEndTest(test::CallTest::kDefaultTimeoutMs),
>+        enable_rrtr_(enable_rrtr),
>+        enable_target_bitrate_(enable_target_bitrate),
>+        enable_zero_target_bitrate_(enable_zero_target_bitrate),
>+        sent_rtcp_sr_(0),
>+        sent_rtcp_rr_(0),
>+        sent_rtcp_rrtr_(0),
>+        sent_rtcp_target_bitrate_(false),
>+        sent_zero_rtcp_target_bitrate_(false),
>+        sent_rtcp_dlrr_(0) {}
>+
>+ private:
>+  // Receive stream should send RR packets (and RRTR packets if enabled).
>+  Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
>+    rtc::CritScope lock(&crit_);
>+    test::RtcpPacketParser parser;
>+    EXPECT_TRUE(parser.Parse(packet, length));
>+
>+    sent_rtcp_rr_ += parser.receiver_report()->num_packets();
>+    EXPECT_EQ(0, parser.sender_report()->num_packets());
>+    EXPECT_GE(1, parser.xr()->num_packets());
>+    if (parser.xr()->num_packets() > 0) {
>+      if (parser.xr()->rrtr())
>+        ++sent_rtcp_rrtr_;
>+      EXPECT_FALSE(parser.xr()->dlrr());
>+    }
>+
>+    return SEND_PACKET;
>+  }
>+  // Send stream should send SR packets (and DLRR packets if enabled).
>+  Action OnSendRtcp(const uint8_t* packet, size_t length) override {
>+    rtc::CritScope lock(&crit_);
>+    test::RtcpPacketParser parser;
>+    EXPECT_TRUE(parser.Parse(packet, length));
>+
>+    sent_rtcp_sr_ += parser.sender_report()->num_packets();
>+    EXPECT_LE(parser.xr()->num_packets(), 1);
>+    if (parser.xr()->num_packets() > 0) {
>+      EXPECT_FALSE(parser.xr()->rrtr());
>+      if (parser.xr()->dlrr())
>+        ++sent_rtcp_dlrr_;
>+      if (parser.xr()->target_bitrate()) {
>+        sent_rtcp_target_bitrate_ = true;
>+        for (const rtcp::TargetBitrate::BitrateItem& item :
>+             parser.xr()->target_bitrate()->GetTargetBitrates()) {
>+          if (item.target_bitrate_kbps == 0) {
>+            sent_zero_rtcp_target_bitrate_ = true;
>+            break;
>+          }
>+        }
>+      }
>+    }
>+
>+    if (sent_rtcp_sr_ > kNumRtcpReportPacketsToObserve &&
>+        sent_rtcp_rr_ > kNumRtcpReportPacketsToObserve &&
>+        (sent_rtcp_target_bitrate_ || !enable_target_bitrate_) &&
>+        (sent_zero_rtcp_target_bitrate_ || !enable_zero_target_bitrate_)) {
>+      if (enable_rrtr_) {
>+        EXPECT_GT(sent_rtcp_rrtr_, 0);
>+        EXPECT_GT(sent_rtcp_dlrr_, 0);
>+      } else {
>+        EXPECT_EQ(sent_rtcp_rrtr_, 0);
>+        EXPECT_EQ(sent_rtcp_dlrr_, 0);
>+      }
>+      EXPECT_EQ(enable_target_bitrate_, sent_rtcp_target_bitrate_);
>+      EXPECT_EQ(enable_zero_target_bitrate_, sent_zero_rtcp_target_bitrate_);
>+      observation_complete_.Set();
>+    }
>+    return SEND_PACKET;
>+  }
>+
>+  size_t GetNumVideoStreams() const override {
>+    // When sending a zero target bitrate, we use two spatial layers so that
>+    // we'll still have a layer with non-zero bitrate.
>+    return enable_zero_target_bitrate_ ? 2 : 1;
>+  }
>+
>+  // This test uses VideoStream settings different from the the default one
>+  // implemented in DefaultVideoStreamFactory, so it implements its own
>+  // VideoEncoderConfig::VideoStreamFactoryInterface which is created
>+  // in ModifyVideoConfigs.
>+  class ZeroTargetVideoStreamFactory
>+      : public VideoEncoderConfig::VideoStreamFactoryInterface {
>+   public:
>+    ZeroTargetVideoStreamFactory() {}
>+
>+   private:
>+    std::vector<VideoStream> CreateEncoderStreams(
>+        int width,
>+        int height,
>+        const VideoEncoderConfig& encoder_config) override {
>+      std::vector<VideoStream> streams =
>+          test::CreateVideoStreams(width, height, encoder_config);
>+      // Set one of the streams' target bitrates to zero to test that a
>+      // bitrate of 0 can be signalled.
>+      streams[encoder_config.number_of_streams - 1].min_bitrate_bps = 0;
>+      streams[encoder_config.number_of_streams - 1].target_bitrate_bps = 0;
>+      streams[encoder_config.number_of_streams - 1].max_bitrate_bps = 0;
>+      return streams;
>+    }
>+  };
>+
>+  void ModifyVideoConfigs(
>+      VideoSendStream::Config* send_config,
>+      std::vector<VideoReceiveStream::Config>* receive_configs,
>+      VideoEncoderConfig* encoder_config) override {
>+    if (enable_zero_target_bitrate_) {
>+      encoder_config->video_stream_factory =
>+          new rtc::RefCountedObject<ZeroTargetVideoStreamFactory>();
>+
>+      // Configure VP8 to be able to use simulcast.
>+      send_config->rtp.payload_name = "VP8";
>+      encoder_config->codec_type = kVideoCodecVP8;
>+      (*receive_configs)[0].decoders.resize(1);
>+      (*receive_configs)[0].decoders[0].payload_type =
>+          send_config->rtp.payload_type;
>+      (*receive_configs)[0].decoders[0].payload_name =
>+          send_config->rtp.payload_name;
>+    }
>+    if (enable_target_bitrate_) {
>+      // TargetBitrate only signaled for screensharing.
>+      encoder_config->content_type = VideoEncoderConfig::ContentType::kScreen;
>+    }
>+    (*receive_configs)[0].rtp.rtcp_mode = RtcpMode::kReducedSize;
>+    (*receive_configs)[0].rtp.rtcp_xr.receiver_reference_time_report =
>+        enable_rrtr_;
>+  }
>+
>+  void PerformTest() override {
>+    EXPECT_TRUE(Wait())
>+        << "Timed out while waiting for RTCP SR/RR packets to be sent.";
>+  }
>+
>+  static const int kNumRtcpReportPacketsToObserve = 5;
>+
>+  rtc::CriticalSection crit_;
>+  const bool enable_rrtr_;
>+  const bool enable_target_bitrate_;
>+  const bool enable_zero_target_bitrate_;
>+  int sent_rtcp_sr_;
>+  int sent_rtcp_rr_ RTC_GUARDED_BY(&crit_);
>+  int sent_rtcp_rrtr_ RTC_GUARDED_BY(&crit_);
>+  bool sent_rtcp_target_bitrate_ RTC_GUARDED_BY(&crit_);
>+  bool sent_zero_rtcp_target_bitrate_ RTC_GUARDED_BY(&crit_);
>+  int sent_rtcp_dlrr_;
>+};
>+
>+TEST_F(ExtendedReportsEndToEndTest,
>+       TestExtendedReportsWithRrtrWithoutTargetBitrate) {
>+  RtcpXrObserver test(/*enable_rrtr=*/true, /*enable_target_bitrate=*/false,
>+                      /*enable_zero_target_bitrate=*/false);
>+  RunBaseTest(&test);
>+}
>+
>+TEST_F(ExtendedReportsEndToEndTest,
>+       TestExtendedReportsWithoutRrtrWithoutTargetBitrate) {
>+  RtcpXrObserver test(/*enable_rrtr=*/false, /*enable_target_bitrate=*/false,
>+                      /*enable_zero_target_bitrate=*/false);
>+  RunBaseTest(&test);
>+}
>+
>+TEST_F(ExtendedReportsEndToEndTest,
>+       TestExtendedReportsWithRrtrWithTargetBitrate) {
>+  RtcpXrObserver test(/*enable_rrtr=*/true, /*enable_target_bitrate=*/true,
>+                      /*enable_zero_target_bitrate=*/false);
>+  RunBaseTest(&test);
>+}
>+
>+TEST_F(ExtendedReportsEndToEndTest,
>+       TestExtendedReportsWithoutRrtrWithTargetBitrate) {
>+  RtcpXrObserver test(/*enable_rrtr=*/false, /*enable_target_bitrate=*/true,
>+                      /*enable_zero_target_bitrate=*/false);
>+  RunBaseTest(&test);
>+}
>+
>+TEST_F(ExtendedReportsEndToEndTest,
>+       TestExtendedReportsCanSignalZeroTargetBitrate) {
>+  RtcpXrObserver test(/*enable_rrtr=*/false, /*enable_target_bitrate=*/true,
>+                      /*enable_zero_target_bitrate=*/true);
>+  RunBaseTest(&test);
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/fec_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/fec_tests.cc
>new file mode 100644
>index 00000000000..4493bde019e
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/fec_tests.cc
>@@ -0,0 +1,492 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/rtp_rtcp/source/byte_io.h"
>+#include "modules/video_coding/codecs/vp8/include/vp8.h"
>+#include "test/call_test.h"
>+#include "test/field_trial.h"
>+#include "test/function_video_encoder_factory.h"
>+#include "test/gtest.h"
>+#include "test/rtcp_packet_parser.h"
>+
>+namespace webrtc {
>+
>+class FecEndToEndTest : public test::CallTest,
>+                        public testing::WithParamInterface<std::string> {
>+ public:
>+  FecEndToEndTest() : field_trial_(GetParam()) {}
>+
>+ private:
>+  test::ScopedFieldTrials field_trial_;
>+};
>+
>+INSTANTIATE_TEST_CASE_P(RoundRobin,
>+                        FecEndToEndTest,
>+                        ::testing::Values("WebRTC-RoundRobinPacing/Disabled/",
>+                                          "WebRTC-RoundRobinPacing/Enabled/"));
>+
>+TEST_P(FecEndToEndTest, ReceivesUlpfec) {
>+  class UlpfecRenderObserver : public test::EndToEndTest,
>+                               public rtc::VideoSinkInterface<VideoFrame> {
>+   public:
>+    UlpfecRenderObserver()
>+        : EndToEndTest(kDefaultTimeoutMs),
>+          encoder_factory_([]() { return VP8Encoder::Create(); }),
>+          random_(0xcafef00d1),
>+          num_packets_sent_(0) {}
>+
>+   private:
>+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
>+      rtc::CritScope lock(&crit_);
>+      RTPHeader header;
>+      EXPECT_TRUE(parser_->Parse(packet, length, &header));
>+
>+      EXPECT_TRUE(header.payloadType == kVideoSendPayloadType ||
>+                  header.payloadType == kRedPayloadType)
>+          << "Unknown payload type received.";
>+      EXPECT_EQ(kVideoSendSsrcs[0], header.ssrc) << "Unknown SSRC received.";
>+
>+      // Parse RED header.
>+      int encapsulated_payload_type = -1;
>+      if (header.payloadType == kRedPayloadType) {
>+        encapsulated_payload_type =
>+            static_cast<int>(packet[header.headerLength]);
>+
>+        EXPECT_TRUE(encapsulated_payload_type == kVideoSendPayloadType ||
>+                    encapsulated_payload_type == kUlpfecPayloadType)
>+            << "Unknown encapsulated payload type received.";
>+      }
>+
>+      // To minimize test flakiness, always let ULPFEC packets through.
>+      if (encapsulated_payload_type == kUlpfecPayloadType) {
>+        return SEND_PACKET;
>+      }
>+
>+      // Simulate 5% video packet loss after rampup period. Record the
>+      // corresponding timestamps that were dropped.
>+      if (num_packets_sent_++ > 100 && random_.Rand(1, 100) <= 5) {
>+        if (encapsulated_payload_type == kVideoSendPayloadType) {
>+          dropped_sequence_numbers_.insert(header.sequenceNumber);
>+          dropped_timestamps_.insert(header.timestamp);
>+        }
>+        return DROP_PACKET;
>+      }
>+
>+      return SEND_PACKET;
>+    }
>+
>+    void OnFrame(const VideoFrame& video_frame) override {
>+      rtc::CritScope lock(&crit_);
>+      // Rendering frame with timestamp of packet that was dropped -> FEC
>+      // protection worked.
>+      auto it = dropped_timestamps_.find(video_frame.timestamp());
>+      if (it != dropped_timestamps_.end()) {
>+        observation_complete_.Set();
>+      }
>+    }
>+
>+    void ModifyVideoConfigs(
>+        VideoSendStream::Config* send_config,
>+        std::vector<VideoReceiveStream::Config>* receive_configs,
>+        VideoEncoderConfig* encoder_config) override {
>+      // Use VP8 instead of FAKE, since the latter does not have PictureID
>+      // in the packetization headers.
>+      send_config->encoder_settings.encoder_factory = &encoder_factory_;
>+      send_config->rtp.payload_name = "VP8";
>+      send_config->rtp.payload_type = kVideoSendPayloadType;
>+      encoder_config->codec_type = kVideoCodecVP8;
>+      VideoReceiveStream::Decoder decoder =
>+          test::CreateMatchingDecoder(*send_config);
>+      decoder_.reset(decoder.decoder);
>+      (*receive_configs)[0].decoders.clear();
>+      (*receive_configs)[0].decoders.push_back(decoder);
>+
>+      // Enable ULPFEC over RED.
>+      send_config->rtp.ulpfec.red_payload_type = kRedPayloadType;
>+      send_config->rtp.ulpfec.ulpfec_payload_type = kUlpfecPayloadType;
>+      (*receive_configs)[0].rtp.red_payload_type = kRedPayloadType;
>+      (*receive_configs)[0].rtp.ulpfec_payload_type = kUlpfecPayloadType;
>+
>+      (*receive_configs)[0].renderer = this;
>+    }
>+
>+    void PerformTest() override {
>+      EXPECT_TRUE(Wait())
>+          << "Timed out waiting for dropped frames to be rendered.";
>+    }
>+
>+    rtc::CriticalSection crit_;
>+    std::unique_ptr<VideoEncoder> encoder_;
>+    test::FunctionVideoEncoderFactory encoder_factory_;
>+    std::unique_ptr<VideoDecoder> decoder_;
>+    std::set<uint32_t> dropped_sequence_numbers_ RTC_GUARDED_BY(crit_);
>+    // Several packets can have the same timestamp.
>+    std::multiset<uint32_t> dropped_timestamps_ RTC_GUARDED_BY(crit_);
>+    Random random_;
>+    int num_packets_sent_ RTC_GUARDED_BY(crit_);
>+  } test;
>+
>+  RunBaseTest(&test);
>+}
>+
>+class FlexfecRenderObserver : public test::EndToEndTest,
>+                              public rtc::VideoSinkInterface<VideoFrame> {
>+ public:
>+  static constexpr uint32_t kVideoLocalSsrc = 123;
>+  static constexpr uint32_t kFlexfecLocalSsrc = 456;
>+
>+  explicit FlexfecRenderObserver(bool enable_nack, bool expect_flexfec_rtcp)
>+      : test::EndToEndTest(test::CallTest::kDefaultTimeoutMs),
>+        enable_nack_(enable_nack),
>+        expect_flexfec_rtcp_(expect_flexfec_rtcp),
>+        received_flexfec_rtcp_(false),
>+        random_(0xcafef00d1),
>+        num_packets_sent_(0) {}
>+
>+  size_t GetNumFlexfecStreams() const override { return 1; }
>+
>+ private:
>+  Action OnSendRtp(const uint8_t* packet, size_t length) override {
>+    rtc::CritScope lock(&crit_);
>+    RTPHeader header;
>+    EXPECT_TRUE(parser_->Parse(packet, length, &header));
>+
>+    EXPECT_TRUE(header.payloadType ==
>+                    test::CallTest::kFakeVideoSendPayloadType ||
>+                header.payloadType == test::CallTest::kFlexfecPayloadType ||
>+                (enable_nack_ &&
>+                 header.payloadType == test::CallTest::kSendRtxPayloadType))
>+        << "Unknown payload type received.";
>+    EXPECT_TRUE(
>+        header.ssrc == test::CallTest::kVideoSendSsrcs[0] ||
>+        header.ssrc == test::CallTest::kFlexfecSendSsrc ||
>+        (enable_nack_ && header.ssrc == test::CallTest::kSendRtxSsrcs[0]))
>+        << "Unknown SSRC received.";
>+
>+    // To reduce test flakiness, always let FlexFEC packets through.
>+    if (header.payloadType == test::CallTest::kFlexfecPayloadType) {
>+      EXPECT_EQ(test::CallTest::kFlexfecSendSsrc, header.ssrc);
>+
>+      return SEND_PACKET;
>+    }
>+
>+    // To reduce test flakiness, always let RTX packets through.
>+    if (header.payloadType == test::CallTest::kSendRtxPayloadType) {
>+      EXPECT_EQ(test::CallTest::kSendRtxSsrcs[0], header.ssrc);
>+
>+      // Parse RTX header.
>+      uint16_t original_sequence_number =
>+          ByteReader<uint16_t>::ReadBigEndian(&packet[header.headerLength]);
>+
>+      // From the perspective of FEC, a retransmitted packet is no longer
>+      // dropped, so remove it from list of dropped packets.
>+      auto seq_num_it =
>+          dropped_sequence_numbers_.find(original_sequence_number);
>+      if (seq_num_it != dropped_sequence_numbers_.end()) {
>+        dropped_sequence_numbers_.erase(seq_num_it);
>+        auto ts_it = dropped_timestamps_.find(header.timestamp);
>+        EXPECT_NE(ts_it, dropped_timestamps_.end());
>+        dropped_timestamps_.erase(ts_it);
>+      }
>+
>+      return SEND_PACKET;
>+    }
>+
>+    // Simulate 5% video packet loss after rampup period. Record the
>+    // corresponding timestamps that were dropped.
>+    if (num_packets_sent_++ > 100 && random_.Rand(1, 100) <= 5) {
>+      EXPECT_EQ(test::CallTest::kFakeVideoSendPayloadType, header.payloadType);
>+      EXPECT_EQ(test::CallTest::kVideoSendSsrcs[0], header.ssrc);
>+
>+      dropped_sequence_numbers_.insert(header.sequenceNumber);
>+      dropped_timestamps_.insert(header.timestamp);
>+
>+      return DROP_PACKET;
>+    }
>+
>+    return SEND_PACKET;
>+  }
>+
>+  Action OnReceiveRtcp(const uint8_t* data, size_t length) override {
>+    test::RtcpPacketParser parser;
>+
>+    parser.Parse(data, length);
>+    if (parser.sender_ssrc() == kFlexfecLocalSsrc) {
>+      EXPECT_EQ(1, parser.receiver_report()->num_packets());
>+      const std::vector<rtcp::ReportBlock>& report_blocks =
>+          parser.receiver_report()->report_blocks();
>+      if (!report_blocks.empty()) {
>+        EXPECT_EQ(1U, report_blocks.size());
>+        EXPECT_EQ(test::CallTest::kFlexfecSendSsrc,
>+                  report_blocks[0].source_ssrc());
>+        rtc::CritScope lock(&crit_);
>+        received_flexfec_rtcp_ = true;
>+      }
>+    }
>+
>+    return SEND_PACKET;
>+  }
>+
>+  test::PacketTransport* CreateSendTransport(
>+      test::SingleThreadedTaskQueueForTesting* task_queue,
>+      Call* sender_call) override {
>+    // At low RTT (< kLowRttNackMs) -> NACK only, no FEC.
>+    const int kNetworkDelayMs = 100;
>+    FakeNetworkPipe::Config config;
>+    config.queue_delay_ms = kNetworkDelayMs;
>+    return new test::PacketTransport(task_queue, sender_call, this,
>+                                     test::PacketTransport::kSender,
>+                                     test::CallTest::payload_type_map_, config);
>+  }
>+
>+  void OnFrame(const VideoFrame& video_frame) override {
>+    EXPECT_EQ(kVideoRotation_90, video_frame.rotation());
>+
>+    rtc::CritScope lock(&crit_);
>+    // Rendering frame with timestamp of packet that was dropped -> FEC
>+    // protection worked.
>+    auto it = dropped_timestamps_.find(video_frame.timestamp());
>+    if (it != dropped_timestamps_.end()) {
>+      if (!expect_flexfec_rtcp_ || received_flexfec_rtcp_) {
>+        observation_complete_.Set();
>+      }
>+    }
>+  }
>+
>+  void ModifyVideoConfigs(
>+      VideoSendStream::Config* send_config,
>+      std::vector<VideoReceiveStream::Config>* receive_configs,
>+      VideoEncoderConfig* encoder_config) override {
>+    (*receive_configs)[0].rtp.local_ssrc = kVideoLocalSsrc;
>+    (*receive_configs)[0].renderer = this;
>+
>+    if (enable_nack_) {
>+      send_config->rtp.nack.rtp_history_ms = test::CallTest::kNackRtpHistoryMs;
>+      send_config->rtp.rtx.ssrcs.push_back(test::CallTest::kSendRtxSsrcs[0]);
>+      send_config->rtp.rtx.payload_type = test::CallTest::kSendRtxPayloadType;
>+
>+      (*receive_configs)[0].rtp.nack.rtp_history_ms =
>+          test::CallTest::kNackRtpHistoryMs;
>+      (*receive_configs)[0].rtp.rtx_ssrc = test::CallTest::kSendRtxSsrcs[0];
>+      (*receive_configs)[0]
>+          .rtp
>+          .rtx_associated_payload_types[test::CallTest::kSendRtxPayloadType] =
>+          test::CallTest::kVideoSendPayloadType;
>+    }
>+  }
>+
>+  void OnFrameGeneratorCapturerCreated(
>+      test::FrameGeneratorCapturer* frame_generator_capturer) override {
>+    frame_generator_capturer->SetFakeRotation(kVideoRotation_90);
>+  }
>+
>+  void ModifyFlexfecConfigs(
>+      std::vector<FlexfecReceiveStream::Config>* receive_configs) override {
>+    (*receive_configs)[0].local_ssrc = kFlexfecLocalSsrc;
>+  }
>+
>+  void PerformTest() override {
>+    EXPECT_TRUE(Wait())
>+        << "Timed out waiting for dropped frames to be rendered.";
>+  }
>+
>+  rtc::CriticalSection crit_;
>+  std::set<uint32_t> dropped_sequence_numbers_ RTC_GUARDED_BY(crit_);
>+  // Several packets can have the same timestamp.
>+  std::multiset<uint32_t> dropped_timestamps_ RTC_GUARDED_BY(crit_);
>+  const bool enable_nack_;
>+  const bool expect_flexfec_rtcp_;
>+  bool received_flexfec_rtcp_ RTC_GUARDED_BY(crit_);
>+  Random random_;
>+  int num_packets_sent_;
>+};
>+
>+TEST_P(FecEndToEndTest, RecoversWithFlexfec) {
>+  FlexfecRenderObserver test(false, false);
>+  RunBaseTest(&test);
>+}
>+
>+TEST_P(FecEndToEndTest, RecoversWithFlexfecAndNack) {
>+  FlexfecRenderObserver test(true, false);
>+  RunBaseTest(&test);
>+}
>+
>+TEST_P(FecEndToEndTest, RecoversWithFlexfecAndSendsCorrespondingRtcp) {
>+  FlexfecRenderObserver test(false, true);
>+  RunBaseTest(&test);
>+}
>+
>+TEST_P(FecEndToEndTest, ReceivedUlpfecPacketsNotNacked) {
>+  class UlpfecNackObserver : public test::EndToEndTest {
>+   public:
>+    UlpfecNackObserver()
>+        : EndToEndTest(kDefaultTimeoutMs),
>+          state_(kFirstPacket),
>+          ulpfec_sequence_number_(0),
>+          has_last_sequence_number_(false),
>+          last_sequence_number_(0),
>+          encoder_factory_([]() { return VP8Encoder::Create(); }),
>+          decoder_(VP8Decoder::Create()) {}
>+
>+   private:
>+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
>+      rtc::CritScope lock_(&crit_);
>+      RTPHeader header;
>+      EXPECT_TRUE(parser_->Parse(packet, length, &header));
>+
>+      int encapsulated_payload_type = -1;
>+      if (header.payloadType == kRedPayloadType) {
>+        encapsulated_payload_type =
>+            static_cast<int>(packet[header.headerLength]);
>+        if (encapsulated_payload_type != kFakeVideoSendPayloadType)
>+          EXPECT_EQ(kUlpfecPayloadType, encapsulated_payload_type);
>+      } else {
>+        EXPECT_EQ(kFakeVideoSendPayloadType, header.payloadType);
>+      }
>+
>+      if (has_last_sequence_number_ &&
>+          !IsNewerSequenceNumber(header.sequenceNumber,
>+                                 last_sequence_number_)) {
>+        // Drop retransmitted packets.
>+        return DROP_PACKET;
>+      }
>+      last_sequence_number_ = header.sequenceNumber;
>+      has_last_sequence_number_ = true;
>+
>+      bool ulpfec_packet = encapsulated_payload_type == kUlpfecPayloadType;
>+      switch (state_) {
>+        case kFirstPacket:
>+          state_ = kDropEveryOtherPacketUntilUlpfec;
>+          break;
>+        case kDropEveryOtherPacketUntilUlpfec:
>+          if (ulpfec_packet) {
>+            state_ = kDropAllMediaPacketsUntilUlpfec;
>+          } else if (header.sequenceNumber % 2 == 0) {
>+            return DROP_PACKET;
>+          }
>+          break;
>+        case kDropAllMediaPacketsUntilUlpfec:
>+          if (!ulpfec_packet)
>+            return DROP_PACKET;
>+          ulpfec_sequence_number_ = header.sequenceNumber;
>+          state_ = kDropOneMediaPacket;
>+          break;
>+        case kDropOneMediaPacket:
>+          if (ulpfec_packet)
>+            return DROP_PACKET;
>+          state_ = kPassOneMediaPacket;
>+          return DROP_PACKET;
>+          break;
>+        case kPassOneMediaPacket:
>+          if (ulpfec_packet)
>+            return DROP_PACKET;
>+          // Pass one media packet after dropped packet after last FEC,
>+          // otherwise receiver might never see a seq_no after
>+          // |ulpfec_sequence_number_|
>+          state_ = kVerifyUlpfecPacketNotInNackList;
>+          break;
>+        case kVerifyUlpfecPacketNotInNackList:
>+          // Continue to drop packets. Make sure no frame can be decoded.
>+          if (ulpfec_packet || header.sequenceNumber % 2 == 0)
>+            return DROP_PACKET;
>+          break;
>+      }
>+      return SEND_PACKET;
>+    }
>+
>+    Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
>+      rtc::CritScope lock_(&crit_);
>+      if (state_ == kVerifyUlpfecPacketNotInNackList) {
>+        test::RtcpPacketParser rtcp_parser;
>+        rtcp_parser.Parse(packet, length);
>+        const std::vector<uint16_t>& nacks = rtcp_parser.nack()->packet_ids();
>+        EXPECT_TRUE(std::find(nacks.begin(), nacks.end(),
>+                              ulpfec_sequence_number_) == nacks.end())
>+            << "Got nack for ULPFEC packet";
>+        if (!nacks.empty() &&
>+            IsNewerSequenceNumber(nacks.back(), ulpfec_sequence_number_)) {
>+          observation_complete_.Set();
>+        }
>+      }
>+      return SEND_PACKET;
>+    }
>+
>+    test::PacketTransport* CreateSendTransport(
>+        test::SingleThreadedTaskQueueForTesting* task_queue,
>+        Call* sender_call) override {
>+      // At low RTT (< kLowRttNackMs) -> NACK only, no FEC.
>+      // Configure some network delay.
>+      const int kNetworkDelayMs = 50;
>+      FakeNetworkPipe::Config config;
>+      config.queue_delay_ms = kNetworkDelayMs;
>+      return new test::PacketTransport(task_queue, sender_call, this,
>+                                       test::PacketTransport::kSender,
>+                                       payload_type_map_, config);
>+    }
>+
>+    // TODO(holmer): Investigate why we don't send FEC packets when the bitrate
>+    // is 10 kbps.
>+    void ModifySenderCallConfig(Call::Config* config) override {
>+      const int kMinBitrateBps = 30000;
>+      config->bitrate_config.min_bitrate_bps = kMinBitrateBps;
>+    }
>+
>+    void ModifyVideoConfigs(
>+        VideoSendStream::Config* send_config,
>+        std::vector<VideoReceiveStream::Config>* receive_configs,
>+        VideoEncoderConfig* encoder_config) override {
>+      // Configure hybrid NACK/FEC.
>+      send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
>+      send_config->rtp.ulpfec.red_payload_type = kRedPayloadType;
>+      send_config->rtp.ulpfec.ulpfec_payload_type = kUlpfecPayloadType;
>+      // Set codec to VP8, otherwise NACK/FEC hybrid will be disabled.
>+      send_config->encoder_settings.encoder_factory = &encoder_factory_;
>+      send_config->rtp.payload_name = "VP8";
>+      send_config->rtp.payload_type = kFakeVideoSendPayloadType;
>+      encoder_config->codec_type = kVideoCodecVP8;
>+
>+      (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
>+      (*receive_configs)[0].rtp.red_payload_type = kRedPayloadType;
>+      (*receive_configs)[0].rtp.ulpfec_payload_type = kUlpfecPayloadType;
>+
>+      (*receive_configs)[0].decoders.resize(1);
>+      (*receive_configs)[0].decoders[0].payload_type =
>+          send_config->rtp.payload_type;
>+      (*receive_configs)[0].decoders[0].payload_name =
>+          send_config->rtp.payload_name;
>+      (*receive_configs)[0].decoders[0].decoder = decoder_.get();
>+    }
>+
>+    void PerformTest() override {
>+      EXPECT_TRUE(Wait())
>+          << "Timed out while waiting for FEC packets to be received.";
>+    }
>+
>+    enum {
>+      kFirstPacket,
>+      kDropEveryOtherPacketUntilUlpfec,
>+      kDropAllMediaPacketsUntilUlpfec,
>+      kDropOneMediaPacket,
>+      kPassOneMediaPacket,
>+      kVerifyUlpfecPacketNotInNackList,
>+    } state_;
>+
>+    rtc::CriticalSection crit_;
>+    uint16_t ulpfec_sequence_number_ RTC_GUARDED_BY(&crit_);
>+    bool has_last_sequence_number_;
>+    uint16_t last_sequence_number_;
>+    std::unique_ptr<webrtc::VideoEncoder> encoder_;
>+    test::FunctionVideoEncoderFactory encoder_factory_;
>+    std::unique_ptr<webrtc::VideoDecoder> decoder_;
>+  } test;
>+
>+  RunBaseTest(&test);
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/histogram_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/histogram_tests.cc
>new file mode 100644
>index 00000000000..04255a826c2
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/histogram_tests.cc
>@@ -0,0 +1,270 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "absl/types/optional.h"
>+#include "modules/video_coding/codecs/vp8/include/vp8.h"
>+#include "system_wrappers/include/metrics.h"
>+#include "system_wrappers/include/metrics_default.h"
>+#include "test/call_test.h"
>+#include "test/function_video_encoder_factory.h"
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+
>+class HistogramTest : public test::CallTest {
>+ protected:
>+  void VerifyHistogramStats(bool use_rtx, bool use_fec, bool screenshare);
>+};
>+
>+void HistogramTest::VerifyHistogramStats(bool use_rtx,
>+                                         bool use_fec,
>+                                         bool screenshare) {
>+  class FrameObserver : public test::EndToEndTest,
>+                        public rtc::VideoSinkInterface<VideoFrame> {
>+   public:
>+    FrameObserver(bool use_rtx, bool use_fec, bool screenshare)
>+        : EndToEndTest(kLongTimeoutMs),
>+          use_rtx_(use_rtx),
>+          use_fec_(use_fec),
>+          screenshare_(screenshare),
>+          // This test uses NACK, so to send FEC we can't use a fake encoder.
>+          encoder_factory_([]() { return VP8Encoder::Create(); }),
>+          num_frames_received_(0) {}
>+
>+   private:
>+    void OnFrame(const VideoFrame& video_frame) override {
>+      // The RTT is needed to estimate |ntp_time_ms| which is used by
>+      // end-to-end delay stats. Therefore, start counting received frames once
>+      // |ntp_time_ms| is valid.
>+      if (video_frame.ntp_time_ms() > 0 &&
>+          Clock::GetRealTimeClock()->CurrentNtpInMilliseconds() >=
>+              video_frame.ntp_time_ms()) {
>+        rtc::CritScope lock(&crit_);
>+        ++num_frames_received_;
>+      }
>+    }
>+
>+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
>+      if (MinMetricRunTimePassed() && MinNumberOfFramesReceived())
>+        observation_complete_.Set();
>+
>+      return SEND_PACKET;
>+    }
>+
>+    bool MinMetricRunTimePassed() {
>+      int64_t now_ms = Clock::GetRealTimeClock()->TimeInMilliseconds();
>+      if (!start_runtime_ms_)
>+        start_runtime_ms_ = now_ms;
>+
>+      int64_t elapsed_sec = (now_ms - *start_runtime_ms_) / 1000;
>+      return elapsed_sec > metrics::kMinRunTimeInSeconds * 2;
>+    }
>+
>+    bool MinNumberOfFramesReceived() const {
>+      const int kMinRequiredHistogramSamples = 200;
>+      rtc::CritScope lock(&crit_);
>+      return num_frames_received_ > kMinRequiredHistogramSamples;
>+    }
>+
>+    void ModifyVideoConfigs(
>+        VideoSendStream::Config* send_config,
>+        std::vector<VideoReceiveStream::Config>* receive_configs,
>+        VideoEncoderConfig* encoder_config) override {
>+      // NACK
>+      send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
>+      (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
>+      (*receive_configs)[0].renderer = this;
>+      // FEC
>+      if (use_fec_) {
>+        send_config->rtp.ulpfec.ulpfec_payload_type = kUlpfecPayloadType;
>+        send_config->rtp.ulpfec.red_payload_type = kRedPayloadType;
>+        send_config->encoder_settings.encoder_factory = &encoder_factory_;
>+        send_config->rtp.payload_name = "VP8";
>+        encoder_config->codec_type = kVideoCodecVP8;
>+        (*receive_configs)[0].decoders[0].payload_name = "VP8";
>+        (*receive_configs)[0].rtp.red_payload_type = kRedPayloadType;
>+        (*receive_configs)[0].rtp.ulpfec_payload_type = kUlpfecPayloadType;
>+      }
>+      // RTX
>+      if (use_rtx_) {
>+        send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[0]);
>+        send_config->rtp.rtx.payload_type = kSendRtxPayloadType;
>+        (*receive_configs)[0].rtp.rtx_ssrc = kSendRtxSsrcs[0];
>+        (*receive_configs)[0]
>+            .rtp.rtx_associated_payload_types[kSendRtxPayloadType] =
>+            kFakeVideoSendPayloadType;
>+        if (use_fec_) {
>+          send_config->rtp.ulpfec.red_rtx_payload_type = kRtxRedPayloadType;
>+          (*receive_configs)[0]
>+              .rtp.rtx_associated_payload_types[kRtxRedPayloadType] =
>+              kSendRtxPayloadType;
>+        }
>+      }
>+      // RTT needed for RemoteNtpTimeEstimator for the receive stream.
>+      (*receive_configs)[0].rtp.rtcp_xr.receiver_reference_time_report = true;
>+      encoder_config->content_type =
>+          screenshare_ ? VideoEncoderConfig::ContentType::kScreen
>+                       : VideoEncoderConfig::ContentType::kRealtimeVideo;
>+    }
>+
>+    void PerformTest() override {
>+      EXPECT_TRUE(Wait()) << "Timed out waiting for min frames to be received.";
>+    }
>+
>+    rtc::CriticalSection crit_;
>+    const bool use_rtx_;
>+    const bool use_fec_;
>+    const bool screenshare_;
>+    test::FunctionVideoEncoderFactory encoder_factory_;
>+    absl::optional<int64_t> start_runtime_ms_;
>+    int num_frames_received_ RTC_GUARDED_BY(&crit_);
>+  } test(use_rtx, use_fec, screenshare);
>+
>+  metrics::Reset();
>+  RunBaseTest(&test);
>+
>+  const std::string video_prefix =
>+      screenshare ? "WebRTC.Video.Screenshare." : "WebRTC.Video.";
>+  // The content type extension is disabled in non screenshare test,
>+  // therefore no slicing on simulcast id should be present.
>+  const std::string video_suffix = screenshare ? ".S0" : "";
>+
>+  // Verify that stats have been updated once.
>+  EXPECT_EQ(2, metrics::NumSamples("WebRTC.Call.LifetimeInSeconds"));
>+  EXPECT_EQ(1, metrics::NumSamples(
>+                   "WebRTC.Call.TimeReceivingVideoRtpPacketsInSeconds"));
>+  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Call.VideoBitrateReceivedInKbps"));
>+  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Call.RtcpBitrateReceivedInBps"));
>+  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Call.BitrateReceivedInKbps"));
>+  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Call.EstimatedSendBitrateInKbps"));
>+  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Call.PacerBitrateInKbps"));
>+
>+  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.SendStreamLifetimeInSeconds"));
>+  EXPECT_EQ(1,
>+            metrics::NumSamples("WebRTC.Video.ReceiveStreamLifetimeInSeconds"));
>+
>+  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.NackPacketsSentPerMinute"));
>+  EXPECT_EQ(1,
>+            metrics::NumSamples(video_prefix + "NackPacketsReceivedPerMinute"));
>+  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.FirPacketsSentPerMinute"));
>+  EXPECT_EQ(1,
>+            metrics::NumSamples(video_prefix + "FirPacketsReceivedPerMinute"));
>+  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.PliPacketsSentPerMinute"));
>+  EXPECT_EQ(1,
>+            metrics::NumSamples(video_prefix + "PliPacketsReceivedPerMinute"));
>+
>+  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "KeyFramesSentInPermille"));
>+  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.KeyFramesReceivedInPermille"));
>+
>+  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "SentPacketsLostInPercent"));
>+  EXPECT_EQ(1,
>+            metrics::NumSamples("WebRTC.Video.ReceivedPacketsLostInPercent"));
>+
>+  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "InputWidthInPixels"));
>+  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "InputHeightInPixels"));
>+  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "SentWidthInPixels"));
>+  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "SentHeightInPixels"));
>+  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "ReceivedWidthInPixels"));
>+  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "ReceivedHeightInPixels"));
>+
>+  EXPECT_EQ(1, metrics::NumEvents(video_prefix + "InputWidthInPixels",
>+                                  kDefaultWidth));
>+  EXPECT_EQ(1, metrics::NumEvents(video_prefix + "InputHeightInPixels",
>+                                  kDefaultHeight));
>+  EXPECT_EQ(
>+      1, metrics::NumEvents(video_prefix + "SentWidthInPixels", kDefaultWidth));
>+  EXPECT_EQ(1, metrics::NumEvents(video_prefix + "SentHeightInPixels",
>+                                  kDefaultHeight));
>+  EXPECT_EQ(1, metrics::NumEvents(video_prefix + "ReceivedWidthInPixels",
>+                                  kDefaultWidth));
>+  EXPECT_EQ(1, metrics::NumEvents(video_prefix + "ReceivedHeightInPixels",
>+                                  kDefaultHeight));
>+
>+  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "InputFramesPerSecond"));
>+  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "SentFramesPerSecond"));
>+  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.DecodedFramesPerSecond"));
>+  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.RenderFramesPerSecond"));
>+  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.DelayedFramesToRenderer"));
>+
>+  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.JitterBufferDelayInMs"));
>+  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.TargetDelayInMs"));
>+  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.CurrentDelayInMs"));
>+  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.OnewayDelayInMs"));
>+
>+  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "EndToEndDelayInMs" +
>+                                   video_suffix));
>+  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "EndToEndDelayMaxInMs" +
>+                                   video_suffix));
>+  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "InterframeDelayInMs" +
>+                                   video_suffix));
>+  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "InterframeDelayMaxInMs" +
>+                                   video_suffix));
>+
>+  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.RenderSqrtPixelsPerSecond"));
>+
>+  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "EncodeTimeInMs"));
>+  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.DecodeTimeInMs"));
>+
>+  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "NumberOfPauseEvents"));
>+  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "PausedTimeInPercent"));
>+
>+  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "BitrateSentInKbps"));
>+  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.BitrateReceivedInKbps"));
>+  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "MediaBitrateSentInKbps"));
>+  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.MediaBitrateReceivedInKbps"));
>+  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "PaddingBitrateSentInKbps"));
>+  EXPECT_EQ(1,
>+            metrics::NumSamples("WebRTC.Video.PaddingBitrateReceivedInKbps"));
>+  EXPECT_EQ(
>+      1, metrics::NumSamples(video_prefix + "RetransmittedBitrateSentInKbps"));
>+  EXPECT_EQ(1, metrics::NumSamples(
>+                   "WebRTC.Video.RetransmittedBitrateReceivedInKbps"));
>+
>+  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.SendDelayInMs"));
>+  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "SendSideDelayInMs"));
>+  EXPECT_EQ(1, metrics::NumSamples(video_prefix + "SendSideDelayMaxInMs"));
>+
>+  int num_rtx_samples = use_rtx ? 1 : 0;
>+  EXPECT_EQ(num_rtx_samples,
>+            metrics::NumSamples("WebRTC.Video.RtxBitrateSentInKbps"));
>+  EXPECT_EQ(num_rtx_samples,
>+            metrics::NumSamples("WebRTC.Video.RtxBitrateReceivedInKbps"));
>+
>+  int num_red_samples = use_fec ? 1 : 0;
>+  EXPECT_EQ(num_red_samples,
>+            metrics::NumSamples("WebRTC.Video.FecBitrateSentInKbps"));
>+  EXPECT_EQ(num_red_samples,
>+            metrics::NumSamples("WebRTC.Video.FecBitrateReceivedInKbps"));
>+  EXPECT_EQ(num_red_samples,
>+            metrics::NumSamples("WebRTC.Video.ReceivedFecPacketsInPercent"));
>+}
>+
>+TEST_F(HistogramTest, VerifyStatsWithRtx) {
>+  const bool kEnabledRtx = true;
>+  const bool kEnabledRed = false;
>+  const bool kScreenshare = false;
>+  VerifyHistogramStats(kEnabledRtx, kEnabledRed, kScreenshare);
>+}
>+
>+TEST_F(HistogramTest, VerifyStatsWithRed) {
>+  const bool kEnabledRtx = false;
>+  const bool kEnabledRed = true;
>+  const bool kScreenshare = false;
>+  VerifyHistogramStats(kEnabledRtx, kEnabledRed, kScreenshare);
>+}
>+
>+TEST_F(HistogramTest, VerifyStatsWithScreenshare) {
>+  const bool kEnabledRtx = false;
>+  const bool kEnabledRed = false;
>+  const bool kScreenshare = true;
>+  VerifyHistogramStats(kEnabledRtx, kEnabledRed, kScreenshare);
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/log_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/log_tests.cc
>new file mode 100644
>index 00000000000..9f0e0195701
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/log_tests.cc
>@@ -0,0 +1,119 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/video_coding/codecs/vp8/include/vp8.h"
>+#include "rtc_base/file.h"
>+#include "test/call_test.h"
>+#include "test/function_video_encoder_factory.h"
>+#include "test/testsupport/fileutils.h"
>+
>+namespace webrtc {
>+
>+class LogEndToEndTest : public test::CallTest {
>+  void SetUp() { paths_.clear(); }
>+  void TearDown() {
>+    for (const auto& path : paths_) {
>+      rtc::RemoveFile(path);
>+    }
>+  }
>+
>+ public:
>+  int AddFile() {
>+    paths_.push_back(test::TempFilename(test::OutputPath(), "test_file"));
>+    return static_cast<int>(paths_.size()) - 1;
>+  }
>+
>+  rtc::PlatformFile OpenFile(int idx) {
>+    return rtc::OpenPlatformFile(paths_[idx]);
>+  }
>+
>+  void LogSend(bool open) {
>+    if (open) {
>+      GetVideoSendStream()->EnableEncodedFrameRecording(
>+          std::vector<rtc::PlatformFile>(1, OpenFile(AddFile())), 0);
>+    } else {
>+      GetVideoSendStream()->DisableEncodedFrameRecording();
>+    }
>+  }
>+  void LogReceive(bool open) {
>+    if (open) {
>+      video_receive_streams_[0]->EnableEncodedFrameRecording(
>+          OpenFile(AddFile()), 0);
>+    } else {
>+      video_receive_streams_[0]->DisableEncodedFrameRecording();
>+    }
>+  }
>+
>+  std::vector<std::string> paths_;
>+};
>+
>+TEST_F(LogEndToEndTest, LogsEncodedFramesWhenRequested) {
>+  static const int kNumFramesToRecord = 10;
>+  class LogEncodingObserver : public test::EndToEndTest,
>+                              public EncodedFrameObserver {
>+   public:
>+    explicit LogEncodingObserver(LogEndToEndTest* fixture)
>+        : EndToEndTest(kDefaultTimeoutMs),
>+          fixture_(fixture),
>+          encoder_factory_([]() { return VP8Encoder::Create(); }),
>+          recorded_frames_(0) {}
>+
>+    void PerformTest() override {
>+      fixture_->LogSend(true);
>+      fixture_->LogReceive(true);
>+      ASSERT_TRUE(Wait()) << "Timed out while waiting for frame logging.";
>+    }
>+
>+    void ModifyVideoConfigs(
>+        VideoSendStream::Config* send_config,
>+        std::vector<VideoReceiveStream::Config>* receive_configs,
>+        VideoEncoderConfig* encoder_config) override {
>+      decoder_ = VP8Decoder::Create();
>+
>+      send_config->post_encode_callback = this;
>+      send_config->rtp.payload_name = "VP8";
>+      send_config->encoder_settings.encoder_factory = &encoder_factory_;
>+      encoder_config->codec_type = kVideoCodecVP8;
>+
>+      (*receive_configs)[0].decoders.resize(1);
>+      (*receive_configs)[0].decoders[0].payload_type =
>+          send_config->rtp.payload_type;
>+      (*receive_configs)[0].decoders[0].payload_name =
>+          send_config->rtp.payload_name;
>+      (*receive_configs)[0].decoders[0].decoder = decoder_.get();
>+    }
>+
>+    void EncodedFrameCallback(const EncodedFrame& encoded_frame) override {
>+      rtc::CritScope lock(&crit_);
>+      if (recorded_frames_++ > kNumFramesToRecord) {
>+        fixture_->LogSend(false);
>+        fixture_->LogReceive(false);
>+        rtc::File send_file(fixture_->OpenFile(0));
>+        rtc::File receive_file(fixture_->OpenFile(1));
>+        uint8_t out[100];
>+        // If logging has worked correctly neither file should be empty, i.e.
>+        // we should be able to read something from them.
>+        EXPECT_LT(0u, send_file.Read(out, 100));
>+        EXPECT_LT(0u, receive_file.Read(out, 100));
>+        observation_complete_.Set();
>+      }
>+    }
>+
>+   private:
>+    LogEndToEndTest* const fixture_;
>+    test::FunctionVideoEncoderFactory encoder_factory_;
>+    std::unique_ptr<VideoDecoder> decoder_;
>+    rtc::CriticalSection crit_;
>+    int recorded_frames_ RTC_GUARDED_BY(crit_);
>+  } test(this);
>+
>+  RunBaseTest(&test);
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/multi_codec_receive_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/multi_codec_receive_tests.cc
>new file mode 100644
>index 00000000000..d4ab3ee194b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/multi_codec_receive_tests.cc
>@@ -0,0 +1,327 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/video_coding/codecs/h264/include/h264.h"
>+#include "modules/video_coding/codecs/vp8/include/vp8.h"
>+#include "modules/video_coding/codecs/vp9/include/vp9.h"
>+#include "test/call_test.h"
>+#include "test/function_video_encoder_factory.h"
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+namespace {
>+constexpr int kWidth = 1280;
>+constexpr int kHeight = 720;
>+constexpr int kFps = 30;
>+constexpr int kFramesToObserve = 10;
>+
>+uint8_t PayloadNameToPayloadType(const std::string& payload_name) {
>+  if (payload_name == "VP8") {
>+    return test::CallTest::kPayloadTypeVP8;
>+  } else if (payload_name == "VP9") {
>+    return test::CallTest::kPayloadTypeVP9;
>+  } else if (payload_name == "H264") {
>+    return test::CallTest::kPayloadTypeH264;
>+  } else {
>+    RTC_NOTREACHED();
>+    return 0;
>+  }
>+}
>+
>+int RemoveOlderOrEqual(uint32_t timestamp, std::vector<uint32_t>* timestamps) {
>+  int num_removed = 0;
>+  while (!timestamps->empty()) {
>+    auto it = timestamps->begin();
>+    if (IsNewerTimestamp(*it, timestamp))
>+      break;
>+
>+    timestamps->erase(it);
>+    ++num_removed;
>+  }
>+  return num_removed;
>+}
>+
>+class VideoStreamFactoryTest
>+    : public VideoEncoderConfig::VideoStreamFactoryInterface {
>+ public:
>+  explicit VideoStreamFactoryTest(size_t num_temporal_layers)
>+      : num_temporal_layers_(num_temporal_layers) {}
>+
>+ private:
>+  std::vector<VideoStream> CreateEncoderStreams(
>+      int width,
>+      int height,
>+      const VideoEncoderConfig& encoder_config) override {
>+    std::vector<VideoStream> streams =
>+        test::CreateVideoStreams(width, height, encoder_config);
>+
>+    for (size_t i = 0; i < encoder_config.number_of_streams; ++i)
>+      streams[i].num_temporal_layers = num_temporal_layers_;
>+
>+    return streams;
>+  }
>+
>+  const size_t num_temporal_layers_;
>+};
>+
>+class FrameObserver : public test::RtpRtcpObserver,
>+                      public rtc::VideoSinkInterface<VideoFrame> {
>+ public:
>+  FrameObserver() : test::RtpRtcpObserver(test::CallTest::kDefaultTimeoutMs) {}
>+
>+  void Reset() {
>+    rtc::CritScope lock(&crit_);
>+    num_sent_frames_ = 0;
>+    num_rendered_frames_ = 0;
>+  }
>+
>+ private:
>+  // Sends kFramesToObserve.
>+  Action OnSendRtp(const uint8_t* packet, size_t length) override {
>+    rtc::CritScope lock(&crit_);
>+
>+    RTPHeader header;
>+    EXPECT_TRUE(parser_->Parse(packet, length, &header));
>+    EXPECT_EQ(header.ssrc, test::CallTest::kVideoSendSsrcs[0]);
>+    EXPECT_GE(length, header.headerLength + header.paddingLength);
>+    if ((length - header.headerLength) == header.paddingLength)
>+      return SEND_PACKET;  // Skip padding, may be sent after OnFrame is called.
>+
>+    if (!last_timestamp_ || header.timestamp != *last_timestamp_) {
>+      // New frame.
>+      if (last_payload_type_) {
>+        bool new_payload_type = header.payloadType != *last_payload_type_;
>+        EXPECT_EQ(num_sent_frames_ == 0, new_payload_type)
>+            << "Payload type should change after reset.";
>+      }
>+      // Sent enough frames?
>+      if (num_sent_frames_ >= kFramesToObserve)
>+        return DROP_PACKET;
>+
>+      ++num_sent_frames_;
>+      sent_timestamps_.push_back(header.timestamp);
>+    }
>+
>+    last_timestamp_ = header.timestamp;
>+    last_payload_type_ = header.payloadType;
>+    return SEND_PACKET;
>+  }
>+
>+  // Verifies that all sent frames are decoded and rendered.
>+  void OnFrame(const VideoFrame& rendered_frame) override {
>+    rtc::CritScope lock(&crit_);
>+    EXPECT_NE(std::find(sent_timestamps_.begin(), sent_timestamps_.end(),
>+                        rendered_frame.timestamp()),
>+              sent_timestamps_.end());
>+
>+    // Remove old timestamps too, only the newest decoded frame is rendered.
>+    num_rendered_frames_ +=
>+        RemoveOlderOrEqual(rendered_frame.timestamp(), &sent_timestamps_);
>+
>+    if (num_rendered_frames_ >= kFramesToObserve) {
>+      EXPECT_TRUE(sent_timestamps_.empty()) << "All sent frames not decoded.";
>+      observation_complete_.Set();
>+    }
>+  }
>+
>+  rtc::CriticalSection crit_;
>+  absl::optional<uint32_t> last_timestamp_;
>+  absl::optional<uint8_t> last_payload_type_;
>+  int num_sent_frames_ RTC_GUARDED_BY(crit_) = 0;
>+  int num_rendered_frames_ RTC_GUARDED_BY(crit_) = 0;
>+  std::vector<uint32_t> sent_timestamps_ RTC_GUARDED_BY(crit_);
>+};
>+}  // namespace
>+
>+class MultiCodecReceiveTest : public test::CallTest {
>+ public:
>+  MultiCodecReceiveTest() {
>+    task_queue_.SendTask([this]() {
>+      CreateCalls();
>+
>+      send_transport_.reset(new test::PacketTransport(
>+          &task_queue_, sender_call_.get(), &observer_,
>+          test::PacketTransport::kSender, kPayloadTypeMap,
>+          FakeNetworkPipe::Config()));
>+      send_transport_->SetReceiver(receiver_call_->Receiver());
>+
>+      receive_transport_.reset(new test::PacketTransport(
>+          &task_queue_, receiver_call_.get(), &observer_,
>+          test::PacketTransport::kReceiver, kPayloadTypeMap,
>+          FakeNetworkPipe::Config()));
>+      receive_transport_->SetReceiver(sender_call_->Receiver());
>+    });
>+  }
>+
>+  virtual ~MultiCodecReceiveTest() {
>+    task_queue_.SendTask([this]() {
>+      send_transport_.reset();
>+      receive_transport_.reset();
>+      DestroyCalls();
>+    });
>+  }
>+
>+  struct CodecConfig {
>+    std::string payload_name;
>+    VideoEncoderFactory* encoder_factory;
>+    size_t num_temporal_layers;
>+  };
>+
>+  void ConfigureEncoder(const CodecConfig& config);
>+  void ConfigureDecoders(const std::vector<CodecConfig>& configs);
>+  void RunTestWithCodecs(const std::vector<CodecConfig>& configs);
>+
>+ private:
>+  const std::map<uint8_t, MediaType> kPayloadTypeMap = {
>+      {CallTest::kPayloadTypeVP8, MediaType::VIDEO},
>+      {CallTest::kPayloadTypeVP9, MediaType::VIDEO},
>+      {CallTest::kPayloadTypeH264, MediaType::VIDEO}};
>+  FrameObserver observer_;
>+};
>+
>+void MultiCodecReceiveTest::ConfigureDecoders(
>+    const std::vector<CodecConfig>& configs) {
>+  // Placing the payload names in a std::set retains the unique names only.
>+  std::set<std::string> unique_payload_names;
>+  for (const auto& config : configs)
>+    unique_payload_names.insert(config.payload_name);
>+
>+  video_receive_configs_[0].decoders.clear();
>+  for (const auto& payload_name : unique_payload_names) {
>+    VideoReceiveStream::Decoder decoder = test::CreateMatchingDecoder(
>+        PayloadNameToPayloadType(payload_name), payload_name);
>+    allocated_decoders_.push_back(
>+        std::unique_ptr<VideoDecoder>(decoder.decoder));
>+    video_receive_configs_[0].decoders.push_back(decoder);
>+  }
>+}
>+
>+void MultiCodecReceiveTest::ConfigureEncoder(const CodecConfig& config) {
>+  GetVideoSendConfig()->encoder_settings.encoder_factory =
>+      config.encoder_factory;
>+  GetVideoSendConfig()->rtp.payload_name = config.payload_name;
>+  GetVideoSendConfig()->rtp.payload_type =
>+      PayloadNameToPayloadType(config.payload_name);
>+  GetVideoEncoderConfig()->codec_type =
>+      PayloadStringToCodecType(config.payload_name);
>+  GetVideoEncoderConfig()->video_stream_factory =
>+      new rtc::RefCountedObject<VideoStreamFactoryTest>(
>+          config.num_temporal_layers);
>+}
>+
>+void MultiCodecReceiveTest::RunTestWithCodecs(
>+    const std::vector<CodecConfig>& configs) {
>+  EXPECT_TRUE(!configs.empty());
>+
>+  // Create and start call.
>+  task_queue_.SendTask([this, &configs]() {
>+    CreateSendConfig(1, 0, 0, send_transport_.get());
>+    ConfigureEncoder(configs[0]);
>+    CreateMatchingReceiveConfigs(receive_transport_.get());
>+    video_receive_configs_[0].renderer = &observer_;
>+    ConfigureDecoders(configs);
>+    CreateVideoStreams();
>+    CreateFrameGeneratorCapturer(kFps, kWidth, kHeight);
>+    Start();
>+  });
>+  EXPECT_TRUE(observer_.Wait()) << "Timed out waiting for frames.";
>+
>+  for (size_t i = 1; i < configs.size(); ++i) {
>+    // Recreate VideoSendStream with new config (codec, temporal layers).
>+    task_queue_.SendTask([this, i, &configs]() {
>+      frame_generator_capturer_->Stop();
>+      DestroyVideoSendStreams();
>+      observer_.Reset();
>+
>+      ConfigureEncoder(configs[i]);
>+      CreateVideoSendStreams();
>+      GetVideoSendStream()->Start();
>+      CreateFrameGeneratorCapturer(kFps, kWidth / 2, kHeight / 2);
>+      ConnectVideoSourcesToStreams();
>+      frame_generator_capturer_->Start();
>+    });
>+    EXPECT_TRUE(observer_.Wait()) << "Timed out waiting for frames.";
>+  }
>+
>+  task_queue_.SendTask([this]() {
>+    Stop();
>+    DestroyStreams();
>+  });
>+}
>+
>+TEST_F(MultiCodecReceiveTest, SingleStreamReceivesVp8Vp9) {
>+  test::FunctionVideoEncoderFactory vp8_encoder_factory(
>+      []() { return VP8Encoder::Create(); });
>+  test::FunctionVideoEncoderFactory vp9_encoder_factory(
>+      []() { return VP9Encoder::Create(); });
>+  RunTestWithCodecs({{"VP8", &vp8_encoder_factory, 1},
>+                     {"VP9", &vp9_encoder_factory, 1},
>+                     {"VP8", &vp8_encoder_factory, 1}});
>+}
>+
>+TEST_F(MultiCodecReceiveTest, SingleStreamReceivesVp8Vp9WithTl) {
>+  test::FunctionVideoEncoderFactory vp8_encoder_factory(
>+      []() { return VP8Encoder::Create(); });
>+  test::FunctionVideoEncoderFactory vp9_encoder_factory(
>+      []() { return VP9Encoder::Create(); });
>+  RunTestWithCodecs({{"VP8", &vp8_encoder_factory, 2},
>+                     {"VP9", &vp9_encoder_factory, 2},
>+                     {"VP8", &vp8_encoder_factory, 2}});
>+}
>+
>+#if defined(WEBRTC_USE_H264)
>+TEST_F(MultiCodecReceiveTest, SingleStreamReceivesVp8H264) {
>+  test::FunctionVideoEncoderFactory vp8_encoder_factory(
>+      []() { return VP8Encoder::Create(); });
>+  test::FunctionVideoEncoderFactory h264_encoder_factory(
>+      []() { return H264Encoder::Create(cricket::VideoCodec("H264")); });
>+  RunTestWithCodecs({{"VP8", &vp8_encoder_factory, 1},
>+                     {"H264", &h264_encoder_factory, 1},
>+                     {"VP8", &vp8_encoder_factory, 1}});
>+}
>+
>+TEST_F(MultiCodecReceiveTest, SingleStreamReceivesVp8H264WithTl) {
>+  test::FunctionVideoEncoderFactory vp8_encoder_factory(
>+      []() { return VP8Encoder::Create(); });
>+  test::FunctionVideoEncoderFactory h264_encoder_factory(
>+      []() { return H264Encoder::Create(cricket::VideoCodec("H264")); });
>+  RunTestWithCodecs({{"VP8", &vp8_encoder_factory, 3},
>+                     {"H264", &h264_encoder_factory, 1},
>+                     {"VP8", &vp8_encoder_factory, 3}});
>+}
>+
>+TEST_F(MultiCodecReceiveTest, SingleStreamReceivesVp8Vp9H264) {
>+  test::FunctionVideoEncoderFactory vp8_encoder_factory(
>+      []() { return VP8Encoder::Create(); });
>+  test::FunctionVideoEncoderFactory vp9_encoder_factory(
>+      []() { return VP9Encoder::Create(); });
>+  test::FunctionVideoEncoderFactory h264_encoder_factory(
>+      []() { return H264Encoder::Create(cricket::VideoCodec("H264")); });
>+  RunTestWithCodecs({{"VP8", &vp8_encoder_factory, 1},
>+                     {"VP9", &vp9_encoder_factory, 1},
>+                     {"H264", &h264_encoder_factory, 1},
>+                     {"VP9", &vp9_encoder_factory, 1}});
>+}
>+
>+TEST_F(MultiCodecReceiveTest, SingleStreamReceivesVp8Vp9H264WithTl) {
>+  test::FunctionVideoEncoderFactory vp8_encoder_factory(
>+      []() { return VP8Encoder::Create(); });
>+  test::FunctionVideoEncoderFactory vp9_encoder_factory(
>+      []() { return VP9Encoder::Create(); });
>+  test::FunctionVideoEncoderFactory h264_encoder_factory(
>+      []() { return H264Encoder::Create(cricket::VideoCodec("H264")); });
>+  RunTestWithCodecs({{"VP8", &vp8_encoder_factory, 3},
>+                     {"VP9", &vp9_encoder_factory, 2},
>+                     {"H264", &h264_encoder_factory, 1},
>+                     {"VP9", &vp9_encoder_factory, 3}});
>+}
>+#endif  // defined(WEBRTC_USE_H264)
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/multi_stream_tester.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/multi_stream_tester.cc
>new file mode 100644
>index 00000000000..57387aa4172
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/multi_stream_tester.cc
>@@ -0,0 +1,150 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "video/end_to_end_tests/multi_stream_tester.h"
>+
>+#include <memory>
>+#include <utility>
>+#include <vector>
>+
>+#include "logging/rtc_event_log/rtc_event_log.h"
>+#include "modules/video_coding/codecs/vp8/include/vp8.h"
>+#include "test/call_test.h"
>+#include "test/encoder_settings.h"
>+#include "test/function_video_encoder_factory.h"
>+
>+namespace webrtc {
>+
>+MultiStreamTester::MultiStreamTester(
>+    test::SingleThreadedTaskQueueForTesting* task_queue)
>+    : task_queue_(task_queue) {
>+  // TODO(sprang): Cleanup when msvc supports explicit initializers for array.
>+  codec_settings[0] = {1, 640, 480};
>+  codec_settings[1] = {2, 320, 240};
>+  codec_settings[2] = {3, 240, 160};
>+  class multi_stream_test {
>+   public:
>+    multi_stream_test();
>+  };
>+}
>+
>+MultiStreamTester::~MultiStreamTester() {}
>+
>+void MultiStreamTester::RunTest() {
>+  webrtc::RtcEventLogNullImpl event_log;
>+  Call::Config config(&event_log);
>+  std::unique_ptr<Call> sender_call;
>+  std::unique_ptr<Call> receiver_call;
>+  std::unique_ptr<test::DirectTransport> sender_transport;
>+  std::unique_ptr<test::DirectTransport> receiver_transport;
>+
>+  VideoSendStream* send_streams[kNumStreams];
>+  VideoReceiveStream* receive_streams[kNumStreams];
>+  test::FrameGeneratorCapturer* frame_generators[kNumStreams];
>+  std::vector<std::unique_ptr<VideoDecoder>> allocated_decoders;
>+  test::FunctionVideoEncoderFactory encoder_factory(
>+      []() { return VP8Encoder::Create(); });
>+
>+  task_queue_->SendTask([&]() {
>+    sender_call = absl::WrapUnique(Call::Create(config));
>+    receiver_call = absl::WrapUnique(Call::Create(config));
>+    sender_transport =
>+        absl::WrapUnique(CreateSendTransport(task_queue_, sender_call.get()));
>+    receiver_transport = absl::WrapUnique(
>+        CreateReceiveTransport(task_queue_, receiver_call.get()));
>+
>+    sender_transport->SetReceiver(receiver_call->Receiver());
>+    receiver_transport->SetReceiver(sender_call->Receiver());
>+
>+    for (size_t i = 0; i < kNumStreams; ++i) {
>+      uint32_t ssrc = codec_settings[i].ssrc;
>+      int width = codec_settings[i].width;
>+      int height = codec_settings[i].height;
>+
>+      VideoSendStream::Config send_config(sender_transport.get());
>+      send_config.rtp.ssrcs.push_back(ssrc);
>+      send_config.encoder_settings.encoder_factory = &encoder_factory;
>+      send_config.rtp.payload_name = "VP8";
>+      send_config.rtp.payload_type = kVideoPayloadType;
>+      VideoEncoderConfig encoder_config;
>+      test::FillEncoderConfiguration(kVideoCodecVP8, 1, &encoder_config);
>+      encoder_config.max_bitrate_bps = 100000;
>+
>+      UpdateSendConfig(i, &send_config, &encoder_config, &frame_generators[i]);
>+
>+      send_streams[i] = sender_call->CreateVideoSendStream(
>+          send_config.Copy(), encoder_config.Copy());
>+      send_streams[i]->Start();
>+
>+      VideoReceiveStream::Config receive_config(receiver_transport.get());
>+      receive_config.rtp.remote_ssrc = ssrc;
>+      receive_config.rtp.local_ssrc = test::CallTest::kReceiverLocalVideoSsrc;
>+      VideoReceiveStream::Decoder decoder =
>+          test::CreateMatchingDecoder(send_config);
>+      allocated_decoders.push_back(
>+          std::unique_ptr<VideoDecoder>(decoder.decoder));
>+      receive_config.decoders.push_back(decoder);
>+
>+      UpdateReceiveConfig(i, &receive_config);
>+
>+      receive_streams[i] =
>+          receiver_call->CreateVideoReceiveStream(std::move(receive_config));
>+      receive_streams[i]->Start();
>+
>+      frame_generators[i] = test::FrameGeneratorCapturer::Create(
>+          width, height, absl::nullopt, absl::nullopt, 30,
>+          Clock::GetRealTimeClock());
>+      send_streams[i]->SetSource(frame_generators[i],
>+                                 DegradationPreference::MAINTAIN_FRAMERATE);
>+      frame_generators[i]->Start();
>+    }
>+  });
>+
>+  Wait();
>+
>+  task_queue_->SendTask([&]() {
>+    for (size_t i = 0; i < kNumStreams; ++i) {
>+      frame_generators[i]->Stop();
>+      sender_call->DestroyVideoSendStream(send_streams[i]);
>+      receiver_call->DestroyVideoReceiveStream(receive_streams[i]);
>+      delete frame_generators[i];
>+    }
>+
>+    sender_transport.reset();
>+    receiver_transport.reset();
>+
>+    sender_call.reset();
>+    receiver_call.reset();
>+  });
>+}
>+
>+void MultiStreamTester::UpdateSendConfig(
>+    size_t stream_index,
>+    VideoSendStream::Config* send_config,
>+    VideoEncoderConfig* encoder_config,
>+    test::FrameGeneratorCapturer** frame_generator) {}
>+
>+void MultiStreamTester::UpdateReceiveConfig(
>+    size_t stream_index,
>+    VideoReceiveStream::Config* receive_config) {}
>+
>+test::DirectTransport* MultiStreamTester::CreateSendTransport(
>+    test::SingleThreadedTaskQueueForTesting* task_queue,
>+    Call* sender_call) {
>+  return new test::DirectTransport(task_queue, sender_call, payload_type_map_);
>+}
>+
>+test::DirectTransport* MultiStreamTester::CreateReceiveTransport(
>+    test::SingleThreadedTaskQueueForTesting* task_queue,
>+    Call* receiver_call) {
>+  return new test::DirectTransport(task_queue, receiver_call,
>+                                   payload_type_map_);
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/multi_stream_tester.h b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/multi_stream_tester.h
>new file mode 100644
>index 00000000000..5bdef1191a2
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/multi_stream_tester.h
>@@ -0,0 +1,64 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef VIDEO_END_TO_END_TESTS_MULTI_STREAM_TESTER_H_
>+#define VIDEO_END_TO_END_TESTS_MULTI_STREAM_TESTER_H_
>+
>+#include <map>
>+
>+#include "call/call.h"
>+#include "test/direct_transport.h"
>+#include "test/frame_generator_capturer.h"
>+
>+namespace webrtc {
>+// Test sets up a Call multiple senders with different resolutions and SSRCs.
>+// Another is set up to receive all three of these with different renderers.
>+class MultiStreamTester {
>+ public:
>+  static constexpr size_t kNumStreams = 3;
>+  const uint8_t kVideoPayloadType = 124;
>+  const std::map<uint8_t, MediaType> payload_type_map_ = {
>+      {kVideoPayloadType, MediaType::VIDEO}};
>+
>+  struct CodecSettings {
>+    uint32_t ssrc;
>+    int width;
>+    int height;
>+  } codec_settings[kNumStreams];
>+
>+  explicit MultiStreamTester(
>+      test::SingleThreadedTaskQueueForTesting* task_queue);
>+
>+  virtual ~MultiStreamTester();
>+
>+  void RunTest();
>+
>+ protected:
>+  virtual void Wait() = 0;
>+  // Note: frame_generator is a point-to-pointer, since the actual instance
>+  // hasn't been created at the time of this call. Only when packets/frames
>+  // start flowing should this be dereferenced.
>+  virtual void UpdateSendConfig(size_t stream_index,
>+                                VideoSendStream::Config* send_config,
>+                                VideoEncoderConfig* encoder_config,
>+                                test::FrameGeneratorCapturer** frame_generator);
>+  virtual void UpdateReceiveConfig(size_t stream_index,
>+                                   VideoReceiveStream::Config* receive_config);
>+  virtual test::DirectTransport* CreateSendTransport(
>+      test::SingleThreadedTaskQueueForTesting* task_queue,
>+      Call* sender_call);
>+  virtual test::DirectTransport* CreateReceiveTransport(
>+      test::SingleThreadedTaskQueueForTesting* task_queue,
>+      Call* receiver_call);
>+
>+  test::SingleThreadedTaskQueueForTesting* const task_queue_;
>+};
>+}  // namespace webrtc
>+#endif  // VIDEO_END_TO_END_TESTS_MULTI_STREAM_TESTER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/multi_stream_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/multi_stream_tests.cc
>new file mode 100644
>index 00000000000..0c925b36e57
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/multi_stream_tests.cc
>@@ -0,0 +1,102 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "logging/rtc_event_log/rtc_event_log.h"
>+#include "modules/video_coding/codecs/vp8/include/vp8.h"
>+#include "test/call_test.h"
>+#include "test/encoder_settings.h"
>+#include "test/field_trial.h"
>+#include "test/gtest.h"
>+#include "video/end_to_end_tests/multi_stream_tester.h"
>+
>+namespace webrtc {
>+class MultiStreamEndToEndTest
>+    : public test::CallTest,
>+      public testing::WithParamInterface<std::string> {
>+ public:
>+  MultiStreamEndToEndTest() : field_trial_(GetParam()) {}
>+
>+ private:
>+  test::ScopedFieldTrials field_trial_;
>+};
>+
>+INSTANTIATE_TEST_CASE_P(RoundRobin,
>+                        MultiStreamEndToEndTest,
>+                        ::testing::Values("WebRTC-RoundRobinPacing/Disabled/",
>+                                          "WebRTC-RoundRobinPacing/Enabled/"));
>+
>+// Each renderer verifies that it receives the expected resolution, and as soon
>+// as every renderer has received a frame, the test finishes.
>+TEST_P(MultiStreamEndToEndTest, SendsAndReceivesMultipleStreams) {
>+  class VideoOutputObserver : public rtc::VideoSinkInterface<VideoFrame> {
>+   public:
>+    VideoOutputObserver(const MultiStreamTester::CodecSettings& settings,
>+                        uint32_t ssrc,
>+                        test::FrameGeneratorCapturer** frame_generator)
>+        : settings_(settings),
>+          ssrc_(ssrc),
>+          frame_generator_(frame_generator),
>+          done_(false, false) {}
>+
>+    void OnFrame(const VideoFrame& video_frame) override {
>+      EXPECT_EQ(settings_.width, video_frame.width());
>+      EXPECT_EQ(settings_.height, video_frame.height());
>+      (*frame_generator_)->Stop();
>+      done_.Set();
>+    }
>+
>+    uint32_t Ssrc() { return ssrc_; }
>+
>+    bool Wait() { return done_.Wait(kDefaultTimeoutMs); }
>+
>+   private:
>+    const MultiStreamTester::CodecSettings& settings_;
>+    const uint32_t ssrc_;
>+    test::FrameGeneratorCapturer** const frame_generator_;
>+    rtc::Event done_;
>+  };
>+
>+  class Tester : public MultiStreamTester {
>+   public:
>+    explicit Tester(test::SingleThreadedTaskQueueForTesting* task_queue)
>+        : MultiStreamTester(task_queue) {}
>+    virtual ~Tester() {}
>+
>+   protected:
>+    void Wait() override {
>+      for (const auto& observer : observers_) {
>+        EXPECT_TRUE(observer->Wait())
>+            << "Time out waiting for from on ssrc " << observer->Ssrc();
>+      }
>+    }
>+
>+    void UpdateSendConfig(
>+        size_t stream_index,
>+        VideoSendStream::Config* send_config,
>+        VideoEncoderConfig* encoder_config,
>+        test::FrameGeneratorCapturer** frame_generator) override {
>+      observers_[stream_index].reset(new VideoOutputObserver(
>+          codec_settings[stream_index], send_config->rtp.ssrcs.front(),
>+          frame_generator));
>+    }
>+
>+    void UpdateReceiveConfig(
>+        size_t stream_index,
>+        VideoReceiveStream::Config* receive_config) override {
>+      receive_config->renderer = observers_[stream_index].get();
>+    }
>+
>+   private:
>+    std::unique_ptr<VideoOutputObserver> observers_[kNumStreams];
>+  } tester(&task_queue_);
>+
>+  tester.RunTest();
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/network_state_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/network_state_tests.cc
>new file mode 100644
>index 00000000000..8d681612ba6
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/network_state_tests.cc
>@@ -0,0 +1,414 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "system_wrappers/include/sleep.h"
>+#include "test/call_test.h"
>+#include "test/encoder_proxy_factory.h"
>+#include "test/fake_encoder.h"
>+#include "test/gtest.h"
>+
>+namespace webrtc {
>+namespace {
>+constexpr int kSilenceTimeoutMs = 2000;
>+}
>+
>+class NetworkStateEndToEndTest : public test::CallTest {
>+ protected:
>+  class UnusedTransport : public Transport {
>+   private:
>+    bool SendRtp(const uint8_t* packet,
>+                 size_t length,
>+                 const PacketOptions& options) override {
>+      ADD_FAILURE() << "Unexpected RTP sent.";
>+      return false;
>+    }
>+
>+    bool SendRtcp(const uint8_t* packet, size_t length) override {
>+      ADD_FAILURE() << "Unexpected RTCP sent.";
>+      return false;
>+    }
>+  };
>+  class RequiredTransport : public Transport {
>+   public:
>+    RequiredTransport(bool rtp_required, bool rtcp_required)
>+        : need_rtp_(rtp_required), need_rtcp_(rtcp_required) {}
>+    ~RequiredTransport() {
>+      if (need_rtp_) {
>+        ADD_FAILURE() << "Expected RTP packet not sent.";
>+      }
>+      if (need_rtcp_) {
>+        ADD_FAILURE() << "Expected RTCP packet not sent.";
>+      }
>+    }
>+
>+   private:
>+    bool SendRtp(const uint8_t* packet,
>+                 size_t length,
>+                 const PacketOptions& options) override {
>+      rtc::CritScope lock(&crit_);
>+      need_rtp_ = false;
>+      return true;
>+    }
>+
>+    bool SendRtcp(const uint8_t* packet, size_t length) override {
>+      rtc::CritScope lock(&crit_);
>+      need_rtcp_ = false;
>+      return true;
>+    }
>+    bool need_rtp_;
>+    bool need_rtcp_;
>+    rtc::CriticalSection crit_;
>+  };
>+  void VerifyNewVideoSendStreamsRespectNetworkState(
>+      MediaType network_to_bring_up,
>+      VideoEncoder* encoder,
>+      Transport* transport);
>+  void VerifyNewVideoReceiveStreamsRespectNetworkState(
>+      MediaType network_to_bring_up,
>+      Transport* transport);
>+};
>+
>+void NetworkStateEndToEndTest::VerifyNewVideoSendStreamsRespectNetworkState(
>+    MediaType network_to_bring_up,
>+    VideoEncoder* encoder,
>+    Transport* transport) {
>+  test::EncoderProxyFactory encoder_factory(encoder);
>+
>+  task_queue_.SendTask([this, network_to_bring_up, &encoder_factory,
>+                        transport]() {
>+    CreateSenderCall(Call::Config(send_event_log_.get()));
>+    sender_call_->SignalChannelNetworkState(network_to_bring_up, kNetworkUp);
>+
>+    CreateSendConfig(1, 0, 0, transport);
>+    GetVideoSendConfig()->encoder_settings.encoder_factory = &encoder_factory;
>+    CreateVideoStreams();
>+    CreateFrameGeneratorCapturer(kDefaultFramerate, kDefaultWidth,
>+                                 kDefaultHeight);
>+
>+    Start();
>+  });
>+
>+  SleepMs(kSilenceTimeoutMs);
>+
>+  task_queue_.SendTask([this]() {
>+    Stop();
>+    DestroyStreams();
>+    DestroyCalls();
>+  });
>+}
>+
>+void NetworkStateEndToEndTest::VerifyNewVideoReceiveStreamsRespectNetworkState(
>+    MediaType network_to_bring_up,
>+    Transport* transport) {
>+  std::unique_ptr<test::DirectTransport> sender_transport;
>+
>+  task_queue_.SendTask([this, &sender_transport, network_to_bring_up,
>+                        transport]() {
>+    CreateCalls();
>+    receiver_call_->SignalChannelNetworkState(network_to_bring_up, kNetworkUp);
>+    sender_transport = absl::make_unique<test::DirectTransport>(
>+        &task_queue_, sender_call_.get(), payload_type_map_);
>+    sender_transport->SetReceiver(receiver_call_->Receiver());
>+    CreateSendConfig(1, 0, 0, sender_transport.get());
>+    CreateMatchingReceiveConfigs(transport);
>+    CreateVideoStreams();
>+    CreateFrameGeneratorCapturer(kDefaultFramerate, kDefaultWidth,
>+                                 kDefaultHeight);
>+    Start();
>+  });
>+
>+  SleepMs(kSilenceTimeoutMs);
>+
>+  task_queue_.SendTask([this, &sender_transport]() {
>+    Stop();
>+    DestroyStreams();
>+    sender_transport.reset();
>+    DestroyCalls();
>+  });
>+}
>+
>+TEST_F(NetworkStateEndToEndTest, RespectsNetworkState) {
>+  // TODO(pbos): Remove accepted downtime packets etc. when signaling network
>+  // down blocks until no more packets will be sent.
>+
>+  // Pacer will send from its packet list and then send required padding before
>+  // checking paused_ again. This should be enough for one round of pacing,
>+  // otherwise increase.
>+  static const int kNumAcceptedDowntimeRtp = 5;
>+  // A single RTCP may be in the pipeline.
>+  static const int kNumAcceptedDowntimeRtcp = 1;
>+  class NetworkStateTest : public test::EndToEndTest, public test::FakeEncoder {
>+   public:
>+    explicit NetworkStateTest(
>+        test::SingleThreadedTaskQueueForTesting* task_queue)
>+        : EndToEndTest(kDefaultTimeoutMs),
>+          FakeEncoder(Clock::GetRealTimeClock()),
>+          task_queue_(task_queue),
>+          encoded_frames_(false, false),
>+          packet_event_(false, false),
>+          sender_call_(nullptr),
>+          receiver_call_(nullptr),
>+          encoder_factory_(this),
>+          sender_state_(kNetworkUp),
>+          sender_rtp_(0),
>+          sender_padding_(0),
>+          sender_rtcp_(0),
>+          receiver_rtcp_(0),
>+          down_frames_(0) {}
>+
>+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
>+      rtc::CritScope lock(&test_crit_);
>+      RTPHeader header;
>+      EXPECT_TRUE(parser_->Parse(packet, length, &header));
>+      if (length == header.headerLength + header.paddingLength)
>+        ++sender_padding_;
>+      ++sender_rtp_;
>+      packet_event_.Set();
>+      return SEND_PACKET;
>+    }
>+
>+    Action OnSendRtcp(const uint8_t* packet, size_t length) override {
>+      rtc::CritScope lock(&test_crit_);
>+      ++sender_rtcp_;
>+      packet_event_.Set();
>+      return SEND_PACKET;
>+    }
>+
>+    Action OnReceiveRtp(const uint8_t* packet, size_t length) override {
>+      ADD_FAILURE() << "Unexpected receiver RTP, should not be sending.";
>+      return SEND_PACKET;
>+    }
>+
>+    Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
>+      rtc::CritScope lock(&test_crit_);
>+      ++receiver_rtcp_;
>+      packet_event_.Set();
>+      return SEND_PACKET;
>+    }
>+
>+    void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
>+      sender_call_ = sender_call;
>+      receiver_call_ = receiver_call;
>+    }
>+
>+    void ModifyVideoConfigs(
>+        VideoSendStream::Config* send_config,
>+        std::vector<VideoReceiveStream::Config>* receive_configs,
>+        VideoEncoderConfig* encoder_config) override {
>+      send_config->encoder_settings.encoder_factory = &encoder_factory_;
>+    }
>+
>+    void PerformTest() override {
>+      EXPECT_TRUE(encoded_frames_.Wait(kDefaultTimeoutMs))
>+          << "No frames received by the encoder.";
>+
>+      task_queue_->SendTask([this]() {
>+        // Wait for packets from both sender/receiver.
>+        WaitForPacketsOrSilence(false, false);
>+
>+        // Sender-side network down for audio; there should be no effect on
>+        // video
>+        sender_call_->SignalChannelNetworkState(MediaType::AUDIO, kNetworkDown);
>+        WaitForPacketsOrSilence(false, false);
>+
>+        // Receiver-side network down for audio; no change expected
>+        receiver_call_->SignalChannelNetworkState(MediaType::AUDIO,
>+                                                  kNetworkDown);
>+        WaitForPacketsOrSilence(false, false);
>+
>+        // Sender-side network down.
>+        sender_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkDown);
>+        {
>+          rtc::CritScope lock(&test_crit_);
>+          // After network goes down we shouldn't be encoding more frames.
>+          sender_state_ = kNetworkDown;
>+        }
>+        // Wait for receiver-packets and no sender packets.
>+        WaitForPacketsOrSilence(true, false);
>+
>+        // Receiver-side network down.
>+        receiver_call_->SignalChannelNetworkState(MediaType::VIDEO,
>+                                                  kNetworkDown);
>+        WaitForPacketsOrSilence(true, true);
>+
>+        // Network up for audio for both sides; video is still not expected to
>+        // start
>+        sender_call_->SignalChannelNetworkState(MediaType::AUDIO, kNetworkUp);
>+        receiver_call_->SignalChannelNetworkState(MediaType::AUDIO, kNetworkUp);
>+        WaitForPacketsOrSilence(true, true);
>+
>+        // Network back up again for both.
>+        {
>+          rtc::CritScope lock(&test_crit_);
>+          // It's OK to encode frames again, as we're about to bring up the
>+          // network.
>+          sender_state_ = kNetworkUp;
>+        }
>+        sender_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
>+        receiver_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
>+        WaitForPacketsOrSilence(false, false);
>+
>+        // TODO(skvlad): add tests to verify that the audio streams are stopped
>+        // when the network goes down for audio once the workaround in
>+        // paced_sender.cc is removed.
>+      });
>+    }
>+
>+    int32_t Encode(const VideoFrame& input_image,
>+                   const CodecSpecificInfo* codec_specific_info,
>+                   const std::vector<FrameType>* frame_types) override {
>+      {
>+        rtc::CritScope lock(&test_crit_);
>+        if (sender_state_ == kNetworkDown) {
>+          ++down_frames_;
>+          EXPECT_LE(down_frames_, 1)
>+              << "Encoding more than one frame while network is down.";
>+          if (down_frames_ > 1)
>+            encoded_frames_.Set();
>+        } else {
>+          encoded_frames_.Set();
>+        }
>+      }
>+      return test::FakeEncoder::Encode(input_image, codec_specific_info,
>+                                       frame_types);
>+    }
>+
>+   private:
>+    void WaitForPacketsOrSilence(bool sender_down, bool receiver_down) {
>+      int64_t initial_time_ms = clock_->TimeInMilliseconds();
>+      int initial_sender_rtp;
>+      int initial_sender_rtcp;
>+      int initial_receiver_rtcp;
>+      {
>+        rtc::CritScope lock(&test_crit_);
>+        initial_sender_rtp = sender_rtp_;
>+        initial_sender_rtcp = sender_rtcp_;
>+        initial_receiver_rtcp = receiver_rtcp_;
>+      }
>+      bool sender_done = false;
>+      bool receiver_done = false;
>+      while (!sender_done || !receiver_done) {
>+        packet_event_.Wait(kSilenceTimeoutMs);
>+        int64_t time_now_ms = clock_->TimeInMilliseconds();
>+        rtc::CritScope lock(&test_crit_);
>+        if (sender_down) {
>+          ASSERT_LE(sender_rtp_ - initial_sender_rtp - sender_padding_,
>+                    kNumAcceptedDowntimeRtp)
>+              << "RTP sent during sender-side downtime.";
>+          ASSERT_LE(sender_rtcp_ - initial_sender_rtcp,
>+                    kNumAcceptedDowntimeRtcp)
>+              << "RTCP sent during sender-side downtime.";
>+          if (time_now_ms - initial_time_ms >=
>+              static_cast<int64_t>(kSilenceTimeoutMs)) {
>+            sender_done = true;
>+          }
>+        } else {
>+          if (sender_rtp_ > initial_sender_rtp + kNumAcceptedDowntimeRtp)
>+            sender_done = true;
>+        }
>+        if (receiver_down) {
>+          ASSERT_LE(receiver_rtcp_ - initial_receiver_rtcp,
>+                    kNumAcceptedDowntimeRtcp)
>+              << "RTCP sent during receiver-side downtime.";
>+          if (time_now_ms - initial_time_ms >=
>+              static_cast<int64_t>(kSilenceTimeoutMs)) {
>+            receiver_done = true;
>+          }
>+        } else {
>+          if (receiver_rtcp_ > initial_receiver_rtcp + kNumAcceptedDowntimeRtcp)
>+            receiver_done = true;
>+        }
>+      }
>+    }
>+
>+    test::SingleThreadedTaskQueueForTesting* const task_queue_;
>+    rtc::CriticalSection test_crit_;
>+    rtc::Event encoded_frames_;
>+    rtc::Event packet_event_;
>+    Call* sender_call_;
>+    Call* receiver_call_;
>+    test::EncoderProxyFactory encoder_factory_;
>+    NetworkState sender_state_ RTC_GUARDED_BY(test_crit_);
>+    int sender_rtp_ RTC_GUARDED_BY(test_crit_);
>+    int sender_padding_ RTC_GUARDED_BY(test_crit_);
>+    int sender_rtcp_ RTC_GUARDED_BY(test_crit_);
>+    int receiver_rtcp_ RTC_GUARDED_BY(test_crit_);
>+    int down_frames_ RTC_GUARDED_BY(test_crit_);
>+  } test(&task_queue_);
>+
>+  RunBaseTest(&test);
>+}
>+
>+TEST_F(NetworkStateEndToEndTest, NewVideoSendStreamsRespectVideoNetworkDown) {
>+  class UnusedEncoder : public test::FakeEncoder {
>+   public:
>+    UnusedEncoder() : FakeEncoder(Clock::GetRealTimeClock()) {}
>+
>+    int32_t InitEncode(const VideoCodec* config,
>+                       int32_t number_of_cores,
>+                       size_t max_payload_size) override {
>+      EXPECT_GT(config->startBitrate, 0u);
>+      return 0;
>+    }
>+    int32_t Encode(const VideoFrame& input_image,
>+                   const CodecSpecificInfo* codec_specific_info,
>+                   const std::vector<FrameType>* frame_types) override {
>+      ADD_FAILURE() << "Unexpected frame encode.";
>+      return test::FakeEncoder::Encode(input_image, codec_specific_info,
>+                                       frame_types);
>+    }
>+  };
>+
>+  UnusedEncoder unused_encoder;
>+  UnusedTransport unused_transport;
>+  VerifyNewVideoSendStreamsRespectNetworkState(
>+      MediaType::AUDIO, &unused_encoder, &unused_transport);
>+}
>+
>+TEST_F(NetworkStateEndToEndTest, NewVideoSendStreamsIgnoreAudioNetworkDown) {
>+  class RequiredEncoder : public test::FakeEncoder {
>+   public:
>+    RequiredEncoder()
>+        : FakeEncoder(Clock::GetRealTimeClock()), encoded_frame_(false) {}
>+    ~RequiredEncoder() {
>+      if (!encoded_frame_) {
>+        ADD_FAILURE() << "Didn't encode an expected frame";
>+      }
>+    }
>+    int32_t Encode(const VideoFrame& input_image,
>+                   const CodecSpecificInfo* codec_specific_info,
>+                   const std::vector<FrameType>* frame_types) override {
>+      encoded_frame_ = true;
>+      return test::FakeEncoder::Encode(input_image, codec_specific_info,
>+                                       frame_types);
>+    }
>+
>+   private:
>+    bool encoded_frame_;
>+  };
>+
>+  RequiredTransport required_transport(true /*rtp*/, false /*rtcp*/);
>+  RequiredEncoder required_encoder;
>+  VerifyNewVideoSendStreamsRespectNetworkState(
>+      MediaType::VIDEO, &required_encoder, &required_transport);
>+}
>+
>+TEST_F(NetworkStateEndToEndTest,
>+       NewVideoReceiveStreamsRespectVideoNetworkDown) {
>+  UnusedTransport transport;
>+  VerifyNewVideoReceiveStreamsRespectNetworkState(MediaType::AUDIO, &transport);
>+}
>+
>+TEST_F(NetworkStateEndToEndTest, NewVideoReceiveStreamsIgnoreAudioNetworkDown) {
>+  RequiredTransport transport(false /*rtp*/, true /*rtcp*/);
>+  VerifyNewVideoReceiveStreamsRespectNetworkState(MediaType::VIDEO, &transport);
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/probing_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/probing_tests.cc
>new file mode 100644
>index 00000000000..4d0009f41ca
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/probing_tests.cc
>@@ -0,0 +1,303 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "test/call_test.h"
>+#include "test/field_trial.h"
>+#include "test/gtest.h"
>+namespace webrtc {
>+
>+class ProbingEndToEndTest : public test::CallTest,
>+                            public testing::WithParamInterface<std::string> {
>+ public:
>+  ProbingEndToEndTest() : field_trial_(GetParam()) {}
>+
>+  virtual ~ProbingEndToEndTest() {
>+  }
>+
>+ private:
>+  test::ScopedFieldTrials field_trial_;
>+};
>+INSTANTIATE_TEST_CASE_P(
>+    FieldTrials,
>+    ProbingEndToEndTest,
>+    ::testing::Values("WebRTC-RoundRobinPacing/Disabled/",
>+                      "WebRTC-RoundRobinPacing/Enabled/",
>+                      "WebRTC-TaskQueueCongestionControl/Enabled/"));
>+
>+class ProbingTest : public test::EndToEndTest {
>+ public:
>+  explicit ProbingTest(int start_bitrate_bps)
>+      : clock_(Clock::GetRealTimeClock()),
>+        start_bitrate_bps_(start_bitrate_bps),
>+        state_(0),
>+        sender_call_(nullptr) {}
>+
>+  void ModifySenderCallConfig(Call::Config* config) override {
>+    config->bitrate_config.start_bitrate_bps = start_bitrate_bps_;
>+  }
>+
>+  void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
>+    sender_call_ = sender_call;
>+  }
>+
>+ protected:
>+  Clock* const clock_;
>+  const int start_bitrate_bps_;
>+  int state_;
>+  Call* sender_call_;
>+};
>+
>+// Flaky under MemorySanitizer: bugs.webrtc.org/7419
>+// Flaky on iOS bots: bugs.webrtc.org/7851
>+#if defined(MEMORY_SANITIZER)
>+TEST_P(ProbingEndToEndTest, DISABLED_InitialProbing) {
>+#elif defined(TARGET_IPHONE_SIMULATOR) && TARGET_IPHONE_SIMULATOR
>+TEST_P(ProbingEndToEndTest, DISABLED_InitialProbing) {
>+#else
>+TEST_P(ProbingEndToEndTest, InitialProbing) {
>+#endif
>+  class InitialProbingTest : public ProbingTest {
>+   public:
>+    explicit InitialProbingTest(bool* success)
>+        : ProbingTest(300000), success_(success) {
>+      *success_ = false;
>+    }
>+
>+    void PerformTest() override {
>+      int64_t start_time_ms = clock_->TimeInMilliseconds();
>+      do {
>+        if (clock_->TimeInMilliseconds() - start_time_ms > kTimeoutMs)
>+          break;
>+
>+        Call::Stats stats = sender_call_->GetStats();
>+        // Initial probing is done with a x3 and x6 multiplier of the start
>+        // bitrate, so a x4 multiplier is a high enough threshold.
>+        if (stats.send_bandwidth_bps > 4 * 300000) {
>+          *success_ = true;
>+          break;
>+        }
>+      } while (!observation_complete_.Wait(20));
>+    }
>+
>+   private:
>+    const int kTimeoutMs = 1000;
>+    bool* const success_;
>+  };
>+
>+  bool success = false;
>+  const int kMaxAttempts = 3;
>+  for (int i = 0; i < kMaxAttempts; ++i) {
>+    InitialProbingTest test(&success);
>+    RunBaseTest(&test);
>+    if (success)
>+      return;
>+  }
>+  EXPECT_TRUE(success) << "Failed to perform mid initial probing ("
>+                       << kMaxAttempts << " attempts).";
>+}
>+
>+// Fails on Linux MSan: bugs.webrtc.org/7428
>+#if defined(MEMORY_SANITIZER)
>+TEST_P(ProbingEndToEndTest, DISABLED_TriggerMidCallProbing) {
>+// Fails on iOS bots: bugs.webrtc.org/7851
>+#elif defined(TARGET_IPHONE_SIMULATOR) && TARGET_IPHONE_SIMULATOR
>+TEST_P(ProbingEndToEndTest, DISABLED_TriggerMidCallProbing) {
>+#else
>+TEST_P(ProbingEndToEndTest, TriggerMidCallProbing) {
>+#endif
>+
>+  class TriggerMidCallProbingTest : public ProbingTest {
>+   public:
>+    TriggerMidCallProbingTest(
>+        test::SingleThreadedTaskQueueForTesting* task_queue,
>+        bool* success)
>+        : ProbingTest(300000), success_(success), task_queue_(task_queue) {}
>+
>+    void PerformTest() override {
>+      *success_ = false;
>+      int64_t start_time_ms = clock_->TimeInMilliseconds();
>+      do {
>+        if (clock_->TimeInMilliseconds() - start_time_ms > kTimeoutMs)
>+          break;
>+
>+        Call::Stats stats = sender_call_->GetStats();
>+
>+        switch (state_) {
>+          case 0:
>+            if (stats.send_bandwidth_bps > 5 * 300000) {
>+              BitrateConstraints bitrate_config;
>+              bitrate_config.max_bitrate_bps = 100000;
>+              task_queue_->SendTask([this, &bitrate_config]() {
>+                sender_call_->GetTransportControllerSend()
>+                    ->SetSdpBitrateParameters(bitrate_config);
>+              });
>+              ++state_;
>+            }
>+            break;
>+          case 1:
>+            if (stats.send_bandwidth_bps < 110000) {
>+              BitrateConstraints bitrate_config;
>+              bitrate_config.max_bitrate_bps = 2500000;
>+              task_queue_->SendTask([this, &bitrate_config]() {
>+                sender_call_->GetTransportControllerSend()
>+                    ->SetSdpBitrateParameters(bitrate_config);
>+              });
>+              ++state_;
>+            }
>+            break;
>+          case 2:
>+            // During high cpu load the pacer will not be able to pace packets
>+            // at the correct speed, but if we go from 110 to 1250 kbps
>+            // in 5 seconds then it is due to probing.
>+            if (stats.send_bandwidth_bps > 1250000) {
>+              *success_ = true;
>+              observation_complete_.Set();
>+            }
>+            break;
>+        }
>+      } while (!observation_complete_.Wait(20));
>+    }
>+
>+   private:
>+    const int kTimeoutMs = 5000;
>+    bool* const success_;
>+    test::SingleThreadedTaskQueueForTesting* const task_queue_;
>+  };
>+
>+  bool success = false;
>+  const int kMaxAttempts = 3;
>+  for (int i = 0; i < kMaxAttempts; ++i) {
>+    TriggerMidCallProbingTest test(&task_queue_, &success);
>+    RunBaseTest(&test);
>+    if (success)
>+      return;
>+  }
>+  EXPECT_TRUE(success) << "Failed to perform mid call probing (" << kMaxAttempts
>+                       << " attempts).";
>+}
>+
>+#if defined(MEMORY_SANITIZER)
>+TEST_P(ProbingEndToEndTest, DISABLED_ProbeOnVideoEncoderReconfiguration) {
>+#elif defined(TARGET_IPHONE_SIMULATOR) && TARGET_IPHONE_SIMULATOR
>+TEST_P(ProbingEndToEndTest, DISABLED_ProbeOnVideoEncoderReconfiguration) {
>+#else
>+TEST_P(ProbingEndToEndTest, ProbeOnVideoEncoderReconfiguration) {
>+#endif
>+
>+  class ReconfigureTest : public ProbingTest {
>+   public:
>+    ReconfigureTest(test::SingleThreadedTaskQueueForTesting* task_queue,
>+                    bool* success)
>+        : ProbingTest(50000), task_queue_(task_queue), success_(success) {}
>+
>+    void ModifyVideoConfigs(
>+        VideoSendStream::Config* send_config,
>+        std::vector<VideoReceiveStream::Config>* receive_configs,
>+        VideoEncoderConfig* encoder_config) override {
>+      encoder_config_ = encoder_config;
>+    }
>+
>+    void OnVideoStreamsCreated(
>+        VideoSendStream* send_stream,
>+        const std::vector<VideoReceiveStream*>& receive_streams) override {
>+      send_stream_ = send_stream;
>+    }
>+
>+    void OnRtpTransportControllerSendCreated(
>+        RtpTransportControllerSend* transport_controller) override {
>+      transport_controller_ = transport_controller;
>+    }
>+
>+    test::PacketTransport* CreateSendTransport(
>+        test::SingleThreadedTaskQueueForTesting* task_queue,
>+        Call* sender_call) override {
>+      send_transport_ = new test::PacketTransport(
>+          task_queue, sender_call, this, test::PacketTransport::kSender,
>+          CallTest::payload_type_map_, FakeNetworkPipe::Config());
>+      return send_transport_;
>+    }
>+
>+    void PerformTest() override {
>+      *success_ = false;
>+      int64_t start_time_ms = clock_->TimeInMilliseconds();
>+      do {
>+        if (clock_->TimeInMilliseconds() - start_time_ms > kTimeoutMs)
>+          break;
>+
>+        Call::Stats stats = sender_call_->GetStats();
>+
>+        switch (state_) {
>+          case 0:
>+            // Wait until initial probing has been completed (6 times start
>+            // bitrate).
>+            if (stats.send_bandwidth_bps >= 250000 &&
>+                stats.send_bandwidth_bps <= 350000) {
>+              FakeNetworkPipe::Config config;
>+              config.link_capacity_kbps = 200;
>+              send_transport_->SetConfig(config);
>+
>+              // In order to speed up the test we can interrupt exponential
>+              // probing by toggling the network availability. The alternative
>+              // is to wait for it to time out (1000 ms).
>+              transport_controller_->OnNetworkAvailability(false);
>+              transport_controller_->OnNetworkAvailability(true);
>+
>+              ++state_;
>+            }
>+            break;
>+          case 1:
>+            if (stats.send_bandwidth_bps <= 210000) {
>+              FakeNetworkPipe::Config config;
>+              config.link_capacity_kbps = 5000;
>+              send_transport_->SetConfig(config);
>+
>+              encoder_config_->max_bitrate_bps = 2000000;
>+              encoder_config_->simulcast_layers[0].max_bitrate_bps = 1200000;
>+              task_queue_->SendTask([this]() {
>+                send_stream_->ReconfigureVideoEncoder(encoder_config_->Copy());
>+              });
>+
>+              ++state_;
>+            }
>+            break;
>+          case 2:
>+            if (stats.send_bandwidth_bps >= 1000000) {
>+              *success_ = true;
>+              observation_complete_.Set();
>+            }
>+            break;
>+        }
>+      } while (!observation_complete_.Wait(20));
>+    }
>+
>+   private:
>+    const int kTimeoutMs = 3000;
>+    test::SingleThreadedTaskQueueForTesting* const task_queue_;
>+    bool* const success_;
>+    test::PacketTransport* send_transport_;
>+    VideoSendStream* send_stream_;
>+    VideoEncoderConfig* encoder_config_;
>+    RtpTransportControllerSend* transport_controller_;
>+  };
>+
>+  bool success = false;
>+  const int kMaxAttempts = 3;
>+  for (int i = 0; i < kMaxAttempts; ++i) {
>+    ReconfigureTest test(&task_queue_, &success);
>+    RunBaseTest(&test);
>+    if (success) {
>+      return;
>+    }
>+  }
>+  EXPECT_TRUE(success) << "Failed to perform mid call probing (" << kMaxAttempts
>+                       << " attempts).";
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/receive_time_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/receive_time_tests.cc
>new file mode 100644
>index 00000000000..9e025cb553a
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/receive_time_tests.cc
>@@ -0,0 +1,141 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "rtc_base/criticalsection.h"
>+#include "rtc_base/timeutils.h"
>+#include "test/call_test.h"
>+#include "test/field_trial.h"
>+#include "test/rtcp_packet_parser.h"
>+
>+namespace webrtc {
>+namespace {
>+
>+// This tester simulates a series of clock reset events where different offsets
>+// are added to the receive time. It detects jumps in the resulting reported
>+// receive times of more than 200 ms.
>+class ReportedReceiveTimeTester : public test::EndToEndTest {
>+ public:
>+  struct TimeJump {
>+    int64_t at_send_time_ms;
>+    int64_t add_offset_ms;
>+    static constexpr int64_t kStop = 0;
>+  };
>+
>+  ReportedReceiveTimeTester()
>+      : EndToEndTest(test::CallTest::kDefaultTimeoutMs) {
>+    // These should be let trough without correction and filtered if correction
>+    // is enabled.
>+    jumps_.push({500, 2000});
>+    jumps_.push({1000, -400});
>+    jumps_.push({1500, 2000000});
>+    jumps_.push({1700, TimeJump::kStop});
>+  }
>+  bool JumpInReportedTimes() { return jump_in_reported_times_; }
>+
>+ protected:
>+  Action OnReceiveRtcp(const uint8_t* data, size_t length) override {
>+    test::RtcpPacketParser parser;
>+    EXPECT_TRUE(parser.Parse(data, length));
>+    const auto& fb = parser.transport_feedback();
>+    if (fb->num_packets() > 0) {
>+      int64_t arrival_time_us = fb->GetBaseTimeUs();
>+      for (const auto& pkt : fb->GetReceivedPackets()) {
>+        arrival_time_us += pkt.delta_us();
>+        if (last_arrival_time_us_ != 0) {
>+          int64_t delta_us = arrival_time_us - last_arrival_time_us_;
>+          rtc::CritScope crit(&send_times_crit_);
>+          if (send_times_us_.size() >= 2) {
>+            int64_t ground_truth_delta_us =
>+                send_times_us_[1] - send_times_us_[0];
>+            send_times_us_.pop_front();
>+            int64_t delta_diff_ms = (delta_us - ground_truth_delta_us) / 1000;
>+            if (std::abs(delta_diff_ms) > 200) {
>+              jump_in_reported_times_ = true;
>+              observation_complete_.Set();
>+            }
>+          }
>+        }
>+        last_arrival_time_us_ = arrival_time_us;
>+      }
>+    }
>+    return SEND_PACKET;
>+  }
>+  Action OnSendRtp(const uint8_t* data, size_t length) override {
>+    {
>+      rtc::CritScope crit(&send_times_crit_);
>+      send_times_us_.push_back(rtc::TimeMicros());
>+    }
>+    int64_t now_ms = rtc::TimeMillis();
>+    if (!first_send_time_ms_)
>+      first_send_time_ms_ = now_ms;
>+    int64_t send_time_ms = now_ms - first_send_time_ms_;
>+    if (send_time_ms >= jumps_.front().at_send_time_ms) {
>+      if (jumps_.front().add_offset_ms == TimeJump::kStop) {
>+        observation_complete_.Set();
>+        jumps_.pop();
>+        return SEND_PACKET;
>+      }
>+      clock_offset_ms_ += jumps_.front().add_offset_ms;
>+      send_transport_->SetClockOffset(clock_offset_ms_);
>+      jumps_.pop();
>+    }
>+    return SEND_PACKET;
>+  }
>+  test::PacketTransport* CreateSendTransport(
>+      test::SingleThreadedTaskQueueForTesting* task_queue,
>+      Call* sender_call) override {
>+    return send_transport_ = new test::PacketTransport(
>+               task_queue, sender_call, this, test::PacketTransport::kSender,
>+               test::CallTest::payload_type_map_, FakeNetworkPipe::Config());
>+  }
>+  void PerformTest() override {
>+    observation_complete_.Wait(test::CallTest::kDefaultTimeoutMs);
>+  }
>+  size_t GetNumVideoStreams() const override { return 1; }
>+  size_t GetNumAudioStreams() const override { return 0; }
>+
>+ private:
>+  int64_t last_arrival_time_us_ = 0;
>+  int64_t first_send_time_ms_ = 0;
>+  rtc::CriticalSection send_times_crit_;
>+  std::deque<int64_t> send_times_us_ RTC_GUARDED_BY(send_times_crit_);
>+  bool jump_in_reported_times_ = false;
>+  test::PacketTransport* send_transport_;
>+  int64_t clock_offset_ms_ = 0;
>+  std::queue<TimeJump> jumps_;
>+};
>+}  // namespace
>+
>+class ReceiveTimeEndToEndTest : public test::CallTest {
>+ public:
>+  ReceiveTimeEndToEndTest() {}
>+
>+  virtual ~ReceiveTimeEndToEndTest() {}
>+};
>+
>+TEST_F(ReceiveTimeEndToEndTest, ReceiveTimeJumpsWithoutFieldTrial) {
>+  // Without the field trial, the jumps in clock offset should be let trough and
>+  // be detected.
>+  ReportedReceiveTimeTester test;
>+  RunBaseTest(&test);
>+  EXPECT_TRUE(test.JumpInReportedTimes());
>+}
>+
>+TEST_F(ReceiveTimeEndToEndTest, ReceiveTimeSteadyWithFieldTrial) {
>+  // Since all the added jumps by the tester are outside the interval of -100 ms
>+  // to 1000 ms, they should all be filtered by the field trial below, and no
>+  // jumps should be detected.
>+  test::ScopedFieldTrials field_trial(
>+      "WebRTC-BweReceiveTimeCorrection/Enabled,-100,1000/");
>+  ReportedReceiveTimeTester test;
>+  RunBaseTest(&test);
>+  EXPECT_FALSE(test.JumpInReportedTimes());
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/retransmission_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/retransmission_tests.cc
>new file mode 100644
>index 00000000000..7145b022114
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/retransmission_tests.cc
>@@ -0,0 +1,494 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/video_coding/codecs/vp8/include/vp8.h"
>+#include "system_wrappers/include/sleep.h"
>+#include "test/call_test.h"
>+#include "test/field_trial.h"
>+#include "test/function_video_encoder_factory.h"
>+#include "test/gtest.h"
>+#include "test/rtcp_packet_parser.h"
>+
>+namespace webrtc {
>+class RetransmissionEndToEndTest
>+    : public test::CallTest,
>+      public testing::WithParamInterface<std::string> {
>+ public:
>+  RetransmissionEndToEndTest() : field_trial_(GetParam()) {}
>+
>+ protected:
>+  void DecodesRetransmittedFrame(bool enable_rtx, bool enable_red);
>+  void ReceivesPliAndRecovers(int rtp_history_ms);
>+
>+ private:
>+ private:
>+  test::ScopedFieldTrials field_trial_;
>+};
>+
>+INSTANTIATE_TEST_CASE_P(RoundRobin,
>+                        RetransmissionEndToEndTest,
>+                        ::testing::Values("WebRTC-RoundRobinPacing/Disabled/",
>+                                          "WebRTC-RoundRobinPacing/Enabled/"));
>+
>+TEST_P(RetransmissionEndToEndTest, ReceivesAndRetransmitsNack) {
>+  static const int kNumberOfNacksToObserve = 2;
>+  static const int kLossBurstSize = 2;
>+  static const int kPacketsBetweenLossBursts = 9;
>+  class NackObserver : public test::EndToEndTest {
>+   public:
>+    NackObserver()
>+        : EndToEndTest(kLongTimeoutMs),
>+          sent_rtp_packets_(0),
>+          packets_left_to_drop_(0),
>+          nacks_left_(kNumberOfNacksToObserve) {}
>+
>+   private:
>+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
>+      rtc::CritScope lock(&crit_);
>+      RTPHeader header;
>+      EXPECT_TRUE(parser_->Parse(packet, length, &header));
>+
>+      // Never drop retransmitted packets.
>+      if (dropped_packets_.find(header.sequenceNumber) !=
>+          dropped_packets_.end()) {
>+        retransmitted_packets_.insert(header.sequenceNumber);
>+        return SEND_PACKET;
>+      }
>+
>+      if (nacks_left_ <= 0 &&
>+          retransmitted_packets_.size() == dropped_packets_.size()) {
>+        observation_complete_.Set();
>+      }
>+
>+      ++sent_rtp_packets_;
>+
>+      // Enough NACKs received, stop dropping packets.
>+      if (nacks_left_ <= 0)
>+        return SEND_PACKET;
>+
>+      // Check if it's time for a new loss burst.
>+      if (sent_rtp_packets_ % kPacketsBetweenLossBursts == 0)
>+        packets_left_to_drop_ = kLossBurstSize;
>+
>+      // Never drop padding packets as those won't be retransmitted.
>+      if (packets_left_to_drop_ > 0 && header.paddingLength == 0) {
>+        --packets_left_to_drop_;
>+        dropped_packets_.insert(header.sequenceNumber);
>+        return DROP_PACKET;
>+      }
>+
>+      return SEND_PACKET;
>+    }
>+
>+    Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
>+      rtc::CritScope lock(&crit_);
>+      test::RtcpPacketParser parser;
>+      EXPECT_TRUE(parser.Parse(packet, length));
>+      nacks_left_ -= parser.nack()->num_packets();
>+      return SEND_PACKET;
>+    }
>+
>+    void ModifyVideoConfigs(
>+        VideoSendStream::Config* send_config,
>+        std::vector<VideoReceiveStream::Config>* receive_configs,
>+        VideoEncoderConfig* encoder_config) override {
>+      send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
>+      (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
>+    }
>+
>+    void PerformTest() override {
>+      EXPECT_TRUE(Wait())
>+          << "Timed out waiting for packets to be NACKed, retransmitted and "
>+             "rendered.";
>+    }
>+
>+    rtc::CriticalSection crit_;
>+    std::set<uint16_t> dropped_packets_;
>+    std::set<uint16_t> retransmitted_packets_;
>+    uint64_t sent_rtp_packets_;
>+    int packets_left_to_drop_;
>+    int nacks_left_ RTC_GUARDED_BY(&crit_);
>+  } test;
>+
>+  RunBaseTest(&test);
>+}
>+
>+TEST_P(RetransmissionEndToEndTest, ReceivesNackAndRetransmitsAudio) {
>+  class NackObserver : public test::EndToEndTest {
>+   public:
>+    NackObserver()
>+        : EndToEndTest(kLongTimeoutMs),
>+          local_ssrc_(0),
>+          remote_ssrc_(0),
>+          receive_transport_(nullptr) {}
>+
>+   private:
>+    size_t GetNumVideoStreams() const override { return 0; }
>+    size_t GetNumAudioStreams() const override { return 1; }
>+
>+    test::PacketTransport* CreateReceiveTransport(
>+        test::SingleThreadedTaskQueueForTesting* task_queue) override {
>+      test::PacketTransport* receive_transport = new test::PacketTransport(
>+          task_queue, nullptr, this, test::PacketTransport::kReceiver,
>+          payload_type_map_, FakeNetworkPipe::Config());
>+      receive_transport_ = receive_transport;
>+      return receive_transport;
>+    }
>+
>+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
>+      RTPHeader header;
>+      EXPECT_TRUE(parser_->Parse(packet, length, &header));
>+
>+      if (!sequence_number_to_retransmit_) {
>+        sequence_number_to_retransmit_ = header.sequenceNumber;
>+
>+        // Don't ask for retransmission straight away, may be deduped in pacer.
>+      } else if (header.sequenceNumber == *sequence_number_to_retransmit_) {
>+        observation_complete_.Set();
>+      } else {
>+        // Send a NACK as often as necessary until retransmission is received.
>+        rtcp::Nack nack;
>+        nack.SetSenderSsrc(local_ssrc_);
>+        nack.SetMediaSsrc(remote_ssrc_);
>+        uint16_t nack_list[] = {*sequence_number_to_retransmit_};
>+        nack.SetPacketIds(nack_list, 1);
>+        rtc::Buffer buffer = nack.Build();
>+
>+        EXPECT_TRUE(receive_transport_->SendRtcp(buffer.data(), buffer.size()));
>+      }
>+
>+      return SEND_PACKET;
>+    }
>+
>+    void ModifyAudioConfigs(
>+        AudioSendStream::Config* send_config,
>+        std::vector<AudioReceiveStream::Config>* receive_configs) override {
>+      send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
>+      (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
>+      local_ssrc_ = (*receive_configs)[0].rtp.local_ssrc;
>+      remote_ssrc_ = (*receive_configs)[0].rtp.remote_ssrc;
>+    }
>+
>+    void PerformTest() override {
>+      EXPECT_TRUE(Wait())
>+          << "Timed out waiting for packets to be NACKed, retransmitted and "
>+             "rendered.";
>+    }
>+
>+    uint32_t local_ssrc_;
>+    uint32_t remote_ssrc_;
>+    Transport* receive_transport_;
>+    absl::optional<uint16_t> sequence_number_to_retransmit_;
>+  } test;
>+
>+  RunBaseTest(&test);
>+}
>+
>+TEST_P(RetransmissionEndToEndTest,
>+       StopSendingKeyframeRequestsForInactiveStream) {
>+  class KeyframeRequestObserver : public test::EndToEndTest {
>+   public:
>+    explicit KeyframeRequestObserver(
>+        test::SingleThreadedTaskQueueForTesting* task_queue)
>+        : clock_(Clock::GetRealTimeClock()), task_queue_(task_queue) {}
>+
>+    void OnVideoStreamsCreated(
>+        VideoSendStream* send_stream,
>+        const std::vector<VideoReceiveStream*>& receive_streams) override {
>+      RTC_DCHECK_EQ(1, receive_streams.size());
>+      send_stream_ = send_stream;
>+      receive_stream_ = receive_streams[0];
>+    }
>+
>+    void PerformTest() override {
>+      bool frame_decoded = false;
>+      int64_t start_time = clock_->TimeInMilliseconds();
>+      while (clock_->TimeInMilliseconds() - start_time <= 5000) {
>+        if (receive_stream_->GetStats().frames_decoded > 0) {
>+          frame_decoded = true;
>+          break;
>+        }
>+        SleepMs(100);
>+      }
>+      ASSERT_TRUE(frame_decoded);
>+      task_queue_->SendTask([this]() { send_stream_->Stop(); });
>+      SleepMs(10000);
>+      ASSERT_EQ(
>+          1U, receive_stream_->GetStats().rtcp_packet_type_counts.pli_packets);
>+    }
>+
>+   private:
>+    Clock* clock_;
>+    VideoSendStream* send_stream_;
>+    VideoReceiveStream* receive_stream_;
>+    test::SingleThreadedTaskQueueForTesting* const task_queue_;
>+  } test(&task_queue_);
>+
>+  RunBaseTest(&test);
>+}
>+
>+void RetransmissionEndToEndTest::ReceivesPliAndRecovers(int rtp_history_ms) {
>+  static const int kPacketsToDrop = 1;
>+
>+  class PliObserver : public test::EndToEndTest,
>+                      public rtc::VideoSinkInterface<VideoFrame> {
>+   public:
>+    explicit PliObserver(int rtp_history_ms)
>+        : EndToEndTest(kLongTimeoutMs),
>+          rtp_history_ms_(rtp_history_ms),
>+          nack_enabled_(rtp_history_ms > 0),
>+          highest_dropped_timestamp_(0),
>+          frames_to_drop_(0),
>+          received_pli_(false) {}
>+
>+   private:
>+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
>+      rtc::CritScope lock(&crit_);
>+      RTPHeader header;
>+      EXPECT_TRUE(parser_->Parse(packet, length, &header));
>+
>+      // Drop all retransmitted packets to force a PLI.
>+      if (header.timestamp <= highest_dropped_timestamp_)
>+        return DROP_PACKET;
>+
>+      if (frames_to_drop_ > 0) {
>+        highest_dropped_timestamp_ = header.timestamp;
>+        --frames_to_drop_;
>+        return DROP_PACKET;
>+      }
>+
>+      return SEND_PACKET;
>+    }
>+
>+    Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
>+      rtc::CritScope lock(&crit_);
>+      test::RtcpPacketParser parser;
>+      EXPECT_TRUE(parser.Parse(packet, length));
>+      if (!nack_enabled_)
>+        EXPECT_EQ(0, parser.nack()->num_packets());
>+      if (parser.pli()->num_packets() > 0)
>+        received_pli_ = true;
>+      return SEND_PACKET;
>+    }
>+
>+    void OnFrame(const VideoFrame& video_frame) override {
>+      rtc::CritScope lock(&crit_);
>+      if (received_pli_ &&
>+          video_frame.timestamp() > highest_dropped_timestamp_) {
>+        observation_complete_.Set();
>+      }
>+      if (!received_pli_)
>+        frames_to_drop_ = kPacketsToDrop;
>+    }
>+
>+    void ModifyVideoConfigs(
>+        VideoSendStream::Config* send_config,
>+        std::vector<VideoReceiveStream::Config>* receive_configs,
>+        VideoEncoderConfig* encoder_config) override {
>+      send_config->rtp.nack.rtp_history_ms = rtp_history_ms_;
>+      (*receive_configs)[0].rtp.nack.rtp_history_ms = rtp_history_ms_;
>+      (*receive_configs)[0].renderer = this;
>+    }
>+
>+    void PerformTest() override {
>+      EXPECT_TRUE(Wait()) << "Timed out waiting for PLI to be "
>+                             "received and a frame to be "
>+                             "rendered afterwards.";
>+    }
>+
>+    rtc::CriticalSection crit_;
>+    int rtp_history_ms_;
>+    bool nack_enabled_;
>+    uint32_t highest_dropped_timestamp_ RTC_GUARDED_BY(&crit_);
>+    int frames_to_drop_ RTC_GUARDED_BY(&crit_);
>+    bool received_pli_ RTC_GUARDED_BY(&crit_);
>+  } test(rtp_history_ms);
>+
>+  RunBaseTest(&test);
>+}
>+
>+TEST_P(RetransmissionEndToEndTest, ReceivesPliAndRecoversWithNack) {
>+  ReceivesPliAndRecovers(1000);
>+}
>+
>+TEST_P(RetransmissionEndToEndTest, ReceivesPliAndRecoversWithoutNack) {
>+  ReceivesPliAndRecovers(0);
>+}
>+// This test drops second RTP packet with a marker bit set, makes sure it's
>+// retransmitted and renders. Retransmission SSRCs are also checked.
>+void RetransmissionEndToEndTest::DecodesRetransmittedFrame(bool enable_rtx,
>+                                                           bool enable_red) {
>+  static const int kDroppedFrameNumber = 10;
>+  class RetransmissionObserver : public test::EndToEndTest,
>+                                 public rtc::VideoSinkInterface<VideoFrame> {
>+   public:
>+    RetransmissionObserver(bool enable_rtx, bool enable_red)
>+        : EndToEndTest(kDefaultTimeoutMs),
>+          payload_type_(GetPayloadType(false, enable_red)),
>+          retransmission_ssrc_(enable_rtx ? kSendRtxSsrcs[0]
>+                                          : kVideoSendSsrcs[0]),
>+          retransmission_payload_type_(GetPayloadType(enable_rtx, enable_red)),
>+          encoder_factory_([]() { return VP8Encoder::Create(); }),
>+          marker_bits_observed_(0),
>+          retransmitted_timestamp_(0) {}
>+
>+   private:
>+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
>+      rtc::CritScope lock(&crit_);
>+      RTPHeader header;
>+      EXPECT_TRUE(parser_->Parse(packet, length, &header));
>+
>+      // Ignore padding-only packets over RTX.
>+      if (header.payloadType != payload_type_) {
>+        EXPECT_EQ(retransmission_ssrc_, header.ssrc);
>+        if (length == header.headerLength + header.paddingLength)
>+          return SEND_PACKET;
>+      }
>+
>+      if (header.timestamp == retransmitted_timestamp_) {
>+        EXPECT_EQ(retransmission_ssrc_, header.ssrc);
>+        EXPECT_EQ(retransmission_payload_type_, header.payloadType);
>+        return SEND_PACKET;
>+      }
>+
>+      // Found the final packet of the frame to inflict loss to, drop this and
>+      // expect a retransmission.
>+      if (header.payloadType == payload_type_ && header.markerBit &&
>+          ++marker_bits_observed_ == kDroppedFrameNumber) {
>+        // This should be the only dropped packet.
>+        EXPECT_EQ(0u, retransmitted_timestamp_);
>+        retransmitted_timestamp_ = header.timestamp;
>+        if (std::find(rendered_timestamps_.begin(), rendered_timestamps_.end(),
>+                      retransmitted_timestamp_) != rendered_timestamps_.end()) {
>+          // Frame was rendered before last packet was scheduled for sending.
>+          // This is extremly rare but possible scenario because prober able to
>+          // resend packet before it was send.
>+          // TODO(danilchap): Remove this corner case when prober would not be
>+          // able to sneak in between packet saved to history for resending and
>+          // pacer notified about existance of that packet for sending.
>+          // See https://bugs.chromium.org/p/webrtc/issues/detail?id=5540 for
>+          // details.
>+          observation_complete_.Set();
>+        }
>+        return DROP_PACKET;
>+      }
>+
>+      return SEND_PACKET;
>+    }
>+
>+    void OnFrame(const VideoFrame& frame) override {
>+      EXPECT_EQ(kVideoRotation_90, frame.rotation());
>+      {
>+        rtc::CritScope lock(&crit_);
>+        if (frame.timestamp() == retransmitted_timestamp_)
>+          observation_complete_.Set();
>+        rendered_timestamps_.push_back(frame.timestamp());
>+      }
>+      orig_renderer_->OnFrame(frame);
>+    }
>+
>+    void ModifyVideoConfigs(
>+        VideoSendStream::Config* send_config,
>+        std::vector<VideoReceiveStream::Config>* receive_configs,
>+        VideoEncoderConfig* encoder_config) override {
>+      send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
>+
>+      // Insert ourselves into the rendering pipeline.
>+      RTC_DCHECK(!orig_renderer_);
>+      orig_renderer_ = (*receive_configs)[0].renderer;
>+      RTC_DCHECK(orig_renderer_);
>+      (*receive_configs)[0].disable_prerenderer_smoothing = true;
>+      (*receive_configs)[0].renderer = this;
>+
>+      (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
>+
>+      if (payload_type_ == kRedPayloadType) {
>+        send_config->rtp.ulpfec.ulpfec_payload_type = kUlpfecPayloadType;
>+        send_config->rtp.ulpfec.red_payload_type = kRedPayloadType;
>+        if (retransmission_ssrc_ == kSendRtxSsrcs[0])
>+          send_config->rtp.ulpfec.red_rtx_payload_type = kRtxRedPayloadType;
>+        (*receive_configs)[0].rtp.ulpfec_payload_type =
>+            send_config->rtp.ulpfec.ulpfec_payload_type;
>+        (*receive_configs)[0].rtp.red_payload_type =
>+            send_config->rtp.ulpfec.red_payload_type;
>+      }
>+
>+      if (retransmission_ssrc_ == kSendRtxSsrcs[0]) {
>+        send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[0]);
>+        send_config->rtp.rtx.payload_type = kSendRtxPayloadType;
>+        (*receive_configs)[0].rtp.rtx_ssrc = kSendRtxSsrcs[0];
>+        (*receive_configs)[0]
>+            .rtp.rtx_associated_payload_types[(payload_type_ == kRedPayloadType)
>+                                                  ? kRtxRedPayloadType
>+                                                  : kSendRtxPayloadType] =
>+            payload_type_;
>+      }
>+      // Configure encoding and decoding with VP8, since generic packetization
>+      // doesn't support FEC with NACK.
>+      RTC_DCHECK_EQ(1, (*receive_configs)[0].decoders.size());
>+      send_config->encoder_settings.encoder_factory = &encoder_factory_;
>+      send_config->rtp.payload_name = "VP8";
>+      encoder_config->codec_type = kVideoCodecVP8;
>+      (*receive_configs)[0].decoders[0].payload_name = "VP8";
>+    }
>+
>+    void OnFrameGeneratorCapturerCreated(
>+        test::FrameGeneratorCapturer* frame_generator_capturer) override {
>+      frame_generator_capturer->SetFakeRotation(kVideoRotation_90);
>+    }
>+
>+    void PerformTest() override {
>+      EXPECT_TRUE(Wait())
>+          << "Timed out while waiting for retransmission to render.";
>+    }
>+
>+    int GetPayloadType(bool use_rtx, bool use_fec) {
>+      if (use_fec) {
>+        if (use_rtx)
>+          return kRtxRedPayloadType;
>+        return kRedPayloadType;
>+      }
>+      if (use_rtx)
>+        return kSendRtxPayloadType;
>+      return kFakeVideoSendPayloadType;
>+    }
>+
>+    rtc::CriticalSection crit_;
>+    rtc::VideoSinkInterface<VideoFrame>* orig_renderer_ = nullptr;
>+    const int payload_type_;
>+    const uint32_t retransmission_ssrc_;
>+    const int retransmission_payload_type_;
>+    test::FunctionVideoEncoderFactory encoder_factory_;
>+    const std::string payload_name_;
>+    int marker_bits_observed_;
>+    uint32_t retransmitted_timestamp_ RTC_GUARDED_BY(&crit_);
>+    std::vector<uint32_t> rendered_timestamps_ RTC_GUARDED_BY(&crit_);
>+  } test(enable_rtx, enable_red);
>+
>+  RunBaseTest(&test);
>+}
>+
>+TEST_P(RetransmissionEndToEndTest, DecodesRetransmittedFrame) {
>+  DecodesRetransmittedFrame(false, false);
>+}
>+
>+TEST_P(RetransmissionEndToEndTest, DecodesRetransmittedFrameOverRtx) {
>+  DecodesRetransmittedFrame(true, false);
>+}
>+
>+TEST_P(RetransmissionEndToEndTest, DecodesRetransmittedFrameByRed) {
>+  DecodesRetransmittedFrame(false, true);
>+}
>+
>+TEST_P(RetransmissionEndToEndTest, DecodesRetransmittedFrameByRedOverRtx) {
>+  DecodesRetransmittedFrame(true, true);
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/rtp_rtcp_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/rtp_rtcp_tests.cc
>new file mode 100644
>index 00000000000..c8eb802ce9d
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/rtp_rtcp_tests.cc
>@@ -0,0 +1,568 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/video_coding/codecs/vp8/include/vp8.h"
>+#include "test/call_test.h"
>+#include "test/gtest.h"
>+#include "test/rtcp_packet_parser.h"
>+
>+namespace webrtc {
>+
>+class RtpRtcpEndToEndTest : public test::CallTest {
>+ protected:
>+  void RespectsRtcpMode(RtcpMode rtcp_mode);
>+  void TestRtpStatePreservation(bool use_rtx, bool provoke_rtcpsr_before_rtp);
>+};
>+
>+void RtpRtcpEndToEndTest::RespectsRtcpMode(RtcpMode rtcp_mode) {
>+  static const int kNumCompoundRtcpPacketsToObserve = 10;
>+  class RtcpModeObserver : public test::EndToEndTest {
>+   public:
>+    explicit RtcpModeObserver(RtcpMode rtcp_mode)
>+        : EndToEndTest(kDefaultTimeoutMs),
>+          rtcp_mode_(rtcp_mode),
>+          sent_rtp_(0),
>+          sent_rtcp_(0) {}
>+
>+   private:
>+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
>+      rtc::CritScope lock(&crit_);
>+      if (++sent_rtp_ % 3 == 0)
>+        return DROP_PACKET;
>+
>+      return SEND_PACKET;
>+    }
>+
>+    Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
>+      rtc::CritScope lock(&crit_);
>+      ++sent_rtcp_;
>+      test::RtcpPacketParser parser;
>+      EXPECT_TRUE(parser.Parse(packet, length));
>+
>+      EXPECT_EQ(0, parser.sender_report()->num_packets());
>+
>+      switch (rtcp_mode_) {
>+        case RtcpMode::kCompound:
>+          // TODO(holmer): We shouldn't send transport feedback alone if
>+          // compound RTCP is negotiated.
>+          if (parser.receiver_report()->num_packets() == 0 &&
>+              parser.transport_feedback()->num_packets() == 0) {
>+            ADD_FAILURE() << "Received RTCP packet without receiver report for "
>+                             "RtcpMode::kCompound.";
>+            observation_complete_.Set();
>+          }
>+
>+          if (sent_rtcp_ >= kNumCompoundRtcpPacketsToObserve)
>+            observation_complete_.Set();
>+
>+          break;
>+        case RtcpMode::kReducedSize:
>+          if (parser.receiver_report()->num_packets() == 0)
>+            observation_complete_.Set();
>+          break;
>+        case RtcpMode::kOff:
>+          RTC_NOTREACHED();
>+          break;
>+      }
>+
>+      return SEND_PACKET;
>+    }
>+
>+    void ModifyVideoConfigs(
>+        VideoSendStream::Config* send_config,
>+        std::vector<VideoReceiveStream::Config>* receive_configs,
>+        VideoEncoderConfig* encoder_config) override {
>+      send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
>+      (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
>+      (*receive_configs)[0].rtp.rtcp_mode = rtcp_mode_;
>+    }
>+
>+    void PerformTest() override {
>+      EXPECT_TRUE(Wait())
>+          << (rtcp_mode_ == RtcpMode::kCompound
>+                  ? "Timed out before observing enough compound packets."
>+                  : "Timed out before receiving a non-compound RTCP packet.");
>+    }
>+
>+    RtcpMode rtcp_mode_;
>+    rtc::CriticalSection crit_;
>+    // Must be protected since RTCP can be sent by both the process thread
>+    // and the pacer thread.
>+    int sent_rtp_ RTC_GUARDED_BY(&crit_);
>+    int sent_rtcp_ RTC_GUARDED_BY(&crit_);
>+  } test(rtcp_mode);
>+
>+  RunBaseTest(&test);
>+}
>+
>+TEST_F(RtpRtcpEndToEndTest, UsesRtcpCompoundMode) {
>+  RespectsRtcpMode(RtcpMode::kCompound);
>+}
>+
>+TEST_F(RtpRtcpEndToEndTest, UsesRtcpReducedSizeMode) {
>+  RespectsRtcpMode(RtcpMode::kReducedSize);
>+}
>+
>+void RtpRtcpEndToEndTest::TestRtpStatePreservation(
>+    bool use_rtx,
>+    bool provoke_rtcpsr_before_rtp) {
>+  // This test uses other VideoStream settings than the the default settings
>+  // implemented in DefaultVideoStreamFactory. Therefore this test implements
>+  // its own VideoEncoderConfig::VideoStreamFactoryInterface which is created
>+  // in ModifyVideoConfigs.
>+  class VideoStreamFactory
>+      : public VideoEncoderConfig::VideoStreamFactoryInterface {
>+   public:
>+    VideoStreamFactory() {}
>+
>+   private:
>+    std::vector<VideoStream> CreateEncoderStreams(
>+        int width,
>+        int height,
>+        const VideoEncoderConfig& encoder_config) override {
>+      std::vector<VideoStream> streams =
>+          test::CreateVideoStreams(width, height, encoder_config);
>+
>+      if (encoder_config.number_of_streams > 1) {
>+        // Lower bitrates so that all streams send initially.
>+        RTC_DCHECK_EQ(3, encoder_config.number_of_streams);
>+        for (size_t i = 0; i < encoder_config.number_of_streams; ++i) {
>+          streams[i].min_bitrate_bps = 10000;
>+          streams[i].target_bitrate_bps = 15000;
>+          streams[i].max_bitrate_bps = 20000;
>+        }
>+      } else {
>+        // Use the same total bitrates when sending a single stream to avoid
>+        // lowering
>+        // the bitrate estimate and requiring a subsequent rampup.
>+        streams[0].min_bitrate_bps = 3 * 10000;
>+        streams[0].target_bitrate_bps = 3 * 15000;
>+        streams[0].max_bitrate_bps = 3 * 20000;
>+      }
>+      return streams;
>+    }
>+  };
>+
>+  class RtpSequenceObserver : public test::RtpRtcpObserver {
>+   public:
>+    explicit RtpSequenceObserver(bool use_rtx)
>+        : test::RtpRtcpObserver(kDefaultTimeoutMs),
>+          ssrcs_to_observe_(kNumSimulcastStreams) {
>+      for (size_t i = 0; i < kNumSimulcastStreams; ++i) {
>+        ssrc_is_rtx_[kVideoSendSsrcs[i]] = false;
>+        if (use_rtx)
>+          ssrc_is_rtx_[kSendRtxSsrcs[i]] = true;
>+      }
>+    }
>+
>+    void ResetExpectedSsrcs(size_t num_expected_ssrcs) {
>+      rtc::CritScope lock(&crit_);
>+      ssrc_observed_.clear();
>+      ssrcs_to_observe_ = num_expected_ssrcs;
>+    }
>+
>+   private:
>+    void ValidateTimestampGap(uint32_t ssrc,
>+                              uint32_t timestamp,
>+                              bool only_padding)
>+        RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_) {
>+      static const int32_t kMaxTimestampGap = kDefaultTimeoutMs * 90;
>+      auto timestamp_it = last_observed_timestamp_.find(ssrc);
>+      if (timestamp_it == last_observed_timestamp_.end()) {
>+        EXPECT_FALSE(only_padding);
>+        last_observed_timestamp_[ssrc] = timestamp;
>+      } else {
>+        // Verify timestamps are reasonably close.
>+        uint32_t latest_observed = timestamp_it->second;
>+        // Wraparound handling is unnecessary here as long as an int variable
>+        // is used to store the result.
>+        int32_t timestamp_gap = timestamp - latest_observed;
>+        EXPECT_LE(std::abs(timestamp_gap), kMaxTimestampGap)
>+            << "Gap in timestamps (" << latest_observed << " -> " << timestamp
>+            << ") too large for SSRC: " << ssrc << ".";
>+        timestamp_it->second = timestamp;
>+      }
>+    }
>+
>+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
>+      RTPHeader header;
>+      EXPECT_TRUE(parser_->Parse(packet, length, &header));
>+      const uint32_t ssrc = header.ssrc;
>+      const int64_t sequence_number =
>+          seq_numbers_unwrapper_.Unwrap(header.sequenceNumber);
>+      const uint32_t timestamp = header.timestamp;
>+      const bool only_padding =
>+          header.headerLength + header.paddingLength == length;
>+
>+      EXPECT_TRUE(ssrc_is_rtx_.find(ssrc) != ssrc_is_rtx_.end())
>+          << "Received SSRC that wasn't configured: " << ssrc;
>+
>+      static const int64_t kMaxSequenceNumberGap = 100;
>+      std::list<int64_t>* seq_numbers = &last_observed_seq_numbers_[ssrc];
>+      if (seq_numbers->empty()) {
>+        seq_numbers->push_back(sequence_number);
>+      } else {
>+        // We shouldn't get replays of previous sequence numbers.
>+        for (int64_t observed : *seq_numbers) {
>+          EXPECT_NE(observed, sequence_number)
>+              << "Received sequence number " << sequence_number << " for SSRC "
>+              << ssrc << " 2nd time.";
>+        }
>+        // Verify sequence numbers are reasonably close.
>+        int64_t latest_observed = seq_numbers->back();
>+        int64_t sequence_number_gap = sequence_number - latest_observed;
>+        EXPECT_LE(std::abs(sequence_number_gap), kMaxSequenceNumberGap)
>+            << "Gap in sequence numbers (" << latest_observed << " -> "
>+            << sequence_number << ") too large for SSRC: " << ssrc << ".";
>+        seq_numbers->push_back(sequence_number);
>+        if (seq_numbers->size() >= kMaxSequenceNumberGap) {
>+          seq_numbers->pop_front();
>+        }
>+      }
>+
>+      if (!ssrc_is_rtx_[ssrc]) {
>+        rtc::CritScope lock(&crit_);
>+        ValidateTimestampGap(ssrc, timestamp, only_padding);
>+
>+        // Wait for media packets on all ssrcs.
>+        if (!ssrc_observed_[ssrc] && !only_padding) {
>+          ssrc_observed_[ssrc] = true;
>+          if (--ssrcs_to_observe_ == 0)
>+            observation_complete_.Set();
>+        }
>+      }
>+
>+      return SEND_PACKET;
>+    }
>+
>+    Action OnSendRtcp(const uint8_t* packet, size_t length) override {
>+      test::RtcpPacketParser rtcp_parser;
>+      rtcp_parser.Parse(packet, length);
>+      if (rtcp_parser.sender_report()->num_packets() > 0) {
>+        uint32_t ssrc = rtcp_parser.sender_report()->sender_ssrc();
>+        uint32_t rtcp_timestamp = rtcp_parser.sender_report()->rtp_timestamp();
>+
>+        rtc::CritScope lock(&crit_);
>+        ValidateTimestampGap(ssrc, rtcp_timestamp, false);
>+      }
>+      return SEND_PACKET;
>+    }
>+
>+    SequenceNumberUnwrapper seq_numbers_unwrapper_;
>+    std::map<uint32_t, std::list<int64_t>> last_observed_seq_numbers_;
>+    std::map<uint32_t, uint32_t> last_observed_timestamp_;
>+    std::map<uint32_t, bool> ssrc_is_rtx_;
>+
>+    rtc::CriticalSection crit_;
>+    size_t ssrcs_to_observe_ RTC_GUARDED_BY(crit_);
>+    std::map<uint32_t, bool> ssrc_observed_ RTC_GUARDED_BY(crit_);
>+  } observer(use_rtx);
>+
>+  std::unique_ptr<test::PacketTransport> send_transport;
>+  std::unique_ptr<test::PacketTransport> receive_transport;
>+
>+  VideoEncoderConfig one_stream;
>+
>+  task_queue_.SendTask([this, &observer, &send_transport, &receive_transport,
>+                        &one_stream, use_rtx]() {
>+    CreateCalls();
>+
>+    send_transport = absl::make_unique<test::PacketTransport>(
>+        &task_queue_, sender_call_.get(), &observer,
>+        test::PacketTransport::kSender, payload_type_map_,
>+        FakeNetworkPipe::Config());
>+    receive_transport = absl::make_unique<test::PacketTransport>(
>+        &task_queue_, nullptr, &observer, test::PacketTransport::kReceiver,
>+        payload_type_map_, FakeNetworkPipe::Config());
>+    send_transport->SetReceiver(receiver_call_->Receiver());
>+    receive_transport->SetReceiver(sender_call_->Receiver());
>+
>+    CreateSendConfig(kNumSimulcastStreams, 0, 0, send_transport.get());
>+
>+    if (use_rtx) {
>+      for (size_t i = 0; i < kNumSimulcastStreams; ++i) {
>+        GetVideoSendConfig()->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[i]);
>+      }
>+      GetVideoSendConfig()->rtp.rtx.payload_type = kSendRtxPayloadType;
>+    }
>+
>+    GetVideoEncoderConfig()->video_stream_factory =
>+        new rtc::RefCountedObject<VideoStreamFactory>();
>+    // Use the same total bitrates when sending a single stream to avoid
>+    // lowering the bitrate estimate and requiring a subsequent rampup.
>+    one_stream = GetVideoEncoderConfig()->Copy();
>+    // one_stream.streams.resize(1);
>+    one_stream.number_of_streams = 1;
>+    CreateMatchingReceiveConfigs(receive_transport.get());
>+
>+    CreateVideoStreams();
>+    CreateFrameGeneratorCapturer(30, 1280, 720);
>+
>+    Start();
>+  });
>+
>+  EXPECT_TRUE(observer.Wait())
>+      << "Timed out waiting for all SSRCs to send packets.";
>+
>+  // Test stream resetting more than once to make sure that the state doesn't
>+  // get set once (this could be due to using std::map::insert for instance).
>+  for (size_t i = 0; i < 3; ++i) {
>+    task_queue_.SendTask([&]() {
>+      frame_generator_capturer_->Stop();
>+      DestroyVideoSendStreams();
>+
>+      // Re-create VideoSendStream with only one stream.
>+      CreateVideoSendStream(one_stream);
>+      GetVideoSendStream()->Start();
>+      if (provoke_rtcpsr_before_rtp) {
>+        // Rapid Resync Request forces sending RTCP Sender Report back.
>+        // Using this request speeds up this test because then there is no need
>+        // to wait for a second for periodic Sender Report.
>+        rtcp::RapidResyncRequest force_send_sr_back_request;
>+        rtc::Buffer packet = force_send_sr_back_request.Build();
>+        static_cast<webrtc::test::DirectTransport*>(receive_transport.get())
>+            ->SendRtcp(packet.data(), packet.size());
>+      }
>+      CreateFrameGeneratorCapturer(30, 1280, 720);
>+      frame_generator_capturer_->Start();
>+    });
>+
>+    observer.ResetExpectedSsrcs(1);
>+    EXPECT_TRUE(observer.Wait()) << "Timed out waiting for single RTP packet.";
>+
>+    // Reconfigure back to use all streams.
>+    task_queue_.SendTask([this]() {
>+      GetVideoSendStream()->ReconfigureVideoEncoder(
>+          GetVideoEncoderConfig()->Copy());
>+    });
>+    observer.ResetExpectedSsrcs(kNumSimulcastStreams);
>+    EXPECT_TRUE(observer.Wait())
>+        << "Timed out waiting for all SSRCs to send packets.";
>+
>+    // Reconfigure down to one stream.
>+    task_queue_.SendTask([this, &one_stream]() {
>+      GetVideoSendStream()->ReconfigureVideoEncoder(one_stream.Copy());
>+    });
>+    observer.ResetExpectedSsrcs(1);
>+    EXPECT_TRUE(observer.Wait()) << "Timed out waiting for single RTP packet.";
>+
>+    // Reconfigure back to use all streams.
>+    task_queue_.SendTask([this]() {
>+      GetVideoSendStream()->ReconfigureVideoEncoder(
>+          GetVideoEncoderConfig()->Copy());
>+    });
>+    observer.ResetExpectedSsrcs(kNumSimulcastStreams);
>+    EXPECT_TRUE(observer.Wait())
>+        << "Timed out waiting for all SSRCs to send packets.";
>+  }
>+
>+  task_queue_.SendTask([this, &send_transport, &receive_transport]() {
>+    Stop();
>+    DestroyStreams();
>+    send_transport.reset();
>+    receive_transport.reset();
>+    DestroyCalls();
>+  });
>+}
>+
>+TEST_F(RtpRtcpEndToEndTest, RestartingSendStreamPreservesRtpState) {
>+  TestRtpStatePreservation(false, false);
>+}
>+
>+TEST_F(RtpRtcpEndToEndTest, RestartingSendStreamPreservesRtpStatesWithRtx) {
>+  TestRtpStatePreservation(true, false);
>+}
>+
>+TEST_F(RtpRtcpEndToEndTest,
>+       RestartingSendStreamKeepsRtpAndRtcpTimestampsSynced) {
>+  TestRtpStatePreservation(true, true);
>+}
>+
>+TEST_F(RtpRtcpEndToEndTest, TestFlexfecRtpStatePreservation) {
>+  class RtpSequenceObserver : public test::RtpRtcpObserver {
>+   public:
>+    RtpSequenceObserver()
>+        : test::RtpRtcpObserver(kDefaultTimeoutMs),
>+          num_flexfec_packets_sent_(0) {}
>+
>+    void ResetPacketCount() {
>+      rtc::CritScope lock(&crit_);
>+      num_flexfec_packets_sent_ = 0;
>+    }
>+
>+   private:
>+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
>+      rtc::CritScope lock(&crit_);
>+
>+      RTPHeader header;
>+      EXPECT_TRUE(parser_->Parse(packet, length, &header));
>+      const uint16_t sequence_number = header.sequenceNumber;
>+      const uint32_t timestamp = header.timestamp;
>+      const uint32_t ssrc = header.ssrc;
>+
>+      if (ssrc == kVideoSendSsrcs[0] || ssrc == kSendRtxSsrcs[0]) {
>+        return SEND_PACKET;
>+      }
>+      EXPECT_EQ(kFlexfecSendSsrc, ssrc) << "Unknown SSRC sent.";
>+
>+      ++num_flexfec_packets_sent_;
>+
>+      // If this is the first packet, we have nothing to compare to.
>+      if (!last_observed_sequence_number_) {
>+        last_observed_sequence_number_.emplace(sequence_number);
>+        last_observed_timestamp_.emplace(timestamp);
>+
>+        return SEND_PACKET;
>+      }
>+
>+      // Verify continuity and monotonicity of RTP sequence numbers.
>+      EXPECT_EQ(static_cast<uint16_t>(*last_observed_sequence_number_ + 1),
>+                sequence_number);
>+      last_observed_sequence_number_.emplace(sequence_number);
>+
>+      // Timestamps should be non-decreasing...
>+      const bool timestamp_is_same_or_newer =
>+          timestamp == *last_observed_timestamp_ ||
>+          IsNewerTimestamp(timestamp, *last_observed_timestamp_);
>+      EXPECT_TRUE(timestamp_is_same_or_newer);
>+      // ...but reasonably close in time.
>+      const int k10SecondsInRtpTimestampBase = 10 * kVideoPayloadTypeFrequency;
>+      EXPECT_TRUE(IsNewerTimestamp(
>+          *last_observed_timestamp_ + k10SecondsInRtpTimestampBase, timestamp));
>+      last_observed_timestamp_.emplace(timestamp);
>+
>+      // Pass test when enough packets have been let through.
>+      if (num_flexfec_packets_sent_ >= 10) {
>+        observation_complete_.Set();
>+      }
>+
>+      return SEND_PACKET;
>+    }
>+
>+    absl::optional<uint16_t> last_observed_sequence_number_
>+        RTC_GUARDED_BY(crit_);
>+    absl::optional<uint32_t> last_observed_timestamp_ RTC_GUARDED_BY(crit_);
>+    size_t num_flexfec_packets_sent_ RTC_GUARDED_BY(crit_);
>+    rtc::CriticalSection crit_;
>+  } observer;
>+
>+  static constexpr int kFrameMaxWidth = 320;
>+  static constexpr int kFrameMaxHeight = 180;
>+  static constexpr int kFrameRate = 15;
>+
>+  std::unique_ptr<test::PacketTransport> send_transport;
>+  std::unique_ptr<test::PacketTransport> receive_transport;
>+  test::FunctionVideoEncoderFactory encoder_factory(
>+      []() { return VP8Encoder::Create(); });
>+
>+  task_queue_.SendTask([&]() {
>+    CreateCalls();
>+
>+    FakeNetworkPipe::Config lossy_delayed_link;
>+    lossy_delayed_link.loss_percent = 2;
>+    lossy_delayed_link.queue_delay_ms = 50;
>+
>+    send_transport = absl::make_unique<test::PacketTransport>(
>+        &task_queue_, sender_call_.get(), &observer,
>+        test::PacketTransport::kSender, payload_type_map_, lossy_delayed_link);
>+    send_transport->SetReceiver(receiver_call_->Receiver());
>+
>+    FakeNetworkPipe::Config flawless_link;
>+    receive_transport = absl::make_unique<test::PacketTransport>(
>+        &task_queue_, nullptr, &observer, test::PacketTransport::kReceiver,
>+        payload_type_map_, flawless_link);
>+    receive_transport->SetReceiver(sender_call_->Receiver());
>+
>+    // For reduced flakyness, we use a real VP8 encoder together with NACK
>+    // and RTX.
>+    const int kNumVideoStreams = 1;
>+    const int kNumFlexfecStreams = 1;
>+    CreateSendConfig(kNumVideoStreams, 0, kNumFlexfecStreams,
>+                     send_transport.get());
>+
>+    GetVideoSendConfig()->encoder_settings.encoder_factory = &encoder_factory;
>+    GetVideoSendConfig()->rtp.payload_name = "VP8";
>+    GetVideoSendConfig()->rtp.payload_type = kVideoSendPayloadType;
>+    GetVideoSendConfig()->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
>+    GetVideoSendConfig()->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[0]);
>+    GetVideoSendConfig()->rtp.rtx.payload_type = kSendRtxPayloadType;
>+    GetVideoEncoderConfig()->codec_type = kVideoCodecVP8;
>+
>+    CreateMatchingReceiveConfigs(receive_transport.get());
>+    video_receive_configs_[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
>+    video_receive_configs_[0].rtp.rtx_ssrc = kSendRtxSsrcs[0];
>+    video_receive_configs_[0]
>+        .rtp.rtx_associated_payload_types[kSendRtxPayloadType] =
>+        kVideoSendPayloadType;
>+
>+    // The matching FlexFEC receive config is not created by
>+    // CreateMatchingReceiveConfigs since this is not a test::BaseTest.
>+    // Set up the receive config manually instead.
>+    FlexfecReceiveStream::Config flexfec_receive_config(
>+        receive_transport.get());
>+    flexfec_receive_config.payload_type =
>+        GetVideoSendConfig()->rtp.flexfec.payload_type;
>+    flexfec_receive_config.remote_ssrc = GetVideoSendConfig()->rtp.flexfec.ssrc;
>+    flexfec_receive_config.protected_media_ssrcs =
>+        GetVideoSendConfig()->rtp.flexfec.protected_media_ssrcs;
>+    flexfec_receive_config.local_ssrc = kReceiverLocalVideoSsrc;
>+    flexfec_receive_config.transport_cc = true;
>+    flexfec_receive_config.rtp_header_extensions.emplace_back(
>+        RtpExtension::kTransportSequenceNumberUri,
>+        test::kTransportSequenceNumberExtensionId);
>+    flexfec_receive_configs_.push_back(flexfec_receive_config);
>+
>+    CreateFlexfecStreams();
>+    CreateVideoStreams();
>+
>+    // RTCP might be disabled if the network is "down".
>+    sender_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
>+    receiver_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
>+
>+    CreateFrameGeneratorCapturer(kFrameRate, kFrameMaxWidth, kFrameMaxHeight);
>+
>+    Start();
>+  });
>+
>+  // Initial test.
>+  EXPECT_TRUE(observer.Wait()) << "Timed out waiting for packets.";
>+
>+  task_queue_.SendTask([this, &observer]() {
>+    // Ensure monotonicity when the VideoSendStream is restarted.
>+    Stop();
>+    observer.ResetPacketCount();
>+    Start();
>+  });
>+
>+  EXPECT_TRUE(observer.Wait()) << "Timed out waiting for packets.";
>+
>+  task_queue_.SendTask([this, &observer]() {
>+    // Ensure monotonicity when the VideoSendStream is recreated.
>+    frame_generator_capturer_->Stop();
>+    DestroyVideoSendStreams();
>+    observer.ResetPacketCount();
>+    CreateVideoSendStreams();
>+    GetVideoSendStream()->Start();
>+    CreateFrameGeneratorCapturer(kFrameRate, kFrameMaxWidth, kFrameMaxHeight);
>+    frame_generator_capturer_->Start();
>+  });
>+
>+  EXPECT_TRUE(observer.Wait()) << "Timed out waiting for packets.";
>+
>+  // Cleanup.
>+  task_queue_.SendTask([this, &send_transport, &receive_transport]() {
>+    Stop();
>+    DestroyStreams();
>+    send_transport.reset();
>+    receive_transport.reset();
>+    DestroyCalls();
>+  });
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/ssrc_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/ssrc_tests.cc
>new file mode 100644
>index 00000000000..69aafec5c81
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/ssrc_tests.cc
>@@ -0,0 +1,336 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "test/call_test.h"
>+#include "test/gtest.h"
>+#include "test/rtcp_packet_parser.h"
>+
>+namespace webrtc {
>+class SsrcEndToEndTest : public test::CallTest {
>+ protected:
>+  void TestSendsSetSsrcs(size_t num_ssrcs, bool send_single_ssrc_first);
>+};
>+
>+TEST_F(SsrcEndToEndTest, ReceiverUsesLocalSsrc) {
>+  class SyncRtcpObserver : public test::EndToEndTest {
>+   public:
>+    SyncRtcpObserver() : EndToEndTest(kDefaultTimeoutMs) {}
>+
>+    Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
>+      test::RtcpPacketParser parser;
>+      EXPECT_TRUE(parser.Parse(packet, length));
>+      EXPECT_EQ(kReceiverLocalVideoSsrc, parser.sender_ssrc());
>+      observation_complete_.Set();
>+
>+      return SEND_PACKET;
>+    }
>+
>+    void PerformTest() override {
>+      EXPECT_TRUE(Wait())
>+          << "Timed out while waiting for a receiver RTCP packet to be sent.";
>+    }
>+  } test;
>+
>+  RunBaseTest(&test);
>+}
>+
>+TEST_F(SsrcEndToEndTest, UnknownRtpPacketGivesUnknownSsrcReturnCode) {
>+  class PacketInputObserver : public PacketReceiver {
>+   public:
>+    explicit PacketInputObserver(PacketReceiver* receiver)
>+        : receiver_(receiver), delivered_packet_(false, false) {}
>+
>+    bool Wait() { return delivered_packet_.Wait(kDefaultTimeoutMs); }
>+
>+   private:
>+    DeliveryStatus DeliverPacket(MediaType media_type,
>+                                 rtc::CopyOnWriteBuffer packet,
>+                                 int64_t packet_time_us) override {
>+      if (RtpHeaderParser::IsRtcp(packet.cdata(), packet.size())) {
>+        return receiver_->DeliverPacket(media_type, std::move(packet),
>+                                        packet_time_us);
>+      } else {
>+        DeliveryStatus delivery_status = receiver_->DeliverPacket(
>+            media_type, std::move(packet), packet_time_us);
>+        EXPECT_EQ(DELIVERY_UNKNOWN_SSRC, delivery_status);
>+        delivered_packet_.Set();
>+        return delivery_status;
>+      }
>+    }
>+
>+    PacketReceiver* receiver_;
>+    rtc::Event delivered_packet_;
>+  };
>+
>+  std::unique_ptr<test::DirectTransport> send_transport;
>+  std::unique_ptr<test::DirectTransport> receive_transport;
>+  std::unique_ptr<PacketInputObserver> input_observer;
>+
>+  task_queue_.SendTask([this, &send_transport, &receive_transport,
>+                        &input_observer]() {
>+    CreateCalls();
>+
>+    send_transport = absl::make_unique<test::DirectTransport>(
>+        &task_queue_, sender_call_.get(), payload_type_map_);
>+    receive_transport = absl::make_unique<test::DirectTransport>(
>+        &task_queue_, receiver_call_.get(), payload_type_map_);
>+    input_observer =
>+        absl::make_unique<PacketInputObserver>(receiver_call_->Receiver());
>+    send_transport->SetReceiver(input_observer.get());
>+    receive_transport->SetReceiver(sender_call_->Receiver());
>+
>+    CreateSendConfig(1, 0, 0, send_transport.get());
>+    CreateMatchingReceiveConfigs(receive_transport.get());
>+
>+    CreateVideoStreams();
>+    CreateFrameGeneratorCapturer(kDefaultFramerate, kDefaultWidth,
>+                                 kDefaultHeight);
>+    Start();
>+
>+    receiver_call_->DestroyVideoReceiveStream(video_receive_streams_[0]);
>+    video_receive_streams_.clear();
>+  });
>+
>+  // Wait() waits for a received packet.
>+  EXPECT_TRUE(input_observer->Wait());
>+
>+  task_queue_.SendTask([this, &send_transport, &receive_transport]() {
>+    Stop();
>+    DestroyStreams();
>+    send_transport.reset();
>+    receive_transport.reset();
>+    DestroyCalls();
>+  });
>+}
>+
>+void SsrcEndToEndTest::TestSendsSetSsrcs(size_t num_ssrcs,
>+                                         bool send_single_ssrc_first) {
>+  class SendsSetSsrcs : public test::EndToEndTest {
>+   public:
>+    SendsSetSsrcs(const uint32_t* ssrcs,
>+                  size_t num_ssrcs,
>+                  bool send_single_ssrc_first)
>+        : EndToEndTest(kDefaultTimeoutMs),
>+          num_ssrcs_(num_ssrcs),
>+          send_single_ssrc_first_(send_single_ssrc_first),
>+          ssrcs_to_observe_(num_ssrcs),
>+          expect_single_ssrc_(send_single_ssrc_first),
>+          send_stream_(nullptr) {
>+      for (size_t i = 0; i < num_ssrcs; ++i)
>+        valid_ssrcs_[ssrcs[i]] = true;
>+    }
>+
>+   private:
>+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
>+      RTPHeader header;
>+      EXPECT_TRUE(parser_->Parse(packet, length, &header));
>+
>+      EXPECT_TRUE(valid_ssrcs_[header.ssrc])
>+          << "Received unknown SSRC: " << header.ssrc;
>+
>+      if (!valid_ssrcs_[header.ssrc])
>+        observation_complete_.Set();
>+
>+      if (!is_observed_[header.ssrc]) {
>+        is_observed_[header.ssrc] = true;
>+        --ssrcs_to_observe_;
>+        if (expect_single_ssrc_) {
>+          expect_single_ssrc_ = false;
>+          observation_complete_.Set();
>+        }
>+      }
>+
>+      if (ssrcs_to_observe_ == 0)
>+        observation_complete_.Set();
>+
>+      return SEND_PACKET;
>+    }
>+
>+    size_t GetNumVideoStreams() const override { return num_ssrcs_; }
>+
>+    // This test use other VideoStream settings than the the default settings
>+    // implemented in DefaultVideoStreamFactory. Therefore  this test implement
>+    // its own VideoEncoderConfig::VideoStreamFactoryInterface which is created
>+    // in ModifyVideoConfigs.
>+    class VideoStreamFactory
>+        : public VideoEncoderConfig::VideoStreamFactoryInterface {
>+     public:
>+      VideoStreamFactory() {}
>+
>+     private:
>+      std::vector<VideoStream> CreateEncoderStreams(
>+          int width,
>+          int height,
>+          const VideoEncoderConfig& encoder_config) override {
>+        std::vector<VideoStream> streams =
>+            test::CreateVideoStreams(width, height, encoder_config);
>+        // Set low simulcast bitrates to not have to wait for bandwidth ramp-up.
>+        for (size_t i = 0; i < encoder_config.number_of_streams; ++i) {
>+          streams[i].min_bitrate_bps = 10000;
>+          streams[i].target_bitrate_bps = 15000;
>+          streams[i].max_bitrate_bps = 20000;
>+        }
>+        return streams;
>+      }
>+    };
>+
>+    void ModifyVideoConfigs(
>+        VideoSendStream::Config* send_config,
>+        std::vector<VideoReceiveStream::Config>* receive_configs,
>+        VideoEncoderConfig* encoder_config) override {
>+      encoder_config->video_stream_factory =
>+          new rtc::RefCountedObject<VideoStreamFactory>();
>+      video_encoder_config_all_streams_ = encoder_config->Copy();
>+      if (send_single_ssrc_first_)
>+        encoder_config->number_of_streams = 1;
>+    }
>+
>+    void OnVideoStreamsCreated(
>+        VideoSendStream* send_stream,
>+        const std::vector<VideoReceiveStream*>& receive_streams) override {
>+      send_stream_ = send_stream;
>+    }
>+
>+    void PerformTest() override {
>+      EXPECT_TRUE(Wait()) << "Timed out while waiting for "
>+                          << (send_single_ssrc_first_ ? "first SSRC."
>+                                                      : "SSRCs.");
>+
>+      if (send_single_ssrc_first_) {
>+        // Set full simulcast and continue with the rest of the SSRCs.
>+        send_stream_->ReconfigureVideoEncoder(
>+            std::move(video_encoder_config_all_streams_));
>+        EXPECT_TRUE(Wait()) << "Timed out while waiting on additional SSRCs.";
>+      }
>+    }
>+
>+   private:
>+    std::map<uint32_t, bool> valid_ssrcs_;
>+    std::map<uint32_t, bool> is_observed_;
>+
>+    const size_t num_ssrcs_;
>+    const bool send_single_ssrc_first_;
>+
>+    size_t ssrcs_to_observe_;
>+    bool expect_single_ssrc_;
>+
>+    VideoSendStream* send_stream_;
>+    VideoEncoderConfig video_encoder_config_all_streams_;
>+  } test(kVideoSendSsrcs, num_ssrcs, send_single_ssrc_first);
>+
>+  RunBaseTest(&test);
>+}
>+
>+TEST_F(SsrcEndToEndTest, SendsSetSsrc) {
>+  TestSendsSetSsrcs(1, false);
>+}
>+
>+TEST_F(SsrcEndToEndTest, SendsSetSimulcastSsrcs) {
>+  TestSendsSetSsrcs(kNumSimulcastStreams, false);
>+}
>+
>+TEST_F(SsrcEndToEndTest, CanSwitchToUseAllSsrcs) {
>+  TestSendsSetSsrcs(kNumSimulcastStreams, true);
>+}
>+
>+TEST_F(SsrcEndToEndTest, DISABLED_RedundantPayloadsTransmittedOnAllSsrcs) {
>+  class ObserveRedundantPayloads : public test::EndToEndTest {
>+   public:
>+    ObserveRedundantPayloads()
>+        : EndToEndTest(kDefaultTimeoutMs),
>+          ssrcs_to_observe_(kNumSimulcastStreams) {
>+      for (size_t i = 0; i < kNumSimulcastStreams; ++i) {
>+        registered_rtx_ssrc_[kSendRtxSsrcs[i]] = true;
>+      }
>+    }
>+
>+   private:
>+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
>+      RTPHeader header;
>+      EXPECT_TRUE(parser_->Parse(packet, length, &header));
>+
>+      if (!registered_rtx_ssrc_[header.ssrc])
>+        return SEND_PACKET;
>+
>+      EXPECT_LE(header.headerLength + header.paddingLength, length);
>+      const bool packet_is_redundant_payload =
>+          header.headerLength + header.paddingLength < length;
>+
>+      if (!packet_is_redundant_payload)
>+        return SEND_PACKET;
>+
>+      if (!observed_redundant_retransmission_[header.ssrc]) {
>+        observed_redundant_retransmission_[header.ssrc] = true;
>+        if (--ssrcs_to_observe_ == 0)
>+          observation_complete_.Set();
>+      }
>+
>+      return SEND_PACKET;
>+    }
>+
>+    size_t GetNumVideoStreams() const override { return kNumSimulcastStreams; }
>+
>+    // This test use other VideoStream settings than the the default settings
>+    // implemented in DefaultVideoStreamFactory. Therefore  this test implement
>+    // its own VideoEncoderConfig::VideoStreamFactoryInterface which is created
>+    // in ModifyVideoConfigs.
>+    class VideoStreamFactory
>+        : public VideoEncoderConfig::VideoStreamFactoryInterface {
>+     public:
>+      VideoStreamFactory() {}
>+
>+     private:
>+      std::vector<VideoStream> CreateEncoderStreams(
>+          int width,
>+          int height,
>+          const VideoEncoderConfig& encoder_config) override {
>+        std::vector<VideoStream> streams =
>+            test::CreateVideoStreams(width, height, encoder_config);
>+        // Set low simulcast bitrates to not have to wait for bandwidth ramp-up.
>+        for (size_t i = 0; i < encoder_config.number_of_streams; ++i) {
>+          streams[i].min_bitrate_bps = 10000;
>+          streams[i].target_bitrate_bps = 15000;
>+          streams[i].max_bitrate_bps = 20000;
>+        }
>+        return streams;
>+      }
>+    };
>+
>+    void ModifyVideoConfigs(
>+        VideoSendStream::Config* send_config,
>+        std::vector<VideoReceiveStream::Config>* receive_configs,
>+        VideoEncoderConfig* encoder_config) override {
>+      // Set low simulcast bitrates to not have to wait for bandwidth ramp-up.
>+      encoder_config->video_stream_factory =
>+          new rtc::RefCountedObject<VideoStreamFactory>();
>+      send_config->rtp.rtx.payload_type = kSendRtxPayloadType;
>+
>+      for (size_t i = 0; i < kNumSimulcastStreams; ++i)
>+        send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[i]);
>+
>+      // Significantly higher than max bitrates for all video streams -> forcing
>+      // padding to trigger redundant padding on all RTX SSRCs.
>+      encoder_config->min_transmit_bitrate_bps = 100000;
>+    }
>+
>+    void PerformTest() override {
>+      EXPECT_TRUE(Wait())
>+          << "Timed out while waiting for redundant payloads on all SSRCs.";
>+    }
>+
>+   private:
>+    size_t ssrcs_to_observe_;
>+    std::map<uint32_t, bool> observed_redundant_retransmission_;
>+    std::map<uint32_t, bool> registered_rtx_ssrc_;
>+  } test;
>+
>+  RunBaseTest(&test);
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/stats_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/stats_tests.cc
>new file mode 100644
>index 00000000000..ae259111d89
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/stats_tests.cc
>@@ -0,0 +1,753 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "modules/rtp_rtcp/source/rtp_utility.h"
>+#include "modules/video_coding/include/video_coding_defines.h"
>+#include "system_wrappers/include/metrics.h"
>+#include "system_wrappers/include/metrics_default.h"
>+#include "system_wrappers/include/sleep.h"
>+#include "test/call_test.h"
>+#include "test/fake_encoder.h"
>+#include "test/function_video_encoder_factory.h"
>+#include "test/gtest.h"
>+#include "test/rtcp_packet_parser.h"
>+
>+namespace webrtc {
>+class StatsEndToEndTest : public test::CallTest {};
>+
>+TEST_F(StatsEndToEndTest, GetStats) {
>+  static const int kStartBitrateBps = 3000000;
>+  static const int kExpectedRenderDelayMs = 20;
>+
>+  class ReceiveStreamRenderer : public rtc::VideoSinkInterface<VideoFrame> {
>+   public:
>+    ReceiveStreamRenderer() {}
>+
>+   private:
>+    void OnFrame(const VideoFrame& video_frame) override {}
>+  };
>+
>+  class StatsObserver : public test::EndToEndTest,
>+                        public rtc::VideoSinkInterface<VideoFrame> {
>+   public:
>+    StatsObserver()
>+        : EndToEndTest(kLongTimeoutMs),
>+          encoder_factory_([]() {
>+            return absl::make_unique<test::DelayedEncoder>(
>+                Clock::GetRealTimeClock(), 10);
>+          }),
>+          send_stream_(nullptr),
>+          expected_send_ssrcs_(),
>+          check_stats_event_(false, false) {}
>+
>+   private:
>+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
>+      // Drop every 25th packet => 4% loss.
>+      static const int kPacketLossFrac = 25;
>+      RTPHeader header;
>+      RtpUtility::RtpHeaderParser parser(packet, length);
>+      if (parser.Parse(&header) &&
>+          expected_send_ssrcs_.find(header.ssrc) !=
>+              expected_send_ssrcs_.end() &&
>+          header.sequenceNumber % kPacketLossFrac == 0) {
>+        return DROP_PACKET;
>+      }
>+      check_stats_event_.Set();
>+      return SEND_PACKET;
>+    }
>+
>+    Action OnSendRtcp(const uint8_t* packet, size_t length) override {
>+      check_stats_event_.Set();
>+      return SEND_PACKET;
>+    }
>+
>+    Action OnReceiveRtp(const uint8_t* packet, size_t length) override {
>+      check_stats_event_.Set();
>+      return SEND_PACKET;
>+    }
>+
>+    Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
>+      check_stats_event_.Set();
>+      return SEND_PACKET;
>+    }
>+
>+    void OnFrame(const VideoFrame& video_frame) override {
>+      // Ensure that we have at least 5ms send side delay.
>+      SleepMs(5);
>+    }
>+
>+    bool CheckReceiveStats() {
>+      for (size_t i = 0; i < receive_streams_.size(); ++i) {
>+        VideoReceiveStream::Stats stats = receive_streams_[i]->GetStats();
>+        EXPECT_EQ(expected_receive_ssrcs_[i], stats.ssrc);
>+
>+        // Make sure all fields have been populated.
>+        // TODO(pbos): Use CompoundKey if/when we ever know that all stats are
>+        // always filled for all receivers.
>+        receive_stats_filled_["IncomingRate"] |=
>+            stats.network_frame_rate != 0 || stats.total_bitrate_bps != 0;
>+
>+        send_stats_filled_["DecoderImplementationName"] |=
>+            stats.decoder_implementation_name ==
>+            test::FakeDecoder::kImplementationName;
>+        receive_stats_filled_["RenderDelayAsHighAsExpected"] |=
>+            stats.render_delay_ms >= kExpectedRenderDelayMs;
>+
>+        receive_stats_filled_["FrameCallback"] |= stats.decode_frame_rate != 0;
>+
>+        receive_stats_filled_["FrameRendered"] |= stats.render_frame_rate != 0;
>+
>+        receive_stats_filled_["StatisticsUpdated"] |=
>+            stats.rtcp_stats.packets_lost != 0 ||
>+            stats.rtcp_stats.extended_highest_sequence_number != 0 ||
>+            stats.rtcp_stats.fraction_lost != 0 || stats.rtcp_stats.jitter != 0;
>+
>+        receive_stats_filled_["DataCountersUpdated"] |=
>+            stats.rtp_stats.transmitted.payload_bytes != 0 ||
>+            stats.rtp_stats.fec.packets != 0 ||
>+            stats.rtp_stats.transmitted.header_bytes != 0 ||
>+            stats.rtp_stats.transmitted.packets != 0 ||
>+            stats.rtp_stats.transmitted.padding_bytes != 0 ||
>+            stats.rtp_stats.retransmitted.packets != 0;
>+
>+        receive_stats_filled_["CodecStats"] |=
>+            stats.target_delay_ms != 0 || stats.discarded_packets != 0;
>+
>+        receive_stats_filled_["FrameCounts"] |=
>+            stats.frame_counts.key_frames != 0 ||
>+            stats.frame_counts.delta_frames != 0;
>+
>+        receive_stats_filled_["CName"] |= !stats.c_name.empty();
>+
>+        receive_stats_filled_["RtcpPacketTypeCount"] |=
>+            stats.rtcp_packet_type_counts.fir_packets != 0 ||
>+            stats.rtcp_packet_type_counts.nack_packets != 0 ||
>+            stats.rtcp_packet_type_counts.pli_packets != 0 ||
>+            stats.rtcp_packet_type_counts.nack_requests != 0 ||
>+            stats.rtcp_packet_type_counts.unique_nack_requests != 0;
>+
>+        assert(stats.current_payload_type == -1 ||
>+               stats.current_payload_type == kFakeVideoSendPayloadType);
>+        receive_stats_filled_["IncomingPayloadType"] |=
>+            stats.current_payload_type == kFakeVideoSendPayloadType;
>+      }
>+
>+      return AllStatsFilled(receive_stats_filled_);
>+    }
>+
>+    bool CheckSendStats() {
>+      RTC_DCHECK(send_stream_);
>+      VideoSendStream::Stats stats = send_stream_->GetStats();
>+
>+      size_t expected_num_streams =
>+          kNumSimulcastStreams + expected_send_ssrcs_.size();
>+      send_stats_filled_["NumStreams"] |=
>+          stats.substreams.size() == expected_num_streams;
>+
>+      send_stats_filled_["CpuOveruseMetrics"] |=
>+          stats.avg_encode_time_ms != 0 && stats.encode_usage_percent != 0;
>+
>+      send_stats_filled_["EncoderImplementationName"] |=
>+          stats.encoder_implementation_name ==
>+          test::FakeEncoder::kImplementationName;
>+
>+      for (std::map<uint32_t, VideoSendStream::StreamStats>::const_iterator it =
>+               stats.substreams.begin();
>+           it != stats.substreams.end(); ++it) {
>+        if (expected_send_ssrcs_.find(it->first) == expected_send_ssrcs_.end())
>+          continue;  // Probably RTX.
>+
>+        send_stats_filled_[CompoundKey("CapturedFrameRate", it->first)] |=
>+            stats.input_frame_rate != 0;
>+
>+        const VideoSendStream::StreamStats& stream_stats = it->second;
>+
>+        send_stats_filled_[CompoundKey("StatisticsUpdated", it->first)] |=
>+            stream_stats.rtcp_stats.packets_lost != 0 ||
>+            stream_stats.rtcp_stats.extended_highest_sequence_number != 0 ||
>+            stream_stats.rtcp_stats.fraction_lost != 0;
>+
>+        send_stats_filled_[CompoundKey("DataCountersUpdated", it->first)] |=
>+            stream_stats.rtp_stats.fec.packets != 0 ||
>+            stream_stats.rtp_stats.transmitted.padding_bytes != 0 ||
>+            stream_stats.rtp_stats.retransmitted.packets != 0 ||
>+            stream_stats.rtp_stats.transmitted.packets != 0;
>+
>+        send_stats_filled_[CompoundKey("BitrateStatisticsObserver.Total",
>+                                       it->first)] |=
>+            stream_stats.total_bitrate_bps != 0;
>+
>+        send_stats_filled_[CompoundKey("BitrateStatisticsObserver.Retransmit",
>+                                       it->first)] |=
>+            stream_stats.retransmit_bitrate_bps != 0;
>+
>+        send_stats_filled_[CompoundKey("FrameCountObserver", it->first)] |=
>+            stream_stats.frame_counts.delta_frames != 0 ||
>+            stream_stats.frame_counts.key_frames != 0;
>+
>+        send_stats_filled_[CompoundKey("OutgoingRate", it->first)] |=
>+            stats.encode_frame_rate != 0;
>+
>+        send_stats_filled_[CompoundKey("Delay", it->first)] |=
>+            stream_stats.avg_delay_ms != 0 || stream_stats.max_delay_ms != 0;
>+
>+        // TODO(pbos): Use CompoundKey when the test makes sure that all SSRCs
>+        // report dropped packets.
>+        send_stats_filled_["RtcpPacketTypeCount"] |=
>+            stream_stats.rtcp_packet_type_counts.fir_packets != 0 ||
>+            stream_stats.rtcp_packet_type_counts.nack_packets != 0 ||
>+            stream_stats.rtcp_packet_type_counts.pli_packets != 0 ||
>+            stream_stats.rtcp_packet_type_counts.nack_requests != 0 ||
>+            stream_stats.rtcp_packet_type_counts.unique_nack_requests != 0;
>+      }
>+
>+      return AllStatsFilled(send_stats_filled_);
>+    }
>+
>+    std::string CompoundKey(const char* name, uint32_t ssrc) {
>+      std::ostringstream oss;
>+      oss << name << "_" << ssrc;
>+      return oss.str();
>+    }
>+
>+    bool AllStatsFilled(const std::map<std::string, bool>& stats_map) {
>+      for (const auto& stat : stats_map) {
>+        if (!stat.second)
>+          return false;
>+      }
>+      return true;
>+    }
>+
>+    test::PacketTransport* CreateSendTransport(
>+        test::SingleThreadedTaskQueueForTesting* task_queue,
>+        Call* sender_call) override {
>+      FakeNetworkPipe::Config network_config;
>+      network_config.loss_percent = 5;
>+      return new test::PacketTransport(task_queue, sender_call, this,
>+                                       test::PacketTransport::kSender,
>+                                       payload_type_map_, network_config);
>+    }
>+    void ModifySenderCallConfig(Call::Config* config) override {
>+      config->bitrate_config.start_bitrate_bps = kStartBitrateBps;
>+    }
>+
>+    // This test use other VideoStream settings than the the default settings
>+    // implemented in DefaultVideoStreamFactory. Therefore  this test implement
>+    // its own VideoEncoderConfig::VideoStreamFactoryInterface which is created
>+    // in ModifyVideoConfigs.
>+    class VideoStreamFactory
>+        : public VideoEncoderConfig::VideoStreamFactoryInterface {
>+     public:
>+      VideoStreamFactory() {}
>+
>+     private:
>+      std::vector<VideoStream> CreateEncoderStreams(
>+          int width,
>+          int height,
>+          const VideoEncoderConfig& encoder_config) override {
>+        std::vector<VideoStream> streams =
>+            test::CreateVideoStreams(width, height, encoder_config);
>+        // Set low simulcast bitrates to not have to wait for bandwidth ramp-up.
>+        for (size_t i = 0; i < encoder_config.number_of_streams; ++i) {
>+          streams[i].min_bitrate_bps = 10000;
>+          streams[i].target_bitrate_bps = 15000;
>+          streams[i].max_bitrate_bps = 20000;
>+        }
>+        return streams;
>+      }
>+    };
>+
>+    void ModifyVideoConfigs(
>+        VideoSendStream::Config* send_config,
>+        std::vector<VideoReceiveStream::Config>* receive_configs,
>+        VideoEncoderConfig* encoder_config) override {
>+      encoder_config->video_stream_factory =
>+          new rtc::RefCountedObject<VideoStreamFactory>();
>+      send_config->pre_encode_callback = this;  // Used to inject delay.
>+      expected_cname_ = send_config->rtp.c_name = "SomeCName";
>+
>+      send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
>+      send_config->rtp.rtx.payload_type = kSendRtxPayloadType;
>+
>+      const std::vector<uint32_t>& ssrcs = send_config->rtp.ssrcs;
>+      for (size_t i = 0; i < ssrcs.size(); ++i) {
>+        expected_send_ssrcs_.insert(ssrcs[i]);
>+        expected_receive_ssrcs_.push_back(
>+            (*receive_configs)[i].rtp.remote_ssrc);
>+        (*receive_configs)[i].render_delay_ms = kExpectedRenderDelayMs;
>+        (*receive_configs)[i].renderer = &receive_stream_renderer_;
>+        (*receive_configs)[i].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
>+
>+        (*receive_configs)[i].rtp.rtx_ssrc = kSendRtxSsrcs[i];
>+        (*receive_configs)[i]
>+            .rtp.rtx_associated_payload_types[kSendRtxPayloadType] =
>+            kFakeVideoSendPayloadType;
>+      }
>+
>+      for (size_t i = 0; i < kNumSimulcastStreams; ++i)
>+        send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[i]);
>+
>+      // Use a delayed encoder to make sure we see CpuOveruseMetrics stats that
>+      // are non-zero.
>+      send_config->encoder_settings.encoder_factory = &encoder_factory_;
>+    }
>+
>+    size_t GetNumVideoStreams() const override { return kNumSimulcastStreams; }
>+
>+    void OnVideoStreamsCreated(
>+        VideoSendStream* send_stream,
>+        const std::vector<VideoReceiveStream*>& receive_streams) override {
>+      send_stream_ = send_stream;
>+      receive_streams_ = receive_streams;
>+    }
>+
>+    void PerformTest() override {
>+      Clock* clock = Clock::GetRealTimeClock();
>+      int64_t now = clock->TimeInMilliseconds();
>+      int64_t stop_time = now + test::CallTest::kLongTimeoutMs;
>+      bool receive_ok = false;
>+      bool send_ok = false;
>+
>+      while (now < stop_time) {
>+        if (!receive_ok)
>+          receive_ok = CheckReceiveStats();
>+        if (!send_ok)
>+          send_ok = CheckSendStats();
>+
>+        if (receive_ok && send_ok)
>+          return;
>+
>+        int64_t time_until_timout_ = stop_time - now;
>+        if (time_until_timout_ > 0)
>+          check_stats_event_.Wait(time_until_timout_);
>+        now = clock->TimeInMilliseconds();
>+      }
>+
>+      ADD_FAILURE() << "Timed out waiting for filled stats.";
>+      for (std::map<std::string, bool>::const_iterator it =
>+               receive_stats_filled_.begin();
>+           it != receive_stats_filled_.end(); ++it) {
>+        if (!it->second) {
>+          ADD_FAILURE() << "Missing receive stats: " << it->first;
>+        }
>+      }
>+
>+      for (std::map<std::string, bool>::const_iterator it =
>+               send_stats_filled_.begin();
>+           it != send_stats_filled_.end(); ++it) {
>+        if (!it->second) {
>+          ADD_FAILURE() << "Missing send stats: " << it->first;
>+        }
>+      }
>+    }
>+
>+    test::FunctionVideoEncoderFactory encoder_factory_;
>+    std::vector<VideoReceiveStream*> receive_streams_;
>+    std::map<std::string, bool> receive_stats_filled_;
>+
>+    VideoSendStream* send_stream_;
>+    std::map<std::string, bool> send_stats_filled_;
>+
>+    std::vector<uint32_t> expected_receive_ssrcs_;
>+    std::set<uint32_t> expected_send_ssrcs_;
>+    std::string expected_cname_;
>+
>+    rtc::Event check_stats_event_;
>+    ReceiveStreamRenderer receive_stream_renderer_;
>+  } test;
>+
>+  RunBaseTest(&test);
>+}
>+
>+TEST_F(StatsEndToEndTest, TimingFramesAreReported) {
>+  static const int kExtensionId = 5;
>+
>+  class StatsObserver : public test::EndToEndTest {
>+   public:
>+    StatsObserver() : EndToEndTest(kLongTimeoutMs) {}
>+
>+   private:
>+    void ModifyVideoConfigs(
>+        VideoSendStream::Config* send_config,
>+        std::vector<VideoReceiveStream::Config>* receive_configs,
>+        VideoEncoderConfig* encoder_config) override {
>+      send_config->rtp.extensions.clear();
>+      send_config->rtp.extensions.push_back(
>+          RtpExtension(RtpExtension::kVideoTimingUri, kExtensionId));
>+      for (size_t i = 0; i < receive_configs->size(); ++i) {
>+        (*receive_configs)[i].rtp.extensions.clear();
>+        (*receive_configs)[i].rtp.extensions.push_back(
>+            RtpExtension(RtpExtension::kVideoTimingUri, kExtensionId));
>+      }
>+    }
>+
>+    void OnVideoStreamsCreated(
>+        VideoSendStream* send_stream,
>+        const std::vector<VideoReceiveStream*>& receive_streams) override {
>+      receive_streams_ = receive_streams;
>+    }
>+
>+    void PerformTest() override {
>+      // No frames reported initially.
>+      for (size_t i = 0; i < receive_streams_.size(); ++i) {
>+        EXPECT_FALSE(receive_streams_[i]->GetStats().timing_frame_info);
>+      }
>+      // Wait for at least one timing frame to be sent with 100ms grace period.
>+      SleepMs(kDefaultTimingFramesDelayMs + 100);
>+      // Check that timing frames are reported for each stream.
>+      for (size_t i = 0; i < receive_streams_.size(); ++i) {
>+        EXPECT_TRUE(receive_streams_[i]->GetStats().timing_frame_info);
>+      }
>+    }
>+
>+    std::vector<VideoReceiveStream*> receive_streams_;
>+  } test;
>+
>+  RunBaseTest(&test);
>+}
>+
>+TEST_F(StatsEndToEndTest, TestReceivedRtpPacketStats) {
>+  static const size_t kNumRtpPacketsToSend = 5;
>+  class ReceivedRtpStatsObserver : public test::EndToEndTest {
>+   public:
>+    ReceivedRtpStatsObserver()
>+        : EndToEndTest(kDefaultTimeoutMs),
>+          receive_stream_(nullptr),
>+          sent_rtp_(0) {}
>+
>+   private:
>+    void OnVideoStreamsCreated(
>+        VideoSendStream* send_stream,
>+        const std::vector<VideoReceiveStream*>& receive_streams) override {
>+      receive_stream_ = receive_streams[0];
>+    }
>+
>+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
>+      if (sent_rtp_ >= kNumRtpPacketsToSend) {
>+        VideoReceiveStream::Stats stats = receive_stream_->GetStats();
>+        if (kNumRtpPacketsToSend == stats.rtp_stats.transmitted.packets) {
>+          observation_complete_.Set();
>+        }
>+        return DROP_PACKET;
>+      }
>+      ++sent_rtp_;
>+      return SEND_PACKET;
>+    }
>+
>+    void PerformTest() override {
>+      EXPECT_TRUE(Wait())
>+          << "Timed out while verifying number of received RTP packets.";
>+    }
>+
>+    VideoReceiveStream* receive_stream_;
>+    uint32_t sent_rtp_;
>+  } test;
>+
>+  RunBaseTest(&test);
>+}
>+
>+#if defined(WEBRTC_WIN)
>+// Disabled due to flakiness on Windows (bugs.webrtc.org/7483).
>+#define MAYBE_ContentTypeSwitches DISABLED_ContentTypeSwitches
>+#else
>+#define MAYBE_ContentTypeSwitches ContentTypeSwitches
>+#endif
>+TEST_F(StatsEndToEndTest, MAYBE_ContentTypeSwitches) {
>+  class StatsObserver : public test::BaseTest,
>+                        public rtc::VideoSinkInterface<VideoFrame> {
>+   public:
>+    StatsObserver() : BaseTest(kLongTimeoutMs), num_frames_received_(0) {}
>+
>+    bool ShouldCreateReceivers() const override { return true; }
>+
>+    void OnFrame(const VideoFrame& video_frame) override {
>+      // The RTT is needed to estimate |ntp_time_ms| which is used by
>+      // end-to-end delay stats. Therefore, start counting received frames once
>+      // |ntp_time_ms| is valid.
>+      if (video_frame.ntp_time_ms() > 0 &&
>+          Clock::GetRealTimeClock()->CurrentNtpInMilliseconds() >=
>+              video_frame.ntp_time_ms()) {
>+        rtc::CritScope lock(&crit_);
>+        ++num_frames_received_;
>+      }
>+    }
>+
>+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
>+      if (MinNumberOfFramesReceived())
>+        observation_complete_.Set();
>+      return SEND_PACKET;
>+    }
>+
>+    bool MinNumberOfFramesReceived() const {
>+      // Have some room for frames with wrong content type during switch.
>+      const int kMinRequiredHistogramSamples = 200 + 50;
>+      rtc::CritScope lock(&crit_);
>+      return num_frames_received_ > kMinRequiredHistogramSamples;
>+    }
>+
>+    // May be called several times.
>+    void PerformTest() override {
>+      EXPECT_TRUE(Wait()) << "Timed out waiting for enough packets.";
>+      // Reset frame counter so next PerformTest() call will do something.
>+      {
>+        rtc::CritScope lock(&crit_);
>+        num_frames_received_ = 0;
>+      }
>+    }
>+
>+    rtc::CriticalSection crit_;
>+    int num_frames_received_ RTC_GUARDED_BY(&crit_);
>+  } test;
>+
>+  metrics::Reset();
>+
>+  Call::Config send_config(send_event_log_.get());
>+  test.ModifySenderCallConfig(&send_config);
>+  Call::Config recv_config(recv_event_log_.get());
>+  test.ModifyReceiverCallConfig(&recv_config);
>+
>+  VideoEncoderConfig encoder_config_with_screenshare;
>+
>+  task_queue_.SendTask([this, &test, &send_config, &recv_config,
>+                        &encoder_config_with_screenshare]() {
>+    CreateSenderCall(send_config);
>+    CreateReceiverCall(recv_config);
>+
>+    receive_transport_.reset(test.CreateReceiveTransport(&task_queue_));
>+    send_transport_.reset(
>+        test.CreateSendTransport(&task_queue_, sender_call_.get()));
>+    send_transport_->SetReceiver(receiver_call_->Receiver());
>+    receive_transport_->SetReceiver(sender_call_->Receiver());
>+
>+    receiver_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
>+    CreateSendConfig(1, 0, 0, send_transport_.get());
>+    CreateMatchingReceiveConfigs(receive_transport_.get());
>+
>+    // Modify send and receive configs.
>+    GetVideoSendConfig()->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
>+    video_receive_configs_[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
>+    video_receive_configs_[0].renderer = &test;
>+    // RTT needed for RemoteNtpTimeEstimator for the receive stream.
>+    video_receive_configs_[0].rtp.rtcp_xr.receiver_reference_time_report = true;
>+    // Start with realtime video.
>+    GetVideoEncoderConfig()->content_type =
>+        VideoEncoderConfig::ContentType::kRealtimeVideo;
>+    // Second encoder config for the second part of the test uses screenshare
>+    encoder_config_with_screenshare = GetVideoEncoderConfig()->Copy();
>+    encoder_config_with_screenshare.content_type =
>+        VideoEncoderConfig::ContentType::kScreen;
>+
>+    CreateVideoStreams();
>+    CreateFrameGeneratorCapturer(kDefaultFramerate, kDefaultWidth,
>+                                 kDefaultHeight);
>+    Start();
>+  });
>+
>+  test.PerformTest();
>+
>+  // Replace old send stream.
>+  task_queue_.SendTask([this, &encoder_config_with_screenshare]() {
>+    DestroyVideoSendStreams();
>+    CreateVideoSendStream(encoder_config_with_screenshare);
>+    SetVideoDegradation(DegradationPreference::BALANCED);
>+    GetVideoSendStream()->Start();
>+  });
>+
>+  // Continue to run test but now with screenshare.
>+  test.PerformTest();
>+
>+  task_queue_.SendTask([this]() {
>+    Stop();
>+    DestroyStreams();
>+    send_transport_.reset();
>+    receive_transport_.reset();
>+    DestroyCalls();
>+  });
>+
>+  // Verify that stats have been updated for both screenshare and video.
>+  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.EndToEndDelayInMs"));
>+  EXPECT_EQ(1,
>+            metrics::NumSamples("WebRTC.Video.Screenshare.EndToEndDelayInMs"));
>+  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.EndToEndDelayMaxInMs"));
>+  EXPECT_EQ(
>+      1, metrics::NumSamples("WebRTC.Video.Screenshare.EndToEndDelayMaxInMs"));
>+  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.InterframeDelayInMs"));
>+  EXPECT_EQ(
>+      1, metrics::NumSamples("WebRTC.Video.Screenshare.InterframeDelayInMs"));
>+  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.InterframeDelayMaxInMs"));
>+  EXPECT_EQ(1, metrics::NumSamples(
>+                   "WebRTC.Video.Screenshare.InterframeDelayMaxInMs"));
>+}
>+
>+TEST_F(StatsEndToEndTest, VerifyNackStats) {
>+  static const int kPacketNumberToDrop = 200;
>+  class NackObserver : public test::EndToEndTest {
>+   public:
>+    NackObserver()
>+        : EndToEndTest(kLongTimeoutMs),
>+          sent_rtp_packets_(0),
>+          dropped_rtp_packet_(0),
>+          dropped_rtp_packet_requested_(false),
>+          send_stream_(nullptr),
>+          start_runtime_ms_(-1) {}
>+
>+   private:
>+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
>+      rtc::CritScope lock(&crit_);
>+      if (++sent_rtp_packets_ == kPacketNumberToDrop) {
>+        std::unique_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
>+        RTPHeader header;
>+        EXPECT_TRUE(parser->Parse(packet, length, &header));
>+        dropped_rtp_packet_ = header.sequenceNumber;
>+        return DROP_PACKET;
>+      }
>+      VerifyStats();
>+      return SEND_PACKET;
>+    }
>+
>+    Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
>+      rtc::CritScope lock(&crit_);
>+      test::RtcpPacketParser rtcp_parser;
>+      rtcp_parser.Parse(packet, length);
>+      const std::vector<uint16_t>& nacks = rtcp_parser.nack()->packet_ids();
>+      if (!nacks.empty() && std::find(nacks.begin(), nacks.end(),
>+                                      dropped_rtp_packet_) != nacks.end()) {
>+        dropped_rtp_packet_requested_ = true;
>+      }
>+      return SEND_PACKET;
>+    }
>+
>+    void VerifyStats() RTC_EXCLUSIVE_LOCKS_REQUIRED(&crit_) {
>+      if (!dropped_rtp_packet_requested_)
>+        return;
>+      int send_stream_nack_packets = 0;
>+      int receive_stream_nack_packets = 0;
>+      VideoSendStream::Stats stats = send_stream_->GetStats();
>+      for (std::map<uint32_t, VideoSendStream::StreamStats>::const_iterator it =
>+               stats.substreams.begin();
>+           it != stats.substreams.end(); ++it) {
>+        const VideoSendStream::StreamStats& stream_stats = it->second;
>+        send_stream_nack_packets +=
>+            stream_stats.rtcp_packet_type_counts.nack_packets;
>+      }
>+      for (size_t i = 0; i < receive_streams_.size(); ++i) {
>+        VideoReceiveStream::Stats stats = receive_streams_[i]->GetStats();
>+        receive_stream_nack_packets +=
>+            stats.rtcp_packet_type_counts.nack_packets;
>+      }
>+      if (send_stream_nack_packets >= 1 && receive_stream_nack_packets >= 1) {
>+        // NACK packet sent on receive stream and received on sent stream.
>+        if (MinMetricRunTimePassed())
>+          observation_complete_.Set();
>+      }
>+    }
>+
>+    bool MinMetricRunTimePassed() {
>+      int64_t now = Clock::GetRealTimeClock()->TimeInMilliseconds();
>+      if (start_runtime_ms_ == -1) {
>+        start_runtime_ms_ = now;
>+        return false;
>+      }
>+      int64_t elapsed_sec = (now - start_runtime_ms_) / 1000;
>+      return elapsed_sec > metrics::kMinRunTimeInSeconds;
>+    }
>+
>+    void ModifyVideoConfigs(
>+        VideoSendStream::Config* send_config,
>+        std::vector<VideoReceiveStream::Config>* receive_configs,
>+        VideoEncoderConfig* encoder_config) override {
>+      send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
>+      (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
>+      (*receive_configs)[0].renderer = &fake_renderer_;
>+    }
>+
>+    void OnVideoStreamsCreated(
>+        VideoSendStream* send_stream,
>+        const std::vector<VideoReceiveStream*>& receive_streams) override {
>+      send_stream_ = send_stream;
>+      receive_streams_ = receive_streams;
>+    }
>+
>+    void PerformTest() override {
>+      EXPECT_TRUE(Wait()) << "Timed out waiting for packet to be NACKed.";
>+    }
>+
>+    test::FakeVideoRenderer fake_renderer_;
>+    rtc::CriticalSection crit_;
>+    uint64_t sent_rtp_packets_;
>+    uint16_t dropped_rtp_packet_ RTC_GUARDED_BY(&crit_);
>+    bool dropped_rtp_packet_requested_ RTC_GUARDED_BY(&crit_);
>+    std::vector<VideoReceiveStream*> receive_streams_;
>+    VideoSendStream* send_stream_;
>+    int64_t start_runtime_ms_;
>+  } test;
>+
>+  metrics::Reset();
>+  RunBaseTest(&test);
>+
>+  EXPECT_EQ(
>+      1, metrics::NumSamples("WebRTC.Video.UniqueNackRequestsSentInPercent"));
>+  EXPECT_EQ(1, metrics::NumSamples(
>+                   "WebRTC.Video.UniqueNackRequestsReceivedInPercent"));
>+  EXPECT_GT(metrics::MinSample("WebRTC.Video.NackPacketsSentPerMinute"), 0);
>+}
>+
>+TEST_F(StatsEndToEndTest, CallReportsRttForSender) {
>+  static const int kSendDelayMs = 30;
>+  static const int kReceiveDelayMs = 70;
>+
>+  std::unique_ptr<test::DirectTransport> sender_transport;
>+  std::unique_ptr<test::DirectTransport> receiver_transport;
>+
>+  task_queue_.SendTask([this, &sender_transport, &receiver_transport]() {
>+    FakeNetworkPipe::Config config;
>+    config.queue_delay_ms = kSendDelayMs;
>+    CreateCalls();
>+    sender_transport = absl::make_unique<test::DirectTransport>(
>+        &task_queue_, config, sender_call_.get(), payload_type_map_);
>+    config.queue_delay_ms = kReceiveDelayMs;
>+    receiver_transport = absl::make_unique<test::DirectTransport>(
>+        &task_queue_, config, receiver_call_.get(), payload_type_map_);
>+    sender_transport->SetReceiver(receiver_call_->Receiver());
>+    receiver_transport->SetReceiver(sender_call_->Receiver());
>+
>+    CreateSendConfig(1, 0, 0, sender_transport.get());
>+    CreateMatchingReceiveConfigs(receiver_transport.get());
>+
>+    CreateVideoStreams();
>+    CreateFrameGeneratorCapturer(kDefaultFramerate, kDefaultWidth,
>+                                 kDefaultHeight);
>+    Start();
>+  });
>+
>+  int64_t start_time_ms = clock_->TimeInMilliseconds();
>+  while (true) {
>+    Call::Stats stats = sender_call_->GetStats();
>+    ASSERT_GE(start_time_ms + kDefaultTimeoutMs, clock_->TimeInMilliseconds())
>+        << "No RTT stats before timeout!";
>+    if (stats.rtt_ms != -1) {
>+      // To avoid failures caused by rounding or minor ntp clock adjustments,
>+      // relax expectation by 1ms.
>+      constexpr int kAllowedErrorMs = 1;
>+      EXPECT_GE(stats.rtt_ms, kSendDelayMs + kReceiveDelayMs - kAllowedErrorMs);
>+      break;
>+    }
>+    SleepMs(10);
>+  }
>+
>+  task_queue_.SendTask([this, &sender_transport, &receiver_transport]() {
>+    Stop();
>+    DestroyStreams();
>+    sender_transport.reset();
>+    receiver_transport.reset();
>+    DestroyCalls();
>+  });
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/transport_feedback_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/transport_feedback_tests.cc
>new file mode 100644
>index 00000000000..525e661e734
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/end_to_end_tests/transport_feedback_tests.cc
>@@ -0,0 +1,506 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "test/call_test.h"
>+#include "test/field_trial.h"
>+#include "test/gtest.h"
>+
>+#include "modules/rtp_rtcp/source/byte_io.h"
>+#include "test/rtcp_packet_parser.h"
>+#include "video/end_to_end_tests/multi_stream_tester.h"
>+
>+namespace webrtc {
>+
>+class TransportFeedbackEndToEndTest
>+    : public test::CallTest,
>+      public testing::WithParamInterface<std::string> {
>+ public:
>+  TransportFeedbackEndToEndTest() : field_trial_(GetParam()) {}
>+
>+  virtual ~TransportFeedbackEndToEndTest() {
>+  }
>+
>+ private:
>+  test::ScopedFieldTrials field_trial_;
>+};
>+
>+INSTANTIATE_TEST_CASE_P(
>+    FieldTrials,
>+    TransportFeedbackEndToEndTest,
>+    ::testing::Values("WebRTC-RoundRobinPacing/Disabled/",
>+                      "WebRTC-RoundRobinPacing/Enabled/",
>+                      "WebRTC-TaskQueueCongestionControl/Enabled/"));
>+
>+TEST_P(TransportFeedbackEndToEndTest, AssignsTransportSequenceNumbers) {
>+  static const int kExtensionId = 5;
>+
>+  class RtpExtensionHeaderObserver : public test::DirectTransport {
>+   public:
>+    RtpExtensionHeaderObserver(
>+        test::SingleThreadedTaskQueueForTesting* task_queue,
>+        Call* sender_call,
>+        const uint32_t& first_media_ssrc,
>+        const std::map<uint32_t, uint32_t>& ssrc_map,
>+        const std::map<uint8_t, MediaType>& payload_type_map)
>+        : DirectTransport(task_queue, sender_call, payload_type_map),
>+          done_(false, false),
>+          parser_(RtpHeaderParser::Create()),
>+          first_media_ssrc_(first_media_ssrc),
>+          rtx_to_media_ssrcs_(ssrc_map),
>+          padding_observed_(false),
>+          rtx_padding_observed_(false),
>+          retransmit_observed_(false),
>+          started_(false) {
>+      parser_->RegisterRtpHeaderExtension(kRtpExtensionTransportSequenceNumber,
>+                                          kExtensionId);
>+    }
>+    virtual ~RtpExtensionHeaderObserver() {}
>+
>+    bool SendRtp(const uint8_t* data,
>+                 size_t length,
>+                 const PacketOptions& options) override {
>+      {
>+        rtc::CritScope cs(&lock_);
>+
>+        if (IsDone())
>+          return false;
>+
>+        if (started_) {
>+          RTPHeader header;
>+          EXPECT_TRUE(parser_->Parse(data, length, &header));
>+          bool drop_packet = false;
>+
>+          EXPECT_TRUE(header.extension.hasTransportSequenceNumber);
>+          EXPECT_EQ(options.packet_id,
>+                    header.extension.transportSequenceNumber);
>+          if (!streams_observed_.empty()) {
>+            // Unwrap packet id and verify uniqueness.
>+            int64_t packet_id = unwrapper_.Unwrap(options.packet_id);
>+            EXPECT_TRUE(received_packed_ids_.insert(packet_id).second);
>+          }
>+
>+          // Drop (up to) every 17th packet, so we get retransmits.
>+          // Only drop media, and not on the first stream (otherwise it will be
>+          // hard to distinguish from padding, which is always sent on the first
>+          // stream).
>+          if (header.payloadType != kSendRtxPayloadType &&
>+              header.ssrc != first_media_ssrc_ &&
>+              header.extension.transportSequenceNumber % 17 == 0) {
>+            dropped_seq_[header.ssrc].insert(header.sequenceNumber);
>+            drop_packet = true;
>+          }
>+
>+          if (header.payloadType == kSendRtxPayloadType) {
>+            uint16_t original_sequence_number =
>+                ByteReader<uint16_t>::ReadBigEndian(&data[header.headerLength]);
>+            uint32_t original_ssrc =
>+                rtx_to_media_ssrcs_.find(header.ssrc)->second;
>+            std::set<uint16_t>* seq_no_map = &dropped_seq_[original_ssrc];
>+            auto it = seq_no_map->find(original_sequence_number);
>+            if (it != seq_no_map->end()) {
>+              retransmit_observed_ = true;
>+              seq_no_map->erase(it);
>+            } else {
>+              rtx_padding_observed_ = true;
>+            }
>+          } else {
>+            streams_observed_.insert(header.ssrc);
>+          }
>+
>+          if (IsDone())
>+            done_.Set();
>+
>+          if (drop_packet)
>+            return true;
>+        }
>+      }
>+
>+      return test::DirectTransport::SendRtp(data, length, options);
>+    }
>+
>+    bool IsDone() {
>+      bool observed_types_ok =
>+          streams_observed_.size() == MultiStreamTester::kNumStreams &&
>+          retransmit_observed_ && rtx_padding_observed_;
>+      if (!observed_types_ok)
>+        return false;
>+      // We should not have any gaps in the sequence number range.
>+      size_t seqno_range =
>+          *received_packed_ids_.rbegin() - *received_packed_ids_.begin() + 1;
>+      return seqno_range == received_packed_ids_.size();
>+    }
>+
>+    bool Wait() {
>+      {
>+        // Can't be sure until this point that rtx_to_media_ssrcs_ etc have
>+        // been initialized and are OK to read.
>+        rtc::CritScope cs(&lock_);
>+        started_ = true;
>+      }
>+      return done_.Wait(kDefaultTimeoutMs);
>+    }
>+
>+    rtc::CriticalSection lock_;
>+    rtc::Event done_;
>+    std::unique_ptr<RtpHeaderParser> parser_;
>+    SequenceNumberUnwrapper unwrapper_;
>+    std::set<int64_t> received_packed_ids_;
>+    std::set<uint32_t> streams_observed_;
>+    std::map<uint32_t, std::set<uint16_t>> dropped_seq_;
>+    const uint32_t& first_media_ssrc_;
>+    const std::map<uint32_t, uint32_t>& rtx_to_media_ssrcs_;
>+    bool padding_observed_;
>+    bool rtx_padding_observed_;
>+    bool retransmit_observed_;
>+    bool started_;
>+  };
>+
>+  class TransportSequenceNumberTester : public MultiStreamTester {
>+   public:
>+    explicit TransportSequenceNumberTester(
>+        test::SingleThreadedTaskQueueForTesting* task_queue)
>+        : MultiStreamTester(task_queue),
>+          first_media_ssrc_(0),
>+          observer_(nullptr) {}
>+    virtual ~TransportSequenceNumberTester() {}
>+
>+   protected:
>+    void Wait() override {
>+      RTC_DCHECK(observer_);
>+      EXPECT_TRUE(observer_->Wait());
>+    }
>+
>+    void UpdateSendConfig(
>+        size_t stream_index,
>+        VideoSendStream::Config* send_config,
>+        VideoEncoderConfig* encoder_config,
>+        test::FrameGeneratorCapturer** frame_generator) override {
>+      send_config->rtp.extensions.clear();
>+      send_config->rtp.extensions.push_back(RtpExtension(
>+          RtpExtension::kTransportSequenceNumberUri, kExtensionId));
>+
>+      // Force some padding to be sent. Note that since we do send media
>+      // packets we can not guarantee that a padding only packet is sent.
>+      // Instead, padding will most likely be send as an RTX packet.
>+      const int kPaddingBitrateBps = 50000;
>+      encoder_config->max_bitrate_bps = 200000;
>+      encoder_config->min_transmit_bitrate_bps =
>+          encoder_config->max_bitrate_bps + kPaddingBitrateBps;
>+
>+      // Configure RTX for redundant payload padding.
>+      send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
>+      send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[stream_index]);
>+      send_config->rtp.rtx.payload_type = kSendRtxPayloadType;
>+      rtx_to_media_ssrcs_[kSendRtxSsrcs[stream_index]] =
>+          send_config->rtp.ssrcs[0];
>+
>+      if (stream_index == 0)
>+        first_media_ssrc_ = send_config->rtp.ssrcs[0];
>+    }
>+
>+    void UpdateReceiveConfig(
>+        size_t stream_index,
>+        VideoReceiveStream::Config* receive_config) override {
>+      receive_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
>+      receive_config->rtp.extensions.clear();
>+      receive_config->rtp.extensions.push_back(RtpExtension(
>+          RtpExtension::kTransportSequenceNumberUri, kExtensionId));
>+      receive_config->renderer = &fake_renderer_;
>+    }
>+
>+    test::DirectTransport* CreateSendTransport(
>+        test::SingleThreadedTaskQueueForTesting* task_queue,
>+        Call* sender_call) override {
>+      std::map<uint8_t, MediaType> payload_type_map =
>+          MultiStreamTester::payload_type_map_;
>+      RTC_DCHECK(payload_type_map.find(kSendRtxPayloadType) ==
>+                 payload_type_map.end());
>+      payload_type_map[kSendRtxPayloadType] = MediaType::VIDEO;
>+      observer_ = new RtpExtensionHeaderObserver(
>+          task_queue, sender_call, first_media_ssrc_, rtx_to_media_ssrcs_,
>+          payload_type_map);
>+      return observer_;
>+    }
>+
>+   private:
>+    test::FakeVideoRenderer fake_renderer_;
>+    uint32_t first_media_ssrc_;
>+    std::map<uint32_t, uint32_t> rtx_to_media_ssrcs_;
>+    RtpExtensionHeaderObserver* observer_;
>+  } tester(&task_queue_);
>+
>+  tester.RunTest();
>+}
>+
>+class TransportFeedbackTester : public test::EndToEndTest {
>+ public:
>+  TransportFeedbackTester(bool feedback_enabled,
>+                          size_t num_video_streams,
>+                          size_t num_audio_streams)
>+      : EndToEndTest(
>+            ::webrtc::TransportFeedbackEndToEndTest::kDefaultTimeoutMs),
>+        feedback_enabled_(feedback_enabled),
>+        num_video_streams_(num_video_streams),
>+        num_audio_streams_(num_audio_streams),
>+        receiver_call_(nullptr) {
>+    // Only one stream of each supported for now.
>+    EXPECT_LE(num_video_streams, 1u);
>+    EXPECT_LE(num_audio_streams, 1u);
>+  }
>+
>+ protected:
>+  Action OnSendRtcp(const uint8_t* data, size_t length) override {
>+    EXPECT_FALSE(HasTransportFeedback(data, length));
>+    return SEND_PACKET;
>+  }
>+
>+  Action OnReceiveRtcp(const uint8_t* data, size_t length) override {
>+    if (HasTransportFeedback(data, length))
>+      observation_complete_.Set();
>+    return SEND_PACKET;
>+  }
>+
>+  bool HasTransportFeedback(const uint8_t* data, size_t length) const {
>+    test::RtcpPacketParser parser;
>+    EXPECT_TRUE(parser.Parse(data, length));
>+    return parser.transport_feedback()->num_packets() > 0;
>+  }
>+
>+  void PerformTest() override {
>+    const int64_t kDisabledFeedbackTimeoutMs = 5000;
>+    EXPECT_EQ(feedback_enabled_,
>+              observation_complete_.Wait(feedback_enabled_
>+                                             ? test::CallTest::kDefaultTimeoutMs
>+                                             : kDisabledFeedbackTimeoutMs));
>+  }
>+
>+  void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
>+    receiver_call_ = receiver_call;
>+  }
>+
>+  size_t GetNumVideoStreams() const override { return num_video_streams_; }
>+  size_t GetNumAudioStreams() const override { return num_audio_streams_; }
>+
>+  void ModifyVideoConfigs(
>+      VideoSendStream::Config* send_config,
>+      std::vector<VideoReceiveStream::Config>* receive_configs,
>+      VideoEncoderConfig* encoder_config) override {
>+    (*receive_configs)[0].rtp.transport_cc = feedback_enabled_;
>+  }
>+
>+  void ModifyAudioConfigs(
>+      AudioSendStream::Config* send_config,
>+      std::vector<AudioReceiveStream::Config>* receive_configs) override {
>+    send_config->rtp.extensions.clear();
>+    send_config->rtp.extensions.push_back(
>+        RtpExtension(RtpExtension::kTransportSequenceNumberUri, kExtensionId));
>+    (*receive_configs)[0].rtp.extensions.clear();
>+    (*receive_configs)[0].rtp.extensions = send_config->rtp.extensions;
>+    (*receive_configs)[0].rtp.transport_cc = feedback_enabled_;
>+  }
>+
>+ private:
>+  static const int kExtensionId = 5;
>+  const bool feedback_enabled_;
>+  const size_t num_video_streams_;
>+  const size_t num_audio_streams_;
>+  Call* receiver_call_;
>+};
>+
>+TEST_P(TransportFeedbackEndToEndTest, VideoReceivesTransportFeedback) {
>+  TransportFeedbackTester test(true, 1, 0);
>+  RunBaseTest(&test);
>+}
>+
>+TEST_P(TransportFeedbackEndToEndTest, VideoTransportFeedbackNotConfigured) {
>+  TransportFeedbackTester test(false, 1, 0);
>+  RunBaseTest(&test);
>+}
>+
>+TEST_P(TransportFeedbackEndToEndTest, AudioReceivesTransportFeedback) {
>+  TransportFeedbackTester test(true, 0, 1);
>+  RunBaseTest(&test);
>+}
>+
>+TEST_P(TransportFeedbackEndToEndTest, AudioTransportFeedbackNotConfigured) {
>+  TransportFeedbackTester test(false, 0, 1);
>+  RunBaseTest(&test);
>+}
>+
>+TEST_P(TransportFeedbackEndToEndTest, AudioVideoReceivesTransportFeedback) {
>+  TransportFeedbackTester test(true, 1, 1);
>+  RunBaseTest(&test);
>+}
>+
>+TEST_P(TransportFeedbackEndToEndTest,
>+       StopsAndResumesMediaWhenCongestionWindowFull) {
>+  test::ScopedFieldTrials override_field_trials(
>+      "WebRTC-CwndExperiment/Enabled-250/");
>+
>+  class TransportFeedbackTester : public test::EndToEndTest {
>+   public:
>+    TransportFeedbackTester(size_t num_video_streams, size_t num_audio_streams)
>+        : EndToEndTest(
>+              ::webrtc::TransportFeedbackEndToEndTest::kDefaultTimeoutMs),
>+          num_video_streams_(num_video_streams),
>+          num_audio_streams_(num_audio_streams),
>+          media_sent_(0),
>+          media_sent_before_(0),
>+          padding_sent_(0) {
>+      // Only one stream of each supported for now.
>+      EXPECT_LE(num_video_streams, 1u);
>+      EXPECT_LE(num_audio_streams, 1u);
>+    }
>+
>+   protected:
>+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
>+      RTPHeader header;
>+      EXPECT_TRUE(parser_->Parse(packet, length, &header));
>+      const bool only_padding =
>+          header.headerLength + header.paddingLength == length;
>+      rtc::CritScope lock(&crit_);
>+      // Padding is expected in congested state to probe for connectivity when
>+      // packets has been dropped.
>+      if (only_padding) {
>+        media_sent_before_ = media_sent_;
>+        ++padding_sent_;
>+      } else {
>+        ++media_sent_;
>+        if (padding_sent_ == 0) {
>+          ++media_sent_before_;
>+          EXPECT_LT(media_sent_, 40)
>+              << "Media sent without feedback when congestion window is full.";
>+        } else if (media_sent_ > media_sent_before_) {
>+          observation_complete_.Set();
>+        }
>+      }
>+      return SEND_PACKET;
>+    }
>+
>+    Action OnReceiveRtcp(const uint8_t* data, size_t length) override {
>+      rtc::CritScope lock(&crit_);
>+      // To fill up the congestion window we drop feedback on packets after 20
>+      // packets have been sent. This means that any packets that has not yet
>+      // received feedback after that will be considered as oustanding data and
>+      // therefore filling up the congestion window. In the congested state, the
>+      // pacer should send padding packets to trigger feedback in case all
>+      // feedback of previous traffic was lost. This test listens for the
>+      // padding packets and when 2 padding packets have been received, feedback
>+      // will be let trough again. This should cause the pacer to continue
>+      // sending meadia yet again.
>+      if (media_sent_ > 20 && HasTransportFeedback(data, length) &&
>+          padding_sent_ < 2) {
>+        return DROP_PACKET;
>+      }
>+      return SEND_PACKET;
>+    }
>+
>+    bool HasTransportFeedback(const uint8_t* data, size_t length) const {
>+      test::RtcpPacketParser parser;
>+      EXPECT_TRUE(parser.Parse(data, length));
>+      return parser.transport_feedback()->num_packets() > 0;
>+    }
>+    void ModifySenderCallConfig(Call::Config* config) override {
>+      config->bitrate_config.max_bitrate_bps = 300000;
>+    }
>+
>+    void PerformTest() override {
>+      const int64_t kFailureTimeoutMs = 10000;
>+      EXPECT_TRUE(observation_complete_.Wait(kFailureTimeoutMs))
>+          << "Stream not continued after congestion window full.";
>+    }
>+
>+    size_t GetNumVideoStreams() const override { return num_video_streams_; }
>+    size_t GetNumAudioStreams() const override { return num_audio_streams_; }
>+
>+   private:
>+    const size_t num_video_streams_;
>+    const size_t num_audio_streams_;
>+    rtc::CriticalSection crit_;
>+    int media_sent_ RTC_GUARDED_BY(crit_);
>+    int media_sent_before_ RTC_GUARDED_BY(crit_);
>+    int padding_sent_ RTC_GUARDED_BY(crit_);
>+  } test(1, 0);
>+  RunBaseTest(&test);
>+}
>+
>+TEST_P(TransportFeedbackEndToEndTest, TransportSeqNumOnAudioAndVideo) {
>+  static constexpr int kExtensionId = 8;
>+  static constexpr size_t kMinPacketsToWaitFor = 50;
>+  class TransportSequenceNumberTest : public test::EndToEndTest {
>+   public:
>+    TransportSequenceNumberTest()
>+        : EndToEndTest(kDefaultTimeoutMs),
>+          video_observed_(false),
>+          audio_observed_(false) {
>+      parser_->RegisterRtpHeaderExtension(kRtpExtensionTransportSequenceNumber,
>+                                          kExtensionId);
>+    }
>+
>+    size_t GetNumVideoStreams() const override { return 1; }
>+    size_t GetNumAudioStreams() const override { return 1; }
>+
>+    void ModifyAudioConfigs(
>+        AudioSendStream::Config* send_config,
>+        std::vector<AudioReceiveStream::Config>* receive_configs) override {
>+      send_config->rtp.extensions.clear();
>+      send_config->rtp.extensions.push_back(RtpExtension(
>+          RtpExtension::kTransportSequenceNumberUri, kExtensionId));
>+      (*receive_configs)[0].rtp.extensions.clear();
>+      (*receive_configs)[0].rtp.extensions = send_config->rtp.extensions;
>+    }
>+
>+    Action OnSendRtp(const uint8_t* packet, size_t length) override {
>+      RTPHeader header;
>+      EXPECT_TRUE(parser_->Parse(packet, length, &header));
>+      EXPECT_TRUE(header.extension.hasTransportSequenceNumber);
>+      // Unwrap packet id and verify uniqueness.
>+      int64_t packet_id =
>+          unwrapper_.Unwrap(header.extension.transportSequenceNumber);
>+      EXPECT_TRUE(received_packet_ids_.insert(packet_id).second);
>+
>+      if (header.ssrc == kVideoSendSsrcs[0])
>+        video_observed_ = true;
>+      if (header.ssrc == kAudioSendSsrc)
>+        audio_observed_ = true;
>+      if (audio_observed_ && video_observed_ &&
>+          received_packet_ids_.size() >= kMinPacketsToWaitFor) {
>+        size_t packet_id_range =
>+            *received_packet_ids_.rbegin() - *received_packet_ids_.begin() + 1;
>+        EXPECT_EQ(received_packet_ids_.size(), packet_id_range);
>+        observation_complete_.Set();
>+      }
>+      return SEND_PACKET;
>+    }
>+
>+    void PerformTest() override {
>+      EXPECT_TRUE(Wait()) << "Timed out while waiting for audio and video "
>+                             "packets with transport sequence number.";
>+    }
>+
>+    void ExpectSuccessful() {
>+      EXPECT_TRUE(video_observed_);
>+      EXPECT_TRUE(audio_observed_);
>+      EXPECT_GE(received_packet_ids_.size(), kMinPacketsToWaitFor);
>+    }
>+
>+   private:
>+    bool video_observed_;
>+    bool audio_observed_;
>+    SequenceNumberUnwrapper unwrapper_;
>+    std::set<int64_t> received_packet_ids_;
>+  } test;
>+
>+  RunBaseTest(&test);
>+  // Double check conditions for successful test to produce better error
>+  // message when the test fail.
>+  test.ExpectSuccessful();
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/full_stack_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/full_stack_tests.cc
>index 9c8d3a625e8..dc8527ebde4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/full_stack_tests.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/full_stack_tests.cc
>@@ -9,35 +9,70 @@
>  */
> #include <stdio.h>
> 
>+#include "media/base/vp9_profile.h"
>+#include "modules/video_coding/codecs/vp9/include/vp9.h"
> #include "rtc_base/experiments/alr_experiment.h"
>+#include "rtc_base/flags.h"
>+#include "system_wrappers/include/field_trial_default.h"
> #include "test/field_trial.h"
> #include "test/gtest.h"
> #include "video/video_quality_test.h"
> 
>+namespace webrtc {
>+namespace flags {
>+
>+DEFINE_bool(logs, false, "print logs to stderr");
>+
>+DEFINE_string(rtc_event_log_name,
>+              "",
>+              "Filename for rtc event log. Two files "
>+              "with \"_send\" and \"_recv\" suffixes will be created.");
>+std::string RtcEventLogName() {
>+  return static_cast<std::string>(FLAG_rtc_event_log_name);
>+}
>+DEFINE_string(rtp_dump_name, "", "Filename for dumped received RTP stream.");
>+std::string RtpDumpName() {
>+  return static_cast<std::string>(FLAG_rtp_dump_name);
>+}
>+DEFINE_string(encoded_frame_path,
>+              "",
>+              "The base path for encoded frame logs. Created files will have "
>+              "the form <encoded_frame_path>.<n>.(recv|send.<m>).ivf");
>+std::string EncodedFramePath() {
>+  return static_cast<std::string>(FLAG_encoded_frame_path);
>+}
>+}  // namespace flags
>+}  // namespace webrtc
>+
> namespace webrtc {
> 
> namespace {
> static const int kFullStackTestDurationSecs = 45;
>-}  // namespace
>+const char kRoundRobinPacingQueueExperiment[] =
>+    "WebRTC-RoundRobinPacing/Enabled/";
>+const char kPacerPushBackExperiment[] =
>+    "WebRTC-PacerPushbackExperiment/Enabled/";
> 
>-class FullStackTest : public VideoQualityTest {
>+struct ParamsWithLogging : public VideoQualityTest::Params {
>  public:
>-  void RunTest(const VideoQualityTest::Params &params) {
>-    RunWithAnalyzer(params);
>+  ParamsWithLogging() {
>+    // Use these logging flags by default, for everything.
>+    logging = {flags::RtcEventLogName(), flags::RtpDumpName(),
>+               flags::EncodedFramePath()};
>   }
>-
>- protected:
>-  const std::string kScreenshareSimulcastExperiment =
>-      "WebRTC-SimulcastScreenshare/Enabled/";
>-  const std::string kAlrProbingExperiment =
>-      std::string(AlrExperimentSettings::kScreenshareProbingBweExperimentName) +
>-      "/1.1,2875,85,20,-20,0/";
>-  const std::string kRoundRobinPacingQueueExperiment =
>-      "WebRTC-RoundRobinPacing/Enabled/";
>-  const std::string kPacerPushBackExperiment =
>-      "WebRTC-PacerPushbackExperiment/Enabled/";
> };
> 
>+std::unique_ptr<VideoQualityTestFixtureInterface>
>+CreateVideoQualityTestFixture() {
>+  return absl::make_unique<VideoQualityTest>(nullptr);
>+}
>+
>+// Takes the current active field trials set, and appends some new trials.
>+std::string AppendFieldTrials(std::string new_trial_string) {
>+  return std::string(field_trial::GetFieldTrialString()) + new_trial_string;
>+}
>+}  // namespace
>+
> // VideoQualityTest::Params params = {
> //   { ... },      // Common.
> //   { ... },      // Video-specific settings.
>@@ -49,501 +84,612 @@ class FullStackTest : public VideoQualityTest {
> // };
> 
> #if !defined(RTC_DISABLE_VP9)
>-TEST_F(FullStackTest, ForemanCifWithoutPacketLossVp9) {
>-  // TODO(pbos): Decide on psnr/ssim thresholds for foreman_cif.
>-  VideoQualityTest::Params foreman_cif;
>+TEST(FullStackTest, ForemanCifWithoutPacketLossVp9) {
>+  auto fixture = CreateVideoQualityTestFixture();
>+  ParamsWithLogging foreman_cif;
>   foreman_cif.call.send_side_bwe = true;
>   foreman_cif.video[0] = {true,   352,    288,   30,    700000,
>                           700000, 700000, false, "VP9", 1,
>-                          0,      0,      false, false, "foreman_cif"};
>+                          0,      0,      false, false, false, "foreman_cif"};
>   foreman_cif.analyzer = {"foreman_cif_net_delay_0_0_plr_0_VP9", 0.0, 0.0,
>                           kFullStackTestDurationSecs};
>-  RunTest(foreman_cif);
>+  fixture->RunWithAnalyzer(foreman_cif);
> }
> 
>-TEST_F(FullStackTest, ForemanCifPlr5Vp9) {
>-  VideoQualityTest::Params foreman_cif;
>+TEST(FullStackTest, ForemanCifPlr5Vp9) {
>+  auto fixture = CreateVideoQualityTestFixture();
>+  ParamsWithLogging foreman_cif;
>   foreman_cif.call.send_side_bwe = true;
>   foreman_cif.video[0] = {true,   352,     288,   30,    30000,
>                           500000, 2000000, false, "VP9", 1,
>-                          0,      0,       false, false, "foreman_cif"};
>+                          0,      0,       false, false, false, "foreman_cif"};
>   foreman_cif.analyzer = {"foreman_cif_delay_50_0_plr_5_VP9", 0.0, 0.0,
>                           kFullStackTestDurationSecs};
>   foreman_cif.pipe.loss_percent = 5;
>   foreman_cif.pipe.queue_delay_ms = 50;
>-  RunTest(foreman_cif);
>+  fixture->RunWithAnalyzer(foreman_cif);
>+}
>+
>+TEST(FullStackTest, GeneratorWithoutPacketLossVp9Profile2) {
>+  // Profile 2 might not be available on some platforms until
>+  // https://bugs.chromium.org/p/webm/issues/detail?id=1544 is solved.
>+  bool profile_2_is_supported = false;
>+  for (const auto& codec : SupportedVP9Codecs()) {
>+    if (ParseSdpForVP9Profile(codec.parameters)
>+            .value_or(VP9Profile::kProfile0) == VP9Profile::kProfile2) {
>+      profile_2_is_supported = true;
>+    }
>+  }
>+  if (!profile_2_is_supported)
>+    return;
>+  auto fixture = CreateVideoQualityTestFixture();
>+
>+  SdpVideoFormat::Parameters vp92 = {
>+      {kVP9FmtpProfileId, VP9ProfileToString(VP9Profile::kProfile2)}};
>+  ParamsWithLogging generator;
>+  generator.call.send_side_bwe = true;
>+  generator.video[0] = {
>+      true, 352, 288, 30,    700000, 700000, 700000,          false, "VP9",
>+      1,    0,   0,   false, false,  false,  "GeneratorI010", 0,     vp92};
>+  generator.analyzer = {"generator_net_delay_0_0_plr_0_VP9Profile2", 0.0, 0.0,
>+                        kFullStackTestDurationSecs};
>+  fixture->RunWithAnalyzer(generator);
>+}
>+
>+TEST(FullStackTest, ForemanCifWithoutPacketLossMultiplexI420Frame) {
>+  auto fixture = CreateVideoQualityTestFixture();
>+  ParamsWithLogging foreman_cif;
>+  foreman_cif.call.send_side_bwe = true;
>+  foreman_cif.video[0] = {true,   352,    288,   30,          700000,
>+                          700000, 700000, false, "multiplex", 1,
>+                          0,      0,      false, false,       false,
>+                          "foreman_cif"};
>+  foreman_cif.analyzer = {"foreman_cif_net_delay_0_0_plr_0_Multiplex", 0.0, 0.0,
>+                          kFullStackTestDurationSecs};
>+  fixture->RunWithAnalyzer(foreman_cif);
> }
>+
>+TEST(FullStackTest, GeneratorWithoutPacketLossMultiplexI420AFrame) {
>+  auto fixture = CreateVideoQualityTestFixture();
>+
>+  ParamsWithLogging generator;
>+  generator.call.send_side_bwe = true;
>+  generator.video[0] = {true,   352,    288,   30,          700000,
>+                        700000, 700000, false, "multiplex", 1,
>+                        0,      0,      false, false,       false,
>+                        "GeneratorI420A"};
>+  generator.analyzer = {"generator_net_delay_0_0_plr_0_Multiplex", 0.0, 0.0,
>+                        kFullStackTestDurationSecs};
>+  fixture->RunWithAnalyzer(generator);
>+}
>+
> #endif  // !defined(RTC_DISABLE_VP9)
> 
>-TEST_F(FullStackTest, ParisQcifWithoutPacketLoss) {
>-  VideoQualityTest::Params paris_qcif;
>+#if defined(WEBRTC_LINUX)
>+// Crashes on the linux trusty perf bot: bugs.webrtc.org/9129.
>+#define MAYBE_ParisQcifWithoutPacketLoss DISABLED_ParisQcifWithoutPacketLoss
>+#else
>+#define MAYBE_ParisQcifWithoutPacketLoss ParisQcifWithoutPacketLoss
>+#endif
>+TEST(FullStackTest, MAYBE_ParisQcifWithoutPacketLoss) {
>+  auto fixture = CreateVideoQualityTestFixture();
>+  ParamsWithLogging paris_qcif;
>   paris_qcif.call.send_side_bwe = true;
>   paris_qcif.video[0] = {true,   176,    144,   30,    300000,
>                          300000, 300000, false, "VP8", 1,
>-                         0,      0,      false, false, "paris_qcif"};
>+                         0,      0,      false, false, false, "paris_qcif"};
>   paris_qcif.analyzer = {"net_delay_0_0_plr_0", 36.0, 0.96,
>                          kFullStackTestDurationSecs};
>-  RunTest(paris_qcif);
>+  fixture->RunWithAnalyzer(paris_qcif);
> }
> 
>-TEST_F(FullStackTest, ForemanCifWithoutPacketLoss) {
>+TEST(FullStackTest, ForemanCifWithoutPacketLoss) {
>+  auto fixture = CreateVideoQualityTestFixture();
>   // TODO(pbos): Decide on psnr/ssim thresholds for foreman_cif.
>-  VideoQualityTest::Params foreman_cif;
>+  ParamsWithLogging foreman_cif;
>   foreman_cif.call.send_side_bwe = true;
>   foreman_cif.video[0] = {true,   352,    288,   30,    700000,
>                           700000, 700000, false, "VP8", 1,
>-                          0,      0,      false, false, "foreman_cif"};
>+                          0,      0,      false, false, false, "foreman_cif"};
>   foreman_cif.analyzer = {"foreman_cif_net_delay_0_0_plr_0", 0.0, 0.0,
>                           kFullStackTestDurationSecs};
>-  RunTest(foreman_cif);
>+  fixture->RunWithAnalyzer(foreman_cif);
> }
> 
>-TEST_F(FullStackTest, ForemanCif30kbpsWithoutPacketLoss) {
>-  VideoQualityTest::Params foreman_cif;
>+TEST(FullStackTest, ForemanCif30kbpsWithoutPacketLoss) {
>+  auto fixture = CreateVideoQualityTestFixture();
>+  ParamsWithLogging foreman_cif;
>   foreman_cif.call.send_side_bwe = true;
>   foreman_cif.video[0] = {true,  352,   288,   10,    30000,
>                           30000, 30000, false, "VP8", 1,
>-                          0,     0,     false, false, "foreman_cif"};
>+                          0,     0,     false, false, false, "foreman_cif"};
>   foreman_cif.analyzer = {"foreman_cif_30kbps_net_delay_0_0_plr_0", 0.0, 0.0,
>                           kFullStackTestDurationSecs};
>-  RunTest(foreman_cif);
>+  fixture->RunWithAnalyzer(foreman_cif);
> }
> 
>-TEST_F(FullStackTest, ForemanCifPlr5) {
>+// Link capacity below default start rate. Automatic down scaling enabled.
>+TEST(FullStackTest, ForemanCifLink150kbpsWithoutPacketLoss) {
>+  auto fixture = CreateVideoQualityTestFixture();
>   VideoQualityTest::Params foreman_cif;
>   foreman_cif.call.send_side_bwe = true;
>   foreman_cif.video[0] = {true,   352,     288,   30,    30000,
>                           500000, 2000000, false, "VP8", 1,
>-                          0,      0,       false, false, "foreman_cif"};
>+                          0,      0,       false,  false, true, "foreman_cif"};
>+  foreman_cif.analyzer = {"foreman_cif_link_150kbps_net_delay_0_0_plr_0",
>+                          0.0, 0.0,
>+                          kFullStackTestDurationSecs};
>+  foreman_cif.pipe.link_capacity_kbps = 150;
>+  fixture->RunWithAnalyzer(foreman_cif);
>+}
>+
>+TEST(FullStackTest, ForemanCifPlr5) {
>+  auto fixture = CreateVideoQualityTestFixture();
>+  ParamsWithLogging foreman_cif;
>+  foreman_cif.call.send_side_bwe = true;
>+  foreman_cif.video[0] = {true,   352,     288,   30,    30000,
>+                          500000, 2000000, false, "VP8", 1,
>+                          0,      0,       false, false, false, "foreman_cif"};
>   foreman_cif.analyzer = {"foreman_cif_delay_50_0_plr_5", 0.0, 0.0,
>                           kFullStackTestDurationSecs};
>   foreman_cif.pipe.loss_percent = 5;
>   foreman_cif.pipe.queue_delay_ms = 50;
>-  RunTest(foreman_cif);
>+  fixture->RunWithAnalyzer(foreman_cif);
> }
> 
>-TEST_F(FullStackTest, ForemanCifPlr5Ulpfec) {
>-  VideoQualityTest::Params foreman_cif;
>+TEST(FullStackTest, ForemanCifPlr5Ulpfec) {
>+  auto fixture = CreateVideoQualityTestFixture();
>+  ParamsWithLogging foreman_cif;
>   foreman_cif.call.send_side_bwe = true;
>   foreman_cif.video[0] = {true,   352,     288,   30,    30000,
>                           500000, 2000000, false, "VP8", 1,
>-                          0,      0,       true,  false, "foreman_cif"};
>+                          0,      0,       true,  false, false, "foreman_cif"};
>   foreman_cif.analyzer = {"foreman_cif_delay_50_0_plr_5_ulpfec", 0.0, 0.0,
>                           kFullStackTestDurationSecs};
>   foreman_cif.pipe.loss_percent = 5;
>   foreman_cif.pipe.queue_delay_ms = 50;
>-  RunTest(foreman_cif);
>+  fixture->RunWithAnalyzer(foreman_cif);
> }
> 
>-TEST_F(FullStackTest, ForemanCifPlr5Flexfec) {
>-  VideoQualityTest::Params foreman_cif;
>+TEST(FullStackTest, ForemanCifPlr5Flexfec) {
>+  auto fixture = CreateVideoQualityTestFixture();
>+  ParamsWithLogging foreman_cif;
>   foreman_cif.call.send_side_bwe = true;
>   foreman_cif.video[0] = {true,   352,     288,   30,    30000,
>                           500000, 2000000, false, "VP8", 1,
>-                          0,      0,       false, true,  "foreman_cif"};
>+                          0,      0,       false, true,  false, "foreman_cif"};
>   foreman_cif.analyzer = {"foreman_cif_delay_50_0_plr_5_flexfec", 0.0, 0.0,
>                           kFullStackTestDurationSecs};
>   foreman_cif.pipe.loss_percent = 5;
>   foreman_cif.pipe.queue_delay_ms = 50;
>-  RunTest(foreman_cif);
>+  fixture->RunWithAnalyzer(foreman_cif);
> }
> 
>-TEST_F(FullStackTest, ForemanCif500kbpsPlr3Flexfec) {
>-  VideoQualityTest::Params foreman_cif;
>+TEST(FullStackTest, ForemanCif500kbpsPlr3Flexfec) {
>+  auto fixture = CreateVideoQualityTestFixture();
>+  ParamsWithLogging foreman_cif;
>   foreman_cif.call.send_side_bwe = true;
>   foreman_cif.video[0] = {true,   352,     288,   30,    30000,
>                           500000, 2000000, false, "VP8", 1,
>-                          0,      0,       false, true,  "foreman_cif"};
>+                          0,      0,       false, true,  false, "foreman_cif"};
>   foreman_cif.analyzer = {"foreman_cif_500kbps_delay_50_0_plr_3_flexfec", 0.0,
>                           0.0, kFullStackTestDurationSecs};
>   foreman_cif.pipe.loss_percent = 3;
>   foreman_cif.pipe.link_capacity_kbps = 500;
>   foreman_cif.pipe.queue_delay_ms = 50;
>-  RunTest(foreman_cif);
>+  fixture->RunWithAnalyzer(foreman_cif);
> }
> 
>-TEST_F(FullStackTest, ForemanCif500kbpsPlr3Ulpfec) {
>-  VideoQualityTest::Params foreman_cif;
>+TEST(FullStackTest, ForemanCif500kbpsPlr3Ulpfec) {
>+  auto fixture = CreateVideoQualityTestFixture();
>+  ParamsWithLogging foreman_cif;
>   foreman_cif.call.send_side_bwe = true;
>   foreman_cif.video[0] = {true,   352,     288,   30,    30000,
>                           500000, 2000000, false, "VP8", 1,
>-                          0,      0,       true,  false, "foreman_cif"};
>+                          0,      0,       true,  false, false, "foreman_cif"};
>   foreman_cif.analyzer = {"foreman_cif_500kbps_delay_50_0_plr_3_ulpfec", 0.0,
>                           0.0, kFullStackTestDurationSecs};
>   foreman_cif.pipe.loss_percent = 3;
>   foreman_cif.pipe.link_capacity_kbps = 500;
>   foreman_cif.pipe.queue_delay_ms = 50;
>-  RunTest(foreman_cif);
>+  fixture->RunWithAnalyzer(foreman_cif);
> }
> 
> #if defined(WEBRTC_USE_H264)
>-TEST_F(FullStackTest, ForemanCifWithoutPacketlossH264) {
>+TEST(FullStackTest, ForemanCifWithoutPacketlossH264) {
>+  auto fixture = CreateVideoQualityTestFixture();
>   // TODO(pbos): Decide on psnr/ssim thresholds for foreman_cif.
>-  VideoQualityTest::Params foreman_cif;
>+  ParamsWithLogging foreman_cif;
>   foreman_cif.call.send_side_bwe = true;
>   foreman_cif.video[0] = {true,   352,    288,   30,     700000,
>                           700000, 700000, false, "H264", 1,
>-                          0,      0,      false, false,  "foreman_cif"};
>+                          0,      0,      false, false,  false, "foreman_cif"};
>   foreman_cif.analyzer = {"foreman_cif_net_delay_0_0_plr_0_H264", 0.0, 0.0,
>                           kFullStackTestDurationSecs};
>-  RunTest(foreman_cif);
>+  fixture->RunWithAnalyzer(foreman_cif);
> }
> 
>-TEST_F(FullStackTest, ForemanCif30kbpsWithoutPacketlossH264) {
>-  VideoQualityTest::Params foreman_cif;
>+TEST(FullStackTest, ForemanCif30kbpsWithoutPacketlossH264) {
>+  auto fixture = CreateVideoQualityTestFixture();
>+  ParamsWithLogging foreman_cif;
>   foreman_cif.call.send_side_bwe = true;
>   foreman_cif.video[0] = {true,  352,   288,   10,     30000,
>                           30000, 30000, false, "H264", 1,
>-                          0,     0,     false, false,  "foreman_cif"};
>+                          0,     0,     false, false,  false, "foreman_cif"};
>   foreman_cif.analyzer = {"foreman_cif_30kbps_net_delay_0_0_plr_0_H264", 0.0,
>                           0.0, kFullStackTestDurationSecs};
>-  RunTest(foreman_cif);
>+  fixture->RunWithAnalyzer(foreman_cif);
> }
> 
>-TEST_F(FullStackTest, ForemanCifPlr5H264) {
>-  VideoQualityTest::Params foreman_cif;
>+TEST(FullStackTest, ForemanCifPlr5H264) {
>+  auto fixture = CreateVideoQualityTestFixture();
>+  ParamsWithLogging foreman_cif;
>   foreman_cif.call.send_side_bwe = true;
>   foreman_cif.video[0] = {true,   352,     288,   30,     30000,
>                           500000, 2000000, false, "H264", 1,
>-                          0,      0,       false, false,  "foreman_cif"};
>+                          0,      0,       false, false,  false, "foreman_cif"};
>   foreman_cif.analyzer = {"foreman_cif_delay_50_0_plr_5_H264", 0.0, 0.0,
>                           kFullStackTestDurationSecs};
>   foreman_cif.pipe.loss_percent = 5;
>   foreman_cif.pipe.queue_delay_ms = 50;
>-  RunTest(foreman_cif);
>+  fixture->RunWithAnalyzer(foreman_cif);
> }
> 
>-TEST_F(FullStackTest, ForemanCifPlr5H264SpsPpsIdrIsKeyframe) {
>+TEST(FullStackTest, ForemanCifPlr5H264SpsPpsIdrIsKeyframe) {
>+  auto fixture = CreateVideoQualityTestFixture();
>   test::ScopedFieldTrials override_field_trials(
>-      "WebRTC-SpsPpsIdrIsH264Keyframe/Enabled/");
>+      AppendFieldTrials("WebRTC-SpsPpsIdrIsH264Keyframe/Enabled/"));
> 
>-  VideoQualityTest::Params foreman_cif;
>+  ParamsWithLogging foreman_cif;
>   foreman_cif.call.send_side_bwe = true;
>   foreman_cif.video[0] = {true,   352,     288,   30,     30000,
>                           500000, 2000000, false, "H264", 1,
>-                          0,      0,       false, false,  "foreman_cif"};
>+                          0,      0,       false, false,  false, "foreman_cif"};
>   foreman_cif.analyzer = {"foreman_cif_delay_50_0_plr_5_H264_sps_pps_idr", 0.0,
>                           0.0, kFullStackTestDurationSecs};
>   foreman_cif.pipe.loss_percent = 5;
>   foreman_cif.pipe.queue_delay_ms = 50;
>-  RunTest(foreman_cif);
>+  fixture->RunWithAnalyzer(foreman_cif);
> }
> 
> // Verify that this is worth the bot time, before enabling.
>-TEST_F(FullStackTest, ForemanCifPlr5H264Flexfec) {
>-  VideoQualityTest::Params foreman_cif;
>+TEST(FullStackTest, ForemanCifPlr5H264Flexfec) {
>+  auto fixture = CreateVideoQualityTestFixture();
>+  ParamsWithLogging foreman_cif;
>   foreman_cif.call.send_side_bwe = true;
>   foreman_cif.video[0] = {true,   352,     288,   30,     30000,
>                           500000, 2000000, false, "H264", 1,
>-                          0,      0,       false, true,   "foreman_cif"};
>+                          0,      0,       false, true,   false, "foreman_cif"};
>   foreman_cif.analyzer = {"foreman_cif_delay_50_0_plr_5_H264_flexfec", 0.0, 0.0,
>                           kFullStackTestDurationSecs};
>   foreman_cif.pipe.loss_percent = 5;
>   foreman_cif.pipe.queue_delay_ms = 50;
>-  RunTest(foreman_cif);
>+  fixture->RunWithAnalyzer(foreman_cif);
> }
> 
> // Ulpfec with H264 is an unsupported combination, so this test is only useful
> // for debugging. It is therefore disabled by default.
>-TEST_F(FullStackTest, DISABLED_ForemanCifPlr5H264Ulpfec) {
>-  VideoQualityTest::Params foreman_cif;
>+TEST(FullStackTest, DISABLED_ForemanCifPlr5H264Ulpfec) {
>+  auto fixture = CreateVideoQualityTestFixture();
>+  ParamsWithLogging foreman_cif;
>   foreman_cif.call.send_side_bwe = true;
>   foreman_cif.video[0] = {true,   352,     288,   30,     30000,
>                           500000, 2000000, false, "H264", 1,
>-                          0,      0,       true,  false,  "foreman_cif"};
>+                          0,      0,       true,  false,  false, "foreman_cif"};
>   foreman_cif.analyzer = {"foreman_cif_delay_50_0_plr_5_H264_ulpfec", 0.0, 0.0,
>                           kFullStackTestDurationSecs};
>   foreman_cif.pipe.loss_percent = 5;
>   foreman_cif.pipe.queue_delay_ms = 50;
>-  RunTest(foreman_cif);
>+  fixture->RunWithAnalyzer(foreman_cif);
> }
> #endif  // defined(WEBRTC_USE_H264)
> 
>-TEST_F(FullStackTest, ForemanCif500kbps) {
>-  VideoQualityTest::Params foreman_cif;
>+TEST(FullStackTest, ForemanCif500kbps) {
>+  auto fixture = CreateVideoQualityTestFixture();
>+  ParamsWithLogging foreman_cif;
>   foreman_cif.call.send_side_bwe = true;
>   foreman_cif.video[0] = {true,   352,     288,   30,    30000,
>                           500000, 2000000, false, "VP8", 1,
>-                          0,      0,       false, false, "foreman_cif"};
>+                          0,      0,       false, false, false, "foreman_cif"};
>   foreman_cif.analyzer = {"foreman_cif_500kbps", 0.0, 0.0,
>                           kFullStackTestDurationSecs};
>   foreman_cif.pipe.queue_length_packets = 0;
>   foreman_cif.pipe.queue_delay_ms = 0;
>   foreman_cif.pipe.link_capacity_kbps = 500;
>-  RunTest(foreman_cif);
>+  fixture->RunWithAnalyzer(foreman_cif);
> }
> 
>-TEST_F(FullStackTest, ForemanCif500kbpsLimitedQueue) {
>-  VideoQualityTest::Params foreman_cif;
>+TEST(FullStackTest, ForemanCif500kbpsLimitedQueue) {
>+  auto fixture = CreateVideoQualityTestFixture();
>+  ParamsWithLogging foreman_cif;
>   foreman_cif.call.send_side_bwe = true;
>   foreman_cif.video[0] = {true,   352,     288,   30,    30000,
>                           500000, 2000000, false, "VP8", 1,
>-                          0,      0,       false, false, "foreman_cif"};
>+                          0,      0,       false, false, false, "foreman_cif"};
>   foreman_cif.analyzer = {"foreman_cif_500kbps_32pkts_queue", 0.0, 0.0,
>                           kFullStackTestDurationSecs};
>   foreman_cif.pipe.queue_length_packets = 32;
>   foreman_cif.pipe.queue_delay_ms = 0;
>   foreman_cif.pipe.link_capacity_kbps = 500;
>-  RunTest(foreman_cif);
>+  fixture->RunWithAnalyzer(foreman_cif);
> }
> 
>-TEST_F(FullStackTest, ForemanCif500kbps100ms) {
>-  VideoQualityTest::Params foreman_cif;
>+TEST(FullStackTest, ForemanCif500kbps100ms) {
>+  auto fixture = CreateVideoQualityTestFixture();
>+  ParamsWithLogging foreman_cif;
>   foreman_cif.call.send_side_bwe = true;
>   foreman_cif.video[0] = {true,   352,     288,   30,    30000,
>                           500000, 2000000, false, "VP8", 1,
>-                          0,      0,       false, false, "foreman_cif"};
>+                          0,      0,       false, false, false, "foreman_cif"};
>   foreman_cif.analyzer = {"foreman_cif_500kbps_100ms", 0.0, 0.0,
>                           kFullStackTestDurationSecs};
>   foreman_cif.pipe.queue_length_packets = 0;
>   foreman_cif.pipe.queue_delay_ms = 100;
>   foreman_cif.pipe.link_capacity_kbps = 500;
>-  RunTest(foreman_cif);
>+  fixture->RunWithAnalyzer(foreman_cif);
> }
> 
>-TEST_F(FullStackTest, ForemanCif500kbps100msLimitedQueue) {
>-  VideoQualityTest::Params foreman_cif;
>+TEST(FullStackTest, ForemanCif500kbps100msLimitedQueue) {
>+  auto fixture = CreateVideoQualityTestFixture();
>+  ParamsWithLogging foreman_cif;
>   foreman_cif.call.send_side_bwe = true;
>   foreman_cif.video[0] = {true,   352,     288,   30,    30000,
>                           500000, 2000000, false, "VP8", 1,
>-                          0,      0,       false, false, "foreman_cif"};
>+                          0,      0,       false, false, false, "foreman_cif"};
>   foreman_cif.analyzer = {"foreman_cif_500kbps_100ms_32pkts_queue", 0.0, 0.0,
>                           kFullStackTestDurationSecs};
>   foreman_cif.pipe.queue_length_packets = 32;
>   foreman_cif.pipe.queue_delay_ms = 100;
>   foreman_cif.pipe.link_capacity_kbps = 500;
>-  RunTest(foreman_cif);
>+  fixture->RunWithAnalyzer(foreman_cif);
> }
> 
>-TEST_F(FullStackTest, ForemanCif500kbps100msLimitedQueueRecvBwe) {
>-  VideoQualityTest::Params foreman_cif;
>+TEST(FullStackTest, ForemanCif500kbps100msLimitedQueueRecvBwe) {
>+  auto fixture = CreateVideoQualityTestFixture();
>+  ParamsWithLogging foreman_cif;
>   foreman_cif.call.send_side_bwe = false;
>   foreman_cif.video[0] = {true,   352,     288,   30,    30000,
>                           500000, 2000000, false, "VP8", 1,
>-                          0,      0,       false, false, "foreman_cif"};
>+                          0,      0,       false, false, false, "foreman_cif"};
>   foreman_cif.analyzer = {"foreman_cif_500kbps_100ms_32pkts_queue_recv_bwe",
>                           0.0, 0.0, kFullStackTestDurationSecs};
>   foreman_cif.pipe.queue_length_packets = 32;
>   foreman_cif.pipe.queue_delay_ms = 100;
>   foreman_cif.pipe.link_capacity_kbps = 500;
>-  RunTest(foreman_cif);
>+  fixture->RunWithAnalyzer(foreman_cif);
> }
> 
>-TEST_F(FullStackTest, ForemanCif1000kbps100msLimitedQueue) {
>-  VideoQualityTest::Params foreman_cif;
>+TEST(FullStackTest, ForemanCif1000kbps100msLimitedQueue) {
>+  auto fixture = CreateVideoQualityTestFixture();
>+  ParamsWithLogging foreman_cif;
>   foreman_cif.call.send_side_bwe = true;
>   foreman_cif.video[0] = {true,    352,     288,   30,    30000,
>                           2000000, 2000000, false, "VP8", 1,
>-                          0,       0,       false, false, "foreman_cif"};
>+                          0,       0,       false, false, false, "foreman_cif"};
>   foreman_cif.analyzer = {"foreman_cif_1000kbps_100ms_32pkts_queue", 0.0, 0.0,
>                           kFullStackTestDurationSecs};
>   foreman_cif.pipe.queue_length_packets = 32;
>   foreman_cif.pipe.queue_delay_ms = 100;
>   foreman_cif.pipe.link_capacity_kbps = 1000;
>-  RunTest(foreman_cif);
>+  fixture->RunWithAnalyzer(foreman_cif);
> }
> 
> // TODO(sprang): Remove this if we have the similar ModerateLimits below?
>-TEST_F(FullStackTest, ConferenceMotionHd2000kbps100msLimitedQueue) {
>-  VideoQualityTest::Params conf_motion_hd;
>+TEST(FullStackTest, ConferenceMotionHd2000kbps100msLimitedQueue) {
>+  auto fixture = CreateVideoQualityTestFixture();
>+  ParamsWithLogging conf_motion_hd;
>   conf_motion_hd.call.send_side_bwe = true;
>   conf_motion_hd.video[0] = {
>       true,    1280,    720,   50,    30000,
>       3000000, 3000000, false, "VP8", 1,
>-      0,       0,       false, false, "ConferenceMotion_1280_720_50"};
>+      0,       0,       false, false, false, "ConferenceMotion_1280_720_50"};
>   conf_motion_hd.analyzer = {"conference_motion_hd_2000kbps_100ms_32pkts_queue",
>                              0.0, 0.0, kFullStackTestDurationSecs};
>   conf_motion_hd.pipe.queue_length_packets = 32;
>   conf_motion_hd.pipe.queue_delay_ms = 100;
>   conf_motion_hd.pipe.link_capacity_kbps = 2000;
>-  RunTest(conf_motion_hd);
>+  fixture->RunWithAnalyzer(conf_motion_hd);
> }
> 
>-TEST_F(FullStackTest, ConferenceMotionHd1TLModerateLimits) {
>-  VideoQualityTest::Params conf_motion_hd;
>+TEST(FullStackTest, ConferenceMotionHd1TLModerateLimits) {
>+  auto fixture = CreateVideoQualityTestFixture();
>+  ParamsWithLogging conf_motion_hd;
>   conf_motion_hd.call.send_side_bwe = true;
>   conf_motion_hd.video[0] = {
>       true,    1280,    720,   50,    30000,
>       3000000, 3000000, false, "VP8", 1,
>-      -1,      0,       false, false, "ConferenceMotion_1280_720_50"};
>+      -1,      0,       false, false, false, "ConferenceMotion_1280_720_50"};
>   conf_motion_hd.analyzer = {"conference_motion_hd_1tl_moderate_limits", 0.0,
>                              0.0, kFullStackTestDurationSecs};
>   conf_motion_hd.pipe.queue_length_packets = 50;
>   conf_motion_hd.pipe.loss_percent = 3;
>   conf_motion_hd.pipe.queue_delay_ms = 100;
>   conf_motion_hd.pipe.link_capacity_kbps = 2000;
>-  RunTest(conf_motion_hd);
>+  fixture->RunWithAnalyzer(conf_motion_hd);
> }
> 
>-TEST_F(FullStackTest, ConferenceMotionHd2TLModerateLimits) {
>-  VideoQualityTest::Params conf_motion_hd;
>+TEST(FullStackTest, ConferenceMotionHd2TLModerateLimits) {
>+  auto fixture = CreateVideoQualityTestFixture();
>+  ParamsWithLogging conf_motion_hd;
>   conf_motion_hd.call.send_side_bwe = true;
>   conf_motion_hd.video[0] = {
>       true,    1280,    720,   50,    30000,
>       3000000, 3000000, false, "VP8", 2,
>-      -1,      0,       false, false, "ConferenceMotion_1280_720_50"};
>+      -1,      0,       false, false, false, "ConferenceMotion_1280_720_50"};
>   conf_motion_hd.analyzer = {"conference_motion_hd_2tl_moderate_limits", 0.0,
>                              0.0, kFullStackTestDurationSecs};
>   conf_motion_hd.pipe.queue_length_packets = 50;
>   conf_motion_hd.pipe.loss_percent = 3;
>   conf_motion_hd.pipe.queue_delay_ms = 100;
>   conf_motion_hd.pipe.link_capacity_kbps = 2000;
>-  RunTest(conf_motion_hd);
>+  fixture->RunWithAnalyzer(conf_motion_hd);
> }
> 
>-TEST_F(FullStackTest, ConferenceMotionHd3TLModerateLimits) {
>-  VideoQualityTest::Params conf_motion_hd;
>+TEST(FullStackTest, ConferenceMotionHd3TLModerateLimits) {
>+  auto fixture = CreateVideoQualityTestFixture();
>+  ParamsWithLogging conf_motion_hd;
>   conf_motion_hd.call.send_side_bwe = true;
>   conf_motion_hd.video[0] = {
>       true,    1280,    720,   50,    30000,
>       3000000, 3000000, false, "VP8", 3,
>-      -1,      0,       false, false, "ConferenceMotion_1280_720_50"};
>+      -1,      0,       false, false, false, "ConferenceMotion_1280_720_50"};
>   conf_motion_hd.analyzer = {"conference_motion_hd_3tl_moderate_limits", 0.0,
>                              0.0, kFullStackTestDurationSecs};
>   conf_motion_hd.pipe.queue_length_packets = 50;
>   conf_motion_hd.pipe.loss_percent = 3;
>   conf_motion_hd.pipe.queue_delay_ms = 100;
>   conf_motion_hd.pipe.link_capacity_kbps = 2000;
>-  RunTest(conf_motion_hd);
>+  fixture->RunWithAnalyzer(conf_motion_hd);
> }
> 
>-TEST_F(FullStackTest, ConferenceMotionHd4TLModerateLimits) {
>-  VideoQualityTest::Params conf_motion_hd;
>+TEST(FullStackTest, ConferenceMotionHd4TLModerateLimits) {
>+  auto fixture = CreateVideoQualityTestFixture();
>+  ParamsWithLogging conf_motion_hd;
>   conf_motion_hd.call.send_side_bwe = true;
>   conf_motion_hd.video[0] = {
>       true,    1280,    720,   50,    30000,
>       3000000, 3000000, false, "VP8", 4,
>-      -1,      0,       false, false, "ConferenceMotion_1280_720_50"};
>+      -1,      0,       false, false, false, "ConferenceMotion_1280_720_50"};
>   conf_motion_hd.analyzer = {"conference_motion_hd_4tl_moderate_limits", 0.0,
>                              0.0, kFullStackTestDurationSecs};
>   conf_motion_hd.pipe.queue_length_packets = 50;
>   conf_motion_hd.pipe.loss_percent = 3;
>   conf_motion_hd.pipe.queue_delay_ms = 100;
>   conf_motion_hd.pipe.link_capacity_kbps = 2000;
>-  RunTest(conf_motion_hd);
>+  fixture->RunWithAnalyzer(conf_motion_hd);
> }
> 
>-TEST_F(FullStackTest, ConferenceMotionHd3TLModerateLimitsAltTLPattern) {
>-  test::ScopedFieldTrials field_trial("WebRTC-UseShortVP8TL3Pattern/Enabled/");
>-  VideoQualityTest::Params conf_motion_hd;
>+TEST(FullStackTest, ConferenceMotionHd3TLModerateLimitsAltTLPattern) {
>+  auto fixture = CreateVideoQualityTestFixture();
>+  test::ScopedFieldTrials field_trial(
>+      AppendFieldTrials("WebRTC-UseShortVP8TL3Pattern/Enabled/"));
>+  ParamsWithLogging conf_motion_hd;
>   conf_motion_hd.call.send_side_bwe = true;
>   conf_motion_hd.video[0] = {
>       true,    1280,    720,   50,    30000,
>       3000000, 3000000, false, "VP8", 3,
>-      -1,      0,       false, false, "ConferenceMotion_1280_720_50"};
>+      -1,      0,       false, false, false, "ConferenceMotion_1280_720_50"};
>   conf_motion_hd.analyzer = {"conference_motion_hd_3tl_alt_moderate_limits",
>                              0.0, 0.0, kFullStackTestDurationSecs};
>   conf_motion_hd.pipe.queue_length_packets = 50;
>   conf_motion_hd.pipe.loss_percent = 3;
>   conf_motion_hd.pipe.queue_delay_ms = 100;
>   conf_motion_hd.pipe.link_capacity_kbps = 2000;
>-  RunTest(conf_motion_hd);
>+  fixture->RunWithAnalyzer(conf_motion_hd);
> }
> 
> #if !defined(RTC_DISABLE_VP9)
>-TEST_F(FullStackTest, ConferenceMotionHd2000kbps100msLimitedQueueVP9) {
>-  VideoQualityTest::Params conf_motion_hd;
>+TEST(FullStackTest, ConferenceMotionHd2000kbps100msLimitedQueueVP9) {
>+  auto fixture = CreateVideoQualityTestFixture();
>+  ParamsWithLogging conf_motion_hd;
>   conf_motion_hd.call.send_side_bwe = true;
>   conf_motion_hd.video[0] = {
>       true,    1280,    720,   50,    30000,
>       3000000, 3000000, false, "VP9", 1,
>-      0,       0,       false, false, "ConferenceMotion_1280_720_50"};
>+      0,       0,       false, false, false, "ConferenceMotion_1280_720_50"};
>   conf_motion_hd.analyzer = {
>       "conference_motion_hd_2000kbps_100ms_32pkts_queue_vp9", 0.0, 0.0,
>       kFullStackTestDurationSecs};
>   conf_motion_hd.pipe.queue_length_packets = 32;
>   conf_motion_hd.pipe.queue_delay_ms = 100;
>   conf_motion_hd.pipe.link_capacity_kbps = 2000;
>-  RunTest(conf_motion_hd);
>+  fixture->RunWithAnalyzer(conf_motion_hd);
> }
> #endif
> 
>-TEST_F(FullStackTest, ScreenshareSlidesVP8_2TL) {
>-  VideoQualityTest::Params screenshare;
>+TEST(FullStackTest, ScreenshareSlidesVP8_2TL) {
>+  auto fixture = CreateVideoQualityTestFixture();
>+  ParamsWithLogging screenshare;
>   screenshare.call.send_side_bwe = true;
>-  screenshare.video[0] = {true,   1850,    1110,  5,     50000,
>-                          200000, 2000000, false, "VP8", 2,
>-                          1,      400000,  false, false, ""};
>+  screenshare.video[0] = {true,    1850,  1110,  5, 50000, 200000,
>+                          1000000, false, "VP8", 2, 1,     400000,
>+                          false,   false, false, ""};
>   screenshare.screenshare[0] = {true, false, 10};
>   screenshare.analyzer = {"screenshare_slides", 0.0, 0.0,
>                           kFullStackTestDurationSecs};
>-  RunTest(screenshare);
>+  fixture->RunWithAnalyzer(screenshare);
> }
> 
>-TEST_F(FullStackTest, ScreenshareSlidesVP8_3TL_Simulcast) {
>-  test::ScopedFieldTrials field_trial(kScreenshareSimulcastExperiment);
>-  VideoQualityTest::Params screenshare;
>+TEST(FullStackTest, ScreenshareSlidesVP8_3TL_Simulcast) {
>+  auto fixture = CreateVideoQualityTestFixture();
>+  ParamsWithLogging screenshare;
>   screenshare.call.send_side_bwe = true;
>   screenshare.screenshare[0] = {true, false, 10};
>   screenshare.video[0] = {true,    1850,    1110,  5,     800000,
>                           2500000, 2500000, false, "VP8", 3,
>-                          2,       400000,  false, false, ""};
>+                          2,       400000,  false, false, false, ""};
>   screenshare.analyzer = {"screenshare_slides_simulcast", 0.0, 0.0,
>                           kFullStackTestDurationSecs};
>-  VideoQualityTest::Params screenshare_params_high;
>-  screenshare_params_high.video[0] = {true,    1850,    1110,  5,     800000,
>-                                      2500000, 2500000, false, "VP8", 3,
>-                                      0,       400000,  false, false, ""};
>+  ParamsWithLogging screenshare_params_high;
>+  screenshare_params_high.video[0] = {true,    1850,  1110,  5, 400000, 1000000,
>+                                      1000000, false, "VP8", 3, 0,      400000,
>+                                      false,   false, false, ""};
>   VideoQualityTest::Params screenshare_params_low;
>-  screenshare_params_low.video[0] = {true,   1850,    1110,  5,     50000,
>-                                     200000, 2000000, false, "VP8", 2,
>-                                     0,      400000,  false, false, ""};
>+  screenshare_params_low.video[0] = {true,    1850,  1110,  5, 50000, 200000,
>+                                     1000000, false, "VP8", 2, 0,     400000,
>+                                     false,   false, false, ""};
> 
>   std::vector<VideoStream> streams = {
>-      DefaultVideoStream(screenshare_params_low, 0),
>-      DefaultVideoStream(screenshare_params_high, 0)};
>-  screenshare.ss[0] = {streams, 1, 1, 0, std::vector<SpatialLayer>(), false};
>-  RunTest(screenshare);
>+      VideoQualityTest::DefaultVideoStream(screenshare_params_low, 0),
>+      VideoQualityTest::DefaultVideoStream(screenshare_params_high, 0)};
>+  screenshare.ss[0] = {
>+      streams, 1, 1, 0, InterLayerPredMode::kOn, std::vector<SpatialLayer>(),
>+      false};
>+  fixture->RunWithAnalyzer(screenshare);
> }
> 
>-TEST_F(FullStackTest, ScreenshareSlidesVP8_2TL_Scroll) {
>-  VideoQualityTest::Params config;
>+TEST(FullStackTest, ScreenshareSlidesVP8_2TL_Scroll) {
>+  auto fixture = CreateVideoQualityTestFixture();
>+  ParamsWithLogging config;
>   config.call.send_side_bwe = true;
>-  config.video[0] = {true,   1850,    1110 / 2, 5,     50000,
>-                     200000, 2000000, false,    "VP8", 2,
>-                     1,      400000,  false,    false, ""};
>+  config.video[0] = {true,    1850,  1110 / 2, 5, 50000, 200000,
>+                     1000000, false, "VP8",    2, 1,     400000,
>+                     false,   false, false,    ""};
>   config.screenshare[0] = {true, false, 10, 2};
>   config.analyzer = {"screenshare_slides_scrolling", 0.0, 0.0,
>                      kFullStackTestDurationSecs};
>-  RunTest(config);
>+  fixture->RunWithAnalyzer(config);
> }
> 
>-TEST_F(FullStackTest, ScreenshareSlidesVP8_2TL_LossyNet) {
>-  VideoQualityTest::Params screenshare;
>+TEST(FullStackTest, ScreenshareSlidesVP8_2TL_LossyNet) {
>+  auto fixture = CreateVideoQualityTestFixture();
>+  ParamsWithLogging screenshare;
>   screenshare.call.send_side_bwe = true;
>-  screenshare.video[0] = {true,   1850,    1110,  5,     50000,
>-                          200000, 2000000, false, "VP8", 2,
>-                          1,      400000,  false, false, ""};
>+  screenshare.video[0] = {true,    1850,  1110,  5, 50000, 200000,
>+                          1000000, false, "VP8", 2, 1,     400000,
>+                          false,   false, false, ""};
>   screenshare.screenshare[0] = {true, false, 10};
>   screenshare.analyzer = {"screenshare_slides_lossy_net", 0.0, 0.0,
>                           kFullStackTestDurationSecs};
>   screenshare.pipe.loss_percent = 5;
>   screenshare.pipe.queue_delay_ms = 200;
>   screenshare.pipe.link_capacity_kbps = 500;
>-  RunTest(screenshare);
>+  fixture->RunWithAnalyzer(screenshare);
> }
> 
>-TEST_F(FullStackTest, ScreenshareSlidesVP8_2TL_VeryLossyNet) {
>-  VideoQualityTest::Params screenshare;
>+TEST(FullStackTest, ScreenshareSlidesVP8_2TL_VeryLossyNet) {
>+  auto fixture = CreateVideoQualityTestFixture();
>+  ParamsWithLogging screenshare;
>   screenshare.call.send_side_bwe = true;
>-  screenshare.video[0] = {true,   1850,    1110,  5,     50000,
>-                          200000, 2000000, false, "VP8", 2,
>-                          1,      400000,  false, false, ""};
>+  screenshare.video[0] = {true,    1850,  1110,  5, 50000, 200000,
>+                          1000000, false, "VP8", 2, 1,     400000,
>+                          false,   false, false, ""};
>   screenshare.screenshare[0] = {true, false, 10};
>   screenshare.analyzer = {"screenshare_slides_very_lossy", 0.0, 0.0,
>                           kFullStackTestDurationSecs};
>   screenshare.pipe.loss_percent = 10;
>   screenshare.pipe.queue_delay_ms = 200;
>   screenshare.pipe.link_capacity_kbps = 500;
>-  RunTest(screenshare);
>+  fixture->RunWithAnalyzer(screenshare);
> }
> 
>-TEST_F(FullStackTest, ScreenshareSlidesVP8_2TL_LossyNetRestrictedQueue) {
>-  VideoQualityTest::Params screenshare;
>+TEST(FullStackTest, ScreenshareSlidesVP8_2TL_LossyNetRestrictedQueue) {
>+  auto fixture = CreateVideoQualityTestFixture();
>+  ParamsWithLogging screenshare;
>   screenshare.call.send_side_bwe = true;
>-  screenshare.video[0] = {true,   1850,    1110,  5,     50000,
>-                          200000, 2000000, false, "VP8", 2,
>-                          1,      400000,  false, false, ""};
>+  screenshare.video[0] = {true,    1850,  1110,  5, 50000, 200000,
>+                          1000000, false, "VP8", 2, 1,     400000,
>+                          false,   false, false, ""};
>   screenshare.screenshare[0] = {true, false, 10};
>   screenshare.analyzer = {"screenshare_slides_lossy_limited", 0.0, 0.0,
>                           kFullStackTestDurationSecs};
>@@ -551,15 +697,16 @@ TEST_F(FullStackTest, ScreenshareSlidesVP8_2TL_LossyNetRestrictedQueue) {
>   screenshare.pipe.link_capacity_kbps = 200;
>   screenshare.pipe.queue_length_packets = 30;
> 
>-  RunTest(screenshare);
>+  fixture->RunWithAnalyzer(screenshare);
> }
> 
>-TEST_F(FullStackTest, ScreenshareSlidesVP8_2TL_ModeratelyRestricted) {
>-  VideoQualityTest::Params screenshare;
>+TEST(FullStackTest, ScreenshareSlidesVP8_2TL_ModeratelyRestricted) {
>+  auto fixture = CreateVideoQualityTestFixture();
>+  ParamsWithLogging screenshare;
>   screenshare.call.send_side_bwe = true;
>-  screenshare.video[0] = {true,   1850,    1110,  5,     50000,
>-                          200000, 2000000, false, "VP8", 2,
>-                          1,      400000,  false, false, ""};
>+  screenshare.video[0] = {true,    1850,  1110,  5, 50000, 200000,
>+                          1000000, false, "VP8", 2, 1,     400000,
>+                          false,   false, false, ""};
>   screenshare.screenshare[0] = {true, false, 10};
>   screenshare.analyzer = {"screenshare_slides_moderately_restricted", 0.0, 0.0,
>                           kFullStackTestDurationSecs};
>@@ -567,269 +714,279 @@ TEST_F(FullStackTest, ScreenshareSlidesVP8_2TL_ModeratelyRestricted) {
>   screenshare.pipe.link_capacity_kbps = 1200;
>   screenshare.pipe.queue_length_packets = 30;
> 
>-  RunTest(screenshare);
>+  fixture->RunWithAnalyzer(screenshare);
> }
> 
>-// TODO(sprang): Retire these tests once experiment is removed.
>-TEST_F(FullStackTest, ScreenshareSlidesVP8_2TL_LossyNetRestrictedQueue_ALR) {
>-  test::ScopedFieldTrials field_trial(kAlrProbingExperiment);
>-  VideoQualityTest::Params screenshare;
>-  screenshare.call.send_side_bwe = true;
>-  screenshare.video[0] = {true,   1850,    1110,  5,     50000,
>-                          200000, 2000000, false, "VP8", 2,
>-                          1,      400000,  false, false, ""};
>-  screenshare.screenshare[0] = {true, false, 10};
>-  screenshare.analyzer = {"screenshare_slides_lossy_limited_ALR", 0.0, 0.0,
>-                          kFullStackTestDurationSecs};
>-  screenshare.pipe.loss_percent = 5;
>-  screenshare.pipe.link_capacity_kbps = 200;
>-  screenshare.pipe.queue_length_packets = 30;
>-
>-  RunTest(screenshare);
>-}
>-
>-TEST_F(FullStackTest, ScreenshareSlidesVP8_2TL_ALR) {
>-  test::ScopedFieldTrials field_trial(kAlrProbingExperiment);
>-  VideoQualityTest::Params screenshare;
>-  screenshare.call.send_side_bwe = true;
>-  screenshare.video[0] = {true,   1850,    1110,  5,     50000,
>-                          200000, 2000000, false, "VP8", 2,
>-                          1,      400000,  false, false, ""};
>-  screenshare.screenshare[0] = {true, false, 10};
>-  screenshare.analyzer = {"screenshare_slides_ALR", 0.0, 0.0,
>-                          kFullStackTestDurationSecs};
>-  RunTest(screenshare);
>-}
>-
>-TEST_F(FullStackTest, ScreenshareSlidesVP8_2TL_ModeratelyRestricted_ALR) {
>-  test::ScopedFieldTrials field_trial(kAlrProbingExperiment);
>-  VideoQualityTest::Params screenshare;
>-  screenshare.call.send_side_bwe = true;
>-  screenshare.video[0] = {true,   1850,    1110,  5,     50000,
>-                          200000, 2000000, false, "VP8", 2,
>-                          1,      400000,  false, false, ""};
>-  screenshare.screenshare[0] = {true, false, 10};
>-  screenshare.analyzer = {"screenshare_slides_moderately_restricted_ALR", 0.0,
>-                          0.0, kFullStackTestDurationSecs};
>-  screenshare.pipe.loss_percent = 1;
>-  screenshare.pipe.link_capacity_kbps = 1200;
>-  screenshare.pipe.queue_length_packets = 30;
>-
>-  RunTest(screenshare);
>-}
>-
>-TEST_F(FullStackTest, ScreenshareSlidesVP8_3TL_Simulcast_ALR) {
>-  test::ScopedFieldTrials field_trial(kScreenshareSimulcastExperiment +
>-                                      kAlrProbingExperiment);
>-  VideoQualityTest::Params screenshare;
>-  screenshare.call.send_side_bwe = true;
>-  screenshare.screenshare[0] = {true, false, 10};
>-  screenshare.video[0] = {true,    1850,    1110,  5,     800000,
>-                          2500000, 2500000, false, "VP8", 3,
>-                          2,       400000,  false, false, ""};
>-  screenshare.analyzer = {"screenshare_slides_simulcast_alr", 0.0, 0.0,
>-                          kFullStackTestDurationSecs};
>-  VideoQualityTest::Params screenshare_params_high;
>-  screenshare_params_high.video[0] = {true,    1850,    1110,  5,     800000,
>-                                      2500000, 2500000, false, "VP8", 3,
>-                                      0,       400000,  false, false, ""};
>-  VideoQualityTest::Params screenshare_params_low;
>-  screenshare_params_low.video[0] = {true,   1850,    1110,  5,     50000,
>-                                     200000, 2000000, false, "VP8", 2,
>-                                     0,      400000,  false, false, ""};
>-
>-  std::vector<VideoStream> streams = {
>-      DefaultVideoStream(screenshare_params_low, 0),
>-      DefaultVideoStream(screenshare_params_high, 0)};
>-  screenshare.ss[0] = {streams, 1, 1, 0, std::vector<SpatialLayer>(), false};
>-  RunTest(screenshare);
>-}
>-
>-const VideoQualityTest::Params::Video kSvcVp9Video = {
>+const ParamsWithLogging::Video kSvcVp9Video = {
>     true,    1280,    720,   30,    800000,
>     2500000, 2500000, false, "VP9", 3,
>-    2,       400000,  false, false, "ConferenceMotion_1280_720_50"};
>+    2,       400000,  false, false, false, "ConferenceMotion_1280_720_50"};
> 
>-const VideoQualityTest::Params::Video kSimulcastVp8VideoHigh = {
>+const ParamsWithLogging::Video kSimulcastVp8VideoHigh = {
>     true,    1280,    720,   30,    800000,
>     2500000, 2500000, false, "VP8", 3,
>-    2,       400000,  false, false, "ConferenceMotion_1280_720_50"};
>+    2,       400000,  false, false, false, "ConferenceMotion_1280_720_50"};
> 
>-const VideoQualityTest::Params::Video kSimulcastVp8VideoMedium = {
>+const ParamsWithLogging::Video kSimulcastVp8VideoMedium = {
>     true,   640,    360,   30,    150000,
>     500000, 700000, false, "VP8", 3,
>-    2,      400000, false, false, "ConferenceMotion_1280_720_50"};
>+    2,      400000, false, false, false, "ConferenceMotion_1280_720_50"};
> 
>-const VideoQualityTest::Params::Video kSimulcastVp8VideoLow = {
>+const ParamsWithLogging::Video kSimulcastVp8VideoLow = {
>     true,   320,    180,   30,    30000,
>     150000, 200000, false, "VP8", 3,
>-    2,      400000, false, false, "ConferenceMotion_1280_720_50"};
>+    2,      400000, false, false, false, "ConferenceMotion_1280_720_50"};
> 
> #if !defined(RTC_DISABLE_VP9)
>-TEST_F(FullStackTest, ScreenshareSlidesVP9_2SL) {
>-  VideoQualityTest::Params screenshare;
>+TEST(FullStackTest, ScreenshareSlidesVP9_2SL) {
>+  auto fixture = CreateVideoQualityTestFixture();
>+  ParamsWithLogging screenshare;
>   screenshare.call.send_side_bwe = true;
>   screenshare.video[0] = {true,   1850,    1110,  5,     50000,
>                           200000, 2000000, false, "VP9", 1,
>-                          0,      400000,  false, false, ""};
>+                          0,      400000,  false, false, false, ""};
>   screenshare.screenshare[0] = {true, false, 10};
>   screenshare.analyzer = {"screenshare_slides_vp9_2sl", 0.0, 0.0,
>                           kFullStackTestDurationSecs};
>-  screenshare.ss[0] = {std::vector<VideoStream>(),  0,    2, 1,
>-                       std::vector<SpatialLayer>(), false};
>-  RunTest(screenshare);
>+  screenshare.ss[0] = {
>+      std::vector<VideoStream>(),  0,    2, 1, InterLayerPredMode::kOn,
>+      std::vector<SpatialLayer>(), false};
>+  fixture->RunWithAnalyzer(screenshare);
> }
> 
>-TEST_F(FullStackTest, VP9SVC_3SL_High) {
>-  VideoQualityTest::Params simulcast;
>+TEST(FullStackTest, VP9SVC_3SL_High) {
>+  auto fixture = CreateVideoQualityTestFixture();
>+  ParamsWithLogging simulcast;
>   simulcast.call.send_side_bwe = true;
>   simulcast.video[0] = kSvcVp9Video;
>   simulcast.analyzer = {"vp9svc_3sl_high", 0.0, 0.0,
>                         kFullStackTestDurationSecs};
>-  simulcast.ss[0] = {std::vector<VideoStream>(),  0,    3, 2,
>-                     std::vector<SpatialLayer>(), false};
>-  RunTest(simulcast);
>+
>+  simulcast.ss[0] = {
>+      std::vector<VideoStream>(),  0,    3, 2, InterLayerPredMode::kOn,
>+      std::vector<SpatialLayer>(), false};
>+  fixture->RunWithAnalyzer(simulcast);
> }
> 
>-TEST_F(FullStackTest, VP9SVC_3SL_Medium) {
>-  VideoQualityTest::Params simulcast;
>+TEST(FullStackTest, VP9SVC_3SL_Medium) {
>+  auto fixture = CreateVideoQualityTestFixture();
>+  ParamsWithLogging simulcast;
>   simulcast.call.send_side_bwe = true;
>   simulcast.video[0] = kSvcVp9Video;
>   simulcast.analyzer = {"vp9svc_3sl_medium", 0.0, 0.0,
>                         kFullStackTestDurationSecs};
>-  simulcast.ss[0] = {std::vector<VideoStream>(),  0,    3, 1,
>-                     std::vector<SpatialLayer>(), false};
>-  RunTest(simulcast);
>+  simulcast.ss[0] = {
>+      std::vector<VideoStream>(),  0,    3, 1, InterLayerPredMode::kOn,
>+      std::vector<SpatialLayer>(), false};
>+  fixture->RunWithAnalyzer(simulcast);
> }
> 
>-TEST_F(FullStackTest, VP9SVC_3SL_Low) {
>-  VideoQualityTest::Params simulcast;
>+TEST(FullStackTest, VP9SVC_3SL_Low) {
>+  auto fixture = CreateVideoQualityTestFixture();
>+  ParamsWithLogging simulcast;
>   simulcast.call.send_side_bwe = true;
>   simulcast.video[0] = kSvcVp9Video;
>   simulcast.analyzer = {"vp9svc_3sl_low", 0.0, 0.0, kFullStackTestDurationSecs};
>-  simulcast.ss[0] = {std::vector<VideoStream>(),  0,    3, 0,
>-                     std::vector<SpatialLayer>(), false};
>-  RunTest(simulcast);
>+  simulcast.ss[0] = {
>+      std::vector<VideoStream>(),  0,    3, 0, InterLayerPredMode::kOn,
>+      std::vector<SpatialLayer>(), false};
>+  fixture->RunWithAnalyzer(simulcast);
>+}
>+
>+// bugs.webrtc.org/9506
>+#if !defined(WEBRTC_MAC)
>+
>+TEST(FullStackTest, VP9KSVC_3SL_High) {
>+  auto fixture = CreateVideoQualityTestFixture();
>+  ParamsWithLogging simulcast;
>+  simulcast.call.send_side_bwe = true;
>+  simulcast.video[0] = kSvcVp9Video;
>+  simulcast.analyzer = {"vp9ksvc_3sl_high", 0.0, 0.0,
>+                        kFullStackTestDurationSecs};
>+  simulcast.ss[0] = {
>+      std::vector<VideoStream>(),  0,    3, 2, InterLayerPredMode::kOnKeyPic,
>+      std::vector<SpatialLayer>(), false};
>+  fixture->RunWithAnalyzer(simulcast);
> }
>+
>+TEST(FullStackTest, VP9KSVC_3SL_Medium) {
>+  auto fixture = CreateVideoQualityTestFixture();
>+  ParamsWithLogging simulcast;
>+  simulcast.call.send_side_bwe = true;
>+  simulcast.video[0] = kSvcVp9Video;
>+  simulcast.analyzer = {"vp9ksvc_3sl_medium", 0.0, 0.0,
>+                        kFullStackTestDurationSecs};
>+  simulcast.ss[0] = {
>+      std::vector<VideoStream>(),  0,    3, 1, InterLayerPredMode::kOnKeyPic,
>+      std::vector<SpatialLayer>(), false};
>+  fixture->RunWithAnalyzer(simulcast);
>+}
>+
>+TEST(FullStackTest, VP9KSVC_3SL_Low) {
>+  auto fixture = CreateVideoQualityTestFixture();
>+  ParamsWithLogging simulcast;
>+  simulcast.call.send_side_bwe = true;
>+  simulcast.video[0] = kSvcVp9Video;
>+  simulcast.analyzer = {"vp9ksvc_3sl_low", 0.0, 0.0,
>+                        kFullStackTestDurationSecs};
>+  simulcast.ss[0] = {
>+      std::vector<VideoStream>(),  0,    3, 0, InterLayerPredMode::kOnKeyPic,
>+      std::vector<SpatialLayer>(), false};
>+  fixture->RunWithAnalyzer(simulcast);
>+}
>+
>+TEST(FullStackTest, VP9KSVC_3SL_Medium_Network_Restricted) {
>+  auto fixture = CreateVideoQualityTestFixture();
>+  VideoQualityTest::Params simulcast;
>+  simulcast.call.send_side_bwe = true;
>+  simulcast.video[0] = kSvcVp9Video;
>+  simulcast.analyzer = {"vp9ksvc_3sl_medium_network_restricted", 0.0, 0.0,
>+                        kFullStackTestDurationSecs};
>+  simulcast.ss[0] = {
>+      std::vector<VideoStream>(),  0,    3, 1, InterLayerPredMode::kOnKeyPic,
>+      std::vector<SpatialLayer>(), false};
>+  simulcast.pipe.link_capacity_kbps = 1000;
>+  fixture->RunWithAnalyzer(simulcast);
>+}
>+#endif  // !defined(WEBRTC_MAC)
>+
> #endif  // !defined(RTC_DISABLE_VP9)
> 
> // Android bots can't handle FullHD, so disable the test.
>-#if defined(WEBRTC_ANDROID)
>+// TODO(bugs.webrtc.org/9220): Investigate source of flakiness on Mac.
>+#if defined(WEBRTC_ANDROID) || defined(WEBRTC_MAC)
> #define MAYBE_SimulcastFullHdOveruse DISABLED_SimulcastFullHdOveruse
> #else
> #define MAYBE_SimulcastFullHdOveruse SimulcastFullHdOveruse
> #endif
> 
>-TEST_F(FullStackTest, MAYBE_SimulcastFullHdOveruse) {
>-  VideoQualityTest::Params simulcast;
>+TEST(FullStackTest, MAYBE_SimulcastFullHdOveruse) {
>+  auto fixture = CreateVideoQualityTestFixture();
>+  ParamsWithLogging simulcast;
>   simulcast.call.send_side_bwe = true;
>   simulcast.video[0] = {true,    1920,    1080,  30,    800000,
>                         2500000, 2500000, false, "VP8", 3,
>-                        2,       400000,  false, false, "Generator"};
>+                        2,       400000,  false, false, false, "Generator"};
>   simulcast.analyzer = {"simulcast_HD_high", 0.0, 0.0,
>                         kFullStackTestDurationSecs};
>   simulcast.pipe.loss_percent = 0;
>   simulcast.pipe.queue_delay_ms = 100;
>-  std::vector<VideoStream> streams = {DefaultVideoStream(simulcast, 0),
>-                                      DefaultVideoStream(simulcast, 0),
>-                                      DefaultVideoStream(simulcast, 0)};
>-  simulcast.ss[0] = {streams, 2, 1, 0, std::vector<SpatialLayer>(), true};
>-  webrtc::test::ScopedFieldTrials override_trials(
>-      "WebRTC-ForceSimulatedOveruseIntervalMs/1000-50000-300/");
>-  RunTest(simulcast);
>+  std::vector<VideoStream> streams = {
>+    VideoQualityTest::DefaultVideoStream(simulcast, 0),
>+    VideoQualityTest::DefaultVideoStream(simulcast, 0),
>+    VideoQualityTest::DefaultVideoStream(simulcast, 0)
>+  };
>+  simulcast.ss[0] = {
>+      streams, 2, 1, 0, InterLayerPredMode::kOn, std::vector<SpatialLayer>(),
>+      true};
>+  webrtc::test::ScopedFieldTrials override_trials(AppendFieldTrials(
>+      "WebRTC-ForceSimulatedOveruseIntervalMs/1000-50000-300/"));
>+  fixture->RunWithAnalyzer(simulcast);
> }
> 
>-TEST_F(FullStackTest, SimulcastVP8_3SL_High) {
>-  VideoQualityTest::Params simulcast;
>+TEST(FullStackTest, SimulcastVP8_3SL_High) {
>+  auto fixture = CreateVideoQualityTestFixture();
>+  ParamsWithLogging simulcast;
>   simulcast.call.send_side_bwe = true;
>   simulcast.video[0] = kSimulcastVp8VideoHigh;
>   simulcast.analyzer = {"simulcast_vp8_3sl_high", 0.0, 0.0,
>                         kFullStackTestDurationSecs};
>   simulcast.pipe.loss_percent = 0;
>   simulcast.pipe.queue_delay_ms = 100;
>-  VideoQualityTest::Params video_params_high;
>+  ParamsWithLogging video_params_high;
>   video_params_high.video[0] = kSimulcastVp8VideoHigh;
>-  VideoQualityTest::Params video_params_medium;
>+  ParamsWithLogging video_params_medium;
>   video_params_medium.video[0] = kSimulcastVp8VideoMedium;
>-  VideoQualityTest::Params video_params_low;
>+  ParamsWithLogging video_params_low;
>   video_params_low.video[0] = kSimulcastVp8VideoLow;
> 
>   std::vector<VideoStream> streams = {
>-      DefaultVideoStream(video_params_low, 0),
>-      DefaultVideoStream(video_params_medium, 0),
>-      DefaultVideoStream(video_params_high, 0)};
>-  simulcast.ss[0] = {streams, 2, 1, 0, std::vector<SpatialLayer>(), false};
>-  RunTest(simulcast);
>+      VideoQualityTest::DefaultVideoStream(video_params_low, 0),
>+      VideoQualityTest::DefaultVideoStream(video_params_medium, 0),
>+      VideoQualityTest::DefaultVideoStream(video_params_high, 0)};
>+  simulcast.ss[0] = {
>+      streams, 2, 1, 0, InterLayerPredMode::kOn, std::vector<SpatialLayer>(),
>+      false};
>+  fixture->RunWithAnalyzer(simulcast);
> }
> 
>-TEST_F(FullStackTest, SimulcastVP8_3SL_Medium) {
>-  VideoQualityTest::Params simulcast;
>+TEST(FullStackTest, SimulcastVP8_3SL_Medium) {
>+  auto fixture = CreateVideoQualityTestFixture();
>+  ParamsWithLogging simulcast;
>   simulcast.call.send_side_bwe = true;
>   simulcast.video[0] = kSimulcastVp8VideoHigh;
>   simulcast.analyzer = {"simulcast_vp8_3sl_medium", 0.0, 0.0,
>                         kFullStackTestDurationSecs};
>   simulcast.pipe.loss_percent = 0;
>   simulcast.pipe.queue_delay_ms = 100;
>-  VideoQualityTest::Params video_params_high;
>+  ParamsWithLogging video_params_high;
>   video_params_high.video[0] = kSimulcastVp8VideoHigh;
>-  VideoQualityTest::Params video_params_medium;
>+  ParamsWithLogging video_params_medium;
>   video_params_medium.video[0] = kSimulcastVp8VideoMedium;
>-  VideoQualityTest::Params video_params_low;
>+  ParamsWithLogging video_params_low;
>   video_params_low.video[0] = kSimulcastVp8VideoLow;
> 
>   std::vector<VideoStream> streams = {
>-      DefaultVideoStream(video_params_low, 0),
>-      DefaultVideoStream(video_params_medium, 0),
>-      DefaultVideoStream(video_params_high, 0)};
>-  simulcast.ss[0] = {streams, 1, 1, 0, std::vector<SpatialLayer>(), false};
>-  RunTest(simulcast);
>+      VideoQualityTest::DefaultVideoStream(video_params_low, 0),
>+      VideoQualityTest::DefaultVideoStream(video_params_medium, 0),
>+      VideoQualityTest::DefaultVideoStream(video_params_high, 0)};
>+  simulcast.ss[0] = {
>+      streams, 1, 1, 0, InterLayerPredMode::kOn, std::vector<SpatialLayer>(),
>+      false};
>+  fixture->RunWithAnalyzer(simulcast);
> }
> 
>-TEST_F(FullStackTest, SimulcastVP8_3SL_Low) {
>-  VideoQualityTest::Params simulcast;
>+TEST(FullStackTest, SimulcastVP8_3SL_Low) {
>+  auto fixture = CreateVideoQualityTestFixture();
>+  ParamsWithLogging simulcast;
>   simulcast.call.send_side_bwe = true;
>   simulcast.video[0] = kSimulcastVp8VideoHigh;
>   simulcast.analyzer = {"simulcast_vp8_3sl_low", 0.0, 0.0,
>                         kFullStackTestDurationSecs};
>   simulcast.pipe.loss_percent = 0;
>   simulcast.pipe.queue_delay_ms = 100;
>-  VideoQualityTest::Params video_params_high;
>+  ParamsWithLogging video_params_high;
>   video_params_high.video[0] = kSimulcastVp8VideoHigh;
>-  VideoQualityTest::Params video_params_medium;
>+  ParamsWithLogging video_params_medium;
>   video_params_medium.video[0] = kSimulcastVp8VideoMedium;
>-  VideoQualityTest::Params video_params_low;
>+  ParamsWithLogging video_params_low;
>   video_params_low.video[0] = kSimulcastVp8VideoLow;
> 
>   std::vector<VideoStream> streams = {
>-      DefaultVideoStream(video_params_low, 0),
>-      DefaultVideoStream(video_params_medium, 0),
>-      DefaultVideoStream(video_params_high, 0)};
>-  simulcast.ss[0] = {streams, 0, 1, 0, std::vector<SpatialLayer>(), false};
>-  RunTest(simulcast);
>+      VideoQualityTest::DefaultVideoStream(video_params_low, 0),
>+      VideoQualityTest::DefaultVideoStream(video_params_medium, 0),
>+      VideoQualityTest::DefaultVideoStream(video_params_high, 0)};
>+  simulcast.ss[0] = {
>+      streams, 0, 1, 0, InterLayerPredMode::kOn, std::vector<SpatialLayer>(),
>+      false};
>+  fixture->RunWithAnalyzer(simulcast);
> }
> 
>-TEST_F(FullStackTest, LargeRoomVP8_5thumb) {
>-  VideoQualityTest::Params large_room;
>+TEST(FullStackTest, LargeRoomVP8_5thumb) {
>+  auto fixture = CreateVideoQualityTestFixture();
>+  ParamsWithLogging large_room;
>   large_room.call.send_side_bwe = true;
>   large_room.video[0] = kSimulcastVp8VideoHigh;
>   large_room.analyzer = {"largeroom_5thumb", 0.0, 0.0,
>                          kFullStackTestDurationSecs};
>   large_room.pipe.loss_percent = 0;
>   large_room.pipe.queue_delay_ms = 100;
>-  VideoQualityTest::Params video_params_high;
>+  ParamsWithLogging video_params_high;
>   video_params_high.video[0] = kSimulcastVp8VideoHigh;
>-  VideoQualityTest::Params video_params_medium;
>+  ParamsWithLogging video_params_medium;
>   video_params_medium.video[0] = kSimulcastVp8VideoMedium;
>-  VideoQualityTest::Params video_params_low;
>+  ParamsWithLogging video_params_low;
>   video_params_low.video[0] = kSimulcastVp8VideoLow;
> 
>   std::vector<VideoStream> streams = {
>-      DefaultVideoStream(video_params_low, 0),
>-      DefaultVideoStream(video_params_medium, 0),
>-      DefaultVideoStream(video_params_high, 0)};
>+      VideoQualityTest::DefaultVideoStream(video_params_low, 0),
>+      VideoQualityTest::DefaultVideoStream(video_params_medium, 0),
>+      VideoQualityTest::DefaultVideoStream(video_params_high, 0)};
>   large_room.call.num_thumbnails = 5;
>-  large_room.ss[0] = {streams, 2, 1, 0, std::vector<SpatialLayer>(), false};
>-  RunTest(large_room);
>+  large_room.ss[0] = {
>+      streams, 2, 1, 0, InterLayerPredMode::kOn, std::vector<SpatialLayer>(),
>+      false};
>+  fixture->RunWithAnalyzer(large_room);
> }
> 
> #if defined(WEBRTC_ANDROID) || defined(WEBRTC_IOS)
>@@ -842,104 +999,112 @@ TEST_F(FullStackTest, LargeRoomVP8_5thumb) {
> #define MAYBE_LargeRoomVP8_15thumb LargeRoomVP8_15thumb
> #endif
> 
>-TEST_F(FullStackTest, MAYBE_LargeRoomVP8_15thumb) {
>-  VideoQualityTest::Params large_room;
>+TEST(FullStackTest, MAYBE_LargeRoomVP8_15thumb) {
>+  auto fixture = CreateVideoQualityTestFixture();
>+  ParamsWithLogging large_room;
>   large_room.call.send_side_bwe = true;
>   large_room.video[0] = kSimulcastVp8VideoHigh;
>   large_room.analyzer = {"largeroom_15thumb", 0.0, 0.0,
>                          kFullStackTestDurationSecs};
>   large_room.pipe.loss_percent = 0;
>   large_room.pipe.queue_delay_ms = 100;
>-  VideoQualityTest::Params video_params_high;
>+  ParamsWithLogging video_params_high;
>   video_params_high.video[0] = kSimulcastVp8VideoHigh;
>-  VideoQualityTest::Params video_params_medium;
>+  ParamsWithLogging video_params_medium;
>   video_params_medium.video[0] = kSimulcastVp8VideoMedium;
>-  VideoQualityTest::Params video_params_low;
>+  ParamsWithLogging video_params_low;
>   video_params_low.video[0] = kSimulcastVp8VideoLow;
> 
>   std::vector<VideoStream> streams = {
>-      DefaultVideoStream(video_params_low, 0),
>-      DefaultVideoStream(video_params_medium, 0),
>-      DefaultVideoStream(video_params_high, 0)};
>+      VideoQualityTest::DefaultVideoStream(video_params_low, 0),
>+      VideoQualityTest::DefaultVideoStream(video_params_medium, 0),
>+      VideoQualityTest::DefaultVideoStream(video_params_high, 0)};
>   large_room.call.num_thumbnails = 15;
>-  large_room.ss[0] = {streams, 2, 1, 0, std::vector<SpatialLayer>(), false};
>-  RunTest(large_room);
>+  large_room.ss[0] = {
>+      streams, 2, 1, 0, InterLayerPredMode::kOn, std::vector<SpatialLayer>(),
>+      false};
>+  fixture->RunWithAnalyzer(large_room);
> }
> 
>-TEST_F(FullStackTest, MAYBE_LargeRoomVP8_50thumb) {
>-  VideoQualityTest::Params large_room;
>+TEST(FullStackTest, MAYBE_LargeRoomVP8_50thumb) {
>+  auto fixture = CreateVideoQualityTestFixture();
>+  ParamsWithLogging large_room;
>   large_room.call.send_side_bwe = true;
>   large_room.video[0] = kSimulcastVp8VideoHigh;
>   large_room.analyzer = {"largeroom_50thumb", 0.0, 0.0,
>                          kFullStackTestDurationSecs};
>   large_room.pipe.loss_percent = 0;
>   large_room.pipe.queue_delay_ms = 100;
>-  VideoQualityTest::Params video_params_high;
>+  ParamsWithLogging video_params_high;
>   video_params_high.video[0] = kSimulcastVp8VideoHigh;
>-  VideoQualityTest::Params video_params_medium;
>+  ParamsWithLogging video_params_medium;
>   video_params_medium.video[0] = kSimulcastVp8VideoMedium;
>-  VideoQualityTest::Params video_params_low;
>+  ParamsWithLogging video_params_low;
>   video_params_low.video[0] = kSimulcastVp8VideoLow;
> 
>   std::vector<VideoStream> streams = {
>-      DefaultVideoStream(video_params_low, 0),
>-      DefaultVideoStream(video_params_medium, 0),
>-      DefaultVideoStream(video_params_high, 0)};
>+      VideoQualityTest::DefaultVideoStream(video_params_low, 0),
>+      VideoQualityTest::DefaultVideoStream(video_params_medium, 0),
>+      VideoQualityTest::DefaultVideoStream(video_params_high, 0)};
>   large_room.call.num_thumbnails = 50;
>-  large_room.ss[0] = {streams, 2, 1, 0, std::vector<SpatialLayer>(), false};
>-  RunTest(large_room);
>+  large_room.ss[0] = {
>+      streams, 2, 1, 0, InterLayerPredMode::kOn, std::vector<SpatialLayer>(),
>+      false};
>+  fixture->RunWithAnalyzer(large_room);
> }
> 
>-class DualStreamsTest : public FullStackTest,
>-                        public ::testing::WithParamInterface<int> {};
>+class DualStreamsTest : public ::testing::TestWithParam<int> {};
> 
> // Disable dual video test on mobile device becuase it's too heavy.
> #if !defined(WEBRTC_ANDROID) && !defined(WEBRTC_IOS)
> TEST_P(DualStreamsTest,
>        ModeratelyRestricted_SlidesVp8_3TL_Simulcast_Video_Simulcast_High) {
>   test::ScopedFieldTrials field_trial(
>-      kScreenshareSimulcastExperiment + kAlrProbingExperiment +
>-      kRoundRobinPacingQueueExperiment + kPacerPushBackExperiment);
>+      AppendFieldTrials(std::string(kRoundRobinPacingQueueExperiment) +
>+                        std::string(kPacerPushBackExperiment)));
>   const int first_stream = GetParam();
>-  VideoQualityTest::Params dual_streams;
>+  ParamsWithLogging dual_streams;
> 
>   // Screenshare Settings.
>   dual_streams.screenshare[first_stream] = {true, false, 10};
>   dual_streams.video[first_stream] = {true,    1850,    1110,  5,     800000,
>                                       2500000, 2500000, false, "VP8", 3,
>-                                      2,       400000,  false, false, ""};
>+                                      2,       400000,  false, false, false,
>+                                      ""};
> 
>-  VideoQualityTest::Params screenshare_params_high;
>-  screenshare_params_high.video[0] = {true,    1850,    1110,  5,     800000,
>-                                      2500000, 2500000, false, "VP8", 3,
>-                                      0,       400000,  false, false, ""};
>+  ParamsWithLogging screenshare_params_high;
>+  screenshare_params_high.video[0] = {true,    1850,  1110,  5, 400000, 1000000,
>+                                      1000000, false, "VP8", 3, 0,      400000,
>+                                      false,   false, false, ""};
>   VideoQualityTest::Params screenshare_params_low;
>-  screenshare_params_low.video[0] = {true,   1850,    1110,  5,     50000,
>-                                     200000, 2000000, false, "VP8", 2,
>-                                     0,      400000,  false, false, ""};
>+  screenshare_params_low.video[0] = {true,    1850,  1110,  5, 50000, 200000,
>+                                     1000000, false, "VP8", 2, 0,     400000,
>+                                     false,   false, false, ""};
>   std::vector<VideoStream> screenhsare_streams = {
>-      DefaultVideoStream(screenshare_params_low, 0),
>-      DefaultVideoStream(screenshare_params_high, 0)};
>+      VideoQualityTest::DefaultVideoStream(screenshare_params_low, 0),
>+      VideoQualityTest::DefaultVideoStream(screenshare_params_high, 0)};
> 
>-  dual_streams.ss[first_stream] = {screenhsare_streams,         1,    1, 0,
>-                                   std::vector<SpatialLayer>(), false};
>+  dual_streams.ss[first_stream] = {
>+      screenhsare_streams,         1,    1, 0, InterLayerPredMode::kOn,
>+      std::vector<SpatialLayer>(), false};
> 
>   // Video settings.
>   dual_streams.video[1 - first_stream] = kSimulcastVp8VideoHigh;
> 
>-  VideoQualityTest::Params video_params_high;
>+  ParamsWithLogging video_params_high;
>   video_params_high.video[0] = kSimulcastVp8VideoHigh;
>-  VideoQualityTest::Params video_params_medium;
>+  ParamsWithLogging video_params_medium;
>   video_params_medium.video[0] = kSimulcastVp8VideoMedium;
>-  VideoQualityTest::Params video_params_low;
>+  ParamsWithLogging video_params_low;
>   video_params_low.video[0] = kSimulcastVp8VideoLow;
>   std::vector<VideoStream> streams = {
>-      DefaultVideoStream(video_params_low, 0),
>-      DefaultVideoStream(video_params_medium, 0),
>-      DefaultVideoStream(video_params_high, 0)};
>+      VideoQualityTest::DefaultVideoStream(video_params_low, 0),
>+      VideoQualityTest::DefaultVideoStream(video_params_medium, 0),
>+      VideoQualityTest::DefaultVideoStream(video_params_high, 0)};
> 
>   dual_streams.ss[1 - first_stream] = {
>-      streams, 2, 1, 0, std::vector<SpatialLayer>(), false};
>+      streams, 2, 1, 0, InterLayerPredMode::kOn, std::vector<SpatialLayer>(),
>+      false};
> 
>   // Call settings.
>   dual_streams.call.send_side_bwe = true;
>@@ -952,26 +1117,29 @@ TEST_P(DualStreamsTest,
>   dual_streams.pipe.queue_length_packets = 30;
>   dual_streams.pipe.queue_delay_ms = 100;
> 
>-  RunTest(dual_streams);
>+  auto fixture = CreateVideoQualityTestFixture();
>+  fixture->RunWithAnalyzer(dual_streams);
> }
> #endif  // !defined(WEBRTC_ANDROID) && !defined(WEBRTC_IOS)
> 
> TEST_P(DualStreamsTest, Conference_Restricted) {
>-  test::ScopedFieldTrials field_trial(kRoundRobinPacingQueueExperiment +
>-                                      kPacerPushBackExperiment);
>+  test::ScopedFieldTrials field_trial(
>+      AppendFieldTrials(std::string(kRoundRobinPacingQueueExperiment) +
>+                        std::string(kPacerPushBackExperiment)));
>   const int first_stream = GetParam();
>-  VideoQualityTest::Params dual_streams;
>+  ParamsWithLogging dual_streams;
> 
>   // Screenshare Settings.
>   dual_streams.screenshare[first_stream] = {true, false, 10};
>   dual_streams.video[first_stream] = {true,    1850,    1110,  5,     800000,
>                                       2500000, 2500000, false, "VP8", 3,
>-                                      2,       400000,  false, false, ""};
>+                                      2,       400000,  false, false, false,
>+                                      ""};
>   // Video settings.
>   dual_streams.video[1 - first_stream] = {
>       true,   1280,   720,   30,    150000,
>       500000, 700000, false, "VP8", 3,
>-      2,      400000, false, false, "ConferenceMotion_1280_720_50"};
>+      2,      400000, false, false, false, "ConferenceMotion_1280_720_50"};
> 
>   // Call settings.
>   dual_streams.call.send_side_bwe = true;
>@@ -984,7 +1152,8 @@ TEST_P(DualStreamsTest, Conference_Restricted) {
>   dual_streams.pipe.queue_length_packets = 30;
>   dual_streams.pipe.queue_delay_ms = 100;
> 
>-  RunTest(dual_streams);
>+  auto fixture = CreateVideoQualityTestFixture();
>+  fixture->RunWithAnalyzer(dual_streams);
> }
> 
> INSTANTIATE_TEST_CASE_P(FullStackTest,
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/full_stack_tests_plot.py b/Source/ThirdParty/libwebrtc/Source/webrtc/video/full_stack_tests_plot.py
>index 3b324da67bb..f50c297b17e 100755
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/full_stack_tests_plot.py
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/full_stack_tests_plot.py
>@@ -196,9 +196,9 @@ def AverageOverCycle(values, length):
> 
>   total = [0.0] * length
>   count = [0] * length
>-  for k in range(len(values)):
>-    if values[k] is not None:
>-      total[k % length] += values[k]
>+  for k, val in enumerate(values):
>+    if val is not None:
>+      total[k % length] += val
>       count[k % length] += 1
> 
>   result = [0.0] * length
>@@ -386,11 +386,11 @@ def PlotConfigsFromArgs(args):
>   #   argparse.ArgumentParser, modified to remember the order of arguments.
>   #   Then we traverse the argument list and fill the PlotConfig.
>   args = itertools.groupby(args, lambda x: x in ["-n", "--next"])
>-  args = list(list(group) for match, group in args if not match)
>+  prep_args = list(list(group) for match, group in args if not match)
> 
>   parser = GetParser()
>   plot_configs = []
>-  for index, raw_args in enumerate(args):
>+  for index, raw_args in enumerate(prep_args):
>     graph_args = parser.parse_args(raw_args).ordered_args
>     plot_configs.append(_PlotConfigFromArgs(graph_args, index))
>   return plot_configs
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/overuse_frame_detector.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/overuse_frame_detector.cc
>index 71e3bd68cd2..b4f75eb9441 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/overuse_frame_detector.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/overuse_frame_detector.cc
>@@ -12,6 +12,7 @@
> 
> #include <assert.h>
> #include <math.h>
>+#include <stdio.h>
> 
> #include <algorithm>
> #include <list>
>@@ -109,12 +110,12 @@ class SendProcessingUsage1 : public OveruseFrameDetector::ProcessingUsage {
>                                         time_when_first_seen_us));
>   }
> 
>-  rtc::Optional<int> FrameSent(
>+  absl::optional<int> FrameSent(
>       uint32_t timestamp,
>       int64_t time_sent_in_us,
>       int64_t /* capture_time_us */,
>-      rtc::Optional<int> /* encode_duration_us */) override {
>-    rtc::Optional<int> encode_duration_us;
>+      absl::optional<int> /* encode_duration_us */) override {
>+    absl::optional<int> encode_duration_us;
>     // Delay before reporting actual encoding time, used to have the ability to
>     // detect total encoding time when encoding more than one layer. Encoding is
>     // here assumed to finish within a second (or that we get enough long-time
>@@ -194,7 +195,8 @@ class SendProcessingUsage1 : public OveruseFrameDetector::ProcessingUsage {
>   float InitialUsageInPercent() const {
>     // Start in between the underuse and overuse threshold.
>     return (options_.low_encode_usage_threshold_percent +
>-            options_.high_encode_usage_threshold_percent) / 2.0f;
>+            options_.high_encode_usage_threshold_percent) /
>+           2.0f;
>   }
> 
>   float InitialProcessingMs() const {
>@@ -240,13 +242,24 @@ class SendProcessingUsage2 : public OveruseFrameDetector::ProcessingUsage {
>                      int64_t time_when_first_seen_us,
>                      int64_t last_capture_time_us) override {}
> 
>-  rtc::Optional<int> FrameSent(uint32_t timestamp,
>-                               int64_t time_sent_in_us,
>-                               int64_t capture_time_us,
>-                               rtc::Optional<int> encode_duration_us) override {
>+  absl::optional<int> FrameSent(
>+      uint32_t /* timestamp */,
>+      int64_t /* time_sent_in_us */,
>+      int64_t capture_time_us,
>+      absl::optional<int> encode_duration_us) override {
>     if (encode_duration_us) {
>+      int duration_per_frame_us =
>+          DurationPerInputFrame(capture_time_us, *encode_duration_us);
>       if (prev_time_us_ != -1) {
>-        AddSample(1e-6 * (*encode_duration_us),
>+        if (capture_time_us < prev_time_us_) {
>+          // The weighting in AddSample assumes that samples are processed with
>+          // non-decreasing measurement timestamps. We could implement
>+          // appropriate weights for samples arriving late, but since it is a
>+          // rare case, keep things simple, by just pushing those measurements a
>+          // bit forward in time.
>+          capture_time_us = prev_time_us_;
>+        }
>+        AddSample(1e-6 * duration_per_frame_us,
>                   1e-6 * (capture_time_us - prev_time_us_));
>       }
>     }
>@@ -276,12 +289,43 @@ class SendProcessingUsage2 : public OveruseFrameDetector::ProcessingUsage {
>     load_estimate_ = c * encode_time + exp(-e) * load_estimate_;
>   }
> 
>+  int64_t DurationPerInputFrame(int64_t capture_time_us,
>+                                int64_t encode_time_us) {
>+    // Discard data on old frames; limit 2 seconds.
>+    static constexpr int64_t kMaxAge = 2 * rtc::kNumMicrosecsPerSec;
>+    for (auto it = max_encode_time_per_input_frame_.begin();
>+         it != max_encode_time_per_input_frame_.end() &&
>+         it->first < capture_time_us - kMaxAge;) {
>+      it = max_encode_time_per_input_frame_.erase(it);
>+    }
>+
>+    std::map<int64_t, int>::iterator it;
>+    bool inserted;
>+    std::tie(it, inserted) = max_encode_time_per_input_frame_.emplace(
>+        capture_time_us, encode_time_us);
>+    if (inserted) {
>+      // First encoded frame for this input frame.
>+      return encode_time_us;
>+    }
>+    if (encode_time_us <= it->second) {
>+      // Shorter encode time than previous frame (unlikely). Count it as being
>+      // done in parallel.
>+      return 0;
>+    }
>+    // Record new maximum encode time, and return increase from previous max.
>+    int increase = encode_time_us - it->second;
>+    it->second = encode_time_us;
>+    return increase;
>+  }
>+
>   int Value() override {
>     return static_cast<int>(100.0 * load_estimate_ + 0.5);
>   }
> 
>- private:
>   const CpuOveruseOptions options_;
>+  // Indexed by the capture timestamp, used as frame id.
>+  std::map<int64_t, int> max_encode_time_per_input_frame_;
>+
>   int64_t prev_time_us_ = -1;
>   double load_estimate_;
> };
>@@ -320,13 +364,13 @@ class OverdoseInjector : public OveruseFrameDetector::ProcessingUsage {
>     usage_->FrameCaptured(frame, time_when_first_seen_us, last_capture_time_us);
>   }
> 
>-  rtc::Optional<int> FrameSent(
>+  absl::optional<int> FrameSent(
>       // These two argument used by old estimator.
>       uint32_t timestamp,
>       int64_t time_sent_in_us,
>       // And these two by the new estimator.
>       int64_t capture_time_us,
>-      rtc::Optional<int> encode_duration_us) override {
>+      absl::optional<int> encode_duration_us) override {
>     return usage_->FrameSent(timestamp, time_sent_in_us, capture_time_us,
>                              encode_duration_us);
>   }
>@@ -361,7 +405,7 @@ class OverdoseInjector : public OveruseFrameDetector::ProcessingUsage {
>       }
>     }
> 
>-    rtc::Optional<int> overried_usage_value;
>+    absl::optional<int> overried_usage_value;
>     switch (state_) {
>       case State::kNormal:
>         break;
>@@ -440,13 +484,12 @@ CpuOveruseOptions::CpuOveruseOptions()
> }
> 
> std::unique_ptr<OveruseFrameDetector::ProcessingUsage>
>-OveruseFrameDetector::CreateProcessingUsage(
>-    const CpuOveruseOptions& options) {
>+OveruseFrameDetector::CreateProcessingUsage(const CpuOveruseOptions& options) {
>   std::unique_ptr<ProcessingUsage> instance;
>   if (options.filter_time_ms > 0) {
>-    instance = rtc::MakeUnique<SendProcessingUsage2>(options);
>+    instance = absl::make_unique<SendProcessingUsage2>(options);
>   } else {
>-    instance = rtc::MakeUnique<SendProcessingUsage1>(options);
>+    instance = absl::make_unique<SendProcessingUsage1>(options);
>   }
>   std::string toggling_interval =
>       field_trial::FindFullName("WebRTC-ForceSimulatedOveruseIntervalMs");
>@@ -458,9 +501,9 @@ OveruseFrameDetector::CreateProcessingUsage(
>                &overuse_period_ms, &underuse_period_ms) == 3) {
>       if (normal_period_ms > 0 && overuse_period_ms > 0 &&
>           underuse_period_ms > 0) {
>-        instance = rtc::MakeUnique<OverdoseInjector>(
>-            std::move(instance), normal_period_ms,
>-            overuse_period_ms, underuse_period_ms);
>+        instance = absl::make_unique<OverdoseInjector>(
>+            std::move(instance), normal_period_ms, overuse_period_ms,
>+            underuse_period_ms);
>       } else {
>         RTC_LOG(LS_WARNING)
>             << "Invalid (non-positive) normal/overuse/underuse periods: "
>@@ -477,8 +520,9 @@ OveruseFrameDetector::CreateProcessingUsage(
> 
> class OveruseFrameDetector::CheckOveruseTask : public rtc::QueuedTask {
>  public:
>-  explicit CheckOveruseTask(OveruseFrameDetector* overuse_detector)
>-      : overuse_detector_(overuse_detector) {
>+  CheckOveruseTask(OveruseFrameDetector* overuse_detector,
>+                   AdaptationObserverInterface* observer)
>+      : overuse_detector_(overuse_detector), observer_(observer) {
>     rtc::TaskQueue::Current()->PostDelayedTask(
>         std::unique_ptr<rtc::QueuedTask>(this), kTimeToFirstCheckForOveruseMs);
>   }
>@@ -493,7 +537,7 @@ class OveruseFrameDetector::CheckOveruseTask : public rtc::QueuedTask {
>     RTC_CHECK(task_checker_.CalledSequentially());
>     if (!overuse_detector_)
>       return true;  // This will make the task queue delete this task.
>-    overuse_detector_->CheckForOveruse();
>+    overuse_detector_->CheckForOveruse(observer_);
> 
>     rtc::TaskQueue::Current()->PostDelayedTask(
>         std::unique_ptr<rtc::QueuedTask>(this), kCheckForOveruseIntervalMs);
>@@ -503,18 +547,16 @@ class OveruseFrameDetector::CheckOveruseTask : public rtc::QueuedTask {
>   }
>   rtc::SequencedTaskChecker task_checker_;
>   OveruseFrameDetector* overuse_detector_;
>+  // Observer getting overuse reports.
>+  AdaptationObserverInterface* observer_;
> };
> 
> OveruseFrameDetector::OveruseFrameDetector(
>-    const CpuOveruseOptions& options,
>-    AdaptationObserverInterface* observer,
>     CpuOveruseMetricsObserver* metrics_observer)
>     : check_overuse_task_(nullptr),
>-      options_(options),
>-      observer_(observer),
>       metrics_observer_(metrics_observer),
>       num_process_times_(0),
>-      // TODO(nisse): Use rtc::Optional
>+      // TODO(nisse): Use absl::optional
>       last_capture_time_us_(-1),
>       num_pixels_(0),
>       max_framerate_(kDefaultFrameRate),
>@@ -523,8 +565,7 @@ OveruseFrameDetector::OveruseFrameDetector(
>       num_overuse_detections_(0),
>       last_rampup_time_ms_(-1),
>       in_quick_rampup_(false),
>-      current_rampup_delay_ms_(kStandardRampUpDelayMs),
>-      usage_(CreateProcessingUsage(options)) {
>+      current_rampup_delay_ms_(kStandardRampUpDelayMs) {
>   task_checker_.Detach();
> }
> 
>@@ -532,24 +573,30 @@ OveruseFrameDetector::~OveruseFrameDetector() {
>   RTC_DCHECK(!check_overuse_task_) << "StopCheckForOverUse must be called.";
> }
> 
>-void OveruseFrameDetector::StartCheckForOveruse() {
>+void OveruseFrameDetector::StartCheckForOveruse(
>+    const CpuOveruseOptions& options,
>+    AdaptationObserverInterface* overuse_observer) {
>   RTC_DCHECK_CALLED_SEQUENTIALLY(&task_checker_);
>   RTC_DCHECK(!check_overuse_task_);
>-  check_overuse_task_ = new CheckOveruseTask(this);
>+  RTC_DCHECK(overuse_observer != nullptr);
>+
>+  SetOptions(options);
>+  check_overuse_task_ = new CheckOveruseTask(this, overuse_observer);
> }
> void OveruseFrameDetector::StopCheckForOveruse() {
>   RTC_DCHECK_CALLED_SEQUENTIALLY(&task_checker_);
>-  check_overuse_task_->Stop();
>-  check_overuse_task_ = nullptr;
>+  if (check_overuse_task_) {
>+    check_overuse_task_->Stop();
>+    check_overuse_task_ = nullptr;
>+  }
> }
> 
> void OveruseFrameDetector::EncodedFrameTimeMeasured(int encode_duration_ms) {
>   RTC_DCHECK_CALLED_SEQUENTIALLY(&task_checker_);
>-  if (!metrics_)
>-    metrics_ = CpuOveruseMetrics();
>-  metrics_->encode_usage_percent = usage_->Value();
>+  encode_usage_percent_ = usage_->Value();
> 
>-  metrics_observer_->OnEncodedFrameTimeMeasured(encode_duration_ms, *metrics_);
>+  metrics_observer_->OnEncodedFrameTimeMeasured(encode_duration_ms,
>+                                                *encode_usage_percent_);
> }
> 
> bool OveruseFrameDetector::FrameSizeChanged(int num_pixels) const {
>@@ -565,7 +612,7 @@ bool OveruseFrameDetector::FrameTimeoutDetected(int64_t now_us) const {
>   if (last_capture_time_us_ == -1)
>     return false;
>   return (now_us - last_capture_time_us_) >
>-      options_.frame_timeout_interval_ms * rtc::kNumMicrosecsPerMillisec;
>+         options_.frame_timeout_interval_ms * rtc::kNumMicrosecsPerMillisec;
> }
> 
> void OveruseFrameDetector::ResetAll(int num_pixels) {
>@@ -576,7 +623,7 @@ void OveruseFrameDetector::ResetAll(int num_pixels) {
>   usage_->Reset();
>   last_capture_time_us_ = -1;
>   num_process_times_ = 0;
>-  metrics_ = rtc::nullopt;
>+  encode_usage_percent_ = absl::nullopt;
>   OnTargetFramerateUpdated(max_framerate_);
> }
> 
>@@ -604,7 +651,7 @@ void OveruseFrameDetector::FrameCaptured(const VideoFrame& frame,
> void OveruseFrameDetector::FrameSent(uint32_t timestamp,
>                                      int64_t time_sent_in_us,
>                                      int64_t capture_time_us,
>-                                     rtc::Optional<int> encode_duration_us) {
>+                                     absl::optional<int> encode_duration_us) {
>   RTC_DCHECK_CALLED_SEQUENTIALLY(&task_checker_);
>   encode_duration_us = usage_->FrameSent(timestamp, time_sent_in_us,
>                                          capture_time_us, encode_duration_us);
>@@ -615,15 +662,18 @@ void OveruseFrameDetector::FrameSent(uint32_t timestamp,
>   }
> }
> 
>-void OveruseFrameDetector::CheckForOveruse() {
>+void OveruseFrameDetector::CheckForOveruse(
>+    AdaptationObserverInterface* observer) {
>   RTC_DCHECK_CALLED_SEQUENTIALLY(&task_checker_);
>+  RTC_DCHECK(observer);
>   ++num_process_times_;
>-  if (num_process_times_ <= options_.min_process_count || !metrics_)
>+  if (num_process_times_ <= options_.min_process_count ||
>+      !encode_usage_percent_)
>     return;
> 
>   int64_t now_ms = rtc::TimeMillis();
> 
>-  if (IsOverusing(*metrics_)) {
>+  if (IsOverusing(*encode_usage_percent_)) {
>     // If the last thing we did was going up, and now have to back down, we need
>     // to check if this peak was short. If so we should back off to avoid going
>     // back and forth between this load, the system doesn't seem to handle it.
>@@ -646,30 +696,35 @@ void OveruseFrameDetector::CheckForOveruse() {
>     checks_above_threshold_ = 0;
>     ++num_overuse_detections_;
> 
>-    if (observer_)
>-      observer_->AdaptDown(kScaleReasonCpu);
>-  } else if (IsUnderusing(*metrics_, now_ms)) {
>+    observer->AdaptDown(kScaleReasonCpu);
>+  } else if (IsUnderusing(*encode_usage_percent_, now_ms)) {
>     last_rampup_time_ms_ = now_ms;
>     in_quick_rampup_ = true;
> 
>-    if (observer_)
>-      observer_->AdaptUp(kScaleReasonCpu);
>+    observer->AdaptUp(kScaleReasonCpu);
>   }
> 
>   int rampup_delay =
>       in_quick_rampup_ ? kQuickRampUpDelayMs : current_rampup_delay_ms_;
> 
>   RTC_LOG(LS_VERBOSE) << " Frame stats: "
>-                      << " encode usage " << metrics_->encode_usage_percent
>+                      << " encode usage " << *encode_usage_percent_
>                       << " overuse detections " << num_overuse_detections_
>                       << " rampup delay " << rampup_delay;
> }
> 
>-bool OveruseFrameDetector::IsOverusing(const CpuOveruseMetrics& metrics) {
>+void OveruseFrameDetector::SetOptions(const CpuOveruseOptions& options) {
>+  RTC_DCHECK_CALLED_SEQUENTIALLY(&task_checker_);
>+  options_ = options;
>+  // Force reset with next frame.
>+  num_pixels_ = 0;
>+  usage_ = CreateProcessingUsage(options);
>+}
>+
>+bool OveruseFrameDetector::IsOverusing(int usage_percent) {
>   RTC_DCHECK_CALLED_SEQUENTIALLY(&task_checker_);
> 
>-  if (metrics.encode_usage_percent >=
>-      options_.high_encode_usage_threshold_percent) {
>+  if (usage_percent >= options_.high_encode_usage_threshold_percent) {
>     ++checks_above_threshold_;
>   } else {
>     checks_above_threshold_ = 0;
>@@ -677,14 +732,12 @@ bool OveruseFrameDetector::IsOverusing(const CpuOveruseMetrics& metrics) {
>   return checks_above_threshold_ >= options_.high_threshold_consecutive_count;
> }
> 
>-bool OveruseFrameDetector::IsUnderusing(const CpuOveruseMetrics& metrics,
>-                                        int64_t time_now) {
>+bool OveruseFrameDetector::IsUnderusing(int usage_percent, int64_t time_now) {
>   RTC_DCHECK_CALLED_SEQUENTIALLY(&task_checker_);
>   int delay = in_quick_rampup_ ? kQuickRampUpDelayMs : current_rampup_delay_ms_;
>   if (time_now < last_rampup_time_ms_ + delay)
>     return false;
> 
>-  return metrics.encode_usage_percent <
>-         options_.low_encode_usage_threshold_percent;
>+  return usage_percent < options_.low_encode_usage_threshold_percent;
> }
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/overuse_frame_detector.h b/Source/ThirdParty/libwebrtc/Source/webrtc/video/overuse_frame_detector.h
>index f7ebdb2b0d9..ea0d76dd901 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/overuse_frame_detector.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/overuse_frame_detector.h
>@@ -14,7 +14,8 @@
> #include <list>
> #include <memory>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
>+#include "api/video/video_stream_encoder_observer.h"
> #include "modules/video_coding/utility/quality_scaler.h"
> #include "rtc_base/constructormagic.h"
> #include "rtc_base/numerics/exp_filter.h"
>@@ -34,7 +35,7 @@ struct CpuOveruseOptions {
>   // General settings.
>   int frame_timeout_interval_ms;  // The maximum allowed interval between two
>                                   // frames before resetting estimations.
>-  int min_frame_samples;  // The minimum number of frames required.
>+  int min_frame_samples;          // The minimum number of frames required.
>   int min_process_count;  // The number of initial process times required before
>                           // triggering an overuse/underuse.
>   int high_threshold_consecutive_count;  // The number of consecutive checks
>@@ -44,20 +45,6 @@ struct CpuOveruseOptions {
>   int filter_time_ms;  // Time constant for averaging
> };
> 
>-struct CpuOveruseMetrics {
>-  CpuOveruseMetrics() : encode_usage_percent(-1) {}
>-
>-  int encode_usage_percent;  // Average encode time divided by the average time
>-                             // difference between incoming captured frames.
>-};
>-
>-class CpuOveruseMetricsObserver {
>- public:
>-  virtual ~CpuOveruseMetricsObserver() {}
>-  virtual void OnEncodedFrameTimeMeasured(int encode_duration_ms,
>-                                          const CpuOveruseMetrics& metrics) = 0;
>-};
>-
> // Use to detect system overuse based on the send-side processing time of
> // incoming frames. All methods must be called on a single task queue but it can
> // be created and destroyed on an arbitrary thread.
>@@ -65,13 +52,12 @@ class CpuOveruseMetricsObserver {
> // check for overuse.
> class OveruseFrameDetector {
>  public:
>-  OveruseFrameDetector(const CpuOveruseOptions& options,
>-                       AdaptationObserverInterface* overuse_observer,
>-                       CpuOveruseMetricsObserver* metrics_observer);
>+  explicit OveruseFrameDetector(CpuOveruseMetricsObserver* metrics_observer);
>   virtual ~OveruseFrameDetector();
> 
>   // Start to periodically check for overuse.
>-  void StartCheckForOveruse();
>+  void StartCheckForOveruse(const CpuOveruseOptions& options,
>+                            AdaptationObserverInterface* overuse_observer);
> 
>   // StopCheckForOveruse must be called before destruction if
>   // StartCheckForOveruse has been called.
>@@ -91,7 +77,7 @@ class OveruseFrameDetector {
>   void FrameSent(uint32_t timestamp,
>                  int64_t time_sent_in_us,
>                  int64_t capture_time_us,
>-                 rtc::Optional<int> encode_duration_us);
>+                 absl::optional<int> encode_duration_us);
> 
>   // Interface for cpu load estimation. Intended for internal use only.
>   class ProcessingUsage {
>@@ -102,27 +88,31 @@ class OveruseFrameDetector {
>                                int64_t time_when_first_seen_us,
>                                int64_t last_capture_time_us) = 0;
>     // Returns encode_time in us, if there's a new measurement.
>-    virtual rtc::Optional<int> FrameSent(
>+    virtual absl::optional<int> FrameSent(
>         // These two argument used by old estimator.
>         uint32_t timestamp,
>         int64_t time_sent_in_us,
>         // And these two by the new estimator.
>         int64_t capture_time_us,
>-        rtc::Optional<int> encode_duration_us) = 0;
>+        absl::optional<int> encode_duration_us) = 0;
> 
>     virtual int Value() = 0;
>     virtual ~ProcessingUsage() = default;
>   };
> 
>  protected:
>-  void CheckForOveruse();  // Protected for test purposes.
>+  // Protected for test purposes.
>+  void CheckForOveruse(AdaptationObserverInterface* overuse_observer);
>+  void SetOptions(const CpuOveruseOptions& options);
>+
>+  CpuOveruseOptions options_;
> 
>  private:
>   class CheckOveruseTask;
> 
>   void EncodedFrameTimeMeasured(int encode_duration_ms);
>-  bool IsOverusing(const CpuOveruseMetrics& metrics);
>-  bool IsUnderusing(const CpuOveruseMetrics& metrics, int64_t time_now);
>+  bool IsOverusing(int encode_usage_percent);
>+  bool IsUnderusing(int encode_usage_percent, int64_t time_now);
> 
>   bool FrameTimeoutDetected(int64_t now) const;
>   bool FrameSizeChanged(int num_pixels) const;
>@@ -136,14 +126,9 @@ class OveruseFrameDetector {
>   // Owned by the task queue from where StartCheckForOveruse is called.
>   CheckOveruseTask* check_overuse_task_;
> 
>-  const CpuOveruseOptions options_;
>-
>-  // Observer getting overuse reports.
>-  AdaptationObserverInterface* const observer_;
>-
>   // Stats metrics.
>   CpuOveruseMetricsObserver* const metrics_observer_;
>-  rtc::Optional<CpuOveruseMetrics> metrics_ RTC_GUARDED_BY(task_checker_);
>+  absl::optional<int> encode_usage_percent_ RTC_GUARDED_BY(task_checker_);
> 
>   int64_t num_process_times_ RTC_GUARDED_BY(task_checker_);
> 
>@@ -159,8 +144,7 @@ class OveruseFrameDetector {
>   bool in_quick_rampup_ RTC_GUARDED_BY(task_checker_);
>   int current_rampup_delay_ms_ RTC_GUARDED_BY(task_checker_);
> 
>-  const std::unique_ptr<ProcessingUsage> usage_
>-      RTC_PT_GUARDED_BY(task_checker_);
>+  std::unique_ptr<ProcessingUsage> usage_ RTC_PT_GUARDED_BY(task_checker_);
> 
>   RTC_DISALLOW_COPY_AND_ASSIGN(OveruseFrameDetector);
> };
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/overuse_frame_detector_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/overuse_frame_detector_unittest.cc
>index cefc91c2ef8..ac019754c54 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/overuse_frame_detector_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/overuse_frame_detector_unittest.cc
>@@ -26,11 +26,11 @@ using ::testing::InvokeWithoutArgs;
> using ::testing::_;
> 
> namespace {
>-  const int kWidth = 640;
>-  const int kHeight = 480;
>-  // Corresponds to load of 15%
>-  const int kFrameIntervalUs = 33 * rtc::kNumMicrosecsPerMillisec;
>-  const int kProcessTimeUs = 5 * rtc::kNumMicrosecsPerMillisec;
>+const int kWidth = 640;
>+const int kHeight = 480;
>+// Corresponds to load of 15%
>+const int kFrameIntervalUs = 33 * rtc::kNumMicrosecsPerMillisec;
>+const int kProcessTimeUs = 5 * rtc::kNumMicrosecsPerMillisec;
> }  // namespace
> 
> class MockCpuOveruseObserver : public AdaptationObserverInterface {
>@@ -44,9 +44,7 @@ class MockCpuOveruseObserver : public AdaptationObserverInterface {
> 
> class CpuOveruseObserverImpl : public AdaptationObserverInterface {
>  public:
>-  CpuOveruseObserverImpl() :
>-    overuse_(0),
>-    normaluse_(0) {}
>+  CpuOveruseObserverImpl() : overuse_(0), normaluse_(0) {}
>   virtual ~CpuOveruseObserverImpl() {}
> 
>   void AdaptDown(AdaptReason) { ++overuse_; }
>@@ -58,39 +56,36 @@ class CpuOveruseObserverImpl : public AdaptationObserverInterface {
> 
> class OveruseFrameDetectorUnderTest : public OveruseFrameDetector {
>  public:
>-  OveruseFrameDetectorUnderTest(const CpuOveruseOptions& options,
>-                                AdaptationObserverInterface* overuse_observer,
>-                                CpuOveruseMetricsObserver* metrics_observer)
>-      : OveruseFrameDetector(options,
>-                             overuse_observer,
>-                             metrics_observer) {}
>+  explicit OveruseFrameDetectorUnderTest(
>+      CpuOveruseMetricsObserver* metrics_observer)
>+      : OveruseFrameDetector(metrics_observer) {}
>   ~OveruseFrameDetectorUnderTest() {}
> 
>   using OveruseFrameDetector::CheckForOveruse;
>+  using OveruseFrameDetector::SetOptions;
> };
> 
> class OveruseFrameDetectorTest : public ::testing::Test,
>                                  public CpuOveruseMetricsObserver {
>  protected:
>   void SetUp() override {
>-    observer_.reset(new MockCpuOveruseObserver());
>+    observer_ = &mock_observer_;
>     options_.min_process_count = 0;
>-    ReinitializeOveruseDetector();
>-  }
>-
>-  void ReinitializeOveruseDetector() {
>-    overuse_detector_.reset(new OveruseFrameDetectorUnderTest(
>-        options_, observer_.get(), this));
>+    overuse_detector_ = absl::make_unique<OveruseFrameDetectorUnderTest>(this);
>+    // Unfortunately, we can't call SetOptions here, since that would break
>+    // single-threading requirements in the RunOnTqNormalUsage test.
>   }
> 
>   void OnEncodedFrameTimeMeasured(int encode_time_ms,
>-                                  const CpuOveruseMetrics& metrics) override {
>-    metrics_ = metrics;
>+                                  int encode_usage_percent) override {
>+    encode_usage_percent_ = encode_usage_percent;
>   }
> 
>   int InitialUsage() {
>     return ((options_.low_encode_usage_threshold_percent +
>-             options_.high_encode_usage_threshold_percent) / 2.0f) + 0.5;
>+             options_.high_encode_usage_threshold_percent) /
>+            2.0f) +
>+           0.5;
>   }
> 
>   virtual void InsertAndSendFramesWithInterval(int num_frames,
>@@ -113,6 +108,36 @@ class OveruseFrameDetectorTest : public ::testing::Test,
>     }
>   }
> 
>+  virtual void InsertAndSendSimulcastFramesWithInterval(
>+      int num_frames,
>+      int interval_us,
>+      int width,
>+      int height,
>+      // One element per layer
>+      rtc::ArrayView<const int> delays_us) {
>+    VideoFrame frame(I420Buffer::Create(width, height),
>+                     webrtc::kVideoRotation_0, 0);
>+    uint32_t timestamp = 0;
>+    while (num_frames-- > 0) {
>+      frame.set_timestamp(timestamp);
>+      int64_t capture_time_us = rtc::TimeMicros();
>+      overuse_detector_->FrameCaptured(frame, capture_time_us);
>+      int max_delay_us = 0;
>+      for (int delay_us : delays_us) {
>+        if (delay_us > max_delay_us) {
>+          clock_.AdvanceTimeMicros(delay_us - max_delay_us);
>+          max_delay_us = delay_us;
>+        }
>+
>+        overuse_detector_->FrameSent(timestamp, rtc::TimeMicros(),
>+                                     capture_time_us, delay_us);
>+      }
>+      overuse_detector_->CheckForOveruse(observer_);
>+      clock_.AdvanceTimeMicros(interval_us - max_delay_us);
>+      timestamp += interval_us * 90 / 1000;
>+    }
>+  }
>+
>   virtual void InsertAndSendFramesWithRandomInterval(int num_frames,
>                                                      int min_interval_us,
>                                                      int max_interval_us,
>@@ -132,9 +157,9 @@ class OveruseFrameDetectorTest : public ::testing::Test,
>       clock_.AdvanceTimeMicros(delay_us);
>       overuse_detector_->FrameSent(timestamp, rtc::TimeMicros(),
>                                    capture_time_us,
>-                                   rtc::Optional<int>(delay_us));
>+                                   absl::optional<int>(delay_us));
> 
>-      overuse_detector_->CheckForOveruse();
>+      overuse_detector_->CheckForOveruse(observer_);
>       // Avoid turning clock backwards.
>       if (interval_us > delay_us)
>         clock_.AdvanceTimeMicros(interval_us - delay_us);
>@@ -148,29 +173,29 @@ class OveruseFrameDetectorTest : public ::testing::Test,
>     // the usage. From the tests where these are used, adding another sample
>     // doesn't affect the expected outcome (this is mainly to check initial
>     // values and whether the overuse detector has been reset or not).
>-    InsertAndSendFramesWithInterval(2, rtc::kNumMicrosecsPerSec,
>-                                    width, height, kFrameIntervalUs);
>+    InsertAndSendFramesWithInterval(2, rtc::kNumMicrosecsPerSec, width, height,
>+                                    kFrameIntervalUs);
>   }
>   void TriggerOveruse(int num_times) {
>     const int kDelayUs = 32 * rtc::kNumMicrosecsPerMillisec;
>     for (int i = 0; i < num_times; ++i) {
>-      InsertAndSendFramesWithInterval(
>-          1000, kFrameIntervalUs, kWidth, kHeight, kDelayUs);
>-      overuse_detector_->CheckForOveruse();
>+      InsertAndSendFramesWithInterval(1000, kFrameIntervalUs, kWidth, kHeight,
>+                                      kDelayUs);
>+      overuse_detector_->CheckForOveruse(observer_);
>     }
>   }
> 
>   void TriggerUnderuse() {
>     const int kDelayUs1 = 5000;
>     const int kDelayUs2 = 6000;
>-    InsertAndSendFramesWithInterval(
>-        1300, kFrameIntervalUs, kWidth, kHeight, kDelayUs1);
>-    InsertAndSendFramesWithInterval(
>-        1, kFrameIntervalUs, kWidth, kHeight, kDelayUs2);
>-    overuse_detector_->CheckForOveruse();
>+    InsertAndSendFramesWithInterval(1300, kFrameIntervalUs, kWidth, kHeight,
>+                                    kDelayUs1);
>+    InsertAndSendFramesWithInterval(1, kFrameIntervalUs, kWidth, kHeight,
>+                                    kDelayUs2);
>+    overuse_detector_->CheckForOveruse(observer_);
>   }
> 
>-  int UsagePercent() { return metrics_.encode_usage_percent; }
>+  int UsagePercent() { return encode_usage_percent_; }
> 
>   int64_t OveruseProcessingTimeLimitForFramerate(int fps) const {
>     int64_t frame_interval = rtc::kNumMicrosecsPerSec / fps;
>@@ -188,45 +213,39 @@ class OveruseFrameDetectorTest : public ::testing::Test,
> 
>   CpuOveruseOptions options_;
>   rtc::ScopedFakeClock clock_;
>-  std::unique_ptr<MockCpuOveruseObserver> observer_;
>+  MockCpuOveruseObserver mock_observer_;
>+  AdaptationObserverInterface* observer_;
>   std::unique_ptr<OveruseFrameDetectorUnderTest> overuse_detector_;
>-  CpuOveruseMetrics metrics_;
>+  int encode_usage_percent_ = -1;
> 
>   static const auto reason_ = AdaptationObserverInterface::AdaptReason::kCpu;
> };
> 
>-
> // UsagePercent() > high_encode_usage_threshold_percent => overuse.
> // UsagePercent() < low_encode_usage_threshold_percent => underuse.
> TEST_F(OveruseFrameDetectorTest, TriggerOveruse) {
>   // usage > high => overuse
>-  EXPECT_CALL(*(observer_.get()), AdaptDown(reason_)).Times(1);
>+  overuse_detector_->SetOptions(options_);
>+  EXPECT_CALL(mock_observer_, AdaptDown(reason_)).Times(1);
>   TriggerOveruse(options_.high_threshold_consecutive_count);
> }
> 
> TEST_F(OveruseFrameDetectorTest, OveruseAndRecover) {
>   // usage > high => overuse
>-  EXPECT_CALL(*(observer_.get()), AdaptDown(reason_)).Times(1);
>+  overuse_detector_->SetOptions(options_);
>+  EXPECT_CALL(mock_observer_, AdaptDown(reason_)).Times(1);
>   TriggerOveruse(options_.high_threshold_consecutive_count);
>   // usage < low => underuse
>-  EXPECT_CALL(*(observer_.get()), AdaptUp(reason_)).Times(testing::AtLeast(1));
>-  TriggerUnderuse();
>-}
>-
>-TEST_F(OveruseFrameDetectorTest, OveruseAndRecoverWithNoObserver) {
>-  overuse_detector_.reset(new OveruseFrameDetectorUnderTest(
>-      options_, nullptr, this));
>-  EXPECT_CALL(*(observer_.get()), AdaptDown(reason_)).Times(0);
>-  TriggerOveruse(options_.high_threshold_consecutive_count);
>-  EXPECT_CALL(*(observer_.get()), AdaptUp(reason_)).Times(0);
>+  EXPECT_CALL(mock_observer_, AdaptUp(reason_)).Times(testing::AtLeast(1));
>   TriggerUnderuse();
> }
> 
> TEST_F(OveruseFrameDetectorTest, DoubleOveruseAndRecover) {
>-  EXPECT_CALL(*(observer_.get()), AdaptDown(reason_)).Times(2);
>+  overuse_detector_->SetOptions(options_);
>+  EXPECT_CALL(mock_observer_, AdaptDown(reason_)).Times(2);
>   TriggerOveruse(options_.high_threshold_consecutive_count);
>   TriggerOveruse(options_.high_threshold_consecutive_count);
>-  EXPECT_CALL(*(observer_.get()), AdaptUp(reason_)).Times(testing::AtLeast(1));
>+  EXPECT_CALL(mock_observer_, AdaptUp(reason_)).Times(testing::AtLeast(1));
>   TriggerUnderuse();
> }
> 
>@@ -234,50 +253,55 @@ TEST_F(OveruseFrameDetectorTest, TriggerUnderuseWithMinProcessCount) {
>   const int kProcessIntervalUs = 5 * rtc::kNumMicrosecsPerSec;
>   options_.min_process_count = 1;
>   CpuOveruseObserverImpl overuse_observer;
>-  overuse_detector_.reset(new OveruseFrameDetectorUnderTest(
>-      options_, &overuse_observer, this));
>-  InsertAndSendFramesWithInterval(
>-      1200, kFrameIntervalUs, kWidth, kHeight, kProcessTimeUs);
>-  overuse_detector_->CheckForOveruse();
>+  observer_ = nullptr;
>+  overuse_detector_->SetOptions(options_);
>+  InsertAndSendFramesWithInterval(1200, kFrameIntervalUs, kWidth, kHeight,
>+                                  kProcessTimeUs);
>+  overuse_detector_->CheckForOveruse(&overuse_observer);
>   EXPECT_EQ(0, overuse_observer.normaluse_);
>   clock_.AdvanceTimeMicros(kProcessIntervalUs);
>-  overuse_detector_->CheckForOveruse();
>+  overuse_detector_->CheckForOveruse(&overuse_observer);
>   EXPECT_EQ(1, overuse_observer.normaluse_);
> }
> 
> TEST_F(OveruseFrameDetectorTest, ConstantOveruseGivesNoNormalUsage) {
>-  EXPECT_CALL(*(observer_.get()), AdaptUp(reason_)).Times(0);
>-  EXPECT_CALL(*(observer_.get()), AdaptDown(reason_)).Times(64);
>+  overuse_detector_->SetOptions(options_);
>+  EXPECT_CALL(mock_observer_, AdaptUp(reason_)).Times(0);
>+  EXPECT_CALL(mock_observer_, AdaptDown(reason_)).Times(64);
>   for (size_t i = 0; i < 64; ++i) {
>     TriggerOveruse(options_.high_threshold_consecutive_count);
>   }
> }
> 
> TEST_F(OveruseFrameDetectorTest, ConsecutiveCountTriggersOveruse) {
>-  EXPECT_CALL(*(observer_.get()), AdaptDown(reason_)).Times(1);
>+  overuse_detector_->SetOptions(options_);
>+  EXPECT_CALL(mock_observer_, AdaptDown(reason_)).Times(1);
>   options_.high_threshold_consecutive_count = 2;
>-  ReinitializeOveruseDetector();
>+  overuse_detector_->SetOptions(options_);
>   TriggerOveruse(2);
> }
> 
> TEST_F(OveruseFrameDetectorTest, IncorrectConsecutiveCountTriggersNoOveruse) {
>-  EXPECT_CALL(*(observer_.get()), AdaptDown(reason_)).Times(0);
>+  overuse_detector_->SetOptions(options_);
>+  EXPECT_CALL(mock_observer_, AdaptDown(reason_)).Times(0);
>   options_.high_threshold_consecutive_count = 2;
>-  ReinitializeOveruseDetector();
>+  overuse_detector_->SetOptions(options_);
>   TriggerOveruse(1);
> }
> 
> TEST_F(OveruseFrameDetectorTest, ProcessingUsage) {
>-  InsertAndSendFramesWithInterval(
>-      1000, kFrameIntervalUs, kWidth, kHeight, kProcessTimeUs);
>+  overuse_detector_->SetOptions(options_);
>+  InsertAndSendFramesWithInterval(1000, kFrameIntervalUs, kWidth, kHeight,
>+                                  kProcessTimeUs);
>   EXPECT_EQ(kProcessTimeUs * 100 / kFrameIntervalUs, UsagePercent());
> }
> 
> TEST_F(OveruseFrameDetectorTest, ResetAfterResolutionChange) {
>+  overuse_detector_->SetOptions(options_);
>   ForceUpdate(kWidth, kHeight);
>   EXPECT_EQ(InitialUsage(), UsagePercent());
>-  InsertAndSendFramesWithInterval(
>-      1000, kFrameIntervalUs, kWidth, kHeight, kProcessTimeUs);
>+  InsertAndSendFramesWithInterval(1000, kFrameIntervalUs, kWidth, kHeight,
>+                                  kProcessTimeUs);
>   EXPECT_NE(InitialUsage(), UsagePercent());
>   // Verify reset (with new width/height).
>   ForceUpdate(kWidth, kHeight + 1);
>@@ -285,28 +309,30 @@ TEST_F(OveruseFrameDetectorTest, ResetAfterResolutionChange) {
> }
> 
> TEST_F(OveruseFrameDetectorTest, ResetAfterFrameTimeout) {
>+  overuse_detector_->SetOptions(options_);
>   ForceUpdate(kWidth, kHeight);
>   EXPECT_EQ(InitialUsage(), UsagePercent());
>-  InsertAndSendFramesWithInterval(
>-      1000, kFrameIntervalUs, kWidth, kHeight, kProcessTimeUs);
>+  InsertAndSendFramesWithInterval(1000, kFrameIntervalUs, kWidth, kHeight,
>+                                  kProcessTimeUs);
>   EXPECT_NE(InitialUsage(), UsagePercent());
>   InsertAndSendFramesWithInterval(
>-      2, options_.frame_timeout_interval_ms *
>-      rtc::kNumMicrosecsPerMillisec, kWidth, kHeight, kProcessTimeUs);
>+      2, options_.frame_timeout_interval_ms * rtc::kNumMicrosecsPerMillisec,
>+      kWidth, kHeight, kProcessTimeUs);
>   EXPECT_NE(InitialUsage(), UsagePercent());
>   // Verify reset.
>   InsertAndSendFramesWithInterval(
>-      2, (options_.frame_timeout_interval_ms + 1) *
>-      rtc::kNumMicrosecsPerMillisec, kWidth, kHeight, kProcessTimeUs);
>+      2,
>+      (options_.frame_timeout_interval_ms + 1) * rtc::kNumMicrosecsPerMillisec,
>+      kWidth, kHeight, kProcessTimeUs);
>   ForceUpdate(kWidth, kHeight);
>   EXPECT_EQ(InitialUsage(), UsagePercent());
> }
> 
> TEST_F(OveruseFrameDetectorTest, MinFrameSamplesBeforeUpdating) {
>   options_.min_frame_samples = 40;
>-  ReinitializeOveruseDetector();
>-  InsertAndSendFramesWithInterval(
>-      40, kFrameIntervalUs, kWidth, kHeight, kProcessTimeUs);
>+  overuse_detector_->SetOptions(options_);
>+  InsertAndSendFramesWithInterval(40, kFrameIntervalUs, kWidth, kHeight,
>+                                  kProcessTimeUs);
>   EXPECT_EQ(InitialUsage(), UsagePercent());
>   // Pass time far enough to digest all previous samples.
>   clock_.AdvanceTimeMicros(rtc::kNumMicrosecsPerSec);
>@@ -317,19 +343,20 @@ TEST_F(OveruseFrameDetectorTest, MinFrameSamplesBeforeUpdating) {
> 
>   // Pass time far enough to digest all previous samples, 41 in total.
>   clock_.AdvanceTimeMicros(rtc::kNumMicrosecsPerSec);
>-  InsertAndSendFramesWithInterval(
>-      1, kFrameIntervalUs, kWidth, kHeight, kProcessTimeUs);
>+  InsertAndSendFramesWithInterval(1, kFrameIntervalUs, kWidth, kHeight,
>+                                  kProcessTimeUs);
>   EXPECT_NE(InitialUsage(), UsagePercent());
> }
> 
> TEST_F(OveruseFrameDetectorTest, InitialProcessingUsage) {
>+  overuse_detector_->SetOptions(options_);
>   ForceUpdate(kWidth, kHeight);
>   EXPECT_EQ(InitialUsage(), UsagePercent());
> }
> 
> TEST_F(OveruseFrameDetectorTest, MeasuresMultipleConcurrentSamples) {
>-  EXPECT_CALL(*(observer_.get()), AdaptDown(reason_))
>-      .Times(testing::AtLeast(1));
>+  overuse_detector_->SetOptions(options_);
>+  EXPECT_CALL(mock_observer_, AdaptDown(reason_)).Times(testing::AtLeast(1));
>   static const int kIntervalUs = 33 * rtc::kNumMicrosecsPerMillisec;
>   static const size_t kNumFramesEncodingDelay = 3;
>   VideoFrame frame(I420Buffer::Create(kWidth, kHeight),
>@@ -345,14 +372,14 @@ TEST_F(OveruseFrameDetectorTest, MeasuresMultipleConcurrentSamples) {
>           static_cast<uint32_t>(i - kNumFramesEncodingDelay), rtc::TimeMicros(),
>           capture_time_us, kIntervalUs);
>     }
>-    overuse_detector_->CheckForOveruse();
>+    overuse_detector_->CheckForOveruse(observer_);
>   }
> }
> 
> TEST_F(OveruseFrameDetectorTest, UpdatesExistingSamples) {
>   // >85% encoding time should trigger overuse.
>-  EXPECT_CALL(*(observer_.get()), AdaptDown(reason_))
>-      .Times(testing::AtLeast(1));
>+  overuse_detector_->SetOptions(options_);
>+  EXPECT_CALL(mock_observer_, AdaptDown(reason_)).Times(testing::AtLeast(1));
>   static const int kIntervalUs = 33 * rtc::kNumMicrosecsPerMillisec;
>   static const int kDelayUs = 30 * rtc::kNumMicrosecsPerMillisec;
>   VideoFrame frame(I420Buffer::Create(kWidth, kHeight),
>@@ -372,7 +399,7 @@ TEST_F(OveruseFrameDetectorTest, UpdatesExistingSamples) {
>                                  kDelayUs);
>     clock_.AdvanceTimeMicros(kIntervalUs - kDelayUs);
>     timestamp += kIntervalUs * 90 / 1000;
>-    overuse_detector_->CheckForOveruse();
>+    overuse_detector_->CheckForOveruse(observer_);
>   }
> }
> 
>@@ -381,14 +408,14 @@ TEST_F(OveruseFrameDetectorTest, RunOnTqNormalUsage) {
> 
>   rtc::Event event(false, false);
>   queue.PostTask([this, &event] {
>-    overuse_detector_->StartCheckForOveruse();
>+    overuse_detector_->StartCheckForOveruse(options_, observer_);
>     event.Set();
>   });
>   event.Wait(rtc::Event::kForever);
> 
>   // Expect NormalUsage(). When called, stop the |overuse_detector_| and then
>   // set |event| to end the test.
>-  EXPECT_CALL(*(observer_.get()), AdaptUp(reason_))
>+  EXPECT_CALL(mock_observer_, AdaptUp(reason_))
>       .WillOnce(InvokeWithoutArgs([this, &event] {
>         overuse_detector_->StopCheckForOveruse();
>         event.Set();
>@@ -410,39 +437,41 @@ TEST_F(OveruseFrameDetectorTest, MaxIntervalScalesWithFramerate) {
>   const int kCapturerMaxFrameRate = 30;
>   const int kEncodeMaxFrameRate = 20;  // Maximum fps the encoder can sustain.
> 
>+  overuse_detector_->SetOptions(options_);
>   // Trigger overuse.
>   int64_t frame_interval_us = rtc::kNumMicrosecsPerSec / kCapturerMaxFrameRate;
>   // Processing time just below over use limit given kEncodeMaxFrameRate.
>   int64_t processing_time_us =
>       (98 * OveruseProcessingTimeLimitForFramerate(kEncodeMaxFrameRate)) / 100;
>-  EXPECT_CALL(*(observer_.get()), AdaptDown(reason_)).Times(1);
>+  EXPECT_CALL(mock_observer_, AdaptDown(reason_)).Times(1);
>   for (int i = 0; i < options_.high_threshold_consecutive_count; ++i) {
>     InsertAndSendFramesWithInterval(1200, frame_interval_us, kWidth, kHeight,
>                                     processing_time_us);
>-    overuse_detector_->CheckForOveruse();
>+    overuse_detector_->CheckForOveruse(observer_);
>   }
> 
>   // Simulate frame rate reduction and normal usage.
>   frame_interval_us = rtc::kNumMicrosecsPerSec / kEncodeMaxFrameRate;
>   overuse_detector_->OnTargetFramerateUpdated(kEncodeMaxFrameRate);
>-  EXPECT_CALL(*(observer_.get()), AdaptDown(reason_)).Times(0);
>+  EXPECT_CALL(mock_observer_, AdaptDown(reason_)).Times(0);
>   for (int i = 0; i < options_.high_threshold_consecutive_count; ++i) {
>     InsertAndSendFramesWithInterval(1200, frame_interval_us, kWidth, kHeight,
>                                     processing_time_us);
>-    overuse_detector_->CheckForOveruse();
>+    overuse_detector_->CheckForOveruse(observer_);
>   }
> 
>   // Reduce processing time to trigger underuse.
>   processing_time_us =
>       (98 * UnderuseProcessingTimeLimitForFramerate(kEncodeMaxFrameRate)) / 100;
>-  EXPECT_CALL(*(observer_.get()), AdaptUp(reason_)).Times(1);
>+  EXPECT_CALL(mock_observer_, AdaptUp(reason_)).Times(1);
>   InsertAndSendFramesWithInterval(1200, frame_interval_us, kWidth, kHeight,
>                                   processing_time_us);
>-  overuse_detector_->CheckForOveruse();
>+  overuse_detector_->CheckForOveruse(observer_);
> }
> 
> TEST_F(OveruseFrameDetectorTest, RespectsMinFramerate) {
>   const int kMinFrameRate = 7;  // Minimum fps allowed by current detector impl.
>+  overuse_detector_->SetOptions(options_);
>   overuse_detector_->OnTargetFramerateUpdated(kMinFrameRate);
> 
>   // Normal usage just at the limit.
>@@ -450,35 +479,36 @@ TEST_F(OveruseFrameDetectorTest, RespectsMinFramerate) {
>   // Processing time just below over use limit given kEncodeMaxFrameRate.
>   int64_t processing_time_us =
>       (98 * OveruseProcessingTimeLimitForFramerate(kMinFrameRate)) / 100;
>-  EXPECT_CALL(*(observer_.get()), AdaptDown(reason_)).Times(0);
>+  EXPECT_CALL(mock_observer_, AdaptDown(reason_)).Times(0);
>   for (int i = 0; i < options_.high_threshold_consecutive_count; ++i) {
>     InsertAndSendFramesWithInterval(1200, frame_interval_us, kWidth, kHeight,
>                                     processing_time_us);
>-    overuse_detector_->CheckForOveruse();
>+    overuse_detector_->CheckForOveruse(observer_);
>   }
> 
>   // Over the limit to overuse.
>   processing_time_us =
>       (102 * OveruseProcessingTimeLimitForFramerate(kMinFrameRate)) / 100;
>-  EXPECT_CALL(*(observer_.get()), AdaptDown(reason_)).Times(1);
>+  EXPECT_CALL(mock_observer_, AdaptDown(reason_)).Times(1);
>   for (int i = 0; i < options_.high_threshold_consecutive_count; ++i) {
>     InsertAndSendFramesWithInterval(1200, frame_interval_us, kWidth, kHeight,
>                                     processing_time_us);
>-    overuse_detector_->CheckForOveruse();
>+    overuse_detector_->CheckForOveruse(observer_);
>   }
> 
>   // Reduce input frame rate. Should still trigger overuse.
>   overuse_detector_->OnTargetFramerateUpdated(kMinFrameRate - 1);
>-  EXPECT_CALL(*(observer_.get()), AdaptDown(reason_)).Times(1);
>+  EXPECT_CALL(mock_observer_, AdaptDown(reason_)).Times(1);
>   for (int i = 0; i < options_.high_threshold_consecutive_count; ++i) {
>     InsertAndSendFramesWithInterval(1200, frame_interval_us, kWidth, kHeight,
>                                     processing_time_us);
>-    overuse_detector_->CheckForOveruse();
>+    overuse_detector_->CheckForOveruse(observer_);
>   }
> }
> 
> TEST_F(OveruseFrameDetectorTest, LimitsMaxFrameInterval) {
>   const int kMaxFrameRate = 20;
>+  overuse_detector_->SetOptions(options_);
>   overuse_detector_->OnTargetFramerateUpdated(kMaxFrameRate);
>   int64_t frame_interval_us = rtc::kNumMicrosecsPerSec / kMaxFrameRate;
>   // Maximum frame interval allowed is 35% above ideal.
>@@ -491,29 +521,29 @@ TEST_F(OveruseFrameDetectorTest, LimitsMaxFrameInterval) {
> 
>   // Processing time just below overuse limit given kMaxFrameRate.
>   int64_t processing_time_us = (98 * max_processing_time_us) / 100;
>-  EXPECT_CALL(*(observer_.get()), AdaptDown(reason_)).Times(0);
>+  EXPECT_CALL(mock_observer_, AdaptDown(reason_)).Times(0);
>   for (int i = 0; i < options_.high_threshold_consecutive_count; ++i) {
>     InsertAndSendFramesWithInterval(1200, max_frame_interval_us, kWidth,
>                                     kHeight, processing_time_us);
>-    overuse_detector_->CheckForOveruse();
>+    overuse_detector_->CheckForOveruse(observer_);
>   }
> 
>   // Go above limit, trigger overuse.
>   processing_time_us = (102 * max_processing_time_us) / 100;
>-  EXPECT_CALL(*(observer_.get()), AdaptDown(reason_)).Times(1);
>+  EXPECT_CALL(mock_observer_, AdaptDown(reason_)).Times(1);
>   for (int i = 0; i < options_.high_threshold_consecutive_count; ++i) {
>     InsertAndSendFramesWithInterval(1200, max_frame_interval_us, kWidth,
>                                     kHeight, processing_time_us);
>-    overuse_detector_->CheckForOveruse();
>+    overuse_detector_->CheckForOveruse(observer_);
>   }
> 
>   // Increase frame interval, should still trigger overuse.
>   max_frame_interval_us *= 2;
>-  EXPECT_CALL(*(observer_.get()), AdaptDown(reason_)).Times(1);
>+  EXPECT_CALL(mock_observer_, AdaptDown(reason_)).Times(1);
>   for (int i = 0; i < options_.high_threshold_consecutive_count; ++i) {
>     InsertAndSendFramesWithInterval(1200, max_frame_interval_us, kWidth,
>                                     kHeight, processing_time_us);
>-    overuse_detector_->CheckForOveruse();
>+    overuse_detector_->CheckForOveruse(observer_);
>   }
> }
> 
>@@ -524,9 +554,10 @@ TEST_F(OveruseFrameDetectorTest, NoOveruseForLargeRandomFrameInterval) {
>   // behavior is improved in this scenario, with only AdaptUp events,
>   // and estimated load closer to the true average.
> 
>-  // EXPECT_CALL(*(observer_.get()), AdaptDown(_)).Times(0);
>-  // EXPECT_CALL(*(observer_.get()), AdaptUp(reason_))
>+  // EXPECT_CALL(mock_observer_, AdaptDown(_)).Times(0);
>+  // EXPECT_CALL(mock_observer_, AdaptUp(reason_))
>   //     .Times(testing::AtLeast(1));
>+  overuse_detector_->SetOptions(options_);
> 
>   const int kNumFrames = 500;
>   const int kEncodeTimeUs = 100 * rtc::kNumMicrosecsPerMillisec;
>@@ -538,9 +569,9 @@ TEST_F(OveruseFrameDetectorTest, NoOveruseForLargeRandomFrameInterval) {
> 
>   overuse_detector_->OnTargetFramerateUpdated(kTargetFramerate);
> 
>-  InsertAndSendFramesWithRandomInterval(kNumFrames,
>-                                        kMinIntervalUs, kMaxIntervalUs,
>-                                        kWidth, kHeight, kEncodeTimeUs);
>+  InsertAndSendFramesWithRandomInterval(kNumFrames, kMinIntervalUs,
>+                                        kMaxIntervalUs, kWidth, kHeight,
>+                                        kEncodeTimeUs);
>   // Average usage 19%. Check that estimate is in the right ball park.
>   // EXPECT_NEAR(UsagePercent(), 20, 10);
>   EXPECT_NEAR(UsagePercent(), 20, 35);
>@@ -551,8 +582,9 @@ TEST_F(OveruseFrameDetectorTest, NoOveruseForLargeRandomFrameInterval) {
> TEST_F(OveruseFrameDetectorTest, NoOveruseForRandomFrameIntervalWithReset) {
>   // TODO(bugs.webrtc.org/8504): When new estimator is relanded,
>   // behavior is improved in this scenario, and we get AdaptUp events.
>-  EXPECT_CALL(*(observer_.get()), AdaptDown(_)).Times(0);
>-  // EXPECT_CALL(*(observer_.get()), AdaptUp(reason_))
>+  overuse_detector_->SetOptions(options_);
>+  EXPECT_CALL(mock_observer_, AdaptDown(_)).Times(0);
>+  // EXPECT_CALL(mock_observer_, AdaptUp(reason_))
>   //     .Times(testing::AtLeast(1));
> 
>   const int kNumFrames = 500;
>@@ -565,9 +597,9 @@ TEST_F(OveruseFrameDetectorTest, NoOveruseForRandomFrameIntervalWithReset) {
> 
>   overuse_detector_->OnTargetFramerateUpdated(kTargetFramerate);
> 
>-  InsertAndSendFramesWithRandomInterval(kNumFrames,
>-                                        kMinIntervalUs, kMaxIntervalUs,
>-                                        kWidth, kHeight, kEncodeTimeUs);
>+  InsertAndSendFramesWithRandomInterval(kNumFrames, kMinIntervalUs,
>+                                        kMaxIntervalUs, kWidth, kHeight,
>+                                        kEncodeTimeUs);
> 
>   // Average usage 6.6%, but since the frame_timeout_interval_ms is
>   // only 1500 ms, we often reset the estimate to the initial value.
>@@ -576,6 +608,27 @@ TEST_F(OveruseFrameDetectorTest, NoOveruseForRandomFrameIntervalWithReset) {
>   EXPECT_LE(UsagePercent(), InitialUsage() + 5);
> }
> 
>+// Models simulcast, with multiple encoded frames for each input frame.
>+// Load estimate should be based on the maximum encode time per input frame.
>+TEST_F(OveruseFrameDetectorTest, NoOveruseForSimulcast) {
>+  overuse_detector_->SetOptions(options_);
>+  EXPECT_CALL(mock_observer_, AdaptDown(_)).Times(0);
>+
>+  constexpr int kNumFrames = 500;
>+  constexpr int kEncodeTimesUs[] = {
>+      10 * rtc::kNumMicrosecsPerMillisec, 8 * rtc::kNumMicrosecsPerMillisec,
>+      12 * rtc::kNumMicrosecsPerMillisec,
>+  };
>+  constexpr int kIntervalUs = 30 * rtc::kNumMicrosecsPerMillisec;
>+
>+  InsertAndSendSimulcastFramesWithInterval(kNumFrames, kIntervalUs, kWidth,
>+                                           kHeight, kEncodeTimesUs);
>+
>+  // Average usage 40%. 12 ms / 30 ms.
>+  EXPECT_GE(UsagePercent(), 35);
>+  EXPECT_LE(UsagePercent(), 45);
>+}
>+
> // Tests using new cpu load estimator
> class OveruseFrameDetectorTest2 : public OveruseFrameDetectorTest {
>  protected:
>@@ -619,7 +672,7 @@ class OveruseFrameDetectorTest2 : public OveruseFrameDetectorTest {
>                                    0 /* ignored send_time_us */,
>                                    capture_time_us, delay_us);
> 
>-      overuse_detector_->CheckForOveruse();
>+      overuse_detector_->CheckForOveruse(observer_);
>       clock_.AdvanceTimeMicros(interval_us);
>     }
>   }
>@@ -636,33 +689,27 @@ class OveruseFrameDetectorTest2 : public OveruseFrameDetectorTest {
> // UsagePercent() < low_encode_usage_threshold_percent => underuse.
> TEST_F(OveruseFrameDetectorTest2, TriggerOveruse) {
>   // usage > high => overuse
>-  EXPECT_CALL(*(observer_.get()), AdaptDown(reason_)).Times(1);
>+  overuse_detector_->SetOptions(options_);
>+  EXPECT_CALL(mock_observer_, AdaptDown(reason_)).Times(1);
>   TriggerOveruse(options_.high_threshold_consecutive_count);
> }
> 
> TEST_F(OveruseFrameDetectorTest2, OveruseAndRecover) {
>   // usage > high => overuse
>-  EXPECT_CALL(*(observer_.get()), AdaptDown(reason_)).Times(1);
>+  overuse_detector_->SetOptions(options_);
>+  EXPECT_CALL(mock_observer_, AdaptDown(reason_)).Times(1);
>   TriggerOveruse(options_.high_threshold_consecutive_count);
>   // usage < low => underuse
>-  EXPECT_CALL(*(observer_.get()), AdaptUp(reason_)).Times(testing::AtLeast(1));
>-  TriggerUnderuse();
>-}
>-
>-TEST_F(OveruseFrameDetectorTest2, OveruseAndRecoverWithNoObserver) {
>-  overuse_detector_.reset(new OveruseFrameDetectorUnderTest(
>-      options_, nullptr, this));
>-  EXPECT_CALL(*(observer_.get()), AdaptDown(reason_)).Times(0);
>-  TriggerOveruse(options_.high_threshold_consecutive_count);
>-  EXPECT_CALL(*(observer_.get()), AdaptUp(reason_)).Times(0);
>+  EXPECT_CALL(mock_observer_, AdaptUp(reason_)).Times(testing::AtLeast(1));
>   TriggerUnderuse();
> }
> 
> TEST_F(OveruseFrameDetectorTest2, DoubleOveruseAndRecover) {
>-  EXPECT_CALL(*(observer_.get()), AdaptDown(reason_)).Times(2);
>+  overuse_detector_->SetOptions(options_);
>+  EXPECT_CALL(mock_observer_, AdaptDown(reason_)).Times(2);
>   TriggerOveruse(options_.high_threshold_consecutive_count);
>   TriggerOveruse(options_.high_threshold_consecutive_count);
>-  EXPECT_CALL(*(observer_.get()), AdaptUp(reason_)).Times(testing::AtLeast(1));
>+  EXPECT_CALL(mock_observer_, AdaptUp(reason_)).Times(testing::AtLeast(1));
>   TriggerUnderuse();
> }
> 
>@@ -670,50 +717,53 @@ TEST_F(OveruseFrameDetectorTest2, TriggerUnderuseWithMinProcessCount) {
>   const int kProcessIntervalUs = 5 * rtc::kNumMicrosecsPerSec;
>   options_.min_process_count = 1;
>   CpuOveruseObserverImpl overuse_observer;
>-  overuse_detector_.reset(new OveruseFrameDetectorUnderTest(
>-      options_, &overuse_observer, this));
>-  InsertAndSendFramesWithInterval(
>-      1200, kFrameIntervalUs, kWidth, kHeight, kProcessTimeUs);
>-  overuse_detector_->CheckForOveruse();
>+  observer_ = nullptr;
>+  overuse_detector_->SetOptions(options_);
>+  InsertAndSendFramesWithInterval(1200, kFrameIntervalUs, kWidth, kHeight,
>+                                  kProcessTimeUs);
>+  overuse_detector_->CheckForOveruse(&overuse_observer);
>   EXPECT_EQ(0, overuse_observer.normaluse_);
>   clock_.AdvanceTimeMicros(kProcessIntervalUs);
>-  overuse_detector_->CheckForOveruse();
>+  overuse_detector_->CheckForOveruse(&overuse_observer);
>   EXPECT_EQ(1, overuse_observer.normaluse_);
> }
> 
> TEST_F(OveruseFrameDetectorTest2, ConstantOveruseGivesNoNormalUsage) {
>-  EXPECT_CALL(*(observer_.get()), AdaptUp(reason_)).Times(0);
>-  EXPECT_CALL(*(observer_.get()), AdaptDown(reason_)).Times(64);
>+  overuse_detector_->SetOptions(options_);
>+  EXPECT_CALL(mock_observer_, AdaptUp(reason_)).Times(0);
>+  EXPECT_CALL(mock_observer_, AdaptDown(reason_)).Times(64);
>   for (size_t i = 0; i < 64; ++i) {
>     TriggerOveruse(options_.high_threshold_consecutive_count);
>   }
> }
> 
> TEST_F(OveruseFrameDetectorTest2, ConsecutiveCountTriggersOveruse) {
>-  EXPECT_CALL(*(observer_.get()), AdaptDown(reason_)).Times(1);
>+  EXPECT_CALL(mock_observer_, AdaptDown(reason_)).Times(1);
>   options_.high_threshold_consecutive_count = 2;
>-  ReinitializeOveruseDetector();
>+  overuse_detector_->SetOptions(options_);
>   TriggerOveruse(2);
> }
> 
> TEST_F(OveruseFrameDetectorTest2, IncorrectConsecutiveCountTriggersNoOveruse) {
>-  EXPECT_CALL(*(observer_.get()), AdaptDown(reason_)).Times(0);
>+  EXPECT_CALL(mock_observer_, AdaptDown(reason_)).Times(0);
>   options_.high_threshold_consecutive_count = 2;
>-  ReinitializeOveruseDetector();
>+  overuse_detector_->SetOptions(options_);
>   TriggerOveruse(1);
> }
> 
> TEST_F(OveruseFrameDetectorTest2, ProcessingUsage) {
>-  InsertAndSendFramesWithInterval(
>-      1000, kFrameIntervalUs, kWidth, kHeight, kProcessTimeUs);
>+  overuse_detector_->SetOptions(options_);
>+  InsertAndSendFramesWithInterval(1000, kFrameIntervalUs, kWidth, kHeight,
>+                                  kProcessTimeUs);
>   EXPECT_EQ(kProcessTimeUs * 100 / kFrameIntervalUs, UsagePercent());
> }
> 
> TEST_F(OveruseFrameDetectorTest2, ResetAfterResolutionChange) {
>+  overuse_detector_->SetOptions(options_);
>   ForceUpdate(kWidth, kHeight);
>   EXPECT_EQ(InitialUsage(), UsagePercent());
>-  InsertAndSendFramesWithInterval(
>-      1000, kFrameIntervalUs, kWidth, kHeight, kProcessTimeUs);
>+  InsertAndSendFramesWithInterval(1000, kFrameIntervalUs, kWidth, kHeight,
>+                                  kProcessTimeUs);
>   EXPECT_NE(InitialUsage(), UsagePercent());
>   // Verify reset (with new width/height).
>   ForceUpdate(kWidth, kHeight + 1);
>@@ -721,24 +771,27 @@ TEST_F(OveruseFrameDetectorTest2, ResetAfterResolutionChange) {
> }
> 
> TEST_F(OveruseFrameDetectorTest2, ResetAfterFrameTimeout) {
>+  overuse_detector_->SetOptions(options_);
>   ForceUpdate(kWidth, kHeight);
>   EXPECT_EQ(InitialUsage(), UsagePercent());
>-  InsertAndSendFramesWithInterval(
>-      1000, kFrameIntervalUs, kWidth, kHeight, kProcessTimeUs);
>+  InsertAndSendFramesWithInterval(1000, kFrameIntervalUs, kWidth, kHeight,
>+                                  kProcessTimeUs);
>   EXPECT_NE(InitialUsage(), UsagePercent());
>   InsertAndSendFramesWithInterval(
>-      2, options_.frame_timeout_interval_ms *
>-      rtc::kNumMicrosecsPerMillisec, kWidth, kHeight, kProcessTimeUs);
>+      2, options_.frame_timeout_interval_ms * rtc::kNumMicrosecsPerMillisec,
>+      kWidth, kHeight, kProcessTimeUs);
>   EXPECT_NE(InitialUsage(), UsagePercent());
>   // Verify reset.
>   InsertAndSendFramesWithInterval(
>-      2, (options_.frame_timeout_interval_ms + 1) *
>-      rtc::kNumMicrosecsPerMillisec, kWidth, kHeight, kProcessTimeUs);
>+      2,
>+      (options_.frame_timeout_interval_ms + 1) * rtc::kNumMicrosecsPerMillisec,
>+      kWidth, kHeight, kProcessTimeUs);
>   ForceUpdate(kWidth, kHeight);
>   EXPECT_EQ(InitialUsage(), UsagePercent());
> }
> 
> TEST_F(OveruseFrameDetectorTest2, ConvergesSlowly) {
>+  overuse_detector_->SetOptions(options_);
>   InsertAndSendFramesWithInterval(1, kFrameIntervalUs, kWidth, kHeight,
>                                   kProcessTimeUs);
>   // No update for the first sample.
>@@ -746,27 +799,28 @@ TEST_F(OveruseFrameDetectorTest2, ConvergesSlowly) {
> 
>   // Total time approximately 40 * 33ms = 1.3s, significantly less
>   // than the 5s time constant.
>-  InsertAndSendFramesWithInterval(
>-      40, kFrameIntervalUs, kWidth, kHeight, kProcessTimeUs);
>+  InsertAndSendFramesWithInterval(40, kFrameIntervalUs, kWidth, kHeight,
>+                                  kProcessTimeUs);
> 
>   // Should have started to approach correct load of 15%, but not very far.
>   EXPECT_LT(UsagePercent(), InitialUsage());
>   EXPECT_GT(UsagePercent(), (InitialUsage() * 3 + 15) / 4);
> 
>   // Run for roughly 10s more, should now be closer.
>-  InsertAndSendFramesWithInterval(
>-      300, kFrameIntervalUs, kWidth, kHeight, kProcessTimeUs);
>+  InsertAndSendFramesWithInterval(300, kFrameIntervalUs, kWidth, kHeight,
>+                                  kProcessTimeUs);
>   EXPECT_NEAR(UsagePercent(), 20, 5);
> }
> 
> TEST_F(OveruseFrameDetectorTest2, InitialProcessingUsage) {
>+  overuse_detector_->SetOptions(options_);
>   ForceUpdate(kWidth, kHeight);
>   EXPECT_EQ(InitialUsage(), UsagePercent());
> }
> 
> TEST_F(OveruseFrameDetectorTest2, MeasuresMultipleConcurrentSamples) {
>-  EXPECT_CALL(*(observer_.get()), AdaptDown(reason_))
>-      .Times(testing::AtLeast(1));
>+  overuse_detector_->SetOptions(options_);
>+  EXPECT_CALL(mock_observer_, AdaptDown(reason_)).Times(testing::AtLeast(1));
>   static const int kIntervalUs = 33 * rtc::kNumMicrosecsPerMillisec;
>   static const size_t kNumFramesEncodingDelay = 3;
>   VideoFrame frame(I420Buffer::Create(kWidth, kHeight),
>@@ -782,14 +836,14 @@ TEST_F(OveruseFrameDetectorTest2, MeasuresMultipleConcurrentSamples) {
>           static_cast<uint32_t>(i - kNumFramesEncodingDelay), rtc::TimeMicros(),
>           capture_time_us, kIntervalUs);
>     }
>-    overuse_detector_->CheckForOveruse();
>+    overuse_detector_->CheckForOveruse(observer_);
>   }
> }
> 
> TEST_F(OveruseFrameDetectorTest2, UpdatesExistingSamples) {
>   // >85% encoding time should trigger overuse.
>-  EXPECT_CALL(*(observer_.get()), AdaptDown(reason_))
>-      .Times(testing::AtLeast(1));
>+  overuse_detector_->SetOptions(options_);
>+  EXPECT_CALL(mock_observer_, AdaptDown(reason_)).Times(testing::AtLeast(1));
>   static const int kIntervalUs = 33 * rtc::kNumMicrosecsPerMillisec;
>   static const int kDelayUs = 30 * rtc::kNumMicrosecsPerMillisec;
>   VideoFrame frame(I420Buffer::Create(kWidth, kHeight),
>@@ -809,7 +863,7 @@ TEST_F(OveruseFrameDetectorTest2, UpdatesExistingSamples) {
>                                  kDelayUs);
>     clock_.AdvanceTimeMicros(kIntervalUs - kDelayUs);
>     timestamp += kIntervalUs * 90 / 1000;
>-    overuse_detector_->CheckForOveruse();
>+    overuse_detector_->CheckForOveruse(observer_);
>   }
> }
> 
>@@ -818,14 +872,14 @@ TEST_F(OveruseFrameDetectorTest2, RunOnTqNormalUsage) {
> 
>   rtc::Event event(false, false);
>   queue.PostTask([this, &event] {
>-    overuse_detector_->StartCheckForOveruse();
>+    overuse_detector_->StartCheckForOveruse(options_, observer_);
>     event.Set();
>   });
>   event.Wait(rtc::Event::kForever);
> 
>   // Expect NormalUsage(). When called, stop the |overuse_detector_| and then
>   // set |event| to end the test.
>-  EXPECT_CALL(*(observer_.get()), AdaptUp(reason_))
>+  EXPECT_CALL(mock_observer_, AdaptUp(reason_))
>       .WillOnce(InvokeWithoutArgs([this, &event] {
>         overuse_detector_->StopCheckForOveruse();
>         event.Set();
>@@ -846,9 +900,9 @@ TEST_F(OveruseFrameDetectorTest2, RunOnTqNormalUsage) {
> // Models screencast, with irregular arrival of frames which are heavy
> // to encode.
> TEST_F(OveruseFrameDetectorTest2, NoOveruseForLargeRandomFrameInterval) {
>-  EXPECT_CALL(*(observer_.get()), AdaptDown(_)).Times(0);
>-  EXPECT_CALL(*(observer_.get()), AdaptUp(reason_))
>-      .Times(testing::AtLeast(1));
>+  overuse_detector_->SetOptions(options_);
>+  EXPECT_CALL(mock_observer_, AdaptDown(_)).Times(0);
>+  EXPECT_CALL(mock_observer_, AdaptUp(reason_)).Times(testing::AtLeast(1));
> 
>   const int kNumFrames = 500;
>   const int kEncodeTimeUs = 100 * rtc::kNumMicrosecsPerMillisec;
>@@ -856,9 +910,9 @@ TEST_F(OveruseFrameDetectorTest2, NoOveruseForLargeRandomFrameInterval) {
>   const int kMinIntervalUs = 30 * rtc::kNumMicrosecsPerMillisec;
>   const int kMaxIntervalUs = 1000 * rtc::kNumMicrosecsPerMillisec;
> 
>-  InsertAndSendFramesWithRandomInterval(kNumFrames,
>-                                        kMinIntervalUs, kMaxIntervalUs,
>-                                        kWidth, kHeight, kEncodeTimeUs);
>+  InsertAndSendFramesWithRandomInterval(kNumFrames, kMinIntervalUs,
>+                                        kMaxIntervalUs, kWidth, kHeight,
>+                                        kEncodeTimeUs);
>   // Average usage 19%. Check that estimate is in the right ball park.
>   EXPECT_NEAR(UsagePercent(), 20, 10);
> }
>@@ -866,9 +920,9 @@ TEST_F(OveruseFrameDetectorTest2, NoOveruseForLargeRandomFrameInterval) {
> // Models screencast, with irregular arrival of frames, often
> // exceeding the timeout interval.
> TEST_F(OveruseFrameDetectorTest2, NoOveruseForRandomFrameIntervalWithReset) {
>-  EXPECT_CALL(*(observer_.get()), AdaptDown(_)).Times(0);
>-  EXPECT_CALL(*(observer_.get()), AdaptUp(reason_))
>-      .Times(testing::AtLeast(1));
>+  overuse_detector_->SetOptions(options_);
>+  EXPECT_CALL(mock_observer_, AdaptDown(_)).Times(0);
>+  EXPECT_CALL(mock_observer_, AdaptUp(reason_)).Times(testing::AtLeast(1));
> 
>   const int kNumFrames = 500;
>   const int kEncodeTimeUs = 100 * rtc::kNumMicrosecsPerMillisec;
>@@ -876,9 +930,9 @@ TEST_F(OveruseFrameDetectorTest2, NoOveruseForRandomFrameIntervalWithReset) {
>   const int kMinIntervalUs = 30 * rtc::kNumMicrosecsPerMillisec;
>   const int kMaxIntervalUs = 3000 * rtc::kNumMicrosecsPerMillisec;
> 
>-  InsertAndSendFramesWithRandomInterval(kNumFrames,
>-                                        kMinIntervalUs, kMaxIntervalUs,
>-                                        kWidth, kHeight, kEncodeTimeUs);
>+  InsertAndSendFramesWithRandomInterval(kNumFrames, kMinIntervalUs,
>+                                        kMaxIntervalUs, kWidth, kHeight,
>+                                        kEncodeTimeUs);
> 
>   // Average usage 6.6%, but since the frame_timeout_interval_ms is
>   // only 1500 ms, we often reset the estimate to the initial value.
>@@ -887,4 +941,40 @@ TEST_F(OveruseFrameDetectorTest2, NoOveruseForRandomFrameIntervalWithReset) {
>   EXPECT_LE(UsagePercent(), InitialUsage() + 5);
> }
> 
>+TEST_F(OveruseFrameDetectorTest2, ToleratesOutOfOrderFrames) {
>+  overuse_detector_->SetOptions(options_);
>+  // Represents a cpu utilization close to 100%. First input frame results in
>+  // three encoded frames, and the last of those isn't finished until after the
>+  // first encoded frame corresponding to the next input frame.
>+  const int kEncodeTimeUs = 30 * rtc::kNumMicrosecsPerMillisec;
>+  const int kCaptureTimesMs[] = { 33, 33, 66, 33 };
>+
>+  for (int capture_time_ms : kCaptureTimesMs) {
>+    overuse_detector_->FrameSent(
>+        0, 0, capture_time_ms * rtc::kNumMicrosecsPerMillisec, kEncodeTimeUs);
>+  }
>+  EXPECT_GE(UsagePercent(), InitialUsage());
>+}
>+
>+// Models simulcast, with multiple encoded frames for each input frame.
>+// Load estimate should be based on the maximum encode time per input frame.
>+TEST_F(OveruseFrameDetectorTest2, NoOveruseForSimulcast) {
>+  overuse_detector_->SetOptions(options_);
>+  EXPECT_CALL(mock_observer_, AdaptDown(_)).Times(0);
>+
>+  constexpr int kNumFrames = 500;
>+  constexpr int kEncodeTimesUs[] = {
>+      10 * rtc::kNumMicrosecsPerMillisec, 8 * rtc::kNumMicrosecsPerMillisec,
>+      12 * rtc::kNumMicrosecsPerMillisec,
>+  };
>+  constexpr int kIntervalUs = 30 * rtc::kNumMicrosecsPerMillisec;
>+
>+  InsertAndSendSimulcastFramesWithInterval(kNumFrames, kIntervalUs, kWidth,
>+                                           kHeight, kEncodeTimesUs);
>+
>+  // Average usage 40%. 12 ms / 30 ms.
>+  EXPECT_GE(UsagePercent(), 35);
>+  EXPECT_LE(UsagePercent(), 45);
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/payload_router.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/payload_router.cc
>deleted file mode 100644
>index f22183287d9..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/payload_router.cc
>+++ /dev/null
>@@ -1,275 +0,0 @@
>-/*
>- *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include "video/payload_router.h"
>-
>-#include "modules/rtp_rtcp/include/rtp_rtcp.h"
>-#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
>-#include "modules/video_coding/include/video_codec_interface.h"
>-#include "rtc_base/checks.h"
>-#include "rtc_base/random.h"
>-#include "rtc_base/timeutils.h"
>-#include "system_wrappers/include/field_trial.h"
>-
>-namespace webrtc {
>-
>-namespace {
>-// Map information from info into rtp.
>-void CopyCodecSpecific(const CodecSpecificInfo* info, RTPVideoHeader* rtp) {
>-  RTC_DCHECK(info);
>-  switch (info->codecType) {
>-    case kVideoCodecVP8: {
>-      rtp->codec = kRtpVideoVp8;
>-      rtp->codecHeader.VP8.InitRTPVideoHeaderVP8();
>-      rtp->codecHeader.VP8.pictureId = info->codecSpecific.VP8.pictureId;
>-      rtp->codecHeader.VP8.nonReference = info->codecSpecific.VP8.nonReference;
>-      rtp->codecHeader.VP8.temporalIdx = info->codecSpecific.VP8.temporalIdx;
>-      rtp->codecHeader.VP8.layerSync = info->codecSpecific.VP8.layerSync;
>-      rtp->codecHeader.VP8.tl0PicIdx = info->codecSpecific.VP8.tl0PicIdx;
>-      rtp->codecHeader.VP8.keyIdx = info->codecSpecific.VP8.keyIdx;
>-      rtp->simulcastIdx = info->codecSpecific.VP8.simulcastIdx;
>-      return;
>-    }
>-    case kVideoCodecVP9: {
>-      rtp->codec = kRtpVideoVp9;
>-      rtp->codecHeader.VP9.InitRTPVideoHeaderVP9();
>-      rtp->codecHeader.VP9.inter_pic_predicted =
>-          info->codecSpecific.VP9.inter_pic_predicted;
>-      rtp->codecHeader.VP9.flexible_mode =
>-          info->codecSpecific.VP9.flexible_mode;
>-      rtp->codecHeader.VP9.ss_data_available =
>-          info->codecSpecific.VP9.ss_data_available;
>-      rtp->codecHeader.VP9.picture_id = info->codecSpecific.VP9.picture_id;
>-      rtp->codecHeader.VP9.tl0_pic_idx = info->codecSpecific.VP9.tl0_pic_idx;
>-      rtp->codecHeader.VP9.temporal_idx = info->codecSpecific.VP9.temporal_idx;
>-      rtp->codecHeader.VP9.spatial_idx = info->codecSpecific.VP9.spatial_idx;
>-      rtp->codecHeader.VP9.temporal_up_switch =
>-          info->codecSpecific.VP9.temporal_up_switch;
>-      rtp->codecHeader.VP9.inter_layer_predicted =
>-          info->codecSpecific.VP9.inter_layer_predicted;
>-      rtp->codecHeader.VP9.gof_idx = info->codecSpecific.VP9.gof_idx;
>-      rtp->codecHeader.VP9.num_spatial_layers =
>-          info->codecSpecific.VP9.num_spatial_layers;
>-
>-      if (info->codecSpecific.VP9.ss_data_available) {
>-        rtp->codecHeader.VP9.spatial_layer_resolution_present =
>-            info->codecSpecific.VP9.spatial_layer_resolution_present;
>-        if (info->codecSpecific.VP9.spatial_layer_resolution_present) {
>-          for (size_t i = 0; i < info->codecSpecific.VP9.num_spatial_layers;
>-               ++i) {
>-            rtp->codecHeader.VP9.width[i] = info->codecSpecific.VP9.width[i];
>-            rtp->codecHeader.VP9.height[i] = info->codecSpecific.VP9.height[i];
>-          }
>-        }
>-        rtp->codecHeader.VP9.gof.CopyGofInfoVP9(info->codecSpecific.VP9.gof);
>-      }
>-
>-      rtp->codecHeader.VP9.num_ref_pics = info->codecSpecific.VP9.num_ref_pics;
>-      for (int i = 0; i < info->codecSpecific.VP9.num_ref_pics; ++i)
>-        rtp->codecHeader.VP9.pid_diff[i] = info->codecSpecific.VP9.p_diff[i];
>-      return;
>-    }
>-    case kVideoCodecH264:
>-      rtp->codec = kRtpVideoH264;
>-      rtp->codecHeader.H264.packetization_mode =
>-          info->codecSpecific.H264.packetization_mode;
>-      return;
>-    case kVideoCodecStereo: {
>-      rtp->codec = kRtpVideoStereo;
>-      RtpVideoCodecTypes associated_codec_type = kRtpVideoNone;
>-      switch (info->codecSpecific.stereo.associated_codec_type) {
>-        case kVideoCodecVP8:
>-          associated_codec_type = kRtpVideoVp8;
>-          break;
>-        case kVideoCodecVP9:
>-          associated_codec_type = kRtpVideoVp9;
>-          break;
>-        case kVideoCodecH264:
>-          associated_codec_type = kRtpVideoH264;
>-          break;
>-        default:
>-          RTC_NOTREACHED();
>-      }
>-      rtp->codecHeader.stereo.associated_codec_type = associated_codec_type;
>-      rtp->codecHeader.stereo.indices = info->codecSpecific.stereo.indices;
>-      return;
>-    }
>-    case kVideoCodecGeneric:
>-      rtp->codec = kRtpVideoGeneric;
>-      rtp->simulcastIdx = info->codecSpecific.generic.simulcast_idx;
>-      return;
>-    default:
>-      return;
>-  }
>-}
>-
>-}  // namespace
>-
>-// Currently only used if forced fallback for VP8 is enabled.
>-// Consider adding tl0idx and set for VP8 and VP9.
>-// Make picture id not codec specific.
>-class PayloadRouter::RtpPayloadParams final {
>- public:
>-  RtpPayloadParams(const uint32_t ssrc, const RtpPayloadState* state)
>-      : ssrc_(ssrc) {
>-    Random random(rtc::TimeMicros());
>-    state_.picture_id =
>-        state ? state->picture_id : (random.Rand<int16_t>() & 0x7FFF);
>-  }
>-  ~RtpPayloadParams() {}
>-
>-  void Set(RTPVideoHeader* rtp_video_header) {
>-    if (rtp_video_header->codec == kRtpVideoVp8 &&
>-        rtp_video_header->codecHeader.VP8.pictureId != kNoPictureId) {
>-      rtp_video_header->codecHeader.VP8.pictureId = state_.picture_id;
>-      state_.picture_id = (state_.picture_id + 1) & 0x7FFF;
>-    }
>-  }
>-
>-  uint32_t ssrc() const { return ssrc_; }
>-
>-  RtpPayloadState state() const { return state_; }
>-
>- private:
>-  const uint32_t ssrc_;
>-  RtpPayloadState state_;
>-};
>-
>-PayloadRouter::PayloadRouter(const std::vector<RtpRtcp*>& rtp_modules,
>-                             const std::vector<uint32_t>& ssrcs,
>-                             int payload_type,
>-                             const std::map<uint32_t, RtpPayloadState>& states)
>-    : active_(false),
>-      rtp_modules_(rtp_modules),
>-      payload_type_(payload_type),
>-      forced_fallback_enabled_((webrtc::field_trial::IsEnabled(
>-          "WebRTC-VP8-Forced-Fallback-Encoder-v2"))) {
>-  RTC_DCHECK_EQ(ssrcs.size(), rtp_modules.size());
>-  // SSRCs are assumed to be sorted in the same order as |rtp_modules|.
>-  for (uint32_t ssrc : ssrcs) {
>-    // Restore state if it previously existed.
>-    const RtpPayloadState* state = nullptr;
>-    auto it = states.find(ssrc);
>-    if (it != states.end()) {
>-      state = &it->second;
>-    }
>-    params_.push_back(RtpPayloadParams(ssrc, state));
>-  }
>-}
>-
>-PayloadRouter::~PayloadRouter() {}
>-
>-void PayloadRouter::SetActive(bool active) {
>-  rtc::CritScope lock(&crit_);
>-  if (active_ == active)
>-    return;
>-  active_ = active;
>-
>-  for (auto& module : rtp_modules_) {
>-    module->SetSendingStatus(active_);
>-    module->SetSendingMediaStatus(active_);
>-  }
>-}
>-
>-bool PayloadRouter::IsActive() {
>-  rtc::CritScope lock(&crit_);
>-  return active_ && !rtp_modules_.empty();
>-}
>-
>-std::map<uint32_t, RtpPayloadState> PayloadRouter::GetRtpPayloadStates() const {
>-  rtc::CritScope lock(&crit_);
>-  std::map<uint32_t, RtpPayloadState> payload_states;
>-  for (const auto& param : params_) {
>-    payload_states[param.ssrc()] = param.state();
>-  }
>-  return payload_states;
>-}
>-
>-EncodedImageCallback::Result PayloadRouter::OnEncodedImage(
>-    const EncodedImage& encoded_image,
>-    const CodecSpecificInfo* codec_specific_info,
>-    const RTPFragmentationHeader* fragmentation) {
>-  rtc::CritScope lock(&crit_);
>-  RTC_DCHECK(!rtp_modules_.empty());
>-  if (!active_)
>-    return Result(Result::ERROR_SEND_FAILED);
>-
>-  RTPVideoHeader rtp_video_header;
>-  memset(&rtp_video_header, 0, sizeof(RTPVideoHeader));
>-  if (codec_specific_info)
>-    CopyCodecSpecific(codec_specific_info, &rtp_video_header);
>-  rtp_video_header.rotation = encoded_image.rotation_;
>-  rtp_video_header.content_type = encoded_image.content_type_;
>-  if (encoded_image.timing_.flags != TimingFrameFlags::kInvalid &&
>-      encoded_image.timing_.flags != TimingFrameFlags::kNotTriggered) {
>-    rtp_video_header.video_timing.encode_start_delta_ms =
>-        VideoSendTiming::GetDeltaCappedMs(
>-            encoded_image.capture_time_ms_,
>-            encoded_image.timing_.encode_start_ms);
>-    rtp_video_header.video_timing.encode_finish_delta_ms =
>-        VideoSendTiming::GetDeltaCappedMs(
>-            encoded_image.capture_time_ms_,
>-            encoded_image.timing_.encode_finish_ms);
>-    rtp_video_header.video_timing.packetization_finish_delta_ms = 0;
>-    rtp_video_header.video_timing.pacer_exit_delta_ms = 0;
>-    rtp_video_header.video_timing.network_timestamp_delta_ms = 0;
>-    rtp_video_header.video_timing.network2_timestamp_delta_ms = 0;
>-    rtp_video_header.video_timing.flags = encoded_image.timing_.flags;
>-  } else {
>-    rtp_video_header.video_timing.flags = TimingFrameFlags::kInvalid;
>-  }
>-  rtp_video_header.playout_delay = encoded_image.playout_delay_;
>-
>-  int stream_index = rtp_video_header.simulcastIdx;
>-  RTC_DCHECK_LT(stream_index, rtp_modules_.size());
>-  if (forced_fallback_enabled_) {
>-    // Sets picture id. The SW and HW encoder have separate picture id
>-    // sequences, set picture id to not cause sequence discontinuties at encoder
>-    // changes.
>-    params_[stream_index].Set(&rtp_video_header);
>-  }
>-  uint32_t frame_id;
>-  bool send_result = rtp_modules_[stream_index]->SendOutgoingData(
>-      encoded_image._frameType, payload_type_, encoded_image._timeStamp,
>-      encoded_image.capture_time_ms_, encoded_image._buffer,
>-      encoded_image._length, fragmentation, &rtp_video_header, &frame_id);
>-  if (!send_result)
>-    return Result(Result::ERROR_SEND_FAILED);
>-
>-  return Result(Result::OK, frame_id);
>-}
>-
>-void PayloadRouter::OnBitrateAllocationUpdated(
>-    const BitrateAllocation& bitrate) {
>-  rtc::CritScope lock(&crit_);
>-  if (IsActive()) {
>-    if (rtp_modules_.size() == 1) {
>-      // If spatial scalability is enabled, it is covered by a single stream.
>-      rtp_modules_[0]->SetVideoBitrateAllocation(bitrate);
>-    } else {
>-      // Simulcast is in use, split the BitrateAllocation into one struct per
>-      // rtp stream, moving over the temporal layer allocation.
>-      for (size_t si = 0; si < rtp_modules_.size(); ++si) {
>-        // Don't send empty TargetBitrate messages on streams not being relayed.
>-        if (!bitrate.IsSpatialLayerUsed(si))
>-          break;
>-
>-        BitrateAllocation layer_bitrate;
>-        for (int tl = 0; tl < kMaxTemporalStreams; ++tl) {
>-          if (bitrate.HasBitrate(si, tl))
>-            layer_bitrate.SetBitrate(0, tl, bitrate.GetBitrate(si, tl));
>-        }
>-        rtp_modules_[si]->SetVideoBitrateAllocation(layer_bitrate);
>-      }
>-    }
>-  }
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/payload_router.h b/Source/ThirdParty/libwebrtc/Source/webrtc/video/payload_router.h
>deleted file mode 100644
>index 13b6cae7ce8..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/payload_router.h
>+++ /dev/null
>@@ -1,76 +0,0 @@
>-/*
>- *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#ifndef VIDEO_PAYLOAD_ROUTER_H_
>-#define VIDEO_PAYLOAD_ROUTER_H_
>-
>-#include <map>
>-#include <vector>
>-
>-#include "api/video_codecs/video_encoder.h"
>-#include "common_types.h"  // NOLINT(build/include)
>-#include "rtc_base/constructormagic.h"
>-#include "rtc_base/criticalsection.h"
>-#include "rtc_base/thread_annotations.h"
>-
>-namespace webrtc {
>-
>-class RTPFragmentationHeader;
>-class RtpRtcp;
>-struct RTPVideoHeader;
>-
>-// PayloadRouter routes outgoing data to the correct sending RTP module, based
>-// on the simulcast layer in RTPVideoHeader.
>-class PayloadRouter : public EncodedImageCallback {
>- public:
>-  // Rtp modules are assumed to be sorted in simulcast index order.
>-  PayloadRouter(const std::vector<RtpRtcp*>& rtp_modules,
>-                const std::vector<uint32_t>& ssrcs,
>-                int payload_type,
>-                const std::map<uint32_t, RtpPayloadState>& states);
>-  ~PayloadRouter();
>-
>-  // PayloadRouter will only route packets if being active, all packets will be
>-  // dropped otherwise.
>-  void SetActive(bool active);
>-  bool IsActive();
>-
>-  std::map<uint32_t, RtpPayloadState> GetRtpPayloadStates() const;
>-
>-  // Implements EncodedImageCallback.
>-  // Returns 0 if the packet was routed / sent, -1 otherwise.
>-  EncodedImageCallback::Result OnEncodedImage(
>-      const EncodedImage& encoded_image,
>-      const CodecSpecificInfo* codec_specific_info,
>-      const RTPFragmentationHeader* fragmentation) override;
>-
>-  void OnBitrateAllocationUpdated(const BitrateAllocation& bitrate);
>-
>- private:
>-  class RtpPayloadParams;
>-
>-  void UpdateModuleSendingState() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
>-
>-  rtc::CriticalSection crit_;
>-  bool active_ RTC_GUARDED_BY(crit_);
>-
>-  // Rtp modules are assumed to be sorted in simulcast index order. Not owned.
>-  const std::vector<RtpRtcp*> rtp_modules_;
>-  const int payload_type_;
>-
>-  const bool forced_fallback_enabled_;
>-  std::vector<RtpPayloadParams> params_ RTC_GUARDED_BY(crit_);
>-
>-  RTC_DISALLOW_COPY_AND_ASSIGN(PayloadRouter);
>-};
>-
>-}  // namespace webrtc
>-
>-#endif  // VIDEO_PAYLOAD_ROUTER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/payload_router_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/payload_router_unittest.cc
>deleted file mode 100644
>index d670124ddf2..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/payload_router_unittest.cc
>+++ /dev/null
>@@ -1,504 +0,0 @@
>-/*
>- *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
>- *
>- *  Use of this source code is governed by a BSD-style license
>- *  that can be found in the LICENSE file in the root of the source
>- *  tree. An additional intellectual property rights grant can be found
>- *  in the file PATENTS.  All contributing project authors may
>- *  be found in the AUTHORS file in the root of the source tree.
>- */
>-
>-#include <memory>
>-#include <string>
>-
>-#include "call/video_config.h"
>-#include "modules/rtp_rtcp/include/rtp_rtcp.h"
>-#include "modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
>-#include "modules/video_coding/include/video_codec_interface.h"
>-#include "test/field_trial.h"
>-#include "test/gmock.h"
>-#include "test/gtest.h"
>-#include "video/payload_router.h"
>-
>-using ::testing::_;
>-using ::testing::AnyNumber;
>-using ::testing::Invoke;
>-using ::testing::NiceMock;
>-using ::testing::Return;
>-using ::testing::Unused;
>-
>-namespace webrtc {
>-namespace {
>-const int8_t kPayloadType = 96;
>-const uint32_t kSsrc1 = 12345;
>-const uint32_t kSsrc2 = 23456;
>-const int16_t kPictureId = 123;
>-const int16_t kTl0PicIdx = 20;
>-const uint8_t kTemporalIdx = 1;
>-const int16_t kInitialPictureId1 = 222;
>-const int16_t kInitialPictureId2 = 44;
>-}  // namespace
>-
>-TEST(PayloadRouterTest, SendOnOneModule) {
>-  NiceMock<MockRtpRtcp> rtp;
>-  std::vector<RtpRtcp*> modules(1, &rtp);
>-
>-  uint8_t payload = 'a';
>-  EncodedImage encoded_image;
>-  encoded_image._timeStamp = 1;
>-  encoded_image.capture_time_ms_ = 2;
>-  encoded_image._frameType = kVideoFrameKey;
>-  encoded_image._buffer = &payload;
>-  encoded_image._length = 1;
>-
>-  PayloadRouter payload_router(modules, {kSsrc1}, kPayloadType, {});
>-
>-  EXPECT_CALL(rtp, SendOutgoingData(encoded_image._frameType, kPayloadType,
>-                                    encoded_image._timeStamp,
>-                                    encoded_image.capture_time_ms_, &payload,
>-                                    encoded_image._length, nullptr, _, _))
>-      .Times(0);
>-  EXPECT_NE(
>-      EncodedImageCallback::Result::OK,
>-      payload_router.OnEncodedImage(encoded_image, nullptr, nullptr).error);
>-
>-  payload_router.SetActive(true);
>-  EXPECT_CALL(rtp, SendOutgoingData(encoded_image._frameType, kPayloadType,
>-                                    encoded_image._timeStamp,
>-                                    encoded_image.capture_time_ms_, &payload,
>-                                    encoded_image._length, nullptr, _, _))
>-      .Times(1)
>-      .WillOnce(Return(true));
>-  EXPECT_EQ(
>-      EncodedImageCallback::Result::OK,
>-      payload_router.OnEncodedImage(encoded_image, nullptr, nullptr).error);
>-
>-  payload_router.SetActive(false);
>-  EXPECT_CALL(rtp, SendOutgoingData(encoded_image._frameType, kPayloadType,
>-                                    encoded_image._timeStamp,
>-                                    encoded_image.capture_time_ms_, &payload,
>-                                    encoded_image._length, nullptr, _, _))
>-      .Times(0);
>-  EXPECT_NE(
>-      EncodedImageCallback::Result::OK,
>-      payload_router.OnEncodedImage(encoded_image, nullptr, nullptr).error);
>-
>-  payload_router.SetActive(true);
>-  EXPECT_CALL(rtp, SendOutgoingData(encoded_image._frameType, kPayloadType,
>-                                    encoded_image._timeStamp,
>-                                    encoded_image.capture_time_ms_, &payload,
>-                                    encoded_image._length, nullptr, _, _))
>-      .Times(1)
>-      .WillOnce(Return(true));
>-  EXPECT_EQ(
>-      EncodedImageCallback::Result::OK,
>-      payload_router.OnEncodedImage(encoded_image, nullptr, nullptr).error);
>-}
>-
>-TEST(PayloadRouterTest, SendSimulcast) {
>-  NiceMock<MockRtpRtcp> rtp_1;
>-  NiceMock<MockRtpRtcp> rtp_2;
>-  std::vector<RtpRtcp*> modules = {&rtp_1, &rtp_2};
>-
>-  uint8_t payload = 'a';
>-  EncodedImage encoded_image;
>-  encoded_image._timeStamp = 1;
>-  encoded_image.capture_time_ms_ = 2;
>-  encoded_image._frameType = kVideoFrameKey;
>-  encoded_image._buffer = &payload;
>-  encoded_image._length = 1;
>-
>-  PayloadRouter payload_router(modules, {kSsrc1, kSsrc2}, kPayloadType, {});
>-
>-  CodecSpecificInfo codec_info_1;
>-  memset(&codec_info_1, 0, sizeof(CodecSpecificInfo));
>-  codec_info_1.codecType = kVideoCodecVP8;
>-  codec_info_1.codecSpecific.VP8.simulcastIdx = 0;
>-
>-  payload_router.SetActive(true);
>-  EXPECT_CALL(rtp_1, SendOutgoingData(encoded_image._frameType, kPayloadType,
>-                                      encoded_image._timeStamp,
>-                                      encoded_image.capture_time_ms_, &payload,
>-                                      encoded_image._length, nullptr, _, _))
>-      .Times(1)
>-      .WillOnce(Return(true));
>-  EXPECT_CALL(rtp_2, SendOutgoingData(_, _, _, _, _, _, _, _, _)).Times(0);
>-  EXPECT_EQ(EncodedImageCallback::Result::OK,
>-            payload_router.OnEncodedImage(encoded_image, &codec_info_1, nullptr)
>-                .error);
>-
>-  CodecSpecificInfo codec_info_2;
>-  memset(&codec_info_2, 0, sizeof(CodecSpecificInfo));
>-  codec_info_2.codecType = kVideoCodecVP8;
>-  codec_info_2.codecSpecific.VP8.simulcastIdx = 1;
>-
>-  EXPECT_CALL(rtp_2, SendOutgoingData(encoded_image._frameType, kPayloadType,
>-                                      encoded_image._timeStamp,
>-                                      encoded_image.capture_time_ms_, &payload,
>-                                      encoded_image._length, nullptr, _, _))
>-      .Times(1)
>-      .WillOnce(Return(true));
>-  EXPECT_CALL(rtp_1, SendOutgoingData(_, _, _, _, _, _, _, _, _))
>-      .Times(0);
>-  EXPECT_EQ(EncodedImageCallback::Result::OK,
>-            payload_router.OnEncodedImage(encoded_image, &codec_info_2, nullptr)
>-                .error);
>-
>-  // Inactive.
>-  payload_router.SetActive(false);
>-  EXPECT_CALL(rtp_1, SendOutgoingData(_, _, _, _, _, _, _, _, _))
>-      .Times(0);
>-  EXPECT_CALL(rtp_2, SendOutgoingData(_, _, _, _, _, _, _, _, _))
>-      .Times(0);
>-  EXPECT_NE(EncodedImageCallback::Result::OK,
>-            payload_router.OnEncodedImage(encoded_image, &codec_info_1, nullptr)
>-                .error);
>-  EXPECT_NE(EncodedImageCallback::Result::OK,
>-            payload_router.OnEncodedImage(encoded_image, &codec_info_2, nullptr)
>-                .error);
>-}
>-
>-TEST(PayloadRouterTest, SimulcastTargetBitrate) {
>-  NiceMock<MockRtpRtcp> rtp_1;
>-  NiceMock<MockRtpRtcp> rtp_2;
>-  std::vector<RtpRtcp*> modules = {&rtp_1, &rtp_2};
>-
>-  PayloadRouter payload_router(modules, {kSsrc1, kSsrc2}, kPayloadType, {});
>-  payload_router.SetActive(true);
>-
>-  BitrateAllocation bitrate;
>-  bitrate.SetBitrate(0, 0, 10000);
>-  bitrate.SetBitrate(0, 1, 20000);
>-  bitrate.SetBitrate(1, 0, 40000);
>-  bitrate.SetBitrate(1, 1, 80000);
>-
>-  BitrateAllocation layer0_bitrate;
>-  layer0_bitrate.SetBitrate(0, 0, 10000);
>-  layer0_bitrate.SetBitrate(0, 1, 20000);
>-
>-  BitrateAllocation layer1_bitrate;
>-  layer1_bitrate.SetBitrate(0, 0, 40000);
>-  layer1_bitrate.SetBitrate(0, 1, 80000);
>-
>-  EXPECT_CALL(rtp_1, SetVideoBitrateAllocation(layer0_bitrate)).Times(1);
>-  EXPECT_CALL(rtp_2, SetVideoBitrateAllocation(layer1_bitrate)).Times(1);
>-
>-  payload_router.OnBitrateAllocationUpdated(bitrate);
>-}
>-
>-TEST(PayloadRouterTest, SimulcastTargetBitrateWithInactiveStream) {
>-  // Set up two active rtp modules.
>-  NiceMock<MockRtpRtcp> rtp_1;
>-  NiceMock<MockRtpRtcp> rtp_2;
>-  std::vector<RtpRtcp*> modules = {&rtp_1, &rtp_2};
>-  PayloadRouter payload_router(modules, {kSsrc1, kSsrc2}, kPayloadType, {});
>-  payload_router.SetActive(true);
>-
>-  // Create bitrate allocation with bitrate only for the first stream.
>-  BitrateAllocation bitrate;
>-  bitrate.SetBitrate(0, 0, 10000);
>-  bitrate.SetBitrate(0, 1, 20000);
>-
>-  // Expect only the first rtp module to be asked to send a TargetBitrate
>-  // message. (No target bitrate with 0bps sent from the second one.)
>-  EXPECT_CALL(rtp_1, SetVideoBitrateAllocation(bitrate)).Times(1);
>-  EXPECT_CALL(rtp_2, SetVideoBitrateAllocation(_)).Times(0);
>-
>-  payload_router.OnBitrateAllocationUpdated(bitrate);
>-}
>-
>-TEST(PayloadRouterTest, SvcTargetBitrate) {
>-  NiceMock<MockRtpRtcp> rtp_1;
>-  std::vector<RtpRtcp*> modules = {&rtp_1};
>-  PayloadRouter payload_router(modules, {kSsrc1}, kPayloadType, {});
>-  payload_router.SetActive(true);
>-
>-  BitrateAllocation bitrate;
>-  bitrate.SetBitrate(0, 0, 10000);
>-  bitrate.SetBitrate(0, 1, 20000);
>-  bitrate.SetBitrate(1, 0, 40000);
>-  bitrate.SetBitrate(1, 1, 80000);
>-
>-  EXPECT_CALL(rtp_1, SetVideoBitrateAllocation(bitrate)).Times(1);
>-
>-  payload_router.OnBitrateAllocationUpdated(bitrate);
>-}
>-
>-TEST(PayloadRouterTest, InfoMappedToRtpVideoHeader_Vp8) {
>-  NiceMock<MockRtpRtcp> rtp1;
>-  NiceMock<MockRtpRtcp> rtp2;
>-  std::vector<RtpRtcp*> modules = {&rtp1, &rtp2};
>-  PayloadRouter payload_router(modules, {kSsrc1, kSsrc2}, kPayloadType, {});
>-  payload_router.SetActive(true);
>-
>-  EncodedImage encoded_image;
>-  encoded_image.rotation_ = kVideoRotation_90;
>-  encoded_image.content_type_ = VideoContentType::SCREENSHARE;
>-
>-  CodecSpecificInfo codec_info;
>-  memset(&codec_info, 0, sizeof(CodecSpecificInfo));
>-  codec_info.codecType = kVideoCodecVP8;
>-  codec_info.codecSpecific.VP8.simulcastIdx = 1;
>-  codec_info.codecSpecific.VP8.pictureId = kPictureId;
>-  codec_info.codecSpecific.VP8.temporalIdx = kTemporalIdx;
>-  codec_info.codecSpecific.VP8.tl0PicIdx = kTl0PicIdx;
>-  codec_info.codecSpecific.VP8.keyIdx = kNoKeyIdx;
>-  codec_info.codecSpecific.VP8.layerSync = true;
>-  codec_info.codecSpecific.VP8.nonReference = true;
>-
>-  EXPECT_CALL(rtp2, SendOutgoingData(_, _, _, _, _, _, nullptr, _, _))
>-      .WillOnce(Invoke([](Unused, Unused, Unused, Unused, Unused, Unused,
>-                          Unused, const RTPVideoHeader* header, Unused) {
>-        EXPECT_EQ(kVideoRotation_90, header->rotation);
>-        EXPECT_EQ(VideoContentType::SCREENSHARE, header->content_type);
>-        EXPECT_EQ(1, header->simulcastIdx);
>-        EXPECT_EQ(kRtpVideoVp8, header->codec);
>-        EXPECT_EQ(kPictureId, header->codecHeader.VP8.pictureId);
>-        EXPECT_EQ(kTemporalIdx, header->codecHeader.VP8.temporalIdx);
>-        EXPECT_EQ(kTl0PicIdx, header->codecHeader.VP8.tl0PicIdx);
>-        EXPECT_EQ(kNoKeyIdx, header->codecHeader.VP8.keyIdx);
>-        EXPECT_TRUE(header->codecHeader.VP8.layerSync);
>-        EXPECT_TRUE(header->codecHeader.VP8.nonReference);
>-        return true;
>-      }));
>-
>-  EXPECT_EQ(
>-      EncodedImageCallback::Result::OK,
>-      payload_router.OnEncodedImage(encoded_image, &codec_info, nullptr).error);
>-}
>-
>-TEST(PayloadRouterTest, InfoMappedToRtpVideoHeader_H264) {
>-  NiceMock<MockRtpRtcp> rtp1;
>-  std::vector<RtpRtcp*> modules = {&rtp1};
>-  PayloadRouter payload_router(modules, {kSsrc1}, kPayloadType, {});
>-  payload_router.SetActive(true);
>-
>-  EncodedImage encoded_image;
>-  CodecSpecificInfo codec_info;
>-  memset(&codec_info, 0, sizeof(CodecSpecificInfo));
>-  codec_info.codecType = kVideoCodecH264;
>-  codec_info.codecSpecific.H264.packetization_mode =
>-      H264PacketizationMode::SingleNalUnit;
>-
>-  EXPECT_CALL(rtp1, SendOutgoingData(_, _, _, _, _, _, nullptr, _, _))
>-      .WillOnce(Invoke([](Unused, Unused, Unused, Unused, Unused, Unused,
>-                          Unused, const RTPVideoHeader* header, Unused) {
>-        EXPECT_EQ(0, header->simulcastIdx);
>-        EXPECT_EQ(kRtpVideoH264, header->codec);
>-        EXPECT_EQ(H264PacketizationMode::SingleNalUnit,
>-                  header->codecHeader.H264.packetization_mode);
>-        return true;
>-      }));
>-
>-  EXPECT_EQ(
>-      EncodedImageCallback::Result::OK,
>-      payload_router.OnEncodedImage(encoded_image, &codec_info, nullptr).error);
>-}
>-
>-TEST(PayloadRouterTest, CreateWithNoPreviousStates) {
>-  NiceMock<MockRtpRtcp> rtp1;
>-  NiceMock<MockRtpRtcp> rtp2;
>-  std::vector<RtpRtcp*> modules = {&rtp1, &rtp2};
>-  PayloadRouter payload_router(modules, {kSsrc1, kSsrc2}, kPayloadType, {});
>-  payload_router.SetActive(true);
>-
>-  std::map<uint32_t, RtpPayloadState> initial_states =
>-      payload_router.GetRtpPayloadStates();
>-  EXPECT_EQ(2u, initial_states.size());
>-  EXPECT_NE(initial_states.find(kSsrc1), initial_states.end());
>-  EXPECT_NE(initial_states.find(kSsrc2), initial_states.end());
>-}
>-
>-TEST(PayloadRouterTest, CreateWithPreviousStates) {
>-  RtpPayloadState state1;
>-  state1.picture_id = kInitialPictureId1;
>-  RtpPayloadState state2;
>-  state2.picture_id = kInitialPictureId2;
>-  std::map<uint32_t, RtpPayloadState> states = {{kSsrc1, state1},
>-                                                {kSsrc2, state2}};
>-
>-  NiceMock<MockRtpRtcp> rtp1;
>-  NiceMock<MockRtpRtcp> rtp2;
>-  std::vector<RtpRtcp*> modules = {&rtp1, &rtp2};
>-  PayloadRouter payload_router(modules, {kSsrc1, kSsrc2}, kPayloadType, states);
>-  payload_router.SetActive(true);
>-
>-  std::map<uint32_t, RtpPayloadState> initial_states =
>-      payload_router.GetRtpPayloadStates();
>-  EXPECT_EQ(2u, initial_states.size());
>-  EXPECT_EQ(kInitialPictureId1, initial_states[kSsrc1].picture_id);
>-  EXPECT_EQ(kInitialPictureId2, initial_states[kSsrc2].picture_id);
>-}
>-
>-class PayloadRouterTest : public ::testing::Test {
>- public:
>-  explicit PayloadRouterTest(const std::string& field_trials)
>-      : override_field_trials_(field_trials) {}
>-  virtual ~PayloadRouterTest() {}
>-
>- protected:
>-  virtual void SetUp() { memset(&codec_info_, 0, sizeof(CodecSpecificInfo)); }
>-
>-  test::ScopedFieldTrials override_field_trials_;
>-  EncodedImage image_;
>-  CodecSpecificInfo codec_info_;
>-};
>-
>-class TestWithForcedFallbackDisabled : public PayloadRouterTest {
>- public:
>-  TestWithForcedFallbackDisabled()
>-      : PayloadRouterTest("WebRTC-VP8-Forced-Fallback-Encoder-v2/Disabled/") {}
>-};
>-
>-class TestWithForcedFallbackEnabled : public PayloadRouterTest {
>- public:
>-  TestWithForcedFallbackEnabled()
>-      : PayloadRouterTest(
>-            "WebRTC-VP8-Forced-Fallback-Encoder-v2/Enabled-1,2,3/") {}
>-};
>-
>-TEST_F(TestWithForcedFallbackDisabled, PictureIdIsNotChangedForVp8) {
>-  NiceMock<MockRtpRtcp> rtp;
>-  std::vector<RtpRtcp*> modules = {&rtp};
>-  PayloadRouter router(modules, {kSsrc1}, kPayloadType, {});
>-  router.SetActive(true);
>-
>-  codec_info_.codecType = kVideoCodecVP8;
>-  codec_info_.codecSpecific.VP8.pictureId = kPictureId;
>-
>-  EXPECT_CALL(rtp, SendOutgoingData(_, _, _, _, _, _, nullptr, _, _))
>-      .WillOnce(Invoke([](Unused, Unused, Unused, Unused, Unused, Unused,
>-                          Unused, const RTPVideoHeader* header, Unused) {
>-        EXPECT_EQ(kRtpVideoVp8, header->codec);
>-        EXPECT_EQ(kPictureId, header->codecHeader.VP8.pictureId);
>-        return true;
>-      }));
>-
>-  EXPECT_EQ(EncodedImageCallback::Result::OK,
>-            router.OnEncodedImage(image_, &codec_info_, nullptr).error);
>-}
>-
>-TEST_F(TestWithForcedFallbackEnabled, PictureIdIsSetForVp8) {
>-  RtpPayloadState state1;
>-  state1.picture_id = kInitialPictureId1;
>-  RtpPayloadState state2;
>-  state2.picture_id = kInitialPictureId2;
>-  std::map<uint32_t, RtpPayloadState> states = {{kSsrc1, state1},
>-                                                {kSsrc2, state2}};
>-
>-  NiceMock<MockRtpRtcp> rtp1;
>-  NiceMock<MockRtpRtcp> rtp2;
>-  std::vector<RtpRtcp*> modules = {&rtp1, &rtp2};
>-  PayloadRouter router(modules, {kSsrc1, kSsrc2}, kPayloadType, states);
>-  router.SetActive(true);
>-
>-  // OnEncodedImage, simulcastIdx: 0.
>-  codec_info_.codecType = kVideoCodecVP8;
>-  codec_info_.codecSpecific.VP8.pictureId = kPictureId;
>-  codec_info_.codecSpecific.VP8.simulcastIdx = 0;
>-
>-  EXPECT_CALL(rtp1, SendOutgoingData(_, _, _, _, _, _, nullptr, _, _))
>-      .WillOnce(Invoke([](Unused, Unused, Unused, Unused, Unused, Unused,
>-                          Unused, const RTPVideoHeader* header, Unused) {
>-        EXPECT_EQ(kRtpVideoVp8, header->codec);
>-        EXPECT_EQ(kInitialPictureId1, header->codecHeader.VP8.pictureId);
>-        return true;
>-      }));
>-
>-  EXPECT_EQ(EncodedImageCallback::Result::OK,
>-            router.OnEncodedImage(image_, &codec_info_, nullptr).error);
>-
>-  // OnEncodedImage, simulcastIdx: 1.
>-  codec_info_.codecSpecific.VP8.pictureId = kPictureId;
>-  codec_info_.codecSpecific.VP8.simulcastIdx = 1;
>-
>-  EXPECT_CALL(rtp2, SendOutgoingData(_, _, _, _, _, _, nullptr, _, _))
>-      .WillOnce(Invoke([](Unused, Unused, Unused, Unused, Unused, Unused,
>-                          Unused, const RTPVideoHeader* header, Unused) {
>-        EXPECT_EQ(kRtpVideoVp8, header->codec);
>-        EXPECT_EQ(kInitialPictureId2, header->codecHeader.VP8.pictureId);
>-        return true;
>-      }));
>-
>-  EXPECT_EQ(EncodedImageCallback::Result::OK,
>-            router.OnEncodedImage(image_, &codec_info_, nullptr).error);
>-
>-  // State should hold next picture id to use.
>-  states = router.GetRtpPayloadStates();
>-  EXPECT_EQ(2u, states.size());
>-  EXPECT_EQ(kInitialPictureId1 + 1, states[kSsrc1].picture_id);
>-  EXPECT_EQ(kInitialPictureId2 + 1, states[kSsrc2].picture_id);
>-}
>-
>-TEST_F(TestWithForcedFallbackEnabled, PictureIdWraps) {
>-  RtpPayloadState state1;
>-  state1.picture_id = kMaxTwoBytePictureId;
>-
>-  NiceMock<MockRtpRtcp> rtp;
>-  std::vector<RtpRtcp*> modules = {&rtp};
>-  PayloadRouter router(modules, {kSsrc1}, kPayloadType, {{kSsrc1, state1}});
>-  router.SetActive(true);
>-
>-  codec_info_.codecType = kVideoCodecVP8;
>-  codec_info_.codecSpecific.VP8.pictureId = kPictureId;
>-
>-  EXPECT_CALL(rtp, SendOutgoingData(_, _, _, _, _, _, nullptr, _, _))
>-      .WillOnce(Invoke([](Unused, Unused, Unused, Unused, Unused, Unused,
>-                          Unused, const RTPVideoHeader* header, Unused) {
>-        EXPECT_EQ(kRtpVideoVp8, header->codec);
>-        EXPECT_EQ(kMaxTwoBytePictureId, header->codecHeader.VP8.pictureId);
>-        return true;
>-      }));
>-
>-  EXPECT_EQ(EncodedImageCallback::Result::OK,
>-            router.OnEncodedImage(image_, &codec_info_, nullptr).error);
>-
>-  // State should hold next picture id to use.
>-  std::map<uint32_t, RtpPayloadState> states = router.GetRtpPayloadStates();
>-  EXPECT_EQ(1u, states.size());
>-  EXPECT_EQ(0, states[kSsrc1].picture_id);  // Wrapped.
>-}
>-
>-TEST_F(TestWithForcedFallbackEnabled, PictureIdIsNotSetIfNoPictureId) {
>-  NiceMock<MockRtpRtcp> rtp;
>-  std::vector<RtpRtcp*> modules = {&rtp};
>-  PayloadRouter router(modules, {kSsrc1}, kPayloadType, {});
>-  router.SetActive(true);
>-
>-  codec_info_.codecType = kVideoCodecVP8;
>-  codec_info_.codecSpecific.VP8.pictureId = kNoPictureId;
>-
>-  EXPECT_CALL(rtp, SendOutgoingData(_, _, _, _, _, _, nullptr, _, _))
>-      .WillOnce(Invoke([](Unused, Unused, Unused, Unused, Unused, Unused,
>-                          Unused, const RTPVideoHeader* header, Unused) {
>-        EXPECT_EQ(kRtpVideoVp8, header->codec);
>-        EXPECT_EQ(kNoPictureId, header->codecHeader.VP8.pictureId);
>-        return true;
>-      }));
>-
>-  EXPECT_EQ(EncodedImageCallback::Result::OK,
>-            router.OnEncodedImage(image_, &codec_info_, nullptr).error);
>-}
>-
>-TEST_F(TestWithForcedFallbackEnabled, PictureIdIsNotSetForVp9) {
>-  NiceMock<MockRtpRtcp> rtp;
>-  std::vector<RtpRtcp*> modules = {&rtp};
>-  PayloadRouter router(modules, {kSsrc1}, kPayloadType, {});
>-  router.SetActive(true);
>-
>-  codec_info_.codecType = kVideoCodecVP9;
>-  codec_info_.codecSpecific.VP9.picture_id = kPictureId;
>-
>-  EXPECT_CALL(rtp, SendOutgoingData(_, _, _, _, _, _, nullptr, _, _))
>-      .WillOnce(Invoke([](Unused, Unused, Unused, Unused, Unused, Unused,
>-                          Unused, const RTPVideoHeader* header, Unused) {
>-        EXPECT_EQ(kRtpVideoVp9, header->codec);
>-        EXPECT_EQ(kPictureId, header->codecHeader.VP9.picture_id);
>-        return true;
>-      }));
>-
>-  EXPECT_EQ(EncodedImageCallback::Result::OK,
>-            router.OnEncodedImage(image_, &codec_info_, nullptr).error);
>-}
>-
>-}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/picture_id_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/picture_id_tests.cc
>index f238d224abd..3c43d7d1122 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/picture_id_tests.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/picture_id_tests.cc
>@@ -10,11 +10,12 @@
> #include "media/engine/internalencoderfactory.h"
> #include "media/engine/simulcast_encoder_adapter.h"
> #include "modules/rtp_rtcp/source/rtp_format.h"
>+#include "modules/video_coding/codecs/vp8/include/vp8.h"
> #include "modules/video_coding/codecs/vp9/include/vp9.h"
> #include "rtc_base/numerics/safe_conversions.h"
> #include "rtc_base/numerics/sequence_number_util.h"
> #include "test/call_test.h"
>-#include "test/field_trial.h"
>+#include "test/function_video_encoder_factory.h"
> 
> namespace webrtc {
> namespace {
>@@ -28,25 +29,11 @@ const int kEncoderBitrateBps = 300000;
> const uint32_t kPictureIdWraparound = (1 << 15);
> const size_t kNumTemporalLayers[] = {1, 2, 3};
> 
>-const char kVp8ForcedFallbackEncoderEnabled[] =
>-    "WebRTC-VP8-Forced-Fallback-Encoder-v2/Enabled/";
>-
>-RtpVideoCodecTypes PayloadNameToRtpVideoCodecType(
>-    const std::string& payload_name) {
>-  if (payload_name == "VP8") {
>-    return kRtpVideoVp8;
>-  } else if (payload_name == "VP9") {
>-    return kRtpVideoVp9;
>-  } else {
>-    RTC_NOTREACHED();
>-    return kRtpVideoNone;
>-  }
>-}
> }  // namespace
> 
> class PictureIdObserver : public test::RtpRtcpObserver {
>  public:
>-  explicit PictureIdObserver(RtpVideoCodecTypes codec_type)
>+  explicit PictureIdObserver(VideoCodecType codec_type)
>       : test::RtpRtcpObserver(test::CallTest::kDefaultTimeoutMs),
>         codec_type_(codec_type),
>         max_expected_picture_id_gap_(0),
>@@ -109,22 +96,19 @@ class PictureIdObserver : public test::RtpRtcpObserver {
>         &parsed_payload, &packet[header.headerLength], payload_length));
> 
>     switch (codec_type_) {
>-      case kRtpVideoVp8:
>-        parsed->picture_id =
>-            parsed_payload.type.Video.codecHeader.VP8.pictureId;
>-        parsed->tl0_pic_idx =
>-            parsed_payload.type.Video.codecHeader.VP8.tl0PicIdx;
>-        parsed->temporal_idx =
>-            parsed_payload.type.Video.codecHeader.VP8.temporalIdx;
>+      case kVideoCodecVP8:
>+        parsed->picture_id = parsed_payload.video_header().vp8().pictureId;
>+        parsed->tl0_pic_idx = parsed_payload.video_header().vp8().tl0PicIdx;
>+        parsed->temporal_idx = parsed_payload.video_header().vp8().temporalIdx;
>         break;
>-      case kRtpVideoVp9:
>-        parsed->picture_id =
>-            parsed_payload.type.Video.codecHeader.VP9.picture_id;
>-        parsed->tl0_pic_idx =
>-            parsed_payload.type.Video.codecHeader.VP9.tl0_pic_idx;
>-        parsed->temporal_idx =
>-            parsed_payload.type.Video.codecHeader.VP9.temporal_idx;
>+      case kVideoCodecVP9: {
>+        const auto& vp9_header = absl::get<RTPVideoHeaderVP9>(
>+            parsed_payload.video_header().video_type_header);
>+        parsed->picture_id = vp9_header.picture_id;
>+        parsed->tl0_pic_idx = vp9_header.tl0_pic_idx;
>+        parsed->temporal_idx = vp9_header.temporal_idx;
>         break;
>+      }
>       default:
>         RTC_NOTREACHED();
>         break;
>@@ -214,7 +198,7 @@ class PictureIdObserver : public test::RtpRtcpObserver {
>   }
> 
>   rtc::CriticalSection crit_;
>-  const RtpVideoCodecTypes codec_type_;
>+  const VideoCodecType codec_type_;
>   std::map<uint32_t, ParsedPacket> last_observed_packet_ RTC_GUARDED_BY(crit_);
>   std::map<uint32_t, size_t> num_packets_sent_ RTC_GUARDED_BY(crit_);
>   int max_expected_picture_id_gap_ RTC_GUARDED_BY(crit_);
>@@ -224,17 +208,11 @@ class PictureIdObserver : public test::RtpRtcpObserver {
> };
> 
> class PictureIdTest : public test::CallTest,
>-                      public ::testing::WithParamInterface<
>-                          ::testing::tuple<std::string, size_t>> {
>+                      public ::testing::WithParamInterface<size_t> {
>  public:
>-  PictureIdTest()
>-      : scoped_field_trial_(::testing::get<0>(GetParam())),
>-        num_temporal_layers_(::testing::get<1>(GetParam())) {}
>+  PictureIdTest() : num_temporal_layers_(GetParam()) {}
> 
>   virtual ~PictureIdTest() {
>-    EXPECT_EQ(nullptr, video_send_stream_);
>-    EXPECT_TRUE(video_receive_streams_.empty());
>-
>     task_queue_.SendTask([this]() {
>       send_transport_.reset();
>       receive_transport_.reset();
>@@ -242,23 +220,21 @@ class PictureIdTest : public test::CallTest,
>     });
>   }
> 
>-  void SetupEncoder(VideoEncoder* encoder, const std::string& payload_name);
>+  void SetupEncoder(VideoEncoderFactory* encoder_factory,
>+                    const std::string& payload_name);
>   void TestPictureIdContinuousAfterReconfigure(
>       const std::vector<int>& ssrc_counts);
>   void TestPictureIdIncreaseAfterRecreateStreams(
>       const std::vector<int>& ssrc_counts);
> 
>  private:
>-  test::ScopedFieldTrials scoped_field_trial_;
>   const size_t num_temporal_layers_;
>   std::unique_ptr<PictureIdObserver> observer_;
> };
> 
>-INSTANTIATE_TEST_CASE_P(
>-    TestWithForcedFallbackEncoderEnabled,
>-    PictureIdTest,
>-    ::testing::Combine(::testing::Values(kVp8ForcedFallbackEncoderEnabled, ""),
>-                       ::testing::ValuesIn(kNumTemporalLayers)));
>+INSTANTIATE_TEST_CASE_P(TemporalLayers,
>+                        PictureIdTest,
>+                        ::testing::ValuesIn(kNumTemporalLayers));
> 
> // Use a special stream factory to ensure that all simulcast streams are being
> // sent.
>@@ -278,15 +254,15 @@ class VideoStreamFactory
> 
>     // Use the same total bitrates when sending a single stream to avoid
>     // lowering the bitrate estimate and requiring a subsequent rampup.
>-    const int encoder_stream_bps = kEncoderBitrateBps / rtc::checked_cast<int>(
>-        encoder_config.number_of_streams);
>+    const int encoder_stream_bps =
>+        kEncoderBitrateBps /
>+        rtc::checked_cast<int>(encoder_config.number_of_streams);
> 
>     for (size_t i = 0; i < encoder_config.number_of_streams; ++i) {
>       streams[i].min_bitrate_bps = encoder_stream_bps;
>       streams[i].target_bitrate_bps = encoder_stream_bps;
>       streams[i].max_bitrate_bps = encoder_stream_bps;
>-      streams[i].temporal_layer_thresholds_bps.resize(num_of_temporal_layers_ -
>-                                                      1);
>+      streams[i].num_temporal_layers = num_of_temporal_layers_;
>       // test::CreateVideoStreams does not return frame sizes for the lower
>       // streams that are accepted by VP8Impl::InitEncode.
>       // TODO(brandtr): Fix the problem in test::CreateVideoStreams, rather
>@@ -303,14 +279,13 @@ class VideoStreamFactory
>   const size_t num_of_temporal_layers_;
> };
> 
>-void PictureIdTest::SetupEncoder(VideoEncoder* encoder,
>+void PictureIdTest::SetupEncoder(VideoEncoderFactory* encoder_factory,
>                                  const std::string& payload_name) {
>   observer_.reset(
>-      new PictureIdObserver(PayloadNameToRtpVideoCodecType(payload_name)));
>+      new PictureIdObserver(PayloadStringToCodecType(payload_name)));
> 
>-  task_queue_.SendTask([this, &encoder, payload_name]() {
>-    Call::Config config(event_log_.get());
>-    CreateCalls(config, config);
>+  task_queue_.SendTask([this, encoder_factory, payload_name]() {
>+    CreateCalls();
> 
>     send_transport_.reset(new test::PacketTransport(
>         &task_queue_, sender_call_.get(), observer_.get(),
>@@ -318,11 +293,13 @@ void PictureIdTest::SetupEncoder(VideoEncoder* encoder,
>         FakeNetworkPipe::Config()));
> 
>     CreateSendConfig(kNumSimulcastStreams, 0, 0, send_transport_.get());
>-    video_send_config_.encoder_settings.encoder = encoder;
>-    video_send_config_.encoder_settings.payload_name = payload_name;
>-    video_encoder_config_.video_stream_factory =
>+    GetVideoSendConfig()->encoder_settings.encoder_factory = encoder_factory;
>+    GetVideoSendConfig()->rtp.payload_name = payload_name;
>+    GetVideoEncoderConfig()->codec_type =
>+        PayloadStringToCodecType(payload_name);
>+    GetVideoEncoderConfig()->video_stream_factory =
>         new rtc::RefCountedObject<VideoStreamFactory>(num_temporal_layers_);
>-    video_encoder_config_.number_of_streams = 1;
>+    GetVideoEncoderConfig()->number_of_streams = 1;
>   });
> }
> 
>@@ -342,12 +319,13 @@ void PictureIdTest::TestPictureIdContinuousAfterReconfigure(
>   // Expect continuously increasing picture id, equivalent to no gaps.
>   observer_->SetMaxExpectedPictureIdGap(0);
>   for (int ssrc_count : ssrc_counts) {
>-    video_encoder_config_.number_of_streams = ssrc_count;
>+    GetVideoEncoderConfig()->number_of_streams = ssrc_count;
>     observer_->SetExpectedSsrcs(ssrc_count);
>     observer_->ResetObservedSsrcs();
>     // Make sure the picture_id sequence is continuous on reinit and recreate.
>     task_queue_.SendTask([this]() {
>-      video_send_stream_->ReconfigureVideoEncoder(video_encoder_config_.Copy());
>+      GetVideoSendStream()->ReconfigureVideoEncoder(
>+          GetVideoEncoderConfig()->Copy());
>     });
>     EXPECT_TRUE(observer_->Wait()) << "Timed out waiting for packets.";
>   }
>@@ -377,15 +355,14 @@ void PictureIdTest::TestPictureIdIncreaseAfterRecreateStreams(
>   for (int ssrc_count : ssrc_counts) {
>     task_queue_.SendTask([this, &ssrc_count]() {
>       frame_generator_capturer_->Stop();
>-      sender_call_->DestroyVideoSendStream(video_send_stream_);
>+      DestroyVideoSendStreams();
> 
>-      video_encoder_config_.number_of_streams = ssrc_count;
>+      GetVideoEncoderConfig()->number_of_streams = ssrc_count;
>       observer_->SetExpectedSsrcs(ssrc_count);
>       observer_->ResetObservedSsrcs();
> 
>-      video_send_stream_ = sender_call_->CreateVideoSendStream(
>-          video_send_config_.Copy(), video_encoder_config_.Copy());
>-      video_send_stream_->Start();
>+      CreateVideoSendStreams();
>+      GetVideoSendStream()->Start();
>       CreateFrameGeneratorCapturer(kFrameRate, kFrameMaxWidth, kFrameMaxHeight);
>       frame_generator_capturer_->Start();
>     });
>@@ -400,60 +377,67 @@ void PictureIdTest::TestPictureIdIncreaseAfterRecreateStreams(
> }
> 
> TEST_P(PictureIdTest, ContinuousAfterReconfigureVp8) {
>-  std::unique_ptr<VideoEncoder> encoder(VP8Encoder::Create());
>-  SetupEncoder(encoder.get(), "VP8");
>+  test::FunctionVideoEncoderFactory encoder_factory(
>+      []() { return VP8Encoder::Create(); });
>+  SetupEncoder(&encoder_factory, "VP8");
>   TestPictureIdContinuousAfterReconfigure({1, 3, 3, 1, 1});
> }
> 
> TEST_P(PictureIdTest, IncreasingAfterRecreateStreamVp8) {
>-  std::unique_ptr<VideoEncoder> encoder(VP8Encoder::Create());
>-  SetupEncoder(encoder.get(), "VP8");
>+  test::FunctionVideoEncoderFactory encoder_factory(
>+      []() { return VP8Encoder::Create(); });
>+  SetupEncoder(&encoder_factory, "VP8");
>   TestPictureIdIncreaseAfterRecreateStreams({1, 3, 3, 1, 1});
> }
> 
> TEST_P(PictureIdTest, ContinuousAfterStreamCountChangeVp8) {
>-  std::unique_ptr<VideoEncoder> encoder(VP8Encoder::Create());
>+  test::FunctionVideoEncoderFactory encoder_factory(
>+      []() { return VP8Encoder::Create(); });
>   // Make sure that the picture id is not reset if the stream count goes
>   // down and then up.
>-  SetupEncoder(encoder.get(), "VP8");
>+  SetupEncoder(&encoder_factory, "VP8");
>   TestPictureIdContinuousAfterReconfigure({3, 1, 3});
> }
> 
> TEST_P(PictureIdTest, ContinuousAfterReconfigureSimulcastEncoderAdapter) {
>   InternalEncoderFactory internal_encoder_factory;
>-  SimulcastEncoderAdapter simulcast_encoder_adapter(&internal_encoder_factory);
>-  SetupEncoder(&simulcast_encoder_adapter, "VP8");
>+  test::FunctionVideoEncoderFactory encoder_factory(
>+      [&internal_encoder_factory]() {
>+        return absl::make_unique<SimulcastEncoderAdapter>(
>+            &internal_encoder_factory, SdpVideoFormat("VP8"));
>+      });
>+  SetupEncoder(&encoder_factory, "VP8");
>   TestPictureIdContinuousAfterReconfigure({1, 3, 3, 1, 1});
> }
> 
> TEST_P(PictureIdTest, IncreasingAfterRecreateStreamSimulcastEncoderAdapter) {
>   InternalEncoderFactory internal_encoder_factory;
>-  SimulcastEncoderAdapter simulcast_encoder_adapter(&internal_encoder_factory);
>-  SetupEncoder(&simulcast_encoder_adapter, "VP8");
>+  test::FunctionVideoEncoderFactory encoder_factory(
>+      [&internal_encoder_factory]() {
>+        return absl::make_unique<SimulcastEncoderAdapter>(
>+            &internal_encoder_factory, SdpVideoFormat("VP8"));
>+      });
>+  SetupEncoder(&encoder_factory, "VP8");
>   TestPictureIdIncreaseAfterRecreateStreams({1, 3, 3, 1, 1});
> }
> 
>-// When using the simulcast encoder adapter, the picture id is randomly set
>-// when the ssrc count is reduced and then increased. This means that we are
>-// not spec compliant in that particular case.
> TEST_P(PictureIdTest, ContinuousAfterStreamCountChangeSimulcastEncoderAdapter) {
>-  // If forced fallback is enabled, the picture id is set in the PayloadRouter
>-  // and the sequence should be continuous.
>-  if (::testing::get<0>(GetParam()) == kVp8ForcedFallbackEncoderEnabled &&
>-      ::testing::get<1>(GetParam()) == 1) {  // No temporal layers.
>-    InternalEncoderFactory internal_encoder_factory;
>-    SimulcastEncoderAdapter simulcast_encoder_adapter(
>-        &internal_encoder_factory);
>-    // Make sure that the picture id is not reset if the stream count goes
>-    // down and then up.
>-    SetupEncoder(&simulcast_encoder_adapter, "VP8");
>-    TestPictureIdContinuousAfterReconfigure({3, 1, 3});
>-  }
>+  InternalEncoderFactory internal_encoder_factory;
>+  test::FunctionVideoEncoderFactory encoder_factory(
>+      [&internal_encoder_factory]() {
>+        return absl::make_unique<SimulcastEncoderAdapter>(
>+            &internal_encoder_factory, SdpVideoFormat("VP8"));
>+      });
>+  // Make sure that the picture id is not reset if the stream count goes
>+  // down and then up.
>+  SetupEncoder(&encoder_factory, "VP8");
>+  TestPictureIdContinuousAfterReconfigure({3, 1, 3});
> }
> 
> TEST_P(PictureIdTest, IncreasingAfterRecreateStreamVp9) {
>-  std::unique_ptr<VideoEncoder> encoder(VP9Encoder::Create());
>-  SetupEncoder(encoder.get(), "VP9");
>+  test::FunctionVideoEncoderFactory encoder_factory(
>+      []() { return VP9Encoder::Create(); });
>+  SetupEncoder(&encoder_factory, "VP9");
>   TestPictureIdIncreaseAfterRecreateStreams({1, 1});
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/quality_scaling_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/quality_scaling_tests.cc
>new file mode 100644
>index 00000000000..bb12edcc82c
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/quality_scaling_tests.cc
>@@ -0,0 +1,263 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <string>
>+
>+#include "media/engine/internalencoderfactory.h"
>+#include "modules/video_coding/codecs/h264/include/h264.h"
>+#include "modules/video_coding/codecs/vp8/include/vp8.h"
>+#include "modules/video_coding/codecs/vp9/include/vp9.h"
>+#include "test/call_test.h"
>+#include "test/field_trial.h"
>+#include "test/frame_generator_capturer.h"
>+#include "test/function_video_encoder_factory.h"
>+
>+namespace webrtc {
>+namespace {
>+constexpr int kWidth = 1280;
>+constexpr int kHeight = 720;
>+constexpr int kLowStartBps = 100000;
>+constexpr int kHighStartBps = 600000;
>+constexpr size_t kTimeoutMs = 10000;  // Some tests are expected to time out.
>+
>+void SetEncoderSpecific(VideoEncoderConfig* encoder_config,
>+                        VideoCodecType type,
>+                        bool automatic_resize,
>+                        bool frame_dropping) {
>+  if (type == kVideoCodecVP8) {
>+    VideoCodecVP8 vp8 = VideoEncoder::GetDefaultVp8Settings();
>+    vp8.automaticResizeOn = automatic_resize;
>+    vp8.frameDroppingOn = frame_dropping;
>+    encoder_config->encoder_specific_settings = new rtc::RefCountedObject<
>+        VideoEncoderConfig::Vp8EncoderSpecificSettings>(vp8);
>+  } else if (type == kVideoCodecVP9) {
>+    VideoCodecVP9 vp9 = VideoEncoder::GetDefaultVp9Settings();
>+    vp9.automaticResizeOn = automatic_resize;
>+    vp9.frameDroppingOn = frame_dropping;
>+    encoder_config->encoder_specific_settings = new rtc::RefCountedObject<
>+        VideoEncoderConfig::Vp9EncoderSpecificSettings>(vp9);
>+  } else if (type == kVideoCodecH264) {
>+    VideoCodecH264 h264 = VideoEncoder::GetDefaultH264Settings();
>+    h264.frameDroppingOn = frame_dropping;
>+    encoder_config->encoder_specific_settings = new rtc::RefCountedObject<
>+        VideoEncoderConfig::H264EncoderSpecificSettings>(h264);
>+  }
>+}
>+}  // namespace
>+
>+class QualityScalingTest : public test::CallTest {
>+ protected:
>+  void RunTest(VideoEncoderFactory* encoder_factory,
>+               const std::string& payload_name,
>+               int start_bps,
>+               bool automatic_resize,
>+               bool frame_dropping,
>+               bool expect_adaptation);
>+
>+  const std::string kPrefix = "WebRTC-Video-QualityScaling/Enabled-";
>+  const std::string kEnd = ",0,0,0.9995,0.9999,1/";
>+};
>+
>+void QualityScalingTest::RunTest(VideoEncoderFactory* encoder_factory,
>+                                 const std::string& payload_name,
>+                                 int start_bps,
>+                                 bool automatic_resize,
>+                                 bool frame_dropping,
>+                                 bool expect_adaptation) {
>+  class ScalingObserver
>+      : public test::SendTest,
>+        public test::FrameGeneratorCapturer::SinkWantsObserver {
>+   public:
>+    ScalingObserver(VideoEncoderFactory* encoder_factory,
>+                    const std::string& payload_name,
>+                    int start_bps,
>+                    bool automatic_resize,
>+                    bool frame_dropping,
>+                    bool expect_adaptation)
>+        : SendTest(expect_adaptation ? kDefaultTimeoutMs : kTimeoutMs),
>+          encoder_factory_(encoder_factory),
>+          payload_name_(payload_name),
>+          start_bps_(start_bps),
>+          automatic_resize_(automatic_resize),
>+          frame_dropping_(frame_dropping),
>+          expect_adaptation_(expect_adaptation) {}
>+
>+   private:
>+    void OnFrameGeneratorCapturerCreated(
>+        test::FrameGeneratorCapturer* frame_generator_capturer) override {
>+      frame_generator_capturer->SetSinkWantsObserver(this);
>+      // Set initial resolution.
>+      frame_generator_capturer->ChangeResolution(kWidth, kHeight);
>+    }
>+
>+    // Called when FrameGeneratorCapturer::AddOrUpdateSink is called.
>+    void OnSinkWantsChanged(rtc::VideoSinkInterface<VideoFrame>* sink,
>+                            const rtc::VideoSinkWants& wants) override {
>+      EXPECT_LT(wants.max_pixel_count, kWidth * kHeight) << "Not a downscale.";
>+      observation_complete_.Set();
>+    }
>+    void ModifySenderCallConfig(Call::Config* config) override {
>+      config->bitrate_config.start_bitrate_bps = start_bps_;
>+    }
>+
>+    void ModifyVideoConfigs(
>+        VideoSendStream::Config* send_config,
>+        std::vector<VideoReceiveStream::Config>* receive_configs,
>+        VideoEncoderConfig* encoder_config) override {
>+      send_config->encoder_settings.encoder_factory = encoder_factory_;
>+      send_config->rtp.payload_name = payload_name_;
>+      send_config->rtp.payload_type = kVideoSendPayloadType;
>+      const VideoCodecType codec_type = PayloadStringToCodecType(payload_name_);
>+      encoder_config->codec_type = codec_type;
>+      encoder_config->max_bitrate_bps = start_bps_;
>+      SetEncoderSpecific(encoder_config, codec_type, automatic_resize_,
>+                         frame_dropping_);
>+    }
>+
>+    void PerformTest() override {
>+      EXPECT_EQ(expect_adaptation_, Wait())
>+          << "Timed out while waiting for a scale down.";
>+    }
>+
>+    VideoEncoderFactory* const encoder_factory_;
>+    const std::string payload_name_;
>+    const int start_bps_;
>+    const bool automatic_resize_;
>+    const bool frame_dropping_;
>+    const bool expect_adaptation_;
>+  } test(encoder_factory, payload_name, start_bps, automatic_resize,
>+         frame_dropping, expect_adaptation);
>+
>+  RunBaseTest(&test);
>+}
>+
>+TEST_F(QualityScalingTest, AdaptsDownForHighQp_Vp8) {
>+  // VP8 QP thresholds, low:1, high:1 -> high QP.
>+  test::ScopedFieldTrials field_trials(kPrefix + "1,1,0,0,0,0" + kEnd);
>+
>+  // QualityScaler enabled.
>+  const bool kAutomaticResize = true;
>+  const bool kFrameDropping = true;
>+  const bool kExpectAdapt = true;
>+
>+  test::FunctionVideoEncoderFactory encoder_factory(
>+      []() { return VP8Encoder::Create(); });
>+  RunTest(&encoder_factory, "VP8", kHighStartBps, kAutomaticResize,
>+          kFrameDropping, kExpectAdapt);
>+}
>+
>+TEST_F(QualityScalingTest, NoAdaptDownForHighQpWithResizeOff_Vp8) {
>+  // VP8 QP thresholds, low:1, high:1 -> high QP.
>+  test::ScopedFieldTrials field_trials(kPrefix + "1,1,0,0,0,0" + kEnd);
>+
>+  // QualityScaler disabled.
>+  const bool kAutomaticResize = false;
>+  const bool kFrameDropping = true;
>+  const bool kExpectAdapt = false;
>+
>+  test::FunctionVideoEncoderFactory encoder_factory(
>+      []() { return VP8Encoder::Create(); });
>+  RunTest(&encoder_factory, "VP8", kHighStartBps, kAutomaticResize,
>+          kFrameDropping, kExpectAdapt);
>+}
>+
>+TEST_F(QualityScalingTest, NoAdaptDownForHighQpWithFrameDroppingOff_Vp8) {
>+  // VP8 QP thresholds, low:1, high:1 -> high QP.
>+  test::ScopedFieldTrials field_trials(kPrefix + "1,1,0,0,0,0" + kEnd);
>+
>+  // QualityScaler disabled.
>+  const bool kAutomaticResize = true;
>+  const bool kFrameDropping = false;
>+  const bool kExpectAdapt = false;
>+
>+  test::FunctionVideoEncoderFactory encoder_factory(
>+      []() { return VP8Encoder::Create(); });
>+  RunTest(&encoder_factory, "VP8", kHighStartBps, kAutomaticResize,
>+          kFrameDropping, kExpectAdapt);
>+}
>+
>+TEST_F(QualityScalingTest, NoAdaptDownForNormalQp_Vp8) {
>+  // VP8 QP thresholds, low:1, high:127 -> normal QP.
>+  test::ScopedFieldTrials field_trials(kPrefix + "1,127,0,0,0,0" + kEnd);
>+
>+  // QualityScaler enabled.
>+  const bool kAutomaticResize = true;
>+  const bool kFrameDropping = true;
>+  const bool kExpectAdapt = false;
>+
>+  test::FunctionVideoEncoderFactory encoder_factory(
>+      []() { return VP8Encoder::Create(); });
>+  RunTest(&encoder_factory, "VP8", kHighStartBps, kAutomaticResize,
>+          kFrameDropping, kExpectAdapt);
>+}
>+
>+TEST_F(QualityScalingTest, AdaptsDownForLowStartBitrate) {
>+  // VP8 QP thresholds, low:1, high:127 -> normal QP.
>+  test::ScopedFieldTrials field_trials(kPrefix + "1,127,0,0,0,0" + kEnd);
>+
>+  // QualityScaler enabled.
>+  const bool kAutomaticResize = true;
>+  const bool kFrameDropping = true;
>+  const bool kExpectAdapt = true;
>+
>+  test::FunctionVideoEncoderFactory encoder_factory(
>+      []() { return VP8Encoder::Create(); });
>+  RunTest(&encoder_factory, "VP8", kLowStartBps, kAutomaticResize,
>+          kFrameDropping, kExpectAdapt);
>+}
>+
>+TEST_F(QualityScalingTest, NoAdaptDownForLowStartBitrateWithScalingOff) {
>+  // VP8 QP thresholds, low:1, high:127 -> normal QP.
>+  test::ScopedFieldTrials field_trials(kPrefix + "1,127,0,0,0,0" + kEnd);
>+
>+  // QualityScaler disabled.
>+  const bool kAutomaticResize = false;
>+  const bool kFrameDropping = true;
>+  const bool kExpectAdapt = false;
>+
>+  test::FunctionVideoEncoderFactory encoder_factory(
>+      []() { return VP8Encoder::Create(); });
>+  RunTest(&encoder_factory, "VP8", kLowStartBps, kAutomaticResize,
>+          kFrameDropping, kExpectAdapt);
>+}
>+
>+TEST_F(QualityScalingTest, NoAdaptDownForHighQp_Vp9) {
>+  // VP9 QP thresholds, low:1, high:1 -> high QP.
>+  test::ScopedFieldTrials field_trials(kPrefix + "0,0,1,1,0,0" + kEnd);
>+
>+  // QualityScaler always disabled.
>+  const bool kAutomaticResize = true;
>+  const bool kFrameDropping = true;
>+  const bool kExpectAdapt = false;
>+
>+  test::FunctionVideoEncoderFactory encoder_factory(
>+      []() { return VP9Encoder::Create(); });
>+  RunTest(&encoder_factory, "VP9", kHighStartBps, kAutomaticResize,
>+          kFrameDropping, kExpectAdapt);
>+}
>+
>+#if defined(WEBRTC_USE_H264)
>+TEST_F(QualityScalingTest, AdaptsDownForHighQp_H264) {
>+  // H264 QP thresholds, low:1, high:1 -> high QP.
>+  test::ScopedFieldTrials field_trials(kPrefix + "0,0,0,0,1,1" + kEnd);
>+
>+  // QualityScaler always enabled.
>+  const bool kAutomaticResize = false;
>+  const bool kFrameDropping = false;
>+  const bool kExpectAdapt = true;
>+
>+  test::FunctionVideoEncoderFactory encoder_factory(
>+      []() { return H264Encoder::Create(cricket::VideoCodec("H264")); });
>+  RunTest(&encoder_factory, "H264", kHighStartBps, kAutomaticResize,
>+          kFrameDropping, kExpectAdapt);
>+}
>+#endif  // defined(WEBRTC_USE_H264)
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/quality_threshold.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/quality_threshold.cc
>index fc8787a3ff4..2b2999280c3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/quality_threshold.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/quality_threshold.cc
>@@ -74,13 +74,13 @@ void QualityThreshold::AddMeasurement(int measurement) {
>   }
> }
> 
>-rtc::Optional<bool> QualityThreshold::IsHigh() const {
>+absl::optional<bool> QualityThreshold::IsHigh() const {
>   return is_high_;
> }
> 
>-rtc::Optional<double> QualityThreshold::CalculateVariance() const {
>+absl::optional<double> QualityThreshold::CalculateVariance() const {
>   if (until_full_ > 0) {
>-    return rtc::nullopt;
>+    return absl::nullopt;
>   }
> 
>   double variance = 0;
>@@ -91,11 +91,11 @@ rtc::Optional<double> QualityThreshold::CalculateVariance() const {
>   return variance / (max_measurements_ - 1);
> }
> 
>-rtc::Optional<double> QualityThreshold::FractionHigh(
>+absl::optional<double> QualityThreshold::FractionHigh(
>     int min_required_samples) const {
>   RTC_DCHECK_GT(min_required_samples, 0);
>   if (num_certain_states_ < min_required_samples)
>-    return rtc::nullopt;
>+    return absl::nullopt;
> 
>   return static_cast<double>(num_high_states_) / num_certain_states_;
> }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/quality_threshold.h b/Source/ThirdParty/libwebrtc/Source/webrtc/video/quality_threshold.h
>index 519e2c33def..4fad555ce6f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/quality_threshold.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/quality_threshold.h
>@@ -13,7 +13,7 @@
> 
> #include <memory>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> 
> namespace webrtc {
> 
>@@ -27,9 +27,9 @@ class QualityThreshold {
>                    int max_measurements);
> 
>   void AddMeasurement(int measurement);
>-  rtc::Optional<bool> IsHigh() const;
>-  rtc::Optional<double> CalculateVariance() const;
>-  rtc::Optional<double> FractionHigh(int min_required_samples) const;
>+  absl::optional<bool> IsHigh() const;
>+  absl::optional<double> CalculateVariance() const;
>+  absl::optional<double> FractionHigh(int min_required_samples) const;
> 
>  private:
>   const std::unique_ptr<int[]> buffer_;
>@@ -39,7 +39,7 @@ class QualityThreshold {
>   const int high_threshold_;
>   int until_full_;
>   int next_index_;
>-  rtc::Optional<bool> is_high_;
>+  absl::optional<bool> is_high_;
>   int sum_;
>   int count_low_;
>   int count_high_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/receive_statistics_proxy.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/receive_statistics_proxy.cc
>index 7bde5d00d97..dada4455970 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/receive_statistics_proxy.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/receive_statistics_proxy.cc
>@@ -12,12 +12,13 @@
> 
> #include <algorithm>
> #include <cmath>
>-#include <sstream>
> #include <utility>
> 
> #include "modules/video_coding/include/video_codec_interface.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>+#include "rtc_base/strings/string_builder.h"
>+#include "rtc_base/timeutils.h"
> #include "system_wrappers/include/clock.h"
> #include "system_wrappers/include/metrics.h"
> 
>@@ -56,17 +57,15 @@ const int kRateStatisticsWindowSizeMs = 1000;
> // values above - in the map.
> const int kMaxCommonInterframeDelayMs = 500;
> 
>-std::string UmaPrefixForContentType(VideoContentType content_type) {
>-  std::stringstream ss;
>-  ss << "WebRTC.Video";
>-  if (videocontenttypehelpers::IsScreenshare(content_type)) {
>-    ss << ".Screenshare";
>-  }
>-  return ss.str();
>+const char* UmaPrefixForContentType(VideoContentType content_type) {
>+  if (videocontenttypehelpers::IsScreenshare(content_type))
>+    return "WebRTC.Video.Screenshare";
>+  return "WebRTC.Video";
> }
> 
> std::string UmaSuffixForContentType(VideoContentType content_type) {
>-  std::stringstream ss;
>+  char ss_buf[1024];
>+  rtc::SimpleStringBuilder ss(ss_buf);
>   int simulcast_id = videocontenttypehelpers::GetSimulcastId(content_type);
>   if (simulcast_id > 0) {
>     ss << ".S" << simulcast_id - 1;
>@@ -77,6 +76,7 @@ std::string UmaSuffixForContentType(VideoContentType content_type) {
>   }
>   return ss.str();
> }
>+
> }  // namespace
> 
> ReceiveStatisticsProxy::ReceiveStatisticsProxy(
>@@ -106,12 +106,19 @@ ReceiveStatisticsProxy::ReceiveStatisticsProxy(
>       render_fps_tracker_(100, 10u),
>       render_pixel_tracker_(100, 10u),
>       total_byte_tracker_(100, 10u),  // bucket_interval_ms, bucket_count
>+      video_quality_observer_(
>+          new VideoQualityObserver(VideoContentType::UNSPECIFIED)),
>       interframe_delay_max_moving_(kMovingMaxWindowMs),
>       freq_offset_counter_(clock, nullptr, kFreqOffsetProcessIntervalMs),
>       first_report_block_time_ms_(-1),
>       avg_rtt_ms_(0),
>       last_content_type_(VideoContentType::UNSPECIFIED),
>+      last_codec_type_(kVideoCodecVP8),
>+      num_delayed_frames_rendered_(0),
>+      sum_missed_render_deadline_ms_(0),
>       timing_frame_info_counter_(kMovingMaxWindowMs) {
>+  decode_thread_.DetachFromThread();
>+  network_thread_.DetachFromThread();
>   stats_.ssrc = config_.rtp.remote_ssrc;
>   // TODO(brandtr): Replace |rtx_stats_| with a single instance of
>   // StreamDataCounters.
>@@ -121,17 +128,38 @@ ReceiveStatisticsProxy::ReceiveStatisticsProxy(
> }
> 
> ReceiveStatisticsProxy::~ReceiveStatisticsProxy() {
>+  RTC_DCHECK_RUN_ON(&main_thread_);
>+  // In case you're reading this wondering "hmm... we're on the main thread but
>+  // calling a method that needs to be called on the decoder thread...", then
>+  // here's what's going on:
>+  // - The decoder thread has been stopped and DecoderThreadStopped() has been
>+  //   called.
>+  // - The decode_thread_ thread checker has been detached, and will now become
>+  //   attached to the current thread, which is OK since we're in the dtor.
>   UpdateHistograms();
> }
> 
> void ReceiveStatisticsProxy::UpdateHistograms() {
>+  RTC_DCHECK_RUN_ON(&decode_thread_);
>+  char log_stream_buf[8 * 1024];
>+  rtc::SimpleStringBuilder log_stream(log_stream_buf);
>   int stream_duration_sec = (clock_->TimeInMilliseconds() - start_ms_) / 1000;
>   if (stats_.frame_counts.key_frames > 0 ||
>       stats_.frame_counts.delta_frames > 0) {
>     RTC_HISTOGRAM_COUNTS_100000("WebRTC.Video.ReceiveStreamLifetimeInSeconds",
>                                 stream_duration_sec);
>-    RTC_LOG(LS_INFO) << "WebRTC.Video.ReceiveStreamLifetimeInSeconds "
>-                     << stream_duration_sec;
>+    log_stream << "WebRTC.Video.ReceiveStreamLifetimeInSeconds "
>+               << stream_duration_sec << '\n';
>+  }
>+
>+  log_stream << "Frames decoded " << stats_.frames_decoded << '\n';
>+
>+  if (num_unique_frames_) {
>+    int num_dropped_frames = *num_unique_frames_ - stats_.frames_decoded;
>+    RTC_HISTOGRAM_COUNTS_1000("WebRTC.Video.DroppedFrames.Receiver",
>+                              num_dropped_frames);
>+    log_stream << "WebRTC.Video.DroppedFrames.Receiver " << num_dropped_frames
>+               << '\n';
>   }
> 
>   if (first_report_block_time_ms_ != -1 &&
>@@ -141,8 +169,33 @@ void ReceiveStatisticsProxy::UpdateHistograms() {
>     if (fraction_lost != -1) {
>       RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.ReceivedPacketsLostInPercent",
>                                fraction_lost);
>-      RTC_LOG(LS_INFO) << "WebRTC.Video.ReceivedPacketsLostInPercent "
>-                       << fraction_lost;
>+      log_stream << "WebRTC.Video.ReceivedPacketsLostInPercent "
>+                 << fraction_lost << '\n';
>+    }
>+  }
>+
>+  if (first_decoded_frame_time_ms_) {
>+    const int64_t elapsed_ms =
>+        (clock_->TimeInMilliseconds() - *first_decoded_frame_time_ms_);
>+    if (elapsed_ms >=
>+        metrics::kMinRunTimeInSeconds * rtc::kNumMillisecsPerSec) {
>+      RTC_HISTOGRAM_COUNTS_100(
>+          "WebRTC.Video.DecodedFramesPerSecond",
>+          static_cast<int>((stats_.frames_decoded * 1000.0f / elapsed_ms) +
>+                           0.5f));
>+
>+      const uint32_t frames_rendered = stats_.frames_rendered;
>+      if (frames_rendered > 0) {
>+        RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.DelayedFramesToRenderer",
>+                                 static_cast<int>(num_delayed_frames_rendered_ *
>+                                                  100 / frames_rendered));
>+        if (num_delayed_frames_rendered_ > 0) {
>+          RTC_HISTOGRAM_COUNTS_1000(
>+              "WebRTC.Video.DelayedFramesToRenderer_AvgDelayInMs",
>+              static_cast<int>(sum_missed_render_deadline_ms_ /
>+                               num_delayed_frames_rendered_));
>+        }
>+      }
>     }
>   }
> 
>@@ -156,17 +209,19 @@ void ReceiveStatisticsProxy::UpdateHistograms() {
>         round(render_pixel_tracker_.ComputeTotalRate()));
>   }
> 
>-  int sync_offset_ms = sync_offset_counter_.Avg(kMinRequiredSamples);
>-  if (sync_offset_ms != -1) {
>-    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.AVSyncOffsetInMs", sync_offset_ms);
>-    RTC_LOG(LS_INFO) << "WebRTC.Video.AVSyncOffsetInMs " << sync_offset_ms;
>+  absl::optional<int> sync_offset_ms =
>+      sync_offset_counter_.Avg(kMinRequiredSamples);
>+  if (sync_offset_ms) {
>+    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.AVSyncOffsetInMs",
>+                               *sync_offset_ms);
>+    log_stream << "WebRTC.Video.AVSyncOffsetInMs " << *sync_offset_ms << '\n';
>   }
>   AggregatedStats freq_offset_stats = freq_offset_counter_.GetStats();
>   if (freq_offset_stats.num_samples > 0) {
>     RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.RtpToNtpFreqOffsetInKhz",
>                                freq_offset_stats.average);
>-    RTC_LOG(LS_INFO) << "WebRTC.Video.RtpToNtpFreqOffsetInKhz, "
>-                     << freq_offset_stats.ToString();
>+    log_stream << "WebRTC.Video.RtpToNtpFreqOffsetInKhz "
>+               << freq_offset_stats.ToString() << '\n';
>   }
> 
>   int num_total_frames =
>@@ -177,41 +232,45 @@ void ReceiveStatisticsProxy::UpdateHistograms() {
>         (num_key_frames * 1000 + num_total_frames / 2) / num_total_frames;
>     RTC_HISTOGRAM_COUNTS_1000("WebRTC.Video.KeyFramesReceivedInPermille",
>                               key_frames_permille);
>-    RTC_LOG(LS_INFO) << "WebRTC.Video.KeyFramesReceivedInPermille "
>-                     << key_frames_permille;
>+    log_stream << "WebRTC.Video.KeyFramesReceivedInPermille "
>+               << key_frames_permille << '\n';
>   }
> 
>-  int qp = qp_counters_.vp8.Avg(kMinRequiredSamples);
>-  if (qp != -1) {
>-    RTC_HISTOGRAM_COUNTS_200("WebRTC.Video.Decoded.Vp8.Qp", qp);
>-    RTC_LOG(LS_INFO) << "WebRTC.Video.Decoded.Vp8.Qp " << qp;
>+  absl::optional<int> qp = qp_counters_.vp8.Avg(kMinRequiredSamples);
>+  if (qp) {
>+    RTC_HISTOGRAM_COUNTS_200("WebRTC.Video.Decoded.Vp8.Qp", *qp);
>+    log_stream << "WebRTC.Video.Decoded.Vp8.Qp " << *qp << '\n';
>   }
>-  int decode_ms = decode_time_counter_.Avg(kMinRequiredSamples);
>-  if (decode_ms != -1) {
>-    RTC_HISTOGRAM_COUNTS_1000("WebRTC.Video.DecodeTimeInMs", decode_ms);
>-    RTC_LOG(LS_INFO) << "WebRTC.Video.DecodeTimeInMs " << decode_ms;
>+  absl::optional<int> decode_ms = decode_time_counter_.Avg(kMinRequiredSamples);
>+  if (decode_ms) {
>+    RTC_HISTOGRAM_COUNTS_1000("WebRTC.Video.DecodeTimeInMs", *decode_ms);
>+    log_stream << "WebRTC.Video.DecodeTimeInMs " << *decode_ms << '\n';
>   }
>-  int jb_delay_ms = jitter_buffer_delay_counter_.Avg(kMinRequiredSamples);
>-  if (jb_delay_ms != -1) {
>+  absl::optional<int> jb_delay_ms =
>+      jitter_buffer_delay_counter_.Avg(kMinRequiredSamples);
>+  if (jb_delay_ms) {
>     RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.JitterBufferDelayInMs",
>-                               jb_delay_ms);
>-    RTC_LOG(LS_INFO) << "WebRTC.Video.JitterBufferDelayInMs " << jb_delay_ms;
>+                               *jb_delay_ms);
>+    log_stream << "WebRTC.Video.JitterBufferDelayInMs " << *jb_delay_ms << '\n';
>   }
> 
>-  int target_delay_ms = target_delay_counter_.Avg(kMinRequiredSamples);
>-  if (target_delay_ms != -1) {
>-    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.TargetDelayInMs", target_delay_ms);
>-    RTC_LOG(LS_INFO) << "WebRTC.Video.TargetDelayInMs " << target_delay_ms;
>+  absl::optional<int> target_delay_ms =
>+      target_delay_counter_.Avg(kMinRequiredSamples);
>+  if (target_delay_ms) {
>+    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.TargetDelayInMs",
>+                               *target_delay_ms);
>+    log_stream << "WebRTC.Video.TargetDelayInMs " << *target_delay_ms << '\n';
>   }
>-  int current_delay_ms = current_delay_counter_.Avg(kMinRequiredSamples);
>-  if (current_delay_ms != -1) {
>+  absl::optional<int> current_delay_ms =
>+      current_delay_counter_.Avg(kMinRequiredSamples);
>+  if (current_delay_ms) {
>     RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.CurrentDelayInMs",
>-                               current_delay_ms);
>-    RTC_LOG(LS_INFO) << "WebRTC.Video.CurrentDelayInMs " << current_delay_ms;
>+                               *current_delay_ms);
>+    log_stream << "WebRTC.Video.CurrentDelayInMs " << *current_delay_ms << '\n';
>   }
>-  int delay_ms = delay_counter_.Avg(kMinRequiredSamples);
>-  if (delay_ms != -1)
>-    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.OnewayDelayInMs", delay_ms);
>+  absl::optional<int> delay_ms = delay_counter_.Avg(kMinRequiredSamples);
>+  if (delay_ms)
>+    RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.OnewayDelayInMs", *delay_ms);
> 
>   // Aggregate content_specific_stats_ by removing experiment or simulcast
>   // information;
>@@ -255,62 +314,64 @@ void ReceiveStatisticsProxy::UpdateHistograms() {
>     RTC_DCHECK(videocontenttypehelpers::GetExperimentId(content_type) == 0 ||
>                videocontenttypehelpers::GetSimulcastId(content_type) == 0);
> 
>-    int e2e_delay_ms = stats.e2e_delay_counter.Avg(kMinRequiredSamples);
>-    if (e2e_delay_ms != -1) {
>+    absl::optional<int> e2e_delay_ms =
>+        stats.e2e_delay_counter.Avg(kMinRequiredSamples);
>+    if (e2e_delay_ms) {
>       RTC_HISTOGRAM_COUNTS_SPARSE_10000(
>-          uma_prefix + ".EndToEndDelayInMs" + uma_suffix, e2e_delay_ms);
>-      RTC_LOG(LS_INFO) << uma_prefix << ".EndToEndDelayInMs" << uma_suffix
>-                       << " " << e2e_delay_ms;
>+          uma_prefix + ".EndToEndDelayInMs" + uma_suffix, *e2e_delay_ms);
>+      log_stream << uma_prefix << ".EndToEndDelayInMs" << uma_suffix << " "
>+                 << *e2e_delay_ms << '\n';
>     }
>-    int e2e_delay_max_ms = stats.e2e_delay_counter.Max();
>-    if (e2e_delay_max_ms != -1 && e2e_delay_ms != -1) {
>+    absl::optional<int> e2e_delay_max_ms = stats.e2e_delay_counter.Max();
>+    if (e2e_delay_max_ms && e2e_delay_ms) {
>       RTC_HISTOGRAM_COUNTS_SPARSE_100000(
>-          uma_prefix + ".EndToEndDelayMaxInMs" + uma_suffix, e2e_delay_max_ms);
>-      RTC_LOG(LS_INFO) << uma_prefix << ".EndToEndDelayMaxInMs" << uma_suffix
>-                       << " " << e2e_delay_max_ms;
>+          uma_prefix + ".EndToEndDelayMaxInMs" + uma_suffix, *e2e_delay_max_ms);
>+      log_stream << uma_prefix << ".EndToEndDelayMaxInMs" << uma_suffix << " "
>+                 << *e2e_delay_max_ms << '\n';
>     }
>-    int interframe_delay_ms =
>+    absl::optional<int> interframe_delay_ms =
>         stats.interframe_delay_counter.Avg(kMinRequiredSamples);
>-    if (interframe_delay_ms != -1) {
>+    if (interframe_delay_ms) {
>       RTC_HISTOGRAM_COUNTS_SPARSE_10000(
>           uma_prefix + ".InterframeDelayInMs" + uma_suffix,
>-          interframe_delay_ms);
>-      RTC_LOG(LS_INFO) << uma_prefix << ".InterframeDelayInMs" << uma_suffix
>-                       << " " << interframe_delay_ms;
>+          *interframe_delay_ms);
>+      log_stream << uma_prefix << ".InterframeDelayInMs" << uma_suffix << " "
>+                 << *interframe_delay_ms << '\n';
>     }
>-    int interframe_delay_max_ms = stats.interframe_delay_counter.Max();
>-    if (interframe_delay_max_ms != -1 && interframe_delay_ms != -1) {
>+    absl::optional<int> interframe_delay_max_ms =
>+        stats.interframe_delay_counter.Max();
>+    if (interframe_delay_max_ms && interframe_delay_ms) {
>       RTC_HISTOGRAM_COUNTS_SPARSE_10000(
>           uma_prefix + ".InterframeDelayMaxInMs" + uma_suffix,
>-          interframe_delay_max_ms);
>-      RTC_LOG(LS_INFO) << uma_prefix << ".InterframeDelayMaxInMs" << uma_suffix
>-                       << " " << interframe_delay_max_ms;
>+          *interframe_delay_max_ms);
>+      log_stream << uma_prefix << ".InterframeDelayMaxInMs" << uma_suffix << " "
>+                 << *interframe_delay_max_ms << '\n';
>     }
> 
>-    rtc::Optional<uint32_t> interframe_delay_95p_ms =
>+    absl::optional<uint32_t> interframe_delay_95p_ms =
>         stats.interframe_delay_percentiles.GetPercentile(0.95f);
>     if (interframe_delay_95p_ms && interframe_delay_ms != -1) {
>       RTC_HISTOGRAM_COUNTS_SPARSE_10000(
>           uma_prefix + ".InterframeDelay95PercentileInMs" + uma_suffix,
>           *interframe_delay_95p_ms);
>-      RTC_LOG(LS_INFO) << uma_prefix << ".InterframeDelay95PercentileInMs"
>-                       << uma_suffix << " " << *interframe_delay_95p_ms;
>+      log_stream << uma_prefix << ".InterframeDelay95PercentileInMs"
>+                 << uma_suffix << " " << *interframe_delay_95p_ms << '\n';
>     }
> 
>-    int width = stats.received_width.Avg(kMinRequiredSamples);
>-    if (width != -1) {
>+    absl::optional<int> width = stats.received_width.Avg(kMinRequiredSamples);
>+    if (width) {
>       RTC_HISTOGRAM_COUNTS_SPARSE_10000(
>-          uma_prefix + ".ReceivedWidthInPixels" + uma_suffix, width);
>-      RTC_LOG(LS_INFO) << uma_prefix << ".ReceivedWidthInPixels" << uma_suffix
>-                       << " " << width;
>+          uma_prefix + ".ReceivedWidthInPixels" + uma_suffix, *width);
>+      log_stream << uma_prefix << ".ReceivedWidthInPixels" << uma_suffix << " "
>+                 << *width << '\n';
>     }
> 
>-    int height = stats.received_height.Avg(kMinRequiredSamples);
>-    if (height != -1) {
>+    absl::optional<int> height = stats.received_height.Avg(kMinRequiredSamples);
>+    if (height) {
>       RTC_HISTOGRAM_COUNTS_SPARSE_10000(
>-          uma_prefix + ".ReceivedHeightInPixels" + uma_suffix, height);
>-      RTC_LOG(LS_INFO) << uma_prefix << ".ReceivedHeightInPixels" << uma_suffix
>-                       << " " << height;
>+          uma_prefix + ".ReceivedHeightInPixels" + uma_suffix, *height);
>+      log_stream << uma_prefix << ".ReceivedHeightInPixels" << uma_suffix << " "
>+                 << *height << '\n';
>     }
> 
>     if (content_type != VideoContentType::UNSPECIFIED) {
>@@ -323,8 +384,8 @@ void ReceiveStatisticsProxy::UpdateHistograms() {
>         RTC_HISTOGRAM_COUNTS_SPARSE_10000(
>             uma_prefix + ".MediaBitrateReceivedInKbps" + uma_suffix,
>             media_bitrate_kbps);
>-        RTC_LOG(LS_INFO) << uma_prefix << ".MediaBitrateReceivedInKbps"
>-                         << uma_suffix << " " << media_bitrate_kbps;
>+        log_stream << uma_prefix << ".MediaBitrateReceivedInKbps" << uma_suffix
>+                   << " " << media_bitrate_kbps << '\n';
>       }
> 
>       int num_total_frames =
>@@ -336,16 +397,16 @@ void ReceiveStatisticsProxy::UpdateHistograms() {
>         RTC_HISTOGRAM_COUNTS_SPARSE_1000(
>             uma_prefix + ".KeyFramesReceivedInPermille" + uma_suffix,
>             key_frames_permille);
>-        RTC_LOG(LS_INFO) << uma_prefix << ".KeyFramesReceivedInPermille"
>-                         << uma_suffix << " " << key_frames_permille;
>+        log_stream << uma_prefix << ".KeyFramesReceivedInPermille" << uma_suffix
>+                   << " " << key_frames_permille << '\n';
>       }
> 
>-      int qp = stats.qp_counter.Avg(kMinRequiredSamples);
>-      if (qp != -1) {
>+      absl::optional<int> qp = stats.qp_counter.Avg(kMinRequiredSamples);
>+      if (qp) {
>         RTC_HISTOGRAM_COUNTS_SPARSE_200(
>-            uma_prefix + ".Decoded.Vp8.Qp" + uma_suffix, qp);
>-        RTC_LOG(LS_INFO) << uma_prefix << ".Decoded.Vp8.Qp" << uma_suffix << " "
>-                         << qp;
>+            uma_prefix + ".Decoded.Vp8.Qp" + uma_suffix, *qp);
>+        log_stream << uma_prefix << ".Decoded.Vp8.Qp" << uma_suffix << " "
>+                   << *qp << '\n';
>       }
>     }
>   }
>@@ -367,8 +428,8 @@ void ReceiveStatisticsProxy::UpdateHistograms() {
>         static_cast<int>(rtp.MediaPayloadBytes() * 8 / elapsed_sec / 1000);
>     RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.MediaBitrateReceivedInKbps",
>                                media_bitrate_kbs);
>-    RTC_LOG(LS_INFO) << "WebRTC.Video.MediaBitrateReceivedInKbps "
>-                     << media_bitrate_kbs;
>+    log_stream << "WebRTC.Video.MediaBitrateReceivedInKbps "
>+               << media_bitrate_kbs << '\n';
>     RTC_HISTOGRAM_COUNTS_10000(
>         "WebRTC.Video.PaddingBitrateReceivedInKbps",
>         static_cast<int>(rtp_rtx.transmitted.padding_bytes * 8 / elapsed_sec /
>@@ -404,34 +465,38 @@ void ReceiveStatisticsProxy::UpdateHistograms() {
>     RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.BadCall.Any",
>                              100 * num_bad_states_ / num_certain_states_);
>   }
>-  rtc::Optional<double> fps_fraction =
>+  absl::optional<double> fps_fraction =
>       fps_threshold_.FractionHigh(kBadCallMinRequiredSamples);
>   if (fps_fraction) {
>     RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.BadCall.FrameRate",
>                              static_cast<int>(100 * (1 - *fps_fraction)));
>   }
>-  rtc::Optional<double> variance_fraction =
>+  absl::optional<double> variance_fraction =
>       variance_threshold_.FractionHigh(kBadCallMinRequiredSamples);
>   if (variance_fraction) {
>     RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.BadCall.FrameRateVariance",
>                              static_cast<int>(100 * *variance_fraction));
>   }
>-  rtc::Optional<double> qp_fraction =
>+  absl::optional<double> qp_fraction =
>       qp_threshold_.FractionHigh(kBadCallMinRequiredSamples);
>   if (qp_fraction) {
>     RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.BadCall.Qp",
>                              static_cast<int>(100 * *qp_fraction));
>   }
>+
>+  RTC_LOG(LS_INFO) << log_stream.str();
> }
> 
> void ReceiveStatisticsProxy::QualitySample() {
>+  RTC_DCHECK_RUN_ON(&network_thread_);
>+
>   int64_t now = clock_->TimeInMilliseconds();
>   if (last_sample_time_ + kMinSampleLengthMs > now)
>     return;
> 
>   double fps =
>       render_fps_tracker_.ComputeRateForInterval(now - last_sample_time_);
>-  int qp = qp_sample_.Avg(1);
>+  absl::optional<int> qp = qp_sample_.Avg(1);
> 
>   bool prev_fps_bad = !fps_threshold_.IsHigh().value_or(true);
>   bool prev_qp_bad = qp_threshold_.IsHigh().value_or(false);
>@@ -439,9 +504,9 @@ void ReceiveStatisticsProxy::QualitySample() {
>   bool prev_any_bad = prev_fps_bad || prev_qp_bad || prev_variance_bad;
> 
>   fps_threshold_.AddMeasurement(static_cast<int>(fps));
>-  if (qp != -1)
>-    qp_threshold_.AddMeasurement(qp);
>-  rtc::Optional<double> fps_variance_opt = fps_threshold_.CalculateVariance();
>+  if (qp)
>+    qp_threshold_.AddMeasurement(*qp);
>+  absl::optional<double> fps_variance_opt = fps_threshold_.CalculateVariance();
>   double fps_variance = fps_variance_opt.value_or(0);
>   if (fps_variance_opt) {
>     variance_threshold_.AddMeasurement(static_cast<int>(fps_variance));
>@@ -478,7 +543,7 @@ void ReceiveStatisticsProxy::QualitySample() {
> 
>   RTC_LOG(LS_VERBOSE) << "SAMPLE: sample_length: " << (now - last_sample_time_)
>                       << " fps: " << fps << " fps_bad: " << fps_bad
>-                      << " qp: " << qp << " qp_bad: " << qp_bad
>+                      << " qp: " << qp.value_or(-1) << " qp_bad: " << qp_bad
>                       << " variance_bad: " << variance_bad
>                       << " fps_variance: " << fps_variance;
> 
>@@ -534,6 +599,7 @@ void ReceiveStatisticsProxy::OnDecoderImplementationName(
> }
> void ReceiveStatisticsProxy::OnIncomingRate(unsigned int framerate,
>                                             unsigned int bitrate_bps) {
>+  RTC_DCHECK_RUN_ON(&network_thread_);
>   rtc::CritScope lock(&crit_);
>   if (stats_.rtp_stats.first_packet_time_ms != -1)
>     QualitySample();
>@@ -564,6 +630,11 @@ void ReceiveStatisticsProxy::OnFrameBufferTimingsUpdated(
>   delay_counter_.Add(target_delay_ms + avg_rtt_ms_ / 2);
> }
> 
>+void ReceiveStatisticsProxy::OnUniqueFramesCounted(int num_unique_frames) {
>+  rtc::CritScope lock(&crit_);
>+  num_unique_frames_.emplace(num_unique_frames);
>+}
>+
> void ReceiveStatisticsProxy::OnTimingFrameInfoUpdated(
>     const TimingFrameInfo& info) {
>   rtc::CritScope lock(&crit_);
>@@ -628,12 +699,24 @@ void ReceiveStatisticsProxy::DataCountersUpdated(
>     total_byte_tracker_.AddSamples(total_bytes - last_total_bytes);
> }
> 
>-void ReceiveStatisticsProxy::OnDecodedFrame(rtc::Optional<uint8_t> qp,
>+void ReceiveStatisticsProxy::OnDecodedFrame(absl::optional<uint8_t> qp,
>+                                            int width,
>+                                            int height,
>                                             VideoContentType content_type) {
>   rtc::CritScope lock(&crit_);
> 
>   uint64_t now = clock_->TimeInMilliseconds();
> 
>+  if (videocontenttypehelpers::IsScreenshare(content_type) !=
>+      videocontenttypehelpers::IsScreenshare(last_content_type_)) {
>+    // Reset the quality observer if content type is switched. This will
>+    // report stats for the previous part of the call.
>+    video_quality_observer_.reset(new VideoQualityObserver(content_type));
>+  }
>+
>+  video_quality_observer_->OnDecodedFrame(qp, width, height, now,
>+                                          last_codec_type_);
>+
>   ContentSpecificStats* content_specific_stats =
>       &content_specific_stats_[content_type];
>   ++stats_.frames_decoded;
>@@ -651,7 +734,7 @@ void ReceiveStatisticsProxy::OnDecodedFrame(rtc::Optional<uint8_t> qp,
>   } else if (stats_.qp_sum) {
>     RTC_LOG(LS_WARNING)
>         << "QP sum was already set and no QP was given for a frame.";
>-    stats_.qp_sum = rtc::nullopt;
>+    stats_.qp_sum = absl::nullopt;
>   }
>   last_content_type_ = content_type;
>   decode_fps_estimator_.Update(1, now);
>@@ -664,6 +747,8 @@ void ReceiveStatisticsProxy::OnDecodedFrame(rtc::Optional<uint8_t> qp,
>         interframe_delay_ms);
>     content_specific_stats->flow_duration_ms += interframe_delay_ms;
>   }
>+  if (stats_.frames_decoded == 1)
>+    first_decoded_frame_time_ms_.emplace(now);
>   last_decoded_frame_time_ms_.emplace(now);
> }
> 
>@@ -672,11 +757,11 @@ void ReceiveStatisticsProxy::OnRenderedFrame(const VideoFrame& frame) {
>   int height = frame.height();
>   RTC_DCHECK_GT(width, 0);
>   RTC_DCHECK_GT(height, 0);
>-  uint64_t now = clock_->TimeInMilliseconds();
>+  int64_t now_ms = clock_->TimeInMilliseconds();
>   rtc::CritScope lock(&crit_);
>   ContentSpecificStats* content_specific_stats =
>       &content_specific_stats_[last_content_type_];
>-  renders_fps_estimator_.Update(1, now);
>+  renders_fps_estimator_.Update(1, now_ms);
>   ++stats_.frames_rendered;
>   stats_.width = width;
>   stats_.height = height;
>@@ -685,6 +770,13 @@ void ReceiveStatisticsProxy::OnRenderedFrame(const VideoFrame& frame) {
>   content_specific_stats->received_width.Add(width);
>   content_specific_stats->received_height.Add(height);
> 
>+  // Consider taking stats_.render_delay_ms into account.
>+  const int64_t time_until_rendering_ms = frame.render_time_ms() - now_ms;
>+  if (time_until_rendering_ms < 0) {
>+    sum_missed_render_deadline_ms_ += -time_until_rendering_ms;
>+    ++num_delayed_frames_rendered_;
>+  }
>+
>   if (frame.ntp_time_ms() > 0) {
>     int64_t delay_ms = clock_->CurrentNtpInMilliseconds() - frame.ntp_time_ms();
>     if (delay_ms >= 0) {
>@@ -709,8 +801,7 @@ void ReceiveStatisticsProxy::OnSyncOffsetUpdated(int64_t sync_offset_ms,
> }
> 
> void ReceiveStatisticsProxy::OnReceiveRatesUpdated(uint32_t bitRate,
>-                                                   uint32_t frameRate) {
>-}
>+                                                   uint32_t frameRate) {}
> 
> void ReceiveStatisticsProxy::OnCompleteFrame(bool is_keyframe,
>                                              size_t size_bytes,
>@@ -751,12 +842,14 @@ void ReceiveStatisticsProxy::OnDiscardedPacketsUpdated(int discarded_packets) {
> void ReceiveStatisticsProxy::OnPreDecode(
>     const EncodedImage& encoded_image,
>     const CodecSpecificInfo* codec_specific_info) {
>+  RTC_DCHECK_RUN_ON(&decode_thread_);
>   if (!codec_specific_info || encoded_image.qp_ == -1) {
>     return;
>   }
>-  if (codec_specific_info->codecType == kVideoCodecVP8) {
>+  rtc::CritScope lock(&crit_);
>+  last_codec_type_ = codec_specific_info->codecType;
>+  if (last_codec_type_ == kVideoCodecVP8) {
>     qp_counters_.vp8.Add(encoded_image.qp_);
>-    rtc::CritScope lock(&crit_);
>     qp_sample_.Add(encoded_image.qp_);
>   }
> }
>@@ -767,38 +860,7 @@ void ReceiveStatisticsProxy::OnStreamInactive() {
>   rtc::CritScope lock(&crit_);
>   // Don't report inter-frame delay if stream was paused.
>   last_decoded_frame_time_ms_.reset();
>-}
>-
>-void ReceiveStatisticsProxy::SampleCounter::Add(int sample) {
>-  sum += sample;
>-  ++num_samples;
>-  if (!max || sample > *max) {
>-    max.emplace(sample);
>-  }
>-}
>-
>-void ReceiveStatisticsProxy::SampleCounter::Add(const SampleCounter& other) {
>-  sum += other.sum;
>-  num_samples += other.num_samples;
>-  if (other.max && (!max || *max < *other.max))
>-    max = other.max;
>-}
>-
>-int ReceiveStatisticsProxy::SampleCounter::Avg(
>-    int64_t min_required_samples) const {
>-  if (num_samples < min_required_samples || num_samples == 0)
>-    return -1;
>-  return static_cast<int>(sum / num_samples);
>-}
>-
>-int ReceiveStatisticsProxy::SampleCounter::Max() const {
>-  return max.value_or(-1);
>-}
>-
>-void ReceiveStatisticsProxy::SampleCounter::Reset() {
>-  num_samples = 0;
>-  sum = 0;
>-  max.reset();
>+  video_quality_observer_->OnStreamInactive();
> }
> 
> void ReceiveStatisticsProxy::OnRttUpdate(int64_t avg_rtt_ms,
>@@ -807,6 +869,15 @@ void ReceiveStatisticsProxy::OnRttUpdate(int64_t avg_rtt_ms,
>   avg_rtt_ms_ = avg_rtt_ms;
> }
> 
>+void ReceiveStatisticsProxy::DecoderThreadStarting() {
>+  RTC_DCHECK_RUN_ON(&main_thread_);
>+}
>+
>+void ReceiveStatisticsProxy::DecoderThreadStopped() {
>+  RTC_DCHECK_RUN_ON(&main_thread_);
>+  decode_thread_.DetachFromThread();
>+}
>+
> ReceiveStatisticsProxy::ContentSpecificStats::ContentSpecificStats()
>     : interframe_delay_percentiles(kMaxCommonInterframeDelayMs) {}
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/receive_statistics_proxy.h b/Source/ThirdParty/libwebrtc/Source/webrtc/video/receive_statistics_proxy.h
>index 0e0c5b45b58..2ccfc938d14 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/receive_statistics_proxy.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/receive_statistics_proxy.h
>@@ -12,10 +12,11 @@
> #define VIDEO_RECEIVE_STATISTICS_PROXY_H_
> 
> #include <map>
>+#include <memory>
> #include <string>
> #include <vector>
> 
>-#include "api/optional.h"
>+#include "absl/types/optional.h"
> #include "call/video_receive_stream.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "common_video/include/frame_callback.h"
>@@ -23,19 +24,20 @@
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/numerics/histogram_percentile_counter.h"
> #include "rtc_base/numerics/moving_max_counter.h"
>+#include "rtc_base/numerics/sample_counter.h"
> #include "rtc_base/rate_statistics.h"
> #include "rtc_base/ratetracker.h"
> #include "rtc_base/thread_annotations.h"
>+#include "rtc_base/thread_checker.h"
> #include "video/quality_threshold.h"
> #include "video/report_block_stats.h"
> #include "video/stats_counter.h"
>+#include "video/video_quality_observer.h"
> #include "video/video_stream_decoder.h"
> 
> namespace webrtc {
> 
> class Clock;
>-class ViECodec;
>-class ViEDecoderObserver;
> struct CodecSpecificInfo;
> 
> class ReceiveStatisticsProxy : public VCMReceiveStatisticsCallback,
>@@ -50,7 +52,10 @@ class ReceiveStatisticsProxy : public VCMReceiveStatisticsCallback,
> 
>   VideoReceiveStream::Stats GetStats() const;
> 
>-  void OnDecodedFrame(rtc::Optional<uint8_t> qp, VideoContentType content_type);
>+  void OnDecodedFrame(absl::optional<uint8_t> qp,
>+                      int width,
>+                      int height,
>+                      VideoContentType content_type);
>   void OnSyncOffsetUpdated(int64_t sync_offset_ms, double estimated_freq_khz);
>   void OnRenderedFrame(const VideoFrame& frame);
>   void OnIncomingPayloadType(int payload_type);
>@@ -60,6 +65,8 @@ class ReceiveStatisticsProxy : public VCMReceiveStatisticsCallback,
>   void OnPreDecode(const EncodedImage& encoded_image,
>                    const CodecSpecificInfo* codec_specific_info);
> 
>+  void OnUniqueFramesCounted(int num_unique_frames);
>+
>   // Indicates video stream has been paused (no incoming packets).
>   void OnStreamInactive();
> 
>@@ -96,23 +103,14 @@ class ReceiveStatisticsProxy : public VCMReceiveStatisticsCallback,
>   // Implements CallStatsObserver.
>   void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) override;
> 
>- private:
>-  struct SampleCounter {
>-    SampleCounter() : sum(0), num_samples(0) {}
>-    void Add(int sample);
>-    int Avg(int64_t min_required_samples) const;
>-    int Max() const;
>-    void Reset();
>-    void Add(const SampleCounter& other);
>-
>-   private:
>-    int64_t sum;
>-    int64_t num_samples;
>-    rtc::Optional<int> max;
>-  };
>+  // Notification methods that are used to check our internal state and validate
>+  // threading assumptions. These are called by VideoReceiveStream.
>+  void DecoderThreadStarting();
>+  void DecoderThreadStopped();
> 
>+ private:
>   struct QpCounters {
>-    SampleCounter vp8;
>+    rtc::SampleCounter vp8;
>   };
> 
>   struct ContentSpecificStats {
>@@ -120,13 +118,13 @@ class ReceiveStatisticsProxy : public VCMReceiveStatisticsCallback,
> 
>     void Add(const ContentSpecificStats& other);
> 
>-    SampleCounter e2e_delay_counter;
>-    SampleCounter interframe_delay_counter;
>+    rtc::SampleCounter e2e_delay_counter;
>+    rtc::SampleCounter interframe_delay_counter;
>     int64_t flow_duration_ms = 0;
>     int64_t total_media_bytes = 0;
>-    SampleCounter received_width;
>-    SampleCounter received_height;
>-    SampleCounter qp_counter;
>+    rtc::SampleCounter received_width;
>+    rtc::SampleCounter received_height;
>+    rtc::SampleCounter qp_counter;
>     FrameCounts frame_counts;
>     rtc::HistogramPercentileCounter interframe_delay_percentiles;
>   };
>@@ -155,7 +153,7 @@ class ReceiveStatisticsProxy : public VCMReceiveStatisticsCallback,
>   QualityThreshold fps_threshold_ RTC_GUARDED_BY(crit_);
>   QualityThreshold qp_threshold_ RTC_GUARDED_BY(crit_);
>   QualityThreshold variance_threshold_ RTC_GUARDED_BY(crit_);
>-  SampleCounter qp_sample_ RTC_GUARDED_BY(crit_);
>+  rtc::SampleCounter qp_sample_ RTC_GUARDED_BY(crit_);
>   int num_bad_states_ RTC_GUARDED_BY(crit_);
>   int num_certain_states_ RTC_GUARDED_BY(crit_);
>   mutable VideoReceiveStream::Stats stats_ RTC_GUARDED_BY(crit_);
>@@ -164,12 +162,14 @@ class ReceiveStatisticsProxy : public VCMReceiveStatisticsCallback,
>   rtc::RateTracker render_fps_tracker_ RTC_GUARDED_BY(crit_);
>   rtc::RateTracker render_pixel_tracker_ RTC_GUARDED_BY(crit_);
>   rtc::RateTracker total_byte_tracker_ RTC_GUARDED_BY(crit_);
>-  SampleCounter sync_offset_counter_ RTC_GUARDED_BY(crit_);
>-  SampleCounter decode_time_counter_ RTC_GUARDED_BY(crit_);
>-  SampleCounter jitter_buffer_delay_counter_ RTC_GUARDED_BY(crit_);
>-  SampleCounter target_delay_counter_ RTC_GUARDED_BY(crit_);
>-  SampleCounter current_delay_counter_ RTC_GUARDED_BY(crit_);
>-  SampleCounter delay_counter_ RTC_GUARDED_BY(crit_);
>+  rtc::SampleCounter sync_offset_counter_ RTC_GUARDED_BY(crit_);
>+  rtc::SampleCounter decode_time_counter_ RTC_GUARDED_BY(crit_);
>+  rtc::SampleCounter jitter_buffer_delay_counter_ RTC_GUARDED_BY(crit_);
>+  rtc::SampleCounter target_delay_counter_ RTC_GUARDED_BY(crit_);
>+  rtc::SampleCounter current_delay_counter_ RTC_GUARDED_BY(crit_);
>+  rtc::SampleCounter delay_counter_ RTC_GUARDED_BY(crit_);
>+  std::unique_ptr<VideoQualityObserver> video_quality_observer_
>+      RTC_GUARDED_BY(crit_);
>   mutable rtc::MovingMaxCounter<int> interframe_delay_max_moving_
>       RTC_GUARDED_BY(crit_);
>   std::map<VideoContentType, ContentSpecificStats> content_specific_stats_
>@@ -177,16 +177,24 @@ class ReceiveStatisticsProxy : public VCMReceiveStatisticsCallback,
>   MaxCounter freq_offset_counter_ RTC_GUARDED_BY(crit_);
>   int64_t first_report_block_time_ms_ RTC_GUARDED_BY(crit_);
>   ReportBlockStats report_block_stats_ RTC_GUARDED_BY(crit_);
>-  QpCounters qp_counters_;  // Only accessed on the decoding thread.
>+  QpCounters qp_counters_ RTC_GUARDED_BY(decode_thread_);
>   std::map<uint32_t, StreamDataCounters> rtx_stats_ RTC_GUARDED_BY(crit_);
>   int64_t avg_rtt_ms_ RTC_GUARDED_BY(crit_);
>   mutable std::map<int64_t, size_t> frame_window_ RTC_GUARDED_BY(&crit_);
>   VideoContentType last_content_type_ RTC_GUARDED_BY(&crit_);
>-  rtc::Optional<int64_t> last_decoded_frame_time_ms_ RTC_GUARDED_BY(&crit_);
>+  VideoCodecType last_codec_type_ RTC_GUARDED_BY(&crit_);
>+  absl::optional<int64_t> first_decoded_frame_time_ms_ RTC_GUARDED_BY(&crit_);
>+  absl::optional<int64_t> last_decoded_frame_time_ms_ RTC_GUARDED_BY(&crit_);
>+  size_t num_delayed_frames_rendered_ RTC_GUARDED_BY(&crit_);
>+  int64_t sum_missed_render_deadline_ms_ RTC_GUARDED_BY(&crit_);
>   // Mutable because calling Max() on MovingMaxCounter is not const. Yet it is
>   // called from const GetStats().
>   mutable rtc::MovingMaxCounter<TimingFrameInfo> timing_frame_info_counter_
>       RTC_GUARDED_BY(&crit_);
>+  absl::optional<int> num_unique_frames_ RTC_GUARDED_BY(crit_);
>+  rtc::ThreadChecker decode_thread_;
>+  rtc::ThreadChecker network_thread_;
>+  rtc::ThreadChecker main_thread_;
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/receive_statistics_proxy_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/receive_statistics_proxy_unittest.cc
>index 2a536907ecd..c3b9ae752c4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/receive_statistics_proxy_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/receive_statistics_proxy_unittest.cc
>@@ -28,6 +28,9 @@ const int64_t kFreqOffsetProcessIntervalInMs = 40000;
> const uint32_t kLocalSsrc = 123;
> const uint32_t kRemoteSsrc = 456;
> const int kMinRequiredSamples = 200;
>+const int kWidth = 1280;
>+const int kHeight = 720;
>+
> }  // namespace
> 
> // TODO(sakal): ReceiveStatisticsProxy is lacking unittesting.
>@@ -57,7 +60,16 @@ class ReceiveStatisticsProxyTest
>   }
> 
>   VideoFrame CreateFrame(int width, int height) {
>-    VideoFrame frame(I420Buffer::Create(width, height), 0, 0, kVideoRotation_0);
>+    return CreateVideoFrame(width, height, 0);
>+  }
>+
>+  VideoFrame CreateFrameWithRenderTimeMs(int64_t render_time_ms) {
>+    return CreateVideoFrame(kWidth, kHeight, render_time_ms);
>+  }
>+
>+  VideoFrame CreateVideoFrame(int width, int height, int64_t render_time_ms) {
>+    VideoFrame frame(I420Buffer::Create(width, height), 0, render_time_ms,
>+                     kVideoRotation_0);
>     frame.set_ntp_time_ms(fake_clock_.CurrentNtpInMilliseconds());
>     return frame;
>   }
>@@ -70,28 +82,76 @@ class ReceiveStatisticsProxyTest
> TEST_F(ReceiveStatisticsProxyTest, OnDecodedFrameIncreasesFramesDecoded) {
>   EXPECT_EQ(0u, statistics_proxy_->GetStats().frames_decoded);
>   for (uint32_t i = 1; i <= 3; ++i) {
>-    statistics_proxy_->OnDecodedFrame(rtc::nullopt,
>+    statistics_proxy_->OnDecodedFrame(absl::nullopt, kWidth, kHeight,
>                                       VideoContentType::UNSPECIFIED);
>     EXPECT_EQ(i, statistics_proxy_->GetStats().frames_decoded);
>   }
> }
> 
>+TEST_F(ReceiveStatisticsProxyTest, DecodedFpsIsReported) {
>+  const int kFps = 20;
>+  const int kRequiredSamples = metrics::kMinRunTimeInSeconds * kFps;
>+  for (int i = 0; i < kRequiredSamples; ++i) {
>+    statistics_proxy_->OnDecodedFrame(absl::optional<uint8_t>(), kWidth,
>+                                      kHeight, VideoContentType::UNSPECIFIED);
>+    fake_clock_.AdvanceTimeMilliseconds(1000 / kFps);
>+  }
>+  statistics_proxy_.reset();
>+  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.DecodedFramesPerSecond"));
>+  EXPECT_EQ(1, metrics::NumEvents("WebRTC.Video.DecodedFramesPerSecond", kFps));
>+}
>+
>+TEST_F(ReceiveStatisticsProxyTest, DecodedFpsIsNotReportedForTooFewSamples) {
>+  const int kFps = 20;
>+  const int kRequiredSamples = metrics::kMinRunTimeInSeconds * kFps;
>+  for (int i = 0; i < kRequiredSamples - 1; ++i) {
>+    statistics_proxy_->OnDecodedFrame(absl::optional<uint8_t>(), kWidth,
>+                                      kHeight, VideoContentType::UNSPECIFIED);
>+    fake_clock_.AdvanceTimeMilliseconds(1000 / kFps);
>+  }
>+  statistics_proxy_.reset();
>+  EXPECT_EQ(0, metrics::NumSamples("WebRTC.Video.DecodedFramesPerSecond"));
>+}
>+
>+TEST_F(ReceiveStatisticsProxyTest, DecodedFpsIsReportedWithQpReset) {
>+  const int kFps1 = 10;
>+  for (int i = 0; i < metrics::kMinRunTimeInSeconds * kFps1; ++i) {
>+    statistics_proxy_->OnDecodedFrame(absl::optional<uint8_t>(), kWidth,
>+                                      kHeight, VideoContentType::UNSPECIFIED);
>+    fake_clock_.AdvanceTimeMilliseconds(1000 / kFps1);
>+  }
>+  // First QP value received, resets frames decoded.
>+  const int kFps2 = 20;
>+  for (int i = 0; i < metrics::kMinRunTimeInSeconds * kFps2; ++i) {
>+    statistics_proxy_->OnDecodedFrame(absl::optional<uint8_t>(1u), kWidth,
>+                                      kHeight, VideoContentType::UNSPECIFIED);
>+    fake_clock_.AdvanceTimeMilliseconds(1000 / kFps2);
>+  }
>+  statistics_proxy_.reset();
>+  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.DecodedFramesPerSecond"));
>+  EXPECT_EQ(1,
>+            metrics::NumEvents("WebRTC.Video.DecodedFramesPerSecond", kFps2));
>+}
>+
> TEST_F(ReceiveStatisticsProxyTest, OnDecodedFrameWithQpResetsFramesDecoded) {
>   EXPECT_EQ(0u, statistics_proxy_->GetStats().frames_decoded);
>   for (uint32_t i = 1; i <= 3; ++i) {
>-    statistics_proxy_->OnDecodedFrame(rtc::nullopt,
>+    statistics_proxy_->OnDecodedFrame(absl::nullopt, kWidth, kHeight,
>                                       VideoContentType::UNSPECIFIED);
>     EXPECT_EQ(i, statistics_proxy_->GetStats().frames_decoded);
>   }
>-  statistics_proxy_->OnDecodedFrame(1u, VideoContentType::UNSPECIFIED);
>+  statistics_proxy_->OnDecodedFrame(1u, kWidth, kHeight,
>+                                    VideoContentType::UNSPECIFIED);
>   EXPECT_EQ(1u, statistics_proxy_->GetStats().frames_decoded);
> }
> 
> TEST_F(ReceiveStatisticsProxyTest, OnDecodedFrameIncreasesQpSum) {
>-  EXPECT_EQ(rtc::nullopt, statistics_proxy_->GetStats().qp_sum);
>-  statistics_proxy_->OnDecodedFrame(3u, VideoContentType::UNSPECIFIED);
>+  EXPECT_EQ(absl::nullopt, statistics_proxy_->GetStats().qp_sum);
>+  statistics_proxy_->OnDecodedFrame(3u, kWidth, kHeight,
>+                                    VideoContentType::UNSPECIFIED);
>   EXPECT_EQ(3u, statistics_proxy_->GetStats().qp_sum);
>-  statistics_proxy_->OnDecodedFrame(127u, VideoContentType::UNSPECIFIED);
>+  statistics_proxy_->OnDecodedFrame(127u, kWidth, kHeight,
>+                                    VideoContentType::UNSPECIFIED);
>   EXPECT_EQ(130u, statistics_proxy_->GetStats().qp_sum);
> }
> 
>@@ -100,10 +160,12 @@ TEST_F(ReceiveStatisticsProxyTest, ReportsContentType) {
>   const std::string kScreenshareString("screen");
>   EXPECT_EQ(kRealtimeString, videocontenttypehelpers::ToString(
>                             statistics_proxy_->GetStats().content_type));
>-  statistics_proxy_->OnDecodedFrame(3u, VideoContentType::SCREENSHARE);
>+  statistics_proxy_->OnDecodedFrame(3u, kWidth, kHeight,
>+                                    VideoContentType::SCREENSHARE);
>   EXPECT_EQ(kScreenshareString, videocontenttypehelpers::ToString(
>                           statistics_proxy_->GetStats().content_type));
>-  statistics_proxy_->OnDecodedFrame(3u, VideoContentType::UNSPECIFIED);
>+  statistics_proxy_->OnDecodedFrame(3u, kWidth, kHeight,
>+                                    VideoContentType::UNSPECIFIED);
>   EXPECT_EQ(kRealtimeString, videocontenttypehelpers::ToString(
>                             statistics_proxy_->GetStats().content_type));
> }
>@@ -113,21 +175,25 @@ TEST_F(ReceiveStatisticsProxyTest, ReportsMaxInterframeDelay) {
>   const int64_t kInterframeDelayMs2 = 200;
>   const int64_t kInterframeDelayMs3 = 100;
>   EXPECT_EQ(-1, statistics_proxy_->GetStats().interframe_delay_max_ms);
>-  statistics_proxy_->OnDecodedFrame(3u, VideoContentType::UNSPECIFIED);
>+  statistics_proxy_->OnDecodedFrame(3u, kWidth, kHeight,
>+                                    VideoContentType::UNSPECIFIED);
>   EXPECT_EQ(-1, statistics_proxy_->GetStats().interframe_delay_max_ms);
> 
>   fake_clock_.AdvanceTimeMilliseconds(kInterframeDelayMs1);
>-  statistics_proxy_->OnDecodedFrame(127u, VideoContentType::UNSPECIFIED);
>+  statistics_proxy_->OnDecodedFrame(127u, kWidth, kHeight,
>+                                    VideoContentType::UNSPECIFIED);
>   EXPECT_EQ(kInterframeDelayMs1,
>             statistics_proxy_->GetStats().interframe_delay_max_ms);
> 
>   fake_clock_.AdvanceTimeMilliseconds(kInterframeDelayMs2);
>-  statistics_proxy_->OnDecodedFrame(127u, VideoContentType::UNSPECIFIED);
>+  statistics_proxy_->OnDecodedFrame(127u, kWidth, kHeight,
>+                                    VideoContentType::UNSPECIFIED);
>   EXPECT_EQ(kInterframeDelayMs2,
>             statistics_proxy_->GetStats().interframe_delay_max_ms);
> 
>   fake_clock_.AdvanceTimeMilliseconds(kInterframeDelayMs3);
>-  statistics_proxy_->OnDecodedFrame(127u, VideoContentType::UNSPECIFIED);
>+  statistics_proxy_->OnDecodedFrame(127u, kWidth, kHeight,
>+                                    VideoContentType::UNSPECIFIED);
>   // kInterframeDelayMs3 is smaller than kInterframeDelayMs2.
>   EXPECT_EQ(kInterframeDelayMs2,
>             statistics_proxy_->GetStats().interframe_delay_max_ms);
>@@ -138,41 +204,46 @@ TEST_F(ReceiveStatisticsProxyTest, ReportInterframeDelayInWindow) {
>   const int64_t kInterframeDelayMs2 = 750;
>   const int64_t kInterframeDelayMs3 = 700;
>   EXPECT_EQ(-1, statistics_proxy_->GetStats().interframe_delay_max_ms);
>-  statistics_proxy_->OnDecodedFrame(3u, VideoContentType::UNSPECIFIED);
>+  statistics_proxy_->OnDecodedFrame(3u, kWidth, kHeight,
>+                                    VideoContentType::UNSPECIFIED);
>   EXPECT_EQ(-1, statistics_proxy_->GetStats().interframe_delay_max_ms);
> 
>   fake_clock_.AdvanceTimeMilliseconds(kInterframeDelayMs1);
>-  statistics_proxy_->OnDecodedFrame(127u, VideoContentType::UNSPECIFIED);
>+  statistics_proxy_->OnDecodedFrame(127u, kWidth, kHeight,
>+                                    VideoContentType::UNSPECIFIED);
>   EXPECT_EQ(kInterframeDelayMs1,
>             statistics_proxy_->GetStats().interframe_delay_max_ms);
> 
>   fake_clock_.AdvanceTimeMilliseconds(kInterframeDelayMs2);
>-  statistics_proxy_->OnDecodedFrame(127u, VideoContentType::UNSPECIFIED);
>+  statistics_proxy_->OnDecodedFrame(127u, kWidth, kHeight,
>+                                    VideoContentType::UNSPECIFIED);
>   // Still first delay is the maximum
>   EXPECT_EQ(kInterframeDelayMs1,
>             statistics_proxy_->GetStats().interframe_delay_max_ms);
> 
>   fake_clock_.AdvanceTimeMilliseconds(kInterframeDelayMs3);
>-  statistics_proxy_->OnDecodedFrame(127u, VideoContentType::UNSPECIFIED);
>+  statistics_proxy_->OnDecodedFrame(127u, kWidth, kHeight,
>+                                    VideoContentType::UNSPECIFIED);
>   // Now the first sample is out of the window, so the second is the maximum.
>   EXPECT_EQ(kInterframeDelayMs2,
>             statistics_proxy_->GetStats().interframe_delay_max_ms);
> }
> 
> TEST_F(ReceiveStatisticsProxyTest, OnDecodedFrameWithoutQpQpSumWontExist) {
>-  EXPECT_EQ(rtc::nullopt, statistics_proxy_->GetStats().qp_sum);
>-  statistics_proxy_->OnDecodedFrame(rtc::nullopt,
>+  EXPECT_EQ(absl::nullopt, statistics_proxy_->GetStats().qp_sum);
>+  statistics_proxy_->OnDecodedFrame(absl::nullopt, kWidth, kHeight,
>                                     VideoContentType::UNSPECIFIED);
>-  EXPECT_EQ(rtc::nullopt, statistics_proxy_->GetStats().qp_sum);
>+  EXPECT_EQ(absl::nullopt, statistics_proxy_->GetStats().qp_sum);
> }
> 
> TEST_F(ReceiveStatisticsProxyTest, OnDecodedFrameWithoutQpResetsQpSum) {
>-  EXPECT_EQ(rtc::nullopt, statistics_proxy_->GetStats().qp_sum);
>-  statistics_proxy_->OnDecodedFrame(3u, VideoContentType::UNSPECIFIED);
>+  EXPECT_EQ(absl::nullopt, statistics_proxy_->GetStats().qp_sum);
>+  statistics_proxy_->OnDecodedFrame(3u, kWidth, kHeight,
>+                                    VideoContentType::UNSPECIFIED);
>   EXPECT_EQ(3u, statistics_proxy_->GetStats().qp_sum);
>-  statistics_proxy_->OnDecodedFrame(rtc::nullopt,
>+  statistics_proxy_->OnDecodedFrame(absl::nullopt, kWidth, kHeight,
>                                     VideoContentType::UNSPECIFIED);
>-  EXPECT_EQ(rtc::nullopt, statistics_proxy_->GetStats().qp_sum);
>+  EXPECT_EQ(absl::nullopt, statistics_proxy_->GetStats().qp_sum);
> }
> 
> TEST_F(ReceiveStatisticsProxyTest, OnRenderedFrameIncreasesFramesRendered) {
>@@ -316,7 +387,7 @@ TEST_F(ReceiveStatisticsProxyTest,
>   const int64_t kLongEndToEndDelay = 100;
>   const uint32_t kExpectedRtpTimestamp = 2;
>   TimingFrameInfo info;
>-  rtc::Optional<TimingFrameInfo> result;
>+  absl::optional<TimingFrameInfo> result;
>   info.rtp_timestamp = kExpectedRtpTimestamp - 1;
>   info.capture_time_ms = 0;
>   info.decode_finish_ms = kShortEndToEndDelay;
>@@ -341,7 +412,7 @@ TEST_F(ReceiveStatisticsProxyTest, RespectsReportingIntervalForTimingFrames) {
>   const uint32_t kExpectedRtpTimestamp = 2;
>   const int64_t kShortDelayMs = 1000;
>   const int64_t kLongDelayMs = 10000;
>-  rtc::Optional<TimingFrameInfo> result;
>+  absl::optional<TimingFrameInfo> result;
>   info.rtp_timestamp = kExpectedRtpTimestamp;
>   info.capture_time_ms = 0;
>   info.decode_finish_ms = kShortEndToEndDelay;
>@@ -662,9 +733,6 @@ TEST_F(ReceiveStatisticsProxyTest, TimingHistogramsAreUpdated) {
> 
> TEST_F(ReceiveStatisticsProxyTest, DoesNotReportStaleFramerates) {
>   const int kDefaultFps = 30;
>-  const int kWidth = 320;
>-  const int kHeight = 240;
>-
>   rtc::scoped_refptr<VideoFrameBuffer> video_frame_buffer(
>       I420Buffer::Create(kWidth, kHeight));
>   VideoFrame frame(video_frame_buffer, kVideoRotation_0, 0);
>@@ -673,7 +741,7 @@ TEST_F(ReceiveStatisticsProxyTest, DoesNotReportStaleFramerates) {
>     // Since OnRenderedFrame is never called the fps in each sample will be 0,
>     // i.e. bad
>     frame.set_ntp_time_ms(fake_clock_.CurrentNtpInMilliseconds());
>-    statistics_proxy_->OnDecodedFrame(rtc::nullopt,
>+    statistics_proxy_->OnDecodedFrame(absl::nullopt, kWidth, kHeight,
>                                       VideoContentType::UNSPECIFIED);
>     statistics_proxy_->OnRenderedFrame(frame);
>     fake_clock_.AdvanceTimeMilliseconds(1000 / kDefaultFps);
>@@ -689,8 +757,6 @@ TEST_F(ReceiveStatisticsProxyTest, DoesNotReportStaleFramerates) {
> }
> 
> TEST_F(ReceiveStatisticsProxyTest, GetStatsReportsReceivedFrameStats) {
>-  const int kWidth = 160;
>-  const int kHeight = 120;
>   EXPECT_EQ(0, statistics_proxy_->GetStats().width);
>   EXPECT_EQ(0, statistics_proxy_->GetStats().height);
>   EXPECT_EQ(0u, statistics_proxy_->GetStats().frames_rendered);
>@@ -704,9 +770,6 @@ TEST_F(ReceiveStatisticsProxyTest, GetStatsReportsReceivedFrameStats) {
> 
> TEST_F(ReceiveStatisticsProxyTest,
>        ReceivedFrameHistogramsAreNotUpdatedForTooFewSamples) {
>-  const int kWidth = 160;
>-  const int kHeight = 120;
>-
>   for (int i = 0; i < kMinRequiredSamples - 1; ++i)
>     statistics_proxy_->OnRenderedFrame(CreateFrame(kWidth, kHeight));
> 
>@@ -718,9 +781,6 @@ TEST_F(ReceiveStatisticsProxyTest,
> }
> 
> TEST_F(ReceiveStatisticsProxyTest, ReceivedFrameHistogramsAreUpdated) {
>-  const int kWidth = 160;
>-  const int kHeight = 120;
>-
>   for (int i = 0; i < kMinRequiredSamples; ++i)
>     statistics_proxy_->OnRenderedFrame(CreateFrame(kWidth, kHeight));
> 
>@@ -735,6 +795,95 @@ TEST_F(ReceiveStatisticsProxyTest, ReceivedFrameHistogramsAreUpdated) {
>             metrics::NumEvents("WebRTC.Video.ReceivedHeightInPixels", kHeight));
> }
> 
>+TEST_F(ReceiveStatisticsProxyTest, ZeroDelayReportedIfFrameNotDelayed) {
>+  statistics_proxy_->OnDecodedFrame(absl::nullopt, kWidth, kHeight,
>+                                    VideoContentType::UNSPECIFIED);
>+
>+  // Frame not delayed, delayed frames to render: 0%.
>+  const int64_t kNowMs = fake_clock_.TimeInMilliseconds();
>+  statistics_proxy_->OnRenderedFrame(CreateFrameWithRenderTimeMs(kNowMs));
>+
>+  // Min run time has passed.
>+  fake_clock_.AdvanceTimeMilliseconds((metrics::kMinRunTimeInSeconds * 1000));
>+  statistics_proxy_.reset();
>+  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.DelayedFramesToRenderer"));
>+  EXPECT_EQ(1, metrics::NumEvents("WebRTC.Video.DelayedFramesToRenderer", 0));
>+  EXPECT_EQ(0, metrics::NumSamples(
>+                   "WebRTC.Video.DelayedFramesToRenderer_AvgDelayInMs"));
>+}
>+
>+TEST_F(ReceiveStatisticsProxyTest,
>+       DelayedFrameHistogramsAreNotUpdatedIfMinRuntimeHasNotPassed) {
>+  statistics_proxy_->OnDecodedFrame(absl::nullopt, kWidth, kHeight,
>+                                    VideoContentType::UNSPECIFIED);
>+
>+  // Frame not delayed, delayed frames to render: 0%.
>+  const int64_t kNowMs = fake_clock_.TimeInMilliseconds();
>+  statistics_proxy_->OnRenderedFrame(CreateFrameWithRenderTimeMs(kNowMs));
>+
>+  // Min run time has not passed.
>+  fake_clock_.AdvanceTimeMilliseconds((metrics::kMinRunTimeInSeconds * 1000) -
>+                                      1);
>+  statistics_proxy_.reset();
>+  EXPECT_EQ(0, metrics::NumSamples("WebRTC.Video.DelayedFramesToRenderer"));
>+  EXPECT_EQ(0, metrics::NumSamples(
>+                   "WebRTC.Video.DelayedFramesToRenderer_AvgDelayInMs"));
>+}
>+
>+TEST_F(ReceiveStatisticsProxyTest,
>+       DelayedFramesHistogramsAreNotUpdatedIfNoRenderedFrames) {
>+  statistics_proxy_->OnDecodedFrame(absl::nullopt, kWidth, kHeight,
>+                                    VideoContentType::UNSPECIFIED);
>+
>+  // Min run time has passed. No rendered frames.
>+  fake_clock_.AdvanceTimeMilliseconds((metrics::kMinRunTimeInSeconds * 1000));
>+  statistics_proxy_.reset();
>+  EXPECT_EQ(0, metrics::NumSamples("WebRTC.Video.DelayedFramesToRenderer"));
>+  EXPECT_EQ(0, metrics::NumSamples(
>+                   "WebRTC.Video.DelayedFramesToRenderer_AvgDelayInMs"));
>+}
>+
>+TEST_F(ReceiveStatisticsProxyTest, DelayReportedIfFrameIsDelayed) {
>+  statistics_proxy_->OnDecodedFrame(absl::nullopt, kWidth, kHeight,
>+                                    VideoContentType::UNSPECIFIED);
>+
>+  // Frame delayed 1 ms, delayed frames to render: 100%.
>+  const int64_t kNowMs = fake_clock_.TimeInMilliseconds();
>+  statistics_proxy_->OnRenderedFrame(CreateFrameWithRenderTimeMs(kNowMs - 1));
>+
>+  // Min run time has passed.
>+  fake_clock_.AdvanceTimeMilliseconds((metrics::kMinRunTimeInSeconds * 1000));
>+  statistics_proxy_.reset();
>+  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.DelayedFramesToRenderer"));
>+  EXPECT_EQ(1, metrics::NumEvents("WebRTC.Video.DelayedFramesToRenderer", 100));
>+  EXPECT_EQ(1, metrics::NumSamples(
>+                   "WebRTC.Video.DelayedFramesToRenderer_AvgDelayInMs"));
>+  EXPECT_EQ(1, metrics::NumEvents(
>+                   "WebRTC.Video.DelayedFramesToRenderer_AvgDelayInMs", 1));
>+}
>+
>+TEST_F(ReceiveStatisticsProxyTest, AverageDelayOfDelayedFramesIsReported) {
>+  statistics_proxy_->OnDecodedFrame(absl::nullopt, kWidth, kHeight,
>+                                    VideoContentType::UNSPECIFIED);
>+
>+  // Two frames delayed (6 ms, 10 ms), delayed frames to render: 50%.
>+  const int64_t kNowMs = fake_clock_.TimeInMilliseconds();
>+  statistics_proxy_->OnRenderedFrame(CreateFrameWithRenderTimeMs(kNowMs - 10));
>+  statistics_proxy_->OnRenderedFrame(CreateFrameWithRenderTimeMs(kNowMs - 6));
>+  statistics_proxy_->OnRenderedFrame(CreateFrameWithRenderTimeMs(kNowMs));
>+  statistics_proxy_->OnRenderedFrame(CreateFrameWithRenderTimeMs(kNowMs + 1));
>+
>+  // Min run time has passed.
>+  fake_clock_.AdvanceTimeMilliseconds((metrics::kMinRunTimeInSeconds * 1000));
>+  statistics_proxy_.reset();
>+  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.DelayedFramesToRenderer"));
>+  EXPECT_EQ(1, metrics::NumEvents("WebRTC.Video.DelayedFramesToRenderer", 50));
>+  EXPECT_EQ(1, metrics::NumSamples(
>+                   "WebRTC.Video.DelayedFramesToRenderer_AvgDelayInMs"));
>+  EXPECT_EQ(1, metrics::NumEvents(
>+                   "WebRTC.Video.DelayedFramesToRenderer_AvgDelayInMs", 8));
>+}
>+
> TEST_F(ReceiveStatisticsProxyTest,
>        RtcpHistogramsNotUpdatedIfMinRuntimeHasNotPassed) {
>   InsertFirstRtpPacket(kRemoteSsrc);
>@@ -788,12 +937,14 @@ TEST_P(ReceiveStatisticsProxyTest, InterFrameDelaysAreReported) {
>   const VideoContentType content_type = GetParam();
>   const int kInterFrameDelayMs = 33;
>   for (int i = 0; i < kMinRequiredSamples; ++i) {
>-    statistics_proxy_->OnDecodedFrame(rtc::nullopt, content_type);
>+    statistics_proxy_->OnDecodedFrame(absl::nullopt, kWidth, kHeight,
>+                                      content_type);
>     fake_clock_.AdvanceTimeMilliseconds(kInterFrameDelayMs);
>   }
>   // One extra with double the interval.
>   fake_clock_.AdvanceTimeMilliseconds(kInterFrameDelayMs);
>-  statistics_proxy_->OnDecodedFrame(rtc::nullopt, content_type);
>+  statistics_proxy_->OnDecodedFrame(absl::nullopt, kWidth, kHeight,
>+                                    content_type);
> 
>   statistics_proxy_.reset();
>   const int kExpectedInterFrame =
>@@ -821,16 +972,19 @@ TEST_P(ReceiveStatisticsProxyTest, InterFrameDelaysPercentilesAreReported) {
>   const int kLastFivePercentsSamples = kMinRequiredSamples * 5 / 100;
>   for (int i = 0; i <= kMinRequiredSamples - kLastFivePercentsSamples; ++i) {
>     fake_clock_.AdvanceTimeMilliseconds(kInterFrameDelayMs);
>-    statistics_proxy_->OnDecodedFrame(rtc::nullopt, content_type);
>+    statistics_proxy_->OnDecodedFrame(absl::nullopt, kWidth, kHeight,
>+                                      content_type);
>   }
>   // Last 5% of intervals are double in size.
>   for (int i = 0; i < kLastFivePercentsSamples; ++i) {
>     fake_clock_.AdvanceTimeMilliseconds(2 * kInterFrameDelayMs);
>-    statistics_proxy_->OnDecodedFrame(rtc::nullopt, content_type);
>+    statistics_proxy_->OnDecodedFrame(absl::nullopt, kWidth, kHeight,
>+                                      content_type);
>   }
>   // Final sample is outlier and 10 times as big.
>   fake_clock_.AdvanceTimeMilliseconds(10 * kInterFrameDelayMs);
>-  statistics_proxy_->OnDecodedFrame(rtc::nullopt, content_type);
>+  statistics_proxy_->OnDecodedFrame(absl::nullopt, kWidth, kHeight,
>+                                    content_type);
> 
>   statistics_proxy_.reset();
>   const int kExpectedInterFrame = kInterFrameDelayMs * 2;
>@@ -849,7 +1003,8 @@ TEST_P(ReceiveStatisticsProxyTest, MaxInterFrameDelayOnlyWithValidAverage) {
>   const VideoContentType content_type = GetParam();
>   const int kInterFrameDelayMs = 33;
>   for (int i = 0; i < kMinRequiredSamples; ++i) {
>-    statistics_proxy_->OnDecodedFrame(rtc::nullopt, content_type);
>+    statistics_proxy_->OnDecodedFrame(absl::nullopt, kWidth, kHeight,
>+                                      content_type);
>     fake_clock_.AdvanceTimeMilliseconds(kInterFrameDelayMs);
>   }
> 
>@@ -868,7 +1023,8 @@ TEST_P(ReceiveStatisticsProxyTest, MaxInterFrameDelayOnlyWithPause) {
>   const VideoContentType content_type = GetParam();
>   const int kInterFrameDelayMs = 33;
>   for (int i = 0; i <= kMinRequiredSamples; ++i) {
>-    statistics_proxy_->OnDecodedFrame(rtc::nullopt, content_type);
>+    statistics_proxy_->OnDecodedFrame(absl::nullopt, kWidth, kHeight,
>+                                      content_type);
>     fake_clock_.AdvanceTimeMilliseconds(kInterFrameDelayMs);
>   }
> 
>@@ -879,9 +1035,11 @@ TEST_P(ReceiveStatisticsProxyTest, MaxInterFrameDelayOnlyWithPause) {
> 
>   // Insert two more frames. The interval during the pause should be disregarded
>   // in the stats.
>-  statistics_proxy_->OnDecodedFrame(rtc::nullopt, content_type);
>+  statistics_proxy_->OnDecodedFrame(absl::nullopt, kWidth, kHeight,
>+                                    content_type);
>   fake_clock_.AdvanceTimeMilliseconds(kInterFrameDelayMs);
>-  statistics_proxy_->OnDecodedFrame(rtc::nullopt, content_type);
>+  statistics_proxy_->OnDecodedFrame(absl::nullopt, kWidth, kHeight,
>+                                    content_type);
> 
>   statistics_proxy_.reset();
>   if (videocontenttypehelpers::IsScreenshare(content_type)) {
>@@ -905,6 +1063,201 @@ TEST_P(ReceiveStatisticsProxyTest, MaxInterFrameDelayOnlyWithPause) {
>   }
> }
> 
>+TEST_P(ReceiveStatisticsProxyTest, FreezesAreReported) {
>+  const VideoContentType content_type = GetParam();
>+  const int kInterFrameDelayMs = 33;
>+  const int kFreezeDelayMs = 200;
>+  for (int i = 0; i < kMinRequiredSamples; ++i) {
>+    statistics_proxy_->OnDecodedFrame(absl::nullopt, kWidth, kHeight,
>+                                      content_type);
>+    fake_clock_.AdvanceTimeMilliseconds(kInterFrameDelayMs);
>+  }
>+  // Add extra freeze.
>+  fake_clock_.AdvanceTimeMilliseconds(kFreezeDelayMs);
>+  statistics_proxy_->OnDecodedFrame(absl::nullopt, kWidth, kHeight,
>+                                    content_type);
>+
>+  statistics_proxy_.reset();
>+  const int kExpectedTimeBetweenFreezes =
>+      kInterFrameDelayMs * (kMinRequiredSamples - 1);
>+  if (videocontenttypehelpers::IsScreenshare(content_type)) {
>+    EXPECT_EQ(
>+        kFreezeDelayMs + kInterFrameDelayMs,
>+        metrics::MinSample("WebRTC.Video.Screenshare.MeanFreezeDurationMs"));
>+    EXPECT_EQ(kExpectedTimeBetweenFreezes,
>+              metrics::MinSample(
>+                  "WebRTC.Video.Screenshare.MeanTimeBetweenFreezesMs"));
>+  } else {
>+    EXPECT_EQ(kFreezeDelayMs + kInterFrameDelayMs,
>+              metrics::MinSample("WebRTC.Video.MeanFreezeDurationMs"));
>+    EXPECT_EQ(kExpectedTimeBetweenFreezes,
>+              metrics::MinSample("WebRTC.Video.MeanTimeBetweenFreezesMs"));
>+  }
>+}
>+
>+TEST_P(ReceiveStatisticsProxyTest, PausesAreIgnored) {
>+  const VideoContentType content_type = GetParam();
>+  const int kInterFrameDelayMs = 33;
>+  const int kPauseDurationMs = 10000;
>+  for (int i = 0; i <= kMinRequiredSamples; ++i) {
>+    statistics_proxy_->OnDecodedFrame(absl::nullopt, kWidth, kHeight,
>+                                      content_type);
>+    fake_clock_.AdvanceTimeMilliseconds(kInterFrameDelayMs);
>+  }
>+  // Add a pause.
>+  fake_clock_.AdvanceTimeMilliseconds(kPauseDurationMs);
>+  statistics_proxy_->OnStreamInactive();
>+
>+  // Second playback interval with triple the length.
>+  for (int i = 0; i <= kMinRequiredSamples * 3; ++i) {
>+    statistics_proxy_->OnDecodedFrame(absl::nullopt, kWidth, kHeight,
>+                                      content_type);
>+    fake_clock_.AdvanceTimeMilliseconds(kInterFrameDelayMs);
>+  }
>+
>+  statistics_proxy_.reset();
>+  // Average of two playback intervals.
>+  const int kExpectedTimeBetweenFreezes =
>+      kInterFrameDelayMs * kMinRequiredSamples * 2;
>+  if (videocontenttypehelpers::IsScreenshare(content_type)) {
>+    EXPECT_EQ(-1, metrics::MinSample(
>+                      "WebRTC.Video.Screenshare.MeanFreezeDurationMs"));
>+    EXPECT_EQ(kExpectedTimeBetweenFreezes,
>+              metrics::MinSample(
>+                  "WebRTC.Video.Screenshare.MeanTimeBetweenFreezesMs"));
>+  } else {
>+    EXPECT_EQ(-1, metrics::MinSample("WebRTC.Video.MeanFreezeDurationMs"));
>+    EXPECT_EQ(kExpectedTimeBetweenFreezes,
>+              metrics::MinSample("WebRTC.Video.MeanTimeBetweenFreezesMs"));
>+  }
>+}
>+
>+TEST_P(ReceiveStatisticsProxyTest, ManyPausesAtTheBeginning) {
>+  const VideoContentType content_type = GetParam();
>+  const int kInterFrameDelayMs = 33;
>+  const int kPauseDurationMs = 10000;
>+  for (int i = 0; i <= kMinRequiredSamples; ++i) {
>+    statistics_proxy_->OnDecodedFrame(absl::nullopt, kWidth, kHeight,
>+                                      content_type);
>+    fake_clock_.AdvanceTimeMilliseconds(kInterFrameDelayMs);
>+
>+    statistics_proxy_->OnStreamInactive();
>+    fake_clock_.AdvanceTimeMilliseconds(kPauseDurationMs);
>+
>+    statistics_proxy_->OnDecodedFrame(absl::nullopt, kWidth, kHeight,
>+                                      content_type);
>+    fake_clock_.AdvanceTimeMilliseconds(kInterFrameDelayMs);
>+  }
>+
>+  statistics_proxy_.reset();
>+  // No freezes should be detected, as all long inter-frame delays were pauses.
>+  if (videocontenttypehelpers::IsScreenshare(content_type)) {
>+    EXPECT_EQ(-1, metrics::MinSample(
>+                      "WebRTC.Video.Screenshare.MeanFreezeDurationMs"));
>+  } else {
>+    EXPECT_EQ(-1, metrics::MinSample("WebRTC.Video.MeanFreezeDurationMs"));
>+  }
>+}
>+
>+TEST_P(ReceiveStatisticsProxyTest, TimeInHdReported) {
>+  const VideoContentType content_type = GetParam();
>+  const int kInterFrameDelayMs = 20;
>+  // HD frames.
>+  for (int i = 0; i < kMinRequiredSamples; ++i) {
>+    statistics_proxy_->OnDecodedFrame(absl::nullopt, kWidth, kHeight,
>+                                      content_type);
>+    fake_clock_.AdvanceTimeMilliseconds(kInterFrameDelayMs);
>+  }
>+  // SD frames.
>+  for (int i = 0; i < 2 * kMinRequiredSamples; ++i) {
>+    statistics_proxy_->OnDecodedFrame(absl::nullopt, kWidth / 2, kHeight / 2,
>+                                      content_type);
>+    fake_clock_.AdvanceTimeMilliseconds(kInterFrameDelayMs);
>+  }
>+  // Extra last frame.
>+  statistics_proxy_->OnDecodedFrame(absl::nullopt, kWidth / 2, kHeight / 2,
>+                                    content_type);
>+  statistics_proxy_.reset();
>+  const int kExpectedTimeInHdPercents = 33;
>+  if (videocontenttypehelpers::IsScreenshare(content_type)) {
>+    EXPECT_EQ(
>+        kExpectedTimeInHdPercents,
>+        metrics::MinSample("WebRTC.Video.Screenshare.TimeInHdPercentage"));
>+  } else {
>+    EXPECT_EQ(kExpectedTimeInHdPercents,
>+              metrics::MinSample("WebRTC.Video.TimeInHdPercentage"));
>+  }
>+}
>+
>+TEST_P(ReceiveStatisticsProxyTest, TimeInBlockyVideoReported) {
>+  const VideoContentType content_type = GetParam();
>+  const int kInterFrameDelayMs = 20;
>+  const int kHighQp = 80;
>+  const int kLowQp = 30;
>+  // High quality frames.
>+  for (int i = 0; i < kMinRequiredSamples; ++i) {
>+    statistics_proxy_->OnDecodedFrame(kLowQp, kWidth, kHeight, content_type);
>+    fake_clock_.AdvanceTimeMilliseconds(kInterFrameDelayMs);
>+  }
>+  // Blocky frames.
>+  for (int i = 0; i < 2 * kMinRequiredSamples; ++i) {
>+    statistics_proxy_->OnDecodedFrame(kHighQp, kWidth, kHeight, content_type);
>+    fake_clock_.AdvanceTimeMilliseconds(kInterFrameDelayMs);
>+  }
>+  // Extra last frame.
>+  statistics_proxy_->OnDecodedFrame(kHighQp, kWidth, kHeight, content_type);
>+  statistics_proxy_.reset();
>+  const int kExpectedTimeInHdPercents = 66;
>+  if (videocontenttypehelpers::IsScreenshare(content_type)) {
>+    EXPECT_EQ(kExpectedTimeInHdPercents,
>+              metrics::MinSample(
>+                  "WebRTC.Video.Screenshare.TimeInBlockyVideoPercentage"));
>+  } else {
>+    EXPECT_EQ(kExpectedTimeInHdPercents,
>+              metrics::MinSample("WebRTC.Video.TimeInBlockyVideoPercentage"));
>+  }
>+}
>+
>+TEST_P(ReceiveStatisticsProxyTest, DownscalesReported) {
>+  const VideoContentType content_type = GetParam();
>+  const int kInterFrameDelayMs = 1000;  // To ensure long enough call duration.
>+  const int kLowQp = 30;
>+
>+  statistics_proxy_->OnDecodedFrame(kLowQp, kWidth / 2, kHeight / 2,
>+                                    content_type);
>+  fake_clock_.AdvanceTimeMilliseconds(kInterFrameDelayMs);
>+
>+  // Downscale.
>+  statistics_proxy_->OnDecodedFrame(kLowQp, kWidth, kHeight, content_type);
>+  fake_clock_.AdvanceTimeMilliseconds(kInterFrameDelayMs);
>+
>+  statistics_proxy_->OnDecodedFrame(kLowQp, kWidth / 2, kHeight / 2,
>+                                    content_type);
>+  fake_clock_.AdvanceTimeMilliseconds(kInterFrameDelayMs);
>+
>+  statistics_proxy_->OnDecodedFrame(kLowQp, kWidth / 2, kHeight / 2,
>+                                    content_type);
>+  fake_clock_.AdvanceTimeMilliseconds(kInterFrameDelayMs);
>+
>+  // Downscale.
>+  statistics_proxy_->OnDecodedFrame(kLowQp, kWidth / 4, kHeight / 4,
>+                                    content_type);
>+  fake_clock_.AdvanceTimeMilliseconds(kInterFrameDelayMs);
>+
>+  statistics_proxy_.reset();
>+  const int kExpectedDownscales = 30;  // 2 per 5 seconds = 30 per minute.
>+  if (videocontenttypehelpers::IsScreenshare(content_type)) {
>+    EXPECT_EQ(
>+        kExpectedDownscales,
>+        metrics::MinSample(
>+            "WebRTC.Video.Screenshare.NumberResolutionDownswitchesPerMinute"));
>+  } else {
>+    EXPECT_EQ(kExpectedDownscales,
>+              metrics::MinSample(
>+                  "WebRTC.Video.NumberResolutionDownswitchesPerMinute"));
>+  }
>+}
>+
> TEST_P(ReceiveStatisticsProxyTest, StatsAreSlicedOnSimulcastAndExperiment) {
>   VideoContentType content_type = GetParam();
>   const uint8_t experiment_id = 1;
>@@ -915,13 +1268,15 @@ TEST_P(ReceiveStatisticsProxyTest, StatsAreSlicedOnSimulcastAndExperiment) {
>   videocontenttypehelpers::SetSimulcastId(&content_type, 1);
>   for (int i = 0; i <= kMinRequiredSamples; ++i) {
>     fake_clock_.AdvanceTimeMilliseconds(kInterFrameDelayMs1);
>-    statistics_proxy_->OnDecodedFrame(rtc::nullopt, content_type);
>+    statistics_proxy_->OnDecodedFrame(absl::nullopt, kWidth, kHeight,
>+                                      content_type);
>   }
> 
>   videocontenttypehelpers::SetSimulcastId(&content_type, 2);
>   for (int i = 0; i <= kMinRequiredSamples; ++i) {
>     fake_clock_.AdvanceTimeMilliseconds(kInterFrameDelayMs2);
>-    statistics_proxy_->OnDecodedFrame(rtc::nullopt, content_type);
>+    statistics_proxy_->OnDecodedFrame(absl::nullopt, kWidth, kHeight,
>+                                      content_type);
>   }
>   statistics_proxy_.reset();
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/replay.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/replay.cc
>index a79c3f984ff..a84c3fe85cd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/replay.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/replay.cc
>@@ -36,7 +36,6 @@
> #include "test/testsupport/frame_writer.h"
> #include "test/video_capturer.h"
> #include "test/video_renderer.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace {
> 
>@@ -107,8 +106,8 @@ static int RedPayloadTypeRtx() {
> 
> // Flag for SSRC.
> const std::string& DefaultSsrc() {
>-  static const std::string ssrc = std::to_string(
>-      test::CallTest::kVideoSendSsrcs[0]);
>+  static const std::string ssrc =
>+      std::to_string(test::CallTest::kVideoSendSsrcs[0]);
>   return ssrc;
> }
> DEFINE_string(ssrc, DefaultSsrc().c_str(), "Incoming SSRC");
>@@ -117,8 +116,8 @@ static uint32_t Ssrc() {
> }
> 
> const std::string& DefaultSsrcRtx() {
>-  static const std::string ssrc_rtx = std::to_string(
>-      test::CallTest::kSendRtxSsrcs[0]);
>+  static const std::string ssrc_rtx =
>+      std::to_string(test::CallTest::kSendRtxSsrcs[0]);
>   return ssrc_rtx;
> }
> DEFINE_string(ssrc_rtx, DefaultSsrcRtx().c_str(), "Incoming RTX SSRC");
>@@ -128,7 +127,9 @@ static uint32_t SsrcRtx() {
> 
> // Flag for abs-send-time id.
> DEFINE_int(abs_send_time_id, -1, "RTP extension ID for abs-send-time");
>-static int AbsSendTimeId() { return static_cast<int>(FLAG_abs_send_time_id); }
>+static int AbsSendTimeId() {
>+  return static_cast<int>(FLAG_abs_send_time_id);
>+}
> 
> // Flag for transmission-offset id.
> DEFINE_int(transmission_offset_id,
>@@ -157,7 +158,9 @@ static std::string DecoderBitstreamFilename() {
> 
> // Flag for video codec.
> DEFINE_string(codec, "VP8", "Video codec");
>-static std::string Codec() { return static_cast<std::string>(FLAG_codec); }
>+static std::string Codec() {
>+  return static_cast<std::string>(FLAG_codec);
>+}
> 
> DEFINE_bool(help, false, "Print this message.");
> }  // namespace flags
>@@ -197,7 +200,7 @@ class FileRenderPassthrough : public rtc::VideoSinkInterface<VideoFrame> {
>   size_t count_;
> };
> 
>-class DecoderBitstreamFileWriter : public EncodedFrameObserver {
>+class DecoderBitstreamFileWriter : public test::FakeDecoder {
>  public:
>   explicit DecoderBitstreamFileWriter(const char* filename)
>       : file_(fopen(filename, "wb")) {
>@@ -205,8 +208,16 @@ class DecoderBitstreamFileWriter : public EncodedFrameObserver {
>   }
>   ~DecoderBitstreamFileWriter() { fclose(file_); }
> 
>-  virtual void EncodedFrameCallback(const EncodedFrame& encoded_frame) {
>-    fwrite(encoded_frame.data_, 1, encoded_frame.length_, file_);
>+  int32_t Decode(const EncodedImage& encoded_frame,
>+                      bool /* missing_frames */,
>+                      const CodecSpecificInfo* /* codec_specific_info */,
>+                      int64_t /* render_time_ms */) override {
>+    if (fwrite(encoded_frame._buffer, 1, encoded_frame._length, file_)
>+        < encoded_frame._length) {
>+      RTC_LOG_ERR(LS_ERROR) << "fwrite of encoded frame failed.";
>+      return WEBRTC_VIDEO_CODEC_ERROR;
>+    }
>+    return WEBRTC_VIDEO_CODEC_OK;
>   }
> 
>  private:
>@@ -247,22 +258,15 @@ void RtpReplay() {
>   }
>   receive_config.renderer = &file_passthrough;
> 
>-  VideoSendStream::Config::EncoderSettings encoder_settings;
>-  encoder_settings.payload_name = flags::Codec();
>-  encoder_settings.payload_type = flags::MediaPayloadType();
>   VideoReceiveStream::Decoder decoder;
>-  std::unique_ptr<DecoderBitstreamFileWriter> bitstream_writer;
>+  decoder =
>+      test::CreateMatchingDecoder(flags::MediaPayloadType(), flags::Codec());
>   if (!flags::DecoderBitstreamFilename().empty()) {
>-    bitstream_writer.reset(new DecoderBitstreamFileWriter(
>-        flags::DecoderBitstreamFilename().c_str()));
>-    receive_config.pre_decode_callback = bitstream_writer.get();
>-  }
>-  decoder = test::CreateMatchingDecoder(encoder_settings);
>-  if (!flags::DecoderBitstreamFilename().empty()) {
>-    // Replace with a null decoder if we're writing the bitstream to a file
>+    // Replace decoder with file writer if we're writing the bitstream to a file
>     // instead.
>     delete decoder.decoder;
>-    decoder.decoder = new test::FakeNullDecoder();
>+    decoder.decoder = new DecoderBitstreamFileWriter(
>+        flags::DecoderBitstreamFilename().c_str());
>   }
>   receive_config.decoders.push_back(decoder);
> 
>@@ -309,7 +313,8 @@ void RtpReplay() {
>     ++num_packets;
>     switch (call->Receiver()->DeliverPacket(
>         webrtc::MediaType::VIDEO,
>-        rtc::CopyOnWriteBuffer(packet.data, packet.length), PacketTime())) {
>+        rtc::CopyOnWriteBuffer(packet.data, packet.length),
>+        /* packet_time_us */ -1)) {
>       case PacketReceiver::DELIVERY_OK:
>         break;
>       case PacketReceiver::DELIVERY_UNKNOWN_SSRC: {
>@@ -336,10 +341,9 @@ void RtpReplay() {
>   fprintf(stderr, "num_packets: %d\n", num_packets);
> 
>   for (std::map<uint32_t, int>::const_iterator it = unknown_packets.begin();
>-       it != unknown_packets.end();
>-       ++it) {
>-    fprintf(
>-        stderr, "Packets for unknown ssrc '%u': %d\n", it->first, it->second);
>+       it != unknown_packets.end(); ++it) {
>+    fprintf(stderr, "Packets for unknown ssrc '%u': %d\n", it->first,
>+            it->second);
>   }
> 
>   call->DestroyVideoReceiveStream(receive_stream);
>@@ -368,8 +372,8 @@ int main(int argc, char* argv[]) {
>   RTC_CHECK(ValidateSsrc(webrtc::flags::FLAG_ssrc));
>   RTC_CHECK(ValidateSsrc(webrtc::flags::FLAG_ssrc_rtx));
>   RTC_CHECK(ValidateRtpHeaderExtensionId(webrtc::flags::FLAG_abs_send_time_id));
>-  RTC_CHECK(ValidateRtpHeaderExtensionId(
>-      webrtc::flags::FLAG_transmission_offset_id));
>+  RTC_CHECK(
>+      ValidateRtpHeaderExtensionId(webrtc::flags::FLAG_transmission_offset_id));
>   RTC_CHECK(ValidateInputFilenameNotEmpty(webrtc::flags::FLAG_input_file));
> 
>   webrtc::test::RunTest(webrtc::RtpReplay);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/report_block_stats.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/report_block_stats.cc
>index 4726a467456..6005043dfa3 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/report_block_stats.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/report_block_stats.cc
>@@ -10,6 +10,8 @@
> 
> #include "video/report_block_stats.h"
> 
>+#include <algorithm>
>+
> namespace webrtc {
> 
> namespace {
>@@ -19,16 +21,15 @@ int FractionLost(uint32_t num_lost_sequence_numbers,
>     return 0;
>   }
>   return ((num_lost_sequence_numbers * 255) + (num_sequence_numbers / 2)) /
>-      num_sequence_numbers;
>+         num_sequence_numbers;
> }
> }  // namespace
> 
>-
> // Helper class for rtcp statistics.
> ReportBlockStats::ReportBlockStats()
>-    : num_sequence_numbers_(0),
>-      num_lost_sequence_numbers_(0) {
>-}
>+    : num_sequence_numbers_(0), num_lost_sequence_numbers_(0) {}
>+
>+ReportBlockStats::~ReportBlockStats() {}
> 
> void ReportBlockStats::Store(const RtcpStatistics& rtcp_stats,
>                              uint32_t remote_ssrc,
>@@ -43,8 +44,8 @@ void ReportBlockStats::Store(const RtcpStatistics& rtcp_stats,
>   block.source_ssrc = source_ssrc;
>   uint32_t num_sequence_numbers = 0;
>   uint32_t num_lost_sequence_numbers = 0;
>-  StoreAndAddPacketIncrement(
>-      block, &num_sequence_numbers, &num_lost_sequence_numbers);
>+  StoreAndAddPacketIncrement(block, &num_sequence_numbers,
>+                             &num_lost_sequence_numbers);
> }
> 
> RTCPReportBlock ReportBlockStats::AggregateAndStore(
>@@ -59,8 +60,7 @@ RTCPReportBlock ReportBlockStats::AggregateAndStore(
>   for (; report_block != report_blocks.end(); ++report_block) {
>     aggregate.packets_lost += report_block->packets_lost;
>     aggregate.jitter += report_block->jitter;
>-    StoreAndAddPacketIncrement(*report_block,
>-                               &num_sequence_numbers,
>+    StoreAndAddPacketIncrement(*report_block, &num_sequence_numbers,
>                                &num_lost_sequence_numbers);
>   }
> 
>@@ -105,9 +105,8 @@ int ReportBlockStats::FractionLostInPercent() const {
>   if (num_sequence_numbers_ == 0) {
>     return -1;
>   }
>-  return FractionLost(
>-      num_lost_sequence_numbers_, num_sequence_numbers_) * 100 / 255;
>+  return FractionLost(num_lost_sequence_numbers_, num_sequence_numbers_) * 100 /
>+         255;
> }
> 
> }  // namespace webrtc
>-
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/report_block_stats.h b/Source/ThirdParty/libwebrtc/Source/webrtc/video/report_block_stats.h
>index 033ba7deb54..241fec76a69 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/report_block_stats.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/report_block_stats.h
>@@ -25,7 +25,7 @@ class ReportBlockStats {
>   typedef std::map<uint32_t, RTCPReportBlock> ReportBlockMap;
>   typedef std::vector<RTCPReportBlock> ReportBlockVector;
>   ReportBlockStats();
>-  ~ReportBlockStats() {}
>+  ~ReportBlockStats();
> 
>   // Updates stats and stores report blocks.
>   // Returns an aggregate of the |report_blocks|.
>@@ -59,4 +59,3 @@ class ReportBlockStats {
> }  // namespace webrtc
> 
> #endif  // VIDEO_REPORT_BLOCK_STATS_H_
>-
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/report_block_stats_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/report_block_stats_unittest.cc
>index 983dcfcf344..f80ae161600 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/report_block_stats_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/report_block_stats_unittest.cc
>@@ -8,8 +8,8 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "test/gtest.h"
> #include "video/report_block_stats.h"
>+#include "test/gtest.h"
> 
> namespace webrtc {
> 
>@@ -54,8 +54,7 @@ class ReportBlockStatsTest : public ::testing::Test {
>     ssrc12block2_.push_back(block2_2_);
>   }
> 
>-  RtcpStatistics RtcpReportBlockToRtcpStatistics(
>-      const RTCPReportBlock& stats) {
>+  RtcpStatistics RtcpReportBlockToRtcpStatistics(const RTCPReportBlock& stats) {
>     RtcpStatistics block;
>     block.packets_lost = stats.packets_lost;
>     block.fraction_lost = stats.fraction_lost;
>@@ -83,7 +82,7 @@ TEST_F(ReportBlockStatsTest, AggregateAndStore_NoSsrc) {
>   std::vector<RTCPReportBlock> empty;
>   RTCPReportBlock aggregated = stats.AggregateAndStore(empty);
>   EXPECT_EQ(0U, aggregated.fraction_lost);
>-  EXPECT_EQ(0U, aggregated.packets_lost);
>+  EXPECT_EQ(0, aggregated.packets_lost);
>   EXPECT_EQ(0U, aggregated.jitter);
>   EXPECT_EQ(0U, aggregated.extended_highest_sequence_number);
> }
>@@ -93,13 +92,13 @@ TEST_F(ReportBlockStatsTest, AggregateAndStore_OneSsrc) {
>   RTCPReportBlock aggregated = stats.AggregateAndStore(ssrc1block1_);
>   // One ssrc, no aggregation done.
>   EXPECT_EQ(123U, aggregated.fraction_lost);
>-  EXPECT_EQ(10U, aggregated.packets_lost);
>+  EXPECT_EQ(10, aggregated.packets_lost);
>   EXPECT_EQ(777U, aggregated.jitter);
>   EXPECT_EQ(24000U, aggregated.extended_highest_sequence_number);
> 
>   aggregated = stats.AggregateAndStore(ssrc1block2_);
>   EXPECT_EQ(0U, aggregated.fraction_lost);
>-  EXPECT_EQ(15U, aggregated.packets_lost);
>+  EXPECT_EQ(15, aggregated.packets_lost);
>   EXPECT_EQ(222U, aggregated.jitter);
>   EXPECT_EQ(24100U, aggregated.extended_highest_sequence_number);
> 
>@@ -111,14 +110,14 @@ TEST_F(ReportBlockStatsTest, AggregateAndStore_TwoSsrcs) {
>   ReportBlockStats stats;
>   RTCPReportBlock aggregated = stats.AggregateAndStore(ssrc12block1_);
>   EXPECT_EQ(0U, aggregated.fraction_lost);
>-  EXPECT_EQ(10U + 111U, aggregated.packets_lost);
>+  EXPECT_EQ(10 + 111, aggregated.packets_lost);
>   EXPECT_EQ((777U + 555U) / 2, aggregated.jitter);
>   EXPECT_EQ(0U, aggregated.extended_highest_sequence_number);
> 
>   aggregated = stats.AggregateAndStore(ssrc12block2_);
>   // fl: 255 * ((15-10) + (136-111)) / ((24100-24000) + (8800-8500)) = 19
>   EXPECT_EQ(19U, aggregated.fraction_lost);
>-  EXPECT_EQ(15U + 136U, aggregated.packets_lost);
>+  EXPECT_EQ(15 + 136, aggregated.packets_lost);
>   EXPECT_EQ((222U + 888U) / 2, aggregated.jitter);
>   EXPECT_EQ(0U, aggregated.extended_highest_sequence_number);
> 
>@@ -143,4 +142,3 @@ TEST_F(ReportBlockStatsTest, StoreAndGetFractionLost) {
> }
> 
> }  // namespace webrtc
>-
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/rtp_streams_synchronizer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/rtp_streams_synchronizer.cc
>index c330a4434a1..e46a2200c55 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/rtp_streams_synchronizer.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/rtp_streams_synchronizer.cc
>@@ -25,10 +25,9 @@ bool UpdateMeasurements(StreamSynchronization::Measurements* stream,
>   stream->latest_timestamp = info.latest_received_capture_timestamp;
>   stream->latest_receive_time_ms = info.latest_receive_time_ms;
>   bool new_rtcp_sr = false;
>-  if (!stream->rtp_to_ntp.UpdateMeasurements(info.capture_time_ntp_secs,
>-                                             info.capture_time_ntp_frac,
>-                                             info.capture_time_source_clock,
>-                                             &new_rtcp_sr)) {
>+  if (!stream->rtp_to_ntp.UpdateMeasurements(
>+          info.capture_time_ntp_secs, info.capture_time_ntp_frac,
>+          info.capture_time_source_clock, &new_rtcp_sr)) {
>     return false;
>   }
>   return true;
>@@ -63,7 +62,7 @@ int64_t RtpStreamsSynchronizer::TimeUntilNextProcess() {
>   RTC_DCHECK_RUN_ON(&process_thread_checker_);
>   const int64_t kSyncIntervalMs = 1000;
>   return kSyncIntervalMs -
>-      (rtc::TimeNanos() - last_sync_time_) / rtc::kNumNanosecsPerMillisec;
>+         (rtc::TimeNanos() - last_sync_time_) / rtc::kNumNanosecsPerMillisec;
> }
> 
> void RtpStreamsSynchronizer::Process() {
>@@ -76,13 +75,13 @@ void RtpStreamsSynchronizer::Process() {
>   }
>   RTC_DCHECK(sync_.get());
> 
>-  rtc::Optional<Syncable::Info> audio_info = syncable_audio_->GetInfo();
>+  absl::optional<Syncable::Info> audio_info = syncable_audio_->GetInfo();
>   if (!audio_info || !UpdateMeasurements(&audio_measurement_, *audio_info)) {
>     return;
>   }
> 
>   int64_t last_video_receive_ms = video_measurement_.latest_receive_time_ms;
>-  rtc::Optional<Syncable::Info> video_info = syncable_video_->GetInfo();
>+  absl::optional<Syncable::Info> video_info = syncable_video_->GetInfo();
>   if (!video_info || !UpdateMeasurements(&video_measurement_, *video_info)) {
>     return;
>   }
>@@ -100,18 +99,16 @@ void RtpStreamsSynchronizer::Process() {
>   }
> 
>   TRACE_COUNTER1("webrtc", "SyncCurrentVideoDelay",
>-      video_info->current_delay_ms);
>+                 video_info->current_delay_ms);
>   TRACE_COUNTER1("webrtc", "SyncCurrentAudioDelay",
>-      audio_info->current_delay_ms);
>+                 audio_info->current_delay_ms);
>   TRACE_COUNTER1("webrtc", "SyncRelativeDelay", relative_delay_ms);
>   int target_audio_delay_ms = 0;
>   int target_video_delay_ms = video_info->current_delay_ms;
>   // Calculate the necessary extra audio delay and desired total video
>   // delay to get the streams in sync.
>-  if (!sync_->ComputeDelays(relative_delay_ms,
>-                            audio_info->current_delay_ms,
>-                            &target_audio_delay_ms,
>-                            &target_video_delay_ms)) {
>+  if (!sync_->ComputeDelays(relative_delay_ms, audio_info->current_delay_ms,
>+                            &target_audio_delay_ms, &target_video_delay_ms)) {
>     return;
>   }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/rtp_streams_synchronizer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/video/rtp_streams_synchronizer.h
>index 0d3b205b419..1a24391d2a8 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/rtp_streams_synchronizer.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/rtp_streams_synchronizer.h
>@@ -25,10 +25,6 @@ namespace webrtc {
> 
> class Syncable;
> 
>-namespace vcm {
>-class VideoReceiver;
>-}  // namespace vcm
>-
> class RtpStreamsSynchronizer : public Module {
>  public:
>   explicit RtpStreamsSynchronizer(Syncable* syncable_video);
>@@ -58,7 +54,7 @@ class RtpStreamsSynchronizer : public Module {
>   StreamSynchronization::Measurements video_measurement_ RTC_GUARDED_BY(crit_);
> 
>   rtc::ThreadChecker process_thread_checker_;
>-  int64_t last_sync_time_ RTC_ACCESS_ON(&process_thread_checker_);
>+  int64_t last_sync_time_ RTC_GUARDED_BY(&process_thread_checker_);
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/rtp_video_stream_receiver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/rtp_video_stream_receiver.cc
>index 238eaf34740..32a4c67eb48 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/rtp_video_stream_receiver.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/rtp_video_stream_receiver.cc
>@@ -14,29 +14,32 @@
> #include <utility>
> #include <vector>
> 
>-#include "call/video_config.h"
>+#include "absl/memory/memory.h"
>+
> #include "common_types.h"  // NOLINT(build/include)
> #include "media/base/mediaconstants.h"
> #include "modules/pacing/packet_router.h"
> #include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
> #include "modules/rtp_rtcp/include/receive_statistics.h"
> #include "modules/rtp_rtcp/include/rtp_cvo.h"
>-#include "modules/rtp_rtcp/include/rtp_receiver.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp.h"
> #include "modules/rtp_rtcp/include/ulpfec_receiver.h"
>+#include "modules/rtp_rtcp/source/rtp_format.h"
> #include "modules/rtp_rtcp/source/rtp_header_extensions.h"
> #include "modules/rtp_rtcp/source/rtp_packet_received.h"
>+#include "modules/rtp_rtcp/source/rtp_rtcp_config.h"
> #include "modules/video_coding/frame_object.h"
> #include "modules/video_coding/h264_sprop_parameter_sets.h"
> #include "modules/video_coding/h264_sps_pps_tracker.h"
>+#include "modules/video_coding/nack_module.h"
> #include "modules/video_coding/packet_buffer.h"
> #include "modules/video_coding/video_coding_impl.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/location.h"
> #include "rtc_base/logging.h"
>+#include "rtc_base/system/fallthrough.h"
> #include "system_wrappers/include/field_trial.h"
> #include "system_wrappers/include/metrics.h"
>-#include "system_wrappers/include/timestamp_extrapolator.h"
> #include "video/receive_statistics_proxy.h"
> 
> namespace webrtc {
>@@ -46,7 +49,7 @@ namespace {
> //                 crbug.com/752886
> constexpr int kPacketBufferStartSize = 512;
> constexpr int kPacketBufferMaxSixe = 2048;
>-}
>+}  // namespace
> 
> std::unique_ptr<RtpRtcp> CreateRtpRtcpModule(
>     ReceiveStatistics* receive_statistics,
>@@ -90,18 +93,13 @@ RtpVideoStreamReceiver::RtpVideoStreamReceiver(
>     ProcessThread* process_thread,
>     NackSender* nack_sender,
>     KeyFrameRequestSender* keyframe_request_sender,
>-    video_coding::OnCompleteFrameCallback* complete_frame_callback,
>-    VCMTiming* timing)
>+    video_coding::OnCompleteFrameCallback* complete_frame_callback)
>     : clock_(Clock::GetRealTimeClock()),
>       config_(*config),
>       packet_router_(packet_router),
>       process_thread_(process_thread),
>       ntp_estimator_(clock_),
>       rtp_header_extensions_(config_.rtp.extensions),
>-      rtp_receiver_(RtpReceiver::CreateVideoReceiver(clock_,
>-                                                     this,
>-                                                     this,
>-                                                     &rtp_payload_registry_)),
>       rtp_receive_statistics_(rtp_receive_statistics),
>       ulpfec_receiver_(UlpfecReceiver::Create(config->rtp.remote_ssrc, this)),
>       receiving_(false),
>@@ -113,7 +111,6 @@ RtpVideoStreamReceiver::RtpVideoStreamReceiver(
>                                     packet_router)),
>       complete_frame_callback_(complete_frame_callback),
>       keyframe_request_sender_(keyframe_request_sender),
>-      timing_(timing),
>       has_received_frame_(false) {
>   constexpr bool remb_candidate = true;
>   packet_router_->AddReceiveRtpModule(rtp_rtcp_.get(), remb_candidate);
>@@ -139,27 +136,6 @@ RtpVideoStreamReceiver::RtpVideoStreamReceiver(
>                                            : kDefaultMaxReorderingThreshold;
>   rtp_receive_statistics_->SetMaxReorderingThreshold(max_reordering_threshold);
> 
>-  if (config_.rtp.rtx_ssrc) {
>-    // Needed for rtp_payload_registry_.RtxEnabled().
>-    rtp_payload_registry_.SetRtxSsrc(config_.rtp.rtx_ssrc);
>-  }
>-
>-  if (IsUlpfecEnabled()) {
>-    VideoCodec ulpfec_codec = {};
>-    ulpfec_codec.codecType = kVideoCodecULPFEC;
>-    strncpy(ulpfec_codec.plName, "ulpfec", sizeof(ulpfec_codec.plName));
>-    ulpfec_codec.plType = config_.rtp.ulpfec_payload_type;
>-    RTC_CHECK(AddReceiveCodec(ulpfec_codec));
>-  }
>-
>-  if (IsRedEnabled()) {
>-    VideoCodec red_codec = {};
>-    red_codec.codecType = kVideoCodecRED;
>-    strncpy(red_codec.plName, "red", sizeof(red_codec.plName));
>-    red_codec.plType = config_.rtp.red_payload_type;
>-    RTC_CHECK(AddReceiveCodec(red_codec));
>-  }
>-
>   if (config_.rtp.rtcp_xr.receiver_reference_time_report)
>     rtp_rtcp_->SetRtcpXrRrtrStatus(true);
> 
>@@ -192,32 +168,28 @@ RtpVideoStreamReceiver::~RtpVideoStreamReceiver() {
>   UpdateHistograms();
> }
> 
>-bool RtpVideoStreamReceiver::AddReceiveCodec(
>+void RtpVideoStreamReceiver::AddReceiveCodec(
>     const VideoCodec& video_codec,
>     const std::map<std::string, std::string>& codec_params) {
>-  pt_codec_params_.insert(make_pair(video_codec.plType, codec_params));
>-  return AddReceiveCodec(video_codec);
>+  pt_codec_type_.emplace(video_codec.plType, video_codec.codecType);
>+  pt_codec_params_.emplace(video_codec.plType, codec_params);
> }
> 
>-bool RtpVideoStreamReceiver::AddReceiveCodec(const VideoCodec& video_codec) {
>-  int8_t old_pltype = -1;
>-  if (rtp_payload_registry_.ReceivePayloadType(video_codec, &old_pltype) !=
>-      -1) {
>-    rtp_payload_registry_.DeRegisterReceivePayload(old_pltype);
>+absl::optional<Syncable::Info> RtpVideoStreamReceiver::GetSyncInfo() const {
>+  if (!last_received_rtp_timestamp_ || !last_received_rtp_system_time_ms_) {
>+    return absl::nullopt;
>   }
>-  return rtp_payload_registry_.RegisterReceivePayload(video_codec) == 0;
>-}
>-
>-uint32_t RtpVideoStreamReceiver::GetRemoteSsrc() const {
>-  return config_.rtp.remote_ssrc;
>-}
>-
>-int RtpVideoStreamReceiver::GetCsrcs(uint32_t* csrcs) const {
>-  return rtp_receiver_->CSRCs(csrcs);
>-}
>+  Syncable::Info info;
>+  if (rtp_rtcp_->RemoteNTP(&info.capture_time_ntp_secs,
>+                           &info.capture_time_ntp_frac, nullptr, nullptr,
>+                           &info.capture_time_source_clock) != 0) {
>+    return absl::nullopt;
>+  }
>+  info.latest_received_capture_timestamp = *last_received_rtp_timestamp_;
>+  info.latest_receive_time_ms = *last_received_rtp_system_time_ms_;
> 
>-RtpReceiver* RtpVideoStreamReceiver::GetRtpReceiver() const {
>-  return rtp_receiver_.get();
>+  // Leaves info.current_delay_ms uninitialized.
>+  return info;
> }
> 
> int32_t RtpVideoStreamReceiver::OnReceivedPayloadData(
>@@ -232,12 +204,8 @@ int32_t RtpVideoStreamReceiver::OnReceivedPayloadData(
>       nack_module_ ? nack_module_->OnReceivedPacket(packet) : -1;
>   packet.receive_time_ms = clock_->TimeInMilliseconds();
> 
>-  // In the case of a video stream without picture ids and no rtx the
>-  // RtpFrameReferenceFinder will need to know about padding to
>-  // correctly calculate frame references.
>   if (packet.sizeBytes == 0) {
>-    reference_finder_->PaddingReceived(packet.seqNum);
>-    packet_buffer_->PaddingReceived(packet.seqNum);
>+    NotifyReceiverOfEmptyPacket(packet.seqNum);
>     return 0;
>   }
> 
>@@ -253,7 +221,7 @@ int32_t RtpVideoStreamReceiver::OnReceivedPayloadData(
>     switch (tracker_.CopyAndFixBitstream(&packet)) {
>       case video_coding::H264SpsPpsTracker::kRequestKeyframe:
>         keyframe_request_sender_->RequestKeyFrame();
>-        FALLTHROUGH();
>+        RTC_FALLTHROUGH();
>       case video_coding::H264SpsPpsTracker::kDrop:
>         return 0;
>       case video_coding::H264SpsPpsTracker::kInsert:
>@@ -275,22 +243,20 @@ void RtpVideoStreamReceiver::OnRecoveredPacket(const uint8_t* rtp_packet,
>   RtpPacketReceived packet;
>   if (!packet.Parse(rtp_packet, rtp_packet_length))
>     return;
>+  if (packet.PayloadType() == config_.rtp.red_payload_type) {
>+    RTC_LOG(LS_WARNING) << "Discarding recovered packet with RED encapsulation";
>+    return;
>+  }
>+
>   packet.IdentifyExtensions(rtp_header_extensions_);
>   packet.set_payload_type_frequency(kVideoPayloadTypeFrequency);
>+  // TODO(nisse): UlpfecReceiverImpl::ProcessReceivedFec passes both
>+  // original (decapsulated) media packets and recovered packets to
>+  // this callback. We need a way to distinguish, for setting
>+  // packet.recovered() correctly. Ideally, move RED decapsulation out
>+  // of the Ulpfec implementation.
> 
>-  RTPHeader header;
>-  packet.GetHeader(&header);
>-  ReceivePacket(rtp_packet, rtp_packet_length, header);
>-}
>-
>-// TODO(pbos): Remove as soon as audio can handle a changing payload type
>-// without this callback.
>-int32_t RtpVideoStreamReceiver::OnInitializeDecoder(
>-    const int payload_type,
>-    const SdpAudioFormat& audio_format,
>-    const uint32_t rate) {
>-  RTC_NOTREACHED();
>-  return 0;
>+  ReceivePacket(packet);
> }
> 
> // This method handles both regular RTP packets and packets recovered
>@@ -304,6 +270,9 @@ void RtpVideoStreamReceiver::OnRtpPacket(const RtpPacketReceived& packet) {
> 
>   if (!packet.recovered()) {
>     int64_t now_ms = clock_->TimeInMilliseconds();
>+    // TODO(nisse): Exclude out-of-order packets?
>+    last_received_rtp_timestamp_ = packet.Timestamp();
>+    last_received_rtp_system_time_ms_ = now_ms;
> 
>     // Periodically log the RTP header of incoming packets.
>     if (now_ms - last_packet_log_ms_ > kPacketLogIntervalMs) {
>@@ -326,27 +295,18 @@ void RtpVideoStreamReceiver::OnRtpPacket(const RtpPacketReceived& packet) {
>     }
>   }
> 
>-  // TODO(nisse): Delete use of GetHeader, but needs refactoring of
>-  // ReceivePacket and IncomingPacket methods below.
>-  RTPHeader header;
>-  packet.GetHeader(&header);
>-
>-  header.payload_type_frequency = kVideoPayloadTypeFrequency;
>+  ReceivePacket(packet);
> 
>-  bool in_order = IsPacketInOrder(header);
>-  if (!packet.recovered()) {
>-    // TODO(nisse): Why isn't this done for recovered packets?
>-    rtp_payload_registry_.SetIncomingPayloadType(header);
>-  }
>-  ReceivePacket(packet.data(), packet.size(), header);
>   // Update receive statistics after ReceivePacket.
>   // Receive statistics will be reset if the payload type changes (make sure
>   // that the first packet is included in the stats).
>   if (!packet.recovered()) {
>+    RTPHeader header;
>+    packet.GetHeader(&header);
>     // TODO(nisse): We should pass a recovered flag to stats, to aid
>     // fixing bug bugs.webrtc.org/6339.
>-    rtp_receive_statistics_->IncomingPacket(
>-        header, packet.size(), IsPacketRetransmitted(header, in_order));
>+    rtp_receive_statistics_->IncomingPacket(header, packet.size(),
>+                                            IsPacketRetransmitted(header));
>   }
> 
>   for (RtpPacketSinkInterface* secondary_sink : secondary_sinks_) {
>@@ -362,10 +322,6 @@ bool RtpVideoStreamReceiver::IsUlpfecEnabled() const {
>   return config_.rtp.ulpfec_payload_type != -1;
> }
> 
>-bool RtpVideoStreamReceiver::IsRedEnabled() const {
>-  return config_.rtp.red_payload_type != -1;
>-}
>-
> bool RtpVideoStreamReceiver::IsRetransmissionsEnabled() const {
>   return config_.rtp.nack.rtp_history_ms > 0;
> }
>@@ -388,33 +344,31 @@ void RtpVideoStreamReceiver::OnReceivedFrame(
>       keyframe_request_sender_->RequestKeyFrame();
>   }
> 
>-  if (!frame->delayed_by_retransmission())
>-    timing_->IncomingTimestamp(frame->timestamp, clock_->TimeInMilliseconds());
>   reference_finder_->ManageFrame(std::move(frame));
> }
> 
> void RtpVideoStreamReceiver::OnCompleteFrame(
>-    std::unique_ptr<video_coding::FrameObject> frame) {
>+    std::unique_ptr<video_coding::EncodedFrame> frame) {
>   {
>     rtc::CritScope lock(&last_seq_num_cs_);
>     video_coding::RtpFrameObject* rtp_frame =
>         static_cast<video_coding::RtpFrameObject*>(frame.get());
>-    last_seq_num_for_pic_id_[rtp_frame->picture_id] = rtp_frame->last_seq_num();
>+    last_seq_num_for_pic_id_[rtp_frame->id.picture_id] =
>+        rtp_frame->last_seq_num();
>   }
>   complete_frame_callback_->OnCompleteFrame(std::move(frame));
> }
> 
>-void RtpVideoStreamReceiver::OnRttUpdate(int64_t avg_rtt_ms,
>-                                         int64_t max_rtt_ms) {
>+void RtpVideoStreamReceiver::UpdateRtt(int64_t max_rtt_ms) {
>   if (nack_module_)
>     nack_module_->UpdateRtt(max_rtt_ms);
> }
> 
>-rtc::Optional<int64_t> RtpVideoStreamReceiver::LastReceivedPacketMs() const {
>+absl::optional<int64_t> RtpVideoStreamReceiver::LastReceivedPacketMs() const {
>   return packet_buffer_->LastReceivedPacketMs();
> }
> 
>-rtc::Optional<int64_t> RtpVideoStreamReceiver::LastReceivedKeyframePacketMs()
>+absl::optional<int64_t> RtpVideoStreamReceiver::LastReceivedKeyframePacketMs()
>     const {
>   return packet_buffer_->LastReceivedKeyframePacketMs();
> }
>@@ -440,78 +394,102 @@ void RtpVideoStreamReceiver::RemoveSecondarySink(
>   secondary_sinks_.erase(it);
> }
> 
>-void RtpVideoStreamReceiver::ReceivePacket(const uint8_t* packet,
>-                                           size_t packet_length,
>-                                           const RTPHeader& header) {
>-  if (rtp_payload_registry_.IsRed(header)) {
>-    ParseAndHandleEncapsulatingHeader(packet, packet_length, header);
>+void RtpVideoStreamReceiver::ReceivePacket(const RtpPacketReceived& packet) {
>+  if (packet.payload_size() == 0) {
>+    // Keep-alive packet.
>+    // TODO(nisse): Could drop empty packets earlier, but need to figure out how
>+    // they should be counted in stats.
>+    return;
>+  }
>+  if (packet.PayloadType() == config_.rtp.red_payload_type) {
>+    RTPHeader header;
>+    packet.GetHeader(&header);
>+    ParseAndHandleEncapsulatingHeader(packet.data(), packet.size(), header);
>+    return;
>+  }
>+
>+  const auto codec_type_it = pt_codec_type_.find(packet.PayloadType());
>+  if (codec_type_it == pt_codec_type_.end()) {
>+    return;
>+  }
>+  auto depacketizer =
>+      absl::WrapUnique(RtpDepacketizer::Create(codec_type_it->second));
>+
>+  if (!depacketizer) {
>+    RTC_LOG(LS_ERROR) << "Failed to create depacketizer.";
>     return;
>   }
>-  const uint8_t* payload = packet + header.headerLength;
>-  assert(packet_length >= header.headerLength);
>-  size_t payload_length = packet_length - header.headerLength;
>-  const auto pl =
>-      rtp_payload_registry_.PayloadTypeToPayload(header.payloadType);
>-  if (pl) {
>-    rtp_receiver_->IncomingRtpPacket(header, payload, payload_length,
>-                                     pl->typeSpecific);
>+  RtpDepacketizer::ParsedPayload parsed_payload;
>+  if (!depacketizer->Parse(&parsed_payload, packet.payload().data(),
>+                           packet.payload().size())) {
>+    RTC_LOG(LS_WARNING) << "Failed parsing payload.";
>+    return;
>   }
>+
>+  WebRtcRTPHeader webrtc_rtp_header = {};
>+  packet.GetHeader(&webrtc_rtp_header.header);
>+
>+  webrtc_rtp_header.frameType = parsed_payload.frame_type;
>+  webrtc_rtp_header.video_header() = parsed_payload.video_header();
>+  webrtc_rtp_header.video_header().rotation = kVideoRotation_0;
>+  webrtc_rtp_header.video_header().content_type = VideoContentType::UNSPECIFIED;
>+  webrtc_rtp_header.video_header().video_timing.flags =
>+      VideoSendTiming::kInvalid;
>+  webrtc_rtp_header.video_header().playout_delay.min_ms = -1;
>+  webrtc_rtp_header.video_header().playout_delay.max_ms = -1;
>+
>+  // Retrieve the video rotation information.
>+  packet.GetExtension<VideoOrientation>(
>+      &webrtc_rtp_header.video_header().rotation);
>+
>+  packet.GetExtension<VideoContentTypeExtension>(
>+      &webrtc_rtp_header.video_header().content_type);
>+  packet.GetExtension<VideoTimingExtension>(
>+      &webrtc_rtp_header.video_header().video_timing);
>+  packet.GetExtension<PlayoutDelayLimits>(
>+      &webrtc_rtp_header.video_header().playout_delay);
>+
>+  OnReceivedPayloadData(parsed_payload.payload, parsed_payload.payload_length,
>+                        &webrtc_rtp_header);
> }
> 
> void RtpVideoStreamReceiver::ParseAndHandleEncapsulatingHeader(
>-    const uint8_t* packet, size_t packet_length, const RTPHeader& header) {
>+    const uint8_t* packet,
>+    size_t packet_length,
>+    const RTPHeader& header) {
>   RTC_DCHECK_CALLED_SEQUENTIALLY(&worker_task_checker_);
>-  if (rtp_payload_registry_.IsRed(header)) {
>-    int8_t ulpfec_pt = rtp_payload_registry_.ulpfec_payload_type();
>-    if (packet[header.headerLength] == ulpfec_pt) {
>+  if (header.payloadType == config_.rtp.red_payload_type &&
>+      packet_length > header.headerLength + header.paddingLength) {
>+    if (packet[header.headerLength] == config_.rtp.ulpfec_payload_type) {
>       rtp_receive_statistics_->FecPacketReceived(header, packet_length);
>       // Notify video_receiver about received FEC packets to avoid NACKing these
>       // packets.
>       NotifyReceiverOfFecPacket(header);
>     }
>-    if (ulpfec_receiver_->AddReceivedRedPacket(header, packet, packet_length,
>-                                               ulpfec_pt) != 0) {
>+    if (ulpfec_receiver_->AddReceivedRedPacket(
>+            header, packet, packet_length, config_.rtp.ulpfec_payload_type) !=
>+        0) {
>       return;
>     }
>     ulpfec_receiver_->ProcessReceivedFec();
>   }
> }
> 
>+// In the case of a video stream without picture ids and no rtx the
>+// RtpFrameReferenceFinder will need to know about padding to
>+// correctly calculate frame references.
>+void RtpVideoStreamReceiver::NotifyReceiverOfEmptyPacket(uint16_t seq_num) {
>+  reference_finder_->PaddingReceived(seq_num);
>+  packet_buffer_->PaddingReceived(seq_num);
>+}
>+
> void RtpVideoStreamReceiver::NotifyReceiverOfFecPacket(
>     const RTPHeader& header) {
>-  int8_t last_media_payload_type =
>-      rtp_payload_registry_.last_received_media_payload_type();
>-  if (last_media_payload_type < 0) {
>-    RTC_LOG(LS_WARNING) << "Failed to get last media payload type.";
>-    return;
>-  }
>-  // Fake an empty media packet.
>-  WebRtcRTPHeader rtp_header = {};
>-  rtp_header.header = header;
>-  rtp_header.header.payloadType = last_media_payload_type;
>-  rtp_header.header.paddingLength = 0;
>-  const auto pl =
>-      rtp_payload_registry_.PayloadTypeToPayload(last_media_payload_type);
>-  if (!pl) {
>-    RTC_LOG(LS_WARNING) << "Failed to get payload specifics.";
>-    return;
>-  }
>-  rtp_header.type.Video.codec = pl->typeSpecific.video_payload().videoCodecType;
>-  rtp_header.type.Video.rotation = kVideoRotation_0;
>-  if (header.extension.hasVideoRotation) {
>-    rtp_header.type.Video.rotation = header.extension.videoRotation;
>-  }
>-  rtp_header.type.Video.content_type = VideoContentType::UNSPECIFIED;
>-  if (header.extension.hasVideoContentType) {
>-    rtp_header.type.Video.content_type = header.extension.videoContentType;
>-  }
>-  rtp_header.type.Video.video_timing = {0u, 0u, 0u, 0u, 0u, 0u, false};
>-  if (header.extension.has_video_timing) {
>-    rtp_header.type.Video.video_timing = header.extension.video_timing;
>+  if (nack_module_) {
>+    nack_module_->OnReceivedPacket(header.sequenceNumber,
>+                                   /* is_keyframe = */ false);
>   }
>-  rtp_header.type.Video.playout_delay = header.extension.playout_delay;
>-
>-  OnReceivedPayloadData(nullptr, 0, &rtp_header);
>+  NotifyReceiverOfEmptyPacket(header.sequenceNumber);
> }
> 
> bool RtpVideoStreamReceiver::DeliverRtcp(const uint8_t* rtcp_packet,
>@@ -525,7 +503,7 @@ bool RtpVideoStreamReceiver::DeliverRtcp(const uint8_t* rtcp_packet,
>   rtp_rtcp_->IncomingRtcpPacket(rtcp_packet, rtcp_packet_length);
> 
>   int64_t rtt = 0;
>-  rtp_rtcp_->RTT(rtp_receiver_->SSRC(), &rtt, nullptr, nullptr, nullptr);
>+  rtp_rtcp_->RTT(config_.rtp.remote_ssrc, &rtt, nullptr, nullptr, nullptr);
>   if (rtt == 0) {
>     // Waiting for valid rtt.
>     return true;
>@@ -588,6 +566,10 @@ void RtpVideoStreamReceiver::SignalNetworkState(NetworkState state) {
>                                                : RtcpMode::kOff);
> }
> 
>+int RtpVideoStreamReceiver::GetUniqueFramesSeen() const {
>+  return packet_buffer_->GetUniqueFramesSeen();
>+}
>+
> void RtpVideoStreamReceiver::StartReceive() {
>   RTC_DCHECK_CALLED_SEQUENTIALLY(&worker_task_checker_);
>   receiving_ = true;
>@@ -598,28 +580,16 @@ void RtpVideoStreamReceiver::StopReceive() {
>   receiving_ = false;
> }
> 
>-bool RtpVideoStreamReceiver::IsPacketInOrder(const RTPHeader& header) const {
>-  StreamStatistician* statistician =
>-      rtp_receive_statistics_->GetStatistician(header.ssrc);
>-  if (!statistician)
>-    return false;
>-  return statistician->IsPacketInOrder(header.sequenceNumber);
>-}
>-
>-bool RtpVideoStreamReceiver::IsPacketRetransmitted(const RTPHeader& header,
>-                                                   bool in_order) const {
>+bool RtpVideoStreamReceiver::IsPacketRetransmitted(
>+    const RTPHeader& header) const {
>   // Retransmissions are handled separately if RTX is enabled.
>-  if (rtp_payload_registry_.RtxEnabled())
>+  if (config_.rtp.rtx_ssrc != 0)
>     return false;
>   StreamStatistician* statistician =
>       rtp_receive_statistics_->GetStatistician(header.ssrc);
>   if (!statistician)
>     return false;
>-  // Check if this is a retransmission.
>-  int64_t min_rtt = 0;
>-  rtp_rtcp_->RTT(config_.rtp.remote_ssrc, nullptr, nullptr, &min_rtt, nullptr);
>-  return !in_order &&
>-      statistician->IsRetransmitOfOldPacket(header, min_rtt);
>+  return statistician->IsRetransmitOfOldPacket(header);
> }
> 
> void RtpVideoStreamReceiver::UpdateHistograms() {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/rtp_video_stream_receiver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/video/rtp_video_stream_receiver.h
>index aa5b7817898..b647bae8184 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/rtp_video_stream_receiver.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/rtp_video_stream_receiver.h
>@@ -17,13 +17,16 @@
> #include <string>
> #include <vector>
> 
>+#include "absl/types/optional.h"
>+
>+#include "api/video_codecs/video_codec.h"
> #include "call/rtp_packet_sink_interface.h"
>+#include "call/syncable.h"
> #include "call/video_receive_stream.h"
> #include "modules/include/module_common_types.h"
> #include "modules/rtp_rtcp/include/receive_statistics.h"
> #include "modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
> #include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
>-#include "modules/rtp_rtcp/include/rtp_payload_registry.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
> #include "modules/video_coding/h264_sps_pps_tracker.h"
>@@ -34,7 +37,6 @@
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/numerics/sequence_number_util.h"
> #include "rtc_base/sequenced_task_checker.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>@@ -48,21 +50,14 @@ class RtpPacketReceived;
> class RtpReceiver;
> class Transport;
> class UlpfecReceiver;
>-class VCMTiming;
>-
>-namespace vcm {
>-class VideoReceiver;
>-}  // namespace vcm
> 
> class RtpVideoStreamReceiver : public RtpData,
>                                public RecoveredPacketReceiver,
>-                               public RtpFeedback,
>                                public RtpPacketSinkInterface,
>                                public VCMFrameTypeCallback,
>                                public VCMPacketRequestCallback,
>                                public video_coding::OnReceivedFrameCallback,
>-                               public video_coding::OnCompleteFrameCallback,
>-                               public CallStatsObserver {
>+                               public video_coding::OnCompleteFrameCallback {
>  public:
>   RtpVideoStreamReceiver(
>       Transport* transport,
>@@ -74,21 +69,18 @@ class RtpVideoStreamReceiver : public RtpData,
>       ProcessThread* process_thread,
>       NackSender* nack_sender,
>       KeyFrameRequestSender* keyframe_request_sender,
>-      video_coding::OnCompleteFrameCallback* complete_frame_callback,
>-      VCMTiming* timing);
>+      video_coding::OnCompleteFrameCallback* complete_frame_callback);
>   ~RtpVideoStreamReceiver();
> 
>-  bool AddReceiveCodec(const VideoCodec& video_codec,
>+  void AddReceiveCodec(const VideoCodec& video_codec,
>                        const std::map<std::string, std::string>& codec_params);
>-  uint32_t GetRemoteSsrc() const;
>-  int GetCsrcs(uint32_t* csrcs) const;
>-
>-  RtpReceiver* GetRtpReceiver() const;
>-  RtpRtcp* rtp_rtcp() const { return rtp_rtcp_.get(); }
> 
>   void StartReceive();
>   void StopReceive();
> 
>+  // Produces the transport-related timestamps; current_delay_ms is left unset.
>+  absl::optional<Syncable::Info> GetSyncInfo() const;
>+
>   bool DeliverRtcp(const uint8_t* rtcp_packet, size_t rtcp_packet_length);
> 
>   void FrameContinuous(int64_t seq_num);
>@@ -97,6 +89,9 @@ class RtpVideoStreamReceiver : public RtpData,
> 
>   void SignalNetworkState(NetworkState state);
> 
>+  // Returns number of different frames seen in the packet buffer.
>+  int GetUniqueFramesSeen() const;
>+
>   // Implements RtpPacketSinkInterface.
>   void OnRtpPacket(const RtpPacketReceived& packet) override;
> 
>@@ -107,13 +102,6 @@ class RtpVideoStreamReceiver : public RtpData,
>   // Implements RecoveredPacketReceiver.
>   void OnRecoveredPacket(const uint8_t* packet, size_t packet_length) override;
> 
>-  // Implements RtpFeedback.
>-  int32_t OnInitializeDecoder(int payload_type,
>-                              const SdpAudioFormat& audio_format,
>-                              uint32_t rate) override;
>-  void OnIncomingSSRCChanged(uint32_t ssrc) override {}
>-  void OnIncomingCSRCChanged(uint32_t CSRC, bool added) override {}
>-
>   // Implements VCMFrameTypeCallback.
>   int32_t RequestKeyFrame() override;
> 
>@@ -132,12 +120,13 @@ class RtpVideoStreamReceiver : public RtpData,
> 
>   // Implements OnCompleteFrameCallback.
>   void OnCompleteFrame(
>-      std::unique_ptr<video_coding::FrameObject> frame) override;
>+      std::unique_ptr<video_coding::EncodedFrame> frame) override;
> 
>-  void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) override;
>+  // Called by VideoReceiveStream when stats are updated.
>+  void UpdateRtt(int64_t max_rtt_ms);
> 
>-  rtc::Optional<int64_t> LastReceivedPacketMs() const;
>-  rtc::Optional<int64_t> LastReceivedKeyframePacketMs() const;
>+  absl::optional<int64_t> LastReceivedPacketMs() const;
>+  absl::optional<int64_t> LastReceivedKeyframePacketMs() const;
> 
>   // RtpDemuxer only forwards a given RTP packet to one sink. However, some
>   // sinks, such as FlexFEC, might wish to be informed of all of the packets
>@@ -147,18 +136,17 @@ class RtpVideoStreamReceiver : public RtpData,
>   void RemoveSecondarySink(const RtpPacketSinkInterface* sink);
> 
>  private:
>-  bool AddReceiveCodec(const VideoCodec& video_codec);
>-  void ReceivePacket(const uint8_t* packet,
>-                     size_t packet_length,
>-                     const RTPHeader& header);
>-  // Parses and handles for instance RTX and RED headers.
>+  // Entry point doing non-stats work for a received packet. Called
>+  // for the same packet both before and after RED decapsulation.
>+  void ReceivePacket(const RtpPacketReceived& packet);
>+  // Parses and handles RED headers.
>   // This function assumes that it's being called from only one thread.
>   void ParseAndHandleEncapsulatingHeader(const uint8_t* packet,
>                                          size_t packet_length,
>                                          const RTPHeader& header);
>+  void NotifyReceiverOfEmptyPacket(uint16_t seq_num);
>   void NotifyReceiverOfFecPacket(const RTPHeader& header);
>-  bool IsPacketInOrder(const RTPHeader& header) const;
>-  bool IsPacketRetransmitted(const RTPHeader& header, bool in_order) const;
>+  bool IsPacketRetransmitted(const RTPHeader& header) const;
>   void UpdateHistograms();
>   bool IsRedEnabled() const;
>   void InsertSpsPpsIntoTracker(uint8_t payload_type);
>@@ -170,10 +158,8 @@ class RtpVideoStreamReceiver : public RtpData,
>   ProcessThread* const process_thread_;
> 
>   RemoteNtpTimeEstimator ntp_estimator_;
>-  RTPPayloadRegistry rtp_payload_registry_;
> 
>   RtpHeaderExtensionMap rtp_header_extensions_;
>-  const std::unique_ptr<RtpReceiver> rtp_receiver_;
>   ReceiveStatistics* const rtp_receive_statistics_;
>   std::unique_ptr<UlpfecReceiver> ulpfec_receiver_;
> 
>@@ -186,7 +172,6 @@ class RtpVideoStreamReceiver : public RtpData,
>   // Members for the new jitter buffer experiment.
>   video_coding::OnCompleteFrameCallback* complete_frame_callback_;
>   KeyFrameRequestSender* keyframe_request_sender_;
>-  VCMTiming* timing_;
>   std::unique_ptr<NackModule> nack_module_;
>   rtc::scoped_refptr<video_coding::PacketBuffer> packet_buffer_;
>   std::unique_ptr<video_coding::RtpFrameReferenceFinder> reference_finder_;
>@@ -194,6 +179,10 @@ class RtpVideoStreamReceiver : public RtpData,
>   std::map<int64_t, uint16_t> last_seq_num_for_pic_id_
>       RTC_GUARDED_BY(last_seq_num_cs_);
>   video_coding::H264SpsPpsTracker tracker_;
>+
>+  absl::optional<uint32_t> last_received_rtp_timestamp_;
>+  absl::optional<int64_t> last_received_rtp_system_time_ms_;
>+  std::map<uint8_t, VideoCodecType> pt_codec_type_;
>   // TODO(johan): Remove pt_codec_params_ once
>   // https://bugs.chromium.org/p/webrtc/issues/detail?id=6883 is resolved.
>   // Maps a payload type to a map of out-of-band supplied codec parameters.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/rtp_video_stream_receiver_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/rtp_video_stream_receiver_unittest.cc
>index 297218205f2..d74d4628d0d 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/rtp_video_stream_receiver_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/rtp_video_stream_receiver_unittest.cc
>@@ -8,9 +8,10 @@
>  *  be found in the AUTHORS file in the root of the source tree.
>  */
> 
>-#include "test/gtest.h"
> #include "test/gmock.h"
>+#include "test/gtest.h"
> 
>+#include "absl/memory/memory.h"
> #include "common_video/h264/h264_common.h"
> #include "media/base/mediaconstants.h"
> #include "modules/pacing/packet_router.h"
>@@ -20,10 +21,8 @@
> #include "modules/video_coding/include/video_coding_defines.h"
> #include "modules/video_coding/packet.h"
> #include "modules/video_coding/rtp_frame_reference_finder.h"
>-#include "modules/video_coding/timing.h"
> #include "rtc_base/bytebuffer.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/ptr_util.h"
> #include "system_wrappers/include/clock.h"
> #include "system_wrappers/include/field_trial_default.h"
> #include "test/field_trial.h"
>@@ -61,14 +60,14 @@ class MockOnCompleteFrameCallback
>  public:
>   MockOnCompleteFrameCallback() : buffer_(rtc::ByteBuffer::ORDER_NETWORK) {}
> 
>-  MOCK_METHOD1(DoOnCompleteFrame, void(video_coding::FrameObject* frame));
>+  MOCK_METHOD1(DoOnCompleteFrame, void(video_coding::EncodedFrame* frame));
>   MOCK_METHOD1(DoOnCompleteFrameFailNullptr,
>-               void(video_coding::FrameObject* frame));
>+               void(video_coding::EncodedFrame* frame));
>   MOCK_METHOD1(DoOnCompleteFrameFailLength,
>-               void(video_coding::FrameObject* frame));
>+               void(video_coding::EncodedFrame* frame));
>   MOCK_METHOD1(DoOnCompleteFrameFailBitstream,
>-               void(video_coding::FrameObject* frame));
>-  void OnCompleteFrame(std::unique_ptr<video_coding::FrameObject> frame) {
>+               void(video_coding::EncodedFrame* frame));
>+  void OnCompleteFrame(std::unique_ptr<video_coding::EncodedFrame> frame) {
>     if (!frame) {
>       DoOnCompleteFrameFailNullptr(nullptr);
>       return;
>@@ -103,7 +102,7 @@ constexpr uint16_t kSequenceNumber = 222;
> std::unique_ptr<RtpPacketReceived> CreateRtpPacketReceived(
>     uint32_t ssrc = kSsrc,
>     uint16_t sequence_number = kSequenceNumber) {
>-  auto packet = rtc::MakeUnique<RtpPacketReceived>();
>+  auto packet = absl::make_unique<RtpPacketReceived>();
>   packet->SetSsrc(ssrc);
>   packet->SetSequenceNumber(sequence_number);
>   return packet;
>@@ -122,24 +121,22 @@ class RtpVideoStreamReceiverTest : public testing::Test {
>   explicit RtpVideoStreamReceiverTest(std::string field_trials)
>       : override_field_trials_(field_trials),
>         config_(CreateConfig()),
>-        timing_(Clock::GetRealTimeClock()),
>         process_thread_(ProcessThread::Create("TestThread")) {}
> 
>   void SetUp() {
>     rtp_receive_statistics_ =
>-        rtc::WrapUnique(ReceiveStatistics::Create(Clock::GetRealTimeClock()));
>-    rtp_video_stream_receiver_ = rtc::MakeUnique<RtpVideoStreamReceiver>(
>+        absl::WrapUnique(ReceiveStatistics::Create(Clock::GetRealTimeClock()));
>+    rtp_video_stream_receiver_ = absl::make_unique<RtpVideoStreamReceiver>(
>         &mock_transport_, nullptr, &packet_router_, &config_,
>         rtp_receive_statistics_.get(), nullptr, process_thread_.get(),
>-        &mock_nack_sender_,
>-        &mock_key_frame_request_sender_, &mock_on_complete_frame_callback_,
>-        &timing_);
>+        &mock_nack_sender_, &mock_key_frame_request_sender_,
>+        &mock_on_complete_frame_callback_);
>   }
> 
>   WebRtcRTPHeader GetDefaultPacket() {
>-    WebRtcRTPHeader packet;
>-    memset(&packet, 0, sizeof(packet));
>-    packet.type.Video.codec = kRtpVideoH264;
>+    WebRtcRTPHeader packet = {};
>+    packet.video_header().codec = kVideoCodecH264;
>+    packet.video_header().video_type_header.emplace<RTPVideoHeaderH264>();
>     return packet;
>   }
> 
>@@ -154,8 +151,9 @@ class RtpVideoStreamReceiverTest : public testing::Test {
>     info.pps_id = -1;
>     data->push_back(H264::NaluType::kSps);
>     data->push_back(sps_id);
>-    packet->type.Video.codecHeader.H264
>-        .nalus[packet->type.Video.codecHeader.H264.nalus_length++] = info;
>+    auto& h264 =
>+        absl::get<RTPVideoHeaderH264>(packet->video_header().video_type_header);
>+    h264.nalus[h264.nalus_length++] = info;
>   }
> 
>   void AddPps(WebRtcRTPHeader* packet,
>@@ -168,8 +166,9 @@ class RtpVideoStreamReceiverTest : public testing::Test {
>     info.pps_id = pps_id;
>     data->push_back(H264::NaluType::kPps);
>     data->push_back(pps_id);
>-    packet->type.Video.codecHeader.H264
>-        .nalus[packet->type.Video.codecHeader.H264.nalus_length++] = info;
>+    auto& h264 =
>+        absl::get<RTPVideoHeaderH264>(packet->video_header().video_type_header);
>+    h264.nalus[h264.nalus_length++] = info;
>   }
> 
>   void AddIdr(WebRtcRTPHeader* packet, int pps_id) {
>@@ -177,8 +176,9 @@ class RtpVideoStreamReceiverTest : public testing::Test {
>     info.type = H264::NaluType::kIdr;
>     info.sps_id = -1;
>     info.pps_id = pps_id;
>-    packet->type.Video.codecHeader.H264
>-        .nalus[packet->type.Video.codecHeader.H264.nalus_length++] = info;
>+    auto& h264 =
>+        absl::get<RTPVideoHeaderH264>(packet->video_header().video_type_header);
>+    h264.nalus[h264.nalus_length++] = info;
>   }
> 
>  protected:
>@@ -196,21 +196,19 @@ class RtpVideoStreamReceiverTest : public testing::Test {
>   MockTransport mock_transport_;
>   MockOnCompleteFrameCallback mock_on_complete_frame_callback_;
>   PacketRouter packet_router_;
>-  VCMTiming timing_;
>   std::unique_ptr<ProcessThread> process_thread_;
>   std::unique_ptr<ReceiveStatistics> rtp_receive_statistics_;
>   std::unique_ptr<RtpVideoStreamReceiver> rtp_video_stream_receiver_;
> };
> 
> TEST_F(RtpVideoStreamReceiverTest, GenericKeyFrame) {
>-  WebRtcRTPHeader rtp_header;
>+  WebRtcRTPHeader rtp_header = {};
>   const std::vector<uint8_t> data({1, 2, 3, 4});
>-  memset(&rtp_header, 0, sizeof(rtp_header));
>   rtp_header.header.sequenceNumber = 1;
>   rtp_header.header.markerBit = 1;
>-  rtp_header.type.Video.is_first_packet_in_frame = true;
>+  rtp_header.video_header().is_first_packet_in_frame = true;
>   rtp_header.frameType = kVideoFrameKey;
>-  rtp_header.type.Video.codec = kRtpVideoGeneric;
>+  rtp_header.video_header().codec = kVideoCodecGeneric;
>   mock_on_complete_frame_callback_.AppendExpectedBitstream(data.data(),
>                                                            data.size());
>   EXPECT_CALL(mock_on_complete_frame_callback_, DoOnCompleteFrame(_));
>@@ -218,15 +216,56 @@ TEST_F(RtpVideoStreamReceiverTest, GenericKeyFrame) {
>                                                     &rtp_header);
> }
> 
>+TEST_F(RtpVideoStreamReceiverTest, NoInfiniteRecursionOnEncapsulatedRedPacket) {
>+  const uint8_t kRedPayloadType = 125;
>+  VideoCodec codec;
>+  codec.plType = kRedPayloadType;
>+  rtp_video_stream_receiver_->AddReceiveCodec(codec, {});
>+  const std::vector<uint8_t> data({
>+      0x80,              // RTP version.
>+      kRedPayloadType,   // Payload type.
>+      0, 0, 0, 0, 0, 0,  // Don't care.
>+      0, 0, 0x4, 0x57,   // SSRC
>+      kRedPayloadType,   // RED header.
>+      0, 0, 0, 0, 0      // Don't care.
>+  });
>+  RtpPacketReceived packet;
>+  EXPECT_TRUE(packet.Parse(data.data(), data.size()));
>+  rtp_video_stream_receiver_->StartReceive();
>+  rtp_video_stream_receiver_->OnRtpPacket(packet);
>+}
>+
>+TEST_F(RtpVideoStreamReceiverTest,
>+       DropsPacketWithRedPayloadTypeAndEmptyPayload) {
>+  const uint8_t kRedPayloadType = 125;
>+  config_.rtp.red_payload_type = kRedPayloadType;
>+  SetUp();  // re-create rtp_video_stream_receiver with red payload type.
>+  // clang-format off
>+  const uint8_t data[] = {
>+      0x80,              // RTP version.
>+      kRedPayloadType,   // Payload type.
>+      0, 0, 0, 0, 0, 0,  // Don't care.
>+      0, 0, 0x4, 0x57,   // SSRC
>+      // Empty rtp payload.
>+  };
>+  // clang-format on
>+  RtpPacketReceived packet;
>+  // Manually convert to CopyOnWriteBuffer to be sure capacity == size
>+  // and asan bot can catch read buffer overflow.
>+  EXPECT_TRUE(packet.Parse(rtc::CopyOnWriteBuffer(data)));
>+  rtp_video_stream_receiver_->StartReceive();
>+  rtp_video_stream_receiver_->OnRtpPacket(packet);
>+  // Expect asan doesn't find anything.
>+}
>+
> TEST_F(RtpVideoStreamReceiverTest, GenericKeyFrameBitstreamError) {
>-  WebRtcRTPHeader rtp_header;
>+  WebRtcRTPHeader rtp_header = {};
>   const std::vector<uint8_t> data({1, 2, 3, 4});
>-  memset(&rtp_header, 0, sizeof(rtp_header));
>   rtp_header.header.sequenceNumber = 1;
>   rtp_header.header.markerBit = 1;
>-  rtp_header.type.Video.is_first_packet_in_frame = true;
>+  rtp_header.video_header().is_first_packet_in_frame = true;
>   rtp_header.frameType = kVideoFrameKey;
>-  rtp_header.type.Video.codec = kRtpVideoGeneric;
>+  rtp_header.video_header().codec = kVideoCodecGeneric;
>   constexpr uint8_t expected_bitsteam[] = {1, 2, 3, 0xff};
>   mock_on_complete_frame_callback_.AppendExpectedBitstream(
>       expected_bitsteam, sizeof(expected_bitsteam));
>@@ -253,7 +292,7 @@ TEST_P(RtpVideoStreamReceiverTestH264, InBandSpsPps) {
>   WebRtcRTPHeader sps_packet = GetDefaultPacket();
>   AddSps(&sps_packet, 0, &sps_data);
>   sps_packet.header.sequenceNumber = 0;
>-  sps_packet.type.Video.is_first_packet_in_frame = true;
>+  sps_packet.video_header().is_first_packet_in_frame = true;
>   mock_on_complete_frame_callback_.AppendExpectedBitstream(
>       kH264StartCode, sizeof(kH264StartCode));
>   mock_on_complete_frame_callback_.AppendExpectedBitstream(sps_data.data(),
>@@ -265,7 +304,7 @@ TEST_P(RtpVideoStreamReceiverTestH264, InBandSpsPps) {
>   WebRtcRTPHeader pps_packet = GetDefaultPacket();
>   AddPps(&pps_packet, 0, 1, &pps_data);
>   pps_packet.header.sequenceNumber = 1;
>-  pps_packet.type.Video.is_first_packet_in_frame = true;
>+  pps_packet.video_header().is_first_packet_in_frame = true;
>   mock_on_complete_frame_callback_.AppendExpectedBitstream(
>       kH264StartCode, sizeof(kH264StartCode));
>   mock_on_complete_frame_callback_.AppendExpectedBitstream(pps_data.data(),
>@@ -276,7 +315,7 @@ TEST_P(RtpVideoStreamReceiverTestH264, InBandSpsPps) {
>   std::vector<uint8_t> idr_data;
>   WebRtcRTPHeader idr_packet = GetDefaultPacket();
>   AddIdr(&idr_packet, 1);
>-  idr_packet.type.Video.is_first_packet_in_frame = true;
>+  idr_packet.video_header().is_first_packet_in_frame = true;
>   idr_packet.header.sequenceNumber = 2;
>   idr_packet.header.markerBit = 1;
>   idr_packet.frameType = kVideoFrameKey;
>@@ -316,12 +355,12 @@ TEST_P(RtpVideoStreamReceiverTestH264, OutOfBandFmtpSpsPps) {
>   WebRtcRTPHeader idr_packet = GetDefaultPacket();
>   AddIdr(&idr_packet, 0);
>   idr_packet.header.payloadType = kPayloadType;
>-  idr_packet.type.Video.is_first_packet_in_frame = true;
>+  idr_packet.video_header().is_first_packet_in_frame = true;
>   idr_packet.header.sequenceNumber = 2;
>   idr_packet.header.markerBit = 1;
>-  idr_packet.type.Video.is_first_packet_in_frame = true;
>+  idr_packet.video_header().is_first_packet_in_frame = true;
>   idr_packet.frameType = kVideoFrameKey;
>-  idr_packet.type.Video.codec = kRtpVideoH264;
>+  idr_packet.video_header().codec = kVideoCodecH264;
>   data.insert(data.end(), {1, 2, 3});
>   mock_on_complete_frame_callback_.AppendExpectedBitstream(
>       kH264StartCode, sizeof(kH264StartCode));
>@@ -337,11 +376,11 @@ TEST_F(RtpVideoStreamReceiverTest, PaddingInMediaStream) {
>   std::vector<uint8_t> data;
>   data.insert(data.end(), {1, 2, 3});
>   header.header.payloadType = 99;
>-  header.type.Video.is_first_packet_in_frame = true;
>+  header.video_header().is_first_packet_in_frame = true;
>   header.header.sequenceNumber = 2;
>   header.header.markerBit = true;
>   header.frameType = kVideoFrameKey;
>-  header.type.Video.codec = kRtpVideoGeneric;
>+  header.video_header().codec = kVideoCodecGeneric;
>   mock_on_complete_frame_callback_.AppendExpectedBitstream(data.data(),
>                                                            data.size());
> 
>@@ -368,14 +407,13 @@ TEST_F(RtpVideoStreamReceiverTest, PaddingInMediaStream) {
> }
> 
> TEST_F(RtpVideoStreamReceiverTest, RequestKeyframeIfFirstFrameIsDelta) {
>-  WebRtcRTPHeader rtp_header;
>+  WebRtcRTPHeader rtp_header = {};
>   const std::vector<uint8_t> data({1, 2, 3, 4});
>-  memset(&rtp_header, 0, sizeof(rtp_header));
>   rtp_header.header.sequenceNumber = 1;
>   rtp_header.header.markerBit = 1;
>-  rtp_header.type.Video.is_first_packet_in_frame = true;
>+  rtp_header.video_header().is_first_packet_in_frame = true;
>   rtp_header.frameType = kVideoFrameDelta;
>-  rtp_header.type.Video.codec = kRtpVideoGeneric;
>+  rtp_header.video_header().codec = kVideoCodecGeneric;
> 
>   EXPECT_CALL(mock_key_frame_request_sender_, RequestKeyFrame());
>   rtp_video_stream_receiver_->OnReceivedPayloadData(data.data(), data.size(),
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/screenshare_loopback.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/screenshare_loopback.cc
>index 8a0b247752a..ddc190d7eab 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/screenshare_loopback.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/screenshare_loopback.cc
>@@ -11,7 +11,9 @@
> #include <stdio.h>
> 
> #include "rtc_base/flags.h"
>+#include "rtc_base/logging.h"
> #include "rtc_base/stringencode.h"
>+#include "system_wrappers/include/field_trial_default.h"
> #include "test/field_trial.h"
> #include "test/gtest.h"
> #include "test/run_test.h"
>@@ -41,9 +43,7 @@ int MinBitrateKbps() {
>   return static_cast<int>(FLAG_min_bitrate);
> }
> 
>-DEFINE_int(start_bitrate,
>-           Call::Config::kDefaultStartBitrateBps / 1000,
>-           "Call start bitrate in kbps.");
>+DEFINE_int(start_bitrate, 300, "Call start bitrate in kbps.");
> int StartBitrateKbps() {
>   return static_cast<int>(FLAG_start_bitrate);
> }
>@@ -53,7 +53,7 @@ int TargetBitrateKbps() {
>   return static_cast<int>(FLAG_target_bitrate);
> }
> 
>-DEFINE_int(max_bitrate, 2000, "Call and stream max bitrate in kbps.");
>+DEFINE_int(max_bitrate, 1000, "Call and stream max bitrate in kbps.");
> int MaxBitrateKbps() {
>   return static_cast<int>(FLAG_max_bitrate);
> }
>@@ -158,6 +158,21 @@ int NumSpatialLayers() {
>   return static_cast<int>(FLAG_num_spatial_layers);
> }
> 
>+DEFINE_int(inter_layer_pred,
>+           0,
>+           "Inter-layer prediction mode. "
>+           "0 - enabled, 1 - disabled, 2 - enabled only for key pictures.");
>+InterLayerPredMode InterLayerPred() {
>+  if (FLAG_inter_layer_pred == 0) {
>+    return InterLayerPredMode::kOn;
>+  } else if (FLAG_inter_layer_pred == 1) {
>+    return InterLayerPredMode::kOff;
>+  } else {
>+    RTC_DCHECK_EQ(FLAG_inter_layer_pred, 2);
>+    return InterLayerPredMode::kOnKeyPic;
>+  }
>+}
>+
> DEFINE_int(selected_sl,
>            -1,
>            "Spatial layer to show or analyze. -1 to disable filtering.");
>@@ -221,9 +236,10 @@ int MinTransmitBitrateKbps() {
>   return FLAG_min_transmit_bitrate;
> }
> 
>-DEFINE_bool(generate_slides,
>-           false,
>-           "Whether to use randomly generated slides or read them from files.");
>+DEFINE_bool(
>+    generate_slides,
>+    false,
>+    "Whether to use randomly generated slides or read them from files.");
> bool GenerateSlides() {
>   return static_cast<int>(FLAG_generate_slides);
> }
>@@ -266,10 +282,10 @@ void Loopback() {
>   pipe_config.delay_standard_deviation_ms = flags::StdPropagationDelayMs();
>   pipe_config.allow_reordering = flags::FLAG_allow_reordering;
> 
>-  Call::Config::BitrateConfig call_bitrate_config;
>+  BitrateConstraints call_bitrate_config;
>   call_bitrate_config.min_bitrate_bps = flags::MinBitrateKbps() * 1000;
>   call_bitrate_config.start_bitrate_bps = flags::StartBitrateKbps() * 1000;
>-  call_bitrate_config.max_bitrate_bps = flags::MaxBitrateKbps() * 1000;
>+  call_bitrate_config.max_bitrate_bps = -1;  // Don't cap bandwidth estimate.
> 
>   VideoQualityTest::Params params;
>   params.call = {flags::FLAG_send_side_bwe, call_bitrate_config};
>@@ -287,15 +303,20 @@ void Loopback() {
>                      flags::MinTransmitBitrateKbps() * 1000,
>                      false,  // ULPFEC disabled.
>                      false,  // FlexFEC disabled.
>+                     false,  // Automatic scaling disabled.
>                      ""};
>   params.screenshare[0] = {true, flags::GenerateSlides(),
>                            flags::SlideChangeInterval(),
>                            flags::ScrollDuration(), flags::Slides()};
>-  params.analyzer = {"screenshare", 0.0, 0.0, flags::DurationSecs(),
>-      flags::OutputFilename(), flags::GraphTitle()};
>+  params.analyzer = {"screenshare",
>+                     0.0,
>+                     0.0,
>+                     flags::DurationSecs(),
>+                     flags::OutputFilename(),
>+                     flags::GraphTitle()};
>   params.pipe = pipe_config;
>-  params.logging = {flags::FLAG_logs, flags::RtcEventLogName(),
>-                    flags::RtpDumpName(), flags::EncodedFramePath()};
>+  params.logging = {flags::RtcEventLogName(), flags::RtpDumpName(),
>+                    flags::EncodedFramePath()};
> 
>   if (flags::NumStreams() > 1 && flags::Stream0().empty() &&
>       flags::Stream1().empty()) {
>@@ -311,13 +332,13 @@ void Loopback() {
>   VideoQualityTest::FillScalabilitySettings(
>       &params, 0, stream_descriptors, flags::NumStreams(),
>       flags::SelectedStream(), flags::NumSpatialLayers(), flags::SelectedSL(),
>-      SL_descriptors);
>+      flags::InterLayerPred(), SL_descriptors);
> 
>-  VideoQualityTest test;
>+  auto fixture = absl::make_unique<VideoQualityTest>(nullptr);
>   if (flags::DurationSecs()) {
>-    test.RunWithAnalyzer(params);
>+    fixture->RunWithAnalyzer(params);
>   } else {
>-    test.RunWithRenderers(params);
>+    fixture->RunWithRenderers(params);
>   }
> }
> }  // namespace webrtc
>@@ -330,10 +351,14 @@ int main(int argc, char* argv[]) {
>     return 0;
>   }
> 
>-  // InitFieldTrialsFromString needs a reference to an std::string instance,
>-  // with a scope that outlives the test.
>-  std::string field_trials = webrtc::flags::FLAG_force_fieldtrials;
>-  webrtc::test::InitFieldTrialsFromString(field_trials);
>+  rtc::LogMessage::SetLogToStderr(webrtc::flags::FLAG_logs);
>+
>+  webrtc::test::ValidateFieldTrialsStringOrDie(
>+      webrtc::flags::FLAG_force_fieldtrials);
>+  // InitFieldTrialsFromString stores the char*, so the char array must outlive
>+  // the application.
>+  webrtc::field_trial::InitFieldTrialsFromString(
>+      webrtc::flags::FLAG_force_fieldtrials);
> 
>   webrtc::test::RunTest(webrtc::Loopback);
>   return 0;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/send_delay_stats.h b/Source/ThirdParty/libwebrtc/Source/webrtc/video/send_delay_stats.h
>index 71bca2d37d6..81442bcbd1c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/send_delay_stats.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/send_delay_stats.h
>@@ -15,20 +15,20 @@
> #include <memory>
> #include <set>
> 
>+#include "call/video_send_stream.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/include/module_common_types.h"
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/thread_annotations.h"
> #include "system_wrappers/include/clock.h"
> #include "video/stats_counter.h"
>-#include "call/video_send_stream.h"
> 
> namespace webrtc {
> 
> class SendDelayStats : public SendPacketObserver {
>  public:
>   explicit SendDelayStats(Clock* clock);
>-  virtual ~SendDelayStats();
>+  ~SendDelayStats() override;
> 
>   // Adds the configured ssrcs for the rtp streams.
>   // Stats will be calculated for these streams.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/send_delay_stats_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/send_delay_stats_unittest.cc
>index 174e70df105..65ae961a8d1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/send_delay_stats_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/send_delay_stats_unittest.cc
>@@ -107,7 +107,8 @@ TEST_F(SendDelayStatsTest, HistogramsAreUpdated) {
>   const int64_t kDelayMs1 = 5;
>   const int64_t kDelayMs2 = 15;
>   const int kNumSamples = kMinRequiredPeriodicSamples * kProcessIntervalMs /
>-                          (kDelayMs1 + kDelayMs2) + 1;
>+                              (kDelayMs1 + kDelayMs2) +
>+                          1;
> 
>   uint16_t id = 0;
>   for (int i = 0; i < kNumSamples; ++i) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/send_statistics_proxy.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/send_statistics_proxy.cc
>index 4375e840c38..2fb955c3dc9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/send_statistics_proxy.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/send_statistics_proxy.cc
>@@ -12,12 +12,15 @@
> 
> #include <algorithm>
> #include <cmath>
>+#include <limits>
> #include <utility>
> 
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/video_coding/include/video_codec_interface.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
>+#include "rtc_base/numerics/mod_ops.h"
>+#include "rtc_base/strings/string_builder.h"
> #include "system_wrappers/include/field_trial.h"
> #include "system_wrappers/include/metrics.h"
> 
>@@ -26,6 +29,7 @@ namespace {
> const float kEncodeTimeWeigthFactor = 0.5f;
> const size_t kMaxEncodedFrameMapSize = 150;
> const int64_t kMaxEncodedFrameWindowMs = 800;
>+const uint32_t kMaxEncodedFrameTimestampDiff = 900000;  // 10 sec.
> const int64_t kBucketSizeMs = 100;
> const size_t kBucketCount = 10;
> 
>@@ -84,40 +88,39 @@ bool IsForcedFallbackPossible(const CodecSpecificInfo* codec_info) {
>           codec_info->codecSpecific.VP8.temporalIdx == kNoTemporalIdx);
> }
> 
>-rtc::Optional<int> GetFallbackMaxPixels(const std::string& group) {
>+absl::optional<int> GetFallbackMaxPixels(const std::string& group) {
>   if (group.empty())
>-    return rtc::nullopt;
>+    return absl::nullopt;
> 
>   int min_pixels;
>   int max_pixels;
>   int min_bps;
>   if (sscanf(group.c_str(), "-%d,%d,%d", &min_pixels, &max_pixels, &min_bps) !=
>       3) {
>-    return rtc::Optional<int>();
>+    return absl::optional<int>();
>   }
> 
>   if (min_pixels <= 0 || max_pixels <= 0 || max_pixels < min_pixels)
>-    return rtc::Optional<int>();
>+    return absl::optional<int>();
> 
>-  return rtc::Optional<int>(max_pixels);
>+  return absl::optional<int>(max_pixels);
> }
> 
>-rtc::Optional<int> GetFallbackMaxPixelsIfFieldTrialEnabled() {
>+absl::optional<int> GetFallbackMaxPixelsIfFieldTrialEnabled() {
>   std::string group =
>       webrtc::field_trial::FindFullName(kVp8ForcedFallbackEncoderFieldTrial);
>   return (group.find("Enabled") == 0) ? GetFallbackMaxPixels(group.substr(7))
>-                                      : rtc::Optional<int>();
>+                                      : absl::optional<int>();
> }
> 
>-rtc::Optional<int> GetFallbackMaxPixelsIfFieldTrialDisabled() {
>+absl::optional<int> GetFallbackMaxPixelsIfFieldTrialDisabled() {
>   std::string group =
>       webrtc::field_trial::FindFullName(kVp8ForcedFallbackEncoderFieldTrial);
>   return (group.find("Disabled") == 0) ? GetFallbackMaxPixels(group.substr(8))
>-                                       : rtc::Optional<int>();
>+                                       : absl::optional<int>();
> }
> }  // namespace
> 
>-
> const int SendStatisticsProxy::kStatsTimeoutMs = 5000;
> 
> SendStatisticsProxy::SendStatisticsProxy(
>@@ -125,7 +128,7 @@ SendStatisticsProxy::SendStatisticsProxy(
>     const VideoSendStream::Config& config,
>     VideoEncoderConfig::ContentType content_type)
>     : clock_(clock),
>-      payload_name_(config.encoder_settings.payload_name),
>+      payload_name_(config.rtp.payload_name),
>       rtp_config_(config.rtp),
>       fallback_max_pixels_(GetFallbackMaxPixelsIfFieldTrialEnabled()),
>       fallback_max_pixels_disabled_(GetFallbackMaxPixelsIfFieldTrialDisabled()),
>@@ -173,6 +176,9 @@ SendStatisticsProxy::UmaSamplesContainer::UmaSamplesContainer(
>       num_streams_(0),
>       num_pixels_highest_stream_(0) {
>   InitializeBitrateCounters(stats);
>+  static_assert(
>+      kMaxEncodedFrameTimestampDiff < std::numeric_limits<uint32_t>::max() / 2,
>+      "has to be smaller than half range");
> }
> 
> SendStatisticsProxy::UmaSamplesContainer::~UmaSamplesContainer() {}
>@@ -241,6 +247,17 @@ bool SendStatisticsProxy::UmaSamplesContainer::InsertEncodedFrame(
>     encoded_frames_.clear();
>   }
> 
>+  // Check for jump in timestamp.
>+  if (!encoded_frames_.empty()) {
>+    uint32_t oldest_timestamp = encoded_frames_.begin()->first;
>+    if (ForwardDiff(oldest_timestamp, encoded_frame._timeStamp) >
>+        kMaxEncodedFrameTimestampDiff) {
>+      // Gap detected, clear frames to have a sequence where newest timestamp
>+      // is not too far away from oldest in order to distinguish old and new.
>+      encoded_frames_.clear();
>+    }
>+  }
>+
>   auto it = encoded_frames_.find(encoded_frame._timeStamp);
>   if (it == encoded_frames_.end()) {
>     // First frame with this timestamp.
>@@ -262,11 +279,13 @@ bool SendStatisticsProxy::UmaSamplesContainer::InsertEncodedFrame(
> }
> 
> void SendStatisticsProxy::UmaSamplesContainer::UpdateHistograms(
>-    const VideoSendStream::Config::Rtp& rtp_config,
>+    const RtpConfig& rtp_config,
>     const VideoSendStream::Stats& current_stats) {
>   RTC_DCHECK(uma_prefix_ == kRealtimePrefix || uma_prefix_ == kScreenPrefix);
>   const int kIndex = uma_prefix_ == kScreenPrefix ? 1 : 0;
>   const int kMinRequiredPeriodicSamples = 6;
>+  char log_stream_buf[8 * 1024];
>+  rtc::SimpleStringBuilder log_stream(log_stream_buf);
>   int in_width = input_width_counter_.Avg(kMinRequiredMetricsSamples);
>   int in_height = input_height_counter_.Avg(kMinRequiredMetricsSamples);
>   if (in_width != -1) {
>@@ -274,15 +293,15 @@ void SendStatisticsProxy::UmaSamplesContainer::UpdateHistograms(
>                                 in_width);
>     RTC_HISTOGRAMS_COUNTS_10000(kIndex, uma_prefix_ + "InputHeightInPixels",
>                                 in_height);
>-    RTC_LOG(LS_INFO) << uma_prefix_ << "InputWidthInPixels " << in_width;
>-    RTC_LOG(LS_INFO) << uma_prefix_ << "InputHeightInPixels " << in_height;
>+    log_stream << uma_prefix_ << "InputWidthInPixels " << in_width << "\n"
>+               << uma_prefix_ << "InputHeightInPixels " << in_height << "\n";
>   }
>   AggregatedStats in_fps = input_fps_counter_.GetStats();
>   if (in_fps.num_samples >= kMinRequiredPeriodicSamples) {
>     RTC_HISTOGRAMS_COUNTS_100(kIndex, uma_prefix_ + "InputFramesPerSecond",
>                               in_fps.average);
>-    RTC_LOG(LS_INFO) << uma_prefix_ + "InputFramesPerSecond, "
>-                     << in_fps.ToString();
>+    log_stream << uma_prefix_ << "InputFramesPerSecond " << in_fps.ToString()
>+               << "\n";
>   }
> 
>   int sent_width = sent_width_counter_.Avg(kMinRequiredMetricsSamples);
>@@ -292,15 +311,15 @@ void SendStatisticsProxy::UmaSamplesContainer::UpdateHistograms(
>                                 sent_width);
>     RTC_HISTOGRAMS_COUNTS_10000(kIndex, uma_prefix_ + "SentHeightInPixels",
>                                 sent_height);
>-    RTC_LOG(LS_INFO) << uma_prefix_ << "SentWidthInPixels " << sent_width;
>-    RTC_LOG(LS_INFO) << uma_prefix_ << "SentHeightInPixels " << sent_height;
>+    log_stream << uma_prefix_ << "SentWidthInPixels " << sent_width << "\n"
>+               << uma_prefix_ << "SentHeightInPixels " << sent_height << "\n";
>   }
>   AggregatedStats sent_fps = sent_fps_counter_.GetStats();
>   if (sent_fps.num_samples >= kMinRequiredPeriodicSamples) {
>     RTC_HISTOGRAMS_COUNTS_100(kIndex, uma_prefix_ + "SentFramesPerSecond",
>                               sent_fps.average);
>-    RTC_LOG(LS_INFO) << uma_prefix_ + "SentFramesPerSecond, "
>-                     << sent_fps.ToString();
>+    log_stream << uma_prefix_ << "SentFramesPerSecond " << sent_fps.ToString()
>+               << "\n";
>   }
> 
>   if (in_fps.num_samples > kMinRequiredPeriodicSamples &&
>@@ -318,8 +337,8 @@ void SendStatisticsProxy::UmaSamplesContainer::UpdateHistograms(
>       RTC_HISTOGRAMS_PERCENTAGE(kIndex,
>                                 uma_prefix_ + "SentToInputFpsRatioPercent",
>                                 sent_to_in_fps_ratio_percent);
>-      RTC_LOG(LS_INFO) << uma_prefix_ << "SentToInputFpsRatioPercent "
>-                       << sent_to_in_fps_ratio_percent;
>+      log_stream << uma_prefix_ << "SentToInputFpsRatioPercent "
>+                 << sent_to_in_fps_ratio_percent << "\n";
>     }
>   }
> 
>@@ -327,15 +346,15 @@ void SendStatisticsProxy::UmaSamplesContainer::UpdateHistograms(
>   if (encode_ms != -1) {
>     RTC_HISTOGRAMS_COUNTS_1000(kIndex, uma_prefix_ + "EncodeTimeInMs",
>                                encode_ms);
>-    RTC_LOG(LS_INFO) << uma_prefix_ << "EncodeTimeInMs " << encode_ms;
>+    log_stream << uma_prefix_ << "EncodeTimeInMs " << encode_ms << "\n";
>   }
>   int key_frames_permille =
>       key_frame_counter_.Permille(kMinRequiredMetricsSamples);
>   if (key_frames_permille != -1) {
>     RTC_HISTOGRAMS_COUNTS_1000(kIndex, uma_prefix_ + "KeyFramesSentInPermille",
>                                key_frames_permille);
>-    RTC_LOG(LS_INFO) << uma_prefix_ << "KeyFramesSentInPermille "
>-                     << key_frames_permille;
>+    log_stream << uma_prefix_ << "KeyFramesSentInPermille "
>+               << key_frames_permille << "\n";
>   }
>   int quality_limited =
>       quality_limited_frame_counter_.Percent(kMinRequiredMetricsSamples);
>@@ -343,8 +362,8 @@ void SendStatisticsProxy::UmaSamplesContainer::UpdateHistograms(
>     RTC_HISTOGRAMS_PERCENTAGE(kIndex,
>                               uma_prefix_ + "QualityLimitedResolutionInPercent",
>                               quality_limited);
>-    RTC_LOG(LS_INFO) << uma_prefix_ << "QualityLimitedResolutionInPercent "
>-                     << quality_limited;
>+    log_stream << uma_prefix_ << "QualityLimitedResolutionInPercent "
>+               << quality_limited << "\n";
>   }
>   int downscales = quality_downscales_counter_.Avg(kMinRequiredMetricsSamples);
>   if (downscales != -1) {
>@@ -427,9 +446,22 @@ void SendStatisticsProxy::UmaSamplesContainer::UpdateHistograms(
>     int qp_h264 = it.second.h264.Avg(kMinRequiredMetricsSamples);
>     if (qp_h264 != -1) {
>       int spatial_idx = it.first;
>-      RTC_DCHECK_EQ(-1, spatial_idx);
>-      RTC_HISTOGRAMS_COUNTS_100(kIndex, uma_prefix_ + "Encoded.Qp.H264",
>-                                qp_h264);
>+      if (spatial_idx == -1) {
>+        RTC_HISTOGRAMS_COUNTS_200(kIndex, uma_prefix_ + "Encoded.Qp.H264",
>+                                  qp_h264);
>+      } else if (spatial_idx == 0) {
>+        RTC_HISTOGRAMS_COUNTS_200(kIndex, uma_prefix_ + "Encoded.Qp.H264.S0",
>+                                  qp_h264);
>+      } else if (spatial_idx == 1) {
>+        RTC_HISTOGRAMS_COUNTS_200(kIndex, uma_prefix_ + "Encoded.Qp.H264.S1",
>+                                  qp_h264);
>+      } else if (spatial_idx == 2) {
>+        RTC_HISTOGRAMS_COUNTS_200(kIndex, uma_prefix_ + "Encoded.Qp.H264.S2",
>+                                  qp_h264);
>+      } else {
>+        RTC_LOG(LS_WARNING)
>+            << "QP stats not recorded for H264 spatial idx " << spatial_idx;
>+      }
>     }
>   }
> 
>@@ -467,8 +499,8 @@ void SendStatisticsProxy::UmaSamplesContainer::UpdateHistograms(
>       if (fraction_lost != -1) {
>         RTC_HISTOGRAMS_PERCENTAGE(
>             kIndex, uma_prefix_ + "SentPacketsLostInPercent", fraction_lost);
>-        RTC_LOG(LS_INFO) << uma_prefix_ << "SentPacketsLostInPercent "
>-                         << fraction_lost;
>+        log_stream << uma_prefix_ << "SentPacketsLostInPercent "
>+                   << fraction_lost;
>       }
> 
>       // The RTCP packet type counters, delivered via the
>@@ -515,16 +547,16 @@ void SendStatisticsProxy::UmaSamplesContainer::UpdateHistograms(
>     if (elapsed_sec >= metrics::kMinRunTimeInSeconds) {
>       RTC_HISTOGRAMS_COUNTS_100(kIndex, uma_prefix_ + "NumberOfPauseEvents",
>                                 target_rate_updates_.pause_resume_events);
>-      RTC_LOG(LS_INFO) << uma_prefix_ << "NumberOfPauseEvents "
>-                       << target_rate_updates_.pause_resume_events;
>+      log_stream << uma_prefix_ << "NumberOfPauseEvents "
>+                 << target_rate_updates_.pause_resume_events << "\n";
> 
>       int paused_time_percent =
>           paused_time_counter_.Percent(metrics::kMinRunTimeInSeconds * 1000);
>       if (paused_time_percent != -1) {
>         RTC_HISTOGRAMS_PERCENTAGE(kIndex, uma_prefix_ + "PausedTimeInPercent",
>                                   paused_time_percent);
>-        RTC_LOG(LS_INFO) << uma_prefix_ << "PausedTimeInPercent "
>-                         << paused_time_percent;
>+        log_stream << uma_prefix_ << "PausedTimeInPercent "
>+                   << paused_time_percent << "\n";
>       }
>     }
>   }
>@@ -548,23 +580,23 @@ void SendStatisticsProxy::UmaSamplesContainer::UpdateHistograms(
>   if (total_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) {
>     RTC_HISTOGRAMS_COUNTS_10000(kIndex, uma_prefix_ + "BitrateSentInKbps",
>                                 total_bytes_per_sec.average * 8 / 1000);
>-    RTC_LOG(LS_INFO) << uma_prefix_ << "BitrateSentInBps, "
>-                     << total_bytes_per_sec.ToStringWithMultiplier(8);
>+    log_stream << uma_prefix_ << "BitrateSentInBps "
>+               << total_bytes_per_sec.ToStringWithMultiplier(8) << "\n";
>   }
>   AggregatedStats media_bytes_per_sec = media_byte_counter_.GetStats();
>   if (media_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) {
>     RTC_HISTOGRAMS_COUNTS_10000(kIndex, uma_prefix_ + "MediaBitrateSentInKbps",
>                                 media_bytes_per_sec.average * 8 / 1000);
>-    RTC_LOG(LS_INFO) << uma_prefix_ << "MediaBitrateSentInBps, "
>-                     << media_bytes_per_sec.ToStringWithMultiplier(8);
>+    log_stream << uma_prefix_ << "MediaBitrateSentInBps "
>+               << media_bytes_per_sec.ToStringWithMultiplier(8) << "\n";
>   }
>   AggregatedStats padding_bytes_per_sec = padding_byte_counter_.GetStats();
>   if (padding_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) {
>     RTC_HISTOGRAMS_COUNTS_10000(kIndex,
>                                 uma_prefix_ + "PaddingBitrateSentInKbps",
>                                 padding_bytes_per_sec.average * 8 / 1000);
>-    RTC_LOG(LS_INFO) << uma_prefix_ << "PaddingBitrateSentInBps, "
>-                     << padding_bytes_per_sec.ToStringWithMultiplier(8);
>+    log_stream << uma_prefix_ << "PaddingBitrateSentInBps "
>+               << padding_bytes_per_sec.ToStringWithMultiplier(8) << "\n";
>   }
>   AggregatedStats retransmit_bytes_per_sec =
>       retransmit_byte_counter_.GetStats();
>@@ -572,16 +604,16 @@ void SendStatisticsProxy::UmaSamplesContainer::UpdateHistograms(
>     RTC_HISTOGRAMS_COUNTS_10000(kIndex,
>                                 uma_prefix_ + "RetransmittedBitrateSentInKbps",
>                                 retransmit_bytes_per_sec.average * 8 / 1000);
>-    RTC_LOG(LS_INFO) << uma_prefix_ << "RetransmittedBitrateSentInBps, "
>-                     << retransmit_bytes_per_sec.ToStringWithMultiplier(8);
>+    log_stream << uma_prefix_ << "RetransmittedBitrateSentInBps "
>+               << retransmit_bytes_per_sec.ToStringWithMultiplier(8) << "\n";
>   }
>   if (!rtp_config.rtx.ssrcs.empty()) {
>     AggregatedStats rtx_bytes_per_sec = rtx_byte_counter_.GetStats();
>     int rtx_bytes_per_sec_avg = -1;
>     if (rtx_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) {
>       rtx_bytes_per_sec_avg = rtx_bytes_per_sec.average;
>-      RTC_LOG(LS_INFO) << uma_prefix_ << "RtxBitrateSentInBps, "
>-                       << rtx_bytes_per_sec.ToStringWithMultiplier(8);
>+      log_stream << uma_prefix_ << "RtxBitrateSentInBps "
>+                 << rtx_bytes_per_sec.ToStringWithMultiplier(8) << "\n";
>     } else if (total_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) {
>       rtx_bytes_per_sec_avg = 0;  // RTX enabled but no RTX data sent, record 0.
>     }
>@@ -596,35 +628,35 @@ void SendStatisticsProxy::UmaSamplesContainer::UpdateHistograms(
>     if (fec_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) {
>       RTC_HISTOGRAMS_COUNTS_10000(kIndex, uma_prefix_ + "FecBitrateSentInKbps",
>                                   fec_bytes_per_sec.average * 8 / 1000);
>-      RTC_LOG(LS_INFO) << uma_prefix_ << "FecBitrateSentInBps, "
>-                       << fec_bytes_per_sec.ToStringWithMultiplier(8);
>+      log_stream << uma_prefix_ << "FecBitrateSentInBps "
>+                 << fec_bytes_per_sec.ToStringWithMultiplier(8) << "\n";
>     }
>   }
>-  RTC_LOG(LS_INFO) << "Frames encoded " << current_stats.frames_encoded;
>-  RTC_LOG(LS_INFO) << uma_prefix_ << "DroppedFrames.Capturer "
>-                   << current_stats.frames_dropped_by_capturer;
>+  log_stream << "Frames encoded " << current_stats.frames_encoded << "\n"
>+             << uma_prefix_ << "DroppedFrames.Capturer "
>+             << current_stats.frames_dropped_by_capturer << "\n";
>   RTC_HISTOGRAMS_COUNTS_1000(kIndex, uma_prefix_ + "DroppedFrames.Capturer",
>                              current_stats.frames_dropped_by_capturer);
>-  RTC_LOG(LS_INFO) << uma_prefix_ << "DroppedFrames.EncoderQueue "
>-                   << current_stats.frames_dropped_by_encoder_queue;
>+  log_stream << uma_prefix_ << "DroppedFrames.EncoderQueue "
>+             << current_stats.frames_dropped_by_encoder_queue << "\n";
>   RTC_HISTOGRAMS_COUNTS_1000(kIndex, uma_prefix_ + "DroppedFrames.EncoderQueue",
>                              current_stats.frames_dropped_by_encoder_queue);
>-  RTC_LOG(LS_INFO) << uma_prefix_ << "DroppedFrames.Encoder "
>-                   << current_stats.frames_dropped_by_encoder;
>+  log_stream << uma_prefix_ << "DroppedFrames.Encoder "
>+             << current_stats.frames_dropped_by_encoder << "\n";
>   RTC_HISTOGRAMS_COUNTS_1000(kIndex, uma_prefix_ + "DroppedFrames.Encoder",
>                              current_stats.frames_dropped_by_encoder);
>-  RTC_LOG(LS_INFO) << uma_prefix_ << "DroppedFrames.Ratelimiter "
>-                   << current_stats.frames_dropped_by_rate_limiter;
>+  log_stream << uma_prefix_ << "DroppedFrames.Ratelimiter "
>+             << current_stats.frames_dropped_by_rate_limiter;
>   RTC_HISTOGRAMS_COUNTS_1000(kIndex, uma_prefix_ + "DroppedFrames.Ratelimiter",
>                              current_stats.frames_dropped_by_rate_limiter);
>+
>+  RTC_LOG(LS_INFO) << log_stream.str();
> }
> 
> void SendStatisticsProxy::OnEncoderReconfigured(
>     const VideoEncoderConfig& config,
>-    const std::vector<VideoStream>& streams,
>-    uint32_t preferred_bitrate_bps) {
>+    const std::vector<VideoStream>& streams) {
>   rtc::CritScope lock(&crit_);
>-  stats_.preferred_media_bitrate_bps = preferred_bitrate_bps;
> 
>   if (content_type_ != config.content_type) {
>     uma_container_->UpdateHistograms(rtp_config_, stats_);
>@@ -638,14 +670,13 @@ void SendStatisticsProxy::OnEncoderReconfigured(
>       streams.empty() ? 0 : (streams.back().width * streams.back().height);
> }
> 
>-void SendStatisticsProxy::OnEncodedFrameTimeMeasured(
>-    int encode_time_ms,
>-    const CpuOveruseMetrics& metrics) {
>+void SendStatisticsProxy::OnEncodedFrameTimeMeasured(int encode_time_ms,
>+                                                     int encode_usage_percent) {
>   rtc::CritScope lock(&crit_);
>   uma_container_->encode_time_counter_.Add(encode_time_ms);
>   encode_time_.Apply(1.0f, encode_time_ms);
>   stats_.avg_encode_time_ms = round(encode_time_.filtered());
>-  stats_.encode_usage_percent = metrics.encode_usage_percent;
>+  stats_.encode_usage_percent = encode_usage_percent;
> }
> 
> void SendStatisticsProxy::OnSuspendChange(bool is_suspended) {
>@@ -856,6 +887,8 @@ void SendStatisticsProxy::OnSendEncodedImage(
>   if (codec_info) {
>     if (codec_info->codecType == kVideoCodecVP8) {
>       simulcast_idx = codec_info->codecSpecific.VP8.simulcastIdx;
>+    } else if (codec_info->codecType == kVideoCodecH264) {
>+      simulcast_idx = codec_info->codecSpecific.H264.simulcast_idx;
>     } else if (codec_info->codecType == kVideoCodecGeneric) {
>       simulcast_idx = codec_info->codecSpecific.generic.simulcast_idx;
>     }
>@@ -878,6 +911,7 @@ void SendStatisticsProxy::OnSendEncodedImage(
>   if (!stats)
>     return;
> 
>+  // TODO(ssilkin): Fix stats reporting for spatial layers in SVC.
>   stats->width = encoded_image._encodedWidth;
>   stats->height = encoded_image._encodedHeight;
>   update_times_[ssrc].resolution_update_ms = clock_->TimeInMilliseconds();
>@@ -903,12 +937,25 @@ void SendStatisticsProxy::OnSendEncodedImage(
>                 : codec_info->codecSpecific.VP9.spatial_idx;
>         uma_container_->qp_counters_[spatial_idx].vp9.Add(encoded_image.qp_);
>       } else if (codec_info->codecType == kVideoCodecH264) {
>-        int spatial_idx = -1;
>+        int spatial_idx = (rtp_config_.ssrcs.size() == 1)
>+                              ? -1
>+                              : static_cast<int>(simulcast_idx);
>         uma_container_->qp_counters_[spatial_idx].h264.Add(encoded_image.qp_);
>       }
>     }
>   }
> 
>+  // If any of the simulcast streams have a huge frame, it should be counted
>+  // as a single difficult input frame.
>+  // https://w3c.github.io/webrtc-stats/#dom-rtcvideosenderstats-hugeframessent
>+  if (encoded_image.timing_.flags & VideoSendTiming::kTriggeredBySize) {
>+    if (!last_outlier_timestamp_ ||
>+        *last_outlier_timestamp_ < encoded_image.capture_time_ms_) {
>+      last_outlier_timestamp_.emplace(encoded_image.capture_time_ms_);
>+      ++stats_.huge_frames_sent;
>+    }
>+  }
>+
>   media_byte_rate_tracker_.AddSamples(encoded_image._length);
> 
>   // Initialize to current since |is_limited_in_resolution| is only updated
>@@ -929,6 +976,11 @@ void SendStatisticsProxy::OnSendEncodedImage(
>   }
> }
> 
>+int SendStatisticsProxy::GetInputFrameRate() const {
>+  rtc::CritScope lock(&crit_);
>+  return round(uma_container_->input_frame_rate_tracker_.ComputeRate());
>+}
>+
> int SendStatisticsProxy::GetSendFrameRate() const {
>   rtc::CritScope lock(&crit_);
>   return round(encoded_frame_rate_tracker_.ComputeRate());
>@@ -951,62 +1003,55 @@ void SendStatisticsProxy::OnIncomingFrame(int width, int height) {
>   }
> }
> 
>-void SendStatisticsProxy::OnFrameDroppedBySource() {
>-  rtc::CritScope lock(&crit_);
>-  ++stats_.frames_dropped_by_capturer;
>-}
>-
>-void SendStatisticsProxy::OnFrameDroppedInEncoderQueue() {
>+void SendStatisticsProxy::OnFrameDropped(DropReason reason) {
>   rtc::CritScope lock(&crit_);
>-  ++stats_.frames_dropped_by_encoder_queue;
>-}
>-
>-void SendStatisticsProxy::OnFrameDroppedByEncoder() {
>-  rtc::CritScope lock(&crit_);
>-  ++stats_.frames_dropped_by_encoder;
>-}
>-
>-void SendStatisticsProxy::OnFrameDroppedByMediaOptimizations() {
>-  rtc::CritScope lock(&crit_);
>-  ++stats_.frames_dropped_by_rate_limiter;
>-}
>-
>-void SendStatisticsProxy::SetAdaptationStats(
>-    const VideoStreamEncoder::AdaptCounts& cpu_counts,
>-    const VideoStreamEncoder::AdaptCounts& quality_counts) {
>-  rtc::CritScope lock(&crit_);
>-  SetAdaptTimer(cpu_counts, &uma_container_->cpu_adapt_timer_);
>-  SetAdaptTimer(quality_counts, &uma_container_->quality_adapt_timer_);
>-  UpdateAdaptationStats(cpu_counts, quality_counts);
>-}
>-
>-void SendStatisticsProxy::OnCpuAdaptationChanged(
>-    const VideoStreamEncoder::AdaptCounts& cpu_counts,
>-    const VideoStreamEncoder::AdaptCounts& quality_counts) {
>-  rtc::CritScope lock(&crit_);
>-  ++stats_.number_of_cpu_adapt_changes;
>-  UpdateAdaptationStats(cpu_counts, quality_counts);
>+  switch (reason) {
>+    case DropReason::kSource:
>+      ++stats_.frames_dropped_by_capturer;
>+      break;
>+    case DropReason::kEncoderQueue:
>+      ++stats_.frames_dropped_by_encoder_queue;
>+      break;
>+    case DropReason::kEncoder:
>+      ++stats_.frames_dropped_by_encoder;
>+      break;
>+    case DropReason::kMediaOptimization:
>+      ++stats_.frames_dropped_by_rate_limiter;
>+      break;
>+  }
> }
> 
>-void SendStatisticsProxy::OnQualityAdaptationChanged(
>-    const VideoStreamEncoder::AdaptCounts& cpu_counts,
>-    const VideoStreamEncoder::AdaptCounts& quality_counts) {
>+void SendStatisticsProxy::OnAdaptationChanged(
>+    AdaptationReason reason,
>+    const AdaptationSteps& cpu_counts,
>+    const AdaptationSteps& quality_counts) {
>   rtc::CritScope lock(&crit_);
>-  TryUpdateInitialQualityResolutionAdaptUp(quality_counts);
>-  ++stats_.number_of_quality_adapt_changes;
>+  switch (reason) {
>+    case AdaptationReason::kNone:
>+      SetAdaptTimer(cpu_counts, &uma_container_->cpu_adapt_timer_);
>+      SetAdaptTimer(quality_counts, &uma_container_->quality_adapt_timer_);
>+      break;
>+    case AdaptationReason::kCpu:
>+      ++stats_.number_of_cpu_adapt_changes;
>+      break;
>+    case AdaptationReason::kQuality:
>+      TryUpdateInitialQualityResolutionAdaptUp(quality_counts);
>+      ++stats_.number_of_quality_adapt_changes;
>+      break;
>+  }
>   UpdateAdaptationStats(cpu_counts, quality_counts);
> }
> 
> void SendStatisticsProxy::UpdateAdaptationStats(
>-    const VideoStreamEncoder::AdaptCounts& cpu_counts,
>-    const VideoStreamEncoder::AdaptCounts& quality_counts) {
>-  cpu_downscales_ = cpu_counts.resolution;
>-  quality_downscales_ = quality_counts.resolution;
>-
>-  stats_.cpu_limited_resolution = cpu_counts.resolution > 0;
>-  stats_.cpu_limited_framerate = cpu_counts.fps > 0;
>-  stats_.bw_limited_resolution = quality_counts.resolution > 0;
>-  stats_.bw_limited_framerate = quality_counts.fps > 0;
>+    const AdaptationSteps& cpu_counts,
>+    const AdaptationSteps& quality_counts) {
>+  cpu_downscales_ = cpu_counts.num_resolution_reductions.value_or(-1);
>+  quality_downscales_ = quality_counts.num_resolution_reductions.value_or(-1);
>+
>+  stats_.cpu_limited_resolution = cpu_counts.num_resolution_reductions > 0;
>+  stats_.cpu_limited_framerate = cpu_counts.num_framerate_reductions > 0;
>+  stats_.bw_limited_resolution = quality_counts.num_resolution_reductions > 0;
>+  stats_.bw_limited_framerate = quality_counts.num_framerate_reductions > 0;
> }
> 
> // TODO(asapersson): Include fps changes.
>@@ -1016,12 +1061,13 @@ void SendStatisticsProxy::OnInitialQualityResolutionAdaptDown() {
> }
> 
> void SendStatisticsProxy::TryUpdateInitialQualityResolutionAdaptUp(
>-    const VideoStreamEncoder::AdaptCounts& quality_counts) {
>+    const AdaptationSteps& quality_counts) {
>   if (uma_container_->initial_quality_changes_.down == 0)
>     return;
> 
>   if (quality_downscales_ > 0 &&
>-      quality_counts.resolution < quality_downscales_) {
>+      quality_counts.num_resolution_reductions.value_or(-1) <
>+          quality_downscales_) {
>     // Adapting up in quality.
>     if (uma_container_->initial_quality_changes_.down >
>         uma_container_->initial_quality_changes_.up) {
>@@ -1030,10 +1076,9 @@ void SendStatisticsProxy::TryUpdateInitialQualityResolutionAdaptUp(
>   }
> }
> 
>-void SendStatisticsProxy::SetAdaptTimer(
>-    const VideoStreamEncoder::AdaptCounts& counts,
>-    StatsTimer* timer) {
>-  if (counts.resolution >= 0 || counts.fps >= 0) {
>+void SendStatisticsProxy::SetAdaptTimer(const AdaptationSteps& counts,
>+                                        StatsTimer* timer) {
>+  if (counts.num_resolution_reductions || counts.num_framerate_reductions) {
>     // Adaptation enabled.
>     if (!stats_.suspended)
>       timer->Start(clock_->TimeInMilliseconds());
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/send_statistics_proxy.h b/Source/ThirdParty/libwebrtc/Source/webrtc/video/send_statistics_proxy.h
>index 1aa836ebca2..97b4a4afe62 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/send_statistics_proxy.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/send_statistics_proxy.h
>@@ -16,6 +16,8 @@
> #include <string>
> #include <vector>
> 
>+#include "api/video/video_stream_encoder_observer.h"
>+#include "call/video_send_stream.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "modules/video_coding/include/video_codec_interface.h"
> #include "modules/video_coding/include/video_coding_defines.h"
>@@ -24,15 +26,12 @@
> #include "rtc_base/ratetracker.h"
> #include "rtc_base/thread_annotations.h"
> #include "system_wrappers/include/clock.h"
>-#include "video/overuse_frame_detector.h"
> #include "video/report_block_stats.h"
> #include "video/stats_counter.h"
>-#include "video/video_stream_encoder.h"
>-#include "call/video_send_stream.h"
> 
> namespace webrtc {
> 
>-class SendStatisticsProxy : public CpuOveruseMetricsObserver,
>+class SendStatisticsProxy : public VideoStreamEncoderObserver,
>                             public RtcpStatisticsCallback,
>                             public RtcpPacketTypeCounterObserver,
>                             public StreamDataCountersCallback,
>@@ -48,50 +47,42 @@ class SendStatisticsProxy : public CpuOveruseMetricsObserver,
>   SendStatisticsProxy(Clock* clock,
>                       const VideoSendStream::Config& config,
>                       VideoEncoderConfig::ContentType content_type);
>-  virtual ~SendStatisticsProxy();
>+  ~SendStatisticsProxy() override;
> 
>   virtual VideoSendStream::Stats GetStats();
> 
>-  virtual void OnSendEncodedImage(const EncodedImage& encoded_image,
>-                                  const CodecSpecificInfo* codec_info);
>+  void OnSendEncodedImage(const EncodedImage& encoded_image,
>+                          const CodecSpecificInfo* codec_info) override;
>   // Used to update incoming frame rate.
>-  void OnIncomingFrame(int width, int height);
>+  void OnIncomingFrame(int width, int height) override;
> 
>   // Dropped frame stats.
>-  void OnFrameDroppedBySource();
>-  void OnFrameDroppedInEncoderQueue();
>-  void OnFrameDroppedByEncoder();
>-  void OnFrameDroppedByMediaOptimizations();
>+  void OnFrameDropped(DropReason) override;
> 
>   // Adaptation stats.
>-  void SetAdaptationStats(
>-      const VideoStreamEncoder::AdaptCounts& cpu_counts,
>-      const VideoStreamEncoder::AdaptCounts& quality_counts);
>-  void OnCpuAdaptationChanged(
>-      const VideoStreamEncoder::AdaptCounts& cpu_counts,
>-      const VideoStreamEncoder::AdaptCounts& quality_counts);
>-  void OnQualityAdaptationChanged(
>-      const VideoStreamEncoder::AdaptCounts& cpu_counts,
>-      const VideoStreamEncoder::AdaptCounts& quality_counts);
>-  void OnMinPixelLimitReached();
>-  void OnInitialQualityResolutionAdaptDown();
>-
>-  void OnSuspendChange(bool is_suspended);
>+  void OnAdaptationChanged(AdaptationReason reason,
>+                           const AdaptationSteps& cpu_counts,
>+                           const AdaptationSteps& quality_counts) override;
>+
>+  void OnMinPixelLimitReached() override;
>+  void OnInitialQualityResolutionAdaptDown() override;
>+
>+  void OnSuspendChange(bool is_suspended) override;
>   void OnInactiveSsrc(uint32_t ssrc);
> 
>   // Used to indicate change in content type, which may require a change in
>-  // how stats are collected and set the configured preferred media bitrate.
>+  // how stats are collected.
>   void OnEncoderReconfigured(const VideoEncoderConfig& encoder_config,
>-                             const std::vector<VideoStream>& streams,
>-                             uint32_t preferred_bitrate_bps);
>+                             const std::vector<VideoStream>& streams) override;
> 
>   // Used to update the encoder target rate.
>   void OnSetEncoderTargetRate(uint32_t bitrate_bps);
> 
>   // Implements CpuOveruseMetricsObserver.
>   void OnEncodedFrameTimeMeasured(int encode_time_ms,
>-                                  const CpuOveruseMetrics& metrics) override;
>+                                  int encode_usage_percent) override;
> 
>+  int GetInputFrameRate() const override;
>   int GetSendFrameRate() const;
> 
>  protected:
>@@ -159,11 +150,12 @@ class SendStatisticsProxy : public CpuOveruseMetricsObserver,
>     int64_t last_ms;
>   };
>   struct FallbackEncoderInfo {
>+    FallbackEncoderInfo() = default;
>     bool is_possible = true;
>     bool is_active = false;
>     int on_off_events = 0;
>     int64_t elapsed_ms = 0;
>-    rtc::Optional<int64_t> last_update_ms;
>+    absl::optional<int64_t> last_update_ms;
>     const int max_frame_diff_ms = 2000;
>   };
>   struct FallbackEncoderInfoDisabled {
>@@ -216,14 +208,13 @@ class SendStatisticsProxy : public CpuOveruseMetricsObserver,
>   VideoSendStream::StreamStats* GetStatsEntry(uint32_t ssrc)
>       RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
> 
>-  void SetAdaptTimer(const VideoStreamEncoder::AdaptCounts& counts,
>-                     StatsTimer* timer) RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
>-  void UpdateAdaptationStats(
>-      const VideoStreamEncoder::AdaptCounts& cpu_counts,
>-      const VideoStreamEncoder::AdaptCounts& quality_counts)
>+  void SetAdaptTimer(const AdaptationSteps& counts, StatsTimer* timer)
>+      RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
>+  void UpdateAdaptationStats(const AdaptationSteps& cpu_counts,
>+                             const AdaptationSteps& quality_counts)
>       RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
>   void TryUpdateInitialQualityResolutionAdaptUp(
>-      const VideoStreamEncoder::AdaptCounts& quality_counts)
>+      const AdaptationSteps& quality_counts)
>       RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
> 
>   void UpdateEncoderFallbackStats(const CodecSpecificInfo* codec_info,
>@@ -235,9 +226,9 @@ class SendStatisticsProxy : public CpuOveruseMetricsObserver,
> 
>   Clock* const clock_;
>   const std::string payload_name_;
>-  const VideoSendStream::Config::Rtp rtp_config_;
>-  const rtc::Optional<int> fallback_max_pixels_;
>-  const rtc::Optional<int> fallback_max_pixels_disabled_;
>+  const RtpConfig rtp_config_;
>+  const absl::optional<int> fallback_max_pixels_;
>+  const absl::optional<int> fallback_max_pixels_disabled_;
>   rtc::CriticalSection crit_;
>   VideoEncoderConfig::ContentType content_type_ RTC_GUARDED_BY(crit_);
>   const int64_t start_ms_;
>@@ -249,6 +240,8 @@ class SendStatisticsProxy : public CpuOveruseMetricsObserver,
>   rtc::RateTracker media_byte_rate_tracker_ RTC_GUARDED_BY(crit_);
>   rtc::RateTracker encoded_frame_rate_tracker_ RTC_GUARDED_BY(crit_);
> 
>+  absl::optional<int64_t> last_outlier_timestamp_ RTC_GUARDED_BY(crit_);
>+
>   // Contains stats used for UMA histograms. These stats will be reset if
>   // content type changes between real-time video and screenshare, since these
>   // will be reported separately.
>@@ -258,7 +251,7 @@ class SendStatisticsProxy : public CpuOveruseMetricsObserver,
>                         Clock* clock);
>     ~UmaSamplesContainer();
> 
>-    void UpdateHistograms(const VideoSendStream::Config::Rtp& rtp_config,
>+    void UpdateHistograms(const RtpConfig& rtp_config,
>                           const VideoSendStream::Stats& current_stats);
> 
>     void InitializeBitrateCounters(const VideoSendStream::Stats& stats);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/send_statistics_proxy_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/send_statistics_proxy_unittest.cc
>index 01cbfe8dda4..6942e36b5dd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/send_statistics_proxy_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/send_statistics_proxy_unittest.cc
>@@ -28,7 +28,6 @@ const uint32_t kFirstRtxSsrc = 18;
> const uint32_t kSecondRtxSsrc = 43;
> const uint32_t kFlexFecSsrc = 55;
> const int kFpsPeriodicIntervalMs = 2000;
>-const int kPreferredBps = 50000;
> const int kWidth = 640;
> const int kHeight = 480;
> const int kQpIdx0 = 21;
>@@ -106,8 +105,6 @@ class SendStatisticsProxyTest : public ::testing::Test {
>     EXPECT_EQ(one.input_frame_rate, other.input_frame_rate);
>     EXPECT_EQ(one.encode_frame_rate, other.encode_frame_rate);
>     EXPECT_EQ(one.media_bitrate_bps, other.media_bitrate_bps);
>-    EXPECT_EQ(one.preferred_media_bitrate_bps,
>-              other.preferred_media_bitrate_bps);
>     EXPECT_EQ(one.suspended, other.suspended);
> 
>     EXPECT_EQ(one.substreams.size(), other.substreams.size());
>@@ -312,23 +309,13 @@ TEST_F(SendStatisticsProxyTest, SendSideDelay) {
> 
> TEST_F(SendStatisticsProxyTest, OnEncodedFrameTimeMeasured) {
>   const int kEncodeTimeMs = 11;
>-  CpuOveruseMetrics metrics;
>-  metrics.encode_usage_percent = 80;
>-  statistics_proxy_->OnEncodedFrameTimeMeasured(kEncodeTimeMs, metrics);
>+  int encode_usage_percent = 80;
>+  statistics_proxy_->OnEncodedFrameTimeMeasured(kEncodeTimeMs,
>+                                                encode_usage_percent);
> 
>   VideoSendStream::Stats stats = statistics_proxy_->GetStats();
>   EXPECT_EQ(kEncodeTimeMs, stats.avg_encode_time_ms);
>-  EXPECT_EQ(metrics.encode_usage_percent, stats.encode_usage_percent);
>-}
>-
>-TEST_F(SendStatisticsProxyTest, OnEncoderReconfiguredChangePreferredBitrate) {
>-  VideoSendStream::Stats stats = statistics_proxy_->GetStats();
>-  EXPECT_EQ(0, stats.preferred_media_bitrate_bps);
>-
>-  VideoEncoderConfig config;
>-  statistics_proxy_->OnEncoderReconfigured(config, {}, kPreferredBps);
>-  stats = statistics_proxy_->GetStats();
>-  EXPECT_EQ(kPreferredBps, stats.preferred_media_bitrate_bps);
>+  EXPECT_EQ(encode_usage_percent, stats.encode_usage_percent);
> }
> 
> TEST_F(SendStatisticsProxyTest, OnSendEncodedImageIncreasesFramesEncoded) {
>@@ -344,7 +331,7 @@ TEST_F(SendStatisticsProxyTest, OnSendEncodedImageIncreasesFramesEncoded) {
> TEST_F(SendStatisticsProxyTest, OnSendEncodedImageIncreasesQpSum) {
>   EncodedImage encoded_image;
>   CodecSpecificInfo codec_info;
>-  EXPECT_EQ(rtc::nullopt, statistics_proxy_->GetStats().qp_sum);
>+  EXPECT_EQ(absl::nullopt, statistics_proxy_->GetStats().qp_sum);
>   encoded_image.qp_ = 3;
>   statistics_proxy_->OnSendEncodedImage(encoded_image, &codec_info);
>   EXPECT_EQ(3u, statistics_proxy_->GetStats().qp_sum);
>@@ -357,78 +344,98 @@ TEST_F(SendStatisticsProxyTest, OnSendEncodedImageWithoutQpQpSumWontExist) {
>   EncodedImage encoded_image;
>   CodecSpecificInfo codec_info;
>   encoded_image.qp_ = -1;
>-  EXPECT_EQ(rtc::nullopt, statistics_proxy_->GetStats().qp_sum);
>+  EXPECT_EQ(absl::nullopt, statistics_proxy_->GetStats().qp_sum);
>   statistics_proxy_->OnSendEncodedImage(encoded_image, &codec_info);
>-  EXPECT_EQ(rtc::nullopt, statistics_proxy_->GetStats().qp_sum);
>+  EXPECT_EQ(absl::nullopt, statistics_proxy_->GetStats().qp_sum);
> }
> 
> TEST_F(SendStatisticsProxyTest, GetCpuAdaptationStats) {
>-  VideoStreamEncoder::AdaptCounts cpu_counts;
>-  VideoStreamEncoder::AdaptCounts quality_counts;
>+  SendStatisticsProxy::AdaptationSteps cpu_counts;
>+  SendStatisticsProxy::AdaptationSteps quality_counts;
>   EXPECT_FALSE(statistics_proxy_->GetStats().cpu_limited_framerate);
>   EXPECT_FALSE(statistics_proxy_->GetStats().cpu_limited_resolution);
>-  cpu_counts.fps = 1;
>-  cpu_counts.resolution = 0;
>-  statistics_proxy_->SetAdaptationStats(cpu_counts, quality_counts);
>+  cpu_counts.num_framerate_reductions = 1;
>+  cpu_counts.num_resolution_reductions = 0;
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kNone, cpu_counts,
>+      quality_counts);
>   EXPECT_TRUE(statistics_proxy_->GetStats().cpu_limited_framerate);
>   EXPECT_FALSE(statistics_proxy_->GetStats().cpu_limited_resolution);
>-  cpu_counts.fps = 0;
>-  cpu_counts.resolution = 1;
>-  statistics_proxy_->SetAdaptationStats(cpu_counts, quality_counts);
>+  cpu_counts.num_framerate_reductions = 0;
>+  cpu_counts.num_resolution_reductions = 1;
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kNone, cpu_counts,
>+      quality_counts);
>   EXPECT_FALSE(statistics_proxy_->GetStats().cpu_limited_framerate);
>   EXPECT_TRUE(statistics_proxy_->GetStats().cpu_limited_resolution);
>-  cpu_counts.fps = 1;
>-  cpu_counts.resolution = -1;
>-  statistics_proxy_->SetAdaptationStats(cpu_counts, quality_counts);
>+  cpu_counts.num_framerate_reductions = 1;
>+  cpu_counts.num_resolution_reductions = absl::nullopt;
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kNone, cpu_counts,
>+      quality_counts);
>   EXPECT_TRUE(statistics_proxy_->GetStats().cpu_limited_framerate);
>   EXPECT_FALSE(statistics_proxy_->GetStats().cpu_limited_resolution);
>-  cpu_counts.fps = -1;
>-  cpu_counts.resolution = -1;
>-  statistics_proxy_->SetAdaptationStats(cpu_counts, quality_counts);
>+  cpu_counts.num_framerate_reductions = absl::nullopt;
>+  cpu_counts.num_resolution_reductions = absl::nullopt;
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kNone, cpu_counts,
>+      quality_counts);
>   EXPECT_FALSE(statistics_proxy_->GetStats().cpu_limited_framerate);
>   EXPECT_FALSE(statistics_proxy_->GetStats().cpu_limited_resolution);
> }
> 
> TEST_F(SendStatisticsProxyTest, GetQualityAdaptationStats) {
>-  VideoStreamEncoder::AdaptCounts cpu_counts;
>-  VideoStreamEncoder::AdaptCounts quality_counts;
>+  SendStatisticsProxy::AdaptationSteps cpu_counts;
>+  SendStatisticsProxy::AdaptationSteps quality_counts;
>   EXPECT_FALSE(statistics_proxy_->GetStats().bw_limited_framerate);
>   EXPECT_FALSE(statistics_proxy_->GetStats().bw_limited_resolution);
>-  quality_counts.fps = 1;
>-  quality_counts.resolution = 0;
>-  statistics_proxy_->SetAdaptationStats(cpu_counts, quality_counts);
>+  quality_counts.num_framerate_reductions = 1;
>+  quality_counts.num_resolution_reductions = 0;
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kNone, cpu_counts,
>+      quality_counts);
>   EXPECT_TRUE(statistics_proxy_->GetStats().bw_limited_framerate);
>   EXPECT_FALSE(statistics_proxy_->GetStats().bw_limited_resolution);
>-  quality_counts.fps = 0;
>-  quality_counts.resolution = 1;
>-  statistics_proxy_->SetAdaptationStats(cpu_counts, quality_counts);
>+  quality_counts.num_framerate_reductions = 0;
>+  quality_counts.num_resolution_reductions = 1;
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kNone, cpu_counts,
>+      quality_counts);
>   EXPECT_FALSE(statistics_proxy_->GetStats().bw_limited_framerate);
>   EXPECT_TRUE(statistics_proxy_->GetStats().bw_limited_resolution);
>-  quality_counts.fps = 1;
>-  quality_counts.resolution = -1;
>-  statistics_proxy_->SetAdaptationStats(cpu_counts, quality_counts);
>+  quality_counts.num_framerate_reductions = 1;
>+  quality_counts.num_resolution_reductions = absl::nullopt;
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kNone, cpu_counts,
>+      quality_counts);
>   EXPECT_TRUE(statistics_proxy_->GetStats().bw_limited_framerate);
>   EXPECT_FALSE(statistics_proxy_->GetStats().bw_limited_resolution);
>-  quality_counts.fps = -1;
>-  quality_counts.resolution = -1;
>-  statistics_proxy_->SetAdaptationStats(cpu_counts, quality_counts);
>+  quality_counts.num_framerate_reductions = absl::nullopt;
>+  quality_counts.num_resolution_reductions = absl::nullopt;
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kNone, cpu_counts,
>+      quality_counts);
>   EXPECT_FALSE(statistics_proxy_->GetStats().bw_limited_framerate);
>   EXPECT_FALSE(statistics_proxy_->GetStats().bw_limited_resolution);
> }
> 
> TEST_F(SendStatisticsProxyTest, GetStatsReportsCpuAdaptChanges) {
>-  VideoStreamEncoder::AdaptCounts cpu_counts;
>-  VideoStreamEncoder::AdaptCounts quality_counts;
>+  SendStatisticsProxy::AdaptationSteps cpu_counts;
>+  SendStatisticsProxy::AdaptationSteps quality_counts;
>   EXPECT_EQ(0, statistics_proxy_->GetStats().number_of_cpu_adapt_changes);
> 
>-  cpu_counts.resolution = 1;
>-  statistics_proxy_->OnCpuAdaptationChanged(cpu_counts, quality_counts);
>+  cpu_counts.num_resolution_reductions = 1;
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kCpu, cpu_counts,
>+      quality_counts);
>   EXPECT_FALSE(statistics_proxy_->GetStats().cpu_limited_framerate);
>   EXPECT_TRUE(statistics_proxy_->GetStats().cpu_limited_resolution);
>   EXPECT_EQ(1, statistics_proxy_->GetStats().number_of_cpu_adapt_changes);
> 
>-  cpu_counts.resolution = 2;
>-  statistics_proxy_->OnCpuAdaptationChanged(cpu_counts, quality_counts);
>+  cpu_counts.num_resolution_reductions = 2;
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kCpu, cpu_counts,
>+      quality_counts);
>   EXPECT_FALSE(statistics_proxy_->GetStats().cpu_limited_framerate);
>   EXPECT_TRUE(statistics_proxy_->GetStats().cpu_limited_resolution);
>   EXPECT_EQ(2, statistics_proxy_->GetStats().number_of_cpu_adapt_changes);
>@@ -436,18 +443,22 @@ TEST_F(SendStatisticsProxyTest, GetStatsReportsCpuAdaptChanges) {
> }
> 
> TEST_F(SendStatisticsProxyTest, GetStatsReportsQualityAdaptChanges) {
>-  VideoStreamEncoder::AdaptCounts cpu_counts;
>-  VideoStreamEncoder::AdaptCounts quality_counts;
>+  SendStatisticsProxy::AdaptationSteps cpu_counts;
>+  SendStatisticsProxy::AdaptationSteps quality_counts;
>   EXPECT_EQ(0, statistics_proxy_->GetStats().number_of_quality_adapt_changes);
> 
>-  quality_counts.fps = 1;
>-  statistics_proxy_->OnQualityAdaptationChanged(cpu_counts, quality_counts);
>+  quality_counts.num_framerate_reductions = 1;
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kQuality, cpu_counts,
>+      quality_counts);
>   EXPECT_TRUE(statistics_proxy_->GetStats().bw_limited_framerate);
>   EXPECT_FALSE(statistics_proxy_->GetStats().bw_limited_resolution);
>   EXPECT_EQ(1, statistics_proxy_->GetStats().number_of_quality_adapt_changes);
> 
>-  quality_counts.fps = 0;
>-  statistics_proxy_->OnQualityAdaptationChanged(cpu_counts, quality_counts);
>+  quality_counts.num_framerate_reductions = 0;
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kQuality, cpu_counts,
>+      quality_counts);
>   EXPECT_FALSE(statistics_proxy_->GetStats().bw_limited_framerate);
>   EXPECT_FALSE(statistics_proxy_->GetStats().bw_limited_resolution);
>   EXPECT_EQ(2, statistics_proxy_->GetStats().number_of_quality_adapt_changes);
>@@ -469,9 +480,11 @@ TEST_F(SendStatisticsProxyTest, AdaptChangesNotReported_MinRuntimeNotPassed) {
>   // First RTP packet sent.
>   UpdateDataCounters(kFirstSsrc);
>   // Enable adaptation.
>-  VideoStreamEncoder::AdaptCounts cpu_counts;
>-  VideoStreamEncoder::AdaptCounts quality_counts;
>-  statistics_proxy_->SetAdaptationStats(cpu_counts, quality_counts);
>+  SendStatisticsProxy::AdaptationSteps cpu_counts;
>+  SendStatisticsProxy::AdaptationSteps quality_counts;
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kNone, cpu_counts,
>+      quality_counts);
>   // Min runtime has not passed.
>   fake_clock_.AdvanceTimeMilliseconds(metrics::kMinRunTimeInSeconds * 1000 - 1);
>   statistics_proxy_.reset();
>@@ -484,9 +497,11 @@ TEST_F(SendStatisticsProxyTest, ZeroAdaptChangesReported) {
>   // First RTP packet sent.
>   UpdateDataCounters(kFirstSsrc);
>   // Enable adaptation.
>-  VideoStreamEncoder::AdaptCounts cpu_counts;
>-  VideoStreamEncoder::AdaptCounts quality_counts;
>-  statistics_proxy_->SetAdaptationStats(cpu_counts, quality_counts);
>+  SendStatisticsProxy::AdaptationSteps cpu_counts;
>+  SendStatisticsProxy::AdaptationSteps quality_counts;
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kNone, cpu_counts,
>+      quality_counts);
>   // Min runtime has passed.
>   fake_clock_.AdvanceTimeMilliseconds(metrics::kMinRunTimeInSeconds * 1000);
>   statistics_proxy_.reset();
>@@ -502,11 +517,15 @@ TEST_F(SendStatisticsProxyTest, CpuAdaptChangesReported) {
>   // First RTP packet sent.
>   UpdateDataCounters(kFirstSsrc);
>   // Enable adaptation.
>-  VideoStreamEncoder::AdaptCounts cpu_counts;
>-  VideoStreamEncoder::AdaptCounts quality_counts;
>-  statistics_proxy_->SetAdaptationStats(cpu_counts, quality_counts);
>+  SendStatisticsProxy::AdaptationSteps cpu_counts;
>+  SendStatisticsProxy::AdaptationSteps quality_counts;
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kNone, cpu_counts,
>+      quality_counts);
>   // Adapt changes: 1, elapsed time: 10 sec => 6 per minute.
>-  statistics_proxy_->OnCpuAdaptationChanged(cpu_counts, quality_counts);
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kCpu, cpu_counts,
>+      quality_counts);
>   fake_clock_.AdvanceTimeMilliseconds(10000);
>   statistics_proxy_.reset();
>   EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.AdaptChangesPerMinute.Cpu"));
>@@ -517,11 +536,15 @@ TEST_F(SendStatisticsProxyTest, ExcludesInitialQualityAdaptDownChange) {
>   // First RTP packet sent.
>   UpdateDataCounters(kFirstSsrc);
>   // Enable adaptation.
>-  VideoStreamEncoder::AdaptCounts cpu_counts;
>-  VideoStreamEncoder::AdaptCounts quality_counts;
>-  statistics_proxy_->SetAdaptationStats(cpu_counts, quality_counts);
>+  SendStatisticsProxy::AdaptationSteps cpu_counts;
>+  SendStatisticsProxy::AdaptationSteps quality_counts;
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kNone, cpu_counts,
>+      quality_counts);
>   // Adapt changes: 1 (1 initial) = 0, elapsed time: 10 sec => 0 per minute.
>-  statistics_proxy_->OnQualityAdaptationChanged(cpu_counts, quality_counts);
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kQuality, cpu_counts,
>+      quality_counts);
>   statistics_proxy_->OnInitialQualityResolutionAdaptDown();
>   fake_clock_.AdvanceTimeMilliseconds(10000);
>   statistics_proxy_.reset();
>@@ -535,18 +558,26 @@ TEST_F(SendStatisticsProxyTest, ExcludesInitialQualityAdaptDownChanges) {
>   // First RTP packet sent.
>   UpdateDataCounters(kFirstSsrc);
>   // Enable adaptation.
>-  VideoStreamEncoder::AdaptCounts cpu_counts;
>-  VideoStreamEncoder::AdaptCounts quality_counts;
>-  statistics_proxy_->SetAdaptationStats(cpu_counts, quality_counts);
>+  SendStatisticsProxy::AdaptationSteps cpu_counts;
>+  SendStatisticsProxy::AdaptationSteps quality_counts;
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kNone, cpu_counts,
>+      quality_counts);
>   // Adapt changes: 3 (2 initial) = 1, elapsed time: 10 sec => 6 per minute.
>-  quality_counts.resolution = 1;
>-  statistics_proxy_->OnQualityAdaptationChanged(cpu_counts, quality_counts);
>+  quality_counts.num_resolution_reductions = 1;
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kQuality, cpu_counts,
>+      quality_counts);
>   statistics_proxy_->OnInitialQualityResolutionAdaptDown();
>-  quality_counts.resolution = 2;
>-  statistics_proxy_->OnQualityAdaptationChanged(cpu_counts, quality_counts);
>+  quality_counts.num_resolution_reductions = 2;
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kQuality, cpu_counts,
>+      quality_counts);
>   statistics_proxy_->OnInitialQualityResolutionAdaptDown();
>-  quality_counts.resolution = 3;
>-  statistics_proxy_->OnQualityAdaptationChanged(cpu_counts, quality_counts);
>+  quality_counts.num_resolution_reductions = 3;
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kQuality, cpu_counts,
>+      quality_counts);
>   fake_clock_.AdvanceTimeMilliseconds(10000);
>   statistics_proxy_.reset();
>   EXPECT_EQ(1,
>@@ -559,11 +590,15 @@ TEST_F(SendStatisticsProxyTest, InitialQualityAdaptChangesNotExcludedOnError) {
>   // First RTP packet sent.
>   UpdateDataCounters(kFirstSsrc);
>   // Enable adaptation.
>-  VideoStreamEncoder::AdaptCounts cpu_counts;
>-  VideoStreamEncoder::AdaptCounts quality_counts;
>-  statistics_proxy_->SetAdaptationStats(cpu_counts, quality_counts);
>+  SendStatisticsProxy::AdaptationSteps cpu_counts;
>+  SendStatisticsProxy::AdaptationSteps quality_counts;
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kNone, cpu_counts,
>+      quality_counts);
>   // Adapt changes: 1 (2 initial) = 1, elapsed time: 10 sec => 6 per minute.
>-  statistics_proxy_->OnQualityAdaptationChanged(cpu_counts, quality_counts);
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kQuality, cpu_counts,
>+      quality_counts);
>   statistics_proxy_->OnInitialQualityResolutionAdaptDown();
>   statistics_proxy_->OnInitialQualityResolutionAdaptDown();
>   fake_clock_.AdvanceTimeMilliseconds(10000);
>@@ -578,28 +613,46 @@ TEST_F(SendStatisticsProxyTest, ExcludesInitialQualityAdaptDownAndUpChanges) {
>   // First RTP packet sent.
>   UpdateDataCounters(kFirstSsrc);
>   // Enable adaptation.
>-  VideoStreamEncoder::AdaptCounts cpu_counts;
>-  VideoStreamEncoder::AdaptCounts quality_counts;
>-  statistics_proxy_->SetAdaptationStats(cpu_counts, quality_counts);
>+  SendStatisticsProxy::AdaptationSteps cpu_counts;
>+  SendStatisticsProxy::AdaptationSteps quality_counts;
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kNone, cpu_counts,
>+      quality_counts);
>   // Adapt changes: 8 (4 initial) = 4, elapsed time: 10 sec => 24 per minute.
>-  quality_counts.resolution = 1;
>-  statistics_proxy_->OnQualityAdaptationChanged(cpu_counts, quality_counts);
>+  quality_counts.num_resolution_reductions = 1;
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kQuality, cpu_counts,
>+      quality_counts);
>   statistics_proxy_->OnInitialQualityResolutionAdaptDown();
>-  quality_counts.resolution = 2;
>-  statistics_proxy_->OnQualityAdaptationChanged(cpu_counts, quality_counts);
>+  quality_counts.num_resolution_reductions = 2;
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kQuality, cpu_counts,
>+      quality_counts);
>   statistics_proxy_->OnInitialQualityResolutionAdaptDown();
>-  quality_counts.resolution = 3;
>-  statistics_proxy_->OnQualityAdaptationChanged(cpu_counts, quality_counts);
>-  quality_counts.fps = 1;
>-  statistics_proxy_->OnQualityAdaptationChanged(cpu_counts, quality_counts);
>-  quality_counts.fps = 0;
>-  statistics_proxy_->OnQualityAdaptationChanged(cpu_counts, quality_counts);
>-  quality_counts.resolution = 2;  // Initial resolution up.
>-  statistics_proxy_->OnQualityAdaptationChanged(cpu_counts, quality_counts);
>-  quality_counts.resolution = 1;  // Initial resolution up.
>-  statistics_proxy_->OnQualityAdaptationChanged(cpu_counts, quality_counts);
>-  quality_counts.resolution = 0;
>-  statistics_proxy_->OnQualityAdaptationChanged(cpu_counts, quality_counts);
>+  quality_counts.num_resolution_reductions = 3;
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kQuality, cpu_counts,
>+      quality_counts);
>+  quality_counts.num_framerate_reductions = 1;
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kQuality, cpu_counts,
>+      quality_counts);
>+  quality_counts.num_framerate_reductions = 0;
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kQuality, cpu_counts,
>+      quality_counts);
>+  quality_counts.num_resolution_reductions = 2;  // Initial resolution up.
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kQuality, cpu_counts,
>+      quality_counts);
>+  quality_counts.num_resolution_reductions = 1;  // Initial resolution up.
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kQuality, cpu_counts,
>+      quality_counts);
>+  quality_counts.num_resolution_reductions = 0;
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kQuality, cpu_counts,
>+      quality_counts);
> 
>   fake_clock_.AdvanceTimeMilliseconds(10000);
>   statistics_proxy_.reset();
>@@ -614,41 +667,61 @@ TEST_F(SendStatisticsProxyTest, AdaptChangesStatsExcludesDisabledTime) {
>   UpdateDataCounters(kFirstSsrc);
> 
>   // Disable quality adaptation.
>-  VideoStreamEncoder::AdaptCounts cpu_counts;
>-  VideoStreamEncoder::AdaptCounts quality_counts;
>-  quality_counts.fps = -1;
>-  quality_counts.resolution = -1;
>-  statistics_proxy_->SetAdaptationStats(cpu_counts, quality_counts);
>+  SendStatisticsProxy::AdaptationSteps cpu_counts;
>+  SendStatisticsProxy::AdaptationSteps quality_counts;
>+  quality_counts.num_framerate_reductions = absl::nullopt;
>+  quality_counts.num_resolution_reductions = absl::nullopt;
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kNone, cpu_counts,
>+      quality_counts);
>   fake_clock_.AdvanceTimeMilliseconds(10000);
> 
>   // Enable quality adaptation.
>   // Adapt changes: 2, elapsed time: 20 sec.
>-  quality_counts.fps = 0;
>-  statistics_proxy_->SetAdaptationStats(cpu_counts, quality_counts);
>+  quality_counts.num_framerate_reductions = 0;
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kNone, cpu_counts,
>+      quality_counts);
>   fake_clock_.AdvanceTimeMilliseconds(5000);
>-  statistics_proxy_->SetAdaptationStats(cpu_counts, quality_counts);
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kNone, cpu_counts,
>+      quality_counts);
>   fake_clock_.AdvanceTimeMilliseconds(9000);
>-  statistics_proxy_->OnQualityAdaptationChanged(cpu_counts, quality_counts);
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kQuality, cpu_counts,
>+      quality_counts);
>   fake_clock_.AdvanceTimeMilliseconds(6000);
>-  statistics_proxy_->OnQualityAdaptationChanged(cpu_counts, quality_counts);
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kQuality, cpu_counts,
>+      quality_counts);
> 
>   // Disable quality adaptation.
>-  quality_counts.fps = -1;
>-  statistics_proxy_->SetAdaptationStats(cpu_counts, quality_counts);
>+  quality_counts.num_framerate_reductions = absl::nullopt;
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kNone, cpu_counts,
>+      quality_counts);
>   fake_clock_.AdvanceTimeMilliseconds(30000);
> 
>   // Enable quality adaptation.
>   // Adapt changes: 1, elapsed time: 10 sec.
>-  quality_counts.resolution = 0;
>-  statistics_proxy_->SetAdaptationStats(cpu_counts, quality_counts);
>-  statistics_proxy_->OnQualityAdaptationChanged(cpu_counts, quality_counts);
>+  quality_counts.num_resolution_reductions = 0;
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kNone, cpu_counts,
>+      quality_counts);
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kQuality, cpu_counts,
>+      quality_counts);
>   fake_clock_.AdvanceTimeMilliseconds(10000);
> 
>   // Disable quality adaptation.
>-  quality_counts.resolution = -1;
>-  statistics_proxy_->SetAdaptationStats(cpu_counts, quality_counts);
>+  quality_counts.num_resolution_reductions = absl::nullopt;
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kNone, cpu_counts,
>+      quality_counts);
>   fake_clock_.AdvanceTimeMilliseconds(5000);
>-  statistics_proxy_->SetAdaptationStats(cpu_counts, quality_counts);
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kNone, cpu_counts,
>+      quality_counts);
>   fake_clock_.AdvanceTimeMilliseconds(20000);
> 
>   // Adapt changes: 3, elapsed time: 30 sec => 6 per minute.
>@@ -682,13 +755,19 @@ TEST_F(SendStatisticsProxyTest, QualityAdaptChangesStatsExcludesSuspendedTime) {
>   UpdateDataCounters(kFirstSsrc);
> 
>   // Enable adaptation.
>-  VideoStreamEncoder::AdaptCounts cpu_counts;
>-  VideoStreamEncoder::AdaptCounts quality_counts;
>+  SendStatisticsProxy::AdaptationSteps cpu_counts;
>+  SendStatisticsProxy::AdaptationSteps quality_counts;
>   // Adapt changes: 2, elapsed time: 20 sec.
>-  statistics_proxy_->SetAdaptationStats(cpu_counts, quality_counts);
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kNone, cpu_counts,
>+      quality_counts);
>   fake_clock_.AdvanceTimeMilliseconds(20000);
>-  statistics_proxy_->OnQualityAdaptationChanged(cpu_counts, quality_counts);
>-  statistics_proxy_->OnQualityAdaptationChanged(cpu_counts, quality_counts);
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kQuality, cpu_counts,
>+      quality_counts);
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kQuality, cpu_counts,
>+      quality_counts);
> 
>   // Suspend and resume video.
>   statistics_proxy_->OnSuspendChange(true);
>@@ -696,7 +775,9 @@ TEST_F(SendStatisticsProxyTest, QualityAdaptChangesStatsExcludesSuspendedTime) {
>   statistics_proxy_->OnSuspendChange(false);
> 
>   // Adapt changes: 1, elapsed time: 10 sec.
>-  statistics_proxy_->OnQualityAdaptationChanged(cpu_counts, quality_counts);
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kQuality, cpu_counts,
>+      quality_counts);
>   fake_clock_.AdvanceTimeMilliseconds(10000);
> 
>   // Adapt changes: 3, elapsed time: 30 sec => 6 per minute.
>@@ -716,21 +797,27 @@ TEST_F(SendStatisticsProxyTest, CpuAdaptChangesStatsExcludesSuspendedTime) {
>   fake_clock_.AdvanceTimeMilliseconds(30000);
> 
>   // Enable adaptation.
>-  VideoStreamEncoder::AdaptCounts cpu_counts;
>-  VideoStreamEncoder::AdaptCounts quality_counts;
>+  SendStatisticsProxy::AdaptationSteps cpu_counts;
>+  SendStatisticsProxy::AdaptationSteps quality_counts;
>   // Adapt changes: 1, elapsed time: 20 sec.
>-  statistics_proxy_->SetAdaptationStats(cpu_counts, quality_counts);
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kNone, cpu_counts,
>+      quality_counts);
>   fake_clock_.AdvanceTimeMilliseconds(10000);
>-  statistics_proxy_->OnCpuAdaptationChanged(cpu_counts, quality_counts);
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kCpu, cpu_counts,
>+      quality_counts);
> 
>   // Video not suspended, stats time already started.
>   statistics_proxy_->OnSuspendChange(false);
>   fake_clock_.AdvanceTimeMilliseconds(10000);
> 
>   // Disable adaptation.
>-  cpu_counts.fps = -1;
>-  cpu_counts.resolution = -1;
>-  statistics_proxy_->SetAdaptationStats(cpu_counts, quality_counts);
>+  cpu_counts.num_framerate_reductions = absl::nullopt;
>+  cpu_counts.num_resolution_reductions = absl::nullopt;
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kNone, cpu_counts,
>+      quality_counts);
>   fake_clock_.AdvanceTimeMilliseconds(30000);
> 
>   // Suspend and resume video, stats time not started when scaling not enabled.
>@@ -741,11 +828,15 @@ TEST_F(SendStatisticsProxyTest, CpuAdaptChangesStatsExcludesSuspendedTime) {
> 
>   // Enable adaptation.
>   // Adapt changes: 1, elapsed time: 10 sec.
>-  cpu_counts.fps = 0;
>-  cpu_counts.resolution = 0;
>-  statistics_proxy_->SetAdaptationStats(cpu_counts, quality_counts);
>+  cpu_counts.num_framerate_reductions = 0;
>+  cpu_counts.num_resolution_reductions = 0;
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kNone, cpu_counts,
>+      quality_counts);
>   fake_clock_.AdvanceTimeMilliseconds(10000);
>-  statistics_proxy_->OnCpuAdaptationChanged(cpu_counts, quality_counts);
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kCpu, cpu_counts,
>+      quality_counts);
> 
>   // Adapt changes: 2, elapsed time: 30 sec => 4 per minute.
>   statistics_proxy_.reset();
>@@ -761,16 +852,20 @@ TEST_F(SendStatisticsProxyTest, AdaptChangesStatsNotStartedIfVideoSuspended) {
>   statistics_proxy_->OnSuspendChange(true);
> 
>   // Enable adaptation, stats time not started when suspended.
>-  VideoStreamEncoder::AdaptCounts cpu_counts;
>-  VideoStreamEncoder::AdaptCounts quality_counts;
>-  statistics_proxy_->SetAdaptationStats(cpu_counts, quality_counts);
>+  SendStatisticsProxy::AdaptationSteps cpu_counts;
>+  SendStatisticsProxy::AdaptationSteps quality_counts;
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kNone, cpu_counts,
>+      quality_counts);
>   fake_clock_.AdvanceTimeMilliseconds(10000);
> 
>   // Resume video, stats time started.
>   // Adapt changes: 1, elapsed time: 10 sec.
>   statistics_proxy_->OnSuspendChange(false);
>   fake_clock_.AdvanceTimeMilliseconds(10000);
>-  statistics_proxy_->OnCpuAdaptationChanged(cpu_counts, quality_counts);
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kCpu, cpu_counts,
>+      quality_counts);
> 
>   // Adapt changes: 1, elapsed time: 10 sec => 6 per minute.
>   statistics_proxy_.reset();
>@@ -781,15 +876,19 @@ TEST_F(SendStatisticsProxyTest, AdaptChangesStatsNotStartedIfVideoSuspended) {
> TEST_F(SendStatisticsProxyTest, AdaptChangesStatsRestartsOnFirstSentPacket) {
>   // Send first packet, adaptation enabled.
>   // Elapsed time before first packet is sent should be excluded.
>-  VideoStreamEncoder::AdaptCounts cpu_counts;
>-  VideoStreamEncoder::AdaptCounts quality_counts;
>-  statistics_proxy_->SetAdaptationStats(cpu_counts, quality_counts);
>+  SendStatisticsProxy::AdaptationSteps cpu_counts;
>+  SendStatisticsProxy::AdaptationSteps quality_counts;
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kNone, cpu_counts,
>+      quality_counts);
>   fake_clock_.AdvanceTimeMilliseconds(10000);
>   UpdateDataCounters(kFirstSsrc);
> 
>   // Adapt changes: 1, elapsed time: 10 sec.
>   fake_clock_.AdvanceTimeMilliseconds(10000);
>-  statistics_proxy_->OnQualityAdaptationChanged(cpu_counts, quality_counts);
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kQuality, cpu_counts,
>+      quality_counts);
>   UpdateDataCounters(kFirstSsrc);
> 
>   // Adapt changes: 1, elapsed time: 10 sec => 6 per minute.
>@@ -802,13 +901,17 @@ TEST_F(SendStatisticsProxyTest, AdaptChangesStatsRestartsOnFirstSentPacket) {
> 
> TEST_F(SendStatisticsProxyTest, AdaptChangesStatsStartedAfterFirstSentPacket) {
>   // Enable and disable adaptation.
>-  VideoStreamEncoder::AdaptCounts cpu_counts;
>-  VideoStreamEncoder::AdaptCounts quality_counts;
>-  statistics_proxy_->SetAdaptationStats(cpu_counts, quality_counts);
>+  SendStatisticsProxy::AdaptationSteps cpu_counts;
>+  SendStatisticsProxy::AdaptationSteps quality_counts;
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kNone, cpu_counts,
>+      quality_counts);
>   fake_clock_.AdvanceTimeMilliseconds(60000);
>-  cpu_counts.fps = -1;
>-  cpu_counts.resolution = -1;
>-  statistics_proxy_->SetAdaptationStats(cpu_counts, quality_counts);
>+  cpu_counts.num_framerate_reductions = absl::nullopt;
>+  cpu_counts.num_resolution_reductions = absl::nullopt;
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kNone, cpu_counts,
>+      quality_counts);
> 
>   // Send first packet, scaling disabled.
>   // Elapsed time before first packet is sent should be excluded.
>@@ -816,14 +919,18 @@ TEST_F(SendStatisticsProxyTest, AdaptChangesStatsStartedAfterFirstSentPacket) {
>   fake_clock_.AdvanceTimeMilliseconds(60000);
> 
>   // Enable adaptation.
>-  cpu_counts.resolution = 0;
>-  statistics_proxy_->SetAdaptationStats(cpu_counts, quality_counts);
>+  cpu_counts.num_resolution_reductions = 0;
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kNone, cpu_counts,
>+      quality_counts);
>   fake_clock_.AdvanceTimeMilliseconds(10000);
>   UpdateDataCounters(kFirstSsrc);
> 
>   // Adapt changes: 1, elapsed time: 20 sec.
>   fake_clock_.AdvanceTimeMilliseconds(10000);
>-  statistics_proxy_->OnCpuAdaptationChanged(cpu_counts, quality_counts);
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kCpu, cpu_counts,
>+      quality_counts);
> 
>   // Adapt changes: 1, elapsed time: 20 sec => 3 per minute.
>   statistics_proxy_.reset();
>@@ -834,22 +941,28 @@ TEST_F(SendStatisticsProxyTest, AdaptChangesStatsStartedAfterFirstSentPacket) {
> TEST_F(SendStatisticsProxyTest, AdaptChangesReportedAfterContentSwitch) {
>   // First RTP packet sent, cpu adaptation enabled.
>   UpdateDataCounters(kFirstSsrc);
>-  VideoStreamEncoder::AdaptCounts cpu_counts;
>-  VideoStreamEncoder::AdaptCounts quality_counts;
>-  quality_counts.fps = -1;
>-  quality_counts.resolution = -1;
>-  statistics_proxy_->SetAdaptationStats(cpu_counts, quality_counts);
>+  SendStatisticsProxy::AdaptationSteps cpu_counts;
>+  SendStatisticsProxy::AdaptationSteps quality_counts;
>+  quality_counts.num_framerate_reductions = absl::nullopt;
>+  quality_counts.num_resolution_reductions = absl::nullopt;
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kNone, cpu_counts,
>+      quality_counts);
> 
>   // Adapt changes: 2, elapsed time: 15 sec => 8 per minute.
>-  statistics_proxy_->OnCpuAdaptationChanged(cpu_counts, quality_counts);
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kCpu, cpu_counts,
>+      quality_counts);
>   fake_clock_.AdvanceTimeMilliseconds(6000);
>-  statistics_proxy_->OnCpuAdaptationChanged(cpu_counts, quality_counts);
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kCpu, cpu_counts,
>+      quality_counts);
>   fake_clock_.AdvanceTimeMilliseconds(9000);
> 
>   // Switch content type, real-time stats should be updated.
>   VideoEncoderConfig config;
>   config.content_type = VideoEncoderConfig::ContentType::kScreen;
>-  statistics_proxy_->OnEncoderReconfigured(config, {}, kPreferredBps);
>+  statistics_proxy_->OnEncoderReconfigured(config, {});
>   EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.AdaptChangesPerMinute.Cpu"));
>   EXPECT_EQ(1, metrics::NumEvents("WebRTC.Video.AdaptChangesPerMinute.Cpu", 8));
>   EXPECT_EQ(0,
>@@ -857,13 +970,23 @@ TEST_F(SendStatisticsProxyTest, AdaptChangesReportedAfterContentSwitch) {
> 
>   // First RTP packet sent, scaling enabled.
>   UpdateDataCounters(kFirstSsrc);
>-  statistics_proxy_->SetAdaptationStats(cpu_counts, quality_counts);
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kNone, cpu_counts,
>+      quality_counts);
> 
>   // Adapt changes: 4, elapsed time: 120 sec => 2 per minute.
>-  statistics_proxy_->OnCpuAdaptationChanged(cpu_counts, quality_counts);
>-  statistics_proxy_->OnCpuAdaptationChanged(cpu_counts, quality_counts);
>-  statistics_proxy_->OnCpuAdaptationChanged(cpu_counts, quality_counts);
>-  statistics_proxy_->OnCpuAdaptationChanged(cpu_counts, quality_counts);
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kCpu, cpu_counts,
>+      quality_counts);
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kCpu, cpu_counts,
>+      quality_counts);
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kCpu, cpu_counts,
>+      quality_counts);
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kCpu, cpu_counts,
>+      quality_counts);
>   fake_clock_.AdvanceTimeMilliseconds(120000);
> 
>   statistics_proxy_.reset();
>@@ -882,12 +1005,12 @@ TEST_F(SendStatisticsProxyTest, SwitchContentTypeUpdatesHistograms) {
>   // No switch, stats should not be updated.
>   VideoEncoderConfig config;
>   config.content_type = VideoEncoderConfig::ContentType::kRealtimeVideo;
>-  statistics_proxy_->OnEncoderReconfigured(config, {}, kPreferredBps);
>+  statistics_proxy_->OnEncoderReconfigured(config, {});
>   EXPECT_EQ(0, metrics::NumSamples("WebRTC.Video.InputWidthInPixels"));
> 
>   // Switch to screenshare, real-time stats should be updated.
>   config.content_type = VideoEncoderConfig::ContentType::kScreen;
>-  statistics_proxy_->OnEncoderReconfigured(config, {}, kPreferredBps);
>+  statistics_proxy_->OnEncoderReconfigured(config, {});
>   EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.InputWidthInPixels"));
> }
> 
>@@ -904,7 +1027,7 @@ TEST_F(SendStatisticsProxyTest, InputResolutionHistogramsAreUpdated) {
> 
> TEST_F(SendStatisticsProxyTest, SentResolutionHistogramsAreUpdated) {
>   const int64_t kMaxEncodedFrameWindowMs = 800;
>-  const int kFps = 20;
>+  const int kFps = 5;
>   const int kNumFramesPerWindow = kFps * kMaxEncodedFrameWindowMs / 1000;
>   const int kMinSamples =  // Sample added when removed from EncodedFrameMap.
>       SendStatisticsProxy::kMinRequiredMetricsSamples + kNumFramesPerWindow;
>@@ -913,7 +1036,7 @@ TEST_F(SendStatisticsProxyTest, SentResolutionHistogramsAreUpdated) {
>   // Not enough samples, stats should not be updated.
>   for (int i = 0; i < kMinSamples - 1; ++i) {
>     fake_clock_.AdvanceTimeMilliseconds(1000 / kFps);
>-    ++encoded_image._timeStamp;
>+    encoded_image._timeStamp += 90 * 1000 / kFps;
>     statistics_proxy_->OnSendEncodedImage(encoded_image, nullptr);
>   }
>   SetUp();  // Reset stats proxy also causes histograms to be reported.
>@@ -921,10 +1044,10 @@ TEST_F(SendStatisticsProxyTest, SentResolutionHistogramsAreUpdated) {
>   EXPECT_EQ(0, metrics::NumSamples("WebRTC.Video.SentHeightInPixels"));
> 
>   // Enough samples, max resolution per frame should be reported.
>-  encoded_image._timeStamp = 0xfffffff0;  // Will wrap.
>+  encoded_image._timeStamp = 0xffff0000;  // Will wrap.
>   for (int i = 0; i < kMinSamples; ++i) {
>     fake_clock_.AdvanceTimeMilliseconds(1000 / kFps);
>-    ++encoded_image._timeStamp;
>+    encoded_image._timeStamp += 90 * 1000 / kFps;
>     encoded_image._encodedWidth = kWidth;
>     encoded_image._encodedHeight = kHeight;
>     statistics_proxy_->OnSendEncodedImage(encoded_image, nullptr);
>@@ -1020,10 +1143,12 @@ TEST_F(SendStatisticsProxyTest, SentFpsHistogramExcludesSuspendedTime) {
> }
> 
> TEST_F(SendStatisticsProxyTest, CpuLimitedHistogramNotUpdatedWhenDisabled) {
>-  VideoStreamEncoder::AdaptCounts cpu_counts;
>-  VideoStreamEncoder::AdaptCounts quality_counts;
>-  cpu_counts.resolution = -1;
>-  statistics_proxy_->SetAdaptationStats(cpu_counts, quality_counts);
>+  SendStatisticsProxy::AdaptationSteps cpu_counts;
>+  SendStatisticsProxy::AdaptationSteps quality_counts;
>+  cpu_counts.num_resolution_reductions = absl::nullopt;
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kNone, cpu_counts,
>+      quality_counts);
> 
>   for (int i = 0; i < SendStatisticsProxy::kMinRequiredMetricsSamples; ++i)
>     statistics_proxy_->OnIncomingFrame(kWidth, kHeight);
>@@ -1034,16 +1159,20 @@ TEST_F(SendStatisticsProxyTest, CpuLimitedHistogramNotUpdatedWhenDisabled) {
> }
> 
> TEST_F(SendStatisticsProxyTest, CpuLimitedHistogramUpdated) {
>-  VideoStreamEncoder::AdaptCounts cpu_counts;
>-  VideoStreamEncoder::AdaptCounts quality_counts;
>-  cpu_counts.resolution = 0;
>-  statistics_proxy_->SetAdaptationStats(cpu_counts, quality_counts);
>+  SendStatisticsProxy::AdaptationSteps cpu_counts;
>+  SendStatisticsProxy::AdaptationSteps quality_counts;
>+  cpu_counts.num_resolution_reductions = 0;
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kNone, cpu_counts,
>+      quality_counts);
> 
>   for (int i = 0; i < SendStatisticsProxy::kMinRequiredMetricsSamples; ++i)
>     statistics_proxy_->OnIncomingFrame(kWidth, kHeight);
> 
>-  cpu_counts.resolution = 1;
>-  statistics_proxy_->OnCpuAdaptationChanged(cpu_counts, quality_counts);
>+  cpu_counts.num_resolution_reductions = 1;
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kCpu, cpu_counts,
>+      quality_counts);
> 
>   for (int i = 0; i < SendStatisticsProxy::kMinRequiredMetricsSamples; ++i)
>     statistics_proxy_->OnIncomingFrame(kWidth, kHeight);
>@@ -1266,12 +1395,18 @@ TEST_F(SendStatisticsProxyTest, VerifyQpHistogramStats_H264) {
>   codec_info.codecType = kVideoCodecH264;
> 
>   for (int i = 0; i < SendStatisticsProxy::kMinRequiredMetricsSamples; ++i) {
>+    codec_info.codecSpecific.H264.simulcast_idx = 0;
>     encoded_image.qp_ = kQpIdx0;
>     statistics_proxy_->OnSendEncodedImage(encoded_image, &codec_info);
>+    codec_info.codecSpecific.H264.simulcast_idx = 1;
>+    encoded_image.qp_ = kQpIdx1;
>+    statistics_proxy_->OnSendEncodedImage(encoded_image, &codec_info);
>   }
>   statistics_proxy_.reset();
>-  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.Encoded.Qp.H264"));
>-  EXPECT_EQ(1, metrics::NumEvents("WebRTC.Video.Encoded.Qp.H264", kQpIdx0));
>+  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.Encoded.Qp.H264.S0"));
>+  EXPECT_EQ(1, metrics::NumEvents("WebRTC.Video.Encoded.Qp.H264.S0", kQpIdx0));
>+  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.Encoded.Qp.H264.S1"));
>+  EXPECT_EQ(1, metrics::NumEvents("WebRTC.Video.Encoded.Qp.H264.S1", kQpIdx1));
> }
> 
> TEST_F(SendStatisticsProxyTest,
>@@ -1282,7 +1417,7 @@ TEST_F(SendStatisticsProxyTest,
>   VideoStream stream1;
>   stream1.width = kWidth;
>   stream1.height = kHeight;
>-  statistics_proxy_->OnEncoderReconfigured(config, {stream1}, kPreferredBps);
>+  statistics_proxy_->OnEncoderReconfigured(config, {stream1});
> 
>   const int64_t kMaxEncodedFrameWindowMs = 800;
>   const int kFps = 20;
>@@ -1319,8 +1454,7 @@ TEST_F(SendStatisticsProxyTest,
>   VideoStream stream2;
>   stream2.width = kWidth;
>   stream2.height = kHeight;
>-  statistics_proxy_->OnEncoderReconfigured(config, {stream1, stream2},
>-                                           kPreferredBps);
>+  statistics_proxy_->OnEncoderReconfigured(config, {stream1, stream2});
> 
>   const int64_t kMaxEncodedFrameWindowMs = 800;
>   const int kFps = 20;
>@@ -1363,8 +1497,7 @@ TEST_F(SendStatisticsProxyTest,
>   VideoStream stream2;
>   stream2.width = kWidth;
>   stream2.height = kHeight;
>-  statistics_proxy_->OnEncoderReconfigured(config, {stream1, stream2},
>-                                           kPreferredBps);
>+  statistics_proxy_->OnEncoderReconfigured(config, {stream1, stream2});
> 
>   const int64_t kMaxEncodedFrameWindowMs = 800;
>   const int kFps = 20;
>@@ -1397,10 +1530,12 @@ TEST_F(SendStatisticsProxyTest,
> 
> TEST_F(SendStatisticsProxyTest,
>        QualityLimitedHistogramsNotUpdatedWhenDisabled) {
>-  VideoStreamEncoder::AdaptCounts cpu_counts;
>-  VideoStreamEncoder::AdaptCounts quality_counts;
>-  quality_counts.resolution = -1;
>-  statistics_proxy_->SetAdaptationStats(cpu_counts, quality_counts);
>+  SendStatisticsProxy::AdaptationSteps cpu_counts;
>+  SendStatisticsProxy::AdaptationSteps quality_counts;
>+  quality_counts.num_resolution_reductions = absl::nullopt;
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kNone, cpu_counts,
>+      quality_counts);
>   EncodedImage encoded_image;
>   for (int i = 0; i < SendStatisticsProxy::kMinRequiredMetricsSamples; ++i)
>     statistics_proxy_->OnSendEncodedImage(encoded_image, &kDefaultCodecInfo);
>@@ -1415,10 +1550,12 @@ TEST_F(SendStatisticsProxyTest,
> 
> TEST_F(SendStatisticsProxyTest,
>        QualityLimitedHistogramsUpdatedWhenEnabled_NoResolutionDownscale) {
>-  VideoStreamEncoder::AdaptCounts cpu_counts;
>-  VideoStreamEncoder::AdaptCounts quality_counts;
>-  quality_counts.resolution = 0;
>-  statistics_proxy_->SetAdaptationStats(cpu_counts, quality_counts);
>+  SendStatisticsProxy::AdaptationSteps cpu_counts;
>+  SendStatisticsProxy::AdaptationSteps quality_counts;
>+  quality_counts.num_resolution_reductions = 0;
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kNone, cpu_counts,
>+      quality_counts);
>   EncodedImage encoded_image;
>   for (int i = 0; i < SendStatisticsProxy::kMinRequiredMetricsSamples; ++i)
>     statistics_proxy_->OnSendEncodedImage(encoded_image, &kDefaultCodecInfo);
>@@ -1437,10 +1574,12 @@ TEST_F(SendStatisticsProxyTest,
> TEST_F(SendStatisticsProxyTest,
>        QualityLimitedHistogramsUpdatedWhenEnabled_TwoResolutionDownscales) {
>   const int kDownscales = 2;
>-  VideoStreamEncoder::AdaptCounts cpu_counts;
>-  VideoStreamEncoder::AdaptCounts quality_counts;
>-  quality_counts.resolution = kDownscales;
>-  statistics_proxy_->SetAdaptationStats(cpu_counts, quality_counts);
>+  SendStatisticsProxy::AdaptationSteps cpu_counts;
>+  SendStatisticsProxy::AdaptationSteps quality_counts;
>+  quality_counts.num_resolution_reductions = kDownscales;
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kNone, cpu_counts,
>+      quality_counts);
>   EncodedImage encoded_image;
>   for (int i = 0; i < SendStatisticsProxy::kMinRequiredMetricsSamples; ++i)
>     statistics_proxy_->OnSendEncodedImage(encoded_image, &kDefaultCodecInfo);
>@@ -1471,8 +1610,7 @@ TEST_F(SendStatisticsProxyTest, GetStatsReportsBandwidthLimitedResolution) {
>   VideoStream stream2;
>   stream2.width = kWidth;
>   stream2.height = kHeight;
>-  statistics_proxy_->OnEncoderReconfigured(config, {stream1, stream2},
>-                                           kPreferredBps);
>+  statistics_proxy_->OnEncoderReconfigured(config, {stream1, stream2});
> 
>   const int64_t kMaxEncodedFrameWindowMs = 800;
>   const int kFps = 20;
>@@ -1517,10 +1655,12 @@ TEST_F(SendStatisticsProxyTest, GetStatsReportsBandwidthLimitedResolution) {
>   EXPECT_FALSE(statistics_proxy_->GetStats().bw_limited_resolution);
> 
>   // Resolution scaled due to quality.
>-  VideoStreamEncoder::AdaptCounts cpu_counts;
>-  VideoStreamEncoder::AdaptCounts quality_counts;
>-  quality_counts.resolution = 1;
>-  statistics_proxy_->OnQualityAdaptationChanged(cpu_counts, quality_counts);
>+  SendStatisticsProxy::AdaptationSteps cpu_counts;
>+  SendStatisticsProxy::AdaptationSteps quality_counts;
>+  quality_counts.num_resolution_reductions = 1;
>+  statistics_proxy_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kQuality, cpu_counts,
>+      quality_counts);
>   statistics_proxy_->OnSendEncodedImage(encoded_image, nullptr);
>   EXPECT_TRUE(statistics_proxy_->GetStats().bw_limited_resolution);
> }
>@@ -1675,7 +1815,7 @@ TEST_F(SendStatisticsProxyTest, ResetsRtcpCountersOnContentChange) {
>   // Changing content type causes histograms to be reported.
>   VideoEncoderConfig config;
>   config.content_type = VideoEncoderConfig::ContentType::kScreen;
>-  statistics_proxy_->OnEncoderReconfigured(config, {}, kPreferredBps);
>+  statistics_proxy_->OnEncoderReconfigured(config, {});
> 
>   EXPECT_EQ(1,
>             metrics::NumSamples("WebRTC.Video.NackPacketsReceivedPerMinute"));
>@@ -1839,7 +1979,7 @@ TEST_F(SendStatisticsProxyTest, ResetsRtpCountersOnContentChange) {
>   // Changing content type causes histograms to be reported.
>   VideoEncoderConfig config;
>   config.content_type = VideoEncoderConfig::ContentType::kScreen;
>-  statistics_proxy_->OnEncoderReconfigured(config, {}, kPreferredBps);
>+  statistics_proxy_->OnEncoderReconfigured(config, {});
> 
>   // Interval: 3500 bytes * 4 / 2 sec = 7000 bytes / sec  = 56 kbps
>   EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.BitrateSentInKbps"));
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/stats_counter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/video/stats_counter.h
>index 5a52e86bc96..68017391b8f 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/stats_counter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/stats_counter.h
>@@ -15,7 +15,6 @@
> #include <string>
> 
> #include "rtc_base/constructormagic.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/stats_counter_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/stats_counter_unittest.cc
>index 3f00d24eec4..32f8c8e5e1c 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/stats_counter_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/stats_counter_unittest.cc
>@@ -32,8 +32,7 @@ class StatsCounterObserverImpl : public StatsCounterObserver {
> 
> class StatsCounterTest : public ::testing::Test {
>  protected:
>-  StatsCounterTest()
>-      : clock_(1234) {}
>+  StatsCounterTest() : clock_(1234) {}
> 
>   void AddSampleAndAdvance(int sample, int interval_ms, AvgCounter* counter) {
>     counter->Add(sample);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/stream_synchronization.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/stream_synchronization.cc
>index 6b800d11193..e39f122f936 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/stream_synchronization.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/stream_synchronization.cc
>@@ -31,8 +31,7 @@ StreamSynchronization::StreamSynchronization(int video_stream_id,
>     : video_stream_id_(video_stream_id),
>       audio_stream_id_(audio_stream_id),
>       base_target_delay_ms_(0),
>-      avg_diff_ms_(0) {
>-}
>+      avg_diff_ms_(0) {}
> 
> bool StreamSynchronization::ComputeRelativeDelay(
>     const Measurements& audio_measurement,
>@@ -53,7 +52,8 @@ bool StreamSynchronization::ComputeRelativeDelay(
>     return false;
>   }
>   // Positive diff means that video_measurement is behind audio_measurement.
>-  *relative_delay_ms = video_measurement.latest_receive_time_ms -
>+  *relative_delay_ms =
>+      video_measurement.latest_receive_time_ms -
>       audio_measurement.latest_receive_time_ms -
>       (video_last_capture_time_ms - audio_last_capture_time_ms);
>   if (*relative_delay_ms > kMaxDeltaDelayMs ||
>@@ -75,11 +75,11 @@ bool StreamSynchronization::ComputeDelays(int relative_delay_ms,
>                       << " for stream " << audio_stream_id_;
>   // Calculate the difference between the lowest possible video delay and
>   // the current audio delay.
>-  int current_diff_ms = current_video_delay_ms - current_audio_delay_ms +
>-      relative_delay_ms;
>+  int current_diff_ms =
>+      current_video_delay_ms - current_audio_delay_ms + relative_delay_ms;
> 
>-  avg_diff_ms_ = ((kFilterLength - 1) * avg_diff_ms_ +
>-      current_diff_ms) / kFilterLength;
>+  avg_diff_ms_ =
>+      ((kFilterLength - 1) * avg_diff_ms_ + current_diff_ms) / kFilterLength;
>   if (abs(avg_diff_ms_) < kMinDeltaMs) {
>     // Don't adjust if the diff is within our margin.
>     return false;
>@@ -124,8 +124,8 @@ bool StreamSynchronization::ComputeDelays(int relative_delay_ms,
>   }
> 
>   // Make sure that video is never below our target.
>-  channel_delay_.extra_video_delay_ms = std::max(
>-      channel_delay_.extra_video_delay_ms, base_target_delay_ms_);
>+  channel_delay_.extra_video_delay_ms =
>+      std::max(channel_delay_.extra_video_delay_ms, base_target_delay_ms_);
> 
>   int new_video_delay_ms;
>   if (channel_delay_.extra_video_delay_ms > base_target_delay_ms_) {
>@@ -137,8 +137,8 @@ bool StreamSynchronization::ComputeDelays(int relative_delay_ms,
>   }
> 
>   // Make sure that we don't go below the extra video delay.
>-  new_video_delay_ms = std::max(
>-      new_video_delay_ms, channel_delay_.extra_video_delay_ms);
>+  new_video_delay_ms =
>+      std::max(new_video_delay_ms, channel_delay_.extra_video_delay_ms);
> 
>   // Verify we don't go above the maximum allowed video delay.
>   new_video_delay_ms =
>@@ -154,8 +154,8 @@ bool StreamSynchronization::ComputeDelays(int relative_delay_ms,
>   }
> 
>   // Make sure that we don't go below the extra audio delay.
>-  new_audio_delay_ms = std::max(
>-      new_audio_delay_ms, channel_delay_.extra_audio_delay_ms);
>+  new_audio_delay_ms =
>+      std::max(new_audio_delay_ms, channel_delay_.extra_audio_delay_ms);
> 
>   // Verify we don't go above the maximum allowed audio delay.
>   new_audio_delay_ms =
>@@ -181,13 +181,11 @@ void StreamSynchronization::SetTargetBufferingDelay(int target_delay_ms) {
>   // Initial extra delay for audio (accounting for existing extra delay).
>   channel_delay_.extra_audio_delay_ms +=
>       target_delay_ms - base_target_delay_ms_;
>-  channel_delay_.last_audio_delay_ms +=
>-      target_delay_ms - base_target_delay_ms_;
>+  channel_delay_.last_audio_delay_ms += target_delay_ms - base_target_delay_ms_;
> 
>   // The video delay is compared to the last value (and how much we can update
>   // is limited by that as well).
>-  channel_delay_.last_video_delay_ms +=
>-      target_delay_ms - base_target_delay_ms_;
>+  channel_delay_.last_video_delay_ms += target_delay_ms - base_target_delay_ms_;
> 
>   channel_delay_.extra_video_delay_ms +=
>       target_delay_ms - base_target_delay_ms_;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/stream_synchronization.h b/Source/ThirdParty/libwebrtc/Source/webrtc/video/stream_synchronization.h
>index 52b8bde21dd..78fcf7286f5 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/stream_synchronization.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/stream_synchronization.h
>@@ -14,7 +14,6 @@
> #include <list>
> 
> #include "system_wrappers/include/rtp_to_ntp_estimator.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/stream_synchronization_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/stream_synchronization_unittest.cc
>index f9ae37d0bb6..d3e8e1088d4 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/stream_synchronization_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/stream_synchronization_unittest.cc
>@@ -31,8 +31,7 @@ static const int kSmoothingFilter = 4 * 2;
> class Time {
>  public:
>   explicit Time(int64_t offset)
>-      : kNtpJan1970(2208988800UL),
>-        time_now_ms_(offset) {}
>+      : kNtpJan1970(2208988800UL), time_now_ms_(offset) {}
> 
>   NtpTime GetNowNtp() const {
>     uint32_t ntp_secs = time_now_ms_ / 1000 + kNtpJan1970;
>@@ -46,13 +45,9 @@ class Time {
>     return frequency * time_now_ms_ / 1000 + offset;
>   }
> 
>-  void IncreaseTimeMs(int64_t inc) {
>-    time_now_ms_ += inc;
>-  }
>+  void IncreaseTimeMs(int64_t inc) { time_now_ms_ += inc; }
> 
>-  int64_t time_now_ms() const {
>-    return time_now_ms_;
>-  }
>+  int64_t time_now_ms() const { return time_now_ms_; }
> 
>  private:
>   // January 1970, in NTP seconds.
>@@ -87,11 +82,11 @@ class StreamSynchronizationTest : public ::testing::Test {
>                       int current_audio_delay_ms,
>                       int* extra_audio_delay_ms,
>                       int* total_video_delay_ms) {
>-    int audio_frequency = static_cast<int>(kDefaultAudioFrequency *
>-                                           audio_clock_drift_ + 0.5);
>+    int audio_frequency =
>+        static_cast<int>(kDefaultAudioFrequency * audio_clock_drift_ + 0.5);
>     int audio_offset = 0;
>-    int video_frequency = static_cast<int>(kDefaultVideoFrequency *
>-                                           video_clock_drift_ + 0.5);
>+    int video_frequency =
>+        static_cast<int>(kDefaultVideoFrequency * video_clock_drift_ + 0.5);
>     bool new_sr;
>     int video_offset = 0;
>     StreamSynchronization::Measurements audio;
>@@ -147,10 +142,8 @@ class StreamSynchronizationTest : public ::testing::Test {
>     StreamSynchronization::ComputeRelativeDelay(audio, video,
>                                                 &relative_delay_ms);
>     EXPECT_EQ(video_delay_ms - audio_delay_ms, relative_delay_ms);
>-    return sync_->ComputeDelays(relative_delay_ms,
>-                                current_audio_delay_ms,
>-                                extra_audio_delay_ms,
>-                                total_video_delay_ms);
>+    return sync_->ComputeDelays(relative_delay_ms, current_audio_delay_ms,
>+                                extra_audio_delay_ms, total_video_delay_ms);
>   }
> 
>   // Simulate audio playback 300 ms after capture and video rendering 100 ms
>@@ -170,38 +163,32 @@ class StreamSynchronizationTest : public ::testing::Test {
>     const int kNeteqDelayIncrease = 50;
>     const int kNeteqDelayDecrease = 10;
> 
>-    EXPECT_TRUE(DelayedStreams(audio_delay_ms,
>-                               video_delay_ms,
>-                               current_audio_delay_ms,
>-                               &extra_audio_delay_ms,
>+    EXPECT_TRUE(DelayedStreams(audio_delay_ms, video_delay_ms,
>+                               current_audio_delay_ms, &extra_audio_delay_ms,
>                                &total_video_delay_ms));
>     EXPECT_EQ(base_target_delay + filtered_move, total_video_delay_ms);
>     EXPECT_EQ(base_target_delay, extra_audio_delay_ms);
>     current_audio_delay_ms = extra_audio_delay_ms;
> 
>     send_time_->IncreaseTimeMs(1000);
>-    receive_time_->IncreaseTimeMs(1000 - std::max(audio_delay_ms,
>-                                                  video_delay_ms));
>+    receive_time_->IncreaseTimeMs(1000 -
>+                                  std::max(audio_delay_ms, video_delay_ms));
>     // Simulate base_target_delay minimum delay in the VCM.
>     total_video_delay_ms = base_target_delay;
>-    EXPECT_TRUE(DelayedStreams(audio_delay_ms,
>-                               video_delay_ms,
>-                               current_audio_delay_ms,
>-                               &extra_audio_delay_ms,
>+    EXPECT_TRUE(DelayedStreams(audio_delay_ms, video_delay_ms,
>+                               current_audio_delay_ms, &extra_audio_delay_ms,
>                                &total_video_delay_ms));
>     EXPECT_EQ(base_target_delay + 2 * filtered_move, total_video_delay_ms);
>     EXPECT_EQ(base_target_delay, extra_audio_delay_ms);
>     current_audio_delay_ms = extra_audio_delay_ms;
> 
>     send_time_->IncreaseTimeMs(1000);
>-    receive_time_->IncreaseTimeMs(1000 - std::max(audio_delay_ms,
>-                                                  video_delay_ms));
>+    receive_time_->IncreaseTimeMs(1000 -
>+                                  std::max(audio_delay_ms, video_delay_ms));
>     // Simulate base_target_delay minimum delay in the VCM.
>     total_video_delay_ms = base_target_delay;
>-    EXPECT_TRUE(DelayedStreams(audio_delay_ms,
>-                               video_delay_ms,
>-                               current_audio_delay_ms,
>-                               &extra_audio_delay_ms,
>+    EXPECT_TRUE(DelayedStreams(audio_delay_ms, video_delay_ms,
>+                               current_audio_delay_ms, &extra_audio_delay_ms,
>                                &total_video_delay_ms));
>     EXPECT_EQ(base_target_delay + 3 * filtered_move, total_video_delay_ms);
>     EXPECT_EQ(base_target_delay, extra_audio_delay_ms);
>@@ -209,37 +196,33 @@ class StreamSynchronizationTest : public ::testing::Test {
>     // Simulate that NetEQ introduces some audio delay.
>     current_audio_delay_ms = base_target_delay + kNeteqDelayIncrease;
>     send_time_->IncreaseTimeMs(1000);
>-    receive_time_->IncreaseTimeMs(1000 - std::max(audio_delay_ms,
>-                                                  video_delay_ms));
>+    receive_time_->IncreaseTimeMs(1000 -
>+                                  std::max(audio_delay_ms, video_delay_ms));
>     // Simulate base_target_delay minimum delay in the VCM.
>     total_video_delay_ms = base_target_delay;
>-    EXPECT_TRUE(DelayedStreams(audio_delay_ms,
>-                               video_delay_ms,
>-                               current_audio_delay_ms,
>-                               &extra_audio_delay_ms,
>+    EXPECT_TRUE(DelayedStreams(audio_delay_ms, video_delay_ms,
>+                               current_audio_delay_ms, &extra_audio_delay_ms,
>                                &total_video_delay_ms));
>     filtered_move = 3 * filtered_move +
>-        (kNeteqDelayIncrease + audio_delay_ms - video_delay_ms) /
>-        kSmoothingFilter;
>+                    (kNeteqDelayIncrease + audio_delay_ms - video_delay_ms) /
>+                        kSmoothingFilter;
>     EXPECT_EQ(base_target_delay + filtered_move, total_video_delay_ms);
>     EXPECT_EQ(base_target_delay, extra_audio_delay_ms);
> 
>     // Simulate that NetEQ reduces its delay.
>     current_audio_delay_ms = base_target_delay + kNeteqDelayDecrease;
>     send_time_->IncreaseTimeMs(1000);
>-    receive_time_->IncreaseTimeMs(1000 - std::max(audio_delay_ms,
>-                                                  video_delay_ms));
>+    receive_time_->IncreaseTimeMs(1000 -
>+                                  std::max(audio_delay_ms, video_delay_ms));
>     // Simulate base_target_delay minimum delay in the VCM.
>     total_video_delay_ms = base_target_delay;
>-    EXPECT_TRUE(DelayedStreams(audio_delay_ms,
>-                               video_delay_ms,
>-                               current_audio_delay_ms,
>-                               &extra_audio_delay_ms,
>+    EXPECT_TRUE(DelayedStreams(audio_delay_ms, video_delay_ms,
>+                               current_audio_delay_ms, &extra_audio_delay_ms,
>                                &total_video_delay_ms));
> 
>     filtered_move = filtered_move +
>-        (kNeteqDelayDecrease + audio_delay_ms - video_delay_ms) /
>-        kSmoothingFilter;
>+                    (kNeteqDelayDecrease + audio_delay_ms - video_delay_ms) /
>+                        kSmoothingFilter;
> 
>     EXPECT_EQ(base_target_delay + filtered_move, total_video_delay_ms);
>     EXPECT_EQ(base_target_delay, extra_audio_delay_ms);
>@@ -252,10 +235,8 @@ class StreamSynchronizationTest : public ::testing::Test {
>     int extra_audio_delay_ms = 0;
>     int total_video_delay_ms = base_target_delay;
> 
>-    EXPECT_TRUE(DelayedStreams(audio_delay_ms,
>-                               video_delay_ms,
>-                               current_audio_delay_ms,
>-                               &extra_audio_delay_ms,
>+    EXPECT_TRUE(DelayedStreams(audio_delay_ms, video_delay_ms,
>+                               current_audio_delay_ms, &extra_audio_delay_ms,
>                                &total_video_delay_ms));
>     EXPECT_EQ(base_target_delay, total_video_delay_ms);
>     // The audio delay is not allowed to change more than this in 1 second.
>@@ -265,77 +246,73 @@ class StreamSynchronizationTest : public ::testing::Test {
> 
>     send_time_->IncreaseTimeMs(1000);
>     receive_time_->IncreaseTimeMs(800);
>-    EXPECT_TRUE(DelayedStreams(audio_delay_ms,
>-                               video_delay_ms,
>-                               current_audio_delay_ms,
>-                               &extra_audio_delay_ms,
>+    EXPECT_TRUE(DelayedStreams(audio_delay_ms, video_delay_ms,
>+                               current_audio_delay_ms, &extra_audio_delay_ms,
>                                &total_video_delay_ms));
>     EXPECT_EQ(base_target_delay, total_video_delay_ms);
>     // The audio delay is not allowed to change more than the half of the
>     // required change in delay.
>-    EXPECT_EQ(current_extra_delay_ms + MaxAudioDelayIncrease(
>-        current_audio_delay_ms,
>-        base_target_delay + video_delay_ms - audio_delay_ms),
>-        extra_audio_delay_ms);
>+    EXPECT_EQ(current_extra_delay_ms +
>+                  MaxAudioDelayIncrease(
>+                      current_audio_delay_ms,
>+                      base_target_delay + video_delay_ms - audio_delay_ms),
>+              extra_audio_delay_ms);
>     current_audio_delay_ms = extra_audio_delay_ms;
>     current_extra_delay_ms = extra_audio_delay_ms;
> 
>     send_time_->IncreaseTimeMs(1000);
>     receive_time_->IncreaseTimeMs(800);
>-    EXPECT_TRUE(DelayedStreams(audio_delay_ms,
>-                               video_delay_ms,
>-                               current_audio_delay_ms,
>-                               &extra_audio_delay_ms,
>+    EXPECT_TRUE(DelayedStreams(audio_delay_ms, video_delay_ms,
>+                               current_audio_delay_ms, &extra_audio_delay_ms,
>                                &total_video_delay_ms));
>     EXPECT_EQ(base_target_delay, total_video_delay_ms);
>     // The audio delay is not allowed to change more than the half of the
>     // required change in delay.
>-    EXPECT_EQ(current_extra_delay_ms + MaxAudioDelayIncrease(
>-        current_audio_delay_ms,
>-        base_target_delay + video_delay_ms - audio_delay_ms),
>-        extra_audio_delay_ms);
>+    EXPECT_EQ(current_extra_delay_ms +
>+                  MaxAudioDelayIncrease(
>+                      current_audio_delay_ms,
>+                      base_target_delay + video_delay_ms - audio_delay_ms),
>+              extra_audio_delay_ms);
>     current_extra_delay_ms = extra_audio_delay_ms;
> 
>     // Simulate that NetEQ for some reason reduced the delay.
>     current_audio_delay_ms = base_target_delay + 10;
>     send_time_->IncreaseTimeMs(1000);
>     receive_time_->IncreaseTimeMs(800);
>-    EXPECT_TRUE(DelayedStreams(audio_delay_ms,
>-                               video_delay_ms,
>-                               current_audio_delay_ms,
>-                               &extra_audio_delay_ms,
>+    EXPECT_TRUE(DelayedStreams(audio_delay_ms, video_delay_ms,
>+                               current_audio_delay_ms, &extra_audio_delay_ms,
>                                &total_video_delay_ms));
>     EXPECT_EQ(base_target_delay, total_video_delay_ms);
>     // Since we only can ask NetEQ for a certain amount of extra delay, and
>     // we only measure the total NetEQ delay, we will ask for additional delay
>     // here to try to stay in sync.
>-    EXPECT_EQ(current_extra_delay_ms + MaxAudioDelayIncrease(
>-        current_audio_delay_ms,
>-        base_target_delay + video_delay_ms - audio_delay_ms),
>-        extra_audio_delay_ms);
>+    EXPECT_EQ(current_extra_delay_ms +
>+                  MaxAudioDelayIncrease(
>+                      current_audio_delay_ms,
>+                      base_target_delay + video_delay_ms - audio_delay_ms),
>+              extra_audio_delay_ms);
>     current_extra_delay_ms = extra_audio_delay_ms;
> 
>     // Simulate that NetEQ for some reason significantly increased the delay.
>     current_audio_delay_ms = base_target_delay + 350;
>     send_time_->IncreaseTimeMs(1000);
>     receive_time_->IncreaseTimeMs(800);
>-    EXPECT_TRUE(DelayedStreams(audio_delay_ms,
>-                               video_delay_ms,
>-                               current_audio_delay_ms,
>-                               &extra_audio_delay_ms,
>+    EXPECT_TRUE(DelayedStreams(audio_delay_ms, video_delay_ms,
>+                               current_audio_delay_ms, &extra_audio_delay_ms,
>                                &total_video_delay_ms));
>     EXPECT_EQ(base_target_delay, total_video_delay_ms);
>     // The audio delay is not allowed to change more than the half of the
>     // required change in delay.
>-    EXPECT_EQ(current_extra_delay_ms + MaxAudioDelayIncrease(
>-        current_audio_delay_ms,
>-        base_target_delay + video_delay_ms - audio_delay_ms),
>-        extra_audio_delay_ms);
>+    EXPECT_EQ(current_extra_delay_ms +
>+                  MaxAudioDelayIncrease(
>+                      current_audio_delay_ms,
>+                      base_target_delay + video_delay_ms - audio_delay_ms),
>+              extra_audio_delay_ms);
>   }
> 
>   int MaxAudioDelayIncrease(int current_audio_delay_ms, int delay_ms) {
>     return std::min((delay_ms - current_audio_delay_ms) / kSmoothingFilter,
>-                     static_cast<int>(kMaxAudioDiffMs));
>+                    static_cast<int>(kMaxAudioDiffMs));
>   }
> 
>   int MaxAudioDelayDecrease(int current_audio_delay_ms, int delay_ms) {
>@@ -347,7 +324,7 @@ class StreamSynchronizationTest : public ::testing::Test {
>   enum { kReceiveTimeOffsetMs = 43210 };
> 
>   StreamSynchronization* sync_;
>-  Time* send_time_;  // The simulated clock at the sender.
>+  Time* send_time_;     // The simulated clock at the sender.
>   Time* receive_time_;  // The simulated clock at the receiver.
>   double audio_clock_drift_;
>   double video_clock_drift_;
>@@ -418,7 +395,7 @@ TEST_F(StreamSynchronizationTest, AudioDelay) {
>   // The audio delay is not allowed to change more than the half of the required
>   // change in delay.
>   EXPECT_EQ(current_extra_delay_ms +
>-            MaxAudioDelayIncrease(current_audio_delay_ms, delay_ms),
>+                MaxAudioDelayIncrease(current_audio_delay_ms, delay_ms),
>             extra_audio_delay_ms);
>   current_audio_delay_ms = extra_audio_delay_ms;
>   current_extra_delay_ms = extra_audio_delay_ms;
>@@ -431,7 +408,7 @@ TEST_F(StreamSynchronizationTest, AudioDelay) {
>   // The audio delay is not allowed to change more than the half of the required
>   // change in delay.
>   EXPECT_EQ(current_extra_delay_ms +
>-            MaxAudioDelayIncrease(current_audio_delay_ms, delay_ms),
>+                MaxAudioDelayIncrease(current_audio_delay_ms, delay_ms),
>             extra_audio_delay_ms);
>   current_extra_delay_ms = extra_audio_delay_ms;
> 
>@@ -446,7 +423,7 @@ TEST_F(StreamSynchronizationTest, AudioDelay) {
>   // we only measure the total NetEQ delay, we will ask for additional delay
>   // here to try to
>   EXPECT_EQ(current_extra_delay_ms +
>-            MaxAudioDelayIncrease(current_audio_delay_ms, delay_ms),
>+                MaxAudioDelayIncrease(current_audio_delay_ms, delay_ms),
>             extra_audio_delay_ms);
>   current_extra_delay_ms = extra_audio_delay_ms;
> 
>@@ -460,7 +437,7 @@ TEST_F(StreamSynchronizationTest, AudioDelay) {
>   // The audio delay is not allowed to change more than the half of the required
>   // change in delay.
>   EXPECT_EQ(current_extra_delay_ms +
>-            MaxAudioDelayDecrease(current_audio_delay_ms, delay_ms),
>+                MaxAudioDelayDecrease(current_audio_delay_ms, delay_ms),
>             extra_audio_delay_ms);
> }
> 
>@@ -500,8 +477,8 @@ TEST_F(StreamSynchronizationTest, BaseDelay) {
>   sync_->SetTargetBufferingDelay(base_target_delay_ms);
>   // We are in sync don't change.
>   EXPECT_FALSE(DelayedStreams(base_target_delay_ms, base_target_delay_ms,
>-                              current_audio_delay_ms,
>-                              &extra_audio_delay_ms, &total_video_delay_ms));
>+                              current_audio_delay_ms, &extra_audio_delay_ms,
>+                              &total_video_delay_ms));
>   // Triggering another call with the same values. Delay should not be modified.
>   base_target_delay_ms = 2000;
>   current_audio_delay_ms = base_target_delay_ms;
>@@ -509,8 +486,8 @@ TEST_F(StreamSynchronizationTest, BaseDelay) {
>   sync_->SetTargetBufferingDelay(base_target_delay_ms);
>   // We are in sync don't change.
>   EXPECT_FALSE(DelayedStreams(base_target_delay_ms, base_target_delay_ms,
>-                              current_audio_delay_ms,
>-                              &extra_audio_delay_ms, &total_video_delay_ms));
>+                              current_audio_delay_ms, &extra_audio_delay_ms,
>+                              &total_video_delay_ms));
>   // Changing delay value - intended to test this module only. In practice it
>   // would take VoE time to adapt.
>   base_target_delay_ms = 5000;
>@@ -519,8 +496,8 @@ TEST_F(StreamSynchronizationTest, BaseDelay) {
>   sync_->SetTargetBufferingDelay(base_target_delay_ms);
>   // We are in sync don't change.
>   EXPECT_FALSE(DelayedStreams(base_target_delay_ms, base_target_delay_ms,
>-                              current_audio_delay_ms,
>-                              &extra_audio_delay_ms, &total_video_delay_ms));
>+                              current_audio_delay_ms, &extra_audio_delay_ms,
>+                              &total_video_delay_ms));
> }
> 
> TEST_F(StreamSynchronizationTest, BothDelayedAudioLaterWithBaseDelay) {
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/sv_loopback.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/sv_loopback.cc
>index 24d3bedf328..b2a1e2e2603 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/sv_loopback.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/sv_loopback.cc
>@@ -11,7 +11,9 @@
> #include <stdio.h>
> 
> #include "rtc_base/flags.h"
>+#include "rtc_base/logging.h"
> #include "rtc_base/stringencode.h"
>+#include "system_wrappers/include/field_trial_default.h"
> #include "test/field_trial.h"
> #include "test/gtest.h"
> #include "test/run_test.h"
>@@ -20,6 +22,17 @@
> namespace webrtc {
> namespace flags {
> 
>+InterLayerPredMode IntToInterLayerPredMode(int inter_layer_pred) {
>+  if (inter_layer_pred == 0) {
>+    return InterLayerPredMode::kOn;
>+  } else if (inter_layer_pred == 1) {
>+    return InterLayerPredMode::kOff;
>+  } else {
>+    RTC_DCHECK_EQ(inter_layer_pred, 2);
>+    return InterLayerPredMode::kOnKeyPic;
>+  }
>+}
>+
> // Flags for video.
> DEFINE_int(vwidth, 640, "Video width.");
> size_t VideoWidth() {
>@@ -79,6 +92,14 @@ int VideoNumSpatialLayers() {
>   return static_cast<int>(FLAG_vnum_spatial_layers);
> }
> 
>+DEFINE_int(vinter_layer_pred,
>+           2,
>+           "Video inter-layer prediction mode. "
>+           "0 - enabled, 1 - disabled, 2 - enabled only for key pictures.");
>+InterLayerPredMode VideoInterLayerPred() {
>+  return IntToInterLayerPredMode(FLAG_vinter_layer_pred);
>+}
>+
> DEFINE_string(
>     vstream0,
>     "",
>@@ -189,11 +210,19 @@ int ScreenshareNumStreams() {
> 
> DEFINE_int(snum_spatial_layers,
>            1,
>-           "Number of screemshare spatial layers to use.");
>+           "Number of screenshare spatial layers to use.");
> int ScreenshareNumSpatialLayers() {
>   return static_cast<int>(FLAG_snum_spatial_layers);
> }
> 
>+DEFINE_int(sinter_layer_pred,
>+           0,
>+           "Screenshare inter-layer prediction mode. "
>+           "0 - enabled, 1 - disabled, 2 - enabled only for key pictures.");
>+InterLayerPredMode ScreenshareInterLayerPred() {
>+  return IntToInterLayerPredMode(FLAG_sinter_layer_pred);
>+}
>+
> DEFINE_string(
>     sstream0,
>     "",
>@@ -453,7 +482,7 @@ void Loopback() {
>   pipe_config.delay_standard_deviation_ms = flags::StdPropagationDelayMs();
>   pipe_config.allow_reordering = flags::FLAG_allow_reordering;
> 
>-  Call::Config::BitrateConfig call_bitrate_config;
>+  BitrateConstraints call_bitrate_config;
>   call_bitrate_config.min_bitrate_bps =
>       (flags::ScreenshareMinBitrateKbps() + flags::VideoMinBitrateKbps()) *
>       1000;
>@@ -480,6 +509,7 @@ void Loopback() {
>       flags::ScreenshareMinTransmitBitrateKbps() * 1000,
>       false,  // ULPFEC disabled.
>       false,  // FlexFEC disabled.
>+      false,  // Automatic scaling disabled
>       ""};
>   params.video[camera_idx] = {flags::FLAG_video,
>                               flags::VideoWidth(),
>@@ -495,12 +525,13 @@ void Loopback() {
>                               0,  // No min transmit bitrate.
>                               flags::FLAG_use_ulpfec,
>                               flags::FLAG_use_flexfec,
>+                              false,
>                               flags::VideoClip(),
>                               flags::GetCaptureDevice()};
>   params.audio = {flags::FLAG_audio, flags::FLAG_audio_video_sync,
>                   flags::FLAG_audio_dtx};
>-  params.logging = {flags::FLAG_logs, flags::FLAG_rtc_event_log_name,
>-                    flags::FLAG_rtp_dump_name, flags::FLAG_encoded_frame_path};
>+  params.logging = {flags::FLAG_rtc_event_log_name, flags::FLAG_rtp_dump_name,
>+                    flags::FLAG_encoded_frame_path};
>   params.analyzer = {"dual_streams",
>                      0.0,
>                      0.0,
>@@ -535,7 +566,7 @@ void Loopback() {
>       &params, screenshare_idx, stream_descriptors,
>       flags::ScreenshareNumStreams(), flags::ScreenshareSelectedStream(),
>       flags::ScreenshareNumSpatialLayers(), flags::ScreenshareSelectedSL(),
>-      SL_descriptors);
>+      flags::ScreenshareInterLayerPred(), SL_descriptors);
> 
>   stream_descriptors.clear();
>   stream_descriptors.push_back(flags::VideoStream0());
>@@ -546,13 +577,13 @@ void Loopback() {
>   VideoQualityTest::FillScalabilitySettings(
>       &params, camera_idx, stream_descriptors, flags::VideoNumStreams(),
>       flags::VideoSelectedStream(), flags::VideoNumSpatialLayers(),
>-      flags::VideoSelectedSL(), SL_descriptors);
>+      flags::VideoSelectedSL(), flags::VideoInterLayerPred(), SL_descriptors);
> 
>-  VideoQualityTest test;
>+  auto fixture = absl::make_unique<VideoQualityTest>(nullptr);
>   if (flags::DurationSecs()) {
>-    test.RunWithAnalyzer(params);
>+    fixture->RunWithAnalyzer(params);
>   } else {
>-    test.RunWithRenderers(params);
>+    fixture->RunWithRenderers(params);
>   }
> }
> }  // namespace webrtc
>@@ -568,10 +599,14 @@ int main(int argc, char* argv[]) {
>     return 0;
>   }
> 
>-  // InitFieldTrialsFromString needs a reference to an std::string instance,
>-  // with a scope that outlives the test.
>-  std::string field_trials = webrtc::flags::FLAG_force_fieldtrials;
>-  webrtc::test::InitFieldTrialsFromString(field_trials);
>+  rtc::LogMessage::SetLogToStderr(webrtc::flags::FLAG_logs);
>+
>+  webrtc::test::ValidateFieldTrialsStringOrDie(
>+      webrtc::flags::FLAG_force_fieldtrials);
>+  // InitFieldTrialsFromString stores the char*, so the char array must outlive
>+  // the application.
>+  webrtc::field_trial::InitFieldTrialsFromString(
>+      webrtc::flags::FLAG_force_fieldtrials);
> 
>   webrtc::test::RunTest(webrtc::Loopback);
>   return 0;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/test/mock_video_stream_encoder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/video/test/mock_video_stream_encoder.h
>new file mode 100644
>index 00000000000..8ce6f41246b
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/test/mock_video_stream_encoder.h
>@@ -0,0 +1,43 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#ifndef VIDEO_TEST_MOCK_VIDEO_STREAM_ENCODER_H_
>+#define VIDEO_TEST_MOCK_VIDEO_STREAM_ENCODER_H_
>+
>+#include "api/video/video_stream_encoder_interface.h"
>+#include "test/gmock.h"
>+
>+namespace webrtc {
>+
>+class MockVideoStreamEncoder : public VideoStreamEncoderInterface {
>+ public:
>+  MOCK_METHOD2(SetSource,
>+               void(rtc::VideoSourceInterface<VideoFrame>*,
>+                    const DegradationPreference&));
>+  MOCK_METHOD2(SetSink, void(EncoderSink*, bool));
>+  MOCK_METHOD1(SetStartBitrate, void(int));
>+  MOCK_METHOD0(SendKeyFrame, void());
>+  MOCK_METHOD3(OnBitrateUpdated, void(uint32_t, uint8_t, int64_t));
>+  MOCK_METHOD1(OnFrame, void(const VideoFrame&));
>+  MOCK_METHOD1(SetBitrateAllocationObserver,
>+               void(VideoBitrateAllocationObserver*));
>+  MOCK_METHOD0(Stop, void());
>+
>+  MOCK_METHOD2(MockedConfigureEncoder, void(const VideoEncoderConfig&, size_t));
>+  // gtest generates implicit copy which is not allowed on VideoEncoderConfig,
>+  // so we can't mock ConfigureEncoder directly.
>+  void ConfigureEncoder(VideoEncoderConfig config,
>+                        size_t max_data_payload_length) {
>+    MockedConfigureEncoder(config, max_data_payload_length);
>+  }
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // VIDEO_TEST_MOCK_VIDEO_STREAM_ENCODER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/transport_adapter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/video/transport_adapter.h
>index 0168cc5b865..dd9964f3630 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/transport_adapter.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/transport_adapter.h
>@@ -31,7 +31,7 @@ class TransportAdapter : public Transport {
>   void Disable();
> 
>  private:
>-  Transport *transport_;
>+  Transport* transport_;
>   std::atomic<bool> enabled_;
> };
> }  // namespace internal
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_analyzer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_analyzer.cc
>new file mode 100644
>index 00000000000..04b5c766c87
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_analyzer.cc
>@@ -0,0 +1,897 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#include "video/video_analyzer.h"
>+
>+#include <algorithm>
>+#include <utility>
>+
>+#include "modules/rtp_rtcp/source/rtp_format.h"
>+#include "modules/rtp_rtcp/source/rtp_utility.h"
>+#include "rtc_base/cpu_time.h"
>+#include "rtc_base/flags.h"
>+#include "rtc_base/format_macros.h"
>+#include "rtc_base/memory_usage.h"
>+#include "rtc_base/pathutils.h"
>+#include "system_wrappers/include/cpu_info.h"
>+#include "test/call_test.h"
>+#include "test/testsupport/frame_writer.h"
>+#include "test/testsupport/perf_test.h"
>+#include "test/testsupport/test_artifacts.h"
>+
>+DEFINE_bool(save_worst_frame,
>+            false,
>+            "Enable saving a frame with the lowest PSNR to a jpeg file in the "
>+            "test_artifacts_dir");
>+
>+namespace webrtc {
>+namespace {
>+constexpr int kSendStatsPollingIntervalMs = 1000;
>+constexpr size_t kMaxComparisons = 10;
>+
>+bool IsFlexfec(int payload_type) {
>+  return payload_type == test::CallTest::kFlexfecPayloadType;
>+}
>+}  // namespace
>+
>+VideoAnalyzer::VideoAnalyzer(test::LayerFilteringTransport* transport,
>+                             const std::string& test_label,
>+                             double avg_psnr_threshold,
>+                             double avg_ssim_threshold,
>+                             int duration_frames,
>+                             FILE* graph_data_output_file,
>+                             const std::string& graph_title,
>+                             uint32_t ssrc_to_analyze,
>+                             uint32_t rtx_ssrc_to_analyze,
>+                             size_t selected_stream,
>+                             int selected_sl,
>+                             int selected_tl,
>+                             bool is_quick_test_enabled,
>+                             Clock* clock,
>+                             std::string rtp_dump_name)
>+    : transport_(transport),
>+      receiver_(nullptr),
>+      call_(nullptr),
>+      send_stream_(nullptr),
>+      receive_stream_(nullptr),
>+      audio_receive_stream_(nullptr),
>+      captured_frame_forwarder_(this, clock),
>+      test_label_(test_label),
>+      graph_data_output_file_(graph_data_output_file),
>+      graph_title_(graph_title),
>+      ssrc_to_analyze_(ssrc_to_analyze),
>+      rtx_ssrc_to_analyze_(rtx_ssrc_to_analyze),
>+      selected_stream_(selected_stream),
>+      selected_sl_(selected_sl),
>+      selected_tl_(selected_tl),
>+      pre_encode_proxy_(this),
>+      last_fec_bytes_(0),
>+      frames_to_process_(duration_frames),
>+      frames_recorded_(0),
>+      frames_processed_(0),
>+      dropped_frames_(0),
>+      dropped_frames_before_first_encode_(0),
>+      dropped_frames_before_rendering_(0),
>+      last_render_time_(0),
>+      last_render_delta_ms_(0),
>+      last_unfreeze_time_ms_(0),
>+      rtp_timestamp_delta_(0),
>+      total_media_bytes_(0),
>+      first_sending_time_(0),
>+      last_sending_time_(0),
>+      cpu_time_(0),
>+      wallclock_time_(0),
>+      avg_psnr_threshold_(avg_psnr_threshold),
>+      avg_ssim_threshold_(avg_ssim_threshold),
>+      is_quick_test_enabled_(is_quick_test_enabled),
>+      stats_polling_thread_(&PollStatsThread, this, "StatsPoller"),
>+      comparison_available_event_(false, false),
>+      done_(true, false),
>+      clock_(clock),
>+      start_ms_(clock->TimeInMilliseconds()) {
>+  // Create thread pool for CPU-expensive PSNR/SSIM calculations.
>+
>+  // Try to use about as many threads as cores, but leave kMinCoresLeft alone,
>+  // so that we don't accidentally starve "real" worker threads (codec etc).
>+  // Also, don't allocate more than kMaxComparisonThreads, even if there are
>+  // spare cores.
>+
>+  uint32_t num_cores = CpuInfo::DetectNumberOfCores();
>+  RTC_DCHECK_GE(num_cores, 1);
>+  static const uint32_t kMinCoresLeft = 4;
>+  static const uint32_t kMaxComparisonThreads = 8;
>+
>+  if (num_cores <= kMinCoresLeft) {
>+    num_cores = 1;
>+  } else {
>+    num_cores -= kMinCoresLeft;
>+    num_cores = std::min(num_cores, kMaxComparisonThreads);
>+  }
>+
>+  for (uint32_t i = 0; i < num_cores; ++i) {
>+    rtc::PlatformThread* thread =
>+        new rtc::PlatformThread(&FrameComparisonThread, this, "Analyzer");
>+    thread->Start();
>+    comparison_thread_pool_.push_back(thread);
>+  }
>+
>+  if (!rtp_dump_name.empty()) {
>+    fprintf(stdout, "Writing rtp dump to %s\n", rtp_dump_name.c_str());
>+    rtp_file_writer_.reset(test::RtpFileWriter::Create(
>+        test::RtpFileWriter::kRtpDump, rtp_dump_name));
>+  }
>+}
>+
>+VideoAnalyzer::~VideoAnalyzer() {
>+  for (rtc::PlatformThread* thread : comparison_thread_pool_) {
>+    thread->Stop();
>+    delete thread;
>+  }
>+}
>+
>+void VideoAnalyzer::SetReceiver(PacketReceiver* receiver) {
>+  receiver_ = receiver;
>+}
>+
>+void VideoAnalyzer::SetSource(test::VideoCapturer* video_capturer,
>+                              bool respect_sink_wants) {
>+  if (respect_sink_wants)
>+    captured_frame_forwarder_.SetSource(video_capturer);
>+  rtc::VideoSinkWants wants;
>+  video_capturer->AddOrUpdateSink(InputInterface(), wants);
>+}
>+
>+void VideoAnalyzer::SetCall(Call* call) {
>+  rtc::CritScope lock(&crit_);
>+  RTC_DCHECK(!call_);
>+  call_ = call;
>+}
>+
>+void VideoAnalyzer::SetSendStream(VideoSendStream* stream) {
>+  rtc::CritScope lock(&crit_);
>+  RTC_DCHECK(!send_stream_);
>+  send_stream_ = stream;
>+}
>+
>+void VideoAnalyzer::SetReceiveStream(VideoReceiveStream* stream) {
>+  rtc::CritScope lock(&crit_);
>+  RTC_DCHECK(!receive_stream_);
>+  receive_stream_ = stream;
>+}
>+
>+void VideoAnalyzer::SetAudioReceiveStream(AudioReceiveStream* recv_stream) {
>+  rtc::CritScope lock(&crit_);
>+  RTC_CHECK(!audio_receive_stream_);
>+  audio_receive_stream_ = recv_stream;
>+}
>+
>+rtc::VideoSinkInterface<VideoFrame>* VideoAnalyzer::InputInterface() {
>+  return &captured_frame_forwarder_;
>+}
>+
>+rtc::VideoSourceInterface<VideoFrame>* VideoAnalyzer::OutputInterface() {
>+  return &captured_frame_forwarder_;
>+}
>+
>+PacketReceiver::DeliveryStatus VideoAnalyzer::DeliverPacket(
>+    MediaType media_type,
>+    rtc::CopyOnWriteBuffer packet,
>+    int64_t packet_time_us) {
>+  // Ignore timestamps of RTCP packets. They're not synchronized with
>+  // RTP packet timestamps and so they would confuse wrap_handler_.
>+  if (RtpHeaderParser::IsRtcp(packet.cdata(), packet.size())) {
>+    return receiver_->DeliverPacket(media_type, std::move(packet),
>+                                    packet_time_us);
>+  }
>+
>+  if (rtp_file_writer_) {
>+    test::RtpPacket p;
>+    memcpy(p.data, packet.cdata(), packet.size());
>+    p.length = packet.size();
>+    p.original_length = packet.size();
>+    p.time_ms = clock_->TimeInMilliseconds() - start_ms_;
>+    rtp_file_writer_->WritePacket(&p);
>+  }
>+
>+  RtpUtility::RtpHeaderParser parser(packet.cdata(), packet.size());
>+  RTPHeader header;
>+  parser.Parse(&header);
>+  if (!IsFlexfec(header.payloadType) && (header.ssrc == ssrc_to_analyze_ ||
>+                                         header.ssrc == rtx_ssrc_to_analyze_)) {
>+    // Ignore FlexFEC timestamps, to avoid collisions with media timestamps.
>+    // (FlexFEC and media are sent on different SSRCs, which have different
>+    // timestamps spaces.)
>+    // Also ignore packets from wrong SSRC, but include retransmits.
>+    rtc::CritScope lock(&crit_);
>+    int64_t timestamp =
>+        wrap_handler_.Unwrap(header.timestamp - rtp_timestamp_delta_);
>+    recv_times_[timestamp] =
>+        Clock::GetRealTimeClock()->CurrentNtpInMilliseconds();
>+  }
>+
>+  return receiver_->DeliverPacket(media_type, std::move(packet),
>+                                  packet_time_us);
>+}
>+
>+void VideoAnalyzer::PreEncodeOnFrame(const VideoFrame& video_frame) {
>+  rtc::CritScope lock(&crit_);
>+  if (!first_encoded_timestamp_) {
>+    while (frames_.front().timestamp() != video_frame.timestamp()) {
>+      ++dropped_frames_before_first_encode_;
>+      frames_.pop_front();
>+      RTC_CHECK(!frames_.empty());
>+    }
>+    first_encoded_timestamp_ = video_frame.timestamp();
>+  }
>+}
>+
>+void VideoAnalyzer::EncodedFrameCallback(const EncodedFrame& encoded_frame) {
>+  rtc::CritScope lock(&crit_);
>+  if (!first_sent_timestamp_ && encoded_frame.stream_id_ == selected_stream_) {
>+    first_sent_timestamp_ = encoded_frame.timestamp_;
>+  }
>+}
>+
>+bool VideoAnalyzer::SendRtp(const uint8_t* packet,
>+                            size_t length,
>+                            const PacketOptions& options) {
>+  RtpUtility::RtpHeaderParser parser(packet, length);
>+  RTPHeader header;
>+  parser.Parse(&header);
>+
>+  int64_t current_time = Clock::GetRealTimeClock()->CurrentNtpInMilliseconds();
>+
>+  bool result = transport_->SendRtp(packet, length, options);
>+  {
>+    rtc::CritScope lock(&crit_);
>+    if (rtp_timestamp_delta_ == 0 && header.ssrc == ssrc_to_analyze_) {
>+      RTC_CHECK(static_cast<bool>(first_sent_timestamp_));
>+      rtp_timestamp_delta_ = header.timestamp - *first_sent_timestamp_;
>+    }
>+
>+    if (!IsFlexfec(header.payloadType) && header.ssrc == ssrc_to_analyze_) {
>+      // Ignore FlexFEC timestamps, to avoid collisions with media timestamps.
>+      // (FlexFEC and media are sent on different SSRCs, which have different
>+      // timestamps spaces.)
>+      // Also ignore packets from wrong SSRC and retransmits.
>+      int64_t timestamp =
>+          wrap_handler_.Unwrap(header.timestamp - rtp_timestamp_delta_);
>+      send_times_[timestamp] = current_time;
>+
>+      if (IsInSelectedSpatialAndTemporalLayer(packet, length, header)) {
>+        encoded_frame_sizes_[timestamp] +=
>+            length - (header.headerLength + header.paddingLength);
>+        total_media_bytes_ +=
>+            length - (header.headerLength + header.paddingLength);
>+      }
>+      if (first_sending_time_ == 0)
>+        first_sending_time_ = current_time;
>+      last_sending_time_ = current_time;
>+    }
>+  }
>+  return result;
>+}
>+
>+bool VideoAnalyzer::SendRtcp(const uint8_t* packet, size_t length) {
>+  return transport_->SendRtcp(packet, length);
>+}
>+
>+void VideoAnalyzer::OnFrame(const VideoFrame& video_frame) {
>+  int64_t render_time_ms =
>+      Clock::GetRealTimeClock()->CurrentNtpInMilliseconds();
>+
>+  rtc::CritScope lock(&crit_);
>+
>+  StartExcludingCpuThreadTime();
>+
>+  int64_t send_timestamp =
>+      wrap_handler_.Unwrap(video_frame.timestamp() - rtp_timestamp_delta_);
>+
>+  while (wrap_handler_.Unwrap(frames_.front().timestamp()) < send_timestamp) {
>+    if (!last_rendered_frame_) {
>+      // No previous frame rendered, this one was dropped after sending but
>+      // before rendering.
>+      ++dropped_frames_before_rendering_;
>+    } else {
>+      AddFrameComparison(frames_.front(), *last_rendered_frame_, true,
>+                         render_time_ms);
>+    }
>+    frames_.pop_front();
>+    RTC_DCHECK(!frames_.empty());
>+  }
>+
>+  VideoFrame reference_frame = frames_.front();
>+  frames_.pop_front();
>+  int64_t reference_timestamp =
>+      wrap_handler_.Unwrap(reference_frame.timestamp());
>+  if (send_timestamp == reference_timestamp - 1) {
>+    // TODO(ivica): Make this work for > 2 streams.
>+    // Look at RTPSender::BuildRTPHeader.
>+    ++send_timestamp;
>+  }
>+  ASSERT_EQ(reference_timestamp, send_timestamp);
>+
>+  AddFrameComparison(reference_frame, video_frame, false, render_time_ms);
>+
>+  last_rendered_frame_ = video_frame;
>+
>+  StopExcludingCpuThreadTime();
>+}
>+
>+void VideoAnalyzer::Wait() {
>+  // Frame comparisons can be very expensive. Wait for test to be done, but
>+  // at time-out check if frames_processed is going up. If so, give it more
>+  // time, otherwise fail. Hopefully this will reduce test flakiness.
>+
>+  stats_polling_thread_.Start();
>+
>+  int last_frames_processed = -1;
>+  int iteration = 0;
>+  while (!done_.Wait(test::CallTest::kDefaultTimeoutMs)) {
>+    int frames_processed;
>+    {
>+      rtc::CritScope crit(&comparison_lock_);
>+      frames_processed = frames_processed_;
>+    }
>+
>+    // Print some output so test infrastructure won't think we've crashed.
>+    const char* kKeepAliveMessages[3] = {
>+        "Uh, I'm-I'm not quite dead, sir.",
>+        "Uh, I-I think uh, I could pull through, sir.",
>+        "Actually, I think I'm all right to come with you--"};
>+    printf("- %s\n", kKeepAliveMessages[iteration++ % 3]);
>+
>+    if (last_frames_processed == -1) {
>+      last_frames_processed = frames_processed;
>+      continue;
>+    }
>+    if (frames_processed == last_frames_processed) {
>+      EXPECT_GT(frames_processed, last_frames_processed)
>+          << "Analyzer stalled while waiting for test to finish.";
>+      done_.Set();
>+      break;
>+    }
>+    last_frames_processed = frames_processed;
>+  }
>+
>+  if (iteration > 0)
>+    printf("- Farewell, sweet Concorde!\n");
>+
>+  stats_polling_thread_.Stop();
>+}
>+
>+rtc::VideoSinkInterface<VideoFrame>* VideoAnalyzer::pre_encode_proxy() {
>+  return &pre_encode_proxy_;
>+}
>+
>+void VideoAnalyzer::StartMeasuringCpuProcessTime() {
>+  rtc::CritScope lock(&cpu_measurement_lock_);
>+  cpu_time_ -= rtc::GetProcessCpuTimeNanos();
>+  wallclock_time_ -= rtc::SystemTimeNanos();
>+}
>+
>+void VideoAnalyzer::StopMeasuringCpuProcessTime() {
>+  rtc::CritScope lock(&cpu_measurement_lock_);
>+  cpu_time_ += rtc::GetProcessCpuTimeNanos();
>+  wallclock_time_ += rtc::SystemTimeNanos();
>+}
>+
>+void VideoAnalyzer::StartExcludingCpuThreadTime() {
>+  rtc::CritScope lock(&cpu_measurement_lock_);
>+  cpu_time_ += rtc::GetThreadCpuTimeNanos();
>+}
>+
>+void VideoAnalyzer::StopExcludingCpuThreadTime() {
>+  rtc::CritScope lock(&cpu_measurement_lock_);
>+  cpu_time_ -= rtc::GetThreadCpuTimeNanos();
>+}
>+
>+double VideoAnalyzer::GetCpuUsagePercent() {
>+  rtc::CritScope lock(&cpu_measurement_lock_);
>+  return static_cast<double>(cpu_time_) / wallclock_time_ * 100.0;
>+}
>+
>+bool VideoAnalyzer::IsInSelectedSpatialAndTemporalLayer(
>+    const uint8_t* packet,
>+    size_t length,
>+    const RTPHeader& header) {
>+  if (header.payloadType != test::CallTest::kPayloadTypeVP9 &&
>+      header.payloadType != test::CallTest::kPayloadTypeVP8) {
>+    return true;
>+  } else {
>+    // Get VP8 and VP9 specific header to check layers indexes.
>+    const uint8_t* payload = packet + header.headerLength;
>+    const size_t payload_length = length - header.headerLength;
>+    const size_t payload_data_length = payload_length - header.paddingLength;
>+    const bool is_vp8 = header.payloadType == test::CallTest::kPayloadTypeVP8;
>+    std::unique_ptr<RtpDepacketizer> depacketizer(
>+        RtpDepacketizer::Create(is_vp8 ? kVideoCodecVP8 : kVideoCodecVP9));
>+    RtpDepacketizer::ParsedPayload parsed_payload;
>+    bool result =
>+        depacketizer->Parse(&parsed_payload, payload, payload_data_length);
>+    RTC_DCHECK(result);
>+
>+    int temporal_idx;
>+    int spatial_idx;
>+    if (is_vp8) {
>+      temporal_idx = parsed_payload.video_header().vp8().temporalIdx;
>+      spatial_idx = kNoTemporalIdx;
>+    } else {
>+      const auto& vp9_header = absl::get<RTPVideoHeaderVP9>(
>+          parsed_payload.video_header().video_type_header);
>+      temporal_idx = vp9_header.temporal_idx;
>+      spatial_idx = vp9_header.spatial_idx;
>+    }
>+
>+    return (selected_tl_ < 0 || temporal_idx == kNoTemporalIdx ||
>+            temporal_idx <= selected_tl_) &&
>+           (selected_sl_ < 0 || spatial_idx == kNoSpatialIdx ||
>+            spatial_idx <= selected_sl_);
>+  }
>+}
>+
>+void VideoAnalyzer::PollStatsThread(void* obj) {
>+  static_cast<VideoAnalyzer*>(obj)->PollStats();
>+}
>+
>+void VideoAnalyzer::PollStats() {
>+  while (!done_.Wait(kSendStatsPollingIntervalMs)) {
>+    rtc::CritScope crit(&comparison_lock_);
>+
>+    Call::Stats call_stats = call_->GetStats();
>+    send_bandwidth_bps_.AddSample(call_stats.send_bandwidth_bps);
>+
>+    VideoSendStream::Stats send_stats = send_stream_->GetStats();
>+    // It's not certain that we yet have estimates for any of these stats.
>+    // Check that they are positive before mixing them in.
>+    if (send_stats.encode_frame_rate > 0)
>+      encode_frame_rate_.AddSample(send_stats.encode_frame_rate);
>+    if (send_stats.avg_encode_time_ms > 0)
>+      encode_time_ms_.AddSample(send_stats.avg_encode_time_ms);
>+    if (send_stats.encode_usage_percent > 0)
>+      encode_usage_percent_.AddSample(send_stats.encode_usage_percent);
>+    if (send_stats.media_bitrate_bps > 0)
>+      media_bitrate_bps_.AddSample(send_stats.media_bitrate_bps);
>+    size_t fec_bytes = 0;
>+    for (auto kv : send_stats.substreams) {
>+      fec_bytes += kv.second.rtp_stats.fec.payload_bytes +
>+                   kv.second.rtp_stats.fec.padding_bytes;
>+    }
>+    fec_bitrate_bps_.AddSample((fec_bytes - last_fec_bytes_) * 8);
>+    last_fec_bytes_ = fec_bytes;
>+
>+    if (receive_stream_ != nullptr) {
>+      VideoReceiveStream::Stats receive_stats = receive_stream_->GetStats();
>+      if (receive_stats.decode_ms > 0)
>+        decode_time_ms_.AddSample(receive_stats.decode_ms);
>+      if (receive_stats.max_decode_ms > 0)
>+        decode_time_max_ms_.AddSample(receive_stats.max_decode_ms);
>+    }
>+
>+    if (audio_receive_stream_ != nullptr) {
>+      AudioReceiveStream::Stats receive_stats =
>+          audio_receive_stream_->GetStats();
>+      audio_expand_rate_.AddSample(receive_stats.expand_rate);
>+      audio_accelerate_rate_.AddSample(receive_stats.accelerate_rate);
>+      audio_jitter_buffer_ms_.AddSample(receive_stats.jitter_buffer_ms);
>+    }
>+
>+    memory_usage_.AddSample(rtc::GetProcessResidentSizeBytes());
>+  }
>+}
>+
>+bool VideoAnalyzer::FrameComparisonThread(void* obj) {
>+  return static_cast<VideoAnalyzer*>(obj)->CompareFrames();
>+}
>+
>+bool VideoAnalyzer::CompareFrames() {
>+  if (AllFramesRecorded())
>+    return false;
>+
>+  FrameComparison comparison;
>+
>+  if (!PopComparison(&comparison)) {
>+    // Wait until new comparison task is available, or test is done.
>+    // If done, wake up remaining threads waiting.
>+    comparison_available_event_.Wait(1000);
>+    if (AllFramesRecorded()) {
>+      comparison_available_event_.Set();
>+      return false;
>+    }
>+    return true;  // Try again.
>+  }
>+
>+  StartExcludingCpuThreadTime();
>+
>+  PerformFrameComparison(comparison);
>+
>+  StopExcludingCpuThreadTime();
>+
>+  if (FrameProcessed()) {
>+    PrintResults();
>+    if (graph_data_output_file_)
>+      PrintSamplesToFile();
>+    done_.Set();
>+    comparison_available_event_.Set();
>+    return false;
>+  }
>+
>+  return true;
>+}
>+
>+bool VideoAnalyzer::PopComparison(VideoAnalyzer::FrameComparison* comparison) {
>+  rtc::CritScope crit(&comparison_lock_);
>+  // If AllFramesRecorded() is true, it means we have already popped
>+  // frames_to_process_ frames from comparisons_, so there is no more work
>+  // for this thread to be done. frames_processed_ might still be lower if
>+  // all comparisons are not done, but those frames are currently being
>+  // worked on by other threads.
>+  if (comparisons_.empty() || AllFramesRecorded())
>+    return false;
>+
>+  *comparison = comparisons_.front();
>+  comparisons_.pop_front();
>+
>+  FrameRecorded();
>+  return true;
>+}
>+
>+void VideoAnalyzer::FrameRecorded() {
>+  rtc::CritScope crit(&comparison_lock_);
>+  ++frames_recorded_;
>+}
>+
>+bool VideoAnalyzer::AllFramesRecorded() {
>+  rtc::CritScope crit(&comparison_lock_);
>+  assert(frames_recorded_ <= frames_to_process_);
>+  return frames_recorded_ == frames_to_process_;
>+}
>+
>+bool VideoAnalyzer::FrameProcessed() {
>+  rtc::CritScope crit(&comparison_lock_);
>+  ++frames_processed_;
>+  assert(frames_processed_ <= frames_to_process_);
>+  return frames_processed_ == frames_to_process_;
>+}
>+
>+void VideoAnalyzer::PrintResults() {
>+  StopMeasuringCpuProcessTime();
>+  rtc::CritScope crit(&comparison_lock_);
>+  // Record the time from the last freeze until the last rendered frame to
>+  // ensure we cover the full timespan of the session. Otherwise the metric
>+  // would penalize an early freeze followed by no freezes until the end.
>+  time_between_freezes_.AddSample(last_render_time_ - last_unfreeze_time_ms_);
>+  PrintResult("psnr", psnr_, " dB");
>+  PrintResult("ssim", ssim_, " score");
>+  PrintResult("sender_time", sender_time_, " ms");
>+  PrintResult("receiver_time", receiver_time_, " ms");
>+  PrintResult("network_time", network_time_, " ms");
>+  PrintResult("total_delay_incl_network", end_to_end_, " ms");
>+  PrintResult("time_between_rendered_frames", rendered_delta_, " ms");
>+  PrintResult("encode_frame_rate", encode_frame_rate_, " fps");
>+  PrintResult("encode_time", encode_time_ms_, " ms");
>+  PrintResult("media_bitrate", media_bitrate_bps_, " bps");
>+  PrintResult("fec_bitrate", fec_bitrate_bps_, " bps");
>+  PrintResult("send_bandwidth", send_bandwidth_bps_, " bps");
>+  PrintResult("time_between_freezes", time_between_freezes_, " ms");
>+
>+  if (worst_frame_) {
>+    test::PrintResult("min_psnr", "", test_label_.c_str(), worst_frame_->psnr,
>+                      "dB", false);
>+  }
>+
>+  if (receive_stream_ != nullptr) {
>+    PrintResult("decode_time", decode_time_ms_, " ms");
>+  }
>+
>+  test::PrintResult("dropped_frames", "", test_label_.c_str(), dropped_frames_,
>+                    "frames", false);
>+  test::PrintResult("cpu_usage", "", test_label_.c_str(), GetCpuUsagePercent(),
>+                    "%", false);
>+
>+#if defined(WEBRTC_WIN)
>+  // On Linux and Mac in Resident Set some unused pages may be counted.
>+  // Therefore this metric will depend on order in which tests are run and
>+  // will be flaky.
>+  PrintResult("memory_usage", memory_usage_, " bytes");
>+#endif
>+
>+  // Saving only the worst frame for manual analysis. Intention here is to
>+  // only detect video corruptions and not to track picture quality. Thus,
>+  // jpeg is used here.
>+  if (FLAG_save_worst_frame && worst_frame_) {
>+    std::string output_dir;
>+    test::GetTestArtifactsDir(&output_dir);
>+    std::string output_path =
>+        rtc::Pathname(output_dir, test_label_ + ".jpg").pathname();
>+    RTC_LOG(LS_INFO) << "Saving worst frame to " << output_path;
>+    test::JpegFrameWriter frame_writer(output_path);
>+    RTC_CHECK(
>+        frame_writer.WriteFrame(worst_frame_->frame, 100 /*best quality*/));
>+  }
>+
>+  if (audio_receive_stream_ != nullptr) {
>+    PrintResult("audio_expand_rate", audio_expand_rate_, "");
>+    PrintResult("audio_accelerate_rate", audio_accelerate_rate_, "");
>+    PrintResult("audio_jitter_buffer", audio_jitter_buffer_ms_, " ms");
>+  }
>+
>+  //  Disable quality check for quick test, as quality checks may fail
>+  //  because too few samples were collected.
>+  if (!is_quick_test_enabled_) {
>+    EXPECT_GT(psnr_.Mean(), avg_psnr_threshold_);
>+    EXPECT_GT(ssim_.Mean(), avg_ssim_threshold_);
>+  }
>+}
>+
>+void VideoAnalyzer::PerformFrameComparison(
>+    const VideoAnalyzer::FrameComparison& comparison) {
>+  // Perform expensive psnr and ssim calculations while not holding lock.
>+  double psnr = -1.0;
>+  double ssim = -1.0;
>+  if (comparison.reference && !comparison.dropped) {
>+    psnr = I420PSNR(&*comparison.reference, &*comparison.render);
>+    ssim = I420SSIM(&*comparison.reference, &*comparison.render);
>+  }
>+
>+  rtc::CritScope crit(&comparison_lock_);
>+
>+  if (psnr >= 0.0 && (!worst_frame_ || worst_frame_->psnr > psnr)) {
>+    worst_frame_.emplace(FrameWithPsnr{psnr, *comparison.render});
>+  }
>+
>+  if (graph_data_output_file_) {
>+    samples_.push_back(Sample(comparison.dropped, comparison.input_time_ms,
>+                              comparison.send_time_ms, comparison.recv_time_ms,
>+                              comparison.render_time_ms,
>+                              comparison.encoded_frame_size, psnr, ssim));
>+  }
>+  if (psnr >= 0.0)
>+    psnr_.AddSample(psnr);
>+  if (ssim >= 0.0)
>+    ssim_.AddSample(ssim);
>+
>+  if (comparison.dropped) {
>+    ++dropped_frames_;
>+    return;
>+  }
>+  if (last_unfreeze_time_ms_ == 0)
>+    last_unfreeze_time_ms_ = comparison.render_time_ms;
>+  if (last_render_time_ != 0) {
>+    const int64_t render_delta_ms =
>+        comparison.render_time_ms - last_render_time_;
>+    rendered_delta_.AddSample(render_delta_ms);
>+    if (last_render_delta_ms_ != 0 &&
>+        render_delta_ms - last_render_delta_ms_ > 150) {
>+      time_between_freezes_.AddSample(last_render_time_ -
>+                                      last_unfreeze_time_ms_);
>+      last_unfreeze_time_ms_ = comparison.render_time_ms;
>+    }
>+    last_render_delta_ms_ = render_delta_ms;
>+  }
>+  last_render_time_ = comparison.render_time_ms;
>+
>+  sender_time_.AddSample(comparison.send_time_ms - comparison.input_time_ms);
>+  if (comparison.recv_time_ms > 0) {
>+    // If recv_time_ms == 0, this frame consisted of a packets which were all
>+    // lost in the transport. Since we were able to render the frame, however,
>+    // the dropped packets were recovered by FlexFEC. The FlexFEC recovery
>+    // happens internally in Call, and we can therefore here not know which
>+    // FEC packets that protected the lost media packets. Consequently, we
>+    // were not able to record a meaningful recv_time_ms. We therefore skip
>+    // this sample.
>+    //
>+    // The reasoning above does not hold for ULPFEC and RTX, as for those
>+    // strategies the timestamp of the received packets is set to the
>+    // timestamp of the protected/retransmitted media packet. I.e., then
>+    // recv_time_ms != 0, even though the media packets were lost.
>+    receiver_time_.AddSample(comparison.render_time_ms -
>+                             comparison.recv_time_ms);
>+    network_time_.AddSample(comparison.recv_time_ms - comparison.send_time_ms);
>+  }
>+  end_to_end_.AddSample(comparison.render_time_ms - comparison.input_time_ms);
>+  encoded_frame_size_.AddSample(comparison.encoded_frame_size);
>+}
>+
>+void VideoAnalyzer::PrintResult(const char* result_type,
>+                                test::Statistics stats,
>+                                const char* unit) {
>+  test::PrintResultMeanAndError(result_type, "", test_label_.c_str(),
>+                                stats.Mean(), stats.StandardDeviation(), unit,
>+                                false);
>+}
>+
>+void VideoAnalyzer::PrintSamplesToFile() {
>+  FILE* out = graph_data_output_file_;
>+  rtc::CritScope crit(&comparison_lock_);
>+  std::sort(samples_.begin(), samples_.end(),
>+            [](const Sample& A, const Sample& B) -> bool {
>+              return A.input_time_ms < B.input_time_ms;
>+            });
>+
>+  fprintf(out, "%s\n", graph_title_.c_str());
>+  fprintf(out, "%" PRIuS "\n", samples_.size());
>+  fprintf(out,
>+          "dropped "
>+          "input_time_ms "
>+          "send_time_ms "
>+          "recv_time_ms "
>+          "render_time_ms "
>+          "encoded_frame_size "
>+          "psnr "
>+          "ssim "
>+          "encode_time_ms\n");
>+  for (const Sample& sample : samples_) {
>+    fprintf(out,
>+            "%d %" PRId64 " %" PRId64 " %" PRId64 " %" PRId64 " %" PRIuS
>+            " %lf %lf\n",
>+            sample.dropped, sample.input_time_ms, sample.send_time_ms,
>+            sample.recv_time_ms, sample.render_time_ms,
>+            sample.encoded_frame_size, sample.psnr, sample.ssim);
>+  }
>+}
>+
>+double VideoAnalyzer::GetAverageMediaBitrateBps() {
>+  if (last_sending_time_ == first_sending_time_) {
>+    return 0;
>+  } else {
>+    return static_cast<double>(total_media_bytes_) * 8 /
>+           (last_sending_time_ - first_sending_time_) *
>+           rtc::kNumMillisecsPerSec;
>+  }
>+}
>+
>+void VideoAnalyzer::AddCapturedFrameForComparison(
>+    const VideoFrame& video_frame) {
>+  rtc::CritScope lock(&crit_);
>+  frames_.push_back(video_frame);
>+}
>+
>+void VideoAnalyzer::AddFrameComparison(const VideoFrame& reference,
>+                                       const VideoFrame& render,
>+                                       bool dropped,
>+                                       int64_t render_time_ms) {
>+  int64_t reference_timestamp = wrap_handler_.Unwrap(reference.timestamp());
>+  int64_t send_time_ms = send_times_[reference_timestamp];
>+  send_times_.erase(reference_timestamp);
>+  int64_t recv_time_ms = recv_times_[reference_timestamp];
>+  recv_times_.erase(reference_timestamp);
>+
>+  // TODO(ivica): Make this work for > 2 streams.
>+  auto it = encoded_frame_sizes_.find(reference_timestamp);
>+  if (it == encoded_frame_sizes_.end())
>+    it = encoded_frame_sizes_.find(reference_timestamp - 1);
>+  size_t encoded_size = it == encoded_frame_sizes_.end() ? 0 : it->second;
>+  if (it != encoded_frame_sizes_.end())
>+    encoded_frame_sizes_.erase(it);
>+
>+  rtc::CritScope crit(&comparison_lock_);
>+  if (comparisons_.size() < kMaxComparisons) {
>+    comparisons_.push_back(FrameComparison(
>+        reference, render, dropped, reference.ntp_time_ms(), send_time_ms,
>+        recv_time_ms, render_time_ms, encoded_size));
>+  } else {
>+    comparisons_.push_back(FrameComparison(dropped, reference.ntp_time_ms(),
>+                                           send_time_ms, recv_time_ms,
>+                                           render_time_ms, encoded_size));
>+  }
>+  comparison_available_event_.Set();
>+}
>+
>+VideoAnalyzer::FrameComparison::FrameComparison()
>+    : dropped(false),
>+      input_time_ms(0),
>+      send_time_ms(0),
>+      recv_time_ms(0),
>+      render_time_ms(0),
>+      encoded_frame_size(0) {}
>+
>+VideoAnalyzer::FrameComparison::FrameComparison(const VideoFrame& reference,
>+                                                const VideoFrame& render,
>+                                                bool dropped,
>+                                                int64_t input_time_ms,
>+                                                int64_t send_time_ms,
>+                                                int64_t recv_time_ms,
>+                                                int64_t render_time_ms,
>+                                                size_t encoded_frame_size)
>+    : reference(reference),
>+      render(render),
>+      dropped(dropped),
>+      input_time_ms(input_time_ms),
>+      send_time_ms(send_time_ms),
>+      recv_time_ms(recv_time_ms),
>+      render_time_ms(render_time_ms),
>+      encoded_frame_size(encoded_frame_size) {}
>+
>+VideoAnalyzer::FrameComparison::FrameComparison(bool dropped,
>+                                                int64_t input_time_ms,
>+                                                int64_t send_time_ms,
>+                                                int64_t recv_time_ms,
>+                                                int64_t render_time_ms,
>+                                                size_t encoded_frame_size)
>+    : dropped(dropped),
>+      input_time_ms(input_time_ms),
>+      send_time_ms(send_time_ms),
>+      recv_time_ms(recv_time_ms),
>+      render_time_ms(render_time_ms),
>+      encoded_frame_size(encoded_frame_size) {}
>+
>+VideoAnalyzer::Sample::Sample(int dropped,
>+                              int64_t input_time_ms,
>+                              int64_t send_time_ms,
>+                              int64_t recv_time_ms,
>+                              int64_t render_time_ms,
>+                              size_t encoded_frame_size,
>+                              double psnr,
>+                              double ssim)
>+    : dropped(dropped),
>+      input_time_ms(input_time_ms),
>+      send_time_ms(send_time_ms),
>+      recv_time_ms(recv_time_ms),
>+      render_time_ms(render_time_ms),
>+      encoded_frame_size(encoded_frame_size),
>+      psnr(psnr),
>+      ssim(ssim) {}
>+
>+VideoAnalyzer::PreEncodeProxy::PreEncodeProxy(VideoAnalyzer* parent)
>+    : parent_(parent) {}
>+
>+void VideoAnalyzer::PreEncodeProxy::OnFrame(const VideoFrame& video_frame) {
>+  parent_->PreEncodeOnFrame(video_frame);
>+}
>+
>+VideoAnalyzer::CapturedFrameForwarder::CapturedFrameForwarder(
>+    VideoAnalyzer* analyzer,
>+    Clock* clock)
>+    : analyzer_(analyzer),
>+      send_stream_input_(nullptr),
>+      video_capturer_(nullptr),
>+      clock_(clock) {}
>+
>+void VideoAnalyzer::CapturedFrameForwarder::SetSource(
>+    test::VideoCapturer* video_capturer) {
>+  video_capturer_ = video_capturer;
>+}
>+
>+void VideoAnalyzer::CapturedFrameForwarder::OnFrame(
>+    const VideoFrame& video_frame) {
>+  VideoFrame copy = video_frame;
>+  // Frames from the capturer does not have a rtp timestamp.
>+  // Create one so it can be used for comparison.
>+  RTC_DCHECK_EQ(0, video_frame.timestamp());
>+  if (video_frame.ntp_time_ms() == 0)
>+    copy.set_ntp_time_ms(clock_->CurrentNtpInMilliseconds());
>+  copy.set_timestamp(copy.ntp_time_ms() * 90);
>+  analyzer_->AddCapturedFrameForComparison(copy);
>+  rtc::CritScope lock(&crit_);
>+  if (send_stream_input_)
>+    send_stream_input_->OnFrame(copy);
>+}
>+
>+void VideoAnalyzer::CapturedFrameForwarder::AddOrUpdateSink(
>+    rtc::VideoSinkInterface<VideoFrame>* sink,
>+    const rtc::VideoSinkWants& wants) {
>+  {
>+    rtc::CritScope lock(&crit_);
>+    RTC_DCHECK(!send_stream_input_ || send_stream_input_ == sink);
>+    send_stream_input_ = sink;
>+  }
>+  if (video_capturer_) {
>+    video_capturer_->AddOrUpdateSink(this, wants);
>+  }
>+}
>+
>+void VideoAnalyzer::CapturedFrameForwarder::RemoveSink(
>+    rtc::VideoSinkInterface<VideoFrame>* sink) {
>+  rtc::CritScope lock(&crit_);
>+  RTC_DCHECK(sink == send_stream_input_);
>+  send_stream_input_ = nullptr;
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_analyzer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_analyzer.h
>new file mode 100644
>index 00000000000..b4d31dbaea1
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_analyzer.h
>@@ -0,0 +1,296 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#ifndef VIDEO_VIDEO_ANALYZER_H_
>+#define VIDEO_VIDEO_ANALYZER_H_
>+
>+#include <deque>
>+#include <map>
>+#include <memory>
>+#include <string>
>+#include <vector>
>+
>+#include "test/layer_filtering_transport.h"
>+#include "test/rtp_file_writer.h"
>+#include "test/statistics.h"
>+#include "test/vcm_capturer.h"
>+
>+namespace webrtc {
>+
>+class VideoAnalyzer : public PacketReceiver,
>+                      public Transport,
>+                      public rtc::VideoSinkInterface<VideoFrame>,
>+                      public EncodedFrameObserver {
>+ public:
>+  VideoAnalyzer(test::LayerFilteringTransport* transport,
>+                const std::string& test_label,
>+                double avg_psnr_threshold,
>+                double avg_ssim_threshold,
>+                int duration_frames,
>+                FILE* graph_data_output_file,
>+                const std::string& graph_title,
>+                uint32_t ssrc_to_analyze,
>+                uint32_t rtx_ssrc_to_analyze,
>+                size_t selected_stream,
>+                int selected_sl,
>+                int selected_tl,
>+                bool is_quick_test_enabled,
>+                Clock* clock,
>+                std::string rtp_dump_name);
>+  ~VideoAnalyzer();
>+
>+  virtual void SetReceiver(PacketReceiver* receiver);
>+  void SetSource(test::VideoCapturer* video_capturer, bool respect_sink_wants);
>+  void SetCall(Call* call);
>+  void SetSendStream(VideoSendStream* stream);
>+  void SetReceiveStream(VideoReceiveStream* stream);
>+  void SetAudioReceiveStream(AudioReceiveStream* recv_stream);
>+
>+  rtc::VideoSinkInterface<VideoFrame>* InputInterface();
>+  rtc::VideoSourceInterface<VideoFrame>* OutputInterface();
>+
>+  DeliveryStatus DeliverPacket(MediaType media_type,
>+                               rtc::CopyOnWriteBuffer packet,
>+                               int64_t packet_time_us) override;
>+
>+  void PreEncodeOnFrame(const VideoFrame& video_frame);
>+
>+  // EncodedFrameObserver implementation, wired to post_encode_callback.
>+  void EncodedFrameCallback(const EncodedFrame& encoded_frame) override;
>+
>+  bool SendRtp(const uint8_t* packet,
>+               size_t length,
>+               const PacketOptions& options) override;
>+
>+  bool SendRtcp(const uint8_t* packet, size_t length) override;
>+  void OnFrame(const VideoFrame& video_frame) override;
>+  void Wait();
>+
>+  rtc::VideoSinkInterface<VideoFrame>* pre_encode_proxy();
>+
>+  void StartMeasuringCpuProcessTime();
>+  void StopMeasuringCpuProcessTime();
>+  void StartExcludingCpuThreadTime();
>+  void StopExcludingCpuThreadTime();
>+  double GetCpuUsagePercent();
>+
>+  test::LayerFilteringTransport* const transport_;
>+  PacketReceiver* receiver_;
>+
>+ private:
>+  struct FrameComparison {
>+    FrameComparison();
>+    FrameComparison(const VideoFrame& reference,
>+                    const VideoFrame& render,
>+                    bool dropped,
>+                    int64_t input_time_ms,
>+                    int64_t send_time_ms,
>+                    int64_t recv_time_ms,
>+                    int64_t render_time_ms,
>+                    size_t encoded_frame_size);
>+    FrameComparison(bool dropped,
>+                    int64_t input_time_ms,
>+                    int64_t send_time_ms,
>+                    int64_t recv_time_ms,
>+                    int64_t render_time_ms,
>+                    size_t encoded_frame_size);
>+
>+    absl::optional<VideoFrame> reference;
>+    absl::optional<VideoFrame> render;
>+    bool dropped;
>+    int64_t input_time_ms;
>+    int64_t send_time_ms;
>+    int64_t recv_time_ms;
>+    int64_t render_time_ms;
>+    size_t encoded_frame_size;
>+  };
>+
>+  struct Sample {
>+    Sample(int dropped,
>+           int64_t input_time_ms,
>+           int64_t send_time_ms,
>+           int64_t recv_time_ms,
>+           int64_t render_time_ms,
>+           size_t encoded_frame_size,
>+           double psnr,
>+           double ssim);
>+
>+    int dropped;
>+    int64_t input_time_ms;
>+    int64_t send_time_ms;
>+    int64_t recv_time_ms;
>+    int64_t render_time_ms;
>+    size_t encoded_frame_size;
>+    double psnr;
>+    double ssim;
>+  };
>+
>+  // This class receives the send-side OnFrame callback and is provided to not
>+  // conflict with the receiver-side renderer callback.
>+  class PreEncodeProxy : public rtc::VideoSinkInterface<VideoFrame> {
>+   public:
>+    explicit PreEncodeProxy(VideoAnalyzer* parent);
>+    void OnFrame(const VideoFrame& video_frame) override;
>+
>+   private:
>+    VideoAnalyzer* const parent_;
>+  };
>+
>+  // Implements VideoSinkInterface to receive captured frames from a
>+  // FrameGeneratorCapturer. Implements VideoSourceInterface to be able to act
>+  // as a source to VideoSendStream.
>+  // It forwards all input frames to the VideoAnalyzer for later comparison and
>+  // forwards the captured frames to the VideoSendStream.
>+  class CapturedFrameForwarder : public rtc::VideoSinkInterface<VideoFrame>,
>+                                 public rtc::VideoSourceInterface<VideoFrame> {
>+   public:
>+    explicit CapturedFrameForwarder(VideoAnalyzer* analyzer, Clock* clock);
>+    void SetSource(test::VideoCapturer* video_capturer);
>+
>+   private:
>+    void OnFrame(const VideoFrame& video_frame) override;
>+
>+    // Called when |send_stream_.SetSource()| is called.
>+    void AddOrUpdateSink(rtc::VideoSinkInterface<VideoFrame>* sink,
>+                         const rtc::VideoSinkWants& wants) override;
>+
>+    // Called by |send_stream_| when |send_stream_.SetSource()| is called.
>+    void RemoveSink(rtc::VideoSinkInterface<VideoFrame>* sink) override;
>+
>+    VideoAnalyzer* const analyzer_;
>+    rtc::CriticalSection crit_;
>+    rtc::VideoSinkInterface<VideoFrame>* send_stream_input_
>+        RTC_GUARDED_BY(crit_);
>+    test::VideoCapturer* video_capturer_;
>+    Clock* clock_;
>+  };
>+
>+  struct FrameWithPsnr {
>+    double psnr;
>+    VideoFrame frame;
>+  };
>+
>+  bool IsInSelectedSpatialAndTemporalLayer(const uint8_t* packet,
>+                                           size_t length,
>+                                           const RTPHeader& header);
>+
>+  void AddFrameComparison(const VideoFrame& reference,
>+                          const VideoFrame& render,
>+                          bool dropped,
>+                          int64_t render_time_ms)
>+      RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_);
>+
>+  static void PollStatsThread(void* obj);
>+  void PollStats();
>+  static bool FrameComparisonThread(void* obj);
>+  bool CompareFrames();
>+  bool PopComparison(FrameComparison* comparison);
>+  // Increment counter for number of frames received for comparison.
>+  void FrameRecorded();
>+  // Returns true if all frames to be compared have been taken from the queue.
>+  bool AllFramesRecorded();
>+  // Increase count of number of frames processed. Returns true if this was the
>+  // last frame to be processed.
>+  bool FrameProcessed();
>+  void PrintResults();
>+  void PerformFrameComparison(const FrameComparison& comparison);
>+  void PrintResult(const char* result_type,
>+                   test::Statistics stats,
>+                   const char* unit);
>+  void PrintSamplesToFile(void);
>+  double GetAverageMediaBitrateBps();
>+  void AddCapturedFrameForComparison(const VideoFrame& video_frame);
>+
>+  Call* call_;
>+  VideoSendStream* send_stream_;
>+  VideoReceiveStream* receive_stream_;
>+  AudioReceiveStream* audio_receive_stream_;
>+  CapturedFrameForwarder captured_frame_forwarder_;
>+  const std::string test_label_;
>+  FILE* const graph_data_output_file_;
>+  const std::string graph_title_;
>+  const uint32_t ssrc_to_analyze_;
>+  const uint32_t rtx_ssrc_to_analyze_;
>+  const size_t selected_stream_;
>+  const int selected_sl_;
>+  const int selected_tl_;
>+  PreEncodeProxy pre_encode_proxy_;
>+
>+  rtc::CriticalSection comparison_lock_;
>+  std::vector<Sample> samples_ RTC_GUARDED_BY(comparison_lock_);
>+  test::Statistics sender_time_ RTC_GUARDED_BY(comparison_lock_);
>+  test::Statistics receiver_time_ RTC_GUARDED_BY(comparison_lock_);
>+  test::Statistics network_time_ RTC_GUARDED_BY(comparison_lock_);
>+  test::Statistics psnr_ RTC_GUARDED_BY(comparison_lock_);
>+  test::Statistics ssim_ RTC_GUARDED_BY(comparison_lock_);
>+  test::Statistics end_to_end_ RTC_GUARDED_BY(comparison_lock_);
>+  test::Statistics rendered_delta_ RTC_GUARDED_BY(comparison_lock_);
>+  test::Statistics encoded_frame_size_ RTC_GUARDED_BY(comparison_lock_);
>+  test::Statistics encode_frame_rate_ RTC_GUARDED_BY(comparison_lock_);
>+  test::Statistics encode_time_ms_ RTC_GUARDED_BY(comparison_lock_);
>+  test::Statistics encode_usage_percent_ RTC_GUARDED_BY(comparison_lock_);
>+  test::Statistics decode_time_ms_ RTC_GUARDED_BY(comparison_lock_);
>+  test::Statistics decode_time_max_ms_ RTC_GUARDED_BY(comparison_lock_);
>+  test::Statistics media_bitrate_bps_ RTC_GUARDED_BY(comparison_lock_);
>+  test::Statistics fec_bitrate_bps_ RTC_GUARDED_BY(comparison_lock_);
>+  test::Statistics send_bandwidth_bps_ RTC_GUARDED_BY(comparison_lock_);
>+  test::Statistics memory_usage_ RTC_GUARDED_BY(comparison_lock_);
>+  test::Statistics time_between_freezes_ RTC_GUARDED_BY(comparison_lock_);
>+  test::Statistics audio_expand_rate_ RTC_GUARDED_BY(comparison_lock_);
>+  test::Statistics audio_accelerate_rate_ RTC_GUARDED_BY(comparison_lock_);
>+  test::Statistics audio_jitter_buffer_ms_ RTC_GUARDED_BY(comparison_lock_);
>+  // Rendered frame with worst PSNR is saved for further analysis.
>+  absl::optional<FrameWithPsnr> worst_frame_ RTC_GUARDED_BY(comparison_lock_);
>+
>+  size_t last_fec_bytes_;
>+
>+  const int frames_to_process_;
>+  int frames_recorded_;
>+  int frames_processed_;
>+  int dropped_frames_;
>+  int dropped_frames_before_first_encode_;
>+  int dropped_frames_before_rendering_;
>+  int64_t last_render_time_;
>+  int64_t last_render_delta_ms_;
>+  int64_t last_unfreeze_time_ms_;
>+  uint32_t rtp_timestamp_delta_;
>+  int64_t total_media_bytes_;
>+  int64_t first_sending_time_;
>+  int64_t last_sending_time_;
>+
>+  rtc::CriticalSection cpu_measurement_lock_;
>+  int64_t cpu_time_ RTC_GUARDED_BY(cpu_measurement_lock_);
>+  int64_t wallclock_time_ RTC_GUARDED_BY(cpu_measurement_lock_);
>+
>+  rtc::CriticalSection crit_;
>+  std::deque<VideoFrame> frames_ RTC_GUARDED_BY(crit_);
>+  absl::optional<VideoFrame> last_rendered_frame_ RTC_GUARDED_BY(crit_);
>+  rtc::TimestampWrapAroundHandler wrap_handler_ RTC_GUARDED_BY(crit_);
>+  std::map<int64_t, int64_t> send_times_ RTC_GUARDED_BY(crit_);
>+  std::map<int64_t, int64_t> recv_times_ RTC_GUARDED_BY(crit_);
>+  std::map<int64_t, size_t> encoded_frame_sizes_ RTC_GUARDED_BY(crit_);
>+  absl::optional<uint32_t> first_encoded_timestamp_ RTC_GUARDED_BY(crit_);
>+  absl::optional<uint32_t> first_sent_timestamp_ RTC_GUARDED_BY(crit_);
>+  const double avg_psnr_threshold_;
>+  const double avg_ssim_threshold_;
>+  bool is_quick_test_enabled_;
>+
>+  std::vector<rtc::PlatformThread*> comparison_thread_pool_;
>+  rtc::PlatformThread stats_polling_thread_;
>+  rtc::Event comparison_available_event_;
>+  std::deque<FrameComparison> comparisons_ RTC_GUARDED_BY(comparison_lock_);
>+  rtc::Event done_;
>+
>+  std::unique_ptr<test::RtpFileWriter> rtp_file_writer_;
>+  Clock* const clock_;
>+  const int64_t start_ms_;
>+};
>+
>+}  // namespace webrtc
>+#endif  // VIDEO_VIDEO_ANALYZER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_loopback.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_loopback.cc
>index 37793cce833..a4d4547dfc9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_loopback.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_loopback.cc
>@@ -11,6 +11,8 @@
> #include <stdio.h>
> 
> #include "rtc_base/flags.h"
>+#include "rtc_base/logging.h"
>+#include "system_wrappers/include/field_trial_default.h"
> #include "test/field_trial.h"
> #include "test/gtest.h"
> #include "test/run_test.h"
>@@ -71,6 +73,21 @@ int NumTemporalLayers() {
>   return static_cast<int>(FLAG_num_temporal_layers);
> }
> 
>+DEFINE_int(inter_layer_pred,
>+           2,
>+           "Inter-layer prediction mode. "
>+           "0 - enabled, 1 - disabled, 2 - enabled only for key pictures.");
>+InterLayerPredMode InterLayerPred() {
>+  if (FLAG_inter_layer_pred == 0) {
>+    return InterLayerPredMode::kOn;
>+  } else if (FLAG_inter_layer_pred == 1) {
>+    return InterLayerPredMode::kOff;
>+  } else {
>+    RTC_DCHECK_EQ(FLAG_inter_layer_pred, 2);
>+    return InterLayerPredMode::kOnKeyPic;
>+  }
>+}
>+
> // Flags common with screenshare loopback, with equal default values.
> DEFINE_string(codec, "VP8", "Video codec to use.");
> std::string Codec() {
>@@ -226,8 +243,10 @@ DEFINE_bool(use_flexfec, false, "Use FlexFEC forward error correction.");
> 
> DEFINE_bool(audio, false, "Add audio stream");
> 
>-DEFINE_bool(audio_video_sync, false, "Sync audio and video stream (no effect if"
>-    " audio is false)");
>+DEFINE_bool(audio_video_sync,
>+            false,
>+            "Sync audio and video stream (no effect if"
>+            " audio is false)");
> 
> DEFINE_bool(audio_dtx, false, "Enable audio DTX (no effect if audio is false)");
> 
>@@ -263,10 +282,10 @@ void Loopback() {
>   pipe_config.delay_standard_deviation_ms = flags::StdPropagationDelayMs();
>   pipe_config.allow_reordering = flags::FLAG_allow_reordering;
> 
>-  Call::Config::BitrateConfig call_bitrate_config;
>+  BitrateConstraints call_bitrate_config;
>   call_bitrate_config.min_bitrate_bps = flags::MinBitrateKbps() * 1000;
>   call_bitrate_config.start_bitrate_bps = flags::StartBitrateKbps() * 1000;
>-  call_bitrate_config.max_bitrate_bps = flags::MaxBitrateKbps() * 1000;
>+  call_bitrate_config.max_bitrate_bps = -1;  // Don't cap bandwidth estimate.
> 
>   VideoQualityTest::Params params;
>   params.call = {flags::FLAG_send_side_bwe, call_bitrate_config, 0};
>@@ -284,15 +303,20 @@ void Loopback() {
>                      0,  // No min transmit bitrate.
>                      flags::FLAG_use_ulpfec,
>                      flags::FLAG_use_flexfec,
>+                     false,
>                      flags::Clip(),
>                      flags::GetCaptureDevice()};
>   params.audio = {flags::FLAG_audio, flags::FLAG_audio_video_sync,
>                   flags::FLAG_audio_dtx};
>-  params.logging = {flags::FLAG_logs, flags::FLAG_rtc_event_log_name,
>-                    flags::FLAG_rtp_dump_name, flags::FLAG_encoded_frame_path};
>+  params.logging = {flags::FLAG_rtc_event_log_name, flags::FLAG_rtp_dump_name,
>+                    flags::FLAG_encoded_frame_path};
>   params.screenshare[0].enabled = false;
>-  params.analyzer = {"video", 0.0, 0.0, flags::DurationSecs(),
>-      flags::OutputFilename(), flags::GraphTitle()};
>+  params.analyzer = {"video",
>+                     0.0,
>+                     0.0,
>+                     flags::DurationSecs(),
>+                     flags::OutputFilename(),
>+                     flags::GraphTitle()};
>   params.pipe = pipe_config;
> 
>   if (flags::NumStreams() > 1 && flags::Stream0().empty() &&
>@@ -309,13 +333,13 @@ void Loopback() {
>   VideoQualityTest::FillScalabilitySettings(
>       &params, 0, stream_descriptors, flags::NumStreams(),
>       flags::SelectedStream(), flags::NumSpatialLayers(), flags::SelectedSL(),
>-      SL_descriptors);
>+      flags::InterLayerPred(), SL_descriptors);
> 
>-  VideoQualityTest test;
>+  auto fixture = absl::make_unique<VideoQualityTest>(nullptr);
>   if (flags::DurationSecs()) {
>-    test.RunWithAnalyzer(params);
>+    fixture->RunWithAnalyzer(params);
>   } else {
>-    test.RunWithRenderers(params);
>+    fixture->RunWithRenderers(params);
>   }
> }
> }  // namespace webrtc
>@@ -328,10 +352,14 @@ int main(int argc, char* argv[]) {
>     return 0;
>   }
> 
>-  // InitFieldTrialsFromString needs a reference to an std::string instance,
>-  // with a scope that outlives the test.
>-  std::string field_trials = webrtc::flags::FLAG_force_fieldtrials;
>-  webrtc::test::InitFieldTrialsFromString(field_trials);
>+  rtc::LogMessage::SetLogToStderr(webrtc::flags::FLAG_logs);
>+
>+  webrtc::test::ValidateFieldTrialsStringOrDie(
>+      webrtc::flags::FLAG_force_fieldtrials);
>+  // InitFieldTrialsFromString stores the char*, so the char array must outlive
>+  // the application.
>+  webrtc::field_trial::InitFieldTrialsFromString(
>+      webrtc::flags::FLAG_force_fieldtrials);
> 
>   webrtc::test::RunTest(webrtc::Loopback);
>   return 0;
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_quality_observer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_quality_observer.cc
>new file mode 100644
>index 00000000000..b35a8c81641
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_quality_observer.cc
>@@ -0,0 +1,194 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "video/video_quality_observer.h"
>+
>+#include <algorithm>
>+#include <string>
>+
>+#include "rtc_base/checks.h"
>+#include "rtc_base/logging.h"
>+#include "rtc_base/strings/string_builder.h"
>+#include "system_wrappers/include/metrics.h"
>+
>+namespace webrtc {
>+
>+namespace {
>+const int kMinFrameSamplesToDetectFreeze = 5;
>+const int kMinCallDurationMs = 3000;
>+const int kMinRequiredSamples = 1;
>+const int kMinIncreaseForFreezeMs = 150;
>+const int kPixelsInHighResolution = 960 * 540;  // CPU-adapted HD still counts.
>+const int kPixelsInMediumResolution = 640 * 360;
>+const int kBlockyQpThresholdVp8 = 70;
>+const int kBlockyQpThresholdVp9 = 60;  // TODO(ilnik): tune this value.
>+// TODO(ilnik): Add H264/HEVC thresholds.
>+}  // namespace
>+
>+VideoQualityObserver::VideoQualityObserver(VideoContentType content_type)
>+    : last_frame_decoded_ms_(-1),
>+      num_frames_decoded_(0),
>+      first_frame_decoded_ms_(-1),
>+      last_frame_pixels_(0),
>+      last_frame_qp_(0),
>+      last_unfreeze_time_(0),
>+      time_in_resolution_ms_(3, 0),
>+      current_resolution_(Resolution::Low),
>+      num_resolution_downgrades_(0),
>+      time_in_blocky_video_ms_(0),
>+      content_type_(content_type),
>+      is_paused_(false) {}
>+
>+VideoQualityObserver::~VideoQualityObserver() {
>+  UpdateHistograms();
>+}
>+
>+void VideoQualityObserver::UpdateHistograms() {
>+  // Don't report anything on an empty video stream.
>+  if (num_frames_decoded_ == 0) {
>+    return;
>+  }
>+
>+  char log_stream_buf[2 * 1024];
>+  rtc::SimpleStringBuilder log_stream(log_stream_buf);
>+
>+  if (last_frame_decoded_ms_ > last_unfreeze_time_) {
>+    smooth_playback_durations_.Add(last_frame_decoded_ms_ -
>+                                   last_unfreeze_time_);
>+  }
>+
>+  std::string uma_prefix = videocontenttypehelpers::IsScreenshare(content_type_)
>+                               ? "WebRTC.Video.Screenshare"
>+                               : "WebRTC.Video";
>+
>+  auto mean_time_between_freezes =
>+      smooth_playback_durations_.Avg(kMinRequiredSamples);
>+  if (mean_time_between_freezes) {
>+    RTC_HISTOGRAM_COUNTS_SPARSE_100000(uma_prefix + ".MeanTimeBetweenFreezesMs",
>+                                       *mean_time_between_freezes);
>+    log_stream << uma_prefix << ".MeanTimeBetweenFreezesMs "
>+               << *mean_time_between_freezes << "\n";
>+  }
>+  auto avg_freeze_length = freezes_durations_.Avg(kMinRequiredSamples);
>+  if (avg_freeze_length) {
>+    RTC_HISTOGRAM_COUNTS_SPARSE_100000(uma_prefix + ".MeanFreezeDurationMs",
>+                                       *avg_freeze_length);
>+    log_stream << uma_prefix << ".MeanFreezeDurationMs " << *avg_freeze_length
>+               << "\n";
>+  }
>+
>+  int64_t call_duration_ms = last_frame_decoded_ms_ - first_frame_decoded_ms_;
>+
>+  if (call_duration_ms >= kMinCallDurationMs) {
>+    int time_spent_in_hd_percentage = static_cast<int>(
>+        time_in_resolution_ms_[Resolution::High] * 100 / call_duration_ms);
>+    int time_with_blocky_video_percentage =
>+        static_cast<int>(time_in_blocky_video_ms_ * 100 / call_duration_ms);
>+
>+    RTC_HISTOGRAM_COUNTS_SPARSE_100(uma_prefix + ".TimeInHdPercentage",
>+                                    time_spent_in_hd_percentage);
>+    log_stream << uma_prefix << ".TimeInHdPercentage "
>+               << time_spent_in_hd_percentage << "\n";
>+    RTC_HISTOGRAM_COUNTS_SPARSE_100(uma_prefix + ".TimeInBlockyVideoPercentage",
>+                                    time_with_blocky_video_percentage);
>+    log_stream << uma_prefix << ".TimeInBlockyVideoPercentage "
>+               << time_with_blocky_video_percentage << "\n";
>+    RTC_HISTOGRAM_COUNTS_SPARSE_100(
>+        uma_prefix + ".NumberResolutionDownswitchesPerMinute",
>+        num_resolution_downgrades_ * 60000 / call_duration_ms);
>+    log_stream << uma_prefix << ".NumberResolutionDownswitchesPerMinute "
>+               << num_resolution_downgrades_ * 60000 / call_duration_ms << "\n";
>+  }
>+  RTC_LOG(LS_INFO) << log_stream.str();
>+}
>+
>+void VideoQualityObserver::OnDecodedFrame(absl::optional<uint8_t> qp,
>+                                          int width,
>+                                          int height,
>+                                          int64_t now_ms,
>+                                          VideoCodecType codec) {
>+  if (num_frames_decoded_ == 0) {
>+    first_frame_decoded_ms_ = now_ms;
>+    last_unfreeze_time_ = now_ms;
>+  }
>+
>+  ++num_frames_decoded_;
>+
>+  if (!is_paused_ && num_frames_decoded_ > 1) {
>+    // Process inter-frame delay.
>+    int64_t interframe_delay_ms = now_ms - last_frame_decoded_ms_;
>+    interframe_delays_.Add(interframe_delay_ms);
>+    absl::optional<int> avg_interframe_delay =
>+        interframe_delays_.Avg(kMinFrameSamplesToDetectFreeze);
>+    // Check if it was a freeze.
>+    if (avg_interframe_delay &&
>+        interframe_delay_ms >=
>+            std::max(3 * *avg_interframe_delay,
>+                     *avg_interframe_delay + kMinIncreaseForFreezeMs)) {
>+      freezes_durations_.Add(interframe_delay_ms);
>+      smooth_playback_durations_.Add(last_frame_decoded_ms_ -
>+                                     last_unfreeze_time_);
>+      last_unfreeze_time_ = now_ms;
>+    } else {
>+      // Only count inter-frame delay as playback time if there
>+      // was no freeze.
>+      time_in_resolution_ms_[current_resolution_] += interframe_delay_ms;
>+      absl::optional<int> qp_blocky_threshold;
>+      // TODO(ilnik): add other codec types when we have QP for them.
>+      switch (codec) {
>+        case kVideoCodecVP8:
>+          qp_blocky_threshold = kBlockyQpThresholdVp8;
>+          break;
>+        case kVideoCodecVP9:
>+          qp_blocky_threshold = kBlockyQpThresholdVp9;
>+          break;
>+        default:
>+          qp_blocky_threshold = absl::nullopt;
>+      }
>+      if (qp_blocky_threshold && qp.value_or(0) > *qp_blocky_threshold) {
>+        time_in_blocky_video_ms_ += interframe_delay_ms;
>+      }
>+    }
>+  }
>+
>+  if (is_paused_) {
>+    // If the stream was paused since the previous frame, do not count the
>+    // pause toward smooth playback. Explicitly count the part before it and
>+    // start the new smooth playback interval from this frame.
>+    is_paused_ = false;
>+    if (last_frame_decoded_ms_ > last_unfreeze_time_) {
>+      smooth_playback_durations_.Add(last_frame_decoded_ms_ -
>+                                     last_unfreeze_time_);
>+    }
>+    last_unfreeze_time_ = now_ms;
>+  }
>+
>+  int64_t pixels = width * height;
>+  if (pixels >= kPixelsInHighResolution) {
>+    current_resolution_ = Resolution::High;
>+  } else if (pixels >= kPixelsInMediumResolution) {
>+    current_resolution_ = Resolution::Medium;
>+  } else {
>+    current_resolution_ = Resolution::Low;
>+  }
>+
>+  if (pixels < last_frame_pixels_) {
>+    ++num_resolution_downgrades_;
>+  }
>+
>+  last_frame_decoded_ms_ = now_ms;
>+  last_frame_qp_ = qp.value_or(0);
>+  last_frame_pixels_ = pixels;
>+}
>+
>+void VideoQualityObserver::OnStreamInactive() {
>+  is_paused_ = true;
>+}
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_quality_observer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_quality_observer.h
>new file mode 100644
>index 00000000000..8ba4af39842
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_quality_observer.h
>@@ -0,0 +1,79 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef VIDEO_VIDEO_QUALITY_OBSERVER_H_
>+#define VIDEO_VIDEO_QUALITY_OBSERVER_H_
>+
>+#include <stdint.h>
>+#include <vector>
>+
>+#include "absl/types/optional.h"
>+#include "api/video/video_content_type.h"
>+#include "common_types.h"  // NOLINT(build/include)
>+#include "rtc_base/numerics/sample_counter.h"
>+
>+namespace webrtc {
>+
>+// Calculates spatial and temporal quality metrics and reports them to UMA
>+// stats.
>+class VideoQualityObserver {
>+ public:
>+  // Use either VideoQualityObserver::kBlockyQpThresholdVp8 or
>+  // VideoQualityObserver::kBlockyQpThresholdVp9.
>+  explicit VideoQualityObserver(VideoContentType content_type);
>+  ~VideoQualityObserver();
>+
>+  void OnDecodedFrame(absl::optional<uint8_t> qp,
>+                      int width,
>+                      int height,
>+                      int64_t now_ms,
>+                      VideoCodecType codec);
>+
>+  void OnStreamInactive();
>+
>+ private:
>+  void UpdateHistograms();
>+
>+  enum Resolution {
>+    Low = 0,
>+    Medium = 1,
>+    High = 2,
>+  };
>+
>+  int64_t last_frame_decoded_ms_;
>+  int64_t num_frames_decoded_;
>+  int64_t first_frame_decoded_ms_;
>+  int64_t last_frame_pixels_;
>+  uint8_t last_frame_qp_;
>+  // Decoded timestamp of the last delayed frame.
>+  int64_t last_unfreeze_time_;
>+  rtc::SampleCounter interframe_delays_;
>+  // An inter-frame delay is counted as a freeze if it's significantly longer
>+  // than average inter-frame delay.
>+  rtc::SampleCounter freezes_durations_;
>+  // Time between freezes.
>+  rtc::SampleCounter smooth_playback_durations_;
>+  // Counters for time spent in different resolutions. Time between each two
>+  // Consecutive frames is counted to bin corresponding to the first frame
>+  // resolution.
>+  std::vector<int64_t> time_in_resolution_ms_;
>+  // Resolution of the last decoded frame. Resolution enum is used as an index.
>+  Resolution current_resolution_;
>+  int num_resolution_downgrades_;
>+  // Similar to resolution, time spent in high-QP video.
>+  int64_t time_in_blocky_video_ms_;
>+  // Content type of the last decoded frame.
>+  VideoContentType content_type_;
>+  bool is_paused_;
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // VIDEO_VIDEO_QUALITY_OBSERVER_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_quality_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_quality_test.cc
>index ec110ff694e..c5bd57e0eea 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_quality_test.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_quality_test.cc
>@@ -19,39 +19,19 @@
> 
> #include "logging/rtc_event_log/output/rtc_event_log_output_file.h"
> #include "media/engine/internalencoderfactory.h"
>+#include "media/engine/vp8_encoder_simulcast_proxy.h"
> #include "media/engine/webrtcvideoengine.h"
> #include "modules/audio_mixer/audio_mixer_impl.h"
>-#include "modules/rtp_rtcp/source/rtp_format.h"
>-#include "modules/rtp_rtcp/source/rtp_utility.h"
> #include "modules/video_coding/codecs/h264/include/h264.h"
>-#include "modules/video_coding/codecs/vp8/include/vp8_common_types.h"
>+#include "modules/video_coding/codecs/multiplex/include/multiplex_encoder_adapter.h"
>+#include "modules/video_coding/codecs/vp8/include/vp8.h"
> #include "modules/video_coding/codecs/vp9/include/vp9.h"
>-#include "rtc_base/cpu_time.h"
>-#include "rtc_base/flags.h"
>-#include "rtc_base/format_macros.h"
>-#include "rtc_base/memory_usage.h"
>-#include "rtc_base/pathutils.h"
>-#include "system_wrappers/include/cpu_info.h"
>-#include "test/rtp_file_writer.h"
> #include "test/run_loop.h"
>-#include "test/statistics.h"
> #include "test/testsupport/fileutils.h"
>-#include "test/testsupport/frame_writer.h"
>-#include "test/testsupport/perf_test.h"
>-#include "test/testsupport/test_artifacts.h"
>-#include "test/vcm_capturer.h"
> #include "test/video_renderer.h"
>-
>-DEFINE_bool(save_worst_frame,
>-            false,
>-            "Enable saving a frame with the lowest PSNR to a jpeg file in the "
>-            "test_artifacts_dir");
>+#include "video/video_analyzer.h"
> 
> namespace {
>-
>-constexpr int kSendStatsPollingIntervalMs = 1000;
>-
>-constexpr size_t kMaxComparisons = 10;
> constexpr char kSyncGroup[] = "av_sync";
> constexpr int kOpusMinBitrateBps = 6000;
> constexpr int kOpusBitrateFbBps = 32000;
>@@ -83,971 +63,30 @@ class VideoStreamFactory
> 
>   std::vector<webrtc::VideoStream> streams_;
> };
>-
>-bool IsFlexfec(int payload_type) {
>-  return payload_type == webrtc::VideoQualityTest::kFlexfecPayloadType;
>-}
>-
> }  // namespace
> 
> namespace webrtc {
> 
>-class VideoAnalyzer : public PacketReceiver,
>-                      public Transport,
>-                      public rtc::VideoSinkInterface<VideoFrame> {
>- public:
>-  VideoAnalyzer(test::LayerFilteringTransport* transport,
>-                const std::string& test_label,
>-                double avg_psnr_threshold,
>-                double avg_ssim_threshold,
>-                int duration_frames,
>-                FILE* graph_data_output_file,
>-                const std::string& graph_title,
>-                uint32_t ssrc_to_analyze,
>-                uint32_t rtx_ssrc_to_analyze,
>-                size_t selected_stream,
>-                int selected_sl,
>-                int selected_tl,
>-                bool is_quick_test_enabled,
>-                Clock* clock,
>-                std::string rtp_dump_name)
>-      : transport_(transport),
>-        receiver_(nullptr),
>-        call_(nullptr),
>-        send_stream_(nullptr),
>-        receive_stream_(nullptr),
>-        captured_frame_forwarder_(this, clock),
>-        test_label_(test_label),
>-        graph_data_output_file_(graph_data_output_file),
>-        graph_title_(graph_title),
>-        ssrc_to_analyze_(ssrc_to_analyze),
>-        rtx_ssrc_to_analyze_(rtx_ssrc_to_analyze),
>-        selected_stream_(selected_stream),
>-        selected_sl_(selected_sl),
>-        selected_tl_(selected_tl),
>-        pre_encode_proxy_(this),
>-        encode_timing_proxy_(this),
>-        last_fec_bytes_(0),
>-        frames_to_process_(duration_frames),
>-        frames_recorded_(0),
>-        frames_processed_(0),
>-        dropped_frames_(0),
>-        dropped_frames_before_first_encode_(0),
>-        dropped_frames_before_rendering_(0),
>-        last_render_time_(0),
>-        rtp_timestamp_delta_(0),
>-        total_media_bytes_(0),
>-        first_sending_time_(0),
>-        last_sending_time_(0),
>-        cpu_time_(0),
>-        wallclock_time_(0),
>-        avg_psnr_threshold_(avg_psnr_threshold),
>-        avg_ssim_threshold_(avg_ssim_threshold),
>-        is_quick_test_enabled_(is_quick_test_enabled),
>-        stats_polling_thread_(&PollStatsThread, this, "StatsPoller"),
>-        comparison_available_event_(false, false),
>-        done_(true, false),
>-        clock_(clock),
>-        start_ms_(clock->TimeInMilliseconds()) {
>-    // Create thread pool for CPU-expensive PSNR/SSIM calculations.
>-
>-    // Try to use about as many threads as cores, but leave kMinCoresLeft alone,
>-    // so that we don't accidentally starve "real" worker threads (codec etc).
>-    // Also, don't allocate more than kMaxComparisonThreads, even if there are
>-    // spare cores.
>-
>-    uint32_t num_cores = CpuInfo::DetectNumberOfCores();
>-    RTC_DCHECK_GE(num_cores, 1);
>-    static const uint32_t kMinCoresLeft = 4;
>-    static const uint32_t kMaxComparisonThreads = 8;
>-
>-    if (num_cores <= kMinCoresLeft) {
>-      num_cores = 1;
>-    } else {
>-      num_cores -= kMinCoresLeft;
>-      num_cores = std::min(num_cores, kMaxComparisonThreads);
>-    }
>-
>-    for (uint32_t i = 0; i < num_cores; ++i) {
>-      rtc::PlatformThread* thread =
>-          new rtc::PlatformThread(&FrameComparisonThread, this, "Analyzer");
>-      thread->Start();
>-      comparison_thread_pool_.push_back(thread);
>-    }
>-
>-    if (!rtp_dump_name.empty()) {
>-      fprintf(stdout, "Writing rtp dump to %s\n", rtp_dump_name.c_str());
>-      rtp_file_writer_.reset(test::RtpFileWriter::Create(
>-          test::RtpFileWriter::kRtpDump, rtp_dump_name));
>-    }
>-  }
>-
>-  ~VideoAnalyzer() {
>-    for (rtc::PlatformThread* thread : comparison_thread_pool_) {
>-      thread->Stop();
>-      delete thread;
>-    }
>-  }
>-
>-  virtual void SetReceiver(PacketReceiver* receiver) { receiver_ = receiver; }
>-
>-  void SetSource(test::VideoCapturer* video_capturer, bool respect_sink_wants) {
>-    if (respect_sink_wants)
>-      captured_frame_forwarder_.SetSource(video_capturer);
>-    rtc::VideoSinkWants wants;
>-    video_capturer->AddOrUpdateSink(InputInterface(), wants);
>-  }
>-
>-  void SetCall(Call* call) {
>-    rtc::CritScope lock(&crit_);
>-    RTC_DCHECK(!call_);
>-    call_ = call;
>-  }
>-
>-  void SetSendStream(VideoSendStream* stream) {
>-    rtc::CritScope lock(&crit_);
>-    RTC_DCHECK(!send_stream_);
>-    send_stream_ = stream;
>-  }
>-
>-  void SetReceiveStream(VideoReceiveStream* stream) {
>-    rtc::CritScope lock(&crit_);
>-    RTC_DCHECK(!receive_stream_);
>-    receive_stream_ = stream;
>-  }
>-
>-  rtc::VideoSinkInterface<VideoFrame>* InputInterface() {
>-    return &captured_frame_forwarder_;
>-  }
>-  rtc::VideoSourceInterface<VideoFrame>* OutputInterface() {
>-    return &captured_frame_forwarder_;
>-  }
>-
>-  DeliveryStatus DeliverPacket(MediaType media_type,
>-                               rtc::CopyOnWriteBuffer packet,
>-                               const PacketTime& packet_time) override {
>-    // Ignore timestamps of RTCP packets. They're not synchronized with
>-    // RTP packet timestamps and so they would confuse wrap_handler_.
>-    if (RtpHeaderParser::IsRtcp(packet.cdata(), packet.size())) {
>-      return receiver_->DeliverPacket(media_type, std::move(packet),
>-                                      packet_time);
>-    }
>-
>-    if (rtp_file_writer_) {
>-      test::RtpPacket p;
>-      memcpy(p.data, packet.cdata(), packet.size());
>-      p.length = packet.size();
>-      p.original_length = packet.size();
>-      p.time_ms = clock_->TimeInMilliseconds() - start_ms_;
>-      rtp_file_writer_->WritePacket(&p);
>-    }
>-
>-    RtpUtility::RtpHeaderParser parser(packet.cdata(), packet.size());
>-    RTPHeader header;
>-    parser.Parse(&header);
>-    if (!IsFlexfec(header.payloadType) &&
>-        (header.ssrc == ssrc_to_analyze_ ||
>-         header.ssrc == rtx_ssrc_to_analyze_)) {
>-      // Ignore FlexFEC timestamps, to avoid collisions with media timestamps.
>-      // (FlexFEC and media are sent on different SSRCs, which have different
>-      // timestamps spaces.)
>-      // Also ignore packets from wrong SSRC, but include retransmits.
>-      rtc::CritScope lock(&crit_);
>-      int64_t timestamp =
>-          wrap_handler_.Unwrap(header.timestamp - rtp_timestamp_delta_);
>-      recv_times_[timestamp] =
>-          Clock::GetRealTimeClock()->CurrentNtpInMilliseconds();
>-    }
>-
>-    return receiver_->DeliverPacket(media_type, std::move(packet), packet_time);
>-  }
>-
>-  void PreEncodeOnFrame(const VideoFrame& video_frame) {
>-    rtc::CritScope lock(&crit_);
>-    if (!first_encoded_timestamp_) {
>-      while (frames_.front().timestamp() != video_frame.timestamp()) {
>-        ++dropped_frames_before_first_encode_;
>-        frames_.pop_front();
>-        RTC_CHECK(!frames_.empty());
>-      }
>-      first_encoded_timestamp_ = video_frame.timestamp();
>-    }
>-  }
>-
>-  void PostEncodeFrameCallback(const EncodedFrame& encoded_frame) {
>-    rtc::CritScope lock(&crit_);
>-    if (!first_sent_timestamp_ &&
>-        encoded_frame.stream_id_ == selected_stream_) {
>-      first_sent_timestamp_ = encoded_frame.timestamp_;
>-    }
>-  }
>-
>-  bool SendRtp(const uint8_t* packet,
>-               size_t length,
>-               const PacketOptions& options) override {
>-    RtpUtility::RtpHeaderParser parser(packet, length);
>-    RTPHeader header;
>-    parser.Parse(&header);
>-
>-    int64_t current_time =
>-        Clock::GetRealTimeClock()->CurrentNtpInMilliseconds();
>-
>-    bool result = transport_->SendRtp(packet, length, options);
>-    {
>-      rtc::CritScope lock(&crit_);
>-      if (rtp_timestamp_delta_ == 0 && header.ssrc == ssrc_to_analyze_) {
>-        RTC_CHECK(static_cast<bool>(first_sent_timestamp_));
>-        rtp_timestamp_delta_ = header.timestamp - *first_sent_timestamp_;
>-      }
>-
>-      if (!IsFlexfec(header.payloadType) && header.ssrc == ssrc_to_analyze_) {
>-        // Ignore FlexFEC timestamps, to avoid collisions with media timestamps.
>-        // (FlexFEC and media are sent on different SSRCs, which have different
>-        // timestamps spaces.)
>-        // Also ignore packets from wrong SSRC and retransmits.
>-        int64_t timestamp =
>-            wrap_handler_.Unwrap(header.timestamp - rtp_timestamp_delta_);
>-        send_times_[timestamp] = current_time;
>-
>-        if (IsInSelectedSpatialAndTemporalLayer(packet, length, header)) {
>-          encoded_frame_sizes_[timestamp] +=
>-              length - (header.headerLength + header.paddingLength);
>-          total_media_bytes_ +=
>-              length - (header.headerLength + header.paddingLength);
>-        }
>-        if (first_sending_time_ == 0)
>-          first_sending_time_ = current_time;
>-        last_sending_time_ = current_time;
>-      }
>-    }
>-    return result;
>-  }
>-
>-  bool SendRtcp(const uint8_t* packet, size_t length) override {
>-    return transport_->SendRtcp(packet, length);
>-  }
>-
>-  void OnFrame(const VideoFrame& video_frame) override {
>-    int64_t render_time_ms =
>-        Clock::GetRealTimeClock()->CurrentNtpInMilliseconds();
>-
>-    rtc::CritScope lock(&crit_);
>-
>-    StartExcludingCpuThreadTime();
>-
>-    int64_t send_timestamp =
>-        wrap_handler_.Unwrap(video_frame.timestamp() - rtp_timestamp_delta_);
>-
>-    while (wrap_handler_.Unwrap(frames_.front().timestamp()) < send_timestamp) {
>-      if (!last_rendered_frame_) {
>-        // No previous frame rendered, this one was dropped after sending but
>-        // before rendering.
>-        ++dropped_frames_before_rendering_;
>-      } else {
>-        AddFrameComparison(frames_.front(), *last_rendered_frame_, true,
>-                           render_time_ms);
>-      }
>-      frames_.pop_front();
>-      RTC_DCHECK(!frames_.empty());
>-    }
>-
>-    VideoFrame reference_frame = frames_.front();
>-    frames_.pop_front();
>-    int64_t reference_timestamp =
>-        wrap_handler_.Unwrap(reference_frame.timestamp());
>-    if (send_timestamp == reference_timestamp - 1) {
>-      // TODO(ivica): Make this work for > 2 streams.
>-      // Look at RTPSender::BuildRTPHeader.
>-      ++send_timestamp;
>-    }
>-    ASSERT_EQ(reference_timestamp, send_timestamp);
>-
>-    AddFrameComparison(reference_frame, video_frame, false, render_time_ms);
>-
>-    last_rendered_frame_ = video_frame;
>-
>-    StopExcludingCpuThreadTime();
>-  }
>-
>-  void Wait() {
>-    // Frame comparisons can be very expensive. Wait for test to be done, but
>-    // at time-out check if frames_processed is going up. If so, give it more
>-    // time, otherwise fail. Hopefully this will reduce test flakiness.
>-
>-    stats_polling_thread_.Start();
>-
>-    int last_frames_processed = -1;
>-    int iteration = 0;
>-    while (!done_.Wait(VideoQualityTest::kDefaultTimeoutMs)) {
>-      int frames_processed;
>-      {
>-        rtc::CritScope crit(&comparison_lock_);
>-        frames_processed = frames_processed_;
>-      }
>-
>-      // Print some output so test infrastructure won't think we've crashed.
>-      const char* kKeepAliveMessages[3] = {
>-          "Uh, I'm-I'm not quite dead, sir.",
>-          "Uh, I-I think uh, I could pull through, sir.",
>-          "Actually, I think I'm all right to come with you--"};
>-      printf("- %s\n", kKeepAliveMessages[iteration++ % 3]);
>-
>-      if (last_frames_processed == -1) {
>-        last_frames_processed = frames_processed;
>-        continue;
>-      }
>-      if (frames_processed == last_frames_processed) {
>-        EXPECT_GT(frames_processed, last_frames_processed)
>-            << "Analyzer stalled while waiting for test to finish.";
>-        done_.Set();
>-        break;
>-      }
>-      last_frames_processed = frames_processed;
>-    }
>-
>-    if (iteration > 0)
>-      printf("- Farewell, sweet Concorde!\n");
>-
>-    stats_polling_thread_.Stop();
>-  }
>-
>-  rtc::VideoSinkInterface<VideoFrame>* pre_encode_proxy() {
>-    return &pre_encode_proxy_;
>-  }
>-  EncodedFrameObserver* encode_timing_proxy() { return &encode_timing_proxy_; }
>-
>-  void StartMeasuringCpuProcessTime() {
>-    rtc::CritScope lock(&cpu_measurement_lock_);
>-    cpu_time_ -= rtc::GetProcessCpuTimeNanos();
>-    wallclock_time_ -= rtc::SystemTimeNanos();
>-  }
>-
>-  void StopMeasuringCpuProcessTime() {
>-    rtc::CritScope lock(&cpu_measurement_lock_);
>-    cpu_time_ += rtc::GetProcessCpuTimeNanos();
>-    wallclock_time_ += rtc::SystemTimeNanos();
>-  }
>-
>-  void StartExcludingCpuThreadTime() {
>-    rtc::CritScope lock(&cpu_measurement_lock_);
>-    cpu_time_ += rtc::GetThreadCpuTimeNanos();
>-  }
>-
>-  void StopExcludingCpuThreadTime() {
>-    rtc::CritScope lock(&cpu_measurement_lock_);
>-    cpu_time_ -= rtc::GetThreadCpuTimeNanos();
>-  }
>-
>-  double GetCpuUsagePercent() {
>-    rtc::CritScope lock(&cpu_measurement_lock_);
>-    return static_cast<double>(cpu_time_) / wallclock_time_ * 100.0;
>-  }
>-
>-  test::LayerFilteringTransport* const transport_;
>-  PacketReceiver* receiver_;
>-
>- private:
>-  struct FrameComparison {
>-    FrameComparison()
>-        : dropped(false),
>-          input_time_ms(0),
>-          send_time_ms(0),
>-          recv_time_ms(0),
>-          render_time_ms(0),
>-          encoded_frame_size(0) {}
>-
>-    FrameComparison(const VideoFrame& reference,
>-                    const VideoFrame& render,
>-                    bool dropped,
>-                    int64_t input_time_ms,
>-                    int64_t send_time_ms,
>-                    int64_t recv_time_ms,
>-                    int64_t render_time_ms,
>-                    size_t encoded_frame_size)
>-        : reference(reference),
>-          render(render),
>-          dropped(dropped),
>-          input_time_ms(input_time_ms),
>-          send_time_ms(send_time_ms),
>-          recv_time_ms(recv_time_ms),
>-          render_time_ms(render_time_ms),
>-          encoded_frame_size(encoded_frame_size) {}
>-
>-    FrameComparison(bool dropped,
>-                    int64_t input_time_ms,
>-                    int64_t send_time_ms,
>-                    int64_t recv_time_ms,
>-                    int64_t render_time_ms,
>-                    size_t encoded_frame_size)
>-        : dropped(dropped),
>-          input_time_ms(input_time_ms),
>-          send_time_ms(send_time_ms),
>-          recv_time_ms(recv_time_ms),
>-          render_time_ms(render_time_ms),
>-          encoded_frame_size(encoded_frame_size) {}
>-
>-    rtc::Optional<VideoFrame> reference;
>-    rtc::Optional<VideoFrame> render;
>-    bool dropped;
>-    int64_t input_time_ms;
>-    int64_t send_time_ms;
>-    int64_t recv_time_ms;
>-    int64_t render_time_ms;
>-    size_t encoded_frame_size;
>-  };
>-
>-  struct Sample {
>-    Sample(int dropped,
>-           int64_t input_time_ms,
>-           int64_t send_time_ms,
>-           int64_t recv_time_ms,
>-           int64_t render_time_ms,
>-           size_t encoded_frame_size,
>-           double psnr,
>-           double ssim)
>-        : dropped(dropped),
>-          input_time_ms(input_time_ms),
>-          send_time_ms(send_time_ms),
>-          recv_time_ms(recv_time_ms),
>-          render_time_ms(render_time_ms),
>-          encoded_frame_size(encoded_frame_size),
>-          psnr(psnr),
>-          ssim(ssim) {}
>-
>-    int dropped;
>-    int64_t input_time_ms;
>-    int64_t send_time_ms;
>-    int64_t recv_time_ms;
>-    int64_t render_time_ms;
>-    size_t encoded_frame_size;
>-    double psnr;
>-    double ssim;
>-  };
>-
>-  // This class receives the send-side OnEncodeTiming and is provided to not
>-  // conflict with the receiver-side pre_decode_callback.
>-  class OnEncodeTimingProxy : public EncodedFrameObserver {
>-   public:
>-    explicit OnEncodeTimingProxy(VideoAnalyzer* parent) : parent_(parent) {}
>-
>-    void EncodedFrameCallback(const EncodedFrame& frame) override {
>-      parent_->PostEncodeFrameCallback(frame);
>-    }
>-
>-   private:
>-    VideoAnalyzer* const parent_;
>-  };
>-
>-  // This class receives the send-side OnFrame callback and is provided to not
>-  // conflict with the receiver-side renderer callback.
>-  class PreEncodeProxy : public rtc::VideoSinkInterface<VideoFrame> {
>-   public:
>-    explicit PreEncodeProxy(VideoAnalyzer* parent) : parent_(parent) {}
>-
>-    void OnFrame(const VideoFrame& video_frame) override {
>-      parent_->PreEncodeOnFrame(video_frame);
>-    }
>-
>-   private:
>-    VideoAnalyzer* const parent_;
>-  };
>-
>-  bool IsInSelectedSpatialAndTemporalLayer(const uint8_t* packet,
>-                                           size_t length,
>-                                           const RTPHeader& header) {
>-    if (header.payloadType != test::CallTest::kPayloadTypeVP9 &&
>-        header.payloadType != test::CallTest::kPayloadTypeVP8) {
>-      return true;
>-    } else {
>-      // Get VP8 and VP9 specific header to check layers indexes.
>-      const uint8_t* payload = packet + header.headerLength;
>-      const size_t payload_length = length - header.headerLength;
>-      const size_t payload_data_length = payload_length - header.paddingLength;
>-      const bool is_vp8 = header.payloadType == test::CallTest::kPayloadTypeVP8;
>-      std::unique_ptr<RtpDepacketizer> depacketizer(
>-          RtpDepacketizer::Create(is_vp8 ? kRtpVideoVp8 : kRtpVideoVp9));
>-      RtpDepacketizer::ParsedPayload parsed_payload;
>-      bool result =
>-          depacketizer->Parse(&parsed_payload, payload, payload_data_length);
>-      RTC_DCHECK(result);
>-      const int temporal_idx = static_cast<int>(
>-          is_vp8 ? parsed_payload.type.Video.codecHeader.VP8.temporalIdx
>-                 : parsed_payload.type.Video.codecHeader.VP9.temporal_idx);
>-      const int spatial_idx = static_cast<int>(
>-          is_vp8 ? kNoSpatialIdx
>-                 : parsed_payload.type.Video.codecHeader.VP9.spatial_idx);
>-      return (selected_tl_ < 0 || temporal_idx == kNoTemporalIdx ||
>-              temporal_idx <= selected_tl_) &&
>-             (selected_sl_ < 0 || spatial_idx == kNoSpatialIdx ||
>-              spatial_idx <= selected_sl_);
>-    }
>-  }
>-
>-  void AddFrameComparison(const VideoFrame& reference,
>-                          const VideoFrame& render,
>-                          bool dropped,
>-                          int64_t render_time_ms)
>-      RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_) {
>-    int64_t reference_timestamp = wrap_handler_.Unwrap(reference.timestamp());
>-    int64_t send_time_ms = send_times_[reference_timestamp];
>-    send_times_.erase(reference_timestamp);
>-    int64_t recv_time_ms = recv_times_[reference_timestamp];
>-    recv_times_.erase(reference_timestamp);
>-
>-    // TODO(ivica): Make this work for > 2 streams.
>-    auto it = encoded_frame_sizes_.find(reference_timestamp);
>-    if (it == encoded_frame_sizes_.end())
>-      it = encoded_frame_sizes_.find(reference_timestamp - 1);
>-    size_t encoded_size = it == encoded_frame_sizes_.end() ? 0 : it->second;
>-    if (it != encoded_frame_sizes_.end())
>-      encoded_frame_sizes_.erase(it);
>-
>-    rtc::CritScope crit(&comparison_lock_);
>-    if (comparisons_.size() < kMaxComparisons) {
>-      comparisons_.push_back(FrameComparison(reference, render, dropped,
>-                                             reference.ntp_time_ms(),
>-                                             send_time_ms, recv_time_ms,
>-                                             render_time_ms, encoded_size));
>-    } else {
>-      comparisons_.push_back(FrameComparison(dropped,
>-                                             reference.ntp_time_ms(),
>-                                             send_time_ms, recv_time_ms,
>-                                             render_time_ms, encoded_size));
>-    }
>-    comparison_available_event_.Set();
>-  }
>-
>-  static void PollStatsThread(void* obj) {
>-    static_cast<VideoAnalyzer*>(obj)->PollStats();
>-  }
>-
>-  void PollStats() {
>-    while (!done_.Wait(kSendStatsPollingIntervalMs)) {
>-      rtc::CritScope crit(&comparison_lock_);
>-
>-      Call::Stats call_stats = call_->GetStats();
>-      send_bandwidth_bps_.AddSample(call_stats.send_bandwidth_bps);
>-
>-      VideoSendStream::Stats send_stats = send_stream_->GetStats();
>-      // It's not certain that we yet have estimates for any of these stats.
>-      // Check that they are positive before mixing them in.
>-      if (send_stats.encode_frame_rate > 0)
>-        encode_frame_rate_.AddSample(send_stats.encode_frame_rate);
>-      if (send_stats.avg_encode_time_ms > 0)
>-        encode_time_ms_.AddSample(send_stats.avg_encode_time_ms);
>-      if (send_stats.encode_usage_percent > 0)
>-        encode_usage_percent_.AddSample(send_stats.encode_usage_percent);
>-      if (send_stats.media_bitrate_bps > 0)
>-        media_bitrate_bps_.AddSample(send_stats.media_bitrate_bps);
>-      size_t fec_bytes = 0;
>-      for (auto kv : send_stats.substreams) {
>-        fec_bytes += kv.second.rtp_stats.fec.payload_bytes +
>-                     kv.second.rtp_stats.fec.padding_bytes;
>-      }
>-      fec_bitrate_bps_.AddSample((fec_bytes - last_fec_bytes_) * 8);
>-      last_fec_bytes_ = fec_bytes;
>-
>-      if (receive_stream_ != nullptr) {
>-        VideoReceiveStream::Stats receive_stats = receive_stream_->GetStats();
>-        if (receive_stats.decode_ms > 0)
>-          decode_time_ms_.AddSample(receive_stats.decode_ms);
>-        if (receive_stats.max_decode_ms > 0)
>-          decode_time_max_ms_.AddSample(receive_stats.max_decode_ms);
>-      }
>-
>-      memory_usage_.AddSample(rtc::GetProcessResidentSizeBytes());
>-    }
>-  }
>-
>-  static bool FrameComparisonThread(void* obj) {
>-    return static_cast<VideoAnalyzer*>(obj)->CompareFrames();
>-  }
>-
>-  bool CompareFrames() {
>-    if (AllFramesRecorded())
>-      return false;
>-
>-    FrameComparison comparison;
>-
>-    if (!PopComparison(&comparison)) {
>-      // Wait until new comparison task is available, or test is done.
>-      // If done, wake up remaining threads waiting.
>-      comparison_available_event_.Wait(1000);
>-      if (AllFramesRecorded()) {
>-        comparison_available_event_.Set();
>-        return false;
>-      }
>-      return true;  // Try again.
>-    }
>-
>-    StartExcludingCpuThreadTime();
>-
>-    PerformFrameComparison(comparison);
>-
>-    StopExcludingCpuThreadTime();
>-
>-    if (FrameProcessed()) {
>-      PrintResults();
>-      if (graph_data_output_file_)
>-        PrintSamplesToFile();
>-      done_.Set();
>-      comparison_available_event_.Set();
>-      return false;
>-    }
>-
>-    return true;
>-  }
>-
>-  bool PopComparison(FrameComparison* comparison) {
>-    rtc::CritScope crit(&comparison_lock_);
>-    // If AllFramesRecorded() is true, it means we have already popped
>-    // frames_to_process_ frames from comparisons_, so there is no more work
>-    // for this thread to be done. frames_processed_ might still be lower if
>-    // all comparisons are not done, but those frames are currently being
>-    // worked on by other threads.
>-    if (comparisons_.empty() || AllFramesRecorded())
>-      return false;
>-
>-    *comparison = comparisons_.front();
>-    comparisons_.pop_front();
>-
>-    FrameRecorded();
>-    return true;
>-  }
>-
>-  // Increment counter for number of frames received for comparison.
>-  void FrameRecorded() {
>-    rtc::CritScope crit(&comparison_lock_);
>-    ++frames_recorded_;
>-  }
>-
>-  // Returns true if all frames to be compared have been taken from the queue.
>-  bool AllFramesRecorded() {
>-    rtc::CritScope crit(&comparison_lock_);
>-    assert(frames_recorded_ <= frames_to_process_);
>-    return frames_recorded_ == frames_to_process_;
>-  }
>-
>-  // Increase count of number of frames processed. Returns true if this was the
>-  // last frame to be processed.
>-  bool FrameProcessed() {
>-    rtc::CritScope crit(&comparison_lock_);
>-    ++frames_processed_;
>-    assert(frames_processed_ <= frames_to_process_);
>-    return frames_processed_ == frames_to_process_;
>-  }
>-
>-  void PrintResults() {
>-    StopMeasuringCpuProcessTime();
>-    rtc::CritScope crit(&comparison_lock_);
>-    PrintResult("psnr", psnr_, " dB");
>-    PrintResult("ssim", ssim_, " score");
>-    PrintResult("sender_time", sender_time_, " ms");
>-    PrintResult("receiver_time", receiver_time_, " ms");
>-    PrintResult("total_delay_incl_network", end_to_end_, " ms");
>-    PrintResult("time_between_rendered_frames", rendered_delta_, " ms");
>-    PrintResult("encode_frame_rate", encode_frame_rate_, " fps");
>-    PrintResult("encode_time", encode_time_ms_, " ms");
>-    PrintResult("media_bitrate", media_bitrate_bps_, " bps");
>-    PrintResult("fec_bitrate", fec_bitrate_bps_, " bps");
>-    PrintResult("send_bandwidth", send_bandwidth_bps_, " bps");
>-
>-    if (worst_frame_) {
>-      test::PrintResult("min_psnr", "", test_label_.c_str(), worst_frame_->psnr,
>-                  "dB", false);
>-    }
>-
>-    if (receive_stream_ != nullptr) {
>-      PrintResult("decode_time", decode_time_ms_, " ms");
>-    }
>-
>-    test::PrintResult("dropped_frames", "", test_label_.c_str(),
>-                  dropped_frames_, "frames", false);
>-    test::PrintResult("cpu_usage", "", test_label_.c_str(),
>-                      GetCpuUsagePercent(), "%", false);
>-
>-#if defined(WEBRTC_WIN)
>-      // On Linux and Mac in Resident Set some unused pages may be counted.
>-      // Therefore this metric will depend on order in which tests are run and
>-      // will be flaky.
>-    PrintResult("memory_usage", memory_usage_, " bytes");
>-#endif
>-
>-    // Saving only the worst frame for manual analysis. Intention here is to
>-    // only detect video corruptions and not to track picture quality. Thus,
>-    // jpeg is used here.
>-    if (FLAG_save_worst_frame && worst_frame_) {
>-      std::string output_dir;
>-      test::GetTestArtifactsDir(&output_dir);
>-      std::string output_path =
>-          rtc::Pathname(output_dir, test_label_ + ".jpg").pathname();
>-      RTC_LOG(LS_INFO) << "Saving worst frame to " << output_path;
>-      test::JpegFrameWriter frame_writer(output_path);
>-      RTC_CHECK(frame_writer.WriteFrame(worst_frame_->frame,
>-                                        100 /*best quality*/));
>-    }
>-
>-    //  Disable quality check for quick test, as quality checks may fail
>-    //  because too few samples were collected.
>-    if (!is_quick_test_enabled_) {
>-      EXPECT_GT(psnr_.Mean(), avg_psnr_threshold_);
>-      EXPECT_GT(ssim_.Mean(), avg_ssim_threshold_);
>-    }
>-  }
>-
>-  void PerformFrameComparison(const FrameComparison& comparison) {
>-    // Perform expensive psnr and ssim calculations while not holding lock.
>-    double psnr = -1.0;
>-    double ssim = -1.0;
>-    if (comparison.reference && !comparison.dropped) {
>-      psnr = I420PSNR(&*comparison.reference, &*comparison.render);
>-      ssim = I420SSIM(&*comparison.reference, &*comparison.render);
>-    }
>-
>-    rtc::CritScope crit(&comparison_lock_);
>-
>-    if (psnr >= 0.0 && (!worst_frame_ || worst_frame_->psnr > psnr)) {
>-      worst_frame_.emplace(FrameWithPsnr{psnr, *comparison.render});
>-    }
>-
>-    if (graph_data_output_file_) {
>-      samples_.push_back(Sample(
>-          comparison.dropped, comparison.input_time_ms, comparison.send_time_ms,
>-          comparison.recv_time_ms, comparison.render_time_ms,
>-          comparison.encoded_frame_size, psnr, ssim));
>-    }
>-    if (psnr >= 0.0)
>-      psnr_.AddSample(psnr);
>-    if (ssim >= 0.0)
>-      ssim_.AddSample(ssim);
>-
>-    if (comparison.dropped) {
>-      ++dropped_frames_;
>-      return;
>-    }
>-    if (last_render_time_ != 0)
>-      rendered_delta_.AddSample(comparison.render_time_ms - last_render_time_);
>-    last_render_time_ = comparison.render_time_ms;
>-
>-    sender_time_.AddSample(comparison.send_time_ms - comparison.input_time_ms);
>-    if (comparison.recv_time_ms > 0) {
>-      // If recv_time_ms == 0, this frame consisted of a packets which were all
>-      // lost in the transport. Since we were able to render the frame, however,
>-      // the dropped packets were recovered by FlexFEC. The FlexFEC recovery
>-      // happens internally in Call, and we can therefore here not know which
>-      // FEC packets that protected the lost media packets. Consequently, we
>-      // were not able to record a meaningful recv_time_ms. We therefore skip
>-      // this sample.
>-      //
>-      // The reasoning above does not hold for ULPFEC and RTX, as for those
>-      // strategies the timestamp of the received packets is set to the
>-      // timestamp of the protected/retransmitted media packet. I.e., then
>-      // recv_time_ms != 0, even though the media packets were lost.
>-      receiver_time_.AddSample(comparison.render_time_ms -
>-                               comparison.recv_time_ms);
>-    }
>-    end_to_end_.AddSample(comparison.render_time_ms - comparison.input_time_ms);
>-    encoded_frame_size_.AddSample(comparison.encoded_frame_size);
>-  }
>-
>-  void PrintResult(const char* result_type,
>-                   test::Statistics stats,
>-                   const char* unit) {
>-    test::PrintResultMeanAndError(result_type, "", test_label_.c_str(),
>-                                  stats.Mean(), stats.StandardDeviation(), unit,
>-                                  false);
>-  }
>-
>-  void PrintSamplesToFile(void) {
>-    FILE* out = graph_data_output_file_;
>-    rtc::CritScope crit(&comparison_lock_);
>-    std::sort(samples_.begin(), samples_.end(),
>-              [](const Sample& A, const Sample& B) -> bool {
>-                return A.input_time_ms < B.input_time_ms;
>-              });
>-
>-    fprintf(out, "%s\n", graph_title_.c_str());
>-    fprintf(out, "%" PRIuS "\n", samples_.size());
>-    fprintf(out,
>-            "dropped "
>-            "input_time_ms "
>-            "send_time_ms "
>-            "recv_time_ms "
>-            "render_time_ms "
>-            "encoded_frame_size "
>-            "psnr "
>-            "ssim "
>-            "encode_time_ms\n");
>-    for (const Sample& sample : samples_) {
>-      fprintf(out, "%d %" PRId64 " %" PRId64 " %" PRId64 " %" PRId64 " %" PRIuS
>-                   " %lf %lf\n",
>-              sample.dropped, sample.input_time_ms, sample.send_time_ms,
>-              sample.recv_time_ms, sample.render_time_ms,
>-              sample.encoded_frame_size, sample.psnr, sample.ssim);
>-    }
>-  }
>-
>-  double GetAverageMediaBitrateBps() {
>-    if (last_sending_time_ == first_sending_time_) {
>-      return 0;
>-    } else {
>-      return static_cast<double>(total_media_bytes_) * 8 /
>-             (last_sending_time_ - first_sending_time_) *
>-             rtc::kNumMillisecsPerSec;
>-    }
>-  }
>-
>-  // Implements VideoSinkInterface to receive captured frames from a
>-  // FrameGeneratorCapturer. Implements VideoSourceInterface to be able to act
>-  // as a source to VideoSendStream.
>-  // It forwards all input frames to the VideoAnalyzer for later comparison and
>-  // forwards the captured frames to the VideoSendStream.
>-  class CapturedFrameForwarder : public rtc::VideoSinkInterface<VideoFrame>,
>-                                 public rtc::VideoSourceInterface<VideoFrame> {
>-   public:
>-    explicit CapturedFrameForwarder(VideoAnalyzer* analyzer, Clock* clock)
>-        : analyzer_(analyzer),
>-          send_stream_input_(nullptr),
>-          video_capturer_(nullptr),
>-          clock_(clock) {}
>-
>-    void SetSource(test::VideoCapturer* video_capturer) {
>-      video_capturer_ = video_capturer;
>-    }
>-
>-   private:
>-    void OnFrame(const VideoFrame& video_frame) override {
>-      VideoFrame copy = video_frame;
>-      // Frames from the capturer does not have a rtp timestamp.
>-      // Create one so it can be used for comparison.
>-      RTC_DCHECK_EQ(0, video_frame.timestamp());
>-      if (video_frame.ntp_time_ms() == 0)
>-        copy.set_ntp_time_ms(clock_->CurrentNtpInMilliseconds());
>-      copy.set_timestamp(copy.ntp_time_ms() * 90);
>-      analyzer_->AddCapturedFrameForComparison(copy);
>-      rtc::CritScope lock(&crit_);
>-      if (send_stream_input_)
>-        send_stream_input_->OnFrame(copy);
>-    }
>-
>-    // Called when |send_stream_.SetSource()| is called.
>-    void AddOrUpdateSink(rtc::VideoSinkInterface<VideoFrame>* sink,
>-                         const rtc::VideoSinkWants& wants) override {
>-      {
>-        rtc::CritScope lock(&crit_);
>-        RTC_DCHECK(!send_stream_input_ || send_stream_input_ == sink);
>-        send_stream_input_ = sink;
>-      }
>-      if (video_capturer_) {
>-        video_capturer_->AddOrUpdateSink(this, wants);
>-      }
>-    }
>-
>-    // Called by |send_stream_| when |send_stream_.SetSource()| is called.
>-    void RemoveSink(rtc::VideoSinkInterface<VideoFrame>* sink) override {
>-      rtc::CritScope lock(&crit_);
>-      RTC_DCHECK(sink == send_stream_input_);
>-      send_stream_input_ = nullptr;
>-    }
>-
>-    VideoAnalyzer* const analyzer_;
>-    rtc::CriticalSection crit_;
>-    rtc::VideoSinkInterface<VideoFrame>* send_stream_input_
>-        RTC_GUARDED_BY(crit_);
>-    test::VideoCapturer* video_capturer_;
>-    Clock* clock_;
>-  };
>-
>-  void AddCapturedFrameForComparison(const VideoFrame& video_frame) {
>-    rtc::CritScope lock(&crit_);
>-    frames_.push_back(video_frame);
>+std::unique_ptr<VideoEncoder> VideoQualityTest::CreateVideoEncoder(
>+    const SdpVideoFormat& format) {
>+  if (format.name == "VP8") {
>+    return absl::make_unique<VP8EncoderSimulcastProxy>(
>+        &internal_encoder_factory_, format);
>+  } else if (format.name == "multiplex") {
>+    return absl::make_unique<MultiplexEncoderAdapter>(
>+        &internal_encoder_factory_, SdpVideoFormat(cricket::kVp9CodecName));
>   }
>+  return internal_encoder_factory_.CreateVideoEncoder(format);
>+}
> 
>-  Call* call_;
>-  VideoSendStream* send_stream_;
>-  VideoReceiveStream* receive_stream_;
>-  CapturedFrameForwarder captured_frame_forwarder_;
>-  const std::string test_label_;
>-  FILE* const graph_data_output_file_;
>-  const std::string graph_title_;
>-  const uint32_t ssrc_to_analyze_;
>-  const uint32_t rtx_ssrc_to_analyze_;
>-  const size_t selected_stream_;
>-  const int selected_sl_;
>-  const int selected_tl_;
>-  PreEncodeProxy pre_encode_proxy_;
>-  OnEncodeTimingProxy encode_timing_proxy_;
>-  std::vector<Sample> samples_ RTC_GUARDED_BY(comparison_lock_);
>-  test::Statistics sender_time_ RTC_GUARDED_BY(comparison_lock_);
>-  test::Statistics receiver_time_ RTC_GUARDED_BY(comparison_lock_);
>-  test::Statistics psnr_ RTC_GUARDED_BY(comparison_lock_);
>-  test::Statistics ssim_ RTC_GUARDED_BY(comparison_lock_);
>-  test::Statistics end_to_end_ RTC_GUARDED_BY(comparison_lock_);
>-  test::Statistics rendered_delta_ RTC_GUARDED_BY(comparison_lock_);
>-  test::Statistics encoded_frame_size_ RTC_GUARDED_BY(comparison_lock_);
>-  test::Statistics encode_frame_rate_ RTC_GUARDED_BY(comparison_lock_);
>-  test::Statistics encode_time_ms_ RTC_GUARDED_BY(comparison_lock_);
>-  test::Statistics encode_usage_percent_ RTC_GUARDED_BY(comparison_lock_);
>-  test::Statistics decode_time_ms_ RTC_GUARDED_BY(comparison_lock_);
>-  test::Statistics decode_time_max_ms_ RTC_GUARDED_BY(comparison_lock_);
>-  test::Statistics media_bitrate_bps_ RTC_GUARDED_BY(comparison_lock_);
>-  test::Statistics fec_bitrate_bps_ RTC_GUARDED_BY(comparison_lock_);
>-  test::Statistics send_bandwidth_bps_ RTC_GUARDED_BY(comparison_lock_);
>-  test::Statistics memory_usage_ RTC_GUARDED_BY(comparison_lock_);
>-
>-  struct FrameWithPsnr {
>-    double psnr;
>-    VideoFrame frame;
>-  };
>-
>-  // Rendered frame with worst PSNR is saved for further analysis.
>-  rtc::Optional<FrameWithPsnr> worst_frame_ RTC_GUARDED_BY(comparison_lock_);
>-
>-  size_t last_fec_bytes_;
>-
>-  const int frames_to_process_;
>-  int frames_recorded_;
>-  int frames_processed_;
>-  int dropped_frames_;
>-  int dropped_frames_before_first_encode_;
>-  int dropped_frames_before_rendering_;
>-  int64_t last_render_time_;
>-  uint32_t rtp_timestamp_delta_;
>-  int64_t total_media_bytes_;
>-  int64_t first_sending_time_;
>-  int64_t last_sending_time_;
>-
>-  int64_t cpu_time_ RTC_GUARDED_BY(cpu_measurement_lock_);
>-  int64_t wallclock_time_ RTC_GUARDED_BY(cpu_measurement_lock_);
>-  rtc::CriticalSection cpu_measurement_lock_;
>-
>-  rtc::CriticalSection crit_;
>-  std::deque<VideoFrame> frames_ RTC_GUARDED_BY(crit_);
>-  rtc::Optional<VideoFrame> last_rendered_frame_ RTC_GUARDED_BY(crit_);
>-  rtc::TimestampWrapAroundHandler wrap_handler_ RTC_GUARDED_BY(crit_);
>-  std::map<int64_t, int64_t> send_times_ RTC_GUARDED_BY(crit_);
>-  std::map<int64_t, int64_t> recv_times_ RTC_GUARDED_BY(crit_);
>-  std::map<int64_t, size_t> encoded_frame_sizes_ RTC_GUARDED_BY(crit_);
>-  rtc::Optional<uint32_t> first_encoded_timestamp_ RTC_GUARDED_BY(crit_);
>-  rtc::Optional<uint32_t> first_sent_timestamp_ RTC_GUARDED_BY(crit_);
>-  const double avg_psnr_threshold_;
>-  const double avg_ssim_threshold_;
>-  bool is_quick_test_enabled_;
>-
>-  rtc::CriticalSection comparison_lock_;
>-  std::vector<rtc::PlatformThread*> comparison_thread_pool_;
>-  rtc::PlatformThread stats_polling_thread_;
>-  rtc::Event comparison_available_event_;
>-  std::deque<FrameComparison> comparisons_ RTC_GUARDED_BY(comparison_lock_);
>-  rtc::Event done_;
>-
>-  std::unique_ptr<test::RtpFileWriter> rtp_file_writer_;
>-  Clock* const clock_;
>-  const int64_t start_ms_;
>-};
>-
>-VideoQualityTest::VideoQualityTest()
>-    : clock_(Clock::GetRealTimeClock()), receive_logs_(0), send_logs_(0) {
>+VideoQualityTest::VideoQualityTest(
>+    std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory)
>+    : clock_(Clock::GetRealTimeClock()),
>+      video_encoder_factory_([this](const SdpVideoFormat& format) {
>+        return this->CreateVideoEncoder(format);
>+      }),
>+      receive_logs_(0),
>+      send_logs_(0) {
>   payload_type_map_ = test::CallTest::payload_type_map_;
>   RTC_DCHECK(payload_type_map_.find(kPayloadTypeH264) ==
>              payload_type_map_.end());
>@@ -1058,21 +97,25 @@ VideoQualityTest::VideoQualityTest()
>   payload_type_map_[kPayloadTypeH264] = webrtc::MediaType::VIDEO;
>   payload_type_map_[kPayloadTypeVP8] = webrtc::MediaType::VIDEO;
>   payload_type_map_[kPayloadTypeVP9] = webrtc::MediaType::VIDEO;
>+
>+  fec_controller_factory_ = std::move(fec_controller_factory);
> }
> 
> VideoQualityTest::Params::Params()
>-    : call({false, Call::Config::BitrateConfig(), 0}),
>+    : call({false, BitrateConstraints(), 0}),
>       video{{false, 640, 480, 30, 50, 800, 800, false, "VP8", 1, -1, 0, false,
>-             false, ""},
>+             false, false, ""},
>             {false, 640, 480, 30, 50, 800, 800, false, "VP8", 1, -1, 0, false,
>-             false, ""}},
>+             false, false, ""}},
>       audio({false, false, false}),
>       screenshare{{false, false, 10, 0}, {false, false, 10, 0}},
>       analyzer({"", 0.0, 0.0, 0, "", ""}),
>       pipe(),
>-      ss{{std::vector<VideoStream>(), 0, 0, -1, std::vector<SpatialLayer>()},
>-         {std::vector<VideoStream>(), 0, 0, -1, std::vector<SpatialLayer>()}},
>-      logging({false, "", "", ""}) {}
>+      ss{{std::vector<VideoStream>(), 0, 0, -1, InterLayerPredMode::kOn,
>+          std::vector<SpatialLayer>()},
>+         {std::vector<VideoStream>(), 0, 0, -1, InterLayerPredMode::kOn,
>+          std::vector<SpatialLayer>()}},
>+      logging({"", "", ""}) {}
> 
> VideoQualityTest::Params::~Params() = default;
> 
>@@ -1200,20 +243,8 @@ VideoStream VideoQualityTest::DefaultVideoStream(const Params& params,
>   stream.target_bitrate_bps = params.video[video_idx].target_bitrate_bps;
>   stream.max_bitrate_bps = params.video[video_idx].max_bitrate_bps;
>   stream.max_qp = kDefaultMaxQp;
>-  // TODO(sprang): Can we make this less of a hack?
>-  if (params.video[video_idx].num_temporal_layers == 2) {
>-    stream.temporal_layer_thresholds_bps.push_back(stream.target_bitrate_bps);
>-  } else if (params.video[video_idx].num_temporal_layers == 3) {
>-    stream.temporal_layer_thresholds_bps.push_back(stream.max_bitrate_bps / 4);
>-    stream.temporal_layer_thresholds_bps.push_back(stream.target_bitrate_bps);
>-  } else {
>-    RTC_CHECK_LE(params.video[video_idx].num_temporal_layers,
>-                 kMaxTemporalStreams);
>-    for (int i = 0; i < params.video[video_idx].num_temporal_layers - 1; ++i) {
>-      stream.temporal_layer_thresholds_bps.push_back(static_cast<int>(
>-          stream.max_bitrate_bps * kVp8LayerRateAlloction[0][i] + 0.5));
>-    }
>-  }
>+  stream.num_temporal_layers = params.video[video_idx].num_temporal_layers;
>+  stream.active = true;
>   return stream;
> }
> 
>@@ -1239,10 +270,13 @@ void VideoQualityTest::FillScalabilitySettings(
>     size_t selected_stream,
>     int num_spatial_layers,
>     int selected_sl,
>+    InterLayerPredMode inter_layer_pred,
>     const std::vector<std::string>& sl_descriptors) {
>   if (params->ss[video_idx].streams.empty() &&
>       params->ss[video_idx].infer_streams) {
>     webrtc::VideoEncoderConfig encoder_config;
>+    encoder_config.codec_type =
>+        PayloadStringToCodecType(params->video[video_idx].codec);
>     encoder_config.content_type =
>         params->screenshare[video_idx].enabled
>             ? webrtc::VideoEncoderConfig::ContentType::kScreen
>@@ -1252,6 +286,7 @@ void VideoQualityTest::FillScalabilitySettings(
>         params->video[video_idx].min_transmit_bps;
>     encoder_config.number_of_streams = num_streams;
>     encoder_config.spatial_layers = params->ss[video_idx].spatial_layers;
>+    encoder_config.simulcast_layers = std::vector<VideoStream>(num_streams);
>     encoder_config.video_stream_factory =
>         new rtc::RefCountedObject<cricket::EncoderStreamFactory>(
>             params->video[video_idx].codec, kDefaultMaxQp,
>@@ -1286,10 +321,8 @@ void VideoQualityTest::FillScalabilitySettings(
>         stream.max_bitrate_bps = v[5];
>       if (v.size() > 6 && v[6] != -1)
>         stream.max_qp = v[6];
>-      if (v.size() > 7) {
>-        stream.temporal_layer_thresholds_bps.clear();
>-        stream.temporal_layer_thresholds_bps.insert(
>-            stream.temporal_layer_thresholds_bps.end(), v.begin() + 7, v.end());
>+      if (v.size() > 7 && v[7] != -1) {
>+        stream.num_temporal_layers = v[7];
>       } else {
>         // Automatic TL thresholds for more than two layers not supported.
>         RTC_CHECK_LE(params->video[video_idx].num_temporal_layers, 2);
>@@ -1302,17 +335,24 @@ void VideoQualityTest::FillScalabilitySettings(
>   params->ss[video_idx].selected_stream = selected_stream;
> 
>   params->ss[video_idx].selected_sl = selected_sl;
>+  params->ss[video_idx].inter_layer_pred = inter_layer_pred;
>   RTC_CHECK(params->ss[video_idx].spatial_layers.empty());
>   for (auto descriptor : sl_descriptors) {
>     if (descriptor.empty())
>       continue;
>     std::vector<int> v = VideoQualityTest::ParseCSV(descriptor);
>-    RTC_CHECK_GT(v[2], 0);
>+    RTC_CHECK_EQ(v.size(), 7);
>+
>+    SpatialLayer layer = {0};
>+    layer.width = v[0];
>+    layer.height = v[1];
>+    layer.numberOfTemporalLayers = v[2];
>+    layer.maxBitrate = v[3];
>+    layer.minBitrate = v[4];
>+    layer.targetBitrate = v[5];
>+    layer.qpMax = v[6];
>+    layer.active = true;
> 
>-    SpatialLayer layer;
>-    layer.scaling_factor_num = v[0] == -1 ? 1 : v[0];
>-    layer.scaling_factor_den = v[1] == -1 ? 1 : v[1];
>-    layer.target_bitrate_bps = v[2];
>     params->ss[video_idx].spatial_layers.push_back(layer);
>   }
> }
>@@ -1320,12 +360,8 @@ void VideoQualityTest::FillScalabilitySettings(
> void VideoQualityTest::SetupVideo(Transport* send_transport,
>                                   Transport* recv_transport) {
>   size_t total_streams_used = 0;
>-  size_t num_flexfec_streams = params_.video[0].flexfec ? 1 : 0;
>-  CreateAudioAndFecSendConfigs(0, num_flexfec_streams, send_transport);
>-  CreateMatchingAudioAndFecConfigs(recv_transport);
>   video_receive_configs_.clear();
>   video_send_configs_.clear();
>-  video_encoders_.clear();
>   video_encoder_configs_.clear();
>   allocated_decoders_.clear();
>   bool decode_all_receive_streams = true;
>@@ -1334,42 +370,32 @@ void VideoQualityTest::SetupVideo(Transport* send_transport,
>   video_encoder_configs_.resize(num_video_streams_);
>   for (size_t video_idx = 0; video_idx < num_video_streams_; ++video_idx) {
>     video_send_configs_.push_back(VideoSendStream::Config(send_transport));
>-    video_encoders_.push_back(nullptr);
>     video_encoder_configs_.push_back(VideoEncoderConfig());
>     num_video_substreams = params_.ss[video_idx].streams.size();
>     RTC_CHECK_GT(num_video_substreams, 0);
>-    CreateVideoSendConfig(&video_send_configs_[video_idx], num_video_substreams,
>-                          total_streams_used, send_transport);
>+    for (size_t i = 0; i < num_video_substreams; ++i)
>+      video_send_configs_[video_idx].rtp.ssrcs.push_back(
>+          kVideoSendSsrcs[total_streams_used + i]);
> 
>     int payload_type;
>     if (params_.video[video_idx].codec == "H264") {
>-      video_encoders_[video_idx] =
>-          H264Encoder::Create(cricket::VideoCodec("H264"));
>       payload_type = kPayloadTypeH264;
>     } else if (params_.video[video_idx].codec == "VP8") {
>-      if (params_.screenshare[video_idx].enabled &&
>-          params_.ss[video_idx].streams.size() > 1) {
>-        // Simulcast screenshare needs a simulcast encoder adapter to work,
>-        // since encoders usually can't natively do simulcast with different
>-        // frame rates for the different layers.
>-        video_encoders_[video_idx].reset(
>-            new SimulcastEncoderAdapter(new InternalEncoderFactory()));
>-      } else {
>-        video_encoders_[video_idx] = VP8Encoder::Create();
>-      }
>       payload_type = kPayloadTypeVP8;
>     } else if (params_.video[video_idx].codec == "VP9") {
>-      video_encoders_[video_idx] = VP9Encoder::Create();
>+      payload_type = kPayloadTypeVP9;
>+    } else if (params_.video[video_idx].codec == "multiplex") {
>       payload_type = kPayloadTypeVP9;
>     } else {
>       RTC_NOTREACHED() << "Codec not supported!";
>       return;
>     }
>-    video_send_configs_[video_idx].encoder_settings.encoder =
>-        video_encoders_[video_idx].get();
>-    video_send_configs_[video_idx].encoder_settings.payload_name =
>+    video_send_configs_[video_idx].encoder_settings.encoder_factory =
>+        &video_encoder_factory_;
>+
>+    video_send_configs_[video_idx].rtp.payload_name =
>         params_.video[video_idx].codec;
>-    video_send_configs_[video_idx].encoder_settings.payload_type = payload_type;
>+    video_send_configs_[video_idx].rtp.payload_type = payload_type;
>     video_send_configs_[video_idx].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
>     video_send_configs_[video_idx].rtp.rtx.payload_type = kSendRtxPayloadType;
>     for (size_t i = 0; i < num_video_substreams; ++i) {
>@@ -1391,6 +417,15 @@ void VideoQualityTest::SetupVideo(Transport* send_transport,
>     video_send_configs_[video_idx].rtp.extensions.push_back(RtpExtension(
>         RtpExtension::kVideoTimingUri, test::kVideoTimingExtensionId));
> 
>+    video_encoder_configs_[video_idx].video_format.name =
>+        params_.video[video_idx].codec;
>+
>+    video_encoder_configs_[video_idx].video_format.parameters =
>+        params_.video[video_idx].sdp_params;
>+
>+    video_encoder_configs_[video_idx].codec_type =
>+        PayloadStringToCodecType(params_.video[video_idx].codec);
>+
>     video_encoder_configs_[video_idx].min_transmit_bitrate_bps =
>         params_.video[video_idx].min_transmit_bps;
> 
>@@ -1405,6 +440,8 @@ void VideoQualityTest::SetupVideo(Transport* send_transport,
>           params_.ss[video_idx].streams[i].max_bitrate_bps;
>     }
>     if (params_.ss[video_idx].infer_streams) {
>+      video_encoder_configs_[video_idx].simulcast_layers =
>+          std::vector<VideoStream>(params_.ss[video_idx].streams.size());
>       video_encoder_configs_[video_idx].video_stream_factory =
>           new rtc::RefCountedObject<cricket::EncoderStreamFactory>(
>               params_.video[video_idx].codec,
>@@ -1419,49 +456,20 @@ void VideoQualityTest::SetupVideo(Transport* send_transport,
> 
>     video_encoder_configs_[video_idx].spatial_layers =
>         params_.ss[video_idx].spatial_layers;
>-
>-    std::vector<VideoReceiveStream::Config> new_receive_configs =
>-        CreateMatchingVideoReceiveConfigs(video_send_configs_[video_idx],
>-                                          recv_transport);
>-
>     decode_all_receive_streams = params_.ss[video_idx].selected_stream ==
>                                  params_.ss[video_idx].streams.size();
>-
>-    for (size_t i = 0; i < num_video_substreams; ++i) {
>-      new_receive_configs[i].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
>-      new_receive_configs[i].rtp.rtx_ssrc =
>-          kSendRtxSsrcs[i + total_streams_used];
>-      new_receive_configs[i]
>-          .rtp.rtx_associated_payload_types[kSendRtxPayloadType] = payload_type;
>-      new_receive_configs[i].rtp.transport_cc = params_.call.send_side_bwe;
>-      new_receive_configs[i].rtp.remb = !params_.call.send_side_bwe;
>-      // Enable RTT calculation so NTP time estimator will work.
>-      new_receive_configs[i].rtp.rtcp_xr.receiver_reference_time_report = true;
>-      // Force fake decoders on non-selected simulcast streams.
>-      if (!decode_all_receive_streams &&
>-          i != params_.ss[video_idx].selected_stream) {
>-        VideoReceiveStream::Decoder decoder;
>-        decoder.decoder = new test::FakeDecoder();
>-        decoder.payload_type =
>-            video_send_configs_[video_idx].encoder_settings.payload_type;
>-        decoder.payload_name =
>-            video_send_configs_[video_idx].encoder_settings.payload_name;
>-        new_receive_configs[i].decoders.clear();
>-        allocated_decoders_.emplace_back(decoder.decoder);
>-        new_receive_configs[i].decoders.push_back(decoder);
>-      }
>-    }
>-
>-    for (VideoReceiveStream::Config& config : new_receive_configs) {
>-      video_receive_configs_.push_back(config.Copy());
>-    }
>+    absl::optional<int> decode_sub_stream;
>+    if (!decode_all_receive_streams)
>+      decode_sub_stream = params_.ss[video_idx].selected_stream;
>+    CreateMatchingVideoReceiveConfigs(
>+        video_send_configs_[video_idx], recv_transport,
>+        params_.call.send_side_bwe, decode_sub_stream, true, kNackRtpHistoryMs);
> 
>     if (params_.screenshare[video_idx].enabled) {
>       // Fill out codec settings.
>       video_encoder_configs_[video_idx].content_type =
>           VideoEncoderConfig::ContentType::kScreen;
>-      degradation_preference_ =
>-          VideoSendStream::DegradationPreference::kMaintainResolution;
>+      degradation_preference_ = DegradationPreference::MAINTAIN_RESOLUTION;
>       if (params_.video[video_idx].codec == "VP8") {
>         VideoCodecVP8 vp8_settings = VideoEncoder::GetDefaultVp8Settings();
>         vp8_settings.denoisingOn = false;
>@@ -1479,6 +487,7 @@ void VideoQualityTest::SetupVideo(Transport* send_transport,
>             params_.video[video_idx].num_temporal_layers);
>         vp9_settings.numberOfSpatialLayers = static_cast<unsigned char>(
>             params_.ss[video_idx].num_spatial_layers);
>+        vp9_settings.interLayerPred = params_.ss[video_idx].inter_layer_pred;
>         video_encoder_configs_[video_idx].encoder_specific_settings =
>             new rtc::RefCountedObject<
>                 VideoEncoderConfig::Vp9EncoderSpecificSettings>(vp9_settings);
>@@ -1491,80 +500,61 @@ void VideoQualityTest::SetupVideo(Transport* send_transport,
>           params_.video[video_idx].num_temporal_layers);
>       vp9_settings.numberOfSpatialLayers =
>           static_cast<unsigned char>(params_.ss[video_idx].num_spatial_layers);
>+      vp9_settings.interLayerPred = params_.ss[video_idx].inter_layer_pred;
>       video_encoder_configs_[video_idx].encoder_specific_settings =
>           new rtc::RefCountedObject<
>               VideoEncoderConfig::Vp9EncoderSpecificSettings>(vp9_settings);
>+    } else if (params_.video[video_idx].automatic_scaling) {
>+      if (params_.video[video_idx].codec == "VP8") {
>+        VideoCodecVP8 vp8_settings = VideoEncoder::GetDefaultVp8Settings();
>+        vp8_settings.automaticResizeOn = true;
>+        video_encoder_configs_[video_idx].encoder_specific_settings =
>+            new rtc::RefCountedObject<
>+                VideoEncoderConfig::Vp8EncoderSpecificSettings>(vp8_settings);
>+      } else if (params_.video[video_idx].codec == "VP9") {
>+        VideoCodecVP9 vp9_settings = VideoEncoder::GetDefaultVp9Settings();
>+        vp9_settings.automaticResizeOn = true;
>+        video_encoder_configs_[video_idx].encoder_specific_settings =
>+            new rtc::RefCountedObject<
>+                VideoEncoderConfig::Vp9EncoderSpecificSettings>(vp9_settings);
>+      } else {
>+        RTC_NOTREACHED() << "Automatic scaling not supported for codec "
>+                         << params_.video[video_idx].codec << ", stream "
>+                         << video_idx;
>+      }
>     }
>     total_streams_used += num_video_substreams;
>   }
> 
>   // FEC supported only for single video stream mode yet.
>   if (params_.video[0].flexfec) {
>-    video_send_configs_[0].rtp.flexfec.payload_type = kFlexfecPayloadType;
>-    video_send_configs_[0].rtp.flexfec.ssrc = kFlexfecSendSsrc;
>     if (decode_all_receive_streams) {
>-      for (uint32_t media_ssrc : video_send_configs_[0].rtp.ssrcs) {
>-        video_send_configs_[0].rtp.flexfec.protected_media_ssrcs.push_back(
>-            media_ssrc);
>-      }
>+      SetSendFecConfig(GetVideoSendConfig()->rtp.ssrcs);
>     } else {
>-      video_send_configs_[0].rtp.flexfec.protected_media_ssrcs = {
>-          kVideoSendSsrcs[params_.ss[0].selected_stream]};
>+      SetSendFecConfig({kVideoSendSsrcs[params_.ss[0].selected_stream]});
>     }
> 
>-    // The matching receive config is _not_ created by
>-    // CreateMatchingReceiveConfigs, since VideoQualityTest is not a BaseTest.
>-    // Set up the receive config manually instead.
>-    FlexfecReceiveStream::Config flexfec_receive_config(recv_transport);
>-    flexfec_receive_config.payload_type =
>-        video_send_configs_[0].rtp.flexfec.payload_type;
>-    flexfec_receive_config.remote_ssrc =
>-        video_send_configs_[0].rtp.flexfec.ssrc;
>-    flexfec_receive_config.protected_media_ssrcs =
>-        video_send_configs_[0].rtp.flexfec.protected_media_ssrcs;
>-    flexfec_receive_config.local_ssrc = kReceiverLocalVideoSsrc;
>-    flexfec_receive_config.transport_cc = params_.call.send_side_bwe;
>+    CreateMatchingFecConfig(recv_transport, *GetVideoSendConfig());
>+    GetFlexFecConfig()->transport_cc = params_.call.send_side_bwe;
>     if (params_.call.send_side_bwe) {
>-      flexfec_receive_config.rtp_header_extensions.push_back(
>+      GetFlexFecConfig()->rtp_header_extensions.push_back(
>           RtpExtension(RtpExtension::kTransportSequenceNumberUri,
>                        test::kTransportSequenceNumberExtensionId));
>     } else {
>-      flexfec_receive_config.rtp_header_extensions.push_back(RtpExtension(
>+      GetFlexFecConfig()->rtp_header_extensions.push_back(RtpExtension(
>           RtpExtension::kAbsSendTimeUri, test::kAbsSendTimeExtensionId));
>     }
>-    flexfec_receive_configs_.push_back(flexfec_receive_config);
>-    if (num_video_substreams > 0) {
>-      video_receive_configs_[0].rtp.protected_by_flexfec = true;
>-    }
>   }
> 
>   if (params_.video[0].ulpfec) {
>-    video_send_configs_[0].rtp.ulpfec.red_payload_type = kRedPayloadType;
>-    video_send_configs_[0].rtp.ulpfec.ulpfec_payload_type = kUlpfecPayloadType;
>-    video_send_configs_[0].rtp.ulpfec.red_rtx_payload_type = kRtxRedPayloadType;
>-
>+    SetSendUlpFecConfig(GetVideoSendConfig());
>     if (decode_all_receive_streams) {
>-      for (auto it = video_receive_configs_.begin();
>-           it != video_receive_configs_.end(); ++it) {
>-        it->rtp.red_payload_type =
>-            video_send_configs_[0].rtp.ulpfec.red_payload_type;
>-        it->rtp.ulpfec_payload_type =
>-            video_send_configs_[0].rtp.ulpfec.ulpfec_payload_type;
>-        it->rtp.rtx_associated_payload_types
>-            [video_send_configs_[0].rtp.ulpfec.red_rtx_payload_type] =
>-            video_send_configs_[0].rtp.ulpfec.red_payload_type;
>+      for (auto& receive_config : video_receive_configs_) {
>+        SetReceiveUlpFecConfig(&receive_config);
>       }
>     } else {
>-      video_receive_configs_[params_.ss[0].selected_stream]
>-          .rtp.red_payload_type =
>-          video_send_configs_[0].rtp.ulpfec.red_payload_type;
>-      video_receive_configs_[params_.ss[0].selected_stream]
>-          .rtp.ulpfec_payload_type =
>-          video_send_configs_[0].rtp.ulpfec.ulpfec_payload_type;
>-      video_receive_configs_[params_.ss[0].selected_stream]
>-          .rtp.rtx_associated_payload_types
>-              [video_send_configs_[0].rtp.ulpfec.red_rtx_payload_type] =
>-          video_send_configs_[0].rtp.ulpfec.red_payload_type;
>+      SetReceiveUlpFecConfig(
>+          &video_receive_configs_[params_.ss[0].selected_stream]);
>     }
>   }
> }
>@@ -1572,17 +562,15 @@ void VideoQualityTest::SetupVideo(Transport* send_transport,
> void VideoQualityTest::SetupThumbnails(Transport* send_transport,
>                                        Transport* recv_transport) {
>   for (int i = 0; i < params_.call.num_thumbnails; ++i) {
>-    thumbnail_encoders_.emplace_back(VP8Encoder::Create());
>-
>     // Thumbnails will be send in the other way: from receiver_call to
>     // sender_call.
>     VideoSendStream::Config thumbnail_send_config(recv_transport);
>     thumbnail_send_config.rtp.ssrcs.push_back(kThumbnailSendSsrcStart + i);
>-    thumbnail_send_config.encoder_settings.encoder =
>-        thumbnail_encoders_.back().get();
>-    thumbnail_send_config.encoder_settings.payload_name =
>-        params_.video[0].codec;
>-    thumbnail_send_config.encoder_settings.payload_type = kPayloadTypeVP8;
>+    // TODO(nisse): Could use a simpler VP8-only encoder factory.
>+    thumbnail_send_config.encoder_settings.encoder_factory =
>+        &video_encoder_factory_;
>+    thumbnail_send_config.rtp.payload_name = params_.video[0].codec;
>+    thumbnail_send_config.rtp.payload_type = kPayloadTypeVP8;
>     thumbnail_send_config.rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
>     thumbnail_send_config.rtp.rtx.payload_type = kSendRtxPayloadType;
>     thumbnail_send_config.rtp.rtx.ssrcs.push_back(kThumbnailRtxSsrcStart + i);
>@@ -1597,6 +585,8 @@ void VideoQualityTest::SetupThumbnails(Transport* send_transport,
>     }
> 
>     VideoEncoderConfig thumbnail_encoder_config;
>+    thumbnail_encoder_config.codec_type = kVideoCodecVP8;
>+    thumbnail_encoder_config.video_format.name = "VP8";
>     thumbnail_encoder_config.min_transmit_bitrate_bps = 7500;
>     thumbnail_send_config.suspend_below_min_bitrate =
>         params_.video[0].suspend_below_min_bitrate;
>@@ -1606,6 +596,7 @@ void VideoQualityTest::SetupThumbnails(Transport* send_transport,
>       thumbnail_encoder_config.video_stream_factory =
>           new rtc::RefCountedObject<VideoStreamFactory>(params_.ss[0].streams);
>     } else {
>+      thumbnail_encoder_config.simulcast_layers = std::vector<VideoStream>(1);
>       thumbnail_encoder_config.video_stream_factory =
>           new rtc::RefCountedObject<cricket::EncoderStreamFactory>(
>               params_.video[0].codec, params_.ss[0].streams[0].max_qp,
>@@ -1613,39 +604,19 @@ void VideoQualityTest::SetupThumbnails(Transport* send_transport,
>     }
>     thumbnail_encoder_config.spatial_layers = params_.ss[0].spatial_layers;
> 
>-    VideoReceiveStream::Config thumbnail_receive_config(send_transport);
>-    thumbnail_receive_config.rtp.remb = false;
>-    thumbnail_receive_config.rtp.transport_cc = true;
>-    thumbnail_receive_config.rtp.local_ssrc = kReceiverLocalVideoSsrc;
>-    for (const RtpExtension& extension : thumbnail_send_config.rtp.extensions)
>-      thumbnail_receive_config.rtp.extensions.push_back(extension);
>-    thumbnail_receive_config.renderer = &fake_renderer_;
>-
>-    VideoReceiveStream::Decoder decoder =
>-        test::CreateMatchingDecoder(thumbnail_send_config.encoder_settings);
>-    allocated_decoders_.push_back(
>-        std::unique_ptr<VideoDecoder>(decoder.decoder));
>-    thumbnail_receive_config.decoders.clear();
>-    thumbnail_receive_config.decoders.push_back(decoder);
>-    thumbnail_receive_config.rtp.remote_ssrc =
>-        thumbnail_send_config.rtp.ssrcs[0];
>-
>-    thumbnail_receive_config.rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
>-    thumbnail_receive_config.rtp.rtx_ssrc = kThumbnailRtxSsrcStart + i;
>-    thumbnail_receive_config.rtp
>-        .rtx_associated_payload_types[kSendRtxPayloadType] = kPayloadTypeVP8;
>-    thumbnail_receive_config.rtp.transport_cc = params_.call.send_side_bwe;
>-    thumbnail_receive_config.rtp.remb = !params_.call.send_side_bwe;
>-
>     thumbnail_encoder_configs_.push_back(thumbnail_encoder_config.Copy());
>     thumbnail_send_configs_.push_back(thumbnail_send_config.Copy());
>-    thumbnail_receive_configs_.push_back(thumbnail_receive_config.Copy());
>-  }
> 
>-  for (int i = 0; i < params_.call.num_thumbnails; ++i) {
>+    AddMatchingVideoReceiveConfigs(
>+        &thumbnail_receive_configs_, thumbnail_send_config, send_transport,
>+        params_.call.send_side_bwe, absl::nullopt, false, kNackRtpHistoryMs);
>+  }
>+  for (size_t i = 0; i < thumbnail_send_configs_.size(); ++i) {
>     thumbnail_send_streams_.push_back(receiver_call_->CreateVideoSendStream(
>         thumbnail_send_configs_[i].Copy(),
>         thumbnail_encoder_configs_[i].Copy()));
>+  }
>+  for (size_t i = 0; i < thumbnail_receive_configs_.size(); ++i) {
>     thumbnail_receive_streams_.push_back(sender_call_->CreateVideoReceiveStream(
>         thumbnail_receive_configs_[i].Copy()));
>   }
>@@ -1673,7 +644,7 @@ void VideoQualityTest::SetupThumbnailCapturers(size_t num_thumbnail_streams) {
>   for (size_t i = 0; i < num_thumbnail_streams; ++i) {
>     thumbnail_capturers_.emplace_back(test::FrameGeneratorCapturer::Create(
>         static_cast<int>(thumbnail.width), static_cast<int>(thumbnail.height),
>-        thumbnail.max_framerate, clock_));
>+        absl::nullopt, absl::nullopt, thumbnail.max_framerate, clock_));
>     RTC_DCHECK(thumbnail_capturers_.back());
>   }
> }
>@@ -1725,6 +696,8 @@ std::unique_ptr<test::FrameGenerator> VideoQualityTest::CreateFrameGenerator(
> }
> 
> void VideoQualityTest::CreateCapturers() {
>+  RTC_DCHECK(video_sources_.empty());
>+  RTC_DCHECK(video_capturers_.empty());
>   video_capturers_.resize(num_video_streams_);
>   for (size_t video_idx = 0; video_idx < num_video_streams_; ++video_idx) {
>     if (params_.screenshare[video_idx].enabled) {
>@@ -1737,9 +710,21 @@ void VideoQualityTest::CreateCapturers() {
>       video_capturers_[video_idx].reset(frame_generator_capturer);
>     } else {
>       if (params_.video[video_idx].clip_name == "Generator") {
>+        video_capturers_[video_idx].reset(test::FrameGeneratorCapturer::Create(
>+            static_cast<int>(params_.video[video_idx].width),
>+            static_cast<int>(params_.video[video_idx].height), absl::nullopt,
>+            absl::nullopt, params_.video[video_idx].fps, clock_));
>+      } else if (params_.video[video_idx].clip_name == "GeneratorI420A") {
>         video_capturers_[video_idx].reset(test::FrameGeneratorCapturer::Create(
>             static_cast<int>(params_.video[video_idx].width),
>             static_cast<int>(params_.video[video_idx].height),
>+            test::FrameGenerator::OutputType::I420A, absl::nullopt,
>+            params_.video[video_idx].fps, clock_));
>+      } else if (params_.video[video_idx].clip_name == "GeneratorI010") {
>+        video_capturers_[video_idx].reset(test::FrameGeneratorCapturer::Create(
>+            static_cast<int>(params_.video[video_idx].width),
>+            static_cast<int>(params_.video[video_idx].height),
>+            test::FrameGenerator::OutputType::I010, absl::nullopt,
>             params_.video[video_idx].fps, clock_));
>       } else if (params_.video[video_idx].clip_name.empty()) {
>         video_capturers_[video_idx].reset(test::VcmCapturer::Create(
>@@ -1752,7 +737,8 @@ void VideoQualityTest::CreateCapturers() {
>               test::FrameGeneratorCapturer::Create(
>                   static_cast<int>(params_.video[video_idx].width),
>                   static_cast<int>(params_.video[video_idx].height),
>-                  params_.video[video_idx].fps, clock_));
>+                  absl::nullopt, absl::nullopt, params_.video[video_idx].fps,
>+                  clock_));
>         }
>       } else {
>         video_capturers_[video_idx].reset(
>@@ -1767,12 +753,43 @@ void VideoQualityTest::CreateCapturers() {
>       }
>     }
>     RTC_DCHECK(video_capturers_[video_idx].get());
>+    video_sources_.push_back(video_capturers_[video_idx].get());
>   }
> }
> 
>+void VideoQualityTest::StartAudioStreams() {
>+  audio_send_stream_->Start();
>+  for (AudioReceiveStream* audio_recv_stream : audio_receive_streams_)
>+    audio_recv_stream->Start();
>+}
>+
>+void VideoQualityTest::StartThumbnailCapture() {
>+  for (std::unique_ptr<test::VideoCapturer>& capturer : thumbnail_capturers_)
>+    capturer->Start();
>+}
>+
>+void VideoQualityTest::StopThumbnailCapture() {
>+  for (std::unique_ptr<test::VideoCapturer>& capturer : thumbnail_capturers_)
>+    capturer->Stop();
>+}
>+
>+void VideoQualityTest::StartThumbnails() {
>+  for (VideoSendStream* send_stream : thumbnail_send_streams_)
>+    send_stream->Start();
>+  for (VideoReceiveStream* receive_stream : thumbnail_receive_streams_)
>+    receive_stream->Start();
>+}
>+
>+void VideoQualityTest::StopThumbnails() {
>+  for (VideoReceiveStream* receive_stream : thumbnail_receive_streams_)
>+    receive_stream->Stop();
>+  for (VideoSendStream* send_stream : thumbnail_send_streams_)
>+    send_stream->Stop();
>+}
>+
> std::unique_ptr<test::LayerFilteringTransport>
> VideoQualityTest::CreateSendTransport() {
>-  return rtc::MakeUnique<test::LayerFilteringTransport>(
>+  return absl::make_unique<test::LayerFilteringTransport>(
>       &task_queue_, params_.pipe, sender_call_.get(), kPayloadTypeVP8,
>       kPayloadTypeVP9, params_.video[0].selected_tl, params_.ss[0].selected_sl,
>       payload_type_map_, kVideoSendSsrcs[0],
>@@ -1782,33 +799,10 @@ VideoQualityTest::CreateSendTransport() {
> 
> std::unique_ptr<test::DirectTransport>
> VideoQualityTest::CreateReceiveTransport() {
>-  return rtc::MakeUnique<test::DirectTransport>(
>+  return absl::make_unique<test::DirectTransport>(
>       &task_queue_, params_.pipe, receiver_call_.get(), payload_type_map_);
> }
> 
>-void VideoQualityTest::CreateVideoStreams() {
>-  RTC_DCHECK(video_send_streams_.empty());
>-  RTC_DCHECK(video_receive_streams_.empty());
>-  RTC_DCHECK_EQ(video_send_configs_.size(), num_video_streams_);
>-  for (size_t i = 0; i < video_send_configs_.size(); ++i) {
>-    video_send_streams_.push_back(sender_call_->CreateVideoSendStream(
>-        video_send_configs_[i].Copy(), video_encoder_configs_[i].Copy()));
>-  }
>-  for (size_t i = 0; i < video_receive_configs_.size(); ++i) {
>-    video_receive_streams_.push_back(receiver_call_->CreateVideoReceiveStream(
>-        video_receive_configs_[i].Copy()));
>-  }
>-
>-  AssociateFlexfecStreamsWithVideoStreams();
>-}
>-
>-void VideoQualityTest::DestroyStreams() {
>-  CallTest::DestroyStreams();
>-
>-  for (VideoSendStream* video_send_stream : video_send_streams_)
>-    sender_call_->DestroyVideoSendStream(video_send_stream);
>-}
>-
> void VideoQualityTest::RunWithAnalyzer(const Params& params) {
>   num_video_streams_ = params.call.dual_video ? 2 : 1;
>   std::unique_ptr<test::LayerFilteringTransport> send_transport;
>@@ -1817,8 +811,6 @@ void VideoQualityTest::RunWithAnalyzer(const Params& params) {
>   std::unique_ptr<VideoAnalyzer> analyzer;
> 
>   params_ = params;
>-
>-  RTC_CHECK(!params_.audio.enabled);
>   // TODO(ivica): Merge with RunWithRenderer and use a flag / argument to
>   // differentiate between the analyzer and the renderer case.
>   CheckParams();
>@@ -1832,16 +824,14 @@ void VideoQualityTest::RunWithAnalyzer(const Params& params) {
>   }
> 
>   if (!params.logging.rtc_event_log_name.empty()) {
>-    send_event_log_ =
>-        RtcEventLog::Create(clock_, RtcEventLog::EncodingType::Legacy);
>-    recv_event_log_ =
>-        RtcEventLog::Create(clock_, RtcEventLog::EncodingType::Legacy);
>+    send_event_log_ = RtcEventLog::Create(RtcEventLog::EncodingType::Legacy);
>+    recv_event_log_ = RtcEventLog::Create(RtcEventLog::EncodingType::Legacy);
>     std::unique_ptr<RtcEventLogOutputFile> send_output(
>-        rtc::MakeUnique<RtcEventLogOutputFile>(
>+        absl::make_unique<RtcEventLogOutputFile>(
>             params.logging.rtc_event_log_name + "_send",
>             RtcEventLog::kUnlimitedOutput));
>     std::unique_ptr<RtcEventLogOutputFile> recv_output(
>-        rtc::MakeUnique<RtcEventLogOutputFile>(
>+        absl::make_unique<RtcEventLogOutputFile>(
>             params.logging.rtc_event_log_name + "_recv",
>             RtcEventLog::kUnlimitedOutput));
>     bool event_log_started =
>@@ -1862,6 +852,9 @@ void VideoQualityTest::RunWithAnalyzer(const Params& params) {
> 
>   task_queue_.SendTask([this, &send_call_config, &recv_call_config,
>                         &send_transport, &recv_transport]() {
>+    if (params_.audio.enabled)
>+      InitializeAudioDevice(&send_call_config, &recv_call_config);
>+
>     CreateCalls(send_call_config, recv_call_config);
>     send_transport = CreateSendTransport();
>     recv_transport = CreateReceiveTransport();
>@@ -1871,7 +864,7 @@ void VideoQualityTest::RunWithAnalyzer(const Params& params) {
>   if (graph_title.empty())
>     graph_title = VideoQualityTest::GenerateGraphTitle();
>   bool is_quick_test_enabled = field_trial::IsEnabled("WebRTC-QuickPerfTest");
>-  analyzer = rtc::MakeUnique<VideoAnalyzer>(
>+  analyzer = absl::make_unique<VideoAnalyzer>(
>       send_transport.get(), params_.analyzer.test_label,
>       params_.analyzer.avg_psnr_threshold, params_.analyzer.avg_ssim_threshold,
>       is_quick_test_enabled
>@@ -1894,10 +887,9 @@ void VideoQualityTest::RunWithAnalyzer(const Params& params) {
>     SetupThumbnails(analyzer.get(), recv_transport.get());
>     video_receive_configs_[params_.ss[0].selected_stream].renderer =
>         analyzer.get();
>-    video_send_configs_[0].pre_encode_callback = analyzer->pre_encode_proxy();
>-    RTC_DCHECK(!video_send_configs_[0].post_encode_callback);
>-    video_send_configs_[0].post_encode_callback =
>-        analyzer->encode_timing_proxy();
>+    GetVideoSendConfig()->pre_encode_callback = analyzer->pre_encode_proxy();
>+    RTC_DCHECK(!GetVideoSendConfig()->post_encode_callback);
>+    GetVideoSendConfig()->post_encode_callback = analyzer.get();
> 
>     CreateFlexfecStreams();
>     CreateVideoStreams();
>@@ -1905,8 +897,8 @@ void VideoQualityTest::RunWithAnalyzer(const Params& params) {
>     if (video_receive_streams_.size() == 1)
>       analyzer->SetReceiveStream(video_receive_streams_[0]);
> 
>-    video_send_streams_[0]->SetSource(analyzer->OutputInterface(),
>-                                      degradation_preference_);
>+    GetVideoSendStream()->SetSource(analyzer->OutputInterface(),
>+                                    degradation_preference_);
>     SetupThumbnailCapturers(params_.call.num_thumbnails);
>     for (size_t i = 0; i < thumbnail_send_streams_.size(); ++i) {
>       thumbnail_send_streams_[i]->SetSource(thumbnail_capturers_[i].get(),
>@@ -1922,50 +914,28 @@ void VideoQualityTest::RunWithAnalyzer(const Params& params) {
>           video_capturers_[video_idx].get(), degradation_preference_);
>     }
> 
>-    StartEncodedFrameLogs(video_send_streams_[0]);
>+    StartEncodedFrameLogs(GetVideoSendStream());
>     StartEncodedFrameLogs(
>         video_receive_streams_[params_.ss[0].selected_stream]);
>-    for (VideoSendStream* video_send_stream : video_send_streams_)
>-      video_send_stream->Start();
>-    for (VideoSendStream* thumbnail_send_stream : thumbnail_send_streams_)
>-      thumbnail_send_stream->Start();
>-    for (VideoReceiveStream* receive_stream : video_receive_streams_)
>-      receive_stream->Start();
>-    for (VideoReceiveStream* thumbnail_receive_stream :
>-         thumbnail_receive_streams_)
>-      thumbnail_receive_stream->Start();
>-
>-    analyzer->StartMeasuringCpuProcessTime();
> 
>-    for (size_t video_idx = 0; video_idx < num_video_streams_; ++video_idx) {
>-      video_capturers_[video_idx]->Start();
>-    }
>-    for (std::unique_ptr<test::VideoCapturer>& video_caputurer :
>-         thumbnail_capturers_) {
>-      video_caputurer->Start();
>+    if (params_.audio.enabled) {
>+      SetupAudio(send_transport.get());
>+      StartAudioStreams();
>+      analyzer->SetAudioReceiveStream(audio_receive_streams_[0]);
>     }
>+    StartVideoStreams();
>+    StartThumbnails();
>+    analyzer->StartMeasuringCpuProcessTime();
>+    StartVideoCapture();
>+    StartThumbnailCapture();
>   });
> 
>   analyzer->Wait();
> 
>-  event_log_->StopLogging();
>-
>   task_queue_.SendTask([&]() {
>-    for (std::unique_ptr<test::VideoCapturer>& video_caputurer :
>-         thumbnail_capturers_)
>-      video_caputurer->Stop();
>-    for (size_t video_idx = 0; video_idx < num_video_streams_; ++video_idx) {
>-      video_capturers_[video_idx]->Stop();
>-    }
>-    for (VideoReceiveStream* thumbnail_receive_stream :
>-         thumbnail_receive_streams_)
>-      thumbnail_receive_stream->Stop();
>-    for (VideoReceiveStream* receive_stream : video_receive_streams_)
>-      receive_stream->Stop();
>-    for (VideoSendStream* thumbnail_send_stream : thumbnail_send_streams_)
>-      thumbnail_send_stream->Stop();
>-    for (VideoSendStream* video_send_stream : video_send_streams_)
>-      video_send_stream->Stop();
>+    StopThumbnailCapture();
>+    StopThumbnails();
>+    Stop();
> 
>     DestroyStreams();
>     DestroyThumbnailStreams();
>@@ -1981,42 +951,55 @@ void VideoQualityTest::RunWithAnalyzer(const Params& params) {
>   });
> }
> 
>-void VideoQualityTest::SetupAudio(Transport* transport,
>-                                  AudioReceiveStream** audio_receive_stream) {
>-  audio_send_config_ = AudioSendStream::Config(transport);
>-  audio_send_config_.rtp.ssrc = kAudioSendSsrc;
>+void VideoQualityTest::InitializeAudioDevice(Call::Config* send_call_config,
>+                                             Call::Config* recv_call_config) {
>+  rtc::scoped_refptr<TestAudioDeviceModule> fake_audio_device =
>+      TestAudioDeviceModule::CreateTestAudioDeviceModule(
>+          TestAudioDeviceModule::CreatePulsedNoiseCapturer(32000, 48000),
>+          TestAudioDeviceModule::CreateDiscardRenderer(48000), 1.f);
>+
>+  AudioState::Config audio_state_config;
>+  audio_state_config.audio_mixer = AudioMixerImpl::Create();
>+  audio_state_config.audio_processing = AudioProcessingBuilder().Create();
>+  audio_state_config.audio_device_module = fake_audio_device;
>+  send_call_config->audio_state = AudioState::Create(audio_state_config);
>+  RTC_CHECK(fake_audio_device->RegisterAudioCallback(
>+                send_call_config->audio_state->audio_transport()) == 0);
>+  recv_call_config->audio_state = AudioState::Create(audio_state_config);
>+  fake_audio_device->Init();
>+}
>+
>+void VideoQualityTest::SetupAudio(Transport* transport) {
>+  AudioSendStream::Config audio_send_config(transport);
>+  audio_send_config.rtp.ssrc = kAudioSendSsrc;
> 
>   // Add extension to enable audio send side BWE, and allow audio bit rate
>   // adaptation.
>-  audio_send_config_.rtp.extensions.clear();
>+  audio_send_config.rtp.extensions.clear();
>+  audio_send_config.send_codec_spec = AudioSendStream::Config::SendCodecSpec(
>+      kAudioSendPayloadType,
>+      {"OPUS",
>+       48000,
>+       2,
>+       {{"usedtx", (params_.audio.dtx ? "1" : "0")}, {"stereo", "1"}}});
>+
>   if (params_.call.send_side_bwe) {
>-    audio_send_config_.rtp.extensions.push_back(
>+    audio_send_config.rtp.extensions.push_back(
>         webrtc::RtpExtension(webrtc::RtpExtension::kTransportSequenceNumberUri,
>                              test::kTransportSequenceNumberExtensionId));
>-    audio_send_config_.min_bitrate_bps = kOpusMinBitrateBps;
>-    audio_send_config_.max_bitrate_bps = kOpusBitrateFbBps;
>+    audio_send_config.min_bitrate_bps = kOpusMinBitrateBps;
>+    audio_send_config.max_bitrate_bps = kOpusBitrateFbBps;
>+    audio_send_config.send_codec_spec->transport_cc_enabled = true;
>   }
>-  audio_send_config_.send_codec_spec = AudioSendStream::Config::SendCodecSpec(
>-      kAudioSendPayloadType,
>-      {"OPUS", 48000, 2,
>-       {{"usedtx", (params_.audio.dtx ? "1" : "0")},
>-         {"stereo", "1"}}});
>-  audio_send_config_.encoder_factory = encoder_factory_;
>-  audio_send_stream_ = sender_call_->CreateAudioSendStream(audio_send_config_);
>-
>-  AudioReceiveStream::Config audio_config;
>-  audio_config.rtp.local_ssrc = kReceiverLocalAudioSsrc;
>-  audio_config.rtcp_send_transport = transport;
>-  audio_config.rtp.remote_ssrc = audio_send_config_.rtp.ssrc;
>-  audio_config.rtp.transport_cc = params_.call.send_side_bwe;
>-  audio_config.rtp.extensions = audio_send_config_.rtp.extensions;
>-  audio_config.decoder_factory = decoder_factory_;
>-  audio_config.decoder_map = {{kAudioSendPayloadType, {"OPUS", 48000, 2}}};
>+  audio_send_config.encoder_factory = audio_encoder_factory_;
>+  SetAudioConfig(audio_send_config);
>+
>+  std::string sync_group;
>   if (params_.video[0].enabled && params_.audio.sync_video)
>-    audio_config.sync_group = kSyncGroup;
>+    sync_group = kSyncGroup;
> 
>-  *audio_receive_stream =
>-      receiver_call_->CreateAudioReceiveStream(audio_config);
>+  CreateMatchingAudioConfigs(transport, sync_group);
>+  CreateAudioStreams();
> }
> 
> void VideoQualityTest::RunWithRenderers(const Params& params) {
>@@ -2025,7 +1008,7 @@ void VideoQualityTest::RunWithRenderers(const Params& params) {
>   std::unique_ptr<test::DirectTransport> recv_transport;
>   std::unique_ptr<test::VideoRenderer> local_preview;
>   std::vector<std::unique_ptr<test::VideoRenderer>> loopback_renderers;
>-  AudioReceiveStream* audio_receive_stream = nullptr;
>+  RtcEventLogNullImpl null_event_log;
> 
>   task_queue_.SendTask([&]() {
>     params_ = params;
>@@ -2033,26 +1016,14 @@ void VideoQualityTest::RunWithRenderers(const Params& params) {
> 
>     // TODO(ivica): Remove bitrate_config and use the default Call::Config(), to
>     // match the full stack tests.
>-    Call::Config call_config(event_log_.get());
>-    call_config.bitrate_config = params_.call.call_bitrate_config;
>-
>-    rtc::scoped_refptr<test::FakeAudioDevice> fake_audio_device =
>-        new rtc::RefCountedObject<test::FakeAudioDevice>(
>-            test::FakeAudioDevice::CreatePulsedNoiseCapturer(32000, 48000),
>-            test::FakeAudioDevice::CreateDiscardRenderer(48000),
>-            1.f);
>+    Call::Config send_call_config(&null_event_log);
>+    send_call_config.bitrate_config = params_.call.call_bitrate_config;
>+    Call::Config recv_call_config(&null_event_log);
> 
>-    if (params_.audio.enabled) {
>-      AudioState::Config audio_state_config;
>-      audio_state_config.audio_mixer = AudioMixerImpl::Create();
>-      audio_state_config.audio_processing = AudioProcessingBuilder().Create();
>-      audio_state_config.audio_device_module = fake_audio_device;
>-      call_config.audio_state = AudioState::Create(audio_state_config);
>-      fake_audio_device->RegisterAudioCallback(
>-          call_config.audio_state->audio_transport());
>-    }
>+    if (params_.audio.enabled)
>+      InitializeAudioDevice(&send_call_config, &recv_call_config);
> 
>-    CreateCalls(call_config, call_config);
>+    CreateCalls(send_call_config, recv_call_config);
> 
>     // TODO(minyue): consider if this is a good transport even for audio only
>     // calls.
>@@ -2072,7 +1043,7 @@ void VideoQualityTest::RunWithRenderers(const Params& params) {
>           "Local Preview", params_.video[0].width, params_.video[0].height));
> 
>       SetupVideo(send_transport.get(), recv_transport.get());
>-      video_send_configs_[0].pre_encode_callback = local_preview.get();
>+      GetVideoSendConfig()->pre_encode_callback = local_preview.get();
> 
>       size_t num_streams_processed = 0;
>       for (size_t video_idx = 0; video_idx < num_video_streams_; ++video_idx) {
>@@ -2112,58 +1083,24 @@ void VideoQualityTest::RunWithRenderers(const Params& params) {
>       CreateVideoStreams();
> 
>       CreateCapturers();
>-      for (size_t video_idx = 0; video_idx < num_video_streams_; ++video_idx) {
>-        video_send_streams_[video_idx]->SetSource(
>-            video_capturers_[video_idx].get(), degradation_preference_);
>-      }
>+      ConnectVideoSourcesToStreams();
>     }
> 
>     if (params_.audio.enabled) {
>-      SetupAudio(send_transport.get(), &audio_receive_stream);
>+      SetupAudio(send_transport.get());
>     }
> 
>     for (VideoReceiveStream* receive_stream : video_receive_streams_)
>       StartEncodedFrameLogs(receive_stream);
>-    StartEncodedFrameLogs(video_send_streams_[0]);
>-
>-    // Start sending and receiving video.
>-    if (params_.video[0].enabled) {
>-      for (VideoReceiveStream* video_receive_stream : video_receive_streams_)
>-        video_receive_stream->Start();
>-      for (VideoSendStream* video_send_stream : video_send_streams_)
>-        video_send_stream->Start();
>-      for (auto& video_capturer : video_capturers_)
>-        video_capturer->Start();
>-    }
>-
>-    if (params_.audio.enabled) {
>-      // Start receiving audio.
>-      audio_receive_stream->Start();
>-
>-      // Start sending audio.
>-      audio_send_stream_->Start();
>-    }
>+    StartEncodedFrameLogs(GetVideoSendStream());
>+    Start();
>   });
> 
>   test::PressEnterToContinue();
> 
>   task_queue_.SendTask([&]() {
>-    if (params_.audio.enabled) {
>-      // Stop sending audio.
>-      audio_send_stream_->Stop();
>-
>-      // Stop receiving audio.
>-      audio_receive_stream->Stop();
>-    }
>-
>-    // Stop receiving and sending video.
>-    if (params_.video[0].enabled) {
>-      for (auto& video_capturer : video_capturers_)
>-        video_capturer->Stop();
>-      for (VideoSendStream* video_send_stream : video_send_streams_)
>-        video_send_stream->Stop();
>-      DestroyStreams();
>-    }
>+    Stop();
>+    DestroyStreams();
> 
>     video_capturers_.clear();
>     send_transport.reset();
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_quality_test.h b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_quality_test.h
>index 20f899d124f..5bef9483e24 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_quality_test.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_quality_test.h
>@@ -15,89 +15,28 @@
> #include <string>
> #include <vector>
> 
>-#include "media/engine/simulcast_encoder_adapter.h"
>+#include "api/fec_controller.h"
>+#include "api/test/video_quality_test_fixture.h"
>+#include "call/fake_network_pipe.h"
>+#include "media/engine/internalencoderfactory.h"
> #include "test/call_test.h"
> #include "test/frame_generator.h"
> #include "test/layer_filtering_transport.h"
> 
> namespace webrtc {
> 
>-class VideoQualityTest : public test::CallTest {
>+class VideoQualityTest :
>+    public test::CallTest, public VideoQualityTestFixtureInterface {
>  public:
>-  // Parameters are grouped into smaller structs to make it easier to set
>-  // the desired elements and skip unused, using aggregate initialization.
>-  // Unfortunately, C++11 (as opposed to C11) doesn't support unnamed structs,
>-  // which makes the implementation of VideoQualityTest a bit uglier.
>-  struct Params {
>-    Params();
>-    ~Params();
>-    struct CallConfig {
>-      bool send_side_bwe;
>-      Call::Config::BitrateConfig call_bitrate_config;
>-      int num_thumbnails;
>-      // Indicates if secondary_(video|ss|screenshare) structures are used.
>-      bool dual_video;
>-    } call;
>-    struct Video {
>-      bool enabled;
>-      size_t width;
>-      size_t height;
>-      int32_t fps;
>-      int min_bitrate_bps;
>-      int target_bitrate_bps;
>-      int max_bitrate_bps;
>-      bool suspend_below_min_bitrate;
>-      std::string codec;
>-      int num_temporal_layers;
>-      int selected_tl;
>-      int min_transmit_bps;
>-      bool ulpfec;
>-      bool flexfec;
>-      std::string clip_name;  // "Generator" to generate frames instead.
>-      size_t capture_device_index;
>-    } video[2];
>-    struct Audio {
>-      bool enabled;
>-      bool sync_video;
>-      bool dtx;
>-    } audio;
>-    struct Screenshare {
>-      bool enabled;
>-      bool generate_slides;
>-      int32_t slide_change_interval;
>-      int32_t scroll_duration;
>-      std::vector<std::string> slides;
>-    } screenshare[2];
>-    struct Analyzer {
>-      std::string test_label;
>-      double avg_psnr_threshold;  // (*)
>-      double avg_ssim_threshold;  // (*)
>-      int test_durations_secs;
>-      std::string graph_data_output_filename;
>-      std::string graph_title;
>-    } analyzer;
>-    FakeNetworkPipe::Config pipe;
>-    struct SS {                          // Spatial scalability.
>-      std::vector<VideoStream> streams;  // If empty, one stream is assumed.
>-      size_t selected_stream;
>-      int num_spatial_layers;
>-      int selected_sl;
>-      // If empty, bitrates are generated in VP9Impl automatically.
>-      std::vector<SpatialLayer> spatial_layers;
>-      // If set, default parameters will be used instead of |streams|.
>-      bool infer_streams;
>-    } ss[2];
>-    struct Logging {
>-      bool logs;
>-      std::string rtc_event_log_name;
>-      std::string rtp_dump_name;
>-      std::string encoded_frame_base_path;
>-    } logging;
>-  };
>-
>-  VideoQualityTest();
>-  void RunWithAnalyzer(const Params& params);
>-  void RunWithRenderers(const Params& params);
>+  explicit VideoQualityTest(
>+      std::unique_ptr<FecControllerFactoryInterface> fec_controller_factory);
>+
>+  void RunWithAnalyzer(const Params& params) override;
>+  void RunWithRenderers(const Params& params) override;
>+
>+  const std::map<uint8_t, webrtc::MediaType>& payload_type_map() override {
>+    return payload_type_map_;
>+  }
> 
>   static void FillScalabilitySettings(
>       Params* params,
>@@ -107,8 +46,14 @@ class VideoQualityTest : public test::CallTest {
>       size_t selected_stream,
>       int num_spatial_layers,
>       int selected_sl,
>+      InterLayerPredMode inter_layer_pred,
>       const std::vector<std::string>& sl_descriptors);
> 
>+  // Helper static methods.
>+  static VideoStream DefaultVideoStream(const Params& params, size_t video_idx);
>+  static VideoStream DefaultThumbnailStream();
>+  static std::vector<int> ParseCSV(const std::string& str);
>+
>  protected:
>   std::map<uint8_t, webrtc::MediaType> payload_type_map_;
> 
>@@ -120,22 +65,23 @@ class VideoQualityTest : public test::CallTest {
>   std::string GenerateGraphTitle() const;
>   void CheckParams();
> 
>-  // Helper static methods.
>-  static VideoStream DefaultVideoStream(const Params& params, size_t video_idx);
>-  static VideoStream DefaultThumbnailStream();
>-  static std::vector<int> ParseCSV(const std::string& str);
>-
>   // Helper methods for setting up the call.
>-  void CreateVideoStreams();
>-  void DestroyStreams();
>   void CreateCapturers();
>   std::unique_ptr<test::FrameGenerator> CreateFrameGenerator(size_t video_idx);
>   void SetupThumbnailCapturers(size_t num_thumbnail_streams);
>+  std::unique_ptr<VideoEncoder> CreateVideoEncoder(
>+      const SdpVideoFormat& format);
>   void SetupVideo(Transport* send_transport, Transport* recv_transport);
>   void SetupThumbnails(Transport* send_transport, Transport* recv_transport);
>+  void StartAudioStreams();
>+  void StartThumbnailCapture();
>+  void StopThumbnailCapture();
>+  void StartThumbnails();
>+  void StopThumbnails();
>   void DestroyThumbnailStreams();
>-  void SetupAudio(Transport* transport,
>-                  AudioReceiveStream** audio_receive_stream);
>+  void InitializeAudioDevice(Call::Config* send_call_config,
>+                             Call::Config* recv_call_config);
>+  void SetupAudio(Transport* transport);
> 
>   void StartEncodedFrameLogs(VideoSendStream* stream);
>   void StartEncodedFrameLogs(VideoReceiveStream* stream);
>@@ -143,33 +89,25 @@ class VideoQualityTest : public test::CallTest {
>   virtual std::unique_ptr<test::LayerFilteringTransport> CreateSendTransport();
>   virtual std::unique_ptr<test::DirectTransport> CreateReceiveTransport();
> 
>-  std::vector<std::unique_ptr<test::VideoCapturer>> video_capturers_;
>   std::vector<std::unique_ptr<test::VideoCapturer>> thumbnail_capturers_;
>-  std::vector<std::unique_ptr<VideoEncoder>> video_encoders_;
>+  Clock* const clock_;
> 
>-  std::vector<std::unique_ptr<VideoEncoder>> thumbnail_encoders_;
>+  test::FunctionVideoEncoderFactory video_encoder_factory_;
>+  InternalEncoderFactory internal_encoder_factory_;
>   std::vector<VideoSendStream::Config> thumbnail_send_configs_;
>   std::vector<VideoEncoderConfig> thumbnail_encoder_configs_;
>   std::vector<VideoSendStream*> thumbnail_send_streams_;
>   std::vector<VideoReceiveStream::Config> thumbnail_receive_configs_;
>   std::vector<VideoReceiveStream*> thumbnail_receive_streams_;
> 
>-  std::vector<VideoSendStream::Config> video_send_configs_;
>-  std::vector<VideoEncoderConfig> video_encoder_configs_;
>-  std::vector<VideoSendStream*> video_send_streams_;
>-
>-  Clock* const clock_;
>-
>   int receive_logs_;
>   int send_logs_;
> 
>-  VideoSendStream::DegradationPreference degradation_preference_ =
>-      VideoSendStream::DegradationPreference::kMaintainFramerate;
>   Params params_;
> 
>-  std::unique_ptr<webrtc::RtcEventLog> recv_event_log_;
>-  std::unique_ptr<webrtc::RtcEventLog> send_event_log_;
>-
>+  // Note: not same as similarly named member in CallTest. This is the number of
>+  // separate send streams, the one in CallTest is the number of substreams for
>+  // a single send stream.
>   size_t num_video_streams_;
> };
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_receive_stream.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_receive_stream.cc
>index 17ddb4864f2..088c0eb0968 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_receive_stream.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_receive_stream.cc
>@@ -16,14 +16,14 @@
> #include <string>
> #include <utility>
> 
>-#include "api/optional.h"
>+#include "absl/memory/memory.h"
>+#include "absl/types/optional.h"
> #include "call/rtp_stream_receiver_controller_interface.h"
> #include "call/rtx_receive_stream.h"
> #include "common_types.h"  // NOLINT(build/include)
> #include "common_video/h264/profile_level_id.h"
>-#include "common_video/libyuv/include/webrtc_libyuv.h"
> #include "common_video/include/incoming_video_stream.h"
>-#include "modules/rtp_rtcp/include/rtp_receiver.h"
>+#include "common_video/libyuv/include/webrtc_libyuv.h"
> #include "modules/rtp_rtcp/include/rtp_rtcp.h"
> #include "modules/utility/include/process_thread.h"
> #include "modules/video_coding/frame_object.h"
>@@ -34,7 +34,6 @@
> #include "rtc_base/checks.h"
> #include "rtc_base/location.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/ptr_util.h"
> #include "rtc_base/trace_event.h"
> #include "system_wrappers/include/clock.h"
> #include "system_wrappers/include/field_trial.h"
>@@ -49,7 +48,6 @@ VideoCodec CreateDecoderVideoCodec(const VideoReceiveStream::Decoder& decoder) {
>   memset(&codec, 0, sizeof(codec));
> 
>   codec.plType = decoder.payload_type;
>-  strncpy(codec.plName, decoder.payload_name.c_str(), sizeof(codec.plName));
>   codec.codecType = PayloadStringToCodecType(decoder.payload_name);
> 
>   if (codec.codecType == kVideoCodecVP8) {
>@@ -60,13 +58,11 @@ VideoCodec CreateDecoderVideoCodec(const VideoReceiveStream::Decoder& decoder) {
>     *(codec.H264()) = VideoEncoder::GetDefaultH264Settings();
>     codec.H264()->profile =
>         H264::ParseSdpProfileLevelId(decoder.codec_params)->profile;
>-  } else if (codec.codecType == kVideoCodecStereo) {
>+  } else if (codec.codecType == kVideoCodecMultiplex) {
>     VideoReceiveStream::Decoder associated_decoder = decoder;
>     associated_decoder.payload_name = CodecTypeToPayloadString(kVideoCodecVP9);
>     VideoCodec associated_codec = CreateDecoderVideoCodec(associated_decoder);
>-    associated_codec.codecType = kVideoCodecStereo;
>-    strncpy(associated_codec.plName, decoder.payload_name.c_str(),
>-            sizeof(associated_codec.plName));
>+    associated_codec.codecType = kVideoCodecMultiplex;
>     return associated_codec;
>   }
> 
>@@ -104,16 +100,15 @@ VideoReceiveStream::VideoReceiveStream(
>       video_receiver_(clock_, nullptr, this, timing_.get(), this, this),
>       stats_proxy_(&config_, clock_),
>       rtp_video_stream_receiver_(&transport_adapter_,
>-                                 call_stats_->rtcp_rtt_stats(),
>+                                 call_stats,
>                                  packet_router,
>                                  &config_,
>                                  rtp_receive_statistics_.get(),
>                                  &stats_proxy_,
>                                  process_thread_,
>-                                 this,  // NackSender
>-                                 this,  // KeyFrameRequestSender
>-                                 this,  // OnCompleteFrameCallback
>-                                 timing_.get()),
>+                                 this,   // NackSender
>+                                 this,   // KeyFrameRequestSender
>+                                 this),  // OnCompleteFrameCallback
>       rtp_stream_sync_(this) {
>   RTC_LOG(LS_INFO) << "VideoReceiveStream: " << config_.ToString();
> 
>@@ -145,7 +140,7 @@ VideoReceiveStream::VideoReceiveStream(
>   media_receiver_ = receiver_controller->CreateReceiver(
>       config_.rtp.remote_ssrc, &rtp_video_stream_receiver_);
>   if (config_.rtp.rtx_ssrc) {
>-    rtx_receive_stream_ = rtc::MakeUnique<RtxReceiveStream>(
>+    rtx_receive_stream_ = absl::make_unique<RtxReceiveStream>(
>         &rtp_video_stream_receiver_, config.rtp.rtx_associated_payload_types,
>         config_.rtp.remote_ssrc, rtp_receive_statistics_.get());
>     rtx_receiver_ = receiver_controller->CreateReceiver(
>@@ -184,8 +179,6 @@ void VideoReceiveStream::Start() {
>                           rtp_video_stream_receiver_.IsUlpfecEnabled();
> 
>   frame_buffer_->Start();
>-  call_stats_->RegisterStatsObserver(&rtp_video_stream_receiver_);
>-  call_stats_->RegisterStatsObserver(this);
> 
>   if (rtp_video_stream_receiver_.IsRetransmissionsEnabled() &&
>       protected_by_fec) {
>@@ -209,8 +202,7 @@ void VideoReceiveStream::Start() {
>     video_receiver_.RegisterExternalDecoder(decoder.decoder,
>                                             decoder.payload_type);
>     VideoCodec codec = CreateDecoderVideoCodec(decoder);
>-    RTC_CHECK(rtp_video_stream_receiver_.AddReceiveCodec(codec,
>-                                                         decoder.codec_params));
>+    rtp_video_stream_receiver_.AddReceiveCodec(codec, decoder.codec_params);
>     RTC_CHECK_EQ(VCM_OK, video_receiver_.RegisterReceiveCodec(
>                              &codec, num_cpu_cores_, false));
>   }
>@@ -220,12 +212,16 @@ void VideoReceiveStream::Start() {
>       &rtp_video_stream_receiver_,
>       rtp_video_stream_receiver_.IsRetransmissionsEnabled(), protected_by_fec,
>       &stats_proxy_, renderer));
>-  // Register the channel to receive stats updates.
>-  call_stats_->RegisterStatsObserver(video_stream_decoder_.get());
>+
>+  // Make sure we register as a stats observer *after* we've prepared the
>+  // |video_stream_decoder_|.
>+  call_stats_->RegisterStatsObserver(this);
> 
>   process_thread_->RegisterModule(&video_receiver_, RTC_FROM_HERE);
> 
>   // Start the decode thread
>+  video_receiver_.DecoderThreadStarting();
>+  stats_proxy_.DecoderThreadStarting();
>   decode_thread_.Start();
>   rtp_video_stream_receiver_.StartReceive();
> }
>@@ -234,9 +230,11 @@ void VideoReceiveStream::Stop() {
>   RTC_DCHECK_CALLED_SEQUENTIALLY(&worker_sequence_checker_);
>   rtp_video_stream_receiver_.StopReceive();
> 
>+  stats_proxy_.OnUniqueFramesCounted(
>+      rtp_video_stream_receiver_.GetUniqueFramesSeen());
>+
>   frame_buffer_->Stop();
>   call_stats_->DeregisterStatsObserver(this);
>-  call_stats_->DeregisterStatsObserver(&rtp_video_stream_receiver_);
>   process_thread_->DeRegisterModule(&video_receiver_);
> 
>   if (decode_thread_.IsRunning()) {
>@@ -246,6 +244,8 @@ void VideoReceiveStream::Stop() {
>     video_receiver_.TriggerDecoderShutdown();
> 
>     decode_thread_.Stop();
>+    video_receiver_.DecoderThreadStopped();
>+    stats_proxy_.DecoderThreadStopped();
>     // Deregister external decoders so they are no longer running during
>     // destruction. This effectively stops the VCM since the decoder thread is
>     // stopped, the VCM is deregistered and no asynchronous decoder threads are
>@@ -254,7 +254,6 @@ void VideoReceiveStream::Stop() {
>       video_receiver_.RegisterExternalDecoder(nullptr, decoder.payload_type);
>   }
> 
>-  call_stats_->DeregisterStatsObserver(video_stream_decoder_.get());
>   video_stream_decoder_.reset();
>   incoming_video_stream_.reset();
>   transport_adapter_.Disable();
>@@ -299,10 +298,9 @@ void VideoReceiveStream::OnFrame(const VideoFrame& video_frame) {
>   // function itself, another in GetChannel() and a third in
>   // GetPlayoutTimestamp.  Seems excessive.  Anyhow, I'm assuming the function
>   // succeeds most of the time, which leads to grabbing a fourth lock.
>-  if (rtp_stream_sync_.GetStreamSyncOffsetInMs(video_frame.timestamp(),
>-                                               video_frame.render_time_ms(),
>-                                               &sync_offset_ms,
>-                                               &estimated_freq_khz)) {
>+  if (rtp_stream_sync_.GetStreamSyncOffsetInMs(
>+          video_frame.timestamp(), video_frame.render_time_ms(),
>+          &sync_offset_ms, &estimated_freq_khz)) {
>     // TODO(tommi): OnSyncOffsetUpdated grabs a lock.
>     stats_proxy_.OnSyncOffsetUpdated(sync_offset_ms, estimated_freq_khz);
>   }
>@@ -324,11 +322,6 @@ EncodedImageCallback::Result VideoReceiveStream::OnEncodedImage(
>   if (codec_specific_info->codecType == kVideoCodecVP8) {
>     simulcast_idx = codec_specific_info->codecSpecific.VP8.simulcastIdx;
>   }
>-  if (config_.pre_decode_callback) {
>-    config_.pre_decode_callback->EncodedFrameCallback(EncodedFrame(
>-        encoded_image._buffer, encoded_image._length, encoded_image._frameType,
>-        simulcast_idx, encoded_image._timeStamp));
>-  }
>   {
>     rtc::CritScope lock(&ivf_writer_lock_);
>     if (ivf_writer_.get()) {
>@@ -352,14 +345,21 @@ void VideoReceiveStream::RequestKeyFrame() {
> }
> 
> void VideoReceiveStream::OnCompleteFrame(
>-    std::unique_ptr<video_coding::FrameObject> frame) {
>+    std::unique_ptr<video_coding::EncodedFrame> frame) {
>+  // TODO(webrtc:9249): Workaround to allow decoding of VP9 SVC stream with
>+  // partially enabled inter-layer prediction.
>+  frame->id.spatial_layer = 0;
>+
>   int64_t last_continuous_pid = frame_buffer_->InsertFrame(std::move(frame));
>   if (last_continuous_pid != -1)
>     rtp_video_stream_receiver_.FrameContinuous(last_continuous_pid);
> }
> 
> void VideoReceiveStream::OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) {
>+  RTC_DCHECK_CALLED_SEQUENTIALLY(&module_process_sequence_checker_);
>   frame_buffer_->UpdateRtt(max_rtt_ms);
>+  rtp_video_stream_receiver_.UpdateRtt(max_rtt_ms);
>+  video_stream_decoder_->UpdateRtt(max_rtt_ms);
> }
> 
> int VideoReceiveStream::id() const {
>@@ -367,28 +367,15 @@ int VideoReceiveStream::id() const {
>   return config_.rtp.remote_ssrc;
> }
> 
>-rtc::Optional<Syncable::Info> VideoReceiveStream::GetInfo() const {
>+absl::optional<Syncable::Info> VideoReceiveStream::GetInfo() const {
>   RTC_DCHECK_CALLED_SEQUENTIALLY(&module_process_sequence_checker_);
>-  Syncable::Info info;
>-
>-  RtpReceiver* rtp_receiver = rtp_video_stream_receiver_.GetRtpReceiver();
>-  RTC_DCHECK(rtp_receiver);
>-  if (!rtp_receiver->GetLatestTimestamps(
>-          &info.latest_received_capture_timestamp,
>-          &info.latest_receive_time_ms))
>-    return rtc::nullopt;
>-
>-  RtpRtcp* rtp_rtcp = rtp_video_stream_receiver_.rtp_rtcp();
>-  RTC_DCHECK(rtp_rtcp);
>-  if (rtp_rtcp->RemoteNTP(&info.capture_time_ntp_secs,
>-                          &info.capture_time_ntp_frac,
>-                          nullptr,
>-                          nullptr,
>-                          &info.capture_time_source_clock) != 0) {
>-    return rtc::nullopt;
>-  }
>+  absl::optional<Syncable::Info> info =
>+      rtp_video_stream_receiver_.GetSyncInfo();
>+
>+  if (!info)
>+    return absl::nullopt;
> 
>-  info.current_delay_ms = video_receiver_.Delay();
>+  info->current_delay_ms = video_receiver_.Delay();
>   return info;
> }
> 
>@@ -413,14 +400,13 @@ bool VideoReceiveStream::Decode() {
>   static const int kMaxWaitForKeyFrameMs = 200;
> 
>   int wait_ms = keyframe_required_ ? kMaxWaitForKeyFrameMs : kMaxWaitForFrameMs;
>-  std::unique_ptr<video_coding::FrameObject> frame;
>+  std::unique_ptr<video_coding::EncodedFrame> frame;
>   // TODO(philipel): Call NextFrame with |keyframe_required| argument when
>   //                 downstream project has been fixed.
>   video_coding::FrameBuffer::ReturnReason res =
>       frame_buffer_->NextFrame(wait_ms, &frame);
> 
>   if (res == video_coding::FrameBuffer::ReturnReason::kStopped) {
>-    video_receiver_.DecodingStopped();
>     return false;
>   }
> 
>@@ -432,7 +418,7 @@ bool VideoReceiveStream::Decode() {
>         decode_result == WEBRTC_VIDEO_CODEC_OK_REQUEST_KEYFRAME) {
>       keyframe_required_ = false;
>       frame_decoded_ = true;
>-      rtp_video_stream_receiver_.FrameDecoded(frame->picture_id);
>+      rtp_video_stream_receiver_.FrameDecoded(frame->id.picture_id);
> 
>       if (decode_result == WEBRTC_VIDEO_CODEC_OK_REQUEST_KEYFRAME)
>         RequestKeyFrame();
>@@ -447,9 +433,9 @@ bool VideoReceiveStream::Decode() {
>   } else {
>     RTC_DCHECK_EQ(res, video_coding::FrameBuffer::ReturnReason::kTimeout);
>     int64_t now_ms = clock_->TimeInMilliseconds();
>-    rtc::Optional<int64_t> last_packet_ms =
>+    absl::optional<int64_t> last_packet_ms =
>         rtp_video_stream_receiver_.LastReceivedPacketMs();
>-    rtc::Optional<int64_t> last_keyframe_packet_ms =
>+    absl::optional<int64_t> last_keyframe_packet_ms =
>         rtp_video_stream_receiver_.LastReceivedKeyframePacketMs();
> 
>     // To avoid spamming keyframe requests for a stream that is not active we
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_receive_stream.h b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_receive_stream.h
>index 60afd8d532a..5b1289443be 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_receive_stream.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_receive_stream.h
>@@ -101,14 +101,14 @@ class VideoReceiveStream : public webrtc::VideoReceiveStream,
> 
>   // Implements video_coding::OnCompleteFrameCallback.
>   void OnCompleteFrame(
>-      std::unique_ptr<video_coding::FrameObject> frame) override;
>+      std::unique_ptr<video_coding::EncodedFrame> frame) override;
> 
>   // Implements CallStatsObserver::OnRttUpdate
>   void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) override;
> 
>   // Implements Syncable.
>   int id() const override;
>-  rtc::Optional<Syncable::Info> GetInfo() const override;
>+  absl::optional<Syncable::Info> GetInfo() const override;
>   uint32_t GetPlayoutTimestamp() const override;
>   void SetMinimumPlayoutDelay(int delay_ms) override;
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_receive_stream_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_receive_stream_unittest.cc
>index 97a447ba771..081bbb566dc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_receive_stream_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_receive_stream_unittest.cc
>@@ -10,8 +10,8 @@
> 
> #include <vector>
> 
>-#include "test/gtest.h"
> #include "test/gmock.h"
>+#include "test/gtest.h"
> 
> #include "api/video_codecs/video_decoder.h"
> #include "call/rtp_stream_receiver_controller.h"
>@@ -50,10 +50,9 @@ class MockVideoDecoder : public VideoDecoder {
>  public:
>   MOCK_METHOD2(InitDecode,
>                int32_t(const VideoCodec* config, int32_t number_of_cores));
>-  MOCK_METHOD5(Decode,
>+  MOCK_METHOD4(Decode,
>                int32_t(const EncodedImage& input,
>                        bool missing_frames,
>-                       const RTPFragmentationHeader* fragmentation,
>                        const CodecSpecificInfo* codec_specific_info,
>                        int64_t render_time_ms));
>   MOCK_METHOD1(RegisterDecodeCompleteCallback,
>@@ -67,10 +66,10 @@ class MockVideoDecoder : public VideoDecoder {
> class VideoReceiveStreamTest : public testing::Test {
>  public:
>   VideoReceiveStreamTest()
>-      : override_field_trials_(kNewJitterBufferFieldTrialEnabled),
>+      : process_thread_(ProcessThread::Create("TestThread")),
>+        override_field_trials_(kNewJitterBufferFieldTrialEnabled),
>         config_(&mock_transport_),
>-        call_stats_(Clock::GetRealTimeClock()),
>-        process_thread_(ProcessThread::Create("TestThread")) {}
>+        call_stats_(Clock::GetRealTimeClock(), process_thread_.get()) {}
> 
>   void SetUp() {
>     constexpr int kDefaultNumCpuCores = 2;
>@@ -91,11 +90,12 @@ class VideoReceiveStreamTest : public testing::Test {
>     config_.decoders.push_back(null_decoder);
> 
>     video_receive_stream_.reset(new webrtc::internal::VideoReceiveStream(
>-        &rtp_stream_receiver_controller_, kDefaultNumCpuCores,
>-        &packet_router_, config_.Copy(), process_thread_.get(), &call_stats_));
>+        &rtp_stream_receiver_controller_, kDefaultNumCpuCores, &packet_router_,
>+        config_.Copy(), process_thread_.get(), &call_stats_));
>   }
> 
>  protected:
>+  std::unique_ptr<ProcessThread> process_thread_;
>   webrtc::test::ScopedFieldTrials override_field_trials_;
>   VideoReceiveStream::Config config_;
>   CallStats call_stats_;
>@@ -104,7 +104,6 @@ class VideoReceiveStreamTest : public testing::Test {
>   cricket::FakeVideoRenderer fake_renderer_;
>   MockTransport mock_transport_;
>   PacketRouter packet_router_;
>-  std::unique_ptr<ProcessThread> process_thread_;
>   RtpStreamReceiverController rtp_stream_receiver_controller_;
>   std::unique_ptr<webrtc::internal::VideoReceiveStream> video_receive_stream_;
> };
>@@ -128,7 +127,7 @@ TEST_F(VideoReceiveStreamTest, CreateFrameFromH264FmtpSpropAndIdr) {
>       }));
>   EXPECT_CALL(mock_h264_video_decoder_, RegisterDecodeCompleteCallback(_));
>   video_receive_stream_->Start();
>-  EXPECT_CALL(mock_h264_video_decoder_, Decode(_, false, _, _, _));
>+  EXPECT_CALL(mock_h264_video_decoder_, Decode(_, false, _, _));
>   RtpPacketReceived parsed_packet;
>   ASSERT_TRUE(parsed_packet.Parse(rtppacket.data(), rtppacket.size()));
>   rtp_stream_receiver_controller_.OnRtpPacket(parsed_packet);
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream.cc
>index a4d657be34e..9271f764cf9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream.cc
>@@ -9,545 +9,52 @@
>  */
> #include "video/video_send_stream.h"
> 
>-#include <algorithm>
>-#include <cmath>
>-#include <sstream>
>-#include <string>
> #include <utility>
>-#include <vector>
> 
>-#include "call/rtp_transport_controller_send_interface.h"
>-#include "call/video_send_stream.h"
>-#include "common_types.h"  // NOLINT(build/include)
>-#include "common_video/include/video_bitrate_allocator.h"
>-#include "modules/bitrate_controller/include/bitrate_controller.h"
>-#include "modules/congestion_controller/include/send_side_congestion_controller.h"
>-#include "modules/pacing/packet_router.h"
>-#include "modules/rtp_rtcp/include/rtp_rtcp.h"
>+#include "api/video/video_stream_encoder_create.h"
> #include "modules/rtp_rtcp/source/rtp_sender.h"
>-#include "modules/utility/include/process_thread.h"
>-#include "modules/video_coding/utility/ivf_file_writer.h"
>-#include "rtc_base/checks.h"
>-#include "rtc_base/experiments/alr_experiment.h"
>-#include "rtc_base/file.h"
>-#include "rtc_base/location.h"
> #include "rtc_base/logging.h"
>-#include "rtc_base/trace_event.h"
>-#include "rtc_base/weak_ptr.h"
>-#include "system_wrappers/include/field_trial.h"
>-#include "video/call_stats.h"
>-#include "video/payload_router.h"
>+#include "video/video_send_stream_impl.h"
> 
> namespace webrtc {
> 
>-static const int kMinSendSidePacketHistorySize = 600;
> namespace {
> 
>-// We don't do MTU discovery, so assume that we have the standard ethernet MTU.
>-const size_t kPathMTU = 1500;
>-
>-std::vector<RtpRtcp*> CreateRtpRtcpModules(
>-    Transport* outgoing_transport,
>-    RtcpIntraFrameObserver* intra_frame_callback,
>-    RtcpBandwidthObserver* bandwidth_callback,
>-    RtpTransportControllerSendInterface* transport,
>-    RtcpRttStats* rtt_stats,
>-    FlexfecSender* flexfec_sender,
>-    SendStatisticsProxy* stats_proxy,
>-    SendDelayStats* send_delay_stats,
>-    RtcEventLog* event_log,
>-    RateLimiter* retransmission_rate_limiter,
>-    OverheadObserver* overhead_observer,
>-    size_t num_modules,
>-    RtpKeepAliveConfig keepalive_config) {
>-  RTC_DCHECK_GT(num_modules, 0);
>-  RtpRtcp::Configuration configuration;
>-  configuration.audio = false;
>-  configuration.receiver_only = false;
>-  configuration.flexfec_sender = flexfec_sender;
>-  configuration.outgoing_transport = outgoing_transport;
>-  configuration.intra_frame_callback = intra_frame_callback;
>-  configuration.bandwidth_callback = bandwidth_callback;
>-  configuration.transport_feedback_callback =
>-      transport->transport_feedback_observer();
>-  configuration.rtt_stats = rtt_stats;
>-  configuration.rtcp_packet_type_counter_observer = stats_proxy;
>-  configuration.paced_sender = transport->packet_sender();
>-  configuration.transport_sequence_number_allocator =
>-      transport->packet_router();
>-  configuration.send_bitrate_observer = stats_proxy;
>-  configuration.send_frame_count_observer = stats_proxy;
>-  configuration.send_side_delay_observer = stats_proxy;
>-  configuration.send_packet_observer = send_delay_stats;
>-  configuration.event_log = event_log;
>-  configuration.retransmission_rate_limiter = retransmission_rate_limiter;
>-  configuration.overhead_observer = overhead_observer;
>-  configuration.keepalive_config = keepalive_config;
>-  std::vector<RtpRtcp*> modules;
>-  for (size_t i = 0; i < num_modules; ++i) {
>-    RtpRtcp* rtp_rtcp = RtpRtcp::CreateRtpRtcp(configuration);
>-    rtp_rtcp->SetSendingStatus(false);
>-    rtp_rtcp->SetSendingMediaStatus(false);
>-    rtp_rtcp->SetRTCPStatus(RtcpMode::kCompound);
>-    modules.push_back(rtp_rtcp);
>-  }
>-  return modules;
>-}
>-
>-// TODO(brandtr): Update this function when we support multistream protection.
>-std::unique_ptr<FlexfecSender> MaybeCreateFlexfecSender(
>-    const VideoSendStream::Config& config,
>-    const std::map<uint32_t, RtpState>& suspended_ssrcs) {
>-  if (config.rtp.flexfec.payload_type < 0) {
>-    return nullptr;
>-  }
>-  RTC_DCHECK_GE(config.rtp.flexfec.payload_type, 0);
>-  RTC_DCHECK_LE(config.rtp.flexfec.payload_type, 127);
>-  if (config.rtp.flexfec.ssrc == 0) {
>-    RTC_LOG(LS_WARNING) << "FlexFEC is enabled, but no FlexFEC SSRC given. "
>-                           "Therefore disabling FlexFEC.";
>-    return nullptr;
>-  }
>-  if (config.rtp.flexfec.protected_media_ssrcs.empty()) {
>-    RTC_LOG(LS_WARNING)
>-        << "FlexFEC is enabled, but no protected media SSRC given. "
>-           "Therefore disabling FlexFEC.";
>-    return nullptr;
>-  }
>-
>-  if (config.rtp.ssrcs.size() > 1) {
>-    RTC_LOG(LS_WARNING)
>-        << "Both FlexFEC and simulcast are enabled. This "
>-           "combination is however not supported by our current "
>-           "FlexFEC implementation. Therefore disabling FlexFEC.";
>-    return nullptr;
>-  }
>-
>-  if (config.rtp.flexfec.protected_media_ssrcs.size() > 1) {
>-    RTC_LOG(LS_WARNING)
>-        << "The supplied FlexfecConfig contained multiple protected "
>-           "media streams, but our implementation currently only "
>-           "supports protecting a single media stream. "
>-           "To avoid confusion, disabling FlexFEC completely.";
>-    return nullptr;
>-  }
>-
>-  const RtpState* rtp_state = nullptr;
>-  auto it = suspended_ssrcs.find(config.rtp.flexfec.ssrc);
>-  if (it != suspended_ssrcs.end()) {
>-    rtp_state = &it->second;
>-  }
>-
>-  RTC_DCHECK_EQ(1U, config.rtp.flexfec.protected_media_ssrcs.size());
>-  return std::unique_ptr<FlexfecSender>(new FlexfecSender(
>-      config.rtp.flexfec.payload_type, config.rtp.flexfec.ssrc,
>-      config.rtp.flexfec.protected_media_ssrcs[0], config.rtp.extensions,
>-      RTPSender::FecExtensionSizes(), rtp_state, Clock::GetRealTimeClock()));
>-}
>-
>-bool TransportSeqNumExtensionConfigured(const VideoSendStream::Config& config) {
>-  const std::vector<RtpExtension>& extensions = config.rtp.extensions;
>-  return std::find_if(
>-             extensions.begin(), extensions.end(), [](const RtpExtension& ext) {
>-               return ext.uri == RtpExtension::kTransportSequenceNumberUri;
>-             }) != extensions.end();
>-}
>-
>-const char kForcedFallbackFieldTrial[] =
>-    "WebRTC-VP8-Forced-Fallback-Encoder-v2";
>-
>-rtc::Optional<int> GetFallbackMinBpsFromFieldTrial() {
>-  if (!webrtc::field_trial::IsEnabled(kForcedFallbackFieldTrial))
>-    return rtc::nullopt;
>-
>-  std::string group =
>-      webrtc::field_trial::FindFullName(kForcedFallbackFieldTrial);
>-  if (group.empty())
>-    return rtc::nullopt;
>-
>-  int min_pixels;
>-  int max_pixels;
>-  int min_bps;
>-  if (sscanf(group.c_str(), "Enabled-%d,%d,%d", &min_pixels, &max_pixels,
>-             &min_bps) != 3) {
>-    return rtc::nullopt;
>-  }
>-
>-  if (min_bps <= 0)
>-    return rtc::nullopt;
>-
>-  return min_bps;
>-}
>-
>-int GetEncoderMinBitrateBps() {
>-  const int kDefaultEncoderMinBitrateBps = 30000;
>-  return GetFallbackMinBpsFromFieldTrial().value_or(
>-      kDefaultEncoderMinBitrateBps);
>-}
>-
>-bool PayloadTypeSupportsSkippingFecPackets(const std::string& payload_name) {
>-  const VideoCodecType codecType = PayloadStringToCodecType(payload_name);
>-  if (codecType == kVideoCodecVP8 || codecType == kVideoCodecVP9) {
>-    return true;
>-  }
>-  return false;
>-}
>-
>-int CalculateMaxPadBitrateBps(std::vector<VideoStream> streams,
>-                              int min_transmit_bitrate_bps,
>-                              bool pad_to_min_bitrate) {
>-  int pad_up_to_bitrate_bps = 0;
>-  // Calculate max padding bitrate for a multi layer codec.
>-  if (streams.size() > 1) {
>-    // Pad to min bitrate of the highest layer.
>-    pad_up_to_bitrate_bps = streams[streams.size() - 1].min_bitrate_bps;
>-    // Add target_bitrate_bps of the lower layers.
>-    for (size_t i = 0; i < streams.size() - 1; ++i)
>-      pad_up_to_bitrate_bps += streams[i].target_bitrate_bps;
>-  } else if (pad_to_min_bitrate) {
>-    pad_up_to_bitrate_bps = streams[0].min_bitrate_bps;
>+size_t CalculateMaxHeaderSize(const RtpConfig& config) {
>+  size_t header_size = kRtpHeaderSize;
>+  size_t extensions_size = 0;
>+  size_t fec_extensions_size = 0;
>+  if (config.extensions.size() > 0) {
>+    RtpHeaderExtensionMap extensions_map(config.extensions);
>+    extensions_size =
>+        extensions_map.GetTotalLengthInBytes(RTPSender::VideoExtensionSizes());
>+    fec_extensions_size =
>+        extensions_map.GetTotalLengthInBytes(RTPSender::FecExtensionSizes());
>+  }
>+  header_size += extensions_size;
>+  if (config.flexfec.payload_type >= 0) {
>+    // All FEC extensions again plus maximum FlexFec overhead.
>+    header_size += fec_extensions_size + 32;
>+  } else {
>+    if (config.ulpfec.ulpfec_payload_type >= 0) {
>+      // Header with all the FEC extensions will be repeated plus maximum
>+      // UlpFec overhead.
>+      header_size += fec_extensions_size + 18;
>+    }
>+    if (config.ulpfec.red_payload_type >= 0) {
>+      header_size += 1;  // RED header.
>+    }
>   }
>-
>-  pad_up_to_bitrate_bps =
>-      std::max(pad_up_to_bitrate_bps, min_transmit_bitrate_bps);
>-
>-  return pad_up_to_bitrate_bps;
>-}
>-
>-uint32_t CalculateOverheadRateBps(int packets_per_second,
>-                                  size_t overhead_bytes_per_packet,
>-                                  uint32_t max_overhead_bps) {
>-  uint32_t overhead_bps =
>-      static_cast<uint32_t>(8 * overhead_bytes_per_packet * packets_per_second);
>-  return std::min(overhead_bps, max_overhead_bps);
>-}
>-
>-int CalculatePacketRate(uint32_t bitrate_bps, size_t packet_size_bytes) {
>-  size_t packet_size_bits = 8 * packet_size_bytes;
>-  // Ceil for int value of bitrate_bps / packet_size_bits.
>-  return static_cast<int>((bitrate_bps + packet_size_bits - 1) /
>-                          packet_size_bits);
>+  // Additional room for Rtx.
>+  if (config.rtx.payload_type >= 0)
>+    header_size += kRtxHeaderSize;
>+  return header_size;
> }
> 
> }  // namespace
> 
> namespace internal {
> 
>-// VideoSendStreamImpl implements internal::VideoSendStream.
>-// It is created and destroyed on |worker_queue|. The intent is to decrease the
>-// need for locking and to ensure methods are called in sequence.
>-// Public methods except |DeliverRtcp| must be called on |worker_queue|.
>-// DeliverRtcp is called on the libjingle worker thread or a network thread.
>-// An encoder may deliver frames through the EncodedImageCallback on an
>-// arbitrary thread.
>-class VideoSendStreamImpl : public webrtc::BitrateAllocatorObserver,
>-                            public webrtc::OverheadObserver,
>-                            public webrtc::VCMProtectionCallback,
>-                            public VideoStreamEncoder::EncoderSink,
>-                            public VideoBitrateAllocationObserver {
>- public:
>-  VideoSendStreamImpl(
>-      SendStatisticsProxy* stats_proxy,
>-      rtc::TaskQueue* worker_queue,
>-      CallStats* call_stats,
>-      RtpTransportControllerSendInterface* transport,
>-      BitrateAllocator* bitrate_allocator,
>-      SendDelayStats* send_delay_stats,
>-      VideoStreamEncoder* video_stream_encoder,
>-      RtcEventLog* event_log,
>-      const VideoSendStream::Config* config,
>-      int initial_encoder_max_bitrate,
>-      double initial_encoder_bitrate_priority,
>-      std::map<uint32_t, RtpState> suspended_ssrcs,
>-      std::map<uint32_t, RtpPayloadState> suspended_payload_states,
>-      VideoEncoderConfig::ContentType content_type);
>-  ~VideoSendStreamImpl() override;
>-
>-  // RegisterProcessThread register |module_process_thread| with those objects
>-  // that use it. Registration has to happen on the thread were
>-  // |module_process_thread| was created (libjingle's worker thread).
>-  // TODO(perkj): Replace the use of |module_process_thread| with a TaskQueue,
>-  // maybe |worker_queue|.
>-  void RegisterProcessThread(ProcessThread* module_process_thread);
>-  void DeRegisterProcessThread();
>-
>-  void SignalNetworkState(NetworkState state);
>-  bool DeliverRtcp(const uint8_t* packet, size_t length);
>-  void Start();
>-  void Stop();
>-
>-  VideoSendStream::RtpStateMap GetRtpStates() const;
>-  VideoSendStream::RtpPayloadStateMap GetRtpPayloadStates() const;
>-
>-  void EnableEncodedFrameRecording(const std::vector<rtc::PlatformFile>& files,
>-                                   size_t byte_limit);
>-
>-  void SetTransportOverhead(size_t transport_overhead_per_packet);
>-
>- private:
>-  class CheckEncoderActivityTask;
>-  class EncoderReconfiguredTask;
>-
>-  // Implements BitrateAllocatorObserver.
>-  uint32_t OnBitrateUpdated(uint32_t bitrate_bps,
>-                            uint8_t fraction_loss,
>-                            int64_t rtt,
>-                            int64_t probing_interval_ms) override;
>-
>-  // Implements webrtc::VCMProtectionCallback.
>-  int ProtectionRequest(const FecProtectionParams* delta_params,
>-                        const FecProtectionParams* key_params,
>-                        uint32_t* sent_video_rate_bps,
>-                        uint32_t* sent_nack_rate_bps,
>-                        uint32_t* sent_fec_rate_bps) override;
>-
>-  // Implements OverheadObserver.
>-  void OnOverheadChanged(size_t overhead_bytes_per_packet) override;
>-
>-  void OnEncoderConfigurationChanged(std::vector<VideoStream> streams,
>-                                     int min_transmit_bitrate_bps) override;
>-
>-  // Implements EncodedImageCallback. The implementation routes encoded frames
>-  // to the |payload_router_| and |config.pre_encode_callback| if set.
>-  // Called on an arbitrary encoder callback thread.
>-  EncodedImageCallback::Result OnEncodedImage(
>-      const EncodedImage& encoded_image,
>-      const CodecSpecificInfo* codec_specific_info,
>-      const RTPFragmentationHeader* fragmentation) override;
>-
>-  // Implements VideoBitrateAllocationObserver.
>-  void OnBitrateAllocationUpdated(const BitrateAllocation& allocation) override;
>-
>-  void ConfigureProtection();
>-  void ConfigureSsrcs();
>-  void SignalEncoderTimedOut();
>-  void SignalEncoderActive();
>-
>-  const bool send_side_bwe_with_overhead_;
>-
>-  SendStatisticsProxy* const stats_proxy_;
>-  const VideoSendStream::Config* const config_;
>-  std::map<uint32_t, RtpState> suspended_ssrcs_;
>-
>-  ProcessThread* module_process_thread_;
>-  rtc::ThreadChecker module_process_thread_checker_;
>-  rtc::TaskQueue* const worker_queue_;
>-
>-  rtc::CriticalSection encoder_activity_crit_sect_;
>-  CheckEncoderActivityTask* check_encoder_activity_task_
>-      RTC_GUARDED_BY(encoder_activity_crit_sect_);
>-
>-  CallStats* const call_stats_;
>-  RtpTransportControllerSendInterface* const transport_;
>-  BitrateAllocator* const bitrate_allocator_;
>-
>-  // TODO(brandtr): Move ownership to PayloadRouter.
>-  std::unique_ptr<FlexfecSender> flexfec_sender_;
>-
>-  rtc::CriticalSection ivf_writers_crit_;
>-  std::unique_ptr<IvfFileWriter>
>-      file_writers_[kMaxSimulcastStreams] RTC_GUARDED_BY(ivf_writers_crit_);
>-
>-  int max_padding_bitrate_;
>-  int encoder_min_bitrate_bps_;
>-  uint32_t encoder_max_bitrate_bps_;
>-  uint32_t encoder_target_rate_bps_;
>-  double encoder_bitrate_priority_;
>-
>-  VideoStreamEncoder* const video_stream_encoder_;
>-  EncoderRtcpFeedback encoder_feedback_;
>-  ProtectionBitrateCalculator protection_bitrate_calculator_;
>-
>-  RtcpBandwidthObserver* const bandwidth_observer_;
>-  // RtpRtcp modules, declared here as they use other members on construction.
>-  const std::vector<RtpRtcp*> rtp_rtcp_modules_;
>-  PayloadRouter payload_router_;
>-
>-  // |weak_ptr_| to our self. This is used since we can not call
>-  // |weak_ptr_factory_.GetWeakPtr| from multiple sequences but it is ok to copy
>-  // an existing WeakPtr.
>-  rtc::WeakPtr<VideoSendStreamImpl> weak_ptr_;
>-  // |weak_ptr_factory_| must be declared last to make sure all WeakPtr's are
>-  // invalidated before any other members are destroyed.
>-  rtc::WeakPtrFactory<VideoSendStreamImpl> weak_ptr_factory_;
>-
>-  rtc::CriticalSection overhead_bytes_per_packet_crit_;
>-  size_t overhead_bytes_per_packet_
>-      RTC_GUARDED_BY(overhead_bytes_per_packet_crit_);
>-  size_t transport_overhead_bytes_per_packet_;
>-};
>-
>-// TODO(tommi): See if there's a more elegant way to create a task that creates
>-// an object on the correct task queue.
>-class VideoSendStream::ConstructionTask : public rtc::QueuedTask {
>- public:
>-  ConstructionTask(
>-      std::unique_ptr<VideoSendStreamImpl>* send_stream,
>-      rtc::Event* done_event,
>-      SendStatisticsProxy* stats_proxy,
>-      VideoStreamEncoder* video_stream_encoder,
>-      ProcessThread* module_process_thread,
>-      CallStats* call_stats,
>-      RtpTransportControllerSendInterface* transport,
>-      BitrateAllocator* bitrate_allocator,
>-      SendDelayStats* send_delay_stats,
>-      RtcEventLog* event_log,
>-      const VideoSendStream::Config* config,
>-      int initial_encoder_max_bitrate,
>-      double initial_encoder_bitrate_priority,
>-      const std::map<uint32_t, RtpState>& suspended_ssrcs,
>-      const std::map<uint32_t, RtpPayloadState>& suspended_payload_states,
>-      VideoEncoderConfig::ContentType content_type)
>-      : send_stream_(send_stream),
>-        done_event_(done_event),
>-        stats_proxy_(stats_proxy),
>-        video_stream_encoder_(video_stream_encoder),
>-        call_stats_(call_stats),
>-        transport_(transport),
>-        bitrate_allocator_(bitrate_allocator),
>-        send_delay_stats_(send_delay_stats),
>-        event_log_(event_log),
>-        config_(config),
>-        initial_encoder_max_bitrate_(initial_encoder_max_bitrate),
>-        initial_encoder_bitrate_priority_(initial_encoder_bitrate_priority),
>-        suspended_ssrcs_(suspended_ssrcs),
>-        suspended_payload_states_(suspended_payload_states),
>-        content_type_(content_type) {}
>-
>-  ~ConstructionTask() override { done_event_->Set(); }
>-
>- private:
>-  bool Run() override {
>-    send_stream_->reset(new VideoSendStreamImpl(
>-        stats_proxy_, rtc::TaskQueue::Current(), call_stats_, transport_,
>-        bitrate_allocator_, send_delay_stats_, video_stream_encoder_,
>-        event_log_, config_, initial_encoder_max_bitrate_,
>-        initial_encoder_bitrate_priority_, std::move(suspended_ssrcs_),
>-        std::move(suspended_payload_states_), content_type_));
>-    return true;
>-  }
>-
>-  std::unique_ptr<VideoSendStreamImpl>* const send_stream_;
>-  rtc::Event* const done_event_;
>-  SendStatisticsProxy* const stats_proxy_;
>-  VideoStreamEncoder* const video_stream_encoder_;
>-  CallStats* const call_stats_;
>-  RtpTransportControllerSendInterface* const transport_;
>-  BitrateAllocator* const bitrate_allocator_;
>-  SendDelayStats* const send_delay_stats_;
>-  RtcEventLog* const event_log_;
>-  const VideoSendStream::Config* config_;
>-  int initial_encoder_max_bitrate_;
>-  double initial_encoder_bitrate_priority_;
>-  std::map<uint32_t, RtpState> suspended_ssrcs_;
>-  std::map<uint32_t, RtpPayloadState> suspended_payload_states_;
>-  const VideoEncoderConfig::ContentType content_type_;
>-};
>-
>-class VideoSendStream::DestructAndGetRtpStateTask : public rtc::QueuedTask {
>- public:
>-  DestructAndGetRtpStateTask(
>-      VideoSendStream::RtpStateMap* state_map,
>-      VideoSendStream::RtpPayloadStateMap* payload_state_map,
>-      std::unique_ptr<VideoSendStreamImpl> send_stream,
>-      rtc::Event* done_event)
>-      : state_map_(state_map),
>-        payload_state_map_(payload_state_map),
>-        send_stream_(std::move(send_stream)),
>-        done_event_(done_event) {}
>-
>-  ~DestructAndGetRtpStateTask() override { RTC_CHECK(!send_stream_); }
>-
>- private:
>-  bool Run() override {
>-    send_stream_->Stop();
>-    *state_map_ = send_stream_->GetRtpStates();
>-    *payload_state_map_ = send_stream_->GetRtpPayloadStates();
>-    send_stream_.reset();
>-    done_event_->Set();
>-    return true;
>-  }
>-
>-  VideoSendStream::RtpStateMap* state_map_;
>-  VideoSendStream::RtpPayloadStateMap* payload_state_map_;
>-  std::unique_ptr<VideoSendStreamImpl> send_stream_;
>-  rtc::Event* done_event_;
>-};
>-
>-// CheckEncoderActivityTask is used for tracking when the encoder last produced
>-// and encoded video frame. If the encoder has not produced anything the last
>-// kEncoderTimeOutMs we also want to stop sending padding.
>-class VideoSendStreamImpl::CheckEncoderActivityTask : public rtc::QueuedTask {
>- public:
>-  static const int kEncoderTimeOutMs = 2000;
>-  explicit CheckEncoderActivityTask(
>-      const rtc::WeakPtr<VideoSendStreamImpl>& send_stream)
>-      : activity_(0), send_stream_(std::move(send_stream)), timed_out_(false) {}
>-
>-  void Stop() {
>-    RTC_CHECK(task_checker_.CalledSequentially());
>-    send_stream_.reset();
>-  }
>-
>-  void UpdateEncoderActivity() {
>-    // UpdateEncoderActivity is called from VideoSendStreamImpl::Encoded on
>-    // whatever thread the real encoder implementation run on. In the case of
>-    // hardware encoders, there might be several encoders
>-    // running in parallel on different threads.
>-    rtc::AtomicOps::ReleaseStore(&activity_, 1);
>-  }
>-
>- private:
>-  bool Run() override {
>-    RTC_CHECK(task_checker_.CalledSequentially());
>-    if (!send_stream_)
>-      return true;
>-    if (!rtc::AtomicOps::AcquireLoad(&activity_)) {
>-      if (!timed_out_) {
>-        send_stream_->SignalEncoderTimedOut();
>-      }
>-      timed_out_ = true;
>-    } else if (timed_out_) {
>-      send_stream_->SignalEncoderActive();
>-      timed_out_ = false;
>-    }
>-    rtc::AtomicOps::ReleaseStore(&activity_, 0);
>-
>-    rtc::TaskQueue::Current()->PostDelayedTask(
>-        std::unique_ptr<rtc::QueuedTask>(this), kEncoderTimeOutMs);
>-    // Return false to prevent this task from being deleted. Ownership has been
>-    // transferred to the task queue when PostDelayedTask was called.
>-    return false;
>-  }
>-  volatile int activity_;
>-
>-  rtc::SequencedTaskChecker task_checker_;
>-  rtc::WeakPtr<VideoSendStreamImpl> send_stream_;
>-  bool timed_out_;
>-};
>-
>-class VideoSendStreamImpl::EncoderReconfiguredTask : public rtc::QueuedTask {
>- public:
>-  EncoderReconfiguredTask(const rtc::WeakPtr<VideoSendStreamImpl>& send_stream,
>-                          std::vector<VideoStream> streams,
>-                          int min_transmit_bitrate_bps)
>-      : send_stream_(std::move(send_stream)),
>-        streams_(std::move(streams)),
>-        min_transmit_bitrate_bps_(min_transmit_bitrate_bps) {}
>-
>- private:
>-  bool Run() override {
>-    if (send_stream_)
>-      send_stream_->OnEncoderConfigurationChanged(std::move(streams_),
>-                                                  min_transmit_bitrate_bps_);
>-    return true;
>-  }
>-
>-  rtc::WeakPtr<VideoSendStreamImpl> send_stream_;
>-  std::vector<VideoStream> streams_;
>-  int min_transmit_bitrate_bps_;
>-};
>-
> VideoSendStream::VideoSendStream(
>     int num_cpu_cores,
>     ProcessThread* module_process_thread,
>@@ -560,7 +67,8 @@ VideoSendStream::VideoSendStream(
>     VideoSendStream::Config config,
>     VideoEncoderConfig encoder_config,
>     const std::map<uint32_t, RtpState>& suspended_ssrcs,
>-    const std::map<uint32_t, RtpPayloadState>& suspended_payload_states)
>+    const std::map<uint32_t, RtpPayloadState>& suspended_payload_states,
>+    std::unique_ptr<FecController> fec_controller)
>     : worker_queue_(worker_queue),
>       thread_sync_event_(false /* manual_reset */, false),
>       stats_proxy_(Clock::GetRealTimeClock(),
>@@ -568,17 +76,27 @@ VideoSendStream::VideoSendStream(
>                    encoder_config.content_type),
>       config_(std::move(config)),
>       content_type_(encoder_config.content_type) {
>-  video_stream_encoder_.reset(
>-      new VideoStreamEncoder(num_cpu_cores, &stats_proxy_,
>-                             config_.encoder_settings,
>-                             config_.pre_encode_callback,
>-                             std::unique_ptr<OveruseFrameDetector>()));
>-  worker_queue_->PostTask(std::unique_ptr<rtc::QueuedTask>(new ConstructionTask(
>-      &send_stream_, &thread_sync_event_, &stats_proxy_,
>-      video_stream_encoder_.get(), module_process_thread, call_stats, transport,
>-      bitrate_allocator, send_delay_stats, event_log, &config_,
>-      encoder_config.max_bitrate_bps, encoder_config.bitrate_priority,
>-      suspended_ssrcs, suspended_payload_states, encoder_config.content_type)));
>+  RTC_DCHECK(config_.encoder_settings.encoder_factory);
>+
>+  video_stream_encoder_ = CreateVideoStreamEncoder(num_cpu_cores, &stats_proxy_,
>+                                                   config_.encoder_settings,
>+                                                   config_.pre_encode_callback);
>+  // TODO(srte): Initialization should not be done posted on a task queue.
>+  // Note that the posted task must not outlive this scope since the closure
>+  // references local variables.
>+  worker_queue_->PostTask(rtc::NewClosure(
>+      [this, call_stats, transport, bitrate_allocator, send_delay_stats,
>+       event_log, &suspended_ssrcs, &encoder_config, &suspended_payload_states,
>+       &fec_controller]() {
>+        send_stream_.reset(new VideoSendStreamImpl(
>+            &stats_proxy_, worker_queue_, call_stats, transport,
>+            bitrate_allocator, send_delay_stats, video_stream_encoder_.get(),
>+            event_log, &config_, encoder_config.max_bitrate_bps,
>+            encoder_config.bitrate_priority, suspended_ssrcs,
>+            suspended_payload_states, encoder_config.content_type,
>+            std::move(fec_controller)));
>+      },
>+      [this]() { thread_sync_event_.Set(); }));
> 
>   // Wait for ConstructionTask to complete so that |send_stream_| can be used.
>   // |module_process_thread| must be registered and deregistered on the thread
>@@ -588,7 +106,7 @@ VideoSendStream::VideoSendStream(
>   // TODO(sprang): Enable this also for regular video calls if it works well.
>   if (encoder_config.content_type == VideoEncoderConfig::ContentType::kScreen) {
>     // Only signal target bitrate for screenshare streams, for now.
>-    video_stream_encoder_->SetBitrateObserver(send_stream_.get());
>+    video_stream_encoder_->SetBitrateAllocationObserver(send_stream_.get());
>   }
> 
>   ReconfigureVideoEncoder(std::move(encoder_config));
>@@ -599,6 +117,19 @@ VideoSendStream::~VideoSendStream() {
>   RTC_DCHECK(!send_stream_);
> }
> 
>+void VideoSendStream::UpdateActiveSimulcastLayers(
>+    const std::vector<bool> active_layers) {
>+  RTC_DCHECK_RUN_ON(&thread_checker_);
>+  RTC_LOG(LS_INFO) << "VideoSendStream::UpdateActiveSimulcastLayers";
>+  VideoSendStreamImpl* send_stream = send_stream_.get();
>+  worker_queue_->PostTask([this, send_stream, active_layers] {
>+    send_stream->UpdateActiveSimulcastLayers(active_layers);
>+    thread_sync_event_.Set();
>+  });
>+
>+  thread_sync_event_.Wait(rtc::Event::kForever);
>+}
>+
> void VideoSendStream::Start() {
>   RTC_DCHECK_RUN_ON(&thread_checker_);
>   RTC_LOG(LS_INFO) << "VideoSendStream::Start";
>@@ -633,9 +164,9 @@ void VideoSendStream::ReconfigureVideoEncoder(VideoEncoderConfig config) {
>   // ReconfigureVideoEncoder from the network thread.
>   // RTC_DCHECK_RUN_ON(&thread_checker_);
>   RTC_DCHECK(content_type_ == config.content_type);
>-  video_stream_encoder_->ConfigureEncoder(std::move(config),
>-                                          config_.rtp.max_packet_size,
>-                                          config_.rtp.nack.rtp_history_ms > 0);
>+  video_stream_encoder_->ConfigureEncoder(
>+      std::move(config),
>+      config_.rtp.max_packet_size - CalculateMaxHeaderSize(config_.rtp));
> }
> 
> VideoSendStream::Stats VideoSendStream::GetStats() {
>@@ -645,11 +176,8 @@ VideoSendStream::Stats VideoSendStream::GetStats() {
>   return stats_proxy_.GetStats();
> }
> 
>-void VideoSendStream::SignalNetworkState(NetworkState state) {
>-  RTC_DCHECK_RUN_ON(&thread_checker_);
>-  VideoSendStreamImpl* send_stream = send_stream_.get();
>-  worker_queue_->PostTask(
>-      [send_stream, state] { send_stream->SignalNetworkState(state); });
>+absl::optional<float> VideoSendStream::GetPacingFactorOverride() const {
>+  return send_stream_->configured_pacing_factor_;
> }
> 
> void VideoSendStream::StopPermanentlyAndGetRtpStates(
>@@ -658,10 +186,13 @@ void VideoSendStream::StopPermanentlyAndGetRtpStates(
>   RTC_DCHECK_RUN_ON(&thread_checker_);
>   video_stream_encoder_->Stop();
>   send_stream_->DeRegisterProcessThread();
>-  worker_queue_->PostTask(
>-      std::unique_ptr<rtc::QueuedTask>(new DestructAndGetRtpStateTask(
>-          rtp_state_map, payload_state_map, std::move(send_stream_),
>-          &thread_sync_event_)));
>+  worker_queue_->PostTask([this, rtp_state_map, payload_state_map]() {
>+    send_stream_->Stop();
>+    *rtp_state_map = send_stream_->GetRtpStates();
>+    *payload_state_map = send_stream_->GetRtpPayloadStates();
>+    send_stream_.reset();
>+    thread_sync_event_.Set();
>+  });
>   thread_sync_event_.Wait(rtc::Event::kForever);
> }
> 
>@@ -685,649 +216,5 @@ void VideoSendStream::EnableEncodedFrameRecording(
>   send_stream_->EnableEncodedFrameRecording(files, byte_limit);
> }
> 
>-VideoSendStreamImpl::VideoSendStreamImpl(
>-    SendStatisticsProxy* stats_proxy,
>-    rtc::TaskQueue* worker_queue,
>-    CallStats* call_stats,
>-    RtpTransportControllerSendInterface* transport,
>-    BitrateAllocator* bitrate_allocator,
>-    SendDelayStats* send_delay_stats,
>-    VideoStreamEncoder* video_stream_encoder,
>-    RtcEventLog* event_log,
>-    const VideoSendStream::Config* config,
>-    int initial_encoder_max_bitrate,
>-    double initial_encoder_bitrate_priority,
>-    std::map<uint32_t, RtpState> suspended_ssrcs,
>-    std::map<uint32_t, RtpPayloadState> suspended_payload_states,
>-    VideoEncoderConfig::ContentType content_type)
>-    : send_side_bwe_with_overhead_(
>-          webrtc::field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")),
>-      stats_proxy_(stats_proxy),
>-      config_(config),
>-      suspended_ssrcs_(std::move(suspended_ssrcs)),
>-      module_process_thread_(nullptr),
>-      worker_queue_(worker_queue),
>-      check_encoder_activity_task_(nullptr),
>-      call_stats_(call_stats),
>-      transport_(transport),
>-      bitrate_allocator_(bitrate_allocator),
>-      flexfec_sender_(MaybeCreateFlexfecSender(*config_, suspended_ssrcs_)),
>-      max_padding_bitrate_(0),
>-      encoder_min_bitrate_bps_(0),
>-      encoder_max_bitrate_bps_(initial_encoder_max_bitrate),
>-      encoder_target_rate_bps_(0),
>-      encoder_bitrate_priority_(initial_encoder_bitrate_priority),
>-      video_stream_encoder_(video_stream_encoder),
>-      encoder_feedback_(Clock::GetRealTimeClock(),
>-                        config_->rtp.ssrcs,
>-                        video_stream_encoder),
>-      protection_bitrate_calculator_(Clock::GetRealTimeClock(), this),
>-      bandwidth_observer_(transport->send_side_cc()->GetBandwidthObserver()),
>-      rtp_rtcp_modules_(CreateRtpRtcpModules(
>-          config_->send_transport,
>-          &encoder_feedback_,
>-          bandwidth_observer_,
>-          transport,
>-          call_stats_->rtcp_rtt_stats(),
>-          flexfec_sender_.get(),
>-          stats_proxy_,
>-          send_delay_stats,
>-          event_log,
>-          transport->send_side_cc()->GetRetransmissionRateLimiter(),
>-          this,
>-          config_->rtp.ssrcs.size(),
>-          transport->keepalive_config())),
>-      payload_router_(rtp_rtcp_modules_,
>-                      config_->rtp.ssrcs,
>-                      config_->encoder_settings.payload_type,
>-                      suspended_payload_states),
>-      weak_ptr_factory_(this),
>-      overhead_bytes_per_packet_(0),
>-      transport_overhead_bytes_per_packet_(0) {
>-  RTC_DCHECK_RUN_ON(worker_queue_);
>-  RTC_LOG(LS_INFO) << "VideoSendStreamInternal: " << config_->ToString();
>-  weak_ptr_ = weak_ptr_factory_.GetWeakPtr();
>-  module_process_thread_checker_.DetachFromThread();
>-
>-  RTC_DCHECK(!config_->rtp.ssrcs.empty());
>-  RTC_DCHECK(call_stats_);
>-  RTC_DCHECK(transport_);
>-  RTC_DCHECK(transport_->send_side_cc());
>-  RTC_DCHECK_GT(encoder_max_bitrate_bps_, 0);
>-
>-  RTC_CHECK(AlrExperimentSettings::MaxOneFieldTrialEnabled());
>-  // If send-side BWE is enabled, check if we should apply updated probing and
>-  // pacing settings.
>-  if (TransportSeqNumExtensionConfigured(*config_)) {
>-    rtc::Optional<AlrExperimentSettings> alr_settings;
>-    if (content_type == VideoEncoderConfig::ContentType::kScreen) {
>-      alr_settings = AlrExperimentSettings::CreateFromFieldTrial(
>-          AlrExperimentSettings::kScreenshareProbingBweExperimentName);
>-    } else {
>-      alr_settings = AlrExperimentSettings::CreateFromFieldTrial(
>-          AlrExperimentSettings::kStrictPacingAndProbingExperimentName);
>-    }
>-    if (alr_settings) {
>-      transport->send_side_cc()->EnablePeriodicAlrProbing(true);
>-      transport->pacer()->SetPacingFactor(alr_settings->pacing_factor);
>-      transport->pacer()->SetQueueTimeLimit(alr_settings->max_paced_queue_time);
>-    }
>-  }
>-
>-  if (config_->periodic_alr_bandwidth_probing) {
>-    transport->send_side_cc()->EnablePeriodicAlrProbing(true);
>-  }
>-
>-  // RTP/RTCP initialization.
>-
>-  // We add the highest spatial layer first to ensure it'll be prioritized
>-  // when sending padding, with the hope that the packet rate will be smaller,
>-  // and that it's more important to protect than the lower layers.
>-  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
>-    constexpr bool remb_candidate = true;
>-    transport->packet_router()->AddSendRtpModule(rtp_rtcp, remb_candidate);
>-  }
>-
>-  for (size_t i = 0; i < config_->rtp.extensions.size(); ++i) {
>-    const std::string& extension = config_->rtp.extensions[i].uri;
>-    int id = config_->rtp.extensions[i].id;
>-    // One-byte-extension local identifiers are in the range 1-14 inclusive.
>-    RTC_DCHECK_GE(id, 1);
>-    RTC_DCHECK_LE(id, 14);
>-    RTC_DCHECK(RtpExtension::IsSupportedForVideo(extension));
>-    for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
>-      RTC_CHECK_EQ(0, rtp_rtcp->RegisterSendRtpHeaderExtension(
>-                          StringToRtpExtensionType(extension), id));
>-    }
>-  }
>-
>-  ConfigureProtection();
>-  ConfigureSsrcs();
>-
>-  // TODO(pbos): Should we set CNAME on all RTP modules?
>-  rtp_rtcp_modules_.front()->SetCNAME(config_->rtp.c_name.c_str());
>-
>-  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
>-    rtp_rtcp->RegisterRtcpStatisticsCallback(stats_proxy_);
>-    rtp_rtcp->RegisterSendChannelRtpStatisticsCallback(stats_proxy_);
>-    rtp_rtcp->SetMaxRtpPacketSize(config_->rtp.max_packet_size);
>-    rtp_rtcp->RegisterVideoSendPayload(
>-        config_->encoder_settings.payload_type,
>-        config_->encoder_settings.payload_name.c_str());
>-  }
>-
>-  RTC_DCHECK(config_->encoder_settings.encoder);
>-  RTC_DCHECK_GE(config_->encoder_settings.payload_type, 0);
>-  RTC_DCHECK_LE(config_->encoder_settings.payload_type, 127);
>-
>-  video_stream_encoder_->SetStartBitrate(
>-      bitrate_allocator_->GetStartBitrate(this));
>-
>-  // Only request rotation at the source when we positively know that the remote
>-  // side doesn't support the rotation extension. This allows us to prepare the
>-  // encoder in the expectation that rotation is supported - which is the common
>-  // case.
>-  bool rotation_applied =
>-      std::find_if(config_->rtp.extensions.begin(),
>-                   config_->rtp.extensions.end(),
>-                   [](const RtpExtension& extension) {
>-                     return extension.uri == RtpExtension::kVideoRotationUri;
>-                   }) == config_->rtp.extensions.end();
>-
>-  video_stream_encoder_->SetSink(this, rotation_applied);
>-}
>-
>-void VideoSendStreamImpl::RegisterProcessThread(
>-    ProcessThread* module_process_thread) {
>-  RTC_DCHECK_RUN_ON(&module_process_thread_checker_);
>-  RTC_DCHECK(!module_process_thread_);
>-  module_process_thread_ = module_process_thread;
>-
>-  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
>-    module_process_thread_->RegisterModule(rtp_rtcp, RTC_FROM_HERE);
>-}
>-
>-void VideoSendStreamImpl::DeRegisterProcessThread() {
>-  RTC_DCHECK_RUN_ON(&module_process_thread_checker_);
>-  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
>-    module_process_thread_->DeRegisterModule(rtp_rtcp);
>-}
>-
>-VideoSendStreamImpl::~VideoSendStreamImpl() {
>-  RTC_DCHECK_RUN_ON(worker_queue_);
>-  RTC_DCHECK(!payload_router_.IsActive())
>-      << "VideoSendStreamImpl::Stop not called";
>-  RTC_LOG(LS_INFO) << "~VideoSendStreamInternal: " << config_->ToString();
>-
>-  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
>-    transport_->packet_router()->RemoveSendRtpModule(rtp_rtcp);
>-    delete rtp_rtcp;
>-  }
>-}
>-
>-bool VideoSendStreamImpl::DeliverRtcp(const uint8_t* packet, size_t length) {
>-  // Runs on a network thread.
>-  RTC_DCHECK(!worker_queue_->IsCurrent());
>-  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
>-    rtp_rtcp->IncomingRtcpPacket(packet, length);
>-  return true;
>-}
>-
>-void VideoSendStreamImpl::Start() {
>-  RTC_DCHECK_RUN_ON(worker_queue_);
>-  RTC_LOG(LS_INFO) << "VideoSendStream::Start";
>-  if (payload_router_.IsActive())
>-    return;
>-  TRACE_EVENT_INSTANT0("webrtc", "VideoSendStream::Start");
>-  payload_router_.SetActive(true);
>-
>-  bitrate_allocator_->AddObserver(
>-      this, encoder_min_bitrate_bps_, encoder_max_bitrate_bps_,
>-      max_padding_bitrate_, !config_->suspend_below_min_bitrate,
>-      config_->track_id, encoder_bitrate_priority_);
>-
>-  // Start monitoring encoder activity.
>-  {
>-    rtc::CritScope lock(&encoder_activity_crit_sect_);
>-    RTC_DCHECK(!check_encoder_activity_task_);
>-    check_encoder_activity_task_ = new CheckEncoderActivityTask(weak_ptr_);
>-    worker_queue_->PostDelayedTask(
>-        std::unique_ptr<rtc::QueuedTask>(check_encoder_activity_task_),
>-        CheckEncoderActivityTask::kEncoderTimeOutMs);
>-  }
>-
>-  video_stream_encoder_->SendKeyFrame();
>-}
>-
>-void VideoSendStreamImpl::Stop() {
>-  RTC_DCHECK_RUN_ON(worker_queue_);
>-  RTC_LOG(LS_INFO) << "VideoSendStream::Stop";
>-  if (!payload_router_.IsActive())
>-    return;
>-  TRACE_EVENT_INSTANT0("webrtc", "VideoSendStream::Stop");
>-  payload_router_.SetActive(false);
>-  bitrate_allocator_->RemoveObserver(this);
>-  {
>-    rtc::CritScope lock(&encoder_activity_crit_sect_);
>-    check_encoder_activity_task_->Stop();
>-    check_encoder_activity_task_ = nullptr;
>-  }
>-  video_stream_encoder_->OnBitrateUpdated(0, 0, 0);
>-  stats_proxy_->OnSetEncoderTargetRate(0);
>-}
>-
>-void VideoSendStreamImpl::SignalEncoderTimedOut() {
>-  RTC_DCHECK_RUN_ON(worker_queue_);
>-  // If the encoder has not produced anything the last kEncoderTimeOutMs and it
>-  // is supposed to, deregister as BitrateAllocatorObserver. This can happen
>-  // if a camera stops producing frames.
>-  if (encoder_target_rate_bps_ > 0) {
>-    RTC_LOG(LS_INFO) << "SignalEncoderTimedOut, Encoder timed out.";
>-    bitrate_allocator_->RemoveObserver(this);
>-  }
>-}
>-
>-void VideoSendStreamImpl::OnBitrateAllocationUpdated(
>-    const BitrateAllocation& allocation) {
>-  payload_router_.OnBitrateAllocationUpdated(allocation);
>-}
>-
>-void VideoSendStreamImpl::SignalEncoderActive() {
>-  RTC_DCHECK_RUN_ON(worker_queue_);
>-  RTC_LOG(LS_INFO) << "SignalEncoderActive, Encoder is active.";
>-  bitrate_allocator_->AddObserver(
>-      this, encoder_min_bitrate_bps_, encoder_max_bitrate_bps_,
>-      max_padding_bitrate_, !config_->suspend_below_min_bitrate,
>-      config_->track_id, encoder_bitrate_priority_);
>-}
>-
>-void VideoSendStreamImpl::OnEncoderConfigurationChanged(
>-    std::vector<VideoStream> streams,
>-    int min_transmit_bitrate_bps) {
>-  if (!worker_queue_->IsCurrent()) {
>-    worker_queue_->PostTask(
>-        std::unique_ptr<rtc::QueuedTask>(new EncoderReconfiguredTask(
>-            weak_ptr_, std::move(streams), min_transmit_bitrate_bps)));
>-    return;
>-  }
>-  RTC_DCHECK_GE(config_->rtp.ssrcs.size(), streams.size());
>-  TRACE_EVENT0("webrtc", "VideoSendStream::OnEncoderConfigurationChanged");
>-  RTC_DCHECK_GE(config_->rtp.ssrcs.size(), streams.size());
>-  RTC_DCHECK_RUN_ON(worker_queue_);
>-
>-  encoder_min_bitrate_bps_ =
>-      std::max(streams[0].min_bitrate_bps, GetEncoderMinBitrateBps());
>-  encoder_max_bitrate_bps_ = 0;
>-  double stream_bitrate_priority_sum = 0;
>-  for (const auto& stream : streams) {
>-    encoder_max_bitrate_bps_ += stream.max_bitrate_bps;
>-    if (stream.bitrate_priority) {
>-      RTC_DCHECK_GT(*stream.bitrate_priority, 0);
>-      stream_bitrate_priority_sum += *stream.bitrate_priority;
>-    }
>-  }
>-  RTC_DCHECK_GT(stream_bitrate_priority_sum, 0);
>-  encoder_bitrate_priority_ = stream_bitrate_priority_sum;
>-  max_padding_bitrate_ = CalculateMaxPadBitrateBps(
>-      streams, min_transmit_bitrate_bps, config_->suspend_below_min_bitrate);
>-
>-  // Clear stats for disabled layers.
>-  for (size_t i = streams.size(); i < config_->rtp.ssrcs.size(); ++i) {
>-    stats_proxy_->OnInactiveSsrc(config_->rtp.ssrcs[i]);
>-  }
>-
>-  size_t number_of_temporal_layers =
>-      streams.back().temporal_layer_thresholds_bps.size() + 1;
>-  protection_bitrate_calculator_.SetEncodingData(
>-      streams[0].width, streams[0].height, number_of_temporal_layers,
>-      config_->rtp.max_packet_size);
>-
>-  if (payload_router_.IsActive()) {
>-    // The send stream is started already. Update the allocator with new bitrate
>-    // limits.
>-    bitrate_allocator_->AddObserver(
>-        this, encoder_min_bitrate_bps_, encoder_max_bitrate_bps_,
>-        max_padding_bitrate_, !config_->suspend_below_min_bitrate,
>-        config_->track_id, encoder_bitrate_priority_);
>-  }
>-}
>-
>-EncodedImageCallback::Result VideoSendStreamImpl::OnEncodedImage(
>-    const EncodedImage& encoded_image,
>-    const CodecSpecificInfo* codec_specific_info,
>-    const RTPFragmentationHeader* fragmentation) {
>-  // Encoded is called on whatever thread the real encoder implementation run
>-  // on. In the case of hardware encoders, there might be several encoders
>-  // running in parallel on different threads.
>-  size_t simulcast_idx = 0;
>-  if (codec_specific_info->codecType == kVideoCodecVP8) {
>-    simulcast_idx = codec_specific_info->codecSpecific.VP8.simulcastIdx;
>-  }
>-  if (config_->post_encode_callback) {
>-    config_->post_encode_callback->EncodedFrameCallback(EncodedFrame(
>-        encoded_image._buffer, encoded_image._length, encoded_image._frameType,
>-        simulcast_idx, encoded_image._timeStamp));
>-  }
>-  {
>-    rtc::CritScope lock(&encoder_activity_crit_sect_);
>-    if (check_encoder_activity_task_)
>-      check_encoder_activity_task_->UpdateEncoderActivity();
>-  }
>-
>-  protection_bitrate_calculator_.UpdateWithEncodedData(encoded_image);
>-  EncodedImageCallback::Result result = payload_router_.OnEncodedImage(
>-      encoded_image, codec_specific_info, fragmentation);
>-
>-  RTC_DCHECK(codec_specific_info);
>-
>-  int layer = codec_specific_info->codecType == kVideoCodecVP8
>-                  ? codec_specific_info->codecSpecific.VP8.simulcastIdx
>-                  : 0;
>-  {
>-    rtc::CritScope lock(&ivf_writers_crit_);
>-    if (file_writers_[layer].get()) {
>-      bool ok = file_writers_[layer]->WriteFrame(
>-          encoded_image, codec_specific_info->codecType);
>-      RTC_DCHECK(ok);
>-    }
>-  }
>-
>-  return result;
>-}
>-
>-void VideoSendStreamImpl::ConfigureProtection() {
>-  RTC_DCHECK_RUN_ON(worker_queue_);
>-
>-  // Consistency of FlexFEC parameters is checked in MaybeCreateFlexfecSender.
>-  const bool flexfec_enabled = (flexfec_sender_ != nullptr);
>-
>-  // Consistency of NACK and RED+ULPFEC parameters is checked in this function.
>-  const bool nack_enabled = config_->rtp.nack.rtp_history_ms > 0;
>-  int red_payload_type = config_->rtp.ulpfec.red_payload_type;
>-  int ulpfec_payload_type = config_->rtp.ulpfec.ulpfec_payload_type;
>-
>-  // Shorthands.
>-  auto IsRedEnabled = [&]() { return red_payload_type >= 0; };
>-  auto DisableRed = [&]() { red_payload_type = -1; };
>-  auto IsUlpfecEnabled = [&]() { return ulpfec_payload_type >= 0; };
>-  auto DisableUlpfec = [&]() { ulpfec_payload_type = -1; };
>-
>-  if (webrtc::field_trial::IsEnabled("WebRTC-DisableUlpFecExperiment")) {
>-    RTC_LOG(LS_INFO) << "Experiment to disable sending ULPFEC is enabled.";
>-    DisableUlpfec();
>-  }
>-
>-  // If enabled, FlexFEC takes priority over RED+ULPFEC.
>-  if (flexfec_enabled) {
>-    // We can safely disable RED here, because if the remote supports FlexFEC,
>-    // we know that it has a receiver without the RED/RTX workaround.
>-    // See http://crbug.com/webrtc/6650 for more information.
>-    if (IsRedEnabled()) {
>-      RTC_LOG(LS_INFO) << "Both FlexFEC and RED are configured. Disabling RED.";
>-      DisableRed();
>-    }
>-    if (IsUlpfecEnabled()) {
>-      RTC_LOG(LS_INFO)
>-          << "Both FlexFEC and ULPFEC are configured. Disabling ULPFEC.";
>-      DisableUlpfec();
>-    }
>-  }
>-
>-  // Payload types without picture ID cannot determine that a stream is complete
>-  // without retransmitting FEC, so using ULPFEC + NACK for H.264 (for instance)
>-  // is a waste of bandwidth since FEC packets still have to be transmitted.
>-  // Note that this is not the case with FlexFEC.
>-  if (nack_enabled && IsUlpfecEnabled() &&
>-      !PayloadTypeSupportsSkippingFecPackets(
>-          config_->encoder_settings.payload_name)) {
>-    RTC_LOG(LS_WARNING)
>-        << "Transmitting payload type without picture ID using "
>-           "NACK+ULPFEC is a waste of bandwidth since ULPFEC packets "
>-           "also have to be retransmitted. Disabling ULPFEC.";
>-    DisableUlpfec();
>-  }
>-
>-  // Verify payload types.
>-  //
>-  // Due to how old receivers work, we need to always send RED if it has been
>-  // negotiated. This is a remnant of an old RED/RTX workaround, see
>-  // https://codereview.webrtc.org/2469093003.
>-  // TODO(brandtr): This change went into M56, so we can remove it in ~M59.
>-  // At that time, we can disable RED whenever ULPFEC is disabled, as there is
>-  // no point in using RED without ULPFEC.
>-  if (IsRedEnabled()) {
>-    RTC_DCHECK_GE(red_payload_type, 0);
>-    RTC_DCHECK_LE(red_payload_type, 127);
>-  }
>-  if (IsUlpfecEnabled()) {
>-    RTC_DCHECK_GE(ulpfec_payload_type, 0);
>-    RTC_DCHECK_LE(ulpfec_payload_type, 127);
>-    if (!IsRedEnabled()) {
>-      RTC_LOG(LS_WARNING)
>-          << "ULPFEC is enabled but RED is disabled. Disabling ULPFEC.";
>-      DisableUlpfec();
>-    }
>-  }
>-
>-  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
>-    // Set NACK.
>-    rtp_rtcp->SetStorePacketsStatus(
>-        true,
>-        kMinSendSidePacketHistorySize);
>-    // Set RED/ULPFEC information.
>-    rtp_rtcp->SetUlpfecConfig(red_payload_type, ulpfec_payload_type);
>-  }
>-
>-  // Currently, both ULPFEC and FlexFEC use the same FEC rate calculation logic,
>-  // so enable that logic if either of those FEC schemes are enabled.
>-  protection_bitrate_calculator_.SetProtectionMethod(
>-      flexfec_enabled || IsUlpfecEnabled(), nack_enabled);
>-}
>-
>-void VideoSendStreamImpl::ConfigureSsrcs() {
>-  RTC_DCHECK_RUN_ON(worker_queue_);
>-  // Configure regular SSRCs.
>-  for (size_t i = 0; i < config_->rtp.ssrcs.size(); ++i) {
>-    uint32_t ssrc = config_->rtp.ssrcs[i];
>-    RtpRtcp* const rtp_rtcp = rtp_rtcp_modules_[i];
>-    rtp_rtcp->SetSSRC(ssrc);
>-
>-    // Restore RTP state if previous existed.
>-    VideoSendStream::RtpStateMap::iterator it = suspended_ssrcs_.find(ssrc);
>-    if (it != suspended_ssrcs_.end())
>-      rtp_rtcp->SetRtpState(it->second);
>-  }
>-
>-  // Set up RTX if available.
>-  if (config_->rtp.rtx.ssrcs.empty())
>-    return;
>-
>-  // Configure RTX SSRCs.
>-  RTC_DCHECK_EQ(config_->rtp.rtx.ssrcs.size(), config_->rtp.ssrcs.size());
>-  for (size_t i = 0; i < config_->rtp.rtx.ssrcs.size(); ++i) {
>-    uint32_t ssrc = config_->rtp.rtx.ssrcs[i];
>-    RtpRtcp* const rtp_rtcp = rtp_rtcp_modules_[i];
>-    rtp_rtcp->SetRtxSsrc(ssrc);
>-    VideoSendStream::RtpStateMap::iterator it = suspended_ssrcs_.find(ssrc);
>-    if (it != suspended_ssrcs_.end())
>-      rtp_rtcp->SetRtxState(it->second);
>-  }
>-
>-  // Configure RTX payload types.
>-  RTC_DCHECK_GE(config_->rtp.rtx.payload_type, 0);
>-  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
>-    rtp_rtcp->SetRtxSendPayloadType(config_->rtp.rtx.payload_type,
>-                                    config_->encoder_settings.payload_type);
>-    rtp_rtcp->SetRtxSendStatus(kRtxRetransmitted | kRtxRedundantPayloads);
>-  }
>-  if (config_->rtp.ulpfec.red_payload_type != -1 &&
>-      config_->rtp.ulpfec.red_rtx_payload_type != -1) {
>-    for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
>-      rtp_rtcp->SetRtxSendPayloadType(config_->rtp.ulpfec.red_rtx_payload_type,
>-                                      config_->rtp.ulpfec.red_payload_type);
>-    }
>-  }
>-}
>-
>-std::map<uint32_t, RtpState> VideoSendStreamImpl::GetRtpStates() const {
>-  RTC_DCHECK_RUN_ON(worker_queue_);
>-  std::map<uint32_t, RtpState> rtp_states;
>-
>-  for (size_t i = 0; i < config_->rtp.ssrcs.size(); ++i) {
>-    uint32_t ssrc = config_->rtp.ssrcs[i];
>-    RTC_DCHECK_EQ(ssrc, rtp_rtcp_modules_[i]->SSRC());
>-    rtp_states[ssrc] = rtp_rtcp_modules_[i]->GetRtpState();
>-  }
>-
>-  for (size_t i = 0; i < config_->rtp.rtx.ssrcs.size(); ++i) {
>-    uint32_t ssrc = config_->rtp.rtx.ssrcs[i];
>-    rtp_states[ssrc] = rtp_rtcp_modules_[i]->GetRtxState();
>-  }
>-
>-  if (flexfec_sender_) {
>-    uint32_t ssrc = config_->rtp.flexfec.ssrc;
>-    rtp_states[ssrc] = flexfec_sender_->GetRtpState();
>-  }
>-
>-  return rtp_states;
>-}
>-
>-std::map<uint32_t, RtpPayloadState> VideoSendStreamImpl::GetRtpPayloadStates()
>-    const {
>-  RTC_DCHECK_RUN_ON(worker_queue_);
>-  return payload_router_.GetRtpPayloadStates();
>-}
>-
>-void VideoSendStreamImpl::SignalNetworkState(NetworkState state) {
>-  RTC_DCHECK_RUN_ON(worker_queue_);
>-  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
>-    rtp_rtcp->SetRTCPStatus(state == kNetworkUp ? config_->rtp.rtcp_mode
>-                                                : RtcpMode::kOff);
>-  }
>-}
>-
>-uint32_t VideoSendStreamImpl::OnBitrateUpdated(uint32_t bitrate_bps,
>-                                               uint8_t fraction_loss,
>-                                               int64_t rtt,
>-                                               int64_t probing_interval_ms) {
>-  RTC_DCHECK_RUN_ON(worker_queue_);
>-  RTC_DCHECK(payload_router_.IsActive())
>-      << "VideoSendStream::Start has not been called.";
>-
>-  // Substract overhead from bitrate.
>-  rtc::CritScope lock(&overhead_bytes_per_packet_crit_);
>-  uint32_t payload_bitrate_bps = bitrate_bps;
>-  if (send_side_bwe_with_overhead_) {
>-    payload_bitrate_bps -= CalculateOverheadRateBps(
>-        CalculatePacketRate(bitrate_bps,
>-                            config_->rtp.max_packet_size +
>-                                transport_overhead_bytes_per_packet_),
>-        overhead_bytes_per_packet_ + transport_overhead_bytes_per_packet_,
>-        bitrate_bps);
>-  }
>-
>-  // Get the encoder target rate. It is the estimated network rate -
>-  // protection overhead.
>-  encoder_target_rate_bps_ = protection_bitrate_calculator_.SetTargetRates(
>-      payload_bitrate_bps, stats_proxy_->GetSendFrameRate(), fraction_loss,
>-      rtt);
>-
>-  uint32_t encoder_overhead_rate_bps =
>-      send_side_bwe_with_overhead_
>-          ? CalculateOverheadRateBps(
>-                CalculatePacketRate(encoder_target_rate_bps_,
>-                                    config_->rtp.max_packet_size +
>-                                        transport_overhead_bytes_per_packet_ -
>-                                        overhead_bytes_per_packet_),
>-                overhead_bytes_per_packet_ +
>-                    transport_overhead_bytes_per_packet_,
>-                bitrate_bps - encoder_target_rate_bps_)
>-          : 0;
>-
>-  // When the field trial "WebRTC-SendSideBwe-WithOverhead" is enabled
>-  // protection_bitrate includes overhead.
>-  uint32_t protection_bitrate =
>-      bitrate_bps - (encoder_target_rate_bps_ + encoder_overhead_rate_bps);
>-
>-  encoder_target_rate_bps_ =
>-      std::min(encoder_max_bitrate_bps_, encoder_target_rate_bps_);
>-  video_stream_encoder_->OnBitrateUpdated(encoder_target_rate_bps_,
>-                                          fraction_loss, rtt);
>-  stats_proxy_->OnSetEncoderTargetRate(encoder_target_rate_bps_);
>-  return protection_bitrate;
>-}
>-
>-void VideoSendStreamImpl::EnableEncodedFrameRecording(
>-    const std::vector<rtc::PlatformFile>& files,
>-    size_t byte_limit) {
>-  {
>-    rtc::CritScope lock(&ivf_writers_crit_);
>-    for (unsigned int i = 0; i < kMaxSimulcastStreams; ++i) {
>-      if (i < files.size()) {
>-        file_writers_[i] = IvfFileWriter::Wrap(rtc::File(files[i]), byte_limit);
>-      } else {
>-        file_writers_[i].reset();
>-      }
>-    }
>-  }
>-
>-  if (!files.empty()) {
>-    // Make a keyframe appear as early as possible in the logs, to give actually
>-    // decodable output.
>-    video_stream_encoder_->SendKeyFrame();
>-  }
>-}
>-
>-int VideoSendStreamImpl::ProtectionRequest(
>-    const FecProtectionParams* delta_params,
>-    const FecProtectionParams* key_params,
>-    uint32_t* sent_video_rate_bps,
>-    uint32_t* sent_nack_rate_bps,
>-    uint32_t* sent_fec_rate_bps) {
>-  RTC_DCHECK_RUN_ON(worker_queue_);
>-  *sent_video_rate_bps = 0;
>-  *sent_nack_rate_bps = 0;
>-  *sent_fec_rate_bps = 0;
>-  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
>-    uint32_t not_used = 0;
>-    uint32_t module_video_rate = 0;
>-    uint32_t module_fec_rate = 0;
>-    uint32_t module_nack_rate = 0;
>-    rtp_rtcp->SetFecParameters(*delta_params, *key_params);
>-    rtp_rtcp->BitrateSent(&not_used, &module_video_rate, &module_fec_rate,
>-                          &module_nack_rate);
>-    *sent_video_rate_bps += module_video_rate;
>-    *sent_nack_rate_bps += module_nack_rate;
>-    *sent_fec_rate_bps += module_fec_rate;
>-  }
>-  return 0;
>-}
>-
>-void VideoSendStreamImpl::OnOverheadChanged(size_t overhead_bytes_per_packet) {
>-  rtc::CritScope lock(&overhead_bytes_per_packet_crit_);
>-  overhead_bytes_per_packet_ = overhead_bytes_per_packet;
>-}
>-
>-void VideoSendStreamImpl::SetTransportOverhead(
>-    size_t transport_overhead_bytes_per_packet) {
>-  if (transport_overhead_bytes_per_packet >= static_cast<int>(kPathMTU)) {
>-    RTC_LOG(LS_ERROR) << "Transport overhead exceeds size of ethernet frame";
>-    return;
>-  }
>-
>-  transport_overhead_bytes_per_packet_ = transport_overhead_bytes_per_packet;
>-
>-  transport_->send_side_cc()->SetTransportOverhead(
>-      transport_overhead_bytes_per_packet_);
>-
>-  size_t rtp_packet_size =
>-      std::min(config_->rtp.max_packet_size,
>-               kPathMTU - transport_overhead_bytes_per_packet_);
>-
>-  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
>-    rtp_rtcp->SetMaxRtpPacketSize(rtp_packet_size);
>-  }
>-}
>-
> }  // namespace internal
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream.h b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream.h
>index 998250cd6e4..9ca73c0abf1 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream.h
>@@ -15,25 +15,28 @@
> #include <memory>
> #include <vector>
> 
>+#include "api/fec_controller.h"
>+#include "api/video/video_stream_encoder_interface.h"
> #include "call/bitrate_allocator.h"
> #include "call/video_receive_stream.h"
> #include "call/video_send_stream.h"
> #include "common_video/libyuv/include/webrtc_libyuv.h"
>-#include "modules/video_coding/protection_bitrate_calculator.h"
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/event.h"
> #include "rtc_base/task_queue.h"
>-#include "video/encoder_rtcp_feedback.h"
> #include "video/send_delay_stats.h"
> #include "video/send_statistics_proxy.h"
>-#include "video/video_stream_encoder.h"
> 
> namespace webrtc {
>+namespace test {
>+class VideoSendStreamPeer;
>+}  // namespace test
> 
> class CallStats;
> class SendSideCongestionController;
> class IvfFileWriter;
> class ProcessThread;
>+class RateLimiter;
> class RtpRtcp;
> class RtpTransportControllerSendInterface;
> class RtcEventLog;
>@@ -48,6 +51,9 @@ class VideoSendStreamImpl;
> // |worker_queue|.
> class VideoSendStream : public webrtc::VideoSendStream {
>  public:
>+  using RtpStateMap = std::map<uint32_t, RtpState>;
>+  using RtpPayloadStateMap = std::map<uint32_t, RtpPayloadState>;
>+
>   VideoSendStream(
>       int num_cpu_cores,
>       ProcessThread* module_process_thread,
>@@ -60,14 +66,16 @@ class VideoSendStream : public webrtc::VideoSendStream {
>       VideoSendStream::Config config,
>       VideoEncoderConfig encoder_config,
>       const std::map<uint32_t, RtpState>& suspended_ssrcs,
>-      const std::map<uint32_t, RtpPayloadState>& suspended_payload_states);
>+      const std::map<uint32_t, RtpPayloadState>& suspended_payload_states,
>+      std::unique_ptr<FecController> fec_controller);
> 
>   ~VideoSendStream() override;
> 
>-  void SignalNetworkState(NetworkState state);
>   bool DeliverRtcp(const uint8_t* packet, size_t length);
> 
>   // webrtc::VideoSendStream implementation.
>+  void UpdateActiveSimulcastLayers(
>+      const std::vector<bool> active_layers) override;
>   void Start() override;
>   void Stop() override;
> 
>@@ -77,9 +85,6 @@ class VideoSendStream : public webrtc::VideoSendStream {
>   void ReconfigureVideoEncoder(VideoEncoderConfig) override;
>   Stats GetStats() override;
> 
>-  typedef std::map<uint32_t, RtpState> RtpStateMap;
>-  typedef std::map<uint32_t, RtpPayloadState> RtpPayloadStateMap;
>-
>   // Takes ownership of each file, is responsible for closing them later.
>   // Calling this method will close and finalize any current logs.
>   // Giving rtc::kInvalidPlatformFileValue in any position disables logging
>@@ -95,8 +100,11 @@ class VideoSendStream : public webrtc::VideoSendStream {
>   void SetTransportOverhead(size_t transport_overhead_per_packet);
> 
>  private:
>+  friend class test::VideoSendStreamPeer;
>+
>   class ConstructionTask;
>-  class DestructAndGetRtpStateTask;
>+
>+  absl::optional<float> GetPacingFactorOverride() const;
> 
>   rtc::ThreadChecker thread_checker_;
>   rtc::TaskQueue* const worker_queue_;
>@@ -106,7 +114,7 @@ class VideoSendStream : public webrtc::VideoSendStream {
>   const VideoSendStream::Config config_;
>   const VideoEncoderConfig::ContentType content_type_;
>   std::unique_ptr<VideoSendStreamImpl> send_stream_;
>-  std::unique_ptr<VideoStreamEncoder> video_stream_encoder_;
>+  std::unique_ptr<VideoStreamEncoderInterface> video_stream_encoder_;
> };
> 
> }  // namespace internal
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream_impl.cc
>new file mode 100644
>index 00000000000..f6607255862
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream_impl.cc
>@@ -0,0 +1,721 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#include "video/video_send_stream_impl.h"
>+
>+#include <algorithm>
>+#include <string>
>+#include <utility>
>+
>+#include "call/rtp_transport_controller_send_interface.h"
>+#include "modules/pacing/packet_router.h"
>+#include "modules/rtp_rtcp/source/rtp_sender.h"
>+#include "rtc_base/checks.h"
>+#include "rtc_base/experiments/alr_experiment.h"
>+#include "rtc_base/file.h"
>+#include "rtc_base/location.h"
>+#include "rtc_base/logging.h"
>+#include "rtc_base/numerics/safe_conversions.h"
>+#include "rtc_base/trace_event.h"
>+#include "system_wrappers/include/field_trial.h"
>+
>+namespace webrtc {
>+namespace internal {
>+namespace {
>+// Assume an average video stream has around 3 packets per frame (1 mbps / 30
>+// fps / 1400B) A sequence number set with size 5500 will be able to store
>+// packet sequence number for at least last 60 seconds.
>+static const int kSendSideSeqNumSetMaxSize = 5500;
>+
>+// We don't do MTU discovery, so assume that we have the standard ethernet MTU.
>+const size_t kPathMTU = 1500;
>+
>+bool TransportSeqNumExtensionConfigured(const VideoSendStream::Config& config) {
>+  const std::vector<RtpExtension>& extensions = config.rtp.extensions;
>+  return std::find_if(
>+             extensions.begin(), extensions.end(), [](const RtpExtension& ext) {
>+               return ext.uri == RtpExtension::kTransportSequenceNumberUri;
>+             }) != extensions.end();
>+}
>+
>+const char kForcedFallbackFieldTrial[] =
>+    "WebRTC-VP8-Forced-Fallback-Encoder-v2";
>+
>+absl::optional<int> GetFallbackMinBpsFromFieldTrial() {
>+  if (!webrtc::field_trial::IsEnabled(kForcedFallbackFieldTrial))
>+    return absl::nullopt;
>+
>+  std::string group =
>+      webrtc::field_trial::FindFullName(kForcedFallbackFieldTrial);
>+  if (group.empty())
>+    return absl::nullopt;
>+
>+  int min_pixels;
>+  int max_pixels;
>+  int min_bps;
>+  if (sscanf(group.c_str(), "Enabled-%d,%d,%d", &min_pixels, &max_pixels,
>+             &min_bps) != 3) {
>+    return absl::nullopt;
>+  }
>+
>+  if (min_bps <= 0)
>+    return absl::nullopt;
>+
>+  return min_bps;
>+}
>+
>+int GetEncoderMinBitrateBps() {
>+  const int kDefaultEncoderMinBitrateBps = 30000;
>+  return GetFallbackMinBpsFromFieldTrial().value_or(
>+      kDefaultEncoderMinBitrateBps);
>+}
>+
>+int CalculateMaxPadBitrateBps(std::vector<VideoStream> streams,
>+                              int min_transmit_bitrate_bps,
>+                              bool pad_to_min_bitrate) {
>+  int pad_up_to_bitrate_bps = 0;
>+  // Calculate max padding bitrate for a multi layer codec.
>+  if (streams.size() > 1) {
>+    // Pad to min bitrate of the highest layer.
>+    pad_up_to_bitrate_bps = streams[streams.size() - 1].min_bitrate_bps;
>+    // Add target_bitrate_bps of the lower layers.
>+    for (size_t i = 0; i < streams.size() - 1; ++i)
>+      pad_up_to_bitrate_bps += streams[i].target_bitrate_bps;
>+  } else if (pad_to_min_bitrate) {
>+    pad_up_to_bitrate_bps = streams[0].min_bitrate_bps;
>+  }
>+
>+  pad_up_to_bitrate_bps =
>+      std::max(pad_up_to_bitrate_bps, min_transmit_bitrate_bps);
>+
>+  return pad_up_to_bitrate_bps;
>+}
>+
>+uint32_t CalculateOverheadRateBps(int packets_per_second,
>+                                  size_t overhead_bytes_per_packet,
>+                                  uint32_t max_overhead_bps) {
>+  uint32_t overhead_bps =
>+      static_cast<uint32_t>(8 * overhead_bytes_per_packet * packets_per_second);
>+  return std::min(overhead_bps, max_overhead_bps);
>+}
>+
>+int CalculatePacketRate(uint32_t bitrate_bps, size_t packet_size_bytes) {
>+  size_t packet_size_bits = 8 * packet_size_bytes;
>+  // Ceil for int value of bitrate_bps / packet_size_bits.
>+  return static_cast<int>((bitrate_bps + packet_size_bits - 1) /
>+                          packet_size_bits);
>+}
>+
>+RtpSenderObservers CreateObservers(CallStats* call_stats,
>+                                   EncoderRtcpFeedback* encoder_feedback,
>+                                   SendStatisticsProxy* stats_proxy,
>+                                   SendDelayStats* send_delay_stats,
>+                                   OverheadObserver* overhead_observer) {
>+  RtpSenderObservers observers;
>+  observers.rtcp_rtt_stats = call_stats;
>+  observers.intra_frame_callback = encoder_feedback;
>+  observers.rtcp_stats = stats_proxy;
>+  observers.rtp_stats = stats_proxy;
>+  observers.bitrate_observer = stats_proxy;
>+  observers.frame_count_observer = stats_proxy;
>+  observers.rtcp_type_observer = stats_proxy;
>+  observers.send_delay_observer = stats_proxy;
>+  observers.send_packet_observer = send_delay_stats;
>+  observers.overhead_observer = overhead_observer;
>+  return observers;
>+}
>+}  // namespace
>+
>+// CheckEncoderActivityTask is used for tracking when the encoder last produced
>+// and encoded video frame. If the encoder has not produced anything the last
>+// kEncoderTimeOutMs we also want to stop sending padding.
>+class VideoSendStreamImpl::CheckEncoderActivityTask : public rtc::QueuedTask {
>+ public:
>+  static const int kEncoderTimeOutMs = 2000;
>+  explicit CheckEncoderActivityTask(
>+      const rtc::WeakPtr<VideoSendStreamImpl>& send_stream)
>+      : activity_(0), send_stream_(std::move(send_stream)), timed_out_(false) {}
>+
>+  void Stop() {
>+    RTC_CHECK(task_checker_.CalledSequentially());
>+    send_stream_.reset();
>+  }
>+
>+  void UpdateEncoderActivity() {
>+    // UpdateEncoderActivity is called from VideoSendStreamImpl::Encoded on
>+    // whatever thread the real encoder implementation run on. In the case of
>+    // hardware encoders, there might be several encoders
>+    // running in parallel on different threads.
>+    rtc::AtomicOps::ReleaseStore(&activity_, 1);
>+  }
>+
>+ private:
>+  bool Run() override {
>+    RTC_CHECK(task_checker_.CalledSequentially());
>+    if (!send_stream_)
>+      return true;
>+    if (!rtc::AtomicOps::AcquireLoad(&activity_)) {
>+      if (!timed_out_) {
>+        send_stream_->SignalEncoderTimedOut();
>+      }
>+      timed_out_ = true;
>+    } else if (timed_out_) {
>+      send_stream_->SignalEncoderActive();
>+      timed_out_ = false;
>+    }
>+    rtc::AtomicOps::ReleaseStore(&activity_, 0);
>+
>+    rtc::TaskQueue::Current()->PostDelayedTask(
>+        std::unique_ptr<rtc::QueuedTask>(this), kEncoderTimeOutMs);
>+    // Return false to prevent this task from being deleted. Ownership has been
>+    // transferred to the task queue when PostDelayedTask was called.
>+    return false;
>+  }
>+  volatile int activity_;
>+
>+  rtc::SequencedTaskChecker task_checker_;
>+  rtc::WeakPtr<VideoSendStreamImpl> send_stream_;
>+  bool timed_out_;
>+};
>+
>+VideoSendStreamImpl::VideoSendStreamImpl(
>+    SendStatisticsProxy* stats_proxy,
>+    rtc::TaskQueue* worker_queue,
>+    CallStats* call_stats,
>+    RtpTransportControllerSendInterface* transport,
>+    BitrateAllocatorInterface* bitrate_allocator,
>+    SendDelayStats* send_delay_stats,
>+    VideoStreamEncoderInterface* video_stream_encoder,
>+    RtcEventLog* event_log,
>+    const VideoSendStream::Config* config,
>+    int initial_encoder_max_bitrate,
>+    double initial_encoder_bitrate_priority,
>+    std::map<uint32_t, RtpState> suspended_ssrcs,
>+    std::map<uint32_t, RtpPayloadState> suspended_payload_states,
>+    VideoEncoderConfig::ContentType content_type,
>+    std::unique_ptr<FecController> fec_controller)
>+    : send_side_bwe_with_overhead_(
>+          webrtc::field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")),
>+      stats_proxy_(stats_proxy),
>+      config_(config),
>+      fec_controller_(std::move(fec_controller)),
>+      worker_queue_(worker_queue),
>+      check_encoder_activity_task_(nullptr),
>+      call_stats_(call_stats),
>+      transport_(transport),
>+      bitrate_allocator_(bitrate_allocator),
>+      max_padding_bitrate_(0),
>+      encoder_min_bitrate_bps_(0),
>+      encoder_target_rate_bps_(0),
>+      encoder_bitrate_priority_(initial_encoder_bitrate_priority),
>+      has_packet_feedback_(false),
>+      video_stream_encoder_(video_stream_encoder),
>+      encoder_feedback_(Clock::GetRealTimeClock(),
>+                        config_->rtp.ssrcs,
>+                        video_stream_encoder),
>+      bandwidth_observer_(transport->GetBandwidthObserver()),
>+      rtp_video_sender_(
>+          transport_->CreateRtpVideoSender(config_->rtp.ssrcs,
>+                                           suspended_ssrcs,
>+                                           suspended_payload_states,
>+                                           config_->rtp,
>+                                           config_->rtcp,
>+                                           config_->send_transport,
>+                                           CreateObservers(call_stats,
>+                                                           &encoder_feedback_,
>+                                                           stats_proxy_,
>+                                                           send_delay_stats,
>+                                                           this),
>+                                           event_log)),
>+      weak_ptr_factory_(this),
>+      overhead_bytes_per_packet_(0),
>+      transport_overhead_bytes_per_packet_(0) {
>+  RTC_DCHECK_RUN_ON(worker_queue_);
>+  RTC_LOG(LS_INFO) << "VideoSendStreamInternal: " << config_->ToString();
>+  weak_ptr_ = weak_ptr_factory_.GetWeakPtr();
>+
>+  RTC_DCHECK(!config_->rtp.ssrcs.empty());
>+  RTC_DCHECK(call_stats_);
>+  RTC_DCHECK(transport_);
>+  RTC_DCHECK_NE(initial_encoder_max_bitrate, 0);
>+
>+  if (initial_encoder_max_bitrate > 0) {
>+    encoder_max_bitrate_bps_ =
>+        rtc::dchecked_cast<uint32_t>(initial_encoder_max_bitrate);
>+  } else {
>+    // TODO(srte): Make sure max bitrate is not set to negative values. We don't
>+    // have any way to handle unset values in downstream code, such as the
>+    // bitrate allocator. Previously -1 was implicitly casted to UINT32_MAX, a
>+    // behaviour that is not safe. Converting to 10 Mbps should be safe for
>+    // reasonable use cases as it allows adding the max of multiple streams
>+    // without wrappping around.
>+    const int kFallbackMaxBitrateBps = 10000000;
>+    RTC_DLOG(LS_ERROR) << "ERROR: Initial encoder max bitrate = "
>+                       << initial_encoder_max_bitrate << " which is <= 0!";
>+    RTC_DLOG(LS_INFO) << "Using default encoder max bitrate = 10 Mbps";
>+    encoder_max_bitrate_bps_ = kFallbackMaxBitrateBps;
>+  }
>+
>+  RTC_CHECK(AlrExperimentSettings::MaxOneFieldTrialEnabled());
>+  // If send-side BWE is enabled, check if we should apply updated probing and
>+  // pacing settings.
>+  if (TransportSeqNumExtensionConfigured(*config_)) {
>+    has_packet_feedback_ = true;
>+
>+    absl::optional<AlrExperimentSettings> alr_settings;
>+    if (content_type == VideoEncoderConfig::ContentType::kScreen) {
>+      alr_settings = AlrExperimentSettings::CreateFromFieldTrial(
>+          AlrExperimentSettings::kScreenshareProbingBweExperimentName);
>+    } else {
>+      alr_settings = AlrExperimentSettings::CreateFromFieldTrial(
>+          AlrExperimentSettings::kStrictPacingAndProbingExperimentName);
>+    }
>+    if (alr_settings) {
>+      transport->EnablePeriodicAlrProbing(true);
>+      transport->SetPacingFactor(alr_settings->pacing_factor);
>+      configured_pacing_factor_ = alr_settings->pacing_factor;
>+      transport->SetQueueTimeLimit(alr_settings->max_paced_queue_time);
>+    } else {
>+      transport->EnablePeriodicAlrProbing(false);
>+      transport->SetPacingFactor(PacedSender::kDefaultPaceMultiplier);
>+      configured_pacing_factor_ = PacedSender::kDefaultPaceMultiplier;
>+      transport->SetQueueTimeLimit(PacedSender::kMaxQueueLengthMs);
>+    }
>+  }
>+
>+  if (config_->periodic_alr_bandwidth_probing) {
>+    transport->EnablePeriodicAlrProbing(true);
>+  }
>+
>+  // Currently, both ULPFEC and FlexFEC use the same FEC rate calculation logic,
>+  // so enable that logic if either of those FEC schemes are enabled.
>+  fec_controller_->SetProtectionMethod(rtp_video_sender_->FecEnabled(),
>+                                       rtp_video_sender_->NackEnabled());
>+
>+  fec_controller_->SetProtectionCallback(this);
>+  // Signal congestion controller this object is ready for OnPacket* callbacks.
>+  if (fec_controller_->UseLossVectorMask()) {
>+    transport_->RegisterPacketFeedbackObserver(this);
>+  }
>+
>+  RTC_DCHECK_GE(config_->rtp.payload_type, 0);
>+  RTC_DCHECK_LE(config_->rtp.payload_type, 127);
>+
>+  video_stream_encoder_->SetStartBitrate(
>+      bitrate_allocator_->GetStartBitrate(this));
>+
>+  // Only request rotation at the source when we positively know that the remote
>+  // side doesn't support the rotation extension. This allows us to prepare the
>+  // encoder in the expectation that rotation is supported - which is the common
>+  // case.
>+  bool rotation_applied =
>+      std::find_if(config_->rtp.extensions.begin(),
>+                   config_->rtp.extensions.end(),
>+                   [](const RtpExtension& extension) {
>+                     return extension.uri == RtpExtension::kVideoRotationUri;
>+                   }) == config_->rtp.extensions.end();
>+
>+  video_stream_encoder_->SetSink(this, rotation_applied);
>+}
>+
>+VideoSendStreamImpl::~VideoSendStreamImpl() {
>+  RTC_DCHECK_RUN_ON(worker_queue_);
>+  RTC_DCHECK(!rtp_video_sender_->IsActive())
>+      << "VideoSendStreamImpl::Stop not called";
>+  RTC_LOG(LS_INFO) << "~VideoSendStreamInternal: " << config_->ToString();
>+  if (fec_controller_->UseLossVectorMask()) {
>+    transport_->DeRegisterPacketFeedbackObserver(this);
>+  }
>+  transport_->DestroyRtpVideoSender(rtp_video_sender_);
>+}
>+
>+void VideoSendStreamImpl::RegisterProcessThread(
>+    ProcessThread* module_process_thread) {
>+  rtp_video_sender_->RegisterProcessThread(module_process_thread);
>+}
>+
>+void VideoSendStreamImpl::DeRegisterProcessThread() {
>+  rtp_video_sender_->DeRegisterProcessThread();
>+}
>+
>+bool VideoSendStreamImpl::DeliverRtcp(const uint8_t* packet, size_t length) {
>+  // Runs on a network thread.
>+  RTC_DCHECK(!worker_queue_->IsCurrent());
>+  rtp_video_sender_->DeliverRtcp(packet, length);
>+  return true;
>+}
>+
>+void VideoSendStreamImpl::UpdateActiveSimulcastLayers(
>+    const std::vector<bool> active_layers) {
>+  RTC_DCHECK_RUN_ON(worker_queue_);
>+  RTC_LOG(LS_INFO) << "VideoSendStream::UpdateActiveSimulcastLayers";
>+  bool previously_active = rtp_video_sender_->IsActive();
>+  rtp_video_sender_->SetActiveModules(active_layers);
>+  if (!rtp_video_sender_->IsActive() && previously_active) {
>+    // Payload router switched from active to inactive.
>+    StopVideoSendStream();
>+  } else if (rtp_video_sender_->IsActive() && !previously_active) {
>+    // Payload router switched from inactive to active.
>+    StartupVideoSendStream();
>+  }
>+}
>+
>+void VideoSendStreamImpl::Start() {
>+  RTC_DCHECK_RUN_ON(worker_queue_);
>+  RTC_LOG(LS_INFO) << "VideoSendStream::Start";
>+  if (rtp_video_sender_->IsActive())
>+    return;
>+  TRACE_EVENT_INSTANT0("webrtc", "VideoSendStream::Start");
>+  rtp_video_sender_->SetActive(true);
>+  StartupVideoSendStream();
>+}
>+
>+void VideoSendStreamImpl::StartupVideoSendStream() {
>+  RTC_DCHECK_RUN_ON(worker_queue_);
>+  bitrate_allocator_->AddObserver(
>+      this,
>+      MediaStreamAllocationConfig{
>+          static_cast<uint32_t>(encoder_min_bitrate_bps_),
>+          encoder_max_bitrate_bps_, static_cast<uint32_t>(max_padding_bitrate_),
>+          !config_->suspend_below_min_bitrate, config_->track_id,
>+          encoder_bitrate_priority_, has_packet_feedback_});
>+  // Start monitoring encoder activity.
>+  {
>+    rtc::CritScope lock(&encoder_activity_crit_sect_);
>+    RTC_DCHECK(!check_encoder_activity_task_);
>+    check_encoder_activity_task_ = new CheckEncoderActivityTask(weak_ptr_);
>+    worker_queue_->PostDelayedTask(
>+        std::unique_ptr<rtc::QueuedTask>(check_encoder_activity_task_),
>+        CheckEncoderActivityTask::kEncoderTimeOutMs);
>+  }
>+
>+  video_stream_encoder_->SendKeyFrame();
>+}
>+
>+void VideoSendStreamImpl::Stop() {
>+  RTC_DCHECK_RUN_ON(worker_queue_);
>+  RTC_LOG(LS_INFO) << "VideoSendStream::Stop";
>+  if (!rtp_video_sender_->IsActive())
>+    return;
>+  TRACE_EVENT_INSTANT0("webrtc", "VideoSendStream::Stop");
>+  rtp_video_sender_->SetActive(false);
>+  StopVideoSendStream();
>+}
>+
>+void VideoSendStreamImpl::StopVideoSendStream() {
>+  bitrate_allocator_->RemoveObserver(this);
>+  {
>+    rtc::CritScope lock(&encoder_activity_crit_sect_);
>+    check_encoder_activity_task_->Stop();
>+    check_encoder_activity_task_ = nullptr;
>+  }
>+  video_stream_encoder_->OnBitrateUpdated(0, 0, 0);
>+  stats_proxy_->OnSetEncoderTargetRate(0);
>+}
>+
>+void VideoSendStreamImpl::SignalEncoderTimedOut() {
>+  RTC_DCHECK_RUN_ON(worker_queue_);
>+  // If the encoder has not produced anything the last kEncoderTimeOutMs and it
>+  // is supposed to, deregister as BitrateAllocatorObserver. This can happen
>+  // if a camera stops producing frames.
>+  if (encoder_target_rate_bps_ > 0) {
>+    RTC_LOG(LS_INFO) << "SignalEncoderTimedOut, Encoder timed out.";
>+    bitrate_allocator_->RemoveObserver(this);
>+  }
>+}
>+
>+void VideoSendStreamImpl::OnBitrateAllocationUpdated(
>+    const VideoBitrateAllocation& allocation) {
>+  rtp_video_sender_->OnBitrateAllocationUpdated(allocation);
>+}
>+
>+void VideoSendStreamImpl::SignalEncoderActive() {
>+  RTC_DCHECK_RUN_ON(worker_queue_);
>+  RTC_LOG(LS_INFO) << "SignalEncoderActive, Encoder is active.";
>+  bitrate_allocator_->AddObserver(
>+      this,
>+      MediaStreamAllocationConfig{
>+          static_cast<uint32_t>(encoder_min_bitrate_bps_),
>+          encoder_max_bitrate_bps_, static_cast<uint32_t>(max_padding_bitrate_),
>+          !config_->suspend_below_min_bitrate, config_->track_id,
>+          encoder_bitrate_priority_, has_packet_feedback_});
>+}
>+
>+void VideoSendStreamImpl::OnEncoderConfigurationChanged(
>+    std::vector<VideoStream> streams,
>+    int min_transmit_bitrate_bps) {
>+  if (!worker_queue_->IsCurrent()) {
>+    rtc::WeakPtr<VideoSendStreamImpl> send_stream = weak_ptr_;
>+    worker_queue_->PostTask([send_stream, streams, min_transmit_bitrate_bps]() {
>+      if (send_stream)
>+        send_stream->OnEncoderConfigurationChanged(std::move(streams),
>+                                                   min_transmit_bitrate_bps);
>+    });
>+    return;
>+  }
>+  RTC_DCHECK_GE(config_->rtp.ssrcs.size(), streams.size());
>+  TRACE_EVENT0("webrtc", "VideoSendStream::OnEncoderConfigurationChanged");
>+  RTC_DCHECK_GE(config_->rtp.ssrcs.size(), streams.size());
>+  RTC_DCHECK_RUN_ON(worker_queue_);
>+
>+  encoder_min_bitrate_bps_ =
>+      std::max(streams[0].min_bitrate_bps, GetEncoderMinBitrateBps());
>+  encoder_max_bitrate_bps_ = 0;
>+  double stream_bitrate_priority_sum = 0;
>+  for (const auto& stream : streams) {
>+    // We don't want to allocate more bitrate than needed to inactive streams.
>+    encoder_max_bitrate_bps_ += stream.active ? stream.max_bitrate_bps : 0;
>+    if (stream.bitrate_priority) {
>+      RTC_DCHECK_GT(*stream.bitrate_priority, 0);
>+      stream_bitrate_priority_sum += *stream.bitrate_priority;
>+    }
>+  }
>+  RTC_DCHECK_GT(stream_bitrate_priority_sum, 0);
>+  encoder_bitrate_priority_ = stream_bitrate_priority_sum;
>+  encoder_max_bitrate_bps_ =
>+      std::max(static_cast<uint32_t>(encoder_min_bitrate_bps_),
>+               encoder_max_bitrate_bps_);
>+
>+  const VideoCodecType codec_type =
>+      PayloadStringToCodecType(config_->rtp.payload_name);
>+  if (codec_type == kVideoCodecVP9) {
>+    max_padding_bitrate_ = streams[0].target_bitrate_bps;
>+  } else {
>+    max_padding_bitrate_ = CalculateMaxPadBitrateBps(
>+        streams, min_transmit_bitrate_bps, config_->suspend_below_min_bitrate);
>+  }
>+
>+  // Clear stats for disabled layers.
>+  for (size_t i = streams.size(); i < config_->rtp.ssrcs.size(); ++i) {
>+    stats_proxy_->OnInactiveSsrc(config_->rtp.ssrcs[i]);
>+  }
>+
>+  const size_t num_temporal_layers =
>+      streams.back().num_temporal_layers.value_or(1);
>+  fec_controller_->SetEncodingData(streams[0].width, streams[0].height,
>+                                   num_temporal_layers,
>+                                   config_->rtp.max_packet_size);
>+
>+  if (rtp_video_sender_->IsActive()) {
>+    // The send stream is started already. Update the allocator with new bitrate
>+    // limits.
>+    bitrate_allocator_->AddObserver(
>+        this, MediaStreamAllocationConfig{
>+                  static_cast<uint32_t>(encoder_min_bitrate_bps_),
>+                  encoder_max_bitrate_bps_,
>+                  static_cast<uint32_t>(max_padding_bitrate_),
>+                  !config_->suspend_below_min_bitrate, config_->track_id,
>+                  encoder_bitrate_priority_, has_packet_feedback_});
>+  }
>+}
>+
>+EncodedImageCallback::Result VideoSendStreamImpl::OnEncodedImage(
>+    const EncodedImage& encoded_image,
>+    const CodecSpecificInfo* codec_specific_info,
>+    const RTPFragmentationHeader* fragmentation) {
>+  // Encoded is called on whatever thread the real encoder implementation run
>+  // on. In the case of hardware encoders, there might be several encoders
>+  // running in parallel on different threads.
>+  size_t simulcast_idx = 0;
>+  if (codec_specific_info->codecType == kVideoCodecVP8) {
>+    simulcast_idx = codec_specific_info->codecSpecific.VP8.simulcastIdx;
>+  }
>+  if (config_->post_encode_callback) {
>+    config_->post_encode_callback->EncodedFrameCallback(EncodedFrame(
>+        encoded_image._buffer, encoded_image._length, encoded_image._frameType,
>+        simulcast_idx, encoded_image._timeStamp));
>+  }
>+  {
>+    rtc::CritScope lock(&encoder_activity_crit_sect_);
>+    if (check_encoder_activity_task_)
>+      check_encoder_activity_task_->UpdateEncoderActivity();
>+  }
>+
>+  fec_controller_->UpdateWithEncodedData(encoded_image._length,
>+                                         encoded_image._frameType);
>+  EncodedImageCallback::Result result = rtp_video_sender_->OnEncodedImage(
>+      encoded_image, codec_specific_info, fragmentation);
>+
>+  RTC_DCHECK(codec_specific_info);
>+
>+  int layer = codec_specific_info->codecType == kVideoCodecVP8
>+                  ? codec_specific_info->codecSpecific.VP8.simulcastIdx
>+                  : 0;
>+  {
>+    rtc::CritScope lock(&ivf_writers_crit_);
>+    if (file_writers_[layer].get()) {
>+      bool ok = file_writers_[layer]->WriteFrame(
>+          encoded_image, codec_specific_info->codecType);
>+      RTC_DCHECK(ok);
>+    }
>+  }
>+
>+  return result;
>+}
>+
>+std::map<uint32_t, RtpState> VideoSendStreamImpl::GetRtpStates() const {
>+  return rtp_video_sender_->GetRtpStates();
>+}
>+
>+std::map<uint32_t, RtpPayloadState> VideoSendStreamImpl::GetRtpPayloadStates()
>+    const {
>+  return rtp_video_sender_->GetRtpPayloadStates();
>+}
>+
>+uint32_t VideoSendStreamImpl::OnBitrateUpdated(uint32_t bitrate_bps,
>+                                               uint8_t fraction_loss,
>+                                               int64_t rtt,
>+                                               int64_t probing_interval_ms) {
>+  RTC_DCHECK_RUN_ON(worker_queue_);
>+  RTC_DCHECK(rtp_video_sender_->IsActive())
>+      << "VideoSendStream::Start has not been called.";
>+
>+  // Substract overhead from bitrate.
>+  rtc::CritScope lock(&overhead_bytes_per_packet_crit_);
>+  uint32_t payload_bitrate_bps = bitrate_bps;
>+  if (send_side_bwe_with_overhead_) {
>+    payload_bitrate_bps -= CalculateOverheadRateBps(
>+        CalculatePacketRate(bitrate_bps,
>+                            config_->rtp.max_packet_size +
>+                                transport_overhead_bytes_per_packet_),
>+        overhead_bytes_per_packet_ + transport_overhead_bytes_per_packet_,
>+        bitrate_bps);
>+  }
>+
>+  // Get the encoder target rate. It is the estimated network rate -
>+  // protection overhead.
>+  encoder_target_rate_bps_ = fec_controller_->UpdateFecRates(
>+      payload_bitrate_bps, stats_proxy_->GetSendFrameRate(), fraction_loss,
>+      loss_mask_vector_, rtt);
>+  loss_mask_vector_.clear();
>+
>+  uint32_t encoder_overhead_rate_bps =
>+      send_side_bwe_with_overhead_
>+          ? CalculateOverheadRateBps(
>+                CalculatePacketRate(encoder_target_rate_bps_,
>+                                    config_->rtp.max_packet_size +
>+                                        transport_overhead_bytes_per_packet_ -
>+                                        overhead_bytes_per_packet_),
>+                overhead_bytes_per_packet_ +
>+                    transport_overhead_bytes_per_packet_,
>+                bitrate_bps - encoder_target_rate_bps_)
>+          : 0;
>+
>+  // When the field trial "WebRTC-SendSideBwe-WithOverhead" is enabled
>+  // protection_bitrate includes overhead.
>+  uint32_t protection_bitrate =
>+      bitrate_bps - (encoder_target_rate_bps_ + encoder_overhead_rate_bps);
>+
>+  encoder_target_rate_bps_ =
>+      std::min(encoder_max_bitrate_bps_, encoder_target_rate_bps_);
>+  video_stream_encoder_->OnBitrateUpdated(encoder_target_rate_bps_,
>+                                          fraction_loss, rtt);
>+  stats_proxy_->OnSetEncoderTargetRate(encoder_target_rate_bps_);
>+  return protection_bitrate;
>+}
>+
>+void VideoSendStreamImpl::EnableEncodedFrameRecording(
>+    const std::vector<rtc::PlatformFile>& files,
>+    size_t byte_limit) {
>+  {
>+    rtc::CritScope lock(&ivf_writers_crit_);
>+    for (unsigned int i = 0; i < kMaxSimulcastStreams; ++i) {
>+      if (i < files.size()) {
>+        file_writers_[i] = IvfFileWriter::Wrap(rtc::File(files[i]), byte_limit);
>+      } else {
>+        file_writers_[i].reset();
>+      }
>+    }
>+  }
>+
>+  if (!files.empty()) {
>+    // Make a keyframe appear as early as possible in the logs, to give actually
>+    // decodable output.
>+    video_stream_encoder_->SendKeyFrame();
>+  }
>+}
>+
>+int VideoSendStreamImpl::ProtectionRequest(
>+    const FecProtectionParams* delta_params,
>+    const FecProtectionParams* key_params,
>+    uint32_t* sent_video_rate_bps,
>+    uint32_t* sent_nack_rate_bps,
>+    uint32_t* sent_fec_rate_bps) {
>+  RTC_DCHECK_RUN_ON(worker_queue_);
>+  rtp_video_sender_->ProtectionRequest(delta_params, key_params,
>+                                       sent_video_rate_bps, sent_nack_rate_bps,
>+                                       sent_fec_rate_bps);
>+  return 0;
>+}
>+
>+void VideoSendStreamImpl::OnOverheadChanged(size_t overhead_bytes_per_packet) {
>+  rtc::CritScope lock(&overhead_bytes_per_packet_crit_);
>+  overhead_bytes_per_packet_ = overhead_bytes_per_packet;
>+}
>+
>+void VideoSendStreamImpl::SetTransportOverhead(
>+    size_t transport_overhead_bytes_per_packet) {
>+  if (transport_overhead_bytes_per_packet >= static_cast<int>(kPathMTU)) {
>+    RTC_LOG(LS_ERROR) << "Transport overhead exceeds size of ethernet frame";
>+    return;
>+  }
>+
>+  transport_overhead_bytes_per_packet_ = transport_overhead_bytes_per_packet;
>+
>+  size_t rtp_packet_size =
>+      std::min(config_->rtp.max_packet_size,
>+               kPathMTU - transport_overhead_bytes_per_packet_);
>+
>+  rtp_video_sender_->SetMaxRtpPacketSize(rtp_packet_size);
>+}
>+
>+void VideoSendStreamImpl::OnPacketAdded(uint32_t ssrc, uint16_t seq_num) {
>+  if (!worker_queue_->IsCurrent()) {
>+    auto ptr = weak_ptr_;
>+    worker_queue_->PostTask([=] {
>+      if (!ptr.get())
>+        return;
>+      ptr->OnPacketAdded(ssrc, seq_num);
>+    });
>+    return;
>+  }
>+  const auto ssrcs = config_->rtp.ssrcs;
>+  if (std::find(ssrcs.begin(), ssrcs.end(), ssrc) != ssrcs.end()) {
>+    feedback_packet_seq_num_set_.insert(seq_num);
>+    if (feedback_packet_seq_num_set_.size() > kSendSideSeqNumSetMaxSize) {
>+      RTC_LOG(LS_WARNING) << "Feedback packet sequence number set exceed it's "
>+                             "max size', will get reset.";
>+      feedback_packet_seq_num_set_.clear();
>+    }
>+  }
>+}
>+
>+void VideoSendStreamImpl::OnPacketFeedbackVector(
>+    const std::vector<PacketFeedback>& packet_feedback_vector) {
>+  if (!worker_queue_->IsCurrent()) {
>+    auto ptr = weak_ptr_;
>+    worker_queue_->PostTask([=] {
>+      if (!ptr.get())
>+        return;
>+      ptr->OnPacketFeedbackVector(packet_feedback_vector);
>+    });
>+    return;
>+  }
>+  // Lost feedbacks are not considered to be lost packets.
>+  for (const PacketFeedback& packet : packet_feedback_vector) {
>+    if (auto it = feedback_packet_seq_num_set_.find(packet.sequence_number) !=
>+                  feedback_packet_seq_num_set_.end()) {
>+      const bool lost = packet.arrival_time_ms == PacketFeedback::kNotReceived;
>+      loss_mask_vector_.push_back(lost);
>+      feedback_packet_seq_num_set_.erase(it);
>+    }
>+  }
>+}
>+}  // namespace internal
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream_impl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream_impl.h
>new file mode 100644
>index 00000000000..9e4359929da
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream_impl.h
>@@ -0,0 +1,193 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+#ifndef VIDEO_VIDEO_SEND_STREAM_IMPL_H_
>+#define VIDEO_VIDEO_SEND_STREAM_IMPL_H_
>+
>+#include <map>
>+#include <memory>
>+#include <unordered_set>
>+#include <vector>
>+
>+#include "api/video/video_bitrate_allocator.h"
>+#include "api/video/video_stream_encoder_interface.h"
>+#include "call/bitrate_allocator.h"
>+#include "call/rtp_video_sender_interface.h"
>+#include "common_types.h"  // NOLINT(build/include)
>+#include "modules/utility/include/process_thread.h"
>+#include "modules/video_coding/utility/ivf_file_writer.h"
>+#include "rtc_base/weak_ptr.h"
>+#include "video/call_stats.h"
>+#include "video/encoder_rtcp_feedback.h"
>+#include "video/send_delay_stats.h"
>+#include "video/send_statistics_proxy.h"
>+#include "video/video_send_stream.h"
>+
>+namespace webrtc {
>+namespace internal {
>+
>+// VideoSendStreamImpl implements internal::VideoSendStream.
>+// It is created and destroyed on |worker_queue|. The intent is to decrease the
>+// need for locking and to ensure methods are called in sequence.
>+// Public methods except |DeliverRtcp| must be called on |worker_queue|.
>+// DeliverRtcp is called on the libjingle worker thread or a network thread.
>+// An encoder may deliver frames through the EncodedImageCallback on an
>+// arbitrary thread.
>+class VideoSendStreamImpl : public webrtc::BitrateAllocatorObserver,
>+                            public webrtc::OverheadObserver,
>+                            public webrtc::VCMProtectionCallback,
>+                            public VideoStreamEncoderInterface::EncoderSink,
>+                            public VideoBitrateAllocationObserver,
>+                            public webrtc::PacketFeedbackObserver {
>+ public:
>+  VideoSendStreamImpl(
>+      SendStatisticsProxy* stats_proxy,
>+      rtc::TaskQueue* worker_queue,
>+      CallStats* call_stats,
>+      RtpTransportControllerSendInterface* transport,
>+      BitrateAllocatorInterface* bitrate_allocator,
>+      SendDelayStats* send_delay_stats,
>+      VideoStreamEncoderInterface* video_stream_encoder,
>+      RtcEventLog* event_log,
>+      const VideoSendStream::Config* config,
>+      int initial_encoder_max_bitrate,
>+      double initial_encoder_bitrate_priority,
>+      std::map<uint32_t, RtpState> suspended_ssrcs,
>+      std::map<uint32_t, RtpPayloadState> suspended_payload_states,
>+      VideoEncoderConfig::ContentType content_type,
>+      std::unique_ptr<FecController> fec_controller);
>+  ~VideoSendStreamImpl() override;
>+
>+  // RegisterProcessThread register |module_process_thread| with those objects
>+  // that use it. Registration has to happen on the thread were
>+  // |module_process_thread| was created (libjingle's worker thread).
>+  // TODO(perkj): Replace the use of |module_process_thread| with a TaskQueue,
>+  // maybe |worker_queue|.
>+  void RegisterProcessThread(ProcessThread* module_process_thread);
>+  void DeRegisterProcessThread();
>+
>+  bool DeliverRtcp(const uint8_t* packet, size_t length);
>+  void UpdateActiveSimulcastLayers(const std::vector<bool> active_layers);
>+  void Start();
>+  void Stop();
>+
>+  // TODO(holmer): Move these to RtpTransportControllerSend.
>+  std::map<uint32_t, RtpState> GetRtpStates() const;
>+
>+  std::map<uint32_t, RtpPayloadState> GetRtpPayloadStates() const;
>+
>+  void EnableEncodedFrameRecording(const std::vector<rtc::PlatformFile>& files,
>+                                   size_t byte_limit);
>+
>+  void SetTransportOverhead(size_t transport_overhead_per_packet);
>+
>+  absl::optional<float> configured_pacing_factor_;
>+
>+  // From PacketFeedbackObserver.
>+  void OnPacketAdded(uint32_t ssrc, uint16_t seq_num) override;
>+  void OnPacketFeedbackVector(
>+      const std::vector<PacketFeedback>& packet_feedback_vector) override;
>+
>+ private:
>+  class CheckEncoderActivityTask;
>+
>+  // Implements BitrateAllocatorObserver.
>+  uint32_t OnBitrateUpdated(uint32_t bitrate_bps,
>+                            uint8_t fraction_loss,
>+                            int64_t rtt,
>+                            int64_t probing_interval_ms) override;
>+
>+  // Implements webrtc::VCMProtectionCallback.
>+  int ProtectionRequest(const FecProtectionParams* delta_params,
>+                        const FecProtectionParams* key_params,
>+                        uint32_t* sent_video_rate_bps,
>+                        uint32_t* sent_nack_rate_bps,
>+                        uint32_t* sent_fec_rate_bps) override;
>+
>+  // Implements OverheadObserver.
>+  void OnOverheadChanged(size_t overhead_bytes_per_packet) override;
>+
>+  void OnEncoderConfigurationChanged(std::vector<VideoStream> streams,
>+                                     int min_transmit_bitrate_bps) override;
>+
>+  // Implements EncodedImageCallback. The implementation routes encoded frames
>+  // to the |payload_router_| and |config.pre_encode_callback| if set.
>+  // Called on an arbitrary encoder callback thread.
>+  EncodedImageCallback::Result OnEncodedImage(
>+      const EncodedImage& encoded_image,
>+      const CodecSpecificInfo* codec_specific_info,
>+      const RTPFragmentationHeader* fragmentation) override;
>+
>+  // Implements VideoBitrateAllocationObserver.
>+  void OnBitrateAllocationUpdated(
>+      const VideoBitrateAllocation& allocation) override;
>+
>+  // Starts monitoring and sends a keyframe.
>+  void StartupVideoSendStream();
>+  // Removes the bitrate observer, stops monitoring and notifies the video
>+  // encoder of the bitrate update.
>+  void StopVideoSendStream();
>+
>+  void ConfigureProtection();
>+  void ConfigureSsrcs();
>+  void SignalEncoderTimedOut();
>+  void SignalEncoderActive();
>+
>+  const bool send_side_bwe_with_overhead_;
>+
>+  SendStatisticsProxy* const stats_proxy_;
>+  const VideoSendStream::Config* const config_;
>+
>+  std::unique_ptr<FecController> fec_controller_;
>+  rtc::TaskQueue* const worker_queue_;
>+
>+  rtc::CriticalSection encoder_activity_crit_sect_;
>+  CheckEncoderActivityTask* check_encoder_activity_task_
>+      RTC_GUARDED_BY(encoder_activity_crit_sect_);
>+
>+  CallStats* const call_stats_;
>+  RtpTransportControllerSendInterface* const transport_;
>+  BitrateAllocatorInterface* const bitrate_allocator_;
>+
>+  rtc::CriticalSection ivf_writers_crit_;
>+  std::unique_ptr<IvfFileWriter>
>+      file_writers_[kMaxSimulcastStreams] RTC_GUARDED_BY(ivf_writers_crit_);
>+
>+  int max_padding_bitrate_;
>+  int encoder_min_bitrate_bps_;
>+  uint32_t encoder_max_bitrate_bps_;
>+  uint32_t encoder_target_rate_bps_;
>+  double encoder_bitrate_priority_;
>+  bool has_packet_feedback_;
>+
>+  VideoStreamEncoderInterface* const video_stream_encoder_;
>+  EncoderRtcpFeedback encoder_feedback_;
>+
>+  RtcpBandwidthObserver* const bandwidth_observer_;
>+  RtpVideoSenderInterface* const rtp_video_sender_;
>+
>+  // |weak_ptr_| to our self. This is used since we can not call
>+  // |weak_ptr_factory_.GetWeakPtr| from multiple sequences but it is ok to copy
>+  // an existing WeakPtr.
>+  rtc::WeakPtr<VideoSendStreamImpl> weak_ptr_;
>+  // |weak_ptr_factory_| must be declared last to make sure all WeakPtr's are
>+  // invalidated before any other members are destroyed.
>+  rtc::WeakPtrFactory<VideoSendStreamImpl> weak_ptr_factory_;
>+
>+  rtc::CriticalSection overhead_bytes_per_packet_crit_;
>+  size_t overhead_bytes_per_packet_
>+      RTC_GUARDED_BY(overhead_bytes_per_packet_crit_);
>+  size_t transport_overhead_bytes_per_packet_;
>+
>+  std::unordered_set<uint16_t> feedback_packet_seq_num_set_;
>+  std::vector<bool> loss_mask_vector_;
>+};
>+}  // namespace internal
>+}  // namespace webrtc
>+#endif  // VIDEO_VIDEO_SEND_STREAM_IMPL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream_impl_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream_impl_unittest.cc
>new file mode 100644
>index 00000000000..78a20c038cc
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream_impl_unittest.cc
>@@ -0,0 +1,219 @@
>+/*
>+ *  Copyright 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include <string>
>+
>+#include "call/rtp_video_sender.h"
>+#include "call/test/mock_bitrate_allocator.h"
>+#include "call/test/mock_rtp_transport_controller_send.h"
>+#include "logging/rtc_event_log/rtc_event_log.h"
>+#include "modules/utility/include/process_thread.h"
>+#include "modules/video_coding/fec_controller_default.h"
>+#include "rtc_base/experiments/alr_experiment.h"
>+#include "rtc_base/task_queue_for_test.h"
>+#include "test/field_trial.h"
>+#include "test/gmock.h"
>+#include "test/gtest.h"
>+#include "test/mock_transport.h"
>+#include "video/test/mock_video_stream_encoder.h"
>+#include "video/video_send_stream_impl.h"
>+
>+namespace webrtc {
>+namespace internal {
>+namespace {
>+using testing::NiceMock;
>+using testing::StrictMock;
>+using testing::ReturnRef;
>+using testing::Return;
>+using testing::Invoke;
>+using testing::_;
>+
>+constexpr int64_t kDefaultInitialBitrateBps = 333000;
>+const double kDefaultBitratePriority = 0.5;
>+
>+const float kAlrProbingExperimentPaceMultiplier = 1.0f;
>+std::string GetAlrProbingExperimentString() {
>+  return std::string(
>+             AlrExperimentSettings::kScreenshareProbingBweExperimentName) +
>+         "/1.0,2875,80,40,-60,3/";
>+}
>+class MockPayloadRouter : public RtpVideoSenderInterface {
>+ public:
>+  MOCK_METHOD1(RegisterProcessThread, void(ProcessThread*));
>+  MOCK_METHOD0(DeRegisterProcessThread, void());
>+  MOCK_METHOD1(SetActive, void(bool));
>+  MOCK_METHOD1(SetActiveModules, void(const std::vector<bool>));
>+  MOCK_METHOD0(IsActive, bool());
>+  MOCK_METHOD1(OnNetworkAvailability, void(bool));
>+  MOCK_CONST_METHOD0(GetRtpStates, std::map<uint32_t, RtpState>());
>+  MOCK_CONST_METHOD0(GetRtpPayloadStates,
>+                     std::map<uint32_t, RtpPayloadState>());
>+  MOCK_CONST_METHOD0(FecEnabled, bool());
>+  MOCK_CONST_METHOD0(NackEnabled, bool());
>+  MOCK_METHOD2(DeliverRtcp, void(const uint8_t*, size_t));
>+  MOCK_METHOD5(ProtectionRequest,
>+               void(const FecProtectionParams*,
>+                    const FecProtectionParams*,
>+                    uint32_t*,
>+                    uint32_t*,
>+                    uint32_t*));
>+  MOCK_METHOD1(SetMaxRtpPacketSize, void(size_t));
>+  MOCK_METHOD1(OnBitrateAllocationUpdated, void(const VideoBitrateAllocation&));
>+  MOCK_METHOD3(OnEncodedImage,
>+               EncodedImageCallback::Result(const EncodedImage&,
>+                                            const CodecSpecificInfo*,
>+                                            const RTPFragmentationHeader*));
>+};
>+}  // namespace
>+
>+class VideoSendStreamImplTest : public ::testing::Test {
>+ protected:
>+  VideoSendStreamImplTest()
>+      : clock_(1000 * 1000 * 1000),
>+        config_(&transport_),
>+        send_delay_stats_(&clock_),
>+        test_queue_("test_queue"),
>+        process_thread_(ProcessThread::Create("test_thread")),
>+        call_stats_(&clock_, process_thread_.get()),
>+        stats_proxy_(&clock_,
>+                     config_,
>+                     VideoEncoderConfig::ContentType::kRealtimeVideo) {
>+    config_.rtp.ssrcs.push_back(8080);
>+    config_.rtp.payload_type = 1;
>+
>+    EXPECT_CALL(transport_controller_, keepalive_config())
>+        .WillRepeatedly(ReturnRef(keepalive_config_));
>+    EXPECT_CALL(transport_controller_, packet_router())
>+        .WillRepeatedly(Return(&packet_router_));
>+    EXPECT_CALL(transport_controller_,
>+                CreateRtpVideoSender(_, _, _, _, _, _, _, _))
>+        .WillRepeatedly(Return(&payload_router_));
>+    EXPECT_CALL(payload_router_, SetActive(_))
>+        .WillRepeatedly(testing::Invoke(
>+            [&](bool active) { payload_router_active_ = active; }));
>+    EXPECT_CALL(payload_router_, IsActive())
>+        .WillRepeatedly(
>+            testing::Invoke([&]() { return payload_router_active_; }));
>+  }
>+  ~VideoSendStreamImplTest() {}
>+
>+  std::unique_ptr<VideoSendStreamImpl> CreateVideoSendStreamImpl(
>+      int initial_encoder_max_bitrate,
>+      double initial_encoder_bitrate_priority,
>+      VideoEncoderConfig::ContentType content_type) {
>+    EXPECT_CALL(bitrate_allocator_, GetStartBitrate(_))
>+        .WillOnce(Return(123000));
>+    std::map<uint32_t, RtpState> suspended_ssrcs;
>+    std::map<uint32_t, RtpPayloadState> suspended_payload_states;
>+    return absl::make_unique<VideoSendStreamImpl>(
>+        &stats_proxy_, &test_queue_, &call_stats_, &transport_controller_,
>+        &bitrate_allocator_, &send_delay_stats_, &video_stream_encoder_,
>+        &event_log_, &config_, initial_encoder_max_bitrate,
>+        initial_encoder_bitrate_priority, suspended_ssrcs,
>+        suspended_payload_states, content_type,
>+        absl::make_unique<FecControllerDefault>(&clock_));
>+  }
>+
>+ protected:
>+  NiceMock<MockTransport> transport_;
>+  NiceMock<MockRtpTransportControllerSend> transport_controller_;
>+  NiceMock<MockBitrateAllocator> bitrate_allocator_;
>+  NiceMock<MockVideoStreamEncoder> video_stream_encoder_;
>+  NiceMock<MockPayloadRouter> payload_router_;
>+
>+  bool payload_router_active_ = false;
>+  SimulatedClock clock_;
>+  RtcEventLogNullImpl event_log_;
>+  VideoSendStream::Config config_;
>+  SendDelayStats send_delay_stats_;
>+  rtc::test::TaskQueueForTest test_queue_;
>+  std::unique_ptr<ProcessThread> process_thread_;
>+  CallStats call_stats_;
>+  SendStatisticsProxy stats_proxy_;
>+  PacketRouter packet_router_;
>+  RtpKeepAliveConfig keepalive_config_;
>+};
>+
>+TEST_F(VideoSendStreamImplTest, RegistersAsBitrateObserverOnStart) {
>+  test_queue_.SendTask([this] {
>+    config_.track_id = "test";
>+    const bool kSuspend = false;
>+    config_.suspend_below_min_bitrate = kSuspend;
>+    auto vss_impl = CreateVideoSendStreamImpl(
>+        kDefaultInitialBitrateBps, kDefaultBitratePriority,
>+        VideoEncoderConfig::ContentType::kRealtimeVideo);
>+    EXPECT_CALL(bitrate_allocator_, AddObserver(vss_impl.get(), _))
>+        .WillOnce(Invoke(
>+            [&](BitrateAllocatorObserver*, MediaStreamAllocationConfig config) {
>+              EXPECT_EQ(config.min_bitrate_bps, 0u);
>+              EXPECT_EQ(config.max_bitrate_bps, kDefaultInitialBitrateBps);
>+              EXPECT_EQ(config.pad_up_bitrate_bps, 0u);
>+              EXPECT_EQ(config.enforce_min_bitrate, !kSuspend);
>+              EXPECT_EQ(config.track_id, "test");
>+              EXPECT_EQ(config.bitrate_priority, kDefaultBitratePriority);
>+              EXPECT_EQ(config.has_packet_feedback, false);
>+            }));
>+    vss_impl->Start();
>+    EXPECT_CALL(bitrate_allocator_, RemoveObserver(vss_impl.get())).Times(1);
>+    vss_impl->Stop();
>+  });
>+}
>+
>+TEST_F(VideoSendStreamImplTest, ReportFeedbackAvailability) {
>+  test_queue_.SendTask([this] {
>+    config_.rtp.extensions.emplace_back(
>+        RtpExtension::kTransportSequenceNumberUri,
>+        RtpExtension::kTransportSequenceNumberDefaultId);
>+
>+    auto vss_impl = CreateVideoSendStreamImpl(
>+        kDefaultInitialBitrateBps, kDefaultBitratePriority,
>+        VideoEncoderConfig::ContentType::kRealtimeVideo);
>+    EXPECT_CALL(bitrate_allocator_, AddObserver(vss_impl.get(), _))
>+        .WillOnce(Invoke(
>+            [&](BitrateAllocatorObserver*, MediaStreamAllocationConfig config) {
>+              EXPECT_EQ(config.has_packet_feedback, true);
>+            }));
>+    vss_impl->Start();
>+    EXPECT_CALL(bitrate_allocator_, RemoveObserver(vss_impl.get())).Times(1);
>+    vss_impl->Stop();
>+  });
>+}
>+
>+TEST_F(VideoSendStreamImplTest, SetsScreensharePacingFactorWithFeedback) {
>+  test::ScopedFieldTrials alr_experiment(GetAlrProbingExperimentString());
>+
>+  test_queue_.SendTask([this] {
>+    config_.rtp.extensions.emplace_back(
>+        RtpExtension::kTransportSequenceNumberUri,
>+        RtpExtension::kTransportSequenceNumberDefaultId);
>+    EXPECT_CALL(transport_controller_,
>+                SetPacingFactor(kAlrProbingExperimentPaceMultiplier))
>+        .Times(1);
>+    auto vss_impl = CreateVideoSendStreamImpl(
>+        kDefaultInitialBitrateBps, kDefaultBitratePriority,
>+        VideoEncoderConfig::ContentType::kScreen);
>+    vss_impl->Start();
>+    vss_impl->Stop();
>+  });
>+}
>+
>+TEST_F(VideoSendStreamImplTest, DoesNotSetPacingFactorWithoutFeedback) {
>+  test::ScopedFieldTrials alr_experiment(GetAlrProbingExperimentString());
>+  test_queue_.SendTask([this] {
>+    EXPECT_CALL(transport_controller_, SetPacingFactor(_)).Times(0);
>+    auto vss_impl = CreateVideoSendStreamImpl(
>+        kDefaultInitialBitrateBps, kDefaultBitratePriority,
>+        VideoEncoderConfig::ContentType::kScreen);
>+    vss_impl->Start();
>+    vss_impl->Stop();
>+  });
>+}
>+}  // namespace internal
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream_tests.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream_tests.cc
>index f3b92fbc7ee..4bf5b99035a 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream_tests.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_send_stream_tests.cc
>@@ -33,23 +33,44 @@
> #include "system_wrappers/include/sleep.h"
> #include "test/call_test.h"
> #include "test/configurable_frame_size_encoder.h"
>+#include "test/encoder_proxy_factory.h"
>+#include "test/fake_encoder.h"
> #include "test/fake_texture_frame.h"
> #include "test/field_trial.h"
> #include "test/frame_generator.h"
> #include "test/frame_generator_capturer.h"
> #include "test/frame_utils.h"
>+#include "test/gmock.h"
> #include "test/gtest.h"
> #include "test/null_transport.h"
> #include "test/rtcp_packet_parser.h"
> #include "test/testsupport/perf_test.h"
> 
>+#include "call/video_send_stream.h"
> #include "video/send_statistics_proxy.h"
> #include "video/transport_adapter.h"
>-#include "call/video_send_stream.h"
>+#include "video/video_send_stream.h"
> 
> namespace webrtc {
>+namespace test {
>+class VideoSendStreamPeer {
>+ public:
>+  explicit VideoSendStreamPeer(webrtc::VideoSendStream* base_class_stream)
>+      : internal_stream_(
>+            static_cast<internal::VideoSendStream*>(base_class_stream)) {}
>+  absl::optional<float> GetPacingFactorOverride() const {
>+    return internal_stream_->GetPacingFactorOverride();
>+  }
> 
>-enum VideoFormat { kGeneric, kVP8, };
>+ private:
>+  internal::VideoSendStream const* const internal_stream_;
>+};
>+}  // namespace test
>+
>+enum VideoFormat {
>+  kGeneric,
>+  kVP8,
>+};
> 
> void ExpectEqualFramesVector(const std::vector<VideoFrame>& frames1,
>                              const std::vector<VideoFrame>& frames2);
>@@ -69,13 +90,13 @@ class VideoSendStreamTest : public test::CallTest {
> 
> TEST_F(VideoSendStreamTest, CanStartStartedStream) {
>   task_queue_.SendTask([this]() {
>-    CreateSenderCall(Call::Config(event_log_.get()));
>+    CreateSenderCall();
> 
>     test::NullTransport transport;
>     CreateSendConfig(1, 0, 0, &transport);
>     CreateVideoStreams();
>-    video_send_stream_->Start();
>-    video_send_stream_->Start();
>+    GetVideoSendStream()->Start();
>+    GetVideoSendStream()->Start();
>     DestroyStreams();
>     DestroyCalls();
>   });
>@@ -83,13 +104,13 @@ TEST_F(VideoSendStreamTest, CanStartStartedStream) {
> 
> TEST_F(VideoSendStreamTest, CanStopStoppedStream) {
>   task_queue_.SendTask([this]() {
>-    CreateSenderCall(Call::Config(event_log_.get()));
>+    CreateSenderCall();
> 
>     test::NullTransport transport;
>     CreateSendConfig(1, 0, 0, &transport);
>     CreateVideoStreams();
>-    video_send_stream_->Stop();
>-    video_send_stream_->Stop();
>+    GetVideoSendStream()->Stop();
>+    GetVideoSendStream()->Stop();
>     DestroyStreams();
>     DestroyCalls();
>   });
>@@ -182,8 +203,10 @@ TEST_F(VideoSendStreamTest, SupportsTransmissionTimeOffset) {
>   class TransmissionTimeOffsetObserver : public test::SendTest {
>    public:
>     TransmissionTimeOffsetObserver()
>-        : SendTest(kDefaultTimeoutMs),
>-          encoder_(Clock::GetRealTimeClock(), kEncodeDelayMs) {
>+        : SendTest(kDefaultTimeoutMs), encoder_factory_([]() {
>+            return absl::make_unique<test::DelayedEncoder>(
>+                Clock::GetRealTimeClock(), kEncodeDelayMs);
>+          }) {
>       EXPECT_TRUE(parser_->RegisterRtpHeaderExtension(
>           kRtpExtensionTransmissionTimeOffset, test::kTOffsetExtensionId));
>     }
>@@ -206,7 +229,7 @@ TEST_F(VideoSendStreamTest, SupportsTransmissionTimeOffset) {
>         VideoSendStream::Config* send_config,
>         std::vector<VideoReceiveStream::Config>* receive_configs,
>         VideoEncoderConfig* encoder_config) override {
>-      send_config->encoder_settings.encoder = &encoder_;
>+      send_config->encoder_settings.encoder_factory = &encoder_factory_;
>       send_config->rtp.extensions.clear();
>       send_config->rtp.extensions.push_back(RtpExtension(
>           RtpExtension::kTimestampOffsetUri, test::kTOffsetExtensionId));
>@@ -216,7 +239,7 @@ TEST_F(VideoSendStreamTest, SupportsTransmissionTimeOffset) {
>       EXPECT_TRUE(Wait()) << "Timed out while waiting for a single RTP packet.";
>     }
> 
>-    test::DelayedEncoder encoder_;
>+    test::FunctionVideoEncoderFactory encoder_factory_;
>   } test;
> 
>   RunBaseTest(&test);
>@@ -227,7 +250,10 @@ TEST_F(VideoSendStreamTest, SupportsTransportWideSequenceNumbers) {
>   class TransportWideSequenceNumberObserver : public test::SendTest {
>    public:
>     TransportWideSequenceNumberObserver()
>-        : SendTest(kDefaultTimeoutMs), encoder_(Clock::GetRealTimeClock()) {
>+        : SendTest(kDefaultTimeoutMs), encoder_factory_([]() {
>+            return absl::make_unique<test::FakeEncoder>(
>+                Clock::GetRealTimeClock());
>+          }) {
>       EXPECT_TRUE(parser_->RegisterRtpHeaderExtension(
>           kRtpExtensionTransportSequenceNumber, kExtensionId));
>     }
>@@ -250,14 +276,14 @@ TEST_F(VideoSendStreamTest, SupportsTransportWideSequenceNumbers) {
>         VideoSendStream::Config* send_config,
>         std::vector<VideoReceiveStream::Config>* receive_configs,
>         VideoEncoderConfig* encoder_config) override {
>-      send_config->encoder_settings.encoder = &encoder_;
>+      send_config->encoder_settings.encoder_factory = &encoder_factory_;
>     }
> 
>     void PerformTest() override {
>       EXPECT_TRUE(Wait()) << "Timed out while waiting for a single RTP packet.";
>     }
> 
>-    test::FakeEncoder encoder_;
>+    test::FunctionVideoEncoderFactory encoder_factory_;
>   } test;
> 
>   RunBaseTest(&test);
>@@ -308,7 +334,8 @@ TEST_F(VideoSendStreamTest, SupportsVideoRotation) {
> TEST_F(VideoSendStreamTest, SupportsVideoContentType) {
>   class VideoContentTypeObserver : public test::SendTest {
>    public:
>-    VideoContentTypeObserver() : SendTest(kDefaultTimeoutMs) {
>+    VideoContentTypeObserver()
>+        : SendTest(kDefaultTimeoutMs), first_frame_sent_(false) {
>       EXPECT_TRUE(parser_->RegisterRtpHeaderExtension(
>           kRtpExtensionVideoContentType, test::kVideoContentTypeExtensionId));
>     }
>@@ -316,9 +343,11 @@ TEST_F(VideoSendStreamTest, SupportsVideoContentType) {
>     Action OnSendRtp(const uint8_t* packet, size_t length) override {
>       RTPHeader header;
>       EXPECT_TRUE(parser_->Parse(packet, length, &header));
>-      // Only the last packet of the frame must have extension.
>-      if (!header.markerBit)
>+      // Only the last packet of the key-frame must have extension.
>+      if (!header.markerBit || first_frame_sent_)
>         return SEND_PACKET;
>+      // First marker bit seen means that the first frame is sent.
>+      first_frame_sent_ = true;
>       EXPECT_TRUE(header.extension.hasVideoContentType);
>       EXPECT_TRUE(videocontenttypehelpers::IsScreenshare(
>           header.extension.videoContentType));
>@@ -340,6 +369,9 @@ TEST_F(VideoSendStreamTest, SupportsVideoContentType) {
>     void PerformTest() override {
>       EXPECT_TRUE(Wait()) << "Timed out while waiting for single RTP packet.";
>     }
>+
>+   private:
>+    bool first_frame_sent_;
>   } test;
> 
>   RunBaseTest(&test);
>@@ -409,9 +441,9 @@ class UlpfecObserver : public test::EndToEndTest {
>                  bool expect_red,
>                  bool expect_ulpfec,
>                  const std::string& codec,
>-                 VideoEncoder* encoder)
>+                 VideoEncoderFactory* encoder_factory)
>       : EndToEndTest(kTimeoutMs),
>-        encoder_(encoder),
>+        encoder_factory_(encoder_factory),
>         payload_name_(codec),
>         use_nack_(use_nack),
>         expect_red_(expect_red),
>@@ -509,8 +541,8 @@ class UlpfecObserver : public test::EndToEndTest {
>           (*receive_configs)[0].rtp.nack.rtp_history_ms =
>               VideoSendStreamTest::kNackRtpHistoryMs;
>     }
>-    send_config->encoder_settings.encoder = encoder_;
>-    send_config->encoder_settings.payload_name = payload_name_;
>+    send_config->encoder_settings.encoder_factory = encoder_factory_;
>+    send_config->rtp.payload_name = payload_name_;
>     send_config->rtp.ulpfec.red_payload_type =
>         VideoSendStreamTest::kRedPayloadType;
>     send_config->rtp.ulpfec.ulpfec_payload_type =
>@@ -522,6 +554,7 @@ class UlpfecObserver : public test::EndToEndTest {
>       send_config->rtp.extensions.push_back(RtpExtension(
>           RtpExtension::kAbsSendTimeUri, test::kAbsSendTimeExtensionId));
>     }
>+    encoder_config->codec_type = PayloadStringToCodecType(payload_name_);
>     (*receive_configs)[0].rtp.red_payload_type =
>         send_config->rtp.ulpfec.red_payload_type;
>     (*receive_configs)[0].rtp.ulpfec_payload_type =
>@@ -533,7 +566,7 @@ class UlpfecObserver : public test::EndToEndTest {
>         << "Timed out waiting for ULPFEC and/or media packets.";
>   }
> 
>-  VideoEncoder* const encoder_;
>+  VideoEncoderFactory* encoder_factory_;
>   std::string payload_name_;
>   const bool use_nack_;
>   const bool expect_red_;
>@@ -545,14 +578,16 @@ class UlpfecObserver : public test::EndToEndTest {
> };
> 
> TEST_F(VideoSendStreamTest, SupportsUlpfecWithExtensions) {
>-  std::unique_ptr<VideoEncoder> encoder(VP8Encoder::Create());
>-  UlpfecObserver test(true, false, true, true, "VP8", encoder.get());
>+  test::FunctionVideoEncoderFactory encoder_factory(
>+      []() { return VP8Encoder::Create(); });
>+  UlpfecObserver test(true, false, true, true, "VP8", &encoder_factory);
>   RunBaseTest(&test);
> }
> 
> TEST_F(VideoSendStreamTest, SupportsUlpfecWithoutExtensions) {
>-  std::unique_ptr<VideoEncoder> encoder(VP8Encoder::Create());
>-  UlpfecObserver test(false, false, true, true, "VP8", encoder.get());
>+  test::FunctionVideoEncoderFactory encoder_factory(
>+      []() { return VP8Encoder::Create(); });
>+  UlpfecObserver test(false, false, true, true, "VP8", &encoder_factory);
>   RunBaseTest(&test);
> }
> 
>@@ -565,8 +600,9 @@ class VideoSendStreamWithoutUlpfecTest : public VideoSendStreamTest {
> };
> 
> TEST_F(VideoSendStreamWithoutUlpfecTest, NoUlpfecIfDisabledThroughFieldTrial) {
>-  std::unique_ptr<VideoEncoder> encoder(VP8Encoder::Create());
>-  UlpfecObserver test(false, false, true, false, "VP8", encoder.get());
>+  test::FunctionVideoEncoderFactory encoder_factory(
>+      []() { return VP8Encoder::Create(); });
>+  UlpfecObserver test(false, false, false, false, "VP8", &encoder_factory);
>   RunBaseTest(&test);
> }
> 
>@@ -575,40 +611,46 @@ TEST_F(VideoSendStreamWithoutUlpfecTest, NoUlpfecIfDisabledThroughFieldTrial) {
> // bandwidth since the receiver has to wait for FEC retransmissions to determine
> // that the received state is actually decodable.
> TEST_F(VideoSendStreamTest, DoesNotUtilizeUlpfecForH264WithNackEnabled) {
>-  std::unique_ptr<VideoEncoder> encoder(
>-      new test::FakeH264Encoder(Clock::GetRealTimeClock()));
>-  UlpfecObserver test(false, true, true, false, "H264", encoder.get());
>+  test::FunctionVideoEncoderFactory encoder_factory([]() {
>+    return absl::make_unique<test::FakeH264Encoder>(Clock::GetRealTimeClock());
>+  });
>+  UlpfecObserver test(false, true, false, false, "H264", &encoder_factory);
>   RunBaseTest(&test);
> }
> 
> // Without retransmissions FEC for H264 is fine.
> TEST_F(VideoSendStreamTest, DoesUtilizeUlpfecForH264WithoutNackEnabled) {
>-  std::unique_ptr<VideoEncoder> encoder(
>-      new test::FakeH264Encoder(Clock::GetRealTimeClock()));
>-  UlpfecObserver test(false, false, true, true, "H264", encoder.get());
>+  test::FunctionVideoEncoderFactory encoder_factory([]() {
>+    return absl::make_unique<test::FakeH264Encoder>(Clock::GetRealTimeClock());
>+  });
>+  UlpfecObserver test(false, false, true, true, "H264", &encoder_factory);
>   RunBaseTest(&test);
> }
> 
> // Disabled as flaky, see https://crbug.com/webrtc/7285 for details.
> TEST_F(VideoSendStreamTest, DISABLED_DoesUtilizeUlpfecForVp8WithNackEnabled) {
>-  std::unique_ptr<VideoEncoder> encoder(VP8Encoder::Create());
>-  UlpfecObserver test(false, true, true, true, "VP8", encoder.get());
>+  test::FunctionVideoEncoderFactory encoder_factory(
>+      []() { return VP8Encoder::Create(); });
>+  UlpfecObserver test(false, true, true, true, "VP8", &encoder_factory);
>   RunBaseTest(&test);
> }
> 
> #if !defined(RTC_DISABLE_VP9)
> // Disabled as flaky, see https://crbug.com/webrtc/7285 for details.
> TEST_F(VideoSendStreamTest, DISABLED_DoesUtilizeUlpfecForVp9WithNackEnabled) {
>-  std::unique_ptr<VideoEncoder> encoder(VP9Encoder::Create());
>-  UlpfecObserver test(false, true, true, true, "VP9", encoder.get());
>+  test::FunctionVideoEncoderFactory encoder_factory(
>+      []() { return VP9Encoder::Create(); });
>+  UlpfecObserver test(false, true, true, true, "VP9", &encoder_factory);
>   RunBaseTest(&test);
> }
> #endif  // !defined(RTC_DISABLE_VP9)
> 
> TEST_F(VideoSendStreamTest, SupportsUlpfecWithMultithreadedH264) {
>-  std::unique_ptr<VideoEncoder> encoder(
>-      new test::MultithreadedFakeH264Encoder(Clock::GetRealTimeClock()));
>-  UlpfecObserver test(false, false, true, true, "H264", encoder.get());
>+  test::FunctionVideoEncoderFactory encoder_factory([]() {
>+    return absl::make_unique<test::MultithreadedFakeH264Encoder>(
>+        Clock::GetRealTimeClock());
>+  });
>+  UlpfecObserver test(false, false, true, true, "H264", &encoder_factory);
>   RunBaseTest(&test);
> }
> 
>@@ -619,16 +661,19 @@ class FlexfecObserver : public test::EndToEndTest {
>   FlexfecObserver(bool header_extensions_enabled,
>                   bool use_nack,
>                   const std::string& codec,
>-                  VideoEncoder* encoder)
>+                  VideoEncoderFactory* encoder_factory,
>+                  size_t num_video_streams)
>       : EndToEndTest(VideoSendStreamTest::kDefaultTimeoutMs),
>-        encoder_(encoder),
>+        encoder_factory_(encoder_factory),
>         payload_name_(codec),
>         use_nack_(use_nack),
>         sent_media_(false),
>         sent_flexfec_(false),
>-        header_extensions_enabled_(header_extensions_enabled) {}
>+        header_extensions_enabled_(header_extensions_enabled),
>+        num_video_streams_(num_video_streams) {}
> 
>   size_t GetNumFlexfecStreams() const override { return 1; }
>+  size_t GetNumVideoStreams() const override { return num_video_streams_; }
> 
>  private:
>   Action OnSendRtp(const uint8_t* packet, size_t length) override {
>@@ -641,7 +686,9 @@ class FlexfecObserver : public test::EndToEndTest {
>     } else {
>       EXPECT_EQ(VideoSendStreamTest::kFakeVideoSendPayloadType,
>                 header.payloadType);
>-      EXPECT_EQ(VideoSendStreamTest::kVideoSendSsrcs[0], header.ssrc);
>+      EXPECT_THAT(testing::make_tuple(VideoSendStreamTest::kVideoSendSsrcs,
>+                                      num_video_streams_),
>+                  testing::Contains(header.ssrc));
>       sent_media_ = true;
>     }
> 
>@@ -681,8 +728,8 @@ class FlexfecObserver : public test::EndToEndTest {
>           (*receive_configs)[0].rtp.nack.rtp_history_ms =
>               VideoSendStreamTest::kNackRtpHistoryMs;
>     }
>-    send_config->encoder_settings.encoder = encoder_;
>-    send_config->encoder_settings.payload_name = payload_name_;
>+    send_config->encoder_settings.encoder_factory = encoder_factory_;
>+    send_config->rtp.payload_name = payload_name_;
>     if (header_extensions_enabled_) {
>       send_config->rtp.extensions.push_back(RtpExtension(
>           RtpExtension::kAbsSendTimeUri, test::kAbsSendTimeExtensionId));
>@@ -691,6 +738,7 @@ class FlexfecObserver : public test::EndToEndTest {
>     } else {
>       send_config->rtp.extensions.clear();
>     }
>+    encoder_config->codec_type = PayloadStringToCodecType(payload_name_);
>   }
> 
>   void PerformTest() override {
>@@ -698,64 +746,82 @@ class FlexfecObserver : public test::EndToEndTest {
>         << "Timed out waiting for FlexFEC and/or media packets.";
>   }
> 
>-  VideoEncoder* const encoder_;
>+  VideoEncoderFactory* encoder_factory_;
>   std::string payload_name_;
>   const bool use_nack_;
>   bool sent_media_;
>   bool sent_flexfec_;
>-  bool header_extensions_enabled_;
>+  const bool header_extensions_enabled_;
>+  const size_t num_video_streams_;
> };
> 
> TEST_F(VideoSendStreamTest, SupportsFlexfecVp8) {
>-  std::unique_ptr<VideoEncoder> encoder(VP8Encoder::Create());
>-  FlexfecObserver test(false, false, "VP8", encoder.get());
>+  test::FunctionVideoEncoderFactory encoder_factory(
>+      []() { return VP8Encoder::Create(); });
>+  FlexfecObserver test(false, false, "VP8", &encoder_factory, 1);
>+  RunBaseTest(&test);
>+}
>+
>+TEST_F(VideoSendStreamTest, SupportsFlexfecSimulcastVp8) {
>+  test::FunctionVideoEncoderFactory encoder_factory(
>+      []() { return VP8Encoder::Create(); });
>+  FlexfecObserver test(false, false, "VP8", &encoder_factory, 2);
>   RunBaseTest(&test);
> }
> 
> TEST_F(VideoSendStreamTest, SupportsFlexfecWithNackVp8) {
>-  std::unique_ptr<VideoEncoder> encoder(VP8Encoder::Create());
>-  FlexfecObserver test(false, true, "VP8", encoder.get());
>+  test::FunctionVideoEncoderFactory encoder_factory(
>+      []() { return VP8Encoder::Create(); });
>+  FlexfecObserver test(false, true, "VP8", &encoder_factory, 1);
>   RunBaseTest(&test);
> }
> 
> TEST_F(VideoSendStreamTest, SupportsFlexfecWithRtpExtensionsVp8) {
>-  std::unique_ptr<VideoEncoder> encoder(VP8Encoder::Create());
>-  FlexfecObserver test(true, false, "VP8", encoder.get());
>+  test::FunctionVideoEncoderFactory encoder_factory(
>+      []() { return VP8Encoder::Create(); });
>+  FlexfecObserver test(true, false, "VP8", &encoder_factory, 1);
>   RunBaseTest(&test);
> }
> 
> #if !defined(RTC_DISABLE_VP9)
> TEST_F(VideoSendStreamTest, SupportsFlexfecVp9) {
>-  std::unique_ptr<VideoEncoder> encoder(VP9Encoder::Create());
>-  FlexfecObserver test(false, false, "VP9", encoder.get());
>+  test::FunctionVideoEncoderFactory encoder_factory(
>+      []() { return VP9Encoder::Create(); });
>+  FlexfecObserver test(false, false, "VP9", &encoder_factory, 1);
>   RunBaseTest(&test);
> }
> 
> TEST_F(VideoSendStreamTest, SupportsFlexfecWithNackVp9) {
>-  std::unique_ptr<VideoEncoder> encoder(VP9Encoder::Create());
>-  FlexfecObserver test(false, true, "VP9", encoder.get());
>+  test::FunctionVideoEncoderFactory encoder_factory(
>+      []() { return VP9Encoder::Create(); });
>+  FlexfecObserver test(false, true, "VP9", &encoder_factory, 1);
>   RunBaseTest(&test);
> }
> #endif  // defined(RTC_DISABLE_VP9)
> 
> TEST_F(VideoSendStreamTest, SupportsFlexfecH264) {
>-  std::unique_ptr<VideoEncoder> encoder(
>-      new test::FakeH264Encoder(Clock::GetRealTimeClock()));
>-  FlexfecObserver test(false, false, "H264", encoder.get());
>+  test::FunctionVideoEncoderFactory encoder_factory([]() {
>+    return absl::make_unique<test::FakeH264Encoder>(Clock::GetRealTimeClock());
>+  });
>+  FlexfecObserver test(false, false, "H264", &encoder_factory, 1);
>   RunBaseTest(&test);
> }
> 
> TEST_F(VideoSendStreamTest, SupportsFlexfecWithNackH264) {
>-  std::unique_ptr<VideoEncoder> encoder(
>-      new test::FakeH264Encoder(Clock::GetRealTimeClock()));
>-  FlexfecObserver test(false, true, "H264", encoder.get());
>+  test::FunctionVideoEncoderFactory encoder_factory([]() {
>+    return absl::make_unique<test::FakeH264Encoder>(Clock::GetRealTimeClock());
>+  });
>+  FlexfecObserver test(false, true, "H264", &encoder_factory, 1);
>   RunBaseTest(&test);
> }
> 
> TEST_F(VideoSendStreamTest, SupportsFlexfecWithMultithreadedH264) {
>-  std::unique_ptr<VideoEncoder> encoder(
>-      new test::MultithreadedFakeH264Encoder(Clock::GetRealTimeClock()));
>-  FlexfecObserver test(false, false, "H264", encoder.get());
>+  test::FunctionVideoEncoderFactory encoder_factory([]() {
>+    return absl::make_unique<test::MultithreadedFakeH264Encoder>(
>+        Clock::GetRealTimeClock());
>+  });
>+
>+  FlexfecObserver test(false, false, "H264", &encoder_factory, 1);
>   RunBaseTest(&test);
> }
> 
>@@ -768,31 +834,38 @@ void VideoSendStreamTest::TestNackRetransmission(
>                           uint8_t retransmit_payload_type)
>         : SendTest(kDefaultTimeoutMs),
>           send_count_(0),
>+          retransmit_count_(0),
>           retransmit_ssrc_(retransmit_ssrc),
>-          retransmit_payload_type_(retransmit_payload_type),
>-          nacked_sequence_number_(-1) {
>-    }
>+          retransmit_payload_type_(retransmit_payload_type) {}
> 
>    private:
>     Action OnSendRtp(const uint8_t* packet, size_t length) override {
>       RTPHeader header;
>       EXPECT_TRUE(parser_->Parse(packet, length, &header));
> 
>-      // Nack second packet after receiving the third one.
>-      if (++send_count_ == 3) {
>-        uint16_t nack_sequence_number = header.sequenceNumber - 1;
>-        nacked_sequence_number_ = nack_sequence_number;
>+      int kRetransmitTarget = 6;
>+      ++send_count_;
>+      if (send_count_ == 5 || send_count_ == 25) {
>+        nacked_sequence_numbers_.push_back(
>+            static_cast<uint16_t>(header.sequenceNumber - 3));
>+        nacked_sequence_numbers_.push_back(
>+            static_cast<uint16_t>(header.sequenceNumber - 2));
>+        nacked_sequence_numbers_.push_back(
>+            static_cast<uint16_t>(header.sequenceNumber - 1));
>+
>         RTCPSender rtcp_sender(false, Clock::GetRealTimeClock(), nullptr,
>-                               nullptr, nullptr, transport_adapter_.get());
>+                               nullptr, nullptr, transport_adapter_.get(),
>+                               RtcpIntervalConfig{});
> 
>         rtcp_sender.SetRTCPStatus(RtcpMode::kReducedSize);
>         rtcp_sender.SetRemoteSSRC(kVideoSendSsrcs[0]);
> 
>         RTCPSender::FeedbackState feedback_state;
> 
>-        EXPECT_EQ(0,
>-                  rtcp_sender.SendRTCP(
>-                      feedback_state, kRtcpNack, 1, &nack_sequence_number));
>+        EXPECT_EQ(0, rtcp_sender.SendRTCP(
>+                         feedback_state, kRtcpNack,
>+                         static_cast<int>(nacked_sequence_numbers_.size()),
>+                         &nacked_sequence_numbers_.front()));
>       }
> 
>       uint16_t sequence_number = header.sequenceNumber;
>@@ -804,11 +877,16 @@ void VideoSendStreamTest::TestNackRetransmission(
>         const uint8_t* rtx_header = packet + header.headerLength;
>         sequence_number = (rtx_header[0] << 8) + rtx_header[1];
>       }
>-
>-      if (sequence_number == nacked_sequence_number_) {
>-        EXPECT_EQ(retransmit_ssrc_, header.ssrc);
>-        EXPECT_EQ(retransmit_payload_type_, header.payloadType);
>-        observation_complete_.Set();
>+      auto found = std::find(nacked_sequence_numbers_.begin(),
>+                             nacked_sequence_numbers_.end(), sequence_number);
>+      if (found != nacked_sequence_numbers_.end()) {
>+        nacked_sequence_numbers_.erase(found);
>+
>+        if (++retransmit_count_ == kRetransmitTarget) {
>+          EXPECT_EQ(retransmit_ssrc_, header.ssrc);
>+          EXPECT_EQ(retransmit_payload_type_, header.payloadType);
>+          observation_complete_.Set();
>+        }
>       }
> 
>       return SEND_PACKET;
>@@ -833,9 +911,10 @@ void VideoSendStreamTest::TestNackRetransmission(
> 
>     std::unique_ptr<internal::TransportAdapter> transport_adapter_;
>     int send_count_;
>+    int retransmit_count_;
>     uint32_t retransmit_ssrc_;
>     uint8_t retransmit_payload_type_;
>-    int nacked_sequence_number_;
>+    std::vector<uint16_t> nacked_sequence_numbers_;
>   } test(retransmit_ssrc, retransmit_payload_type);
> 
>   RunBaseTest(&test);
>@@ -872,11 +951,15 @@ void VideoSendStreamTest::TestPacketFragmentationSize(VideoFormat format,
>                            bool use_fec)
>         : SendTest(kLongTimeoutMs),
>           encoder_(stop),
>+          encoder_factory_(&encoder_),
>           max_packet_size_(max_packet_size),
>           stop_size_(stop_size),
>           test_generic_packetization_(test_generic_packetization),
>           use_fec_(use_fec),
>           packet_count_(0),
>+          packets_lost_(0),
>+          last_packet_count_(0),
>+          last_packets_lost_(0),
>           accumulated_size_(0),
>           accumulated_payload_(0),
>           fec_packet_received_(false),
>@@ -968,14 +1051,21 @@ void VideoSendStreamTest::TestPacketFragmentationSize(VideoFormat format,
>     void TriggerLossReport(const RTPHeader& header) {
>       // Send lossy receive reports to trigger FEC enabling.
>       const int kLossPercent = 5;
>-      if (packet_count_++ % (100 / kLossPercent) != 0) {
>+      if (++packet_count_ % (100 / kLossPercent) == 0) {
>+        packets_lost_++;
>+        int loss_delta = packets_lost_ - last_packets_lost_;
>+        int packets_delta = packet_count_ - last_packet_count_;
>+        last_packet_count_ = packet_count_;
>+        last_packets_lost_ = packets_lost_;
>+        uint8_t loss_ratio =
>+            static_cast<uint8_t>(loss_delta * 255 / packets_delta);
>         FakeReceiveStatistics lossy_receive_stats(
>             kVideoSendSsrcs[0], header.sequenceNumber,
>-            (packet_count_ * (100 - kLossPercent)) / 100,  // Cumulative lost.
>-            static_cast<uint8_t>((255 * kLossPercent) / 100));  // Loss percent.
>+            packets_lost_,  // Cumulative lost.
>+            loss_ratio);    // Loss percent.
>         RTCPSender rtcp_sender(false, Clock::GetRealTimeClock(),
>                                &lossy_receive_stats, nullptr, nullptr,
>-                               transport_adapter_.get());
>+                               transport_adapter_.get(), RtcpIntervalConfig{});
> 
>         rtcp_sender.SetRTCPStatus(RtcpMode::kReducedSize);
>         rtcp_sender.SetRemoteSSRC(kVideoSendSsrcs[0]);
>@@ -995,12 +1085,9 @@ void VideoSendStreamTest::TestPacketFragmentationSize(VideoFormat format,
>       }
>       encoder_.SetFrameSize(static_cast<size_t>(current_size_frame_));
>     }
>-
>-    Call::Config GetSenderCallConfig() override {
>-      Call::Config config(event_log_.get());
>+    void ModifySenderCallConfig(Call::Config* config) override {
>       const int kMinBitrateBps = 30000;
>-      config.bitrate_config.min_bitrate_bps = kMinBitrateBps;
>-      return config;
>+      config->bitrate_config.min_bitrate_bps = kMinBitrateBps;
>     }
> 
>     void ModifyVideoConfigs(
>@@ -1016,9 +1103,8 @@ void VideoSendStreamTest::TestPacketFragmentationSize(VideoFormat format,
>       }
> 
>       if (!test_generic_packetization_)
>-        send_config->encoder_settings.payload_name = "VP8";
>-
>-      send_config->encoder_settings.encoder = &encoder_;
>+        send_config->rtp.payload_name = "VP8";
>+      send_config->encoder_settings.encoder_factory = &encoder_factory_;
>       send_config->rtp.max_packet_size = kMaxPacketSize;
>       send_config->post_encode_callback = this;
> 
>@@ -1043,8 +1129,7 @@ void VideoSendStreamTest::TestPacketFragmentationSize(VideoFormat format,
>           std::vector<VideoStream> streams =
>               test::CreateVideoStreams(width, height, encoder_config);
>           for (VideoStream& stream : streams) {
>-            stream.temporal_layer_thresholds_bps.resize(num_temporal_layers_ -
>-                                                        1);
>+            stream.num_temporal_layers = num_temporal_layers_;
>           }
>           return streams;
>         }
>@@ -1062,6 +1147,7 @@ void VideoSendStreamTest::TestPacketFragmentationSize(VideoFormat format,
> 
>     std::unique_ptr<internal::TransportAdapter> transport_adapter_;
>     test::ConfigurableFrameSizeEncoder encoder_;
>+    test::EncoderProxyFactory encoder_factory_;
> 
>     const size_t max_packet_size_;
>     const size_t stop_size_;
>@@ -1069,6 +1155,9 @@ void VideoSendStreamTest::TestPacketFragmentationSize(VideoFormat format,
>     const bool use_fec_;
> 
>     uint32_t packet_count_;
>+    uint32_t packets_lost_;
>+    uint32_t last_packet_count_;
>+    uint32_t last_packets_lost_;
>     size_t accumulated_size_;
>     size_t accumulated_payload_;
>     bool fec_packet_received_;
>@@ -1080,8 +1169,8 @@ void VideoSendStreamTest::TestPacketFragmentationSize(VideoFormat format,
> 
>   // Don't auto increment if FEC is used; continue sending frame size until
>   // a FEC packet has been received.
>-  FrameFragmentationTest test(
>-      kMaxPacketSize, start, stop, format == kGeneric, with_fec);
>+  FrameFragmentationTest test(kMaxPacketSize, start, stop, format == kGeneric,
>+                              with_fec);
> 
>   RunBaseTest(&test);
> }
>@@ -1232,7 +1321,7 @@ TEST_F(VideoSendStreamTest, SuspendBelowMinBitrate) {
>       FakeReceiveStatistics receive_stats(kVideoSendSsrcs[0],
>                                           last_sequence_number_, rtp_count_, 0);
>       RTCPSender rtcp_sender(false, clock_, &receive_stats, nullptr, nullptr,
>-                             transport_adapter_.get());
>+                             transport_adapter_.get(), RtcpIntervalConfig{});
> 
>       rtcp_sender.SetRTCPStatus(RtcpMode::kReducedSize);
>       rtcp_sender.SetRemoteSSRC(kVideoSendSsrcs[0]);
>@@ -1268,8 +1357,7 @@ TEST_F(VideoSendStreamTest, NoPaddingWhenVideoIsMuted) {
>         : SendTest(kDefaultTimeoutMs),
>           clock_(Clock::GetRealTimeClock()),
>           last_packet_time_ms_(-1),
>-          capturer_(nullptr) {
>-    }
>+          capturer_(nullptr) {}
> 
>    private:
>     Action OnSendRtp(const uint8_t* packet, size_t length) override {
>@@ -1441,12 +1529,9 @@ TEST_F(VideoSendStreamTest, MinTransmitBitrateRespectsRemb) {
>         EXPECT_EQ(1u, stats.substreams.size());
>         int total_bitrate_bps =
>             stats.substreams.begin()->second.total_bitrate_bps;
>-        test::PrintResult("bitrate_stats_",
>-                          "min_transmit_bitrate_low_remb",
>-                          "bitrate_bps",
>-                          static_cast<size_t>(total_bitrate_bps),
>-                          "bps",
>-                          false);
>+        test::PrintResult("bitrate_stats_", "min_transmit_bitrate_low_remb",
>+                          "bitrate_bps", static_cast<size_t>(total_bitrate_bps),
>+                          "bps", false);
>         if (total_bitrate_bps > kHighBitrateBps) {
>           rtp_rtcp_->SetRemb(kRembBitrateBps,
>                              std::vector<uint32_t>(1, header.ssrc));
>@@ -1547,13 +1632,18 @@ TEST_F(VideoSendStreamTest, ChangingNetworkRoute) {
>     }
> 
>     void PerformTest() override {
>-      rtc::NetworkRoute new_route(true, 10, 20, -1);
>-      Call::Config::BitrateConfig bitrate_config;
>+      rtc::NetworkRoute new_route;
>+      new_route.connected = true;
>+      new_route.local_network_id = 10;
>+      new_route.remote_network_id = 20;
>+      BitrateConstraints bitrate_config;
> 
>       task_queue_->SendTask([this, &new_route, &bitrate_config]() {
>-        call_->OnNetworkRouteChanged("transport", new_route);
>+        call_->GetTransportControllerSend()->OnNetworkRouteChanged("transport",
>+                                                                   new_route);
>         bitrate_config.start_bitrate_bps = kStartBitrateBps;
>-        call_->SetBitrateConfig(bitrate_config);
>+        call_->GetTransportControllerSend()->SetSdpBitrateParameters(
>+            bitrate_config);
>       });
> 
>       EXPECT_TRUE(Wait())
>@@ -1562,11 +1652,13 @@ TEST_F(VideoSendStreamTest, ChangingNetworkRoute) {
>       task_queue_->SendTask([this, &new_route, &bitrate_config]() {
>         bitrate_config.start_bitrate_bps = -1;
>         bitrate_config.max_bitrate_bps = kNewMaxBitrateBps;
>-        call_->SetBitrateConfig(bitrate_config);
>+        call_->GetTransportControllerSend()->SetSdpBitrateParameters(
>+            bitrate_config);
>         // TODO(holmer): We should set the last sent packet id here and verify
>         // that we correctly ignore any packet loss reported prior to that id.
>         ++new_route.local_network_id;
>-        call_->OnNetworkRouteChanged("transport", new_route);
>+        call_->GetTransportControllerSend()->OnNetworkRouteChanged("transport",
>+                                                                   new_route);
>         EXPECT_GE(call_->GetStats().send_bandwidth_bps, kStartBitrateBps);
>       });
>     }
>@@ -1766,14 +1858,11 @@ TEST_F(VideoSendStreamTest, RespectsMinTransmitBitrateAfterContentSwitch) {
>                           const VideoEncoderConfig& encoder_config) {
>     task_queue_.SendTask([this, &send_stream_config, &encoder_config]() {
>       Stop();
>-      sender_call_->DestroyVideoSendStream(video_send_stream_);
>-      video_send_config_ = send_stream_config.Copy();
>-      video_encoder_config_ = encoder_config.Copy();
>-      video_send_stream_ = sender_call_->CreateVideoSendStream(
>-          video_send_config_.Copy(), video_encoder_config_.Copy());
>-      video_send_stream_->SetSource(
>-          frame_generator_capturer_.get(),
>-          VideoSendStream::DegradationPreference::kMaintainResolution);
>+      DestroyVideoSendStreams();
>+      SetVideoSendConfig(send_stream_config);
>+      SetVideoEncoderConfig(encoder_config);
>+      CreateVideoSendStreams();
>+      SetVideoDegradation(DegradationPreference::MAINTAIN_RESOLUTION);
>       Start();
>     });
>   };
>@@ -1843,11 +1932,12 @@ TEST_F(VideoSendStreamTest,
> 
>   test::NullTransport transport;
>   EncoderObserver encoder;
>+  test::EncoderProxyFactory encoder_factory(&encoder);
> 
>-  task_queue_.SendTask([this, &transport, &encoder]() {
>-    CreateSenderCall(Call::Config(event_log_.get()));
>+  task_queue_.SendTask([this, &transport, &encoder_factory]() {
>+    CreateSenderCall();
>     CreateSendConfig(1, 0, 0, &transport);
>-    video_send_config_.encoder_settings.encoder = &encoder;
>+    GetVideoSendConfig()->encoder_settings.encoder_factory = &encoder_factory;
>     CreateVideoStreams();
>     CreateFrameGeneratorCapturer(kDefaultFramerate, kDefaultWidth,
>                                  kDefaultHeight);
>@@ -1908,29 +1998,34 @@ TEST_F(VideoSendStreamTest, CanReconfigureToUseStartBitrateAbovePreviousMax) {
>     int start_bitrate_kbps_ RTC_GUARDED_BY(crit_);
>   };
> 
>-  CreateSenderCall(Call::Config(event_log_.get()));
>+  CreateSenderCall();
> 
>   test::NullTransport transport;
>   CreateSendConfig(1, 0, 0, &transport);
> 
>-  Call::Config::BitrateConfig bitrate_config;
>-  bitrate_config.start_bitrate_bps = 2 * video_encoder_config_.max_bitrate_bps;
>-  sender_call_->SetBitrateConfig(bitrate_config);
>+  BitrateConstraints bitrate_config;
>+  bitrate_config.start_bitrate_bps =
>+      2 * GetVideoEncoderConfig()->max_bitrate_bps;
>+  sender_call_->GetTransportControllerSend()->SetSdpBitrateParameters(
>+      bitrate_config);
> 
>   StartBitrateObserver encoder;
>-  video_send_config_.encoder_settings.encoder = &encoder;
>+  test::EncoderProxyFactory encoder_factory(&encoder);
>   // Since this test does not use a capturer, set |internal_source| = true.
>   // Encoder configuration is otherwise updated on the next video frame.
>-  video_send_config_.encoder_settings.internal_source = true;
>+  encoder_factory.SetHasInternalSource(true);
>+  GetVideoSendConfig()->encoder_settings.encoder_factory = &encoder_factory;
> 
>   CreateVideoStreams();
> 
>   EXPECT_TRUE(encoder.WaitForStartBitrate());
>-  EXPECT_EQ(video_encoder_config_.max_bitrate_bps / 1000,
>+  EXPECT_EQ(GetVideoEncoderConfig()->max_bitrate_bps / 1000,
>             encoder.GetStartBitrateKbps());
> 
>-  video_encoder_config_.max_bitrate_bps = 2 * bitrate_config.start_bitrate_bps;
>-  video_send_stream_->ReconfigureVideoEncoder(video_encoder_config_.Copy());
>+  GetVideoEncoderConfig()->max_bitrate_bps =
>+      2 * bitrate_config.start_bitrate_bps;
>+  GetVideoSendStream()->ReconfigureVideoEncoder(
>+      GetVideoEncoderConfig()->Copy());
> 
>   // New bitrate should be reconfigured above the previous max. As there's no
>   // network connection this shouldn't be flaky, as no bitrate should've been
>@@ -1942,100 +2037,167 @@ TEST_F(VideoSendStreamTest, CanReconfigureToUseStartBitrateAbovePreviousMax) {
>   DestroyStreams();
> }
> 
>+class StartStopBitrateObserver : public test::FakeEncoder {
>+ public:
>+  StartStopBitrateObserver()
>+      : FakeEncoder(Clock::GetRealTimeClock()),
>+        encoder_init_(false, false),
>+        bitrate_changed_(false, false) {}
>+  int32_t InitEncode(const VideoCodec* config,
>+                     int32_t number_of_cores,
>+                     size_t max_payload_size) override {
>+    rtc::CritScope lock(&crit_);
>+    encoder_init_.Set();
>+    return FakeEncoder::InitEncode(config, number_of_cores, max_payload_size);
>+  }
>+
>+  int32_t SetRateAllocation(const VideoBitrateAllocation& bitrate,
>+                            uint32_t framerate) override {
>+    rtc::CritScope lock(&crit_);
>+    bitrate_kbps_ = bitrate.get_sum_kbps();
>+    bitrate_changed_.Set();
>+    return FakeEncoder::SetRateAllocation(bitrate, framerate);
>+  }
>+
>+  bool WaitForEncoderInit() {
>+    return encoder_init_.Wait(VideoSendStreamTest::kDefaultTimeoutMs);
>+  }
>+
>+  bool WaitBitrateChanged(bool non_zero) {
>+    do {
>+      absl::optional<int> bitrate_kbps;
>+      {
>+        rtc::CritScope lock(&crit_);
>+        bitrate_kbps = bitrate_kbps_;
>+      }
>+      if (!bitrate_kbps)
>+        continue;
>+
>+      if ((non_zero && *bitrate_kbps > 0) ||
>+          (!non_zero && *bitrate_kbps == 0)) {
>+        return true;
>+      }
>+    } while (bitrate_changed_.Wait(VideoSendStreamTest::kDefaultTimeoutMs));
>+    return false;
>+  }
>+
>+ private:
>+  rtc::CriticalSection crit_;
>+  rtc::Event encoder_init_;
>+  rtc::Event bitrate_changed_;
>+  absl::optional<int> bitrate_kbps_ RTC_GUARDED_BY(crit_);
>+};
>+
> // This test that if the encoder use an internal source, VideoEncoder::SetRates
> // will be called with zero bitrate during initialization and that
> // VideoSendStream::Stop also triggers VideoEncoder::SetRates Start to be called
> // with zero bitrate.
> TEST_F(VideoSendStreamTest, VideoSendStreamStopSetEncoderRateToZero) {
>-  class StartStopBitrateObserver : public test::FakeEncoder {
>-   public:
>-    StartStopBitrateObserver()
>-        : FakeEncoder(Clock::GetRealTimeClock()),
>-          encoder_init_(false, false),
>-          bitrate_changed_(false, false) {}
>-    int32_t InitEncode(const VideoCodec* config,
>-                       int32_t number_of_cores,
>-                       size_t max_payload_size) override {
>-      rtc::CritScope lock(&crit_);
>-      encoder_init_.Set();
>-      return FakeEncoder::InitEncode(config, number_of_cores, max_payload_size);
>-    }
>+  test::NullTransport transport;
>+  StartStopBitrateObserver encoder;
>+  test::EncoderProxyFactory encoder_factory(&encoder);
>+  encoder_factory.SetHasInternalSource(true);
>+  test::FrameForwarder forwarder;
> 
>-    int32_t SetRateAllocation(const BitrateAllocation& bitrate,
>-                              uint32_t framerate) override {
>-      rtc::CritScope lock(&crit_);
>-      bitrate_kbps_ = bitrate.get_sum_kbps();
>-      bitrate_changed_.Set();
>-      return FakeEncoder::SetRateAllocation(bitrate, framerate);
>-    }
>+  task_queue_.SendTask([this, &transport, &encoder_factory, &forwarder]() {
>+    CreateSenderCall();
>+    CreateSendConfig(1, 0, 0, &transport);
> 
>-    bool WaitForEncoderInit() {
>-      return encoder_init_.Wait(VideoSendStreamTest::kDefaultTimeoutMs);
>-    }
>+    sender_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
>+    GetVideoSendConfig()->encoder_settings.encoder_factory = &encoder_factory;
> 
>-    bool WaitBitrateChanged(bool non_zero) {
>-      do {
>-        rtc::Optional<int> bitrate_kbps;
>-        {
>-          rtc::CritScope lock(&crit_);
>-          bitrate_kbps = bitrate_kbps_;
>-        }
>-        if (!bitrate_kbps)
>-          continue;
>+    CreateVideoStreams();
>+    // Inject a frame, to force encoder creation.
>+    GetVideoSendStream()->Start();
>+    GetVideoSendStream()->SetSource(&forwarder,
>+                                    DegradationPreference::DISABLED);
>+    forwarder.IncomingCapturedFrame(CreateVideoFrame(640, 480, 4));
>+  });
> 
>-        if ((non_zero && *bitrate_kbps > 0) ||
>-            (!non_zero && *bitrate_kbps == 0)) {
>-          return true;
>-        }
>-      } while (bitrate_changed_.Wait(VideoSendStreamTest::kDefaultTimeoutMs));
>-      return false;
>-    }
>+  EXPECT_TRUE(encoder.WaitForEncoderInit());
> 
>-   private:
>-    rtc::CriticalSection crit_;
>-    rtc::Event encoder_init_;
>-    rtc::Event bitrate_changed_;
>-    rtc::Optional<int> bitrate_kbps_ RTC_GUARDED_BY(crit_);
>-  };
>+  task_queue_.SendTask([this]() { GetVideoSendStream()->Start(); });
>+  EXPECT_TRUE(encoder.WaitBitrateChanged(true));
>+
>+  task_queue_.SendTask([this]() { GetVideoSendStream()->Stop(); });
>+  EXPECT_TRUE(encoder.WaitBitrateChanged(false));
>+
>+  task_queue_.SendTask([this]() { GetVideoSendStream()->Start(); });
>+  EXPECT_TRUE(encoder.WaitBitrateChanged(true));
> 
>+  task_queue_.SendTask([this]() {
>+    DestroyStreams();
>+    DestroyCalls();
>+  });
>+}
>+
>+// Tests that when the encoder uses an internal source, the VideoEncoder will
>+// be updated with a new bitrate when turning the VideoSendStream on/off with
>+// VideoSendStream::UpdateActiveSimulcastLayers, and when the VideoStreamEncoder
>+// is reconfigured with new active layers.
>+TEST_F(VideoSendStreamTest, VideoSendStreamUpdateActiveSimulcastLayers) {
>   test::NullTransport transport;
>   StartStopBitrateObserver encoder;
>+  test::EncoderProxyFactory encoder_factory(&encoder);
>+  encoder_factory.SetHasInternalSource(true);
>+  test::FrameForwarder forwarder;
> 
>-  task_queue_.SendTask([this, &transport, &encoder]() {
>-    CreateSenderCall(Call::Config(event_log_.get()));
>-    CreateSendConfig(1, 0, 0, &transport);
>+  task_queue_.SendTask([this, &transport, &encoder_factory, &forwarder]() {
>+    CreateSenderCall();
>+    // Create two simulcast streams.
>+    CreateSendConfig(2, 0, 0, &transport);
> 
>     sender_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
>+    GetVideoSendConfig()->encoder_settings.encoder_factory = &encoder_factory;
> 
>-    video_send_config_.encoder_settings.encoder = &encoder;
>-    video_send_config_.encoder_settings.internal_source = true;
>+    GetVideoSendConfig()->rtp.payload_name = "VP8";
> 
>     CreateVideoStreams();
>+
>+    // Inject a frame, to force encoder creation.
>+    GetVideoSendStream()->Start();
>+    GetVideoSendStream()->SetSource(&forwarder,
>+                                    DegradationPreference::DISABLED);
>+    forwarder.IncomingCapturedFrame(CreateVideoFrame(640, 480, 4));
>   });
> 
>   EXPECT_TRUE(encoder.WaitForEncoderInit());
> 
>+  // When we turn on the simulcast layers it will update the BitrateAllocator,
>+  // which in turn updates the VideoEncoder's bitrate.
>   task_queue_.SendTask([this]() {
>-    video_send_stream_->Start();
>+    GetVideoSendStream()->UpdateActiveSimulcastLayers({true, true});
>   });
>   EXPECT_TRUE(encoder.WaitBitrateChanged(true));
> 
>+  GetVideoEncoderConfig()->simulcast_layers[0].active = true;
>+  GetVideoEncoderConfig()->simulcast_layers[1].active = false;
>   task_queue_.SendTask([this]() {
>-    video_send_stream_->Stop();
>+    GetVideoSendStream()->ReconfigureVideoEncoder(
>+        GetVideoEncoderConfig()->Copy());
>   });
>-  EXPECT_TRUE(encoder.WaitBitrateChanged(false));
>+  // TODO(bugs.webrtc.org/8807): Currently we require a hard reconfiguration to
>+  // update the VideoBitrateAllocator and BitrateAllocator of which layers are
>+  // active. Once the change is made for a "soft" reconfiguration we can remove
>+  // the expecation for an encoder init. We can also test that bitrate changes
>+  // when just updating individual active layers, which should change the
>+  // bitrate set to the video encoder.
>+  EXPECT_TRUE(encoder.WaitForEncoderInit());
>+  EXPECT_TRUE(encoder.WaitBitrateChanged(true));
> 
>+  // Turning off both simulcast layers should trigger a bitrate change of 0.
>+  GetVideoEncoderConfig()->simulcast_layers[0].active = false;
>+  GetVideoEncoderConfig()->simulcast_layers[1].active = false;
>   task_queue_.SendTask([this]() {
>-    video_send_stream_->Start();
>+    GetVideoSendStream()->UpdateActiveSimulcastLayers({false, false});
>   });
>-  EXPECT_TRUE(encoder.WaitBitrateChanged(true));
>+  EXPECT_TRUE(encoder.WaitBitrateChanged(false));
> 
>   task_queue_.SendTask([this]() {
>     DestroyStreams();
>     DestroyCalls();
>   });
> }
>-
> TEST_F(VideoSendStreamTest, CapturesTextureAndVideoFrames) {
>   class FrameObserver : public rtc::VideoSinkInterface<VideoFrame> {
>    public:
>@@ -2070,10 +2232,10 @@ TEST_F(VideoSendStreamTest, CapturesTextureAndVideoFrames) {
> 
>   task_queue_.SendTask([this, &transport, &observer, &input_frames]() {
>     // Initialize send stream.
>-    CreateSenderCall(Call::Config(event_log_.get()));
>+    CreateSenderCall();
> 
>     CreateSendConfig(1, 0, 0, &transport);
>-    video_send_config_.pre_encode_callback = &observer;
>+    GetVideoSendConfig()->pre_encode_callback = &observer;
>     CreateVideoStreams();
> 
>     // Prepare five input frames. Send ordinary VideoFrame and texture frames
>@@ -2090,19 +2252,19 @@ TEST_F(VideoSendStreamTest, CapturesTextureAndVideoFrames) {
>     input_frames.push_back(test::FakeNativeBuffer::CreateFrame(
>         width, height, 5, 5, kVideoRotation_0));
> 
>-    video_send_stream_->Start();
>+    GetVideoSendStream()->Start();
>     test::FrameForwarder forwarder;
>-    video_send_stream_->SetSource(
>-        &forwarder, VideoSendStream::DegradationPreference::kMaintainFramerate);
>+    GetVideoSendStream()->SetSource(&forwarder,
>+                                    DegradationPreference::MAINTAIN_FRAMERATE);
>     for (size_t i = 0; i < input_frames.size(); i++) {
>       forwarder.IncomingCapturedFrame(input_frames[i]);
>       // Wait until the output frame is received before sending the next input
>       // frame. Or the previous input frame may be replaced without delivering.
>       observer.WaitOutputFrame();
>     }
>-    video_send_stream_->Stop();
>-    video_send_stream_->SetSource(
>-        nullptr, VideoSendStream::DegradationPreference::kMaintainFramerate);
>+    GetVideoSendStream()->Stop();
>+    GetVideoSendStream()->SetSource(nullptr,
>+                                    DegradationPreference::MAINTAIN_FRAMERATE);
>   });
> 
>   // Test if the input and output frames are the same. render_time_ms and
>@@ -2146,7 +2308,8 @@ TEST_F(VideoSendStreamTest, EncoderIsProperlyInitializedAndDestroyed) {
>           initialized_(false),
>           callback_registered_(false),
>           num_releases_(0),
>-          released_(false) {}
>+          released_(false),
>+          encoder_factory_(this) {}
> 
>     bool IsReleased() {
>       rtc::CritScope lock(&crit_);
>@@ -2222,7 +2385,7 @@ TEST_F(VideoSendStreamTest, EncoderIsProperlyInitializedAndDestroyed) {
>         VideoSendStream::Config* send_config,
>         std::vector<VideoReceiveStream::Config>* receive_configs,
>         VideoEncoderConfig* encoder_config) override {
>-      send_config->encoder_settings.encoder = this;
>+      send_config->encoder_settings.encoder_factory = &encoder_factory_;
>       encoder_config_ = encoder_config->Copy();
>     }
> 
>@@ -2251,6 +2414,7 @@ TEST_F(VideoSendStreamTest, EncoderIsProperlyInitializedAndDestroyed) {
>     bool callback_registered_ RTC_GUARDED_BY(crit_);
>     size_t num_releases_ RTC_GUARDED_BY(crit_);
>     bool released_ RTC_GUARDED_BY(crit_);
>+    test::EncoderProxyFactory encoder_factory_;
>     VideoEncoderConfig encoder_config_;
>   } test_encoder(&task_queue_);
> 
>@@ -2269,14 +2433,15 @@ TEST_F(VideoSendStreamTest, EncoderSetupPropagatesCommonEncoderConfigValues) {
>           FakeEncoder(Clock::GetRealTimeClock()),
>           init_encode_event_(false, false),
>           num_initializations_(0),
>-          stream_(nullptr) {}
>+          stream_(nullptr),
>+          encoder_factory_(this) {}
> 
>    private:
>     void ModifyVideoConfigs(
>         VideoSendStream::Config* send_config,
>         std::vector<VideoReceiveStream::Config>* receive_configs,
>         VideoEncoderConfig* encoder_config) override {
>-      send_config->encoder_settings.encoder = this;
>+      send_config->encoder_settings.encoder_factory = &encoder_factory_;
>       encoder_config->max_bitrate_bps = kFirstMaxBitrateBps;
>       encoder_config_ = encoder_config->Copy();
>     }
>@@ -2320,13 +2485,14 @@ TEST_F(VideoSendStreamTest, EncoderSetupPropagatesCommonEncoderConfigValues) {
>     rtc::Event init_encode_event_;
>     size_t num_initializations_;
>     VideoSendStream* stream_;
>+    test::EncoderProxyFactory encoder_factory_;
>     VideoEncoderConfig encoder_config_;
>   } test;
> 
>   RunBaseTest(&test);
> }
> 
>-static const size_t kVideoCodecConfigObserverNumberOfTemporalLayers = 4;
>+static const size_t kVideoCodecConfigObserverNumberOfTemporalLayers = 3;
> template <typename T>
> class VideoCodecConfigObserver : public test::SendTest,
>                                  public test::FakeEncoder {
>@@ -2339,8 +2505,9 @@ class VideoCodecConfigObserver : public test::SendTest,
>         codec_name_(codec_name),
>         init_encode_event_(false, false),
>         num_initializations_(0),
>-        stream_(nullptr) {
>-    memset(&encoder_settings_, 0, sizeof(encoder_settings_));
>+        stream_(nullptr),
>+        encoder_factory_(this) {
>+    InitCodecSpecifics();
>   }
> 
>  private:
>@@ -2357,8 +2524,8 @@ class VideoCodecConfigObserver : public test::SendTest,
>       std::vector<VideoStream> streams =
>           test::CreateVideoStreams(width, height, encoder_config);
>       for (size_t i = 0; i < streams.size(); ++i) {
>-        streams[i].temporal_layer_thresholds_bps.resize(
>-            kVideoCodecConfigObserverNumberOfTemporalLayers - 1);
>+        streams[i].num_temporal_layers =
>+            kVideoCodecConfigObserverNumberOfTemporalLayers;
>       }
>       return streams;
>     }
>@@ -2368,9 +2535,10 @@ class VideoCodecConfigObserver : public test::SendTest,
>       VideoSendStream::Config* send_config,
>       std::vector<VideoReceiveStream::Config>* receive_configs,
>       VideoEncoderConfig* encoder_config) override {
>-    send_config->encoder_settings.encoder = this;
>-    send_config->encoder_settings.payload_name = codec_name_;
>+    send_config->encoder_settings.encoder_factory = &encoder_factory_;
>+    send_config->rtp.payload_name = codec_name_;
> 
>+    encoder_config->codec_type = video_codec_type_;
>     encoder_config->encoder_specific_settings = GetEncoderSpecificSettings();
>     encoder_config->video_stream_factory =
>         new rtc::RefCountedObject<VideoStreamFactory>();
>@@ -2393,6 +2561,7 @@ class VideoCodecConfigObserver : public test::SendTest,
>     return FakeEncoder::InitEncode(config, number_of_cores, max_payload_size);
>   }
> 
>+  void InitCodecSpecifics();
>   void VerifyCodecSpecifics(const VideoCodec& config) const;
>   rtc::scoped_refptr<VideoEncoderConfig::EncoderSpecificSettings>
>   GetEncoderSpecificSettings() const;
>@@ -2402,7 +2571,8 @@ class VideoCodecConfigObserver : public test::SendTest,
>         init_encode_event_.Wait(VideoSendStreamTest::kDefaultTimeoutMs));
>     ASSERT_EQ(1u, num_initializations_) << "VideoEncoder not initialized.";
> 
>-    encoder_settings_.frameDroppingOn = true;
>+    // Change encoder settings to actually trigger reconfiguration.
>+    encoder_settings_.frameDroppingOn = !encoder_settings_.frameDroppingOn;
>     encoder_config_.encoder_specific_settings = GetEncoderSpecificSettings();
>     stream_->ReconfigureVideoEncoder(std::move(encoder_config_));
>     ASSERT_TRUE(
>@@ -2425,9 +2595,15 @@ class VideoCodecConfigObserver : public test::SendTest,
>   rtc::Event init_encode_event_;
>   size_t num_initializations_;
>   VideoSendStream* stream_;
>+  test::EncoderProxyFactory encoder_factory_;
>   VideoEncoderConfig encoder_config_;
> };
> 
>+template <>
>+void VideoCodecConfigObserver<VideoCodecH264>::InitCodecSpecifics() {
>+  encoder_settings_ = VideoEncoder::GetDefaultH264Settings();
>+}
>+
> template <>
> void VideoCodecConfigObserver<VideoCodecH264>::VerifyCodecSpecifics(
>     const VideoCodec& config) const {
>@@ -2442,6 +2618,11 @@ VideoCodecConfigObserver<VideoCodecH264>::GetEncoderSpecificSettings() const {
>       VideoEncoderConfig::H264EncoderSpecificSettings>(encoder_settings_);
> }
> 
>+template <>
>+void VideoCodecConfigObserver<VideoCodecVP8>::InitCodecSpecifics() {
>+  encoder_settings_ = VideoEncoder::GetDefaultVp8Settings();
>+}
>+
> template <>
> void VideoCodecConfigObserver<VideoCodecVP8>::VerifyCodecSpecifics(
>     const VideoCodec& config) const {
>@@ -2456,12 +2637,10 @@ void VideoCodecConfigObserver<VideoCodecVP8>::VerifyCodecSpecifics(
>   }
> 
>   // Set expected temporal layers as they should have been set when
>-  // reconfiguring the encoder and not match the set config. Also copy the
>-  // TemporalLayersFactory pointer that has been injected by VideoStreamEncoder.
>+  // reconfiguring the encoder and not match the set config.
>   VideoCodecVP8 encoder_settings = encoder_settings_;
>   encoder_settings.numberOfTemporalLayers =
>       kVideoCodecConfigObserverNumberOfTemporalLayers;
>-  encoder_settings.tl_factory = config.VP8().tl_factory;
>   EXPECT_EQ(
>       0, memcmp(&config.VP8(), &encoder_settings, sizeof(encoder_settings_)));
> }
>@@ -2473,6 +2652,11 @@ VideoCodecConfigObserver<VideoCodecVP8>::GetEncoderSpecificSettings() const {
>       VideoEncoderConfig::Vp8EncoderSpecificSettings>(encoder_settings_);
> }
> 
>+template <>
>+void VideoCodecConfigObserver<VideoCodecVP9>::InitCodecSpecifics() {
>+  encoder_settings_ = VideoEncoder::GetDefaultVp9Settings();
>+}
>+
> template <>
> void VideoCodecConfigObserver<VideoCodecVP9>::VerifyCodecSpecifics(
>     const VideoCodec& config) const {
>@@ -2520,9 +2704,10 @@ TEST_F(VideoSendStreamTest, EncoderSetupPropagatesH264Config) {
> TEST_F(VideoSendStreamTest, RtcpSenderReportContainsMediaBytesSent) {
>   class RtcpSenderReportTest : public test::SendTest {
>    public:
>-    RtcpSenderReportTest() : SendTest(kDefaultTimeoutMs),
>-                             rtp_packets_sent_(0),
>-                             media_bytes_sent_(0) {}
>+    RtcpSenderReportTest()
>+        : SendTest(kDefaultTimeoutMs),
>+          rtp_packets_sent_(0),
>+          media_bytes_sent_(0) {}
> 
>    private:
>     Action OnSendRtp(const uint8_t* packet, size_t length) override {
>@@ -2568,7 +2753,7 @@ TEST_F(VideoSendStreamTest, RtcpSenderReportContainsMediaBytesSent) {
> }
> 
> TEST_F(VideoSendStreamTest, TranslatesTwoLayerScreencastToTargetBitrate) {
>-  static const int kScreencastTargetBitrateKbps = 200;
>+  static const int kScreencastMaxTargetBitrateDeltaKbps = 1;
> 
>   class VideoStreamFactory
>       : public VideoEncoderConfig::VideoStreamFactoryInterface {
>@@ -2582,9 +2767,13 @@ TEST_F(VideoSendStreamTest, TranslatesTwoLayerScreencastToTargetBitrate) {
>         const VideoEncoderConfig& encoder_config) override {
>       std::vector<VideoStream> streams =
>           test::CreateVideoStreams(width, height, encoder_config);
>-      EXPECT_TRUE(streams[0].temporal_layer_thresholds_bps.empty());
>-      streams[0].temporal_layer_thresholds_bps.push_back(
>-          kScreencastTargetBitrateKbps * 1000);
>+      EXPECT_FALSE(streams[0].num_temporal_layers.has_value());
>+      streams[0].num_temporal_layers = 2;
>+      RTC_CHECK_GT(streams[0].max_bitrate_bps,
>+                   kScreencastMaxTargetBitrateDeltaKbps);
>+      streams[0].target_bitrate_bps =
>+          streams[0].max_bitrate_bps -
>+          kScreencastMaxTargetBitrateDeltaKbps * 1000;
>       return streams;
>     }
>   };
>@@ -2594,23 +2783,26 @@ TEST_F(VideoSendStreamTest, TranslatesTwoLayerScreencastToTargetBitrate) {
>    public:
>     ScreencastTargetBitrateTest()
>         : SendTest(kDefaultTimeoutMs),
>-          test::FakeEncoder(Clock::GetRealTimeClock()) {}
>+          test::FakeEncoder(Clock::GetRealTimeClock()),
>+          encoder_factory_(this) {}
> 
>    private:
>     int32_t InitEncode(const VideoCodec* config,
>                        int32_t number_of_cores,
>                        size_t max_payload_size) override {
>-      EXPECT_EQ(static_cast<unsigned int>(kScreencastTargetBitrateKbps),
>-                config->targetBitrate);
>+      EXPECT_EQ(config->numberOfSimulcastStreams, 1);
>+      EXPECT_EQ(static_cast<unsigned int>(kScreencastMaxTargetBitrateDeltaKbps),
>+                config->simulcastStream[0].maxBitrate -
>+                    config->simulcastStream[0].targetBitrate);
>       observation_complete_.Set();
>-      return test::FakeEncoder::InitEncode(
>-          config, number_of_cores, max_payload_size);
>+      return test::FakeEncoder::InitEncode(config, number_of_cores,
>+                                           max_payload_size);
>     }
>     void ModifyVideoConfigs(
>         VideoSendStream::Config* send_config,
>         std::vector<VideoReceiveStream::Config>* receive_configs,
>         VideoEncoderConfig* encoder_config) override {
>-      send_config->encoder_settings.encoder = this;
>+      send_config->encoder_settings.encoder_factory = &encoder_factory_;
>       EXPECT_EQ(1u, encoder_config->number_of_streams);
>       encoder_config->video_stream_factory =
>           new rtc::RefCountedObject<VideoStreamFactory>();
>@@ -2621,6 +2813,7 @@ TEST_F(VideoSendStreamTest, TranslatesTwoLayerScreencastToTargetBitrate) {
>       EXPECT_TRUE(Wait())
>           << "Timed out while waiting for the encoder to be initialized.";
>     }
>+    test::EncoderProxyFactory encoder_factory_;
>   } test;
> 
>   RunBaseTest(&test);
>@@ -2648,7 +2841,8 @@ TEST_F(VideoSendStreamTest, ReconfigureBitratesSetsEncoderBitratesCorrectly) {
>           target_bitrate_(0),
>           num_initializations_(0),
>           call_(nullptr),
>-          send_stream_(nullptr) {}
>+          send_stream_(nullptr),
>+          encoder_factory_(this) {}
> 
>    private:
>     int32_t InitEncode(const VideoCodec* codecSettings,
>@@ -2685,7 +2879,7 @@ TEST_F(VideoSendStreamTest, ReconfigureBitratesSetsEncoderBitratesCorrectly) {
>                                      maxPayloadSize);
>     }
> 
>-    int32_t SetRateAllocation(const BitrateAllocation& bitrate,
>+    int32_t SetRateAllocation(const VideoBitrateAllocation& bitrate,
>                               uint32_t frameRate) override {
>       {
>         rtc::CritScope lock(&crit_);
>@@ -2706,12 +2900,10 @@ TEST_F(VideoSendStreamTest, ReconfigureBitratesSetsEncoderBitratesCorrectly) {
>       EXPECT_EQ(expected_bitrate, target_bitrate_);
>     }
> 
>-    Call::Config GetSenderCallConfig() override {
>-      Call::Config config(event_log_.get());
>-      config.bitrate_config.min_bitrate_bps = kMinBitrateKbps * 1000;
>-      config.bitrate_config.start_bitrate_bps = kStartBitrateKbps * 1000;
>-      config.bitrate_config.max_bitrate_bps = kMaxBitrateKbps * 1000;
>-      return config;
>+    void ModifySenderCallConfig(Call::Config* config) override {
>+      config->bitrate_config.min_bitrate_bps = kMinBitrateKbps * 1000;
>+      config->bitrate_config.start_bitrate_bps = kStartBitrateKbps * 1000;
>+      config->bitrate_config.max_bitrate_bps = kMaxBitrateKbps * 1000;
>     }
> 
>     class VideoStreamFactory
>@@ -2738,7 +2930,7 @@ TEST_F(VideoSendStreamTest, ReconfigureBitratesSetsEncoderBitratesCorrectly) {
>         VideoSendStream::Config* send_config,
>         std::vector<VideoReceiveStream::Config>* receive_configs,
>         VideoEncoderConfig* encoder_config) override {
>-      send_config->encoder_settings.encoder = this;
>+      send_config->encoder_settings.encoder_factory = &encoder_factory_;
>       // Set bitrates lower/higher than min/max to make sure they are properly
>       // capped.
>       encoder_config->max_bitrate_bps = kMaxBitrateKbps * 1000;
>@@ -2764,11 +2956,12 @@ TEST_F(VideoSendStreamTest, ReconfigureBitratesSetsEncoderBitratesCorrectly) {
>           init_encode_event_.Wait(VideoSendStreamTest::kDefaultTimeoutMs))
>           << "Timed out while waiting for encoder to be configured.";
>       WaitForSetRates(kStartBitrateKbps);
>-      Call::Config::BitrateConfig bitrate_config;
>+      BitrateConstraints bitrate_config;
>       bitrate_config.start_bitrate_bps = kIncreasedStartBitrateKbps * 1000;
>       bitrate_config.max_bitrate_bps = kIncreasedMaxBitrateKbps * 1000;
>       task_queue_->SendTask([this, &bitrate_config]() {
>-        call_->SetBitrateConfig(bitrate_config);
>+        call_->GetTransportControllerSend()->SetSdpBitrateParameters(
>+            bitrate_config);
>       });
>       // Encoder rate is capped by EncoderConfig max_bitrate_bps.
>       WaitForSetRates(kMaxBitrateKbps);
>@@ -2787,7 +2980,7 @@ TEST_F(VideoSendStreamTest, ReconfigureBitratesSetsEncoderBitratesCorrectly) {
>       EXPECT_EQ(3, num_initializations_)
>           << "Encoder should have been reconfigured with the new value.";
>       // Expected target bitrate is the start bitrate set in the call to
>-      // call_->SetBitrateConfig.
>+      // call_->GetTransportControllerSend()->SetSdpBitrateParameters.
>       WaitForSetRates(kIncreasedStartBitrateKbps);
>     }
> 
>@@ -2800,6 +2993,7 @@ TEST_F(VideoSendStreamTest, ReconfigureBitratesSetsEncoderBitratesCorrectly) {
>     int num_initializations_;
>     webrtc::Call* call_;
>     webrtc::VideoSendStream* send_stream_;
>+    test::EncoderProxyFactory encoder_factory_;
>     webrtc::VideoEncoderConfig encoder_config_;
>   } test(&task_queue_);
> 
>@@ -2812,15 +3006,15 @@ TEST_F(VideoSendStreamTest, ReportsSentResolution) {
>   static const struct {
>     int width;
>     int height;
>-  } kEncodedResolution[kNumStreams] = {
>-      {241, 181}, {300, 121}, {121, 221}};
>+  } kEncodedResolution[kNumStreams] = {{241, 181}, {300, 121}, {121, 221}};
>   class ScreencastTargetBitrateTest : public test::SendTest,
>                                       public test::FakeEncoder {
>    public:
>     ScreencastTargetBitrateTest()
>         : SendTest(kDefaultTimeoutMs),
>           test::FakeEncoder(Clock::GetRealTimeClock()),
>-          send_stream_(nullptr) {}
>+          send_stream_(nullptr),
>+          encoder_factory_(this) {}
> 
>    private:
>     int32_t Encode(const VideoFrame& input_image,
>@@ -2858,7 +3052,7 @@ TEST_F(VideoSendStreamTest, ReportsSentResolution) {
>         VideoSendStream::Config* send_config,
>         std::vector<VideoReceiveStream::Config>* receive_configs,
>         VideoEncoderConfig* encoder_config) override {
>-      send_config->encoder_settings.encoder = this;
>+      send_config->encoder_settings.encoder_factory = &encoder_factory_;
>       EXPECT_EQ(kNumStreams, encoder_config->number_of_streams);
>     }
> 
>@@ -2888,6 +3082,7 @@ TEST_F(VideoSendStreamTest, ReportsSentResolution) {
>     }
> 
>     VideoSendStream* send_stream_;
>+    test::EncoderProxyFactory encoder_factory_;
>   } test;
> 
>   RunBaseTest(&test);
>@@ -2898,7 +3093,7 @@ class Vp9HeaderObserver : public test::SendTest {
>  public:
>   Vp9HeaderObserver()
>       : SendTest(VideoSendStreamTest::kLongTimeoutMs),
>-        vp9_encoder_(VP9Encoder::Create()),
>+        encoder_factory_([]() { return VP9Encoder::Create(); }),
>         vp9_settings_(VideoEncoder::GetDefaultVp9Settings()),
>         packets_sent_(0),
>         frames_sent_(0),
>@@ -2928,8 +3123,7 @@ class Vp9HeaderObserver : public test::SendTest {
>         const VideoEncoderConfig& encoder_config) override {
>       std::vector<VideoStream> streams =
>           test::CreateVideoStreams(width, height, encoder_config);
>-      streams[0].temporal_layer_thresholds_bps.resize(
>-          number_of_temporal_layers_ - 1);
>+      streams.back().num_temporal_layers = number_of_temporal_layers_;
>       return streams;
>     }
> 
>@@ -2940,12 +3134,12 @@ class Vp9HeaderObserver : public test::SendTest {
>       VideoSendStream::Config* send_config,
>       std::vector<VideoReceiveStream::Config>* receive_configs,
>       VideoEncoderConfig* encoder_config) override {
>-    send_config->encoder_settings.encoder = vp9_encoder_.get();
>-    send_config->encoder_settings.payload_name = "VP9";
>-    send_config->encoder_settings.payload_type = kVp9PayloadType;
>+    send_config->encoder_settings.encoder_factory = &encoder_factory_;
>+    send_config->rtp.payload_name = "VP9";
>+    send_config->rtp.payload_type = kVp9PayloadType;
>     ModifyVideoConfigsHook(send_config, receive_configs, encoder_config);
>     encoder_config->encoder_specific_settings = new rtc::RefCountedObject<
>-      VideoEncoderConfig::Vp9EncoderSpecificSettings>(vp9_settings_);
>+        VideoEncoderConfig::Vp9EncoderSpecificSettings>(vp9_settings_);
>     EXPECT_EQ(1u, encoder_config->number_of_streams);
>     encoder_config->video_stream_factory =
>         new rtc::RefCountedObject<VideoStreamFactory>(
>@@ -2980,19 +3174,21 @@ class Vp9HeaderObserver : public test::SendTest {
>       RtpDepacketizer::ParsedPayload parsed;
>       RtpDepacketizerVp9 depacketizer;
>       EXPECT_TRUE(depacketizer.Parse(&parsed, payload, payload_length));
>-      EXPECT_EQ(RtpVideoCodecTypes::kRtpVideoVp9, parsed.type.Video.codec);
>+      EXPECT_EQ(VideoCodecType::kVideoCodecVP9, parsed.video_header().codec);
>       // Verify common fields for all configurations.
>-      VerifyCommonHeader(parsed.type.Video.codecHeader.VP9);
>-      CompareConsecutiveFrames(header, parsed.type.Video);
>+      const auto& vp9_header =
>+          absl::get<RTPVideoHeaderVP9>(parsed.video_header().video_type_header);
>+      VerifyCommonHeader(vp9_header);
>+      CompareConsecutiveFrames(header, parsed.video_header());
>       // Verify configuration specific settings.
>-      InspectHeader(parsed.type.Video.codecHeader.VP9);
>+      InspectHeader(vp9_header);
> 
>       ++packets_sent_;
>       if (header.markerBit) {
>         ++frames_sent_;
>       }
>       last_header_ = header;
>-      last_vp9_ = parsed.type.Video.codecHeader.VP9;
>+      last_vp9_ = vp9_header;
>     }
>     return SEND_PACKET;
>   }
>@@ -3074,7 +3270,8 @@ class Vp9HeaderObserver : public test::SendTest {
>           EXPECT_TRUE(vp9.temporal_up_switch);
>           break;
>         case 2:
>-          EXPECT_EQ(last_vp9_.temporal_idx == 0, vp9.temporal_up_switch);
>+          EXPECT_LT(last_vp9_.temporal_idx, 2);
>+          EXPECT_TRUE(vp9.temporal_up_switch);
>           break;
>       }
>     }
>@@ -3176,7 +3373,8 @@ class Vp9HeaderObserver : public test::SendTest {
> 
>   void CompareConsecutiveFrames(const RTPHeader& header,
>                                 const RTPVideoHeader& video) const {
>-    const RTPVideoHeaderVP9& vp9 = video.codecHeader.VP9;
>+    const auto& vp9_header =
>+        absl::get<RTPVideoHeaderVP9>(video.video_type_header);
> 
>     bool new_frame = packets_sent_ == 0 ||
>                      IsNewerTimestamp(header.timestamp, last_header_.timestamp);
>@@ -3184,25 +3382,25 @@ class Vp9HeaderObserver : public test::SendTest {
>     if (!new_frame) {
>       EXPECT_FALSE(last_header_.markerBit);
>       EXPECT_EQ(last_header_.timestamp, header.timestamp);
>-      EXPECT_EQ(last_vp9_.picture_id, vp9.picture_id);
>-      EXPECT_EQ(last_vp9_.temporal_idx, vp9.temporal_idx);
>-      EXPECT_EQ(last_vp9_.tl0_pic_idx, vp9.tl0_pic_idx);
>-      VerifySpatialIdxWithinFrame(vp9);
>+      EXPECT_EQ(last_vp9_.picture_id, vp9_header.picture_id);
>+      EXPECT_EQ(last_vp9_.temporal_idx, vp9_header.temporal_idx);
>+      EXPECT_EQ(last_vp9_.tl0_pic_idx, vp9_header.tl0_pic_idx);
>+      VerifySpatialIdxWithinFrame(vp9_header);
>       return;
>     }
>     // New frame.
>-    EXPECT_TRUE(vp9.beginning_of_frame);
>+    EXPECT_TRUE(vp9_header.beginning_of_frame);
> 
>     // Compare with last packet in previous frame.
>     if (frames_sent_ == 0)
>       return;
>     EXPECT_TRUE(last_vp9_.end_of_frame);
>     EXPECT_TRUE(last_header_.markerBit);
>-    EXPECT_TRUE(ContinuousPictureId(vp9));
>-    VerifyTl0Idx(vp9);
>+    EXPECT_TRUE(ContinuousPictureId(vp9_header));
>+    VerifyTl0Idx(vp9_header);
>   }
> 
>-  std::unique_ptr<VP9Encoder> vp9_encoder_;
>+  test::FunctionVideoEncoderFactory encoder_factory_;
>   VideoCodecVP9 vp9_settings_;
>   webrtc::VideoEncoderConfig encoder_config_;
>   RTPHeader last_header_;
>@@ -3255,16 +3453,31 @@ void VideoSendStreamTest::TestVp9NonFlexMode(uint8_t num_temporal_layers,
>   // Set to < kNumFramesToSend and coprime to length of temporal layer
>   // structures to verify temporal id reset on key frame.
>   static const int kKeyFrameInterval = 31;
>+
>+  static const int kWidth = kMinVp9SpatialLayerWidth;
>+  static const int kHeight = kMinVp9SpatialLayerHeight;
>+  static const float kGoodBitsPerPixel = 0.1f;
>   class NonFlexibleMode : public Vp9HeaderObserver {
>    public:
>     NonFlexibleMode(uint8_t num_temporal_layers, uint8_t num_spatial_layers)
>         : num_temporal_layers_(num_temporal_layers),
>           num_spatial_layers_(num_spatial_layers),
>           l_field_(num_temporal_layers > 1 || num_spatial_layers > 1) {}
>+
>     void ModifyVideoConfigsHook(
>         VideoSendStream::Config* send_config,
>         std::vector<VideoReceiveStream::Config>* receive_configs,
>         VideoEncoderConfig* encoder_config) override {
>+      encoder_config->codec_type = kVideoCodecVP9;
>+      int bitrate_bps = 0;
>+      for (int sl_idx = 0; sl_idx < num_spatial_layers_; ++sl_idx) {
>+        const int width = kWidth << sl_idx;
>+        const int height = kHeight << sl_idx;
>+        const float bpp = kGoodBitsPerPixel / (1 << sl_idx);
>+        bitrate_bps += static_cast<int>(width * height * bpp * 30);
>+      }
>+      encoder_config->max_bitrate_bps = bitrate_bps * 2;
>+
>       vp9_settings_.flexibleMode = false;
>       vp9_settings_.frameDroppingOn = false;
>       vp9_settings_.keyFrameInterval = kKeyFrameInterval;
>@@ -3272,6 +3485,15 @@ void VideoSendStreamTest::TestVp9NonFlexMode(uint8_t num_temporal_layers,
>       vp9_settings_.numberOfSpatialLayers = num_spatial_layers_;
>     }
> 
>+    void ModifyVideoCaptureStartResolution(int* width,
>+                                           int* height,
>+                                           int* frame_rate) override {
>+      expected_width_ = kWidth << (num_spatial_layers_ - 1);
>+      expected_height_ = kHeight << (num_spatial_layers_ - 1);
>+      *width = expected_width_;
>+      *height = expected_height_;
>+    }
>+
>     void InspectHeader(const RTPVideoHeaderVP9& vp9) override {
>       bool ss_data_expected =
>           !vp9.inter_pic_predicted && vp9.beginning_of_frame &&
>@@ -3319,6 +3541,7 @@ TEST_F(VideoSendStreamTest, Vp9NonFlexModeSmallResolution) {
>         VideoSendStream::Config* send_config,
>         std::vector<VideoReceiveStream::Config>* receive_configs,
>         VideoEncoderConfig* encoder_config) override {
>+      encoder_config->codec_type = kVideoCodecVP9;
>       vp9_settings_.flexibleMode = false;
>       vp9_settings_.numberOfTemporalLayers = 1;
>       vp9_settings_.numberOfSpatialLayers = 1;
>@@ -3348,7 +3571,10 @@ TEST_F(VideoSendStreamTest, Vp9NonFlexModeSmallResolution) {
> // Crashes on Android; bugs.webrtc.org/7401
> #define MAYBE_Vp9FlexModeRefCount DISABLED_Vp9FlexModeRefCount
> #else
>-#define MAYBE_Vp9FlexModeRefCount Vp9FlexModeRefCount
>+// TODO(webrtc:9270): Support of flexible mode is temporarily disabled. Enable
>+// the test after webrtc:9270 is implemented.
>+#define MAYBE_Vp9FlexModeRefCount DISABLED_Vp9FlexModeRefCount
>+// #define MAYBE_Vp9FlexModeRefCount Vp9FlexModeRefCount
> #endif
> TEST_F(VideoSendStreamTest, MAYBE_Vp9FlexModeRefCount) {
>   class FlexibleMode : public Vp9HeaderObserver {
>@@ -3356,6 +3582,7 @@ TEST_F(VideoSendStreamTest, MAYBE_Vp9FlexModeRefCount) {
>         VideoSendStream::Config* send_config,
>         std::vector<VideoReceiveStream::Config>* receive_configs,
>         VideoEncoderConfig* encoder_config) override {
>+      encoder_config->codec_type = kVideoCodecVP9;
>       encoder_config->content_type = VideoEncoderConfig::ContentType::kScreen;
>       vp9_settings_.flexibleMode = true;
>       vp9_settings_.numberOfTemporalLayers = 1;
>@@ -3378,20 +3605,20 @@ TEST_F(VideoSendStreamTest, MAYBE_Vp9FlexModeRefCount) {
> 
> void VideoSendStreamTest::TestRequestSourceRotateVideo(
>     bool support_orientation_ext) {
>-  CreateSenderCall(Call::Config(event_log_.get()));
>+  CreateSenderCall();
> 
>   test::NullTransport transport;
>   CreateSendConfig(1, 0, 0, &transport);
>-  video_send_config_.rtp.extensions.clear();
>+  GetVideoSendConfig()->rtp.extensions.clear();
>   if (support_orientation_ext) {
>-    video_send_config_.rtp.extensions.push_back(
>+    GetVideoSendConfig()->rtp.extensions.push_back(
>         RtpExtension(RtpExtension::kVideoRotationUri, 1));
>   }
> 
>   CreateVideoStreams();
>   test::FrameForwarder forwarder;
>-  video_send_stream_->SetSource(
>-      &forwarder, VideoSendStream::DegradationPreference::kMaintainFramerate);
>+  GetVideoSendStream()->SetSource(&forwarder,
>+                                  DegradationPreference::MAINTAIN_FRAMERATE);
> 
>   EXPECT_TRUE(forwarder.sink_wants().rotation_applied !=
>               support_orientation_ext);
>@@ -3423,12 +3650,13 @@ TEST_F(VideoSendStreamTest, RemoveOverheadFromBandwidth) {
>         : EndToEndTest(test::CallTest::kDefaultTimeoutMs),
>           FakeEncoder(Clock::GetRealTimeClock()),
>           task_queue_(task_queue),
>+          encoder_factory_(this),
>           call_(nullptr),
>           max_bitrate_bps_(0),
>           first_packet_sent_(false),
>           bitrate_changed_event_(false, false) {}
> 
>-    int32_t SetRateAllocation(const BitrateAllocation& bitrate,
>+    int32_t SetRateAllocation(const VideoBitrateAllocation& bitrate,
>                               uint32_t frameRate) override {
>       rtc::CritScope lock(&crit_);
>       // Wait for the first sent packet so that videosendstream knows
>@@ -3449,7 +3677,7 @@ TEST_F(VideoSendStreamTest, RemoveOverheadFromBandwidth) {
>         std::vector<VideoReceiveStream::Config>* receive_configs,
>         VideoEncoderConfig* encoder_config) override {
>       send_config->rtp.max_packet_size = 1200;
>-      send_config->encoder_settings.encoder = this;
>+      send_config->encoder_settings.encoder_factory = &encoder_factory_;
>       EXPECT_FALSE(send_config->rtp.extensions.empty());
>     }
> 
>@@ -3460,7 +3688,7 @@ TEST_F(VideoSendStreamTest, RemoveOverheadFromBandwidth) {
>     }
> 
>     void PerformTest() override {
>-      Call::Config::BitrateConfig bitrate_config;
>+      BitrateConstraints bitrate_config;
>       constexpr int kStartBitrateBps = 60000;
>       constexpr int kMaxBitrateBps = 60000;
>       constexpr int kMinBitrateBps = 10000;
>@@ -3468,14 +3696,16 @@ TEST_F(VideoSendStreamTest, RemoveOverheadFromBandwidth) {
>       bitrate_config.max_bitrate_bps = kMaxBitrateBps;
>       bitrate_config.min_bitrate_bps = kMinBitrateBps;
>       task_queue_->SendTask([this, &bitrate_config]() {
>-        call_->SetBitrateConfig(bitrate_config);
>+        call_->GetTransportControllerSend()->SetSdpBitrateParameters(
>+            bitrate_config);
>         call_->OnTransportOverheadChanged(webrtc::MediaType::VIDEO, 40);
>       });
> 
>       // At a bitrate of 60kbps with a packet size of 1200B video and an
>       // overhead of 40B per packet video produces 2240bps overhead.
>       // So the encoder BW should be set to 57760bps.
>-      bitrate_changed_event_.Wait(VideoSendStreamTest::kDefaultTimeoutMs);
>+      EXPECT_TRUE(
>+          bitrate_changed_event_.Wait(VideoSendStreamTest::kDefaultTimeoutMs));
>       {
>         rtc::CritScope lock(&crit_);
>         EXPECT_LE(max_bitrate_bps_, 57760u);
>@@ -3484,6 +3714,7 @@ TEST_F(VideoSendStreamTest, RemoveOverheadFromBandwidth) {
> 
>    private:
>     test::SingleThreadedTaskQueueForTesting* const task_queue_;
>+    test::EncoderProxyFactory encoder_factory_;
>     Call* call_;
>     rtc::CriticalSection crit_;
>     uint32_t max_bitrate_bps_ RTC_GUARDED_BY(&crit_);
>@@ -3539,81 +3770,231 @@ TEST_F(VideoSendStreamTest, SendsKeepAlive) {
>   RunBaseTest(&test);
> }
> 
>-TEST_F(VideoSendStreamTest, ConfiguresAlrWhenSendSideOn) {
>-  const std::string kAlrProbingExperiment =
>-      std::string(AlrExperimentSettings::kScreenshareProbingBweExperimentName) +
>-      "/1.0,2875,80,40,-60,3/";
>-  test::ScopedFieldTrials alr_experiment(kAlrProbingExperiment);
>-  class PacingFactorObserver : public test::SendTest {
>-   public:
>-    PacingFactorObserver(bool configure_send_side, float expected_pacing_factor)
>-        : test::SendTest(kDefaultTimeoutMs),
>-          configure_send_side_(configure_send_side),
>-          expected_pacing_factor_(expected_pacing_factor),
>-          paced_sender_(nullptr) {}
>+class PacingFactorObserver : public test::SendTest {
>+ public:
>+  PacingFactorObserver(bool configure_send_side,
>+                       absl::optional<float> expected_pacing_factor)
>+      : test::SendTest(VideoSendStreamTest::kDefaultTimeoutMs),
>+        configure_send_side_(configure_send_side),
>+        expected_pacing_factor_(expected_pacing_factor) {}
> 
>-    void ModifyVideoConfigs(
>-        VideoSendStream::Config* send_config,
>-        std::vector<VideoReceiveStream::Config>* receive_configs,
>-        VideoEncoderConfig* encoder_config) override {
>-      // Check if send-side bwe extension is already present, and remove it if
>-      // it is not desired.
>-      bool has_send_side = false;
>-      for (auto it = send_config->rtp.extensions.begin();
>-           it != send_config->rtp.extensions.end(); ++it) {
>-        if (it->uri == RtpExtension::kTransportSequenceNumberUri) {
>-          if (configure_send_side_) {
>-            has_send_side = true;
>-          } else {
>-            send_config->rtp.extensions.erase(it);
>-          }
>-          break;
>+  void ModifyVideoConfigs(
>+      VideoSendStream::Config* send_config,
>+      std::vector<VideoReceiveStream::Config>* receive_configs,
>+      VideoEncoderConfig* encoder_config) override {
>+    // Check if send-side bwe extension is already present, and remove it if
>+    // it is not desired.
>+    bool has_send_side = false;
>+    for (auto it = send_config->rtp.extensions.begin();
>+         it != send_config->rtp.extensions.end(); ++it) {
>+      if (it->uri == RtpExtension::kTransportSequenceNumberUri) {
>+        if (configure_send_side_) {
>+          has_send_side = true;
>+        } else {
>+          send_config->rtp.extensions.erase(it);
>         }
>+        break;
>       }
>-
>-      if (configure_send_side_ && !has_send_side) {
>-        // Want send side, not present by default, so add it.
>-        send_config->rtp.extensions.emplace_back(
>-            RtpExtension::kTransportSequenceNumberUri,
>-            RtpExtension::kTransportSequenceNumberDefaultId);
>-      }
>-
>-      // ALR only enabled for screenshare.
>-      encoder_config->content_type = VideoEncoderConfig::ContentType::kScreen;
>     }
> 
>-    void OnRtpTransportControllerSendCreated(
>-        RtpTransportControllerSend* controller) override {
>-      // Grab a reference to the pacer.
>-      paced_sender_ = controller->pacer();
>+    if (configure_send_side_ && !has_send_side) {
>+      // Want send side, not present by default, so add it.
>+      send_config->rtp.extensions.emplace_back(
>+          RtpExtension::kTransportSequenceNumberUri,
>+          RtpExtension::kTransportSequenceNumberDefaultId);
>     }
> 
>-    void OnVideoStreamsCreated(
>-        VideoSendStream* send_stream,
>-        const std::vector<VideoReceiveStream*>& receive_streams) override {
>-      // Video streams created, check that pacer is correctly configured.
>-      EXPECT_EQ(expected_pacing_factor_, paced_sender_->GetPacingFactor());
>-      observation_complete_.Set();
>-    }
>+    // ALR only enabled for screenshare.
>+    encoder_config->content_type = VideoEncoderConfig::ContentType::kScreen;
>+  }
> 
>-    void PerformTest() override {
>-      EXPECT_TRUE(Wait()) << "Timed out while waiting for pacer config.";
>-    }
>+  void OnVideoStreamsCreated(
>+      VideoSendStream* send_stream,
>+      const std::vector<VideoReceiveStream*>& receive_streams) override {
>+    auto internal_send_peer = test::VideoSendStreamPeer(send_stream);
>+    // Video streams created, check that pacing factor is correctly configured.
>+    EXPECT_EQ(expected_pacing_factor_,
>+              internal_send_peer.GetPacingFactorOverride());
>+    observation_complete_.Set();
>+  }
> 
>-   private:
>-    const bool configure_send_side_;
>-    const float expected_pacing_factor_;
>-    const PacedSender* paced_sender_;
>-  };
>+  void PerformTest() override {
>+    EXPECT_TRUE(Wait()) << "Timed out while waiting for stream creation.";
>+  }
>+
>+ private:
>+  const bool configure_send_side_;
>+  const absl::optional<float> expected_pacing_factor_;
>+};
>+
>+std::string GetAlrProbingExperimentString() {
>+  return std::string(
>+             AlrExperimentSettings::kScreenshareProbingBweExperimentName) +
>+         "/1.0,2875,80,40,-60,3/";
>+}
>+const float kAlrProbingExperimentPaceMultiplier = 1.0f;
> 
>+TEST_F(VideoSendStreamTest, AlrConfiguredWhenSendSideOn) {
>+  test::ScopedFieldTrials alr_experiment(GetAlrProbingExperimentString());
>   // Send-side bwe on, use pacing factor from |kAlrProbingExperiment| above.
>-  PacingFactorObserver test_with_send_side(true, 1.0f);
>+  PacingFactorObserver test_with_send_side(true,
>+                                           kAlrProbingExperimentPaceMultiplier);
>   RunBaseTest(&test_with_send_side);
>+}
> 
>-  // Send-side bwe off, use default pacing factor.
>-  PacingFactorObserver test_without_send_side(
>-      false, PacedSender::kDefaultPaceMultiplier);
>+TEST_F(VideoSendStreamTest, AlrNotConfiguredWhenSendSideOff) {
>+  test::ScopedFieldTrials alr_experiment(GetAlrProbingExperimentString());
>+  // Send-side bwe off, use configuration should not be overridden.
>+  PacingFactorObserver test_without_send_side(false, absl::nullopt);
>   RunBaseTest(&test_without_send_side);
> }
> 
>+// Test class takes as argument a function pointer to reset the send
>+// stream and call OnVideoStreamsCreated. This is necessary since you cannot
>+// change the content type of a VideoSendStream, you need to recreate it.
>+// Stopping and recreating the stream can only be done on the main thread and in
>+// the context of VideoSendStreamTest (not BaseTest). The test switches from
>+// realtime to screenshare and back.
>+template <typename T>
>+class ContentSwitchTest : public test::SendTest {
>+ public:
>+  enum class StreamState {
>+    kBeforeSwitch = 0,
>+    kInScreenshare = 1,
>+    kAfterSwitchBack = 2,
>+  };
>+  static const uint32_t kMinPacketsToSend = 50;
>+
>+  explicit ContentSwitchTest(T* stream_reset_fun)
>+      : SendTest(test::CallTest::kDefaultTimeoutMs),
>+        content_switch_event_(false, false),
>+        call_(nullptr),
>+        state_(StreamState::kBeforeSwitch),
>+        send_stream_(nullptr),
>+        send_stream_config_(nullptr),
>+        packets_sent_(0),
>+        stream_resetter_(stream_reset_fun) {
>+    RTC_DCHECK(stream_resetter_);
>+  }
>+
>+  void OnVideoStreamsCreated(
>+      VideoSendStream* send_stream,
>+      const std::vector<VideoReceiveStream*>& receive_streams) override {
>+    rtc::CritScope lock(&crit_);
>+    send_stream_ = send_stream;
>+  }
>+
>+  void ModifyVideoConfigs(
>+      VideoSendStream::Config* send_config,
>+      std::vector<VideoReceiveStream::Config>* receive_configs,
>+      VideoEncoderConfig* encoder_config) override {
>+    RTC_DCHECK_EQ(1, encoder_config->number_of_streams);
>+    encoder_config->min_transmit_bitrate_bps = 0;
>+    encoder_config->content_type =
>+        VideoEncoderConfig::ContentType::kRealtimeVideo;
>+    send_stream_config_ = send_config->Copy();
>+    encoder_config_ = encoder_config->Copy();
>+  }
>+
>+  void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
>+    call_ = sender_call;
>+  }
>+
>+  Action OnSendRtp(const uint8_t* packet, size_t length) override {
>+    rtc::CritScope lock(&crit_);
>+
>+    auto internal_send_peer = test::VideoSendStreamPeer(send_stream_);
>+    float pacing_factor =
>+        internal_send_peer.GetPacingFactorOverride().value_or(0.0f);
>+    float expected_pacing_factor = PacedSender::kDefaultPaceMultiplier;
>+    if (send_stream_->GetStats().content_type ==
>+        webrtc::VideoContentType::SCREENSHARE) {
>+      expected_pacing_factor = 1.0f;  // Currently used pacing factor in ALR.
>+    }
>+
>+    EXPECT_NEAR(expected_pacing_factor, pacing_factor, 1e-6);
>+
>+    // Wait until at least kMinPacketsToSend packets to be sent, so that
>+    // some frames would be encoded.
>+    if (++packets_sent_ < kMinPacketsToSend)
>+      return SEND_PACKET;
>+
>+    if (state_ != StreamState::kAfterSwitchBack) {
>+      // We've sent kMinPacketsToSend packets, switch the content type and move
>+      // move to the next state.
>+      // Note that we need to recreate the stream if changing content type.
>+      packets_sent_ = 0;
>+      if (encoder_config_.content_type ==
>+          VideoEncoderConfig::ContentType::kRealtimeVideo) {
>+        encoder_config_.content_type = VideoEncoderConfig::ContentType::kScreen;
>+      } else {
>+        encoder_config_.content_type =
>+            VideoEncoderConfig::ContentType::kRealtimeVideo;
>+      }
>+      switch (state_) {
>+        case StreamState::kBeforeSwitch:
>+          state_ = StreamState::kInScreenshare;
>+          break;
>+        case StreamState::kInScreenshare:
>+          state_ = StreamState::kAfterSwitchBack;
>+          break;
>+        case StreamState::kAfterSwitchBack:
>+          RTC_NOTREACHED();
>+          break;
>+      }
>+      content_switch_event_.Set();
>+      return SEND_PACKET;
>+    }
>+
>+    observation_complete_.Set();
>+    return SEND_PACKET;
>+  }
>+
>+  void PerformTest() override {
>+    while (GetStreamState() != StreamState::kAfterSwitchBack) {
>+      ASSERT_TRUE(
>+          content_switch_event_.Wait(test::CallTest::kDefaultTimeoutMs));
>+      (*stream_resetter_)(send_stream_config_, encoder_config_, this);
>+    }
>+
>+    ASSERT_TRUE(Wait())
>+        << "Timed out waiting for a frame sent after switch back";
>+  }
>+
>+ private:
>+  StreamState GetStreamState() {
>+    rtc::CritScope lock(&crit_);
>+    return state_;
>+  }
>+
>+  rtc::CriticalSection crit_;
>+  rtc::Event content_switch_event_;
>+  Call* call_;
>+  StreamState state_ RTC_GUARDED_BY(crit_);
>+  VideoSendStream* send_stream_ RTC_GUARDED_BY(crit_);
>+  VideoSendStream::Config send_stream_config_;
>+  VideoEncoderConfig encoder_config_;
>+  uint32_t packets_sent_ RTC_GUARDED_BY(crit_);
>+  T* stream_resetter_;
>+};
>+
>+TEST_F(VideoSendStreamTest, SwitchesToScreenshareAndBack) {
>+  auto reset_fun = [this](const VideoSendStream::Config& send_stream_config,
>+                          const VideoEncoderConfig& encoder_config,
>+                          test::BaseTest* test) {
>+    task_queue_.SendTask([this, &send_stream_config, &encoder_config, &test]() {
>+      Stop();
>+      DestroyVideoSendStreams();
>+      SetVideoSendConfig(send_stream_config);
>+      SetVideoEncoderConfig(encoder_config);
>+      CreateVideoSendStreams();
>+      SetVideoDegradation(DegradationPreference::MAINTAIN_RESOLUTION);
>+      test->OnVideoStreamsCreated(GetVideoSendStream(), video_receive_streams_);
>+      Start();
>+    });
>+  };
>+  ContentSwitchTest<decltype(reset_fun)> test(&reset_fun);
>+  RunBaseTest(&test);
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_stream_decoder.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_stream_decoder.cc
>index b27a52ea892..c1442172239 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_stream_decoder.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_stream_decoder.cc
>@@ -14,13 +14,11 @@
> #include <map>
> #include <vector>
> 
>-#include "common_video/include/frame_callback.h"
> #include "modules/video_coding/video_coding_impl.h"
> #include "rtc_base/checks.h"
> #include "rtc_base/logging.h"
> #include "system_wrappers/include/metrics.h"
> #include "video/call_stats.h"
>-#include "video/payload_router.h"
> #include "video/receive_statistics_proxy.h"
> 
> namespace webrtc {
>@@ -35,14 +33,12 @@ VideoStreamDecoder::VideoStreamDecoder(
>     rtc::VideoSinkInterface<VideoFrame>* incoming_video_stream)
>     : video_receiver_(video_receiver),
>       receive_stats_callback_(receive_statistics_proxy),
>-      incoming_video_stream_(incoming_video_stream),
>-      last_rtt_ms_(0) {
>+      incoming_video_stream_(incoming_video_stream) {
>   RTC_DCHECK(video_receiver_);
> 
>   static const int kMaxPacketAgeToNack = 450;
>   static const int kMaxNackListSize = 250;
>-  video_receiver_->SetNackSettings(kMaxNackListSize,
>-                                   kMaxPacketAgeToNack, 0);
>+  video_receiver_->SetNackSettings(kMaxNackListSize, kMaxPacketAgeToNack, 0);
>   video_receiver_->RegisterReceiveCallback(this);
>   video_receiver_->RegisterFrameTypeCallback(vcm_frame_type_callback);
>   video_receiver_->RegisterReceiveStatisticsCallback(this);
>@@ -76,15 +72,16 @@ VideoStreamDecoder::~VideoStreamDecoder() {
> // thread may have held the lock when calling VideoDecoder::Decode, Reset, or
> // Release. Acquiring the same lock in the path of decode callback can deadlock.
> int32_t VideoStreamDecoder::FrameToRender(VideoFrame& video_frame,
>-                                          rtc::Optional<uint8_t> qp,
>+                                          absl::optional<uint8_t> qp,
>                                           VideoContentType content_type) {
>-  receive_stats_callback_->OnDecodedFrame(qp, content_type);
>+  receive_stats_callback_->OnDecodedFrame(qp, video_frame.width(),
>+                                          video_frame.height(), content_type);
>   incoming_video_stream_->OnFrame(video_frame);
>   return 0;
> }
> 
> int32_t VideoStreamDecoder::ReceivedDecodedReferenceFrame(
>-  const uint64_t picture_id) {
>+    const uint64_t picture_id) {
>   RTC_NOTREACHED();
>   return 0;
> }
>@@ -126,10 +123,7 @@ void VideoStreamDecoder::OnCompleteFrame(bool is_keyframe,
>                                          size_t size_bytes,
>                                          VideoContentType content_type) {}
> 
>-void VideoStreamDecoder::OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) {
>+void VideoStreamDecoder::UpdateRtt(int64_t max_rtt_ms) {
>   video_receiver_->SetReceiveChannelParameters(max_rtt_ms);
>-
>-  rtc::CritScope lock(&crit_);
>-  last_rtt_ms_ = avg_rtt_ms;
> }
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_stream_decoder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_stream_decoder.h
>index 812f47b6a3d..dd8601a5e61 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_stream_decoder.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_stream_decoder.h
>@@ -16,21 +16,16 @@
> #include <memory>
> #include <vector>
> 
>-#include "api/videosinkinterface.h"
>+#include "api/video/video_sink_interface.h"
> #include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
> #include "modules/video_coding/include/video_coding_defines.h"
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/platform_thread.h"
> #include "rtc_base/scoped_ref_ptr.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> 
> namespace webrtc {
> 
>-class CallStatsObserver;
>-class ChannelStatsObserver;
>-class EncodedImageCallback;
> class ReceiveStatisticsProxy;
>-class VideoRenderCallback;
> 
> namespace vcm {
> class VideoReceiver;
>@@ -42,11 +37,8 @@ enum StreamType {
> };
> 
> class VideoStreamDecoder : public VCMReceiveCallback,
>-                           public VCMReceiveStatisticsCallback,
>-                           public CallStatsObserver {
>+                           public VCMReceiveStatisticsCallback {
>  public:
>-  friend class ChannelStatsObserver;
>-
>   VideoStreamDecoder(
>       vcm::VideoReceiver* video_receiver,
>       VCMFrameTypeCallback* vcm_frame_type_callback,
>@@ -59,7 +51,7 @@ class VideoStreamDecoder : public VCMReceiveCallback,
> 
>   // Implements VCMReceiveCallback.
>   int32_t FrameToRender(VideoFrame& video_frame,
>-                        rtc::Optional<uint8_t> qp,
>+                        absl::optional<uint8_t> qp,
>                         VideoContentType content_type) override;
>   int32_t ReceivedDecodedReferenceFrame(const uint64_t picture_id) override;
>   void OnIncomingPayloadType(int payload_type) override;
>@@ -85,8 +77,8 @@ class VideoStreamDecoder : public VCMReceiveCallback,
>   void RegisterReceiveStatisticsProxy(
>       ReceiveStatisticsProxy* receive_statistics_proxy);
> 
>-  // Implements StatsObserver.
>-  void OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) override;
>+  // Called by VideoReceiveStream when stats are updated.
>+  void UpdateRtt(int64_t max_rtt_ms);
> 
>  private:
>   // Used for all registered callbacks except rendering.
>@@ -96,8 +88,6 @@ class VideoStreamDecoder : public VCMReceiveCallback,
> 
>   ReceiveStatisticsProxy* const receive_stats_callback_;
>   rtc::VideoSinkInterface<VideoFrame>* const incoming_video_stream_;
>-
>-  int64_t last_rtt_ms_ RTC_GUARDED_BY(crit_);
> };
> 
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_stream_decoder_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_stream_decoder_impl.cc
>new file mode 100644
>index 00000000000..5313d6a59e4
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_stream_decoder_impl.cc
>@@ -0,0 +1,280 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#include "video/video_stream_decoder_impl.h"
>+
>+#include "absl/memory/memory.h"
>+#include "rtc_base/logging.h"
>+#include "rtc_base/numerics/mod_ops.h"
>+
>+namespace webrtc {
>+
>+VideoStreamDecoderImpl::VideoStreamDecoderImpl(
>+    VideoStreamDecoder::Callbacks* callbacks,
>+    VideoDecoderFactory* decoder_factory,
>+    std::map<int, std::pair<SdpVideoFormat, int>> decoder_settings)
>+    : callbacks_(callbacks),
>+      decoder_factory_(decoder_factory),
>+      decoder_settings_(std::move(decoder_settings)),
>+      bookkeeping_queue_("video_stream_decoder_bookkeeping_queue"),
>+      decode_thread_(&DecodeLoop,
>+                     this,
>+                     "video_stream_decoder_decode_thread",
>+                     rtc::kHighestPriority),
>+      jitter_estimator_(Clock::GetRealTimeClock()),
>+      timing_(Clock::GetRealTimeClock()),
>+      frame_buffer_(Clock::GetRealTimeClock(),
>+                    &jitter_estimator_,
>+                    &timing_,
>+                    nullptr),
>+      next_frame_timestamps_index_(0) {
>+  frame_timestamps_.fill({-1, -1, -1});
>+  decode_thread_.Start();
>+}
>+
>+VideoStreamDecoderImpl::~VideoStreamDecoderImpl() {
>+  frame_buffer_.Stop();
>+  decode_thread_.Stop();
>+}
>+
>+void VideoStreamDecoderImpl::OnFrame(
>+    std::unique_ptr<video_coding::EncodedFrame> frame) {
>+  if (!bookkeeping_queue_.IsCurrent()) {
>+    struct OnFrameTask : rtc::QueuedTask {
>+      OnFrameTask(std::unique_ptr<video_coding::EncodedFrame> frame,
>+                  VideoStreamDecoderImpl* video_stream_decoder)
>+          : frame_(std::move(frame)),
>+            video_stream_decoder_(video_stream_decoder) {}
>+
>+      bool Run() override {
>+        video_stream_decoder_->OnFrame(std::move(frame_));
>+        return true;
>+      }
>+
>+      std::unique_ptr<video_coding::EncodedFrame> frame_;
>+      VideoStreamDecoderImpl* video_stream_decoder_;
>+    };
>+
>+    bookkeeping_queue_.PostTask(
>+        absl::make_unique<OnFrameTask>(std::move(frame), this));
>+    return;
>+  }
>+
>+  RTC_DCHECK_RUN_ON(&bookkeeping_queue_);
>+
>+  uint64_t continuous_pid = frame_buffer_.InsertFrame(std::move(frame));
>+  video_coding::VideoLayerFrameId continuous_id(continuous_pid, 0);
>+  if (last_continuous_id_ < continuous_id) {
>+    last_continuous_id_ = continuous_id;
>+    callbacks_->OnContinuousUntil(last_continuous_id_);
>+  }
>+}
>+
>+VideoDecoder* VideoStreamDecoderImpl::GetDecoder(int payload_type) {
>+  if (current_payload_type_ == payload_type) {
>+    RTC_DCHECK(decoder_);
>+    return decoder_.get();
>+  }
>+
>+  current_payload_type_.reset();
>+  decoder_.reset();
>+
>+  auto decoder_settings_it = decoder_settings_.find(payload_type);
>+  if (decoder_settings_it == decoder_settings_.end()) {
>+    RTC_LOG(LS_WARNING) << "Payload type " << payload_type
>+                        << " not registered.";
>+    return nullptr;
>+  }
>+
>+  const SdpVideoFormat& video_format = decoder_settings_it->second.first;
>+  std::unique_ptr<VideoDecoder> decoder =
>+      decoder_factory_->CreateVideoDecoder(video_format);
>+  if (!decoder) {
>+    RTC_LOG(LS_WARNING) << "Failed to create decoder for payload type "
>+                        << payload_type << ".";
>+    return nullptr;
>+  }
>+
>+  int num_cores = decoder_settings_it->second.second;
>+  int32_t init_result = decoder->InitDecode(nullptr, num_cores);
>+  if (init_result != WEBRTC_VIDEO_CODEC_OK) {
>+    RTC_LOG(LS_WARNING) << "Failed to initialize decoder for payload type "
>+                        << payload_type << ".";
>+    return nullptr;
>+  }
>+
>+  int32_t register_result = decoder->RegisterDecodeCompleteCallback(this);
>+  if (register_result != WEBRTC_VIDEO_CODEC_OK) {
>+    RTC_LOG(LS_WARNING) << "Failed to register decode callback.";
>+    return nullptr;
>+  }
>+
>+  current_payload_type_.emplace(payload_type);
>+  decoder_ = std::move(decoder);
>+  return decoder_.get();
>+}
>+
>+// static
>+void VideoStreamDecoderImpl::DecodeLoop(void* ptr) {
>+  // TODO(philipel): Remove this and use rtc::Event::kForever when it's
>+  //                 supported by the |frame_buffer_|.
>+  static constexpr int kForever = 100000000;
>+
>+  int max_wait_time_ms = kForever;
>+  bool keyframe_required = true;
>+  auto* vs_decoder = static_cast<VideoStreamDecoderImpl*>(ptr);
>+  while (true) {
>+    DecodeResult decode_result =
>+        vs_decoder->DecodeNextFrame(max_wait_time_ms, keyframe_required);
>+
>+    switch (decode_result) {
>+      case kOk: {
>+        max_wait_time_ms = kForever;
>+        keyframe_required = false;
>+        break;
>+      }
>+      case kDecodeFailure: {
>+        max_wait_time_ms = 0;
>+        keyframe_required = true;
>+        break;
>+      }
>+      case kNoFrame: {
>+        max_wait_time_ms = kForever;
>+        // If we end up here it means that we got a decoding error and there is
>+        // no keyframe available in the |frame_buffer_|.
>+        vs_decoder->bookkeeping_queue_.PostTask([vs_decoder]() {
>+          RTC_DCHECK_RUN_ON(&vs_decoder->bookkeeping_queue_);
>+          vs_decoder->callbacks_->OnNonDecodableState();
>+        });
>+        break;
>+      }
>+      case kNoDecoder: {
>+        max_wait_time_ms = kForever;
>+        break;
>+      }
>+      case kShutdown: {
>+        return;
>+      }
>+    }
>+  }
>+}
>+
>+VideoStreamDecoderImpl::DecodeResult VideoStreamDecoderImpl::DecodeNextFrame(
>+    int max_wait_time_ms,
>+    bool keyframe_required) {
>+  std::unique_ptr<video_coding::EncodedFrame> frame;
>+  video_coding::FrameBuffer::ReturnReason res =
>+      frame_buffer_.NextFrame(max_wait_time_ms, &frame, keyframe_required);
>+
>+  if (res == video_coding::FrameBuffer::ReturnReason::kStopped)
>+    return kShutdown;
>+
>+  if (frame) {
>+    VideoDecoder* decoder = GetDecoder(frame->PayloadType());
>+    if (!decoder) {
>+      RTC_LOG(LS_WARNING) << "Failed to get decoder, dropping frame ("
>+                          << frame->id.picture_id << ":"
>+                          << frame->id.spatial_layer << ").";
>+      return kNoDecoder;
>+    }
>+
>+    int64_t decode_start_time_ms = rtc::TimeMillis();
>+    int64_t timestamp = frame->timestamp;
>+    int64_t render_time_us = frame->RenderTimeMs() * 1000;
>+    bookkeeping_queue_.PostTask(
>+        [this, decode_start_time_ms, timestamp, render_time_us]() {
>+          RTC_DCHECK_RUN_ON(&bookkeeping_queue_);
>+          // Saving decode start time this way wont work if we decode spatial
>+          // layers sequentially.
>+          FrameTimestamps* frame_timestamps =
>+              &frame_timestamps_[next_frame_timestamps_index_];
>+          frame_timestamps->timestamp = timestamp;
>+          frame_timestamps->decode_start_time_ms = decode_start_time_ms;
>+          frame_timestamps->render_time_us = render_time_us;
>+
>+          next_frame_timestamps_index_ =
>+              Add<kFrameTimestampsMemory>(next_frame_timestamps_index_, 1);
>+        });
>+
>+    int32_t decode_result = decoder->Decode(frame->EncodedImage(),
>+                                            false,    // missing_frame
>+                                            nullptr,  // codec specific info
>+                                            frame->RenderTimeMs());
>+
>+    return decode_result == WEBRTC_VIDEO_CODEC_OK ? kOk : kDecodeFailure;
>+  }
>+
>+  return kNoFrame;
>+}
>+
>+VideoStreamDecoderImpl::FrameTimestamps*
>+VideoStreamDecoderImpl::GetFrameTimestamps(int64_t timestamp) {
>+  RTC_DCHECK_RUN_ON(&bookkeeping_queue_);
>+
>+  int start_time_index = next_frame_timestamps_index_;
>+  for (int i = 0; i < kFrameTimestampsMemory; ++i) {
>+    start_time_index = Subtract<kFrameTimestampsMemory>(start_time_index, 1);
>+
>+    if (frame_timestamps_[start_time_index].timestamp == timestamp)
>+      return &frame_timestamps_[start_time_index];
>+  }
>+
>+  return nullptr;
>+}
>+
>+// VideoDecoder::DecodedImageCallback
>+int32_t VideoStreamDecoderImpl::Decoded(VideoFrame& decoded_image) {
>+  Decoded(decoded_image, absl::nullopt, absl::nullopt);
>+  return WEBRTC_VIDEO_CODEC_OK;
>+}
>+
>+// VideoDecoder::DecodedImageCallback
>+int32_t VideoStreamDecoderImpl::Decoded(VideoFrame& decoded_image,
>+                                        int64_t decode_time_ms) {
>+  Decoded(decoded_image, decode_time_ms, absl::nullopt);
>+  return WEBRTC_VIDEO_CODEC_OK;
>+}
>+
>+// VideoDecoder::DecodedImageCallback
>+void VideoStreamDecoderImpl::Decoded(VideoFrame& decoded_image,
>+                                     absl::optional<int32_t> decode_time_ms,
>+                                     absl::optional<uint8_t> qp) {
>+  int64_t decode_stop_time_ms = rtc::TimeMillis();
>+
>+  bookkeeping_queue_.PostTask([this, decode_stop_time_ms, decoded_image,
>+                               decode_time_ms, qp]() {
>+    RTC_DCHECK_RUN_ON(&bookkeeping_queue_);
>+
>+    FrameTimestamps* frame_timestamps =
>+        GetFrameTimestamps(decoded_image.timestamp());
>+    if (!frame_timestamps) {
>+      RTC_LOG(LS_ERROR) << "No frame information found for frame with timestamp"
>+                        << decoded_image.timestamp();
>+      return;
>+    }
>+
>+    absl::optional<int> casted_qp;
>+    if (qp)
>+      casted_qp.emplace(*qp);
>+
>+    absl::optional<int> casted_decode_time_ms(decode_time_ms.value_or(
>+        decode_stop_time_ms - frame_timestamps->decode_start_time_ms));
>+
>+    timing_.StopDecodeTimer(0, *casted_decode_time_ms, decode_stop_time_ms,
>+                            frame_timestamps->render_time_us / 1000);
>+
>+    callbacks_->OnDecodedFrame(
>+        VideoFrame(decoded_image.video_frame_buffer(), decoded_image.rotation(),
>+                   frame_timestamps->render_time_us),
>+        casted_decode_time_ms, casted_qp);
>+  });
>+}
>+
>+}  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_stream_decoder_impl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_stream_decoder_impl.h
>new file mode 100644
>index 00000000000..cd1f953a6d5
>--- /dev/null
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_stream_decoder_impl.h
>@@ -0,0 +1,99 @@
>+/*
>+ *  Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
>+ *
>+ *  Use of this source code is governed by a BSD-style license
>+ *  that can be found in the LICENSE file in the root of the source
>+ *  tree. An additional intellectual property rights grant can be found
>+ *  in the file PATENTS.  All contributing project authors may
>+ *  be found in the AUTHORS file in the root of the source tree.
>+ */
>+
>+#ifndef VIDEO_VIDEO_STREAM_DECODER_IMPL_H_
>+#define VIDEO_VIDEO_STREAM_DECODER_IMPL_H_
>+
>+#include <map>
>+#include <memory>
>+#include <utility>
>+
>+#include "absl/types/optional.h"
>+#include "api/video/video_stream_decoder.h"
>+#include "modules/video_coding/frame_buffer2.h"
>+#include "modules/video_coding/jitter_estimator.h"
>+#include "modules/video_coding/timing.h"
>+#include "rtc_base/platform_thread.h"
>+#include "rtc_base/task_queue.h"
>+#include "rtc_base/thread_checker.h"
>+#include "system_wrappers/include/clock.h"
>+
>+namespace webrtc {
>+
>+class VideoStreamDecoderImpl : public VideoStreamDecoder,
>+                               private DecodedImageCallback {
>+ public:
>+  VideoStreamDecoderImpl(
>+      VideoStreamDecoder::Callbacks* callbacks,
>+      VideoDecoderFactory* decoder_factory,
>+      std::map<int, std::pair<SdpVideoFormat, int>> decoder_settings);
>+
>+  ~VideoStreamDecoderImpl() override;
>+
>+  void OnFrame(std::unique_ptr<video_coding::EncodedFrame> frame) override;
>+
>+ private:
>+  enum DecodeResult {
>+    kOk,
>+    kDecodeFailure,
>+    kNoFrame,
>+    kNoDecoder,
>+    kShutdown,
>+  };
>+
>+  struct FrameTimestamps {
>+    int64_t timestamp;
>+    int64_t decode_start_time_ms;
>+    int64_t render_time_us;
>+  };
>+
>+  VideoDecoder* GetDecoder(int payload_type);
>+  static void DecodeLoop(void* ptr);
>+  DecodeResult DecodeNextFrame(int max_wait_time_ms, bool keyframe_required);
>+
>+  FrameTimestamps* GetFrameTimestamps(int64_t timestamp);
>+
>+  // Implements DecodedImageCallback interface
>+  int32_t Decoded(VideoFrame& decodedImage) override;
>+  int32_t Decoded(VideoFrame& decodedImage, int64_t decode_time_ms) override;
>+  void Decoded(VideoFrame& decodedImage,
>+               absl::optional<int32_t> decode_time_ms,
>+               absl::optional<uint8_t> qp) override;
>+
>+  VideoStreamDecoder::Callbacks* const callbacks_
>+      RTC_PT_GUARDED_BY(bookkeeping_queue_);
>+  VideoDecoderFactory* const decoder_factory_;
>+  std::map<int, std::pair<SdpVideoFormat, int>> decoder_settings_;
>+
>+  // The |bookkeeping_queue_| is used to:
>+  //  - Make |callbacks_|.
>+  //  - Insert/extract frames from the |frame_buffer_|
>+  //  - Synchronize with whatever thread that makes the Decoded callback.
>+  rtc::TaskQueue bookkeeping_queue_;
>+
>+  rtc::PlatformThread decode_thread_;
>+  VCMJitterEstimator jitter_estimator_;
>+  VCMTiming timing_;
>+  video_coding::FrameBuffer frame_buffer_;
>+  video_coding::VideoLayerFrameId last_continuous_id_;
>+  absl::optional<int> current_payload_type_;
>+  std::unique_ptr<VideoDecoder> decoder_;
>+
>+  // Some decoders are pipelined so it is not sufficient to save frame info
>+  // for the last frame only.
>+  static constexpr int kFrameTimestampsMemory = 8;
>+  std::array<FrameTimestamps, kFrameTimestampsMemory> frame_timestamps_
>+      RTC_GUARDED_BY(bookkeeping_queue_);
>+  int next_frame_timestamps_index_ RTC_GUARDED_BY(bookkeeping_queue_);
>+};
>+
>+}  // namespace webrtc
>+
>+#endif  // VIDEO_VIDEO_STREAM_DECODER_IMPL_H_
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_stream_encoder.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_stream_encoder.cc
>index 7ec7ab3cbce..4bf3a860167 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_stream_encoder.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_stream_encoder.cc
>@@ -16,21 +16,19 @@
> #include <utility>
> 
> #include "api/video/i420_buffer.h"
>-#include "common_video/include/video_bitrate_allocator.h"
> #include "common_video/include/video_frame.h"
>-#include "modules/pacing/paced_sender.h"
>-#include "modules/video_coding/codecs/vp8/temporal_layers.h"
> #include "modules/video_coding/include/video_codec_initializer.h"
> #include "modules/video_coding/include/video_coding.h"
>-#include "modules/video_coding/include/video_coding_defines.h"
> #include "rtc_base/arraysize.h"
> #include "rtc_base/checks.h"
>+#include "rtc_base/experiments/quality_scaling_experiment.h"
> #include "rtc_base/location.h"
> #include "rtc_base/logging.h"
>+#include "rtc_base/system/fallthrough.h"
> #include "rtc_base/timeutils.h"
> #include "rtc_base/trace_event.h"
>+#include "system_wrappers/include/field_trial.h"
> #include "video/overuse_frame_detector.h"
>-#include "video/send_statistics_proxy.h"
> 
> namespace webrtc {
> 
>@@ -41,22 +39,19 @@ const int64_t kFrameLogIntervalMs = 60000;
> const int kMinFramerateFps = 2;
> const int kMaxFramerateFps = 120;
> 
>+// Time to keep a single cached pending frame in paused state.
>+const int64_t kPendingFrameTimeoutMs = 1000;
>+
>+const char kInitialFramedropFieldTrial[] = "WebRTC-InitialFramedrop";
>+
> // The maximum number of frames to drop at beginning of stream
> // to try and achieve desired bitrate.
> const int kMaxInitialFramedrop = 4;
>+// When the first change in BWE above this threshold occurs,
>+// enable DropFrameDueToSize logic.
>+const float kFramedropThreshold = 0.3;
> 
>-uint32_t MaximumFrameSizeForBitrate(uint32_t kbps) {
>-  if (kbps > 0) {
>-    if (kbps < 300 /* qvga */) {
>-      return 320 * 240;
>-    } else if (kbps < 500 /* vga */) {
>-      return 640 * 480;
>-    }
>-  }
>-  return std::numeric_limits<uint32_t>::max();
>-}
>-
>-// Initial limits for kBalanced degradation preference.
>+// Initial limits for BALANCED degradation preference.
> int MinFps(int pixels) {
>   if (pixels <= 320 * 240) {
>     return 7;
>@@ -79,95 +74,41 @@ int MaxFps(int pixels) {
>   }
> }
> 
>-bool IsResolutionScalingEnabled(
>-    VideoSendStream::DegradationPreference degradation_preference) {
>-  return degradation_preference ==
>-             VideoSendStream::DegradationPreference::kMaintainFramerate ||
>-         degradation_preference ==
>-             VideoSendStream::DegradationPreference::kBalanced;
>+uint32_t abs_diff(uint32_t a, uint32_t b) {
>+  return (a < b) ? b - a : a - b;
> }
> 
>-bool IsFramerateScalingEnabled(
>-    VideoSendStream::DegradationPreference degradation_preference) {
>-  return degradation_preference ==
>-             VideoSendStream::DegradationPreference::kMaintainResolution ||
>-         degradation_preference ==
>-             VideoSendStream::DegradationPreference::kBalanced;
>+bool IsResolutionScalingEnabled(DegradationPreference degradation_preference) {
>+  return degradation_preference == DegradationPreference::MAINTAIN_FRAMERATE ||
>+         degradation_preference == DegradationPreference::BALANCED;
> }
> 
>-}  //  namespace
>+bool IsFramerateScalingEnabled(DegradationPreference degradation_preference) {
>+  return degradation_preference == DegradationPreference::MAINTAIN_RESOLUTION ||
>+         degradation_preference == DegradationPreference::BALANCED;
>+}
> 
>-class VideoStreamEncoder::ConfigureEncoderTask : public rtc::QueuedTask {
>- public:
>-  ConfigureEncoderTask(VideoStreamEncoder* video_stream_encoder,
>-                       VideoEncoderConfig config,
>-                       size_t max_data_payload_length,
>-                       bool nack_enabled)
>-      : video_stream_encoder_(video_stream_encoder),
>-        config_(std::move(config)),
>-        max_data_payload_length_(max_data_payload_length),
>-        nack_enabled_(nack_enabled) {}
>+// TODO(pbos): Lower these thresholds (to closer to 100%) when we handle
>+// pipelining encoders better (multiple input frames before something comes
>+// out). This should effectively turn off CPU adaptations for systems that
>+// remotely cope with the load right now.
>+CpuOveruseOptions GetCpuOveruseOptions(
>+    const VideoStreamEncoderSettings& settings,
>+    bool full_overuse_time) {
>+  CpuOveruseOptions options;
> 
>- private:
>-  bool Run() override {
>-    video_stream_encoder_->ConfigureEncoderOnTaskQueue(
>-        std::move(config_), max_data_payload_length_, nack_enabled_);
>-    return true;
>+  if (full_overuse_time) {
>+    options.low_encode_usage_threshold_percent = 150;
>+    options.high_encode_usage_threshold_percent = 200;
>   }
>-
>-  VideoStreamEncoder* const video_stream_encoder_;
>-  VideoEncoderConfig config_;
>-  size_t max_data_payload_length_;
>-  bool nack_enabled_;
>-};
>-
>-class VideoStreamEncoder::EncodeTask : public rtc::QueuedTask {
>- public:
>-  EncodeTask(const VideoFrame& frame,
>-             VideoStreamEncoder* video_stream_encoder,
>-             int64_t time_when_posted_us,
>-             bool log_stats)
>-      : frame_(frame),
>-        video_stream_encoder_(video_stream_encoder),
>-        time_when_posted_us_(time_when_posted_us),
>-        log_stats_(log_stats) {
>-    ++video_stream_encoder_->posted_frames_waiting_for_encode_;
>+  if (settings.experiment_cpu_load_estimator) {
>+    options.filter_time_ms = 5 * rtc::kNumMillisecsPerSec;
>   }
> 
>- private:
>-  bool Run() override {
>-    RTC_DCHECK_RUN_ON(&video_stream_encoder_->encoder_queue_);
>-    video_stream_encoder_->stats_proxy_->OnIncomingFrame(frame_.width(),
>-                                                         frame_.height());
>-    ++video_stream_encoder_->captured_frame_count_;
>-    const int posted_frames_waiting_for_encode =
>-        video_stream_encoder_->posted_frames_waiting_for_encode_.fetch_sub(1);
>-    RTC_DCHECK_GT(posted_frames_waiting_for_encode, 0);
>-    if (posted_frames_waiting_for_encode == 1) {
>-      video_stream_encoder_->EncodeVideoFrame(frame_, time_when_posted_us_);
>-    } else {
>-      // There is a newer frame in flight. Do not encode this frame.
>-      RTC_LOG(LS_VERBOSE)
>-          << "Incoming frame dropped due to that the encoder is blocked.";
>-      ++video_stream_encoder_->dropped_frame_count_;
>-      video_stream_encoder_->stats_proxy_->OnFrameDroppedInEncoderQueue();
>-    }
>-    if (log_stats_) {
>-      RTC_LOG(LS_INFO) << "Number of frames: captured "
>-                       << video_stream_encoder_->captured_frame_count_
>-                       << ", dropped (due to encoder blocked) "
>-                       << video_stream_encoder_->dropped_frame_count_
>-                       << ", interval_ms " << kFrameLogIntervalMs;
>-      video_stream_encoder_->captured_frame_count_ = 0;
>-      video_stream_encoder_->dropped_frame_count_ = 0;
>-    }
>-    return true;
>-  }
>-  VideoFrame frame_;
>-  VideoStreamEncoder* const video_stream_encoder_;
>-  const int64_t time_when_posted_us_;
>-  const bool log_stats_;
>-};
>+  return options;
>+}
>+
>+}  //  namespace
> 
> // VideoSourceProxy is responsible ensuring thread safety between calls to
> // VideoStreamEncoder::SetSource that will happen on libjingle's worker thread
>@@ -177,13 +118,11 @@ class VideoStreamEncoder::VideoSourceProxy {
>  public:
>   explicit VideoSourceProxy(VideoStreamEncoder* video_stream_encoder)
>       : video_stream_encoder_(video_stream_encoder),
>-        degradation_preference_(
>-            VideoSendStream::DegradationPreference::kDegradationDisabled),
>+        degradation_preference_(DegradationPreference::DISABLED),
>         source_(nullptr) {}
> 
>-  void SetSource(
>-      rtc::VideoSourceInterface<VideoFrame>* source,
>-      const VideoSendStream::DegradationPreference& degradation_preference) {
>+  void SetSource(rtc::VideoSourceInterface<VideoFrame>* source,
>+                 const DegradationPreference& degradation_preference) {
>     // Called on libjingle's worker thread.
>     RTC_DCHECK_CALLED_SEQUENTIALLY(&main_checker_);
>     rtc::VideoSourceInterface<VideoFrame>* old_source = nullptr;
>@@ -251,7 +190,7 @@ class VideoStreamEncoder::VideoSourceProxy {
>     RTC_LOG(LS_INFO) << "Scaling down resolution, max pixels: "
>                      << pixels_wanted;
>     sink_wants_.max_pixel_count = pixels_wanted;
>-    sink_wants_.target_pixel_count = rtc::nullopt;
>+    sink_wants_.target_pixel_count = absl::nullopt;
>     source_->AddOrUpdateSink(video_stream_encoder_,
>                              GetActiveSinkWantsInternal());
>     return true;
>@@ -354,16 +293,16 @@ class VideoStreamEncoder::VideoSourceProxy {
>     // Clear any constraints from the current sink wants that don't apply to
>     // the used degradation_preference.
>     switch (degradation_preference_) {
>-      case VideoSendStream::DegradationPreference::kBalanced:
>+      case DegradationPreference::BALANCED:
>         break;
>-      case VideoSendStream::DegradationPreference::kMaintainFramerate:
>+      case DegradationPreference::MAINTAIN_FRAMERATE:
>         wants.max_framerate_fps = std::numeric_limits<int>::max();
>         break;
>-      case VideoSendStream::DegradationPreference::kMaintainResolution:
>+      case DegradationPreference::MAINTAIN_RESOLUTION:
>         wants.max_pixel_count = std::numeric_limits<int>::max();
>         wants.target_pixel_count.reset();
>         break;
>-      case VideoSendStream::DegradationPreference::kDegradationDisabled:
>+      case DegradationPreference::DISABLED:
>         wants.max_pixel_count = std::numeric_limits<int>::max();
>         wants.target_pixel_count.reset();
>         wants.max_framerate_fps = std::numeric_limits<int>::max();
>@@ -375,8 +314,7 @@ class VideoStreamEncoder::VideoSourceProxy {
>   rtc::SequencedTaskChecker main_checker_;
>   VideoStreamEncoder* const video_stream_encoder_;
>   rtc::VideoSinkWants sink_wants_ RTC_GUARDED_BY(&crit_);
>-  VideoSendStream::DegradationPreference degradation_preference_
>-      RTC_GUARDED_BY(&crit_);
>+  DegradationPreference degradation_preference_ RTC_GUARDED_BY(&crit_);
>   rtc::VideoSourceInterface<VideoFrame>* source_ RTC_GUARDED_BY(&crit_);
> 
>   RTC_DISALLOW_COPY_AND_ASSIGN(VideoSourceProxy);
>@@ -384,37 +322,32 @@ class VideoStreamEncoder::VideoSourceProxy {
> 
> VideoStreamEncoder::VideoStreamEncoder(
>     uint32_t number_of_cores,
>-    SendStatisticsProxy* stats_proxy,
>-    const VideoSendStream::Config::EncoderSettings& settings,
>+    VideoStreamEncoderObserver* encoder_stats_observer,
>+    const VideoStreamEncoderSettings& settings,
>     rtc::VideoSinkInterface<VideoFrame>* pre_encode_callback,
>     std::unique_ptr<OveruseFrameDetector> overuse_detector)
>     : shutdown_event_(true /* manual_reset */, false),
>       number_of_cores_(number_of_cores),
>-      initial_rampup_(0),
>+      initial_framedrop_(0),
>+      initial_framedrop_on_bwe_enabled_(
>+          webrtc::field_trial::IsEnabled(kInitialFramedropFieldTrial)),
>+      quality_scaling_experiment_enabled_(QualityScalingExperiment::Enabled()),
>       source_proxy_(new VideoSourceProxy(this)),
>       sink_(nullptr),
>       settings_(settings),
>-      codec_type_(PayloadStringToCodecType(settings.payload_name)),
>       video_sender_(Clock::GetRealTimeClock(), this),
>-      overuse_detector_(
>-          overuse_detector.get()
>-              ? overuse_detector.release()
>-              : new OveruseFrameDetector(
>-                    GetCpuOveruseOptions(settings.full_overuse_time),
>-                    this,
>-                    stats_proxy)),
>-      stats_proxy_(stats_proxy),
>+      overuse_detector_(std::move(overuse_detector)),
>+      encoder_stats_observer_(encoder_stats_observer),
>       pre_encode_callback_(pre_encode_callback),
>       max_framerate_(-1),
>       pending_encoder_reconfiguration_(false),
>+      pending_encoder_creation_(false),
>       encoder_start_bitrate_bps_(0),
>       max_data_payload_length_(0),
>-      nack_enabled_(false),
>       last_observed_bitrate_bps_(0),
>       encoder_paused_and_dropped_frame_(false),
>       clock_(Clock::GetRealTimeClock()),
>-      degradation_preference_(
>-          VideoSendStream::DegradationPreference::kDegradationDisabled),
>+      degradation_preference_(DegradationPreference::DISABLED),
>       posted_frames_waiting_for_encode_(0),
>       last_captured_timestamp_(0),
>       delta_ntp_internal_ms_(clock_->CurrentNtpInMilliseconds() -
>@@ -424,13 +357,8 @@ VideoStreamEncoder::VideoStreamEncoder(
>       dropped_frame_count_(0),
>       bitrate_observer_(nullptr),
>       encoder_queue_("EncoderQueue") {
>-  RTC_DCHECK(stats_proxy);
>-  encoder_queue_.PostTask([this] {
>-    RTC_DCHECK_RUN_ON(&encoder_queue_);
>-    overuse_detector_->StartCheckForOveruse();
>-    video_sender_.RegisterExternalEncoder(
>-        settings_.encoder, settings_.payload_type, settings_.internal_source);
>-  });
>+  RTC_DCHECK(encoder_stats_observer);
>+  RTC_DCHECK(overuse_detector_);
> }
> 
> VideoStreamEncoder::~VideoStreamEncoder() {
>@@ -439,30 +367,15 @@ VideoStreamEncoder::~VideoStreamEncoder() {
>       << "Must call ::Stop() before destruction.";
> }
> 
>-// TODO(pbos): Lower these thresholds (to closer to 100%) when we handle
>-// pipelining encoders better (multiple input frames before something comes
>-// out). This should effectively turn off CPU adaptations for systems that
>-// remotely cope with the load right now.
>-CpuOveruseOptions VideoStreamEncoder::GetCpuOveruseOptions(
>-    bool full_overuse_time) {
>-  CpuOveruseOptions options;
>-  if (full_overuse_time) {
>-    options.low_encode_usage_threshold_percent = 150;
>-    options.high_encode_usage_threshold_percent = 200;
>-  }
>-  return options;
>-}
>-
> void VideoStreamEncoder::Stop() {
>   RTC_DCHECK_RUN_ON(&thread_checker_);
>-  source_proxy_->SetSource(nullptr, VideoSendStream::DegradationPreference());
>+  source_proxy_->SetSource(nullptr, DegradationPreference());
>   encoder_queue_.PostTask([this] {
>     RTC_DCHECK_RUN_ON(&encoder_queue_);
>     overuse_detector_->StopCheckForOveruse();
>     rate_allocator_.reset();
>     bitrate_observer_ = nullptr;
>-    video_sender_.RegisterExternalEncoder(nullptr, settings_.payload_type,
>-                                          false);
>+    video_sender_.RegisterExternalEncoder(nullptr, false);
>     quality_scaler_ = nullptr;
>     shutdown_event_.Set();
>   });
>@@ -470,7 +383,7 @@ void VideoStreamEncoder::Stop() {
>   shutdown_event_.Wait(rtc::Event::kForever);
> }
> 
>-void VideoStreamEncoder::SetBitrateObserver(
>+void VideoStreamEncoder::SetBitrateAllocationObserver(
>     VideoBitrateAllocationObserver* bitrate_observer) {
>   RTC_DCHECK_RUN_ON(&thread_checker_);
>   encoder_queue_.PostTask([this, bitrate_observer] {
>@@ -482,7 +395,7 @@ void VideoStreamEncoder::SetBitrateObserver(
> 
> void VideoStreamEncoder::SetSource(
>     rtc::VideoSourceInterface<VideoFrame>* source,
>-    const VideoSendStream::DegradationPreference& degradation_preference) {
>+    const DegradationPreference& degradation_preference) {
>   RTC_DCHECK_RUN_ON(&thread_checker_);
>   source_proxy_->SetSource(source, degradation_preference);
>   encoder_queue_.PostTask([this, degradation_preference] {
>@@ -491,10 +404,8 @@ void VideoStreamEncoder::SetSource(
>       // Reset adaptation state, so that we're not tricked into thinking there's
>       // an already pending request of the same type.
>       last_adaptation_request_.reset();
>-      if (degradation_preference ==
>-              VideoSendStream::DegradationPreference::kBalanced ||
>-          degradation_preference_ ==
>-              VideoSendStream::DegradationPreference::kBalanced) {
>+      if (degradation_preference == DegradationPreference::BALANCED ||
>+          degradation_preference_ == DegradationPreference::BALANCED) {
>         // TODO(asapersson): Consider removing |adapt_counters_| map and use one
>         // AdaptCounter for all modes.
>         source_proxy_->ResetPixelFpsCount();
>@@ -502,9 +413,10 @@ void VideoStreamEncoder::SetSource(
>       }
>     }
>     degradation_preference_ = degradation_preference;
>-    bool allow_scaling = IsResolutionScalingEnabled(degradation_preference_);
>-    initial_rampup_ = allow_scaling ? 0 : kMaxInitialFramedrop;
>-    ConfigureQualityScaler();
>+
>+    if (encoder_)
>+      ConfigureQualityScaler();
>+
>     if (!IsFramerateScalingEnabled(degradation_preference) &&
>         max_framerate_ != -1) {
>       // If frame rate scaling is no longer allowed, remove any potential
>@@ -530,23 +442,32 @@ void VideoStreamEncoder::SetStartBitrate(int start_bitrate_bps) {
> }
> 
> void VideoStreamEncoder::ConfigureEncoder(VideoEncoderConfig config,
>-                                          size_t max_data_payload_length,
>-                                          bool nack_enabled) {
>+                                          size_t max_data_payload_length) {
>+  // TODO(srte): This struct should be replaced by a lambda with move capture
>+  // when C++14 lambda is allowed.
>+  struct ConfigureEncoderTask {
>+    void operator()() {
>+      encoder->ConfigureEncoderOnTaskQueue(std::move(config),
>+                                           max_data_payload_length);
>+    }
>+    VideoStreamEncoder* encoder;
>+    VideoEncoderConfig config;
>+    size_t max_data_payload_length;
>+  };
>   encoder_queue_.PostTask(
>-      std::unique_ptr<rtc::QueuedTask>(new ConfigureEncoderTask(
>-          this, std::move(config), max_data_payload_length, nack_enabled)));
>+      ConfigureEncoderTask{this, std::move(config), max_data_payload_length});
> }
> 
> void VideoStreamEncoder::ConfigureEncoderOnTaskQueue(
>     VideoEncoderConfig config,
>-    size_t max_data_payload_length,
>-    bool nack_enabled) {
>+    size_t max_data_payload_length) {
>   RTC_DCHECK_RUN_ON(&encoder_queue_);
>   RTC_DCHECK(sink_);
>   RTC_LOG(LS_INFO) << "ConfigureEncoder requested.";
> 
>   max_data_payload_length_ = max_data_payload_length;
>-  nack_enabled_ = nack_enabled;
>+  pending_encoder_creation_ =
>+      (!encoder_ || encoder_config_.video_format != config.video_format);
>   encoder_config_ = std::move(config);
>   pending_encoder_reconfiguration_ = true;
> 
>@@ -556,14 +477,19 @@ void VideoStreamEncoder::ConfigureEncoderOnTaskQueue(
>   // The codec configuration depends on incoming video frame size.
>   if (last_frame_info_) {
>     ReconfigureEncoder();
>-  } else if (settings_.internal_source) {
>+  } else if (settings_.encoder_factory
>+                 ->QueryVideoEncoder(encoder_config_.video_format)
>+                 .has_internal_source) {
>     last_frame_info_ = VideoFrameInfo(176, 144, false);
>     ReconfigureEncoder();
>   }
> }
> 
>+// TODO(bugs.webrtc.org/8807): Currently this always does a hard
>+// reconfiguration, but this isn't always necessary. Add in logic to only update
>+// the VideoBitrateAllocator and call OnEncoderConfigurationChanged with a
>+// "soft" reconfiguration.
> void VideoStreamEncoder::ReconfigureEncoder() {
>-  RTC_DCHECK_RUN_ON(&encoder_queue_);
>   RTC_DCHECK(pending_encoder_reconfiguration_);
>   std::vector<VideoStream> streams =
>       encoder_config_.video_stream_factory->CreateEncoderStreams(
>@@ -584,12 +510,33 @@ void VideoStreamEncoder::ReconfigureEncoder() {
>   crop_height_ = last_frame_info_->height - highest_stream_height;
> 
>   VideoCodec codec;
>-  if (!VideoCodecInitializer::SetupCodec(encoder_config_, settings_, streams,
>-                                         nack_enabled_, &codec,
>+  if (!VideoCodecInitializer::SetupCodec(encoder_config_, streams, &codec,
>                                          &rate_allocator_)) {
>     RTC_LOG(LS_ERROR) << "Failed to create encoder configuration.";
>   }
> 
>+  // Set min_bitrate_bps, max_bitrate_bps, and max padding bit rate for VP9.
>+  if (encoder_config_.codec_type == kVideoCodecVP9) {
>+    RTC_DCHECK_EQ(1U, streams.size());
>+    int max_encoder_bitrate_kbps = 0;
>+    for (int i = 0; i < codec.VP9()->numberOfSpatialLayers; ++i) {
>+      max_encoder_bitrate_kbps += codec.spatialLayers[i].maxBitrate;
>+    }
>+    // Lower max bitrate to the level codec actually can produce.
>+    streams[0].max_bitrate_bps =
>+        std::min(streams[0].max_bitrate_bps, max_encoder_bitrate_kbps * 1000);
>+    streams[0].min_bitrate_bps = codec.spatialLayers[0].minBitrate * 1000;
>+    // Pass along the value of maximum padding bit rate from
>+    // spatialLayers[].targetBitrate to streams[0].target_bitrate_bps.
>+    // TODO(ssilkin): There should be some margin between max padding bitrate
>+    // and max encoder bitrate. With the current logic they can be equal.
>+    streams[0].target_bitrate_bps =
>+        std::min(static_cast<unsigned int>(streams[0].max_bitrate_bps),
>+                 codec.spatialLayers[codec.VP9()->numberOfSpatialLayers - 1]
>+                         .targetBitrate *
>+                     1000);
>+  }
>+
>   codec.startBitrate =
>       std::max(encoder_start_bitrate_bps_ / 1000, codec.minBitrate);
>   codec.startBitrate = std::min(codec.startBitrate, codec.maxBitrate);
>@@ -597,6 +544,32 @@ void VideoStreamEncoder::ReconfigureEncoder() {
>   max_framerate_ = codec.maxFramerate;
>   RTC_DCHECK_LE(max_framerate_, kMaxFramerateFps);
> 
>+  // Keep the same encoder, as long as the video_format is unchanged.
>+  if (pending_encoder_creation_) {
>+    pending_encoder_creation_ = false;
>+    if (encoder_) {
>+      video_sender_.RegisterExternalEncoder(nullptr, false);
>+    }
>+
>+    encoder_ = settings_.encoder_factory->CreateVideoEncoder(
>+        encoder_config_.video_format);
>+    // TODO(nisse): What to do if creating the encoder fails? Crash,
>+    // or just discard incoming frames?
>+    RTC_CHECK(encoder_);
>+
>+    const webrtc::VideoEncoderFactory::CodecInfo info =
>+        settings_.encoder_factory->QueryVideoEncoder(
>+            encoder_config_.video_format);
>+
>+    overuse_detector_->StopCheckForOveruse();
>+    overuse_detector_->StartCheckForOveruse(
>+        GetCpuOveruseOptions(settings_, info.is_hardware_accelerated), this);
>+
>+    video_sender_.RegisterExternalEncoder(encoder_.get(),
>+                                          info.has_internal_source);
>+  }
>+  // RegisterSendCodec implies an unconditional call to
>+  // encoder_->InitEncode().
>   bool success = video_sender_.RegisterSendCodec(
>                      &codec, number_of_cores_,
>                      static_cast<uint32_t>(max_data_payload_length_)) == VCM_OK;
>@@ -605,19 +578,10 @@ void VideoStreamEncoder::ReconfigureEncoder() {
>     rate_allocator_.reset();
>   }
> 
>-  video_sender_.UpdateChannelParemeters(rate_allocator_.get(),
>+  video_sender_.UpdateChannelParameters(rate_allocator_.get(),
>                                         bitrate_observer_);
> 
>-  // Get the current actual framerate, as measured by the stats proxy. This is
>-  // used to get the correct bitrate layer allocation.
>-  int current_framerate = stats_proxy_->GetSendFrameRate();
>-  if (current_framerate == 0)
>-    current_framerate = codec.maxFramerate;
>-  stats_proxy_->OnEncoderReconfigured(
>-      encoder_config_, streams,
>-      rate_allocator_.get()
>-          ? rate_allocator_->GetPreferredBitrateBps(current_framerate)
>-          : codec.maxBitrate);
>+  encoder_stats_observer_->OnEncoderReconfigured(encoder_config_, streams);
> 
>   pending_encoder_reconfiguration_ = false;
> 
>@@ -638,29 +602,38 @@ void VideoStreamEncoder::ReconfigureEncoder() {
> 
> void VideoStreamEncoder::ConfigureQualityScaler() {
>   RTC_DCHECK_RUN_ON(&encoder_queue_);
>-  const auto scaling_settings = settings_.encoder->GetScalingSettings();
>+  const auto scaling_settings = encoder_->GetScalingSettings();
>   const bool quality_scaling_allowed =
>       IsResolutionScalingEnabled(degradation_preference_) &&
>-      scaling_settings.enabled;
>+      scaling_settings.thresholds;
> 
>   if (quality_scaling_allowed) {
>     if (quality_scaler_.get() == nullptr) {
>       // Quality scaler has not already been configured.
>-      // Drop frames and scale down until desired quality is achieved.
>-      if (scaling_settings.thresholds) {
>-        quality_scaler_.reset(
>-            new QualityScaler(this, *(scaling_settings.thresholds)));
>-      } else {
>-        quality_scaler_.reset(new QualityScaler(this, codec_type_));
>+
>+      // Use experimental thresholds if available.
>+      absl::optional<VideoEncoder::QpThresholds> experimental_thresholds;
>+      if (quality_scaling_experiment_enabled_) {
>+        experimental_thresholds = QualityScalingExperiment::GetQpThresholds(
>+            encoder_config_.codec_type);
>       }
>+      // Since the interface is non-public, absl::make_unique can't do this
>+      // upcast.
>+      AdaptationObserverInterface* observer = this;
>+      quality_scaler_ = absl::make_unique<QualityScaler>(
>+          observer, experimental_thresholds ? *experimental_thresholds
>+                                            : *(scaling_settings.thresholds));
>+      has_seen_first_significant_bwe_change_ = false;
>+      initial_framedrop_ = 0;
>     }
>   } else {
>     quality_scaler_.reset(nullptr);
>-    initial_rampup_ = kMaxInitialFramedrop;
>+    initial_framedrop_ = kMaxInitialFramedrop;
>   }
> 
>-  stats_proxy_->SetAdaptationStats(GetActiveCounts(kCpu),
>-                                   GetActiveCounts(kQuality));
>+  encoder_stats_observer_->OnAdaptationChanged(
>+      VideoStreamEncoderObserver::AdaptationReason::kNone,
>+      GetActiveCounts(kCpu), GetActiveCounts(kQuality));
> }
> 
> void VideoStreamEncoder::OnFrame(const VideoFrame& video_frame) {
>@@ -709,12 +682,44 @@ void VideoStreamEncoder::OnFrame(const VideoFrame& video_frame) {
>   }
> 
>   last_captured_timestamp_ = incoming_frame.ntp_time_ms();
>-  encoder_queue_.PostTask(std::unique_ptr<rtc::QueuedTask>(new EncodeTask(
>-      incoming_frame, this, rtc::TimeMicros(), log_stats)));
>+
>+  int64_t post_time_us = rtc::TimeMicros();
>+  ++posted_frames_waiting_for_encode_;
>+
>+  encoder_queue_.PostTask(
>+      [this, incoming_frame, post_time_us, log_stats]() {
>+        RTC_DCHECK_RUN_ON(&encoder_queue_);
>+        encoder_stats_observer_->OnIncomingFrame(incoming_frame.width(),
>+                                                 incoming_frame.height());
>+        ++captured_frame_count_;
>+        const int posted_frames_waiting_for_encode =
>+            posted_frames_waiting_for_encode_.fetch_sub(1);
>+        RTC_DCHECK_GT(posted_frames_waiting_for_encode, 0);
>+        if (posted_frames_waiting_for_encode == 1) {
>+          MaybeEncodeVideoFrame(incoming_frame, post_time_us);
>+        } else {
>+          // There is a newer frame in flight. Do not encode this frame.
>+          RTC_LOG(LS_VERBOSE)
>+              << "Incoming frame dropped due to that the encoder is blocked.";
>+          ++dropped_frame_count_;
>+          encoder_stats_observer_->OnFrameDropped(
>+              VideoStreamEncoderObserver::DropReason::kEncoderQueue);
>+        }
>+        if (log_stats) {
>+          RTC_LOG(LS_INFO) << "Number of frames: captured "
>+                           << captured_frame_count_
>+                           << ", dropped (due to encoder blocked) "
>+                           << dropped_frame_count_ << ", interval_ms "
>+                           << kFrameLogIntervalMs;
>+          captured_frame_count_ = 0;
>+          dropped_frame_count_ = 0;
>+        }
>+      });
> }
> 
> void VideoStreamEncoder::OnDiscardedFrame() {
>-  stats_proxy_->OnFrameDroppedBySource();
>+  encoder_stats_observer_->OnFrameDropped(
>+      VideoStreamEncoderObserver::DropReason::kSource);
> }
> 
> bool VideoStreamEncoder::EncoderPaused() const {
>@@ -744,8 +749,8 @@ void VideoStreamEncoder::TraceFrameDropEnd() {
>   encoder_paused_and_dropped_frame_ = false;
> }
> 
>-void VideoStreamEncoder::EncodeVideoFrame(const VideoFrame& video_frame,
>-                                          int64_t time_when_posted_us) {
>+void VideoStreamEncoder::MaybeEncodeVideoFrame(const VideoFrame& video_frame,
>+                                               int64_t time_when_posted_us) {
>   RTC_DCHECK_RUN_ON(&encoder_queue_);
> 
>   if (pre_encode_callback_)
>@@ -763,19 +768,11 @@ void VideoStreamEncoder::EncodeVideoFrame(const VideoFrame& video_frame,
>                      << ", texture=" << last_frame_info_->is_texture << ".";
>   }
> 
>-  if (initial_rampup_ < kMaxInitialFramedrop &&
>-      video_frame.size() >
>-          MaximumFrameSizeForBitrate(encoder_start_bitrate_bps_ / 1000)) {
>-    RTC_LOG(LS_INFO) << "Dropping frame. Too large for target bitrate.";
>-    int count = GetConstAdaptCounter().ResolutionCount(kQuality);
>-    AdaptDown(kQuality);
>-    if (GetConstAdaptCounter().ResolutionCount(kQuality) > count) {
>-      stats_proxy_->OnInitialQualityResolutionAdaptDown();
>-    }
>-    ++initial_rampup_;
>-    return;
>-  }
>-  initial_rampup_ = kMaxInitialFramedrop;
>+  // We have to create then encoder before the frame drop logic,
>+  // because the latter depends on encoder_->GetScalingSettings.
>+  // According to the testcase
>+  // InitialFrameDropOffWhenEncoderDisabledScaling, the return value
>+  // from GetScalingSettings should enable or disable the frame drop.
> 
>   int64_t now_ms = clock_->TimeInMilliseconds();
>   if (pending_encoder_reconfiguration_) {
>@@ -784,15 +781,55 @@ void VideoStreamEncoder::EncodeVideoFrame(const VideoFrame& video_frame,
>   } else if (!last_parameters_update_ms_ ||
>              now_ms - *last_parameters_update_ms_ >=
>                  vcm::VCMProcessTimer::kDefaultProcessIntervalMs) {
>-    video_sender_.UpdateChannelParemeters(rate_allocator_.get(),
>+    video_sender_.UpdateChannelParameters(rate_allocator_.get(),
>                                           bitrate_observer_);
>     last_parameters_update_ms_.emplace(now_ms);
>   }
> 
>+  if (DropDueToSize(video_frame.size())) {
>+    RTC_LOG(LS_INFO) << "Dropping frame. Too large for target bitrate.";
>+    int count = GetConstAdaptCounter().ResolutionCount(kQuality);
>+    AdaptDown(kQuality);
>+    if (GetConstAdaptCounter().ResolutionCount(kQuality) > count) {
>+      encoder_stats_observer_->OnInitialQualityResolutionAdaptDown();
>+    }
>+    ++initial_framedrop_;
>+    // Storing references to a native buffer risks blocking frame capture.
>+    if (video_frame.video_frame_buffer()->type() !=
>+        VideoFrameBuffer::Type::kNative) {
>+      pending_frame_ = video_frame;
>+      pending_frame_post_time_us_ = time_when_posted_us;
>+    } else {
>+      // Ensure that any previously stored frame is dropped.
>+      pending_frame_.reset();
>+    }
>+    return;
>+  }
>+  initial_framedrop_ = kMaxInitialFramedrop;
>+
>   if (EncoderPaused()) {
>-    TraceFrameDropStart();
>+    // Storing references to a native buffer risks blocking frame capture.
>+    if (video_frame.video_frame_buffer()->type() !=
>+        VideoFrameBuffer::Type::kNative) {
>+      if (pending_frame_)
>+        TraceFrameDropStart();
>+      pending_frame_ = video_frame;
>+      pending_frame_post_time_us_ = time_when_posted_us;
>+    } else {
>+      // Ensure that any previously stored frame is dropped.
>+      pending_frame_.reset();
>+      TraceFrameDropStart();
>+    }
>     return;
>   }
>+
>+  pending_frame_.reset();
>+  EncodeVideoFrame(video_frame, time_when_posted_us);
>+}
>+
>+void VideoStreamEncoder::EncodeVideoFrame(const VideoFrame& video_frame,
>+                                          int64_t time_when_posted_us) {
>+  RTC_DCHECK_RUN_ON(&encoder_queue_);
>   TraceFrameDropEnd();
> 
>   VideoFrame out_frame(video_frame);
>@@ -832,6 +869,7 @@ void VideoStreamEncoder::SendKeyFrame() {
>     return;
>   }
>   RTC_DCHECK_RUN_ON(&encoder_queue_);
>+  TRACE_EVENT0("webrtc", "OnKeyFrameRequest");
>   video_sender_.IntraFrameRequest(0);
> }
> 
>@@ -842,7 +880,8 @@ EncodedImageCallback::Result VideoStreamEncoder::OnEncodedImage(
>   // Encoded is called on whatever thread the real encoder implementation run
>   // on. In the case of hardware encoders, there might be several encoders
>   // running in parallel on different threads.
>-  stats_proxy_->OnSendEncodedImage(encoded_image, codec_specific_info);
>+  encoder_stats_observer_->OnSendEncodedImage(encoded_image,
>+                                              codec_specific_info);
> 
>   EncodedImageCallback::Result result =
>       sink_->OnEncodedImage(encoded_image, codec_specific_info, fragmentation);
>@@ -853,8 +892,8 @@ EncodedImageCallback::Result VideoStreamEncoder::OnEncodedImage(
>   int64_t capture_time_us =
>       encoded_image.capture_time_ms_ * rtc::kNumMicrosecsPerMillisec;
> 
>-  rtc::Optional<int> encode_duration_us;
>-  if (encoded_image.timing_.flags != TimingFrameFlags::kInvalid) {
>+  absl::optional<int> encode_duration_us;
>+  if (encoded_image.timing_.flags != VideoSendTiming::kInvalid) {
>     encode_duration_us.emplace(
>         // TODO(nisse): Maybe use capture_time_ms_ rather than encode_start_ms_?
>         rtc::kNumMicrosecsPerMillisec *
>@@ -868,7 +907,7 @@ EncodedImageCallback::Result VideoStreamEncoder::OnEncodedImage(
>         overuse_detector_->FrameSent(timestamp, time_sent_us, capture_time_us,
>                                      encode_duration_us);
>         if (quality_scaler_ && qp >= 0)
>-          quality_scaler_->ReportQP(qp);
>+          quality_scaler_->ReportQp(qp);
>       });
> 
>   return result;
>@@ -877,31 +916,26 @@ EncodedImageCallback::Result VideoStreamEncoder::OnEncodedImage(
> void VideoStreamEncoder::OnDroppedFrame(DropReason reason) {
>   switch (reason) {
>     case DropReason::kDroppedByMediaOptimizations:
>-      stats_proxy_->OnFrameDroppedByMediaOptimizations();
>+      encoder_stats_observer_->OnFrameDropped(
>+          VideoStreamEncoderObserver::DropReason::kMediaOptimization);
>       encoder_queue_.PostTask([this] {
>         RTC_DCHECK_RUN_ON(&encoder_queue_);
>         if (quality_scaler_)
>-          quality_scaler_->ReportDroppedFrame();
>+          quality_scaler_->ReportDroppedFrameByMediaOpt();
>       });
>       break;
>     case DropReason::kDroppedByEncoder:
>-      stats_proxy_->OnFrameDroppedByEncoder();
>+      encoder_stats_observer_->OnFrameDropped(
>+          VideoStreamEncoderObserver::DropReason::kEncoder);
>+      encoder_queue_.PostTask([this] {
>+        RTC_DCHECK_RUN_ON(&encoder_queue_);
>+        if (quality_scaler_)
>+          quality_scaler_->ReportDroppedFrameByEncoder();
>+      });
>       break;
>   }
> }
> 
>-void VideoStreamEncoder::OnReceivedIntraFrameRequest(size_t stream_index) {
>-  if (!encoder_queue_.IsCurrent()) {
>-    encoder_queue_.PostTask(
>-        [this, stream_index] { OnReceivedIntraFrameRequest(stream_index); });
>-    return;
>-  }
>-  RTC_DCHECK_RUN_ON(&encoder_queue_);
>-  // Key frame request from remote side, signal to VCM.
>-  TRACE_EVENT0("webrtc", "OnKeyFrameRequest");
>-  video_sender_.IntraFrameRequest(stream_index);
>-}
>-
> void VideoStreamEncoder::OnBitrateUpdated(uint32_t bitrate_bps,
>                                           uint8_t fraction_lost,
>                                           int64_t round_trip_time_ms) {
>@@ -918,6 +952,19 @@ void VideoStreamEncoder::OnBitrateUpdated(uint32_t bitrate_bps,
>   RTC_LOG(LS_VERBOSE) << "OnBitrateUpdated, bitrate " << bitrate_bps
>                       << " packet loss " << static_cast<int>(fraction_lost)
>                       << " rtt " << round_trip_time_ms;
>+  // On significant changes to BWE at the start of the call,
>+  // enable frame drops to quickly react to jumps in available bandwidth.
>+  if (encoder_start_bitrate_bps_ != 0 &&
>+      !has_seen_first_significant_bwe_change_ && quality_scaler_ &&
>+      initial_framedrop_on_bwe_enabled_ &&
>+      abs_diff(bitrate_bps, encoder_start_bitrate_bps_) >=
>+          kFramedropThreshold * encoder_start_bitrate_bps_) {
>+    // Reset initial framedrop feature when first real BW estimate arrives.
>+    // TODO(kthelgason): Update BitrateAllocator to not call OnBitrateUpdated
>+    // without an actual BW estimate.
>+    initial_framedrop_ = 0;
>+    has_seen_first_significant_bwe_change_ = true;
>+  }
> 
>   video_sender_.SetChannelParameters(bitrate_bps, fraction_lost,
>                                      round_trip_time_ms, rate_allocator_.get(),
>@@ -932,15 +979,34 @@ void VideoStreamEncoder::OnBitrateUpdated(uint32_t bitrate_bps,
>   if (video_suspension_changed) {
>     RTC_LOG(LS_INFO) << "Video suspend state changed to: "
>                      << (video_is_suspended ? "suspended" : "not suspended");
>-    stats_proxy_->OnSuspendChange(video_is_suspended);
>+    encoder_stats_observer_->OnSuspendChange(video_is_suspended);
>+  }
>+  if (video_suspension_changed && !video_is_suspended && pending_frame_ &&
>+      !DropDueToSize(pending_frame_->size())) {
>+    int64_t pending_time_us = rtc::TimeMicros() - pending_frame_post_time_us_;
>+    if (pending_time_us < kPendingFrameTimeoutMs * 1000)
>+      EncodeVideoFrame(*pending_frame_, pending_frame_post_time_us_);
>+    pending_frame_.reset();
>+  }
>+}
>+
>+bool VideoStreamEncoder::DropDueToSize(uint32_t pixel_count) const {
>+  if (initial_framedrop_ < kMaxInitialFramedrop &&
>+      encoder_start_bitrate_bps_ > 0) {
>+    if (encoder_start_bitrate_bps_ < 300000 /* qvga */) {
>+      return pixel_count > 320 * 240;
>+    } else if (encoder_start_bitrate_bps_ < 500000 /* vga */) {
>+      return pixel_count > 640 * 480;
>+    }
>   }
>+  return false;
> }
> 
> void VideoStreamEncoder::AdaptDown(AdaptReason reason) {
>   RTC_DCHECK_RUN_ON(&encoder_queue_);
>   AdaptationRequest adaptation_request = {
>       last_frame_info_->pixel_count(),
>-      stats_proxy_->GetStats().input_frame_rate,
>+      encoder_stats_observer_->GetInputFrameRate(),
>       AdaptationRequest::Mode::kAdaptDown};
> 
>   bool downgrade_requested =
>@@ -948,9 +1014,9 @@ void VideoStreamEncoder::AdaptDown(AdaptReason reason) {
>       last_adaptation_request_->mode_ == AdaptationRequest::Mode::kAdaptDown;
> 
>   switch (degradation_preference_) {
>-    case VideoSendStream::DegradationPreference::kBalanced:
>+    case DegradationPreference::BALANCED:
>       break;
>-    case VideoSendStream::DegradationPreference::kMaintainFramerate:
>+    case DegradationPreference::MAINTAIN_FRAMERATE:
>       if (downgrade_requested &&
>           adaptation_request.input_pixel_count_ >=
>               last_adaptation_request_->input_pixel_count_) {
>@@ -959,7 +1025,7 @@ void VideoStreamEncoder::AdaptDown(AdaptReason reason) {
>         return;
>       }
>       break;
>-    case VideoSendStream::DegradationPreference::kMaintainResolution:
>+    case DegradationPreference::MAINTAIN_RESOLUTION:
>       if (adaptation_request.framerate_fps_ <= 0 ||
>           (downgrade_requested &&
>            adaptation_request.framerate_fps_ < kMinFramerateFps)) {
>@@ -972,12 +1038,12 @@ void VideoStreamEncoder::AdaptDown(AdaptReason reason) {
>         return;
>       }
>       break;
>-    case VideoSendStream::DegradationPreference::kDegradationDisabled:
>+    case DegradationPreference::DISABLED:
>       return;
>   }
> 
>   switch (degradation_preference_) {
>-    case VideoSendStream::DegradationPreference::kBalanced: {
>+    case DegradationPreference::BALANCED: {
>       // Try scale down framerate, if lower.
>       int fps = MinFps(last_frame_info_->pixel_count());
>       if (source_proxy_->RestrictFramerate(fps)) {
>@@ -985,23 +1051,23 @@ void VideoStreamEncoder::AdaptDown(AdaptReason reason) {
>         break;
>       }
>       // Scale down resolution.
>-      FALLTHROUGH();
>+      RTC_FALLTHROUGH();
>     }
>-    case VideoSendStream::DegradationPreference::kMaintainFramerate: {
>+    case DegradationPreference::MAINTAIN_FRAMERATE: {
>       // Scale down resolution.
>       bool min_pixels_reached = false;
>       if (!source_proxy_->RequestResolutionLowerThan(
>               adaptation_request.input_pixel_count_,
>-              settings_.encoder->GetScalingSettings().min_pixels_per_frame,
>+              encoder_->GetScalingSettings().min_pixels_per_frame,
>               &min_pixels_reached)) {
>         if (min_pixels_reached)
>-          stats_proxy_->OnMinPixelLimitReached();
>+          encoder_stats_observer_->OnMinPixelLimitReached();
>         return;
>       }
>       GetAdaptCounter().IncrementResolution(reason);
>       break;
>     }
>-    case VideoSendStream::DegradationPreference::kMaintainResolution: {
>+    case DegradationPreference::MAINTAIN_RESOLUTION: {
>       // Scale down framerate.
>       const int requested_framerate = source_proxy_->RequestFramerateLowerThan(
>           adaptation_request.framerate_fps_);
>@@ -1013,7 +1079,7 @@ void VideoStreamEncoder::AdaptDown(AdaptReason reason) {
>       GetAdaptCounter().IncrementFramerate(reason);
>       break;
>     }
>-    case VideoSendStream::DegradationPreference::kDegradationDisabled:
>+    case DegradationPreference::DISABLED:
>       RTC_NOTREACHED();
>   }
> 
>@@ -1035,15 +1101,14 @@ void VideoStreamEncoder::AdaptUp(AdaptReason reason) {
> 
>   AdaptationRequest adaptation_request = {
>       last_frame_info_->pixel_count(),
>-      stats_proxy_->GetStats().input_frame_rate,
>+      encoder_stats_observer_->GetInputFrameRate(),
>       AdaptationRequest::Mode::kAdaptUp};
> 
>   bool adapt_up_requested =
>       last_adaptation_request_ &&
>       last_adaptation_request_->mode_ == AdaptationRequest::Mode::kAdaptUp;
> 
>-  if (degradation_preference_ ==
>-      VideoSendStream::DegradationPreference::kMaintainFramerate) {
>+  if (degradation_preference_ == DegradationPreference::MAINTAIN_FRAMERATE) {
>     if (adapt_up_requested &&
>         adaptation_request.input_pixel_count_ <=
>             last_adaptation_request_->input_pixel_count_) {
>@@ -1054,7 +1119,7 @@ void VideoStreamEncoder::AdaptUp(AdaptReason reason) {
>   }
> 
>   switch (degradation_preference_) {
>-    case VideoSendStream::DegradationPreference::kBalanced: {
>+    case DegradationPreference::BALANCED: {
>       // Try scale up framerate, if higher.
>       int fps = MaxFps(last_frame_info_->pixel_count());
>       if (source_proxy_->IncreaseFramerate(fps)) {
>@@ -1068,9 +1133,9 @@ void VideoStreamEncoder::AdaptUp(AdaptReason reason) {
>         break;
>       }
>       // Scale up resolution.
>-      FALLTHROUGH();
>+      RTC_FALLTHROUGH();
>     }
>-    case VideoSendStream::DegradationPreference::kMaintainFramerate: {
>+    case DegradationPreference::MAINTAIN_FRAMERATE: {
>       // Scale up resolution.
>       int pixel_count = adaptation_request.input_pixel_count_;
>       if (adapt_counter.ResolutionCount() == 1) {
>@@ -1082,7 +1147,7 @@ void VideoStreamEncoder::AdaptUp(AdaptReason reason) {
>       GetAdaptCounter().DecrementResolution(reason);
>       break;
>     }
>-    case VideoSendStream::DegradationPreference::kMaintainResolution: {
>+    case DegradationPreference::MAINTAIN_RESOLUTION: {
>       // Scale up framerate.
>       int fps = adaptation_request.framerate_fps_;
>       if (adapt_counter.FramerateCount() == 1) {
>@@ -1101,7 +1166,7 @@ void VideoStreamEncoder::AdaptUp(AdaptReason reason) {
>       GetAdaptCounter().DecrementFramerate(reason);
>       break;
>     }
>-    case VideoSendStream::DegradationPreference::kDegradationDisabled:
>+    case DegradationPreference::DISABLED:
>       return;
>   }
> 
>@@ -1112,38 +1177,41 @@ void VideoStreamEncoder::AdaptUp(AdaptReason reason) {
>   RTC_LOG(LS_INFO) << adapt_counter.ToString();
> }
> 
>+// TODO(nisse): Delete, once AdaptReason and AdaptationReason are merged.
> void VideoStreamEncoder::UpdateAdaptationStats(AdaptReason reason) {
>   switch (reason) {
>     case kCpu:
>-      stats_proxy_->OnCpuAdaptationChanged(GetActiveCounts(kCpu),
>-                                           GetActiveCounts(kQuality));
>+      encoder_stats_observer_->OnAdaptationChanged(
>+          VideoStreamEncoderObserver::AdaptationReason::kCpu,
>+          GetActiveCounts(kCpu), GetActiveCounts(kQuality));
>       break;
>     case kQuality:
>-      stats_proxy_->OnQualityAdaptationChanged(GetActiveCounts(kCpu),
>-                                               GetActiveCounts(kQuality));
>+      encoder_stats_observer_->OnAdaptationChanged(
>+          VideoStreamEncoderObserver::AdaptationReason::kQuality,
>+          GetActiveCounts(kCpu), GetActiveCounts(kQuality));
>       break;
>   }
> }
> 
>-VideoStreamEncoder::AdaptCounts VideoStreamEncoder::GetActiveCounts(
>+VideoStreamEncoderObserver::AdaptationSteps VideoStreamEncoder::GetActiveCounts(
>     AdaptReason reason) {
>-  VideoStreamEncoder::AdaptCounts counts =
>+  VideoStreamEncoderObserver::AdaptationSteps counts =
>       GetConstAdaptCounter().Counts(reason);
>   switch (reason) {
>     case kCpu:
>       if (!IsFramerateScalingEnabled(degradation_preference_))
>-        counts.fps = -1;
>+        counts.num_framerate_reductions = absl::nullopt;
>       if (!IsResolutionScalingEnabled(degradation_preference_))
>-        counts.resolution = -1;
>+        counts.num_resolution_reductions = absl::nullopt;
>       break;
>     case kQuality:
>       if (!IsFramerateScalingEnabled(degradation_preference_) ||
>           !quality_scaler_) {
>-        counts.fps = -1;
>+        counts.num_framerate_reductions = absl::nullopt;
>       }
>       if (!IsResolutionScalingEnabled(degradation_preference_) ||
>           !quality_scaler_) {
>-        counts.resolution = -1;
>+        counts.num_resolution_reductions = absl::nullopt;
>       }
>       break;
>   }
>@@ -1175,11 +1243,11 @@ std::string VideoStreamEncoder::AdaptCounter::ToString() const {
>   return ss.str();
> }
> 
>-VideoStreamEncoder::AdaptCounts VideoStreamEncoder::AdaptCounter::Counts(
>-    int reason) const {
>-  AdaptCounts counts;
>-  counts.fps = fps_counters_[reason];
>-  counts.resolution = resolution_counters_[reason];
>+VideoStreamEncoderObserver::AdaptationSteps
>+VideoStreamEncoder::AdaptCounter::Counts(int reason) const {
>+  VideoStreamEncoderObserver::AdaptationSteps counts;
>+  counts.num_framerate_reductions = fps_counters_[reason];
>+  counts.num_resolution_reductions = resolution_counters_[reason];
>   return counts;
> }
> 
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_stream_encoder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_stream_encoder.h
>index cbf4787ab51..4008426bac9 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_stream_encoder.h
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_stream_encoder.h
>@@ -17,28 +17,23 @@
> #include <string>
> #include <vector>
> 
>+#include "api/video/video_bitrate_allocator.h"
> #include "api/video/video_rotation.h"
>+#include "api/video/video_sink_interface.h"
>+#include "api/video/video_stream_encoder_interface.h"
>+#include "api/video/video_stream_encoder_observer.h"
>+#include "api/video/video_stream_encoder_settings.h"
> #include "api/video_codecs/video_encoder.h"
>-#include "api/videosinkinterface.h"
>-#include "call/call.h"
>-#include "common_types.h"  // NOLINT(build/include)
>-#include "common_video/include/video_bitrate_allocator.h"
>-#include "modules/video_coding/include/video_coding_defines.h"
> #include "modules/video_coding/utility/quality_scaler.h"
> #include "modules/video_coding/video_coding_impl.h"
> #include "rtc_base/criticalsection.h"
> #include "rtc_base/event.h"
> #include "rtc_base/sequenced_task_checker.h"
> #include "rtc_base/task_queue.h"
>-#include "typedefs.h"  // NOLINT(build/include)
> #include "video/overuse_frame_detector.h"
>-#include "call/video_send_stream.h"
> 
> namespace webrtc {
> 
>-class SendStatisticsProxy;
>-class VideoBitrateAllocationObserver;
>-
> // VideoStreamEncoder represent a video encoder that accepts raw video frames as
> // input and produces an encoded bit stream.
> // Usage:
>@@ -47,65 +42,41 @@ class VideoBitrateAllocationObserver;
> //  Call SetSource.
> //  Call ConfigureEncoder with the codec settings.
> //  Call Stop() when done.
>-class VideoStreamEncoder : public rtc::VideoSinkInterface<VideoFrame>,
>-                           public EncodedImageCallback,
>-                           public AdaptationObserverInterface {
>+class VideoStreamEncoder : public VideoStreamEncoderInterface,
>+                           private EncodedImageCallback,
>+                           // Protected only to provide access to tests.
>+                           protected AdaptationObserverInterface {
>  public:
>-  // Interface for receiving encoded video frames and notifications about
>-  // configuration changes.
>-  class EncoderSink : public EncodedImageCallback {
>-   public:
>-    virtual void OnEncoderConfigurationChanged(
>-        std::vector<VideoStream> streams,
>-        int min_transmit_bitrate_bps) = 0;
>-  };
>-
>-  // Number of resolution and framerate reductions (-1: disabled).
>-  struct AdaptCounts {
>-    int resolution = 0;
>-    int fps = 0;
>-  };
>-
>   VideoStreamEncoder(uint32_t number_of_cores,
>-                     SendStatisticsProxy* stats_proxy,
>-                     const VideoSendStream::Config::EncoderSettings& settings,
>+                     VideoStreamEncoderObserver* encoder_stats_observer,
>+                     const VideoStreamEncoderSettings& settings,
>                      rtc::VideoSinkInterface<VideoFrame>* pre_encode_callback,
>                      std::unique_ptr<OveruseFrameDetector> overuse_detector);
>-  ~VideoStreamEncoder();
>+  ~VideoStreamEncoder() override;
> 
>-  // Sets the source that will provide I420 video frames.
>-  // |degradation_preference| control whether or not resolution or frame rate
>-  // may be reduced.
>-  void SetSource(
>-      rtc::VideoSourceInterface<VideoFrame>* source,
>-      const VideoSendStream::DegradationPreference& degradation_preference);
>+  void SetSource(rtc::VideoSourceInterface<VideoFrame>* source,
>+                 const DegradationPreference& degradation_preference) override;
> 
>-  // Sets the |sink| that gets the encoded frames. |rotation_applied| means
>-  // that the source must support rotation. Only set |rotation_applied| if the
>-  // remote side does not support the rotation extension.
>-  void SetSink(EncoderSink* sink, bool rotation_applied);
>+  void SetSink(EncoderSink* sink, bool rotation_applied) override;
> 
>   // TODO(perkj): Can we remove VideoCodec.startBitrate ?
>-  void SetStartBitrate(int start_bitrate_bps);
>+  void SetStartBitrate(int start_bitrate_bps) override;
> 
>-  void SetBitrateObserver(VideoBitrateAllocationObserver* bitrate_observer);
>+  void SetBitrateAllocationObserver(
>+      VideoBitrateAllocationObserver* bitrate_observer) override;
> 
>   void ConfigureEncoder(VideoEncoderConfig config,
>-                        size_t max_data_payload_length,
>-                        bool nack_enabled);
>+                        size_t max_data_payload_length) override;
> 
>   // Permanently stop encoding. After this method has returned, it is
>   // guaranteed that no encoded frames will be delivered to the sink.
>-  void Stop();
>-
>-  void SendKeyFrame();
>+  void Stop() override;
> 
>-  // virtual to test EncoderStateFeedback with mocks.
>-  virtual void OnReceivedIntraFrameRequest(size_t stream_index);
>+  void SendKeyFrame() override;
> 
>   void OnBitrateUpdated(uint32_t bitrate_bps,
>                         uint8_t fraction_lost,
>-                        int64_t round_trip_time_ms);
>+                        int64_t round_trip_time_ms) override;
> 
>  protected:
>   // Used for testing. For example the |ScalingObserverInterface| methods must
>@@ -116,21 +87,14 @@ class VideoStreamEncoder : public rtc::VideoSinkInterface<VideoFrame>,
>   // These methods are protected for easier testing.
>   void AdaptUp(AdaptReason reason) override;
>   void AdaptDown(AdaptReason reason) override;
>-  static CpuOveruseOptions GetCpuOveruseOptions(bool full_overuse_time);
> 
>  private:
>-  class ConfigureEncoderTask;
>-  class EncodeTask;
>   class VideoSourceProxy;
> 
>   class VideoFrameInfo {
>    public:
>-    VideoFrameInfo(int width,
>-                   int height,
>-                   bool is_texture)
>-        : width(width),
>-          height(height),
>-          is_texture(is_texture) {}
>+    VideoFrameInfo(int width, int height, bool is_texture)
>+        : width(width), height(height), is_texture(is_texture) {}
>     int width;
>     int height;
>     bool is_texture;
>@@ -138,9 +102,8 @@ class VideoStreamEncoder : public rtc::VideoSinkInterface<VideoFrame>,
>   };
> 
>   void ConfigureEncoderOnTaskQueue(VideoEncoderConfig config,
>-                                   size_t max_data_payload_length,
>-                                   bool nack_enabled);
>-  void ReconfigureEncoder();
>+                                   size_t max_data_payload_length);
>+  void ReconfigureEncoder() RTC_RUN_ON(&encoder_queue_);
> 
>   void ConfigureQualityScaler();
> 
>@@ -148,8 +111,14 @@ class VideoStreamEncoder : public rtc::VideoSinkInterface<VideoFrame>,
>   void OnFrame(const VideoFrame& video_frame) override;
>   void OnDiscardedFrame() override;
> 
>+  void MaybeEncodeVideoFrame(const VideoFrame& frame,
>+                             int64_t time_when_posted_in_ms);
>+
>   void EncodeVideoFrame(const VideoFrame& frame,
>                         int64_t time_when_posted_in_ms);
>+  // Indicates wether frame should be dropped because the pixel count is too
>+  // large for the current bitrate configuration.
>+  bool DropDueToSize(uint32_t pixel_count) const RTC_RUN_ON(&encoder_queue_);
> 
>   // Implements EncodedImageCallback.
>   EncodedImageCallback::Result OnEncodedImage(
>@@ -170,7 +139,7 @@ class VideoStreamEncoder : public rtc::VideoSinkInterface<VideoFrame>,
>     ~AdaptCounter();
> 
>     // Get number of adaptation downscales for |reason|.
>-    AdaptCounts Counts(int reason) const;
>+    VideoStreamEncoderObserver::AdaptationSteps Counts(int reason) const;
> 
>     std::string ToString() const;
> 
>@@ -203,59 +172,68 @@ class VideoStreamEncoder : public rtc::VideoSinkInterface<VideoFrame>,
>   AdaptCounter& GetAdaptCounter() RTC_RUN_ON(&encoder_queue_);
>   const AdaptCounter& GetConstAdaptCounter() RTC_RUN_ON(&encoder_queue_);
>   void UpdateAdaptationStats(AdaptReason reason) RTC_RUN_ON(&encoder_queue_);
>-  AdaptCounts GetActiveCounts(AdaptReason reason) RTC_RUN_ON(&encoder_queue_);
>+  VideoStreamEncoderObserver::AdaptationSteps GetActiveCounts(
>+      AdaptReason reason) RTC_RUN_ON(&encoder_queue_);
> 
>   rtc::Event shutdown_event_;
> 
>   const uint32_t number_of_cores_;
>-  // Counts how many frames we've dropped in the initial rampup phase.
>-  int initial_rampup_;
>+  // Counts how many frames we've dropped in the initial framedrop phase.
>+  int initial_framedrop_;
>+  const bool initial_framedrop_on_bwe_enabled_;
>+  bool has_seen_first_significant_bwe_change_ = false;
>+
>+  const bool quality_scaling_experiment_enabled_;
> 
>   const std::unique_ptr<VideoSourceProxy> source_proxy_;
>   EncoderSink* sink_;
>-  const VideoSendStream::Config::EncoderSettings settings_;
>-  const VideoCodecType codec_type_;
>+  const VideoStreamEncoderSettings settings_;
> 
>-  vcm::VideoSender video_sender_ RTC_ACCESS_ON(&encoder_queue_);
>-  std::unique_ptr<OveruseFrameDetector> overuse_detector_
>-      RTC_ACCESS_ON(&encoder_queue_);
>-  std::unique_ptr<QualityScaler> quality_scaler_ RTC_ACCESS_ON(&encoder_queue_);
>+  vcm::VideoSender video_sender_ RTC_GUARDED_BY(&encoder_queue_);
>+  const std::unique_ptr<OveruseFrameDetector> overuse_detector_
>+      RTC_PT_GUARDED_BY(&encoder_queue_);
>+  std::unique_ptr<QualityScaler> quality_scaler_ RTC_GUARDED_BY(&encoder_queue_)
>+      RTC_PT_GUARDED_BY(&encoder_queue_);
> 
>-  SendStatisticsProxy* const stats_proxy_;
>+  VideoStreamEncoderObserver* const encoder_stats_observer_;
>   rtc::VideoSinkInterface<VideoFrame>* const pre_encode_callback_;
>   // |thread_checker_| checks that public methods that are related to lifetime
>   // of VideoStreamEncoder are called on the same thread.
>   rtc::ThreadChecker thread_checker_;
> 
>-  VideoEncoderConfig encoder_config_ RTC_ACCESS_ON(&encoder_queue_);
>+  VideoEncoderConfig encoder_config_ RTC_GUARDED_BY(&encoder_queue_);
>+  std::unique_ptr<VideoEncoder> encoder_ RTC_GUARDED_BY(&encoder_queue_)
>+      RTC_PT_GUARDED_BY(&encoder_queue_);
>   std::unique_ptr<VideoBitrateAllocator> rate_allocator_
>-      RTC_ACCESS_ON(&encoder_queue_);
>+      RTC_GUARDED_BY(&encoder_queue_) RTC_PT_GUARDED_BY(&encoder_queue_);
>   // The maximum frame rate of the current codec configuration, as determined
>   // at the last ReconfigureEncoder() call.
>-  int max_framerate_ RTC_ACCESS_ON(&encoder_queue_);
>+  int max_framerate_ RTC_GUARDED_BY(&encoder_queue_);
> 
>   // Set when ConfigureEncoder has been called in order to lazy reconfigure the
>   // encoder on the next frame.
>-  bool pending_encoder_reconfiguration_ RTC_ACCESS_ON(&encoder_queue_);
>-  rtc::Optional<VideoFrameInfo> last_frame_info_ RTC_ACCESS_ON(&encoder_queue_);
>-  int crop_width_ RTC_ACCESS_ON(&encoder_queue_);
>-  int crop_height_ RTC_ACCESS_ON(&encoder_queue_);
>-  uint32_t encoder_start_bitrate_bps_ RTC_ACCESS_ON(&encoder_queue_);
>-  size_t max_data_payload_length_ RTC_ACCESS_ON(&encoder_queue_);
>-  bool nack_enabled_ RTC_ACCESS_ON(&encoder_queue_);
>-  uint32_t last_observed_bitrate_bps_ RTC_ACCESS_ON(&encoder_queue_);
>-  bool encoder_paused_and_dropped_frame_ RTC_ACCESS_ON(&encoder_queue_);
>+  bool pending_encoder_reconfiguration_ RTC_GUARDED_BY(&encoder_queue_);
>+  // Set when configuration must create a new encoder object, e.g.,
>+  // because of a codec change.
>+  bool pending_encoder_creation_ RTC_GUARDED_BY(&encoder_queue_);
>+  absl::optional<VideoFrameInfo> last_frame_info_
>+      RTC_GUARDED_BY(&encoder_queue_);
>+  int crop_width_ RTC_GUARDED_BY(&encoder_queue_);
>+  int crop_height_ RTC_GUARDED_BY(&encoder_queue_);
>+  uint32_t encoder_start_bitrate_bps_ RTC_GUARDED_BY(&encoder_queue_);
>+  size_t max_data_payload_length_ RTC_GUARDED_BY(&encoder_queue_);
>+  uint32_t last_observed_bitrate_bps_ RTC_GUARDED_BY(&encoder_queue_);
>+  bool encoder_paused_and_dropped_frame_ RTC_GUARDED_BY(&encoder_queue_);
>   Clock* const clock_;
>   // Counters used for deciding if the video resolution or framerate is
>   // currently restricted, and if so, why, on a per degradation preference
>   // basis.
>   // TODO(sprang): Replace this with a state holding a relative overuse measure
>   // instead, that can be translated into suitable down-scale or fps limit.
>-  std::map<const VideoSendStream::DegradationPreference, AdaptCounter>
>-      adapt_counters_ RTC_ACCESS_ON(&encoder_queue_);
>+  std::map<const DegradationPreference, AdaptCounter> adapt_counters_
>+      RTC_GUARDED_BY(&encoder_queue_);
>   // Set depending on degradation preferences.
>-  VideoSendStream::DegradationPreference degradation_preference_
>-      RTC_ACCESS_ON(&encoder_queue_);
>+  DegradationPreference degradation_preference_ RTC_GUARDED_BY(&encoder_queue_);
> 
>   struct AdaptationRequest {
>     // The pixel count produced by the source at the time of the adaptation.
>@@ -267,8 +245,8 @@ class VideoStreamEncoder : public rtc::VideoSinkInterface<VideoFrame>,
>   };
>   // Stores a snapshot of the last adaptation request triggered by an AdaptUp
>   // or AdaptDown signal.
>-  rtc::Optional<AdaptationRequest> last_adaptation_request_
>-      RTC_ACCESS_ON(&encoder_queue_);
>+  absl::optional<AdaptationRequest> last_adaptation_request_
>+      RTC_GUARDED_BY(&encoder_queue_);
> 
>   rtc::RaceChecker incoming_frame_race_checker_
>       RTC_GUARDED_BY(incoming_frame_race_checker_);
>@@ -280,13 +258,15 @@ class VideoStreamEncoder : public rtc::VideoSinkInterface<VideoFrame>,
>       RTC_GUARDED_BY(incoming_frame_race_checker_);
> 
>   int64_t last_frame_log_ms_ RTC_GUARDED_BY(incoming_frame_race_checker_);
>-  int captured_frame_count_ RTC_ACCESS_ON(&encoder_queue_);
>-  int dropped_frame_count_ RTC_ACCESS_ON(&encoder_queue_);
>+  int captured_frame_count_ RTC_GUARDED_BY(&encoder_queue_);
>+  int dropped_frame_count_ RTC_GUARDED_BY(&encoder_queue_);
>+  absl::optional<VideoFrame> pending_frame_ RTC_GUARDED_BY(&encoder_queue_);
>+  int64_t pending_frame_post_time_us_ RTC_GUARDED_BY(&encoder_queue_);
> 
>   VideoBitrateAllocationObserver* bitrate_observer_
>-      RTC_ACCESS_ON(&encoder_queue_);
>-  rtc::Optional<int64_t> last_parameters_update_ms_
>-      RTC_ACCESS_ON(&encoder_queue_);
>+      RTC_GUARDED_BY(&encoder_queue_);
>+  absl::optional<int64_t> last_parameters_update_ms_
>+      RTC_GUARDED_BY(&encoder_queue_);
> 
>   // All public methods are proxied to |encoder_queue_|. It must must be
>   // destroyed first to make sure no tasks are run that use other members.
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_stream_encoder_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_stream_encoder_unittest.cc
>index afeef02b66e..cda8a05edcc 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_stream_encoder_unittest.cc
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/video/video_stream_encoder_unittest.cc
>@@ -15,13 +15,17 @@
> #include "api/video/i420_buffer.h"
> #include "media/base/videoadapter.h"
> #include "modules/video_coding/codecs/vp8/temporal_layers.h"
>+#include "modules/video_coding/codecs/vp9/include/vp9_globals.h"
> #include "modules/video_coding/utility/default_video_bitrate_allocator.h"
> #include "rtc_base/fakeclock.h"
> #include "rtc_base/logging.h"
>+#include "rtc_base/refcountedobject.h"
> #include "system_wrappers/include/metrics_default.h"
> #include "system_wrappers/include/sleep.h"
>+#include "test/encoder_proxy_factory.h"
> #include "test/encoder_settings.h"
> #include "test/fake_encoder.h"
>+#include "test/field_trial.h"
> #include "test/frame_generator.h"
> #include "test/gmock.h"
> #include "test/gtest.h"
>@@ -33,12 +37,10 @@ const int kMinPixelsPerFrame = 320 * 180;
> const int kMinFramerateFps = 2;
> const int kMinBalancedFramerateFps = 7;
> const int64_t kFrameTimeoutMs = 100;
>-const unsigned char kNumSlDummy = 0;
> }  // namespace
> 
> namespace webrtc {
> 
>-using DegredationPreference = VideoSendStream::DegradationPreference;
> using ScaleReason = AdaptationObserverInterface::AdaptReason;
> using ::testing::_;
> using ::testing::Return;
>@@ -66,12 +68,8 @@ class TestBuffer : public webrtc::I420Buffer {
> 
> class CpuOveruseDetectorProxy : public OveruseFrameDetector {
>  public:
>-  CpuOveruseDetectorProxy(const CpuOveruseOptions& options,
>-                          AdaptationObserverInterface* overuse_observer,
>-                          CpuOveruseMetricsObserver* metrics_observer)
>-      : OveruseFrameDetector(options,
>-                             overuse_observer,
>-                             metrics_observer),
>+  explicit CpuOveruseDetectorProxy(CpuOveruseMetricsObserver* metrics_observer)
>+      : OveruseFrameDetector(metrics_observer),
>         last_target_framerate_fps_(-1) {}
>   virtual ~CpuOveruseDetectorProxy() {}
> 
>@@ -86,6 +84,8 @@ class CpuOveruseDetectorProxy : public OveruseFrameDetector {
>     return last_target_framerate_fps_;
>   }
> 
>+  CpuOveruseOptions GetOptions() { return options_; }
>+
>  private:
>   rtc::CriticalSection lock_;
>   int last_target_framerate_fps_ RTC_GUARDED_BY(lock_);
>@@ -94,17 +94,14 @@ class CpuOveruseDetectorProxy : public OveruseFrameDetector {
> class VideoStreamEncoderUnderTest : public VideoStreamEncoder {
>  public:
>   VideoStreamEncoderUnderTest(SendStatisticsProxy* stats_proxy,
>-                      const VideoSendStream::Config::EncoderSettings& settings)
>-      : VideoStreamEncoder(
>-            1 /* number_of_cores */,
>-            stats_proxy,
>-            settings,
>-            nullptr /* pre_encode_callback */,
>-            std::unique_ptr<OveruseFrameDetector>(
>-                overuse_detector_proxy_ = new CpuOveruseDetectorProxy(
>-                    GetCpuOveruseOptions(settings.full_overuse_time),
>-                    this,
>-                    stats_proxy))) {}
>+                              const VideoStreamEncoderSettings& settings)
>+      : VideoStreamEncoder(1 /* number_of_cores */,
>+                           stats_proxy,
>+                           settings,
>+                           nullptr /* pre_encode_callback */,
>+                           std::unique_ptr<OveruseFrameDetector>(
>+                               overuse_detector_proxy_ =
>+                                   new CpuOveruseDetectorProxy(stats_proxy))) {}
> 
>   void PostTaskAndWait(bool down, AdaptReason reason) {
>     rtc::Event event(false, false);
>@@ -119,9 +116,7 @@ class VideoStreamEncoderUnderTest : public VideoStreamEncoder {
>   // encoder queue is not blocked before we start sending it frames.
>   void WaitUntilTaskQueueIsIdle() {
>     rtc::Event event(false, false);
>-    encoder_queue()->PostTask([&event] {
>-      event.Set();
>-    });
>+    encoder_queue()->PostTask([&event] { event.Set(); });
>     ASSERT_TRUE(event.Wait(5000));
>   }
> 
>@@ -153,7 +148,7 @@ class VideoStreamFactory
>     std::vector<VideoStream> streams =
>         test::CreateVideoStreams(width, height, encoder_config);
>     for (VideoStream& stream : streams) {
>-      stream.temporal_layer_thresholds_bps.resize(num_temporal_layers_ - 1);
>+      stream.num_temporal_layers = num_temporal_layers_;
>       stream.max_framerate = framerate_;
>     }
>     return streams;
>@@ -163,7 +158,6 @@ class VideoStreamFactory
>   const int framerate_;
> };
> 
>-
> class AdaptingFrameForwarder : public test::FrameForwarder {
>  public:
>   AdaptingFrameForwarder() : adaptation_enabled_(false) {}
>@@ -184,8 +178,8 @@ class AdaptingFrameForwarder : public test::FrameForwarder {
>     return last_wants_;
>   }
> 
>-  rtc::Optional<int> last_sent_width() const { return last_width_; }
>-  rtc::Optional<int> last_sent_height() const { return last_height_; }
>+  absl::optional<int> last_sent_width() const { return last_width_; }
>+  absl::optional<int> last_sent_height() const { return last_height_; }
> 
>   void IncomingCapturedFrame(const VideoFrame& video_frame) override {
>     int cropped_width = 0;
>@@ -205,8 +199,8 @@ class AdaptingFrameForwarder : public test::FrameForwarder {
>         last_width_.emplace(adapted_frame.width());
>         last_height_.emplace(adapted_frame.height());
>       } else {
>-        last_width_ = rtc::nullopt;
>-        last_height_ = rtc::nullopt;
>+        last_width_ = absl::nullopt;
>+        last_height_ = absl::nullopt;
>       }
>     } else {
>       test::FrameForwarder::IncomingCapturedFrame(video_frame);
>@@ -227,10 +221,11 @@ class AdaptingFrameForwarder : public test::FrameForwarder {
>   cricket::VideoAdapter adapter_;
>   bool adaptation_enabled_ RTC_GUARDED_BY(crit_);
>   rtc::VideoSinkWants last_wants_ RTC_GUARDED_BY(crit_);
>-  rtc::Optional<int> last_width_;
>-  rtc::Optional<int> last_height_;
>+  absl::optional<int> last_width_;
>+  absl::optional<int> last_height_;
> };
> 
>+// TODO(nisse): Mock only VideoStreamEncoderObserver.
> class MockableSendStatisticsProxy : public SendStatisticsProxy {
>  public:
>   MockableSendStatisticsProxy(Clock* clock,
>@@ -245,6 +240,12 @@ class MockableSendStatisticsProxy : public SendStatisticsProxy {
>     return SendStatisticsProxy::GetStats();
>   }
> 
>+  int GetInputFrameRate() const override {
>+    rtc::CritScope cs(&lock_);
>+    if (mock_stats_)
>+      return mock_stats_->input_frame_rate;
>+    return SendStatisticsProxy::GetInputFrameRate();
>+  }
>   void SetMockStats(const VideoSendStream::Stats& stats) {
>     rtc::CritScope cs(&lock_);
>     mock_stats_.emplace(stats);
>@@ -257,12 +258,12 @@ class MockableSendStatisticsProxy : public SendStatisticsProxy {
> 
>  private:
>   rtc::CriticalSection lock_;
>-  rtc::Optional<VideoSendStream::Stats> mock_stats_ RTC_GUARDED_BY(lock_);
>+  absl::optional<VideoSendStream::Stats> mock_stats_ RTC_GUARDED_BY(lock_);
> };
> 
> class MockBitrateObserver : public VideoBitrateAllocationObserver {
>  public:
>-  MOCK_METHOD1(OnBitrateAllocationUpdated, void(const BitrateAllocation&));
>+  MOCK_METHOD1(OnBitrateAllocationUpdated, void(const VideoBitrateAllocation&));
> };
> 
> }  // namespace
>@@ -277,6 +278,7 @@ class VideoStreamEncoderTest : public ::testing::Test {
>         codec_height_(240),
>         max_framerate_(30),
>         fake_encoder_(),
>+        encoder_factory_(&fake_encoder_),
>         stats_proxy_(new MockableSendStatisticsProxy(
>             Clock::GetRealTimeClock(),
>             video_send_config_,
>@@ -286,12 +288,12 @@ class VideoStreamEncoderTest : public ::testing::Test {
>   void SetUp() override {
>     metrics::Reset();
>     video_send_config_ = VideoSendStream::Config(nullptr);
>-    video_send_config_.encoder_settings.encoder = &fake_encoder_;
>-    video_send_config_.encoder_settings.payload_name = "FAKE";
>-    video_send_config_.encoder_settings.payload_type = 125;
>+    video_send_config_.encoder_settings.encoder_factory = &encoder_factory_;
>+    video_send_config_.rtp.payload_name = "FAKE";
>+    video_send_config_.rtp.payload_type = 125;
> 
>     VideoEncoderConfig video_encoder_config;
>-    test::FillEncoderConfiguration(1, &video_encoder_config);
>+    test::FillEncoderConfiguration(kVideoCodecVP8, 1, &video_encoder_config);
>     video_encoder_config.video_stream_factory =
>         new rtc::RefCountedObject<VideoStreamFactory>(1, max_framerate_);
>     video_encoder_config_ = video_encoder_config.Copy();
>@@ -303,22 +305,20 @@ class VideoStreamEncoderTest : public ::testing::Test {
>     max_framerate_ = streams[0].max_framerate;
>     fake_clock_.SetTimeMicros(1234);
> 
>-    ConfigureEncoder(std::move(video_encoder_config), true /* nack_enabled */);
>+    ConfigureEncoder(std::move(video_encoder_config));
>   }
> 
>-  void ConfigureEncoder(VideoEncoderConfig video_encoder_config,
>-                        bool nack_enabled) {
>+  void ConfigureEncoder(VideoEncoderConfig video_encoder_config) {
>     if (video_stream_encoder_)
>       video_stream_encoder_->Stop();
>     video_stream_encoder_.reset(new VideoStreamEncoderUnderTest(
>         stats_proxy_.get(), video_send_config_.encoder_settings));
>     video_stream_encoder_->SetSink(&sink_, false /* rotation_applied */);
>     video_stream_encoder_->SetSource(
>-        &video_source_,
>-        VideoSendStream::DegradationPreference::kMaintainFramerate);
>+        &video_source_, webrtc::DegradationPreference::MAINTAIN_FRAMERATE);
>     video_stream_encoder_->SetStartBitrate(kTargetBitrateBps);
>     video_stream_encoder_->ConfigureEncoder(std::move(video_encoder_config),
>-                                            kMaxPayloadLength, nack_enabled);
>+                                            kMaxPayloadLength);
>     video_stream_encoder_->WaitUntilTaskQueueIsIdle();
>   }
> 
>@@ -326,11 +326,11 @@ class VideoStreamEncoderTest : public ::testing::Test {
>                     size_t num_streams,
>                     size_t num_temporal_layers,
>                     unsigned char num_spatial_layers,
>-                    bool nack_enabled,
>                     bool screenshare) {
>-    video_send_config_.encoder_settings.payload_name = payload_name;
>+    video_send_config_.rtp.payload_name = payload_name;
> 
>     VideoEncoderConfig video_encoder_config;
>+    video_encoder_config.codec_type = PayloadStringToCodecType(payload_name);
>     video_encoder_config.number_of_streams = num_streams;
>     video_encoder_config.max_bitrate_bps = kTargetBitrateBps;
>     video_encoder_config.video_stream_factory =
>@@ -346,7 +346,7 @@ class VideoStreamEncoderTest : public ::testing::Test {
>           new rtc::RefCountedObject<
>               VideoEncoderConfig::Vp9EncoderSpecificSettings>(vp9_settings);
>     }
>-    ConfigureEncoder(std::move(video_encoder_config), nack_enabled);
>+    ConfigureEncoder(std::move(video_encoder_config));
>   }
> 
>   VideoFrame CreateFrame(int64_t ntp_time_ms,
>@@ -505,8 +505,8 @@ class VideoStreamEncoderTest : public ::testing::Test {
>     VideoEncoder::ScalingSettings GetScalingSettings() const override {
>       rtc::CritScope lock(&local_crit_sect_);
>       if (quality_scaling_)
>-        return VideoEncoder::ScalingSettings(true, 1, 2, kMinPixelsPerFrame);
>-      return VideoEncoder::ScalingSettings(false);
>+        return VideoEncoder::ScalingSettings(1, 2, kMinPixelsPerFrame);
>+      return VideoEncoder::ScalingSettings::kOff;
>     }
> 
>     void ContinueEncode() { continue_encode_event_.Set(); }
>@@ -559,15 +559,13 @@ class VideoStreamEncoderTest : public ::testing::Test {
>       int res =
>           FakeEncoder::InitEncode(config, number_of_cores, max_payload_size);
>       rtc::CritScope lock(&local_crit_sect_);
>-      if (config->codecType == kVideoCodecVP8 && config->VP8().tl_factory) {
>+      if (config->codecType == kVideoCodecVP8) {
>         // Simulate setting up temporal layers, in order to validate the life
>         // cycle of these objects.
>         int num_streams = std::max<int>(1, config->numberOfSimulcastStreams);
>-        int num_temporal_layers =
>-            std::max<int>(1, config->VP8().numberOfTemporalLayers);
>         for (int i = 0; i < num_streams; ++i) {
>           allocated_temporal_layers_.emplace_back(
>-              config->VP8().tl_factory->Create(i, num_temporal_layers, 42));
>+              TemporalLayers::CreateTemporalLayers(*config, i));
>         }
>       }
>       if (force_init_encode_failed_)
>@@ -689,6 +687,7 @@ class VideoStreamEncoderTest : public ::testing::Test {
>   int codec_height_;
>   int max_framerate_;
>   TestEncoder fake_encoder_;
>+  test::EncoderProxyFactory encoder_factory_;
>   std::unique_ptr<MockableSendStatisticsProxy> stats_proxy_;
>   TestSink sink_;
>   AdaptingFrameForwarder video_source_;
>@@ -708,13 +707,20 @@ TEST_F(VideoStreamEncoderTest, EncodeOneFrame) {
> TEST_F(VideoStreamEncoderTest, DropsFramesBeforeFirstOnBitrateUpdated) {
>   // Dropped since no target bitrate has been set.
>   rtc::Event frame_destroyed_event(false, false);
>+  // The encoder will cache up to one frame for a short duration. Adding two
>+  // frames means that the first frame will be dropped and the second frame will
>+  // be sent when the encoder is enabled.
>   video_source_.IncomingCapturedFrame(CreateFrame(1, &frame_destroyed_event));
>+  video_source_.IncomingCapturedFrame(CreateFrame(2, nullptr));
>   EXPECT_TRUE(frame_destroyed_event.Wait(kDefaultTimeoutMs));
> 
>   video_stream_encoder_->OnBitrateUpdated(kTargetBitrateBps, 0, 0);
> 
>-  video_source_.IncomingCapturedFrame(CreateFrame(2, nullptr));
>+  // The pending frame should be received.
>   WaitForEncodedFrame(2);
>+  video_source_.IncomingCapturedFrame(CreateFrame(3, nullptr));
>+
>+  WaitForEncodedFrame(3);
>   video_stream_encoder_->Stop();
> }
> 
>@@ -724,12 +730,16 @@ TEST_F(VideoStreamEncoderTest, DropsFramesWhenRateSetToZero) {
>   WaitForEncodedFrame(1);
> 
>   video_stream_encoder_->OnBitrateUpdated(0, 0, 0);
>-  // Dropped since bitrate is zero.
>+  // The encoder will cache up to one frame for a short duration. Adding two
>+  // frames means that the first frame will be dropped and the second frame will
>+  // be sent when the encoder is resumed.
>   video_source_.IncomingCapturedFrame(CreateFrame(2, nullptr));
>+  video_source_.IncomingCapturedFrame(CreateFrame(3, nullptr));
> 
>   video_stream_encoder_->OnBitrateUpdated(kTargetBitrateBps, 0, 0);
>-  video_source_.IncomingCapturedFrame(CreateFrame(3, nullptr));
>   WaitForEncodedFrame(3);
>+  video_source_.IncomingCapturedFrame(CreateFrame(4, nullptr));
>+  WaitForEncodedFrame(4);
>   video_stream_encoder_->Stop();
> }
> 
>@@ -788,11 +798,10 @@ TEST_F(VideoStreamEncoderTest,
>   EXPECT_EQ(1, sink_.number_of_reconfigurations());
> 
>   VideoEncoderConfig video_encoder_config;
>-  test::FillEncoderConfiguration(1, &video_encoder_config);
>+  test::FillEncoderConfiguration(kVideoCodecVP8, 1, &video_encoder_config);
>   video_encoder_config.min_transmit_bitrate_bps = 9999;
>   video_stream_encoder_->ConfigureEncoder(std::move(video_encoder_config),
>-                                          kMaxPayloadLength,
>-                                          true /* nack_enabled */);
>+                                          kMaxPayloadLength);
> 
>   // Capture a frame and wait for it to synchronize with the encoder thread.
>   video_source_.IncomingCapturedFrame(CreateFrame(2, nullptr));
>@@ -827,180 +836,11 @@ TEST_F(VideoStreamEncoderTest, FrameResolutionChangeReconfigureEncoder) {
>   video_stream_encoder_->Stop();
> }
> 
>-TEST_F(VideoStreamEncoderTest, Vp8ResilienceIsOffFor1S1TLWithNackEnabled) {
>-  const bool kNackEnabled = true;
>-  const size_t kNumStreams = 1;
>-  const size_t kNumTl = 1;
>-  ResetEncoder("VP8", kNumStreams, kNumTl, kNumSlDummy, kNackEnabled, false);
>-  video_stream_encoder_->OnBitrateUpdated(kTargetBitrateBps, 0, 0);
>-
>-  // Capture a frame and wait for it to synchronize with the encoder thread.
>-  video_source_.IncomingCapturedFrame(CreateFrame(1, nullptr));
>-  WaitForEncodedFrame(1);
>-  // The encoder have been configured once when the first frame is received.
>-  EXPECT_EQ(1, sink_.number_of_reconfigurations());
>-  EXPECT_EQ(kVideoCodecVP8, fake_encoder_.codec_config().codecType);
>-  EXPECT_EQ(kNumStreams, fake_encoder_.codec_config().numberOfSimulcastStreams);
>-  EXPECT_EQ(kNumTl, fake_encoder_.codec_config().VP8()->numberOfTemporalLayers);
>-  // Resilience is off for no temporal layers with nack on.
>-  EXPECT_EQ(kResilienceOff, fake_encoder_.codec_config().VP8()->resilience);
>-  video_stream_encoder_->Stop();
>-}
>-
>-TEST_F(VideoStreamEncoderTest, Vp8ResilienceIsOffFor2S1TlWithNackEnabled) {
>-  const bool kNackEnabled = true;
>-  const size_t kNumStreams = 2;
>-  const size_t kNumTl = 1;
>-  ResetEncoder("VP8", kNumStreams, kNumTl, kNumSlDummy, kNackEnabled, false);
>-  video_stream_encoder_->OnBitrateUpdated(kTargetBitrateBps, 0, 0);
>-
>-  // Capture a frame and wait for it to synchronize with the encoder thread.
>-  video_source_.IncomingCapturedFrame(CreateFrame(1, nullptr));
>-  WaitForEncodedFrame(1);
>-  // The encoder have been configured once when the first frame is received.
>-  EXPECT_EQ(1, sink_.number_of_reconfigurations());
>-  EXPECT_EQ(kVideoCodecVP8, fake_encoder_.codec_config().codecType);
>-  EXPECT_EQ(kNumStreams, fake_encoder_.codec_config().numberOfSimulcastStreams);
>-  EXPECT_EQ(kNumTl, fake_encoder_.codec_config().VP8()->numberOfTemporalLayers);
>-  // Resilience is off for no temporal layers and >1 streams with nack on.
>-  EXPECT_EQ(kResilienceOff, fake_encoder_.codec_config().VP8()->resilience);
>-  video_stream_encoder_->Stop();
>-}
>-
>-TEST_F(VideoStreamEncoderTest, Vp8ResilienceIsOnFor1S1TLWithNackDisabled) {
>-  const bool kNackEnabled = false;
>-  const size_t kNumStreams = 1;
>-  const size_t kNumTl = 1;
>-  ResetEncoder("VP8", kNumStreams, kNumTl, kNumSlDummy, kNackEnabled, false);
>-  video_stream_encoder_->OnBitrateUpdated(kTargetBitrateBps, 0, 0);
>-
>-  // Capture a frame and wait for it to synchronize with the encoder thread.
>-  video_source_.IncomingCapturedFrame(CreateFrame(1, nullptr));
>-  WaitForEncodedFrame(1);
>-  // The encoder have been configured once when the first frame is received.
>-  EXPECT_EQ(1, sink_.number_of_reconfigurations());
>-  EXPECT_EQ(kVideoCodecVP8, fake_encoder_.codec_config().codecType);
>-  EXPECT_EQ(kNumStreams, fake_encoder_.codec_config().numberOfSimulcastStreams);
>-  EXPECT_EQ(kNumTl, fake_encoder_.codec_config().VP8()->numberOfTemporalLayers);
>-  // Resilience is on for no temporal layers with nack off.
>-  EXPECT_EQ(kResilientStream, fake_encoder_.codec_config().VP8()->resilience);
>-  video_stream_encoder_->Stop();
>-}
>-
>-TEST_F(VideoStreamEncoderTest, Vp8ResilienceIsOnFor1S2TlWithNackEnabled) {
>-  const bool kNackEnabled = true;
>-  const size_t kNumStreams = 1;
>-  const size_t kNumTl = 2;
>-  ResetEncoder("VP8", kNumStreams, kNumTl, kNumSlDummy, kNackEnabled, false);
>-  video_stream_encoder_->OnBitrateUpdated(kTargetBitrateBps, 0, 0);
>-
>-  // Capture a frame and wait for it to synchronize with the encoder thread.
>-  video_source_.IncomingCapturedFrame(CreateFrame(1, nullptr));
>-  WaitForEncodedFrame(1);
>-  // The encoder have been configured once when the first frame is received.
>-  EXPECT_EQ(1, sink_.number_of_reconfigurations());
>-  EXPECT_EQ(kVideoCodecVP8, fake_encoder_.codec_config().codecType);
>-  EXPECT_EQ(kNumStreams, fake_encoder_.codec_config().numberOfSimulcastStreams);
>-  EXPECT_EQ(kNumTl, fake_encoder_.codec_config().VP8()->numberOfTemporalLayers);
>-  // Resilience is on for temporal layers.
>-  EXPECT_EQ(kResilientStream, fake_encoder_.codec_config().VP8()->resilience);
>-  video_stream_encoder_->Stop();
>-}
>-
>-TEST_F(VideoStreamEncoderTest, Vp9ResilienceIsOffFor1SL1TLWithNackEnabled) {
>-  const bool kNackEnabled = true;
>-  const size_t kNumStreams = 1;
>-  const size_t kNumTl = 1;
>-  const unsigned char kNumSl = 1;
>-  ResetEncoder("VP9", kNumStreams, kNumTl, kNumSl, kNackEnabled, false);
>-  video_stream_encoder_->OnBitrateUpdated(kTargetBitrateBps, 0, 0);
>-
>-  // Capture a frame and wait for it to synchronize with the encoder thread.
>-  video_source_.IncomingCapturedFrame(CreateFrame(1, nullptr));
>-  sink_.WaitForEncodedFrame(1);
>-  // The encoder have been configured once when the first frame is received.
>-  EXPECT_EQ(1, sink_.number_of_reconfigurations());
>-  EXPECT_EQ(kVideoCodecVP9, fake_encoder_.codec_config().codecType);
>-  EXPECT_EQ(kNumStreams, fake_encoder_.codec_config().numberOfSimulcastStreams);
>-  EXPECT_EQ(kNumTl, fake_encoder_.codec_config().VP9()->numberOfTemporalLayers);
>-  EXPECT_EQ(kNumSl, fake_encoder_.codec_config().VP9()->numberOfSpatialLayers);
>-  // Resilience is off for no spatial and temporal layers with nack on.
>-  EXPECT_FALSE(fake_encoder_.codec_config().VP9()->resilienceOn);
>-  video_stream_encoder_->Stop();
>-}
>-
>-TEST_F(VideoStreamEncoderTest, Vp9ResilienceIsOnFor1SL1TLWithNackDisabled) {
>-  const bool kNackEnabled = false;
>-  const size_t kNumStreams = 1;
>-  const size_t kNumTl = 1;
>-  const unsigned char kNumSl = 1;
>-  ResetEncoder("VP9", kNumStreams, kNumTl, kNumSl, kNackEnabled, false);
>-  video_stream_encoder_->OnBitrateUpdated(kTargetBitrateBps, 0, 0);
>-
>-  // Capture a frame and wait for it to synchronize with the encoder thread.
>-  video_source_.IncomingCapturedFrame(CreateFrame(1, nullptr));
>-  sink_.WaitForEncodedFrame(1);
>-  // The encoder have been configured once when the first frame is received.
>-  EXPECT_EQ(1, sink_.number_of_reconfigurations());
>-  EXPECT_EQ(kVideoCodecVP9, fake_encoder_.codec_config().codecType);
>-  EXPECT_EQ(kNumStreams, fake_encoder_.codec_config().numberOfSimulcastStreams);
>-  EXPECT_EQ(kNumTl, fake_encoder_.codec_config().VP9()->numberOfTemporalLayers);
>-  EXPECT_EQ(kNumSl, fake_encoder_.codec_config().VP9()->numberOfSpatialLayers);
>-  // Resilience is on if nack is off.
>-  EXPECT_TRUE(fake_encoder_.codec_config().VP9()->resilienceOn);
>-  video_stream_encoder_->Stop();
>-}
>-
>-TEST_F(VideoStreamEncoderTest, Vp9ResilienceIsOnFor2SL1TLWithNackEnabled) {
>-  const bool kNackEnabled = true;
>-  const size_t kNumStreams = 1;
>-  const size_t kNumTl = 1;
>-  const unsigned char kNumSl = 2;
>-  ResetEncoder("VP9", kNumStreams, kNumTl, kNumSl, kNackEnabled, false);
>-  video_stream_encoder_->OnBitrateUpdated(kTargetBitrateBps, 0, 0);
>-
>-  // Capture a frame and wait for it to synchronize with the encoder thread.
>-  video_source_.IncomingCapturedFrame(CreateFrame(1, nullptr));
>-  sink_.WaitForEncodedFrame(1);
>-  // The encoder have been configured once when the first frame is received.
>-  EXPECT_EQ(1, sink_.number_of_reconfigurations());
>-  EXPECT_EQ(kVideoCodecVP9, fake_encoder_.codec_config().codecType);
>-  EXPECT_EQ(kNumStreams, fake_encoder_.codec_config().numberOfSimulcastStreams);
>-  EXPECT_EQ(kNumTl, fake_encoder_.codec_config().VP9()->numberOfTemporalLayers);
>-  EXPECT_EQ(kNumSl, fake_encoder_.codec_config().VP9()->numberOfSpatialLayers);
>-  // Resilience is on for spatial layers.
>-  EXPECT_TRUE(fake_encoder_.codec_config().VP9()->resilienceOn);
>-  video_stream_encoder_->Stop();
>-}
>-
>-TEST_F(VideoStreamEncoderTest, Vp9ResilienceIsOnFor1SL2TLWithNackEnabled) {
>-  const bool kNackEnabled = true;
>-  const size_t kNumStreams = 1;
>-  const size_t kNumTl = 2;
>-  const unsigned char kNumSl = 1;
>-  ResetEncoder("VP9", kNumStreams, kNumTl, kNumSl, kNackEnabled, false);
>-  video_stream_encoder_->OnBitrateUpdated(kTargetBitrateBps, 0, 0);
>-
>-  // Capture a frame and wait for it to synchronize with the encoder thread.
>-  video_source_.IncomingCapturedFrame(CreateFrame(1, nullptr));
>-  sink_.WaitForEncodedFrame(1);
>-  // The encoder have been configured once when the first frame is received.
>-  EXPECT_EQ(1, sink_.number_of_reconfigurations());
>-  EXPECT_EQ(kVideoCodecVP9, fake_encoder_.codec_config().codecType);
>-  EXPECT_EQ(kNumStreams, fake_encoder_.codec_config().numberOfSimulcastStreams);
>-  EXPECT_EQ(kNumTl, fake_encoder_.codec_config().VP9()->numberOfTemporalLayers);
>-  EXPECT_EQ(kNumSl, fake_encoder_.codec_config().VP9()->numberOfSpatialLayers);
>-  // Resilience is on for temporal layers.
>-  EXPECT_TRUE(fake_encoder_.codec_config().VP9()->resilienceOn);
>-  video_stream_encoder_->Stop();
>-}
>-
> TEST_F(VideoStreamEncoderTest, SwitchSourceDeregisterEncoderAsSink) {
>   EXPECT_TRUE(video_source_.has_sinks());
>   test::FrameForwarder new_video_source;
>   video_stream_encoder_->SetSource(
>-      &new_video_source,
>-      VideoSendStream::DegradationPreference::kMaintainFramerate);
>+      &new_video_source, webrtc::DegradationPreference::MAINTAIN_FRAMERATE);
>   EXPECT_FALSE(video_source_.has_sinks());
>   EXPECT_TRUE(new_video_source.has_sinks());
> 
>@@ -1023,11 +863,10 @@ TEST_F(VideoStreamEncoderTest, TestCpuDowngrades_BalancedMode) {
>   // adaptation manually by mocking the stats proxy.
>   video_source_.set_adaptation_enabled(true);
> 
>-  // Enable kBalanced preference, no initial limitation.
>+  // Enable BALANCED preference, no initial limitation.
>   video_stream_encoder_->OnBitrateUpdated(kTargetBitrateBps, 0, 0);
>-  video_stream_encoder_->SetSource(
>-      &video_source_,
>-      VideoSendStream::DegradationPreference::kBalanced);
>+  video_stream_encoder_->SetSource(&video_source_,
>+                                   webrtc::DegradationPreference::BALANCED);
>   VerifyNoLimitation(video_source_.sink_wants());
>   EXPECT_FALSE(stats_proxy_->GetStats().cpu_limited_resolution);
>   EXPECT_FALSE(stats_proxy_->GetStats().cpu_limited_framerate);
>@@ -1139,8 +978,7 @@ TEST_F(VideoStreamEncoderTest, SinkWantsStoredByDegradationPreference) {
>   // Set new source, switch to maintain-resolution.
>   test::FrameForwarder new_video_source;
>   video_stream_encoder_->SetSource(
>-      &new_video_source,
>-      VideoSendStream::DegradationPreference::kMaintainResolution);
>+      &new_video_source, webrtc::DegradationPreference::MAINTAIN_RESOLUTION);
> 
>   // Initially no degradation registered.
>   VerifyNoLimitation(new_video_source.sink_wants());
>@@ -1165,9 +1003,8 @@ TEST_F(VideoStreamEncoderTest, SinkWantsStoredByDegradationPreference) {
>   EXPECT_LT(new_video_source.sink_wants().max_framerate_fps, kInputFps);
> 
>   // Turn off degradation completely.
>-  video_stream_encoder_->SetSource(
>-      &new_video_source,
>-      VideoSendStream::DegradationPreference::kDegradationDisabled);
>+  video_stream_encoder_->SetSource(&new_video_source,
>+                                   webrtc::DegradationPreference::DISABLED);
>   VerifyNoLimitation(new_video_source.sink_wants());
> 
>   video_stream_encoder_->TriggerCpuOveruse();
>@@ -1181,8 +1018,7 @@ TEST_F(VideoStreamEncoderTest, SinkWantsStoredByDegradationPreference) {
> 
>   // Calling SetSource with resolution scaling enabled apply the old SinkWants.
>   video_stream_encoder_->SetSource(
>-      &new_video_source,
>-      VideoSendStream::DegradationPreference::kMaintainFramerate);
>+      &new_video_source, webrtc::DegradationPreference::MAINTAIN_FRAMERATE);
>   EXPECT_LT(new_video_source.sink_wants().max_pixel_count,
>             kFrameWidth * kFrameHeight);
>   EXPECT_FALSE(new_video_source.sink_wants().target_pixel_count);
>@@ -1191,8 +1027,7 @@ TEST_F(VideoStreamEncoderTest, SinkWantsStoredByDegradationPreference) {
> 
>   // Calling SetSource with framerate scaling enabled apply the old SinkWants.
>   video_stream_encoder_->SetSource(
>-      &new_video_source,
>-      VideoSendStream::DegradationPreference::kMaintainResolution);
>+      &new_video_source, webrtc::DegradationPreference::MAINTAIN_RESOLUTION);
>   EXPECT_FALSE(new_video_source.sink_wants().target_pixel_count);
>   EXPECT_EQ(std::numeric_limits<int>::max(),
>             new_video_source.sink_wants().max_pixel_count);
>@@ -1291,8 +1126,7 @@ TEST_F(VideoStreamEncoderTest, SwitchingSourceKeepsCpuAdaptation) {
>   // Set new source with adaptation still enabled.
>   test::FrameForwarder new_video_source;
>   video_stream_encoder_->SetSource(
>-      &new_video_source,
>-      VideoSendStream::DegradationPreference::kMaintainFramerate);
>+      &new_video_source, webrtc::DegradationPreference::MAINTAIN_FRAMERATE);
> 
>   new_video_source.IncomingCapturedFrame(CreateFrame(3, kWidth, kHeight));
>   WaitForEncodedFrame(3);
>@@ -1302,9 +1136,8 @@ TEST_F(VideoStreamEncoderTest, SwitchingSourceKeepsCpuAdaptation) {
>   EXPECT_EQ(1, stats.number_of_cpu_adapt_changes);
> 
>   // Set adaptation disabled.
>-  video_stream_encoder_->SetSource(
>-      &new_video_source,
>-      VideoSendStream::DegradationPreference::kDegradationDisabled);
>+  video_stream_encoder_->SetSource(&new_video_source,
>+                                   webrtc::DegradationPreference::DISABLED);
> 
>   new_video_source.IncomingCapturedFrame(CreateFrame(4, kWidth, kHeight));
>   WaitForEncodedFrame(4);
>@@ -1315,8 +1148,7 @@ TEST_F(VideoStreamEncoderTest, SwitchingSourceKeepsCpuAdaptation) {
> 
>   // Set adaptation back to enabled.
>   video_stream_encoder_->SetSource(
>-      &new_video_source,
>-      VideoSendStream::DegradationPreference::kMaintainFramerate);
>+      &new_video_source, webrtc::DegradationPreference::MAINTAIN_FRAMERATE);
> 
>   new_video_source.IncomingCapturedFrame(CreateFrame(5, kWidth, kHeight));
>   WaitForEncodedFrame(5);
>@@ -1352,9 +1184,8 @@ TEST_F(VideoStreamEncoderTest, SwitchingSourceKeepsQualityAdaptation) {
> 
>   // Set new source with adaptation still enabled.
>   test::FrameForwarder new_video_source;
>-  video_stream_encoder_->SetSource(
>-      &new_video_source,
>-      VideoSendStream::DegradationPreference::kBalanced);
>+  video_stream_encoder_->SetSource(&new_video_source,
>+                                   webrtc::DegradationPreference::BALANCED);
> 
>   new_video_source.IncomingCapturedFrame(CreateFrame(2, kWidth, kHeight));
>   WaitForEncodedFrame(2);
>@@ -1373,9 +1204,8 @@ TEST_F(VideoStreamEncoderTest, SwitchingSourceKeepsQualityAdaptation) {
>   EXPECT_EQ(1, stats.number_of_quality_adapt_changes);
> 
>   // Set new source with adaptation still enabled.
>-  video_stream_encoder_->SetSource(
>-      &new_video_source,
>-      VideoSendStream::DegradationPreference::kBalanced);
>+  video_stream_encoder_->SetSource(&new_video_source,
>+                                   webrtc::DegradationPreference::BALANCED);
> 
>   new_video_source.IncomingCapturedFrame(CreateFrame(4, kWidth, kHeight));
>   WaitForEncodedFrame(4);
>@@ -1386,8 +1216,7 @@ TEST_F(VideoStreamEncoderTest, SwitchingSourceKeepsQualityAdaptation) {
> 
>   // Disable resolution scaling.
>   video_stream_encoder_->SetSource(
>-      &new_video_source,
>-      VideoSendStream::DegradationPreference::kMaintainResolution);
>+      &new_video_source, webrtc::DegradationPreference::MAINTAIN_RESOLUTION);
> 
>   new_video_source.IncomingCapturedFrame(CreateFrame(5, kWidth, kHeight));
>   WaitForEncodedFrame(5);
>@@ -1429,11 +1258,15 @@ TEST_F(VideoStreamEncoderTest,
>   EXPECT_TRUE(stats_proxy_->GetStats().bw_limited_resolution);
>   EXPECT_EQ(1, stats_proxy_->GetStats().number_of_cpu_adapt_changes);
> 
>-  // Set source with adaptation still enabled but quality scaler is off.
>+  // Leave source unchanged, but disable quality scaler.
>   fake_encoder_.SetQualityScaling(false);
>-  video_stream_encoder_->SetSource(
>-      &video_source_,
>-      VideoSendStream::DegradationPreference::kMaintainFramerate);
>+
>+  VideoEncoderConfig video_encoder_config;
>+  test::FillEncoderConfiguration(kVideoCodecVP8, 1, &video_encoder_config);
>+  // Make format different, to force recreation of encoder.
>+  video_encoder_config.video_format.parameters["foo"] = "foo";
>+  video_stream_encoder_->ConfigureEncoder(std::move(video_encoder_config),
>+                                          kMaxPayloadLength);
> 
>   video_source_.IncomingCapturedFrame(CreateFrame(4, kWidth, kHeight));
>   WaitForEncodedFrame(4);
>@@ -1471,8 +1304,7 @@ TEST_F(VideoStreamEncoderTest,
>   // Set new source with adaptation still enabled.
>   test::FrameForwarder new_video_source;
>   video_stream_encoder_->SetSource(
>-      &new_video_source,
>-      VideoSendStream::DegradationPreference::kMaintainFramerate);
>+      &new_video_source, webrtc::DegradationPreference::MAINTAIN_FRAMERATE);
> 
>   new_video_source.IncomingCapturedFrame(
>       CreateFrame(sequence, kWidth, kHeight));
>@@ -1484,8 +1316,7 @@ TEST_F(VideoStreamEncoderTest,
> 
>   // Set cpu adaptation by frame dropping.
>   video_stream_encoder_->SetSource(
>-      &new_video_source,
>-      VideoSendStream::DegradationPreference::kMaintainResolution);
>+      &new_video_source, webrtc::DegradationPreference::MAINTAIN_RESOLUTION);
>   new_video_source.IncomingCapturedFrame(
>       CreateFrame(sequence, kWidth, kHeight));
>   WaitForEncodedFrame(sequence++);
>@@ -1514,9 +1345,8 @@ TEST_F(VideoStreamEncoderTest,
>   EXPECT_EQ(2, stats.number_of_cpu_adapt_changes);
> 
>   // Disable CPU adaptation.
>-  video_stream_encoder_->SetSource(
>-      &new_video_source,
>-      VideoSendStream::DegradationPreference::kDegradationDisabled);
>+  video_stream_encoder_->SetSource(&new_video_source,
>+                                   webrtc::DegradationPreference::DISABLED);
>   new_video_source.IncomingCapturedFrame(
>       CreateFrame(sequence, kWidth, kHeight));
>   WaitForEncodedFrame(sequence++);
>@@ -1538,8 +1368,7 @@ TEST_F(VideoStreamEncoderTest,
> 
>   // Switch back the source with resolution adaptation enabled.
>   video_stream_encoder_->SetSource(
>-      &video_source_,
>-      VideoSendStream::DegradationPreference::kMaintainFramerate);
>+      &video_source_, webrtc::DegradationPreference::MAINTAIN_FRAMERATE);
>   video_source_.IncomingCapturedFrame(CreateFrame(sequence, kWidth, kHeight));
>   WaitForEncodedFrame(sequence++);
>   stats = stats_proxy_->GetStats();
>@@ -1558,8 +1387,7 @@ TEST_F(VideoStreamEncoderTest,
> 
>   // Back to the source with adaptation off, set it back to maintain-resolution.
>   video_stream_encoder_->SetSource(
>-      &new_video_source,
>-      VideoSendStream::DegradationPreference::kMaintainResolution);
>+      &new_video_source, webrtc::DegradationPreference::MAINTAIN_RESOLUTION);
>   new_video_source.IncomingCapturedFrame(
>       CreateFrame(sequence, kWidth, kHeight));
>   WaitForEncodedFrame(sequence++);
>@@ -1583,21 +1411,6 @@ TEST_F(VideoStreamEncoderTest,
>   video_stream_encoder_->Stop();
> }
> 
>-TEST_F(VideoStreamEncoderTest, StatsTracksPreferredBitrate) {
>-  video_stream_encoder_->OnBitrateUpdated(kTargetBitrateBps, 0, 0);
>-
>-  const int kWidth = 1280;
>-  const int kHeight = 720;
>-  video_source_.IncomingCapturedFrame(CreateFrame(1, kWidth, kHeight));
>-  WaitForEncodedFrame(1);
>-
>-  VideoSendStream::Stats stats = stats_proxy_->GetStats();
>-  EXPECT_EQ(video_encoder_config_.max_bitrate_bps,
>-            stats.preferred_media_bitrate_bps);
>-
>-  video_stream_encoder_->Stop();
>-}
>-
> TEST_F(VideoStreamEncoderTest,
>        ScalingUpAndDownDoesNothingWithMaintainResolution) {
>   const int kWidth = 1280;
>@@ -1623,8 +1436,7 @@ TEST_F(VideoStreamEncoderTest,
>   // Set resolution scaling disabled.
>   test::FrameForwarder new_video_source;
>   video_stream_encoder_->SetSource(
>-      &new_video_source,
>-      VideoSendStream::DegradationPreference::kMaintainResolution);
>+      &new_video_source, webrtc::DegradationPreference::MAINTAIN_RESOLUTION);
> 
>   // Trigger scale down.
>   video_stream_encoder_->TriggerQualityLow();
>@@ -1653,10 +1465,10 @@ TEST_F(VideoStreamEncoderTest,
>   const int kHeight = 720;
>   video_stream_encoder_->OnBitrateUpdated(kTargetBitrateBps, 0, 0);
> 
>-  // Enable kMaintainFramerate preference, no initial limitation.
>+  // Enable MAINTAIN_FRAMERATE preference, no initial limitation.
>   test::FrameForwarder source;
>   video_stream_encoder_->SetSource(
>-      &source, VideoSendStream::DegradationPreference::kMaintainFramerate);
>+      &source, webrtc::DegradationPreference::MAINTAIN_FRAMERATE);
> 
>   source.IncomingCapturedFrame(CreateFrame(1, kWidth, kHeight));
>   WaitForEncodedFrame(1);
>@@ -1685,11 +1497,10 @@ TEST_F(VideoStreamEncoderTest, SkipsSameOrLargerAdaptDownRequest_BalancedMode) {
>   const int kHeight = 720;
>   video_stream_encoder_->OnBitrateUpdated(kTargetBitrateBps, 0, 0);
> 
>-  // Enable kBalanced preference, no initial limitation.
>+  // Enable BALANCED preference, no initial limitation.
>   test::FrameForwarder source;
>-  video_stream_encoder_->SetSource(
>-      &source,
>-      VideoSendStream::DegradationPreference::kBalanced);
>+  video_stream_encoder_->SetSource(&source,
>+                                   webrtc::DegradationPreference::BALANCED);
>   source.IncomingCapturedFrame(CreateFrame(1, kWidth, kHeight));
>   sink_.WaitForEncodedFrame(1);
>   VerifyNoLimitation(source.sink_wants());
>@@ -1726,10 +1537,10 @@ TEST_F(VideoStreamEncoderTest,
>   const int kHeight = 720;
>   video_stream_encoder_->OnBitrateUpdated(kTargetBitrateBps, 0, 0);
> 
>-  // Enable kMaintainFramerate preference, no initial limitation.
>+  // Enable MAINTAIN_FRAMERATE preference, no initial limitation.
>   test::FrameForwarder source;
>   video_stream_encoder_->SetSource(
>-      &source, VideoSendStream::DegradationPreference::kMaintainFramerate);
>+      &source, webrtc::DegradationPreference::MAINTAIN_FRAMERATE);
> 
>   source.IncomingCapturedFrame(CreateFrame(1, kWidth, kHeight));
>   WaitForEncodedFrame(kWidth, kHeight);
>@@ -1752,10 +1563,10 @@ TEST_F(VideoStreamEncoderTest,
>   const int kHeight = 720;
>   video_stream_encoder_->OnBitrateUpdated(kTargetBitrateBps, 0, 0);
> 
>-  // Enable kMaintainResolution preference, no initial limitation.
>+  // Enable MAINTAIN_RESOLUTION preference, no initial limitation.
>   test::FrameForwarder source;
>   video_stream_encoder_->SetSource(
>-      &source, VideoSendStream::DegradationPreference::kMaintainResolution);
>+      &source, webrtc::DegradationPreference::MAINTAIN_RESOLUTION);
> 
>   source.IncomingCapturedFrame(CreateFrame(1, kWidth, kHeight));
>   WaitForEncodedFrame(kWidth, kHeight);
>@@ -1777,11 +1588,10 @@ TEST_F(VideoStreamEncoderTest, NoChangeForInitialNormalUsage_BalancedMode) {
>   const int kHeight = 720;
>   video_stream_encoder_->OnBitrateUpdated(kTargetBitrateBps, 0, 0);
> 
>-  // Enable kBalanced preference, no initial limitation.
>+  // Enable BALANCED preference, no initial limitation.
>   test::FrameForwarder source;
>-  video_stream_encoder_->SetSource(
>-      &source,
>-      VideoSendStream::DegradationPreference::kBalanced);
>+  video_stream_encoder_->SetSource(&source,
>+                                   webrtc::DegradationPreference::BALANCED);
> 
>   source.IncomingCapturedFrame(CreateFrame(1, kWidth, kHeight));
>   sink_.WaitForEncodedFrame(kWidth, kHeight);
>@@ -1805,10 +1615,10 @@ TEST_F(VideoStreamEncoderTest, NoChangeForInitialNormalUsage_DisabledMode) {
>   const int kHeight = 720;
>   video_stream_encoder_->OnBitrateUpdated(kTargetBitrateBps, 0, 0);
> 
>-  // Enable kDegradationDisabled preference, no initial limitation.
>+  // Enable DISABLED preference, no initial limitation.
>   test::FrameForwarder source;
>-  video_stream_encoder_->SetSource(
>-      &source, VideoSendStream::DegradationPreference::kDegradationDisabled);
>+  video_stream_encoder_->SetSource(&source,
>+                                   webrtc::DegradationPreference::DISABLED);
> 
>   source.IncomingCapturedFrame(CreateFrame(1, kWidth, kHeight));
>   sink_.WaitForEncodedFrame(kWidth, kHeight);
>@@ -1833,11 +1643,11 @@ TEST_F(VideoStreamEncoderTest,
>   const int kHeight = 720;
>   video_stream_encoder_->OnBitrateUpdated(kTargetBitrateBps, 0, 0);
> 
>-  // Enable kMaintainFramerate preference, no initial limitation.
>+  // Enable MAINTAIN_FRAMERATE preference, no initial limitation.
>   AdaptingFrameForwarder source;
>   source.set_adaptation_enabled(true);
>   video_stream_encoder_->SetSource(
>-      &source, VideoSendStream::DegradationPreference::kMaintainFramerate);
>+      &source, webrtc::DegradationPreference::MAINTAIN_FRAMERATE);
> 
>   source.IncomingCapturedFrame(CreateFrame(1, kWidth, kHeight));
>   WaitForEncodedFrame(1);
>@@ -1874,7 +1684,7 @@ TEST_F(VideoStreamEncoderTest,
>   stats.input_frame_rate = kInputFps;
>   stats_proxy_->SetMockStats(stats);
> 
>-  // Expect no scaling to begin with (preference: kMaintainFramerate).
>+  // Expect no scaling to begin with (preference: MAINTAIN_FRAMERATE).
>   video_source_.IncomingCapturedFrame(CreateFrame(1, kWidth, kHeight));
>   sink_.WaitForEncodedFrame(1);
>   VerifyNoLimitation(video_source_.sink_wants());
>@@ -1885,11 +1695,10 @@ TEST_F(VideoStreamEncoderTest,
>   sink_.WaitForEncodedFrame(2);
>   VerifyFpsMaxResolutionLt(video_source_.sink_wants(), kWidth * kHeight);
> 
>-  // Enable kMaintainResolution preference.
>+  // Enable MAINTAIN_RESOLUTION preference.
>   test::FrameForwarder new_video_source;
>   video_stream_encoder_->SetSource(
>-      &new_video_source,
>-      VideoSendStream::DegradationPreference::kMaintainResolution);
>+      &new_video_source, webrtc::DegradationPreference::MAINTAIN_RESOLUTION);
>   VerifyNoLimitation(new_video_source.sink_wants());
> 
>   // Trigger adapt down, expect reduced framerate.
>@@ -1946,11 +1755,11 @@ TEST_F(VideoStreamEncoderTest,
>   const int kHeight = 720;
>   video_stream_encoder_->OnBitrateUpdated(kTargetBitrateBps, 0, 0);
> 
>-  // Enable kMaintainFramerate preference, no initial limitation.
>+  // Enable MAINTAIN_FRAMERATE preference, no initial limitation.
>   AdaptingFrameForwarder source;
>   source.set_adaptation_enabled(true);
>   video_stream_encoder_->SetSource(
>-      &source, VideoSendStream::DegradationPreference::kMaintainFramerate);
>+      &source, webrtc::DegradationPreference::MAINTAIN_FRAMERATE);
> 
>   source.IncomingCapturedFrame(CreateFrame(1, kWidth, kHeight));
>   WaitForEncodedFrame(kWidth, kHeight);
>@@ -1999,12 +1808,11 @@ TEST_F(VideoStreamEncoderTest,
>   const int kHeight = 720;
>   video_stream_encoder_->OnBitrateUpdated(kTargetBitrateBps, 0, 0);
> 
>-  // Enable kBalanced preference, no initial limitation.
>+  // Enable BALANCED preference, no initial limitation.
>   AdaptingFrameForwarder source;
>   source.set_adaptation_enabled(true);
>-  video_stream_encoder_->SetSource(
>-      &source,
>-      VideoSendStream::DegradationPreference::kBalanced);
>+  video_stream_encoder_->SetSource(&source,
>+                                   webrtc::DegradationPreference::BALANCED);
> 
>   source.IncomingCapturedFrame(CreateFrame(1, kWidth, kHeight));
>   sink_.WaitForEncodedFrame(kWidth, kHeight);
>@@ -2053,11 +1861,11 @@ TEST_F(VideoStreamEncoderTest,
>   const int kHeight = 720;
>   video_stream_encoder_->OnBitrateUpdated(kTargetBitrateBps, 0, 0);
> 
>-  // Enable kMaintainFramerate preference, no initial limitation.
>+  // Enable MAINTAIN_FRAMERATE preference, no initial limitation.
>   AdaptingFrameForwarder source;
>   source.set_adaptation_enabled(true);
>   video_stream_encoder_->SetSource(
>-      &source, VideoSendStream::DegradationPreference::kMaintainFramerate);
>+      &source, webrtc::DegradationPreference::MAINTAIN_FRAMERATE);
> 
>   source.IncomingCapturedFrame(CreateFrame(1, kWidth, kHeight));
>   WaitForEncodedFrame(kWidth, kHeight);
>@@ -2207,9 +2015,8 @@ TEST_F(VideoStreamEncoderTest,
>   const int kWidth = 640;
>   const int kHeight = 360;
> 
>-  video_stream_encoder_->SetSource(
>-      &video_source_,
>-      VideoSendStream::DegradationPreference::kDegradationDisabled);
>+  video_stream_encoder_->SetSource(&video_source_,
>+                                   webrtc::DegradationPreference::DISABLED);
> 
>   for (int i = 1; i <= SendStatisticsProxy::kMinRequiredMetricsSamples; ++i) {
>     video_source_.IncomingCapturedFrame(CreateFrame(i, kWidth, kHeight));
>@@ -2226,10 +2033,10 @@ TEST_F(VideoStreamEncoderTest,
> 
> TEST_F(VideoStreamEncoderTest, CallsBitrateObserver) {
>   MockBitrateObserver bitrate_observer;
>-  video_stream_encoder_->SetBitrateObserver(&bitrate_observer);
>+  video_stream_encoder_->SetBitrateAllocationObserver(&bitrate_observer);
> 
>   const int kDefaultFps = 30;
>-  const BitrateAllocation expected_bitrate =
>+  const VideoBitrateAllocation expected_bitrate =
>       DefaultVideoBitrateAllocator(fake_encoder_.codec_config())
>           .GetAllocation(kLowTargetBitrateBps, kDefaultFps);
> 
>@@ -2272,7 +2079,7 @@ TEST_F(VideoStreamEncoderTest, OveruseDetectorUpdatedOnReconfigureAndAdaption) {
>   video_stream_encoder_->OnBitrateUpdated(kTargetBitrateBps, 0, 0);
>   test::FrameForwarder source;
>   video_stream_encoder_->SetSource(
>-      &source, VideoSendStream::DegradationPreference::kMaintainResolution);
>+      &source, webrtc::DegradationPreference::MAINTAIN_RESOLUTION);
> 
>   // Insert a single frame, triggering initial configuration.
>   source.IncomingCapturedFrame(CreateFrame(1, kFrameWidth, kFrameHeight));
>@@ -2284,12 +2091,13 @@ TEST_F(VideoStreamEncoderTest, OveruseDetectorUpdatedOnReconfigureAndAdaption) {
> 
>   // Trigger reconfigure encoder (without resetting the entire instance).
>   VideoEncoderConfig video_encoder_config;
>+  video_encoder_config.codec_type = kVideoCodecVP8;
>   video_encoder_config.max_bitrate_bps = kTargetBitrateBps;
>   video_encoder_config.number_of_streams = 1;
>   video_encoder_config.video_stream_factory =
>       new rtc::RefCountedObject<VideoStreamFactory>(1, kFramerate);
>   video_stream_encoder_->ConfigureEncoder(std::move(video_encoder_config),
>-                                          kMaxPayloadLength, false);
>+                                          kMaxPayloadLength);
>   video_stream_encoder_->WaitUntilTaskQueueIsIdle();
> 
>   // Detector should be updated with fps limit from codec config.
>@@ -2331,17 +2139,18 @@ TEST_F(VideoStreamEncoderTest,
>   video_stream_encoder_->OnBitrateUpdated(kTargetBitrateBps, 0, 0);
>   test::FrameForwarder source;
>   video_stream_encoder_->SetSource(
>-      &source, VideoSendStream::DegradationPreference::kMaintainResolution);
>+      &source, webrtc::DegradationPreference::MAINTAIN_RESOLUTION);
> 
>   // Trigger initial configuration.
>   VideoEncoderConfig video_encoder_config;
>+  video_encoder_config.codec_type = kVideoCodecVP8;
>   video_encoder_config.max_bitrate_bps = kTargetBitrateBps;
>   video_encoder_config.number_of_streams = 1;
>   video_encoder_config.video_stream_factory =
>       new rtc::RefCountedObject<VideoStreamFactory>(1, kLowFramerate);
>   source.IncomingCapturedFrame(CreateFrame(1, kFrameWidth, kFrameHeight));
>   video_stream_encoder_->ConfigureEncoder(std::move(video_encoder_config),
>-                                          kMaxPayloadLength, false);
>+                                          kMaxPayloadLength);
>   video_stream_encoder_->WaitUntilTaskQueueIsIdle();
> 
>   EXPECT_EQ(
>@@ -2364,7 +2173,7 @@ TEST_F(VideoStreamEncoderTest,
>       new rtc::RefCountedObject<VideoStreamFactory>(1, kHighFramerate);
>   source.IncomingCapturedFrame(CreateFrame(1, kFrameWidth, kFrameHeight));
>   video_stream_encoder_->ConfigureEncoder(std::move(video_encoder_config),
>-                                          kMaxPayloadLength, false);
>+                                          kMaxPayloadLength);
>   video_stream_encoder_->WaitUntilTaskQueueIsIdle();
> 
>   EXPECT_EQ(
>@@ -2393,17 +2202,18 @@ TEST_F(VideoStreamEncoderTest,
>   video_stream_encoder_->OnBitrateUpdated(kTargetBitrateBps, 0, 0);
>   test::FrameForwarder source;
>   video_stream_encoder_->SetSource(
>-      &source, VideoSendStream::DegradationPreference::kMaintainResolution);
>+      &source, webrtc::DegradationPreference::MAINTAIN_RESOLUTION);
> 
>   // Trigger initial configuration.
>   VideoEncoderConfig video_encoder_config;
>+  video_encoder_config.codec_type = kVideoCodecVP8;
>   video_encoder_config.max_bitrate_bps = kTargetBitrateBps;
>   video_encoder_config.number_of_streams = 1;
>   video_encoder_config.video_stream_factory =
>       new rtc::RefCountedObject<VideoStreamFactory>(1, kFramerate);
>   source.IncomingCapturedFrame(CreateFrame(1, kFrameWidth, kFrameHeight));
>   video_stream_encoder_->ConfigureEncoder(std::move(video_encoder_config),
>-                                          kMaxPayloadLength, false);
>+                                          kMaxPayloadLength);
>   video_stream_encoder_->WaitUntilTaskQueueIsIdle();
> 
>   EXPECT_EQ(
>@@ -2423,7 +2233,7 @@ TEST_F(VideoStreamEncoderTest,
>   // Change degradation preference to not enable framerate scaling. Target
>   // framerate should be changed to codec defined limit.
>   video_stream_encoder_->SetSource(
>-      &source, VideoSendStream::DegradationPreference::kMaintainFramerate);
>+      &source, webrtc::DegradationPreference::MAINTAIN_FRAMERATE);
>   video_stream_encoder_->WaitUntilTaskQueueIsIdle();
>   EXPECT_EQ(
>       video_stream_encoder_->overuse_detector_proxy_->GetLastTargetFramerate(),
>@@ -2491,8 +2301,7 @@ TEST_F(VideoStreamEncoderTest,
> 
>   // Set degradation preference.
>   video_stream_encoder_->SetSource(
>-      &video_source_,
>-      VideoSendStream::DegradationPreference::kMaintainResolution);
>+      &video_source_, webrtc::DegradationPreference::MAINTAIN_RESOLUTION);
> 
>   video_source_.IncomingCapturedFrame(CreateFrame(1, kWidth, kHeight));
>   // Frame should not be dropped, even if it's too large.
>@@ -2505,12 +2314,18 @@ TEST_F(VideoStreamEncoderTest, InitialFrameDropOffWhenEncoderDisabledScaling) {
>   const int kWidth = 640;
>   const int kHeight = 360;
>   fake_encoder_.SetQualityScaling(false);
>+
>+  VideoEncoderConfig video_encoder_config;
>+  test::FillEncoderConfiguration(kVideoCodecVP8, 1, &video_encoder_config);
>+  // Make format different, to force recreation of encoder.
>+  video_encoder_config.video_format.parameters["foo"] = "foo";
>+  video_stream_encoder_->ConfigureEncoder(std::move(video_encoder_config),
>+                                          kMaxPayloadLength);
>   video_stream_encoder_->OnBitrateUpdated(kLowTargetBitrateBps, 0, 0);
> 
>   // Force quality scaler reconfiguration by resetting the source.
>-  video_stream_encoder_->SetSource(
>-      &video_source_,
>-      VideoSendStream::DegradationPreference::kBalanced);
>+  video_stream_encoder_->SetSource(&video_source_,
>+                                   webrtc::DegradationPreference::BALANCED);
> 
>   video_source_.IncomingCapturedFrame(CreateFrame(1, kWidth, kHeight));
>   // Frame should not be dropped, even if it's too large.
>@@ -2520,16 +2335,40 @@ TEST_F(VideoStreamEncoderTest, InitialFrameDropOffWhenEncoderDisabledScaling) {
>   fake_encoder_.SetQualityScaling(true);
> }
> 
>+TEST_F(VideoStreamEncoderTest, InitialFrameDropActivatesWhenBWEstimateReady) {
>+  webrtc::test::ScopedFieldTrials field_trials(
>+      "WebRTC-InitialFramedrop/Enabled/");
>+  // Reset encoder for field trials to take effect.
>+  ConfigureEncoder(video_encoder_config_.Copy());
>+  const int kTooLowBitrateForFrameSizeBps = 10000;
>+  const int kWidth = 640;
>+  const int kHeight = 360;
>+
>+  video_stream_encoder_->OnBitrateUpdated(kTargetBitrateBps, 0, 0);
>+  video_source_.IncomingCapturedFrame(CreateFrame(1, kWidth, kHeight));
>+  // Frame should not be dropped.
>+  WaitForEncodedFrame(1);
>+
>+  video_stream_encoder_->OnBitrateUpdated(kTooLowBitrateForFrameSizeBps, 0, 0);
>+  video_source_.IncomingCapturedFrame(CreateFrame(2, kWidth, kHeight));
>+  // Expect to drop this frame, the wait should time out.
>+  ExpectDroppedFrame();
>+
>+  // Expect the sink_wants to specify a scaled frame.
>+  EXPECT_LT(video_source_.sink_wants().max_pixel_count, kWidth * kHeight);
>+  video_stream_encoder_->Stop();
>+}
>+
> TEST_F(VideoStreamEncoderTest,
>        ResolutionNotAdaptedForTooSmallFrame_MaintainFramerateMode) {
>   const int kTooSmallWidth = 10;
>   const int kTooSmallHeight = 10;
>   video_stream_encoder_->OnBitrateUpdated(kTargetBitrateBps, 0, 0);
> 
>-  // Enable kMaintainFramerate preference, no initial limitation.
>+  // Enable MAINTAIN_FRAMERATE preference, no initial limitation.
>   test::FrameForwarder source;
>   video_stream_encoder_->SetSource(
>-      &source, VideoSendStream::DegradationPreference::kMaintainFramerate);
>+      &source, webrtc::DegradationPreference::MAINTAIN_FRAMERATE);
>   VerifyNoLimitation(source.sink_wants());
>   EXPECT_FALSE(stats_proxy_->GetStats().cpu_limited_resolution);
> 
>@@ -2551,11 +2390,10 @@ TEST_F(VideoStreamEncoderTest,
>   const int kFpsLimit = 7;
>   video_stream_encoder_->OnBitrateUpdated(kTargetBitrateBps, 0, 0);
> 
>-  // Enable kBalanced preference, no initial limitation.
>+  // Enable BALANCED preference, no initial limitation.
>   test::FrameForwarder source;
>-  video_stream_encoder_->SetSource(
>-      &source,
>-      VideoSendStream::DegradationPreference::kBalanced);
>+  video_stream_encoder_->SetSource(&source,
>+                                   webrtc::DegradationPreference::BALANCED);
>   VerifyNoLimitation(source.sink_wants());
>   EXPECT_FALSE(stats_proxy_->GetStats().bw_limited_resolution);
>   EXPECT_FALSE(stats_proxy_->GetStats().bw_limited_framerate);
>@@ -2584,7 +2422,7 @@ TEST_F(VideoStreamEncoderTest,
> TEST_F(VideoStreamEncoderTest, FailingInitEncodeDoesntCauseCrash) {
>   fake_encoder_.ForceInitEncodeFailure(true);
>   video_stream_encoder_->OnBitrateUpdated(kTargetBitrateBps, 0, 0);
>-  ResetEncoder("VP8", 2, 1, 1, true, false);
>+  ResetEncoder("VP8", 2, 1, 1, false);
>   const int kFrameWidth = 1280;
>   const int kFrameHeight = 720;
>   video_source_.IncomingCapturedFrame(
>@@ -2632,8 +2470,7 @@ TEST_F(VideoStreamEncoderTest,
> 
>   video_stream_encoder_->OnBitrateUpdated(kTargetBitrateBps, 0, 0);
>   video_stream_encoder_->SetSource(
>-      &video_source_,
>-      VideoSendStream::DegradationPreference::kMaintainResolution);
>+      &video_source_, webrtc::DegradationPreference::MAINTAIN_RESOLUTION);
>   video_source_.set_adaptation_enabled(true);
> 
>   int64_t timestamp_ms = fake_clock_.TimeNanos() / rtc::kNumNanosecsPerMillisec;
>@@ -2730,12 +2567,11 @@ TEST_F(VideoStreamEncoderTest, DoesntAdaptDownPastMinFramerate) {
> 
>   // Reconfigure encoder with two temporal layers and screensharing, which will
>   // disable frame dropping and make testing easier.
>-  ResetEncoder("VP8", 1, 2, 1, true, true);
>+  ResetEncoder("VP8", 1, 2, 1, true);
> 
>   video_stream_encoder_->OnBitrateUpdated(kTargetBitrateBps, 0, 0);
>   video_stream_encoder_->SetSource(
>-      &video_source_,
>-      VideoSendStream::DegradationPreference::kMaintainResolution);
>+      &video_source_, webrtc::DegradationPreference::MAINTAIN_RESOLUTION);
>   video_source_.set_adaptation_enabled(true);
> 
>   int64_t timestamp_ms = fake_clock_.TimeNanos() / rtc::kNumNanosecsPerMillisec;
>@@ -2753,8 +2589,8 @@ TEST_F(VideoStreamEncoderTest, DoesntAdaptDownPastMinFramerate) {
>         sink_.WaitForEncodedFrame(timestamp_ms);
>       }
>       timestamp_ms += kFrameIntervalMs;
>-      fake_clock_.AdvanceTimeMicros(
>-          kFrameIntervalMs * rtc::kNumMicrosecsPerMillisec);
>+      fake_clock_.AdvanceTimeMicros(kFrameIntervalMs *
>+                                    rtc::kNumMicrosecsPerMillisec);
>     }
>     // ...and then try to adapt again.
>     video_stream_encoder_->TriggerCpuOveruse();
>@@ -2774,12 +2610,11 @@ TEST_F(VideoStreamEncoderTest,
>   int64_t timestamp_ms = kFrameIntervalMs;
>   video_stream_encoder_->OnBitrateUpdated(kTargetBitrateBps, 0, 0);
> 
>-  // Enable kBalanced preference, no initial limitation.
>+  // Enable BALANCED preference, no initial limitation.
>   AdaptingFrameForwarder source;
>   source.set_adaptation_enabled(true);
>-  video_stream_encoder_->SetSource(
>-      &source,
>-      VideoSendStream::DegradationPreference::kBalanced);
>+  video_stream_encoder_->SetSource(&source,
>+                                   webrtc::DegradationPreference::BALANCED);
>   timestamp_ms += kFrameIntervalMs;
>   source.IncomingCapturedFrame(CreateFrame(timestamp_ms, kWidth, kHeight));
>   WaitForEncodedFrame(kWidth, kHeight);
>@@ -2955,12 +2790,11 @@ TEST_F(VideoStreamEncoderTest, AdaptWithTwoReasonsAndDifferentOrder_Framerate) {
>   int64_t timestamp_ms = kFrameIntervalMs;
>   video_stream_encoder_->OnBitrateUpdated(kTargetBitrateBps, 0, 0);
> 
>-  // Enable kBalanced preference, no initial limitation.
>+  // Enable BALANCED preference, no initial limitation.
>   AdaptingFrameForwarder source;
>   source.set_adaptation_enabled(true);
>-  video_stream_encoder_->SetSource(
>-      &source,
>-      VideoSendStream::DegradationPreference::kBalanced);
>+  video_stream_encoder_->SetSource(&source,
>+                                   webrtc::DegradationPreference::BALANCED);
>   timestamp_ms += kFrameIntervalMs;
>   source.IncomingCapturedFrame(CreateFrame(timestamp_ms, kWidth, kHeight));
>   WaitForEncodedFrame(kWidth, kHeight);
>@@ -3069,12 +2903,11 @@ TEST_F(VideoStreamEncoderTest,
>   int64_t timestamp_ms = kFrameIntervalMs;
>   video_stream_encoder_->OnBitrateUpdated(kTargetBitrateBps, 0, 0);
> 
>-  // Enable kBalanced preference, no initial limitation.
>+  // Enable BALANCED preference, no initial limitation.
>   AdaptingFrameForwarder source;
>   source.set_adaptation_enabled(true);
>-  video_stream_encoder_->SetSource(
>-      &source,
>-      VideoSendStream::DegradationPreference::kBalanced);
>+  video_stream_encoder_->SetSource(&source,
>+                                   webrtc::DegradationPreference::BALANCED);
>   timestamp_ms += kFrameIntervalMs;
>   source.IncomingCapturedFrame(CreateFrame(timestamp_ms, kWidth, kHeight));
>   WaitForEncodedFrame(kWidth, kHeight);
>@@ -3165,11 +2998,10 @@ TEST_F(VideoStreamEncoderTest, AcceptsFullHdAdaptedDownSimulcastFrames) {
>         int width,
>         int height,
>         const VideoEncoderConfig& encoder_config) override {
>-      std::vector<VideoStream> streams =
>-          test::CreateVideoStreams(width - width % 4, height - height % 4,
>-                                   encoder_config);
>+      std::vector<VideoStream> streams = test::CreateVideoStreams(
>+          width - width % 4, height - height % 4, encoder_config);
>       for (VideoStream& stream : streams) {
>-        stream.temporal_layer_thresholds_bps.resize(num_temporal_layers_ - 1);
>+        stream.num_temporal_layers = num_temporal_layers_;
>         stream.max_framerate = framerate_;
>       }
>       return streams;
>@@ -3190,12 +3022,13 @@ TEST_F(VideoStreamEncoderTest, AcceptsFullHdAdaptedDownSimulcastFrames) {
>   video_stream_encoder_->OnBitrateUpdated(kTargetBitrateBps, 0, 0);
>   // Trigger reconfigure encoder (without resetting the entire instance).
>   VideoEncoderConfig video_encoder_config;
>+  video_encoder_config.codec_type = kVideoCodecVP8;
>   video_encoder_config.max_bitrate_bps = kTargetBitrateBps;
>   video_encoder_config.number_of_streams = 1;
>   video_encoder_config.video_stream_factory =
>       new rtc::RefCountedObject<CroppingVideoStreamFactory>(1, kFramerate);
>   video_stream_encoder_->ConfigureEncoder(std::move(video_encoder_config),
>-                                          kMaxPayloadLength, false);
>+                                          kMaxPayloadLength);
>   video_stream_encoder_->WaitUntilTaskQueueIsIdle();
> 
>   video_source_.set_adaptation_enabled(true);
>@@ -3270,7 +3103,7 @@ TEST_F(VideoStreamEncoderTest, DoesNotUpdateBitrateAllocationWhenSuspended) {
>   const int kTargetBitrateBps = 1000000;
> 
>   MockBitrateObserver bitrate_observer;
>-  video_stream_encoder_->SetBitrateObserver(&bitrate_observer);
>+  video_stream_encoder_->SetBitrateAllocationObserver(&bitrate_observer);
> 
>   EXPECT_CALL(bitrate_observer, OnBitrateAllocationUpdated(_)).Times(1);
>   // Initial bitrate update.
>@@ -3302,4 +3135,44 @@ TEST_F(VideoStreamEncoderTest, DoesNotUpdateBitrateAllocationWhenSuspended) {
>   video_stream_encoder_->Stop();
> }
> 
>+TEST_F(VideoStreamEncoderTest,
>+       DefaultCpuAdaptationThresholdsForSoftwareEncoder) {
>+  const int kFrameWidth = 1280;
>+  const int kFrameHeight = 720;
>+  const CpuOveruseOptions default_options;
>+  video_stream_encoder_->OnBitrateUpdated(kTargetBitrateBps, 0, 0);
>+  video_source_.IncomingCapturedFrame(
>+      CreateFrame(1, kFrameWidth, kFrameHeight));
>+  WaitForEncodedFrame(1);
>+  EXPECT_EQ(video_stream_encoder_->overuse_detector_proxy_->GetOptions()
>+                .low_encode_usage_threshold_percent,
>+            default_options.low_encode_usage_threshold_percent);
>+  EXPECT_EQ(video_stream_encoder_->overuse_detector_proxy_->GetOptions()
>+                .high_encode_usage_threshold_percent,
>+            default_options.high_encode_usage_threshold_percent);
>+  video_stream_encoder_->Stop();
>+}
>+
>+TEST_F(VideoStreamEncoderTest,
>+       HigherCpuAdaptationThresholdsForHardwareEncoder) {
>+  const int kFrameWidth = 1280;
>+  const int kFrameHeight = 720;
>+  CpuOveruseOptions hardware_options;
>+  hardware_options.low_encode_usage_threshold_percent = 150;
>+  hardware_options.high_encode_usage_threshold_percent = 200;
>+  encoder_factory_.SetIsHardwareAccelerated(true);
>+
>+  video_stream_encoder_->OnBitrateUpdated(kTargetBitrateBps, 0, 0);
>+  video_source_.IncomingCapturedFrame(
>+      CreateFrame(1, kFrameWidth, kFrameHeight));
>+  WaitForEncodedFrame(1);
>+  EXPECT_EQ(video_stream_encoder_->overuse_detector_proxy_->GetOptions()
>+                .low_encode_usage_threshold_percent,
>+            hardware_options.low_encode_usage_threshold_percent);
>+  EXPECT_EQ(video_stream_encoder_->overuse_detector_proxy_->GetOptions()
>+                .high_encode_usage_threshold_percent,
>+            hardware_options.high_encode_usage_threshold_percent);
>+  video_stream_encoder_->Stop();
>+}
>+
> }  // namespace webrtc
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/voice_engine/BUILD.gn b/Source/ThirdParty/libwebrtc/Source/webrtc/voice_engine/BUILD.gn
>deleted file mode 100644
>index d0de9c64af2..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/voice_engine/BUILD.gn
>+++ /dev/null
>@@ -1,148 +0,0 @@
>-# Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
>-#
>-# Use of this source code is governed by a BSD-style license
>-# that can be found in the LICENSE file in the root of the source
>-# tree. An additional intellectual property rights grant can be found
>-# in the file PATENTS.  All contributing project authors may
>-# be found in the AUTHORS file in the root of the source tree.
>-
>-import("../webrtc.gni")
>-
>-visibility = [ ":*" ]
>-
>-rtc_static_library("voice_engine") {
>-  visibility += [ "*" ]
>-  sources = [
>-    "channel.cc",
>-    "channel.h",
>-    "channel_proxy.cc",
>-    "channel_proxy.h",
>-    "transport_feedback_packet_loss_tracker.cc",
>-    "transport_feedback_packet_loss_tracker.h",
>-    "utility.cc",
>-    "utility.h",
>-  ]
>-
>-  if (is_win) {
>-    cflags = [
>-      # TODO(kjellander): Bug 261: fix this warning.
>-      "/wd4373",  # Virtual function override.
>-    ]
>-  }
>-
>-  if (!build_with_chromium && is_clang) {
>-    # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>-    suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-  }
>-
>-  deps = [
>-    ":audio_level",
>-    "..:webrtc_common",
>-    "../:typedefs",
>-    "../api:array_view",
>-    "../api:audio_mixer_api",
>-    "../api:call_api",
>-    "../api:libjingle_peerconnection_api",
>-    "../api:optional",
>-    "../api:refcountedbase",
>-    "../api:transport_api",
>-    "../api/audio_codecs:audio_codecs_api",
>-    "../audio/utility:audio_frame_operations",
>-    "../call:rtp_interfaces",
>-    "../common_audio",
>-    "../logging:rtc_event_log_api",
>-    "../modules:module_api",
>-    "../modules/audio_coding",
>-    "../modules/audio_coding:audio_format_conversion",
>-    "../modules/audio_coding:audio_network_adaptor_config",
>-    "../modules/audio_device",
>-    "../modules/audio_processing",
>-    "../modules/bitrate_controller",
>-    "../modules/media_file",
>-    "../modules/pacing",
>-    "../modules/rtp_rtcp",
>-    "../modules/rtp_rtcp:rtp_rtcp_format",
>-    "../modules/utility",
>-    "../rtc_base:checks",
>-    "../rtc_base:rate_limiter",
>-    "../rtc_base:rtc_base_approved",
>-    "../rtc_base:rtc_task_queue",
>-    "../system_wrappers",
>-    "../system_wrappers:field_trial_api",
>-    "../system_wrappers:metrics_api",
>-  ]
>-}
>-
>-rtc_static_library("audio_level") {
>-  visibility += [
>-    ":voice_engine",
>-    "../audio:audio",
>-  ]
>-  sources = [
>-    "audio_level.cc",
>-    "audio_level.h",
>-  ]
>-
>-  deps = [
>-    "..:webrtc_common",
>-    "../:typedefs",
>-    "../common_audio",
>-    "../modules:module_api",
>-    "../rtc_base:rtc_base_approved",
>-  ]
>-}
>-
>-if (rtc_include_tests) {
>-  rtc_test("voice_engine_unittests") {
>-    visibility += webrtc_default_visibility
>-    deps = [
>-      ":voice_engine",
>-      "../api/audio_codecs:builtin_audio_decoder_factory",
>-      "../common_audio",
>-      "../modules:module_api",
>-      "../modules/audio_coding",
>-      "../modules/audio_device",
>-      "../modules/audio_processing",
>-      "../modules/media_file",
>-      "../modules/rtp_rtcp:rtp_rtcp_format",
>-      "../modules/utility",
>-      "../modules/video_capture:video_capture",
>-      "../rtc_base:checks",
>-      "../rtc_base:rtc_base_approved",
>-      "../rtc_base:rtc_base_tests_utils",
>-      "../system_wrappers",
>-      "../test:test_common",
>-      "../test:test_main",
>-      "../test:video_test_common",
>-      "//testing/gmock",
>-      "//testing/gtest",
>-    ]
>-
>-    if (is_android) {
>-      deps += [ "//testing/android/native_test:native_test_native_code" ]
>-      shard_timeout = 900
>-    }
>-
>-    sources = [
>-      "transport_feedback_packet_loss_tracker_unittest.cc",
>-      "utility_unittest.cc",
>-    ]
>-
>-    data = [
>-      "../resources/utility/encapsulated_pcm16b_8khz.wav",
>-      "../resources/utility/encapsulated_pcmu_8khz.wav",
>-    ]
>-
>-    if (is_win) {
>-      cflags = [
>-        # TODO(kjellander): Bug 261: fix this warning.
>-        "/wd4373",  # Virtual function override.
>-      ]
>-    }
>-
>-    if (!build_with_chromium && is_clang) {
>-      # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
>-      suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
>-    }
>-  }
>-}
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/voice_engine/DEPS b/Source/ThirdParty/libwebrtc/Source/webrtc/voice_engine/DEPS
>deleted file mode 100644
>index c8e9a1cac68..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/voice_engine/DEPS
>+++ /dev/null
>@@ -1,14 +0,0 @@
>-include_rules = [
>-  "+audio/utility/audio_frame_operations.h",
>-  "+call",
>-  "+common_audio",
>-  "+logging/rtc_event_log",
>-  "+modules/audio_coding",
>-  "+modules/audio_device",
>-  "+modules/audio_processing",
>-  "+modules/media_file",
>-  "+modules/pacing",
>-  "+modules/rtp_rtcp",
>-  "+modules/utility",
>-  "+system_wrappers",
>-]
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/voice_engine/OWNERS b/Source/ThirdParty/libwebrtc/Source/webrtc/voice_engine/OWNERS
>deleted file mode 100644
>index 0430ede769b..00000000000
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/voice_engine/OWNERS
>+++ /dev/null
>@@ -1,10 +0,0 @@
>-henrikg@webrtc.org
>-henrika@webrtc.org
>-niklas.enbom@webrtc.org
>-solenberg@webrtc.org
>-
>-
>-# These are for the common case of adding or renaming files. If you're doing
>-# structural changes, please get a review from a reviewer in this file.
>-per-file *.gn=*
>-per-file *.gni=*
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/webrtc.gni b/Source/ThirdParty/libwebrtc/Source/webrtc/webrtc.gni
>index fb1d77cadfc..eeb25c0edbd 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/webrtc.gni
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/webrtc.gni
>@@ -31,6 +31,11 @@ if (is_ios) {
> }
> 
> declare_args() {
>+  # Setting this to false will require the API user to pass in their own
>+  # SSLCertificateVerifier to verify the certificates presented from a
>+  # TLS-TURN server. In return disabling this saves around 100kb in the binary.
>+  rtc_builtin_ssl_root_certificates = true
>+
>   # Include the iLBC audio codec?
>   rtc_include_ilbc = true
> 
>@@ -81,6 +86,11 @@ declare_args() {
>   # Enable to use the Mozilla internal settings.
>   build_with_mozilla = false
> 
>+  # Enable use of Android AAudio which requires Android SDK 26 or above and
>+  # NDK r16 or above.
>+  rtc_enable_android_aaudio = false
>+
>+  # TODO(henrika): can this flag be removed?
>   rtc_enable_android_opensl = false
> 
>   # Link-Time Optimizations.
>@@ -112,12 +122,10 @@ declare_args() {
>   # all platforms except Android and iOS. Because FFmpeg can be built
>   # with/without H.264 support, |ffmpeg_branding| has to separately be set to a
>   # value that includes H.264, for example "Chrome". If FFmpeg is built without
>-  # H.264, compilation succeeds but |H264DecoderImpl| fails to initialize. See
>-  # also: |rtc_initialize_ffmpeg|.
>+  # H.264, compilation succeeds but |H264DecoderImpl| fails to initialize.
>   # CHECK THE OPENH264, FFMPEG AND H.264 LICENSES/PATENTS BEFORE BUILDING.
>   # http://www.openh264.org, https://www.ffmpeg.org/
>-  rtc_use_h264 =
>-      is_chrome_branded && proprietary_codecs && !is_android && !is_ios
>+  rtc_use_h264 = proprietary_codecs && !is_android && !is_ios
> 
>   # By default, use normal platform audio support or dummy audio, but don't
>   # use file-based audio playout and record.
>@@ -129,16 +137,14 @@ declare_args() {
>   # real audio samples.
>   rtc_audio_device_plays_sinus_tone = false
> 
>-  # When set to true, test targets will declare the files needed to run memcheck
>-  # as data dependencies. This is to enable memcheck execution on swarming bots.
>-  rtc_use_memcheck = false
>+  # Disable this to build without support for built-in software codecs.
>+  rtc_use_builtin_sw_codecs = true
> 
>-  # FFmpeg must be initialized for |H264DecoderImpl| to work. This can be done
>-  # by WebRTC during |H264DecoderImpl::InitDecode| or externally. FFmpeg must
>-  # only be initialized once. Projects that initialize FFmpeg externally, such
>-  # as Chromium, must turn this flag off so that WebRTC does not also
>-  # initialize.
>-  rtc_initialize_ffmpeg = !build_with_chromium
>+  if (is_ios) {
>+    # Build broadcast extension in AppRTCMobile for iOS. This results in the
>+    # binary only running on iOS 11+, which is why it is disabled by default.
>+    rtc_apprtcmobile_broadcast_extension = false
>+  }
> }
> 
> if (!build_with_mozilla) {
>@@ -160,7 +166,6 @@ declare_args() {
>   rtc_build_libsrtp = !build_with_mozilla
>   rtc_build_libvpx = !build_with_mozilla
>   rtc_libvpx_build_vp9 = !build_with_mozilla
>-  rtc_build_openmax_dl = !build_with_mozilla
>   rtc_build_opus = !build_with_mozilla
>   rtc_build_ssl = !build_with_mozilla
>   rtc_build_usrsctp = !build_with_mozilla
>@@ -168,7 +173,7 @@ declare_args() {
>   # Enable libevent task queues on platforms that support it.
>   # rtc_link_task_queue_impl must be set to true for this to
>   # have an effect.
>-  if (is_win || is_mac || is_ios || is_nacl) {
>+  if (is_win || is_mac || is_ios || is_nacl || is_fuchsia) {
>     rtc_enable_libevent = false
>     rtc_build_libevent = false
>   } else {
>@@ -176,19 +181,10 @@ declare_args() {
>     rtc_build_libevent = !build_with_mozilla
>   }
> 
>-  if (!is_ios && (current_cpu != "arm" || arm_version >= 7) &&
>-      current_cpu != "mips64el" && !build_with_mozilla) {
>-    rtc_use_openmax_dl = true
>-  } else {
>-    rtc_use_openmax_dl = false
>-  }
>-
>   # Build sources requiring GTK. NOTICE: This is not present in Chrome OS
>   # build environments, even if available for Chromium builds.
>   rtc_use_gtk = !build_with_chromium && !build_with_mozilla
> 
>-  rtc_restrict_logging = build_with_chromium || build_with_mozilla
>-
>   # Excluded in Chromium since its prerequisites don't require Pulse Audio.
>   rtc_include_pulse_audio = !build_with_chromium
> 
>@@ -273,6 +269,47 @@ if (build_with_chromium) {
>   webrtc_default_visibility += [ webrtc_root + "/../webrtc_overrides/*" ]
> }
> 
>+# ---- Poisons ----
>+#
>+# The general idea is that some targets declare that they contain some
>+# kind of poison, which makes it impossible for other targets to
>+# depend on them (even transitively) unless they declare themselves
>+# immune to that particular type of poison.
>+#
>+# Targets that *contain* poison of type foo should contain the line
>+#
>+#   poisonous = [ "foo" ]
>+#
>+# and targets that *are immune but arent't themselves poisonous*
>+# should contain
>+#
>+#   allow_poison = [ "foo" ]
>+#
>+# This useful in cases where we have some large target or set of
>+# targets and want to ensure that most other targets do not
>+# transitively depend on them. For example, almost no high-level
>+# target should depend on the audio codecs, since we want WebRTC users
>+# to be able to inject any subset of them and actually end up with a
>+# binary that doesn't include the codecs they didn't inject.
>+#
>+# Test-only targets (`testonly` set to true) and non-public targets
>+# (`visibility` not containing "*") are automatically immune to all
>+# types of poison.
>+#
>+# Here's the complete list of all types of poison. It must be kept in
>+# 1:1 correspondence with the set of //:poison_* targets.
>+#
>+all_poison_types = [
>+  # Encoders and decoders for specific audio codecs such as Opus and iSAC.
>+  "audio_codecs",
>+
>+  # Software video codecs (VP8 and VP9 through libvpx).
>+  "software_video_codecs",
>+]
>+
>+absl_include_config = "//third_party/abseil-cpp:absl_include_config"
>+absl_define_config = "//third_party/abseil-cpp:absl_define_config"
>+
> template("rtc_test") {
>   test(target_name) {
>     forward_variables_from(invoker,
>@@ -291,7 +328,11 @@ template("rtc_test") {
>     configs += invoker.configs
>     configs -= rtc_remove_configs
>     configs -= invoker.suppressed_configs
>-    public_configs = [ rtc_common_inherited_config ]
>+    public_configs = [
>+      rtc_common_inherited_config,
>+      absl_include_config,
>+      absl_define_config,
>+    ]
>     if (defined(invoker.public_configs)) {
>       public_configs += invoker.public_configs
>     }
>@@ -316,10 +357,53 @@ template("rtc_source_set") {
>     if (!defined(visibility)) {
>       visibility = webrtc_default_visibility
>     }
>+
>+    # What's your poison?
>+    if (defined(testonly) && testonly) {
>+      assert(!defined(poisonous))
>+      assert(!defined(allow_poison))
>+    } else {
>+      if (!defined(poisonous)) {
>+        poisonous = []
>+      }
>+      if (!defined(allow_poison)) {
>+        allow_poison = []
>+      }
>+      if (!defined(assert_no_deps)) {
>+        assert_no_deps = []
>+      }
>+      if (!defined(deps)) {
>+        deps = []
>+      }
>+      foreach(p, poisonous) {
>+        deps += [ webrtc_root + ":poison_" + p ]
>+      }
>+      foreach(poison_type, all_poison_types) {
>+        allow_dep = true
>+        foreach(v, visibility) {
>+          if (v == "*") {
>+            allow_dep = false
>+          }
>+        }
>+        foreach(p, allow_poison + poisonous) {
>+          if (p == poison_type) {
>+            allow_dep = true
>+          }
>+        }
>+        if (!allow_dep) {
>+          assert_no_deps += [ webrtc_root + ":poison_" + poison_type ]
>+        }
>+      }
>+    }
>+
>     configs += invoker.configs
>     configs -= rtc_remove_configs
>     configs -= invoker.suppressed_configs
>-    public_configs = [ rtc_common_inherited_config ]
>+    public_configs = [
>+      rtc_common_inherited_config,
>+      absl_include_config,
>+      absl_define_config,
>+    ]
>     if (defined(invoker.public_configs)) {
>       public_configs += invoker.public_configs
>     }
>@@ -344,12 +428,13 @@ template("rtc_executable") {
>     configs += invoker.configs
>     configs -= rtc_remove_configs
>     configs -= invoker.suppressed_configs
>-    deps = [
>-      "//build/config:exe_and_shlib_deps",
>-    ]
>-    deps += invoker.deps
>+    deps = invoker.deps
> 
>-    public_configs = [ rtc_common_inherited_config ]
>+    public_configs = [
>+      rtc_common_inherited_config,
>+      absl_include_config,
>+      absl_define_config,
>+    ]
>     if (defined(invoker.public_configs)) {
>       public_configs += invoker.public_configs
>     }
>@@ -370,10 +455,53 @@ template("rtc_static_library") {
>     if (!defined(visibility)) {
>       visibility = webrtc_default_visibility
>     }
>+
>+    # What's your poison?
>+    if (defined(testonly) && testonly) {
>+      assert(!defined(poisonous))
>+      assert(!defined(allow_poison))
>+    } else {
>+      if (!defined(poisonous)) {
>+        poisonous = []
>+      }
>+      if (!defined(allow_poison)) {
>+        allow_poison = []
>+      }
>+      if (!defined(assert_no_deps)) {
>+        assert_no_deps = []
>+      }
>+      if (!defined(deps)) {
>+        deps = []
>+      }
>+      foreach(p, poisonous) {
>+        deps += [ webrtc_root + ":poison_" + p ]
>+      }
>+      foreach(poison_type, all_poison_types) {
>+        allow_dep = true
>+        foreach(v, visibility) {
>+          if (v == "*") {
>+            allow_dep = false
>+          }
>+        }
>+        foreach(p, allow_poison + poisonous) {
>+          if (p == poison_type) {
>+            allow_dep = true
>+          }
>+        }
>+        if (!allow_dep) {
>+          assert_no_deps += [ webrtc_root + ":poison_" + poison_type ]
>+        }
>+      }
>+    }
>+
>     configs += invoker.configs
>     configs -= rtc_remove_configs
>     configs -= invoker.suppressed_configs
>-    public_configs = [ rtc_common_inherited_config ]
>+    public_configs = [
>+      rtc_common_inherited_config,
>+      absl_include_config,
>+      absl_define_config,
>+    ]
>     if (defined(invoker.public_configs)) {
>       public_configs += invoker.public_configs
>     }
>@@ -394,10 +522,53 @@ template("rtc_shared_library") {
>     if (!defined(visibility)) {
>       visibility = webrtc_default_visibility
>     }
>+
>+    # What's your poison?
>+    if (defined(testonly) && testonly) {
>+      assert(!defined(poisonous))
>+      assert(!defined(allow_poison))
>+    } else {
>+      if (!defined(poisonous)) {
>+        poisonous = []
>+      }
>+      if (!defined(allow_poison)) {
>+        allow_poison = []
>+      }
>+      if (!defined(assert_no_deps)) {
>+        assert_no_deps = []
>+      }
>+      if (!defined(deps)) {
>+        deps = []
>+      }
>+      foreach(p, poisonous) {
>+        deps += [ webrtc_root + ":poison_" + p ]
>+      }
>+      foreach(poison_type, all_poison_types) {
>+        allow_dep = true
>+        foreach(v, visibility) {
>+          if (v == "*") {
>+            allow_dep = false
>+          }
>+        }
>+        foreach(p, allow_poison + poisonous) {
>+          if (p == poison_type) {
>+            allow_dep = true
>+          }
>+        }
>+        if (!allow_dep) {
>+          assert_no_deps += [ webrtc_root + ":poison_" + poison_type ]
>+        }
>+      }
>+    }
>+
>     configs += invoker.configs
>     configs -= rtc_remove_configs
>     configs -= invoker.suppressed_configs
>-    public_configs = [ rtc_common_inherited_config ]
>+    public_configs = [
>+      rtc_common_inherited_config,
>+      absl_include_config,
>+      absl_define_config,
>+    ]
>     if (defined(invoker.public_configs)) {
>       public_configs += invoker.public_configs
>     }
>@@ -491,10 +662,30 @@ if (is_android) {
>                                "visibility",
>                              ])
> 
>+      javac_args = []
>+
>       # Treat warnings as errors.
>-      javac_args = [ "-Werror" ]
>+      javac_args += [ "-Werror" ]
>+
>+      # TODO(crbug.com/824679): Find out why this fails in Chromium
>+      if (!build_with_chromium) {
>+        javac_args += [
>+          "-Xep:ParameterNotNullable:ERROR",
>+          "-Xep:FieldMissingNullable:ERROR",
>+          "-Xep:ReturnMissingNullable:ERROR",
>+        ]
>+      }
>+
>+      # Add any arguments defined by the invoker.
>+      if (defined(invoker.javac_args)) {
>+        javac_args += invoker.javac_args
>+      }
>+
>+      if (!defined(deps)) {
>+        deps = []
>+      }
>+      deps += [ "//third_party/jsr-305:jsr_305_javalib" ]
> 
>-      # TODO(sakal): Fix build hooks crbug.com/webrtc/8168
>       no_build_hooks = true
>     }
>   }
>@@ -513,7 +704,20 @@ if (is_android) {
>       # Treat warnings as errors.
>       javac_args = [ "-Werror" ]
> 
>-      # TODO(sakal): Fix build hooks crbug.com/webrtc/8168
>+      # TODO(crbug.com/824679): Find out why this fails in Chromium
>+      if (!build_with_chromium) {
>+        javac_args += [
>+          "-Xep:ParameterNotNullable:ERROR",
>+          "-Xep:FieldMissingNullable:ERROR",
>+          "-Xep:ReturnMissingNullable:ERROR",
>+        ]
>+      }
>+
>+      if (!defined(deps)) {
>+        deps = []
>+      }
>+      deps += [ "//third_party/jsr-305:jsr_305_javalib" ]
>+
>       no_build_hooks = true
>     }
>   }
>@@ -532,7 +736,20 @@ if (is_android) {
>       # Treat warnings as errors.
>       javac_args = [ "-Werror" ]
> 
>-      # TODO(sakal): Fix build hooks crbug.com/webrtc/8168
>+      # TODO(crbug.com/824679): Find out why this fails in Chromium
>+      if (!build_with_chromium) {
>+        javac_args += [
>+          "-Xep:ParameterNotNullable:ERROR",
>+          "-Xep:FieldMissingNullable:ERROR",
>+          "-Xep:ReturnMissingNullable:ERROR",
>+        ]
>+      }
>+
>+      if (!defined(deps)) {
>+        deps = []
>+      }
>+      deps += [ "//third_party/jsr-305:jsr_305_javalib" ]
>+
>       no_build_hooks = true
>     }
>   }
>diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/whitespace.txt b/Source/ThirdParty/libwebrtc/Source/webrtc/whitespace.txt
>index fbd953182f2..ad522932040 100644
>--- a/Source/ThirdParty/libwebrtc/Source/webrtc/whitespace.txt
>+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/whitespace.txt
>@@ -2,4 +2,3 @@ You can modify this file to create no-op changelists.
> Try to write something funny. And please don't add trailing whitespace.
> 
> Once upon a time there was an elephant in Stockholm.
>-
>diff --git a/Source/ThirdParty/libwebrtc/WebKit/0001-Adapting-libwebrtc-H264-codec.patch b/Source/ThirdParty/libwebrtc/WebKit/0001-Adapting-libwebrtc-H264-codec.patch
>deleted file mode 100644
>index 03638614812..00000000000
>--- a/Source/ThirdParty/libwebrtc/WebKit/0001-Adapting-libwebrtc-H264-codec.patch
>+++ /dev/null
>@@ -1,118 +0,0 @@
>-From c5debd83bb3933413d0a3dd2417f5fd9d64a3205 Mon Sep 17 00:00:00 2001
>-From: Youenn Fablet <youenn@apple.com>
>-Date: Thu, 22 Mar 2018 16:38:14 -0700
>-Subject: [PATCH] Adapting-libwebrtc-H264-codec
>-
>----
>- .../sdk/objc/Framework/Classes/Video/RTCCVPixelBuffer.mm      |  1 +
>- .../Framework/Classes/VideoToolbox/RTCVideoDecoderH264.mm     | 11 +++++++----
>- .../Framework/Classes/VideoToolbox/RTCVideoEncoderH264.mm     | 10 ++++++++--
>- 3 files changed, 16 insertions(+), 6 deletions(-)
>-
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCCVPixelBuffer.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCCVPixelBuffer.mm
>-index 203896758e8..537a65e17d3 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCCVPixelBuffer.mm
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCCVPixelBuffer.mm
>-@@ -73,6 +73,7 @@ - (instancetype)initWithPixelBuffer:(CVPixelBufferRef)pixelBuffer
>- 
>- - (void)dealloc {
>-   CVBufferRelease(_pixelBuffer);
>-+  [super dealloc];
>- }
>- 
>- - (int)width {
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoDecoderH264.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoDecoderH264.mm
>-index 2f4b6bd72c6..34d08db0e3b 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoDecoderH264.mm
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoDecoderH264.mm
>-@@ -90,6 +90,10 @@ - (instancetype)init {
>- - (void)dealloc {
>-   [self destroyDecompressionSession];
>-   [self setVideoFormat:nullptr];
>-+  if (_callback) {
>-+    Block_release(_callback);
>-+  }
>-+  [super dealloc];
>- }
>- 
>- - (NSInteger)startDecodeWithSettings:(RTCVideoEncoderSettings *)settings
>-@@ -178,7 +182,7 @@ - (NSInteger)decode:(RTCEncodedImage *)inputImage
>- }
>- 
>- - (void)setCallback:(RTCVideoDecoderCallback)callback {
>--  _callback = callback;
>-+  _callback = Block_copy(callback);
>- }
>- 
>- - (void)setError:(OSStatus)error {
>-@@ -190,6 +194,7 @@ - (NSInteger)releaseDecoder {
>-   // is safe to null out the callback.
>-   [self destroyDecompressionSession];
>-   [self setVideoFormat:nullptr];
>-+  Block_release(_callback);
>-   _callback = nullptr;
>-   return WEBRTC_VIDEO_CODEC_OK;
>- }
>-@@ -260,9 +265,7 @@ - (void)configureDecompressionSession {
>- - (void)destroyDecompressionSession {
>-   if (_decompressionSession) {
>- #if defined(WEBRTC_IOS)
>--    if ([UIDevice isIOS11OrLater]) {
>--      VTDecompressionSessionWaitForAsynchronousFrames(_decompressionSession);
>--    }
>-+    VTDecompressionSessionWaitForAsynchronousFrames(_decompressionSession);
>- #endif
>-     VTDecompressionSessionInvalidate(_decompressionSession);
>-     CFRelease(_decompressionSession);
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoEncoderH264.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoEncoderH264.mm
>-index a818c27d1e6..809948011a6 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoEncoderH264.mm
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoEncoderH264.mm
>-@@ -34,6 +34,7 @@
>- #include "sdk/objc/Framework/Classes/VideoToolbox/nalu_rewriter.h"
>- #include "system_wrappers/include/clock.h"
>- #include "third_party/libyuv/include/libyuv/convert_from.h"
>-+#include "sdk/WebKit/WebKitUtilities.h"
>- 
>- @interface RTCVideoEncoderH264 ()
>- 
>-@@ -318,6 +319,10 @@ - (instancetype)initWithCodecInfo:(RTCVideoCodecInfo *)codecInfo {
>- 
>- - (void)dealloc {
>-   [self destroyCompressionSession];
>-+  if (_callback) {
>-+    Block_release(_callback);
>-+  }
>-+  [super dealloc];
>- }
>- 
>- - (NSInteger)startEncodeWithSettings:(RTCVideoEncoderSettings *)settings
>-@@ -463,7 +468,7 @@ - (NSInteger)encode:(RTCVideoFrame *)frame
>- }
>- 
>- - (void)setCallback:(RTCVideoEncoderCallback)callback {
>--  _callback = callback;
>-+  _callback = Block_copy(callback);
>- }
>- 
>- - (int)setBitrate:(uint32_t)bitrateKbit framerate:(uint32_t)framerate {
>-@@ -480,6 +485,7 @@ - (NSInteger)releaseEncoder {
>-   // callback anymore. Do not remove callback until the session is invalidated
>-   // since async encoder callbacks can occur until invalidation.
>-   [self destroyCompressionSession];
>-+  Block_release(_callback);
>-   _callback = nullptr;
>-   return WEBRTC_VIDEO_CODEC_OK;
>- }
>-@@ -573,7 +579,7 @@ - (int)resetCompressionSessionWithPixelFormat:(OSType)framePixelFormat {
>-       nullptr, 1, &kCFTypeDictionaryKeyCallBacks, &kCFTypeDictionaryValueCallBacks);
>-   CFDictionarySetValue(encoder_specs,
>-                        kVTVideoEncoderSpecification_EnableHardwareAcceleratedVideoEncoder,
>--                       kCFBooleanTrue);
>-+                       webrtc::isH264HardwareEncoderAllowed() ? kCFBooleanTrue : kCFBooleanFalse);
>- #endif
>-   OSStatus status =
>-       VTCompressionSessionCreate(nullptr,  // use default allocator
>--- 
>-2.15.1 (Apple Git-101)
>-
>diff --git a/Source/ThirdParty/libwebrtc/WebKit/0001-Disable-SIGPIPE-for-WebRTC-sockets.patch b/Source/ThirdParty/libwebrtc/WebKit/0001-Disable-SIGPIPE-for-WebRTC-sockets.patch
>deleted file mode 100644
>index 3a12db8e198..00000000000
>--- a/Source/ThirdParty/libwebrtc/WebKit/0001-Disable-SIGPIPE-for-WebRTC-sockets.patch
>+++ /dev/null
>@@ -1,25 +0,0 @@
>-From ca7c3a4683bbafaaf8a7bbe9fcf7b99cd0a55441 Mon Sep 17 00:00:00 2001
>-From: Youenn Fablet <youenn@apple.com>
>-Date: Mon, 9 Apr 2018 13:27:57 -0700
>-Subject: [PATCH] Disable SIGPIPE for WebRTC sockets
>-
>----
>- .../ThirdParty/libwebrtc/Source/webrtc/rtc_base/physicalsocketserver.cc | 2 +-
>- 1 file changed, 1 insertion(+), 1 deletion(-)
>-
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/physicalsocketserver.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/physicalsocketserver.cc
>-index 0d5fc525242..bba921b6692 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/physicalsocketserver.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/physicalsocketserver.cc
>-@@ -632,7 +632,7 @@ bool SocketDispatcher::Initialize() {
>- #elif defined(WEBRTC_POSIX)
>-   fcntl(s_, F_SETFL, fcntl(s_, F_GETFL, 0) | O_NONBLOCK);
>- #endif
>--#if defined(WEBRTC_IOS)
>-+#if defined(WEBRTC_IOS) || (defined(WEBRTC_MAC) && defined(WEBRTC_WEBKIT_BUILD))
>-   // iOS may kill sockets when the app is moved to the background
>-   // (specifically, if the app doesn't use the "voip" UIBackgroundMode). When
>-   // we attempt to write to such a socket, SIGPIPE will be raised, which by
>--- 
>-2.15.1 (Apple Git-101)
>-
>diff --git a/Source/ThirdParty/libwebrtc/WebKit/0001-Update-RTCVideoEncoderH264.mm-for-WebKit.patch b/Source/ThirdParty/libwebrtc/WebKit/0001-Update-RTCVideoEncoderH264.mm-for-WebKit.patch
>deleted file mode 100644
>index 168be45c14f..00000000000
>--- a/Source/ThirdParty/libwebrtc/WebKit/0001-Update-RTCVideoEncoderH264.mm-for-WebKit.patch
>+++ /dev/null
>@@ -1,274 +0,0 @@
>-From 6526d1daf054748cd0983f994ec0f988df235dc1 Mon Sep 17 00:00:00 2001
>-From: Youenn Fablet <youenn@apple.com>
>-Date: Mon, 30 Apr 2018 11:26:26 -0700
>-Subject: [PATCH] Update RTCVideoEncoderH264 for WebKit
>-
>----
>- .../Classes/VideoToolbox/RTCVideoEncoderH264.mm    | 159 ++++++++++++++++++---
>- 1 file changed, 140 insertions(+), 19 deletions(-)
>-
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoEncoderH264.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoEncoderH264.mm
>-index a818c27d1e6..68118f8f56a 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoEncoderH264.mm
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoEncoderH264.mm
>-@@ -35,6 +35,43 @@
>- #include "system_wrappers/include/clock.h"
>- #include "third_party/libyuv/include/libyuv/convert_from.h"
>- 
>-+#include "sdk/WebKit/EncoderUtilities.h"
>-+#include "sdk/WebKit/WebKitUtilities.h"
>-+
>-+#import <dlfcn.h>
>-+#import <objc/runtime.h>
>-+
>-+SOFT_LINK_FRAMEWORK_OPTIONAL(VideoToolBox)
>-+SOFT_LINK_POINTER_OPTIONAL(VideoToolBox, kVTVideoEncoderSpecification_Usage, NSString *)
>-+
>-+#if !ENABLE_VCP_ENCODER && !defined(WEBRTC_IOS)
>-+static inline bool isStandardFrameSize(int32_t width, int32_t height)
>-+{
>-+    // FIXME: Envision relaxing this rule, something like width and height dividable by 4 or 8 should be good enough.
>-+    if (width == 1280)
>-+        return height == 720;
>-+    if (width == 720)
>-+        return height == 1280;
>-+    if (width == 960)
>-+        return height == 540;
>-+    if (width == 540)
>-+        return height == 960;
>-+    if (width == 640)
>-+        return height == 480;
>-+    if (width == 480)
>-+        return height == 640;
>-+    if (width == 288)
>-+        return height == 352;
>-+    if (width == 352)
>-+        return height == 288;
>-+    if (width == 320)
>-+        return height == 240;
>-+    if (width == 240)
>-+        return height == 320;
>-+    return false;
>-+}
>-+#endif
>-+
>- @interface RTCVideoEncoderH264 ()
>- 
>- - (void)frameWasEncoded:(OSStatus)status
>-@@ -285,7 +322,7 @@ @implementation RTCVideoEncoderH264 {
>-   RTCVideoEncoderCallback _callback;
>-   int32_t _width;
>-   int32_t _height;
>--  VTCompressionSessionRef _compressionSession;
>-+  CompressionSessionRef _compressionSession;
>-   RTCVideoCodecMode _mode;
>- 
>-   webrtc::H264BitstreamParser _h264BitstreamParser;
>-@@ -318,6 +355,10 @@ - (instancetype)initWithCodecInfo:(RTCVideoCodecInfo *)codecInfo {
>- 
>- - (void)dealloc {
>-   [self destroyCompressionSession];
>-+  if (_callback) {
>-+    Block_release(_callback);
>-+  }
>-+  [super dealloc];
>- }
>- 
>- - (NSInteger)startEncodeWithSettings:(RTCVideoEncoderSettings *)settings
>-@@ -358,9 +399,9 @@ - (NSInteger)encode:(RTCVideoFrame *)frame
>- 
>-   // Get a pixel buffer from the pool and copy frame data over.
>-   CVPixelBufferPoolRef pixelBufferPool =
>--      VTCompressionSessionGetPixelBufferPool(_compressionSession);
>-+      CompressionSessionGetPixelBufferPool(_compressionSession);
>-   if ([self resetCompressionSessionIfNeededForPool:pixelBufferPool withFrame:frame]) {
>--    pixelBufferPool = VTCompressionSessionGetPixelBufferPool(_compressionSession);
>-+    pixelBufferPool = CompressionSessionGetPixelBufferPool(_compressionSession);
>-     isKeyframeRequired = YES;
>-   }
>- 
>-@@ -442,7 +483,7 @@ - (NSInteger)encode:(RTCVideoFrame *)frame
>-   // Update the bitrate if needed.
>-   [self setBitrateBps:_bitrateAdjuster->GetAdjustedBitrateBps()];
>- 
>--  OSStatus status = VTCompressionSessionEncodeFrame(_compressionSession,
>-+  OSStatus status = CompressionSessionEncodeFrame(_compressionSession,
>-                                                     pixelBuffer,
>-                                                     presentationTimeStamp,
>-                                                     kCMTimeInvalid,
>-@@ -463,7 +504,7 @@ - (NSInteger)encode:(RTCVideoFrame *)frame
>- }
>- 
>- - (void)setCallback:(RTCVideoEncoderCallback)callback {
>--  _callback = callback;
>-+  _callback = Block_copy(callback);
>- }
>- 
>- - (int)setBitrate:(uint32_t)bitrateKbit framerate:(uint32_t)framerate {
>-@@ -480,6 +521,7 @@ - (NSInteger)releaseEncoder {
>-   // callback anymore. Do not remove callback until the session is invalidated
>-   // since async encoder callbacks can occur until invalidation.
>-   [self destroyCompressionSession];
>-+  Block_release(_callback);
>-   _callback = nullptr;
>-   return WEBRTC_VIDEO_CODEC_OK;
>- }
>-@@ -565,22 +607,21 @@ - (int)resetCompressionSessionWithPixelFormat:(OSType)framePixelFormat {
>-     CFRelease(pixelFormat);
>-     pixelFormat = nullptr;
>-   }
>--  CFMutableDictionaryRef encoder_specs = nullptr;
>-+  CFDictionaryRef encoderSpecs = nullptr;
>- #if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
>-+  auto useHardwareEncoder = webrtc::isH264HardwareEncoderAllowed() ? kCFBooleanTrue : kCFBooleanFalse;
>-   // Currently hw accl is supported above 360p on mac, below 360p
>-   // the compression session will be created with hw accl disabled.
>--  encoder_specs = CFDictionaryCreateMutable(
>--      nullptr, 1, &kCFTypeDictionaryKeyCallBacks, &kCFTypeDictionaryValueCallBacks);
>--  CFDictionarySetValue(encoder_specs,
>--                       kVTVideoEncoderSpecification_EnableHardwareAcceleratedVideoEncoder,
>--                       kCFBooleanTrue);
>-+  CFTypeRef sessionKeys[] = {kVTVideoEncoderSpecification_EnableHardwareAcceleratedVideoEncoder, kVTVideoEncoderSpecification_RequireHardwareAcceleratedVideoEncoder, kVTCompressionPropertyKey_RealTime };
>-+  CFTypeRef sessionValues[] = {  useHardwareEncoder, useHardwareEncoder, kCFBooleanTrue };
>-+  encoderSpecs = CFDictionaryCreate(kCFAllocatorDefault, sessionKeys, sessionValues, 3, &kCFTypeDictionaryKeyCallBacks, &kCFTypeDictionaryValueCallBacks);
>- #endif
>-   OSStatus status =
>--      VTCompressionSessionCreate(nullptr,  // use default allocator
>-+      CompressionSessionCreate(nullptr,  // use default allocator
>-                                  _width,
>-                                  _height,
>--                                 kCMVideoCodecType_H264,
>--                                 encoder_specs,  // use hardware accelerated encoder if available
>-+                                 kCodecTypeH264,
>-+                                 encoderSpecs,  // use hardware accelerated encoder if available
>-                                  sourceAttributes,
>-                                  nullptr,  // use default compressed data allocator
>-                                  compressionOutputCallback,
>-@@ -590,24 +631,101 @@ - (int)resetCompressionSessionWithPixelFormat:(OSType)framePixelFormat {
>-     CFRelease(sourceAttributes);
>-     sourceAttributes = nullptr;
>-   }
>--  if (encoder_specs) {
>--    CFRelease(encoder_specs);
>--    encoder_specs = nullptr;
>-+  if (encoderSpecs) {
>-+    CFRelease(encoderSpecs);
>-+    encoderSpecs = nullptr;
>-   }
>-+
>-+#if ENABLE_VCP_ENCODER || defined(WEBRTC_IOS)
>-   if (status != noErr) {
>-     RTC_LOG(LS_ERROR) << "Failed to create compression session: " << status;
>-     return WEBRTC_VIDEO_CODEC_ERROR;
>-   }
>-+#endif
>- #if defined(WEBRTC_MAC) && !defined(WEBRTC_IOS)
>-   CFBooleanRef hwaccl_enabled = nullptr;
>--  status = VTSessionCopyProperty(_compressionSession,
>-+  if (status == noErr) {
>-+    status = VTSessionCopyProperty(_compressionSession,
>-                                  kVTCompressionPropertyKey_UsingHardwareAcceleratedVideoEncoder,
>-                                  nullptr,
>-                                  &hwaccl_enabled);
>-+  }
>-   if (status == noErr && (CFBooleanGetValue(hwaccl_enabled))) {
>-     RTC_LOG(LS_INFO) << "Compression session created with hw accl enabled";
>-   } else {
>-     RTC_LOG(LS_INFO) << "Compression session created with hw accl disabled";
>-+
>-+#if !ENABLE_VCP_ENCODER && !defined(WEBRTC_IOS)
>-+    if (!isStandardFrameSize(_width, _height)) {
>-+      RTC_LOG(LS_ERROR) << "Using H264 software encoder with non standard size is not supported";
>-+      return WEBRTC_VIDEO_CODEC_ERROR;
>-+    }
>-+
>-+    if (!getkVTVideoEncoderSpecification_Usage()) {
>-+      RTC_LOG(LS_ERROR) << "RTCVideoEncoderH264 cannot create a H264 software encoder";
>-+      return WEBRTC_VIDEO_CODEC_ERROR;
>-+    }
>-+
>-+    CFDictionaryRef ioSurfaceValue = CreateCFTypeDictionary(nullptr, nullptr, 0);
>-+    int64_t pixelFormatType = framePixelFormat;
>-+    CFNumberRef pixelFormat = CFNumberCreate(nullptr, kCFNumberLongType, &pixelFormatType);
>-+
>-+    const size_t attributesSize = 3;
>-+    CFTypeRef keys[attributesSize] = {
>-+      kCVPixelBufferOpenGLCompatibilityKey,
>-+      kCVPixelBufferIOSurfacePropertiesKey,
>-+      kCVPixelBufferPixelFormatTypeKey
>-+    };
>-+    CFTypeRef values[attributesSize] = {
>-+      kCFBooleanTrue,
>-+      ioSurfaceValue,
>-+      pixelFormat};
>-+    CFDictionaryRef sourceAttributes = CreateCFTypeDictionary(keys, values, attributesSize);
>-+
>-+    if (ioSurfaceValue) {
>-+      CFRelease(ioSurfaceValue);
>-+      ioSurfaceValue = nullptr;
>-+    }
>-+    if (pixelFormat) {
>-+      CFRelease(pixelFormat);
>-+      pixelFormat = nullptr;
>-+    }
>-+
>-+    CFMutableDictionaryRef encoderSpecs = CFDictionaryCreateMutable(nullptr, 2, &kCFTypeDictionaryKeyCallBacks, &kCFTypeDictionaryValueCallBacks);
>-+    CFDictionarySetValue(encoderSpecs, kVTVideoEncoderSpecification_EnableHardwareAcceleratedVideoEncoder, kCFBooleanFalse);
>-+    int usageValue = 1;
>-+    CFNumberRef usage = CFNumberCreate(nullptr, kCFNumberIntType, &usageValue);
>-+    CFDictionarySetValue(encoderSpecs, getkVTVideoEncoderSpecification_Usage(), usage);
>-+    if (usage) {
>-+      CFRelease(usage);
>-+      usage = nullptr;
>-+    }
>-+
>-+    [self destroyCompressionSession];
>-+
>-+    OSStatus status =
>-+      CompressionSessionCreate(nullptr,  // use default allocator
>-+                                 _width,
>-+                                 _height,
>-+                                 kCodecTypeH264,
>-+                                 encoderSpecs,
>-+                                 sourceAttributes,
>-+                                 nullptr,  // use default compressed data allocator
>-+                                 compressionOutputCallback,
>-+                                 nullptr,
>-+                                 &_compressionSession);
>-+    if (sourceAttributes) {
>-+      CFRelease(sourceAttributes);
>-+      sourceAttributes = nullptr;
>-+    }
>-+    if (encoderSpecs) {
>-+      CFRelease(encoderSpecs);
>-+      encoderSpecs = nullptr;
>-+    }
>-+    if (status != noErr) {
>-+      return WEBRTC_VIDEO_CODEC_ERROR;
>-+    }
>-+#endif
>-   }
>- #endif
>-   [self configureCompressionSession];
>-@@ -619,6 +737,9 @@ - (void)configureCompressionSession {
>-   SetVTSessionProperty(_compressionSession, kVTCompressionPropertyKey_RealTime, true);
>-   SetVTSessionProperty(_compressionSession, kVTCompressionPropertyKey_ProfileLevel, _profile);
>-   SetVTSessionProperty(_compressionSession, kVTCompressionPropertyKey_AllowFrameReordering, false);
>-+#if ENABLE_VCP_ENCODER
>-+  SetVTSessionProperty(_compressionSession, (__bridge CFStringRef)getkVTVideoEncoderSpecification_Usage(), 1);
>-+#endif
>-   [self setEncoderBitrateBps:_targetBitrateBps];
>-   // TODO(tkchin): Look at entropy mode and colorspace matrices.
>-   // TODO(tkchin): Investigate to see if there's any way to make this work.
>-@@ -636,7 +757,7 @@ - (void)configureCompressionSession {
>- 
>- - (void)destroyCompressionSession {
>-   if (_compressionSession) {
>--    VTCompressionSessionInvalidate(_compressionSession);
>-+    CompressionSessionInvalidate(_compressionSession);
>-     CFRelease(_compressionSession);
>-     _compressionSession = nullptr;
>-   }
>--- 
>-2.16.1 (Apple Git-102)
>-
>diff --git a/Source/ThirdParty/libwebrtc/WebKit/0001-Using-VCP.patch b/Source/ThirdParty/libwebrtc/WebKit/0001-Using-VCP.patch
>deleted file mode 100644
>index 5d08b2a0961..00000000000
>--- a/Source/ThirdParty/libwebrtc/WebKit/0001-Using-VCP.patch
>+++ /dev/null
>@@ -1,77 +0,0 @@
>-From f6d216e76cc576b7cac5a980c74d4cf60f28cb17 Mon Sep 17 00:00:00 2001
>-From: Youenn Fablet <youenn@apple.com>
>-Date: Thu, 22 Mar 2018 15:44:50 -0700
>-Subject: [PATCH] Using VCP
>-
>----
>- .../Classes/VideoToolbox/RTCVideoEncoderH264.mm          | 16 +++++++++-------
>- 1 file changed, 9 insertions(+), 7 deletions(-)
>-
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoEncoderH264.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoEncoderH264.mm
>-index d1af6e3c632..138d44573b8 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoEncoderH264.mm
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoEncoderH264.mm
>-@@ -34,6 +34,8 @@
>- #include "sdk/objc/Framework/Classes/VideoToolbox/nalu_rewriter.h"
>- #include "system_wrappers/include/clock.h"
>- #include "third_party/libyuv/include/libyuv/convert_from.h"
>-+
>-+#include "sdk/WebKit/EncoderUtilities.h"
>- #include "sdk/WebKit/WebKitUtilities.h"
>- 
>- @interface RTCVideoEncoderH264 ()
>-@@ -286,7 +288,7 @@ @implementation RTCVideoEncoderH264 {
>-   RTCVideoEncoderCallback _callback;
>-   int32_t _width;
>-   int32_t _height;
>--  VTCompressionSessionRef _compressionSession;
>-+  CompressionSessionRef _compressionSession;
>-   RTCVideoCodecMode _mode;
>- 
>-   webrtc::H264BitstreamParser _h264BitstreamParser;
>-@@ -360,9 +362,9 @@ - (NSInteger)encode:(RTCVideoFrame *)frame
>- 
>-   // Get a pixel buffer from the pool and copy frame data over.
>-   CVPixelBufferPoolRef pixelBufferPool =
>--      VTCompressionSessionGetPixelBufferPool(_compressionSession);
>-+      CompressionSessionGetPixelBufferPool(_compressionSession);
>-   if ([self resetCompressionSessionIfNeededForPool:pixelBufferPool withFrame:frame]) {
>--    pixelBufferPool = VTCompressionSessionGetPixelBufferPool(_compressionSession);
>-+    pixelBufferPool = CompressionSessionGetPixelBufferPool(_compressionSession);
>-     isKeyframeRequired = YES;
>-   }
>- 
>-@@ -444,7 +446,7 @@ - (NSInteger)encode:(RTCVideoFrame *)frame
>-   // Update the bitrate if needed.
>-   [self setBitrateBps:_bitrateAdjuster->GetAdjustedBitrateBps()];
>- 
>--  OSStatus status = VTCompressionSessionEncodeFrame(_compressionSession,
>-+  OSStatus status = CompressionSessionEncodeFrame(_compressionSession,
>-                                                     pixelBuffer,
>-                                                     presentationTimeStamp,
>-                                                     kCMTimeInvalid,
>-@@ -579,10 +581,10 @@ - (int)resetCompressionSessionWithPixelFormat:(OSType)framePixelFormat {
>-                        webrtc::isH264HardwareEncoderAllowed() ? kCFBooleanTrue : kCFBooleanFalse);
>- #endif
>-   OSStatus status =
>--      VTCompressionSessionCreate(nullptr,  // use default allocator
>-+      CompressionSessionCreate(nullptr,  // use default allocator
>-                                  _width,
>-                                  _height,
>--                                 kCMVideoCodecType_H264,
>-+                                 kCodecTypeH264,
>-                                  encoder_specs,  // use hardware accelerated encoder if available
>-                                  sourceAttributes,
>-                                  nullptr,  // use default compressed data allocator
>-@@ -639,7 +641,7 @@ - (void)configureCompressionSession {
>- 
>- - (void)destroyCompressionSession {
>-   if (_compressionSession) {
>--    VTCompressionSessionInvalidate(_compressionSession);
>-+    CompressionSessionInvalidate(_compressionSession);
>-     CFRelease(_compressionSession);
>-     _compressionSession = nullptr;
>-   }
>--- 
>-2.15.1 (Apple Git-101)
>-
>diff --git a/Source/ThirdParty/libwebrtc/WebKit/0003-Fixing-VP8-files.patch b/Source/ThirdParty/libwebrtc/WebKit/0003-Fixing-VP8-files.patch
>deleted file mode 100644
>index 06bb0099b02..00000000000
>--- a/Source/ThirdParty/libwebrtc/WebKit/0003-Fixing-VP8-files.patch
>+++ /dev/null
>@@ -1,3072 +0,0 @@
>-From be6c0b3edb476e45bf282046e8d1e41c6d4e17cf Mon Sep 17 00:00:00 2001
>-From: Youenn Fablet <youenn@apple.com>
>-Date: Thu, 14 Dec 2017 13:47:41 -0800
>-Subject: [PATCH 3/8] Fixing VP8 files
>-
>----
>- .../codecs/vp8/default_temporal_layers.cc          | 354 +++++++++---
>- .../codecs/vp8/default_temporal_layers.h           |  59 +-
>- .../codecs/vp8/default_temporal_layers_unittest.cc | 276 +++++++--
>- .../modules/video_coding/codecs/vp8/include/vp8.h  |  16 +-
>- .../codecs/vp8/include/vp8_common_types.h          |  10 +-
>- .../video_coding/codecs/vp8/include/vp8_globals.h  |   8 +-
>- .../video_coding/codecs/vp8/screenshare_layers.cc  | 103 ++--
>- .../video_coding/codecs/vp8/screenshare_layers.h   |  33 +-
>- .../codecs/vp8/screenshare_layers_unittest.cc      |  74 ++-
>- .../codecs/vp8/simulcast_rate_allocator.cc         |   4 +-
>- .../codecs/vp8/simulcast_rate_allocator.h          |  14 +-
>- .../video_coding/codecs/vp8/simulcast_unittest.cc  |  18 +-
>- .../video_coding/codecs/vp8/temporal_layers.h      | 119 +++-
>- .../codecs/vp8/test/vp8_impl_unittest.cc           | 628 +++++++++++----------
>- .../modules/video_coding/codecs/vp8/vp8_impl.cc    | 246 +++++---
>- .../modules/video_coding/codecs/vp8/vp8_impl.h     |  34 +-
>- .../modules/video_coding/codecs/vp8/vp8_noop.cc    |   8 +-
>- 17 files changed, 1364 insertions(+), 640 deletions(-)
>-
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.cc
>-index 3573aa8f1cb..f070750e7bc 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.cc
>-@@ -1,31 +1,36 @@
>- /* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
>--*
>--*  Use of this source code is governed by a BSD-style license
>--*  that can be found in the LICENSE file in the root of the source
>--*  tree. An additional intellectual property rights grant can be found
>--*  in the file PATENTS.  All contributing project authors may
>--*  be found in the AUTHORS file in the root of the source tree.
>--*/
>-+ *
>-+ *  Use of this source code is governed by a BSD-style license
>-+ *  that can be found in the LICENSE file in the root of the source
>-+ *  tree. An additional intellectual property rights grant can be found
>-+ *  in the file PATENTS.  All contributing project authors may
>-+ *  be found in the AUTHORS file in the root of the source tree.
>-+ */
>- 
>--#include "webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.h"
>-+#include "modules/video_coding/codecs/vp8/default_temporal_layers.h"
>- 
>- #include <stdlib.h>
>- #include <string.h>
>- 
>- #include <algorithm>
>-+#include <memory>
>-+#include <set>
>- #include <vector>
>- 
>--#include "webrtc/base/checks.h"
>--#include "webrtc/modules/include/module_common_types.h"
>--#include "webrtc/modules/video_coding/include/video_codec_interface.h"
>--#include "webrtc/modules/video_coding/codecs/vp8/include/vp8_common_types.h"
>-+#include "modules/include/module_common_types.h"
>-+#include "modules/video_coding/codecs/vp8/include/vp8_common_types.h"
>-+#include "modules/video_coding/include/video_codec_interface.h"
>-+#include "rtc_base/checks.h"
>-+#include "rtc_base/logging.h"
>-+#include "system_wrappers/include/field_trial.h"
>- 
>--#include "vpx/vpx_encoder.h"
>- #include "vpx/vp8cx.h"
>-+#include "vpx/vpx_encoder.h"
>- 
>- namespace webrtc {
>- 
>--TemporalLayers::FrameConfig::FrameConfig() {}
>-+TemporalLayers::FrameConfig::FrameConfig()
>-+    : FrameConfig(kNone, kNone, kNone, false) {}
>- 
>- TemporalLayers::FrameConfig::FrameConfig(TemporalLayers::BufferFlags last,
>-                                          TemporalLayers::BufferFlags golden,
>-@@ -48,6 +53,9 @@ TemporalLayers::FrameConfig::FrameConfig(TemporalLayers::BufferFlags last,
>-       last_buffer_flags(last),
>-       golden_buffer_flags(golden),
>-       arf_buffer_flags(arf),
>-+      encoder_layer_id(0),
>-+      packetizer_temporal_idx(kNoTemporalIdx),
>-+      layer_sync(false),
>-       freeze_entropy(freeze_entropy) {}
>- 
>- namespace {
>-@@ -91,16 +99,19 @@ std::vector<bool> GetTemporalLayerSync(size_t num_layers) {
>-     case 2:
>-       return {false, true, false, false, false, false, false, false};
>-     case 3:
>--      return {false, true, true, false, false, false, false, false};
>-+      if (field_trial::IsEnabled("WebRTC-UseShortVP8TL3Pattern")) {
>-+        return {false, true, true, false};
>-+      } else {
>-+        return {false, true, true, false, false, false, false, false};
>-+      }
>-     case 4:
>--      return {false, true, true,  true, true,  true, false, true,
>--              false, true, false, true, false, true, false, true};
>-+      return {false, true,  true,  false, true,  false, false, false,
>-+              false, false, false, false, false, false, false, false};
>-     default:
>--      RTC_NOTREACHED();
>-       break;
>-   }
>--  RTC_NOTREACHED();
>--  return {false};
>-+  RTC_NOTREACHED() << num_layers;
>-+  return {};
>- }
>- 
>- std::vector<TemporalLayers::FrameConfig> GetTemporalPattern(size_t num_layers) {
>-@@ -129,7 +140,7 @@ std::vector<TemporalLayers::FrameConfig> GetTemporalPattern(size_t num_layers) {
>-       // TL0 also references and updates the 'last' buffer.
>-       // TL1 also references 'last' and references and updates 'golden'.
>-       return {TemporalLayers::FrameConfig(TemporalLayers::kReferenceAndUpdate,
>--                                          TemporalLayers::kUpdate,
>-+                                          TemporalLayers::kNone,
>-                                           TemporalLayers::kReference),
>-               TemporalLayers::FrameConfig(TemporalLayers::kReference,
>-                                           TemporalLayers::kUpdate,
>-@@ -153,35 +164,68 @@ std::vector<TemporalLayers::FrameConfig> GetTemporalPattern(size_t num_layers) {
>-                   TemporalLayers::kReference, TemporalLayers::kReference,
>-                   TemporalLayers::kReference, TemporalLayers::kFreezeEntropy)};
>-     case 3:
>--      // All layers can reference but not update the 'alt' buffer, this means
>--      // that the 'alt' buffer reference is effectively the last keyframe.
>--      // TL0 also references and updates the 'last' buffer.
>--      // TL1 also references 'last' and references and updates 'golden'.
>--      // TL2 references both 'last' and 'golden' but updates no buffer.
>--      return {TemporalLayers::FrameConfig(TemporalLayers::kReferenceAndUpdate,
>--                                          TemporalLayers::kUpdate,
>--                                          TemporalLayers::kReference),
>--              TemporalLayers::FrameConfig(
>--                  TemporalLayers::kReference, TemporalLayers::kNone,
>--                  TemporalLayers::kReference, TemporalLayers::kFreezeEntropy),
>--              TemporalLayers::FrameConfig(TemporalLayers::kReference,
>--                                          TemporalLayers::kUpdate,
>--                                          TemporalLayers::kReference),
>--              TemporalLayers::FrameConfig(
>--                  TemporalLayers::kReference, TemporalLayers::kReference,
>--                  TemporalLayers::kReference, TemporalLayers::kFreezeEntropy),
>--              TemporalLayers::FrameConfig(TemporalLayers::kReferenceAndUpdate,
>--                                          TemporalLayers::kNone,
>--                                          TemporalLayers::kReference),
>--              TemporalLayers::FrameConfig(
>--                  TemporalLayers::kReference, TemporalLayers::kReference,
>--                  TemporalLayers::kReference, TemporalLayers::kFreezeEntropy),
>--              TemporalLayers::FrameConfig(TemporalLayers::kReference,
>--                                          TemporalLayers::kReferenceAndUpdate,
>--                                          TemporalLayers::kReference),
>--              TemporalLayers::FrameConfig(
>--                  TemporalLayers::kReference, TemporalLayers::kReference,
>--                  TemporalLayers::kReference, TemporalLayers::kFreezeEntropy)};
>-+      if (field_trial::IsEnabled("WebRTC-UseShortVP8TL3Pattern")) {
>-+        // This field trial is intended to check if it is worth using a shorter
>-+        // temporal pattern, trading some coding efficiency for less risk of
>-+        // dropped frames.
>-+        // The coding efficiency will decrease somewhat since the higher layer
>-+        // state is more volatile, but it will be offset slightly by updating
>-+        // the altref buffer with TL2 frames, instead of just referencing lower
>-+        // layers.
>-+        // If a frame is dropped in a higher layer, the jitter
>-+        // buffer on the receive side won't be able to decode any higher layer
>-+        // frame until the next sync frame. So we expect a noticeable decrease
>-+        // in frame drops on links with high packet loss.
>-+
>-+        // TL0 references and updates the 'last' buffer.
>-+        // TL1  references 'last' and references and updates 'golden'.
>-+        // TL2 references both 'last' & 'golden' and references and updates
>-+        // 'arf'.
>-+        return {TemporalLayers::FrameConfig(TemporalLayers::kReferenceAndUpdate,
>-+                                            TemporalLayers::kNone,
>-+                                            TemporalLayers::kNone),
>-+                TemporalLayers::FrameConfig(TemporalLayers::kReference,
>-+                                            TemporalLayers::kNone,
>-+                                            TemporalLayers::kUpdate),
>-+                TemporalLayers::FrameConfig(TemporalLayers::kReference,
>-+                                            TemporalLayers::kUpdate,
>-+                                            TemporalLayers::kNone),
>-+                TemporalLayers::FrameConfig(TemporalLayers::kReference,
>-+                                            TemporalLayers::kReference,
>-+                                            TemporalLayers::kReference,
>-+                                            TemporalLayers::kFreezeEntropy)};
>-+      } else {
>-+        // All layers can reference but not update the 'alt' buffer, this means
>-+        // that the 'alt' buffer reference is effectively the last keyframe.
>-+        // TL0 also references and updates the 'last' buffer.
>-+        // TL1 also references 'last' and references and updates 'golden'.
>-+        // TL2 references both 'last' and 'golden' but updates no buffer.
>-+        return {TemporalLayers::FrameConfig(TemporalLayers::kReferenceAndUpdate,
>-+                                            TemporalLayers::kNone,
>-+                                            TemporalLayers::kReference),
>-+                TemporalLayers::FrameConfig(
>-+                    TemporalLayers::kReference, TemporalLayers::kNone,
>-+                    TemporalLayers::kReference, TemporalLayers::kFreezeEntropy),
>-+                TemporalLayers::FrameConfig(TemporalLayers::kReference,
>-+                                            TemporalLayers::kUpdate,
>-+                                            TemporalLayers::kReference),
>-+                TemporalLayers::FrameConfig(
>-+                    TemporalLayers::kReference, TemporalLayers::kReference,
>-+                    TemporalLayers::kReference, TemporalLayers::kFreezeEntropy),
>-+                TemporalLayers::FrameConfig(TemporalLayers::kReferenceAndUpdate,
>-+                                            TemporalLayers::kNone,
>-+                                            TemporalLayers::kReference),
>-+                TemporalLayers::FrameConfig(
>-+                    TemporalLayers::kReference, TemporalLayers::kReference,
>-+                    TemporalLayers::kReference, TemporalLayers::kFreezeEntropy),
>-+                TemporalLayers::FrameConfig(TemporalLayers::kReference,
>-+                                            TemporalLayers::kReferenceAndUpdate,
>-+                                            TemporalLayers::kReference),
>-+                TemporalLayers::FrameConfig(TemporalLayers::kReference,
>-+                                            TemporalLayers::kReference,
>-+                                            TemporalLayers::kReference,
>-+                                            TemporalLayers::kFreezeEntropy)};
>-+      }
>-     case 4:
>-       // TL0 references and updates only the 'last' buffer.
>-       // TL1 references 'last' and updates and references 'golden'.
>-@@ -191,13 +235,13 @@ std::vector<TemporalLayers::FrameConfig> GetTemporalPattern(size_t num_layers) {
>-                                           TemporalLayers::kNone,
>-                                           TemporalLayers::kNone),
>-               TemporalLayers::FrameConfig(
>--                  TemporalLayers::kReference, TemporalLayers::kReference,
>--                  TemporalLayers::kReference, TemporalLayers::kFreezeEntropy),
>-+                  TemporalLayers::kReference, TemporalLayers::kNone,
>-+                  TemporalLayers::kNone, TemporalLayers::kFreezeEntropy),
>-               TemporalLayers::FrameConfig(TemporalLayers::kReference,
>-                                           TemporalLayers::kNone,
>-                                           TemporalLayers::kUpdate),
>-               TemporalLayers::FrameConfig(
>--                  TemporalLayers::kReference, TemporalLayers::kReference,
>-+                  TemporalLayers::kReference, TemporalLayers::kNone,
>-                   TemporalLayers::kReference, TemporalLayers::kFreezeEntropy),
>-               TemporalLayers::FrameConfig(TemporalLayers::kReference,
>-                                           TemporalLayers::kUpdate,
>-@@ -244,6 +288,17 @@ std::vector<TemporalLayers::FrameConfig> GetTemporalPattern(size_t num_layers) {
>-       TemporalLayers::kNone, TemporalLayers::kNone, TemporalLayers::kNone)};
>- }
>- 
>-+// Temporary fix for forced SW fallback.
>-+// For VP8 SW codec, |TemporalLayers| is created and reported to
>-+// SimulcastRateAllocator::OnTemporalLayersCreated but not for VP8 HW.
>-+// Causes an issue when going from forced SW -> HW as |TemporalLayers| is not
>-+// deregistred when deleted by SW codec (tl factory might not exist, owned by
>-+// SimulcastRateAllocator).
>-+bool ExcludeOnTemporalLayersCreated(int num_temporal_layers) {
>-+  return webrtc::field_trial::IsEnabled(
>-+             "WebRTC-VP8-Forced-Fallback-Encoder-v2") &&
>-+         num_temporal_layers == 1;
>-+}
>- }  // namespace
>- 
>- DefaultTemporalLayers::DefaultTemporalLayers(int number_of_temporal_layers,
>-@@ -254,7 +309,6 @@ DefaultTemporalLayers::DefaultTemporalLayers(int number_of_temporal_layers,
>-       temporal_pattern_(GetTemporalPattern(num_layers_)),
>-       tl0_pic_idx_(initial_tl0_pic_idx),
>-       pattern_idx_(255),
>--      timestamp_(0),
>-       last_base_layer_sync_(false) {
>-   RTC_DCHECK_EQ(temporal_pattern_.size(), temporal_layer_sync_.size());
>-   RTC_CHECK_GE(kMaxTemporalStreams, number_of_temporal_layers);
>-@@ -266,12 +320,6 @@ DefaultTemporalLayers::DefaultTemporalLayers(int number_of_temporal_layers,
>-   RTC_DCHECK_LE(temporal_ids_.size(), temporal_pattern_.size());
>- }
>- 
>--int DefaultTemporalLayers::GetTemporalLayerId(
>--    const TemporalLayers::FrameConfig& tl_config) const {
>--  RTC_DCHECK(!tl_config.drop_frame);
>--  return temporal_ids_[tl_config.pattern_idx % temporal_ids_.size()];
>--}
>--
>- uint8_t DefaultTemporalLayers::Tl0PicIdx() const {
>-   return tl0_pic_idx_;
>- }
>-@@ -332,7 +380,10 @@ TemporalLayers::FrameConfig DefaultTemporalLayers::UpdateLayerConfig(
>-   RTC_DCHECK_LT(0, temporal_pattern_.size());
>-   pattern_idx_ = (pattern_idx_ + 1) % temporal_pattern_.size();
>-   TemporalLayers::FrameConfig tl_config = temporal_pattern_[pattern_idx_];
>--  tl_config.pattern_idx = pattern_idx_;
>-+  tl_config.layer_sync =
>-+      temporal_layer_sync_[pattern_idx_ % temporal_layer_sync_.size()];
>-+  tl_config.encoder_layer_id = tl_config.packetizer_temporal_idx =
>-+      temporal_ids_[pattern_idx_ % temporal_ids_.size()];
>-   return tl_config;
>- }
>- 
>-@@ -348,24 +399,19 @@ void DefaultTemporalLayers::PopulateCodecSpecific(
>-     vp8_info->layerSync = false;
>-     vp8_info->tl0PicIdx = kNoTl0PicIdx;
>-   } else {
>-+    vp8_info->temporalIdx = tl_config.packetizer_temporal_idx;
>-+    vp8_info->layerSync = tl_config.layer_sync;
>-     if (frame_is_keyframe) {
>-       vp8_info->temporalIdx = 0;
>-       vp8_info->layerSync = true;
>--    } else {
>--      vp8_info->temporalIdx = GetTemporalLayerId(tl_config);
>--
>--      vp8_info->layerSync = temporal_layer_sync_[tl_config.pattern_idx %
>--                                                 temporal_layer_sync_.size()];
>-     }
>-     if (last_base_layer_sync_ && vp8_info->temporalIdx != 0) {
>-       // Regardless of pattern the frame after a base layer sync will always
>-       // be a layer sync.
>-       vp8_info->layerSync = true;
>-     }
>--    if (vp8_info->temporalIdx == 0 && timestamp != timestamp_) {
>--      timestamp_ = timestamp;
>-+    if (vp8_info->temporalIdx == 0)
>-       tl0_pic_idx_++;
>--    }
>-     last_base_layer_sync_ = frame_is_keyframe;
>-     vp8_info->tl0PicIdx = tl0_pic_idx_;
>-   }
>-@@ -377,13 +423,181 @@ TemporalLayers* TemporalLayersFactory::Create(
>-     uint8_t initial_tl0_pic_idx) const {
>-   TemporalLayers* tl =
>-       new DefaultTemporalLayers(temporal_layers, initial_tl0_pic_idx);
>--  if (listener_)
>-+  if (listener_ && !ExcludeOnTemporalLayersCreated(temporal_layers))
>-     listener_->OnTemporalLayersCreated(simulcast_id, tl);
>-   return tl;
>- }
>- 
>-+std::unique_ptr<TemporalLayersChecker> TemporalLayersFactory::CreateChecker(
>-+    int /*simulcast_id*/,
>-+    int temporal_layers,
>-+    uint8_t initial_tl0_pic_idx) const {
>-+  TemporalLayersChecker* tlc =
>-+      new DefaultTemporalLayersChecker(temporal_layers, initial_tl0_pic_idx);
>-+  return std::unique_ptr<TemporalLayersChecker>(tlc);
>-+}
>-+
>- void TemporalLayersFactory::SetListener(TemporalLayersListener* listener) {
>-   listener_ = listener;
>- }
>- 
>-+// Returns list of temporal dependencies for each frame in the temporal pattern.
>-+// Values are lists of indecies in the pattern.
>-+std::vector<std::set<uint8_t>> GetTemporalDependencies(
>-+    int num_temporal_layers) {
>-+  switch (num_temporal_layers) {
>-+    case 1:
>-+      return {{0}};
>-+    case 2:
>-+      return {{6}, {0}, {0}, {1, 2}, {2}, {3, 4}, {4}, {5, 6}};
>-+    case 3:
>-+      if (field_trial::IsEnabled("WebRTC-UseShortVP8TL3Pattern")) {
>-+        return {{0}, {0}, {0}, {0, 1, 2}};
>-+      } else {
>-+        return {{4}, {0}, {0}, {0, 2}, {0}, {2, 4}, {2, 4}, {4, 6}};
>-+      }
>-+    case 4:
>-+      return {{8},    {0},         {0},         {0, 2},
>-+              {0},    {0, 2, 4},   {0, 2, 4},   {0, 4, 6},
>-+              {0},    {4, 6, 8},   {4, 6, 8},   {4, 8, 10},
>-+              {4, 8}, {8, 10, 12}, {8, 10, 12}, {8, 12, 14}};
>-+    default:
>-+      RTC_NOTREACHED();
>-+      return {};
>-+  }
>-+}
>-+
>-+DefaultTemporalLayersChecker::DefaultTemporalLayersChecker(
>-+    int num_temporal_layers,
>-+    uint8_t initial_tl0_pic_idx)
>-+    : TemporalLayersChecker(num_temporal_layers, initial_tl0_pic_idx),
>-+      num_layers_(std::max(1, num_temporal_layers)),
>-+      temporal_ids_(GetTemporalIds(num_layers_)),
>-+      temporal_dependencies_(GetTemporalDependencies(num_layers_)),
>-+      pattern_idx_(255) {
>-+  int i = 0;
>-+  while (temporal_ids_.size() < temporal_dependencies_.size()) {
>-+    temporal_ids_.push_back(temporal_ids_[i++]);
>-+  }
>-+}
>-+
>-+bool DefaultTemporalLayersChecker::CheckTemporalConfig(
>-+    bool frame_is_keyframe,
>-+    const TemporalLayers::FrameConfig& frame_config) {
>-+  if (!TemporalLayersChecker::CheckTemporalConfig(frame_is_keyframe,
>-+                                                  frame_config)) {
>-+    return false;
>-+  }
>-+  if (frame_config.drop_frame) {
>-+    return true;
>-+  }
>-+  ++pattern_idx_;
>-+  if (pattern_idx_ == temporal_ids_.size()) {
>-+    // All non key-frame buffers should be updated each pattern cycle.
>-+    if (!last_.is_keyframe && !last_.is_updated_this_cycle) {
>-+      RTC_LOG(LS_ERROR) << "Last buffer was not updated during pattern cycle.";
>-+      return false;
>-+    }
>-+    if (!arf_.is_keyframe && !arf_.is_updated_this_cycle) {
>-+      RTC_LOG(LS_ERROR) << "Arf buffer was not updated during pattern cycle.";
>-+      return false;
>-+    }
>-+    if (!golden_.is_keyframe && !golden_.is_updated_this_cycle) {
>-+      RTC_LOG(LS_ERROR)
>-+          << "Golden buffer was not updated during pattern cycle.";
>-+      return false;
>-+    }
>-+    last_.is_updated_this_cycle = false;
>-+    arf_.is_updated_this_cycle = false;
>-+    golden_.is_updated_this_cycle = false;
>-+    pattern_idx_ = 0;
>-+  }
>-+  uint8_t expected_tl_idx = temporal_ids_[pattern_idx_];
>-+  if (frame_config.packetizer_temporal_idx != expected_tl_idx) {
>-+    RTC_LOG(LS_ERROR) << "Frame has an incorrect temporal index. Expected: "
>-+                      << static_cast<int>(expected_tl_idx) << " Actual: "
>-+                      << static_cast<int>(frame_config.packetizer_temporal_idx);
>-+    return false;
>-+  }
>-+
>-+  bool need_sync = temporal_ids_[pattern_idx_] > 0 &&
>-+                   temporal_ids_[pattern_idx_] != kNoTemporalIdx;
>-+  std::vector<int> dependencies;
>-+
>-+  if (frame_config.last_buffer_flags &
>-+      TemporalLayers::BufferFlags::kReference) {
>-+    uint8_t referenced_layer = temporal_ids_[last_.pattern_idx];
>-+    if (referenced_layer > 0) {
>-+      need_sync = false;
>-+    }
>-+    if (!last_.is_keyframe) {
>-+      dependencies.push_back(last_.pattern_idx);
>-+    }
>-+  }
>-+
>-+  if (frame_config.arf_buffer_flags & TemporalLayers::BufferFlags::kReference) {
>-+    uint8_t referenced_layer = temporal_ids_[arf_.pattern_idx];
>-+    if (referenced_layer > 0) {
>-+      need_sync = false;
>-+    }
>-+    if (!arf_.is_keyframe) {
>-+      dependencies.push_back(arf_.pattern_idx);
>-+    }
>-+  }
>-+
>-+  if (frame_config.golden_buffer_flags &
>-+      TemporalLayers::BufferFlags::kReference) {
>-+    uint8_t referenced_layer = temporal_ids_[golden_.pattern_idx];
>-+    if (referenced_layer > 0) {
>-+      need_sync = false;
>-+    }
>-+    if (!golden_.is_keyframe) {
>-+      dependencies.push_back(golden_.pattern_idx);
>-+    }
>-+  }
>-+
>-+  if (need_sync != frame_config.layer_sync) {
>-+    RTC_LOG(LS_ERROR) << "Sync bit is set incorrectly on a frame. Expected: "
>-+                      << need_sync << " Actual: " << frame_config.layer_sync;
>-+    return false;
>-+  }
>-+
>-+  if (!frame_is_keyframe) {
>-+    size_t i;
>-+    for (i = 0; i < dependencies.size(); ++i) {
>-+      if (temporal_dependencies_[pattern_idx_].find(dependencies[i]) ==
>-+          temporal_dependencies_[pattern_idx_].end()) {
>-+        RTC_LOG(LS_ERROR)
>-+            << "Illegal temporal dependency out of defined pattern "
>-+               "from position "
>-+            << static_cast<int>(pattern_idx_) << " to position "
>-+            << static_cast<int>(dependencies[i]);
>-+        return false;
>-+      }
>-+    }
>-+  }
>-+
>-+  if (frame_config.last_buffer_flags & TemporalLayers::BufferFlags::kUpdate) {
>-+    last_.is_updated_this_cycle = true;
>-+    last_.pattern_idx = pattern_idx_;
>-+    last_.is_keyframe = false;
>-+  }
>-+  if (frame_config.arf_buffer_flags & TemporalLayers::BufferFlags::kUpdate) {
>-+    arf_.is_updated_this_cycle = true;
>-+    arf_.pattern_idx = pattern_idx_;
>-+    arf_.is_keyframe = false;
>-+  }
>-+  if (frame_config.golden_buffer_flags & TemporalLayers::BufferFlags::kUpdate) {
>-+    golden_.is_updated_this_cycle = true;
>-+    golden_.pattern_idx = pattern_idx_;
>-+    golden_.is_keyframe = false;
>-+  }
>-+  if (frame_is_keyframe) {
>-+    last_.is_keyframe = true;
>-+    arf_.is_keyframe = true;
>-+    golden_.is_keyframe = true;
>-+  }
>-+  return true;
>-+}
>-+
>- }  // namespace webrtc
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.h
>-index 4681fe9dc84..f1925569209 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.h
>-@@ -1,22 +1,23 @@
>- /* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
>--*
>--*  Use of this source code is governed by a BSD-style license
>--*  that can be found in the LICENSE file in the root of the source
>--*  tree. An additional intellectual property rights grant can be found
>--*  in the file PATENTS.  All contributing project authors may
>--*  be found in the AUTHORS file in the root of the source tree.
>--*/
>-+ *
>-+ *  Use of this source code is governed by a BSD-style license
>-+ *  that can be found in the LICENSE file in the root of the source
>-+ *  tree. An additional intellectual property rights grant can be found
>-+ *  in the file PATENTS.  All contributing project authors may
>-+ *  be found in the AUTHORS file in the root of the source tree.
>-+ */
>- /*
>--* This file defines classes for doing temporal layers with VP8.
>--*/
>--#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_DEFAULT_TEMPORAL_LAYERS_H_
>--#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_DEFAULT_TEMPORAL_LAYERS_H_
>-+ * This file defines classes for doing temporal layers with VP8.
>-+ */
>-+#ifndef MODULES_VIDEO_CODING_CODECS_VP8_DEFAULT_TEMPORAL_LAYERS_H_
>-+#define MODULES_VIDEO_CODING_CODECS_VP8_DEFAULT_TEMPORAL_LAYERS_H_
>- 
>-+#include <set>
>- #include <vector>
>- 
>--#include "webrtc/modules/video_coding/codecs/vp8/temporal_layers.h"
>-+#include "modules/video_coding/codecs/vp8/temporal_layers.h"
>- 
>--#include "webrtc/base/optional.h"
>-+#include "api/optional.h"
>- 
>- namespace webrtc {
>- 
>-@@ -45,9 +46,6 @@ class DefaultTemporalLayers : public TemporalLayers {
>- 
>-   void FrameEncoded(unsigned int size, int qp) override {}
>- 
>--  int GetTemporalLayerId(
>--      const TemporalLayers::FrameConfig& references) const override;
>--
>-   uint8_t Tl0PicIdx() const override;
>- 
>-  private:
>-@@ -58,10 +56,35 @@ class DefaultTemporalLayers : public TemporalLayers {
>- 
>-   uint8_t tl0_pic_idx_;
>-   uint8_t pattern_idx_;
>--  uint32_t timestamp_;
>-   bool last_base_layer_sync_;
>-   rtc::Optional<std::vector<uint32_t>> new_bitrates_kbps_;
>- };
>- 
>-+class DefaultTemporalLayersChecker : public TemporalLayersChecker {
>-+ public:
>-+  DefaultTemporalLayersChecker(int number_of_temporal_layers,
>-+                               uint8_t initial_tl0_pic_idx);
>-+  bool CheckTemporalConfig(
>-+      bool frame_is_keyframe,
>-+      const TemporalLayers::FrameConfig& frame_config) override;
>-+
>-+ private:
>-+  struct BufferState {
>-+    BufferState()
>-+        : is_updated_this_cycle(false), is_keyframe(true), pattern_idx(0) {}
>-+
>-+    bool is_updated_this_cycle;
>-+    bool is_keyframe;
>-+    uint8_t pattern_idx;
>-+  };
>-+  const size_t num_layers_;
>-+  std::vector<unsigned int> temporal_ids_;
>-+  const std::vector<std::set<uint8_t>> temporal_dependencies_;
>-+  BufferState last_;
>-+  BufferState arf_;
>-+  BufferState golden_;
>-+  uint8_t pattern_idx_;
>-+};
>-+
>- }  // namespace webrtc
>--#endif  // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_DEFAULT_TEMPORAL_LAYERS_H_
>-+#endif  // MODULES_VIDEO_CODING_CODECS_VP8_DEFAULT_TEMPORAL_LAYERS_H_
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers_unittest.cc
>-index edab34c16dd..1c1d63f1f8d 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers_unittest.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/default_temporal_layers_unittest.cc
>-@@ -8,19 +8,20 @@
>-  *  be found in the AUTHORS file in the root of the source tree.
>-  */
>- 
>--#include "webrtc/modules/video_coding/codecs/vp8/default_temporal_layers.h"
>-+#include "modules/video_coding/codecs/vp8/default_temporal_layers.h"
>-+#include "modules/video_coding/codecs/vp8/vp8_impl.h"
>-+#include "modules/video_coding/include/video_codec_interface.h"
>-+#include "test/field_trial.h"
>-+#include "test/gtest.h"
>- #include "vpx/vp8cx.h"
>- #include "vpx/vpx_encoder.h"
>--#include "webrtc/modules/video_coding/codecs/vp8/vp8_impl.h"
>--#include "webrtc/modules/video_coding/include/video_codec_interface.h"
>--#include "webrtc/test/gtest.h"
>- 
>- namespace webrtc {
>-+namespace test {
>- 
>- enum {
>-   kTemporalUpdateLast = VP8_EFLAG_NO_UPD_GF | VP8_EFLAG_NO_UPD_ARF |
>--                        VP8_EFLAG_NO_REF_GF |
>--                        VP8_EFLAG_NO_REF_ARF,
>-+                        VP8_EFLAG_NO_REF_GF | VP8_EFLAG_NO_REF_ARF,
>-   kTemporalUpdateGoldenWithoutDependency =
>-       VP8_EFLAG_NO_REF_GF | VP8_EFLAG_NO_REF_ARF | VP8_EFLAG_NO_UPD_ARF |
>-       VP8_EFLAG_NO_UPD_LAST,
>-@@ -31,16 +32,16 @@ enum {
>-       VP8_EFLAG_NO_UPD_LAST,
>-   kTemporalUpdateAltref = VP8_EFLAG_NO_UPD_GF | VP8_EFLAG_NO_UPD_LAST,
>-   kTemporalUpdateNone = VP8_EFLAG_NO_UPD_GF | VP8_EFLAG_NO_UPD_ARF |
>--                        VP8_EFLAG_NO_UPD_LAST |
>--                        VP8_EFLAG_NO_UPD_ENTROPY,
>--  kTemporalUpdateNoneNoRefAltRef = VP8_EFLAG_NO_REF_ARF | VP8_EFLAG_NO_UPD_GF |
>--                                   VP8_EFLAG_NO_UPD_ARF |
>--                                   VP8_EFLAG_NO_UPD_LAST |
>--                                   VP8_EFLAG_NO_UPD_ENTROPY,
>--  kTemporalUpdateNoneNoRefGolden = VP8_EFLAG_NO_REF_GF | VP8_EFLAG_NO_UPD_GF |
>--                                   VP8_EFLAG_NO_UPD_ARF |
>--                                   VP8_EFLAG_NO_UPD_LAST |
>--                                   VP8_EFLAG_NO_UPD_ENTROPY,
>-+                        VP8_EFLAG_NO_UPD_LAST | VP8_EFLAG_NO_UPD_ENTROPY,
>-+  kTemporalUpdateNoneNoRefAltRef =
>-+      VP8_EFLAG_NO_REF_ARF | VP8_EFLAG_NO_UPD_GF | VP8_EFLAG_NO_UPD_ARF |
>-+      VP8_EFLAG_NO_UPD_LAST | VP8_EFLAG_NO_UPD_ENTROPY,
>-+  kTemporalUpdateNoneNoRefGolden =
>-+      VP8_EFLAG_NO_REF_GF | VP8_EFLAG_NO_UPD_GF | VP8_EFLAG_NO_UPD_ARF |
>-+      VP8_EFLAG_NO_UPD_LAST | VP8_EFLAG_NO_UPD_ENTROPY,
>-+  kTemporalUpdateNoneNoRefGoldenAltRef =
>-+      VP8_EFLAG_NO_REF_GF | VP8_EFLAG_NO_UPD_GF | VP8_EFLAG_NO_REF_ARF |
>-+      VP8_EFLAG_NO_UPD_ARF | VP8_EFLAG_NO_UPD_LAST | VP8_EFLAG_NO_UPD_ENTROPY,
>-   kTemporalUpdateGoldenWithoutDependencyRefAltRef =
>-       VP8_EFLAG_NO_REF_GF | VP8_EFLAG_NO_UPD_ARF | VP8_EFLAG_NO_UPD_LAST,
>-   kTemporalUpdateGoldenRefAltRef = VP8_EFLAG_NO_UPD_ARF | VP8_EFLAG_NO_UPD_LAST,
>-@@ -52,13 +53,14 @@ enum {
>- 
>- TEST(TemporalLayersTest, 2Layers) {
>-   DefaultTemporalLayers tl(2, 0);
>-+  DefaultTemporalLayersChecker checker(2, 0);
>-   vpx_codec_enc_cfg_t cfg;
>-   CodecSpecificInfoVP8 vp8_info;
>-   tl.OnRatesUpdated(500, 500, 30);
>-   tl.UpdateConfiguration(&cfg);
>- 
>-   int expected_flags[16] = {
>--      kTemporalUpdateLastAndGoldenRefAltRef,
>-+      kTemporalUpdateLastRefAltRef,
>-       kTemporalUpdateGoldenWithoutDependencyRefAltRef,
>-       kTemporalUpdateLastRefAltRef,
>-       kTemporalUpdateGoldenRefAltRef,
>-@@ -66,7 +68,7 @@ TEST(TemporalLayersTest, 2Layers) {
>-       kTemporalUpdateGoldenRefAltRef,
>-       kTemporalUpdateLastRefAltRef,
>-       kTemporalUpdateNone,
>--      kTemporalUpdateLastAndGoldenRefAltRef,
>-+      kTemporalUpdateLastRefAltRef,
>-       kTemporalUpdateGoldenWithoutDependencyRefAltRef,
>-       kTemporalUpdateLastRefAltRef,
>-       kTemporalUpdateGoldenRefAltRef,
>-@@ -85,23 +87,28 @@ TEST(TemporalLayersTest, 2Layers) {
>-   uint32_t timestamp = 0;
>-   for (int i = 0; i < 16; ++i) {
>-     TemporalLayers::FrameConfig tl_config = tl.UpdateLayerConfig(timestamp);
>--    EXPECT_EQ(expected_flags[i], VP8EncoderImpl::EncodeFlags(tl_config));
>--    tl.PopulateCodecSpecific(false, tl_config, &vp8_info, 0);
>-+    EXPECT_EQ(expected_flags[i], VP8EncoderImpl::EncodeFlags(tl_config)) << i;
>-+    tl.PopulateCodecSpecific(i == 0, tl_config, &vp8_info, 0);
>-+    EXPECT_TRUE(checker.CheckTemporalConfig(i == 0, tl_config));
>-     EXPECT_EQ(expected_temporal_idx[i], vp8_info.temporalIdx);
>--    EXPECT_EQ(expected_layer_sync[i], vp8_info.layerSync);
>-+    EXPECT_EQ(expected_temporal_idx[i], tl_config.packetizer_temporal_idx);
>-+    EXPECT_EQ(expected_temporal_idx[i], tl_config.encoder_layer_id);
>-+    EXPECT_EQ(i == 0 || expected_layer_sync[i], vp8_info.layerSync);
>-+    EXPECT_EQ(expected_layer_sync[i], tl_config.layer_sync);
>-     timestamp += 3000;
>-   }
>- }
>- 
>- TEST(TemporalLayersTest, 3Layers) {
>-   DefaultTemporalLayers tl(3, 0);
>-+  DefaultTemporalLayersChecker checker(3, 0);
>-   vpx_codec_enc_cfg_t cfg;
>-   CodecSpecificInfoVP8 vp8_info;
>-   tl.OnRatesUpdated(500, 500, 30);
>-   tl.UpdateConfiguration(&cfg);
>- 
>-   int expected_flags[16] = {
>--      kTemporalUpdateLastAndGoldenRefAltRef,
>-+      kTemporalUpdateLastRefAltRef,
>-       kTemporalUpdateNoneNoRefGolden,
>-       kTemporalUpdateGoldenWithoutDependencyRefAltRef,
>-       kTemporalUpdateNone,
>-@@ -109,7 +116,7 @@ TEST(TemporalLayersTest, 3Layers) {
>-       kTemporalUpdateNone,
>-       kTemporalUpdateGoldenRefAltRef,
>-       kTemporalUpdateNone,
>--      kTemporalUpdateLastAndGoldenRefAltRef,
>-+      kTemporalUpdateLastRefAltRef,
>-       kTemporalUpdateNoneNoRefGolden,
>-       kTemporalUpdateGoldenWithoutDependencyRefAltRef,
>-       kTemporalUpdateNone,
>-@@ -128,25 +135,67 @@ TEST(TemporalLayersTest, 3Layers) {
>-   unsigned int timestamp = 0;
>-   for (int i = 0; i < 16; ++i) {
>-     TemporalLayers::FrameConfig tl_config = tl.UpdateLayerConfig(timestamp);
>--    EXPECT_EQ(expected_flags[i], VP8EncoderImpl::EncodeFlags(tl_config));
>--    tl.PopulateCodecSpecific(false, tl_config, &vp8_info, 0);
>-+    EXPECT_EQ(expected_flags[i], VP8EncoderImpl::EncodeFlags(tl_config)) << i;
>-+    tl.PopulateCodecSpecific(i == 0, tl_config, &vp8_info, 0);
>-+    EXPECT_TRUE(checker.CheckTemporalConfig(i == 0, tl_config));
>-+    EXPECT_EQ(expected_temporal_idx[i], vp8_info.temporalIdx);
>-+    EXPECT_EQ(expected_temporal_idx[i], tl_config.packetizer_temporal_idx);
>-+    EXPECT_EQ(expected_temporal_idx[i], tl_config.encoder_layer_id);
>-+    EXPECT_EQ(i == 0 || expected_layer_sync[i], vp8_info.layerSync);
>-+    EXPECT_EQ(expected_layer_sync[i], tl_config.layer_sync);
>-+    timestamp += 3000;
>-+  }
>-+}
>-+
>-+TEST(TemporalLayersTest, Alternative3Layers) {
>-+  ScopedFieldTrials field_trial("WebRTC-UseShortVP8TL3Pattern/Enabled/");
>-+  DefaultTemporalLayers tl(3, 0);
>-+  DefaultTemporalLayersChecker checker(3, 0);
>-+  vpx_codec_enc_cfg_t cfg;
>-+  CodecSpecificInfoVP8 vp8_info;
>-+  tl.OnRatesUpdated(500, 500, 30);
>-+  tl.UpdateConfiguration(&cfg);
>-+
>-+  int expected_flags[8] = {kTemporalUpdateLast,
>-+                           kTemporalUpdateAltrefWithoutDependency,
>-+                           kTemporalUpdateGoldenWithoutDependency,
>-+                           kTemporalUpdateNone,
>-+                           kTemporalUpdateLast,
>-+                           kTemporalUpdateAltrefWithoutDependency,
>-+                           kTemporalUpdateGoldenWithoutDependency,
>-+                           kTemporalUpdateNone};
>-+  int expected_temporal_idx[8] = {0, 2, 1, 2, 0, 2, 1, 2};
>-+
>-+  bool expected_layer_sync[8] = {false, true, true, false,
>-+                                 false, true, true, false};
>-+
>-+  unsigned int timestamp = 0;
>-+  for (int i = 0; i < 8; ++i) {
>-+    TemporalLayers::FrameConfig tl_config = tl.UpdateLayerConfig(timestamp);
>-+    EXPECT_EQ(expected_flags[i], VP8EncoderImpl::EncodeFlags(tl_config)) << i;
>-+    tl.PopulateCodecSpecific(i == 0, tl_config, &vp8_info, 0);
>-+    EXPECT_TRUE(checker.CheckTemporalConfig(i == 0, tl_config));
>-     EXPECT_EQ(expected_temporal_idx[i], vp8_info.temporalIdx);
>--    EXPECT_EQ(expected_layer_sync[i], vp8_info.layerSync);
>-+    EXPECT_EQ(expected_temporal_idx[i], tl_config.packetizer_temporal_idx);
>-+    EXPECT_EQ(expected_temporal_idx[i], tl_config.encoder_layer_id);
>-+    EXPECT_EQ(i == 0 || expected_layer_sync[i], vp8_info.layerSync);
>-+    EXPECT_EQ(expected_layer_sync[i], tl_config.layer_sync);
>-     timestamp += 3000;
>-   }
>- }
>- 
>- TEST(TemporalLayersTest, 4Layers) {
>-   DefaultTemporalLayers tl(4, 0);
>-+  DefaultTemporalLayersChecker checker(4, 0);
>-   vpx_codec_enc_cfg_t cfg;
>-   CodecSpecificInfoVP8 vp8_info;
>-   tl.OnRatesUpdated(500, 500, 30);
>-   tl.UpdateConfiguration(&cfg);
>-   int expected_flags[16] = {
>-       kTemporalUpdateLast,
>--      kTemporalUpdateNone,
>-+      kTemporalUpdateNoneNoRefGoldenAltRef,
>-       kTemporalUpdateAltrefWithoutDependency,
>--      kTemporalUpdateNone,
>-+      kTemporalUpdateNoneNoRefGolden,
>-       kTemporalUpdateGoldenWithoutDependency,
>-       kTemporalUpdateNone,
>-       kTemporalUpdateAltref,
>-@@ -163,30 +212,35 @@ TEST(TemporalLayersTest, 4Layers) {
>-   int expected_temporal_idx[16] = {0, 3, 2, 3, 1, 3, 2, 3,
>-                                    0, 3, 2, 3, 1, 3, 2, 3};
>- 
>--  bool expected_layer_sync[16] = {false, true, true,  true, true,  true,
>--                                  false, true, false, true, false, true,
>--                                  false, true, false, true};
>-+  bool expected_layer_sync[16] = {false, true,  true,  false, true,  false,
>-+                                  false, false, false, false, false, false,
>-+                                  false, false, false, false};
>- 
>-   uint32_t timestamp = 0;
>-   for (int i = 0; i < 16; ++i) {
>-     TemporalLayers::FrameConfig tl_config = tl.UpdateLayerConfig(timestamp);
>--    EXPECT_EQ(expected_flags[i], VP8EncoderImpl::EncodeFlags(tl_config));
>--    tl.PopulateCodecSpecific(false, tl_config, &vp8_info, 0);
>-+    EXPECT_EQ(expected_flags[i], VP8EncoderImpl::EncodeFlags(tl_config)) << i;
>-+    tl.PopulateCodecSpecific(i == 0, tl_config, &vp8_info, 0);
>-+    EXPECT_TRUE(checker.CheckTemporalConfig(i == 0, tl_config));
>-     EXPECT_EQ(expected_temporal_idx[i], vp8_info.temporalIdx);
>--    EXPECT_EQ(expected_layer_sync[i], vp8_info.layerSync);
>-+    EXPECT_EQ(expected_temporal_idx[i], tl_config.packetizer_temporal_idx);
>-+    EXPECT_EQ(expected_temporal_idx[i], tl_config.encoder_layer_id);
>-+    EXPECT_EQ(i == 0 || expected_layer_sync[i], vp8_info.layerSync);
>-+    EXPECT_EQ(expected_layer_sync[i], tl_config.layer_sync);
>-     timestamp += 3000;
>-   }
>- }
>- 
>- TEST(TemporalLayersTest, KeyFrame) {
>-   DefaultTemporalLayers tl(3, 0);
>-+  DefaultTemporalLayersChecker checker(3, 0);
>-   vpx_codec_enc_cfg_t cfg;
>-   CodecSpecificInfoVP8 vp8_info;
>-   tl.OnRatesUpdated(500, 500, 30);
>-   tl.UpdateConfiguration(&cfg);
>- 
>-   int expected_flags[8] = {
>--      kTemporalUpdateLastAndGoldenRefAltRef,
>-+      kTemporalUpdateLastRefAltRef,
>-       kTemporalUpdateNoneNoRefGolden,
>-       kTemporalUpdateGoldenWithoutDependencyRefAltRef,
>-       kTemporalUpdateNone,
>-@@ -195,21 +249,161 @@ TEST(TemporalLayersTest, KeyFrame) {
>-       kTemporalUpdateGoldenRefAltRef,
>-       kTemporalUpdateNone,
>-   };
>--  int expected_temporal_idx[8] = {0, 0, 0, 0, 0, 0, 0, 2};
>-+  int expected_temporal_idx[8] = {0, 2, 1, 2, 0, 2, 1, 2};
>-+  bool expected_layer_sync[8] = {false, true,  true,  false,
>-+                                 false, false, false, false};
>- 
>-   uint32_t timestamp = 0;
>-   for (int i = 0; i < 7; ++i) {
>-     TemporalLayers::FrameConfig tl_config = tl.UpdateLayerConfig(timestamp);
>--    EXPECT_EQ(expected_flags[i], VP8EncoderImpl::EncodeFlags(tl_config));
>-+    EXPECT_EQ(expected_flags[i], VP8EncoderImpl::EncodeFlags(tl_config)) << i;
>-     tl.PopulateCodecSpecific(true, tl_config, &vp8_info, 0);
>--    EXPECT_EQ(expected_temporal_idx[i], vp8_info.temporalIdx);
>--    EXPECT_EQ(true, vp8_info.layerSync);
>-+    EXPECT_TRUE(checker.CheckTemporalConfig(true, tl_config));
>-+    EXPECT_EQ(expected_temporal_idx[i], tl_config.packetizer_temporal_idx);
>-+    EXPECT_EQ(expected_temporal_idx[i], tl_config.encoder_layer_id);
>-+    EXPECT_EQ(0, vp8_info.temporalIdx)
>-+        << "Key frame should always be packetized as layer 0";
>-+    EXPECT_EQ(expected_layer_sync[i], tl_config.layer_sync);
>-+    EXPECT_TRUE(vp8_info.layerSync) << "Key frame should be marked layer sync.";
>-     timestamp += 3000;
>-   }
>-   TemporalLayers::FrameConfig tl_config = tl.UpdateLayerConfig(timestamp);
>-   EXPECT_EQ(expected_flags[7], VP8EncoderImpl::EncodeFlags(tl_config));
>-   tl.PopulateCodecSpecific(false, tl_config, &vp8_info, 0);
>--  EXPECT_EQ(expected_temporal_idx[7], vp8_info.temporalIdx);
>--  EXPECT_EQ(true, vp8_info.layerSync);
>-+  EXPECT_TRUE(checker.CheckTemporalConfig(false, tl_config));
>-+  EXPECT_NE(0, vp8_info.temporalIdx)
>-+      << "To test something useful, this frame should not use layer 0.";
>-+  EXPECT_EQ(expected_temporal_idx[7], vp8_info.temporalIdx)
>-+      << "Non-keyframe, should use frame temporal index.";
>-+  EXPECT_EQ(expected_temporal_idx[7], tl_config.packetizer_temporal_idx);
>-+  EXPECT_EQ(expected_temporal_idx[7], tl_config.encoder_layer_id);
>-+  EXPECT_FALSE(tl_config.layer_sync);
>-+  EXPECT_TRUE(vp8_info.layerSync) << "Frame after keyframe should always be "
>-+                                     "marked layer sync since it only depends "
>-+                                     "on the base layer.";
>-+}
>-+
>-+class TemporalLayersReferenceTest : public ::testing::TestWithParam<int> {
>-+ public:
>-+  TemporalLayersReferenceTest()
>-+      : timestamp_(1),
>-+        last_sync_timestamp_(timestamp_),
>-+        tl0_reference_(nullptr) {}
>-+  virtual ~TemporalLayersReferenceTest() {}
>-+
>-+ protected:
>-+  static const int kMaxPatternLength = 32;
>-+
>-+  struct BufferState {
>-+    BufferState() : BufferState(-1, 0, false) {}
>-+    BufferState(int temporal_idx, uint32_t timestamp, bool sync)
>-+        : temporal_idx(temporal_idx), timestamp(timestamp), sync(sync) {}
>-+    int temporal_idx;
>-+    uint32_t timestamp;
>-+    bool sync;
>-+  };
>-+
>-+  bool UpdateSyncRefState(const TemporalLayers::BufferFlags& flags,
>-+                          BufferState* buffer_state) {
>-+    if (flags & TemporalLayers::kReference) {
>-+      if (buffer_state->temporal_idx == -1)
>-+        return true;  // References key-frame.
>-+      if (buffer_state->temporal_idx == 0) {
>-+        // No more than one reference to TL0 frame.
>-+        EXPECT_EQ(nullptr, tl0_reference_);
>-+        tl0_reference_ = buffer_state;
>-+        return true;
>-+      }
>-+      return false;  // References higher layer.
>-+    }
>-+    return true;  // No reference, does not affect sync frame status.
>-+  }
>-+
>-+  void ValidateReference(const TemporalLayers::BufferFlags& flags,
>-+                         const BufferState& buffer_state,
>-+                         int temporal_layer) {
>-+    if (flags & TemporalLayers::kReference) {
>-+      if (temporal_layer > 0 && buffer_state.timestamp > 0) {
>-+        // Check that high layer reference does not go past last sync frame.
>-+        EXPECT_GE(buffer_state.timestamp, last_sync_timestamp_);
>-+      }
>-+      // No reference to buffer in higher layer.
>-+      EXPECT_LE(buffer_state.temporal_idx, temporal_layer);
>-+    }
>-+  }
>-+
>-+  uint32_t timestamp_ = 1;
>-+  uint32_t last_sync_timestamp_ = timestamp_;
>-+  BufferState* tl0_reference_;
>-+
>-+  BufferState last_state;
>-+  BufferState golden_state;
>-+  BufferState altref_state;
>-+};
>-+
>-+INSTANTIATE_TEST_CASE_P(DefaultTemporalLayersTest,
>-+                        TemporalLayersReferenceTest,
>-+                        ::testing::Range(1, kMaxTemporalStreams + 1));
>-+
>-+TEST_P(TemporalLayersReferenceTest, ValidFrameConfigs) {
>-+  const int num_layers = GetParam();
>-+  DefaultTemporalLayers tl(num_layers, 0);
>-+  vpx_codec_enc_cfg_t cfg;
>-+  tl.OnRatesUpdated(500, 500, 30);
>-+  tl.UpdateConfiguration(&cfg);
>-+
>-+  // Run through the pattern and store the frame dependencies, plus keep track
>-+  // of the buffer state; which buffers references which temporal layers (if
>-+  // (any). If a given buffer is never updated, it is legal to reference it
>-+  // even for sync frames. In order to be general, don't assume TL0 always
>-+  // updates |last|.
>-+  std::vector<TemporalLayers::FrameConfig> tl_configs(kMaxPatternLength);
>-+  for (int i = 0; i < kMaxPatternLength; ++i) {
>-+    TemporalLayers::FrameConfig tl_config = tl.UpdateLayerConfig(timestamp_++);
>-+    EXPECT_FALSE(tl_config.drop_frame);
>-+    tl_configs.push_back(tl_config);
>-+    int temporal_idx = tl_config.encoder_layer_id;
>-+    // For the default layers, always keep encoder and rtp layers in sync.
>-+    EXPECT_EQ(tl_config.packetizer_temporal_idx, temporal_idx);
>-+
>-+    // Determine if this frame is in a higher layer but references only TL0
>-+    // or untouched buffers, if so verify it is marked as a layer sync.
>-+    bool is_sync_frame = true;
>-+    tl0_reference_ = nullptr;
>-+    if (temporal_idx <= 0) {
>-+      is_sync_frame = false;  // TL0 by definition not a sync frame.
>-+    } else if (!UpdateSyncRefState(tl_config.last_buffer_flags, &last_state)) {
>-+      is_sync_frame = false;
>-+    } else if (!UpdateSyncRefState(tl_config.golden_buffer_flags,
>-+                                   &golden_state)) {
>-+      is_sync_frame = false;
>-+    } else if (!UpdateSyncRefState(tl_config.arf_buffer_flags, &altref_state)) {
>-+      is_sync_frame = false;
>-+    }
>-+    if (is_sync_frame) {
>-+      // Cache timestamp for last found sync frame, so that we can verify no
>-+      // references back past this frame.
>-+      ASSERT_TRUE(tl0_reference_);
>-+      last_sync_timestamp_ = tl0_reference_->timestamp;
>-+    }
>-+    EXPECT_EQ(tl_config.layer_sync, is_sync_frame);
>-+
>-+    // Validate no reference from lower to high temporal layer, or backwards
>-+    // past last reference frame.
>-+    ValidateReference(tl_config.last_buffer_flags, last_state, temporal_idx);
>-+    ValidateReference(tl_config.golden_buffer_flags, golden_state,
>-+                      temporal_idx);
>-+    ValidateReference(tl_config.arf_buffer_flags, altref_state, temporal_idx);
>-+
>-+    // Update the current layer state.
>-+    BufferState state = {temporal_idx, timestamp_, is_sync_frame};
>-+    if (tl_config.last_buffer_flags & TemporalLayers::kUpdate)
>-+      last_state = state;
>-+    if (tl_config.golden_buffer_flags & TemporalLayers::kUpdate)
>-+      golden_state = state;
>-+    if (tl_config.arf_buffer_flags & TemporalLayers::kUpdate)
>-+      altref_state = state;
>-+  }
>- }
>-+}  // namespace test
>- }  // namespace webrtc
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/include/vp8.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/include/vp8.h
>-index c9e45ee5b65..06003613adf 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/include/vp8.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/include/vp8.h
>-@@ -10,28 +10,30 @@
>-  *  WEBRTC VP8 wrapper interface
>-  */
>- 
>--#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_H_
>--#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_H_
>-+#ifndef MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_H_
>-+#define MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_H_
>- 
>--#include "webrtc/modules/video_coding/include/video_codec_interface.h"
>-+#include <memory>
>-+
>-+#include "modules/video_coding/include/video_codec_interface.h"
>- 
>- namespace webrtc {
>- 
>- class VP8Encoder : public VideoEncoder {
>-  public:
>-   static bool IsSupported();
>--  static VP8Encoder* Create();
>-+  static std::unique_ptr<VP8Encoder> Create();
>- 
>-   virtual ~VP8Encoder() {}
>- };  // end of VP8Encoder class
>- 
>- class VP8Decoder : public VideoDecoder {
>-  public:
>--  static bool IsSupported();
>--  static VP8Decoder* Create();
>-+    static bool IsSupported();
>-+    static std::unique_ptr<VP8Decoder> Create();
>- 
>-   virtual ~VP8Decoder() {}
>- };  // end of VP8Decoder class
>- }  // namespace webrtc
>- 
>--#endif  // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_H_
>-+#endif  // MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_H_
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/include/vp8_common_types.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/include/vp8_common_types.h
>-index 7a27e4429a9..dff70ac3328 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/include/vp8_common_types.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/include/vp8_common_types.h
>-@@ -8,17 +8,17 @@
>-  *  be found in the AUTHORS file in the root of the source tree.
>-  */
>- 
>--#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_COMMON_TYPES_H_
>--#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_COMMON_TYPES_H_
>-+#ifndef MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_COMMON_TYPES_H_
>-+#define MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_COMMON_TYPES_H_
>- 
>--#include "webrtc/common_types.h"
>-+#include "common_types.h"  // NOLINT(build/include)
>- 
>- namespace webrtc {
>- 
>- // Ratio allocation between temporal streams:
>- // Values as required for the VP8 codec (accumulating).
>- static const float
>--    kVp8LayerRateAlloction[kMaxTemporalStreams][kMaxTemporalStreams] = {
>-+    kVp8LayerRateAlloction[kMaxSimulcastStreams][kMaxTemporalStreams] = {
>-         {1.0f, 1.0f, 1.0f, 1.0f},  // 1 layer
>-         {0.6f, 1.0f, 1.0f, 1.0f},  // 2 layers {60%, 40%}
>-         {0.4f, 0.6f, 1.0f, 1.0f},  // 3 layers {40%, 20%, 40%}
>-@@ -26,4 +26,4 @@ static const float
>- };
>- 
>- }  // namespace webrtc
>--#endif  // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_COMMON_TYPES_H_
>-+#endif  // MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_COMMON_TYPES_H_
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/include/vp8_globals.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/include/vp8_globals.h
>-index 938e1991269..1fab5f45a6f 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/include/vp8_globals.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/include/vp8_globals.h
>-@@ -11,10 +11,10 @@
>- // This file contains codec dependent definitions that are needed in
>- // order to compile the WebRTC codebase, even if this codec is not used.
>- 
>--#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_GLOBALS_H_
>--#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_GLOBALS_H_
>-+#ifndef MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_GLOBALS_H_
>-+#define MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_GLOBALS_H_
>- 
>--#include "webrtc/modules/video_coding/codecs/interface/common_constants.h"
>-+#include "modules/video_coding/codecs/interface/common_constants.h"
>- 
>- namespace webrtc {
>- 
>-@@ -46,4 +46,4 @@ struct RTPVideoHeaderVP8 {
>- 
>- }  // namespace webrtc
>- 
>--#endif  // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_GLOBALS_H_
>-+#endif  // MODULES_VIDEO_CODING_CODECS_VP8_INCLUDE_VP8_GLOBALS_H_
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/screenshare_layers.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/screenshare_layers.cc
>-index a30679a36cf..782dc772fca 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/screenshare_layers.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/screenshare_layers.cc
>-@@ -1,30 +1,32 @@
>- /* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
>--*
>--*  Use of this source code is governed by a BSD-style license
>--*  that can be found in the LICENSE file in the root of the source
>--*  tree. An additional intellectual property rights grant can be found
>--*  in the file PATENTS.  All contributing project authors may
>--*  be found in the AUTHORS file in the root of the source tree.
>--*/
>-+ *
>-+ *  Use of this source code is governed by a BSD-style license
>-+ *  that can be found in the LICENSE file in the root of the source
>-+ *  tree. An additional intellectual property rights grant can be found
>-+ *  in the file PATENTS.  All contributing project authors may
>-+ *  be found in the AUTHORS file in the root of the source tree.
>-+ */
>- 
>--#include "webrtc/modules/video_coding/codecs/vp8/screenshare_layers.h"
>-+#include "modules/video_coding/codecs/vp8/screenshare_layers.h"
>- 
>- #include <stdlib.h>
>- 
>- #include <algorithm>
>-+#include <memory>
>- 
>--#include "webrtc/base/checks.h"
>--#include "vpx/vpx_encoder.h"
>-+#include "modules/video_coding/include/video_codec_interface.h"
>-+#include "rtc_base/checks.h"
>-+#include "rtc_base/logging.h"
>-+#include "system_wrappers/include/clock.h"
>-+#include "system_wrappers/include/metrics.h"
>- #include "vpx/vp8cx.h"
>--#include "webrtc/modules/video_coding/include/video_codec_interface.h"
>--#include "webrtc/system_wrappers/include/clock.h"
>--#include "webrtc/system_wrappers/include/metrics.h"
>-+#include "vpx/vpx_encoder.h"
>- 
>- namespace webrtc {
>- 
>- static const int kOneSecond90Khz = 90000;
>--static const int kMinTimeBetweenSyncs = kOneSecond90Khz * 5;
>--static const int kMaxTimeBetweenSyncs = kOneSecond90Khz * 10;
>-+static const int kMinTimeBetweenSyncs = kOneSecond90Khz * 2;
>-+static const int kMaxTimeBetweenSyncs = kOneSecond90Khz * 4;
>- static const int kQpDeltaThresholdForSync = 8;
>- static const int kMinBitrateKbpsForQpBoost = 500;
>- 
>-@@ -54,6 +56,23 @@ webrtc::TemporalLayers* ScreenshareTemporalLayersFactory::Create(
>-   return tl;
>- }
>- 
>-+std::unique_ptr<webrtc::TemporalLayersChecker>
>-+ScreenshareTemporalLayersFactory::CreateChecker(
>-+    int simulcast_id,
>-+    int temporal_layers,
>-+    uint8_t initial_tl0_pic_idx) const {
>-+  webrtc::TemporalLayersChecker* tlc;
>-+  if (simulcast_id == 0) {
>-+    tlc =
>-+        new webrtc::TemporalLayersChecker(temporal_layers, initial_tl0_pic_idx);
>-+  } else {
>-+    TemporalLayersFactory rt_tl_factory;
>-+    return rt_tl_factory.CreateChecker(simulcast_id, temporal_layers,
>-+                                       initial_tl0_pic_idx);
>-+  }
>-+  return std::unique_ptr<webrtc::TemporalLayersChecker>(tlc);
>-+}
>-+
>- ScreenshareLayers::ScreenshareLayers(int num_temporal_layers,
>-                                      uint8_t initial_tl0_pic_idx,
>-                                      Clock* clock)
>-@@ -79,12 +98,6 @@ ScreenshareLayers::~ScreenshareLayers() {
>-   UpdateHistograms();
>- }
>- 
>--int ScreenshareLayers::GetTemporalLayerId(
>--    const TemporalLayers::FrameConfig& tl_config) const {
>--  // Codec does not use temporal layers for screenshare.
>--  return 0;
>--}
>--
>- uint8_t ScreenshareLayers::Tl0PicIdx() const {
>-   return tl0_pic_idx_;
>- }
>-@@ -96,7 +109,6 @@ TemporalLayers::FrameConfig ScreenshareLayers::UpdateLayerConfig(
>-     // TODO(pbos): Consider updating only last, and not all buffers.
>-     TemporalLayers::FrameConfig tl_config(
>-         kReferenceAndUpdate, kReferenceAndUpdate, kReferenceAndUpdate);
>--    tl_config.pattern_idx = static_cast<int>(TemporalLayerState::kTl1);
>-     return tl_config;
>-   }
>- 
>-@@ -152,11 +164,17 @@ TemporalLayers::FrameConfig ScreenshareLayers::UpdateLayerConfig(
>-       last_emitted_tl0_timestamp_ = unwrapped_timestamp;
>-       break;
>-     case 1:
>--      if (TimeToSync(unwrapped_timestamp)) {
>--        last_sync_timestamp_ = unwrapped_timestamp;
>--        layer_state = TemporalLayerState::kTl1Sync;
>-+      if (layers_[1].state != TemporalLayer::State::kDropped) {
>-+        if (TimeToSync(unwrapped_timestamp)) {
>-+          last_sync_timestamp_ = unwrapped_timestamp;
>-+          layer_state = TemporalLayerState::kTl1Sync;
>-+        } else {
>-+          layer_state = TemporalLayerState::kTl1;
>-+        }
>-       } else {
>--        layer_state = TemporalLayerState::kTl1;
>-+        layer_state = last_sync_timestamp_ == unwrapped_timestamp
>-+                          ? TemporalLayerState::kTl1Sync
>-+                          : TemporalLayerState::kTl1;
>-       }
>-       break;
>-     case -1:
>-@@ -178,21 +196,24 @@ TemporalLayers::FrameConfig ScreenshareLayers::UpdateLayerConfig(
>-       // TL0 only references and updates 'last'.
>-       tl_config =
>-           TemporalLayers::FrameConfig(kReferenceAndUpdate, kNone, kNone);
>-+      tl_config.packetizer_temporal_idx = 0;
>-       break;
>-     case TemporalLayerState::kTl1:
>-       // TL1 references both 'last' and 'golden' but only updates 'golden'.
>-       tl_config =
>-           TemporalLayers::FrameConfig(kReference, kReferenceAndUpdate, kNone);
>-+      tl_config.packetizer_temporal_idx = 1;
>-       break;
>-     case TemporalLayerState::kTl1Sync:
>-       // Predict from only TL0 to allow participants to switch to the high
>-       // bitrate stream. Updates 'golden' so that TL1 can continue to refer to
>-       // and update 'golden' from this point on.
>-       tl_config = TemporalLayers::FrameConfig(kReference, kUpdate, kNone);
>-+      tl_config.packetizer_temporal_idx = 1;
>-       break;
>-   }
>- 
>--  tl_config.pattern_idx = static_cast<int>(layer_state);
>-+  tl_config.layer_sync = layer_state == TemporalLayerState::kTl1Sync;
>-   return tl_config;
>- }
>- 
>-@@ -269,36 +290,24 @@ void ScreenshareLayers::PopulateCodecSpecific(
>-     const TemporalLayers::FrameConfig& tl_config,
>-     CodecSpecificInfoVP8* vp8_info,
>-     uint32_t timestamp) {
>--  int64_t unwrapped_timestamp = time_wrap_handler_.Unwrap(timestamp);
>-   if (number_of_temporal_layers_ == 1) {
>-     vp8_info->temporalIdx = kNoTemporalIdx;
>-     vp8_info->layerSync = false;
>-     vp8_info->tl0PicIdx = kNoTl0PicIdx;
>-   } else {
>--    TemporalLayerState layer_state =
>--        static_cast<TemporalLayerState>(tl_config.pattern_idx);
>--    switch (layer_state) {
>--      case TemporalLayerState::kDrop:
>--        RTC_NOTREACHED();
>--        break;
>--      case TemporalLayerState::kTl0:
>--        vp8_info->temporalIdx = 0;
>--        break;
>--      case TemporalLayerState::kTl1:
>--      case TemporalLayerState::kTl1Sync:
>--        vp8_info->temporalIdx = 1;
>--        break;
>--    }
>-+    int64_t unwrapped_timestamp = time_wrap_handler_.Unwrap(timestamp);
>-+    vp8_info->temporalIdx = tl_config.packetizer_temporal_idx;
>-+    vp8_info->layerSync = tl_config.layer_sync;
>-     if (frame_is_keyframe) {
>-       vp8_info->temporalIdx = 0;
>-       last_sync_timestamp_ = unwrapped_timestamp;
>-+      vp8_info->layerSync = true;
>-     } else if (last_base_layer_sync_ && vp8_info->temporalIdx != 0) {
>-       // Regardless of pattern the frame after a base layer sync will always
>-       // be a layer sync.
>-       last_sync_timestamp_ = unwrapped_timestamp;
>-+      vp8_info->layerSync = true;
>-     }
>--    vp8_info->layerSync = last_sync_timestamp_ != -1 &&
>--                          last_sync_timestamp_ == unwrapped_timestamp;
>-     if (vp8_info->temporalIdx == 0) {
>-       tl0_pic_idx_++;
>-     }
>-@@ -443,8 +452,9 @@ void ScreenshareLayers::UpdateHistograms() {
>-     int total_frames = stats_.num_tl0_frames_ + stats_.num_tl1_frames_;
>-     RTC_HISTOGRAM_COUNTS_10000(
>-         "WebRTC.Video.Screenshare.FramesPerDrop",
>--        (stats_.num_dropped_frames_ == 0 ? 0 : total_frames /
>--                                                   stats_.num_dropped_frames_));
>-+        (stats_.num_dropped_frames_ == 0
>-+             ? 0
>-+             : total_frames / stats_.num_dropped_frames_));
>-     RTC_HISTOGRAM_COUNTS_10000(
>-         "WebRTC.Video.Screenshare.FramesPerOvershoot",
>-         (stats_.num_overshoots_ == 0 ? 0
>-@@ -465,5 +475,4 @@ void ScreenshareLayers::UpdateHistograms() {
>-     }
>-   }
>- }
>--
>- }  // namespace webrtc
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/screenshare_layers.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/screenshare_layers.h
>-index 873846e8b22..81db90abe6b 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/screenshare_layers.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/screenshare_layers.h
>-@@ -1,21 +1,21 @@
>- /* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
>--*
>--*  Use of this source code is governed by a BSD-style license
>--*  that can be found in the LICENSE file in the root of the source
>--*  tree. An additional intellectual property rights grant can be found
>--*  in the file PATENTS.  All contributing project authors may
>--*  be found in the AUTHORS file in the root of the source tree.
>--*/
>--#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_SCREENSHARE_LAYERS_H_
>--#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_SCREENSHARE_LAYERS_H_
>-+ *
>-+ *  Use of this source code is governed by a BSD-style license
>-+ *  that can be found in the LICENSE file in the root of the source
>-+ *  tree. An additional intellectual property rights grant can be found
>-+ *  in the file PATENTS.  All contributing project authors may
>-+ *  be found in the AUTHORS file in the root of the source tree.
>-+ */
>-+#ifndef MODULES_VIDEO_CODING_CODECS_VP8_SCREENSHARE_LAYERS_H_
>-+#define MODULES_VIDEO_CODING_CODECS_VP8_SCREENSHARE_LAYERS_H_
>- 
>- #include <vector>
>- 
>--#include "webrtc/base/rate_statistics.h"
>--#include "webrtc/base/timeutils.h"
>--#include "webrtc/modules/video_coding/codecs/vp8/temporal_layers.h"
>--#include "webrtc/modules/video_coding/utility/frame_dropper.h"
>--#include "webrtc/typedefs.h"
>-+#include "modules/video_coding/codecs/vp8/temporal_layers.h"
>-+#include "modules/video_coding/utility/frame_dropper.h"
>-+#include "rtc_base/rate_statistics.h"
>-+#include "rtc_base/timeutils.h"
>-+#include "typedefs.h"  // NOLINT(build/include)
>- 
>- namespace webrtc {
>- 
>-@@ -54,9 +54,6 @@ class ScreenshareLayers : public TemporalLayers {
>- 
>-   void FrameEncoded(unsigned int size, int qp) override;
>- 
>--  int GetTemporalLayerId(
>--      const TemporalLayers::FrameConfig& tl_config) const override;
>--
>-   uint8_t Tl0PicIdx() const override;
>- 
>-  private:
>-@@ -127,4 +124,4 @@ class ScreenshareLayers : public TemporalLayers {
>- };
>- }  // namespace webrtc
>- 
>--#endif  // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_SCREENSHARE_LAYERS_H_
>-+#endif  // MODULES_VIDEO_CODING_CODECS_VP8_SCREENSHARE_LAYERS_H_
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/screenshare_layers_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/screenshare_layers_unittest.cc
>-index 9cac0b0a420..754235ab0b3 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/screenshare_layers_unittest.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/screenshare_layers_unittest.cc
>-@@ -13,14 +13,14 @@
>- 
>- #include "vpx/vp8cx.h"
>- #include "vpx/vpx_encoder.h"
>--#include "webrtc/modules/video_coding/codecs/vp8/screenshare_layers.h"
>--#include "webrtc/modules/video_coding/codecs/vp8/vp8_impl.h"
>--#include "webrtc/modules/video_coding/include/video_codec_interface.h"
>--#include "webrtc/modules/video_coding/utility/mock/mock_frame_dropper.h"
>--#include "webrtc/system_wrappers/include/clock.h"
>--#include "webrtc/system_wrappers/include/metrics.h"
>--#include "webrtc/system_wrappers/include/metrics_default.h"
>--#include "webrtc/test/gtest.h"
>-+#include "modules/video_coding/codecs/vp8/screenshare_layers.h"
>-+#include "modules/video_coding/codecs/vp8/vp8_impl.h"
>-+#include "modules/video_coding/include/video_codec_interface.h"
>-+#include "modules/video_coding/utility/mock/mock_frame_dropper.h"
>-+#include "system_wrappers/include/clock.h"
>-+#include "system_wrappers/include/metrics.h"
>-+#include "system_wrappers/include/metrics_default.h"
>-+#include "test/gtest.h"
>- 
>- using ::testing::_;
>- using ::testing::ElementsAre;
>-@@ -35,8 +35,8 @@ const int kDefaultQp = 54;
>- const int kDefaultTl0BitrateKbps = 200;
>- const int kDefaultTl1BitrateKbps = 2000;
>- const int kFrameRate = 5;
>--const int kSyncPeriodSeconds = 5;
>--const int kMaxSyncPeriodSeconds = 10;
>-+const int kSyncPeriodSeconds = 2;
>-+const int kMaxSyncPeriodSeconds = 4;
>- 
>- // Expected flags for corresponding temporal layers.
>- const int kTl0Flags = VP8_EFLAG_NO_UPD_GF | VP8_EFLAG_NO_UPD_ARF |
>-@@ -71,6 +71,10 @@ class ScreenshareLayerTest : public ::testing::Test {
>- 
>-   int ConfigureFrame(bool key_frame) {
>-     tl_config_ = layers_->UpdateLayerConfig(timestamp_);
>-+    EXPECT_EQ(0, tl_config_.encoder_layer_id)
>-+        << "ScreenshareLayers always encodes using the bitrate allocator for "
>-+           "layer 0, but may reference different buffers and packetize "
>-+           "differently.";
>-     if (tl_config_.drop_frame) {
>-       return -1;
>-     }
>-@@ -126,18 +130,29 @@ class ScreenshareLayerTest : public ::testing::Test {
>- 
>-   // Adds frames until we get one in the specified temporal layer. The last
>-   // FrameEncoded() call will be omitted and needs to be done by the caller.
>--  void SkipUntilTl(int layer) {
>--    for (int i = 0; i < 5; ++i) {
>--      ConfigureFrame(false);
>-+  // Returns the flags for the last frame.
>-+  int SkipUntilTl(int layer) {
>-+    return SkipUntilTlAndSync(layer, rtc::nullopt);
>-+  }
>-+
>-+  // Same as SkipUntilTl, but also waits until the sync bit condition is met.
>-+  int SkipUntilTlAndSync(int layer, rtc::Optional<bool> sync) {
>-+    int flags = 0;
>-+    const int kMaxFramesToSkip =
>-+        1 + (sync.value_or(false) ? kMaxSyncPeriodSeconds : 1) * kFrameRate;
>-+    for (int i = 0; i < kMaxFramesToSkip; ++i) {
>-+      flags = ConfigureFrame(false);
>-       timestamp_ += kTimestampDelta5Fps;
>--      if (vp8_info_.temporalIdx != layer) {
>-+      if (vp8_info_.temporalIdx != layer ||
>-+          (sync && *sync != vp8_info_.layerSync)) {
>-         layers_->FrameEncoded(frame_size_, kDefaultQp);
>-       } else {
>--        // Found frame form sought layer.
>--        return;
>-+        // Found frame from sought after layer.
>-+        return flags;
>-       }
>-     }
>-     ADD_FAILURE() << "Did not get a frame of TL" << layer << " in time.";
>-+    return -1;
>-   }
>- 
>-   int min_qp_;
>-@@ -562,4 +577,31 @@ TEST_F(ScreenshareLayerTest, RespectsConfiguredFramerate) {
>-   // Allow for some rounding errors in the measurements.
>-   EXPECT_NEAR(num_discarded_frames, num_input_frames / 2, 2);
>- }
>-+
>-+TEST_F(ScreenshareLayerTest, 2LayersSyncAtOvershootDrop) {
>-+  // Run grace period so we have existing frames in both TL0 and Tl1.
>-+  EXPECT_TRUE(RunGracePeriod());
>-+
>-+  // Move ahead until we have a sync frame in TL1.
>-+  EXPECT_EQ(kTl1SyncFlags, SkipUntilTlAndSync(1, true));
>-+  ASSERT_TRUE(vp8_info_.layerSync);
>-+
>-+  // Simulate overshoot of this frame.
>-+  layers_->FrameEncoded(0, -1);
>-+
>-+  // Reencode, frame config, flags and codec specific info should remain the
>-+  // same as for the dropped frame.
>-+  timestamp_ -= kTimestampDelta5Fps;  // Undo last timestamp increment.
>-+  TemporalLayers::FrameConfig new_tl_config =
>-+      layers_->UpdateLayerConfig(timestamp_);
>-+  EXPECT_EQ(tl_config_, new_tl_config);
>-+
>-+  config_updated_ = layers_->UpdateConfiguration(&cfg_);
>-+  EXPECT_EQ(kTl1SyncFlags, VP8EncoderImpl::EncodeFlags(tl_config_));
>-+
>-+  CodecSpecificInfoVP8 new_vp8_info;
>-+  layers_->PopulateCodecSpecific(false, tl_config_, &new_vp8_info, timestamp_);
>-+  EXPECT_TRUE(new_vp8_info.layerSync);
>-+}
>-+
>- }  // namespace webrtc
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/simulcast_rate_allocator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/simulcast_rate_allocator.cc
>-index 39c43c9b6d0..b2b3334815f 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/simulcast_rate_allocator.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/simulcast_rate_allocator.cc
>-@@ -8,14 +8,14 @@
>-  *  be found in the AUTHORS file in the root of the source tree.
>-  */
>- 
>--#include "webrtc/modules/video_coding/codecs/vp8/simulcast_rate_allocator.h"
>-+#include "modules/video_coding/codecs/vp8/simulcast_rate_allocator.h"
>- 
>- #include <algorithm>
>- #include <memory>
>- #include <vector>
>- #include <utility>
>- 
>--#include "webrtc/base/checks.h"
>-+#include "rtc_base/checks.h"
>- 
>- namespace webrtc {
>- 
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/simulcast_rate_allocator.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/simulcast_rate_allocator.h
>-index e558290e6b4..929abba4132 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/simulcast_rate_allocator.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/simulcast_rate_allocator.h
>-@@ -8,18 +8,18 @@
>-  *  be found in the AUTHORS file in the root of the source tree.
>-  */
>- 
>--#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_SIMULCAST_RATE_ALLOCATOR_H_
>--#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_SIMULCAST_RATE_ALLOCATOR_H_
>-+#ifndef MODULES_VIDEO_CODING_CODECS_VP8_SIMULCAST_RATE_ALLOCATOR_H_
>-+#define MODULES_VIDEO_CODING_CODECS_VP8_SIMULCAST_RATE_ALLOCATOR_H_
>- 
>- #include <stdint.h>
>- 
>- #include <map>
>- #include <memory>
>- 
>--#include "webrtc/api/video_codecs/video_encoder.h"
>--#include "webrtc/base/constructormagic.h"
>--#include "webrtc/common_video/include/video_bitrate_allocator.h"
>--#include "webrtc/modules/video_coding/codecs/vp8/temporal_layers.h"
>-+#include "api/video_codecs/video_encoder.h"
>-+#include "common_video/include/video_bitrate_allocator.h"
>-+#include "modules/video_coding/codecs/vp8/temporal_layers.h"
>-+#include "rtc_base/constructormagic.h"
>- 
>- namespace webrtc {
>- 
>-@@ -48,4 +48,4 @@ class SimulcastRateAllocator : public VideoBitrateAllocator,
>- 
>- }  // namespace webrtc
>- 
>--#endif  // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_SIMULCAST_RATE_ALLOCATOR_H_
>-+#endif  // MODULES_VIDEO_CODING_CODECS_VP8_SIMULCAST_RATE_ALLOCATOR_H_
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/simulcast_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/simulcast_unittest.cc
>-index 9d919cdd5bd..b1dd794f291 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/simulcast_unittest.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/simulcast_unittest.cc
>-@@ -8,19 +8,19 @@
>-  *  be found in the AUTHORS file in the root of the source tree.
>-  */
>- 
>--#include "webrtc/modules/video_coding/codecs/vp8/simulcast_unittest.h"
>-+#include "modules/video_coding/codecs/vp8/simulcast_test_utility.h"
>- 
>- namespace webrtc {
>- namespace testing {
>- 
>- class TestVp8Impl : public TestVp8Simulcast {
>-- public:
>--  TestVp8Impl()
>--      : TestVp8Simulcast(VP8Encoder::Create(), VP8Decoder::Create()) {}
>--
>-  protected:
>--  virtual void SetUp() { TestVp8Simulcast::SetUp(); }
>--  virtual void TearDown() { TestVp8Simulcast::TearDown(); }
>-+  std::unique_ptr<VP8Encoder> CreateEncoder() override {
>-+    return VP8Encoder::Create();
>-+  }
>-+  std::unique_ptr<VP8Decoder> CreateDecoder() override {
>-+    return VP8Decoder::Create();
>-+  }
>- };
>- 
>- TEST_F(TestVp8Impl, TestKeyFrameRequestsOnAllStreams) {
>-@@ -71,10 +71,6 @@ TEST_F(TestVp8Impl, TestSaptioTemporalLayers333PatternEncoder) {
>-   TestVp8Simulcast::TestSaptioTemporalLayers333PatternEncoder();
>- }
>- 
>--TEST_F(TestVp8Impl, TestSpatioTemporalLayers321PatternEncoder) {
>--  TestVp8Simulcast::TestSpatioTemporalLayers321PatternEncoder();
>--}
>--
>- TEST_F(TestVp8Impl, TestStrideEncodeDecode) {
>-   TestVp8Simulcast::TestStrideEncodeDecode();
>- }
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/temporal_layers.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/temporal_layers.h
>-index f0561076a0e..dad718cd0f3 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/temporal_layers.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/temporal_layers.h
>-@@ -1,21 +1,21 @@
>- /* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
>--*
>--*  Use of this source code is governed by a BSD-style license
>--*  that can be found in the LICENSE file in the root of the source
>--*  tree. An additional intellectual property rights grant can be found
>--*  in the file PATENTS.  All contributing project authors may
>--*  be found in the AUTHORS file in the root of the source tree.
>--*/
>-+ *
>-+ *  Use of this source code is governed by a BSD-style license
>-+ *  that can be found in the LICENSE file in the root of the source
>-+ *  tree. An additional intellectual property rights grant can be found
>-+ *  in the file PATENTS.  All contributing project authors may
>-+ *  be found in the AUTHORS file in the root of the source tree.
>-+ */
>- /*
>--* This file defines the interface for doing temporal layers with VP8.
>--*/
>--#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_TEMPORAL_LAYERS_H_
>--#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_TEMPORAL_LAYERS_H_
>-+ * This file defines the interface for doing temporal layers with VP8.
>-+ */
>-+#ifndef MODULES_VIDEO_CODING_CODECS_VP8_TEMPORAL_LAYERS_H_
>-+#define MODULES_VIDEO_CODING_CODECS_VP8_TEMPORAL_LAYERS_H_
>- 
>- #include <vector>
>-+#include <memory>
>- 
>--#include "webrtc/typedefs.h"
>--
>-+#include "typedefs.h"  // NOLINT(build/include)
>- struct vpx_codec_enc_cfg;
>- typedef struct vpx_codec_enc_cfg vpx_codec_enc_cfg_t;
>- 
>-@@ -47,13 +47,34 @@ class TemporalLayers {
>-     BufferFlags golden_buffer_flags;
>-     BufferFlags arf_buffer_flags;
>- 
>--    // TODO(pbos): Consider breaking these out of here and returning only a
>--    // pattern index that needs to be returned to fill CodecSpecificInfoVP8 or
>--    // EncodeFlags.
>-+    // The encoder layer ID is used to utilize the correct bitrate allocator
>-+    // inside the encoder. It does not control references nor determine which
>-+    // "actual" temporal layer this is. The packetizer temporal index determines
>-+    // which layer the encoded frame should be packetized into.
>-+    // Normally these are the same, but current temporal-layer strategies for
>-+    // screenshare use one bitrate allocator for all layers, but attempt to
>-+    // packetize / utilize references to split a stream into multiple layers,
>-+    // with different quantizer settings, to hit target bitrate.
>-+    // TODO(pbos): Screenshare layers are being reconsidered at the time of
>-+    // writing, we might be able to remove this distinction, and have a temporal
>-+    // layer imply both (the normal case).
>-+    int encoder_layer_id;
>-+    int packetizer_temporal_idx;
>-+
>-     bool layer_sync;
>-+
>-     bool freeze_entropy;
>- 
>--    int pattern_idx;
>-+    bool operator==(const FrameConfig& o) const {
>-+      return drop_frame == o.drop_frame &&
>-+             last_buffer_flags == o.last_buffer_flags &&
>-+             golden_buffer_flags == o.golden_buffer_flags &&
>-+             arf_buffer_flags == o.arf_buffer_flags &&
>-+             layer_sync == o.layer_sync && freeze_entropy == o.freeze_entropy &&
>-+             encoder_layer_id == o.encoder_layer_id &&
>-+             packetizer_temporal_idx == o.packetizer_temporal_idx;
>-+    }
>-+    bool operator!=(const FrameConfig& o) const { return !(*this == o); }
>- 
>-    private:
>-     FrameConfig(BufferFlags last,
>-@@ -91,18 +112,27 @@ class TemporalLayers {
>-   // Returns the current tl0_pic_idx, so it can be reused in future
>-   // instantiations.
>-   virtual uint8_t Tl0PicIdx() const = 0;
>--  virtual int GetTemporalLayerId(
>--      const TemporalLayers::FrameConfig& tl_config) const = 0;
>- };
>- 
>- class TemporalLayersListener;
>-+class TemporalLayersChecker;
>-+
>- class TemporalLayersFactory {
>-  public:
>-   TemporalLayersFactory() : listener_(nullptr) {}
>-   virtual ~TemporalLayersFactory() {}
>--  virtual TemporalLayers* Create(int /* simulcast_id */,
>--                                 int /* temporal_layers */,
>--                                 uint8_t /* initial_tl0_pic_idx */) const { return nullptr; }
>-+  virtual TemporalLayers* Create(int simulcast_id,
>-+                                 int temporal_layers,
>-+                                 uint8_t initial_tl0_pic_idx) const;
>-+
>-+  // Creates helper class which performs online checks of a correctness of
>-+  // temporal layers dependencies returned by TemporalLayers class created in
>-+  // the same factory.
>-+  virtual std::unique_ptr<TemporalLayersChecker> CreateChecker(
>-+      int simulcast_id,
>-+      int temporal_layers,
>-+      uint8_t initial_tl0_pic_idx) const;
>-+
>-   void SetListener(TemporalLayersListener* listener);
>- 
>-  protected:
>-@@ -117,6 +147,14 @@ class ScreenshareTemporalLayersFactory : public webrtc::TemporalLayersFactory {
>-   webrtc::TemporalLayers* Create(int simulcast_id,
>-                                  int num_temporal_layers,
>-                                  uint8_t initial_tl0_pic_idx) const override;
>-+
>-+  // Creates helper class which performs online checks of a correctness of
>-+  // temporal layers dependencies returned by TemporalLayers class created in
>-+  // the same factory.
>-+  std::unique_ptr<webrtc::TemporalLayersChecker> CreateChecker(
>-+      int simulcast_id,
>-+      int temporal_layers,
>-+      uint8_t initial_tl0_pic_idx) const override;
>- };
>- 
>- class TemporalLayersListener {
>-@@ -128,5 +166,40 @@ class TemporalLayersListener {
>-                                        TemporalLayers* layers) = 0;
>- };
>- 
>-+// Used only inside RTC_DCHECK(). It checks correctness of temporal layers
>-+// dependencies and sync bits. The only method of this class is called after
>-+// each UpdateLayersConfig() of a corresponding TemporalLayers class.
>-+class TemporalLayersChecker {
>-+ public:
>-+  TemporalLayersChecker(int num_temporal_layers, uint8_t initial_tl0_pic_idx);
>-+  virtual ~TemporalLayersChecker() {}
>-+
>-+  virtual bool CheckTemporalConfig(
>-+      bool frame_is_keyframe,
>-+      const TemporalLayers::FrameConfig& frame_config);
>-+
>-+ private:
>-+  struct BufferState {
>-+    BufferState() : is_keyframe(true), temporal_layer(0), sequence_number(0) {}
>-+    bool is_keyframe;
>-+    uint8_t temporal_layer;
>-+    uint32_t sequence_number;
>-+  };
>-+  bool CheckAndUpdateBufferState(BufferState* state,
>-+                                 bool* need_sync,
>-+                                 bool frame_is_keyframe,
>-+                                 uint8_t temporal_layer,
>-+                                 webrtc::TemporalLayers::BufferFlags flags,
>-+                                 uint32_t sequence_number,
>-+                                 uint32_t* lowest_sequence_referenced);
>-+  BufferState last_;
>-+  BufferState arf_;
>-+  BufferState golden_;
>-+  int num_temporal_layers_;
>-+  uint32_t sequence_number_;
>-+  uint32_t last_sync_sequence_number_;
>-+  uint32_t last_tl0_sequence_number_;
>-+};
>-+
>- }  // namespace webrtc
>--#endif  // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_TEMPORAL_LAYERS_H_
>-+#endif  // MODULES_VIDEO_CODING_CODECS_VP8_TEMPORAL_LAYERS_H_
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc
>-index a3db4fdad69..5d6abec76d8 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/test/vp8_impl_unittest.cc
>-@@ -12,99 +12,88 @@
>- 
>- #include <memory>
>- 
>--#include "webrtc/api/video/i420_buffer.h"
>--#include "webrtc/base/checks.h"
>--#include "webrtc/base/optional.h"
>--#include "webrtc/base/timeutils.h"
>--#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
>--#include "webrtc/modules/video_coding/codecs/test/video_codec_test.h"
>--#include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
>--#include "webrtc/modules/video_coding/codecs/vp8/temporal_layers.h"
>--#include "webrtc/test/frame_utils.h"
>--#include "webrtc/test/gtest.h"
>--#include "webrtc/test/testsupport/fileutils.h"
>-+#include "api/optional.h"
>-+#include "api/video/i420_buffer.h"
>-+#include "common_video/libyuv/include/webrtc_libyuv.h"
>-+#include "modules/video_coding/codecs/test/video_codec_test.h"
>-+#include "modules/video_coding/codecs/vp8/include/vp8.h"
>-+#include "modules/video_coding/codecs/vp8/temporal_layers.h"
>-+#include "rtc_base/checks.h"
>-+#include "rtc_base/timeutils.h"
>-+#include "test/field_trial.h"
>-+#include "test/frame_utils.h"
>-+#include "test/gtest.h"
>-+#include "test/testsupport/fileutils.h"
>-+#include "test/video_codec_settings.h"
>- 
>- namespace webrtc {
>- 
>- namespace {
>-+constexpr uint32_t kInitialTimestampRtp = 123;
>-+constexpr int64_t kTestNtpTimeMs = 456;
>-+constexpr int64_t kInitialTimestampMs = 789;
>-+constexpr uint32_t kTimestampIncrement = 3000;
>-+constexpr int kNumCores = 1;
>-+constexpr size_t kMaxPayloadSize = 1440;
>-+constexpr int kDefaultMinPixelsPerFrame = 320 * 180;
>-+constexpr int kWidth = 172;
>-+constexpr int kHeight = 144;
>- 
>- void Calc16ByteAlignedStride(int width, int* stride_y, int* stride_uv) {
>-   *stride_y = 16 * ((width + 15) / 16);
>-   *stride_uv = 16 * ((width + 31) / 32);
>- }
>--
>--enum { kMaxWaitEncTimeMs = 100 };
>--enum { kMaxWaitDecTimeMs = 25 };
>--
>--constexpr uint32_t kTestTimestamp = 123;
>--constexpr int64_t kTestNtpTimeMs = 456;
>--constexpr uint32_t kTimestampIncrementPerFrame = 3000;
>--
>- }  // namespace
>- 
>--// TODO(mikhal): Replace these with mocks.
>--class Vp8UnitTestEncodeCompleteCallback : public webrtc::EncodedImageCallback {
>-+class EncodedImageCallbackTestImpl : public webrtc::EncodedImageCallback {
>-  public:
>--  Vp8UnitTestEncodeCompleteCallback(EncodedImage* frame,
>--                                    CodecSpecificInfo* codec_specific_info,
>--                                    unsigned int decoderSpecificSize,
>--                                    void* decoderSpecificInfo)
>--      : encoded_frame_(frame),
>--        codec_specific_info_(codec_specific_info),
>--        encode_complete_(false) {}
>--
>--  Result OnEncodedImage(const EncodedImage& encoded_frame_,
>-+  Result OnEncodedImage(const EncodedImage& encoded_frame,
>-                         const CodecSpecificInfo* codec_specific_info,
>--                        const RTPFragmentationHeader* fragmentation) override;
>--  bool EncodeComplete();
>-+                        const RTPFragmentationHeader* fragmentation) override {
>-+    EXPECT_GT(encoded_frame._length, 0u);
>-+    VerifyQpParser(encoded_frame);
>- 
>-- private:
>--  EncodedImage* const encoded_frame_;
>--  CodecSpecificInfo* const codec_specific_info_;
>--  std::unique_ptr<uint8_t[]> frame_buffer_;
>--  bool encode_complete_;
>--};
>-+    if (encoded_frame_._size != encoded_frame._size) {
>-+      delete[] encoded_frame_._buffer;
>-+      frame_buffer_.reset(new uint8_t[encoded_frame._size]);
>-+    }
>-+    RTC_DCHECK(frame_buffer_);
>-+    memcpy(frame_buffer_.get(), encoded_frame._buffer, encoded_frame._length);
>-+    encoded_frame_ = encoded_frame;
>-+    encoded_frame_._buffer = frame_buffer_.get();
>-+
>-+    // Skip |codec_name|, to avoid allocating.
>-+    EXPECT_STREQ("libvpx", codec_specific_info->codec_name);
>-+    EXPECT_EQ(kVideoCodecVP8, codec_specific_info->codecType);
>-+    EXPECT_EQ(0u, codec_specific_info->codecSpecific.VP8.simulcastIdx);
>-+    codec_specific_info_.codecType = codec_specific_info->codecType;
>-+    codec_specific_info_.codecSpecific = codec_specific_info->codecSpecific;
>-+    complete_ = true;
>-+    return Result(Result::OK, 0);
>-+  }
>- 
>--webrtc::EncodedImageCallback::Result
>--Vp8UnitTestEncodeCompleteCallback::OnEncodedImage(
>--    const EncodedImage& encoded_frame,
>--    const CodecSpecificInfo* codec_specific_info,
>--    const RTPFragmentationHeader* fragmentation) {
>--  if (encoded_frame_->_size < encoded_frame._length) {
>--    delete[] encoded_frame_->_buffer;
>--    frame_buffer_.reset(new uint8_t[encoded_frame._length]);
>--    encoded_frame_->_buffer = frame_buffer_.get();
>--    encoded_frame_->_size = encoded_frame._length;
>-+  void VerifyQpParser(const EncodedImage& encoded_frame) const {
>-+    int qp;
>-+    ASSERT_TRUE(vp8::GetQp(encoded_frame._buffer, encoded_frame._length, &qp));
>-+    EXPECT_EQ(encoded_frame.qp_, qp) << "Encoder QP != parsed bitstream QP.";
>-   }
>--  memcpy(encoded_frame_->_buffer, encoded_frame._buffer, encoded_frame._length);
>--  encoded_frame_->_length = encoded_frame._length;
>--  encoded_frame_->_encodedWidth = encoded_frame._encodedWidth;
>--  encoded_frame_->_encodedHeight = encoded_frame._encodedHeight;
>--  encoded_frame_->_timeStamp = encoded_frame._timeStamp;
>--  encoded_frame_->_frameType = encoded_frame._frameType;
>--  encoded_frame_->_completeFrame = encoded_frame._completeFrame;
>--  encoded_frame_->rotation_ = encoded_frame.rotation_;
>--  encoded_frame_->qp_ = encoded_frame.qp_;
>--  codec_specific_info_->codecType = codec_specific_info->codecType;
>--  // Skip |codec_name|, to avoid allocating.
>--  codec_specific_info_->codecSpecific = codec_specific_info->codecSpecific;
>--  encode_complete_ = true;
>--  return Result(Result::OK, 0);
>--}
>- 
>--bool Vp8UnitTestEncodeCompleteCallback::EncodeComplete() {
>--  if (encode_complete_) {
>--    encode_complete_ = false;
>--    return true;
>-+  bool EncodeComplete() {
>-+    if (complete_) {
>-+      complete_ = false;
>-+      return true;
>-+    }
>-+    return false;
>-   }
>--  return false;
>--}
>- 
>--class Vp8UnitTestDecodeCompleteCallback : public webrtc::DecodedImageCallback {
>-+  EncodedImage encoded_frame_;
>-+  CodecSpecificInfo codec_specific_info_;
>-+  std::unique_ptr<uint8_t[]> frame_buffer_;
>-+  bool complete_ = false;
>-+};
>-+
>-+class DecodedImageCallbackTestImpl : public webrtc::DecodedImageCallback {
>-  public:
>--  explicit Vp8UnitTestDecodeCompleteCallback(rtc::Optional<VideoFrame>* frame,
>--                                             rtc::Optional<uint8_t>* qp)
>--      : decoded_frame_(frame), decoded_qp_(qp), decode_complete(false) {}
>-   int32_t Decoded(VideoFrame& frame) override {
>-     RTC_NOTREACHED();
>-     return -1;
>-@@ -115,163 +104,158 @@ class Vp8UnitTestDecodeCompleteCallback : public webrtc::DecodedImageCallback {
>-   }
>-   void Decoded(VideoFrame& frame,
>-                rtc::Optional<int32_t> decode_time_ms,
>--               rtc::Optional<uint8_t> qp) override;
>--  bool DecodeComplete();
>--
>-- private:
>--  rtc::Optional<VideoFrame>* decoded_frame_;
>--  rtc::Optional<uint8_t>* decoded_qp_;
>--  bool decode_complete;
>--};
>-+               rtc::Optional<uint8_t> qp) override {
>-+    EXPECT_GT(frame.width(), 0);
>-+    EXPECT_GT(frame.height(), 0);
>-+    EXPECT_TRUE(qp);
>-+    frame_ = frame;
>-+    qp_ = qp;
>-+    complete_ = true;
>-+  }
>- 
>--bool Vp8UnitTestDecodeCompleteCallback::DecodeComplete() {
>--  if (decode_complete) {
>--    decode_complete = false;
>--    return true;
>-+  bool DecodeComplete() {
>-+    if (complete_) {
>-+      complete_ = false;
>-+      return true;
>-+    }
>-+    return false;
>-   }
>--  return false;
>--}
>- 
>--void Vp8UnitTestDecodeCompleteCallback::Decoded(
>--    VideoFrame& frame,
>--    rtc::Optional<int32_t> decode_time_ms,
>--    rtc::Optional<uint8_t> qp) {
>--  *decoded_frame_ = rtc::Optional<VideoFrame>(frame);
>--  *decoded_qp_ = qp;
>--  decode_complete = true;
>--}
>-+  rtc::Optional<VideoFrame> frame_;
>-+  rtc::Optional<uint8_t> qp_;
>-+  bool complete_ = false;
>-+};
>- 
>- class TestVp8Impl : public ::testing::Test {
>-+ public:
>-+  TestVp8Impl() : TestVp8Impl("") {}
>-+  explicit TestVp8Impl(const std::string& field_trials)
>-+      : override_field_trials_(field_trials),
>-+        encoder_(VP8Encoder::Create()),
>-+        decoder_(VP8Decoder::Create()) {}
>-+  virtual ~TestVp8Impl() {}
>-+
>-  protected:
>-   virtual void SetUp() {
>--    encoder_.reset(VP8Encoder::Create());
>--    decoder_.reset(VP8Decoder::Create());
>--    memset(&codec_settings_, 0, sizeof(codec_settings_));
>--    encode_complete_callback_.reset(new Vp8UnitTestEncodeCompleteCallback(
>--        &encoded_frame_, &codec_specific_info_, 0, nullptr));
>--    decode_complete_callback_.reset(
>--        new Vp8UnitTestDecodeCompleteCallback(&decoded_frame_, &decoded_qp_));
>--    encoder_->RegisterEncodeCompleteCallback(encode_complete_callback_.get());
>--    decoder_->RegisterDecodeCompleteCallback(decode_complete_callback_.get());
>-+    encoder_->RegisterEncodeCompleteCallback(&encoded_cb_);
>-+    decoder_->RegisterDecodeCompleteCallback(&decoded_cb_);
>-+    SetupCodecSettings();
>-+    SetupInputFrame();
>-+  }
>-+
>-+  void SetupInputFrame() {
>-     // Using a QCIF image (aligned stride (u,v planes) > width).
>-     // Processing only one frame.
>--    source_file_ = fopen(test::ResourcePath("paris_qcif", "yuv").c_str(), "rb");
>--    ASSERT_TRUE(source_file_ != nullptr);
>-+    FILE* file = fopen(test::ResourcePath("paris_qcif", "yuv").c_str(), "rb");
>-+    ASSERT_TRUE(file != nullptr);
>-     rtc::scoped_refptr<I420BufferInterface> compact_buffer(
>--        test::ReadI420Buffer(kWidth, kHeight, source_file_));
>-+        test::ReadI420Buffer(kWidth, kHeight, file));
>-     ASSERT_TRUE(compact_buffer);
>--    codec_settings_.width = kWidth;
>--    codec_settings_.height = kHeight;
>--    const int kFramerate = 30;
>--    codec_settings_.maxFramerate = kFramerate;
>-+
>-     // Setting aligned stride values.
>-     int stride_uv;
>-     int stride_y;
>--    Calc16ByteAlignedStride(codec_settings_.width, &stride_y, &stride_uv);
>-+    Calc16ByteAlignedStride(kWidth, &stride_y, &stride_uv);
>-     EXPECT_EQ(stride_y, 176);
>-     EXPECT_EQ(stride_uv, 96);
>--
>-     rtc::scoped_refptr<I420Buffer> stride_buffer(
>-         I420Buffer::Create(kWidth, kHeight, stride_y, stride_uv, stride_uv));
>- 
>-     // No scaling in our case, just a copy, to add stride to the image.
>-     stride_buffer->ScaleFrom(*compact_buffer);
>- 
>--    input_frame_.reset(
>--        new VideoFrame(stride_buffer, kVideoRotation_0, 0));
>--    input_frame_->set_timestamp(kTestTimestamp);
>-+    input_frame_.reset(new VideoFrame(stride_buffer, kInitialTimestampRtp,
>-+                                      kInitialTimestampMs, kVideoRotation_0));
>-+    fclose(file);
>-   }
>- 
>--  void SetUpEncodeDecode() {
>--    codec_settings_.startBitrate = 300;
>-+  void SetupCodecSettings() {
>-+    webrtc::test::CodecSettings(kVideoCodecVP8, &codec_settings_);
>-     codec_settings_.maxBitrate = 4000;
>--    codec_settings_.qpMax = 56;
>-+    codec_settings_.width = kWidth;
>-+    codec_settings_.height = kHeight;
>-     codec_settings_.VP8()->denoisingOn = true;
>-+    codec_settings_.VP8()->frameDroppingOn = false;
>-+    codec_settings_.VP8()->automaticResizeOn = false;
>-+    codec_settings_.VP8()->complexity = kComplexityNormal;
>-     codec_settings_.VP8()->tl_factory = &tl_factory_;
>--    codec_settings_.VP8()->numberOfTemporalLayers = 1;
>--
>--    EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>--              encoder_->InitEncode(&codec_settings_, 1, 1440));
>--    EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, decoder_->InitDecode(&codec_settings_, 1));
>-   }
>- 
>--  size_t WaitForEncodedFrame() const {
>--    int64_t startTime = rtc::TimeMillis();
>--    while (rtc::TimeMillis() - startTime < kMaxWaitEncTimeMs) {
>--      if (encode_complete_callback_->EncodeComplete()) {
>--        return encoded_frame_._length;
>--      }
>--    }
>--    return 0;
>-+  void InitEncodeDecode() {
>-+    EXPECT_EQ(
>-+        WEBRTC_VIDEO_CODEC_OK,
>-+        encoder_->InitEncode(&codec_settings_, kNumCores, kMaxPayloadSize));
>-+    EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>-+              decoder_->InitDecode(&codec_settings_, kNumCores));
>-   }
>- 
>--  size_t WaitForDecodedFrame() const {
>--    int64_t startTime = rtc::TimeMillis();
>--    while (rtc::TimeMillis() - startTime < kMaxWaitDecTimeMs) {
>--      if (decode_complete_callback_->DecodeComplete()) {
>--        return CalcBufferSize(VideoType::kI420, decoded_frame_->width(),
>--                              decoded_frame_->height());
>--      }
>--    }
>--    return 0;
>-+  void EncodeFrame() {
>-+    EXPECT_FALSE(encoded_cb_.EncodeComplete());
>-+    EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>-+              encoder_->Encode(*input_frame_, nullptr, nullptr));
>-+    EXPECT_TRUE(encoded_cb_.EncodeComplete());
>-   }
>- 
>-   void ExpectFrameWith(int16_t picture_id,
>-                        int tl0_pic_idx,
>-                        uint8_t temporal_idx) {
>--    ASSERT_TRUE(WaitForEncodedFrame());
>--    EXPECT_EQ(picture_id, codec_specific_info_.codecSpecific.VP8.pictureId);
>--    EXPECT_EQ(tl0_pic_idx, codec_specific_info_.codecSpecific.VP8.tl0PicIdx);
>--    EXPECT_EQ(temporal_idx, codec_specific_info_.codecSpecific.VP8.temporalIdx);
>-+    EXPECT_EQ(picture_id % (1 << 15),
>-+              encoded_cb_.codec_specific_info_.codecSpecific.VP8.pictureId);
>-+    EXPECT_EQ(tl0_pic_idx % (1 << 8),
>-+              encoded_cb_.codec_specific_info_.codecSpecific.VP8.tl0PicIdx);
>-+    EXPECT_EQ(temporal_idx,
>-+              encoded_cb_.codec_specific_info_.codecSpecific.VP8.temporalIdx);
>-   }
>- 
>--  const int kWidth = 172;
>--  const int kHeight = 144;
>--
>--  std::unique_ptr<Vp8UnitTestEncodeCompleteCallback> encode_complete_callback_;
>--  std::unique_ptr<Vp8UnitTestDecodeCompleteCallback> decode_complete_callback_;
>--  std::unique_ptr<uint8_t[]> source_buffer_;
>--  FILE* source_file_;
>-+  test::ScopedFieldTrials override_field_trials_;
>-+  EncodedImageCallbackTestImpl encoded_cb_;
>-+  DecodedImageCallbackTestImpl decoded_cb_;
>-   std::unique_ptr<VideoFrame> input_frame_;
>--  std::unique_ptr<VideoEncoder> encoder_;
>--  std::unique_ptr<VideoDecoder> decoder_;
>--  EncodedImage encoded_frame_;
>--  CodecSpecificInfo codec_specific_info_;
>--  rtc::Optional<VideoFrame> decoded_frame_;
>--  rtc::Optional<uint8_t> decoded_qp_;
>-+  const std::unique_ptr<VideoEncoder> encoder_;
>-+  const std::unique_ptr<VideoDecoder> decoder_;
>-   VideoCodec codec_settings_;
>-   TemporalLayersFactory tl_factory_;
>- };
>- 
>--TEST_F(TestVp8Impl, EncoderParameterTest) {
>--  strncpy(codec_settings_.plName, "VP8", 31);
>--  codec_settings_.plType = 126;
>--  codec_settings_.maxBitrate = 0;
>--  codec_settings_.minBitrate = 0;
>--  codec_settings_.width = 1440;
>--  codec_settings_.height = 1080;
>--  codec_settings_.maxFramerate = 30;
>--  codec_settings_.startBitrate = 300;
>--  codec_settings_.qpMax = 56;
>--  codec_settings_.VP8()->complexity = kComplexityNormal;
>--  codec_settings_.VP8()->numberOfTemporalLayers = 1;
>--  codec_settings_.VP8()->tl_factory = &tl_factory_;
>--  // Calls before InitEncode().
>--  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Release());
>--  int bit_rate = 300;
>-+TEST_F(TestVp8Impl, SetRateAllocation) {
>-+  const int kBitrateBps = 300000;
>-   BitrateAllocation bitrate_allocation;
>--  bitrate_allocation.SetBitrate(0, 0, bit_rate * 1000);
>-+  bitrate_allocation.SetBitrate(0, 0, kBitrateBps);
>-   EXPECT_EQ(WEBRTC_VIDEO_CODEC_UNINITIALIZED,
>-             encoder_->SetRateAllocation(bitrate_allocation,
>-                                         codec_settings_.maxFramerate));
>-+  InitEncodeDecode();
>-+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>-+            encoder_->SetRateAllocation(bitrate_allocation,
>-+                                        codec_settings_.maxFramerate));
>-+}
>- 
>-+TEST_F(TestVp8Impl, EncodeFrameAndRelease) {
>-+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Release());
>-   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>--            encoder_->InitEncode(&codec_settings_, 1, 1440));
>-+            encoder_->InitEncode(&codec_settings_, kNumCores, kMaxPayloadSize));
>-+  EncodeFrame();
>-+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Release());
>-+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_UNINITIALIZED,
>-+            encoder_->Encode(*input_frame_, nullptr, nullptr));
>-+}
>- 
>--  // Decoder parameter tests.
>--  // Calls before InitDecode().
>-+TEST_F(TestVp8Impl, InitDecode) {
>-   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, decoder_->Release());
>--  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, decoder_->InitDecode(&codec_settings_, 1));
>-+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>-+            decoder_->InitDecode(&codec_settings_, kNumCores));
>-+}
>-+
>-+TEST_F(TestVp8Impl, OnEncodedImageReportsInfo) {
>-+  InitEncodeDecode();
>-+  EncodeFrame();
>-+  EXPECT_EQ(kInitialTimestampRtp, encoded_cb_.encoded_frame_._timeStamp);
>-+  EXPECT_EQ(kInitialTimestampMs, encoded_cb_.encoded_frame_.capture_time_ms_);
>-+  EXPECT_EQ(kWidth, static_cast<int>(encoded_cb_.encoded_frame_._encodedWidth));
>-+  EXPECT_EQ(kHeight,
>-+            static_cast<int>(encoded_cb_.encoded_frame_._encodedHeight));
>-+  EXPECT_EQ(-1,  // Disabled for single stream.
>-+            encoded_cb_.encoded_frame_.adapt_reason_.bw_resolutions_disabled);
>- }
>- 
>- // We only test the encoder here, since the decoded frame rotation is set based
>-@@ -279,46 +263,84 @@ TEST_F(TestVp8Impl, EncoderParameterTest) {
>- // TODO(brandtr): Consider passing through the rotation flag through the decoder
>- // in the same way as done in the encoder.
>- TEST_F(TestVp8Impl, EncodedRotationEqualsInputRotation) {
>--  SetUpEncodeDecode();
>--
>-+  InitEncodeDecode();
>-   input_frame_->set_rotation(kVideoRotation_0);
>--  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>--            encoder_->Encode(*input_frame_, nullptr, nullptr));
>--  WaitForEncodedFrame();
>--  EXPECT_EQ(kVideoRotation_0, encoded_frame_.rotation_);
>-+  EncodeFrame();
>-+  EXPECT_EQ(kVideoRotation_0, encoded_cb_.encoded_frame_.rotation_);
>- 
>-   input_frame_->set_rotation(kVideoRotation_90);
>--  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>--            encoder_->Encode(*input_frame_, nullptr, nullptr));
>--  WaitForEncodedFrame();
>--  EXPECT_EQ(kVideoRotation_90, encoded_frame_.rotation_);
>-+  EncodeFrame();
>-+  EXPECT_EQ(kVideoRotation_90, encoded_cb_.encoded_frame_.rotation_);
>- }
>- 
>- TEST_F(TestVp8Impl, DecodedQpEqualsEncodedQp) {
>--  SetUpEncodeDecode();
>--  encoder_->Encode(*input_frame_, nullptr, nullptr);
>--  EXPECT_GT(WaitForEncodedFrame(), 0u);
>-+  InitEncodeDecode();
>-+  EncodeFrame();
>-   // First frame should be a key frame.
>--  encoded_frame_._frameType = kVideoFrameKey;
>-+  encoded_cb_.encoded_frame_._frameType = kVideoFrameKey;
>-   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>--            decoder_->Decode(encoded_frame_, false, nullptr));
>--  EXPECT_GT(WaitForDecodedFrame(), 0u);
>--  ASSERT_TRUE(decoded_frame_);
>--  EXPECT_GT(I420PSNR(input_frame_.get(), &*decoded_frame_), 36);
>--  ASSERT_TRUE(decoded_qp_);
>--  EXPECT_EQ(encoded_frame_.qp_, *decoded_qp_);
>-+            decoder_->Decode(encoded_cb_.encoded_frame_, false, nullptr));
>-+  EXPECT_TRUE(decoded_cb_.DecodeComplete());
>-+  EXPECT_GT(I420PSNR(input_frame_.get(), &*decoded_cb_.frame_), 36);
>-+  EXPECT_EQ(encoded_cb_.encoded_frame_.qp_, *decoded_cb_.qp_);
>- }
>- 
>--TEST_F(TestVp8Impl, ParserQpEqualsEncodedQp) {
>--  SetUpEncodeDecode();
>-+TEST_F(TestVp8Impl, ChecksSimulcastSettings) {
>-+  codec_settings_.numberOfSimulcastStreams = 2;
>-+  // Reslutions are not scaled by 2, temporal layers do not match.
>-+  codec_settings_.simulcastStream[0] = {kWidth, kHeight, 2, 4000,
>-+                                        3000,   2000,    80};
>-+  codec_settings_.simulcastStream[1] = {kWidth, kHeight, 3, 4000,
>-+                                        3000,   2000,    80};
>-+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_ERR_SIMULCAST_PARAMETERS_NOT_SUPPORTED,
>-+            encoder_->InitEncode(&codec_settings_, kNumCores, kMaxPayloadSize));
>-+  codec_settings_.numberOfSimulcastStreams = 3;
>-+  // Reslutions are not scaled by 2.
>-+  codec_settings_.simulcastStream[0] = {kWidth / 2, kHeight / 2, 1, 4000,
>-+                                        3000,       2000,        80};
>-+  codec_settings_.simulcastStream[1] = {kWidth / 2, kHeight / 2, 1, 4000,
>-+                                        3000,       2000,        80};
>-+  codec_settings_.simulcastStream[2] = {kWidth, kHeight, 1, 4000,
>-+                                        3000,   2000,    80};
>-+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_ERR_SIMULCAST_PARAMETERS_NOT_SUPPORTED,
>-+            encoder_->InitEncode(&codec_settings_, kNumCores, kMaxPayloadSize));
>-+  // Reslutions are not scaled by 2.
>-+  codec_settings_.simulcastStream[0] = {kWidth, kHeight, 1, 4000,
>-+                                        3000,   2000,    80};
>-+  codec_settings_.simulcastStream[1] = {kWidth, kHeight, 1, 4000,
>-+                                        3000,   2000,    80};
>-+  codec_settings_.simulcastStream[2] = {kWidth, kHeight, 1, 4000,
>-+                                        3000,   2000,    80};
>-+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_ERR_SIMULCAST_PARAMETERS_NOT_SUPPORTED,
>-+            encoder_->InitEncode(&codec_settings_, kNumCores, kMaxPayloadSize));
>-+  // Temporal layers do not match.
>-+  codec_settings_.simulcastStream[0] = {kWidth / 4, kHeight / 4, 1, 4000,
>-+                                        3000,       2000,        80};
>-+  codec_settings_.simulcastStream[1] = {kWidth / 2, kHeight / 2, 2, 4000,
>-+                                        3000,       2000,        80};
>-+  codec_settings_.simulcastStream[2] = {kWidth, kHeight, 3, 4000,
>-+                                        3000,   2000,    80};
>-+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_ERR_SIMULCAST_PARAMETERS_NOT_SUPPORTED,
>-+            encoder_->InitEncode(&codec_settings_, kNumCores, kMaxPayloadSize));
>-+  // Resolutions do not match codec config.
>-+  codec_settings_.simulcastStream[0] = {
>-+      kWidth / 4 + 1, kHeight / 4 + 1, 1, 4000, 3000, 2000, 80};
>-+  codec_settings_.simulcastStream[1] = {
>-+      kWidth / 2 + 2, kHeight / 2 + 2, 1, 4000, 3000, 2000, 80};
>-+  codec_settings_.simulcastStream[2] = {kWidth + 4, kHeight + 4, 1, 4000,
>-+                                        3000,       2000,        80};
>-+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_ERR_SIMULCAST_PARAMETERS_NOT_SUPPORTED,
>-+            encoder_->InitEncode(&codec_settings_, kNumCores, kMaxPayloadSize));
>-+  // Everything fine: scaling by 2, top resolution matches video, temporal
>-+  // settings are the same for all layers.
>-+  codec_settings_.simulcastStream[0] = {kWidth / 4, kHeight / 4, 1, 4000,
>-+                                        3000,       2000,        80};
>-+  codec_settings_.simulcastStream[1] = {kWidth / 2, kHeight / 2, 1, 4000,
>-+                                        3000,       2000,        80};
>-+  codec_settings_.simulcastStream[2] = {kWidth, kHeight, 1, 4000,
>-+                                        3000,   2000,    80};
>-   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>--            encoder_->Encode(*input_frame_, nullptr, nullptr));
>--  EXPECT_GT(WaitForEncodedFrame(), 0u);
>--
>--  int qp = 0;
>--  ASSERT_TRUE(vp8::GetQp(encoded_frame_._buffer, encoded_frame_._length, &qp));
>--
>--  EXPECT_EQ(encoded_frame_.qp_, qp);
>-+            encoder_->InitEncode(&codec_settings_, kNumCores, kMaxPayloadSize));
>- }
>- 
>- #if defined(WEBRTC_ANDROID)
>-@@ -327,20 +349,18 @@ TEST_F(TestVp8Impl, ParserQpEqualsEncodedQp) {
>- #define MAYBE_AlignedStrideEncodeDecode AlignedStrideEncodeDecode
>- #endif
>- TEST_F(TestVp8Impl, MAYBE_AlignedStrideEncodeDecode) {
>--  SetUpEncodeDecode();
>--  encoder_->Encode(*input_frame_, nullptr, nullptr);
>--  EXPECT_GT(WaitForEncodedFrame(), 0u);
>-+  InitEncodeDecode();
>-+  EncodeFrame();
>-   // First frame should be a key frame.
>--  encoded_frame_._frameType = kVideoFrameKey;
>--  encoded_frame_.ntp_time_ms_ = kTestNtpTimeMs;
>-+  encoded_cb_.encoded_frame_._frameType = kVideoFrameKey;
>-+  encoded_cb_.encoded_frame_.ntp_time_ms_ = kTestNtpTimeMs;
>-   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>--            decoder_->Decode(encoded_frame_, false, nullptr));
>--  EXPECT_GT(WaitForDecodedFrame(), 0u);
>--  ASSERT_TRUE(decoded_frame_);
>-+            decoder_->Decode(encoded_cb_.encoded_frame_, false, nullptr));
>-+  EXPECT_TRUE(decoded_cb_.DecodeComplete());
>-   // Compute PSNR on all planes (faster than SSIM).
>--  EXPECT_GT(I420PSNR(input_frame_.get(), &*decoded_frame_), 36);
>--  EXPECT_EQ(kTestTimestamp, decoded_frame_->timestamp());
>--  EXPECT_EQ(kTestNtpTimeMs, decoded_frame_->ntp_time_ms());
>-+  EXPECT_GT(I420PSNR(input_frame_.get(), &*decoded_cb_.frame_), 36);
>-+  EXPECT_EQ(kInitialTimestampRtp, decoded_cb_.frame_->timestamp());
>-+  EXPECT_EQ(kTestNtpTimeMs, decoded_cb_.frame_->ntp_time_ms());
>- }
>- 
>- #if defined(WEBRTC_ANDROID)
>-@@ -349,100 +369,138 @@ TEST_F(TestVp8Impl, MAYBE_AlignedStrideEncodeDecode) {
>- #define MAYBE_DecodeWithACompleteKeyFrame DecodeWithACompleteKeyFrame
>- #endif
>- TEST_F(TestVp8Impl, MAYBE_DecodeWithACompleteKeyFrame) {
>--  SetUpEncodeDecode();
>--  encoder_->Encode(*input_frame_, nullptr, nullptr);
>--  EXPECT_GT(WaitForEncodedFrame(), 0u);
>-+  InitEncodeDecode();
>-+  EncodeFrame();
>-   // Setting complete to false -> should return an error.
>--  encoded_frame_._completeFrame = false;
>-+  encoded_cb_.encoded_frame_._completeFrame = false;
>-   EXPECT_EQ(WEBRTC_VIDEO_CODEC_ERROR,
>--            decoder_->Decode(encoded_frame_, false, nullptr));
>-+            decoder_->Decode(encoded_cb_.encoded_frame_, false, nullptr));
>-   // Setting complete back to true. Forcing a delta frame.
>--  encoded_frame_._frameType = kVideoFrameDelta;
>--  encoded_frame_._completeFrame = true;
>-+  encoded_cb_.encoded_frame_._frameType = kVideoFrameDelta;
>-+  encoded_cb_.encoded_frame_._completeFrame = true;
>-   EXPECT_EQ(WEBRTC_VIDEO_CODEC_ERROR,
>--            decoder_->Decode(encoded_frame_, false, nullptr));
>-+            decoder_->Decode(encoded_cb_.encoded_frame_, false, nullptr));
>-   // Now setting a key frame.
>--  encoded_frame_._frameType = kVideoFrameKey;
>-+  encoded_cb_.encoded_frame_._frameType = kVideoFrameKey;
>-   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>--            decoder_->Decode(encoded_frame_, false, nullptr));
>--  ASSERT_TRUE(decoded_frame_);
>--  EXPECT_GT(I420PSNR(input_frame_.get(), &*decoded_frame_), 36);
>-+            decoder_->Decode(encoded_cb_.encoded_frame_, false, nullptr));
>-+  ASSERT_TRUE(decoded_cb_.frame_);
>-+  EXPECT_GT(I420PSNR(input_frame_.get(), &*decoded_cb_.frame_), 36);
>- }
>- 
>--TEST_F(TestVp8Impl, EncoderRetainsRtpStateAfterRelease) {
>--  SetUpEncodeDecode();
>--  // Override default settings.
>-+TEST_F(TestVp8Impl, EncoderWith2TemporalLayersRetainsRtpStateAfterRelease) {
>-   codec_settings_.VP8()->numberOfTemporalLayers = 2;
>--  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>--            encoder_->InitEncode(&codec_settings_, 1, 1440));
>-+  InitEncodeDecode();
>- 
>-   // Temporal layer 0.
>--  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>--            encoder_->Encode(*input_frame_, nullptr, nullptr));
>--  ASSERT_TRUE(WaitForEncodedFrame());
>--  EXPECT_EQ(0, codec_specific_info_.codecSpecific.VP8.temporalIdx);
>--  int16_t picture_id = codec_specific_info_.codecSpecific.VP8.pictureId;
>--  int tl0_pic_idx = codec_specific_info_.codecSpecific.VP8.tl0PicIdx;
>--
>-+  EncodeFrame();
>-+  EXPECT_EQ(0, encoded_cb_.codec_specific_info_.codecSpecific.VP8.temporalIdx);
>-+  int16_t picture_id =
>-+      encoded_cb_.codec_specific_info_.codecSpecific.VP8.pictureId;
>-+  int tl0_pic_idx =
>-+      encoded_cb_.codec_specific_info_.codecSpecific.VP8.tl0PicIdx;
>-   // Temporal layer 1.
>--  input_frame_->set_timestamp(input_frame_->timestamp() +
>--                              kTimestampIncrementPerFrame);
>--  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>--            encoder_->Encode(*input_frame_, nullptr, nullptr));
>--  ExpectFrameWith((picture_id + 1) % (1 << 15), tl0_pic_idx, 1);
>--
>-+  input_frame_->set_timestamp(input_frame_->timestamp() + kTimestampIncrement);
>-+  EncodeFrame();
>-+  ExpectFrameWith(picture_id + 1, tl0_pic_idx + 0, 1);
>-   // Temporal layer 0.
>--  input_frame_->set_timestamp(input_frame_->timestamp() +
>--                              kTimestampIncrementPerFrame);
>--  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>--            encoder_->Encode(*input_frame_, nullptr, nullptr));
>--  ExpectFrameWith((picture_id + 2) % (1 << 15), (tl0_pic_idx + 1) % (1 << 8),
>--                  0);
>--
>-+  input_frame_->set_timestamp(input_frame_->timestamp() + kTimestampIncrement);
>-+  EncodeFrame();
>-+  ExpectFrameWith(picture_id + 2, tl0_pic_idx + 1, 0);
>-   // Temporal layer 1.
>--  input_frame_->set_timestamp(input_frame_->timestamp() +
>--                              kTimestampIncrementPerFrame);
>--  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>--            encoder_->Encode(*input_frame_, nullptr, nullptr));
>--  ExpectFrameWith((picture_id + 3) % (1 << 15), (tl0_pic_idx + 1) % (1 << 8),
>--                  1);
>-+  input_frame_->set_timestamp(input_frame_->timestamp() + kTimestampIncrement);
>-+  EncodeFrame();
>-+  ExpectFrameWith(picture_id + 3, tl0_pic_idx + 1, 1);
>- 
>-   // Reinit.
>-   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Release());
>-   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>--            encoder_->InitEncode(&codec_settings_, 1, 1440));
>-+            encoder_->InitEncode(&codec_settings_, kNumCores, kMaxPayloadSize));
>- 
>-   // Temporal layer 0.
>--  input_frame_->set_timestamp(input_frame_->timestamp() +
>--                              kTimestampIncrementPerFrame);
>--  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>--            encoder_->Encode(*input_frame_, nullptr, nullptr));
>--  ExpectFrameWith((picture_id + 4) % (1 << 15), (tl0_pic_idx + 2) % (1 << 8),
>--                  0);
>-+  input_frame_->set_timestamp(input_frame_->timestamp() + kTimestampIncrement);
>-+  EncodeFrame();
>-+  ExpectFrameWith(picture_id + 4, tl0_pic_idx + 2, 0);
>-+  // Temporal layer 1.
>-+  input_frame_->set_timestamp(input_frame_->timestamp() + kTimestampIncrement);
>-+  EncodeFrame();
>-+  ExpectFrameWith(picture_id + 5, tl0_pic_idx + 2, 1);
>-+  // Temporal layer 0.
>-+  input_frame_->set_timestamp(input_frame_->timestamp() + kTimestampIncrement);
>-+  EncodeFrame();
>-+  ExpectFrameWith(picture_id + 6, tl0_pic_idx + 3, 0);
>-+  // Temporal layer 1.
>-+  input_frame_->set_timestamp(input_frame_->timestamp() + kTimestampIncrement);
>-+  EncodeFrame();
>-+  ExpectFrameWith(picture_id + 7, tl0_pic_idx + 3, 1);
>-+}
>- 
>-+TEST_F(TestVp8Impl, EncoderWith3TemporalLayersRetainsRtpStateAfterRelease) {
>-+  codec_settings_.VP8()->numberOfTemporalLayers = 3;
>-+  InitEncodeDecode();
>-+
>-+  // Temporal layer 0.
>-+  EncodeFrame();
>-+  EXPECT_EQ(0, encoded_cb_.codec_specific_info_.codecSpecific.VP8.temporalIdx);
>-+  int16_t picture_id =
>-+      encoded_cb_.codec_specific_info_.codecSpecific.VP8.pictureId;
>-+  int tl0_pic_idx =
>-+      encoded_cb_.codec_specific_info_.codecSpecific.VP8.tl0PicIdx;
>-+  // Temporal layer 2.
>-+  input_frame_->set_timestamp(input_frame_->timestamp() + kTimestampIncrement);
>-+  EncodeFrame();
>-+  ExpectFrameWith(picture_id + 1, tl0_pic_idx + 0, 2);
>-   // Temporal layer 1.
>--  input_frame_->set_timestamp(input_frame_->timestamp() +
>--                              kTimestampIncrementPerFrame);
>-+  input_frame_->set_timestamp(input_frame_->timestamp() + kTimestampIncrement);
>-+  EncodeFrame();
>-+  ExpectFrameWith(picture_id + 2, tl0_pic_idx + 0, 1);
>-+  // Temporal layer 2.
>-+  input_frame_->set_timestamp(input_frame_->timestamp() + kTimestampIncrement);
>-+  EncodeFrame();
>-+  ExpectFrameWith(picture_id + 3, tl0_pic_idx + 0, 2);
>-+
>-+  // Reinit.
>-+  EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Release());
>-   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>--            encoder_->Encode(*input_frame_, nullptr, nullptr));
>--  ExpectFrameWith((picture_id + 5) % (1 << 15), (tl0_pic_idx + 2) % (1 << 8),
>--                  1);
>-+            encoder_->InitEncode(&codec_settings_, kNumCores, kMaxPayloadSize));
>- 
>-   // Temporal layer 0.
>--  input_frame_->set_timestamp(input_frame_->timestamp() +
>--                              kTimestampIncrementPerFrame);
>-+  input_frame_->set_timestamp(input_frame_->timestamp() + kTimestampIncrement);
>-+  EncodeFrame();
>-+  ExpectFrameWith(picture_id + 4, tl0_pic_idx + 1, 0);
>-+  // Temporal layer 2.
>-+  input_frame_->set_timestamp(input_frame_->timestamp() + kTimestampIncrement);
>-+  EncodeFrame();
>-+  ExpectFrameWith(picture_id + 5, tl0_pic_idx + 1, 2);
>-+  // Temporal layer 1.
>-+  input_frame_->set_timestamp(input_frame_->timestamp() + kTimestampIncrement);
>-+  EncodeFrame();
>-+  ExpectFrameWith(picture_id + 6, tl0_pic_idx + 1, 1);
>-+  // Temporal layer 2.
>-+  input_frame_->set_timestamp(input_frame_->timestamp() + kTimestampIncrement);
>-+  EncodeFrame();
>-+  ExpectFrameWith(picture_id + 7, tl0_pic_idx + 1, 2);
>-+}
>-+
>-+TEST_F(TestVp8Impl, ScalingDisabledIfAutomaticResizeOff) {
>-+  codec_settings_.VP8()->frameDroppingOn = true;
>-+  codec_settings_.VP8()->automaticResizeOn = false;
>-   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>--            encoder_->Encode(*input_frame_, nullptr, nullptr));
>--  ExpectFrameWith((picture_id + 6) % (1 << 15), (tl0_pic_idx + 3) % (1 << 8),
>--                  0);
>-+            encoder_->InitEncode(&codec_settings_, kNumCores, kMaxPayloadSize));
>- 
>--  // Temporal layer 1.
>--  input_frame_->set_timestamp(input_frame_->timestamp() +
>--                              kTimestampIncrementPerFrame);
>-+  VideoEncoder::ScalingSettings settings = encoder_->GetScalingSettings();
>-+  EXPECT_FALSE(settings.enabled);
>-+}
>-+
>-+TEST_F(TestVp8Impl, ScalingEnabledIfAutomaticResizeOn) {
>-+  codec_settings_.VP8()->frameDroppingOn = true;
>-+  codec_settings_.VP8()->automaticResizeOn = true;
>-   EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK,
>--            encoder_->Encode(*input_frame_, nullptr, nullptr));
>--  ExpectFrameWith((picture_id + 7) % (1 << 15), (tl0_pic_idx + 3) % (1 << 8),
>--                  1);
>-+            encoder_->InitEncode(&codec_settings_, kNumCores, kMaxPayloadSize));
>-+
>-+  VideoEncoder::ScalingSettings settings = encoder_->GetScalingSettings();
>-+  EXPECT_TRUE(settings.enabled);
>-+  EXPECT_EQ(kDefaultMinPixelsPerFrame, settings.min_pixels_per_frame);
>- }
>- 
>- }  // namespace webrtc
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
>-index 5038e8edf49..9e89696d465 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/vp8_impl.cc
>-@@ -8,7 +8,7 @@
>-  *  be found in the AUTHORS file in the root of the source tree.
>-  */
>- 
>--#include "webrtc/modules/video_coding/codecs/vp8/vp8_impl.h"
>-+#include "modules/video_coding/codecs/vp8/vp8_impl.h"
>- 
>- #include <stdlib.h>
>- #include <string.h>
>-@@ -17,29 +17,31 @@
>- #include <string>
>- 
>- // NOTE(ajm): Path provided by gyp.
>--#include "libyuv/scale.h"    // NOLINT
>- #include "libyuv/convert.h"  // NOLINT
>-+#include "libyuv/scale.h"    // NOLINT
>- 
>--#include "webrtc/base/checks.h"
>--#include "webrtc/base/random.h"
>--#include "webrtc/base/timeutils.h"
>--#include "webrtc/base/trace_event.h"
>--#include "webrtc/common_types.h"
>--#include "webrtc/common_video/libyuv/include/webrtc_libyuv.h"
>--#include "webrtc/modules/include/module_common_types.h"
>--#include "webrtc/modules/video_coding/codecs/vp8/include/vp8_common_types.h"
>--#include "webrtc/modules/video_coding/codecs/vp8/screenshare_layers.h"
>--#include "webrtc/modules/video_coding/codecs/vp8/simulcast_rate_allocator.h"
>--#include "webrtc/modules/video_coding/codecs/vp8/temporal_layers.h"
>--#include "webrtc/modules/video_coding/include/video_codec_interface.h"
>--#include "webrtc/system_wrappers/include/clock.h"
>--#include "webrtc/system_wrappers/include/field_trial.h"
>--#include "webrtc/system_wrappers/include/metrics.h"
>-+#include "common_types.h"  // NOLINT(build/include)
>-+#include "common_video/libyuv/include/webrtc_libyuv.h"
>-+#include "modules/include/module_common_types.h"
>-+#include "modules/video_coding/codecs/vp8/include/vp8_common_types.h"
>-+#include "modules/video_coding/codecs/vp8/screenshare_layers.h"
>-+#include "modules/video_coding/codecs/vp8/simulcast_rate_allocator.h"
>-+#include "modules/video_coding/codecs/vp8/temporal_layers.h"
>-+#include "modules/video_coding/include/video_codec_interface.h"
>-+#include "rtc_base/checks.h"
>-+#include "rtc_base/numerics/exp_filter.h"
>-+#include "rtc_base/ptr_util.h"
>-+#include "rtc_base/random.h"
>-+#include "rtc_base/timeutils.h"
>-+#include "rtc_base/trace_event.h"
>-+#include "system_wrappers/include/clock.h"
>-+#include "system_wrappers/include/field_trial.h"
>-+#include "system_wrappers/include/metrics.h"
>- 
>- namespace webrtc {
>- namespace {
>- 
>--const char kVp8PostProcArmFieldTrial[] = "WebRTC-VP8-Postproc-Arm";
>-+const char kVp8PostProcArmFieldTrial[] = "WebRTC-VP8-Postproc-Config-Arm";
>- const char kVp8GfBoostFieldTrial[] = "WebRTC-VP8-GfBoost";
>- 
>- const int kTokenPartitions = VP8_ONE_TOKENPARTITION;
>-@@ -97,6 +99,21 @@ bool ValidSimulcastResolutions(const VideoCodec& codec, int num_streams) {
>-       return false;
>-     }
>-   }
>-+  for (int i = 1; i < num_streams; ++i) {
>-+    if (codec.simulcastStream[i].width !=
>-+        codec.simulcastStream[i - 1].width * 2) {
>-+      return false;
>-+    }
>-+  }
>-+  return true;
>-+}
>-+
>-+bool ValidSimulcastTemporalLayers(const VideoCodec& codec, int num_streams) {
>-+  for (int i = 0; i < num_streams - 1; ++i) {
>-+    if (codec.simulcastStream[i].numberOfTemporalLayers !=
>-+        codec.simulcastStream[i + 1].numberOfTemporalLayers)
>-+      return false;
>-+  }
>-   return true;
>- }
>- 
>-@@ -122,14 +139,36 @@ bool GetGfBoostPercentageFromFieldTrialGroup(int* boost_percentage) {
>- 
>-   return true;
>- }
>-+
>-+void GetPostProcParamsFromFieldTrialGroup(
>-+    VP8DecoderImpl::DeblockParams* deblock_params) {
>-+  std::string group =
>-+      webrtc::field_trial::FindFullName(kVp8PostProcArmFieldTrial);
>-+  if (group.empty())
>-+    return;
>-+
>-+  VP8DecoderImpl::DeblockParams params;
>-+  if (sscanf(group.c_str(), "Enabled-%d,%d,%d", &params.max_level,
>-+             &params.min_qp, &params.degrade_qp) != 3)
>-+    return;
>-+
>-+  if (params.max_level < 0 || params.max_level > 16)
>-+    return;
>-+
>-+  if (params.min_qp < 0 || params.degrade_qp <= params.min_qp)
>-+    return;
>-+
>-+  *deblock_params = params;
>-+}
>-+
>- }  // namespace
>- 
>--VP8Encoder* VP8Encoder::Create() {
>--  return new VP8EncoderImpl();
>-+std::unique_ptr<VP8Encoder> VP8Encoder::Create() {
>-+  return rtc::MakeUnique<VP8EncoderImpl>();
>- }
>- 
>--VP8Decoder* VP8Decoder::Create() {
>--  return new VP8DecoderImpl();
>-+std::unique_ptr<VP8Decoder> VP8Decoder::Create() {
>-+  return rtc::MakeUnique<VP8DecoderImpl>();
>- }
>- 
>- vpx_enc_frame_flags_t VP8EncoderImpl::EncodeFlags(
>-@@ -173,6 +212,7 @@ VP8EncoderImpl::VP8EncoderImpl()
>-     tl0_pic_idx_.push_back(random.Rand<uint8_t>());
>-   }
>-   temporal_layers_.reserve(kMaxSimulcastStreams);
>-+  temporal_layers_checkers_.reserve(kMaxSimulcastStreams);
>-   raw_images_.reserve(kMaxSimulcastStreams);
>-   encoded_images_.reserve(kMaxSimulcastStreams);
>-   send_stream_.reserve(kMaxSimulcastStreams);
>-@@ -212,6 +252,7 @@ int VP8EncoderImpl::Release() {
>-     tl0_pic_idx_[i] = temporal_layers_[i]->Tl0PicIdx();
>-   }
>-   temporal_layers_.clear();
>-+  temporal_layers_checkers_.clear();
>-   inited_ = false;
>-   return ret_val;
>- }
>-@@ -281,8 +322,7 @@ const char* VP8EncoderImpl::ImplementationName() const {
>-   return "libvpx";
>- }
>- 
>--void VP8EncoderImpl::SetStreamState(bool send_stream,
>--                                            int stream_idx) {
>-+void VP8EncoderImpl::SetStreamState(bool send_stream, int stream_idx) {
>-   if (send_stream && !send_stream_[stream_idx]) {
>-     // Need a key frame if we have not sent this stream before.
>-     key_frame_request_[stream_idx] = true;
>-@@ -298,6 +338,8 @@ void VP8EncoderImpl::SetupTemporalLayers(int num_streams,
>-   if (num_streams == 1) {
>-     temporal_layers_.emplace_back(
>-         tl_factory->Create(0, num_temporal_layers, tl0_pic_idx_[0]));
>-+    temporal_layers_checkers_.emplace_back(
>-+        tl_factory->CreateChecker(0, num_temporal_layers, tl0_pic_idx_[0]));
>-   } else {
>-     for (int i = 0; i < num_streams; ++i) {
>-       RTC_CHECK_GT(num_temporal_layers, 0);
>-@@ -305,6 +347,8 @@ void VP8EncoderImpl::SetupTemporalLayers(int num_streams,
>-                             codec.simulcastStream[i].numberOfTemporalLayers);
>-       temporal_layers_.emplace_back(
>-           tl_factory->Create(i, layers, tl0_pic_idx_[i]));
>-+      temporal_layers_checkers_.emplace_back(
>-+          tl_factory->CreateChecker(i, layers, tl0_pic_idx_[i]));
>-     }
>-   }
>- }
>-@@ -339,8 +383,10 @@ int VP8EncoderImpl::InitEncode(const VideoCodec* inst,
>-   int number_of_streams = NumberOfStreams(*inst);
>-   bool doing_simulcast = (number_of_streams > 1);
>- 
>--  if (doing_simulcast && !ValidSimulcastResolutions(*inst, number_of_streams)) {
>--    return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
>-+  if (doing_simulcast &&
>-+      (!ValidSimulcastResolutions(*inst, number_of_streams) ||
>-+       !ValidSimulcastTemporalLayers(*inst, number_of_streams))) {
>-+    return WEBRTC_VIDEO_CODEC_ERR_SIMULCAST_PARAMETERS_NOT_SUPPORTED;
>-   }
>- 
>-   int num_temporal_layers =
>-@@ -524,7 +570,8 @@ int VP8EncoderImpl::InitEncode(const VideoCodec* inst,
>- }
>- 
>- int VP8EncoderImpl::SetCpuSpeed(int width, int height) {
>--#if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64) || defined(ANDROID)
>-+#if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64) \
>-+  || defined(WEBRTC_ANDROID)
>-   // On mobile platform, use a lower speed setting for lower resolutions for
>-   // CPUs with 4 or more cores.
>-   RTC_DCHECK_GT(number_of_cores_, 0);
>-@@ -549,7 +596,7 @@ int VP8EncoderImpl::SetCpuSpeed(int width, int height) {
>- }
>- 
>- int VP8EncoderImpl::NumberOfThreads(int width, int height, int cpus) {
>--#if defined(ANDROID)
>-+#if defined(WEBRTC_ANDROID)
>-   if (width * height >= 320 * 180) {
>-     if (cpus >= 4) {
>-       // 3 threads for CPUs with 4 and more cores since most of times only 4
>-@@ -603,7 +650,8 @@ int VP8EncoderImpl::InitAndSetControlSettings() {
>-   // when encoding lower resolution streams. Would it work with the
>-   // multi-res encoding feature?
>-   denoiserState denoiser_state = kDenoiserOnYOnly;
>--#if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64) || defined(ANDROID)
>-+#if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64) \
>-+  || defined(WEBRTC_ANDROID)
>-   denoiser_state = kDenoiserOnYOnly;
>- #else
>-   denoiser_state = kDenoiserOnAdaptive;
>-@@ -703,17 +751,6 @@ int VP8EncoderImpl::Encode(const VideoFrame& frame,
>-         raw_images_[i].stride[VPX_PLANE_V], raw_images_[i].d_w,
>-         raw_images_[i].d_h, libyuv::kFilterBilinear);
>-   }
>--  vpx_enc_frame_flags_t flags[kMaxSimulcastStreams];
>--  TemporalLayers::FrameConfig tl_configs[kMaxSimulcastStreams];
>--  for (size_t i = 0; i < encoders_.size(); ++i) {
>--    tl_configs[i] = temporal_layers_[i]->UpdateLayerConfig(frame.timestamp());
>--
>--    if (tl_configs[i].drop_frame) {
>--      // Drop this frame.
>--      return WEBRTC_VIDEO_CODEC_OK;
>--    }
>--    flags[i] = EncodeFlags(tl_configs[i]);
>--  }
>-   bool send_key_frame = false;
>-   for (size_t i = 0; i < key_frame_request_.size() && i < send_stream_.size();
>-        ++i) {
>-@@ -731,6 +768,18 @@ int VP8EncoderImpl::Encode(const VideoFrame& frame,
>-       }
>-     }
>-   }
>-+  vpx_enc_frame_flags_t flags[kMaxSimulcastStreams];
>-+  TemporalLayers::FrameConfig tl_configs[kMaxSimulcastStreams];
>-+  for (size_t i = 0; i < encoders_.size(); ++i) {
>-+    tl_configs[i] = temporal_layers_[i]->UpdateLayerConfig(frame.timestamp());
>-+    RTC_DCHECK(temporal_layers_checkers_[i]->CheckTemporalConfig(
>-+        send_key_frame, tl_configs[i]));
>-+    if (tl_configs[i].drop_frame) {
>-+      // Drop this frame.
>-+      return WEBRTC_VIDEO_CODEC_OK;
>-+    }
>-+    flags[i] = EncodeFlags(tl_configs[i]);
>-+  }
>-   if (send_key_frame) {
>-     // Adapt the size of the key frame when in screenshare with 1 temporal
>-     // layer.
>-@@ -765,9 +814,8 @@ int VP8EncoderImpl::Encode(const VideoFrame& frame,
>-     }
>- 
>-     vpx_codec_control(&encoders_[i], VP8E_SET_FRAME_FLAGS, flags[stream_idx]);
>--    vpx_codec_control(
>--        &encoders_[i], VP8E_SET_TEMPORAL_LAYER_ID,
>--        temporal_layers_[stream_idx]->GetTemporalLayerId(tl_configs[i]));
>-+    vpx_codec_control(&encoders_[i], VP8E_SET_TEMPORAL_LAYER_ID,
>-+                      tl_configs[i].encoder_layer_id);
>-   }
>-   // TODO(holmer): Ideally the duration should be the timestamp diff of this
>-   // frame and the next frame to be encoded, which we don't have. Instead we
>-@@ -777,20 +825,31 @@ int VP8EncoderImpl::Encode(const VideoFrame& frame,
>-   assert(codec_.maxFramerate > 0);
>-   uint32_t duration = 90000 / codec_.maxFramerate;
>- 
>--  // Note we must pass 0 for |flags| field in encode call below since they are
>--  // set above in |vpx_codec_control| function for each encoder/spatial layer.
>--  int error = vpx_codec_encode(&encoders_[0], &raw_images_[0], timestamp_,
>--                               duration, 0, VPX_DL_REALTIME);
>--  // Reset specific intra frame thresholds, following the key frame.
>--  if (send_key_frame) {
>--    vpx_codec_control(&(encoders_[0]), VP8E_SET_MAX_INTRA_BITRATE_PCT,
>--                      rc_max_intra_target_);
>-+  int error = WEBRTC_VIDEO_CODEC_OK;
>-+  int num_tries = 0;
>-+  // If the first try returns WEBRTC_VIDEO_CODEC_TARGET_BITRATE_OVERSHOOT
>-+  // the frame must be reencoded with the same parameters again because
>-+  // target bitrate is exceeded and encoder state has been reset.
>-+  while (num_tries == 0 ||
>-+         (num_tries == 1 &&
>-+          error == WEBRTC_VIDEO_CODEC_TARGET_BITRATE_OVERSHOOT)) {
>-+    ++num_tries;
>-+    // Note we must pass 0 for |flags| field in encode call below since they are
>-+    // set above in |vpx_codec_control| function for each encoder/spatial layer.
>-+    error = vpx_codec_encode(&encoders_[0], &raw_images_[0], timestamp_,
>-+                             duration, 0, VPX_DL_REALTIME);
>-+    // Reset specific intra frame thresholds, following the key frame.
>-+    if (send_key_frame) {
>-+      vpx_codec_control(&(encoders_[0]), VP8E_SET_MAX_INTRA_BITRATE_PCT,
>-+                        rc_max_intra_target_);
>-+    }
>-+    if (error)
>-+      return WEBRTC_VIDEO_CODEC_ERROR;
>-+    timestamp_ += duration;
>-+    // Examines frame timestamps only.
>-+    error = GetEncodedPartitions(tl_configs, frame);
>-   }
>--  if (error)
>--    return WEBRTC_VIDEO_CODEC_ERROR;
>--  timestamp_ += duration;
>--  // Examines frame timestamps only.
>--  return GetEncodedPartitions(tl_configs, frame);
>-+  return error;
>- }
>- 
>- void VP8EncoderImpl::PopulateCodecSpecific(
>-@@ -878,7 +937,7 @@ int VP8EncoderImpl::GetEncodedPartitions(
>-     encoded_images_[encoder_idx].content_type_ =
>-         (codec_.mode == kScreensharing) ? VideoContentType::SCREENSHARE
>-                                         : VideoContentType::UNSPECIFIED;
>--    encoded_images_[encoder_idx].timing_.is_timing_frame = false;
>-+    encoded_images_[encoder_idx].timing_.flags = TimingFrameFlags::kInvalid;
>- 
>-     int qp = -1;
>-     vpx_codec_control(&encoders_[encoder_idx], VP8E_GET_LAST_QUANTIZER_64, &qp);
>-@@ -926,9 +985,33 @@ int VP8EncoderImpl::RegisterEncodeCompleteCallback(
>-   return WEBRTC_VIDEO_CODEC_OK;
>- }
>- 
>-+class VP8DecoderImpl::QpSmoother {
>-+ public:
>-+  QpSmoother() : last_sample_ms_(rtc::TimeMillis()), smoother_(kAlpha) {}
>-+
>-+  int GetAvg() const {
>-+    float value = smoother_.filtered();
>-+    return (value == rtc::ExpFilter::kValueUndefined) ? 0
>-+                                                      : static_cast<int>(value);
>-+  }
>-+
>-+  void Add(float sample) {
>-+    int64_t now_ms = rtc::TimeMillis();
>-+    smoother_.Apply(static_cast<float>(now_ms - last_sample_ms_), sample);
>-+    last_sample_ms_ = now_ms;
>-+  }
>-+
>-+  void Reset() { smoother_.Reset(kAlpha); }
>-+
>-+ private:
>-+  const float kAlpha = 0.95f;
>-+  int64_t last_sample_ms_;
>-+  rtc::ExpFilter smoother_;
>-+};
>-+
>- VP8DecoderImpl::VP8DecoderImpl()
>--    : use_postproc_arm_(webrtc::field_trial::FindFullName(
>--                            kVp8PostProcArmFieldTrial) == "Enabled"),
>-+    : use_postproc_arm_(
>-+          webrtc::field_trial::IsEnabled(kVp8PostProcArmFieldTrial)),
>-       buffer_pool_(false, 300 /* max_number_of_buffers*/),
>-       decode_complete_callback_(NULL),
>-       inited_(false),
>-@@ -936,7 +1019,11 @@ VP8DecoderImpl::VP8DecoderImpl()
>-       propagation_cnt_(-1),
>-       last_frame_width_(0),
>-       last_frame_height_(0),
>--      key_frame_required_(true) {}
>-+      key_frame_required_(true),
>-+      qp_smoother_(use_postproc_arm_ ? new QpSmoother() : nullptr) {
>-+  if (use_postproc_arm_)
>-+    GetPostProcParamsFromFieldTrialGroup(&deblock_);
>-+}
>- 
>- VP8DecoderImpl::~VP8DecoderImpl() {
>-   inited_ = true;  // in order to do the actual release
>-@@ -957,7 +1044,8 @@ int VP8DecoderImpl::InitDecode(const VideoCodec* inst, int number_of_cores) {
>-   cfg.threads = 1;
>-   cfg.h = cfg.w = 0;  // set after decode
>- 
>--#if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64) || defined(ANDROID)
>-+#if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64) \
>-+  || defined(WEBRTC_ANDROID)
>-   vpx_codec_flags_t flags = use_postproc_arm_ ? VPX_CODEC_USE_POSTPROC : 0;
>- #else
>-   vpx_codec_flags_t flags = VPX_CODEC_USE_POSTPROC;
>-@@ -996,16 +1084,28 @@ int VP8DecoderImpl::Decode(const EncodedImage& input_image,
>-   }
>- 
>- // Post process configurations.
>--#if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64) || defined(ANDROID)
>-+#if defined(WEBRTC_ARCH_ARM) || defined(WEBRTC_ARCH_ARM64) \
>-+  || defined(WEBRTC_ANDROID)
>-   if (use_postproc_arm_) {
>-     vp8_postproc_cfg_t ppcfg;
>-     ppcfg.post_proc_flag = VP8_MFQE;
>--    // For low resolutions, use stronger deblocking filter and enable the
>--    // deblock and demacroblocker.
>-+    // For low resolutions, use stronger deblocking filter.
>-     int last_width_x_height = last_frame_width_ * last_frame_height_;
>-     if (last_width_x_height > 0 && last_width_x_height <= 320 * 240) {
>--      ppcfg.deblocking_level = 6;  // Only affects VP8_DEMACROBLOCK.
>--      ppcfg.post_proc_flag |= VP8_DEBLOCK | VP8_DEMACROBLOCK;
>-+      // Enable the deblock and demacroblocker based on qp thresholds.
>-+      RTC_DCHECK(qp_smoother_);
>-+      int qp = qp_smoother_->GetAvg();
>-+      if (qp > deblock_.min_qp) {
>-+        int level = deblock_.max_level;
>-+        if (qp < deblock_.degrade_qp) {
>-+          // Use lower level.
>-+          level = deblock_.max_level * (qp - deblock_.min_qp) /
>-+                  (deblock_.degrade_qp - deblock_.min_qp);
>-+        }
>-+        // Deblocking level only affects VP8_DEMACROBLOCK.
>-+        ppcfg.deblocking_level = std::max(level, 1);
>-+        ppcfg.post_proc_flag |= VP8_DEBLOCK | VP8_DEMACROBLOCK;
>-+      }
>-     }
>-     vpx_codec_control(decoder_, VP8_SET_POSTPROC, &ppcfg);
>-   }
>-@@ -1105,6 +1205,13 @@ int VP8DecoderImpl::ReturnFrame(const vpx_image_t* img,
>-     // Decoder OK and NULL image => No show frame
>-     return WEBRTC_VIDEO_CODEC_NO_OUTPUT;
>-   }
>-+  if (qp_smoother_) {
>-+    if (last_frame_width_ != static_cast<int>(img->d_w) ||
>-+        last_frame_height_ != static_cast<int>(img->d_h)) {
>-+      qp_smoother_->Reset();
>-+    }
>-+    qp_smoother_->Add(qp);
>-+  }
>-   last_frame_width_ = img->d_w;
>-   last_frame_height_ = img->d_h;
>-   // Allocate memory for decoded image.
>-@@ -1122,13 +1229,12 @@ int VP8DecoderImpl::ReturnFrame(const vpx_image_t* img,
>-                    img->planes[VPX_PLANE_V], img->stride[VPX_PLANE_V],
>-                    buffer->MutableDataY(), buffer->StrideY(),
>-                    buffer->MutableDataU(), buffer->StrideU(),
>--                   buffer->MutableDataV(), buffer->StrideV(),
>--                   img->d_w, img->d_h);
>-+                   buffer->MutableDataV(), buffer->StrideV(), img->d_w,
>-+                   img->d_h);
>- 
>-   VideoFrame decoded_image(buffer, timestamp, 0, kVideoRotation_0);
>-   decoded_image.set_ntp_time_ms(ntp_time_ms);
>--  decode_complete_callback_->Decoded(decoded_image, rtc::Optional<int32_t>(),
>--                                     rtc::Optional<uint8_t>(qp));
>-+  decode_complete_callback_->Decoded(decoded_image, rtc::nullopt, qp);
>- 
>-   return WEBRTC_VIDEO_CODEC_OK;
>- }
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/vp8_impl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/vp8_impl.h
>-index 376dde12808..e5d5d81e35c 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/vp8_impl.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/vp8_impl.h
>-@@ -10,26 +10,26 @@
>-  * WEBRTC VP8 wrapper interface
>-  */
>- 
>--#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_VP8_IMPL_H_
>--#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_VP8_IMPL_H_
>-+#ifndef MODULES_VIDEO_CODING_CODECS_VP8_VP8_IMPL_H_
>-+#define MODULES_VIDEO_CODING_CODECS_VP8_VP8_IMPL_H_
>- 
>- #include <memory>
>- #include <vector>
>- 
>- // NOTE: This include order must remain to avoid compile errors, even though
>- //       it breaks the style guide.
>--#include "vpx/vpx_encoder.h"
>--#include "vpx/vpx_decoder.h"
>- #include "vpx/vp8cx.h"
>- #include "vpx/vp8dx.h"
>-+#include "vpx/vpx_decoder.h"
>-+#include "vpx/vpx_encoder.h"
>- 
>--#include "webrtc/api/video/video_frame.h"
>--#include "webrtc/common_video/include/i420_buffer_pool.h"
>--#include "webrtc/common_video/include/video_frame.h"
>--#include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
>--#include "webrtc/modules/video_coding/codecs/vp8/temporal_layers.h"
>--#include "webrtc/modules/video_coding/include/video_codec_interface.h"
>--#include "webrtc/modules/video_coding/utility/quality_scaler.h"
>-+#include "api/video/video_frame.h"
>-+#include "common_video/include/i420_buffer_pool.h"
>-+#include "common_video/include/video_frame.h"
>-+#include "modules/video_coding/codecs/vp8/include/vp8.h"
>-+#include "modules/video_coding/codecs/vp8/temporal_layers.h"
>-+#include "modules/video_coding/include/video_codec_interface.h"
>-+#include "modules/video_coding/utility/quality_scaler.h"
>- 
>- namespace webrtc {
>- 
>-@@ -104,6 +104,7 @@ class VP8EncoderImpl : public VP8Encoder {
>-   int number_of_cores_;
>-   uint32_t rc_max_intra_target_;
>-   std::vector<std::unique_ptr<TemporalLayers>> temporal_layers_;
>-+  std::vector<std::unique_ptr<TemporalLayersChecker>> temporal_layers_checkers_;
>-   std::vector<uint16_t> picture_id_;
>-   std::vector<uint8_t> tl0_pic_idx_;
>-   std::vector<bool> key_frame_request_;
>-@@ -135,7 +136,14 @@ class VP8DecoderImpl : public VP8Decoder {
>- 
>-   const char* ImplementationName() const override;
>- 
>-+  struct DeblockParams {
>-+    int max_level = 6;   // Deblocking strength: [0, 16].
>-+    int degrade_qp = 1;  // If QP value is below, start lowering |max_level|.
>-+    int min_qp = 0;      // If QP value is below, turn off deblocking.
>-+  };
>-+
>-  private:
>-+  class QpSmoother;
>-   int ReturnFrame(const vpx_image_t* img,
>-                   uint32_t timeStamp,
>-                   int64_t ntp_time_ms,
>-@@ -151,7 +159,9 @@ class VP8DecoderImpl : public VP8Decoder {
>-   int last_frame_width_;
>-   int last_frame_height_;
>-   bool key_frame_required_;
>-+  DeblockParams deblock_;
>-+  const std::unique_ptr<QpSmoother> qp_smoother_;
>- };
>- }  // namespace webrtc
>- 
>--#endif  // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_VP8_IMPL_H_
>-+#endif  // MODULES_VIDEO_CODING_CODECS_VP8_VP8_IMPL_H_
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/vp8_noop.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/vp8_noop.cc
>-index a61c3655190..3267344204c 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/vp8_noop.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/codecs/vp8/vp8_noop.cc
>-@@ -13,8 +13,8 @@
>- #error
>- #endif  // !defined(RTC_DISABLE_VP8)
>- 
>--#include "webrtc/base/checks.h"
>--#include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
>-+#include "rtc_base/checks.h"
>-+#include "modules/video_coding/codecs/vp8/include/vp8.h"
>- 
>- namespace webrtc {
>- 
>-@@ -22,7 +22,7 @@ bool VP8Encoder::IsSupported() {
>-   return false;
>- }
>- 
>--VP8Encoder* VP8Encoder::Create() {
>-+std::unique_ptr<VP8Encoder> VP8Encoder::Create() {
>-   RTC_NOTREACHED();
>-   return nullptr;
>- }
>-@@ -31,7 +31,7 @@ bool VP8Decoder::IsSupported() {
>-   return false;
>- }
>- 
>--VP8Decoder* VP8Decoder::Create() {
>-+std::unique_ptr<VP8Decoder> VP8Decoder::Create() {
>-   RTC_NOTREACHED();
>-   return nullptr;
>- }
>--- 
>-2.13.5 (Apple Git-94)
>-
>diff --git a/Source/ThirdParty/libwebrtc/WebKit/0004-Removing-parameter-names-from-files-included-from-We.patch b/Source/ThirdParty/libwebrtc/WebKit/0004-Removing-parameter-names-from-files-included-from-We.patch
>deleted file mode 100644
>index 359833fc7ee..00000000000
>--- a/Source/ThirdParty/libwebrtc/WebKit/0004-Removing-parameter-names-from-files-included-from-We.patch
>+++ /dev/null
>@@ -1,682 +0,0 @@
>-From f848b7acea46bb74d5a41b9da5b2e48b31203a30 Mon Sep 17 00:00:00 2001
>-From: Youenn Fablet <youenn@apple.com>
>-Date: Thu, 14 Dec 2017 13:50:22 -0800
>-Subject: [PATCH 4/8] Removing parameter names from files included from WebKit
>-
>----
>- .../libwebrtc/Source/webrtc/api/array_view.h       |  4 +-
>- .../Source/webrtc/api/datachannelinterface.h       |  2 +-
>- .../ThirdParty/libwebrtc/Source/webrtc/api/jsep.h  |  2 +-
>- .../Source/webrtc/api/mediastreaminterface.h       | 20 ++---
>- .../Source/webrtc/api/peerconnectioninterface.h    | 87 +++++++++++-----------
>- .../libwebrtc/Source/webrtc/api/umametrics.h       |  6 +-
>- .../Source/webrtc/api/video_codecs/video_decoder.h |  8 +-
>- .../Source/webrtc/api/video_codecs/video_encoder.h |  2 +-
>- .../webrtc/common_video/include/frame_callback.h   |  2 +-
>- .../webrtc/logging/rtc_event_log/rtc_event_log.h   |  8 +-
>- .../Source/webrtc/media/base/mediachannel.h        |  6 +-
>- .../modules/audio_device/include/audio_device.h    | 16 ++--
>- .../Source/webrtc/p2p/base/icetransportinternal.h  |  2 +-
>- .../libwebrtc/Source/webrtc/p2p/base/port.h        |  2 +-
>- .../libwebrtc/Source/webrtc/p2p/base/stun.h        |  2 +-
>- .../libwebrtc/Source/webrtc/p2p/base/stunrequest.h |  6 +-
>- .../Source/webrtc/rtc_base/messagehandler.h        |  4 +-
>- .../libwebrtc/Source/webrtc/rtc_base/network.h     |  2 +-
>- .../rtc_base/numerics/safe_conversions_impl.h      |  2 +-
>- .../Source/webrtc/rtc_base/socketserver.h          |  2 +-
>- .../libwebrtc/Source/webrtc/rtc_base/stream.h      |  4 +-
>- 21 files changed, 94 insertions(+), 95 deletions(-)
>-
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/array_view.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/array_view.h
>-index d951d0f02d3..b27bd29a8e2 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/array_view.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/array_view.h
>-@@ -94,7 +94,7 @@ class ArrayViewBase {
>-   static_assert(Size > 0, "ArrayView size must be variable or non-negative");
>- 
>-  public:
>--  ArrayViewBase(T* data, size_t size) : data_(data) {}
>-+  ArrayViewBase(T* data, size_t) : data_(data) {}
>- 
>-   static constexpr size_t size() { return Size; }
>-   static constexpr bool empty() { return false; }
>-@@ -111,7 +111,7 @@ class ArrayViewBase {
>- template <typename T>
>- class ArrayViewBase<T, 0> {
>-  public:
>--  explicit ArrayViewBase(T* data, size_t size) {}
>-+  explicit ArrayViewBase(T*, size_t) {}
>- 
>-   static constexpr size_t size() { return 0; }
>-   static constexpr bool empty() { return true; }
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/datachannelinterface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/datachannelinterface.h
>-index 4ab7efb7407..4b5ce9a7333 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/datachannelinterface.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/datachannelinterface.h
>-@@ -90,7 +90,7 @@ class DataChannelObserver {
>-   //  A data buffer was successfully received.
>-   virtual void OnMessage(const DataBuffer& buffer) = 0;
>-   // The data channel's buffered_amount has changed.
>--  virtual void OnBufferedAmountChange(uint64_t previous_amount) {}
>-+  virtual void OnBufferedAmountChange(uint64_t) {}
>- 
>-  protected:
>-   virtual ~DataChannelObserver() {}
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/jsep.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/jsep.h
>-index acfffd4d0a7..59064d85758 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/jsep.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/jsep.h
>-@@ -126,7 +126,7 @@ class SessionDescriptionInterface {
>-   //
>-   // Returns the number of candidates removed.
>-   virtual size_t RemoveCandidates(
>--      const std::vector<cricket::Candidate>& candidates) { return 0; }
>-+      const std::vector<cricket::Candidate>&) { return 0; }
>- 
>-   // Returns the number of m= sections in the session description.
>-   virtual size_t number_of_mediasections() const = 0;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediastreaminterface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediastreaminterface.h
>-index 80595e09abe..aae5a032937 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediastreaminterface.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediastreaminterface.h
>-@@ -167,14 +167,14 @@ class VideoTrackInterface
>- 
>-   // Register a video sink for this track. Used to connect the track to the
>-   // underlying video engine.
>--  void AddOrUpdateSink(rtc::VideoSinkInterface<VideoFrame>* sink,
>--                       const rtc::VideoSinkWants& wants) override {}
>--  void RemoveSink(rtc::VideoSinkInterface<VideoFrame>* sink) override {}
>-+  void AddOrUpdateSink(rtc::VideoSinkInterface<VideoFrame>*,
>-+                       const rtc::VideoSinkWants&) override {}
>-+  void RemoveSink(rtc::VideoSinkInterface<VideoFrame>*) override {}
>- 
>-   virtual VideoTrackSourceInterface* GetSource() const = 0;
>- 
>-   virtual ContentHint content_hint() const { return ContentHint::kNone; }
>--  virtual void set_content_hint(ContentHint hint) {}
>-+  virtual void set_content_hint(ContentHint) {}
>- 
>-  protected:
>-   virtual ~VideoTrackInterface() {}
>-@@ -211,15 +211,15 @@ class AudioSourceInterface : public MediaSourceInterface {
>-   // Sets the volume of the source. |volume| is in  the range of [0, 10].
>-   // TODO(tommi): This method should be on the track and ideally volume should
>-   // be applied in the track in a way that does not affect clones of the track.
>--  virtual void SetVolume(double volume) {}
>-+  virtual void SetVolume(double) {}
>- 
>-   // Registers/unregisters observers to the audio source.
>--  virtual void RegisterAudioObserver(AudioObserver* observer) {}
>--  virtual void UnregisterAudioObserver(AudioObserver* observer) {}
>-+  virtual void RegisterAudioObserver(AudioObserver*) {}
>-+  virtual void UnregisterAudioObserver(AudioObserver*) {}
>- 
>-   // TODO(tommi): Make pure virtual.
>--  virtual void AddSink(AudioTrackSinkInterface* sink) {}
>--  virtual void RemoveSink(AudioTrackSinkInterface* sink) {}
>-+  virtual void AddSink(AudioTrackSinkInterface*) {}
>-+  virtual void RemoveSink(AudioTrackSinkInterface*) {}
>- };
>- 
>- // Interface of the audio processor used by the audio track to collect
>-@@ -288,7 +288,7 @@ class AudioTrackInterface : public MediaStreamTrackInterface {
>-   // Return true on success, otherwise false.
>-   // TODO(deadbeef): Change the interface to int GetSignalLevel() and pure
>-   // virtual after it's implemented in chromium.
>--  virtual bool GetSignalLevel(int* level) { return false; }
>-+  virtual bool GetSignalLevel(int*) { return false; }
>- 
>-   // Get the audio processor used by the audio track. Return null if the track
>-   // does not have any processor.
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/peerconnectioninterface.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/peerconnectioninterface.h
>-index a1e7d7ee2fb..76f7b28945c 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/peerconnectioninterface.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/peerconnectioninterface.h
>-@@ -629,12 +629,12 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>-   // Errors:
>-   // - INVALID_PARAMETER: |track| is null.
>-   virtual RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
>--  AddTransceiver(rtc::scoped_refptr<MediaStreamTrackInterface> track) {
>-+  AddTransceiver(rtc::scoped_refptr<MediaStreamTrackInterface>) {
>-     return RTCError(RTCErrorType::INTERNAL_ERROR, "not implemented");
>-   }
>-   virtual RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
>--  AddTransceiver(rtc::scoped_refptr<MediaStreamTrackInterface> track,
>--                 const RtpTransceiverInit& init) {
>-+  AddTransceiver(rtc::scoped_refptr<MediaStreamTrackInterface>,
>-+                 const RtpTransceiverInit&) {
>-     return RTCError(RTCErrorType::INTERNAL_ERROR, "not implemented");
>-   }
>- 
>-@@ -644,12 +644,12 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>-   // - INVALID_PARAMETER: |media_type| is not MEDIA_TYPE_AUDIO or
>-   //                      MEDIA_TYPE_VIDEO.
>-   virtual RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
>--  AddTransceiver(cricket::MediaType media_type) {
>-+  AddTransceiver(cricket::MediaType) {
>-     return RTCError(RTCErrorType::INTERNAL_ERROR, "not implemented");
>-   }
>-   virtual RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
>--  AddTransceiver(cricket::MediaType media_type,
>--                 const RtpTransceiverInit& init) {
>-+  AddTransceiver(cricket::MediaType,
>-+                 const RtpTransceiverInit&) {
>-     return RTCError(RTCErrorType::INTERNAL_ERROR, "not implemented");
>-   }
>- 
>-@@ -675,8 +675,8 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>-   // |stream_id| is used to populate the msid attribute; if empty, one will
>-   // be generated automatically.
>-   virtual rtc::scoped_refptr<RtpSenderInterface> CreateSender(
>--      const std::string& kind,
>--      const std::string& stream_id) {
>-+      const std::string&,
>-+      const std::string&) {
>-     return rtc::scoped_refptr<RtpSenderInterface>();
>-   }
>- 
>-@@ -719,7 +719,7 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>-   // TODO(hbos): Default implementation that does nothing only exists as to not
>-   // break third party projects. As soon as they have been updated this should
>-   // be changed to "= 0;".
>--  virtual void GetStats(RTCStatsCollectorCallback* callback) {}
>-+  virtual void GetStats(RTCStatsCollectorCallback*) {}
>- 
>-   // Create a data channel with the provided config, or default config if none
>-   // is provided. Note that an offer/answer negotiation is still necessary
>-@@ -760,23 +760,23 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>- 
>-   // Create a new offer.
>-   // The CreateSessionDescriptionObserver callback will be called when done.
>--  virtual void CreateOffer(CreateSessionDescriptionObserver* observer,
>--                           const MediaConstraintsInterface* constraints) {}
>-+  virtual void CreateOffer(CreateSessionDescriptionObserver*,
>-+                           const MediaConstraintsInterface*) {}
>- 
>-   // TODO(jiayl): remove the default impl and the old interface when chromium
>-   // code is updated.
>--  virtual void CreateOffer(CreateSessionDescriptionObserver* observer,
>--                           const RTCOfferAnswerOptions& options) {}
>-+  virtual void CreateOffer(CreateSessionDescriptionObserver*,
>-+                           const RTCOfferAnswerOptions&) {}
>- 
>-   // Create an answer to an offer.
>-   // The CreateSessionDescriptionObserver callback will be called when done.
>--  virtual void CreateAnswer(CreateSessionDescriptionObserver* observer,
>--                            const RTCOfferAnswerOptions& options) {}
>-+  virtual void CreateAnswer(CreateSessionDescriptionObserver*,
>-+                            const RTCOfferAnswerOptions&) {}
>-   // Deprecated - use version above.
>-   // TODO(hta): Remove and remove default implementations when all callers
>-   // are updated.
>--  virtual void CreateAnswer(CreateSessionDescriptionObserver* observer,
>--                            const MediaConstraintsInterface* constraints) {}
>-+  virtual void CreateAnswer(CreateSessionDescriptionObserver*,
>-+                            const MediaConstraintsInterface*) {}
>- 
>-   // Sets the local session description.
>-   // The PeerConnection takes the ownership of |desc| even if it fails.
>-@@ -789,19 +789,19 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>-   // The PeerConnection takes the ownership of |desc| even if it fails.
>-   // The |observer| callback will be called when done.
>-   // TODO(hbos): Remove when Chrome implements the new signature.
>--  virtual void SetRemoteDescription(SetSessionDescriptionObserver* observer,
>--                                    SessionDescriptionInterface* desc) {}
>-+  virtual void SetRemoteDescription(SetSessionDescriptionObserver*,
>-+                                    SessionDescriptionInterface*) {}
>-   // TODO(hbos): Make pure virtual when Chrome has updated its signature.
>-   virtual void SetRemoteDescription(
>--      std::unique_ptr<SessionDescriptionInterface> desc,
>--      rtc::scoped_refptr<SetRemoteDescriptionObserverInterface> observer) {}
>-+      std::unique_ptr<SessionDescriptionInterface>,
>-+      rtc::scoped_refptr<SetRemoteDescriptionObserverInterface>) {}
>-   // Deprecated; Replaced by SetConfiguration.
>-   // TODO(deadbeef): Remove once Chrome is moved over to SetConfiguration.
>--  virtual bool UpdateIce(const IceServers& configuration,
>--                         const MediaConstraintsInterface* constraints) {
>-+  virtual bool UpdateIce(const IceServers&,
>-+                         const MediaConstraintsInterface*) {
>-     return false;
>-   }
>--  virtual bool UpdateIce(const IceServers& configuration) { return false; }
>-+  virtual bool UpdateIce(const IceServers&) { return false; }
>- 
>-   // TODO(deadbeef): Make this pure virtual once all Chrome subclasses of
>-   // PeerConnectionInterface implement it.
>-@@ -833,14 +833,14 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>-   // TODO(deadbeef): Make this pure virtual once all Chrome subclasses of
>-   // PeerConnectionInterface implement it.
>-   virtual bool SetConfiguration(
>--      const PeerConnectionInterface::RTCConfiguration& config,
>--      RTCError* error) {
>-+      const PeerConnectionInterface::RTCConfiguration&,
>-+      RTCError*) {
>-     return false;
>-   }
>-   // Version without error output param for backwards compatibility.
>-   // TODO(deadbeef): Remove once chromium is updated.
>-   virtual bool SetConfiguration(
>--      const PeerConnectionInterface::RTCConfiguration& config) {
>-+      const PeerConnectionInterface::RTCConfiguration&) {
>-     return false;
>-   }
>- 
>-@@ -854,7 +854,7 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>-   // continual gathering, to avoid an ever-growing list of candidates as
>-   // networks come and go.
>-   virtual bool RemoveIceCandidates(
>--      const std::vector<cricket::Candidate>& candidates) {
>-+      const std::vector<cricket::Candidate>&) {
>-     return false;
>-   }
>- 
>-@@ -884,8 +884,7 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>-   // ownership is passed with std::unique_ptr
>-   // TODO(alexnarest): Make this pure virtual when tests will be updated
>-   virtual void SetBitrateAllocationStrategy(
>--      std::unique_ptr<rtc::BitrateAllocationStrategy>
>--          bitrate_allocation_strategy) {}
>-+      std::unique_ptr<rtc::BitrateAllocationStrategy>) {}
>- 
>-   // Enable/disable playout of received audio streams. Enabled by default. Note
>-   // that even if playout is enabled, streams will only be played out if the
>-@@ -894,13 +893,13 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>-   // for audio data every 10ms to ensure that audio processing happens and the
>-   // audio statistics are updated.
>-   // TODO(henrika): deprecate and remove this.
>--  virtual void SetAudioPlayout(bool playout) {}
>-+  virtual void SetAudioPlayout(bool) {}
>- 
>-   // Enable/disable recording of transmitted audio streams. Enabled by default.
>-   // Note that even if recording is enabled, streams will only be recorded if
>-   // the appropriate SDP is also applied.
>-   // TODO(henrika): deprecate and remove this.
>--  virtual void SetAudioRecording(bool recording) {}
>-+  virtual void SetAudioRecording(bool) {}
>- 
>-   // Returns the current SignalingState.
>-   virtual SignalingState signaling_state() = 0;
>-@@ -919,8 +918,8 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>-   // automatically after 10 minutes have passed, or when the StopRtcEventLog
>-   // function is called.
>-   // TODO(eladalon): Deprecate and remove this.
>--  virtual bool StartRtcEventLog(rtc::PlatformFile file,
>--                                int64_t max_size_bytes) {
>-+  virtual bool StartRtcEventLog(rtc::PlatformFile,
>-+                                int64_t) {
>-     return false;
>-   }
>- 
>-@@ -928,8 +927,8 @@ class PeerConnectionInterface : public rtc::RefCountInterface {
>-   // |output| and passes it on to Call, which will take the ownership. If the
>-   // operation fails the output will be closed and deallocated. The event log
>-   // will send serialized events to the output object every |output_period_ms|.
>--  virtual bool StartRtcEventLog(std::unique_ptr<RtcEventLogOutput> output,
>--                                int64_t output_period_ms) {
>-+  virtual bool StartRtcEventLog(std::unique_ptr<RtcEventLogOutput>,
>-+                                int64_t) {
>-     return false;
>-   }
>- 
>-@@ -1002,16 +1001,16 @@ class PeerConnectionObserver {
>-   // TODO(honghaiz): Make this a pure virtual method when all its subclasses
>-   // implement it.
>-   virtual void OnIceCandidatesRemoved(
>--      const std::vector<cricket::Candidate>& candidates) {}
>-+      const std::vector<cricket::Candidate>&) {}
>- 
>-   // Called when the ICE connection receiving status changes.
>--  virtual void OnIceConnectionReceivingChange(bool receiving) {}
>-+  virtual void OnIceConnectionReceivingChange(bool) {}
>- 
>-   // This is called when a receiver and its track is created.
>-   // TODO(zhihuang): Make this pure virtual when all subclasses implement it.
>-   virtual void OnAddTrack(
>--      rtc::scoped_refptr<RtpReceiverInterface> receiver,
>--      const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& streams) {}
>-+      rtc::scoped_refptr<RtpReceiverInterface>,
>-+      const std::vector<rtc::scoped_refptr<MediaStreamInterface>>&) {}
>- 
>-   // TODO(hbos,deadbeef): Add |OnAssociatedStreamsUpdated| with |receiver| and
>-   // |streams| as arguments. This should be called when an existing receiver its
>-@@ -1030,7 +1029,7 @@ class PeerConnectionObserver {
>-   // no longer removed, deprecate and remove this callback.
>-   // TODO(hbos,deadbeef): Make pure virtual when all subclasses implement it.
>-   virtual void OnRemoveTrack(
>--      rtc::scoped_refptr<RtpReceiverInterface> receiver) {}
>-+      rtc::scoped_refptr<RtpReceiverInterface>) {}
>- 
>-  protected:
>-   // Dtor protected as objects shouldn't be deleted via this interface.
>-@@ -1133,7 +1132,7 @@ class PeerConnectionFactoryInterface : public rtc::RefCountInterface {
>-   // TODO(deadbeef): Make pure virtual once downstream mock PC factory classes
>-   // are updated.
>-   virtual rtc::scoped_refptr<VideoTrackSourceInterface> CreateVideoSource(
>--      std::unique_ptr<cricket::VideoCapturer> capturer) {
>-+      std::unique_ptr<cricket::VideoCapturer>) {
>-     return nullptr;
>-   }
>- 
>-@@ -1144,8 +1143,8 @@ class PeerConnectionFactoryInterface : public rtc::RefCountInterface {
>-   // |constraints| is only used for the invocation of this method, and can
>-   // safely be destroyed afterwards.
>-   virtual rtc::scoped_refptr<VideoTrackSourceInterface> CreateVideoSource(
>--      std::unique_ptr<cricket::VideoCapturer> capturer,
>--      const MediaConstraintsInterface* constraints) {
>-+      std::unique_ptr<cricket::VideoCapturer>,
>-+      const MediaConstraintsInterface*) {
>-     return nullptr;
>-   }
>- 
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/umametrics.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/umametrics.h
>-index c5125985233..0d66a5e51fe 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/umametrics.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/umametrics.h
>-@@ -116,9 +116,9 @@ class MetricsObserverInterface : public rtc::RefCountInterface {
>-   // |type| is the type of the enum counter to be incremented. |counter|
>-   // is the particular counter in that type. |counter_max| is the next sequence
>-   // number after the highest counter.
>--  virtual void IncrementEnumCounter(PeerConnectionEnumCounterType type,
>--                                    int counter,
>--                                    int counter_max) {}
>-+  virtual void IncrementEnumCounter(PeerConnectionEnumCounterType,
>-+                                    int,
>-+                                    int) {}
>- 
>-   // This is used to handle sparse counters like SSL cipher suites.
>-   // TODO(guoweis): Remove the implementation once the dependency's interface
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_decoder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_decoder.h
>-index 5897901346c..7d3a7f59781 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_decoder.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_decoder.h
>-@@ -36,7 +36,7 @@ class DecodedImageCallback {
>-   // decode time excluding waiting time for any previous pending frame to
>-   // return. This is necessary for breaking positive feedback in the delay
>-   // estimation when the decoder has a single output buffer.
>--  virtual int32_t Decoded(VideoFrame& decodedImage, int64_t decode_time_ms) {
>-+  virtual int32_t Decoded(VideoFrame& decodedImage, int64_t /* decode_time_ms */) {
>-     // The default implementation ignores custom decode time value.
>-     return Decoded(decodedImage);
>-   }
>-@@ -44,16 +44,16 @@ class DecodedImageCallback {
>-   // updated.
>-   virtual void Decoded(VideoFrame& decodedImage,
>-                        rtc::Optional<int32_t> decode_time_ms,
>--                       rtc::Optional<uint8_t> qp) {
>-+                       rtc::Optional<uint8_t> /* qp */) {
>-     Decoded(decodedImage,
>-             decode_time_ms ? static_cast<int32_t>(*decode_time_ms) : -1);
>-   }
>- 
>--  virtual int32_t ReceivedDecodedReferenceFrame(const uint64_t pictureId) {
>-+  virtual int32_t ReceivedDecodedReferenceFrame(const uint64_t /* pictureId */) {
>-     return -1;
>-   }
>- 
>--  virtual int32_t ReceivedDecodedFrame(const uint64_t pictureId) { return -1; }
>-+  virtual int32_t ReceivedDecodedFrame(const uint64_t /* pictureId */) { return -1; }
>- };
>- 
>- class VideoDecoder {
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_encoder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_encoder.h
>-index eb401afded9..64e33069099 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_encoder.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/video_codecs/video_encoder.h
>-@@ -74,7 +74,7 @@ class EncodedImageCallback {
>-   // Deprecated. TODO(ilnik): Remove this in few weeks.
>-   virtual void OnDroppedFrame() {}
>- 
>--  virtual void OnDroppedFrame(DropReason reason) {}
>-+  virtual void OnDroppedFrame(DropReason) {}
>- };
>- 
>- class VideoEncoder {
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/frame_callback.h b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/frame_callback.h
>-index 21019652870..0ab06a9a026 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/frame_callback.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/include/frame_callback.h
>-@@ -49,7 +49,7 @@ struct EncodedFrame {
>- class EncodedFrameObserver {
>-  public:
>-   virtual void EncodedFrameCallback(const EncodedFrame& encoded_frame) = 0;
>--  virtual void OnEncodeTiming(int64_t capture_ntp_ms, int encode_duration_ms) {}
>-+  virtual void OnEncodeTiming(int64_t, int) {}
>- 
>-  protected:
>-   virtual ~EncodedFrameObserver() {}
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log.h b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log.h
>-index 89b4fc1a086..89ace2c9e95 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/logging/rtc_event_log/rtc_event_log.h
>-@@ -41,7 +41,7 @@ class RtcEventLog {
>-   // TODO(nisse): webrtc::Clock is deprecated. Delete this method and
>-   // above forward declaration of Clock when
>-   // webrtc/system_wrappers/include/clock.h is deleted.
>--  static std::unique_ptr<RtcEventLog> Create(const Clock* clock,
>-+  static std::unique_ptr<RtcEventLog> Create(const Clock*,
>-                                              EncodingType encoding_type) {
>-     return Create(encoding_type);
>-   }
>-@@ -65,12 +65,12 @@ class RtcEventLog {
>- // No-op implementation is used if flag is not set, or in tests.
>- class RtcEventLogNullImpl : public RtcEventLog {
>-  public:
>--  bool StartLogging(std::unique_ptr<RtcEventLogOutput> output,
>--                    int64_t output_period_ms) override {
>-+  bool StartLogging(std::unique_ptr<RtcEventLogOutput>,
>-+                    int64_t) override {
>-     return false;
>-   }
>-   void StopLogging() override {}
>--  void Log(std::unique_ptr<RtcEvent> event) override {}
>-+  void Log(std::unique_ptr<RtcEvent>) override {}
>- };
>- 
>- }  // namespace webrtc
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediachannel.h b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediachannel.h
>-index 4826b2dc4a7..1a409615467 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediachannel.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/base/mediachannel.h
>-@@ -1228,13 +1228,13 @@ class DataMediaChannel : public MediaChannel {
>-   virtual bool SetRecvParameters(const DataRecvParameters& params) = 0;
>- 
>-   // TODO(pthatcher): Implement this.
>--  virtual bool GetStats(DataMediaInfo* info) { return true; }
>-+  virtual bool GetStats(DataMediaInfo*) { return true; }
>- 
>-   virtual bool SetSend(bool send) = 0;
>-   virtual bool SetReceive(bool receive) = 0;
>- 
>--  virtual void OnNetworkRouteChanged(const std::string& transport_name,
>--                                     const rtc::NetworkRoute& network_route) {}
>-+  virtual void OnNetworkRouteChanged(const std::string&,
>-+                                     const rtc::NetworkRoute&) {}
>- 
>-   virtual bool SendData(
>-       const SendDataParams& params,
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/audio_device.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/audio_device.h
>-index c096a8398cd..754c4e16ef9 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/audio_device.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/include/audio_device.h
>-@@ -143,29 +143,29 @@ class AudioDeviceModule : public rtc::RefCountInterface {
>-   virtual int32_t SetStereoRecording(bool enable) = 0;
>-   virtual int32_t StereoRecording(bool* enabled) const = 0;
>-   // TODO(bugs.webrtc.org/7306): deprecated.
>--  virtual int32_t SetRecordingChannel(const ChannelType channel) { return -1; }
>--  virtual int32_t RecordingChannel(ChannelType* channel) const { return -1; }
>-+  virtual int32_t SetRecordingChannel(const ChannelType) { return -1; }
>-+  virtual int32_t RecordingChannel(ChannelType*) const { return -1; }
>- 
>-   // Playout delay
>-   virtual int32_t PlayoutDelay(uint16_t* delayMS) const = 0;
>- 
>-   // TODO(bugs.webrtc.org/7306): deprecated (to be removed).
>--  virtual int32_t SetRecordingSampleRate(const uint32_t samplesPerSec) {
>-+  virtual int32_t SetRecordingSampleRate(const uint32_t) {
>-     return -1;
>-   }
>--  virtual int32_t RecordingSampleRate(uint32_t* samplesPerSec) const {
>-+  virtual int32_t RecordingSampleRate(uint32_t*) const {
>-     return -1;
>-   }
>--  virtual int32_t SetPlayoutSampleRate(const uint32_t samplesPerSec) {
>-+  virtual int32_t SetPlayoutSampleRate(const uint32_t) {
>-     return -1;
>-   }
>--  virtual int32_t PlayoutSampleRate(uint32_t* samplesPerSec) const {
>-+  virtual int32_t PlayoutSampleRate(uint32_t*) const {
>-     return -1;
>-   }
>- 
>-   // TODO(bugs.webrtc.org/7306): deprecated (to be removed).
>--  virtual int32_t SetLoudspeakerStatus(bool enable) { return -1; }
>--  virtual int32_t GetLoudspeakerStatus(bool* enabled) const { return -1; }
>-+  virtual int32_t SetLoudspeakerStatus(bool) { return -1; }
>-+  virtual int32_t GetLoudspeakerStatus(bool*) const { return -1; }
>- 
>-   // Only supported on Android.
>-   virtual bool BuiltInAECIsAvailable() const = 0;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/icetransportinternal.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/icetransportinternal.h
>-index 0a71162eea1..16f6b699a9b 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/icetransportinternal.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/icetransportinternal.h
>-@@ -62,7 +62,7 @@ class IceTransportInternal : public rtc::PacketTransportInternal {
>- 
>-   // TODO(zhihuang): Remove this once it's no longer called in
>-   // remoting/protocol/libjingle_transport_factory.cc
>--  virtual void SetIceProtocolType(IceProtocolType type) {}
>-+  virtual void SetIceProtocolType(IceProtocolType) {}
>- 
>-   virtual void SetIceCredentials(const std::string& ice_ufrag,
>-                                  const std::string& ice_pwd);
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/port.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/port.h
>-index 2957fd8b742..380fd896933 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/port.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/port.h
>-@@ -380,7 +380,7 @@ class Port : public PortInterface, public rtc::MessageHandler,
>-   }
>- 
>-   // Extra work to be done in subclasses when a connection is destroyed.
>--  virtual void HandleConnectionDestroyed(Connection* conn) {}
>-+  virtual void HandleConnectionDestroyed(Connection*) {}
>- 
>-  private:
>-   void Construct();
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stun.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stun.h
>-index 6ff07b5766e..6ecd40d1779 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stun.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stun.h
>-@@ -218,7 +218,7 @@ class StunAttribute {
>-   virtual StunAttributeValueType value_type() const = 0;
>- 
>-   // Only XorAddressAttribute needs this so far.
>--  virtual void SetOwner(StunMessage* owner) {}
>-+  virtual void SetOwner(StunMessage*) {}
>- 
>-   // Reads the body (not the type or length) for this type of attribute from
>-   // the given buffer.  Return value is true if successful.
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunrequest.h b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunrequest.h
>-index 7bb04664089..9bca20ae20b 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunrequest.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/p2p/base/stunrequest.h
>-@@ -118,11 +118,11 @@ class StunRequest : public rtc::MessageHandler {
>- 
>-   // Fills in a request object to be sent.  Note that request's transaction ID
>-   // will already be set and cannot be changed.
>--  virtual void Prepare(StunMessage* request) {}
>-+  virtual void Prepare(StunMessage*) {}
>- 
>-   // Called when the message receives a response or times out.
>--  virtual void OnResponse(StunMessage* response) {}
>--  virtual void OnErrorResponse(StunMessage* response) {}
>-+  virtual void OnResponse(StunMessage*) {}
>-+  virtual void OnErrorResponse(StunMessage*) {}
>-   virtual void OnTimeout() {}
>-   // Called when the message is sent.
>-   virtual void OnSent();
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagehandler.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagehandler.h
>-index ff953f7898d..dec3b8b86a0 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagehandler.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/messagehandler.h
>-@@ -40,7 +40,7 @@ class FunctorMessageHandler : public MessageHandler {
>-  public:
>-   explicit FunctorMessageHandler(FunctorT&& functor)
>-       : functor_(std::forward<FunctorT>(functor)) {}
>--  virtual void OnMessage(Message* msg) {
>-+  virtual void OnMessage(Message*) {
>-     result_ = functor_();
>-   }
>-   const ReturnT& result() const { return result_; }
>-@@ -60,7 +60,7 @@ class FunctorMessageHandler<void, FunctorT> : public MessageHandler {
>-  public:
>-   explicit FunctorMessageHandler(const FunctorT& functor)
>-       : functor_(functor) {}
>--  virtual void OnMessage(Message* msg) {
>-+  virtual void OnMessage(Message*) {
>-     functor_();
>-   }
>-   void result() const {}
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/network.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/network.h
>-index 5755699fcdf..a1b5c4fa68a 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/network.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/network.h
>-@@ -117,7 +117,7 @@ class NetworkManager : public DefaultLocalAddressProvider {
>-   // can optionally be called after GetNetworks.
>-   //
>-   // TODO(guoweis): remove this body when chromium implements this.
>--  virtual void GetAnyAddressNetworks(NetworkList* networks) {}
>-+  virtual void GetAnyAddressNetworks(NetworkList*) {}
>- 
>-   // Dumps the current list of networks in the network manager.
>-   virtual void DumpNetworks() {}
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/safe_conversions_impl.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/safe_conversions_impl.h
>-index 9b4f1c6483d..0174a83d2f5 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/safe_conversions_impl.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/safe_conversions_impl.h
>-@@ -101,7 +101,7 @@ struct RangeCheckImpl {};
>- // Dst range always contains the result: nothing to check.
>- template <typename Dst, typename Src, DstSign IsDstSigned, SrcSign IsSrcSigned>
>- struct RangeCheckImpl<Dst, Src, IsDstSigned, IsSrcSigned, CONTAINS_RANGE> {
>--  static RangeCheckResult Check(Src value) { return TYPE_VALID; }
>-+  static RangeCheckResult Check(Src) { return TYPE_VALID; }
>- };
>- 
>- // Signed to signed narrowing.
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketserver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketserver.h
>-index e29884e6b3c..38543693d37 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketserver.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/socketserver.h
>-@@ -35,7 +35,7 @@ class SocketServer : public SocketFactory {
>-   // When the socket server is installed into a Thread, this function is
>-   // called to allow the socket server to use the thread's message queue for
>-   // any messaging that it might need to perform.
>--  virtual void SetMessageQueue(MessageQueue* queue) {}
>-+  virtual void SetMessageQueue(MessageQueue*) {}
>- 
>-   // Sleeps until:
>-   //  1) cms milliseconds have elapsed (unless cms == kForever)
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stream.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stream.h
>-index 171a09f1e00..c959c228ea8 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stream.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stream.h
>-@@ -132,7 +132,7 @@ class StreamInterface : public MessageHandler {
>-   // processed.  Read and ConsumeReadData invalidate the buffer returned by
>-   // GetReadData.
>-   virtual const void* GetReadData(size_t* data_len);
>--  virtual void ConsumeReadData(size_t used) {}
>-+  virtual void ConsumeReadData(size_t) {}
>- 
>-   // GetWriteBuffer returns a pointer to a buffer which is owned by the stream.
>-   // The buffer has a capacity of buf_len bytes.  null is returned if there is
>-@@ -146,7 +146,7 @@ class StreamInterface : public MessageHandler {
>-   // when it is available.  If the requested amount is too large, return an
>-   // error.
>-   virtual void* GetWriteBuffer(size_t* buf_len);
>--  virtual void ConsumeWriteBuffer(size_t used) {}
>-+  virtual void ConsumeWriteBuffer(size_t) {}
>- 
>-   // Write data_len bytes found in data, circumventing any throttling which
>-   // would could cause SR_BLOCK to be returned.  Returns true if all the data
>--- 
>-2.13.5 (Apple Git-94)
>-
>diff --git a/Source/ThirdParty/libwebrtc/WebKit/0005-Fix-RTC_FATAL.patch b/Source/ThirdParty/libwebrtc/WebKit/0005-Fix-RTC_FATAL.patch
>deleted file mode 100644
>index 43aa5a00dcd..00000000000
>--- a/Source/ThirdParty/libwebrtc/WebKit/0005-Fix-RTC_FATAL.patch
>+++ /dev/null
>@@ -1,537 +0,0 @@
>-From 62e7f44af723f2f7f28038a9f08355a29cb0dc24 Mon Sep 17 00:00:00 2001
>-From: Youenn Fablet <youenn@apple.com>
>-Date: Thu, 14 Dec 2017 13:56:05 -0800
>-Subject: [PATCH 5/8] Fix RTC_FATAL
>-
>----
>- .../api/audio_codecs/ilbc/audio_encoder_ilbc.cc    |  2 +-
>- .../libwebrtc/Source/webrtc/api/mediatypes.cc      |  4 +--
>- .../webrtc/common_video/video_frame_buffer.cc      |  2 +-
>- .../audio_coding/acm2/audio_coding_module.cc       |  2 +-
>- .../modules/audio_coding/acm2/rent_a_codec.cc      |  4 +--
>- .../audio_coding/codecs/cng/audio_encoder_cng.cc   |  2 +-
>- .../audio_coding/codecs/ilbc/audio_encoder_ilbc.cc |  4 +--
>- .../audio_coding/neteq/tools/neteq_rtpplay.cc      |  2 +-
>- .../modules/audio_coding/neteq/tools/neteq_test.cc |  4 +--
>- .../audio_coding/neteq/tools/rtp_file_source.cc    |  2 +-
>- .../audio_device/android/audio_device_template.h   | 30 +++++++++++-----------
>- .../audio_device/android/audio_record_jni.cc       |  2 +-
>- .../quality_assessment/fake_polqa.cc               |  2 +-
>- .../audio_processing/test/wav_based_simulator.cc   |  2 +-
>- .../libwebrtc/Source/webrtc/ortc/ortcfactory.cc    |  4 +--
>- .../Source/webrtc/ortc/rtpparametersconversion.cc  |  2 +-
>- .../libwebrtc/Source/webrtc/rtc_base/checks.h      |  6 ++---
>- .../libwebrtc/Source/webrtc/rtc_base/flags.cc      |  6 ++---
>- .../webrtc/rtc_base/numerics/safe_conversions.h    |  4 +--
>- .../Source/webrtc/rtc_base/opensslidentity.cc      |  8 +++---
>- .../Source/webrtc/test/rtp_file_reader.cc          |  4 +--
>- .../Source/webrtc/voice_engine/utility.cc          |  4 +--
>- 22 files changed, 51 insertions(+), 51 deletions(-)
>-
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.cc
>-index a7c68ffcf02..6db88402e39 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/audio_codecs/ilbc/audio_encoder_ilbc.cc
>-@@ -33,7 +33,7 @@ int GetIlbcBitrate(int ptime) {
>-       // 50 bytes per frame of 30 ms => (approx) 13333 bits/s.
>-       return 13333;
>-     default:
>--      FATAL();
>-+      RTC_FATAL();
>-   }
>- }
>- }  // namespace
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediatypes.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediatypes.cc
>-index 599542db08f..140d0ae9625 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediatypes.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/api/mediatypes.cc
>-@@ -28,7 +28,7 @@ std::string MediaTypeToString(MediaType type) {
>-     case MEDIA_TYPE_DATA:
>-       return kMediaTypeData;
>-   }
>--  FATAL();
>-+  RTC_FATAL();
>-   // Not reachable; avoids compile warning.
>-   return "";
>- }
>-@@ -41,7 +41,7 @@ MediaType MediaTypeFromString(const std::string& type_str) {
>-   } else if (type_str == kMediaTypeData) {
>-     return MEDIA_TYPE_DATA;
>-   }
>--  FATAL();
>-+  RTC_FATAL();
>-   // Not reachable; avoids compile warning.
>-   return static_cast<MediaType>(-1);
>- }
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/video_frame_buffer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/video_frame_buffer.cc
>-index 88c8ca4a5d4..c05f72e6d17 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/video_frame_buffer.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/common_video/video_frame_buffer.cc
>-@@ -253,7 +253,7 @@ rtc::scoped_refptr<PlanarYuvBuffer> WrapYuvBuffer(
>-       return WrapI444Buffer(width, height, y_plane, y_stride, u_plane, u_stride,
>-                             v_plane, v_stride, no_longer_used);
>-     default:
>--      FATAL() << "Unexpected frame buffer type.";
>-+      RTC_FATAL() << "Unexpected frame buffer type.";
>-       return nullptr;
>-   }
>- }
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/audio_coding_module.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
>-index 18325f41590..f9c9690e98e 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
>-@@ -1145,7 +1145,7 @@ int AudioCodingModuleImpl::SetOpusApplication(OpusApplicationMode application) {
>-       app = AudioEncoder::Application::kAudio;
>-       break;
>-     default:
>--      FATAL();
>-+      RTC_FATAL();
>-       return 0;
>-   }
>-   return encoder_stack_->SetApplication(app) ? 0 : -1;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/rent_a_codec.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/rent_a_codec.cc
>-index 78db38dad60..4a1746e3da9 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/rent_a_codec.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/acm2/rent_a_codec.cc
>-@@ -215,7 +215,7 @@ std::unique_ptr<AudioEncoder> CreateCngEncoder(
>-       config.vad_mode = Vad::kVadVeryAggressive;
>-       break;
>-     default:
>--      FATAL();
>-+      RTC_FATAL();
>-   }
>-   return std::unique_ptr<AudioEncoder>(new AudioEncoderCng(std::move(config)));
>- }
>-@@ -230,7 +230,7 @@ std::unique_ptr<AudioDecoder> CreateIsacDecoder(
>-   return std::unique_ptr<AudioDecoder>(
>-       new AudioDecoderIsacFloatImpl(sample_rate_hz, bwinfo));
>- #else
>--  FATAL() << "iSAC is not supported.";
>-+  RTC_FATAL() << "iSAC is not supported.";
>-   return std::unique_ptr<AudioDecoder>();
>- #endif
>- }
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
>-index 78148ab3504..5a043ecbccb 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
>-@@ -139,7 +139,7 @@ AudioEncoder::EncodedInfo AudioEncoderCng::EncodeImpl(
>-       break;
>-     }
>-     case Vad::kError: {
>--      FATAL();  // Fails only if fed invalid data.
>-+      RTC_FATAL();  // Fails only if fed invalid data.
>-       break;
>-     }
>-   }
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
>-index 6ddc078f095..e108c412ceb 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.cc
>-@@ -40,7 +40,7 @@ int GetIlbcBitrate(int ptime) {
>-       // 50 bytes per frame of 30 ms => (approx) 13333 bits/s.
>-       return 13333;
>-     default:
>--      FATAL();
>-+      RTC_FATAL();
>-   }
>- }
>- 
>-@@ -148,7 +148,7 @@ size_t AudioEncoderIlbcImpl::RequiredOutputSizeBytes() const {
>-     case 3:   return 50;
>-     case 4:   return 2 * 38;
>-     case 6:   return 2 * 50;
>--    default:  FATAL();
>-+    default:  RTC_FATAL();
>-   }
>- }
>- 
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
>-index 2657bfaeb5b..4137f8e0c4f 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc
>-@@ -171,7 +171,7 @@ std::string CodecName(NetEqDecoder codec) {
>-     case NetEqDecoder::kDecoderCNGswb48kHz:
>-       return "comfort noise (48 kHz)";
>-     default:
>--      FATAL();
>-+      RTC_FATAL();
>-       return "undefined";
>-   }
>- }
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_test.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_test.cc
>-index 39d95495779..295111a97d9 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_test.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/neteq_test.cc
>-@@ -21,12 +21,12 @@ void DefaultNetEqTestErrorCallback::OnInsertPacketError(
>-     const NetEqInput::PacketData& packet) {
>-   std::cerr << "InsertPacket returned an error." << std::endl;
>-   std::cerr << "Packet data: " << packet.ToString() << std::endl;
>--  FATAL();
>-+  RTC_FATAL();
>- }
>- 
>- void DefaultNetEqTestErrorCallback::OnGetAudioError() {
>-   std::cerr << "GetAudio returned an error." << std::endl;
>--  FATAL();
>-+  RTC_FATAL();
>- }
>- 
>- NetEqTest::NetEqTest(const NetEq::Config& config,
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
>-index c9ae5f2e714..d1b1008d704 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_coding/neteq/tools/rtp_file_source.cc
>-@@ -93,7 +93,7 @@ bool RtpFileSource::OpenFile(const std::string& file_name) {
>-     return true;
>-   rtp_reader_.reset(RtpFileReader::Create(RtpFileReader::kPcap, file_name));
>-   if (!rtp_reader_) {
>--    FATAL() << "Couldn't open input file as either a rtpdump or .pcap. Note "
>-+    RTC_FATAL() << "Couldn't open input file as either a rtpdump or .pcap. Note "
>-                "that .pcapng is not supported.";
>-   }
>-   return true;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/audio_device_template.h b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/audio_device_template.h
>-index 75dbaf315f5..a4ef5dfa310 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/audio_device_template.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/audio_device_template.h
>-@@ -103,14 +103,14 @@ class AudioDeviceTemplate : public AudioDeviceGeneric {
>-   int32_t PlayoutDeviceName(uint16_t index,
>-                             char name[kAdmMaxDeviceNameSize],
>-                             char guid[kAdmMaxGuidSize]) override {
>--    FATAL() << "Should never be called";
>-+    RTC_FATAL() << "Should never be called";
>-     return -1;
>-   }
>- 
>-   int32_t RecordingDeviceName(uint16_t index,
>-                               char name[kAdmMaxDeviceNameSize],
>-                               char guid[kAdmMaxGuidSize]) override {
>--    FATAL() << "Should never be called";
>-+    RTC_FATAL() << "Should never be called";
>-     return -1;
>-   }
>- 
>-@@ -123,7 +123,7 @@ class AudioDeviceTemplate : public AudioDeviceGeneric {
>- 
>-   int32_t SetPlayoutDevice(
>-       AudioDeviceModule::WindowsDeviceType device) override {
>--    FATAL() << "Should never be called";
>-+    RTC_FATAL() << "Should never be called";
>-     return -1;
>-   }
>- 
>-@@ -136,7 +136,7 @@ class AudioDeviceTemplate : public AudioDeviceGeneric {
>- 
>-   int32_t SetRecordingDevice(
>-       AudioDeviceModule::WindowsDeviceType device) override {
>--    FATAL() << "Should never be called";
>-+    RTC_FATAL() << "Should never be called";
>-     return -1;
>-   }
>- 
>-@@ -217,7 +217,7 @@ class AudioDeviceTemplate : public AudioDeviceGeneric {
>- 
>-   int32_t SetAGC(bool enable) override {
>-     if (enable) {
>--      FATAL() << "Should never be called";
>-+      RTC_FATAL() << "Should never be called";
>-     }
>-     return -1;
>-   }
>-@@ -278,52 +278,52 @@ class AudioDeviceTemplate : public AudioDeviceGeneric {
>-   }
>- 
>-   int32_t SetMicrophoneVolume(uint32_t volume) override {
>--    FATAL() << "Should never be called";
>-+    RTC_FATAL() << "Should never be called";
>-     return -1;
>-   }
>- 
>-   int32_t MicrophoneVolume(uint32_t& volume) const override {
>--    FATAL() << "Should never be called";
>-+    RTC_FATAL() << "Should never be called";
>-     return -1;
>-   }
>- 
>-   int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const override {
>--    FATAL() << "Should never be called";
>-+    RTC_FATAL() << "Should never be called";
>-     return -1;
>-   }
>- 
>-   int32_t MinMicrophoneVolume(uint32_t& minVolume) const override {
>--    FATAL() << "Should never be called";
>-+    RTC_FATAL() << "Should never be called";
>-     return -1;
>-   }
>- 
>-   int32_t SpeakerMuteIsAvailable(bool& available) override {
>--    FATAL() << "Should never be called";
>-+    RTC_FATAL() << "Should never be called";
>-     return -1;
>-   }
>- 
>-   int32_t SetSpeakerMute(bool enable) override {
>--    FATAL() << "Should never be called";
>-+    RTC_FATAL() << "Should never be called";
>-     return -1;
>-   }
>- 
>-   int32_t SpeakerMute(bool& enabled) const override {
>--    FATAL() << "Should never be called";
>-+    RTC_FATAL() << "Should never be called";
>-     return -1;
>-   }
>- 
>-   int32_t MicrophoneMuteIsAvailable(bool& available) override {
>--    FATAL() << "Not implemented";
>-+    RTC_FATAL() << "Not implemented";
>-     return -1;
>-   }
>- 
>-   int32_t SetMicrophoneMute(bool enable) override {
>--    FATAL() << "Not implemented";
>-+    RTC_FATAL() << "Not implemented";
>-     return -1;
>-   }
>- 
>-   int32_t MicrophoneMute(bool& enabled) const override {
>--    FATAL() << "Not implemented";
>-+    RTC_FATAL() << "Not implemented";
>-     return -1;
>-   }
>- 
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/audio_record_jni.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/audio_record_jni.cc
>-index 79f8c8b0f9b..b8a564f06f2 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/audio_record_jni.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/android/audio_record_jni.cc
>-@@ -196,7 +196,7 @@ int32_t AudioRecordJni::EnableBuiltInAEC(bool enable) {
>- 
>- int32_t AudioRecordJni::EnableBuiltInAGC(bool enable) {
>-   // TODO(henrika): possibly remove when no longer used by any client.
>--  FATAL() << "Should never be called";
>-+  RTC_FATAL() << "Should never be called";
>-   return -1;
>- }
>- 
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/fake_polqa.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/fake_polqa.cc
>-index 62d8ebb84d9..bae652e2835 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/fake_polqa.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/py_quality_assessment/quality_assessment/fake_polqa.cc
>-@@ -43,7 +43,7 @@ int main(int argc, char* argv[]) {
>-       return 0;
>-     }
>-   }
>--  FATAL() << kErrorMessage;
>-+  RTC_FATAL() << kErrorMessage;
>- }
>- 
>- }  // namespace test
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/wav_based_simulator.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/wav_based_simulator.cc
>-index f53d1e5d8d9..6ed8e529416 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/wav_based_simulator.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_processing/test/wav_based_simulator.cc
>-@@ -40,7 +40,7 @@ WavBasedSimulator::GetCustomEventChain(const std::string& filename) {
>-       case '\n':
>-         break;
>-       default:
>--        FATAL() << "Incorrect custom call order file, reverting to using the "
>-+        RTC_FATAL() << "Incorrect custom call order file, reverting to using the "
>-                    "default call order";
>-         fclose(stream);
>-         return WavBasedSimulator::GetDefaultEventChain();
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcfactory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcfactory.cc
>-index 7492dd47aae..b9f0f61e5d8 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcfactory.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/ortcfactory.cc
>-@@ -332,7 +332,7 @@ RtpCapabilities OrtcFactory::GetRtpSenderCapabilities(
>-       return RtpCapabilities();
>-   }
>-   // Not reached; avoids compile warning.
>--  FATAL();
>-+  RTC_FATAL();
>- }
>- 
>- RTCErrorOr<std::unique_ptr<OrtcRtpSenderInterface>>
>-@@ -397,7 +397,7 @@ RtpCapabilities OrtcFactory::GetRtpReceiverCapabilities(
>-       return RtpCapabilities();
>-   }
>-   // Not reached; avoids compile warning.
>--  FATAL();
>-+  RTC_FATAL();
>- }
>- 
>- RTCErrorOr<std::unique_ptr<OrtcRtpReceiverInterface>>
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtpparametersconversion.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtpparametersconversion.cc
>-index 0a657eca8f3..bc702909e34 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtpparametersconversion.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/ortc/rtpparametersconversion.cc
>-@@ -62,7 +62,7 @@ RTCErrorOr<cricket::FeedbackParam> ToCricketFeedbackParam(
>-       return cricket::FeedbackParam(cricket::kRtcpFbParamTransportCc);
>-   }
>-   // Not reached; avoids compile warning.
>--  FATAL();
>-+  RTC_FATAL();
>- }
>- 
>- template <typename C>
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/checks.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/checks.h
>-index e9a19e6051f..cf2965fd77e 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/checks.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/checks.h
>-@@ -73,7 +73,7 @@ NO_RETURN void rtc_FatalMessage(const char* file, int line, const char* msg);
>- //   messages if the condition doesn't hold. Prefer them to raw RTC_CHECK and
>- //   RTC_DCHECK.
>- //
>--// - FATAL() aborts unconditionally.
>-+// - RTC_FATAL() aborts unconditionally.
>- //
>- // TODO(ajm): Ideally, checks.h would be combined with logging.h, but
>- // consolidation with system_wrappers/logging.h should happen first.
>-@@ -135,7 +135,7 @@ std::string* MakeCheckOpString(const t1& v1, const t2& v2, const char* names) {
>- }
>- 
>- // MSVC doesn't like complex extern templates and DLLs.
>--#if !defined(COMPILER_MSVC)
>-+#if 0
>- // Commonly used instantiations of MakeCheckOpString<>. Explicitly instantiated
>- // in logging.cc.
>- extern template std::string* MakeCheckOpString<int, int>(
>-@@ -222,7 +222,7 @@ class FatalMessageVoidify {
>- #define RTC_NOTREACHED() RTC_DCHECK(RTC_UNREACHABLE_CODE_HIT)
>- 
>- // TODO(bugs.webrtc.org/8454): Add an RTC_ prefix or rename differently.
>--#define FATAL() rtc::FatalMessage(__FILE__, __LINE__).stream()
>-+#define RTC_FATAL() rtc::FatalMessage(__FILE__, __LINE__).stream()
>- // TODO(ajm): Consider adding RTC_NOTIMPLEMENTED macro when
>- // base/logging.h and system_wrappers/logging.h are consolidated such that we
>- // can match the Chromium behavior.
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/flags.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/flags.cc
>-index 4f2b11a9624..1c24a76ace4 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/flags.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/flags.cc
>-@@ -57,7 +57,7 @@ void Flag::SetToDefault() {
>-       variable_->s = default_.s;
>-       return;
>-   }
>--  FATAL() << "unreachable code";
>-+  RTC_FATAL() << "unreachable code";
>- }
>- 
>- 
>-@@ -68,7 +68,7 @@ static const char* Type2String(Flag::Type type) {
>-     case Flag::FLOAT: return "float";
>-     case Flag::STRING: return "string";
>-   }
>--  FATAL() << "unreachable code";
>-+  RTC_FATAL() << "unreachable code";
>- }
>- 
>- 
>-@@ -87,7 +87,7 @@ static void PrintFlagValue(Flag::Type type, FlagValue* p) {
>-       printf("%s", p->s);
>-       return;
>-   }
>--  FATAL() << "unreachable code";
>-+  RTC_FATAL() << "unreachable code";
>- }
>- 
>- 
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/safe_conversions.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/safe_conversions.h
>-index 58efcaa746a..48c212e4d49 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/safe_conversions.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/numerics/safe_conversions.h
>-@@ -63,11 +63,11 @@ inline Dst saturated_cast(Src value) {
>- 
>-     // Should fail only on attempting to assign NaN to a saturated integer.
>-     case internal::TYPE_INVALID:
>--      FATAL();
>-+      RTC_FATAL();
>-       return std::numeric_limits<Dst>::max();
>-   }
>- 
>--  FATAL();
>-+  RTC_FATAL();
>-   return static_cast<Dst>(value);
>- }
>- 
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslidentity.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslidentity.cc
>-index 85ef176de09..bfb167aacef 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslidentity.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/opensslidentity.cc
>-@@ -411,11 +411,11 @@ OpenSSLCertificate* OpenSSLCertificate::GetReference() const {
>- std::string OpenSSLCertificate::ToPEMString() const {
>-   BIO* bio = BIO_new(BIO_s_mem());
>-   if (!bio) {
>--    FATAL() << "unreachable code";
>-+    RTC_FATAL() << "unreachable code";
>-   }
>-   if (!PEM_write_bio_X509(bio, x509_)) {
>-     BIO_free(bio);
>--    FATAL() << "unreachable code";
>-+    RTC_FATAL() << "unreachable code";
>-   }
>-   BIO_write(bio, "\0", 1);
>-   char* buffer;
>-@@ -432,11 +432,11 @@ void OpenSSLCertificate::ToDER(Buffer* der_buffer) const {
>-   // Calculates the DER representation of the certificate, from scratch.
>-   BIO* bio = BIO_new(BIO_s_mem());
>-   if (!bio) {
>--    FATAL() << "unreachable code";
>-+    RTC_FATAL() << "unreachable code";
>-   }
>-   if (!i2d_X509_bio(bio, x509_)) {
>-     BIO_free(bio);
>--    FATAL() << "unreachable code";
>-+    RTC_FATAL() << "unreachable code";
>-   }
>-   char* data;
>-   size_t length = BIO_get_mem_data(bio, &data);
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/test/rtp_file_reader.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/test/rtp_file_reader.cc
>-index 03674875432..8d7207d4e8a 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/test/rtp_file_reader.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/test/rtp_file_reader.cc
>-@@ -90,7 +90,7 @@ class InterleavedRtpFileReader : public RtpFileReaderImpl {
>-     uint32_t len = 0;
>-     TRY(ReadUint32(&len, file_));
>-     if (packet->length < len) {
>--      FATAL() << "Packet is too large to fit: " << len << " bytes vs "
>-+      RTC_FATAL() << "Packet is too large to fit: " << len << " bytes vs "
>-               << packet->length
>-               << " bytes allocated. Consider increasing the buffer "
>-                  "size";
>-@@ -178,7 +178,7 @@ class RtpDumpReader : public RtpFileReaderImpl {
>-     // Use 'len' here because a 'plen' of 0 specifies rtcp.
>-     len -= kPacketHeaderSize;
>-     if (packet->length < len) {
>--      FATAL() << "Packet is too large to fit: " << len << " bytes vs "
>-+      RTC_FATAL() << "Packet is too large to fit: " << len << " bytes vs "
>-               << packet->length
>-               << " bytes allocated. Consider increasing the buffer "
>-                  "size";
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/voice_engine/utility.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/voice_engine/utility.cc
>-index 939870245f1..5caa73ae451 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/voice_engine/utility.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/voice_engine/utility.cc
>-@@ -58,7 +58,7 @@ void RemixAndResample(const int16_t* src_data,
>- 
>-   if (resampler->InitializeIfNeeded(sample_rate_hz, dst_frame->sample_rate_hz_,
>-                                     audio_ptr_num_channels) == -1) {
>--    FATAL() << "InitializeIfNeeded failed: sample_rate_hz = " << sample_rate_hz
>-+    RTC_FATAL() << "InitializeIfNeeded failed: sample_rate_hz = " << sample_rate_hz
>-             << ", dst_frame->sample_rate_hz_ = " << dst_frame->sample_rate_hz_
>-             << ", audio_ptr_num_channels = " << audio_ptr_num_channels;
>-   }
>-@@ -72,7 +72,7 @@ void RemixAndResample(const int16_t* src_data,
>-                                        dst_frame->mutable_data(),
>-                                        AudioFrame::kMaxDataSizeSamples);
>-   if (out_length == -1) {
>--    FATAL() << "Resample failed: audio_ptr = " << audio_ptr
>-+    RTC_FATAL() << "Resample failed: audio_ptr = " << audio_ptr
>-             << ", src_length = " << src_length
>-             << ", dst_frame->mutable_data() = " << dst_frame->mutable_data();
>-   }
>--- 
>-2.13.5 (Apple Git-94)
>-
>diff --git a/Source/ThirdParty/libwebrtc/WebKit/0006-Disabling-VP8.patch b/Source/ThirdParty/libwebrtc/WebKit/0006-Disabling-VP8.patch
>deleted file mode 100644
>index 9773bf0dab5..00000000000
>--- a/Source/ThirdParty/libwebrtc/WebKit/0006-Disabling-VP8.patch
>+++ /dev/null
>@@ -1,100 +0,0 @@
>-From 201c211d31aa1e9e3b3cef70feff291512efde6d Mon Sep 17 00:00:00 2001
>-From: Youenn Fablet <youenn@apple.com>
>-Date: Thu, 14 Dec 2017 13:56:52 -0800
>-Subject: [PATCH 6/8] Disabling VP8
>-
>----
>- .../libwebrtc/Source/webrtc/media/engine/internaldecoderfactory.cc  | 3 ++-
>- .../libwebrtc/Source/webrtc/media/engine/internalencoderfactory.cc  | 3 ++-
>- .../libwebrtc/Source/webrtc/modules/video_coding/qp_parser.cc       | 6 +++++-
>- .../Source/webrtc/modules/video_coding/video_codec_initializer.cc   | 6 ++++--
>- 4 files changed, 13 insertions(+), 5 deletions(-)
>-
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/internaldecoderfactory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/internaldecoderfactory.cc
>-index e8cecb79a43..57daca256c4 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/internaldecoderfactory.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/internaldecoderfactory.cc
>-@@ -22,7 +22,8 @@ namespace webrtc {
>- std::vector<SdpVideoFormat> InternalDecoderFactory::GetSupportedFormats()
>-     const {
>-   std::vector<SdpVideoFormat> formats;
>--  formats.push_back(SdpVideoFormat(cricket::kVp8CodecName));
>-+  if (VP8Decoder::IsSupported())
>-+    formats.push_back(SdpVideoFormat(cricket::kVp8CodecName));
>-   if (VP9Decoder::IsSupported())
>-     formats.push_back(SdpVideoFormat(cricket::kVp9CodecName));
>-   for (const SdpVideoFormat& h264_format : SupportedH264Codecs())
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/internalencoderfactory.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/internalencoderfactory.cc
>-index 5f8479cfb82..0390646add4 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/internalencoderfactory.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/media/engine/internalencoderfactory.cc
>-@@ -22,7 +22,8 @@ namespace webrtc {
>- std::vector<SdpVideoFormat> InternalEncoderFactory::GetSupportedFormats()
>-     const {
>-   std::vector<SdpVideoFormat> supported_codecs;
>--  supported_codecs.push_back(SdpVideoFormat(cricket::kVp8CodecName));
>-+  if (VP8Encoder::IsSupported())
>-+    supported_codecs.push_back(SdpVideoFormat(cricket::kVp8CodecName));
>-   if (webrtc::VP9Encoder::IsSupported())
>-     supported_codecs.push_back(SdpVideoFormat(cricket::kVp9CodecName));
>- 
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/qp_parser.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/qp_parser.cc
>-index 2adbca9b712..418b8b7e5ff 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/qp_parser.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/qp_parser.cc
>-@@ -17,13 +17,17 @@
>- namespace webrtc {
>- 
>- bool QpParser::GetQp(const VCMEncodedFrame& frame, int* qp) {
>--  switch (frame.CodecSpecific()->codecType) {
>-+    switch (frame.CodecSpecific()->codecType) {
>-+#if !RTC_DISABLE_VP8
>-     case kVideoCodecVP8:
>-       // QP range: [0, 127].
>-       return vp8::GetQp(frame.Buffer(), frame.Length(), qp);
>-+#endif
>-+#if !RTC_DISABLE_VP9
>-     case kVideoCodecVP9:
>-       // QP range: [0, 255].
>-       return vp9::GetQp(frame.Buffer(), frame.Length(), qp);
>-+#endif
>-     default:
>-       return false;
>-   }
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_codec_initializer.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_codec_initializer.cc
>-index 3e1b0e1163b..89f6f0bad9d 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_codec_initializer.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/video_coding/video_codec_initializer.cc
>-@@ -42,7 +42,7 @@ bool VideoCodecInitializer::SetupCodec(
>-   *codec =
>-       VideoEncoderConfigToVideoCodec(config, streams, settings.payload_name,
>-                                      settings.payload_type, nack_enabled);
>--
>-+#if !defined(RTC_DISABLE_VP8)
>-   std::unique_ptr<TemporalLayersFactory> tl_factory;
>-   switch (codec->codecType) {
>-     case kVideoCodecVP8: {
>-@@ -68,7 +68,7 @@ bool VideoCodecInitializer::SetupCodec(
>-     }
>-   }
>-   *bitrate_allocator = CreateBitrateAllocator(*codec, std::move(tl_factory));
>--
>-+#endif
>-   return true;
>- }
>- 
>-@@ -80,9 +80,11 @@ VideoCodecInitializer::CreateBitrateAllocator(
>- 
>-   switch (codec.codecType) {
>-     case kVideoCodecVP8: {
>-+#if !defined(RTC_DISABLE_VP8)
>-       // Set up default VP8 temporal layer factory, if not provided.
>-       rate_allocator.reset(
>-           new SimulcastRateAllocator(codec, std::move(tl_factory)));
>-+#endif
>-     } break;
>-     default:
>-       rate_allocator.reset(new DefaultVideoBitrateAllocator(codec));
>--- 
>-2.13.5 (Apple Git-94)
>-
>diff --git a/Source/ThirdParty/libwebrtc/WebKit/0007-Fix-RTC_STRINGIZE.patch b/Source/ThirdParty/libwebrtc/WebKit/0007-Fix-RTC_STRINGIZE.patch
>deleted file mode 100644
>index 2a56c84bb4e..00000000000
>--- a/Source/ThirdParty/libwebrtc/WebKit/0007-Fix-RTC_STRINGIZE.patch
>+++ /dev/null
>@@ -1,38 +0,0 @@
>-From efa6c8d735872ad827a9ec5c3c7e09ab5cfac1c1 Mon Sep 17 00:00:00 2001
>-From: Youenn Fablet <youenn@apple.com>
>-Date: Thu, 14 Dec 2017 13:57:16 -0800
>-Subject: [PATCH 7/8] Fix RTC_STRINGIZE
>-
>----
>- Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/location.h         | 2 +-
>- Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringize_macros.h | 2 +-
>- 2 files changed, 2 insertions(+), 2 deletions(-)
>-
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/location.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/location.h
>-index 513bc263651..718d9589348 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/location.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/location.h
>-@@ -50,7 +50,7 @@ class Location {
>- #define RTC_FROM_HERE RTC_FROM_HERE_WITH_FUNCTION(__FUNCTION__)
>- 
>- #define RTC_FROM_HERE_WITH_FUNCTION(function_name) \
>--  ::rtc::Location(function_name, __FILE__ ":" STRINGIZE(__LINE__))
>-+  ::rtc::Location(function_name, __FILE__ ":" RTC_STRINGIZE(__LINE__))
>- 
>- }  // namespace rtc
>- 
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringize_macros.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringize_macros.h
>-index aee8d14551d..acf0ba404ca 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringize_macros.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/stringize_macros.h
>-@@ -33,6 +33,6 @@
>- // Then:
>- //   STRINGIZE(A) produces "FOO"
>- //   STRINGIZE(B(y)) produces "myobj->FunctionCall(y)"
>--#define STRINGIZE(x) STRINGIZE_NO_EXPANSION(x)
>-+#define RTC_STRINGIZE(x) STRINGIZE_NO_EXPANSION(x)
>- 
>- #endif  // RTC_BASE_STRINGIZE_MACROS_H_
>--- 
>-2.13.5 (Apple Git-94)
>-
>diff --git a/Source/ThirdParty/libwebrtc/WebKit/0008-Fix-sanitizer.patch b/Source/ThirdParty/libwebrtc/WebKit/0008-Fix-sanitizer.patch
>deleted file mode 100644
>index 18ae7715c54..00000000000
>--- a/Source/ThirdParty/libwebrtc/WebKit/0008-Fix-sanitizer.patch
>+++ /dev/null
>@@ -1,60 +0,0 @@
>-From 826df71603f4e35466cb5330bd18920d503e80b2 Mon Sep 17 00:00:00 2001
>-From: Youenn Fablet <youenn@apple.com>
>-Date: Thu, 14 Dec 2017 13:57:39 -0800
>-Subject: [PATCH 8/8] Fix sanitizer
>-
>----
>- .../libwebrtc/Source/webrtc/rtc_base/sanitizer.h         | 16 ++++++++++++++++
>- 1 file changed, 16 insertions(+)
>-
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sanitizer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sanitizer.h
>-index 1b94e1edf47..96d024883fa 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sanitizer.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/sanitizer.h
>-@@ -52,6 +52,10 @@ static inline void rtc_AsanPoison(const volatile void* ptr,
>-                                   size_t num_elements) {
>- #if RTC_HAS_ASAN
>-   ASAN_POISON_MEMORY_REGION(ptr, element_size * num_elements);
>-+#else
>-+  (void)ptr;
>-+  (void)element_size;
>-+  (void)num_elements;
>- #endif
>- }
>- 
>-@@ -63,6 +67,10 @@ static inline void rtc_AsanUnpoison(const volatile void* ptr,
>-                                     size_t num_elements) {
>- #if RTC_HAS_ASAN
>-   ASAN_UNPOISON_MEMORY_REGION(ptr, element_size * num_elements);
>-+#else
>-+  (void)ptr;
>-+  (void)element_size;
>-+  (void)num_elements;
>- #endif
>- }
>- 
>-@@ -73,6 +81,10 @@ static inline void rtc_MsanMarkUninitialized(const volatile void* ptr,
>-                                              size_t num_elements) {
>- #if RTC_HAS_MSAN
>-   __msan_poison(ptr, element_size * num_elements);
>-+#else
>-+  (void)ptr;
>-+  (void)element_size;
>-+  (void)num_elements;
>- #endif
>- }
>- 
>-@@ -84,6 +96,10 @@ static inline void rtc_MsanCheckInitialized(const volatile void* ptr,
>-                                             size_t num_elements) {
>- #if RTC_HAS_MSAN
>-   __msan_check_mem_is_initialized(ptr, element_size * num_elements);
>-+#else
>-+  (void)ptr;
>-+  (void)element_size;
>-+  (void)num_elements;
>- #endif
>- }
>- 
>--- 
>-2.13.5 (Apple Git-94)
>-
>diff --git a/Source/ThirdParty/libwebrtc/WebKit/0009-Remove-dispatch_set_target_queue.patch b/Source/ThirdParty/libwebrtc/WebKit/0009-Remove-dispatch_set_target_queue.patch
>deleted file mode 100644
>index c61406951b6..00000000000
>--- a/Source/ThirdParty/libwebrtc/WebKit/0009-Remove-dispatch_set_target_queue.patch
>+++ /dev/null
>@@ -1,23 +0,0 @@
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_gcd.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_gcd.cc
>-index a13e088ede6..675b8005722 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_gcd.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/rtc_base/task_queue_gcd.cc
>-@@ -139,7 +139,7 @@ class TaskQueue::Impl : public RefCountInterface {
>- TaskQueue::Impl::Impl(const char* queue_name,
>-                       TaskQueue* task_queue,
>-                       Priority priority)
>--    : queue_(dispatch_queue_create(queue_name, DISPATCH_QUEUE_SERIAL)),
>-+    : queue_(dispatch_queue_create_with_target(queue_name, DISPATCH_QUEUE_SERIAL, dispatch_get_global_queue(TaskQueuePriorityToGCD(priority), 0))),
>-       context_(new QueueContext(task_queue)) {
>-   RTC_DCHECK(queue_name);
>-   RTC_CHECK(queue_);
>-@@ -148,9 +148,6 @@ TaskQueue::Impl::Impl(const char* queue_name,
>-   // to the queue is released.  This may run after the TaskQueue object has
>-   // been deleted.
>-   dispatch_set_finalizer_f(queue_, &QueueContext::DeleteContext);
>--
>--  dispatch_set_target_queue(
>--      queue_, dispatch_get_global_queue(TaskQueuePriorityToGCD(priority), 0));
>- }
>- 
>- TaskQueue::Impl::~Impl() {
>diff --git a/Source/ThirdParty/libwebrtc/WebKit/0010-Fix-RTCVideoEncoderH264-CVPixelBuffer-leak.patch b/Source/ThirdParty/libwebrtc/WebKit/0010-Fix-RTCVideoEncoderH264-CVPixelBuffer-leak.patch
>deleted file mode 100644
>index 83098c349c6..00000000000
>--- a/Source/ThirdParty/libwebrtc/WebKit/0010-Fix-RTCVideoEncoderH264-CVPixelBuffer-leak.patch
>+++ /dev/null
>@@ -1,12 +0,0 @@
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoEncoderH264.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoEncoderH264.mm
>-index 42f940426ae..16a841d9458 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoEncoderH264.mm
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/RTCVideoEncoderH264.mm
>-@@ -429,6 +429,7 @@ - (NSInteger)encode:(RTCVideoFrame *)frame
>-       }
>-       _frameScaleBuffer.shrink_to_fit();
>-       if (![rtcPixelBuffer cropAndScaleTo:pixelBuffer withTempBuffer:_frameScaleBuffer.data()]) {
>-+        CVBufferRelease(pixelBuffer);
>-         return WEBRTC_VIDEO_CODEC_ERROR;
>-       }
>-     }
>diff --git a/Source/ThirdParty/libwebrtc/WebKit/0011-Fix-AudioDeviceID-array-leak.patch b/Source/ThirdParty/libwebrtc/WebKit/0011-Fix-AudioDeviceID-array-leak.patch
>deleted file mode 100644
>index 0bed7f48b6c..00000000000
>--- a/Source/ThirdParty/libwebrtc/WebKit/0011-Fix-AudioDeviceID-array-leak.patch
>+++ /dev/null
>@@ -1,56 +0,0 @@
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/mac/audio_device_mac.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/mac/audio_device_mac.cc
>-index c585c32cafc..89ea9317c1f 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/mac/audio_device_mac.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/modules/audio_device/mac/audio_device_mac.cc
>-@@ -19,6 +19,7 @@
>- #include <ApplicationServices/ApplicationServices.h>
>- #include <libkern/OSAtomic.h>  // OSAtomicCompareAndSwap()
>- #include <mach/mach.h>         // mach_task_self()
>-+#include <memory>              // std::make_unique<>() and std::unique_ptr<>()
>- #include <sys/sysctl.h>        // sysctlbyname()
>- 
>- namespace webrtc {
>-@@ -1564,8 +1565,8 @@ int32_t AudioDeviceMac::GetNumberDevices(const AudioObjectPropertyScope scope,
>-     return 0;
>-   }
>- 
>--  AudioDeviceID* deviceIds = (AudioDeviceID*)malloc(size);
>-   UInt32 numberDevices = size / sizeof(AudioDeviceID);
>-+  auto deviceIds = std::make_unique<AudioDeviceID[]>(numberDevices);
>-   AudioBufferList* bufferList = NULL;
>-   UInt32 numberScopedDevices = 0;
>- 
>-@@ -1597,7 +1598,7 @@ int32_t AudioDeviceMac::GetNumberDevices(const AudioObjectPropertyScope scope,
>-   bool listOK = true;
>- 
>-   WEBRTC_CA_LOG_ERR(AudioObjectGetPropertyData(
>--      kAudioObjectSystemObject, &propertyAddress, 0, NULL, &size, deviceIds));
>-+      kAudioObjectSystemObject, &propertyAddress, 0, NULL, &size, deviceIds.get()));
>-   if (err != noErr) {
>-     listOK = false;
>-   } else {
>-@@ -1641,11 +1642,6 @@ int32_t AudioDeviceMac::GetNumberDevices(const AudioObjectPropertyScope scope,
>-   }
>- 
>-   if (!listOK) {
>--    if (deviceIds) {
>--      free(deviceIds);
>--      deviceIds = NULL;
>--    }
>--
>-     if (bufferList) {
>-       free(bufferList);
>-       bufferList = NULL;
>-@@ -1654,12 +1650,6 @@ int32_t AudioDeviceMac::GetNumberDevices(const AudioObjectPropertyScope scope,
>-     return -1;
>-   }
>- 
>--  // Happy ending
>--  if (deviceIds) {
>--    free(deviceIds);
>--    deviceIds = NULL;
>--  }
>--
>-   return numberScopedDevices;
>- }
>- 
>diff --git a/Source/ThirdParty/libwebrtc/WebKit/0012-Add-WK-prefix-to-Objective-C-classes-and-protocols.patch b/Source/ThirdParty/libwebrtc/WebKit/0012-Add-WK-prefix-to-Objective-C-classes-and-protocols.patch
>deleted file mode 100644
>index 5d78ef037a3..00000000000
>--- a/Source/ThirdParty/libwebrtc/WebKit/0012-Add-WK-prefix-to-Objective-C-classes-and-protocols.patch
>+++ /dev/null
>@@ -1,1076 +0,0 @@
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/WebKitUtilities.mm b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/WebKitUtilities.mm
>-index 1490c89ac5f..5e65230f321 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/WebKitUtilities.mm
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/WebKit/WebKitUtilities.mm
>-@@ -33,6 +33,7 @@
>- #include <webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoFrameBuffer.h>
>- 
>- #if !defined(WEBRTC_IOS)
>-+__attribute__((objc_runtime_name("WK_RTCUIApplicationStatusObserver")))
>- @interface RTCUIApplicationStatusObserver : NSObject
>- 
>- + (instancetype)sharedInstance;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Common/RTCUIApplicationStatusObserver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Common/RTCUIApplicationStatusObserver.h
>-index a2064df5db9..8ee12b62499 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Common/RTCUIApplicationStatusObserver.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Common/RTCUIApplicationStatusObserver.h
>-@@ -12,6 +12,7 @@
>- 
>- #import <Foundation/Foundation.h>
>- 
>-+__attribute__((objc_runtime_name("WK_RTCUIApplicationStatusObserver")))
>- @interface RTCUIApplicationStatusObserver : NSObject
>- 
>- + (instancetype)sharedInstance;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLI420Renderer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLI420Renderer.h
>-index e5987fe22a2..315b1f5a8da 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLI420Renderer.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLI420Renderer.h
>-@@ -13,5 +13,6 @@
>- #import "RTCMTLRenderer.h"
>- 
>- NS_AVAILABLE(10_11, 9_0)
>-+__attribute__((objc_runtime_name("WK_RTCMTLI420Renderer")))
>- @interface RTCMTLI420Renderer : RTCMTLRenderer
>- @end
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLNV12Renderer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLNV12Renderer.h
>-index 866b7ea17ee..8bbcb204d25 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLNV12Renderer.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLNV12Renderer.h
>-@@ -13,6 +13,7 @@
>- #import "RTCMTLRenderer.h"
>- 
>- NS_AVAILABLE(10_11, 9_0)
>-+__attribute__((objc_runtime_name("WK_RTCMTLNV12Renderer")))
>- @interface RTCMTLNV12Renderer : RTCMTLRenderer
>- 
>- @end
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLRenderer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLRenderer.h
>-index e279b098b55..110e2124656 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLRenderer.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Metal/RTCMTLRenderer.h
>-@@ -21,6 +21,7 @@ NS_ASSUME_NONNULL_BEGIN
>- /**
>-  * Protocol defining ability to render RTCVideoFrame in Metal enabled views.
>-  */
>-+__attribute__((objc_runtime_name("WK_RTCMTLRenderer")))
>- @protocol RTCMTLRenderer<NSObject>
>- 
>- /**
>-@@ -49,6 +50,7 @@ NS_ASSUME_NONNULL_BEGIN
>-  * Implementation of RTCMTLRenderer protocol for rendering native nv12 video frames.
>-  */
>- NS_AVAILABLE(10_11, 9_0)
>-+__attribute__((objc_runtime_name("WK_RTCMTLRenderer")))
>- @interface RTCMTLRenderer : NSObject<RTCMTLRenderer>
>- @end
>- 
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCDtmfSender+Private.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCDtmfSender+Private.h
>-index 197aeb83a42..04f10e8b399 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCDtmfSender+Private.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCDtmfSender+Private.h
>-@@ -14,6 +14,7 @@
>- 
>- NS_ASSUME_NONNULL_BEGIN
>- 
>-+__attribute__((objc_runtime_name("WK_RTCDtmfSender")))
>- @interface RTCDtmfSender : NSObject <RTCDtmfSender>
>- 
>- @property(nonatomic, readonly) rtc::scoped_refptr<webrtc::DtmfSenderInterface> nativeDtmfSender;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoRendererAdapter.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoRendererAdapter.h
>-index b0b6f044884..5b4570310d4 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoRendererAdapter.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCVideoRendererAdapter.h
>-@@ -18,6 +18,7 @@ NS_ASSUME_NONNULL_BEGIN
>-  * adapter adapts calls made to that interface to the RTCVideoRenderer supplied
>-  * during construction.
>-  */
>-+__attribute__((objc_runtime_name("WK_RTCVideoRendererAdapter")))
>- @interface RTCVideoRendererAdapter : NSObject
>- 
>- - (instancetype)init NS_UNAVAILABLE;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCWrappedNativeVideoDecoder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCWrappedNativeVideoDecoder.h
>-index 74cfaeb861f..cb29b692640 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCWrappedNativeVideoDecoder.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCWrappedNativeVideoDecoder.h
>-@@ -14,6 +14,7 @@
>- #include "api/video_codecs/video_decoder.h"
>- #include "media/base/codec.h"
>- 
>-+__attribute__((objc_runtime_name("WK_RTCWrappedNativeVideoDecoder")))
>- @interface RTCWrappedNativeVideoDecoder : NSObject <RTCVideoDecoder>
>- 
>- - (instancetype)initWithNativeDecoder:(std::unique_ptr<webrtc::VideoDecoder>)decoder;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCWrappedNativeVideoEncoder.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCWrappedNativeVideoEncoder.h
>-index 5b95d33eba9..29eff125a54 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCWrappedNativeVideoEncoder.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/PeerConnection/RTCWrappedNativeVideoEncoder.h
>-@@ -15,6 +15,7 @@
>- #include "api/video_codecs/video_encoder.h"
>- #include "media/base/codec.h"
>- 
>-+__attribute__((objc_runtime_name("WK_RTCWrappedNativeVideoEncoder")))
>- @interface RTCWrappedNativeVideoEncoder : NSObject <RTCVideoEncoder>
>- 
>- - (instancetype)initWithNativeEncoder:(std::unique_ptr<webrtc::VideoEncoder>)encoder;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/UI/RTCEAGLVideoView.m b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/UI/RTCEAGLVideoView.m
>-index 8f379c864ae..8f4da302fab 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/UI/RTCEAGLVideoView.m
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/UI/RTCEAGLVideoView.m
>-@@ -23,6 +23,7 @@
>- // refreshes, which should be 30fps. We wrap the display link in order to avoid
>- // a retain cycle since CADisplayLink takes a strong reference onto its target.
>- // The timer is paused by default.
>-+__attribute__((objc_runtime_name("WK_RTCDisplayLinkTimer")))
>- @interface RTCDisplayLinkTimer : NSObject
>- 
>- @property(nonatomic) BOOL isPaused;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCAVFoundationVideoCapturerInternal.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCAVFoundationVideoCapturerInternal.h
>-index 8b73a808639..217437aff2d 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCAVFoundationVideoCapturerInternal.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCAVFoundationVideoCapturerInternal.h
>-@@ -21,6 +21,7 @@ NS_ASSUME_NONNULL_BEGIN
>- // It is meant to be owned by an instance of AVFoundationVideoCapturer.
>- // The reason for this is because other webrtc objects own cricket::VideoCapturer, which is not
>- // ref counted. To prevent bad behavior we do not expose this class directly.
>-+__attribute__((objc_runtime_name("WK_RTCAVFoundationVideoCapturerInternal")))
>- @interface RTCAVFoundationVideoCapturerInternal
>-     : NSObject <AVCaptureVideoDataOutputSampleBufferDelegate>
>- 
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCDefaultShader.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCDefaultShader.h
>-index ed5125e2e8e..cc2ff4e46be 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCDefaultShader.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCDefaultShader.h
>-@@ -16,6 +16,7 @@ NS_ASSUME_NONNULL_BEGIN
>-  *  RTCEAGLVideoView if no external shader is specified. This shader will render
>-  *  the video in a rectangle without any color or geometric transformations.
>-  */
>-+__attribute__((objc_runtime_name("WK_RTCDefaultShader")))
>- @interface RTCDefaultShader : NSObject<RTCVideoViewShading>
>- 
>- @end
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCI420TextureCache.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCI420TextureCache.h
>-index 19f1bc99934..6726a8842b7 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCI420TextureCache.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCI420TextureCache.h
>-@@ -11,6 +11,7 @@
>- #import "RTCOpenGLDefines.h"
>- #import "WebRTC/RTCVideoFrame.h"
>- 
>-+__attribute__((objc_runtime_name("WK_RTCI420TextureCache")))
>- @interface RTCI420TextureCache : NSObject
>- 
>- @property(nonatomic, readonly) GLuint yTexture;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCNV12TextureCache.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCNV12TextureCache.h
>-index 9cba8232714..d59b73c9a93 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCNV12TextureCache.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/RTCNV12TextureCache.h
>-@@ -14,6 +14,7 @@
>- 
>- NS_ASSUME_NONNULL_BEGIN
>- 
>-+__attribute__((objc_runtime_name("WK_RTCNV12TextureCache")))
>- @interface RTCNV12TextureCache : NSObject
>- 
>- @property(nonatomic, readonly) GLuint yTexture;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/objc_frame_buffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/objc_frame_buffer.h
>-index 85a89e6dd9a..163028eb29d 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/objc_frame_buffer.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/Video/objc_frame_buffer.h
>-@@ -15,6 +15,7 @@
>- 
>- #include "common_video/include/video_frame_buffer.h"
>- 
>-+__attribute__((objc_runtime_name("WK_RTCVideoFrameBuffer")))
>- @protocol RTCVideoFrameBuffer;
>- 
>- namespace webrtc {
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/objc_video_decoder_factory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/objc_video_decoder_factory.h
>-index a0106ff83fe..13467e3898e 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/objc_video_decoder_factory.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/objc_video_decoder_factory.h
>-@@ -15,6 +15,7 @@
>- #include "media/base/codec.h"
>- #include "media/engine/webrtcvideodecoderfactory.h"
>- 
>-+__attribute__((objc_runtime_name("WK_RTCVideoDecoderFactory")))
>- @protocol RTCVideoDecoderFactory;
>- 
>- namespace webrtc {
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/objc_video_encoder_factory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/objc_video_encoder_factory.h
>-index 6cdfc56f548..727922bfe4e 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/objc_video_encoder_factory.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Classes/VideoToolbox/objc_video_encoder_factory.h
>-@@ -16,6 +16,7 @@
>- #include "api/video_codecs/video_encoder_factory.h"
>- #include "media/engine/webrtcvideoencoderfactory.h"
>- 
>-+__attribute__((objc_runtime_name("WK_RTCVideoEncoderFactory")))
>- @protocol RTCVideoEncoderFactory;
>- 
>- namespace webrtc {
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAVFoundationVideoSource.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAVFoundationVideoSource.h
>-index 6d369b34034..de73b477099 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAVFoundationVideoSource.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAVFoundationVideoSource.h
>-@@ -27,6 +27,7 @@ NS_ASSUME_NONNULL_BEGIN
>-  * counted video source interface.
>-  */
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCAVFoundationVideoSource")))
>- @interface RTCAVFoundationVideoSource : RTCVideoSource
>- 
>- - (instancetype)init NS_UNAVAILABLE;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSession.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSession.h
>-index 354634ec39f..e44aa39ae55 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSession.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSession.h
>-@@ -28,6 +28,7 @@ extern NSInteger const kRTCAudioSessionErrorConfiguration;
>- // from AVAudioSession and handle them before calling these delegate methods,
>- // at which point applications can perform additional processing if required.
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCAudioSessionDelegate")))
>- @protocol RTCAudioSessionDelegate <NSObject>
>- 
>- @optional
>-@@ -91,6 +92,7 @@ RTC_EXPORT
>-  *  case of this is when CallKit activates the audio session for the application
>-  */
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCAudioSessionActivationDelegate")))
>- @protocol RTCAudioSessionActivationDelegate <NSObject>
>- 
>- /** Called when the audio session is activated outside of the app by iOS. */
>-@@ -109,6 +111,7 @@ RTC_EXPORT
>-  *  activated only once. See |setActive:error:|.
>-  */
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCAudioSession")))
>- @interface RTCAudioSession : NSObject <RTCAudioSessionActivationDelegate>
>- 
>- /** Convenience property to access the AVAudioSession singleton. Callers should
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSessionConfiguration.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSessionConfiguration.h
>-index 6a02751d294..0097246e7c8 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSessionConfiguration.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSessionConfiguration.h
>-@@ -23,6 +23,7 @@ extern const double kRTCAudioSessionLowComplexityIOBufferDuration;
>- 
>- // Struct to hold configuration values.
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCAudioSessionConfiguration")))
>- @interface RTCAudioSessionConfiguration : NSObject
>- 
>- @property(nonatomic, strong) NSString *category;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSource.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSource.h
>-index 3db31573cb8..7287c31a738 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSource.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioSource.h
>-@@ -16,6 +16,7 @@
>- NS_ASSUME_NONNULL_BEGIN
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCAudioSource")))
>- @interface RTCAudioSource : RTCMediaSource
>- 
>- - (instancetype)init NS_UNAVAILABLE;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioTrack.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioTrack.h
>-index b83eede528c..bab1b1a0020 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioTrack.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCAudioTrack.h
>-@@ -16,6 +16,7 @@ NS_ASSUME_NONNULL_BEGIN
>- @class RTCAudioSource;
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCAudioTrack")))
>- @interface RTCAudioTrack : RTCMediaStreamTrack
>- 
>- - (instancetype)init NS_UNAVAILABLE;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCCameraPreviewView.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCCameraPreviewView.h
>-index 9018aec3d18..9f9a59fc87d 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCCameraPreviewView.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCCameraPreviewView.h
>-@@ -20,6 +20,7 @@
>-  *  AVCaptureSession.
>-  */
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCCameraPreviewView")))
>- @interface RTCCameraPreviewView : UIView
>- 
>- /** The capture session being rendered in the view. Capture session
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCCameraVideoCapturer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCCameraVideoCapturer.h
>-index 31b405c5426..b2e012778f3 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCCameraVideoCapturer.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCCameraVideoCapturer.h
>-@@ -19,6 +19,7 @@ NS_ASSUME_NONNULL_BEGIN
>- RTC_EXPORT
>- // Camera capture that implements RTCVideoCapturer. Delivers frames to a RTCVideoCapturerDelegate
>- // (usually RTCVideoSource).
>-+__attribute__((objc_runtime_name("WK_RTCCameraVideoCapturer")))
>- @interface RTCCameraVideoCapturer : RTCVideoCapturer
>- 
>- // Capture session that is used for capturing. Valid from initialization to dealloc.
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCConfiguration.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCConfiguration.h
>-index fcb62b94168..bf8e50c0504 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCConfiguration.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCConfiguration.h
>-@@ -66,6 +66,7 @@ typedef NS_ENUM(NSInteger, RTCEncryptionKeyType) {
>- NS_ASSUME_NONNULL_BEGIN
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCConfiguration")))
>- @interface RTCConfiguration : NSObject
>- 
>- /** An array of Ice Servers available to be used by ICE. */
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDataChannel.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDataChannel.h
>-index 893bd0a3365..1785513753a 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDataChannel.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDataChannel.h
>-@@ -16,6 +16,7 @@
>- NS_ASSUME_NONNULL_BEGIN
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCDataBuffer")))
>- @interface RTCDataBuffer : NSObject
>- 
>- /** NSData representation of the underlying buffer. */
>-@@ -37,6 +38,7 @@ RTC_EXPORT
>- 
>- @class RTCDataChannel;
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCDataChannelDelegate")))
>- @protocol RTCDataChannelDelegate <NSObject>
>- 
>- /** The data channel state changed. */
>-@@ -63,6 +65,7 @@ typedef NS_ENUM(NSInteger, RTCDataChannelState) {
>- };
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCDataChannel")))
>- @interface RTCDataChannel : NSObject
>- 
>- /**
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDataChannelConfiguration.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDataChannelConfiguration.h
>-index 65f59318447..0b8913ce549 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDataChannelConfiguration.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDataChannelConfiguration.h
>-@@ -16,6 +16,7 @@
>- NS_ASSUME_NONNULL_BEGIN
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCDataChannelConfiguration")))
>- @interface RTCDataChannelConfiguration : NSObject
>- 
>- /** Set to YES if ordered delivery is required. */
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDispatcher.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDispatcher.h
>-index 3dddead60ef..2166277bc17 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDispatcher.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDispatcher.h
>-@@ -26,6 +26,7 @@ typedef NS_ENUM(NSInteger, RTCDispatcherQueueType) {
>-  *  shared dispatch queue.
>-  */
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCDispatcher")))
>- @interface RTCDispatcher : NSObject
>- 
>- - (instancetype)init NS_UNAVAILABLE;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDtmfSender.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDtmfSender.h
>-index c69af026b7c..b4b6c6cf7d6 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDtmfSender.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCDtmfSender.h
>-@@ -15,6 +15,7 @@
>- NS_ASSUME_NONNULL_BEGIN
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCDtmfSender")))
>- @protocol RTCDtmfSender <NSObject>
>- 
>- /**
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCEAGLVideoView.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCEAGLVideoView.h
>-index 60c624e0d25..d4d3b54fee7 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCEAGLVideoView.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCEAGLVideoView.h
>-@@ -19,6 +19,7 @@ NS_ASSUME_NONNULL_BEGIN
>- 
>- @class RTCEAGLVideoView;
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCEAGLVideoViewDelegate")))
>- @protocol RTCEAGLVideoViewDelegate
>- 
>- - (void)videoView:(RTCEAGLVideoView *)videoView didChangeVideoSize:(CGSize)size;
>-@@ -30,6 +31,7 @@ RTC_EXPORT
>-  * bounds using OpenGLES 2.0 or OpenGLES 3.0.
>-  */
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCEAGLVideoView")))
>- @interface RTCEAGLVideoView : UIView <RTCVideoRenderer>
>- 
>- @property(nonatomic, weak) id<RTCEAGLVideoViewDelegate> delegate;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCFileLogger.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCFileLogger.h
>-index 1e8a591b73f..33df4efcf0f 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCFileLogger.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCFileLogger.h
>-@@ -34,6 +34,7 @@ NS_ASSUME_NONNULL_BEGIN
>- // For kRTCFileLoggerTypeApp, the oldest log is overwritten.
>- // This class is not threadsafe.
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCFileLogger")))
>- @interface RTCFileLogger : NSObject
>- 
>- // The severity level to capture. The default is kRTCFileLoggerSeverityInfo.
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCFileVideoCapturer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCFileVideoCapturer.h
>-index 7b898e494cf..70f482e35c7 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCFileVideoCapturer.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCFileVideoCapturer.h
>-@@ -26,6 +26,7 @@ typedef void (^RTCFileVideoCapturerErrorBlock)(NSError *error);
>- RTC_EXPORT
>- 
>- NS_CLASS_AVAILABLE_IOS(10)
>-+__attribute__((objc_runtime_name("WK_RTCFileVideoCapturer")))
>- @interface RTCFileVideoCapturer : RTCVideoCapturer
>- 
>- /**
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCIceCandidate.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCIceCandidate.h
>-index 3b7f3e26e5a..902925203ea 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCIceCandidate.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCIceCandidate.h
>-@@ -15,6 +15,7 @@
>- NS_ASSUME_NONNULL_BEGIN
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCIceCandidate")))
>- @interface RTCIceCandidate : NSObject
>- 
>- /**
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCIceServer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCIceServer.h
>-index 727da8acbeb..d1feda40fe7 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCIceServer.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCIceServer.h
>-@@ -20,6 +20,7 @@ typedef NS_ENUM(NSUInteger, RTCTlsCertPolicy) {
>- NS_ASSUME_NONNULL_BEGIN
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCIceServer")))
>- @interface RTCIceServer : NSObject
>- 
>- /** URI(s) for this server represented as NSStrings. */
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCIntervalRange.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCIntervalRange.h
>-index 9384b153683..11eb4dc6233 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCIntervalRange.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCIntervalRange.h
>-@@ -12,6 +12,7 @@
>- 
>- NS_ASSUME_NONNULL_BEGIN
>- 
>-+__attribute__((objc_runtime_name("WK_RTCIntervalRange")))
>- @interface RTCIntervalRange : NSObject
>- 
>- @property(nonatomic, readonly) NSInteger min;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCLegacyStatsReport.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCLegacyStatsReport.h
>-index 14d996c1cec..bbc4c035f78 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCLegacyStatsReport.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCLegacyStatsReport.h
>-@@ -16,6 +16,7 @@ NS_ASSUME_NONNULL_BEGIN
>- 
>- /** This does not currently conform to the spec. */
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCLegacyStatsReport")))
>- @interface RTCLegacyStatsReport : NSObject
>- 
>- /** Time since 1970-01-01T00:00:00Z in milliseconds. */
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMTLNSVideoView.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMTLNSVideoView.h
>-index 4990e2529ef..b18fc777402 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMTLNSVideoView.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMTLNSVideoView.h
>-@@ -12,6 +12,7 @@
>- #import "WebRTC/RTCVideoRenderer.h"
>- 
>- NS_AVAILABLE_MAC(10.11)
>-+__attribute__((objc_runtime_name("WK_RTCMTLNSVideoView")))
>- @interface RTCMTLNSVideoView : NSView<RTCVideoRenderer>
>- + (BOOL)isMetalAvailable;
>- @end
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMTLVideoView.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMTLVideoView.h
>-index 226c62a6a17..b62d0889b06 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMTLVideoView.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMTLVideoView.h
>-@@ -31,6 +31,7 @@ NS_ASSUME_NONNULL_BEGIN
>- NS_CLASS_AVAILABLE_IOS(9)
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCMTLVideoView")))
>- @interface RTCMTLVideoView : UIView <RTCVideoRenderer>
>- 
>- @end
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaConstraints.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaConstraints.h
>-index 26a116888a4..e6321940c44 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaConstraints.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaConstraints.h
>-@@ -40,6 +40,7 @@ RTC_EXTERN NSString * const kRTCMediaConstraintsValueTrue;
>- RTC_EXTERN NSString * const kRTCMediaConstraintsValueFalse;
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCMediaConstraints")))
>- @interface RTCMediaConstraints : NSObject
>- 
>- - (instancetype)init NS_UNAVAILABLE;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaSource.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaSource.h
>-index 6cfcb7d2ce8..1a4003b5283 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaSource.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaSource.h
>-@@ -22,6 +22,7 @@ typedef NS_ENUM(NSInteger, RTCSourceState) {
>- NS_ASSUME_NONNULL_BEGIN
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCMediaSource")))
>- @interface RTCMediaSource : NSObject
>- 
>- /** The current state of the RTCMediaSource. */
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaStream.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaStream.h
>-index b97960d436c..9cefde13c5f 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaStream.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaStream.h
>-@@ -19,6 +19,7 @@ NS_ASSUME_NONNULL_BEGIN
>- @class RTCVideoTrack;
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCMediaStream")))
>- @interface RTCMediaStream : NSObject
>- 
>- /** The audio tracks in this stream. */
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaStreamTrack.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaStreamTrack.h
>-index c42f0cfb1a0..a1161c2ac2c 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaStreamTrack.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMediaStreamTrack.h
>-@@ -26,6 +26,7 @@ RTC_EXTERN NSString * const kRTCMediaStreamTrackKindAudio;
>- RTC_EXTERN NSString * const kRTCMediaStreamTrackKindVideo;
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCMediaStreamTrack")))
>- @interface RTCMediaStreamTrack : NSObject
>- 
>- /**
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMetricsSampleInfo.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMetricsSampleInfo.h
>-index f01bea9b71b..28b71d1f9e4 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMetricsSampleInfo.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCMetricsSampleInfo.h
>-@@ -15,6 +15,7 @@
>- NS_ASSUME_NONNULL_BEGIN
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCMetricsSampleInfo")))
>- @interface RTCMetricsSampleInfo : NSObject
>- 
>- /**
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCNSGLVideoView.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCNSGLVideoView.h
>-index 9e6a9ce9483..e29ec9f3c9d 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCNSGLVideoView.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCNSGLVideoView.h
>-@@ -20,12 +20,14 @@
>- NS_ASSUME_NONNULL_BEGIN
>- 
>- @class RTCNSGLVideoView;
>-+__attribute__((objc_runtime_name("WK_RTCNSGLVideoViewDelegate")))
>- @protocol RTCNSGLVideoViewDelegate
>- 
>- - (void)videoView:(RTCNSGLVideoView *)videoView didChangeVideoSize:(CGSize)size;
>- 
>- @end
>- 
>-+__attribute__((objc_runtime_name("WK_RTCNSGLVideoView")))
>- @interface RTCNSGLVideoView : NSOpenGLView <RTCVideoRenderer>
>- 
>- @property(nonatomic, weak) id<RTCNSGLVideoViewDelegate> delegate;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnection.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnection.h
>-index a4c113b1d65..041232e341b 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnection.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnection.h
>-@@ -69,6 +69,7 @@ typedef NS_ENUM(NSInteger, RTCStatsOutputLevel) {
>- @class RTCPeerConnection;
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCPeerConnectionDelegate")))
>- @protocol RTCPeerConnectionDelegate <NSObject>
>- 
>- /** Called when the SignalingState changed. */
>-@@ -109,6 +110,7 @@ RTC_EXPORT
>- @end
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCPeerConnection")))
>- @interface RTCPeerConnection : NSObject
>- 
>- /** The object that will be notifed about events such as state changes and
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnectionFactory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnectionFactory.h
>-index 56c399e27a8..8acaf43ce26 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnectionFactory.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnectionFactory.h
>-@@ -24,11 +24,15 @@ NS_ASSUME_NONNULL_BEGIN
>- @class RTCVideoSource;
>- @class RTCVideoTrack;
>- @class RTCPeerConnectionFactoryOptions;
>-+__attribute__((objc_runtime_name("WK_RTCPeerConnectionDelegate")))
>- @protocol RTCPeerConnectionDelegate;
>-+__attribute__((objc_runtime_name("WK_RTCVideoDecoderFactory")))
>- @protocol RTCVideoDecoderFactory;
>-+__attribute__((objc_runtime_name("WK_RTCVideoEncoderFactory")))
>- @protocol RTCVideoEncoderFactory;
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCPeerConnectionFactory")))
>- @interface RTCPeerConnectionFactory : NSObject
>- 
>- /* Initialize object with default H264 video encoder/decoder factories */
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnectionFactoryOptions.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnectionFactoryOptions.h
>-index a65abc6c8b5..9a91ae218ae 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnectionFactoryOptions.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCPeerConnectionFactoryOptions.h
>-@@ -15,6 +15,7 @@
>- NS_ASSUME_NONNULL_BEGIN
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCPeerConnectionFactoryOptions")))
>- @interface RTCPeerConnectionFactoryOptions : NSObject
>- 
>- @property(nonatomic, assign) BOOL disableEncryption;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpCodecParameters.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpCodecParameters.h
>-index c7ca2f59c00..2725e53d31e 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpCodecParameters.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpCodecParameters.h
>-@@ -33,6 +33,7 @@ RTC_EXTERN const NSString * const kRTCH264CodecName;
>- 
>- /** Defined in http://w3c.github.io/webrtc-pc/#idl-def-RTCRtpCodecParameters */
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCRtpCodecParameters")))
>- @interface RTCRtpCodecParameters : NSObject
>- 
>- /** The RTP payload type. */
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpEncodingParameters.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpEncodingParameters.h
>-index a956f0d4c37..4d2e1e67152 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpEncodingParameters.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpEncodingParameters.h
>-@@ -15,6 +15,7 @@
>- NS_ASSUME_NONNULL_BEGIN
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCRtpEncodingParameters")))
>- @interface RTCRtpEncodingParameters : NSObject
>- 
>- /** Controls whether the encoding is currently transmitted. */
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpParameters.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpParameters.h
>-index bdebf84884c..1972749d9a1 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpParameters.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpParameters.h
>-@@ -17,6 +17,7 @@
>- NS_ASSUME_NONNULL_BEGIN
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCRtpParameters")))
>- @interface RTCRtpParameters : NSObject
>- 
>- /** The currently active encodings in the order of preference. */
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpReceiver.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpReceiver.h
>-index 6156db9f932..324271876b7 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpReceiver.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpReceiver.h
>-@@ -26,6 +26,7 @@ typedef NS_ENUM(NSInteger, RTCRtpMediaType) {
>- @class RTCRtpReceiver;
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCRtpReceiverDelegate")))
>- @protocol RTCRtpReceiverDelegate <NSObject>
>- 
>- /** Called when the first RTP packet is received.
>-@@ -46,6 +47,7 @@ RTC_EXPORT
>- @end
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCRtpReceiver")))
>- @protocol RTCRtpReceiver <NSObject>
>- 
>- /** A unique identifier for this receiver. */
>-@@ -73,6 +75,7 @@ RTC_EXPORT
>- @end
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCRtpReceiver")))
>- @interface RTCRtpReceiver : NSObject <RTCRtpReceiver>
>- 
>- - (instancetype)init NS_UNAVAILABLE;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpSender.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpSender.h
>-index f6e91250267..bf2fbc6ac51 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpSender.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCRtpSender.h
>-@@ -18,6 +18,7 @@
>- NS_ASSUME_NONNULL_BEGIN
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCRtpSender")))
>- @protocol RTCRtpSender <NSObject>
>- 
>- /** A unique identifier for this sender. */
>-@@ -41,6 +42,7 @@ RTC_EXPORT
>- @end
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCRtpSender")))
>- @interface RTCRtpSender : NSObject <RTCRtpSender>
>- 
>- - (instancetype)init NS_UNAVAILABLE;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCSessionDescription.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCSessionDescription.h
>-index 41439804a8a..ba05e22066f 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCSessionDescription.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCSessionDescription.h
>-@@ -25,6 +25,7 @@ typedef NS_ENUM(NSInteger, RTCSdpType) {
>- NS_ASSUME_NONNULL_BEGIN
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCSessionDescription")))
>- @interface RTCSessionDescription : NSObject
>- 
>- /** The type of session description. */
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCapturer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCapturer.h
>-index 3addec83b1d..72215872bb9 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCapturer.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCapturer.h
>-@@ -15,11 +15,13 @@ NS_ASSUME_NONNULL_BEGIN
>- @class RTCVideoCapturer;
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoCapturerDelegate")))
>- @protocol RTCVideoCapturerDelegate <NSObject>
>- - (void)capturer:(RTCVideoCapturer *)capturer didCaptureVideoFrame:(RTCVideoFrame *)frame;
>- @end
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoCapturer")))
>- @interface RTCVideoCapturer : NSObject
>- 
>- @property(nonatomic, readonly, weak) id<RTCVideoCapturerDelegate> delegate;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodec.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodec.h
>-index 470eb023260..345c22e53c2 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodec.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodec.h
>-@@ -37,6 +37,7 @@ typedef NS_ENUM(NSUInteger, RTCVideoContentType) {
>- 
>- /** Represents an encoded frame. Corresponds to webrtc::EncodedImage. */
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCEncodedImage")))
>- @interface RTCEncodedImage : NSObject
>- 
>- @property(nonatomic, strong) NSData *buffer;
>-@@ -58,6 +59,7 @@ RTC_EXPORT
>- 
>- /** Information for header. Corresponds to webrtc::RTPFragmentationHeader. */
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCRtpFragmentationHeader")))
>- @interface RTCRtpFragmentationHeader : NSObject
>- 
>- @property(nonatomic, strong) NSArray<NSNumber *> *fragmentationOffset;
>-@@ -71,6 +73,7 @@ RTC_EXPORT
>-  *  Corresponds to webrtc::CodecSpecificInfo.
>-  */
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCCodecSpecificInfo")))
>- @protocol RTCCodecSpecificInfo <NSObject>
>- 
>- @end
>-@@ -90,6 +93,7 @@ typedef NS_ENUM(NSUInteger, RTCVideoCodecMode) {
>- 
>- /** Holds information to identify a codec. Corresponds to cricket::VideoCodec. */
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoCodecInfo")))
>- @interface RTCVideoCodecInfo : NSObject <NSCoding>
>- 
>- - (instancetype)init NS_UNAVAILABLE;
>-@@ -109,6 +113,7 @@ RTC_EXPORT
>- 
>- /** Settings for encoder. Corresponds to webrtc::VideoCodec. */
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoEncoderSettings")))
>- @interface RTCVideoEncoderSettings : NSObject
>- 
>- @property(nonatomic, strong) NSString *name;
>-@@ -130,6 +135,7 @@ RTC_EXPORT
>- 
>- /** QP thresholds for encoder. Corresponds to webrtc::VideoEncoder::QpThresholds. */
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoEncoderQpThresholds")))
>- @interface RTCVideoEncoderQpThresholds : NSObject
>- 
>- - (instancetype)initWithThresholdsLow:(NSInteger)low high:(NSInteger)high;
>-@@ -141,6 +147,7 @@ RTC_EXPORT
>- 
>- /** Protocol for encoder implementations. */
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoEncoder")))
>- @protocol RTCVideoEncoder <NSObject>
>- 
>- - (void)setCallback:(RTCVideoEncoderCallback)callback;
>-@@ -162,6 +169,7 @@ RTC_EXPORT
>- 
>- /** Protocol for decoder implementations. */
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoDecoder")))
>- @protocol RTCVideoDecoder <NSObject>
>- 
>- - (void)setCallback:(RTCVideoDecoderCallback)callback;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodecFactory.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodecFactory.h
>-index 0ef2c8a655c..9fb1cb71d99 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodecFactory.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodecFactory.h
>-@@ -17,6 +17,7 @@ NS_ASSUME_NONNULL_BEGIN
>- 
>- /** RTCVideoEncoderFactory is an Objective-C version of webrtc::VideoEncoderFactory. */
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoEncoderFactory")))
>- @protocol RTCVideoEncoderFactory <NSObject>
>- 
>- - (nullable id<RTCVideoEncoder>)createEncoder:(RTCVideoCodecInfo *)info;
>-@@ -26,6 +27,7 @@ RTC_EXPORT
>- 
>- /** RTCVideoDecoderFactory is an Objective-C version of webrtc::VideoDecoderFactory. */
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoDecoderFactory")))
>- @protocol RTCVideoDecoderFactory <NSObject>
>- 
>- - (nullable id<RTCVideoDecoder>)createDecoder:(RTCVideoCodecInfo *)info;
>-@@ -39,6 +41,7 @@ RTC_EXPORT
>-  *  codecs, create custom implementations of RTCVideoEncoderFactory and RTCVideoDecoderFactory.
>-  */
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCDefaultVideoEncoderFactory")))
>- @interface RTCDefaultVideoEncoderFactory : NSObject <RTCVideoEncoderFactory>
>- 
>- @property(nonatomic, retain) RTCVideoCodecInfo *preferredCodec;
>-@@ -48,6 +51,7 @@ RTC_EXPORT
>- @end
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCDefaultVideoDecoderFactory")))
>- @interface RTCDefaultVideoDecoderFactory : NSObject <RTCVideoDecoderFactory>
>- @end
>- 
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodecH264.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodecH264.h
>-index 1c307ebe84a..c8d7fc007f1 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodecH264.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoCodecH264.h
>-@@ -20,6 +20,7 @@ typedef NS_ENUM(NSUInteger, RTCH264PacketizationMode) {
>- };
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCCodecSpecificInfoH264")))
>- @interface RTCCodecSpecificInfoH264 : NSObject <RTCCodecSpecificInfo>
>- 
>- @property(nonatomic, assign) RTCH264PacketizationMode packetizationMode;
>-@@ -28,6 +29,7 @@ RTC_EXPORT
>- 
>- /** Encoder. */
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoEncoderH264")))
>- @interface RTCVideoEncoderH264 : NSObject<RTCVideoEncoder>
>- 
>- - (instancetype)initWithCodecInfo:(RTCVideoCodecInfo *)codecInfo;
>-@@ -36,15 +38,18 @@ RTC_EXPORT
>- 
>- /** Decoder. */
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoDecoderH264")))
>- @interface RTCVideoDecoderH264 : NSObject<RTCVideoDecoder>
>- @end
>- 
>- /** Encoder factory. */
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoEncoderFactoryH264")))
>- @interface RTCVideoEncoderFactoryH264 : NSObject<RTCVideoEncoderFactory>
>- @end
>- 
>- /** Decoder factory. */
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoDecoderFactoryH264")))
>- @interface RTCVideoDecoderFactoryH264 : NSObject<RTCVideoDecoderFactory>
>- @end
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoDecoderVP8.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoDecoderVP8.h
>-index 962b6312f7f..3e19ef8909c 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoDecoderVP8.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoDecoderVP8.h
>-@@ -14,6 +14,7 @@
>- #import <WebRTC/RTCVideoCodec.h>
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoDecoderVP8")))
>- @interface RTCVideoDecoderVP8 : NSObject
>- 
>- /* This returns a VP8 decoder that can be returned from a RTCVideoDecoderFactory injected into
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoDecoderVP9.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoDecoderVP9.h
>-index faf66ef065b..3fe79bc4a52 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoDecoderVP9.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoDecoderVP9.h
>-@@ -14,6 +14,7 @@
>- #import <WebRTC/RTCVideoCodec.h>
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoDecoderVP9")))
>- @interface RTCVideoDecoderVP9 : NSObject
>- 
>- /* This returns a VP9 decoder that can be returned from a RTCVideoDecoderFactory injected into
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoEncoderVP8.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoEncoderVP8.h
>-index e63b24dde75..52155b52551 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoEncoderVP8.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoEncoderVP8.h
>-@@ -14,6 +14,7 @@
>- #import <WebRTC/RTCVideoCodec.h>
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoEncoderVP8")))
>- @interface RTCVideoEncoderVP8 : NSObject
>- 
>- /* This returns a VP8 encoder that can be returned from a RTCVideoEncoderFactory injected into
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoEncoderVP9.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoEncoderVP9.h
>-index cef8c82e96f..866e626a05e 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoEncoderVP9.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoEncoderVP9.h
>-@@ -14,6 +14,7 @@
>- #import <WebRTC/RTCVideoCodec.h>
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoEncoderVP9")))
>- @interface RTCVideoEncoderVP9 : NSObject
>- 
>- /* This returns a VP9 encoder that can be returned from a RTCVideoEncoderFactory injected into
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoFrame.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoFrame.h
>-index dcc4c6e7b5b..5b8fba602df 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoFrame.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoFrame.h
>-@@ -22,10 +22,12 @@ typedef NS_ENUM(NSInteger, RTCVideoRotation) {
>-   RTCVideoRotation_270 = 270,
>- };
>- 
>-+__attribute__((objc_runtime_name("WK_RTCVideoFrameBuffer")))
>- @protocol RTCVideoFrameBuffer;
>- 
>- // RTCVideoFrame is an ObjectiveC version of webrtc::VideoFrame.
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoFrame")))
>- @interface RTCVideoFrame : NSObject
>- 
>- /** Width without rotation applied. */
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoFrameBuffer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoFrameBuffer.h
>-index 4a683b0c926..8b825d8050c 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoFrameBuffer.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoFrameBuffer.h
>-@@ -13,10 +13,12 @@
>- 
>- NS_ASSUME_NONNULL_BEGIN
>- 
>-+__attribute__((objc_runtime_name("WK_RTCI420Buffer")))
>- @protocol RTCI420Buffer;
>- 
>- // RTCVideoFrameBuffer is an ObjectiveC version of webrtc::VideoFrameBuffer.
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoFrameBuffer")))
>- @protocol RTCVideoFrameBuffer <NSObject>
>- 
>- @property(nonatomic, readonly) int width;
>-@@ -27,6 +29,7 @@ RTC_EXPORT
>- @end
>- 
>- /** Protocol for RTCVideoFrameBuffers containing YUV planar data. */
>-+__attribute__((objc_runtime_name("WK_RTCYUVPlanarBuffer")))
>- @protocol RTCYUVPlanarBuffer <RTCVideoFrameBuffer>
>- 
>- @property(nonatomic, readonly) int chromaWidth;
>-@@ -48,6 +51,7 @@ RTC_EXPORT
>- @end
>- 
>- /** Extension of the YUV planar data buffer with mutable data access */
>-+__attribute__((objc_runtime_name("WK_RTCMutableYUVPlanarBuffer")))
>- @protocol RTCMutableYUVPlanarBuffer <RTCYUVPlanarBuffer>
>- 
>- @property(nonatomic, readonly) uint8_t *mutableDataY;
>-@@ -57,15 +61,18 @@ RTC_EXPORT
>- @end
>- 
>- /** Protocol for RTCYUVPlanarBuffers containing I420 data */
>-+__attribute__((objc_runtime_name("WK_RTCI420Buffer")))
>- @protocol RTCI420Buffer <RTCYUVPlanarBuffer>
>- @end
>- 
>- /** Extension of the I420 buffer with mutable data access */
>-+__attribute__((objc_runtime_name("WK_RTCMutableI420Buffer")))
>- @protocol RTCMutableI420Buffer <RTCI420Buffer, RTCMutableYUVPlanarBuffer>
>- @end
>- 
>- /** RTCVideoFrameBuffer containing a CVPixelBufferRef */
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCCVPixelBuffer")))
>- @interface RTCCVPixelBuffer : NSObject <RTCVideoFrameBuffer>
>- 
>- @property(nonatomic, readonly) CVPixelBufferRef pixelBuffer;
>-@@ -95,11 +102,13 @@ RTC_EXPORT
>- 
>- /** RTCI420Buffer implements the RTCI420Buffer protocol */
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCI420Buffer")))
>- @interface RTCI420Buffer : NSObject <RTCI420Buffer>
>- @end
>- 
>- /** Mutable version of RTCI420Buffer */
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCMutableI420Buffer")))
>- @interface RTCMutableI420Buffer : RTCI420Buffer <RTCMutableI420Buffer>
>- @end
>- 
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoRenderer.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoRenderer.h
>-index 5e2e820918a..c821a1e3008 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoRenderer.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoRenderer.h
>-@@ -20,6 +20,7 @@ NS_ASSUME_NONNULL_BEGIN
>- @class RTCVideoFrame;
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoRenderer")))
>- @protocol RTCVideoRenderer <NSObject>
>- 
>- /** The size of the frame. */
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoSource.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoSource.h
>-index a9ebc06c88e..60075e3435a 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoSource.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoSource.h
>-@@ -18,6 +18,7 @@ NS_ASSUME_NONNULL_BEGIN
>- 
>- RTC_EXPORT
>- 
>-+__attribute__((objc_runtime_name("WK_RTCVideoSource")))
>- @interface RTCVideoSource : RTCMediaSource <RTCVideoCapturerDelegate>
>- 
>- - (instancetype)init NS_UNAVAILABLE;
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoTrack.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoTrack.h
>-index 899d7c34780..ecbdbb93c68 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoTrack.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoTrack.h
>-@@ -14,11 +14,13 @@
>- 
>- NS_ASSUME_NONNULL_BEGIN
>- 
>-+__attribute__((objc_runtime_name("WK_RTCVideoRenderer")))
>- @protocol RTCVideoRenderer;
>- @class RTCPeerConnectionFactory;
>- @class RTCVideoSource;
>- 
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoTrack")))
>- @interface RTCVideoTrack : RTCMediaStreamTrack
>- 
>- /** The video source for this video track. */
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoViewShading.h b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoViewShading.h
>-index 6f3ece94a1f..5e33ff235ca 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoViewShading.h
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/sdk/objc/Framework/Headers/WebRTC/RTCVideoViewShading.h
>-@@ -19,6 +19,7 @@ NS_ASSUME_NONNULL_BEGIN
>-  * rendering for the RTCEAGLVideoView/RTCNSGLVideoView.
>-  */
>- RTC_EXPORT
>-+__attribute__((objc_runtime_name("WK_RTCVideoViewShading")))
>- @protocol RTCVideoViewShading <NSObject>
>- 
>- /** Callback for I420 frames. Each plane is given as a texture. */
>diff --git a/Source/ThirdParty/libwebrtc/WebKit/0013-Fix-SafeSetError-use-after-move.patch b/Source/ThirdParty/libwebrtc/WebKit/0013-Fix-SafeSetError-use-after-move.patch
>deleted file mode 100644
>index db7ce41afe6..00000000000
>--- a/Source/ThirdParty/libwebrtc/WebKit/0013-Fix-SafeSetError-use-after-move.patch
>+++ /dev/null
>@@ -1,27 +0,0 @@
>-diff --git a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection.cc b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection.cc
>-index 6281d57e3db..1bebb0509a1 100644
>---- a/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection.cc
>-+++ b/Source/ThirdParty/libwebrtc/Source/webrtc/pc/peerconnection.cc
>-@@ -226,18 +226,19 @@ uint32_t ConvertIceTransportTypeToCandidateFilter(
>- }
>- 
>- // Helper to set an error and return from a method.
>--bool SafeSetError(webrtc::RTCErrorType type, webrtc::RTCError* error) {
>-+static bool SafeSetError(webrtc::RTCErrorType type, webrtc::RTCError* error) {
>-   if (error) {
>-     error->set_type(type);
>-   }
>-   return type == webrtc::RTCErrorType::NONE;
>- }
>- 
>--bool SafeSetError(webrtc::RTCError error, webrtc::RTCError* error_out) {
>-+static bool SafeSetError(webrtc::RTCError&& error, webrtc::RTCError* error_out) {
>-+  bool result = error.ok();
>-   if (error_out) {
>-     *error_out = std::move(error);
>-   }
>--  return error.ok();
>-+  return result;
>- }
>- 
>- std::string GetSignalingStateString(
>diff --git a/Source/ThirdParty/libwebrtc/libwebrtc.xcodeproj/project.pbxproj b/Source/ThirdParty/libwebrtc/libwebrtc.xcodeproj/project.pbxproj
>index df2a5f92b34..28d8367d8be 100644
>--- a/Source/ThirdParty/libwebrtc/libwebrtc.xcodeproj/project.pbxproj
>+++ b/Source/ThirdParty/libwebrtc/libwebrtc.xcodeproj/project.pbxproj
>@@ -21,6 +21,51 @@
> /* End PBXAggregateTarget section */
> 
> /* Begin PBXBuildFile section */
>+		4102F68D21273206006AE8D7 /* rtcstatstraversal.h in Headers */ = {isa = PBXBuildFile; fileRef = 4102F68421273204006AE8D7 /* rtcstatstraversal.h */; };
>+		4102F68E21273206006AE8D7 /* jseptransportcontroller.h in Headers */ = {isa = PBXBuildFile; fileRef = 4102F68521273204006AE8D7 /* jseptransportcontroller.h */; };
>+		4102F68F21273206006AE8D7 /* transportstats.cc in Sources */ = {isa = PBXBuildFile; fileRef = 4102F68621273205006AE8D7 /* transportstats.cc */; };
>+		4102F69021273206006AE8D7 /* jseptransportcontroller.cc in Sources */ = {isa = PBXBuildFile; fileRef = 4102F68721273205006AE8D7 /* jseptransportcontroller.cc */; };
>+		4102F69121273206006AE8D7 /* rtpparametersconversion.h in Headers */ = {isa = PBXBuildFile; fileRef = 4102F68821273205006AE8D7 /* rtpparametersconversion.h */; };
>+		4102F69221273206006AE8D7 /* rtcstatstraversal.cc in Sources */ = {isa = PBXBuildFile; fileRef = 4102F68921273205006AE8D7 /* rtcstatstraversal.cc */; };
>+		4102F69321273206006AE8D7 /* rtpparametersconversion.cc in Sources */ = {isa = PBXBuildFile; fileRef = 4102F68A21273206006AE8D7 /* rtpparametersconversion.cc */; };
>+		4102F69421273206006AE8D7 /* transportstats.h in Headers */ = {isa = PBXBuildFile; fileRef = 4102F68B21273206006AE8D7 /* transportstats.h */; };
>+		4102F69521273206006AE8D7 /* peerconnectioninternal.h in Headers */ = {isa = PBXBuildFile; fileRef = 4102F68C21273206006AE8D7 /* peerconnectioninternal.h */; };
>+		4102F6A021273262006AE8D7 /* transport_feedback_packet_loss_tracker.h in Headers */ = {isa = PBXBuildFile; fileRef = 4102F69621273260006AE8D7 /* transport_feedback_packet_loss_tracker.h */; };
>+		4102F6A121273262006AE8D7 /* remix_resample.cc in Sources */ = {isa = PBXBuildFile; fileRef = 4102F69721273260006AE8D7 /* remix_resample.cc */; };
>+		4102F6A221273262006AE8D7 /* audio_level.h in Headers */ = {isa = PBXBuildFile; fileRef = 4102F69821273260006AE8D7 /* audio_level.h */; };
>+		4102F6A321273262006AE8D7 /* channel_proxy.h in Headers */ = {isa = PBXBuildFile; fileRef = 4102F69921273261006AE8D7 /* channel_proxy.h */; };
>+		4102F6A421273262006AE8D7 /* audio_level.cc in Sources */ = {isa = PBXBuildFile; fileRef = 4102F69A21273261006AE8D7 /* audio_level.cc */; };
>+		4102F6A521273262006AE8D7 /* channel.h in Headers */ = {isa = PBXBuildFile; fileRef = 4102F69B21273261006AE8D7 /* channel.h */; };
>+		4102F6A621273262006AE8D7 /* channel_proxy.cc in Sources */ = {isa = PBXBuildFile; fileRef = 4102F69C21273261006AE8D7 /* channel_proxy.cc */; };
>+		4102F6A721273262006AE8D7 /* remix_resample.h in Headers */ = {isa = PBXBuildFile; fileRef = 4102F69D21273261006AE8D7 /* remix_resample.h */; };
>+		4102F6A821273262006AE8D7 /* channel.cc in Sources */ = {isa = PBXBuildFile; fileRef = 4102F69E21273262006AE8D7 /* channel.cc */; };
>+		4102F6A921273262006AE8D7 /* transport_feedback_packet_loss_tracker.cc in Sources */ = {isa = PBXBuildFile; fileRef = 4102F69F21273262006AE8D7 /* transport_feedback_packet_loss_tracker.cc */; };
>+		4102F6AB212732B1006AE8D7 /* rtp_payload_params.cc in Sources */ = {isa = PBXBuildFile; fileRef = 4102F6AA212732B0006AE8D7 /* rtp_payload_params.cc */; };
>+		4102F6AE212732E7006AE8D7 /* expand_uma_logger.cc in Sources */ = {isa = PBXBuildFile; fileRef = 4102F6AC212732E6006AE8D7 /* expand_uma_logger.cc */; };
>+		4102F6AF212732E7006AE8D7 /* expand_uma_logger.h in Headers */ = {isa = PBXBuildFile; fileRef = 4102F6AD212732E7006AE8D7 /* expand_uma_logger.h */; };
>+		4102F6B221273331006AE8D7 /* mediaconfig.h in Headers */ = {isa = PBXBuildFile; fileRef = 4102F6B021273330006AE8D7 /* mediaconfig.h */; };
>+		4102F6B321273331006AE8D7 /* mediachannel.cc in Sources */ = {isa = PBXBuildFile; fileRef = 4102F6B121273330006AE8D7 /* mediachannel.cc */; };
>+		4102F6B921273382006AE8D7 /* packet_receiver.h in Headers */ = {isa = PBXBuildFile; fileRef = 4102F6B521273381006AE8D7 /* packet_receiver.h */; };
>+		4102F6BA21273382006AE8D7 /* audio_receive_stream.cc in Sources */ = {isa = PBXBuildFile; fileRef = 4102F6B621273381006AE8D7 /* audio_receive_stream.cc */; };
>+		4102F6BB21273382006AE8D7 /* degraded_call.cc in Sources */ = {isa = PBXBuildFile; fileRef = 4102F6B721273381006AE8D7 /* degraded_call.cc */; };
>+		4102F6BC21273382006AE8D7 /* flexfec_receive_stream.cc in Sources */ = {isa = PBXBuildFile; fileRef = 4102F6B821273381006AE8D7 /* flexfec_receive_stream.cc */; };
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>+		4102F6C9212733B7006AE8D7 /* video_quality_observer.cc in Sources */ = {isa = PBXBuildFile; fileRef = 4102F6BF212733B6006AE8D7 /* video_quality_observer.cc */; };
>+		4102F6CA212733B7006AE8D7 /* video_stream_encoder.h in Headers */ = {isa = PBXBuildFile; fileRef = 4102F6C0212733B6006AE8D7 /* video_stream_encoder.h */; };
>+		4102F6CB212733B7006AE8D7 /* video_stream_decoder_impl.cc in Sources */ = {isa = PBXBuildFile; fileRef = 4102F6C1212733B6006AE8D7 /* video_stream_decoder_impl.cc */; };
>+		4102F6CC212733B7006AE8D7 /* video_send_stream_impl.cc in Sources */ = {isa = PBXBuildFile; fileRef = 4102F6C2212733B6006AE8D7 /* video_send_stream_impl.cc */; };
>+		4102F6CE212733B7006AE8D7 /* video_stream_decoder_impl.h in Headers */ = {isa = PBXBuildFile; fileRef = 4102F6C4212733B6006AE8D7 /* video_stream_decoder_impl.h */; };
>+		4102F6CF212733B7006AE8D7 /* video_quality_observer.h in Headers */ = {isa = PBXBuildFile; fileRef = 4102F6C5212733B6006AE8D7 /* video_quality_observer.h */; };
>+		4102F6DB21273416006AE8D7 /* video_stream_decoder_create.cc in Sources */ = {isa = PBXBuildFile; fileRef = 4102F6D121273414006AE8D7 /* video_stream_decoder_create.cc */; };
>+		4102F6DC21273416006AE8D7 /* color_space.cc in Sources */ = {isa = PBXBuildFile; fileRef = 4102F6D221273414006AE8D7 /* color_space.cc */; };
>+		4102F6DD21273416006AE8D7 /* video_sink_interface.h in Headers */ = {isa = PBXBuildFile; fileRef = 4102F6D321273415006AE8D7 /* video_sink_interface.h */; };
>+		4102F6DE21273416006AE8D7 /* video_stream_decoder_create.h in Headers */ = {isa = PBXBuildFile; fileRef = 4102F6D421273415006AE8D7 /* video_stream_decoder_create.h */; };
>+		4102F6DF21273416006AE8D7 /* video_stream_encoder_create.cc in Sources */ = {isa = PBXBuildFile; fileRef = 4102F6D521273415006AE8D7 /* video_stream_encoder_create.cc */; };
>+		4102F6E021273416006AE8D7 /* i010_buffer.cc in Sources */ = {isa = PBXBuildFile; fileRef = 4102F6D621273415006AE8D7 /* i010_buffer.cc */; };
>+		4102F6E121273416006AE8D7 /* video_stream_encoder_settings.h in Headers */ = {isa = PBXBuildFile; fileRef = 4102F6D721273415006AE8D7 /* video_stream_encoder_settings.h */; };
>+		4102F6E221273416006AE8D7 /* color_space.h in Headers */ = {isa = PBXBuildFile; fileRef = 4102F6D821273416006AE8D7 /* color_space.h */; };
>+		4102F6E321273416006AE8D7 /* video_bitrate_allocation.cc in Sources */ = {isa = PBXBuildFile; fileRef = 4102F6D921273416006AE8D7 /* video_bitrate_allocation.cc */; };
>+		4102F6E421273416006AE8D7 /* video_bitrate_allocator_factory.h in Headers */ = {isa = PBXBuildFile; fileRef = 4102F6DA21273416006AE8D7 /* video_bitrate_allocator_factory.h */; };
> 		41109AAD1E5FA19200C0955A /* i420_buffer_pool.h in Headers */ = {isa = PBXBuildFile; fileRef = 41109AA61E5FA19200C0955A /* i420_buffer_pool.h */; };
> 		41109AAE1E5FA19200C0955A /* video_frame_buffer.h in Headers */ = {isa = PBXBuildFile; fileRef = 41109AA71E5FA19200C0955A /* video_frame_buffer.h */; };
> 		41109AAF1E5FA19200C0955A /* frame_callback.h in Headers */ = {isa = PBXBuildFile; fileRef = 41109AA81E5FA19200C0955A /* frame_callback.h */; };
>@@ -31,6 +76,14 @@
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> 		412455571EF887FB00F11809 /* packetlossestimator.h in Headers */ = {isa = PBXBuildFile; fileRef = 412455551EF887FA00F11809 /* packetlossestimator.h */; };
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>+		41299B932127367B00B3414B /* pitch_filter.h in Headers */ = {isa = PBXBuildFile; fileRef = 41299B902127367B00B3414B /* pitch_filter.h */; };
>+		41299B962127369100B3414B /* fake_network_pipe.cc in Sources */ = {isa = PBXBuildFile; fileRef = 41299B942127369000B3414B /* fake_network_pipe.cc */; };
>+		41299B972127369100B3414B /* fake_network_pipe.h in Headers */ = {isa = PBXBuildFile; fileRef = 41299B952127369100B3414B /* fake_network_pipe.h */; };
> 		413091F31EF8CF9200757C55 /* rtp_video_stream_receiver.h in Headers */ = {isa = PBXBuildFile; fileRef = 413091F11EF8CF9100757C55 /* rtp_video_stream_receiver.h */; };
> 		413091F41EF8CF9200757C55 /* rtp_video_stream_receiver.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413091F21EF8CF9100757C55 /* rtp_video_stream_receiver.cc */; };
> 		413091F71EF8CFBD00757C55 /* iceserverparsing.h in Headers */ = {isa = PBXBuildFile; fileRef = 413091F51EF8CFBD00757C55 /* iceserverparsing.h */; };
>@@ -40,21 +93,14 @@
> 		413091FE1EF8D06D00757C55 /* receive_side_congestion_controller.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413091FD1EF8D06D00757C55 /* receive_side_congestion_controller.cc */; };
> 		413092011EF8D0A600757C55 /* rtp_demuxer.h in Headers */ = {isa = PBXBuildFile; fileRef = 413091FF1EF8D0A400757C55 /* rtp_demuxer.h */; };
> 		413092021EF8D0A600757C55 /* rtp_demuxer.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413092001EF8D0A600757C55 /* rtp_demuxer.cc */; };
>-		413092051EF8D0C900757C55 /* audio_level.h in Headers */ = {isa = PBXBuildFile; fileRef = 413092031EF8D0C800757C55 /* audio_level.h */; };
>-		413092061EF8D0C900757C55 /* audio_level.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413092041EF8D0C800757C55 /* audio_level.cc */; };
> 		413092091EF8D11200757C55 /* qp_parser.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413092071EF8D11100757C55 /* qp_parser.cc */; };
> 		4130920A1EF8D11200757C55 /* qp_parser.h in Headers */ = {isa = PBXBuildFile; fileRef = 413092081EF8D11100757C55 /* qp_parser.h */; };
>-		4130920E1EF8D40100757C55 /* transport_feedback_packet_loss_tracker.h in Headers */ = {isa = PBXBuildFile; fileRef = 4130920C1EF8D40100757C55 /* transport_feedback_packet_loss_tracker.h */; };
>-		4130920F1EF8D40100757C55 /* transport_feedback_packet_loss_tracker.cc in Sources */ = {isa = PBXBuildFile; fileRef = 4130920D1EF8D40100757C55 /* transport_feedback_packet_loss_tracker.cc */; };
> 		413092121EF8D51100757C55 /* aec_dump.h in Headers */ = {isa = PBXBuildFile; fileRef = 413092101EF8D50C00757C55 /* aec_dump.h */; };
> 		413092131EF8D51100757C55 /* aec_dump.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413092111EF8D51000757C55 /* aec_dump.cc */; };
> 		413092151EF8D5EE00757C55 /* send_side_congestion_controller.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413092141EF8D5ED00757C55 /* send_side_congestion_controller.cc */; };
> 		413092191EF8D63F00757C55 /* rtp_transport_controller_send.h in Headers */ = {isa = PBXBuildFile; fileRef = 413092161EF8D63400757C55 /* rtp_transport_controller_send.h */; };
> 		4130921A1EF8D63F00757C55 /* rtp_transport_controller_send_interface.h in Headers */ = {isa = PBXBuildFile; fileRef = 413092171EF8D63900757C55 /* rtp_transport_controller_send_interface.h */; };
> 		4130921B1EF8D63F00757C55 /* rtp_transport_controller_send.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413092181EF8D63F00757C55 /* rtp_transport_controller_send.cc */; };
>-		413092201EF8D6A200757C55 /* gain_controller2.h in Headers */ = {isa = PBXBuildFile; fileRef = 4130921C1EF8D69000757C55 /* gain_controller2.h */; };
>-		413092231EF8D6A200757C55 /* gain_controller2.cc in Sources */ = {isa = PBXBuildFile; fileRef = 4130921F1EF8D6A200757C55 /* gain_controller2.cc */; };
>-		413092281EF8D6E600757C55 /* builtin_audio_encoder_factory.h in Headers */ = {isa = PBXBuildFile; fileRef = 413092251EF8D6E600757C55 /* builtin_audio_encoder_factory.h */; };
> 		413092301EF8D77100757C55 /* aec_dump_factory.h in Headers */ = {isa = PBXBuildFile; fileRef = 4130922D1EF8D77100757C55 /* aec_dump_factory.h */; };
> 		413A213D1FE0F0EF00373E99 /* srtp_priv.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A21381FE0F0EE00373E99 /* srtp_priv.h */; };
> 		413A213E1FE0F0EF00373E99 /* ekt.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A21391FE0F0EF00373E99 /* ekt.h */; };
>@@ -77,7 +123,6 @@
> 		413A22AC1FE18E0600373E99 /* crc32.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A21511FE18D5200373E99 /* crc32.h */; };
> 		413A22B01FE18E0700373E99 /* asyncinvoker.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A21551FE18D5400373E99 /* asyncinvoker.h */; };
> 		413A22B11FE18E0700373E99 /* messagequeue.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A21561FE18D5400373E99 /* messagequeue.cc */; };
>-		413A22B21FE18E0700373E99 /* md5digest.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A21571FE18D5500373E99 /* md5digest.cc */; };
> 		413A22B31FE18E0700373E99 /* nethelper.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A21591FE18D5600373E99 /* nethelper.cc */; };
> 		413A22B51FE18E0700373E99 /* stringencode.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A215B1FE18D5700373E99 /* stringencode.cc */; };
> 		413A22B61FE18E0700373E99 /* asyncudpsocket.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A215C1FE18D5800373E99 /* asyncudpsocket.h */; };
>@@ -100,7 +145,6 @@
> 		413A22D01FE18E0700373E99 /* platform_file.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A21791FE18D7400373E99 /* platform_file.h */; };
> 		413A22D31FE18E0700373E99 /* socketstream.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A217C1FE18D7700373E99 /* socketstream.cc */; };
> 		413A22D41FE18E0700373E99 /* sequenced_task_checker.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A217E1FE18D7800373E99 /* sequenced_task_checker.h */; };
>-		413A22D51FE18E0700373E99 /* sha1.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A217F1FE18D7900373E99 /* sha1.h */; };
> 		413A22D71FE18E0700373E99 /* file.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A21821FE18D7C00373E99 /* file.h */; };
> 		413A22D81FE18E0700373E99 /* helpers.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A21831FE18D7D00373E99 /* helpers.h */; };
> 		413A22D91FE18E0700373E99 /* natsocketfactory.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A21841FE18D7E00373E99 /* natsocketfactory.h */; };
>@@ -117,16 +161,11 @@
> 		413A22EA1FE18E0700373E99 /* gunit.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A21951FE18D8800373E99 /* gunit.h */; };
> 		413A22EB1FE18E0700373E99 /* stringencode.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A21961FE18D8900373E99 /* stringencode.h */; };
> 		413A22ED1FE18E0700373E99 /* atomicops.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A21981FE18D8A00373E99 /* atomicops.h */; };
>-		413A22EE1FE18E0700373E99 /* ratelimiter.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A21991FE18D8A00373E99 /* ratelimiter.cc */; };
> 		413A22F01FE18E0700373E99 /* socketadapters.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A219B1FE18D8B00373E99 /* socketadapters.cc */; };
>-		413A22F11FE18E0700373E99 /* basictypes.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A219C1FE18D8C00373E99 /* basictypes.h */; };
>-		413A22F51FE18E0700373E99 /* sha1digest.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A21A01FE18D8E00373E99 /* sha1digest.cc */; };
>-		413A22F61FE18E0700373E99 /* base64.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A21A11FE18D8F00373E99 /* base64.h */; };
> 		413A22F81FE18E0700373E99 /* fakesslidentity.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A21A31FE18D9000373E99 /* fakesslidentity.h */; };
> 		413A22F91FE18E0700373E99 /* messagedigest.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A21A41FE18D9000373E99 /* messagedigest.cc */; };
> 		413A22FA1FE18E0700373E99 /* messagedigest.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A21A51FE18D9100373E99 /* messagedigest.h */; };
> 		413A22FB1FE18E0700373E99 /* race_checker.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A21A61FE18D9100373E99 /* race_checker.cc */; };
>-		413A22FC1FE18E0700373E99 /* sha1digest.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A21A71FE18D9100373E99 /* sha1digest.h */; };
> 		413A22FD1FE18E0700373E99 /* stream.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A21A81FE18D9200373E99 /* stream.h */; };
> 		413A22FE1FE18E0700373E99 /* template_util.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A21A91FE18D9200373E99 /* template_util.h */; };
> 		413A23001FE18E0700373E99 /* trace_event.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A21AB1FE18D9300373E99 /* trace_event.h */; };
>@@ -137,7 +176,6 @@
> 		413A23051FE18E0700373E99 /* scoped_ref_ptr.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A21B01FE18D9600373E99 /* scoped_ref_ptr.h */; };
> 		413A23071FE18E0700373E99 /* httpbase.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A21B21FE18D9700373E99 /* httpbase.cc */; };
> 		413A23091FE18E0700373E99 /* memory_usage.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A21B41FE18D9700373E99 /* memory_usage.h */; };
>-		413A230A1FE18E0700373E99 /* ptr_util.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A21B51FE18D9800373E99 /* ptr_util.h */; };
> 		413A230D1FE18E0700373E99 /* rate_statistics.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A21B91FE18D9A00373E99 /* rate_statistics.h */; };
> 		413A230F1FE18E0700373E99 /* rtccertificate.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A21BB1FE18D9B00373E99 /* rtccertificate.cc */; };
> 		413A23101FE18E0700373E99 /* fakeclock.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A21BC1FE18D9B00373E99 /* fakeclock.cc */; };
>@@ -189,7 +227,6 @@
> 		413A23551FE18E0700373E99 /* asyncresolverinterface.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A22021FE18DBD00373E99 /* asyncresolverinterface.cc */; };
> 		413A23571FE18E0700373E99 /* network.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A22041FE18DBE00373E99 /* network.h */; };
> 		413A23581FE18E0700373E99 /* rate_statistics.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A22051FE18DBE00373E99 /* rate_statistics.cc */; };
>-		413A23591FE18E0700373E99 /* transformadapter.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A22061FE18DBF00373E99 /* transformadapter.cc */; };
> 		413A235A1FE18E0700373E99 /* platform_file.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A22071FE18DBF00373E99 /* platform_file.cc */; };
> 		413A235B1FE18E0700373E99 /* signalthread.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A22081FE18DC000373E99 /* signalthread.cc */; };
> 		413A235C1FE18E0700373E99 /* bind.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A22091FE18DC000373E99 /* bind.h */; };
>@@ -208,12 +245,10 @@
> 		413A236C1FE18E0700373E99 /* constructormagic.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A22191FE18DC600373E99 /* constructormagic.h */; };
> 		413A236D1FE18E0700373E99 /* httpcommon.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A221A1FE18DC700373E99 /* httpcommon.cc */; };
> 		413A236E1FE18E0700373E99 /* helpers.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A221B1FE18DC700373E99 /* helpers.cc */; };
>-		413A23711FE18E0700373E99 /* sigslot.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A221E1FE18DC800373E99 /* sigslot.cc */; };
> 		413A23731FE18E0700373E99 /* checks.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A22201FE18DCA00373E99 /* checks.cc */; };
> 		413A23741FE18E0700373E99 /* ifaddrs_converter.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A22211FE18DCB00373E99 /* ifaddrs_converter.cc */; };
> 		413A23761FE18E0700373E99 /* network_constants.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A22231FE18DCC00373E99 /* network_constants.h */; };
> 		413A23771FE18E0700373E99 /* bitbuffer.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A22241FE18DCD00373E99 /* bitbuffer.h */; };
>-		413A237A1FE18E0700373E99 /* transformadapter.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A22271FE18DCE00373E99 /* transformadapter.h */; };
> 		413A237B1FE18E0700373E99 /* timestampaligner.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A22281FE18DCE00373E99 /* timestampaligner.cc */; };
> 		413A237C1FE18E0700373E99 /* networkmonitor.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A22291FE18DCF00373E99 /* networkmonitor.h */; };
> 		413A237D1FE18E0700373E99 /* task_queue_posix.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A222A1FE18DCF00373E99 /* task_queue_posix.h */; };
>@@ -230,16 +265,12 @@
> 		413A238E1FE18E0700373E99 /* buffer.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A223B1FE18DD900373E99 /* buffer.h */; };
> 		413A238F1FE18E0700373E99 /* event_tracer.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A223C1FE18DD900373E99 /* event_tracer.cc */; };
> 		413A23901FE18E0800373E99 /* pathutils.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A223D1FE18DD900373E99 /* pathutils.h */; };
>-		413A23911FE18E0800373E99 /* md5.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A223E1FE18DDA00373E99 /* md5.h */; };
> 		413A23921FE18E0800373E99 /* sslroots.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A223F1FE18DDA00373E99 /* sslroots.h */; };
>-		413A23941FE18E0800373E99 /* ratelimiter.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A22411FE18DDB00373E99 /* ratelimiter.h */; };
> 		413A23951FE18E0800373E99 /* httpcommon.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A22421FE18DDB00373E99 /* httpcommon.h */; };
> 		413A23961FE18E0800373E99 /* asyncsocket.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A22431FE18DDC00373E99 /* asyncsocket.cc */; };
> 		413A23971FE18E0800373E99 /* macutils.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A22441FE18DDC00373E99 /* macutils.h */; };
>-		413A23981FE18E0800373E99 /* md5.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A22451FE18DDC00373E99 /* md5.cc */; };
> 		413A239B1FE18E0800373E99 /* bufferqueue.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A22481FE18DDE00373E99 /* bufferqueue.cc */; };
> 		413A239D1FE18E0800373E99 /* sequenced_task_checker_impl.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A224A1FE18DDF00373E99 /* sequenced_task_checker_impl.cc */; };
>-		413A23A01FE18E0800373E99 /* md5digest.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A224D1FE18DE000373E99 /* md5digest.h */; };
> 		413A23A11FE18E0800373E99 /* nethelpers.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A224E1FE18DE000373E99 /* nethelpers.cc */; };
> 		413A23A41FE18E0800373E99 /* bytebuffer.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A22511FE18DE100373E99 /* bytebuffer.h */; };
> 		413A23A51FE18E0800373E99 /* event.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A22521FE18DE200373E99 /* event.h */; };
>@@ -279,16 +310,13 @@
> 		413A23D81FE18E0800373E99 /* file_posix.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A22851FE18DFC00373E99 /* file_posix.cc */; };
> 		413A23D91FE18E0800373E99 /* testutils.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A22861FE18DFC00373E99 /* testutils.h */; };
> 		413A23DA1FE18E0800373E99 /* filerotatingstream.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A22871FE18DFD00373E99 /* filerotatingstream.cc */; };
>-		413A23DC1FE18E0800373E99 /* base64.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A22891FE18DFD00373E99 /* base64.cc */; };
> 		413A23DD1FE18E0800373E99 /* thread_darwin.mm in Sources */ = {isa = PBXBuildFile; fileRef = 413A228A1FE18DFE00373E99 /* thread_darwin.mm */; };
> 		413A23DE1FE18E0800373E99 /* task_queue_gcd.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A228B1FE18DFE00373E99 /* task_queue_gcd.cc */; };
> 		413A23DF1FE18E0800373E99 /* firewallsocketserver.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A228C1FE18DFE00373E99 /* firewallsocketserver.h */; };
>-		413A23E01FE18E0800373E99 /* timedelta.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A228D1FE18DFF00373E99 /* timedelta.h */; };
> 		413A23E21FE18E0800373E99 /* cpu_time.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A22901FE18E0000373E99 /* cpu_time.h */; };
> 		413A23E41FE18E0800373E99 /* arraysize.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A22921FE18E0000373E99 /* arraysize.h */; };
> 		413A23E51FE18E0800373E99 /* ignore_wundef.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A22931FE18E0100373E99 /* ignore_wundef.h */; };
> 		413A23E61FE18E0800373E99 /* network.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A22941FE18E0100373E99 /* network.cc */; };
>-		413A23E71FE18E0800373E99 /* sha1.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A22951FE18E0200373E99 /* sha1.cc */; };
> 		413A23EA1FE18E0800373E99 /* physicalsocketserver.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A22981FE18E0300373E99 /* physicalsocketserver.h */; };
> 		413A23EB1FE18E0800373E99 /* location.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A22991FE18E0300373E99 /* location.cc */; };
> 		413A23EC1FE18E0800373E99 /* location.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A229A1FE18E0400373E99 /* location.h */; };
>@@ -297,8 +325,6 @@
> 		413A23EF1FE18E0800373E99 /* race_checker.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A229D1FE18E0500373E99 /* race_checker.h */; };
> 		413A23F11FE18E0800373E99 /* sslfingerprint.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A229F1FE18E0600373E99 /* sslfingerprint.cc */; };
> 		413A23F21FE18E0800373E99 /* bytebuffer.cc in Sources */ = {isa = PBXBuildFile; fileRef = 413A22A01FE18E0600373E99 /* bytebuffer.cc */; };
>-		413A23FC1FE1987F00373E99 /* objc_video_decoder_factory.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A23F61FE1987B00373E99 /* objc_video_decoder_factory.h */; };
>-		413A23FD1FE1987F00373E99 /* objc_video_encoder_factory.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A23F71FE1987C00373E99 /* objc_video_encoder_factory.h */; };
> 		413A243A1FE1991A00373E99 /* RTCVideoViewShading.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A24021FE1990300373E99 /* RTCVideoViewShading.h */; };
> 		413A243B1FE1991A00373E99 /* RTCRtpReceiver.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A24031FE1990300373E99 /* RTCRtpReceiver.h */; };
> 		413A243C1FE1991A00373E99 /* RTCVideoCodecH264.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A24041FE1990300373E99 /* RTCVideoCodecH264.h */; };
>@@ -349,26 +375,22 @@
> 		413A246A1FE1991A00373E99 /* RTCMetricsSampleInfo.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A24321FE1991600373E99 /* RTCMetricsSampleInfo.h */; };
> 		413A246B1FE1991A00373E99 /* UIDevice+RTCDevice.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A24331FE1991600373E99 /* UIDevice+RTCDevice.h */; };
> 		413A246C1FE1991A00373E99 /* RTCRtpCodecParameters.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A24341FE1991700373E99 /* RTCRtpCodecParameters.h */; };
>-		413A246D1FE1991A00373E99 /* RTCAVFoundationVideoSource.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A24351FE1991700373E99 /* RTCAVFoundationVideoSource.h */; };
> 		413A246E1FE1991A00373E99 /* RTCVideoEncoderVP8.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A24361FE1991800373E99 /* RTCVideoEncoderVP8.h */; };
> 		413A246F1FE1991A00373E99 /* RTCVideoRenderer.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A24371FE1991800373E99 /* RTCVideoRenderer.h */; };
> 		413A24701FE1991A00373E99 /* RTCDataChannelConfiguration.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A24381FE1991900373E99 /* RTCDataChannelConfiguration.h */; };
> 		413A24711FE1991A00373E99 /* RTCMTLVideoView.h in Headers */ = {isa = PBXBuildFile; fileRef = 413A24391FE1991900373E99 /* RTCMTLVideoView.h */; };
>+		413AD1A321265B10003F7263 /* algorithm.h in Headers */ = {isa = PBXBuildFile; fileRef = 413AD1A121265B0F003F7263 /* algorithm.h */; };
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>+		419241812127497100634FCF /* bad_variant_access.cc in Sources */ = {isa = PBXBuildFile; fileRef = 4192417F2127497100634FCF /* bad_variant_access.cc */; };
>+		419241822127497100634FCF /* bad_variant_access.h in Headers */ = {isa = PBXBuildFile; fileRef = 419241802127497100634FCF /* bad_variant_access.h */; };
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>+		419241862127499300634FCF /* packet_queue_interface.cc in Sources */ = {isa = PBXBuildFile; fileRef = 419241842127499300634FCF /* packet_queue_interface.cc */; };
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>+		4192418C212749C800634FCF /* send_side_congestion_controller.cc in Sources */ = {isa = PBXBuildFile; fileRef = 41924188212749C700634FCF /* send_side_congestion_controller.cc */; };
>+		4192418D212749C800634FCF /* pacer_controller.h in Headers */ = {isa = PBXBuildFile; fileRef = 41924189212749C800634FCF /* pacer_controller.h */; };
>+		4192418E212749C800634FCF /* transport_feedback_adapter.cc in Sources */ = {isa = PBXBuildFile; fileRef = 4192418A212749C800634FCF /* transport_feedback_adapter.cc */; };
>+		419241902127581000634FCF /* fec_private_tables_bursty.cc in Sources */ = {isa = PBXBuildFile; fileRef = 4192418F2127581000634FCF /* fec_private_tables_bursty.cc */; };
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>+		419241992127586500634FCF /* rnn_vad_weights.cc in Sources */ = {isa = PBXBuildFile; fileRef = 419241942127586400634FCF /* rnn_vad_weights.cc */; };
>+		4192419A2127586500634FCF /* kiss_fft.cc in Sources */ = {isa = PBXBuildFile; fileRef = 419241952127586500634FCF /* kiss_fft.cc */; };
>+		4192419B2127586500634FCF /* kiss_fft.h in Headers */ = {isa = PBXBuildFile; fileRef = 419241962127586500634FCF /* kiss_fft.h */; };
>+		4192419C2127586500634FCF /* rnn_activations.h in Headers */ = {isa = PBXBuildFile; fileRef = 419241972127586500634FCF /* rnn_activations.h */; };
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>+		419241A02127588300634FCF /* audio_codec_pair_id.cc in Sources */ = {isa = PBXBuildFile; fileRef = 4192419E2127588200634FCF /* audio_codec_pair_id.cc */; };
>+		419241A8212758D300634FCF /* spectral_features_internal.cc in Sources */ = {isa = PBXBuildFile; fileRef = 419241A1212758D200634FCF /* spectral_features_internal.cc */; };
>+		419241A9212758D300634FCF /* symmetric_matrix_buffer.h in Headers */ = {isa = PBXBuildFile; fileRef = 419241A2212758D200634FCF /* symmetric_matrix_buffer.h */; };
>+		419241AA212758D300634FCF /* rnn.h in Headers */ = {isa = PBXBuildFile; fileRef = 419241A3212758D200634FCF /* rnn.h */; };
>+		419241AB212758D300634FCF /* spectral_features.h in Headers */ = {isa = PBXBuildFile; fileRef = 419241A4212758D200634FCF /* spectral_features.h */; };
>+		419241AC212758D300634FCF /* sequence_buffer.h in Headers */ = {isa = PBXBuildFile; fileRef = 419241A5212758D200634FCF /* sequence_buffer.h */; };
>+		419241AD212758D300634FCF /* spectral_features.cc in Sources */ = {isa = PBXBuildFile; fileRef = 419241A6212758D200634FCF /* spectral_features.cc */; };
>+		419241AE212758D300634FCF /* ring_buffer.h in Headers */ = {isa = PBXBuildFile; fileRef = 419241A7212758D300634FCF /* ring_buffer.h */; };
>+		419241B02127590200634FCF /* rtp_generic_frame_descriptor_extension.cc in Sources */ = {isa = PBXBuildFile; fileRef = 419241AF2127590200634FCF /* rtp_generic_frame_descriptor_extension.cc */; };
>+		419241C02127593F00634FCF /* loss_rate_filter.h in Headers */ = {isa = PBXBuildFile; fileRef = 419241B22127593C00634FCF /* loss_rate_filter.h */; };
>+		419241C12127593F00634FCF /* data_transfer_tracker.cc in Sources */ = {isa = PBXBuildFile; fileRef = 419241B32127593D00634FCF /* data_transfer_tracker.cc */; };
>+		419241C22127593F00634FCF /* windowed_filter.h in Headers */ = {isa = PBXBuildFile; fileRef = 419241B42127593D00634FCF /* windowed_filter.h */; };
>+		419241C32127593F00634FCF /* bbr_network_controller.h in Headers */ = {isa = PBXBuildFile; fileRef = 419241B52127593D00634FCF /* bbr_network_controller.h */; };
>+		419241C42127593F00634FCF /* bandwidth_sampler.cc in Sources */ = {isa = PBXBuildFile; fileRef = 419241B62127593D00634FCF /* bandwidth_sampler.cc */; };
>+		419241C52127593F00634FCF /* loss_rate_filter.cc in Sources */ = {isa = PBXBuildFile; fileRef = 419241B72127593D00634FCF /* loss_rate_filter.cc */; };
>+		419241C62127593F00634FCF /* bbr_factory.h in Headers */ = {isa = PBXBuildFile; fileRef = 419241B82127593E00634FCF /* bbr_factory.h */; };
>+		419241C72127593F00634FCF /* bbr_network_controller.cc in Sources */ = {isa = PBXBuildFile; fileRef = 419241B92127593E00634FCF /* bbr_network_controller.cc */; };
>+		419241C82127593F00634FCF /* packet_number_indexed_queue.h in Headers */ = {isa = PBXBuildFile; fileRef = 419241BA2127593E00634FCF /* packet_number_indexed_queue.h */; };
>+		419241C92127593F00634FCF /* bbr_factory.cc in Sources */ = {isa = PBXBuildFile; fileRef = 419241BB2127593E00634FCF /* bbr_factory.cc */; };
>+		419241CA2127593F00634FCF /* rtt_stats.cc in Sources */ = {isa = PBXBuildFile; fileRef = 419241BC2127593E00634FCF /* rtt_stats.cc */; };
>+		419241CB2127593F00634FCF /* rtt_stats.h in Headers */ = {isa = PBXBuildFile; fileRef = 419241BD2127593E00634FCF /* rtt_stats.h */; };
>+		419241CC2127593F00634FCF /* bandwidth_sampler.h in Headers */ = {isa = PBXBuildFile; fileRef = 419241BE2127593F00634FCF /* bandwidth_sampler.h */; };
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>+		419241D02127597D00634FCF /* simulated_network.cc in Sources */ = {isa = PBXBuildFile; fileRef = 419241CE2127597C00634FCF /* simulated_network.cc */; };
>+		419241D12127597D00634FCF /* simulated_network.h in Headers */ = {isa = PBXBuildFile; fileRef = 419241CF2127597C00634FCF /* simulated_network.h */; };
>+		419241D3212759A100634FCF /* fec_private_tables_random.cc in Sources */ = {isa = PBXBuildFile; fileRef = 419241D2212759A100634FCF /* fec_private_tables_random.cc */; };
>+		419241DD21275A3000634FCF /* time_delta.h in Headers */ = {isa = PBXBuildFile; fileRef = 419241D521275A2F00634FCF /* time_delta.h */; };
>+		419241DE21275A3000634FCF /* data_rate.h in Headers */ = {isa = PBXBuildFile; fileRef = 419241D621275A2F00634FCF /* data_rate.h */; };
>+		419241DF21275A3000634FCF /* data_size.cc in Sources */ = {isa = PBXBuildFile; fileRef = 419241D721275A2F00634FCF /* data_size.cc */; };
>+		419241E021275A3000634FCF /* timestamp.h in Headers */ = {isa = PBXBuildFile; fileRef = 419241D821275A2F00634FCF /* timestamp.h */; };
>+		419241E121275A3000634FCF /* data_rate.cc in Sources */ = {isa = PBXBuildFile; fileRef = 419241D921275A2F00634FCF /* data_rate.cc */; };
>+		419241E221275A3000634FCF /* time_delta.cc in Sources */ = {isa = PBXBuildFile; fileRef = 419241DA21275A3000634FCF /* time_delta.cc */; };
>+		419241E321275A3000634FCF /* timestamp.cc in Sources */ = {isa = PBXBuildFile; fileRef = 419241DB21275A3000634FCF /* timestamp.cc */; };
>+		419241E421275A3000634FCF /* data_size.h in Headers */ = {isa = PBXBuildFile; fileRef = 419241DC21275A3000634FCF /* data_size.h */; };
>+		419241E721275A7600634FCF /* field_trial_units.h in Headers */ = {isa = PBXBuildFile; fileRef = 419241E521275A7600634FCF /* field_trial_units.h */; };
>+		419241E821275A7600634FCF /* field_trial_units.cc in Sources */ = {isa = PBXBuildFile; fileRef = 419241E621275A7600634FCF /* field_trial_units.cc */; };
>+		419241ED21275AFA00634FCF /* simulcast_utility.h in Headers */ = {isa = PBXBuildFile; fileRef = 419241E921275AF900634FCF /* simulcast_utility.h */; };
>+		419241EE21275AFA00634FCF /* simulcast_rate_allocator.cc in Sources */ = {isa = PBXBuildFile; fileRef = 419241EA21275AFA00634FCF /* simulcast_rate_allocator.cc */; };
>+		419241EF21275AFA00634FCF /* simulcast_utility.cc in Sources */ = {isa = PBXBuildFile; fileRef = 419241EB21275AFA00634FCF /* simulcast_utility.cc */; };
>+		419241F021275AFA00634FCF /* simulcast_rate_allocator.h in Headers */ = {isa = PBXBuildFile; fileRef = 419241EC21275AFA00634FCF /* simulcast_rate_allocator.h */; };
>+		419241F121275B2B00634FCF /* simulcast_encoder_adapter.cc in Sources */ = {isa = PBXBuildFile; fileRef = 41F9BFB12051C93500ABF0B9 /* simulcast_encoder_adapter.cc */; };
>+		419241F421275C3200634FCF /* vp8_encoder_simulcast_proxy.h in Headers */ = {isa = PBXBuildFile; fileRef = 419241F221275C3100634FCF /* vp8_encoder_simulcast_proxy.h */; };
>+		419241F521275C3200634FCF /* vp8_encoder_simulcast_proxy.cc in Sources */ = {isa = PBXBuildFile; fileRef = 419241F321275C3200634FCF /* vp8_encoder_simulcast_proxy.cc */; };
>+		4192423D212768BD00634FCF /* objc_frame_buffer.mm in Sources */ = {isa = PBXBuildFile; fileRef = 4192421D2127664D00634FCF /* objc_frame_buffer.mm */; };
>+		4192423E21276B5A00634FCF /* VideoToolBoxEncoderFactory.cpp in Sources */ = {isa = PBXBuildFile; fileRef = 419C84AE1FE304C60040C30F /* VideoToolBoxEncoderFactory.cpp */; };
>+		4192424121276F0200634FCF /* VideoToolBoxDecoderFactory.cpp in Sources */ = {isa = PBXBuildFile; fileRef = 419C84AF1FE304C70040C30F /* VideoToolBoxDecoderFactory.cpp */; };
>+		419242422127718D00634FCF /* video_encoder_factory.mm in Sources */ = {isa = PBXBuildFile; fileRef = 419242052127664500634FCF /* video_encoder_factory.mm */; };
>+		419242432127719300634FCF /* video_decoder_factory.mm in Sources */ = {isa = PBXBuildFile; fileRef = 419242032127664200634FCF /* video_decoder_factory.mm */; };
>+		41924244212771B700634FCF /* objc_video_encoder_factory.mm in Sources */ = {isa = PBXBuildFile; fileRef = 419242172127664B00634FCF /* objc_video_encoder_factory.mm */; };
>+		41924245212771BB00634FCF /* objc_video_decoder_factory.mm in Sources */ = {isa = PBXBuildFile; fileRef = 4192421A2127664C00634FCF /* objc_video_decoder_factory.mm */; };
>+		41924246212771DC00634FCF /* objc_video_frame.mm in Sources */ = {isa = PBXBuildFile; fileRef = 4192421E2127664D00634FCF /* objc_video_frame.mm */; };
> 		419C829A1FE20CA10040C30F /* interval_budget.cc in Sources */ = {isa = PBXBuildFile; fileRef = 419C82991FE20CA10040C30F /* interval_budget.cc */; };
> 		419C829D1FE20D1C0040C30F /* audio_processing_statistics.cc in Sources */ = {isa = PBXBuildFile; fileRef = 419C829B1FE20D1B0040C30F /* audio_processing_statistics.cc */; };
> 		419C829E1FE20D1C0040C30F /* audio_processing_statistics.h in Headers */ = {isa = PBXBuildFile; fileRef = 419C829C1FE20D1C0040C30F /* audio_processing_statistics.h */; };
>@@ -469,7 +640,6 @@
> 		419C82B81FE20DCD0040C30F /* rtp_config.h in Headers */ = {isa = PBXBuildFile; fileRef = 419C82A51FE20DC80040C30F /* rtp_config.h */; };
> 		419C82B91FE20DCD0040C30F /* rtcp_demuxer.cc in Sources */ = {isa = PBXBuildFile; fileRef = 419C82A61FE20DC80040C30F /* rtcp_demuxer.cc */; };
> 		419C82BA1FE20DCD0040C30F /* video_send_stream.h in Headers */ = {isa = PBXBuildFile; fileRef = 419C82A71FE20DC90040C30F /* video_send_stream.h */; };
>-		419C82BB1FE20DCD0040C30F /* video_config.cc in Sources */ = {isa = PBXBuildFile; fileRef = 419C82A81FE20DCA0040C30F /* video_config.cc */; };
> 		419C82BC1FE20DCD0040C30F /* rtp_stream_receiver_controller.cc in Sources */ = {isa = PBXBuildFile; fileRef = 419C82A91FE20DCA0040C30F /* rtp_stream_receiver_controller.cc */; };
> 		419C82BD1FE20DCD0040C30F /* rtcp_packet_sink_interface.h in Headers */ = {isa = PBXBuildFile; fileRef = 419C82AA1FE20DCA0040C30F /* rtcp_packet_sink_interface.h */; };
> 		419C82BE1FE20DCD0040C30F /* video_receive_stream.cc in Sources */ = {isa = PBXBuildFile; fileRef = 419C82AB1FE20DCB0040C30F /* video_receive_stream.cc */; };
>@@ -477,11 +647,9 @@
> 		419C82C01FE20DCD0040C30F /* rtp_stream_receiver_controller_interface.h in Headers */ = {isa = PBXBuildFile; fileRef = 419C82AD1FE20DCB0040C30F /* rtp_stream_receiver_controller_interface.h */; };
> 		419C82C11FE20DCD0040C30F /* rtp_stream_receiver_controller.h in Headers */ = {isa = PBXBuildFile; fileRef = 419C82AE1FE20DCC0040C30F /* rtp_stream_receiver_controller.h */; };
> 		419C82C21FE20DCD0040C30F /* rtx_receive_stream.h in Headers */ = {isa = PBXBuildFile; fileRef = 419C82AF1FE20DCC0040C30F /* rtx_receive_stream.h */; };
>-		419C82C31FE20DCD0040C30F /* video_config.h in Headers */ = {isa = PBXBuildFile; fileRef = 419C82B01FE20DCC0040C30F /* video_config.h */; };
> 		419C82C41FE20DCD0040C30F /* video_receive_stream.h in Headers */ = {isa = PBXBuildFile; fileRef = 419C82B11FE20DCD0040C30F /* video_receive_stream.h */; };
> 		419C82C91FE20E2C0040C30F /* rtpmediautils.cc in Sources */ = {isa = PBXBuildFile; fileRef = 419C82C71FE20E2B0040C30F /* rtpmediautils.cc */; };
> 		419C82CA1FE20E2C0040C30F /* rtpmediautils.h in Headers */ = {isa = PBXBuildFile; fileRef = 419C82C81FE20E2C0040C30F /* rtpmediautils.h */; };
>-		419C82D91FE20E590040C30F /* transportcontroller.h in Headers */ = {isa = PBXBuildFile; fileRef = 419C82CB1FE20E540040C30F /* transportcontroller.h */; };
> 		419C82DA1FE20E590040C30F /* rtptransceiver.h in Headers */ = {isa = PBXBuildFile; fileRef = 419C82CC1FE20E540040C30F /* rtptransceiver.h */; };
> 		419C82DB1FE20E590040C30F /* rtptransportinternaladapter.h in Headers */ = {isa = PBXBuildFile; fileRef = 419C82CD1FE20E550040C30F /* rtptransportinternaladapter.h */; };
> 		419C82DC1FE20E590040C30F /* sdputils.cc in Sources */ = {isa = PBXBuildFile; fileRef = 419C82CE1FE20E550040C30F /* sdputils.cc */; };
>@@ -491,11 +659,9 @@
> 		419C82E01FE20E590040C30F /* dtlssrtptransport.h in Headers */ = {isa = PBXBuildFile; fileRef = 419C82D21FE20E570040C30F /* dtlssrtptransport.h */; };
> 		419C82E11FE20E590040C30F /* srtpsession.h in Headers */ = {isa = PBXBuildFile; fileRef = 419C82D31FE20E570040C30F /* srtpsession.h */; };
> 		419C82E21FE20E590040C30F /* srtptransport.h in Headers */ = {isa = PBXBuildFile; fileRef = 419C82D41FE20E570040C30F /* srtptransport.h */; };
>-		419C82E31FE20E590040C30F /* transportcontroller.cc in Sources */ = {isa = PBXBuildFile; fileRef = 419C82D51FE20E580040C30F /* transportcontroller.cc */; };
> 		419C82E41FE20E590040C30F /* dtlssrtptransport.cc in Sources */ = {isa = PBXBuildFile; fileRef = 419C82D61FE20E580040C30F /* dtlssrtptransport.cc */; };
> 		419C82E51FE20E590040C30F /* rtptransportinternal.h in Headers */ = {isa = PBXBuildFile; fileRef = 419C82D71FE20E580040C30F /* rtptransportinternal.h */; };
> 		419C82E61FE20E590040C30F /* rtptransceiver.cc in Sources */ = {isa = PBXBuildFile; fileRef = 419C82D81FE20E590040C30F /* rtptransceiver.cc */; };
>-		419C82E91FE20E8E0040C30F /* optional.cc in Sources */ = {isa = PBXBuildFile; fileRef = 419C82E81FE20E8E0040C30F /* optional.cc */; };
> 		419C82F01FE20EB50040C30F /* audio_encoder_opus.h in Headers */ = {isa = PBXBuildFile; fileRef = 419C82EA1FE20EB40040C30F /* audio_encoder_opus.h */; };
> 		419C82F11FE20EB50040C30F /* audio_encoder_opus_config.h in Headers */ = {isa = PBXBuildFile; fileRef = 419C82EB1FE20EB40040C30F /* audio_encoder_opus_config.h */; };
> 		419C82F21FE20EB50040C30F /* audio_decoder_opus.h in Headers */ = {isa = PBXBuildFile; fileRef = 419C82EC1FE20EB40040C30F /* audio_decoder_opus.h */; };
>@@ -507,16 +673,13 @@
> 		419C83011FE20F020040C30F /* rtcp_transceiver_config.h in Headers */ = {isa = PBXBuildFile; fileRef = 419C82F81FE20EFC0040C30F /* rtcp_transceiver_config.h */; };
> 		419C83021FE20F020040C30F /* rtcp_transceiver_config.cc in Sources */ = {isa = PBXBuildFile; fileRef = 419C82F91FE20EFD0040C30F /* rtcp_transceiver_config.cc */; };
> 		419C83031FE20F020040C30F /* rtcp_transceiver.h in Headers */ = {isa = PBXBuildFile; fileRef = 419C82FA1FE20EFD0040C30F /* rtcp_transceiver.h */; };
>-		419C83041FE20F020040C30F /* rtp_format_video_stereo.h in Headers */ = {isa = PBXBuildFile; fileRef = 419C82FB1FE20EFE0040C30F /* rtp_format_video_stereo.h */; };
> 		419C83051FE20F020040C30F /* rtcp_transceiver_impl.h in Headers */ = {isa = PBXBuildFile; fileRef = 419C82FC1FE20EFE0040C30F /* rtcp_transceiver_impl.h */; };
>-		419C83061FE20F020040C30F /* rtp_format_video_stereo.cc in Sources */ = {isa = PBXBuildFile; fileRef = 419C82FD1FE20EFE0040C30F /* rtp_format_video_stereo.cc */; };
> 		419C83071FE20F020040C30F /* rtp_packet_received.cc in Sources */ = {isa = PBXBuildFile; fileRef = 419C82FE1FE20F010040C30F /* rtp_packet_received.cc */; };
> 		419C830B1FE20F3C0040C30F /* fec_controller_rplr_based.h in Headers */ = {isa = PBXBuildFile; fileRef = 419C83081FE20F3B0040C30F /* fec_controller_rplr_based.h */; };
> 		419C830C1FE20F3C0040C30F /* fec_controller_rplr_based.cc in Sources */ = {isa = PBXBuildFile; fileRef = 419C83091FE20F3B0040C30F /* fec_controller_rplr_based.cc */; };
> 		419C830D1FE20F3C0040C30F /* audio_network_adaptor_config.cc in Sources */ = {isa = PBXBuildFile; fileRef = 419C830A1FE20F3B0040C30F /* audio_network_adaptor_config.cc */; };
> 		419C83121FE242800040C30F /* candidate.cc in Sources */ = {isa = PBXBuildFile; fileRef = 419C830E1FE2427F0040C30F /* candidate.cc */; };
> 		419C83131FE242800040C30F /* proxy.cc in Sources */ = {isa = PBXBuildFile; fileRef = 419C830F1FE2427F0040C30F /* proxy.cc */; };
>-		419C83141FE242800040C30F /* umametrics.cc in Sources */ = {isa = PBXBuildFile; fileRef = 419C83101FE2427F0040C30F /* umametrics.cc */; };
> 		419C83151FE242800040C30F /* rtpparameters.cc in Sources */ = {isa = PBXBuildFile; fileRef = 419C83111FE2427F0040C30F /* rtpparameters.cc */; };
> 		419C831A1FE242B20040C30F /* fir_filter_c.h in Headers */ = {isa = PBXBuildFile; fileRef = 419C83161FE242B10040C30F /* fir_filter_c.h */; };
> 		419C831B1FE242B20040C30F /* fir_filter_c.cc in Sources */ = {isa = PBXBuildFile; fileRef = 419C83171FE242B20040C30F /* fir_filter_c.cc */; };
>@@ -526,9 +689,7 @@
> 		419C83241FE245280040C30F /* rtc_event_rtp_packet_incoming.cc in Sources */ = {isa = PBXBuildFile; fileRef = 419C83221FE245280040C30F /* rtc_event_rtp_packet_incoming.cc */; };
> 		419C83251FE245280040C30F /* rtc_event_rtp_packet_incoming.h in Headers */ = {isa = PBXBuildFile; fileRef = 419C83231FE245280040C30F /* rtc_event_rtp_packet_incoming.h */; };
> 		419C832C1FE245CD0040C30F /* packet_queue.cc in Sources */ = {isa = PBXBuildFile; fileRef = 419C83261FE245CB0040C30F /* packet_queue.cc */; };
>-		419C832D1FE245CD0040C30F /* packet_queue2.cc in Sources */ = {isa = PBXBuildFile; fileRef = 419C83271FE245CB0040C30F /* packet_queue2.cc */; };
> 		419C832E1FE245CD0040C30F /* pacer.h in Headers */ = {isa = PBXBuildFile; fileRef = 419C83281FE245CB0040C30F /* pacer.h */; };
>-		419C832F1FE245CD0040C30F /* packet_queue2.h in Headers */ = {isa = PBXBuildFile; fileRef = 419C83291FE245CC0040C30F /* packet_queue2.h */; };
> 		419C83301FE245CD0040C30F /* packet_queue.h in Headers */ = {isa = PBXBuildFile; fileRef = 419C832A1FE245CC0040C30F /* packet_queue.h */; };
> 		419C83311FE245CD0040C30F /* interval_budget.h in Headers */ = {isa = PBXBuildFile; fileRef = 419C832B1FE245CC0040C30F /* interval_budget.h */; };
> 		419C83341FE245EA0040C30F /* time_interval.h in Headers */ = {isa = PBXBuildFile; fileRef = 419C83321FE245E90040C30F /* time_interval.h */; };
>@@ -625,10 +786,6 @@
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> 		5C119FFB1E457400004F0987 /* audio_device_buffer.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5C119FF01E457400004F0987 /* audio_device_buffer.cc */; };
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>@@ -997,9 +1291,6 @@
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> 		5C11A00D1E457448004F0987 /* audio_mixer_manager_mac.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5C11A0091E457448004F0987 /* audio_mixer_manager_mac.cc */; };
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> 		5C11A01F1E457578004F0987 /* file_audio_device_factory.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5C11A0191E457578004F0987 /* file_audio_device_factory.cc */; };
>@@ -1206,8 +1497,6 @@
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> 		5C4B48861E42C1BA002651C8 /* simulcast.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5C4B485F1E42C1BA002651C8 /* simulcast.cc */; };
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>-		5C4B48891E42C1BA002651C8 /* videoencodersoftwarefallbackwrapper.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5C4B48621E42C1BA002651C8 /* videoencodersoftwarefallbackwrapper.cc */; };
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> 		5C4B488D1E42C1BA002651C8 /* webrtcmediaengine.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5C4B48661E42C1BA002651C8 /* webrtcmediaengine.cc */; };
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> 		5C4B48901E42C1BA002651C8 /* webrtcvideocapturer.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5C4B48691E42C1BA002651C8 /* webrtcvideocapturer.cc */; };
>@@ -1284,8 +1573,6 @@
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> 		5C4B4C641E431F9C002651C8 /* channel_buffer.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5C4B4C301E431F9C002651C8 /* channel_buffer.cc */; };
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>-		5C4B4C661E431F9C002651C8 /* fft4g.c in Sources */ = {isa = PBXBuildFile; fileRef = 5C4B4C341E431F9C002651C8 /* fft4g.c */; };
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> 		5C4B4C6A1E431F9C002651C8 /* fir_filter_sse.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5C4B4C381E431F9C002651C8 /* fir_filter_sse.cc */; };
> 		5C4B4C6B1E431F9C002651C8 /* fir_filter_sse.h in Headers */ = {isa = PBXBuildFile; fileRef = 5C4B4C391E431F9C002651C8 /* fir_filter_sse.h */; };
> 		5C4B4C6E1E431F9C002651C8 /* fir_filter.h in Headers */ = {isa = PBXBuildFile; fileRef = 5C4B4C3C1E431F9C002651C8 /* fir_filter.h */; };
>@@ -1305,7 +1592,6 @@
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> 		5C4B4C861E431F9C002651C8 /* window_generator.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5C4B4C561E431F9C002651C8 /* window_generator.cc */; };
> 		5C4B4C871E431F9C002651C8 /* window_generator.h in Headers */ = {isa = PBXBuildFile; fileRef = 5C4B4C571E431F9C002651C8 /* window_generator.h */; };
>-		5C4B4CC11E4320A9002651C8 /* aligned_malloc.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5C4B4C8F1E4320A9002651C8 /* aligned_malloc.cc */; };
> 		5C4B4CC61E4320A9002651C8 /* clock.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5C4B4C941E4320A9002651C8 /* clock.cc */; };
> 		5C4B4CCA1E4320A9002651C8 /* cpu_features.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5C4B4C981E4320A9002651C8 /* cpu_features.cc */; };
> 		5C4B4CCB1E4320A9002651C8 /* cpu_info.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5C4B4C991E4320A9002651C8 /* cpu_info.cc */; };
>@@ -1313,19 +1599,12 @@
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> 		5C4B4CD21E4320A9002651C8 /* event.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5C4B4CA01E4320A9002651C8 /* event.cc */; };
> 		5C4B4CD31E4320A9002651C8 /* field_trial_default.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5C4B4CA11E4320A9002651C8 /* field_trial_default.cc */; };
>-		5C4B4CD41E4320A9002651C8 /* file_impl.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5C4B4CA21E4320A9002651C8 /* file_impl.cc */; };
> 		5C4B4CD91E4320A9002651C8 /* metrics_default.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5C4B4CA71E4320A9002651C8 /* metrics_default.cc */; };
>-		5C4B4CDE1E4320A9002651C8 /* rw_lock_posix.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5C4B4CAD1E4320A9002651C8 /* rw_lock_posix.cc */; };
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> 		5C4B4CE51E4320A9002651C8 /* sleep.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5C4B4CB41E4320A9002651C8 /* sleep.cc */; };
>-		5C4B4CE91E4320A9002651C8 /* timestamp_extrapolator.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5C4B4CB81E4320A9002651C8 /* timestamp_extrapolator.cc */; };
> 		5C4B4D021E432185002651C8 /* spake25519.c in Sources */ = {isa = PBXBuildFile; fileRef = 5C4B4CF61E432168002651C8 /* spake25519.c */; };
> 		5C4B4D031E432185002651C8 /* x25519-x86_64.c in Sources */ = {isa = PBXBuildFile; fileRef = 5C4B4CF81E432168002651C8 /* x25519-x86_64.c */; };
> 		5C4B4D091E4322F6002651C8 /* basicportallocator.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5C4B4D051E4322F6002651C8 /* basicportallocator.cc */; };
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>-		5C4B4D0B1E4322F6002651C8 /* socketmonitor.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5C4B4D071E4322F6002651C8 /* socketmonitor.cc */; };
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> 		5C5F40971E978FDB00D94279 /* NSQ_del_dec_sse.c in Sources */ = {isa = PBXBuildFile; fileRef = 5C5F408D1E978FD200D94279 /* NSQ_del_dec_sse.c */; };
> 		5C5F40981E978FDB00D94279 /* NSQ_sse.c in Sources */ = {isa = PBXBuildFile; fileRef = 5C5F408E1E978FD200D94279 /* NSQ_sse.c */; };
> 		5C5F40991E978FDB00D94279 /* SigProc_FIX_sse.h in Headers */ = {isa = PBXBuildFile; fileRef = 5C5F408F1E978FD200D94279 /* SigProc_FIX_sse.h */; };
>@@ -1400,8 +1679,6 @@
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> 		5C63F93D1E41737B002CA531 /* dtmfsenderinterface.h in Headers */ = {isa = PBXBuildFile; fileRef = 5C63F8EB1E41737B002CA531 /* dtmfsenderinterface.h */; };
>-		5C63F93F1E41737B002CA531 /* fakemetricsobserver.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5C63F8ED1E41737B002CA531 /* fakemetricsobserver.cc */; };
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> 		5C63F9411E41737B002CA531 /* jsep.h in Headers */ = {isa = PBXBuildFile; fileRef = 5C63F8EF1E41737B002CA531 /* jsep.h */; };
> 		5C63F9431E41737B002CA531 /* jsepicecandidate.h in Headers */ = {isa = PBXBuildFile; fileRef = 5C63F8F11E41737B002CA531 /* jsepicecandidate.h */; };
> 		5C63F9451E41737B002CA531 /* jsepsessiondescription.h in Headers */ = {isa = PBXBuildFile; fileRef = 5C63F8F31E41737B002CA531 /* jsepsessiondescription.h */; };
>@@ -1427,7 +1704,6 @@
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> 		5C63F9E51E4174F6002CA531 /* candidatepairinterface.h in Headers */ = {isa = PBXBuildFile; fileRef = 5C63F98F1E4174F6002CA531 /* candidatepairinterface.h */; };
>-		5C63F9E61E4174F6002CA531 /* common.h in Headers */ = {isa = PBXBuildFile; fileRef = 5C63F9901E4174F6002CA531 /* common.h */; };
> 		5C63F9EB1E4174F6002CA531 /* fakeportallocator.h in Headers */ = {isa = PBXBuildFile; fileRef = 5C63F9951E4174F6002CA531 /* fakeportallocator.h */; };
> 		5C63F9ED1E4174F6002CA531 /* p2pconstants.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5C63F9971E4174F6002CA531 /* p2pconstants.cc */; };
> 		5C63F9EE1E4174F6002CA531 /* p2pconstants.h in Headers */ = {isa = PBXBuildFile; fileRef = 5C63F9981E4174F6002CA531 /* p2pconstants.h */; };
>@@ -1446,8 +1722,6 @@
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> 		5C63FA041E4174F6002CA531 /* relayserver.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5C63F9AE1E4174F6002CA531 /* relayserver.cc */; };
> 		5C63FA051E4174F6002CA531 /* relayserver.h in Headers */ = {isa = PBXBuildFile; fileRef = 5C63F9AF1E4174F6002CA531 /* relayserver.h */; };
>-		5C63FA061E4174F6002CA531 /* session.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5C63F9B01E4174F6002CA531 /* session.cc */; };
>-		5C63FA071E4174F6002CA531 /* session.h in Headers */ = {isa = PBXBuildFile; fileRef = 5C63F9B11E4174F6002CA531 /* session.h */; };
> 		5C63FA0B1E4174F6002CA531 /* stun.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5C63F9B51E4174F6002CA531 /* stun.cc */; };
> 		5C63FA0C1E4174F6002CA531 /* stun.h in Headers */ = {isa = PBXBuildFile; fileRef = 5C63F9B61E4174F6002CA531 /* stun.h */; };
> 		5C63FA0E1E4174F6002CA531 /* stunport.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5C63F9B81E4174F6002CA531 /* stunport.cc */; };
>@@ -1475,9 +1749,7 @@
> 		5C63FA771E417AED002CA531 /* rtcstats_objects.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5C63FA741E417AED002CA531 /* rtcstats_objects.cc */; };
> 		5C63FA781E417AED002CA531 /* rtcstats.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5C63FA751E417AED002CA531 /* rtcstats.cc */; };
> 		5C63FA791E417AED002CA531 /* rtcstatsreport.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5C63FA761E417AED002CA531 /* rtcstatsreport.cc */; };
>-		5C6CDD4B1E413598009754E3 /* common_types.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5C6CDD3E1E413598009754E3 /* common_types.cc */; };
> 		5C6CDD4C1E413598009754E3 /* common_types.h in Headers */ = {isa = PBXBuildFile; fileRef = 5C6CDD3F1E413598009754E3 /* common_types.h */; };
>-		5C6CDD501E413598009754E3 /* typedefs.h in Headers */ = {isa = PBXBuildFile; fileRef = 5C6CDD431E413598009754E3 /* typedefs.h */; };
> 		5CD284671E6A57F40094FDC8 /* i420_buffer.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5CD284611E6A57F40094FDC8 /* i420_buffer.cc */; };
> 		5CD284681E6A57F40094FDC8 /* i420_buffer.h in Headers */ = {isa = PBXBuildFile; fileRef = 5CD284621E6A57F40094FDC8 /* i420_buffer.h */; };
> 		5CD284691E6A57F40094FDC8 /* video_frame_buffer.h in Headers */ = {isa = PBXBuildFile; fileRef = 5CD284631E6A57F40094FDC8 /* video_frame_buffer.h */; };
>@@ -1608,8 +1880,6 @@
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> 		5CD285CF1E6A63430094FDC8 /* matched_filter.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5CD2858B1E6A63430094FDC8 /* matched_filter.cc */; };
> 		5CD285D01E6A63430094FDC8 /* matched_filter.h in Headers */ = {isa = PBXBuildFile; fileRef = 5CD2858C1E6A63430094FDC8 /* matched_filter.h */; };
>-		5CD285D11E6A63430094FDC8 /* output_selector.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5CD2858D1E6A63430094FDC8 /* output_selector.cc */; };
>-		5CD285D21E6A63430094FDC8 /* output_selector.h in Headers */ = {isa = PBXBuildFile; fileRef = 5CD2858E1E6A63430094FDC8 /* output_selector.h */; };
> 		5CD285D51E6A63430094FDC8 /* render_delay_buffer.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5CD285911E6A63430094FDC8 /* render_delay_buffer.cc */; };
> 		5CD285D61E6A63430094FDC8 /* render_delay_buffer.h in Headers */ = {isa = PBXBuildFile; fileRef = 5CD285921E6A63430094FDC8 /* render_delay_buffer.h */; };
> 		5CD285D71E6A63430094FDC8 /* render_delay_controller_metrics.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5CD285931E6A63430094FDC8 /* render_delay_controller_metrics.cc */; };
>@@ -1637,17 +1907,10 @@
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> 		5CD285FA1E6A64520094FDC8 /* rtcp_nack_stats.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5CD285F81E6A64520094FDC8 /* rtcp_nack_stats.cc */; };
> 		5CD285FB1E6A64520094FDC8 /* rtcp_nack_stats.h in Headers */ = {isa = PBXBuildFile; fileRef = 5CD285F91E6A64520094FDC8 /* rtcp_nack_stats.h */; };
>-		5CD2860B1E6A64840094FDC8 /* transport_feedback_packet_loss_tracker.h in Headers */ = {isa = PBXBuildFile; fileRef = 5CD286021E6A64840094FDC8 /* transport_feedback_packet_loss_tracker.h */; };
> 		5CD286101E6A64C90094FDC8 /* moving_max.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5CD2860E1E6A64C90094FDC8 /* moving_max.cc */; };
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>-		5CD286241E6A666D0094FDC8 /* videodecodersoftwarefallbackwrapper.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5CD286221E6A666D0094FDC8 /* videodecodersoftwarefallbackwrapper.cc */; };
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> 		5CD286271E6A669D0094FDC8 /* rtp_to_ntp_estimator.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5CD286261E6A669D0094FDC8 /* rtp_to_ntp_estimator.cc */; };
> 		5CD2862A1E6A66C80094FDC8 /* quality_threshold.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5CD286281E6A66C80094FDC8 /* quality_threshold.cc */; };
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>@@ -1679,8 +1942,6 @@
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>@@ -1720,8 +1981,6 @@
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>@@ -1742,14 +2001,8 @@
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>@@ -1806,8 +2059,6 @@
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> 		5CDD85AA1E43B5C000621E92 /* receive_statistics_proxy.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5CDD856D1E43B5C000621E92 /* receive_statistics_proxy.cc */; };
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>@@ -1962,12 +2212,6 @@
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>@@ -1986,11 +2230,8 @@
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>@@ -2101,7 +2342,6 @@
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> 		5CDD893E1E43BF3A00621E92 /* bye.cc in Sources */ = {isa = PBXBuildFile; fileRef = 5CDD88F01E43BF3A00621E92 /* bye.cc */; };
>@@ -2166,10 +2406,6 @@
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>@@ -2657,20 +2893,12 @@
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>@@ -2688,7 +2916,6 @@
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>@@ -2702,27 +2929,6 @@
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>-		4130920D1EF8D40100757C55 /* transport_feedback_packet_loss_tracker.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = transport_feedback_packet_loss_tracker.cc; sourceTree = "<group>"; };
> 		413092101EF8D50C00757C55 /* aec_dump.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = aec_dump.h; path = include/aec_dump.h; sourceTree = "<group>"; };
> 		413092111EF8D51000757C55 /* aec_dump.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = aec_dump.cc; path = include/aec_dump.cc; sourceTree = "<group>"; };
> 		413092141EF8D5ED00757C55 /* send_side_congestion_controller.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = send_side_congestion_controller.cc; sourceTree = "<group>"; };
> 		413092161EF8D63400757C55 /* rtp_transport_controller_send.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = rtp_transport_controller_send.h; path = call/rtp_transport_controller_send.h; sourceTree = "<group>"; };
> 		413092171EF8D63900757C55 /* rtp_transport_controller_send_interface.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = rtp_transport_controller_send_interface.h; path = call/rtp_transport_controller_send_interface.h; sourceTree = "<group>"; };
> 		413092181EF8D63F00757C55 /* rtp_transport_controller_send.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = rtp_transport_controller_send.cc; path = call/rtp_transport_controller_send.cc; sourceTree = "<group>"; };
>-		4130921C1EF8D69000757C55 /* gain_controller2.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = gain_controller2.h; path = agc2/gain_controller2.h; sourceTree = "<group>"; };
>-		4130921F1EF8D6A200757C55 /* gain_controller2.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = gain_controller2.cc; path = agc2/gain_controller2.cc; sourceTree = "<group>"; };
>-		413092251EF8D6E600757C55 /* builtin_audio_encoder_factory.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = builtin_audio_encoder_factory.h; sourceTree = "<group>"; };
> 		4130922B1EF8D76100757C55 /* aec_dump_impl.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = aec_dump_impl.cc; path = Source/webrtc/modules/audio_processing/aec_dump/aec_dump_impl.cc; sourceTree = SOURCE_ROOT; };
> 		4130922C1EF8D76A00757C55 /* aec_dump_impl.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = aec_dump_impl.h; path = Source/webrtc/modules/audio_processing/aec_dump/aec_dump_impl.h; sourceTree = SOURCE_ROOT; };
> 		4130922D1EF8D77100757C55 /* aec_dump_factory.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = aec_dump_factory.h; path = Source/webrtc/modules/audio_processing/aec_dump/aec_dump_factory.h; sourceTree = SOURCE_ROOT; };
>@@ -3023,7 +3275,6 @@
> 		413A213C1FE0F0EF00373E99 /* getopt_s.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = getopt_s.h; path = include/getopt_s.h; sourceTree = "<group>"; };
> 		413A21441FE18D4C00373E99 /* testbase64.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = testbase64.h; path = rtc_base/testbase64.h; sourceTree = "<group>"; };
> 		413A21451FE18D4C00373E99 /* function_view.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = function_view.h; path = rtc_base/function_view.h; sourceTree = "<group>"; };
>-		413A21461FE18D4C00373E99 /* numerics */ = {isa = PBXFileReference; lastKnownFileType = folder; name = numerics; path = rtc_base/numerics; sourceTree = "<group>"; };
> 		413A21471FE18D4D00373E99 /* format_macros.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = format_macros.h; path = rtc_base/format_macros.h; sourceTree = "<group>"; };
> 		413A21481FE18D4D00373E99 /* unixfilesystem.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = unixfilesystem.h; path = rtc_base/unixfilesystem.h; sourceTree = "<group>"; };
> 		413A21491FE18D4E00373E99 /* testechoserver.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = testechoserver.cc; path = rtc_base/testechoserver.cc; sourceTree = "<group>"; };
>@@ -3040,10 +3291,8 @@
> 		413A21541FE18D5300373E99 /* httpserver_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = httpserver_unittest.cc; path = rtc_base/httpserver_unittest.cc; sourceTree = "<group>"; };
> 		413A21551FE18D5400373E99 /* asyncinvoker.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = asyncinvoker.h; path = rtc_base/asyncinvoker.h; sourceTree = "<group>"; };
> 		413A21561FE18D5400373E99 /* messagequeue.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = messagequeue.cc; path = rtc_base/messagequeue.cc; sourceTree = "<group>"; };
>-		413A21571FE18D5500373E99 /* md5digest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = md5digest.cc; path = rtc_base/md5digest.cc; sourceTree = "<group>"; };
> 		413A21581FE18D5500373E99 /* asyncpacketsocket.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = asyncpacketsocket.cc; path = rtc_base/asyncpacketsocket.cc; sourceTree = "<group>"; };
> 		413A21591FE18D5600373E99 /* nethelper.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = nethelper.cc; path = rtc_base/nethelper.cc; sourceTree = "<group>"; };
>-		413A215A1FE18D5600373E99 /* ptr_util_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = ptr_util_unittest.cc; path = rtc_base/ptr_util_unittest.cc; sourceTree = "<group>"; };
> 		413A215B1FE18D5700373E99 /* stringencode.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = stringencode.cc; path = rtc_base/stringencode.cc; sourceTree = "<group>"; };
> 		413A215C1FE18D5800373E99 /* asyncudpsocket.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = asyncudpsocket.h; path = rtc_base/asyncudpsocket.h; sourceTree = "<group>"; };
> 		413A215D1FE18D5900373E99 /* fakenetwork.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = fakenetwork.h; path = rtc_base/fakenetwork.h; sourceTree = "<group>"; };
>@@ -3077,7 +3326,6 @@
> 		413A217B1FE18D7600373E99 /* rollingaccumulator_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = rollingaccumulator_unittest.cc; path = rtc_base/rollingaccumulator_unittest.cc; sourceTree = "<group>"; };
> 		413A217C1FE18D7700373E99 /* socketstream.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = socketstream.cc; path = rtc_base/socketstream.cc; sourceTree = "<group>"; };
> 		413A217E1FE18D7800373E99 /* sequenced_task_checker.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = sequenced_task_checker.h; path = rtc_base/sequenced_task_checker.h; sourceTree = "<group>"; };
>-		413A217F1FE18D7900373E99 /* sha1.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = sha1.h; path = rtc_base/sha1.h; sourceTree = "<group>"; };
> 		413A21801FE18D7A00373E99 /* DEPS */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = text; name = DEPS; path = rtc_base/DEPS; sourceTree = "<group>"; };
> 		413A21811FE18D7B00373E99 /* base64_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = base64_unittest.cc; path = rtc_base/base64_unittest.cc; sourceTree = "<group>"; };
> 		413A21821FE18D7C00373E99 /* file.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = file.h; path = rtc_base/file.h; sourceTree = "<group>"; };
>@@ -3103,21 +3351,16 @@
> 		413A21961FE18D8900373E99 /* stringencode.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = stringencode.h; path = rtc_base/stringencode.h; sourceTree = "<group>"; };
> 		413A21971FE18D8900373E99 /* swap_queue_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = swap_queue_unittest.cc; path = rtc_base/swap_queue_unittest.cc; sourceTree = "<group>"; };
> 		413A21981FE18D8A00373E99 /* atomicops.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = atomicops.h; path = rtc_base/atomicops.h; sourceTree = "<group>"; };
>-		413A21991FE18D8A00373E99 /* ratelimiter.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = ratelimiter.cc; path = rtc_base/ratelimiter.cc; sourceTree = "<group>"; };
> 		413A219A1FE18D8B00373E99 /* openssladapter.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = openssladapter.cc; path = rtc_base/openssladapter.cc; sourceTree = "<group>"; };
> 		413A219B1FE18D8B00373E99 /* socketadapters.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = socketadapters.cc; path = rtc_base/socketadapters.cc; sourceTree = "<group>"; };
>-		413A219C1FE18D8C00373E99 /* basictypes.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = basictypes.h; path = rtc_base/basictypes.h; sourceTree = "<group>"; };
> 		413A219D1FE18D8D00373E99 /* pathutils_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = pathutils_unittest.cc; path = rtc_base/pathutils_unittest.cc; sourceTree = "<group>"; };
> 		413A219E1FE18D8D00373E99 /* task_queue_win.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = task_queue_win.cc; path = rtc_base/task_queue_win.cc; sourceTree = "<group>"; };
> 		413A219F1FE18D8E00373E99 /* testutils.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = testutils.cc; path = rtc_base/testutils.cc; sourceTree = "<group>"; };
>-		413A21A01FE18D8E00373E99 /* sha1digest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = sha1digest.cc; path = rtc_base/sha1digest.cc; sourceTree = "<group>"; };
>-		413A21A11FE18D8F00373E99 /* base64.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = base64.h; path = rtc_base/base64.h; sourceTree = "<group>"; };
> 		413A21A21FE18D8F00373E99 /* sigslot_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = sigslot_unittest.cc; path = rtc_base/sigslot_unittest.cc; sourceTree = "<group>"; };
> 		413A21A31FE18D9000373E99 /* fakesslidentity.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = fakesslidentity.h; path = rtc_base/fakesslidentity.h; sourceTree = "<group>"; };
> 		413A21A41FE18D9000373E99 /* messagedigest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = messagedigest.cc; path = rtc_base/messagedigest.cc; sourceTree = "<group>"; };
> 		413A21A51FE18D9100373E99 /* messagedigest.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = messagedigest.h; path = rtc_base/messagedigest.h; sourceTree = "<group>"; };
> 		413A21A61FE18D9100373E99 /* race_checker.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = race_checker.cc; path = rtc_base/race_checker.cc; sourceTree = "<group>"; };
>-		413A21A71FE18D9100373E99 /* sha1digest.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = sha1digest.h; path = rtc_base/sha1digest.h; sourceTree = "<group>"; };
> 		413A21A81FE18D9200373E99 /* stream.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = stream.h; path = rtc_base/stream.h; sourceTree = "<group>"; };
> 		413A21A91FE18D9200373E99 /* template_util.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = template_util.h; path = rtc_base/template_util.h; sourceTree = "<group>"; };
> 		413A21AA1FE18D9300373E99 /* sslstreamadapter.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = sslstreamadapter.cc; path = rtc_base/sslstreamadapter.cc; sourceTree = "<group>"; };
>@@ -3131,7 +3374,6 @@
> 		413A21B21FE18D9700373E99 /* httpbase.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = httpbase.cc; path = rtc_base/httpbase.cc; sourceTree = "<group>"; };
> 		413A21B31FE18D9700373E99 /* httpcommon_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = httpcommon_unittest.cc; path = rtc_base/httpcommon_unittest.cc; sourceTree = "<group>"; };
> 		413A21B41FE18D9700373E99 /* memory_usage.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = memory_usage.h; path = rtc_base/memory_usage.h; sourceTree = "<group>"; };
>-		413A21B51FE18D9800373E99 /* ptr_util.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = ptr_util.h; path = rtc_base/ptr_util.h; sourceTree = "<group>"; };
> 		413A21B61FE18D9800373E99 /* thread_checker_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = thread_checker_unittest.cc; path = rtc_base/thread_checker_unittest.cc; sourceTree = "<group>"; };
> 		413A21B71FE18D9900373E99 /* callback.h.pump */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = text; name = callback.h.pump; path = rtc_base/callback.h.pump; sourceTree = "<group>"; };
> 		413A21B81FE18D9900373E99 /* cpu_time.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = cpu_time.cc; path = rtc_base/cpu_time.cc; sourceTree = "<group>"; };
>@@ -3146,7 +3388,6 @@
> 		413A21C11FE18D9D00373E99 /* copyonwritebuffer.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = copyonwritebuffer.cc; path = rtc_base/copyonwritebuffer.cc; sourceTree = "<group>"; };
> 		413A21C21FE18D9E00373E99 /* asyncudpsocket.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = asyncudpsocket.cc; path = rtc_base/asyncudpsocket.cc; sourceTree = "<group>"; };
> 		413A21C31FE18D9E00373E99 /* callback.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = callback.h; path = rtc_base/callback.h; sourceTree = "<group>"; };
>-		413A21C41FE18D9F00373E99 /* event_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = event_unittest.cc; path = rtc_base/event_unittest.cc; sourceTree = "<group>"; };
> 		413A21C51FE18D9F00373E99 /* helpers_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = helpers_unittest.cc; path = rtc_base/helpers_unittest.cc; sourceTree = "<group>"; };
> 		413A21C61FE18DA000373E99 /* natsocketfactory.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = natsocketfactory.cc; path = rtc_base/natsocketfactory.cc; sourceTree = "<group>"; };
> 		413A21C71FE18DA100373E99 /* optionsfile_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = optionsfile_unittest.cc; path = rtc_base/optionsfile_unittest.cc; sourceTree = "<group>"; };
>@@ -3171,13 +3412,11 @@
> 		413A21DA1FE18DAC00373E99 /* virtualsocket_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = virtualsocket_unittest.cc; path = rtc_base/virtualsocket_unittest.cc; sourceTree = "<group>"; };
> 		413A21DB1FE18DAC00373E99 /* openssladapter.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = openssladapter.h; path = rtc_base/openssladapter.h; sourceTree = "<group>"; };
> 		413A21DC1FE18DAD00373E99 /* json.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = json.cc; path = rtc_base/json.cc; sourceTree = "<group>"; };
>-		413A21DD1FE18DAD00373E99 /* basictypes_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = basictypes_unittest.cc; path = rtc_base/basictypes_unittest.cc; sourceTree = "<group>"; };
> 		413A21DE1FE18DAE00373E99 /* logging.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = logging.h; path = rtc_base/logging.h; sourceTree = "<group>"; };
> 		413A21DF1FE18DAE00373E99 /* sigslotrepeater.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = sigslotrepeater.h; path = rtc_base/sigslotrepeater.h; sourceTree = "<group>"; };
> 		413A21E01FE18DAE00373E99 /* logging_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = logging_unittest.cc; path = rtc_base/logging_unittest.cc; sourceTree = "<group>"; };
> 		413A21E11FE18DAF00373E99 /* messagedigest_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = messagedigest_unittest.cc; path = rtc_base/messagedigest_unittest.cc; sourceTree = "<group>"; };
> 		413A21E21FE18DAF00373E99 /* onetimeevent.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = onetimeevent.h; path = rtc_base/onetimeevent.h; sourceTree = "<group>"; };
>-		413A21E31FE18DB000373E99 /* sha1digest_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = sha1digest_unittest.cc; path = rtc_base/sha1digest_unittest.cc; sourceTree = "<group>"; };
> 		413A21E41FE18DB000373E99 /* macutils.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = macutils.cc; path = rtc_base/macutils.cc; sourceTree = "<group>"; };
> 		413A21E51FE18DB100373E99 /* task_queue.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = task_queue.h; path = rtc_base/task_queue.h; sourceTree = "<group>"; };
> 		413A21E61FE18DB100373E99 /* httpbase_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = httpbase_unittest.cc; path = rtc_base/httpbase_unittest.cc; sourceTree = "<group>"; };
>@@ -3187,7 +3426,6 @@
> 		413A21EA1FE18DB300373E99 /* byteorder.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = byteorder.h; path = rtc_base/byteorder.h; sourceTree = "<group>"; };
> 		413A21EB1FE18DB300373E99 /* sslstreamadapter_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = sslstreamadapter_unittest.cc; path = rtc_base/sslstreamadapter_unittest.cc; sourceTree = "<group>"; };
> 		413A21EC1FE18DB400373E99 /* json.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = json.h; path = rtc_base/json.h; sourceTree = "<group>"; };
>-		413A21ED1FE18DB400373E99 /* ratelimiter_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = ratelimiter_unittest.cc; path = rtc_base/ratelimiter_unittest.cc; sourceTree = "<group>"; };
> 		413A21EE1FE18DB400373E99 /* thread.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = thread.h; path = rtc_base/thread.h; sourceTree = "<group>"; };
> 		413A21EF1FE18DB500373E99 /* crc32.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = crc32.cc; path = rtc_base/crc32.cc; sourceTree = "<group>"; };
> 		413A21F01FE18DB500373E99 /* memory_usage_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = memory_usage_unittest.cc; path = rtc_base/memory_usage_unittest.cc; sourceTree = "<group>"; };
>@@ -3212,7 +3450,6 @@
> 		413A22031FE18DBE00373E99 /* stringencode_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = stringencode_unittest.cc; path = rtc_base/stringencode_unittest.cc; sourceTree = "<group>"; };
> 		413A22041FE18DBE00373E99 /* network.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = network.h; path = rtc_base/network.h; sourceTree = "<group>"; };
> 		413A22051FE18DBE00373E99 /* rate_statistics.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = rate_statistics.cc; path = rtc_base/rate_statistics.cc; sourceTree = "<group>"; };
>-		413A22061FE18DBF00373E99 /* transformadapter.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = transformadapter.cc; path = rtc_base/transformadapter.cc; sourceTree = "<group>"; };
> 		413A22071FE18DBF00373E99 /* platform_file.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = platform_file.cc; path = rtc_base/platform_file.cc; sourceTree = "<group>"; };
> 		413A22081FE18DC000373E99 /* signalthread.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = signalthread.cc; path = rtc_base/signalthread.cc; sourceTree = "<group>"; };
> 		413A22091FE18DC000373E99 /* bind.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = bind.h; path = rtc_base/bind.h; sourceTree = "<group>"; };
>@@ -3235,8 +3472,6 @@
> 		413A221A1FE18DC700373E99 /* httpcommon.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = httpcommon.cc; path = rtc_base/httpcommon.cc; sourceTree = "<group>"; };
> 		413A221B1FE18DC700373E99 /* helpers.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = helpers.cc; path = rtc_base/helpers.cc; sourceTree = "<group>"; };
> 		413A221C1FE18DC800373E99 /* bitrateallocationstrategy_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = bitrateallocationstrategy_unittest.cc; path = rtc_base/bitrateallocationstrategy_unittest.cc; sourceTree = "<group>"; };
>-		413A221D1FE18DC800373E99 /* md5digest_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = md5digest_unittest.cc; path = rtc_base/md5digest_unittest.cc; sourceTree = "<group>"; };
>-		413A221E1FE18DC800373E99 /* sigslot.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = sigslot.cc; path = rtc_base/sigslot.cc; sourceTree = "<group>"; };
> 		413A221F1FE18DC900373E99 /* ratetracker_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = ratetracker_unittest.cc; path = rtc_base/ratetracker_unittest.cc; sourceTree = "<group>"; };
> 		413A22201FE18DCA00373E99 /* checks.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = checks.cc; path = rtc_base/checks.cc; sourceTree = "<group>"; };
> 		413A22211FE18DCB00373E99 /* ifaddrs_converter.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = ifaddrs_converter.cc; path = rtc_base/ifaddrs_converter.cc; sourceTree = "<group>"; };
>@@ -3245,7 +3480,6 @@
> 		413A22241FE18DCD00373E99 /* bitbuffer.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = bitbuffer.h; path = rtc_base/bitbuffer.h; sourceTree = "<group>"; };
> 		413A22251FE18DCD00373E99 /* ssladapter.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = ssladapter.cc; path = rtc_base/ssladapter.cc; sourceTree = "<group>"; };
> 		413A22261FE18DCD00373E99 /* thread_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = thread_unittest.cc; path = rtc_base/thread_unittest.cc; sourceTree = "<group>"; };
>-		413A22271FE18DCE00373E99 /* transformadapter.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = transformadapter.h; path = rtc_base/transformadapter.h; sourceTree = "<group>"; };
> 		413A22281FE18DCE00373E99 /* timestampaligner.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = timestampaligner.cc; path = rtc_base/timestampaligner.cc; sourceTree = "<group>"; };
> 		413A22291FE18DCF00373E99 /* networkmonitor.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = networkmonitor.h; path = rtc_base/networkmonitor.h; sourceTree = "<group>"; };
> 		413A222A1FE18DCF00373E99 /* task_queue_posix.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = task_queue_posix.h; path = rtc_base/task_queue_posix.h; sourceTree = "<group>"; };
>@@ -3267,14 +3501,11 @@
> 		413A223B1FE18DD900373E99 /* buffer.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = buffer.h; path = rtc_base/buffer.h; sourceTree = "<group>"; };
> 		413A223C1FE18DD900373E99 /* event_tracer.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = event_tracer.cc; path = rtc_base/event_tracer.cc; sourceTree = "<group>"; };
> 		413A223D1FE18DD900373E99 /* pathutils.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = pathutils.h; path = rtc_base/pathutils.h; sourceTree = "<group>"; };
>-		413A223E1FE18DDA00373E99 /* md5.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = md5.h; path = rtc_base/md5.h; sourceTree = "<group>"; };
> 		413A223F1FE18DDA00373E99 /* sslroots.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = sslroots.h; path = rtc_base/sslroots.h; sourceTree = "<group>"; };
> 		413A22401FE18DDB00373E99 /* openssldigest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = openssldigest.cc; path = rtc_base/openssldigest.cc; sourceTree = "<group>"; };
>-		413A22411FE18DDB00373E99 /* ratelimiter.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = ratelimiter.h; path = rtc_base/ratelimiter.h; sourceTree = "<group>"; };
> 		413A22421FE18DDB00373E99 /* httpcommon.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = httpcommon.h; path = rtc_base/httpcommon.h; sourceTree = "<group>"; };
> 		413A22431FE18DDC00373E99 /* asyncsocket.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = asyncsocket.cc; path = rtc_base/asyncsocket.cc; sourceTree = "<group>"; };
> 		413A22441FE18DDC00373E99 /* macutils.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = macutils.h; path = rtc_base/macutils.h; sourceTree = "<group>"; };
>-		413A22451FE18DDC00373E99 /* md5.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = md5.cc; path = rtc_base/md5.cc; sourceTree = "<group>"; };
> 		413A22461FE18DDD00373E99 /* sigslottester_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = sigslottester_unittest.cc; path = rtc_base/sigslottester_unittest.cc; sourceTree = "<group>"; };
> 		413A22471FE18DDD00373E99 /* stringize_macros_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = stringize_macros_unittest.cc; path = rtc_base/stringize_macros_unittest.cc; sourceTree = "<group>"; };
> 		413A22481FE18DDE00373E99 /* bufferqueue.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = bufferqueue.cc; path = rtc_base/bufferqueue.cc; sourceTree = "<group>"; };
>@@ -3282,7 +3513,6 @@
> 		413A224A1FE18DDF00373E99 /* sequenced_task_checker_impl.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = sequenced_task_checker_impl.cc; path = rtc_base/sequenced_task_checker_impl.cc; sourceTree = "<group>"; };
> 		413A224B1FE18DDF00373E99 /* timestampaligner_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = timestampaligner_unittest.cc; path = rtc_base/timestampaligner_unittest.cc; sourceTree = "<group>"; };
> 		413A224C1FE18DE000373E99 /* unittest_main.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = unittest_main.cc; path = rtc_base/unittest_main.cc; sourceTree = "<group>"; };
>-		413A224D1FE18DE000373E99 /* md5digest.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = md5digest.h; path = rtc_base/md5digest.h; sourceTree = "<group>"; };
> 		413A224E1FE18DE000373E99 /* nethelpers.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = nethelpers.cc; path = rtc_base/nethelpers.cc; sourceTree = "<group>"; };
> 		413A224F1FE18DE100373E99 /* ssladapter.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = ssladapter.h; path = rtc_base/ssladapter.h; sourceTree = "<group>"; };
> 		413A22501FE18DE100373E99 /* stream_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = stream_unittest.cc; path = rtc_base/stream_unittest.cc; sourceTree = "<group>"; };
>@@ -3330,7 +3560,6 @@
> 		413A227A1FE18DF800373E99 /* ifaddrs_converter.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = ifaddrs_converter.h; path = rtc_base/ifaddrs_converter.h; sourceTree = "<group>"; };
> 		413A227B1FE18DF800373E99 /* physicalsocketserver_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = physicalsocketserver_unittest.cc; path = rtc_base/physicalsocketserver_unittest.cc; sourceTree = "<group>"; };
> 		413A227C1FE18DF800373E99 /* nullsocketserver_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = nullsocketserver_unittest.cc; path = rtc_base/nullsocketserver_unittest.cc; sourceTree = "<group>"; };
>-		413A227D1FE18DF900373E99 /* event_tracer_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = event_tracer_unittest.cc; path = rtc_base/event_tracer_unittest.cc; sourceTree = "<group>"; };
> 		413A227E1FE18DF900373E99 /* openssladapter_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = openssladapter_unittest.cc; path = rtc_base/openssladapter_unittest.cc; sourceTree = "<group>"; };
> 		413A227F1FE18DF900373E99 /* nullsocketserver.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = nullsocketserver.cc; path = rtc_base/nullsocketserver.cc; sourceTree = "<group>"; };
> 		413A22801FE18DFA00373E99 /* stream.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = stream.cc; path = rtc_base/stream.cc; sourceTree = "<group>"; };
>@@ -3342,11 +3571,9 @@
> 		413A22861FE18DFC00373E99 /* testutils.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = testutils.h; path = rtc_base/testutils.h; sourceTree = "<group>"; };
> 		413A22871FE18DFD00373E99 /* filerotatingstream.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = filerotatingstream.cc; path = rtc_base/filerotatingstream.cc; sourceTree = "<group>"; };
> 		413A22881FE18DFD00373E99 /* rtccertificate_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = rtccertificate_unittest.cc; path = rtc_base/rtccertificate_unittest.cc; sourceTree = "<group>"; };
>-		413A22891FE18DFD00373E99 /* base64.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = base64.cc; path = rtc_base/base64.cc; sourceTree = "<group>"; };
> 		413A228A1FE18DFE00373E99 /* thread_darwin.mm */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.objcpp; name = thread_darwin.mm; path = rtc_base/thread_darwin.mm; sourceTree = "<group>"; };
> 		413A228B1FE18DFE00373E99 /* task_queue_gcd.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = task_queue_gcd.cc; path = rtc_base/task_queue_gcd.cc; sourceTree = "<group>"; };
> 		413A228C1FE18DFE00373E99 /* firewallsocketserver.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = firewallsocketserver.h; path = rtc_base/firewallsocketserver.h; sourceTree = "<group>"; };
>-		413A228D1FE18DFF00373E99 /* timedelta.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = timedelta.h; path = rtc_base/timedelta.h; sourceTree = "<group>"; };
> 		413A228E1FE18DFF00373E99 /* timeutils_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = timeutils_unittest.cc; path = rtc_base/timeutils_unittest.cc; sourceTree = "<group>"; };
> 		413A228F1FE18DFF00373E99 /* BUILD.gn */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = text; name = BUILD.gn; path = rtc_base/BUILD.gn; sourceTree = "<group>"; };
> 		413A22901FE18E0000373E99 /* cpu_time.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = cpu_time.h; path = rtc_base/cpu_time.h; sourceTree = "<group>"; };
>@@ -3354,7 +3581,6 @@
> 		413A22921FE18E0000373E99 /* arraysize.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = arraysize.h; path = rtc_base/arraysize.h; sourceTree = "<group>"; };
> 		413A22931FE18E0100373E99 /* ignore_wundef.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = ignore_wundef.h; path = rtc_base/ignore_wundef.h; sourceTree = "<group>"; };
> 		413A22941FE18E0100373E99 /* network.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = network.cc; path = rtc_base/network.cc; sourceTree = "<group>"; };
>-		413A22951FE18E0200373E99 /* sha1.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = sha1.cc; path = rtc_base/sha1.cc; sourceTree = "<group>"; };
> 		413A22961FE18E0200373E99 /* copyonwritebuffer_unittest.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = copyonwritebuffer_unittest.cc; path = rtc_base/copyonwritebuffer_unittest.cc; sourceTree = "<group>"; };
> 		413A22971FE18E0200373E99 /* ifaddrs-android.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = "ifaddrs-android.cc"; path = "rtc_base/ifaddrs-android.cc"; sourceTree = "<group>"; };
> 		413A22981FE18E0300373E99 /* physicalsocketserver.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = physicalsocketserver.h; path = rtc_base/physicalsocketserver.h; sourceTree = "<group>"; };
>@@ -3367,11 +3593,7 @@
> 		413A229F1FE18E0600373E99 /* sslfingerprint.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = sslfingerprint.cc; path = rtc_base/sslfingerprint.cc; sourceTree = "<group>"; };
> 		413A22A01FE18E0600373E99 /* bytebuffer.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = bytebuffer.cc; path = rtc_base/bytebuffer.cc; sourceTree = "<group>"; };
> 		413A23F41FE1987B00373E99 /* RTCVideoEncoderH264.mm */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.objcpp; path = RTCVideoEncoderH264.mm; sourceTree = "<group>"; };
>-		413A23F51FE1987B00373E99 /* objc_video_encoder_factory.mm */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.objcpp; path = objc_video_encoder_factory.mm; sourceTree = "<group>"; };
>-		413A23F61FE1987B00373E99 /* objc_video_decoder_factory.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = objc_video_decoder_factory.h; sourceTree = "<group>"; };
>-		413A23F71FE1987C00373E99 /* objc_video_encoder_factory.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = objc_video_encoder_factory.h; sourceTree = "<group>"; };
> 		413A23F81FE1987C00373E99 /* RTCVideoDecoderH264.mm */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.objcpp; path = RTCVideoDecoderH264.mm; sourceTree = "<group>"; };
>-		413A23F91FE1987C00373E99 /* objc_video_decoder_factory.mm */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.objcpp; path = objc_video_decoder_factory.mm; sourceTree = "<group>"; };
> 		413A24021FE1990300373E99 /* RTCVideoViewShading.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = RTCVideoViewShading.h; path = sdk/objc/Framework/Headers/WebRTC/RTCVideoViewShading.h; sourceTree = "<group>"; };
> 		413A24031FE1990300373E99 /* RTCRtpReceiver.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = RTCRtpReceiver.h; path = sdk/objc/Framework/Headers/WebRTC/RTCRtpReceiver.h; sourceTree = "<group>"; };
> 		413A24041FE1990300373E99 /* RTCVideoCodecH264.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = RTCVideoCodecH264.h; path = sdk/objc/Framework/Headers/WebRTC/RTCVideoCodecH264.h; sourceTree = "<group>"; };
>@@ -3422,26 +3644,26 @@
> 		413A24321FE1991600373E99 /* RTCMetricsSampleInfo.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = RTCMetricsSampleInfo.h; path = sdk/objc/Framework/Headers/WebRTC/RTCMetricsSampleInfo.h; sourceTree = "<group>"; };
> 		413A24331FE1991600373E99 /* UIDevice+RTCDevice.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = "UIDevice+RTCDevice.h"; path = "sdk/objc/Framework/Headers/WebRTC/UIDevice+RTCDevice.h"; sourceTree = "<group>"; };
> 		413A24341FE1991700373E99 /* RTCRtpCodecParameters.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = RTCRtpCodecParameters.h; path = sdk/objc/Framework/Headers/WebRTC/RTCRtpCodecParameters.h; sourceTree = "<group>"; };
>-		413A24351FE1991700373E99 /* RTCAVFoundationVideoSource.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = RTCAVFoundationVideoSource.h; path = sdk/objc/Framework/Headers/WebRTC/RTCAVFoundationVideoSource.h; sourceTree = "<group>"; };
> 		413A24361FE1991800373E99 /* RTCVideoEncoderVP8.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = RTCVideoEncoderVP8.h; path = sdk/objc/Framework/Headers/WebRTC/RTCVideoEncoderVP8.h; sourceTree = "<group>"; };
> 		413A24371FE1991800373E99 /* RTCVideoRenderer.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = RTCVideoRenderer.h; path = sdk/objc/Framework/Headers/WebRTC/RTCVideoRenderer.h; sourceTree = "<group>"; };
> 		413A24381FE1991900373E99 /* RTCDataChannelConfiguration.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = RTCDataChannelConfiguration.h; path = sdk/objc/Framework/Headers/WebRTC/RTCDataChannelConfiguration.h; sourceTree = "<group>"; };
> 		413A24391FE1991900373E99 /* RTCMTLVideoView.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = RTCMTLVideoView.h; path = sdk/objc/Framework/Headers/WebRTC/RTCMTLVideoView.h; sourceTree = "<group>"; };
>+		413AD1A121265B0F003F7263 /* algorithm.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = algorithm.h; path = "Source/third_party/abseil-cpp/absl/algorithm/algorithm.h"; sourceTree = SOURCE_ROOT; };
>+		413AD1A221265B0F003F7263 /* container.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = container.h; path = "Source/third_party/abseil-cpp/absl/algorithm/container.h"; sourceTree = SOURCE_ROOT; };
>+		413AD1A721265B59003F7263 /* optimization.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; name = optimization.h; path = "Source/third_party/abseil-cpp/absl/base/optimization.h"; sourceTree = SOURCE_ROOT; };
>+		413AD1A821265B59003F7263 /* dynamic_annotations.cc */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.cpp.cpp; name = dynamic_annotations.cc; path = "Source/third_party/abseil-cpp/absl/base/dynamic_annotations.cc"; sourceTree = SOURCE_ROOT; };
>+		413AD1A921265B59003F7263 /* casts.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; name = casts.h; path = "Source/third_party/abseil-cpp/absl/base/casts.h"; sourceTree = SOURCE_ROOT; };
>+		413AD1AA21265B59003F7263 /* macros.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; name = macros.h; path = "Source/third_party/abseil-cpp/absl/base/macros.h"; sourceTree = SOURCE_ROOT; };
> 		4140B8181E4E3383007409E6 /* audio_encoder_pcm.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = audio_encoder_pcm.cc; path = g711/audio_encoder_pcm.cc; sourceTree = "<group>"; };
> 		4140B8191E4E3383007409E6 /* audio_encoder_pcm.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = audio_encoder_pcm.h; path = g711/audio_encoder_pcm.h; sourceTree = "<group>"; };
> 		4140B81A1E4E3383007409E6 /* audio_decoder_pcm.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = audio_decoder_pcm.cc; path = g711/audio_decoder_pcm.cc; sourceTree = "<group>"; };
> 		4140B81B1E4E3383007409E6 /* audio_decoder_pcm.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = audio_decoder_pcm.h; path = g711/audio_decoder_pcm.h; sourceTree = "<group>"; };
> 		4140B81C1E4E3383007409E6 /* g711_interface.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = g711_interface.c; path = g711/g711_interface.c; sourceTree = "<group>"; };
> 		4140B81D1E4E3383007409E6 /* g711_interface.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = g711_interface.h; path = g711/g711_interface.h; sourceTree = "<group>"; };
>-		4140B81E1E4E3383007409E6 /* g711.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = g711.c; path = g711/g711.c; sourceTree = "<group>"; };
>-		4140B81F1E4E3383007409E6 /* g711.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = g711.h; path = g711/g711.h; sourceTree = "<group>"; };
> 		4140B8281E4E3396007409E6 /* audio_decoder_g722.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = audio_decoder_g722.cc; path = g722/audio_decoder_g722.cc; sourceTree = "<group>"; };
> 		4140B8291E4E3396007409E6 /* audio_decoder_g722.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = audio_decoder_g722.h; path = g722/audio_decoder_g722.h; sourceTree = "<group>"; };
> 		4140B82A1E4E3396007409E6 /* audio_encoder_g722.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = audio_encoder_g722.cc; path = g722/audio_encoder_g722.cc; sourceTree = "<group>"; };
> 		4140B82B1E4E3396007409E6 /* audio_encoder_g722.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = audio_encoder_g722.h; path = g722/audio_encoder_g722.h; sourceTree = "<group>"; };
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>+		419241B72127593D00634FCF /* loss_rate_filter.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = loss_rate_filter.cc; sourceTree = "<group>"; };
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>+		419241B92127593E00634FCF /* bbr_network_controller.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = bbr_network_controller.cc; sourceTree = "<group>"; };
>+		419241BA2127593E00634FCF /* packet_number_indexed_queue.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = packet_number_indexed_queue.h; sourceTree = "<group>"; };
>+		419241BB2127593E00634FCF /* bbr_factory.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = bbr_factory.cc; sourceTree = "<group>"; };
>+		419241BC2127593E00634FCF /* rtt_stats.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = rtt_stats.cc; sourceTree = "<group>"; };
>+		419241BD2127593E00634FCF /* rtt_stats.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = rtt_stats.h; sourceTree = "<group>"; };
>+		419241BE2127593F00634FCF /* bandwidth_sampler.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = bandwidth_sampler.h; sourceTree = "<group>"; };
>+		419241BF2127593F00634FCF /* data_transfer_tracker.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = data_transfer_tracker.h; sourceTree = "<group>"; };
>+		419241CE2127597C00634FCF /* simulated_network.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = simulated_network.cc; path = call/simulated_network.cc; sourceTree = "<group>"; };
>+		419241CF2127597C00634FCF /* simulated_network.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = simulated_network.h; path = call/simulated_network.h; sourceTree = "<group>"; };
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>+		419241D721275A2F00634FCF /* data_size.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = data_size.cc; sourceTree = "<group>"; };
>+		419241D821275A2F00634FCF /* timestamp.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = timestamp.h; sourceTree = "<group>"; };
>+		419241D921275A2F00634FCF /* data_rate.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = data_rate.cc; sourceTree = "<group>"; };
>+		419241DA21275A3000634FCF /* time_delta.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = time_delta.cc; sourceTree = "<group>"; };
>+		419241DB21275A3000634FCF /* timestamp.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = timestamp.cc; sourceTree = "<group>"; };
>+		419241DC21275A3000634FCF /* data_size.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = data_size.h; sourceTree = "<group>"; };
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>+		419241E621275A7600634FCF /* field_trial_units.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = field_trial_units.cc; sourceTree = "<group>"; };
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>+		419241EA21275AFA00634FCF /* simulcast_rate_allocator.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = simulcast_rate_allocator.cc; sourceTree = "<group>"; };
>+		419241EB21275AFA00634FCF /* simulcast_utility.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = simulcast_utility.cc; sourceTree = "<group>"; };
>+		419241EC21275AFA00634FCF /* simulcast_rate_allocator.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = simulcast_rate_allocator.h; sourceTree = "<group>"; };
>+		419241F221275C3100634FCF /* vp8_encoder_simulcast_proxy.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = vp8_encoder_simulcast_proxy.h; path = engine/vp8_encoder_simulcast_proxy.h; sourceTree = "<group>"; };
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>+		419241FB2127664100634FCF /* audio_device_module.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = audio_device_module.h; sourceTree = "<group>"; };
>+		419241FC2127664100634FCF /* audio_device_module.mm */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.objcpp; path = audio_device_module.mm; sourceTree = "<group>"; };
>+		419241FD2127664100634FCF /* video_capturer.mm */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.objcpp; path = video_capturer.mm; sourceTree = "<group>"; };
>+		419241FE2127664100634FCF /* video_renderer.mm */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.objcpp; path = video_renderer.mm; sourceTree = "<group>"; };
>+		419241FF2127664100634FCF /* video_frame.mm */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.objcpp; path = video_frame.mm; sourceTree = "<group>"; };
>+		419242002127664200634FCF /* video_frame.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = video_frame.h; sourceTree = "<group>"; };
>+		419242012127664200634FCF /* video_decoder_factory.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = video_decoder_factory.h; sourceTree = "<group>"; };
>+		419242022127664200634FCF /* video_renderer.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = video_renderer.h; sourceTree = "<group>"; };
>+		419242032127664200634FCF /* video_decoder_factory.mm */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.objcpp; path = video_decoder_factory.mm; sourceTree = "<group>"; };
>+		419242042127664500634FCF /* video_encoder_factory.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = video_encoder_factory.h; sourceTree = "<group>"; };
>+		419242052127664500634FCF /* video_encoder_factory.mm */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.objcpp; path = video_encoder_factory.mm; sourceTree = "<group>"; };
>+		419242062127664600634FCF /* video_capturer.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = video_capturer.h; sourceTree = "<group>"; };
>+		419242072127664600634FCF /* video_frame_buffer.mm */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.objcpp; path = video_frame_buffer.mm; sourceTree = "<group>"; };
>+		419242082127664700634FCF /* video_frame_buffer.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = video_frame_buffer.h; sourceTree = "<group>"; };
>+		419242172127664B00634FCF /* objc_video_encoder_factory.mm */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.objcpp; path = objc_video_encoder_factory.mm; sourceTree = "<group>"; };
>+		419242182127664B00634FCF /* objc_video_decoder_factory.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = objc_video_decoder_factory.h; sourceTree = "<group>"; };
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>+		4192421A2127664C00634FCF /* objc_video_decoder_factory.mm */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.objcpp; path = objc_video_decoder_factory.mm; sourceTree = "<group>"; };
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>+		4192421D2127664D00634FCF /* objc_frame_buffer.mm */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.objcpp; path = objc_frame_buffer.mm; sourceTree = "<group>"; };
>+		4192421E2127664D00634FCF /* objc_video_frame.mm */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.objcpp; path = objc_video_frame.mm; sourceTree = "<group>"; };
>+		4192421F2127664D00634FCF /* objc_video_renderer.mm */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.objcpp; path = objc_video_renderer.mm; sourceTree = "<group>"; };
>+		419242202127664D00634FCF /* objc_video_track_source.mm */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.objcpp; path = objc_video_track_source.mm; sourceTree = "<group>"; };
>+		419242212127664E00634FCF /* objc_video_renderer.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = objc_video_renderer.h; sourceTree = "<group>"; };
>+		419242222127664E00634FCF /* objc_video_encoder_factory.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = objc_video_encoder_factory.h; sourceTree = "<group>"; };
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>+		419242342127665A00634FCF /* voice_processing_audio_unit.mm */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.objcpp; name = voice_processing_audio_unit.mm; path = audio/voice_processing_audio_unit.mm; sourceTree = "<group>"; };
>+		419242352127665A00634FCF /* voice_processing_audio_unit.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = voice_processing_audio_unit.h; path = audio/voice_processing_audio_unit.h; sourceTree = "<group>"; };
>+		4192423F21276D4700634FCF /* RTCPeerConnectionFactory.mm */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.objcpp; name = RTCPeerConnectionFactory.mm; path = PeerConnection/RTCPeerConnectionFactory.mm; sourceTree = "<group>"; };
> 		419C82991FE20CA10040C30F /* interval_budget.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = interval_budget.cc; sourceTree = "<group>"; };
> 		419C829B1FE20D1B0040C30F /* audio_processing_statistics.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = audio_processing_statistics.cc; path = include/audio_processing_statistics.cc; sourceTree = "<group>"; };
> 		419C829C1FE20D1C0040C30F /* audio_processing_statistics.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = audio_processing_statistics.h; path = include/audio_processing_statistics.h; sourceTree = "<group>"; };
>@@ -3565,7 +3930,6 @@
> 		419C82A51FE20DC80040C30F /* rtp_config.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = rtp_config.h; path = call/rtp_config.h; sourceTree = "<group>"; };
> 		419C82A61FE20DC80040C30F /* rtcp_demuxer.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = rtcp_demuxer.cc; path = call/rtcp_demuxer.cc; sourceTree = "<group>"; };
> 		419C82A71FE20DC90040C30F /* video_send_stream.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = video_send_stream.h; path = call/video_send_stream.h; sourceTree = "<group>"; };
>-		419C82A81FE20DCA0040C30F /* video_config.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = video_config.cc; path = call/video_config.cc; sourceTree = "<group>"; };
> 		419C82A91FE20DCA0040C30F /* rtp_stream_receiver_controller.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = rtp_stream_receiver_controller.cc; path = call/rtp_stream_receiver_controller.cc; sourceTree = "<group>"; };
> 		419C82AA1FE20DCA0040C30F /* rtcp_packet_sink_interface.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = rtcp_packet_sink_interface.h; path = call/rtcp_packet_sink_interface.h; sourceTree = "<group>"; };
> 		419C82AB1FE20DCB0040C30F /* video_receive_stream.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = video_receive_stream.cc; path = call/video_receive_stream.cc; sourceTree = "<group>"; };
>@@ -3573,11 +3937,9 @@
> 		419C82AD1FE20DCB0040C30F /* rtp_stream_receiver_controller_interface.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = rtp_stream_receiver_controller_interface.h; path = call/rtp_stream_receiver_controller_interface.h; sourceTree = "<group>"; };
> 		419C82AE1FE20DCC0040C30F /* rtp_stream_receiver_controller.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = rtp_stream_receiver_controller.h; path = call/rtp_stream_receiver_controller.h; sourceTree = "<group>"; };
> 		419C82AF1FE20DCC0040C30F /* rtx_receive_stream.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = rtx_receive_stream.h; path = call/rtx_receive_stream.h; sourceTree = "<group>"; };
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> 		419C82B11FE20DCD0040C30F /* video_receive_stream.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = video_receive_stream.h; path = call/video_receive_stream.h; sourceTree = "<group>"; };
> 		419C82C71FE20E2B0040C30F /* rtpmediautils.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = rtpmediautils.cc; sourceTree = "<group>"; };
> 		419C82C81FE20E2C0040C30F /* rtpmediautils.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = rtpmediautils.h; sourceTree = "<group>"; };
>-		419C82CB1FE20E540040C30F /* transportcontroller.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = transportcontroller.h; sourceTree = "<group>"; };
> 		419C82CC1FE20E540040C30F /* rtptransceiver.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = rtptransceiver.h; sourceTree = "<group>"; };
> 		419C82CD1FE20E550040C30F /* rtptransportinternaladapter.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = rtptransportinternaladapter.h; sourceTree = "<group>"; };
> 		419C82CE1FE20E550040C30F /* sdputils.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = sdputils.cc; sourceTree = "<group>"; };
>@@ -3587,11 +3949,9 @@
> 		419C82D21FE20E570040C30F /* dtlssrtptransport.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = dtlssrtptransport.h; sourceTree = "<group>"; };
> 		419C82D31FE20E570040C30F /* srtpsession.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = srtpsession.h; sourceTree = "<group>"; };
> 		419C82D41FE20E570040C30F /* srtptransport.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = srtptransport.h; sourceTree = "<group>"; };
>-		419C82D51FE20E580040C30F /* transportcontroller.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = transportcontroller.cc; sourceTree = "<group>"; };
> 		419C82D61FE20E580040C30F /* dtlssrtptransport.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = dtlssrtptransport.cc; sourceTree = "<group>"; };
> 		419C82D71FE20E580040C30F /* rtptransportinternal.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = rtptransportinternal.h; sourceTree = "<group>"; };
> 		419C82D81FE20E590040C30F /* rtptransceiver.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = rtptransceiver.cc; sourceTree = "<group>"; };
>-		419C82E81FE20E8E0040C30F /* optional.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = optional.cc; sourceTree = "<group>"; };
> 		419C82EA1FE20EB40040C30F /* audio_encoder_opus.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = audio_encoder_opus.h; path = audio_codecs/opus/audio_encoder_opus.h; sourceTree = "<group>"; };
> 		419C82EB1FE20EB40040C30F /* audio_encoder_opus_config.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = audio_encoder_opus_config.h; path = audio_codecs/opus/audio_encoder_opus_config.h; sourceTree = "<group>"; };
> 		419C82EC1FE20EB40040C30F /* audio_decoder_opus.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = audio_decoder_opus.h; path = audio_codecs/opus/audio_decoder_opus.h; sourceTree = "<group>"; };
>@@ -3603,16 +3963,13 @@
> 		419C82F81FE20EFC0040C30F /* rtcp_transceiver_config.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = rtcp_transceiver_config.h; sourceTree = "<group>"; };
> 		419C82F91FE20EFD0040C30F /* rtcp_transceiver_config.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = rtcp_transceiver_config.cc; sourceTree = "<group>"; };
> 		419C82FA1FE20EFD0040C30F /* rtcp_transceiver.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = rtcp_transceiver.h; sourceTree = "<group>"; };
>-		419C82FB1FE20EFE0040C30F /* rtp_format_video_stereo.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = rtp_format_video_stereo.h; sourceTree = "<group>"; };
> 		419C82FC1FE20EFE0040C30F /* rtcp_transceiver_impl.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = rtcp_transceiver_impl.h; sourceTree = "<group>"; };
>-		419C82FD1FE20EFE0040C30F /* rtp_format_video_stereo.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = rtp_format_video_stereo.cc; sourceTree = "<group>"; };
> 		419C82FE1FE20F010040C30F /* rtp_packet_received.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = rtp_packet_received.cc; sourceTree = "<group>"; };
> 		419C83081FE20F3B0040C30F /* fec_controller_rplr_based.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = fec_controller_rplr_based.h; path = audio_network_adaptor/fec_controller_rplr_based.h; sourceTree = "<group>"; };
> 		419C83091FE20F3B0040C30F /* fec_controller_rplr_based.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = fec_controller_rplr_based.cc; path = audio_network_adaptor/fec_controller_rplr_based.cc; sourceTree = "<group>"; };
> 		419C830A1FE20F3B0040C30F /* audio_network_adaptor_config.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = audio_network_adaptor_config.cc; path = audio_network_adaptor/audio_network_adaptor_config.cc; sourceTree = "<group>"; };
> 		419C830E1FE2427F0040C30F /* candidate.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = candidate.cc; sourceTree = "<group>"; };
> 		419C830F1FE2427F0040C30F /* proxy.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = proxy.cc; sourceTree = "<group>"; };
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> 		41A391461EFC446E00C4516A /* sha1-armv4-large.pl */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = text.script.perl; path = "sha1-armv4-large.pl"; sourceTree = "<group>"; };
> 		41A391471EFC446E00C4516A /* sha1-armv8.pl */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = text.script.perl; path = "sha1-armv8.pl"; sourceTree = "<group>"; };
>@@ -3898,6 +4280,100 @@
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>+		41F263A5212680A700274F59 /* fallthrough.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = fallthrough.h; sourceTree = "<group>"; };
>+		41F263A6212680A700274F59 /* inline.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = inline.h; sourceTree = "<group>"; };
>+		41F263A7212680A700274F59 /* unused.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = unused.h; sourceTree = "<group>"; };
>+		41F263A8212680A700274F59 /* BUILD.gn */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = text; path = BUILD.gn; sourceTree = "<group>"; };
>+		41F263A9212680A700274F59 /* asm_defines.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = asm_defines.h; sourceTree = "<group>"; };
>+		41F263AA212680A800274F59 /* file_wrapper.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = file_wrapper.h; sourceTree = "<group>"; };
>+		41F263AB212680A800274F59 /* arch.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = arch.h; sourceTree = "<group>"; };
>+		41F263B52126810B00274F59 /* base64.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = base64.cc; path = base64/base64.cc; sourceTree = "<group>"; };
>+		41F263B62126810C00274F59 /* base64.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = base64.h; path = base64/base64.h; sourceTree = "<group>"; };
>+		41F263B92126811400274F59 /* sigslot.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = sigslot.cc; path = sigslot/sigslot.cc; sourceTree = "<group>"; };
>+		41F263BA2126811400274F59 /* sigslot.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = sigslot.h; path = sigslot/sigslot.h; sourceTree = "<group>"; };
>+		41F263BD2126818800274F59 /* video_stream_encoder_observer.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; path = video_stream_encoder_observer.h; sourceTree = "<group>"; };
>+		41F263BE2126818800274F59 /* video_source_interface.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; path = video_source_interface.h; sourceTree = "<group>"; };
>+		41F263BF2126818900274F59 /* i010_buffer.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; path = i010_buffer.h; sourceTree = "<group>"; };
>+		41F263C02126818900274F59 /* video_stream_encoder_create.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; path = video_stream_encoder_create.h; sourceTree = "<group>"; };
>+		41F263C12126818900274F59 /* video_bitrate_allocator.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; path = video_bitrate_allocator.h; sourceTree = "<group>"; };
>+		41F263C22126818900274F59 /* encoded_frame.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; path = encoded_frame.h; sourceTree = "<group>"; };
>+		41F263C32126818900274F59 /* video_stream_encoder_interface.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; path = video_stream_encoder_interface.h; sourceTree = "<group>"; };
>+		41F263C42126818900274F59 /* video_bitrate_allocation.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; path = video_bitrate_allocation.h; sourceTree = "<group>"; };
>+		41F263C52126818A00274F59 /* video_stream_decoder.h */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.c.h; path = video_stream_decoder.h; sourceTree = "<group>"; };
>+		41F263C62126818A00274F59 /* video_source_interface.cc */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.cpp.cpp; path = video_source_interface.cc; sourceTree = "<group>"; };
>+		41F263C72126818A00274F59 /* encoded_frame.cc */ = {isa = PBXFileReference; lastKnownFileType = sourcecode.cpp.cpp; path = encoded_frame.cc; sourceTree = "<group>"; };
> 		41F411AC1EF8D91E00343C26 /* null_aec_dump_factory.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = null_aec_dump_factory.cc; path = Source/webrtc/modules/audio_processing/aec_dump/null_aec_dump_factory.cc; sourceTree = SOURCE_ROOT; };
> 		41F411AE1EF8DA0000343C26 /* rtcerror.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = rtcerror.cc; sourceTree = "<group>"; };
> 		41F411AF1EF8DA0000343C26 /* rtcerror.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = rtcerror.h; sourceTree = "<group>"; };
>@@ -3994,13 +4517,10 @@
> 		41F9BF782051C7FD00ABF0B9 /* jsep.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = jsep.cc; sourceTree = "<group>"; };
> 		41F9BF792051C7FD00ABF0B9 /* array_view.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = array_view.h; sourceTree = "<group>"; };
> 		41F9BF7A2051C7FE00ABF0B9 /* rtceventlogoutput.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = rtceventlogoutput.h; sourceTree = "<group>"; };
>-		41F9BF7B2051C7FE00ABF0B9 /* videosinkinterface.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = videosinkinterface.h; sourceTree = "<group>"; };
>-		41F9BF7C2051C7FE00ABF0B9 /* videosourceinterface.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = videosourceinterface.h; sourceTree = "<group>"; };
> 		41F9BF7D2051C7FE00ABF0B9 /* rtp_headers.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = rtp_headers.cc; sourceTree = "<group>"; };
> 		41F9BF7E2051C7FF00ABF0B9 /* refcountedbase.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = refcountedbase.h; sourceTree = "<group>"; };
> 		41F9BF7F2051C7FF00ABF0B9 /* rtp_headers.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = rtp_headers.h; sourceTree = "<group>"; };
> 		41F9BF802051C7FF00ABF0B9 /* rtptransceiverinterface.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = rtptransceiverinterface.h; sourceTree = "<group>"; };
>-		41F9BF812051C80000ABF0B9 /* videosourceinterface.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = videosourceinterface.cc; sourceTree = "<group>"; };
> 		41F9BF822051C80000ABF0B9 /* cryptoparams.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = cryptoparams.h; sourceTree = "<group>"; };
> 		41F9BF832051C80000ABF0B9 /* turncustomizer.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = turncustomizer.h; sourceTree = "<group>"; };
> 		41F9BF842051C80000ABF0B9 /* setremotedescriptionobserverinterface.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = setremotedescriptionobserverinterface.h; sourceTree = "<group>"; };
>@@ -4087,9 +4607,6 @@
> 		5C11A0081E457448004F0987 /* audio_device_mac.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = audio_device_mac.h; path = mac/audio_device_mac.h; sourceTree = "<group>"; };
> 		5C11A0091E457448004F0987 /* audio_mixer_manager_mac.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = audio_mixer_manager_mac.cc; path = mac/audio_mixer_manager_mac.cc; sourceTree = "<group>"; };
> 		5C11A00A1E457448004F0987 /* audio_mixer_manager_mac.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = audio_mixer_manager_mac.h; path = mac/audio_mixer_manager_mac.h; sourceTree = "<group>"; };
>-		5C11A0101E4574C5004F0987 /* pa_memorybarrier.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = pa_memorybarrier.h; path = mac/portaudio/pa_memorybarrier.h; sourceTree = "<group>"; };
>-		5C11A0111E4574C5004F0987 /* pa_ringbuffer.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = pa_ringbuffer.c; path = mac/portaudio/pa_ringbuffer.c; sourceTree = "<group>"; };
>-		5C11A0121E4574C5004F0987 /* pa_ringbuffer.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = pa_ringbuffer.h; path = mac/portaudio/pa_ringbuffer.h; sourceTree = "<group>"; };
> 		5C11A0171E457578004F0987 /* audio_device_dummy.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = audio_device_dummy.cc; path = dummy/audio_device_dummy.cc; sourceTree = "<group>"; };
> 		5C11A0181E457578004F0987 /* audio_device_dummy.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = audio_device_dummy.h; path = dummy/audio_device_dummy.h; sourceTree = "<group>"; };
> 		5C11A0191E457578004F0987 /* file_audio_device_factory.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = file_audio_device_factory.cc; path = dummy/file_audio_device_factory.cc; sourceTree = "<group>"; };
>@@ -4299,8 +4816,6 @@
> 		5C4B485D1E42C1BA002651C8 /* payload_type_mapper.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = payload_type_mapper.h; path = engine/payload_type_mapper.h; sourceTree = "<group>"; };
> 		5C4B485F1E42C1BA002651C8 /* simulcast.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = simulcast.cc; path = engine/simulcast.cc; sourceTree = "<group>"; };
> 		5C4B48601E42C1BA002651C8 /* simulcast.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = simulcast.h; path = engine/simulcast.h; sourceTree = "<group>"; };
>-		5C4B48621E42C1BA002651C8 /* videoencodersoftwarefallbackwrapper.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = videoencodersoftwarefallbackwrapper.cc; path = engine/videoencodersoftwarefallbackwrapper.cc; sourceTree = "<group>"; };
>-		5C4B48631E42C1BA002651C8 /* videoencodersoftwarefallbackwrapper.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = videoencodersoftwarefallbackwrapper.h; path = engine/videoencodersoftwarefallbackwrapper.h; sourceTree = "<group>"; };
> 		5C4B48661E42C1BA002651C8 /* webrtcmediaengine.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = webrtcmediaengine.cc; path = engine/webrtcmediaengine.cc; sourceTree = "<group>"; };
> 		5C4B48671E42C1BA002651C8 /* webrtcmediaengine.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = webrtcmediaengine.h; path = engine/webrtcmediaengine.h; sourceTree = "<group>"; };
> 		5C4B48691E42C1BA002651C8 /* webrtcvideocapturer.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = webrtcvideocapturer.cc; path = engine/webrtcvideocapturer.cc; sourceTree = "<group>"; };
>@@ -4440,8 +4955,6 @@
> 		5C4B4C2D1E431F9C002651C8 /* blocker.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = blocker.h; sourceTree = "<group>"; };
> 		5C4B4C301E431F9C002651C8 /* channel_buffer.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = channel_buffer.cc; sourceTree = "<group>"; };
> 		5C4B4C311E431F9C002651C8 /* channel_buffer.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = channel_buffer.h; sourceTree = "<group>"; };
>-		5C4B4C341E431F9C002651C8 /* fft4g.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; path = fft4g.c; sourceTree = "<group>"; };
>-		5C4B4C351E431F9C002651C8 /* fft4g.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = fft4g.h; sourceTree = "<group>"; };
> 		5C4B4C381E431F9C002651C8 /* fir_filter_sse.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = fir_filter_sse.cc; sourceTree = "<group>"; };
> 		5C4B4C391E431F9C002651C8 /* fir_filter_sse.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = fir_filter_sse.h; sourceTree = "<group>"; };
> 		5C4B4C3C1E431F9C002651C8 /* fir_filter.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = fir_filter.h; sourceTree = "<group>"; };
>@@ -4461,7 +4974,6 @@
> 		5C4B4C541E431F9C002651C8 /* wav_header.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = wav_header.h; sourceTree = "<group>"; };
> 		5C4B4C561E431F9C002651C8 /* window_generator.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = window_generator.cc; sourceTree = "<group>"; };
> 		5C4B4C571E431F9C002651C8 /* window_generator.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = window_generator.h; sourceTree = "<group>"; };
>-		5C4B4C8F1E4320A9002651C8 /* aligned_malloc.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = aligned_malloc.cc; path = source/aligned_malloc.cc; sourceTree = "<group>"; };
> 		5C4B4C941E4320A9002651C8 /* clock.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = clock.cc; path = source/clock.cc; sourceTree = "<group>"; };
> 		5C4B4C981E4320A9002651C8 /* cpu_features.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = cpu_features.cc; path = source/cpu_features.cc; sourceTree = "<group>"; };
> 		5C4B4C991E4320A9002651C8 /* cpu_info.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = cpu_info.cc; path = source/cpu_info.cc; sourceTree = "<group>"; };
>@@ -4469,19 +4981,12 @@
> 		5C4B4C9D1E4320A9002651C8 /* event_timer_posix.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = event_timer_posix.h; path = source/event_timer_posix.h; sourceTree = "<group>"; };
> 		5C4B4CA01E4320A9002651C8 /* event.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = event.cc; path = source/event.cc; sourceTree = "<group>"; };
> 		5C4B4CA11E4320A9002651C8 /* field_trial_default.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = field_trial_default.cc; path = source/field_trial_default.cc; sourceTree = "<group>"; };
>-		5C4B4CA21E4320A9002651C8 /* file_impl.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = file_impl.cc; path = source/file_impl.cc; sourceTree = "<group>"; };
> 		5C4B4CA71E4320A9002651C8 /* metrics_default.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = metrics_default.cc; path = source/metrics_default.cc; sourceTree = "<group>"; };
>-		5C4B4CAD1E4320A9002651C8 /* rw_lock_posix.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = rw_lock_posix.cc; path = source/rw_lock_posix.cc; sourceTree = "<group>"; };
>-		5C4B4CAE1E4320A9002651C8 /* rw_lock_posix.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = rw_lock_posix.h; path = source/rw_lock_posix.h; sourceTree = "<group>"; };
>-		5C4B4CB31E4320A9002651C8 /* rw_lock.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = rw_lock.cc; path = source/rw_lock.cc; sourceTree = "<group>"; };
> 		5C4B4CB41E4320A9002651C8 /* sleep.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = sleep.cc; path = source/sleep.cc; sourceTree = "<group>"; };
>-		5C4B4CB81E4320A9002651C8 /* timestamp_extrapolator.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = timestamp_extrapolator.cc; path = source/timestamp_extrapolator.cc; sourceTree = "<group>"; };
> 		5C4B4CF61E432168002651C8 /* spake25519.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = spake25519.c; path = curve25519/spake25519.c; sourceTree = "<group>"; };
> 		5C4B4CF81E432168002651C8 /* x25519-x86_64.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = "x25519-x86_64.c"; path = "curve25519/x25519-x86_64.c"; sourceTree = "<group>"; };
> 		5C4B4D051E4322F6002651C8 /* basicportallocator.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = basicportallocator.cc; sourceTree = "<group>"; };
> 		5C4B4D061E4322F6002651C8 /* basicportallocator.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = basicportallocator.h; sourceTree = "<group>"; };
>-		5C4B4D071E4322F6002651C8 /* socketmonitor.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = socketmonitor.cc; sourceTree = "<group>"; };
>-		5C4B4D081E4322F6002651C8 /* socketmonitor.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = socketmonitor.h; sourceTree = "<group>"; };
> 		5C4B4D2F1E4323D4002651C8 /* compare_common.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = compare_common.cc; path = source/compare_common.cc; sourceTree = "<group>"; };
> 		5C4B4D341E4323D4002651C8 /* compare.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = compare.cc; path = source/compare.cc; sourceTree = "<group>"; };
> 		5C4B4D351E4323D4002651C8 /* convert_argb.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; name = convert_argb.cc; path = source/convert_argb.cc; sourceTree = "<group>"; };
>@@ -4522,8 +5027,6 @@
> 		5C63F8D81E416D53002CA531 /* writer.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = writer.h; path = json/writer.h; sourceTree = "<group>"; };
> 		5C63F8E71E41737B002CA531 /* datachannelinterface.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = datachannelinterface.h; sourceTree = "<group>"; };
> 		5C63F8EB1E41737B002CA531 /* dtmfsenderinterface.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = dtmfsenderinterface.h; sourceTree = "<group>"; };
>-		5C63F8ED1E41737B002CA531 /* fakemetricsobserver.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = fakemetricsobserver.cc; sourceTree = "<group>"; };
>-		5C63F8EE1E41737B002CA531 /* fakemetricsobserver.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = fakemetricsobserver.h; sourceTree = "<group>"; };
> 		5C63F8EF1E41737B002CA531 /* jsep.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = jsep.h; sourceTree = "<group>"; };
> 		5C63F8F11E41737B002CA531 /* jsepicecandidate.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = jsepicecandidate.h; sourceTree = "<group>"; };
> 		5C63F8F31E41737B002CA531 /* jsepsessiondescription.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = jsepsessiondescription.h; sourceTree = "<group>"; };
>@@ -4549,7 +5052,6 @@
> 		5C63F98C1E4174F6002CA531 /* basicpacketsocketfactory.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = basicpacketsocketfactory.cc; sourceTree = "<group>"; };
> 		5C63F98D1E4174F6002CA531 /* basicpacketsocketfactory.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = basicpacketsocketfactory.h; sourceTree = "<group>"; };
> 		5C63F98F1E4174F6002CA531 /* candidatepairinterface.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = candidatepairinterface.h; sourceTree = "<group>"; };
>-		5C63F9901E4174F6002CA531 /* common.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = common.h; sourceTree = "<group>"; };
> 		5C63F9951E4174F6002CA531 /* fakeportallocator.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = fakeportallocator.h; sourceTree = "<group>"; };
> 		5C63F9971E4174F6002CA531 /* p2pconstants.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = p2pconstants.cc; sourceTree = "<group>"; };
> 		5C63F9981E4174F6002CA531 /* p2pconstants.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = p2pconstants.h; sourceTree = "<group>"; };
>@@ -4568,8 +5070,6 @@
> 		5C63F9AC1E4174F6002CA531 /* relayport.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = relayport.h; sourceTree = "<group>"; };
> 		5C63F9AE1E4174F6002CA531 /* relayserver.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = relayserver.cc; sourceTree = "<group>"; };
> 		5C63F9AF1E4174F6002CA531 /* relayserver.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = relayserver.h; sourceTree = "<group>"; };
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>-		5C63F9B11E4174F6002CA531 /* session.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = session.h; sourceTree = "<group>"; };
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> 		5C63F9B61E4174F6002CA531 /* stun.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = stun.h; sourceTree = "<group>"; };
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>@@ -4594,34 +5094,25 @@
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> 		5CD284621E6A57F40094FDC8 /* i420_buffer.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = i420_buffer.h; sourceTree = "<group>"; };
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> 		5CD2858B1E6A63430094FDC8 /* matched_filter.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = matched_filter.cc; sourceTree = "<group>"; };
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> 		5CD285911E6A63430094FDC8 /* render_delay_buffer.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = render_delay_buffer.cc; sourceTree = "<group>"; };
> 		5CD285921E6A63430094FDC8 /* render_delay_buffer.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = render_delay_buffer.h; sourceTree = "<group>"; };
> 		5CD285931E6A63430094FDC8 /* render_delay_controller_metrics.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = render_delay_controller_metrics.cc; sourceTree = "<group>"; };
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> 		5CD285F91E6A64520094FDC8 /* rtcp_nack_stats.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = rtcp_nack_stats.h; sourceTree = "<group>"; };
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>@@ -4824,8 +5306,6 @@
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> 		5CDD83771E439A3500621E92 /* vp8_header_parser.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = vp8_header_parser.cc; sourceTree = "<group>"; };
> 		5CDD83781E439A3500621E92 /* vp8_header_parser.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = vp8_header_parser.h; sourceTree = "<group>"; };
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> 		5CDD83891E439A6F00621E92 /* codec_timer.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = codec_timer.cc; sourceTree = "<group>"; };
> 		5CDD838A1E439A6F00621E92 /* codec_timer.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = codec_timer.h; sourceTree = "<group>"; };
> 		5CDD838B1E439A6F00621E92 /* decoding_state.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = decoding_state.cc; sourceTree = "<group>"; };
>@@ -4865,8 +5345,6 @@
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> 		5CDD83AF1E439A6F00621E92 /* packet.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = packet.cc; sourceTree = "<group>"; };
> 		5CDD83B01E439A6F00621E92 /* packet.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = packet.h; sourceTree = "<group>"; };
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> 		5CDD83B51E439A6F00621E92 /* receiver.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = receiver.cc; sourceTree = "<group>"; };
> 		5CDD83B61E439A6F00621E92 /* receiver.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = receiver.h; sourceTree = "<group>"; };
> 		5CDD83B71E439A6F00621E92 /* rtp_frame_reference_finder.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = rtp_frame_reference_finder.cc; sourceTree = "<group>"; };
>@@ -4887,19 +5365,12 @@
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> 		5CDD840C1E439B2900621E92 /* audio_format_conversion.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = audio_format_conversion.cc; sourceTree = "<group>"; };
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> 		5CDD848A1E43AF1300621E92 /* audio_buffer.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = audio_buffer.h; sourceTree = "<group>"; };
> 		5CDD848B1E43AF1300621E92 /* audio_processing_impl.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = audio_processing_impl.cc; sourceTree = "<group>"; };
>@@ -4956,8 +5427,6 @@
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> 		5CDD85661E43B5C000621E92 /* overuse_frame_detector.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = overuse_frame_detector.cc; sourceTree = "<group>"; };
> 		5CDD85671E43B5C000621E92 /* overuse_frame_detector.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = overuse_frame_detector.h; sourceTree = "<group>"; };
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> 		5CDD856D1E43B5C000621E92 /* receive_statistics_proxy.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = receive_statistics_proxy.cc; sourceTree = "<group>"; };
> 		5CDD856E1E43B5C000621E92 /* receive_statistics_proxy.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = receive_statistics_proxy.h; sourceTree = "<group>"; };
> 		5CDD85711E43B5C000621E92 /* report_block_stats.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = report_block_stats.cc; sourceTree = "<group>"; };
>@@ -5028,7 +5497,6 @@
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> 		5CDD86371E43B8B500621E92 /* spl_inl.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = spl_inl.c; path = signal_processing/spl_inl.c; sourceTree = "<group>"; };
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> 		5CDD863B1E43B8B500621E92 /* spl_sqrt.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = spl_sqrt.c; path = signal_processing/spl_sqrt.c; sourceTree = "<group>"; };
> 		5CDD863C1E43B8B500621E92 /* splitting_filter.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = splitting_filter.c; path = signal_processing/splitting_filter.c; sourceTree = "<group>"; };
> 		5CDD863D1E43B8B500621E92 /* sqrt_of_one_minus_x_squared.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = sqrt_of_one_minus_x_squared.c; path = signal_processing/sqrt_of_one_minus_x_squared.c; sourceTree = "<group>"; };
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> 		5CDD87821E43BC0500621E92 /* entropy_coding.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = entropy_coding.c; path = source/entropy_coding.c; sourceTree = "<group>"; };
> 		5CDD87831E43BC0500621E92 /* entropy_coding.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = entropy_coding.h; path = source/entropy_coding.h; sourceTree = "<group>"; };
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> 		5CDD87891E43BC0500621E92 /* filterbanks.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = filterbanks.c; path = source/filterbanks.c; sourceTree = "<group>"; };
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> 		5CDD886E1E43BE3C00621E92 /* ulpfec_receiver_impl.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = ulpfec_receiver_impl.cc; sourceTree = "<group>"; };
> 		5CDD886F1E43BE3C00621E92 /* ulpfec_receiver_impl.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = ulpfec_receiver_impl.h; sourceTree = "<group>"; };
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> 		5CDD88ED1E43BF3A00621E92 /* app.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = app.cc; sourceTree = "<group>"; };
> 		5CDD88EE1E43BF3A00621E92 /* app.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = app.h; sourceTree = "<group>"; };
> 		5CDD88F01E43BF3A00621E92 /* bye.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = bye.cc; sourceTree = "<group>"; };
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> 		5CDD89AA1E43BFB300621E92 /* decoder_database.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = decoder_database.cc; sourceTree = "<group>"; };
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> 		5CDD8FB01E43CD0700621E92 /* transport_feedback_adapter.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = transport_feedback_adapter.cc; sourceTree = "<group>"; };
> 		5CDD8FB11E43CD0700621E92 /* transport_feedback_adapter.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = transport_feedback_adapter.h; sourceTree = "<group>"; };
> 		5CDD8FBE1E43CD6600621E92 /* aimd_rate_control.cc */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = aimd_rate_control.cc; sourceTree = "<group>"; };
>@@ -5841,7 +6287,6 @@
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>@@ -5883,15 +6307,6 @@
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>-		5CDD90731E43D15C00621E92 /* nonlinear_beamformer.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; name = nonlinear_beamformer.h; path = beamformer/nonlinear_beamformer.h; sourceTree = "<group>"; };
> 		5CDD90871E43D33800621E92 /* ekt.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = ekt.c; path = srtp/ekt.c; sourceTree = "<group>"; };
> 		5CDD90881E43D33800621E92 /* srtp.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = srtp.c; path = srtp/srtp.c; sourceTree = "<group>"; };
> 		5CDD908D1E43D4CC00621E92 /* rdb.c */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.c; name = rdb.c; path = replay/rdb.c; sourceTree = "<group>"; };
>@@ -6009,6 +6424,20 @@
> 		412455581EF88AD900F11809 /* video_codecs */ = {
> 			isa = PBXGroup;
> 			children = (
>+				41E02CC22127358700C27CD6 /* builtin_video_encoder_factory.cc */,
>+				41E02CC32127358800C27CD6 /* video_encoder_config.cc */,
>+				41DDB27321267AC000296D47 /* builtin_video_decoder_factory.cc */,
>+				41DDB27621267AC100296D47 /* builtin_video_decoder_factory.h */,
>+				41DDB27721267AC100296D47 /* builtin_video_encoder_factory.h */,
>+				41DDB27A21267AC200296D47 /* sdp_video_format.cc */,
>+				41DDB27021267AC000296D47 /* video_codec.cc */,
>+				41DDB27121267AC000296D47 /* video_codec.h */,
>+				41DDB27421267AC100296D47 /* video_decoder_software_fallback_wrapper.cc */,
>+				41DDB27821267AC100296D47 /* video_decoder_software_fallback_wrapper.h */,
>+				41DDB27221267AC000296D47 /* video_decoder.cc */,
>+				41DDB27B21267AC200296D47 /* video_encoder_config.h */,
>+				41DDB27521267AC100296D47 /* video_encoder_software_fallback_wrapper.cc */,
>+				41DDB27921267AC200296D47 /* video_encoder_software_fallback_wrapper.h */,
> 				4145F6191FE1F38B00EB9CAF /* sdp_video_format.h */,
> 				4145E48B1EF88B9600FCF6E6 /* video_decoder.h */,
> 				4145F61A1FE1F38C00EB9CAF /* video_decoder_factory.h */,
>@@ -6019,13 +6448,32 @@
> 			name = video_codecs;
> 			sourceTree = "<group>";
> 		};
>-		4130920B1EF8D2CD00757C55 /* agc2 */ = {
>+		41299B98212736B200B3414B /* rnn_vad */ = {
> 			isa = PBXGroup;
> 			children = (
>-				4130921F1EF8D6A200757C55 /* gain_controller2.cc */,
>-				4130921C1EF8D69000757C55 /* gain_controller2.h */,
>+				419241362127372300634FCF /* common.h */,
>+				419241312127372200634FCF /* features_extraction.cc */,
>+				419241322127372200634FCF /* features_extraction.h */,
>+				419241342127372300634FCF /* fft_util.cc */,
>+				419241392127372400634FCF /* fft_util.h */,
>+				419241352127372300634FCF /* lp_residual.cc */,
>+				419241372127372400634FCF /* lp_residual.h */,
>+				41299B9B212736DE00B3414B /* pitch_info.h */,
>+				41299B9A212736DE00B3414B /* pitch_search_internal.cc */,
>+				41299B99212736DE00B3414B /* pitch_search_internal.h */,
>+				419241382127372400634FCF /* pitch_search.cc */,
>+				419241332127372300634FCF /* pitch_search.h */,
>+				419241A7212758D300634FCF /* ring_buffer.h */,
>+				41299B9C212736DE00B3414B /* rnn.cc */,
>+				419241A3212758D200634FCF /* rnn.h */,
>+				419241A5212758D200634FCF /* sequence_buffer.h */,
>+				419241A1212758D200634FCF /* spectral_features_internal.cc */,
>+				41299B9D212736DF00B3414B /* spectral_features_internal.h */,
>+				419241A6212758D200634FCF /* spectral_features.cc */,
>+				419241A4212758D200634FCF /* spectral_features.h */,
>+				419241A2212758D200634FCF /* symmetric_matrix_buffer.h */,
> 			);
>-			name = agc2;
>+			path = rnn_vad;
> 			sourceTree = "<group>";
> 		};
> 		4130922A1EF8D73500757C55 /* aec_dump */ = {
>@@ -6054,11 +6502,11 @@
> 		413A21431FE18D0100373E99 /* rtc_base */ = {
> 			isa = PBXGroup;
> 			children = (
>-				413A21801FE18D7A00373E99 /* DEPS */,
>-				41F9BFC82051DD8E00ABF0B9 /* experiments */,
>-				419C83411FE246380040C30F /* numerics */,
>-				413A21461FE18D4C00373E99 /* numerics */,
>-				413A21FD1FE18DBB00373E99 /* OWNERS */,
>+				4192417A2127390C00634FCF /* time */,
>+				419241652127387900634FCF /* experiments */,
>+				41F2639D2126800000274F59 /* aligned_array.h */,
>+				41F2639B21267FFF00274F59 /* aligned_malloc.cc */,
>+				41F2639C21267FFF00274F59 /* aligned_malloc.h */,
> 				413A22921FE18E0000373E99 /* arraysize.h */,
> 				413A214D1FE18D5000373E99 /* asyncinvoker-inl.h */,
> 				413A22391FE18DD700373E99 /* asyncinvoker.cc */,
>@@ -6069,80 +6517,78 @@
> 				413A21501FE18D5100373E99 /* asyncresolverinterface.h */,
> 				413A22431FE18DDC00373E99 /* asyncsocket.cc */,
> 				413A220B1FE18DC100373E99 /* asyncsocket.h */,
>+				413A22741FE18DF600373E99 /* asynctcpsocket_unittest.cc */,
> 				413A22581FE18DE500373E99 /* asynctcpsocket.cc */,
> 				413A21641FE18D5F00373E99 /* asynctcpsocket.h */,
>-				413A22741FE18DF600373E99 /* asynctcpsocket_unittest.cc */,
>+				413A22911FE18E0000373E99 /* asyncudpsocket_unittest.cc */,
> 				413A21C21FE18D9E00373E99 /* asyncudpsocket.cc */,
> 				413A215C1FE18D5800373E99 /* asyncudpsocket.h */,
>-				413A22911FE18E0000373E99 /* asyncudpsocket_unittest.cc */,
>-				413A21981FE18D8A00373E99 /* atomicops.h */,
> 				413A225D1FE18DE600373E99 /* atomicops_unittest.cc */,
>-				413A22891FE18DFD00373E99 /* base64.cc */,
>-				413A21A11FE18D8F00373E99 /* base64.h */,
>+				413A21981FE18D8A00373E99 /* atomicops.h */,
> 				413A21811FE18D7B00373E99 /* base64_unittest.cc */,
>-				413A219C1FE18D8C00373E99 /* basictypes.h */,
>-				413A21DD1FE18DAD00373E99 /* basictypes_unittest.cc */,
>-				413A22091FE18DC000373E99 /* bind.h */,
> 				413A222F1FE18DD200373E99 /* bind_unittest.cc */,
>+				413A22091FE18DC000373E99 /* bind.h */,
>+				413A22571FE18DE400373E99 /* bitbuffer_unittest.cc */,
> 				413A21E71FE18DB100373E99 /* bitbuffer.cc */,
> 				413A22241FE18DCD00373E99 /* bitbuffer.h */,
>-				413A22571FE18DE400373E99 /* bitbuffer_unittest.cc */,
>+				413A221C1FE18DC800373E99 /* bitrateallocationstrategy_unittest.cc */,
> 				413A214A1FE18D4E00373E99 /* bitrateallocationstrategy.cc */,
> 				413A21D21FE18DA700373E99 /* bitrateallocationstrategy.h */,
>-				413A221C1FE18DC800373E99 /* bitrateallocationstrategy_unittest.cc */,
>-				413A223B1FE18DD900373E99 /* buffer.h */,
> 				413A22131FE18DC400373E99 /* buffer_unittest.cc */,
>+				413A223B1FE18DD900373E99 /* buffer.h */,
>+				413A21711FE18D6D00373E99 /* bufferqueue_unittest.cc */,
> 				413A22481FE18DDE00373E99 /* bufferqueue.cc */,
> 				413A21D71FE18DAA00373E99 /* bufferqueue.h */,
>-				413A21711FE18D6D00373E99 /* bufferqueue_unittest.cc */,
> 				413A228F1FE18DFF00373E99 /* BUILD.gn */,
>+				413A21C01FE18D9D00373E99 /* bytebuffer_unittest.cc */,
> 				413A22A01FE18E0600373E99 /* bytebuffer.cc */,
> 				413A22511FE18DE100373E99 /* bytebuffer.h */,
>-				413A21C01FE18D9D00373E99 /* bytebuffer_unittest.cc */,
>-				413A21EA1FE18DB300373E99 /* byteorder.h */,
> 				413A22361FE18DD500373E99 /* byteorder_unittest.cc */,
>+				413A21EA1FE18DB300373E99 /* byteorder.h */,
>+				413A218A1FE18D8200373E99 /* callback_unittest.cc */,
> 				413A21C31FE18D9E00373E99 /* callback.h */,
> 				413A21B71FE18D9900373E99 /* callback.h.pump */,
>-				413A218A1FE18D8200373E99 /* callback_unittest.cc */,
> 				413A22201FE18DCA00373E99 /* checks.cc */,
> 				413A21FB1FE18DBA00373E99 /* checks.h */,
> 				413A22321FE18DD300373E99 /* compile_assert_c.h */,
> 				413A22191FE18DC600373E99 /* constructormagic.h */,
>+				413A22961FE18E0200373E99 /* copyonwritebuffer_unittest.cc */,
> 				413A21C11FE18D9D00373E99 /* copyonwritebuffer.cc */,
> 				413A22591FE18DE500373E99 /* copyonwritebuffer.h */,
>-				413A22961FE18E0200373E99 /* copyonwritebuffer_unittest.cc */,
>+				413A22611FE18DE800373E99 /* cpu_time_unittest.cc */,
> 				413A21B81FE18D9900373E99 /* cpu_time.cc */,
> 				413A22901FE18E0000373E99 /* cpu_time.h */,
>-				413A22611FE18DE800373E99 /* cpu_time_unittest.cc */,
>+				413A22651FE18DEB00373E99 /* crc32_unittest.cc */,
> 				413A21EF1FE18DB500373E99 /* crc32.cc */,
> 				413A21511FE18D5200373E99 /* crc32.h */,
>-				413A22651FE18DEB00373E99 /* crc32_unittest.cc */,
>+				413A21D01FE18DA700373E99 /* criticalsection_unittest.cc */,
> 				413A22351FE18DD500373E99 /* criticalsection.cc */,
> 				413A22161FE18DC500373E99 /* criticalsection.h */,
>-				413A21D01FE18DA700373E99 /* criticalsection_unittest.cc */,
> 				413A21CE1FE18DA500373E99 /* cryptstring.cc */,
> 				413A21D51FE18DA900373E99 /* cryptstring.h */,
>+				41E02CBB2127356A00C27CD6 /* data_rate_limiter.cc */,
>+				41E02CBC2127356A00C27CD6 /* data_rate_limiter.h */,
> 				413A21FA1FE18DBA00373E99 /* deprecation.h */,
>+				413A21801FE18D7A00373E99 /* DEPS */,
> 				413A21F41FE18DB700373E99 /* dscp.h */,
>-				413A22641FE18DEB00373E99 /* event.cc */,
>-				413A22521FE18DE200373E99 /* event.h */,
> 				413A223C1FE18DD900373E99 /* event_tracer.cc */,
> 				413A222B1FE18DCF00373E99 /* event_tracer.h */,
>-				413A227D1FE18DF900373E99 /* event_tracer_unittest.cc */,
>-				413A21C41FE18D9F00373E99 /* event_unittest.cc */,
>+				413A22641FE18DEB00373E99 /* event.cc */,
>+				413A22521FE18DE200373E99 /* event.h */,
>+				41F9BFC82051DD8E00ABF0B9 /* experiments */,
> 				413A21BC1FE18D9B00373E99 /* fakeclock.cc */,
> 				413A222C1FE18DD000373E99 /* fakeclock.h */,
> 				413A215D1FE18D5900373E99 /* fakenetwork.h */,
> 				413A22791FE18DF800373E99 /* fakesslidentity.cc */,
> 				413A21A31FE18D9000373E99 /* fakesslidentity.h */,
>-				413A21911FE18D8500373E99 /* file.cc */,
>-				413A21821FE18D7C00373E99 /* file.h */,
> 				413A22851FE18DFC00373E99 /* file_posix.cc */,
> 				413A218B1FE18D8200373E99 /* file_unittest.cc */,
> 				413A21521FE18D5200373E99 /* file_win.cc */,
>+				413A21911FE18D8500373E99 /* file.cc */,
>+				413A21821FE18D7C00373E99 /* file.h */,
>+				413A225A1FE18DE500373E99 /* filerotatingstream_unittest.cc */,
> 				413A22871FE18DFD00373E99 /* filerotatingstream.cc */,
> 				413A218C1FE18D8300373E99 /* filerotatingstream.h */,
>-				413A225A1FE18DE500373E99 /* filerotatingstream_unittest.cc */,
> 				413A22621FE18DE900373E99 /* fileutils.cc */,
> 				413A226A1FE18DF000373E99 /* fileutils.h */,
> 				413A22171FE18DC600373E99 /* firewallsocketserver.cc */,
>@@ -6150,63 +6596,58 @@
> 				413A21611FE18D5B00373E99 /* flags.cc */,
> 				413A22691FE18DEF00373E99 /* flags.h */,
> 				413A21471FE18D4D00373E99 /* format_macros.h */,
>-				413A21451FE18D4C00373E99 /* function_view.h */,
> 				413A21891FE18D8100373E99 /* function_view_unittest.cc */,
>+				413A21451FE18D4C00373E99 /* function_view.h */,
> 				413A21AC1FE18D9400373E99 /* gtest_prod_util.h */,
>-				413A21951FE18D8800373E99 /* gunit.h */,
> 				413A220A1FE18DC000373E99 /* gunit_prod.h */,
>+				413A21951FE18D8800373E99 /* gunit.h */,
>+				413A21C51FE18D9F00373E99 /* helpers_unittest.cc */,
> 				413A221B1FE18DC700373E99 /* helpers.cc */,
> 				413A21831FE18D7D00373E99 /* helpers.h */,
>-				413A21C51FE18D9F00373E99 /* helpers_unittest.cc */,
>+				413A21E61FE18DB100373E99 /* httpbase_unittest.cc */,
> 				413A21B21FE18D9700373E99 /* httpbase.cc */,
> 				413A22601FE18DE700373E99 /* httpbase.h */,
>-				413A21E61FE18DB100373E99 /* httpbase_unittest.cc */,
>+				413A21B31FE18D9700373E99 /* httpcommon_unittest.cc */,
> 				413A225F1FE18DE700373E99 /* httpcommon-inl.h */,
> 				413A221A1FE18DC700373E99 /* httpcommon.cc */,
> 				413A22421FE18DDB00373E99 /* httpcommon.h */,
>-				413A21B31FE18D9700373E99 /* httpcommon_unittest.cc */,
>+				413A21541FE18D5300373E99 /* httpserver_unittest.cc */,
> 				413A21FE1FE18DBC00373E99 /* httpserver.cc */,
> 				413A21701FE18D6C00373E99 /* httpserver.h */,
>-				413A21541FE18D5300373E99 /* httpserver_unittest.cc */,
>-				413A22971FE18E0200373E99 /* ifaddrs-android.cc */,
>-				413A216F1FE18D6B00373E99 /* ifaddrs-android.h */,
> 				413A22211FE18DCB00373E99 /* ifaddrs_converter.cc */,
> 				413A227A1FE18DF800373E99 /* ifaddrs_converter.h */,
>+				413A22971FE18E0200373E99 /* ifaddrs-android.cc */,
>+				413A216F1FE18D6B00373E99 /* ifaddrs-android.h */,
> 				413A22931FE18E0100373E99 /* ignore_wundef.h */,
>+				413A216A1FE18D6700373E99 /* ipaddress_unittest.cc */,
> 				413A229B1FE18E0400373E99 /* ipaddress.cc */,
> 				413A214E1FE18D5000373E99 /* ipaddress.h */,
>-				413A216A1FE18D6700373E99 /* ipaddress_unittest.cc */,
>+				413A21F21FE18DB600373E99 /* json_unittest.cc */,
> 				413A21DC1FE18DAD00373E99 /* json.cc */,
> 				413A21EC1FE18DB400373E99 /* json.h */,
>-				413A21F21FE18DB600373E99 /* json_unittest.cc */,
> 				413A22121FE18DC400373E99 /* keep_ref_until_done.h */,
> 				413A22991FE18E0300373E99 /* location.cc */,
> 				413A229A1FE18E0400373E99 /* location.h */,
>-				413A21721FE18D6D00373E99 /* logging.cc */,
>-				413A21DE1FE18DAE00373E99 /* logging.h */,
> 				413A21671FE18D6300373E99 /* logging_mac.mm */,
> 				413A21E01FE18DAE00373E99 /* logging_unittest.cc */,
>+				413A21721FE18D6D00373E99 /* logging.cc */,
>+				413A21DE1FE18DAE00373E99 /* logging.h */,
> 				413A226B1FE18DF000373E99 /* logsinks.cc */,
> 				413A21FF1FE18DBC00373E99 /* logsinks.h */,
> 				413A21D81FE18DAB00373E99 /* macifaddrs_converter.cc */,
> 				413A21E41FE18DB000373E99 /* macutils.cc */,
> 				413A22441FE18DDC00373E99 /* macutils.h */,
>-				413A22451FE18DDC00373E99 /* md5.cc */,
>-				413A223E1FE18DDA00373E99 /* md5.h */,
>-				413A21571FE18D5500373E99 /* md5digest.cc */,
>-				413A224D1FE18DE000373E99 /* md5digest.h */,
>-				413A221D1FE18DC800373E99 /* md5digest_unittest.cc */,
>+				413A21F01FE18DB500373E99 /* memory_usage_unittest.cc */,
> 				413A222D1FE18DD100373E99 /* memory_usage.cc */,
> 				413A21B41FE18D9700373E99 /* memory_usage.h */,
>-				413A21F01FE18DB500373E99 /* memory_usage_unittest.cc */,
>+				413A21E11FE18DAF00373E99 /* messagedigest_unittest.cc */,
> 				413A21A41FE18D9000373E99 /* messagedigest.cc */,
> 				413A21A51FE18D9100373E99 /* messagedigest.h */,
>-				413A21E11FE18DAF00373E99 /* messagedigest_unittest.cc */,
> 				413A22181FE18DC600373E99 /* messagehandler.cc */,
> 				413A215F1FE18D5A00373E99 /* messagehandler.h */,
>+				413A217A1FE18D7500373E99 /* messagequeue_unittest.cc */,
> 				413A21561FE18D5400373E99 /* messagequeue.cc */,
> 				413A22561FE18DE400373E99 /* messagequeue.h */,
>-				413A217A1FE18D7500373E99 /* messagequeue_unittest.cc */,
> 				413A21C81FE18DA100373E99 /* nat_unittest.cc */,
> 				413A21531FE18D5300373E99 /* natserver.cc */,
> 				413A216B1FE18D6800373E99 /* natserver.h */,
>@@ -6218,180 +6659,182 @@
> 				413A214F1FE18D5100373E99 /* nethelper.h */,
> 				413A224E1FE18DE000373E99 /* nethelpers.cc */,
> 				413A22371FE18DD600373E99 /* nethelpers.h */,
>-				413A22941FE18E0100373E99 /* network.cc */,
>-				413A22041FE18DBE00373E99 /* network.h */,
> 				413A22231FE18DCC00373E99 /* network_constants.h */,
> 				413A22681FE18DEE00373E99 /* network_unittest.cc */,
>+				413A22941FE18E0100373E99 /* network.cc */,
>+				413A22041FE18DBE00373E99 /* network.h */,
> 				413A22381FE18DD600373E99 /* networkmonitor.cc */,
> 				413A22291FE18DCF00373E99 /* networkmonitor.h */,
> 				413A21691FE18D6500373E99 /* networkroute.h */,
> 				413A220C1FE18DC100373E99 /* noop.cc */,
> 				413A229C1FE18E0400373E99 /* noop.mm */,
>+				413A227C1FE18DF800373E99 /* nullsocketserver_unittest.cc */,
> 				413A227F1FE18DF900373E99 /* nullsocketserver.cc */,
> 				413A226D1FE18DF100373E99 /* nullsocketserver.h */,
>-				413A227C1FE18DF800373E99 /* nullsocketserver_unittest.cc */,
>-				413A21E21FE18DAF00373E99 /* onetimeevent.h */,
>+				419C83411FE246380040C30F /* numerics */,
> 				413A21621FE18D5D00373E99 /* onetimeevent_unittest.cc */,
>+				413A21E21FE18DAF00373E99 /* onetimeevent.h */,
> 				413A22751FE18DF600373E99 /* openssl.h */,
>+				413A227E1FE18DF900373E99 /* openssladapter_unittest.cc */,
> 				413A219A1FE18D8B00373E99 /* openssladapter.cc */,
> 				413A21DB1FE18DAC00373E99 /* openssladapter.h */,
>-				413A227E1FE18DF900373E99 /* openssladapter_unittest.cc */,
>+				41DDB25E212679A100296D47 /* opensslcertificate.cc */,
>+				41DDB260212679A200296D47 /* opensslcertificate.h */,
> 				413A22401FE18DDB00373E99 /* openssldigest.cc */,
> 				413A218E1FE18D8400373E99 /* openssldigest.h */,
> 				413A21681FE18D6400373E99 /* opensslidentity.cc */,
> 				413A22491FE18DDF00373E99 /* opensslidentity.h */,
>+				41DDB25C212679A000296D47 /* opensslsessioncache.cc */,
>+				41DDB25D212679A000296D47 /* opensslsessioncache.h */,
> 				413A218D1FE18D8300373E99 /* opensslstreamadapter.cc */,
> 				413A22631FE18DEA00373E99 /* opensslstreamadapter.h */,
>+				41DDB25F212679A200296D47 /* opensslutility.cc */,
>+				41DDB25B2126799F00296D47 /* opensslutility.h */,
>+				413A21C71FE18DA100373E99 /* optionsfile_unittest.cc */,
> 				413A21F61FE18DB800373E99 /* optionsfile.cc */,
> 				413A21F11FE18DB600373E99 /* optionsfile.h */,
>-				413A21C71FE18DA100373E99 /* optionsfile_unittest.cc */,
>+				413A21FD1FE18DBB00373E99 /* OWNERS */,
>+				413A219D1FE18D8D00373E99 /* pathutils_unittest.cc */,
> 				413A21E81FE18DB200373E99 /* pathutils.cc */,
> 				413A223D1FE18DD900373E99 /* pathutils.h */,
>-				413A219D1FE18D8D00373E99 /* pathutils_unittest.cc */,
>+				413A227B1FE18DF800373E99 /* physicalsocketserver_unittest.cc */,
> 				413A21E91FE18DB200373E99 /* physicalsocketserver.cc */,
> 				413A22981FE18E0300373E99 /* physicalsocketserver.h */,
>-				413A227B1FE18DF800373E99 /* physicalsocketserver_unittest.cc */,
> 				413A22071FE18DBF00373E99 /* platform_file.cc */,
> 				413A21791FE18D7400373E99 /* platform_file.h */,
>-				413A21CA1FE18DA300373E99 /* platform_thread.cc */,
>-				413A22531FE18DE200373E99 /* platform_thread.h */,
>+				41DDB253212678E300296D47 /* platform_thread_types.cc */,
> 				413A21C91FE18DA200373E99 /* platform_thread_types.h */,
> 				413A21D11FE18DA700373E99 /* platform_thread_unittest.cc */,
>+				413A21CA1FE18DA300373E99 /* platform_thread.cc */,
>+				413A22531FE18DE200373E99 /* platform_thread.h */,
> 				413A22731FE18DF600373E99 /* protobuf_utils.h */,
> 				413A214B1FE18D4E00373E99 /* proxy_unittest.cc */,
> 				413A226F1FE18DF200373E99 /* proxyinfo.cc */,
> 				413A21CB1FE18DA300373E99 /* proxyinfo.h */,
> 				413A214C1FE18D4F00373E99 /* proxyserver.cc */,
> 				413A22721FE18DF500373E99 /* proxyserver.h */,
>-				413A21B51FE18D9800373E99 /* ptr_util.h */,
>-				413A215A1FE18D5600373E99 /* ptr_util_unittest.cc */,
> 				413A21A61FE18D9100373E99 /* race_checker.cc */,
> 				413A229D1FE18E0500373E99 /* race_checker.h */,
>+				413A21661FE18D6200373E99 /* random_unittest.cc */,
> 				413A22781FE18DF700373E99 /* random.cc */,
> 				413A21F81FE18DB900373E99 /* random.h */,
>-				413A21661FE18D6200373E99 /* random_unittest.cc */,
>+				413A229E1FE18E0500373E99 /* rate_limiter_unittest.cc */,
> 				413A223A1FE18DD800373E99 /* rate_limiter.cc */,
> 				413A220D1FE18DC200373E99 /* rate_limiter.h */,
>-				413A229E1FE18E0500373E99 /* rate_limiter_unittest.cc */,
>+				413A21731FE18D6E00373E99 /* rate_statistics_unittest.cc */,
> 				413A22051FE18DBE00373E99 /* rate_statistics.cc */,
> 				413A21B91FE18D9A00373E99 /* rate_statistics.h */,
>-				413A21731FE18D6E00373E99 /* rate_statistics_unittest.cc */,
>-				413A21991FE18D8A00373E99 /* ratelimiter.cc */,
>-				413A22411FE18DDB00373E99 /* ratelimiter.h */,
>-				413A21ED1FE18DB400373E99 /* ratelimiter_unittest.cc */,
>+				413A221F1FE18DC900373E99 /* ratetracker_unittest.cc */,
> 				413A220F1FE18DC200373E99 /* ratetracker.cc */,
> 				413A21AF1FE18D9500373E99 /* ratetracker.h */,
>-				413A221F1FE18DC900373E99 /* ratetracker_unittest.cc */,
> 				413A21AD1FE18D9400373E99 /* refcount.h */,
>-				413A22141FE18DC400373E99 /* refcountedobject.h */,
> 				413A21CD1FE18DA500373E99 /* refcountedobject_unittest.cc */,
>+				413A22141FE18DC400373E99 /* refcountedobject.h */,
> 				413A21CC1FE18DA400373E99 /* refcounter.h */,
>-				413A22541FE18DE300373E99 /* rollingaccumulator.h */,
> 				413A217B1FE18D7600373E99 /* rollingaccumulator_unittest.cc */,
>+				413A22541FE18DE300373E99 /* rollingaccumulator.h */,
>+				413A22881FE18DFD00373E99 /* rtccertificate_unittest.cc */,
> 				413A21BB1FE18D9B00373E99 /* rtccertificate.cc */,
> 				413A21CF1FE18DA600373E99 /* rtccertificate.h */,
>-				413A22881FE18DFD00373E99 /* rtccertificate_unittest.cc */,
>+				413A21851FE18D7E00373E99 /* rtccertificategenerator_unittest.cc */,
> 				413A21651FE18D6100373E99 /* rtccertificategenerator.cc */,
> 				413A21861FE18D7F00373E99 /* rtccertificategenerator.h */,
>-				413A21851FE18D7E00373E99 /* rtccertificategenerator_unittest.cc */,
> 				413A21D41FE18DA900373E99 /* sanitizer.h */,
> 				413A21B01FE18D9600373E99 /* scoped_ref_ptr.h */,
>-				413A217E1FE18D7800373E99 /* sequenced_task_checker.h */,
> 				413A224A1FE18DDF00373E99 /* sequenced_task_checker_impl.cc */,
> 				413A225B1FE18DE600373E99 /* sequenced_task_checker_impl.h */,
> 				413A21BD1FE18D9C00373E99 /* sequenced_task_checker_unittest.cc */,
>-				413A22951FE18E0200373E99 /* sha1.cc */,
>-				413A217F1FE18D7900373E99 /* sha1.h */,
>-				413A21A01FE18D8E00373E99 /* sha1digest.cc */,
>-				413A21A71FE18D9100373E99 /* sha1digest.h */,
>-				413A21E31FE18DB000373E99 /* sha1digest_unittest.cc */,
>+				413A217E1FE18D7800373E99 /* sequenced_task_checker.h */,
>+				413A22221FE18DCB00373E99 /* signalthread_unittest.cc */,
> 				413A22081FE18DC000373E99 /* signalthread.cc */,
> 				413A222E1FE18DD100373E99 /* signalthread.h */,
>-				413A22221FE18DCB00373E99 /* signalthread_unittest.cc */,
>-				413A221E1FE18DC800373E99 /* sigslot.cc */,
>-				413A21761FE18D7100373E99 /* sigslot.h */,
> 				413A21A21FE18D8F00373E99 /* sigslot_unittest.cc */,
>+				413A21761FE18D7100373E99 /* sigslot.h */,
> 				413A21DF1FE18DAE00373E99 /* sigslotrepeater.h */,
>-				413A22671FE18DEC00373E99 /* sigslottester.h */,
> 				413A22461FE18DDD00373E99 /* sigslottester_unittest.cc */,
>-				413A21881FE18D8000373E99 /* socket.h */,
>+				413A22671FE18DEC00373E99 /* sigslottester.h */,
> 				413A21BA1FE18D9A00373E99 /* socket_unittest.cc */,
> 				413A21871FE18D8000373E99 /* socket_unittest.h */,
>+				41DDB2552126790A00296D47 /* socket.cc */,
>+				413A21881FE18D8000373E99 /* socket.h */,
> 				413A219B1FE18D8B00373E99 /* socketadapters.cc */,
> 				413A220E1FE18DC200373E99 /* socketadapters.h */,
>+				413A22101FE18DC300373E99 /* socketaddress_unittest.cc */,
> 				413A22151FE18DC500373E99 /* socketaddress.cc */,
> 				413A22761FE18DF700373E99 /* socketaddress.h */,
>-				413A22101FE18DC300373E99 /* socketaddress_unittest.cc */,
> 				413A21F91FE18DB900373E99 /* socketaddresspair.cc */,
> 				413A22771FE18DF700373E99 /* socketaddresspair.h */,
> 				413A216C1FE18D6900373E99 /* socketfactory.h */,
> 				413A22661FE18DEC00373E99 /* socketserver.h */,
> 				413A217C1FE18D7700373E99 /* socketstream.cc */,
> 				413A22001FE18DBC00373E99 /* socketstream.h */,
>+				413A22341FE18DD400373E99 /* ssladapter_unittest.cc */,
> 				413A22251FE18DCD00373E99 /* ssladapter.cc */,
> 				413A224F1FE18DE100373E99 /* ssladapter.h */,
>-				413A22341FE18DD400373E99 /* ssladapter_unittest.cc */,
>+				41E02CB92127356A00C27CD6 /* sslcertificate.cc */,
>+				41E02CB82127356900C27CD6 /* sslcertificate.h */,
> 				413A229F1FE18E0600373E99 /* sslfingerprint.cc */,
> 				413A225C1FE18DE600373E99 /* sslfingerprint.h */,
>+				413A226C1FE18DF000373E99 /* sslidentity_unittest.cc */,
> 				413A21601FE18D5B00373E99 /* sslidentity.cc */,
> 				413A218F1FE18D8400373E99 /* sslidentity.h */,
>-				413A226C1FE18DF000373E99 /* sslidentity_unittest.cc */,
> 				413A223F1FE18DDA00373E99 /* sslroots.h */,
>+				413A21EB1FE18DB300373E99 /* sslstreamadapter_unittest.cc */,
> 				413A21AA1FE18D9300373E99 /* sslstreamadapter.cc */,
> 				413A226E1FE18DF100373E99 /* sslstreamadapter.h */,
>-				413A21EB1FE18DB300373E99 /* sslstreamadapter_unittest.cc */,
>+				413A22501FE18DE100373E99 /* stream_unittest.cc */,
> 				413A22801FE18DFA00373E99 /* stream.cc */,
> 				413A21A81FE18D9200373E99 /* stream.h */,
>-				413A22501FE18DE100373E99 /* stream_unittest.cc */,
>+				413A21781FE18D7300373E99 /* string_to_number_unittest.cc */,
> 				413A22831FE18DFC00373E99 /* string_to_number.cc */,
> 				413A22111FE18DC300373E99 /* string_to_number.h */,
>-				413A21781FE18D7300373E99 /* string_to_number_unittest.cc */,
>+				413A22031FE18DBE00373E99 /* stringencode_unittest.cc */,
> 				413A215B1FE18D5700373E99 /* stringencode.cc */,
> 				413A21961FE18D8900373E99 /* stringencode.h */,
>-				413A22031FE18DBE00373E99 /* stringencode_unittest.cc */,
>-				413A225E1FE18DE700373E99 /* stringize_macros.h */,
> 				413A22471FE18DDD00373E99 /* stringize_macros_unittest.cc */,
>+				413A225E1FE18DE700373E99 /* stringize_macros.h */,
>+				41DDB267212679AE00296D47 /* strings */,
>+				413A22311FE18DD300373E99 /* stringutils_unittest.cc */,
> 				413A21BF1FE18D9C00373E99 /* stringutils.cc */,
> 				413A21771FE18D7200373E99 /* stringutils.h */,
>-				413A22311FE18DD300373E99 /* stringutils_unittest.cc */,
>-				413A21F31FE18DB700373E99 /* swap_queue.h */,
> 				413A21971FE18D8900373E99 /* swap_queue_unittest.cc */,
>-				413A21E51FE18DB100373E99 /* task_queue.h */,
>+				413A21F31FE18DB700373E99 /* swap_queue.h */,
>+				4192415C2127379200634FCF /* synchronization */,
>+				41F263A22126808800274F59 /* system */,
> 				413A228B1FE18DFE00373E99 /* task_queue_gcd.cc */,
> 				413A21741FE18D7000373E99 /* task_queue_libevent.cc */,
> 				413A21D91FE18DAB00373E99 /* task_queue_posix.cc */,
> 				413A222A1FE18DCF00373E99 /* task_queue_posix.h */,
> 				413A21B11FE18D9600373E99 /* task_queue_unittest.cc */,
> 				413A219E1FE18D8D00373E99 /* task_queue_win.cc */,
>+				413A21E51FE18DB100373E99 /* task_queue.h */,
> 				413A21A91FE18D9200373E99 /* template_util.h */,
> 				413A21441FE18D4C00373E99 /* testbase64.h */,
>+				413A22811FE18DFB00373E99 /* testclient_unittest.cc */,
> 				413A21D31FE18DA800373E99 /* testclient.cc */,
> 				413A22821FE18DFB00373E99 /* testclient.h */,
>-				413A22811FE18DFB00373E99 /* testclient_unittest.cc */,
> 				413A21491FE18D4E00373E99 /* testechoserver.cc */,
> 				413A21D61FE18DAA00373E99 /* testechoserver.h */,
> 				413A219F1FE18D8E00373E99 /* testutils.cc */,
> 				413A22861FE18DFC00373E99 /* testutils.h */,
>-				413A21901FE18D8500373E99 /* thread.cc */,
>-				413A21EE1FE18DB400373E99 /* thread.h */,
>-				413A216E1FE18D6A00373E99 /* thread_annotations.h */,
>+				41F263B4212680E800274F59 /* third_party */,
> 				413A22841FE18DFC00373E99 /* thread_annotations_unittest.cc */,
>-				413A21931FE18D8700373E99 /* thread_checker.h */,
>+				413A216E1FE18D6A00373E99 /* thread_annotations.h */,
> 				413A21FC1FE18DBB00373E99 /* thread_checker_impl.cc */,
> 				413A215E1FE18D5900373E99 /* thread_checker_impl.h */,
> 				413A21B61FE18D9800373E99 /* thread_checker_unittest.cc */,
>+				413A21931FE18D8700373E99 /* thread_checker.h */,
> 				413A228A1FE18DFE00373E99 /* thread_darwin.mm */,
> 				413A22261FE18DCD00373E99 /* thread_unittest.cc */,
>-				413A228D1FE18DFF00373E99 /* timedelta.h */,
>+				413A21901FE18D8500373E99 /* thread.cc */,
>+				413A21EE1FE18DB400373E99 /* thread.h */,
>+				413A224B1FE18DDF00373E99 /* timestampaligner_unittest.cc */,
> 				413A22281FE18DCE00373E99 /* timestampaligner.cc */,
> 				413A22011FE18DBD00373E99 /* timestampaligner.h */,
>-				413A224B1FE18DDF00373E99 /* timestampaligner_unittest.cc */,
>+				413A228E1FE18DFF00373E99 /* timeutils_unittest.cc */,
> 				413A22701FE18DF300373E99 /* timeutils.cc */,
> 				413A21F51FE18DB700373E99 /* timeutils.h */,
>-				413A228E1FE18DFF00373E99 /* timeutils_unittest.cc */,
> 				413A21AB1FE18D9300373E99 /* trace_event.h */,
>-				413A22061FE18DBF00373E99 /* transformadapter.cc */,
>-				413A22271FE18DCE00373E99 /* transformadapter.h */,
> 				413A21F71FE18DB800373E99 /* type_traits.h */,
> 				413A224C1FE18DE000373E99 /* unittest_main.cc */,
> 				413A22331FE18DD400373E99 /* unixfilesystem.cc */,
>@@ -6399,8 +6842,11 @@
> 				413A21DA1FE18DAC00373E99 /* virtualsocket_unittest.cc */,
> 				413A21BE1FE18D9C00373E99 /* virtualsocketserver.cc */,
> 				413A21631FE18D5E00373E99 /* virtualsocketserver.h */,
>-				413A21941FE18D8800373E99 /* weak_ptr.cc */,
> 				413A22711FE18DF400373E99 /* weak_ptr_unittest.cc */,
>+				413A21941FE18D8800373E99 /* weak_ptr.cc */,
>+				41E02CBA2127356A00C27CD6 /* weak_ptr.h */,
>+				41DDB2572126792800296D47 /* zero_memory.cc */,
>+				41DDB2582126792800296D47 /* zero_memory.h */,
> 			);
> 			name = rtc_base;
> 			sourceTree = "<group>";
>@@ -6420,7 +6866,6 @@
> 				413A24201FE1990C00373E99 /* RTCAudioSessionConfiguration.h */,
> 				413A24131FE1990700373E99 /* RTCAudioSource.h */,
> 				413A241F1FE1990B00373E99 /* RTCAudioTrack.h */,
>-				413A24351FE1991700373E99 /* RTCAVFoundationVideoSource.h */,
> 				413A24231FE1990E00373E99 /* RTCCameraPreviewView.h */,
> 				413A24151FE1990700373E99 /* RTCCameraVideoCapturer.h */,
> 				413A240C1FE1990500373E99 /* RTCConfiguration.h */,
>@@ -6475,6 +6920,49 @@
> 			name = WebRTC;
> 			sourceTree = "<group>";
> 		};
>+		413AD19E21265ACB003F7263 /* abseil-cpp */ = {
>+			isa = PBXGroup;
>+			children = (
>+				413AD19F21265ADF003F7263 /* absl */,
>+			);
>+			name = "abseil-cpp";
>+			sourceTree = "<group>";
>+		};
>+		413AD19F21265ADF003F7263 /* absl */ = {
>+			isa = PBXGroup;
>+			children = (
>+				413AD1A021265AFA003F7263 /* algorithm */,
>+				413AD1A621265B35003F7263 /* base */,
>+				41DDB24521265B8200296D47 /* container */,
>+				41A08BB621268A6A001D5D7B /* memory */,
>+				41DDB24A21265BA800296D47 /* types */,
>+			);
>+			name = absl;
>+			path = "abseil-cpp/absl";
>+			sourceTree = "<group>";
>+		};
>+		413AD1A021265AFA003F7263 /* algorithm */ = {
>+			isa = PBXGroup;
>+			children = (
>+				413AD1A121265B0F003F7263 /* algorithm.h */,
>+				413AD1A221265B0F003F7263 /* container.h */,
>+			);
>+			name = algorithm;
>+			sourceTree = "<group>";
>+		};
>+		413AD1A621265B35003F7263 /* base */ = {
>+			isa = PBXGroup;
>+			children = (
>+				41A08BCA2126961F001D5D7B /* raw_logging.cc */,
>+				41A08BC92126961F001D5D7B /* raw_logging.h */,
>+				413AD1A921265B59003F7263 /* casts.h */,
>+				413AD1A821265B59003F7263 /* dynamic_annotations.cc */,
>+				413AD1AA21265B59003F7263 /* macros.h */,
>+				413AD1A721265B59003F7263 /* optimization.h */,
>+			);
>+			name = base;
>+			sourceTree = "<group>";
>+		};
> 		4140B8161E4E335E007409E6 /* g711 */ = {
> 			isa = PBXGroup;
> 			children = (
>@@ -6482,8 +6970,6 @@
> 				4140B81B1E4E3383007409E6 /* audio_decoder_pcm.h */,
> 				4140B8181E4E3383007409E6 /* audio_encoder_pcm.cc */,
> 				4140B8191E4E3383007409E6 /* audio_encoder_pcm.h */,
>-				4140B81E1E4E3383007409E6 /* g711.c */,
>-				4140B81F1E4E3383007409E6 /* g711.h */,
> 				4140B81C1E4E3383007409E6 /* g711_interface.c */,
> 				4140B81D1E4E3383007409E6 /* g711_interface.h */,
> 			);
>@@ -6497,9 +6983,6 @@
> 				4140B8291E4E3396007409E6 /* audio_decoder_g722.h */,
> 				4140B82A1E4E3396007409E6 /* audio_encoder_g722.cc */,
> 				4140B82B1E4E3396007409E6 /* audio_encoder_g722.h */,
>-				4140B82C1E4E3396007409E6 /* g722_decode.c */,
>-				4140B82D1E4E3396007409E6 /* g722_enc_dec.h */,
>-				4140B82E1E4E3396007409E6 /* g722_encode.c */,
> 				4140B82F1E4E3396007409E6 /* g722_interface.c */,
> 				4140B8301E4E3396007409E6 /* g722_interface.h */,
> 			);
>@@ -6597,10 +7080,6 @@
> 		4145E4931EF890E500FCF6E6 /* Video */ = {
> 			isa = PBXGroup;
> 			children = (
>-				4145E4A81EF8933C00FCF6E6 /* corevideo_frame_buffer.cc */,
>-				4145E4A91EF8933C00FCF6E6 /* corevideo_frame_buffer.h */,
>-				41ECEAD220641458009D5141 /* objc_frame_buffer.h */,
>-				41ECEAD120641457009D5141 /* objc_frame_buffer.mm */,
> 				41ECEAC520641302009D5141 /* RTCCVPixelBuffer.mm */,
> 				41ECEACE2064136F009D5141 /* RTCI420Buffer+Private.h */,
> 				41ECEACD2064136F009D5141 /* RTCI420Buffer.mm */,
>@@ -6615,10 +7094,6 @@
> 				41ECEAD52064147C009D5141 /* helpers.h */,
> 				4145E4B21EF8943D00FCF6E6 /* nalu_rewriter.cc */,
> 				4145E4AF1EF8943D00FCF6E6 /* nalu_rewriter.h */,
>-				413A23F61FE1987B00373E99 /* objc_video_decoder_factory.h */,
>-				413A23F91FE1987C00373E99 /* objc_video_decoder_factory.mm */,
>-				413A23F71FE1987C00373E99 /* objc_video_encoder_factory.h */,
>-				413A23F51FE1987B00373E99 /* objc_video_encoder_factory.mm */,
> 				413A23F81FE1987C00373E99 /* RTCVideoDecoderH264.mm */,
> 				413A23F41FE1987B00373E99 /* RTCVideoEncoderH264.mm */,
> 			);
>@@ -6666,6 +7141,158 @@
> 			name = Common;
> 			sourceTree = "<group>";
> 		};
>+		4192415C2127379200634FCF /* synchronization */ = {
>+			isa = PBXGroup;
>+			children = (
>+				419241602127385B00634FCF /* rw_lock_posix.cc */,
>+				4192415E2127385B00634FCF /* rw_lock_posix.h */,
>+				4192415D2127385A00634FCF /* rw_lock_wrapper.cc */,
>+				4192415F2127385B00634FCF /* rw_lock_wrapper.h */,
>+			);
>+			name = synchronization;
>+			path = rtc_base/synchronization;
>+			sourceTree = "<group>";
>+		};
>+		419241652127387900634FCF /* experiments */ = {
>+			isa = PBXGroup;
>+			children = (
>+				4192416B2127389B00634FCF /* congestion_controller_experiment.cc */,
>+				419241672127389A00634FCF /* congestion_controller_experiment.h */,
>+				419241682127389A00634FCF /* field_trial_parser.cc */,
>+				419241692127389B00634FCF /* field_trial_parser.h */,
>+				419241E621275A7600634FCF /* field_trial_units.cc */,
>+				419241E521275A7600634FCF /* field_trial_units.h */,
>+				4192416A2127389B00634FCF /* quality_scaling_experiment.cc */,
>+				419241662127389A00634FCF /* quality_scaling_experiment.h */,
>+			);
>+			name = experiments;
>+			path = rtc_base/experiments;
>+			sourceTree = "<group>";
>+		};
>+		4192417A2127390C00634FCF /* time */ = {
>+			isa = PBXGroup;
>+			children = (
>+				4192417B2127392300634FCF /* timestamp_extrapolator.cc */,
>+				4192417C2127392300634FCF /* timestamp_extrapolator.h */,
>+			);
>+			name = time;
>+			path = rtc_base/time;
>+			sourceTree = "<group>";
>+		};
>+		419241912127583900634FCF /* rnnnoise */ = {
>+			isa = PBXGroup;
>+			children = (
>+				419241922127585000634FCF /* src */,
>+			);
>+			name = rnnnoise;
>+			path = rnnoise;
>+			sourceTree = "<group>";
>+		};
>+		419241922127585000634FCF /* src */ = {
>+			isa = PBXGroup;
>+			children = (
>+				419241952127586500634FCF /* kiss_fft.cc */,
>+				419241962127586500634FCF /* kiss_fft.h */,
>+				419241972127586500634FCF /* rnn_activations.h */,
>+				419241942127586400634FCF /* rnn_vad_weights.cc */,
>+				419241932127586400634FCF /* rnn_vad_weights.h */,
>+			);
>+			name = src;
>+			sourceTree = "<group>";
>+		};
>+		419241B12127591C00634FCF /* bbr */ = {
>+			isa = PBXGroup;
>+			children = (
>+				419241B62127593D00634FCF /* bandwidth_sampler.cc */,
>+				419241BE2127593F00634FCF /* bandwidth_sampler.h */,
>+				419241BB2127593E00634FCF /* bbr_factory.cc */,
>+				419241B82127593E00634FCF /* bbr_factory.h */,
>+				419241B92127593E00634FCF /* bbr_network_controller.cc */,
>+				419241B52127593D00634FCF /* bbr_network_controller.h */,
>+				419241B32127593D00634FCF /* data_transfer_tracker.cc */,
>+				419241BF2127593F00634FCF /* data_transfer_tracker.h */,
>+				419241B72127593D00634FCF /* loss_rate_filter.cc */,
>+				419241B22127593C00634FCF /* loss_rate_filter.h */,
>+				419241BA2127593E00634FCF /* packet_number_indexed_queue.h */,
>+				419241BC2127593E00634FCF /* rtt_stats.cc */,
>+				419241BD2127593E00634FCF /* rtt_stats.h */,
>+				419241B42127593D00634FCF /* windowed_filter.h */,
>+			);
>+			path = bbr;
>+			sourceTree = "<group>";
>+		};
>+		419241D421275A1800634FCF /* units */ = {
>+			isa = PBXGroup;
>+			children = (
>+				419241D921275A2F00634FCF /* data_rate.cc */,
>+				419241D621275A2F00634FCF /* data_rate.h */,
>+				419241D721275A2F00634FCF /* data_size.cc */,
>+				419241DC21275A3000634FCF /* data_size.h */,
>+				419241DA21275A3000634FCF /* time_delta.cc */,
>+				419241D521275A2F00634FCF /* time_delta.h */,
>+				419241DB21275A3000634FCF /* timestamp.cc */,
>+				419241D821275A2F00634FCF /* timestamp.h */,
>+			);
>+			path = units;
>+			sourceTree = "<group>";
>+		};
>+		419241F7212765EE00634FCF /* Native */ = {
>+			isa = PBXGroup;
>+			children = (
>+				419241F82127660E00634FCF /* api */,
>+				419241FA2127661700634FCF /* src */,
>+			);
>+			name = Native;
>+			path = sdk/objc/Framework/Native;
>+			sourceTree = "<group>";
>+		};
>+		419241F82127660E00634FCF /* api */ = {
>+			isa = PBXGroup;
>+			children = (
>+				419241FB2127664100634FCF /* audio_device_module.h */,
>+				419241FC2127664100634FCF /* audio_device_module.mm */,
>+				419242062127664600634FCF /* video_capturer.h */,
>+				419241FD2127664100634FCF /* video_capturer.mm */,
>+				419242012127664200634FCF /* video_decoder_factory.h */,
>+				419242032127664200634FCF /* video_decoder_factory.mm */,
>+				419242042127664500634FCF /* video_encoder_factory.h */,
>+				419242052127664500634FCF /* video_encoder_factory.mm */,
>+				419242082127664700634FCF /* video_frame_buffer.h */,
>+				419242072127664600634FCF /* video_frame_buffer.mm */,
>+				419242002127664200634FCF /* video_frame.h */,
>+				419241FF2127664100634FCF /* video_frame.mm */,
>+				419242022127664200634FCF /* video_renderer.h */,
>+				419241FE2127664100634FCF /* video_renderer.mm */,
>+			);
>+			path = api;
>+			sourceTree = "<group>";
>+		};
>+		419241FA2127661700634FCF /* src */ = {
>+			isa = PBXGroup;
>+			children = (
>+				419242322127665A00634FCF /* audio_device_ios.h */,
>+				4192422F2127665900634FCF /* audio_device_ios.mm */,
>+				419242312127665900634FCF /* audio_device_module_ios.h */,
>+				419242332127665A00634FCF /* audio_device_module_ios.mm */,
>+				419242302127665900634FCF /* audio_session_observer.h */,
>+				419242352127665A00634FCF /* voice_processing_audio_unit.h */,
>+				419242342127665A00634FCF /* voice_processing_audio_unit.mm */,
>+				4192421B2127664C00634FCF /* objc_frame_buffer.h */,
>+				4192421D2127664D00634FCF /* objc_frame_buffer.mm */,
>+				419242182127664B00634FCF /* objc_video_decoder_factory.h */,
>+				4192421A2127664C00634FCF /* objc_video_decoder_factory.mm */,
>+				419242222127664E00634FCF /* objc_video_encoder_factory.h */,
>+				419242172127664B00634FCF /* objc_video_encoder_factory.mm */,
>+				419242192127664B00634FCF /* objc_video_frame.h */,
>+				4192421E2127664D00634FCF /* objc_video_frame.mm */,
>+				419242212127664E00634FCF /* objc_video_renderer.h */,
>+				4192421F2127664D00634FCF /* objc_video_renderer.mm */,
>+				4192421C2127664C00634FCF /* objc_video_track_source.h */,
>+				419242202127664D00634FCF /* objc_video_track_source.mm */,
>+			);
>+			path = src;
>+			sourceTree = "<group>";
>+		};
> 		419C82E71FE20E7D0040C30F /* opus */ = {
> 			isa = PBXGroup;
> 			children = (
>@@ -6746,6 +7373,8 @@
> 				419C834B1FE246640040C30F /* safe_compare.h */,
> 				419C83471FE246630040C30F /* safe_conversions.h */,
> 				419C83481FE246630040C30F /* safe_minmax.h */,
>+				41E02CD22127360700C27CD6 /* sample_counter.cc */,
>+				41E02CD12127360600C27CD6 /* sample_counter.h */,
> 			);
> 			name = numerics;
> 			sourceTree = "<group>";
>@@ -6879,6 +7508,59 @@
> 			path = audio_codecs/ilbc;
> 			sourceTree = "<group>";
> 		};
>+		41A08BB621268A6A001D5D7B /* memory */ = {
>+			isa = PBXGroup;
>+			children = (
>+				41A08BB721268A7D001D5D7B /* memory.h */,
>+			);
>+			name = memory;
>+			sourceTree = "<group>";
>+		};
>+		41A08BD921272F3E001D5D7B /* agc2 */ = {
>+			isa = PBXGroup;
>+			children = (
>+				41299B98212736B200B3414B /* rnn_vad */,
>+				41D6B46E212731A4008F9353 /* adaptive_agc.cc */,
>+				41D6B464212731A2008F9353 /* adaptive_agc.h */,
>+				41A08BE421272F85001D5D7B /* adaptive_digital_gain_applier.cc */,
>+				41A08BDB21272F84001D5D7B /* adaptive_digital_gain_applier.h */,
>+				41A08BE921272F86001D5D7B /* adaptive_mode_level_estimator_agc.cc */,
>+				41D6B462212731A1008F9353 /* adaptive_mode_level_estimator_agc.h */,
>+				41D6B45F212731A1008F9353 /* adaptive_mode_level_estimator.cc */,
>+				41A08BDC21272F84001D5D7B /* adaptive_mode_level_estimator.h */,
>+				41D6B46D212731A4008F9353 /* agc2_common.h */,
>+				41D6B45E212731A0008F9353 /* biquad_filter.cc */,
>+				41D6B463212731A2008F9353 /* biquad_filter.h */,
>+				41A08BE621272F86001D5D7B /* compute_interpolated_gain_curve.cc */,
>+				41A08BDA21272F84001D5D7B /* compute_interpolated_gain_curve.h */,
>+				41A08BE821272F86001D5D7B /* down_sampler.cc */,
>+				41D6B460212731A1008F9353 /* down_sampler.h */,
>+				41A08BE521272F86001D5D7B /* fixed_digital_level_estimator.cc */,
>+				41A08BDF21272F84001D5D7B /* fixed_digital_level_estimator.h */,
>+				41D6B465212731A2008F9353 /* fixed_gain_controller.cc */,
>+				41D6B467212731A3008F9353 /* fixed_gain_controller.h */,
>+				41A08BDE21272F84001D5D7B /* gain_applier.cc */,
>+				41A08BE721272F86001D5D7B /* gain_applier.h */,
>+				41A08BE021272F85001D5D7B /* gain_curve_applier.cc */,
>+				41D6B469212731A3008F9353 /* gain_curve_applier.h */,
>+				41A08BE121272F85001D5D7B /* interpolated_gain_curve.cc */,
>+				41A08BEA21272F86001D5D7B /* limiter.h */,
>+				41D6B470212731A6008F9353 /* limiter.cc */,
>+				41D6B461212731A1008F9353 /* noise_level_estimator.cc */,
>+				41D6B46A212731A3008F9353 /* noise_spectrum_estimator.cc */,
>+				41A08BE221272F85001D5D7B /* noise_spectrum_estimator.h */,
>+				41D6B468212731A3008F9353 /* saturation_protector.cc */,
>+				41D6B46B212731A3008F9353 /* saturation_protector.h */,
>+				41D6B46F212731A4008F9353 /* signal_classifier.cc */,
>+				41A08BEB21272F87001D5D7B /* signal_classifier.h */,
>+				41D6B46C212731A4008F9353 /* vad_with_level.cc */,
>+				41A08BE321272F85001D5D7B /* vad_with_level.h */,
>+				41D6B466212731A2008F9353 /* vector_float_frame.cc */,
>+				41A08BDD21272F84001D5D7B /* vector_float_frame.h */,
>+			);
>+			path = agc2;
>+			sourceTree = "<group>";
>+		};
> 		41A391431EFC446E00C4516A /* sha */ = {
> 			isa = PBXGroup;
> 			children = (
>@@ -7129,6 +7811,93 @@
> 			path = curve25519/asm;
> 			sourceTree = "<group>";
> 		};
>+		41DDB24521265B8200296D47 /* container */ = {
>+			isa = PBXGroup;
>+			children = (
>+				41DDB24B21265BD700296D47 /* fixed_array.h */,
>+				41DDB24C21265BD700296D47 /* inlined_vector.h */,
>+			);
>+			name = container;
>+			sourceTree = "<group>";
>+		};
>+		41DDB24A21265BA800296D47 /* types */ = {
>+			isa = PBXGroup;
>+			children = (
>+				4192417F2127497100634FCF /* bad_variant_access.cc */,
>+				419241802127497100634FCF /* bad_variant_access.h */,
>+				41A08BC6212695DE001D5D7B /* bad_optional_access.cc */,
>+				41A08BC5212695DE001D5D7B /* bad_optional_access.h */,
>+				41DDB25021265BE900296D47 /* optional.cc */,
>+				41DDB24F21265BE900296D47 /* optional.h */,
>+			);
>+			name = types;
>+			sourceTree = "<group>";
>+		};
>+		41DDB267212679AE00296D47 /* strings */ = {
>+			isa = PBXGroup;
>+			children = (
>+				41DDB26B212679D200296D47 /* audio_format_to_string.cc */,
>+				41DDB269212679D100296D47 /* audio_format_to_string.h */,
>+				41DDB26A212679D200296D47 /* string_builder.cc */,
>+				41DDB268212679D100296D47 /* string_builder.h */,
>+			);
>+			name = strings;
>+			path = rtc_base/strings;
>+			sourceTree = "<group>";
>+		};
>+		41E02C8F212734E800C27CD6 /* goog_cc */ = {
>+			isa = PBXGroup;
>+			children = (
>+				41E02C942127352A00C27CD6 /* acknowledged_bitrate_estimator.cc */,
>+				41E02C922127352A00C27CD6 /* acknowledged_bitrate_estimator.h */,
>+				41E02CA12127352C00C27CD6 /* alr_detector.cc */,
>+				41E02C992127352B00C27CD6 /* alr_detector.h */,
>+				41E02C9A2127352B00C27CD6 /* bitrate_estimator.cc */,
>+				41E02C982127352B00C27CD6 /* bitrate_estimator.h */,
>+				41E02C912127352900C27CD6 /* delay_based_bwe.cc */,
>+				41E02C972127352A00C27CD6 /* delay_based_bwe.h */,
>+				41E02CA32127352D00C27CD6 /* delay_increase_detector_interface.h */,
>+				41E02C952127352A00C27CD6 /* goog_cc_factory.cc */,
>+				41E02CA22127352C00C27CD6 /* goog_cc_network_control.cc */,
>+				41E02C962127352A00C27CD6 /* goog_cc_network_control.h */,
>+				41E02C9D2127352B00C27CD6 /* median_slope_estimator.cc */,
>+				41E02C9B2127352B00C27CD6 /* median_slope_estimator.h */,
>+				41E02C9E2127352C00C27CD6 /* probe_bitrate_estimator.cc */,
>+				41E02C932127352A00C27CD6 /* probe_bitrate_estimator.h */,
>+				41E02CA02127352C00C27CD6 /* probe_controller.cc */,
>+				41E02C902127352900C27CD6 /* probe_controller.h */,
>+				41E02C9C2127352B00C27CD6 /* trendline_estimator.cc */,
>+				41E02C9F2127352C00C27CD6 /* trendline_estimator.h */,
>+			);
>+			path = goog_cc;
>+			sourceTree = "<group>";
>+		};
>+		41E02CC62127359F00C27CD6 /* transport */ = {
>+			isa = PBXGroup;
>+			children = (
>+				41E02CC7212735B600C27CD6 /* bitrate_settings.cc */,
>+				41E02CCB212735B700C27CD6 /* bitrate_settings.h */,
>+				41E02CCA212735B700C27CD6 /* network_control.h */,
>+				41E02CC8212735B700C27CD6 /* network_types.cc */,
>+				41E02CC9212735B700C27CD6 /* network_types.h */,
>+			);
>+			path = transport;
>+			sourceTree = "<group>";
>+		};
>+		41E02CD52127362300C27CD6 /* rtp */ = {
>+			isa = PBXGroup;
>+			children = (
>+				41924189212749C800634FCF /* pacer_controller.h */,
>+				41924188212749C700634FCF /* send_side_congestion_controller.cc */,
>+				41924187212749C700634FCF /* send_time_history.h */,
>+				4192418A212749C800634FCF /* transport_feedback_adapter.cc */,
>+				41E02CD62127363C00C27CD6 /* pacer_controller.cc */,
>+				41E02CD82127363D00C27CD6 /* send_time_history.cc */,
>+				41E02CD72127363C00C27CD6 /* transport_feedback_adapter.h */,
>+			);
>+			path = rtp;
>+			sourceTree = "<group>";
>+		};
> 		41EA53A21EFC2BAF002FF04C /* fipsmodule */ = {
> 			isa = PBXGroup;
> 			children = (
>@@ -7284,13 +8053,13 @@
> 		41ECEAB2206300E3009D5141 /* PeerConnection */ = {
> 			isa = PBXGroup;
> 			children = (
>+				4192423F21276D4700634FCF /* RTCPeerConnectionFactory.mm */,
> 				41ECEAC720641328009D5141 /* RTCEncodedImage.mm */,
> 				41ECEAD9206414A6009D5141 /* RTCRtpFragmentationHeader.mm */,
> 				41ECEAB320630107009D5141 /* RTCVideoCodec+Private.h */,
> 				41ECEAB420630107009D5141 /* RTCVideoCodec.mm */,
> 				41ECEAB520630108009D5141 /* RTCVideoCodecH264.mm */,
> 				41ECEAE920641812009D5141 /* RTCVideoEncoderSettings.mm */,
>-				41ECEACA2064134E009D5141 /* RTCVideoFrame+Private.h */,
> 				41ECEAC92064134E009D5141 /* RTCVideoFrame.mm */,
> 				41ECEADF206416FE009D5141 /* RTCWrappedNativeVideoDecoder.h */,
> 				41ECEAE0206416FE009D5141 /* RTCWrappedNativeVideoDecoder.mm */,
>@@ -7300,6 +8069,63 @@
> 			name = PeerConnection;
> 			sourceTree = "<group>";
> 		};
>+		41F2636B21267B2300274F59 /* third_party */ = {
>+			isa = PBXGroup;
>+			children = (
>+				41F2637121267B4A00274F59 /* spl_sqrt_floor.c */,
>+				41F2637021267B4A00274F59 /* spl_sqrt_floor.h */,
>+				41F2636C21267B3D00274F59 /* fft4g.c */,
>+				41F2636D21267B3E00274F59 /* fft4g.h */,
>+			);
>+			path = third_party;
>+			sourceTree = "<group>";
>+		};
>+		41F2638621267F2500274F59 /* third_party */ = {
>+			isa = PBXGroup;
>+			children = (
>+				41F2639521267F5E00274F59 /* pa_memorybarrier.h */,
>+				41F2639721267F5E00274F59 /* pa_ringbuffer.c */,
>+				41F2639621267F5E00274F59 /* pa_ringbuffer.h */,
>+				41F2639121267F5300274F59 /* g722_decode.c */,
>+				41F2639021267F5300274F59 /* g722_enc_dec.h */,
>+				41F2638F21267F5200274F59 /* g722_encode.c */,
>+				41F2638B21267F4900274F59 /* g711.c */,
>+				41F2638C21267F4900274F59 /* g711.h */,
>+				41F2638821267F4000274F59 /* fft.c */,
>+				41F2638721267F4000274F59 /* fft.h */,
>+			);
>+			path = third_party;
>+			sourceTree = "<group>";
>+		};
>+		41F263A22126808800274F59 /* system */ = {
>+			isa = PBXGroup;
>+			children = (
>+				41F263AB212680A800274F59 /* arch.h */,
>+				41F263A9212680A700274F59 /* asm_defines.h */,
>+				41F263A8212680A700274F59 /* BUILD.gn */,
>+				41F263A5212680A700274F59 /* fallthrough.h */,
>+				41F263A3212680A600274F59 /* file_wrapper.cc */,
>+				41F263AA212680A800274F59 /* file_wrapper.h */,
>+				41F263A4212680A600274F59 /* ignore_warnings.h */,
>+				41F263A6212680A700274F59 /* inline.h */,
>+				41F263A7212680A700274F59 /* unused.h */,
>+			);
>+			name = system;
>+			path = rtc_base/system;
>+			sourceTree = "<group>";
>+		};
>+		41F263B4212680E800274F59 /* third_party */ = {
>+			isa = PBXGroup;
>+			children = (
>+				41F263B92126811400274F59 /* sigslot.cc */,
>+				41F263BA2126811400274F59 /* sigslot.h */,
>+				41F263B52126810B00274F59 /* base64.cc */,
>+				41F263B62126810C00274F59 /* base64.h */,
>+			);
>+			name = third_party;
>+			path = rtc_base/third_party;
>+			sourceTree = "<group>";
>+		};
> 		41F411BD1EF8DB8200343C26 /* vp8 */ = {
> 			isa = PBXGroup;
> 			children = (
>@@ -7308,10 +8134,6 @@
> 				419C84311FE24E800040C30F /* default_temporal_layers.h */,
> 				419C84301FE24E7F0040C30F /* screenshare_layers.cc */,
> 				419C842C1FE24E7F0040C30F /* screenshare_layers.h */,
>-				419C84321FE24E800040C30F /* simulcast_encoder_adapter.cc */,
>-				419C842D1FE24E7F0040C30F /* simulcast_encoder_adapter.h */,
>-				419C842E1FE24E7F0040C30F /* simulcast_rate_allocator.cc */,
>-				419C842F1FE24E7F0040C30F /* simulcast_rate_allocator.h */,
> 				419C842A1FE24E7E0040C30F /* temporal_layers.cc */,
> 				419C84331FE24E800040C30F /* temporal_layers.h */,
> 				41F411BE1EF8DBA500343C26 /* vp8_noop.cc */,
>@@ -7385,7 +8207,6 @@
> 		5C11A0061E457431004F0987 /* mac */ = {
> 			isa = PBXGroup;
> 			children = (
>-				5C11A00F1E4574AF004F0987 /* portaudio */,
> 				5C11A0071E457448004F0987 /* audio_device_mac.cc */,
> 				5C11A0081E457448004F0987 /* audio_device_mac.h */,
> 				5C11A0091E457448004F0987 /* audio_mixer_manager_mac.cc */,
>@@ -7394,16 +8215,6 @@
> 			name = mac;
> 			sourceTree = "<group>";
> 		};
>-		5C11A00F1E4574AF004F0987 /* portaudio */ = {
>-			isa = PBXGroup;
>-			children = (
>-				5C11A0101E4574C5004F0987 /* pa_memorybarrier.h */,
>-				5C11A0111E4574C5004F0987 /* pa_ringbuffer.c */,
>-				5C11A0121E4574C5004F0987 /* pa_ringbuffer.h */,
>-			);
>-			name = portaudio;
>-			sourceTree = "<group>";
>-		};
> 		5C11A0161E45756C004F0987 /* dummy */ = {
> 			isa = PBXGroup;
> 			children = (
>@@ -7435,10 +8246,7 @@
> 				5C63FA731E417AD3002CA531 /* stats */,
> 				5C4B4C8B1E43207F002651C8 /* system_wrappers */,
> 				5CDD85591E43B5AE00621E92 /* video */,
>-				5CDD84291E43ADEE00621E92 /* voice_engine */,
>-				5C6CDD3E1E413598009754E3 /* common_types.cc */,
> 				5C6CDD3F1E413598009754E3 /* common_types.h */,
>-				5C6CDD431E413598009754E3 /* typedefs.h */,
> 			);
> 			path = webrtc;
> 			sourceTree = "<group>";
>@@ -7870,7 +8678,9 @@
> 				5C4B48AB1E42C1E3002651C8 /* fakevideorenderer.h */,
> 				4145E4BD1EF894F500FCF6E6 /* h264_profile_level_id.cc */,
> 				4145E4BC1EF894F100FCF6E6 /* h264_profile_level_id.h */,
>+				4102F6B121273330006AE8D7 /* mediachannel.cc */,
> 				5C4B48AD1E42C1E3002651C8 /* mediachannel.h */,
>+				4102F6B021273330006AE8D7 /* mediaconfig.h */,
> 				5C4B48AE1E42C1E3002651C8 /* mediaconstants.cc */,
> 				5C4B48AF1E42C1E3002651C8 /* mediaconstants.h */,
> 				5C4B48B01E42C1E3002651C8 /* mediaengine.cc */,
>@@ -7901,6 +8711,8 @@
> 		5C4B484B1E42C199002651C8 /* engine */ = {
> 			isa = PBXGroup;
> 			children = (
>+				419241F321275C3200634FCF /* vp8_encoder_simulcast_proxy.cc */,
>+				419241F221275C3100634FCF /* vp8_encoder_simulcast_proxy.h */,
> 				41F411C61EF972F100343C26 /* adm_helpers.cc */,
> 				41F411C51EF972EC00343C26 /* adm_helpers.h */,
> 				5CD284C01E6A5FFC0094FDC8 /* apm_helpers.cc */,
>@@ -7924,10 +8736,6 @@
> 				5C4B48601E42C1BA002651C8 /* simulcast.h */,
> 				41F9BFB12051C93500ABF0B9 /* simulcast_encoder_adapter.cc */,
> 				41F9BFB32051C93600ABF0B9 /* simulcast_encoder_adapter.h */,
>-				5CD286221E6A666D0094FDC8 /* videodecodersoftwarefallbackwrapper.cc */,
>-				5CD286231E6A666D0094FDC8 /* videodecodersoftwarefallbackwrapper.h */,
>-				5C4B48621E42C1BA002651C8 /* videoencodersoftwarefallbackwrapper.cc */,
>-				5C4B48631E42C1BA002651C8 /* videoencodersoftwarefallbackwrapper.h */,
> 				5C4B48661E42C1BA002651C8 /* webrtcmediaengine.cc */,
> 				5C4B48671E42C1BA002651C8 /* webrtcmediaengine.h */,
> 				5C4B48691E42C1BA002651C8 /* webrtcvideocapturer.cc */,
>@@ -8103,6 +8911,7 @@
> 			children = (
> 				5C4B4B5A1E431C63002651C8 /* Classes */,
> 				413A24001FE198BE00373E99 /* Headers */,
>+				419241F7212765EE00634FCF /* Native */,
> 			);
> 			name = Framework;
> 			sourceTree = "<group>";
>@@ -8139,6 +8948,7 @@
> 		5C4B4C221E431F81002651C8 /* common_audio */ = {
> 			isa = PBXGroup;
> 			children = (
>+				41F2636B21267B2300274F59 /* third_party */,
> 				5CDD87431E43BAE900621E92 /* resampler */,
> 				5CDD86091E43B8A300621E92 /* signal_processing */,
> 				5CDD86C41E43BA1800621E92 /* vad */,
>@@ -8152,8 +8962,6 @@
> 				5C4B4C2D1E431F9C002651C8 /* blocker.h */,
> 				5C4B4C301E431F9C002651C8 /* channel_buffer.cc */,
> 				5C4B4C311E431F9C002651C8 /* channel_buffer.h */,
>-				5C4B4C341E431F9C002651C8 /* fft4g.c */,
>-				5C4B4C351E431F9C002651C8 /* fft4g.h */,
> 				5C4B4C3C1E431F9C002651C8 /* fir_filter.h */,
> 				419C83171FE242B20040C30F /* fir_filter_c.cc */,
> 				419C83161FE242B10040C30F /* fir_filter_c.h */,
>@@ -8194,7 +9002,6 @@
> 		5C4B4C8C1E43209B002651C8 /* source */ = {
> 			isa = PBXGroup;
> 			children = (
>-				5C4B4C8F1E4320A9002651C8 /* aligned_malloc.cc */,
> 				5C4B4C941E4320A9002651C8 /* clock.cc */,
> 				5C4B4C981E4320A9002651C8 /* cpu_features.cc */,
> 				5C4B4C991E4320A9002651C8 /* cpu_info.cc */,
>@@ -8202,15 +9009,10 @@
> 				5C4B4C9C1E4320A9002651C8 /* event_timer_posix.cc */,
> 				5C4B4C9D1E4320A9002651C8 /* event_timer_posix.h */,
> 				5C4B4CA11E4320A9002651C8 /* field_trial_default.cc */,
>-				5C4B4CA21E4320A9002651C8 /* file_impl.cc */,
> 				5C4B4CA71E4320A9002651C8 /* metrics_default.cc */,
> 				5CD286261E6A669D0094FDC8 /* rtp_to_ntp_estimator.cc */,
> 				41F9BFE52051DECD00ABF0B9 /* runtime_enabled_features_default.cc */,
>-				5C4B4CB31E4320A9002651C8 /* rw_lock.cc */,
>-				5C4B4CAD1E4320A9002651C8 /* rw_lock_posix.cc */,
>-				5C4B4CAE1E4320A9002651C8 /* rw_lock_posix.h */,
> 				5C4B4CB41E4320A9002651C8 /* sleep.cc */,
>-				5C4B4CB81E4320A9002651C8 /* timestamp_extrapolator.cc */,
> 			);
> 			name = source;
> 			sourceTree = "<group>";
>@@ -8230,8 +9032,6 @@
> 			children = (
> 				5C4B4D051E4322F6002651C8 /* basicportallocator.cc */,
> 				5C4B4D061E4322F6002651C8 /* basicportallocator.h */,
>-				5C4B4D071E4322F6002651C8 /* socketmonitor.cc */,
>-				5C4B4D081E4322F6002651C8 /* socketmonitor.h */,
> 				41F9BFE82051DEEA00ABF0B9 /* turnportfactory.cc */,
> 				41F9BFE72051DEE900ABF0B9 /* turnportfactory.h */,
> 			);
>@@ -8292,6 +9092,8 @@
> 		5C63F8CB1E416CC3002CA531 /* third_party */ = {
> 			isa = PBXGroup;
> 			children = (
>+				419241912127583900634FCF /* rnnnoise */,
>+				413AD19E21265ACB003F7263 /* abseil-cpp */,
> 				5C63FC631E4184C0002CA531 /* boringssl */,
> 				5C63F8CC1E416CEB002CA531 /* jsoncpp */,
> 				5CDD90841E43D30300621E92 /* libsrtp */,
>@@ -8345,21 +9147,26 @@
> 		5C63F8E21E41732B002CA531 /* api */ = {
> 			isa = PBXGroup;
> 			children = (
>+				41F9BF792051C7FD00ABF0B9 /* array_view.h */,
> 				5CD2853B1E6A61BE0094FDC8 /* audio_codecs */,
>+				415F1FCA212730F000064CBF /* audio_frame.cc */,
>+				415F1FCB212730F000064CBF /* audio_frame.h */,
>+				415F1FC9212730F000064CBF /* audio_mixer.h */,
>+				41F2637521267B7600274F59 /* audio_options.cc */,
>+				41F2637821267B7600274F59 /* audio_options.h */,
>+				41F2637921267B7700274F59 /* bitrate_constraints.h */,
> 				5CDD8C451E43C57900621E92 /* call */,
>-				4145E4C01EF8962800FCF6E6 /* stats */,
>-				5CD284601E6A57DD0094FDC8 /* video */,
>-				412455581EF88AD900F11809 /* video_codecs */,
>-				41F9BF792051C7FD00ABF0B9 /* array_view.h */,
> 				419C830E1FE2427F0040C30F /* candidate.cc */,
> 				41F9BF772051C7FD00ABF0B9 /* candidate.h */,
> 				41F9BF822051C80000ABF0B9 /* cryptoparams.h */,
>+				41F2637721267B7600274F59 /* datachannelinterface.cc */,
> 				5C63F8E71E41737B002CA531 /* datachannelinterface.h */,
> 				5C63F8EB1E41737B002CA531 /* dtmfsenderinterface.h */,
>-				5C63F8ED1E41737B002CA531 /* fakemetricsobserver.cc */,
>-				5C63F8EE1E41737B002CA531 /* fakemetricsobserver.h */,
>+				415F1FC8212730F000064CBF /* echo_control.h */,
>+				41F2637621267B7600274F59 /* fec_controller.h */,
> 				41F9BF782051C7FD00ABF0B9 /* jsep.cc */,
> 				5C63F8EF1E41737B002CA531 /* jsep.h */,
>+				41F2637B21267B7700274F59 /* jsepicecandidate.cc */,
> 				5C63F8F11E41737B002CA531 /* jsepicecandidate.h */,
> 				5C63F8F31E41737B002CA531 /* jsepsessiondescription.h */,
> 				5C63F8F61E41737B002CA531 /* mediaconstraintsinterface.cc */,
>@@ -8371,8 +9178,8 @@
> 				5CD286301E6A67BF0094FDC8 /* mediatypes.cc */,
> 				5CD286311E6A67BF0094FDC8 /* mediatypes.h */,
> 				5C63F9021E41737B002CA531 /* notifier.h */,
>-				419C82E81FE20E8E0040C30F /* optional.cc */,
> 				5C63F9081E41737B002CA531 /* peerconnectionfactoryproxy.h */,
>+				41F2637421267B7500274F59 /* peerconnectioninterface.cc */,
> 				5C63F9091E41737B002CA531 /* peerconnectioninterface.h */,
> 				5C63F90A1E41737B002CA531 /* peerconnectionproxy.h */,
> 				419C830F1FE2427F0040C30F /* proxy.cc */,
>@@ -8385,18 +9192,21 @@
> 				41F9BF7F2051C7FF00ABF0B9 /* rtp_headers.h */,
> 				419C83111FE2427F0040C30F /* rtpparameters.cc */,
> 				5C63F9151E41737B002CA531 /* rtpparameters.h */,
>+				41F2637A21267B7700274F59 /* rtpreceiverinterface.cc */,
> 				5C63F9181E41737B002CA531 /* rtpreceiverinterface.h */,
> 				5C63F91B1E41737B002CA531 /* rtpsenderinterface.h */,
>+				41F2637C21267B7700274F59 /* rtptransceiverinterface.cc */,
> 				41F9BF802051C7FF00ABF0B9 /* rtptransceiverinterface.h */,
> 				41F9BF842051C80000ABF0B9 /* setremotedescriptionobserverinterface.h */,
>+				4145E4C01EF8962800FCF6E6 /* stats */,
> 				5C63F9211E41737B002CA531 /* statstypes.cc */,
> 				5C63F9221E41737B002CA531 /* statstypes.h */,
>+				41E02CC62127359F00C27CD6 /* transport */,
> 				41F9BF832051C80000ABF0B9 /* turncustomizer.h */,
>-				419C83101FE2427F0040C30F /* umametrics.cc */,
> 				5C63F9241E41737B002CA531 /* umametrics.h */,
>-				41F9BF7B2051C7FE00ABF0B9 /* videosinkinterface.h */,
>-				41F9BF812051C80000ABF0B9 /* videosourceinterface.cc */,
>-				41F9BF7C2051C7FE00ABF0B9 /* videosourceinterface.h */,
>+				419241D421275A1800634FCF /* units */,
>+				5CD284601E6A57DD0094FDC8 /* video */,
>+				412455581EF88AD900F11809 /* video_codecs */,
> 				5C63F9281E41737B002CA531 /* videosourceproxy.h */,
> 			);
> 			path = api;
>@@ -8416,16 +9226,22 @@
> 			children = (
> 				5C63F98A1E4174F6002CA531 /* asyncstuntcpsocket.cc */,
> 				5C63F98B1E4174F6002CA531 /* asyncstuntcpsocket.h */,
>+				4192414A2127376D00634FCF /* basicasyncresolverfactory.cc */,
>+				419241512127376E00634FCF /* basicasyncresolverfactory.h */,
> 				5C63F98C1E4174F6002CA531 /* basicpacketsocketfactory.cc */,
> 				5C63F98D1E4174F6002CA531 /* basicpacketsocketfactory.h */,
> 				5C63F98F1E4174F6002CA531 /* candidatepairinterface.h */,
>-				5C63F9901E4174F6002CA531 /* common.h */,
> 				419C83D81FE247F20040C30F /* dtlstransport.cc */,
> 				419C83D41FE247F10040C30F /* dtlstransport.h */,
> 				419C83D51FE247F10040C30F /* dtlstransportinternal.cc */,
> 				419C83D61FE247F10040C30F /* dtlstransportinternal.h */,
>+				4192414C2127376E00634FCF /* fakecandidatepair.h */,
>+				419241492127376D00634FCF /* fakedtlstransport.h */,
>+				4192414F2127376E00634FCF /* fakeicetransport.h */,
>+				4192414E2127376E00634FCF /* fakepackettransport.h */,
> 				5C63F9951E4174F6002CA531 /* fakeportallocator.h */,
> 				419C83D71FE247F20040C30F /* icetransportinternal.cc */,
>+				419241502127376E00634FCF /* icetransportinternal.h */,
> 				5C63F9971E4174F6002CA531 /* p2pconstants.cc */,
> 				5C63F9981E4174F6002CA531 /* p2pconstants.h */,
> 				5C63F99C1E4174F6002CA531 /* p2ptransportchannel.cc */,
>@@ -8445,12 +9261,12 @@
> 				5C63F9A61E4174F6002CA531 /* portinterface.h */,
> 				5C63F9A81E4174F6002CA531 /* pseudotcp.cc */,
> 				5C63F9A91E4174F6002CA531 /* pseudotcp.h */,
>+				419241482127376D00634FCF /* regatheringcontroller.cc */,
>+				4192414B2127376D00634FCF /* regatheringcontroller.h */,
> 				5C63F9AB1E4174F6002CA531 /* relayport.cc */,
> 				5C63F9AC1E4174F6002CA531 /* relayport.h */,
> 				5C63F9AE1E4174F6002CA531 /* relayserver.cc */,
> 				5C63F9AF1E4174F6002CA531 /* relayserver.h */,
>-				5C63F9B01E4174F6002CA531 /* session.cc */,
>-				5C63F9B11E4174F6002CA531 /* session.h */,
> 				5C63F9B51E4174F6002CA531 /* stun.cc */,
> 				5C63F9B61E4174F6002CA531 /* stun.h */,
> 				5C63F9B81E4174F6002CA531 /* stunport.cc */,
>@@ -8476,6 +9292,7 @@
> 				5C63F9DA1E4174F6002CA531 /* turnserver.h */,
> 				5C63F9DB1E4174F6002CA531 /* udpport.h */,
> 				419C83D31FE247F00040C30F /* udptransport.cc */,
>+				4192414D2127376E00634FCF /* udptransport.h */,
> 			);
> 			path = base;
> 			sourceTree = "<group>";
>@@ -8483,8 +9300,6 @@
> 		5C63FA351E417608002CA531 /* pc */ = {
> 			isa = PBXGroup;
> 			children = (
>-				5C63FA361E41761F002CA531 /* audiomonitor.cc */,
>-				5C63FA371E41761F002CA531 /* audiomonitor.h */,
> 				5CD284C41E6A60570094FDC8 /* audiotrack.cc */,
> 				5CD284C51E6A60570094FDC8 /* audiotrack.h */,
> 				5C63FA3A1E41761F002CA531 /* bundlefilter.cc */,
>@@ -8494,8 +9309,6 @@
> 				5C63FA401E41761F002CA531 /* channelmanager.cc */,
> 				5C63FA411E41761F002CA531 /* channelmanager.h */,
> 				4145E4C91EF8CB3100FCF6E6 /* createpeerconnectionfactory.cc */,
>-				5C63FA431E41761F002CA531 /* currentspeakermonitor.cc */,
>-				5C63FA441E41761F002CA531 /* currentspeakermonitor.h */,
> 				5CD284C61E6A60570094FDC8 /* datachannel.cc */,
> 				5CD284C71E6A60570094FDC8 /* datachannel.h */,
> 				419C82D61FE20E580040C30F /* dtlssrtptransport.cc */,
>@@ -8510,10 +9323,10 @@
> 				5CD284711E6A5D080094FDC8 /* jsepsessiondescription.cc */,
> 				41F9BFDD2051DE1700ABF0B9 /* jseptransport.cc */,
> 				41F9BFDB2051DE1600ABF0B9 /* jseptransport.h */,
>+				4102F68721273205006AE8D7 /* jseptransportcontroller.cc */,
>+				4102F68521273204006AE8D7 /* jseptransportcontroller.h */,
> 				5CD284CA1E6A60570094FDC8 /* localaudiosource.cc */,
> 				5CD284CB1E6A60570094FDC8 /* localaudiosource.h */,
>-				5C63FA481E41761F002CA531 /* mediamonitor.cc */,
>-				5C63FA491E41761F002CA531 /* mediamonitor.h */,
> 				5C63FA4B1E41761F002CA531 /* mediasession.cc */,
> 				5C63FA4C1E41761F002CA531 /* mediasession.h */,
> 				5CD284CE1E6A60570094FDC8 /* mediastream.cc */,
>@@ -8525,6 +9338,7 @@
> 				5CD284D41E6A60570094FDC8 /* peerconnection.h */,
> 				5CD2847D1E6A5E740094FDC8 /* peerconnectionfactory.cc */,
> 				5CD2847E1E6A5E740094FDC8 /* peerconnectionfactory.h */,
>+				4102F68C21273206006AE8D7 /* peerconnectioninternal.h */,
> 				41F9BFDC2051DE1700ABF0B9 /* peerconnectionwrapper.cc */,
> 				41F9BFDA2051DE1600ABF0B9 /* peerconnectionwrapper.h */,
> 				5CD284D91E6A60570094FDC8 /* remoteaudiosource.cc */,
>@@ -8533,8 +9347,12 @@
> 				5C63FA511E41761F002CA531 /* rtcpmuxfilter.h */,
> 				5CD284DB1E6A60570094FDC8 /* rtcstatscollector.cc */,
> 				5CD284DC1E6A60570094FDC8 /* rtcstatscollector.h */,
>+				4102F68921273205006AE8D7 /* rtcstatstraversal.cc */,
>+				4102F68421273204006AE8D7 /* rtcstatstraversal.h */,
> 				419C82C71FE20E2B0040C30F /* rtpmediautils.cc */,
> 				419C82C81FE20E2C0040C30F /* rtpmediautils.h */,
>+				4102F68A21273206006AE8D7 /* rtpparametersconversion.cc */,
>+				4102F68821273205006AE8D7 /* rtpparametersconversion.h */,
> 				5CD284DD1E6A60570094FDC8 /* rtpreceiver.cc */,
> 				5CD284DE1E6A60570094FDC8 /* rtpreceiver.h */,
> 				5CD284DF1E6A60570094FDC8 /* rtpsender.cc */,
>@@ -8562,15 +9380,14 @@
> 				5CD284E51E6A60570094FDC8 /* streamcollection.h */,
> 				5CD284E61E6A60570094FDC8 /* trackmediainfomap.cc */,
> 				5CD284E71E6A60570094FDC8 /* trackmediainfomap.h */,
>-				419C82D51FE20E580040C30F /* transportcontroller.cc */,
>-				419C82CB1FE20E540040C30F /* transportcontroller.h */,
>+				4102F68621273205006AE8D7 /* transportstats.cc */,
>+				4102F68B21273206006AE8D7 /* transportstats.h */,
> 				5CD284E81E6A60570094FDC8 /* videocapturertracksource.cc */,
> 				5CD284E91E6A60570094FDC8 /* videocapturertracksource.h */,
> 				5CD284EA1E6A60570094FDC8 /* videotrack.cc */,
> 				5CD284EB1E6A60570094FDC8 /* videotrack.h */,
> 				5CD284EC1E6A60570094FDC8 /* videotracksource.cc */,
> 				5CD284ED1E6A60570094FDC8 /* videotracksource.h */,
>-				5C63FA561E41761F002CA531 /* voicechannel.h */,
> 				5CD284751E6A5D550094FDC8 /* webrtcsdp.cc */,
> 				5CD284761E6A5D550094FDC8 /* webrtcsdp.h */,
> 				5CD284EE1E6A60570094FDC8 /* webrtcsessiondescriptionfactory.cc */,
>@@ -8683,15 +9500,36 @@
> 		5CD284601E6A57DD0094FDC8 /* video */ = {
> 			isa = PBXGroup;
> 			children = (
>+				4102F6D221273414006AE8D7 /* color_space.cc */,
>+				4102F6D821273416006AE8D7 /* color_space.h */,
>+				41F263C72126818A00274F59 /* encoded_frame.cc */,
>+				41F263C22126818900274F59 /* encoded_frame.h */,
>+				4102F6D621273415006AE8D7 /* i010_buffer.cc */,
>+				41F263BF2126818900274F59 /* i010_buffer.h */,
> 				5CD284611E6A57F40094FDC8 /* i420_buffer.cc */,
> 				5CD284621E6A57F40094FDC8 /* i420_buffer.h */,
>+				4102F6D921273416006AE8D7 /* video_bitrate_allocation.cc */,
>+				41F263C42126818900274F59 /* video_bitrate_allocation.h */,
>+				4102F6DA21273416006AE8D7 /* video_bitrate_allocator_factory.h */,
>+				41F263C12126818900274F59 /* video_bitrate_allocator.h */,
> 				419C840D1FE249AA0040C30F /* video_content_type.cc */,
> 				419C840F1FE249AB0040C30F /* video_content_type.h */,
>-				5CD284641E6A57F40094FDC8 /* video_frame.cc */,
>-				5CD284651E6A57F40094FDC8 /* video_frame.h */,
> 				4124554A1EF8874300F11809 /* video_frame_buffer.cc */,
> 				5CD284631E6A57F40094FDC8 /* video_frame_buffer.h */,
>+				5CD284641E6A57F40094FDC8 /* video_frame.cc */,
>+				5CD284651E6A57F40094FDC8 /* video_frame.h */,
> 				5CD284661E6A57F40094FDC8 /* video_rotation.h */,
>+				4102F6D321273415006AE8D7 /* video_sink_interface.h */,
>+				41F263C62126818A00274F59 /* video_source_interface.cc */,
>+				41F263BE2126818800274F59 /* video_source_interface.h */,
>+				4102F6D121273414006AE8D7 /* video_stream_decoder_create.cc */,
>+				4102F6D421273415006AE8D7 /* video_stream_decoder_create.h */,
>+				41F263C52126818A00274F59 /* video_stream_decoder.h */,
>+				4102F6D521273415006AE8D7 /* video_stream_encoder_create.cc */,
>+				41F263C02126818900274F59 /* video_stream_encoder_create.h */,
>+				41F263C32126818900274F59 /* video_stream_encoder_interface.h */,
>+				41F263BD2126818800274F59 /* video_stream_encoder_observer.h */,
>+				4102F6D721273415006AE8D7 /* video_stream_encoder_settings.h */,
> 				419C840C1FE249A90040C30F /* video_timing.cc */,
> 				419C840E1FE249AA0040C30F /* video_timing.h */,
> 			);
>@@ -8723,6 +9561,8 @@
> 				419C835A1FE246770040C30F /* isac */,
> 				419C83EB1FE248770040C30F /* L16 */,
> 				419C82E71FE20E7D0040C30F /* opus */,
>+				4192419E2127588200634FCF /* audio_codec_pair_id.cc */,
>+				4192419D2127588200634FCF /* audio_codec_pair_id.h */,
> 				5CD2853D1E6A61D20094FDC8 /* audio_decoder.cc */,
> 				5CD2853E1E6A61D20094FDC8 /* audio_decoder.h */,
> 				5CD2853C1E6A61D20094FDC8 /* audio_decoder_factory.h */,
>@@ -8802,8 +9642,6 @@
> 				5CD2858A1E6A63430094FDC8 /* matched_filter_lag_aggregator.h */,
> 				41F9BFCE2051DDE400ABF0B9 /* matrix_buffer.cc */,
> 				41F9BFD02051DDE400ABF0B9 /* matrix_buffer.h */,
>-				5CD2858D1E6A63430094FDC8 /* output_selector.cc */,
>-				5CD2858E1E6A63430094FDC8 /* output_selector.h */,
> 				4145E4D71EF8CC9A00FCF6E6 /* render_buffer.cc */,
> 				4145E4D81EF8CC9A00FCF6E6 /* render_buffer.h */,
> 				5CD285911E6A63430094FDC8 /* render_delay_buffer.cc */,
>@@ -8871,13 +9709,13 @@
> 		5CDD83681E4399E400621E92 /* modules */ = {
> 			isa = PBXGroup;
> 			children = (
>+				41F2638621267F2500274F59 /* third_party */,
> 				5CDD84051E439AF200621E92 /* audio_coding */,
> 				5C119FEF1E4573CA004F0987 /* audio_device */,
> 				5CD284811E6A5F2B0094FDC8 /* audio_mixer */,
> 				5CDD84881E43AEC900621E92 /* audio_processing */,
> 				5CDD90511E43D10400621E92 /* bitrate_controller */,
> 				5CDD8FA61E43CCE100621E92 /* congestion_controller */,
>-				5CDD87601E43BB1D00621E92 /* media_file */,
> 				5CDD8F951E43CCA200621E92 /* pacing */,
> 				5CDD8FBD1E43CD3D00621E92 /* remote_bitrate_estimator */,
> 				5CDD87F91E43BE1C00621E92 /* rtp_rtcp */,
>@@ -8890,10 +9728,16 @@
> 		5CDD83691E4399F400621E92 /* video_coding */ = {
> 			isa = PBXGroup;
> 			children = (
>+				41E02C83212734B800C27CD6 /* decoder_database.cc */,
>+				41E02C7F212734B800C27CD6 /* decoder_database.h */,
>+				41E02C80212734B800C27CD6 /* encoder_database.cc */,
>+				41E02C81212734B800C27CD6 /* encoder_database.h */,
>+				41E02C82212734B800C27CD6 /* fec_controller_default.cc */,
>+				41E02C84212734B900C27CD6 /* fec_controller_default.h */,
>+				41E02C86212734B900C27CD6 /* fec_rate_table.h */,
>+				41E02C85212734B900C27CD6 /* video_coding_defines.cc */,
> 				5CDD853D1E43B3F800621E92 /* codecs */,
> 				5CDD836A1E439A0D00621E92 /* utility */,
>-				5CDD83871E439A6F00621E92 /* codec_database.cc */,
>-				5CDD83881E439A6F00621E92 /* codec_database.h */,
> 				5CDD83891E439A6F00621E92 /* codec_timer.cc */,
> 				5CDD838A1E439A6F00621E92 /* codec_timer.h */,
> 				5CDD838B1E439A6F00621E92 /* decoding_state.cc */,
>@@ -8935,8 +9779,6 @@
> 				5CDD83B01E439A6F00621E92 /* packet.h */,
> 				5CDD83AD1E439A6F00621E92 /* packet_buffer.cc */,
> 				5CDD83AE1E439A6F00621E92 /* packet_buffer.h */,
>-				5CDD83B31E439A6F00621E92 /* protection_bitrate_calculator.cc */,
>-				5CDD83B41E439A6F00621E92 /* protection_bitrate_calculator.h */,
> 				413092071EF8D11100757C55 /* qp_parser.cc */,
> 				413092081EF8D11100757C55 /* qp_parser.h */,
> 				5CDD83B51E439A6F00621E92 /* receiver.cc */,
>@@ -8963,6 +9805,10 @@
> 		5CDD836A1E439A0D00621E92 /* utility */ = {
> 			isa = PBXGroup;
> 			children = (
>+				419241EA21275AFA00634FCF /* simulcast_rate_allocator.cc */,
>+				419241EC21275AFA00634FCF /* simulcast_rate_allocator.h */,
>+				419241EB21275AFA00634FCF /* simulcast_utility.cc */,
>+				419241E921275AF900634FCF /* simulcast_utility.h */,
> 				5CD285371E6A61980094FDC8 /* default_video_bitrate_allocator.cc */,
> 				5CD285381E6A61980094FDC8 /* default_video_bitrate_allocator.h */,
> 				5CDD836B1E439A3500621E92 /* frame_dropper.cc */,
>@@ -9006,44 +9852,22 @@
> 				5CDD840B1E439B2900621E92 /* audio_encoder.h */,
> 				5CDD840C1E439B2900621E92 /* audio_format_conversion.cc */,
> 				5CDD840D1E439B2900621E92 /* audio_format_conversion.h */,
>-				5CDD84111E439B2900621E92 /* builtin_audio_decoder_factory.h */,
>-				413092251EF8D6E600757C55 /* builtin_audio_encoder_factory.h */,
> 				5CDD84121E439B2900621E92 /* legacy_encoded_audio_frame.cc */,
> 				5CDD84131E439B2900621E92 /* legacy_encoded_audio_frame.h */,
> 			);
> 			path = codecs;
> 			sourceTree = "<group>";
> 		};
>-		5CDD84291E43ADEE00621E92 /* voice_engine */ = {
>-			isa = PBXGroup;
>-			children = (
>-				413092041EF8D0C800757C55 /* audio_level.cc */,
>-				413092031EF8D0C800757C55 /* audio_level.h */,
>-				5CDD842E1E43AE2900621E92 /* channel.cc */,
>-				5CDD842F1E43AE2900621E92 /* channel.h */,
>-				5CDD842C1E43AE2900621E92 /* channel_proxy.cc */,
>-				5CDD842D1E43AE2900621E92 /* channel_proxy.h */,
>-				4130920D1EF8D40100757C55 /* transport_feedback_packet_loss_tracker.cc */,
>-				5CD286021E6A64840094FDC8 /* transport_feedback_packet_loss_tracker.h */,
>-				4130920C1EF8D40100757C55 /* transport_feedback_packet_loss_tracker.h */,
>-				5CDD843C1E43AE2900621E92 /* utility.cc */,
>-				5CDD843D1E43AE2900621E92 /* utility.h */,
>-			);
>-			path = voice_engine;
>-			sourceTree = "<group>";
>-		};
> 		5CDD84881E43AEC900621E92 /* audio_processing */ = {
> 			isa = PBXGroup;
> 			children = (
>+				41A08BD921272F3E001D5D7B /* agc2 */,
> 				5CDD86A91E43B9B900621E92 /* aec */,
> 				5CD2855F1E6A63050094FDC8 /* aec3 */,
> 				4130922A1EF8D73500757C55 /* aec_dump */,
> 				5CDD86E51E43BA5B00621E92 /* agc */,
>-				4130920B1EF8D2CD00757C55 /* agc2 */,
>-				5CDD90631E43D14B00621E92 /* beamformer */,
> 				5CDD86961E43B98200621E92 /* echo_detector */,
> 				5CDD84C91E43AF1D00621E92 /* include */,
>-				5CDD90041E43CECE00621E92 /* level_controller */,
> 				5CDD87E21E43BD9100621E92 /* logging */,
> 				5CDD85EC1E43B84200621E92 /* ns */,
> 				5CDD8C6B1E43C64F00621E92 /* transient */,
>@@ -9058,10 +9882,16 @@
> 				5CDD848F1E43AF1300621E92 /* echo_cancellation_impl.h */,
> 				5CDD84901E43AF1300621E92 /* echo_control_mobile_impl.cc */,
> 				5CDD84911E43AF1300621E92 /* echo_control_mobile_impl.h */,
>+				415F1FBC2127304C00064CBF /* echo_cancellation_proxy.cc */,
>+				415F1FBB2127304C00064CBF /* echo_cancellation_proxy.h */,
>+				415F1FBA2127304B00064CBF /* echo_control_mobile_proxy.cc */,
>+				415F1FB92127304B00064CBF /* echo_control_mobile_proxy.h */,
> 				5CDD84921E43AF1300621E92 /* gain_control_for_experimental_agc.cc */,
> 				5CDD84931E43AF1300621E92 /* gain_control_for_experimental_agc.h */,
> 				5CDD84941E43AF1300621E92 /* gain_control_impl.cc */,
> 				5CDD84951E43AF1300621E92 /* gain_control_impl.h */,
>+				41A08BCE21272EE1001D5D7B /* gain_controller2.cc */,
>+				41A08BCD21272EE1001D5D7B /* gain_controller2.h */,
> 				5CDD84981E43AF1300621E92 /* level_estimator_impl.cc */,
> 				5CDD84991E43AF1300621E92 /* level_estimator_impl.h */,
> 				5CD2855B1E6A62ED0094FDC8 /* low_cut_filter.cc */,
>@@ -9141,43 +9971,61 @@
> 		5CDD851A1E43B38F00621E92 /* call */ = {
> 			isa = PBXGroup;
> 			children = (
>+				4102F6B621273381006AE8D7 /* audio_receive_stream.cc */,
> 				5CD2849D1E6A5F9F0094FDC8 /* audio_receive_stream.h */,
> 				5CD2849E1E6A5F9F0094FDC8 /* audio_send_stream.cc */,
> 				5CD2849F1E6A5F9F0094FDC8 /* audio_send_stream.h */,
>+				415F1FC52127308E00064CBF /* audio_state.cc */,
> 				5CD284A01E6A5F9F0094FDC8 /* audio_state.h */,
> 				5CDD851C1E43B39C00621E92 /* bitrate_allocator.cc */,
> 				5CDD851D1E43B39C00621E92 /* bitrate_allocator.h */,
>+				415F1FD72127313F00064CBF /* call_config.cc */,
>+				415F1FDA2127313F00064CBF /* call_config.h */,
> 				5CDD85221E43B39C00621E92 /* call.cc */,
> 				5CD284A61E6A5F9F0094FDC8 /* call.h */,
> 				4145E4E11EF8CD7200FCF6E6 /* callfactory.cc */,
> 				4145E4E31EF8CD7C00FCF6E6 /* callfactory.h */,
>-				4145E4E21EF8CD7700FCF6E6 /* callfactoryinterface.h */,
>-				5CDD85261E43B39C00621E92 /* flexfec_receive_stream.h */,
>+				4102F6B721273381006AE8D7 /* degraded_call.cc */,
>+				415F1FD22127313E00064CBF /* degraded_call.h */,
>+				41299B942127369000B3414B /* fake_network_pipe.cc */,
>+				41299B952127369100B3414B /* fake_network_pipe.h */,
> 				5CD284A81E6A5F9F0094FDC8 /* flexfec_receive_stream_impl.cc */,
> 				5CD284A91E6A5F9F0094FDC8 /* flexfec_receive_stream_impl.h */,
>+				4102F6B821273381006AE8D7 /* flexfec_receive_stream.cc */,
>+				5CDD85261E43B39C00621E92 /* flexfec_receive_stream.h */,
>+				415F1FD82127313F00064CBF /* packet_receiver.cc */,
>+				4102F6B521273381006AE8D7 /* packet_receiver.h */,
>+				415F1FD02127313D00064CBF /* receive_time_calculator.cc */,
>+				415F1FD52127313E00064CBF /* receive_time_calculator.h */,
> 				419C82A61FE20DC80040C30F /* rtcp_demuxer.cc */,
> 				419C829F1FE20DC60040C30F /* rtcp_demuxer.h */,
> 				419C82AA1FE20DCA0040C30F /* rtcp_packet_sink_interface.h */,
>+				415F1FD92127313F00064CBF /* rtp_bitrate_configurator.cc */,
> 				419C82A11FE20DC70040C30F /* rtp_config.cc */,
> 				419C82A51FE20DC80040C30F /* rtp_config.h */,
> 				413092001EF8D0A600757C55 /* rtp_demuxer.cc */,
> 				413091FF1EF8D0A400757C55 /* rtp_demuxer.h */,
> 				419C82A31FE20DC70040C30F /* rtp_packet_sink_interface.h */,
>+				4102F6AA212732B0006AE8D7 /* rtp_payload_params.cc */,
>+				415F1FD12127313E00064CBF /* rtp_payload_params.h */,
> 				419C82AC1FE20DCB0040C30F /* rtp_rtcp_demuxer_helper.cc */,
> 				419C82A21FE20DC70040C30F /* rtp_rtcp_demuxer_helper.h */,
>+				419C82AD1FE20DCB0040C30F /* rtp_stream_receiver_controller_interface.h */,
> 				419C82A91FE20DCA0040C30F /* rtp_stream_receiver_controller.cc */,
> 				419C82AE1FE20DCC0040C30F /* rtp_stream_receiver_controller.h */,
>-				419C82AD1FE20DCB0040C30F /* rtp_stream_receiver_controller_interface.h */,
>+				413092171EF8D63900757C55 /* rtp_transport_controller_send_interface.h */,
> 				413092181EF8D63F00757C55 /* rtp_transport_controller_send.cc */,
> 				413092161EF8D63400757C55 /* rtp_transport_controller_send.h */,
>-				413092171EF8D63900757C55 /* rtp_transport_controller_send_interface.h */,
>+				415F1FD62127313E00064CBF /* rtp_video_sender_interface.h */,
>+				415F1FD32127313E00064CBF /* rtp_video_sender.cc */,
>+				415F1FD42127313E00064CBF /* rtp_video_sender.h */,
> 				419C82A41FE20DC80040C30F /* rtx_receive_stream.cc */,
> 				419C82AF1FE20DCC0040C30F /* rtx_receive_stream.h */,
>+				419241CE2127597C00634FCF /* simulated_network.cc */,
>+				419241CF2127597C00634FCF /* simulated_network.h */,
> 				41F9BF9D2051C86900ABF0B9 /* ssrc_binding_observer.h */,
> 				5CD284AE1E6A5F9F0094FDC8 /* syncable.cc */,
> 				5CD284AF1E6A5F9F0094FDC8 /* syncable.h */,
>-				419C82A81FE20DCA0040C30F /* video_config.cc */,
>-				419C82B01FE20DCC0040C30F /* video_config.h */,
> 				419C82AB1FE20DCB0040C30F /* video_receive_stream.cc */,
> 				419C82B11FE20DCD0040C30F /* video_receive_stream.h */,
> 				419C82A01FE20DC60040C30F /* video_send_stream.cc */,
>@@ -9210,14 +10058,19 @@
> 		5CDD85591E43B5AE00621E92 /* video */ = {
> 			isa = PBXGroup;
> 			children = (
>+				4102F6BF212733B6006AE8D7 /* video_quality_observer.cc */,
>+				4102F6C5212733B6006AE8D7 /* video_quality_observer.h */,
>+				4102F6C2212733B6006AE8D7 /* video_send_stream_impl.cc */,
>+				4102F6BE212733B5006AE8D7 /* video_send_stream_impl.h */,
>+				4102F6C1212733B6006AE8D7 /* video_stream_decoder_impl.cc */,
>+				4102F6C4212733B6006AE8D7 /* video_stream_decoder_impl.h */,
>+				4102F6C0212733B6006AE8D7 /* video_stream_encoder.h */,
> 				5CDD855C1E43B5C000621E92 /* call_stats.cc */,
> 				5CDD855D1E43B5C000621E92 /* call_stats.h */,
> 				5CDD85601E43B5C000621E92 /* encoder_rtcp_feedback.cc */,
> 				5CDD85611E43B5C000621E92 /* encoder_rtcp_feedback.h */,
> 				5CDD85661E43B5C000621E92 /* overuse_frame_detector.cc */,
> 				5CDD85671E43B5C000621E92 /* overuse_frame_detector.h */,
>-				5CDD856A1E43B5C000621E92 /* payload_router.cc */,
>-				5CDD856B1E43B5C000621E92 /* payload_router.h */,
> 				5CD286281E6A66C80094FDC8 /* quality_threshold.cc */,
> 				5CD286291E6A66C80094FDC8 /* quality_threshold.h */,
> 				5CDD856D1E43B5C000621E92 /* receive_statistics_proxy.cc */,
>@@ -9318,7 +10171,6 @@
> 				5CDD86361E43B8B500621E92 /* spl_init.c */,
> 				5CDD86371E43B8B500621E92 /* spl_inl.c */,
> 				5CDD863B1E43B8B500621E92 /* spl_sqrt.c */,
>-				5CDD863A1E43B8B500621E92 /* spl_sqrt_floor.c */,
> 				5CDD863C1E43B8B500621E92 /* splitting_filter.c */,
> 				5CDD863D1E43B8B500621E92 /* sqrt_of_one_minus_x_squared.c */,
> 				5CDD863F1E43B8B500621E92 /* vector_scaling_operations.c */,
>@@ -9466,19 +10318,6 @@
> 			name = resampler;
> 			sourceTree = "<group>";
> 		};
>-		5CDD87601E43BB1D00621E92 /* media_file */ = {
>-			isa = PBXGroup;
>-			children = (
>-				5CDD87661E43BB3000621E92 /* media_file.h */,
>-				5CDD87611E43BB3000621E92 /* media_file_defines.h */,
>-				5CDD87621E43BB3000621E92 /* media_file_impl.cc */,
>-				5CDD87631E43BB3000621E92 /* media_file_impl.h */,
>-				5CDD87641E43BB3000621E92 /* media_file_utility.cc */,
>-				5CDD87651E43BB3000621E92 /* media_file_utility.h */,
>-			);
>-			name = media_file;
>-			sourceTree = "<group>";
>-		};
> 		5CDD876D1E43BB7E00621E92 /* isac */ = {
> 			isa = PBXGroup;
> 			children = (
>@@ -9507,6 +10346,9 @@
> 		5CDD876F1E43BB8D00621E92 /* source */ = {
> 			isa = PBXGroup;
> 			children = (
>+				41299B8E2127367A00B3414B /* isac_vad.c */,
>+				41299B8F2127367B00B3414B /* isac_vad.h */,
>+				41299B902127367B00B3414B /* pitch_filter.h */,
> 				5CDD87731E43BC0500621E92 /* arith_routines.c */,
> 				5CDD87741E43BC0500621E92 /* arith_routines.h */,
> 				5CDD87711E43BC0500621E92 /* arith_routines_hist.c */,
>@@ -9525,11 +10367,8 @@
> 				5CDD87801E43BC0500621E92 /* encode_lpc_swb.h */,
> 				5CDD87821E43BC0500621E92 /* entropy_coding.c */,
> 				5CDD87831E43BC0500621E92 /* entropy_coding.h */,
>-				5CDD87841E43BC0500621E92 /* fft.c */,
> 				5CDD87851E43BC0500621E92 /* fft.h */,
> 				5CDD87861E43BC0500621E92 /* filter_functions.c */,
>-				5CDD87871E43BC0500621E92 /* filterbank_tables.c */,
>-				5CDD87881E43BC0500621E92 /* filterbank_tables.h */,
> 				5CDD87891E43BC0500621E92 /* filterbanks.c */,
> 				5CDD878A1E43BC0500621E92 /* intialize.c */,
> 				5CDD878D1E43BC0500621E92 /* isac.c */,
>@@ -9592,11 +10431,14 @@
> 		5CDD87FA1E43BE2600621E92 /* source */ = {
> 			isa = PBXGroup;
> 			children = (
>-				5CDD88EB1E43BF2500621E92 /* rtcp_packet */,
>+				419241D2212759A100634FCF /* fec_private_tables_random.cc */,
>+				419241AF2127590200634FCF /* rtp_generic_frame_descriptor_extension.cc */,
> 				5CDD87FC1E43BE3C00621E92 /* byte_io.h */,
>+				4102F6EB21273460006AE8D7 /* contributing_sources.cc */,
> 				5CDD87FE1E43BE3C00621E92 /* dtmf_queue.cc */,
> 				5CDD87FF1E43BE3C00621E92 /* dtmf_queue.h */,
> 				5CDD88001E43BE3C00621E92 /* fec_private_tables_bursty.h */,
>+				4192418F2127581000634FCF /* fec_private_tables_bursty.cc */,
> 				5CDD88011E43BE3C00621E92 /* fec_private_tables_random.h */,
> 				5CDD88021E43BE3C00621E92 /* fec_test_helper.cc */,
> 				5CDD88031E43BE3C00621E92 /* fec_test_helper.h */,
>@@ -9604,10 +10446,10 @@
> 				5CDD88061E43BE3C00621E92 /* flexfec_header_reader_writer.h */,
> 				5CDD880A1E43BE3C00621E92 /* flexfec_receiver.cc */,
> 				5CDD880C1E43BE3C00621E92 /* flexfec_sender.cc */,
>-				5CDD880F1E43BE3C00621E92 /* forward_error_correction.cc */,
>-				5CDD88101E43BE3C00621E92 /* forward_error_correction.h */,
> 				5CDD880D1E43BE3C00621E92 /* forward_error_correction_internal.cc */,
> 				5CDD880E1E43BE3C00621E92 /* forward_error_correction_internal.h */,
>+				5CDD880F1E43BE3C00621E92 /* forward_error_correction.cc */,
>+				5CDD88101E43BE3C00621E92 /* forward_error_correction.h */,
> 				5CDD88151E43BE3C00621E92 /* packet_loss_stats.cc */,
> 				5CDD88161E43BE3C00621E92 /* packet_loss_stats.h */,
> 				5CDD88181E43BE3C00621E92 /* playout_delay_oracle.cc */,
>@@ -9617,41 +10459,44 @@
> 				5CDD881E1E43BE3C00621E92 /* remote_ntp_time_estimator.cc */,
> 				5CD285F81E6A64520094FDC8 /* rtcp_nack_stats.cc */,
> 				5CD285F91E6A64520094FDC8 /* rtcp_nack_stats.h */,
>+				5CDD88EB1E43BF2500621E92 /* rtcp_packet */,
> 				5CDD88211E43BE3C00621E92 /* rtcp_packet.cc */,
> 				5CDD88221E43BE3C00621E92 /* rtcp_packet.h */,
> 				5CDD88241E43BE3C00621E92 /* rtcp_receiver.cc */,
> 				5CDD88251E43BE3C00621E92 /* rtcp_receiver.h */,
> 				5CDD88271E43BE3C00621E92 /* rtcp_sender.cc */,
> 				5CDD88281E43BE3C00621E92 /* rtcp_sender.h */,
>-				419C82F61FE20EFB0040C30F /* rtcp_transceiver.cc */,
>-				419C82FA1FE20EFD0040C30F /* rtcp_transceiver.h */,
> 				419C82F91FE20EFD0040C30F /* rtcp_transceiver_config.cc */,
> 				419C82F81FE20EFC0040C30F /* rtcp_transceiver_config.h */,
> 				419C82F71FE20EFC0040C30F /* rtcp_transceiver_impl.cc */,
> 				419C82FC1FE20EFE0040C30F /* rtcp_transceiver_impl.h */,
>-				5CDD883A1E43BE3C00621E92 /* rtp_format.cc */,
>-				5CDD883B1E43BE3C00621E92 /* rtp_format.h */,
>+				419C82F61FE20EFB0040C30F /* rtcp_transceiver.cc */,
>+				419C82FA1FE20EFD0040C30F /* rtcp_transceiver.h */,
> 				5CDD882E1E43BE3C00621E92 /* rtp_format_h264.cc */,
> 				5CDD882F1E43BE3C00621E92 /* rtp_format_h264.h */,
> 				5CDD88301E43BE3C00621E92 /* rtp_format_video_generic.cc */,
> 				5CDD88311E43BE3C00621E92 /* rtp_format_video_generic.h */,
>-				419C82FD1FE20EFE0040C30F /* rtp_format_video_stereo.cc */,
>-				419C82FB1FE20EFE0040C30F /* rtp_format_video_stereo.h */,
> 				5CDD88351E43BE3C00621E92 /* rtp_format_vp8.cc */,
> 				5CDD88361E43BE3C00621E92 /* rtp_format_vp8.h */,
> 				5CDD88381E43BE3C00621E92 /* rtp_format_vp9.cc */,
> 				5CDD88391E43BE3C00621E92 /* rtp_format_vp9.h */,
>+				5CDD883A1E43BE3C00621E92 /* rtp_format.cc */,
>+				5CDD883B1E43BE3C00621E92 /* rtp_format.h */,
>+				4102F6E82127345F006AE8D7 /* rtp_generic_frame_descriptor_extension.h */,
>+				4102F6E52127345E006AE8D7 /* rtp_generic_frame_descriptor.cc */,
>+				4102F6E62127345F006AE8D7 /* rtp_generic_frame_descriptor.h */,
> 				4145E4DB1EF8CCEE00FCF6E6 /* rtp_header_extension_map.cc */,
> 				5CDD883F1E43BE3C00621E92 /* rtp_header_extensions.cc */,
> 				5CDD88401E43BE3C00621E92 /* rtp_header_extensions.h */,
> 				5CDD88411E43BE3C00621E92 /* rtp_header_parser.cc */,
>-				5CDD88481E43BE3C00621E92 /* rtp_packet.cc */,
>-				5CDD88491E43BE3C00621E92 /* rtp_packet.h */,
> 				5CDD88431E43BE3C00621E92 /* rtp_packet_history.cc */,
> 				5CDD88441E43BE3C00621E92 /* rtp_packet_history.h */,
> 				419C82FE1FE20F010040C30F /* rtp_packet_received.cc */,
> 				5CDD88451E43BE3C00621E92 /* rtp_packet_received.h */,
>+				4102F6E92127345F006AE8D7 /* rtp_packet_to_send.cc */,
> 				5CDD88461E43BE3C00621E92 /* rtp_packet_to_send.h */,
>+				5CDD88481E43BE3C00621E92 /* rtp_packet.cc */,
>+				5CDD88491E43BE3C00621E92 /* rtp_packet.h */,
> 				5CDD884B1E43BE3C00621E92 /* rtp_payload_registry.cc */,
> 				5CDD884C1E43BE3C00621E92 /* rtp_receiver_audio.cc */,
> 				5CDD884D1E43BE3C00621E92 /* rtp_receiver_audio.h */,
>@@ -9664,14 +10509,16 @@
> 				5CDD88541E43BE3C00621E92 /* rtp_rtcp_config.h */,
> 				5CDD88561E43BE3C00621E92 /* rtp_rtcp_impl.cc */,
> 				5CDD88571E43BE3C00621E92 /* rtp_rtcp_impl.h */,
>-				5CDD885D1E43BE3C00621E92 /* rtp_sender.cc */,
>-				5CDD885E1E43BE3C00621E92 /* rtp_sender.h */,
> 				5CDD88581E43BE3C00621E92 /* rtp_sender_audio.cc */,
> 				5CDD88591E43BE3C00621E92 /* rtp_sender_audio.h */,
> 				5CDD885B1E43BE3C00621E92 /* rtp_sender_video.cc */,
> 				5CDD885C1E43BE3C00621E92 /* rtp_sender_video.h */,
>+				5CDD885D1E43BE3C00621E92 /* rtp_sender.cc */,
>+				5CDD885E1E43BE3C00621E92 /* rtp_sender.h */,
> 				5CDD885F1E43BE3C00621E92 /* rtp_utility.cc */,
> 				5CDD88601E43BE3C00621E92 /* rtp_utility.h */,
>+				4102F6E72127345F006AE8D7 /* rtp_video_header.cc */,
>+				4102F6EA2127345F006AE8D7 /* rtp_video_header.h */,
> 				5CDD88641E43BE3C00621E92 /* time_util.cc */,
> 				5CDD88651E43BE3C00621E92 /* time_util.h */,
> 				5CDD88661E43BE3C00621E92 /* tmmbr_help.cc */,
>@@ -9682,7 +10529,6 @@
> 				5CDD886D1E43BE3C00621E92 /* ulpfec_header_reader_writer.h */,
> 				5CDD886E1E43BE3C00621E92 /* ulpfec_receiver_impl.cc */,
> 				5CDD886F1E43BE3C00621E92 /* ulpfec_receiver_impl.h */,
>-				5CDD88711E43BE3C00621E92 /* video_codec_information.h */,
> 			);
> 			name = source;
> 			path = rtp_rtcp/source;
>@@ -9748,6 +10594,8 @@
> 		5CDD89881E43BF8A00621E92 /* neteq */ = {
> 			isa = PBXGroup;
> 			children = (
>+				4102F6AC212732E6006AE8D7 /* expand_uma_logger.cc */,
>+				4102F6AD212732E7006AE8D7 /* expand_uma_logger.h */,
> 				5CDD8A5F1E43C00000621E92 /* tools */,
> 				5CDD89891E43BFB300621E92 /* accelerate.cc */,
> 				5CDD898A1E43BFB300621E92 /* accelerate.h */,
>@@ -9765,10 +10613,6 @@
> 				5CDD89A11E43BFB300621E92 /* cross_correlation.h */,
> 				5CDD89A71E43BFB300621E92 /* decision_logic.cc */,
> 				5CDD89A81E43BFB300621E92 /* decision_logic.h */,
>-				5CDD89A21E43BFB300621E92 /* decision_logic_fax.cc */,
>-				5CDD89A31E43BFB300621E92 /* decision_logic_fax.h */,
>-				5CDD89A41E43BFB300621E92 /* decision_logic_normal.cc */,
>-				5CDD89A51E43BFB300621E92 /* decision_logic_normal.h */,
> 				5CDD89AA1E43BFB300621E92 /* decoder_database.cc */,
> 				5CDD89AB1E43BFB300621E92 /* decoder_database.h */,
> 				5CDD89AC1E43BFB300621E92 /* defines.h */,
>@@ -9862,7 +10706,8 @@
> 		5CDD8ABB1E43C22200621E92 /* audio */ = {
> 			isa = PBXGroup;
> 			children = (
>-				5CD2854A1E6A62090094FDC8 /* utility */,
>+				4102F69A21273261006AE8D7 /* audio_level.cc */,
>+				4102F69821273260006AE8D7 /* audio_level.h */,
> 				5CDD8ABD1E43C23900621E92 /* audio_receive_stream.cc */,
> 				5CDD8ABE1E43C23900621E92 /* audio_receive_stream.h */,
> 				5CDD8AC01E43C23900621E92 /* audio_send_stream.cc */,
>@@ -9871,11 +10716,20 @@
> 				5CDD8AC41E43C23900621E92 /* audio_state.h */,
> 				41F9BF992051C84B00ABF0B9 /* audio_transport_impl.cc */,
> 				41F9BF9A2051C84B00ABF0B9 /* audio_transport_impl.h */,
>+				4102F69C21273261006AE8D7 /* channel_proxy.cc */,
>+				4102F69921273261006AE8D7 /* channel_proxy.h */,
>+				4102F69E21273262006AE8D7 /* channel.cc */,
>+				4102F69B21273261006AE8D7 /* channel.h */,
> 				5CDD8AC61E43C23900621E92 /* conversion.h */,
> 				419C83B11FE2472D0040C30F /* null_audio_poller.cc */,
> 				419C83B21FE2472E0040C30F /* null_audio_poller.h */,
>+				4102F69721273260006AE8D7 /* remix_resample.cc */,
>+				4102F69D21273261006AE8D7 /* remix_resample.h */,
> 				419C83331FE245E90040C30F /* time_interval.cc */,
> 				419C83321FE245E90040C30F /* time_interval.h */,
>+				4102F69F21273262006AE8D7 /* transport_feedback_packet_loss_tracker.cc */,
>+				4102F69621273260006AE8D7 /* transport_feedback_packet_loss_tracker.h */,
>+				5CD2854A1E6A62090094FDC8 /* utility */,
> 			);
> 			path = audio;
> 			sourceTree = "<group>";
>@@ -10041,6 +10895,9 @@
> 		5CDD8C451E43C57900621E92 /* call */ = {
> 			isa = PBXGroup;
> 			children = (
>+				41924172212738C200634FCF /* callfactoryinterface.h */,
>+				41924174212738C200634FCF /* transport.cc */,
>+				41924173212738C200634FCF /* transport.h */,
> 				5CDD8C491E43C58E00621E92 /* audio_sink.h */,
> 			);
> 			path = call;
>@@ -10385,8 +11242,8 @@
> 		5CDD8F951E43CCA200621E92 /* pacing */ = {
> 			isa = PBXGroup;
> 			children = (
>-				5CDD8F961E43CCBE00621E92 /* alr_detector.cc */,
>-				5CDD8F971E43CCBE00621E92 /* alr_detector.h */,
>+				41A08BD221272EFA001D5D7B /* round_robin_packet_queue.cc */,
>+				41A08BD121272EFA001D5D7B /* round_robin_packet_queue.h */,
> 				5CDD8F981E43CCBE00621E92 /* bitrate_prober.cc */,
> 				5CDD8F991E43CCBE00621E92 /* bitrate_prober.h */,
> 				419C82991FE20CA10040C30F /* interval_budget.cc */,
>@@ -10396,8 +11253,8 @@
> 				419C83281FE245CB0040C30F /* pacer.h */,
> 				419C83261FE245CB0040C30F /* packet_queue.cc */,
> 				419C832A1FE245CC0040C30F /* packet_queue.h */,
>-				419C83271FE245CB0040C30F /* packet_queue2.cc */,
>-				419C83291FE245CC0040C30F /* packet_queue2.h */,
>+				419241842127499300634FCF /* packet_queue_interface.cc */,
>+				419241832127499300634FCF /* packet_queue_interface.h */,
> 				5CDD8F9C1E43CCBE00621E92 /* packet_router.cc */,
> 				5CDD8F9D1E43CCBE00621E92 /* packet_router.h */,
> 			);
>@@ -10407,24 +11264,15 @@
> 		5CDD8FA61E43CCE100621E92 /* congestion_controller */ = {
> 			isa = PBXGroup;
> 			children = (
>-				419C83FF1FE2498F0040C30F /* acknowledged_bitrate_estimator.cc */,
>-				419C84001FE2498F0040C30F /* acknowledged_bitrate_estimator.h */,
>-				419C84031FE249910040C30F /* bitrate_estimator.cc */,
>-				419C84021FE249900040C30F /* bitrate_estimator.h */,
>-				5CDD8FAA1E43CD0700621E92 /* delay_based_bwe.cc */,
>-				5CDD8FAB1E43CD0700621E92 /* delay_based_bwe.h */,
>-				5CD286161E6A66380094FDC8 /* median_slope_estimator.cc */,
>-				5CD286171E6A66380094FDC8 /* median_slope_estimator.h */,
>-				5CDD8FAC1E43CD0700621E92 /* probe_bitrate_estimator.cc */,
>-				5CDD8FAD1E43CD0700621E92 /* probe_bitrate_estimator.h */,
>-				5CDD8FAE1E43CD0700621E92 /* probe_controller.cc */,
>-				5CDD8FAF1E43CD0700621E92 /* probe_controller.h */,
>+				419241B12127591C00634FCF /* bbr */,
>+				41E02CD52127362300C27CD6 /* rtp */,
>+				41E02C8F212734E800C27CD6 /* goog_cc */,
>+				41A08BD621272F1C001D5D7B /* congestion_window_pushback_controller.cc */,
>+				41A08BD521272F1C001D5D7B /* congestion_window_pushback_controller.h */,
> 				413091FD1EF8D06D00757C55 /* receive_side_congestion_controller.cc */,
> 				413092141EF8D5ED00757C55 /* send_side_congestion_controller.cc */,
> 				5CDD8FB01E43CD0700621E92 /* transport_feedback_adapter.cc */,
> 				5CDD8FB11E43CD0700621E92 /* transport_feedback_adapter.h */,
>-				5CD2861A1E6A66380094FDC8 /* trendline_estimator.cc */,
>-				5CD2861B1E6A66380094FDC8 /* trendline_estimator.h */,
> 			);
> 			path = congestion_controller;
> 			sourceTree = "<group>";
>@@ -10447,7 +11295,6 @@
> 				5CDD8FCB1E43CD6600621E92 /* remote_bitrate_estimator_single_stream.h */,
> 				5CDD8FCE1E43CD6600621E92 /* remote_estimator_proxy.cc */,
> 				5CDD8FCF1E43CD6600621E92 /* remote_estimator_proxy.h */,
>-				5CDD8FD01E43CD6600621E92 /* send_time_history.cc */,
> 			);
> 			path = remote_bitrate_estimator;
> 			sourceTree = "<group>";
>@@ -10474,34 +11321,6 @@
> 			name = pcm16b;
> 			sourceTree = "<group>";
> 		};
>-		5CDD90041E43CECE00621E92 /* level_controller */ = {
>-			isa = PBXGroup;
>-			children = (
>-				5CDD90051E43CEDE00621E92 /* biquad_filter.cc */,
>-				5CDD90061E43CEDE00621E92 /* biquad_filter.h */,
>-				5CDD90071E43CEDE00621E92 /* down_sampler.cc */,
>-				5CDD90081E43CEDE00621E92 /* down_sampler.h */,
>-				5CDD90091E43CEDE00621E92 /* gain_applier.cc */,
>-				5CDD900A1E43CEDE00621E92 /* gain_applier.h */,
>-				5CDD900B1E43CEDE00621E92 /* gain_selector.cc */,
>-				5CDD900C1E43CEDE00621E92 /* gain_selector.h */,
>-				5CDD90101E43CEDE00621E92 /* level_controller.cc */,
>-				5CDD90111E43CEDE00621E92 /* level_controller.h */,
>-				5CDD900E1E43CEDE00621E92 /* level_controller_constants.h */,
>-				5CDD90121E43CEDE00621E92 /* noise_level_estimator.cc */,
>-				5CDD90131E43CEDE00621E92 /* noise_level_estimator.h */,
>-				5CDD90141E43CEDE00621E92 /* noise_spectrum_estimator.cc */,
>-				5CDD90151E43CEDE00621E92 /* noise_spectrum_estimator.h */,
>-				5CDD90161E43CEDE00621E92 /* peak_level_estimator.cc */,
>-				5CDD90171E43CEDE00621E92 /* peak_level_estimator.h */,
>-				5CDD90181E43CEDE00621E92 /* saturating_gain_estimator.cc */,
>-				5CDD90191E43CEDE00621E92 /* saturating_gain_estimator.h */,
>-				5CDD901A1E43CEDE00621E92 /* signal_classifier.cc */,
>-				5CDD901B1E43CEDE00621E92 /* signal_classifier.h */,
>-			);
>-			path = level_controller;
>-			sourceTree = "<group>";
>-		};
> 		5CDD90331E43CF2700621E92 /* logging */ = {
> 			isa = PBXGroup;
> 			children = (
>@@ -10513,6 +11332,13 @@
> 		5CDD90341E43CF3700621E92 /* rtc_event_log */ = {
> 			isa = PBXGroup;
> 			children = (
>+				41924178212738FB00634FCF /* rtc_event_log_impl.cc */,
>+				41A08BC221269576001D5D7B /* icelogger.cc */,
>+				41A08BC121269576001D5D7B /* icelogger.h */,
>+				41A08BB921269552001D5D7B /* rtc_event_ice_candidate_pair_config.cc */,
>+				41A08BBB21269552001D5D7B /* rtc_event_ice_candidate_pair_config.h */,
>+				41A08BBC21269553001D5D7B /* rtc_event_ice_candidate_pair.cc */,
>+				41A08BBA21269552001D5D7B /* rtc_event_ice_candidate_pair.h */,
> 				419C83211FE244FE0040C30F /* events */,
> 				419C84201FE24B960040C30F /* output */,
> 				5CDD903E1E43CF6100621E92 /* rtc_event_log.cc */,
>@@ -10547,22 +11373,6 @@
> 			name = bitrate_controller;
> 			sourceTree = "<group>";
> 		};
>-		5CDD90631E43D14B00621E92 /* beamformer */ = {
>-			isa = PBXGroup;
>-			children = (
>-				5CDD90651E43D15C00621E92 /* array_util.cc */,
>-				5CDD90661E43D15C00621E92 /* array_util.h */,
>-				5CDD90681E43D15C00621E92 /* complex_matrix.h */,
>-				5CDD906A1E43D15C00621E92 /* covariance_matrix_generator.cc */,
>-				5CDD906B1E43D15C00621E92 /* covariance_matrix_generator.h */,
>-				5CDD906E1E43D15C00621E92 /* matrix.h */,
>-				5CDD906F1E43D15C00621E92 /* mock_nonlinear_beamformer.h */,
>-				5CDD90721E43D15C00621E92 /* nonlinear_beamformer.cc */,
>-				5CDD90731E43D15C00621E92 /* nonlinear_beamformer.h */,
>-			);
>-			name = beamformer;
>-			sourceTree = "<group>";
>-		};
> 		5CDD90841E43D30300621E92 /* libsrtp */ = {
> 			isa = PBXGroup;
> 			children = (
>@@ -10734,17 +11544,13 @@
> 			isa = PBXHeadersBuildPhase;
> 			buildActionMask = 2147483647;
> 			files = (
>-				5C088C0F1E4AA44400403995 /* audiomonitor.h in Headers */,
> 				5C088C111E4AA44400403995 /* bundlefilter.h in Headers */,
> 				5C088C131E4AA44400403995 /* channel.h in Headers */,
> 				5C088C151E4AA44400403995 /* channelmanager.h in Headers */,
>-				5C088C171E4AA44400403995 /* currentspeakermonitor.h in Headers */,
> 				5C088C191E4AA44400403995 /* externalhmac.h in Headers */,
>-				5C088C1B1E4AA44400403995 /* mediamonitor.h in Headers */,
> 				5C088C1D1E4AA44400403995 /* mediasession.h in Headers */,
> 				5C088C1F1E4AA44400403995 /* rtcpmuxfilter.h in Headers */,
> 				5C088C211E4AA44400403995 /* srtpfilter.h in Headers */,
>-				5C088C221E4AA44400403995 /* voicechannel.h in Headers */,
> 			);
> 			runOnlyForDeploymentPostprocessing = 0;
> 		};
>@@ -10904,10 +11710,11 @@
> 			files = (
> 				5CDD8B6D1E43C2B500621E92 /* abs_quant.h in Headers */,
> 				5CDD8B6B1E43C2B500621E92 /* abs_quant_loop.h in Headers */,
>+				41F2636A21267ADF00274F59 /* video_encoder_software_fallback_wrapper.h in Headers */,
> 				5CDD89F91E43BFB300621E92 /* accelerate.h in Headers */,
>-				419C84081FE249910040C30F /* acknowledged_bitrate_estimator.h in Headers */,
> 				5CDD85041E43B1EA00621E92 /* acm_codec_database.h in Headers */,
> 				5CDD850A1E43B1EA00621E92 /* acm_receiver.h in Headers */,
>+				41E02CD0212735B700C27CD6 /* bitrate_settings.h in Headers */,
> 				5CDD850C1E43B1EA00621E92 /* acm_resampler.h in Headers */,
> 				5C4B48D81E42C1E3002651C8 /* adaptedvideotracksource.h in Headers */,
> 				5CD285A51E6A63430094FDC8 /* adaptive_fir_filter.h in Headers */,
>@@ -10930,22 +11737,28 @@
> 				5CDD87021E43BA7500621E92 /* agc.h in Headers */,
> 				5CDD87001E43BA7500621E92 /* agc_manager_direct.h in Headers */,
> 				5CDD8FD21E43CD6600621E92 /* aimd_rate_control.h in Headers */,
>+				41D6B45421273159008F9353 /* call_config.h in Headers */,
>+				4192417E2127392300634FCF /* timestamp_extrapolator.h in Headers */,
> 				5C0885281E4A99D200403995 /* alloc.h in Headers */,
>-				5CDD8F9F1E43CCBE00621E92 /* alr_detector.h in Headers */,
> 				41F9BFCC2051DDA200ABF0B9 /* alr_experiment.h in Headers */,
> 				5CDD86ED1E43BA6D00621E92 /* analog_agc.h in Headers */,
> 				5CDD87E61E43BDA100621E92 /* apm_data_dumper.h in Headers */,
> 				5CD284C31E6A5FFC0094FDC8 /* apm_helpers.h in Headers */,
> 				5CDD893C1E43BF3A00621E92 /* app.h in Headers */,
>+				4102F6E221273416006AE8D7 /* color_space.h in Headers */,
> 				5CDD87AA1E43BC0500621E92 /* arith_routines.h in Headers */,
> 				41433D011F79B33400387B4D /* arith_routins.h in Headers */,
>-				5CDD90761E43D15C00621E92 /* array_util.h in Headers */,
> 				41F9BF872051C80100ABF0B9 /* array_view.h in Headers */,
> 				413A23E41FE18E0800373E99 /* arraysize.h in Headers */,
>+				415F1FBF2127304C00064CBF /* echo_cancellation_proxy.h in Headers */,
> 				5C63F8D91E416D53002CA531 /* assertions.h in Headers */,
>+				41E02C8E212734B900C27CD6 /* fec_rate_table.h in Headers */,
>+				41E02CAA2127352D00C27CD6 /* goog_cc_network_control.h in Headers */,
>+				41DDB261212679A300296D47 /* opensslutility.h in Headers */,
> 				413A22A81FE18E0600373E99 /* asyncinvoker-inl.h in Headers */,
> 				413A22B01FE18E0700373E99 /* asyncinvoker.h in Headers */,
> 				413A23A81FE18E0800373E99 /* asyncpacketsocket.h in Headers */,
>+				41A08BAD212681C8001D5D7B /* i010_buffer.h in Headers */,
> 				413A22AB1FE18E0600373E99 /* asyncresolverinterface.h in Headers */,
> 				413A235E1FE18E0700373E99 /* asyncsocket.h in Headers */,
> 				5C63F9E11E4174F6002CA531 /* asyncstuntcpsocket.h in Headers */,
>@@ -10958,9 +11771,13 @@
> 				5CDD84161E439B2900621E92 /* audio_decoder.h in Headers */,
> 				5CD285451E6A61D20094FDC8 /* audio_decoder.h in Headers */,
> 				5CD285431E6A61D20094FDC8 /* audio_decoder_factory.h in Headers */,
>+				41DDB25A2126792800296D47 /* zero_memory.h in Headers */,
> 				41F9BF982051C82100ABF0B9 /* audio_decoder_factory_template.h in Headers */,
>+				41A08BB2212681C8001D5D7B /* video_stream_decoder.h in Headers */,
>+				419241852127499300634FCF /* packet_queue_interface.h in Headers */,
> 				419C83E81FE248350040C30F /* audio_decoder_g711.h in Headers */,
> 				4140B8321E4E3396007409E6 /* audio_decoder_g722.h in Headers */,
>+				413AD1A321265B10003F7263 /* algorithm.h in Headers */,
> 				419C833D1FE246230040C30F /* audio_decoder_g722.h in Headers */,
> 				5CDD8B6F1E43C2B500621E92 /* audio_decoder_ilbc.h in Headers */,
> 				419C84BE1FE319440040C30F /* audio_decoder_ilbc.h in Headers */,
>@@ -10970,6 +11787,7 @@
> 				5CDD8C0A1E43C34600621E92 /* audio_decoder_isac_t.h in Headers */,
> 				5CDD8C091E43C34600621E92 /* audio_decoder_isac_t_impl.h in Headers */,
> 				419C83F01FE2488D0040C30F /* audio_decoder_L16.h in Headers */,
>+				416D3BE5212731C200775F09 /* gain_curve_applier.h in Headers */,
> 				5CDD8C611E43C60900621E92 /* audio_decoder_opus.h in Headers */,
> 				419C82F21FE20EB50040C30F /* audio_decoder_opus.h in Headers */,
> 				4140B8231E4E3383007409E6 /* audio_decoder_pcm.h in Headers */,
>@@ -10978,8 +11796,11 @@
> 				5C119FFC1E457400004F0987 /* audio_device_buffer.h in Headers */,
> 				5C119FFD1E457400004F0987 /* audio_device_config.h in Headers */,
> 				4145F6141FE1E16F00EB9CAF /* audio_device_data_observer.h in Headers */,
>+				41F2638221267B7700274F59 /* bitrate_constraints.h in Headers */,
>+				4192419B2127586500634FCF /* kiss_fft.h in Headers */,
> 				4145F6121FE1E16F00EB9CAF /* audio_device_defines.h in Headers */,
> 				5C11A01E1E457578004F0987 /* audio_device_dummy.h in Headers */,
>+				41D6B45521273159008F9353 /* degraded_call.h in Headers */,
> 				5C119FFF1E457400004F0987 /* audio_device_generic.h in Headers */,
> 				5CFD537C1E4BA4F500482908 /* audio_device_ios.h in Headers */,
> 				5C11A00C1E457448004F0987 /* audio_device_mac.h in Headers */,
>@@ -10994,6 +11815,7 @@
> 				419C83401FE246230040C30F /* audio_encoder_g722.h in Headers */,
> 				419C833C1FE246230040C30F /* audio_encoder_g722_config.h in Headers */,
> 				5CDD8B711E43C2B500621E92 /* audio_encoder_ilbc.h in Headers */,
>+				41E02CC12127356A00C27CD6 /* data_rate_limiter.h in Headers */,
> 				419C84C01FE319440040C30F /* audio_encoder_ilbc.h in Headers */,
> 				419C84BC1FE319440040C30F /* audio_encoder_ilbc_config.h in Headers */,
> 				419C83661FE246CA0040C30F /* audio_encoder_isac.h in Headers */,
>@@ -11008,52 +11830,59 @@
> 				4140B8211E4E3383007409E6 /* audio_encoder_pcm.h in Headers */,
> 				5CDD8FFF1E43CE3A00621E92 /* audio_encoder_pcm16b.h in Headers */,
> 				5CD285471E6A61D20094FDC8 /* audio_format.h in Headers */,
>+				415F1FCD212730F000064CBF /* audio_mixer.h in Headers */,
> 				5CDD841A1E439B2900621E92 /* audio_format_conversion.h in Headers */,
> 				5CD284931E6A5F410094FDC8 /* audio_frame_manipulator.h in Headers */,
> 				5CD2854E1E6A62130094FDC8 /* audio_frame_operations.h in Headers */,
>-				413092051EF8D0C900757C55 /* audio_level.h in Headers */,
> 				5CDD8A901E43C00F00621E92 /* audio_loop.h in Headers */,
> 				5CD284961E6A5F410094FDC8 /* audio_mixer_impl.h in Headers */,
> 				5C11A00E1E457448004F0987 /* audio_mixer_manager_mac.h in Headers */,
> 				5CDD8A021E43BFB300621E92 /* audio_multi_vector.h in Headers */,
> 				5CDD8F7A1E43CBE000621E92 /* audio_network_adaptor_impl.h in Headers */,
> 				5CDD8FEA1E43CDCA00621E92 /* audio_processing.h in Headers */,
>+				419241C32127593F00634FCF /* bbr_network_controller.h in Headers */,
> 				5CDD84AC1E43AF1300621E92 /* audio_processing_impl.h in Headers */,
> 				419C829E1FE20D1C0040C30F /* audio_processing_statistics.h in Headers */,
> 				5CDD8ACD1E43C23900621E92 /* audio_receive_stream.h in Headers */,
>+				41299B972127369100B3414B /* fake_network_pipe.h in Headers */,
> 				5CD284B01E6A5F9F0094FDC8 /* audio_receive_stream.h in Headers */,
>+				4102F6CF212733B7006AE8D7 /* video_quality_observer.h in Headers */,
> 				5C4B4C5D1E431F9C002651C8 /* audio_ring_buffer.h in Headers */,
> 				5CDD8AD01E43C23900621E92 /* audio_send_stream.h in Headers */,
> 				5CD284B21E6A5F9F0094FDC8 /* audio_send_stream.h in Headers */,
> 				5CFD53801E4BA4F500482908 /* audio_session_observer.h in Headers */,
> 				5CDD8C4F1E43C58E00621E92 /* audio_sink.h in Headers */,
>+				41D6B45D21273159008F9353 /* rtp_video_sender.h in Headers */,
> 				5CDD8A921E43C00F00621E92 /* audio_sink.h in Headers */,
> 				5CDD8AD31E43C23900621E92 /* audio_state.h in Headers */,
> 				5CD284B31E6A5F9F0094FDC8 /* audio_state.h in Headers */,
>+				419241472127372800634FCF /* spectral_features_internal.h in Headers */,
> 				41F9BF9C2051C84C00ABF0B9 /* audio_transport_impl.h in Headers */,
> 				41F9BFA02051C88500ABF0B9 /* audio_util.h in Headers */,
> 				5CDD8A051E43BFB300621E92 /* audio_vector.h in Headers */,
> 				5C4B48D91E42C1E3002651C8 /* audiosource.h in Headers */,
>+				415F1FCC212730F000064CBF /* echo_control.h in Headers */,
> 				5CD284F11E6A60570094FDC8 /* audiotrack.h in Headers */,
>+				41F2638E21267F4900274F59 /* g711.h in Headers */,
> 				5CDD8B731E43C2B500621E92 /* augmented_cb_corr.h in Headers */,
> 				5C0885291E4A99D200403995 /* auth.h in Headers */,
> 				5C63F8DA1E416D53002CA531 /* autolink.h in Headers */,
>+				41DDB24E21265BD700296D47 /* inlined_vector.h in Headers */,
>+				41F2639921267F5E00274F59 /* pa_ringbuffer.h in Headers */,
>+				41DDB25121265BE900296D47 /* optional.h in Headers */,
> 				5CDD8A081E43BFB300621E92 /* background_noise.h in Headers */,
> 				5CDD87AF1E43BC0500621E92 /* bandwidth_estimator.h in Headers */,
> 				41433D241F79B33400387B4D /* bandwidth_estimator.h in Headers */,
> 				5CDD8C0D1E43C34600621E92 /* bandwidth_info.h in Headers */,
>-				413A22F61FE18E0700373E99 /* base64.h in Headers */,
> 				5C63F9E31E4174F6002CA531 /* basicpacketsocketfactory.h in Headers */,
> 				5C4B4D0A1E4322F6002651C8 /* basicportallocator.h in Headers */,
>-				413A22F11FE18E0700373E99 /* basictypes.h in Headers */,
>+				41F2639821267F5E00274F59 /* pa_memorybarrier.h in Headers */,
> 				413A235C1FE18E0700373E99 /* bind.h in Headers */,
>-				5CDD901D1E43CEDE00621E92 /* biquad_filter.h in Headers */,
> 				413A23771FE18E0700373E99 /* bitbuffer.h in Headers */,
> 				41109AB01E5FA19200C0955A /* bitrate_adjuster.h in Headers */,
> 				5CDD85301E43B39C00621E92 /* bitrate_allocator.h in Headers */,
> 				5CDD8F7E1E43CBE000621E92 /* bitrate_controller.h in Headers */,
> 				5CDD905E1E43D11200621E92 /* bitrate_controller_impl.h in Headers */,
>-				419C840A1FE249910040C30F /* bitrate_estimator.h in Headers */,
> 				5CDD8FA11E43CCBE00621E92 /* bitrate_prober.h in Headers */,
> 				413A23261FE18E0700373E99 /* bitrateallocationstrategy.h in Headers */,
> 				5CD285AD1E6A63430094FDC8 /* block_framer.h in Headers */,
>@@ -11065,9 +11894,7 @@
> 				5CDD8A0B1E43BFB300621E92 /* buffer_level_filter.h in Headers */,
> 				413A232B1FE18E0700373E99 /* bufferqueue.h in Headers */,
> 				5CD285491E6A61D20094FDC8 /* builtin_audio_decoder_factory.h in Headers */,
>-				5CDD841E1E439B2900621E92 /* builtin_audio_decoder_factory.h in Headers */,
> 				413091FC1EF8CFF800757C55 /* builtin_audio_encoder_factory.h in Headers */,
>-				413092281EF8D6E600757C55 /* builtin_audio_encoder_factory.h in Headers */,
> 				5CDD8B751E43C2B500621E92 /* bw_expand.h in Headers */,
> 				5CDD893F1E43BF3A00621E92 /* bye.h in Headers */,
> 				5CDD88761E43BE3C00621E92 /* byte_io.h in Headers */,
>@@ -11078,9 +11905,9 @@
> 				5CDD859D1E43B5C000621E92 /* call_stats.h in Headers */,
> 				413A23171FE18E0700373E99 /* callback.h in Headers */,
> 				4145E4E61EF8CD7D00FCF6E6 /* callfactory.h in Headers */,
>-				4145E4E51EF8CD7D00FCF6E6 /* callfactoryinterface.h in Headers */,
> 				41F9BF852051C80100ABF0B9 /* candidate.h in Headers */,
> 				5C63F9E51E4174F6002CA531 /* candidatepairinterface.h in Headers */,
>+				4102F6A021273262006AE8D7 /* transport_feedback_packet_loss_tracker.h in Headers */,
> 				5CD285B31E6A63430094FDC8 /* cascaded_biquad_filter.h in Headers */,
> 				5CDD8B771E43C2B500621E92 /* cb_construct.h in Headers */,
> 				5CDD8B7D1E43C2B500621E92 /* cb_mem_energy.h in Headers */,
>@@ -11089,52 +11916,59 @@
> 				5CDD8B811E43C2B500621E92 /* cb_search.h in Headers */,
> 				5CDD8B7F1E43C2B500621E92 /* cb_search_core.h in Headers */,
> 				5CDD8B831E43C2B500621E92 /* cb_update_best_index.h in Headers */,
>-				5CDD845E1E43AE2900621E92 /* channel.h in Headers */,
> 				5C4B4C651E431F9C002651C8 /* channel_buffer.h in Headers */,
> 				5CDD8F811E43CBE000621E92 /* channel_controller.h in Headers */,
>-				5CDD845C1E43AE2900621E92 /* channel_proxy.h in Headers */,
> 				5CDD8B851E43C2B500621E92 /* chebyshev.h in Headers */,
> 				413A234F1FE18E0700373E99 /* checks.h in Headers */,
> 				5C08852A1E4A99D200403995 /* cipher.h in Headers */,
>+				41924175212738C200634FCF /* callfactoryinterface.h in Headers */,
> 				5CDD86A21E43B99400621E92 /* circular_buffer.h in Headers */,
>+				41F2639321267F5300274F59 /* g722_enc_dec.h in Headers */,
> 				5C4B48DC1E42C1E3002651C8 /* codec.h in Headers */,
>+				41E02C7E2127347400C27CD6 /* rtp_video_header.h in Headers */,
> 				5CDD87B01E43BC0500621E92 /* codec.h in Headers */,
> 				41433CF31F79B33400387B4D /* codec.h in Headers */,
>-				5CDD83C71E439A6F00621E92 /* codec_database.h in Headers */,
> 				5CDD85161E43B1EA00621E92 /* codec_manager.h in Headers */,
> 				5CDD83C91E439A6F00621E92 /* codec_timer.h in Headers */,
>+				4102F6CA212733B7006AE8D7 /* video_stream_encoder.h in Headers */,
>+				4102F68D21273206006AE8D7 /* rtcstatstraversal.h in Headers */,
> 				5CDD8A0E1E43BFB300621E92 /* comfort_noise.h in Headers */,
>+				41299B922127367B00B3414B /* isac_vad.h in Headers */,
> 				5CD285B51E6A63430094FDC8 /* comfort_noise_generator.h in Headers */,
>-				5C63F9E61E4174F6002CA531 /* common.h in Headers */,
> 				5CDD87271E43BABE00621E92 /* common.h in Headers */,
> 				5CDD84AD1E43AF1300621E92 /* common.h in Headers */,
> 				5CDD8C861E43C66000621E92 /* common.h in Headers */,
> 				5CDD89421E43BF3A00621E92 /* common_header.h in Headers */,
> 				5C6CDD4C1E413598009754E3 /* common_types.h in Headers */,
>+				416D3BE2212731C200775F09 /* down_sampler.h in Headers */,
> 				5CDD8B871E43C2B500621E92 /* comp_corr.h in Headers */,
>+				41F2636121267ADF00274F59 /* builtin_video_encoder_factory.h in Headers */,
> 				413A23851FE18E0700373E99 /* compile_assert_c.h in Headers */,
> 				5CDD86461E43B8B500621E92 /* complex_fft_tables.h in Headers */,
>-				5CDD90781E43D15C00621E92 /* complex_matrix.h in Headers */,
> 				5CDD89451E43BF3A00621E92 /* compound_packet.h in Headers */,
> 				5CDD8FEC1E43CDCA00621E92 /* config.h in Headers */,
> 				5C63F8DB1E416D53002CA531 /* config.h in Headers */,
>+				4192419C2127586500634FCF /* rnn_activations.h in Headers */,
> 				5CDD8A941E43C00F00621E92 /* constant_pcm_packet_source.h in Headers */,
> 				5C4B48781E42C1BA002651C8 /* constants.h in Headers */,
> 				5CDD8B8A1E43C2B500621E92 /* constants.h in Headers */,
> 				413A236C1FE18E0700373E99 /* constructormagic.h in Headers */,
> 				5CDD8F861E43CBE000621E92 /* controller.h in Headers */,
>+				415F1FB621272FBA00064CBF /* vad_with_level.h in Headers */,
> 				5CDD8F841E43CBE000621E92 /* controller_manager.h in Headers */,
> 				5CDD8AD41E43C23900621E92 /* conversion.h in Headers */,
> 				41F9BFB42051C93600ABF0B9 /* convert_legacy_video_factory.h in Headers */,
> 				413A23AC1FE18E0800373E99 /* copyonwritebuffer.h in Headers */,
>-				4145E4AB1EF8933D00FCF6E6 /* corevideo_frame_buffer.h in Headers */,
>-				5CDD907B1E43D15C00621E92 /* covariance_matrix_generator.h in Headers */,
>+				41E02CA42127352D00C27CD6 /* probe_controller.h in Headers */,
> 				413A23E21FE18E0800373E99 /* cpu_time.h in Headers */,
> 				5CDD87B21E43BC0500621E92 /* crc.h in Headers */,
>+				4102F6DD21273416006AE8D7 /* video_sink_interface.h in Headers */,
> 				413A22AC1FE18E0600373E99 /* crc32.h in Headers */,
> 				5CDD8B8C1E43C2B500621E92 /* create_augmented_vec.h in Headers */,
>+				419241CD2127593F00634FCF /* data_transfer_tracker.h in Headers */,
> 				413A23691FE18E0700373E99 /* criticalsection.h in Headers */,
> 				5CDD8A101E43BFB300621E92 /* cross_correlation.h in Headers */,
>+				419241A9212758D300634FCF /* symmetric_matrix_buffer.h in Headers */,
> 				5C08852B1E4A99D200403995 /* crypto_kernel.h in Headers */,
> 				5C08852C1E4A99D200403995 /* crypto_types.h in Headers */,
> 				5C4B48DD1E42C1E3002651C8 /* cryptoparams.h in Headers */,
>@@ -11147,10 +11981,10 @@
> 				5CDD8F881E43CBE000621E92 /* debug_dump_writer.h in Headers */,
> 				419C83F81FE248F10040C30F /* decimator.h in Headers */,
> 				5CDD8A171E43BFB300621E92 /* decision_logic.h in Headers */,
>-				5CDD8A121E43BFB300621E92 /* decision_logic_fax.h in Headers */,
>-				5CDD8A141E43BFB300621E92 /* decision_logic_normal.h in Headers */,
> 				5CDD8B901E43C2B500621E92 /* decode.h in Headers */,
>+				41A08BB821268A7D001D5D7B /* memory.h in Headers */,
> 				5CDD8B8E1E43C2B500621E92 /* decode_residual.h in Headers */,
>+				4102F69421273206006AE8D7 /* transportstats.h in Headers */,
> 				419C849D1FE2FEC00040C30F /* decoder.h in Headers */,
> 				5CDD8A1A1E43BFB300621E92 /* decoder_database.h in Headers */,
> 				5CDD8B921E43C2B500621E92 /* decoder_interpolate_lsf.h in Headers */,
>@@ -11158,10 +11992,10 @@
> 				5CD284981E6A5F410094FDC8 /* default_output_rate_calculator.h in Headers */,
> 				419C843B1FE24E800040C30F /* default_temporal_layers.h in Headers */,
> 				5CD2853A1E6A61980094FDC8 /* default_video_bitrate_allocator.h in Headers */,
>+				41F2637F21267B7700274F59 /* fec_controller.h in Headers */,
> 				5CDD85FB1E43B84E00621E92 /* defines.h in Headers */,
> 				5CDD8A1B1E43BFB300621E92 /* defines.h in Headers */,
> 				5CDD8B931E43C2B500621E92 /* defines.h in Headers */,
>-				5CDD8FB61E43CD0700621E92 /* delay_based_bwe.h in Headers */,
> 				5CDD868E1E43B93900621E92 /* delay_estimator.h in Headers */,
> 				5CDD86891E43B93900621E92 /* delay_estimator_internal.h in Headers */,
> 				5CDD868C1E43B93900621E92 /* delay_estimator_wrapper.h in Headers */,
>@@ -11169,11 +12003,12 @@
> 				5CDD8A211E43BFB300621E92 /* delay_peak_detector.h in Headers */,
> 				413A234E1FE18E0700373E99 /* deprecation.h in Headers */,
> 				5C4B48DE1E42C1E3002651C8 /* device.h in Headers */,
>+				4192418B212749C800634FCF /* send_time_history.h in Headers */,
> 				5CDD86EF1E43BA6D00621E92 /* digital_agc.h in Headers */,
> 				5CDD89481E43BF3A00621E92 /* dlrr.h in Headers */,
> 				5CDD8B951E43C2B500621E92 /* do_plc.h in Headers */,
> 				412455441EF87C0F00F11809 /* dot_product_with_scale.h in Headers */,
>-				5CDD901F1E43CEDE00621E92 /* down_sampler.h in Headers */,
>+				41F2636621267ADF00274F59 /* video_decoder_software_fallback_wrapper.h in Headers */,
> 				4145E4D51EF8CC7600FCF6E6 /* downsampled_render_buffer.h in Headers */,
> 				413A23481FE18E0700373E99 /* dscp.h in Headers */,
> 				5CDD8A241E43BFB300621E92 /* dsp_helper.h in Headers */,
>@@ -11191,8 +12026,10 @@
> 				5CDD84AF1E43AF1300621E92 /* echo_cancellation_impl.h in Headers */,
> 				5CD285B91E6A63430094FDC8 /* echo_canceller3.h in Headers */,
> 				5CDD85EB1E43B81000621E92 /* echo_control_mobile.h in Headers */,
>+				41E02CCF212735B700C27CD6 /* network_control.h in Headers */,
> 				5CDD84B11E43AF1300621E92 /* echo_control_mobile_impl.h in Headers */,
> 				5CD285BB1E6A63430094FDC8 /* echo_path_delay_estimator.h in Headers */,
>+				419241632127385B00634FCF /* rw_lock_wrapper.h in Headers */,
> 				5CD285BD1E6A63430094FDC8 /* echo_path_variability.h in Headers */,
> 				5CD285C11E6A63430094FDC8 /* echo_remover.h in Headers */,
> 				5CD285BF1E6A63430094FDC8 /* echo_remover_metrics.h in Headers */,
>@@ -11208,6 +12045,7 @@
> 				5CDD8B991E43C2B500621E92 /* energy_inverse.h in Headers */,
> 				5CDD8B9B1E43C2B500621E92 /* enh_upsample.h in Headers */,
> 				5CDD8B9F1E43C2B500621E92 /* enhancer.h in Headers */,
>+				4102F6C8212733B7006AE8D7 /* video_send_stream_impl.h in Headers */,
> 				5CDD8B9D1E43C2B500621E92 /* enhancer_interface.h in Headers */,
> 				41433D131F79B33400387B4D /* entropy_coding.h in Headers */,
> 				5CDD87B91E43BC0500621E92 /* entropy_coding.h in Headers */,
>@@ -11216,6 +12054,7 @@
> 				5C08852E1E4A99D200403995 /* err.h in Headers */,
> 				413A23A51FE18E0800373E99 /* event.h in Headers */,
> 				5CD286151E6A66130094FDC8 /* event_log_writer.h in Headers */,
>+				419241C22127593F00634FCF /* windowed_filter.h in Headers */,
> 				5C4B4CCF1E4320A9002651C8 /* event_timer_posix.h in Headers */,
> 				413A237E1FE18E0700373E99 /* event_tracer.h in Headers */,
> 				419C83501FE246650040C30F /* exp_filter.h in Headers */,
>@@ -11223,24 +12062,30 @@
> 				5CDD894B1E43BF3A00621E92 /* extended_jitter_report.h in Headers */,
> 				5CDD894E1E43BF3A00621E92 /* extended_reports.h in Headers */,
> 				5CDD8A981E43C00F00621E92 /* fake_decode_from_file.h in Headers */,
>+				41E02CCE212735B700C27CD6 /* network_types.h in Headers */,
>+				41F2638921267F4000274F59 /* fft.h in Headers */,
>+				415F1FB321272FBA00064CBF /* limiter.h in Headers */,
> 				413A237F1FE18E0700373E99 /* fakeclock.h in Headers */,
>+				41924176212738C200634FCF /* transport.h in Headers */,
> 				5C4B48DF1E42C1E3002651C8 /* fakemediaengine.h in Headers */,
>-				5C63F9401E41737B002CA531 /* fakemetricsobserver.h in Headers */,
> 				413A22B71FE18E0700373E99 /* fakenetwork.h in Headers */,
> 				5C4B48E01E42C1E3002651C8 /* fakenetworkinterface.h in Headers */,
>+				419241D12127597D00634FCF /* simulated_network.h in Headers */,
> 				5C63F9EB1E4174F6002CA531 /* fakeportallocator.h in Headers */,
> 				5C4B48E11E42C1E3002651C8 /* fakertp.h in Headers */,
> 				413A22F81FE18E0700373E99 /* fakesslidentity.h in Headers */,
> 				5C4B48E21E42C1E3002651C8 /* fakevideocapturer.h in Headers */,
> 				5C4B48E31E42C1E3002651C8 /* fakevideorenderer.h in Headers */,
> 				5C63F8DC1E416D53002CA531 /* features.h in Headers */,
>+				41299B8D2127365100B3414B /* transport_feedback_adapter.h in Headers */,
>+				41E02CB32127352D00C27CD6 /* trendline_estimator.h in Headers */,
> 				419C830B1FE20F3C0040C30F /* fec_controller_rplr_based.h in Headers */,
> 				5CDD88791E43BE3C00621E92 /* fec_private_tables_bursty.h in Headers */,
> 				5CDD887A1E43BE3C00621E92 /* fec_private_tables_random.h in Headers */,
> 				5CDD887C1E43BE3C00621E92 /* fec_test_helper.h in Headers */,
> 				5CDD87BB1E43BC0500621E92 /* fft.h in Headers */,
>+				41F2639F2126800000274F59 /* aligned_malloc.h in Headers */,
> 				41433D221F79B33400387B4D /* fft.h in Headers */,
>-				5C4B4C671E431F9C002651C8 /* fft4g.h in Headers */,
> 				41F9BFD32051DDE500ABF0B9 /* fft_buffer.h in Headers */,
> 				5CD285C81E6A63430094FDC8 /* fft_data.h in Headers */,
> 				413A22D71FE18E0700373E99 /* file.h in Headers */,
>@@ -11251,8 +12096,10 @@
> 				413A23BD1FE18E0800373E99 /* fileutils.h in Headers */,
> 				41433CED1F79B33400387B4D /* filterbank_internal.h in Headers */,
> 				41433D0D1F79B33400387B4D /* filterbank_tables.h in Headers */,
>-				5CDD87BE1E43BC0500621E92 /* filterbank_tables.h in Headers */,
> 				5CDD8BA11E43C2B500621E92 /* filtered_cb_vecs.h in Headers */,
>+				419241AE212758D300634FCF /* ring_buffer.h in Headers */,
>+				4192416C2127389B00634FCF /* quality_scaling_experiment.h in Headers */,
>+				419241CC2127593F00634FCF /* bandwidth_sampler.h in Headers */,
> 				5C11A0041E457400004F0987 /* fine_audio_buffer.h in Headers */,
> 				5CDD89511E43BF3A00621E92 /* fir.h in Headers */,
> 				5C4B4C6E1E431F9C002651C8 /* fir_filter.h in Headers */,
>@@ -11266,6 +12113,7 @@
> 				5CD284BA1E6A5F9F0094FDC8 /* flexfec_receive_stream_impl.h in Headers */,
> 				419C83CB1FE247B40040C30F /* flexfec_receiver.h in Headers */,
> 				419C83CE1FE247B40040C30F /* flexfec_sender.h in Headers */,
>+				419241592127376F00634FCF /* fakeicetransport.h in Headers */,
> 				413A22A31FE18E0600373E99 /* format_macros.h in Headers */,
> 				5CDD88891E43BE3C00621E92 /* forward_error_correction.h in Headers */,
> 				5CDD88871E43BE3C00621E92 /* forward_error_correction_internal.h in Headers */,
>@@ -11275,28 +12123,31 @@
> 				5CDD83D21E439A6F00621E92 /* frame_buffer2.h in Headers */,
> 				41109AAF1E5FA19200C0955A /* frame_callback.h in Headers */,
> 				5CDD8BA31E43C2B500621E92 /* frame_classify.h in Headers */,
>+				41A08BAF212681C8001D5D7B /* video_bitrate_allocator.h in Headers */,
> 				5CD2849B1E6A5F410094FDC8 /* frame_combiner.h in Headers */,
>+				419241562127376F00634FCF /* fakecandidatepair.h in Headers */,
> 				5CDD837A1E439A3500621E92 /* frame_dropper.h in Headers */,
>+				419241552127376F00634FCF /* regatheringcontroller.h in Headers */,
> 				5CDD8F911E43CBE000621E92 /* frame_length_controller.h in Headers */,
>+				419241F421275C3200634FCF /* vp8_encoder_simulcast_proxy.h in Headers */,
> 				5CDD83D41E439A6F00621E92 /* frame_object.h in Headers */,
> 				413A22A21FE18E0600373E99 /* function_view.h in Headers */,
>-				4140B8271E4E3383007409E6 /* g711.h in Headers */,
> 				4140B8251E4E3383007409E6 /* g711_interface.h in Headers */,
>-				4140B8361E4E3396007409E6 /* g722_enc_dec.h in Headers */,
>+				416D3BE1212731C200775F09 /* biquad_filter.h in Headers */,
> 				4140B8391E4E3396007409E6 /* g722_interface.h in Headers */,
>-				5CDD90211E43CEDE00621E92 /* gain_applier.h in Headers */,
>+				41E02CBF2127356A00C27CD6 /* weak_ptr.h in Headers */,
> 				5CDD86F01E43BA6D00621E92 /* gain_control.h in Headers */,
> 				5CDD84B31E43AF1300621E92 /* gain_control_for_experimental_agc.h in Headers */,
> 				5CDD84B51E43AF1300621E92 /* gain_control_impl.h in Headers */,
>-				413092201EF8D6A200757C55 /* gain_controller2.h in Headers */,
> 				5CDD8BA51E43C2B500621E92 /* gain_dequant.h in Headers */,
> 				5CDD87031E43BA7500621E92 /* gain_map_internal.h in Headers */,
> 				5CDD8BA71E43C2B500621E92 /* gain_quant.h in Headers */,
>-				5CDD90231E43CEDE00621E92 /* gain_selector.h in Headers */,
> 				5CDD83D61E439A6F00621E92 /* generic_decoder.h in Headers */,
> 				5CDD83D81E439A6F00621E92 /* generic_encoder.h in Headers */,
> 				5CDD8BA91E43C2B500621E92 /* get_cd_vec.h in Headers */,
>+				415F1FB521272FBA00064CBF /* signal_classifier.h in Headers */,
> 				5CDD8BAB1E43C2B500621E92 /* get_lsp_poly.h in Headers */,
>+				4192416D2127389B00634FCF /* congestion_controller_experiment.h in Headers */,
> 				5CDD8BAD1E43C2B500621E92 /* get_sync_seq.h in Headers */,
> 				413A21411FE0F0EF00373E99 /* getopt_s.h in Headers */,
> 				5CDD872A1E43BABE00621E92 /* gmm.h in Headers */,
>@@ -11310,6 +12161,7 @@
> 				4145E4BE1EF894F600FCF6E6 /* h264_profile_level_id.h in Headers */,
> 				5CD285EB1E6A639F0094FDC8 /* h264_sprop_parameter_sets.h in Headers */,
> 				5CDD83DA1E439A6F00621E92 /* h264_sps_pps_tracker.h in Headers */,
>+				4102F6B921273382006AE8D7 /* packet_receiver.h in Headers */,
> 				413A22D81FE18E0700373E99 /* helpers.h in Headers */,
> 				41ECEAD72064147C009D5141 /* helpers.h in Headers */,
> 				5CDD83DC1E439A6F00621E92 /* histogram.h in Headers */,
>@@ -11317,6 +12169,7 @@
> 				5CDD8BAF1E43C2B500621E92 /* hp_input.h in Headers */,
> 				5CDD8BB11E43C2B500621E92 /* hp_output.h in Headers */,
> 				413A23B31FE18E0800373E99 /* httpbase.h in Headers */,
>+				41F2636821267ADF00274F59 /* video_encoder_config.h in Headers */,
> 				413A23B21FE18E0800373E99 /* httpcommon-inl.h in Headers */,
> 				413A23951FE18E0800373E99 /* httpcommon.h in Headers */,
> 				413A22C81FE18E0700373E99 /* httpserver.h in Headers */,
>@@ -11329,6 +12182,7 @@
> 				5CDD8BB41E43C2B500621E92 /* ilbc.h in Headers */,
> 				41109AB11E5FA19200C0955A /* incoming_video_stream.h in Headers */,
> 				5CDD8BB61E43C2B500621E92 /* index_conv_dec.h in Headers */,
>+				41E02C89212734B900C27CD6 /* encoder_database.h in Headers */,
> 				5CDD8BB81E43C2B500621E92 /* index_conv_enc.h in Headers */,
> 				5CDD8BBA1E43C2B500621E92 /* init_decode.h in Headers */,
> 				5CDD8BBC1E43C2B500621E92 /* init_encode.h in Headers */,
>@@ -11344,8 +12198,11 @@
> 				419C83311FE245CD0040C30F /* interval_budget.h in Headers */,
> 				413A22A91FE18E0600373E99 /* ipaddress.h in Headers */,
> 				41433CFF1F79B33400387B4D /* isac_fix_type.h in Headers */,
>+				41DDB266212679A300296D47 /* opensslcertificate.h in Headers */,
>+				419241402127372400634FCF /* lp_residual.h in Headers */,
> 				5CDD87C11E43BC0500621E92 /* isac_float_type.h in Headers */,
> 				5CDD837C1E439A3500621E92 /* ivf_file_writer.h in Headers */,
>+				41A08BB4212681C8001D5D7B /* video_stream_encoder_interface.h in Headers */,
> 				5CDD83E21E439A6F00621E92 /* jitter_buffer.h in Headers */,
> 				5CDD83E01E439A6F00621E92 /* jitter_buffer_common.h in Headers */,
> 				5CDD83E41E439A6F00621E92 /* jitter_estimator.h in Headers */,
>@@ -11359,8 +12216,6 @@
> 				5C0885301E4A99D200403995 /* key.h in Headers */,
> 				5C4B4C711E431F9C002651C8 /* lapped_transform.h in Headers */,
> 				5CDD84201E439B2900621E92 /* legacy_encoded_audio_frame.h in Headers */,
>-				5CDD90281E43CEDE00621E92 /* level_controller.h in Headers */,
>-				5CDD90251E43CEDE00621E92 /* level_controller_constants.h in Headers */,
> 				5CDD84B91E43AF1300621E92 /* level_estimator_impl.h in Headers */,
> 				5CD284F71E6A60570094FDC8 /* localaudiosource.h in Headers */,
> 				413A23EC1FE18E0800373E99 /* location.h in Headers */,
>@@ -11373,6 +12228,8 @@
> 				5CDD8BC21E43C2B500621E92 /* lpc_encode.h in Headers */,
> 				5CDD87C81E43BC0500621E92 /* lpc_gain_swb_tables.h in Headers */,
> 				41433D181F79B33400387B4D /* lpc_masking_model.h in Headers */,
>+				41F263B1212680A800274F59 /* asm_defines.h in Headers */,
>+				41E02CA72127352D00C27CD6 /* probe_bitrate_estimator.h in Headers */,
> 				5CDD87CA1E43BC0500621E92 /* lpc_shape_swb12_tables.h in Headers */,
> 				5CDD87CC1E43BC0500621E92 /* lpc_shape_swb16_tables.h in Headers */,
> 				41433D281F79B33400387B4D /* lpc_tables.h in Headers */,
>@@ -11381,6 +12238,7 @@
> 				5CDD8BC61E43C2B500621E92 /* lsf_interpolate_to_poly_dec.h in Headers */,
> 				5CDD8BC81E43C2B500621E92 /* lsf_interpolate_to_poly_enc.h in Headers */,
> 				5CDD8BCA1E43C2B500621E92 /* lsf_to_lsp.h in Headers */,
>+				416D3BDF212731C200775F09 /* agc2_common.h in Headers */,
> 				5CDD8BCC1E43C2B500621E92 /* lsf_to_poly.h in Headers */,
> 				5CDD8BCE1E43C2B500621E92 /* lsp_to_lsf.h in Headers */,
> 				413A23971FE18E0800373E99 /* macutils.h in Headers */,
>@@ -11388,26 +12246,19 @@
> 				5CD285D01E6A63430094FDC8 /* matched_filter.h in Headers */,
> 				5CD285CE1E6A63430094FDC8 /* matched_filter_lag_aggregator.h in Headers */,
> 				419C83591FE246650040C30F /* mathutils.h in Headers */,
>-				5CDD907E1E43D15C00621E92 /* matrix.h in Headers */,
> 				41F9BFD62051DDE500ABF0B9 /* matrix_buffer.h in Headers */,
>-				413A23911FE18E0800373E99 /* md5.h in Headers */,
>-				413A23A01FE18E0800373E99 /* md5digest.h in Headers */,
> 				5CDD86A51E43B99400621E92 /* mean_variance_estimator.h in Headers */,
>-				5CDD876C1E43BB3000621E92 /* media_file.h in Headers */,
>-				5CDD87671E43BB3000621E92 /* media_file_defines.h in Headers */,
>-				5CDD87691E43BB3000621E92 /* media_file_impl.h in Headers */,
>-				5CDD876B1E43BB3000621E92 /* media_file_utility.h in Headers */,
> 				5CDD83E61E439A6F00621E92 /* media_opt_util.h in Headers */,
> 				5CDD83E81E439A6F00621E92 /* media_optimization.h in Headers */,
> 				5C4B48E51E42C1E3002651C8 /* mediachannel.h in Headers */,
> 				5C4B48E71E42C1E3002651C8 /* mediaconstants.h in Headers */,
> 				5C63F9491E41737B002CA531 /* mediaconstraintsinterface.h in Headers */,
> 				5C4B48E91E42C1E3002651C8 /* mediaengine.h in Headers */,
>-				5CD2861D1E6A66380094FDC8 /* median_slope_estimator.h in Headers */,
> 				5CD284FB1E6A60570094FDC8 /* mediastream.h in Headers */,
> 				5C63F94E1E41737B002CA531 /* mediastreaminterface.h in Headers */,
> 				5CD284FD1E6A60570094FDC8 /* mediastreamobserver.h in Headers */,
> 				5C63F9511E41737B002CA531 /* mediastreamproxy.h in Headers */,
>+				41A08BAC212681C8001D5D7B /* encoded_frame.h in Headers */,
> 				5CD284FE1E6A60570094FDC8 /* mediastreamtrack.h in Headers */,
> 				5C63F9531E41737B002CA531 /* mediastreamtrackproxy.h in Headers */,
> 				5CD286331E6A67BF0094FDC8 /* mediatypes.h in Headers */,
>@@ -11417,10 +12268,11 @@
> 				413A22B91FE18E0700373E99 /* messagehandler.h in Headers */,
> 				413A23A91FE18E0800373E99 /* messagequeue.h in Headers */,
> 				5CDD87071E43BA7500621E92 /* mock_agc.h in Headers */,
>+				4192413C2127372400634FCF /* pitch_search.h in Headers */,
> 				5C11A0051E457400004F0987 /* mock_audio_device_buffer.h in Headers */,
> 				5CDD8FED1E43CDCA00621E92 /* mock_audio_processing.h in Headers */,
>-				5CDD907F1E43D15C00621E92 /* mock_nonlinear_beamformer.h in Headers */,
> 				419C83561FE246650040C30F /* mod_ops.h in Headers */,
>+				41A08BCF21272EE2001D5D7B /* gain_controller2.h in Headers */,
> 				5CDD837E1E439A3500621E92 /* moving_average.h in Headers */,
> 				5CD286111E6A64C90094FDC8 /* moving_max.h in Headers */,
> 				419C83551FE246650040C30F /* moving_max_counter.h in Headers */,
>@@ -11433,6 +12285,7 @@
> 				5CDD8A331E43BFB300621E92 /* nack_tracker.h in Headers */,
> 				4145E4B71EF8943E00FCF6E6 /* nalu_rewriter.h in Headers */,
> 				413A22C41FE18E0700373E99 /* natserver.h in Headers */,
>+				41E02CAD2127352D00C27CD6 /* alr_detector.h in Headers */,
> 				413A22D91FE18E0700373E99 /* natsocketfactory.h in Headers */,
> 				413A22E71FE18E0700373E99 /* nattypes.h in Headers */,
> 				5CDD8BD21E43C2B500621E92 /* nearest_neighbor.h in Headers */,
>@@ -11446,28 +12299,25 @@
> 				413A23761FE18E0700373E99 /* network_constants.h in Headers */,
> 				413A237C1FE18E0700373E99 /* networkmonitor.h in Headers */,
> 				413A22C21FE18E0700373E99 /* networkroute.h in Headers */,
>+				419241C02127593F00634FCF /* loss_rate_filter.h in Headers */,
> 				5CDD872B1E43BABE00621E92 /* noise_gmm_tables.h in Headers */,
>-				5CDD902A1E43CEDE00621E92 /* noise_level_estimator.h in Headers */,
>-				5CDD902C1E43CEDE00621E92 /* noise_spectrum_estimator.h in Headers */,
> 				5CDD85FF1E43B84E00621E92 /* noise_suppression.h in Headers */,
> 				5CDD84EA1E43B0B600621E92 /* noise_suppression_impl.h in Headers */,
> 				5CDD85FD1E43B84E00621E92 /* noise_suppression_x.h in Headers */,
>-				5CDD90831E43D15C00621E92 /* nonlinear_beamformer.h in Headers */,
> 				5CDD8A3E1E43BFB300621E92 /* normal.h in Headers */,
> 				5CDD86A81E43B99400621E92 /* normalized_covariance_estimator.h in Headers */,
> 				5C63F9541E41737B002CA531 /* notifier.h in Headers */,
> 				5CDD86011E43B84E00621E92 /* ns_core.h in Headers */,
> 				41ECEAC120640F28009D5141 /* NSString+StdString.h in Headers */,
>+				41F263B82126810C00274F59 /* base64.h in Headers */,
> 				5CDD86061E43B84E00621E92 /* nsx_core.h in Headers */,
>+				41A08BCB2126961F001D5D7B /* raw_logging.h in Headers */,
> 				5CDD86071E43B84E00621E92 /* nsx_defines.h in Headers */,
> 				419C83B41FE2472E0040C30F /* null_audio_poller.h in Headers */,
> 				5C0885311E4A99D200403995 /* null_auth.h in Headers */,
> 				5C0885321E4A99D200403995 /* null_cipher.h in Headers */,
> 				413A23C01FE18E0800373E99 /* nullsocketserver.h in Headers */,
> 				5CD285331E6A61110094FDC8 /* nullwebrtcvideoengine.h in Headers */,
>-				41ECEAD420641458009D5141 /* objc_frame_buffer.h in Headers */,
>-				413A23FC1FE1987F00373E99 /* objc_video_decoder_factory.h in Headers */,
>-				413A23FD1FE1987F00373E99 /* objc_video_encoder_factory.h in Headers */,
> 				413A23361FE18E0700373E99 /* onetimeevent.h in Headers */,
> 				5CDD86951E43B93900621E92 /* ooura_fft.h in Headers */,
> 				5CDD86921E43B93900621E92 /* ooura_fft_tables_common.h in Headers */,
>@@ -11477,26 +12327,24 @@
> 				5CDD87CF1E43BC0500621E92 /* os_specific_inline.h in Headers */,
> 				5CDD8AAA1E43C00F00621E92 /* output_audio_file.h in Headers */,
> 				5CD2849C1E6A5F410094FDC8 /* output_rate_calculator.h in Headers */,
>-				5CD285D21E6A63430094FDC8 /* output_selector.h in Headers */,
> 				5CDD8AAB1E43C00F00621E92 /* output_wav_file.h in Headers */,
> 				5CDD8FD81E43CD6600621E92 /* overuse_detector.h in Headers */,
> 				5CDD8FDA1E43CD6600621E92 /* overuse_estimator.h in Headers */,
> 				5CDD85A51E43B5C000621E92 /* overuse_frame_detector.h in Headers */,
> 				5C63F9EE1E4174F6002CA531 /* p2pconstants.h in Headers */,
> 				5C63F9F31E4174F6002CA531 /* p2ptransportchannel.h in Headers */,
>-				5C11A0131E4574C5004F0987 /* pa_memorybarrier.h in Headers */,
>-				5C11A0151E4574C5004F0987 /* pa_ringbuffer.h in Headers */,
>+				415F1FB721272FBA00064CBF /* vector_float_frame.h in Headers */,
> 				5CDD8FA31E43CCBE00621E92 /* paced_sender.h in Headers */,
> 				419C832E1FE245CD0040C30F /* pacer.h in Headers */,
> 				5CDD8BD41E43C2B500621E92 /* pack_bits.h in Headers */,
> 				5CDD83EF1E439A6F00621E92 /* packet.h in Headers */,
> 				5CDD8AB01E43C00F00621E92 /* packet.h in Headers */,
> 				5CDD8A431E43BFB300621E92 /* packet.h in Headers */,
>+				415F1FAE21272FBA00064CBF /* fixed_digital_level_estimator.h in Headers */,
> 				5CDD8A411E43BFB300621E92 /* packet_buffer.h in Headers */,
> 				5CDD83ED1E439A6F00621E92 /* packet_buffer.h in Headers */,
> 				5CDD888D1E43BE3C00621E92 /* packet_loss_stats.h in Headers */,
> 				419C83301FE245CD0040C30F /* packet_queue.h in Headers */,
>-				419C832F1FE245CD0040C30F /* packet_queue2.h in Headers */,
> 				5CDD8FA51E43CCBE00621E92 /* packet_router.h in Headers */,
> 				5CDD8AAD1E43C00F00621E92 /* packet_source.h in Headers */,
> 				412455571EF887FB00F11809 /* packetlossestimator.h in Headers */,
>@@ -11504,10 +12352,9 @@
> 				5C63F9F51E4174F6002CA531 /* packettransportinterface.h in Headers */,
> 				419C83D91FE247F20040C30F /* packettransportinternal.h in Headers */,
> 				413A23901FE18E0800373E99 /* pathutils.h in Headers */,
>-				5CDD85A81E43B5C000621E92 /* payload_router.h in Headers */,
> 				5C4B48841E42C1BA002651C8 /* payload_type_mapper.h in Headers */,
>+				41D6B45B21273159008F9353 /* rtp_video_sender_interface.h in Headers */,
> 				5CDD90011E43CE3A00621E92 /* pcm16b.h in Headers */,
>-				5CDD902E1E43CEDE00621E92 /* peak_level_estimator.h in Headers */,
> 				5CD285001E6A60570094FDC8 /* peerconnection.h in Headers */,
> 				5CD284801E6A5E740094FDC8 /* peerconnectionfactory.h in Headers */,
> 				5C63F9591E41737B002CA531 /* peerconnectionfactoryproxy.h in Headers */,
>@@ -11517,7 +12364,9 @@
> 				419C834E1FE246650040C30F /* percentile_filter.h in Headers */,
> 				413A23EA1FE18E0800373E99 /* physicalsocketserver.h in Headers */,
> 				5CDD872E1E43BABE00621E92 /* pitch_based_vad.h in Headers */,
>+				415F1FB021272FBA00064CBF /* gain_applier.h in Headers */,
> 				41433CF11F79B33400387B4D /* pitch_estimator.h in Headers */,
>+				419241ED21275AFA00634FCF /* simulcast_utility.h in Headers */,
> 				5CDD87D11E43BC0500621E92 /* pitch_estimator.h in Headers */,
> 				5CDD87D41E43BC0600621E92 /* pitch_gain_tables.h in Headers */,
> 				41433D231F79B33400387B4D /* pitch_gain_tables.h in Headers */,
>@@ -11533,40 +12382,45 @@
> 				5CDD8BD61E43C2B500621E92 /* poly_to_lsf.h in Headers */,
> 				5CDD8BD81E43C2B500621E92 /* poly_to_lsp.h in Headers */,
> 				5C63F9F81E4174F6002CA531 /* port.h in Headers */,
>+				415F1FBD2127304C00064CBF /* echo_control_mobile_proxy.h in Headers */,
>+				419241C62127593F00634FCF /* bbr_factory.h in Headers */,
> 				5C63F9FB1E4174F6002CA531 /* portallocator.h in Headers */,
> 				5C63F9FC1E4174F6002CA531 /* portinterface.h in Headers */,
>+				4102F6E421273416006AE8D7 /* video_bitrate_allocator_factory.h in Headers */,
> 				5CDD8A461E43BFB300621E92 /* post_decode_vad.h in Headers */,
> 				5CDD835A1E43257200621E92 /* pps_parser.h in Headers */,
>+				415F1FB421272FBA00064CBF /* noise_spectrum_estimator.h in Headers */,
> 				5CDD8A481E43BFB300621E92 /* preemptive_expand.h in Headers */,
>-				5CDD8FB81E43CD0700621E92 /* probe_bitrate_estimator.h in Headers */,
>-				5CDD8FBA1E43CD0700621E92 /* probe_controller.h in Headers */,
> 				5CDD84E61E43B04D00621E92 /* process_thread_impl.h in Headers */,
> 				5CDD835D1E43257200621E92 /* profile_level_id.h in Headers */,
>-				5CDD83F31E439A6F00621E92 /* protection_bitrate_calculator.h in Headers */,
> 				413A23C61FE18E0800373E99 /* protobuf_utils.h in Headers */,
> 				5C63F95D1E41737B002CA531 /* proxy.h in Headers */,
> 				413A231F1FE18E0700373E99 /* proxyinfo.h in Headers */,
> 				413A23C51FE18E0800373E99 /* proxyserver.h in Headers */,
> 				5C63F9FF1E4174F6002CA531 /* pseudotcp.h in Headers */,
>+				41D6B45A21273159008F9353 /* rtp_payload_params.h in Headers */,
> 				5CDD89591E43BF3A00621E92 /* psfb.h in Headers */,
>-				413A230A1FE18E0700373E99 /* ptr_util.h in Headers */,
> 				5CDD87561E43BAF500621E92 /* push_sinc_resampler.h in Headers */,
> 				4130920A1EF8D11200757C55 /* qp_parser.h in Headers */,
>+				4102F68E21273206006AE8D7 /* jseptransportcontroller.h in Headers */,
> 				5CDD83821E439A3500621E92 /* quality_scaler.h in Headers */,
> 				5CD2862B1E6A66C80094FDC8 /* quality_threshold.h in Headers */,
> 				413A23EF1FE18E0800373E99 /* race_checker.h in Headers */,
> 				413A234C1FE18E0700373E99 /* random.h in Headers */,
>+				41E02C7B2127347400C27CD6 /* rtp_generic_frame_descriptor.h in Headers */,
> 				5CDD8A4B1E43BFB300621E92 /* random_vector.h in Headers */,
> 				5CDD895C1E43BF3A00621E92 /* rapid_resync_request.h in Headers */,
>+				41E02C8C212734B900C27CD6 /* fec_controller_default.h in Headers */,
> 				413A23601FE18E0700373E99 /* rate_limiter.h in Headers */,
> 				413A230D1FE18E0700373E99 /* rate_statistics.h in Headers */,
>-				413A23941FE18E0800373E99 /* ratelimiter.h in Headers */,
> 				413A23041FE18E0700373E99 /* ratetracker.h in Headers */,
> 				5C0885331E4A99D200403995 /* rdb.h in Headers */,
> 				5C0885341E4A99D200403995 /* rdbx.h in Headers */,
>+				4102F69521273206006AE8D7 /* peerconnectioninternal.h in Headers */,
> 				5C63F8DF1E416D53002CA531 /* reader.h in Headers */,
> 				5C4B4C781E431F9C002651C8 /* real_fourier.h in Headers */,
> 				5C4B4C731E431F9C002651C8 /* real_fourier_ooura.h in Headers */,
>+				419241C82127593F00634FCF /* packet_number_indexed_queue.h in Headers */,
> 				419C83CA1FE247B40040C30F /* receive_statistics.h in Headers */,
> 				5CDD88921E43BE3C00621E92 /* receive_statistics_impl.h in Headers */,
> 				5CDD85AB1E43B5C000621E92 /* receive_statistics_proxy.h in Headers */,
>@@ -11578,21 +12432,29 @@
> 				413A23671FE18E0700373E99 /* refcountedobject.h in Headers */,
> 				413A23201FE18E0700373E99 /* refcounter.h in Headers */,
> 				5CDD8BDA1E43C2B500621E92 /* refiner.h in Headers */,
>+				41F263AF212680A800274F59 /* inline.h in Headers */,
> 				5C63FA021E4174F6002CA531 /* relayport.h in Headers */,
> 				5C63FA051E4174F6002CA531 /* relayserver.h in Headers */,
>+				4102F69121273206006AE8D7 /* rtpparametersconversion.h in Headers */,
> 				5CDD89621E43BF3A00621E92 /* remb.h in Headers */,
>+				4192419F2127588300634FCF /* audio_codec_pair_id.h in Headers */,
> 				5CDD8FDC1E43CD6600621E92 /* remote_bitrate_estimator_abs_send_time.h in Headers */,
> 				5CDD8FDE1E43CD6600621E92 /* remote_bitrate_estimator_single_stream.h in Headers */,
>+				41F2636F21267B3E00274F59 /* fft4g.h in Headers */,
> 				5CDD8FE21E43CD6600621E92 /* remote_estimator_proxy.h in Headers */,
> 				419C83C51FE247B40040C30F /* remote_ntp_time_estimator.h in Headers */,
> 				5CD285061E6A60570094FDC8 /* remoteaudiosource.h in Headers */,
> 				4145E4DA1EF8CC9B00FCF6E6 /* render_buffer.h in Headers */,
> 				5CD285D61E6A63430094FDC8 /* render_delay_buffer.h in Headers */,
>+				4192416F2127389B00634FCF /* field_trial_parser.h in Headers */,
> 				5CD285DA1E6A63430094FDC8 /* render_delay_controller.h in Headers */,
> 				5CD285D81E6A63430094FDC8 /* render_delay_controller_metrics.h in Headers */,
>+				4192415B2127376F00634FCF /* basicasyncresolverfactory.h in Headers */,
>+				41A08BC7212695DE001D5D7B /* bad_optional_access.h in Headers */,
> 				5CDD84BC1E43AF1300621E92 /* render_queue_item_verifier.h in Headers */,
> 				5CD285DC1E6A63430094FDC8 /* render_signal_analyzer.h in Headers */,
> 				5CDD85191E43B1EA00621E92 /* rent_a_codec.h in Headers */,
>+				41F263BC2126811500274F59 /* sigslot.h in Headers */,
> 				5CDD89651E43BF3A00621E92 /* report_block.h in Headers */,
> 				5CDD85AF1E43B5C000621E92 /* report_block_stats.h in Headers */,
> 				5CDD86651E43B8B500621E92 /* resample_by_2_internal.h in Headers */,
>@@ -11601,20 +12463,26 @@
> 				5CD285DE1E6A63430094FDC8 /* residual_echo_estimator.h in Headers */,
> 				5C4B4C7B1E431F9C002651C8 /* ring_buffer.h in Headers */,
> 				5CDD84C01E43AF1300621E92 /* rms_level.h in Headers */,
>+				41A08BAE212681C8001D5D7B /* video_bitrate_allocation.h in Headers */,
> 				413A23A71FE18E0800373E99 /* rollingaccumulator.h in Headers */,
> 				5CDD896B1E43BF3A00621E92 /* rrtr.h in Headers */,
>+				4102F6A521273262006AE8D7 /* channel.h in Headers */,
> 				419C83AC1FE247080040C30F /* rtc_event.h in Headers */,
> 				41F9BFA92051C8E200ABF0B9 /* rtc_event_alr_state.h in Headers */,
> 				419C839D1FE246F90040C30F /* rtc_event_audio_network_adaptation.h in Headers */,
>+				41F2638121267B7700274F59 /* audio_options.h in Headers */,
> 				419C83981FE246F90040C30F /* rtc_event_audio_playout.h in Headers */,
> 				419C83A11FE246F90040C30F /* rtc_event_audio_receive_stream_config.h in Headers */,
> 				419C838E1FE246F90040C30F /* rtc_event_audio_send_stream_config.h in Headers */,
> 				419C83931FE246F90040C30F /* rtc_event_bwe_update_delay_based.h in Headers */,
> 				419C83971FE246F90040C30F /* rtc_event_bwe_update_loss_based.h in Headers */,
> 				5CDD904C1E43CF6100621E92 /* rtc_event_log.h in Headers */,
>+				4102F6A721273262006AE8D7 /* remix_resample.h in Headers */,
>+				41E02C87212734B900C27CD6 /* decoder_database.h in Headers */,
> 				41F411BB1EF8DA9600343C26 /* rtc_event_log_factory.h in Headers */,
> 				41F411BA1EF8DA9600343C26 /* rtc_event_log_factory_interface.h in Headers */,
> 				419C84231FE24BA60040C30F /* rtc_event_log_output_file.h in Headers */,
>+				415F1FCF212730F000064CBF /* audio_frame.h in Headers */,
> 				419C839F1FE246F90040C30F /* rtc_event_probe_cluster_created.h in Headers */,
> 				419C83A01FE246F90040C30F /* rtc_event_probe_result_failure.h in Headers */,
> 				419C839A1FE246F90040C30F /* rtc_event_probe_result_success.h in Headers */,
>@@ -11629,7 +12497,6 @@
> 				413A24581FE1991A00373E99 /* RTCAudioSessionConfiguration.h in Headers */,
> 				413A244B1FE1991A00373E99 /* RTCAudioSource.h in Headers */,
> 				413A24571FE1991A00373E99 /* RTCAudioTrack.h in Headers */,
>-				413A246D1FE1991A00373E99 /* RTCAVFoundationVideoSource.h in Headers */,
> 				413A245B1FE1991A00373E99 /* RTCCameraPreviewView.h in Headers */,
> 				413A244D1FE1991A00373E99 /* RTCCameraVideoCapturer.h in Headers */,
> 				413A23231FE18E0700373E99 /* rtccertificate.h in Headers */,
>@@ -11640,6 +12507,7 @@
> 				413A24521FE1991A00373E99 /* RTCDispatcher.h in Headers */,
> 				413A245D1FE1991A00373E99 /* RTCEAGLVideoView.h in Headers */,
> 				41F411B11EF8DA0200343C26 /* rtcerror.h in Headers */,
>+				41DDB24D21265BD700296D47 /* fixed_array.h in Headers */,
> 				41F9BF882051C80100ABF0B9 /* rtceventlogoutput.h in Headers */,
> 				413A24401FE1991A00373E99 /* RTCFieldTrials.h in Headers */,
> 				413A245A1FE1991A00373E99 /* RTCFileLogger.h in Headers */,
>@@ -11653,21 +12521,32 @@
> 				413A24491FE1991A00373E99 /* RTCMacros.h in Headers */,
> 				413A24681FE1991A00373E99 /* RTCMediaConstraints.h in Headers */,
> 				413A24511FE1991A00373E99 /* RTCMediaSource.h in Headers */,
>+				419241452127372800634FCF /* pitch_search_internal.h in Headers */,
> 				413A24541FE1991A00373E99 /* RTCMediaStream.h in Headers */,
>+				415F1FA921272FBA00064CBF /* adaptive_mode_level_estimator.h in Headers */,
> 				413A24471FE1991A00373E99 /* RTCMediaStreamTrack.h in Headers */,
> 				413A24641FE1991A00373E99 /* RTCMetrics.h in Headers */,
> 				413A246A1FE1991A00373E99 /* RTCMetricsSampleInfo.h in Headers */,
> 				413A24671FE1991A00373E99 /* RTCMTLNSVideoView.h in Headers */,
> 				413A24711FE1991A00373E99 /* RTCMTLVideoView.h in Headers */,
>+				4102F6A321273262006AE8D7 /* channel_proxy.h in Headers */,
> 				413A24591FE1991A00373E99 /* RTCNSGLVideoView.h in Headers */,
>+				41E02C792127347400C27CD6 /* rtp_generic_frame_descriptor_extension.h in Headers */,
>+				41F263B2212680A800274F59 /* file_wrapper.h in Headers */,
> 				5CDD8A501E43BFB300621E92 /* rtcp.h in Headers */,
> 				419C82B21FE20DCD0040C30F /* rtcp_demuxer.h in Headers */,
>+				41F2637221267B4A00274F59 /* spl_sqrt_floor.h in Headers */,
> 				5CD285FB1E6A64520094FDC8 /* rtcp_nack_stats.h in Headers */,
> 				5CDD88981E43BE3C00621E92 /* rtcp_packet.h in Headers */,
> 				419C82BD1FE20DCD0040C30F /* rtcp_packet_sink_interface.h in Headers */,
> 				5CDD889B1E43BE3C00621E92 /* rtcp_receiver.h in Headers */,
>+				41F263A02126800000274F59 /* aligned_array.h in Headers */,
>+				4192418D212749C800634FCF /* pacer_controller.h in Headers */,
> 				5CDD889E1E43BE3C00621E92 /* rtcp_sender.h in Headers */,
>+				416D3BEA212731C200775F09 /* saturation_protector.h in Headers */,
>+				41DDB26C212679D200296D47 /* string_builder.h in Headers */,
> 				419C83031FE20F020040C30F /* rtcp_transceiver.h in Headers */,
>+				41A08BD321272EFA001D5D7B /* round_robin_packet_queue.h in Headers */,
> 				419C83011FE20F020040C30F /* rtcp_transceiver_config.h in Headers */,
> 				419C83051FE20F020040C30F /* rtcp_transceiver_impl.h in Headers */,
> 				413A243E1FE1991A00373E99 /* RTCPeerConnection.h in Headers */,
>@@ -11679,8 +12558,11 @@
> 				413A245C1FE1991A00373E99 /* RTCRtpSender.h in Headers */,
> 				413A244A1FE1991A00373E99 /* RTCSessionDescription.h in Headers */,
> 				413A24661FE1991A00373E99 /* RTCSSLAdapter.h in Headers */,
>+				41E02CAF2127352D00C27CD6 /* median_slope_estimator.h in Headers */,
> 				4145E4C81EF896D700FCF6E6 /* rtcstats.h in Headers */,
> 				4145E4C71EF896D700FCF6E6 /* rtcstats_objects.h in Headers */,
>+				419241E721275A7600634FCF /* field_trial_units.h in Headers */,
>+				41A08BC321269577001D5D7B /* icelogger.h in Headers */,
> 				5CD285081E6A60570094FDC8 /* rtcstatscollector.h in Headers */,
> 				4145E4C51EF896D700FCF6E6 /* rtcstatscollectorcallback.h in Headers */,
> 				4145E4C61EF896D700FCF6E6 /* rtcstatsreport.h in Headers */,
>@@ -11695,23 +12577,26 @@
> 				413A24601FE1991A00373E99 /* RTCVideoDecoderVP9.h in Headers */,
> 				413A246E1FE1991A00373E99 /* RTCVideoEncoderVP8.h in Headers */,
> 				413A24461FE1991A00373E99 /* RTCVideoEncoderVP9.h in Headers */,
>-				41ECEACC2064134F009D5141 /* RTCVideoFrame+Private.h in Headers */,
>+				41E02CA62127352D00C27CD6 /* acknowledged_bitrate_estimator.h in Headers */,
> 				413A244F1FE1991A00373E99 /* RTCVideoFrame.h in Headers */,
> 				413A24411FE1991A00373E99 /* RTCVideoFrameBuffer.h in Headers */,
> 				413A246F1FE1991A00373E99 /* RTCVideoRenderer.h in Headers */,
>+				41D6B45821273159008F9353 /* receive_time_calculator.h in Headers */,
> 				413A24611FE1991A00373E99 /* RTCVideoSource.h in Headers */,
> 				413A24501FE1991A00373E99 /* RTCVideoTrack.h in Headers */,
> 				413A243A1FE1991A00373E99 /* RTCVideoViewShading.h in Headers */,
> 				41ECEAE3206416FF009D5141 /* RTCWrappedNativeVideoDecoder.h in Headers */,
> 				41ECEAE1206416FF009D5141 /* RTCWrappedNativeVideoEncoder.h in Headers */,
>+				419241982127586500634FCF /* rnn_vad_weights.h in Headers */,
> 				419C82B81FE20DCD0040C30F /* rtp_config.h in Headers */,
> 				419C83C91FE247B40040C30F /* rtp_cvo.h in Headers */,
> 				413092011EF8D0A600757C55 /* rtp_demuxer.h in Headers */,
> 				5CDD8AB71E43C00F00621E92 /* rtp_file_source.h in Headers */,
>+				41DDB26D212679D200296D47 /* audio_format_to_string.h in Headers */,
>+				41F263AD212680A800274F59 /* ignore_warnings.h in Headers */,
> 				5CDD88B11E43BE3C00621E92 /* rtp_format.h in Headers */,
> 				5CDD88A51E43BE3C00621E92 /* rtp_format_h264.h in Headers */,
> 				5CDD88A71E43BE3C00621E92 /* rtp_format_video_generic.h in Headers */,
>-				419C83041FE20F020040C30F /* rtp_format_video_stereo.h in Headers */,
> 				5CDD88AC1E43BE3C00621E92 /* rtp_format_vp8.h in Headers */,
> 				5CDD88AF1E43BE3C00621E92 /* rtp_format_vp9.h in Headers */,
> 				5CDD83F71E439A6F00621E92 /* rtp_frame_reference_finder.h in Headers */,
>@@ -11721,7 +12606,9 @@
> 				419C83CC1FE247B40040C30F /* rtp_header_parser.h in Headers */,
> 				41F9BF8D2051C80100ABF0B9 /* rtp_headers.h in Headers */,
> 				5CDD88BF1E43BE3D00621E92 /* rtp_packet.h in Headers */,
>+				419241422127372400634FCF /* fft_util.h in Headers */,
> 				5CDD88BA1E43BE3D00621E92 /* rtp_packet_history.h in Headers */,
>+				41F263B0212680A800274F59 /* unused.h in Headers */,
> 				5CDD88BB1E43BE3D00621E92 /* rtp_packet_received.h in Headers */,
> 				419C82B61FE20DCD0040C30F /* rtp_packet_sink_interface.h in Headers */,
> 				5CDD88BC1E43BE3D00621E92 /* rtp_packet_to_send.h in Headers */,
>@@ -11729,9 +12616,12 @@
> 				419C83C71FE247B40040C30F /* rtp_receiver.h in Headers */,
> 				5CDD88C31E43BE3D00621E92 /* rtp_receiver_audio.h in Headers */,
> 				5CDD88C51E43BE3D00621E92 /* rtp_receiver_impl.h in Headers */,
>+				41A08BD721272F1D001D5D7B /* congestion_window_pushback_controller.h in Headers */,
> 				5CDD88C71E43BE3D00621E92 /* rtp_receiver_strategy.h in Headers */,
> 				5CDD88C91E43BE3D00621E92 /* rtp_receiver_video.h in Headers */,
> 				419C83C31FE247B40040C30F /* rtp_rtcp.h in Headers */,
>+				4102F6A221273262006AE8D7 /* audio_level.h in Headers */,
>+				4102F6DE21273416006AE8D7 /* video_stream_decoder_create.h in Headers */,
> 				5CDD88CA1E43BE3D00621E92 /* rtp_rtcp_config.h in Headers */,
> 				419C83C81FE247B40040C30F /* rtp_rtcp_defines.h in Headers */,
> 				419C82B51FE20DCD0040C30F /* rtp_rtcp_demuxer_helper.h in Headers */,
>@@ -11750,31 +12640,33 @@
> 				5CDD896D1E43BF3A00621E92 /* rtpfb.h in Headers */,
> 				419C82CA1FE20E2C0040C30F /* rtpmediautils.h in Headers */,
> 				5C63F9661E41737B002CA531 /* rtpparameters.h in Headers */,
>+				41F2636421267ADF00274F59 /* video_codec.h in Headers */,
> 				5CD2850A1E6A60570094FDC8 /* rtpreceiver.h in Headers */,
> 				5C63F9691E41737B002CA531 /* rtpreceiverinterface.h in Headers */,
> 				5CD2850C1E6A60570094FDC8 /* rtpsender.h in Headers */,
> 				5C63F96C1E41737B002CA531 /* rtpsenderinterface.h in Headers */,
> 				419C82DA1FE20E590040C30F /* rtptransceiver.h in Headers */,
> 				41F9BF8E2051C80100ABF0B9 /* rtptransceiverinterface.h in Headers */,
>+				41DDB263212679A300296D47 /* opensslsessioncache.h in Headers */,
> 				4145E4CE1EF8CB9500FCF6E6 /* rtptransport.h in Headers */,
> 				419C82E51FE20E590040C30F /* rtptransportinternal.h in Headers */,
> 				419C82DB1FE20E590040C30F /* rtptransportinternaladapter.h in Headers */,
> 				5C4B48F21E42C1E3002651C8 /* rtputils.h in Headers */,
> 				5CDD83F91E439A6F00621E92 /* rtt_filter.h in Headers */,
> 				419C82C21FE20DCD0040C30F /* rtx_receive_stream.h in Headers */,
>-				5C4B4CDF1E4320A9002651C8 /* rw_lock_posix.h in Headers */,
> 				419C83571FE246650040C30F /* safe_compare.h in Headers */,
> 				419C83531FE246650040C30F /* safe_conversions.h in Headers */,
> 				419C83541FE246650040C30F /* safe_minmax.h in Headers */,
> 				413A23281FE18E0700373E99 /* sanitizer.h in Headers */,
>-				5CDD90301E43CEDE00621E92 /* saturating_gain_estimator.h in Headers */,
> 				413A23051FE18E0700373E99 /* scoped_ref_ptr.h in Headers */,
> 				41F9BFB62051C93600ABF0B9 /* scopedvideodecoder.h in Headers */,
>+				41A08BB1212681C8001D5D7B /* video_source_interface.h in Headers */,
> 				41F9BFB72051C93600ABF0B9 /* scopedvideoencoder.h in Headers */,
> 				419C84361FE24E800040C30F /* screenshare_layers.h in Headers */,
> 				5CD2863D1E6A681C0094FDC8 /* sctptransport.h in Headers */,
> 				5CD2863E1E6A681C0094FDC8 /* sctptransportinternal.h in Headers */,
> 				5CD2850E1E6A60570094FDC8 /* sctputils.h in Headers */,
>+				41E02CD32127360700C27CD6 /* sample_counter.h in Headers */,
> 				5CDD89701E43BF3A00621E92 /* sdes.h in Headers */,
> 				4145F61D1FE1F38D00EB9CAF /* sdp_video_format.h in Headers */,
> 				419C82DD1FE20E590040C30F /* sdputils.h in Headers */,
>@@ -11784,17 +12676,16 @@
> 				5CDD89731E43BF3A00621E92 /* sender_report.h in Headers */,
> 				413A22D41FE18E0700373E99 /* sequenced_task_checker.h in Headers */,
> 				413A23AE1FE18E0800373E99 /* sequenced_task_checker_impl.h in Headers */,
>-				5C63FA071E4174F6002CA531 /* session.h in Headers */,
> 				5CDD83FC1E439A6F00621E92 /* session_info.h in Headers */,
> 				41F9BFDF2051DE1800ABF0B9 /* sessiondescription.h in Headers */,
> 				41F9BF922051C80100ABF0B9 /* setremotedescriptionobserverinterface.h in Headers */,
>+				41A08BBF21269553001D5D7B /* rtc_event_ice_candidate_pair_config.h in Headers */,
> 				41433D081F79B33400387B4D /* settings.h in Headers */,
> 				5CDD87D71E43BC0600621E92 /* settings.h in Headers */,
>-				413A22D51FE18E0700373E99 /* sha1.h in Headers */,
>-				413A22FC1FE18E0700373E99 /* sha1digest.h in Headers */,
> 				5CD285E01E6A63430094FDC8 /* shadow_filter_update_gain.h in Headers */,
>-				5CDD90321E43CEDE00621E92 /* signal_classifier.h in Headers */,
> 				41F9BFC52051CAE500ABF0B9 /* signal_processing_library.h in Headers */,
>+				415F1FAB21272FBA00064CBF /* compute_interpolated_gain_curve.h in Headers */,
>+				419241F021275AFA00634FCF /* simulcast_rate_allocator.h in Headers */,
> 				413A23811FE18E0700373E99 /* signalthread.h in Headers */,
> 				413A22CD1FE18E0700373E99 /* sigslot.h in Headers */,
> 				413A23331FE18E0700373E99 /* sigslotrepeater.h in Headers */,
>@@ -11804,22 +12695,24 @@
> 				5CDD8BE01E43C2B500621E92 /* simple_lsf_dequant.h in Headers */,
> 				5CDD8BE21E43C2B500621E92 /* simple_lsf_quant.h in Headers */,
> 				5C4B48871E42C1BA002651C8 /* simulcast.h in Headers */,
>-				419C84371FE24E800040C30F /* simulcast_encoder_adapter.h in Headers */,
>-				419C84391FE24E800040C30F /* simulcast_rate_allocator.h in Headers */,
> 				5CDD875D1E43BAF500621E92 /* sinc_resampler.h in Headers */,
> 				5CDD875F1E43BAF500621E92 /* sinusoidal_linear_chirp_source.h in Headers */,
> 				5CDD8BE61E43C2B500621E92 /* smooth.h in Headers */,
> 				5CDD8BE41E43C2B500621E92 /* smooth_out_data.h in Headers */,
> 				5CD285F71E6A63F60094FDC8 /* smoothing_filter.h in Headers */,
>+				419241582127376F00634FCF /* fakepackettransport.h in Headers */,
> 				413A22DD1FE18E0700373E99 /* socket.h in Headers */,
>+				415F1FA721272FBA00064CBF /* adaptive_digital_gain_applier.h in Headers */,
> 				413A22DC1FE18E0700373E99 /* socket_unittest.h in Headers */,
> 				413A23611FE18E0700373E99 /* socketadapters.h in Headers */,
> 				413A23C91FE18E0800373E99 /* socketaddress.h in Headers */,
> 				413A23CA1FE18E0800373E99 /* socketaddresspair.h in Headers */,
> 				413A22C51FE18E0700373E99 /* socketfactory.h in Headers */,
>-				5C4B4D0C1E4322F6002651C8 /* socketmonitor.h in Headers */,
> 				413A23B91FE18E0800373E99 /* socketserver.h in Headers */,
> 				413A23531FE18E0700373E99 /* socketstream.h in Headers */,
>+				419241DE21275A3000634FCF /* data_rate.h in Headers */,
>+				4192413B2127372400634FCF /* features_extraction.h in Headers */,
>+				416D3BDC212731C200775F09 /* adaptive_agc.h in Headers */,
> 				5CDD8BE81E43C2B500621E92 /* sort_sq.h in Headers */,
> 				5C4B4C7E1E431F9C002651C8 /* sparse_fir_filter.h in Headers */,
> 				5CDD87D91E43BC0600621E92 /* spectrum_ar_model_tables.h in Headers */,
>@@ -11834,7 +12727,9 @@
> 				419C82E21FE20E590040C30F /* srtptransport.h in Headers */,
> 				413A23AF1FE18E0800373E99 /* sslfingerprint.h in Headers */,
> 				413A22E41FE18E0700373E99 /* sslidentity.h in Headers */,
>+				419241822127497100634FCF /* bad_variant_access.h in Headers */,
> 				413A23921FE18E0800373E99 /* sslroots.h in Headers */,
>+				416D3BDD212731C200775F09 /* adaptive_mode_level_estimator_agc.h in Headers */,
> 				413A23C11FE18E0800373E99 /* sslstreamadapter.h in Headers */,
> 				41F9BF9E2051C86900ABF0B9 /* ssrc_binding_observer.h in Headers */,
> 				5CDD87371E43BABE00621E92 /* standalone_vad.h in Headers */,
>@@ -11847,14 +12742,17 @@
> 				5C63F9731E41737B002CA531 /* statstypes.h in Headers */,
> 				413A22FD1FE18E0700373E99 /* stream.h in Headers */,
> 				5CDD85C01E43B5C000621E92 /* stream_synchronization.h in Headers */,
>+				4102F6CE212733B7006AE8D7 /* video_stream_decoder_impl.h in Headers */,
> 				5CD285111E6A60570094FDC8 /* streamcollection.h in Headers */,
> 				5C4B48F51E42C1E3002651C8 /* streamparams.h in Headers */,
> 				413A23641FE18E0700373E99 /* string_to_number.h in Headers */,
>+				419241CB2127593F00634FCF /* rtt_stats.h in Headers */,
> 				413A22EB1FE18E0700373E99 /* stringencode.h in Headers */,
> 				413A23B11FE18E0800373E99 /* stringize_macros.h in Headers */,
> 				413A22CE1FE18E0700373E99 /* stringutils.h in Headers */,
> 				41433CF01F79B33400387B4D /* structs.h in Headers */,
> 				5CDD87DA1E43BC0600621E92 /* structs.h in Headers */,
>+				419241AA212758D300634FCF /* rnn.h in Headers */,
> 				5C63FA0C1E4174F6002CA531 /* stun.h in Headers */,
> 				5C63FA0F1E4174F6002CA531 /* stunport.h in Headers */,
> 				5C63FA121E4174F6002CA531 /* stunrequest.h in Headers */,
>@@ -11862,9 +12760,11 @@
> 				5CD285E31E6A63430094FDC8 /* subtractor.h in Headers */,
> 				5CD285E11E6A63430094FDC8 /* subtractor_output.h in Headers */,
> 				5CD285E51E6A63430094FDC8 /* suppression_filter.h in Headers */,
>+				413AD1A421265B10003F7263 /* container.h in Headers */,
> 				5CD285E71E6A63430094FDC8 /* suppression_gain.h in Headers */,
> 				5CDD8BF01E43C2B500621E92 /* swap_bytes.h in Headers */,
> 				413A23471FE18E0700373E99 /* swap_queue.h in Headers */,
>+				4102F6AF212732E7006AE8D7 /* expand_uma_logger.h in Headers */,
> 				5CDD8A551E43BFB300621E92 /* sync_buffer.h in Headers */,
> 				5CD284BF1E6A5F9F0094FDC8 /* syncable.h in Headers */,
> 				5CDD89791E43BF3A00621E92 /* target_bitrate.h in Headers */,
>@@ -11873,23 +12773,30 @@
> 				5C63FA181E4174F6002CA531 /* tcpport.h in Headers */,
> 				413A22FE1FE18E0700373E99 /* template_util.h in Headers */,
> 				419C843D1FE24E800040C30F /* temporal_layers.h in Headers */,
>+				419241E421275A3000634FCF /* data_size.h in Headers */,
> 				413A22A11FE18E0600373E99 /* testbase64.h in Headers */,
> 				413A23D51FE18E0800373E99 /* testclient.h in Headers */,
> 				413A232A1FE18E0700373E99 /* testechoserver.h in Headers */,
> 				5C63FA191E4174F6002CA531 /* testrelayserver.h in Headers */,
>+				41299B932127367B00B3414B /* pitch_filter.h in Headers */,
> 				5C63FA1A1E4174F6002CA531 /* teststunserver.h in Headers */,
> 				5C63FA1B1E4174F6002CA531 /* testturnserver.h in Headers */,
>+				41E02CBD2127356A00C27CD6 /* sslcertificate.h in Headers */,
> 				413A23D91FE18E0800373E99 /* testutils.h in Headers */,
> 				413A23421FE18E0700373E99 /* thread.h in Headers */,
> 				413A22C61FE18E0700373E99 /* thread_annotations.h in Headers */,
>+				41E02CAB2127352D00C27CD6 /* delay_based_bwe.h in Headers */,
>+				41E02CB72127352D00C27CD6 /* delay_increase_detector_interface.h in Headers */,
> 				413A22E81FE18E0700373E99 /* thread_checker.h in Headers */,
>+				4192415A2127376F00634FCF /* icetransportinternal.h in Headers */,
> 				413A22B81FE18E0700373E99 /* thread_checker_impl.h in Headers */,
> 				5CDD84C41E43AF1300621E92 /* three_band_filter_bank.h in Headers */,
> 				5CDD8A581E43BFB300621E92 /* tick_timer.h in Headers */,
> 				419C83341FE245EA0040C30F /* time_interval.h in Headers */,
>+				41F2636021267ADF00274F59 /* builtin_video_decoder_factory.h in Headers */,
>+				419241432127372800634FCF /* pitch_info.h in Headers */,
> 				5CDD8A5B1E43BFB300621E92 /* time_stretch.h in Headers */,
> 				5CDD88DB1E43BE3D00621E92 /* time_util.h in Headers */,
>-				413A23E01FE18E0800373E99 /* timedelta.h in Headers */,
> 				5CDD83FE1E439A6F00621E92 /* timestamp_map.h in Headers */,
> 				5CDD8A5E1E43BFB300621E92 /* timestamp_scaler.h in Headers */,
> 				413A23541FE18E0700373E99 /* timestampaligner.h in Headers */,
>@@ -11898,34 +12805,32 @@
> 				5CDD897B1E43BF3A00621E92 /* tmmb_item.h in Headers */,
> 				5CDD897E1E43BF3A00621E92 /* tmmbn.h in Headers */,
> 				5CDD89811E43BF3A00621E92 /* tmmbr.h in Headers */,
>+				419241AB212758D300634FCF /* spectral_features.h in Headers */,
> 				5CDD88DD1E43BE3D00621E92 /* tmmbr_help.h in Headers */,
> 				413A23001FE18E0700373E99 /* trace_event.h in Headers */,
> 				5CD285131E6A60570094FDC8 /* trackmediainfomap.h in Headers */,
>-				413A237A1FE18E0700373E99 /* transformadapter.h in Headers */,
> 				5CDD8C921E43C66000621E92 /* transient_detector.h in Headers */,
> 				5CDD8C961E43C66000621E92 /* transient_suppressor.h in Headers */,
> 				5C63FA1E1E4174F6002CA531 /* transport.h in Headers */,
> 				5CDD85C21E43B5C000621E92 /* transport_adapter.h in Headers */,
> 				5CDD89841E43BF3A00621E92 /* transport_feedback.h in Headers */,
>+				4192413F2127372400634FCF /* common.h in Headers */,
> 				5CDD8FBC1E43CD0700621E92 /* transport_feedback_adapter.h in Headers */,
>-				4130920E1EF8D40100757C55 /* transport_feedback_packet_loss_tracker.h in Headers */,
>-				5CD2860B1E6A64840094FDC8 /* transport_feedback_packet_loss_tracker.h in Headers */,
>-				419C82D91FE20E590040C30F /* transportcontroller.h in Headers */,
>+				4102F6B221273331006AE8D7 /* mediaconfig.h in Headers */,
> 				5C63FA261E4174F6002CA531 /* transportdescription.h in Headers */,
> 				5C63FA291E4174F6002CA531 /* transportdescriptionfactory.h in Headers */,
> 				5C63FA2A1E4174F6002CA531 /* transportinfo.h in Headers */,
>-				5CD286211E6A66380094FDC8 /* trendline_estimator.h in Headers */,
> 				41F9BF912051C80100ABF0B9 /* turncustomizer.h in Headers */,
> 				5C63FA2D1E4174F6002CA531 /* turnport.h in Headers */,
> 				41F9BFE92051DEEA00ABF0B9 /* turnportfactory.h in Headers */,
> 				5C63FA301E4174F6002CA531 /* turnserver.h in Headers */,
> 				5C4B48FA1E42C1E3002651C8 /* turnutils.h in Headers */,
> 				413A234B1FE18E0700373E99 /* type_traits.h in Headers */,
>-				5C6CDD501E413598009754E3 /* typedefs.h in Headers */,
> 				5CDD84C61E43AF1300621E92 /* typing_detection.h in Headers */,
> 				5C63FA311E4174F6002CA531 /* udpport.h in Headers */,
> 				413A246B1FE1991A00373E99 /* UIDevice+RTCDevice.h in Headers */,
> 				5CDD88E01E43BE3D00621E92 /* ulpfec_generator.h in Headers */,
>+				416D3BE4212731C200775F09 /* fixed_gain_controller.h in Headers */,
> 				5CDD88E31E43BE3D00621E92 /* ulpfec_header_reader_writer.h in Headers */,
> 				419C83C61FE247B40040C30F /* ulpfec_receiver.h in Headers */,
> 				5CDD88E51E43BE3D00621E92 /* ulpfec_receiver_impl.h in Headers */,
>@@ -11934,9 +12839,9 @@
> 				5CDD8BF21E43C2B500621E92 /* unpack_bits.h in Headers */,
> 				413A21401FE0F0EF00373E99 /* ut_sim.h in Headers */,
> 				5CDD87091E43BA7500621E92 /* utility.h in Headers */,
>-				5CDD846C1E43AE2900621E92 /* utility.h in Headers */,
> 				5CDD87DF1E43BC2E00621E92 /* utility.h in Headers */,
> 				5CDD873B1E43BABE00621E92 /* vad_audio_proc.h in Headers */,
>+				41A08BBE21269553001D5D7B /* rtc_event_ice_candidate_pair.h in Headers */,
> 				5CDD87381E43BABE00621E92 /* vad_audio_proc_internal.h in Headers */,
> 				5CDD873E1E43BABE00621E92 /* vad_circular_buffer.h in Headers */,
> 				5CDD86D71E43BA2800621E92 /* vad_core.h in Headers */,
>@@ -11944,10 +12849,9 @@
> 				5CDD86DD1E43BA2800621E92 /* vad_gmm.h in Headers */,
> 				5CDD86E01E43BA2800621E92 /* vad_sp.h in Headers */,
> 				5C63F8E01E416D53002CA531 /* value.h in Headers */,
>+				419241532127376F00634FCF /* fakedtlstransport.h in Headers */,
> 				41F9BFD82051DDE500ABF0B9 /* vector_buffer.h in Headers */,
>-				5CDD88E71E43BE3D00621E92 /* video_codec_information.h in Headers */,
> 				5CDD84021E439A6F00621E92 /* video_coding_impl.h in Headers */,
>-				419C82C31FE20DCD0040C30F /* video_config.h in Headers */,
> 				419C84131FE249AB0040C30F /* video_content_type.h in Headers */,
> 				4145E48D1EF88B9D00FCF6E6 /* video_decoder.h in Headers */,
> 				4145F61E1FE1F38D00EB9CAF /* video_decoder_factory.h in Headers */,
>@@ -11955,33 +12859,38 @@
> 				4145F61F1FE1F38D00EB9CAF /* video_encoder_factory.h in Headers */,
> 				5CD2846B1E6A57F40094FDC8 /* video_frame.h in Headers */,
> 				41109AAE1E5FA19200C0955A /* video_frame_buffer.h in Headers */,
>+				41A08BB3212681C8001D5D7B /* video_stream_encoder_create.h in Headers */,
> 				5CD284691E6A57F40094FDC8 /* video_frame_buffer.h in Headers */,
>+				419241DD21275A3000634FCF /* time_delta.h in Headers */,
> 				419C82C41FE20DCD0040C30F /* video_receive_stream.h in Headers */,
>+				41F263AE212680A800274F59 /* fallthrough.h in Headers */,
>+				41E02CAC2127352D00C27CD6 /* bitrate_estimator.h in Headers */,
> 				5CDD85CB1E43B5C000621E92 /* video_receive_stream.h in Headers */,
> 				5C4B4C211E431F75002651C8 /* video_render_frames.h in Headers */,
> 				5CD2846C1E6A57F40094FDC8 /* video_rotation.h in Headers */,
> 				419C82BA1FE20DCD0040C30F /* video_send_stream.h in Headers */,
> 				5CDD85CE1E43B5C000621E92 /* video_send_stream.h in Headers */,
>+				41A08BB5212681C8001D5D7B /* video_stream_encoder_observer.h in Headers */,
> 				5CDD85D01E43B5C000621E92 /* video_stream_decoder.h in Headers */,
> 				419C84121FE249AB0040C30F /* video_timing.h in Headers */,
>+				419241622127385B00634FCF /* rw_lock_posix.h in Headers */,
> 				5C4B48FD1E42C1E3002651C8 /* videoadapter.h in Headers */,
> 				5C4B49001E42C1E3002651C8 /* videobroadcaster.h in Headers */,
> 				5C4B49031E42C1E3002651C8 /* videocapturer.h in Headers */,
>+				419241E021275A3000634FCF /* timestamp.h in Headers */,
> 				5C4B49041E42C1E3002651C8 /* videocapturerfactory.h in Headers */,
>+				419241572127376F00634FCF /* udptransport.h in Headers */,
> 				5CD285151E6A60570094FDC8 /* videocapturertracksource.h in Headers */,
> 				5C4B49071E42C1E3002651C8 /* videocommon.h in Headers */,
>-				5CD286251E6A666D0094FDC8 /* videodecodersoftwarefallbackwrapper.h in Headers */,
>-				5C4B488A1E42C1BA002651C8 /* videoencodersoftwarefallbackwrapper.h in Headers */,
> 				41ECEAFE20646664009D5141 /* VideoProcessingSoftLink.h in Headers */,
>-				41F9BF892051C80100ABF0B9 /* videosinkinterface.h in Headers */,
> 				5C4B490C1E42C1E3002651C8 /* videosourcebase.h in Headers */,
>-				41F9BF8A2051C80100ABF0B9 /* videosourceinterface.h in Headers */,
> 				5C63F9791E41737B002CA531 /* videosourceproxy.h in Headers */,
> 				419C84B11FE304C70040C30F /* VideoToolBoxDecoderFactory.h in Headers */,
> 				419C84B51FE309050040C30F /* VideoToolBoxEncoderFactory.h in Headers */,
> 				5CD285171E6A60570094FDC8 /* videotrack.h in Headers */,
> 				5CD285191E6A60570094FDC8 /* videotracksource.h in Headers */,
> 				413A22BC1FE18E0700373E99 /* virtualsocketserver.h in Headers */,
>+				41F263B3212680A800274F59 /* arch.h in Headers */,
> 				5CDD87411E43BABE00621E92 /* voice_activity_detector.h in Headers */,
> 				5CDD84C81E43AF1300621E92 /* voice_detection_impl.h in Headers */,
> 				5CDD87421E43BABE00621E92 /* voice_gmm_tables.h in Headers */,
>@@ -11994,10 +12903,12 @@
> 				5C4B4C811E431F9C002651C8 /* wav_file.h in Headers */,
> 				5C4B4C841E431F9C002651C8 /* wav_header.h in Headers */,
> 				41ECEABC206403C2009D5141 /* WebKitUtilities.h in Headers */,
>+				419241AC212758D300634FCF /* sequence_buffer.h in Headers */,
> 				5CDD8CAB1E43C6F700621E92 /* webrtc_cng.h in Headers */,
> 				4145E4911EF88EF700FCF6E6 /* webrtc_libyuv.h in Headers */,
> 				41F9BFC72051DCE900ABF0B9 /* webrtc_vad.h in Headers */,
> 				5C4B488E1E42C1BA002651C8 /* webrtcmediaengine.h in Headers */,
>+				4102F6E121273416006AE8D7 /* video_stream_encoder_settings.h in Headers */,
> 				5CD2847A1E6A5D550094FDC8 /* webrtcsdp.h in Headers */,
> 				5CD2851B1E6A60570094FDC8 /* webrtcsessiondescriptionfactory.h in Headers */,
> 				5C4B48911E42C1BA002651C8 /* webrtcvideocapturer.h in Headers */,
>@@ -12219,7 +13130,7 @@
> 			);
> 			runOnlyForDeploymentPostprocessing = 0;
> 			shellPath = /bin/sh;
>-			shellScript = "PRIVATE_HEADERS_FOLDER_PATH=usr/local/include\n\nif [[ \"${DEPLOYMENT_LOCATION}\" == \"NO\" ]]; then\n    PRIVATE_HEADERS_PATH=\"${TARGET_BUILD_DIR%/}/${PRIVATE_HEADERS_FOLDER_PATH}\"\nelse\n    PRIVATE_HEADERS_PATH=\"${DSTROOT}${INSTALL_PATH_PREFIX%/}/${PRIVATE_HEADERS_FOLDER_PATH}\"\nfi;\n\nmkdir -p \"${PRIVATE_HEADERS_PATH}\"\n\nrsync -av --no-owner --no-group --prune-empty-dirs --exclude \".svn\" --exclude \"usr\" --include \"*/\" --include \"*.h\" --exclude \"*\" \"${SRCROOT}/Source/webrtc\" \"${PRIVATE_HEADERS_PATH}\"\n";
>+			shellScript = "PRIVATE_HEADERS_FOLDER_PATH=usr/local/include\n\nif [[ \"${DEPLOYMENT_LOCATION}\" == \"NO\" ]]; then\n    PRIVATE_HEADERS_PATH=\"${TARGET_BUILD_DIR%/}/${PRIVATE_HEADERS_FOLDER_PATH}\"\nelse\n    PRIVATE_HEADERS_PATH=\"${DSTROOT}${INSTALL_PATH_PREFIX%/}/${PRIVATE_HEADERS_FOLDER_PATH}\"\nfi;\n\nmkdir -p \"${PRIVATE_HEADERS_PATH}\"\n\nrsync -av --no-owner --no-group --prune-empty-dirs --exclude \".svn\" --exclude \"usr\" --include \"*/\" --include \"*.h\" --exclude \"*\" \"${SRCROOT}/Source/webrtc\" \"${PRIVATE_HEADERS_PATH}\"\n\nrsync -av --no-owner --no-group --prune-empty-dirs --exclude \".svn\" --exclude \"usr\" --include \"*/\" --include \"*.h\" --exclude \"*\" \"${SRCROOT}/Source/third_party/abseil-cpp/absl\" \"${PRIVATE_HEADERS_PATH}\"\n";
> 		};
> 		5CD286461E6E154E0094FDC8 /* Check for Weak VTables and Externals */ = {
> 			isa = PBXShellScriptBuildPhase;
>@@ -12303,13 +13214,10 @@
> 			isa = PBXSourcesBuildPhase;
> 			buildActionMask = 2147483647;
> 			files = (
>-				5C088C0E1E4AA44400403995 /* audiomonitor.cc in Sources */,
> 				5C088C101E4AA44400403995 /* bundlefilter.cc in Sources */,
> 				5C088C121E4AA44400403995 /* channel.cc in Sources */,
> 				5C088C141E4AA44400403995 /* channelmanager.cc in Sources */,
>-				5C088C161E4AA44400403995 /* currentspeakermonitor.cc in Sources */,
> 				5C088C181E4AA44400403995 /* externalhmac.cc in Sources */,
>-				5C088C1A1E4AA44400403995 /* mediamonitor.cc in Sources */,
> 				5C088C1C1E4AA44400403995 /* mediasession.cc in Sources */,
> 				5C088C1E1E4AA44400403995 /* rtcpmuxfilter.cc in Sources */,
> 				5C088C201E4AA44400403995 /* srtpfilter.cc in Sources */,
>@@ -12798,45 +13706,54 @@
> 			isa = PBXSourcesBuildPhase;
> 			buildActionMask = 2147483647;
> 			files = (
>+				415F1FB121272FBA00064CBF /* gain_curve_applier.cc in Sources */,
> 				5CDD8B6C1E43C2B500621E92 /* abs_quant.c in Sources */,
> 				5CDD8B6A1E43C2B500621E92 /* abs_quant_loop.c in Sources */,
> 				5CDD89F81E43BFB300621E92 /* accelerate.cc in Sources */,
>-				419C84071FE249910040C30F /* acknowledged_bitrate_estimator.cc in Sources */,
> 				5CDD85031E43B1EA00621E92 /* acm_codec_database.cc in Sources */,
> 				5CDD85091E43B1EA00621E92 /* acm_receiver.cc in Sources */,
> 				5CDD850B1E43B1EA00621E92 /* acm_resampler.cc in Sources */,
> 				5C4B48D71E42C1E3002651C8 /* adaptedvideotracksource.cc in Sources */,
>+				419241C52127593F00634FCF /* loss_rate_filter.cc in Sources */,
>+				419241542127376F00634FCF /* basicasyncresolverfactory.cc in Sources */,
> 				5CD285A41E6A63430094FDC8 /* adaptive_fir_filter.cc in Sources */,
> 				41F411C71EF97BAC00343C26 /* adm_helpers.cc in Sources */,
> 				5CD285A81E6A63430094FDC8 /* aec3_common.cc in Sources */,
> 				5CD285AA1E6A63430094FDC8 /* aec3_fft.cc in Sources */,
>+				41F263B72126810C00274F59 /* base64.cc in Sources */,
> 				5CDD86BC1E43B9C200621E92 /* aec_core.cc in Sources */,
> 				5CDD86BB1E43B9C200621E92 /* aec_core_sse2.cc in Sources */,
> 				413092131EF8D51100757C55 /* aec_dump.cc in Sources */,
> 				5CDD86BE1E43B9C200621E92 /* aec_resampler.cc in Sources */,
> 				5CD285A61E6A63430094FDC8 /* aec_state.cc in Sources */,
>+				415F1FC02127304C00064CBF /* echo_cancellation_proxy.cc in Sources */,
> 				5CDD85E71E43B81000621E92 /* aecm_core.cc in Sources */,
> 				5CDD85E41E43B81000621E92 /* aecm_core_c.cc in Sources */,
> 				5CDD87011E43BA7500621E92 /* agc.cc in Sources */,
>+				41F2639221267F5300274F59 /* g722_encode.c in Sources */,
> 				5CDD86FF1E43BA7500621E92 /* agc_manager_direct.cc in Sources */,
> 				5CDD8FD11E43CD6600621E92 /* aimd_rate_control.cc in Sources */,
>-				5C4B4CC11E4320A9002651C8 /* aligned_malloc.cc in Sources */,
>-				5CDD8F9E1E43CCBE00621E92 /* alr_detector.cc in Sources */,
> 				41F9BFCB2051DDA200ABF0B9 /* alr_experiment.cc in Sources */,
> 				5CDD86EC1E43BA6D00621E92 /* analog_agc.c in Sources */,
> 				5CDD87E51E43BDA100621E92 /* apm_data_dumper.cc in Sources */,
> 				5CD284C21E6A5FFC0094FDC8 /* apm_helpers.cc in Sources */,
> 				5CDD893B1E43BF3A00621E92 /* app.cc in Sources */,
>+				419241642127385B00634FCF /* rw_lock_posix.cc in Sources */,
> 				5CDD87A91E43BC0500621E92 /* arith_routines.c in Sources */,
> 				41433D0A1F79B33400387B4D /* arith_routines.c in Sources */,
>+				4102F6AB212732B1006AE8D7 /* rtp_payload_params.cc in Sources */,
>+				41299B8C2127365100B3414B /* send_time_history.cc in Sources */,
> 				5CDD87A71E43BC0500621E92 /* arith_routines_hist.c in Sources */,
> 				41433D291F79B33400387B4D /* arith_routines_hist.c in Sources */,
>+				41E02C8B212734B900C27CD6 /* decoder_database.cc in Sources */,
>+				41F2638A21267F4000274F59 /* fft.c in Sources */,
> 				5CDD87A81E43BC0500621E92 /* arith_routines_logist.c in Sources */,
>+				41E02C88212734B900C27CD6 /* encoder_database.cc in Sources */,
> 				41433D0F1F79B33400387B4D /* arith_routines_logist.c in Sources */,
>-				5CDD90751E43D15C00621E92 /* array_util.cc in Sources */,
> 				413A238C1FE18E0700373E99 /* asyncinvoker.cc in Sources */,
> 				413A216D1FE18D6A00373E99 /* asyncpacketsocket.cc in Sources */,
> 				413A23551FE18E0700373E99 /* asyncresolverinterface.cc in Sources */,
>+				41299B8B2127365100B3414B /* pacer_controller.cc in Sources */,
> 				413A23961FE18E0800373E99 /* asyncsocket.cc in Sources */,
> 				5C63F9E01E4174F6002CA531 /* asyncstuntcpsocket.cc in Sources */,
> 				413A23AB1FE18E0800373E99 /* asynctcpsocket.cc in Sources */,
>@@ -12844,6 +13761,7 @@
> 				5CDD84A91E43AF1300621E92 /* audio_buffer.cc in Sources */,
> 				5CDD85101E43B1EA00621E92 /* audio_coding_module.cc in Sources */,
> 				5C4B4C591E431F9C002651C8 /* audio_converter.cc in Sources */,
>+				41DDB26E212679D200296D47 /* string_builder.cc in Sources */,
> 				5CD285441E6A61D20094FDC8 /* audio_decoder.cc in Sources */,
> 				419C83E91FE248350040C30F /* audio_decoder_g711.cc in Sources */,
> 				4140B8311E4E3396007409E6 /* audio_decoder_g722.cc in Sources */,
>@@ -12851,7 +13769,9 @@
> 				5CDD8B6E1E43C2B500621E92 /* audio_decoder_ilbc.cc in Sources */,
> 				419C84BF1FE319440040C30F /* audio_decoder_ilbc.cc in Sources */,
> 				5CDD87AB1E43BC0500621E92 /* audio_decoder_isac.cc in Sources */,
>+				4192419A2127586500634FCF /* kiss_fft.cc in Sources */,
> 				419C835C1FE246BB0040C30F /* audio_decoder_isac_fix.cc in Sources */,
>+				419241EE21275AFA00634FCF /* simulcast_rate_allocator.cc in Sources */,
> 				419C836A1FE246CA0040C30F /* audio_decoder_isac_float.cc in Sources */,
> 				41433CF91F79B33400387B4D /* audio_decoder_isacfix.cc in Sources */,
> 				419C83F21FE2488D0040C30F /* audio_decoder_L16.cc in Sources */,
>@@ -12860,6 +13780,8 @@
> 				4140B8221E4E3383007409E6 /* audio_decoder_pcm.cc in Sources */,
> 				5CDD8FFC1E43CE3A00621E92 /* audio_decoder_pcm16b.cc in Sources */,
> 				5C119FFB1E457400004F0987 /* audio_device_buffer.cc in Sources */,
>+				419241E221275A3000634FCF /* time_delta.cc in Sources */,
>+				41F2639421267F5300274F59 /* g722_decode.c in Sources */,
> 				5C11A01D1E457578004F0987 /* audio_device_dummy.cc in Sources */,
> 				5C119FFE1E457400004F0987 /* audio_device_generic.cc in Sources */,
> 				5C11A00B1E457448004F0987 /* audio_device_mac.cc in Sources */,
>@@ -12867,12 +13789,18 @@
> 				5CDD8CA71E43C6F700621E92 /* audio_encoder_cng.cc in Sources */,
> 				5CDD8FF31E43CDF400621E92 /* audio_encoder_copy_red.cc in Sources */,
> 				419C83E71FE248350040C30F /* audio_encoder_g711.cc in Sources */,
>+				41F2636221267ADF00274F59 /* sdp_video_format.cc in Sources */,
>+				419241EF21275AFA00634FCF /* simulcast_utility.cc in Sources */,
> 				4140B8331E4E3396007409E6 /* audio_encoder_g722.cc in Sources */,
> 				419C833E1FE246230040C30F /* audio_encoder_g722.cc in Sources */,
> 				5CDD8B701E43C2B500621E92 /* audio_encoder_ilbc.cc in Sources */,
> 				419C84BD1FE319440040C30F /* audio_encoder_ilbc.cc in Sources */,
> 				5CDD87AD1E43BC0500621E92 /* audio_encoder_isac.cc in Sources */,
> 				419C836C1FE246CA0040C30F /* audio_encoder_isac_fix.cc in Sources */,
>+				416D3BDB212731C200775F09 /* adaptive_agc.cc in Sources */,
>+				416D3BE0212731C200775F09 /* biquad_filter.cc in Sources */,
>+				41924179212738FB00634FCF /* rtc_event_log_impl.cc in Sources */,
>+				4102F6BA21273382006AE8D7 /* audio_receive_stream.cc in Sources */,
> 				419C836B1FE246CA0040C30F /* audio_encoder_isac_float.cc in Sources */,
> 				41433CFB1F79B33400387B4D /* audio_encoder_isacfix.cc in Sources */,
> 				419C83F11FE2488D0040C30F /* audio_encoder_L16.cc in Sources */,
>@@ -12880,22 +13808,25 @@
> 				419C82F51FE20EB50040C30F /* audio_encoder_opus.cc in Sources */,
> 				419C82F31FE20EB50040C30F /* audio_encoder_opus_config.cc in Sources */,
> 				4140B8201E4E3383007409E6 /* audio_encoder_pcm.cc in Sources */,
>+				41F2638321267B7700274F59 /* rtpreceiverinterface.cc in Sources */,
> 				5CDD8FFE1E43CE3A00621E92 /* audio_encoder_pcm16b.cc in Sources */,
> 				5CD285461E6A61D20094FDC8 /* audio_format.cc in Sources */,
> 				5CDD84191E439B2900621E92 /* audio_format_conversion.cc in Sources */,
> 				5CD284921E6A5F410094FDC8 /* audio_frame_manipulator.cc in Sources */,
>+				41D6B45C21273159008F9353 /* rtp_video_sender.cc in Sources */,
> 				5CD2854D1E6A62130094FDC8 /* audio_frame_operations.cc in Sources */,
>-				413092061EF8D0C900757C55 /* audio_level.cc in Sources */,
> 				5CDD8A8F1E43C00F00621E92 /* audio_loop.cc in Sources */,
> 				5CD284951E6A5F410094FDC8 /* audio_mixer_impl.cc in Sources */,
> 				5C11A00D1E457448004F0987 /* audio_mixer_manager_mac.cc in Sources */,
> 				5CDD8A011E43BFB300621E92 /* audio_multi_vector.cc in Sources */,
> 				419C830D1FE20F3C0040C30F /* audio_network_adaptor_config.cc in Sources */,
> 				5CDD8F791E43CBE000621E92 /* audio_network_adaptor_impl.cc in Sources */,
>+				419241C72127593F00634FCF /* bbr_network_controller.cc in Sources */,
> 				5CDD8FE91E43CDCA00621E92 /* audio_processing.cc in Sources */,
> 				5CDD84AB1E43AF1300621E92 /* audio_processing_impl.cc in Sources */,
> 				419C829D1FE20D1C0040C30F /* audio_processing_statistics.cc in Sources */,
> 				5CDD8ACC1E43C23900621E92 /* audio_receive_stream.cc in Sources */,
>+				416D3BED212731C200775F09 /* vector_float_frame.cc in Sources */,
> 				5C4B4C5C1E431F9C002651C8 /* audio_ring_buffer.cc in Sources */,
> 				5CDD8ACF1E43C23900621E92 /* audio_send_stream.cc in Sources */,
> 				5CD284B11E6A5F9F0094FDC8 /* audio_send_stream.cc in Sources */,
>@@ -12904,6 +13835,7 @@
> 				41F9BF9B2051C84C00ABF0B9 /* audio_transport_impl.cc in Sources */,
> 				5C4B4C5F1E431F9C002651C8 /* audio_util.cc in Sources */,
> 				5CDD8A041E43BFB300621E92 /* audio_vector.cc in Sources */,
>+				41E02CAE2127352D00C27CD6 /* bitrate_estimator.cc in Sources */,
> 				5CD284F01E6A60570094FDC8 /* audiotrack.cc in Sources */,
> 				5CDD8B721E43C2B500621E92 /* augmented_cb_corr.c in Sources */,
> 				5CDD86401E43B8B500621E92 /* auto_corr_to_refl_coef.c in Sources */,
>@@ -12911,34 +13843,39 @@
> 				5CDD8A071E43BFB300621E92 /* background_noise.cc in Sources */,
> 				41433CF71F79B33400387B4D /* bandwidth_estimator.c in Sources */,
> 				5CDD87AE1E43BC0500621E92 /* bandwidth_estimator.c in Sources */,
>-				413A23DC1FE18E0800373E99 /* base64.cc in Sources */,
>+				41E02CCC212735B700C27CD6 /* bitrate_settings.cc in Sources */,
> 				5C63F9E21E4174F6002CA531 /* basicpacketsocketfactory.cc in Sources */,
>+				4102F69221273206006AE8D7 /* rtcstatstraversal.cc in Sources */,
> 				5C4B4D091E4322F6002651C8 /* basicportallocator.cc in Sources */,
> 				419C846F1FE255FA0040C30F /* bio_ssl.cc in Sources */,
>-				5CDD901C1E43CEDE00621E92 /* biquad_filter.cc in Sources */,
> 				413A233B1FE18E0700373E99 /* bitbuffer.cc in Sources */,
> 				5C4B4C191E431F75002651C8 /* bitrate_adjuster.cc in Sources */,
> 				5CDD852F1E43B39C00621E92 /* bitrate_allocator.cc in Sources */,
> 				5CDD8F7D1E43CBE000621E92 /* bitrate_controller.cc in Sources */,
>+				4102F6BC21273382006AE8D7 /* flexfec_receive_stream.cc in Sources */,
> 				5CDD905D1E43D11200621E92 /* bitrate_controller_impl.cc in Sources */,
>-				419C840B1FE249910040C30F /* bitrate_estimator.cc in Sources */,
> 				5CDD8FA01E43CCBE00621E92 /* bitrate_prober.cc in Sources */,
> 				413A21751FE18D7100373E99 /* bitrateallocationstrategy.cc in Sources */,
> 				5CD285AC1E6A63430094FDC8 /* block_framer.cc in Sources */,
> 				5CDD86871E43B93900621E92 /* block_mean_calculator.cc in Sources */,
> 				5CD285B01E6A63430094FDC8 /* block_processor.cc in Sources */,
> 				5CD285AE1E6A63430094FDC8 /* block_processor_metrics.cc in Sources */,
>+				4102F69021273206006AE8D7 /* jseptransportcontroller.cc in Sources */,
> 				5C4B4C611E431F9C002651C8 /* blocker.cc in Sources */,
> 				5CDD8A0A1E43BFB300621E92 /* buffer_level_filter.cc in Sources */,
>+				41DDB264212679A300296D47 /* opensslcertificate.cc in Sources */,
> 				413A239B1FE18E0800373E99 /* bufferqueue.cc in Sources */,
> 				5CD285481E6A61D20094FDC8 /* builtin_audio_decoder_factory.cc in Sources */,
>+				41F2637321267B4A00274F59 /* spl_sqrt_floor.c in Sources */,
> 				413091FB1EF8CFF800757C55 /* builtin_audio_encoder_factory.cc in Sources */,
> 				5CDD8B741E43C2B500621E92 /* bw_expand.c in Sources */,
> 				5CDD8FD31E43CD6600621E92 /* bwe_defines.cc in Sources */,
> 				5CDD893E1E43BF3A00621E92 /* bye.cc in Sources */,
>+				4192423E21276B5A00634FCF /* VideoToolBoxEncoderFactory.cpp in Sources */,
> 				413A23F21FE18E0800373E99 /* bytebuffer.cc in Sources */,
> 				5CDD85341E43B39C00621E92 /* call.cc in Sources */,
> 				5CDD85121E43B1EA00621E92 /* call_statistics.cc in Sources */,
>+				419241442127372800634FCF /* pitch_search_internal.cc in Sources */,
> 				5CDD859C1E43B5C000621E92 /* call_stats.cc in Sources */,
> 				4145E4E41EF8CD7D00FCF6E6 /* callfactory.cc in Sources */,
> 				419C83121FE242800040C30F /* candidate.cc in Sources */,
>@@ -12950,22 +13887,23 @@
> 				5CDD8B801E43C2B500621E92 /* cb_search.c in Sources */,
> 				5CDD8B7E1E43C2B500621E92 /* cb_search_core.c in Sources */,
> 				5CDD8B821E43C2B500621E92 /* cb_update_best_index.c in Sources */,
>-				5CDD845D1E43AE2900621E92 /* channel.cc in Sources */,
>+				4102F6DB21273416006AE8D7 /* video_stream_decoder_create.cc in Sources */,
> 				5C4B4C641E431F9C002651C8 /* channel_buffer.cc in Sources */,
> 				5CDD8F801E43CBE000621E92 /* channel_controller.cc in Sources */,
>-				5CDD845B1E43AE2900621E92 /* channel_proxy.cc in Sources */,
> 				5CDD8B841E43C2B500621E92 /* chebyshev.c in Sources */,
> 				413A23731FE18E0700373E99 /* checks.cc in Sources */,
>+				419241AD212758D300634FCF /* spectral_features.cc in Sources */,
> 				5CDD86A11E43B99400621E92 /* circular_buffer.cc in Sources */,
> 				5C4B4CC61E4320A9002651C8 /* clock.cc in Sources */,
> 				5C4B48DB1E42C1E3002651C8 /* codec.cc in Sources */,
>-				5CDD83C61E439A6F00621E92 /* codec_database.cc in Sources */,
> 				5CDD85151E43B1EA00621E92 /* codec_manager.cc in Sources */,
> 				5CDD83C81E439A6F00621E92 /* codec_timer.cc in Sources */,
> 				5CDD8A0D1E43BFB300621E92 /* comfort_noise.cc in Sources */,
> 				5CD285B41E6A63430094FDC8 /* comfort_noise_generator.cc in Sources */,
> 				5CDD89411E43BF3A00621E92 /* common_header.cc in Sources */,
>-				5C6CDD4B1E413598009754E3 /* common_types.cc in Sources */,
>+				419241812127497100634FCF /* bad_variant_access.cc in Sources */,
>+				416D3BEC212731C200775F09 /* vad_with_level.cc in Sources */,
>+				41DDB26F212679D200296D47 /* audio_format_to_string.cc in Sources */,
> 				5CDD8B861E43C2B500621E92 /* comp_corr.c in Sources */,
> 				5CFD538D1E4BD3A300482908 /* compare_neon.cc in Sources */,
> 				5CFD538E1E4BD3A300482908 /* compare_neon64.cc in Sources */,
>@@ -12978,11 +13916,11 @@
> 				419C83FB1FE249370040C30F /* constants.cc in Sources */,
> 				5CDD8F851E43CBE000621E92 /* controller.cc in Sources */,
> 				5CDD8F831E43CBE000621E92 /* controller_manager.cc in Sources */,
>+				41E02C7D2127347400C27CD6 /* rtp_video_header.cc in Sources */,
> 				41F9BFAC2051C91900ABF0B9 /* convert_legacy_video_factory.cc in Sources */,
> 				5CDD86481E43B8B500621E92 /* copy_set_operations.c in Sources */,
> 				413A23151FE18E0700373E99 /* copyonwritebuffer.cc in Sources */,
>-				4145E4AA1EF8933D00FCF6E6 /* corevideo_frame_buffer.cc in Sources */,
>-				5CDD907A1E43D15C00621E92 /* covariance_matrix_generator.cc in Sources */,
>+				41F2636921267ADF00274F59 /* video_encoder_software_fallback_wrapper.cc in Sources */,
> 				5C4B4CCA1E4320A9002651C8 /* cpu_features.cc in Sources */,
> 				5C4B4CCB1E4320A9002651C8 /* cpu_info.cc in Sources */,
> 				5CDD87B11E43BC0500621E92 /* crc.c in Sources */,
>@@ -12993,7 +13931,9 @@
> 				5CDD864B1E43B8B500621E92 /* cross_correlation.c in Sources */,
> 				5CDD8A0F1E43BFB300621E92 /* cross_correlation.cc in Sources */,
> 				413A23221FE18E0700373E99 /* cryptstring.cc in Sources */,
>+				41924246212771DC00634FCF /* objc_video_frame.mm in Sources */,
> 				419C848E1FE2574D0040C30F /* custom_extensions.cc in Sources */,
>+				419241E821275A7600634FCF /* field_trial_units.cc in Sources */,
> 				419C848C1FE2574D0040C30F /* d1_both.cc in Sources */,
> 				419C848D1FE2574D0040C30F /* d1_lib.cc in Sources */,
> 				419C848F1FE2574D0040C30F /* d1_pkt.cc in Sources */,
>@@ -13001,9 +13941,9 @@
> 				5CD284F21E6A60570094FDC8 /* datachannel.cc in Sources */,
> 				5CDD8F871E43CBE000621E92 /* debug_dump_writer.cc in Sources */,
> 				419C83F91FE248F10040C30F /* decimator.cc in Sources */,
>+				4192416E2127389B00634FCF /* field_trial_parser.cc in Sources */,
> 				5CDD8A161E43BFB300621E92 /* decision_logic.cc in Sources */,
>-				5CDD8A111E43BFB300621E92 /* decision_logic_fax.cc in Sources */,
>-				5CDD8A131E43BFB300621E92 /* decision_logic_normal.cc in Sources */,
>+				4102F6A121273262006AE8D7 /* remix_resample.cc in Sources */,
> 				41433D051F79B33400387B4D /* decode.c in Sources */,
> 				5CDD87B41E43BC0500621E92 /* decode.c in Sources */,
> 				5CDD8B8F1E43C2B500621E92 /* decode.c in Sources */,
>@@ -13012,40 +13952,54 @@
> 				41433D0E1F79B33400387B4D /* decode_plc.c in Sources */,
> 				5CDD8B8D1E43C2B500621E92 /* decode_residual.c in Sources */,
> 				5CDD8A191E43BFB300621E92 /* decoder_database.cc in Sources */,
>+				419241E121275A3000634FCF /* data_rate.cc in Sources */,
> 				5CDD8B911E43C2B500621E92 /* decoder_interpolate_lsf.c in Sources */,
>+				419241CA2127593F00634FCF /* rtt_stats.cc in Sources */,
> 				5CDD83CA1E439A6F00621E92 /* decoding_state.cc in Sources */,
> 				5CD284971E6A5F410094FDC8 /* default_output_rate_calculator.cc in Sources */,
>+				41924245212771BB00634FCF /* objc_video_decoder_factory.mm in Sources */,
> 				5CD285391E6A61980094FDC8 /* default_video_bitrate_allocator.cc in Sources */,
>-				5CDD8FB51E43CD0700621E92 /* delay_based_bwe.cc in Sources */,
> 				5CDD868D1E43B93900621E92 /* delay_estimator.cc in Sources */,
> 				5CDD868B1E43B93900621E92 /* delay_estimator_wrapper.cc in Sources */,
> 				5CDD8A1D1E43BFB300621E92 /* delay_manager.cc in Sources */,
> 				5CDD8A201E43BFB300621E92 /* delay_peak_detector.cc in Sources */,
>+				416D3BE7212731C200775F09 /* noise_level_estimator.cc in Sources */,
>+				4102F68F21273206006AE8D7 /* transportstats.cc in Sources */,
> 				5CDD86EE1E43BA6D00621E92 /* digital_agc.c in Sources */,
> 				5CDD864C1E43B8B500621E92 /* division_operations.c in Sources */,
> 				5CDD89471E43BF3A00621E92 /* dlrr.cc in Sources */,
> 				5CDD8B941E43C2B500621E92 /* do_plc.c in Sources */,
> 				412455451EF87C0F00F11809 /* dot_product_with_scale.cc in Sources */,
>-				5CDD901E1E43CEDE00621E92 /* down_sampler.cc in Sources */,
> 				5CDD86501E43B8B500621E92 /* downsample_fast.c in Sources */,
>+				4102F6BB21273382006AE8D7 /* degraded_call.cc in Sources */,
> 				4145E4D61EF8CC7600FCF6E6 /* downsampled_render_buffer.cc in Sources */,
> 				5CDD8A231E43BFB300621E92 /* dsp_helper.cc in Sources */,
> 				419C84881FE2574D0040C30F /* dtls_method.cc in Sources */,
> 				419C848A1FE2574D0040C30F /* dtls_record.cc in Sources */,
> 				419C82E41FE20E590040C30F /* dtlssrtptransport.cc in Sources */,
> 				419C83E11FE247F20040C30F /* dtlstransport.cc in Sources */,
>+				41DDB254212678E400296D47 /* platform_thread_types.cc in Sources */,
> 				419C83DE1FE247F20040C30F /* dtlstransportinternal.cc in Sources */,
>+				41E02CB02127352D00C27CD6 /* trendline_estimator.cc in Sources */,
> 				5CDD8A261E43BFB300621E92 /* dtmf_buffer.cc in Sources */,
>+				4102F6C9212733B7006AE8D7 /* video_quality_observer.cc in Sources */,
> 				5CDD88771E43BE3C00621E92 /* dtmf_queue.cc in Sources */,
> 				5CDD8A291E43BFB300621E92 /* dtmf_tone_generator.cc in Sources */,
> 				5CD284F41E6A60570094FDC8 /* dtmfsender.cc in Sources */,
>+				419241E321275A3000634FCF /* timestamp.cc in Sources */,
> 				5CDD8F8A1E43CBE000621E92 /* dtx_controller.cc in Sources */,
> 				5CDD86C11E43B9C200621E92 /* echo_cancellation.cc in Sources */,
> 				5CDD84AE1E43AF1300621E92 /* echo_cancellation_impl.cc in Sources */,
>+				4102F6A421273262006AE8D7 /* audio_level.cc in Sources */,
>+				419241D3212759A100634FCF /* fec_private_tables_random.cc in Sources */,
> 				5CD285B81E6A63430094FDC8 /* echo_canceller3.cc in Sources */,
>+				41F2639A21267F5E00274F59 /* pa_ringbuffer.c in Sources */,
> 				5CDD85EA1E43B81000621E92 /* echo_control_mobile.cc in Sources */,
>+				419241F121275B2B00634FCF /* simulcast_encoder_adapter.cc in Sources */,
>+				416D3BE8212731C200775F09 /* noise_spectrum_estimator.cc in Sources */,
> 				5CDD84B01E43AF1300621E92 /* echo_control_mobile_impl.cc in Sources */,
> 				5CD285BA1E6A63430094FDC8 /* echo_path_delay_estimator.cc in Sources */,
>+				419241C42127593F00634FCF /* bandwidth_sampler.cc in Sources */,
> 				5CD285BC1E6A63430094FDC8 /* echo_path_variability.cc in Sources */,
> 				5CD285C01E6A63430094FDC8 /* echo_remover.cc in Sources */,
> 				5CD285BE1E6A63430094FDC8 /* echo_remover_metrics.cc in Sources */,
>@@ -13055,10 +14009,12 @@
> 				5CDD87B51E43BC0500621E92 /* encode_lpc_swb.c in Sources */,
> 				5CDD8A951E43C00F00621E92 /* encode_neteq_input.cc in Sources */,
> 				5CDD83CC1E439A6F00621E92 /* encoded_frame.cc in Sources */,
>+				41A08BAB212681C8001D5D7B /* encoded_frame.cc in Sources */,
> 				5CDD859F1E43B5C000621E92 /* encoder_rtcp_feedback.cc in Sources */,
> 				5CDD86511E43B8B500621E92 /* energy.c in Sources */,
> 				5CDD8B981E43C2B500621E92 /* energy_inverse.c in Sources */,
> 				5CDD8B9A1E43C2B500621E92 /* enh_upsample.c in Sources */,
>+				41E02CBE2127356A00C27CD6 /* sslcertificate.cc in Sources */,
> 				5CDD8B9E1E43C2B500621E92 /* enhancer.c in Sources */,
> 				5CDD8B9C1E43C2B500621E92 /* enhancer_interface.c in Sources */,
> 				5CDD87B81E43BC0500621E92 /* entropy_coding.c in Sources */,
>@@ -13066,6 +14022,8 @@
> 				5CD285C21E6A63430094FDC8 /* erl_estimator.cc in Sources */,
> 				5CD285C41E6A63430094FDC8 /* erle_estimator.cc in Sources */,
> 				419C84921FE2591F0040C30F /* err_data.c in Sources */,
>+				4102F6DC21273416006AE8D7 /* color_space.cc in Sources */,
>+				41E02C7A2127347400C27CD6 /* rtp_generic_frame_descriptor.cc in Sources */,
> 				5C4B4CD21E4320A9002651C8 /* event.cc in Sources */,
> 				413A23B71FE18E0800373E99 /* event.cc in Sources */,
> 				5CD286141E6A66130094FDC8 /* event_log_writer.cc in Sources */,
>@@ -13075,45 +14033,45 @@
> 				5CDD8A2C1E43BFB300621E92 /* expand.cc in Sources */,
> 				5CDD894A1E43BF3A00621E92 /* extended_jitter_report.cc in Sources */,
> 				5CDD894D1E43BF3A00621E92 /* extended_reports.cc in Sources */,
>+				41E02C8D212734B900C27CD6 /* video_coding_defines.cc in Sources */,
> 				5CDD8A971E43C00F00621E92 /* fake_decode_from_file.cc in Sources */,
> 				413A23101FE18E0700373E99 /* fakeclock.cc in Sources */,
>-				5C63F93F1E41737B002CA531 /* fakemetricsobserver.cc in Sources */,
> 				413A23CC1FE18E0800373E99 /* fakesslidentity.cc in Sources */,
> 				419C830C1FE20F3C0040C30F /* fec_controller_rplr_based.cc in Sources */,
> 				5CDD887B1E43BE3C00621E92 /* fec_test_helper.cc in Sources */,
>-				5CDD87BA1E43BC0500621E92 /* fft.c in Sources */,
> 				41433CEA1F79B33400387B4D /* fft.c in Sources */,
>-				5C4B4C661E431F9C002651C8 /* fft4g.c in Sources */,
> 				41F9BFD72051DDE500ABF0B9 /* fft_buffer.cc in Sources */,
> 				5C4B4CD31E4320A9002651C8 /* field_trial_default.cc in Sources */,
> 				413A22E61FE18E0700373E99 /* file.cc in Sources */,
> 				5C11A0211E457578004F0987 /* file_audio_device.cc in Sources */,
> 				5C11A01F1E457578004F0987 /* file_audio_device_factory.cc in Sources */,
>-				5C4B4CD41E4320A9002651C8 /* file_impl.cc in Sources */,
> 				413A23D81FE18E0800373E99 /* file_posix.cc in Sources */,
> 				5CDD8C8B1E43C66000621E92 /* file_utils.cc in Sources */,
> 				413A23DA1FE18E0800373E99 /* filerotatingstream.cc in Sources */,
> 				413A23B51FE18E0800373E99 /* fileutils.cc in Sources */,
> 				5CDD86551E43B8B500621E92 /* filter_ar.c in Sources */,
> 				5CDD86541E43B8B500621E92 /* filter_ar_fast_q12.c in Sources */,
>+				419241992127586500634FCF /* rnn_vad_weights.cc in Sources */,
> 				5CDD87BC1E43BC0500621E92 /* filter_functions.c in Sources */,
> 				5CDD86561E43B8B500621E92 /* filter_ma_fast_q12.c in Sources */,
>-				5CDD87BD1E43BC0500621E92 /* filterbank_tables.c in Sources */,
> 				41433D271F79B33400387B4D /* filterbank_tables.c in Sources */,
> 				41433D061F79B33400387B4D /* filterbanks.c in Sources */,
> 				5CDD87BF1E43BC0500621E92 /* filterbanks.c in Sources */,
> 				5CDD8BA01E43C2B500621E92 /* filtered_cb_vecs.c in Sources */,
>+				4192413D2127372400634FCF /* fft_util.cc in Sources */,
> 				41433D041F79B33400387B4D /* filters.c in Sources */,
> 				5C11A0031E457400004F0987 /* fine_audio_buffer.cc in Sources */,
> 				5CDD89501E43BF3A00621E92 /* fir.cc in Sources */,
> 				419C831B1FE242B20040C30F /* fir_filter_c.cc in Sources */,
> 				419C831D1FE242B20040C30F /* fir_filter_factory.cc in Sources */,
> 				5C4B4C6A1E431F9C002651C8 /* fir_filter_sse.cc in Sources */,
>+				4192424121276F0200634FCF /* VideoToolBoxDecoderFactory.cpp in Sources */,
> 				413A217D1FE18D7800373E99 /* flags.cc in Sources */,
> 				5CDD887E1E43BE3C00621E92 /* flexfec_header_reader_writer.cc in Sources */,
> 				5CD284B91E6A5F9F0094FDC8 /* flexfec_receive_stream_impl.cc in Sources */,
> 				5CDD88831E43BE3C00621E92 /* flexfec_receiver.cc in Sources */,
> 				5CDD88851E43BE3C00621E92 /* flexfec_sender.cc in Sources */,
>+				41DDB2592126792800296D47 /* zero_memory.cc in Sources */,
> 				5CDD88881E43BE3C00621E92 /* forward_error_correction.cc in Sources */,
> 				5CDD88861E43BE3C00621E92 /* forward_error_correction_internal.cc in Sources */,
> 				5CD285C91E6A63430094FDC8 /* frame_blocker.cc in Sources */,
>@@ -13124,24 +14082,23 @@
> 				5CDD83791E439A3500621E92 /* frame_dropper.cc in Sources */,
> 				5CDD8F901E43CBE000621E92 /* frame_length_controller.cc in Sources */,
> 				5CDD83D31E439A6F00621E92 /* frame_object.cc in Sources */,
>-				4140B8261E4E3383007409E6 /* g711.c in Sources */,
> 				4140B8241E4E3383007409E6 /* g711_interface.c in Sources */,
>-				4140B8351E4E3396007409E6 /* g722_decode.c in Sources */,
>-				4140B8371E4E3396007409E6 /* g722_encode.c in Sources */,
> 				4140B8381E4E3396007409E6 /* g722_interface.c in Sources */,
>-				5CDD90201E43CEDE00621E92 /* gain_applier.cc in Sources */,
>+				41F2635F21267ADF00274F59 /* builtin_video_decoder_factory.cc in Sources */,
> 				5CDD84B21E43AF1300621E92 /* gain_control_for_experimental_agc.cc in Sources */,
>+				4102F6A921273262006AE8D7 /* transport_feedback_packet_loss_tracker.cc in Sources */,
> 				5CDD84B41E43AF1300621E92 /* gain_control_impl.cc in Sources */,
>-				413092231EF8D6A200757C55 /* gain_controller2.cc in Sources */,
> 				5CDD8BA41E43C2B500621E92 /* gain_dequant.c in Sources */,
> 				5CDD8BA61E43C2B500621E92 /* gain_quant.c in Sources */,
>-				5CDD90221E43CEDE00621E92 /* gain_selector.cc in Sources */,
>+				4192413E2127372400634FCF /* lp_residual.cc in Sources */,
>+				41F2638D21267F4900274F59 /* g711.c in Sources */,
> 				5CDD83D51E439A6F00621E92 /* generic_decoder.cc in Sources */,
> 				5CDD83D71E439A6F00621E92 /* generic_encoder.cc in Sources */,
> 				5CDD8BA81E43C2B500621E92 /* get_cd_vec.c in Sources */,
> 				5CDD86571E43B8B500621E92 /* get_hanning_window.c in Sources */,
> 				5CDD8BAA1E43C2B500621E92 /* get_lsp_poly.c in Sources */,
> 				5CDD86581E43B8B500621E92 /* get_scaling_square.c in Sources */,
>+				419241C92127593F00634FCF /* bbr_factory.cc in Sources */,
> 				5CDD8BAC1E43C2B500621E92 /* get_sync_seq.c in Sources */,
> 				5CDD87291E43BABE00621E92 /* gmm.cc in Sources */,
> 				5CDD85581E43B42B00621E92 /* h264.cc in Sources */,
>@@ -13149,12 +14106,18 @@
> 				5CDD83561E43257200621E92 /* h264_common.cc in Sources */,
> 				4145E4BF1EF894F600FCF6E6 /* h264_profile_level_id.cc in Sources */,
> 				5CD285EA1E6A639F0094FDC8 /* h264_sprop_parameter_sets.cc in Sources */,
>+				4192418E212749C800634FCF /* transport_feedback_adapter.cc in Sources */,
> 				5CDD83D91E439A6F00621E92 /* h264_sps_pps_tracker.cc in Sources */,
>+				41F263BB2126811500274F59 /* sigslot.cc in Sources */,
> 				419C84691FE255FA0040C30F /* handshake.cc in Sources */,
>+				41299B912127367B00B3414B /* isac_vad.c in Sources */,
> 				419C84891FE2574D0040C30F /* handshake_client.cc in Sources */,
>+				41299B962127369100B3414B /* fake_network_pipe.cc in Sources */,
> 				419C848B1FE2574D0040C30F /* handshake_server.cc in Sources */,
> 				41ECEAD82064147C009D5141 /* helpers.cc in Sources */,
> 				413A236E1FE18E0700373E99 /* helpers.cc in Sources */,
>+				419241612127385B00634FCF /* rw_lock_wrapper.cc in Sources */,
>+				41E02CB22127352D00C27CD6 /* probe_bitrate_estimator.cc in Sources */,
> 				5CDD83DB1E439A6F00621E92 /* histogram.cc in Sources */,
> 				419C83511FE246650040C30F /* histogram_percentile_counter.cc in Sources */,
> 				5CDD8BAE1E43C2B500621E92 /* hp_input.c in Sources */,
>@@ -13164,6 +14127,7 @@
> 				5CDD90501E43D0E900621E92 /* i420.cc in Sources */,
> 				5CD284671E6A57F40094FDC8 /* i420_buffer.cc in Sources */,
> 				5C4B4C1B1E431F75002651C8 /* i420_buffer_pool.cc in Sources */,
>+				41924244212771B700634FCF /* objc_video_encoder_factory.mm in Sources */,
> 				413091F81EF8CFBD00757C55 /* iceserverparsing.cc in Sources */,
> 				419C83E01FE247F20040C30F /* icetransportinternal.cc in Sources */,
> 				413A23741FE18E0700373E99 /* ifaddrs_converter.cc in Sources */,
>@@ -13178,6 +14142,7 @@
> 				5CDD8A9A1E43C00F00621E92 /* input_audio_file.cc in Sources */,
> 				5CDD8FD51E43CD6600621E92 /* inter_arrival.cc in Sources */,
> 				5CDD83DD1E439A6F00621E92 /* inter_frame_delay.cc in Sources */,
>+				41F2637E21267B7700274F59 /* audio_options.cc in Sources */,
> 				5CD2852F1E6A61110094FDC8 /* internaldecoderfactory.cc in Sources */,
> 				5CD285311E6A61110094FDC8 /* internalencoderfactory.cc in Sources */,
> 				5CDD8BBF1E43C2B500621E92 /* interpolate.c in Sources */,
>@@ -13194,13 +14159,13 @@
> 				5CD284741E6A5D280094FDC8 /* jsepicecandidate.cc in Sources */,
> 				5CD284721E6A5D080094FDC8 /* jsepsessiondescription.cc in Sources */,
> 				41F9BFE32051DE1800ABF0B9 /* jseptransport.cc in Sources */,
>+				41F2639E2126800000274F59 /* aligned_malloc.cc in Sources */,
> 				413A23301FE18E0700373E99 /* json.cc in Sources */,
> 				5C4B4C701E431F9C002651C8 /* lapped_transform.cc in Sources */,
> 				5CDD87C41E43BC0500621E92 /* lattice.c in Sources */,
> 				41433CF21F79B33400387B4D /* lattice.c in Sources */,
> 				41433D141F79B33400387B4D /* lattice_c.c in Sources */,
> 				5CDD841F1E439B2900621E92 /* legacy_encoded_audio_frame.cc in Sources */,
>-				5CDD90271E43CEDE00621E92 /* level_controller.cc in Sources */,
> 				5CDD84B81E43AF1300621E92 /* level_estimator_impl.cc in Sources */,
> 				5CDD865A1E43B8B500621E92 /* levinson_durbin.c in Sources */,
> 				5CD284F61E6A60570094FDC8 /* localaudiosource.cc in Sources */,
>@@ -13208,6 +14173,7 @@
> 				5CDD8C0F1E43C34600621E92 /* locked_bandwidth_info.cc in Sources */,
> 				413A22CA1FE18E0700373E99 /* logging.cc in Sources */,
> 				413A22C01FE18E0700373E99 /* logging_mac.mm in Sources */,
>+				419241862127499300634FCF /* packet_queue_interface.cc in Sources */,
> 				413A23BE1FE18E0800373E99 /* logsinks.cc in Sources */,
> 				5CDD87051E43BA7500621E92 /* loudness_histogram.cc in Sources */,
> 				5CD2855D1E6A62ED0094FDC8 /* low_cut_filter.cc in Sources */,
>@@ -13215,14 +14181,18 @@
> 				5CDD8BC11E43C2B500621E92 /* lpc_encode.c in Sources */,
> 				5CDD87C71E43BC0500621E92 /* lpc_gain_swb_tables.c in Sources */,
> 				41433CEE1F79B33400387B4D /* lpc_masking_model.c in Sources */,
>+				419241F521275C3200634FCF /* vp8_encoder_simulcast_proxy.cc in Sources */,
> 				5CDD87C91E43BC0500621E92 /* lpc_shape_swb12_tables.c in Sources */,
> 				5CDD87CB1E43BC0500621E92 /* lpc_shape_swb16_tables.c in Sources */,
> 				5CDD87CD1E43BC0500621E92 /* lpc_tables.c in Sources */,
> 				41433D1E1F79B33400387B4D /* lpc_tables.c in Sources */,
> 				5CDD865B1E43B8B500621E92 /* lpc_to_refl_coef.c in Sources */,
> 				5CDD8BC31E43C2B500621E92 /* lsf_check.c in Sources */,
>+				419241522127376F00634FCF /* regatheringcontroller.cc in Sources */,
>+				41E02CB62127352D00C27CD6 /* goog_cc_network_control.cc in Sources */,
> 				5CDD8BC51E43C2B500621E92 /* lsf_interpolate_to_poly_dec.c in Sources */,
> 				5CDD8BC71E43C2B500621E92 /* lsf_interpolate_to_poly_enc.c in Sources */,
>+				4102F6E021273416006AE8D7 /* i010_buffer.cc in Sources */,
> 				5CDD8BC91E43C2B500621E92 /* lsf_to_lsp.c in Sources */,
> 				5CDD8BCB1E43C2B500621E92 /* lsf_to_poly.c in Sources */,
> 				5CDD8BCD1E43C2B500621E92 /* lsp_to_lsf.c in Sources */,
>@@ -13230,19 +14200,15 @@
> 				413A23381FE18E0700373E99 /* macutils.cc in Sources */,
> 				5CD285CB1E6A63430094FDC8 /* main_filter_update_gain.cc in Sources */,
> 				5CD285CF1E6A63430094FDC8 /* matched_filter.cc in Sources */,
>+				416D3BDE212731C200775F09 /* adaptive_mode_level_estimator.cc in Sources */,
> 				5CD285CD1E6A63430094FDC8 /* matched_filter_lag_aggregator.cc in Sources */,
> 				41F9BFD42051DDE500ABF0B9 /* matrix_buffer.cc in Sources */,
>-				413A23981FE18E0800373E99 /* md5.cc in Sources */,
>-				413A22B21FE18E0700373E99 /* md5digest.cc in Sources */,
> 				5CDD86A41E43B99400621E92 /* mean_variance_estimator.cc in Sources */,
>-				5CDD87681E43BB3000621E92 /* media_file_impl.cc in Sources */,
>-				5CDD876A1E43BB3000621E92 /* media_file_utility.cc in Sources */,
> 				5CDD83E51E439A6F00621E92 /* media_opt_util.cc in Sources */,
> 				5CDD83E71E439A6F00621E92 /* media_optimization.cc in Sources */,
> 				5C4B48E61E42C1E3002651C8 /* mediaconstants.cc in Sources */,
> 				5C63F9481E41737B002CA531 /* mediaconstraintsinterface.cc in Sources */,
> 				5C4B48E81E42C1E3002651C8 /* mediaengine.cc in Sources */,
>-				5CD2861C1E6A66380094FDC8 /* median_slope_estimator.cc in Sources */,
> 				5CD284FA1E6A60570094FDC8 /* mediastream.cc in Sources */,
> 				5CD286401E6A683A0094FDC8 /* mediastreaminterface.cc in Sources */,
> 				5CD284FC1E6A60570094FDC8 /* mediastreamobserver.cc in Sources */,
>@@ -13272,62 +14238,59 @@
> 				413A23A11FE18E0800373E99 /* nethelpers.cc in Sources */,
> 				413A23E61FE18E0800373E99 /* network.cc in Sources */,
> 				413A238B1FE18E0700373E99 /* networkmonitor.cc in Sources */,
>-				5CDD90291E43CEDE00621E92 /* noise_level_estimator.cc in Sources */,
>-				5CDD902B1E43CEDE00621E92 /* noise_spectrum_estimator.cc in Sources */,
> 				5CDD85FE1E43B84E00621E92 /* noise_suppression.c in Sources */,
>+				41A08BD421272EFA001D5D7B /* round_robin_packet_queue.cc in Sources */,
> 				5CDD84E91E43B0B600621E92 /* noise_suppression_impl.cc in Sources */,
> 				5CDD85FC1E43B84E00621E92 /* noise_suppression_x.c in Sources */,
>-				5CDD90821E43D15C00621E92 /* nonlinear_beamformer.cc in Sources */,
> 				413A235F1FE18E0700373E99 /* noop.cc in Sources */,
> 				413A23EE1FE18E0800373E99 /* noop.mm in Sources */,
>+				419241702127389B00634FCF /* quality_scaling_experiment.cc in Sources */,
> 				5CDD8A3D1E43BFB300621E92 /* normal.cc in Sources */,
>+				419241A02127588300634FCF /* audio_codec_pair_id.cc in Sources */,
> 				5CDD86A71E43B99400621E92 /* normalized_covariance_estimator.cc in Sources */,
> 				5CDD86001E43B84E00621E92 /* ns_core.c in Sources */,
>+				4102F69321273206006AE8D7 /* rtpparametersconversion.cc in Sources */,
>+				415F1FCE212730F000064CBF /* audio_frame.cc in Sources */,
>+				415F1FA621272FBA00064CBF /* adaptive_digital_gain_applier.cc in Sources */,
> 				41ECEAC220640F28009D5141 /* NSString+StdString.mm in Sources */,
> 				5CDD86051E43B84E00621E92 /* nsx_core.c in Sources */,
> 				5CDD86021E43B84E00621E92 /* nsx_core_c.c in Sources */,
> 				41F411AD1EF8D91F00343C26 /* null_aec_dump_factory.cc in Sources */,
>+				4192423D212768BD00634FCF /* objc_frame_buffer.mm in Sources */,
> 				419C83B31FE2472E0040C30F /* null_audio_poller.cc in Sources */,
> 				413A23D21FE18E0800373E99 /* nullsocketserver.cc in Sources */,
>-				41ECEAD320641458009D5141 /* objc_frame_buffer.mm in Sources */,
>-				41ECEADB206415EA009D5141 /* objc_video_decoder_factory.mm in Sources */,
>-				41ECEADC206415EE009D5141 /* objc_video_encoder_factory.mm in Sources */,
> 				5CDD86941E43B93900621E92 /* ooura_fft.cc in Sources */,
> 				5CDD86911E43B93900621E92 /* ooura_fft_sse2.cc in Sources */,
> 				419C84271FE24BD10040C30F /* openssladapter.cc in Sources */,
> 				419C84261FE24BCF0040C30F /* openssldigest.cc in Sources */,
> 				419C84251FE24BC80040C30F /* opensslidentity.cc in Sources */,
> 				419C843F1FE250B90040C30F /* opensslstreamadapter.cc in Sources */,
>-				419C82E91FE20E8E0040C30F /* optional.cc in Sources */,
> 				413A234A1FE18E0700373E99 /* optionsfile.cc in Sources */,
> 				5CDD8C671E43C60900621E92 /* opus_interface.c in Sources */,
>-				5CD285D11E6A63430094FDC8 /* output_selector.cc in Sources */,
> 				5CDD8FD71E43CD6600621E92 /* overuse_detector.cc in Sources */,
> 				5CDD8FD91E43CD6600621E92 /* overuse_estimator.cc in Sources */,
> 				5CDD85A41E43B5C000621E92 /* overuse_frame_detector.cc in Sources */,
> 				5C63F9ED1E4174F6002CA531 /* p2pconstants.cc in Sources */,
> 				5C63F9F21E4174F6002CA531 /* p2ptransportchannel.cc in Sources */,
>-				5C11A0141E4574C5004F0987 /* pa_ringbuffer.c in Sources */,
> 				5CDD8FA21E43CCBE00621E92 /* paced_sender.cc in Sources */,
> 				5CDD8BD31E43C2B500621E92 /* pack_bits.c in Sources */,
> 				5CDD8A421E43BFB300621E92 /* packet.cc in Sources */,
> 				5CDD83EE1E439A6F00621E92 /* packet.cc in Sources */,
>+				419242422127718D00634FCF /* video_encoder_factory.mm in Sources */,
> 				5CDD8AAF1E43C00F00621E92 /* packet.cc in Sources */,
> 				5CDD8A401E43BFB300621E92 /* packet_buffer.cc in Sources */,
> 				5CDD83EC1E439A6F00621E92 /* packet_buffer.cc in Sources */,
> 				5CDD888C1E43BE3C00621E92 /* packet_loss_stats.cc in Sources */,
> 				419C832C1FE245CD0040C30F /* packet_queue.cc in Sources */,
>-				419C832D1FE245CD0040C30F /* packet_queue2.cc in Sources */,
> 				5CDD8FA41E43CCBE00621E92 /* packet_router.cc in Sources */,
> 				5CDD8AAC1E43C00F00621E92 /* packet_source.cc in Sources */,
> 				412455561EF887FB00F11809 /* packetlossestimator.cc in Sources */,
> 				419C83F51FE248CA0040C30F /* packetsocketfactory.cc in Sources */,
> 				419C83DA1FE247F20040C30F /* packettransportinternal.cc in Sources */,
>+				416D3BE3212731C200775F09 /* fixed_gain_controller.cc in Sources */,
> 				413A233C1FE18E0700373E99 /* pathutils.cc in Sources */,
>-				5CDD85A71E43B5C000621E92 /* payload_router.cc in Sources */,
> 				5C4B48831E42C1BA002651C8 /* payload_type_mapper.cc in Sources */,
> 				5CDD90001E43CE3A00621E92 /* pcm16b.c in Sources */,
>-				5CDD902D1E43CEDE00621E92 /* peak_level_estimator.cc in Sources */,
> 				5CD284FF1E6A60570094FDC8 /* peerconnection.cc in Sources */,
> 				5CD2847F1E6A5E740094FDC8 /* peerconnectionfactory.cc in Sources */,
> 				413A233D1FE18E0700373E99 /* physicalsocketserver.cc in Sources */,
>@@ -13336,6 +14299,7 @@
> 				5CDD87D01E43BC0500621E92 /* pitch_estimator.c in Sources */,
> 				41433CF61F79B33400387B4D /* pitch_estimator_c.c in Sources */,
> 				5CDD87D21E43BC0500621E92 /* pitch_filter.c in Sources */,
>+				41F2636321267ADF00274F59 /* video_codec.cc in Sources */,
> 				41433D111F79B33400387B4D /* pitch_filter.c in Sources */,
> 				41433CF41F79B33400387B4D /* pitch_filter_c.c in Sources */,
> 				41433D101F79B33400387B4D /* pitch_gain_tables.c in Sources */,
>@@ -13356,13 +14320,11 @@
> 				5CDD8A451E43BFB300621E92 /* post_decode_vad.cc in Sources */,
> 				5CDD83591E43257200621E92 /* pps_parser.cc in Sources */,
> 				5CDD8A471E43BFB300621E92 /* preemptive_expand.cc in Sources */,
>-				5CDD8FB71E43CD0700621E92 /* probe_bitrate_estimator.cc in Sources */,
>-				5CDD8FB91E43CD0700621E92 /* probe_controller.cc in Sources */,
> 				5CDD84E51E43B04D00621E92 /* process_thread_impl.cc in Sources */,
>-				5CDD83F21E439A6F00621E92 /* protection_bitrate_calculator.cc in Sources */,
> 				419C83131FE242800040C30F /* proxy.cc in Sources */,
> 				413A23C21FE18E0800373E99 /* proxyinfo.cc in Sources */,
> 				413A22A71FE18E0600373E99 /* proxyserver.cc in Sources */,
>+				4192418C212749C800634FCF /* send_side_congestion_controller.cc in Sources */,
> 				5C63F9FE1E4174F6002CA531 /* pseudotcp.cc in Sources */,
> 				5CDD89581E43BF3A00621E92 /* psfb.cc in Sources */,
> 				5CDD87531E43BAF500621E92 /* push_resampler.cc in Sources */,
>@@ -13370,14 +14332,17 @@
> 				413092091EF8D11200757C55 /* qp_parser.cc in Sources */,
> 				5CDD83811E439A3500621E92 /* quality_scaler.cc in Sources */,
> 				5CD2862A1E6A66C80094FDC8 /* quality_threshold.cc in Sources */,
>+				41F2636521267ADF00274F59 /* video_decoder_software_fallback_wrapper.cc in Sources */,
> 				413A22FB1FE18E0700373E99 /* race_checker.cc in Sources */,
> 				413A23CB1FE18E0800373E99 /* random.cc in Sources */,
> 				5CDD8A4A1E43BFB300621E92 /* random_vector.cc in Sources */,
>+				4102F6AE212732E7006AE8D7 /* expand_uma_logger.cc in Sources */,
> 				5CDD865F1E43B8B500621E92 /* randomization_functions.c in Sources */,
> 				5CDD895B1E43BF3A00621E92 /* rapid_resync_request.cc in Sources */,
> 				413A238D1FE18E0700373E99 /* rate_limiter.cc in Sources */,
> 				413A23581FE18E0700373E99 /* rate_statistics.cc in Sources */,
>-				413A22EE1FE18E0700373E99 /* ratelimiter.cc in Sources */,
>+				4102F6E321273416006AE8D7 /* video_bitrate_allocation.cc in Sources */,
>+				41E02CC52127358800C27CD6 /* video_encoder_config.cc in Sources */,
> 				413A23621FE18E0700373E99 /* ratetracker.cc in Sources */,
> 				5CDD86611E43B8B500621E92 /* real_fft.c in Sources */,
> 				5C4B4C771E431F9C002651C8 /* real_fourier.cc in Sources */,
>@@ -13388,17 +14353,22 @@
> 				5CDD83F41E439A6F00621E92 /* receiver.cc in Sources */,
> 				5CDD895E1E43BF3A00621E92 /* receiver_report.cc in Sources */,
> 				5CDD8A4D1E43BFB300621E92 /* red_payload_splitter.cc in Sources */,
>+				41DDB265212679A300296D47 /* opensslutility.cc in Sources */,
> 				5CDD8BD91E43C2B500621E92 /* refiner.c in Sources */,
>+				419241902127581000634FCF /* fec_private_tables_bursty.cc in Sources */,
> 				5CDD86621E43B8B500621E92 /* refl_coef_to_lpc.c in Sources */,
> 				5C63FA011E4174F6002CA531 /* relayport.cc in Sources */,
> 				5C63FA041E4174F6002CA531 /* relayserver.cc in Sources */,
> 				5CDD89611E43BF3A00621E92 /* remb.cc in Sources */,
>+				41E02CC42127358800C27CD6 /* builtin_video_encoder_factory.cc in Sources */,
> 				5CDD8FDB1E43CD6600621E92 /* remote_bitrate_estimator_abs_send_time.cc in Sources */,
> 				5CDD8FDD1E43CD6600621E92 /* remote_bitrate_estimator_single_stream.cc in Sources */,
> 				5CDD8FE11E43CD6600621E92 /* remote_estimator_proxy.cc in Sources */,
>+				41DDB2562126790A00296D47 /* socket.cc in Sources */,
> 				5CDD88951E43BE3C00621E92 /* remote_ntp_time_estimator.cc in Sources */,
> 				5CD285051E6A60570094FDC8 /* remoteaudiosource.cc in Sources */,
> 				4145E4D91EF8CC9B00FCF6E6 /* render_buffer.cc in Sources */,
>+				419241C12127593F00634FCF /* data_transfer_tracker.cc in Sources */,
> 				5CD285D51E6A63430094FDC8 /* render_delay_buffer.cc in Sources */,
> 				5CD285D91E6A63430094FDC8 /* render_delay_controller.cc in Sources */,
> 				5CD285D71E6A63430094FDC8 /* render_delay_controller_metrics.cc in Sources */,
>@@ -13425,17 +14395,20 @@
> 				41F9BFAA2051C8E200ABF0B9 /* rtc_event_alr_state.cc in Sources */,
> 				419C83A21FE246F90040C30F /* rtc_event_audio_network_adaptation.cc in Sources */,
> 				419C83951FE246F90040C30F /* rtc_event_audio_playout.cc in Sources */,
>+				415F1FAC21272FBA00064CBF /* down_sampler.cc in Sources */,
> 				419C839B1FE246F90040C30F /* rtc_event_audio_receive_stream_config.cc in Sources */,
> 				419C839E1FE246F90040C30F /* rtc_event_audio_send_stream_config.cc in Sources */,
> 				419C838B1FE246F90040C30F /* rtc_event_bwe_update_delay_based.cc in Sources */,
> 				419C838D1FE246F90040C30F /* rtc_event_bwe_update_loss_based.cc in Sources */,
> 				5CDD904B1E43CF6100621E92 /* rtc_event_log.cc in Sources */,
>+				41F2636721267ADF00274F59 /* video_decoder.cc in Sources */,
> 				41F411B91EF8DA9600343C26 /* rtc_event_log_factory.cc in Sources */,
> 				419C84241FE24BA60040C30F /* rtc_event_log_output_file.cc in Sources */,
> 				419C838F1FE246F90040C30F /* rtc_event_probe_cluster_created.cc in Sources */,
> 				419C83991FE246F90040C30F /* rtc_event_probe_result_failure.cc in Sources */,
> 				419C83921FE246F90040C30F /* rtc_event_probe_result_success.cc in Sources */,
> 				419C83911FE246F90040C30F /* rtc_event_rtcp_packet_incoming.cc in Sources */,
>+				41E02C7C2127347400C27CD6 /* rtp_packet_to_send.cc in Sources */,
> 				419C839C1FE246F90040C30F /* rtc_event_rtcp_packet_outgoing.cc in Sources */,
> 				419C83241FE245280040C30F /* rtc_event_rtp_packet_incoming.cc in Sources */,
> 				419C83AA1FE247080040C30F /* rtc_event_rtp_packet_outgoing.cc in Sources */,
>@@ -13448,40 +14421,53 @@
> 				41ECEAC820641328009D5141 /* RTCEncodedImage.mm in Sources */,
> 				41F411B01EF8DA0100343C26 /* rtcerror.cc in Sources */,
> 				41ECEACF20641370009D5141 /* RTCI420Buffer.mm in Sources */,
>+				41E02CA92127352D00C27CD6 /* goog_cc_factory.cc in Sources */,
> 				5CDD8A4F1E43BFB300621E92 /* rtcp.cc in Sources */,
>+				419241DF21275A3000634FCF /* data_size.cc in Sources */,
> 				419C82B91FE20DCD0040C30F /* rtcp_demuxer.cc in Sources */,
> 				5CD285FA1E6A64520094FDC8 /* rtcp_nack_stats.cc in Sources */,
>+				415F1FB221272FBA00064CBF /* interpolated_gain_curve.cc in Sources */,
> 				5CDD88971E43BE3C00621E92 /* rtcp_packet.cc in Sources */,
> 				5CDD889A1E43BE3C00621E92 /* rtcp_receiver.cc in Sources */,
>+				415F1FAA21272FBA00064CBF /* compute_interpolated_gain_curve.cc in Sources */,
> 				5CDD889D1E43BE3C00621E92 /* rtcp_sender.cc in Sources */,
> 				419C82FF1FE20F020040C30F /* rtcp_transceiver.cc in Sources */,
>+				41924177212738C200634FCF /* transport.cc in Sources */,
> 				419C83021FE20F020040C30F /* rtcp_transceiver_config.cc in Sources */,
> 				419C83001FE20F020040C30F /* rtcp_transceiver_impl.cc in Sources */,
>+				41D6B45921273159008F9353 /* rtp_bitrate_configurator.cc in Sources */,
> 				41ECEADA206414A6009D5141 /* RTCRtpFragmentationHeader.mm in Sources */,
> 				5C63FA781E417AED002CA531 /* rtcstats.cc in Sources */,
> 				5C63FA771E417AED002CA531 /* rtcstats_objects.cc in Sources */,
>+				415F1FA821272FBA00064CBF /* adaptive_mode_level_estimator_agc.cc in Sources */,
> 				5CD285071E6A60570094FDC8 /* rtcstatscollector.cc in Sources */,
> 				5C63FA791E417AED002CA531 /* rtcstatsreport.cc in Sources */,
> 				41ECEAB720630108009D5141 /* RTCVideoCodec.mm in Sources */,
>+				41F2636E21267B3E00274F59 /* fft4g.c in Sources */,
> 				41ECEAB820630108009D5141 /* RTCVideoCodecH264.mm in Sources */,
> 				41ECEAC3206411E2009D5141 /* RTCVideoDecoderH264.mm in Sources */,
> 				41ECEAC4206411E5009D5141 /* RTCVideoEncoderH264.mm in Sources */,
> 				41ECEAEA20641812009D5141 /* RTCVideoEncoderSettings.mm in Sources */,
> 				41ECEACB2064134F009D5141 /* RTCVideoFrame.mm in Sources */,
>+				41F2638021267B7700274F59 /* datachannelinterface.cc in Sources */,
> 				41ECEAE4206416FF009D5141 /* RTCWrappedNativeVideoDecoder.mm in Sources */,
>+				419241B02127590200634FCF /* rtp_generic_frame_descriptor_extension.cc in Sources */,
> 				41ECEAE2206416FF009D5141 /* RTCWrappedNativeVideoEncoder.mm in Sources */,
> 				419C82B41FE20DCD0040C30F /* rtp_config.cc in Sources */,
> 				413092021EF8D0A600757C55 /* rtp_demuxer.cc in Sources */,
> 				5CDD88B01E43BE3C00621E92 /* rtp_format.cc in Sources */,
> 				5CDD88A41E43BE3C00621E92 /* rtp_format_h264.cc in Sources */,
> 				5CDD88A61E43BE3C00621E92 /* rtp_format_video_generic.cc in Sources */,
>-				419C83061FE20F020040C30F /* rtp_format_video_stereo.cc in Sources */,
> 				5CDD88AB1E43BE3C00621E92 /* rtp_format_vp8.cc in Sources */,
> 				5CDD88AE1E43BE3C00621E92 /* rtp_format_vp9.cc in Sources */,
> 				5CDD83F61E439A6F00621E92 /* rtp_frame_reference_finder.cc in Sources */,
>+				41E02CCD212735B700C27CD6 /* network_types.cc in Sources */,
> 				5CDD8AB81E43C00F00621E92 /* rtp_generator.cc in Sources */,
>+				416D3BEB212731C200775F09 /* signal_classifier.cc in Sources */,
> 				4145E4DC1EF8CCEF00FCF6E6 /* rtp_header_extension_map.cc in Sources */,
> 				5CDD88B51E43BE3C00621E92 /* rtp_header_extensions.cc in Sources */,
>+				41A08BC8212695DE001D5D7B /* bad_optional_access.cc in Sources */,
>+				419241A8212758D300634FCF /* spectral_features_internal.cc in Sources */,
> 				5CDD88B71E43BE3C00621E92 /* rtp_header_parser.cc in Sources */,
> 				41F9BF8B2051C80100ABF0B9 /* rtp_headers.cc in Sources */,
> 				5CDD88BE1E43BE3D00621E92 /* rtp_packet.cc in Sources */,
>@@ -13494,6 +14480,7 @@
> 				5CDD88C81E43BE3D00621E92 /* rtp_receiver_video.cc in Sources */,
> 				419C83CD1FE247B40040C30F /* rtp_rtcp_defines.cc in Sources */,
> 				419C82BF1FE20DCD0040C30F /* rtp_rtcp_demuxer_helper.cc in Sources */,
>+				4102F6A821273262006AE8D7 /* channel.cc in Sources */,
> 				5CDD88CC1E43BE3D00621E92 /* rtp_rtcp_impl.cc in Sources */,
> 				5CDD88D31E43BE3D00621E92 /* rtp_sender.cc in Sources */,
> 				5CDD88CE1E43BE3D00621E92 /* rtp_sender_audio.cc in Sources */,
>@@ -13502,7 +14489,9 @@
> 				5CDD85B21E43B5C000621E92 /* rtp_streams_synchronizer.cc in Sources */,
> 				5CD286271E6A669D0094FDC8 /* rtp_to_ntp_estimator.cc in Sources */,
> 				4130921B1EF8D63F00757C55 /* rtp_transport_controller_send.cc in Sources */,
>+				41E02CD42127360700C27CD6 /* sample_counter.cc in Sources */,
> 				5CDD88D51E43BE3D00621E92 /* rtp_utility.cc in Sources */,
>+				415F1FAF21272FBA00064CBF /* gain_applier.cc in Sources */,
> 				413091F41EF8CF9200757C55 /* rtp_video_stream_receiver.cc in Sources */,
> 				5C4B48EB1E42C1E3002651C8 /* rtpdataengine.cc in Sources */,
> 				5CDD896C1E43BF3A00621E92 /* rtpfb.cc in Sources */,
>@@ -13514,80 +14503,85 @@
> 				4145E4CC1EF8CB8B00FCF6E6 /* rtptransport.cc in Sources */,
> 				5C4B48F11E42C1E3002651C8 /* rtputils.cc in Sources */,
> 				5CDD83F81E439A6F00621E92 /* rtt_filter.cc in Sources */,
>+				41E02CB52127352D00C27CD6 /* alr_detector.cc in Sources */,
> 				419C82B71FE20DCD0040C30F /* rtx_receive_stream.cc in Sources */,
> 				41F9BFE62051DECD00ABF0B9 /* runtime_enabled_features_default.cc in Sources */,
>-				5C4B4CE41E4320A9002651C8 /* rw_lock.cc in Sources */,
>-				5C4B4CDE1E4320A9002651C8 /* rw_lock_posix.cc in Sources */,
>+				41D6B45721273159008F9353 /* receive_time_calculator.cc in Sources */,
> 				419C84701FE255FA0040C30F /* s3_both.cc in Sources */,
> 				419C84771FE255FA0040C30F /* s3_lib.cc in Sources */,
> 				419C84731FE255FA0040C30F /* s3_pkt.cc in Sources */,
>-				5CDD902F1E43CEDE00621E92 /* saturating_gain_estimator.cc in Sources */,
> 				5CFD53931E4BD3A300482908 /* scale_neon.cc in Sources */,
> 				5CFD53941E4BD3A300482908 /* scale_neon64.cc in Sources */,
> 				41F9BFB52051C93600ABF0B9 /* scopedvideodecoder.cc in Sources */,
>+				4102F6CB212733B7006AE8D7 /* video_stream_decoder_impl.cc in Sources */,
> 				41F9BFB92051C93600ABF0B9 /* scopedvideoencoder.cc in Sources */,
> 				5CD2863C1E6A681C0094FDC8 /* sctptransport.cc in Sources */,
> 				5CD2850D1E6A60570094FDC8 /* sctputils.cc in Sources */,
>+				41E02CB42127352D00C27CD6 /* probe_controller.cc in Sources */,
> 				5CDD896F1E43BF3A00621E92 /* sdes.cc in Sources */,
> 				419C82DC1FE20E590040C30F /* sdputils.cc in Sources */,
>+				41E02CA82127352D00C27CD6 /* acknowledged_bitrate_estimator.cc in Sources */,
> 				5CDD85B61E43B5C000621E92 /* send_delay_stats.cc in Sources */,
> 				5CDD90611E43D11200621E92 /* send_side_bandwidth_estimation.cc in Sources */,
> 				413092151EF8D5EE00757C55 /* send_side_congestion_controller.cc in Sources */,
> 				5CDD85B91E43B5C000621E92 /* send_statistics_proxy.cc in Sources */,
>-				5CDD8FE31E43CD6600621E92 /* send_time_history.cc in Sources */,
> 				5CDD89721E43BF3A00621E92 /* sender_report.cc in Sources */,
> 				413A239D1FE18E0800373E99 /* sequenced_task_checker_impl.cc in Sources */,
>-				5C63FA061E4174F6002CA531 /* session.cc in Sources */,
>+				419241412127372400634FCF /* pitch_search.cc in Sources */,
> 				5CDD83FB1E439A6F00621E92 /* session_info.cc in Sources */,
> 				41F9BFE42051DE1800ABF0B9 /* sessiondescription.cc in Sources */,
>-				413A23E71FE18E0800373E99 /* sha1.cc in Sources */,
>-				413A22F51FE18E0700373E99 /* sha1digest.cc in Sources */,
> 				5CD285DF1E6A63430094FDC8 /* shadow_filter_update_gain.cc in Sources */,
>-				5CDD90311E43CEDE00621E92 /* signal_classifier.cc in Sources */,
> 				413A235B1FE18E0700373E99 /* signalthread.cc in Sources */,
>-				413A23711FE18E0700373E99 /* sigslot.cc in Sources */,
>+				416D3BE9212731C200775F09 /* saturation_protector.cc in Sources */,
> 				5CDD8BDB1E43C2B500621E92 /* simple_interpolate_lsf.c in Sources */,
> 				5CDD8BDD1E43C2B500621E92 /* simple_lpc_analysis.c in Sources */,
> 				5CDD8BDF1E43C2B500621E92 /* simple_lsf_dequant.c in Sources */,
> 				5CDD8BE11E43C2B500621E92 /* simple_lsf_quant.c in Sources */,
> 				5C4B48861E42C1BA002651C8 /* simulcast.cc in Sources */,
>-				419C84381FE24E800040C30F /* simulcast_rate_allocator.cc in Sources */,
> 				5CDD875C1E43BAF500621E92 /* sinc_resampler.cc in Sources */,
> 				5CDD87E11E43BD7000621E92 /* sinc_resampler_sse.cc in Sources */,
> 				5CDD875E1E43BAF500621E92 /* sinusoidal_linear_chirp_source.cc in Sources */,
> 				5C4B4CE51E4320A9002651C8 /* sleep.cc in Sources */,
>+				419242432127719300634FCF /* video_decoder_factory.mm in Sources */,
> 				5CDD8BE51E43C2B500621E92 /* smooth.c in Sources */,
> 				5CDD8BE31E43C2B500621E92 /* smooth_out_data.c in Sources */,
> 				5CD285F61E6A63F60094FDC8 /* smoothing_filter.cc in Sources */,
> 				413A22F01FE18E0700373E99 /* socketadapters.cc in Sources */,
> 				413A23681FE18E0700373E99 /* socketaddress.cc in Sources */,
> 				413A234D1FE18E0700373E99 /* socketaddresspair.cc in Sources */,
>-				5C4B4D0B1E4322F6002651C8 /* socketmonitor.cc in Sources */,
> 				413A22D31FE18E0700373E99 /* socketstream.cc in Sources */,
> 				5CDD8BE71E43C2B500621E92 /* sort_sq.c in Sources */,
> 				5C4B4C7D1E431F9C002651C8 /* sparse_fir_filter.cc in Sources */,
>+				415F1FC62127308F00064CBF /* audio_state.cc in Sources */,
> 				41433D021F79B33400387B4D /* spectrum_ar_model_tables.c in Sources */,
> 				5CDD87D81E43BC0600621E92 /* spectrum_ar_model_tables.c in Sources */,
> 				5CDD866B1E43B8B500621E92 /* spl_init.c in Sources */,
>+				4102F6A621273262006AE8D7 /* channel_proxy.cc in Sources */,
>+				41E02CC02127356A00C27CD6 /* data_rate_limiter.cc in Sources */,
>+				41F2638521267B7700274F59 /* rtptransceiverinterface.cc in Sources */,
> 				5CDD866C1E43B8B500621E92 /* spl_inl.c in Sources */,
> 				5CDD86701E43B8B500621E92 /* spl_sqrt.c in Sources */,
>-				5CDD866F1E43B8B500621E92 /* spl_sqrt_floor.c in Sources */,
> 				5CDD8BE91E43C2B500621E92 /* split_vq.c in Sources */,
> 				5CDD86711E43B8B500621E92 /* splitting_filter.c in Sources */,
> 				5CDD84C11E43AF1300621E92 /* splitting_filter.cc in Sources */,
> 				5CDD835F1E43257200621E92 /* sps_parser.cc in Sources */,
> 				5CDD83621E43257200621E92 /* sps_vui_rewriter.cc in Sources */,
>+				4102F6DF21273416006AE8D7 /* video_stream_encoder_create.cc in Sources */,
> 				5CDD86721E43B8B500621E92 /* sqrt_of_one_minus_x_squared.c in Sources */,
> 				419C82DE1FE20E590040C30F /* srtpsession.cc in Sources */,
> 				419C82DF1FE20E590040C30F /* srtptransport.cc in Sources */,
> 				419C845E1FE255FA0040C30F /* ssl_aead_ctx.cc in Sources */,
> 				419C84741FE255FA0040C30F /* ssl_asn1.cc in Sources */,
>+				41A08BC021269553001D5D7B /* rtc_event_ice_candidate_pair.cc in Sources */,
>+				419241462127372800634FCF /* rnn.cc in Sources */,
> 				419C84601FE255FA0040C30F /* ssl_buffer.cc in Sources */,
> 				419C84751FE255FA0040C30F /* ssl_cert.cc in Sources */,
> 				419C84651FE255FA0040C30F /* ssl_cipher.cc in Sources */,
> 				419C84721FE255FA0040C30F /* ssl_file.cc in Sources */,
> 				419C84781FE255FA0040C30F /* ssl_key_share.cc in Sources */,
>+				41D6B45621273159008F9353 /* packet_receiver.cc in Sources */,
> 				419C84761FE255FA0040C30F /* ssl_lib.cc in Sources */,
>+				41A08BB0212681C8001D5D7B /* video_source_interface.cc in Sources */,
> 				419C846C1FE255FA0040C30F /* ssl_privkey.cc in Sources */,
> 				419C845F1FE255FA0040C30F /* ssl_session.cc in Sources */,
> 				419C84611FE255FA0040C30F /* ssl_stat.cc in Sources */,
>@@ -13598,16 +14592,20 @@
> 				413A23F11FE18E0800373E99 /* sslfingerprint.cc in Sources */,
> 				419C84161FE24A8D0040C30F /* sslidentity.cc in Sources */,
> 				419C84171FE24AA30040C30F /* sslstreamadapter.cc in Sources */,
>+				41F2637D21267B7700274F59 /* peerconnectioninterface.cc in Sources */,
> 				5CDD87361E43BABE00621E92 /* standalone_vad.cc in Sources */,
> 				5CDD8BEB1E43C2B500621E92 /* state_construct.c in Sources */,
> 				5CDD8BED1E43C2B500621E92 /* state_search.c in Sources */,
> 				5CDD8A511E43BFB300621E92 /* statistics_calculator.cc in Sources */,
> 				5CDD85BC1E43B5C000621E92 /* stats_counter.cc in Sources */,
> 				5CD2850F1E6A60570094FDC8 /* statscollector.cc in Sources */,
>+				41DDB25221265BE900296D47 /* optional.cc in Sources */,
> 				5C63F9721E41737B002CA531 /* statstypes.cc in Sources */,
> 				413A23D31FE18E0800373E99 /* stream.cc in Sources */,
>+				4192413A2127372400634FCF /* features_extraction.cc in Sources */,
> 				5CDD85BF1E43B5C000621E92 /* stream_synchronization.cc in Sources */,
> 				5C4B48F41E42C1E3002651C8 /* streamparams.cc in Sources */,
>+				41DDB262212679A300296D47 /* opensslsessioncache.cc in Sources */,
> 				413A23D61FE18E0800373E99 /* string_to_number.cc in Sources */,
> 				413A22B51FE18E0700373E99 /* stringencode.cc in Sources */,
> 				413A23131FE18E0700373E99 /* stringutils.cc in Sources */,
>@@ -13621,6 +14619,7 @@
> 				5CDD8BEF1E43C2B500621E92 /* swap_bytes.c in Sources */,
> 				5CDD8A541E43BFB300621E92 /* sync_buffer.cc in Sources */,
> 				5CD284BE1E6A5F9F0094FDC8 /* syncable.cc in Sources */,
>+				4102F6B321273331006AE8D7 /* mediachannel.cc in Sources */,
> 				419C846D1FE255FA0040C30F /* t1_enc.cc in Sources */,
> 				419C84791FE255FA0040C30F /* t1_lib.cc in Sources */,
> 				5CDD89781E43BF3A00621E92 /* target_bitrate.cc in Sources */,
>@@ -13629,6 +14628,7 @@
> 				5C63FA171E4174F6002CA531 /* tcpport.cc in Sources */,
> 				419C84341FE24E800040C30F /* temporal_layers.cc in Sources */,
> 				413A22A51FE18E0600373E99 /* testechoserver.cc in Sources */,
>+				4192417D2127392300634FCF /* timestamp_extrapolator.cc in Sources */,
> 				413A22E51FE18E0700373E99 /* thread.cc in Sources */,
> 				413A23501FE18E0700373E99 /* thread_checker_impl.cc in Sources */,
> 				413A23DD1FE18E0800373E99 /* thread_darwin.mm in Sources */,
>@@ -13636,15 +14636,19 @@
> 				5CDD8A571E43BFB300621E92 /* tick_timer.cc in Sources */,
> 				419C83351FE245EA0040C30F /* time_interval.cc in Sources */,
> 				5CDD8A5A1E43BFB300621E92 /* time_stretch.cc in Sources */,
>+				41A08BBD21269553001D5D7B /* rtc_event_ice_candidate_pair_config.cc in Sources */,
> 				5CDD88DA1E43BE3D00621E92 /* time_util.cc in Sources */,
>-				5C4B4CE91E4320A9002651C8 /* timestamp_extrapolator.cc in Sources */,
> 				5CDD83FD1E439A6F00621E92 /* timestamp_map.cc in Sources */,
> 				5CDD8A5D1E43BFB300621E92 /* timestamp_scaler.cc in Sources */,
> 				413A237B1FE18E0700373E99 /* timestampaligner.cc in Sources */,
>+				415F1FAD21272FBA00064CBF /* fixed_digital_level_estimator.cc in Sources */,
>+				415F1FBE2127304C00064CBF /* echo_control_mobile_proxy.cc in Sources */,
> 				413A23C31FE18E0800373E99 /* timeutils.cc in Sources */,
> 				5CDD83FF1E439A6F00621E92 /* timing.cc in Sources */,
> 				419C84681FE255FA0040C30F /* tls13_both.cc in Sources */,
> 				419C84671FE255FA0040C30F /* tls13_client.cc in Sources */,
>+				41E02CA52127352D00C27CD6 /* delay_based_bwe.cc in Sources */,
>+				41E02C8A212734B900C27CD6 /* fec_controller_default.cc in Sources */,
> 				419C84631FE255FA0040C30F /* tls13_enc.cc in Sources */,
> 				419C845D1FE255FA0040C30F /* tls13_server.cc in Sources */,
> 				419C846B1FE255FA0040C30F /* tls_method.cc in Sources */,
>@@ -13653,47 +14657,48 @@
> 				5CDD897D1E43BF3A00621E92 /* tmmbn.cc in Sources */,
> 				5CDD89801E43BF3A00621E92 /* tmmbr.cc in Sources */,
> 				5CDD88DC1E43BE3D00621E92 /* tmmbr_help.cc in Sources */,
>+				41F2638421267B7700274F59 /* jsepicecandidate.cc in Sources */,
> 				5CD285121E6A60570094FDC8 /* trackmediainfomap.cc in Sources */,
> 				41433D001F79B33400387B4D /* transform.c in Sources */,
>+				41E02C782127347400C27CD6 /* contributing_sources.cc in Sources */,
> 				5CDD87DB1E43BC0600621E92 /* transform.c in Sources */,
> 				41433D161F79B33400387B4D /* transform_tables.c in Sources */,
>-				413A23591FE18E0700373E99 /* transformadapter.cc in Sources */,
>+				41F263AC212680A800274F59 /* file_wrapper.cc in Sources */,
>+				41A08BCC21269620001D5D7B /* raw_logging.cc in Sources */,
> 				5CDD8C911E43C66000621E92 /* transient_detector.cc in Sources */,
> 				5CDD8C951E43C66000621E92 /* transient_suppressor.cc in Sources */,
> 				5CDD85C11E43B5C000621E92 /* transport_adapter.cc in Sources */,
> 				5CDD89831E43BF3A00621E92 /* transport_feedback.cc in Sources */,
> 				5CDD8FBB1E43CD0700621E92 /* transport_feedback_adapter.cc in Sources */,
>-				4130920F1EF8D40100757C55 /* transport_feedback_packet_loss_tracker.cc in Sources */,
>-				419C82E31FE20E590040C30F /* transportcontroller.cc in Sources */,
> 				5C63FA251E4174F6002CA531 /* transportdescription.cc in Sources */,
> 				5C63FA281E4174F6002CA531 /* transportdescriptionfactory.cc in Sources */,
>-				5CD286201E6A66380094FDC8 /* trendline_estimator.cc in Sources */,
> 				5C63FA2C1E4174F6002CA531 /* turnport.cc in Sources */,
>+				4102F6CC212733B7006AE8D7 /* video_send_stream_impl.cc in Sources */,
> 				41F9BFEA2051DEEA00ABF0B9 /* turnportfactory.cc in Sources */,
> 				5C63FA2F1E4174F6002CA531 /* turnserver.cc in Sources */,
> 				5C4B48F91E42C1E3002651C8 /* turnutils.cc in Sources */,
>+				41A08BD021272EE2001D5D7B /* gain_controller2.cc in Sources */,
> 				5CDD84C51E43AF1300621E92 /* typing_detection.cc in Sources */,
> 				419C83DC1FE247F20040C30F /* udptransport.cc in Sources */,
> 				5CDD88DF1E43BE3D00621E92 /* ulpfec_generator.cc in Sources */,
> 				5CDD88E21E43BE3D00621E92 /* ulpfec_header_reader_writer.cc in Sources */,
> 				5CDD88E41E43BE3D00621E92 /* ulpfec_receiver_impl.cc in Sources */,
>-				419C83141FE242800040C30F /* umametrics.cc in Sources */,
> 				5CDD8BF11E43C2B500621E92 /* unpack_bits.c in Sources */,
>+				41A08BD821272F1D001D5D7B /* congestion_window_pushback_controller.cc in Sources */,
> 				5CDD87DE1E43BC2E00621E92 /* utility.c in Sources */,
>-				419C84291FE24DEC0040C30F /* utility.cc in Sources */,
> 				5CDD87081E43BA7500621E92 /* utility.cc in Sources */,
> 				5CDD86E31E43BA2800621E92 /* vad.cc in Sources */,
> 				5CDD873A1E43BABE00621E92 /* vad_audio_proc.cc in Sources */,
> 				5CDD873D1E43BABE00621E92 /* vad_circular_buffer.cc in Sources */,
> 				5CDD86D61E43BA2800621E92 /* vad_core.c in Sources */,
> 				5CDD86D91E43BA2800621E92 /* vad_filterbank.c in Sources */,
>+				419241712127389B00634FCF /* congestion_controller_experiment.cc in Sources */,
> 				5CDD86DC1E43BA2800621E92 /* vad_gmm.c in Sources */,
> 				5CDD86DF1E43BA2800621E92 /* vad_sp.c in Sources */,
> 				41F9BFD52051DDE500ABF0B9 /* vector_buffer.cc in Sources */,
> 				5CDD86741E43B8B500621E92 /* vector_scaling_operations.c in Sources */,
> 				5CD285351E6A61590094FDC8 /* video_codec_initializer.cc in Sources */,
> 				5CDD84011E439A6F00621E92 /* video_coding_impl.cc in Sources */,
>-				419C82BB1FE20DCD0040C30F /* video_config.cc in Sources */,
> 				419C84111FE249AB0040C30F /* video_content_type.cc in Sources */,
> 				4145F6201FE1F38D00EB9CAF /* video_encoder.cc in Sources */,
> 				5CD2846A1E6A57F40094FDC8 /* video_frame.cc in Sources */,
>@@ -13703,8 +14708,11 @@
> 				419C82BE1FE20DCD0040C30F /* video_receive_stream.cc in Sources */,
> 				5CDD85CA1E43B5C000621E92 /* video_receive_stream.cc in Sources */,
> 				5CDD84031E439A6F00621E92 /* video_receiver.cc in Sources */,
>+				41E02CB12127352D00C27CD6 /* median_slope_estimator.cc in Sources */,
> 				5C4B4C201E431F75002651C8 /* video_render_frames.cc in Sources */,
>+				416D3BE6212731C200775F09 /* limiter.cc in Sources */,
> 				419C82B31FE20DCD0040C30F /* video_send_stream.cc in Sources */,
>+				41A08BC421269577001D5D7B /* icelogger.cc in Sources */,
> 				5CDD85CD1E43B5C000621E92 /* video_send_stream.cc in Sources */,
> 				5CDD84041E439A6F00621E92 /* video_sender.cc in Sources */,
> 				5CDD85CF1E43B5C000621E92 /* video_stream_decoder.cc in Sources */,
>@@ -13715,16 +14723,15 @@
> 				5C4B49021E42C1E3002651C8 /* videocapturer.cc in Sources */,
> 				5CD285141E6A60570094FDC8 /* videocapturertracksource.cc in Sources */,
> 				5C4B49061E42C1E3002651C8 /* videocommon.cc in Sources */,
>-				5CD286241E6A666D0094FDC8 /* videodecodersoftwarefallbackwrapper.cc in Sources */,
>-				5C4B48891E42C1BA002651C8 /* videoencodersoftwarefallbackwrapper.cc in Sources */,
> 				41ECEAFD20646664009D5141 /* VideoProcessingSoftLink.cpp in Sources */,
> 				5C4B490B1E42C1E3002651C8 /* videosourcebase.cc in Sources */,
>-				41F9BF8F2051C80100ABF0B9 /* videosourceinterface.cc in Sources */,
>+				419241D02127597D00634FCF /* simulated_network.cc in Sources */,
> 				5CD285161E6A60570094FDC8 /* videotrack.cc in Sources */,
> 				5CD285181E6A60570094FDC8 /* videotracksource.cc in Sources */,
> 				413A23121FE18E0700373E99 /* virtualsocketserver.cc in Sources */,
> 				5CDD87401E43BABE00621E92 /* voice_activity_detector.cc in Sources */,
> 				5CDD84C71E43AF1300621E92 /* voice_detection_impl.cc in Sources */,
>+				41D6B45321273159008F9353 /* call_config.cc in Sources */,
> 				5CFD53821E4BA4F500482908 /* voice_processing_audio_unit.mm in Sources */,
> 				5CDD89861E43BF3A00621E92 /* voip_metric.cc in Sources */,
> 				5CDD83851E439A3500621E92 /* vp8_header_parser.cc in Sources */,
>@@ -13889,6 +14896,8 @@
> 					Source/third_party/libsrtp/crypto/include/,
> 					Source/webrtc,
> 					Source/third_party/usrsctp,
>+					"Source/third_party/abseil-cpp",
>+					Source/third_party/rnnoise,
> 				);
> 				PRODUCT_NAME = "$(TARGET_NAME)";
> 			};
>@@ -13908,6 +14917,8 @@
> 					Source/third_party/libsrtp/crypto/include/,
> 					Source/webrtc,
> 					Source/third_party/usrsctp,
>+					"Source/third_party/abseil-cpp",
>+					Source/third_party/rnnoise,
> 				);
> 				OTHER_CFLAGS = "";
> 				PRODUCT_NAME = "$(TARGET_NAME)";
>@@ -13928,6 +14939,8 @@
> 					Source/third_party/libsrtp/crypto/include/,
> 					Source/webrtc,
> 					Source/third_party/usrsctp,
>+					"Source/third_party/abseil-cpp",
>+					Source/third_party/rnnoise,
> 				);
> 				OTHER_CFLAGS = "";
> 				PRODUCT_NAME = "$(TARGET_NAME)";
>@@ -14093,6 +15106,7 @@
> 					Source/webrtc/sdk/objc/Framework/Classes/Common,
> 					Source/webrtc/sdk/objc/Framework/Classes/Video,
> 					Source/webrtc/sdk/objc/Framework/Classes/PeerConnection,
>+					"Source/third_party/abseil-cpp",
> 				);
> 				OTHER_CFLAGS = "";
> 			};
>@@ -14145,6 +15159,7 @@
> 					Source/webrtc/sdk/objc/Framework/Classes/Common,
> 					Source/webrtc/sdk/objc/Framework/Classes/Video,
> 					Source/webrtc/sdk/objc/Framework/Classes/PeerConnection,
>+					"Source/third_party/abseil-cpp",
> 				);
> 			};
> 			name = Debug;
>@@ -14181,6 +15196,7 @@
> 					Source/webrtc/sdk/objc/Framework/Classes/Common,
> 					Source/webrtc/sdk/objc/Framework/Classes/Video,
> 					Source/webrtc/sdk/objc/Framework/Classes/PeerConnection,
>+					"Source/third_party/abseil-cpp",
> 				);
> 				OTHER_CFLAGS = "";
> 			};
>diff --git a/Source/WebCore/ChangeLog b/Source/WebCore/ChangeLog
>index 6eb7da9f613..c816535b1e9 100644
>--- a/Source/WebCore/ChangeLog
>+++ b/Source/WebCore/ChangeLog
>@@ -1,3 +1,50 @@
>+2018-08-20  Youenn Fablet  <youenn@apple.com>
>+
>+        Update libwebrtc up to 984f1a80c0
>+        https://bugs.webkit.org/show_bug.cgi?id=188745
>+
>+        Reviewed by NOBODY (OOPS!).
>+
>+        Covered by existing tests.
>+
>+        Updated implementation according new webrtc backend.
>+        Instead of modifying libwebrtc header files, we disable unused parameter warning for such headers included in WebCore.
>+        WebCore implementation is updated according new webrtc API, mostly AddRef/Release.
>+        RealtimeOutgoingVideoSource now uses a pixel convolver to convert RGB pixel buffers as I420.
>+
>+        * Configurations/Base.xcconfig:
>+        * Configurations/WebCoreTestSupport.xcconfig:
>+        * Modules/mediastream/libwebrtc/LibWebRTCDataChannelHandler.h:
>+        * Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp:
>+        (WebCore::LibWebRTCMediaEndpoint::addTrack):
>+        (WebCore::LibWebRTCMediaEndpoint::mediaStreamFromRTCStream):
>+        (WebCore::fillEncodingParameters):
>+        (WebCore::fillRtpParameters):
>+        * Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h:
>+        (WebCore::LibWebRTCMediaEndpoint::Release const):
>+        * WebCore.xcodeproj/project.pbxproj:
>+        * platform/mediastream/RealtimeIncomingAudioSource.h:
>+        * platform/mediastream/RealtimeIncomingVideoSource.h:
>+        * platform/mediastream/RealtimeOutgoingAudioSource.h:
>+        * platform/mediastream/RealtimeOutgoingVideoSource.h:
>+        * platform/mediastream/libwebrtc/LibWebRTCAudioModule.h:
>+        * platform/mediastream/libwebrtc/LibWebRTCProvider.cpp:
>+        * platform/mediastream/libwebrtc/LibWebRTCProvider.h:
>+        * platform/mediastream/mac/MockRealtimeVideoSourceMac.mm:
>+        (WebCore::MockRealtimeVideoSourceMac::CMSampleBufferFromPixelBuffer):
>+        * platform/mediastream/mac/RealtimeIncomingVideoSourceCocoa.mm:
>+        * platform/mediastream/mac/RealtimeOutgoingVideoSourceCocoa.cpp:
>+        (WebCore::ConvertToI420):
>+        (WebCore::RealtimeOutgoingVideoSourceCocoa::sampleBufferUpdated):
>+        * platform/mediastream/mac/RealtimeOutgoingVideoSourceCocoa.h:
>+        * platform/mediastream/mac/RealtimeOutgoingVideoSourceCocoa.mm: Copied from Source/WebCore/platform/mediastream/RealtimeIncomingVideoSource.h.
>+        (WebCore::RealtimeOutgoingVideoSourceCocoa::convertToYUV):
>+        * testing/MockLibWebRTCPeerConnection.cpp:
>+        (WebCore::MockLibWebRTCPeerConnection::AddTrack):
>+        * testing/MockLibWebRTCPeerConnection.h:
>+        (WebCore::MockRtpSender::GetParameters):
>+        (WebCore::MockRtpSender::SetParameters):
>+
> 2018-08-14  Antoine Quint  <graouts@apple.com>
> 
>         [Web Animations] Crash under AnimationTimeline::cancelOrRemoveDeclarativeAnimation()
>diff --git a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCDataChannelHandler.h b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCDataChannelHandler.h
>index 899c4ee1ad9..50bf77624a4 100644
>--- a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCDataChannelHandler.h
>+++ b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCDataChannelHandler.h
>@@ -28,8 +28,14 @@
> 
> #include "LibWebRTCMacros.h"
> #include "RTCDataChannelHandler.h"
>+
>+#pragma clang diagnostic push
>+#pragma clang diagnostic ignored "-Wunused-parameter"
>+
> #include <webrtc/api/datachannelinterface.h>
> 
>+#pragma clang diagnostic pop
>+
> namespace WebCore {
> 
> class RTCDataChannelHandlerClient;
>diff --git a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp
>index ac971961cef..af6a0b7ee51 100644
>--- a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp
>+++ b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.cpp
>@@ -197,27 +197,31 @@ void LibWebRTCMediaEndpoint::addTrack(RTCRtpSender& sender, MediaStreamTrack& tr
> {
>     ASSERT(m_backend);
> 
>-    std::vector<webrtc::MediaStreamInterface*> mediaStreams;
>-    rtc::scoped_refptr<webrtc::MediaStreamInterface> mediaStream = nullptr;
>-    if (mediaStreamIds.size()) {
>-        // libwebrtc is only using the first one if any.
>-        mediaStream = m_peerConnectionFactory.CreateLocalMediaStream(mediaStreamIds[0].utf8().data());
>-        mediaStreams.push_back(mediaStream.get());
>+    String mediaStreamId = mediaStreamIds.isEmpty() ? createCanonicalUUIDString() : mediaStreamIds[0];
>+    rtc::scoped_refptr<webrtc::MediaStreamInterface> mediaStream = m_localStreams.get(mediaStreamId);
>+    if (!mediaStream) {
>+        mediaStream = m_peerConnectionFactory.CreateLocalMediaStream(mediaStreamId.utf8().data());
>+        m_backend->AddStream(mediaStream);
>+        m_localStreams.add(mediaStreamId, mediaStream);
>     }
>-    
>+
>+    std::vector<std::string> ids;
>+    for (auto& id : mediaStreamIds)
>+        ids.push_back(id.utf8().data());
>+
>     switch (track.privateTrack().type()) {
>     case RealtimeMediaSource::Type::Audio: {
>         auto trackSource = RealtimeOutgoingAudioSource::create(track.privateTrack());
>         auto audioTrack = m_peerConnectionFactory.CreateAudioTrack(track.id().utf8().data(), trackSource.ptr());
>         m_peerConnectionBackend.addAudioSource(WTFMove(trackSource));
>-        m_senders.add(&sender, m_backend->AddTrack(audioTrack.get(), WTFMove(mediaStreams)));
>+        m_senders.add(&sender, m_backend->AddTrack(audioTrack.get(), WTFMove(ids)).MoveValue());
>         return;
>     }
>     case RealtimeMediaSource::Type::Video: {
>         auto videoSource = RealtimeOutgoingVideoSource::create(track.privateTrack());
>         auto videoTrack = m_peerConnectionFactory.CreateVideoTrack(track.id().utf8().data(), videoSource.ptr());
>         m_peerConnectionBackend.addVideoSource(WTFMove(videoSource));
>-        m_senders.add(&sender, m_backend->AddTrack(videoTrack.get(), WTFMove(mediaStreams)));
>+        m_senders.add(&sender, m_backend->AddTrack(videoTrack.get(), WTFMove(ids)).MoveValue());
>         return;
>     }
>     case RealtimeMediaSource::Type::None:
>@@ -619,7 +623,7 @@ void LibWebRTCMediaEndpoint::OnSignalingChange(webrtc::PeerConnectionInterface::
> MediaStream& LibWebRTCMediaEndpoint::mediaStreamFromRTCStream(webrtc::MediaStreamInterface& rtcStream)
> {
>     auto mediaStream = m_streams.ensure(&rtcStream, [&rtcStream, this] {
>-        auto label = rtcStream.label();
>+        auto label = rtcStream.id();
>         auto stream = MediaStream::create(*m_peerConnectionBackend.connection().scriptExecutionContext(), MediaStreamPrivate::create({ }, fromStdString(label)));
>         auto streamPointer = stream.ptr();
>         m_peerConnectionBackend.addRemoteStream(WTFMove(stream));
>@@ -968,7 +972,8 @@ static inline RTCRtpParameters::EncodingParameters fillEncodingParameters(const
>     if (rtcParameters.max_framerate)
>         parameters.maxFramerate = *rtcParameters.max_framerate;
>     parameters.rid = fromStdString(rtcParameters.rid);
>-    parameters.scaleResolutionDownBy = rtcParameters.scale_resolution_down_by;
>+    if (rtcParameters.scale_resolution_down_by)
>+        parameters.scaleResolutionDownBy = *rtcParameters.scale_resolution_down_by;
> 
>     return parameters;
> }
>@@ -1010,6 +1015,8 @@ static RTCRtpParameters fillRtpParameters(const webrtc::RtpParameters rtcParamet
>         parameters.codecs.append(fillCodecParameters(codec));
> 
>     switch (rtcParameters.degradation_preference) {
>+    // FIXME: Support DegradationPreference::DISABLED.
>+    case webrtc::DegradationPreference::DISABLED:
>     case webrtc::DegradationPreference::MAINTAIN_FRAMERATE:
>         parameters.degradationPreference = RTCRtpParameters::DegradationPreference::MaintainFramerate;
>         break;
>diff --git a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h
>index 297b60c185e..d409a1b23dd 100644
>--- a/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h
>+++ b/Source/WebCore/Modules/mediastream/libwebrtc/LibWebRTCMediaEndpoint.h
>@@ -32,10 +32,17 @@
> #include "RealtimeOutgoingAudioSource.h"
> #include "RealtimeOutgoingVideoSource.h"
> #include <Timer.h>
>+
>+#pragma clang diagnostic push
>+#pragma clang diagnostic ignored "-Wunused-parameter"
>+
> #include <webrtc/api/jsep.h>
> #include <webrtc/api/peerconnectioninterface.h>
> #include <webrtc/pc/peerconnectionfactory.h>
> #include <webrtc/pc/rtcstatscollector.h>
>+
>+#pragma clang diagnostic pop
>+
> #include <wtf/LoggerHelper.h>
> #include <wtf/ThreadSafeRefCounted.h>
> 
>@@ -129,8 +136,10 @@ private:
>     void AddRef() const { ref(); }
>     rtc::RefCountReleaseStatus Release() const
>     {
>+        auto result = refCount() - 1;
>         deref();
>-        return refCount() ? rtc::RefCountReleaseStatus::kDroppedLastRef : rtc::RefCountReleaseStatus::kOtherRefsRemained;
>+        return result ? rtc::RefCountReleaseStatus::kOtherRefsRemained
>+        : rtc::RefCountReleaseStatus::kDroppedLastRef;
>     }
> 
>     bool shouldOfferAllowToReceiveAudio() const;
>@@ -214,6 +223,8 @@ private:
>     bool m_isInitiator { false };
>     Timer m_statsLogTimer;
> 
>+    HashMap<String, rtc::scoped_refptr<webrtc::MediaStreamInterface>> m_localStreams;
>+
> #if !RELEASE_LOG_DISABLED
>     int64_t m_statsFirstDeliveredTimestamp { 0 };
>     Ref<const Logger> m_logger;
>diff --git a/Source/WebCore/WebCore.xcodeproj/project.pbxproj b/Source/WebCore/WebCore.xcodeproj/project.pbxproj
>index d34f8b00938..11f9e388c19 100644
>--- a/Source/WebCore/WebCore.xcodeproj/project.pbxproj
>+++ b/Source/WebCore/WebCore.xcodeproj/project.pbxproj
>@@ -1106,6 +1106,7 @@
> 		41885B9311B6FDA6003383BB /* FormSubmission.h in Headers */ = {isa = PBXBuildFile; fileRef = 41885B9111B6FDA6003383BB /* FormSubmission.h */; settings = {ATTRIBUTES = (Private, ); }; };
> 		418A06D0133C04D500CD379C /* EventDispatcher.h in Headers */ = {isa = PBXBuildFile; fileRef = 418A06CE133C04D500CD379C /* EventDispatcher.h */; };
> 		418F88050FF957AF0080F045 /* JSAbstractWorker.h in Headers */ = {isa = PBXBuildFile; fileRef = 418F88030FF957AE0080F045 /* JSAbstractWorker.h */; };
>+		419242492127B93E00634FCF /* RealtimeOutgoingVideoSourceCocoa.mm in Sources */ = {isa = PBXBuildFile; fileRef = 419242472127B7CC00634FCF /* RealtimeOutgoingVideoSourceCocoa.mm */; };
> 		419ACF921F97E7DA009F1A83 /* ServiceWorkerFetch.h in Headers */ = {isa = PBXBuildFile; fileRef = 419ACF8E1F97E7D5009F1A83 /* ServiceWorkerFetch.h */; settings = {ATTRIBUTES = (Private, ); }; };
> 		419BC2DF1685329900D64D6D /* VisitedLinkState.h in Headers */ = {isa = PBXBuildFile; fileRef = 419BC2DD1685329900D64D6D /* VisitedLinkState.h */; };
> 		419BE7591BC7F42B00E1C85B /* WebCoreBuiltinNames.h in Headers */ = {isa = PBXBuildFile; fileRef = 419BE7521BC7F3DB00E1C85B /* WebCoreBuiltinNames.h */; };
>@@ -7170,7 +7171,6 @@
> 		3FFFF9AC159D9B060020BBD5 /* ViewportStyleResolver.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = ViewportStyleResolver.h; sourceTree = "<group>"; };
> 		4107908A1FC3E4F20061B27A /* ClientOrigin.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = ClientOrigin.h; sourceTree = "<group>"; };
> 		410B7E711045FAB000D8224F /* JSMessageEventCustom.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = JSMessageEventCustom.cpp; sourceTree = "<group>"; };
>-		D2CDB5ED638F43AF86F07AA2 /* JSErrorEventCustom.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = JSErrorEventCustom.cpp; sourceTree = "<group>"; };
> 		41103AA71E39790A00769F03 /* RealtimeOutgoingAudioSource.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = RealtimeOutgoingAudioSource.cpp; sourceTree = "<group>"; };
> 		41103AA71E39790A00769F14 /* RealtimeOutgoingAudioSourceCocoa.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = RealtimeOutgoingAudioSourceCocoa.cpp; sourceTree = "<group>"; };
> 		41103AA81E39790A00769F03 /* RealtimeOutgoingAudioSource.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = RealtimeOutgoingAudioSource.h; sourceTree = "<group>"; };
>@@ -7287,6 +7287,7 @@
> 		418C395F1C8F0AAB0051C8A3 /* ReadableStreamDefaultController.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = ReadableStreamDefaultController.h; sourceTree = "<group>"; };
> 		418F88020FF957AE0080F045 /* JSAbstractWorker.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = JSAbstractWorker.cpp; sourceTree = "<group>"; };
> 		418F88030FF957AE0080F045 /* JSAbstractWorker.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = JSAbstractWorker.h; sourceTree = "<group>"; };
>+		419242472127B7CC00634FCF /* RealtimeOutgoingVideoSourceCocoa.mm */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.objcpp; path = RealtimeOutgoingVideoSourceCocoa.mm; sourceTree = "<group>"; };
> 		41945694203502A5004BA277 /* ApplicationCacheResourceLoader.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = ApplicationCacheResourceLoader.cpp; sourceTree = "<group>"; };
> 		41945696203502A6004BA277 /* ApplicationCacheResourceLoader.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = ApplicationCacheResourceLoader.h; sourceTree = "<group>"; };
> 		419ACF8E1F97E7D5009F1A83 /* ServiceWorkerFetch.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = ServiceWorkerFetch.h; sourceTree = "<group>"; };
>@@ -9256,7 +9257,6 @@
> 		71EFCEDA202B388F00D7C411 /* AnimationEffectReadOnly.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = AnimationEffectReadOnly.cpp; sourceTree = "<group>"; };
> 		71EFCEDB202B389000D7C411 /* KeyframeEffectReadOnly.idl */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = text; path = KeyframeEffectReadOnly.idl; sourceTree = "<group>"; };
> 		71EFCEDE202B39C700D7C411 /* JSAnimationEffectReadOnlyCustom.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = JSAnimationEffectReadOnlyCustom.cpp; sourceTree = "<group>"; };
>-		A4A69B8BB91B49D0A804C31D /* JSPromiseRejectionEventCustom.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = JSPromiseRejectionEventCustom.cpp; sourceTree = "<group>"; };
> 		71EFCEDF202B3AA100D7C411 /* JSKeyframeEffectReadOnly.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = JSKeyframeEffectReadOnly.h; sourceTree = "<group>"; };
> 		71EFCEE0202B3AA300D7C411 /* JSKeyframeEffectReadOnly.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = JSKeyframeEffectReadOnly.cpp; sourceTree = "<group>"; };
> 		71F936F71DD4F99B00922CC7 /* tracks-support.js */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.javascript; path = "tracks-support.js"; sourceTree = "<group>"; };
>@@ -11178,6 +11178,7 @@
> 		A454424D119B3687009BE912 /* RenderMeter.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = RenderMeter.h; sourceTree = "<group>"; };
> 		A456FA2411AD4A830020B420 /* LabelsNodeList.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = LabelsNodeList.cpp; sourceTree = "<group>"; };
> 		A456FA2511AD4A830020B420 /* LabelsNodeList.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = LabelsNodeList.h; sourceTree = "<group>"; };
>+		A4A69B8BB91B49D0A804C31D /* JSPromiseRejectionEventCustom.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = JSPromiseRejectionEventCustom.cpp; sourceTree = "<group>"; };
> 		A501920C132EBF2E008BFE55 /* Autocapitalize.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = Autocapitalize.h; sourceTree = "<group>"; };
> 		A5071E821C56D079009951BE /* ResourceUsageData.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = ResourceUsageData.h; sourceTree = "<group>"; };
> 		A5071E831C56D079009951BE /* ResourceUsageThread.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = ResourceUsageThread.cpp; sourceTree = "<group>"; };
>@@ -13580,6 +13581,7 @@
> 		D0EDA773143E303C0028E383 /* CachedRawResource.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = CachedRawResource.h; sourceTree = "<group>"; };
> 		D0FF2A5B11F8C45A007E74E0 /* PingLoader.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = PingLoader.cpp; sourceTree = "<group>"; };
> 		D0FF2A5C11F8C45A007E74E0 /* PingLoader.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = PingLoader.h; sourceTree = "<group>"; };
>+		D2CDB5ED638F43AF86F07AA2 /* JSErrorEventCustom.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = JSErrorEventCustom.cpp; sourceTree = "<group>"; };
> 		D302754512A5FE84004BD828 /* RenderDetailsMarker.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = RenderDetailsMarker.cpp; sourceTree = "<group>"; };
> 		D302754612A5FE84004BD828 /* RenderDetailsMarker.h */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.c.h; path = RenderDetailsMarker.h; sourceTree = "<group>"; };
> 		D359D786129CA2710006E5D2 /* HTMLDetailsElement.cpp */ = {isa = PBXFileReference; fileEncoding = 4; lastKnownFileType = sourcecode.cpp.cpp; path = HTMLDetailsElement.cpp; sourceTree = "<group>"; };
>@@ -15324,6 +15326,7 @@
> 				41103AA71E39790A00769F14 /* RealtimeOutgoingAudioSourceCocoa.cpp */,
> 				41103AA81E39790A00769F14 /* RealtimeOutgoingAudioSourceCocoa.h */,
> 				5CDD833B1E4324BB00621B83 /* RealtimeOutgoingVideoSourceCocoa.cpp */,
>+				419242472127B7CC00634FCF /* RealtimeOutgoingVideoSourceCocoa.mm */,
> 				5CDD833C1E4324BB00621B83 /* RealtimeOutgoingVideoSourceCocoa.h */,
> 				070A9F5E1FFECC70003DF649 /* ScreenDisplayCaptureSourceMac.h */,
> 				070A9F601FFECC71003DF649 /* ScreenDisplayCaptureSourceMac.mm */,
>@@ -31728,6 +31731,7 @@
> 				538EC8A11F993F9D004D22A8 /* UnifiedSource48.cpp in Sources */,
> 				DE5F85991FA1ABF4006DB63A /* UnifiedSource49-mm.mm in Sources */,
> 				538EC8A21F993F9D004D22A8 /* UnifiedSource49.cpp in Sources */,
>+				419242492127B93E00634FCF /* RealtimeOutgoingVideoSourceCocoa.mm in Sources */,
> 				DE5F859A1FA1ABF4006DB63A /* UnifiedSource50-mm.mm in Sources */,
> 				538EC8A31F993F9D004D22A8 /* UnifiedSource50.cpp in Sources */,
> 				DE5F86501FA2AF24006DB63A /* UnifiedSource51-mm.mm in Sources */,
>diff --git a/Source/WebCore/platform/mediastream/RealtimeIncomingAudioSource.h b/Source/WebCore/platform/mediastream/RealtimeIncomingAudioSource.h
>index 4c363b62daa..cbf5730ac94 100644
>--- a/Source/WebCore/platform/mediastream/RealtimeIncomingAudioSource.h
>+++ b/Source/WebCore/platform/mediastream/RealtimeIncomingAudioSource.h
>@@ -34,7 +34,14 @@
> 
> #include "LibWebRTCMacros.h"
> #include "RealtimeMediaSource.h"
>+
>+#pragma clang diagnostic push
>+#pragma clang diagnostic ignored "-Wunused-parameter"
>+
> #include <webrtc/api/mediastreaminterface.h>
>+
>+#pragma clang diagnostic pop
>+
> #include <wtf/RetainPtr.h>
> 
> namespace WebCore {
>diff --git a/Source/WebCore/platform/mediastream/RealtimeIncomingVideoSource.h b/Source/WebCore/platform/mediastream/RealtimeIncomingVideoSource.h
>index 85fbfa7e122..4d7cad29a1a 100644
>--- a/Source/WebCore/platform/mediastream/RealtimeIncomingVideoSource.h
>+++ b/Source/WebCore/platform/mediastream/RealtimeIncomingVideoSource.h
>@@ -34,7 +34,14 @@
> 
> #include "LibWebRTCMacros.h"
> #include "RealtimeMediaSource.h"
>+
>+#pragma clang diagnostic push
>+#pragma clang diagnostic ignored "-Wunused-parameter"
>+
> #include <webrtc/api/mediastreaminterface.h>
>+
>+#pragma clang diagnostic pop
>+
> #include <wtf/RetainPtr.h>
> 
> namespace WebCore {
>diff --git a/Source/WebCore/platform/mediastream/RealtimeOutgoingAudioSource.h b/Source/WebCore/platform/mediastream/RealtimeOutgoingAudioSource.h
>index 76a61f8f644..ce2e5b1d9e0 100644
>--- a/Source/WebCore/platform/mediastream/RealtimeOutgoingAudioSource.h
>+++ b/Source/WebCore/platform/mediastream/RealtimeOutgoingAudioSource.h
>@@ -34,7 +34,14 @@
> #include "LibWebRTCMacros.h"
> #include "MediaStreamTrackPrivate.h"
> #include "Timer.h"
>+
>+#pragma clang diagnostic push
>+#pragma clang diagnostic ignored "-Wunused-parameter"
>+
> #include <webrtc/api/mediastreaminterface.h>
>+
>+#pragma clang diagnostic pop
>+
> #include <wtf/ThreadSafeRefCounted.h>
> 
> namespace webrtc {
>diff --git a/Source/WebCore/platform/mediastream/RealtimeOutgoingVideoSource.h b/Source/WebCore/platform/mediastream/RealtimeOutgoingVideoSource.h
>index 029d1478274..c1aa9d11395 100644
>--- a/Source/WebCore/platform/mediastream/RealtimeOutgoingVideoSource.h
>+++ b/Source/WebCore/platform/mediastream/RealtimeOutgoingVideoSource.h
>@@ -33,10 +33,15 @@
> #include "LibWebRTCMacros.h"
> #include "MediaStreamTrackPrivate.h"
> #include <Timer.h>
>+
>+#pragma clang diagnostic push
>+#pragma clang diagnostic ignored "-Wunused-parameter"
>+
> #include <webrtc/api/mediastreaminterface.h>
>-#include <webrtc/api/optional.h>
> #include <webrtc/common_video/include/i420_buffer_pool.h>
>-#include <webrtc/api/videosinkinterface.h>
>+
>+#pragma clang diagnostic pop
>+
> #include <wtf/Optional.h>
> #include <wtf/ThreadSafeRefCounted.h>
> 
>@@ -54,9 +59,8 @@ public:
>     void AddRef() const final { ref(); }
>     rtc::RefCountReleaseStatus Release() const final
>     {
>-        auto result = hasOneRef() ? rtc::RefCountReleaseStatus::kDroppedLastRef : rtc::RefCountReleaseStatus::kOtherRefsRemained;
>         deref();
>-        return result;
>+        return rtc::RefCountReleaseStatus::kOtherRefsRemained;
>     }
> 
>     void setApplyRotation(bool shouldApplyRotation) { m_shouldApplyRotation = shouldApplyRotation; }
>@@ -88,7 +92,7 @@ private:
> 
>     // VideoTrackSourceInterface API
>     bool is_screencast() const final { return false; }
>-    rtc::Optional<bool> needs_denoising() const final { return rtc::Optional<bool>(); }
>+    absl::optional<bool> needs_denoising() const final { return absl::optional<bool>(); }
>     bool GetStats(Stats*) final { return false; };
> 
>     // MediaSourceInterface API
>diff --git a/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCAudioModule.h b/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCAudioModule.h
>index 9875c4fa61d..71ca2ab5dba 100644
>--- a/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCAudioModule.h
>+++ b/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCAudioModule.h
>@@ -28,9 +28,15 @@
> #if USE(LIBWEBRTC)
> 
> #include "LibWebRTCMacros.h"
>+
>+#pragma clang diagnostic push
>+#pragma clang diagnostic ignored "-Wunused-parameter"
>+
>+#include <webrtc/modules/audio_device/include/audio_device.h>
> #include <webrtc/rtc_base/messagehandler.h>
> #include <webrtc/rtc_base/thread.h>
>-#include <webrtc/modules/audio_device/include/audio_device.h>
>+
>+#pragma clang diagnostic pop
> 
> namespace WebCore {
> 
>@@ -47,7 +53,7 @@ private:
>         return value;
>     }
> 
>-    void AddRef() const final { return; }
>+    void AddRef() const final { }
>     rtc::RefCountReleaseStatus Release() const final { return rtc::RefCountReleaseStatus::kOtherRefsRemained; }
>     void OnMessage(rtc::Message*);
> 
>diff --git a/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCProvider.cpp b/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCProvider.cpp
>index f2363ba0ce4..b4bd0a45138 100644
>--- a/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCProvider.cpp
>+++ b/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCProvider.cpp
>@@ -30,6 +30,10 @@
> #include "LibWebRTCAudioModule.h"
> #include "Logging.h"
> #include <dlfcn.h>
>+
>+#pragma clang diagnostic push
>+#pragma clang diagnostic ignored "-Wunused-parameter"
>+
> #include <webrtc/api/audio_codecs/builtin_audio_decoder_factory.h>
> #include <webrtc/api/audio_codecs/builtin_audio_encoder_factory.h>
> #include <webrtc/api/peerconnectionfactoryproxy.h>
>@@ -38,6 +42,9 @@
> #include <webrtc/p2p/client/basicportallocator.h>
> #include <webrtc/pc/peerconnectionfactory.h>
> #include <webrtc/rtc_base/physicalsocketserver.h>
>+
>+#pragma clang diagnostic pop
>+
> #include <wtf/Function.h>
> #include <wtf/NeverDestroyed.h>
> #endif
>diff --git a/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCProvider.h b/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCProvider.h
>index 0b4b9bd5b3a..6991cde0aaf 100644
>--- a/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCProvider.h
>+++ b/Source/WebCore/platform/mediastream/libwebrtc/LibWebRTCProvider.h
>@@ -35,12 +35,15 @@
> 
> #if USE(LIBWEBRTC)
> 
>+#pragma clang diagnostic push
>+#pragma clang diagnostic ignored "-Wunused-parameter"
>+
> #include <webrtc/api/peerconnectioninterface.h>
>-#include <webrtc/api/video_codecs/video_decoder_factory.h>
>-#include <webrtc/api/video_codecs/video_encoder_factory.h>
>-#include <webrtc/rtc_base/scoped_ref_ptr.h>
> #include <webrtc/media/engine/webrtcvideodecoderfactory.h>
> #include <webrtc/media/engine/webrtcvideoencoderfactory.h>
>+#include <webrtc/rtc_base/scoped_ref_ptr.h>
>+
>+#pragma clang diagnostic pop
> 
> namespace rtc {
> class NetworkManager;
>diff --git a/Source/WebCore/platform/mediastream/mac/MockRealtimeVideoSourceMac.mm b/Source/WebCore/platform/mediastream/mac/MockRealtimeVideoSourceMac.mm
>index 2ad4d0133e9..7e7ca7d72e6 100644
>--- a/Source/WebCore/platform/mediastream/mac/MockRealtimeVideoSourceMac.mm
>+++ b/Source/WebCore/platform/mediastream/mac/MockRealtimeVideoSourceMac.mm
>@@ -70,7 +70,7 @@ RetainPtr<CMSampleBufferRef> MockRealtimeVideoSourceMac::CMSampleBufferFromPixel
> {
>     if (!pixelBuffer)
>         return nullptr;
>-
>+    
>     if (!m_pixelBufferConformer)
>         m_pixelBufferConformer = std::make_unique<PixelBufferConformerCV>((__bridge CFDictionaryRef)@{ (__bridge NSString *)kCVPixelBufferPixelFormatTypeKey: @(kCVPixelFormatType_420YpCbCr8Planar) });
> 
>diff --git a/Source/WebCore/platform/mediastream/mac/RealtimeIncomingVideoSourceCocoa.mm b/Source/WebCore/platform/mediastream/mac/RealtimeIncomingVideoSourceCocoa.mm
>index 9e201eb58f9..4b425f9ba62 100644
>--- a/Source/WebCore/platform/mediastream/mac/RealtimeIncomingVideoSourceCocoa.mm
>+++ b/Source/WebCore/platform/mediastream/mac/RealtimeIncomingVideoSourceCocoa.mm
>@@ -33,7 +33,14 @@
> #include "Logging.h"
> #include "MediaSampleAVFObjC.h"
> #include <pal/cf/CoreMediaSoftLink.h>
>+
>+#pragma clang diagnostic push
>+#pragma clang diagnostic ignored "-Wunused-parameter"
>+
> #include <webrtc/sdk/WebKit/WebKitUtilities.h>
>+
>+#pragma clang diagnostic pop
>+
> #include <wtf/cf/TypeCastsCF.h>
> 
> #include "CoreVideoSoftLink.h"
>diff --git a/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingVideoSourceCocoa.cpp b/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingVideoSourceCocoa.cpp
>index 439a1bf1579..cf25fd83f1f 100644
>--- a/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingVideoSourceCocoa.cpp
>+++ b/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingVideoSourceCocoa.cpp
>@@ -29,10 +29,17 @@
> #if USE(LIBWEBRTC)
> 
> #include "Logging.h"
>+
>+#pragma clang diagnostic push
>+#pragma clang diagnostic ignored "-Wunused-parameter"
>+
> #include <webrtc/api/video/i420_buffer.h>
> #include <webrtc/common_video/libyuv/include/webrtc_libyuv.h>
> #include <webrtc/sdk/WebKit/WebKitUtilities.h>
> #include <webrtc/sdk/objc/Framework/Classes/Video/corevideo_frame_buffer.h>
>+
>+#pragma clang diagnostic pop
>+
> #include <pal/cf/CoreMediaSoftLink.h>
> #include "CoreVideoSoftLink.h"
> 
>@@ -50,7 +57,8 @@ typedef enum RotationMode {
>   kRotateCounterClockwise = 270,
> } RotationModeEnum;
> 
>-int ConvertToI420(const uint8_t* src_frame,
>+int ConvertToI420(webrtc::VideoType src_video_type,
>+                  const uint8_t* src_frame,
>                   size_t src_size,
>                   uint8_t* dst_y,
>                   int dst_stride_y,
>@@ -64,8 +72,7 @@ int ConvertToI420(const uint8_t* src_frame,
>                   int src_height,
>                   int crop_width,
>                   int crop_height,
>-                  RotationMode rotation,
>-                  uint32_t format);
>+                  RotationMode rotation);
> }
> }
> 
>@@ -103,7 +110,7 @@ static inline int ConvertToI420(webrtc::VideoType src_video_type,
>   if (rotation == libyuv::kRotate90 || rotation == libyuv::kRotate270) {
>     std::swap(dst_width, dst_height);
>   }
>-  return libyuv::ConvertToI420(
>+  return libyuv::ConvertToI420(src_video_type,
>       src_frame, sample_size,
>       dst_buffer->MutableDataY(), dst_buffer->StrideY(),
>       dst_buffer->MutableDataU(), dst_buffer->StrideU(),
>@@ -111,8 +118,7 @@ static inline int ConvertToI420(webrtc::VideoType src_video_type,
>       crop_x, crop_y,
>       src_width, src_height,
>       dst_width, dst_height,
>-      rotation,
>-      webrtc::ConvertVideoType(src_video_type));
>+      rotation);
> }
> 
> void RealtimeOutgoingVideoSourceCocoa::sampleBufferUpdated(MediaStreamTrackPrivate&, MediaSample& sample)
>@@ -147,41 +153,23 @@ void RealtimeOutgoingVideoSourceCocoa::sampleBufferUpdated(MediaStreamTrackPriva
>     auto pixelBuffer = static_cast<CVPixelBufferRef>(CMSampleBufferGetImageBuffer(sample.platformSample().sample.cmSampleBuffer));
>     auto pixelFormatType = CVPixelBufferGetPixelFormatType(pixelBuffer);
> 
>-    if (pixelFormatType == kCVPixelFormatType_420YpCbCr8Planar || pixelFormatType == kCVPixelFormatType_420YpCbCr8BiPlanarFullRange) {
>-        rtc::scoped_refptr<webrtc::VideoFrameBuffer> buffer = webrtc::pixelBufferToFrame(pixelBuffer);
>-        if (m_shouldApplyRotation && m_currentRotation != webrtc::kVideoRotation_0) {
>-            // FIXME: We should make AVVideoCaptureSource handle the rotation whenever possible.
>-            // This implementation is inefficient, we should rotate on the CMSampleBuffer directly instead of doing this double allocation.
>-            auto rotatedBuffer = buffer->ToI420();
>-            ASSERT(rotatedBuffer);
>-            buffer = webrtc::I420Buffer::Rotate(*rotatedBuffer, m_currentRotation);
>-        }
>-        sendFrame(WTFMove(buffer));
>-        return;
>+    rtc::scoped_refptr<webrtc::VideoFrameBuffer> buffer;
>+    RetainPtr<CVPixelBufferRef> convertedBuffer;
>+    if (pixelFormatType == kCVPixelFormatType_420YpCbCr8Planar || pixelFormatType == kCVPixelFormatType_420YpCbCr8BiPlanarFullRange)
>+        buffer = webrtc::pixelBufferToFrame(pixelBuffer);
>+    else {
>+        convertedBuffer = convertToYUV(pixelBuffer);
>+        buffer = webrtc::pixelBufferToFrame(convertedBuffer.get());
>     }
> 
>-    CVPixelBufferLockBaseAddress(pixelBuffer, 0);
>-    auto* source = reinterpret_cast<uint8_t*>(CVPixelBufferGetBaseAddressOfPlane(pixelBuffer, 0));
>-
>-    ASSERT(m_width);
>-    ASSERT(m_height);
>-
>-    auto newBuffer = m_bufferPool.CreateBuffer(m_width, m_height);
>-    ASSERT(newBuffer);
>-    if (!newBuffer) {
>-        RELEASE_LOG(WebRTC, "RealtimeOutgoingVideoSourceCocoa::videoSampleAvailable unable to allocate buffer for conversion to YUV");
>-        return;
>-    }
>-    if (pixelFormatType == kCVPixelFormatType_32BGRA)
>-        ConvertToI420(webrtc::VideoType::kARGB, source, 0, 0, m_width, m_height, 0, libyuv::kRotate0, newBuffer);
>-    else {
>-        ASSERT(pixelFormatType == kCVPixelFormatType_32ARGB);
>-        ConvertToI420(webrtc::VideoType::kBGRA, source, 0, 0, m_width, m_height, 0, libyuv::kRotate0, newBuffer);
>+    if (m_shouldApplyRotation && m_currentRotation != webrtc::kVideoRotation_0) {
>+        // FIXME: We should make AVVideoCaptureSource handle the rotation whenever possible.
>+        // This implementation is inefficient, we should rotate on the CMSampleBuffer directly instead of doing this double allocation.
>+        auto rotatedBuffer = buffer->ToI420();
>+        ASSERT(rotatedBuffer);
>+        buffer = webrtc::I420Buffer::Rotate(*rotatedBuffer, m_currentRotation);
>     }
>-    CVPixelBufferUnlockBaseAddress(pixelBuffer, 0);
>-    if (m_shouldApplyRotation && m_currentRotation != webrtc::kVideoRotation_0)
>-        newBuffer = webrtc::I420Buffer::Rotate(*newBuffer, m_currentRotation);
>-    sendFrame(WTFMove(newBuffer));
>+    sendFrame(WTFMove(buffer));
> }
> 
> 
>diff --git a/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingVideoSourceCocoa.h b/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingVideoSourceCocoa.h
>index 77bef461660..961d512d7b4 100644
>--- a/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingVideoSourceCocoa.h
>+++ b/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingVideoSourceCocoa.h
>@@ -28,6 +28,7 @@
> #if USE(LIBWEBRTC)
> 
> #include "RealtimeOutgoingVideoSource.h"
>+#include "PixelBufferConformerCV.h"
> 
> namespace WebCore {
> 
>@@ -41,6 +42,10 @@ private:
>     // MediaStreamTrackPrivate::Observer API
>     void sampleBufferUpdated(MediaStreamTrackPrivate&, MediaSample&) final;
> 
>+    RetainPtr<CVPixelBufferRef> convertToYUV(CVPixelBufferRef);
>+
>+    std::unique_ptr<PixelBufferConformerCV> m_pixelBufferConformer;
>+
> #if !RELEASE_LOG_DISABLED
>     size_t m_numberOfFrames { 0 };
> #endif
>diff --git a/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingVideoSourceCocoa.mm b/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingVideoSourceCocoa.mm
>new file mode 100644
>index 00000000000..a06836daf1f
>--- /dev/null
>+++ b/Source/WebCore/platform/mediastream/mac/RealtimeOutgoingVideoSourceCocoa.mm
>@@ -0,0 +1,60 @@
>+/*
>+ * Copyright (C) 2018 Apple Inc.
>+ *
>+ * Redistribution and use in source and binary forms, with or without
>+ * modification, are permitted, provided that the following conditions
>+ * are required to be met:
>+ *
>+ * 1.  Redistributions of source code must retain the above copyright
>+ *     notice, this list of conditions and the following disclaimer.
>+ * 2.  Redistributions in binary form must reproduce the above copyright
>+ *     notice, this list of conditions and the following disclaimer in the
>+ *     documentation and/or other materials provided with the distribution.
>+ *
>+ * THIS SOFTWARE IS PROVIDED BY APPLE INC. AND ITS CONTRIBUTORS "AS IS" AND ANY
>+ * EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
>+ * WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
>+ * DISCLAIMED. IN NO EVENT SHALL APPLE INC. AND ITS CONTRIBUTORS BE LIABLE FOR
>+ * ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL
>+ * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR
>+ * SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER
>+ * CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY,
>+ * OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
>+ * OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
>+ */
>+
>+#import "config.h"
>+#import "RealtimeOutgoingVideoSourceCocoa.h"
>+
>+#if ENABLE(MEDIA_STREAM)
>+#import "GraphicsContextCG.h"
>+#import "ImageBuffer.h"
>+#import "MediaConstraints.h"
>+#import "MediaSampleAVFObjC.h"
>+#import "NotImplemented.h"
>+#import "PlatformLayer.h"
>+#import "RealtimeMediaSourceSettings.h"
>+#import <QuartzCore/CALayer.h>
>+#import <QuartzCore/CATransaction.h>
>+#import <objc/runtime.h>
>+
>+#import <pal/cf/CoreMediaSoftLink.h>
>+#import "CoreVideoSoftLink.h"
>+
>+namespace WebCore {
>+using namespace PAL;
>+
>+RetainPtr<CVPixelBufferRef> RealtimeOutgoingVideoSourceCocoa::convertToYUV(CVPixelBufferRef pixelBuffer)
>+{
>+    if (!pixelBuffer)
>+        return nullptr;
>+
>+    if (!m_pixelBufferConformer)
>+        m_pixelBufferConformer = std::make_unique<PixelBufferConformerCV>((__bridge CFDictionaryRef)@{ (__bridge NSString *)kCVPixelBufferPixelFormatTypeKey: @(kCVPixelFormatType_420YpCbCr8Planar) });
>+
>+    return m_pixelBufferConformer->convert(pixelBuffer);
>+}
>+
>+} // namespace WebCore
>+
>+#endif // ENABLE(MEDIA_STREAM)
>diff --git a/Source/WebCore/testing/MockLibWebRTCPeerConnection.cpp b/Source/WebCore/testing/MockLibWebRTCPeerConnection.cpp
>index b5d0de1c213..35a121f8d1b 100644
>--- a/Source/WebCore/testing/MockLibWebRTCPeerConnection.cpp
>+++ b/Source/WebCore/testing/MockLibWebRTCPeerConnection.cpp
>@@ -239,17 +239,17 @@ rtc::scoped_refptr<webrtc::DataChannelInterface> MockLibWebRTCPeerConnection::Cr
>     return new rtc::RefCountedObject<MockLibWebRTCDataChannel>(std::string(label), parameters.ordered, parameters.reliable, parameters.id);
> }
> 
>-rtc::scoped_refptr<webrtc::RtpSenderInterface> MockLibWebRTCPeerConnection::AddTrack(webrtc::MediaStreamTrackInterface* track, std::vector<webrtc::MediaStreamInterface*> streams)
>+webrtc::RTCErrorOr<rtc::scoped_refptr<webrtc::RtpSenderInterface>> MockLibWebRTCPeerConnection::AddTrack(rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> track, const std::vector<std::string>& streamIds)
> {
>     LibWebRTCProvider::callOnWebRTCSignalingThread([observer = &m_observer] {
>         observer->OnRenegotiationNeeded();
>     });
> 
>-    if (streams.size())
>-        m_streamLabel = streams.front()->label();
>+    if (!streamIds.empty())
>+        m_streamLabel = streamIds.front();
> 
>-    m_senders.append(new rtc::RefCountedObject<MockRtpSender>(rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>(track)));
>-    return m_senders.last().get();
>+    m_senders.append(new rtc::RefCountedObject<MockRtpSender>(WTFMove(track)));
>+    return rtc::scoped_refptr<webrtc::RtpSenderInterface>(m_senders.last().get());
> }
> 
> bool MockLibWebRTCPeerConnection::RemoveTrack(webrtc::RtpSenderInterface* sender)
>diff --git a/Source/WebCore/testing/MockLibWebRTCPeerConnection.h b/Source/WebCore/testing/MockLibWebRTCPeerConnection.h
>index 2b7e0dcd94f..9a80da48fee 100644
>--- a/Source/WebCore/testing/MockLibWebRTCPeerConnection.h
>+++ b/Source/WebCore/testing/MockLibWebRTCPeerConnection.h
>@@ -27,8 +27,15 @@
> #if USE(LIBWEBRTC)
> 
> #include "LibWebRTCMacros.h"
>+
>+#pragma clang diagnostic push
>+#pragma clang diagnostic ignored "-Wunused-parameter"
>+
> #include <webrtc/api/mediastreaminterface.h>
> #include <webrtc/api/peerconnectioninterface.h>
>+
>+#pragma clang diagnostic pop
>+
> #include <wtf/text/WTFString.h>
> 
> namespace WebCore {
>@@ -49,11 +56,9 @@ protected:
> private:
>     rtc::scoped_refptr<webrtc::StreamCollectionInterface> local_streams() override { return nullptr; }
>     rtc::scoped_refptr<webrtc::StreamCollectionInterface> remote_streams() override { return nullptr; }
>-    rtc::scoped_refptr<webrtc::DtmfSenderInterface> CreateDtmfSender(webrtc::AudioTrackInterface*) override { return nullptr; }
>     const webrtc::SessionDescriptionInterface* local_description() const override { return nullptr; }
>     const webrtc::SessionDescriptionInterface* remote_description() const override { return nullptr; }
>     bool AddIceCandidate(const webrtc::IceCandidateInterface*) override { return true; }
>-    void RegisterUMAObserver(webrtc::UMAObserver*) override { }
>     SignalingState signaling_state() override { return kStable; }
>     IceConnectionState ice_connection_state() override { return kIceConnectionNew; }
>     IceGatheringState ice_gathering_state() override { return kIceGatheringNew; }
>@@ -67,7 +72,7 @@ protected:
>     void SetRemoteDescription(webrtc::SetSessionDescriptionObserver*, webrtc::SessionDescriptionInterface*) final;
>     void CreateAnswer(webrtc::CreateSessionDescriptionObserver*, const webrtc::MediaConstraintsInterface*) final;
>     rtc::scoped_refptr<webrtc::DataChannelInterface> CreateDataChannel(const std::string&, const webrtc::DataChannelInit*) final;
>-    rtc::scoped_refptr<webrtc::RtpSenderInterface> AddTrack(webrtc::MediaStreamTrackInterface*, std::vector<webrtc::MediaStreamInterface*> streams) final;
>+    webrtc::RTCErrorOr<rtc::scoped_refptr<webrtc::RtpSenderInterface>> AddTrack(rtc::scoped_refptr<webrtc::MediaStreamTrackInterface>, const std::vector<std::string>& streams) final;
>     bool RemoveTrack(webrtc::RtpSenderInterface*) final;
>     webrtc::RTCError SetBitrate(const BitrateParameters&) final { return { }; }
> 
>@@ -221,8 +226,8 @@ public:
>     cricket::MediaType media_type() const { return cricket::MEDIA_TYPE_VIDEO; }
>     std::string id() const { return ""; }
>     std::vector<std::string> stream_ids() const { return { }; }
>-    webrtc::RtpParameters GetParameters() const { return { }; }
>-    bool SetParameters(const webrtc::RtpParameters&) { return false; }
>+    webrtc::RtpParameters GetParameters() { return { }; }
>+    webrtc::RTCError SetParameters(const webrtc::RtpParameters&) { return { }; }
>     rtc::scoped_refptr<webrtc::DtmfSenderInterface> GetDtmfSender() const { return nullptr; }
> 
> private:
>@@ -245,7 +250,6 @@ private:
> 
>     void SetOptions(const Options&) final { }
>     rtc::scoped_refptr<webrtc::AudioSourceInterface> CreateAudioSource(const cricket::AudioOptions&) final { return nullptr; }
>-    rtc::scoped_refptr<webrtc::AudioSourceInterface> CreateAudioSource(const webrtc::MediaConstraintsInterface*) final { return nullptr; }
> 
>     rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> CreateVideoSource(cricket::VideoCapturer*) final { return nullptr; }
>     rtc::scoped_refptr<webrtc::VideoTrackSourceInterface> CreateVideoSource(cricket::VideoCapturer*, const webrtc::MediaConstraintsInterface*) final { return nullptr; }

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Attachments on bug 188745: 347508 | 347510 | 347512 | 347568 | 347607 | 347679 | 347698 | 347735 | 347769 | 347795 | 347808 | 347839